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authorTimothy Pearson <tpearson@raptorengineering.com>2017-08-23 14:45:25 -0500
committerTimothy Pearson <tpearson@raptorengineering.com>2017-08-23 14:45:25 -0500
commitfcbb27b0ec6dcbc5a5108cb8fb19eae64593d204 (patch)
tree22962a4387943edc841c72a4e636a068c66d58fd /sound
downloadast2050-linux-kernel-fcbb27b0ec6dcbc5a5108cb8fb19eae64593d204.zip
ast2050-linux-kernel-fcbb27b0ec6dcbc5a5108cb8fb19eae64593d204.tar.gz
Initial import of modified Linux 2.6.28 tree
Original upstream URL: git://git.kernel.org/pub/scm/linux/kernel/git/stable/linux-stable.git | branch linux-2.6.28.y
Diffstat (limited to 'sound')
-rw-r--r--sound/Kconfig109
-rw-r--r--sound/Makefile19
-rw-r--r--sound/ac97_bus.c76
-rw-r--r--sound/aoa/Kconfig17
-rw-r--r--sound/aoa/Makefile4
-rw-r--r--sound/aoa/aoa-gpio.h81
-rw-r--r--sound/aoa/aoa.h129
-rw-r--r--sound/aoa/codecs/Kconfig32
-rw-r--r--sound/aoa/codecs/Makefile3
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-onyx.c1118
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-onyx.h75
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h134
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h209
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-tas.c1012
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-tas.h55
-rw-r--r--sound/aoa/codecs/snd-aoa-codec-toonie.c150
-rw-r--r--sound/aoa/core/Makefile5
-rw-r--r--sound/aoa/core/snd-aoa-alsa.c99
-rw-r--r--sound/aoa/core/snd-aoa-alsa.h16
-rw-r--r--sound/aoa/core/snd-aoa-core.c162
-rw-r--r--sound/aoa/core/snd-aoa-gpio-feature.c408
-rw-r--r--sound/aoa/core/snd-aoa-gpio-pmf.c252
-rw-r--r--sound/aoa/fabrics/Kconfig11
-rw-r--r--sound/aoa/fabrics/Makefile1
-rw-r--r--sound/aoa/fabrics/snd-aoa-fabric-layout.c1120
-rw-r--r--sound/aoa/soundbus/Kconfig14
-rw-r--r--sound/aoa/soundbus/Makefile3
-rw-r--r--sound/aoa/soundbus/core.c219
-rw-r--r--sound/aoa/soundbus/i2sbus/Makefile2
-rw-r--r--sound/aoa/soundbus/i2sbus/i2sbus-control.c193
-rw-r--r--sound/aoa/soundbus/i2sbus/i2sbus-core.c450
-rw-r--r--sound/aoa/soundbus/i2sbus/i2sbus-interface.h187
-rw-r--r--sound/aoa/soundbus/i2sbus/i2sbus-pcm.c1062
-rw-r--r--sound/aoa/soundbus/i2sbus/i2sbus.h126
-rw-r--r--sound/aoa/soundbus/soundbus.h204
-rw-r--r--sound/aoa/soundbus/sysfs.c42
-rw-r--r--sound/arm/Kconfig54
-rw-r--r--sound/arm/Makefile19
-rw-r--r--sound/arm/aaci.c1206
-rw-r--r--sound/arm/aaci.h247
-rw-r--r--sound/arm/devdma.c80
-rw-r--r--sound/arm/devdma.h3
-rw-r--r--sound/arm/pxa2xx-ac97-lib.c384
-rw-r--r--sound/arm/pxa2xx-ac97.c260
-rw-r--r--sound/arm/pxa2xx-pcm-lib.c278
-rw-r--r--sound/arm/pxa2xx-pcm.c131
-rw-r--r--sound/arm/pxa2xx-pcm.h30
-rw-r--r--sound/arm/sa11xx-uda1341.c983
-rw-r--r--sound/core/Kconfig186
-rw-r--r--sound/core/Makefile31
-rw-r--r--sound/core/control.c1517
-rw-r--r--sound/core/control_compat.c443
-rw-r--r--sound/core/device.c243
-rw-r--r--sound/core/hwdep.c528
-rw-r--r--sound/core/hwdep_compat.c78
-rw-r--r--sound/core/info.c1015
-rw-r--r--sound/core/info_oss.c138
-rw-r--r--sound/core/init.c850
-rw-r--r--sound/core/isadma.c108
-rw-r--r--sound/core/jack.c166
-rw-r--r--sound/core/memalloc.c542
-rw-r--r--sound/core/memory.c91
-rw-r--r--sound/core/misc.c106
-rw-r--r--sound/core/oss/Makefile13
-rw-r--r--sound/core/oss/copy.c92
-rw-r--r--sound/core/oss/io.c141
-rw-r--r--sound/core/oss/linear.c180
-rw-r--r--sound/core/oss/mixer_oss.c1389
-rw-r--r--sound/core/oss/mulaw.c344
-rw-r--r--sound/core/oss/pcm_oss.c3062
-rw-r--r--sound/core/oss/pcm_plugin.c752
-rw-r--r--sound/core/oss/pcm_plugin.h184
-rw-r--r--sound/core/oss/rate.c348
-rw-r--r--sound/core/oss/route.c110
-rw-r--r--sound/core/pcm.c1142
-rw-r--r--sound/core/pcm_compat.c534
-rw-r--r--sound/core/pcm_lib.c1975
-rw-r--r--sound/core/pcm_memory.c434
-rw-r--r--sound/core/pcm_misc.c470
-rw-r--r--sound/core/pcm_native.c3421
-rw-r--r--sound/core/pcm_timer.c162
-rw-r--r--sound/core/rawmidi.c1692
-rw-r--r--sound/core/rawmidi_compat.c120
-rw-r--r--sound/core/rtctimer.c188
-rw-r--r--sound/core/seq/Makefile40
-rw-r--r--sound/core/seq/oss/Makefile10
-rw-r--r--sound/core/seq/oss/seq_oss.c311
-rw-r--r--sound/core/seq/oss/seq_oss_device.h189
-rw-r--r--sound/core/seq/oss/seq_oss_event.c447
-rw-r--r--sound/core/seq/oss/seq_oss_event.h112
-rw-r--r--sound/core/seq/oss/seq_oss_init.c545
-rw-r--r--sound/core/seq/oss/seq_oss_ioctl.c209
-rw-r--r--sound/core/seq/oss/seq_oss_midi.c711
-rw-r--r--sound/core/seq/oss/seq_oss_midi.h48
-rw-r--r--sound/core/seq/oss/seq_oss_readq.c236
-rw-r--r--sound/core/seq/oss/seq_oss_readq.h56
-rw-r--r--sound/core/seq/oss/seq_oss_rw.c216
-rw-r--r--sound/core/seq/oss/seq_oss_synth.c662
-rw-r--r--sound/core/seq/oss/seq_oss_synth.h51
-rw-r--r--sound/core/seq/oss/seq_oss_timer.c283
-rw-r--r--sound/core/seq/oss/seq_oss_timer.h70
-rw-r--r--sound/core/seq/oss/seq_oss_writeq.c172
-rw-r--r--sound/core/seq/oss/seq_oss_writeq.h50
-rw-r--r--sound/core/seq/seq.c130
-rw-r--r--sound/core/seq/seq_clientmgr.c2587
-rw-r--r--sound/core/seq/seq_clientmgr.h103
-rw-r--r--sound/core/seq/seq_compat.c138
-rw-r--r--sound/core/seq/seq_device.c572
-rw-r--r--sound/core/seq/seq_dummy.c261
-rw-r--r--sound/core/seq/seq_fifo.c272
-rw-r--r--sound/core/seq/seq_fifo.h72
-rw-r--r--sound/core/seq/seq_info.c71
-rw-r--r--sound/core/seq/seq_info.h40
-rw-r--r--sound/core/seq/seq_lock.c48
-rw-r--r--sound/core/seq/seq_lock.h33
-rw-r--r--sound/core/seq/seq_memory.c520
-rw-r--r--sound/core/seq/seq_memory.h103
-rw-r--r--sound/core/seq/seq_midi.c482
-rw-r--r--sound/core/seq/seq_midi_emul.c739
-rw-r--r--sound/core/seq/seq_midi_event.c549
-rw-r--r--sound/core/seq/seq_ports.c684
-rw-r--r--sound/core/seq/seq_ports.h142
-rw-r--r--sound/core/seq/seq_prioq.c452
-rw-r--r--sound/core/seq/seq_prioq.h62
-rw-r--r--sound/core/seq/seq_queue.c795
-rw-r--r--sound/core/seq/seq_queue.h139
-rw-r--r--sound/core/seq/seq_system.c173
-rw-r--r--sound/core/seq/seq_system.h46
-rw-r--r--sound/core/seq/seq_timer.c456
-rw-r--r--sound/core/seq/seq_timer.h148
-rw-r--r--sound/core/seq/seq_virmidi.c542
-rw-r--r--sound/core/sgbuf.c138
-rw-r--r--sound/core/sound.c468
-rw-r--r--sound/core/sound_oss.c278
-rw-r--r--sound/core/timer.c1992
-rw-r--r--sound/core/timer_compat.c127
-rw-r--r--sound/core/vmaster.c371
-rw-r--r--sound/drivers/Kconfig185
-rw-r--r--sound/drivers/Makefile23
-rw-r--r--sound/drivers/dummy.c715
-rw-r--r--sound/drivers/ml403-ac97cr.c1354
-rw-r--r--sound/drivers/mpu401/Makefile12
-rw-r--r--sound/drivers/mpu401/mpu401.c289
-rw-r--r--sound/drivers/mpu401/mpu401_uart.c631
-rw-r--r--sound/drivers/mtpav.c791
-rw-r--r--sound/drivers/mts64.c1088
-rw-r--r--sound/drivers/opl3/Makefile20
-rw-r--r--sound/drivers/opl3/opl3_drums.c226
-rw-r--r--sound/drivers/opl3/opl3_lib.c560
-rw-r--r--sound/drivers/opl3/opl3_midi.c869
-rw-r--r--sound/drivers/opl3/opl3_oss.c283
-rw-r--r--sound/drivers/opl3/opl3_seq.c297
-rw-r--r--sound/drivers/opl3/opl3_synth.c614
-rw-r--r--sound/drivers/opl3/opl3_voice.h52
-rw-r--r--sound/drivers/opl4/Makefile18
-rw-r--r--sound/drivers/opl4/opl4_lib.c279
-rw-r--r--sound/drivers/opl4/opl4_local.h232
-rw-r--r--sound/drivers/opl4/opl4_mixer.c95
-rw-r--r--sound/drivers/opl4/opl4_proc.c165
-rw-r--r--sound/drivers/opl4/opl4_seq.c214
-rw-r--r--sound/drivers/opl4/opl4_synth.c634
-rw-r--r--sound/drivers/opl4/yrw801.c961
-rw-r--r--sound/drivers/pcm-indirect2.c573
-rw-r--r--sound/drivers/pcm-indirect2.h140
-rw-r--r--sound/drivers/pcsp/Makefile2
-rw-r--r--sound/drivers/pcsp/pcsp.c239
-rw-r--r--sound/drivers/pcsp/pcsp.h84
-rw-r--r--sound/drivers/pcsp/pcsp_input.c116
-rw-r--r--sound/drivers/pcsp/pcsp_input.h14
-rw-r--r--sound/drivers/pcsp/pcsp_lib.c321
-rw-r--r--sound/drivers/pcsp/pcsp_mixer.c144
-rw-r--r--sound/drivers/portman2x4.c877
-rw-r--r--sound/drivers/serial-u16550.c1049
-rw-r--r--sound/drivers/virmidi.c195
-rw-r--r--sound/drivers/vx/Makefile8
-rw-r--r--sound/drivers/vx/vx_cmd.c109
-rw-r--r--sound/drivers/vx/vx_cmd.h246
-rw-r--r--sound/drivers/vx/vx_core.c826
-rw-r--r--sound/drivers/vx/vx_hwdep.c270
-rw-r--r--sound/drivers/vx/vx_mixer.c1028
-rw-r--r--sound/drivers/vx/vx_pcm.c1330
-rw-r--r--sound/drivers/vx/vx_uer.c312
-rw-r--r--sound/i2c/Makefile17
-rw-r--r--sound/i2c/cs8427.c626
-rw-r--r--sound/i2c/i2c.c346
-rw-r--r--sound/i2c/l3/Makefile8
-rw-r--r--sound/i2c/l3/uda1341.c935
-rw-r--r--sound/i2c/other/Makefile16
-rw-r--r--sound/i2c/other/ak4114.c626
-rw-r--r--sound/i2c/other/ak4117.c551
-rw-r--r--sound/i2c/other/ak4xxx-adda.c872
-rw-r--r--sound/i2c/other/pt2258.c226
-rw-r--r--sound/i2c/other/tea575x-tuner.c252
-rw-r--r--sound/i2c/tea6330t.c385
-rw-r--r--sound/isa/Kconfig415
-rw-r--r--sound/isa/Makefile30
-rw-r--r--sound/isa/ad1816a/Makefile9
-rw-r--r--sound/isa/ad1816a/ad1816a.c289
-rw-r--r--sound/isa/ad1816a/ad1816a_lib.c977
-rw-r--r--sound/isa/ad1848/Makefile10
-rw-r--r--sound/isa/ad1848/ad1848.c188
-rw-r--r--sound/isa/adlib.c129
-rw-r--r--sound/isa/als100.c327
-rw-r--r--sound/isa/azt2320.c354
-rw-r--r--sound/isa/cmi8330.c711
-rw-r--r--sound/isa/cs423x/Makefile15
-rw-r--r--sound/isa/cs423x/cs4231.c205
-rw-r--r--sound/isa/cs423x/cs4232.c2
-rw-r--r--sound/isa/cs423x/cs4236.c750
-rw-r--r--sound/isa/cs423x/cs4236_lib.c1037
-rw-r--r--sound/isa/dt019x.c320
-rw-r--r--sound/isa/es1688/Makefile11
-rw-r--r--sound/isa/es1688/es1688.c208
-rw-r--r--sound/isa/es1688/es1688_lib.c1053
-rw-r--r--sound/isa/es18xx.c2452
-rw-r--r--sound/isa/gus/Makefile24
-rw-r--r--sound/isa/gus/gus_dma.c243
-rw-r--r--sound/isa/gus/gus_dram.c102
-rw-r--r--sound/isa/gus/gus_instr.c172
-rw-r--r--sound/isa/gus/gus_io.c540
-rw-r--r--sound/isa/gus/gus_irq.c147
-rw-r--r--sound/isa/gus/gus_main.c482
-rw-r--r--sound/isa/gus/gus_mem.c350
-rw-r--r--sound/isa/gus/gus_mem_proc.c134
-rw-r--r--sound/isa/gus/gus_mixer.c193
-rw-r--r--sound/isa/gus/gus_pcm.c896
-rw-r--r--sound/isa/gus/gus_reset.c413
-rw-r--r--sound/isa/gus/gus_tables.h90
-rw-r--r--sound/isa/gus/gus_timer.c203
-rw-r--r--sound/isa/gus/gus_uart.c256
-rw-r--r--sound/isa/gus/gus_volume.c217
-rw-r--r--sound/isa/gus/gusclassic.c245
-rw-r--r--sound/isa/gus/gusextreme.c372
-rw-r--r--sound/isa/gus/gusmax.c387
-rw-r--r--sound/isa/gus/interwave-stb.c2
-rw-r--r--sound/isa/gus/interwave.c944
-rw-r--r--sound/isa/opl3sa2.c963
-rw-r--r--sound/isa/opti9xx/Makefile15
-rw-r--r--sound/isa/opti9xx/miro.c1444
-rw-r--r--sound/isa/opti9xx/miro.h73
-rw-r--r--sound/isa/opti9xx/opti92x-ad1848.c1035
-rw-r--r--sound/isa/opti9xx/opti92x-cs4231.c2
-rw-r--r--sound/isa/opti9xx/opti93x.c3
-rw-r--r--sound/isa/sb/Makefile36
-rw-r--r--sound/isa/sb/emu8000.c1159
-rw-r--r--sound/isa/sb/emu8000_callback.c546
-rw-r--r--sound/isa/sb/emu8000_local.h45
-rw-r--r--sound/isa/sb/emu8000_patch.c305
-rw-r--r--sound/isa/sb/emu8000_pcm.c701
-rw-r--r--sound/isa/sb/emu8000_synth.c135
-rw-r--r--sound/isa/sb/es968.c247
-rw-r--r--sound/isa/sb/sb16.c695
-rw-r--r--sound/isa/sb/sb16_csp.c1200
-rw-r--r--sound/isa/sb/sb16_main.c924
-rw-r--r--sound/isa/sb/sb8.c267
-rw-r--r--sound/isa/sb/sb8_main.c559
-rw-r--r--sound/isa/sb/sb8_midi.c286
-rw-r--r--sound/isa/sb/sb_common.c320
-rw-r--r--sound/isa/sb/sb_mixer.c995
-rw-r--r--sound/isa/sb/sbawe.c2
-rw-r--r--sound/isa/sc6000.c655
-rw-r--r--sound/isa/sgalaxy.c369
-rw-r--r--sound/isa/sscape.c1560
-rw-r--r--sound/isa/wavefront/Makefile9
-rw-r--r--sound/isa/wavefront/wavefront.c687
-rw-r--r--sound/isa/wavefront/wavefront_fx.c303
-rw-r--r--sound/isa/wavefront/wavefront_midi.c577
-rw-r--r--sound/isa/wavefront/wavefront_synth.c2198
-rw-r--r--sound/isa/wavefront/yss225.c2739
-rw-r--r--sound/isa/wss/Makefile10
-rw-r--r--sound/isa/wss/wss_lib.c2322
-rw-r--r--sound/last.c41
-rw-r--r--sound/mips/Kconfig34
-rw-r--r--sound/mips/Makefile12
-rw-r--r--sound/mips/ad1843.c561
-rw-r--r--sound/mips/au1x00.c693
-rw-r--r--sound/mips/hal2.c947
-rw-r--r--sound/mips/hal2.h245
-rw-r--r--sound/mips/sgio2audio.c1006
-rw-r--r--sound/oss/.gitignore4
-rw-r--r--sound/oss/CHANGELOG369
-rw-r--r--sound/oss/Kconfig569
-rw-r--r--sound/oss/Makefile111
-rw-r--r--sound/oss/README.FIRST6
-rw-r--r--sound/oss/ac97_codec.c1206
-rw-r--r--sound/oss/ad1848.c3068
-rw-r--r--sound/oss/ad1848.h24
-rw-r--r--sound/oss/ad1848_mixer.h253
-rw-r--r--sound/oss/aedsp16.c1372
-rw-r--r--sound/oss/au1550_ac97.c2129
-rw-r--r--sound/oss/audio.c983
-rw-r--r--sound/oss/bin2hex.c39
-rw-r--r--sound/oss/coproc.h12
-rw-r--r--sound/oss/dev_table.c256
-rw-r--r--sound/oss/dev_table.h390
-rw-r--r--sound/oss/dmabuf.c1267
-rw-r--r--sound/oss/dmasound/Kconfig45
-rw-r--r--sound/oss/dmasound/Makefile7
-rw-r--r--sound/oss/dmasound/dmasound.h262
-rw-r--r--sound/oss/dmasound/dmasound_atari.c1618
-rw-r--r--sound/oss/dmasound/dmasound_core.c1549
-rw-r--r--sound/oss/dmasound/dmasound_paula.c740
-rw-r--r--sound/oss/dmasound/dmasound_q40.c634
-rw-r--r--sound/oss/hex2hex.c101
-rw-r--r--sound/oss/kahlua.c230
-rw-r--r--sound/oss/midi_ctrl.h22
-rw-r--r--sound/oss/midi_synth.c714
-rw-r--r--sound/oss/midi_synth.h47
-rw-r--r--sound/oss/midibuf.c424
-rw-r--r--sound/oss/mpu401.c1815
-rw-r--r--sound/oss/mpu401.h11
-rw-r--r--sound/oss/msnd.c414
-rw-r--r--sound/oss/msnd.h278
-rw-r--r--sound/oss/msnd_classic.c3
-rw-r--r--sound/oss/msnd_classic.h185
-rw-r--r--sound/oss/msnd_pinnacle.c1916
-rw-r--r--sound/oss/msnd_pinnacle.h246
-rw-r--r--sound/oss/opl3.c1250
-rw-r--r--sound/oss/opl3_hw.h246
-rw-r--r--sound/oss/os.h46
-rw-r--r--sound/oss/pas2.h17
-rw-r--r--sound/oss/pas2_card.c457
-rw-r--r--sound/oss/pas2_midi.c262
-rw-r--r--sound/oss/pas2_mixer.c336
-rw-r--r--sound/oss/pas2_pcm.c437
-rw-r--r--sound/oss/pss.c1266
-rw-r--r--sound/oss/sb.h185
-rw-r--r--sound/oss/sb_audio.c1098
-rw-r--r--sound/oss/sb_card.c353
-rw-r--r--sound/oss/sb_card.h149
-rw-r--r--sound/oss/sb_common.c1291
-rw-r--r--sound/oss/sb_ess.c1832
-rw-r--r--sound/oss/sb_ess.h34
-rw-r--r--sound/oss/sb_midi.c205
-rw-r--r--sound/oss/sb_mixer.c768
-rw-r--r--sound/oss/sb_mixer.h105
-rw-r--r--sound/oss/sequencer.c1674
-rw-r--r--sound/oss/sh_dac_audio.c331
-rw-r--r--sound/oss/sound_calls.h87
-rw-r--r--sound/oss/sound_config.h145
-rw-r--r--sound/oss/sound_firmware.h2
-rw-r--r--sound/oss/sound_timer.c327
-rw-r--r--sound/oss/soundcard.c744
-rw-r--r--sound/oss/soundvers.h2
-rw-r--r--sound/oss/sscape.c1480
-rw-r--r--sound/oss/swarm_cs4297a.c2740
-rw-r--r--sound/oss/sys_timer.c288
-rw-r--r--sound/oss/trix.c525
-rw-r--r--sound/oss/tuning.h23
-rw-r--r--sound/oss/uart401.c481
-rw-r--r--sound/oss/uart6850.c361
-rw-r--r--sound/oss/ulaw.h69
-rw-r--r--sound/oss/v_midi.c289
-rw-r--r--sound/oss/v_midi.h15
-rw-r--r--sound/oss/vidc.c560
-rw-r--r--sound/oss/vidc.h63
-rw-r--r--sound/oss/vidc_fill.S218
-rw-r--r--sound/oss/vwsnd.c3485
-rw-r--r--sound/oss/waveartist.c2032
-rw-r--r--sound/oss/waveartist.h92
-rw-r--r--sound/parisc/Kconfig20
-rw-r--r--sound/parisc/Makefile8
-rw-r--r--sound/parisc/harmony.c1043
-rw-r--r--sound/parisc/harmony.h154
-rw-r--r--sound/pci/Kconfig895
-rw-r--r--sound/pci/Makefile78
-rw-r--r--sound/pci/ac97/Makefile10
-rw-r--r--sound/pci/ac97/ac97_codec.c2903
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-rw-r--r--sound/soc/codecs/wm8580.h42
-rw-r--r--sound/soc/codecs/wm8731.c797
-rw-r--r--sound/soc/codecs/wm8731.h46
-rw-r--r--sound/soc/codecs/wm8750.c1091
-rw-r--r--sound/soc/codecs/wm8750.h69
-rw-r--r--sound/soc/codecs/wm8753.c1882
-rw-r--r--sound/soc/codecs/wm8753.h127
-rw-r--r--sound/soc/codecs/wm8900.c1541
-rw-r--r--sound/soc/codecs/wm8900.h64
-rw-r--r--sound/soc/codecs/wm8903.c1813
-rw-r--r--sound/soc/codecs/wm8903.h1463
-rw-r--r--sound/soc/codecs/wm8971.c941
-rw-r--r--sound/soc/codecs/wm8971.h64
-rw-r--r--sound/soc/codecs/wm8990.c1635
-rw-r--r--sound/soc/codecs/wm8990.h843
-rw-r--r--sound/soc/codecs/wm9712.c750
-rw-r--r--sound/soc/codecs/wm9712.h14
-rw-r--r--sound/soc/codecs/wm9713.c1306
-rw-r--r--sound/soc/codecs/wm9713.h53
-rw-r--r--sound/soc/davinci/Kconfig19
-rw-r--r--sound/soc/davinci/Makefile11
-rw-r--r--sound/soc/davinci/davinci-evm.c202
-rw-r--r--sound/soc/davinci/davinci-i2s.c409
-rw-r--r--sound/soc/davinci/davinci-i2s.h17
-rw-r--r--sound/soc/davinci/davinci-pcm.c389
-rw-r--r--sound/soc/davinci/davinci-pcm.h29
-rw-r--r--sound/soc/fsl/Kconfig27
-rw-r--r--sound/soc/fsl/Makefile11
-rw-r--r--sound/soc/fsl/fsl_dma.c858
-rw-r--r--sound/soc/fsl/fsl_dma.h149
-rw-r--r--sound/soc/fsl/fsl_ssi.c697
-rw-r--r--sound/soc/fsl/fsl_ssi.h224
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c886
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c625
-rw-r--r--sound/soc/fsl/soc-of-simple.c171
-rw-r--r--sound/soc/omap/Kconfig23
-rw-r--r--sound/soc/omap/Makefile13
-rw-r--r--sound/soc/omap/n810.c391
-rw-r--r--sound/soc/omap/omap-mcbsp.c500
-rw-r--r--sound/soc/omap/omap-mcbsp.h55
-rw-r--r--sound/soc/omap/omap-pcm.c359
-rw-r--r--sound/soc/omap/omap-pcm.h35
-rw-r--r--sound/soc/omap/osk5912.c232
-rw-r--r--sound/soc/pxa/Kconfig77
-rw-r--r--sound/soc/pxa/Makefile23
-rw-r--r--sound/soc/pxa/corgi.c372
-rw-r--r--sound/soc/pxa/e800_wm9712.c89
-rw-r--r--sound/soc/pxa/em-x270.c102
-rw-r--r--sound/soc/pxa/poodle.c341
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c232
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.h22
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c410
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.h20
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c123
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.h19
-rw-r--r--sound/soc/pxa/spitz.c375
-rw-r--r--sound/soc/pxa/tosa.c292
-rw-r--r--sound/soc/s3c24xx/Kconfig46
-rw-r--r--sound/soc/s3c24xx/Makefile19
-rw-r--r--sound/soc/s3c24xx/lm4857.h32
-rw-r--r--sound/soc/s3c24xx/ln2440sbc_alc650.c85
-rw-r--r--sound/soc/s3c24xx/neo1973_wm8753.c722
-rw-r--r--sound/soc/s3c24xx/s3c2412-i2s.c745
-rw-r--r--sound/soc/s3c24xx/s3c2412-i2s.h38
-rw-r--r--sound/soc/s3c24xx/s3c2443-ac97.c398
-rw-r--r--sound/soc/s3c24xx/s3c24xx-ac97.h31
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c483
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.h37
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c470
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.h31
-rw-r--r--sound/soc/s3c24xx/smdk2443_wm9710.c81
-rw-r--r--sound/soc/sh/Kconfig38
-rw-r--r--sound/soc/sh/Makefile14
-rw-r--r--sound/soc/sh/dma-sh7760.c353
-rw-r--r--sound/soc/sh/hac.c318
-rw-r--r--sound/soc/sh/sh7760-ac97.c91
-rw-r--r--sound/soc/sh/ssi.c399
-rw-r--r--sound/soc/soc-core.c1891
-rw-r--r--sound/soc/soc-dapm.c1545
-rw-r--r--sound/sound_core.c593
-rw-r--r--sound/sound_firmware.c79
-rw-r--r--sound/sparc/Kconfig41
-rw-r--r--sound/sparc/Makefile12
-rw-r--r--sound/sparc/amd7930.c1094
-rw-r--r--sound/sparc/cs4231.c2129
-rw-r--r--sound/sparc/dbri.c2706
-rw-r--r--sound/spi/Kconfig38
-rw-r--r--sound/spi/Makefile5
-rw-r--r--sound/spi/at73c213.c1131
-rw-r--r--sound/spi/at73c213.h119
-rw-r--r--sound/synth/Makefile20
-rw-r--r--sound/synth/emux/Makefile20
-rw-r--r--sound/synth/emux/emux.c191
-rw-r--r--sound/synth/emux/emux_effect.c310
-rw-r--r--sound/synth/emux/emux_hwdep.c171
-rw-r--r--sound/synth/emux/emux_nrpn.c396
-rw-r--r--sound/synth/emux/emux_oss.c516
-rw-r--r--sound/synth/emux/emux_proc.c133
-rw-r--r--sound/synth/emux/emux_seq.c403
-rw-r--r--sound/synth/emux/emux_synth.c981
-rw-r--r--sound/synth/emux/emux_voice.h96
-rw-r--r--sound/synth/emux/soundfont.c1489
-rw-r--r--sound/synth/util_mem.c210
-rw-r--r--sound/usb/Kconfig83
-rw-r--r--sound/usb/Makefile13
-rw-r--r--sound/usb/caiaq/Makefile4
-rw-r--r--sound/usb/caiaq/caiaq-audio.c697
-rw-r--r--sound/usb/caiaq/caiaq-audio.h7
-rw-r--r--sound/usb/caiaq/caiaq-control.c315
-rw-r--r--sound/usb/caiaq/caiaq-control.h6
-rw-r--r--sound/usb/caiaq/caiaq-device.c506
-rw-r--r--sound/usb/caiaq/caiaq-device.h129
-rw-r--r--sound/usb/caiaq/caiaq-input.c363
-rw-r--r--sound/usb/caiaq/caiaq-input.h8
-rw-r--r--sound/usb/caiaq/caiaq-midi.c180
-rw-r--r--sound/usb/caiaq/caiaq-midi.h8
-rw-r--r--sound/usb/usbaudio.c3783
-rw-r--r--sound/usb/usbaudio.h251
-rw-r--r--sound/usb/usbmidi.c1841
-rw-r--r--sound/usb/usbmixer.c2076
-rw-r--r--sound/usb/usbmixer_maps.c316
-rw-r--r--sound/usb/usbquirks.h2039
-rw-r--r--sound/usb/usx2y/Makefile5
-rw-r--r--sound/usb/usx2y/us122l.c693
-rw-r--r--sound/usb/usx2y/us122l.h27
-rw-r--r--sound/usb/usx2y/usX2Yhwdep.c280
-rw-r--r--sound/usb/usx2y/usX2Yhwdep.h6
-rw-r--r--sound/usb/usx2y/usb_stream.c761
-rw-r--r--sound/usb/usx2y/usb_stream.h112
-rw-r--r--sound/usb/usx2y/usbus428ctldefs.h104
-rw-r--r--sound/usb/usx2y/usbusx2y.c460
-rw-r--r--sound/usb/usx2y/usbusx2y.h83
-rw-r--r--sound/usb/usx2y/usbusx2yaudio.c1024
-rw-r--r--sound/usb/usx2y/usx2y.h51
-rw-r--r--sound/usb/usx2y/usx2yhwdeppcm.c793
-rw-r--r--sound/usb/usx2y/usx2yhwdeppcm.h20
857 files changed, 476713 insertions, 0 deletions
diff --git a/sound/Kconfig b/sound/Kconfig
new file mode 100644
index 0000000..200aca1
--- /dev/null
+++ b/sound/Kconfig
@@ -0,0 +1,109 @@
+# sound/Config.in
+#
+
+menuconfig SOUND
+ tristate "Sound card support"
+ depends on HAS_IOMEM
+ help
+ If you have a sound card in your computer, i.e. if it can say more
+ than an occasional beep, say Y. Be sure to have all the information
+ about your sound card and its configuration down (I/O port,
+ interrupt and DMA channel), because you will be asked for it.
+
+ You want to read the Sound-HOWTO, available from
+ <http://www.tldp.org/docs.html#howto>. General information about
+ the modular sound system is contained in the files
+ <file:Documentation/sound/oss/Introduction>. The file
+ <file:Documentation/sound/oss/README.OSS> contains some slightly
+ outdated but still useful information as well. Newer sound
+ driver documentation is found in <file:Documentation/sound/alsa/*>.
+
+ If you have a PnP sound card and you want to configure it at boot
+ time using the ISA PnP tools (read
+ <http://www.roestock.demon.co.uk/isapnptools/>), then you need to
+ compile the sound card support as a module and load that module
+ after the PnP configuration is finished. To do this, choose M here
+ and read <file:Documentation/sound/oss/README.modules>; the module
+ will be called soundcore.
+
+if SOUND
+
+config SOUND_OSS_CORE
+ bool
+ default n
+
+source "sound/oss/dmasound/Kconfig"
+
+if !M68K
+
+menuconfig SND
+ tristate "Advanced Linux Sound Architecture"
+ help
+ Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture),
+ the new base sound system.
+
+ For more information, see <http://www.alsa-project.org/>
+
+if SND
+
+source "sound/core/Kconfig"
+
+source "sound/drivers/Kconfig"
+
+source "sound/isa/Kconfig"
+
+source "sound/pci/Kconfig"
+
+source "sound/ppc/Kconfig"
+
+source "sound/aoa/Kconfig"
+
+source "sound/arm/Kconfig"
+
+source "sound/spi/Kconfig"
+
+source "sound/mips/Kconfig"
+
+source "sound/sh/Kconfig"
+
+# the following will depend on the order of config.
+# here assuming USB is defined before ALSA
+source "sound/usb/Kconfig"
+
+# the following will depend on the order of config.
+# here assuming PCMCIA is defined before ALSA
+source "sound/pcmcia/Kconfig"
+
+source "sound/sparc/Kconfig"
+
+source "sound/parisc/Kconfig"
+
+source "sound/soc/Kconfig"
+
+endif # SND
+
+menuconfig SOUND_PRIME
+ tristate "Open Sound System (DEPRECATED)"
+ select SOUND_OSS_CORE
+ help
+ Say 'Y' or 'M' to enable Open Sound System drivers.
+
+if SOUND_PRIME
+
+source "sound/oss/Kconfig"
+
+endif # SOUND_PRIME
+
+endif # !M68K
+
+endif # SOUND
+
+# AC97_BUS is used from both sound and ucb1400
+config AC97_BUS
+ tristate
+ help
+ This is used to avoid config and link hard dependencies between the
+ sound subsystem and other function drivers completely unrelated to
+ sound although they're sharing the AC97 bus. Concerned drivers
+ should "select" this.
+
diff --git a/sound/Makefile b/sound/Makefile
new file mode 100644
index 0000000..c76d707
--- /dev/null
+++ b/sound/Makefile
@@ -0,0 +1,19 @@
+# Makefile for the Linux sound card driver
+#
+
+obj-$(CONFIG_SOUND) += soundcore.o
+obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o
+obj-$(CONFIG_SOUND_PRIME) += oss/
+obj-$(CONFIG_DMASOUND) += oss/
+obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
+ sparc/ spi/ parisc/ pcmcia/ mips/ soc/
+obj-$(CONFIG_SND_AOA) += aoa/
+
+# This one must be compilable even if sound is configured out
+obj-$(CONFIG_AC97_BUS) += ac97_bus.o
+
+ifeq ($(CONFIG_SND),y)
+ obj-y += last.o
+endif
+
+soundcore-objs := sound_core.o
diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c
new file mode 100644
index 0000000..7fa37e1
--- /dev/null
+++ b/sound/ac97_bus.c
@@ -0,0 +1,76 @@
+/*
+ * Linux driver model AC97 bus interface
+ *
+ * Author: Nicolas Pitre
+ * Created: Jan 14, 2005
+ * Copyright: (C) MontaVista Software Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/string.h>
+
+/*
+ * Let drivers decide whether they want to support given codec from their
+ * probe method. Drivers have direct access to the struct snd_ac97 structure and may
+ * decide based on the id field amongst other things.
+ */
+static int ac97_bus_match(struct device *dev, struct device_driver *drv)
+{
+ return 1;
+}
+
+#ifdef CONFIG_PM
+static int ac97_bus_suspend(struct device *dev, pm_message_t state)
+{
+ int ret = 0;
+
+ if (dev->driver && dev->driver->suspend)
+ ret = dev->driver->suspend(dev, state);
+
+ return ret;
+}
+
+static int ac97_bus_resume(struct device *dev)
+{
+ int ret = 0;
+
+ if (dev->driver && dev->driver->resume)
+ ret = dev->driver->resume(dev);
+
+ return ret;
+}
+#endif /* CONFIG_PM */
+
+struct bus_type ac97_bus_type = {
+ .name = "ac97",
+ .match = ac97_bus_match,
+#ifdef CONFIG_PM
+ .suspend = ac97_bus_suspend,
+ .resume = ac97_bus_resume,
+#endif /* CONFIG_PM */
+};
+
+static int __init ac97_bus_init(void)
+{
+ return bus_register(&ac97_bus_type);
+}
+
+subsys_initcall(ac97_bus_init);
+
+static void __exit ac97_bus_exit(void)
+{
+ bus_unregister(&ac97_bus_type);
+}
+
+module_exit(ac97_bus_exit);
+
+EXPORT_SYMBOL(ac97_bus_type);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig
new file mode 100644
index 0000000..c081e18
--- /dev/null
+++ b/sound/aoa/Kconfig
@@ -0,0 +1,17 @@
+menuconfig SND_AOA
+ tristate "Apple Onboard Audio driver"
+ depends on PPC_PMAC
+ select SND_PCM
+ ---help---
+ This option enables the new driver for the various
+ Apple Onboard Audio components.
+
+if SND_AOA
+
+source "sound/aoa/fabrics/Kconfig"
+
+source "sound/aoa/codecs/Kconfig"
+
+source "sound/aoa/soundbus/Kconfig"
+
+endif # SND_AOA
diff --git a/sound/aoa/Makefile b/sound/aoa/Makefile
new file mode 100644
index 0000000..a8c037f
--- /dev/null
+++ b/sound/aoa/Makefile
@@ -0,0 +1,4 @@
+obj-$(CONFIG_SND_AOA) += core/
+obj-$(CONFIG_SND_AOA_SOUNDBUS) += soundbus/
+obj-$(CONFIG_SND_AOA) += fabrics/
+obj-$(CONFIG_SND_AOA) += codecs/
diff --git a/sound/aoa/aoa-gpio.h b/sound/aoa/aoa-gpio.h
new file mode 100644
index 0000000..ee64f5d
--- /dev/null
+++ b/sound/aoa/aoa-gpio.h
@@ -0,0 +1,81 @@
+/*
+ * Apple Onboard Audio GPIO definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#ifndef __AOA_GPIO_H
+#define __AOA_GPIO_H
+#include <linux/workqueue.h>
+#include <linux/mutex.h>
+#include <asm/prom.h>
+
+typedef void (*notify_func_t)(void *data);
+
+enum notify_type {
+ AOA_NOTIFY_HEADPHONE,
+ AOA_NOTIFY_LINE_IN,
+ AOA_NOTIFY_LINE_OUT,
+};
+
+struct gpio_runtime;
+struct gpio_methods {
+ /* for initialisation/de-initialisation of the GPIO layer */
+ void (*init)(struct gpio_runtime *rt);
+ void (*exit)(struct gpio_runtime *rt);
+
+ /* turn off headphone, speakers, lineout */
+ void (*all_amps_off)(struct gpio_runtime *rt);
+ /* turn headphone, speakers, lineout back to previous setting */
+ void (*all_amps_restore)(struct gpio_runtime *rt);
+
+ void (*set_headphone)(struct gpio_runtime *rt, int on);
+ void (*set_speakers)(struct gpio_runtime *rt, int on);
+ void (*set_lineout)(struct gpio_runtime *rt, int on);
+
+ int (*get_headphone)(struct gpio_runtime *rt);
+ int (*get_speakers)(struct gpio_runtime *rt);
+ int (*get_lineout)(struct gpio_runtime *rt);
+
+ void (*set_hw_reset)(struct gpio_runtime *rt, int on);
+
+ /* use this to be notified of any events. The notification
+ * function is passed the data, and is called in process
+ * context by the use of schedule_work.
+ * The interface for it is that setting a function to NULL
+ * removes it, and they return 0 if the operation succeeded,
+ * and -EBUSY if the notification is already assigned by
+ * someone else. */
+ int (*set_notify)(struct gpio_runtime *rt,
+ enum notify_type type,
+ notify_func_t notify,
+ void *data);
+ /* returns 0 if not plugged in, 1 if plugged in
+ * or a negative error code */
+ int (*get_detect)(struct gpio_runtime *rt,
+ enum notify_type type);
+};
+
+struct gpio_notification {
+ struct delayed_work work;
+ notify_func_t notify;
+ void *data;
+ void *gpio_private;
+ struct mutex mutex;
+};
+
+struct gpio_runtime {
+ /* to be assigned by fabric */
+ struct device_node *node;
+ /* since everyone needs this pointer anyway... */
+ struct gpio_methods *methods;
+ /* to be used by the gpio implementation */
+ int implementation_private;
+ struct gpio_notification headphone_notify;
+ struct gpio_notification line_in_notify;
+ struct gpio_notification line_out_notify;
+};
+
+#endif /* __AOA_GPIO_H */
diff --git a/sound/aoa/aoa.h b/sound/aoa/aoa.h
new file mode 100644
index 0000000..e087894
--- /dev/null
+++ b/sound/aoa/aoa.h
@@ -0,0 +1,129 @@
+/*
+ * Apple Onboard Audio definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#ifndef __AOA_H
+#define __AOA_H
+#include <asm/prom.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/asound.h>
+#include <sound/control.h>
+#include "aoa-gpio.h"
+#include "soundbus/soundbus.h"
+
+#define MAX_CODEC_NAME_LEN 32
+
+struct aoa_codec {
+ char name[MAX_CODEC_NAME_LEN];
+
+ struct module *owner;
+
+ /* called when the fabric wants to init this codec.
+ * Do alsa card manipulations from here. */
+ int (*init)(struct aoa_codec *codec);
+
+ /* called when the fabric is done with the codec.
+ * The alsa card will be cleaned up so don't bother. */
+ void (*exit)(struct aoa_codec *codec);
+
+ /* May be NULL, but can be used by the fabric.
+ * Refcounting is the codec driver's responsibility */
+ struct device_node *node;
+
+ /* assigned by fabric before init() is called, points
+ * to the soundbus device. Cannot be NULL. */
+ struct soundbus_dev *soundbus_dev;
+
+ /* assigned by the fabric before init() is called, points
+ * to the fabric's gpio runtime record for the relevant
+ * device. */
+ struct gpio_runtime *gpio;
+
+ /* assigned by the fabric before init() is called, contains
+ * a codec specific bitmask of what outputs and inputs are
+ * actually connected */
+ u32 connected;
+
+ /* data the fabric can associate with this structure */
+ void *fabric_data;
+
+ /* private! */
+ struct list_head list;
+ struct aoa_fabric *fabric;
+};
+
+/* return 0 on success */
+extern int
+aoa_codec_register(struct aoa_codec *codec);
+extern void
+aoa_codec_unregister(struct aoa_codec *codec);
+
+#define MAX_LAYOUT_NAME_LEN 32
+
+struct aoa_fabric {
+ char name[MAX_LAYOUT_NAME_LEN];
+
+ struct module *owner;
+
+ /* once codecs register, they are passed here after.
+ * They are of course not initialised, since the
+ * fabric is responsible for initialising some fields
+ * in the codec structure! */
+ int (*found_codec)(struct aoa_codec *codec);
+ /* called for each codec when it is removed,
+ * also in the case that aoa_fabric_unregister
+ * is called and all codecs are removed
+ * from this fabric.
+ * Also called if found_codec returned 0 but
+ * the codec couldn't initialise. */
+ void (*remove_codec)(struct aoa_codec *codec);
+ /* If found_codec returned 0, and the codec
+ * could be initialised, this is called. */
+ void (*attached_codec)(struct aoa_codec *codec);
+};
+
+/* return 0 on success, -EEXIST if another fabric is
+ * registered, -EALREADY if the same fabric is registered.
+ * Passing NULL can be used to test for the presence
+ * of another fabric, if -EALREADY is returned there is
+ * no other fabric present.
+ * In the case that the function returns -EALREADY
+ * and the fabric passed is not NULL, all codecs
+ * that are not assigned yet are passed to the fabric
+ * again for reconsideration. */
+extern int
+aoa_fabric_register(struct aoa_fabric *fabric, struct device *dev);
+
+/* it is vital to call this when the fabric exits!
+ * When calling, the remove_codec will be called
+ * for all codecs, unless it is NULL. */
+extern void
+aoa_fabric_unregister(struct aoa_fabric *fabric);
+
+/* if for some reason you want to get rid of a codec
+ * before the fabric is removed, use this.
+ * Note that remove_codec is called for it! */
+extern void
+aoa_fabric_unlink_codec(struct aoa_codec *codec);
+
+/* alsa help methods */
+struct aoa_card {
+ struct snd_card *alsa_card;
+};
+
+extern int aoa_snd_device_new(snd_device_type_t type,
+ void * device_data, struct snd_device_ops * ops);
+extern struct snd_card *aoa_get_card(void);
+extern int aoa_snd_ctl_add(struct snd_kcontrol* control);
+
+/* GPIO stuff */
+extern struct gpio_methods *pmf_gpio_methods;
+extern struct gpio_methods *ftr_gpio_methods;
+/* extern struct gpio_methods *map_gpio_methods; */
+
+#endif /* __AOA_H */
diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig
new file mode 100644
index 0000000..808eb11
--- /dev/null
+++ b/sound/aoa/codecs/Kconfig
@@ -0,0 +1,32 @@
+config SND_AOA_ONYX
+ tristate "support Onyx chip"
+ select I2C
+ select I2C_POWERMAC
+ ---help---
+ This option enables support for the Onyx (pcm3052)
+ codec chip found in the latest Apple machines
+ (most of those with digital audio output).
+
+#config SND_AOA_TOPAZ
+# tristate "support Topaz chips"
+# ---help---
+# This option enables support for the Topaz (CS84xx)
+# codec chips found in the latest Apple machines,
+# these chips do the digital input and output on
+# some PowerMacs.
+
+config SND_AOA_TAS
+ tristate "support TAS chips"
+ select I2C
+ select I2C_POWERMAC
+ ---help---
+ This option enables support for the tas chips
+ found in a lot of Apple Machines, especially
+ iBooks and PowerBooks without digital.
+
+config SND_AOA_TOONIE
+ tristate "support Toonie chip"
+ ---help---
+ This option enables support for the toonie codec
+ found in the Mac Mini. If you have a Mac Mini and
+ want to hear sound, select this option.
diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile
new file mode 100644
index 0000000..31cbe68
--- /dev/null
+++ b/sound/aoa/codecs/Makefile
@@ -0,0 +1,3 @@
+obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
+obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
+obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o
diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/snd-aoa-codec-onyx.c
new file mode 100644
index 0000000..6a3837d
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c
@@ -0,0 +1,1118 @@
+/*
+ * Apple Onboard Audio driver for Onyx codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ *
+ * This is a driver for the pcm3052 codec chip (codenamed Onyx)
+ * that is present in newer Apple hardware (with digital output).
+ *
+ * The Onyx codec has the following connections (listed by the bit
+ * to be used in aoa_codec.connected):
+ * 0: analog output
+ * 1: digital output
+ * 2: line input
+ * 3: microphone input
+ * Note that even though I know of no machine that has for example
+ * the digital output connected but not the analog, I have handled
+ * all the different cases in the code so that this driver may serve
+ * as a good example of what to do.
+ *
+ * NOTE: This driver assumes that there's at most one chip to be
+ * used with one alsa card, in form of creating all kinds
+ * of mixer elements without regard for their existence.
+ * But snd-aoa assumes that there's at most one card, so
+ * this means you can only have one onyx on a system. This
+ * should probably be fixed by changing the assumption of
+ * having just a single card on a system, and making the
+ * 'card' pointer accessible to anyone who needs it instead
+ * of hiding it in the aoa_snd_* functions...
+ *
+ */
+#include <linux/delay.h>
+#include <linux/module.h>
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa");
+
+#include "snd-aoa-codec-onyx.h"
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-onyx: "
+
+struct onyx {
+ /* cache registers 65 to 80, they are write-only! */
+ u8 cache[16];
+ struct i2c_client i2c;
+ struct aoa_codec codec;
+ u32 initialised:1,
+ spdif_locked:1,
+ analog_locked:1,
+ original_mute:2;
+ int open_count;
+ struct codec_info *codec_info;
+
+ /* mutex serializes concurrent access to the device
+ * and this structure.
+ */
+ struct mutex mutex;
+};
+#define codec_to_onyx(c) container_of(c, struct onyx, codec)
+
+/* both return 0 if all ok, else on error */
+static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value)
+{
+ s32 v;
+
+ if (reg != ONYX_REG_CONTROL) {
+ *value = onyx->cache[reg-FIRSTREGISTER];
+ return 0;
+ }
+ v = i2c_smbus_read_byte_data(&onyx->i2c, reg);
+ if (v < 0)
+ return -1;
+ *value = (u8)v;
+ onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value;
+ return 0;
+}
+
+static int onyx_write_register(struct onyx *onyx, u8 reg, u8 value)
+{
+ int result;
+
+ result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value);
+ if (!result)
+ onyx->cache[reg-FIRSTREGISTER] = value;
+ return result;
+}
+
+/* alsa stuff */
+
+static int onyx_dev_register(struct snd_device *dev)
+{
+ return 0;
+}
+
+static struct snd_device_ops ops = {
+ .dev_register = onyx_dev_register,
+};
+
+/* this is necessary because most alsa mixer programs
+ * can't properly handle the negative range */
+#define VOLUME_RANGE_SHIFT 128
+
+static int onyx_snd_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = -128 + VOLUME_RANGE_SHIFT;
+ uinfo->value.integer.max = -1 + VOLUME_RANGE_SHIFT;
+ return 0;
+}
+
+static int onyx_snd_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ s8 l, r;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] = l + VOLUME_RANGE_SHIFT;
+ ucontrol->value.integer.value[1] = r + VOLUME_RANGE_SHIFT;
+
+ return 0;
+}
+
+static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ s8 l, r;
+
+ if (ucontrol->value.integer.value[0] < -128 + VOLUME_RANGE_SHIFT ||
+ ucontrol->value.integer.value[0] > -1 + VOLUME_RANGE_SHIFT)
+ return -EINVAL;
+ if (ucontrol->value.integer.value[1] < -128 + VOLUME_RANGE_SHIFT ||
+ ucontrol->value.integer.value[1] > -1 + VOLUME_RANGE_SHIFT)
+ return -EINVAL;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
+ onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
+
+ if (l + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[0] &&
+ r + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[1]) {
+ mutex_unlock(&onyx->mutex);
+ return 0;
+ }
+
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT,
+ ucontrol->value.integer.value[0]
+ - VOLUME_RANGE_SHIFT);
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT,
+ ucontrol->value.integer.value[1]
+ - VOLUME_RANGE_SHIFT);
+ mutex_unlock(&onyx->mutex);
+
+ return 1;
+}
+
+static struct snd_kcontrol_new volume_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_vol_info,
+ .get = onyx_snd_vol_get,
+ .put = onyx_snd_vol_put,
+};
+
+/* like above, this is necessary because a lot
+ * of alsa mixer programs don't handle ranges
+ * that don't start at 0 properly.
+ * even alsamixer is one of them... */
+#define INPUTGAIN_RANGE_SHIFT (-3)
+
+static int onyx_snd_inputgain_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 3 + INPUTGAIN_RANGE_SHIFT;
+ uinfo->value.integer.max = 28 + INPUTGAIN_RANGE_SHIFT;
+ return 0;
+}
+
+static int onyx_snd_inputgain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 ig;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &ig);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] =
+ (ig & ONYX_ADC_PGA_GAIN_MASK) + INPUTGAIN_RANGE_SHIFT;
+
+ return 0;
+}
+
+static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v, n;
+
+ if (ucontrol->value.integer.value[0] < 3 + INPUTGAIN_RANGE_SHIFT ||
+ ucontrol->value.integer.value[0] > 28 + INPUTGAIN_RANGE_SHIFT)
+ return -EINVAL;
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+ n = v;
+ n &= ~ONYX_ADC_PGA_GAIN_MASK;
+ n |= (ucontrol->value.integer.value[0] - INPUTGAIN_RANGE_SHIFT)
+ & ONYX_ADC_PGA_GAIN_MASK;
+ onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, n);
+ mutex_unlock(&onyx->mutex);
+
+ return n != v;
+}
+
+static struct snd_kcontrol_new inputgain_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Capture Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_inputgain_info,
+ .get = onyx_snd_inputgain_get,
+ .put = onyx_snd_inputgain_put,
+};
+
+static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "Line-In", "Microphone" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int onyx_snd_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ s8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.enumerated.item[0] = !!(v&ONYX_ADC_INPUT_MIC);
+
+ return 0;
+}
+
+static void onyx_set_capture_source(struct onyx *onyx, int mic)
+{
+ s8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+ v &= ~ONYX_ADC_INPUT_MIC;
+ if (mic)
+ v |= ONYX_ADC_INPUT_MIC;
+ onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, v);
+ mutex_unlock(&onyx->mutex);
+}
+
+static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ onyx_set_capture_source(snd_kcontrol_chip(kcontrol),
+ ucontrol->value.enumerated.item[0]);
+ return 1;
+}
+
+static struct snd_kcontrol_new capture_source_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* If we name this 'Input Source', it properly shows up in
+ * alsamixer as a selection, * but it's shown under the
+ * 'Playback' category.
+ * If I name it 'Capture Source', it shows up in strange
+ * ways (two bools of which one can be selected at a
+ * time) but at least it's shown in the 'Capture'
+ * category.
+ * I was told that this was due to backward compatibility,
+ * but I don't understand then why the mangling is *not*
+ * done when I name it "Input Source".....
+ */
+ .name = "Capture Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_capture_source_info,
+ .get = onyx_snd_capture_source_get,
+ .put = onyx_snd_capture_source_put,
+};
+
+#define onyx_snd_mute_info snd_ctl_boolean_stereo_info
+
+static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 c;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] = !(c & ONYX_MUTE_LEFT);
+ ucontrol->value.integer.value[1] = !(c & ONYX_MUTE_RIGHT);
+
+ return 0;
+}
+
+static int onyx_snd_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v = 0, c = 0;
+ int err = -EBUSY;
+
+ mutex_lock(&onyx->mutex);
+ if (onyx->analog_locked)
+ goto out_unlock;
+
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+ c = v;
+ c &= ~(ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT);
+ if (!ucontrol->value.integer.value[0])
+ c |= ONYX_MUTE_LEFT;
+ if (!ucontrol->value.integer.value[1])
+ c |= ONYX_MUTE_RIGHT;
+ err = onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, c);
+
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? (v != c) : err;
+}
+
+static struct snd_kcontrol_new mute_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = onyx_snd_mute_info,
+ .get = onyx_snd_mute_get,
+ .put = onyx_snd_mute_put,
+};
+
+
+#define onyx_snd_single_bit_info snd_ctl_boolean_mono_info
+
+#define FLAG_POLARITY_INVERT 1
+#define FLAG_SPDIFLOCK 2
+
+static int onyx_snd_single_bit_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 c;
+ long int pv = kcontrol->private_value;
+ u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
+ u8 address = (pv >> 8) & 0xff;
+ u8 mask = pv & 0xff;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, address, &c);
+ mutex_unlock(&onyx->mutex);
+
+ ucontrol->value.integer.value[0] = !!(c & mask) ^ polarity;
+
+ return 0;
+}
+
+static int onyx_snd_single_bit_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v = 0, c = 0;
+ int err;
+ long int pv = kcontrol->private_value;
+ u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
+ u8 spdiflock = (pv >> 16) & FLAG_SPDIFLOCK;
+ u8 address = (pv >> 8) & 0xff;
+ u8 mask = pv & 0xff;
+
+ mutex_lock(&onyx->mutex);
+ if (spdiflock && onyx->spdif_locked) {
+ /* even if alsamixer doesn't care.. */
+ err = -EBUSY;
+ goto out_unlock;
+ }
+ onyx_read_register(onyx, address, &v);
+ c = v;
+ c &= ~(mask);
+ if (!!ucontrol->value.integer.value[0] ^ polarity)
+ c |= mask;
+ err = onyx_write_register(onyx, address, c);
+
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? (v != c) : err;
+}
+
+#define SINGLE_BIT(n, type, description, address, mask, flags) \
+static struct snd_kcontrol_new n##_control = { \
+ .iface = SNDRV_CTL_ELEM_IFACE_##type, \
+ .name = description, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = onyx_snd_single_bit_info, \
+ .get = onyx_snd_single_bit_get, \
+ .put = onyx_snd_single_bit_put, \
+ .private_value = (flags << 16) | (address << 8) | mask \
+}
+
+SINGLE_BIT(spdif,
+ MIXER,
+ SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
+ ONYX_REG_DIG_INFO4,
+ ONYX_SPDIF_ENABLE,
+ FLAG_SPDIFLOCK);
+SINGLE_BIT(ovr1,
+ MIXER,
+ "Oversampling Rate",
+ ONYX_REG_DAC_CONTROL,
+ ONYX_OVR1,
+ 0);
+SINGLE_BIT(flt0,
+ MIXER,
+ "Fast Digital Filter Rolloff",
+ ONYX_REG_DAC_FILTER,
+ ONYX_ROLLOFF_FAST,
+ FLAG_POLARITY_INVERT);
+SINGLE_BIT(hpf,
+ MIXER,
+ "Highpass Filter",
+ ONYX_REG_ADC_HPF_BYPASS,
+ ONYX_HPF_DISABLE,
+ FLAG_POLARITY_INVERT);
+SINGLE_BIT(dm12,
+ MIXER,
+ "Digital De-Emphasis",
+ ONYX_REG_DAC_DEEMPH,
+ ONYX_DIGDEEMPH_CTRL,
+ 0);
+
+static int onyx_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ /* datasheet page 30, all others are 0 */
+ ucontrol->value.iec958.status[0] = 0x3e;
+ ucontrol->value.iec958.status[1] = 0xff;
+
+ ucontrol->value.iec958.status[3] = 0x3f;
+ ucontrol->value.iec958.status[4] = 0x0f;
+
+ return 0;
+}
+
+static struct snd_kcontrol_new onyx_spdif_mask = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = onyx_spdif_info,
+ .get = onyx_spdif_mask_get,
+};
+
+static int onyx_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
+ ucontrol->value.iec958.status[0] = v & 0x3e;
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO2, &v);
+ ucontrol->value.iec958.status[1] = v;
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
+ ucontrol->value.iec958.status[3] = v & 0x3f;
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ ucontrol->value.iec958.status[4] = v & 0x0f;
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+static int onyx_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+ u8 v;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
+ v = (v & ~0x3e) | (ucontrol->value.iec958.status[0] & 0x3e);
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO1, v);
+
+ v = ucontrol->value.iec958.status[1];
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO2, v);
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
+ v = (v & ~0x3f) | (ucontrol->value.iec958.status[3] & 0x3f);
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO3, v);
+
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ v = (v & ~0x0f) | (ucontrol->value.iec958.status[4] & 0x0f);
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
+ mutex_unlock(&onyx->mutex);
+
+ return 1;
+}
+
+static struct snd_kcontrol_new onyx_spdif_ctrl = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = onyx_spdif_info,
+ .get = onyx_spdif_get,
+ .put = onyx_spdif_put,
+};
+
+/* our registers */
+
+static u8 register_map[] = {
+ ONYX_REG_DAC_ATTEN_LEFT,
+ ONYX_REG_DAC_ATTEN_RIGHT,
+ ONYX_REG_CONTROL,
+ ONYX_REG_DAC_CONTROL,
+ ONYX_REG_DAC_DEEMPH,
+ ONYX_REG_DAC_FILTER,
+ ONYX_REG_DAC_OUTPHASE,
+ ONYX_REG_ADC_CONTROL,
+ ONYX_REG_ADC_HPF_BYPASS,
+ ONYX_REG_DIG_INFO1,
+ ONYX_REG_DIG_INFO2,
+ ONYX_REG_DIG_INFO3,
+ ONYX_REG_DIG_INFO4
+};
+
+static u8 initial_values[ARRAY_SIZE(register_map)] = {
+ 0x80, 0x80, /* muted */
+ ONYX_MRST | ONYX_SRST, /* but handled specially! */
+ ONYX_MUTE_LEFT | ONYX_MUTE_RIGHT,
+ 0, /* no deemphasis */
+ ONYX_DAC_FILTER_ALWAYS,
+ ONYX_OUTPHASE_INVERTED,
+ (-1 /*dB*/ + 8) & 0xF, /* line in selected, -1 dB gain*/
+ ONYX_ADC_HPF_ALWAYS,
+ (1<<2), /* pcm audio */
+ 2, /* category: pcm coder */
+ 0, /* sampling frequency 44.1 kHz, clock accuracy level II */
+ 1 /* 24 bit depth */
+};
+
+/* reset registers of chip, either to initial or to previous values */
+static int onyx_register_init(struct onyx *onyx)
+{
+ int i;
+ u8 val;
+ u8 regs[sizeof(initial_values)];
+
+ if (!onyx->initialised) {
+ memcpy(regs, initial_values, sizeof(initial_values));
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &val))
+ return -1;
+ val &= ~ONYX_SILICONVERSION;
+ val |= initial_values[3];
+ regs[3] = val;
+ } else {
+ for (i=0; i<sizeof(register_map); i++)
+ regs[i] = onyx->cache[register_map[i]-FIRSTREGISTER];
+ }
+
+ for (i=0; i<sizeof(register_map); i++) {
+ if (onyx_write_register(onyx, register_map[i], regs[i]))
+ return -1;
+ }
+ onyx->initialised = 1;
+ return 0;
+}
+
+static struct transfer_info onyx_transfers[] = {
+ /* this is first so we can skip it if no input is present...
+ * No hardware exists with that, but it's here as an example
+ * of what to do :) */
+ {
+ /* analog input */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .transfer_in = 1,
+ .must_be_clock_source = 0,
+ .tag = 0,
+ },
+ {
+ /* if analog and digital are currently off, anything should go,
+ * so this entry describes everything we can do... */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+ | SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+#endif
+ ,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .tag = 0,
+ },
+ {
+ /* analog output */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .transfer_in = 0,
+ .must_be_clock_source = 0,
+ .tag = 1,
+ },
+ {
+ /* digital pcm output, also possible for analog out */
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .transfer_in = 0,
+ .must_be_clock_source = 0,
+ .tag = 2,
+ },
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+ /* Once alsa gets supports for this kind of thing we can add it... */
+ {
+ /* digital compressed output */
+ .formats = SNDRV_PCM_FMTBIT_COMPRESSED_16BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .tag = 2,
+ },
+#endif
+ {}
+};
+
+static int onyx_usable(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out)
+{
+ u8 v;
+ struct onyx *onyx = cii->codec_data;
+ int spdif_enabled, analog_enabled;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ spdif_enabled = !!(v & ONYX_SPDIF_ENABLE);
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+ analog_enabled =
+ (v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT))
+ != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT);
+ mutex_unlock(&onyx->mutex);
+
+ switch (ti->tag) {
+ case 0: return 1;
+ case 1: return analog_enabled;
+ case 2: return spdif_enabled;
+ }
+ return 1;
+}
+
+static int onyx_prepare(struct codec_info_item *cii,
+ struct bus_info *bi,
+ struct snd_pcm_substream *substream)
+{
+ u8 v;
+ struct onyx *onyx = cii->codec_data;
+ int err = -EBUSY;
+
+ mutex_lock(&onyx->mutex);
+
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+ if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) {
+ /* mute and lock analog output */
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+ if (onyx_write_register(onyx,
+ ONYX_REG_DAC_CONTROL,
+ v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT))
+ goto out_unlock;
+ onyx->analog_locked = 1;
+ err = 0;
+ goto out_unlock;
+ }
+#endif
+ switch (substream->runtime->rate) {
+ case 32000:
+ case 44100:
+ case 48000:
+ /* these rates are ok for all outputs */
+ /* FIXME: program spdif channel control bits here so that
+ * userspace doesn't have to if it only plays pcm! */
+ err = 0;
+ goto out_unlock;
+ default:
+ /* got some rate that the digital output can't do,
+ * so disable and lock it */
+ onyx_read_register(cii->codec_data, ONYX_REG_DIG_INFO4, &v);
+ if (onyx_write_register(onyx,
+ ONYX_REG_DIG_INFO4,
+ v & ~ONYX_SPDIF_ENABLE))
+ goto out_unlock;
+ onyx->spdif_locked = 1;
+ err = 0;
+ goto out_unlock;
+ }
+
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return err;
+}
+
+static int onyx_open(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream)
+{
+ struct onyx *onyx = cii->codec_data;
+
+ mutex_lock(&onyx->mutex);
+ onyx->open_count++;
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+static int onyx_close(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream)
+{
+ struct onyx *onyx = cii->codec_data;
+
+ mutex_lock(&onyx->mutex);
+ onyx->open_count--;
+ if (!onyx->open_count)
+ onyx->spdif_locked = onyx->analog_locked = 0;
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+static int onyx_switch_clock(struct codec_info_item *cii,
+ enum clock_switch what)
+{
+ struct onyx *onyx = cii->codec_data;
+
+ mutex_lock(&onyx->mutex);
+ /* this *MUST* be more elaborate later... */
+ switch (what) {
+ case CLOCK_SWITCH_PREPARE_SLAVE:
+ onyx->codec.gpio->methods->all_amps_off(onyx->codec.gpio);
+ break;
+ case CLOCK_SWITCH_SLAVE:
+ onyx->codec.gpio->methods->all_amps_restore(onyx->codec.gpio);
+ break;
+ default: /* silence warning */
+ break;
+ }
+ mutex_unlock(&onyx->mutex);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+
+static int onyx_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+ struct onyx *onyx = cii->codec_data;
+ u8 v;
+ int err = -ENXIO;
+
+ mutex_lock(&onyx->mutex);
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
+ goto out_unlock;
+ onyx_write_register(onyx, ONYX_REG_CONTROL, v | ONYX_ADPSV | ONYX_DAPSV);
+ /* Apple does a sleep here but the datasheet says to do it on resume */
+ err = 0;
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return err;
+}
+
+static int onyx_resume(struct codec_info_item *cii)
+{
+ struct onyx *onyx = cii->codec_data;
+ u8 v;
+ int err = -ENXIO;
+
+ mutex_lock(&onyx->mutex);
+
+ /* reset codec */
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+
+ /* take codec out of suspend (if it still is after reset) */
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
+ goto out_unlock;
+ onyx_write_register(onyx, ONYX_REG_CONTROL, v & ~(ONYX_ADPSV | ONYX_DAPSV));
+ /* FIXME: should divide by sample rate, but 8k is the lowest we go */
+ msleep(2205000/8000);
+ /* reset all values */
+ onyx_register_init(onyx);
+ err = 0;
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return err;
+}
+
+#endif /* CONFIG_PM */
+
+static struct codec_info onyx_codec_info = {
+ .transfers = onyx_transfers,
+ .sysclock_factor = 256,
+ .bus_factor = 64,
+ .owner = THIS_MODULE,
+ .usable = onyx_usable,
+ .prepare = onyx_prepare,
+ .open = onyx_open,
+ .close = onyx_close,
+ .switch_clock = onyx_switch_clock,
+#ifdef CONFIG_PM
+ .suspend = onyx_suspend,
+ .resume = onyx_resume,
+#endif
+};
+
+static int onyx_init_codec(struct aoa_codec *codec)
+{
+ struct onyx *onyx = codec_to_onyx(codec);
+ struct snd_kcontrol *ctl;
+ struct codec_info *ci = &onyx_codec_info;
+ u8 v;
+ int err;
+
+ if (!onyx->codec.gpio || !onyx->codec.gpio->methods) {
+ printk(KERN_ERR PFX "gpios not assigned!!\n");
+ return -EINVAL;
+ }
+
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1);
+ msleep(1);
+ onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+ msleep(1);
+
+ if (onyx_register_init(onyx)) {
+ printk(KERN_ERR PFX "failed to initialise onyx registers\n");
+ return -ENODEV;
+ }
+
+ if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, onyx, &ops)) {
+ printk(KERN_ERR PFX "failed to create onyx snd device!\n");
+ return -ENODEV;
+ }
+
+ /* nothing connected? what a joke! */
+ if ((onyx->codec.connected & 0xF) == 0)
+ return -ENOTCONN;
+
+ /* if no inputs are present... */
+ if ((onyx->codec.connected & 0xC) == 0) {
+ if (!onyx->codec_info)
+ onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
+ if (!onyx->codec_info)
+ return -ENOMEM;
+ ci = onyx->codec_info;
+ *ci = onyx_codec_info;
+ ci->transfers++;
+ }
+
+ /* if no outputs are present... */
+ if ((onyx->codec.connected & 3) == 0) {
+ if (!onyx->codec_info)
+ onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
+ if (!onyx->codec_info)
+ return -ENOMEM;
+ ci = onyx->codec_info;
+ /* this is fine as there have to be inputs
+ * if we end up in this part of the code */
+ *ci = onyx_codec_info;
+ ci->transfers[1].formats = 0;
+ }
+
+ if (onyx->codec.soundbus_dev->attach_codec(onyx->codec.soundbus_dev,
+ aoa_get_card(),
+ ci, onyx)) {
+ printk(KERN_ERR PFX "error creating onyx pcm\n");
+ return -ENODEV;
+ }
+#define ADDCTL(n) \
+ do { \
+ ctl = snd_ctl_new1(&n, onyx); \
+ if (ctl) { \
+ ctl->id.device = \
+ onyx->codec.soundbus_dev->pcm->device; \
+ err = aoa_snd_ctl_add(ctl); \
+ if (err) \
+ goto error; \
+ } \
+ } while (0)
+
+ if (onyx->codec.soundbus_dev->pcm) {
+ /* give the user appropriate controls
+ * depending on what inputs are connected */
+ if ((onyx->codec.connected & 0xC) == 0xC)
+ ADDCTL(capture_source_control);
+ else if (onyx->codec.connected & 4)
+ onyx_set_capture_source(onyx, 0);
+ else
+ onyx_set_capture_source(onyx, 1);
+ if (onyx->codec.connected & 0xC)
+ ADDCTL(inputgain_control);
+
+ /* depending on what output is connected,
+ * give the user appropriate controls */
+ if (onyx->codec.connected & 1) {
+ ADDCTL(volume_control);
+ ADDCTL(mute_control);
+ ADDCTL(ovr1_control);
+ ADDCTL(flt0_control);
+ ADDCTL(hpf_control);
+ ADDCTL(dm12_control);
+ /* spdif control defaults to off */
+ }
+ if (onyx->codec.connected & 2) {
+ ADDCTL(onyx_spdif_mask);
+ ADDCTL(onyx_spdif_ctrl);
+ }
+ if ((onyx->codec.connected & 3) == 3)
+ ADDCTL(spdif_control);
+ /* if only S/PDIF is connected, enable it unconditionally */
+ if ((onyx->codec.connected & 3) == 2) {
+ onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+ v |= ONYX_SPDIF_ENABLE;
+ onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
+ }
+ }
+#undef ADDCTL
+ printk(KERN_INFO PFX "attached to onyx codec via i2c\n");
+
+ return 0;
+ error:
+ onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
+ snd_device_free(aoa_get_card(), onyx);
+ return err;
+}
+
+static void onyx_exit_codec(struct aoa_codec *codec)
+{
+ struct onyx *onyx = codec_to_onyx(codec);
+
+ if (!onyx->codec.soundbus_dev) {
+ printk(KERN_ERR PFX "onyx_exit_codec called without soundbus_dev!\n");
+ return;
+ }
+ onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
+}
+
+static struct i2c_driver onyx_driver;
+
+static int onyx_create(struct i2c_adapter *adapter,
+ struct device_node *node,
+ int addr)
+{
+ struct onyx *onyx;
+ u8 dummy;
+
+ onyx = kzalloc(sizeof(struct onyx), GFP_KERNEL);
+
+ if (!onyx)
+ return -ENOMEM;
+
+ mutex_init(&onyx->mutex);
+ onyx->i2c.driver = &onyx_driver;
+ onyx->i2c.adapter = adapter;
+ onyx->i2c.addr = addr & 0x7f;
+ strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE);
+
+ if (i2c_attach_client(&onyx->i2c)) {
+ printk(KERN_ERR PFX "failed to attach to i2c\n");
+ goto fail;
+ }
+
+ /* we try to read from register ONYX_REG_CONTROL
+ * to check if the codec is present */
+ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) {
+ i2c_detach_client(&onyx->i2c);
+ printk(KERN_ERR PFX "failed to read control register\n");
+ goto fail;
+ }
+
+ strlcpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN);
+ onyx->codec.owner = THIS_MODULE;
+ onyx->codec.init = onyx_init_codec;
+ onyx->codec.exit = onyx_exit_codec;
+ onyx->codec.node = of_node_get(node);
+
+ if (aoa_codec_register(&onyx->codec)) {
+ i2c_detach_client(&onyx->i2c);
+ goto fail;
+ }
+ printk(KERN_DEBUG PFX "created and attached onyx instance\n");
+ return 0;
+ fail:
+ kfree(onyx);
+ return -EINVAL;
+}
+
+static int onyx_i2c_attach(struct i2c_adapter *adapter)
+{
+ struct device_node *busnode, *dev = NULL;
+ struct pmac_i2c_bus *bus;
+
+ bus = pmac_i2c_adapter_to_bus(adapter);
+ if (bus == NULL)
+ return -ENODEV;
+ busnode = pmac_i2c_get_bus_node(bus);
+
+ while ((dev = of_get_next_child(busnode, dev)) != NULL) {
+ if (of_device_is_compatible(dev, "pcm3052")) {
+ const u32 *addr;
+ printk(KERN_DEBUG PFX "found pcm3052\n");
+ addr = of_get_property(dev, "reg", NULL);
+ if (!addr)
+ return -ENODEV;
+ return onyx_create(adapter, dev, (*addr)>>1);
+ }
+ }
+
+ /* if that didn't work, try desperate mode for older
+ * machines that have stuff missing from the device tree */
+
+ if (!of_device_is_compatible(busnode, "k2-i2c"))
+ return -ENODEV;
+
+ printk(KERN_DEBUG PFX "found k2-i2c, checking if onyx chip is on it\n");
+ /* probe both possible addresses for the onyx chip */
+ if (onyx_create(adapter, NULL, 0x46) == 0)
+ return 0;
+ return onyx_create(adapter, NULL, 0x47);
+}
+
+static int onyx_i2c_detach(struct i2c_client *client)
+{
+ struct onyx *onyx = container_of(client, struct onyx, i2c);
+ int err;
+
+ if ((err = i2c_detach_client(client)))
+ return err;
+ aoa_codec_unregister(&onyx->codec);
+ of_node_put(onyx->codec.node);
+ if (onyx->codec_info)
+ kfree(onyx->codec_info);
+ kfree(onyx);
+ return 0;
+}
+
+static struct i2c_driver onyx_driver = {
+ .driver = {
+ .name = "aoa_codec_onyx",
+ .owner = THIS_MODULE,
+ },
+ .attach_adapter = onyx_i2c_attach,
+ .detach_client = onyx_i2c_detach,
+};
+
+static int __init onyx_init(void)
+{
+ return i2c_add_driver(&onyx_driver);
+}
+
+static void __exit onyx_exit(void)
+{
+ i2c_del_driver(&onyx_driver);
+}
+
+module_init(onyx_init);
+module_exit(onyx_exit);
diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.h b/sound/aoa/codecs/snd-aoa-codec-onyx.h
new file mode 100644
index 0000000..ffd2025
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.h
@@ -0,0 +1,75 @@
+/*
+ * Apple Onboard Audio driver for Onyx codec (header)
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SND_AOA_CODEC_ONYX_H
+#define __SND_AOA_CODEC_ONYX_H
+#include <stddef.h>
+#include <linux/i2c.h>
+#include <asm/pmac_low_i2c.h>
+#include <asm/prom.h>
+
+/* PCM3052 register definitions */
+
+/* the attenuation registers take values from
+ * -1 (0dB) to -127 (-63.0 dB) or others (muted) */
+#define ONYX_REG_DAC_ATTEN_LEFT 65
+#define FIRSTREGISTER ONYX_REG_DAC_ATTEN_LEFT
+#define ONYX_REG_DAC_ATTEN_RIGHT 66
+
+#define ONYX_REG_CONTROL 67
+# define ONYX_MRST (1<<7)
+# define ONYX_SRST (1<<6)
+# define ONYX_ADPSV (1<<5)
+# define ONYX_DAPSV (1<<4)
+# define ONYX_SILICONVERSION (1<<0)
+/* all others reserved */
+
+#define ONYX_REG_DAC_CONTROL 68
+# define ONYX_OVR1 (1<<6)
+# define ONYX_MUTE_RIGHT (1<<1)
+# define ONYX_MUTE_LEFT (1<<0)
+
+#define ONYX_REG_DAC_DEEMPH 69
+# define ONYX_DIGDEEMPH_SHIFT 5
+# define ONYX_DIGDEEMPH_MASK (3<<ONYX_DIGDEEMPH_SHIFT)
+# define ONYX_DIGDEEMPH_CTRL (1<<4)
+
+#define ONYX_REG_DAC_FILTER 70
+# define ONYX_ROLLOFF_FAST (1<<5)
+# define ONYX_DAC_FILTER_ALWAYS (1<<2)
+
+#define ONYX_REG_DAC_OUTPHASE 71
+# define ONYX_OUTPHASE_INVERTED (1<<0)
+
+#define ONYX_REG_ADC_CONTROL 72
+# define ONYX_ADC_INPUT_MIC (1<<5)
+/* 8 + input gain in dB, valid range for input gain is -4 .. 20 dB */
+# define ONYX_ADC_PGA_GAIN_MASK 0x1f
+
+#define ONYX_REG_ADC_HPF_BYPASS 75
+# define ONYX_HPF_DISABLE (1<<3)
+# define ONYX_ADC_HPF_ALWAYS (1<<2)
+
+#define ONYX_REG_DIG_INFO1 77
+# define ONYX_MASK_DIN_TO_BPZ (1<<7)
+/* bits 1-5 control channel bits 1-5 */
+# define ONYX_DIGOUT_DISABLE (1<<0)
+
+#define ONYX_REG_DIG_INFO2 78
+/* controls channel bits 8-15 */
+
+#define ONYX_REG_DIG_INFO3 79
+/* control channel bits 24-29, high 2 bits reserved */
+
+#define ONYX_REG_DIG_INFO4 80
+# define ONYX_VALIDL (1<<7)
+# define ONYX_VALIDR (1<<6)
+# define ONYX_SPDIF_ENABLE (1<<5)
+/* lower 4 bits control bits 32-35 of channel control and word length */
+# define ONYX_WORDLEN_MASK (0xF)
+
+#endif /* __SND_AOA_CODEC_ONYX_H */
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h b/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h
new file mode 100644
index 0000000..69b6113
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h
@@ -0,0 +1,134 @@
+/*
+ * This file is only included exactly once!
+ *
+ * The tables here are derived from the tas3004 datasheet,
+ * modulo typo corrections and some smoothing...
+ */
+
+#define TAS3004_TREBLE_MIN 0
+#define TAS3004_TREBLE_MAX 72
+#define TAS3004_BASS_MIN 0
+#define TAS3004_BASS_MAX 72
+#define TAS3004_TREBLE_ZERO 36
+#define TAS3004_BASS_ZERO 36
+
+static u8 tas3004_treble_table[] = {
+ 150, /* -18 dB */
+ 149,
+ 148,
+ 147,
+ 146,
+ 145,
+ 144,
+ 143,
+ 142,
+ 141,
+ 140,
+ 139,
+ 138,
+ 137,
+ 136,
+ 135,
+ 134,
+ 133,
+ 132,
+ 131,
+ 130,
+ 129,
+ 128,
+ 127,
+ 126,
+ 125,
+ 124,
+ 123,
+ 122,
+ 121,
+ 120,
+ 119,
+ 118,
+ 117,
+ 116,
+ 115,
+ 114, /* 0 dB */
+ 113,
+ 112,
+ 111,
+ 109,
+ 108,
+ 107,
+ 105,
+ 104,
+ 103,
+ 101,
+ 99,
+ 98,
+ 96,
+ 93,
+ 91,
+ 89,
+ 86,
+ 83,
+ 81,
+ 77,
+ 74,
+ 71,
+ 67,
+ 63,
+ 59,
+ 54,
+ 49,
+ 44,
+ 38,
+ 32,
+ 26,
+ 19,
+ 10,
+ 4,
+ 2,
+ 1, /* +18 dB */
+};
+
+static inline u8 tas3004_treble(int idx)
+{
+ return tas3004_treble_table[idx];
+}
+
+/* I only save the difference here to the treble table
+ * so that the binary is smaller...
+ * I have also ignored completely differences of
+ * +/- 1
+ */
+static s8 tas3004_bass_diff_to_treble[] = {
+ 2, /* 7 dB, offset 50 */
+ 2,
+ 2,
+ 2,
+ 2,
+ 1,
+ 2,
+ 2,
+ 2,
+ 3,
+ 4,
+ 4,
+ 5,
+ 6,
+ 7,
+ 8,
+ 9,
+ 10,
+ 11,
+ 14,
+ 13,
+ 8,
+ 1, /* 18 dB */
+};
+
+static inline u8 tas3004_bass(int idx)
+{
+ u8 result = tas3004_treble_table[idx];
+
+ if (idx >= 50)
+ result += tas3004_bass_diff_to_treble[idx-50];
+ return result;
+}
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h b/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h
new file mode 100644
index 0000000..4cfa675
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h
@@ -0,0 +1,209 @@
+/*
+ This is the program used to generate below table.
+
+#include <stdio.h>
+#include <math.h>
+int main() {
+ int dB2;
+ printf("/" "* This file is only included exactly once!\n");
+ printf(" *\n");
+ printf(" * If they'd only tell us that generating this table was\n");
+ printf(" * as easy as calculating\n");
+ printf(" * hwvalue = 1048576.0*exp(0.057564628*dB*2)\n");
+ printf(" * :) *" "/\n");
+ printf("static int tas_gaintable[] = {\n");
+ printf(" 0x000000, /" "* -infinity dB *" "/\n");
+ for (dB2=-140;dB2<=36;dB2++)
+ printf(" 0x%.6x, /" "* %-02.1f dB *" "/\n", (int)(1048576.0*exp(0.057564628*dB2)), dB2/2.0);
+ printf("};\n\n");
+}
+
+*/
+
+/* This file is only included exactly once!
+ *
+ * If they'd only tell us that generating this table was
+ * as easy as calculating
+ * hwvalue = 1048576.0*exp(0.057564628*dB*2)
+ * :) */
+static int tas_gaintable[] = {
+ 0x000000, /* -infinity dB */
+ 0x00014b, /* -70.0 dB */
+ 0x00015f, /* -69.5 dB */
+ 0x000174, /* -69.0 dB */
+ 0x00018a, /* -68.5 dB */
+ 0x0001a1, /* -68.0 dB */
+ 0x0001ba, /* -67.5 dB */
+ 0x0001d4, /* -67.0 dB */
+ 0x0001f0, /* -66.5 dB */
+ 0x00020d, /* -66.0 dB */
+ 0x00022c, /* -65.5 dB */
+ 0x00024d, /* -65.0 dB */
+ 0x000270, /* -64.5 dB */
+ 0x000295, /* -64.0 dB */
+ 0x0002bc, /* -63.5 dB */
+ 0x0002e6, /* -63.0 dB */
+ 0x000312, /* -62.5 dB */
+ 0x000340, /* -62.0 dB */
+ 0x000372, /* -61.5 dB */
+ 0x0003a6, /* -61.0 dB */
+ 0x0003dd, /* -60.5 dB */
+ 0x000418, /* -60.0 dB */
+ 0x000456, /* -59.5 dB */
+ 0x000498, /* -59.0 dB */
+ 0x0004de, /* -58.5 dB */
+ 0x000528, /* -58.0 dB */
+ 0x000576, /* -57.5 dB */
+ 0x0005c9, /* -57.0 dB */
+ 0x000620, /* -56.5 dB */
+ 0x00067d, /* -56.0 dB */
+ 0x0006e0, /* -55.5 dB */
+ 0x000748, /* -55.0 dB */
+ 0x0007b7, /* -54.5 dB */
+ 0x00082c, /* -54.0 dB */
+ 0x0008a8, /* -53.5 dB */
+ 0x00092b, /* -53.0 dB */
+ 0x0009b6, /* -52.5 dB */
+ 0x000a49, /* -52.0 dB */
+ 0x000ae5, /* -51.5 dB */
+ 0x000b8b, /* -51.0 dB */
+ 0x000c3a, /* -50.5 dB */
+ 0x000cf3, /* -50.0 dB */
+ 0x000db8, /* -49.5 dB */
+ 0x000e88, /* -49.0 dB */
+ 0x000f64, /* -48.5 dB */
+ 0x00104e, /* -48.0 dB */
+ 0x001145, /* -47.5 dB */
+ 0x00124b, /* -47.0 dB */
+ 0x001361, /* -46.5 dB */
+ 0x001487, /* -46.0 dB */
+ 0x0015be, /* -45.5 dB */
+ 0x001708, /* -45.0 dB */
+ 0x001865, /* -44.5 dB */
+ 0x0019d8, /* -44.0 dB */
+ 0x001b60, /* -43.5 dB */
+ 0x001cff, /* -43.0 dB */
+ 0x001eb7, /* -42.5 dB */
+ 0x002089, /* -42.0 dB */
+ 0x002276, /* -41.5 dB */
+ 0x002481, /* -41.0 dB */
+ 0x0026ab, /* -40.5 dB */
+ 0x0028f5, /* -40.0 dB */
+ 0x002b63, /* -39.5 dB */
+ 0x002df5, /* -39.0 dB */
+ 0x0030ae, /* -38.5 dB */
+ 0x003390, /* -38.0 dB */
+ 0x00369e, /* -37.5 dB */
+ 0x0039db, /* -37.0 dB */
+ 0x003d49, /* -36.5 dB */
+ 0x0040ea, /* -36.0 dB */
+ 0x0044c3, /* -35.5 dB */
+ 0x0048d6, /* -35.0 dB */
+ 0x004d27, /* -34.5 dB */
+ 0x0051b9, /* -34.0 dB */
+ 0x005691, /* -33.5 dB */
+ 0x005bb2, /* -33.0 dB */
+ 0x006121, /* -32.5 dB */
+ 0x0066e3, /* -32.0 dB */
+ 0x006cfb, /* -31.5 dB */
+ 0x007370, /* -31.0 dB */
+ 0x007a48, /* -30.5 dB */
+ 0x008186, /* -30.0 dB */
+ 0x008933, /* -29.5 dB */
+ 0x009154, /* -29.0 dB */
+ 0x0099f1, /* -28.5 dB */
+ 0x00a310, /* -28.0 dB */
+ 0x00acba, /* -27.5 dB */
+ 0x00b6f6, /* -27.0 dB */
+ 0x00c1cd, /* -26.5 dB */
+ 0x00cd49, /* -26.0 dB */
+ 0x00d973, /* -25.5 dB */
+ 0x00e655, /* -25.0 dB */
+ 0x00f3fb, /* -24.5 dB */
+ 0x010270, /* -24.0 dB */
+ 0x0111c0, /* -23.5 dB */
+ 0x0121f9, /* -23.0 dB */
+ 0x013328, /* -22.5 dB */
+ 0x01455b, /* -22.0 dB */
+ 0x0158a2, /* -21.5 dB */
+ 0x016d0e, /* -21.0 dB */
+ 0x0182af, /* -20.5 dB */
+ 0x019999, /* -20.0 dB */
+ 0x01b1de, /* -19.5 dB */
+ 0x01cb94, /* -19.0 dB */
+ 0x01e6cf, /* -18.5 dB */
+ 0x0203a7, /* -18.0 dB */
+ 0x022235, /* -17.5 dB */
+ 0x024293, /* -17.0 dB */
+ 0x0264db, /* -16.5 dB */
+ 0x02892c, /* -16.0 dB */
+ 0x02afa3, /* -15.5 dB */
+ 0x02d862, /* -15.0 dB */
+ 0x03038a, /* -14.5 dB */
+ 0x033142, /* -14.0 dB */
+ 0x0361af, /* -13.5 dB */
+ 0x0394fa, /* -13.0 dB */
+ 0x03cb50, /* -12.5 dB */
+ 0x0404de, /* -12.0 dB */
+ 0x0441d5, /* -11.5 dB */
+ 0x048268, /* -11.0 dB */
+ 0x04c6d0, /* -10.5 dB */
+ 0x050f44, /* -10.0 dB */
+ 0x055c04, /* -9.5 dB */
+ 0x05ad50, /* -9.0 dB */
+ 0x06036e, /* -8.5 dB */
+ 0x065ea5, /* -8.0 dB */
+ 0x06bf44, /* -7.5 dB */
+ 0x07259d, /* -7.0 dB */
+ 0x079207, /* -6.5 dB */
+ 0x0804dc, /* -6.0 dB */
+ 0x087e80, /* -5.5 dB */
+ 0x08ff59, /* -5.0 dB */
+ 0x0987d5, /* -4.5 dB */
+ 0x0a1866, /* -4.0 dB */
+ 0x0ab189, /* -3.5 dB */
+ 0x0b53be, /* -3.0 dB */
+ 0x0bff91, /* -2.5 dB */
+ 0x0cb591, /* -2.0 dB */
+ 0x0d765a, /* -1.5 dB */
+ 0x0e4290, /* -1.0 dB */
+ 0x0f1adf, /* -0.5 dB */
+ 0x100000, /* 0.0 dB */
+ 0x10f2b4, /* 0.5 dB */
+ 0x11f3c9, /* 1.0 dB */
+ 0x13041a, /* 1.5 dB */
+ 0x14248e, /* 2.0 dB */
+ 0x15561a, /* 2.5 dB */
+ 0x1699c0, /* 3.0 dB */
+ 0x17f094, /* 3.5 dB */
+ 0x195bb8, /* 4.0 dB */
+ 0x1adc61, /* 4.5 dB */
+ 0x1c73d5, /* 5.0 dB */
+ 0x1e236d, /* 5.5 dB */
+ 0x1fec98, /* 6.0 dB */
+ 0x21d0d9, /* 6.5 dB */
+ 0x23d1cd, /* 7.0 dB */
+ 0x25f125, /* 7.5 dB */
+ 0x2830af, /* 8.0 dB */
+ 0x2a9254, /* 8.5 dB */
+ 0x2d1818, /* 9.0 dB */
+ 0x2fc420, /* 9.5 dB */
+ 0x3298b0, /* 10.0 dB */
+ 0x35982f, /* 10.5 dB */
+ 0x38c528, /* 11.0 dB */
+ 0x3c224c, /* 11.5 dB */
+ 0x3fb278, /* 12.0 dB */
+ 0x4378b0, /* 12.5 dB */
+ 0x477829, /* 13.0 dB */
+ 0x4bb446, /* 13.5 dB */
+ 0x5030a1, /* 14.0 dB */
+ 0x54f106, /* 14.5 dB */
+ 0x59f980, /* 15.0 dB */
+ 0x5f4e52, /* 15.5 dB */
+ 0x64f403, /* 16.0 dB */
+ 0x6aef5e, /* 16.5 dB */
+ 0x714575, /* 17.0 dB */
+ 0x77fbaa, /* 17.5 dB */
+ 0x7f17af, /* 18.0 dB */
+};
+
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c
new file mode 100644
index 0000000..6c515b2
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas.c
@@ -0,0 +1,1012 @@
+/*
+ * Apple Onboard Audio driver for tas codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ * Open questions:
+ * - How to distinguish between 3004 and versions?
+ *
+ * FIXMEs:
+ * - This codec driver doesn't honour the 'connected'
+ * property of the aoa_codec struct, hence if
+ * it is used in machines where not everything is
+ * connected it will display wrong mixer elements.
+ * - Driver assumes that the microphone is always
+ * monaureal and connected to the right channel of
+ * the input. This should also be a codec-dependent
+ * flag, maybe the codec should have 3 different
+ * bits for the three different possibilities how
+ * it can be hooked up...
+ * But as long as I don't see any hardware hooked
+ * up that way...
+ * - As Apple notes in their code, the tas3004 seems
+ * to delay the right channel by one sample. You can
+ * see this when for example recording stereo in
+ * audacity, or recording the tas output via cable
+ * on another machine (use a sinus generator or so).
+ * I tried programming the BiQuads but couldn't
+ * make the delay work, maybe someone can read the
+ * datasheet and fix it. The relevant Apple comment
+ * is in AppleTAS3004Audio.cpp lines 1637 ff. Note
+ * that their comment describing how they program
+ * the filters sucks...
+ *
+ * Other things:
+ * - this should actually register *two* aoa_codec
+ * structs since it has two inputs. Then it must
+ * use the prepare callback to forbid running the
+ * secondary output on a different clock.
+ * Also, whatever bus knows how to do this must
+ * provide two soundbus_dev devices and the fabric
+ * must be able to link them correctly.
+ *
+ * I don't even know if Apple ever uses the second
+ * port on the tas3004 though, I don't think their
+ * i2s controllers can even do it. OTOH, they all
+ * derive the clocks from common clocks, so it
+ * might just be possible. The framework allows the
+ * codec to refine the transfer_info items in the
+ * usable callback, so we can simply remove the
+ * rates the second instance is not using when it
+ * actually is in use.
+ * Maybe we'll need to make the sound busses have
+ * a 'clock group id' value so the codec can
+ * determine if the two outputs can be driven at
+ * the same time. But that is likely overkill, up
+ * to the fabric to not link them up incorrectly,
+ * and up to the hardware designer to not wire
+ * them up in some weird unusable way.
+ */
+#include <stddef.h>
+#include <linux/i2c.h>
+#include <asm/pmac_low_i2c.h>
+#include <asm/prom.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/mutex.h>
+
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("tas codec driver for snd-aoa");
+
+#include "snd-aoa-codec-tas.h"
+#include "snd-aoa-codec-tas-gain-table.h"
+#include "snd-aoa-codec-tas-basstreble.h"
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+#define PFX "snd-aoa-codec-tas: "
+
+
+struct tas {
+ struct aoa_codec codec;
+ struct i2c_client i2c;
+ u32 mute_l:1, mute_r:1 ,
+ controls_created:1 ,
+ drc_enabled:1,
+ hw_enabled:1;
+ u8 cached_volume_l, cached_volume_r;
+ u8 mixer_l[3], mixer_r[3];
+ u8 bass, treble;
+ u8 acr;
+ int drc_range;
+ /* protects hardware access against concurrency from
+ * userspace when hitting controls and during
+ * codec init/suspend/resume */
+ struct mutex mtx;
+};
+
+static int tas_reset_init(struct tas *tas);
+
+static struct tas *codec_to_tas(struct aoa_codec *codec)
+{
+ return container_of(codec, struct tas, codec);
+}
+
+static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data)
+{
+ if (len == 1)
+ return i2c_smbus_write_byte_data(&tas->i2c, reg, *data);
+ else
+ return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data);
+}
+
+static void tas3004_set_drc(struct tas *tas)
+{
+ unsigned char val[6];
+
+ if (tas->drc_enabled)
+ val[0] = 0x50; /* 3:1 above threshold */
+ else
+ val[0] = 0x51; /* disabled */
+ val[1] = 0x02; /* 1:1 below threshold */
+ if (tas->drc_range > 0xef)
+ val[2] = 0xef;
+ else if (tas->drc_range < 0)
+ val[2] = 0x00;
+ else
+ val[2] = tas->drc_range;
+ val[3] = 0xb0;
+ val[4] = 0x60;
+ val[5] = 0xa0;
+
+ tas_write_reg(tas, TAS_REG_DRC, 6, val);
+}
+
+static void tas_set_treble(struct tas *tas)
+{
+ u8 tmp;
+
+ tmp = tas3004_treble(tas->treble);
+ tas_write_reg(tas, TAS_REG_TREBLE, 1, &tmp);
+}
+
+static void tas_set_bass(struct tas *tas)
+{
+ u8 tmp;
+
+ tmp = tas3004_bass(tas->bass);
+ tas_write_reg(tas, TAS_REG_BASS, 1, &tmp);
+}
+
+static void tas_set_volume(struct tas *tas)
+{
+ u8 block[6];
+ int tmp;
+ u8 left, right;
+
+ left = tas->cached_volume_l;
+ right = tas->cached_volume_r;
+
+ if (left > 177) left = 177;
+ if (right > 177) right = 177;
+
+ if (tas->mute_l) left = 0;
+ if (tas->mute_r) right = 0;
+
+ /* analysing the volume and mixer tables shows
+ * that they are similar enough when we shift
+ * the mixer table down by 4 bits. The error
+ * is miniscule, in just one item the error
+ * is 1, at a value of 0x07f17b (mixer table
+ * value is 0x07f17a) */
+ tmp = tas_gaintable[left];
+ block[0] = tmp>>20;
+ block[1] = tmp>>12;
+ block[2] = tmp>>4;
+ tmp = tas_gaintable[right];
+ block[3] = tmp>>20;
+ block[4] = tmp>>12;
+ block[5] = tmp>>4;
+ tas_write_reg(tas, TAS_REG_VOL, 6, block);
+}
+
+static void tas_set_mixer(struct tas *tas)
+{
+ u8 block[9];
+ int tmp, i;
+ u8 val;
+
+ for (i=0;i<3;i++) {
+ val = tas->mixer_l[i];
+ if (val > 177) val = 177;
+ tmp = tas_gaintable[val];
+ block[3*i+0] = tmp>>16;
+ block[3*i+1] = tmp>>8;
+ block[3*i+2] = tmp;
+ }
+ tas_write_reg(tas, TAS_REG_LMIX, 9, block);
+
+ for (i=0;i<3;i++) {
+ val = tas->mixer_r[i];
+ if (val > 177) val = 177;
+ tmp = tas_gaintable[val];
+ block[3*i+0] = tmp>>16;
+ block[3*i+1] = tmp>>8;
+ block[3*i+2] = tmp;
+ }
+ tas_write_reg(tas, TAS_REG_RMIX, 9, block);
+}
+
+/* alsa stuff */
+
+static int tas_dev_register(struct snd_device *dev)
+{
+ return 0;
+}
+
+static struct snd_device_ops ops = {
+ .dev_register = tas_dev_register,
+};
+
+static int tas_snd_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 177;
+ return 0;
+}
+
+static int tas_snd_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->cached_volume_l;
+ ucontrol->value.integer.value[1] = tas->cached_volume_r;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] > 177)
+ return -EINVAL;
+ if (ucontrol->value.integer.value[1] < 0 ||
+ ucontrol->value.integer.value[1] > 177)
+ return -EINVAL;
+
+ mutex_lock(&tas->mtx);
+ if (tas->cached_volume_l == ucontrol->value.integer.value[0]
+ && tas->cached_volume_r == ucontrol->value.integer.value[1]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->cached_volume_l = ucontrol->value.integer.value[0];
+ tas->cached_volume_r = ucontrol->value.integer.value[1];
+ if (tas->hw_enabled)
+ tas_set_volume(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new volume_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_vol_info,
+ .get = tas_snd_vol_get,
+ .put = tas_snd_vol_put,
+};
+
+#define tas_snd_mute_info snd_ctl_boolean_stereo_info
+
+static int tas_snd_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = !tas->mute_l;
+ ucontrol->value.integer.value[1] = !tas->mute_r;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ if (tas->mute_l == !ucontrol->value.integer.value[0]
+ && tas->mute_r == !ucontrol->value.integer.value[1]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->mute_l = !ucontrol->value.integer.value[0];
+ tas->mute_r = !ucontrol->value.integer.value[1];
+ if (tas->hw_enabled)
+ tas_set_volume(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new mute_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_mute_info,
+ .get = tas_snd_mute_get,
+ .put = tas_snd_mute_put,
+};
+
+static int tas_snd_mixer_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 177;
+ return 0;
+}
+
+static int tas_snd_mixer_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+ int idx = kcontrol->private_value;
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->mixer_l[idx];
+ ucontrol->value.integer.value[1] = tas->mixer_r[idx];
+ mutex_unlock(&tas->mtx);
+
+ return 0;
+}
+
+static int tas_snd_mixer_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+ int idx = kcontrol->private_value;
+
+ mutex_lock(&tas->mtx);
+ if (tas->mixer_l[idx] == ucontrol->value.integer.value[0]
+ && tas->mixer_r[idx] == ucontrol->value.integer.value[1]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->mixer_l[idx] = ucontrol->value.integer.value[0];
+ tas->mixer_r[idx] = ucontrol->value.integer.value[1];
+
+ if (tas->hw_enabled)
+ tas_set_mixer(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+#define MIXER_CONTROL(n,descr,idx) \
+static struct snd_kcontrol_new n##_control = { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = descr " Playback Volume", \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = tas_snd_mixer_info, \
+ .get = tas_snd_mixer_get, \
+ .put = tas_snd_mixer_put, \
+ .private_value = idx, \
+}
+
+MIXER_CONTROL(pcm1, "PCM", 0);
+MIXER_CONTROL(monitor, "Monitor", 2);
+
+static int tas_snd_drc_range_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = TAS3004_DRC_MAX;
+ return 0;
+}
+
+static int tas_snd_drc_range_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->drc_range;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] > TAS3004_DRC_MAX)
+ return -EINVAL;
+
+ mutex_lock(&tas->mtx);
+ if (tas->drc_range == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->drc_range = ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas3004_set_drc(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new drc_range_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DRC Range",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_drc_range_info,
+ .get = tas_snd_drc_range_get,
+ .put = tas_snd_drc_range_put,
+};
+
+#define tas_snd_drc_switch_info snd_ctl_boolean_mono_info
+
+static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->drc_enabled;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ if (tas->drc_enabled == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->drc_enabled = !!ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas3004_set_drc(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new drc_switch_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DRC Range Switch",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_drc_switch_info,
+ .get = tas_snd_drc_switch_get,
+ .put = tas_snd_drc_switch_put,
+};
+
+static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "Line-In", "Microphone" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.enumerated.item[0] = !!(tas->acr & TAS_ACR_INPUT_B);
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+ int oldacr;
+
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ mutex_lock(&tas->mtx);
+ oldacr = tas->acr;
+
+ /*
+ * Despite what the data sheet says in one place, the
+ * TAS_ACR_B_MONAUREAL bit forces mono output even when
+ * input A (line in) is selected.
+ */
+ tas->acr &= ~(TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL);
+ if (ucontrol->value.enumerated.item[0])
+ tas->acr |= TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL |
+ TAS_ACR_B_MON_SEL_RIGHT;
+ if (oldacr == tas->acr) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+ if (tas->hw_enabled)
+ tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new capture_source_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* If we name this 'Input Source', it properly shows up in
+ * alsamixer as a selection, * but it's shown under the
+ * 'Playback' category.
+ * If I name it 'Capture Source', it shows up in strange
+ * ways (two bools of which one can be selected at a
+ * time) but at least it's shown in the 'Capture'
+ * category.
+ * I was told that this was due to backward compatibility,
+ * but I don't understand then why the mangling is *not*
+ * done when I name it "Input Source".....
+ */
+ .name = "Capture Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_capture_source_info,
+ .get = tas_snd_capture_source_get,
+ .put = tas_snd_capture_source_put,
+};
+
+static int tas_snd_treble_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = TAS3004_TREBLE_MIN;
+ uinfo->value.integer.max = TAS3004_TREBLE_MAX;
+ return 0;
+}
+
+static int tas_snd_treble_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->treble;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_treble_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < TAS3004_TREBLE_MIN ||
+ ucontrol->value.integer.value[0] > TAS3004_TREBLE_MAX)
+ return -EINVAL;
+ mutex_lock(&tas->mtx);
+ if (tas->treble == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->treble = ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas_set_treble(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new treble_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Treble",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_treble_info,
+ .get = tas_snd_treble_get,
+ .put = tas_snd_treble_put,
+};
+
+static int tas_snd_bass_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = TAS3004_BASS_MIN;
+ uinfo->value.integer.max = TAS3004_BASS_MAX;
+ return 0;
+}
+
+static int tas_snd_bass_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&tas->mtx);
+ ucontrol->value.integer.value[0] = tas->bass;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_snd_bass_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.integer.value[0] < TAS3004_BASS_MIN ||
+ ucontrol->value.integer.value[0] > TAS3004_BASS_MAX)
+ return -EINVAL;
+ mutex_lock(&tas->mtx);
+ if (tas->bass == ucontrol->value.integer.value[0]) {
+ mutex_unlock(&tas->mtx);
+ return 0;
+ }
+
+ tas->bass = ucontrol->value.integer.value[0];
+ if (tas->hw_enabled)
+ tas_set_bass(tas);
+ mutex_unlock(&tas->mtx);
+ return 1;
+}
+
+static struct snd_kcontrol_new bass_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Bass",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = tas_snd_bass_info,
+ .get = tas_snd_bass_get,
+ .put = tas_snd_bass_put,
+};
+
+static struct transfer_info tas_transfers[] = {
+ {
+ /* input */
+ .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .transfer_in = 1,
+ },
+ {
+ /* output */
+ .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .transfer_in = 0,
+ },
+ {}
+};
+
+static int tas_usable(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out)
+{
+ return 1;
+}
+
+static int tas_reset_init(struct tas *tas)
+{
+ u8 tmp;
+
+ tas->codec.gpio->methods->all_amps_off(tas->codec.gpio);
+ msleep(5);
+ tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
+ msleep(5);
+ tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 1);
+ msleep(20);
+ tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
+ msleep(10);
+ tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio);
+
+ tmp = TAS_MCS_SCLK64 | TAS_MCS_SPORT_MODE_I2S | TAS_MCS_SPORT_WL_24BIT;
+ if (tas_write_reg(tas, TAS_REG_MCS, 1, &tmp))
+ goto outerr;
+
+ tas->acr |= TAS_ACR_ANALOG_PDOWN;
+ if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr))
+ goto outerr;
+
+ tmp = 0;
+ if (tas_write_reg(tas, TAS_REG_MCS2, 1, &tmp))
+ goto outerr;
+
+ tas3004_set_drc(tas);
+
+ /* Set treble & bass to 0dB */
+ tas->treble = TAS3004_TREBLE_ZERO;
+ tas->bass = TAS3004_BASS_ZERO;
+ tas_set_treble(tas);
+ tas_set_bass(tas);
+
+ tas->acr &= ~TAS_ACR_ANALOG_PDOWN;
+ if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr))
+ goto outerr;
+
+ return 0;
+ outerr:
+ return -ENODEV;
+}
+
+static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock)
+{
+ struct tas *tas = cii->codec_data;
+
+ switch(clock) {
+ case CLOCK_SWITCH_PREPARE_SLAVE:
+ /* Clocks are going away, mute mute mute */
+ tas->codec.gpio->methods->all_amps_off(tas->codec.gpio);
+ tas->hw_enabled = 0;
+ break;
+ case CLOCK_SWITCH_SLAVE:
+ /* Clocks are back, re-init the codec */
+ mutex_lock(&tas->mtx);
+ tas_reset_init(tas);
+ tas_set_volume(tas);
+ tas_set_mixer(tas);
+ tas->hw_enabled = 1;
+ tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio);
+ mutex_unlock(&tas->mtx);
+ break;
+ default:
+ /* doesn't happen as of now */
+ return -EINVAL;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_PM
+/* we are controlled via i2c and assume that is always up
+ * If that wasn't the case, we'd have to suspend once
+ * our i2c device is suspended, and then take note of that! */
+static int tas_suspend(struct tas *tas)
+{
+ mutex_lock(&tas->mtx);
+ tas->hw_enabled = 0;
+ tas->acr |= TAS_ACR_ANALOG_PDOWN;
+ tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int tas_resume(struct tas *tas)
+{
+ /* reset codec */
+ mutex_lock(&tas->mtx);
+ tas_reset_init(tas);
+ tas_set_volume(tas);
+ tas_set_mixer(tas);
+ tas->hw_enabled = 1;
+ mutex_unlock(&tas->mtx);
+ return 0;
+}
+
+static int _tas_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+ return tas_suspend(cii->codec_data);
+}
+
+static int _tas_resume(struct codec_info_item *cii)
+{
+ return tas_resume(cii->codec_data);
+}
+#else /* CONFIG_PM */
+#define _tas_suspend NULL
+#define _tas_resume NULL
+#endif /* CONFIG_PM */
+
+static struct codec_info tas_codec_info = {
+ .transfers = tas_transfers,
+ /* in theory, we can drive it at 512 too...
+ * but so far the framework doesn't allow
+ * for that and I don't see much point in it. */
+ .sysclock_factor = 256,
+ /* same here, could be 32 for just one 16 bit format */
+ .bus_factor = 64,
+ .owner = THIS_MODULE,
+ .usable = tas_usable,
+ .switch_clock = tas_switch_clock,
+ .suspend = _tas_suspend,
+ .resume = _tas_resume,
+};
+
+static int tas_init_codec(struct aoa_codec *codec)
+{
+ struct tas *tas = codec_to_tas(codec);
+ int err;
+
+ if (!tas->codec.gpio || !tas->codec.gpio->methods) {
+ printk(KERN_ERR PFX "gpios not assigned!!\n");
+ return -EINVAL;
+ }
+
+ mutex_lock(&tas->mtx);
+ if (tas_reset_init(tas)) {
+ printk(KERN_ERR PFX "tas failed to initialise\n");
+ mutex_unlock(&tas->mtx);
+ return -ENXIO;
+ }
+ tas->hw_enabled = 1;
+ mutex_unlock(&tas->mtx);
+
+ if (tas->codec.soundbus_dev->attach_codec(tas->codec.soundbus_dev,
+ aoa_get_card(),
+ &tas_codec_info, tas)) {
+ printk(KERN_ERR PFX "error attaching tas to soundbus\n");
+ return -ENODEV;
+ }
+
+ if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, tas, &ops)) {
+ printk(KERN_ERR PFX "failed to create tas snd device!\n");
+ return -ENODEV;
+ }
+ err = aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&drc_range_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&drc_switch_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&treble_control, tas));
+ if (err)
+ goto error;
+
+ err = aoa_snd_ctl_add(snd_ctl_new1(&bass_control, tas));
+ if (err)
+ goto error;
+
+ return 0;
+ error:
+ tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
+ snd_device_free(aoa_get_card(), tas);
+ return err;
+}
+
+static void tas_exit_codec(struct aoa_codec *codec)
+{
+ struct tas *tas = codec_to_tas(codec);
+
+ if (!tas->codec.soundbus_dev)
+ return;
+ tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
+}
+
+
+static struct i2c_driver tas_driver;
+
+static int tas_create(struct i2c_adapter *adapter,
+ struct device_node *node,
+ int addr)
+{
+ struct tas *tas;
+
+ tas = kzalloc(sizeof(struct tas), GFP_KERNEL);
+
+ if (!tas)
+ return -ENOMEM;
+
+ mutex_init(&tas->mtx);
+ tas->i2c.driver = &tas_driver;
+ tas->i2c.adapter = adapter;
+ tas->i2c.addr = addr;
+ /* seems that half is a saner default */
+ tas->drc_range = TAS3004_DRC_MAX / 2;
+ strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE);
+
+ if (i2c_attach_client(&tas->i2c)) {
+ printk(KERN_ERR PFX "failed to attach to i2c\n");
+ goto fail;
+ }
+
+ strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN);
+ tas->codec.owner = THIS_MODULE;
+ tas->codec.init = tas_init_codec;
+ tas->codec.exit = tas_exit_codec;
+ tas->codec.node = of_node_get(node);
+
+ if (aoa_codec_register(&tas->codec)) {
+ goto detach;
+ }
+ printk(KERN_DEBUG
+ "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n",
+ addr, node->full_name);
+ return 0;
+ detach:
+ i2c_detach_client(&tas->i2c);
+ fail:
+ mutex_destroy(&tas->mtx);
+ kfree(tas);
+ return -EINVAL;
+}
+
+static int tas_i2c_attach(struct i2c_adapter *adapter)
+{
+ struct device_node *busnode, *dev = NULL;
+ struct pmac_i2c_bus *bus;
+
+ bus = pmac_i2c_adapter_to_bus(adapter);
+ if (bus == NULL)
+ return -ENODEV;
+ busnode = pmac_i2c_get_bus_node(bus);
+
+ while ((dev = of_get_next_child(busnode, dev)) != NULL) {
+ if (of_device_is_compatible(dev, "tas3004")) {
+ const u32 *addr;
+ printk(KERN_DEBUG PFX "found tas3004\n");
+ addr = of_get_property(dev, "reg", NULL);
+ if (!addr)
+ continue;
+ return tas_create(adapter, dev, ((*addr) >> 1) & 0x7f);
+ }
+ /* older machines have no 'codec' node with a 'compatible'
+ * property that says 'tas3004', they just have a 'deq'
+ * node without any such property... */
+ if (strcmp(dev->name, "deq") == 0) {
+ const u32 *_addr;
+ u32 addr;
+ printk(KERN_DEBUG PFX "found 'deq' node\n");
+ _addr = of_get_property(dev, "i2c-address", NULL);
+ if (!_addr)
+ continue;
+ addr = ((*_addr) >> 1) & 0x7f;
+ /* now, if the address doesn't match any of the two
+ * that a tas3004 can have, we cannot handle this.
+ * I doubt it ever happens but hey. */
+ if (addr != 0x34 && addr != 0x35)
+ continue;
+ return tas_create(adapter, dev, addr);
+ }
+ }
+ return -ENODEV;
+}
+
+static int tas_i2c_detach(struct i2c_client *client)
+{
+ struct tas *tas = container_of(client, struct tas, i2c);
+ int err;
+ u8 tmp = TAS_ACR_ANALOG_PDOWN;
+
+ if ((err = i2c_detach_client(client)))
+ return err;
+ aoa_codec_unregister(&tas->codec);
+ of_node_put(tas->codec.node);
+
+ /* power down codec chip */
+ tas_write_reg(tas, TAS_REG_ACR, 1, &tmp);
+
+ mutex_destroy(&tas->mtx);
+ kfree(tas);
+ return 0;
+}
+
+static struct i2c_driver tas_driver = {
+ .driver = {
+ .name = "aoa_codec_tas",
+ .owner = THIS_MODULE,
+ },
+ .attach_adapter = tas_i2c_attach,
+ .detach_client = tas_i2c_detach,
+};
+
+static int __init tas_init(void)
+{
+ return i2c_add_driver(&tas_driver);
+}
+
+static void __exit tas_exit(void)
+{
+ i2c_del_driver(&tas_driver);
+}
+
+module_init(tas_init);
+module_exit(tas_exit);
diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.h b/sound/aoa/codecs/snd-aoa-codec-tas.h
new file mode 100644
index 0000000..ae177e3
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas.h
@@ -0,0 +1,55 @@
+/*
+ * Apple Onboard Audio driver for tas codec (header)
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SND_AOA_CODECTASH
+#define __SND_AOA_CODECTASH
+
+#define TAS_REG_MCS 0x01 /* main control */
+# define TAS_MCS_FASTLOAD (1<<7)
+# define TAS_MCS_SCLK64 (1<<6)
+# define TAS_MCS_SPORT_MODE_MASK (3<<4)
+# define TAS_MCS_SPORT_MODE_I2S (2<<4)
+# define TAS_MCS_SPORT_MODE_RJ (1<<4)
+# define TAS_MCS_SPORT_MODE_LJ (0<<4)
+# define TAS_MCS_SPORT_WL_MASK (3<<0)
+# define TAS_MCS_SPORT_WL_16BIT (0<<0)
+# define TAS_MCS_SPORT_WL_18BIT (1<<0)
+# define TAS_MCS_SPORT_WL_20BIT (2<<0)
+# define TAS_MCS_SPORT_WL_24BIT (3<<0)
+
+#define TAS_REG_DRC 0x02
+#define TAS_REG_VOL 0x04
+#define TAS_REG_TREBLE 0x05
+#define TAS_REG_BASS 0x06
+#define TAS_REG_LMIX 0x07
+#define TAS_REG_RMIX 0x08
+
+#define TAS_REG_ACR 0x40 /* analog control */
+# define TAS_ACR_B_MONAUREAL (1<<7)
+# define TAS_ACR_B_MON_SEL_RIGHT (1<<6)
+# define TAS_ACR_DEEMPH_MASK (3<<2)
+# define TAS_ACR_DEEMPH_OFF (0<<2)
+# define TAS_ACR_DEEMPH_48KHz (1<<2)
+# define TAS_ACR_DEEMPH_44KHz (2<<2)
+# define TAS_ACR_INPUT_B (1<<1)
+# define TAS_ACR_ANALOG_PDOWN (1<<0)
+
+#define TAS_REG_MCS2 0x43 /* main control 2 */
+# define TAS_MCS2_ALLPASS (1<<1)
+
+#define TAS_REG_LEFT_BIQUAD6 0x10
+#define TAS_REG_RIGHT_BIQUAD6 0x19
+
+#define TAS_REG_LEFT_LOUDNESS 0x21
+#define TAS_REG_RIGHT_LOUDNESS 0x22
+#define TAS_REG_LEFT_LOUDNESS_GAIN 0x23
+#define TAS_REG_RIGHT_LOUDNESS_GAIN 0x24
+
+#define TAS3001_DRC_MAX 0x5f
+#define TAS3004_DRC_MAX 0xef
+
+#endif /* __SND_AOA_CODECTASH */
diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/snd-aoa-codec-toonie.c
new file mode 100644
index 0000000..3c7d1d8
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-toonie.c
@@ -0,0 +1,150 @@
+/*
+ * Apple Onboard Audio driver for Toonie codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ *
+ * This is a driver for the toonie codec chip. This chip is present
+ * on the Mac Mini and is nothing but a DAC.
+ */
+#include <linux/delay.h>
+#include <linux/module.h>
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("toonie codec driver for snd-aoa");
+
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-toonie: "
+
+struct toonie {
+ struct aoa_codec codec;
+};
+#define codec_to_toonie(c) container_of(c, struct toonie, codec)
+
+static int toonie_dev_register(struct snd_device *dev)
+{
+ return 0;
+}
+
+static struct snd_device_ops ops = {
+ .dev_register = toonie_dev_register,
+};
+
+static struct transfer_info toonie_transfers[] = {
+ /* This thing *only* has analog output,
+ * the rates are taken from Info.plist
+ * from Darwin. */
+ {
+ .formats = SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ },
+ {}
+};
+
+static int toonie_usable(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out)
+{
+ return 1;
+}
+
+#ifdef CONFIG_PM
+static int toonie_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+ /* can we turn it off somehow? */
+ return 0;
+}
+
+static int toonie_resume(struct codec_info_item *cii)
+{
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static struct codec_info toonie_codec_info = {
+ .transfers = toonie_transfers,
+ .sysclock_factor = 256,
+ .bus_factor = 64,
+ .owner = THIS_MODULE,
+ .usable = toonie_usable,
+#ifdef CONFIG_PM
+ .suspend = toonie_suspend,
+ .resume = toonie_resume,
+#endif
+};
+
+static int toonie_init_codec(struct aoa_codec *codec)
+{
+ struct toonie *toonie = codec_to_toonie(codec);
+
+ /* nothing connected? what a joke! */
+ if (toonie->codec.connected != 1)
+ return -ENOTCONN;
+
+ if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, toonie, &ops)) {
+ printk(KERN_ERR PFX "failed to create toonie snd device!\n");
+ return -ENODEV;
+ }
+
+ if (toonie->codec.soundbus_dev->attach_codec(toonie->codec.soundbus_dev,
+ aoa_get_card(),
+ &toonie_codec_info, toonie)) {
+ printk(KERN_ERR PFX "error creating toonie pcm\n");
+ snd_device_free(aoa_get_card(), toonie);
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static void toonie_exit_codec(struct aoa_codec *codec)
+{
+ struct toonie *toonie = codec_to_toonie(codec);
+
+ if (!toonie->codec.soundbus_dev) {
+ printk(KERN_ERR PFX "toonie_exit_codec called without soundbus_dev!\n");
+ return;
+ }
+ toonie->codec.soundbus_dev->detach_codec(toonie->codec.soundbus_dev, toonie);
+}
+
+static struct toonie *toonie;
+
+static int __init toonie_init(void)
+{
+ toonie = kzalloc(sizeof(struct toonie), GFP_KERNEL);
+
+ if (!toonie)
+ return -ENOMEM;
+
+ strlcpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name));
+ toonie->codec.owner = THIS_MODULE;
+ toonie->codec.init = toonie_init_codec;
+ toonie->codec.exit = toonie_exit_codec;
+
+ if (aoa_codec_register(&toonie->codec)) {
+ kfree(toonie);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static void __exit toonie_exit(void)
+{
+ aoa_codec_unregister(&toonie->codec);
+ kfree(toonie);
+}
+
+module_init(toonie_init);
+module_exit(toonie_exit);
diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile
new file mode 100644
index 0000000..62dc728
--- /dev/null
+++ b/sound/aoa/core/Makefile
@@ -0,0 +1,5 @@
+obj-$(CONFIG_SND_AOA) += snd-aoa.o
+snd-aoa-objs := snd-aoa-core.o \
+ snd-aoa-alsa.o \
+ snd-aoa-gpio-pmf.o \
+ snd-aoa-gpio-feature.o
diff --git a/sound/aoa/core/snd-aoa-alsa.c b/sound/aoa/core/snd-aoa-alsa.c
new file mode 100644
index 0000000..17fe689
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-alsa.c
@@ -0,0 +1,99 @@
+/*
+ * Apple Onboard Audio Alsa helpers
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#include <linux/module.h>
+#include "snd-aoa-alsa.h"
+
+static int index = -1;
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "index for AOA sound card.");
+
+static struct aoa_card *aoa_card;
+
+int aoa_alsa_init(char *name, struct module *mod, struct device *dev)
+{
+ struct snd_card *alsa_card;
+ int err;
+
+ if (aoa_card)
+ /* cannot be EEXIST due to usage in aoa_fabric_register */
+ return -EBUSY;
+
+ alsa_card = snd_card_new(index, name, mod, sizeof(struct aoa_card));
+ if (!alsa_card)
+ return -ENOMEM;
+ aoa_card = alsa_card->private_data;
+ aoa_card->alsa_card = alsa_card;
+ alsa_card->dev = dev;
+ strlcpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver));
+ strlcpy(alsa_card->shortname, name, sizeof(alsa_card->shortname));
+ strlcpy(alsa_card->longname, name, sizeof(alsa_card->longname));
+ strlcpy(alsa_card->mixername, name, sizeof(alsa_card->mixername));
+ err = snd_card_register(aoa_card->alsa_card);
+ if (err < 0) {
+ printk(KERN_ERR "snd-aoa: couldn't register alsa card\n");
+ snd_card_free(aoa_card->alsa_card);
+ aoa_card = NULL;
+ return err;
+ }
+ return 0;
+}
+
+struct snd_card *aoa_get_card(void)
+{
+ if (aoa_card)
+ return aoa_card->alsa_card;
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(aoa_get_card);
+
+void aoa_alsa_cleanup(void)
+{
+ if (aoa_card) {
+ snd_card_free(aoa_card->alsa_card);
+ aoa_card = NULL;
+ }
+}
+
+int aoa_snd_device_new(snd_device_type_t type,
+ void * device_data, struct snd_device_ops * ops)
+{
+ struct snd_card *card = aoa_get_card();
+ int err;
+
+ if (!card) return -ENOMEM;
+
+ err = snd_device_new(card, type, device_data, ops);
+ if (err) {
+ printk(KERN_ERR "snd-aoa: failed to create snd device (%d)\n", err);
+ return err;
+ }
+ err = snd_device_register(card, device_data);
+ if (err) {
+ printk(KERN_ERR "snd-aoa: failed to register "
+ "snd device (%d)\n", err);
+ printk(KERN_ERR "snd-aoa: have you forgotten the "
+ "dev_register callback?\n");
+ snd_device_free(card, device_data);
+ }
+ return err;
+}
+EXPORT_SYMBOL_GPL(aoa_snd_device_new);
+
+int aoa_snd_ctl_add(struct snd_kcontrol* control)
+{
+ int err;
+
+ if (!aoa_card) return -ENODEV;
+
+ err = snd_ctl_add(aoa_card->alsa_card, control);
+ if (err)
+ printk(KERN_ERR "snd-aoa: failed to add alsa control (%d)\n",
+ err);
+ return err;
+}
+EXPORT_SYMBOL_GPL(aoa_snd_ctl_add);
diff --git a/sound/aoa/core/snd-aoa-alsa.h b/sound/aoa/core/snd-aoa-alsa.h
new file mode 100644
index 0000000..9669e44
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-alsa.h
@@ -0,0 +1,16 @@
+/*
+ * Apple Onboard Audio Alsa private helpers
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#ifndef __SND_AOA_ALSA_H
+#define __SND_AOA_ALSA_H
+#include "../aoa.h"
+
+extern int aoa_alsa_init(char *name, struct module *mod, struct device *dev);
+extern void aoa_alsa_cleanup(void);
+
+#endif /* __SND_AOA_ALSA_H */
diff --git a/sound/aoa/core/snd-aoa-core.c b/sound/aoa/core/snd-aoa-core.c
new file mode 100644
index 0000000..19fdae4
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-core.c
@@ -0,0 +1,162 @@
+/*
+ * Apple Onboard Audio driver core
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/list.h>
+#include "../aoa.h"
+#include "snd-aoa-alsa.h"
+
+MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver");
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+
+/* We allow only one fabric. This simplifies things,
+ * and more don't really make that much sense */
+static struct aoa_fabric *fabric;
+static LIST_HEAD(codec_list);
+
+static int attach_codec_to_fabric(struct aoa_codec *c)
+{
+ int err;
+
+ if (!try_module_get(c->owner))
+ return -EBUSY;
+ /* found_codec has to be assigned */
+ err = -ENOENT;
+ if (fabric->found_codec)
+ err = fabric->found_codec(c);
+ if (err) {
+ module_put(c->owner);
+ printk(KERN_ERR "snd-aoa: fabric didn't like codec %s\n",
+ c->name);
+ return err;
+ }
+ c->fabric = fabric;
+
+ err = 0;
+ if (c->init)
+ err = c->init(c);
+ if (err) {
+ printk(KERN_ERR "snd-aoa: codec %s didn't init\n", c->name);
+ c->fabric = NULL;
+ if (fabric->remove_codec)
+ fabric->remove_codec(c);
+ module_put(c->owner);
+ return err;
+ }
+ if (fabric->attached_codec)
+ fabric->attached_codec(c);
+ return 0;
+}
+
+int aoa_codec_register(struct aoa_codec *codec)
+{
+ int err = 0;
+
+ /* if there's a fabric already, we can tell if we
+ * will want to have this codec, so propagate error
+ * through. Otherwise, this will happen later... */
+ if (fabric)
+ err = attach_codec_to_fabric(codec);
+ if (!err)
+ list_add(&codec->list, &codec_list);
+ return err;
+}
+EXPORT_SYMBOL_GPL(aoa_codec_register);
+
+void aoa_codec_unregister(struct aoa_codec *codec)
+{
+ list_del(&codec->list);
+ if (codec->fabric && codec->exit)
+ codec->exit(codec);
+ if (fabric && fabric->remove_codec)
+ fabric->remove_codec(codec);
+ codec->fabric = NULL;
+ module_put(codec->owner);
+}
+EXPORT_SYMBOL_GPL(aoa_codec_unregister);
+
+int aoa_fabric_register(struct aoa_fabric *new_fabric, struct device *dev)
+{
+ struct aoa_codec *c;
+ int err;
+
+ /* allow querying for presence of fabric
+ * (i.e. do this test first!) */
+ if (new_fabric == fabric) {
+ err = -EALREADY;
+ goto attach;
+ }
+ if (fabric)
+ return -EEXIST;
+ if (!new_fabric)
+ return -EINVAL;
+
+ err = aoa_alsa_init(new_fabric->name, new_fabric->owner, dev);
+ if (err)
+ return err;
+
+ fabric = new_fabric;
+
+ attach:
+ list_for_each_entry(c, &codec_list, list) {
+ if (c->fabric != fabric)
+ attach_codec_to_fabric(c);
+ }
+ return err;
+}
+EXPORT_SYMBOL_GPL(aoa_fabric_register);
+
+void aoa_fabric_unregister(struct aoa_fabric *old_fabric)
+{
+ struct aoa_codec *c;
+
+ if (fabric != old_fabric)
+ return;
+
+ list_for_each_entry(c, &codec_list, list) {
+ if (c->fabric)
+ aoa_fabric_unlink_codec(c);
+ }
+
+ aoa_alsa_cleanup();
+
+ fabric = NULL;
+}
+EXPORT_SYMBOL_GPL(aoa_fabric_unregister);
+
+void aoa_fabric_unlink_codec(struct aoa_codec *codec)
+{
+ if (!codec->fabric) {
+ printk(KERN_ERR "snd-aoa: fabric unassigned "
+ "in aoa_fabric_unlink_codec\n");
+ dump_stack();
+ return;
+ }
+ if (codec->exit)
+ codec->exit(codec);
+ if (codec->fabric->remove_codec)
+ codec->fabric->remove_codec(codec);
+ codec->fabric = NULL;
+ module_put(codec->owner);
+}
+EXPORT_SYMBOL_GPL(aoa_fabric_unlink_codec);
+
+static int __init aoa_init(void)
+{
+ return 0;
+}
+
+static void __exit aoa_exit(void)
+{
+ aoa_alsa_cleanup();
+}
+
+module_init(aoa_init);
+module_exit(aoa_exit);
diff --git a/sound/aoa/core/snd-aoa-gpio-feature.c b/sound/aoa/core/snd-aoa-gpio-feature.c
new file mode 100644
index 0000000..805dcbf
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-gpio-feature.c
@@ -0,0 +1,408 @@
+/*
+ * Apple Onboard Audio feature call GPIO control
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ * This file contains the GPIO control routines for
+ * direct (through feature calls) access to the GPIO
+ * registers.
+ */
+
+#include <asm/pmac_feature.h>
+#include <linux/interrupt.h>
+#include "../aoa.h"
+
+/* TODO: these are 20 global variables
+ * that aren't used on most machines...
+ * Move them into a dynamically allocated
+ * structure and use that.
+ */
+
+/* these are the GPIO numbers (register addresses as offsets into
+ * the GPIO space) */
+static int headphone_mute_gpio;
+static int amp_mute_gpio;
+static int lineout_mute_gpio;
+static int hw_reset_gpio;
+static int lineout_detect_gpio;
+static int headphone_detect_gpio;
+static int linein_detect_gpio;
+
+/* see the SWITCH_GPIO macro */
+static int headphone_mute_gpio_activestate;
+static int amp_mute_gpio_activestate;
+static int lineout_mute_gpio_activestate;
+static int hw_reset_gpio_activestate;
+static int lineout_detect_gpio_activestate;
+static int headphone_detect_gpio_activestate;
+static int linein_detect_gpio_activestate;
+
+/* node pointers that we save when getting the GPIO number
+ * to get the interrupt later */
+static struct device_node *lineout_detect_node;
+static struct device_node *linein_detect_node;
+static struct device_node *headphone_detect_node;
+
+static int lineout_detect_irq;
+static int linein_detect_irq;
+static int headphone_detect_irq;
+
+static struct device_node *get_gpio(char *name,
+ char *altname,
+ int *gpioptr,
+ int *gpioactiveptr)
+{
+ struct device_node *np, *gpio;
+ const u32 *reg;
+ const char *audio_gpio;
+
+ *gpioptr = -1;
+
+ /* check if we can get it the easy way ... */
+ np = of_find_node_by_name(NULL, name);
+ if (!np) {
+ /* some machines have only gpioX/extint-gpioX nodes,
+ * and an audio-gpio property saying what it is ...
+ * So what we have to do is enumerate all children
+ * of the gpio node and check them all. */
+ gpio = of_find_node_by_name(NULL, "gpio");
+ if (!gpio)
+ return NULL;
+ while ((np = of_get_next_child(gpio, np))) {
+ audio_gpio = of_get_property(np, "audio-gpio", NULL);
+ if (!audio_gpio)
+ continue;
+ if (strcmp(audio_gpio, name) == 0)
+ break;
+ if (altname && (strcmp(audio_gpio, altname) == 0))
+ break;
+ }
+ /* still not found, assume not there */
+ if (!np)
+ return NULL;
+ }
+
+ reg = of_get_property(np, "reg", NULL);
+ if (!reg)
+ return NULL;
+
+ *gpioptr = *reg;
+
+ /* this is a hack, usually the GPIOs 'reg' property
+ * should have the offset based from the GPIO space
+ * which is at 0x50, but apparently not always... */
+ if (*gpioptr < 0x50)
+ *gpioptr += 0x50;
+
+ reg = of_get_property(np, "audio-gpio-active-state", NULL);
+ if (!reg)
+ /* Apple seems to default to 1, but
+ * that doesn't seem right at least on most
+ * machines. So until proven that the opposite
+ * is necessary, we default to 0
+ * (which, incidentally, snd-powermac also does...) */
+ *gpioactiveptr = 0;
+ else
+ *gpioactiveptr = *reg;
+
+ return np;
+}
+
+static void get_irq(struct device_node * np, int *irqptr)
+{
+ if (np)
+ *irqptr = irq_of_parse_and_map(np, 0);
+ else
+ *irqptr = NO_IRQ;
+}
+
+/* 0x4 is outenable, 0x1 is out, thus 4 or 5 */
+#define SWITCH_GPIO(name, v, on) \
+ (((v)&~1) | ((on)? \
+ (name##_gpio_activestate==0?4:5): \
+ (name##_gpio_activestate==0?5:4)))
+
+#define FTR_GPIO(name, bit) \
+static void ftr_gpio_set_##name(struct gpio_runtime *rt, int on)\
+{ \
+ int v; \
+ \
+ if (unlikely(!rt)) return; \
+ \
+ if (name##_mute_gpio < 0) \
+ return; \
+ \
+ v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, \
+ name##_mute_gpio, \
+ 0); \
+ \
+ /* muted = !on... */ \
+ v = SWITCH_GPIO(name##_mute, v, !on); \
+ \
+ pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, \
+ name##_mute_gpio, v); \
+ \
+ rt->implementation_private &= ~(1<<bit); \
+ rt->implementation_private |= (!!on << bit); \
+} \
+static int ftr_gpio_get_##name(struct gpio_runtime *rt) \
+{ \
+ if (unlikely(!rt)) return 0; \
+ return (rt->implementation_private>>bit)&1; \
+}
+
+FTR_GPIO(headphone, 0);
+FTR_GPIO(amp, 1);
+FTR_GPIO(lineout, 2);
+
+static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on)
+{
+ int v;
+
+ if (unlikely(!rt)) return;
+ if (hw_reset_gpio < 0)
+ return;
+
+ v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL,
+ hw_reset_gpio, 0);
+ v = SWITCH_GPIO(hw_reset, v, on);
+ pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL,
+ hw_reset_gpio, v);
+}
+
+static void ftr_gpio_all_amps_off(struct gpio_runtime *rt)
+{
+ int saved;
+
+ if (unlikely(!rt)) return;
+ saved = rt->implementation_private;
+ ftr_gpio_set_headphone(rt, 0);
+ ftr_gpio_set_amp(rt, 0);
+ ftr_gpio_set_lineout(rt, 0);
+ rt->implementation_private = saved;
+}
+
+static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt)
+{
+ int s;
+
+ if (unlikely(!rt)) return;
+ s = rt->implementation_private;
+ ftr_gpio_set_headphone(rt, (s>>0)&1);
+ ftr_gpio_set_amp(rt, (s>>1)&1);
+ ftr_gpio_set_lineout(rt, (s>>2)&1);
+}
+
+static void ftr_handle_notify(struct work_struct *work)
+{
+ struct gpio_notification *notif =
+ container_of(work, struct gpio_notification, work.work);
+
+ mutex_lock(&notif->mutex);
+ if (notif->notify)
+ notif->notify(notif->data);
+ mutex_unlock(&notif->mutex);
+}
+
+static void gpio_enable_dual_edge(int gpio)
+{
+ int v;
+
+ if (gpio == -1)
+ return;
+ v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0);
+ v |= 0x80; /* enable dual edge */
+ pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, gpio, v);
+}
+
+static void ftr_gpio_init(struct gpio_runtime *rt)
+{
+ get_gpio("headphone-mute", NULL,
+ &headphone_mute_gpio,
+ &headphone_mute_gpio_activestate);
+ get_gpio("amp-mute", NULL,
+ &amp_mute_gpio,
+ &amp_mute_gpio_activestate);
+ get_gpio("lineout-mute", NULL,
+ &lineout_mute_gpio,
+ &lineout_mute_gpio_activestate);
+ get_gpio("hw-reset", "audio-hw-reset",
+ &hw_reset_gpio,
+ &hw_reset_gpio_activestate);
+
+ headphone_detect_node = get_gpio("headphone-detect", NULL,
+ &headphone_detect_gpio,
+ &headphone_detect_gpio_activestate);
+ /* go Apple, and thanks for giving these different names
+ * across the board... */
+ lineout_detect_node = get_gpio("lineout-detect", "line-output-detect",
+ &lineout_detect_gpio,
+ &lineout_detect_gpio_activestate);
+ linein_detect_node = get_gpio("linein-detect", "line-input-detect",
+ &linein_detect_gpio,
+ &linein_detect_gpio_activestate);
+
+ gpio_enable_dual_edge(headphone_detect_gpio);
+ gpio_enable_dual_edge(lineout_detect_gpio);
+ gpio_enable_dual_edge(linein_detect_gpio);
+
+ get_irq(headphone_detect_node, &headphone_detect_irq);
+ get_irq(lineout_detect_node, &lineout_detect_irq);
+ get_irq(linein_detect_node, &linein_detect_irq);
+
+ ftr_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+ INIT_DELAYED_WORK(&rt->headphone_notify.work, ftr_handle_notify);
+ INIT_DELAYED_WORK(&rt->line_in_notify.work, ftr_handle_notify);
+ INIT_DELAYED_WORK(&rt->line_out_notify.work, ftr_handle_notify);
+ mutex_init(&rt->headphone_notify.mutex);
+ mutex_init(&rt->line_in_notify.mutex);
+ mutex_init(&rt->line_out_notify.mutex);
+}
+
+static void ftr_gpio_exit(struct gpio_runtime *rt)
+{
+ ftr_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+ if (rt->headphone_notify.notify)
+ free_irq(headphone_detect_irq, &rt->headphone_notify);
+ if (rt->line_in_notify.gpio_private)
+ free_irq(linein_detect_irq, &rt->line_in_notify);
+ if (rt->line_out_notify.gpio_private)
+ free_irq(lineout_detect_irq, &rt->line_out_notify);
+ cancel_delayed_work(&rt->headphone_notify.work);
+ cancel_delayed_work(&rt->line_in_notify.work);
+ cancel_delayed_work(&rt->line_out_notify.work);
+ flush_scheduled_work();
+ mutex_destroy(&rt->headphone_notify.mutex);
+ mutex_destroy(&rt->line_in_notify.mutex);
+ mutex_destroy(&rt->line_out_notify.mutex);
+}
+
+static irqreturn_t ftr_handle_notify_irq(int xx, void *data)
+{
+ struct gpio_notification *notif = data;
+
+ schedule_delayed_work(&notif->work, 0);
+
+ return IRQ_HANDLED;
+}
+
+static int ftr_set_notify(struct gpio_runtime *rt,
+ enum notify_type type,
+ notify_func_t notify,
+ void *data)
+{
+ struct gpio_notification *notif;
+ notify_func_t old;
+ int irq;
+ char *name;
+ int err = -EBUSY;
+
+ switch (type) {
+ case AOA_NOTIFY_HEADPHONE:
+ notif = &rt->headphone_notify;
+ name = "headphone-detect";
+ irq = headphone_detect_irq;
+ break;
+ case AOA_NOTIFY_LINE_IN:
+ notif = &rt->line_in_notify;
+ name = "linein-detect";
+ irq = linein_detect_irq;
+ break;
+ case AOA_NOTIFY_LINE_OUT:
+ notif = &rt->line_out_notify;
+ name = "lineout-detect";
+ irq = lineout_detect_irq;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (irq == NO_IRQ)
+ return -ENODEV;
+
+ mutex_lock(&notif->mutex);
+
+ old = notif->notify;
+
+ if (!old && !notify) {
+ err = 0;
+ goto out_unlock;
+ }
+
+ if (old && notify) {
+ if (old == notify && notif->data == data)
+ err = 0;
+ goto out_unlock;
+ }
+
+ if (old && !notify)
+ free_irq(irq, notif);
+
+ if (!old && notify) {
+ err = request_irq(irq, ftr_handle_notify_irq, 0, name, notif);
+ if (err)
+ goto out_unlock;
+ }
+
+ notif->notify = notify;
+ notif->data = data;
+
+ err = 0;
+ out_unlock:
+ mutex_unlock(&notif->mutex);
+ return err;
+}
+
+static int ftr_get_detect(struct gpio_runtime *rt,
+ enum notify_type type)
+{
+ int gpio, ret, active;
+
+ switch (type) {
+ case AOA_NOTIFY_HEADPHONE:
+ gpio = headphone_detect_gpio;
+ active = headphone_detect_gpio_activestate;
+ break;
+ case AOA_NOTIFY_LINE_IN:
+ gpio = linein_detect_gpio;
+ active = linein_detect_gpio_activestate;
+ break;
+ case AOA_NOTIFY_LINE_OUT:
+ gpio = lineout_detect_gpio;
+ active = lineout_detect_gpio_activestate;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (gpio == -1)
+ return -ENODEV;
+
+ ret = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0);
+ if (ret < 0)
+ return ret;
+ return ((ret >> 1) & 1) == active;
+}
+
+static struct gpio_methods methods = {
+ .init = ftr_gpio_init,
+ .exit = ftr_gpio_exit,
+ .all_amps_off = ftr_gpio_all_amps_off,
+ .all_amps_restore = ftr_gpio_all_amps_restore,
+ .set_headphone = ftr_gpio_set_headphone,
+ .set_speakers = ftr_gpio_set_amp,
+ .set_lineout = ftr_gpio_set_lineout,
+ .set_hw_reset = ftr_gpio_set_hw_reset,
+ .get_headphone = ftr_gpio_get_headphone,
+ .get_speakers = ftr_gpio_get_amp,
+ .get_lineout = ftr_gpio_get_lineout,
+ .set_notify = ftr_set_notify,
+ .get_detect = ftr_get_detect,
+};
+
+struct gpio_methods *ftr_gpio_methods = &methods;
+EXPORT_SYMBOL_GPL(ftr_gpio_methods);
diff --git a/sound/aoa/core/snd-aoa-gpio-pmf.c b/sound/aoa/core/snd-aoa-gpio-pmf.c
new file mode 100644
index 0000000..5ca2220
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-gpio-pmf.c
@@ -0,0 +1,252 @@
+/*
+ * Apple Onboard Audio pmf GPIOs
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <asm/pmac_feature.h>
+#include <asm/pmac_pfunc.h>
+#include "../aoa.h"
+
+#define PMF_GPIO(name, bit) \
+static void pmf_gpio_set_##name(struct gpio_runtime *rt, int on)\
+{ \
+ struct pmf_args args = { .count = 1, .u[0].v = !on }; \
+ int rc; \
+ \
+ if (unlikely(!rt)) return; \
+ rc = pmf_call_function(rt->node, #name "-mute", &args); \
+ if (rc && rc != -ENODEV) \
+ printk(KERN_WARNING "pmf_gpio_set_" #name \
+ " failed, rc: %d\n", rc); \
+ rt->implementation_private &= ~(1<<bit); \
+ rt->implementation_private |= (!!on << bit); \
+} \
+static int pmf_gpio_get_##name(struct gpio_runtime *rt) \
+{ \
+ if (unlikely(!rt)) return 0; \
+ return (rt->implementation_private>>bit)&1; \
+}
+
+PMF_GPIO(headphone, 0);
+PMF_GPIO(amp, 1);
+PMF_GPIO(lineout, 2);
+
+static void pmf_gpio_set_hw_reset(struct gpio_runtime *rt, int on)
+{
+ struct pmf_args args = { .count = 1, .u[0].v = !!on };
+ int rc;
+
+ if (unlikely(!rt)) return;
+ rc = pmf_call_function(rt->node, "hw-reset", &args);
+ if (rc)
+ printk(KERN_WARNING "pmf_gpio_set_hw_reset"
+ " failed, rc: %d\n", rc);
+}
+
+static void pmf_gpio_all_amps_off(struct gpio_runtime *rt)
+{
+ int saved;
+
+ if (unlikely(!rt)) return;
+ saved = rt->implementation_private;
+ pmf_gpio_set_headphone(rt, 0);
+ pmf_gpio_set_amp(rt, 0);
+ pmf_gpio_set_lineout(rt, 0);
+ rt->implementation_private = saved;
+}
+
+static void pmf_gpio_all_amps_restore(struct gpio_runtime *rt)
+{
+ int s;
+
+ if (unlikely(!rt)) return;
+ s = rt->implementation_private;
+ pmf_gpio_set_headphone(rt, (s>>0)&1);
+ pmf_gpio_set_amp(rt, (s>>1)&1);
+ pmf_gpio_set_lineout(rt, (s>>2)&1);
+}
+
+static void pmf_handle_notify(struct work_struct *work)
+{
+ struct gpio_notification *notif =
+ container_of(work, struct gpio_notification, work.work);
+
+ mutex_lock(&notif->mutex);
+ if (notif->notify)
+ notif->notify(notif->data);
+ mutex_unlock(&notif->mutex);
+}
+
+static void pmf_gpio_init(struct gpio_runtime *rt)
+{
+ pmf_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+ INIT_DELAYED_WORK(&rt->headphone_notify.work, pmf_handle_notify);
+ INIT_DELAYED_WORK(&rt->line_in_notify.work, pmf_handle_notify);
+ INIT_DELAYED_WORK(&rt->line_out_notify.work, pmf_handle_notify);
+ mutex_init(&rt->headphone_notify.mutex);
+ mutex_init(&rt->line_in_notify.mutex);
+ mutex_init(&rt->line_out_notify.mutex);
+}
+
+static void pmf_gpio_exit(struct gpio_runtime *rt)
+{
+ pmf_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+
+ if (rt->headphone_notify.gpio_private)
+ pmf_unregister_irq_client(rt->headphone_notify.gpio_private);
+ if (rt->line_in_notify.gpio_private)
+ pmf_unregister_irq_client(rt->line_in_notify.gpio_private);
+ if (rt->line_out_notify.gpio_private)
+ pmf_unregister_irq_client(rt->line_out_notify.gpio_private);
+
+ /* make sure no work is pending before freeing
+ * all things */
+ cancel_delayed_work(&rt->headphone_notify.work);
+ cancel_delayed_work(&rt->line_in_notify.work);
+ cancel_delayed_work(&rt->line_out_notify.work);
+ flush_scheduled_work();
+
+ mutex_destroy(&rt->headphone_notify.mutex);
+ mutex_destroy(&rt->line_in_notify.mutex);
+ mutex_destroy(&rt->line_out_notify.mutex);
+
+ if (rt->headphone_notify.gpio_private)
+ kfree(rt->headphone_notify.gpio_private);
+ if (rt->line_in_notify.gpio_private)
+ kfree(rt->line_in_notify.gpio_private);
+ if (rt->line_out_notify.gpio_private)
+ kfree(rt->line_out_notify.gpio_private);
+}
+
+static void pmf_handle_notify_irq(void *data)
+{
+ struct gpio_notification *notif = data;
+
+ schedule_delayed_work(&notif->work, 0);
+}
+
+static int pmf_set_notify(struct gpio_runtime *rt,
+ enum notify_type type,
+ notify_func_t notify,
+ void *data)
+{
+ struct gpio_notification *notif;
+ notify_func_t old;
+ struct pmf_irq_client *irq_client;
+ char *name;
+ int err = -EBUSY;
+
+ switch (type) {
+ case AOA_NOTIFY_HEADPHONE:
+ notif = &rt->headphone_notify;
+ name = "headphone-detect";
+ break;
+ case AOA_NOTIFY_LINE_IN:
+ notif = &rt->line_in_notify;
+ name = "linein-detect";
+ break;
+ case AOA_NOTIFY_LINE_OUT:
+ notif = &rt->line_out_notify;
+ name = "lineout-detect";
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ mutex_lock(&notif->mutex);
+
+ old = notif->notify;
+
+ if (!old && !notify) {
+ err = 0;
+ goto out_unlock;
+ }
+
+ if (old && notify) {
+ if (old == notify && notif->data == data)
+ err = 0;
+ goto out_unlock;
+ }
+
+ if (old && !notify) {
+ irq_client = notif->gpio_private;
+ pmf_unregister_irq_client(irq_client);
+ kfree(irq_client);
+ notif->gpio_private = NULL;
+ }
+ if (!old && notify) {
+ irq_client = kzalloc(sizeof(struct pmf_irq_client),
+ GFP_KERNEL);
+ irq_client->data = notif;
+ irq_client->handler = pmf_handle_notify_irq;
+ irq_client->owner = THIS_MODULE;
+ err = pmf_register_irq_client(rt->node,
+ name,
+ irq_client);
+ if (err) {
+ printk(KERN_ERR "snd-aoa: gpio layer failed to"
+ " register %s irq (%d)\n", name, err);
+ kfree(irq_client);
+ goto out_unlock;
+ }
+ notif->gpio_private = irq_client;
+ }
+ notif->notify = notify;
+ notif->data = data;
+
+ err = 0;
+ out_unlock:
+ mutex_unlock(&notif->mutex);
+ return err;
+}
+
+static int pmf_get_detect(struct gpio_runtime *rt,
+ enum notify_type type)
+{
+ char *name;
+ int err = -EBUSY, ret;
+ struct pmf_args args = { .count = 1, .u[0].p = &ret };
+
+ switch (type) {
+ case AOA_NOTIFY_HEADPHONE:
+ name = "headphone-detect";
+ break;
+ case AOA_NOTIFY_LINE_IN:
+ name = "linein-detect";
+ break;
+ case AOA_NOTIFY_LINE_OUT:
+ name = "lineout-detect";
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ err = pmf_call_function(rt->node, name, &args);
+ if (err)
+ return err;
+ return ret;
+}
+
+static struct gpio_methods methods = {
+ .init = pmf_gpio_init,
+ .exit = pmf_gpio_exit,
+ .all_amps_off = pmf_gpio_all_amps_off,
+ .all_amps_restore = pmf_gpio_all_amps_restore,
+ .set_headphone = pmf_gpio_set_headphone,
+ .set_speakers = pmf_gpio_set_amp,
+ .set_lineout = pmf_gpio_set_lineout,
+ .set_hw_reset = pmf_gpio_set_hw_reset,
+ .get_headphone = pmf_gpio_get_headphone,
+ .get_speakers = pmf_gpio_get_amp,
+ .get_lineout = pmf_gpio_get_lineout,
+ .set_notify = pmf_set_notify,
+ .get_detect = pmf_get_detect,
+};
+
+struct gpio_methods *pmf_gpio_methods = &methods;
+EXPORT_SYMBOL_GPL(pmf_gpio_methods);
diff --git a/sound/aoa/fabrics/Kconfig b/sound/aoa/fabrics/Kconfig
new file mode 100644
index 0000000..3ca475a
--- /dev/null
+++ b/sound/aoa/fabrics/Kconfig
@@ -0,0 +1,11 @@
+config SND_AOA_FABRIC_LAYOUT
+ tristate "layout-id fabric"
+ select SND_AOA_SOUNDBUS
+ select SND_AOA_SOUNDBUS_I2S
+ ---help---
+ This enables the layout-id fabric for the Apple Onboard
+ Audio driver, the module holding it all together
+ based on the device-tree's layout-id property.
+
+ If you are unsure and have a later Apple machine,
+ compile it as a module.
diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile
new file mode 100644
index 0000000..55fc5e7
--- /dev/null
+++ b/sound/aoa/fabrics/Makefile
@@ -0,0 +1 @@
+obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
new file mode 100644
index 0000000..dea7abb
--- /dev/null
+++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
@@ -0,0 +1,1120 @@
+/*
+ * Apple Onboard Audio driver -- layout fabric
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ *
+ * This fabric module looks for sound codecs
+ * based on the layout-id property in the device tree.
+ *
+ */
+
+#include <asm/prom.h>
+#include <linux/list.h>
+#include <linux/module.h>
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Layout-ID fabric for snd-aoa");
+
+#define MAX_CODECS_PER_BUS 2
+
+/* These are the connections the layout fabric
+ * knows about. It doesn't really care about the
+ * input ones, but I thought I'd separate them
+ * to give them proper names. The thing is that
+ * Apple usually will distinguish the active output
+ * by GPIOs, while the active input is set directly
+ * on the codec. Hence we here tell the codec what
+ * we think is connected. This information is hard-
+ * coded below ... */
+#define CC_SPEAKERS (1<<0)
+#define CC_HEADPHONE (1<<1)
+#define CC_LINEOUT (1<<2)
+#define CC_DIGITALOUT (1<<3)
+#define CC_LINEIN (1<<4)
+#define CC_MICROPHONE (1<<5)
+#define CC_DIGITALIN (1<<6)
+/* pretty bogus but users complain...
+ * This is a flag saying that the LINEOUT
+ * should be renamed to HEADPHONE.
+ * be careful with input detection! */
+#define CC_LINEOUT_LABELLED_HEADPHONE (1<<7)
+
+struct codec_connection {
+ /* CC_ flags from above */
+ int connected;
+ /* codec dependent bit to be set in the aoa_codec.connected field.
+ * This intentionally doesn't have any generic flags because the
+ * fabric has to know the codec anyway and all codecs might have
+ * different connectors */
+ int codec_bit;
+};
+
+struct codec_connect_info {
+ char *name;
+ struct codec_connection *connections;
+};
+
+#define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0)
+
+struct layout {
+ unsigned int layout_id;
+ struct codec_connect_info codecs[MAX_CODECS_PER_BUS];
+ int flags;
+
+ /* if busname is not assigned, we use 'Master' below,
+ * so that our layout table doesn't need to be filled
+ * too much.
+ * We only assign these two if we expect to find more
+ * than one soundbus, i.e. on those machines with
+ * multiple layout-ids */
+ char *busname;
+ int pcmid;
+};
+
+MODULE_ALIAS("sound-layout-36");
+MODULE_ALIAS("sound-layout-41");
+MODULE_ALIAS("sound-layout-45");
+MODULE_ALIAS("sound-layout-47");
+MODULE_ALIAS("sound-layout-48");
+MODULE_ALIAS("sound-layout-49");
+MODULE_ALIAS("sound-layout-50");
+MODULE_ALIAS("sound-layout-51");
+MODULE_ALIAS("sound-layout-56");
+MODULE_ALIAS("sound-layout-57");
+MODULE_ALIAS("sound-layout-58");
+MODULE_ALIAS("sound-layout-60");
+MODULE_ALIAS("sound-layout-61");
+MODULE_ALIAS("sound-layout-62");
+MODULE_ALIAS("sound-layout-64");
+MODULE_ALIAS("sound-layout-65");
+MODULE_ALIAS("sound-layout-66");
+MODULE_ALIAS("sound-layout-67");
+MODULE_ALIAS("sound-layout-68");
+MODULE_ALIAS("sound-layout-69");
+MODULE_ALIAS("sound-layout-70");
+MODULE_ALIAS("sound-layout-72");
+MODULE_ALIAS("sound-layout-76");
+MODULE_ALIAS("sound-layout-80");
+MODULE_ALIAS("sound-layout-82");
+MODULE_ALIAS("sound-layout-84");
+MODULE_ALIAS("sound-layout-86");
+MODULE_ALIAS("sound-layout-90");
+MODULE_ALIAS("sound-layout-92");
+MODULE_ALIAS("sound-layout-94");
+MODULE_ALIAS("sound-layout-96");
+MODULE_ALIAS("sound-layout-98");
+MODULE_ALIAS("sound-layout-100");
+
+/* onyx with all but microphone connected */
+static struct codec_connection onyx_connections_nomic[] = {
+ {
+ .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_DIGITALOUT,
+ .codec_bit = 1,
+ },
+ {
+ .connected = CC_LINEIN,
+ .codec_bit = 2,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+/* onyx on machines without headphone */
+static struct codec_connection onyx_connections_noheadphones[] = {
+ {
+ .connected = CC_SPEAKERS | CC_LINEOUT |
+ CC_LINEOUT_LABELLED_HEADPHONE,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_DIGITALOUT,
+ .codec_bit = 1,
+ },
+ /* FIXME: are these correct? probably not for all the machines
+ * below ... If not this will need separating. */
+ {
+ .connected = CC_LINEIN,
+ .codec_bit = 2,
+ },
+ {
+ .connected = CC_MICROPHONE,
+ .codec_bit = 3,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+/* onyx on machines with real line-out */
+static struct codec_connection onyx_connections_reallineout[] = {
+ {
+ .connected = CC_SPEAKERS | CC_LINEOUT | CC_HEADPHONE,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_DIGITALOUT,
+ .codec_bit = 1,
+ },
+ {
+ .connected = CC_LINEIN,
+ .codec_bit = 2,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines without line out */
+static struct codec_connection tas_connections_nolineout[] = {
+ {
+ .connected = CC_SPEAKERS | CC_HEADPHONE,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_LINEIN,
+ .codec_bit = 2,
+ },
+ {
+ .connected = CC_MICROPHONE,
+ .codec_bit = 3,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines with neither line out nor line in */
+static struct codec_connection tas_connections_noline[] = {
+ {
+ .connected = CC_SPEAKERS | CC_HEADPHONE,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_MICROPHONE,
+ .codec_bit = 3,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines without microphone */
+static struct codec_connection tas_connections_nomic[] = {
+ {
+ .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_LINEIN,
+ .codec_bit = 2,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines with everything connected */
+static struct codec_connection tas_connections_all[] = {
+ {
+ .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_LINEIN,
+ .codec_bit = 2,
+ },
+ {
+ .connected = CC_MICROPHONE,
+ .codec_bit = 3,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection toonie_connections[] = {
+ {
+ .connected = CC_SPEAKERS | CC_HEADPHONE,
+ .codec_bit = 0,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection topaz_input[] = {
+ {
+ .connected = CC_DIGITALIN,
+ .codec_bit = 0,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection topaz_output[] = {
+ {
+ .connected = CC_DIGITALOUT,
+ .codec_bit = 1,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection topaz_inout[] = {
+ {
+ .connected = CC_DIGITALIN,
+ .codec_bit = 0,
+ },
+ {
+ .connected = CC_DIGITALOUT,
+ .codec_bit = 1,
+ },
+ {} /* terminate array by .connected == 0 */
+};
+
+static struct layout layouts[] = {
+ /* last PowerBooks (15" Oct 2005) */
+ { .layout_id = 82,
+ .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ /* PowerMac9,1 */
+ { .layout_id = 60,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_reallineout,
+ },
+ },
+ /* PowerMac9,1 */
+ { .layout_id = 61,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ /* PowerBook5,7 */
+ { .layout_id = 64,
+ .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ },
+ /* PowerBook5,7 */
+ { .layout_id = 65,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ /* PowerBook5,9 [17" Oct 2005] */
+ { .layout_id = 84,
+ .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ /* PowerMac8,1 */
+ { .layout_id = 45,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ /* Quad PowerMac (analog in, analog/digital out) */
+ { .layout_id = 68,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_nomic,
+ },
+ },
+ /* Quad PowerMac (digital in) */
+ { .layout_id = 69,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ .busname = "digital in", .pcmid = 1 },
+ /* Early 2005 PowerBook (PowerBook 5,6) */
+ { .layout_id = 70,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nolineout,
+ },
+ },
+ /* PowerBook 5,4 */
+ { .layout_id = 51,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nolineout,
+ },
+ },
+ /* PowerBook6,7 */
+ { .layout_id = 80,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_noline,
+ },
+ },
+ /* PowerBook6,8 */
+ { .layout_id = 72,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nolineout,
+ },
+ },
+ /* PowerMac8,2 */
+ { .layout_id = 86,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_nomic,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ /* PowerBook6,7 */
+ { .layout_id = 92,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nolineout,
+ },
+ },
+ /* PowerMac10,1 (Mac Mini) */
+ { .layout_id = 58,
+ .codecs[0] = {
+ .name = "toonie",
+ .connections = toonie_connections,
+ },
+ },
+ {
+ .layout_id = 96,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ },
+ /* unknown, untested, but this comes from Apple */
+ { .layout_id = 41,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_all,
+ },
+ },
+ { .layout_id = 36,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nomic,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = topaz_inout,
+ },
+ },
+ { .layout_id = 47,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ },
+ { .layout_id = 48,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ { .layout_id = 49,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_nomic,
+ },
+ },
+ { .layout_id = 50,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ { .layout_id = 56,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ },
+ { .layout_id = 57,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ { .layout_id = 62,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = topaz_output,
+ },
+ },
+ { .layout_id = 66,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ },
+ { .layout_id = 67,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ },
+ { .layout_id = 76,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nomic,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = topaz_inout,
+ },
+ },
+ { .layout_id = 90,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_noline,
+ },
+ },
+ { .layout_id = 94,
+ .codecs[0] = {
+ .name = "onyx",
+ /* but it has an external mic?? how to select? */
+ .connections = onyx_connections_noheadphones,
+ },
+ },
+ { .layout_id = 98,
+ .codecs[0] = {
+ .name = "toonie",
+ .connections = toonie_connections,
+ },
+ },
+ { .layout_id = 100,
+ .codecs[0] = {
+ .name = "topaz",
+ .connections = topaz_input,
+ },
+ .codecs[1] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ },
+ {}
+};
+
+static struct layout *find_layout_by_id(unsigned int id)
+{
+ struct layout *l;
+
+ l = layouts;
+ while (l->layout_id) {
+ if (l->layout_id == id)
+ return l;
+ l++;
+ }
+ return NULL;
+}
+
+static void use_layout(struct layout *l)
+{
+ int i;
+
+ for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+ if (l->codecs[i].name) {
+ request_module("snd-aoa-codec-%s", l->codecs[i].name);
+ }
+ }
+ /* now we wait for the codecs to call us back */
+}
+
+struct layout_dev;
+
+struct layout_dev_ptr {
+ struct layout_dev *ptr;
+};
+
+struct layout_dev {
+ struct list_head list;
+ struct soundbus_dev *sdev;
+ struct device_node *sound;
+ struct aoa_codec *codecs[MAX_CODECS_PER_BUS];
+ struct layout *layout;
+ struct gpio_runtime gpio;
+
+ /* we need these for headphone/lineout detection */
+ struct snd_kcontrol *headphone_ctrl;
+ struct snd_kcontrol *lineout_ctrl;
+ struct snd_kcontrol *speaker_ctrl;
+ struct snd_kcontrol *headphone_detected_ctrl;
+ struct snd_kcontrol *lineout_detected_ctrl;
+
+ struct layout_dev_ptr selfptr_headphone;
+ struct layout_dev_ptr selfptr_lineout;
+
+ u32 have_lineout_detect:1,
+ have_headphone_detect:1,
+ switch_on_headphone:1,
+ switch_on_lineout:1;
+};
+
+static LIST_HEAD(layouts_list);
+static int layouts_list_items;
+/* this can go away but only if we allow multiple cards,
+ * make the fabric handle all the card stuff, etc... */
+static struct layout_dev *layout_device;
+
+#define control_info snd_ctl_boolean_mono_info
+
+#define AMP_CONTROL(n, description) \
+static int n##_control_get(struct snd_kcontrol *kcontrol, \
+ struct snd_ctl_elem_value *ucontrol) \
+{ \
+ struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \
+ if (gpio->methods && gpio->methods->get_##n) \
+ ucontrol->value.integer.value[0] = \
+ gpio->methods->get_##n(gpio); \
+ return 0; \
+} \
+static int n##_control_put(struct snd_kcontrol *kcontrol, \
+ struct snd_ctl_elem_value *ucontrol) \
+{ \
+ struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \
+ if (gpio->methods && gpio->methods->get_##n) \
+ gpio->methods->set_##n(gpio, \
+ !!ucontrol->value.integer.value[0]); \
+ return 1; \
+} \
+static struct snd_kcontrol_new n##_ctl = { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = description, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = control_info, \
+ .get = n##_control_get, \
+ .put = n##_control_put, \
+}
+
+AMP_CONTROL(headphone, "Headphone Switch");
+AMP_CONTROL(speakers, "Speakers Switch");
+AMP_CONTROL(lineout, "Line-Out Switch");
+
+static int detect_choice_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
+
+ switch (kcontrol->private_value) {
+ case 0:
+ ucontrol->value.integer.value[0] = ldev->switch_on_headphone;
+ break;
+ case 1:
+ ucontrol->value.integer.value[0] = ldev->switch_on_lineout;
+ break;
+ default:
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static int detect_choice_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
+
+ switch (kcontrol->private_value) {
+ case 0:
+ ldev->switch_on_headphone = !!ucontrol->value.integer.value[0];
+ break;
+ case 1:
+ ldev->switch_on_lineout = !!ucontrol->value.integer.value[0];
+ break;
+ default:
+ return -ENODEV;
+ }
+ return 1;
+}
+
+static struct snd_kcontrol_new headphone_detect_choice = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Detect Autoswitch",
+ .info = control_info,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .get = detect_choice_get,
+ .put = detect_choice_put,
+ .private_value = 0,
+};
+
+static struct snd_kcontrol_new lineout_detect_choice = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line-Out Detect Autoswitch",
+ .info = control_info,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .get = detect_choice_get,
+ .put = detect_choice_put,
+ .private_value = 1,
+};
+
+static int detected_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
+ int v;
+
+ switch (kcontrol->private_value) {
+ case 0:
+ v = ldev->gpio.methods->get_detect(&ldev->gpio,
+ AOA_NOTIFY_HEADPHONE);
+ break;
+ case 1:
+ v = ldev->gpio.methods->get_detect(&ldev->gpio,
+ AOA_NOTIFY_LINE_OUT);
+ break;
+ default:
+ return -ENODEV;
+ }
+ ucontrol->value.integer.value[0] = v;
+ return 0;
+}
+
+static struct snd_kcontrol_new headphone_detected = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Detected",
+ .info = control_info,
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .get = detected_get,
+ .private_value = 0,
+};
+
+static struct snd_kcontrol_new lineout_detected = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line-Out Detected",
+ .info = control_info,
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .get = detected_get,
+ .private_value = 1,
+};
+
+static int check_codec(struct aoa_codec *codec,
+ struct layout_dev *ldev,
+ struct codec_connect_info *cci)
+{
+ const u32 *ref;
+ char propname[32];
+ struct codec_connection *cc;
+
+ /* if the codec has a 'codec' node, we require a reference */
+ if (codec->node && (strcmp(codec->node->name, "codec") == 0)) {
+ snprintf(propname, sizeof(propname),
+ "platform-%s-codec-ref", codec->name);
+ ref = of_get_property(ldev->sound, propname, NULL);
+ if (!ref) {
+ printk(KERN_INFO "snd-aoa-fabric-layout: "
+ "required property %s not present\n", propname);
+ return -ENODEV;
+ }
+ if (*ref != codec->node->linux_phandle) {
+ printk(KERN_INFO "snd-aoa-fabric-layout: "
+ "%s doesn't match!\n", propname);
+ return -ENODEV;
+ }
+ } else {
+ if (layouts_list_items != 1) {
+ printk(KERN_INFO "snd-aoa-fabric-layout: "
+ "more than one soundbus, but no references.\n");
+ return -ENODEV;
+ }
+ }
+ codec->soundbus_dev = ldev->sdev;
+ codec->gpio = &ldev->gpio;
+
+ cc = cci->connections;
+ if (!cc)
+ return -EINVAL;
+
+ printk(KERN_INFO "snd-aoa-fabric-layout: can use this codec\n");
+
+ codec->connected = 0;
+ codec->fabric_data = cc;
+
+ while (cc->connected) {
+ codec->connected |= 1<<cc->codec_bit;
+ cc++;
+ }
+
+ return 0;
+}
+
+static int layout_found_codec(struct aoa_codec *codec)
+{
+ struct layout_dev *ldev;
+ int i;
+
+ list_for_each_entry(ldev, &layouts_list, list) {
+ for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+ if (!ldev->layout->codecs[i].name)
+ continue;
+ if (strcmp(ldev->layout->codecs[i].name, codec->name) == 0) {
+ if (check_codec(codec,
+ ldev,
+ &ldev->layout->codecs[i]) == 0)
+ return 0;
+ }
+ }
+ }
+ return -ENODEV;
+}
+
+static void layout_remove_codec(struct aoa_codec *codec)
+{
+ int i;
+ /* here remove the codec from the layout dev's
+ * codec reference */
+
+ codec->soundbus_dev = NULL;
+ codec->gpio = NULL;
+ for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+ }
+}
+
+static void layout_notify(void *data)
+{
+ struct layout_dev_ptr *dptr = data;
+ struct layout_dev *ldev;
+ int v, update;
+ struct snd_kcontrol *detected, *c;
+ struct snd_card *card = aoa_get_card();
+
+ ldev = dptr->ptr;
+ if (data == &ldev->selfptr_headphone) {
+ v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_HEADPHONE);
+ detected = ldev->headphone_detected_ctrl;
+ update = ldev->switch_on_headphone;
+ if (update) {
+ ldev->gpio.methods->set_speakers(&ldev->gpio, !v);
+ ldev->gpio.methods->set_headphone(&ldev->gpio, v);
+ ldev->gpio.methods->set_lineout(&ldev->gpio, 0);
+ }
+ } else if (data == &ldev->selfptr_lineout) {
+ v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_LINE_OUT);
+ detected = ldev->lineout_detected_ctrl;
+ update = ldev->switch_on_lineout;
+ if (update) {
+ ldev->gpio.methods->set_speakers(&ldev->gpio, !v);
+ ldev->gpio.methods->set_headphone(&ldev->gpio, 0);
+ ldev->gpio.methods->set_lineout(&ldev->gpio, v);
+ }
+ } else
+ return;
+
+ if (detected)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &detected->id);
+ if (update) {
+ c = ldev->headphone_ctrl;
+ if (c)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
+ c = ldev->speaker_ctrl;
+ if (c)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
+ c = ldev->lineout_ctrl;
+ if (c)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
+ }
+}
+
+static void layout_attached_codec(struct aoa_codec *codec)
+{
+ struct codec_connection *cc;
+ struct snd_kcontrol *ctl;
+ int headphones, lineout;
+ struct layout_dev *ldev = layout_device;
+
+ /* need to add this codec to our codec array! */
+
+ cc = codec->fabric_data;
+
+ headphones = codec->gpio->methods->get_detect(codec->gpio,
+ AOA_NOTIFY_HEADPHONE);
+ lineout = codec->gpio->methods->get_detect(codec->gpio,
+ AOA_NOTIFY_LINE_OUT);
+
+ while (cc->connected) {
+ if (cc->connected & CC_SPEAKERS) {
+ if (headphones <= 0 && lineout <= 0)
+ ldev->gpio.methods->set_speakers(codec->gpio, 1);
+ ctl = snd_ctl_new1(&speakers_ctl, codec->gpio);
+ ldev->speaker_ctrl = ctl;
+ aoa_snd_ctl_add(ctl);
+ }
+ if (cc->connected & CC_HEADPHONE) {
+ if (headphones == 1)
+ ldev->gpio.methods->set_headphone(codec->gpio, 1);
+ ctl = snd_ctl_new1(&headphone_ctl, codec->gpio);
+ ldev->headphone_ctrl = ctl;
+ aoa_snd_ctl_add(ctl);
+ ldev->have_headphone_detect =
+ !ldev->gpio.methods
+ ->set_notify(&ldev->gpio,
+ AOA_NOTIFY_HEADPHONE,
+ layout_notify,
+ &ldev->selfptr_headphone);
+ if (ldev->have_headphone_detect) {
+ ctl = snd_ctl_new1(&headphone_detect_choice,
+ ldev);
+ aoa_snd_ctl_add(ctl);
+ ctl = snd_ctl_new1(&headphone_detected,
+ ldev);
+ ldev->headphone_detected_ctrl = ctl;
+ aoa_snd_ctl_add(ctl);
+ }
+ }
+ if (cc->connected & CC_LINEOUT) {
+ if (lineout == 1)
+ ldev->gpio.methods->set_lineout(codec->gpio, 1);
+ ctl = snd_ctl_new1(&lineout_ctl, codec->gpio);
+ if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
+ strlcpy(ctl->id.name,
+ "Headphone Switch", sizeof(ctl->id.name));
+ ldev->lineout_ctrl = ctl;
+ aoa_snd_ctl_add(ctl);
+ ldev->have_lineout_detect =
+ !ldev->gpio.methods
+ ->set_notify(&ldev->gpio,
+ AOA_NOTIFY_LINE_OUT,
+ layout_notify,
+ &ldev->selfptr_lineout);
+ if (ldev->have_lineout_detect) {
+ ctl = snd_ctl_new1(&lineout_detect_choice,
+ ldev);
+ if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
+ strlcpy(ctl->id.name,
+ "Headphone Detect Autoswitch",
+ sizeof(ctl->id.name));
+ aoa_snd_ctl_add(ctl);
+ ctl = snd_ctl_new1(&lineout_detected,
+ ldev);
+ if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
+ strlcpy(ctl->id.name,
+ "Headphone Detected",
+ sizeof(ctl->id.name));
+ ldev->lineout_detected_ctrl = ctl;
+ aoa_snd_ctl_add(ctl);
+ }
+ }
+ cc++;
+ }
+ /* now update initial state */
+ if (ldev->have_headphone_detect)
+ layout_notify(&ldev->selfptr_headphone);
+ if (ldev->have_lineout_detect)
+ layout_notify(&ldev->selfptr_lineout);
+}
+
+static struct aoa_fabric layout_fabric = {
+ .name = "SoundByLayout",
+ .owner = THIS_MODULE,
+ .found_codec = layout_found_codec,
+ .remove_codec = layout_remove_codec,
+ .attached_codec = layout_attached_codec,
+};
+
+static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
+{
+ struct device_node *sound = NULL;
+ const unsigned int *layout_id;
+ struct layout *layout;
+ struct layout_dev *ldev = NULL;
+ int err;
+
+ /* hm, currently we can only have one ... */
+ if (layout_device)
+ return -ENODEV;
+
+ /* by breaking out we keep a reference */
+ while ((sound = of_get_next_child(sdev->ofdev.node, sound))) {
+ if (sound->type && strcasecmp(sound->type, "soundchip") == 0)
+ break;
+ }
+ if (!sound) return -ENODEV;
+
+ layout_id = of_get_property(sound, "layout-id", NULL);
+ if (!layout_id)
+ goto outnodev;
+ printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d\n",
+ *layout_id);
+
+ layout = find_layout_by_id(*layout_id);
+ if (!layout) {
+ printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n");
+ goto outnodev;
+ }
+
+ ldev = kzalloc(sizeof(struct layout_dev), GFP_KERNEL);
+ if (!ldev)
+ goto outnodev;
+
+ layout_device = ldev;
+ ldev->sdev = sdev;
+ ldev->sound = sound;
+ ldev->layout = layout;
+ ldev->gpio.node = sound->parent;
+ switch (layout->layout_id) {
+ case 41: /* that unknown machine no one seems to have */
+ case 51: /* PowerBook5,4 */
+ case 58: /* Mac Mini */
+ ldev->gpio.methods = ftr_gpio_methods;
+ printk(KERN_DEBUG
+ "snd-aoa-fabric-layout: Using direct GPIOs\n");
+ break;
+ default:
+ ldev->gpio.methods = pmf_gpio_methods;
+ printk(KERN_DEBUG
+ "snd-aoa-fabric-layout: Using PMF GPIOs\n");
+ }
+ ldev->selfptr_headphone.ptr = ldev;
+ ldev->selfptr_lineout.ptr = ldev;
+ sdev->ofdev.dev.driver_data = ldev;
+ list_add(&ldev->list, &layouts_list);
+ layouts_list_items++;
+
+ /* assign these before registering ourselves, so
+ * callbacks that are done during registration
+ * already have the values */
+ sdev->pcmid = ldev->layout->pcmid;
+ if (ldev->layout->busname) {
+ sdev->pcmname = ldev->layout->busname;
+ } else {
+ sdev->pcmname = "Master";
+ }
+
+ ldev->gpio.methods->init(&ldev->gpio);
+
+ err = aoa_fabric_register(&layout_fabric, &sdev->ofdev.dev);
+ if (err && err != -EALREADY) {
+ printk(KERN_INFO "snd-aoa-fabric-layout: can't use,"
+ " another fabric is active!\n");
+ goto outlistdel;
+ }
+
+ use_layout(layout);
+ ldev->switch_on_headphone = 1;
+ ldev->switch_on_lineout = 1;
+ return 0;
+ outlistdel:
+ /* we won't be using these then... */
+ ldev->gpio.methods->exit(&ldev->gpio);
+ /* reset if we didn't use it */
+ sdev->pcmname = NULL;
+ sdev->pcmid = -1;
+ list_del(&ldev->list);
+ layouts_list_items--;
+ outnodev:
+ of_node_put(sound);
+ layout_device = NULL;
+ kfree(ldev);
+ return -ENODEV;
+}
+
+static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
+{
+ struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+ int i;
+
+ for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+ if (ldev->codecs[i]) {
+ aoa_fabric_unlink_codec(ldev->codecs[i]);
+ }
+ ldev->codecs[i] = NULL;
+ }
+ list_del(&ldev->list);
+ layouts_list_items--;
+ of_node_put(ldev->sound);
+
+ ldev->gpio.methods->set_notify(&ldev->gpio,
+ AOA_NOTIFY_HEADPHONE,
+ NULL,
+ NULL);
+ ldev->gpio.methods->set_notify(&ldev->gpio,
+ AOA_NOTIFY_LINE_OUT,
+ NULL,
+ NULL);
+
+ ldev->gpio.methods->exit(&ldev->gpio);
+ layout_device = NULL;
+ kfree(ldev);
+ sdev->pcmid = -1;
+ sdev->pcmname = NULL;
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
+{
+ struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+
+ if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
+ ldev->gpio.methods->all_amps_off(&ldev->gpio);
+
+ return 0;
+}
+
+static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
+{
+ struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+
+ if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
+ ldev->gpio.methods->all_amps_restore(&ldev->gpio);
+
+ return 0;
+}
+#endif
+
+static struct soundbus_driver aoa_soundbus_driver = {
+ .name = "snd_aoa_soundbus_drv",
+ .owner = THIS_MODULE,
+ .probe = aoa_fabric_layout_probe,
+ .remove = aoa_fabric_layout_remove,
+#ifdef CONFIG_PM
+ .suspend = aoa_fabric_layout_suspend,
+ .resume = aoa_fabric_layout_resume,
+#endif
+ .driver = {
+ .owner = THIS_MODULE,
+ }
+};
+
+static int __init aoa_fabric_layout_init(void)
+{
+ int err;
+
+ err = soundbus_register_driver(&aoa_soundbus_driver);
+ if (err)
+ return err;
+ return 0;
+}
+
+static void __exit aoa_fabric_layout_exit(void)
+{
+ soundbus_unregister_driver(&aoa_soundbus_driver);
+ aoa_fabric_unregister(&layout_fabric);
+}
+
+module_init(aoa_fabric_layout_init);
+module_exit(aoa_fabric_layout_exit);
diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig
new file mode 100644
index 0000000..839d113
--- /dev/null
+++ b/sound/aoa/soundbus/Kconfig
@@ -0,0 +1,14 @@
+config SND_AOA_SOUNDBUS
+ tristate "Apple Soundbus support"
+ select SND_PCM
+ ---help---
+ This option enables the generic driver for the soundbus
+ support on Apple machines.
+
+ It is required for the sound bus implementations.
+
+config SND_AOA_SOUNDBUS_I2S
+ tristate "I2S bus support"
+ depends on SND_AOA_SOUNDBUS && PCI
+ ---help---
+ This option enables support for Apple I2S busses.
diff --git a/sound/aoa/soundbus/Makefile b/sound/aoa/soundbus/Makefile
new file mode 100644
index 0000000..0e61f5a
--- /dev/null
+++ b/sound/aoa/soundbus/Makefile
@@ -0,0 +1,3 @@
+obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o
+snd-aoa-soundbus-objs := core.o sysfs.o
+obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/
diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c
new file mode 100644
index 0000000..fa8ab28
--- /dev/null
+++ b/sound/aoa/soundbus/core.c
@@ -0,0 +1,219 @@
+/*
+ * soundbus
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/module.h>
+#include "soundbus.h"
+
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Apple Soundbus");
+
+struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev)
+{
+ struct device *tmp;
+
+ if (!dev)
+ return NULL;
+ tmp = get_device(&dev->ofdev.dev);
+ if (tmp)
+ return to_soundbus_device(tmp);
+ else
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(soundbus_dev_get);
+
+void soundbus_dev_put(struct soundbus_dev *dev)
+{
+ if (dev)
+ put_device(&dev->ofdev.dev);
+}
+EXPORT_SYMBOL_GPL(soundbus_dev_put);
+
+static int soundbus_probe(struct device *dev)
+{
+ int error = -ENODEV;
+ struct soundbus_driver *drv;
+ struct soundbus_dev *soundbus_dev;
+
+ drv = to_soundbus_driver(dev->driver);
+ soundbus_dev = to_soundbus_device(dev);
+
+ if (!drv->probe)
+ return error;
+
+ soundbus_dev_get(soundbus_dev);
+
+ error = drv->probe(soundbus_dev);
+ if (error)
+ soundbus_dev_put(soundbus_dev);
+
+ return error;
+}
+
+
+static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env)
+{
+ struct soundbus_dev * soundbus_dev;
+ struct of_device * of;
+ const char *compat;
+ int retval = 0;
+ int cplen, seen = 0;
+
+ if (!dev)
+ return -ENODEV;
+
+ soundbus_dev = to_soundbus_device(dev);
+ if (!soundbus_dev)
+ return -ENODEV;
+
+ of = &soundbus_dev->ofdev;
+
+ /* stuff we want to pass to /sbin/hotplug */
+ retval = add_uevent_var(env, "OF_NAME=%s", of->node->name);
+ if (retval)
+ return retval;
+
+ retval = add_uevent_var(env, "OF_TYPE=%s", of->node->type);
+ if (retval)
+ return retval;
+
+ /* Since the compatible field can contain pretty much anything
+ * it's not really legal to split it out with commas. We split it
+ * up using a number of environment variables instead. */
+
+ compat = of_get_property(of->node, "compatible", &cplen);
+ while (compat && cplen > 0) {
+ int tmp = env->buflen;
+ retval = add_uevent_var(env, "OF_COMPATIBLE_%d=%s", seen, compat);
+ if (retval)
+ return retval;
+ compat += env->buflen - tmp;
+ cplen -= env->buflen - tmp;
+ seen += 1;
+ }
+
+ retval = add_uevent_var(env, "OF_COMPATIBLE_N=%d", seen);
+ if (retval)
+ return retval;
+ retval = add_uevent_var(env, "MODALIAS=%s", soundbus_dev->modalias);
+
+ return retval;
+}
+
+static int soundbus_device_remove(struct device *dev)
+{
+ struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+ struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+ if (dev->driver && drv->remove)
+ drv->remove(soundbus_dev);
+ soundbus_dev_put(soundbus_dev);
+
+ return 0;
+}
+
+static void soundbus_device_shutdown(struct device *dev)
+{
+ struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+ struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+ if (dev->driver && drv->shutdown)
+ drv->shutdown(soundbus_dev);
+}
+
+#ifdef CONFIG_PM
+
+static int soundbus_device_suspend(struct device *dev, pm_message_t state)
+{
+ struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+ struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+ if (dev->driver && drv->suspend)
+ return drv->suspend(soundbus_dev, state);
+ return 0;
+}
+
+static int soundbus_device_resume(struct device * dev)
+{
+ struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+ struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+ if (dev->driver && drv->resume)
+ return drv->resume(soundbus_dev);
+ return 0;
+}
+
+#endif /* CONFIG_PM */
+
+static struct bus_type soundbus_bus_type = {
+ .name = "aoa-soundbus",
+ .probe = soundbus_probe,
+ .uevent = soundbus_uevent,
+ .remove = soundbus_device_remove,
+ .shutdown = soundbus_device_shutdown,
+#ifdef CONFIG_PM
+ .suspend = soundbus_device_suspend,
+ .resume = soundbus_device_resume,
+#endif
+ .dev_attrs = soundbus_dev_attrs,
+};
+
+int soundbus_add_one(struct soundbus_dev *dev)
+{
+ static int devcount;
+
+ /* sanity checks */
+ if (!dev->attach_codec ||
+ !dev->ofdev.node ||
+ dev->pcmname ||
+ dev->pcmid != -1) {
+ printk(KERN_ERR "soundbus: adding device failed sanity check!\n");
+ return -EINVAL;
+ }
+
+ dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount);
+ dev->ofdev.dev.bus = &soundbus_bus_type;
+ return of_device_register(&dev->ofdev);
+}
+EXPORT_SYMBOL_GPL(soundbus_add_one);
+
+void soundbus_remove_one(struct soundbus_dev *dev)
+{
+ of_device_unregister(&dev->ofdev);
+}
+EXPORT_SYMBOL_GPL(soundbus_remove_one);
+
+int soundbus_register_driver(struct soundbus_driver *drv)
+{
+ /* initialize common driver fields */
+ drv->driver.name = drv->name;
+ drv->driver.bus = &soundbus_bus_type;
+
+ /* register with core */
+ return driver_register(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(soundbus_register_driver);
+
+void soundbus_unregister_driver(struct soundbus_driver *drv)
+{
+ driver_unregister(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(soundbus_unregister_driver);
+
+static int __init soundbus_init(void)
+{
+ return bus_register(&soundbus_bus_type);
+}
+
+static void __exit soundbus_exit(void)
+{
+ bus_unregister(&soundbus_bus_type);
+}
+
+subsys_initcall(soundbus_init);
+module_exit(soundbus_exit);
diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile
new file mode 100644
index 0000000..e57a5cf
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
+snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-control.c b/sound/aoa/soundbus/i2sbus/i2sbus-control.c
new file mode 100644
index 0000000..87beb4a
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-control.c
@@ -0,0 +1,193 @@
+/*
+ * i2sbus driver -- bus control routines
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/kernel.h>
+#include <linux/delay.h>
+
+#include <asm/io.h>
+#include <asm/prom.h>
+#include <asm/macio.h>
+#include <asm/pmac_feature.h>
+#include <asm/pmac_pfunc.h>
+#include <asm/keylargo.h>
+
+#include "i2sbus.h"
+
+int i2sbus_control_init(struct macio_dev* dev, struct i2sbus_control **c)
+{
+ *c = kzalloc(sizeof(struct i2sbus_control), GFP_KERNEL);
+ if (!*c)
+ return -ENOMEM;
+
+ INIT_LIST_HEAD(&(*c)->list);
+
+ (*c)->macio = dev->bus->chip;
+ return 0;
+}
+
+void i2sbus_control_destroy(struct i2sbus_control *c)
+{
+ kfree(c);
+}
+
+/* this is serialised externally */
+int i2sbus_control_add_dev(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev)
+{
+ struct device_node *np;
+
+ np = i2sdev->sound.ofdev.node;
+ i2sdev->enable = pmf_find_function(np, "enable");
+ i2sdev->cell_enable = pmf_find_function(np, "cell-enable");
+ i2sdev->clock_enable = pmf_find_function(np, "clock-enable");
+ i2sdev->cell_disable = pmf_find_function(np, "cell-disable");
+ i2sdev->clock_disable = pmf_find_function(np, "clock-disable");
+
+ /* if the bus number is not 0 or 1 we absolutely need to use
+ * the platform functions -- there's nothing in Darwin that
+ * would allow seeing a system behind what the FCRs are then,
+ * and I don't want to go parsing a bunch of platform functions
+ * by hand to try finding a system... */
+ if (i2sdev->bus_number != 0 && i2sdev->bus_number != 1 &&
+ (!i2sdev->enable ||
+ !i2sdev->cell_enable || !i2sdev->clock_enable ||
+ !i2sdev->cell_disable || !i2sdev->clock_disable)) {
+ pmf_put_function(i2sdev->enable);
+ pmf_put_function(i2sdev->cell_enable);
+ pmf_put_function(i2sdev->clock_enable);
+ pmf_put_function(i2sdev->cell_disable);
+ pmf_put_function(i2sdev->clock_disable);
+ return -ENODEV;
+ }
+
+ list_add(&i2sdev->item, &c->list);
+
+ return 0;
+}
+
+void i2sbus_control_remove_dev(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev)
+{
+ /* this is serialised externally */
+ list_del(&i2sdev->item);
+ if (list_empty(&c->list))
+ i2sbus_control_destroy(c);
+}
+
+int i2sbus_control_enable(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev)
+{
+ struct pmf_args args = { .count = 0 };
+ struct macio_chip *macio = c->macio;
+
+ if (i2sdev->enable)
+ return pmf_call_one(i2sdev->enable, &args);
+
+ if (macio == NULL || macio->base == NULL)
+ return -ENODEV;
+
+ switch (i2sdev->bus_number) {
+ case 0:
+ /* these need to be locked or done through
+ * newly created feature calls! */
+ MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_ENABLE);
+ break;
+ case 1:
+ MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_ENABLE);
+ break;
+ default:
+ return -ENODEV;
+ }
+ return 0;
+}
+
+int i2sbus_control_cell(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev,
+ int enable)
+{
+ struct pmf_args args = { .count = 0 };
+ struct macio_chip *macio = c->macio;
+
+ switch (enable) {
+ case 0:
+ if (i2sdev->cell_disable)
+ return pmf_call_one(i2sdev->cell_disable, &args);
+ break;
+ case 1:
+ if (i2sdev->cell_enable)
+ return pmf_call_one(i2sdev->cell_enable, &args);
+ break;
+ default:
+ printk(KERN_ERR "i2sbus: INVALID CELL ENABLE VALUE\n");
+ return -ENODEV;
+ }
+
+ if (macio == NULL || macio->base == NULL)
+ return -ENODEV;
+
+ switch (i2sdev->bus_number) {
+ case 0:
+ if (enable)
+ MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_CELL_ENABLE);
+ else
+ MACIO_BIC(KEYLARGO_FCR1, KL1_I2S0_CELL_ENABLE);
+ break;
+ case 1:
+ if (enable)
+ MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_CELL_ENABLE);
+ else
+ MACIO_BIC(KEYLARGO_FCR1, KL1_I2S1_CELL_ENABLE);
+ break;
+ default:
+ return -ENODEV;
+ }
+ return 0;
+}
+
+int i2sbus_control_clock(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev,
+ int enable)
+{
+ struct pmf_args args = { .count = 0 };
+ struct macio_chip *macio = c->macio;
+
+ switch (enable) {
+ case 0:
+ if (i2sdev->clock_disable)
+ return pmf_call_one(i2sdev->clock_disable, &args);
+ break;
+ case 1:
+ if (i2sdev->clock_enable)
+ return pmf_call_one(i2sdev->clock_enable, &args);
+ break;
+ default:
+ printk(KERN_ERR "i2sbus: INVALID CLOCK ENABLE VALUE\n");
+ return -ENODEV;
+ }
+
+ if (macio == NULL || macio->base == NULL)
+ return -ENODEV;
+
+ switch (i2sdev->bus_number) {
+ case 0:
+ if (enable)
+ MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_CLK_ENABLE_BIT);
+ else
+ MACIO_BIC(KEYLARGO_FCR1, KL1_I2S0_CLK_ENABLE_BIT);
+ break;
+ case 1:
+ if (enable)
+ MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_CLK_ENABLE_BIT);
+ else
+ MACIO_BIC(KEYLARGO_FCR1, KL1_I2S1_CLK_ENABLE_BIT);
+ break;
+ default:
+ return -ENODEV;
+ }
+ return 0;
+}
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
new file mode 100644
index 0000000..b4590df
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
@@ -0,0 +1,450 @@
+/*
+ * i2sbus driver
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/module.h>
+#include <linux/pci.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+
+#include <asm/macio.h>
+#include <asm/dbdma.h>
+
+#include "../soundbus.h"
+#include "i2sbus.h"
+
+MODULE_LICENSE("GPL");
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_DESCRIPTION("Apple Soundbus: I2S support");
+
+static int force;
+module_param(force, int, 0444);
+MODULE_PARM_DESC(force, "Force loading i2sbus even when"
+ " no layout-id property is present");
+
+static struct of_device_id i2sbus_match[] = {
+ { .name = "i2s" },
+ { }
+};
+
+MODULE_DEVICE_TABLE(of, i2sbus_match);
+
+static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
+ struct dbdma_command_mem *r,
+ int numcmds)
+{
+ /* one more for rounding, one for branch back, one for stop command */
+ r->size = (numcmds + 3) * sizeof(struct dbdma_cmd);
+ /* We use the PCI APIs for now until the generic one gets fixed
+ * enough or until we get some macio-specific versions
+ */
+ r->space = dma_alloc_coherent(
+ &macio_get_pci_dev(i2sdev->macio)->dev,
+ r->size,
+ &r->bus_addr,
+ GFP_KERNEL);
+
+ if (!r->space) return -ENOMEM;
+
+ memset(r->space, 0, r->size);
+ r->cmds = (void*)DBDMA_ALIGN(r->space);
+ r->bus_cmd_start = r->bus_addr +
+ (dma_addr_t)((char*)r->cmds - (char*)r->space);
+
+ return 0;
+}
+
+static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
+ struct dbdma_command_mem *r)
+{
+ if (!r->space) return;
+
+ dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev,
+ r->size, r->space, r->bus_addr);
+}
+
+static void i2sbus_release_dev(struct device *dev)
+{
+ struct i2sbus_dev *i2sdev;
+ int i;
+
+ i2sdev = container_of(dev, struct i2sbus_dev, sound.ofdev.dev);
+
+ if (i2sdev->intfregs) iounmap(i2sdev->intfregs);
+ if (i2sdev->out.dbdma) iounmap(i2sdev->out.dbdma);
+ if (i2sdev->in.dbdma) iounmap(i2sdev->in.dbdma);
+ for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++)
+ if (i2sdev->allocated_resource[i])
+ release_and_free_resource(i2sdev->allocated_resource[i]);
+ free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring);
+ free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring);
+ for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++)
+ free_irq(i2sdev->interrupts[i], i2sdev);
+ i2sbus_control_remove_dev(i2sdev->control, i2sdev);
+ mutex_destroy(&i2sdev->lock);
+ kfree(i2sdev);
+}
+
+static irqreturn_t i2sbus_bus_intr(int irq, void *devid)
+{
+ struct i2sbus_dev *dev = devid;
+ u32 intreg;
+
+ spin_lock(&dev->low_lock);
+ intreg = in_le32(&dev->intfregs->intr_ctl);
+
+ /* acknowledge interrupt reasons */
+ out_le32(&dev->intfregs->intr_ctl, intreg);
+
+ spin_unlock(&dev->low_lock);
+
+ return IRQ_HANDLED;
+}
+
+
+/*
+ * XXX FIXME: We test the layout_id's here to get the proper way of
+ * mapping in various registers, thanks to bugs in Apple device-trees.
+ * We could instead key off the machine model and the name of the i2s
+ * node (i2s-a). This we'll do when we move it all to macio_asic.c
+ * and have that export items for each sub-node too.
+ */
+static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index,
+ int layout, struct resource *res)
+{
+ struct device_node *parent;
+ int pindex, rc = -ENXIO;
+ const u32 *reg;
+
+ /* Machines with layout 76 and 36 (K2 based) have a weird device
+ * tree what we need to special case.
+ * Normal machines just fetch the resource from the i2s-X node.
+ * Darwin further divides normal machines into old and new layouts
+ * with a subtely different code path but that doesn't seem necessary
+ * in practice, they just bloated it. In addition, even on our K2
+ * case the i2s-modem node, if we ever want to handle it, uses the
+ * normal layout
+ */
+ if (layout != 76 && layout != 36)
+ return of_address_to_resource(np, index, res);
+
+ parent = of_get_parent(np);
+ pindex = (index == aoa_resource_i2smmio) ? 0 : 1;
+ rc = of_address_to_resource(parent, pindex, res);
+ if (rc)
+ goto bail;
+ reg = of_get_property(np, "reg", NULL);
+ if (reg == NULL) {
+ rc = -ENXIO;
+ goto bail;
+ }
+ res->start += reg[index * 2];
+ res->end = res->start + reg[index * 2 + 1] - 1;
+ bail:
+ of_node_put(parent);
+ return rc;
+}
+
+/* FIXME: look at device node refcounting */
+static int i2sbus_add_dev(struct macio_dev *macio,
+ struct i2sbus_control *control,
+ struct device_node *np)
+{
+ struct i2sbus_dev *dev;
+ struct device_node *child = NULL, *sound = NULL;
+ struct resource *r;
+ int i, layout = 0, rlen, ok = force;
+ static const char *rnames[] = { "i2sbus: %s (control)",
+ "i2sbus: %s (tx)",
+ "i2sbus: %s (rx)" };
+ static irq_handler_t ints[] = {
+ i2sbus_bus_intr,
+ i2sbus_tx_intr,
+ i2sbus_rx_intr
+ };
+
+ if (strlen(np->name) != 5)
+ return 0;
+ if (strncmp(np->name, "i2s-", 4))
+ return 0;
+
+ dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL);
+ if (!dev)
+ return 0;
+
+ i = 0;
+ while ((child = of_get_next_child(np, child))) {
+ if (strcmp(child->name, "sound") == 0) {
+ i++;
+ sound = child;
+ }
+ }
+ if (i == 1) {
+ const u32 *layout_id =
+ of_get_property(sound, "layout-id", NULL);
+ if (layout_id) {
+ layout = *layout_id;
+ snprintf(dev->sound.modalias, 32,
+ "sound-layout-%d", layout);
+ ok = 1;
+ }
+ }
+ /* for the time being, until we can handle non-layout-id
+ * things in some fabric, refuse to attach if there is no
+ * layout-id property or we haven't been forced to attach.
+ * When there are two i2s busses and only one has a layout-id,
+ * then this depends on the order, but that isn't important
+ * either as the second one in that case is just a modem. */
+ if (!ok) {
+ kfree(dev);
+ return -ENODEV;
+ }
+
+ mutex_init(&dev->lock);
+ spin_lock_init(&dev->low_lock);
+ dev->sound.ofdev.node = np;
+ dev->sound.ofdev.dma_mask = macio->ofdev.dma_mask;
+ dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.dma_mask;
+ dev->sound.ofdev.dev.parent = &macio->ofdev.dev;
+ dev->sound.ofdev.dev.release = i2sbus_release_dev;
+ dev->sound.attach_codec = i2sbus_attach_codec;
+ dev->sound.detach_codec = i2sbus_detach_codec;
+ dev->sound.pcmid = -1;
+ dev->macio = macio;
+ dev->control = control;
+ dev->bus_number = np->name[4] - 'a';
+ INIT_LIST_HEAD(&dev->sound.codec_list);
+
+ for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
+ dev->interrupts[i] = -1;
+ snprintf(dev->rnames[i], sizeof(dev->rnames[i]),
+ rnames[i], np->name);
+ }
+ for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
+ int irq = irq_of_parse_and_map(np, i);
+ if (request_irq(irq, ints[i], 0, dev->rnames[i], dev))
+ goto err;
+ dev->interrupts[i] = irq;
+ }
+
+
+ /* Resource handling is problematic as some device-trees contain
+ * useless crap (ugh ugh ugh). We work around that here by calling
+ * specific functions for calculating the appropriate resources.
+ *
+ * This will all be moved to macio_asic.c at one point
+ */
+ for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
+ if (i2sbus_get_and_fixup_rsrc(np,i,layout,&dev->resources[i]))
+ goto err;
+ /* If only we could use our resource dev->resources[i]...
+ * but request_resource doesn't know about parents and
+ * contained resources...
+ */
+ dev->allocated_resource[i] =
+ request_mem_region(dev->resources[i].start,
+ dev->resources[i].end -
+ dev->resources[i].start + 1,
+ dev->rnames[i]);
+ if (!dev->allocated_resource[i]) {
+ printk(KERN_ERR "i2sbus: failed to claim resource %d!\n", i);
+ goto err;
+ }
+ }
+
+ r = &dev->resources[aoa_resource_i2smmio];
+ rlen = r->end - r->start + 1;
+ if (rlen < sizeof(struct i2s_interface_regs))
+ goto err;
+ dev->intfregs = ioremap(r->start, rlen);
+
+ r = &dev->resources[aoa_resource_txdbdma];
+ rlen = r->end - r->start + 1;
+ if (rlen < sizeof(struct dbdma_regs))
+ goto err;
+ dev->out.dbdma = ioremap(r->start, rlen);
+
+ r = &dev->resources[aoa_resource_rxdbdma];
+ rlen = r->end - r->start + 1;
+ if (rlen < sizeof(struct dbdma_regs))
+ goto err;
+ dev->in.dbdma = ioremap(r->start, rlen);
+
+ if (!dev->intfregs || !dev->out.dbdma || !dev->in.dbdma)
+ goto err;
+
+ if (alloc_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring,
+ MAX_DBDMA_COMMANDS))
+ goto err;
+ if (alloc_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring,
+ MAX_DBDMA_COMMANDS))
+ goto err;
+
+ if (i2sbus_control_add_dev(dev->control, dev)) {
+ printk(KERN_ERR "i2sbus: control layer didn't like bus\n");
+ goto err;
+ }
+
+ if (soundbus_add_one(&dev->sound)) {
+ printk(KERN_DEBUG "i2sbus: device registration error!\n");
+ goto err;
+ }
+
+ /* enable this cell */
+ i2sbus_control_cell(dev->control, dev, 1);
+ i2sbus_control_enable(dev->control, dev);
+ i2sbus_control_clock(dev->control, dev, 1);
+
+ return 1;
+ err:
+ for (i=0;i<3;i++)
+ if (dev->interrupts[i] != -1)
+ free_irq(dev->interrupts[i], dev);
+ free_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring);
+ free_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring);
+ if (dev->intfregs) iounmap(dev->intfregs);
+ if (dev->out.dbdma) iounmap(dev->out.dbdma);
+ if (dev->in.dbdma) iounmap(dev->in.dbdma);
+ for (i=0;i<3;i++)
+ if (dev->allocated_resource[i])
+ release_and_free_resource(dev->allocated_resource[i]);
+ mutex_destroy(&dev->lock);
+ kfree(dev);
+ return 0;
+}
+
+static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match)
+{
+ struct device_node *np = NULL;
+ int got = 0, err;
+ struct i2sbus_control *control = NULL;
+
+ err = i2sbus_control_init(dev, &control);
+ if (err)
+ return err;
+ if (!control) {
+ printk(KERN_ERR "i2sbus_control_init API breakage\n");
+ return -ENODEV;
+ }
+
+ while ((np = of_get_next_child(dev->ofdev.node, np))) {
+ if (of_device_is_compatible(np, "i2sbus") ||
+ of_device_is_compatible(np, "i2s-modem")) {
+ got += i2sbus_add_dev(dev, control, np);
+ }
+ }
+
+ if (!got) {
+ /* found none, clean up */
+ i2sbus_control_destroy(control);
+ return -ENODEV;
+ }
+
+ dev->ofdev.dev.driver_data = control;
+
+ return 0;
+}
+
+static int i2sbus_remove(struct macio_dev* dev)
+{
+ struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct i2sbus_dev *i2sdev, *tmp;
+
+ list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
+ soundbus_remove_one(&i2sdev->sound);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
+{
+ struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct codec_info_item *cii;
+ struct i2sbus_dev* i2sdev;
+ int err, ret = 0;
+
+ list_for_each_entry(i2sdev, &control->list, item) {
+ /* Notify Alsa */
+ if (i2sdev->sound.pcm) {
+ /* Suspend PCM streams */
+ snd_pcm_suspend_all(i2sdev->sound.pcm);
+ }
+
+ /* Notify codecs */
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+ err = 0;
+ if (cii->codec->suspend)
+ err = cii->codec->suspend(cii, state);
+ if (err)
+ ret = err;
+ }
+
+ /* wait until streams are stopped */
+ i2sbus_wait_for_stop_both(i2sdev);
+ }
+
+ return ret;
+}
+
+static int i2sbus_resume(struct macio_dev* dev)
+{
+ struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct codec_info_item *cii;
+ struct i2sbus_dev* i2sdev;
+ int err, ret = 0;
+
+ list_for_each_entry(i2sdev, &control->list, item) {
+ /* reset i2s bus format etc. */
+ i2sbus_pcm_prepare_both(i2sdev);
+
+ /* Notify codecs so they can re-initialize */
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+ err = 0;
+ if (cii->codec->resume)
+ err = cii->codec->resume(cii);
+ if (err)
+ ret = err;
+ }
+ }
+
+ return ret;
+}
+#endif /* CONFIG_PM */
+
+static int i2sbus_shutdown(struct macio_dev* dev)
+{
+ return 0;
+}
+
+static struct macio_driver i2sbus_drv = {
+ .name = "soundbus-i2s",
+ .owner = THIS_MODULE,
+ .match_table = i2sbus_match,
+ .probe = i2sbus_probe,
+ .remove = i2sbus_remove,
+#ifdef CONFIG_PM
+ .suspend = i2sbus_suspend,
+ .resume = i2sbus_resume,
+#endif
+ .shutdown = i2sbus_shutdown,
+};
+
+static int __init soundbus_i2sbus_init(void)
+{
+ return macio_register_driver(&i2sbus_drv);
+}
+
+static void __exit soundbus_i2sbus_exit(void)
+{
+ macio_unregister_driver(&i2sbus_drv);
+}
+
+module_init(soundbus_i2sbus_init);
+module_exit(soundbus_i2sbus_exit);
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-interface.h b/sound/aoa/soundbus/i2sbus/i2sbus-interface.h
new file mode 100644
index 0000000..c6b5f54
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-interface.h
@@ -0,0 +1,187 @@
+/*
+ * i2sbus driver -- interface register definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __I2SBUS_INTERFACE_H
+#define __I2SBUS_INTERFACE_H
+
+/* i2s bus control registers, at least what we know about them */
+
+#define __PAD(m,n) u8 __pad##m[n]
+#define _PAD(line, n) __PAD(line, n)
+#define PAD(n) _PAD(__LINE__, (n))
+struct i2s_interface_regs {
+ __le32 intr_ctl; /* 0x00 */
+ PAD(12);
+ __le32 serial_format; /* 0x10 */
+ PAD(12);
+ __le32 codec_msg_out; /* 0x20 */
+ PAD(12);
+ __le32 codec_msg_in; /* 0x30 */
+ PAD(12);
+ __le32 frame_count; /* 0x40 */
+ PAD(12);
+ __le32 frame_match; /* 0x50 */
+ PAD(12);
+ __le32 data_word_sizes; /* 0x60 */
+ PAD(12);
+ __le32 peak_level_sel; /* 0x70 */
+ PAD(12);
+ __le32 peak_level_in0; /* 0x80 */
+ PAD(12);
+ __le32 peak_level_in1; /* 0x90 */
+ PAD(12);
+ /* total size: 0x100 bytes */
+} __attribute__((__packed__));
+
+/* interrupt register is just a bitfield with
+ * interrupt enable and pending bits */
+#define I2S_REG_INTR_CTL 0x00
+# define I2S_INT_FRAME_COUNT (1<<31)
+# define I2S_PENDING_FRAME_COUNT (1<<30)
+# define I2S_INT_MESSAGE_FLAG (1<<29)
+# define I2S_PENDING_MESSAGE_FLAG (1<<28)
+# define I2S_INT_NEW_PEAK (1<<27)
+# define I2S_PENDING_NEW_PEAK (1<<26)
+# define I2S_INT_CLOCKS_STOPPED (1<<25)
+# define I2S_PENDING_CLOCKS_STOPPED (1<<24)
+# define I2S_INT_EXTERNAL_SYNC_ERROR (1<<23)
+# define I2S_PENDING_EXTERNAL_SYNC_ERROR (1<<22)
+# define I2S_INT_EXTERNAL_SYNC_OK (1<<21)
+# define I2S_PENDING_EXTERNAL_SYNC_OK (1<<20)
+# define I2S_INT_NEW_SAMPLE_RATE (1<<19)
+# define I2S_PENDING_NEW_SAMPLE_RATE (1<<18)
+# define I2S_INT_STATUS_FLAG (1<<17)
+# define I2S_PENDING_STATUS_FLAG (1<<16)
+
+/* serial format register is more interesting :)
+ * It contains:
+ * - clock source
+ * - MClk divisor
+ * - SClk divisor
+ * - SClk master flag
+ * - serial format (sony, i2s 64x, i2s 32x, dav, silabs)
+ * - external sample frequency interrupt (don't understand)
+ * - external sample frequency
+ */
+#define I2S_REG_SERIAL_FORMAT 0x10
+/* clock source. You get either 18.432, 45.1584 or 49.1520 MHz */
+# define I2S_SF_CLOCK_SOURCE_SHIFT 30
+# define I2S_SF_CLOCK_SOURCE_MASK (3<<I2S_SF_CLOCK_SOURCE_SHIFT)
+# define I2S_SF_CLOCK_SOURCE_18MHz (0<<I2S_SF_CLOCK_SOURCE_SHIFT)
+# define I2S_SF_CLOCK_SOURCE_45MHz (1<<I2S_SF_CLOCK_SOURCE_SHIFT)
+# define I2S_SF_CLOCK_SOURCE_49MHz (2<<I2S_SF_CLOCK_SOURCE_SHIFT)
+/* also, let's define the exact clock speeds here, in Hz */
+#define I2S_CLOCK_SPEED_18MHz 18432000
+#define I2S_CLOCK_SPEED_45MHz 45158400
+#define I2S_CLOCK_SPEED_49MHz 49152000
+/* MClk is the clock that drives the codec, usually called its 'system clock'.
+ * It is derived by taking only every 'divisor' tick of the clock.
+ */
+# define I2S_SF_MCLKDIV_SHIFT 24
+# define I2S_SF_MCLKDIV_MASK (0x1F<<I2S_SF_MCLKDIV_SHIFT)
+# define I2S_SF_MCLKDIV_1 (0x14<<I2S_SF_MCLKDIV_SHIFT)
+# define I2S_SF_MCLKDIV_3 (0x13<<I2S_SF_MCLKDIV_SHIFT)
+# define I2S_SF_MCLKDIV_5 (0x12<<I2S_SF_MCLKDIV_SHIFT)
+# define I2S_SF_MCLKDIV_14 (0x0E<<I2S_SF_MCLKDIV_SHIFT)
+# define I2S_SF_MCLKDIV_OTHER(div) (((div/2-1)<<I2S_SF_MCLKDIV_SHIFT)&I2S_SF_MCLKDIV_MASK)
+static inline int i2s_sf_mclkdiv(int div, int *out)
+{
+ int d;
+
+ switch(div) {
+ case 1: *out |= I2S_SF_MCLKDIV_1; return 0;
+ case 3: *out |= I2S_SF_MCLKDIV_3; return 0;
+ case 5: *out |= I2S_SF_MCLKDIV_5; return 0;
+ case 14: *out |= I2S_SF_MCLKDIV_14; return 0;
+ default:
+ if (div%2) return -1;
+ d = div/2-1;
+ if (d == 0x14 || d == 0x13 || d == 0x12 || d == 0x0E)
+ return -1;
+ *out |= I2S_SF_MCLKDIV_OTHER(div);
+ return 0;
+ }
+}
+/* SClk is the clock that drives the i2s wire bus. Note that it is
+ * derived from the MClk above by taking only every 'divisor' tick
+ * of MClk.
+ */
+# define I2S_SF_SCLKDIV_SHIFT 20
+# define I2S_SF_SCLKDIV_MASK (0xF<<I2S_SF_SCLKDIV_SHIFT)
+# define I2S_SF_SCLKDIV_1 (8<<I2S_SF_SCLKDIV_SHIFT)
+# define I2S_SF_SCLKDIV_3 (9<<I2S_SF_SCLKDIV_SHIFT)
+# define I2S_SF_SCLKDIV_OTHER(div) (((div/2-1)<<I2S_SF_SCLKDIV_SHIFT)&I2S_SF_SCLKDIV_MASK)
+static inline int i2s_sf_sclkdiv(int div, int *out)
+{
+ int d;
+
+ switch(div) {
+ case 1: *out |= I2S_SF_SCLKDIV_1; return 0;
+ case 3: *out |= I2S_SF_SCLKDIV_3; return 0;
+ default:
+ if (div%2) return -1;
+ d = div/2-1;
+ if (d == 8 || d == 9) return -1;
+ *out |= I2S_SF_SCLKDIV_OTHER(div);
+ return 0;
+ }
+}
+# define I2S_SF_SCLK_MASTER (1<<19)
+/* serial format is the way the data is put to the i2s wire bus */
+# define I2S_SF_SERIAL_FORMAT_SHIFT 16
+# define I2S_SF_SERIAL_FORMAT_MASK (7<<I2S_SF_SERIAL_FORMAT_SHIFT)
+# define I2S_SF_SERIAL_FORMAT_SONY (0<<I2S_SF_SERIAL_FORMAT_SHIFT)
+# define I2S_SF_SERIAL_FORMAT_I2S_64X (1<<I2S_SF_SERIAL_FORMAT_SHIFT)
+# define I2S_SF_SERIAL_FORMAT_I2S_32X (2<<I2S_SF_SERIAL_FORMAT_SHIFT)
+# define I2S_SF_SERIAL_FORMAT_I2S_DAV (4<<I2S_SF_SERIAL_FORMAT_SHIFT)
+# define I2S_SF_SERIAL_FORMAT_I2S_SILABS (5<<I2S_SF_SERIAL_FORMAT_SHIFT)
+/* unknown */
+# define I2S_SF_EXT_SAMPLE_FREQ_INT_SHIFT 12
+# define I2S_SF_EXT_SAMPLE_FREQ_INT_MASK (0xF<<I2S_SF_SAMPLE_FREQ_INT_SHIFT)
+/* probably gives external frequency? */
+# define I2S_SF_EXT_SAMPLE_FREQ_MASK 0xFFF
+
+/* used to send codec messages, but how isn't clear */
+#define I2S_REG_CODEC_MSG_OUT 0x20
+
+/* used to receive codec messages, but how isn't clear */
+#define I2S_REG_CODEC_MSG_IN 0x30
+
+/* frame count reg isn't clear to me yet, but probably useful */
+#define I2S_REG_FRAME_COUNT 0x40
+
+/* program to some value, and get interrupt if frame count reaches it */
+#define I2S_REG_FRAME_MATCH 0x50
+
+/* this register describes how the bus transfers data */
+#define I2S_REG_DATA_WORD_SIZES 0x60
+/* number of interleaved input channels */
+# define I2S_DWS_NUM_CHANNELS_IN_SHIFT 24
+# define I2S_DWS_NUM_CHANNELS_IN_MASK (0x1F<<I2S_DWS_NUM_CHANNELS_IN_SHIFT)
+/* word size of input data */
+# define I2S_DWS_DATA_IN_SIZE_SHIFT 16
+# define I2S_DWS_DATA_IN_16BIT (0<<I2S_DWS_DATA_IN_SIZE_SHIFT)
+# define I2S_DWS_DATA_IN_24BIT (3<<I2S_DWS_DATA_IN_SIZE_SHIFT)
+/* number of interleaved output channels */
+# define I2S_DWS_NUM_CHANNELS_OUT_SHIFT 8
+# define I2S_DWS_NUM_CHANNELS_OUT_MASK (0x1F<<I2S_DWS_NUM_CHANNELS_OUT_SHIFT)
+/* word size of output data */
+# define I2S_DWS_DATA_OUT_SIZE_SHIFT 0
+# define I2S_DWS_DATA_OUT_16BIT (0<<I2S_DWS_DATA_OUT_SIZE_SHIFT)
+# define I2S_DWS_DATA_OUT_24BIT (3<<I2S_DWS_DATA_OUT_SIZE_SHIFT)
+
+
+/* unknown */
+#define I2S_REG_PEAK_LEVEL_SEL 0x70
+
+/* unknown */
+#define I2S_REG_PEAK_LEVEL_IN0 0x80
+
+/* unknown */
+#define I2S_REG_PEAK_LEVEL_IN1 0x90
+
+#endif /* __I2SBUS_INTERFACE_H */
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c
new file mode 100644
index 0000000..59bacd3
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c
@@ -0,0 +1,1062 @@
+/*
+ * i2sbus driver -- pcm routines
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <asm/macio.h>
+#include <linux/pci.h>
+#include "../soundbus.h"
+#include "i2sbus.h"
+
+static inline void get_pcm_info(struct i2sbus_dev *i2sdev, int in,
+ struct pcm_info **pi, struct pcm_info **other)
+{
+ if (in) {
+ if (pi)
+ *pi = &i2sdev->in;
+ if (other)
+ *other = &i2sdev->out;
+ } else {
+ if (pi)
+ *pi = &i2sdev->out;
+ if (other)
+ *other = &i2sdev->in;
+ }
+}
+
+static int clock_and_divisors(int mclk, int sclk, int rate, int *out)
+{
+ /* sclk must be derived from mclk! */
+ if (mclk % sclk)
+ return -1;
+ /* derive sclk register value */
+ if (i2s_sf_sclkdiv(mclk / sclk, out))
+ return -1;
+
+ if (I2S_CLOCK_SPEED_18MHz % (rate * mclk) == 0) {
+ if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_18MHz / (rate * mclk), out)) {
+ *out |= I2S_SF_CLOCK_SOURCE_18MHz;
+ return 0;
+ }
+ }
+ if (I2S_CLOCK_SPEED_45MHz % (rate * mclk) == 0) {
+ if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_45MHz / (rate * mclk), out)) {
+ *out |= I2S_SF_CLOCK_SOURCE_45MHz;
+ return 0;
+ }
+ }
+ if (I2S_CLOCK_SPEED_49MHz % (rate * mclk) == 0) {
+ if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_49MHz / (rate * mclk), out)) {
+ *out |= I2S_SF_CLOCK_SOURCE_49MHz;
+ return 0;
+ }
+ }
+ return -1;
+}
+
+#define CHECK_RATE(rate) \
+ do { if (rates & SNDRV_PCM_RATE_ ##rate) { \
+ int dummy; \
+ if (clock_and_divisors(sysclock_factor, \
+ bus_factor, rate, &dummy)) \
+ rates &= ~SNDRV_PCM_RATE_ ##rate; \
+ } } while (0)
+
+static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in)
+{
+ struct pcm_info *pi, *other;
+ struct soundbus_dev *sdev;
+ int masks_inited = 0, err;
+ struct codec_info_item *cii, *rev;
+ struct snd_pcm_hardware *hw;
+ u64 formats = 0;
+ unsigned int rates = 0;
+ struct transfer_info v;
+ int result = 0;
+ int bus_factor = 0, sysclock_factor = 0;
+ int found_this;
+
+ mutex_lock(&i2sdev->lock);
+
+ get_pcm_info(i2sdev, in, &pi, &other);
+
+ hw = &pi->substream->runtime->hw;
+ sdev = &i2sdev->sound;
+
+ if (pi->active) {
+ /* alsa messed up */
+ result = -EBUSY;
+ goto out_unlock;
+ }
+
+ /* we now need to assign the hw */
+ list_for_each_entry(cii, &sdev->codec_list, list) {
+ struct transfer_info *ti = cii->codec->transfers;
+ bus_factor = cii->codec->bus_factor;
+ sysclock_factor = cii->codec->sysclock_factor;
+ while (ti->formats && ti->rates) {
+ v = *ti;
+ if (ti->transfer_in == in
+ && cii->codec->usable(cii, ti, &v)) {
+ if (masks_inited) {
+ formats &= v.formats;
+ rates &= v.rates;
+ } else {
+ formats = v.formats;
+ rates = v.rates;
+ masks_inited = 1;
+ }
+ }
+ ti++;
+ }
+ }
+ if (!masks_inited || !bus_factor || !sysclock_factor) {
+ result = -ENODEV;
+ goto out_unlock;
+ }
+ /* bus dependent stuff */
+ hw->info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_JOINT_DUPLEX;
+
+ CHECK_RATE(5512);
+ CHECK_RATE(8000);
+ CHECK_RATE(11025);
+ CHECK_RATE(16000);
+ CHECK_RATE(22050);
+ CHECK_RATE(32000);
+ CHECK_RATE(44100);
+ CHECK_RATE(48000);
+ CHECK_RATE(64000);
+ CHECK_RATE(88200);
+ CHECK_RATE(96000);
+ CHECK_RATE(176400);
+ CHECK_RATE(192000);
+ hw->rates = rates;
+
+ /* well. the codec might want 24 bits only, and we'll
+ * ever only transfer 24 bits, but they are top-aligned!
+ * So for alsa, we claim that we're doing full 32 bit
+ * while in reality we'll ignore the lower 8 bits of
+ * that when doing playback (they're transferred as 0
+ * as far as I know, no codecs we have are 32-bit capable
+ * so I can't really test) and when doing recording we'll
+ * always have those lower 8 bits recorded as 0 */
+ if (formats & SNDRV_PCM_FMTBIT_S24_BE)
+ formats |= SNDRV_PCM_FMTBIT_S32_BE;
+ if (formats & SNDRV_PCM_FMTBIT_U24_BE)
+ formats |= SNDRV_PCM_FMTBIT_U32_BE;
+ /* now mask off what we can support. I suppose we could
+ * also support S24_3LE and some similar formats, but I
+ * doubt there's a codec that would be able to use that,
+ * so we don't support it here. */
+ hw->formats = formats & (SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U16_BE |
+ SNDRV_PCM_FMTBIT_S32_BE |
+ SNDRV_PCM_FMTBIT_U32_BE);
+
+ /* we need to set the highest and lowest rate possible.
+ * These are the highest and lowest rates alsa can
+ * support properly in its bitfield.
+ * Below, we'll use that to restrict to the rate
+ * currently in use (if any). */
+ hw->rate_min = 5512;
+ hw->rate_max = 192000;
+ /* if the other stream is active, then we can only
+ * support what it is currently using.
+ * FIXME: I lied. This comment is wrong. We can support
+ * anything that works with the same serial format, ie.
+ * when recording 24 bit sound we can well play 16 bit
+ * sound at the same time iff using the same transfer mode.
+ */
+ if (other->active) {
+ /* FIXME: is this guaranteed by the alsa api? */
+ hw->formats &= (1ULL << i2sdev->format);
+ /* see above, restrict rates to the one we already have */
+ hw->rate_min = i2sdev->rate;
+ hw->rate_max = i2sdev->rate;
+ }
+
+ hw->channels_min = 2;
+ hw->channels_max = 2;
+ /* these are somewhat arbitrary */
+ hw->buffer_bytes_max = 131072;
+ hw->period_bytes_min = 256;
+ hw->period_bytes_max = 16384;
+ hw->periods_min = 3;
+ hw->periods_max = MAX_DBDMA_COMMANDS;
+ err = snd_pcm_hw_constraint_integer(pi->substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0) {
+ result = err;
+ goto out_unlock;
+ }
+ list_for_each_entry(cii, &sdev->codec_list, list) {
+ if (cii->codec->open) {
+ err = cii->codec->open(cii, pi->substream);
+ if (err) {
+ result = err;
+ /* unwind */
+ found_this = 0;
+ list_for_each_entry_reverse(rev,
+ &sdev->codec_list, list) {
+ if (found_this && rev->codec->close) {
+ rev->codec->close(rev,
+ pi->substream);
+ }
+ if (rev == cii)
+ found_this = 1;
+ }
+ goto out_unlock;
+ }
+ }
+ }
+
+ out_unlock:
+ mutex_unlock(&i2sdev->lock);
+ return result;
+}
+
+#undef CHECK_RATE
+
+static int i2sbus_pcm_close(struct i2sbus_dev *i2sdev, int in)
+{
+ struct codec_info_item *cii;
+ struct pcm_info *pi;
+ int err = 0, tmp;
+
+ mutex_lock(&i2sdev->lock);
+
+ get_pcm_info(i2sdev, in, &pi, NULL);
+
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+ if (cii->codec->close) {
+ tmp = cii->codec->close(cii, pi->substream);
+ if (tmp)
+ err = tmp;
+ }
+ }
+
+ pi->substream = NULL;
+ pi->active = 0;
+ mutex_unlock(&i2sdev->lock);
+ return err;
+}
+
+static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev,
+ struct pcm_info *pi)
+{
+ unsigned long flags;
+ struct completion done;
+ long timeout;
+
+ spin_lock_irqsave(&i2sdev->low_lock, flags);
+ if (pi->dbdma_ring.stopping) {
+ init_completion(&done);
+ pi->stop_completion = &done;
+ spin_unlock_irqrestore(&i2sdev->low_lock, flags);
+ timeout = wait_for_completion_timeout(&done, HZ);
+ spin_lock_irqsave(&i2sdev->low_lock, flags);
+ pi->stop_completion = NULL;
+ if (timeout == 0) {
+ /* timeout expired, stop dbdma forcefully */
+ printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n");
+ /* make sure RUN, PAUSE and S0 bits are cleared */
+ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
+ pi->dbdma_ring.stopping = 0;
+ timeout = 10;
+ while (in_le32(&pi->dbdma->status) & ACTIVE) {
+ if (--timeout <= 0)
+ break;
+ udelay(1);
+ }
+ }
+ }
+ spin_unlock_irqrestore(&i2sdev->low_lock, flags);
+}
+
+#ifdef CONFIG_PM
+void i2sbus_wait_for_stop_both(struct i2sbus_dev *i2sdev)
+{
+ struct pcm_info *pi;
+
+ get_pcm_info(i2sdev, 0, &pi, NULL);
+ i2sbus_wait_for_stop(i2sdev, pi);
+ get_pcm_info(i2sdev, 1, &pi, NULL);
+ i2sbus_wait_for_stop(i2sdev, pi);
+}
+#endif
+
+static int i2sbus_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+}
+
+static inline int i2sbus_hw_free(struct snd_pcm_substream *substream, int in)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+ struct pcm_info *pi;
+
+ get_pcm_info(i2sdev, in, &pi, NULL);
+ if (pi->dbdma_ring.stopping)
+ i2sbus_wait_for_stop(i2sdev, pi);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int i2sbus_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ return i2sbus_hw_free(substream, 0);
+}
+
+static int i2sbus_record_hw_free(struct snd_pcm_substream *substream)
+{
+ return i2sbus_hw_free(substream, 1);
+}
+
+static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in)
+{
+ /* whee. Hard work now. The user has selected a bitrate
+ * and bit format, so now we have to program our
+ * I2S controller appropriately. */
+ struct snd_pcm_runtime *runtime;
+ struct dbdma_cmd *command;
+ int i, periodsize, nperiods;
+ dma_addr_t offset;
+ struct bus_info bi;
+ struct codec_info_item *cii;
+ int sfr = 0; /* serial format register */
+ int dws = 0; /* data word sizes reg */
+ int input_16bit;
+ struct pcm_info *pi, *other;
+ int cnt;
+ int result = 0;
+ unsigned int cmd, stopaddr;
+
+ mutex_lock(&i2sdev->lock);
+
+ get_pcm_info(i2sdev, in, &pi, &other);
+
+ if (pi->dbdma_ring.running) {
+ result = -EBUSY;
+ goto out_unlock;
+ }
+ if (pi->dbdma_ring.stopping)
+ i2sbus_wait_for_stop(i2sdev, pi);
+
+ if (!pi->substream || !pi->substream->runtime) {
+ result = -EINVAL;
+ goto out_unlock;
+ }
+
+ runtime = pi->substream->runtime;
+ pi->active = 1;
+ if (other->active &&
+ ((i2sdev->format != runtime->format)
+ || (i2sdev->rate != runtime->rate))) {
+ result = -EINVAL;
+ goto out_unlock;
+ }
+
+ i2sdev->format = runtime->format;
+ i2sdev->rate = runtime->rate;
+
+ periodsize = snd_pcm_lib_period_bytes(pi->substream);
+ nperiods = pi->substream->runtime->periods;
+ pi->current_period = 0;
+
+ /* generate dbdma command ring first */
+ command = pi->dbdma_ring.cmds;
+ memset(command, 0, (nperiods + 2) * sizeof(struct dbdma_cmd));
+
+ /* commands to DMA to/from the ring */
+ /*
+ * For input, we need to do a graceful stop; if we abort
+ * the DMA, we end up with leftover bytes that corrupt
+ * the next recording. To do this we set the S0 status
+ * bit and wait for the DMA controller to stop. Each
+ * command has a branch condition to
+ * make it branch to a stop command if S0 is set.
+ * On input we also need to wait for the S7 bit to be
+ * set before turning off the DMA controller.
+ * In fact we do the graceful stop for output as well.
+ */
+ offset = runtime->dma_addr;
+ cmd = (in? INPUT_MORE: OUTPUT_MORE) | BR_IFSET | INTR_ALWAYS;
+ stopaddr = pi->dbdma_ring.bus_cmd_start +
+ (nperiods + 1) * sizeof(struct dbdma_cmd);
+ for (i = 0; i < nperiods; i++, command++, offset += periodsize) {
+ command->command = cpu_to_le16(cmd);
+ command->cmd_dep = cpu_to_le32(stopaddr);
+ command->phy_addr = cpu_to_le32(offset);
+ command->req_count = cpu_to_le16(periodsize);
+ }
+
+ /* branch back to beginning of ring */
+ command->command = cpu_to_le16(DBDMA_NOP | BR_ALWAYS);
+ command->cmd_dep = cpu_to_le32(pi->dbdma_ring.bus_cmd_start);
+ command++;
+
+ /* set stop command */
+ command->command = cpu_to_le16(DBDMA_STOP);
+
+ /* ok, let's set the serial format and stuff */
+ switch (runtime->format) {
+ /* 16 bit formats */
+ case SNDRV_PCM_FORMAT_S16_BE:
+ case SNDRV_PCM_FORMAT_U16_BE:
+ /* FIXME: if we add different bus factors we need to
+ * do more here!! */
+ bi.bus_factor = 0;
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+ bi.bus_factor = cii->codec->bus_factor;
+ break;
+ }
+ if (!bi.bus_factor) {
+ result = -ENODEV;
+ goto out_unlock;
+ }
+ input_16bit = 1;
+ break;
+ case SNDRV_PCM_FORMAT_S32_BE:
+ case SNDRV_PCM_FORMAT_U32_BE:
+ /* force 64x bus speed, otherwise the data cannot be
+ * transferred quickly enough! */
+ bi.bus_factor = 64;
+ input_16bit = 0;
+ break;
+ default:
+ result = -EINVAL;
+ goto out_unlock;
+ }
+ /* we assume all sysclocks are the same! */
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+ bi.sysclock_factor = cii->codec->sysclock_factor;
+ break;
+ }
+
+ if (clock_and_divisors(bi.sysclock_factor,
+ bi.bus_factor,
+ runtime->rate,
+ &sfr) < 0) {
+ result = -EINVAL;
+ goto out_unlock;
+ }
+ switch (bi.bus_factor) {
+ case 32:
+ sfr |= I2S_SF_SERIAL_FORMAT_I2S_32X;
+ break;
+ case 64:
+ sfr |= I2S_SF_SERIAL_FORMAT_I2S_64X;
+ break;
+ }
+ /* FIXME: THIS ASSUMES MASTER ALL THE TIME */
+ sfr |= I2S_SF_SCLK_MASTER;
+
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+ int err = 0;
+ if (cii->codec->prepare)
+ err = cii->codec->prepare(cii, &bi, pi->substream);
+ if (err) {
+ result = err;
+ goto out_unlock;
+ }
+ }
+ /* codecs are fine with it, so set our clocks */
+ if (input_16bit)
+ dws = (2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) |
+ (2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) |
+ I2S_DWS_DATA_IN_16BIT | I2S_DWS_DATA_OUT_16BIT;
+ else
+ dws = (2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) |
+ (2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) |
+ I2S_DWS_DATA_IN_24BIT | I2S_DWS_DATA_OUT_24BIT;
+
+ /* early exit if already programmed correctly */
+ /* not locking these is fine since we touch them only in this function */
+ if (in_le32(&i2sdev->intfregs->serial_format) == sfr
+ && in_le32(&i2sdev->intfregs->data_word_sizes) == dws)
+ goto out_unlock;
+
+ /* let's notify the codecs about clocks going away.
+ * For now we only do mastering on the i2s cell... */
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+ if (cii->codec->switch_clock)
+ cii->codec->switch_clock(cii, CLOCK_SWITCH_PREPARE_SLAVE);
+
+ i2sbus_control_enable(i2sdev->control, i2sdev);
+ i2sbus_control_cell(i2sdev->control, i2sdev, 1);
+
+ out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED);
+
+ i2sbus_control_clock(i2sdev->control, i2sdev, 0);
+
+ msleep(1);
+
+ /* wait for clock stopped. This can apparently take a while... */
+ cnt = 100;
+ while (cnt-- &&
+ !(in_le32(&i2sdev->intfregs->intr_ctl) & I2S_PENDING_CLOCKS_STOPPED)) {
+ msleep(5);
+ }
+ out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED);
+
+ /* not locking these is fine since we touch them only in this function */
+ out_le32(&i2sdev->intfregs->serial_format, sfr);
+ out_le32(&i2sdev->intfregs->data_word_sizes, dws);
+
+ i2sbus_control_enable(i2sdev->control, i2sdev);
+ i2sbus_control_cell(i2sdev->control, i2sdev, 1);
+ i2sbus_control_clock(i2sdev->control, i2sdev, 1);
+ msleep(1);
+
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+ if (cii->codec->switch_clock)
+ cii->codec->switch_clock(cii, CLOCK_SWITCH_SLAVE);
+
+ out_unlock:
+ mutex_unlock(&i2sdev->lock);
+ return result;
+}
+
+#ifdef CONFIG_PM
+void i2sbus_pcm_prepare_both(struct i2sbus_dev *i2sdev)
+{
+ i2sbus_pcm_prepare(i2sdev, 0);
+ i2sbus_pcm_prepare(i2sdev, 1);
+}
+#endif
+
+static int i2sbus_pcm_trigger(struct i2sbus_dev *i2sdev, int in, int cmd)
+{
+ struct codec_info_item *cii;
+ struct pcm_info *pi;
+ int result = 0;
+ unsigned long flags;
+
+ spin_lock_irqsave(&i2sdev->low_lock, flags);
+
+ get_pcm_info(i2sdev, in, &pi, NULL);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (pi->dbdma_ring.running) {
+ result = -EALREADY;
+ goto out_unlock;
+ }
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+ if (cii->codec->start)
+ cii->codec->start(cii, pi->substream);
+ pi->dbdma_ring.running = 1;
+
+ if (pi->dbdma_ring.stopping) {
+ /* Clear the S0 bit, then see if we stopped yet */
+ out_le32(&pi->dbdma->control, 1 << 16);
+ if (in_le32(&pi->dbdma->status) & ACTIVE) {
+ /* possible race here? */
+ udelay(10);
+ if (in_le32(&pi->dbdma->status) & ACTIVE) {
+ pi->dbdma_ring.stopping = 0;
+ goto out_unlock; /* keep running */
+ }
+ }
+ }
+
+ /* make sure RUN, PAUSE and S0 bits are cleared */
+ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
+
+ /* set branch condition select register */
+ out_le32(&pi->dbdma->br_sel, (1 << 16) | 1);
+
+ /* write dma command buffer address to the dbdma chip */
+ out_le32(&pi->dbdma->cmdptr, pi->dbdma_ring.bus_cmd_start);
+
+ /* initialize the frame count and current period */
+ pi->current_period = 0;
+ pi->frame_count = in_le32(&i2sdev->intfregs->frame_count);
+
+ /* set the DMA controller running */
+ out_le32(&pi->dbdma->control, (RUN << 16) | RUN);
+
+ /* off you go! */
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (!pi->dbdma_ring.running) {
+ result = -EALREADY;
+ goto out_unlock;
+ }
+ pi->dbdma_ring.running = 0;
+
+ /* Set the S0 bit to make the DMA branch to the stop cmd */
+ out_le32(&pi->dbdma->control, (1 << 16) | 1);
+ pi->dbdma_ring.stopping = 1;
+
+ list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+ if (cii->codec->stop)
+ cii->codec->stop(cii, pi->substream);
+ break;
+ default:
+ result = -EINVAL;
+ goto out_unlock;
+ }
+
+ out_unlock:
+ spin_unlock_irqrestore(&i2sdev->low_lock, flags);
+ return result;
+}
+
+static snd_pcm_uframes_t i2sbus_pcm_pointer(struct i2sbus_dev *i2sdev, int in)
+{
+ struct pcm_info *pi;
+ u32 fc;
+
+ get_pcm_info(i2sdev, in, &pi, NULL);
+
+ fc = in_le32(&i2sdev->intfregs->frame_count);
+ fc = fc - pi->frame_count;
+
+ if (fc >= pi->substream->runtime->buffer_size)
+ fc %= pi->substream->runtime->buffer_size;
+ return fc;
+}
+
+static inline void handle_interrupt(struct i2sbus_dev *i2sdev, int in)
+{
+ struct pcm_info *pi;
+ u32 fc, nframes;
+ u32 status;
+ int timeout, i;
+ int dma_stopped = 0;
+ struct snd_pcm_runtime *runtime;
+
+ spin_lock(&i2sdev->low_lock);
+ get_pcm_info(i2sdev, in, &pi, NULL);
+ if (!pi->dbdma_ring.running && !pi->dbdma_ring.stopping)
+ goto out_unlock;
+
+ i = pi->current_period;
+ runtime = pi->substream->runtime;
+ while (pi->dbdma_ring.cmds[i].xfer_status) {
+ if (le16_to_cpu(pi->dbdma_ring.cmds[i].xfer_status) & BT)
+ /*
+ * BT is the branch taken bit. If it took a branch
+ * it is because we set the S0 bit to make it
+ * branch to the stop command.
+ */
+ dma_stopped = 1;
+ pi->dbdma_ring.cmds[i].xfer_status = 0;
+
+ if (++i >= runtime->periods) {
+ i = 0;
+ pi->frame_count += runtime->buffer_size;
+ }
+ pi->current_period = i;
+
+ /*
+ * Check the frame count. The DMA tends to get a bit
+ * ahead of the frame counter, which confuses the core.
+ */
+ fc = in_le32(&i2sdev->intfregs->frame_count);
+ nframes = i * runtime->period_size;
+ if (fc < pi->frame_count + nframes)
+ pi->frame_count = fc - nframes;
+ }
+
+ if (dma_stopped) {
+ timeout = 1000;
+ for (;;) {
+ status = in_le32(&pi->dbdma->status);
+ if (!(status & ACTIVE) && (!in || (status & 0x80)))
+ break;
+ if (--timeout <= 0) {
+ printk(KERN_ERR "i2sbus: timed out "
+ "waiting for DMA to stop!\n");
+ break;
+ }
+ udelay(1);
+ }
+
+ /* Turn off DMA controller, clear S0 bit */
+ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
+
+ pi->dbdma_ring.stopping = 0;
+ if (pi->stop_completion)
+ complete(pi->stop_completion);
+ }
+
+ if (!pi->dbdma_ring.running)
+ goto out_unlock;
+ spin_unlock(&i2sdev->low_lock);
+ /* may call _trigger again, hence needs to be unlocked */
+ snd_pcm_period_elapsed(pi->substream);
+ return;
+
+ out_unlock:
+ spin_unlock(&i2sdev->low_lock);
+}
+
+irqreturn_t i2sbus_tx_intr(int irq, void *devid)
+{
+ handle_interrupt((struct i2sbus_dev *)devid, 0);
+ return IRQ_HANDLED;
+}
+
+irqreturn_t i2sbus_rx_intr(int irq, void *devid)
+{
+ handle_interrupt((struct i2sbus_dev *)devid, 1);
+ return IRQ_HANDLED;
+}
+
+static int i2sbus_playback_open(struct snd_pcm_substream *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ i2sdev->out.substream = substream;
+ return i2sbus_pcm_open(i2sdev, 0);
+}
+
+static int i2sbus_playback_close(struct snd_pcm_substream *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+ int err;
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->out.substream != substream)
+ return -EINVAL;
+ err = i2sbus_pcm_close(i2sdev, 0);
+ if (!err)
+ i2sdev->out.substream = NULL;
+ return err;
+}
+
+static int i2sbus_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->out.substream != substream)
+ return -EINVAL;
+ return i2sbus_pcm_prepare(i2sdev, 0);
+}
+
+static int i2sbus_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->out.substream != substream)
+ return -EINVAL;
+ return i2sbus_pcm_trigger(i2sdev, 0, cmd);
+}
+
+static snd_pcm_uframes_t i2sbus_playback_pointer(struct snd_pcm_substream
+ *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->out.substream != substream)
+ return 0;
+ return i2sbus_pcm_pointer(i2sdev, 0);
+}
+
+static struct snd_pcm_ops i2sbus_playback_ops = {
+ .open = i2sbus_playback_open,
+ .close = i2sbus_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = i2sbus_hw_params,
+ .hw_free = i2sbus_playback_hw_free,
+ .prepare = i2sbus_playback_prepare,
+ .trigger = i2sbus_playback_trigger,
+ .pointer = i2sbus_playback_pointer,
+};
+
+static int i2sbus_record_open(struct snd_pcm_substream *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ i2sdev->in.substream = substream;
+ return i2sbus_pcm_open(i2sdev, 1);
+}
+
+static int i2sbus_record_close(struct snd_pcm_substream *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+ int err;
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->in.substream != substream)
+ return -EINVAL;
+ err = i2sbus_pcm_close(i2sdev, 1);
+ if (!err)
+ i2sdev->in.substream = NULL;
+ return err;
+}
+
+static int i2sbus_record_prepare(struct snd_pcm_substream *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->in.substream != substream)
+ return -EINVAL;
+ return i2sbus_pcm_prepare(i2sdev, 1);
+}
+
+static int i2sbus_record_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->in.substream != substream)
+ return -EINVAL;
+ return i2sbus_pcm_trigger(i2sdev, 1, cmd);
+}
+
+static snd_pcm_uframes_t i2sbus_record_pointer(struct snd_pcm_substream
+ *substream)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+ if (!i2sdev)
+ return -EINVAL;
+ if (i2sdev->in.substream != substream)
+ return 0;
+ return i2sbus_pcm_pointer(i2sdev, 1);
+}
+
+static struct snd_pcm_ops i2sbus_record_ops = {
+ .open = i2sbus_record_open,
+ .close = i2sbus_record_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = i2sbus_hw_params,
+ .hw_free = i2sbus_record_hw_free,
+ .prepare = i2sbus_record_prepare,
+ .trigger = i2sbus_record_trigger,
+ .pointer = i2sbus_record_pointer,
+};
+
+static void i2sbus_private_free(struct snd_pcm *pcm)
+{
+ struct i2sbus_dev *i2sdev = snd_pcm_chip(pcm);
+ struct codec_info_item *p, *tmp;
+
+ i2sdev->sound.pcm = NULL;
+ i2sdev->out.created = 0;
+ i2sdev->in.created = 0;
+ list_for_each_entry_safe(p, tmp, &i2sdev->sound.codec_list, list) {
+ printk(KERN_ERR "i2sbus: a codec didn't unregister!\n");
+ list_del(&p->list);
+ module_put(p->codec->owner);
+ kfree(p);
+ }
+ soundbus_dev_put(&i2sdev->sound);
+ module_put(THIS_MODULE);
+}
+
+int
+i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card,
+ struct codec_info *ci, void *data)
+{
+ int err, in = 0, out = 0;
+ struct transfer_info *tmp;
+ struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev);
+ struct codec_info_item *cii;
+
+ if (!dev->pcmname || dev->pcmid == -1) {
+ printk(KERN_ERR "i2sbus: pcm name and id must be set!\n");
+ return -EINVAL;
+ }
+
+ list_for_each_entry(cii, &dev->codec_list, list) {
+ if (cii->codec_data == data)
+ return -EALREADY;
+ }
+
+ if (!ci->transfers || !ci->transfers->formats
+ || !ci->transfers->rates || !ci->usable)
+ return -EINVAL;
+
+ /* we currently code the i2s transfer on the clock, and support only
+ * 32 and 64 */
+ if (ci->bus_factor != 32 && ci->bus_factor != 64)
+ return -EINVAL;
+
+ /* If you want to fix this, you need to keep track of what transport infos
+ * are to be used, which codecs they belong to, and then fix all the
+ * sysclock/busclock stuff above to depend on which is usable */
+ list_for_each_entry(cii, &dev->codec_list, list) {
+ if (cii->codec->sysclock_factor != ci->sysclock_factor) {
+ printk(KERN_DEBUG
+ "cannot yet handle multiple different sysclocks!\n");
+ return -EINVAL;
+ }
+ if (cii->codec->bus_factor != ci->bus_factor) {
+ printk(KERN_DEBUG
+ "cannot yet handle multiple different bus clocks!\n");
+ return -EINVAL;
+ }
+ }
+
+ tmp = ci->transfers;
+ while (tmp->formats && tmp->rates) {
+ if (tmp->transfer_in)
+ in = 1;
+ else
+ out = 1;
+ tmp++;
+ }
+
+ cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL);
+ if (!cii) {
+ printk(KERN_DEBUG "i2sbus: failed to allocate cii\n");
+ return -ENOMEM;
+ }
+
+ /* use the private data to point to the codec info */
+ cii->sdev = soundbus_dev_get(dev);
+ cii->codec = ci;
+ cii->codec_data = data;
+
+ if (!cii->sdev) {
+ printk(KERN_DEBUG
+ "i2sbus: failed to get soundbus dev reference\n");
+ err = -ENODEV;
+ goto out_free_cii;
+ }
+
+ if (!try_module_get(THIS_MODULE)) {
+ printk(KERN_DEBUG "i2sbus: failed to get module reference!\n");
+ err = -EBUSY;
+ goto out_put_sdev;
+ }
+
+ if (!try_module_get(ci->owner)) {
+ printk(KERN_DEBUG
+ "i2sbus: failed to get module reference to codec owner!\n");
+ err = -EBUSY;
+ goto out_put_this_module;
+ }
+
+ if (!dev->pcm) {
+ err = snd_pcm_new(card, dev->pcmname, dev->pcmid, 0, 0,
+ &dev->pcm);
+ if (err) {
+ printk(KERN_DEBUG "i2sbus: failed to create pcm\n");
+ goto out_put_ci_module;
+ }
+ dev->pcm->dev = &dev->ofdev.dev;
+ }
+
+ /* ALSA yet again sucks.
+ * If it is ever fixed, remove this line. See below. */
+ out = in = 1;
+
+ if (!i2sdev->out.created && out) {
+ if (dev->pcm->card != card) {
+ /* eh? */
+ printk(KERN_ERR
+ "Can't attach same bus to different cards!\n");
+ err = -EINVAL;
+ goto out_put_ci_module;
+ }
+ err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1);
+ if (err)
+ goto out_put_ci_module;
+ snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &i2sbus_playback_ops);
+ i2sdev->out.created = 1;
+ }
+
+ if (!i2sdev->in.created && in) {
+ if (dev->pcm->card != card) {
+ printk(KERN_ERR
+ "Can't attach same bus to different cards!\n");
+ err = -EINVAL;
+ goto out_put_ci_module;
+ }
+ err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1);
+ if (err)
+ goto out_put_ci_module;
+ snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &i2sbus_record_ops);
+ i2sdev->in.created = 1;
+ }
+
+ /* so we have to register the pcm after adding any substream
+ * to it because alsa doesn't create the devices for the
+ * substreams when we add them later.
+ * Therefore, force in and out on both busses (above) and
+ * register the pcm now instead of just after creating it.
+ */
+ err = snd_device_register(card, dev->pcm);
+ if (err) {
+ printk(KERN_ERR "i2sbus: error registering new pcm\n");
+ goto out_put_ci_module;
+ }
+ /* no errors any more, so let's add this to our list */
+ list_add(&cii->list, &dev->codec_list);
+
+ dev->pcm->private_data = i2sdev;
+ dev->pcm->private_free = i2sbus_private_free;
+
+ /* well, we really should support scatter/gather DMA */
+ snd_pcm_lib_preallocate_pages_for_all(
+ dev->pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(macio_get_pci_dev(i2sdev->macio)),
+ 64 * 1024, 64 * 1024);
+
+ return 0;
+ out_put_ci_module:
+ module_put(ci->owner);
+ out_put_this_module:
+ module_put(THIS_MODULE);
+ out_put_sdev:
+ soundbus_dev_put(dev);
+ out_free_cii:
+ kfree(cii);
+ return err;
+}
+
+void i2sbus_detach_codec(struct soundbus_dev *dev, void *data)
+{
+ struct codec_info_item *cii = NULL, *i;
+
+ list_for_each_entry(i, &dev->codec_list, list) {
+ if (i->codec_data == data) {
+ cii = i;
+ break;
+ }
+ }
+ if (cii) {
+ list_del(&cii->list);
+ module_put(cii->codec->owner);
+ kfree(cii);
+ }
+ /* no more codecs, but still a pcm? */
+ if (list_empty(&dev->codec_list) && dev->pcm) {
+ /* the actual cleanup is done by the callback above! */
+ snd_device_free(dev->pcm->card, dev->pcm);
+ }
+}
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus.h b/sound/aoa/soundbus/i2sbus/i2sbus.h
new file mode 100644
index 0000000..ff29654
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus.h
@@ -0,0 +1,126 @@
+/*
+ * i2sbus driver -- private definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __I2SBUS_H
+#define __I2SBUS_H
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+#include <linux/mutex.h>
+#include <linux/completion.h>
+
+#include <sound/pcm.h>
+
+#include <asm/prom.h>
+#include <asm/pmac_feature.h>
+#include <asm/dbdma.h>
+
+#include "i2sbus-interface.h"
+#include "../soundbus.h"
+
+struct i2sbus_control {
+ struct list_head list;
+ struct macio_chip *macio;
+};
+
+#define MAX_DBDMA_COMMANDS 32
+
+struct dbdma_command_mem {
+ dma_addr_t bus_addr;
+ dma_addr_t bus_cmd_start;
+ struct dbdma_cmd *cmds;
+ void *space;
+ int size;
+ u32 running:1;
+ u32 stopping:1;
+};
+
+struct pcm_info {
+ u32 created:1, /* has this direction been created with alsa? */
+ active:1; /* is this stream active? */
+ /* runtime information */
+ struct snd_pcm_substream *substream;
+ int current_period;
+ u32 frame_count;
+ struct dbdma_command_mem dbdma_ring;
+ volatile struct dbdma_regs __iomem *dbdma;
+ struct completion *stop_completion;
+};
+
+enum {
+ aoa_resource_i2smmio = 0,
+ aoa_resource_txdbdma,
+ aoa_resource_rxdbdma,
+};
+
+struct i2sbus_dev {
+ struct soundbus_dev sound;
+ struct macio_dev *macio;
+ struct i2sbus_control *control;
+ volatile struct i2s_interface_regs __iomem *intfregs;
+
+ struct resource resources[3];
+ struct resource *allocated_resource[3];
+ int interrupts[3];
+ char rnames[3][32];
+
+ /* info about currently active substreams */
+ struct pcm_info out, in;
+ snd_pcm_format_t format;
+ unsigned int rate;
+
+ /* list for a single controller */
+ struct list_head item;
+ /* number of bus on controller */
+ int bus_number;
+ /* for use by control layer */
+ struct pmf_function *enable,
+ *cell_enable,
+ *cell_disable,
+ *clock_enable,
+ *clock_disable;
+
+ /* locks */
+ /* spinlock for low-level interrupt locking */
+ spinlock_t low_lock;
+ /* mutex for high-level consistency */
+ struct mutex lock;
+};
+
+#define soundbus_dev_to_i2sbus_dev(sdev) \
+ container_of(sdev, struct i2sbus_dev, sound)
+
+/* pcm specific functions */
+extern int
+i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card,
+ struct codec_info *ci, void *data);
+extern void
+i2sbus_detach_codec(struct soundbus_dev *dev, void *data);
+extern irqreturn_t
+i2sbus_tx_intr(int irq, void *devid);
+extern irqreturn_t
+i2sbus_rx_intr(int irq, void *devid);
+
+extern void i2sbus_wait_for_stop_both(struct i2sbus_dev *i2sdev);
+extern void i2sbus_pcm_prepare_both(struct i2sbus_dev *i2sdev);
+
+/* control specific functions */
+extern int i2sbus_control_init(struct macio_dev* dev,
+ struct i2sbus_control **c);
+extern void i2sbus_control_destroy(struct i2sbus_control *c);
+extern int i2sbus_control_add_dev(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev);
+extern void i2sbus_control_remove_dev(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev);
+extern int i2sbus_control_enable(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev);
+extern int i2sbus_control_cell(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev,
+ int enable);
+extern int i2sbus_control_clock(struct i2sbus_control *c,
+ struct i2sbus_dev *i2sdev,
+ int enable);
+#endif /* __I2SBUS_H */
diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h
new file mode 100644
index 0000000..a0f223c
--- /dev/null
+++ b/sound/aoa/soundbus/soundbus.h
@@ -0,0 +1,204 @@
+/*
+ * soundbus generic definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SOUNDBUS_H
+#define __SOUNDBUS_H
+
+#include <linux/of_device.h>
+#include <sound/pcm.h>
+#include <linux/list.h>
+
+
+/* When switching from master to slave or the other way around,
+ * you don't want to have the codec chip acting as clock source
+ * while the bus still is.
+ * More importantly, while switch from slave to master, you need
+ * to turn off the chip's master function first, but then there's
+ * no clock for a while and other chips might reset, so we notify
+ * their drivers after having switched.
+ * The constants here are codec-point of view, so when we switch
+ * the soundbus to master we tell the codec we're going to switch
+ * and give it CLOCK_SWITCH_PREPARE_SLAVE!
+ */
+enum clock_switch {
+ CLOCK_SWITCH_PREPARE_SLAVE,
+ CLOCK_SWITCH_PREPARE_MASTER,
+ CLOCK_SWITCH_SLAVE,
+ CLOCK_SWITCH_MASTER,
+ CLOCK_SWITCH_NOTIFY,
+};
+
+/* information on a transfer the codec can take */
+struct transfer_info {
+ u64 formats; /* SNDRV_PCM_FMTBIT_* */
+ unsigned int rates; /* SNDRV_PCM_RATE_* */
+ /* flags */
+ u32 transfer_in:1, /* input = 1, output = 0 */
+ must_be_clock_source:1;
+ /* for codecs to distinguish among their TIs */
+ int tag;
+};
+
+struct codec_info_item {
+ struct codec_info *codec;
+ void *codec_data;
+ struct soundbus_dev *sdev;
+ /* internal, to be used by the soundbus provider */
+ struct list_head list;
+};
+
+/* for prepare, where the codecs need to know
+ * what we're going to drive the bus with */
+struct bus_info {
+ /* see below */
+ int sysclock_factor;
+ int bus_factor;
+};
+
+/* information on the codec itself, plus function pointers */
+struct codec_info {
+ /* the module this lives in */
+ struct module *owner;
+
+ /* supported transfer possibilities, array terminated by
+ * formats or rates being 0. */
+ struct transfer_info *transfers;
+
+ /* Master clock speed factor
+ * to be used (master clock speed = sysclock_factor * sampling freq)
+ * Unused if the soundbus provider has no such notion.
+ */
+ int sysclock_factor;
+
+ /* Bus factor, bus clock speed = bus_factor * sampling freq)
+ * Unused if the soundbus provider has no such notion.
+ */
+ int bus_factor;
+
+ /* operations */
+ /* clock switching, see above */
+ int (*switch_clock)(struct codec_info_item *cii,
+ enum clock_switch clock);
+
+ /* called for each transfer_info when the user
+ * opens the pcm device to determine what the
+ * hardware can support at this point in time.
+ * That can depend on other user-switchable controls.
+ * Return 1 if usable, 0 if not.
+ * out points to another instance of a transfer_info
+ * which is initialised to the values in *ti, and
+ * it's format and rate values can be modified by
+ * the callback if it is necessary to further restrict
+ * the formats that can be used at the moment, for
+ * example when one codec has multiple logical codec
+ * info structs for multiple inputs.
+ */
+ int (*usable)(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out);
+
+ /* called when pcm stream is opened, probably not implemented
+ * most of the time since it isn't too useful */
+ int (*open)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ /* called when the pcm stream is closed, at this point
+ * the user choices can all be unlocked (see below) */
+ int (*close)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ /* if the codec must forbid some user choices because
+ * they are not valid with the substream/transfer info,
+ * it must do so here. Example: no digital output for
+ * incompatible framerate, say 8KHz, on Onyx.
+ * If the selected stuff in the substream is NOT
+ * compatible, you have to reject this call! */
+ int (*prepare)(struct codec_info_item *cii,
+ struct bus_info *bi,
+ struct snd_pcm_substream *substream);
+
+ /* start() is called before data is pushed to the codec.
+ * Note that start() must be atomic! */
+ int (*start)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ /* stop() is called after data is no longer pushed to the codec.
+ * Note that stop() must be atomic! */
+ int (*stop)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ int (*suspend)(struct codec_info_item *cii, pm_message_t state);
+ int (*resume)(struct codec_info_item *cii);
+};
+
+/* information on a soundbus device */
+struct soundbus_dev {
+ /* the bus it belongs to */
+ struct list_head onbuslist;
+
+ /* the of device it represents */
+ struct of_device ofdev;
+
+ /* what modules go by */
+ char modalias[32];
+
+ /* These fields must be before attach_codec can be called.
+ * They should be set by the owner of the alsa card object
+ * that is needed, and whoever sets them must make sure
+ * that they are unique within that alsa card object. */
+ char *pcmname;
+ int pcmid;
+
+ /* this is assigned by the soundbus provider in attach_codec */
+ struct snd_pcm *pcm;
+
+ /* operations */
+ /* attach a codec to this soundbus, give the alsa
+ * card object the PCMs for this soundbus should be in.
+ * The 'data' pointer must be unique, it is used as the
+ * key for detach_codec(). */
+ int (*attach_codec)(struct soundbus_dev *dev, struct snd_card *card,
+ struct codec_info *ci, void *data);
+ void (*detach_codec)(struct soundbus_dev *dev, void *data);
+ /* TODO: suspend/resume */
+
+ /* private for the soundbus provider */
+ struct list_head codec_list;
+ u32 have_out:1, have_in:1;
+};
+#define to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev.dev)
+#define of_to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev)
+
+extern int soundbus_add_one(struct soundbus_dev *dev);
+extern void soundbus_remove_one(struct soundbus_dev *dev);
+
+extern struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev);
+extern void soundbus_dev_put(struct soundbus_dev *dev);
+
+struct soundbus_driver {
+ char *name;
+ struct module *owner;
+
+ /* we don't implement any matching at all */
+
+ int (*probe)(struct soundbus_dev* dev);
+ int (*remove)(struct soundbus_dev* dev);
+
+ int (*suspend)(struct soundbus_dev* dev, pm_message_t state);
+ int (*resume)(struct soundbus_dev* dev);
+ int (*shutdown)(struct soundbus_dev* dev);
+
+ struct device_driver driver;
+};
+#define to_soundbus_driver(drv) container_of(drv,struct soundbus_driver, driver)
+
+extern int soundbus_register_driver(struct soundbus_driver *drv);
+extern void soundbus_unregister_driver(struct soundbus_driver *drv);
+
+extern struct device_attribute soundbus_dev_attrs[];
+
+#endif /* __SOUNDBUS_H */
diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c
new file mode 100644
index 0000000..f580942
--- /dev/null
+++ b/sound/aoa/soundbus/sysfs.c
@@ -0,0 +1,42 @@
+#include <linux/kernel.h>
+#include <linux/stat.h>
+/* FIX UP */
+#include "soundbus.h"
+
+#define soundbus_config_of_attr(field, format_string) \
+static ssize_t \
+field##_show (struct device *dev, struct device_attribute *attr, \
+ char *buf) \
+{ \
+ struct soundbus_dev *mdev = to_soundbus_device (dev); \
+ return sprintf (buf, format_string, mdev->ofdev.node->field); \
+}
+
+static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
+ char *buf)
+{
+ struct soundbus_dev *sdev = to_soundbus_device(dev);
+ struct of_device *of = &sdev->ofdev;
+ int length;
+
+ if (*sdev->modalias) {
+ strlcpy(buf, sdev->modalias, sizeof(sdev->modalias) + 1);
+ strcat(buf, "\n");
+ length = strlen(buf);
+ } else {
+ length = sprintf(buf, "of:N%sT%s\n",
+ of->node->name, of->node->type);
+ }
+
+ return length;
+}
+
+soundbus_config_of_attr (name, "%s\n");
+soundbus_config_of_attr (type, "%s\n");
+
+struct device_attribute soundbus_dev_attrs[] = {
+ __ATTR_RO(name),
+ __ATTR_RO(type),
+ __ATTR_RO(modalias),
+ __ATTR_NULL
+};
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
new file mode 100644
index 0000000..f8e6de4
--- /dev/null
+++ b/sound/arm/Kconfig
@@ -0,0 +1,54 @@
+# ALSA ARM drivers
+
+menuconfig SND_ARM
+ bool "ARM sound devices"
+ depends on ARM
+ default y
+ help
+ Support for sound devices specific to ARM architectures.
+ Drivers that are implemented on ASoC can be found in
+ "ALSA for SoC audio support" section.
+
+if SND_ARM
+
+config SND_SA11XX_UDA1341
+ tristate "SA11xx UDA1341TS driver (iPaq H3600)"
+ depends on ARCH_SA1100 && L3
+ select SND_PCM
+ help
+ Say Y here if you have a Compaq iPaq H3x00 handheld computer
+ and want to use its Philips UDA 1341 audio chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sa11xx-uda1341.
+
+config SND_ARMAACI
+ tristate "ARM PrimeCell PL041 AC Link support"
+ depends on ARM_AMBA
+ select SND_PCM
+ select SND_AC97_CODEC
+
+config SND_PXA2XX_PCM
+ tristate
+ select SND_PCM
+
+config SND_PXA2XX_LIB
+ tristate
+ select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
+
+config SND_PXA2XX_LIB_AC97
+ bool
+
+config SND_PXA2XX_AC97
+ tristate "AC97 driver for the Intel PXA2xx chip"
+ depends on ARCH_PXA
+ select SND_PXA2XX_PCM
+ select SND_AC97_CODEC
+ select SND_PXA2XX_LIB
+ select SND_PXA2XX_LIB_AC97
+ help
+ Say Y or M if you want to support any AC97 codec attached to
+ the PXA2xx AC97 interface.
+
+endif # SND_ARM
+
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
new file mode 100644
index 0000000..2054de1
--- /dev/null
+++ b/sound/arm/Makefile
@@ -0,0 +1,19 @@
+#
+# Makefile for ALSA
+#
+
+obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o
+snd-sa11xx-uda1341-objs := sa11xx-uda1341.o
+
+obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
+snd-aaci-objs := aaci.o devdma.o
+
+obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o
+snd-pxa2xx-pcm-objs := pxa2xx-pcm.o
+
+obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o
+snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o
+snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o
+
+obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o
+snd-pxa2xx-ac97-objs := pxa2xx-ac97.o
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
new file mode 100644
index 0000000..89096e8
--- /dev/null
+++ b/sound/arm/aaci.c
@@ -0,0 +1,1206 @@
+/*
+ * linux/sound/arm/aaci.c - ARM PrimeCell AACI PL041 driver
+ *
+ * Copyright (C) 2003 Deep Blue Solutions Ltd, All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Documentation: ARM DDI 0173B
+ */
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/ioport.h>
+#include <linux/device.h>
+#include <linux/spinlock.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/amba/bus.h>
+
+#include <asm/io.h>
+#include <asm/irq.h>
+#include <asm/sizes.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "aaci.h"
+#include "devdma.h"
+
+#define DRIVER_NAME "aaci-pl041"
+
+/*
+ * PM support is not complete. Turn it off.
+ */
+#undef CONFIG_PM
+
+static void aaci_ac97_select_codec(struct aaci *aaci, struct snd_ac97 *ac97)
+{
+ u32 v, maincr = aaci->maincr | MAINCR_SCRA(ac97->num);
+
+ /*
+ * Ensure that the slot 1/2 RX registers are empty.
+ */
+ v = readl(aaci->base + AACI_SLFR);
+ if (v & SLFR_2RXV)
+ readl(aaci->base + AACI_SL2RX);
+ if (v & SLFR_1RXV)
+ readl(aaci->base + AACI_SL1RX);
+
+ writel(maincr, aaci->base + AACI_MAINCR);
+}
+
+/*
+ * P29:
+ * The recommended use of programming the external codec through slot 1
+ * and slot 2 data is to use the channels during setup routines and the
+ * slot register at any other time. The data written into slot 1, slot 2
+ * and slot 12 registers is transmitted only when their corresponding
+ * SI1TxEn, SI2TxEn and SI12TxEn bits are set in the AACI_MAINCR
+ * register.
+ */
+static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct aaci *aaci = ac97->private_data;
+ u32 v;
+ int timeout = 5000;
+
+ if (ac97->num >= 4)
+ return;
+
+ mutex_lock(&aaci->ac97_sem);
+
+ aaci_ac97_select_codec(aaci, ac97);
+
+ /*
+ * P54: You must ensure that AACI_SL2TX is always written
+ * to, if required, before data is written to AACI_SL1TX.
+ */
+ writel(val << 4, aaci->base + AACI_SL2TX);
+ writel(reg << 12, aaci->base + AACI_SL1TX);
+
+ /*
+ * Wait for the transmission of both slots to complete.
+ */
+ do {
+ v = readl(aaci->base + AACI_SLFR);
+ } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--);
+
+ if (!timeout)
+ dev_err(&aaci->dev->dev,
+ "timeout waiting for write to complete\n");
+
+ mutex_unlock(&aaci->ac97_sem);
+}
+
+/*
+ * Read an AC'97 register.
+ */
+static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct aaci *aaci = ac97->private_data;
+ u32 v;
+ int timeout = 5000;
+ int retries = 10;
+
+ if (ac97->num >= 4)
+ return ~0;
+
+ mutex_lock(&aaci->ac97_sem);
+
+ aaci_ac97_select_codec(aaci, ac97);
+
+ /*
+ * Write the register address to slot 1.
+ */
+ writel((reg << 12) | (1 << 19), aaci->base + AACI_SL1TX);
+
+ /*
+ * Wait for the transmission to complete.
+ */
+ do {
+ v = readl(aaci->base + AACI_SLFR);
+ } while ((v & SLFR_1TXB) && timeout--);
+
+ if (!timeout) {
+ dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
+ v = ~0;
+ goto out;
+ }
+
+ /*
+ * Give the AC'97 codec more than enough time
+ * to respond. (42us = ~2 frames at 48kHz.)
+ */
+ udelay(42);
+
+ /*
+ * Wait for slot 2 to indicate data.
+ */
+ timeout = 5000;
+ do {
+ cond_resched();
+ v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
+ } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--);
+
+ if (!timeout) {
+ dev_err(&aaci->dev->dev, "timeout on RX valid\n");
+ v = ~0;
+ goto out;
+ }
+
+ do {
+ v = readl(aaci->base + AACI_SL1RX) >> 12;
+ if (v == reg) {
+ v = readl(aaci->base + AACI_SL2RX) >> 4;
+ break;
+ } else if (--retries) {
+ dev_warn(&aaci->dev->dev,
+ "ac97 read back fail. retry\n");
+ continue;
+ } else {
+ dev_warn(&aaci->dev->dev,
+ "wrong ac97 register read back (%x != %x)\n",
+ v, reg);
+ v = ~0;
+ }
+ } while (retries);
+ out:
+ mutex_unlock(&aaci->ac97_sem);
+ return v;
+}
+
+static inline void aaci_chan_wait_ready(struct aaci_runtime *aacirun)
+{
+ u32 val;
+ int timeout = 5000;
+
+ do {
+ val = readl(aacirun->base + AACI_SR);
+ } while (val & (SR_TXB|SR_RXB) && timeout--);
+}
+
+
+
+/*
+ * Interrupt support.
+ */
+static void aaci_fifo_irq(struct aaci *aaci, int channel, u32 mask)
+{
+ if (mask & ISR_ORINTR) {
+ dev_warn(&aaci->dev->dev, "RX overrun on chan %d\n", channel);
+ writel(ICLR_RXOEC1 << channel, aaci->base + AACI_INTCLR);
+ }
+
+ if (mask & ISR_RXTOINTR) {
+ dev_warn(&aaci->dev->dev, "RX timeout on chan %d\n", channel);
+ writel(ICLR_RXTOFEC1 << channel, aaci->base + AACI_INTCLR);
+ }
+
+ if (mask & ISR_RXINTR) {
+ struct aaci_runtime *aacirun = &aaci->capture;
+ void *ptr;
+
+ if (!aacirun->substream || !aacirun->start) {
+ dev_warn(&aaci->dev->dev, "RX interrupt???\n");
+ writel(0, aacirun->base + AACI_IE);
+ return;
+ }
+ ptr = aacirun->ptr;
+
+ do {
+ unsigned int len = aacirun->fifosz;
+ u32 val;
+
+ if (aacirun->bytes <= 0) {
+ aacirun->bytes += aacirun->period;
+ aacirun->ptr = ptr;
+ spin_unlock(&aaci->lock);
+ snd_pcm_period_elapsed(aacirun->substream);
+ spin_lock(&aaci->lock);
+ }
+ if (!(aacirun->cr & CR_EN))
+ break;
+
+ val = readl(aacirun->base + AACI_SR);
+ if (!(val & SR_RXHF))
+ break;
+ if (!(val & SR_RXFF))
+ len >>= 1;
+
+ aacirun->bytes -= len;
+
+ /* reading 16 bytes at a time */
+ for( ; len > 0; len -= 16) {
+ asm(
+ "ldmia %1, {r0, r1, r2, r3}\n\t"
+ "stmia %0!, {r0, r1, r2, r3}"
+ : "+r" (ptr)
+ : "r" (aacirun->fifo)
+ : "r0", "r1", "r2", "r3", "cc");
+
+ if (ptr >= aacirun->end)
+ ptr = aacirun->start;
+ }
+ } while(1);
+ aacirun->ptr = ptr;
+ }
+
+ if (mask & ISR_URINTR) {
+ dev_dbg(&aaci->dev->dev, "TX underrun on chan %d\n", channel);
+ writel(ICLR_TXUEC1 << channel, aaci->base + AACI_INTCLR);
+ }
+
+ if (mask & ISR_TXINTR) {
+ struct aaci_runtime *aacirun = &aaci->playback;
+ void *ptr;
+
+ if (!aacirun->substream || !aacirun->start) {
+ dev_warn(&aaci->dev->dev, "TX interrupt???\n");
+ writel(0, aacirun->base + AACI_IE);
+ return;
+ }
+
+ ptr = aacirun->ptr;
+ do {
+ unsigned int len = aacirun->fifosz;
+ u32 val;
+
+ if (aacirun->bytes <= 0) {
+ aacirun->bytes += aacirun->period;
+ aacirun->ptr = ptr;
+ spin_unlock(&aaci->lock);
+ snd_pcm_period_elapsed(aacirun->substream);
+ spin_lock(&aaci->lock);
+ }
+ if (!(aacirun->cr & CR_EN))
+ break;
+
+ val = readl(aacirun->base + AACI_SR);
+ if (!(val & SR_TXHE))
+ break;
+ if (!(val & SR_TXFE))
+ len >>= 1;
+
+ aacirun->bytes -= len;
+
+ /* writing 16 bytes at a time */
+ for ( ; len > 0; len -= 16) {
+ asm(
+ "ldmia %0!, {r0, r1, r2, r3}\n\t"
+ "stmia %1, {r0, r1, r2, r3}"
+ : "+r" (ptr)
+ : "r" (aacirun->fifo)
+ : "r0", "r1", "r2", "r3", "cc");
+
+ if (ptr >= aacirun->end)
+ ptr = aacirun->start;
+ }
+ } while (1);
+
+ aacirun->ptr = ptr;
+ }
+}
+
+static irqreturn_t aaci_irq(int irq, void *devid)
+{
+ struct aaci *aaci = devid;
+ u32 mask;
+ int i;
+
+ spin_lock(&aaci->lock);
+ mask = readl(aaci->base + AACI_ALLINTS);
+ if (mask) {
+ u32 m = mask;
+ for (i = 0; i < 4; i++, m >>= 7) {
+ if (m & 0x7f) {
+ aaci_fifo_irq(aaci, i, m);
+ }
+ }
+ }
+ spin_unlock(&aaci->lock);
+
+ return mask ? IRQ_HANDLED : IRQ_NONE;
+}
+
+
+
+/*
+ * ALSA support.
+ */
+
+struct aaci_stream {
+ unsigned char codec_idx;
+ unsigned char rate_idx;
+};
+
+static struct aaci_stream aaci_streams[] = {
+ [ACSTREAM_FRONT] = {
+ .codec_idx = 0,
+ .rate_idx = AC97_RATES_FRONT_DAC,
+ },
+ [ACSTREAM_SURROUND] = {
+ .codec_idx = 0,
+ .rate_idx = AC97_RATES_SURR_DAC,
+ },
+ [ACSTREAM_LFE] = {
+ .codec_idx = 0,
+ .rate_idx = AC97_RATES_LFE_DAC,
+ },
+};
+
+static inline unsigned int aaci_rate_mask(struct aaci *aaci, int streamid)
+{
+ struct aaci_stream *s = aaci_streams + streamid;
+ return aaci->ac97_bus->codec[s->codec_idx]->rates[s->rate_idx];
+}
+
+static unsigned int rate_list[] = {
+ 5512, 8000, 11025, 16000, 22050, 32000, 44100,
+ 48000, 64000, 88200, 96000, 176400, 192000
+};
+
+/*
+ * Double-rate rule: we can support double rate iff channels == 2
+ * (unimplemented)
+ */
+static int
+aaci_rule_rate_by_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule)
+{
+ struct aaci *aaci = rule->private;
+ unsigned int rate_mask = SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_5512;
+ struct snd_interval *c = hw_param_interval(p, SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ switch (c->max) {
+ case 6:
+ rate_mask &= aaci_rate_mask(aaci, ACSTREAM_LFE);
+ case 4:
+ rate_mask &= aaci_rate_mask(aaci, ACSTREAM_SURROUND);
+ case 2:
+ rate_mask &= aaci_rate_mask(aaci, ACSTREAM_FRONT);
+ }
+
+ return snd_interval_list(hw_param_interval(p, rule->var),
+ ARRAY_SIZE(rate_list), rate_list,
+ rate_mask);
+}
+
+static struct snd_pcm_hardware aaci_hw_info = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME,
+
+ /*
+ * ALSA doesn't support 18-bit or 20-bit packed into 32-bit
+ * words. It also doesn't support 12-bit at all.
+ */
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+
+ /* should this be continuous or knot? */
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_max = 48000,
+ .rate_min = 4000,
+ .channels_min = 2,
+ .channels_max = 6,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 256,
+ .period_bytes_max = PAGE_SIZE,
+ .periods_min = 4,
+ .periods_max = PAGE_SIZE / 16,
+};
+
+static int __aaci_pcm_open(struct aaci *aaci,
+ struct snd_pcm_substream *substream,
+ struct aaci_runtime *aacirun)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+
+ aacirun->substream = substream;
+ runtime->private_data = aacirun;
+ runtime->hw = aaci_hw_info;
+
+ /*
+ * FIXME: ALSA specifies fifo_size in bytes. If we're in normal
+ * mode, each 32-bit word contains one sample. If we're in
+ * compact mode, each 32-bit word contains two samples, effectively
+ * halving the FIFO size. However, we don't know for sure which
+ * we'll be using at this point. We set this to the lower limit.
+ */
+ runtime->hw.fifo_size = aaci->fifosize * 2;
+
+ /*
+ * Add rule describing hardware rate dependency
+ * on the number of channels.
+ */
+ ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ aaci_rule_rate_by_channels, aaci,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (ret)
+ goto out;
+
+ ret = request_irq(aaci->dev->irq[0], aaci_irq, IRQF_SHARED|IRQF_DISABLED,
+ DRIVER_NAME, aaci);
+ if (ret)
+ goto out;
+
+ return 0;
+
+ out:
+ return ret;
+}
+
+
+/*
+ * Common ALSA stuff
+ */
+static int aaci_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+
+ WARN_ON(aacirun->cr & CR_EN);
+
+ aacirun->substream = NULL;
+ free_irq(aaci->dev->irq[0], aaci);
+
+ return 0;
+}
+
+static int aaci_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+
+ /*
+ * This must not be called with the device enabled.
+ */
+ WARN_ON(aacirun->cr & CR_EN);
+
+ if (aacirun->pcm_open)
+ snd_ac97_pcm_close(aacirun->pcm);
+ aacirun->pcm_open = 0;
+
+ /*
+ * Clear out the DMA and any allocated buffers.
+ */
+ devdma_hw_free(NULL, substream);
+
+ return 0;
+}
+
+static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct aaci_runtime *aacirun,
+ struct snd_pcm_hw_params *params)
+{
+ int err;
+
+ aaci_pcm_hw_free(substream);
+
+ err = devdma_hw_alloc(NULL, substream,
+ params_buffer_bytes(params));
+ if (err < 0)
+ goto out;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
+ params_channels(params),
+ aacirun->pcm->r[0].slots);
+ else
+ err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
+ params_channels(params),
+ aacirun->pcm->r[1].slots);
+
+ if (err)
+ goto out;
+
+ aacirun->pcm_open = 1;
+
+ out:
+ return err;
+}
+
+static int aaci_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct aaci_runtime *aacirun = runtime->private_data;
+
+ aacirun->start = (void *)runtime->dma_area;
+ aacirun->end = aacirun->start + runtime->dma_bytes;
+ aacirun->ptr = aacirun->start;
+ aacirun->period =
+ aacirun->bytes = frames_to_bytes(runtime, runtime->period_size);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct aaci_runtime *aacirun = runtime->private_data;
+ ssize_t bytes = aacirun->ptr - aacirun->start;
+
+ return bytes_to_frames(runtime, bytes);
+}
+
+static int aaci_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma)
+{
+ return devdma_mmap(NULL, substream, vma);
+}
+
+
+/*
+ * Playback specific ALSA stuff
+ */
+static const u32 channels_to_txmask[] = {
+ [2] = CR_SL3 | CR_SL4,
+ [4] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8,
+ [6] = CR_SL3 | CR_SL4 | CR_SL7 | CR_SL8 | CR_SL6 | CR_SL9,
+};
+
+/*
+ * We can support two and four channel audio. Unfortunately
+ * six channel audio requires a non-standard channel ordering:
+ * 2 -> FL(3), FR(4)
+ * 4 -> FL(3), FR(4), SL(7), SR(8)
+ * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required)
+ * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual)
+ * This requires an ALSA configuration file to correct.
+ */
+static unsigned int channel_list[] = { 2, 4, 6 };
+
+static int
+aaci_rule_channels(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *rule)
+{
+ struct aaci *aaci = rule->private;
+ unsigned int chan_mask = 1 << 0, slots;
+
+ /*
+ * pcms[0] is the our 5.1 PCM instance.
+ */
+ slots = aaci->ac97_bus->pcms[0].r[0].slots;
+ if (slots & (1 << AC97_SLOT_PCM_SLEFT)) {
+ chan_mask |= 1 << 1;
+ if (slots & (1 << AC97_SLOT_LFE))
+ chan_mask |= 1 << 2;
+ }
+
+ return snd_interval_list(hw_param_interval(p, rule->var),
+ ARRAY_SIZE(channel_list), channel_list,
+ chan_mask);
+}
+
+static int aaci_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct aaci *aaci = substream->private_data;
+ int ret;
+
+ /*
+ * Add rule describing channel dependency.
+ */
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ aaci_rule_channels, aaci,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (ret)
+ return ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = __aaci_pcm_open(aaci, substream, &aaci->playback);
+ } else {
+ ret = __aaci_pcm_open(aaci, substream, &aaci->capture);
+ }
+ return ret;
+}
+
+static int aaci_pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+ unsigned int channels = params_channels(params);
+ int ret;
+
+ WARN_ON(channels >= ARRAY_SIZE(channels_to_txmask) ||
+ !channels_to_txmask[channels]);
+
+ ret = aaci_pcm_hw_params(substream, aacirun, params);
+
+ /*
+ * Enable FIFO, compact mode, 16 bits per sample.
+ * FIXME: double rate slots?
+ */
+ if (ret >= 0) {
+ aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+ aacirun->cr |= channels_to_txmask[channels];
+
+ aacirun->fifosz = aaci->fifosize * 4;
+ if (aacirun->cr & CR_COMPACT)
+ aacirun->fifosz >>= 1;
+ }
+ return ret;
+}
+
+static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun)
+{
+ u32 ie;
+
+ ie = readl(aacirun->base + AACI_IE);
+ ie &= ~(IE_URIE|IE_TXIE);
+ writel(ie, aacirun->base + AACI_IE);
+ aacirun->cr &= ~CR_EN;
+ aaci_chan_wait_ready(aacirun);
+ writel(aacirun->cr, aacirun->base + AACI_TXCR);
+}
+
+static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
+{
+ u32 ie;
+
+ aaci_chan_wait_ready(aacirun);
+ aacirun->cr |= CR_EN;
+
+ ie = readl(aacirun->base + AACI_IE);
+ ie |= IE_URIE | IE_TXIE;
+ writel(ie, aacirun->base + AACI_IE);
+ writel(aacirun->cr, aacirun->base + AACI_TXCR);
+}
+
+static int aaci_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+ unsigned long flags;
+ int ret = 0;
+
+ spin_lock_irqsave(&aaci->lock, flags);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ aaci_pcm_playback_start(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ aaci_pcm_playback_start(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ aaci_pcm_playback_stop(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ aaci_pcm_playback_stop(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+ spin_unlock_irqrestore(&aaci->lock, flags);
+
+ return ret;
+}
+
+static struct snd_pcm_ops aaci_playback_ops = {
+ .open = aaci_pcm_open,
+ .close = aaci_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = aaci_pcm_playback_hw_params,
+ .hw_free = aaci_pcm_hw_free,
+ .prepare = aaci_pcm_prepare,
+ .trigger = aaci_pcm_playback_trigger,
+ .pointer = aaci_pcm_pointer,
+ .mmap = aaci_pcm_mmap,
+};
+
+static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+ int ret;
+
+ ret = aaci_pcm_hw_params(substream, aacirun, params);
+
+ if (ret >= 0) {
+ aacirun->cr = CR_FEN | CR_COMPACT | CR_SZ16;
+
+ /* Line in record: slot 3 and 4 */
+ aacirun->cr |= CR_SL3 | CR_SL4;
+
+ aacirun->fifosz = aaci->fifosize * 4;
+
+ if (aacirun->cr & CR_COMPACT)
+ aacirun->fifosz >>= 1;
+ }
+ return ret;
+}
+
+static void aaci_pcm_capture_stop(struct aaci_runtime *aacirun)
+{
+ u32 ie;
+
+ aaci_chan_wait_ready(aacirun);
+
+ ie = readl(aacirun->base + AACI_IE);
+ ie &= ~(IE_ORIE | IE_RXIE);
+ writel(ie, aacirun->base+AACI_IE);
+
+ aacirun->cr &= ~CR_EN;
+
+ writel(aacirun->cr, aacirun->base + AACI_RXCR);
+}
+
+static void aaci_pcm_capture_start(struct aaci_runtime *aacirun)
+{
+ u32 ie;
+
+ aaci_chan_wait_ready(aacirun);
+
+#ifdef DEBUG
+ /* RX Timeout value: bits 28:17 in RXCR */
+ aacirun->cr |= 0xf << 17;
+#endif
+
+ aacirun->cr |= CR_EN;
+ writel(aacirun->cr, aacirun->base + AACI_RXCR);
+
+ ie = readl(aacirun->base + AACI_IE);
+ ie |= IE_ORIE |IE_RXIE; // overrun and rx interrupt -- half full
+ writel(ie, aacirun->base + AACI_IE);
+}
+
+static int aaci_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+ unsigned long flags;
+ int ret = 0;
+
+ spin_lock_irqsave(&aaci->lock, flags);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ aaci_pcm_capture_start(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ aaci_pcm_capture_start(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ aaci_pcm_capture_stop(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ aaci_pcm_capture_stop(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ spin_unlock_irqrestore(&aaci->lock, flags);
+
+ return ret;
+}
+
+static int aaci_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct aaci *aaci = substream->private_data;
+
+ aaci_pcm_prepare(substream);
+
+ /* allow changing of sample rate */
+ aaci_ac97_write(aaci->ac97, AC97_EXTENDED_STATUS, 0x0001); /* VRA */
+ aaci_ac97_write(aaci->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate);
+ aaci_ac97_write(aaci->ac97, AC97_PCM_MIC_ADC_RATE, runtime->rate);
+
+ /* Record select: Mic: 0, Aux: 3, Line: 4 */
+ aaci_ac97_write(aaci->ac97, AC97_REC_SEL, 0x0404);
+
+ return 0;
+}
+
+static struct snd_pcm_ops aaci_capture_ops = {
+ .open = aaci_pcm_open,
+ .close = aaci_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = aaci_pcm_capture_hw_params,
+ .hw_free = aaci_pcm_hw_free,
+ .prepare = aaci_pcm_capture_prepare,
+ .trigger = aaci_pcm_capture_trigger,
+ .pointer = aaci_pcm_pointer,
+ .mmap = aaci_pcm_mmap,
+};
+
+/*
+ * Power Management.
+ */
+#ifdef CONFIG_PM
+static int aaci_do_suspend(struct snd_card *card, unsigned int state)
+{
+ struct aaci *aaci = card->private_data;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3cold);
+ snd_pcm_suspend_all(aaci->pcm);
+ return 0;
+}
+
+static int aaci_do_resume(struct snd_card *card, unsigned int state)
+{
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+
+static int aaci_suspend(struct amba_device *dev, pm_message_t state)
+{
+ struct snd_card *card = amba_get_drvdata(dev);
+ return card ? aaci_do_suspend(card) : 0;
+}
+
+static int aaci_resume(struct amba_device *dev)
+{
+ struct snd_card *card = amba_get_drvdata(dev);
+ return card ? aaci_do_resume(card) : 0;
+}
+#else
+#define aaci_do_suspend NULL
+#define aaci_do_resume NULL
+#define aaci_suspend NULL
+#define aaci_resume NULL
+#endif
+
+
+static struct ac97_pcm ac97_defs[] __devinitdata = {
+ [0] = { /* Front PCM */
+ .exclusive = 1,
+ .r = {
+ [0] = {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT) |
+ (1 << AC97_SLOT_PCM_CENTER) |
+ (1 << AC97_SLOT_PCM_SLEFT) |
+ (1 << AC97_SLOT_PCM_SRIGHT) |
+ (1 << AC97_SLOT_LFE),
+ },
+ },
+ },
+ [1] = { /* PCM in */
+ .stream = 1,
+ .exclusive = 1,
+ .r = {
+ [0] = {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT),
+ },
+ },
+ },
+ [2] = { /* Mic in */
+ .stream = 1,
+ .exclusive = 1,
+ .r = {
+ [0] = {
+ .slots = (1 << AC97_SLOT_MIC),
+ },
+ },
+ }
+};
+
+static struct snd_ac97_bus_ops aaci_bus_ops = {
+ .write = aaci_ac97_write,
+ .read = aaci_ac97_read,
+};
+
+static int __devinit aaci_probe_ac97(struct aaci *aaci)
+{
+ struct snd_ac97_template ac97_template;
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97;
+ int ret;
+
+ /*
+ * Assert AACIRESET for 2us
+ */
+ writel(0, aaci->base + AACI_RESET);
+ udelay(2);
+ writel(RESET_NRST, aaci->base + AACI_RESET);
+
+ /*
+ * Give the AC'97 codec more than enough time
+ * to wake up. (42us = ~2 frames at 48kHz.)
+ */
+ udelay(42);
+
+ ret = snd_ac97_bus(aaci->card, 0, &aaci_bus_ops, aaci, &ac97_bus);
+ if (ret)
+ goto out;
+
+ ac97_bus->clock = 48000;
+ aaci->ac97_bus = ac97_bus;
+
+ memset(&ac97_template, 0, sizeof(struct snd_ac97_template));
+ ac97_template.private_data = aaci;
+ ac97_template.num = 0;
+ ac97_template.scaps = AC97_SCAP_SKIP_MODEM;
+
+ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97);
+ if (ret)
+ goto out;
+ aaci->ac97 = ac97;
+
+ /*
+ * Disable AC97 PC Beep input on audio codecs.
+ */
+ if (ac97_is_audio(ac97))
+ snd_ac97_write_cache(ac97, AC97_PC_BEEP, 0x801e);
+
+ ret = snd_ac97_pcm_assign(ac97_bus, ARRAY_SIZE(ac97_defs), ac97_defs);
+ if (ret)
+ goto out;
+
+ aaci->playback.pcm = &ac97_bus->pcms[0];
+ aaci->capture.pcm = &ac97_bus->pcms[1];
+
+ out:
+ return ret;
+}
+
+static void aaci_free_card(struct snd_card *card)
+{
+ struct aaci *aaci = card->private_data;
+ if (aaci->base)
+ iounmap(aaci->base);
+}
+
+static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
+{
+ struct aaci *aaci;
+ struct snd_card *card;
+
+ card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+ THIS_MODULE, sizeof(struct aaci));
+ if (card == NULL)
+ return NULL;
+
+ card->private_free = aaci_free_card;
+
+ strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver));
+ strlcpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname));
+ snprintf(card->longname, sizeof(card->longname),
+ "%s at 0x%016llx, irq %d",
+ card->shortname, (unsigned long long)dev->res.start,
+ dev->irq[0]);
+
+ aaci = card->private_data;
+ mutex_init(&aaci->ac97_sem);
+ spin_lock_init(&aaci->lock);
+ aaci->card = card;
+ aaci->dev = dev;
+
+ /* Set MAINCR to allow slot 1 and 2 data IO */
+ aaci->maincr = MAINCR_IE | MAINCR_SL1RXEN | MAINCR_SL1TXEN |
+ MAINCR_SL2RXEN | MAINCR_SL2TXEN;
+
+ return aaci;
+}
+
+static int __devinit aaci_init_pcm(struct aaci *aaci)
+{
+ struct snd_pcm *pcm;
+ int ret;
+
+ ret = snd_pcm_new(aaci->card, "AACI AC'97", 0, 1, 1, &pcm);
+ if (ret == 0) {
+ aaci->pcm = pcm;
+ pcm->private_data = aaci;
+ pcm->info_flags = 0;
+
+ strlcpy(pcm->name, DRIVER_NAME, sizeof(pcm->name));
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops);
+ }
+
+ return ret;
+}
+
+static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
+{
+ struct aaci_runtime *aacirun = &aaci->playback;
+ int i;
+
+ writel(CR_FEN | CR_SZ16 | CR_EN, aacirun->base + AACI_TXCR);
+
+ for (i = 0; !(readl(aacirun->base + AACI_SR) & SR_TXFF) && i < 4096; i++)
+ writel(0, aacirun->fifo);
+
+ writel(0, aacirun->base + AACI_TXCR);
+
+ /*
+ * Re-initialise the AACI after the FIFO depth test, to
+ * ensure that the FIFOs are empty. Unfortunately, merely
+ * disabling the channel doesn't clear the FIFO.
+ */
+ writel(aaci->maincr & ~MAINCR_IE, aaci->base + AACI_MAINCR);
+ writel(aaci->maincr, aaci->base + AACI_MAINCR);
+
+ /*
+ * If we hit 4096, we failed. Go back to the specified
+ * fifo depth.
+ */
+ if (i == 4096)
+ i = 8;
+
+ return i;
+}
+
+static int __devinit aaci_probe(struct amba_device *dev, void *id)
+{
+ struct aaci *aaci;
+ int ret, i;
+
+ ret = amba_request_regions(dev, NULL);
+ if (ret)
+ return ret;
+
+ aaci = aaci_init_card(dev);
+ if (!aaci) {
+ ret = -ENOMEM;
+ goto out;
+ }
+
+ aaci->base = ioremap(dev->res.start, SZ_4K);
+ if (!aaci->base) {
+ ret = -ENOMEM;
+ goto out;
+ }
+
+ /*
+ * Playback uses AACI channel 0
+ */
+ aaci->playback.base = aaci->base + AACI_CSCH1;
+ aaci->playback.fifo = aaci->base + AACI_DR1;
+
+ /*
+ * Capture uses AACI channel 0
+ */
+ aaci->capture.base = aaci->base + AACI_CSCH1;
+ aaci->capture.fifo = aaci->base + AACI_DR1;
+
+ for (i = 0; i < 4; i++) {
+ void __iomem *base = aaci->base + i * 0x14;
+
+ writel(0, base + AACI_IE);
+ writel(0, base + AACI_TXCR);
+ writel(0, base + AACI_RXCR);
+ }
+
+ writel(0x1fff, aaci->base + AACI_INTCLR);
+ writel(aaci->maincr, aaci->base + AACI_MAINCR);
+
+ ret = aaci_probe_ac97(aaci);
+ if (ret)
+ goto out;
+
+ /*
+ * Size the FIFOs (must be multiple of 16).
+ */
+ aaci->fifosize = aaci_size_fifo(aaci);
+ if (aaci->fifosize & 15) {
+ printk(KERN_WARNING "AACI: fifosize = %d not supported\n",
+ aaci->fifosize);
+ ret = -ENODEV;
+ goto out;
+ }
+
+ ret = aaci_init_pcm(aaci);
+ if (ret)
+ goto out;
+
+ snd_card_set_dev(aaci->card, &dev->dev);
+
+ ret = snd_card_register(aaci->card);
+ if (ret == 0) {
+ dev_info(&dev->dev, "%s, fifo %d\n", aaci->card->longname,
+ aaci->fifosize);
+ amba_set_drvdata(dev, aaci->card);
+ return ret;
+ }
+
+ out:
+ if (aaci)
+ snd_card_free(aaci->card);
+ amba_release_regions(dev);
+ return ret;
+}
+
+static int __devexit aaci_remove(struct amba_device *dev)
+{
+ struct snd_card *card = amba_get_drvdata(dev);
+
+ amba_set_drvdata(dev, NULL);
+
+ if (card) {
+ struct aaci *aaci = card->private_data;
+ writel(0, aaci->base + AACI_MAINCR);
+
+ snd_card_free(card);
+ amba_release_regions(dev);
+ }
+
+ return 0;
+}
+
+static struct amba_id aaci_ids[] = {
+ {
+ .id = 0x00041041,
+ .mask = 0x000fffff,
+ },
+ { 0, 0 },
+};
+
+static struct amba_driver aaci_driver = {
+ .drv = {
+ .name = DRIVER_NAME,
+ },
+ .probe = aaci_probe,
+ .remove = __devexit_p(aaci_remove),
+ .suspend = aaci_suspend,
+ .resume = aaci_resume,
+ .id_table = aaci_ids,
+};
+
+static int __init aaci_init(void)
+{
+ return amba_driver_register(&aaci_driver);
+}
+
+static void __exit aaci_exit(void)
+{
+ amba_driver_unregister(&aaci_driver);
+}
+
+module_init(aaci_init);
+module_exit(aaci_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ARM PrimeCell PL041 Advanced Audio CODEC Interface driver");
diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h
new file mode 100644
index 0000000..924f69c
--- /dev/null
+++ b/sound/arm/aaci.h
@@ -0,0 +1,247 @@
+/*
+ * linux/sound/arm/aaci.c - ARM PrimeCell AACI PL041 driver
+ *
+ * Copyright (C) 2003 Deep Blue Solutions, Ltd, All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#ifndef AACI_H
+#define AACI_H
+
+/*
+ * Control and status register offsets
+ * P39.
+ */
+#define AACI_CSCH1 0x000
+#define AACI_CSCH2 0x014
+#define AACI_CSCH3 0x028
+#define AACI_CSCH4 0x03c
+
+#define AACI_RXCR 0x000 /* 29 bits Control Rx FIFO */
+#define AACI_TXCR 0x004 /* 17 bits Control Tx FIFO */
+#define AACI_SR 0x008 /* 12 bits Status */
+#define AACI_ISR 0x00c /* 7 bits Int Status */
+#define AACI_IE 0x010 /* 7 bits Int Enable */
+
+/*
+ * Other registers
+ */
+#define AACI_SL1RX 0x050
+#define AACI_SL1TX 0x054
+#define AACI_SL2RX 0x058
+#define AACI_SL2TX 0x05c
+#define AACI_SL12RX 0x060
+#define AACI_SL12TX 0x064
+#define AACI_SLFR 0x068 /* slot flags */
+#define AACI_SLISTAT 0x06c /* slot interrupt status */
+#define AACI_SLIEN 0x070 /* slot interrupt enable */
+#define AACI_INTCLR 0x074 /* interrupt clear */
+#define AACI_MAINCR 0x078 /* main control */
+#define AACI_RESET 0x07c /* reset control */
+#define AACI_SYNC 0x080 /* sync control */
+#define AACI_ALLINTS 0x084 /* all fifo interrupt status */
+#define AACI_MAINFR 0x088 /* main flag register */
+#define AACI_DR1 0x090 /* data read/written fifo 1 */
+#define AACI_DR2 0x0b0 /* data read/written fifo 2 */
+#define AACI_DR3 0x0d0 /* data read/written fifo 3 */
+#define AACI_DR4 0x0f0 /* data read/written fifo 4 */
+
+/*
+ * TX/RX fifo control register (CR). P48
+ */
+#define CR_FEN (1 << 16) /* fifo enable */
+#define CR_COMPACT (1 << 15) /* compact mode */
+#define CR_SZ16 (0 << 13) /* 16 bits */
+#define CR_SZ18 (1 << 13) /* 18 bits */
+#define CR_SZ20 (2 << 13) /* 20 bits */
+#define CR_SZ12 (3 << 13) /* 12 bits */
+#define CR_SL12 (1 << 12)
+#define CR_SL11 (1 << 11)
+#define CR_SL10 (1 << 10)
+#define CR_SL9 (1 << 9)
+#define CR_SL8 (1 << 8)
+#define CR_SL7 (1 << 7)
+#define CR_SL6 (1 << 6)
+#define CR_SL5 (1 << 5)
+#define CR_SL4 (1 << 4)
+#define CR_SL3 (1 << 3)
+#define CR_SL2 (1 << 2)
+#define CR_SL1 (1 << 1)
+#define CR_EN (1 << 0) /* transmit enable */
+
+/*
+ * status register bits. P49
+ */
+#define SR_RXTOFE (1 << 11) /* rx timeout fifo empty */
+#define SR_TXTO (1 << 10) /* rx timeout fifo nonempty */
+#define SR_TXU (1 << 9) /* tx underrun */
+#define SR_RXO (1 << 8) /* rx overrun */
+#define SR_TXB (1 << 7) /* tx busy */
+#define SR_RXB (1 << 6) /* rx busy */
+#define SR_TXFF (1 << 5) /* tx fifo full */
+#define SR_RXFF (1 << 4) /* rx fifo full */
+#define SR_TXHE (1 << 3) /* tx fifo half empty */
+#define SR_RXHF (1 << 2) /* rx fifo half full */
+#define SR_TXFE (1 << 1) /* tx fifo empty */
+#define SR_RXFE (1 << 0) /* rx fifo empty */
+
+/*
+ * interrupt status register bits.
+ */
+#define ISR_RXTOFEINTR (1 << 6) /* rx fifo empty */
+#define ISR_URINTR (1 << 5) /* tx underflow */
+#define ISR_ORINTR (1 << 4) /* rx overflow */
+#define ISR_RXINTR (1 << 3) /* rx fifo */
+#define ISR_TXINTR (1 << 2) /* tx fifo intr */
+#define ISR_RXTOINTR (1 << 1) /* tx timeout */
+#define ISR_TXCINTR (1 << 0) /* tx complete */
+
+/*
+ * interrupt enable register bits.
+ */
+#define IE_RXTOIE (1 << 6)
+#define IE_URIE (1 << 5)
+#define IE_ORIE (1 << 4)
+#define IE_RXIE (1 << 3)
+#define IE_TXIE (1 << 2)
+#define IE_RXTIE (1 << 1)
+#define IE_TXCIE (1 << 0)
+
+/*
+ * interrupt status. P51
+ */
+#define ISR_RXTOFE (1 << 6) /* rx timeout fifo empty */
+#define ISR_UR (1 << 5) /* tx fifo underrun */
+#define ISR_OR (1 << 4) /* rx fifo overrun */
+#define ISR_RX (1 << 3) /* rx interrupt status */
+#define ISR_TX (1 << 2) /* tx interrupt status */
+#define ISR_RXTO (1 << 1) /* rx timeout */
+#define ISR_TXC (1 << 0) /* tx complete */
+
+/*
+ * interrupt enable. P52
+ */
+#define IE_RXTOFE (1 << 6) /* rx timeout fifo empty */
+#define IE_UR (1 << 5) /* tx fifo underrun */
+#define IE_OR (1 << 4) /* rx fifo overrun */
+#define IE_RX (1 << 3) /* rx interrupt status */
+#define IE_TX (1 << 2) /* tx interrupt status */
+#define IE_RXTO (1 << 1) /* rx timeout */
+#define IE_TXC (1 << 0) /* tx complete */
+
+/*
+ * slot flag register bits. P56
+ */
+#define SLFR_RWIS (1 << 13) /* raw wake-up interrupt status */
+#define SLFR_RGPIOINTR (1 << 12) /* raw gpio interrupt */
+#define SLFR_12TXE (1 << 11) /* slot 12 tx empty */
+#define SLFR_12RXV (1 << 10) /* slot 12 rx valid */
+#define SLFR_2TXE (1 << 9) /* slot 2 tx empty */
+#define SLFR_2RXV (1 << 8) /* slot 2 rx valid */
+#define SLFR_1TXE (1 << 7) /* slot 1 tx empty */
+#define SLFR_1RXV (1 << 6) /* slot 1 rx valid */
+#define SLFR_12TXB (1 << 5) /* slot 12 tx busy */
+#define SLFR_12RXB (1 << 4) /* slot 12 rx busy */
+#define SLFR_2TXB (1 << 3) /* slot 2 tx busy */
+#define SLFR_2RXB (1 << 2) /* slot 2 rx busy */
+#define SLFR_1TXB (1 << 1) /* slot 1 tx busy */
+#define SLFR_1RXB (1 << 0) /* slot 1 rx busy */
+
+/*
+ * Interrupt clear register.
+ */
+#define ICLR_RXTOFEC4 (1 << 12)
+#define ICLR_RXTOFEC3 (1 << 11)
+#define ICLR_RXTOFEC2 (1 << 10)
+#define ICLR_RXTOFEC1 (1 << 9)
+#define ICLR_TXUEC4 (1 << 8)
+#define ICLR_TXUEC3 (1 << 7)
+#define ICLR_TXUEC2 (1 << 6)
+#define ICLR_TXUEC1 (1 << 5)
+#define ICLR_RXOEC4 (1 << 4)
+#define ICLR_RXOEC3 (1 << 3)
+#define ICLR_RXOEC2 (1 << 2)
+#define ICLR_RXOEC1 (1 << 1)
+#define ICLR_WISC (1 << 0)
+
+/*
+ * Main control register bits. P62
+ */
+#define MAINCR_SCRA(x) ((x) << 10) /* secondary codec reg access */
+#define MAINCR_DMAEN (1 << 9) /* dma enable */
+#define MAINCR_SL12TXEN (1 << 8) /* slot 12 transmit enable */
+#define MAINCR_SL12RXEN (1 << 7) /* slot 12 receive enable */
+#define MAINCR_SL2TXEN (1 << 6) /* slot 2 transmit enable */
+#define MAINCR_SL2RXEN (1 << 5) /* slot 2 receive enable */
+#define MAINCR_SL1TXEN (1 << 4) /* slot 1 transmit enable */
+#define MAINCR_SL1RXEN (1 << 3) /* slot 1 receive enable */
+#define MAINCR_LPM (1 << 2) /* low power mode */
+#define MAINCR_LOOPBK (1 << 1) /* loopback */
+#define MAINCR_IE (1 << 0) /* aaci interface enable */
+
+/*
+ * Reset register bits. P65
+ */
+#define RESET_NRST (1 << 0)
+
+/*
+ * Sync register bits. P65
+ */
+#define SYNC_FORCE (1 << 0)
+
+/*
+ * Main flag register bits. P66
+ */
+#define MAINFR_TXB (1 << 1) /* transmit busy */
+#define MAINFR_RXB (1 << 0) /* receive busy */
+
+
+
+struct aaci_runtime {
+ void __iomem *base;
+ void __iomem *fifo;
+
+ struct ac97_pcm *pcm;
+ int pcm_open;
+
+ u32 cr;
+ struct snd_pcm_substream *substream;
+
+ /*
+ * PIO support
+ */
+ void *start;
+ void *end;
+ void *ptr;
+ int bytes;
+ unsigned int period;
+ unsigned int fifosz;
+};
+
+struct aaci {
+ struct amba_device *dev;
+ struct snd_card *card;
+ void __iomem *base;
+ unsigned int fifosize;
+
+ /* AC'97 */
+ struct mutex ac97_sem;
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97;
+
+ u32 maincr;
+ spinlock_t lock;
+
+ struct aaci_runtime playback;
+ struct aaci_runtime capture;
+
+ struct snd_pcm *pcm;
+};
+
+#define ACSTREAM_FRONT 0
+#define ACSTREAM_SURROUND 1
+#define ACSTREAM_LFE 2
+
+#endif
diff --git a/sound/arm/devdma.c b/sound/arm/devdma.c
new file mode 100644
index 0000000..9d1e666
--- /dev/null
+++ b/sound/arm/devdma.c
@@ -0,0 +1,80 @@
+/*
+ * linux/sound/arm/devdma.c
+ *
+ * Copyright (C) 2003-2004 Russell King, All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * ARM DMA shim for ALSA.
+ */
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+#include "devdma.h"
+
+void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+
+ if (runtime->dma_area == NULL)
+ return;
+
+ if (buf != &substream->dma_buffer) {
+ dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, buf->addr);
+ kfree(runtime->dma_buffer_p);
+ }
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+}
+
+int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+ int ret = 0;
+
+ if (buf) {
+ if (buf->bytes >= size)
+ goto out;
+ devdma_hw_free(dev, substream);
+ }
+
+ if (substream->dma_buffer.area != NULL && substream->dma_buffer.bytes >= size) {
+ buf = &substream->dma_buffer;
+ } else {
+ buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL);
+ if (!buf)
+ goto nomem;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = dev;
+ buf->area = dma_alloc_coherent(dev, size, &buf->addr, GFP_KERNEL);
+ buf->bytes = size;
+ buf->private_data = NULL;
+
+ if (!buf->area)
+ goto free;
+ }
+ snd_pcm_set_runtime_buffer(substream, buf);
+ ret = 1;
+ out:
+ runtime->dma_bytes = size;
+ return ret;
+
+ free:
+ kfree(buf);
+ nomem:
+ return -ENOMEM;
+}
+
+int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return dma_mmap_coherent(dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes);
+}
diff --git a/sound/arm/devdma.h b/sound/arm/devdma.h
new file mode 100644
index 0000000..d025329
--- /dev/null
+++ b/sound/arm/devdma.h
@@ -0,0 +1,3 @@
+void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream);
+int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size);
+int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma);
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
new file mode 100644
index 0000000..34c1d94
--- /dev/null
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -0,0 +1,384 @@
+/*
+ * Based on sound/arm/pxa2xx-ac97.c and sound/soc/pxa/pxa2xx-ac97.c
+ * which contain:
+ *
+ * Author: Nicolas Pitre
+ * Created: Dec 02, 2004
+ * Copyright: MontaVista Software Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/kernel.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+
+#include <sound/ac97_codec.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <asm/irq.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
+
+static DEFINE_MUTEX(car_mutex);
+static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
+static volatile long gsr_bits;
+static struct clk *ac97_clk;
+static struct clk *ac97conf_clk;
+
+/*
+ * Beware PXA27x bugs:
+ *
+ * o Slot 12 read from modem space will hang controller.
+ * o CDONE, SDONE interrupt fails after any slot 12 IO.
+ *
+ * We therefore have an hybrid approach for waiting on SDONE (interrupt or
+ * 1 jiffy timeout if interrupt never comes).
+ */
+
+unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ unsigned short val = -1;
+ volatile u32 *reg_addr;
+
+ mutex_lock(&car_mutex);
+
+ /* set up primary or secondary codec space */
+ if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS)
+ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
+ else
+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+ reg_addr += (reg >> 1);
+
+ /* start read access across the ac97 link */
+ GSR = GSR_CDONE | GSR_SDONE;
+ gsr_bits = 0;
+ val = *reg_addr;
+ if (reg == AC97_GPIO_STATUS)
+ goto out;
+ if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1) <= 0 &&
+ !((GSR | gsr_bits) & GSR_SDONE)) {
+ printk(KERN_ERR "%s: read error (ac97_reg=%d GSR=%#lx)\n",
+ __func__, reg, GSR | gsr_bits);
+ val = -1;
+ goto out;
+ }
+
+ /* valid data now */
+ GSR = GSR_CDONE | GSR_SDONE;
+ gsr_bits = 0;
+ val = *reg_addr;
+ /* but we've just started another cycle... */
+ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
+
+out: mutex_unlock(&car_mutex);
+ return val;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_read);
+
+void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ volatile u32 *reg_addr;
+
+ mutex_lock(&car_mutex);
+
+ /* set up primary or secondary codec space */
+ if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS)
+ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
+ else
+ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+ reg_addr += (reg >> 1);
+
+ GSR = GSR_CDONE | GSR_SDONE;
+ gsr_bits = 0;
+ *reg_addr = val;
+ if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1) <= 0 &&
+ !((GSR | gsr_bits) & GSR_CDONE))
+ printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n",
+ __func__, reg, GSR | gsr_bits);
+
+ mutex_unlock(&car_mutex);
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_write);
+
+#ifdef CONFIG_PXA25x
+static inline void pxa_ac97_warm_pxa25x(void)
+{
+ gsr_bits = 0;
+
+ GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN;
+ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
+}
+
+static inline void pxa_ac97_cold_pxa25x(void)
+{
+ GCR &= GCR_COLD_RST; /* clear everything but nCRST */
+ GCR &= ~GCR_COLD_RST; /* then assert nCRST */
+
+ gsr_bits = 0;
+
+ GCR = GCR_COLD_RST;
+ GCR |= GCR_CDONE_IE|GCR_SDONE_IE;
+ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
+}
+#endif
+
+#ifdef CONFIG_PXA27x
+static inline void pxa_ac97_warm_pxa27x(void)
+{
+ gsr_bits = 0;
+
+ /* warm reset broken on Bulverde,
+ so manually keep AC97 reset high */
+ pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH);
+ udelay(10);
+ GCR |= GCR_WARM_RST;
+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+ udelay(500);
+}
+
+static inline void pxa_ac97_cold_pxa27x(void)
+{
+ GCR &= GCR_COLD_RST; /* clear everything but nCRST */
+ GCR &= ~GCR_COLD_RST; /* then assert nCRST */
+
+ gsr_bits = 0;
+
+ /* PXA27x Developers Manual section 13.5.2.2.1 */
+ clk_enable(ac97conf_clk);
+ udelay(5);
+ clk_disable(ac97conf_clk);
+ GCR = GCR_COLD_RST;
+ udelay(50);
+}
+#endif
+
+#ifdef CONFIG_PXA3xx
+static inline void pxa_ac97_warm_pxa3xx(void)
+{
+ int timeout = 100;
+
+ gsr_bits = 0;
+
+ /* Can't use interrupts */
+ GCR |= GCR_WARM_RST;
+ while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
+ mdelay(1);
+}
+
+static inline void pxa_ac97_cold_pxa3xx(void)
+{
+ int timeout = 1000;
+
+ /* Hold CLKBPB for 100us */
+ GCR = 0;
+ GCR = GCR_CLKBPB;
+ udelay(100);
+ GCR = 0;
+
+ GCR &= GCR_COLD_RST; /* clear everything but nCRST */
+ GCR &= ~GCR_COLD_RST; /* then assert nCRST */
+
+ gsr_bits = 0;
+
+ /* Can't use interrupts on PXA3xx */
+ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
+
+ GCR = GCR_WARM_RST | GCR_COLD_RST;
+ while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--)
+ mdelay(10);
+}
+#endif
+
+bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
+{
+#ifdef CONFIG_PXA25x
+ if (cpu_is_pxa25x())
+ pxa_ac97_warm_pxa25x();
+ else
+#endif
+#ifdef CONFIG_PXA27x
+ if (cpu_is_pxa27x())
+ pxa_ac97_warm_pxa27x();
+ else
+#endif
+#ifdef CONFIG_PXA3xx
+ if (cpu_is_pxa3xx())
+ pxa_ac97_warm_pxa3xx();
+ else
+#endif
+ BUG();
+
+ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+ printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
+ __func__, gsr_bits);
+
+ return false;
+ }
+
+ return true;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
+
+bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
+{
+#ifdef CONFIG_PXA25x
+ if (cpu_is_pxa25x())
+ pxa_ac97_cold_pxa25x();
+ else
+#endif
+#ifdef CONFIG_PXA27x
+ if (cpu_is_pxa27x())
+ pxa_ac97_cold_pxa27x();
+ else
+#endif
+#ifdef CONFIG_PXA3xx
+ if (cpu_is_pxa3xx())
+ pxa_ac97_cold_pxa3xx();
+ else
+#endif
+ BUG();
+
+ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+ printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
+ __func__, gsr_bits);
+
+ return false;
+ }
+
+ return true;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_cold_reset);
+
+
+void pxa2xx_ac97_finish_reset(struct snd_ac97 *ac97)
+{
+ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
+ GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_finish_reset);
+
+static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id)
+{
+ long status;
+
+ status = GSR;
+ if (status) {
+ GSR = status;
+ gsr_bits |= status;
+ wake_up(&gsr_wq);
+
+ /* Although we don't use those we still need to clear them
+ since they tend to spuriously trigger when MMC is used
+ (hardware bug? go figure)... */
+ if (cpu_is_pxa27x()) {
+ MISR = MISR_EOC;
+ PISR = PISR_EOC;
+ MCSR = MCSR_EOC;
+ }
+
+ return IRQ_HANDLED;
+ }
+
+ return IRQ_NONE;
+}
+
+#ifdef CONFIG_PM
+int pxa2xx_ac97_hw_suspend(void)
+{
+ GCR |= GCR_ACLINK_OFF;
+ clk_disable(ac97_clk);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend);
+
+int pxa2xx_ac97_hw_resume(void)
+{
+ if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
+ pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
+ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
+ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
+ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
+ }
+ if (cpu_is_pxa27x()) {
+ /* Use GPIO 113 as AC97 Reset on Bulverde */
+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+ }
+ clk_enable(ac97_clk);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume);
+#endif
+
+int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
+{
+ int ret;
+
+ ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL);
+ if (ret < 0)
+ goto err;
+
+ if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
+ pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
+ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
+ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
+ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
+ }
+
+ if (cpu_is_pxa27x()) {
+ /* Use GPIO 113 as AC97 Reset on Bulverde */
+ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+ ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
+ if (IS_ERR(ac97conf_clk)) {
+ ret = PTR_ERR(ac97conf_clk);
+ ac97conf_clk = NULL;
+ goto err_irq;
+ }
+ }
+
+ ac97_clk = clk_get(&dev->dev, "AC97CLK");
+ if (IS_ERR(ac97_clk)) {
+ ret = PTR_ERR(ac97_clk);
+ ac97_clk = NULL;
+ goto err_irq;
+ }
+
+ return clk_enable(ac97_clk);
+
+err_irq:
+ GCR |= GCR_ACLINK_OFF;
+ if (ac97conf_clk) {
+ clk_put(ac97conf_clk);
+ ac97conf_clk = NULL;
+ }
+ free_irq(IRQ_AC97, NULL);
+err:
+ return ret;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe);
+
+void pxa2xx_ac97_hw_remove(struct platform_device *dev)
+{
+ GCR |= GCR_ACLINK_OFF;
+ free_irq(IRQ_AC97, NULL);
+ if (ac97conf_clk) {
+ clk_put(ac97conf_clk);
+ ac97conf_clk = NULL;
+ }
+ clk_disable(ac97_clk);
+ clk_put(ac97_clk);
+ ac97_clk = NULL;
+}
+EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_remove);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel/Marvell PXA sound library");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
new file mode 100644
index 0000000..c2635be
--- /dev/null
+++ b/sound/arm/pxa2xx-ac97.c
@@ -0,0 +1,260 @@
+/*
+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
+ *
+ * Author: Nicolas Pitre
+ * Created: Dec 02, 2004
+ * Copyright: MontaVista Software Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/audio.h>
+
+#include "pxa2xx-pcm.h"
+
+static void pxa2xx_ac97_reset(struct snd_ac97 *ac97)
+{
+ if (!pxa2xx_ac97_try_cold_reset(ac97)) {
+ pxa2xx_ac97_try_warm_reset(ac97);
+ }
+
+ pxa2xx_ac97_finish_reset(ac97);
+}
+
+static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
+ .read = pxa2xx_ac97_read,
+ .write = pxa2xx_ac97_write,
+ .reset = pxa2xx_ac97_reset,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = {
+ .name = "AC97 PCM out",
+ .dev_addr = __PREG(PCDR),
+ .drcmr = &DRCMR(12),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = {
+ .name = "AC97 PCM in",
+ .dev_addr = __PREG(PCDR),
+ .drcmr = &DRCMR(11),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct snd_pcm *pxa2xx_ac97_pcm;
+static struct snd_ac97 *pxa2xx_ac97_ac97;
+
+static int pxa2xx_ac97_pcm_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ pxa2xx_audio_ops_t *platform_ops;
+ int r;
+
+ runtime->hw.channels_min = 2;
+ runtime->hw.channels_max = 2;
+
+ r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ AC97_RATES_FRONT_DAC : AC97_RATES_ADC;
+ runtime->hw.rates = pxa2xx_ac97_ac97->rates[r];
+ snd_pcm_limit_hw_rates(runtime);
+
+ platform_ops = substream->pcm->card->dev->platform_data;
+ if (platform_ops && platform_ops->startup)
+ return platform_ops->startup(substream, platform_ops->priv);
+ else
+ return 0;
+}
+
+static void pxa2xx_ac97_pcm_shutdown(struct snd_pcm_substream *substream)
+{
+ pxa2xx_audio_ops_t *platform_ops;
+
+ platform_ops = substream->pcm->card->dev->platform_data;
+ if (platform_ops && platform_ops->shutdown)
+ platform_ops->shutdown(substream, platform_ops->priv);
+}
+
+static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
+ return snd_ac97_set_rate(pxa2xx_ac97_ac97, reg, runtime->rate);
+}
+
+static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
+ .playback_params = &pxa2xx_ac97_pcm_out,
+ .capture_params = &pxa2xx_ac97_pcm_in,
+ .startup = pxa2xx_ac97_pcm_startup,
+ .shutdown = pxa2xx_ac97_pcm_shutdown,
+ .prepare = pxa2xx_ac97_pcm_prepare,
+};
+
+#ifdef CONFIG_PM
+
+static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state)
+{
+ pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3cold);
+ snd_pcm_suspend_all(pxa2xx_ac97_pcm);
+ snd_ac97_suspend(pxa2xx_ac97_ac97);
+ if (platform_ops && platform_ops->suspend)
+ platform_ops->suspend(platform_ops->priv);
+
+ return pxa2xx_ac97_hw_suspend();
+}
+
+static int pxa2xx_ac97_do_resume(struct snd_card *card)
+{
+ pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data;
+ int rc;
+
+ rc = pxa2xx_ac97_hw_resume();
+ if (rc)
+ return rc;
+
+ if (platform_ops && platform_ops->resume)
+ platform_ops->resume(platform_ops->priv);
+ snd_ac97_resume(pxa2xx_ac97_ac97);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+ return 0;
+}
+
+static int pxa2xx_ac97_suspend(struct platform_device *dev, pm_message_t state)
+{
+ struct snd_card *card = platform_get_drvdata(dev);
+ int ret = 0;
+
+ if (card)
+ ret = pxa2xx_ac97_do_suspend(card, PMSG_SUSPEND);
+
+ return ret;
+}
+
+static int pxa2xx_ac97_resume(struct platform_device *dev)
+{
+ struct snd_card *card = platform_get_drvdata(dev);
+ int ret = 0;
+
+ if (card)
+ ret = pxa2xx_ac97_do_resume(card);
+
+ return ret;
+}
+
+#else
+#define pxa2xx_ac97_suspend NULL
+#define pxa2xx_ac97_resume NULL
+#endif
+
+static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
+{
+ struct snd_card *card;
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97_template ac97_template;
+ int ret;
+
+ ret = -ENOMEM;
+ card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+ THIS_MODULE, 0);
+ if (!card)
+ goto err;
+
+ card->dev = &dev->dev;
+ strncpy(card->driver, dev->dev.driver->name, sizeof(card->driver));
+
+ ret = pxa2xx_pcm_new(card, &pxa2xx_ac97_pcm_client, &pxa2xx_ac97_pcm);
+ if (ret)
+ goto err;
+
+ ret = pxa2xx_ac97_hw_probe(dev);
+ if (ret)
+ goto err;
+
+ ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus);
+ if (ret)
+ goto err_remove;
+ memset(&ac97_template, 0, sizeof(ac97_template));
+ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &pxa2xx_ac97_ac97);
+ if (ret)
+ goto err_remove;
+
+ snprintf(card->shortname, sizeof(card->shortname),
+ "%s", snd_ac97_get_short_name(pxa2xx_ac97_ac97));
+ snprintf(card->longname, sizeof(card->longname),
+ "%s (%s)", dev->dev.driver->name, card->mixername);
+
+ snd_card_set_dev(card, &dev->dev);
+ ret = snd_card_register(card);
+ if (ret == 0) {
+ platform_set_drvdata(dev, card);
+ return 0;
+ }
+
+err_remove:
+ pxa2xx_ac97_hw_remove(dev);
+err:
+ if (card)
+ snd_card_free(card);
+ return ret;
+}
+
+static int __devexit pxa2xx_ac97_remove(struct platform_device *dev)
+{
+ struct snd_card *card = platform_get_drvdata(dev);
+
+ if (card) {
+ snd_card_free(card);
+ platform_set_drvdata(dev, NULL);
+ pxa2xx_ac97_hw_remove(dev);
+ }
+
+ return 0;
+}
+
+static struct platform_driver pxa2xx_ac97_driver = {
+ .probe = pxa2xx_ac97_probe,
+ .remove = __devexit_p(pxa2xx_ac97_remove),
+ .suspend = pxa2xx_ac97_suspend,
+ .resume = pxa2xx_ac97_resume,
+ .driver = {
+ .name = "pxa2xx-ac97",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init pxa2xx_ac97_init(void)
+{
+ return platform_driver_register(&pxa2xx_ac97_driver);
+}
+
+static void __exit pxa2xx_ac97_exit(void)
+{
+ platform_driver_unregister(&pxa2xx_ac97_driver);
+}
+
+module_init(pxa2xx_ac97_init);
+module_exit(pxa2xx_ac97_exit);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-ac97");
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
new file mode 100644
index 0000000..75a0d74
--- /dev/null
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -0,0 +1,278 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <asm/dma.h>
+#include <mach/pxa-regs.h>
+
+#include "pxa2xx-pcm.h"
+
+static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192 - 32,
+ .periods_min = 1,
+ .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc),
+ .buffer_bytes_max = 128 * 1024,
+ .fifo_size = 32,
+};
+
+int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *rtd = runtime->private_data;
+ size_t totsize = params_buffer_bytes(params);
+ size_t period = params_period_bytes(params);
+ pxa_dma_desc *dma_desc;
+ dma_addr_t dma_buff_phys, next_desc_phys;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = totsize;
+
+ dma_desc = rtd->dma_desc_array;
+ next_desc_phys = rtd->dma_desc_array_phys;
+ dma_buff_phys = runtime->dma_addr;
+ do {
+ next_desc_phys += sizeof(pxa_dma_desc);
+ dma_desc->ddadr = next_desc_phys;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dma_desc->dsadr = dma_buff_phys;
+ dma_desc->dtadr = rtd->params->dev_addr;
+ } else {
+ dma_desc->dsadr = rtd->params->dev_addr;
+ dma_desc->dtadr = dma_buff_phys;
+ }
+ if (period > totsize)
+ period = totsize;
+ dma_desc->dcmd = rtd->params->dcmd | period | DCMD_ENDIRQEN;
+ dma_desc++;
+ dma_buff_phys += period;
+ } while (totsize -= period);
+ dma_desc[-1].ddadr = rtd->dma_desc_array_phys;
+
+ return 0;
+}
+EXPORT_SYMBOL(__pxa2xx_pcm_hw_params);
+
+int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
+
+ if (rtd && rtd->params)
+ *rtd->params->drcmr = 0;
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+EXPORT_SYMBOL(__pxa2xx_pcm_hw_free);
+
+int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
+ DCSR(prtd->dma_ch) = DCSR_RUN;
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ DCSR(prtd->dma_ch) &= ~DCSR_RUN;
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ DCSR(prtd->dma_ch) |= DCSR_RUN;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
+ DCSR(prtd->dma_ch) |= DCSR_RUN;
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL(pxa2xx_pcm_trigger);
+
+snd_pcm_uframes_t
+pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *prtd = runtime->private_data;
+
+ dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch);
+ snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+
+ if (x == runtime->buffer_size)
+ x = 0;
+ return x;
+}
+EXPORT_SYMBOL(pxa2xx_pcm_pointer);
+
+int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+ DCSR(prtd->dma_ch) &= ~DCSR_RUN;
+ DCSR(prtd->dma_ch) = 0;
+ DCMD(prtd->dma_ch) = 0;
+ *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD;
+
+ return 0;
+}
+EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
+
+void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
+{
+ struct snd_pcm_substream *substream = dev_id;
+ struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
+ int dcsr;
+
+ dcsr = DCSR(dma_ch);
+ DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN;
+
+ if (dcsr & DCSR_ENDINTR) {
+ snd_pcm_period_elapsed(substream);
+ } else {
+ printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
+ rtd->params->name, dma_ch, dcsr);
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ }
+}
+EXPORT_SYMBOL(pxa2xx_pcm_dma_irq);
+
+int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *rtd;
+ int ret;
+
+ runtime->hw = pxa2xx_pcm_hardware;
+
+ /*
+ * For mysterious reasons (and despite what the manual says)
+ * playback samples are lost if the DMA count is not a multiple
+ * of the DMA burst size. Let's add a rule to enforce that.
+ */
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret)
+ goto out;
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret)
+ goto out;
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ ret = -ENOMEM;
+ rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
+ if (!rtd)
+ goto out;
+ rtd->dma_desc_array =
+ dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE,
+ &rtd->dma_desc_array_phys, GFP_KERNEL);
+ if (!rtd->dma_desc_array)
+ goto err1;
+
+ runtime->private_data = rtd;
+ return 0;
+
+ err1:
+ kfree(rtd);
+ out:
+ return ret;
+}
+EXPORT_SYMBOL(__pxa2xx_pcm_open);
+
+int __pxa2xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *rtd = runtime->private_data;
+
+ dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
+ rtd->dma_desc_array, rtd->dma_desc_array_phys);
+ kfree(rtd);
+ return 0;
+}
+EXPORT_SYMBOL(__pxa2xx_pcm_close);
+
+int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+EXPORT_SYMBOL(pxa2xx_pcm_mmap);
+
+int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = pxa2xx_pcm_hardware.buffer_bytes_max;
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+ return 0;
+}
+EXPORT_SYMBOL(pxa2xx_pcm_preallocate_dma_buffer);
+
+void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+EXPORT_SYMBOL(pxa2xx_pcm_free_dma_buffers);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel PXA2xx sound library");
+MODULE_LICENSE("GPL");
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
new file mode 100644
index 0000000..535704f
--- /dev/null
+++ b/sound/arm/pxa2xx-pcm.c
@@ -0,0 +1,131 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <sound/core.h>
+#include <sound/pxa2xx-lib.h>
+
+#include "pxa2xx-pcm.h"
+
+static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_pcm_client *client = substream->private_data;
+
+ __pxa2xx_pcm_prepare(substream);
+
+ return client->prepare(substream);
+}
+
+static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_pcm_client *client = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *rtd;
+ int ret;
+
+ ret = __pxa2xx_pcm_open(substream);
+ if (ret)
+ goto out;
+
+ rtd = runtime->private_data;
+
+ rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ client->playback_params : client->capture_params;
+ ret = pxa_request_dma(rtd->params->name, DMA_PRIO_LOW,
+ pxa2xx_pcm_dma_irq, substream);
+ if (ret < 0)
+ goto err2;
+ rtd->dma_ch = ret;
+
+ ret = client->startup(substream);
+ if (!ret)
+ goto out;
+
+ pxa_free_dma(rtd->dma_ch);
+ err2:
+ __pxa2xx_pcm_close(substream);
+ out:
+ return ret;
+}
+
+static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_pcm_client *client = substream->private_data;
+ struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
+
+ pxa_free_dma(rtd->dma_ch);
+ client->shutdown(substream);
+
+ return __pxa2xx_pcm_close(substream);
+}
+
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
+ .open = pxa2xx_pcm_open,
+ .close = pxa2xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = __pxa2xx_pcm_hw_params,
+ .hw_free = __pxa2xx_pcm_hw_free,
+ .prepare = pxa2xx_pcm_prepare,
+ .trigger = pxa2xx_pcm_trigger,
+ .pointer = pxa2xx_pcm_pointer,
+ .mmap = pxa2xx_pcm_mmap,
+};
+
+static u64 pxa2xx_pcm_dmamask = 0xffffffff;
+
+int pxa2xx_pcm_new(struct snd_card *card, struct pxa2xx_pcm_client *client,
+ struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int play = client->playback_params ? 1 : 0;
+ int capt = client->capture_params ? 1 : 0;
+ int ret;
+
+ ret = snd_pcm_new(card, "PXA2xx-PCM", 0, play, capt, &pcm);
+ if (ret)
+ goto out;
+
+ pcm->private_data = client;
+ pcm->private_free = pxa2xx_pcm_free_dma_buffers;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &pxa2xx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (play) {
+ int stream = SNDRV_PCM_STREAM_PLAYBACK;
+ snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops);
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
+ if (ret)
+ goto out;
+ }
+ if (capt) {
+ int stream = SNDRV_PCM_STREAM_CAPTURE;
+ snd_pcm_set_ops(pcm, stream, &pxa2xx_pcm_ops);
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, stream);
+ if (ret)
+ goto out;
+ }
+
+ if (rpcm)
+ *rpcm = pcm;
+ ret = 0;
+
+ out:
+ return ret;
+}
+
+EXPORT_SYMBOL(pxa2xx_pcm_new);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h
new file mode 100644
index 0000000..5c4a4d3
--- /dev/null
+++ b/sound/arm/pxa2xx-pcm.h
@@ -0,0 +1,30 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include <asm/dma.h>
+
+struct pxa2xx_runtime_data {
+ int dma_ch;
+ struct pxa2xx_pcm_dma_params *params;
+ pxa_dma_desc *dma_desc_array;
+ dma_addr_t dma_desc_array_phys;
+};
+
+struct pxa2xx_pcm_client {
+ struct pxa2xx_pcm_dma_params *playback_params;
+ struct pxa2xx_pcm_dma_params *capture_params;
+ int (*startup)(struct snd_pcm_substream *);
+ void (*shutdown)(struct snd_pcm_substream *);
+ int (*prepare)(struct snd_pcm_substream *);
+};
+
+extern int pxa2xx_pcm_new(struct snd_card *, struct pxa2xx_pcm_client *, struct snd_pcm **);
+
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
new file mode 100644
index 0000000..1dcd51d
--- /dev/null
+++ b/sound/arm/sa11xx-uda1341.c
@@ -0,0 +1,983 @@
+/*
+ * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
+ * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License.
+ *
+ * History:
+ *
+ * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS
+ * 2002-03-20 Tomas Kasparek playback over ALSA is working
+ * 2002-03-28 Tomas Kasparek playback over OSS emulation is working
+ * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA)
+ * 2002-03-29 Tomas Kasparek capture is working (OSS emulation)
+ * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates)
+ * 2003-02-14 Brian Avery fixed full duplex mode, other updates
+ * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL)
+ * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
+ * working suspend and resume
+ * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again
+ * merged HAL layer (patches from Brian)
+ */
+
+/***************************************************************************************************
+*
+* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
+* available in the Alsa doc section on the website
+*
+* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
+* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
+* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
+* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
+* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
+* is a mem loc that always decodes to 0's w/ no off chip access.
+*
+* Some alsa terminology:
+* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes
+* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
+* buffer and 4 periods in the runtime structure this means we'll get an int every 256
+* bytes or 4 times per buffer.
+* A number of the sizes are in frames rather than bytes, use frames_to_bytes and
+* bytes_to_frames to convert. The easiest way to tell the units is to look at the
+* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
+*
+* Notes about the pointer fxn:
+* The pointer fxn needs to return the offset into the dma buffer in frames.
+* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
+*
+* Notes about pause/resume
+* Implementing this would be complicated so it's skipped. The problem case is:
+* A full duplex connection is going, then play is paused. At this point you need to start xmitting
+* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd
+* need to save off the dma info, and restore it properly on a resume. Yeach!
+*
+* Notes about transfer methods:
+* The async write calls fail. I probably need to implement something else to support them?
+*
+***************************************************************************************************/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/errno.h>
+#include <linux/ioctl.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#ifdef CONFIG_PM
+#include <linux/pm.h>
+#endif
+
+#include <mach/hardware.h>
+#include <mach/h3600.h>
+#include <asm/mach-types.h>
+#include <asm/dma.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+
+#include <linux/l3/l3.h>
+
+#undef DEBUG_MODE
+#undef DEBUG_FUNCTION_NAMES
+#include <sound/uda1341.h>
+
+/*
+ * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
+ * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
+ * module for Familiar 0.6.1
+ */
+
+/* {{{ Type definitions */
+
+MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
+MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
+
+static char *id; /* ID for this card */
+
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
+
+struct audio_stream {
+ char *id; /* identification string */
+ int stream_id; /* numeric identification */
+ dma_device_t dma_dev; /* device identifier for DMA */
+#ifdef HH_VERSION
+ dmach_t dmach; /* dma channel identification */
+#else
+ dma_regs_t *dma_regs; /* points to our DMA registers */
+#endif
+ unsigned int active:1; /* we are using this stream for transfer now */
+ int period; /* current transfer period */
+ int periods; /* current count of periods registerd in the DMA engine */
+ int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
+ unsigned int old_offset;
+ spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
+ struct snd_pcm_substream *stream;
+};
+
+struct sa11xx_uda1341 {
+ struct snd_card *card;
+ struct l3_client *uda1341;
+ struct snd_pcm *pcm;
+ long samplerate;
+ struct audio_stream s[2]; /* playback & capture */
+};
+
+static unsigned int rates[] = {
+ 8000, 10666, 10985, 14647,
+ 16000, 21970, 22050, 24000,
+ 29400, 32000, 44100, 48000,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static struct platform_device *device;
+
+/* }}} */
+
+/* {{{ Clock and sample rate stuff */
+
+/*
+ * Stop-gap solution until rest of hh.org HAL stuff is merged.
+ */
+#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
+#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
+
+#ifdef CONFIG_SA1100_H3XXX
+#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
+#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
+#else
+#error This driver could serve H3x00 handhelds only!
+#endif
+
+static void sa11xx_uda1341_set_audio_clock(long val)
+{
+ switch (val) {
+ case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
+ GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
+ break;
+
+ case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
+ GPSR = GPIO_H3600_CLK_SET0;
+ GPCR = GPIO_H3600_CLK_SET1;
+ break;
+
+ case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
+ GPCR = GPIO_H3600_CLK_SET0;
+ GPSR = GPIO_H3600_CLK_SET1;
+ break;
+
+ case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
+ GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
+ break;
+ }
+}
+
+static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
+{
+ int clk_div = 0;
+ int clk=0;
+
+ /* We don't want to mess with clocks when frames are in flight */
+ Ser4SSCR0 &= ~SSCR0_SSE;
+ /* wait for any frame to complete */
+ udelay(125);
+
+ /*
+ * We have the following clock sources:
+ * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
+ * Those can be divided either by 256, 384 or 512.
+ * This makes up 12 combinations for the following samplerates...
+ */
+ if (rate >= 48000)
+ rate = 48000;
+ else if (rate >= 44100)
+ rate = 44100;
+ else if (rate >= 32000)
+ rate = 32000;
+ else if (rate >= 29400)
+ rate = 29400;
+ else if (rate >= 24000)
+ rate = 24000;
+ else if (rate >= 22050)
+ rate = 22050;
+ else if (rate >= 21970)
+ rate = 21970;
+ else if (rate >= 16000)
+ rate = 16000;
+ else if (rate >= 14647)
+ rate = 14647;
+ else if (rate >= 10985)
+ rate = 10985;
+ else if (rate >= 10666)
+ rate = 10666;
+ else
+ rate = 8000;
+
+ /* Set the external clock generator */
+
+ sa11xx_uda1341_set_audio_clock(rate);
+
+ /* Select the clock divisor */
+ switch (rate) {
+ case 8000:
+ case 10985:
+ case 22050:
+ case 24000:
+ clk = F512;
+ clk_div = SSCR0_SerClkDiv(16);
+ break;
+ case 16000:
+ case 21970:
+ case 44100:
+ case 48000:
+ clk = F256;
+ clk_div = SSCR0_SerClkDiv(8);
+ break;
+ case 10666:
+ case 14647:
+ case 29400:
+ case 32000:
+ clk = F384;
+ clk_div = SSCR0_SerClkDiv(12);
+ break;
+ }
+
+ /* FMT setting should be moved away when other FMTs are added (FIXME) */
+ l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
+
+ l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
+ Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
+ sa11xx_uda1341->samplerate = rate;
+}
+
+/* }}} */
+
+/* {{{ HW init and shutdown */
+
+static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
+{
+ unsigned long flags;
+
+ /* Setup DMA stuff */
+ sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
+ sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
+ sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
+
+ sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
+ sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
+ sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
+
+ /* Initialize the UDA1341 internal state */
+
+ /* Setup the uarts */
+ local_irq_save(flags);
+ GAFR |= (GPIO_SSP_CLK);
+ GPDR &= ~(GPIO_SSP_CLK);
+ Ser4SSCR0 = 0;
+ Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
+ Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
+ Ser4SSCR0 |= SSCR0_SSE;
+ local_irq_restore(flags);
+
+ /* Enable the audio power */
+
+ clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
+ set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
+ set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
+
+ /* Wait for the UDA1341 to wake up */
+ mdelay(1); //FIXME - was removed by Perex - Why?
+
+ /* Initialize the UDA1341 internal state */
+ l3_open(sa11xx_uda1341->uda1341);
+
+ /* external clock configuration (after l3_open - regs must be initialized */
+ sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
+
+ /* Wait for the UDA1341 to wake up */
+ set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
+ mdelay(1);
+
+ /* make the left and right channels unswapped (flip the WS latch) */
+ Ser4SSDR = 0;
+
+ clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
+}
+
+static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
+{
+ /* mute on */
+ set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
+
+ /* disable the audio power and all signals leading to the audio chip */
+ l3_close(sa11xx_uda1341->uda1341);
+ Ser4SSCR0 = 0;
+ clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
+
+ /* power off and mute off */
+ /* FIXME - is muting off necesary??? */
+
+ clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
+ clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
+}
+
+/* }}} */
+
+/* {{{ DMA staff */
+
+/*
+ * these are the address and sizes used to fill the xmit buffer
+ * so we can get a clock in record only mode
+ */
+#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
+#define FORCE_CLOCK_SIZE 4096 // was 2048
+
+// FIXME Why this value exactly - wrote comment
+#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
+
+#ifdef HH_VERSION
+
+static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
+{
+ int ret;
+
+ ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
+ if (ret < 0) {
+ printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
+ return ret;
+ }
+ sa1100_dma_set_callback(s->dmach, callback);
+ return 0;
+}
+
+static inline void audio_dma_free(struct audio_stream *s)
+{
+ sa1100_free_dma(s->dmach);
+ s->dmach = -1;
+}
+
+#else
+
+static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
+{
+ int ret;
+
+ ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
+ if (ret < 0)
+ printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
+ return ret;
+}
+
+static void audio_dma_free(struct audio_stream *s)
+{
+ sa1100_free_dma(s->dma_regs);
+ s->dma_regs = 0;
+}
+
+#endif
+
+static u_int audio_get_dma_pos(struct audio_stream *s)
+{
+ struct snd_pcm_substream *substream = s->stream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int offset;
+ unsigned long flags;
+ dma_addr_t addr;
+
+ // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
+ spin_lock_irqsave(&s->dma_lock, flags);
+#ifdef HH_VERSION
+ sa1100_dma_get_current(s->dmach, NULL, &addr);
+#else
+ addr = sa1100_get_dma_pos((s)->dma_regs);
+#endif
+ offset = addr - runtime->dma_addr;
+ spin_unlock_irqrestore(&s->dma_lock, flags);
+
+ offset = bytes_to_frames(runtime,offset);
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+
+ return offset;
+}
+
+/*
+ * this stops the dma and clears the dma ptrs
+ */
+static void audio_stop_dma(struct audio_stream *s)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&s->dma_lock, flags);
+ s->active = 0;
+ s->period = 0;
+ /* this stops the dma channel and clears the buffer ptrs */
+#ifdef HH_VERSION
+ sa1100_dma_flush_all(s->dmach);
+#else
+ sa1100_clear_dma(s->dma_regs);
+#endif
+ spin_unlock_irqrestore(&s->dma_lock, flags);
+}
+
+static void audio_process_dma(struct audio_stream *s)
+{
+ struct snd_pcm_substream *substream = s->stream;
+ struct snd_pcm_runtime *runtime;
+ unsigned int dma_size;
+ unsigned int offset;
+ int ret;
+
+ /* we are requested to process synchronization DMA transfer */
+ if (s->tx_spin) {
+ if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK))
+ return;
+ /* fill the xmit dma buffers and return */
+#ifdef HH_VERSION
+ sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
+#else
+ while (1) {
+ ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
+ if (ret)
+ return;
+ }
+#endif
+ return;
+ }
+
+ /* must be set here - only valid for running streams, not for forced_clock dma fills */
+ runtime = substream->runtime;
+ while (s->active && s->periods < runtime->periods) {
+ dma_size = frames_to_bytes(runtime, runtime->period_size);
+ if (s->old_offset) {
+ /* a little trick, we need resume from old position */
+ offset = frames_to_bytes(runtime, s->old_offset - 1);
+ s->old_offset = 0;
+ s->periods = 0;
+ s->period = offset / dma_size;
+ offset %= dma_size;
+ dma_size = dma_size - offset;
+ if (!dma_size)
+ continue; /* special case */
+ } else {
+ offset = dma_size * s->period;
+ snd_BUG_ON(dma_size > DMA_BUF_SIZE);
+ }
+#ifdef HH_VERSION
+ ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
+ if (ret)
+ return; //FIXME
+#else
+ ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
+ if (ret) {
+ printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
+ return;
+ }
+#endif
+
+ s->period++;
+ s->period %= runtime->periods;
+ s->periods++;
+ }
+}
+
+#ifdef HH_VERSION
+static void audio_dma_callback(void *data, int size)
+#else
+static void audio_dma_callback(void *data)
+#endif
+{
+ struct audio_stream *s = data;
+
+ /*
+ * If we are getting a callback for an active stream then we inform
+ * the PCM middle layer we've finished a period
+ */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ spin_lock(&s->dma_lock);
+ if (!s->tx_spin && s->periods > 0)
+ s->periods--;
+ audio_process_dma(s);
+ spin_unlock(&s->dma_lock);
+}
+
+/* }}} */
+
+/* {{{ PCM setting */
+
+/* {{{ trigger & timer */
+
+static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
+ int stream_id = substream->pstr->stream;
+ struct audio_stream *s = &chip->s[stream_id];
+ struct audio_stream *s1 = &chip->s[stream_id ^ 1];
+ int err = 0;
+
+ /* note local interrupts are already disabled in the midlevel code */
+ spin_lock(&s->dma_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* now we need to make sure a record only stream has a clock */
+ if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
+ /* we need to force fill the xmit DMA with zeros */
+ s1->tx_spin = 1;
+ audio_process_dma(s1);
+ }
+ /* this case is when you were recording then you turn on a
+ * playback stream so we stop (also clears it) the dma first,
+ * clear the sync flag and then we let it turned on
+ */
+ else {
+ s->tx_spin = 0;
+ }
+
+ /* requested stream startup */
+ s->active = 1;
+ audio_process_dma(s);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ /* requested stream shutdown */
+ audio_stop_dma(s);
+
+ /*
+ * now we need to make sure a record only stream has a clock
+ * so if we're stopping a playback with an active capture
+ * we need to turn the 0 fill dma on for the xmit side
+ */
+ if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
+ /* we need to force fill the xmit DMA with zeros */
+ s->tx_spin = 1;
+ audio_process_dma(s);
+ }
+ /*
+ * we killed a capture only stream, so we should also kill
+ * the zero fill transmit
+ */
+ else {
+ if (s1->tx_spin) {
+ s1->tx_spin = 0;
+ audio_stop_dma(s1);
+ }
+ }
+
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ s->active = 0;
+#ifdef HH_VERSION
+ sa1100_dma_stop(s->dmach);
+#else
+ //FIXME - DMA API
+#endif
+ s->old_offset = audio_get_dma_pos(s) + 1;
+#ifdef HH_VERSION
+ sa1100_dma_flush_all(s->dmach);
+#else
+ //FIXME - DMA API
+#endif
+ s->periods = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ s->active = 1;
+ s->tx_spin = 0;
+ audio_process_dma(s);
+ if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
+ s1->tx_spin = 1;
+ audio_process_dma(s1);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+#ifdef HH_VERSION
+ sa1100_dma_stop(s->dmach);
+#else
+ //FIXME - DMA API
+#endif
+ s->active = 0;
+ if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (s1->active) {
+ s->tx_spin = 1;
+ s->old_offset = audio_get_dma_pos(s) + 1;
+#ifdef HH_VERSION
+ sa1100_dma_flush_all(s->dmach);
+#else
+ //FIXME - DMA API
+#endif
+ audio_process_dma(s);
+ }
+ } else {
+ if (s1->tx_spin) {
+ s1->tx_spin = 0;
+#ifdef HH_VERSION
+ sa1100_dma_flush_all(s1->dmach);
+#else
+ //FIXME - DMA API
+#endif
+ }
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ s->active = 1;
+ if (s->old_offset) {
+ s->tx_spin = 0;
+ audio_process_dma(s);
+ break;
+ }
+ if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
+ s1->tx_spin = 1;
+ audio_process_dma(s1);
+ }
+#ifdef HH_VERSION
+ sa1100_dma_resume(s->dmach);
+#else
+ //FIXME - DMA API
+#endif
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ spin_unlock(&s->dma_lock);
+ return err;
+}
+
+static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
+{
+ struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_stream *s = &chip->s[substream->pstr->stream];
+
+ /* set requested samplerate */
+ sa11xx_uda1341_set_samplerate(chip, runtime->rate);
+
+ /* set requestd format when available */
+ /* set FMT here !!! FIXME */
+
+ s->period = 0;
+ s->periods = 0;
+
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
+{
+ struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
+ return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
+}
+
+/* }}} */
+
+static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
+{
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 64*1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = DMA_BUF_SIZE,
+ .periods_min = 2,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
+{
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 64*1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = DMA_BUF_SIZE,
+ .periods_min = 2,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
+{
+ struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int stream_id = substream->pstr->stream;
+ int err;
+
+ chip->s[stream_id].stream = substream;
+
+ if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
+ runtime->hw = snd_sa11xx_uda1341_playback;
+ else
+ runtime->hw = snd_sa11xx_uda1341_capture;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
+{
+ struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
+
+ chip->s[substream->pstr->stream].stream = NULL;
+ return 0;
+}
+
+/* {{{ HW params & free */
+
+static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* }}} */
+
+static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
+ .open = snd_card_sa11xx_uda1341_open,
+ .close = snd_card_sa11xx_uda1341_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sa11xx_uda1341_hw_params,
+ .hw_free = snd_sa11xx_uda1341_hw_free,
+ .prepare = snd_sa11xx_uda1341_prepare,
+ .trigger = snd_sa11xx_uda1341_trigger,
+ .pointer = snd_sa11xx_uda1341_pointer,
+};
+
+static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
+ .open = snd_card_sa11xx_uda1341_open,
+ .close = snd_card_sa11xx_uda1341_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sa11xx_uda1341_hw_params,
+ .hw_free = snd_sa11xx_uda1341_hw_free,
+ .prepare = snd_sa11xx_uda1341_prepare,
+ .trigger = snd_sa11xx_uda1341_trigger,
+ .pointer = snd_sa11xx_uda1341_pointer,
+};
+
+static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
+ return err;
+
+ /*
+ * this sets up our initial buffers and sets the dma_type to isa.
+ * isa works but I'm not sure why (or if) it's the right choice
+ * this may be too large, trying it for now
+ */
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, 64*1024);
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
+ pcm->private_data = sa11xx_uda1341;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "UDA1341 PCM");
+
+ sa11xx_uda1341_audio_init(sa11xx_uda1341);
+
+ /* setup DMA controller */
+ audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
+ audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
+
+ sa11xx_uda1341->pcm = pcm;
+
+ return 0;
+}
+
+/* }}} */
+
+/* {{{ module init & exit */
+
+#ifdef CONFIG_PM
+
+static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
+ pm_message_t state)
+{
+ struct snd_card *card = platform_get_drvdata(devptr);
+ struct sa11xx_uda1341 *chip = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+#ifdef HH_VERSION
+ sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
+ sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
+#else
+ //FIXME
+#endif
+ l3_command(chip->uda1341, CMD_SUSPEND, NULL);
+ sa11xx_uda1341_audio_shutdown(chip);
+
+ return 0;
+}
+
+static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
+{
+ struct snd_card *card = platform_get_drvdata(devptr);
+ struct sa11xx_uda1341 *chip = card->private_data;
+
+ sa11xx_uda1341_audio_init(chip);
+ l3_command(chip->uda1341, CMD_RESUME, NULL);
+#ifdef HH_VERSION
+ sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
+ sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
+#else
+ //FIXME
+#endif
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* COMFIG_PM */
+
+void snd_sa11xx_uda1341_free(struct snd_card *card)
+{
+ struct sa11xx_uda1341 *chip = card->private_data;
+
+ audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
+ audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
+}
+
+static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr)
+{
+ int err;
+ struct snd_card *card;
+ struct sa11xx_uda1341 *chip;
+
+ /* register the soundcard */
+ card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
+ if (card == NULL)
+ return -ENOMEM;
+
+ chip = card->private_data;
+ spin_lock_init(&chip->s[0].dma_lock);
+ spin_lock_init(&chip->s[1].dma_lock);
+
+ card->private_free = snd_sa11xx_uda1341_free;
+ chip->card = card;
+ chip->samplerate = AUDIO_RATE_DEFAULT;
+
+ // mixer
+ if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
+ goto nodev;
+
+ // PCM
+ if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
+ goto nodev;
+
+ strcpy(card->driver, "UDA1341");
+ strcpy(card->shortname, "H3600 UDA1341TS");
+ sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
+
+ snd_card_set_dev(card, &devptr->dev);
+
+ if ((err = snd_card_register(card)) == 0) {
+ printk( KERN_INFO "iPAQ audio support initialized\n" );
+ platform_set_drvdata(devptr, card);
+ return 0;
+ }
+
+ nodev:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
+
+static struct platform_driver sa11xx_uda1341_driver = {
+ .probe = sa11xx_uda1341_probe,
+ .remove = __devexit_p(sa11xx_uda1341_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_sa11xx_uda1341_suspend,
+ .resume = snd_sa11xx_uda1341_resume,
+#endif
+ .driver = {
+ .name = SA11XX_UDA1341_DRIVER,
+ },
+};
+
+static int __init sa11xx_uda1341_init(void)
+{
+ int err;
+
+ if (!machine_is_h3xxx())
+ return -ENODEV;
+ if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
+ return err;
+ device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
+ if (!IS_ERR(device)) {
+ if (platform_get_drvdata(device))
+ return 0;
+ platform_device_unregister(device);
+ err = -ENODEV;
+ } else
+ err = PTR_ERR(device);
+ platform_driver_unregister(&sa11xx_uda1341_driver);
+ return err;
+}
+
+static void __exit sa11xx_uda1341_exit(void)
+{
+ platform_device_unregister(device);
+ platform_driver_unregister(&sa11xx_uda1341_driver);
+}
+
+module_init(sa11xx_uda1341_init);
+module_exit(sa11xx_uda1341_exit);
+
+/* }}} */
+
+/*
+ * Local variables:
+ * indent-tabs-mode: t
+ * End:
+ */
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
new file mode 100644
index 0000000..66348c9
--- /dev/null
+++ b/sound/core/Kconfig
@@ -0,0 +1,186 @@
+# ALSA soundcard-configuration
+config SND_TIMER
+ tristate
+
+config SND_PCM
+ tristate
+ select SND_TIMER
+
+config SND_HWDEP
+ tristate
+
+config SND_RAWMIDI
+ tristate
+
+# To be effective this also requires INPUT - users should say:
+# select SND_JACK if INPUT=y || INPUT=SND
+# to avoid having to force INPUT on.
+config SND_JACK
+ bool
+
+config SND_SEQUENCER
+ tristate "Sequencer support"
+ select SND_TIMER
+ help
+ Say Y or M to enable MIDI sequencer and router support. This
+ feature allows routing and enqueueing of MIDI events. Events
+ can be processed at a given time.
+
+ Many programs require this feature, so you should enable it
+ unless you know what you're doing.
+
+config SND_SEQ_DUMMY
+ tristate "Sequencer dummy client"
+ depends on SND_SEQUENCER
+ help
+ Say Y here to enable the dummy sequencer client. This client
+ is a simple MIDI-through client: all normal input events are
+ redirected to the output port immediately.
+
+ You don't need this unless you want to connect many MIDI
+ devices or applications together.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-seq-dummy.
+
+config SND_OSSEMUL
+ select SOUND_OSS_CORE
+ bool
+
+config SND_MIXER_OSS
+ tristate "OSS Mixer API"
+ select SND_OSSEMUL
+ help
+ To enable OSS mixer API emulation (/dev/mixer*), say Y here
+ and read <file:Documentation/sound/alsa/OSS-Emulation.txt>.
+
+ Many programs still use the OSS API, so say Y.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-mixer-oss.
+
+config SND_PCM_OSS
+ tristate "OSS PCM (digital audio) API"
+ select SND_OSSEMUL
+ select SND_PCM
+ help
+ To enable OSS digital audio (PCM) emulation (/dev/dsp*), say Y
+ here and read <file:Documentation/sound/alsa/OSS-Emulation.txt>.
+
+ Many programs still use the OSS API, so say Y.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-pcm-oss.
+
+config SND_PCM_OSS_PLUGINS
+ bool "OSS PCM (digital audio) API - Include plugin system"
+ depends on SND_PCM_OSS
+ default y
+ help
+ If you disable this option, the ALSA's OSS PCM API will not
+ support conversion of channels, formats and rates. It will
+ behave like most of new OSS/Free drivers in 2.4/2.6 kernels.
+
+config SND_SEQUENCER_OSS
+ bool "OSS Sequencer API"
+ depends on SND_SEQUENCER
+ select SND_OSSEMUL
+ help
+ Say Y here to enable OSS sequencer emulation (both
+ /dev/sequencer and /dev/music interfaces).
+
+ Many programs still use the OSS API, so say Y.
+
+ If you choose M in "Sequencer support" (SND_SEQUENCER),
+ this will be compiled as a module. The module will be called
+ snd-seq-oss.
+
+config SND_RTCTIMER
+ tristate "RTC Timer support"
+ depends on RTC
+ select SND_TIMER
+ help
+ Say Y here to enable RTC timer support for ALSA. ALSA uses
+ the RTC timer as a precise timing source and maps the RTC
+ timer to ALSA's timer interface. The ALSA sequencer code also
+ can use this timing source.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-rtctimer.
+
+ Note that this option is exclusive with the new RTC drivers
+ (CONFIG_RTC_CLASS) since this requires the old API.
+
+config SND_SEQ_RTCTIMER_DEFAULT
+ bool "Use RTC as default sequencer timer"
+ depends on SND_RTCTIMER && SND_SEQUENCER
+ default y
+ help
+ Say Y here to use the RTC timer as the default sequencer
+ timer. This is strongly recommended because it ensures
+ precise MIDI timing even when the system timer runs at less
+ than 1000 Hz.
+
+ If in doubt, say Y.
+
+config SND_DYNAMIC_MINORS
+ bool "Dynamic device file minor numbers"
+ help
+ If you say Y here, the minor numbers of ALSA device files in
+ /dev/snd/ are allocated dynamically. This allows you to have
+ more than 8 sound cards, but requires a dynamic device file
+ system like udev.
+
+ If you are unsure about this, say N here.
+
+config SND_SUPPORT_OLD_API
+ bool "Support old ALSA API"
+ default y
+ help
+ Say Y here to support the obsolete ALSA PCM API (ver.0.9.0 rc3
+ or older).
+
+config SND_VERBOSE_PROCFS
+ bool "Verbose procfs contents"
+ depends on PROC_FS
+ default y
+ help
+ Say Y here to include code for verbose procfs contents (provides
+ useful information to developers when a problem occurs). On the
+ other side, it makes the ALSA subsystem larger.
+
+config SND_VERBOSE_PRINTK
+ bool "Verbose printk"
+ help
+ Say Y here to enable verbose log messages. These messages
+ will help to identify source file and position containing
+ printed messages.
+
+ You don't need this unless you're debugging ALSA.
+
+config SND_DEBUG
+ bool "Debug"
+ help
+ Say Y here to enable ALSA debug code.
+
+config SND_DEBUG_VERBOSE
+ bool "More verbose debug"
+ depends on SND_DEBUG
+ help
+ Say Y here to enable extra-verbose debugging messages.
+
+ Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages.
+ So, say Y only if you are ready to be annoyed.
+
+config SND_PCM_XRUN_DEBUG
+ bool "Enable PCM ring buffer overrun/underrun debugging"
+ default n
+ depends on SND_DEBUG && SND_VERBOSE_PROCFS
+ help
+ Say Y to enable the PCM ring buffer overrun/underrun debugging.
+ It is usually not required, but if you have trouble with
+ sound clicking when system is loaded, it may help to determine
+ the process or driver which causes the scheduling gaps.
+
+config SND_VMASTER
+ bool
diff --git a/sound/core/Makefile b/sound/core/Makefile
new file mode 100644
index 0000000..d57125a
--- /dev/null
+++ b/sound/core/Makefile
@@ -0,0 +1,31 @@
+#
+# Makefile for ALSA
+# Copyright (c) 1999,2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-y := sound.o init.o memory.o info.o control.o misc.o device.o
+snd-$(CONFIG_ISA_DMA_API) += isadma.o
+snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o
+snd-$(CONFIG_SND_VMASTER) += vmaster.o
+snd-$(CONFIG_SND_JACK) += jack.o
+
+snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
+ pcm_memory.o
+
+snd-page-alloc-y := memalloc.o
+snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o
+
+snd-rawmidi-objs := rawmidi.o
+snd-timer-objs := timer.o
+snd-rtctimer-objs := rtctimer.o
+snd-hwdep-objs := hwdep.o
+
+obj-$(CONFIG_SND) += snd.o
+obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o
+obj-$(CONFIG_SND_TIMER) += snd-timer.o
+obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o
+obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o
+obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o
+
+obj-$(CONFIG_SND_OSSEMUL) += oss/
+obj-$(CONFIG_SND_SEQUENCER) += seq/
diff --git a/sound/core/control.c b/sound/core/control.c
new file mode 100644
index 0000000..636b3b5
--- /dev/null
+++ b/sound/core/control.c
@@ -0,0 +1,1517 @@
+/*
+ * Routines for driver control interface
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/threads.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/info.h>
+#include <sound/control.h>
+
+/* max number of user-defined controls */
+#define MAX_USER_CONTROLS 32
+
+struct snd_kctl_ioctl {
+ struct list_head list; /* list of all ioctls */
+ snd_kctl_ioctl_func_t fioctl;
+};
+
+static DECLARE_RWSEM(snd_ioctl_rwsem);
+static LIST_HEAD(snd_control_ioctls);
+#ifdef CONFIG_COMPAT
+static LIST_HEAD(snd_control_compat_ioctls);
+#endif
+
+static int snd_ctl_open(struct inode *inode, struct file *file)
+{
+ unsigned long flags;
+ struct snd_card *card;
+ struct snd_ctl_file *ctl;
+ int err;
+
+ card = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_CONTROL);
+ if (!card) {
+ err = -ENODEV;
+ goto __error1;
+ }
+ err = snd_card_file_add(card, file);
+ if (err < 0) {
+ err = -ENODEV;
+ goto __error1;
+ }
+ if (!try_module_get(card->module)) {
+ err = -EFAULT;
+ goto __error2;
+ }
+ ctl = kzalloc(sizeof(*ctl), GFP_KERNEL);
+ if (ctl == NULL) {
+ err = -ENOMEM;
+ goto __error;
+ }
+ INIT_LIST_HEAD(&ctl->events);
+ init_waitqueue_head(&ctl->change_sleep);
+ spin_lock_init(&ctl->read_lock);
+ ctl->card = card;
+ ctl->prefer_pcm_subdevice = -1;
+ ctl->prefer_rawmidi_subdevice = -1;
+ ctl->pid = current->pid;
+ file->private_data = ctl;
+ write_lock_irqsave(&card->ctl_files_rwlock, flags);
+ list_add_tail(&ctl->list, &card->ctl_files);
+ write_unlock_irqrestore(&card->ctl_files_rwlock, flags);
+ return 0;
+
+ __error:
+ module_put(card->module);
+ __error2:
+ snd_card_file_remove(card, file);
+ __error1:
+ return err;
+}
+
+static void snd_ctl_empty_read_queue(struct snd_ctl_file * ctl)
+{
+ unsigned long flags;
+ struct snd_kctl_event *cread;
+
+ spin_lock_irqsave(&ctl->read_lock, flags);
+ while (!list_empty(&ctl->events)) {
+ cread = snd_kctl_event(ctl->events.next);
+ list_del(&cread->list);
+ kfree(cread);
+ }
+ spin_unlock_irqrestore(&ctl->read_lock, flags);
+}
+
+static int snd_ctl_release(struct inode *inode, struct file *file)
+{
+ unsigned long flags;
+ struct snd_card *card;
+ struct snd_ctl_file *ctl;
+ struct snd_kcontrol *control;
+ unsigned int idx;
+
+ ctl = file->private_data;
+ file->private_data = NULL;
+ card = ctl->card;
+ write_lock_irqsave(&card->ctl_files_rwlock, flags);
+ list_del(&ctl->list);
+ write_unlock_irqrestore(&card->ctl_files_rwlock, flags);
+ down_write(&card->controls_rwsem);
+ list_for_each_entry(control, &card->controls, list)
+ for (idx = 0; idx < control->count; idx++)
+ if (control->vd[idx].owner == ctl)
+ control->vd[idx].owner = NULL;
+ up_write(&card->controls_rwsem);
+ snd_ctl_empty_read_queue(ctl);
+ kfree(ctl);
+ module_put(card->module);
+ snd_card_file_remove(card, file);
+ return 0;
+}
+
+void snd_ctl_notify(struct snd_card *card, unsigned int mask,
+ struct snd_ctl_elem_id *id)
+{
+ unsigned long flags;
+ struct snd_ctl_file *ctl;
+ struct snd_kctl_event *ev;
+
+ if (snd_BUG_ON(!card || !id))
+ return;
+ read_lock(&card->ctl_files_rwlock);
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+ card->mixer_oss_change_count++;
+#endif
+ list_for_each_entry(ctl, &card->ctl_files, list) {
+ if (!ctl->subscribed)
+ continue;
+ spin_lock_irqsave(&ctl->read_lock, flags);
+ list_for_each_entry(ev, &ctl->events, list) {
+ if (ev->id.numid == id->numid) {
+ ev->mask |= mask;
+ goto _found;
+ }
+ }
+ ev = kzalloc(sizeof(*ev), GFP_ATOMIC);
+ if (ev) {
+ ev->id = *id;
+ ev->mask = mask;
+ list_add_tail(&ev->list, &ctl->events);
+ } else {
+ snd_printk(KERN_ERR "No memory available to allocate event\n");
+ }
+ _found:
+ wake_up(&ctl->change_sleep);
+ spin_unlock_irqrestore(&ctl->read_lock, flags);
+ kill_fasync(&ctl->fasync, SIGIO, POLL_IN);
+ }
+ read_unlock(&card->ctl_files_rwlock);
+}
+
+EXPORT_SYMBOL(snd_ctl_notify);
+
+/**
+ * snd_ctl_new - create a control instance from the template
+ * @control: the control template
+ * @access: the default control access
+ *
+ * Allocates a new struct snd_kcontrol instance and copies the given template
+ * to the new instance. It does not copy volatile data (access).
+ *
+ * Returns the pointer of the new instance, or NULL on failure.
+ */
+static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control,
+ unsigned int access)
+{
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+
+ if (snd_BUG_ON(!control || !control->count))
+ return NULL;
+ kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL);
+ if (kctl == NULL) {
+ snd_printk(KERN_ERR "Cannot allocate control instance\n");
+ return NULL;
+ }
+ *kctl = *control;
+ for (idx = 0; idx < kctl->count; idx++)
+ kctl->vd[idx].access = access;
+ return kctl;
+}
+
+/**
+ * snd_ctl_new1 - create a control instance from the template
+ * @ncontrol: the initialization record
+ * @private_data: the private data to set
+ *
+ * Allocates a new struct snd_kcontrol instance and initialize from the given
+ * template. When the access field of ncontrol is 0, it's assumed as
+ * READWRITE access. When the count field is 0, it's assumes as one.
+ *
+ * Returns the pointer of the newly generated instance, or NULL on failure.
+ */
+struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol,
+ void *private_data)
+{
+ struct snd_kcontrol kctl;
+ unsigned int access;
+
+ if (snd_BUG_ON(!ncontrol || !ncontrol->info))
+ return NULL;
+ memset(&kctl, 0, sizeof(kctl));
+ kctl.id.iface = ncontrol->iface;
+ kctl.id.device = ncontrol->device;
+ kctl.id.subdevice = ncontrol->subdevice;
+ if (ncontrol->name) {
+ strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name));
+ if (strcmp(ncontrol->name, kctl.id.name) != 0)
+ snd_printk(KERN_WARNING
+ "Control name '%s' truncated to '%s'\n",
+ ncontrol->name, kctl.id.name);
+ }
+ kctl.id.index = ncontrol->index;
+ kctl.count = ncontrol->count ? ncontrol->count : 1;
+ access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
+ (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE|
+ SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE|
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK));
+ kctl.info = ncontrol->info;
+ kctl.get = ncontrol->get;
+ kctl.put = ncontrol->put;
+ kctl.tlv.p = ncontrol->tlv.p;
+ kctl.private_value = ncontrol->private_value;
+ kctl.private_data = private_data;
+ return snd_ctl_new(&kctl, access);
+}
+
+EXPORT_SYMBOL(snd_ctl_new1);
+
+/**
+ * snd_ctl_free_one - release the control instance
+ * @kcontrol: the control instance
+ *
+ * Releases the control instance created via snd_ctl_new()
+ * or snd_ctl_new1().
+ * Don't call this after the control was added to the card.
+ */
+void snd_ctl_free_one(struct snd_kcontrol *kcontrol)
+{
+ if (kcontrol) {
+ if (kcontrol->private_free)
+ kcontrol->private_free(kcontrol);
+ kfree(kcontrol);
+ }
+}
+
+EXPORT_SYMBOL(snd_ctl_free_one);
+
+static unsigned int snd_ctl_hole_check(struct snd_card *card,
+ unsigned int count)
+{
+ struct snd_kcontrol *kctl;
+
+ list_for_each_entry(kctl, &card->controls, list) {
+ if ((kctl->id.numid <= card->last_numid &&
+ kctl->id.numid + kctl->count > card->last_numid) ||
+ (kctl->id.numid <= card->last_numid + count - 1 &&
+ kctl->id.numid + kctl->count > card->last_numid + count - 1))
+ return card->last_numid = kctl->id.numid + kctl->count - 1;
+ }
+ return card->last_numid;
+}
+
+static int snd_ctl_find_hole(struct snd_card *card, unsigned int count)
+{
+ unsigned int last_numid, iter = 100000;
+
+ last_numid = card->last_numid;
+ while (last_numid != snd_ctl_hole_check(card, count)) {
+ if (--iter == 0) {
+ /* this situation is very unlikely */
+ snd_printk(KERN_ERR "unable to allocate new control numid\n");
+ return -ENOMEM;
+ }
+ last_numid = card->last_numid;
+ }
+ return 0;
+}
+
+/**
+ * snd_ctl_add - add the control instance to the card
+ * @card: the card instance
+ * @kcontrol: the control instance to add
+ *
+ * Adds the control instance created via snd_ctl_new() or
+ * snd_ctl_new1() to the given card. Assigns also an unique
+ * numid used for fast search.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ *
+ * It frees automatically the control which cannot be added.
+ */
+int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol)
+{
+ struct snd_ctl_elem_id id;
+ unsigned int idx;
+ int err = -EINVAL;
+
+ if (! kcontrol)
+ return err;
+ if (snd_BUG_ON(!card || !kcontrol->info))
+ goto error;
+ id = kcontrol->id;
+ down_write(&card->controls_rwsem);
+ if (snd_ctl_find_id(card, &id)) {
+ up_write(&card->controls_rwsem);
+ snd_printd(KERN_ERR "control %i:%i:%i:%s:%i is already present\n",
+ id.iface,
+ id.device,
+ id.subdevice,
+ id.name,
+ id.index);
+ err = -EBUSY;
+ goto error;
+ }
+ if (snd_ctl_find_hole(card, kcontrol->count) < 0) {
+ up_write(&card->controls_rwsem);
+ err = -ENOMEM;
+ goto error;
+ }
+ list_add_tail(&kcontrol->list, &card->controls);
+ card->controls_count += kcontrol->count;
+ kcontrol->id.numid = card->last_numid + 1;
+ card->last_numid += kcontrol->count;
+ up_write(&card->controls_rwsem);
+ for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id);
+ return 0;
+
+ error:
+ snd_ctl_free_one(kcontrol);
+ return err;
+}
+
+EXPORT_SYMBOL(snd_ctl_add);
+
+/**
+ * snd_ctl_remove - remove the control from the card and release it
+ * @card: the card instance
+ * @kcontrol: the control instance to remove
+ *
+ * Removes the control from the card and then releases the instance.
+ * You don't need to call snd_ctl_free_one(). You must be in
+ * the write lock - down_write(&card->controls_rwsem).
+ *
+ * Returns 0 if successful, or a negative error code on failure.
+ */
+int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol)
+{
+ struct snd_ctl_elem_id id;
+ unsigned int idx;
+
+ if (snd_BUG_ON(!card || !kcontrol))
+ return -EINVAL;
+ list_del(&kcontrol->list);
+ card->controls_count -= kcontrol->count;
+ id = kcontrol->id;
+ for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_REMOVE, &id);
+ snd_ctl_free_one(kcontrol);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ctl_remove);
+
+/**
+ * snd_ctl_remove_id - remove the control of the given id and release it
+ * @card: the card instance
+ * @id: the control id to remove
+ *
+ * Finds the control instance with the given id, removes it from the
+ * card list and releases it.
+ *
+ * Returns 0 if successful, or a negative error code on failure.
+ */
+int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id)
+{
+ struct snd_kcontrol *kctl;
+ int ret;
+
+ down_write(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, id);
+ if (kctl == NULL) {
+ up_write(&card->controls_rwsem);
+ return -ENOENT;
+ }
+ ret = snd_ctl_remove(card, kctl);
+ up_write(&card->controls_rwsem);
+ return ret;
+}
+
+EXPORT_SYMBOL(snd_ctl_remove_id);
+
+/**
+ * snd_ctl_remove_unlocked_id - remove the unlocked control of the given id and release it
+ * @file: active control handle
+ * @id: the control id to remove
+ *
+ * Finds the control instance with the given id, removes it from the
+ * card list and releases it.
+ *
+ * Returns 0 if successful, or a negative error code on failure.
+ */
+static int snd_ctl_remove_unlocked_id(struct snd_ctl_file * file,
+ struct snd_ctl_elem_id *id)
+{
+ struct snd_card *card = file->card;
+ struct snd_kcontrol *kctl;
+ int idx, ret;
+
+ down_write(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, id);
+ if (kctl == NULL) {
+ up_write(&card->controls_rwsem);
+ return -ENOENT;
+ }
+ for (idx = 0; idx < kctl->count; idx++)
+ if (kctl->vd[idx].owner != NULL && kctl->vd[idx].owner != file) {
+ up_write(&card->controls_rwsem);
+ return -EBUSY;
+ }
+ ret = snd_ctl_remove(card, kctl);
+ up_write(&card->controls_rwsem);
+ return ret;
+}
+
+/**
+ * snd_ctl_rename_id - replace the id of a control on the card
+ * @card: the card instance
+ * @src_id: the old id
+ * @dst_id: the new id
+ *
+ * Finds the control with the old id from the card, and replaces the
+ * id with the new one.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id,
+ struct snd_ctl_elem_id *dst_id)
+{
+ struct snd_kcontrol *kctl;
+
+ down_write(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, src_id);
+ if (kctl == NULL) {
+ up_write(&card->controls_rwsem);
+ return -ENOENT;
+ }
+ kctl->id = *dst_id;
+ kctl->id.numid = card->last_numid + 1;
+ card->last_numid += kctl->count;
+ up_write(&card->controls_rwsem);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ctl_rename_id);
+
+/**
+ * snd_ctl_find_numid - find the control instance with the given number-id
+ * @card: the card instance
+ * @numid: the number-id to search
+ *
+ * Finds the control instance with the given number-id from the card.
+ *
+ * Returns the pointer of the instance if found, or NULL if not.
+ *
+ * The caller must down card->controls_rwsem before calling this function
+ * (if the race condition can happen).
+ */
+struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid)
+{
+ struct snd_kcontrol *kctl;
+
+ if (snd_BUG_ON(!card || !numid))
+ return NULL;
+ list_for_each_entry(kctl, &card->controls, list) {
+ if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid)
+ return kctl;
+ }
+ return NULL;
+}
+
+EXPORT_SYMBOL(snd_ctl_find_numid);
+
+/**
+ * snd_ctl_find_id - find the control instance with the given id
+ * @card: the card instance
+ * @id: the id to search
+ *
+ * Finds the control instance with the given id from the card.
+ *
+ * Returns the pointer of the instance if found, or NULL if not.
+ *
+ * The caller must down card->controls_rwsem before calling this function
+ * (if the race condition can happen).
+ */
+struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card,
+ struct snd_ctl_elem_id *id)
+{
+ struct snd_kcontrol *kctl;
+
+ if (snd_BUG_ON(!card || !id))
+ return NULL;
+ if (id->numid != 0)
+ return snd_ctl_find_numid(card, id->numid);
+ list_for_each_entry(kctl, &card->controls, list) {
+ if (kctl->id.iface != id->iface)
+ continue;
+ if (kctl->id.device != id->device)
+ continue;
+ if (kctl->id.subdevice != id->subdevice)
+ continue;
+ if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)))
+ continue;
+ if (kctl->id.index > id->index)
+ continue;
+ if (kctl->id.index + kctl->count <= id->index)
+ continue;
+ return kctl;
+ }
+ return NULL;
+}
+
+EXPORT_SYMBOL(snd_ctl_find_id);
+
+static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl,
+ unsigned int cmd, void __user *arg)
+{
+ struct snd_ctl_card_info *info;
+
+ info = kzalloc(sizeof(*info), GFP_KERNEL);
+ if (! info)
+ return -ENOMEM;
+ down_read(&snd_ioctl_rwsem);
+ info->card = card->number;
+ strlcpy(info->id, card->id, sizeof(info->id));
+ strlcpy(info->driver, card->driver, sizeof(info->driver));
+ strlcpy(info->name, card->shortname, sizeof(info->name));
+ strlcpy(info->longname, card->longname, sizeof(info->longname));
+ strlcpy(info->mixername, card->mixername, sizeof(info->mixername));
+ strlcpy(info->components, card->components, sizeof(info->components));
+ up_read(&snd_ioctl_rwsem);
+ if (copy_to_user(arg, info, sizeof(struct snd_ctl_card_info))) {
+ kfree(info);
+ return -EFAULT;
+ }
+ kfree(info);
+ return 0;
+}
+
+static int snd_ctl_elem_list(struct snd_card *card,
+ struct snd_ctl_elem_list __user *_list)
+{
+ struct list_head *plist;
+ struct snd_ctl_elem_list list;
+ struct snd_kcontrol *kctl;
+ struct snd_ctl_elem_id *dst, *id;
+ unsigned int offset, space, first, jidx;
+
+ if (copy_from_user(&list, _list, sizeof(list)))
+ return -EFAULT;
+ offset = list.offset;
+ space = list.space;
+ first = 0;
+ /* try limit maximum space */
+ if (space > 16384)
+ return -ENOMEM;
+ if (space > 0) {
+ /* allocate temporary buffer for atomic operation */
+ dst = vmalloc(space * sizeof(struct snd_ctl_elem_id));
+ if (dst == NULL)
+ return -ENOMEM;
+ down_read(&card->controls_rwsem);
+ list.count = card->controls_count;
+ plist = card->controls.next;
+ while (plist != &card->controls) {
+ if (offset == 0)
+ break;
+ kctl = snd_kcontrol(plist);
+ if (offset < kctl->count)
+ break;
+ offset -= kctl->count;
+ plist = plist->next;
+ }
+ list.used = 0;
+ id = dst;
+ while (space > 0 && plist != &card->controls) {
+ kctl = snd_kcontrol(plist);
+ for (jidx = offset; space > 0 && jidx < kctl->count; jidx++) {
+ snd_ctl_build_ioff(id, kctl, jidx);
+ id++;
+ space--;
+ list.used++;
+ }
+ plist = plist->next;
+ offset = 0;
+ }
+ up_read(&card->controls_rwsem);
+ if (list.used > 0 &&
+ copy_to_user(list.pids, dst,
+ list.used * sizeof(struct snd_ctl_elem_id))) {
+ vfree(dst);
+ return -EFAULT;
+ }
+ vfree(dst);
+ } else {
+ down_read(&card->controls_rwsem);
+ list.count = card->controls_count;
+ up_read(&card->controls_rwsem);
+ }
+ if (copy_to_user(_list, &list, sizeof(list)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_ctl_elem_info(struct snd_ctl_file *ctl,
+ struct snd_ctl_elem_info *info)
+{
+ struct snd_card *card = ctl->card;
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_volatile *vd;
+ unsigned int index_offset;
+ int result;
+
+ down_read(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, &info->id);
+ if (kctl == NULL) {
+ up_read(&card->controls_rwsem);
+ return -ENOENT;
+ }
+#ifdef CONFIG_SND_DEBUG
+ info->access = 0;
+#endif
+ result = kctl->info(kctl, info);
+ if (result >= 0) {
+ snd_BUG_ON(info->access);
+ index_offset = snd_ctl_get_ioff(kctl, &info->id);
+ vd = &kctl->vd[index_offset];
+ snd_ctl_build_ioff(&info->id, kctl, index_offset);
+ info->access = vd->access;
+ if (vd->owner) {
+ info->access |= SNDRV_CTL_ELEM_ACCESS_LOCK;
+ if (vd->owner == ctl)
+ info->access |= SNDRV_CTL_ELEM_ACCESS_OWNER;
+ info->owner = vd->owner_pid;
+ } else {
+ info->owner = -1;
+ }
+ }
+ up_read(&card->controls_rwsem);
+ return result;
+}
+
+static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl,
+ struct snd_ctl_elem_info __user *_info)
+{
+ struct snd_ctl_elem_info info;
+ int result;
+
+ if (copy_from_user(&info, _info, sizeof(info)))
+ return -EFAULT;
+ snd_power_lock(ctl->card);
+ result = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0);
+ if (result >= 0)
+ result = snd_ctl_elem_info(ctl, &info);
+ snd_power_unlock(ctl->card);
+ if (result >= 0)
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return result;
+}
+
+static int snd_ctl_elem_read(struct snd_card *card,
+ struct snd_ctl_elem_value *control)
+{
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_volatile *vd;
+ unsigned int index_offset;
+ int result;
+
+ down_read(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, &control->id);
+ if (kctl == NULL) {
+ result = -ENOENT;
+ } else {
+ index_offset = snd_ctl_get_ioff(kctl, &control->id);
+ vd = &kctl->vd[index_offset];
+ if ((vd->access & SNDRV_CTL_ELEM_ACCESS_READ) &&
+ kctl->get != NULL) {
+ snd_ctl_build_ioff(&control->id, kctl, index_offset);
+ result = kctl->get(kctl, control);
+ } else
+ result = -EPERM;
+ }
+ up_read(&card->controls_rwsem);
+ return result;
+}
+
+static int snd_ctl_elem_read_user(struct snd_card *card,
+ struct snd_ctl_elem_value __user *_control)
+{
+ struct snd_ctl_elem_value *control;
+ int result;
+
+ control = kmalloc(sizeof(*control), GFP_KERNEL);
+ if (control == NULL)
+ return -ENOMEM;
+ if (copy_from_user(control, _control, sizeof(*control))) {
+ kfree(control);
+ return -EFAULT;
+ }
+ snd_power_lock(card);
+ result = snd_power_wait(card, SNDRV_CTL_POWER_D0);
+ if (result >= 0)
+ result = snd_ctl_elem_read(card, control);
+ snd_power_unlock(card);
+ if (result >= 0)
+ if (copy_to_user(_control, control, sizeof(*control)))
+ result = -EFAULT;
+ kfree(control);
+ return result;
+}
+
+static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file,
+ struct snd_ctl_elem_value *control)
+{
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_volatile *vd;
+ unsigned int index_offset;
+ int result;
+
+ down_read(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, &control->id);
+ if (kctl == NULL) {
+ result = -ENOENT;
+ } else {
+ index_offset = snd_ctl_get_ioff(kctl, &control->id);
+ vd = &kctl->vd[index_offset];
+ if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) ||
+ kctl->put == NULL ||
+ (file && vd->owner && vd->owner != file)) {
+ result = -EPERM;
+ } else {
+ snd_ctl_build_ioff(&control->id, kctl, index_offset);
+ result = kctl->put(kctl, control);
+ }
+ if (result > 0) {
+ up_read(&card->controls_rwsem);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &control->id);
+ return 0;
+ }
+ }
+ up_read(&card->controls_rwsem);
+ return result;
+}
+
+static int snd_ctl_elem_write_user(struct snd_ctl_file *file,
+ struct snd_ctl_elem_value __user *_control)
+{
+ struct snd_ctl_elem_value *control;
+ struct snd_card *card;
+ int result;
+
+ control = kmalloc(sizeof(*control), GFP_KERNEL);
+ if (control == NULL)
+ return -ENOMEM;
+ if (copy_from_user(control, _control, sizeof(*control))) {
+ kfree(control);
+ return -EFAULT;
+ }
+ card = file->card;
+ snd_power_lock(card);
+ result = snd_power_wait(card, SNDRV_CTL_POWER_D0);
+ if (result >= 0)
+ result = snd_ctl_elem_write(card, file, control);
+ snd_power_unlock(card);
+ if (result >= 0)
+ if (copy_to_user(_control, control, sizeof(*control)))
+ result = -EFAULT;
+ kfree(control);
+ return result;
+}
+
+static int snd_ctl_elem_lock(struct snd_ctl_file *file,
+ struct snd_ctl_elem_id __user *_id)
+{
+ struct snd_card *card = file->card;
+ struct snd_ctl_elem_id id;
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_volatile *vd;
+ int result;
+
+ if (copy_from_user(&id, _id, sizeof(id)))
+ return -EFAULT;
+ down_write(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, &id);
+ if (kctl == NULL) {
+ result = -ENOENT;
+ } else {
+ vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)];
+ if (vd->owner != NULL)
+ result = -EBUSY;
+ else {
+ vd->owner = file;
+ vd->owner_pid = current->pid;
+ result = 0;
+ }
+ }
+ up_write(&card->controls_rwsem);
+ return result;
+}
+
+static int snd_ctl_elem_unlock(struct snd_ctl_file *file,
+ struct snd_ctl_elem_id __user *_id)
+{
+ struct snd_card *card = file->card;
+ struct snd_ctl_elem_id id;
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_volatile *vd;
+ int result;
+
+ if (copy_from_user(&id, _id, sizeof(id)))
+ return -EFAULT;
+ down_write(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, &id);
+ if (kctl == NULL) {
+ result = -ENOENT;
+ } else {
+ vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)];
+ if (vd->owner == NULL)
+ result = -EINVAL;
+ else if (vd->owner != file)
+ result = -EPERM;
+ else {
+ vd->owner = NULL;
+ vd->owner_pid = 0;
+ result = 0;
+ }
+ }
+ up_write(&card->controls_rwsem);
+ return result;
+}
+
+struct user_element {
+ struct snd_ctl_elem_info info;
+ void *elem_data; /* element data */
+ unsigned long elem_data_size; /* size of element data in bytes */
+ void *tlv_data; /* TLV data */
+ unsigned long tlv_data_size; /* TLV data size */
+ void *priv_data; /* private data (like strings for enumerated type) */
+ unsigned long priv_data_size; /* size of private data in bytes */
+};
+
+static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct user_element *ue = kcontrol->private_data;
+
+ *uinfo = ue->info;
+ return 0;
+}
+
+static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct user_element *ue = kcontrol->private_data;
+
+ memcpy(&ucontrol->value, ue->elem_data, ue->elem_data_size);
+ return 0;
+}
+
+static int snd_ctl_elem_user_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int change;
+ struct user_element *ue = kcontrol->private_data;
+
+ change = memcmp(&ucontrol->value, ue->elem_data, ue->elem_data_size) != 0;
+ if (change)
+ memcpy(ue->elem_data, &ucontrol->value, ue->elem_data_size);
+ return change;
+}
+
+static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kcontrol,
+ int op_flag,
+ unsigned int size,
+ unsigned int __user *tlv)
+{
+ struct user_element *ue = kcontrol->private_data;
+ int change = 0;
+ void *new_data;
+
+ if (op_flag > 0) {
+ if (size > 1024 * 128) /* sane value */
+ return -EINVAL;
+ new_data = kmalloc(size, GFP_KERNEL);
+ if (new_data == NULL)
+ return -ENOMEM;
+ if (copy_from_user(new_data, tlv, size)) {
+ kfree(new_data);
+ return -EFAULT;
+ }
+ change = ue->tlv_data_size != size;
+ if (!change)
+ change = memcmp(ue->tlv_data, new_data, size);
+ kfree(ue->tlv_data);
+ ue->tlv_data = new_data;
+ ue->tlv_data_size = size;
+ } else {
+ if (! ue->tlv_data_size || ! ue->tlv_data)
+ return -ENXIO;
+ if (size < ue->tlv_data_size)
+ return -ENOSPC;
+ if (copy_to_user(tlv, ue->tlv_data, ue->tlv_data_size))
+ return -EFAULT;
+ }
+ return change;
+}
+
+static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol)
+{
+ struct user_element *ue = kcontrol->private_data;
+ if (ue->tlv_data)
+ kfree(ue->tlv_data);
+ kfree(ue);
+}
+
+static int snd_ctl_elem_add(struct snd_ctl_file *file,
+ struct snd_ctl_elem_info *info, int replace)
+{
+ struct snd_card *card = file->card;
+ struct snd_kcontrol kctl, *_kctl;
+ unsigned int access;
+ long private_size;
+ struct user_element *ue;
+ int idx, err;
+
+ if (card->user_ctl_count >= MAX_USER_CONTROLS)
+ return -ENOMEM;
+ if (info->count > 1024)
+ return -EINVAL;
+ access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
+ (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE|
+ SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE));
+ info->id.numid = 0;
+ memset(&kctl, 0, sizeof(kctl));
+ down_write(&card->controls_rwsem);
+ _kctl = snd_ctl_find_id(card, &info->id);
+ err = 0;
+ if (_kctl) {
+ if (replace)
+ err = snd_ctl_remove(card, _kctl);
+ else
+ err = -EBUSY;
+ } else {
+ if (replace)
+ err = -ENOENT;
+ }
+ up_write(&card->controls_rwsem);
+ if (err < 0)
+ return err;
+ memcpy(&kctl.id, &info->id, sizeof(info->id));
+ kctl.count = info->owner ? info->owner : 1;
+ access |= SNDRV_CTL_ELEM_ACCESS_USER;
+ kctl.info = snd_ctl_elem_user_info;
+ if (access & SNDRV_CTL_ELEM_ACCESS_READ)
+ kctl.get = snd_ctl_elem_user_get;
+ if (access & SNDRV_CTL_ELEM_ACCESS_WRITE)
+ kctl.put = snd_ctl_elem_user_put;
+ if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) {
+ kctl.tlv.c = snd_ctl_elem_user_tlv;
+ access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+ }
+ switch (info->type) {
+ case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+ case SNDRV_CTL_ELEM_TYPE_INTEGER:
+ private_size = sizeof(long);
+ if (info->count > 128)
+ return -EINVAL;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+ private_size = sizeof(long long);
+ if (info->count > 64)
+ return -EINVAL;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_BYTES:
+ private_size = sizeof(unsigned char);
+ if (info->count > 512)
+ return -EINVAL;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_IEC958:
+ private_size = sizeof(struct snd_aes_iec958);
+ if (info->count != 1)
+ return -EINVAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ private_size *= info->count;
+ ue = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL);
+ if (ue == NULL)
+ return -ENOMEM;
+ ue->info = *info;
+ ue->info.access = 0;
+ ue->elem_data = (char *)ue + sizeof(*ue);
+ ue->elem_data_size = private_size;
+ kctl.private_free = snd_ctl_elem_user_free;
+ _kctl = snd_ctl_new(&kctl, access);
+ if (_kctl == NULL) {
+ kfree(ue);
+ return -ENOMEM;
+ }
+ _kctl->private_data = ue;
+ for (idx = 0; idx < _kctl->count; idx++)
+ _kctl->vd[idx].owner = file;
+ err = snd_ctl_add(card, _kctl);
+ if (err < 0)
+ return err;
+
+ down_write(&card->controls_rwsem);
+ card->user_ctl_count++;
+ up_write(&card->controls_rwsem);
+
+ return 0;
+}
+
+static int snd_ctl_elem_add_user(struct snd_ctl_file *file,
+ struct snd_ctl_elem_info __user *_info, int replace)
+{
+ struct snd_ctl_elem_info info;
+ if (copy_from_user(&info, _info, sizeof(info)))
+ return -EFAULT;
+ return snd_ctl_elem_add(file, &info, replace);
+}
+
+static int snd_ctl_elem_remove(struct snd_ctl_file *file,
+ struct snd_ctl_elem_id __user *_id)
+{
+ struct snd_ctl_elem_id id;
+ int err;
+
+ if (copy_from_user(&id, _id, sizeof(id)))
+ return -EFAULT;
+ err = snd_ctl_remove_unlocked_id(file, &id);
+ if (! err) {
+ struct snd_card *card = file->card;
+ down_write(&card->controls_rwsem);
+ card->user_ctl_count--;
+ up_write(&card->controls_rwsem);
+ }
+ return err;
+}
+
+static int snd_ctl_subscribe_events(struct snd_ctl_file *file, int __user *ptr)
+{
+ int subscribe;
+ if (get_user(subscribe, ptr))
+ return -EFAULT;
+ if (subscribe < 0) {
+ subscribe = file->subscribed;
+ if (put_user(subscribe, ptr))
+ return -EFAULT;
+ return 0;
+ }
+ if (subscribe) {
+ file->subscribed = 1;
+ return 0;
+ } else if (file->subscribed) {
+ snd_ctl_empty_read_queue(file);
+ file->subscribed = 0;
+ }
+ return 0;
+}
+
+static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file,
+ struct snd_ctl_tlv __user *_tlv,
+ int op_flag)
+{
+ struct snd_card *card = file->card;
+ struct snd_ctl_tlv tlv;
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_volatile *vd;
+ unsigned int len;
+ int err = 0;
+
+ if (copy_from_user(&tlv, _tlv, sizeof(tlv)))
+ return -EFAULT;
+ if (tlv.length < sizeof(unsigned int) * 3)
+ return -EINVAL;
+ down_read(&card->controls_rwsem);
+ kctl = snd_ctl_find_numid(card, tlv.numid);
+ if (kctl == NULL) {
+ err = -ENOENT;
+ goto __kctl_end;
+ }
+ if (kctl->tlv.p == NULL) {
+ err = -ENXIO;
+ goto __kctl_end;
+ }
+ vd = &kctl->vd[tlv.numid - kctl->id.numid];
+ if ((op_flag == 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) == 0) ||
+ (op_flag > 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) == 0) ||
+ (op_flag < 0 && (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND) == 0)) {
+ err = -ENXIO;
+ goto __kctl_end;
+ }
+ if (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
+ if (file && vd->owner != NULL && vd->owner != file) {
+ err = -EPERM;
+ goto __kctl_end;
+ }
+ err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv);
+ if (err > 0) {
+ up_read(&card->controls_rwsem);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_TLV, &kctl->id);
+ return 0;
+ }
+ } else {
+ if (op_flag) {
+ err = -ENXIO;
+ goto __kctl_end;
+ }
+ len = kctl->tlv.p[1] + 2 * sizeof(unsigned int);
+ if (tlv.length < len) {
+ err = -ENOMEM;
+ goto __kctl_end;
+ }
+ if (copy_to_user(_tlv->tlv, kctl->tlv.p, len))
+ err = -EFAULT;
+ }
+ __kctl_end:
+ up_read(&card->controls_rwsem);
+ return err;
+}
+
+static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_ctl_file *ctl;
+ struct snd_card *card;
+ struct snd_kctl_ioctl *p;
+ void __user *argp = (void __user *)arg;
+ int __user *ip = argp;
+ int err;
+
+ ctl = file->private_data;
+ card = ctl->card;
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ switch (cmd) {
+ case SNDRV_CTL_IOCTL_PVERSION:
+ return put_user(SNDRV_CTL_VERSION, ip) ? -EFAULT : 0;
+ case SNDRV_CTL_IOCTL_CARD_INFO:
+ return snd_ctl_card_info(card, ctl, cmd, argp);
+ case SNDRV_CTL_IOCTL_ELEM_LIST:
+ return snd_ctl_elem_list(card, argp);
+ case SNDRV_CTL_IOCTL_ELEM_INFO:
+ return snd_ctl_elem_info_user(ctl, argp);
+ case SNDRV_CTL_IOCTL_ELEM_READ:
+ return snd_ctl_elem_read_user(card, argp);
+ case SNDRV_CTL_IOCTL_ELEM_WRITE:
+ return snd_ctl_elem_write_user(ctl, argp);
+ case SNDRV_CTL_IOCTL_ELEM_LOCK:
+ return snd_ctl_elem_lock(ctl, argp);
+ case SNDRV_CTL_IOCTL_ELEM_UNLOCK:
+ return snd_ctl_elem_unlock(ctl, argp);
+ case SNDRV_CTL_IOCTL_ELEM_ADD:
+ return snd_ctl_elem_add_user(ctl, argp, 0);
+ case SNDRV_CTL_IOCTL_ELEM_REPLACE:
+ return snd_ctl_elem_add_user(ctl, argp, 1);
+ case SNDRV_CTL_IOCTL_ELEM_REMOVE:
+ return snd_ctl_elem_remove(ctl, argp);
+ case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS:
+ return snd_ctl_subscribe_events(ctl, ip);
+ case SNDRV_CTL_IOCTL_TLV_READ:
+ return snd_ctl_tlv_ioctl(ctl, argp, 0);
+ case SNDRV_CTL_IOCTL_TLV_WRITE:
+ return snd_ctl_tlv_ioctl(ctl, argp, 1);
+ case SNDRV_CTL_IOCTL_TLV_COMMAND:
+ return snd_ctl_tlv_ioctl(ctl, argp, -1);
+ case SNDRV_CTL_IOCTL_POWER:
+ return -ENOPROTOOPT;
+ case SNDRV_CTL_IOCTL_POWER_STATE:
+#ifdef CONFIG_PM
+ return put_user(card->power_state, ip) ? -EFAULT : 0;
+#else
+ return put_user(SNDRV_CTL_POWER_D0, ip) ? -EFAULT : 0;
+#endif
+ }
+ down_read(&snd_ioctl_rwsem);
+ list_for_each_entry(p, &snd_control_ioctls, list) {
+ err = p->fioctl(card, ctl, cmd, arg);
+ if (err != -ENOIOCTLCMD) {
+ up_read(&snd_ioctl_rwsem);
+ return err;
+ }
+ }
+ up_read(&snd_ioctl_rwsem);
+ snd_printdd("unknown ioctl = 0x%x\n", cmd);
+ return -ENOTTY;
+}
+
+static ssize_t snd_ctl_read(struct file *file, char __user *buffer,
+ size_t count, loff_t * offset)
+{
+ struct snd_ctl_file *ctl;
+ int err = 0;
+ ssize_t result = 0;
+
+ ctl = file->private_data;
+ if (snd_BUG_ON(!ctl || !ctl->card))
+ return -ENXIO;
+ if (!ctl->subscribed)
+ return -EBADFD;
+ if (count < sizeof(struct snd_ctl_event))
+ return -EINVAL;
+ spin_lock_irq(&ctl->read_lock);
+ while (count >= sizeof(struct snd_ctl_event)) {
+ struct snd_ctl_event ev;
+ struct snd_kctl_event *kev;
+ while (list_empty(&ctl->events)) {
+ wait_queue_t wait;
+ if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) {
+ err = -EAGAIN;
+ goto __end_lock;
+ }
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&ctl->change_sleep, &wait);
+ set_current_state(TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&ctl->read_lock);
+ schedule();
+ remove_wait_queue(&ctl->change_sleep, &wait);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ spin_lock_irq(&ctl->read_lock);
+ }
+ kev = snd_kctl_event(ctl->events.next);
+ ev.type = SNDRV_CTL_EVENT_ELEM;
+ ev.data.elem.mask = kev->mask;
+ ev.data.elem.id = kev->id;
+ list_del(&kev->list);
+ spin_unlock_irq(&ctl->read_lock);
+ kfree(kev);
+ if (copy_to_user(buffer, &ev, sizeof(struct snd_ctl_event))) {
+ err = -EFAULT;
+ goto __end;
+ }
+ spin_lock_irq(&ctl->read_lock);
+ buffer += sizeof(struct snd_ctl_event);
+ count -= sizeof(struct snd_ctl_event);
+ result += sizeof(struct snd_ctl_event);
+ }
+ __end_lock:
+ spin_unlock_irq(&ctl->read_lock);
+ __end:
+ return result > 0 ? result : err;
+}
+
+static unsigned int snd_ctl_poll(struct file *file, poll_table * wait)
+{
+ unsigned int mask;
+ struct snd_ctl_file *ctl;
+
+ ctl = file->private_data;
+ if (!ctl->subscribed)
+ return 0;
+ poll_wait(file, &ctl->change_sleep, wait);
+
+ mask = 0;
+ if (!list_empty(&ctl->events))
+ mask |= POLLIN | POLLRDNORM;
+
+ return mask;
+}
+
+/*
+ * register the device-specific control-ioctls.
+ * called from each device manager like pcm.c, hwdep.c, etc.
+ */
+static int _snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn, struct list_head *lists)
+{
+ struct snd_kctl_ioctl *pn;
+
+ pn = kzalloc(sizeof(struct snd_kctl_ioctl), GFP_KERNEL);
+ if (pn == NULL)
+ return -ENOMEM;
+ pn->fioctl = fcn;
+ down_write(&snd_ioctl_rwsem);
+ list_add_tail(&pn->list, lists);
+ up_write(&snd_ioctl_rwsem);
+ return 0;
+}
+
+int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn)
+{
+ return _snd_ctl_register_ioctl(fcn, &snd_control_ioctls);
+}
+
+EXPORT_SYMBOL(snd_ctl_register_ioctl);
+
+#ifdef CONFIG_COMPAT
+int snd_ctl_register_ioctl_compat(snd_kctl_ioctl_func_t fcn)
+{
+ return _snd_ctl_register_ioctl(fcn, &snd_control_compat_ioctls);
+}
+
+EXPORT_SYMBOL(snd_ctl_register_ioctl_compat);
+#endif
+
+/*
+ * de-register the device-specific control-ioctls.
+ */
+static int _snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn,
+ struct list_head *lists)
+{
+ struct snd_kctl_ioctl *p;
+
+ if (snd_BUG_ON(!fcn))
+ return -EINVAL;
+ down_write(&snd_ioctl_rwsem);
+ list_for_each_entry(p, lists, list) {
+ if (p->fioctl == fcn) {
+ list_del(&p->list);
+ up_write(&snd_ioctl_rwsem);
+ kfree(p);
+ return 0;
+ }
+ }
+ up_write(&snd_ioctl_rwsem);
+ snd_BUG();
+ return -EINVAL;
+}
+
+int snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn)
+{
+ return _snd_ctl_unregister_ioctl(fcn, &snd_control_ioctls);
+}
+
+EXPORT_SYMBOL(snd_ctl_unregister_ioctl);
+
+#ifdef CONFIG_COMPAT
+int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn)
+{
+ return _snd_ctl_unregister_ioctl(fcn, &snd_control_compat_ioctls);
+}
+
+EXPORT_SYMBOL(snd_ctl_unregister_ioctl_compat);
+#endif
+
+static int snd_ctl_fasync(int fd, struct file * file, int on)
+{
+ struct snd_ctl_file *ctl;
+ int err;
+ ctl = file->private_data;
+ err = fasync_helper(fd, file, on, &ctl->fasync);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * ioctl32 compat
+ */
+#ifdef CONFIG_COMPAT
+#include "control_compat.c"
+#else
+#define snd_ctl_ioctl_compat NULL
+#endif
+
+/*
+ * INIT PART
+ */
+
+static const struct file_operations snd_ctl_f_ops =
+{
+ .owner = THIS_MODULE,
+ .read = snd_ctl_read,
+ .open = snd_ctl_open,
+ .release = snd_ctl_release,
+ .poll = snd_ctl_poll,
+ .unlocked_ioctl = snd_ctl_ioctl,
+ .compat_ioctl = snd_ctl_ioctl_compat,
+ .fasync = snd_ctl_fasync,
+};
+
+/*
+ * registration of the control device
+ */
+static int snd_ctl_dev_register(struct snd_device *device)
+{
+ struct snd_card *card = device->device_data;
+ int err, cardnum;
+ char name[16];
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ cardnum = card->number;
+ if (snd_BUG_ON(cardnum < 0 || cardnum >= SNDRV_CARDS))
+ return -ENXIO;
+ sprintf(name, "controlC%i", cardnum);
+ if ((err = snd_register_device(SNDRV_DEVICE_TYPE_CONTROL, card, -1,
+ &snd_ctl_f_ops, card, name)) < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * disconnection of the control device
+ */
+static int snd_ctl_dev_disconnect(struct snd_device *device)
+{
+ struct snd_card *card = device->device_data;
+ struct snd_ctl_file *ctl;
+ int err, cardnum;
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ cardnum = card->number;
+ if (snd_BUG_ON(cardnum < 0 || cardnum >= SNDRV_CARDS))
+ return -ENXIO;
+
+ read_lock(&card->ctl_files_rwlock);
+ list_for_each_entry(ctl, &card->ctl_files, list) {
+ wake_up(&ctl->change_sleep);
+ kill_fasync(&ctl->fasync, SIGIO, POLL_ERR);
+ }
+ read_unlock(&card->ctl_files_rwlock);
+
+ if ((err = snd_unregister_device(SNDRV_DEVICE_TYPE_CONTROL,
+ card, -1)) < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * free all controls
+ */
+static int snd_ctl_dev_free(struct snd_device *device)
+{
+ struct snd_card *card = device->device_data;
+ struct snd_kcontrol *control;
+
+ down_write(&card->controls_rwsem);
+ while (!list_empty(&card->controls)) {
+ control = snd_kcontrol(card->controls.next);
+ snd_ctl_remove(card, control);
+ }
+ up_write(&card->controls_rwsem);
+ return 0;
+}
+
+/*
+ * create control core:
+ * called from init.c
+ */
+int snd_ctl_create(struct snd_card *card)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ctl_dev_free,
+ .dev_register = snd_ctl_dev_register,
+ .dev_disconnect = snd_ctl_dev_disconnect,
+ };
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ return snd_device_new(card, SNDRV_DEV_CONTROL, card, &ops);
+}
+
+/*
+ * Frequently used control callbacks
+ */
+int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ctl_boolean_mono_info);
+
+int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ctl_boolean_stereo_info);
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
new file mode 100644
index 0000000..368dc9c
--- /dev/null
+++ b/sound/core/control_compat.c
@@ -0,0 +1,443 @@
+/*
+ * compat ioctls for control API
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/* this file included from control.c */
+
+#include <linux/compat.h>
+
+struct snd_ctl_elem_list32 {
+ u32 offset;
+ u32 space;
+ u32 used;
+ u32 count;
+ u32 pids;
+ unsigned char reserved[50];
+} /* don't set packed attribute here */;
+
+static int snd_ctl_elem_list_compat(struct snd_card *card,
+ struct snd_ctl_elem_list32 __user *data32)
+{
+ struct snd_ctl_elem_list __user *data;
+ compat_caddr_t ptr;
+ int err;
+
+ data = compat_alloc_user_space(sizeof(*data));
+
+ /* offset, space, used, count */
+ if (copy_in_user(data, data32, 4 * sizeof(u32)))
+ return -EFAULT;
+ /* pids */
+ if (get_user(ptr, &data32->pids) ||
+ put_user(compat_ptr(ptr), &data->pids))
+ return -EFAULT;
+ err = snd_ctl_elem_list(card, data);
+ if (err < 0)
+ return err;
+ /* copy the result */
+ if (copy_in_user(data32, data, 4 * sizeof(u32)))
+ return -EFAULT;
+ return 0;
+}
+
+/*
+ * control element info
+ * it uses union, so the things are not easy..
+ */
+
+struct snd_ctl_elem_info32 {
+ struct snd_ctl_elem_id id; // the size of struct is same
+ s32 type;
+ u32 access;
+ u32 count;
+ s32 owner;
+ union {
+ struct {
+ s32 min;
+ s32 max;
+ s32 step;
+ } integer;
+ struct {
+ u64 min;
+ u64 max;
+ u64 step;
+ } integer64;
+ struct {
+ u32 items;
+ u32 item;
+ char name[64];
+ } enumerated;
+ unsigned char reserved[128];
+ } value;
+ unsigned char reserved[64];
+} __attribute__((packed));
+
+static int snd_ctl_elem_info_compat(struct snd_ctl_file *ctl,
+ struct snd_ctl_elem_info32 __user *data32)
+{
+ struct snd_ctl_elem_info *data;
+ int err;
+
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (! data)
+ return -ENOMEM;
+
+ err = -EFAULT;
+ /* copy id */
+ if (copy_from_user(&data->id, &data32->id, sizeof(data->id)))
+ goto error;
+ /* we need to copy the item index.
+ * hope this doesn't break anything..
+ */
+ if (get_user(data->value.enumerated.item, &data32->value.enumerated.item))
+ goto error;
+
+ snd_power_lock(ctl->card);
+ err = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0);
+ if (err >= 0)
+ err = snd_ctl_elem_info(ctl, data);
+ snd_power_unlock(ctl->card);
+
+ if (err < 0)
+ goto error;
+ /* restore info to 32bit */
+ err = -EFAULT;
+ /* id, type, access, count */
+ if (copy_to_user(&data32->id, &data->id, sizeof(data->id)) ||
+ copy_to_user(&data32->type, &data->type, 3 * sizeof(u32)))
+ goto error;
+ if (put_user(data->owner, &data32->owner))
+ goto error;
+ switch (data->type) {
+ case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+ case SNDRV_CTL_ELEM_TYPE_INTEGER:
+ if (put_user(data->value.integer.min, &data32->value.integer.min) ||
+ put_user(data->value.integer.max, &data32->value.integer.max) ||
+ put_user(data->value.integer.step, &data32->value.integer.step))
+ goto error;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+ if (copy_to_user(&data32->value.integer64,
+ &data->value.integer64,
+ sizeof(data->value.integer64)))
+ goto error;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_ENUMERATED:
+ if (copy_to_user(&data32->value.enumerated,
+ &data->value.enumerated,
+ sizeof(data->value.enumerated)))
+ goto error;
+ break;
+ default:
+ break;
+ }
+ err = 0;
+ error:
+ kfree(data);
+ return err;
+}
+
+/* read / write */
+struct snd_ctl_elem_value32 {
+ struct snd_ctl_elem_id id;
+ unsigned int indirect; /* bit-field causes misalignment */
+ union {
+ s32 integer[128];
+ unsigned char data[512];
+#ifndef CONFIG_X86_64
+ s64 integer64[64];
+#endif
+ } value;
+ unsigned char reserved[128];
+};
+
+
+/* get the value type and count of the control */
+static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id,
+ int *countp)
+{
+ struct snd_kcontrol *kctl;
+ struct snd_ctl_elem_info *info;
+ int err;
+
+ down_read(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, id);
+ if (! kctl) {
+ up_read(&card->controls_rwsem);
+ return -ENXIO;
+ }
+ info = kzalloc(sizeof(*info), GFP_KERNEL);
+ if (info == NULL) {
+ up_read(&card->controls_rwsem);
+ return -ENOMEM;
+ }
+ info->id = *id;
+ err = kctl->info(kctl, info);
+ up_read(&card->controls_rwsem);
+ if (err >= 0) {
+ err = info->type;
+ *countp = info->count;
+ }
+ kfree(info);
+ return err;
+}
+
+static int get_elem_size(int type, int count)
+{
+ switch (type) {
+ case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+ return sizeof(s64) * count;
+ case SNDRV_CTL_ELEM_TYPE_ENUMERATED:
+ return sizeof(int) * count;
+ case SNDRV_CTL_ELEM_TYPE_BYTES:
+ return 512;
+ case SNDRV_CTL_ELEM_TYPE_IEC958:
+ return sizeof(struct snd_aes_iec958);
+ default:
+ return -1;
+ }
+}
+
+static int copy_ctl_value_from_user(struct snd_card *card,
+ struct snd_ctl_elem_value *data,
+ struct snd_ctl_elem_value32 __user *data32,
+ int *typep, int *countp)
+{
+ int i, type, size;
+ int uninitialized_var(count);
+ unsigned int indirect;
+
+ if (copy_from_user(&data->id, &data32->id, sizeof(data->id)))
+ return -EFAULT;
+ if (get_user(indirect, &data32->indirect))
+ return -EFAULT;
+ if (indirect)
+ return -EINVAL;
+ type = get_ctl_type(card, &data->id, &count);
+ if (type < 0)
+ return type;
+
+ if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN ||
+ type == SNDRV_CTL_ELEM_TYPE_INTEGER) {
+ for (i = 0; i < count; i++) {
+ int val;
+ if (get_user(val, &data32->value.integer[i]))
+ return -EFAULT;
+ data->value.integer.value[i] = val;
+ }
+ } else {
+ size = get_elem_size(type, count);
+ if (size < 0) {
+ printk(KERN_ERR "snd_ioctl32_ctl_elem_value: unknown type %d\n", type);
+ return -EINVAL;
+ }
+ if (copy_from_user(data->value.bytes.data,
+ data32->value.data, size))
+ return -EFAULT;
+ }
+
+ *typep = type;
+ *countp = count;
+ return 0;
+}
+
+/* restore the value to 32bit */
+static int copy_ctl_value_to_user(struct snd_ctl_elem_value32 __user *data32,
+ struct snd_ctl_elem_value *data,
+ int type, int count)
+{
+ int i, size;
+
+ if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN ||
+ type == SNDRV_CTL_ELEM_TYPE_INTEGER) {
+ for (i = 0; i < count; i++) {
+ int val;
+ val = data->value.integer.value[i];
+ if (put_user(val, &data32->value.integer[i]))
+ return -EFAULT;
+ }
+ } else {
+ size = get_elem_size(type, count);
+ if (copy_to_user(data32->value.data,
+ data->value.bytes.data, size))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static int snd_ctl_elem_read_user_compat(struct snd_card *card,
+ struct snd_ctl_elem_value32 __user *data32)
+{
+ struct snd_ctl_elem_value *data;
+ int err, type, count;
+
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (data == NULL)
+ return -ENOMEM;
+
+ if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0)
+ goto error;
+
+ snd_power_lock(card);
+ err = snd_power_wait(card, SNDRV_CTL_POWER_D0);
+ if (err >= 0)
+ err = snd_ctl_elem_read(card, data);
+ snd_power_unlock(card);
+ if (err >= 0)
+ err = copy_ctl_value_to_user(data32, data, type, count);
+ error:
+ kfree(data);
+ return err;
+}
+
+static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file,
+ struct snd_ctl_elem_value32 __user *data32)
+{
+ struct snd_ctl_elem_value *data;
+ struct snd_card *card = file->card;
+ int err, type, count;
+
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (data == NULL)
+ return -ENOMEM;
+
+ if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0)
+ goto error;
+
+ snd_power_lock(card);
+ err = snd_power_wait(card, SNDRV_CTL_POWER_D0);
+ if (err >= 0)
+ err = snd_ctl_elem_write(card, file, data);
+ snd_power_unlock(card);
+ if (err >= 0)
+ err = copy_ctl_value_to_user(data32, data, type, count);
+ error:
+ kfree(data);
+ return err;
+}
+
+/* add or replace a user control */
+static int snd_ctl_elem_add_compat(struct snd_ctl_file *file,
+ struct snd_ctl_elem_info32 __user *data32,
+ int replace)
+{
+ struct snd_ctl_elem_info *data;
+ int err;
+
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (! data)
+ return -ENOMEM;
+
+ err = -EFAULT;
+ /* id, type, access, count */ \
+ if (copy_from_user(&data->id, &data32->id, sizeof(data->id)) ||
+ copy_from_user(&data->type, &data32->type, 3 * sizeof(u32)))
+ goto error;
+ if (get_user(data->owner, &data32->owner) ||
+ get_user(data->type, &data32->type))
+ goto error;
+ switch (data->type) {
+ case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+ case SNDRV_CTL_ELEM_TYPE_INTEGER:
+ if (get_user(data->value.integer.min, &data32->value.integer.min) ||
+ get_user(data->value.integer.max, &data32->value.integer.max) ||
+ get_user(data->value.integer.step, &data32->value.integer.step))
+ goto error;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_INTEGER64:
+ if (copy_from_user(&data->value.integer64,
+ &data32->value.integer64,
+ sizeof(data->value.integer64)))
+ goto error;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_ENUMERATED:
+ if (copy_from_user(&data->value.enumerated,
+ &data32->value.enumerated,
+ sizeof(data->value.enumerated)))
+ goto error;
+ break;
+ default:
+ break;
+ }
+ err = snd_ctl_elem_add(file, data, replace);
+ error:
+ kfree(data);
+ return err;
+}
+
+enum {
+ SNDRV_CTL_IOCTL_ELEM_LIST32 = _IOWR('U', 0x10, struct snd_ctl_elem_list32),
+ SNDRV_CTL_IOCTL_ELEM_INFO32 = _IOWR('U', 0x11, struct snd_ctl_elem_info32),
+ SNDRV_CTL_IOCTL_ELEM_READ32 = _IOWR('U', 0x12, struct snd_ctl_elem_value32),
+ SNDRV_CTL_IOCTL_ELEM_WRITE32 = _IOWR('U', 0x13, struct snd_ctl_elem_value32),
+ SNDRV_CTL_IOCTL_ELEM_ADD32 = _IOWR('U', 0x17, struct snd_ctl_elem_info32),
+ SNDRV_CTL_IOCTL_ELEM_REPLACE32 = _IOWR('U', 0x18, struct snd_ctl_elem_info32),
+};
+
+static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_ctl_file *ctl;
+ struct snd_kctl_ioctl *p;
+ void __user *argp = compat_ptr(arg);
+ int err;
+
+ ctl = file->private_data;
+ if (snd_BUG_ON(!ctl || !ctl->card))
+ return -ENXIO;
+
+ switch (cmd) {
+ case SNDRV_CTL_IOCTL_PVERSION:
+ case SNDRV_CTL_IOCTL_CARD_INFO:
+ case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS:
+ case SNDRV_CTL_IOCTL_POWER:
+ case SNDRV_CTL_IOCTL_POWER_STATE:
+ case SNDRV_CTL_IOCTL_ELEM_LOCK:
+ case SNDRV_CTL_IOCTL_ELEM_UNLOCK:
+ case SNDRV_CTL_IOCTL_ELEM_REMOVE:
+ case SNDRV_CTL_IOCTL_TLV_READ:
+ case SNDRV_CTL_IOCTL_TLV_WRITE:
+ case SNDRV_CTL_IOCTL_TLV_COMMAND:
+ return snd_ctl_ioctl(file, cmd, (unsigned long)argp);
+ case SNDRV_CTL_IOCTL_ELEM_LIST32:
+ return snd_ctl_elem_list_compat(ctl->card, argp);
+ case SNDRV_CTL_IOCTL_ELEM_INFO32:
+ return snd_ctl_elem_info_compat(ctl, argp);
+ case SNDRV_CTL_IOCTL_ELEM_READ32:
+ return snd_ctl_elem_read_user_compat(ctl->card, argp);
+ case SNDRV_CTL_IOCTL_ELEM_WRITE32:
+ return snd_ctl_elem_write_user_compat(ctl, argp);
+ case SNDRV_CTL_IOCTL_ELEM_ADD32:
+ return snd_ctl_elem_add_compat(ctl, argp, 0);
+ case SNDRV_CTL_IOCTL_ELEM_REPLACE32:
+ return snd_ctl_elem_add_compat(ctl, argp, 1);
+ }
+
+ down_read(&snd_ioctl_rwsem);
+ list_for_each_entry(p, &snd_control_compat_ioctls, list) {
+ if (p->fioctl) {
+ err = p->fioctl(ctl->card, ctl, cmd, arg);
+ if (err != -ENOIOCTLCMD) {
+ up_read(&snd_ioctl_rwsem);
+ return err;
+ }
+ }
+ }
+ up_read(&snd_ioctl_rwsem);
+ return -ENOIOCTLCMD;
+}
diff --git a/sound/core/device.c b/sound/core/device.c
new file mode 100644
index 0000000..c58d822
--- /dev/null
+++ b/sound/core/device.c
@@ -0,0 +1,243 @@
+/*
+ * Device management routines
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+
+/**
+ * snd_device_new - create an ALSA device component
+ * @card: the card instance
+ * @type: the device type, SNDRV_DEV_XXX
+ * @device_data: the data pointer of this device
+ * @ops: the operator table
+ *
+ * Creates a new device component for the given data pointer.
+ * The device will be assigned to the card and managed together
+ * by the card.
+ *
+ * The data pointer plays a role as the identifier, too, so the
+ * pointer address must be unique and unchanged.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_device_new(struct snd_card *card, snd_device_type_t type,
+ void *device_data, struct snd_device_ops *ops)
+{
+ struct snd_device *dev;
+
+ if (snd_BUG_ON(!card || !device_data || !ops))
+ return -ENXIO;
+ dev = kzalloc(sizeof(*dev), GFP_KERNEL);
+ if (dev == NULL) {
+ snd_printk(KERN_ERR "Cannot allocate device\n");
+ return -ENOMEM;
+ }
+ dev->card = card;
+ dev->type = type;
+ dev->state = SNDRV_DEV_BUILD;
+ dev->device_data = device_data;
+ dev->ops = ops;
+ list_add(&dev->list, &card->devices); /* add to the head of list */
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_device_new);
+
+/**
+ * snd_device_free - release the device from the card
+ * @card: the card instance
+ * @device_data: the data pointer to release
+ *
+ * Removes the device from the list on the card and invokes the
+ * callbacks, dev_disconnect and dev_free, corresponding to the state.
+ * Then release the device.
+ *
+ * Returns zero if successful, or a negative error code on failure or if the
+ * device not found.
+ */
+int snd_device_free(struct snd_card *card, void *device_data)
+{
+ struct snd_device *dev;
+
+ if (snd_BUG_ON(!card || !device_data))
+ return -ENXIO;
+ list_for_each_entry(dev, &card->devices, list) {
+ if (dev->device_data != device_data)
+ continue;
+ /* unlink */
+ list_del(&dev->list);
+ if (dev->state == SNDRV_DEV_REGISTERED &&
+ dev->ops->dev_disconnect)
+ if (dev->ops->dev_disconnect(dev))
+ snd_printk(KERN_ERR
+ "device disconnect failure\n");
+ if (dev->ops->dev_free) {
+ if (dev->ops->dev_free(dev))
+ snd_printk(KERN_ERR "device free failure\n");
+ }
+ kfree(dev);
+ return 0;
+ }
+ snd_printd("device free %p (from %p), not found\n", device_data,
+ __builtin_return_address(0));
+ return -ENXIO;
+}
+
+EXPORT_SYMBOL(snd_device_free);
+
+/**
+ * snd_device_disconnect - disconnect the device
+ * @card: the card instance
+ * @device_data: the data pointer to disconnect
+ *
+ * Turns the device into the disconnection state, invoking
+ * dev_disconnect callback, if the device was already registered.
+ *
+ * Usually called from snd_card_disconnect().
+ *
+ * Returns zero if successful, or a negative error code on failure or if the
+ * device not found.
+ */
+int snd_device_disconnect(struct snd_card *card, void *device_data)
+{
+ struct snd_device *dev;
+
+ if (snd_BUG_ON(!card || !device_data))
+ return -ENXIO;
+ list_for_each_entry(dev, &card->devices, list) {
+ if (dev->device_data != device_data)
+ continue;
+ if (dev->state == SNDRV_DEV_REGISTERED &&
+ dev->ops->dev_disconnect) {
+ if (dev->ops->dev_disconnect(dev))
+ snd_printk(KERN_ERR "device disconnect failure\n");
+ dev->state = SNDRV_DEV_DISCONNECTED;
+ }
+ return 0;
+ }
+ snd_printd("device disconnect %p (from %p), not found\n", device_data,
+ __builtin_return_address(0));
+ return -ENXIO;
+}
+
+/**
+ * snd_device_register - register the device
+ * @card: the card instance
+ * @device_data: the data pointer to register
+ *
+ * Registers the device which was already created via
+ * snd_device_new(). Usually this is called from snd_card_register(),
+ * but it can be called later if any new devices are created after
+ * invocation of snd_card_register().
+ *
+ * Returns zero if successful, or a negative error code on failure or if the
+ * device not found.
+ */
+int snd_device_register(struct snd_card *card, void *device_data)
+{
+ struct snd_device *dev;
+ int err;
+
+ if (snd_BUG_ON(!card || !device_data))
+ return -ENXIO;
+ list_for_each_entry(dev, &card->devices, list) {
+ if (dev->device_data != device_data)
+ continue;
+ if (dev->state == SNDRV_DEV_BUILD && dev->ops->dev_register) {
+ if ((err = dev->ops->dev_register(dev)) < 0)
+ return err;
+ dev->state = SNDRV_DEV_REGISTERED;
+ return 0;
+ }
+ snd_printd("snd_device_register busy\n");
+ return -EBUSY;
+ }
+ snd_BUG();
+ return -ENXIO;
+}
+
+EXPORT_SYMBOL(snd_device_register);
+
+/*
+ * register all the devices on the card.
+ * called from init.c
+ */
+int snd_device_register_all(struct snd_card *card)
+{
+ struct snd_device *dev;
+ int err;
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ list_for_each_entry(dev, &card->devices, list) {
+ if (dev->state == SNDRV_DEV_BUILD && dev->ops->dev_register) {
+ if ((err = dev->ops->dev_register(dev)) < 0)
+ return err;
+ dev->state = SNDRV_DEV_REGISTERED;
+ }
+ }
+ return 0;
+}
+
+/*
+ * disconnect all the devices on the card.
+ * called from init.c
+ */
+int snd_device_disconnect_all(struct snd_card *card)
+{
+ struct snd_device *dev;
+ int err = 0;
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ list_for_each_entry(dev, &card->devices, list) {
+ if (snd_device_disconnect(card, dev->device_data) < 0)
+ err = -ENXIO;
+ }
+ return err;
+}
+
+/*
+ * release all the devices on the card.
+ * called from init.c
+ */
+int snd_device_free_all(struct snd_card *card, snd_device_cmd_t cmd)
+{
+ struct snd_device *dev;
+ int err;
+ unsigned int range_low, range_high;
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ range_low = cmd * SNDRV_DEV_TYPE_RANGE_SIZE;
+ range_high = range_low + SNDRV_DEV_TYPE_RANGE_SIZE - 1;
+ __again:
+ list_for_each_entry(dev, &card->devices, list) {
+ if (dev->type >= range_low && dev->type <= range_high) {
+ if ((err = snd_device_free(card, dev->device_data)) < 0)
+ return err;
+ goto __again;
+ }
+ }
+ return 0;
+}
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
new file mode 100644
index 0000000..195cafc
--- /dev/null
+++ b/sound/core/hwdep.c
@@ -0,0 +1,528 @@
+/*
+ * Hardware dependent layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/major.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/minors.h>
+#include <sound/hwdep.h>
+#include <sound/info.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Hardware dependent layer");
+MODULE_LICENSE("GPL");
+
+static LIST_HEAD(snd_hwdep_devices);
+static DEFINE_MUTEX(register_mutex);
+
+static int snd_hwdep_free(struct snd_hwdep *hwdep);
+static int snd_hwdep_dev_free(struct snd_device *device);
+static int snd_hwdep_dev_register(struct snd_device *device);
+static int snd_hwdep_dev_disconnect(struct snd_device *device);
+
+
+static struct snd_hwdep *snd_hwdep_search(struct snd_card *card, int device)
+{
+ struct snd_hwdep *hwdep;
+
+ list_for_each_entry(hwdep, &snd_hwdep_devices, list)
+ if (hwdep->card == card && hwdep->device == device)
+ return hwdep;
+ return NULL;
+}
+
+static loff_t snd_hwdep_llseek(struct file * file, loff_t offset, int orig)
+{
+ struct snd_hwdep *hw = file->private_data;
+ if (hw->ops.llseek)
+ return hw->ops.llseek(hw, file, offset, orig);
+ return -ENXIO;
+}
+
+static ssize_t snd_hwdep_read(struct file * file, char __user *buf,
+ size_t count, loff_t *offset)
+{
+ struct snd_hwdep *hw = file->private_data;
+ if (hw->ops.read)
+ return hw->ops.read(hw, buf, count, offset);
+ return -ENXIO;
+}
+
+static ssize_t snd_hwdep_write(struct file * file, const char __user *buf,
+ size_t count, loff_t *offset)
+{
+ struct snd_hwdep *hw = file->private_data;
+ if (hw->ops.write)
+ return hw->ops.write(hw, buf, count, offset);
+ return -ENXIO;
+}
+
+static int snd_hwdep_open(struct inode *inode, struct file * file)
+{
+ int major = imajor(inode);
+ struct snd_hwdep *hw;
+ int err;
+ wait_queue_t wait;
+
+ if (major == snd_major) {
+ hw = snd_lookup_minor_data(iminor(inode),
+ SNDRV_DEVICE_TYPE_HWDEP);
+#ifdef CONFIG_SND_OSSEMUL
+ } else if (major == SOUND_MAJOR) {
+ hw = snd_lookup_oss_minor_data(iminor(inode),
+ SNDRV_OSS_DEVICE_TYPE_DMFM);
+#endif
+ } else
+ return -ENXIO;
+ if (hw == NULL)
+ return -ENODEV;
+
+ if (!hw->ops.open)
+ return -ENXIO;
+
+ if (!try_module_get(hw->card->module))
+ return -EFAULT;
+
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&hw->open_wait, &wait);
+ mutex_lock(&hw->open_mutex);
+ while (1) {
+ if (hw->exclusive && hw->used > 0) {
+ err = -EBUSY;
+ break;
+ }
+ err = hw->ops.open(hw, file);
+ if (err >= 0)
+ break;
+ if (err == -EAGAIN) {
+ if (file->f_flags & O_NONBLOCK) {
+ err = -EBUSY;
+ break;
+ }
+ } else
+ break;
+ set_current_state(TASK_INTERRUPTIBLE);
+ mutex_unlock(&hw->open_mutex);
+ schedule();
+ mutex_lock(&hw->open_mutex);
+ if (signal_pending(current)) {
+ err = -ERESTARTSYS;
+ break;
+ }
+ }
+ remove_wait_queue(&hw->open_wait, &wait);
+ if (err >= 0) {
+ err = snd_card_file_add(hw->card, file);
+ if (err >= 0) {
+ file->private_data = hw;
+ hw->used++;
+ } else {
+ if (hw->ops.release)
+ hw->ops.release(hw, file);
+ }
+ }
+ mutex_unlock(&hw->open_mutex);
+ if (err < 0)
+ module_put(hw->card->module);
+ return err;
+}
+
+static int snd_hwdep_release(struct inode *inode, struct file * file)
+{
+ int err = -ENXIO;
+ struct snd_hwdep *hw = file->private_data;
+ struct module *mod = hw->card->module;
+
+ mutex_lock(&hw->open_mutex);
+ if (hw->ops.release)
+ err = hw->ops.release(hw, file);
+ if (hw->used > 0)
+ hw->used--;
+ mutex_unlock(&hw->open_mutex);
+ wake_up(&hw->open_wait);
+
+ snd_card_file_remove(hw->card, file);
+ module_put(mod);
+ return err;
+}
+
+static unsigned int snd_hwdep_poll(struct file * file, poll_table * wait)
+{
+ struct snd_hwdep *hw = file->private_data;
+ if (hw->ops.poll)
+ return hw->ops.poll(hw, file, wait);
+ return 0;
+}
+
+static int snd_hwdep_info(struct snd_hwdep *hw,
+ struct snd_hwdep_info __user *_info)
+{
+ struct snd_hwdep_info info;
+
+ memset(&info, 0, sizeof(info));
+ info.card = hw->card->number;
+ strlcpy(info.id, hw->id, sizeof(info.id));
+ strlcpy(info.name, hw->name, sizeof(info.name));
+ info.iface = hw->iface;
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_hwdep_dsp_status(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_status __user *_info)
+{
+ struct snd_hwdep_dsp_status info;
+ int err;
+
+ if (! hw->ops.dsp_status)
+ return -ENXIO;
+ memset(&info, 0, sizeof(info));
+ info.dsp_loaded = hw->dsp_loaded;
+ if ((err = hw->ops.dsp_status(hw, &info)) < 0)
+ return err;
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_hwdep_dsp_load(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_image __user *_info)
+{
+ struct snd_hwdep_dsp_image info;
+ int err;
+
+ if (! hw->ops.dsp_load)
+ return -ENXIO;
+ memset(&info, 0, sizeof(info));
+ if (copy_from_user(&info, _info, sizeof(info)))
+ return -EFAULT;
+ /* check whether the dsp was already loaded */
+ if (hw->dsp_loaded & (1 << info.index))
+ return -EBUSY;
+ if (!access_ok(VERIFY_READ, info.image, info.length))
+ return -EFAULT;
+ err = hw->ops.dsp_load(hw, &info);
+ if (err < 0)
+ return err;
+ hw->dsp_loaded |= (1 << info.index);
+ return 0;
+}
+
+static long snd_hwdep_ioctl(struct file * file, unsigned int cmd,
+ unsigned long arg)
+{
+ struct snd_hwdep *hw = file->private_data;
+ void __user *argp = (void __user *)arg;
+ switch (cmd) {
+ case SNDRV_HWDEP_IOCTL_PVERSION:
+ return put_user(SNDRV_HWDEP_VERSION, (int __user *)argp);
+ case SNDRV_HWDEP_IOCTL_INFO:
+ return snd_hwdep_info(hw, argp);
+ case SNDRV_HWDEP_IOCTL_DSP_STATUS:
+ return snd_hwdep_dsp_status(hw, argp);
+ case SNDRV_HWDEP_IOCTL_DSP_LOAD:
+ return snd_hwdep_dsp_load(hw, argp);
+ }
+ if (hw->ops.ioctl)
+ return hw->ops.ioctl(hw, file, cmd, arg);
+ return -ENOTTY;
+}
+
+static int snd_hwdep_mmap(struct file * file, struct vm_area_struct * vma)
+{
+ struct snd_hwdep *hw = file->private_data;
+ if (hw->ops.mmap)
+ return hw->ops.mmap(hw, file, vma);
+ return -ENXIO;
+}
+
+static int snd_hwdep_control_ioctl(struct snd_card *card,
+ struct snd_ctl_file * control,
+ unsigned int cmd, unsigned long arg)
+{
+ switch (cmd) {
+ case SNDRV_CTL_IOCTL_HWDEP_NEXT_DEVICE:
+ {
+ int device;
+
+ if (get_user(device, (int __user *)arg))
+ return -EFAULT;
+ mutex_lock(&register_mutex);
+ device = device < 0 ? 0 : device + 1;
+ while (device < SNDRV_MINOR_HWDEPS) {
+ if (snd_hwdep_search(card, device))
+ break;
+ device++;
+ }
+ if (device >= SNDRV_MINOR_HWDEPS)
+ device = -1;
+ mutex_unlock(&register_mutex);
+ if (put_user(device, (int __user *)arg))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_CTL_IOCTL_HWDEP_INFO:
+ {
+ struct snd_hwdep_info __user *info = (struct snd_hwdep_info __user *)arg;
+ int device, err;
+ struct snd_hwdep *hwdep;
+
+ if (get_user(device, &info->device))
+ return -EFAULT;
+ mutex_lock(&register_mutex);
+ hwdep = snd_hwdep_search(card, device);
+ if (hwdep)
+ err = snd_hwdep_info(hwdep, info);
+ else
+ err = -ENXIO;
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+ }
+ return -ENOIOCTLCMD;
+}
+
+#ifdef CONFIG_COMPAT
+#include "hwdep_compat.c"
+#else
+#define snd_hwdep_ioctl_compat NULL
+#endif
+
+/*
+
+ */
+
+static const struct file_operations snd_hwdep_f_ops =
+{
+ .owner = THIS_MODULE,
+ .llseek = snd_hwdep_llseek,
+ .read = snd_hwdep_read,
+ .write = snd_hwdep_write,
+ .open = snd_hwdep_open,
+ .release = snd_hwdep_release,
+ .poll = snd_hwdep_poll,
+ .unlocked_ioctl = snd_hwdep_ioctl,
+ .compat_ioctl = snd_hwdep_ioctl_compat,
+ .mmap = snd_hwdep_mmap,
+};
+
+/**
+ * snd_hwdep_new - create a new hwdep instance
+ * @card: the card instance
+ * @id: the id string
+ * @device: the device index (zero-based)
+ * @rhwdep: the pointer to store the new hwdep instance
+ *
+ * Creates a new hwdep instance with the given index on the card.
+ * The callbacks (hwdep->ops) must be set on the returned instance
+ * after this call manually by the caller.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_hwdep_new(struct snd_card *card, char *id, int device,
+ struct snd_hwdep **rhwdep)
+{
+ struct snd_hwdep *hwdep;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_hwdep_dev_free,
+ .dev_register = snd_hwdep_dev_register,
+ .dev_disconnect = snd_hwdep_dev_disconnect,
+ };
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ if (rhwdep)
+ *rhwdep = NULL;
+ hwdep = kzalloc(sizeof(*hwdep), GFP_KERNEL);
+ if (hwdep == NULL) {
+ snd_printk(KERN_ERR "hwdep: cannot allocate\n");
+ return -ENOMEM;
+ }
+ hwdep->card = card;
+ hwdep->device = device;
+ if (id)
+ strlcpy(hwdep->id, id, sizeof(hwdep->id));
+#ifdef CONFIG_SND_OSSEMUL
+ hwdep->oss_type = -1;
+#endif
+ if ((err = snd_device_new(card, SNDRV_DEV_HWDEP, hwdep, &ops)) < 0) {
+ snd_hwdep_free(hwdep);
+ return err;
+ }
+ init_waitqueue_head(&hwdep->open_wait);
+ mutex_init(&hwdep->open_mutex);
+ if (rhwdep)
+ *rhwdep = hwdep;
+ return 0;
+}
+
+static int snd_hwdep_free(struct snd_hwdep *hwdep)
+{
+ if (!hwdep)
+ return 0;
+ if (hwdep->private_free)
+ hwdep->private_free(hwdep);
+ kfree(hwdep);
+ return 0;
+}
+
+static int snd_hwdep_dev_free(struct snd_device *device)
+{
+ struct snd_hwdep *hwdep = device->device_data;
+ return snd_hwdep_free(hwdep);
+}
+
+static int snd_hwdep_dev_register(struct snd_device *device)
+{
+ struct snd_hwdep *hwdep = device->device_data;
+ int err;
+ char name[32];
+
+ mutex_lock(&register_mutex);
+ if (snd_hwdep_search(hwdep->card, hwdep->device)) {
+ mutex_unlock(&register_mutex);
+ return -EBUSY;
+ }
+ list_add_tail(&hwdep->list, &snd_hwdep_devices);
+ sprintf(name, "hwC%iD%i", hwdep->card->number, hwdep->device);
+ if ((err = snd_register_device(SNDRV_DEVICE_TYPE_HWDEP,
+ hwdep->card, hwdep->device,
+ &snd_hwdep_f_ops, hwdep, name)) < 0) {
+ snd_printk(KERN_ERR "unable to register hardware dependent device %i:%i\n",
+ hwdep->card->number, hwdep->device);
+ list_del(&hwdep->list);
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+#ifdef CONFIG_SND_OSSEMUL
+ hwdep->ossreg = 0;
+ if (hwdep->oss_type >= 0) {
+ if ((hwdep->oss_type == SNDRV_OSS_DEVICE_TYPE_DMFM) && (hwdep->device != 0)) {
+ snd_printk (KERN_WARNING "only hwdep device 0 can be registered as OSS direct FM device!\n");
+ } else {
+ if (snd_register_oss_device(hwdep->oss_type,
+ hwdep->card, hwdep->device,
+ &snd_hwdep_f_ops, hwdep,
+ hwdep->oss_dev) < 0) {
+ snd_printk(KERN_ERR "unable to register OSS compatibility device %i:%i\n",
+ hwdep->card->number, hwdep->device);
+ } else
+ hwdep->ossreg = 1;
+ }
+ }
+#endif
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+static int snd_hwdep_dev_disconnect(struct snd_device *device)
+{
+ struct snd_hwdep *hwdep = device->device_data;
+
+ if (snd_BUG_ON(!hwdep))
+ return -ENXIO;
+ mutex_lock(&register_mutex);
+ if (snd_hwdep_search(hwdep->card, hwdep->device) != hwdep) {
+ mutex_unlock(&register_mutex);
+ return -EINVAL;
+ }
+#ifdef CONFIG_SND_OSSEMUL
+ if (hwdep->ossreg)
+ snd_unregister_oss_device(hwdep->oss_type, hwdep->card, hwdep->device);
+#endif
+ snd_unregister_device(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card, hwdep->device);
+ list_del_init(&hwdep->list);
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+#ifdef CONFIG_PROC_FS
+/*
+ * Info interface
+ */
+
+static void snd_hwdep_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_hwdep *hwdep;
+
+ mutex_lock(&register_mutex);
+ list_for_each_entry(hwdep, &snd_hwdep_devices, list)
+ snd_iprintf(buffer, "%02i-%02i: %s\n",
+ hwdep->card->number, hwdep->device, hwdep->name);
+ mutex_unlock(&register_mutex);
+}
+
+static struct snd_info_entry *snd_hwdep_proc_entry;
+
+static void __init snd_hwdep_proc_init(void)
+{
+ struct snd_info_entry *entry;
+
+ if ((entry = snd_info_create_module_entry(THIS_MODULE, "hwdep", NULL)) != NULL) {
+ entry->c.text.read = snd_hwdep_proc_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ snd_hwdep_proc_entry = entry;
+}
+
+static void __exit snd_hwdep_proc_done(void)
+{
+ snd_info_free_entry(snd_hwdep_proc_entry);
+}
+#else /* !CONFIG_PROC_FS */
+#define snd_hwdep_proc_init()
+#define snd_hwdep_proc_done()
+#endif /* CONFIG_PROC_FS */
+
+
+/*
+ * ENTRY functions
+ */
+
+static int __init alsa_hwdep_init(void)
+{
+ snd_hwdep_proc_init();
+ snd_ctl_register_ioctl(snd_hwdep_control_ioctl);
+ snd_ctl_register_ioctl_compat(snd_hwdep_control_ioctl);
+ return 0;
+}
+
+static void __exit alsa_hwdep_exit(void)
+{
+ snd_ctl_unregister_ioctl(snd_hwdep_control_ioctl);
+ snd_ctl_unregister_ioctl_compat(snd_hwdep_control_ioctl);
+ snd_hwdep_proc_done();
+}
+
+module_init(alsa_hwdep_init)
+module_exit(alsa_hwdep_exit)
+
+EXPORT_SYMBOL(snd_hwdep_new);
diff --git a/sound/core/hwdep_compat.c b/sound/core/hwdep_compat.c
new file mode 100644
index 0000000..3827c0c
--- /dev/null
+++ b/sound/core/hwdep_compat.c
@@ -0,0 +1,78 @@
+/*
+ * 32bit -> 64bit ioctl wrapper for hwdep API
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* This file is included from hwdep.c */
+
+#include <linux/compat.h>
+
+struct snd_hwdep_dsp_image32 {
+ u32 index;
+ unsigned char name[64];
+ u32 image; /* pointer */
+ u32 length;
+ u32 driver_data;
+} /* don't set packed attribute here */;
+
+static int snd_hwdep_dsp_load_compat(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_image32 __user *src)
+{
+ struct snd_hwdep_dsp_image __user *dst;
+ compat_caddr_t ptr;
+ u32 val;
+
+ dst = compat_alloc_user_space(sizeof(*dst));
+
+ /* index and name */
+ if (copy_in_user(dst, src, 4 + 64))
+ return -EFAULT;
+ if (get_user(ptr, &src->image) ||
+ put_user(compat_ptr(ptr), &dst->image))
+ return -EFAULT;
+ if (get_user(val, &src->length) ||
+ put_user(val, &dst->length))
+ return -EFAULT;
+ if (get_user(val, &src->driver_data) ||
+ put_user(val, &dst->driver_data))
+ return -EFAULT;
+
+ return snd_hwdep_dsp_load(hw, dst);
+}
+
+enum {
+ SNDRV_HWDEP_IOCTL_DSP_LOAD32 = _IOW('H', 0x03, struct snd_hwdep_dsp_image32)
+};
+
+static long snd_hwdep_ioctl_compat(struct file * file, unsigned int cmd,
+ unsigned long arg)
+{
+ struct snd_hwdep *hw = file->private_data;
+ void __user *argp = compat_ptr(arg);
+ switch (cmd) {
+ case SNDRV_HWDEP_IOCTL_PVERSION:
+ case SNDRV_HWDEP_IOCTL_INFO:
+ case SNDRV_HWDEP_IOCTL_DSP_STATUS:
+ return snd_hwdep_ioctl(file, cmd, (unsigned long)argp);
+ case SNDRV_HWDEP_IOCTL_DSP_LOAD32:
+ return snd_hwdep_dsp_load_compat(hw, argp);
+ }
+ if (hw->ops.ioctl_compat)
+ return hw->ops.ioctl_compat(hw, file, cmd, arg);
+ return -ENOIOCTLCMD;
+}
diff --git a/sound/core/info.c b/sound/core/info.c
new file mode 100644
index 0000000..527b207
--- /dev/null
+++ b/sound/core/info.c
@@ -0,0 +1,1015 @@
+/*
+ * Information interface for ALSA driver
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/mm.h>
+#include <linux/smp_lock.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/info.h>
+#include <sound/version.h>
+#include <linux/proc_fs.h>
+#include <linux/mutex.h>
+#include <stdarg.h>
+
+/*
+ *
+ */
+
+#ifdef CONFIG_PROC_FS
+
+int snd_info_check_reserved_words(const char *str)
+{
+ static char *reserved[] =
+ {
+ "version",
+ "meminfo",
+ "memdebug",
+ "detect",
+ "devices",
+ "oss",
+ "cards",
+ "timers",
+ "synth",
+ "pcm",
+ "seq",
+ NULL
+ };
+ char **xstr = reserved;
+
+ while (*xstr) {
+ if (!strcmp(*xstr, str))
+ return 0;
+ xstr++;
+ }
+ if (!strncmp(str, "card", 4))
+ return 0;
+ return 1;
+}
+
+static DEFINE_MUTEX(info_mutex);
+
+struct snd_info_private_data {
+ struct snd_info_buffer *rbuffer;
+ struct snd_info_buffer *wbuffer;
+ struct snd_info_entry *entry;
+ void *file_private_data;
+};
+
+static int snd_info_version_init(void);
+static int snd_info_version_done(void);
+static void snd_info_disconnect(struct snd_info_entry *entry);
+
+
+/* resize the proc r/w buffer */
+static int resize_info_buffer(struct snd_info_buffer *buffer,
+ unsigned int nsize)
+{
+ char *nbuf;
+
+ nsize = PAGE_ALIGN(nsize);
+ nbuf = kmalloc(nsize, GFP_KERNEL);
+ if (! nbuf)
+ return -ENOMEM;
+
+ memcpy(nbuf, buffer->buffer, buffer->len);
+ kfree(buffer->buffer);
+ buffer->buffer = nbuf;
+ buffer->len = nsize;
+ return 0;
+}
+
+/**
+ * snd_iprintf - printf on the procfs buffer
+ * @buffer: the procfs buffer
+ * @fmt: the printf format
+ *
+ * Outputs the string on the procfs buffer just like printf().
+ *
+ * Returns the size of output string.
+ */
+int snd_iprintf(struct snd_info_buffer *buffer, char *fmt,...)
+{
+ va_list args;
+ int len, res;
+ int err = 0;
+
+ might_sleep();
+ if (buffer->stop || buffer->error)
+ return 0;
+ len = buffer->len - buffer->size;
+ va_start(args, fmt);
+ for (;;) {
+ va_list ap;
+ va_copy(ap, args);
+ res = vsnprintf(buffer->buffer + buffer->curr, len, fmt, ap);
+ va_end(ap);
+ if (res < len)
+ break;
+ err = resize_info_buffer(buffer, buffer->len + PAGE_SIZE);
+ if (err < 0)
+ break;
+ len = buffer->len - buffer->size;
+ }
+ va_end(args);
+
+ if (err < 0)
+ return err;
+ buffer->curr += res;
+ buffer->size += res;
+ return res;
+}
+
+EXPORT_SYMBOL(snd_iprintf);
+
+/*
+
+ */
+
+static struct proc_dir_entry *snd_proc_root;
+struct snd_info_entry *snd_seq_root;
+EXPORT_SYMBOL(snd_seq_root);
+
+#ifdef CONFIG_SND_OSSEMUL
+struct snd_info_entry *snd_oss_root;
+#endif
+
+static inline void snd_info_entry_prepare(struct proc_dir_entry *de)
+{
+ de->owner = THIS_MODULE;
+}
+
+static void snd_remove_proc_entry(struct proc_dir_entry *parent,
+ struct proc_dir_entry *de)
+{
+ if (de)
+ remove_proc_entry(de->name, parent);
+}
+
+static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig)
+{
+ struct snd_info_private_data *data;
+ struct snd_info_entry *entry;
+ loff_t ret;
+
+ data = file->private_data;
+ entry = data->entry;
+ lock_kernel();
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_TEXT:
+ switch (orig) {
+ case SEEK_SET:
+ file->f_pos = offset;
+ ret = file->f_pos;
+ goto out;
+ case SEEK_CUR:
+ file->f_pos += offset;
+ ret = file->f_pos;
+ goto out;
+ case SEEK_END:
+ default:
+ ret = -EINVAL;
+ goto out;
+ }
+ break;
+ case SNDRV_INFO_CONTENT_DATA:
+ if (entry->c.ops->llseek) {
+ ret = entry->c.ops->llseek(entry,
+ data->file_private_data,
+ file, offset, orig);
+ goto out;
+ }
+ break;
+ }
+ ret = -ENXIO;
+out:
+ unlock_kernel();
+ return ret;
+}
+
+static ssize_t snd_info_entry_read(struct file *file, char __user *buffer,
+ size_t count, loff_t * offset)
+{
+ struct snd_info_private_data *data;
+ struct snd_info_entry *entry;
+ struct snd_info_buffer *buf;
+ size_t size = 0;
+ loff_t pos;
+
+ data = file->private_data;
+ if (snd_BUG_ON(!data))
+ return -ENXIO;
+ pos = *offset;
+ if (pos < 0 || (long) pos != pos || (ssize_t) count < 0)
+ return -EIO;
+ if ((unsigned long) pos + (unsigned long) count < (unsigned long) pos)
+ return -EIO;
+ entry = data->entry;
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_TEXT:
+ buf = data->rbuffer;
+ if (buf == NULL)
+ return -EIO;
+ if (pos >= buf->size)
+ return 0;
+ size = buf->size - pos;
+ size = min(count, size);
+ if (copy_to_user(buffer, buf->buffer + pos, size))
+ return -EFAULT;
+ break;
+ case SNDRV_INFO_CONTENT_DATA:
+ if (entry->c.ops->read)
+ size = entry->c.ops->read(entry,
+ data->file_private_data,
+ file, buffer, count, pos);
+ break;
+ }
+ if ((ssize_t) size > 0)
+ *offset = pos + size;
+ return size;
+}
+
+static ssize_t snd_info_entry_write(struct file *file, const char __user *buffer,
+ size_t count, loff_t * offset)
+{
+ struct snd_info_private_data *data;
+ struct snd_info_entry *entry;
+ struct snd_info_buffer *buf;
+ ssize_t size = 0;
+ loff_t pos;
+
+ data = file->private_data;
+ if (snd_BUG_ON(!data))
+ return -ENXIO;
+ entry = data->entry;
+ pos = *offset;
+ if (pos < 0 || (long) pos != pos || (ssize_t) count < 0)
+ return -EIO;
+ if ((unsigned long) pos + (unsigned long) count < (unsigned long) pos)
+ return -EIO;
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_TEXT:
+ buf = data->wbuffer;
+ if (buf == NULL)
+ return -EIO;
+ mutex_lock(&entry->access);
+ if (pos + count >= buf->len) {
+ if (resize_info_buffer(buf, pos + count)) {
+ mutex_unlock(&entry->access);
+ return -ENOMEM;
+ }
+ }
+ if (copy_from_user(buf->buffer + pos, buffer, count)) {
+ mutex_unlock(&entry->access);
+ return -EFAULT;
+ }
+ buf->size = pos + count;
+ mutex_unlock(&entry->access);
+ size = count;
+ break;
+ case SNDRV_INFO_CONTENT_DATA:
+ if (entry->c.ops->write)
+ size = entry->c.ops->write(entry,
+ data->file_private_data,
+ file, buffer, count, pos);
+ break;
+ }
+ if ((ssize_t) size > 0)
+ *offset = pos + size;
+ return size;
+}
+
+static int snd_info_entry_open(struct inode *inode, struct file *file)
+{
+ struct snd_info_entry *entry;
+ struct snd_info_private_data *data;
+ struct snd_info_buffer *buffer;
+ struct proc_dir_entry *p;
+ int mode, err;
+
+ mutex_lock(&info_mutex);
+ p = PDE(inode);
+ entry = p == NULL ? NULL : (struct snd_info_entry *)p->data;
+ if (entry == NULL || ! entry->p) {
+ mutex_unlock(&info_mutex);
+ return -ENODEV;
+ }
+ if (!try_module_get(entry->module)) {
+ err = -EFAULT;
+ goto __error1;
+ }
+ mode = file->f_flags & O_ACCMODE;
+ if (mode == O_RDONLY || mode == O_RDWR) {
+ if ((entry->content == SNDRV_INFO_CONTENT_DATA &&
+ entry->c.ops->read == NULL)) {
+ err = -ENODEV;
+ goto __error;
+ }
+ }
+ if (mode == O_WRONLY || mode == O_RDWR) {
+ if ((entry->content == SNDRV_INFO_CONTENT_DATA &&
+ entry->c.ops->write == NULL)) {
+ err = -ENODEV;
+ goto __error;
+ }
+ }
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (data == NULL) {
+ err = -ENOMEM;
+ goto __error;
+ }
+ data->entry = entry;
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_TEXT:
+ if (mode == O_RDONLY || mode == O_RDWR) {
+ buffer = kzalloc(sizeof(*buffer), GFP_KERNEL);
+ if (buffer == NULL)
+ goto __nomem;
+ data->rbuffer = buffer;
+ buffer->len = PAGE_SIZE;
+ buffer->buffer = kmalloc(buffer->len, GFP_KERNEL);
+ if (buffer->buffer == NULL)
+ goto __nomem;
+ }
+ if (mode == O_WRONLY || mode == O_RDWR) {
+ buffer = kzalloc(sizeof(*buffer), GFP_KERNEL);
+ if (buffer == NULL)
+ goto __nomem;
+ data->wbuffer = buffer;
+ buffer->len = PAGE_SIZE;
+ buffer->buffer = kmalloc(buffer->len, GFP_KERNEL);
+ if (buffer->buffer == NULL)
+ goto __nomem;
+ }
+ break;
+ case SNDRV_INFO_CONTENT_DATA: /* data */
+ if (entry->c.ops->open) {
+ if ((err = entry->c.ops->open(entry, mode,
+ &data->file_private_data)) < 0) {
+ kfree(data);
+ goto __error;
+ }
+ }
+ break;
+ }
+ file->private_data = data;
+ mutex_unlock(&info_mutex);
+ if (entry->content == SNDRV_INFO_CONTENT_TEXT &&
+ (mode == O_RDONLY || mode == O_RDWR)) {
+ if (entry->c.text.read) {
+ mutex_lock(&entry->access);
+ entry->c.text.read(entry, data->rbuffer);
+ mutex_unlock(&entry->access);
+ }
+ }
+ return 0;
+
+ __nomem:
+ if (data->rbuffer) {
+ kfree(data->rbuffer->buffer);
+ kfree(data->rbuffer);
+ }
+ if (data->wbuffer) {
+ kfree(data->wbuffer->buffer);
+ kfree(data->wbuffer);
+ }
+ kfree(data);
+ err = -ENOMEM;
+ __error:
+ module_put(entry->module);
+ __error1:
+ mutex_unlock(&info_mutex);
+ return err;
+}
+
+static int snd_info_entry_release(struct inode *inode, struct file *file)
+{
+ struct snd_info_entry *entry;
+ struct snd_info_private_data *data;
+ int mode;
+
+ mode = file->f_flags & O_ACCMODE;
+ data = file->private_data;
+ entry = data->entry;
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_TEXT:
+ if (data->rbuffer) {
+ kfree(data->rbuffer->buffer);
+ kfree(data->rbuffer);
+ }
+ if (data->wbuffer) {
+ if (entry->c.text.write) {
+ entry->c.text.write(entry, data->wbuffer);
+ if (data->wbuffer->error) {
+ snd_printk(KERN_WARNING "data write error to %s (%i)\n",
+ entry->name,
+ data->wbuffer->error);
+ }
+ }
+ kfree(data->wbuffer->buffer);
+ kfree(data->wbuffer);
+ }
+ break;
+ case SNDRV_INFO_CONTENT_DATA:
+ if (entry->c.ops->release)
+ entry->c.ops->release(entry, mode,
+ data->file_private_data);
+ break;
+ }
+ module_put(entry->module);
+ kfree(data);
+ return 0;
+}
+
+static unsigned int snd_info_entry_poll(struct file *file, poll_table * wait)
+{
+ struct snd_info_private_data *data;
+ struct snd_info_entry *entry;
+ unsigned int mask;
+
+ data = file->private_data;
+ if (data == NULL)
+ return 0;
+ entry = data->entry;
+ mask = 0;
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_DATA:
+ if (entry->c.ops->poll)
+ return entry->c.ops->poll(entry,
+ data->file_private_data,
+ file, wait);
+ if (entry->c.ops->read)
+ mask |= POLLIN | POLLRDNORM;
+ if (entry->c.ops->write)
+ mask |= POLLOUT | POLLWRNORM;
+ break;
+ }
+ return mask;
+}
+
+static long snd_info_entry_ioctl(struct file *file, unsigned int cmd,
+ unsigned long arg)
+{
+ struct snd_info_private_data *data;
+ struct snd_info_entry *entry;
+
+ data = file->private_data;
+ if (data == NULL)
+ return 0;
+ entry = data->entry;
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_DATA:
+ if (entry->c.ops->ioctl)
+ return entry->c.ops->ioctl(entry,
+ data->file_private_data,
+ file, cmd, arg);
+ break;
+ }
+ return -ENOTTY;
+}
+
+static int snd_info_entry_mmap(struct file *file, struct vm_area_struct *vma)
+{
+ struct inode *inode = file->f_path.dentry->d_inode;
+ struct snd_info_private_data *data;
+ struct snd_info_entry *entry;
+
+ data = file->private_data;
+ if (data == NULL)
+ return 0;
+ entry = data->entry;
+ switch (entry->content) {
+ case SNDRV_INFO_CONTENT_DATA:
+ if (entry->c.ops->mmap)
+ return entry->c.ops->mmap(entry,
+ data->file_private_data,
+ inode, file, vma);
+ break;
+ }
+ return -ENXIO;
+}
+
+static const struct file_operations snd_info_entry_operations =
+{
+ .owner = THIS_MODULE,
+ .llseek = snd_info_entry_llseek,
+ .read = snd_info_entry_read,
+ .write = snd_info_entry_write,
+ .poll = snd_info_entry_poll,
+ .unlocked_ioctl = snd_info_entry_ioctl,
+ .mmap = snd_info_entry_mmap,
+ .open = snd_info_entry_open,
+ .release = snd_info_entry_release,
+};
+
+/**
+ * snd_create_proc_entry - create a procfs entry
+ * @name: the name of the proc file
+ * @mode: the file permission bits, S_Ixxx
+ * @parent: the parent proc-directory entry
+ *
+ * Creates a new proc file entry with the given name and permission
+ * on the given directory.
+ *
+ * Returns the pointer of new instance or NULL on failure.
+ */
+static struct proc_dir_entry *snd_create_proc_entry(const char *name, mode_t mode,
+ struct proc_dir_entry *parent)
+{
+ struct proc_dir_entry *p;
+ p = create_proc_entry(name, mode, parent);
+ if (p)
+ snd_info_entry_prepare(p);
+ return p;
+}
+
+int __init snd_info_init(void)
+{
+ struct proc_dir_entry *p;
+
+ p = snd_create_proc_entry("asound", S_IFDIR | S_IRUGO | S_IXUGO, NULL);
+ if (p == NULL)
+ return -ENOMEM;
+ snd_proc_root = p;
+#ifdef CONFIG_SND_OSSEMUL
+ {
+ struct snd_info_entry *entry;
+ if ((entry = snd_info_create_module_entry(THIS_MODULE, "oss", NULL)) == NULL)
+ return -ENOMEM;
+ entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ snd_oss_root = entry;
+ }
+#endif
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+ {
+ struct snd_info_entry *entry;
+ if ((entry = snd_info_create_module_entry(THIS_MODULE, "seq", NULL)) == NULL)
+ return -ENOMEM;
+ entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ snd_seq_root = entry;
+ }
+#endif
+ snd_info_version_init();
+ snd_minor_info_init();
+ snd_minor_info_oss_init();
+ snd_card_info_init();
+ return 0;
+}
+
+int __exit snd_info_done(void)
+{
+ snd_card_info_done();
+ snd_minor_info_oss_done();
+ snd_minor_info_done();
+ snd_info_version_done();
+ if (snd_proc_root) {
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+ snd_info_free_entry(snd_seq_root);
+#endif
+#ifdef CONFIG_SND_OSSEMUL
+ snd_info_free_entry(snd_oss_root);
+#endif
+ snd_remove_proc_entry(NULL, snd_proc_root);
+ }
+ return 0;
+}
+
+/*
+
+ */
+
+
+/*
+ * create a card proc file
+ * called from init.c
+ */
+int snd_info_card_create(struct snd_card *card)
+{
+ char str[8];
+ struct snd_info_entry *entry;
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+
+ sprintf(str, "card%i", card->number);
+ if ((entry = snd_info_create_module_entry(card->module, str, NULL)) == NULL)
+ return -ENOMEM;
+ entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ card->proc_root = entry;
+ return 0;
+}
+
+/*
+ * register the card proc file
+ * called from init.c
+ */
+int snd_info_card_register(struct snd_card *card)
+{
+ struct proc_dir_entry *p;
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+
+ if (!strcmp(card->id, card->proc_root->name))
+ return 0;
+
+ p = proc_symlink(card->id, snd_proc_root, card->proc_root->name);
+ if (p == NULL)
+ return -ENOMEM;
+ card->proc_root_link = p;
+ return 0;
+}
+
+/*
+ * de-register the card proc file
+ * called from init.c
+ */
+void snd_info_card_disconnect(struct snd_card *card)
+{
+ if (!card)
+ return;
+ mutex_lock(&info_mutex);
+ if (card->proc_root_link) {
+ snd_remove_proc_entry(snd_proc_root, card->proc_root_link);
+ card->proc_root_link = NULL;
+ }
+ if (card->proc_root)
+ snd_info_disconnect(card->proc_root);
+ mutex_unlock(&info_mutex);
+}
+
+/*
+ * release the card proc file resources
+ * called from init.c
+ */
+int snd_info_card_free(struct snd_card *card)
+{
+ if (!card)
+ return 0;
+ snd_info_free_entry(card->proc_root);
+ card->proc_root = NULL;
+ return 0;
+}
+
+
+/**
+ * snd_info_get_line - read one line from the procfs buffer
+ * @buffer: the procfs buffer
+ * @line: the buffer to store
+ * @len: the max. buffer size - 1
+ *
+ * Reads one line from the buffer and stores the string.
+ *
+ * Returns zero if successful, or 1 if error or EOF.
+ */
+int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
+{
+ int c = -1;
+
+ if (len <= 0 || buffer->stop || buffer->error)
+ return 1;
+ while (--len > 0) {
+ c = buffer->buffer[buffer->curr++];
+ if (c == '\n') {
+ if (buffer->curr >= buffer->size)
+ buffer->stop = 1;
+ break;
+ }
+ *line++ = c;
+ if (buffer->curr >= buffer->size) {
+ buffer->stop = 1;
+ break;
+ }
+ }
+ while (c != '\n' && !buffer->stop) {
+ c = buffer->buffer[buffer->curr++];
+ if (buffer->curr >= buffer->size)
+ buffer->stop = 1;
+ }
+ *line = '\0';
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_info_get_line);
+
+/**
+ * snd_info_get_str - parse a string token
+ * @dest: the buffer to store the string token
+ * @src: the original string
+ * @len: the max. length of token - 1
+ *
+ * Parses the original string and copy a token to the given
+ * string buffer.
+ *
+ * Returns the updated pointer of the original string so that
+ * it can be used for the next call.
+ */
+char *snd_info_get_str(char *dest, char *src, int len)
+{
+ int c;
+
+ while (*src == ' ' || *src == '\t')
+ src++;
+ if (*src == '"' || *src == '\'') {
+ c = *src++;
+ while (--len > 0 && *src && *src != c) {
+ *dest++ = *src++;
+ }
+ if (*src == c)
+ src++;
+ } else {
+ while (--len > 0 && *src && *src != ' ' && *src != '\t') {
+ *dest++ = *src++;
+ }
+ }
+ *dest = 0;
+ while (*src == ' ' || *src == '\t')
+ src++;
+ return src;
+}
+
+EXPORT_SYMBOL(snd_info_get_str);
+
+/**
+ * snd_info_create_entry - create an info entry
+ * @name: the proc file name
+ *
+ * Creates an info entry with the given file name and initializes as
+ * the default state.
+ *
+ * Usually called from other functions such as
+ * snd_info_create_card_entry().
+ *
+ * Returns the pointer of the new instance, or NULL on failure.
+ */
+static struct snd_info_entry *snd_info_create_entry(const char *name)
+{
+ struct snd_info_entry *entry;
+ entry = kzalloc(sizeof(*entry), GFP_KERNEL);
+ if (entry == NULL)
+ return NULL;
+ entry->name = kstrdup(name, GFP_KERNEL);
+ if (entry->name == NULL) {
+ kfree(entry);
+ return NULL;
+ }
+ entry->mode = S_IFREG | S_IRUGO;
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ mutex_init(&entry->access);
+ INIT_LIST_HEAD(&entry->children);
+ INIT_LIST_HEAD(&entry->list);
+ return entry;
+}
+
+/**
+ * snd_info_create_module_entry - create an info entry for the given module
+ * @module: the module pointer
+ * @name: the file name
+ * @parent: the parent directory
+ *
+ * Creates a new info entry and assigns it to the given module.
+ *
+ * Returns the pointer of the new instance, or NULL on failure.
+ */
+struct snd_info_entry *snd_info_create_module_entry(struct module * module,
+ const char *name,
+ struct snd_info_entry *parent)
+{
+ struct snd_info_entry *entry = snd_info_create_entry(name);
+ if (entry) {
+ entry->module = module;
+ entry->parent = parent;
+ }
+ return entry;
+}
+
+EXPORT_SYMBOL(snd_info_create_module_entry);
+
+/**
+ * snd_info_create_card_entry - create an info entry for the given card
+ * @card: the card instance
+ * @name: the file name
+ * @parent: the parent directory
+ *
+ * Creates a new info entry and assigns it to the given card.
+ *
+ * Returns the pointer of the new instance, or NULL on failure.
+ */
+struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
+ const char *name,
+ struct snd_info_entry * parent)
+{
+ struct snd_info_entry *entry = snd_info_create_entry(name);
+ if (entry) {
+ entry->module = card->module;
+ entry->card = card;
+ entry->parent = parent;
+ }
+ return entry;
+}
+
+EXPORT_SYMBOL(snd_info_create_card_entry);
+
+static void snd_info_disconnect(struct snd_info_entry *entry)
+{
+ struct list_head *p, *n;
+ struct proc_dir_entry *root;
+
+ list_for_each_safe(p, n, &entry->children) {
+ snd_info_disconnect(list_entry(p, struct snd_info_entry, list));
+ }
+
+ if (! entry->p)
+ return;
+ list_del_init(&entry->list);
+ root = entry->parent == NULL ? snd_proc_root : entry->parent->p;
+ snd_BUG_ON(!root);
+ snd_remove_proc_entry(root, entry->p);
+ entry->p = NULL;
+}
+
+static int snd_info_dev_free_entry(struct snd_device *device)
+{
+ struct snd_info_entry *entry = device->device_data;
+ snd_info_free_entry(entry);
+ return 0;
+}
+
+static int snd_info_dev_register_entry(struct snd_device *device)
+{
+ struct snd_info_entry *entry = device->device_data;
+ return snd_info_register(entry);
+}
+
+/**
+ * snd_card_proc_new - create an info entry for the given card
+ * @card: the card instance
+ * @name: the file name
+ * @entryp: the pointer to store the new info entry
+ *
+ * Creates a new info entry and assigns it to the given card.
+ * Unlike snd_info_create_card_entry(), this function registers the
+ * info entry as an ALSA device component, so that it can be
+ * unregistered/released without explicit call.
+ * Also, you don't have to register this entry via snd_info_register(),
+ * since this will be registered by snd_card_register() automatically.
+ *
+ * The parent is assumed as card->proc_root.
+ *
+ * For releasing this entry, use snd_device_free() instead of
+ * snd_info_free_entry().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_card_proc_new(struct snd_card *card, const char *name,
+ struct snd_info_entry **entryp)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_info_dev_free_entry,
+ .dev_register = snd_info_dev_register_entry,
+ /* disconnect is done via snd_info_card_disconnect() */
+ };
+ struct snd_info_entry *entry;
+ int err;
+
+ entry = snd_info_create_card_entry(card, name, card->proc_root);
+ if (! entry)
+ return -ENOMEM;
+ if ((err = snd_device_new(card, SNDRV_DEV_INFO, entry, &ops)) < 0) {
+ snd_info_free_entry(entry);
+ return err;
+ }
+ if (entryp)
+ *entryp = entry;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_card_proc_new);
+
+/**
+ * snd_info_free_entry - release the info entry
+ * @entry: the info entry
+ *
+ * Releases the info entry. Don't call this after registered.
+ */
+void snd_info_free_entry(struct snd_info_entry * entry)
+{
+ if (entry == NULL)
+ return;
+ if (entry->p) {
+ mutex_lock(&info_mutex);
+ snd_info_disconnect(entry);
+ mutex_unlock(&info_mutex);
+ }
+ kfree(entry->name);
+ if (entry->private_free)
+ entry->private_free(entry);
+ kfree(entry);
+}
+
+EXPORT_SYMBOL(snd_info_free_entry);
+
+/**
+ * snd_info_register - register the info entry
+ * @entry: the info entry
+ *
+ * Registers the proc info entry.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_info_register(struct snd_info_entry * entry)
+{
+ struct proc_dir_entry *root, *p = NULL;
+
+ if (snd_BUG_ON(!entry))
+ return -ENXIO;
+ root = entry->parent == NULL ? snd_proc_root : entry->parent->p;
+ mutex_lock(&info_mutex);
+ p = snd_create_proc_entry(entry->name, entry->mode, root);
+ if (!p) {
+ mutex_unlock(&info_mutex);
+ return -ENOMEM;
+ }
+ p->owner = entry->module;
+ if (!S_ISDIR(entry->mode))
+ p->proc_fops = &snd_info_entry_operations;
+ p->size = entry->size;
+ p->data = entry;
+ entry->p = p;
+ if (entry->parent)
+ list_add_tail(&entry->list, &entry->parent->children);
+ mutex_unlock(&info_mutex);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_info_register);
+
+/*
+
+ */
+
+static struct snd_info_entry *snd_info_version_entry;
+
+static void snd_info_version_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ snd_iprintf(buffer,
+ "Advanced Linux Sound Architecture Driver Version "
+ CONFIG_SND_VERSION CONFIG_SND_DATE ".\n"
+ );
+}
+
+static int __init snd_info_version_init(void)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_module_entry(THIS_MODULE, "version", NULL);
+ if (entry == NULL)
+ return -ENOMEM;
+ entry->c.text.read = snd_info_version_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ snd_info_version_entry = entry;
+ return 0;
+}
+
+static int __exit snd_info_version_done(void)
+{
+ snd_info_free_entry(snd_info_version_entry);
+ return 0;
+}
+
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c
new file mode 100644
index 0000000..e4af138
--- /dev/null
+++ b/sound/core/info_oss.c
@@ -0,0 +1,138 @@
+/*
+ * Information interface for ALSA driver
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/info.h>
+#include <sound/version.h>
+#include <linux/utsname.h>
+#include <linux/mutex.h>
+
+#if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS)
+
+/*
+ * OSS compatible part
+ */
+
+static DEFINE_MUTEX(strings);
+static char *snd_sndstat_strings[SNDRV_CARDS][SNDRV_OSS_INFO_DEV_COUNT];
+static struct snd_info_entry *snd_sndstat_proc_entry;
+
+int snd_oss_info_register(int dev, int num, char *string)
+{
+ char *x;
+
+ if (snd_BUG_ON(dev < 0 || dev >= SNDRV_OSS_INFO_DEV_COUNT))
+ return -ENXIO;
+ if (snd_BUG_ON(num < 0 || num >= SNDRV_CARDS))
+ return -ENXIO;
+ mutex_lock(&strings);
+ if (string == NULL) {
+ if ((x = snd_sndstat_strings[num][dev]) != NULL) {
+ kfree(x);
+ x = NULL;
+ }
+ } else {
+ x = kstrdup(string, GFP_KERNEL);
+ if (x == NULL) {
+ mutex_unlock(&strings);
+ return -ENOMEM;
+ }
+ }
+ snd_sndstat_strings[num][dev] = x;
+ mutex_unlock(&strings);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_oss_info_register);
+
+static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int dev)
+{
+ int idx, ok = -1;
+ char *str;
+
+ snd_iprintf(buf, "\n%s:", id);
+ mutex_lock(&strings);
+ for (idx = 0; idx < SNDRV_CARDS; idx++) {
+ str = snd_sndstat_strings[idx][dev];
+ if (str) {
+ if (ok < 0) {
+ snd_iprintf(buf, "\n");
+ ok++;
+ }
+ snd_iprintf(buf, "%i: %s\n", idx, str);
+ }
+ }
+ mutex_unlock(&strings);
+ if (ok < 0)
+ snd_iprintf(buf, " NOT ENABLED IN CONFIG\n");
+ return ok;
+}
+
+static void snd_sndstat_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_iprintf(buffer, "Sound Driver:3.8.1a-980706 (ALSA v" CONFIG_SND_VERSION " emulation code)\n");
+ snd_iprintf(buffer, "Kernel: %s %s %s %s %s\n",
+ init_utsname()->sysname,
+ init_utsname()->nodename,
+ init_utsname()->release,
+ init_utsname()->version,
+ init_utsname()->machine);
+ snd_iprintf(buffer, "Config options: 0\n");
+ snd_iprintf(buffer, "\nInstalled drivers: \n");
+ snd_iprintf(buffer, "Type 10: ALSA emulation\n");
+ snd_iprintf(buffer, "\nCard config: \n");
+ snd_card_info_read_oss(buffer);
+ snd_sndstat_show_strings(buffer, "Audio devices", SNDRV_OSS_INFO_DEV_AUDIO);
+ snd_sndstat_show_strings(buffer, "Synth devices", SNDRV_OSS_INFO_DEV_SYNTH);
+ snd_sndstat_show_strings(buffer, "Midi devices", SNDRV_OSS_INFO_DEV_MIDI);
+ snd_sndstat_show_strings(buffer, "Timers", SNDRV_OSS_INFO_DEV_TIMERS);
+ snd_sndstat_show_strings(buffer, "Mixers", SNDRV_OSS_INFO_DEV_MIXERS);
+}
+
+int snd_info_minor_register(void)
+{
+ struct snd_info_entry *entry;
+
+ memset(snd_sndstat_strings, 0, sizeof(snd_sndstat_strings));
+ if ((entry = snd_info_create_module_entry(THIS_MODULE, "sndstat", snd_oss_root)) != NULL) {
+ entry->c.text.read = snd_sndstat_proc_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ snd_sndstat_proc_entry = entry;
+ return 0;
+}
+
+int snd_info_minor_unregister(void)
+{
+ snd_info_free_entry(snd_sndstat_proc_entry);
+ snd_sndstat_proc_entry = NULL;
+ return 0;
+}
+
+#endif /* CONFIG_SND_OSSEMUL */
diff --git a/sound/core/init.c b/sound/core/init.c
new file mode 100644
index 0000000..b47ff8b
--- /dev/null
+++ b/sound/core/init.c
@@ -0,0 +1,850 @@
+/*
+ * Initialization routines
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/sched.h>
+#include <linux/file.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/ctype.h>
+#include <linux/pm.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+
+static DEFINE_SPINLOCK(shutdown_lock);
+static LIST_HEAD(shutdown_files);
+
+static const struct file_operations snd_shutdown_f_ops;
+
+static unsigned int snd_cards_lock; /* locked for registering/using */
+struct snd_card *snd_cards[SNDRV_CARDS];
+EXPORT_SYMBOL(snd_cards);
+
+static DEFINE_MUTEX(snd_card_mutex);
+
+static char *slots[SNDRV_CARDS];
+module_param_array(slots, charp, NULL, 0444);
+MODULE_PARM_DESC(slots, "Module names assigned to the slots.");
+
+/* return non-zero if the given index is reserved for the given
+ * module via slots option
+ */
+static int module_slot_match(struct module *module, int idx)
+{
+ int match = 1;
+#ifdef MODULE
+ const char *s1, *s2;
+
+ if (!module || !module->name || !slots[idx])
+ return 0;
+
+ s1 = module->name;
+ s2 = slots[idx];
+ if (*s2 == '!') {
+ match = 0; /* negative match */
+ s2++;
+ }
+ /* compare module name strings
+ * hyphens are handled as equivalent with underscore
+ */
+ for (;;) {
+ char c1 = *s1++;
+ char c2 = *s2++;
+ if (c1 == '-')
+ c1 = '_';
+ if (c2 == '-')
+ c2 = '_';
+ if (c1 != c2)
+ return !match;
+ if (!c1)
+ break;
+ }
+#endif /* MODULE */
+ return match;
+}
+
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag);
+EXPORT_SYMBOL(snd_mixer_oss_notify_callback);
+#endif
+
+#ifdef CONFIG_PROC_FS
+static void snd_card_id_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_iprintf(buffer, "%s\n", entry->card->id);
+}
+
+static inline int init_info_for_card(struct snd_card *card)
+{
+ int err;
+ struct snd_info_entry *entry;
+
+ if ((err = snd_info_card_register(card)) < 0) {
+ snd_printd("unable to create card info\n");
+ return err;
+ }
+ if ((entry = snd_info_create_card_entry(card, "id", card->proc_root)) == NULL) {
+ snd_printd("unable to create card entry\n");
+ return err;
+ }
+ entry->c.text.read = snd_card_id_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ card->proc_id = entry;
+ return 0;
+}
+#else /* !CONFIG_PROC_FS */
+#define init_info_for_card(card)
+#endif
+
+/**
+ * snd_card_new - create and initialize a soundcard structure
+ * @idx: card index (address) [0 ... (SNDRV_CARDS-1)]
+ * @xid: card identification (ASCII string)
+ * @module: top level module for locking
+ * @extra_size: allocate this extra size after the main soundcard structure
+ *
+ * Creates and initializes a soundcard structure.
+ *
+ * Returns kmallocated snd_card structure. Creates the ALSA control interface
+ * (which is blocked until snd_card_register function is called).
+ */
+struct snd_card *snd_card_new(int idx, const char *xid,
+ struct module *module, int extra_size)
+{
+ struct snd_card *card;
+ int err, idx2;
+
+ if (extra_size < 0)
+ extra_size = 0;
+ card = kzalloc(sizeof(*card) + extra_size, GFP_KERNEL);
+ if (card == NULL)
+ return NULL;
+ if (xid) {
+ if (!snd_info_check_reserved_words(xid))
+ goto __error;
+ strlcpy(card->id, xid, sizeof(card->id));
+ }
+ err = 0;
+ mutex_lock(&snd_card_mutex);
+ if (idx < 0) {
+ for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++)
+ /* idx == -1 == 0xffff means: take any free slot */
+ if (~snd_cards_lock & idx & 1<<idx2) {
+ if (module_slot_match(module, idx2)) {
+ idx = idx2;
+ break;
+ }
+ }
+ }
+ if (idx < 0) {
+ for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++)
+ /* idx == -1 == 0xffff means: take any free slot */
+ if (~snd_cards_lock & idx & 1<<idx2) {
+ if (!slots[idx2] || !*slots[idx2]) {
+ idx = idx2;
+ break;
+ }
+ }
+ }
+ if (idx < 0)
+ err = -ENODEV;
+ else if (idx < snd_ecards_limit) {
+ if (snd_cards_lock & (1 << idx))
+ err = -EBUSY; /* invalid */
+ } else if (idx >= SNDRV_CARDS)
+ err = -ENODEV;
+ if (err < 0) {
+ mutex_unlock(&snd_card_mutex);
+ snd_printk(KERN_ERR "cannot find the slot for index %d (range 0-%i), error: %d\n",
+ idx, snd_ecards_limit - 1, err);
+ goto __error;
+ }
+ snd_cards_lock |= 1 << idx; /* lock it */
+ if (idx >= snd_ecards_limit)
+ snd_ecards_limit = idx + 1; /* increase the limit */
+ mutex_unlock(&snd_card_mutex);
+ card->number = idx;
+ card->module = module;
+ INIT_LIST_HEAD(&card->devices);
+ init_rwsem(&card->controls_rwsem);
+ rwlock_init(&card->ctl_files_rwlock);
+ INIT_LIST_HEAD(&card->controls);
+ INIT_LIST_HEAD(&card->ctl_files);
+ spin_lock_init(&card->files_lock);
+ init_waitqueue_head(&card->shutdown_sleep);
+#ifdef CONFIG_PM
+ mutex_init(&card->power_lock);
+ init_waitqueue_head(&card->power_sleep);
+#endif
+ /* the control interface cannot be accessed from the user space until */
+ /* snd_cards_bitmask and snd_cards are set with snd_card_register */
+ if ((err = snd_ctl_create(card)) < 0) {
+ snd_printd("unable to register control minors\n");
+ goto __error;
+ }
+ if ((err = snd_info_card_create(card)) < 0) {
+ snd_printd("unable to create card info\n");
+ goto __error_ctl;
+ }
+ if (extra_size > 0)
+ card->private_data = (char *)card + sizeof(struct snd_card);
+ return card;
+
+ __error_ctl:
+ snd_device_free_all(card, SNDRV_DEV_CMD_PRE);
+ __error:
+ kfree(card);
+ return NULL;
+}
+
+EXPORT_SYMBOL(snd_card_new);
+
+/* return non-zero if a card is already locked */
+int snd_card_locked(int card)
+{
+ int locked;
+
+ mutex_lock(&snd_card_mutex);
+ locked = snd_cards_lock & (1 << card);
+ mutex_unlock(&snd_card_mutex);
+ return locked;
+}
+
+static loff_t snd_disconnect_llseek(struct file *file, loff_t offset, int orig)
+{
+ return -ENODEV;
+}
+
+static ssize_t snd_disconnect_read(struct file *file, char __user *buf,
+ size_t count, loff_t *offset)
+{
+ return -ENODEV;
+}
+
+static ssize_t snd_disconnect_write(struct file *file, const char __user *buf,
+ size_t count, loff_t *offset)
+{
+ return -ENODEV;
+}
+
+static int snd_disconnect_release(struct inode *inode, struct file *file)
+{
+ struct snd_monitor_file *df = NULL, *_df;
+
+ spin_lock(&shutdown_lock);
+ list_for_each_entry(_df, &shutdown_files, shutdown_list) {
+ if (_df->file == file) {
+ df = _df;
+ break;
+ }
+ }
+ spin_unlock(&shutdown_lock);
+
+ if (likely(df)) {
+ if ((file->f_flags & FASYNC) && df->disconnected_f_op->fasync)
+ df->disconnected_f_op->fasync(-1, file, 0);
+ return df->disconnected_f_op->release(inode, file);
+ }
+
+ panic("%s(%p, %p) failed!", __func__, inode, file);
+}
+
+static unsigned int snd_disconnect_poll(struct file * file, poll_table * wait)
+{
+ return POLLERR | POLLNVAL;
+}
+
+static long snd_disconnect_ioctl(struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return -ENODEV;
+}
+
+static int snd_disconnect_mmap(struct file *file, struct vm_area_struct *vma)
+{
+ return -ENODEV;
+}
+
+static int snd_disconnect_fasync(int fd, struct file *file, int on)
+{
+ return -ENODEV;
+}
+
+static const struct file_operations snd_shutdown_f_ops =
+{
+ .owner = THIS_MODULE,
+ .llseek = snd_disconnect_llseek,
+ .read = snd_disconnect_read,
+ .write = snd_disconnect_write,
+ .release = snd_disconnect_release,
+ .poll = snd_disconnect_poll,
+ .unlocked_ioctl = snd_disconnect_ioctl,
+#ifdef CONFIG_COMPAT
+ .compat_ioctl = snd_disconnect_ioctl,
+#endif
+ .mmap = snd_disconnect_mmap,
+ .fasync = snd_disconnect_fasync
+};
+
+/**
+ * snd_card_disconnect - disconnect all APIs from the file-operations (user space)
+ * @card: soundcard structure
+ *
+ * Disconnects all APIs from the file-operations (user space).
+ *
+ * Returns zero, otherwise a negative error code.
+ *
+ * Note: The current implementation replaces all active file->f_op with special
+ * dummy file operations (they do nothing except release).
+ */
+int snd_card_disconnect(struct snd_card *card)
+{
+ struct snd_monitor_file *mfile;
+ struct file *file;
+ int err;
+
+ if (!card)
+ return -EINVAL;
+
+ spin_lock(&card->files_lock);
+ if (card->shutdown) {
+ spin_unlock(&card->files_lock);
+ return 0;
+ }
+ card->shutdown = 1;
+ spin_unlock(&card->files_lock);
+
+ /* phase 1: disable fops (user space) operations for ALSA API */
+ mutex_lock(&snd_card_mutex);
+ snd_cards[card->number] = NULL;
+ snd_cards_lock &= ~(1 << card->number);
+ mutex_unlock(&snd_card_mutex);
+
+ /* phase 2: replace file->f_op with special dummy operations */
+
+ spin_lock(&card->files_lock);
+ mfile = card->files;
+ while (mfile) {
+ file = mfile->file;
+
+ /* it's critical part, use endless loop */
+ /* we have no room to fail */
+ mfile->disconnected_f_op = mfile->file->f_op;
+
+ spin_lock(&shutdown_lock);
+ list_add(&mfile->shutdown_list, &shutdown_files);
+ spin_unlock(&shutdown_lock);
+
+ mfile->file->f_op = &snd_shutdown_f_ops;
+ fops_get(mfile->file->f_op);
+
+ mfile = mfile->next;
+ }
+ spin_unlock(&card->files_lock);
+
+ /* phase 3: notify all connected devices about disconnection */
+ /* at this point, they cannot respond to any calls except release() */
+
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+ if (snd_mixer_oss_notify_callback)
+ snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_DISCONNECT);
+#endif
+
+ /* notify all devices that we are disconnected */
+ err = snd_device_disconnect_all(card);
+ if (err < 0)
+ snd_printk(KERN_ERR "not all devices for card %i can be disconnected\n", card->number);
+
+ snd_info_card_disconnect(card);
+#ifndef CONFIG_SYSFS_DEPRECATED
+ if (card->card_dev) {
+ device_unregister(card->card_dev);
+ card->card_dev = NULL;
+ }
+#endif
+#ifdef CONFIG_PM
+ wake_up(&card->power_sleep);
+#endif
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_card_disconnect);
+
+/**
+ * snd_card_free - frees given soundcard structure
+ * @card: soundcard structure
+ *
+ * This function releases the soundcard structure and the all assigned
+ * devices automatically. That is, you don't have to release the devices
+ * by yourself.
+ *
+ * Returns zero. Frees all associated devices and frees the control
+ * interface associated to given soundcard.
+ */
+static int snd_card_do_free(struct snd_card *card)
+{
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+ if (snd_mixer_oss_notify_callback)
+ snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE);
+#endif
+ if (snd_device_free_all(card, SNDRV_DEV_CMD_PRE) < 0) {
+ snd_printk(KERN_ERR "unable to free all devices (pre)\n");
+ /* Fatal, but this situation should never occur */
+ }
+ if (snd_device_free_all(card, SNDRV_DEV_CMD_NORMAL) < 0) {
+ snd_printk(KERN_ERR "unable to free all devices (normal)\n");
+ /* Fatal, but this situation should never occur */
+ }
+ if (snd_device_free_all(card, SNDRV_DEV_CMD_POST) < 0) {
+ snd_printk(KERN_ERR "unable to free all devices (post)\n");
+ /* Fatal, but this situation should never occur */
+ }
+ if (card->private_free)
+ card->private_free(card);
+ snd_info_free_entry(card->proc_id);
+ if (snd_info_card_free(card) < 0) {
+ snd_printk(KERN_WARNING "unable to free card info\n");
+ /* Not fatal error */
+ }
+ kfree(card);
+ return 0;
+}
+
+int snd_card_free_when_closed(struct snd_card *card)
+{
+ int free_now = 0;
+ int ret = snd_card_disconnect(card);
+ if (ret)
+ return ret;
+
+ spin_lock(&card->files_lock);
+ if (card->files == NULL)
+ free_now = 1;
+ else
+ card->free_on_last_close = 1;
+ spin_unlock(&card->files_lock);
+
+ if (free_now)
+ snd_card_do_free(card);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_card_free_when_closed);
+
+int snd_card_free(struct snd_card *card)
+{
+ int ret = snd_card_disconnect(card);
+ if (ret)
+ return ret;
+
+ /* wait, until all devices are ready for the free operation */
+ wait_event(card->shutdown_sleep, card->files == NULL);
+ snd_card_do_free(card);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_card_free);
+
+static void choose_default_id(struct snd_card *card)
+{
+ int i, len, idx_flag = 0, loops = SNDRV_CARDS;
+ char *id, *spos;
+
+ id = spos = card->shortname;
+ while (*id != '\0') {
+ if (*id == ' ')
+ spos = id + 1;
+ id++;
+ }
+ id = card->id;
+ while (*spos != '\0' && !isalnum(*spos))
+ spos++;
+ if (isdigit(*spos))
+ *id++ = isalpha(card->shortname[0]) ? card->shortname[0] : 'D';
+ while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) {
+ if (isalnum(*spos))
+ *id++ = *spos;
+ spos++;
+ }
+ *id = '\0';
+
+ id = card->id;
+
+ if (*id == '\0')
+ strcpy(id, "default");
+
+ while (1) {
+ if (loops-- == 0) {
+ snd_printk(KERN_ERR "unable to choose default card id (%s)\n", id);
+ strcpy(card->id, card->proc_root->name);
+ return;
+ }
+ if (!snd_info_check_reserved_words(id))
+ goto __change;
+ for (i = 0; i < snd_ecards_limit; i++) {
+ if (snd_cards[i] && !strcmp(snd_cards[i]->id, id))
+ goto __change;
+ }
+ break;
+
+ __change:
+ len = strlen(id);
+ if (idx_flag) {
+ if (id[len-1] != '9')
+ id[len-1]++;
+ else
+ id[len-1] = 'A';
+ } else if ((size_t)len <= sizeof(card->id) - 3) {
+ strcat(id, "_1");
+ idx_flag++;
+ } else {
+ spos = id + len - 2;
+ if ((size_t)len <= sizeof(card->id) - 2)
+ spos++;
+ *spos++ = '_';
+ *spos++ = '1';
+ *spos++ = '\0';
+ idx_flag++;
+ }
+ }
+}
+
+/**
+ * snd_card_register - register the soundcard
+ * @card: soundcard structure
+ *
+ * This function registers all the devices assigned to the soundcard.
+ * Until calling this, the ALSA control interface is blocked from the
+ * external accesses. Thus, you should call this function at the end
+ * of the initialization of the card.
+ *
+ * Returns zero otherwise a negative error code if the registrain failed.
+ */
+int snd_card_register(struct snd_card *card)
+{
+ int err;
+
+ if (snd_BUG_ON(!card))
+ return -EINVAL;
+#ifndef CONFIG_SYSFS_DEPRECATED
+ if (!card->card_dev) {
+ card->card_dev = device_create(sound_class, card->dev,
+ MKDEV(0, 0), NULL,
+ "card%i", card->number);
+ if (IS_ERR(card->card_dev))
+ card->card_dev = NULL;
+ }
+#endif
+ if ((err = snd_device_register_all(card)) < 0)
+ return err;
+ mutex_lock(&snd_card_mutex);
+ if (snd_cards[card->number]) {
+ /* already registered */
+ mutex_unlock(&snd_card_mutex);
+ return 0;
+ }
+ if (card->id[0] == '\0')
+ choose_default_id(card);
+ snd_cards[card->number] = card;
+ mutex_unlock(&snd_card_mutex);
+ init_info_for_card(card);
+#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+ if (snd_mixer_oss_notify_callback)
+ snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER);
+#endif
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_card_register);
+
+#ifdef CONFIG_PROC_FS
+static struct snd_info_entry *snd_card_info_entry;
+
+static void snd_card_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int idx, count;
+ struct snd_card *card;
+
+ for (idx = count = 0; idx < SNDRV_CARDS; idx++) {
+ mutex_lock(&snd_card_mutex);
+ if ((card = snd_cards[idx]) != NULL) {
+ count++;
+ snd_iprintf(buffer, "%2i [%-15s]: %s - %s\n",
+ idx,
+ card->id,
+ card->driver,
+ card->shortname);
+ snd_iprintf(buffer, " %s\n",
+ card->longname);
+ }
+ mutex_unlock(&snd_card_mutex);
+ }
+ if (!count)
+ snd_iprintf(buffer, "--- no soundcards ---\n");
+}
+
+#ifdef CONFIG_SND_OSSEMUL
+
+void snd_card_info_read_oss(struct snd_info_buffer *buffer)
+{
+ int idx, count;
+ struct snd_card *card;
+
+ for (idx = count = 0; idx < SNDRV_CARDS; idx++) {
+ mutex_lock(&snd_card_mutex);
+ if ((card = snd_cards[idx]) != NULL) {
+ count++;
+ snd_iprintf(buffer, "%s\n", card->longname);
+ }
+ mutex_unlock(&snd_card_mutex);
+ }
+ if (!count) {
+ snd_iprintf(buffer, "--- no soundcards ---\n");
+ }
+}
+
+#endif
+
+#ifdef MODULE
+static struct snd_info_entry *snd_card_module_info_entry;
+static void snd_card_module_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int idx;
+ struct snd_card *card;
+
+ for (idx = 0; idx < SNDRV_CARDS; idx++) {
+ mutex_lock(&snd_card_mutex);
+ if ((card = snd_cards[idx]) != NULL)
+ snd_iprintf(buffer, "%2i %s\n",
+ idx, card->module->name);
+ mutex_unlock(&snd_card_mutex);
+ }
+}
+#endif
+
+int __init snd_card_info_init(void)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_module_entry(THIS_MODULE, "cards", NULL);
+ if (! entry)
+ return -ENOMEM;
+ entry->c.text.read = snd_card_info_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ snd_card_info_entry = entry;
+
+#ifdef MODULE
+ entry = snd_info_create_module_entry(THIS_MODULE, "modules", NULL);
+ if (entry) {
+ entry->c.text.read = snd_card_module_info_read;
+ if (snd_info_register(entry) < 0)
+ snd_info_free_entry(entry);
+ else
+ snd_card_module_info_entry = entry;
+ }
+#endif
+
+ return 0;
+}
+
+int __exit snd_card_info_done(void)
+{
+ snd_info_free_entry(snd_card_info_entry);
+#ifdef MODULE
+ snd_info_free_entry(snd_card_module_info_entry);
+#endif
+ return 0;
+}
+
+#endif /* CONFIG_PROC_FS */
+
+/**
+ * snd_component_add - add a component string
+ * @card: soundcard structure
+ * @component: the component id string
+ *
+ * This function adds the component id string to the supported list.
+ * The component can be referred from the alsa-lib.
+ *
+ * Returns zero otherwise a negative error code.
+ */
+
+int snd_component_add(struct snd_card *card, const char *component)
+{
+ char *ptr;
+ int len = strlen(component);
+
+ ptr = strstr(card->components, component);
+ if (ptr != NULL) {
+ if (ptr[len] == '\0' || ptr[len] == ' ') /* already there */
+ return 1;
+ }
+ if (strlen(card->components) + 1 + len + 1 > sizeof(card->components)) {
+ snd_BUG();
+ return -ENOMEM;
+ }
+ if (card->components[0] != '\0')
+ strcat(card->components, " ");
+ strcat(card->components, component);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_component_add);
+
+/**
+ * snd_card_file_add - add the file to the file list of the card
+ * @card: soundcard structure
+ * @file: file pointer
+ *
+ * This function adds the file to the file linked-list of the card.
+ * This linked-list is used to keep tracking the connection state,
+ * and to avoid the release of busy resources by hotplug.
+ *
+ * Returns zero or a negative error code.
+ */
+int snd_card_file_add(struct snd_card *card, struct file *file)
+{
+ struct snd_monitor_file *mfile;
+
+ mfile = kmalloc(sizeof(*mfile), GFP_KERNEL);
+ if (mfile == NULL)
+ return -ENOMEM;
+ mfile->file = file;
+ mfile->disconnected_f_op = NULL;
+ mfile->next = NULL;
+ spin_lock(&card->files_lock);
+ if (card->shutdown) {
+ spin_unlock(&card->files_lock);
+ kfree(mfile);
+ return -ENODEV;
+ }
+ mfile->next = card->files;
+ card->files = mfile;
+ spin_unlock(&card->files_lock);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_card_file_add);
+
+/**
+ * snd_card_file_remove - remove the file from the file list
+ * @card: soundcard structure
+ * @file: file pointer
+ *
+ * This function removes the file formerly added to the card via
+ * snd_card_file_add() function.
+ * If all files are removed and snd_card_free_when_closed() was
+ * called beforehand, it processes the pending release of
+ * resources.
+ *
+ * Returns zero or a negative error code.
+ */
+int snd_card_file_remove(struct snd_card *card, struct file *file)
+{
+ struct snd_monitor_file *mfile, *pfile = NULL;
+ int last_close = 0;
+
+ spin_lock(&card->files_lock);
+ mfile = card->files;
+ while (mfile) {
+ if (mfile->file == file) {
+ if (pfile)
+ pfile->next = mfile->next;
+ else
+ card->files = mfile->next;
+ break;
+ }
+ pfile = mfile;
+ mfile = mfile->next;
+ }
+ if (mfile && mfile->disconnected_f_op) {
+ fops_put(mfile->disconnected_f_op);
+ spin_lock(&shutdown_lock);
+ list_del(&mfile->shutdown_list);
+ spin_unlock(&shutdown_lock);
+ }
+ if (card->files == NULL)
+ last_close = 1;
+ spin_unlock(&card->files_lock);
+ if (last_close) {
+ wake_up(&card->shutdown_sleep);
+ if (card->free_on_last_close)
+ snd_card_do_free(card);
+ }
+ if (!mfile) {
+ snd_printk(KERN_ERR "ALSA card file remove problem (%p)\n", file);
+ return -ENOENT;
+ }
+ kfree(mfile);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_card_file_remove);
+
+#ifdef CONFIG_PM
+/**
+ * snd_power_wait - wait until the power-state is changed.
+ * @card: soundcard structure
+ * @power_state: expected power state
+ *
+ * Waits until the power-state is changed.
+ *
+ * Note: the power lock must be active before call.
+ */
+int snd_power_wait(struct snd_card *card, unsigned int power_state)
+{
+ wait_queue_t wait;
+ int result = 0;
+
+ /* fastpath */
+ if (snd_power_get_state(card) == power_state)
+ return 0;
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&card->power_sleep, &wait);
+ while (1) {
+ if (card->shutdown) {
+ result = -ENODEV;
+ break;
+ }
+ if (snd_power_get_state(card) == power_state)
+ break;
+ set_current_state(TASK_UNINTERRUPTIBLE);
+ snd_power_unlock(card);
+ schedule_timeout(30 * HZ);
+ snd_power_lock(card);
+ }
+ remove_wait_queue(&card->power_sleep, &wait);
+ return result;
+}
+
+EXPORT_SYMBOL(snd_power_wait);
+#endif /* CONFIG_PM */
diff --git a/sound/core/isadma.c b/sound/core/isadma.c
new file mode 100644
index 0000000..79f0f16
--- /dev/null
+++ b/sound/core/isadma.c
@@ -0,0 +1,108 @@
+/*
+ * ISA DMA support functions
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * Defining following add some delay. Maybe this helps for some broken
+ * ISA DMA controllers.
+ */
+
+#undef HAVE_REALLY_SLOW_DMA_CONTROLLER
+
+#include <sound/core.h>
+#include <asm/dma.h>
+
+/**
+ * snd_dma_program - program an ISA DMA transfer
+ * @dma: the dma number
+ * @addr: the physical address of the buffer
+ * @size: the DMA transfer size
+ * @mode: the DMA transfer mode, DMA_MODE_XXX
+ *
+ * Programs an ISA DMA transfer for the given buffer.
+ */
+void snd_dma_program(unsigned long dma,
+ unsigned long addr, unsigned int size,
+ unsigned short mode)
+{
+ unsigned long flags;
+
+ flags = claim_dma_lock();
+ disable_dma(dma);
+ clear_dma_ff(dma);
+ set_dma_mode(dma, mode);
+ set_dma_addr(dma, addr);
+ set_dma_count(dma, size);
+ if (!(mode & DMA_MODE_NO_ENABLE))
+ enable_dma(dma);
+ release_dma_lock(flags);
+}
+
+EXPORT_SYMBOL(snd_dma_program);
+
+/**
+ * snd_dma_disable - stop the ISA DMA transfer
+ * @dma: the dma number
+ *
+ * Stops the ISA DMA transfer.
+ */
+void snd_dma_disable(unsigned long dma)
+{
+ unsigned long flags;
+
+ flags = claim_dma_lock();
+ clear_dma_ff(dma);
+ disable_dma(dma);
+ release_dma_lock(flags);
+}
+
+EXPORT_SYMBOL(snd_dma_disable);
+
+/**
+ * snd_dma_pointer - return the current pointer to DMA transfer buffer in bytes
+ * @dma: the dma number
+ * @size: the dma transfer size
+ *
+ * Returns the current pointer in DMA tranfer buffer in bytes
+ */
+unsigned int snd_dma_pointer(unsigned long dma, unsigned int size)
+{
+ unsigned long flags;
+ unsigned int result;
+
+ flags = claim_dma_lock();
+ clear_dma_ff(dma);
+ if (!isa_dma_bridge_buggy)
+ disable_dma(dma);
+ result = get_dma_residue(dma);
+ if (!isa_dma_bridge_buggy)
+ enable_dma(dma);
+ release_dma_lock(flags);
+#ifdef CONFIG_SND_DEBUG
+ if (result > size)
+ snd_printk(KERN_ERR "pointer (0x%x) for DMA #%ld is greater than transfer size (0x%x)\n", result, dma, size);
+#endif
+ if (result >= size || result == 0)
+ return 0;
+ else
+ return size - result;
+}
+
+EXPORT_SYMBOL(snd_dma_pointer);
diff --git a/sound/core/jack.c b/sound/core/jack.c
new file mode 100644
index 0000000..bd2d9e6
--- /dev/null
+++ b/sound/core/jack.c
@@ -0,0 +1,166 @@
+/*
+ * Jack abstraction layer
+ *
+ * Copyright 2008 Wolfson Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/input.h>
+#include <sound/jack.h>
+#include <sound/core.h>
+
+static int snd_jack_dev_free(struct snd_device *device)
+{
+ struct snd_jack *jack = device->device_data;
+
+ /* If the input device is registered with the input subsystem
+ * then we need to use a different deallocator. */
+ if (jack->registered)
+ input_unregister_device(jack->input_dev);
+ else
+ input_free_device(jack->input_dev);
+
+ kfree(jack);
+
+ return 0;
+}
+
+static int snd_jack_dev_register(struct snd_device *device)
+{
+ struct snd_jack *jack = device->device_data;
+ struct snd_card *card = device->card;
+ int err;
+
+ snprintf(jack->name, sizeof(jack->name), "%s %s",
+ card->longname, jack->id);
+ jack->input_dev->name = jack->name;
+
+ /* Default to the sound card device. */
+ if (!jack->input_dev->dev.parent)
+ jack->input_dev->dev.parent = card->dev;
+
+ err = input_register_device(jack->input_dev);
+ if (err == 0)
+ jack->registered = 1;
+
+ return err;
+}
+
+/**
+ * snd_jack_new - Create a new jack
+ * @card: the card instance
+ * @id: an identifying string for this jack
+ * @type: a bitmask of enum snd_jack_type values that can be detected by
+ * this jack
+ * @jjack: Used to provide the allocated jack object to the caller.
+ *
+ * Creates a new jack object.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ * On success jjack will be initialised.
+ */
+int snd_jack_new(struct snd_card *card, const char *id, int type,
+ struct snd_jack **jjack)
+{
+ struct snd_jack *jack;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_jack_dev_free,
+ .dev_register = snd_jack_dev_register,
+ };
+
+ jack = kzalloc(sizeof(struct snd_jack), GFP_KERNEL);
+ if (jack == NULL)
+ return -ENOMEM;
+
+ jack->id = id;
+
+ jack->input_dev = input_allocate_device();
+ if (jack->input_dev == NULL) {
+ err = -ENOMEM;
+ goto fail_input;
+ }
+
+ jack->input_dev->phys = "ALSA";
+
+ jack->type = type;
+
+ if (type & SND_JACK_HEADPHONE)
+ input_set_capability(jack->input_dev, EV_SW,
+ SW_HEADPHONE_INSERT);
+ if (type & SND_JACK_MICROPHONE)
+ input_set_capability(jack->input_dev, EV_SW,
+ SW_MICROPHONE_INSERT);
+
+ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops);
+ if (err < 0)
+ goto fail_input;
+
+ *jjack = jack;
+
+ return 0;
+
+fail_input:
+ input_free_device(jack->input_dev);
+ kfree(jack);
+ return err;
+}
+EXPORT_SYMBOL(snd_jack_new);
+
+/**
+ * snd_jack_set_parent - Set the parent device for a jack
+ *
+ * @jack: The jack to configure
+ * @parent: The device to set as parent for the jack.
+ *
+ * Set the parent for the jack input device in the device tree. This
+ * function is only valid prior to registration of the jack. If no
+ * parent is configured then the parent device will be the sound card.
+ */
+void snd_jack_set_parent(struct snd_jack *jack, struct device *parent)
+{
+ WARN_ON(jack->registered);
+
+ jack->input_dev->dev.parent = parent;
+}
+EXPORT_SYMBOL(snd_jack_set_parent);
+
+/**
+ * snd_jack_report - Report the current status of a jack
+ *
+ * @jack: The jack to report status for
+ * @status: The current status of the jack
+ */
+void snd_jack_report(struct snd_jack *jack, int status)
+{
+ if (!jack)
+ return;
+
+ if (jack->type & SND_JACK_HEADPHONE)
+ input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT,
+ status & SND_JACK_HEADPHONE);
+ if (jack->type & SND_JACK_MICROPHONE)
+ input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT,
+ status & SND_JACK_MICROPHONE);
+
+ input_sync(jack->input_dev);
+}
+EXPORT_SYMBOL(snd_jack_report);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("Jack detection support for ALSA");
+MODULE_LICENSE("GPL");
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
new file mode 100644
index 0000000..1b3534d
--- /dev/null
+++ b/sound/core/memalloc.c
@@ -0,0 +1,542 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * Generic memory allocators
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/proc_fs.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/mm.h>
+#include <linux/seq_file.h>
+#include <asm/uaccess.h>
+#include <linux/dma-mapping.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/memalloc.h>
+
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Memory allocator for ALSA system.");
+MODULE_LICENSE("GPL");
+
+
+/*
+ */
+
+static DEFINE_MUTEX(list_mutex);
+static LIST_HEAD(mem_list_head);
+
+/* buffer preservation list */
+struct snd_mem_list {
+ struct snd_dma_buffer buffer;
+ unsigned int id;
+ struct list_head list;
+};
+
+/* id for pre-allocated buffers */
+#define SNDRV_DMA_DEVICE_UNUSED (unsigned int)-1
+
+/*
+ *
+ * Generic memory allocators
+ *
+ */
+
+static long snd_allocated_pages; /* holding the number of allocated pages */
+
+static inline void inc_snd_pages(int order)
+{
+ snd_allocated_pages += 1 << order;
+}
+
+static inline void dec_snd_pages(int order)
+{
+ snd_allocated_pages -= 1 << order;
+}
+
+/**
+ * snd_malloc_pages - allocate pages with the given size
+ * @size: the size to allocate in bytes
+ * @gfp_flags: the allocation conditions, GFP_XXX
+ *
+ * Allocates the physically contiguous pages with the given size.
+ *
+ * Returns the pointer of the buffer, or NULL if no enoguh memory.
+ */
+void *snd_malloc_pages(size_t size, gfp_t gfp_flags)
+{
+ int pg;
+ void *res;
+
+ if (WARN_ON(!size))
+ return NULL;
+ if (WARN_ON(!gfp_flags))
+ return NULL;
+ gfp_flags |= __GFP_COMP; /* compound page lets parts be mapped */
+ pg = get_order(size);
+ if ((res = (void *) __get_free_pages(gfp_flags, pg)) != NULL)
+ inc_snd_pages(pg);
+ return res;
+}
+
+/**
+ * snd_free_pages - release the pages
+ * @ptr: the buffer pointer to release
+ * @size: the allocated buffer size
+ *
+ * Releases the buffer allocated via snd_malloc_pages().
+ */
+void snd_free_pages(void *ptr, size_t size)
+{
+ int pg;
+
+ if (ptr == NULL)
+ return;
+ pg = get_order(size);
+ dec_snd_pages(pg);
+ free_pages((unsigned long) ptr, pg);
+}
+
+/*
+ *
+ * Bus-specific memory allocators
+ *
+ */
+
+#ifdef CONFIG_HAS_DMA
+/* allocate the coherent DMA pages */
+static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *dma)
+{
+ int pg;
+ void *res;
+ gfp_t gfp_flags;
+
+ if (WARN_ON(!dma))
+ return NULL;
+ pg = get_order(size);
+ gfp_flags = GFP_KERNEL
+ | __GFP_COMP /* compound page lets parts be mapped */
+ | __GFP_NORETRY /* don't trigger OOM-killer */
+ | __GFP_NOWARN; /* no stack trace print - this call is non-critical */
+ res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags);
+ if (res != NULL)
+ inc_snd_pages(pg);
+
+ return res;
+}
+
+/* free the coherent DMA pages */
+static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr,
+ dma_addr_t dma)
+{
+ int pg;
+
+ if (ptr == NULL)
+ return;
+ pg = get_order(size);
+ dec_snd_pages(pg);
+ dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma);
+}
+#endif /* CONFIG_HAS_DMA */
+
+/*
+ *
+ * ALSA generic memory management
+ *
+ */
+
+
+/**
+ * snd_dma_alloc_pages - allocate the buffer area according to the given type
+ * @type: the DMA buffer type
+ * @device: the device pointer
+ * @size: the buffer size to allocate
+ * @dmab: buffer allocation record to store the allocated data
+ *
+ * Calls the memory-allocator function for the corresponding
+ * buffer type.
+ *
+ * Returns zero if the buffer with the given size is allocated successfuly,
+ * other a negative value at error.
+ */
+int snd_dma_alloc_pages(int type, struct device *device, size_t size,
+ struct snd_dma_buffer *dmab)
+{
+ if (WARN_ON(!size))
+ return -ENXIO;
+ if (WARN_ON(!dmab))
+ return -ENXIO;
+
+ dmab->dev.type = type;
+ dmab->dev.dev = device;
+ dmab->bytes = 0;
+ switch (type) {
+ case SNDRV_DMA_TYPE_CONTINUOUS:
+ dmab->area = snd_malloc_pages(size, (unsigned long)device);
+ dmab->addr = 0;
+ break;
+#ifdef CONFIG_HAS_DMA
+ case SNDRV_DMA_TYPE_DEV:
+ dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr);
+ break;
+ case SNDRV_DMA_TYPE_DEV_SG:
+ snd_malloc_sgbuf_pages(device, size, dmab, NULL);
+ break;
+#endif
+ default:
+ printk(KERN_ERR "snd-malloc: invalid device type %d\n", type);
+ dmab->area = NULL;
+ dmab->addr = 0;
+ return -ENXIO;
+ }
+ if (! dmab->area)
+ return -ENOMEM;
+ dmab->bytes = size;
+ return 0;
+}
+
+/**
+ * snd_dma_alloc_pages_fallback - allocate the buffer area according to the given type with fallback
+ * @type: the DMA buffer type
+ * @device: the device pointer
+ * @size: the buffer size to allocate
+ * @dmab: buffer allocation record to store the allocated data
+ *
+ * Calls the memory-allocator function for the corresponding
+ * buffer type. When no space is left, this function reduces the size and
+ * tries to allocate again. The size actually allocated is stored in
+ * res_size argument.
+ *
+ * Returns zero if the buffer with the given size is allocated successfuly,
+ * other a negative value at error.
+ */
+int snd_dma_alloc_pages_fallback(int type, struct device *device, size_t size,
+ struct snd_dma_buffer *dmab)
+{
+ int err;
+
+ while ((err = snd_dma_alloc_pages(type, device, size, dmab)) < 0) {
+ size_t aligned_size;
+ if (err != -ENOMEM)
+ return err;
+ if (size <= PAGE_SIZE)
+ return -ENOMEM;
+ aligned_size = PAGE_SIZE << get_order(size);
+ if (size != aligned_size)
+ size = aligned_size;
+ else
+ size >>= 1;
+ }
+ if (! dmab->area)
+ return -ENOMEM;
+ return 0;
+}
+
+
+/**
+ * snd_dma_free_pages - release the allocated buffer
+ * @dmab: the buffer allocation record to release
+ *
+ * Releases the allocated buffer via snd_dma_alloc_pages().
+ */
+void snd_dma_free_pages(struct snd_dma_buffer *dmab)
+{
+ switch (dmab->dev.type) {
+ case SNDRV_DMA_TYPE_CONTINUOUS:
+ snd_free_pages(dmab->area, dmab->bytes);
+ break;
+#ifdef CONFIG_HAS_DMA
+ case SNDRV_DMA_TYPE_DEV:
+ snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr);
+ break;
+ case SNDRV_DMA_TYPE_DEV_SG:
+ snd_free_sgbuf_pages(dmab);
+ break;
+#endif
+ default:
+ printk(KERN_ERR "snd-malloc: invalid device type %d\n", dmab->dev.type);
+ }
+}
+
+
+/**
+ * snd_dma_get_reserved - get the reserved buffer for the given device
+ * @dmab: the buffer allocation record to store
+ * @id: the buffer id
+ *
+ * Looks for the reserved-buffer list and re-uses if the same buffer
+ * is found in the list. When the buffer is found, it's removed from the free list.
+ *
+ * Returns the size of buffer if the buffer is found, or zero if not found.
+ */
+size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id)
+{
+ struct snd_mem_list *mem;
+
+ if (WARN_ON(!dmab))
+ return 0;
+
+ mutex_lock(&list_mutex);
+ list_for_each_entry(mem, &mem_list_head, list) {
+ if (mem->id == id &&
+ (mem->buffer.dev.dev == NULL || dmab->dev.dev == NULL ||
+ ! memcmp(&mem->buffer.dev, &dmab->dev, sizeof(dmab->dev)))) {
+ struct device *dev = dmab->dev.dev;
+ list_del(&mem->list);
+ *dmab = mem->buffer;
+ if (dmab->dev.dev == NULL)
+ dmab->dev.dev = dev;
+ kfree(mem);
+ mutex_unlock(&list_mutex);
+ return dmab->bytes;
+ }
+ }
+ mutex_unlock(&list_mutex);
+ return 0;
+}
+
+/**
+ * snd_dma_reserve_buf - reserve the buffer
+ * @dmab: the buffer to reserve
+ * @id: the buffer id
+ *
+ * Reserves the given buffer as a reserved buffer.
+ *
+ * Returns zero if successful, or a negative code at error.
+ */
+int snd_dma_reserve_buf(struct snd_dma_buffer *dmab, unsigned int id)
+{
+ struct snd_mem_list *mem;
+
+ if (WARN_ON(!dmab))
+ return -EINVAL;
+ mem = kmalloc(sizeof(*mem), GFP_KERNEL);
+ if (! mem)
+ return -ENOMEM;
+ mutex_lock(&list_mutex);
+ mem->buffer = *dmab;
+ mem->id = id;
+ list_add_tail(&mem->list, &mem_list_head);
+ mutex_unlock(&list_mutex);
+ return 0;
+}
+
+/*
+ * purge all reserved buffers
+ */
+static void free_all_reserved_pages(void)
+{
+ struct list_head *p;
+ struct snd_mem_list *mem;
+
+ mutex_lock(&list_mutex);
+ while (! list_empty(&mem_list_head)) {
+ p = mem_list_head.next;
+ mem = list_entry(p, struct snd_mem_list, list);
+ list_del(p);
+ snd_dma_free_pages(&mem->buffer);
+ kfree(mem);
+ }
+ mutex_unlock(&list_mutex);
+}
+
+
+#ifdef CONFIG_PROC_FS
+/*
+ * proc file interface
+ */
+#define SND_MEM_PROC_FILE "driver/snd-page-alloc"
+static struct proc_dir_entry *snd_mem_proc;
+
+static int snd_mem_proc_read(struct seq_file *seq, void *offset)
+{
+ long pages = snd_allocated_pages >> (PAGE_SHIFT-12);
+ struct snd_mem_list *mem;
+ int devno;
+ static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" };
+
+ mutex_lock(&list_mutex);
+ seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n",
+ pages * PAGE_SIZE, pages, PAGE_SIZE / 1024);
+ devno = 0;
+ list_for_each_entry(mem, &mem_list_head, list) {
+ devno++;
+ seq_printf(seq, "buffer %d : ID %08x : type %s\n",
+ devno, mem->id, types[mem->buffer.dev.type]);
+ seq_printf(seq, " addr = 0x%lx, size = %d bytes\n",
+ (unsigned long)mem->buffer.addr,
+ (int)mem->buffer.bytes);
+ }
+ mutex_unlock(&list_mutex);
+ return 0;
+}
+
+static int snd_mem_proc_open(struct inode *inode, struct file *file)
+{
+ return single_open(file, snd_mem_proc_read, NULL);
+}
+
+/* FIXME: for pci only - other bus? */
+#ifdef CONFIG_PCI
+#define gettoken(bufp) strsep(bufp, " \t\n")
+
+static ssize_t snd_mem_proc_write(struct file *file, const char __user * buffer,
+ size_t count, loff_t * ppos)
+{
+ char buf[128];
+ char *token, *p;
+
+ if (count > sizeof(buf) - 1)
+ return -EINVAL;
+ if (copy_from_user(buf, buffer, count))
+ return -EFAULT;
+ buf[count] = '\0';
+
+ p = buf;
+ token = gettoken(&p);
+ if (! token || *token == '#')
+ return count;
+ if (strcmp(token, "add") == 0) {
+ char *endp;
+ int vendor, device, size, buffers;
+ long mask;
+ int i, alloced;
+ struct pci_dev *pci;
+
+ if ((token = gettoken(&p)) == NULL ||
+ (vendor = simple_strtol(token, NULL, 0)) <= 0 ||
+ (token = gettoken(&p)) == NULL ||
+ (device = simple_strtol(token, NULL, 0)) <= 0 ||
+ (token = gettoken(&p)) == NULL ||
+ (mask = simple_strtol(token, NULL, 0)) < 0 ||
+ (token = gettoken(&p)) == NULL ||
+ (size = memparse(token, &endp)) < 64*1024 ||
+ size > 16*1024*1024 /* too big */ ||
+ (token = gettoken(&p)) == NULL ||
+ (buffers = simple_strtol(token, NULL, 0)) <= 0 ||
+ buffers > 4) {
+ printk(KERN_ERR "snd-page-alloc: invalid proc write format\n");
+ return count;
+ }
+ vendor &= 0xffff;
+ device &= 0xffff;
+
+ alloced = 0;
+ pci = NULL;
+ while ((pci = pci_get_device(vendor, device, pci)) != NULL) {
+ if (mask > 0 && mask < 0xffffffff) {
+ if (pci_set_dma_mask(pci, mask) < 0 ||
+ pci_set_consistent_dma_mask(pci, mask) < 0) {
+ printk(KERN_ERR "snd-page-alloc: cannot set DMA mask %lx for pci %04x:%04x\n", mask, vendor, device);
+ pci_dev_put(pci);
+ return count;
+ }
+ }
+ for (i = 0; i < buffers; i++) {
+ struct snd_dma_buffer dmab;
+ memset(&dmab, 0, sizeof(dmab));
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, &dmab) < 0) {
+ printk(KERN_ERR "snd-page-alloc: cannot allocate buffer pages (size = %d)\n", size);
+ pci_dev_put(pci);
+ return count;
+ }
+ snd_dma_reserve_buf(&dmab, snd_dma_pci_buf_id(pci));
+ }
+ alloced++;
+ }
+ if (! alloced) {
+ for (i = 0; i < buffers; i++) {
+ struct snd_dma_buffer dmab;
+ memset(&dmab, 0, sizeof(dmab));
+ /* FIXME: We can allocate only in ZONE_DMA
+ * without a device pointer!
+ */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, NULL,
+ size, &dmab) < 0) {
+ printk(KERN_ERR "snd-page-alloc: cannot allocate buffer pages (size = %d)\n", size);
+ break;
+ }
+ snd_dma_reserve_buf(&dmab, (unsigned int)((vendor << 16) | device));
+ }
+ }
+ } else if (strcmp(token, "erase") == 0)
+ /* FIXME: need for releasing each buffer chunk? */
+ free_all_reserved_pages();
+ else
+ printk(KERN_ERR "snd-page-alloc: invalid proc cmd\n");
+ return count;
+}
+#endif /* CONFIG_PCI */
+
+static const struct file_operations snd_mem_proc_fops = {
+ .owner = THIS_MODULE,
+ .open = snd_mem_proc_open,
+ .read = seq_read,
+#ifdef CONFIG_PCI
+ .write = snd_mem_proc_write,
+#endif
+ .llseek = seq_lseek,
+ .release = single_release,
+};
+
+#endif /* CONFIG_PROC_FS */
+
+/*
+ * module entry
+ */
+
+static int __init snd_mem_init(void)
+{
+#ifdef CONFIG_PROC_FS
+ snd_mem_proc = proc_create(SND_MEM_PROC_FILE, 0644, NULL,
+ &snd_mem_proc_fops);
+#endif
+ return 0;
+}
+
+static void __exit snd_mem_exit(void)
+{
+ remove_proc_entry(SND_MEM_PROC_FILE, NULL);
+ free_all_reserved_pages();
+ if (snd_allocated_pages > 0)
+ printk(KERN_ERR "snd-malloc: Memory leak? pages not freed = %li\n", snd_allocated_pages);
+}
+
+
+module_init(snd_mem_init)
+module_exit(snd_mem_exit)
+
+
+/*
+ * exports
+ */
+EXPORT_SYMBOL(snd_dma_alloc_pages);
+EXPORT_SYMBOL(snd_dma_alloc_pages_fallback);
+EXPORT_SYMBOL(snd_dma_free_pages);
+
+EXPORT_SYMBOL(snd_dma_get_reserved_buf);
+EXPORT_SYMBOL(snd_dma_reserve_buf);
+
+EXPORT_SYMBOL(snd_malloc_pages);
+EXPORT_SYMBOL(snd_free_pages);
diff --git a/sound/core/memory.c b/sound/core/memory.c
new file mode 100644
index 0000000..1161158
--- /dev/null
+++ b/sound/core/memory.c
@@ -0,0 +1,91 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ * Misc memory accessors
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <asm/uaccess.h>
+#include <sound/core.h>
+
+/**
+ * copy_to_user_fromio - copy data from mmio-space to user-space
+ * @dst: the destination pointer on user-space
+ * @src: the source pointer on mmio
+ * @count: the data size to copy in bytes
+ *
+ * Copies the data from mmio-space to user-space.
+ *
+ * Returns zero if successful, or non-zero on failure.
+ */
+int copy_to_user_fromio(void __user *dst, const volatile void __iomem *src, size_t count)
+{
+#if defined(__i386__) || defined(CONFIG_SPARC32)
+ return copy_to_user(dst, (const void __force*)src, count) ? -EFAULT : 0;
+#else
+ char buf[256];
+ while (count) {
+ size_t c = count;
+ if (c > sizeof(buf))
+ c = sizeof(buf);
+ memcpy_fromio(buf, (void __iomem *)src, c);
+ if (copy_to_user(dst, buf, c))
+ return -EFAULT;
+ count -= c;
+ dst += c;
+ src += c;
+ }
+ return 0;
+#endif
+}
+
+EXPORT_SYMBOL(copy_to_user_fromio);
+
+/**
+ * copy_from_user_toio - copy data from user-space to mmio-space
+ * @dst: the destination pointer on mmio-space
+ * @src: the source pointer on user-space
+ * @count: the data size to copy in bytes
+ *
+ * Copies the data from user-space to mmio-space.
+ *
+ * Returns zero if successful, or non-zero on failure.
+ */
+int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size_t count)
+{
+#if defined(__i386__) || defined(CONFIG_SPARC32)
+ return copy_from_user((void __force *)dst, src, count) ? -EFAULT : 0;
+#else
+ char buf[256];
+ while (count) {
+ size_t c = count;
+ if (c > sizeof(buf))
+ c = sizeof(buf);
+ if (copy_from_user(buf, src, c))
+ return -EFAULT;
+ memcpy_toio(dst, buf, c);
+ count -= c;
+ dst += c;
+ src += c;
+ }
+ return 0;
+#endif
+}
+
+EXPORT_SYMBOL(copy_from_user_toio);
diff --git a/sound/core/misc.c b/sound/core/misc.c
new file mode 100644
index 0000000..38524f6
--- /dev/null
+++ b/sound/core/misc.c
@@ -0,0 +1,106 @@
+/*
+ * Misc and compatibility things
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/ioport.h>
+#include <sound/core.h>
+
+void release_and_free_resource(struct resource *res)
+{
+ if (res) {
+ release_resource(res);
+ kfree(res);
+ }
+}
+
+EXPORT_SYMBOL(release_and_free_resource);
+
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+void snd_verbose_printk(const char *file, int line, const char *format, ...)
+{
+ va_list args;
+
+ if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') {
+ char tmp[] = "<0>";
+ tmp[1] = format[1];
+ printk("%sALSA %s:%d: ", tmp, file, line);
+ format += 3;
+ } else {
+ printk("ALSA %s:%d: ", file, line);
+ }
+ va_start(args, format);
+ vprintk(format, args);
+ va_end(args);
+}
+
+EXPORT_SYMBOL(snd_verbose_printk);
+#endif
+
+#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK)
+void snd_verbose_printd(const char *file, int line, const char *format, ...)
+{
+ va_list args;
+
+ if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') {
+ char tmp[] = "<0>";
+ tmp[1] = format[1];
+ printk("%sALSA %s:%d: ", tmp, file, line);
+ format += 3;
+ } else {
+ printk(KERN_DEBUG "ALSA %s:%d: ", file, line);
+ }
+ va_start(args, format);
+ vprintk(format, args);
+ va_end(args);
+
+}
+
+EXPORT_SYMBOL(snd_verbose_printd);
+#endif
+
+#ifdef CONFIG_PCI
+#include <linux/pci.h>
+/**
+ * snd_pci_quirk_lookup - look up a PCI SSID quirk list
+ * @pci: pci_dev handle
+ * @list: quirk list, terminated by a null entry
+ *
+ * Look through the given quirk list and finds a matching entry
+ * with the same PCI SSID. When subdevice is 0, all subdevice
+ * values may match.
+ *
+ * Returns the matched entry pointer, or NULL if nothing matched.
+ */
+const struct snd_pci_quirk *
+snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list)
+{
+ const struct snd_pci_quirk *q;
+
+ for (q = list; q->subvendor; q++)
+ if (q->subvendor == pci->subsystem_vendor &&
+ (!q->subdevice || q->subdevice == pci->subsystem_device))
+ return q;
+ return NULL;
+}
+
+EXPORT_SYMBOL(snd_pci_quirk_lookup);
+#endif
diff --git a/sound/core/oss/Makefile b/sound/core/oss/Makefile
new file mode 100644
index 0000000..10a7945
--- /dev/null
+++ b/sound/core/oss/Makefile
@@ -0,0 +1,13 @@
+#
+# Makefile for ALSA
+# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-mixer-oss-objs := mixer_oss.o
+
+snd-pcm-oss-y := pcm_oss.o
+snd-pcm-oss-$(CONFIG_SND_PCM_OSS_PLUGINS) += pcm_plugin.o \
+ io.o copy.o linear.o mulaw.o route.o rate.o
+
+obj-$(CONFIG_SND_MIXER_OSS) += snd-mixer-oss.o
+obj-$(CONFIG_SND_PCM_OSS) += snd-pcm-oss.o
diff --git a/sound/core/oss/copy.c b/sound/core/oss/copy.c
new file mode 100644
index 0000000..05b58d4
--- /dev/null
+++ b/sound/core/oss/copy.c
@@ -0,0 +1,92 @@
+/*
+ * Linear conversion Plug-In
+ * Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "pcm_plugin.h"
+
+static snd_pcm_sframes_t copy_transfer(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ unsigned int channel;
+ unsigned int nchannels;
+
+ if (snd_BUG_ON(!plugin || !src_channels || !dst_channels))
+ return -ENXIO;
+ if (frames == 0)
+ return 0;
+ nchannels = plugin->src_format.channels;
+ for (channel = 0; channel < nchannels; channel++) {
+ if (snd_BUG_ON(src_channels->area.first % 8 ||
+ src_channels->area.step % 8))
+ return -ENXIO;
+ if (snd_BUG_ON(dst_channels->area.first % 8 ||
+ dst_channels->area.step % 8))
+ return -ENXIO;
+ if (!src_channels->enabled) {
+ if (dst_channels->wanted)
+ snd_pcm_area_silence(&dst_channels->area, 0, frames, plugin->dst_format.format);
+ dst_channels->enabled = 0;
+ continue;
+ }
+ dst_channels->enabled = 1;
+ snd_pcm_area_copy(&src_channels->area, 0, &dst_channels->area, 0, frames, plugin->src_format.format);
+ src_channels++;
+ dst_channels++;
+ }
+ return frames;
+}
+
+int snd_pcm_plugin_build_copy(struct snd_pcm_substream *plug,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin)
+{
+ int err;
+ struct snd_pcm_plugin *plugin;
+ int width;
+
+ if (snd_BUG_ON(!r_plugin))
+ return -ENXIO;
+ *r_plugin = NULL;
+
+ if (snd_BUG_ON(src_format->format != dst_format->format))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->rate != dst_format->rate))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->channels != dst_format->channels))
+ return -ENXIO;
+
+ width = snd_pcm_format_physical_width(src_format->format);
+ if (snd_BUG_ON(width <= 0))
+ return -ENXIO;
+
+ err = snd_pcm_plugin_build(plug, "copy", src_format, dst_format,
+ 0, &plugin);
+ if (err < 0)
+ return err;
+ plugin->transfer = copy_transfer;
+ *r_plugin = plugin;
+ return 0;
+}
diff --git a/sound/core/oss/io.c b/sound/core/oss/io.c
new file mode 100644
index 0000000..6faa1d7
--- /dev/null
+++ b/sound/core/oss/io.c
@@ -0,0 +1,141 @@
+/*
+ * PCM I/O Plug-In Interface
+ * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "pcm_plugin.h"
+
+#define pcm_write(plug,buf,count) snd_pcm_oss_write3(plug,buf,count,1)
+#define pcm_writev(plug,vec,count) snd_pcm_oss_writev3(plug,vec,count,1)
+#define pcm_read(plug,buf,count) snd_pcm_oss_read3(plug,buf,count,1)
+#define pcm_readv(plug,vec,count) snd_pcm_oss_readv3(plug,vec,count,1)
+
+/*
+ * Basic io plugin
+ */
+
+static snd_pcm_sframes_t io_playback_transfer(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ if (snd_BUG_ON(!plugin))
+ return -ENXIO;
+ if (snd_BUG_ON(!src_channels))
+ return -ENXIO;
+ if (plugin->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED) {
+ return pcm_write(plugin->plug, src_channels->area.addr, frames);
+ } else {
+ int channel, channels = plugin->dst_format.channels;
+ void **bufs = (void**)plugin->extra_data;
+ if (snd_BUG_ON(!bufs))
+ return -ENXIO;
+ for (channel = 0; channel < channels; channel++) {
+ if (src_channels[channel].enabled)
+ bufs[channel] = src_channels[channel].area.addr;
+ else
+ bufs[channel] = NULL;
+ }
+ return pcm_writev(plugin->plug, bufs, frames);
+ }
+}
+
+static snd_pcm_sframes_t io_capture_transfer(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ if (snd_BUG_ON(!plugin))
+ return -ENXIO;
+ if (snd_BUG_ON(!dst_channels))
+ return -ENXIO;
+ if (plugin->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED) {
+ return pcm_read(plugin->plug, dst_channels->area.addr, frames);
+ } else {
+ int channel, channels = plugin->dst_format.channels;
+ void **bufs = (void**)plugin->extra_data;
+ if (snd_BUG_ON(!bufs))
+ return -ENXIO;
+ for (channel = 0; channel < channels; channel++) {
+ if (dst_channels[channel].enabled)
+ bufs[channel] = dst_channels[channel].area.addr;
+ else
+ bufs[channel] = NULL;
+ }
+ return pcm_readv(plugin->plug, bufs, frames);
+ }
+ return 0;
+}
+
+static snd_pcm_sframes_t io_src_channels(struct snd_pcm_plugin *plugin,
+ snd_pcm_uframes_t frames,
+ struct snd_pcm_plugin_channel **channels)
+{
+ int err;
+ unsigned int channel;
+ struct snd_pcm_plugin_channel *v;
+ err = snd_pcm_plugin_client_channels(plugin, frames, &v);
+ if (err < 0)
+ return err;
+ *channels = v;
+ if (plugin->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED) {
+ for (channel = 0; channel < plugin->src_format.channels; ++channel, ++v)
+ v->wanted = 1;
+ }
+ return frames;
+}
+
+int snd_pcm_plugin_build_io(struct snd_pcm_substream *plug,
+ struct snd_pcm_hw_params *params,
+ struct snd_pcm_plugin **r_plugin)
+{
+ int err;
+ struct snd_pcm_plugin_format format;
+ struct snd_pcm_plugin *plugin;
+
+ if (snd_BUG_ON(!r_plugin))
+ return -ENXIO;
+ *r_plugin = NULL;
+ if (snd_BUG_ON(!plug || !params))
+ return -ENXIO;
+ format.format = params_format(params);
+ format.rate = params_rate(params);
+ format.channels = params_channels(params);
+ err = snd_pcm_plugin_build(plug, "I/O io",
+ &format, &format,
+ sizeof(void *) * format.channels,
+ &plugin);
+ if (err < 0)
+ return err;
+ plugin->access = params_access(params);
+ if (snd_pcm_plug_stream(plug) == SNDRV_PCM_STREAM_PLAYBACK) {
+ plugin->transfer = io_playback_transfer;
+ if (plugin->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED)
+ plugin->client_channels = io_src_channels;
+ } else {
+ plugin->transfer = io_capture_transfer;
+ }
+
+ *r_plugin = plugin;
+ return 0;
+}
diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c
new file mode 100644
index 0000000..4c1d168
--- /dev/null
+++ b/sound/core/oss/linear.c
@@ -0,0 +1,180 @@
+/*
+ * Linear conversion Plug-In
+ * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>,
+ * Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "pcm_plugin.h"
+
+/*
+ * Basic linear conversion plugin
+ */
+
+struct linear_priv {
+ int cvt_endian; /* need endian conversion? */
+ unsigned int src_ofs; /* byte offset in source format */
+ unsigned int dst_ofs; /* byte soffset in destination format */
+ unsigned int copy_ofs; /* byte offset in temporary u32 data */
+ unsigned int dst_bytes; /* byte size of destination format */
+ unsigned int copy_bytes; /* bytes to copy per conversion */
+ unsigned int flip; /* MSB flip for signeness, done after endian conv */
+};
+
+static inline void do_convert(struct linear_priv *data,
+ unsigned char *dst, unsigned char *src)
+{
+ unsigned int tmp = 0;
+ unsigned char *p = (unsigned char *)&tmp;
+
+ memcpy(p + data->copy_ofs, src + data->src_ofs, data->copy_bytes);
+ if (data->cvt_endian)
+ tmp = swab32(tmp);
+ tmp ^= data->flip;
+ memcpy(dst, p + data->dst_ofs, data->dst_bytes);
+}
+
+static void convert(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ struct linear_priv *data = (struct linear_priv *)plugin->extra_data;
+ int channel;
+ int nchannels = plugin->src_format.channels;
+ for (channel = 0; channel < nchannels; ++channel) {
+ char *src;
+ char *dst;
+ int src_step, dst_step;
+ snd_pcm_uframes_t frames1;
+ if (!src_channels[channel].enabled) {
+ if (dst_channels[channel].wanted)
+ snd_pcm_area_silence(&dst_channels[channel].area, 0, frames, plugin->dst_format.format);
+ dst_channels[channel].enabled = 0;
+ continue;
+ }
+ dst_channels[channel].enabled = 1;
+ src = src_channels[channel].area.addr + src_channels[channel].area.first / 8;
+ dst = dst_channels[channel].area.addr + dst_channels[channel].area.first / 8;
+ src_step = src_channels[channel].area.step / 8;
+ dst_step = dst_channels[channel].area.step / 8;
+ frames1 = frames;
+ while (frames1-- > 0) {
+ do_convert(data, dst, src);
+ src += src_step;
+ dst += dst_step;
+ }
+ }
+}
+
+static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ struct linear_priv *data;
+
+ if (snd_BUG_ON(!plugin || !src_channels || !dst_channels))
+ return -ENXIO;
+ data = (struct linear_priv *)plugin->extra_data;
+ if (frames == 0)
+ return 0;
+#ifdef CONFIG_SND_DEBUG
+ {
+ unsigned int channel;
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ if (snd_BUG_ON(src_channels[channel].area.first % 8 ||
+ src_channels[channel].area.step % 8))
+ return -ENXIO;
+ if (snd_BUG_ON(dst_channels[channel].area.first % 8 ||
+ dst_channels[channel].area.step % 8))
+ return -ENXIO;
+ }
+ }
+#endif
+ convert(plugin, src_channels, dst_channels, frames);
+ return frames;
+}
+
+static void init_data(struct linear_priv *data, int src_format, int dst_format)
+{
+ int src_le, dst_le, src_bytes, dst_bytes;
+
+ src_bytes = snd_pcm_format_width(src_format) / 8;
+ dst_bytes = snd_pcm_format_width(dst_format) / 8;
+ src_le = snd_pcm_format_little_endian(src_format) > 0;
+ dst_le = snd_pcm_format_little_endian(dst_format) > 0;
+
+ data->dst_bytes = dst_bytes;
+ data->cvt_endian = src_le != dst_le;
+ data->copy_bytes = src_bytes < dst_bytes ? src_bytes : dst_bytes;
+ if (src_le) {
+ data->copy_ofs = 4 - data->copy_bytes;
+ data->src_ofs = src_bytes - data->copy_bytes;
+ } else
+ data->src_ofs = snd_pcm_format_physical_width(src_format) / 8 -
+ src_bytes;
+ if (dst_le)
+ data->dst_ofs = 4 - data->dst_bytes;
+ else
+ data->dst_ofs = snd_pcm_format_physical_width(dst_format) / 8 -
+ dst_bytes;
+ if (snd_pcm_format_signed(src_format) !=
+ snd_pcm_format_signed(dst_format)) {
+ if (dst_le)
+ data->flip = cpu_to_le32(0x80000000);
+ else
+ data->flip = cpu_to_be32(0x80000000);
+ }
+}
+
+int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin)
+{
+ int err;
+ struct linear_priv *data;
+ struct snd_pcm_plugin *plugin;
+
+ if (snd_BUG_ON(!r_plugin))
+ return -ENXIO;
+ *r_plugin = NULL;
+
+ if (snd_BUG_ON(src_format->rate != dst_format->rate))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->channels != dst_format->channels))
+ return -ENXIO;
+ if (snd_BUG_ON(!snd_pcm_format_linear(src_format->format) ||
+ !snd_pcm_format_linear(dst_format->format)))
+ return -ENXIO;
+
+ err = snd_pcm_plugin_build(plug, "linear format conversion",
+ src_format, dst_format,
+ sizeof(struct linear_priv), &plugin);
+ if (err < 0)
+ return err;
+ data = (struct linear_priv *)plugin->extra_data;
+ init_data(data, src_format->format, dst_format->format);
+ plugin->transfer = linear_transfer;
+ *r_plugin = plugin;
+ return 0;
+}
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
new file mode 100644
index 0000000..4690b8b
--- /dev/null
+++ b/sound/core/oss/mixer_oss.c
@@ -0,0 +1,1389 @@
+/*
+ * OSS emulation layer for the mixer interface
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/mixer_oss.h>
+#include <linux/soundcard.h>
+
+#define OSS_ALSAEMULVER _SIOR ('M', 249, int)
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Mixer OSS emulation for ALSA.");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_MIXER);
+
+static int snd_mixer_oss_open(struct inode *inode, struct file *file)
+{
+ struct snd_card *card;
+ struct snd_mixer_oss_file *fmixer;
+ int err;
+
+ card = snd_lookup_oss_minor_data(iminor(inode),
+ SNDRV_OSS_DEVICE_TYPE_MIXER);
+ if (card == NULL)
+ return -ENODEV;
+ if (card->mixer_oss == NULL)
+ return -ENODEV;
+ err = snd_card_file_add(card, file);
+ if (err < 0)
+ return err;
+ fmixer = kzalloc(sizeof(*fmixer), GFP_KERNEL);
+ if (fmixer == NULL) {
+ snd_card_file_remove(card, file);
+ return -ENOMEM;
+ }
+ fmixer->card = card;
+ fmixer->mixer = card->mixer_oss;
+ file->private_data = fmixer;
+ if (!try_module_get(card->module)) {
+ kfree(fmixer);
+ snd_card_file_remove(card, file);
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static int snd_mixer_oss_release(struct inode *inode, struct file *file)
+{
+ struct snd_mixer_oss_file *fmixer;
+
+ if (file->private_data) {
+ fmixer = (struct snd_mixer_oss_file *) file->private_data;
+ module_put(fmixer->card->module);
+ snd_card_file_remove(fmixer->card, file);
+ kfree(fmixer);
+ }
+ return 0;
+}
+
+static int snd_mixer_oss_info(struct snd_mixer_oss_file *fmixer,
+ mixer_info __user *_info)
+{
+ struct snd_card *card = fmixer->card;
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct mixer_info info;
+
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, mixer && mixer->id[0] ? mixer->id : card->driver, sizeof(info.id));
+ strlcpy(info.name, mixer && mixer->name[0] ? mixer->name : card->mixername, sizeof(info.name));
+ info.modify_counter = card->mixer_oss_change_count;
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_mixer_oss_info_obsolete(struct snd_mixer_oss_file *fmixer,
+ _old_mixer_info __user *_info)
+{
+ struct snd_card *card = fmixer->card;
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ _old_mixer_info info;
+
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, mixer && mixer->id[0] ? mixer->id : card->driver, sizeof(info.id));
+ strlcpy(info.name, mixer && mixer->name[0] ? mixer->name : card->mixername, sizeof(info.name));
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_mixer_oss_caps(struct snd_mixer_oss_file *fmixer)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ int result = 0;
+
+ if (mixer == NULL)
+ return -EIO;
+ if (mixer->get_recsrc && mixer->put_recsrc)
+ result |= SOUND_CAP_EXCL_INPUT;
+ return result;
+}
+
+static int snd_mixer_oss_devmask(struct snd_mixer_oss_file *fmixer)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct snd_mixer_oss_slot *pslot;
+ int result = 0, chn;
+
+ if (mixer == NULL)
+ return -EIO;
+ for (chn = 0; chn < 31; chn++) {
+ pslot = &mixer->slots[chn];
+ if (pslot->put_volume || pslot->put_recsrc)
+ result |= 1 << chn;
+ }
+ return result;
+}
+
+static int snd_mixer_oss_stereodevs(struct snd_mixer_oss_file *fmixer)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct snd_mixer_oss_slot *pslot;
+ int result = 0, chn;
+
+ if (mixer == NULL)
+ return -EIO;
+ for (chn = 0; chn < 31; chn++) {
+ pslot = &mixer->slots[chn];
+ if (pslot->put_volume && pslot->stereo)
+ result |= 1 << chn;
+ }
+ return result;
+}
+
+static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ int result = 0;
+
+ if (mixer == NULL)
+ return -EIO;
+ if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */
+ result = mixer->mask_recsrc;
+ } else {
+ struct snd_mixer_oss_slot *pslot;
+ int chn;
+ for (chn = 0; chn < 31; chn++) {
+ pslot = &mixer->slots[chn];
+ if (pslot->put_recsrc)
+ result |= 1 << chn;
+ }
+ }
+ return result;
+}
+
+static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ int result = 0;
+
+ if (mixer == NULL)
+ return -EIO;
+ if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */
+ int err;
+ if ((err = mixer->get_recsrc(fmixer, &result)) < 0)
+ return err;
+ result = 1 << result;
+ } else {
+ struct snd_mixer_oss_slot *pslot;
+ int chn;
+ for (chn = 0; chn < 31; chn++) {
+ pslot = &mixer->slots[chn];
+ if (pslot->get_recsrc) {
+ int active = 0;
+ pslot->get_recsrc(fmixer, pslot, &active);
+ if (active)
+ result |= 1 << chn;
+ }
+ }
+ }
+ return mixer->oss_recsrc = result;
+}
+
+static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsrc)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct snd_mixer_oss_slot *pslot;
+ int chn, active;
+ int result = 0;
+
+ if (mixer == NULL)
+ return -EIO;
+ if (mixer->get_recsrc && mixer->put_recsrc) { /* exclusive input */
+ if (recsrc & ~mixer->oss_recsrc)
+ recsrc &= ~mixer->oss_recsrc;
+ mixer->put_recsrc(fmixer, ffz(~recsrc));
+ mixer->get_recsrc(fmixer, &result);
+ result = 1 << result;
+ }
+ for (chn = 0; chn < 31; chn++) {
+ pslot = &mixer->slots[chn];
+ if (pslot->put_recsrc) {
+ active = (recsrc & (1 << chn)) ? 1 : 0;
+ pslot->put_recsrc(fmixer, pslot, active);
+ }
+ }
+ if (! result) {
+ for (chn = 0; chn < 31; chn++) {
+ pslot = &mixer->slots[chn];
+ if (pslot->get_recsrc) {
+ active = 0;
+ pslot->get_recsrc(fmixer, pslot, &active);
+ if (active)
+ result |= 1 << chn;
+ }
+ }
+ }
+ return result;
+}
+
+static int snd_mixer_oss_get_volume(struct snd_mixer_oss_file *fmixer, int slot)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct snd_mixer_oss_slot *pslot;
+ int result = 0, left, right;
+
+ if (mixer == NULL || slot > 30)
+ return -EIO;
+ pslot = &mixer->slots[slot];
+ left = pslot->volume[0];
+ right = pslot->volume[1];
+ if (pslot->get_volume)
+ result = pslot->get_volume(fmixer, pslot, &left, &right);
+ if (!pslot->stereo)
+ right = left;
+ if (snd_BUG_ON(left < 0 || left > 100))
+ return -EIO;
+ if (snd_BUG_ON(right < 0 || right > 100))
+ return -EIO;
+ if (result >= 0) {
+ pslot->volume[0] = left;
+ pslot->volume[1] = right;
+ result = (left & 0xff) | ((right & 0xff) << 8);
+ }
+ return result;
+}
+
+static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer,
+ int slot, int volume)
+{
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct snd_mixer_oss_slot *pslot;
+ int result = 0, left = volume & 0xff, right = (volume >> 8) & 0xff;
+
+ if (mixer == NULL || slot > 30)
+ return -EIO;
+ pslot = &mixer->slots[slot];
+ if (left > 100)
+ left = 100;
+ if (right > 100)
+ right = 100;
+ if (!pslot->stereo)
+ right = left;
+ if (pslot->put_volume)
+ result = pslot->put_volume(fmixer, pslot, left, right);
+ if (result < 0)
+ return result;
+ pslot->volume[0] = left;
+ pslot->volume[1] = right;
+ return (left & 0xff) | ((right & 0xff) << 8);
+}
+
+static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int cmd, unsigned long arg)
+{
+ void __user *argp = (void __user *)arg;
+ int __user *p = argp;
+ int tmp;
+
+ if (snd_BUG_ON(!fmixer))
+ return -ENXIO;
+ if (((cmd >> 8) & 0xff) == 'M') {
+ switch (cmd) {
+ case SOUND_MIXER_INFO:
+ return snd_mixer_oss_info(fmixer, argp);
+ case SOUND_OLD_MIXER_INFO:
+ return snd_mixer_oss_info_obsolete(fmixer, argp);
+ case SOUND_MIXER_WRITE_RECSRC:
+ if (get_user(tmp, p))
+ return -EFAULT;
+ tmp = snd_mixer_oss_set_recsrc(fmixer, tmp);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ case OSS_GETVERSION:
+ return put_user(SNDRV_OSS_VERSION, p);
+ case OSS_ALSAEMULVER:
+ return put_user(1, p);
+ case SOUND_MIXER_READ_DEVMASK:
+ tmp = snd_mixer_oss_devmask(fmixer);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ case SOUND_MIXER_READ_STEREODEVS:
+ tmp = snd_mixer_oss_stereodevs(fmixer);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ case SOUND_MIXER_READ_RECMASK:
+ tmp = snd_mixer_oss_recmask(fmixer);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ case SOUND_MIXER_READ_CAPS:
+ tmp = snd_mixer_oss_caps(fmixer);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ case SOUND_MIXER_READ_RECSRC:
+ tmp = snd_mixer_oss_get_recsrc(fmixer);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ }
+ }
+ if (cmd & SIOC_IN) {
+ if (get_user(tmp, p))
+ return -EFAULT;
+ tmp = snd_mixer_oss_set_volume(fmixer, cmd & 0xff, tmp);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ } else if (cmd & SIOC_OUT) {
+ tmp = snd_mixer_oss_get_volume(fmixer, cmd & 0xff);
+ if (tmp < 0)
+ return tmp;
+ return put_user(tmp, p);
+ }
+ return -ENXIO;
+}
+
+static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg);
+}
+
+int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg)
+{
+ struct snd_mixer_oss_file fmixer;
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ if (card->mixer_oss == NULL)
+ return -ENXIO;
+ memset(&fmixer, 0, sizeof(fmixer));
+ fmixer.card = card;
+ fmixer.mixer = card->mixer_oss;
+ return snd_mixer_oss_ioctl1(&fmixer, cmd, arg);
+}
+
+#ifdef CONFIG_COMPAT
+/* all compatible */
+#define snd_mixer_oss_ioctl_compat snd_mixer_oss_ioctl
+#else
+#define snd_mixer_oss_ioctl_compat NULL
+#endif
+
+/*
+ * REGISTRATION PART
+ */
+
+static const struct file_operations snd_mixer_oss_f_ops =
+{
+ .owner = THIS_MODULE,
+ .open = snd_mixer_oss_open,
+ .release = snd_mixer_oss_release,
+ .unlocked_ioctl = snd_mixer_oss_ioctl,
+ .compat_ioctl = snd_mixer_oss_ioctl_compat,
+};
+
+/*
+ * utilities
+ */
+
+static long snd_mixer_oss_conv(long val, long omin, long omax, long nmin, long nmax)
+{
+ long orange = omax - omin, nrange = nmax - nmin;
+
+ if (orange == 0)
+ return 0;
+ return ((nrange * (val - omin)) + (orange / 2)) / orange + nmin;
+}
+
+/* convert from alsa native to oss values (0-100) */
+static long snd_mixer_oss_conv1(long val, long min, long max, int *old)
+{
+ if (val == snd_mixer_oss_conv(*old, 0, 100, min, max))
+ return *old;
+ return snd_mixer_oss_conv(val, min, max, 0, 100);
+}
+
+/* convert from oss to alsa native values */
+static long snd_mixer_oss_conv2(long val, long min, long max)
+{
+ return snd_mixer_oss_conv(val, 0, 100, min, max);
+}
+
+#if 0
+static void snd_mixer_oss_recsrce_set(struct snd_card *card, int slot)
+{
+ struct snd_mixer_oss *mixer = card->mixer_oss;
+ if (mixer)
+ mixer->mask_recsrc |= 1 << slot;
+}
+
+static int snd_mixer_oss_recsrce_get(struct snd_card *card, int slot)
+{
+ struct snd_mixer_oss *mixer = card->mixer_oss;
+ if (mixer && (mixer->mask_recsrc & (1 << slot)))
+ return 1;
+ return 0;
+}
+#endif
+
+#define SNDRV_MIXER_OSS_SIGNATURE 0x65999250
+
+#define SNDRV_MIXER_OSS_ITEM_GLOBAL 0
+#define SNDRV_MIXER_OSS_ITEM_GSWITCH 1
+#define SNDRV_MIXER_OSS_ITEM_GROUTE 2
+#define SNDRV_MIXER_OSS_ITEM_GVOLUME 3
+#define SNDRV_MIXER_OSS_ITEM_PSWITCH 4
+#define SNDRV_MIXER_OSS_ITEM_PROUTE 5
+#define SNDRV_MIXER_OSS_ITEM_PVOLUME 6
+#define SNDRV_MIXER_OSS_ITEM_CSWITCH 7
+#define SNDRV_MIXER_OSS_ITEM_CROUTE 8
+#define SNDRV_MIXER_OSS_ITEM_CVOLUME 9
+#define SNDRV_MIXER_OSS_ITEM_CAPTURE 10
+
+#define SNDRV_MIXER_OSS_ITEM_COUNT 11
+
+#define SNDRV_MIXER_OSS_PRESENT_GLOBAL (1<<0)
+#define SNDRV_MIXER_OSS_PRESENT_GSWITCH (1<<1)
+#define SNDRV_MIXER_OSS_PRESENT_GROUTE (1<<2)
+#define SNDRV_MIXER_OSS_PRESENT_GVOLUME (1<<3)
+#define SNDRV_MIXER_OSS_PRESENT_PSWITCH (1<<4)
+#define SNDRV_MIXER_OSS_PRESENT_PROUTE (1<<5)
+#define SNDRV_MIXER_OSS_PRESENT_PVOLUME (1<<6)
+#define SNDRV_MIXER_OSS_PRESENT_CSWITCH (1<<7)
+#define SNDRV_MIXER_OSS_PRESENT_CROUTE (1<<8)
+#define SNDRV_MIXER_OSS_PRESENT_CVOLUME (1<<9)
+#define SNDRV_MIXER_OSS_PRESENT_CAPTURE (1<<10)
+
+struct slot {
+ unsigned int signature;
+ unsigned int present;
+ unsigned int channels;
+ unsigned int numid[SNDRV_MIXER_OSS_ITEM_COUNT];
+ unsigned int capture_item;
+ struct snd_mixer_oss_assign_table *assigned;
+ unsigned int allocated: 1;
+};
+
+#define ID_UNKNOWN ((unsigned int)-1)
+
+static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, const char *name, int index)
+{
+ struct snd_card *card = mixer->card;
+ struct snd_ctl_elem_id id;
+
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, name);
+ id.index = index;
+ return snd_ctl_find_id(card, &id);
+}
+
+static void snd_mixer_oss_get_volume1_vol(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ unsigned int numid,
+ int *left, int *right)
+{
+ struct snd_ctl_elem_info *uinfo;
+ struct snd_ctl_elem_value *uctl;
+ struct snd_kcontrol *kctl;
+ struct snd_card *card = fmixer->card;
+
+ if (numid == ID_UNKNOWN)
+ return;
+ down_read(&card->controls_rwsem);
+ if ((kctl = snd_ctl_find_numid(card, numid)) == NULL) {
+ up_read(&card->controls_rwsem);
+ return;
+ }
+ uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL);
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (uinfo == NULL || uctl == NULL)
+ goto __unalloc;
+ if (kctl->info(kctl, uinfo))
+ goto __unalloc;
+ if (kctl->get(kctl, uctl))
+ goto __unalloc;
+ if (uinfo->type == SNDRV_CTL_ELEM_TYPE_BOOLEAN &&
+ uinfo->value.integer.min == 0 && uinfo->value.integer.max == 1)
+ goto __unalloc;
+ *left = snd_mixer_oss_conv1(uctl->value.integer.value[0], uinfo->value.integer.min, uinfo->value.integer.max, &pslot->volume[0]);
+ if (uinfo->count > 1)
+ *right = snd_mixer_oss_conv1(uctl->value.integer.value[1], uinfo->value.integer.min, uinfo->value.integer.max, &pslot->volume[1]);
+ __unalloc:
+ up_read(&card->controls_rwsem);
+ kfree(uctl);
+ kfree(uinfo);
+}
+
+static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ unsigned int numid,
+ int *left, int *right,
+ int route)
+{
+ struct snd_ctl_elem_info *uinfo;
+ struct snd_ctl_elem_value *uctl;
+ struct snd_kcontrol *kctl;
+ struct snd_card *card = fmixer->card;
+
+ if (numid == ID_UNKNOWN)
+ return;
+ down_read(&card->controls_rwsem);
+ if ((kctl = snd_ctl_find_numid(card, numid)) == NULL) {
+ up_read(&card->controls_rwsem);
+ return;
+ }
+ uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL);
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (uinfo == NULL || uctl == NULL)
+ goto __unalloc;
+ if (kctl->info(kctl, uinfo))
+ goto __unalloc;
+ if (kctl->get(kctl, uctl))
+ goto __unalloc;
+ if (!uctl->value.integer.value[0]) {
+ *left = 0;
+ if (uinfo->count == 1)
+ *right = 0;
+ }
+ if (uinfo->count > 1 && !uctl->value.integer.value[route ? 3 : 1])
+ *right = 0;
+ __unalloc:
+ up_read(&card->controls_rwsem);
+ kfree(uctl);
+ kfree(uinfo);
+}
+
+static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ int *left, int *right)
+{
+ struct slot *slot = (struct slot *)pslot->private_data;
+
+ *left = *right = 100;
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) {
+ snd_mixer_oss_get_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
+ snd_mixer_oss_get_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) {
+ snd_mixer_oss_get_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GLOBAL], left, right);
+ }
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) {
+ snd_mixer_oss_get_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) {
+ snd_mixer_oss_get_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) {
+ snd_mixer_oss_get_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) {
+ snd_mixer_oss_get_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1);
+ }
+ return 0;
+}
+
+static void snd_mixer_oss_put_volume1_vol(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ unsigned int numid,
+ int left, int right)
+{
+ struct snd_ctl_elem_info *uinfo;
+ struct snd_ctl_elem_value *uctl;
+ struct snd_kcontrol *kctl;
+ struct snd_card *card = fmixer->card;
+ int res;
+
+ if (numid == ID_UNKNOWN)
+ return;
+ down_read(&card->controls_rwsem);
+ if ((kctl = snd_ctl_find_numid(card, numid)) == NULL)
+ return;
+ uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL);
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (uinfo == NULL || uctl == NULL)
+ goto __unalloc;
+ if (kctl->info(kctl, uinfo))
+ goto __unalloc;
+ if (uinfo->type == SNDRV_CTL_ELEM_TYPE_BOOLEAN &&
+ uinfo->value.integer.min == 0 && uinfo->value.integer.max == 1)
+ goto __unalloc;
+ uctl->value.integer.value[0] = snd_mixer_oss_conv2(left, uinfo->value.integer.min, uinfo->value.integer.max);
+ if (uinfo->count > 1)
+ uctl->value.integer.value[1] = snd_mixer_oss_conv2(right, uinfo->value.integer.min, uinfo->value.integer.max);
+ if ((res = kctl->put(kctl, uctl)) < 0)
+ goto __unalloc;
+ if (res > 0)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id);
+ __unalloc:
+ up_read(&card->controls_rwsem);
+ kfree(uctl);
+ kfree(uinfo);
+}
+
+static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ unsigned int numid,
+ int left, int right,
+ int route)
+{
+ struct snd_ctl_elem_info *uinfo;
+ struct snd_ctl_elem_value *uctl;
+ struct snd_kcontrol *kctl;
+ struct snd_card *card = fmixer->card;
+ int res;
+
+ if (numid == ID_UNKNOWN)
+ return;
+ down_read(&card->controls_rwsem);
+ if ((kctl = snd_ctl_find_numid(card, numid)) == NULL) {
+ up_read(&fmixer->card->controls_rwsem);
+ return;
+ }
+ uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL);
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (uinfo == NULL || uctl == NULL)
+ goto __unalloc;
+ if (kctl->info(kctl, uinfo))
+ goto __unalloc;
+ if (uinfo->count > 1) {
+ uctl->value.integer.value[0] = left > 0 ? 1 : 0;
+ uctl->value.integer.value[route ? 3 : 1] = right > 0 ? 1 : 0;
+ if (route) {
+ uctl->value.integer.value[1] =
+ uctl->value.integer.value[2] = 0;
+ }
+ } else {
+ uctl->value.integer.value[0] = (left > 0 || right > 0) ? 1 : 0;
+ }
+ if ((res = kctl->put(kctl, uctl)) < 0)
+ goto __unalloc;
+ if (res > 0)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id);
+ __unalloc:
+ up_read(&card->controls_rwsem);
+ kfree(uctl);
+ kfree(uinfo);
+}
+
+static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ int left, int right)
+{
+ struct slot *slot = (struct slot *)pslot->private_data;
+
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) {
+ snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME)
+ snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
+ snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) {
+ snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GLOBAL], left, right);
+ }
+ if (left || right) {
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH)
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0);
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH)
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0);
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE)
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1);
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE)
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1);
+ } else {
+ if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) {
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) {
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) {
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1);
+ } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) {
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1);
+ }
+ }
+ return 0;
+}
+
+static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ int *active)
+{
+ struct slot *slot = (struct slot *)pslot->private_data;
+ int left, right;
+
+ left = right = 1;
+ snd_mixer_oss_get_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], &left, &right, 0);
+ *active = (left || right) ? 1 : 0;
+ return 0;
+}
+
+static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ int *active)
+{
+ struct slot *slot = (struct slot *)pslot->private_data;
+ int left, right;
+
+ left = right = 1;
+ snd_mixer_oss_get_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], &left, &right, 1);
+ *active = (left || right) ? 1 : 0;
+ return 0;
+}
+
+static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ int active)
+{
+ struct slot *slot = (struct slot *)pslot->private_data;
+
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0);
+ return 0;
+}
+
+static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer,
+ struct snd_mixer_oss_slot *pslot,
+ int active)
+{
+ struct slot *slot = (struct slot *)pslot->private_data;
+
+ snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1);
+ return 0;
+}
+
+static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned int *active_index)
+{
+ struct snd_card *card = fmixer->card;
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct snd_kcontrol *kctl;
+ struct snd_mixer_oss_slot *pslot;
+ struct slot *slot;
+ struct snd_ctl_elem_info *uinfo;
+ struct snd_ctl_elem_value *uctl;
+ int err, idx;
+
+ uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL);
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (uinfo == NULL || uctl == NULL) {
+ err = -ENOMEM;
+ goto __unlock;
+ }
+ down_read(&card->controls_rwsem);
+ kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0);
+ if (! kctl) {
+ err = -ENOENT;
+ goto __unlock;
+ }
+ if ((err = kctl->info(kctl, uinfo)) < 0)
+ goto __unlock;
+ if ((err = kctl->get(kctl, uctl)) < 0)
+ goto __unlock;
+ for (idx = 0; idx < 32; idx++) {
+ if (!(mixer->mask_recsrc & (1 << idx)))
+ continue;
+ pslot = &mixer->slots[idx];
+ slot = (struct slot *)pslot->private_data;
+ if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE)
+ continue;
+ if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE))
+ continue;
+ if (slot->capture_item == uctl->value.enumerated.item[0]) {
+ *active_index = idx;
+ break;
+ }
+ }
+ err = 0;
+ __unlock:
+ up_read(&card->controls_rwsem);
+ kfree(uctl);
+ kfree(uinfo);
+ return err;
+}
+
+static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned int active_index)
+{
+ struct snd_card *card = fmixer->card;
+ struct snd_mixer_oss *mixer = fmixer->mixer;
+ struct snd_kcontrol *kctl;
+ struct snd_mixer_oss_slot *pslot;
+ struct slot *slot = NULL;
+ struct snd_ctl_elem_info *uinfo;
+ struct snd_ctl_elem_value *uctl;
+ int err;
+ unsigned int idx;
+
+ uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL);
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (uinfo == NULL || uctl == NULL) {
+ err = -ENOMEM;
+ goto __unlock;
+ }
+ down_read(&card->controls_rwsem);
+ kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0);
+ if (! kctl) {
+ err = -ENOENT;
+ goto __unlock;
+ }
+ if ((err = kctl->info(kctl, uinfo)) < 0)
+ goto __unlock;
+ for (idx = 0; idx < 32; idx++) {
+ if (!(mixer->mask_recsrc & (1 << idx)))
+ continue;
+ pslot = &mixer->slots[idx];
+ slot = (struct slot *)pslot->private_data;
+ if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE)
+ continue;
+ if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE))
+ continue;
+ if (idx == active_index)
+ break;
+ slot = NULL;
+ }
+ if (! slot)
+ goto __unlock;
+ for (idx = 0; idx < uinfo->count; idx++)
+ uctl->value.enumerated.item[idx] = slot->capture_item;
+ err = kctl->put(kctl, uctl);
+ if (err > 0)
+ snd_ctl_notify(fmixer->card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id);
+ err = 0;
+ __unlock:
+ up_read(&card->controls_rwsem);
+ kfree(uctl);
+ kfree(uinfo);
+ return err;
+}
+
+struct snd_mixer_oss_assign_table {
+ int oss_id;
+ const char *name;
+ int index;
+};
+
+static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *slot, const char *name, int index, int item)
+{
+ struct snd_ctl_elem_info *info;
+ struct snd_kcontrol *kcontrol;
+ struct snd_card *card = mixer->card;
+ int err;
+
+ down_read(&card->controls_rwsem);
+ kcontrol = snd_mixer_oss_test_id(mixer, name, index);
+ if (kcontrol == NULL) {
+ up_read(&card->controls_rwsem);
+ return 0;
+ }
+ info = kmalloc(sizeof(*info), GFP_KERNEL);
+ if (! info) {
+ up_read(&card->controls_rwsem);
+ return -ENOMEM;
+ }
+ if ((err = kcontrol->info(kcontrol, info)) < 0) {
+ up_read(&card->controls_rwsem);
+ kfree(info);
+ return err;
+ }
+ slot->numid[item] = kcontrol->id.numid;
+ up_read(&card->controls_rwsem);
+ if (info->count > slot->channels)
+ slot->channels = info->count;
+ slot->present |= 1 << item;
+ kfree(info);
+ return 0;
+}
+
+static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn)
+{
+ struct slot *p = (struct slot *)chn->private_data;
+ if (p) {
+ if (p->allocated && p->assigned) {
+ kfree(p->assigned->name);
+ kfree(p->assigned);
+ }
+ kfree(p);
+ }
+}
+
+static void mixer_slot_clear(struct snd_mixer_oss_slot *rslot)
+{
+ int idx = rslot->number; /* remember this */
+ if (rslot->private_free)
+ rslot->private_free(rslot);
+ memset(rslot, 0, sizeof(*rslot));
+ rslot->number = idx;
+}
+
+/* In a separate function to keep gcc 3.2 happy - do NOT merge this in
+ snd_mixer_oss_build_input! */
+static int snd_mixer_oss_build_test_all(struct snd_mixer_oss *mixer,
+ struct snd_mixer_oss_assign_table *ptr,
+ struct slot *slot)
+{
+ char str[64];
+ int err;
+
+ err = snd_mixer_oss_build_test(mixer, slot, ptr->name, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_GLOBAL);
+ if (err)
+ return err;
+ sprintf(str, "%s Switch", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_GSWITCH);
+ if (err)
+ return err;
+ sprintf(str, "%s Route", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_GROUTE);
+ if (err)
+ return err;
+ sprintf(str, "%s Volume", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_GVOLUME);
+ if (err)
+ return err;
+ sprintf(str, "%s Playback Switch", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_PSWITCH);
+ if (err)
+ return err;
+ sprintf(str, "%s Playback Route", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_PROUTE);
+ if (err)
+ return err;
+ sprintf(str, "%s Playback Volume", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_PVOLUME);
+ if (err)
+ return err;
+ sprintf(str, "%s Capture Switch", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_CSWITCH);
+ if (err)
+ return err;
+ sprintf(str, "%s Capture Route", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_CROUTE);
+ if (err)
+ return err;
+ sprintf(str, "%s Capture Volume", ptr->name);
+ err = snd_mixer_oss_build_test(mixer, slot, str, ptr->index,
+ SNDRV_MIXER_OSS_ITEM_CVOLUME);
+ if (err)
+ return err;
+
+ return 0;
+}
+
+/*
+ * build an OSS mixer element.
+ * ptr_allocated means the entry is dynamically allocated (change via proc file).
+ * when replace_old = 1, the old entry is replaced with the new one.
+ */
+static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mixer_oss_assign_table *ptr, int ptr_allocated, int replace_old)
+{
+ struct slot slot;
+ struct slot *pslot;
+ struct snd_kcontrol *kctl;
+ struct snd_mixer_oss_slot *rslot;
+ char str[64];
+
+ /* check if already assigned */
+ if (mixer->slots[ptr->oss_id].get_volume && ! replace_old)
+ return 0;
+
+ memset(&slot, 0, sizeof(slot));
+ memset(slot.numid, 0xff, sizeof(slot.numid)); /* ID_UNKNOWN */
+ if (snd_mixer_oss_build_test_all(mixer, ptr, &slot))
+ return 0;
+ down_read(&mixer->card->controls_rwsem);
+ if (ptr->index == 0 && (kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0)) != NULL) {
+ struct snd_ctl_elem_info *uinfo;
+
+ uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL);
+ if (! uinfo) {
+ up_read(&mixer->card->controls_rwsem);
+ return -ENOMEM;
+ }
+
+ if (kctl->info(kctl, uinfo)) {
+ up_read(&mixer->card->controls_rwsem);
+ return 0;
+ }
+ strcpy(str, ptr->name);
+ if (!strcmp(str, "Master"))
+ strcpy(str, "Mix");
+ if (!strcmp(str, "Master Mono"))
+ strcpy(str, "Mix Mono");
+ slot.capture_item = 0;
+ if (!strcmp(uinfo->value.enumerated.name, str)) {
+ slot.present |= SNDRV_MIXER_OSS_PRESENT_CAPTURE;
+ } else {
+ for (slot.capture_item = 1; slot.capture_item < uinfo->value.enumerated.items; slot.capture_item++) {
+ uinfo->value.enumerated.item = slot.capture_item;
+ if (kctl->info(kctl, uinfo)) {
+ up_read(&mixer->card->controls_rwsem);
+ return 0;
+ }
+ if (!strcmp(uinfo->value.enumerated.name, str)) {
+ slot.present |= SNDRV_MIXER_OSS_PRESENT_CAPTURE;
+ break;
+ }
+ }
+ }
+ kfree(uinfo);
+ }
+ up_read(&mixer->card->controls_rwsem);
+ if (slot.present != 0) {
+ pslot = kmalloc(sizeof(slot), GFP_KERNEL);
+ if (! pslot)
+ return -ENOMEM;
+ *pslot = slot;
+ pslot->signature = SNDRV_MIXER_OSS_SIGNATURE;
+ pslot->assigned = ptr;
+ pslot->allocated = ptr_allocated;
+ rslot = &mixer->slots[ptr->oss_id];
+ mixer_slot_clear(rslot);
+ rslot->stereo = slot.channels > 1 ? 1 : 0;
+ rslot->get_volume = snd_mixer_oss_get_volume1;
+ rslot->put_volume = snd_mixer_oss_put_volume1;
+ /* note: ES18xx have both Capture Source and XX Capture Volume !!! */
+ if (slot.present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) {
+ rslot->get_recsrc = snd_mixer_oss_get_recsrc1_sw;
+ rslot->put_recsrc = snd_mixer_oss_put_recsrc1_sw;
+ } else if (slot.present & SNDRV_MIXER_OSS_PRESENT_CROUTE) {
+ rslot->get_recsrc = snd_mixer_oss_get_recsrc1_route;
+ rslot->put_recsrc = snd_mixer_oss_put_recsrc1_route;
+ } else if (slot.present & SNDRV_MIXER_OSS_PRESENT_CAPTURE) {
+ mixer->mask_recsrc |= 1 << ptr->oss_id;
+ }
+ rslot->private_data = pslot;
+ rslot->private_free = snd_mixer_oss_slot_free;
+ return 1;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_PROC_FS
+/*
+ */
+#define MIXER_VOL(name) [SOUND_MIXER_##name] = #name
+static char *oss_mixer_names[SNDRV_OSS_MAX_MIXERS] = {
+ MIXER_VOL(VOLUME),
+ MIXER_VOL(BASS),
+ MIXER_VOL(TREBLE),
+ MIXER_VOL(SYNTH),
+ MIXER_VOL(PCM),
+ MIXER_VOL(SPEAKER),
+ MIXER_VOL(LINE),
+ MIXER_VOL(MIC),
+ MIXER_VOL(CD),
+ MIXER_VOL(IMIX),
+ MIXER_VOL(ALTPCM),
+ MIXER_VOL(RECLEV),
+ MIXER_VOL(IGAIN),
+ MIXER_VOL(OGAIN),
+ MIXER_VOL(LINE1),
+ MIXER_VOL(LINE2),
+ MIXER_VOL(LINE3),
+ MIXER_VOL(DIGITAL1),
+ MIXER_VOL(DIGITAL2),
+ MIXER_VOL(DIGITAL3),
+ MIXER_VOL(PHONEIN),
+ MIXER_VOL(PHONEOUT),
+ MIXER_VOL(VIDEO),
+ MIXER_VOL(RADIO),
+ MIXER_VOL(MONITOR),
+};
+
+/*
+ * /proc interface
+ */
+
+static void snd_mixer_oss_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_mixer_oss *mixer = entry->private_data;
+ int i;
+
+ mutex_lock(&mixer->reg_mutex);
+ for (i = 0; i < SNDRV_OSS_MAX_MIXERS; i++) {
+ struct slot *p;
+
+ if (! oss_mixer_names[i])
+ continue;
+ p = (struct slot *)mixer->slots[i].private_data;
+ snd_iprintf(buffer, "%s ", oss_mixer_names[i]);
+ if (p && p->assigned)
+ snd_iprintf(buffer, "\"%s\" %d\n",
+ p->assigned->name,
+ p->assigned->index);
+ else
+ snd_iprintf(buffer, "\"\" 0\n");
+ }
+ mutex_unlock(&mixer->reg_mutex);
+}
+
+static void snd_mixer_oss_proc_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_mixer_oss *mixer = entry->private_data;
+ char line[128], str[32], idxstr[16], *cptr;
+ int ch, idx;
+ struct snd_mixer_oss_assign_table *tbl;
+ struct slot *slot;
+
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ cptr = snd_info_get_str(str, line, sizeof(str));
+ for (ch = 0; ch < SNDRV_OSS_MAX_MIXERS; ch++)
+ if (oss_mixer_names[ch] && strcmp(oss_mixer_names[ch], str) == 0)
+ break;
+ if (ch >= SNDRV_OSS_MAX_MIXERS) {
+ snd_printk(KERN_ERR "mixer_oss: invalid OSS volume '%s'\n", str);
+ continue;
+ }
+ cptr = snd_info_get_str(str, cptr, sizeof(str));
+ if (! *str) {
+ /* remove the entry */
+ mutex_lock(&mixer->reg_mutex);
+ mixer_slot_clear(&mixer->slots[ch]);
+ mutex_unlock(&mixer->reg_mutex);
+ continue;
+ }
+ snd_info_get_str(idxstr, cptr, sizeof(idxstr));
+ idx = simple_strtoul(idxstr, NULL, 10);
+ if (idx >= 0x4000) { /* too big */
+ snd_printk(KERN_ERR "mixer_oss: invalid index %d\n", idx);
+ continue;
+ }
+ mutex_lock(&mixer->reg_mutex);
+ slot = (struct slot *)mixer->slots[ch].private_data;
+ if (slot && slot->assigned &&
+ slot->assigned->index == idx && ! strcmp(slot->assigned->name, str))
+ /* not changed */
+ goto __unlock;
+ tbl = kmalloc(sizeof(*tbl), GFP_KERNEL);
+ if (! tbl) {
+ snd_printk(KERN_ERR "mixer_oss: no memory\n");
+ goto __unlock;
+ }
+ tbl->oss_id = ch;
+ tbl->name = kstrdup(str, GFP_KERNEL);
+ if (! tbl->name) {
+ kfree(tbl);
+ goto __unlock;
+ }
+ tbl->index = idx;
+ if (snd_mixer_oss_build_input(mixer, tbl, 1, 1) <= 0) {
+ kfree(tbl->name);
+ kfree(tbl);
+ }
+ __unlock:
+ mutex_unlock(&mixer->reg_mutex);
+ }
+}
+
+static void snd_mixer_oss_proc_init(struct snd_mixer_oss *mixer)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_card_entry(mixer->card, "oss_mixer",
+ mixer->card->proc_root);
+ if (! entry)
+ return;
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = snd_mixer_oss_proc_read;
+ entry->c.text.write = snd_mixer_oss_proc_write;
+ entry->private_data = mixer;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ mixer->proc_entry = entry;
+}
+
+static void snd_mixer_oss_proc_done(struct snd_mixer_oss *mixer)
+{
+ snd_info_free_entry(mixer->proc_entry);
+ mixer->proc_entry = NULL;
+}
+#else /* !CONFIG_PROC_FS */
+#define snd_mixer_oss_proc_init(mix)
+#define snd_mixer_oss_proc_done(mix)
+#endif /* CONFIG_PROC_FS */
+
+static void snd_mixer_oss_build(struct snd_mixer_oss *mixer)
+{
+ static struct snd_mixer_oss_assign_table table[] = {
+ { SOUND_MIXER_VOLUME, "Master", 0 },
+ { SOUND_MIXER_VOLUME, "Front", 0 }, /* fallback */
+ { SOUND_MIXER_BASS, "Tone Control - Bass", 0 },
+ { SOUND_MIXER_TREBLE, "Tone Control - Treble", 0 },
+ { SOUND_MIXER_SYNTH, "Synth", 0 },
+ { SOUND_MIXER_SYNTH, "FM", 0 }, /* fallback */
+ { SOUND_MIXER_SYNTH, "Music", 0 }, /* fallback */
+ { SOUND_MIXER_PCM, "PCM", 0 },
+ { SOUND_MIXER_SPEAKER, "PC Speaker", 0 },
+ { SOUND_MIXER_LINE, "Line", 0 },
+ { SOUND_MIXER_MIC, "Mic", 0 },
+ { SOUND_MIXER_CD, "CD", 0 },
+ { SOUND_MIXER_IMIX, "Monitor Mix", 0 },
+ { SOUND_MIXER_ALTPCM, "PCM", 1 },
+ { SOUND_MIXER_ALTPCM, "Headphone", 0 }, /* fallback */
+ { SOUND_MIXER_ALTPCM, "Wave", 0 }, /* fallback */
+ { SOUND_MIXER_RECLEV, "-- nothing --", 0 },
+ { SOUND_MIXER_IGAIN, "Capture", 0 },
+ { SOUND_MIXER_OGAIN, "Playback", 0 },
+ { SOUND_MIXER_LINE1, "Aux", 0 },
+ { SOUND_MIXER_LINE2, "Aux", 1 },
+ { SOUND_MIXER_LINE3, "Aux", 2 },
+ { SOUND_MIXER_DIGITAL1, "Digital", 0 },
+ { SOUND_MIXER_DIGITAL1, "IEC958", 0 }, /* fallback */
+ { SOUND_MIXER_DIGITAL1, "IEC958 Optical", 0 }, /* fallback */
+ { SOUND_MIXER_DIGITAL1, "IEC958 Coaxial", 0 }, /* fallback */
+ { SOUND_MIXER_DIGITAL2, "Digital", 1 },
+ { SOUND_MIXER_DIGITAL3, "Digital", 2 },
+ { SOUND_MIXER_PHONEIN, "Phone", 0 },
+ { SOUND_MIXER_PHONEOUT, "Master Mono", 0 },
+ { SOUND_MIXER_PHONEOUT, "Speaker", 0 }, /*fallback*/
+ { SOUND_MIXER_PHONEOUT, "Mono", 0 }, /*fallback*/
+ { SOUND_MIXER_PHONEOUT, "Phone", 0 }, /* fallback */
+ { SOUND_MIXER_VIDEO, "Video", 0 },
+ { SOUND_MIXER_RADIO, "Radio", 0 },
+ { SOUND_MIXER_MONITOR, "Monitor", 0 }
+ };
+ unsigned int idx;
+
+ for (idx = 0; idx < ARRAY_SIZE(table); idx++)
+ snd_mixer_oss_build_input(mixer, &table[idx], 0, 0);
+ if (mixer->mask_recsrc) {
+ mixer->get_recsrc = snd_mixer_oss_get_recsrc2;
+ mixer->put_recsrc = snd_mixer_oss_put_recsrc2;
+ }
+}
+
+/*
+ *
+ */
+
+static int snd_mixer_oss_free1(void *private)
+{
+ struct snd_mixer_oss *mixer = private;
+ struct snd_card *card;
+ int idx;
+
+ if (!mixer)
+ return 0;
+ card = mixer->card;
+ if (snd_BUG_ON(mixer != card->mixer_oss))
+ return -ENXIO;
+ card->mixer_oss = NULL;
+ for (idx = 0; idx < SNDRV_OSS_MAX_MIXERS; idx++) {
+ struct snd_mixer_oss_slot *chn = &mixer->slots[idx];
+ if (chn->private_free)
+ chn->private_free(chn);
+ }
+ kfree(mixer);
+ return 0;
+}
+
+static int snd_mixer_oss_notify_handler(struct snd_card *card, int cmd)
+{
+ struct snd_mixer_oss *mixer;
+
+ if (cmd == SND_MIXER_OSS_NOTIFY_REGISTER) {
+ char name[128];
+ int idx, err;
+
+ mixer = kcalloc(2, sizeof(*mixer), GFP_KERNEL);
+ if (mixer == NULL)
+ return -ENOMEM;
+ mutex_init(&mixer->reg_mutex);
+ sprintf(name, "mixer%i%i", card->number, 0);
+ if ((err = snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_MIXER,
+ card, 0,
+ &snd_mixer_oss_f_ops, card,
+ name)) < 0) {
+ snd_printk(KERN_ERR "unable to register OSS mixer device %i:%i\n",
+ card->number, 0);
+ kfree(mixer);
+ return err;
+ }
+ mixer->oss_dev_alloc = 1;
+ mixer->card = card;
+ if (*card->mixername)
+ strlcpy(mixer->name, card->mixername, sizeof(mixer->name));
+ else
+ strlcpy(mixer->name, name, sizeof(mixer->name));
+#ifdef SNDRV_OSS_INFO_DEV_MIXERS
+ snd_oss_info_register(SNDRV_OSS_INFO_DEV_MIXERS,
+ card->number,
+ mixer->name);
+#endif
+ for (idx = 0; idx < SNDRV_OSS_MAX_MIXERS; idx++)
+ mixer->slots[idx].number = idx;
+ card->mixer_oss = mixer;
+ snd_mixer_oss_build(mixer);
+ snd_mixer_oss_proc_init(mixer);
+ } else {
+ mixer = card->mixer_oss;
+ if (mixer == NULL)
+ return 0;
+ if (mixer->oss_dev_alloc) {
+#ifdef SNDRV_OSS_INFO_DEV_MIXERS
+ snd_oss_info_unregister(SNDRV_OSS_INFO_DEV_MIXERS, mixer->card->number);
+#endif
+ snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_MIXER, mixer->card, 0);
+ mixer->oss_dev_alloc = 0;
+ }
+ if (cmd == SND_MIXER_OSS_NOTIFY_DISCONNECT)
+ return 0;
+ snd_mixer_oss_proc_done(mixer);
+ return snd_mixer_oss_free1(mixer);
+ }
+ return 0;
+}
+
+static int __init alsa_mixer_oss_init(void)
+{
+ int idx;
+
+ snd_mixer_oss_notify_callback = snd_mixer_oss_notify_handler;
+ for (idx = 0; idx < SNDRV_CARDS; idx++) {
+ if (snd_cards[idx])
+ snd_mixer_oss_notify_handler(snd_cards[idx], SND_MIXER_OSS_NOTIFY_REGISTER);
+ }
+ return 0;
+}
+
+static void __exit alsa_mixer_oss_exit(void)
+{
+ int idx;
+
+ snd_mixer_oss_notify_callback = NULL;
+ for (idx = 0; idx < SNDRV_CARDS; idx++) {
+ if (snd_cards[idx])
+ snd_mixer_oss_notify_handler(snd_cards[idx], SND_MIXER_OSS_NOTIFY_FREE);
+ }
+}
+
+module_init(alsa_mixer_oss_init)
+module_exit(alsa_mixer_oss_exit)
+
+EXPORT_SYMBOL(snd_mixer_oss_ioctl_card);
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
new file mode 100644
index 0000000..f7649d4
--- /dev/null
+++ b/sound/core/oss/mulaw.c
@@ -0,0 +1,344 @@
+/*
+ * Mu-Law conversion Plug-In Interface
+ * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
+ * Uros Bizjak <uros@kss-loka.si>
+ *
+ * Based on reference implementation by Sun Microsystems, Inc.
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "pcm_plugin.h"
+
+#define SIGN_BIT (0x80) /* Sign bit for a u-law byte. */
+#define QUANT_MASK (0xf) /* Quantization field mask. */
+#define NSEGS (8) /* Number of u-law segments. */
+#define SEG_SHIFT (4) /* Left shift for segment number. */
+#define SEG_MASK (0x70) /* Segment field mask. */
+
+static inline int val_seg(int val)
+{
+ int r = 0;
+ val >>= 7;
+ if (val & 0xf0) {
+ val >>= 4;
+ r += 4;
+ }
+ if (val & 0x0c) {
+ val >>= 2;
+ r += 2;
+ }
+ if (val & 0x02)
+ r += 1;
+ return r;
+}
+
+#define BIAS (0x84) /* Bias for linear code. */
+
+/*
+ * linear2ulaw() - Convert a linear PCM value to u-law
+ *
+ * In order to simplify the encoding process, the original linear magnitude
+ * is biased by adding 33 which shifts the encoding range from (0 - 8158) to
+ * (33 - 8191). The result can be seen in the following encoding table:
+ *
+ * Biased Linear Input Code Compressed Code
+ * ------------------------ ---------------
+ * 00000001wxyza 000wxyz
+ * 0000001wxyzab 001wxyz
+ * 000001wxyzabc 010wxyz
+ * 00001wxyzabcd 011wxyz
+ * 0001wxyzabcde 100wxyz
+ * 001wxyzabcdef 101wxyz
+ * 01wxyzabcdefg 110wxyz
+ * 1wxyzabcdefgh 111wxyz
+ *
+ * Each biased linear code has a leading 1 which identifies the segment
+ * number. The value of the segment number is equal to 7 minus the number
+ * of leading 0's. The quantization interval is directly available as the
+ * four bits wxyz. * The trailing bits (a - h) are ignored.
+ *
+ * Ordinarily the complement of the resulting code word is used for
+ * transmission, and so the code word is complemented before it is returned.
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+static unsigned char linear2ulaw(int pcm_val) /* 2's complement (16-bit range) */
+{
+ int mask;
+ int seg;
+ unsigned char uval;
+
+ /* Get the sign and the magnitude of the value. */
+ if (pcm_val < 0) {
+ pcm_val = BIAS - pcm_val;
+ mask = 0x7F;
+ } else {
+ pcm_val += BIAS;
+ mask = 0xFF;
+ }
+ if (pcm_val > 0x7FFF)
+ pcm_val = 0x7FFF;
+
+ /* Convert the scaled magnitude to segment number. */
+ seg = val_seg(pcm_val);
+
+ /*
+ * Combine the sign, segment, quantization bits;
+ * and complement the code word.
+ */
+ uval = (seg << 4) | ((pcm_val >> (seg + 3)) & 0xF);
+ return uval ^ mask;
+}
+
+/*
+ * ulaw2linear() - Convert a u-law value to 16-bit linear PCM
+ *
+ * First, a biased linear code is derived from the code word. An unbiased
+ * output can then be obtained by subtracting 33 from the biased code.
+ *
+ * Note that this function expects to be passed the complement of the
+ * original code word. This is in keeping with ISDN conventions.
+ */
+static int ulaw2linear(unsigned char u_val)
+{
+ int t;
+
+ /* Complement to obtain normal u-law value. */
+ u_val = ~u_val;
+
+ /*
+ * Extract and bias the quantization bits. Then
+ * shift up by the segment number and subtract out the bias.
+ */
+ t = ((u_val & QUANT_MASK) << 3) + BIAS;
+ t <<= ((unsigned)u_val & SEG_MASK) >> SEG_SHIFT;
+
+ return ((u_val & SIGN_BIT) ? (BIAS - t) : (t - BIAS));
+}
+
+/*
+ * Basic Mu-Law plugin
+ */
+
+typedef void (*mulaw_f)(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames);
+
+struct mulaw_priv {
+ mulaw_f func;
+ int cvt_endian; /* need endian conversion? */
+ unsigned int native_ofs; /* byte offset in native format */
+ unsigned int copy_ofs; /* byte offset in s16 format */
+ unsigned int native_bytes; /* byte size of the native format */
+ unsigned int copy_bytes; /* bytes to copy per conversion */
+ u16 flip; /* MSB flip for signedness, done after endian conversion */
+};
+
+static inline void cvt_s16_to_native(struct mulaw_priv *data,
+ unsigned char *dst, u16 sample)
+{
+ sample ^= data->flip;
+ if (data->cvt_endian)
+ sample = swab16(sample);
+ if (data->native_bytes > data->copy_bytes)
+ memset(dst, 0, data->native_bytes);
+ memcpy(dst + data->native_ofs, (char *)&sample + data->copy_ofs,
+ data->copy_bytes);
+}
+
+static void mulaw_decode(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data;
+ int channel;
+ int nchannels = plugin->src_format.channels;
+ for (channel = 0; channel < nchannels; ++channel) {
+ char *src;
+ char *dst;
+ int src_step, dst_step;
+ snd_pcm_uframes_t frames1;
+ if (!src_channels[channel].enabled) {
+ if (dst_channels[channel].wanted)
+ snd_pcm_area_silence(&dst_channels[channel].area, 0, frames, plugin->dst_format.format);
+ dst_channels[channel].enabled = 0;
+ continue;
+ }
+ dst_channels[channel].enabled = 1;
+ src = src_channels[channel].area.addr + src_channels[channel].area.first / 8;
+ dst = dst_channels[channel].area.addr + dst_channels[channel].area.first / 8;
+ src_step = src_channels[channel].area.step / 8;
+ dst_step = dst_channels[channel].area.step / 8;
+ frames1 = frames;
+ while (frames1-- > 0) {
+ signed short sample = ulaw2linear(*src);
+ cvt_s16_to_native(data, dst, sample);
+ src += src_step;
+ dst += dst_step;
+ }
+ }
+}
+
+static inline signed short cvt_native_to_s16(struct mulaw_priv *data,
+ unsigned char *src)
+{
+ u16 sample = 0;
+ memcpy((char *)&sample + data->copy_ofs, src + data->native_ofs,
+ data->copy_bytes);
+ if (data->cvt_endian)
+ sample = swab16(sample);
+ sample ^= data->flip;
+ return (signed short)sample;
+}
+
+static void mulaw_encode(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ struct mulaw_priv *data = (struct mulaw_priv *)plugin->extra_data;
+ int channel;
+ int nchannels = plugin->src_format.channels;
+ for (channel = 0; channel < nchannels; ++channel) {
+ char *src;
+ char *dst;
+ int src_step, dst_step;
+ snd_pcm_uframes_t frames1;
+ if (!src_channels[channel].enabled) {
+ if (dst_channels[channel].wanted)
+ snd_pcm_area_silence(&dst_channels[channel].area, 0, frames, plugin->dst_format.format);
+ dst_channels[channel].enabled = 0;
+ continue;
+ }
+ dst_channels[channel].enabled = 1;
+ src = src_channels[channel].area.addr + src_channels[channel].area.first / 8;
+ dst = dst_channels[channel].area.addr + dst_channels[channel].area.first / 8;
+ src_step = src_channels[channel].area.step / 8;
+ dst_step = dst_channels[channel].area.step / 8;
+ frames1 = frames;
+ while (frames1-- > 0) {
+ signed short sample = cvt_native_to_s16(data, src);
+ *dst = linear2ulaw(sample);
+ src += src_step;
+ dst += dst_step;
+ }
+ }
+}
+
+static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ struct mulaw_priv *data;
+
+ if (snd_BUG_ON(!plugin || !src_channels || !dst_channels))
+ return -ENXIO;
+ if (frames == 0)
+ return 0;
+#ifdef CONFIG_SND_DEBUG
+ {
+ unsigned int channel;
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ if (snd_BUG_ON(src_channels[channel].area.first % 8 ||
+ src_channels[channel].area.step % 8))
+ return -ENXIO;
+ if (snd_BUG_ON(dst_channels[channel].area.first % 8 ||
+ dst_channels[channel].area.step % 8))
+ return -ENXIO;
+ }
+ }
+#endif
+ data = (struct mulaw_priv *)plugin->extra_data;
+ data->func(plugin, src_channels, dst_channels, frames);
+ return frames;
+}
+
+static void init_data(struct mulaw_priv *data, int format)
+{
+#ifdef SNDRV_LITTLE_ENDIAN
+ data->cvt_endian = snd_pcm_format_big_endian(format) > 0;
+#else
+ data->cvt_endian = snd_pcm_format_little_endian(format) > 0;
+#endif
+ if (!snd_pcm_format_signed(format))
+ data->flip = 0x8000;
+ data->native_bytes = snd_pcm_format_physical_width(format) / 8;
+ data->copy_bytes = data->native_bytes < 2 ? 1 : 2;
+ if (snd_pcm_format_little_endian(format)) {
+ data->native_ofs = data->native_bytes - data->copy_bytes;
+ data->copy_ofs = 2 - data->copy_bytes;
+ } else {
+ /* S24 in 4bytes need an 1 byte offset */
+ data->native_ofs = data->native_bytes -
+ snd_pcm_format_width(format) / 8;
+ }
+}
+
+int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin)
+{
+ int err;
+ struct mulaw_priv *data;
+ struct snd_pcm_plugin *plugin;
+ struct snd_pcm_plugin_format *format;
+ mulaw_f func;
+
+ if (snd_BUG_ON(!r_plugin))
+ return -ENXIO;
+ *r_plugin = NULL;
+
+ if (snd_BUG_ON(src_format->rate != dst_format->rate))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->channels != dst_format->channels))
+ return -ENXIO;
+
+ if (dst_format->format == SNDRV_PCM_FORMAT_MU_LAW) {
+ format = src_format;
+ func = mulaw_encode;
+ }
+ else if (src_format->format == SNDRV_PCM_FORMAT_MU_LAW) {
+ format = dst_format;
+ func = mulaw_decode;
+ }
+ else {
+ snd_BUG();
+ return -EINVAL;
+ }
+ if (snd_BUG_ON(!snd_pcm_format_linear(format->format)))
+ return -ENXIO;
+
+ err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion",
+ src_format, dst_format,
+ sizeof(struct mulaw_priv), &plugin);
+ if (err < 0)
+ return err;
+ data = (struct mulaw_priv *)plugin->extra_data;
+ data->func = func;
+ init_data(data, format->format);
+ plugin->transfer = mulaw_transfer;
+ *r_plugin = plugin;
+ return 0;
+}
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
new file mode 100644
index 0000000..f04ff35
--- /dev/null
+++ b/sound/core/oss/pcm_oss.c
@@ -0,0 +1,3062 @@
+/*
+ * Digital Audio (PCM) abstract layer / OSS compatible
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#if 0
+#define PLUGIN_DEBUG
+#endif
+#if 0
+#define OSS_DEBUG
+#endif
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/vmalloc.h>
+#include <linux/moduleparam.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "pcm_plugin.h"
+#include <sound/info.h>
+#include <linux/soundcard.h>
+#include <sound/initval.h>
+
+#define OSS_ALSAEMULVER _SIOR ('M', 249, int)
+
+static int dsp_map[SNDRV_CARDS];
+static int adsp_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1};
+static int nonblock_open = 1;
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>");
+MODULE_DESCRIPTION("PCM OSS emulation for ALSA.");
+MODULE_LICENSE("GPL");
+module_param_array(dsp_map, int, NULL, 0444);
+MODULE_PARM_DESC(dsp_map, "PCM device number assigned to 1st OSS device.");
+module_param_array(adsp_map, int, NULL, 0444);
+MODULE_PARM_DESC(adsp_map, "PCM device number assigned to 2nd OSS device.");
+module_param(nonblock_open, bool, 0644);
+MODULE_PARM_DESC(nonblock_open, "Don't block opening busy PCM devices.");
+MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM);
+MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM1);
+
+extern int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg);
+static int snd_pcm_oss_get_rate(struct snd_pcm_oss_file *pcm_oss_file);
+static int snd_pcm_oss_get_channels(struct snd_pcm_oss_file *pcm_oss_file);
+static int snd_pcm_oss_get_format(struct snd_pcm_oss_file *pcm_oss_file);
+
+static inline mm_segment_t snd_enter_user(void)
+{
+ mm_segment_t fs = get_fs();
+ set_fs(get_ds());
+ return fs;
+}
+
+static inline void snd_leave_user(mm_segment_t fs)
+{
+ set_fs(fs);
+}
+
+/*
+ * helper functions to process hw_params
+ */
+static int snd_interval_refine_min(struct snd_interval *i, unsigned int min, int openmin)
+{
+ int changed = 0;
+ if (i->min < min) {
+ i->min = min;
+ i->openmin = openmin;
+ changed = 1;
+ } else if (i->min == min && !i->openmin && openmin) {
+ i->openmin = 1;
+ changed = 1;
+ }
+ if (i->integer) {
+ if (i->openmin) {
+ i->min++;
+ i->openmin = 0;
+ }
+ }
+ if (snd_interval_checkempty(i)) {
+ snd_interval_none(i);
+ return -EINVAL;
+ }
+ return changed;
+}
+
+static int snd_interval_refine_max(struct snd_interval *i, unsigned int max, int openmax)
+{
+ int changed = 0;
+ if (i->max > max) {
+ i->max = max;
+ i->openmax = openmax;
+ changed = 1;
+ } else if (i->max == max && !i->openmax && openmax) {
+ i->openmax = 1;
+ changed = 1;
+ }
+ if (i->integer) {
+ if (i->openmax) {
+ i->max--;
+ i->openmax = 0;
+ }
+ }
+ if (snd_interval_checkempty(i)) {
+ snd_interval_none(i);
+ return -EINVAL;
+ }
+ return changed;
+}
+
+static int snd_interval_refine_set(struct snd_interval *i, unsigned int val)
+{
+ struct snd_interval t;
+ t.empty = 0;
+ t.min = t.max = val;
+ t.openmin = t.openmax = 0;
+ t.integer = 1;
+ return snd_interval_refine(i, &t);
+}
+
+/**
+ * snd_pcm_hw_param_value_min
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @dir: pointer to the direction (-1,0,1) or NULL
+ *
+ * Return the minimum value for field PAR.
+ */
+static unsigned int
+snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir)
+{
+ if (hw_is_mask(var)) {
+ if (dir)
+ *dir = 0;
+ return snd_mask_min(hw_param_mask_c(params, var));
+ }
+ if (hw_is_interval(var)) {
+ const struct snd_interval *i = hw_param_interval_c(params, var);
+ if (dir)
+ *dir = i->openmin;
+ return snd_interval_min(i);
+ }
+ return -EINVAL;
+}
+
+/**
+ * snd_pcm_hw_param_value_max
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @dir: pointer to the direction (-1,0,1) or NULL
+ *
+ * Return the maximum value for field PAR.
+ */
+static unsigned int
+snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir)
+{
+ if (hw_is_mask(var)) {
+ if (dir)
+ *dir = 0;
+ return snd_mask_max(hw_param_mask_c(params, var));
+ }
+ if (hw_is_interval(var)) {
+ const struct snd_interval *i = hw_param_interval_c(params, var);
+ if (dir)
+ *dir = - (int) i->openmax;
+ return snd_interval_max(i);
+ }
+ return -EINVAL;
+}
+
+static int _snd_pcm_hw_param_mask(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var,
+ const struct snd_mask *val)
+{
+ int changed;
+ changed = snd_mask_refine(hw_param_mask(params, var), val);
+ if (changed) {
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ }
+ return changed;
+}
+
+static int snd_pcm_hw_param_mask(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var,
+ const struct snd_mask *val)
+{
+ int changed = _snd_pcm_hw_param_mask(params, var, val);
+ if (changed < 0)
+ return changed;
+ if (params->rmask) {
+ int err = snd_pcm_hw_refine(pcm, params);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int _snd_pcm_hw_param_min(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int dir)
+{
+ int changed;
+ int open = 0;
+ if (dir) {
+ if (dir > 0) {
+ open = 1;
+ } else if (dir < 0) {
+ if (val > 0) {
+ open = 1;
+ val--;
+ }
+ }
+ }
+ if (hw_is_mask(var))
+ changed = snd_mask_refine_min(hw_param_mask(params, var),
+ val + !!open);
+ else if (hw_is_interval(var))
+ changed = snd_interval_refine_min(hw_param_interval(params, var),
+ val, open);
+ else
+ return -EINVAL;
+ if (changed) {
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ }
+ return changed;
+}
+
+/**
+ * snd_pcm_hw_param_min
+ * @pcm: PCM instance
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @val: minimal value
+ * @dir: pointer to the direction (-1,0,1) or NULL
+ *
+ * Inside configuration space defined by PARAMS remove from PAR all
+ * values < VAL. Reduce configuration space accordingly.
+ * Return new minimum or -EINVAL if the configuration space is empty
+ */
+static int snd_pcm_hw_param_min(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int *dir)
+{
+ int changed = _snd_pcm_hw_param_min(params, var, val, dir ? *dir : 0);
+ if (changed < 0)
+ return changed;
+ if (params->rmask) {
+ int err = snd_pcm_hw_refine(pcm, params);
+ if (err < 0)
+ return err;
+ }
+ return snd_pcm_hw_param_value_min(params, var, dir);
+}
+
+static int _snd_pcm_hw_param_max(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int dir)
+{
+ int changed;
+ int open = 0;
+ if (dir) {
+ if (dir < 0) {
+ open = 1;
+ } else if (dir > 0) {
+ open = 1;
+ val++;
+ }
+ }
+ if (hw_is_mask(var)) {
+ if (val == 0 && open) {
+ snd_mask_none(hw_param_mask(params, var));
+ changed = -EINVAL;
+ } else
+ changed = snd_mask_refine_max(hw_param_mask(params, var),
+ val - !!open);
+ } else if (hw_is_interval(var))
+ changed = snd_interval_refine_max(hw_param_interval(params, var),
+ val, open);
+ else
+ return -EINVAL;
+ if (changed) {
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ }
+ return changed;
+}
+
+/**
+ * snd_pcm_hw_param_max
+ * @pcm: PCM instance
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @val: maximal value
+ * @dir: pointer to the direction (-1,0,1) or NULL
+ *
+ * Inside configuration space defined by PARAMS remove from PAR all
+ * values >= VAL + 1. Reduce configuration space accordingly.
+ * Return new maximum or -EINVAL if the configuration space is empty
+ */
+static int snd_pcm_hw_param_max(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int *dir)
+{
+ int changed = _snd_pcm_hw_param_max(params, var, val, dir ? *dir : 0);
+ if (changed < 0)
+ return changed;
+ if (params->rmask) {
+ int err = snd_pcm_hw_refine(pcm, params);
+ if (err < 0)
+ return err;
+ }
+ return snd_pcm_hw_param_value_max(params, var, dir);
+}
+
+static int boundary_sub(int a, int adir,
+ int b, int bdir,
+ int *c, int *cdir)
+{
+ adir = adir < 0 ? -1 : (adir > 0 ? 1 : 0);
+ bdir = bdir < 0 ? -1 : (bdir > 0 ? 1 : 0);
+ *c = a - b;
+ *cdir = adir - bdir;
+ if (*cdir == -2) {
+ (*c)--;
+ } else if (*cdir == 2) {
+ (*c)++;
+ }
+ return 0;
+}
+
+static int boundary_lt(unsigned int a, int adir,
+ unsigned int b, int bdir)
+{
+ if (adir < 0) {
+ a--;
+ adir = 1;
+ } else if (adir > 0)
+ adir = 1;
+ if (bdir < 0) {
+ b--;
+ bdir = 1;
+ } else if (bdir > 0)
+ bdir = 1;
+ return a < b || (a == b && adir < bdir);
+}
+
+/* Return 1 if min is nearer to best than max */
+static int boundary_nearer(int min, int mindir,
+ int best, int bestdir,
+ int max, int maxdir)
+{
+ int dmin, dmindir;
+ int dmax, dmaxdir;
+ boundary_sub(best, bestdir, min, mindir, &dmin, &dmindir);
+ boundary_sub(max, maxdir, best, bestdir, &dmax, &dmaxdir);
+ return boundary_lt(dmin, dmindir, dmax, dmaxdir);
+}
+
+/**
+ * snd_pcm_hw_param_near
+ * @pcm: PCM instance
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @best: value to set
+ * @dir: pointer to the direction (-1,0,1) or NULL
+ *
+ * Inside configuration space defined by PARAMS set PAR to the available value
+ * nearest to VAL. Reduce configuration space accordingly.
+ * This function cannot be called for SNDRV_PCM_HW_PARAM_ACCESS,
+ * SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_SUBFORMAT.
+ * Return the value found.
+ */
+static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, unsigned int best,
+ int *dir)
+{
+ struct snd_pcm_hw_params *save = NULL;
+ int v;
+ unsigned int saved_min;
+ int last = 0;
+ int min, max;
+ int mindir, maxdir;
+ int valdir = dir ? *dir : 0;
+ /* FIXME */
+ if (best > INT_MAX)
+ best = INT_MAX;
+ min = max = best;
+ mindir = maxdir = valdir;
+ if (maxdir > 0)
+ maxdir = 0;
+ else if (maxdir == 0)
+ maxdir = -1;
+ else {
+ maxdir = 1;
+ max--;
+ }
+ save = kmalloc(sizeof(*save), GFP_KERNEL);
+ if (save == NULL)
+ return -ENOMEM;
+ *save = *params;
+ saved_min = min;
+ min = snd_pcm_hw_param_min(pcm, params, var, min, &mindir);
+ if (min >= 0) {
+ struct snd_pcm_hw_params *params1;
+ if (max < 0)
+ goto _end;
+ if ((unsigned int)min == saved_min && mindir == valdir)
+ goto _end;
+ params1 = kmalloc(sizeof(*params1), GFP_KERNEL);
+ if (params1 == NULL) {
+ kfree(save);
+ return -ENOMEM;
+ }
+ *params1 = *save;
+ max = snd_pcm_hw_param_max(pcm, params1, var, max, &maxdir);
+ if (max < 0) {
+ kfree(params1);
+ goto _end;
+ }
+ if (boundary_nearer(max, maxdir, best, valdir, min, mindir)) {
+ *params = *params1;
+ last = 1;
+ }
+ kfree(params1);
+ } else {
+ *params = *save;
+ max = snd_pcm_hw_param_max(pcm, params, var, max, &maxdir);
+ if (max < 0)
+ return max;
+ last = 1;
+ }
+ _end:
+ kfree(save);
+ if (last)
+ v = snd_pcm_hw_param_last(pcm, params, var, dir);
+ else
+ v = snd_pcm_hw_param_first(pcm, params, var, dir);
+ snd_BUG_ON(v < 0);
+ return v;
+}
+
+static int _snd_pcm_hw_param_set(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int dir)
+{
+ int changed;
+ if (hw_is_mask(var)) {
+ struct snd_mask *m = hw_param_mask(params, var);
+ if (val == 0 && dir < 0) {
+ changed = -EINVAL;
+ snd_mask_none(m);
+ } else {
+ if (dir > 0)
+ val++;
+ else if (dir < 0)
+ val--;
+ changed = snd_mask_refine_set(hw_param_mask(params, var), val);
+ }
+ } else if (hw_is_interval(var)) {
+ struct snd_interval *i = hw_param_interval(params, var);
+ if (val == 0 && dir < 0) {
+ changed = -EINVAL;
+ snd_interval_none(i);
+ } else if (dir == 0)
+ changed = snd_interval_refine_set(i, val);
+ else {
+ struct snd_interval t;
+ t.openmin = 1;
+ t.openmax = 1;
+ t.empty = 0;
+ t.integer = 0;
+ if (dir < 0) {
+ t.min = val - 1;
+ t.max = val;
+ } else {
+ t.min = val;
+ t.max = val+1;
+ }
+ changed = snd_interval_refine(i, &t);
+ }
+ } else
+ return -EINVAL;
+ if (changed) {
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ }
+ return changed;
+}
+
+/**
+ * snd_pcm_hw_param_set
+ * @pcm: PCM instance
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @val: value to set
+ * @dir: pointer to the direction (-1,0,1) or NULL
+ *
+ * Inside configuration space defined by PARAMS remove from PAR all
+ * values != VAL. Reduce configuration space accordingly.
+ * Return VAL or -EINVAL if the configuration space is empty
+ */
+static int snd_pcm_hw_param_set(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int dir)
+{
+ int changed = _snd_pcm_hw_param_set(params, var, val, dir);
+ if (changed < 0)
+ return changed;
+ if (params->rmask) {
+ int err = snd_pcm_hw_refine(pcm, params);
+ if (err < 0)
+ return err;
+ }
+ return snd_pcm_hw_param_value(params, var, NULL);
+}
+
+static int _snd_pcm_hw_param_setinteger(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ int changed;
+ changed = snd_interval_setinteger(hw_param_interval(params, var));
+ if (changed) {
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ }
+ return changed;
+}
+
+/*
+ * plugin
+ */
+
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+static int snd_pcm_oss_plugin_clear(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_plugin *plugin, *next;
+
+ plugin = runtime->oss.plugin_first;
+ while (plugin) {
+ next = plugin->next;
+ snd_pcm_plugin_free(plugin);
+ plugin = next;
+ }
+ runtime->oss.plugin_first = runtime->oss.plugin_last = NULL;
+ return 0;
+}
+
+static int snd_pcm_plugin_insert(struct snd_pcm_plugin *plugin)
+{
+ struct snd_pcm_runtime *runtime = plugin->plug->runtime;
+ plugin->next = runtime->oss.plugin_first;
+ plugin->prev = NULL;
+ if (runtime->oss.plugin_first) {
+ runtime->oss.plugin_first->prev = plugin;
+ runtime->oss.plugin_first = plugin;
+ } else {
+ runtime->oss.plugin_last =
+ runtime->oss.plugin_first = plugin;
+ }
+ return 0;
+}
+
+int snd_pcm_plugin_append(struct snd_pcm_plugin *plugin)
+{
+ struct snd_pcm_runtime *runtime = plugin->plug->runtime;
+ plugin->next = NULL;
+ plugin->prev = runtime->oss.plugin_last;
+ if (runtime->oss.plugin_last) {
+ runtime->oss.plugin_last->next = plugin;
+ runtime->oss.plugin_last = plugin;
+ } else {
+ runtime->oss.plugin_last =
+ runtime->oss.plugin_first = plugin;
+ }
+ return 0;
+}
+#endif /* CONFIG_SND_PCM_OSS_PLUGINS */
+
+static long snd_pcm_oss_bytes(struct snd_pcm_substream *substream, long frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ long buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ long bytes = frames_to_bytes(runtime, frames);
+ if (buffer_size == runtime->oss.buffer_bytes)
+ return bytes;
+#if BITS_PER_LONG >= 64
+ return runtime->oss.buffer_bytes * bytes / buffer_size;
+#else
+ {
+ u64 bsize = (u64)runtime->oss.buffer_bytes * (u64)bytes;
+ u32 rem;
+ div64_32(&bsize, buffer_size, &rem);
+ return (long)bsize;
+ }
+#endif
+}
+
+static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ long buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ if (buffer_size == runtime->oss.buffer_bytes)
+ return bytes_to_frames(runtime, bytes);
+ return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes);
+}
+
+/* define extended formats in the recent OSS versions (if any) */
+/* linear formats */
+#define AFMT_S32_LE 0x00001000
+#define AFMT_S32_BE 0x00002000
+#define AFMT_S24_LE 0x00008000
+#define AFMT_S24_BE 0x00010000
+#define AFMT_S24_PACKED 0x00040000
+
+/* other supported formats */
+#define AFMT_FLOAT 0x00004000
+#define AFMT_SPDIF_RAW 0x00020000
+
+/* unsupported formats */
+#define AFMT_AC3 0x00000400
+#define AFMT_VORBIS 0x00000800
+
+static int snd_pcm_oss_format_from(int format)
+{
+ switch (format) {
+ case AFMT_MU_LAW: return SNDRV_PCM_FORMAT_MU_LAW;
+ case AFMT_A_LAW: return SNDRV_PCM_FORMAT_A_LAW;
+ case AFMT_IMA_ADPCM: return SNDRV_PCM_FORMAT_IMA_ADPCM;
+ case AFMT_U8: return SNDRV_PCM_FORMAT_U8;
+ case AFMT_S16_LE: return SNDRV_PCM_FORMAT_S16_LE;
+ case AFMT_S16_BE: return SNDRV_PCM_FORMAT_S16_BE;
+ case AFMT_S8: return SNDRV_PCM_FORMAT_S8;
+ case AFMT_U16_LE: return SNDRV_PCM_FORMAT_U16_LE;
+ case AFMT_U16_BE: return SNDRV_PCM_FORMAT_U16_BE;
+ case AFMT_MPEG: return SNDRV_PCM_FORMAT_MPEG;
+ case AFMT_S32_LE: return SNDRV_PCM_FORMAT_S32_LE;
+ case AFMT_S32_BE: return SNDRV_PCM_FORMAT_S32_BE;
+ case AFMT_S24_LE: return SNDRV_PCM_FORMAT_S24_LE;
+ case AFMT_S24_BE: return SNDRV_PCM_FORMAT_S24_BE;
+ case AFMT_S24_PACKED: return SNDRV_PCM_FORMAT_S24_3LE;
+ case AFMT_FLOAT: return SNDRV_PCM_FORMAT_FLOAT;
+ case AFMT_SPDIF_RAW: return SNDRV_PCM_FORMAT_IEC958_SUBFRAME;
+ default: return SNDRV_PCM_FORMAT_U8;
+ }
+}
+
+static int snd_pcm_oss_format_to(int format)
+{
+ switch (format) {
+ case SNDRV_PCM_FORMAT_MU_LAW: return AFMT_MU_LAW;
+ case SNDRV_PCM_FORMAT_A_LAW: return AFMT_A_LAW;
+ case SNDRV_PCM_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
+ case SNDRV_PCM_FORMAT_U8: return AFMT_U8;
+ case SNDRV_PCM_FORMAT_S16_LE: return AFMT_S16_LE;
+ case SNDRV_PCM_FORMAT_S16_BE: return AFMT_S16_BE;
+ case SNDRV_PCM_FORMAT_S8: return AFMT_S8;
+ case SNDRV_PCM_FORMAT_U16_LE: return AFMT_U16_LE;
+ case SNDRV_PCM_FORMAT_U16_BE: return AFMT_U16_BE;
+ case SNDRV_PCM_FORMAT_MPEG: return AFMT_MPEG;
+ case SNDRV_PCM_FORMAT_S32_LE: return AFMT_S32_LE;
+ case SNDRV_PCM_FORMAT_S32_BE: return AFMT_S32_BE;
+ case SNDRV_PCM_FORMAT_S24_LE: return AFMT_S24_LE;
+ case SNDRV_PCM_FORMAT_S24_BE: return AFMT_S24_BE;
+ case SNDRV_PCM_FORMAT_S24_3LE: return AFMT_S24_PACKED;
+ case SNDRV_PCM_FORMAT_FLOAT: return AFMT_FLOAT;
+ case SNDRV_PCM_FORMAT_IEC958_SUBFRAME: return AFMT_SPDIF_RAW;
+ default: return -EINVAL;
+ }
+}
+
+static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *oss_params,
+ struct snd_pcm_hw_params *slave_params)
+{
+ size_t s;
+ size_t oss_buffer_size, oss_period_size, oss_periods;
+ size_t min_period_size, max_period_size;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ size_t oss_frame_size;
+
+ oss_frame_size = snd_pcm_format_physical_width(params_format(oss_params)) *
+ params_channels(oss_params) / 8;
+
+ oss_buffer_size = snd_pcm_plug_client_size(substream,
+ snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size;
+ oss_buffer_size = 1 << ld2(oss_buffer_size);
+ if (atomic_read(&substream->mmap_count)) {
+ if (oss_buffer_size > runtime->oss.mmap_bytes)
+ oss_buffer_size = runtime->oss.mmap_bytes;
+ }
+
+ if (substream->oss.setup.period_size > 16)
+ oss_period_size = substream->oss.setup.period_size;
+ else if (runtime->oss.fragshift) {
+ oss_period_size = 1 << runtime->oss.fragshift;
+ if (oss_period_size > oss_buffer_size / 2)
+ oss_period_size = oss_buffer_size / 2;
+ } else {
+ int sd;
+ size_t bytes_per_sec = params_rate(oss_params) * snd_pcm_format_physical_width(params_format(oss_params)) * params_channels(oss_params) / 8;
+
+ oss_period_size = oss_buffer_size;
+ do {
+ oss_period_size /= 2;
+ } while (oss_period_size > bytes_per_sec);
+ if (runtime->oss.subdivision == 0) {
+ sd = 4;
+ if (oss_period_size / sd > 4096)
+ sd *= 2;
+ if (oss_period_size / sd < 4096)
+ sd = 1;
+ } else
+ sd = runtime->oss.subdivision;
+ oss_period_size /= sd;
+ if (oss_period_size < 16)
+ oss_period_size = 16;
+ }
+
+ min_period_size = snd_pcm_plug_client_size(substream,
+ snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
+ min_period_size *= oss_frame_size;
+ min_period_size = 1 << (ld2(min_period_size - 1) + 1);
+ if (oss_period_size < min_period_size)
+ oss_period_size = min_period_size;
+
+ max_period_size = snd_pcm_plug_client_size(substream,
+ snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
+ max_period_size *= oss_frame_size;
+ max_period_size = 1 << ld2(max_period_size);
+ if (oss_period_size > max_period_size)
+ oss_period_size = max_period_size;
+
+ oss_periods = oss_buffer_size / oss_period_size;
+
+ if (substream->oss.setup.periods > 1)
+ oss_periods = substream->oss.setup.periods;
+
+ s = snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIODS, NULL);
+ if (runtime->oss.maxfrags && s > runtime->oss.maxfrags)
+ s = runtime->oss.maxfrags;
+ if (oss_periods > s)
+ oss_periods = s;
+
+ s = snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIODS, NULL);
+ if (s < 2)
+ s = 2;
+ if (oss_periods < s)
+ oss_periods = s;
+
+ while (oss_period_size * oss_periods > oss_buffer_size)
+ oss_period_size /= 2;
+
+ if (oss_period_size < 16)
+ return -EINVAL;
+ runtime->oss.period_bytes = oss_period_size;
+ runtime->oss.period_frames = 1;
+ runtime->oss.periods = oss_periods;
+ return 0;
+}
+
+static int choose_rate(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, unsigned int best_rate)
+{
+ struct snd_interval *it;
+ struct snd_pcm_hw_params *save;
+ unsigned int rate, prev;
+
+ save = kmalloc(sizeof(*save), GFP_KERNEL);
+ if (save == NULL)
+ return -ENOMEM;
+ *save = *params;
+ it = hw_param_interval(save, SNDRV_PCM_HW_PARAM_RATE);
+
+ /* try multiples of the best rate */
+ rate = best_rate;
+ for (;;) {
+ if (it->max < rate || (it->max == rate && it->openmax))
+ break;
+ if (it->min < rate || (it->min == rate && !it->openmin)) {
+ int ret;
+ ret = snd_pcm_hw_param_set(substream, params,
+ SNDRV_PCM_HW_PARAM_RATE,
+ rate, 0);
+ if (ret == (int)rate) {
+ kfree(save);
+ return rate;
+ }
+ *params = *save;
+ }
+ prev = rate;
+ rate += best_rate;
+ if (rate <= prev)
+ break;
+ }
+
+ /* not found, use the nearest rate */
+ kfree(save);
+ return snd_pcm_hw_param_near(substream, params, SNDRV_PCM_HW_PARAM_RATE, best_rate, NULL);
+}
+
+static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_hw_params *params, *sparams;
+ struct snd_pcm_sw_params *sw_params;
+ ssize_t oss_buffer_size, oss_period_size;
+ size_t oss_frame_size;
+ int err;
+ int direct;
+ int format, sformat, n;
+ struct snd_mask sformat_mask;
+ struct snd_mask mask;
+
+ if (mutex_lock_interruptible(&runtime->oss.params_lock))
+ return -EINTR;
+ sw_params = kmalloc(sizeof(*sw_params), GFP_KERNEL);
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ sparams = kmalloc(sizeof(*sparams), GFP_KERNEL);
+ if (!sw_params || !params || !sparams) {
+ snd_printd("No memory\n");
+ err = -ENOMEM;
+ goto failure;
+ }
+
+ if (atomic_read(&substream->mmap_count))
+ direct = 1;
+ else
+ direct = substream->oss.setup.direct;
+
+ _snd_pcm_hw_params_any(sparams);
+ _snd_pcm_hw_param_setinteger(sparams, SNDRV_PCM_HW_PARAM_PERIODS);
+ _snd_pcm_hw_param_min(sparams, SNDRV_PCM_HW_PARAM_PERIODS, 2, 0);
+ snd_mask_none(&mask);
+ if (atomic_read(&substream->mmap_count))
+ snd_mask_set(&mask, SNDRV_PCM_ACCESS_MMAP_INTERLEAVED);
+ else {
+ snd_mask_set(&mask, SNDRV_PCM_ACCESS_RW_INTERLEAVED);
+ if (!direct)
+ snd_mask_set(&mask, SNDRV_PCM_ACCESS_RW_NONINTERLEAVED);
+ }
+ err = snd_pcm_hw_param_mask(substream, sparams, SNDRV_PCM_HW_PARAM_ACCESS, &mask);
+ if (err < 0) {
+ snd_printd("No usable accesses\n");
+ err = -EINVAL;
+ goto failure;
+ }
+ choose_rate(substream, sparams, runtime->oss.rate);
+ snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_CHANNELS, runtime->oss.channels, NULL);
+
+ format = snd_pcm_oss_format_from(runtime->oss.format);
+
+ sformat_mask = *hw_param_mask(sparams, SNDRV_PCM_HW_PARAM_FORMAT);
+ if (direct)
+ sformat = format;
+ else
+ sformat = snd_pcm_plug_slave_format(format, &sformat_mask);
+
+ if (sformat < 0 || !snd_mask_test(&sformat_mask, sformat)) {
+ for (sformat = 0; sformat <= SNDRV_PCM_FORMAT_LAST; sformat++) {
+ if (snd_mask_test(&sformat_mask, sformat) &&
+ snd_pcm_oss_format_to(sformat) >= 0)
+ break;
+ }
+ if (sformat > SNDRV_PCM_FORMAT_LAST) {
+ snd_printd("Cannot find a format!!!\n");
+ err = -EINVAL;
+ goto failure;
+ }
+ }
+ err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, sformat, 0);
+ if (err < 0)
+ goto failure;
+
+ if (direct) {
+ memcpy(params, sparams, sizeof(*params));
+ } else {
+ _snd_pcm_hw_params_any(params);
+ _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_ACCESS,
+ SNDRV_PCM_ACCESS_RW_INTERLEAVED, 0);
+ _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_FORMAT,
+ snd_pcm_oss_format_from(runtime->oss.format), 0);
+ _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_CHANNELS,
+ runtime->oss.channels, 0);
+ _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_RATE,
+ runtime->oss.rate, 0);
+ pdprintf("client: access = %i, format = %i, channels = %i, rate = %i\n",
+ params_access(params), params_format(params),
+ params_channels(params), params_rate(params));
+ }
+ pdprintf("slave: access = %i, format = %i, channels = %i, rate = %i\n",
+ params_access(sparams), params_format(sparams),
+ params_channels(sparams), params_rate(sparams));
+
+ oss_frame_size = snd_pcm_format_physical_width(params_format(params)) *
+ params_channels(params) / 8;
+
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+ snd_pcm_oss_plugin_clear(substream);
+ if (!direct) {
+ /* add necessary plugins */
+ snd_pcm_oss_plugin_clear(substream);
+ if ((err = snd_pcm_plug_format_plugins(substream,
+ params,
+ sparams)) < 0) {
+ snd_printd("snd_pcm_plug_format_plugins failed: %i\n", err);
+ snd_pcm_oss_plugin_clear(substream);
+ goto failure;
+ }
+ if (runtime->oss.plugin_first) {
+ struct snd_pcm_plugin *plugin;
+ if ((err = snd_pcm_plugin_build_io(substream, sparams, &plugin)) < 0) {
+ snd_printd("snd_pcm_plugin_build_io failed: %i\n", err);
+ snd_pcm_oss_plugin_clear(substream);
+ goto failure;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ err = snd_pcm_plugin_append(plugin);
+ } else {
+ err = snd_pcm_plugin_insert(plugin);
+ }
+ if (err < 0) {
+ snd_pcm_oss_plugin_clear(substream);
+ goto failure;
+ }
+ }
+ }
+#endif
+
+ err = snd_pcm_oss_period_size(substream, params, sparams);
+ if (err < 0)
+ goto failure;
+
+ n = snd_pcm_plug_slave_size(substream, runtime->oss.period_bytes / oss_frame_size);
+ err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, n, NULL);
+ if (err < 0)
+ goto failure;
+
+ err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIODS,
+ runtime->oss.periods, NULL);
+ if (err < 0)
+ goto failure;
+
+ snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
+
+ if ((err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, sparams)) < 0) {
+ snd_printd("HW_PARAMS failed: %i\n", err);
+ goto failure;
+ }
+
+ memset(sw_params, 0, sizeof(*sw_params));
+ if (runtime->oss.trigger) {
+ sw_params->start_threshold = 1;
+ } else {
+ sw_params->start_threshold = runtime->boundary;
+ }
+ if (atomic_read(&substream->mmap_count) ||
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ sw_params->stop_threshold = runtime->boundary;
+ else
+ sw_params->stop_threshold = runtime->buffer_size;
+ sw_params->tstamp_mode = SNDRV_PCM_TSTAMP_NONE;
+ sw_params->period_step = 1;
+ sw_params->avail_min = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ 1 : runtime->period_size;
+ if (atomic_read(&substream->mmap_count) ||
+ substream->oss.setup.nosilence) {
+ sw_params->silence_threshold = 0;
+ sw_params->silence_size = 0;
+ } else {
+ snd_pcm_uframes_t frames;
+ frames = runtime->period_size + 16;
+ if (frames > runtime->buffer_size)
+ frames = runtime->buffer_size;
+ sw_params->silence_threshold = frames;
+ sw_params->silence_size = frames;
+ }
+
+ if ((err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_SW_PARAMS, sw_params)) < 0) {
+ snd_printd("SW_PARAMS failed: %i\n", err);
+ goto failure;
+ }
+
+ runtime->oss.periods = params_periods(sparams);
+ oss_period_size = snd_pcm_plug_client_size(substream, params_period_size(sparams));
+ if (oss_period_size < 0) {
+ err = -EINVAL;
+ goto failure;
+ }
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+ if (runtime->oss.plugin_first) {
+ err = snd_pcm_plug_alloc(substream, oss_period_size);
+ if (err < 0)
+ goto failure;
+ }
+#endif
+ oss_period_size *= oss_frame_size;
+
+ oss_buffer_size = oss_period_size * runtime->oss.periods;
+ if (oss_buffer_size < 0) {
+ err = -EINVAL;
+ goto failure;
+ }
+
+ runtime->oss.period_bytes = oss_period_size;
+ runtime->oss.buffer_bytes = oss_buffer_size;
+
+ pdprintf("oss: period bytes = %i, buffer bytes = %i\n",
+ runtime->oss.period_bytes,
+ runtime->oss.buffer_bytes);
+ pdprintf("slave: period_size = %i, buffer_size = %i\n",
+ params_period_size(sparams),
+ params_buffer_size(sparams));
+
+ runtime->oss.format = snd_pcm_oss_format_to(params_format(params));
+ runtime->oss.channels = params_channels(params);
+ runtime->oss.rate = params_rate(params);
+
+ runtime->oss.params = 0;
+ runtime->oss.prepare = 1;
+ vfree(runtime->oss.buffer);
+ runtime->oss.buffer = vmalloc(runtime->oss.period_bytes);
+ runtime->oss.buffer_used = 0;
+ if (runtime->dma_area)
+ snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes));
+
+ runtime->oss.period_frames = snd_pcm_alsa_frames(substream, oss_period_size);
+
+ err = 0;
+failure:
+ kfree(sw_params);
+ kfree(params);
+ kfree(sparams);
+ mutex_unlock(&runtime->oss.params_lock);
+ return err;
+}
+
+static int snd_pcm_oss_get_active_substream(struct snd_pcm_oss_file *pcm_oss_file, struct snd_pcm_substream **r_substream)
+{
+ int idx, err;
+ struct snd_pcm_substream *asubstream = NULL, *substream;
+
+ for (idx = 0; idx < 2; idx++) {
+ substream = pcm_oss_file->streams[idx];
+ if (substream == NULL)
+ continue;
+ if (asubstream == NULL)
+ asubstream = substream;
+ if (substream->runtime->oss.params) {
+ err = snd_pcm_oss_change_params(substream);
+ if (err < 0)
+ return err;
+ }
+ }
+ if (!asubstream)
+ return -EIO;
+ if (r_substream)
+ *r_substream = asubstream;
+ return 0;
+}
+
+static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream)
+{
+ int err;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_PREPARE, NULL);
+ if (err < 0) {
+ snd_printd("snd_pcm_oss_prepare: SNDRV_PCM_IOCTL_PREPARE failed\n");
+ return err;
+ }
+ runtime->oss.prepare = 0;
+ runtime->oss.prev_hw_ptr_interrupt = 0;
+ runtime->oss.period_ptr = 0;
+ runtime->oss.buffer_used = 0;
+
+ return 0;
+}
+
+static int snd_pcm_oss_make_ready(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+ int err;
+
+ if (substream == NULL)
+ return 0;
+ runtime = substream->runtime;
+ if (runtime->oss.params) {
+ err = snd_pcm_oss_change_params(substream);
+ if (err < 0)
+ return err;
+ }
+ if (runtime->oss.prepare) {
+ err = snd_pcm_oss_prepare(substream);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int snd_pcm_oss_capture_position_fixup(struct snd_pcm_substream *substream, snd_pcm_sframes_t *delay)
+{
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_uframes_t frames;
+ int err = 0;
+
+ while (1) {
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DELAY, delay);
+ if (err < 0)
+ break;
+ runtime = substream->runtime;
+ if (*delay <= (snd_pcm_sframes_t)runtime->buffer_size)
+ break;
+ /* in case of overrun, skip whole periods like OSS/Linux driver does */
+ /* until avail(delay) <= buffer_size */
+ frames = (*delay - runtime->buffer_size) + runtime->period_size - 1;
+ frames /= runtime->period_size;
+ frames *= runtime->period_size;
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_FORWARD, &frames);
+ if (err < 0)
+ break;
+ }
+ return err;
+}
+
+snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const char *ptr, snd_pcm_uframes_t frames, int in_kernel)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+ while (1) {
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN ||
+ runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
+#ifdef OSS_DEBUG
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
+ printk("pcm_oss: write: recovering from XRUN\n");
+ else
+ printk("pcm_oss: write: recovering from SUSPEND\n");
+#endif
+ ret = snd_pcm_oss_prepare(substream);
+ if (ret < 0)
+ break;
+ }
+ if (in_kernel) {
+ mm_segment_t fs;
+ fs = snd_enter_user();
+ ret = snd_pcm_lib_write(substream, (void __user *)ptr, frames);
+ snd_leave_user(fs);
+ } else {
+ ret = snd_pcm_lib_write(substream, (void __user *)ptr, frames);
+ }
+ if (ret != -EPIPE && ret != -ESTRPIPE)
+ break;
+ /* test, if we can't store new data, because the stream */
+ /* has not been started */
+ if (runtime->status->state == SNDRV_PCM_STATE_PREPARED)
+ return -EAGAIN;
+ }
+ return ret;
+}
+
+snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *ptr, snd_pcm_uframes_t frames, int in_kernel)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t delay;
+ int ret;
+ while (1) {
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN ||
+ runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
+#ifdef OSS_DEBUG
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
+ printk("pcm_oss: read: recovering from XRUN\n");
+ else
+ printk("pcm_oss: read: recovering from SUSPEND\n");
+#endif
+ ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL);
+ if (ret < 0)
+ break;
+ } else if (runtime->status->state == SNDRV_PCM_STATE_SETUP) {
+ ret = snd_pcm_oss_prepare(substream);
+ if (ret < 0)
+ break;
+ }
+ ret = snd_pcm_oss_capture_position_fixup(substream, &delay);
+ if (ret < 0)
+ break;
+ if (in_kernel) {
+ mm_segment_t fs;
+ fs = snd_enter_user();
+ ret = snd_pcm_lib_read(substream, (void __user *)ptr, frames);
+ snd_leave_user(fs);
+ } else {
+ ret = snd_pcm_lib_read(substream, (void __user *)ptr, frames);
+ }
+ if (ret == -EPIPE) {
+ if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
+ ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
+ if (ret < 0)
+ break;
+ }
+ continue;
+ }
+ if (ret != -ESTRPIPE)
+ break;
+ }
+ return ret;
+}
+
+snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames, int in_kernel)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+ while (1) {
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN ||
+ runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
+#ifdef OSS_DEBUG
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
+ printk("pcm_oss: writev: recovering from XRUN\n");
+ else
+ printk("pcm_oss: writev: recovering from SUSPEND\n");
+#endif
+ ret = snd_pcm_oss_prepare(substream);
+ if (ret < 0)
+ break;
+ }
+ if (in_kernel) {
+ mm_segment_t fs;
+ fs = snd_enter_user();
+ ret = snd_pcm_lib_writev(substream, (void __user **)bufs, frames);
+ snd_leave_user(fs);
+ } else {
+ ret = snd_pcm_lib_writev(substream, (void __user **)bufs, frames);
+ }
+ if (ret != -EPIPE && ret != -ESTRPIPE)
+ break;
+
+ /* test, if we can't store new data, because the stream */
+ /* has not been started */
+ if (runtime->status->state == SNDRV_PCM_STATE_PREPARED)
+ return -EAGAIN;
+ }
+ return ret;
+}
+
+snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames, int in_kernel)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+ while (1) {
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN ||
+ runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
+#ifdef OSS_DEBUG
+ if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
+ printk("pcm_oss: readv: recovering from XRUN\n");
+ else
+ printk("pcm_oss: readv: recovering from SUSPEND\n");
+#endif
+ ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL);
+ if (ret < 0)
+ break;
+ } else if (runtime->status->state == SNDRV_PCM_STATE_SETUP) {
+ ret = snd_pcm_oss_prepare(substream);
+ if (ret < 0)
+ break;
+ }
+ if (in_kernel) {
+ mm_segment_t fs;
+ fs = snd_enter_user();
+ ret = snd_pcm_lib_readv(substream, (void __user **)bufs, frames);
+ snd_leave_user(fs);
+ } else {
+ ret = snd_pcm_lib_readv(substream, (void __user **)bufs, frames);
+ }
+ if (ret != -EPIPE && ret != -ESTRPIPE)
+ break;
+ }
+ return ret;
+}
+
+static ssize_t snd_pcm_oss_write2(struct snd_pcm_substream *substream, const char *buf, size_t bytes, int in_kernel)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t frames, frames1;
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+ if (runtime->oss.plugin_first) {
+ struct snd_pcm_plugin_channel *channels;
+ size_t oss_frame_bytes = (runtime->oss.plugin_first->src_width * runtime->oss.plugin_first->src_format.channels) / 8;
+ if (!in_kernel) {
+ if (copy_from_user(runtime->oss.buffer, (const char __user *)buf, bytes))
+ return -EFAULT;
+ buf = runtime->oss.buffer;
+ }
+ frames = bytes / oss_frame_bytes;
+ frames1 = snd_pcm_plug_client_channels_buf(substream, (char *)buf, frames, &channels);
+ if (frames1 < 0)
+ return frames1;
+ frames1 = snd_pcm_plug_write_transfer(substream, channels, frames1);
+ if (frames1 <= 0)
+ return frames1;
+ bytes = frames1 * oss_frame_bytes;
+ } else
+#endif
+ {
+ frames = bytes_to_frames(runtime, bytes);
+ frames1 = snd_pcm_oss_write3(substream, buf, frames, in_kernel);
+ if (frames1 <= 0)
+ return frames1;
+ bytes = frames_to_bytes(runtime, frames1);
+ }
+ return bytes;
+}
+
+static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const char __user *buf, size_t bytes)
+{
+ size_t xfer = 0;
+ ssize_t tmp;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (atomic_read(&substream->mmap_count))
+ return -ENXIO;
+
+ if ((tmp = snd_pcm_oss_make_ready(substream)) < 0)
+ return tmp;
+ mutex_lock(&runtime->oss.params_lock);
+ while (bytes > 0) {
+ if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) {
+ tmp = bytes;
+ if (tmp + runtime->oss.buffer_used > runtime->oss.period_bytes)
+ tmp = runtime->oss.period_bytes - runtime->oss.buffer_used;
+ if (tmp > 0) {
+ if (copy_from_user(runtime->oss.buffer + runtime->oss.buffer_used, buf, tmp)) {
+ tmp = -EFAULT;
+ goto err;
+ }
+ }
+ runtime->oss.buffer_used += tmp;
+ buf += tmp;
+ bytes -= tmp;
+ xfer += tmp;
+ if (substream->oss.setup.partialfrag ||
+ runtime->oss.buffer_used == runtime->oss.period_bytes) {
+ tmp = snd_pcm_oss_write2(substream, runtime->oss.buffer + runtime->oss.period_ptr,
+ runtime->oss.buffer_used - runtime->oss.period_ptr, 1);
+ if (tmp <= 0)
+ goto err;
+ runtime->oss.bytes += tmp;
+ runtime->oss.period_ptr += tmp;
+ runtime->oss.period_ptr %= runtime->oss.period_bytes;
+ if (runtime->oss.period_ptr == 0 ||
+ runtime->oss.period_ptr == runtime->oss.buffer_used)
+ runtime->oss.buffer_used = 0;
+ else if ((substream->f_flags & O_NONBLOCK) != 0) {
+ tmp = -EAGAIN;
+ goto err;
+ }
+ }
+ } else {
+ tmp = snd_pcm_oss_write2(substream,
+ (const char __force *)buf,
+ runtime->oss.period_bytes, 0);
+ if (tmp <= 0)
+ goto err;
+ runtime->oss.bytes += tmp;
+ buf += tmp;
+ bytes -= tmp;
+ xfer += tmp;
+ if ((substream->f_flags & O_NONBLOCK) != 0 &&
+ tmp != runtime->oss.period_bytes)
+ break;
+ }
+ }
+ mutex_unlock(&runtime->oss.params_lock);
+ return xfer;
+
+ err:
+ mutex_unlock(&runtime->oss.params_lock);
+ return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp;
+}
+
+static ssize_t snd_pcm_oss_read2(struct snd_pcm_substream *substream, char *buf, size_t bytes, int in_kernel)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t frames, frames1;
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+ char __user *final_dst = (char __user *)buf;
+ if (runtime->oss.plugin_first) {
+ struct snd_pcm_plugin_channel *channels;
+ size_t oss_frame_bytes = (runtime->oss.plugin_last->dst_width * runtime->oss.plugin_last->dst_format.channels) / 8;
+ if (!in_kernel)
+ buf = runtime->oss.buffer;
+ frames = bytes / oss_frame_bytes;
+ frames1 = snd_pcm_plug_client_channels_buf(substream, buf, frames, &channels);
+ if (frames1 < 0)
+ return frames1;
+ frames1 = snd_pcm_plug_read_transfer(substream, channels, frames1);
+ if (frames1 <= 0)
+ return frames1;
+ bytes = frames1 * oss_frame_bytes;
+ if (!in_kernel && copy_to_user(final_dst, buf, bytes))
+ return -EFAULT;
+ } else
+#endif
+ {
+ frames = bytes_to_frames(runtime, bytes);
+ frames1 = snd_pcm_oss_read3(substream, buf, frames, in_kernel);
+ if (frames1 <= 0)
+ return frames1;
+ bytes = frames_to_bytes(runtime, frames1);
+ }
+ return bytes;
+}
+
+static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __user *buf, size_t bytes)
+{
+ size_t xfer = 0;
+ ssize_t tmp;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (atomic_read(&substream->mmap_count))
+ return -ENXIO;
+
+ if ((tmp = snd_pcm_oss_make_ready(substream)) < 0)
+ return tmp;
+ mutex_lock(&runtime->oss.params_lock);
+ while (bytes > 0) {
+ if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) {
+ if (runtime->oss.buffer_used == 0) {
+ tmp = snd_pcm_oss_read2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1);
+ if (tmp <= 0)
+ goto err;
+ runtime->oss.bytes += tmp;
+ runtime->oss.period_ptr = tmp;
+ runtime->oss.buffer_used = tmp;
+ }
+ tmp = bytes;
+ if ((size_t) tmp > runtime->oss.buffer_used)
+ tmp = runtime->oss.buffer_used;
+ if (copy_to_user(buf, runtime->oss.buffer + (runtime->oss.period_ptr - runtime->oss.buffer_used), tmp)) {
+ tmp = -EFAULT;
+ goto err;
+ }
+ buf += tmp;
+ bytes -= tmp;
+ xfer += tmp;
+ runtime->oss.buffer_used -= tmp;
+ } else {
+ tmp = snd_pcm_oss_read2(substream, (char __force *)buf,
+ runtime->oss.period_bytes, 0);
+ if (tmp <= 0)
+ goto err;
+ runtime->oss.bytes += tmp;
+ buf += tmp;
+ bytes -= tmp;
+ xfer += tmp;
+ }
+ }
+ mutex_unlock(&runtime->oss.params_lock);
+ return xfer;
+
+ err:
+ mutex_unlock(&runtime->oss.params_lock);
+ return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp;
+}
+
+static int snd_pcm_oss_reset(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream != NULL) {
+ snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
+ substream->runtime->oss.prepare = 1;
+ }
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+ if (substream != NULL) {
+ snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
+ substream->runtime->oss.prepare = 1;
+ }
+ return 0;
+}
+
+static int snd_pcm_oss_post(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+ int err;
+
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream != NULL) {
+ if ((err = snd_pcm_oss_make_ready(substream)) < 0)
+ return err;
+ snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_START, NULL);
+ }
+ /* note: all errors from the start action are ignored */
+ /* OSS apps do not know, how to handle them */
+ return 0;
+}
+
+static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size)
+{
+ struct snd_pcm_runtime *runtime;
+ ssize_t result = 0;
+ long res;
+ wait_queue_t wait;
+
+ runtime = substream->runtime;
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&runtime->sleep, &wait);
+#ifdef OSS_DEBUG
+ printk("sync1: size = %li\n", size);
+#endif
+ while (1) {
+ result = snd_pcm_oss_write2(substream, runtime->oss.buffer, size, 1);
+ if (result > 0) {
+ runtime->oss.buffer_used = 0;
+ result = 0;
+ break;
+ }
+ if (result != 0 && result != -EAGAIN)
+ break;
+ result = 0;
+ set_current_state(TASK_INTERRUPTIBLE);
+ snd_pcm_stream_lock_irq(substream);
+ res = runtime->status->state;
+ snd_pcm_stream_unlock_irq(substream);
+ if (res != SNDRV_PCM_STATE_RUNNING) {
+ set_current_state(TASK_RUNNING);
+ break;
+ }
+ res = schedule_timeout(10 * HZ);
+ if (signal_pending(current)) {
+ result = -ERESTARTSYS;
+ break;
+ }
+ if (res == 0) {
+ snd_printk(KERN_ERR "OSS sync error - DMA timeout\n");
+ result = -EIO;
+ break;
+ }
+ }
+ remove_wait_queue(&runtime->sleep, &wait);
+ return result;
+}
+
+static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ int err = 0;
+ unsigned int saved_f_flags;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_format_t format;
+ unsigned long width;
+ size_t size;
+
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream != NULL) {
+ runtime = substream->runtime;
+ if (atomic_read(&substream->mmap_count))
+ goto __direct;
+ if ((err = snd_pcm_oss_make_ready(substream)) < 0)
+ return err;
+ format = snd_pcm_oss_format_from(runtime->oss.format);
+ width = snd_pcm_format_physical_width(format);
+ mutex_lock(&runtime->oss.params_lock);
+ if (runtime->oss.buffer_used > 0) {
+#ifdef OSS_DEBUG
+ printk("sync: buffer_used\n");
+#endif
+ size = (8 * (runtime->oss.period_bytes - runtime->oss.buffer_used) + 7) / width;
+ snd_pcm_format_set_silence(format,
+ runtime->oss.buffer + runtime->oss.buffer_used,
+ size);
+ err = snd_pcm_oss_sync1(substream, runtime->oss.period_bytes);
+ if (err < 0) {
+ mutex_unlock(&runtime->oss.params_lock);
+ return err;
+ }
+ } else if (runtime->oss.period_ptr > 0) {
+#ifdef OSS_DEBUG
+ printk("sync: period_ptr\n");
+#endif
+ size = runtime->oss.period_bytes - runtime->oss.period_ptr;
+ snd_pcm_format_set_silence(format,
+ runtime->oss.buffer,
+ size * 8 / width);
+ err = snd_pcm_oss_sync1(substream, size);
+ if (err < 0) {
+ mutex_unlock(&runtime->oss.params_lock);
+ return err;
+ }
+ }
+ /*
+ * The ALSA's period might be a bit large than OSS one.
+ * Fill the remain portion of ALSA period with zeros.
+ */
+ size = runtime->control->appl_ptr % runtime->period_size;
+ if (size > 0) {
+ size = runtime->period_size - size;
+ if (runtime->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED) {
+ size = (runtime->frame_bits * size) / 8;
+ while (size > 0) {
+ mm_segment_t fs;
+ size_t size1 = size < runtime->oss.period_bytes ? size : runtime->oss.period_bytes;
+ size -= size1;
+ size1 *= 8;
+ size1 /= runtime->sample_bits;
+ snd_pcm_format_set_silence(runtime->format,
+ runtime->oss.buffer,
+ size1);
+ size1 /= runtime->channels; /* frames */
+ fs = snd_enter_user();
+ snd_pcm_lib_write(substream, (void __user *)runtime->oss.buffer, size1);
+ snd_leave_user(fs);
+ }
+ } else if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED) {
+ void __user *buffers[runtime->channels];
+ memset(buffers, 0, runtime->channels * sizeof(void *));
+ snd_pcm_lib_writev(substream, buffers, size);
+ }
+ }
+ mutex_unlock(&runtime->oss.params_lock);
+ /*
+ * finish sync: drain the buffer
+ */
+ __direct:
+ saved_f_flags = substream->f_flags;
+ substream->f_flags &= ~O_NONBLOCK;
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL);
+ substream->f_flags = saved_f_flags;
+ if (err < 0)
+ return err;
+ runtime->oss.prepare = 1;
+ }
+
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+ if (substream != NULL) {
+ if ((err = snd_pcm_oss_make_ready(substream)) < 0)
+ return err;
+ runtime = substream->runtime;
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
+ if (err < 0)
+ return err;
+ runtime->oss.buffer_used = 0;
+ runtime->oss.prepare = 1;
+ }
+ return 0;
+}
+
+static int snd_pcm_oss_set_rate(struct snd_pcm_oss_file *pcm_oss_file, int rate)
+{
+ int idx;
+
+ for (idx = 1; idx >= 0; --idx) {
+ struct snd_pcm_substream *substream = pcm_oss_file->streams[idx];
+ struct snd_pcm_runtime *runtime;
+ if (substream == NULL)
+ continue;
+ runtime = substream->runtime;
+ if (rate < 1000)
+ rate = 1000;
+ else if (rate > 192000)
+ rate = 192000;
+ if (runtime->oss.rate != rate) {
+ runtime->oss.params = 1;
+ runtime->oss.rate = rate;
+ }
+ }
+ return snd_pcm_oss_get_rate(pcm_oss_file);
+}
+
+static int snd_pcm_oss_get_rate(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+ int err;
+
+ if ((err = snd_pcm_oss_get_active_substream(pcm_oss_file, &substream)) < 0)
+ return err;
+ return substream->runtime->oss.rate;
+}
+
+static int snd_pcm_oss_set_channels(struct snd_pcm_oss_file *pcm_oss_file, unsigned int channels)
+{
+ int idx;
+ if (channels < 1)
+ channels = 1;
+ if (channels > 128)
+ return -EINVAL;
+ for (idx = 1; idx >= 0; --idx) {
+ struct snd_pcm_substream *substream = pcm_oss_file->streams[idx];
+ struct snd_pcm_runtime *runtime;
+ if (substream == NULL)
+ continue;
+ runtime = substream->runtime;
+ if (runtime->oss.channels != channels) {
+ runtime->oss.params = 1;
+ runtime->oss.channels = channels;
+ }
+ }
+ return snd_pcm_oss_get_channels(pcm_oss_file);
+}
+
+static int snd_pcm_oss_get_channels(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+ int err;
+
+ if ((err = snd_pcm_oss_get_active_substream(pcm_oss_file, &substream)) < 0)
+ return err;
+ return substream->runtime->oss.channels;
+}
+
+static int snd_pcm_oss_get_block_size(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+ int err;
+
+ if ((err = snd_pcm_oss_get_active_substream(pcm_oss_file, &substream)) < 0)
+ return err;
+ return substream->runtime->oss.period_bytes;
+}
+
+static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+ int err;
+ int direct;
+ struct snd_pcm_hw_params *params;
+ unsigned int formats = 0;
+ struct snd_mask format_mask;
+ int fmt;
+
+ if ((err = snd_pcm_oss_get_active_substream(pcm_oss_file, &substream)) < 0)
+ return err;
+ if (atomic_read(&substream->mmap_count))
+ direct = 1;
+ else
+ direct = substream->oss.setup.direct;
+ if (!direct)
+ return AFMT_MU_LAW | AFMT_U8 |
+ AFMT_S16_LE | AFMT_S16_BE |
+ AFMT_S8 | AFMT_U16_LE |
+ AFMT_U16_BE |
+ AFMT_S32_LE | AFMT_S32_BE |
+ AFMT_S24_LE | AFMT_S24_LE |
+ AFMT_S24_PACKED;
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ if (!params)
+ return -ENOMEM;
+ _snd_pcm_hw_params_any(params);
+ err = snd_pcm_hw_refine(substream, params);
+ format_mask = *hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ kfree(params);
+ if (err < 0)
+ return err;
+ for (fmt = 0; fmt < 32; ++fmt) {
+ if (snd_mask_test(&format_mask, fmt)) {
+ int f = snd_pcm_oss_format_to(fmt);
+ if (f >= 0)
+ formats |= f;
+ }
+ }
+ return formats;
+}
+
+static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int format)
+{
+ int formats, idx;
+
+ if (format != AFMT_QUERY) {
+ formats = snd_pcm_oss_get_formats(pcm_oss_file);
+ if (formats < 0)
+ return formats;
+ if (!(formats & format))
+ format = AFMT_U8;
+ for (idx = 1; idx >= 0; --idx) {
+ struct snd_pcm_substream *substream = pcm_oss_file->streams[idx];
+ struct snd_pcm_runtime *runtime;
+ if (substream == NULL)
+ continue;
+ runtime = substream->runtime;
+ if (runtime->oss.format != format) {
+ runtime->oss.params = 1;
+ runtime->oss.format = format;
+ }
+ }
+ }
+ return snd_pcm_oss_get_format(pcm_oss_file);
+}
+
+static int snd_pcm_oss_get_format(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+ int err;
+
+ if ((err = snd_pcm_oss_get_active_substream(pcm_oss_file, &substream)) < 0)
+ return err;
+ return substream->runtime->oss.format;
+}
+
+static int snd_pcm_oss_set_subdivide1(struct snd_pcm_substream *substream, int subdivide)
+{
+ struct snd_pcm_runtime *runtime;
+
+ if (substream == NULL)
+ return 0;
+ runtime = substream->runtime;
+ if (subdivide == 0) {
+ subdivide = runtime->oss.subdivision;
+ if (subdivide == 0)
+ subdivide = 1;
+ return subdivide;
+ }
+ if (runtime->oss.subdivision || runtime->oss.fragshift)
+ return -EINVAL;
+ if (subdivide != 1 && subdivide != 2 && subdivide != 4 &&
+ subdivide != 8 && subdivide != 16)
+ return -EINVAL;
+ runtime->oss.subdivision = subdivide;
+ runtime->oss.params = 1;
+ return subdivide;
+}
+
+static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int subdivide)
+{
+ int err = -EINVAL, idx;
+
+ for (idx = 1; idx >= 0; --idx) {
+ struct snd_pcm_substream *substream = pcm_oss_file->streams[idx];
+ if (substream == NULL)
+ continue;
+ if ((err = snd_pcm_oss_set_subdivide1(substream, subdivide)) < 0)
+ return err;
+ }
+ return err;
+}
+
+static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsigned int val)
+{
+ struct snd_pcm_runtime *runtime;
+
+ if (substream == NULL)
+ return 0;
+ runtime = substream->runtime;
+ if (runtime->oss.subdivision || runtime->oss.fragshift)
+ return -EINVAL;
+ runtime->oss.fragshift = val & 0xffff;
+ runtime->oss.maxfrags = (val >> 16) & 0xffff;
+ if (runtime->oss.fragshift < 4) /* < 16 */
+ runtime->oss.fragshift = 4;
+ if (runtime->oss.maxfrags < 2)
+ runtime->oss.maxfrags = 2;
+ runtime->oss.params = 1;
+ return 0;
+}
+
+static int snd_pcm_oss_set_fragment(struct snd_pcm_oss_file *pcm_oss_file, unsigned int val)
+{
+ int err = -EINVAL, idx;
+
+ for (idx = 1; idx >= 0; --idx) {
+ struct snd_pcm_substream *substream = pcm_oss_file->streams[idx];
+ if (substream == NULL)
+ continue;
+ if ((err = snd_pcm_oss_set_fragment1(substream, val)) < 0)
+ return err;
+ }
+ return err;
+}
+
+static int snd_pcm_oss_nonblock(struct file * file)
+{
+ file->f_flags |= O_NONBLOCK;
+ return 0;
+}
+
+static int snd_pcm_oss_get_caps1(struct snd_pcm_substream *substream, int res)
+{
+
+ if (substream == NULL) {
+ res &= ~DSP_CAP_DUPLEX;
+ return res;
+ }
+#ifdef DSP_CAP_MULTI
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->pstr->substream_count > 1)
+ res |= DSP_CAP_MULTI;
+#endif
+ /* DSP_CAP_REALTIME is set all times: */
+ /* all ALSA drivers can return actual pointer in ring buffer */
+#if defined(DSP_CAP_REALTIME) && 0
+ {
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->info & (SNDRV_PCM_INFO_BLOCK_TRANSFER|SNDRV_PCM_INFO_BATCH))
+ res &= ~DSP_CAP_REALTIME;
+ }
+#endif
+ return res;
+}
+
+static int snd_pcm_oss_get_caps(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ int result, idx;
+
+ result = DSP_CAP_TRIGGER | DSP_CAP_MMAP | DSP_CAP_DUPLEX | DSP_CAP_REALTIME;
+ for (idx = 0; idx < 2; idx++) {
+ struct snd_pcm_substream *substream = pcm_oss_file->streams[idx];
+ result = snd_pcm_oss_get_caps1(substream, result);
+ }
+ result |= 0x0001; /* revision - same as SB AWE 64 */
+ return result;
+}
+
+static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, snd_pcm_uframes_t hw_ptr)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t appl_ptr;
+ appl_ptr = hw_ptr + runtime->buffer_size;
+ appl_ptr %= runtime->boundary;
+ runtime->control->appl_ptr = appl_ptr;
+}
+
+static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int trigger)
+{
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm_substream *psubstream = NULL, *csubstream = NULL;
+ int err, cmd;
+
+#ifdef OSS_DEBUG
+ printk("pcm_oss: trigger = 0x%x\n", trigger);
+#endif
+
+ psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ csubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+
+ if (psubstream) {
+ if ((err = snd_pcm_oss_make_ready(psubstream)) < 0)
+ return err;
+ }
+ if (csubstream) {
+ if ((err = snd_pcm_oss_make_ready(csubstream)) < 0)
+ return err;
+ }
+ if (psubstream) {
+ runtime = psubstream->runtime;
+ if (trigger & PCM_ENABLE_OUTPUT) {
+ if (runtime->oss.trigger)
+ goto _skip1;
+ if (atomic_read(&psubstream->mmap_count))
+ snd_pcm_oss_simulate_fill(psubstream, runtime->hw_ptr_interrupt);
+ runtime->oss.trigger = 1;
+ runtime->start_threshold = 1;
+ cmd = SNDRV_PCM_IOCTL_START;
+ } else {
+ if (!runtime->oss.trigger)
+ goto _skip1;
+ runtime->oss.trigger = 0;
+ runtime->start_threshold = runtime->boundary;
+ cmd = SNDRV_PCM_IOCTL_DROP;
+ runtime->oss.prepare = 1;
+ }
+ err = snd_pcm_kernel_ioctl(psubstream, cmd, NULL);
+ if (err < 0)
+ return err;
+ }
+ _skip1:
+ if (csubstream) {
+ runtime = csubstream->runtime;
+ if (trigger & PCM_ENABLE_INPUT) {
+ if (runtime->oss.trigger)
+ goto _skip2;
+ runtime->oss.trigger = 1;
+ runtime->start_threshold = 1;
+ cmd = SNDRV_PCM_IOCTL_START;
+ } else {
+ if (!runtime->oss.trigger)
+ goto _skip2;
+ runtime->oss.trigger = 0;
+ runtime->start_threshold = runtime->boundary;
+ cmd = SNDRV_PCM_IOCTL_DROP;
+ runtime->oss.prepare = 1;
+ }
+ err = snd_pcm_kernel_ioctl(csubstream, cmd, NULL);
+ if (err < 0)
+ return err;
+ }
+ _skip2:
+ return 0;
+}
+
+static int snd_pcm_oss_get_trigger(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *psubstream = NULL, *csubstream = NULL;
+ int result = 0;
+
+ psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ csubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+ if (psubstream && psubstream->runtime && psubstream->runtime->oss.trigger)
+ result |= PCM_ENABLE_OUTPUT;
+ if (csubstream && csubstream->runtime && csubstream->runtime->oss.trigger)
+ result |= PCM_ENABLE_INPUT;
+ return result;
+}
+
+static int snd_pcm_oss_get_odelay(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_sframes_t delay;
+ int err;
+
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream == NULL)
+ return -EINVAL;
+ if ((err = snd_pcm_oss_make_ready(substream)) < 0)
+ return err;
+ runtime = substream->runtime;
+ if (runtime->oss.params || runtime->oss.prepare)
+ return 0;
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DELAY, &delay);
+ if (err == -EPIPE)
+ delay = 0; /* hack for broken OSS applications */
+ else if (err < 0)
+ return err;
+ return snd_pcm_oss_bytes(substream, delay);
+}
+
+static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream, struct count_info __user * _info)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_sframes_t delay;
+ int fixup;
+ struct count_info info;
+ int err;
+
+ if (_info == NULL)
+ return -EFAULT;
+ substream = pcm_oss_file->streams[stream];
+ if (substream == NULL)
+ return -EINVAL;
+ if ((err = snd_pcm_oss_make_ready(substream)) < 0)
+ return err;
+ runtime = substream->runtime;
+ if (runtime->oss.params || runtime->oss.prepare) {
+ memset(&info, 0, sizeof(info));
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ }
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DELAY, &delay);
+ if (err == -EPIPE || err == -ESTRPIPE || (! err && delay < 0)) {
+ err = 0;
+ delay = 0;
+ fixup = 0;
+ } else {
+ fixup = runtime->oss.buffer_used;
+ }
+ } else {
+ err = snd_pcm_oss_capture_position_fixup(substream, &delay);
+ fixup = -runtime->oss.buffer_used;
+ }
+ if (err < 0)
+ return err;
+ info.ptr = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr % runtime->buffer_size);
+ if (atomic_read(&substream->mmap_count)) {
+ snd_pcm_sframes_t n;
+ n = (delay = runtime->hw_ptr_interrupt) - runtime->oss.prev_hw_ptr_interrupt;
+ if (n < 0)
+ n += runtime->boundary;
+ info.blocks = n / runtime->period_size;
+ runtime->oss.prev_hw_ptr_interrupt = delay;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_pcm_oss_simulate_fill(substream, delay);
+ info.bytes = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr) & INT_MAX;
+ } else {
+ delay = snd_pcm_oss_bytes(substream, delay);
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (substream->oss.setup.buggyptr)
+ info.blocks = (runtime->oss.buffer_bytes - delay - fixup) / runtime->oss.period_bytes;
+ else
+ info.blocks = (delay + fixup) / runtime->oss.period_bytes;
+ info.bytes = (runtime->oss.bytes - delay) & INT_MAX;
+ } else {
+ delay += fixup;
+ info.blocks = delay / runtime->oss.period_bytes;
+ info.bytes = (runtime->oss.bytes + delay) & INT_MAX;
+ }
+ }
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_pcm_oss_get_space(struct snd_pcm_oss_file *pcm_oss_file, int stream, struct audio_buf_info __user *_info)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_sframes_t avail;
+ int fixup;
+ struct audio_buf_info info;
+ int err;
+
+ if (_info == NULL)
+ return -EFAULT;
+ substream = pcm_oss_file->streams[stream];
+ if (substream == NULL)
+ return -EINVAL;
+ runtime = substream->runtime;
+
+ if (runtime->oss.params &&
+ (err = snd_pcm_oss_change_params(substream)) < 0)
+ return err;
+
+ info.fragsize = runtime->oss.period_bytes;
+ info.fragstotal = runtime->periods;
+ if (runtime->oss.prepare) {
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ info.bytes = runtime->oss.period_bytes * runtime->oss.periods;
+ info.fragments = runtime->oss.periods;
+ } else {
+ info.bytes = 0;
+ info.fragments = 0;
+ }
+ } else {
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DELAY, &avail);
+ if (err == -EPIPE || err == -ESTRPIPE || (! err && avail < 0)) {
+ avail = runtime->buffer_size;
+ err = 0;
+ fixup = 0;
+ } else {
+ avail = runtime->buffer_size - avail;
+ fixup = -runtime->oss.buffer_used;
+ }
+ } else {
+ err = snd_pcm_oss_capture_position_fixup(substream, &avail);
+ fixup = runtime->oss.buffer_used;
+ }
+ if (err < 0)
+ return err;
+ info.bytes = snd_pcm_oss_bytes(substream, avail) + fixup;
+ info.fragments = info.bytes / runtime->oss.period_bytes;
+ }
+
+#ifdef OSS_DEBUG
+ printk("pcm_oss: space: bytes = %i, fragments = %i, fragstotal = %i, fragsize = %i\n", info.bytes, info.fragments, info.fragstotal, info.fragsize);
+#endif
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_pcm_oss_get_mapbuf(struct snd_pcm_oss_file *pcm_oss_file, int stream, struct buffmem_desc __user * _info)
+{
+ // it won't be probably implemented
+ // snd_printd("TODO: snd_pcm_oss_get_mapbuf\n");
+ return -EINVAL;
+}
+
+static const char *strip_task_path(const char *path)
+{
+ const char *ptr, *ptrl = NULL;
+ for (ptr = path; *ptr; ptr++) {
+ if (*ptr == '/')
+ ptrl = ptr + 1;
+ }
+ return ptrl;
+}
+
+static void snd_pcm_oss_look_for_setup(struct snd_pcm *pcm, int stream,
+ const char *task_name,
+ struct snd_pcm_oss_setup *rsetup)
+{
+ struct snd_pcm_oss_setup *setup;
+
+ mutex_lock(&pcm->streams[stream].oss.setup_mutex);
+ do {
+ for (setup = pcm->streams[stream].oss.setup_list; setup;
+ setup = setup->next) {
+ if (!strcmp(setup->task_name, task_name))
+ goto out;
+ }
+ } while ((task_name = strip_task_path(task_name)) != NULL);
+ out:
+ if (setup)
+ *rsetup = *setup;
+ mutex_unlock(&pcm->streams[stream].oss.setup_mutex);
+}
+
+static void snd_pcm_oss_release_substream(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+ runtime = substream->runtime;
+ vfree(runtime->oss.buffer);
+ runtime->oss.buffer = NULL;
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+ snd_pcm_oss_plugin_clear(substream);
+#endif
+ substream->oss.oss = 0;
+}
+
+static void snd_pcm_oss_init_substream(struct snd_pcm_substream *substream,
+ struct snd_pcm_oss_setup *setup,
+ int minor)
+{
+ struct snd_pcm_runtime *runtime;
+
+ substream->oss.oss = 1;
+ substream->oss.setup = *setup;
+ if (setup->nonblock)
+ substream->f_flags |= O_NONBLOCK;
+ else if (setup->block)
+ substream->f_flags &= ~O_NONBLOCK;
+ runtime = substream->runtime;
+ runtime->oss.params = 1;
+ runtime->oss.trigger = 1;
+ runtime->oss.rate = 8000;
+ mutex_init(&runtime->oss.params_lock);
+ switch (SNDRV_MINOR_OSS_DEVICE(minor)) {
+ case SNDRV_MINOR_OSS_PCM_8:
+ runtime->oss.format = AFMT_U8;
+ break;
+ case SNDRV_MINOR_OSS_PCM_16:
+ runtime->oss.format = AFMT_S16_LE;
+ break;
+ default:
+ runtime->oss.format = AFMT_MU_LAW;
+ }
+ runtime->oss.channels = 1;
+ runtime->oss.fragshift = 0;
+ runtime->oss.maxfrags = 0;
+ runtime->oss.subdivision = 0;
+ substream->pcm_release = snd_pcm_oss_release_substream;
+}
+
+static int snd_pcm_oss_release_file(struct snd_pcm_oss_file *pcm_oss_file)
+{
+ int cidx;
+ if (!pcm_oss_file)
+ return 0;
+ for (cidx = 0; cidx < 2; ++cidx) {
+ struct snd_pcm_substream *substream = pcm_oss_file->streams[cidx];
+ if (substream)
+ snd_pcm_release_substream(substream);
+ }
+ kfree(pcm_oss_file);
+ return 0;
+}
+
+static int snd_pcm_oss_open_file(struct file *file,
+ struct snd_pcm *pcm,
+ struct snd_pcm_oss_file **rpcm_oss_file,
+ int minor,
+ struct snd_pcm_oss_setup *setup)
+{
+ int idx, err;
+ struct snd_pcm_oss_file *pcm_oss_file;
+ struct snd_pcm_substream *substream;
+ fmode_t f_mode = file->f_mode;
+
+ if (rpcm_oss_file)
+ *rpcm_oss_file = NULL;
+
+ pcm_oss_file = kzalloc(sizeof(*pcm_oss_file), GFP_KERNEL);
+ if (pcm_oss_file == NULL)
+ return -ENOMEM;
+
+ if ((f_mode & (FMODE_WRITE|FMODE_READ)) == (FMODE_WRITE|FMODE_READ) &&
+ (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX))
+ f_mode = FMODE_WRITE;
+
+ file->f_flags &= ~O_APPEND;
+ for (idx = 0; idx < 2; idx++) {
+ if (setup[idx].disable)
+ continue;
+ if (! pcm->streams[idx].substream_count)
+ continue; /* no matching substream */
+ if (idx == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (! (f_mode & FMODE_WRITE))
+ continue;
+ } else {
+ if (! (f_mode & FMODE_READ))
+ continue;
+ }
+ err = snd_pcm_open_substream(pcm, idx, file, &substream);
+ if (err < 0) {
+ snd_pcm_oss_release_file(pcm_oss_file);
+ return err;
+ }
+
+ pcm_oss_file->streams[idx] = substream;
+ substream->file = pcm_oss_file;
+ snd_pcm_oss_init_substream(substream, &setup[idx], minor);
+ }
+
+ if (!pcm_oss_file->streams[0] && !pcm_oss_file->streams[1]) {
+ snd_pcm_oss_release_file(pcm_oss_file);
+ return -EINVAL;
+ }
+
+ file->private_data = pcm_oss_file;
+ if (rpcm_oss_file)
+ *rpcm_oss_file = pcm_oss_file;
+ return 0;
+}
+
+
+static int snd_task_name(struct task_struct *task, char *name, size_t size)
+{
+ unsigned int idx;
+
+ if (snd_BUG_ON(!task || !name || size < 2))
+ return -EINVAL;
+ for (idx = 0; idx < sizeof(task->comm) && idx + 1 < size; idx++)
+ name[idx] = task->comm[idx];
+ name[idx] = '\0';
+ return 0;
+}
+
+static int snd_pcm_oss_open(struct inode *inode, struct file *file)
+{
+ int err;
+ char task_name[32];
+ struct snd_pcm *pcm;
+ struct snd_pcm_oss_file *pcm_oss_file;
+ struct snd_pcm_oss_setup setup[2];
+ int nonblock;
+ wait_queue_t wait;
+
+ pcm = snd_lookup_oss_minor_data(iminor(inode),
+ SNDRV_OSS_DEVICE_TYPE_PCM);
+ if (pcm == NULL) {
+ err = -ENODEV;
+ goto __error1;
+ }
+ err = snd_card_file_add(pcm->card, file);
+ if (err < 0)
+ goto __error1;
+ if (!try_module_get(pcm->card->module)) {
+ err = -EFAULT;
+ goto __error2;
+ }
+ if (snd_task_name(current, task_name, sizeof(task_name)) < 0) {
+ err = -EFAULT;
+ goto __error;
+ }
+ memset(setup, 0, sizeof(setup));
+ if (file->f_mode & FMODE_WRITE)
+ snd_pcm_oss_look_for_setup(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ task_name, &setup[0]);
+ if (file->f_mode & FMODE_READ)
+ snd_pcm_oss_look_for_setup(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ task_name, &setup[1]);
+
+ nonblock = !!(file->f_flags & O_NONBLOCK);
+ if (!nonblock)
+ nonblock = nonblock_open;
+
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&pcm->open_wait, &wait);
+ mutex_lock(&pcm->open_mutex);
+ while (1) {
+ err = snd_pcm_oss_open_file(file, pcm, &pcm_oss_file,
+ iminor(inode), setup);
+ if (err >= 0)
+ break;
+ if (err == -EAGAIN) {
+ if (nonblock) {
+ err = -EBUSY;
+ break;
+ }
+ } else
+ break;
+ set_current_state(TASK_INTERRUPTIBLE);
+ mutex_unlock(&pcm->open_mutex);
+ schedule();
+ mutex_lock(&pcm->open_mutex);
+ if (signal_pending(current)) {
+ err = -ERESTARTSYS;
+ break;
+ }
+ }
+ remove_wait_queue(&pcm->open_wait, &wait);
+ mutex_unlock(&pcm->open_mutex);
+ if (err < 0)
+ goto __error;
+ return err;
+
+ __error:
+ module_put(pcm->card->module);
+ __error2:
+ snd_card_file_remove(pcm->card, file);
+ __error1:
+ return err;
+}
+
+static int snd_pcm_oss_release(struct inode *inode, struct file *file)
+{
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_oss_file *pcm_oss_file;
+
+ pcm_oss_file = file->private_data;
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream == NULL)
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+ if (snd_BUG_ON(!substream))
+ return -ENXIO;
+ pcm = substream->pcm;
+ if (!pcm->card->shutdown)
+ snd_pcm_oss_sync(pcm_oss_file);
+ mutex_lock(&pcm->open_mutex);
+ snd_pcm_oss_release_file(pcm_oss_file);
+ mutex_unlock(&pcm->open_mutex);
+ wake_up(&pcm->open_wait);
+ module_put(pcm->card->module);
+ snd_card_file_remove(pcm->card, file);
+ return 0;
+}
+
+static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_pcm_oss_file *pcm_oss_file;
+ int __user *p = (int __user *)arg;
+ int res;
+
+ pcm_oss_file = file->private_data;
+ if (cmd == OSS_GETVERSION)
+ return put_user(SNDRV_OSS_VERSION, p);
+ if (cmd == OSS_ALSAEMULVER)
+ return put_user(1, p);
+#if defined(CONFIG_SND_MIXER_OSS) || (defined(MODULE) && defined(CONFIG_SND_MIXER_OSS_MODULE))
+ if (((cmd >> 8) & 0xff) == 'M') { /* mixer ioctl - for OSS compatibility */
+ struct snd_pcm_substream *substream;
+ int idx;
+ for (idx = 0; idx < 2; ++idx) {
+ substream = pcm_oss_file->streams[idx];
+ if (substream != NULL)
+ break;
+ }
+ if (snd_BUG_ON(idx >= 2))
+ return -ENXIO;
+ return snd_mixer_oss_ioctl_card(substream->pcm->card, cmd, arg);
+ }
+#endif
+ if (((cmd >> 8) & 0xff) != 'P')
+ return -EINVAL;
+#ifdef OSS_DEBUG
+ printk("pcm_oss: ioctl = 0x%x\n", cmd);
+#endif
+ switch (cmd) {
+ case SNDCTL_DSP_RESET:
+ return snd_pcm_oss_reset(pcm_oss_file);
+ case SNDCTL_DSP_SYNC:
+ return snd_pcm_oss_sync(pcm_oss_file);
+ case SNDCTL_DSP_SPEED:
+ if (get_user(res, p))
+ return -EFAULT;
+ if ((res = snd_pcm_oss_set_rate(pcm_oss_file, res))<0)
+ return res;
+ return put_user(res, p);
+ case SOUND_PCM_READ_RATE:
+ res = snd_pcm_oss_get_rate(pcm_oss_file);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SNDCTL_DSP_STEREO:
+ if (get_user(res, p))
+ return -EFAULT;
+ res = res > 0 ? 2 : 1;
+ if ((res = snd_pcm_oss_set_channels(pcm_oss_file, res)) < 0)
+ return res;
+ return put_user(--res, p);
+ case SNDCTL_DSP_GETBLKSIZE:
+ res = snd_pcm_oss_get_block_size(pcm_oss_file);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SNDCTL_DSP_SETFMT:
+ if (get_user(res, p))
+ return -EFAULT;
+ res = snd_pcm_oss_set_format(pcm_oss_file, res);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SOUND_PCM_READ_BITS:
+ res = snd_pcm_oss_get_format(pcm_oss_file);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SNDCTL_DSP_CHANNELS:
+ if (get_user(res, p))
+ return -EFAULT;
+ res = snd_pcm_oss_set_channels(pcm_oss_file, res);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SOUND_PCM_READ_CHANNELS:
+ res = snd_pcm_oss_get_channels(pcm_oss_file);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SOUND_PCM_WRITE_FILTER:
+ case SOUND_PCM_READ_FILTER:
+ return -EIO;
+ case SNDCTL_DSP_POST:
+ return snd_pcm_oss_post(pcm_oss_file);
+ case SNDCTL_DSP_SUBDIVIDE:
+ if (get_user(res, p))
+ return -EFAULT;
+ res = snd_pcm_oss_set_subdivide(pcm_oss_file, res);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SNDCTL_DSP_SETFRAGMENT:
+ if (get_user(res, p))
+ return -EFAULT;
+ return snd_pcm_oss_set_fragment(pcm_oss_file, res);
+ case SNDCTL_DSP_GETFMTS:
+ res = snd_pcm_oss_get_formats(pcm_oss_file);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SNDCTL_DSP_GETOSPACE:
+ case SNDCTL_DSP_GETISPACE:
+ return snd_pcm_oss_get_space(pcm_oss_file,
+ cmd == SNDCTL_DSP_GETISPACE ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK,
+ (struct audio_buf_info __user *) arg);
+ case SNDCTL_DSP_NONBLOCK:
+ return snd_pcm_oss_nonblock(file);
+ case SNDCTL_DSP_GETCAPS:
+ res = snd_pcm_oss_get_caps(pcm_oss_file);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SNDCTL_DSP_GETTRIGGER:
+ res = snd_pcm_oss_get_trigger(pcm_oss_file);
+ if (res < 0)
+ return res;
+ return put_user(res, p);
+ case SNDCTL_DSP_SETTRIGGER:
+ if (get_user(res, p))
+ return -EFAULT;
+ return snd_pcm_oss_set_trigger(pcm_oss_file, res);
+ case SNDCTL_DSP_GETIPTR:
+ case SNDCTL_DSP_GETOPTR:
+ return snd_pcm_oss_get_ptr(pcm_oss_file,
+ cmd == SNDCTL_DSP_GETIPTR ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK,
+ (struct count_info __user *) arg);
+ case SNDCTL_DSP_MAPINBUF:
+ case SNDCTL_DSP_MAPOUTBUF:
+ return snd_pcm_oss_get_mapbuf(pcm_oss_file,
+ cmd == SNDCTL_DSP_MAPINBUF ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK,
+ (struct buffmem_desc __user *) arg);
+ case SNDCTL_DSP_SETSYNCRO:
+ /* stop DMA now.. */
+ return 0;
+ case SNDCTL_DSP_SETDUPLEX:
+ if (snd_pcm_oss_get_caps(pcm_oss_file) & DSP_CAP_DUPLEX)
+ return 0;
+ return -EIO;
+ case SNDCTL_DSP_GETODELAY:
+ res = snd_pcm_oss_get_odelay(pcm_oss_file);
+ if (res < 0) {
+ /* it's for sure, some broken apps don't check for error codes */
+ put_user(0, p);
+ return res;
+ }
+ return put_user(res, p);
+ case SNDCTL_DSP_PROFILE:
+ return 0; /* silently ignore */
+ default:
+ snd_printd("pcm_oss: unknown command = 0x%x\n", cmd);
+ }
+ return -EINVAL;
+}
+
+#ifdef CONFIG_COMPAT
+/* all compatible */
+#define snd_pcm_oss_ioctl_compat snd_pcm_oss_ioctl
+#else
+#define snd_pcm_oss_ioctl_compat NULL
+#endif
+
+static ssize_t snd_pcm_oss_read(struct file *file, char __user *buf, size_t count, loff_t *offset)
+{
+ struct snd_pcm_oss_file *pcm_oss_file;
+ struct snd_pcm_substream *substream;
+
+ pcm_oss_file = file->private_data;
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+ if (substream == NULL)
+ return -ENXIO;
+ substream->f_flags = file->f_flags & O_NONBLOCK;
+#ifndef OSS_DEBUG
+ return snd_pcm_oss_read1(substream, buf, count);
+#else
+ {
+ ssize_t res = snd_pcm_oss_read1(substream, buf, count);
+ printk("pcm_oss: read %li bytes (returned %li bytes)\n", (long)count, (long)res);
+ return res;
+ }
+#endif
+}
+
+static ssize_t snd_pcm_oss_write(struct file *file, const char __user *buf, size_t count, loff_t *offset)
+{
+ struct snd_pcm_oss_file *pcm_oss_file;
+ struct snd_pcm_substream *substream;
+ long result;
+
+ pcm_oss_file = file->private_data;
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream == NULL)
+ return -ENXIO;
+ substream->f_flags = file->f_flags & O_NONBLOCK;
+ result = snd_pcm_oss_write1(substream, buf, count);
+#ifdef OSS_DEBUG
+ printk("pcm_oss: write %li bytes (wrote %li bytes)\n", (long)count, (long)result);
+#endif
+ return result;
+}
+
+static int snd_pcm_oss_playback_ready(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (atomic_read(&substream->mmap_count))
+ return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt;
+ else
+ return snd_pcm_playback_avail(runtime) >= runtime->oss.period_frames;
+}
+
+static int snd_pcm_oss_capture_ready(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (atomic_read(&substream->mmap_count))
+ return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt;
+ else
+ return snd_pcm_capture_avail(runtime) >= runtime->oss.period_frames;
+}
+
+static unsigned int snd_pcm_oss_poll(struct file *file, poll_table * wait)
+{
+ struct snd_pcm_oss_file *pcm_oss_file;
+ unsigned int mask;
+ struct snd_pcm_substream *psubstream = NULL, *csubstream = NULL;
+
+ pcm_oss_file = file->private_data;
+
+ psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ csubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+
+ mask = 0;
+ if (psubstream != NULL) {
+ struct snd_pcm_runtime *runtime = psubstream->runtime;
+ poll_wait(file, &runtime->sleep, wait);
+ snd_pcm_stream_lock_irq(psubstream);
+ if (runtime->status->state != SNDRV_PCM_STATE_DRAINING &&
+ (runtime->status->state != SNDRV_PCM_STATE_RUNNING ||
+ snd_pcm_oss_playback_ready(psubstream)))
+ mask |= POLLOUT | POLLWRNORM;
+ snd_pcm_stream_unlock_irq(psubstream);
+ }
+ if (csubstream != NULL) {
+ struct snd_pcm_runtime *runtime = csubstream->runtime;
+ snd_pcm_state_t ostate;
+ poll_wait(file, &runtime->sleep, wait);
+ snd_pcm_stream_lock_irq(csubstream);
+ if ((ostate = runtime->status->state) != SNDRV_PCM_STATE_RUNNING ||
+ snd_pcm_oss_capture_ready(csubstream))
+ mask |= POLLIN | POLLRDNORM;
+ snd_pcm_stream_unlock_irq(csubstream);
+ if (ostate != SNDRV_PCM_STATE_RUNNING && runtime->oss.trigger) {
+ struct snd_pcm_oss_file ofile;
+ memset(&ofile, 0, sizeof(ofile));
+ ofile.streams[SNDRV_PCM_STREAM_CAPTURE] = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+ runtime->oss.trigger = 0;
+ snd_pcm_oss_set_trigger(&ofile, PCM_ENABLE_INPUT);
+ }
+ }
+
+ return mask;
+}
+
+static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area)
+{
+ struct snd_pcm_oss_file *pcm_oss_file;
+ struct snd_pcm_substream *substream = NULL;
+ struct snd_pcm_runtime *runtime;
+ int err;
+
+#ifdef OSS_DEBUG
+ printk("pcm_oss: mmap begin\n");
+#endif
+ pcm_oss_file = file->private_data;
+ switch ((area->vm_flags & (VM_READ | VM_WRITE))) {
+ case VM_READ | VM_WRITE:
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (substream)
+ break;
+ /* Fall through */
+ case VM_READ:
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
+ break;
+ case VM_WRITE:
+ substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ break;
+ default:
+ return -EINVAL;
+ }
+ /* set VM_READ access as well to fix memset() routines that do
+ reads before writes (to improve performance) */
+ area->vm_flags |= VM_READ;
+ if (substream == NULL)
+ return -ENXIO;
+ runtime = substream->runtime;
+ if (!(runtime->info & SNDRV_PCM_INFO_MMAP_VALID))
+ return -EIO;
+ if (runtime->info & SNDRV_PCM_INFO_INTERLEAVED)
+ runtime->access = SNDRV_PCM_ACCESS_MMAP_INTERLEAVED;
+ else
+ return -EIO;
+
+ if (runtime->oss.params) {
+ if ((err = snd_pcm_oss_change_params(substream)) < 0)
+ return err;
+ }
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+ if (runtime->oss.plugin_first != NULL)
+ return -EIO;
+#endif
+
+ if (area->vm_pgoff != 0)
+ return -EINVAL;
+
+ err = snd_pcm_mmap_data(substream, file, area);
+ if (err < 0)
+ return err;
+ runtime->oss.mmap_bytes = area->vm_end - area->vm_start;
+ runtime->silence_threshold = 0;
+ runtime->silence_size = 0;
+#ifdef OSS_DEBUG
+ printk("pcm_oss: mmap ok, bytes = 0x%x\n", runtime->oss.mmap_bytes);
+#endif
+ /* In mmap mode we never stop */
+ runtime->stop_threshold = runtime->boundary;
+
+ return 0;
+}
+
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+/*
+ * /proc interface
+ */
+
+static void snd_pcm_oss_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_str *pstr = entry->private_data;
+ struct snd_pcm_oss_setup *setup = pstr->oss.setup_list;
+ mutex_lock(&pstr->oss.setup_mutex);
+ while (setup) {
+ snd_iprintf(buffer, "%s %u %u%s%s%s%s%s%s\n",
+ setup->task_name,
+ setup->periods,
+ setup->period_size,
+ setup->disable ? " disable" : "",
+ setup->direct ? " direct" : "",
+ setup->block ? " block" : "",
+ setup->nonblock ? " non-block" : "",
+ setup->partialfrag ? " partial-frag" : "",
+ setup->nosilence ? " no-silence" : "");
+ setup = setup->next;
+ }
+ mutex_unlock(&pstr->oss.setup_mutex);
+}
+
+static void snd_pcm_oss_proc_free_setup_list(struct snd_pcm_str * pstr)
+{
+ struct snd_pcm_oss_setup *setup, *setupn;
+
+ for (setup = pstr->oss.setup_list, pstr->oss.setup_list = NULL;
+ setup; setup = setupn) {
+ setupn = setup->next;
+ kfree(setup->task_name);
+ kfree(setup);
+ }
+ pstr->oss.setup_list = NULL;
+}
+
+static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_str *pstr = entry->private_data;
+ char line[128], str[32], task_name[32], *ptr;
+ int idx1;
+ struct snd_pcm_oss_setup *setup, *setup1, template;
+
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ mutex_lock(&pstr->oss.setup_mutex);
+ memset(&template, 0, sizeof(template));
+ ptr = snd_info_get_str(task_name, line, sizeof(task_name));
+ if (!strcmp(task_name, "clear") || !strcmp(task_name, "erase")) {
+ snd_pcm_oss_proc_free_setup_list(pstr);
+ mutex_unlock(&pstr->oss.setup_mutex);
+ continue;
+ }
+ for (setup = pstr->oss.setup_list; setup; setup = setup->next) {
+ if (!strcmp(setup->task_name, task_name)) {
+ template = *setup;
+ break;
+ }
+ }
+ ptr = snd_info_get_str(str, ptr, sizeof(str));
+ template.periods = simple_strtoul(str, NULL, 10);
+ ptr = snd_info_get_str(str, ptr, sizeof(str));
+ template.period_size = simple_strtoul(str, NULL, 10);
+ for (idx1 = 31; idx1 >= 0; idx1--)
+ if (template.period_size & (1 << idx1))
+ break;
+ for (idx1--; idx1 >= 0; idx1--)
+ template.period_size &= ~(1 << idx1);
+ do {
+ ptr = snd_info_get_str(str, ptr, sizeof(str));
+ if (!strcmp(str, "disable")) {
+ template.disable = 1;
+ } else if (!strcmp(str, "direct")) {
+ template.direct = 1;
+ } else if (!strcmp(str, "block")) {
+ template.block = 1;
+ } else if (!strcmp(str, "non-block")) {
+ template.nonblock = 1;
+ } else if (!strcmp(str, "partial-frag")) {
+ template.partialfrag = 1;
+ } else if (!strcmp(str, "no-silence")) {
+ template.nosilence = 1;
+ } else if (!strcmp(str, "buggy-ptr")) {
+ template.buggyptr = 1;
+ }
+ } while (*str);
+ if (setup == NULL) {
+ setup = kmalloc(sizeof(*setup), GFP_KERNEL);
+ if (! setup) {
+ buffer->error = -ENOMEM;
+ mutex_unlock(&pstr->oss.setup_mutex);
+ return;
+ }
+ if (pstr->oss.setup_list == NULL)
+ pstr->oss.setup_list = setup;
+ else {
+ for (setup1 = pstr->oss.setup_list;
+ setup1->next; setup1 = setup1->next);
+ setup1->next = setup;
+ }
+ template.task_name = kstrdup(task_name, GFP_KERNEL);
+ if (! template.task_name) {
+ kfree(setup);
+ buffer->error = -ENOMEM;
+ mutex_unlock(&pstr->oss.setup_mutex);
+ return;
+ }
+ }
+ *setup = template;
+ mutex_unlock(&pstr->oss.setup_mutex);
+ }
+}
+
+static void snd_pcm_oss_proc_init(struct snd_pcm *pcm)
+{
+ int stream;
+ for (stream = 0; stream < 2; ++stream) {
+ struct snd_info_entry *entry;
+ struct snd_pcm_str *pstr = &pcm->streams[stream];
+ if (pstr->substream_count == 0)
+ continue;
+ if ((entry = snd_info_create_card_entry(pcm->card, "oss", pstr->proc_root)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = snd_pcm_oss_proc_read;
+ entry->c.text.write = snd_pcm_oss_proc_write;
+ entry->private_data = pstr;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ pstr->oss.proc_entry = entry;
+ }
+}
+
+static void snd_pcm_oss_proc_done(struct snd_pcm *pcm)
+{
+ int stream;
+ for (stream = 0; stream < 2; ++stream) {
+ struct snd_pcm_str *pstr = &pcm->streams[stream];
+ snd_info_free_entry(pstr->oss.proc_entry);
+ pstr->oss.proc_entry = NULL;
+ snd_pcm_oss_proc_free_setup_list(pstr);
+ }
+}
+#else /* !CONFIG_SND_VERBOSE_PROCFS */
+#define snd_pcm_oss_proc_init(pcm)
+#define snd_pcm_oss_proc_done(pcm)
+#endif /* CONFIG_SND_VERBOSE_PROCFS */
+
+/*
+ * ENTRY functions
+ */
+
+static const struct file_operations snd_pcm_oss_f_reg =
+{
+ .owner = THIS_MODULE,
+ .read = snd_pcm_oss_read,
+ .write = snd_pcm_oss_write,
+ .open = snd_pcm_oss_open,
+ .release = snd_pcm_oss_release,
+ .poll = snd_pcm_oss_poll,
+ .unlocked_ioctl = snd_pcm_oss_ioctl,
+ .compat_ioctl = snd_pcm_oss_ioctl_compat,
+ .mmap = snd_pcm_oss_mmap,
+};
+
+static void register_oss_dsp(struct snd_pcm *pcm, int index)
+{
+ char name[128];
+ sprintf(name, "dsp%i%i", pcm->card->number, pcm->device);
+ if (snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_PCM,
+ pcm->card, index, &snd_pcm_oss_f_reg,
+ pcm, name) < 0) {
+ snd_printk(KERN_ERR "unable to register OSS PCM device %i:%i\n",
+ pcm->card->number, pcm->device);
+ }
+}
+
+static int snd_pcm_oss_register_minor(struct snd_pcm *pcm)
+{
+ pcm->oss.reg = 0;
+ if (dsp_map[pcm->card->number] == (int)pcm->device) {
+ char name[128];
+ int duplex;
+ register_oss_dsp(pcm, 0);
+ duplex = (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream_count > 0 &&
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream_count &&
+ !(pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX));
+ sprintf(name, "%s%s", pcm->name, duplex ? " (DUPLEX)" : "");
+#ifdef SNDRV_OSS_INFO_DEV_AUDIO
+ snd_oss_info_register(SNDRV_OSS_INFO_DEV_AUDIO,
+ pcm->card->number,
+ name);
+#endif
+ pcm->oss.reg++;
+ pcm->oss.reg_mask |= 1;
+ }
+ if (adsp_map[pcm->card->number] == (int)pcm->device) {
+ register_oss_dsp(pcm, 1);
+ pcm->oss.reg++;
+ pcm->oss.reg_mask |= 2;
+ }
+
+ if (pcm->oss.reg)
+ snd_pcm_oss_proc_init(pcm);
+
+ return 0;
+}
+
+static int snd_pcm_oss_disconnect_minor(struct snd_pcm *pcm)
+{
+ if (pcm->oss.reg) {
+ if (pcm->oss.reg_mask & 1) {
+ pcm->oss.reg_mask &= ~1;
+ snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_PCM,
+ pcm->card, 0);
+ }
+ if (pcm->oss.reg_mask & 2) {
+ pcm->oss.reg_mask &= ~2;
+ snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_PCM,
+ pcm->card, 1);
+ }
+ if (dsp_map[pcm->card->number] == (int)pcm->device) {
+#ifdef SNDRV_OSS_INFO_DEV_AUDIO
+ snd_oss_info_unregister(SNDRV_OSS_INFO_DEV_AUDIO, pcm->card->number);
+#endif
+ }
+ pcm->oss.reg = 0;
+ }
+ return 0;
+}
+
+static int snd_pcm_oss_unregister_minor(struct snd_pcm *pcm)
+{
+ snd_pcm_oss_disconnect_minor(pcm);
+ snd_pcm_oss_proc_done(pcm);
+ return 0;
+}
+
+static struct snd_pcm_notify snd_pcm_oss_notify =
+{
+ .n_register = snd_pcm_oss_register_minor,
+ .n_disconnect = snd_pcm_oss_disconnect_minor,
+ .n_unregister = snd_pcm_oss_unregister_minor,
+};
+
+static int __init alsa_pcm_oss_init(void)
+{
+ int i;
+ int err;
+
+ /* check device map table */
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ if (dsp_map[i] < 0 || dsp_map[i] >= SNDRV_PCM_DEVICES) {
+ snd_printk(KERN_ERR "invalid dsp_map[%d] = %d\n",
+ i, dsp_map[i]);
+ dsp_map[i] = 0;
+ }
+ if (adsp_map[i] < 0 || adsp_map[i] >= SNDRV_PCM_DEVICES) {
+ snd_printk(KERN_ERR "invalid adsp_map[%d] = %d\n",
+ i, adsp_map[i]);
+ adsp_map[i] = 1;
+ }
+ }
+ if ((err = snd_pcm_notify(&snd_pcm_oss_notify, 0)) < 0)
+ return err;
+ return 0;
+}
+
+static void __exit alsa_pcm_oss_exit(void)
+{
+ snd_pcm_notify(&snd_pcm_oss_notify, 1);
+}
+
+module_init(alsa_pcm_oss_init)
+module_exit(alsa_pcm_oss_exit)
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
new file mode 100644
index 0000000..6751daa
--- /dev/null
+++ b/sound/core/oss/pcm_plugin.c
@@ -0,0 +1,752 @@
+/*
+ * PCM Plug-In shared (kernel/library) code
+ * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
+ * Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#if 0
+#define PLUGIN_DEBUG
+#endif
+
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/vmalloc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "pcm_plugin.h"
+
+#define snd_pcm_plug_first(plug) ((plug)->runtime->oss.plugin_first)
+#define snd_pcm_plug_last(plug) ((plug)->runtime->oss.plugin_last)
+
+/*
+ * because some cards might have rates "very close", we ignore
+ * all "resampling" requests within +-5%
+ */
+static int rate_match(unsigned int src_rate, unsigned int dst_rate)
+{
+ unsigned int low = (src_rate * 95) / 100;
+ unsigned int high = (src_rate * 105) / 100;
+ return dst_rate >= low && dst_rate <= high;
+}
+
+static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_plugin_format *format;
+ ssize_t width;
+ size_t size;
+ unsigned int channel;
+ struct snd_pcm_plugin_channel *c;
+
+ if (plugin->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ format = &plugin->src_format;
+ } else {
+ format = &plugin->dst_format;
+ }
+ if ((width = snd_pcm_format_physical_width(format->format)) < 0)
+ return width;
+ size = frames * format->channels * width;
+ if (snd_BUG_ON(size % 8))
+ return -ENXIO;
+ size /= 8;
+ if (plugin->buf_frames < frames) {
+ vfree(plugin->buf);
+ plugin->buf = vmalloc(size);
+ plugin->buf_frames = frames;
+ }
+ if (!plugin->buf) {
+ plugin->buf_frames = 0;
+ return -ENOMEM;
+ }
+ c = plugin->buf_channels;
+ if (plugin->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED) {
+ for (channel = 0; channel < format->channels; channel++, c++) {
+ c->frames = frames;
+ c->enabled = 1;
+ c->wanted = 0;
+ c->area.addr = plugin->buf;
+ c->area.first = channel * width;
+ c->area.step = format->channels * width;
+ }
+ } else if (plugin->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED) {
+ if (snd_BUG_ON(size % format->channels))
+ return -EINVAL;
+ size /= format->channels;
+ for (channel = 0; channel < format->channels; channel++, c++) {
+ c->frames = frames;
+ c->enabled = 1;
+ c->wanted = 0;
+ c->area.addr = plugin->buf + (channel * size);
+ c->area.first = 0;
+ c->area.step = width;
+ }
+ } else
+ return -EINVAL;
+ return 0;
+}
+
+int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames)
+{
+ int err;
+ if (snd_BUG_ON(!snd_pcm_plug_first(plug)))
+ return -ENXIO;
+ if (snd_pcm_plug_stream(plug) == SNDRV_PCM_STREAM_PLAYBACK) {
+ struct snd_pcm_plugin *plugin = snd_pcm_plug_first(plug);
+ while (plugin->next) {
+ if (plugin->dst_frames)
+ frames = plugin->dst_frames(plugin, frames);
+ if (snd_BUG_ON(frames <= 0))
+ return -ENXIO;
+ plugin = plugin->next;
+ err = snd_pcm_plugin_alloc(plugin, frames);
+ if (err < 0)
+ return err;
+ }
+ } else {
+ struct snd_pcm_plugin *plugin = snd_pcm_plug_last(plug);
+ while (plugin->prev) {
+ if (plugin->src_frames)
+ frames = plugin->src_frames(plugin, frames);
+ if (snd_BUG_ON(frames <= 0))
+ return -ENXIO;
+ plugin = plugin->prev;
+ err = snd_pcm_plugin_alloc(plugin, frames);
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+
+snd_pcm_sframes_t snd_pcm_plugin_client_channels(struct snd_pcm_plugin *plugin,
+ snd_pcm_uframes_t frames,
+ struct snd_pcm_plugin_channel **channels)
+{
+ *channels = plugin->buf_channels;
+ return frames;
+}
+
+int snd_pcm_plugin_build(struct snd_pcm_substream *plug,
+ const char *name,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ size_t extra,
+ struct snd_pcm_plugin **ret)
+{
+ struct snd_pcm_plugin *plugin;
+ unsigned int channels;
+
+ if (snd_BUG_ON(!plug))
+ return -ENXIO;
+ if (snd_BUG_ON(!src_format || !dst_format))
+ return -ENXIO;
+ plugin = kzalloc(sizeof(*plugin) + extra, GFP_KERNEL);
+ if (plugin == NULL)
+ return -ENOMEM;
+ plugin->name = name;
+ plugin->plug = plug;
+ plugin->stream = snd_pcm_plug_stream(plug);
+ plugin->access = SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+ plugin->src_format = *src_format;
+ plugin->src_width = snd_pcm_format_physical_width(src_format->format);
+ snd_BUG_ON(plugin->src_width <= 0);
+ plugin->dst_format = *dst_format;
+ plugin->dst_width = snd_pcm_format_physical_width(dst_format->format);
+ snd_BUG_ON(plugin->dst_width <= 0);
+ if (plugin->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ channels = src_format->channels;
+ else
+ channels = dst_format->channels;
+ plugin->buf_channels = kcalloc(channels, sizeof(*plugin->buf_channels), GFP_KERNEL);
+ if (plugin->buf_channels == NULL) {
+ snd_pcm_plugin_free(plugin);
+ return -ENOMEM;
+ }
+ plugin->client_channels = snd_pcm_plugin_client_channels;
+ *ret = plugin;
+ return 0;
+}
+
+int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin)
+{
+ if (! plugin)
+ return 0;
+ if (plugin->private_free)
+ plugin->private_free(plugin);
+ kfree(plugin->buf_channels);
+ vfree(plugin->buf);
+ kfree(plugin);
+ return 0;
+}
+
+snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
+{
+ struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
+ int stream = snd_pcm_plug_stream(plug);
+
+ if (snd_BUG_ON(!plug))
+ return -ENXIO;
+ if (drv_frames == 0)
+ return 0;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ plugin = snd_pcm_plug_last(plug);
+ while (plugin && drv_frames > 0) {
+ plugin_prev = plugin->prev;
+ if (plugin->src_frames)
+ drv_frames = plugin->src_frames(plugin, drv_frames);
+ plugin = plugin_prev;
+ }
+ } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ plugin = snd_pcm_plug_first(plug);
+ while (plugin && drv_frames > 0) {
+ plugin_next = plugin->next;
+ if (plugin->dst_frames)
+ drv_frames = plugin->dst_frames(plugin, drv_frames);
+ plugin = plugin_next;
+ }
+ } else
+ snd_BUG();
+ return drv_frames;
+}
+
+snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames)
+{
+ struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
+ snd_pcm_sframes_t frames;
+ int stream = snd_pcm_plug_stream(plug);
+
+ if (snd_BUG_ON(!plug))
+ return -ENXIO;
+ if (clt_frames == 0)
+ return 0;
+ frames = clt_frames;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ plugin = snd_pcm_plug_first(plug);
+ while (plugin && frames > 0) {
+ plugin_next = plugin->next;
+ if (plugin->dst_frames) {
+ frames = plugin->dst_frames(plugin, frames);
+ if (frames < 0)
+ return frames;
+ }
+ plugin = plugin_next;
+ }
+ } else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ plugin = snd_pcm_plug_last(plug);
+ while (plugin) {
+ plugin_prev = plugin->prev;
+ if (plugin->src_frames) {
+ frames = plugin->src_frames(plugin, frames);
+ if (frames < 0)
+ return frames;
+ }
+ plugin = plugin_prev;
+ }
+ } else
+ snd_BUG();
+ return frames;
+}
+
+static int snd_pcm_plug_formats(struct snd_mask *mask, int format)
+{
+ struct snd_mask formats = *mask;
+ u64 linfmts = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_U24_BE | SNDRV_PCM_FMTBIT_S24_BE |
+ SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE |
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE);
+ snd_mask_set(&formats, SNDRV_PCM_FORMAT_MU_LAW);
+
+ if (formats.bits[0] & (u32)linfmts)
+ formats.bits[0] |= (u32)linfmts;
+ if (formats.bits[1] & (u32)(linfmts >> 32))
+ formats.bits[1] |= (u32)(linfmts >> 32);
+ return snd_mask_test(&formats, format);
+}
+
+static int preferred_formats[] = {
+ SNDRV_PCM_FORMAT_S16_LE,
+ SNDRV_PCM_FORMAT_S16_BE,
+ SNDRV_PCM_FORMAT_U16_LE,
+ SNDRV_PCM_FORMAT_U16_BE,
+ SNDRV_PCM_FORMAT_S24_3LE,
+ SNDRV_PCM_FORMAT_S24_3BE,
+ SNDRV_PCM_FORMAT_U24_3LE,
+ SNDRV_PCM_FORMAT_U24_3BE,
+ SNDRV_PCM_FORMAT_S24_LE,
+ SNDRV_PCM_FORMAT_S24_BE,
+ SNDRV_PCM_FORMAT_U24_LE,
+ SNDRV_PCM_FORMAT_U24_BE,
+ SNDRV_PCM_FORMAT_S32_LE,
+ SNDRV_PCM_FORMAT_S32_BE,
+ SNDRV_PCM_FORMAT_U32_LE,
+ SNDRV_PCM_FORMAT_U32_BE,
+ SNDRV_PCM_FORMAT_S8,
+ SNDRV_PCM_FORMAT_U8
+};
+
+int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask)
+{
+ int i;
+
+ if (snd_mask_test(format_mask, format))
+ return format;
+ if (! snd_pcm_plug_formats(format_mask, format))
+ return -EINVAL;
+ if (snd_pcm_format_linear(format)) {
+ unsigned int width = snd_pcm_format_width(format);
+ int unsignd = snd_pcm_format_unsigned(format) > 0;
+ int big = snd_pcm_format_big_endian(format) > 0;
+ unsigned int badness, best = -1;
+ int best_format = -1;
+ for (i = 0; i < ARRAY_SIZE(preferred_formats); i++) {
+ int f = preferred_formats[i];
+ unsigned int w;
+ if (!snd_mask_test(format_mask, f))
+ continue;
+ w = snd_pcm_format_width(f);
+ if (w >= width)
+ badness = w - width;
+ else
+ badness = width - w + 32;
+ badness += snd_pcm_format_unsigned(f) != unsignd;
+ badness += snd_pcm_format_big_endian(f) != big;
+ if (badness < best) {
+ best_format = f;
+ best = badness;
+ }
+ }
+ return best_format >= 0 ? best_format : -EINVAL;
+ } else {
+ switch (format) {
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ for (i = 0; i < ARRAY_SIZE(preferred_formats); ++i) {
+ int format1 = preferred_formats[i];
+ if (snd_mask_test(format_mask, format1))
+ return format1;
+ }
+ default:
+ return -EINVAL;
+ }
+ }
+}
+
+int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug,
+ struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_params *slave_params)
+{
+ struct snd_pcm_plugin_format tmpformat;
+ struct snd_pcm_plugin_format dstformat;
+ struct snd_pcm_plugin_format srcformat;
+ int src_access, dst_access;
+ struct snd_pcm_plugin *plugin = NULL;
+ int err;
+ int stream = snd_pcm_plug_stream(plug);
+ int slave_interleaved = (params_channels(slave_params) == 1 ||
+ params_access(slave_params) == SNDRV_PCM_ACCESS_RW_INTERLEAVED);
+
+ switch (stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ dstformat.format = params_format(slave_params);
+ dstformat.rate = params_rate(slave_params);
+ dstformat.channels = params_channels(slave_params);
+ srcformat.format = params_format(params);
+ srcformat.rate = params_rate(params);
+ srcformat.channels = params_channels(params);
+ src_access = SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+ dst_access = (slave_interleaved ? SNDRV_PCM_ACCESS_RW_INTERLEAVED :
+ SNDRV_PCM_ACCESS_RW_NONINTERLEAVED);
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ dstformat.format = params_format(params);
+ dstformat.rate = params_rate(params);
+ dstformat.channels = params_channels(params);
+ srcformat.format = params_format(slave_params);
+ srcformat.rate = params_rate(slave_params);
+ srcformat.channels = params_channels(slave_params);
+ src_access = (slave_interleaved ? SNDRV_PCM_ACCESS_RW_INTERLEAVED :
+ SNDRV_PCM_ACCESS_RW_NONINTERLEAVED);
+ dst_access = SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+ tmpformat = srcformat;
+
+ pdprintf("srcformat: format=%i, rate=%i, channels=%i\n",
+ srcformat.format,
+ srcformat.rate,
+ srcformat.channels);
+ pdprintf("dstformat: format=%i, rate=%i, channels=%i\n",
+ dstformat.format,
+ dstformat.rate,
+ dstformat.channels);
+
+ /* Format change (linearization) */
+ if (! rate_match(srcformat.rate, dstformat.rate) &&
+ ! snd_pcm_format_linear(srcformat.format)) {
+ if (srcformat.format != SNDRV_PCM_FORMAT_MU_LAW)
+ return -EINVAL;
+ tmpformat.format = SNDRV_PCM_FORMAT_S16;
+ err = snd_pcm_plugin_build_mulaw(plug,
+ &srcformat, &tmpformat,
+ &plugin);
+ if (err < 0)
+ return err;
+ err = snd_pcm_plugin_append(plugin);
+ if (err < 0) {
+ snd_pcm_plugin_free(plugin);
+ return err;
+ }
+ srcformat = tmpformat;
+ src_access = dst_access;
+ }
+
+ /* channels reduction */
+ if (srcformat.channels > dstformat.channels) {
+ tmpformat.channels = dstformat.channels;
+ err = snd_pcm_plugin_build_route(plug, &srcformat, &tmpformat, &plugin);
+ pdprintf("channels reduction: src=%i, dst=%i returns %i\n", srcformat.channels, tmpformat.channels, err);
+ if (err < 0)
+ return err;
+ err = snd_pcm_plugin_append(plugin);
+ if (err < 0) {
+ snd_pcm_plugin_free(plugin);
+ return err;
+ }
+ srcformat = tmpformat;
+ src_access = dst_access;
+ }
+
+ /* rate resampling */
+ if (!rate_match(srcformat.rate, dstformat.rate)) {
+ if (srcformat.format != SNDRV_PCM_FORMAT_S16) {
+ /* convert to S16 for resampling */
+ tmpformat.format = SNDRV_PCM_FORMAT_S16;
+ err = snd_pcm_plugin_build_linear(plug,
+ &srcformat, &tmpformat,
+ &plugin);
+ if (err < 0)
+ return err;
+ err = snd_pcm_plugin_append(plugin);
+ if (err < 0) {
+ snd_pcm_plugin_free(plugin);
+ return err;
+ }
+ srcformat = tmpformat;
+ src_access = dst_access;
+ }
+ tmpformat.rate = dstformat.rate;
+ err = snd_pcm_plugin_build_rate(plug,
+ &srcformat, &tmpformat,
+ &plugin);
+ pdprintf("rate down resampling: src=%i, dst=%i returns %i\n", srcformat.rate, tmpformat.rate, err);
+ if (err < 0)
+ return err;
+ err = snd_pcm_plugin_append(plugin);
+ if (err < 0) {
+ snd_pcm_plugin_free(plugin);
+ return err;
+ }
+ srcformat = tmpformat;
+ src_access = dst_access;
+ }
+
+ /* format change */
+ if (srcformat.format != dstformat.format) {
+ tmpformat.format = dstformat.format;
+ if (srcformat.format == SNDRV_PCM_FORMAT_MU_LAW ||
+ tmpformat.format == SNDRV_PCM_FORMAT_MU_LAW) {
+ err = snd_pcm_plugin_build_mulaw(plug,
+ &srcformat, &tmpformat,
+ &plugin);
+ }
+ else if (snd_pcm_format_linear(srcformat.format) &&
+ snd_pcm_format_linear(tmpformat.format)) {
+ err = snd_pcm_plugin_build_linear(plug,
+ &srcformat, &tmpformat,
+ &plugin);
+ }
+ else
+ return -EINVAL;
+ pdprintf("format change: src=%i, dst=%i returns %i\n", srcformat.format, tmpformat.format, err);
+ if (err < 0)
+ return err;
+ err = snd_pcm_plugin_append(plugin);
+ if (err < 0) {
+ snd_pcm_plugin_free(plugin);
+ return err;
+ }
+ srcformat = tmpformat;
+ src_access = dst_access;
+ }
+
+ /* channels extension */
+ if (srcformat.channels < dstformat.channels) {
+ tmpformat.channels = dstformat.channels;
+ err = snd_pcm_plugin_build_route(plug, &srcformat, &tmpformat, &plugin);
+ pdprintf("channels extension: src=%i, dst=%i returns %i\n", srcformat.channels, tmpformat.channels, err);
+ if (err < 0)
+ return err;
+ err = snd_pcm_plugin_append(plugin);
+ if (err < 0) {
+ snd_pcm_plugin_free(plugin);
+ return err;
+ }
+ srcformat = tmpformat;
+ src_access = dst_access;
+ }
+
+ /* de-interleave */
+ if (src_access != dst_access) {
+ err = snd_pcm_plugin_build_copy(plug,
+ &srcformat,
+ &tmpformat,
+ &plugin);
+ pdprintf("interleave change (copy: returns %i)\n", err);
+ if (err < 0)
+ return err;
+ err = snd_pcm_plugin_append(plugin);
+ if (err < 0) {
+ snd_pcm_plugin_free(plugin);
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+snd_pcm_sframes_t snd_pcm_plug_client_channels_buf(struct snd_pcm_substream *plug,
+ char *buf,
+ snd_pcm_uframes_t count,
+ struct snd_pcm_plugin_channel **channels)
+{
+ struct snd_pcm_plugin *plugin;
+ struct snd_pcm_plugin_channel *v;
+ struct snd_pcm_plugin_format *format;
+ int width, nchannels, channel;
+ int stream = snd_pcm_plug_stream(plug);
+
+ if (snd_BUG_ON(!buf))
+ return -ENXIO;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ plugin = snd_pcm_plug_first(plug);
+ format = &plugin->src_format;
+ } else {
+ plugin = snd_pcm_plug_last(plug);
+ format = &plugin->dst_format;
+ }
+ v = plugin->buf_channels;
+ *channels = v;
+ if ((width = snd_pcm_format_physical_width(format->format)) < 0)
+ return width;
+ nchannels = format->channels;
+ if (snd_BUG_ON(plugin->access != SNDRV_PCM_ACCESS_RW_INTERLEAVED &&
+ format->channels > 1))
+ return -ENXIO;
+ for (channel = 0; channel < nchannels; channel++, v++) {
+ v->frames = count;
+ v->enabled = 1;
+ v->wanted = (stream == SNDRV_PCM_STREAM_CAPTURE);
+ v->area.addr = buf;
+ v->area.first = channel * width;
+ v->area.step = nchannels * width;
+ }
+ return count;
+}
+
+snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *src_channels, snd_pcm_uframes_t size)
+{
+ struct snd_pcm_plugin *plugin, *next;
+ struct snd_pcm_plugin_channel *dst_channels;
+ int err;
+ snd_pcm_sframes_t frames = size;
+
+ plugin = snd_pcm_plug_first(plug);
+ while (plugin && frames > 0) {
+ if ((next = plugin->next) != NULL) {
+ snd_pcm_sframes_t frames1 = frames;
+ if (plugin->dst_frames)
+ frames1 = plugin->dst_frames(plugin, frames);
+ if ((err = next->client_channels(next, frames1, &dst_channels)) < 0) {
+ return err;
+ }
+ if (err != frames1) {
+ frames = err;
+ if (plugin->src_frames)
+ frames = plugin->src_frames(plugin, frames1);
+ }
+ } else
+ dst_channels = NULL;
+ pdprintf("write plugin: %s, %li\n", plugin->name, frames);
+ if ((frames = plugin->transfer(plugin, src_channels, dst_channels, frames)) < 0)
+ return frames;
+ src_channels = dst_channels;
+ plugin = next;
+ }
+ return snd_pcm_plug_client_size(plug, frames);
+}
+
+snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *dst_channels_final, snd_pcm_uframes_t size)
+{
+ struct snd_pcm_plugin *plugin, *next;
+ struct snd_pcm_plugin_channel *src_channels, *dst_channels;
+ snd_pcm_sframes_t frames = size;
+ int err;
+
+ frames = snd_pcm_plug_slave_size(plug, frames);
+ if (frames < 0)
+ return frames;
+
+ src_channels = NULL;
+ plugin = snd_pcm_plug_first(plug);
+ while (plugin && frames > 0) {
+ if ((next = plugin->next) != NULL) {
+ if ((err = plugin->client_channels(plugin, frames, &dst_channels)) < 0) {
+ return err;
+ }
+ frames = err;
+ } else {
+ dst_channels = dst_channels_final;
+ }
+ pdprintf("read plugin: %s, %li\n", plugin->name, frames);
+ if ((frames = plugin->transfer(plugin, src_channels, dst_channels, frames)) < 0)
+ return frames;
+ plugin = next;
+ src_channels = dst_channels;
+ }
+ return frames;
+}
+
+int snd_pcm_area_silence(const struct snd_pcm_channel_area *dst_area, size_t dst_offset,
+ size_t samples, int format)
+{
+ /* FIXME: sub byte resolution and odd dst_offset */
+ unsigned char *dst;
+ unsigned int dst_step;
+ int width;
+ const unsigned char *silence;
+ if (!dst_area->addr)
+ return 0;
+ dst = dst_area->addr + (dst_area->first + dst_area->step * dst_offset) / 8;
+ width = snd_pcm_format_physical_width(format);
+ if (width <= 0)
+ return -EINVAL;
+ if (dst_area->step == (unsigned int) width && width >= 8)
+ return snd_pcm_format_set_silence(format, dst, samples);
+ silence = snd_pcm_format_silence_64(format);
+ if (! silence)
+ return -EINVAL;
+ dst_step = dst_area->step / 8;
+ if (width == 4) {
+ /* Ima ADPCM */
+ int dstbit = dst_area->first % 8;
+ int dstbit_step = dst_area->step % 8;
+ while (samples-- > 0) {
+ if (dstbit)
+ *dst &= 0xf0;
+ else
+ *dst &= 0x0f;
+ dst += dst_step;
+ dstbit += dstbit_step;
+ if (dstbit == 8) {
+ dst++;
+ dstbit = 0;
+ }
+ }
+ } else {
+ width /= 8;
+ while (samples-- > 0) {
+ memcpy(dst, silence, width);
+ dst += dst_step;
+ }
+ }
+ return 0;
+}
+
+int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_area, size_t src_offset,
+ const struct snd_pcm_channel_area *dst_area, size_t dst_offset,
+ size_t samples, int format)
+{
+ /* FIXME: sub byte resolution and odd dst_offset */
+ char *src, *dst;
+ int width;
+ int src_step, dst_step;
+ src = src_area->addr + (src_area->first + src_area->step * src_offset) / 8;
+ if (!src_area->addr)
+ return snd_pcm_area_silence(dst_area, dst_offset, samples, format);
+ dst = dst_area->addr + (dst_area->first + dst_area->step * dst_offset) / 8;
+ if (!dst_area->addr)
+ return 0;
+ width = snd_pcm_format_physical_width(format);
+ if (width <= 0)
+ return -EINVAL;
+ if (src_area->step == (unsigned int) width &&
+ dst_area->step == (unsigned int) width && width >= 8) {
+ size_t bytes = samples * width / 8;
+ memcpy(dst, src, bytes);
+ return 0;
+ }
+ src_step = src_area->step / 8;
+ dst_step = dst_area->step / 8;
+ if (width == 4) {
+ /* Ima ADPCM */
+ int srcbit = src_area->first % 8;
+ int srcbit_step = src_area->step % 8;
+ int dstbit = dst_area->first % 8;
+ int dstbit_step = dst_area->step % 8;
+ while (samples-- > 0) {
+ unsigned char srcval;
+ if (srcbit)
+ srcval = *src & 0x0f;
+ else
+ srcval = (*src & 0xf0) >> 4;
+ if (dstbit)
+ *dst = (*dst & 0xf0) | srcval;
+ else
+ *dst = (*dst & 0x0f) | (srcval << 4);
+ src += src_step;
+ srcbit += srcbit_step;
+ if (srcbit == 8) {
+ src++;
+ srcbit = 0;
+ }
+ dst += dst_step;
+ dstbit += dstbit_step;
+ if (dstbit == 8) {
+ dst++;
+ dstbit = 0;
+ }
+ }
+ } else {
+ width /= 8;
+ while (samples-- > 0) {
+ memcpy(dst, src, width);
+ src += src_step;
+ dst += dst_step;
+ }
+ }
+ return 0;
+}
diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h
new file mode 100644
index 0000000..ca2f4c3
--- /dev/null
+++ b/sound/core/oss/pcm_plugin.h
@@ -0,0 +1,184 @@
+#ifndef __PCM_PLUGIN_H
+#define __PCM_PLUGIN_H
+
+/*
+ * Digital Audio (Plugin interface) abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifdef CONFIG_SND_PCM_OSS_PLUGINS
+
+#define snd_pcm_plug_stream(plug) ((plug)->stream)
+
+enum snd_pcm_plugin_action {
+ INIT = 0,
+ PREPARE = 1,
+};
+
+struct snd_pcm_channel_area {
+ void *addr; /* base address of channel samples */
+ unsigned int first; /* offset to first sample in bits */
+ unsigned int step; /* samples distance in bits */
+};
+
+struct snd_pcm_plugin_channel {
+ void *aptr; /* pointer to the allocated area */
+ struct snd_pcm_channel_area area;
+ snd_pcm_uframes_t frames; /* allocated frames */
+ unsigned int enabled:1; /* channel need to be processed */
+ unsigned int wanted:1; /* channel is wanted */
+};
+
+struct snd_pcm_plugin_format {
+ int format;
+ unsigned int rate;
+ unsigned int channels;
+};
+
+struct snd_pcm_plugin {
+ const char *name; /* plug-in name */
+ int stream;
+ struct snd_pcm_plugin_format src_format; /* source format */
+ struct snd_pcm_plugin_format dst_format; /* destination format */
+ int src_width; /* sample width in bits */
+ int dst_width; /* sample width in bits */
+ int access;
+ snd_pcm_sframes_t (*src_frames)(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t dst_frames);
+ snd_pcm_sframes_t (*dst_frames)(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t src_frames);
+ snd_pcm_sframes_t (*client_channels)(struct snd_pcm_plugin *plugin,
+ snd_pcm_uframes_t frames,
+ struct snd_pcm_plugin_channel **channels);
+ snd_pcm_sframes_t (*transfer)(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames);
+ int (*action)(struct snd_pcm_plugin *plugin,
+ enum snd_pcm_plugin_action action,
+ unsigned long data);
+ struct snd_pcm_plugin *prev;
+ struct snd_pcm_plugin *next;
+ struct snd_pcm_substream *plug;
+ void *private_data;
+ void (*private_free)(struct snd_pcm_plugin *plugin);
+ char *buf;
+ snd_pcm_uframes_t buf_frames;
+ struct snd_pcm_plugin_channel *buf_channels;
+ char extra_data[0];
+};
+
+int snd_pcm_plugin_build(struct snd_pcm_substream *handle,
+ const char *name,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ size_t extra,
+ struct snd_pcm_plugin **ret);
+int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin);
+int snd_pcm_plugin_clear(struct snd_pcm_plugin **first);
+int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames);
+snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size);
+snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size);
+
+#define FULL ROUTE_PLUGIN_RESOLUTION
+#define HALF ROUTE_PLUGIN_RESOLUTION / 2
+
+int snd_pcm_plugin_build_io(struct snd_pcm_substream *handle,
+ struct snd_pcm_hw_params *params,
+ struct snd_pcm_plugin **r_plugin);
+int snd_pcm_plugin_build_linear(struct snd_pcm_substream *handle,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin);
+int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *handle,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin);
+int snd_pcm_plugin_build_rate(struct snd_pcm_substream *handle,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin);
+int snd_pcm_plugin_build_route(struct snd_pcm_substream *handle,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin);
+int snd_pcm_plugin_build_copy(struct snd_pcm_substream *handle,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin);
+
+int snd_pcm_plug_format_plugins(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_params *slave_params);
+
+int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask);
+
+int snd_pcm_plugin_append(struct snd_pcm_plugin *plugin);
+
+snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *handle,
+ struct snd_pcm_plugin_channel *src_channels,
+ snd_pcm_uframes_t size);
+snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *handle,
+ struct snd_pcm_plugin_channel *dst_channels_final,
+ snd_pcm_uframes_t size);
+
+snd_pcm_sframes_t snd_pcm_plug_client_channels_buf(struct snd_pcm_substream *handle,
+ char *buf, snd_pcm_uframes_t count,
+ struct snd_pcm_plugin_channel **channels);
+
+snd_pcm_sframes_t snd_pcm_plugin_client_channels(struct snd_pcm_plugin *plugin,
+ snd_pcm_uframes_t frames,
+ struct snd_pcm_plugin_channel **channels);
+
+int snd_pcm_area_silence(const struct snd_pcm_channel_area *dst_channel,
+ size_t dst_offset,
+ size_t samples, int format);
+int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel,
+ size_t src_offset,
+ const struct snd_pcm_channel_area *dst_channel,
+ size_t dst_offset,
+ size_t samples, int format);
+
+void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size);
+void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr);
+snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream,
+ const char *ptr, snd_pcm_uframes_t size,
+ int in_kernel);
+snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream,
+ char *ptr, snd_pcm_uframes_t size, int in_kernel);
+snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream,
+ void **bufs, snd_pcm_uframes_t frames,
+ int in_kernel);
+snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream,
+ void **bufs, snd_pcm_uframes_t frames,
+ int in_kernel);
+
+#else
+
+static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; }
+static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; }
+static inline int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask) { return format; }
+
+#endif
+
+#ifdef PLUGIN_DEBUG
+#define pdprintf( fmt, args... ) printk( "plugin: " fmt, ##args)
+#else
+#define pdprintf( fmt, args... )
+#endif
+
+#endif /* __PCM_PLUGIN_H */
diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
new file mode 100644
index 0000000..2fa9299
--- /dev/null
+++ b/sound/core/oss/rate.c
@@ -0,0 +1,348 @@
+/*
+ * Rate conversion Plug-In
+ * Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "pcm_plugin.h"
+
+#define SHIFT 11
+#define BITS (1<<SHIFT)
+#define R_MASK (BITS-1)
+
+/*
+ * Basic rate conversion plugin
+ */
+
+struct rate_channel {
+ signed short last_S1;
+ signed short last_S2;
+};
+
+typedef void (*rate_f)(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ int src_frames, int dst_frames);
+
+struct rate_priv {
+ unsigned int pitch;
+ unsigned int pos;
+ rate_f func;
+ snd_pcm_sframes_t old_src_frames, old_dst_frames;
+ struct rate_channel channels[0];
+};
+
+static void rate_init(struct snd_pcm_plugin *plugin)
+{
+ unsigned int channel;
+ struct rate_priv *data = (struct rate_priv *)plugin->extra_data;
+ data->pos = 0;
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ data->channels[channel].last_S1 = 0;
+ data->channels[channel].last_S2 = 0;
+ }
+}
+
+static void resample_expand(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ int src_frames, int dst_frames)
+{
+ unsigned int pos = 0;
+ signed int val;
+ signed short S1, S2;
+ signed short *src, *dst;
+ unsigned int channel;
+ int src_step, dst_step;
+ int src_frames1, dst_frames1;
+ struct rate_priv *data = (struct rate_priv *)plugin->extra_data;
+ struct rate_channel *rchannels = data->channels;
+
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ pos = data->pos;
+ S1 = rchannels->last_S1;
+ S2 = rchannels->last_S2;
+ if (!src_channels[channel].enabled) {
+ if (dst_channels[channel].wanted)
+ snd_pcm_area_silence(&dst_channels[channel].area, 0, dst_frames, plugin->dst_format.format);
+ dst_channels[channel].enabled = 0;
+ continue;
+ }
+ dst_channels[channel].enabled = 1;
+ src = (signed short *)src_channels[channel].area.addr +
+ src_channels[channel].area.first / 8 / 2;
+ dst = (signed short *)dst_channels[channel].area.addr +
+ dst_channels[channel].area.first / 8 / 2;
+ src_step = src_channels[channel].area.step / 8 / 2;
+ dst_step = dst_channels[channel].area.step / 8 / 2;
+ src_frames1 = src_frames;
+ dst_frames1 = dst_frames;
+ while (dst_frames1-- > 0) {
+ if (pos & ~R_MASK) {
+ pos &= R_MASK;
+ S1 = S2;
+ if (src_frames1-- > 0) {
+ S2 = *src;
+ src += src_step;
+ }
+ }
+ val = S1 + ((S2 - S1) * (signed int)pos) / BITS;
+ if (val < -32768)
+ val = -32768;
+ else if (val > 32767)
+ val = 32767;
+ *dst = val;
+ dst += dst_step;
+ pos += data->pitch;
+ }
+ rchannels->last_S1 = S1;
+ rchannels->last_S2 = S2;
+ rchannels++;
+ }
+ data->pos = pos;
+}
+
+static void resample_shrink(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ int src_frames, int dst_frames)
+{
+ unsigned int pos = 0;
+ signed int val;
+ signed short S1, S2;
+ signed short *src, *dst;
+ unsigned int channel;
+ int src_step, dst_step;
+ int src_frames1, dst_frames1;
+ struct rate_priv *data = (struct rate_priv *)plugin->extra_data;
+ struct rate_channel *rchannels = data->channels;
+
+ for (channel = 0; channel < plugin->src_format.channels; ++channel) {
+ pos = data->pos;
+ S1 = rchannels->last_S1;
+ S2 = rchannels->last_S2;
+ if (!src_channels[channel].enabled) {
+ if (dst_channels[channel].wanted)
+ snd_pcm_area_silence(&dst_channels[channel].area, 0, dst_frames, plugin->dst_format.format);
+ dst_channels[channel].enabled = 0;
+ continue;
+ }
+ dst_channels[channel].enabled = 1;
+ src = (signed short *)src_channels[channel].area.addr +
+ src_channels[channel].area.first / 8 / 2;
+ dst = (signed short *)dst_channels[channel].area.addr +
+ dst_channels[channel].area.first / 8 / 2;
+ src_step = src_channels[channel].area.step / 8 / 2;
+ dst_step = dst_channels[channel].area.step / 8 / 2;
+ src_frames1 = src_frames;
+ dst_frames1 = dst_frames;
+ while (dst_frames1 > 0) {
+ S1 = S2;
+ if (src_frames1-- > 0) {
+ S2 = *src;
+ src += src_step;
+ }
+ if (pos & ~R_MASK) {
+ pos &= R_MASK;
+ val = S1 + ((S2 - S1) * (signed int)pos) / BITS;
+ if (val < -32768)
+ val = -32768;
+ else if (val > 32767)
+ val = 32767;
+ *dst = val;
+ dst += dst_step;
+ dst_frames1--;
+ }
+ pos += data->pitch;
+ }
+ rchannels->last_S1 = S1;
+ rchannels->last_S2 = S2;
+ rchannels++;
+ }
+ data->pos = pos;
+}
+
+static snd_pcm_sframes_t rate_src_frames(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t frames)
+{
+ struct rate_priv *data;
+ snd_pcm_sframes_t res;
+
+ if (snd_BUG_ON(!plugin))
+ return -ENXIO;
+ if (frames == 0)
+ return 0;
+ data = (struct rate_priv *)plugin->extra_data;
+ if (plugin->src_format.rate < plugin->dst_format.rate) {
+ res = (((frames * data->pitch) + (BITS/2)) >> SHIFT);
+ } else {
+ res = (((frames << SHIFT) + (data->pitch / 2)) / data->pitch);
+ }
+ if (data->old_src_frames > 0) {
+ snd_pcm_sframes_t frames1 = frames, res1 = data->old_dst_frames;
+ while (data->old_src_frames < frames1) {
+ frames1 >>= 1;
+ res1 <<= 1;
+ }
+ while (data->old_src_frames > frames1) {
+ frames1 <<= 1;
+ res1 >>= 1;
+ }
+ if (data->old_src_frames == frames1)
+ return res1;
+ }
+ data->old_src_frames = frames;
+ data->old_dst_frames = res;
+ return res;
+}
+
+static snd_pcm_sframes_t rate_dst_frames(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t frames)
+{
+ struct rate_priv *data;
+ snd_pcm_sframes_t res;
+
+ if (snd_BUG_ON(!plugin))
+ return -ENXIO;
+ if (frames == 0)
+ return 0;
+ data = (struct rate_priv *)plugin->extra_data;
+ if (plugin->src_format.rate < plugin->dst_format.rate) {
+ res = (((frames << SHIFT) + (data->pitch / 2)) / data->pitch);
+ } else {
+ res = (((frames * data->pitch) + (BITS/2)) >> SHIFT);
+ }
+ if (data->old_dst_frames > 0) {
+ snd_pcm_sframes_t frames1 = frames, res1 = data->old_src_frames;
+ while (data->old_dst_frames < frames1) {
+ frames1 >>= 1;
+ res1 <<= 1;
+ }
+ while (data->old_dst_frames > frames1) {
+ frames1 <<= 1;
+ res1 >>= 1;
+ }
+ if (data->old_dst_frames == frames1)
+ return res1;
+ }
+ data->old_dst_frames = frames;
+ data->old_src_frames = res;
+ return res;
+}
+
+static snd_pcm_sframes_t rate_transfer(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ snd_pcm_uframes_t dst_frames;
+ struct rate_priv *data;
+
+ if (snd_BUG_ON(!plugin || !src_channels || !dst_channels))
+ return -ENXIO;
+ if (frames == 0)
+ return 0;
+#ifdef CONFIG_SND_DEBUG
+ {
+ unsigned int channel;
+ for (channel = 0; channel < plugin->src_format.channels; channel++) {
+ if (snd_BUG_ON(src_channels[channel].area.first % 8 ||
+ src_channels[channel].area.step % 8))
+ return -ENXIO;
+ if (snd_BUG_ON(dst_channels[channel].area.first % 8 ||
+ dst_channels[channel].area.step % 8))
+ return -ENXIO;
+ }
+ }
+#endif
+
+ dst_frames = rate_dst_frames(plugin, frames);
+ if (dst_frames > dst_channels[0].frames)
+ dst_frames = dst_channels[0].frames;
+ data = (struct rate_priv *)plugin->extra_data;
+ data->func(plugin, src_channels, dst_channels, frames, dst_frames);
+ return dst_frames;
+}
+
+static int rate_action(struct snd_pcm_plugin *plugin,
+ enum snd_pcm_plugin_action action,
+ unsigned long udata)
+{
+ if (snd_BUG_ON(!plugin))
+ return -ENXIO;
+ switch (action) {
+ case INIT:
+ case PREPARE:
+ rate_init(plugin);
+ break;
+ default:
+ break;
+ }
+ return 0; /* silenty ignore other actions */
+}
+
+int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin)
+{
+ int err;
+ struct rate_priv *data;
+ struct snd_pcm_plugin *plugin;
+
+ if (snd_BUG_ON(!r_plugin))
+ return -ENXIO;
+ *r_plugin = NULL;
+
+ if (snd_BUG_ON(src_format->channels != dst_format->channels))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->channels <= 0))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->format != SNDRV_PCM_FORMAT_S16))
+ return -ENXIO;
+ if (snd_BUG_ON(dst_format->format != SNDRV_PCM_FORMAT_S16))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->rate == dst_format->rate))
+ return -ENXIO;
+
+ err = snd_pcm_plugin_build(plug, "rate conversion",
+ src_format, dst_format,
+ sizeof(struct rate_priv) +
+ src_format->channels * sizeof(struct rate_channel),
+ &plugin);
+ if (err < 0)
+ return err;
+ data = (struct rate_priv *)plugin->extra_data;
+ if (src_format->rate < dst_format->rate) {
+ data->pitch = ((src_format->rate << SHIFT) + (dst_format->rate >> 1)) / dst_format->rate;
+ data->func = resample_expand;
+ } else {
+ data->pitch = ((dst_format->rate << SHIFT) + (src_format->rate >> 1)) / src_format->rate;
+ data->func = resample_shrink;
+ }
+ data->pos = 0;
+ rate_init(plugin);
+ data->old_src_frames = data->old_dst_frames = 0;
+ plugin->transfer = rate_transfer;
+ plugin->src_frames = rate_src_frames;
+ plugin->dst_frames = rate_dst_frames;
+ plugin->action = rate_action;
+ *r_plugin = plugin;
+ return 0;
+}
diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c
new file mode 100644
index 0000000..0dcc287
--- /dev/null
+++ b/sound/core/oss/route.c
@@ -0,0 +1,110 @@
+/*
+ * Route Plug-In
+ * Copyright (c) 2000 by Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "pcm_plugin.h"
+
+static void zero_areas(struct snd_pcm_plugin_channel *dvp, int ndsts,
+ snd_pcm_uframes_t frames, int format)
+{
+ int dst = 0;
+ for (; dst < ndsts; ++dst) {
+ if (dvp->wanted)
+ snd_pcm_area_silence(&dvp->area, 0, frames, format);
+ dvp->enabled = 0;
+ dvp++;
+ }
+}
+
+static inline void copy_area(const struct snd_pcm_plugin_channel *src_channel,
+ struct snd_pcm_plugin_channel *dst_channel,
+ snd_pcm_uframes_t frames, int format)
+{
+ dst_channel->enabled = 1;
+ snd_pcm_area_copy(&src_channel->area, 0, &dst_channel->area, 0, frames, format);
+}
+
+static snd_pcm_sframes_t route_transfer(struct snd_pcm_plugin *plugin,
+ const struct snd_pcm_plugin_channel *src_channels,
+ struct snd_pcm_plugin_channel *dst_channels,
+ snd_pcm_uframes_t frames)
+{
+ int nsrcs, ndsts, dst;
+ struct snd_pcm_plugin_channel *dvp;
+ int format;
+
+ if (snd_BUG_ON(!plugin || !src_channels || !dst_channels))
+ return -ENXIO;
+ if (frames == 0)
+ return 0;
+
+ nsrcs = plugin->src_format.channels;
+ ndsts = plugin->dst_format.channels;
+
+ format = plugin->dst_format.format;
+ dvp = dst_channels;
+ if (nsrcs <= 1) {
+ /* expand to all channels */
+ for (dst = 0; dst < ndsts; ++dst) {
+ copy_area(src_channels, dvp, frames, format);
+ dvp++;
+ }
+ return frames;
+ }
+
+ for (dst = 0; dst < ndsts && dst < nsrcs; ++dst) {
+ copy_area(src_channels, dvp, frames, format);
+ dvp++;
+ src_channels++;
+ }
+ if (dst < ndsts)
+ zero_areas(dvp, ndsts - dst, frames, format);
+ return frames;
+}
+
+int snd_pcm_plugin_build_route(struct snd_pcm_substream *plug,
+ struct snd_pcm_plugin_format *src_format,
+ struct snd_pcm_plugin_format *dst_format,
+ struct snd_pcm_plugin **r_plugin)
+{
+ struct snd_pcm_plugin *plugin;
+ int err;
+
+ if (snd_BUG_ON(!r_plugin))
+ return -ENXIO;
+ *r_plugin = NULL;
+ if (snd_BUG_ON(src_format->rate != dst_format->rate))
+ return -ENXIO;
+ if (snd_BUG_ON(src_format->format != dst_format->format))
+ return -ENXIO;
+
+ err = snd_pcm_plugin_build(plug, "route conversion",
+ src_format, dst_format, 0, &plugin);
+ if (err < 0)
+ return err;
+
+ plugin->transfer = route_transfer;
+ *r_plugin = plugin;
+ return 0;
+}
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
new file mode 100644
index 0000000..192a433
--- /dev/null
+++ b/sound/core/pcm.c
@@ -0,0 +1,1142 @@
+/*
+ * Digital Audio (PCM) abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/info.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Abramo Bagnara <abramo@alsa-project.org>");
+MODULE_DESCRIPTION("Midlevel PCM code for ALSA.");
+MODULE_LICENSE("GPL");
+
+static LIST_HEAD(snd_pcm_devices);
+static LIST_HEAD(snd_pcm_notify_list);
+static DEFINE_MUTEX(register_mutex);
+
+static int snd_pcm_free(struct snd_pcm *pcm);
+static int snd_pcm_dev_free(struct snd_device *device);
+static int snd_pcm_dev_register(struct snd_device *device);
+static int snd_pcm_dev_disconnect(struct snd_device *device);
+
+static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device)
+{
+ struct snd_pcm *pcm;
+
+ list_for_each_entry(pcm, &snd_pcm_devices, list) {
+ if (pcm->card == card && pcm->device == device)
+ return pcm;
+ }
+ return NULL;
+}
+
+static int snd_pcm_next(struct snd_card *card, int device)
+{
+ struct snd_pcm *pcm;
+
+ list_for_each_entry(pcm, &snd_pcm_devices, list) {
+ if (pcm->card == card && pcm->device > device)
+ return pcm->device;
+ else if (pcm->card->number > card->number)
+ return -1;
+ }
+ return -1;
+}
+
+static int snd_pcm_add(struct snd_pcm *newpcm)
+{
+ struct snd_pcm *pcm;
+
+ list_for_each_entry(pcm, &snd_pcm_devices, list) {
+ if (pcm->card == newpcm->card && pcm->device == newpcm->device)
+ return -EBUSY;
+ if (pcm->card->number > newpcm->card->number ||
+ (pcm->card == newpcm->card &&
+ pcm->device > newpcm->device)) {
+ list_add(&newpcm->list, pcm->list.prev);
+ return 0;
+ }
+ }
+ list_add_tail(&newpcm->list, &snd_pcm_devices);
+ return 0;
+}
+
+static int snd_pcm_control_ioctl(struct snd_card *card,
+ struct snd_ctl_file *control,
+ unsigned int cmd, unsigned long arg)
+{
+ switch (cmd) {
+ case SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE:
+ {
+ int device;
+
+ if (get_user(device, (int __user *)arg))
+ return -EFAULT;
+ mutex_lock(&register_mutex);
+ device = snd_pcm_next(card, device);
+ mutex_unlock(&register_mutex);
+ if (put_user(device, (int __user *)arg))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_CTL_IOCTL_PCM_INFO:
+ {
+ struct snd_pcm_info __user *info;
+ unsigned int device, subdevice;
+ int stream;
+ struct snd_pcm *pcm;
+ struct snd_pcm_str *pstr;
+ struct snd_pcm_substream *substream;
+ int err;
+
+ info = (struct snd_pcm_info __user *)arg;
+ if (get_user(device, &info->device))
+ return -EFAULT;
+ if (get_user(stream, &info->stream))
+ return -EFAULT;
+ if (stream < 0 || stream > 1)
+ return -EINVAL;
+ if (get_user(subdevice, &info->subdevice))
+ return -EFAULT;
+ mutex_lock(&register_mutex);
+ pcm = snd_pcm_get(card, device);
+ if (pcm == NULL) {
+ err = -ENXIO;
+ goto _error;
+ }
+ pstr = &pcm->streams[stream];
+ if (pstr->substream_count == 0) {
+ err = -ENOENT;
+ goto _error;
+ }
+ if (subdevice >= pstr->substream_count) {
+ err = -ENXIO;
+ goto _error;
+ }
+ for (substream = pstr->substream; substream;
+ substream = substream->next)
+ if (substream->number == (int)subdevice)
+ break;
+ if (substream == NULL) {
+ err = -ENXIO;
+ goto _error;
+ }
+ err = snd_pcm_info_user(substream, info);
+ _error:
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+ case SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE:
+ {
+ int val;
+
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ control->prefer_pcm_subdevice = val;
+ return 0;
+ }
+ }
+ return -ENOIOCTLCMD;
+}
+
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+
+#define STATE(v) [SNDRV_PCM_STATE_##v] = #v
+#define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v
+#define READY(v) [SNDRV_PCM_READY_##v] = #v
+#define XRUN(v) [SNDRV_PCM_XRUN_##v] = #v
+#define SILENCE(v) [SNDRV_PCM_SILENCE_##v] = #v
+#define TSTAMP(v) [SNDRV_PCM_TSTAMP_##v] = #v
+#define ACCESS(v) [SNDRV_PCM_ACCESS_##v] = #v
+#define START(v) [SNDRV_PCM_START_##v] = #v
+#define FORMAT(v) [SNDRV_PCM_FORMAT_##v] = #v
+#define SUBFORMAT(v) [SNDRV_PCM_SUBFORMAT_##v] = #v
+
+static char *snd_pcm_format_names[] = {
+ FORMAT(S8),
+ FORMAT(U8),
+ FORMAT(S16_LE),
+ FORMAT(S16_BE),
+ FORMAT(U16_LE),
+ FORMAT(U16_BE),
+ FORMAT(S24_LE),
+ FORMAT(S24_BE),
+ FORMAT(U24_LE),
+ FORMAT(U24_BE),
+ FORMAT(S32_LE),
+ FORMAT(S32_BE),
+ FORMAT(U32_LE),
+ FORMAT(U32_BE),
+ FORMAT(FLOAT_LE),
+ FORMAT(FLOAT_BE),
+ FORMAT(FLOAT64_LE),
+ FORMAT(FLOAT64_BE),
+ FORMAT(IEC958_SUBFRAME_LE),
+ FORMAT(IEC958_SUBFRAME_BE),
+ FORMAT(MU_LAW),
+ FORMAT(A_LAW),
+ FORMAT(IMA_ADPCM),
+ FORMAT(MPEG),
+ FORMAT(GSM),
+ FORMAT(SPECIAL),
+ FORMAT(S24_3LE),
+ FORMAT(S24_3BE),
+ FORMAT(U24_3LE),
+ FORMAT(U24_3BE),
+ FORMAT(S20_3LE),
+ FORMAT(S20_3BE),
+ FORMAT(U20_3LE),
+ FORMAT(U20_3BE),
+ FORMAT(S18_3LE),
+ FORMAT(S18_3BE),
+ FORMAT(U18_3LE),
+ FORMAT(U18_3BE),
+};
+
+static const char *snd_pcm_format_name(snd_pcm_format_t format)
+{
+ return snd_pcm_format_names[format];
+}
+
+static char *snd_pcm_stream_names[] = {
+ STREAM(PLAYBACK),
+ STREAM(CAPTURE),
+};
+
+static char *snd_pcm_state_names[] = {
+ STATE(OPEN),
+ STATE(SETUP),
+ STATE(PREPARED),
+ STATE(RUNNING),
+ STATE(XRUN),
+ STATE(DRAINING),
+ STATE(PAUSED),
+ STATE(SUSPENDED),
+};
+
+static char *snd_pcm_access_names[] = {
+ ACCESS(MMAP_INTERLEAVED),
+ ACCESS(MMAP_NONINTERLEAVED),
+ ACCESS(MMAP_COMPLEX),
+ ACCESS(RW_INTERLEAVED),
+ ACCESS(RW_NONINTERLEAVED),
+};
+
+static char *snd_pcm_subformat_names[] = {
+ SUBFORMAT(STD),
+};
+
+static char *snd_pcm_tstamp_mode_names[] = {
+ TSTAMP(NONE),
+ TSTAMP(ENABLE),
+};
+
+static const char *snd_pcm_stream_name(int stream)
+{
+ return snd_pcm_stream_names[stream];
+}
+
+static const char *snd_pcm_access_name(snd_pcm_access_t access)
+{
+ return snd_pcm_access_names[access];
+}
+
+static const char *snd_pcm_subformat_name(snd_pcm_subformat_t subformat)
+{
+ return snd_pcm_subformat_names[subformat];
+}
+
+static const char *snd_pcm_tstamp_mode_name(int mode)
+{
+ return snd_pcm_tstamp_mode_names[mode];
+}
+
+static const char *snd_pcm_state_name(snd_pcm_state_t state)
+{
+ return snd_pcm_state_names[state];
+}
+
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#include <linux/soundcard.h>
+
+static const char *snd_pcm_oss_format_name(int format)
+{
+ switch (format) {
+ case AFMT_MU_LAW:
+ return "MU_LAW";
+ case AFMT_A_LAW:
+ return "A_LAW";
+ case AFMT_IMA_ADPCM:
+ return "IMA_ADPCM";
+ case AFMT_U8:
+ return "U8";
+ case AFMT_S16_LE:
+ return "S16_LE";
+ case AFMT_S16_BE:
+ return "S16_BE";
+ case AFMT_S8:
+ return "S8";
+ case AFMT_U16_LE:
+ return "U16_LE";
+ case AFMT_U16_BE:
+ return "U16_BE";
+ case AFMT_MPEG:
+ return "MPEG";
+ default:
+ return "unknown";
+ }
+}
+#endif
+
+static void snd_pcm_proc_info_read(struct snd_pcm_substream *substream,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_info *info;
+ int err;
+
+ if (! substream)
+ return;
+
+ info = kmalloc(sizeof(*info), GFP_KERNEL);
+ if (! info) {
+ printk(KERN_DEBUG "snd_pcm_proc_info_read: cannot malloc\n");
+ return;
+ }
+
+ err = snd_pcm_info(substream, info);
+ if (err < 0) {
+ snd_iprintf(buffer, "error %d\n", err);
+ kfree(info);
+ return;
+ }
+ snd_iprintf(buffer, "card: %d\n", info->card);
+ snd_iprintf(buffer, "device: %d\n", info->device);
+ snd_iprintf(buffer, "subdevice: %d\n", info->subdevice);
+ snd_iprintf(buffer, "stream: %s\n", snd_pcm_stream_name(info->stream));
+ snd_iprintf(buffer, "id: %s\n", info->id);
+ snd_iprintf(buffer, "name: %s\n", info->name);
+ snd_iprintf(buffer, "subname: %s\n", info->subname);
+ snd_iprintf(buffer, "class: %d\n", info->dev_class);
+ snd_iprintf(buffer, "subclass: %d\n", info->dev_subclass);
+ snd_iprintf(buffer, "subdevices_count: %d\n", info->subdevices_count);
+ snd_iprintf(buffer, "subdevices_avail: %d\n", info->subdevices_avail);
+ kfree(info);
+}
+
+static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_pcm_proc_info_read(((struct snd_pcm_str *)entry->private_data)->substream,
+ buffer);
+}
+
+static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data,
+ buffer);
+}
+
+static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_substream *substream = entry->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (!runtime) {
+ snd_iprintf(buffer, "closed\n");
+ return;
+ }
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ snd_iprintf(buffer, "no setup\n");
+ return;
+ }
+ snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access));
+ snd_iprintf(buffer, "format: %s\n", snd_pcm_format_name(runtime->format));
+ snd_iprintf(buffer, "subformat: %s\n", snd_pcm_subformat_name(runtime->subformat));
+ snd_iprintf(buffer, "channels: %u\n", runtime->channels);
+ snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den);
+ snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size);
+ snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size);
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ if (substream->oss.oss) {
+ snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format));
+ snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels);
+ snd_iprintf(buffer, "OSS rate: %u\n", runtime->oss.rate);
+ snd_iprintf(buffer, "OSS period bytes: %lu\n", (unsigned long)runtime->oss.period_bytes);
+ snd_iprintf(buffer, "OSS periods: %u\n", runtime->oss.periods);
+ snd_iprintf(buffer, "OSS period frames: %lu\n", (unsigned long)runtime->oss.period_frames);
+ }
+#endif
+}
+
+static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_substream *substream = entry->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (!runtime) {
+ snd_iprintf(buffer, "closed\n");
+ return;
+ }
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ snd_iprintf(buffer, "no setup\n");
+ return;
+ }
+ snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode));
+ snd_iprintf(buffer, "period_step: %u\n", runtime->period_step);
+ snd_iprintf(buffer, "avail_min: %lu\n", runtime->control->avail_min);
+ snd_iprintf(buffer, "start_threshold: %lu\n", runtime->start_threshold);
+ snd_iprintf(buffer, "stop_threshold: %lu\n", runtime->stop_threshold);
+ snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold);
+ snd_iprintf(buffer, "silence_size: %lu\n", runtime->silence_size);
+ snd_iprintf(buffer, "boundary: %lu\n", runtime->boundary);
+}
+
+static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_substream *substream = entry->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_status status;
+ int err;
+ if (!runtime) {
+ snd_iprintf(buffer, "closed\n");
+ return;
+ }
+ memset(&status, 0, sizeof(status));
+ err = snd_pcm_status(substream, &status);
+ if (err < 0) {
+ snd_iprintf(buffer, "error %d\n", err);
+ return;
+ }
+ snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state));
+ snd_iprintf(buffer, "trigger_time: %ld.%09ld\n",
+ status.trigger_tstamp.tv_sec, status.trigger_tstamp.tv_nsec);
+ snd_iprintf(buffer, "tstamp : %ld.%09ld\n",
+ status.tstamp.tv_sec, status.tstamp.tv_nsec);
+ snd_iprintf(buffer, "delay : %ld\n", status.delay);
+ snd_iprintf(buffer, "avail : %ld\n", status.avail);
+ snd_iprintf(buffer, "avail_max : %ld\n", status.avail_max);
+ snd_iprintf(buffer, "-----\n");
+ snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr);
+ snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr);
+}
+
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+static void snd_pcm_xrun_debug_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_str *pstr = entry->private_data;
+ snd_iprintf(buffer, "%d\n", pstr->xrun_debug);
+}
+
+static void snd_pcm_xrun_debug_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_str *pstr = entry->private_data;
+ char line[64];
+ if (!snd_info_get_line(buffer, line, sizeof(line)))
+ pstr->xrun_debug = simple_strtoul(line, NULL, 10);
+}
+#endif
+
+static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr)
+{
+ struct snd_pcm *pcm = pstr->pcm;
+ struct snd_info_entry *entry;
+ char name[16];
+
+ sprintf(name, "pcm%i%c", pcm->device,
+ pstr->stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c');
+ if ((entry = snd_info_create_card_entry(pcm->card, name, pcm->card->proc_root)) == NULL)
+ return -ENOMEM;
+ entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ pstr->proc_root = entry;
+
+ if ((entry = snd_info_create_card_entry(pcm->card, "info", pstr->proc_root)) != NULL) {
+ snd_info_set_text_ops(entry, pstr, snd_pcm_stream_proc_info_read);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ pstr->proc_info_entry = entry;
+
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ if ((entry = snd_info_create_card_entry(pcm->card, "xrun_debug",
+ pstr->proc_root)) != NULL) {
+ entry->c.text.read = snd_pcm_xrun_debug_read;
+ entry->c.text.write = snd_pcm_xrun_debug_write;
+ entry->mode |= S_IWUSR;
+ entry->private_data = pstr;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ pstr->proc_xrun_debug_entry = entry;
+#endif
+ return 0;
+}
+
+static int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr)
+{
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ snd_info_free_entry(pstr->proc_xrun_debug_entry);
+ pstr->proc_xrun_debug_entry = NULL;
+#endif
+ snd_info_free_entry(pstr->proc_info_entry);
+ pstr->proc_info_entry = NULL;
+ snd_info_free_entry(pstr->proc_root);
+ pstr->proc_root = NULL;
+ return 0;
+}
+
+static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream)
+{
+ struct snd_info_entry *entry;
+ struct snd_card *card;
+ char name[16];
+
+ card = substream->pcm->card;
+
+ sprintf(name, "sub%i", substream->number);
+ if ((entry = snd_info_create_card_entry(card, name, substream->pstr->proc_root)) == NULL)
+ return -ENOMEM;
+ entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ substream->proc_root = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "info", substream->proc_root)) != NULL) {
+ snd_info_set_text_ops(entry, substream,
+ snd_pcm_substream_proc_info_read);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ substream->proc_info_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "hw_params", substream->proc_root)) != NULL) {
+ snd_info_set_text_ops(entry, substream,
+ snd_pcm_substream_proc_hw_params_read);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ substream->proc_hw_params_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "sw_params", substream->proc_root)) != NULL) {
+ snd_info_set_text_ops(entry, substream,
+ snd_pcm_substream_proc_sw_params_read);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ substream->proc_sw_params_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "status", substream->proc_root)) != NULL) {
+ snd_info_set_text_ops(entry, substream,
+ snd_pcm_substream_proc_status_read);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ substream->proc_status_entry = entry;
+
+ return 0;
+}
+
+static int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream)
+{
+ snd_info_free_entry(substream->proc_info_entry);
+ substream->proc_info_entry = NULL;
+ snd_info_free_entry(substream->proc_hw_params_entry);
+ substream->proc_hw_params_entry = NULL;
+ snd_info_free_entry(substream->proc_sw_params_entry);
+ substream->proc_sw_params_entry = NULL;
+ snd_info_free_entry(substream->proc_status_entry);
+ substream->proc_status_entry = NULL;
+ snd_info_free_entry(substream->proc_root);
+ substream->proc_root = NULL;
+ return 0;
+}
+#else /* !CONFIG_SND_VERBOSE_PROCFS */
+static inline int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) { return 0; }
+static inline int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { return 0; }
+static inline int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) { return 0; }
+static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) { return 0; }
+#endif /* CONFIG_SND_VERBOSE_PROCFS */
+
+/**
+ * snd_pcm_new_stream - create a new PCM stream
+ * @pcm: the pcm instance
+ * @stream: the stream direction, SNDRV_PCM_STREAM_XXX
+ * @substream_count: the number of substreams
+ *
+ * Creates a new stream for the pcm.
+ * The corresponding stream on the pcm must have been empty before
+ * calling this, i.e. zero must be given to the argument of
+ * snd_pcm_new().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
+{
+ int idx, err;
+ struct snd_pcm_str *pstr = &pcm->streams[stream];
+ struct snd_pcm_substream *substream, *prev;
+
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ mutex_init(&pstr->oss.setup_mutex);
+#endif
+ pstr->stream = stream;
+ pstr->pcm = pcm;
+ pstr->substream_count = substream_count;
+ if (substream_count > 0) {
+ err = snd_pcm_stream_proc_init(pstr);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
+ return err;
+ }
+ }
+ prev = NULL;
+ for (idx = 0, prev = NULL; idx < substream_count; idx++) {
+ substream = kzalloc(sizeof(*substream), GFP_KERNEL);
+ if (substream == NULL) {
+ snd_printk(KERN_ERR "Cannot allocate PCM substream\n");
+ return -ENOMEM;
+ }
+ substream->pcm = pcm;
+ substream->pstr = pstr;
+ substream->number = idx;
+ substream->stream = stream;
+ sprintf(substream->name, "subdevice #%i", idx);
+ snprintf(substream->latency_id, sizeof(substream->latency_id),
+ "ALSA-PCM%d-%d%c%d", pcm->card->number, pcm->device,
+ (stream ? 'c' : 'p'), idx);
+ substream->buffer_bytes_max = UINT_MAX;
+ if (prev == NULL)
+ pstr->substream = substream;
+ else
+ prev->next = substream;
+ err = snd_pcm_substream_proc_init(substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
+ if (prev == NULL)
+ pstr->substream = NULL;
+ else
+ prev->next = NULL;
+ kfree(substream);
+ return err;
+ }
+ substream->group = &substream->self_group;
+ spin_lock_init(&substream->self_group.lock);
+ INIT_LIST_HEAD(&substream->self_group.substreams);
+ list_add_tail(&substream->link_list, &substream->self_group.substreams);
+ spin_lock_init(&substream->timer_lock);
+ atomic_set(&substream->mmap_count, 0);
+ prev = substream;
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_new_stream);
+
+/**
+ * snd_pcm_new - create a new PCM instance
+ * @card: the card instance
+ * @id: the id string
+ * @device: the device index (zero based)
+ * @playback_count: the number of substreams for playback
+ * @capture_count: the number of substreams for capture
+ * @rpcm: the pointer to store the new pcm instance
+ *
+ * Creates a new PCM instance.
+ *
+ * The pcm operators have to be set afterwards to the new instance
+ * via snd_pcm_set_ops().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_new(struct snd_card *card, char *id, int device,
+ int playback_count, int capture_count,
+ struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_pcm_dev_free,
+ .dev_register = snd_pcm_dev_register,
+ .dev_disconnect = snd_pcm_dev_disconnect,
+ };
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ if (rpcm)
+ *rpcm = NULL;
+ pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
+ if (pcm == NULL) {
+ snd_printk(KERN_ERR "Cannot allocate PCM\n");
+ return -ENOMEM;
+ }
+ pcm->card = card;
+ pcm->device = device;
+ if (id)
+ strlcpy(pcm->id, id, sizeof(pcm->id));
+ if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) {
+ snd_pcm_free(pcm);
+ return err;
+ }
+ if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count)) < 0) {
+ snd_pcm_free(pcm);
+ return err;
+ }
+ mutex_init(&pcm->open_mutex);
+ init_waitqueue_head(&pcm->open_wait);
+ if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) {
+ snd_pcm_free(pcm);
+ return err;
+ }
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_new);
+
+static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
+{
+ struct snd_pcm_substream *substream, *substream_next;
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ struct snd_pcm_oss_setup *setup, *setupn;
+#endif
+ substream = pstr->substream;
+ while (substream) {
+ substream_next = substream->next;
+ snd_pcm_timer_done(substream);
+ snd_pcm_substream_proc_done(substream);
+ kfree(substream);
+ substream = substream_next;
+ }
+ snd_pcm_stream_proc_done(pstr);
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ for (setup = pstr->oss.setup_list; setup; setup = setupn) {
+ setupn = setup->next;
+ kfree(setup->task_name);
+ kfree(setup);
+ }
+#endif
+}
+
+static int snd_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_notify *notify;
+
+ if (!pcm)
+ return 0;
+ list_for_each_entry(notify, &snd_pcm_notify_list, list) {
+ notify->n_unregister(pcm);
+ }
+ if (pcm->private_free)
+ pcm->private_free(pcm);
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+ snd_pcm_free_stream(&pcm->streams[SNDRV_PCM_STREAM_PLAYBACK]);
+ snd_pcm_free_stream(&pcm->streams[SNDRV_PCM_STREAM_CAPTURE]);
+ kfree(pcm);
+ return 0;
+}
+
+static int snd_pcm_dev_free(struct snd_device *device)
+{
+ struct snd_pcm *pcm = device->device_data;
+ return snd_pcm_free(pcm);
+}
+
+int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
+ struct file *file,
+ struct snd_pcm_substream **rsubstream)
+{
+ struct snd_pcm_str * pstr;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ struct snd_ctl_file *kctl;
+ struct snd_card *card;
+ int prefer_subdevice = -1;
+ size_t size;
+
+ if (snd_BUG_ON(!pcm || !rsubstream))
+ return -ENXIO;
+ *rsubstream = NULL;
+ pstr = &pcm->streams[stream];
+ if (pstr->substream == NULL || pstr->substream_count == 0)
+ return -ENODEV;
+
+ card = pcm->card;
+ read_lock(&card->ctl_files_rwlock);
+ list_for_each_entry(kctl, &card->ctl_files, list) {
+ if (kctl->pid == current->pid) {
+ prefer_subdevice = kctl->prefer_pcm_subdevice;
+ if (prefer_subdevice != -1)
+ break;
+ }
+ }
+ read_unlock(&card->ctl_files_rwlock);
+
+ switch (stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) {
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; substream; substream = substream->next) {
+ if (SUBSTREAM_BUSY(substream))
+ return -EAGAIN;
+ }
+ }
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) {
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) {
+ if (SUBSTREAM_BUSY(substream))
+ return -EAGAIN;
+ }
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (file->f_flags & O_APPEND) {
+ if (prefer_subdevice < 0) {
+ if (pstr->substream_count > 1)
+ return -EINVAL; /* must be unique */
+ substream = pstr->substream;
+ } else {
+ for (substream = pstr->substream; substream;
+ substream = substream->next)
+ if (substream->number == prefer_subdevice)
+ break;
+ }
+ if (! substream)
+ return -ENODEV;
+ if (! SUBSTREAM_BUSY(substream))
+ return -EBADFD;
+ substream->ref_count++;
+ *rsubstream = substream;
+ return 0;
+ }
+
+ if (prefer_subdevice >= 0) {
+ for (substream = pstr->substream; substream; substream = substream->next)
+ if (!SUBSTREAM_BUSY(substream) && substream->number == prefer_subdevice)
+ goto __ok;
+ }
+ for (substream = pstr->substream; substream; substream = substream->next)
+ if (!SUBSTREAM_BUSY(substream))
+ break;
+ __ok:
+ if (substream == NULL)
+ return -EAGAIN;
+
+ runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
+ if (runtime == NULL)
+ return -ENOMEM;
+
+ size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status));
+ runtime->status = snd_malloc_pages(size, GFP_KERNEL);
+ if (runtime->status == NULL) {
+ kfree(runtime);
+ return -ENOMEM;
+ }
+ memset((void*)runtime->status, 0, size);
+
+ size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control));
+ runtime->control = snd_malloc_pages(size, GFP_KERNEL);
+ if (runtime->control == NULL) {
+ snd_free_pages((void*)runtime->status,
+ PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status)));
+ kfree(runtime);
+ return -ENOMEM;
+ }
+ memset((void*)runtime->control, 0, size);
+
+ init_waitqueue_head(&runtime->sleep);
+
+ runtime->status->state = SNDRV_PCM_STATE_OPEN;
+
+ substream->runtime = runtime;
+ substream->private_data = pcm->private_data;
+ substream->ref_count = 1;
+ substream->f_flags = file->f_flags;
+ pstr->substream_opened++;
+ *rsubstream = substream;
+ return 0;
+}
+
+void snd_pcm_detach_substream(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return;
+ runtime = substream->runtime;
+ if (runtime->private_free != NULL)
+ runtime->private_free(runtime);
+ snd_free_pages((void*)runtime->status,
+ PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status)));
+ snd_free_pages((void*)runtime->control,
+ PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)));
+ kfree(runtime->hw_constraints.rules);
+ kfree(runtime);
+ substream->runtime = NULL;
+ substream->pstr->substream_opened--;
+}
+
+static ssize_t show_pcm_class(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct snd_pcm *pcm;
+ const char *str;
+ static const char *strs[SNDRV_PCM_CLASS_LAST + 1] = {
+ [SNDRV_PCM_CLASS_GENERIC] = "generic",
+ [SNDRV_PCM_CLASS_MULTI] = "multi",
+ [SNDRV_PCM_CLASS_MODEM] = "modem",
+ [SNDRV_PCM_CLASS_DIGITIZER] = "digitizer",
+ };
+
+ if (! (pcm = dev_get_drvdata(dev)) ||
+ pcm->dev_class > SNDRV_PCM_CLASS_LAST)
+ str = "none";
+ else
+ str = strs[pcm->dev_class];
+ return snprintf(buf, PAGE_SIZE, "%s\n", str);
+}
+
+static struct device_attribute pcm_attrs =
+ __ATTR(pcm_class, S_IRUGO, show_pcm_class, NULL);
+
+static int snd_pcm_dev_register(struct snd_device *device)
+{
+ int cidx, err;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_notify *notify;
+ char str[16];
+ struct snd_pcm *pcm = device->device_data;
+ struct device *dev;
+
+ if (snd_BUG_ON(!pcm || !device))
+ return -ENXIO;
+ mutex_lock(&register_mutex);
+ err = snd_pcm_add(pcm);
+ if (err) {
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+ for (cidx = 0; cidx < 2; cidx++) {
+ int devtype = -1;
+ if (pcm->streams[cidx].substream == NULL)
+ continue;
+ switch (cidx) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ sprintf(str, "pcmC%iD%ip", pcm->card->number, pcm->device);
+ devtype = SNDRV_DEVICE_TYPE_PCM_PLAYBACK;
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ sprintf(str, "pcmC%iD%ic", pcm->card->number, pcm->device);
+ devtype = SNDRV_DEVICE_TYPE_PCM_CAPTURE;
+ break;
+ }
+ /* device pointer to use, pcm->dev takes precedence if
+ * it is assigned, otherwise fall back to card's device
+ * if possible */
+ dev = pcm->dev;
+ if (!dev)
+ dev = snd_card_get_device_link(pcm->card);
+ /* register pcm */
+ err = snd_register_device_for_dev(devtype, pcm->card,
+ pcm->device,
+ &snd_pcm_f_ops[cidx],
+ pcm, str, dev);
+ if (err < 0) {
+ list_del(&pcm->list);
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+ snd_add_device_sysfs_file(devtype, pcm->card, pcm->device,
+ &pcm_attrs);
+ for (substream = pcm->streams[cidx].substream; substream; substream = substream->next)
+ snd_pcm_timer_init(substream);
+ }
+
+ list_for_each_entry(notify, &snd_pcm_notify_list, list)
+ notify->n_register(pcm);
+
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+static int snd_pcm_dev_disconnect(struct snd_device *device)
+{
+ struct snd_pcm *pcm = device->device_data;
+ struct snd_pcm_notify *notify;
+ struct snd_pcm_substream *substream;
+ int cidx, devtype;
+
+ mutex_lock(&register_mutex);
+ if (list_empty(&pcm->list))
+ goto unlock;
+
+ list_del_init(&pcm->list);
+ for (cidx = 0; cidx < 2; cidx++)
+ for (substream = pcm->streams[cidx].substream; substream; substream = substream->next)
+ if (substream->runtime)
+ substream->runtime->status->state = SNDRV_PCM_STATE_DISCONNECTED;
+ list_for_each_entry(notify, &snd_pcm_notify_list, list) {
+ notify->n_disconnect(pcm);
+ }
+ for (cidx = 0; cidx < 2; cidx++) {
+ devtype = -1;
+ switch (cidx) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ devtype = SNDRV_DEVICE_TYPE_PCM_PLAYBACK;
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ devtype = SNDRV_DEVICE_TYPE_PCM_CAPTURE;
+ break;
+ }
+ snd_unregister_device(devtype, pcm->card, pcm->device);
+ }
+ unlock:
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree)
+{
+ struct snd_pcm *pcm;
+
+ if (snd_BUG_ON(!notify ||
+ !notify->n_register ||
+ !notify->n_unregister ||
+ !notify->n_disconnect))
+ return -EINVAL;
+ mutex_lock(&register_mutex);
+ if (nfree) {
+ list_del(&notify->list);
+ list_for_each_entry(pcm, &snd_pcm_devices, list)
+ notify->n_unregister(pcm);
+ } else {
+ list_add_tail(&notify->list, &snd_pcm_notify_list);
+ list_for_each_entry(pcm, &snd_pcm_devices, list)
+ notify->n_register(pcm);
+ }
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_notify);
+
+#ifdef CONFIG_PROC_FS
+/*
+ * Info interface
+ */
+
+static void snd_pcm_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm *pcm;
+
+ mutex_lock(&register_mutex);
+ list_for_each_entry(pcm, &snd_pcm_devices, list) {
+ snd_iprintf(buffer, "%02i-%02i: %s : %s",
+ pcm->card->number, pcm->device, pcm->id, pcm->name);
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream)
+ snd_iprintf(buffer, " : playback %i",
+ pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream_count);
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream)
+ snd_iprintf(buffer, " : capture %i",
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream_count);
+ snd_iprintf(buffer, "\n");
+ }
+ mutex_unlock(&register_mutex);
+}
+
+static struct snd_info_entry *snd_pcm_proc_entry;
+
+static void snd_pcm_proc_init(void)
+{
+ struct snd_info_entry *entry;
+
+ if ((entry = snd_info_create_module_entry(THIS_MODULE, "pcm", NULL)) != NULL) {
+ snd_info_set_text_ops(entry, NULL, snd_pcm_proc_read);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ snd_pcm_proc_entry = entry;
+}
+
+static void snd_pcm_proc_done(void)
+{
+ snd_info_free_entry(snd_pcm_proc_entry);
+}
+
+#else /* !CONFIG_PROC_FS */
+#define snd_pcm_proc_init()
+#define snd_pcm_proc_done()
+#endif /* CONFIG_PROC_FS */
+
+
+/*
+ * ENTRY functions
+ */
+
+static int __init alsa_pcm_init(void)
+{
+ snd_ctl_register_ioctl(snd_pcm_control_ioctl);
+ snd_ctl_register_ioctl_compat(snd_pcm_control_ioctl);
+ snd_pcm_proc_init();
+ return 0;
+}
+
+static void __exit alsa_pcm_exit(void)
+{
+ snd_ctl_unregister_ioctl(snd_pcm_control_ioctl);
+ snd_ctl_unregister_ioctl_compat(snd_pcm_control_ioctl);
+ snd_pcm_proc_done();
+}
+
+module_init(alsa_pcm_init)
+module_exit(alsa_pcm_exit)
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
new file mode 100644
index 0000000..36d7a59
--- /dev/null
+++ b/sound/core/pcm_compat.c
@@ -0,0 +1,534 @@
+/*
+ * 32bit -> 64bit ioctl wrapper for PCM API
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* This file included from pcm_native.c */
+
+#include <linux/compat.h>
+
+static int snd_pcm_ioctl_delay_compat(struct snd_pcm_substream *substream,
+ s32 __user *src)
+{
+ snd_pcm_sframes_t delay;
+ mm_segment_t fs;
+ int err;
+
+ fs = snd_enter_user();
+ err = snd_pcm_delay(substream, &delay);
+ snd_leave_user(fs);
+ if (err < 0)
+ return err;
+ if (put_user(delay, src))
+ return -EFAULT;
+ return err;
+}
+
+static int snd_pcm_ioctl_rewind_compat(struct snd_pcm_substream *substream,
+ u32 __user *src)
+{
+ snd_pcm_uframes_t frames;
+ int err;
+
+ if (get_user(frames, src))
+ return -EFAULT;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ err = snd_pcm_playback_rewind(substream, frames);
+ else
+ err = snd_pcm_capture_rewind(substream, frames);
+ if (put_user(err, src))
+ return -EFAULT;
+ return err < 0 ? err : 0;
+}
+
+static int snd_pcm_ioctl_forward_compat(struct snd_pcm_substream *substream,
+ u32 __user *src)
+{
+ snd_pcm_uframes_t frames;
+ int err;
+
+ if (get_user(frames, src))
+ return -EFAULT;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ err = snd_pcm_playback_forward(substream, frames);
+ else
+ err = snd_pcm_capture_forward(substream, frames);
+ if (put_user(err, src))
+ return -EFAULT;
+ return err < 0 ? err : 0;
+}
+
+struct snd_pcm_hw_params32 {
+ u32 flags;
+ struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; /* this must be identical */
+ struct snd_mask mres[5]; /* reserved masks */
+ struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1];
+ struct snd_interval ires[9]; /* reserved intervals */
+ u32 rmask;
+ u32 cmask;
+ u32 info;
+ u32 msbits;
+ u32 rate_num;
+ u32 rate_den;
+ u32 fifo_size;
+ unsigned char reserved[64];
+};
+
+struct snd_pcm_sw_params32 {
+ s32 tstamp_mode;
+ u32 period_step;
+ u32 sleep_min;
+ u32 avail_min;
+ u32 xfer_align;
+ u32 start_threshold;
+ u32 stop_threshold;
+ u32 silence_threshold;
+ u32 silence_size;
+ u32 boundary;
+ unsigned char reserved[64];
+};
+
+/* recalcuate the boundary within 32bit */
+static snd_pcm_uframes_t recalculate_boundary(struct snd_pcm_runtime *runtime)
+{
+ snd_pcm_uframes_t boundary;
+
+ if (! runtime->buffer_size)
+ return 0;
+ boundary = runtime->buffer_size;
+ while (boundary * 2 <= 0x7fffffffUL - runtime->buffer_size)
+ boundary *= 2;
+ return boundary;
+}
+
+static int snd_pcm_ioctl_sw_params_compat(struct snd_pcm_substream *substream,
+ struct snd_pcm_sw_params32 __user *src)
+{
+ struct snd_pcm_sw_params params;
+ snd_pcm_uframes_t boundary;
+ int err;
+
+ memset(&params, 0, sizeof(params));
+ if (get_user(params.tstamp_mode, &src->tstamp_mode) ||
+ get_user(params.period_step, &src->period_step) ||
+ get_user(params.sleep_min, &src->sleep_min) ||
+ get_user(params.avail_min, &src->avail_min) ||
+ get_user(params.xfer_align, &src->xfer_align) ||
+ get_user(params.start_threshold, &src->start_threshold) ||
+ get_user(params.stop_threshold, &src->stop_threshold) ||
+ get_user(params.silence_threshold, &src->silence_threshold) ||
+ get_user(params.silence_size, &src->silence_size))
+ return -EFAULT;
+ /*
+ * Check silent_size parameter. Since we have 64bit boundary,
+ * silence_size must be compared with the 32bit boundary.
+ */
+ boundary = recalculate_boundary(substream->runtime);
+ if (boundary && params.silence_size >= boundary)
+ params.silence_size = substream->runtime->boundary;
+ err = snd_pcm_sw_params(substream, &params);
+ if (err < 0)
+ return err;
+ if (boundary && put_user(boundary, &src->boundary))
+ return -EFAULT;
+ return err;
+}
+
+struct snd_pcm_channel_info32 {
+ u32 channel;
+ u32 offset;
+ u32 first;
+ u32 step;
+};
+
+static int snd_pcm_ioctl_channel_info_compat(struct snd_pcm_substream *substream,
+ struct snd_pcm_channel_info32 __user *src)
+{
+ struct snd_pcm_channel_info info;
+ int err;
+
+ if (get_user(info.channel, &src->channel) ||
+ get_user(info.offset, &src->offset) ||
+ get_user(info.first, &src->first) ||
+ get_user(info.step, &src->step))
+ return -EFAULT;
+ err = snd_pcm_channel_info(substream, &info);
+ if (err < 0)
+ return err;
+ if (put_user(info.channel, &src->channel) ||
+ put_user(info.offset, &src->offset) ||
+ put_user(info.first, &src->first) ||
+ put_user(info.step, &src->step))
+ return -EFAULT;
+ return err;
+}
+
+struct snd_pcm_status32 {
+ s32 state;
+ struct compat_timespec trigger_tstamp;
+ struct compat_timespec tstamp;
+ u32 appl_ptr;
+ u32 hw_ptr;
+ s32 delay;
+ u32 avail;
+ u32 avail_max;
+ u32 overrange;
+ s32 suspended_state;
+ unsigned char reserved[60];
+} __attribute__((packed));
+
+
+static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream,
+ struct snd_pcm_status32 __user *src)
+{
+ struct snd_pcm_status status;
+ int err;
+
+ err = snd_pcm_status(substream, &status);
+ if (err < 0)
+ return err;
+
+ if (put_user(status.state, &src->state) ||
+ put_user(status.trigger_tstamp.tv_sec, &src->trigger_tstamp.tv_sec) ||
+ put_user(status.trigger_tstamp.tv_nsec, &src->trigger_tstamp.tv_nsec) ||
+ put_user(status.tstamp.tv_sec, &src->tstamp.tv_sec) ||
+ put_user(status.tstamp.tv_nsec, &src->tstamp.tv_nsec) ||
+ put_user(status.appl_ptr, &src->appl_ptr) ||
+ put_user(status.hw_ptr, &src->hw_ptr) ||
+ put_user(status.delay, &src->delay) ||
+ put_user(status.avail, &src->avail) ||
+ put_user(status.avail_max, &src->avail_max) ||
+ put_user(status.overrange, &src->overrange) ||
+ put_user(status.suspended_state, &src->suspended_state))
+ return -EFAULT;
+
+ return err;
+}
+
+/* both for HW_PARAMS and HW_REFINE */
+static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream,
+ int refine,
+ struct snd_pcm_hw_params32 __user *data32)
+{
+ struct snd_pcm_hw_params *data;
+ struct snd_pcm_runtime *runtime;
+ int err;
+
+ if (! (runtime = substream->runtime))
+ return -ENOTTY;
+
+ data = kmalloc(sizeof(*data), GFP_KERNEL);
+ if (data == NULL)
+ return -ENOMEM;
+ /* only fifo_size is different, so just copy all */
+ if (copy_from_user(data, data32, sizeof(*data32))) {
+ err = -EFAULT;
+ goto error;
+ }
+ if (refine)
+ err = snd_pcm_hw_refine(substream, data);
+ else
+ err = snd_pcm_hw_params(substream, data);
+ if (err < 0)
+ goto error;
+ if (copy_to_user(data32, data, sizeof(*data32)) ||
+ put_user(data->fifo_size, &data32->fifo_size)) {
+ err = -EFAULT;
+ goto error;
+ }
+
+ if (! refine) {
+ unsigned int new_boundary = recalculate_boundary(runtime);
+ if (new_boundary)
+ runtime->boundary = new_boundary;
+ }
+ error:
+ kfree(data);
+ return err;
+}
+
+
+/*
+ */
+struct snd_xferi32 {
+ s32 result;
+ u32 buf;
+ u32 frames;
+};
+
+static int snd_pcm_ioctl_xferi_compat(struct snd_pcm_substream *substream,
+ int dir, struct snd_xferi32 __user *data32)
+{
+ compat_caddr_t buf;
+ u32 frames;
+ int err;
+
+ if (! substream->runtime)
+ return -ENOTTY;
+ if (substream->stream != dir)
+ return -EINVAL;
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+
+ if (get_user(buf, &data32->buf) ||
+ get_user(frames, &data32->frames))
+ return -EFAULT;
+
+ if (dir == SNDRV_PCM_STREAM_PLAYBACK)
+ err = snd_pcm_lib_write(substream, compat_ptr(buf), frames);
+ else
+ err = snd_pcm_lib_read(substream, compat_ptr(buf), frames);
+ if (err < 0)
+ return err;
+ /* copy the result */
+ if (put_user(err, &data32->result))
+ return -EFAULT;
+ return 0;
+}
+
+
+/* snd_xfern needs remapping of bufs */
+struct snd_xfern32 {
+ s32 result;
+ u32 bufs; /* this is void **; */
+ u32 frames;
+};
+
+/*
+ * xfern ioctl nees to copy (up to) 128 pointers on stack.
+ * although we may pass the copied pointers through f_op->ioctl, but the ioctl
+ * handler there expands again the same 128 pointers on stack, so it is better
+ * to handle the function (calling pcm_readv/writev) directly in this handler.
+ */
+static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream,
+ int dir, struct snd_xfern32 __user *data32)
+{
+ compat_caddr_t buf;
+ compat_caddr_t __user *bufptr;
+ u32 frames;
+ void __user **bufs;
+ int err, ch, i;
+
+ if (! substream->runtime)
+ return -ENOTTY;
+ if (substream->stream != dir)
+ return -EINVAL;
+
+ if ((ch = substream->runtime->channels) > 128)
+ return -EINVAL;
+ if (get_user(buf, &data32->bufs) ||
+ get_user(frames, &data32->frames))
+ return -EFAULT;
+ bufptr = compat_ptr(buf);
+ bufs = kmalloc(sizeof(void __user *) * ch, GFP_KERNEL);
+ if (bufs == NULL)
+ return -ENOMEM;
+ for (i = 0; i < ch; i++) {
+ u32 ptr;
+ if (get_user(ptr, bufptr)) {
+ kfree(bufs);
+ return -EFAULT;
+ }
+ bufs[ch] = compat_ptr(ptr);
+ bufptr++;
+ }
+ if (dir == SNDRV_PCM_STREAM_PLAYBACK)
+ err = snd_pcm_lib_writev(substream, bufs, frames);
+ else
+ err = snd_pcm_lib_readv(substream, bufs, frames);
+ if (err >= 0) {
+ if (put_user(err, &data32->result))
+ err = -EFAULT;
+ }
+ kfree(bufs);
+ return err;
+}
+
+
+struct snd_pcm_mmap_status32 {
+ s32 state;
+ s32 pad1;
+ u32 hw_ptr;
+ struct compat_timespec tstamp;
+ s32 suspended_state;
+} __attribute__((packed));
+
+struct snd_pcm_mmap_control32 {
+ u32 appl_ptr;
+ u32 avail_min;
+};
+
+struct snd_pcm_sync_ptr32 {
+ u32 flags;
+ union {
+ struct snd_pcm_mmap_status32 status;
+ unsigned char reserved[64];
+ } s;
+ union {
+ struct snd_pcm_mmap_control32 control;
+ unsigned char reserved[64];
+ } c;
+} __attribute__((packed));
+
+static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream,
+ struct snd_pcm_sync_ptr32 __user *src)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ volatile struct snd_pcm_mmap_status *status;
+ volatile struct snd_pcm_mmap_control *control;
+ u32 sflags;
+ struct snd_pcm_mmap_control scontrol;
+ struct snd_pcm_mmap_status sstatus;
+ snd_pcm_uframes_t boundary;
+ int err;
+
+ if (snd_BUG_ON(!runtime))
+ return -EINVAL;
+
+ if (get_user(sflags, &src->flags) ||
+ get_user(scontrol.appl_ptr, &src->c.control.appl_ptr) ||
+ get_user(scontrol.avail_min, &src->c.control.avail_min))
+ return -EFAULT;
+ if (sflags & SNDRV_PCM_SYNC_PTR_HWSYNC) {
+ err = snd_pcm_hwsync(substream);
+ if (err < 0)
+ return err;
+ }
+ status = runtime->status;
+ control = runtime->control;
+ boundary = recalculate_boundary(runtime);
+ if (! boundary)
+ boundary = 0x7fffffff;
+ snd_pcm_stream_lock_irq(substream);
+ /* FIXME: we should consider the boundary for the sync from app */
+ if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL))
+ control->appl_ptr = scontrol.appl_ptr;
+ else
+ scontrol.appl_ptr = control->appl_ptr % boundary;
+ if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN))
+ control->avail_min = scontrol.avail_min;
+ else
+ scontrol.avail_min = control->avail_min;
+ sstatus.state = status->state;
+ sstatus.hw_ptr = status->hw_ptr % boundary;
+ sstatus.tstamp = status->tstamp;
+ sstatus.suspended_state = status->suspended_state;
+ snd_pcm_stream_unlock_irq(substream);
+ if (put_user(sstatus.state, &src->s.status.state) ||
+ put_user(sstatus.hw_ptr, &src->s.status.hw_ptr) ||
+ put_user(sstatus.tstamp.tv_sec, &src->s.status.tstamp.tv_sec) ||
+ put_user(sstatus.tstamp.tv_nsec, &src->s.status.tstamp.tv_nsec) ||
+ put_user(sstatus.suspended_state, &src->s.status.suspended_state) ||
+ put_user(scontrol.appl_ptr, &src->c.control.appl_ptr) ||
+ put_user(scontrol.avail_min, &src->c.control.avail_min))
+ return -EFAULT;
+
+ return 0;
+}
+
+
+/*
+ */
+enum {
+ SNDRV_PCM_IOCTL_HW_REFINE32 = _IOWR('A', 0x10, struct snd_pcm_hw_params32),
+ SNDRV_PCM_IOCTL_HW_PARAMS32 = _IOWR('A', 0x11, struct snd_pcm_hw_params32),
+ SNDRV_PCM_IOCTL_SW_PARAMS32 = _IOWR('A', 0x13, struct snd_pcm_sw_params32),
+ SNDRV_PCM_IOCTL_STATUS32 = _IOR('A', 0x20, struct snd_pcm_status32),
+ SNDRV_PCM_IOCTL_DELAY32 = _IOR('A', 0x21, s32),
+ SNDRV_PCM_IOCTL_CHANNEL_INFO32 = _IOR('A', 0x32, struct snd_pcm_channel_info32),
+ SNDRV_PCM_IOCTL_REWIND32 = _IOW('A', 0x46, u32),
+ SNDRV_PCM_IOCTL_FORWARD32 = _IOW('A', 0x49, u32),
+ SNDRV_PCM_IOCTL_WRITEI_FRAMES32 = _IOW('A', 0x50, struct snd_xferi32),
+ SNDRV_PCM_IOCTL_READI_FRAMES32 = _IOR('A', 0x51, struct snd_xferi32),
+ SNDRV_PCM_IOCTL_WRITEN_FRAMES32 = _IOW('A', 0x52, struct snd_xfern32),
+ SNDRV_PCM_IOCTL_READN_FRAMES32 = _IOR('A', 0x53, struct snd_xfern32),
+ SNDRV_PCM_IOCTL_SYNC_PTR32 = _IOWR('A', 0x23, struct snd_pcm_sync_ptr32),
+
+};
+
+static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ void __user *argp = compat_ptr(arg);
+
+ pcm_file = file->private_data;
+ if (! pcm_file)
+ return -ENOTTY;
+ substream = pcm_file->substream;
+ if (! substream)
+ return -ENOTTY;
+
+ /*
+ * When PCM is used on 32bit mode, we need to disable
+ * mmap of PCM status/control records because of the size
+ * incompatibility.
+ */
+ pcm_file->no_compat_mmap = 1;
+
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL_PVERSION:
+ case SNDRV_PCM_IOCTL_INFO:
+ case SNDRV_PCM_IOCTL_TSTAMP:
+ case SNDRV_PCM_IOCTL_TTSTAMP:
+ case SNDRV_PCM_IOCTL_HWSYNC:
+ case SNDRV_PCM_IOCTL_PREPARE:
+ case SNDRV_PCM_IOCTL_RESET:
+ case SNDRV_PCM_IOCTL_START:
+ case SNDRV_PCM_IOCTL_DROP:
+ case SNDRV_PCM_IOCTL_DRAIN:
+ case SNDRV_PCM_IOCTL_PAUSE:
+ case SNDRV_PCM_IOCTL_HW_FREE:
+ case SNDRV_PCM_IOCTL_RESUME:
+ case SNDRV_PCM_IOCTL_XRUN:
+ case SNDRV_PCM_IOCTL_LINK:
+ case SNDRV_PCM_IOCTL_UNLINK:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return snd_pcm_playback_ioctl1(file, substream, cmd, argp);
+ else
+ return snd_pcm_capture_ioctl1(file, substream, cmd, argp);
+ case SNDRV_PCM_IOCTL_HW_REFINE32:
+ return snd_pcm_ioctl_hw_params_compat(substream, 1, argp);
+ case SNDRV_PCM_IOCTL_HW_PARAMS32:
+ return snd_pcm_ioctl_hw_params_compat(substream, 0, argp);
+ case SNDRV_PCM_IOCTL_SW_PARAMS32:
+ return snd_pcm_ioctl_sw_params_compat(substream, argp);
+ case SNDRV_PCM_IOCTL_STATUS32:
+ return snd_pcm_status_user_compat(substream, argp);
+ case SNDRV_PCM_IOCTL_SYNC_PTR32:
+ return snd_pcm_ioctl_sync_ptr_compat(substream, argp);
+ case SNDRV_PCM_IOCTL_CHANNEL_INFO32:
+ return snd_pcm_ioctl_channel_info_compat(substream, argp);
+ case SNDRV_PCM_IOCTL_WRITEI_FRAMES32:
+ return snd_pcm_ioctl_xferi_compat(substream, SNDRV_PCM_STREAM_PLAYBACK, argp);
+ case SNDRV_PCM_IOCTL_READI_FRAMES32:
+ return snd_pcm_ioctl_xferi_compat(substream, SNDRV_PCM_STREAM_CAPTURE, argp);
+ case SNDRV_PCM_IOCTL_WRITEN_FRAMES32:
+ return snd_pcm_ioctl_xfern_compat(substream, SNDRV_PCM_STREAM_PLAYBACK, argp);
+ case SNDRV_PCM_IOCTL_READN_FRAMES32:
+ return snd_pcm_ioctl_xfern_compat(substream, SNDRV_PCM_STREAM_CAPTURE, argp);
+ case SNDRV_PCM_IOCTL_DELAY32:
+ return snd_pcm_ioctl_delay_compat(substream, argp);
+ case SNDRV_PCM_IOCTL_REWIND32:
+ return snd_pcm_ioctl_rewind_compat(substream, argp);
+ case SNDRV_PCM_IOCTL_FORWARD32:
+ return snd_pcm_ioctl_forward_compat(substream, argp);
+ }
+
+ return -ENOIOCTLCMD;
+}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
new file mode 100644
index 0000000..9216910
--- /dev/null
+++ b/sound/core/pcm_lib.c
@@ -0,0 +1,1975 @@
+/*
+ * Digital Audio (PCM) abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/timer.h>
+
+/*
+ * fill ring buffer with silence
+ * runtime->silence_start: starting pointer to silence area
+ * runtime->silence_filled: size filled with silence
+ * runtime->silence_threshold: threshold from application
+ * runtime->silence_size: maximal size from application
+ *
+ * when runtime->silence_size >= runtime->boundary - fill processed area with silence immediately
+ */
+void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t frames, ofs, transfer;
+
+ if (runtime->silence_size < runtime->boundary) {
+ snd_pcm_sframes_t noise_dist, n;
+ if (runtime->silence_start != runtime->control->appl_ptr) {
+ n = runtime->control->appl_ptr - runtime->silence_start;
+ if (n < 0)
+ n += runtime->boundary;
+ if ((snd_pcm_uframes_t)n < runtime->silence_filled)
+ runtime->silence_filled -= n;
+ else
+ runtime->silence_filled = 0;
+ runtime->silence_start = runtime->control->appl_ptr;
+ }
+ if (runtime->silence_filled >= runtime->buffer_size)
+ return;
+ noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled;
+ if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold)
+ return;
+ frames = runtime->silence_threshold - noise_dist;
+ if (frames > runtime->silence_size)
+ frames = runtime->silence_size;
+ } else {
+ if (new_hw_ptr == ULONG_MAX) { /* initialization */
+ snd_pcm_sframes_t avail = snd_pcm_playback_hw_avail(runtime);
+ runtime->silence_filled = avail > 0 ? avail : 0;
+ runtime->silence_start = (runtime->status->hw_ptr +
+ runtime->silence_filled) %
+ runtime->boundary;
+ } else {
+ ofs = runtime->status->hw_ptr;
+ frames = new_hw_ptr - ofs;
+ if ((snd_pcm_sframes_t)frames < 0)
+ frames += runtime->boundary;
+ runtime->silence_filled -= frames;
+ if ((snd_pcm_sframes_t)runtime->silence_filled < 0) {
+ runtime->silence_filled = 0;
+ runtime->silence_start = new_hw_ptr;
+ } else {
+ runtime->silence_start = ofs;
+ }
+ }
+ frames = runtime->buffer_size - runtime->silence_filled;
+ }
+ if (snd_BUG_ON(frames > runtime->buffer_size))
+ return;
+ if (frames == 0)
+ return;
+ ofs = runtime->silence_start % runtime->buffer_size;
+ while (frames > 0) {
+ transfer = ofs + frames > runtime->buffer_size ? runtime->buffer_size - ofs : frames;
+ if (runtime->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED ||
+ runtime->access == SNDRV_PCM_ACCESS_MMAP_INTERLEAVED) {
+ if (substream->ops->silence) {
+ int err;
+ err = substream->ops->silence(substream, -1, ofs, transfer);
+ snd_BUG_ON(err < 0);
+ } else {
+ char *hwbuf = runtime->dma_area + frames_to_bytes(runtime, ofs);
+ snd_pcm_format_set_silence(runtime->format, hwbuf, transfer * runtime->channels);
+ }
+ } else {
+ unsigned int c;
+ unsigned int channels = runtime->channels;
+ if (substream->ops->silence) {
+ for (c = 0; c < channels; ++c) {
+ int err;
+ err = substream->ops->silence(substream, c, ofs, transfer);
+ snd_BUG_ON(err < 0);
+ }
+ } else {
+ size_t dma_csize = runtime->dma_bytes / channels;
+ for (c = 0; c < channels; ++c) {
+ char *hwbuf = runtime->dma_area + (c * dma_csize) + samples_to_bytes(runtime, ofs);
+ snd_pcm_format_set_silence(runtime->format, hwbuf, transfer);
+ }
+ }
+ }
+ runtime->silence_filled += transfer;
+ frames -= transfer;
+ ofs = 0;
+ }
+}
+
+static void xrun(struct snd_pcm_substream *substream)
+{
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ if (substream->pstr->xrun_debug) {
+ snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n",
+ substream->pcm->card->number,
+ substream->pcm->device,
+ substream->stream ? 'c' : 'p');
+ if (substream->pstr->xrun_debug > 1)
+ dump_stack();
+ }
+#endif
+}
+
+static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream,
+ struct snd_pcm_runtime *runtime)
+{
+ snd_pcm_uframes_t pos;
+
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
+ pos = substream->ops->pointer(substream);
+ if (pos == SNDRV_PCM_POS_XRUN)
+ return pos; /* XRUN */
+#ifdef CONFIG_SND_DEBUG
+ if (pos >= runtime->buffer_size) {
+ snd_printk(KERN_ERR "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size);
+ }
+#endif
+ pos -= pos % runtime->min_align;
+ return pos;
+}
+
+static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream,
+ struct snd_pcm_runtime *runtime)
+{
+ snd_pcm_uframes_t avail;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ avail = snd_pcm_playback_avail(runtime);
+ else
+ avail = snd_pcm_capture_avail(runtime);
+ if (avail > runtime->avail_max)
+ runtime->avail_max = avail;
+ if (avail >= runtime->stop_threshold) {
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING)
+ snd_pcm_drain_done(substream);
+ else
+ xrun(substream);
+ return -EPIPE;
+ }
+ if (avail >= runtime->control->avail_min)
+ wake_up(&runtime->sleep);
+ return 0;
+}
+
+static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t pos;
+ snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt;
+ snd_pcm_sframes_t delta;
+
+ pos = snd_pcm_update_hw_ptr_pos(substream, runtime);
+ if (pos == SNDRV_PCM_POS_XRUN) {
+ xrun(substream);
+ return -EPIPE;
+ }
+ if (runtime->period_size == runtime->buffer_size)
+ goto __next_buf;
+ new_hw_ptr = runtime->hw_ptr_base + pos;
+ hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;
+
+ delta = hw_ptr_interrupt - new_hw_ptr;
+ if (delta > 0) {
+ if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) {
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ if (runtime->periods > 1 && substream->pstr->xrun_debug) {
+ snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2);
+ if (substream->pstr->xrun_debug > 1)
+ dump_stack();
+ }
+#endif
+ return 0;
+ }
+ __next_buf:
+ runtime->hw_ptr_base += runtime->buffer_size;
+ if (runtime->hw_ptr_base == runtime->boundary)
+ runtime->hw_ptr_base = 0;
+ new_hw_ptr = runtime->hw_ptr_base + pos;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ runtime->silence_size > 0)
+ snd_pcm_playback_silence(substream, new_hw_ptr);
+
+ runtime->status->hw_ptr = new_hw_ptr;
+ runtime->hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size;
+
+ return snd_pcm_update_hw_ptr_post(substream, runtime);
+}
+
+/* CAUTION: call it with irq disabled */
+int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t pos;
+ snd_pcm_uframes_t old_hw_ptr, new_hw_ptr;
+ snd_pcm_sframes_t delta;
+
+ old_hw_ptr = runtime->status->hw_ptr;
+ pos = snd_pcm_update_hw_ptr_pos(substream, runtime);
+ if (pos == SNDRV_PCM_POS_XRUN) {
+ xrun(substream);
+ return -EPIPE;
+ }
+ new_hw_ptr = runtime->hw_ptr_base + pos;
+
+ delta = old_hw_ptr - new_hw_ptr;
+ if (delta > 0) {
+ if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) {
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+ if (runtime->periods > 2 && substream->pstr->xrun_debug) {
+ snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2);
+ if (substream->pstr->xrun_debug > 1)
+ dump_stack();
+ }
+#endif
+ return 0;
+ }
+ runtime->hw_ptr_base += runtime->buffer_size;
+ if (runtime->hw_ptr_base == runtime->boundary)
+ runtime->hw_ptr_base = 0;
+ new_hw_ptr = runtime->hw_ptr_base + pos;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ runtime->silence_size > 0)
+ snd_pcm_playback_silence(substream, new_hw_ptr);
+
+ runtime->status->hw_ptr = new_hw_ptr;
+
+ return snd_pcm_update_hw_ptr_post(substream, runtime);
+}
+
+/**
+ * snd_pcm_set_ops - set the PCM operators
+ * @pcm: the pcm instance
+ * @direction: stream direction, SNDRV_PCM_STREAM_XXX
+ * @ops: the operator table
+ *
+ * Sets the given PCM operators to the pcm instance.
+ */
+void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, struct snd_pcm_ops *ops)
+{
+ struct snd_pcm_str *stream = &pcm->streams[direction];
+ struct snd_pcm_substream *substream;
+
+ for (substream = stream->substream; substream != NULL; substream = substream->next)
+ substream->ops = ops;
+}
+
+EXPORT_SYMBOL(snd_pcm_set_ops);
+
+/**
+ * snd_pcm_sync - set the PCM sync id
+ * @substream: the pcm substream
+ *
+ * Sets the PCM sync identifier for the card.
+ */
+void snd_pcm_set_sync(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->sync.id32[0] = substream->pcm->card->number;
+ runtime->sync.id32[1] = -1;
+ runtime->sync.id32[2] = -1;
+ runtime->sync.id32[3] = -1;
+}
+
+EXPORT_SYMBOL(snd_pcm_set_sync);
+
+/*
+ * Standard ioctl routine
+ */
+
+static inline unsigned int div32(unsigned int a, unsigned int b,
+ unsigned int *r)
+{
+ if (b == 0) {
+ *r = 0;
+ return UINT_MAX;
+ }
+ *r = a % b;
+ return a / b;
+}
+
+static inline unsigned int div_down(unsigned int a, unsigned int b)
+{
+ if (b == 0)
+ return UINT_MAX;
+ return a / b;
+}
+
+static inline unsigned int div_up(unsigned int a, unsigned int b)
+{
+ unsigned int r;
+ unsigned int q;
+ if (b == 0)
+ return UINT_MAX;
+ q = div32(a, b, &r);
+ if (r)
+ ++q;
+ return q;
+}
+
+static inline unsigned int mul(unsigned int a, unsigned int b)
+{
+ if (a == 0)
+ return 0;
+ if (div_down(UINT_MAX, a) < b)
+ return UINT_MAX;
+ return a * b;
+}
+
+static inline unsigned int muldiv32(unsigned int a, unsigned int b,
+ unsigned int c, unsigned int *r)
+{
+ u_int64_t n = (u_int64_t) a * b;
+ if (c == 0) {
+ snd_BUG_ON(!n);
+ *r = 0;
+ return UINT_MAX;
+ }
+ div64_32(&n, c, r);
+ if (n >= UINT_MAX) {
+ *r = 0;
+ return UINT_MAX;
+ }
+ return n;
+}
+
+/**
+ * snd_interval_refine - refine the interval value of configurator
+ * @i: the interval value to refine
+ * @v: the interval value to refer to
+ *
+ * Refines the interval value with the reference value.
+ * The interval is changed to the range satisfying both intervals.
+ * The interval status (min, max, integer, etc.) are evaluated.
+ *
+ * Returns non-zero if the value is changed, zero if not changed.
+ */
+int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v)
+{
+ int changed = 0;
+ if (snd_BUG_ON(snd_interval_empty(i)))
+ return -EINVAL;
+ if (i->min < v->min) {
+ i->min = v->min;
+ i->openmin = v->openmin;
+ changed = 1;
+ } else if (i->min == v->min && !i->openmin && v->openmin) {
+ i->openmin = 1;
+ changed = 1;
+ }
+ if (i->max > v->max) {
+ i->max = v->max;
+ i->openmax = v->openmax;
+ changed = 1;
+ } else if (i->max == v->max && !i->openmax && v->openmax) {
+ i->openmax = 1;
+ changed = 1;
+ }
+ if (!i->integer && v->integer) {
+ i->integer = 1;
+ changed = 1;
+ }
+ if (i->integer) {
+ if (i->openmin) {
+ i->min++;
+ i->openmin = 0;
+ }
+ if (i->openmax) {
+ i->max--;
+ i->openmax = 0;
+ }
+ } else if (!i->openmin && !i->openmax && i->min == i->max)
+ i->integer = 1;
+ if (snd_interval_checkempty(i)) {
+ snd_interval_none(i);
+ return -EINVAL;
+ }
+ return changed;
+}
+
+EXPORT_SYMBOL(snd_interval_refine);
+
+static int snd_interval_refine_first(struct snd_interval *i)
+{
+ if (snd_BUG_ON(snd_interval_empty(i)))
+ return -EINVAL;
+ if (snd_interval_single(i))
+ return 0;
+ i->max = i->min;
+ i->openmax = i->openmin;
+ if (i->openmax)
+ i->max++;
+ return 1;
+}
+
+static int snd_interval_refine_last(struct snd_interval *i)
+{
+ if (snd_BUG_ON(snd_interval_empty(i)))
+ return -EINVAL;
+ if (snd_interval_single(i))
+ return 0;
+ i->min = i->max;
+ i->openmin = i->openmax;
+ if (i->openmin)
+ i->min--;
+ return 1;
+}
+
+void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c)
+{
+ if (a->empty || b->empty) {
+ snd_interval_none(c);
+ return;
+ }
+ c->empty = 0;
+ c->min = mul(a->min, b->min);
+ c->openmin = (a->openmin || b->openmin);
+ c->max = mul(a->max, b->max);
+ c->openmax = (a->openmax || b->openmax);
+ c->integer = (a->integer && b->integer);
+}
+
+/**
+ * snd_interval_div - refine the interval value with division
+ * @a: dividend
+ * @b: divisor
+ * @c: quotient
+ *
+ * c = a / b
+ *
+ * Returns non-zero if the value is changed, zero if not changed.
+ */
+void snd_interval_div(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c)
+{
+ unsigned int r;
+ if (a->empty || b->empty) {
+ snd_interval_none(c);
+ return;
+ }
+ c->empty = 0;
+ c->min = div32(a->min, b->max, &r);
+ c->openmin = (r || a->openmin || b->openmax);
+ if (b->min > 0) {
+ c->max = div32(a->max, b->min, &r);
+ if (r) {
+ c->max++;
+ c->openmax = 1;
+ } else
+ c->openmax = (a->openmax || b->openmin);
+ } else {
+ c->max = UINT_MAX;
+ c->openmax = 0;
+ }
+ c->integer = 0;
+}
+
+/**
+ * snd_interval_muldivk - refine the interval value
+ * @a: dividend 1
+ * @b: dividend 2
+ * @k: divisor (as integer)
+ * @c: result
+ *
+ * c = a * b / k
+ *
+ * Returns non-zero if the value is changed, zero if not changed.
+ */
+void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interval *b,
+ unsigned int k, struct snd_interval *c)
+{
+ unsigned int r;
+ if (a->empty || b->empty) {
+ snd_interval_none(c);
+ return;
+ }
+ c->empty = 0;
+ c->min = muldiv32(a->min, b->min, k, &r);
+ c->openmin = (r || a->openmin || b->openmin);
+ c->max = muldiv32(a->max, b->max, k, &r);
+ if (r) {
+ c->max++;
+ c->openmax = 1;
+ } else
+ c->openmax = (a->openmax || b->openmax);
+ c->integer = 0;
+}
+
+/**
+ * snd_interval_mulkdiv - refine the interval value
+ * @a: dividend 1
+ * @k: dividend 2 (as integer)
+ * @b: divisor
+ * @c: result
+ *
+ * c = a * k / b
+ *
+ * Returns non-zero if the value is changed, zero if not changed.
+ */
+void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
+ const struct snd_interval *b, struct snd_interval *c)
+{
+ unsigned int r;
+ if (a->empty || b->empty) {
+ snd_interval_none(c);
+ return;
+ }
+ c->empty = 0;
+ c->min = muldiv32(a->min, k, b->max, &r);
+ c->openmin = (r || a->openmin || b->openmax);
+ if (b->min > 0) {
+ c->max = muldiv32(a->max, k, b->min, &r);
+ if (r) {
+ c->max++;
+ c->openmax = 1;
+ } else
+ c->openmax = (a->openmax || b->openmin);
+ } else {
+ c->max = UINT_MAX;
+ c->openmax = 0;
+ }
+ c->integer = 0;
+}
+
+/* ---- */
+
+
+/**
+ * snd_interval_ratnum - refine the interval value
+ * @i: interval to refine
+ * @rats_count: number of ratnum_t
+ * @rats: ratnum_t array
+ * @nump: pointer to store the resultant numerator
+ * @denp: pointer to store the resultant denominator
+ *
+ * Returns non-zero if the value is changed, zero if not changed.
+ */
+int snd_interval_ratnum(struct snd_interval *i,
+ unsigned int rats_count, struct snd_ratnum *rats,
+ unsigned int *nump, unsigned int *denp)
+{
+ unsigned int best_num, best_diff, best_den;
+ unsigned int k;
+ struct snd_interval t;
+ int err;
+
+ best_num = best_den = best_diff = 0;
+ for (k = 0; k < rats_count; ++k) {
+ unsigned int num = rats[k].num;
+ unsigned int den;
+ unsigned int q = i->min;
+ int diff;
+ if (q == 0)
+ q = 1;
+ den = div_down(num, q);
+ if (den < rats[k].den_min)
+ continue;
+ if (den > rats[k].den_max)
+ den = rats[k].den_max;
+ else {
+ unsigned int r;
+ r = (den - rats[k].den_min) % rats[k].den_step;
+ if (r != 0)
+ den -= r;
+ }
+ diff = num - q * den;
+ if (best_num == 0 ||
+ diff * best_den < best_diff * den) {
+ best_diff = diff;
+ best_den = den;
+ best_num = num;
+ }
+ }
+ if (best_den == 0) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ t.min = div_down(best_num, best_den);
+ t.openmin = !!(best_num % best_den);
+
+ best_num = best_den = best_diff = 0;
+ for (k = 0; k < rats_count; ++k) {
+ unsigned int num = rats[k].num;
+ unsigned int den;
+ unsigned int q = i->max;
+ int diff;
+ if (q == 0) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ den = div_up(num, q);
+ if (den > rats[k].den_max)
+ continue;
+ if (den < rats[k].den_min)
+ den = rats[k].den_min;
+ else {
+ unsigned int r;
+ r = (den - rats[k].den_min) % rats[k].den_step;
+ if (r != 0)
+ den += rats[k].den_step - r;
+ }
+ diff = q * den - num;
+ if (best_num == 0 ||
+ diff * best_den < best_diff * den) {
+ best_diff = diff;
+ best_den = den;
+ best_num = num;
+ }
+ }
+ if (best_den == 0) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ t.max = div_up(best_num, best_den);
+ t.openmax = !!(best_num % best_den);
+ t.integer = 0;
+ err = snd_interval_refine(i, &t);
+ if (err < 0)
+ return err;
+
+ if (snd_interval_single(i)) {
+ if (nump)
+ *nump = best_num;
+ if (denp)
+ *denp = best_den;
+ }
+ return err;
+}
+
+EXPORT_SYMBOL(snd_interval_ratnum);
+
+/**
+ * snd_interval_ratden - refine the interval value
+ * @i: interval to refine
+ * @rats_count: number of struct ratden
+ * @rats: struct ratden array
+ * @nump: pointer to store the resultant numerator
+ * @denp: pointer to store the resultant denominator
+ *
+ * Returns non-zero if the value is changed, zero if not changed.
+ */
+static int snd_interval_ratden(struct snd_interval *i,
+ unsigned int rats_count, struct snd_ratden *rats,
+ unsigned int *nump, unsigned int *denp)
+{
+ unsigned int best_num, best_diff, best_den;
+ unsigned int k;
+ struct snd_interval t;
+ int err;
+
+ best_num = best_den = best_diff = 0;
+ for (k = 0; k < rats_count; ++k) {
+ unsigned int num;
+ unsigned int den = rats[k].den;
+ unsigned int q = i->min;
+ int diff;
+ num = mul(q, den);
+ if (num > rats[k].num_max)
+ continue;
+ if (num < rats[k].num_min)
+ num = rats[k].num_max;
+ else {
+ unsigned int r;
+ r = (num - rats[k].num_min) % rats[k].num_step;
+ if (r != 0)
+ num += rats[k].num_step - r;
+ }
+ diff = num - q * den;
+ if (best_num == 0 ||
+ diff * best_den < best_diff * den) {
+ best_diff = diff;
+ best_den = den;
+ best_num = num;
+ }
+ }
+ if (best_den == 0) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ t.min = div_down(best_num, best_den);
+ t.openmin = !!(best_num % best_den);
+
+ best_num = best_den = best_diff = 0;
+ for (k = 0; k < rats_count; ++k) {
+ unsigned int num;
+ unsigned int den = rats[k].den;
+ unsigned int q = i->max;
+ int diff;
+ num = mul(q, den);
+ if (num < rats[k].num_min)
+ continue;
+ if (num > rats[k].num_max)
+ num = rats[k].num_max;
+ else {
+ unsigned int r;
+ r = (num - rats[k].num_min) % rats[k].num_step;
+ if (r != 0)
+ num -= r;
+ }
+ diff = q * den - num;
+ if (best_num == 0 ||
+ diff * best_den < best_diff * den) {
+ best_diff = diff;
+ best_den = den;
+ best_num = num;
+ }
+ }
+ if (best_den == 0) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ t.max = div_up(best_num, best_den);
+ t.openmax = !!(best_num % best_den);
+ t.integer = 0;
+ err = snd_interval_refine(i, &t);
+ if (err < 0)
+ return err;
+
+ if (snd_interval_single(i)) {
+ if (nump)
+ *nump = best_num;
+ if (denp)
+ *denp = best_den;
+ }
+ return err;
+}
+
+/**
+ * snd_interval_list - refine the interval value from the list
+ * @i: the interval value to refine
+ * @count: the number of elements in the list
+ * @list: the value list
+ * @mask: the bit-mask to evaluate
+ *
+ * Refines the interval value from the list.
+ * When mask is non-zero, only the elements corresponding to bit 1 are
+ * evaluated.
+ *
+ * Returns non-zero if the value is changed, zero if not changed.
+ */
+int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask)
+{
+ unsigned int k;
+ int changed = 0;
+
+ if (!count) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ for (k = 0; k < count; k++) {
+ if (mask && !(mask & (1 << k)))
+ continue;
+ if (i->min == list[k] && !i->openmin)
+ goto _l1;
+ if (i->min < list[k]) {
+ i->min = list[k];
+ i->openmin = 0;
+ changed = 1;
+ goto _l1;
+ }
+ }
+ i->empty = 1;
+ return -EINVAL;
+ _l1:
+ for (k = count; k-- > 0;) {
+ if (mask && !(mask & (1 << k)))
+ continue;
+ if (i->max == list[k] && !i->openmax)
+ goto _l2;
+ if (i->max > list[k]) {
+ i->max = list[k];
+ i->openmax = 0;
+ changed = 1;
+ goto _l2;
+ }
+ }
+ i->empty = 1;
+ return -EINVAL;
+ _l2:
+ if (snd_interval_checkempty(i)) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ return changed;
+}
+
+EXPORT_SYMBOL(snd_interval_list);
+
+static int snd_interval_step(struct snd_interval *i, unsigned int min, unsigned int step)
+{
+ unsigned int n;
+ int changed = 0;
+ n = (i->min - min) % step;
+ if (n != 0 || i->openmin) {
+ i->min += step - n;
+ changed = 1;
+ }
+ n = (i->max - min) % step;
+ if (n != 0 || i->openmax) {
+ i->max -= n;
+ changed = 1;
+ }
+ if (snd_interval_checkempty(i)) {
+ i->empty = 1;
+ return -EINVAL;
+ }
+ return changed;
+}
+
+/* Info constraints helpers */
+
+/**
+ * snd_pcm_hw_rule_add - add the hw-constraint rule
+ * @runtime: the pcm runtime instance
+ * @cond: condition bits
+ * @var: the variable to evaluate
+ * @func: the evaluation function
+ * @private: the private data pointer passed to function
+ * @dep: the dependent variables
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond,
+ int var,
+ snd_pcm_hw_rule_func_t func, void *private,
+ int dep, ...)
+{
+ struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints;
+ struct snd_pcm_hw_rule *c;
+ unsigned int k;
+ va_list args;
+ va_start(args, dep);
+ if (constrs->rules_num >= constrs->rules_all) {
+ struct snd_pcm_hw_rule *new;
+ unsigned int new_rules = constrs->rules_all + 16;
+ new = kcalloc(new_rules, sizeof(*c), GFP_KERNEL);
+ if (!new)
+ return -ENOMEM;
+ if (constrs->rules) {
+ memcpy(new, constrs->rules,
+ constrs->rules_num * sizeof(*c));
+ kfree(constrs->rules);
+ }
+ constrs->rules = new;
+ constrs->rules_all = new_rules;
+ }
+ c = &constrs->rules[constrs->rules_num];
+ c->cond = cond;
+ c->func = func;
+ c->var = var;
+ c->private = private;
+ k = 0;
+ while (1) {
+ if (snd_BUG_ON(k >= ARRAY_SIZE(c->deps)))
+ return -EINVAL;
+ c->deps[k++] = dep;
+ if (dep < 0)
+ break;
+ dep = va_arg(args, int);
+ }
+ constrs->rules_num++;
+ va_end(args);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_rule_add);
+
+/**
+ * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint
+ * @runtime: PCM runtime instance
+ * @var: hw_params variable to apply the mask
+ * @mask: the bitmap mask
+ *
+ * Apply the constraint of the given bitmap mask to a 32-bit mask parameter.
+ */
+int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
+ u_int32_t mask)
+{
+ struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints;
+ struct snd_mask *maskp = constrs_mask(constrs, var);
+ *maskp->bits &= mask;
+ memset(maskp->bits + 1, 0, (SNDRV_MASK_MAX-32) / 8); /* clear rest */
+ if (*maskp->bits == 0)
+ return -EINVAL;
+ return 0;
+}
+
+/**
+ * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint
+ * @runtime: PCM runtime instance
+ * @var: hw_params variable to apply the mask
+ * @mask: the 64bit bitmap mask
+ *
+ * Apply the constraint of the given bitmap mask to a 64-bit mask parameter.
+ */
+int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
+ u_int64_t mask)
+{
+ struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints;
+ struct snd_mask *maskp = constrs_mask(constrs, var);
+ maskp->bits[0] &= (u_int32_t)mask;
+ maskp->bits[1] &= (u_int32_t)(mask >> 32);
+ memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */
+ if (! maskp->bits[0] && ! maskp->bits[1])
+ return -EINVAL;
+ return 0;
+}
+
+/**
+ * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval
+ * @runtime: PCM runtime instance
+ * @var: hw_params variable to apply the integer constraint
+ *
+ * Apply the constraint of integer to an interval parameter.
+ */
+int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var)
+{
+ struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints;
+ return snd_interval_setinteger(constrs_interval(constrs, var));
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_integer);
+
+/**
+ * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval
+ * @runtime: PCM runtime instance
+ * @var: hw_params variable to apply the range
+ * @min: the minimal value
+ * @max: the maximal value
+ *
+ * Apply the min/max range constraint to an interval parameter.
+ */
+int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var,
+ unsigned int min, unsigned int max)
+{
+ struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints;
+ struct snd_interval t;
+ t.min = min;
+ t.max = max;
+ t.openmin = t.openmax = 0;
+ t.integer = 0;
+ return snd_interval_refine(constrs_interval(constrs, var), &t);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_minmax);
+
+static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pcm_hw_constraint_list *list = rule->private;
+ return snd_interval_list(hw_param_interval(params, rule->var), list->count, list->list, list->mask);
+}
+
+
+/**
+ * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter
+ * @runtime: PCM runtime instance
+ * @cond: condition bits
+ * @var: hw_params variable to apply the list constraint
+ * @l: list
+ *
+ * Apply the list of constraints to an interval parameter.
+ */
+int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ struct snd_pcm_hw_constraint_list *l)
+{
+ return snd_pcm_hw_rule_add(runtime, cond, var,
+ snd_pcm_hw_rule_list, l,
+ var, -1);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_list);
+
+static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pcm_hw_constraint_ratnums *r = rule->private;
+ unsigned int num = 0, den = 0;
+ int err;
+ err = snd_interval_ratnum(hw_param_interval(params, rule->var),
+ r->nrats, r->rats, &num, &den);
+ if (err >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) {
+ params->rate_num = num;
+ params->rate_den = den;
+ }
+ return err;
+}
+
+/**
+ * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter
+ * @runtime: PCM runtime instance
+ * @cond: condition bits
+ * @var: hw_params variable to apply the ratnums constraint
+ * @r: struct snd_ratnums constriants
+ */
+int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ struct snd_pcm_hw_constraint_ratnums *r)
+{
+ return snd_pcm_hw_rule_add(runtime, cond, var,
+ snd_pcm_hw_rule_ratnums, r,
+ var, -1);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums);
+
+static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pcm_hw_constraint_ratdens *r = rule->private;
+ unsigned int num = 0, den = 0;
+ int err = snd_interval_ratden(hw_param_interval(params, rule->var),
+ r->nrats, r->rats, &num, &den);
+ if (err >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) {
+ params->rate_num = num;
+ params->rate_den = den;
+ }
+ return err;
+}
+
+/**
+ * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter
+ * @runtime: PCM runtime instance
+ * @cond: condition bits
+ * @var: hw_params variable to apply the ratdens constraint
+ * @r: struct snd_ratdens constriants
+ */
+int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ struct snd_pcm_hw_constraint_ratdens *r)
+{
+ return snd_pcm_hw_rule_add(runtime, cond, var,
+ snd_pcm_hw_rule_ratdens, r,
+ var, -1);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_ratdens);
+
+static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ unsigned int l = (unsigned long) rule->private;
+ int width = l & 0xffff;
+ unsigned int msbits = l >> 16;
+ struct snd_interval *i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
+ if (snd_interval_single(i) && snd_interval_value(i) == width)
+ params->msbits = msbits;
+ return 0;
+}
+
+/**
+ * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule
+ * @runtime: PCM runtime instance
+ * @cond: condition bits
+ * @width: sample bits width
+ * @msbits: msbits width
+ */
+int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ unsigned int width,
+ unsigned int msbits)
+{
+ unsigned long l = (msbits << 16) | width;
+ return snd_pcm_hw_rule_add(runtime, cond, -1,
+ snd_pcm_hw_rule_msbits,
+ (void*) l,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_msbits);
+
+static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ unsigned long step = (unsigned long) rule->private;
+ return snd_interval_step(hw_param_interval(params, rule->var), 0, step);
+}
+
+/**
+ * snd_pcm_hw_constraint_step - add a hw constraint step rule
+ * @runtime: PCM runtime instance
+ * @cond: condition bits
+ * @var: hw_params variable to apply the step constraint
+ * @step: step size
+ */
+int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var,
+ unsigned long step)
+{
+ return snd_pcm_hw_rule_add(runtime, cond, var,
+ snd_pcm_hw_rule_step, (void *) step,
+ var, -1);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_step);
+
+static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule)
+{
+ static unsigned int pow2_sizes[] = {
+ 1<<0, 1<<1, 1<<2, 1<<3, 1<<4, 1<<5, 1<<6, 1<<7,
+ 1<<8, 1<<9, 1<<10, 1<<11, 1<<12, 1<<13, 1<<14, 1<<15,
+ 1<<16, 1<<17, 1<<18, 1<<19, 1<<20, 1<<21, 1<<22, 1<<23,
+ 1<<24, 1<<25, 1<<26, 1<<27, 1<<28, 1<<29, 1<<30
+ };
+ return snd_interval_list(hw_param_interval(params, rule->var),
+ ARRAY_SIZE(pow2_sizes), pow2_sizes, 0);
+}
+
+/**
+ * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule
+ * @runtime: PCM runtime instance
+ * @cond: condition bits
+ * @var: hw_params variable to apply the power-of-2 constraint
+ */
+int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
+ unsigned int cond,
+ snd_pcm_hw_param_t var)
+{
+ return snd_pcm_hw_rule_add(runtime, cond, var,
+ snd_pcm_hw_rule_pow2, NULL,
+ var, -1);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_constraint_pow2);
+
+static void _snd_pcm_hw_param_any(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ if (hw_is_mask(var)) {
+ snd_mask_any(hw_param_mask(params, var));
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ return;
+ }
+ if (hw_is_interval(var)) {
+ snd_interval_any(hw_param_interval(params, var));
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ return;
+ }
+ snd_BUG();
+}
+
+void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params)
+{
+ unsigned int k;
+ memset(params, 0, sizeof(*params));
+ for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++)
+ _snd_pcm_hw_param_any(params, k);
+ for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++)
+ _snd_pcm_hw_param_any(params, k);
+ params->info = ~0U;
+}
+
+EXPORT_SYMBOL(_snd_pcm_hw_params_any);
+
+/**
+ * snd_pcm_hw_param_value - return @params field @var value
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @dir: pointer to the direction (-1,0,1) or %NULL
+ *
+ * Return the value for field @var if it's fixed in configuration space
+ * defined by @params. Return -%EINVAL otherwise.
+ */
+int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir)
+{
+ if (hw_is_mask(var)) {
+ const struct snd_mask *mask = hw_param_mask_c(params, var);
+ if (!snd_mask_single(mask))
+ return -EINVAL;
+ if (dir)
+ *dir = 0;
+ return snd_mask_value(mask);
+ }
+ if (hw_is_interval(var)) {
+ const struct snd_interval *i = hw_param_interval_c(params, var);
+ if (!snd_interval_single(i))
+ return -EINVAL;
+ if (dir)
+ *dir = i->openmin;
+ return snd_interval_value(i);
+ }
+ return -EINVAL;
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_param_value);
+
+void _snd_pcm_hw_param_setempty(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ if (hw_is_mask(var)) {
+ snd_mask_none(hw_param_mask(params, var));
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ } else if (hw_is_interval(var)) {
+ snd_interval_none(hw_param_interval(params, var));
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ } else {
+ snd_BUG();
+ }
+}
+
+EXPORT_SYMBOL(_snd_pcm_hw_param_setempty);
+
+static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ int changed;
+ if (hw_is_mask(var))
+ changed = snd_mask_refine_first(hw_param_mask(params, var));
+ else if (hw_is_interval(var))
+ changed = snd_interval_refine_first(hw_param_interval(params, var));
+ else
+ return -EINVAL;
+ if (changed) {
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ }
+ return changed;
+}
+
+
+/**
+ * snd_pcm_hw_param_first - refine config space and return minimum value
+ * @pcm: PCM instance
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @dir: pointer to the direction (-1,0,1) or %NULL
+ *
+ * Inside configuration space defined by @params remove from @var all
+ * values > minimum. Reduce configuration space accordingly.
+ * Return the minimum.
+ */
+int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir)
+{
+ int changed = _snd_pcm_hw_param_first(params, var);
+ if (changed < 0)
+ return changed;
+ if (params->rmask) {
+ int err = snd_pcm_hw_refine(pcm, params);
+ if (snd_BUG_ON(err < 0))
+ return err;
+ }
+ return snd_pcm_hw_param_value(params, var, dir);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_param_first);
+
+static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var)
+{
+ int changed;
+ if (hw_is_mask(var))
+ changed = snd_mask_refine_last(hw_param_mask(params, var));
+ else if (hw_is_interval(var))
+ changed = snd_interval_refine_last(hw_param_interval(params, var));
+ else
+ return -EINVAL;
+ if (changed) {
+ params->cmask |= 1 << var;
+ params->rmask |= 1 << var;
+ }
+ return changed;
+}
+
+
+/**
+ * snd_pcm_hw_param_last - refine config space and return maximum value
+ * @pcm: PCM instance
+ * @params: the hw_params instance
+ * @var: parameter to retrieve
+ * @dir: pointer to the direction (-1,0,1) or %NULL
+ *
+ * Inside configuration space defined by @params remove from @var all
+ * values < maximum. Reduce configuration space accordingly.
+ * Return the maximum.
+ */
+int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params,
+ snd_pcm_hw_param_t var, int *dir)
+{
+ int changed = _snd_pcm_hw_param_last(params, var);
+ if (changed < 0)
+ return changed;
+ if (params->rmask) {
+ int err = snd_pcm_hw_refine(pcm, params);
+ if (snd_BUG_ON(err < 0))
+ return err;
+ }
+ return snd_pcm_hw_param_value(params, var, dir);
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_param_last);
+
+/**
+ * snd_pcm_hw_param_choose - choose a configuration defined by @params
+ * @pcm: PCM instance
+ * @params: the hw_params instance
+ *
+ * Choose one configuration from configuration space defined by @params.
+ * The configuration chosen is that obtained fixing in this order:
+ * first access, first format, first subformat, min channels,
+ * min rate, min period time, max buffer size, min tick time
+ */
+int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm,
+ struct snd_pcm_hw_params *params)
+{
+ static int vars[] = {
+ SNDRV_PCM_HW_PARAM_ACCESS,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ SNDRV_PCM_HW_PARAM_SUBFORMAT,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ SNDRV_PCM_HW_PARAM_RATE,
+ SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ SNDRV_PCM_HW_PARAM_TICK_TIME,
+ -1
+ };
+ int err, *v;
+
+ for (v = vars; *v != -1; v++) {
+ if (*v != SNDRV_PCM_HW_PARAM_BUFFER_SIZE)
+ err = snd_pcm_hw_param_first(pcm, params, *v, NULL);
+ else
+ err = snd_pcm_hw_param_last(pcm, params, *v, NULL);
+ if (snd_BUG_ON(err < 0))
+ return err;
+ }
+ return 0;
+}
+
+static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream,
+ void *arg)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long flags;
+ snd_pcm_stream_lock_irqsave(substream, flags);
+ if (snd_pcm_running(substream) &&
+ snd_pcm_update_hw_ptr(substream) >= 0)
+ runtime->status->hw_ptr %= runtime->buffer_size;
+ else
+ runtime->status->hw_ptr = 0;
+ snd_pcm_stream_unlock_irqrestore(substream, flags);
+ return 0;
+}
+
+static int snd_pcm_lib_ioctl_channel_info(struct snd_pcm_substream *substream,
+ void *arg)
+{
+ struct snd_pcm_channel_info *info = arg;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int width;
+ if (!(runtime->info & SNDRV_PCM_INFO_MMAP)) {
+ info->offset = -1;
+ return 0;
+ }
+ width = snd_pcm_format_physical_width(runtime->format);
+ if (width < 0)
+ return width;
+ info->offset = 0;
+ switch (runtime->access) {
+ case SNDRV_PCM_ACCESS_MMAP_INTERLEAVED:
+ case SNDRV_PCM_ACCESS_RW_INTERLEAVED:
+ info->first = info->channel * width;
+ info->step = runtime->channels * width;
+ break;
+ case SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED:
+ case SNDRV_PCM_ACCESS_RW_NONINTERLEAVED:
+ {
+ size_t size = runtime->dma_bytes / runtime->channels;
+ info->first = info->channel * size * 8;
+ info->step = width;
+ break;
+ }
+ default:
+ snd_BUG();
+ break;
+ }
+ return 0;
+}
+
+/**
+ * snd_pcm_lib_ioctl - a generic PCM ioctl callback
+ * @substream: the pcm substream instance
+ * @cmd: ioctl command
+ * @arg: ioctl argument
+ *
+ * Processes the generic ioctl commands for PCM.
+ * Can be passed as the ioctl callback for PCM ops.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL1_INFO:
+ return 0;
+ case SNDRV_PCM_IOCTL1_RESET:
+ return snd_pcm_lib_ioctl_reset(substream, arg);
+ case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
+ return snd_pcm_lib_ioctl_channel_info(substream, arg);
+ }
+ return -ENXIO;
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_ioctl);
+
+/**
+ * snd_pcm_period_elapsed - update the pcm status for the next period
+ * @substream: the pcm substream instance
+ *
+ * This function is called from the interrupt handler when the
+ * PCM has processed the period size. It will update the current
+ * pointer, wake up sleepers, etc.
+ *
+ * Even if more than one periods have elapsed since the last call, you
+ * have to call this only once.
+ */
+void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+ unsigned long flags;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return;
+ runtime = substream->runtime;
+
+ if (runtime->transfer_ack_begin)
+ runtime->transfer_ack_begin(substream);
+
+ snd_pcm_stream_lock_irqsave(substream, flags);
+ if (!snd_pcm_running(substream) ||
+ snd_pcm_update_hw_ptr_interrupt(substream) < 0)
+ goto _end;
+
+ if (substream->timer_running)
+ snd_timer_interrupt(substream->timer, 1);
+ _end:
+ snd_pcm_stream_unlock_irqrestore(substream, flags);
+ if (runtime->transfer_ack_end)
+ runtime->transfer_ack_end(substream);
+ kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
+}
+
+EXPORT_SYMBOL(snd_pcm_period_elapsed);
+
+/*
+ * Wait until avail_min data becomes available
+ * Returns a negative error code if any error occurs during operation.
+ * The available space is stored on availp. When err = 0 and avail = 0
+ * on the capture stream, it indicates the stream is in DRAINING state.
+ */
+static int wait_for_avail_min(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t *availp)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ wait_queue_t wait;
+ int err = 0;
+ snd_pcm_uframes_t avail = 0;
+ long tout;
+
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&runtime->sleep, &wait);
+ for (;;) {
+ if (signal_pending(current)) {
+ err = -ERESTARTSYS;
+ break;
+ }
+ set_current_state(TASK_INTERRUPTIBLE);
+ snd_pcm_stream_unlock_irq(substream);
+ tout = schedule_timeout(msecs_to_jiffies(10000));
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_SUSPENDED:
+ err = -ESTRPIPE;
+ goto _endloop;
+ case SNDRV_PCM_STATE_XRUN:
+ err = -EPIPE;
+ goto _endloop;
+ case SNDRV_PCM_STATE_DRAINING:
+ if (is_playback)
+ err = -EPIPE;
+ else
+ avail = 0; /* indicate draining */
+ goto _endloop;
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_DISCONNECTED:
+ err = -EBADFD;
+ goto _endloop;
+ }
+ if (!tout) {
+ snd_printd("%s write error (DMA or IRQ trouble?)\n",
+ is_playback ? "playback" : "capture");
+ err = -EIO;
+ break;
+ }
+ if (is_playback)
+ avail = snd_pcm_playback_avail(runtime);
+ else
+ avail = snd_pcm_capture_avail(runtime);
+ if (avail >= runtime->control->avail_min)
+ break;
+ }
+ _endloop:
+ remove_wait_queue(&runtime->sleep, &wait);
+ *availp = avail;
+ return err;
+}
+
+static int snd_pcm_lib_write_transfer(struct snd_pcm_substream *substream,
+ unsigned int hwoff,
+ unsigned long data, unsigned int off,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ char __user *buf = (char __user *) data + frames_to_bytes(runtime, off);
+ if (substream->ops->copy) {
+ if ((err = substream->ops->copy(substream, -1, hwoff, buf, frames)) < 0)
+ return err;
+ } else {
+ char *hwbuf = runtime->dma_area + frames_to_bytes(runtime, hwoff);
+ if (copy_from_user(hwbuf, buf, frames_to_bytes(runtime, frames)))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+typedef int (*transfer_f)(struct snd_pcm_substream *substream, unsigned int hwoff,
+ unsigned long data, unsigned int off,
+ snd_pcm_uframes_t size);
+
+static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream,
+ unsigned long data,
+ snd_pcm_uframes_t size,
+ int nonblock,
+ transfer_f transfer)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t xfer = 0;
+ snd_pcm_uframes_t offset = 0;
+ int err = 0;
+
+ if (size == 0)
+ return 0;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_RUNNING:
+ case SNDRV_PCM_STATE_PAUSED:
+ break;
+ case SNDRV_PCM_STATE_XRUN:
+ err = -EPIPE;
+ goto _end_unlock;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ err = -ESTRPIPE;
+ goto _end_unlock;
+ default:
+ err = -EBADFD;
+ goto _end_unlock;
+ }
+
+ while (size > 0) {
+ snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
+ snd_pcm_uframes_t avail;
+ snd_pcm_uframes_t cont;
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_update_hw_ptr(substream);
+ avail = snd_pcm_playback_avail(runtime);
+ if (!avail) {
+ if (nonblock) {
+ err = -EAGAIN;
+ goto _end_unlock;
+ }
+ err = wait_for_avail_min(substream, &avail);
+ if (err < 0)
+ goto _end_unlock;
+ }
+ frames = size > avail ? avail : size;
+ cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size;
+ if (frames > cont)
+ frames = cont;
+ if (snd_BUG_ON(!frames)) {
+ snd_pcm_stream_unlock_irq(substream);
+ return -EINVAL;
+ }
+ appl_ptr = runtime->control->appl_ptr;
+ appl_ofs = appl_ptr % runtime->buffer_size;
+ snd_pcm_stream_unlock_irq(substream);
+ if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0)
+ goto _end;
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_XRUN:
+ err = -EPIPE;
+ goto _end_unlock;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ err = -ESTRPIPE;
+ goto _end_unlock;
+ default:
+ break;
+ }
+ appl_ptr += frames;
+ if (appl_ptr >= runtime->boundary)
+ appl_ptr -= runtime->boundary;
+ runtime->control->appl_ptr = appl_ptr;
+ if (substream->ops->ack)
+ substream->ops->ack(substream);
+
+ offset += frames;
+ size -= frames;
+ xfer += frames;
+ if (runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
+ snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) {
+ err = snd_pcm_start(substream);
+ if (err < 0)
+ goto _end_unlock;
+ }
+ }
+ _end_unlock:
+ snd_pcm_stream_unlock_irq(substream);
+ _end:
+ return xfer > 0 ? (snd_pcm_sframes_t)xfer : err;
+}
+
+/* sanity-check for read/write methods */
+static int pcm_sanity_check(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ if (snd_BUG_ON(!substream->ops->copy && !runtime->dma_area))
+ return -EINVAL;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ return 0;
+}
+
+snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, const void __user *buf, snd_pcm_uframes_t size)
+{
+ struct snd_pcm_runtime *runtime;
+ int nonblock;
+ int err;
+
+ err = pcm_sanity_check(substream);
+ if (err < 0)
+ return err;
+ runtime = substream->runtime;
+ nonblock = !!(substream->f_flags & O_NONBLOCK);
+
+ if (runtime->access != SNDRV_PCM_ACCESS_RW_INTERLEAVED &&
+ runtime->channels > 1)
+ return -EINVAL;
+ return snd_pcm_lib_write1(substream, (unsigned long)buf, size, nonblock,
+ snd_pcm_lib_write_transfer);
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_write);
+
+static int snd_pcm_lib_writev_transfer(struct snd_pcm_substream *substream,
+ unsigned int hwoff,
+ unsigned long data, unsigned int off,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ void __user **bufs = (void __user **)data;
+ int channels = runtime->channels;
+ int c;
+ if (substream->ops->copy) {
+ if (snd_BUG_ON(!substream->ops->silence))
+ return -EINVAL;
+ for (c = 0; c < channels; ++c, ++bufs) {
+ if (*bufs == NULL) {
+ if ((err = substream->ops->silence(substream, c, hwoff, frames)) < 0)
+ return err;
+ } else {
+ char __user *buf = *bufs + samples_to_bytes(runtime, off);
+ if ((err = substream->ops->copy(substream, c, hwoff, buf, frames)) < 0)
+ return err;
+ }
+ }
+ } else {
+ /* default transfer behaviour */
+ size_t dma_csize = runtime->dma_bytes / channels;
+ for (c = 0; c < channels; ++c, ++bufs) {
+ char *hwbuf = runtime->dma_area + (c * dma_csize) + samples_to_bytes(runtime, hwoff);
+ if (*bufs == NULL) {
+ snd_pcm_format_set_silence(runtime->format, hwbuf, frames);
+ } else {
+ char __user *buf = *bufs + samples_to_bytes(runtime, off);
+ if (copy_from_user(hwbuf, buf, samples_to_bytes(runtime, frames)))
+ return -EFAULT;
+ }
+ }
+ }
+ return 0;
+}
+
+snd_pcm_sframes_t snd_pcm_lib_writev(struct snd_pcm_substream *substream,
+ void __user **bufs,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime;
+ int nonblock;
+ int err;
+
+ err = pcm_sanity_check(substream);
+ if (err < 0)
+ return err;
+ runtime = substream->runtime;
+ nonblock = !!(substream->f_flags & O_NONBLOCK);
+
+ if (runtime->access != SNDRV_PCM_ACCESS_RW_NONINTERLEAVED)
+ return -EINVAL;
+ return snd_pcm_lib_write1(substream, (unsigned long)bufs, frames,
+ nonblock, snd_pcm_lib_writev_transfer);
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_writev);
+
+static int snd_pcm_lib_read_transfer(struct snd_pcm_substream *substream,
+ unsigned int hwoff,
+ unsigned long data, unsigned int off,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ char __user *buf = (char __user *) data + frames_to_bytes(runtime, off);
+ if (substream->ops->copy) {
+ if ((err = substream->ops->copy(substream, -1, hwoff, buf, frames)) < 0)
+ return err;
+ } else {
+ char *hwbuf = runtime->dma_area + frames_to_bytes(runtime, hwoff);
+ if (copy_to_user(buf, hwbuf, frames_to_bytes(runtime, frames)))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream,
+ unsigned long data,
+ snd_pcm_uframes_t size,
+ int nonblock,
+ transfer_f transfer)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t xfer = 0;
+ snd_pcm_uframes_t offset = 0;
+ int err = 0;
+
+ if (size == 0)
+ return 0;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_PREPARED:
+ if (size >= runtime->start_threshold) {
+ err = snd_pcm_start(substream);
+ if (err < 0)
+ goto _end_unlock;
+ }
+ break;
+ case SNDRV_PCM_STATE_DRAINING:
+ case SNDRV_PCM_STATE_RUNNING:
+ case SNDRV_PCM_STATE_PAUSED:
+ break;
+ case SNDRV_PCM_STATE_XRUN:
+ err = -EPIPE;
+ goto _end_unlock;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ err = -ESTRPIPE;
+ goto _end_unlock;
+ default:
+ err = -EBADFD;
+ goto _end_unlock;
+ }
+
+ while (size > 0) {
+ snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
+ snd_pcm_uframes_t avail;
+ snd_pcm_uframes_t cont;
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_update_hw_ptr(substream);
+ avail = snd_pcm_capture_avail(runtime);
+ if (!avail) {
+ if (runtime->status->state ==
+ SNDRV_PCM_STATE_DRAINING) {
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
+ goto _end_unlock;
+ }
+ if (nonblock) {
+ err = -EAGAIN;
+ goto _end_unlock;
+ }
+ err = wait_for_avail_min(substream, &avail);
+ if (err < 0)
+ goto _end_unlock;
+ if (!avail)
+ continue; /* draining */
+ }
+ frames = size > avail ? avail : size;
+ cont = runtime->buffer_size - runtime->control->appl_ptr % runtime->buffer_size;
+ if (frames > cont)
+ frames = cont;
+ if (snd_BUG_ON(!frames)) {
+ snd_pcm_stream_unlock_irq(substream);
+ return -EINVAL;
+ }
+ appl_ptr = runtime->control->appl_ptr;
+ appl_ofs = appl_ptr % runtime->buffer_size;
+ snd_pcm_stream_unlock_irq(substream);
+ if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0)
+ goto _end;
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_XRUN:
+ err = -EPIPE;
+ goto _end_unlock;
+ case SNDRV_PCM_STATE_SUSPENDED:
+ err = -ESTRPIPE;
+ goto _end_unlock;
+ default:
+ break;
+ }
+ appl_ptr += frames;
+ if (appl_ptr >= runtime->boundary)
+ appl_ptr -= runtime->boundary;
+ runtime->control->appl_ptr = appl_ptr;
+ if (substream->ops->ack)
+ substream->ops->ack(substream);
+
+ offset += frames;
+ size -= frames;
+ xfer += frames;
+ }
+ _end_unlock:
+ snd_pcm_stream_unlock_irq(substream);
+ _end:
+ return xfer > 0 ? (snd_pcm_sframes_t)xfer : err;
+}
+
+snd_pcm_sframes_t snd_pcm_lib_read(struct snd_pcm_substream *substream, void __user *buf, snd_pcm_uframes_t size)
+{
+ struct snd_pcm_runtime *runtime;
+ int nonblock;
+ int err;
+
+ err = pcm_sanity_check(substream);
+ if (err < 0)
+ return err;
+ runtime = substream->runtime;
+ nonblock = !!(substream->f_flags & O_NONBLOCK);
+ if (runtime->access != SNDRV_PCM_ACCESS_RW_INTERLEAVED)
+ return -EINVAL;
+ return snd_pcm_lib_read1(substream, (unsigned long)buf, size, nonblock, snd_pcm_lib_read_transfer);
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_read);
+
+static int snd_pcm_lib_readv_transfer(struct snd_pcm_substream *substream,
+ unsigned int hwoff,
+ unsigned long data, unsigned int off,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ void __user **bufs = (void __user **)data;
+ int channels = runtime->channels;
+ int c;
+ if (substream->ops->copy) {
+ for (c = 0; c < channels; ++c, ++bufs) {
+ char __user *buf;
+ if (*bufs == NULL)
+ continue;
+ buf = *bufs + samples_to_bytes(runtime, off);
+ if ((err = substream->ops->copy(substream, c, hwoff, buf, frames)) < 0)
+ return err;
+ }
+ } else {
+ snd_pcm_uframes_t dma_csize = runtime->dma_bytes / channels;
+ for (c = 0; c < channels; ++c, ++bufs) {
+ char *hwbuf;
+ char __user *buf;
+ if (*bufs == NULL)
+ continue;
+
+ hwbuf = runtime->dma_area + (c * dma_csize) + samples_to_bytes(runtime, hwoff);
+ buf = *bufs + samples_to_bytes(runtime, off);
+ if (copy_to_user(buf, hwbuf, samples_to_bytes(runtime, frames)))
+ return -EFAULT;
+ }
+ }
+ return 0;
+}
+
+snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream,
+ void __user **bufs,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime;
+ int nonblock;
+ int err;
+
+ err = pcm_sanity_check(substream);
+ if (err < 0)
+ return err;
+ runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+
+ nonblock = !!(substream->f_flags & O_NONBLOCK);
+ if (runtime->access != SNDRV_PCM_ACCESS_RW_NONINTERLEAVED)
+ return -EINVAL;
+ return snd_pcm_lib_read1(substream, (unsigned long)bufs, frames, nonblock, snd_pcm_lib_readv_transfer);
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_readv);
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
new file mode 100644
index 0000000..a6d4280
--- /dev/null
+++ b/sound/core/pcm_memory.c
@@ -0,0 +1,434 @@
+/*
+ * Digital Audio (PCM) abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/time.h>
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/initval.h>
+
+static int preallocate_dma = 1;
+module_param(preallocate_dma, int, 0444);
+MODULE_PARM_DESC(preallocate_dma, "Preallocate DMA memory when the PCM devices are initialized.");
+
+static int maximum_substreams = 4;
+module_param(maximum_substreams, int, 0444);
+MODULE_PARM_DESC(maximum_substreams, "Maximum substreams with preallocated DMA memory.");
+
+static const size_t snd_minimum_buffer = 16384;
+
+
+/*
+ * try to allocate as the large pages as possible.
+ * stores the resultant memory size in *res_size.
+ *
+ * the minimum size is snd_minimum_buffer. it should be power of 2.
+ */
+static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t size)
+{
+ struct snd_dma_buffer *dmab = &substream->dma_buffer;
+ int err;
+
+ /* already reserved? */
+ if (snd_dma_get_reserved_buf(dmab, substream->dma_buf_id) > 0) {
+ if (dmab->bytes >= size)
+ return 0; /* yes */
+ /* no, free the reserved block */
+ snd_dma_free_pages(dmab);
+ dmab->bytes = 0;
+ }
+
+ do {
+ if ((err = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev,
+ size, dmab)) < 0) {
+ if (err != -ENOMEM)
+ return err; /* fatal error */
+ } else
+ return 0;
+ size >>= 1;
+ } while (size >= snd_minimum_buffer);
+ dmab->bytes = 0; /* tell error */
+ return 0;
+}
+
+/*
+ * release the preallocated buffer if not yet done.
+ */
+static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream)
+{
+ if (substream->dma_buffer.area == NULL)
+ return;
+ if (substream->dma_buf_id)
+ snd_dma_reserve_buf(&substream->dma_buffer, substream->dma_buf_id);
+ else
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+}
+
+/**
+ * snd_pcm_lib_preallocate_free - release the preallocated buffer of the specified substream.
+ * @substream: the pcm substream instance
+ *
+ * Releases the pre-allocated buffer of the given substream.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_preallocate_dma_free(substream);
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+ snd_info_free_entry(substream->proc_prealloc_max_entry);
+ substream->proc_prealloc_max_entry = NULL;
+ snd_info_free_entry(substream->proc_prealloc_entry);
+ substream->proc_prealloc_entry = NULL;
+#endif
+ return 0;
+}
+
+/**
+ * snd_pcm_lib_preallocate_free_for_all - release all pre-allocated buffers on the pcm
+ * @pcm: the pcm instance
+ *
+ * Releases all the pre-allocated buffers on the given pcm.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++)
+ for (substream = pcm->streams[stream].substream; substream; substream = substream->next)
+ snd_pcm_lib_preallocate_free(substream);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_preallocate_free_for_all);
+
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+/*
+ * read callback for prealloc proc file
+ *
+ * prints the current allocated size in kB.
+ */
+static void snd_pcm_lib_preallocate_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_substream *substream = entry->private_data;
+ snd_iprintf(buffer, "%lu\n", (unsigned long) substream->dma_buffer.bytes / 1024);
+}
+
+/*
+ * read callback for prealloc_max proc file
+ *
+ * prints the maximum allowed size in kB.
+ */
+static void snd_pcm_lib_preallocate_max_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_substream *substream = entry->private_data;
+ snd_iprintf(buffer, "%lu\n", (unsigned long) substream->dma_max / 1024);
+}
+
+/*
+ * write callback for prealloc proc file
+ *
+ * accepts the preallocation size in kB.
+ */
+static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pcm_substream *substream = entry->private_data;
+ char line[64], str[64];
+ size_t size;
+ struct snd_dma_buffer new_dmab;
+
+ if (substream->runtime) {
+ buffer->error = -EBUSY;
+ return;
+ }
+ if (!snd_info_get_line(buffer, line, sizeof(line))) {
+ snd_info_get_str(str, line, sizeof(str));
+ size = simple_strtoul(str, NULL, 10) * 1024;
+ if ((size != 0 && size < 8192) || size > substream->dma_max) {
+ buffer->error = -EINVAL;
+ return;
+ }
+ if (substream->dma_buffer.bytes == size)
+ return;
+ memset(&new_dmab, 0, sizeof(new_dmab));
+ new_dmab.dev = substream->dma_buffer.dev;
+ if (size > 0) {
+ if (snd_dma_alloc_pages(substream->dma_buffer.dev.type,
+ substream->dma_buffer.dev.dev,
+ size, &new_dmab) < 0) {
+ buffer->error = -ENOMEM;
+ return;
+ }
+ substream->buffer_bytes_max = size;
+ } else {
+ substream->buffer_bytes_max = UINT_MAX;
+ }
+ if (substream->dma_buffer.area)
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer = new_dmab;
+ } else {
+ buffer->error = -EINVAL;
+ }
+}
+
+static inline void preallocate_info_init(struct snd_pcm_substream *substream)
+{
+ struct snd_info_entry *entry;
+
+ if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc", substream->proc_root)) != NULL) {
+ entry->c.text.read = snd_pcm_lib_preallocate_proc_read;
+ entry->c.text.write = snd_pcm_lib_preallocate_proc_write;
+ entry->mode |= S_IWUSR;
+ entry->private_data = substream;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ substream->proc_prealloc_entry = entry;
+ if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc_max", substream->proc_root)) != NULL) {
+ entry->c.text.read = snd_pcm_lib_preallocate_max_proc_read;
+ entry->private_data = substream;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ substream->proc_prealloc_max_entry = entry;
+}
+
+#else /* !CONFIG_SND_VERBOSE_PROCFS */
+#define preallocate_info_init(s)
+#endif /* CONFIG_SND_VERBOSE_PROCFS */
+
+/*
+ * pre-allocate the buffer and create a proc file for the substream
+ */
+static int snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream,
+ size_t size, size_t max)
+{
+
+ if (size > 0 && preallocate_dma && substream->number < maximum_substreams)
+ preallocate_pcm_pages(substream, size);
+
+ if (substream->dma_buffer.bytes > 0)
+ substream->buffer_bytes_max = substream->dma_buffer.bytes;
+ substream->dma_max = max;
+ preallocate_info_init(substream);
+ return 0;
+}
+
+
+/**
+ * snd_pcm_lib_preallocate_pages - pre-allocation for the given DMA type
+ * @substream: the pcm substream instance
+ * @type: DMA type (SNDRV_DMA_TYPE_*)
+ * @data: DMA type dependant data
+ * @size: the requested pre-allocation size in bytes
+ * @max: the max. allowed pre-allocation size
+ *
+ * Do pre-allocation for the given DMA buffer type.
+ *
+ * When substream->dma_buf_id is set, the function tries to look for
+ * the reserved buffer, and the buffer is not freed but reserved at
+ * destruction time. The dma_buf_id must be unique for all systems
+ * (in the same DMA buffer type) e.g. using snd_dma_pci_buf_id().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream,
+ int type, struct device *data,
+ size_t size, size_t max)
+{
+ substream->dma_buffer.dev.type = type;
+ substream->dma_buffer.dev.dev = data;
+ return snd_pcm_lib_preallocate_pages1(substream, size, max);
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages);
+
+/**
+ * snd_pcm_lib_preallocate_pages_for_all - pre-allocation for continous memory type (all substreams)
+ * @pcm: the pcm instance
+ * @type: DMA type (SNDRV_DMA_TYPE_*)
+ * @data: DMA type dependant data
+ * @size: the requested pre-allocation size in bytes
+ * @max: the max. allowed pre-allocation size
+ *
+ * Do pre-allocation to all substreams of the given pcm for the
+ * specified DMA type.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
+ int type, void *data,
+ size_t size, size_t max)
+{
+ struct snd_pcm_substream *substream;
+ int stream, err;
+
+ for (stream = 0; stream < 2; stream++)
+ for (substream = pcm->streams[stream].substream; substream; substream = substream->next)
+ if ((err = snd_pcm_lib_preallocate_pages(substream, type, data, size, max)) < 0)
+ return err;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all);
+
+/**
+ * snd_pcm_sgbuf_ops_page - get the page struct at the given offset
+ * @substream: the pcm substream instance
+ * @offset: the buffer offset
+ *
+ * Returns the page struct at the given buffer offset.
+ * Used as the page callback of PCM ops.
+ */
+struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned long offset)
+{
+ struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream);
+
+ unsigned int idx = offset >> PAGE_SHIFT;
+ if (idx >= (unsigned int)sgbuf->pages)
+ return NULL;
+ return sgbuf->page_table[idx];
+}
+
+EXPORT_SYMBOL(snd_pcm_sgbuf_ops_page);
+
+/*
+ * compute the max chunk size with continuous pages on sg-buffer
+ */
+unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
+ unsigned int ofs, unsigned int size)
+{
+ struct snd_sg_buf *sg = snd_pcm_substream_sgbuf(substream);
+ unsigned int start, end, pg;
+
+ start = ofs >> PAGE_SHIFT;
+ end = (ofs + size - 1) >> PAGE_SHIFT;
+ /* check page continuity */
+ pg = sg->table[start].addr >> PAGE_SHIFT;
+ for (;;) {
+ start++;
+ if (start > end)
+ break;
+ pg++;
+ if ((sg->table[start].addr >> PAGE_SHIFT) != pg)
+ return (start << PAGE_SHIFT) - ofs;
+ }
+ /* ok, all on continuous pages */
+ return size;
+}
+EXPORT_SYMBOL(snd_pcm_sgbuf_get_chunk_size);
+
+/**
+ * snd_pcm_lib_malloc_pages - allocate the DMA buffer
+ * @substream: the substream to allocate the DMA buffer to
+ * @size: the requested buffer size in bytes
+ *
+ * Allocates the DMA buffer on the BUS type given earlier to
+ * snd_pcm_lib_preallocate_xxx_pages().
+ *
+ * Returns 1 if the buffer is changed, 0 if not changed, or a negative
+ * code on failure.
+ */
+int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size)
+{
+ struct snd_pcm_runtime *runtime;
+ struct snd_dma_buffer *dmab = NULL;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return -EINVAL;
+ if (snd_BUG_ON(substream->dma_buffer.dev.type ==
+ SNDRV_DMA_TYPE_UNKNOWN))
+ return -EINVAL;
+ runtime = substream->runtime;
+
+ if (runtime->dma_buffer_p) {
+ /* perphaps, we might free the large DMA memory region
+ to save some space here, but the actual solution
+ costs us less time */
+ if (runtime->dma_buffer_p->bytes >= size) {
+ runtime->dma_bytes = size;
+ return 0; /* ok, do not change */
+ }
+ snd_pcm_lib_free_pages(substream);
+ }
+ if (substream->dma_buffer.area != NULL &&
+ substream->dma_buffer.bytes >= size) {
+ dmab = &substream->dma_buffer; /* use the pre-allocated buffer */
+ } else {
+ dmab = kzalloc(sizeof(*dmab), GFP_KERNEL);
+ if (! dmab)
+ return -ENOMEM;
+ dmab->dev = substream->dma_buffer.dev;
+ if (snd_dma_alloc_pages(substream->dma_buffer.dev.type,
+ substream->dma_buffer.dev.dev,
+ size, dmab) < 0) {
+ kfree(dmab);
+ return -ENOMEM;
+ }
+ }
+ snd_pcm_set_runtime_buffer(substream, dmab);
+ runtime->dma_bytes = size;
+ return 1; /* area was changed */
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_malloc_pages);
+
+/**
+ * snd_pcm_lib_free_pages - release the allocated DMA buffer.
+ * @substream: the substream to release the DMA buffer
+ *
+ * Releases the DMA buffer allocated via snd_pcm_lib_malloc_pages().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return -EINVAL;
+ runtime = substream->runtime;
+ if (runtime->dma_area == NULL)
+ return 0;
+ if (runtime->dma_buffer_p != &substream->dma_buffer) {
+ /* it's a newly allocated buffer. release it now. */
+ snd_dma_free_pages(runtime->dma_buffer_p);
+ kfree(runtime->dma_buffer_p);
+ }
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_free_pages);
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
new file mode 100644
index 0000000..ea2bf82
--- /dev/null
+++ b/sound/core/pcm_misc.c
@@ -0,0 +1,470 @@
+/*
+ * PCM Interface - misc routines
+ * Copyright (c) 1998 by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This library is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU Library General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#define SND_PCM_FORMAT_UNKNOWN (-1)
+
+/* NOTE: "signed" prefix must be given below since the default char is
+ * unsigned on some architectures!
+ */
+struct pcm_format_data {
+ unsigned char width; /* bit width */
+ unsigned char phys; /* physical bit width */
+ signed char le; /* 0 = big-endian, 1 = little-endian, -1 = others */
+ signed char signd; /* 0 = unsigned, 1 = signed, -1 = others */
+ unsigned char silence[8]; /* silence data to fill */
+};
+
+static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = {
+ [SNDRV_PCM_FORMAT_S8] = {
+ .width = 8, .phys = 8, .le = -1, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_U8] = {
+ .width = 8, .phys = 8, .le = -1, .signd = 0,
+ .silence = { 0x80 },
+ },
+ [SNDRV_PCM_FORMAT_S16_LE] = {
+ .width = 16, .phys = 16, .le = 1, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_S16_BE] = {
+ .width = 16, .phys = 16, .le = 0, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_U16_LE] = {
+ .width = 16, .phys = 16, .le = 1, .signd = 0,
+ .silence = { 0x00, 0x80 },
+ },
+ [SNDRV_PCM_FORMAT_U16_BE] = {
+ .width = 16, .phys = 16, .le = 0, .signd = 0,
+ .silence = { 0x80, 0x00 },
+ },
+ [SNDRV_PCM_FORMAT_S24_LE] = {
+ .width = 24, .phys = 32, .le = 1, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_S24_BE] = {
+ .width = 24, .phys = 32, .le = 0, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_U24_LE] = {
+ .width = 24, .phys = 32, .le = 1, .signd = 0,
+ .silence = { 0x00, 0x00, 0x80 },
+ },
+ [SNDRV_PCM_FORMAT_U24_BE] = {
+ .width = 24, .phys = 32, .le = 0, .signd = 0,
+ .silence = { 0x00, 0x80, 0x00, 0x00 },
+ },
+ [SNDRV_PCM_FORMAT_S32_LE] = {
+ .width = 32, .phys = 32, .le = 1, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_S32_BE] = {
+ .width = 32, .phys = 32, .le = 0, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_U32_LE] = {
+ .width = 32, .phys = 32, .le = 1, .signd = 0,
+ .silence = { 0x00, 0x00, 0x00, 0x80 },
+ },
+ [SNDRV_PCM_FORMAT_U32_BE] = {
+ .width = 32, .phys = 32, .le = 0, .signd = 0,
+ .silence = { 0x80, 0x00, 0x00, 0x00 },
+ },
+ [SNDRV_PCM_FORMAT_FLOAT_LE] = {
+ .width = 32, .phys = 32, .le = 1, .signd = -1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_FLOAT_BE] = {
+ .width = 32, .phys = 32, .le = 0, .signd = -1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_FLOAT64_LE] = {
+ .width = 64, .phys = 64, .le = 1, .signd = -1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_FLOAT64_BE] = {
+ .width = 64, .phys = 64, .le = 0, .signd = -1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE] = {
+ .width = 32, .phys = 32, .le = 1, .signd = -1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE] = {
+ .width = 32, .phys = 32, .le = 0, .signd = -1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_MU_LAW] = {
+ .width = 8, .phys = 8, .le = -1, .signd = -1,
+ .silence = { 0x7f },
+ },
+ [SNDRV_PCM_FORMAT_A_LAW] = {
+ .width = 8, .phys = 8, .le = -1, .signd = -1,
+ .silence = { 0x55 },
+ },
+ [SNDRV_PCM_FORMAT_IMA_ADPCM] = {
+ .width = 4, .phys = 4, .le = -1, .signd = -1,
+ .silence = {},
+ },
+ /* FIXME: the following three formats are not defined properly yet */
+ [SNDRV_PCM_FORMAT_MPEG] = {
+ .le = -1, .signd = -1,
+ },
+ [SNDRV_PCM_FORMAT_GSM] = {
+ .le = -1, .signd = -1,
+ },
+ [SNDRV_PCM_FORMAT_SPECIAL] = {
+ .le = -1, .signd = -1,
+ },
+ [SNDRV_PCM_FORMAT_S24_3LE] = {
+ .width = 24, .phys = 24, .le = 1, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_S24_3BE] = {
+ .width = 24, .phys = 24, .le = 0, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_U24_3LE] = {
+ .width = 24, .phys = 24, .le = 1, .signd = 0,
+ .silence = { 0x00, 0x00, 0x80 },
+ },
+ [SNDRV_PCM_FORMAT_U24_3BE] = {
+ .width = 24, .phys = 24, .le = 0, .signd = 0,
+ .silence = { 0x80, 0x00, 0x00 },
+ },
+ [SNDRV_PCM_FORMAT_S20_3LE] = {
+ .width = 20, .phys = 24, .le = 1, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_S20_3BE] = {
+ .width = 20, .phys = 24, .le = 0, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_U20_3LE] = {
+ .width = 20, .phys = 24, .le = 1, .signd = 0,
+ .silence = { 0x00, 0x00, 0x08 },
+ },
+ [SNDRV_PCM_FORMAT_U20_3BE] = {
+ .width = 20, .phys = 24, .le = 0, .signd = 0,
+ .silence = { 0x08, 0x00, 0x00 },
+ },
+ [SNDRV_PCM_FORMAT_S18_3LE] = {
+ .width = 18, .phys = 24, .le = 1, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_S18_3BE] = {
+ .width = 18, .phys = 24, .le = 0, .signd = 1,
+ .silence = {},
+ },
+ [SNDRV_PCM_FORMAT_U18_3LE] = {
+ .width = 18, .phys = 24, .le = 1, .signd = 0,
+ .silence = { 0x00, 0x00, 0x02 },
+ },
+ [SNDRV_PCM_FORMAT_U18_3BE] = {
+ .width = 18, .phys = 24, .le = 0, .signd = 0,
+ .silence = { 0x02, 0x00, 0x00 },
+ },
+};
+
+
+/**
+ * snd_pcm_format_signed - Check the PCM format is signed linear
+ * @format: the format to check
+ *
+ * Returns 1 if the given PCM format is signed linear, 0 if unsigned
+ * linear, and a negative error code for non-linear formats.
+ */
+int snd_pcm_format_signed(snd_pcm_format_t format)
+{
+ int val;
+ if (format < 0 || format > SNDRV_PCM_FORMAT_LAST)
+ return -EINVAL;
+ if ((val = pcm_formats[format].signd) < 0)
+ return -EINVAL;
+ return val;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_signed);
+
+/**
+ * snd_pcm_format_unsigned - Check the PCM format is unsigned linear
+ * @format: the format to check
+ *
+ * Returns 1 if the given PCM format is unsigned linear, 0 if signed
+ * linear, and a negative error code for non-linear formats.
+ */
+int snd_pcm_format_unsigned(snd_pcm_format_t format)
+{
+ int val;
+
+ val = snd_pcm_format_signed(format);
+ if (val < 0)
+ return val;
+ return !val;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_unsigned);
+
+/**
+ * snd_pcm_format_linear - Check the PCM format is linear
+ * @format: the format to check
+ *
+ * Returns 1 if the given PCM format is linear, 0 if not.
+ */
+int snd_pcm_format_linear(snd_pcm_format_t format)
+{
+ return snd_pcm_format_signed(format) >= 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_linear);
+
+/**
+ * snd_pcm_format_little_endian - Check the PCM format is little-endian
+ * @format: the format to check
+ *
+ * Returns 1 if the given PCM format is little-endian, 0 if
+ * big-endian, or a negative error code if endian not specified.
+ */
+int snd_pcm_format_little_endian(snd_pcm_format_t format)
+{
+ int val;
+ if (format < 0 || format > SNDRV_PCM_FORMAT_LAST)
+ return -EINVAL;
+ if ((val = pcm_formats[format].le) < 0)
+ return -EINVAL;
+ return val;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_little_endian);
+
+/**
+ * snd_pcm_format_big_endian - Check the PCM format is big-endian
+ * @format: the format to check
+ *
+ * Returns 1 if the given PCM format is big-endian, 0 if
+ * little-endian, or a negative error code if endian not specified.
+ */
+int snd_pcm_format_big_endian(snd_pcm_format_t format)
+{
+ int val;
+
+ val = snd_pcm_format_little_endian(format);
+ if (val < 0)
+ return val;
+ return !val;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_big_endian);
+
+/**
+ * snd_pcm_format_width - return the bit-width of the format
+ * @format: the format to check
+ *
+ * Returns the bit-width of the format, or a negative error code
+ * if unknown format.
+ */
+int snd_pcm_format_width(snd_pcm_format_t format)
+{
+ int val;
+ if (format < 0 || format > SNDRV_PCM_FORMAT_LAST)
+ return -EINVAL;
+ if ((val = pcm_formats[format].width) == 0)
+ return -EINVAL;
+ return val;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_width);
+
+/**
+ * snd_pcm_format_physical_width - return the physical bit-width of the format
+ * @format: the format to check
+ *
+ * Returns the physical bit-width of the format, or a negative error code
+ * if unknown format.
+ */
+int snd_pcm_format_physical_width(snd_pcm_format_t format)
+{
+ int val;
+ if (format < 0 || format > SNDRV_PCM_FORMAT_LAST)
+ return -EINVAL;
+ if ((val = pcm_formats[format].phys) == 0)
+ return -EINVAL;
+ return val;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_physical_width);
+
+/**
+ * snd_pcm_format_size - return the byte size of samples on the given format
+ * @format: the format to check
+ * @samples: sampling rate
+ *
+ * Returns the byte size of the given samples for the format, or a
+ * negative error code if unknown format.
+ */
+ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples)
+{
+ int phys_width = snd_pcm_format_physical_width(format);
+ if (phys_width < 0)
+ return -EINVAL;
+ return samples * phys_width / 8;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_size);
+
+/**
+ * snd_pcm_format_silence_64 - return the silent data in 8 bytes array
+ * @format: the format to check
+ *
+ * Returns the format pattern to fill or NULL if error.
+ */
+const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format)
+{
+ if (format < 0 || format > SNDRV_PCM_FORMAT_LAST)
+ return NULL;
+ if (! pcm_formats[format].phys)
+ return NULL;
+ return pcm_formats[format].silence;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_silence_64);
+
+/**
+ * snd_pcm_format_set_silence - set the silence data on the buffer
+ * @format: the PCM format
+ * @data: the buffer pointer
+ * @samples: the number of samples to set silence
+ *
+ * Sets the silence data on the buffer for the given samples.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int samples)
+{
+ int width;
+ unsigned char *dst, *pat;
+
+ if (format < 0 || format > SNDRV_PCM_FORMAT_LAST)
+ return -EINVAL;
+ if (samples == 0)
+ return 0;
+ width = pcm_formats[format].phys; /* physical width */
+ pat = pcm_formats[format].silence;
+ if (! width)
+ return -EINVAL;
+ /* signed or 1 byte data */
+ if (pcm_formats[format].signd == 1 || width <= 8) {
+ unsigned int bytes = samples * width / 8;
+ memset(data, *pat, bytes);
+ return 0;
+ }
+ /* non-zero samples, fill using a loop */
+ width /= 8;
+ dst = data;
+#if 0
+ while (samples--) {
+ memcpy(dst, pat, width);
+ dst += width;
+ }
+#else
+ /* a bit optimization for constant width */
+ switch (width) {
+ case 2:
+ while (samples--) {
+ memcpy(dst, pat, 2);
+ dst += 2;
+ }
+ break;
+ case 3:
+ while (samples--) {
+ memcpy(dst, pat, 3);
+ dst += 3;
+ }
+ break;
+ case 4:
+ while (samples--) {
+ memcpy(dst, pat, 4);
+ dst += 4;
+ }
+ break;
+ case 8:
+ while (samples--) {
+ memcpy(dst, pat, 8);
+ dst += 8;
+ }
+ break;
+ }
+#endif
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_format_set_silence);
+
+/**
+ * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields
+ * @runtime: the runtime instance
+ *
+ * Determines the rate_min and rate_max fields from the rates bits of
+ * the given runtime->hw.
+ *
+ * Returns zero if successful.
+ */
+int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime)
+{
+ int i;
+ for (i = 0; i < (int)snd_pcm_known_rates.count; i++) {
+ if (runtime->hw.rates & (1 << i)) {
+ runtime->hw.rate_min = snd_pcm_known_rates.list[i];
+ break;
+ }
+ }
+ for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) {
+ if (runtime->hw.rates & (1 << i)) {
+ runtime->hw.rate_max = snd_pcm_known_rates.list[i];
+ break;
+ }
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_limit_hw_rates);
+
+/**
+ * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit
+ * @rate: the sample rate to convert
+ *
+ * Returns the SNDRV_PCM_RATE_xxx flag that corresponds to the given rate, or
+ * SNDRV_PCM_RATE_KNOT for an unknown rate.
+ */
+unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate)
+{
+ unsigned int i;
+
+ for (i = 0; i < snd_pcm_known_rates.count; i++)
+ if (snd_pcm_known_rates.list[i] == rate)
+ return 1u << i;
+ return SNDRV_PCM_RATE_KNOT;
+}
+EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
new file mode 100644
index 0000000..a789efc
--- /dev/null
+++ b/sound/core/pcm_native.c
@@ -0,0 +1,3421 @@
+/*
+ * Digital Audio (PCM) abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/mm.h>
+#include <linux/file.h>
+#include <linux/slab.h>
+#include <linux/smp_lock.h>
+#include <linux/time.h>
+#include <linux/pm_qos_params.h>
+#include <linux/uio.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/timer.h>
+#include <sound/minors.h>
+#include <asm/io.h>
+
+/*
+ * Compatibility
+ */
+
+struct snd_pcm_hw_params_old {
+ unsigned int flags;
+ unsigned int masks[SNDRV_PCM_HW_PARAM_SUBFORMAT -
+ SNDRV_PCM_HW_PARAM_ACCESS + 1];
+ struct snd_interval intervals[SNDRV_PCM_HW_PARAM_TICK_TIME -
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS + 1];
+ unsigned int rmask;
+ unsigned int cmask;
+ unsigned int info;
+ unsigned int msbits;
+ unsigned int rate_num;
+ unsigned int rate_den;
+ snd_pcm_uframes_t fifo_size;
+ unsigned char reserved[64];
+};
+
+#ifdef CONFIG_SND_SUPPORT_OLD_API
+#define SNDRV_PCM_IOCTL_HW_REFINE_OLD _IOWR('A', 0x10, struct snd_pcm_hw_params_old)
+#define SNDRV_PCM_IOCTL_HW_PARAMS_OLD _IOWR('A', 0x11, struct snd_pcm_hw_params_old)
+
+static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params_old __user * _oparams);
+static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params_old __user * _oparams);
+#endif
+static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream);
+
+/*
+ *
+ */
+
+DEFINE_RWLOCK(snd_pcm_link_rwlock);
+EXPORT_SYMBOL(snd_pcm_link_rwlock);
+
+static DECLARE_RWSEM(snd_pcm_link_rwsem);
+
+static inline mm_segment_t snd_enter_user(void)
+{
+ mm_segment_t fs = get_fs();
+ set_fs(get_ds());
+ return fs;
+}
+
+static inline void snd_leave_user(mm_segment_t fs)
+{
+ set_fs(fs);
+}
+
+
+
+int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info)
+{
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm *pcm = substream->pcm;
+ struct snd_pcm_str *pstr = substream->pstr;
+
+ memset(info, 0, sizeof(*info));
+ info->card = pcm->card->number;
+ info->device = pcm->device;
+ info->stream = substream->stream;
+ info->subdevice = substream->number;
+ strlcpy(info->id, pcm->id, sizeof(info->id));
+ strlcpy(info->name, pcm->name, sizeof(info->name));
+ info->dev_class = pcm->dev_class;
+ info->dev_subclass = pcm->dev_subclass;
+ info->subdevices_count = pstr->substream_count;
+ info->subdevices_avail = pstr->substream_count - pstr->substream_opened;
+ strlcpy(info->subname, substream->name, sizeof(info->subname));
+ runtime = substream->runtime;
+ /* AB: FIXME!!! This is definitely nonsense */
+ if (runtime) {
+ info->sync = runtime->sync;
+ substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_INFO, info);
+ }
+ return 0;
+}
+
+int snd_pcm_info_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_info __user * _info)
+{
+ struct snd_pcm_info *info;
+ int err;
+
+ info = kmalloc(sizeof(*info), GFP_KERNEL);
+ if (! info)
+ return -ENOMEM;
+ err = snd_pcm_info(substream, info);
+ if (err >= 0) {
+ if (copy_to_user(_info, info, sizeof(*info)))
+ err = -EFAULT;
+ }
+ kfree(info);
+ return err;
+}
+
+#undef RULES_DEBUG
+
+#ifdef RULES_DEBUG
+#define HW_PARAM(v) [SNDRV_PCM_HW_PARAM_##v] = #v
+char *snd_pcm_hw_param_names[] = {
+ HW_PARAM(ACCESS),
+ HW_PARAM(FORMAT),
+ HW_PARAM(SUBFORMAT),
+ HW_PARAM(SAMPLE_BITS),
+ HW_PARAM(FRAME_BITS),
+ HW_PARAM(CHANNELS),
+ HW_PARAM(RATE),
+ HW_PARAM(PERIOD_TIME),
+ HW_PARAM(PERIOD_SIZE),
+ HW_PARAM(PERIOD_BYTES),
+ HW_PARAM(PERIODS),
+ HW_PARAM(BUFFER_TIME),
+ HW_PARAM(BUFFER_SIZE),
+ HW_PARAM(BUFFER_BYTES),
+ HW_PARAM(TICK_TIME),
+};
+#endif
+
+int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned int k;
+ struct snd_pcm_hardware *hw;
+ struct snd_interval *i = NULL;
+ struct snd_mask *m = NULL;
+ struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints;
+ unsigned int rstamps[constrs->rules_num];
+ unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1];
+ unsigned int stamp = 2;
+ int changed, again;
+
+ params->info = 0;
+ params->fifo_size = 0;
+ if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS))
+ params->msbits = 0;
+ if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) {
+ params->rate_num = 0;
+ params->rate_den = 0;
+ }
+
+ for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) {
+ m = hw_param_mask(params, k);
+ if (snd_mask_empty(m))
+ return -EINVAL;
+ if (!(params->rmask & (1 << k)))
+ continue;
+#ifdef RULES_DEBUG
+ printk("%s = ", snd_pcm_hw_param_names[k]);
+ printk("%04x%04x%04x%04x -> ", m->bits[3], m->bits[2], m->bits[1], m->bits[0]);
+#endif
+ changed = snd_mask_refine(m, constrs_mask(constrs, k));
+#ifdef RULES_DEBUG
+ printk("%04x%04x%04x%04x\n", m->bits[3], m->bits[2], m->bits[1], m->bits[0]);
+#endif
+ if (changed)
+ params->cmask |= 1 << k;
+ if (changed < 0)
+ return changed;
+ }
+
+ for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) {
+ i = hw_param_interval(params, k);
+ if (snd_interval_empty(i))
+ return -EINVAL;
+ if (!(params->rmask & (1 << k)))
+ continue;
+#ifdef RULES_DEBUG
+ printk("%s = ", snd_pcm_hw_param_names[k]);
+ if (i->empty)
+ printk("empty");
+ else
+ printk("%c%u %u%c",
+ i->openmin ? '(' : '[', i->min,
+ i->max, i->openmax ? ')' : ']');
+ printk(" -> ");
+#endif
+ changed = snd_interval_refine(i, constrs_interval(constrs, k));
+#ifdef RULES_DEBUG
+ if (i->empty)
+ printk("empty\n");
+ else
+ printk("%c%u %u%c\n",
+ i->openmin ? '(' : '[', i->min,
+ i->max, i->openmax ? ')' : ']');
+#endif
+ if (changed)
+ params->cmask |= 1 << k;
+ if (changed < 0)
+ return changed;
+ }
+
+ for (k = 0; k < constrs->rules_num; k++)
+ rstamps[k] = 0;
+ for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++)
+ vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0;
+ do {
+ again = 0;
+ for (k = 0; k < constrs->rules_num; k++) {
+ struct snd_pcm_hw_rule *r = &constrs->rules[k];
+ unsigned int d;
+ int doit = 0;
+ if (r->cond && !(r->cond & params->flags))
+ continue;
+ for (d = 0; r->deps[d] >= 0; d++) {
+ if (vstamps[r->deps[d]] > rstamps[k]) {
+ doit = 1;
+ break;
+ }
+ }
+ if (!doit)
+ continue;
+#ifdef RULES_DEBUG
+ printk("Rule %d [%p]: ", k, r->func);
+ if (r->var >= 0) {
+ printk("%s = ", snd_pcm_hw_param_names[r->var]);
+ if (hw_is_mask(r->var)) {
+ m = hw_param_mask(params, r->var);
+ printk("%x", *m->bits);
+ } else {
+ i = hw_param_interval(params, r->var);
+ if (i->empty)
+ printk("empty");
+ else
+ printk("%c%u %u%c",
+ i->openmin ? '(' : '[', i->min,
+ i->max, i->openmax ? ')' : ']');
+ }
+ }
+#endif
+ changed = r->func(params, r);
+#ifdef RULES_DEBUG
+ if (r->var >= 0) {
+ printk(" -> ");
+ if (hw_is_mask(r->var))
+ printk("%x", *m->bits);
+ else {
+ if (i->empty)
+ printk("empty");
+ else
+ printk("%c%u %u%c",
+ i->openmin ? '(' : '[', i->min,
+ i->max, i->openmax ? ')' : ']');
+ }
+ }
+ printk("\n");
+#endif
+ rstamps[k] = stamp;
+ if (changed && r->var >= 0) {
+ params->cmask |= (1 << r->var);
+ vstamps[r->var] = stamp;
+ again = 1;
+ }
+ if (changed < 0)
+ return changed;
+ stamp++;
+ }
+ } while (again);
+ if (!params->msbits) {
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
+ if (snd_interval_single(i))
+ params->msbits = snd_interval_value(i);
+ }
+
+ if (!params->rate_den) {
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (snd_interval_single(i)) {
+ params->rate_num = snd_interval_value(i);
+ params->rate_den = 1;
+ }
+ }
+
+ hw = &substream->runtime->hw;
+ if (!params->info)
+ params->info = hw->info;
+ if (!params->fifo_size)
+ params->fifo_size = hw->fifo_size;
+ params->rmask = 0;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_hw_refine);
+
+static int snd_pcm_hw_refine_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params __user * _params)
+{
+ struct snd_pcm_hw_params *params;
+ int err;
+
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ if (!params) {
+ err = -ENOMEM;
+ goto out;
+ }
+ if (copy_from_user(params, _params, sizeof(*params))) {
+ err = -EFAULT;
+ goto out;
+ }
+ err = snd_pcm_hw_refine(substream, params);
+ if (copy_to_user(_params, params, sizeof(*params))) {
+ if (!err)
+ err = -EFAULT;
+ }
+out:
+ kfree(params);
+ return err;
+}
+
+static int period_to_usecs(struct snd_pcm_runtime *runtime)
+{
+ int usecs;
+
+ if (! runtime->rate)
+ return -1; /* invalid */
+
+ /* take 75% of period time as the deadline */
+ usecs = (750000 / runtime->rate) * runtime->period_size;
+ usecs += ((750000 % runtime->rate) * runtime->period_size) /
+ runtime->rate;
+
+ return usecs;
+}
+
+static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime;
+ int err, usecs;
+ unsigned int bits;
+ snd_pcm_uframes_t frames;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
+ break;
+ default:
+ snd_pcm_stream_unlock_irq(substream);
+ return -EBADFD;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ if (!substream->oss.oss)
+#endif
+ if (atomic_read(&substream->mmap_count))
+ return -EBADFD;
+
+ params->rmask = ~0U;
+ err = snd_pcm_hw_refine(substream, params);
+ if (err < 0)
+ goto _error;
+
+ err = snd_pcm_hw_params_choose(substream, params);
+ if (err < 0)
+ goto _error;
+
+ if (substream->ops->hw_params != NULL) {
+ err = substream->ops->hw_params(substream, params);
+ if (err < 0)
+ goto _error;
+ }
+
+ runtime->access = params_access(params);
+ runtime->format = params_format(params);
+ runtime->subformat = params_subformat(params);
+ runtime->channels = params_channels(params);
+ runtime->rate = params_rate(params);
+ runtime->period_size = params_period_size(params);
+ runtime->periods = params_periods(params);
+ runtime->buffer_size = params_buffer_size(params);
+ runtime->info = params->info;
+ runtime->rate_num = params->rate_num;
+ runtime->rate_den = params->rate_den;
+
+ bits = snd_pcm_format_physical_width(runtime->format);
+ runtime->sample_bits = bits;
+ bits *= runtime->channels;
+ runtime->frame_bits = bits;
+ frames = 1;
+ while (bits % 8 != 0) {
+ bits *= 2;
+ frames *= 2;
+ }
+ runtime->byte_align = bits / 8;
+ runtime->min_align = frames;
+
+ /* Default sw params */
+ runtime->tstamp_mode = SNDRV_PCM_TSTAMP_NONE;
+ runtime->period_step = 1;
+ runtime->control->avail_min = runtime->period_size;
+ runtime->start_threshold = 1;
+ runtime->stop_threshold = runtime->buffer_size;
+ runtime->silence_threshold = 0;
+ runtime->silence_size = 0;
+ runtime->boundary = runtime->buffer_size;
+ while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
+ runtime->boundary *= 2;
+
+ snd_pcm_timer_resolution_change(substream);
+ runtime->status->state = SNDRV_PCM_STATE_SETUP;
+
+ pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY,
+ substream->latency_id);
+ if ((usecs = period_to_usecs(runtime)) >= 0)
+ pm_qos_add_requirement(PM_QOS_CPU_DMA_LATENCY,
+ substream->latency_id, usecs);
+ return 0;
+ _error:
+ /* hardware might be unuseable from this time,
+ so we force application to retry to set
+ the correct hardware parameter settings */
+ runtime->status->state = SNDRV_PCM_STATE_OPEN;
+ if (substream->ops->hw_free != NULL)
+ substream->ops->hw_free(substream);
+ return err;
+}
+
+static int snd_pcm_hw_params_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params __user * _params)
+{
+ struct snd_pcm_hw_params *params;
+ int err;
+
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ if (!params) {
+ err = -ENOMEM;
+ goto out;
+ }
+ if (copy_from_user(params, _params, sizeof(*params))) {
+ err = -EFAULT;
+ goto out;
+ }
+ err = snd_pcm_hw_params(substream, params);
+ if (copy_to_user(_params, params, sizeof(*params))) {
+ if (!err)
+ err = -EFAULT;
+ }
+out:
+ kfree(params);
+ return err;
+}
+
+static int snd_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+ int result = 0;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
+ break;
+ default:
+ snd_pcm_stream_unlock_irq(substream);
+ return -EBADFD;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ if (atomic_read(&substream->mmap_count))
+ return -EBADFD;
+ if (substream->ops->hw_free)
+ result = substream->ops->hw_free(substream);
+ runtime->status->state = SNDRV_PCM_STATE_OPEN;
+ pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY,
+ substream->latency_id);
+ return result;
+}
+
+static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_sw_params *params)
+{
+ struct snd_pcm_runtime *runtime;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ snd_pcm_stream_lock_irq(substream);
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ snd_pcm_stream_unlock_irq(substream);
+ return -EBADFD;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+
+ if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
+ return -EINVAL;
+ if (params->avail_min == 0)
+ return -EINVAL;
+ if (params->silence_size >= runtime->boundary) {
+ if (params->silence_threshold != 0)
+ return -EINVAL;
+ } else {
+ if (params->silence_size > params->silence_threshold)
+ return -EINVAL;
+ if (params->silence_threshold > runtime->buffer_size)
+ return -EINVAL;
+ }
+ snd_pcm_stream_lock_irq(substream);
+ runtime->tstamp_mode = params->tstamp_mode;
+ runtime->period_step = params->period_step;
+ runtime->control->avail_min = params->avail_min;
+ runtime->start_threshold = params->start_threshold;
+ runtime->stop_threshold = params->stop_threshold;
+ runtime->silence_threshold = params->silence_threshold;
+ runtime->silence_size = params->silence_size;
+ params->boundary = runtime->boundary;
+ if (snd_pcm_running(substream)) {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ runtime->silence_size > 0)
+ snd_pcm_playback_silence(substream, ULONG_MAX);
+ wake_up(&runtime->sleep);
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ return 0;
+}
+
+static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_sw_params __user * _params)
+{
+ struct snd_pcm_sw_params params;
+ int err;
+ if (copy_from_user(&params, _params, sizeof(params)))
+ return -EFAULT;
+ err = snd_pcm_sw_params(substream, &params);
+ if (copy_to_user(_params, &params, sizeof(params)))
+ return -EFAULT;
+ return err;
+}
+
+int snd_pcm_status(struct snd_pcm_substream *substream,
+ struct snd_pcm_status *status)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_pcm_stream_lock_irq(substream);
+ status->state = runtime->status->state;
+ status->suspended_state = runtime->status->suspended_state;
+ if (status->state == SNDRV_PCM_STATE_OPEN)
+ goto _end;
+ status->trigger_tstamp = runtime->trigger_tstamp;
+ if (snd_pcm_running(substream)) {
+ snd_pcm_update_hw_ptr(substream);
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) {
+ status->tstamp = runtime->status->tstamp;
+ goto _tstamp_end;
+ }
+ }
+ snd_pcm_gettime(runtime, &status->tstamp);
+ _tstamp_end:
+ status->appl_ptr = runtime->control->appl_ptr;
+ status->hw_ptr = runtime->status->hw_ptr;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ status->avail = snd_pcm_playback_avail(runtime);
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
+ runtime->status->state == SNDRV_PCM_STATE_DRAINING)
+ status->delay = runtime->buffer_size - status->avail;
+ else
+ status->delay = 0;
+ } else {
+ status->avail = snd_pcm_capture_avail(runtime);
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ status->delay = status->avail;
+ else
+ status->delay = 0;
+ }
+ status->avail_max = runtime->avail_max;
+ status->overrange = runtime->overrange;
+ runtime->avail_max = 0;
+ runtime->overrange = 0;
+ _end:
+ snd_pcm_stream_unlock_irq(substream);
+ return 0;
+}
+
+static int snd_pcm_status_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_status __user * _status)
+{
+ struct snd_pcm_status status;
+ int res;
+
+ memset(&status, 0, sizeof(status));
+ res = snd_pcm_status(substream, &status);
+ if (res < 0)
+ return res;
+ if (copy_to_user(_status, &status, sizeof(status)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_pcm_channel_info(struct snd_pcm_substream *substream,
+ struct snd_pcm_channel_info * info)
+{
+ struct snd_pcm_runtime *runtime;
+ unsigned int channel;
+
+ channel = info->channel;
+ runtime = substream->runtime;
+ snd_pcm_stream_lock_irq(substream);
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ snd_pcm_stream_unlock_irq(substream);
+ return -EBADFD;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ if (channel >= runtime->channels)
+ return -EINVAL;
+ memset(info, 0, sizeof(*info));
+ info->channel = channel;
+ return substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_CHANNEL_INFO, info);
+}
+
+static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_channel_info __user * _info)
+{
+ struct snd_pcm_channel_info info;
+ int res;
+
+ if (copy_from_user(&info, _info, sizeof(info)))
+ return -EFAULT;
+ res = snd_pcm_channel_info(substream, &info);
+ if (res < 0)
+ return res;
+ if (copy_to_user(_info, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static void snd_pcm_trigger_tstamp(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->trigger_master == NULL)
+ return;
+ if (runtime->trigger_master == substream) {
+ snd_pcm_gettime(runtime, &runtime->trigger_tstamp);
+ } else {
+ snd_pcm_trigger_tstamp(runtime->trigger_master);
+ runtime->trigger_tstamp = runtime->trigger_master->runtime->trigger_tstamp;
+ }
+ runtime->trigger_master = NULL;
+}
+
+struct action_ops {
+ int (*pre_action)(struct snd_pcm_substream *substream, int state);
+ int (*do_action)(struct snd_pcm_substream *substream, int state);
+ void (*undo_action)(struct snd_pcm_substream *substream, int state);
+ void (*post_action)(struct snd_pcm_substream *substream, int state);
+};
+
+/*
+ * this functions is core for handling of linked stream
+ * Note: the stream state might be changed also on failure
+ * Note2: call with calling stream lock + link lock
+ */
+static int snd_pcm_action_group(struct action_ops *ops,
+ struct snd_pcm_substream *substream,
+ int state, int do_lock)
+{
+ struct snd_pcm_substream *s = NULL;
+ struct snd_pcm_substream *s1;
+ int res = 0;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (do_lock && s != substream)
+ spin_lock_nested(&s->self_group.lock,
+ SINGLE_DEPTH_NESTING);
+ res = ops->pre_action(s, state);
+ if (res < 0)
+ goto _unlock;
+ }
+ snd_pcm_group_for_each_entry(s, substream) {
+ res = ops->do_action(s, state);
+ if (res < 0) {
+ if (ops->undo_action) {
+ snd_pcm_group_for_each_entry(s1, substream) {
+ if (s1 == s) /* failed stream */
+ break;
+ ops->undo_action(s1, state);
+ }
+ }
+ s = NULL; /* unlock all */
+ goto _unlock;
+ }
+ }
+ snd_pcm_group_for_each_entry(s, substream) {
+ ops->post_action(s, state);
+ }
+ _unlock:
+ if (do_lock) {
+ /* unlock streams */
+ snd_pcm_group_for_each_entry(s1, substream) {
+ if (s1 != substream)
+ spin_unlock(&s1->self_group.lock);
+ if (s1 == s) /* end */
+ break;
+ }
+ }
+ return res;
+}
+
+/*
+ * Note: call with stream lock
+ */
+static int snd_pcm_action_single(struct action_ops *ops,
+ struct snd_pcm_substream *substream,
+ int state)
+{
+ int res;
+
+ res = ops->pre_action(substream, state);
+ if (res < 0)
+ return res;
+ res = ops->do_action(substream, state);
+ if (res == 0)
+ ops->post_action(substream, state);
+ else if (ops->undo_action)
+ ops->undo_action(substream, state);
+ return res;
+}
+
+/*
+ * Note: call with stream lock
+ */
+static int snd_pcm_action(struct action_ops *ops,
+ struct snd_pcm_substream *substream,
+ int state)
+{
+ int res;
+
+ if (snd_pcm_stream_linked(substream)) {
+ if (!spin_trylock(&substream->group->lock)) {
+ spin_unlock(&substream->self_group.lock);
+ spin_lock(&substream->group->lock);
+ spin_lock(&substream->self_group.lock);
+ }
+ res = snd_pcm_action_group(ops, substream, state, 1);
+ spin_unlock(&substream->group->lock);
+ } else {
+ res = snd_pcm_action_single(ops, substream, state);
+ }
+ return res;
+}
+
+/*
+ * Note: don't use any locks before
+ */
+static int snd_pcm_action_lock_irq(struct action_ops *ops,
+ struct snd_pcm_substream *substream,
+ int state)
+{
+ int res;
+
+ read_lock_irq(&snd_pcm_link_rwlock);
+ if (snd_pcm_stream_linked(substream)) {
+ spin_lock(&substream->group->lock);
+ spin_lock(&substream->self_group.lock);
+ res = snd_pcm_action_group(ops, substream, state, 1);
+ spin_unlock(&substream->self_group.lock);
+ spin_unlock(&substream->group->lock);
+ } else {
+ spin_lock(&substream->self_group.lock);
+ res = snd_pcm_action_single(ops, substream, state);
+ spin_unlock(&substream->self_group.lock);
+ }
+ read_unlock_irq(&snd_pcm_link_rwlock);
+ return res;
+}
+
+/*
+ */
+static int snd_pcm_action_nonatomic(struct action_ops *ops,
+ struct snd_pcm_substream *substream,
+ int state)
+{
+ int res;
+
+ down_read(&snd_pcm_link_rwsem);
+ if (snd_pcm_stream_linked(substream))
+ res = snd_pcm_action_group(ops, substream, state, 0);
+ else
+ res = snd_pcm_action_single(ops, substream, state);
+ up_read(&snd_pcm_link_rwsem);
+ return res;
+}
+
+/*
+ * start callbacks
+ */
+static int snd_pcm_pre_start(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->status->state != SNDRV_PCM_STATE_PREPARED)
+ return -EBADFD;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ !snd_pcm_playback_data(substream))
+ return -EPIPE;
+ runtime->trigger_master = substream;
+ return 0;
+}
+
+static int snd_pcm_do_start(struct snd_pcm_substream *substream, int state)
+{
+ if (substream->runtime->trigger_master != substream)
+ return 0;
+ return substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_START);
+}
+
+static void snd_pcm_undo_start(struct snd_pcm_substream *substream, int state)
+{
+ if (substream->runtime->trigger_master == substream)
+ substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP);
+}
+
+static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_trigger_tstamp(substream);
+ runtime->status->state = state;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ runtime->silence_size > 0)
+ snd_pcm_playback_silence(substream, ULONG_MAX);
+ if (substream->timer)
+ snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART,
+ &runtime->trigger_tstamp);
+}
+
+static struct action_ops snd_pcm_action_start = {
+ .pre_action = snd_pcm_pre_start,
+ .do_action = snd_pcm_do_start,
+ .undo_action = snd_pcm_undo_start,
+ .post_action = snd_pcm_post_start
+};
+
+/**
+ * snd_pcm_start - start all linked streams
+ * @substream: the PCM substream instance
+ */
+int snd_pcm_start(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_action(&snd_pcm_action_start, substream,
+ SNDRV_PCM_STATE_RUNNING);
+}
+
+/*
+ * stop callbacks
+ */
+static int snd_pcm_pre_stop(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ runtime->trigger_master = substream;
+ return 0;
+}
+
+static int snd_pcm_do_stop(struct snd_pcm_substream *substream, int state)
+{
+ if (substream->runtime->trigger_master == substream &&
+ snd_pcm_running(substream))
+ substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP);
+ return 0; /* unconditonally stop all substreams */
+}
+
+static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->status->state != state) {
+ snd_pcm_trigger_tstamp(substream);
+ if (substream->timer)
+ snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP,
+ &runtime->trigger_tstamp);
+ runtime->status->state = state;
+ }
+ wake_up(&runtime->sleep);
+}
+
+static struct action_ops snd_pcm_action_stop = {
+ .pre_action = snd_pcm_pre_stop,
+ .do_action = snd_pcm_do_stop,
+ .post_action = snd_pcm_post_stop
+};
+
+/**
+ * snd_pcm_stop - try to stop all running streams in the substream group
+ * @substream: the PCM substream instance
+ * @state: PCM state after stopping the stream
+ *
+ * The state of each stream is then changed to the given state unconditionally.
+ */
+int snd_pcm_stop(struct snd_pcm_substream *substream, int state)
+{
+ return snd_pcm_action(&snd_pcm_action_stop, substream, state);
+}
+
+EXPORT_SYMBOL(snd_pcm_stop);
+
+/**
+ * snd_pcm_drain_done - stop the DMA only when the given stream is playback
+ * @substream: the PCM substream
+ *
+ * After stopping, the state is changed to SETUP.
+ * Unlike snd_pcm_stop(), this affects only the given stream.
+ */
+int snd_pcm_drain_done(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_action_single(&snd_pcm_action_stop, substream,
+ SNDRV_PCM_STATE_SETUP);
+}
+
+/*
+ * pause callbacks
+ */
+static int snd_pcm_pre_pause(struct snd_pcm_substream *substream, int push)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (!(runtime->info & SNDRV_PCM_INFO_PAUSE))
+ return -ENOSYS;
+ if (push) {
+ if (runtime->status->state != SNDRV_PCM_STATE_RUNNING)
+ return -EBADFD;
+ } else if (runtime->status->state != SNDRV_PCM_STATE_PAUSED)
+ return -EBADFD;
+ runtime->trigger_master = substream;
+ return 0;
+}
+
+static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
+{
+ if (substream->runtime->trigger_master != substream)
+ return 0;
+ return substream->ops->trigger(substream,
+ push ? SNDRV_PCM_TRIGGER_PAUSE_PUSH :
+ SNDRV_PCM_TRIGGER_PAUSE_RELEASE);
+}
+
+static void snd_pcm_undo_pause(struct snd_pcm_substream *substream, int push)
+{
+ if (substream->runtime->trigger_master == substream)
+ substream->ops->trigger(substream,
+ push ? SNDRV_PCM_TRIGGER_PAUSE_RELEASE :
+ SNDRV_PCM_TRIGGER_PAUSE_PUSH);
+}
+
+static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_trigger_tstamp(substream);
+ if (push) {
+ runtime->status->state = SNDRV_PCM_STATE_PAUSED;
+ if (substream->timer)
+ snd_timer_notify(substream->timer,
+ SNDRV_TIMER_EVENT_MPAUSE,
+ &runtime->trigger_tstamp);
+ wake_up(&runtime->sleep);
+ } else {
+ runtime->status->state = SNDRV_PCM_STATE_RUNNING;
+ if (substream->timer)
+ snd_timer_notify(substream->timer,
+ SNDRV_TIMER_EVENT_MCONTINUE,
+ &runtime->trigger_tstamp);
+ }
+}
+
+static struct action_ops snd_pcm_action_pause = {
+ .pre_action = snd_pcm_pre_pause,
+ .do_action = snd_pcm_do_pause,
+ .undo_action = snd_pcm_undo_pause,
+ .post_action = snd_pcm_post_pause
+};
+
+/*
+ * Push/release the pause for all linked streams.
+ */
+static int snd_pcm_pause(struct snd_pcm_substream *substream, int push)
+{
+ return snd_pcm_action(&snd_pcm_action_pause, substream, push);
+}
+
+#ifdef CONFIG_PM
+/* suspend */
+
+static int snd_pcm_pre_suspend(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED)
+ return -EBUSY;
+ runtime->trigger_master = substream;
+ return 0;
+}
+
+static int snd_pcm_do_suspend(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->trigger_master != substream)
+ return 0;
+ if (! snd_pcm_running(substream))
+ return 0;
+ substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_SUSPEND);
+ return 0; /* suspend unconditionally */
+}
+
+static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_trigger_tstamp(substream);
+ if (substream->timer)
+ snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND,
+ &runtime->trigger_tstamp);
+ runtime->status->suspended_state = runtime->status->state;
+ runtime->status->state = SNDRV_PCM_STATE_SUSPENDED;
+ wake_up(&runtime->sleep);
+}
+
+static struct action_ops snd_pcm_action_suspend = {
+ .pre_action = snd_pcm_pre_suspend,
+ .do_action = snd_pcm_do_suspend,
+ .post_action = snd_pcm_post_suspend
+};
+
+/**
+ * snd_pcm_suspend - trigger SUSPEND to all linked streams
+ * @substream: the PCM substream
+ *
+ * After this call, all streams are changed to SUSPENDED state.
+ */
+int snd_pcm_suspend(struct snd_pcm_substream *substream)
+{
+ int err;
+ unsigned long flags;
+
+ if (! substream)
+ return 0;
+
+ snd_pcm_stream_lock_irqsave(substream, flags);
+ err = snd_pcm_action(&snd_pcm_action_suspend, substream, 0);
+ snd_pcm_stream_unlock_irqrestore(substream, flags);
+ return err;
+}
+
+EXPORT_SYMBOL(snd_pcm_suspend);
+
+/**
+ * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm
+ * @pcm: the PCM instance
+ *
+ * After this call, all streams are changed to SUSPENDED state.
+ */
+int snd_pcm_suspend_all(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ int stream, err = 0;
+
+ if (! pcm)
+ return 0;
+
+ for (stream = 0; stream < 2; stream++) {
+ for (substream = pcm->streams[stream].substream;
+ substream; substream = substream->next) {
+ /* FIXME: the open/close code should lock this as well */
+ if (substream->runtime == NULL)
+ continue;
+ err = snd_pcm_suspend(substream);
+ if (err < 0 && err != -EBUSY)
+ return err;
+ }
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_suspend_all);
+
+/* resume */
+
+static int snd_pcm_pre_resume(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (!(runtime->info & SNDRV_PCM_INFO_RESUME))
+ return -ENOSYS;
+ runtime->trigger_master = substream;
+ return 0;
+}
+
+static int snd_pcm_do_resume(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->trigger_master != substream)
+ return 0;
+ /* DMA not running previously? */
+ if (runtime->status->suspended_state != SNDRV_PCM_STATE_RUNNING &&
+ (runtime->status->suspended_state != SNDRV_PCM_STATE_DRAINING ||
+ substream->stream != SNDRV_PCM_STREAM_PLAYBACK))
+ return 0;
+ return substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_RESUME);
+}
+
+static void snd_pcm_undo_resume(struct snd_pcm_substream *substream, int state)
+{
+ if (substream->runtime->trigger_master == substream &&
+ snd_pcm_running(substream))
+ substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_SUSPEND);
+}
+
+static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_trigger_tstamp(substream);
+ if (substream->timer)
+ snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME,
+ &runtime->trigger_tstamp);
+ runtime->status->state = runtime->status->suspended_state;
+}
+
+static struct action_ops snd_pcm_action_resume = {
+ .pre_action = snd_pcm_pre_resume,
+ .do_action = snd_pcm_do_resume,
+ .undo_action = snd_pcm_undo_resume,
+ .post_action = snd_pcm_post_resume
+};
+
+static int snd_pcm_resume(struct snd_pcm_substream *substream)
+{
+ struct snd_card *card = substream->pcm->card;
+ int res;
+
+ snd_power_lock(card);
+ if ((res = snd_power_wait(card, SNDRV_CTL_POWER_D0)) >= 0)
+ res = snd_pcm_action_lock_irq(&snd_pcm_action_resume, substream, 0);
+ snd_power_unlock(card);
+ return res;
+}
+
+#else
+
+static int snd_pcm_resume(struct snd_pcm_substream *substream)
+{
+ return -ENOSYS;
+}
+
+#endif /* CONFIG_PM */
+
+/*
+ * xrun ioctl
+ *
+ * Change the RUNNING stream(s) to XRUN state.
+ */
+static int snd_pcm_xrun(struct snd_pcm_substream *substream)
+{
+ struct snd_card *card = substream->pcm->card;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int result;
+
+ snd_power_lock(card);
+ if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
+ result = snd_power_wait(card, SNDRV_CTL_POWER_D0);
+ if (result < 0)
+ goto _unlock;
+ }
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_XRUN:
+ result = 0; /* already there */
+ break;
+ case SNDRV_PCM_STATE_RUNNING:
+ result = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ break;
+ default:
+ result = -EBADFD;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ _unlock:
+ snd_power_unlock(card);
+ return result;
+}
+
+/*
+ * reset ioctl
+ */
+static int snd_pcm_pre_reset(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_RUNNING:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
+ case SNDRV_PCM_STATE_SUSPENDED:
+ return 0;
+ default:
+ return -EBADFD;
+ }
+}
+
+static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err = substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL);
+ if (err < 0)
+ return err;
+ runtime->hw_ptr_base = 0;
+ runtime->hw_ptr_interrupt = runtime->status->hw_ptr -
+ runtime->status->hw_ptr % runtime->period_size;
+ runtime->silence_start = runtime->status->hw_ptr;
+ runtime->silence_filled = 0;
+ return 0;
+}
+
+static void snd_pcm_post_reset(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ runtime->control->appl_ptr = runtime->status->hw_ptr;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ runtime->silence_size > 0)
+ snd_pcm_playback_silence(substream, ULONG_MAX);
+}
+
+static struct action_ops snd_pcm_action_reset = {
+ .pre_action = snd_pcm_pre_reset,
+ .do_action = snd_pcm_do_reset,
+ .post_action = snd_pcm_post_reset
+};
+
+static int snd_pcm_reset(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_action_nonatomic(&snd_pcm_action_reset, substream, 0);
+}
+
+/*
+ * prepare ioctl
+ */
+/* we use the second argument for updating f_flags */
+static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream,
+ int f_flags)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN ||
+ runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED)
+ return -EBADFD;
+ if (snd_pcm_running(substream))
+ return -EBUSY;
+ substream->f_flags = f_flags;
+ return 0;
+}
+
+static int snd_pcm_do_prepare(struct snd_pcm_substream *substream, int state)
+{
+ int err;
+ err = substream->ops->prepare(substream);
+ if (err < 0)
+ return err;
+ return snd_pcm_do_reset(substream, 0);
+}
+
+static void snd_pcm_post_prepare(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ runtime->control->appl_ptr = runtime->status->hw_ptr;
+ runtime->status->state = SNDRV_PCM_STATE_PREPARED;
+}
+
+static struct action_ops snd_pcm_action_prepare = {
+ .pre_action = snd_pcm_pre_prepare,
+ .do_action = snd_pcm_do_prepare,
+ .post_action = snd_pcm_post_prepare
+};
+
+/**
+ * snd_pcm_prepare - prepare the PCM substream to be triggerable
+ * @substream: the PCM substream instance
+ * @file: file to refer f_flags
+ */
+static int snd_pcm_prepare(struct snd_pcm_substream *substream,
+ struct file *file)
+{
+ int res;
+ struct snd_card *card = substream->pcm->card;
+ int f_flags;
+
+ if (file)
+ f_flags = file->f_flags;
+ else
+ f_flags = substream->f_flags;
+
+ snd_power_lock(card);
+ if ((res = snd_power_wait(card, SNDRV_CTL_POWER_D0)) >= 0)
+ res = snd_pcm_action_nonatomic(&snd_pcm_action_prepare,
+ substream, f_flags);
+ snd_power_unlock(card);
+ return res;
+}
+
+/*
+ * drain ioctl
+ */
+
+static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state)
+{
+ if (substream->f_flags & O_NONBLOCK)
+ return -EAGAIN;
+ substream->runtime->trigger_master = substream;
+ return 0;
+}
+
+static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_PREPARED:
+ /* start playback stream if possible */
+ if (! snd_pcm_playback_empty(substream)) {
+ snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING);
+ snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING);
+ }
+ break;
+ case SNDRV_PCM_STATE_RUNNING:
+ runtime->status->state = SNDRV_PCM_STATE_DRAINING;
+ break;
+ default:
+ break;
+ }
+ } else {
+ /* stop running stream */
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) {
+ int new_state = snd_pcm_capture_avail(runtime) > 0 ?
+ SNDRV_PCM_STATE_DRAINING : SNDRV_PCM_STATE_SETUP;
+ snd_pcm_do_stop(substream, new_state);
+ snd_pcm_post_stop(substream, new_state);
+ }
+ }
+ return 0;
+}
+
+static void snd_pcm_post_drain_init(struct snd_pcm_substream *substream, int state)
+{
+}
+
+static struct action_ops snd_pcm_action_drain_init = {
+ .pre_action = snd_pcm_pre_drain_init,
+ .do_action = snd_pcm_do_drain_init,
+ .post_action = snd_pcm_post_drain_init
+};
+
+struct drain_rec {
+ struct snd_pcm_substream *substream;
+ wait_queue_t wait;
+ snd_pcm_uframes_t stop_threshold;
+};
+
+static int snd_pcm_drop(struct snd_pcm_substream *substream);
+
+/*
+ * Drain the stream(s).
+ * When the substream is linked, sync until the draining of all playback streams
+ * is finished.
+ * After this call, all streams are supposed to be either SETUP or DRAINING
+ * (capture only) state.
+ */
+static int snd_pcm_drain(struct snd_pcm_substream *substream)
+{
+ struct snd_card *card;
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm_substream *s;
+ int result = 0;
+ int i, num_drecs;
+ struct drain_rec *drec, drec_tmp, *d;
+
+ card = substream->pcm->card;
+ runtime = substream->runtime;
+
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+
+ snd_power_lock(card);
+ if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
+ result = snd_power_wait(card, SNDRV_CTL_POWER_D0);
+ if (result < 0) {
+ snd_power_unlock(card);
+ return result;
+ }
+ }
+
+ /* allocate temporary record for drain sync */
+ down_read(&snd_pcm_link_rwsem);
+ if (snd_pcm_stream_linked(substream)) {
+ drec = kmalloc(substream->group->count * sizeof(*drec), GFP_KERNEL);
+ if (! drec) {
+ up_read(&snd_pcm_link_rwsem);
+ snd_power_unlock(card);
+ return -ENOMEM;
+ }
+ } else
+ drec = &drec_tmp;
+
+ /* count only playback streams */
+ num_drecs = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ runtime = s->runtime;
+ if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ d = &drec[num_drecs++];
+ d->substream = s;
+ init_waitqueue_entry(&d->wait, current);
+ add_wait_queue(&runtime->sleep, &d->wait);
+ /* stop_threshold fixup to avoid endless loop when
+ * stop_threshold > buffer_size
+ */
+ d->stop_threshold = runtime->stop_threshold;
+ if (runtime->stop_threshold > runtime->buffer_size)
+ runtime->stop_threshold = runtime->buffer_size;
+ }
+ }
+ up_read(&snd_pcm_link_rwsem);
+
+ snd_pcm_stream_lock_irq(substream);
+ /* resume pause */
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED)
+ snd_pcm_pause(substream, 0);
+
+ /* pre-start/stop - all running streams are changed to DRAINING state */
+ result = snd_pcm_action(&snd_pcm_action_drain_init, substream, 0);
+ if (result < 0) {
+ snd_pcm_stream_unlock_irq(substream);
+ goto _error;
+ }
+
+ for (;;) {
+ long tout;
+ if (signal_pending(current)) {
+ result = -ERESTARTSYS;
+ break;
+ }
+ /* all finished? */
+ for (i = 0; i < num_drecs; i++) {
+ runtime = drec[i].substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_DRAINING)
+ break;
+ }
+ if (i == num_drecs)
+ break; /* yes, all drained */
+
+ set_current_state(TASK_INTERRUPTIBLE);
+ snd_pcm_stream_unlock_irq(substream);
+ snd_power_unlock(card);
+ tout = schedule_timeout(10 * HZ);
+ snd_power_lock(card);
+ snd_pcm_stream_lock_irq(substream);
+ if (tout == 0) {
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_SUSPENDED)
+ result = -ESTRPIPE;
+ else {
+ snd_printd("playback drain error (DMA or IRQ trouble?)\n");
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
+ result = -EIO;
+ }
+ break;
+ }
+ }
+
+ snd_pcm_stream_unlock_irq(substream);
+
+ _error:
+ for (i = 0; i < num_drecs; i++) {
+ d = &drec[i];
+ runtime = d->substream->runtime;
+ remove_wait_queue(&runtime->sleep, &d->wait);
+ runtime->stop_threshold = d->stop_threshold;
+ }
+
+ if (drec != &drec_tmp)
+ kfree(drec);
+ snd_power_unlock(card);
+
+ return result;
+}
+
+/*
+ * drop ioctl
+ *
+ * Immediately put all linked substreams into SETUP state.
+ */
+static int snd_pcm_drop(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime;
+ struct snd_card *card;
+ int result = 0;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ card = substream->pcm->card;
+
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN ||
+ runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED ||
+ runtime->status->state == SNDRV_PCM_STATE_SUSPENDED)
+ return -EBADFD;
+
+ snd_pcm_stream_lock_irq(substream);
+ /* resume pause */
+ if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
+ snd_pcm_pause(substream, 0);
+
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
+ /* runtime->control->appl_ptr = runtime->status->hw_ptr; */
+ snd_pcm_stream_unlock_irq(substream);
+
+ return result;
+}
+
+
+/* WARNING: Don't forget to fput back the file */
+static struct file *snd_pcm_file_fd(int fd)
+{
+ struct file *file;
+ struct inode *inode;
+ unsigned int minor;
+
+ file = fget(fd);
+ if (!file)
+ return NULL;
+ inode = file->f_path.dentry->d_inode;
+ if (!S_ISCHR(inode->i_mode) ||
+ imajor(inode) != snd_major) {
+ fput(file);
+ return NULL;
+ }
+ minor = iminor(inode);
+ if (!snd_lookup_minor_data(minor, SNDRV_DEVICE_TYPE_PCM_PLAYBACK) &&
+ !snd_lookup_minor_data(minor, SNDRV_DEVICE_TYPE_PCM_CAPTURE)) {
+ fput(file);
+ return NULL;
+ }
+ return file;
+}
+
+/*
+ * PCM link handling
+ */
+static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
+{
+ int res = 0;
+ struct file *file;
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream1;
+
+ file = snd_pcm_file_fd(fd);
+ if (!file)
+ return -EBADFD;
+ pcm_file = file->private_data;
+ substream1 = pcm_file->substream;
+ down_write(&snd_pcm_link_rwsem);
+ write_lock_irq(&snd_pcm_link_rwlock);
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
+ substream->runtime->status->state != substream1->runtime->status->state) {
+ res = -EBADFD;
+ goto _end;
+ }
+ if (snd_pcm_stream_linked(substream1)) {
+ res = -EALREADY;
+ goto _end;
+ }
+ if (!snd_pcm_stream_linked(substream)) {
+ substream->group = kmalloc(sizeof(struct snd_pcm_group), GFP_ATOMIC);
+ if (substream->group == NULL) {
+ res = -ENOMEM;
+ goto _end;
+ }
+ spin_lock_init(&substream->group->lock);
+ INIT_LIST_HEAD(&substream->group->substreams);
+ list_add_tail(&substream->link_list, &substream->group->substreams);
+ substream->group->count = 1;
+ }
+ list_add_tail(&substream1->link_list, &substream->group->substreams);
+ substream->group->count++;
+ substream1->group = substream->group;
+ _end:
+ write_unlock_irq(&snd_pcm_link_rwlock);
+ up_write(&snd_pcm_link_rwsem);
+ fput(file);
+ return res;
+}
+
+static void relink_to_local(struct snd_pcm_substream *substream)
+{
+ substream->group = &substream->self_group;
+ INIT_LIST_HEAD(&substream->self_group.substreams);
+ list_add_tail(&substream->link_list, &substream->self_group.substreams);
+}
+
+static int snd_pcm_unlink(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_substream *s;
+ int res = 0;
+
+ down_write(&snd_pcm_link_rwsem);
+ write_lock_irq(&snd_pcm_link_rwlock);
+ if (!snd_pcm_stream_linked(substream)) {
+ res = -EALREADY;
+ goto _end;
+ }
+ list_del(&substream->link_list);
+ substream->group->count--;
+ if (substream->group->count == 1) { /* detach the last stream, too */
+ snd_pcm_group_for_each_entry(s, substream) {
+ relink_to_local(s);
+ break;
+ }
+ kfree(substream->group);
+ }
+ relink_to_local(substream);
+ _end:
+ write_unlock_irq(&snd_pcm_link_rwlock);
+ up_write(&snd_pcm_link_rwsem);
+ return res;
+}
+
+/*
+ * hw configurator
+ */
+static int snd_pcm_hw_rule_mul(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval t;
+ snd_interval_mul(hw_param_interval_c(params, rule->deps[0]),
+ hw_param_interval_c(params, rule->deps[1]), &t);
+ return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+static int snd_pcm_hw_rule_div(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval t;
+ snd_interval_div(hw_param_interval_c(params, rule->deps[0]),
+ hw_param_interval_c(params, rule->deps[1]), &t);
+ return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+static int snd_pcm_hw_rule_muldivk(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval t;
+ snd_interval_muldivk(hw_param_interval_c(params, rule->deps[0]),
+ hw_param_interval_c(params, rule->deps[1]),
+ (unsigned long) rule->private, &t);
+ return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval t;
+ snd_interval_mulkdiv(hw_param_interval_c(params, rule->deps[0]),
+ (unsigned long) rule->private,
+ hw_param_interval_c(params, rule->deps[1]), &t);
+ return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ unsigned int k;
+ struct snd_interval *i = hw_param_interval(params, rule->deps[0]);
+ struct snd_mask m;
+ struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ snd_mask_any(&m);
+ for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+ int bits;
+ if (! snd_mask_test(mask, k))
+ continue;
+ bits = snd_pcm_format_physical_width(k);
+ if (bits <= 0)
+ continue; /* ignore invalid formats */
+ if ((unsigned)bits < i->min || (unsigned)bits > i->max)
+ snd_mask_reset(&m, k);
+ }
+ return snd_mask_refine(mask, &m);
+}
+
+static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval t;
+ unsigned int k;
+ t.min = UINT_MAX;
+ t.max = 0;
+ t.openmin = 0;
+ t.openmax = 0;
+ for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) {
+ int bits;
+ if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k))
+ continue;
+ bits = snd_pcm_format_physical_width(k);
+ if (bits <= 0)
+ continue; /* ignore invalid formats */
+ if (t.min > (unsigned)bits)
+ t.min = bits;
+ if (t.max < (unsigned)bits)
+ t.max = bits;
+ }
+ t.integer = 1;
+ return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12
+#error "Change this table"
+#endif
+
+static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100,
+ 48000, 64000, 88200, 96000, 176400, 192000 };
+
+const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+};
+
+static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pcm_hardware *hw = rule->private;
+ return snd_interval_list(hw_param_interval(params, rule->var),
+ snd_pcm_known_rates.count,
+ snd_pcm_known_rates.list, hw->rates);
+}
+
+static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval t;
+ struct snd_pcm_substream *substream = rule->private;
+ t.min = 0;
+ t.max = substream->buffer_bytes_max;
+ t.openmin = 0;
+ t.openmax = 0;
+ t.integer = 1;
+ return snd_interval_refine(hw_param_interval(params, rule->var), &t);
+}
+
+int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_hw_constraints *constrs = &runtime->hw_constraints;
+ int k, err;
+
+ for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) {
+ snd_mask_any(constrs_mask(constrs, k));
+ }
+
+ for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) {
+ snd_interval_any(constrs_interval(constrs, k));
+ }
+
+ snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_CHANNELS));
+ snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_SIZE));
+ snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_BYTES));
+ snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_SAMPLE_BITS));
+ snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_FRAME_BITS));
+
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ snd_pcm_hw_rule_format, NULL,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ snd_pcm_hw_rule_sample_bits, NULL,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ snd_pcm_hw_rule_div, NULL,
+ SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS,
+ snd_pcm_hw_rule_mul, NULL,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS,
+ snd_pcm_hw_rule_mulkdiv, (void*) 8,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS,
+ snd_pcm_hw_rule_mulkdiv, (void*) 8,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_pcm_hw_rule_div, NULL,
+ SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_pcm_hw_rule_mulkdiv, (void*) 1000000,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_TIME, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_pcm_hw_rule_mulkdiv, (void*) 1000000,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_BUFFER_TIME, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS,
+ snd_pcm_hw_rule_div, NULL,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ snd_pcm_hw_rule_div, NULL,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ snd_pcm_hw_rule_mulkdiv, (void*) 8,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ snd_pcm_hw_rule_muldivk, (void*) 1000000,
+ SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ snd_pcm_hw_rule_mul, NULL,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ snd_pcm_hw_rule_mulkdiv, (void*) 8,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ snd_pcm_hw_rule_muldivk, (void*) 1000000,
+ SNDRV_PCM_HW_PARAM_BUFFER_TIME, SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ snd_pcm_hw_rule_muldivk, (void*) 8,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ snd_pcm_hw_rule_muldivk, (void*) 8,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+ snd_pcm_hw_rule_mulkdiv, (void*) 1000000,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME,
+ snd_pcm_hw_rule_mulkdiv, (void*) 1000000,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_hardware *hw = &runtime->hw;
+ int err;
+ unsigned int mask = 0;
+
+ if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
+ mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED;
+ if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
+ mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED;
+ if (hw->info & SNDRV_PCM_INFO_MMAP) {
+ if (hw->info & SNDRV_PCM_INFO_INTERLEAVED)
+ mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED;
+ if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED)
+ mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED;
+ if (hw->info & SNDRV_PCM_INFO_COMPLEX)
+ mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX;
+ }
+ err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, hw->formats);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw->channels_min, hw->channels_max);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
+ hw->rate_min, hw->rate_max);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ hw->period_bytes_min, hw->period_bytes_max);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS,
+ hw->periods_min, hw->periods_max);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ hw->period_bytes_min, hw->buffer_bytes_max);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ snd_pcm_hw_rule_buffer_bytes_max, substream,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1);
+ if (err < 0)
+ return err;
+
+ /* FIXME: remove */
+ if (runtime->dma_bytes) {
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, runtime->dma_bytes);
+ if (err < 0)
+ return -EINVAL;
+ }
+
+ if (!(hw->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))) {
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_pcm_hw_rule_rate, hw,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+ }
+
+ /* FIXME: this belong to lowlevel */
+ snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+
+ return 0;
+}
+
+static void pcm_release_private(struct snd_pcm_substream *substream)
+{
+ snd_pcm_unlink(substream);
+}
+
+void snd_pcm_release_substream(struct snd_pcm_substream *substream)
+{
+ substream->ref_count--;
+ if (substream->ref_count > 0)
+ return;
+
+ snd_pcm_drop(substream);
+ if (substream->hw_opened) {
+ if (substream->ops->hw_free != NULL)
+ substream->ops->hw_free(substream);
+ substream->ops->close(substream);
+ substream->hw_opened = 0;
+ }
+ if (substream->pcm_release) {
+ substream->pcm_release(substream);
+ substream->pcm_release = NULL;
+ }
+ snd_pcm_detach_substream(substream);
+}
+
+EXPORT_SYMBOL(snd_pcm_release_substream);
+
+int snd_pcm_open_substream(struct snd_pcm *pcm, int stream,
+ struct file *file,
+ struct snd_pcm_substream **rsubstream)
+{
+ struct snd_pcm_substream *substream;
+ int err;
+
+ err = snd_pcm_attach_substream(pcm, stream, file, &substream);
+ if (err < 0)
+ return err;
+ if (substream->ref_count > 1) {
+ *rsubstream = substream;
+ return 0;
+ }
+
+ err = snd_pcm_hw_constraints_init(substream);
+ if (err < 0) {
+ snd_printd("snd_pcm_hw_constraints_init failed\n");
+ goto error;
+ }
+
+ if ((err = substream->ops->open(substream)) < 0)
+ goto error;
+
+ substream->hw_opened = 1;
+
+ err = snd_pcm_hw_constraints_complete(substream);
+ if (err < 0) {
+ snd_printd("snd_pcm_hw_constraints_complete failed\n");
+ goto error;
+ }
+
+ *rsubstream = substream;
+ return 0;
+
+ error:
+ snd_pcm_release_substream(substream);
+ return err;
+}
+
+EXPORT_SYMBOL(snd_pcm_open_substream);
+
+static int snd_pcm_open_file(struct file *file,
+ struct snd_pcm *pcm,
+ int stream,
+ struct snd_pcm_file **rpcm_file)
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_str *str;
+ int err;
+
+ if (rpcm_file)
+ *rpcm_file = NULL;
+
+ err = snd_pcm_open_substream(pcm, stream, file, &substream);
+ if (err < 0)
+ return err;
+
+ pcm_file = kzalloc(sizeof(*pcm_file), GFP_KERNEL);
+ if (pcm_file == NULL) {
+ snd_pcm_release_substream(substream);
+ return -ENOMEM;
+ }
+ pcm_file->substream = substream;
+ if (substream->ref_count == 1) {
+ str = substream->pstr;
+ substream->file = pcm_file;
+ substream->pcm_release = pcm_release_private;
+ }
+ file->private_data = pcm_file;
+ if (rpcm_file)
+ *rpcm_file = pcm_file;
+ return 0;
+}
+
+static int snd_pcm_playback_open(struct inode *inode, struct file *file)
+{
+ struct snd_pcm *pcm;
+
+ pcm = snd_lookup_minor_data(iminor(inode),
+ SNDRV_DEVICE_TYPE_PCM_PLAYBACK);
+ return snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK);
+}
+
+static int snd_pcm_capture_open(struct inode *inode, struct file *file)
+{
+ struct snd_pcm *pcm;
+
+ pcm = snd_lookup_minor_data(iminor(inode),
+ SNDRV_DEVICE_TYPE_PCM_CAPTURE);
+ return snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE);
+}
+
+static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream)
+{
+ int err;
+ struct snd_pcm_file *pcm_file;
+ wait_queue_t wait;
+
+ if (pcm == NULL) {
+ err = -ENODEV;
+ goto __error1;
+ }
+ err = snd_card_file_add(pcm->card, file);
+ if (err < 0)
+ goto __error1;
+ if (!try_module_get(pcm->card->module)) {
+ err = -EFAULT;
+ goto __error2;
+ }
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&pcm->open_wait, &wait);
+ mutex_lock(&pcm->open_mutex);
+ while (1) {
+ err = snd_pcm_open_file(file, pcm, stream, &pcm_file);
+ if (err >= 0)
+ break;
+ if (err == -EAGAIN) {
+ if (file->f_flags & O_NONBLOCK) {
+ err = -EBUSY;
+ break;
+ }
+ } else
+ break;
+ set_current_state(TASK_INTERRUPTIBLE);
+ mutex_unlock(&pcm->open_mutex);
+ schedule();
+ mutex_lock(&pcm->open_mutex);
+ if (signal_pending(current)) {
+ err = -ERESTARTSYS;
+ break;
+ }
+ }
+ remove_wait_queue(&pcm->open_wait, &wait);
+ mutex_unlock(&pcm->open_mutex);
+ if (err < 0)
+ goto __error;
+ return err;
+
+ __error:
+ module_put(pcm->card->module);
+ __error2:
+ snd_card_file_remove(pcm->card, file);
+ __error1:
+ return err;
+}
+
+static int snd_pcm_release(struct inode *inode, struct file *file)
+{
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_file *pcm_file;
+
+ pcm_file = file->private_data;
+ substream = pcm_file->substream;
+ if (snd_BUG_ON(!substream))
+ return -ENXIO;
+ pcm = substream->pcm;
+ mutex_lock(&pcm->open_mutex);
+ snd_pcm_release_substream(substream);
+ kfree(pcm_file);
+ mutex_unlock(&pcm->open_mutex);
+ wake_up(&pcm->open_wait);
+ module_put(pcm->card->module);
+ snd_card_file_remove(pcm->card, file);
+ return 0;
+}
+
+static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t appl_ptr;
+ snd_pcm_sframes_t ret;
+ snd_pcm_sframes_t hw_avail;
+
+ if (frames == 0)
+ return 0;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_PREPARED:
+ break;
+ case SNDRV_PCM_STATE_DRAINING:
+ case SNDRV_PCM_STATE_RUNNING:
+ if (snd_pcm_update_hw_ptr(substream) >= 0)
+ break;
+ /* Fall through */
+ case SNDRV_PCM_STATE_XRUN:
+ ret = -EPIPE;
+ goto __end;
+ default:
+ ret = -EBADFD;
+ goto __end;
+ }
+
+ hw_avail = snd_pcm_playback_hw_avail(runtime);
+ if (hw_avail <= 0) {
+ ret = 0;
+ goto __end;
+ }
+ if (frames > (snd_pcm_uframes_t)hw_avail)
+ frames = hw_avail;
+ appl_ptr = runtime->control->appl_ptr - frames;
+ if (appl_ptr < 0)
+ appl_ptr += runtime->boundary;
+ runtime->control->appl_ptr = appl_ptr;
+ ret = frames;
+ __end:
+ snd_pcm_stream_unlock_irq(substream);
+ return ret;
+}
+
+static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t appl_ptr;
+ snd_pcm_sframes_t ret;
+ snd_pcm_sframes_t hw_avail;
+
+ if (frames == 0)
+ return 0;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_DRAINING:
+ break;
+ case SNDRV_PCM_STATE_RUNNING:
+ if (snd_pcm_update_hw_ptr(substream) >= 0)
+ break;
+ /* Fall through */
+ case SNDRV_PCM_STATE_XRUN:
+ ret = -EPIPE;
+ goto __end;
+ default:
+ ret = -EBADFD;
+ goto __end;
+ }
+
+ hw_avail = snd_pcm_capture_hw_avail(runtime);
+ if (hw_avail <= 0) {
+ ret = 0;
+ goto __end;
+ }
+ if (frames > (snd_pcm_uframes_t)hw_avail)
+ frames = hw_avail;
+ appl_ptr = runtime->control->appl_ptr - frames;
+ if (appl_ptr < 0)
+ appl_ptr += runtime->boundary;
+ runtime->control->appl_ptr = appl_ptr;
+ ret = frames;
+ __end:
+ snd_pcm_stream_unlock_irq(substream);
+ return ret;
+}
+
+static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t appl_ptr;
+ snd_pcm_sframes_t ret;
+ snd_pcm_sframes_t avail;
+
+ if (frames == 0)
+ return 0;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
+ break;
+ case SNDRV_PCM_STATE_DRAINING:
+ case SNDRV_PCM_STATE_RUNNING:
+ if (snd_pcm_update_hw_ptr(substream) >= 0)
+ break;
+ /* Fall through */
+ case SNDRV_PCM_STATE_XRUN:
+ ret = -EPIPE;
+ goto __end;
+ default:
+ ret = -EBADFD;
+ goto __end;
+ }
+
+ avail = snd_pcm_playback_avail(runtime);
+ if (avail <= 0) {
+ ret = 0;
+ goto __end;
+ }
+ if (frames > (snd_pcm_uframes_t)avail)
+ frames = avail;
+ appl_ptr = runtime->control->appl_ptr + frames;
+ if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary)
+ appl_ptr -= runtime->boundary;
+ runtime->control->appl_ptr = appl_ptr;
+ ret = frames;
+ __end:
+ snd_pcm_stream_unlock_irq(substream);
+ return ret;
+}
+
+static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t frames)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t appl_ptr;
+ snd_pcm_sframes_t ret;
+ snd_pcm_sframes_t avail;
+
+ if (frames == 0)
+ return 0;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_DRAINING:
+ case SNDRV_PCM_STATE_PAUSED:
+ break;
+ case SNDRV_PCM_STATE_RUNNING:
+ if (snd_pcm_update_hw_ptr(substream) >= 0)
+ break;
+ /* Fall through */
+ case SNDRV_PCM_STATE_XRUN:
+ ret = -EPIPE;
+ goto __end;
+ default:
+ ret = -EBADFD;
+ goto __end;
+ }
+
+ avail = snd_pcm_capture_avail(runtime);
+ if (avail <= 0) {
+ ret = 0;
+ goto __end;
+ }
+ if (frames > (snd_pcm_uframes_t)avail)
+ frames = avail;
+ appl_ptr = runtime->control->appl_ptr + frames;
+ if (appl_ptr >= (snd_pcm_sframes_t)runtime->boundary)
+ appl_ptr -= runtime->boundary;
+ runtime->control->appl_ptr = appl_ptr;
+ ret = frames;
+ __end:
+ snd_pcm_stream_unlock_irq(substream);
+ return ret;
+}
+
+static int snd_pcm_hwsync(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_DRAINING:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ goto __badfd;
+ case SNDRV_PCM_STATE_RUNNING:
+ if ((err = snd_pcm_update_hw_ptr(substream)) < 0)
+ break;
+ /* Fall through */
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_SUSPENDED:
+ err = 0;
+ break;
+ case SNDRV_PCM_STATE_XRUN:
+ err = -EPIPE;
+ break;
+ default:
+ __badfd:
+ err = -EBADFD;
+ break;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ return err;
+}
+
+static int snd_pcm_delay(struct snd_pcm_substream *substream,
+ snd_pcm_sframes_t __user *res)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ snd_pcm_sframes_t n = 0;
+
+ snd_pcm_stream_lock_irq(substream);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_DRAINING:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ goto __badfd;
+ case SNDRV_PCM_STATE_RUNNING:
+ if ((err = snd_pcm_update_hw_ptr(substream)) < 0)
+ break;
+ /* Fall through */
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_SUSPENDED:
+ err = 0;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ n = snd_pcm_playback_hw_avail(runtime);
+ else
+ n = snd_pcm_capture_avail(runtime);
+ break;
+ case SNDRV_PCM_STATE_XRUN:
+ err = -EPIPE;
+ break;
+ default:
+ __badfd:
+ err = -EBADFD;
+ break;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ if (!err)
+ if (put_user(n, res))
+ err = -EFAULT;
+ return err;
+}
+
+static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream,
+ struct snd_pcm_sync_ptr __user *_sync_ptr)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_pcm_sync_ptr sync_ptr;
+ volatile struct snd_pcm_mmap_status *status;
+ volatile struct snd_pcm_mmap_control *control;
+ int err;
+
+ memset(&sync_ptr, 0, sizeof(sync_ptr));
+ if (get_user(sync_ptr.flags, (unsigned __user *)&(_sync_ptr->flags)))
+ return -EFAULT;
+ if (copy_from_user(&sync_ptr.c.control, &(_sync_ptr->c.control), sizeof(struct snd_pcm_mmap_control)))
+ return -EFAULT;
+ status = runtime->status;
+ control = runtime->control;
+ if (sync_ptr.flags & SNDRV_PCM_SYNC_PTR_HWSYNC) {
+ err = snd_pcm_hwsync(substream);
+ if (err < 0)
+ return err;
+ }
+ snd_pcm_stream_lock_irq(substream);
+ if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL))
+ control->appl_ptr = sync_ptr.c.control.appl_ptr;
+ else
+ sync_ptr.c.control.appl_ptr = control->appl_ptr;
+ if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN))
+ control->avail_min = sync_ptr.c.control.avail_min;
+ else
+ sync_ptr.c.control.avail_min = control->avail_min;
+ sync_ptr.s.status.state = status->state;
+ sync_ptr.s.status.hw_ptr = status->hw_ptr;
+ sync_ptr.s.status.tstamp = status->tstamp;
+ sync_ptr.s.status.suspended_state = status->suspended_state;
+ snd_pcm_stream_unlock_irq(substream);
+ if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int arg;
+
+ if (get_user(arg, _arg))
+ return -EFAULT;
+ if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST)
+ return -EINVAL;
+ runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY;
+ if (arg == SNDRV_PCM_TSTAMP_TYPE_MONOTONIC)
+ runtime->tstamp_type = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC;
+ return 0;
+}
+
+static int snd_pcm_common_ioctl1(struct file *file,
+ struct snd_pcm_substream *substream,
+ unsigned int cmd, void __user *arg)
+{
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL_PVERSION:
+ return put_user(SNDRV_PCM_VERSION, (int __user *)arg) ? -EFAULT : 0;
+ case SNDRV_PCM_IOCTL_INFO:
+ return snd_pcm_info_user(substream, arg);
+ case SNDRV_PCM_IOCTL_TSTAMP: /* just for compatibility */
+ return 0;
+ case SNDRV_PCM_IOCTL_TTSTAMP:
+ return snd_pcm_tstamp(substream, arg);
+ case SNDRV_PCM_IOCTL_HW_REFINE:
+ return snd_pcm_hw_refine_user(substream, arg);
+ case SNDRV_PCM_IOCTL_HW_PARAMS:
+ return snd_pcm_hw_params_user(substream, arg);
+ case SNDRV_PCM_IOCTL_HW_FREE:
+ return snd_pcm_hw_free(substream);
+ case SNDRV_PCM_IOCTL_SW_PARAMS:
+ return snd_pcm_sw_params_user(substream, arg);
+ case SNDRV_PCM_IOCTL_STATUS:
+ return snd_pcm_status_user(substream, arg);
+ case SNDRV_PCM_IOCTL_CHANNEL_INFO:
+ return snd_pcm_channel_info_user(substream, arg);
+ case SNDRV_PCM_IOCTL_PREPARE:
+ return snd_pcm_prepare(substream, file);
+ case SNDRV_PCM_IOCTL_RESET:
+ return snd_pcm_reset(substream);
+ case SNDRV_PCM_IOCTL_START:
+ return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING);
+ case SNDRV_PCM_IOCTL_LINK:
+ return snd_pcm_link(substream, (int)(unsigned long) arg);
+ case SNDRV_PCM_IOCTL_UNLINK:
+ return snd_pcm_unlink(substream);
+ case SNDRV_PCM_IOCTL_RESUME:
+ return snd_pcm_resume(substream);
+ case SNDRV_PCM_IOCTL_XRUN:
+ return snd_pcm_xrun(substream);
+ case SNDRV_PCM_IOCTL_HWSYNC:
+ return snd_pcm_hwsync(substream);
+ case SNDRV_PCM_IOCTL_DELAY:
+ return snd_pcm_delay(substream, arg);
+ case SNDRV_PCM_IOCTL_SYNC_PTR:
+ return snd_pcm_sync_ptr(substream, arg);
+#ifdef CONFIG_SND_SUPPORT_OLD_API
+ case SNDRV_PCM_IOCTL_HW_REFINE_OLD:
+ return snd_pcm_hw_refine_old_user(substream, arg);
+ case SNDRV_PCM_IOCTL_HW_PARAMS_OLD:
+ return snd_pcm_hw_params_old_user(substream, arg);
+#endif
+ case SNDRV_PCM_IOCTL_DRAIN:
+ return snd_pcm_drain(substream);
+ case SNDRV_PCM_IOCTL_DROP:
+ return snd_pcm_drop(substream);
+ case SNDRV_PCM_IOCTL_PAUSE:
+ {
+ int res;
+ snd_pcm_stream_lock_irq(substream);
+ res = snd_pcm_pause(substream, (int)(unsigned long)arg);
+ snd_pcm_stream_unlock_irq(substream);
+ return res;
+ }
+ }
+ snd_printd("unknown ioctl = 0x%x\n", cmd);
+ return -ENOTTY;
+}
+
+static int snd_pcm_playback_ioctl1(struct file *file,
+ struct snd_pcm_substream *substream,
+ unsigned int cmd, void __user *arg)
+{
+ if (snd_BUG_ON(!substream))
+ return -ENXIO;
+ if (snd_BUG_ON(substream->stream != SNDRV_PCM_STREAM_PLAYBACK))
+ return -EINVAL;
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL_WRITEI_FRAMES:
+ {
+ struct snd_xferi xferi;
+ struct snd_xferi __user *_xferi = arg;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t result;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (put_user(0, &_xferi->result))
+ return -EFAULT;
+ if (copy_from_user(&xferi, _xferi, sizeof(xferi)))
+ return -EFAULT;
+ result = snd_pcm_lib_write(substream, xferi.buf, xferi.frames);
+ __put_user(result, &_xferi->result);
+ return result < 0 ? result : 0;
+ }
+ case SNDRV_PCM_IOCTL_WRITEN_FRAMES:
+ {
+ struct snd_xfern xfern;
+ struct snd_xfern __user *_xfern = arg;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ void __user **bufs;
+ snd_pcm_sframes_t result;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (runtime->channels > 128)
+ return -EINVAL;
+ if (put_user(0, &_xfern->result))
+ return -EFAULT;
+ if (copy_from_user(&xfern, _xfern, sizeof(xfern)))
+ return -EFAULT;
+ bufs = kmalloc(sizeof(void *) * runtime->channels, GFP_KERNEL);
+ if (bufs == NULL)
+ return -ENOMEM;
+ if (copy_from_user(bufs, xfern.bufs, sizeof(void *) * runtime->channels)) {
+ kfree(bufs);
+ return -EFAULT;
+ }
+ result = snd_pcm_lib_writev(substream, bufs, xfern.frames);
+ kfree(bufs);
+ __put_user(result, &_xfern->result);
+ return result < 0 ? result : 0;
+ }
+ case SNDRV_PCM_IOCTL_REWIND:
+ {
+ snd_pcm_uframes_t frames;
+ snd_pcm_uframes_t __user *_frames = arg;
+ snd_pcm_sframes_t result;
+ if (get_user(frames, _frames))
+ return -EFAULT;
+ if (put_user(0, _frames))
+ return -EFAULT;
+ result = snd_pcm_playback_rewind(substream, frames);
+ __put_user(result, _frames);
+ return result < 0 ? result : 0;
+ }
+ case SNDRV_PCM_IOCTL_FORWARD:
+ {
+ snd_pcm_uframes_t frames;
+ snd_pcm_uframes_t __user *_frames = arg;
+ snd_pcm_sframes_t result;
+ if (get_user(frames, _frames))
+ return -EFAULT;
+ if (put_user(0, _frames))
+ return -EFAULT;
+ result = snd_pcm_playback_forward(substream, frames);
+ __put_user(result, _frames);
+ return result < 0 ? result : 0;
+ }
+ }
+ return snd_pcm_common_ioctl1(file, substream, cmd, arg);
+}
+
+static int snd_pcm_capture_ioctl1(struct file *file,
+ struct snd_pcm_substream *substream,
+ unsigned int cmd, void __user *arg)
+{
+ if (snd_BUG_ON(!substream))
+ return -ENXIO;
+ if (snd_BUG_ON(substream->stream != SNDRV_PCM_STREAM_CAPTURE))
+ return -EINVAL;
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL_READI_FRAMES:
+ {
+ struct snd_xferi xferi;
+ struct snd_xferi __user *_xferi = arg;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_sframes_t result;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (put_user(0, &_xferi->result))
+ return -EFAULT;
+ if (copy_from_user(&xferi, _xferi, sizeof(xferi)))
+ return -EFAULT;
+ result = snd_pcm_lib_read(substream, xferi.buf, xferi.frames);
+ __put_user(result, &_xferi->result);
+ return result < 0 ? result : 0;
+ }
+ case SNDRV_PCM_IOCTL_READN_FRAMES:
+ {
+ struct snd_xfern xfern;
+ struct snd_xfern __user *_xfern = arg;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ void *bufs;
+ snd_pcm_sframes_t result;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (runtime->channels > 128)
+ return -EINVAL;
+ if (put_user(0, &_xfern->result))
+ return -EFAULT;
+ if (copy_from_user(&xfern, _xfern, sizeof(xfern)))
+ return -EFAULT;
+ bufs = kmalloc(sizeof(void *) * runtime->channels, GFP_KERNEL);
+ if (bufs == NULL)
+ return -ENOMEM;
+ if (copy_from_user(bufs, xfern.bufs, sizeof(void *) * runtime->channels)) {
+ kfree(bufs);
+ return -EFAULT;
+ }
+ result = snd_pcm_lib_readv(substream, bufs, xfern.frames);
+ kfree(bufs);
+ __put_user(result, &_xfern->result);
+ return result < 0 ? result : 0;
+ }
+ case SNDRV_PCM_IOCTL_REWIND:
+ {
+ snd_pcm_uframes_t frames;
+ snd_pcm_uframes_t __user *_frames = arg;
+ snd_pcm_sframes_t result;
+ if (get_user(frames, _frames))
+ return -EFAULT;
+ if (put_user(0, _frames))
+ return -EFAULT;
+ result = snd_pcm_capture_rewind(substream, frames);
+ __put_user(result, _frames);
+ return result < 0 ? result : 0;
+ }
+ case SNDRV_PCM_IOCTL_FORWARD:
+ {
+ snd_pcm_uframes_t frames;
+ snd_pcm_uframes_t __user *_frames = arg;
+ snd_pcm_sframes_t result;
+ if (get_user(frames, _frames))
+ return -EFAULT;
+ if (put_user(0, _frames))
+ return -EFAULT;
+ result = snd_pcm_capture_forward(substream, frames);
+ __put_user(result, _frames);
+ return result < 0 ? result : 0;
+ }
+ }
+ return snd_pcm_common_ioctl1(file, substream, cmd, arg);
+}
+
+static long snd_pcm_playback_ioctl(struct file *file, unsigned int cmd,
+ unsigned long arg)
+{
+ struct snd_pcm_file *pcm_file;
+
+ pcm_file = file->private_data;
+
+ if (((cmd >> 8) & 0xff) != 'A')
+ return -ENOTTY;
+
+ return snd_pcm_playback_ioctl1(file, pcm_file->substream, cmd,
+ (void __user *)arg);
+}
+
+static long snd_pcm_capture_ioctl(struct file *file, unsigned int cmd,
+ unsigned long arg)
+{
+ struct snd_pcm_file *pcm_file;
+
+ pcm_file = file->private_data;
+
+ if (((cmd >> 8) & 0xff) != 'A')
+ return -ENOTTY;
+
+ return snd_pcm_capture_ioctl1(file, pcm_file->substream, cmd,
+ (void __user *)arg);
+}
+
+int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ mm_segment_t fs;
+ int result;
+
+ fs = snd_enter_user();
+ switch (substream->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ result = snd_pcm_playback_ioctl1(NULL, substream, cmd,
+ (void __user *)arg);
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ result = snd_pcm_capture_ioctl1(NULL, substream, cmd,
+ (void __user *)arg);
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ snd_leave_user(fs);
+ return result;
+}
+
+EXPORT_SYMBOL(snd_pcm_kernel_ioctl);
+
+static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count,
+ loff_t * offset)
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_sframes_t result;
+
+ pcm_file = file->private_data;
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (!frame_aligned(runtime, count))
+ return -EINVAL;
+ count = bytes_to_frames(runtime, count);
+ result = snd_pcm_lib_read(substream, buf, count);
+ if (result > 0)
+ result = frames_to_bytes(runtime, result);
+ return result;
+}
+
+static ssize_t snd_pcm_write(struct file *file, const char __user *buf,
+ size_t count, loff_t * offset)
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_sframes_t result;
+
+ pcm_file = file->private_data;
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (!frame_aligned(runtime, count))
+ return -EINVAL;
+ count = bytes_to_frames(runtime, count);
+ result = snd_pcm_lib_write(substream, buf, count);
+ if (result > 0)
+ result = frames_to_bytes(runtime, result);
+ return result;
+}
+
+static ssize_t snd_pcm_aio_read(struct kiocb *iocb, const struct iovec *iov,
+ unsigned long nr_segs, loff_t pos)
+
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_sframes_t result;
+ unsigned long i;
+ void __user **bufs;
+ snd_pcm_uframes_t frames;
+
+ pcm_file = iocb->ki_filp->private_data;
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (nr_segs > 1024 || nr_segs != runtime->channels)
+ return -EINVAL;
+ if (!frame_aligned(runtime, iov->iov_len))
+ return -EINVAL;
+ frames = bytes_to_samples(runtime, iov->iov_len);
+ bufs = kmalloc(sizeof(void *) * nr_segs, GFP_KERNEL);
+ if (bufs == NULL)
+ return -ENOMEM;
+ for (i = 0; i < nr_segs; ++i)
+ bufs[i] = iov[i].iov_base;
+ result = snd_pcm_lib_readv(substream, bufs, frames);
+ if (result > 0)
+ result = frames_to_bytes(runtime, result);
+ kfree(bufs);
+ return result;
+}
+
+static ssize_t snd_pcm_aio_write(struct kiocb *iocb, const struct iovec *iov,
+ unsigned long nr_segs, loff_t pos)
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_sframes_t result;
+ unsigned long i;
+ void __user **bufs;
+ snd_pcm_uframes_t frames;
+
+ pcm_file = iocb->ki_filp->private_data;
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (nr_segs > 128 || nr_segs != runtime->channels ||
+ !frame_aligned(runtime, iov->iov_len))
+ return -EINVAL;
+ frames = bytes_to_samples(runtime, iov->iov_len);
+ bufs = kmalloc(sizeof(void *) * nr_segs, GFP_KERNEL);
+ if (bufs == NULL)
+ return -ENOMEM;
+ for (i = 0; i < nr_segs; ++i)
+ bufs[i] = iov[i].iov_base;
+ result = snd_pcm_lib_writev(substream, bufs, frames);
+ if (result > 0)
+ result = frames_to_bytes(runtime, result);
+ kfree(bufs);
+ return result;
+}
+
+static unsigned int snd_pcm_playback_poll(struct file *file, poll_table * wait)
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ unsigned int mask;
+ snd_pcm_uframes_t avail;
+
+ pcm_file = file->private_data;
+
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+
+ poll_wait(file, &runtime->sleep, wait);
+
+ snd_pcm_stream_lock_irq(substream);
+ avail = snd_pcm_playback_avail(runtime);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_RUNNING:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
+ if (avail >= runtime->control->avail_min) {
+ mask = POLLOUT | POLLWRNORM;
+ break;
+ }
+ /* Fall through */
+ case SNDRV_PCM_STATE_DRAINING:
+ mask = 0;
+ break;
+ default:
+ mask = POLLOUT | POLLWRNORM | POLLERR;
+ break;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ return mask;
+}
+
+static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait)
+{
+ struct snd_pcm_file *pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ unsigned int mask;
+ snd_pcm_uframes_t avail;
+
+ pcm_file = file->private_data;
+
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+
+ poll_wait(file, &runtime->sleep, wait);
+
+ snd_pcm_stream_lock_irq(substream);
+ avail = snd_pcm_capture_avail(runtime);
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_RUNNING:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
+ if (avail >= runtime->control->avail_min) {
+ mask = POLLIN | POLLRDNORM;
+ break;
+ }
+ mask = 0;
+ break;
+ case SNDRV_PCM_STATE_DRAINING:
+ if (avail > 0) {
+ mask = POLLIN | POLLRDNORM;
+ break;
+ }
+ /* Fall through */
+ default:
+ mask = POLLIN | POLLRDNORM | POLLERR;
+ break;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+ return mask;
+}
+
+/*
+ * mmap support
+ */
+
+/*
+ * Only on coherent architectures, we can mmap the status and the control records
+ * for effcient data transfer. On others, we have to use HWSYNC ioctl...
+ */
+#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA)
+/*
+ * mmap status record
+ */
+static int snd_pcm_mmap_status_fault(struct vm_area_struct *area,
+ struct vm_fault *vmf)
+{
+ struct snd_pcm_substream *substream = area->vm_private_data;
+ struct snd_pcm_runtime *runtime;
+
+ if (substream == NULL)
+ return VM_FAULT_SIGBUS;
+ runtime = substream->runtime;
+ vmf->page = virt_to_page(runtime->status);
+ get_page(vmf->page);
+ return 0;
+}
+
+static struct vm_operations_struct snd_pcm_vm_ops_status =
+{
+ .fault = snd_pcm_mmap_status_fault,
+};
+
+static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file,
+ struct vm_area_struct *area)
+{
+ struct snd_pcm_runtime *runtime;
+ long size;
+ if (!(area->vm_flags & VM_READ))
+ return -EINVAL;
+ runtime = substream->runtime;
+ size = area->vm_end - area->vm_start;
+ if (size != PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status)))
+ return -EINVAL;
+ area->vm_ops = &snd_pcm_vm_ops_status;
+ area->vm_private_data = substream;
+ area->vm_flags |= VM_RESERVED;
+ return 0;
+}
+
+/*
+ * mmap control record
+ */
+static int snd_pcm_mmap_control_fault(struct vm_area_struct *area,
+ struct vm_fault *vmf)
+{
+ struct snd_pcm_substream *substream = area->vm_private_data;
+ struct snd_pcm_runtime *runtime;
+
+ if (substream == NULL)
+ return VM_FAULT_SIGBUS;
+ runtime = substream->runtime;
+ vmf->page = virt_to_page(runtime->control);
+ get_page(vmf->page);
+ return 0;
+}
+
+static struct vm_operations_struct snd_pcm_vm_ops_control =
+{
+ .fault = snd_pcm_mmap_control_fault,
+};
+
+static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file,
+ struct vm_area_struct *area)
+{
+ struct snd_pcm_runtime *runtime;
+ long size;
+ if (!(area->vm_flags & VM_READ))
+ return -EINVAL;
+ runtime = substream->runtime;
+ size = area->vm_end - area->vm_start;
+ if (size != PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)))
+ return -EINVAL;
+ area->vm_ops = &snd_pcm_vm_ops_control;
+ area->vm_private_data = substream;
+ area->vm_flags |= VM_RESERVED;
+ return 0;
+}
+#else /* ! coherent mmap */
+/*
+ * don't support mmap for status and control records.
+ */
+static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file,
+ struct vm_area_struct *area)
+{
+ return -ENXIO;
+}
+static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file,
+ struct vm_area_struct *area)
+{
+ return -ENXIO;
+}
+#endif /* coherent mmap */
+
+/*
+ * fault callback for mmapping a RAM page
+ */
+static int snd_pcm_mmap_data_fault(struct vm_area_struct *area,
+ struct vm_fault *vmf)
+{
+ struct snd_pcm_substream *substream = area->vm_private_data;
+ struct snd_pcm_runtime *runtime;
+ unsigned long offset;
+ struct page * page;
+ void *vaddr;
+ size_t dma_bytes;
+
+ if (substream == NULL)
+ return VM_FAULT_SIGBUS;
+ runtime = substream->runtime;
+ offset = vmf->pgoff << PAGE_SHIFT;
+ dma_bytes = PAGE_ALIGN(runtime->dma_bytes);
+ if (offset > dma_bytes - PAGE_SIZE)
+ return VM_FAULT_SIGBUS;
+ if (substream->ops->page) {
+ page = substream->ops->page(substream, offset);
+ if (!page)
+ return VM_FAULT_SIGBUS;
+ } else {
+ vaddr = runtime->dma_area + offset;
+ page = virt_to_page(vaddr);
+ }
+ get_page(page);
+ vmf->page = page;
+ return 0;
+}
+
+static struct vm_operations_struct snd_pcm_vm_ops_data =
+{
+ .open = snd_pcm_mmap_data_open,
+ .close = snd_pcm_mmap_data_close,
+ .fault = snd_pcm_mmap_data_fault,
+};
+
+/*
+ * mmap the DMA buffer on RAM
+ */
+static int snd_pcm_default_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area)
+{
+ area->vm_ops = &snd_pcm_vm_ops_data;
+ area->vm_private_data = substream;
+ area->vm_flags |= VM_RESERVED;
+ atomic_inc(&substream->mmap_count);
+ return 0;
+}
+
+/*
+ * mmap the DMA buffer on I/O memory area
+ */
+#if SNDRV_PCM_INFO_MMAP_IOMEM
+static struct vm_operations_struct snd_pcm_vm_ops_data_mmio =
+{
+ .open = snd_pcm_mmap_data_open,
+ .close = snd_pcm_mmap_data_close,
+};
+
+int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area)
+{
+ long size;
+ unsigned long offset;
+
+#ifdef pgprot_noncached
+ area->vm_page_prot = pgprot_noncached(area->vm_page_prot);
+#endif
+ area->vm_ops = &snd_pcm_vm_ops_data_mmio;
+ area->vm_private_data = substream;
+ area->vm_flags |= VM_IO;
+ size = area->vm_end - area->vm_start;
+ offset = area->vm_pgoff << PAGE_SHIFT;
+ if (io_remap_pfn_range(area, area->vm_start,
+ (substream->runtime->dma_addr + offset) >> PAGE_SHIFT,
+ size, area->vm_page_prot))
+ return -EAGAIN;
+ atomic_inc(&substream->mmap_count);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem);
+#endif /* SNDRV_PCM_INFO_MMAP */
+
+/*
+ * mmap DMA buffer
+ */
+int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file,
+ struct vm_area_struct *area)
+{
+ struct snd_pcm_runtime *runtime;
+ long size;
+ unsigned long offset;
+ size_t dma_bytes;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (!(area->vm_flags & (VM_WRITE|VM_READ)))
+ return -EINVAL;
+ } else {
+ if (!(area->vm_flags & VM_READ))
+ return -EINVAL;
+ }
+ runtime = substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
+ return -EBADFD;
+ if (!(runtime->info & SNDRV_PCM_INFO_MMAP))
+ return -ENXIO;
+ if (runtime->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED ||
+ runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED)
+ return -EINVAL;
+ size = area->vm_end - area->vm_start;
+ offset = area->vm_pgoff << PAGE_SHIFT;
+ dma_bytes = PAGE_ALIGN(runtime->dma_bytes);
+ if ((size_t)size > dma_bytes)
+ return -EINVAL;
+ if (offset > dma_bytes - size)
+ return -EINVAL;
+
+ if (substream->ops->mmap)
+ return substream->ops->mmap(substream, area);
+ else
+ return snd_pcm_default_mmap(substream, area);
+}
+
+EXPORT_SYMBOL(snd_pcm_mmap_data);
+
+static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area)
+{
+ struct snd_pcm_file * pcm_file;
+ struct snd_pcm_substream *substream;
+ unsigned long offset;
+
+ pcm_file = file->private_data;
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+
+ offset = area->vm_pgoff << PAGE_SHIFT;
+ switch (offset) {
+ case SNDRV_PCM_MMAP_OFFSET_STATUS:
+ if (pcm_file->no_compat_mmap)
+ return -ENXIO;
+ return snd_pcm_mmap_status(substream, file, area);
+ case SNDRV_PCM_MMAP_OFFSET_CONTROL:
+ if (pcm_file->no_compat_mmap)
+ return -ENXIO;
+ return snd_pcm_mmap_control(substream, file, area);
+ default:
+ return snd_pcm_mmap_data(substream, file, area);
+ }
+ return 0;
+}
+
+static int snd_pcm_fasync(int fd, struct file * file, int on)
+{
+ struct snd_pcm_file * pcm_file;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ int err = -ENXIO;
+
+ lock_kernel();
+ pcm_file = file->private_data;
+ substream = pcm_file->substream;
+ if (PCM_RUNTIME_CHECK(substream))
+ goto out;
+ runtime = substream->runtime;
+ err = fasync_helper(fd, file, on, &runtime->fasync);
+out:
+ unlock_kernel();
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * ioctl32 compat
+ */
+#ifdef CONFIG_COMPAT
+#include "pcm_compat.c"
+#else
+#define snd_pcm_ioctl_compat NULL
+#endif
+
+/*
+ * To be removed helpers to keep binary compatibility
+ */
+
+#ifdef CONFIG_SND_SUPPORT_OLD_API
+#define __OLD_TO_NEW_MASK(x) ((x&7)|((x&0x07fffff8)<<5))
+#define __NEW_TO_OLD_MASK(x) ((x&7)|((x&0xffffff00)>>5))
+
+static void snd_pcm_hw_convert_from_old_params(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_params_old *oparams)
+{
+ unsigned int i;
+
+ memset(params, 0, sizeof(*params));
+ params->flags = oparams->flags;
+ for (i = 0; i < ARRAY_SIZE(oparams->masks); i++)
+ params->masks[i].bits[0] = oparams->masks[i];
+ memcpy(params->intervals, oparams->intervals, sizeof(oparams->intervals));
+ params->rmask = __OLD_TO_NEW_MASK(oparams->rmask);
+ params->cmask = __OLD_TO_NEW_MASK(oparams->cmask);
+ params->info = oparams->info;
+ params->msbits = oparams->msbits;
+ params->rate_num = oparams->rate_num;
+ params->rate_den = oparams->rate_den;
+ params->fifo_size = oparams->fifo_size;
+}
+
+static void snd_pcm_hw_convert_to_old_params(struct snd_pcm_hw_params_old *oparams,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned int i;
+
+ memset(oparams, 0, sizeof(*oparams));
+ oparams->flags = params->flags;
+ for (i = 0; i < ARRAY_SIZE(oparams->masks); i++)
+ oparams->masks[i] = params->masks[i].bits[0];
+ memcpy(oparams->intervals, params->intervals, sizeof(oparams->intervals));
+ oparams->rmask = __NEW_TO_OLD_MASK(params->rmask);
+ oparams->cmask = __NEW_TO_OLD_MASK(params->cmask);
+ oparams->info = params->info;
+ oparams->msbits = params->msbits;
+ oparams->rate_num = params->rate_num;
+ oparams->rate_den = params->rate_den;
+ oparams->fifo_size = params->fifo_size;
+}
+
+static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params_old __user * _oparams)
+{
+ struct snd_pcm_hw_params *params;
+ struct snd_pcm_hw_params_old *oparams = NULL;
+ int err;
+
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ if (!params) {
+ err = -ENOMEM;
+ goto out;
+ }
+ oparams = kmalloc(sizeof(*oparams), GFP_KERNEL);
+ if (!oparams) {
+ err = -ENOMEM;
+ goto out;
+ }
+
+ if (copy_from_user(oparams, _oparams, sizeof(*oparams))) {
+ err = -EFAULT;
+ goto out;
+ }
+ snd_pcm_hw_convert_from_old_params(params, oparams);
+ err = snd_pcm_hw_refine(substream, params);
+ snd_pcm_hw_convert_to_old_params(oparams, params);
+ if (copy_to_user(_oparams, oparams, sizeof(*oparams))) {
+ if (!err)
+ err = -EFAULT;
+ }
+out:
+ kfree(params);
+ kfree(oparams);
+ return err;
+}
+
+static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params_old __user * _oparams)
+{
+ struct snd_pcm_hw_params *params;
+ struct snd_pcm_hw_params_old *oparams = NULL;
+ int err;
+
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ if (!params) {
+ err = -ENOMEM;
+ goto out;
+ }
+ oparams = kmalloc(sizeof(*oparams), GFP_KERNEL);
+ if (!oparams) {
+ err = -ENOMEM;
+ goto out;
+ }
+ if (copy_from_user(oparams, _oparams, sizeof(*oparams))) {
+ err = -EFAULT;
+ goto out;
+ }
+ snd_pcm_hw_convert_from_old_params(params, oparams);
+ err = snd_pcm_hw_params(substream, params);
+ snd_pcm_hw_convert_to_old_params(oparams, params);
+ if (copy_to_user(_oparams, oparams, sizeof(*oparams))) {
+ if (!err)
+ err = -EFAULT;
+ }
+out:
+ kfree(params);
+ kfree(oparams);
+ return err;
+}
+#endif /* CONFIG_SND_SUPPORT_OLD_API */
+
+#ifndef CONFIG_MMU
+unsigned long dummy_get_unmapped_area(struct file *file, unsigned long addr,
+ unsigned long len, unsigned long pgoff,
+ unsigned long flags)
+{
+ return 0;
+}
+#else
+# define dummy_get_unmapped_area NULL
+#endif
+
+/*
+ * Register section
+ */
+
+const struct file_operations snd_pcm_f_ops[2] = {
+ {
+ .owner = THIS_MODULE,
+ .write = snd_pcm_write,
+ .aio_write = snd_pcm_aio_write,
+ .open = snd_pcm_playback_open,
+ .release = snd_pcm_release,
+ .poll = snd_pcm_playback_poll,
+ .unlocked_ioctl = snd_pcm_playback_ioctl,
+ .compat_ioctl = snd_pcm_ioctl_compat,
+ .mmap = snd_pcm_mmap,
+ .fasync = snd_pcm_fasync,
+ .get_unmapped_area = dummy_get_unmapped_area,
+ },
+ {
+ .owner = THIS_MODULE,
+ .read = snd_pcm_read,
+ .aio_read = snd_pcm_aio_read,
+ .open = snd_pcm_capture_open,
+ .release = snd_pcm_release,
+ .poll = snd_pcm_capture_poll,
+ .unlocked_ioctl = snd_pcm_capture_ioctl,
+ .compat_ioctl = snd_pcm_ioctl_compat,
+ .mmap = snd_pcm_mmap,
+ .fasync = snd_pcm_fasync,
+ .get_unmapped_area = dummy_get_unmapped_area,
+ }
+};
diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c
new file mode 100644
index 0000000..2c89c04
--- /dev/null
+++ b/sound/core/pcm_timer.c
@@ -0,0 +1,162 @@
+/*
+ * Digital Audio (PCM) abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/timer.h>
+
+/*
+ * Timer functions
+ */
+
+/* Greatest common divisor */
+static unsigned long gcd(unsigned long a, unsigned long b)
+{
+ unsigned long r;
+ if (a < b) {
+ r = a;
+ a = b;
+ b = r;
+ }
+ while ((r = a % b) != 0) {
+ a = b;
+ b = r;
+ }
+ return b;
+}
+
+void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream)
+{
+ unsigned long rate, mult, fsize, l, post;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ mult = 1000000000;
+ rate = runtime->rate;
+ if (snd_BUG_ON(!rate))
+ return;
+ l = gcd(mult, rate);
+ mult /= l;
+ rate /= l;
+ fsize = runtime->period_size;
+ if (snd_BUG_ON(!fsize))
+ return;
+ l = gcd(rate, fsize);
+ rate /= l;
+ fsize /= l;
+ post = 1;
+ while ((mult * fsize) / fsize != mult) {
+ mult /= 2;
+ post *= 2;
+ }
+ if (rate == 0) {
+ snd_printk(KERN_ERR "pcm timer resolution out of range (rate = %u, period_size = %lu)\n", runtime->rate, runtime->period_size);
+ runtime->timer_resolution = -1;
+ return;
+ }
+ runtime->timer_resolution = (mult * fsize / rate) * post;
+}
+
+static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer)
+{
+ struct snd_pcm_substream *substream;
+
+ substream = timer->private_data;
+ return substream->runtime ? substream->runtime->timer_resolution : 0;
+}
+
+static int snd_pcm_timer_start(struct snd_timer * timer)
+{
+ unsigned long flags;
+ struct snd_pcm_substream *substream;
+
+ substream = snd_timer_chip(timer);
+ spin_lock_irqsave(&substream->timer_lock, flags);
+ substream->timer_running = 1;
+ spin_unlock_irqrestore(&substream->timer_lock, flags);
+ return 0;
+}
+
+static int snd_pcm_timer_stop(struct snd_timer * timer)
+{
+ unsigned long flags;
+ struct snd_pcm_substream *substream;
+
+ substream = snd_timer_chip(timer);
+ spin_lock_irqsave(&substream->timer_lock, flags);
+ substream->timer_running = 0;
+ spin_unlock_irqrestore(&substream->timer_lock, flags);
+ return 0;
+}
+
+static struct snd_timer_hardware snd_pcm_timer =
+{
+ .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_SLAVE,
+ .resolution = 0,
+ .ticks = 1,
+ .c_resolution = snd_pcm_timer_resolution,
+ .start = snd_pcm_timer_start,
+ .stop = snd_pcm_timer_stop,
+};
+
+/*
+ * Init functions
+ */
+
+static void snd_pcm_timer_free(struct snd_timer *timer)
+{
+ struct snd_pcm_substream *substream = timer->private_data;
+ substream->timer = NULL;
+}
+
+void snd_pcm_timer_init(struct snd_pcm_substream *substream)
+{
+ struct snd_timer_id tid;
+ struct snd_timer *timer;
+
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.dev_class = SNDRV_TIMER_CLASS_PCM;
+ tid.card = substream->pcm->card->number;
+ tid.device = substream->pcm->device;
+ tid.subdevice = (substream->number << 1) | (substream->stream & 1);
+ if (snd_timer_new(substream->pcm->card, "PCM", &tid, &timer) < 0)
+ return;
+ sprintf(timer->name, "PCM %s %i-%i-%i",
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE ?
+ "capture" : "playback",
+ tid.card, tid.device, tid.subdevice);
+ timer->hw = snd_pcm_timer;
+ if (snd_device_register(timer->card, timer) < 0) {
+ snd_device_free(timer->card, timer);
+ return;
+ }
+ timer->private_data = substream;
+ timer->private_free = snd_pcm_timer_free;
+ substream->timer = timer;
+}
+
+void snd_pcm_timer_done(struct snd_pcm_substream *substream)
+{
+ if (substream->timer) {
+ snd_device_free(substream->pcm->card, substream->timer);
+ substream->timer = NULL;
+ }
+}
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
new file mode 100644
index 0000000..39672f6
--- /dev/null
+++ b/sound/core/rawmidi.c
@@ -0,0 +1,1692 @@
+/*
+ * Abstract layer for MIDI v1.0 stream
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <linux/major.h>
+#include <linux/init.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/mutex.h>
+#include <linux/moduleparam.h>
+#include <linux/delay.h>
+#include <sound/rawmidi.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/minors.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Midlevel RawMidi code for ALSA.");
+MODULE_LICENSE("GPL");
+
+#ifdef CONFIG_SND_OSSEMUL
+static int midi_map[SNDRV_CARDS];
+static int amidi_map[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1};
+module_param_array(midi_map, int, NULL, 0444);
+MODULE_PARM_DESC(midi_map, "Raw MIDI device number assigned to 1st OSS device.");
+module_param_array(amidi_map, int, NULL, 0444);
+MODULE_PARM_DESC(amidi_map, "Raw MIDI device number assigned to 2nd OSS device.");
+#endif /* CONFIG_SND_OSSEMUL */
+
+static int snd_rawmidi_free(struct snd_rawmidi *rawmidi);
+static int snd_rawmidi_dev_free(struct snd_device *device);
+static int snd_rawmidi_dev_register(struct snd_device *device);
+static int snd_rawmidi_dev_disconnect(struct snd_device *device);
+
+static LIST_HEAD(snd_rawmidi_devices);
+static DEFINE_MUTEX(register_mutex);
+
+static struct snd_rawmidi *snd_rawmidi_search(struct snd_card *card, int device)
+{
+ struct snd_rawmidi *rawmidi;
+
+ list_for_each_entry(rawmidi, &snd_rawmidi_devices, list)
+ if (rawmidi->card == card && rawmidi->device == device)
+ return rawmidi;
+ return NULL;
+}
+
+static inline unsigned short snd_rawmidi_file_flags(struct file *file)
+{
+ switch (file->f_mode & (FMODE_READ | FMODE_WRITE)) {
+ case FMODE_WRITE:
+ return SNDRV_RAWMIDI_LFLG_OUTPUT;
+ case FMODE_READ:
+ return SNDRV_RAWMIDI_LFLG_INPUT;
+ default:
+ return SNDRV_RAWMIDI_LFLG_OPEN;
+ }
+}
+
+static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream)
+{
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+ return runtime->avail >= runtime->avail_min;
+}
+
+static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream,
+ size_t count)
+{
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+ return runtime->avail >= runtime->avail_min &&
+ (!substream->append || runtime->avail >= count);
+}
+
+static void snd_rawmidi_input_event_tasklet(unsigned long data)
+{
+ struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
+ substream->runtime->event(substream);
+}
+
+static void snd_rawmidi_output_trigger_tasklet(unsigned long data)
+{
+ struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *)data;
+ substream->ops->trigger(substream, 1);
+}
+
+static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
+{
+ struct snd_rawmidi_runtime *runtime;
+
+ if ((runtime = kzalloc(sizeof(*runtime), GFP_KERNEL)) == NULL)
+ return -ENOMEM;
+ spin_lock_init(&runtime->lock);
+ init_waitqueue_head(&runtime->sleep);
+ if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT)
+ tasklet_init(&runtime->tasklet,
+ snd_rawmidi_input_event_tasklet,
+ (unsigned long)substream);
+ else
+ tasklet_init(&runtime->tasklet,
+ snd_rawmidi_output_trigger_tasklet,
+ (unsigned long)substream);
+ runtime->event = NULL;
+ runtime->buffer_size = PAGE_SIZE;
+ runtime->avail_min = 1;
+ if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT)
+ runtime->avail = 0;
+ else
+ runtime->avail = runtime->buffer_size;
+ if ((runtime->buffer = kmalloc(runtime->buffer_size, GFP_KERNEL)) == NULL) {
+ kfree(runtime);
+ return -ENOMEM;
+ }
+ runtime->appl_ptr = runtime->hw_ptr = 0;
+ substream->runtime = runtime;
+ return 0;
+}
+
+static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream)
+{
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ kfree(runtime->buffer);
+ kfree(runtime);
+ substream->runtime = NULL;
+ return 0;
+}
+
+static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up)
+{
+ if (!substream->opened)
+ return;
+ if (up) {
+ tasklet_hi_schedule(&substream->runtime->tasklet);
+ } else {
+ tasklet_kill(&substream->runtime->tasklet);
+ substream->ops->trigger(substream, 0);
+ }
+}
+
+static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ if (!substream->opened)
+ return;
+ substream->ops->trigger(substream, up);
+ if (!up && substream->runtime->event)
+ tasklet_kill(&substream->runtime->tasklet);
+}
+
+int snd_rawmidi_drop_output(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ snd_rawmidi_output_trigger(substream, 0);
+ runtime->drain = 0;
+ spin_lock_irqsave(&runtime->lock, flags);
+ runtime->appl_ptr = runtime->hw_ptr = 0;
+ runtime->avail = runtime->buffer_size;
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return 0;
+}
+
+int snd_rawmidi_drain_output(struct snd_rawmidi_substream *substream)
+{
+ int err;
+ long timeout;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ err = 0;
+ runtime->drain = 1;
+ timeout = wait_event_interruptible_timeout(runtime->sleep,
+ (runtime->avail >= runtime->buffer_size),
+ 10*HZ);
+ if (signal_pending(current))
+ err = -ERESTARTSYS;
+ if (runtime->avail < runtime->buffer_size && !timeout) {
+ snd_printk(KERN_WARNING "rawmidi drain error (avail = %li, buffer_size = %li)\n", (long)runtime->avail, (long)runtime->buffer_size);
+ err = -EIO;
+ }
+ runtime->drain = 0;
+ if (err != -ERESTARTSYS) {
+ /* we need wait a while to make sure that Tx FIFOs are empty */
+ if (substream->ops->drain)
+ substream->ops->drain(substream);
+ else
+ msleep(50);
+ snd_rawmidi_drop_output(substream);
+ }
+ return err;
+}
+
+int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ snd_rawmidi_input_trigger(substream, 0);
+ runtime->drain = 0;
+ spin_lock_irqsave(&runtime->lock, flags);
+ runtime->appl_ptr = runtime->hw_ptr = 0;
+ runtime->avail = 0;
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return 0;
+}
+
+int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice,
+ int mode, struct snd_rawmidi_file * rfile)
+{
+ struct snd_rawmidi *rmidi;
+ struct list_head *list1, *list2;
+ struct snd_rawmidi_substream *sinput = NULL, *soutput = NULL;
+ struct snd_rawmidi_runtime *input = NULL, *output = NULL;
+ int err;
+
+ if (rfile)
+ rfile->input = rfile->output = NULL;
+ mutex_lock(&register_mutex);
+ rmidi = snd_rawmidi_search(card, device);
+ mutex_unlock(&register_mutex);
+ if (rmidi == NULL) {
+ err = -ENODEV;
+ goto __error1;
+ }
+ if (!try_module_get(rmidi->card->module)) {
+ err = -EFAULT;
+ goto __error1;
+ }
+ if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK))
+ mutex_lock(&rmidi->open_mutex);
+ if (mode & SNDRV_RAWMIDI_LFLG_INPUT) {
+ if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_INPUT)) {
+ err = -ENXIO;
+ goto __error;
+ }
+ if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) {
+ err = -ENODEV;
+ goto __error;
+ }
+ if (rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened >=
+ rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) {
+ err = -EAGAIN;
+ goto __error;
+ }
+ }
+ if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
+ if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT)) {
+ err = -ENXIO;
+ goto __error;
+ }
+ if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) {
+ err = -ENODEV;
+ goto __error;
+ }
+ if (rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened >=
+ rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) {
+ err = -EAGAIN;
+ goto __error;
+ }
+ }
+ list1 = rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams.next;
+ while (1) {
+ if (list1 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) {
+ sinput = NULL;
+ if (mode & SNDRV_RAWMIDI_LFLG_INPUT) {
+ err = -EAGAIN;
+ goto __error;
+ }
+ break;
+ }
+ sinput = list_entry(list1, struct snd_rawmidi_substream, list);
+ if ((mode & SNDRV_RAWMIDI_LFLG_INPUT) && sinput->opened)
+ goto __nexti;
+ if (subdevice < 0 || (subdevice >= 0 && subdevice == sinput->number))
+ break;
+ __nexti:
+ list1 = list1->next;
+ }
+ list2 = rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams.next;
+ while (1) {
+ if (list2 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) {
+ soutput = NULL;
+ if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
+ err = -EAGAIN;
+ goto __error;
+ }
+ break;
+ }
+ soutput = list_entry(list2, struct snd_rawmidi_substream, list);
+ if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
+ if (mode & SNDRV_RAWMIDI_LFLG_APPEND) {
+ if (soutput->opened && !soutput->append)
+ goto __nexto;
+ } else {
+ if (soutput->opened)
+ goto __nexto;
+ }
+ }
+ if (subdevice < 0 || (subdevice >= 0 && subdevice == soutput->number))
+ break;
+ __nexto:
+ list2 = list2->next;
+ }
+ if (mode & SNDRV_RAWMIDI_LFLG_INPUT) {
+ if ((err = snd_rawmidi_runtime_create(sinput)) < 0)
+ goto __error;
+ input = sinput->runtime;
+ if ((err = sinput->ops->open(sinput)) < 0)
+ goto __error;
+ sinput->opened = 1;
+ rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened++;
+ } else {
+ sinput = NULL;
+ }
+ if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
+ if (soutput->opened)
+ goto __skip_output;
+ if ((err = snd_rawmidi_runtime_create(soutput)) < 0) {
+ if (mode & SNDRV_RAWMIDI_LFLG_INPUT)
+ sinput->ops->close(sinput);
+ goto __error;
+ }
+ output = soutput->runtime;
+ if ((err = soutput->ops->open(soutput)) < 0) {
+ if (mode & SNDRV_RAWMIDI_LFLG_INPUT)
+ sinput->ops->close(sinput);
+ goto __error;
+ }
+ __skip_output:
+ soutput->opened = 1;
+ if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
+ soutput->append = 1;
+ if (soutput->use_count++ == 0)
+ soutput->active_sensing = 1;
+ rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened++;
+ } else {
+ soutput = NULL;
+ }
+ if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK))
+ mutex_unlock(&rmidi->open_mutex);
+ if (rfile) {
+ rfile->rmidi = rmidi;
+ rfile->input = sinput;
+ rfile->output = soutput;
+ }
+ return 0;
+
+ __error:
+ if (input != NULL)
+ snd_rawmidi_runtime_free(sinput);
+ if (output != NULL)
+ snd_rawmidi_runtime_free(soutput);
+ module_put(rmidi->card->module);
+ if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK))
+ mutex_unlock(&rmidi->open_mutex);
+ __error1:
+ return err;
+}
+
+static int snd_rawmidi_open(struct inode *inode, struct file *file)
+{
+ int maj = imajor(inode);
+ struct snd_card *card;
+ int subdevice;
+ unsigned short fflags;
+ int err;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_file *rawmidi_file;
+ wait_queue_t wait;
+ struct snd_ctl_file *kctl;
+
+ if (maj == snd_major) {
+ rmidi = snd_lookup_minor_data(iminor(inode),
+ SNDRV_DEVICE_TYPE_RAWMIDI);
+#ifdef CONFIG_SND_OSSEMUL
+ } else if (maj == SOUND_MAJOR) {
+ rmidi = snd_lookup_oss_minor_data(iminor(inode),
+ SNDRV_OSS_DEVICE_TYPE_MIDI);
+#endif
+ } else
+ return -ENXIO;
+
+ if (rmidi == NULL)
+ return -ENODEV;
+ if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK))
+ return -EINVAL; /* invalid combination */
+ card = rmidi->card;
+ err = snd_card_file_add(card, file);
+ if (err < 0)
+ return -ENODEV;
+ fflags = snd_rawmidi_file_flags(file);
+ if ((file->f_flags & O_APPEND) || maj == SOUND_MAJOR) /* OSS emul? */
+ fflags |= SNDRV_RAWMIDI_LFLG_APPEND;
+ fflags |= SNDRV_RAWMIDI_LFLG_NOOPENLOCK;
+ rawmidi_file = kmalloc(sizeof(*rawmidi_file), GFP_KERNEL);
+ if (rawmidi_file == NULL) {
+ snd_card_file_remove(card, file);
+ return -ENOMEM;
+ }
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&rmidi->open_wait, &wait);
+ mutex_lock(&rmidi->open_mutex);
+ while (1) {
+ subdevice = -1;
+ read_lock(&card->ctl_files_rwlock);
+ list_for_each_entry(kctl, &card->ctl_files, list) {
+ if (kctl->pid == current->pid) {
+ subdevice = kctl->prefer_rawmidi_subdevice;
+ if (subdevice != -1)
+ break;
+ }
+ }
+ read_unlock(&card->ctl_files_rwlock);
+ err = snd_rawmidi_kernel_open(rmidi->card, rmidi->device,
+ subdevice, fflags, rawmidi_file);
+ if (err >= 0)
+ break;
+ if (err == -EAGAIN) {
+ if (file->f_flags & O_NONBLOCK) {
+ err = -EBUSY;
+ break;
+ }
+ } else
+ break;
+ set_current_state(TASK_INTERRUPTIBLE);
+ mutex_unlock(&rmidi->open_mutex);
+ schedule();
+ mutex_lock(&rmidi->open_mutex);
+ if (signal_pending(current)) {
+ err = -ERESTARTSYS;
+ break;
+ }
+ }
+#ifdef CONFIG_SND_OSSEMUL
+ if (rawmidi_file->input && rawmidi_file->input->runtime)
+ rawmidi_file->input->runtime->oss = (maj == SOUND_MAJOR);
+ if (rawmidi_file->output && rawmidi_file->output->runtime)
+ rawmidi_file->output->runtime->oss = (maj == SOUND_MAJOR);
+#endif
+ remove_wait_queue(&rmidi->open_wait, &wait);
+ if (err >= 0) {
+ file->private_data = rawmidi_file;
+ } else {
+ snd_card_file_remove(card, file);
+ kfree(rawmidi_file);
+ }
+ mutex_unlock(&rmidi->open_mutex);
+ return err;
+}
+
+int snd_rawmidi_kernel_release(struct snd_rawmidi_file * rfile)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream;
+ struct snd_rawmidi_runtime *runtime;
+
+ if (snd_BUG_ON(!rfile))
+ return -ENXIO;
+ rmidi = rfile->rmidi;
+ mutex_lock(&rmidi->open_mutex);
+ if (rfile->input != NULL) {
+ substream = rfile->input;
+ rfile->input = NULL;
+ runtime = substream->runtime;
+ snd_rawmidi_input_trigger(substream, 0);
+ substream->ops->close(substream);
+ if (runtime->private_free != NULL)
+ runtime->private_free(substream);
+ snd_rawmidi_runtime_free(substream);
+ substream->opened = 0;
+ rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened--;
+ }
+ if (rfile->output != NULL) {
+ substream = rfile->output;
+ rfile->output = NULL;
+ if (--substream->use_count == 0) {
+ runtime = substream->runtime;
+ if (substream->active_sensing) {
+ unsigned char buf = 0xfe;
+ /* sending single active sensing message to shut the device up */
+ snd_rawmidi_kernel_write(substream, &buf, 1);
+ }
+ if (snd_rawmidi_drain_output(substream) == -ERESTARTSYS)
+ snd_rawmidi_output_trigger(substream, 0);
+ substream->ops->close(substream);
+ if (runtime->private_free != NULL)
+ runtime->private_free(substream);
+ snd_rawmidi_runtime_free(substream);
+ substream->opened = 0;
+ substream->append = 0;
+ }
+ rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened--;
+ }
+ mutex_unlock(&rmidi->open_mutex);
+ module_put(rmidi->card->module);
+ return 0;
+}
+
+static int snd_rawmidi_release(struct inode *inode, struct file *file)
+{
+ struct snd_rawmidi_file *rfile;
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ rfile = file->private_data;
+ err = snd_rawmidi_kernel_release(rfile);
+ rmidi = rfile->rmidi;
+ wake_up(&rmidi->open_wait);
+ kfree(rfile);
+ snd_card_file_remove(rmidi->card, file);
+ return err;
+}
+
+static int snd_rawmidi_info(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_info *info)
+{
+ struct snd_rawmidi *rmidi;
+
+ if (substream == NULL)
+ return -ENODEV;
+ rmidi = substream->rmidi;
+ memset(info, 0, sizeof(*info));
+ info->card = rmidi->card->number;
+ info->device = rmidi->device;
+ info->subdevice = substream->number;
+ info->stream = substream->stream;
+ info->flags = rmidi->info_flags;
+ strcpy(info->id, rmidi->id);
+ strcpy(info->name, rmidi->name);
+ strcpy(info->subname, substream->name);
+ info->subdevices_count = substream->pstr->substream_count;
+ info->subdevices_avail = (substream->pstr->substream_count -
+ substream->pstr->substream_opened);
+ return 0;
+}
+
+static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_info __user * _info)
+{
+ struct snd_rawmidi_info info;
+ int err;
+ if ((err = snd_rawmidi_info(substream, &info)) < 0)
+ return err;
+ if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info)))
+ return -EFAULT;
+ return 0;
+}
+
+int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_str *pstr;
+ struct snd_rawmidi_substream *substream;
+
+ mutex_lock(&register_mutex);
+ rmidi = snd_rawmidi_search(card, info->device);
+ mutex_unlock(&register_mutex);
+ if (!rmidi)
+ return -ENXIO;
+ if (info->stream < 0 || info->stream > 1)
+ return -EINVAL;
+ pstr = &rmidi->streams[info->stream];
+ if (pstr->substream_count == 0)
+ return -ENOENT;
+ if (info->subdevice >= pstr->substream_count)
+ return -ENXIO;
+ list_for_each_entry(substream, &pstr->substreams, list) {
+ if ((unsigned int)substream->number == info->subdevice)
+ return snd_rawmidi_info(substream, info);
+ }
+ return -ENXIO;
+}
+
+static int snd_rawmidi_info_select_user(struct snd_card *card,
+ struct snd_rawmidi_info __user *_info)
+{
+ int err;
+ struct snd_rawmidi_info info;
+ if (get_user(info.device, &_info->device))
+ return -EFAULT;
+ if (get_user(info.stream, &_info->stream))
+ return -EFAULT;
+ if (get_user(info.subdevice, &_info->subdevice))
+ return -EFAULT;
+ if ((err = snd_rawmidi_info_select(card, &info)) < 0)
+ return err;
+ if (copy_to_user(_info, &info, sizeof(struct snd_rawmidi_info)))
+ return -EFAULT;
+ return 0;
+}
+
+int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_params * params)
+{
+ char *newbuf;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ if (substream->append && substream->use_count > 1)
+ return -EBUSY;
+ snd_rawmidi_drain_output(substream);
+ if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) {
+ return -EINVAL;
+ }
+ if (params->avail_min < 1 || params->avail_min > params->buffer_size) {
+ return -EINVAL;
+ }
+ if (params->buffer_size != runtime->buffer_size) {
+ newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
+ if (!newbuf)
+ return -ENOMEM;
+ kfree(runtime->buffer);
+ runtime->buffer = newbuf;
+ runtime->buffer_size = params->buffer_size;
+ runtime->avail = runtime->buffer_size;
+ }
+ runtime->avail_min = params->avail_min;
+ substream->active_sensing = !params->no_active_sensing;
+ return 0;
+}
+
+int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_params * params)
+{
+ char *newbuf;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ snd_rawmidi_drain_input(substream);
+ if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) {
+ return -EINVAL;
+ }
+ if (params->avail_min < 1 || params->avail_min > params->buffer_size) {
+ return -EINVAL;
+ }
+ if (params->buffer_size != runtime->buffer_size) {
+ newbuf = kmalloc(params->buffer_size, GFP_KERNEL);
+ if (!newbuf)
+ return -ENOMEM;
+ kfree(runtime->buffer);
+ runtime->buffer = newbuf;
+ runtime->buffer_size = params->buffer_size;
+ }
+ runtime->avail_min = params->avail_min;
+ return 0;
+}
+
+static int snd_rawmidi_output_status(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_status * status)
+{
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ memset(status, 0, sizeof(*status));
+ status->stream = SNDRV_RAWMIDI_STREAM_OUTPUT;
+ spin_lock_irq(&runtime->lock);
+ status->avail = runtime->avail;
+ spin_unlock_irq(&runtime->lock);
+ return 0;
+}
+
+static int snd_rawmidi_input_status(struct snd_rawmidi_substream *substream,
+ struct snd_rawmidi_status * status)
+{
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ memset(status, 0, sizeof(*status));
+ status->stream = SNDRV_RAWMIDI_STREAM_INPUT;
+ spin_lock_irq(&runtime->lock);
+ status->avail = runtime->avail;
+ status->xruns = runtime->xruns;
+ runtime->xruns = 0;
+ spin_unlock_irq(&runtime->lock);
+ return 0;
+}
+
+static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_rawmidi_file *rfile;
+ void __user *argp = (void __user *)arg;
+
+ rfile = file->private_data;
+ if (((cmd >> 8) & 0xff) != 'W')
+ return -ENOTTY;
+ switch (cmd) {
+ case SNDRV_RAWMIDI_IOCTL_PVERSION:
+ return put_user(SNDRV_RAWMIDI_VERSION, (int __user *)argp) ? -EFAULT : 0;
+ case SNDRV_RAWMIDI_IOCTL_INFO:
+ {
+ int stream;
+ struct snd_rawmidi_info __user *info = argp;
+ if (get_user(stream, &info->stream))
+ return -EFAULT;
+ switch (stream) {
+ case SNDRV_RAWMIDI_STREAM_INPUT:
+ return snd_rawmidi_info_user(rfile->input, info);
+ case SNDRV_RAWMIDI_STREAM_OUTPUT:
+ return snd_rawmidi_info_user(rfile->output, info);
+ default:
+ return -EINVAL;
+ }
+ }
+ case SNDRV_RAWMIDI_IOCTL_PARAMS:
+ {
+ struct snd_rawmidi_params params;
+ if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params)))
+ return -EFAULT;
+ switch (params.stream) {
+ case SNDRV_RAWMIDI_STREAM_OUTPUT:
+ if (rfile->output == NULL)
+ return -EINVAL;
+ return snd_rawmidi_output_params(rfile->output, &params);
+ case SNDRV_RAWMIDI_STREAM_INPUT:
+ if (rfile->input == NULL)
+ return -EINVAL;
+ return snd_rawmidi_input_params(rfile->input, &params);
+ default:
+ return -EINVAL;
+ }
+ }
+ case SNDRV_RAWMIDI_IOCTL_STATUS:
+ {
+ int err = 0;
+ struct snd_rawmidi_status status;
+ if (copy_from_user(&status, argp, sizeof(struct snd_rawmidi_status)))
+ return -EFAULT;
+ switch (status.stream) {
+ case SNDRV_RAWMIDI_STREAM_OUTPUT:
+ if (rfile->output == NULL)
+ return -EINVAL;
+ err = snd_rawmidi_output_status(rfile->output, &status);
+ break;
+ case SNDRV_RAWMIDI_STREAM_INPUT:
+ if (rfile->input == NULL)
+ return -EINVAL;
+ err = snd_rawmidi_input_status(rfile->input, &status);
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (err < 0)
+ return err;
+ if (copy_to_user(argp, &status, sizeof(struct snd_rawmidi_status)))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_RAWMIDI_IOCTL_DROP:
+ {
+ int val;
+ if (get_user(val, (int __user *) argp))
+ return -EFAULT;
+ switch (val) {
+ case SNDRV_RAWMIDI_STREAM_OUTPUT:
+ if (rfile->output == NULL)
+ return -EINVAL;
+ return snd_rawmidi_drop_output(rfile->output);
+ default:
+ return -EINVAL;
+ }
+ }
+ case SNDRV_RAWMIDI_IOCTL_DRAIN:
+ {
+ int val;
+ if (get_user(val, (int __user *) argp))
+ return -EFAULT;
+ switch (val) {
+ case SNDRV_RAWMIDI_STREAM_OUTPUT:
+ if (rfile->output == NULL)
+ return -EINVAL;
+ return snd_rawmidi_drain_output(rfile->output);
+ case SNDRV_RAWMIDI_STREAM_INPUT:
+ if (rfile->input == NULL)
+ return -EINVAL;
+ return snd_rawmidi_drain_input(rfile->input);
+ default:
+ return -EINVAL;
+ }
+ }
+#ifdef CONFIG_SND_DEBUG
+ default:
+ snd_printk(KERN_WARNING "rawmidi: unknown command = 0x%x\n", cmd);
+#endif
+ }
+ return -ENOTTY;
+}
+
+static int snd_rawmidi_control_ioctl(struct snd_card *card,
+ struct snd_ctl_file *control,
+ unsigned int cmd,
+ unsigned long arg)
+{
+ void __user *argp = (void __user *)arg;
+
+ switch (cmd) {
+ case SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE:
+ {
+ int device;
+
+ if (get_user(device, (int __user *)argp))
+ return -EFAULT;
+ mutex_lock(&register_mutex);
+ device = device < 0 ? 0 : device + 1;
+ while (device < SNDRV_RAWMIDI_DEVICES) {
+ if (snd_rawmidi_search(card, device))
+ break;
+ device++;
+ }
+ if (device == SNDRV_RAWMIDI_DEVICES)
+ device = -1;
+ mutex_unlock(&register_mutex);
+ if (put_user(device, (int __user *)argp))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE:
+ {
+ int val;
+
+ if (get_user(val, (int __user *)argp))
+ return -EFAULT;
+ control->prefer_rawmidi_subdevice = val;
+ return 0;
+ }
+ case SNDRV_CTL_IOCTL_RAWMIDI_INFO:
+ return snd_rawmidi_info_select_user(card, argp);
+ }
+ return -ENOIOCTLCMD;
+}
+
+/**
+ * snd_rawmidi_receive - receive the input data from the device
+ * @substream: the rawmidi substream
+ * @buffer: the buffer pointer
+ * @count: the data size to read
+ *
+ * Reads the data from the internal buffer.
+ *
+ * Returns the size of read data, or a negative error code on failure.
+ */
+int snd_rawmidi_receive(struct snd_rawmidi_substream *substream,
+ const unsigned char *buffer, int count)
+{
+ unsigned long flags;
+ int result = 0, count1;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ if (!substream->opened)
+ return -EBADFD;
+ if (runtime->buffer == NULL) {
+ snd_printd("snd_rawmidi_receive: input is not active!!!\n");
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&runtime->lock, flags);
+ if (count == 1) { /* special case, faster code */
+ substream->bytes++;
+ if (runtime->avail < runtime->buffer_size) {
+ runtime->buffer[runtime->hw_ptr++] = buffer[0];
+ runtime->hw_ptr %= runtime->buffer_size;
+ runtime->avail++;
+ result++;
+ } else {
+ runtime->xruns++;
+ }
+ } else {
+ substream->bytes += count;
+ count1 = runtime->buffer_size - runtime->hw_ptr;
+ if (count1 > count)
+ count1 = count;
+ if (count1 > (int)(runtime->buffer_size - runtime->avail))
+ count1 = runtime->buffer_size - runtime->avail;
+ memcpy(runtime->buffer + runtime->hw_ptr, buffer, count1);
+ runtime->hw_ptr += count1;
+ runtime->hw_ptr %= runtime->buffer_size;
+ runtime->avail += count1;
+ count -= count1;
+ result += count1;
+ if (count > 0) {
+ buffer += count1;
+ count1 = count;
+ if (count1 > (int)(runtime->buffer_size - runtime->avail)) {
+ count1 = runtime->buffer_size - runtime->avail;
+ runtime->xruns += count - count1;
+ }
+ if (count1 > 0) {
+ memcpy(runtime->buffer, buffer, count1);
+ runtime->hw_ptr = count1;
+ runtime->avail += count1;
+ result += count1;
+ }
+ }
+ }
+ if (result > 0) {
+ if (runtime->event)
+ tasklet_hi_schedule(&runtime->tasklet);
+ else if (snd_rawmidi_ready(substream))
+ wake_up(&runtime->sleep);
+ }
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return result;
+}
+
+static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream,
+ unsigned char __user *userbuf,
+ unsigned char *kernelbuf, long count)
+{
+ unsigned long flags;
+ long result = 0, count1;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ while (count > 0 && runtime->avail) {
+ count1 = runtime->buffer_size - runtime->appl_ptr;
+ if (count1 > count)
+ count1 = count;
+ spin_lock_irqsave(&runtime->lock, flags);
+ if (count1 > (int)runtime->avail)
+ count1 = runtime->avail;
+ if (kernelbuf)
+ memcpy(kernelbuf + result, runtime->buffer + runtime->appl_ptr, count1);
+ if (userbuf) {
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ if (copy_to_user(userbuf + result,
+ runtime->buffer + runtime->appl_ptr, count1)) {
+ return result > 0 ? result : -EFAULT;
+ }
+ spin_lock_irqsave(&runtime->lock, flags);
+ }
+ runtime->appl_ptr += count1;
+ runtime->appl_ptr %= runtime->buffer_size;
+ runtime->avail -= count1;
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ result += count1;
+ count -= count1;
+ }
+ return result;
+}
+
+long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream,
+ unsigned char *buf, long count)
+{
+ snd_rawmidi_input_trigger(substream, 1);
+ return snd_rawmidi_kernel_read1(substream, NULL/*userbuf*/, buf, count);
+}
+
+static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t count,
+ loff_t *offset)
+{
+ long result;
+ int count1;
+ struct snd_rawmidi_file *rfile;
+ struct snd_rawmidi_substream *substream;
+ struct snd_rawmidi_runtime *runtime;
+
+ rfile = file->private_data;
+ substream = rfile->input;
+ if (substream == NULL)
+ return -EIO;
+ runtime = substream->runtime;
+ snd_rawmidi_input_trigger(substream, 1);
+ result = 0;
+ while (count > 0) {
+ spin_lock_irq(&runtime->lock);
+ while (!snd_rawmidi_ready(substream)) {
+ wait_queue_t wait;
+ if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) {
+ spin_unlock_irq(&runtime->lock);
+ return result > 0 ? result : -EAGAIN;
+ }
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&runtime->sleep, &wait);
+ set_current_state(TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&runtime->lock);
+ schedule();
+ remove_wait_queue(&runtime->sleep, &wait);
+ if (signal_pending(current))
+ return result > 0 ? result : -ERESTARTSYS;
+ if (!runtime->avail)
+ return result > 0 ? result : -EIO;
+ spin_lock_irq(&runtime->lock);
+ }
+ spin_unlock_irq(&runtime->lock);
+ count1 = snd_rawmidi_kernel_read1(substream,
+ (unsigned char __user *)buf,
+ NULL/*kernelbuf*/,
+ count);
+ if (count1 < 0)
+ return result > 0 ? result : count1;
+ result += count1;
+ buf += count1;
+ count -= count1;
+ }
+ return result;
+}
+
+/**
+ * snd_rawmidi_transmit_empty - check whether the output buffer is empty
+ * @substream: the rawmidi substream
+ *
+ * Returns 1 if the internal output buffer is empty, 0 if not.
+ */
+int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream)
+{
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+ int result;
+ unsigned long flags;
+
+ if (runtime->buffer == NULL) {
+ snd_printd("snd_rawmidi_transmit_empty: output is not active!!!\n");
+ return 1;
+ }
+ spin_lock_irqsave(&runtime->lock, flags);
+ result = runtime->avail >= runtime->buffer_size;
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return result;
+}
+
+/**
+ * snd_rawmidi_transmit_peek - copy data from the internal buffer
+ * @substream: the rawmidi substream
+ * @buffer: the buffer pointer
+ * @count: data size to transfer
+ *
+ * Copies data from the internal output buffer to the given buffer.
+ *
+ * Call this in the interrupt handler when the midi output is ready,
+ * and call snd_rawmidi_transmit_ack() after the transmission is
+ * finished.
+ *
+ * Returns the size of copied data, or a negative error code on failure.
+ */
+int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
+ unsigned char *buffer, int count)
+{
+ unsigned long flags;
+ int result, count1;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ if (runtime->buffer == NULL) {
+ snd_printd("snd_rawmidi_transmit_peek: output is not active!!!\n");
+ return -EINVAL;
+ }
+ result = 0;
+ spin_lock_irqsave(&runtime->lock, flags);
+ if (runtime->avail >= runtime->buffer_size) {
+ /* warning: lowlevel layer MUST trigger down the hardware */
+ goto __skip;
+ }
+ if (count == 1) { /* special case, faster code */
+ *buffer = runtime->buffer[runtime->hw_ptr];
+ result++;
+ } else {
+ count1 = runtime->buffer_size - runtime->hw_ptr;
+ if (count1 > count)
+ count1 = count;
+ if (count1 > (int)(runtime->buffer_size - runtime->avail))
+ count1 = runtime->buffer_size - runtime->avail;
+ memcpy(buffer, runtime->buffer + runtime->hw_ptr, count1);
+ count -= count1;
+ result += count1;
+ if (count > 0) {
+ if (count > (int)(runtime->buffer_size - runtime->avail - count1))
+ count = runtime->buffer_size - runtime->avail - count1;
+ memcpy(buffer + count1, runtime->buffer, count);
+ result += count;
+ }
+ }
+ __skip:
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return result;
+}
+
+/**
+ * snd_rawmidi_transmit_ack - acknowledge the transmission
+ * @substream: the rawmidi substream
+ * @count: the tranferred count
+ *
+ * Advances the hardware pointer for the internal output buffer with
+ * the given size and updates the condition.
+ * Call after the transmission is finished.
+ *
+ * Returns the advanced size if successful, or a negative error code on failure.
+ */
+int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count)
+{
+ unsigned long flags;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ if (runtime->buffer == NULL) {
+ snd_printd("snd_rawmidi_transmit_ack: output is not active!!!\n");
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&runtime->lock, flags);
+ snd_BUG_ON(runtime->avail + count > runtime->buffer_size);
+ runtime->hw_ptr += count;
+ runtime->hw_ptr %= runtime->buffer_size;
+ runtime->avail += count;
+ substream->bytes += count;
+ if (count > 0) {
+ if (runtime->drain || snd_rawmidi_ready(substream))
+ wake_up(&runtime->sleep);
+ }
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return count;
+}
+
+/**
+ * snd_rawmidi_transmit - copy from the buffer to the device
+ * @substream: the rawmidi substream
+ * @buffer: the buffer pointer
+ * @count: the data size to transfer
+ *
+ * Copies data from the buffer to the device and advances the pointer.
+ *
+ * Returns the copied size if successful, or a negative error code on failure.
+ */
+int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
+ unsigned char *buffer, int count)
+{
+ if (!substream->opened)
+ return -EBADFD;
+ count = snd_rawmidi_transmit_peek(substream, buffer, count);
+ if (count < 0)
+ return count;
+ return snd_rawmidi_transmit_ack(substream, count);
+}
+
+static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream,
+ const unsigned char __user *userbuf,
+ const unsigned char *kernelbuf,
+ long count)
+{
+ unsigned long flags;
+ long count1, result;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+
+ if (snd_BUG_ON(!kernelbuf && !userbuf))
+ return -EINVAL;
+ if (snd_BUG_ON(!runtime->buffer))
+ return -EINVAL;
+
+ result = 0;
+ spin_lock_irqsave(&runtime->lock, flags);
+ if (substream->append) {
+ if ((long)runtime->avail < count) {
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return -EAGAIN;
+ }
+ }
+ while (count > 0 && runtime->avail > 0) {
+ count1 = runtime->buffer_size - runtime->appl_ptr;
+ if (count1 > count)
+ count1 = count;
+ if (count1 > (long)runtime->avail)
+ count1 = runtime->avail;
+ if (kernelbuf)
+ memcpy(runtime->buffer + runtime->appl_ptr,
+ kernelbuf + result, count1);
+ else if (userbuf) {
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ if (copy_from_user(runtime->buffer + runtime->appl_ptr,
+ userbuf + result, count1)) {
+ spin_lock_irqsave(&runtime->lock, flags);
+ result = result > 0 ? result : -EFAULT;
+ goto __end;
+ }
+ spin_lock_irqsave(&runtime->lock, flags);
+ }
+ runtime->appl_ptr += count1;
+ runtime->appl_ptr %= runtime->buffer_size;
+ runtime->avail -= count1;
+ result += count1;
+ count -= count1;
+ }
+ __end:
+ count1 = runtime->avail < runtime->buffer_size;
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ if (count1)
+ snd_rawmidi_output_trigger(substream, 1);
+ return result;
+}
+
+long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream,
+ const unsigned char *buf, long count)
+{
+ return snd_rawmidi_kernel_write1(substream, NULL, buf, count);
+}
+
+static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf,
+ size_t count, loff_t *offset)
+{
+ long result, timeout;
+ int count1;
+ struct snd_rawmidi_file *rfile;
+ struct snd_rawmidi_runtime *runtime;
+ struct snd_rawmidi_substream *substream;
+
+ rfile = file->private_data;
+ substream = rfile->output;
+ runtime = substream->runtime;
+ /* we cannot put an atomic message to our buffer */
+ if (substream->append && count > runtime->buffer_size)
+ return -EIO;
+ result = 0;
+ while (count > 0) {
+ spin_lock_irq(&runtime->lock);
+ while (!snd_rawmidi_ready_append(substream, count)) {
+ wait_queue_t wait;
+ if (file->f_flags & O_NONBLOCK) {
+ spin_unlock_irq(&runtime->lock);
+ return result > 0 ? result : -EAGAIN;
+ }
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&runtime->sleep, &wait);
+ set_current_state(TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&runtime->lock);
+ timeout = schedule_timeout(30 * HZ);
+ remove_wait_queue(&runtime->sleep, &wait);
+ if (signal_pending(current))
+ return result > 0 ? result : -ERESTARTSYS;
+ if (!runtime->avail && !timeout)
+ return result > 0 ? result : -EIO;
+ spin_lock_irq(&runtime->lock);
+ }
+ spin_unlock_irq(&runtime->lock);
+ count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count);
+ if (count1 < 0)
+ return result > 0 ? result : count1;
+ result += count1;
+ buf += count1;
+ if ((size_t)count1 < count && (file->f_flags & O_NONBLOCK))
+ break;
+ count -= count1;
+ }
+ if (file->f_flags & O_SYNC) {
+ spin_lock_irq(&runtime->lock);
+ while (runtime->avail != runtime->buffer_size) {
+ wait_queue_t wait;
+ unsigned int last_avail = runtime->avail;
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&runtime->sleep, &wait);
+ set_current_state(TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&runtime->lock);
+ timeout = schedule_timeout(30 * HZ);
+ remove_wait_queue(&runtime->sleep, &wait);
+ if (signal_pending(current))
+ return result > 0 ? result : -ERESTARTSYS;
+ if (runtime->avail == last_avail && !timeout)
+ return result > 0 ? result : -EIO;
+ spin_lock_irq(&runtime->lock);
+ }
+ spin_unlock_irq(&runtime->lock);
+ }
+ return result;
+}
+
+static unsigned int snd_rawmidi_poll(struct file *file, poll_table * wait)
+{
+ struct snd_rawmidi_file *rfile;
+ struct snd_rawmidi_runtime *runtime;
+ unsigned int mask;
+
+ rfile = file->private_data;
+ if (rfile->input != NULL) {
+ runtime = rfile->input->runtime;
+ snd_rawmidi_input_trigger(rfile->input, 1);
+ poll_wait(file, &runtime->sleep, wait);
+ }
+ if (rfile->output != NULL) {
+ runtime = rfile->output->runtime;
+ poll_wait(file, &runtime->sleep, wait);
+ }
+ mask = 0;
+ if (rfile->input != NULL) {
+ if (snd_rawmidi_ready(rfile->input))
+ mask |= POLLIN | POLLRDNORM;
+ }
+ if (rfile->output != NULL) {
+ if (snd_rawmidi_ready(rfile->output))
+ mask |= POLLOUT | POLLWRNORM;
+ }
+ return mask;
+}
+
+/*
+ */
+#ifdef CONFIG_COMPAT
+#include "rawmidi_compat.c"
+#else
+#define snd_rawmidi_ioctl_compat NULL
+#endif
+
+/*
+
+ */
+
+static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream;
+ struct snd_rawmidi_runtime *runtime;
+
+ rmidi = entry->private_data;
+ snd_iprintf(buffer, "%s\n\n", rmidi->name);
+ mutex_lock(&rmidi->open_mutex);
+ if (rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT) {
+ list_for_each_entry(substream,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams,
+ list) {
+ snd_iprintf(buffer,
+ "Output %d\n"
+ " Tx bytes : %lu\n",
+ substream->number,
+ (unsigned long) substream->bytes);
+ if (substream->opened) {
+ runtime = substream->runtime;
+ snd_iprintf(buffer,
+ " Mode : %s\n"
+ " Buffer size : %lu\n"
+ " Avail : %lu\n",
+ runtime->oss ? "OSS compatible" : "native",
+ (unsigned long) runtime->buffer_size,
+ (unsigned long) runtime->avail);
+ }
+ }
+ }
+ if (rmidi->info_flags & SNDRV_RAWMIDI_INFO_INPUT) {
+ list_for_each_entry(substream,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams,
+ list) {
+ snd_iprintf(buffer,
+ "Input %d\n"
+ " Rx bytes : %lu\n",
+ substream->number,
+ (unsigned long) substream->bytes);
+ if (substream->opened) {
+ runtime = substream->runtime;
+ snd_iprintf(buffer,
+ " Buffer size : %lu\n"
+ " Avail : %lu\n"
+ " Overruns : %lu\n",
+ (unsigned long) runtime->buffer_size,
+ (unsigned long) runtime->avail,
+ (unsigned long) runtime->xruns);
+ }
+ }
+ }
+ mutex_unlock(&rmidi->open_mutex);
+}
+
+/*
+ * Register functions
+ */
+
+static const struct file_operations snd_rawmidi_f_ops =
+{
+ .owner = THIS_MODULE,
+ .read = snd_rawmidi_read,
+ .write = snd_rawmidi_write,
+ .open = snd_rawmidi_open,
+ .release = snd_rawmidi_release,
+ .poll = snd_rawmidi_poll,
+ .unlocked_ioctl = snd_rawmidi_ioctl,
+ .compat_ioctl = snd_rawmidi_ioctl_compat,
+};
+
+static int snd_rawmidi_alloc_substreams(struct snd_rawmidi *rmidi,
+ struct snd_rawmidi_str *stream,
+ int direction,
+ int count)
+{
+ struct snd_rawmidi_substream *substream;
+ int idx;
+
+ for (idx = 0; idx < count; idx++) {
+ substream = kzalloc(sizeof(*substream), GFP_KERNEL);
+ if (substream == NULL) {
+ snd_printk(KERN_ERR "rawmidi: cannot allocate substream\n");
+ return -ENOMEM;
+ }
+ substream->stream = direction;
+ substream->number = idx;
+ substream->rmidi = rmidi;
+ substream->pstr = stream;
+ list_add_tail(&substream->list, &stream->substreams);
+ stream->substream_count++;
+ }
+ return 0;
+}
+
+/**
+ * snd_rawmidi_new - create a rawmidi instance
+ * @card: the card instance
+ * @id: the id string
+ * @device: the device index
+ * @output_count: the number of output streams
+ * @input_count: the number of input streams
+ * @rrawmidi: the pointer to store the new rawmidi instance
+ *
+ * Creates a new rawmidi instance.
+ * Use snd_rawmidi_set_ops() to set the operators to the new instance.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_rawmidi_new(struct snd_card *card, char *id, int device,
+ int output_count, int input_count,
+ struct snd_rawmidi ** rrawmidi)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_rawmidi_dev_free,
+ .dev_register = snd_rawmidi_dev_register,
+ .dev_disconnect = snd_rawmidi_dev_disconnect,
+ };
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ rmidi = kzalloc(sizeof(*rmidi), GFP_KERNEL);
+ if (rmidi == NULL) {
+ snd_printk(KERN_ERR "rawmidi: cannot allocate\n");
+ return -ENOMEM;
+ }
+ rmidi->card = card;
+ rmidi->device = device;
+ mutex_init(&rmidi->open_mutex);
+ init_waitqueue_head(&rmidi->open_wait);
+ INIT_LIST_HEAD(&rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams);
+ INIT_LIST_HEAD(&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams);
+
+ if (id != NULL)
+ strlcpy(rmidi->id, id, sizeof(rmidi->id));
+ if ((err = snd_rawmidi_alloc_substreams(rmidi,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT],
+ SNDRV_RAWMIDI_STREAM_INPUT,
+ input_count)) < 0) {
+ snd_rawmidi_free(rmidi);
+ return err;
+ }
+ if ((err = snd_rawmidi_alloc_substreams(rmidi,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT],
+ SNDRV_RAWMIDI_STREAM_OUTPUT,
+ output_count)) < 0) {
+ snd_rawmidi_free(rmidi);
+ return err;
+ }
+ if ((err = snd_device_new(card, SNDRV_DEV_RAWMIDI, rmidi, &ops)) < 0) {
+ snd_rawmidi_free(rmidi);
+ return err;
+ }
+ if (rrawmidi)
+ *rrawmidi = rmidi;
+ return 0;
+}
+
+static void snd_rawmidi_free_substreams(struct snd_rawmidi_str *stream)
+{
+ struct snd_rawmidi_substream *substream;
+
+ while (!list_empty(&stream->substreams)) {
+ substream = list_entry(stream->substreams.next, struct snd_rawmidi_substream, list);
+ list_del(&substream->list);
+ kfree(substream);
+ }
+}
+
+static int snd_rawmidi_free(struct snd_rawmidi *rmidi)
+{
+ if (!rmidi)
+ return 0;
+
+ snd_info_free_entry(rmidi->proc_entry);
+ rmidi->proc_entry = NULL;
+ mutex_lock(&register_mutex);
+ if (rmidi->ops && rmidi->ops->dev_unregister)
+ rmidi->ops->dev_unregister(rmidi);
+ mutex_unlock(&register_mutex);
+
+ snd_rawmidi_free_substreams(&rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]);
+ snd_rawmidi_free_substreams(&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]);
+ if (rmidi->private_free)
+ rmidi->private_free(rmidi);
+ kfree(rmidi);
+ return 0;
+}
+
+static int snd_rawmidi_dev_free(struct snd_device *device)
+{
+ struct snd_rawmidi *rmidi = device->device_data;
+ return snd_rawmidi_free(rmidi);
+}
+
+#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))
+static void snd_rawmidi_dev_seq_free(struct snd_seq_device *device)
+{
+ struct snd_rawmidi *rmidi = device->private_data;
+ rmidi->seq_dev = NULL;
+}
+#endif
+
+static int snd_rawmidi_dev_register(struct snd_device *device)
+{
+ int err;
+ struct snd_info_entry *entry;
+ char name[16];
+ struct snd_rawmidi *rmidi = device->device_data;
+
+ if (rmidi->device >= SNDRV_RAWMIDI_DEVICES)
+ return -ENOMEM;
+ mutex_lock(&register_mutex);
+ if (snd_rawmidi_search(rmidi->card, rmidi->device)) {
+ mutex_unlock(&register_mutex);
+ return -EBUSY;
+ }
+ list_add_tail(&rmidi->list, &snd_rawmidi_devices);
+ sprintf(name, "midiC%iD%i", rmidi->card->number, rmidi->device);
+ if ((err = snd_register_device(SNDRV_DEVICE_TYPE_RAWMIDI,
+ rmidi->card, rmidi->device,
+ &snd_rawmidi_f_ops, rmidi, name)) < 0) {
+ snd_printk(KERN_ERR "unable to register rawmidi device %i:%i\n", rmidi->card->number, rmidi->device);
+ list_del(&rmidi->list);
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+ if (rmidi->ops && rmidi->ops->dev_register &&
+ (err = rmidi->ops->dev_register(rmidi)) < 0) {
+ snd_unregister_device(SNDRV_DEVICE_TYPE_RAWMIDI, rmidi->card, rmidi->device);
+ list_del(&rmidi->list);
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+#ifdef CONFIG_SND_OSSEMUL
+ rmidi->ossreg = 0;
+ if ((int)rmidi->device == midi_map[rmidi->card->number]) {
+ if (snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_MIDI,
+ rmidi->card, 0, &snd_rawmidi_f_ops,
+ rmidi, name) < 0) {
+ snd_printk(KERN_ERR "unable to register OSS rawmidi device %i:%i\n", rmidi->card->number, 0);
+ } else {
+ rmidi->ossreg++;
+#ifdef SNDRV_OSS_INFO_DEV_MIDI
+ snd_oss_info_register(SNDRV_OSS_INFO_DEV_MIDI, rmidi->card->number, rmidi->name);
+#endif
+ }
+ }
+ if ((int)rmidi->device == amidi_map[rmidi->card->number]) {
+ if (snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_MIDI,
+ rmidi->card, 1, &snd_rawmidi_f_ops,
+ rmidi, name) < 0) {
+ snd_printk(KERN_ERR "unable to register OSS rawmidi device %i:%i\n", rmidi->card->number, 1);
+ } else {
+ rmidi->ossreg++;
+ }
+ }
+#endif /* CONFIG_SND_OSSEMUL */
+ mutex_unlock(&register_mutex);
+ sprintf(name, "midi%d", rmidi->device);
+ entry = snd_info_create_card_entry(rmidi->card, name, rmidi->card->proc_root);
+ if (entry) {
+ entry->private_data = rmidi;
+ entry->c.text.read = snd_rawmidi_proc_info_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ rmidi->proc_entry = entry;
+#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))
+ if (!rmidi->ops || !rmidi->ops->dev_register) { /* own registration mechanism */
+ if (snd_seq_device_new(rmidi->card, rmidi->device, SNDRV_SEQ_DEV_ID_MIDISYNTH, 0, &rmidi->seq_dev) >= 0) {
+ rmidi->seq_dev->private_data = rmidi;
+ rmidi->seq_dev->private_free = snd_rawmidi_dev_seq_free;
+ sprintf(rmidi->seq_dev->name, "MIDI %d-%d", rmidi->card->number, rmidi->device);
+ snd_device_register(rmidi->card, rmidi->seq_dev);
+ }
+ }
+#endif
+ return 0;
+}
+
+static int snd_rawmidi_dev_disconnect(struct snd_device *device)
+{
+ struct snd_rawmidi *rmidi = device->device_data;
+
+ mutex_lock(&register_mutex);
+ list_del_init(&rmidi->list);
+#ifdef CONFIG_SND_OSSEMUL
+ if (rmidi->ossreg) {
+ if ((int)rmidi->device == midi_map[rmidi->card->number]) {
+ snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_MIDI, rmidi->card, 0);
+#ifdef SNDRV_OSS_INFO_DEV_MIDI
+ snd_oss_info_unregister(SNDRV_OSS_INFO_DEV_MIDI, rmidi->card->number);
+#endif
+ }
+ if ((int)rmidi->device == amidi_map[rmidi->card->number])
+ snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_MIDI, rmidi->card, 1);
+ rmidi->ossreg = 0;
+ }
+#endif /* CONFIG_SND_OSSEMUL */
+ snd_unregister_device(SNDRV_DEVICE_TYPE_RAWMIDI, rmidi->card, rmidi->device);
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+/**
+ * snd_rawmidi_set_ops - set the rawmidi operators
+ * @rmidi: the rawmidi instance
+ * @stream: the stream direction, SNDRV_RAWMIDI_STREAM_XXX
+ * @ops: the operator table
+ *
+ * Sets the rawmidi operators for the given stream direction.
+ */
+void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream,
+ struct snd_rawmidi_ops *ops)
+{
+ struct snd_rawmidi_substream *substream;
+
+ list_for_each_entry(substream, &rmidi->streams[stream].substreams, list)
+ substream->ops = ops;
+}
+
+/*
+ * ENTRY functions
+ */
+
+static int __init alsa_rawmidi_init(void)
+{
+
+ snd_ctl_register_ioctl(snd_rawmidi_control_ioctl);
+ snd_ctl_register_ioctl_compat(snd_rawmidi_control_ioctl);
+#ifdef CONFIG_SND_OSSEMUL
+ { int i;
+ /* check device map table */
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ if (midi_map[i] < 0 || midi_map[i] >= SNDRV_RAWMIDI_DEVICES) {
+ snd_printk(KERN_ERR "invalid midi_map[%d] = %d\n", i, midi_map[i]);
+ midi_map[i] = 0;
+ }
+ if (amidi_map[i] < 0 || amidi_map[i] >= SNDRV_RAWMIDI_DEVICES) {
+ snd_printk(KERN_ERR "invalid amidi_map[%d] = %d\n", i, amidi_map[i]);
+ amidi_map[i] = 1;
+ }
+ }
+ }
+#endif /* CONFIG_SND_OSSEMUL */
+ return 0;
+}
+
+static void __exit alsa_rawmidi_exit(void)
+{
+ snd_ctl_unregister_ioctl(snd_rawmidi_control_ioctl);
+ snd_ctl_unregister_ioctl_compat(snd_rawmidi_control_ioctl);
+}
+
+module_init(alsa_rawmidi_init)
+module_exit(alsa_rawmidi_exit)
+
+EXPORT_SYMBOL(snd_rawmidi_output_params);
+EXPORT_SYMBOL(snd_rawmidi_input_params);
+EXPORT_SYMBOL(snd_rawmidi_drop_output);
+EXPORT_SYMBOL(snd_rawmidi_drain_output);
+EXPORT_SYMBOL(snd_rawmidi_drain_input);
+EXPORT_SYMBOL(snd_rawmidi_receive);
+EXPORT_SYMBOL(snd_rawmidi_transmit_empty);
+EXPORT_SYMBOL(snd_rawmidi_transmit_peek);
+EXPORT_SYMBOL(snd_rawmidi_transmit_ack);
+EXPORT_SYMBOL(snd_rawmidi_transmit);
+EXPORT_SYMBOL(snd_rawmidi_new);
+EXPORT_SYMBOL(snd_rawmidi_set_ops);
+EXPORT_SYMBOL(snd_rawmidi_info_select);
+EXPORT_SYMBOL(snd_rawmidi_kernel_open);
+EXPORT_SYMBOL(snd_rawmidi_kernel_release);
+EXPORT_SYMBOL(snd_rawmidi_kernel_read);
+EXPORT_SYMBOL(snd_rawmidi_kernel_write);
diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c
new file mode 100644
index 0000000..5268c1f
--- /dev/null
+++ b/sound/core/rawmidi_compat.c
@@ -0,0 +1,120 @@
+/*
+ * 32bit -> 64bit ioctl wrapper for raw MIDI API
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* This file included from rawmidi.c */
+
+#include <linux/compat.h>
+
+struct snd_rawmidi_params32 {
+ s32 stream;
+ u32 buffer_size;
+ u32 avail_min;
+ unsigned int no_active_sensing; /* avoid bit-field */
+ unsigned char reserved[16];
+} __attribute__((packed));
+
+static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile,
+ struct snd_rawmidi_params32 __user *src)
+{
+ struct snd_rawmidi_params params;
+ unsigned int val;
+
+ if (rfile->output == NULL)
+ return -EINVAL;
+ if (get_user(params.stream, &src->stream) ||
+ get_user(params.buffer_size, &src->buffer_size) ||
+ get_user(params.avail_min, &src->avail_min) ||
+ get_user(val, &src->no_active_sensing))
+ return -EFAULT;
+ params.no_active_sensing = val;
+ switch (params.stream) {
+ case SNDRV_RAWMIDI_STREAM_OUTPUT:
+ return snd_rawmidi_output_params(rfile->output, &params);
+ case SNDRV_RAWMIDI_STREAM_INPUT:
+ return snd_rawmidi_input_params(rfile->input, &params);
+ }
+ return -EINVAL;
+}
+
+struct snd_rawmidi_status32 {
+ s32 stream;
+ struct compat_timespec tstamp;
+ u32 avail;
+ u32 xruns;
+ unsigned char reserved[16];
+} __attribute__((packed));
+
+static int snd_rawmidi_ioctl_status_compat(struct snd_rawmidi_file *rfile,
+ struct snd_rawmidi_status32 __user *src)
+{
+ int err;
+ struct snd_rawmidi_status status;
+
+ if (rfile->output == NULL)
+ return -EINVAL;
+ if (get_user(status.stream, &src->stream))
+ return -EFAULT;
+
+ switch (status.stream) {
+ case SNDRV_RAWMIDI_STREAM_OUTPUT:
+ err = snd_rawmidi_output_status(rfile->output, &status);
+ break;
+ case SNDRV_RAWMIDI_STREAM_INPUT:
+ err = snd_rawmidi_input_status(rfile->input, &status);
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (err < 0)
+ return err;
+
+ if (put_user(status.tstamp.tv_sec, &src->tstamp.tv_sec) ||
+ put_user(status.tstamp.tv_nsec, &src->tstamp.tv_nsec) ||
+ put_user(status.avail, &src->avail) ||
+ put_user(status.xruns, &src->xruns))
+ return -EFAULT;
+
+ return 0;
+}
+
+enum {
+ SNDRV_RAWMIDI_IOCTL_PARAMS32 = _IOWR('W', 0x10, struct snd_rawmidi_params32),
+ SNDRV_RAWMIDI_IOCTL_STATUS32 = _IOWR('W', 0x20, struct snd_rawmidi_status32),
+};
+
+static long snd_rawmidi_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_rawmidi_file *rfile;
+ void __user *argp = compat_ptr(arg);
+
+ rfile = file->private_data;
+ switch (cmd) {
+ case SNDRV_RAWMIDI_IOCTL_PVERSION:
+ case SNDRV_RAWMIDI_IOCTL_INFO:
+ case SNDRV_RAWMIDI_IOCTL_DROP:
+ case SNDRV_RAWMIDI_IOCTL_DRAIN:
+ return snd_rawmidi_ioctl(file, cmd, (unsigned long)argp);
+ case SNDRV_RAWMIDI_IOCTL_PARAMS32:
+ return snd_rawmidi_ioctl_params_compat(rfile, argp);
+ case SNDRV_RAWMIDI_IOCTL_STATUS32:
+ return snd_rawmidi_ioctl_status_compat(rfile, argp);
+ }
+ return -ENOIOCTLCMD;
+}
diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
new file mode 100644
index 0000000..51e64e3
--- /dev/null
+++ b/sound/core/rtctimer.c
@@ -0,0 +1,188 @@
+/*
+ * RTC based high-frequency timer
+ *
+ * Copyright (C) 2000 Takashi Iwai
+ * based on rtctimer.c by Steve Ratcliffe
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/log2.h>
+#include <sound/core.h>
+#include <sound/timer.h>
+
+#if defined(CONFIG_RTC) || defined(CONFIG_RTC_MODULE)
+
+#include <linux/mc146818rtc.h>
+
+#define RTC_FREQ 1024 /* default frequency */
+#define NANO_SEC 1000000000L /* 10^9 in sec */
+
+/*
+ * prototypes
+ */
+static int rtctimer_open(struct snd_timer *t);
+static int rtctimer_close(struct snd_timer *t);
+static int rtctimer_start(struct snd_timer *t);
+static int rtctimer_stop(struct snd_timer *t);
+
+
+/*
+ * The hardware dependent description for this timer.
+ */
+static struct snd_timer_hardware rtc_hw = {
+ .flags = SNDRV_TIMER_HW_AUTO |
+ SNDRV_TIMER_HW_FIRST |
+ SNDRV_TIMER_HW_TASKLET,
+ .ticks = 100000000L, /* FIXME: XXX */
+ .open = rtctimer_open,
+ .close = rtctimer_close,
+ .start = rtctimer_start,
+ .stop = rtctimer_stop,
+};
+
+static int rtctimer_freq = RTC_FREQ; /* frequency */
+static struct snd_timer *rtctimer;
+static struct tasklet_struct rtc_tasklet;
+static rtc_task_t rtc_task;
+
+
+static int
+rtctimer_open(struct snd_timer *t)
+{
+ int err;
+
+ err = rtc_register(&rtc_task);
+ if (err < 0)
+ return err;
+ t->private_data = &rtc_task;
+ return 0;
+}
+
+static int
+rtctimer_close(struct snd_timer *t)
+{
+ rtc_task_t *rtc = t->private_data;
+ if (rtc) {
+ rtc_unregister(rtc);
+ tasklet_kill(&rtc_tasklet);
+ t->private_data = NULL;
+ }
+ return 0;
+}
+
+static int
+rtctimer_start(struct snd_timer *timer)
+{
+ rtc_task_t *rtc = timer->private_data;
+ if (snd_BUG_ON(!rtc))
+ return -EINVAL;
+ rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq);
+ rtc_control(rtc, RTC_PIE_ON, 0);
+ return 0;
+}
+
+static int
+rtctimer_stop(struct snd_timer *timer)
+{
+ rtc_task_t *rtc = timer->private_data;
+ if (snd_BUG_ON(!rtc))
+ return -EINVAL;
+ rtc_control(rtc, RTC_PIE_OFF, 0);
+ return 0;
+}
+
+static void rtctimer_tasklet(unsigned long data)
+{
+ snd_timer_interrupt((struct snd_timer *)data, 1);
+}
+
+/*
+ * interrupt
+ */
+static void rtctimer_interrupt(void *private_data)
+{
+ tasklet_hi_schedule(private_data);
+}
+
+
+/*
+ * ENTRY functions
+ */
+static int __init rtctimer_init(void)
+{
+ int err;
+ struct snd_timer *timer;
+
+ if (rtctimer_freq < 2 || rtctimer_freq > 8192 ||
+ !is_power_of_2(rtctimer_freq)) {
+ snd_printk(KERN_ERR "rtctimer: invalid frequency %d\n",
+ rtctimer_freq);
+ return -EINVAL;
+ }
+
+ /* Create a new timer and set up the fields */
+ err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer);
+ if (err < 0)
+ return err;
+
+ timer->module = THIS_MODULE;
+ strcpy(timer->name, "RTC timer");
+ timer->hw = rtc_hw;
+ timer->hw.resolution = NANO_SEC / rtctimer_freq;
+
+ tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer);
+
+ /* set up RTC callback */
+ rtc_task.func = rtctimer_interrupt;
+ rtc_task.private_data = &rtc_tasklet;
+
+ err = snd_timer_global_register(timer);
+ if (err < 0) {
+ snd_timer_global_free(timer);
+ return err;
+ }
+ rtctimer = timer; /* remember this */
+
+ return 0;
+}
+
+static void __exit rtctimer_exit(void)
+{
+ if (rtctimer) {
+ snd_timer_global_free(rtctimer);
+ rtctimer = NULL;
+ }
+}
+
+
+/*
+ * exported stuff
+ */
+module_init(rtctimer_init)
+module_exit(rtctimer_exit)
+
+module_param(rtctimer_freq, int, 0444);
+MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz");
+
+MODULE_LICENSE("GPL");
+
+MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC));
+
+#endif /* CONFIG_RTC || CONFIG_RTC_MODULE */
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
new file mode 100644
index 0000000..0695937
--- /dev/null
+++ b/sound/core/seq/Makefile
@@ -0,0 +1,40 @@
+#
+# Makefile for ALSA
+# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
+#
+
+ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
+ obj-$(CONFIG_SND_SEQUENCER) += oss/
+endif
+
+snd-seq-device-objs := seq_device.o
+snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \
+ seq_fifo.o seq_prioq.o seq_timer.o \
+ seq_system.o seq_ports.o seq_info.o
+snd-seq-midi-objs := seq_midi.o
+snd-seq-midi-emul-objs := seq_midi_emul.o
+snd-seq-midi-event-objs := seq_midi_event.o
+snd-seq-dummy-objs := seq_dummy.o
+snd-seq-virmidi-objs := seq_virmidi.o
+
+#
+# this function returns:
+# "m" - CONFIG_SND_SEQUENCER is m
+# <empty string> - CONFIG_SND_SEQUENCER is undefined
+# otherwise parameter #1 value
+#
+sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+
+obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o
+ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
+obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
+endif
+obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o
+obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o
+obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o
+obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o
+obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o
+obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o
diff --git a/sound/core/seq/oss/Makefile b/sound/core/seq/oss/Makefile
new file mode 100644
index 0000000..b38406b
--- /dev/null
+++ b/sound/core/seq/oss/Makefile
@@ -0,0 +1,10 @@
+#
+# Makefile for ALSA
+# Copyright (c) 1999 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-seq-oss-objs := seq_oss.o seq_oss_init.o seq_oss_timer.o seq_oss_ioctl.o \
+ seq_oss_event.o seq_oss_rw.o seq_oss_synth.o \
+ seq_oss_midi.o seq_oss_readq.o seq_oss_writeq.o
+
+obj-$(CONFIG_SND_SEQUENCER) += snd-seq-oss.o
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
new file mode 100644
index 0000000..f25e3cc
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss.c
@@ -0,0 +1,311 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * registration of device and proc
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/initval.h>
+#include "seq_oss_device.h"
+#include "seq_oss_synth.h"
+
+/*
+ * module option
+ */
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("OSS-compatible sequencer module");
+MODULE_LICENSE("GPL");
+/* Takashi says this is really only for sound-service-0-, but this is OK. */
+MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_SEQUENCER);
+MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_MUSIC);
+
+#ifdef SNDRV_SEQ_OSS_DEBUG
+module_param(seq_oss_debug, int, 0644);
+MODULE_PARM_DESC(seq_oss_debug, "debug option");
+int seq_oss_debug = 0;
+#endif
+
+
+/*
+ * prototypes
+ */
+static int register_device(void);
+static void unregister_device(void);
+#ifdef CONFIG_PROC_FS
+static int register_proc(void);
+static void unregister_proc(void);
+#else
+static inline int register_proc(void) { return 0; }
+static inline void unregister_proc(void) {}
+#endif
+
+static int odev_open(struct inode *inode, struct file *file);
+static int odev_release(struct inode *inode, struct file *file);
+static ssize_t odev_read(struct file *file, char __user *buf, size_t count, loff_t *offset);
+static ssize_t odev_write(struct file *file, const char __user *buf, size_t count, loff_t *offset);
+static long odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg);
+static unsigned int odev_poll(struct file *file, poll_table * wait);
+
+
+/*
+ * module interface
+ */
+
+static int __init alsa_seq_oss_init(void)
+{
+ int rc;
+ static struct snd_seq_dev_ops ops = {
+ snd_seq_oss_synth_register,
+ snd_seq_oss_synth_unregister,
+ };
+
+ snd_seq_autoload_lock();
+ if ((rc = register_device()) < 0)
+ goto error;
+ if ((rc = register_proc()) < 0) {
+ unregister_device();
+ goto error;
+ }
+ if ((rc = snd_seq_oss_create_client()) < 0) {
+ unregister_proc();
+ unregister_device();
+ goto error;
+ }
+
+ if ((rc = snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OSS, &ops,
+ sizeof(struct snd_seq_oss_reg))) < 0) {
+ snd_seq_oss_delete_client();
+ unregister_proc();
+ unregister_device();
+ goto error;
+ }
+
+ /* success */
+ snd_seq_oss_synth_init();
+
+ error:
+ snd_seq_autoload_unlock();
+ return rc;
+}
+
+static void __exit alsa_seq_oss_exit(void)
+{
+ snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OSS);
+ snd_seq_oss_delete_client();
+ unregister_proc();
+ unregister_device();
+}
+
+module_init(alsa_seq_oss_init)
+module_exit(alsa_seq_oss_exit)
+
+/*
+ * ALSA minor device interface
+ */
+
+static DEFINE_MUTEX(register_mutex);
+
+static int
+odev_open(struct inode *inode, struct file *file)
+{
+ int level, rc;
+
+ if (iminor(inode) == SNDRV_MINOR_OSS_MUSIC)
+ level = SNDRV_SEQ_OSS_MODE_MUSIC;
+ else
+ level = SNDRV_SEQ_OSS_MODE_SYNTH;
+
+ mutex_lock(&register_mutex);
+ rc = snd_seq_oss_open(file, level);
+ mutex_unlock(&register_mutex);
+
+ return rc;
+}
+
+static int
+odev_release(struct inode *inode, struct file *file)
+{
+ struct seq_oss_devinfo *dp;
+
+ if ((dp = file->private_data) == NULL)
+ return 0;
+
+ snd_seq_oss_drain_write(dp);
+
+ mutex_lock(&register_mutex);
+ snd_seq_oss_release(dp);
+ mutex_unlock(&register_mutex);
+
+ return 0;
+}
+
+static ssize_t
+odev_read(struct file *file, char __user *buf, size_t count, loff_t *offset)
+{
+ struct seq_oss_devinfo *dp;
+ dp = file->private_data;
+ if (snd_BUG_ON(!dp))
+ return -ENXIO;
+ return snd_seq_oss_read(dp, buf, count);
+}
+
+
+static ssize_t
+odev_write(struct file *file, const char __user *buf, size_t count, loff_t *offset)
+{
+ struct seq_oss_devinfo *dp;
+ dp = file->private_data;
+ if (snd_BUG_ON(!dp))
+ return -ENXIO;
+ return snd_seq_oss_write(dp, buf, count, file);
+}
+
+static long
+odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct seq_oss_devinfo *dp;
+ dp = file->private_data;
+ if (snd_BUG_ON(!dp))
+ return -ENXIO;
+ return snd_seq_oss_ioctl(dp, cmd, arg);
+}
+
+#ifdef CONFIG_COMPAT
+#define odev_ioctl_compat odev_ioctl
+#else
+#define odev_ioctl_compat NULL
+#endif
+
+static unsigned int
+odev_poll(struct file *file, poll_table * wait)
+{
+ struct seq_oss_devinfo *dp;
+ dp = file->private_data;
+ if (snd_BUG_ON(!dp))
+ return -ENXIO;
+ return snd_seq_oss_poll(dp, file, wait);
+}
+
+/*
+ * registration of sequencer minor device
+ */
+
+static const struct file_operations seq_oss_f_ops =
+{
+ .owner = THIS_MODULE,
+ .read = odev_read,
+ .write = odev_write,
+ .open = odev_open,
+ .release = odev_release,
+ .poll = odev_poll,
+ .unlocked_ioctl = odev_ioctl,
+ .compat_ioctl = odev_ioctl_compat,
+};
+
+static int __init
+register_device(void)
+{
+ int rc;
+
+ mutex_lock(&register_mutex);
+ if ((rc = snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_SEQUENCER,
+ NULL, 0,
+ &seq_oss_f_ops, NULL,
+ SNDRV_SEQ_OSS_DEVNAME)) < 0) {
+ snd_printk(KERN_ERR "can't register device seq\n");
+ mutex_unlock(&register_mutex);
+ return rc;
+ }
+ if ((rc = snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_MUSIC,
+ NULL, 0,
+ &seq_oss_f_ops, NULL,
+ SNDRV_SEQ_OSS_DEVNAME)) < 0) {
+ snd_printk(KERN_ERR "can't register device music\n");
+ snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_SEQUENCER, NULL, 0);
+ mutex_unlock(&register_mutex);
+ return rc;
+ }
+ debug_printk(("device registered\n"));
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+static void
+unregister_device(void)
+{
+ mutex_lock(&register_mutex);
+ debug_printk(("device unregistered\n"));
+ if (snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_MUSIC, NULL, 0) < 0)
+ snd_printk(KERN_ERR "error unregister device music\n");
+ if (snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_SEQUENCER, NULL, 0) < 0)
+ snd_printk(KERN_ERR "error unregister device seq\n");
+ mutex_unlock(&register_mutex);
+}
+
+/*
+ * /proc interface
+ */
+
+#ifdef CONFIG_PROC_FS
+
+static struct snd_info_entry *info_entry;
+
+static void
+info_read(struct snd_info_entry *entry, struct snd_info_buffer *buf)
+{
+ mutex_lock(&register_mutex);
+ snd_iprintf(buf, "OSS sequencer emulation version %s\n", SNDRV_SEQ_OSS_VERSION_STR);
+ snd_seq_oss_system_info_read(buf);
+ snd_seq_oss_synth_info_read(buf);
+ snd_seq_oss_midi_info_read(buf);
+ mutex_unlock(&register_mutex);
+}
+
+
+static int __init
+register_proc(void)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_module_entry(THIS_MODULE, SNDRV_SEQ_OSS_PROCNAME, snd_seq_root);
+ if (entry == NULL)
+ return -ENOMEM;
+
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = NULL;
+ entry->c.text.read = info_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return -ENOMEM;
+ }
+ info_entry = entry;
+ return 0;
+}
+
+static void
+unregister_proc(void)
+{
+ snd_info_free_entry(info_entry);
+ info_entry = NULL;
+}
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h
new file mode 100644
index 0000000..bf8d2b4
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_device.h
@@ -0,0 +1,189 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SEQ_OSS_DEVICE_H
+#define __SEQ_OSS_DEVICE_H
+
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/slab.h>
+#include <linux/sched.h>
+#include <sound/core.h>
+#include <sound/seq_oss.h>
+#include <sound/rawmidi.h>
+#include <sound/seq_kernel.h>
+#include <sound/info.h>
+
+/* enable debug print */
+#define SNDRV_SEQ_OSS_DEBUG
+
+/* max. applications */
+#define SNDRV_SEQ_OSS_MAX_CLIENTS 16
+#define SNDRV_SEQ_OSS_MAX_SYNTH_DEVS 16
+#define SNDRV_SEQ_OSS_MAX_MIDI_DEVS 32
+
+/* version */
+#define SNDRV_SEQ_OSS_MAJOR_VERSION 0
+#define SNDRV_SEQ_OSS_MINOR_VERSION 1
+#define SNDRV_SEQ_OSS_TINY_VERSION 8
+#define SNDRV_SEQ_OSS_VERSION_STR "0.1.8"
+
+/* device and proc interface name */
+#define SNDRV_SEQ_OSS_DEVNAME "seq_oss"
+#define SNDRV_SEQ_OSS_PROCNAME "oss"
+
+
+/*
+ * type definitions
+ */
+
+typedef unsigned int reltime_t;
+typedef unsigned int abstime_t;
+
+
+/*
+ * synthesizer channel information
+ */
+struct seq_oss_chinfo {
+ int note, vel;
+};
+
+/*
+ * synthesizer information
+ */
+struct seq_oss_synthinfo {
+ struct snd_seq_oss_arg arg;
+ struct seq_oss_chinfo *ch;
+ struct seq_oss_synth_sysex *sysex;
+ int nr_voices;
+ int opened;
+ int is_midi;
+ int midi_mapped;
+};
+
+
+/*
+ * sequencer client information
+ */
+
+struct seq_oss_devinfo {
+
+ int index; /* application index */
+ int cseq; /* sequencer client number */
+ int port; /* sequencer port number */
+ int queue; /* sequencer queue number */
+
+ struct snd_seq_addr addr; /* address of this device */
+
+ int seq_mode; /* sequencer mode */
+ int file_mode; /* file access */
+
+ /* midi device table */
+ int max_mididev;
+
+ /* synth device table */
+ int max_synthdev;
+ struct seq_oss_synthinfo synths[SNDRV_SEQ_OSS_MAX_SYNTH_DEVS];
+ int synth_opened;
+
+ /* output queue */
+ struct seq_oss_writeq *writeq;
+
+ /* midi input queue */
+ struct seq_oss_readq *readq;
+
+ /* timer */
+ struct seq_oss_timer *timer;
+};
+
+
+/*
+ * function prototypes
+ */
+
+/* create/delete OSS sequencer client */
+int snd_seq_oss_create_client(void);
+int snd_seq_oss_delete_client(void);
+
+/* device file interface */
+int snd_seq_oss_open(struct file *file, int level);
+void snd_seq_oss_release(struct seq_oss_devinfo *dp);
+int snd_seq_oss_ioctl(struct seq_oss_devinfo *dp, unsigned int cmd, unsigned long arg);
+int snd_seq_oss_read(struct seq_oss_devinfo *dev, char __user *buf, int count);
+int snd_seq_oss_write(struct seq_oss_devinfo *dp, const char __user *buf, int count, struct file *opt);
+unsigned int snd_seq_oss_poll(struct seq_oss_devinfo *dp, struct file *file, poll_table * wait);
+
+void snd_seq_oss_reset(struct seq_oss_devinfo *dp);
+void snd_seq_oss_drain_write(struct seq_oss_devinfo *dp);
+
+/* */
+void snd_seq_oss_process_queue(struct seq_oss_devinfo *dp, abstime_t time);
+
+
+/* proc interface */
+void snd_seq_oss_system_info_read(struct snd_info_buffer *buf);
+void snd_seq_oss_midi_info_read(struct snd_info_buffer *buf);
+void snd_seq_oss_synth_info_read(struct snd_info_buffer *buf);
+void snd_seq_oss_readq_info_read(struct seq_oss_readq *q, struct snd_info_buffer *buf);
+
+/* file mode macros */
+#define is_read_mode(mode) ((mode) & SNDRV_SEQ_OSS_FILE_READ)
+#define is_write_mode(mode) ((mode) & SNDRV_SEQ_OSS_FILE_WRITE)
+#define is_nonblock_mode(mode) ((mode) & SNDRV_SEQ_OSS_FILE_NONBLOCK)
+
+/* dispatch event */
+static inline int
+snd_seq_oss_dispatch(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, int atomic, int hop)
+{
+ return snd_seq_kernel_client_dispatch(dp->cseq, ev, atomic, hop);
+}
+
+/* ioctl */
+static inline int
+snd_seq_oss_control(struct seq_oss_devinfo *dp, unsigned int type, void *arg)
+{
+ return snd_seq_kernel_client_ctl(dp->cseq, type, arg);
+}
+
+/* fill the addresses in header */
+static inline void
+snd_seq_oss_fill_addr(struct seq_oss_devinfo *dp, struct snd_seq_event *ev,
+ int dest_client, int dest_port)
+{
+ ev->queue = dp->queue;
+ ev->source = dp->addr;
+ ev->dest.client = dest_client;
+ ev->dest.port = dest_port;
+}
+
+
+/* misc. functions for proc interface */
+char *enabled_str(int bool);
+
+
+/* for debug */
+#ifdef SNDRV_SEQ_OSS_DEBUG
+extern int seq_oss_debug;
+#define debug_printk(x) do { if (seq_oss_debug > 0) snd_printk x; } while (0)
+#else
+#define debug_printk(x) /**/
+#endif
+
+#endif /* __SEQ_OSS_DEVICE_H */
diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c
new file mode 100644
index 0000000..066f5f3
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_event.c
@@ -0,0 +1,447 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_device.h"
+#include "seq_oss_synth.h"
+#include "seq_oss_midi.h"
+#include "seq_oss_event.h"
+#include "seq_oss_timer.h"
+#include <sound/seq_oss_legacy.h>
+#include "seq_oss_readq.h"
+#include "seq_oss_writeq.h"
+
+
+/*
+ * prototypes
+ */
+static int extended_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev);
+static int chn_voice_event(struct seq_oss_devinfo *dp, union evrec *event_rec, struct snd_seq_event *ev);
+static int chn_common_event(struct seq_oss_devinfo *dp, union evrec *event_rec, struct snd_seq_event *ev);
+static int timing_event(struct seq_oss_devinfo *dp, union evrec *event_rec, struct snd_seq_event *ev);
+static int local_event(struct seq_oss_devinfo *dp, union evrec *event_rec, struct snd_seq_event *ev);
+static int old_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev);
+static int note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev);
+static int note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev);
+static int set_note_event(struct seq_oss_devinfo *dp, int dev, int type, int ch, int note, int vel, struct snd_seq_event *ev);
+static int set_control_event(struct seq_oss_devinfo *dp, int dev, int type, int ch, int param, int val, struct snd_seq_event *ev);
+static int set_echo_event(struct seq_oss_devinfo *dp, union evrec *rec, struct snd_seq_event *ev);
+
+
+/*
+ * convert an OSS event to ALSA event
+ * return 0 : enqueued
+ * non-zero : invalid - ignored
+ */
+
+int
+snd_seq_oss_process_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev)
+{
+ switch (q->s.code) {
+ case SEQ_EXTENDED:
+ return extended_event(dp, q, ev);
+
+ case EV_CHN_VOICE:
+ return chn_voice_event(dp, q, ev);
+
+ case EV_CHN_COMMON:
+ return chn_common_event(dp, q, ev);
+
+ case EV_TIMING:
+ return timing_event(dp, q, ev);
+
+ case EV_SEQ_LOCAL:
+ return local_event(dp, q, ev);
+
+ case EV_SYSEX:
+ return snd_seq_oss_synth_sysex(dp, q->x.dev, q->x.buf, ev);
+
+ case SEQ_MIDIPUTC:
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC)
+ return -EINVAL;
+ /* put a midi byte */
+ if (! is_write_mode(dp->file_mode))
+ break;
+ if (snd_seq_oss_midi_open(dp, q->s.dev, SNDRV_SEQ_OSS_FILE_WRITE))
+ break;
+ if (snd_seq_oss_midi_filemode(dp, q->s.dev) & SNDRV_SEQ_OSS_FILE_WRITE)
+ return snd_seq_oss_midi_putc(dp, q->s.dev, q->s.parm1, ev);
+ break;
+
+ case SEQ_ECHO:
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC)
+ return -EINVAL;
+ return set_echo_event(dp, q, ev);
+
+ case SEQ_PRIVATE:
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC)
+ return -EINVAL;
+ return snd_seq_oss_synth_raw_event(dp, q->c[1], q->c, ev);
+
+ default:
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC)
+ return -EINVAL;
+ return old_event(dp, q, ev);
+ }
+ return -EINVAL;
+}
+
+/* old type events: mode1 only */
+static int
+old_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev)
+{
+ switch (q->s.code) {
+ case SEQ_NOTEOFF:
+ return note_off_event(dp, 0, q->n.chn, q->n.note, q->n.vel, ev);
+
+ case SEQ_NOTEON:
+ return note_on_event(dp, 0, q->n.chn, q->n.note, q->n.vel, ev);
+
+ case SEQ_WAIT:
+ /* skip */
+ break;
+
+ case SEQ_PGMCHANGE:
+ return set_control_event(dp, 0, SNDRV_SEQ_EVENT_PGMCHANGE,
+ q->n.chn, 0, q->n.note, ev);
+
+ case SEQ_SYNCTIMER:
+ return snd_seq_oss_timer_reset(dp->timer);
+ }
+
+ return -EINVAL;
+}
+
+/* 8bytes extended event: mode1 only */
+static int
+extended_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev)
+{
+ int val;
+
+ switch (q->e.cmd) {
+ case SEQ_NOTEOFF:
+ return note_off_event(dp, q->e.dev, q->e.chn, q->e.p1, q->e.p2, ev);
+
+ case SEQ_NOTEON:
+ return note_on_event(dp, q->e.dev, q->e.chn, q->e.p1, q->e.p2, ev);
+
+ case SEQ_PGMCHANGE:
+ return set_control_event(dp, q->e.dev, SNDRV_SEQ_EVENT_PGMCHANGE,
+ q->e.chn, 0, q->e.p1, ev);
+
+ case SEQ_AFTERTOUCH:
+ return set_control_event(dp, q->e.dev, SNDRV_SEQ_EVENT_CHANPRESS,
+ q->e.chn, 0, q->e.p1, ev);
+
+ case SEQ_BALANCE:
+ /* convert -128:127 to 0:127 */
+ val = (char)q->e.p1;
+ val = (val + 128) / 2;
+ return set_control_event(dp, q->e.dev, SNDRV_SEQ_EVENT_CONTROLLER,
+ q->e.chn, CTL_PAN, val, ev);
+
+ case SEQ_CONTROLLER:
+ val = ((short)q->e.p3 << 8) | (short)q->e.p2;
+ switch (q->e.p1) {
+ case CTRL_PITCH_BENDER: /* SEQ1 V2 control */
+ /* -0x2000:0x1fff */
+ return set_control_event(dp, q->e.dev,
+ SNDRV_SEQ_EVENT_PITCHBEND,
+ q->e.chn, 0, val, ev);
+ case CTRL_PITCH_BENDER_RANGE:
+ /* conversion: 100/semitone -> 128/semitone */
+ return set_control_event(dp, q->e.dev,
+ SNDRV_SEQ_EVENT_REGPARAM,
+ q->e.chn, 0, val*128/100, ev);
+ default:
+ return set_control_event(dp, q->e.dev,
+ SNDRV_SEQ_EVENT_CONTROL14,
+ q->e.chn, q->e.p1, val, ev);
+ }
+
+ case SEQ_VOLMODE:
+ return snd_seq_oss_synth_raw_event(dp, q->e.dev, q->c, ev);
+
+ }
+ return -EINVAL;
+}
+
+/* channel voice events: mode1 and 2 */
+static int
+chn_voice_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev)
+{
+ if (q->v.chn >= 32)
+ return -EINVAL;
+ switch (q->v.cmd) {
+ case MIDI_NOTEON:
+ return note_on_event(dp, q->v.dev, q->v.chn, q->v.note, q->v.parm, ev);
+
+ case MIDI_NOTEOFF:
+ return note_off_event(dp, q->v.dev, q->v.chn, q->v.note, q->v.parm, ev);
+
+ case MIDI_KEY_PRESSURE:
+ return set_note_event(dp, q->v.dev, SNDRV_SEQ_EVENT_KEYPRESS,
+ q->v.chn, q->v.note, q->v.parm, ev);
+
+ }
+ return -EINVAL;
+}
+
+/* channel common events: mode1 and 2 */
+static int
+chn_common_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev)
+{
+ if (q->l.chn >= 32)
+ return -EINVAL;
+ switch (q->l.cmd) {
+ case MIDI_PGM_CHANGE:
+ return set_control_event(dp, q->l.dev, SNDRV_SEQ_EVENT_PGMCHANGE,
+ q->l.chn, 0, q->l.p1, ev);
+
+ case MIDI_CTL_CHANGE:
+ return set_control_event(dp, q->l.dev, SNDRV_SEQ_EVENT_CONTROLLER,
+ q->l.chn, q->l.p1, q->l.val, ev);
+
+ case MIDI_PITCH_BEND:
+ /* conversion: 0:0x3fff -> -0x2000:0x1fff */
+ return set_control_event(dp, q->l.dev, SNDRV_SEQ_EVENT_PITCHBEND,
+ q->l.chn, 0, q->l.val - 8192, ev);
+
+ case MIDI_CHN_PRESSURE:
+ return set_control_event(dp, q->l.dev, SNDRV_SEQ_EVENT_CHANPRESS,
+ q->l.chn, 0, q->l.val, ev);
+ }
+ return -EINVAL;
+}
+
+/* timer events: mode1 and mode2 */
+static int
+timing_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev)
+{
+ switch (q->t.cmd) {
+ case TMR_ECHO:
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC)
+ return set_echo_event(dp, q, ev);
+ else {
+ union evrec tmp;
+ memset(&tmp, 0, sizeof(tmp));
+ /* XXX: only for little-endian! */
+ tmp.echo = (q->t.time << 8) | SEQ_ECHO;
+ return set_echo_event(dp, &tmp, ev);
+ }
+
+ case TMR_STOP:
+ if (dp->seq_mode)
+ return snd_seq_oss_timer_stop(dp->timer);
+ return 0;
+
+ case TMR_CONTINUE:
+ if (dp->seq_mode)
+ return snd_seq_oss_timer_continue(dp->timer);
+ return 0;
+
+ case TMR_TEMPO:
+ if (dp->seq_mode)
+ return snd_seq_oss_timer_tempo(dp->timer, q->t.time);
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
+/* local events: mode1 and 2 */
+static int
+local_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev)
+{
+ return -EINVAL;
+}
+
+/*
+ * process note-on event for OSS synth
+ * three different modes are available:
+ * - SNDRV_SEQ_OSS_PROCESS_EVENTS (for one-voice per channel mode)
+ * Accept note 255 as volume change.
+ * - SNDRV_SEQ_OSS_PASS_EVENTS
+ * Pass all events to lowlevel driver anyway
+ * - SNDRV_SEQ_OSS_PROCESS_KEYPRESS (mostly for Emu8000)
+ * Use key-pressure if note >= 128
+ */
+static int
+note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
+{
+ struct seq_oss_synthinfo *info = &dp->synths[dev];
+ switch (info->arg.event_passing) {
+ case SNDRV_SEQ_OSS_PROCESS_EVENTS:
+ if (! info->ch || ch < 0 || ch >= info->nr_voices) {
+ /* pass directly */
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEON, ch, note, vel, ev);
+ }
+
+ if (note == 255 && info->ch[ch].note >= 0) {
+ /* volume control */
+ int type;
+ //if (! vel)
+ /* set volume to zero -- note off */
+ // type = SNDRV_SEQ_EVENT_NOTEOFF;
+ //else
+ if (info->ch[ch].vel)
+ /* sample already started -- volume change */
+ type = SNDRV_SEQ_EVENT_KEYPRESS;
+ else
+ /* sample not started -- start now */
+ type = SNDRV_SEQ_EVENT_NOTEON;
+ info->ch[ch].vel = vel;
+ return set_note_event(dp, dev, type, ch, info->ch[ch].note, vel, ev);
+ } else if (note >= 128)
+ return -EINVAL; /* invalid */
+
+ if (note != info->ch[ch].note && info->ch[ch].note >= 0)
+ /* note changed - note off at beginning */
+ set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEOFF, ch, info->ch[ch].note, 0, ev);
+ /* set current status */
+ info->ch[ch].note = note;
+ info->ch[ch].vel = vel;
+ if (vel) /* non-zero velocity - start the note now */
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEON, ch, note, vel, ev);
+ return -EINVAL;
+
+ case SNDRV_SEQ_OSS_PASS_EVENTS:
+ /* pass the event anyway */
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEON, ch, note, vel, ev);
+
+ case SNDRV_SEQ_OSS_PROCESS_KEYPRESS:
+ if (note >= 128) /* key pressure: shifted by 128 */
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_KEYPRESS, ch, note - 128, vel, ev);
+ else /* normal note-on event */
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEON, ch, note, vel, ev);
+ }
+ return -EINVAL;
+}
+
+/*
+ * process note-off event for OSS synth
+ */
+static int
+note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, struct snd_seq_event *ev)
+{
+ struct seq_oss_synthinfo *info = &dp->synths[dev];
+ switch (info->arg.event_passing) {
+ case SNDRV_SEQ_OSS_PROCESS_EVENTS:
+ if (! info->ch || ch < 0 || ch >= info->nr_voices) {
+ /* pass directly */
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEON, ch, note, vel, ev);
+ }
+
+ if (info->ch[ch].note >= 0) {
+ note = info->ch[ch].note;
+ info->ch[ch].vel = 0;
+ info->ch[ch].note = -1;
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEOFF, ch, note, vel, ev);
+ }
+ return -EINVAL; /* invalid */
+
+ case SNDRV_SEQ_OSS_PASS_EVENTS:
+ case SNDRV_SEQ_OSS_PROCESS_KEYPRESS:
+ /* pass the event anyway */
+ return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEOFF, ch, note, vel, ev);
+
+ }
+ return -EINVAL;
+}
+
+/*
+ * create a note event
+ */
+static int
+set_note_event(struct seq_oss_devinfo *dp, int dev, int type, int ch, int note, int vel, struct snd_seq_event *ev)
+{
+ if (! snd_seq_oss_synth_is_valid(dp, dev))
+ return -ENXIO;
+
+ ev->type = type;
+ snd_seq_oss_synth_addr(dp, dev, ev);
+ ev->data.note.channel = ch;
+ ev->data.note.note = note;
+ ev->data.note.velocity = vel;
+
+ return 0;
+}
+
+/*
+ * create a control event
+ */
+static int
+set_control_event(struct seq_oss_devinfo *dp, int dev, int type, int ch, int param, int val, struct snd_seq_event *ev)
+{
+ if (! snd_seq_oss_synth_is_valid(dp, dev))
+ return -ENXIO;
+
+ ev->type = type;
+ snd_seq_oss_synth_addr(dp, dev, ev);
+ ev->data.control.channel = ch;
+ ev->data.control.param = param;
+ ev->data.control.value = val;
+
+ return 0;
+}
+
+/*
+ * create an echo event
+ */
+static int
+set_echo_event(struct seq_oss_devinfo *dp, union evrec *rec, struct snd_seq_event *ev)
+{
+ ev->type = SNDRV_SEQ_EVENT_ECHO;
+ /* echo back to itself */
+ snd_seq_oss_fill_addr(dp, ev, dp->addr.client, dp->addr.port);
+ memcpy(&ev->data, rec, LONG_EVENT_SIZE);
+ return 0;
+}
+
+/*
+ * event input callback from ALSA sequencer:
+ * the echo event is processed here.
+ */
+int
+snd_seq_oss_event_input(struct snd_seq_event *ev, int direct, void *private_data,
+ int atomic, int hop)
+{
+ struct seq_oss_devinfo *dp = (struct seq_oss_devinfo *)private_data;
+ union evrec *rec;
+
+ if (ev->type != SNDRV_SEQ_EVENT_ECHO)
+ return snd_seq_oss_midi_input(ev, direct, private_data);
+
+ if (ev->source.client != dp->cseq)
+ return 0; /* ignored */
+
+ rec = (union evrec*)&ev->data;
+ if (rec->s.code == SEQ_SYNCTIMER) {
+ /* sync echo back */
+ snd_seq_oss_writeq_wakeup(dp->writeq, rec->t.time);
+
+ } else {
+ /* echo back event */
+ if (dp->readq == NULL)
+ return 0;
+ snd_seq_oss_readq_put_event(dp->readq, rec);
+ }
+ return 0;
+}
+
diff --git a/sound/core/seq/oss/seq_oss_event.h b/sound/core/seq/oss/seq_oss_event.h
new file mode 100644
index 0000000..9a4d9ad
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_event.h
@@ -0,0 +1,112 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * seq_oss_event.h - OSS event queue record
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SEQ_OSS_EVENT_H
+#define __SEQ_OSS_EVENT_H
+
+#include "seq_oss_device.h"
+
+#define SHORT_EVENT_SIZE 4
+#define LONG_EVENT_SIZE 8
+
+/* short event (4bytes) */
+struct evrec_short {
+ unsigned char code;
+ unsigned char parm1;
+ unsigned char dev;
+ unsigned char parm2;
+};
+
+/* short note events (4bytes) */
+struct evrec_note {
+ unsigned char code;
+ unsigned char chn;
+ unsigned char note;
+ unsigned char vel;
+};
+
+/* long timer events (8bytes) */
+struct evrec_timer {
+ unsigned char code;
+ unsigned char cmd;
+ unsigned char dummy1, dummy2;
+ unsigned int time;
+};
+
+/* long extended events (8bytes) */
+struct evrec_extended {
+ unsigned char code;
+ unsigned char cmd;
+ unsigned char dev;
+ unsigned char chn;
+ unsigned char p1, p2, p3, p4;
+};
+
+/* long channel events (8bytes) */
+struct evrec_long {
+ unsigned char code;
+ unsigned char dev;
+ unsigned char cmd;
+ unsigned char chn;
+ unsigned char p1, p2;
+ unsigned short val;
+};
+
+/* channel voice events (8bytes) */
+struct evrec_voice {
+ unsigned char code;
+ unsigned char dev;
+ unsigned char cmd;
+ unsigned char chn;
+ unsigned char note, parm;
+ unsigned short dummy;
+};
+
+/* sysex events (8bytes) */
+struct evrec_sysex {
+ unsigned char code;
+ unsigned char dev;
+ unsigned char buf[6];
+};
+
+/* event record */
+union evrec {
+ struct evrec_short s;
+ struct evrec_note n;
+ struct evrec_long l;
+ struct evrec_voice v;
+ struct evrec_timer t;
+ struct evrec_extended e;
+ struct evrec_sysex x;
+ unsigned int echo;
+ unsigned char c[LONG_EVENT_SIZE];
+};
+
+#define ev_is_long(ev) ((ev)->s.code >= 128)
+#define ev_length(ev) ((ev)->s.code >= 128 ? LONG_EVENT_SIZE : SHORT_EVENT_SIZE)
+
+int snd_seq_oss_process_event(struct seq_oss_devinfo *dp, union evrec *q, struct snd_seq_event *ev);
+int snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *q);
+int snd_seq_oss_event_input(struct snd_seq_event *ev, int direct, void *private_data, int atomic, int hop);
+
+
+#endif /* __SEQ_OSS_EVENT_H */
diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c
new file mode 100644
index 0000000..d0d721c
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_init.c
@@ -0,0 +1,545 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * open/close and reset interface
+ *
+ * Copyright (C) 1998-1999 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_device.h"
+#include "seq_oss_synth.h"
+#include "seq_oss_midi.h"
+#include "seq_oss_writeq.h"
+#include "seq_oss_readq.h"
+#include "seq_oss_timer.h"
+#include "seq_oss_event.h"
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+
+/*
+ * common variables
+ */
+static int maxqlen = SNDRV_SEQ_OSS_MAX_QLEN;
+module_param(maxqlen, int, 0444);
+MODULE_PARM_DESC(maxqlen, "maximum queue length");
+
+static int system_client = -1; /* ALSA sequencer client number */
+static int system_port = -1;
+
+static int num_clients;
+static struct seq_oss_devinfo *client_table[SNDRV_SEQ_OSS_MAX_CLIENTS];
+
+
+/*
+ * prototypes
+ */
+static int receive_announce(struct snd_seq_event *ev, int direct, void *private, int atomic, int hop);
+static int translate_mode(struct file *file);
+static int create_port(struct seq_oss_devinfo *dp);
+static int delete_port(struct seq_oss_devinfo *dp);
+static int alloc_seq_queue(struct seq_oss_devinfo *dp);
+static int delete_seq_queue(int queue);
+static void free_devinfo(void *private);
+
+#define call_ctl(type,rec) snd_seq_kernel_client_ctl(system_client, type, rec)
+
+
+/*
+ * create sequencer client for OSS sequencer
+ */
+int __init
+snd_seq_oss_create_client(void)
+{
+ int rc;
+ struct snd_seq_port_info *port;
+ struct snd_seq_port_callback port_callback;
+
+ port = kmalloc(sizeof(*port), GFP_KERNEL);
+ if (!port) {
+ rc = -ENOMEM;
+ goto __error;
+ }
+
+ /* create ALSA client */
+ rc = snd_seq_create_kernel_client(NULL, SNDRV_SEQ_CLIENT_OSS,
+ "OSS sequencer");
+ if (rc < 0)
+ goto __error;
+
+ system_client = rc;
+ debug_printk(("new client = %d\n", rc));
+
+ /* look up midi devices */
+ snd_seq_oss_midi_lookup_ports(system_client);
+
+ /* create annoucement receiver port */
+ memset(port, 0, sizeof(*port));
+ strcpy(port->name, "Receiver");
+ port->addr.client = system_client;
+ port->capability = SNDRV_SEQ_PORT_CAP_WRITE; /* receive only */
+ port->type = 0;
+
+ memset(&port_callback, 0, sizeof(port_callback));
+ /* don't set port_callback.owner here. otherwise the module counter
+ * is incremented and we can no longer release the module..
+ */
+ port_callback.event_input = receive_announce;
+ port->kernel = &port_callback;
+
+ call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+ if ((system_port = port->addr.port) >= 0) {
+ struct snd_seq_port_subscribe subs;
+
+ memset(&subs, 0, sizeof(subs));
+ subs.sender.client = SNDRV_SEQ_CLIENT_SYSTEM;
+ subs.sender.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE;
+ subs.dest.client = system_client;
+ subs.dest.port = system_port;
+ call_ctl(SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &subs);
+ }
+ rc = 0;
+
+ __error:
+ kfree(port);
+ return rc;
+}
+
+
+/*
+ * receive annoucement from system port, and check the midi device
+ */
+static int
+receive_announce(struct snd_seq_event *ev, int direct, void *private, int atomic, int hop)
+{
+ struct snd_seq_port_info pinfo;
+
+ if (atomic)
+ return 0; /* it must not happen */
+
+ switch (ev->type) {
+ case SNDRV_SEQ_EVENT_PORT_START:
+ case SNDRV_SEQ_EVENT_PORT_CHANGE:
+ if (ev->data.addr.client == system_client)
+ break; /* ignore myself */
+ memset(&pinfo, 0, sizeof(pinfo));
+ pinfo.addr = ev->data.addr;
+ if (call_ctl(SNDRV_SEQ_IOCTL_GET_PORT_INFO, &pinfo) >= 0)
+ snd_seq_oss_midi_check_new_port(&pinfo);
+ break;
+
+ case SNDRV_SEQ_EVENT_PORT_EXIT:
+ if (ev->data.addr.client == system_client)
+ break; /* ignore myself */
+ snd_seq_oss_midi_check_exit_port(ev->data.addr.client,
+ ev->data.addr.port);
+ break;
+ }
+ return 0;
+}
+
+
+/*
+ * delete OSS sequencer client
+ */
+int
+snd_seq_oss_delete_client(void)
+{
+ if (system_client >= 0)
+ snd_seq_delete_kernel_client(system_client);
+
+ snd_seq_oss_midi_clear_all();
+
+ return 0;
+}
+
+
+/*
+ * open sequencer device
+ */
+int
+snd_seq_oss_open(struct file *file, int level)
+{
+ int i, rc;
+ struct seq_oss_devinfo *dp;
+
+ dp = kzalloc(sizeof(*dp), GFP_KERNEL);
+ if (!dp) {
+ snd_printk(KERN_ERR "can't malloc device info\n");
+ return -ENOMEM;
+ }
+ debug_printk(("oss_open: dp = %p\n", dp));
+
+ dp->cseq = system_client;
+ dp->port = -1;
+ dp->queue = -1;
+
+ for (i = 0; i < SNDRV_SEQ_OSS_MAX_CLIENTS; i++) {
+ if (client_table[i] == NULL)
+ break;
+ }
+
+ dp->index = i;
+ if (i >= SNDRV_SEQ_OSS_MAX_CLIENTS) {
+ snd_printk(KERN_ERR "too many applications\n");
+ rc = -ENOMEM;
+ goto _error;
+ }
+
+ /* look up synth and midi devices */
+ snd_seq_oss_synth_setup(dp);
+ snd_seq_oss_midi_setup(dp);
+
+ if (dp->synth_opened == 0 && dp->max_mididev == 0) {
+ /* snd_printk(KERN_ERR "no device found\n"); */
+ rc = -ENODEV;
+ goto _error;
+ }
+
+ /* create port */
+ debug_printk(("create new port\n"));
+ rc = create_port(dp);
+ if (rc < 0) {
+ snd_printk(KERN_ERR "can't create port\n");
+ goto _error;
+ }
+
+ /* allocate queue */
+ debug_printk(("allocate queue\n"));
+ rc = alloc_seq_queue(dp);
+ if (rc < 0)
+ goto _error;
+
+ /* set address */
+ dp->addr.client = dp->cseq;
+ dp->addr.port = dp->port;
+ /*dp->addr.queue = dp->queue;*/
+ /*dp->addr.channel = 0;*/
+
+ dp->seq_mode = level;
+
+ /* set up file mode */
+ dp->file_mode = translate_mode(file);
+
+ /* initialize read queue */
+ debug_printk(("initialize read queue\n"));
+ if (is_read_mode(dp->file_mode)) {
+ dp->readq = snd_seq_oss_readq_new(dp, maxqlen);
+ if (!dp->readq) {
+ rc = -ENOMEM;
+ goto _error;
+ }
+ }
+
+ /* initialize write queue */
+ debug_printk(("initialize write queue\n"));
+ if (is_write_mode(dp->file_mode)) {
+ dp->writeq = snd_seq_oss_writeq_new(dp, maxqlen);
+ if (!dp->writeq) {
+ rc = -ENOMEM;
+ goto _error;
+ }
+ }
+
+ /* initialize timer */
+ debug_printk(("initialize timer\n"));
+ dp->timer = snd_seq_oss_timer_new(dp);
+ if (!dp->timer) {
+ snd_printk(KERN_ERR "can't alloc timer\n");
+ rc = -ENOMEM;
+ goto _error;
+ }
+ debug_printk(("timer initialized\n"));
+
+ /* set private data pointer */
+ file->private_data = dp;
+
+ /* set up for mode2 */
+ if (level == SNDRV_SEQ_OSS_MODE_MUSIC)
+ snd_seq_oss_synth_setup_midi(dp);
+ else if (is_read_mode(dp->file_mode))
+ snd_seq_oss_midi_open_all(dp, SNDRV_SEQ_OSS_FILE_READ);
+
+ client_table[dp->index] = dp;
+ num_clients++;
+
+ debug_printk(("open done\n"));
+ return 0;
+
+ _error:
+ snd_seq_oss_writeq_delete(dp->writeq);
+ snd_seq_oss_readq_delete(dp->readq);
+ snd_seq_oss_synth_cleanup(dp);
+ snd_seq_oss_midi_cleanup(dp);
+ delete_port(dp);
+ delete_seq_queue(dp->queue);
+ kfree(dp);
+
+ return rc;
+}
+
+/*
+ * translate file flags to private mode
+ */
+static int
+translate_mode(struct file *file)
+{
+ int file_mode = 0;
+ if ((file->f_flags & O_ACCMODE) != O_RDONLY)
+ file_mode |= SNDRV_SEQ_OSS_FILE_WRITE;
+ if ((file->f_flags & O_ACCMODE) != O_WRONLY)
+ file_mode |= SNDRV_SEQ_OSS_FILE_READ;
+ if (file->f_flags & O_NONBLOCK)
+ file_mode |= SNDRV_SEQ_OSS_FILE_NONBLOCK;
+ return file_mode;
+}
+
+
+/*
+ * create sequencer port
+ */
+static int
+create_port(struct seq_oss_devinfo *dp)
+{
+ int rc;
+ struct snd_seq_port_info port;
+ struct snd_seq_port_callback callback;
+
+ memset(&port, 0, sizeof(port));
+ port.addr.client = dp->cseq;
+ sprintf(port.name, "Sequencer-%d", dp->index);
+ port.capability = SNDRV_SEQ_PORT_CAP_READ|SNDRV_SEQ_PORT_CAP_WRITE; /* no subscription */
+ port.type = SNDRV_SEQ_PORT_TYPE_SPECIFIC;
+ port.midi_channels = 128;
+ port.synth_voices = 128;
+
+ memset(&callback, 0, sizeof(callback));
+ callback.owner = THIS_MODULE;
+ callback.private_data = dp;
+ callback.event_input = snd_seq_oss_event_input;
+ callback.private_free = free_devinfo;
+ port.kernel = &callback;
+
+ rc = call_ctl(SNDRV_SEQ_IOCTL_CREATE_PORT, &port);
+ if (rc < 0)
+ return rc;
+
+ dp->port = port.addr.port;
+ debug_printk(("new port = %d\n", port.addr.port));
+
+ return 0;
+}
+
+/*
+ * delete ALSA port
+ */
+static int
+delete_port(struct seq_oss_devinfo *dp)
+{
+ if (dp->port < 0)
+ return 0;
+
+ debug_printk(("delete_port %i\n", dp->port));
+ return snd_seq_event_port_detach(dp->cseq, dp->port);
+}
+
+/*
+ * allocate a queue
+ */
+static int
+alloc_seq_queue(struct seq_oss_devinfo *dp)
+{
+ struct snd_seq_queue_info qinfo;
+ int rc;
+
+ memset(&qinfo, 0, sizeof(qinfo));
+ qinfo.owner = system_client;
+ qinfo.locked = 1;
+ strcpy(qinfo.name, "OSS Sequencer Emulation");
+ if ((rc = call_ctl(SNDRV_SEQ_IOCTL_CREATE_QUEUE, &qinfo)) < 0)
+ return rc;
+ dp->queue = qinfo.queue;
+ return 0;
+}
+
+/*
+ * release queue
+ */
+static int
+delete_seq_queue(int queue)
+{
+ struct snd_seq_queue_info qinfo;
+ int rc;
+
+ if (queue < 0)
+ return 0;
+ memset(&qinfo, 0, sizeof(qinfo));
+ qinfo.queue = queue;
+ rc = call_ctl(SNDRV_SEQ_IOCTL_DELETE_QUEUE, &qinfo);
+ if (rc < 0)
+ printk(KERN_ERR "seq-oss: unable to delete queue %d (%d)\n", queue, rc);
+ return rc;
+}
+
+
+/*
+ * free device informations - private_free callback of port
+ */
+static void
+free_devinfo(void *private)
+{
+ struct seq_oss_devinfo *dp = (struct seq_oss_devinfo *)private;
+
+ if (dp->timer)
+ snd_seq_oss_timer_delete(dp->timer);
+
+ if (dp->writeq)
+ snd_seq_oss_writeq_delete(dp->writeq);
+
+ if (dp->readq)
+ snd_seq_oss_readq_delete(dp->readq);
+
+ kfree(dp);
+}
+
+
+/*
+ * close sequencer device
+ */
+void
+snd_seq_oss_release(struct seq_oss_devinfo *dp)
+{
+ int queue;
+
+ client_table[dp->index] = NULL;
+ num_clients--;
+
+ debug_printk(("resetting..\n"));
+ snd_seq_oss_reset(dp);
+
+ debug_printk(("cleaning up..\n"));
+ snd_seq_oss_synth_cleanup(dp);
+ snd_seq_oss_midi_cleanup(dp);
+
+ /* clear slot */
+ debug_printk(("releasing resource..\n"));
+ queue = dp->queue;
+ if (dp->port >= 0)
+ delete_port(dp);
+ delete_seq_queue(queue);
+
+ debug_printk(("release done\n"));
+}
+
+
+/*
+ * Wait until the queue is empty (if we don't have nonblock)
+ */
+void
+snd_seq_oss_drain_write(struct seq_oss_devinfo *dp)
+{
+ if (! dp->timer->running)
+ return;
+ if (is_write_mode(dp->file_mode) && !is_nonblock_mode(dp->file_mode) &&
+ dp->writeq) {
+ debug_printk(("syncing..\n"));
+ while (snd_seq_oss_writeq_sync(dp->writeq))
+ ;
+ }
+}
+
+
+/*
+ * reset sequencer devices
+ */
+void
+snd_seq_oss_reset(struct seq_oss_devinfo *dp)
+{
+ int i;
+
+ /* reset all synth devices */
+ for (i = 0; i < dp->max_synthdev; i++)
+ snd_seq_oss_synth_reset(dp, i);
+
+ /* reset all midi devices */
+ if (dp->seq_mode != SNDRV_SEQ_OSS_MODE_MUSIC) {
+ for (i = 0; i < dp->max_mididev; i++)
+ snd_seq_oss_midi_reset(dp, i);
+ }
+
+ /* remove queues */
+ if (dp->readq)
+ snd_seq_oss_readq_clear(dp->readq);
+ if (dp->writeq)
+ snd_seq_oss_writeq_clear(dp->writeq);
+
+ /* reset timer */
+ snd_seq_oss_timer_stop(dp->timer);
+}
+
+
+#ifdef CONFIG_PROC_FS
+/*
+ * misc. functions for proc interface
+ */
+char *
+enabled_str(int bool)
+{
+ return bool ? "enabled" : "disabled";
+}
+
+static char *
+filemode_str(int val)
+{
+ static char *str[] = {
+ "none", "read", "write", "read/write",
+ };
+ return str[val & SNDRV_SEQ_OSS_FILE_ACMODE];
+}
+
+
+/*
+ * proc interface
+ */
+void
+snd_seq_oss_system_info_read(struct snd_info_buffer *buf)
+{
+ int i;
+ struct seq_oss_devinfo *dp;
+
+ snd_iprintf(buf, "ALSA client number %d\n", system_client);
+ snd_iprintf(buf, "ALSA receiver port %d\n", system_port);
+
+ snd_iprintf(buf, "\nNumber of applications: %d\n", num_clients);
+ for (i = 0; i < num_clients; i++) {
+ snd_iprintf(buf, "\nApplication %d: ", i);
+ if ((dp = client_table[i]) == NULL) {
+ snd_iprintf(buf, "*empty*\n");
+ continue;
+ }
+ snd_iprintf(buf, "port %d : queue %d\n", dp->port, dp->queue);
+ snd_iprintf(buf, " sequencer mode = %s : file open mode = %s\n",
+ (dp->seq_mode ? "music" : "synth"),
+ filemode_str(dp->file_mode));
+ if (dp->seq_mode)
+ snd_iprintf(buf, " timer tempo = %d, timebase = %d\n",
+ dp->timer->oss_tempo, dp->timer->oss_timebase);
+ snd_iprintf(buf, " max queue length %d\n", maxqlen);
+ if (is_read_mode(dp->file_mode) && dp->readq)
+ snd_seq_oss_readq_info_read(dp->readq, buf);
+ }
+}
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/core/seq/oss/seq_oss_ioctl.c b/sound/core/seq/oss/seq_oss_ioctl.c
new file mode 100644
index 0000000..5ac701c
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_ioctl.c
@@ -0,0 +1,209 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * OSS compatible i/o control
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_device.h"
+#include "seq_oss_readq.h"
+#include "seq_oss_writeq.h"
+#include "seq_oss_timer.h"
+#include "seq_oss_synth.h"
+#include "seq_oss_midi.h"
+#include "seq_oss_event.h"
+
+static int snd_seq_oss_synth_info_user(struct seq_oss_devinfo *dp, void __user *arg)
+{
+ struct synth_info info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+ if (snd_seq_oss_synth_make_info(dp, info.device, &info) < 0)
+ return -EINVAL;
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_seq_oss_midi_info_user(struct seq_oss_devinfo *dp, void __user *arg)
+{
+ struct midi_info info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+ if (snd_seq_oss_midi_make_info(dp, info.device, &info) < 0)
+ return -EINVAL;
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_seq_oss_oob_user(struct seq_oss_devinfo *dp, void __user *arg)
+{
+ unsigned char ev[8];
+ struct snd_seq_event tmpev;
+
+ if (copy_from_user(ev, arg, 8))
+ return -EFAULT;
+ memset(&tmpev, 0, sizeof(tmpev));
+ snd_seq_oss_fill_addr(dp, &tmpev, dp->addr.port, dp->addr.client);
+ tmpev.time.tick = 0;
+ if (! snd_seq_oss_process_event(dp, (union evrec *)ev, &tmpev)) {
+ snd_seq_oss_dispatch(dp, &tmpev, 0, 0);
+ }
+ return 0;
+}
+
+int
+snd_seq_oss_ioctl(struct seq_oss_devinfo *dp, unsigned int cmd, unsigned long carg)
+{
+ int dev, val;
+ void __user *arg = (void __user *)carg;
+ int __user *p = arg;
+
+ switch (cmd) {
+ case SNDCTL_TMR_TIMEBASE:
+ case SNDCTL_TMR_TEMPO:
+ case SNDCTL_TMR_START:
+ case SNDCTL_TMR_STOP:
+ case SNDCTL_TMR_CONTINUE:
+ case SNDCTL_TMR_METRONOME:
+ case SNDCTL_TMR_SOURCE:
+ case SNDCTL_TMR_SELECT:
+ case SNDCTL_SEQ_CTRLRATE:
+ return snd_seq_oss_timer_ioctl(dp->timer, cmd, arg);
+
+ case SNDCTL_SEQ_PANIC:
+ debug_printk(("panic\n"));
+ snd_seq_oss_reset(dp);
+ return -EINVAL;
+
+ case SNDCTL_SEQ_SYNC:
+ debug_printk(("sync\n"));
+ if (! is_write_mode(dp->file_mode) || dp->writeq == NULL)
+ return 0;
+ while (snd_seq_oss_writeq_sync(dp->writeq))
+ ;
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ return 0;
+
+ case SNDCTL_SEQ_RESET:
+ debug_printk(("reset\n"));
+ snd_seq_oss_reset(dp);
+ return 0;
+
+ case SNDCTL_SEQ_TESTMIDI:
+ debug_printk(("test midi\n"));
+ if (get_user(dev, p))
+ return -EFAULT;
+ return snd_seq_oss_midi_open(dp, dev, dp->file_mode);
+
+ case SNDCTL_SEQ_GETINCOUNT:
+ debug_printk(("get in count\n"));
+ if (dp->readq == NULL || ! is_read_mode(dp->file_mode))
+ return 0;
+ return put_user(dp->readq->qlen, p) ? -EFAULT : 0;
+
+ case SNDCTL_SEQ_GETOUTCOUNT:
+ debug_printk(("get out count\n"));
+ if (! is_write_mode(dp->file_mode) || dp->writeq == NULL)
+ return 0;
+ return put_user(snd_seq_oss_writeq_get_free_size(dp->writeq), p) ? -EFAULT : 0;
+
+ case SNDCTL_SEQ_GETTIME:
+ debug_printk(("get time\n"));
+ return put_user(snd_seq_oss_timer_cur_tick(dp->timer), p) ? -EFAULT : 0;
+
+ case SNDCTL_SEQ_RESETSAMPLES:
+ debug_printk(("reset samples\n"));
+ if (get_user(dev, p))
+ return -EFAULT;
+ return snd_seq_oss_synth_ioctl(dp, dev, cmd, carg);
+
+ case SNDCTL_SEQ_NRSYNTHS:
+ debug_printk(("nr synths\n"));
+ return put_user(dp->max_synthdev, p) ? -EFAULT : 0;
+
+ case SNDCTL_SEQ_NRMIDIS:
+ debug_printk(("nr midis\n"));
+ return put_user(dp->max_mididev, p) ? -EFAULT : 0;
+
+ case SNDCTL_SYNTH_MEMAVL:
+ debug_printk(("mem avail\n"));
+ if (get_user(dev, p))
+ return -EFAULT;
+ val = snd_seq_oss_synth_ioctl(dp, dev, cmd, carg);
+ return put_user(val, p) ? -EFAULT : 0;
+
+ case SNDCTL_FM_4OP_ENABLE:
+ debug_printk(("4op\n"));
+ if (get_user(dev, p))
+ return -EFAULT;
+ snd_seq_oss_synth_ioctl(dp, dev, cmd, carg);
+ return 0;
+
+ case SNDCTL_SYNTH_INFO:
+ case SNDCTL_SYNTH_ID:
+ debug_printk(("synth info\n"));
+ return snd_seq_oss_synth_info_user(dp, arg);
+
+ case SNDCTL_SEQ_OUTOFBAND:
+ debug_printk(("out of band\n"));
+ return snd_seq_oss_oob_user(dp, arg);
+
+ case SNDCTL_MIDI_INFO:
+ debug_printk(("midi info\n"));
+ return snd_seq_oss_midi_info_user(dp, arg);
+
+ case SNDCTL_SEQ_THRESHOLD:
+ debug_printk(("threshold\n"));
+ if (! is_write_mode(dp->file_mode))
+ return 0;
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val < 1)
+ val = 1;
+ if (val >= dp->writeq->maxlen)
+ val = dp->writeq->maxlen - 1;
+ snd_seq_oss_writeq_set_output(dp->writeq, val);
+ return 0;
+
+ case SNDCTL_MIDI_PRETIME:
+ debug_printk(("pretime\n"));
+ if (dp->readq == NULL || !is_read_mode(dp->file_mode))
+ return 0;
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val <= 0)
+ val = -1;
+ else
+ val = (HZ * val) / 10;
+ dp->readq->pre_event_timeout = val;
+ return put_user(val, p) ? -EFAULT : 0;
+
+ default:
+ debug_printk(("others\n"));
+ if (! is_write_mode(dp->file_mode))
+ return -EIO;
+ return snd_seq_oss_synth_ioctl(dp, 0, cmd, carg);
+ }
+ return 0;
+}
+
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
new file mode 100644
index 0000000..0a711d2
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_midi.c
@@ -0,0 +1,711 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * MIDI device handlers
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_midi.h"
+#include "seq_oss_readq.h"
+#include "seq_oss_timer.h"
+#include "seq_oss_event.h"
+#include <sound/seq_midi_event.h>
+#include "../seq_lock.h"
+#include <linux/init.h>
+
+
+/*
+ * constants
+ */
+#define SNDRV_SEQ_OSS_MAX_MIDI_NAME 30
+
+/*
+ * definition of midi device record
+ */
+struct seq_oss_midi {
+ int seq_device; /* device number */
+ int client; /* sequencer client number */
+ int port; /* sequencer port number */
+ unsigned int flags; /* port capability */
+ int opened; /* flag for opening */
+ unsigned char name[SNDRV_SEQ_OSS_MAX_MIDI_NAME];
+ struct snd_midi_event *coder; /* MIDI event coder */
+ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */
+ snd_use_lock_t use_lock;
+};
+
+
+/*
+ * midi device table
+ */
+static int max_midi_devs;
+static struct seq_oss_midi *midi_devs[SNDRV_SEQ_OSS_MAX_MIDI_DEVS];
+
+static DEFINE_SPINLOCK(register_lock);
+
+/*
+ * prototypes
+ */
+static struct seq_oss_midi *get_mdev(int dev);
+static struct seq_oss_midi *get_mididev(struct seq_oss_devinfo *dp, int dev);
+static int send_synth_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, int dev);
+static int send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, struct seq_oss_midi *mdev);
+
+/*
+ * look up the existing ports
+ * this looks a very exhausting job.
+ */
+int __init
+snd_seq_oss_midi_lookup_ports(int client)
+{
+ struct snd_seq_client_info *clinfo;
+ struct snd_seq_port_info *pinfo;
+
+ clinfo = kzalloc(sizeof(*clinfo), GFP_KERNEL);
+ pinfo = kzalloc(sizeof(*pinfo), GFP_KERNEL);
+ if (! clinfo || ! pinfo) {
+ kfree(clinfo);
+ kfree(pinfo);
+ return -ENOMEM;
+ }
+ clinfo->client = -1;
+ while (snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT, clinfo) == 0) {
+ if (clinfo->client == client)
+ continue; /* ignore myself */
+ pinfo->addr.client = clinfo->client;
+ pinfo->addr.port = -1;
+ while (snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_QUERY_NEXT_PORT, pinfo) == 0)
+ snd_seq_oss_midi_check_new_port(pinfo);
+ }
+ kfree(clinfo);
+ kfree(pinfo);
+ return 0;
+}
+
+
+/*
+ */
+static struct seq_oss_midi *
+get_mdev(int dev)
+{
+ struct seq_oss_midi *mdev;
+ unsigned long flags;
+
+ spin_lock_irqsave(&register_lock, flags);
+ mdev = midi_devs[dev];
+ if (mdev)
+ snd_use_lock_use(&mdev->use_lock);
+ spin_unlock_irqrestore(&register_lock, flags);
+ return mdev;
+}
+
+/*
+ * look for the identical slot
+ */
+static struct seq_oss_midi *
+find_slot(int client, int port)
+{
+ int i;
+ struct seq_oss_midi *mdev;
+ unsigned long flags;
+
+ spin_lock_irqsave(&register_lock, flags);
+ for (i = 0; i < max_midi_devs; i++) {
+ mdev = midi_devs[i];
+ if (mdev && mdev->client == client && mdev->port == port) {
+ /* found! */
+ snd_use_lock_use(&mdev->use_lock);
+ spin_unlock_irqrestore(&register_lock, flags);
+ return mdev;
+ }
+ }
+ spin_unlock_irqrestore(&register_lock, flags);
+ return NULL;
+}
+
+
+#define PERM_WRITE (SNDRV_SEQ_PORT_CAP_WRITE|SNDRV_SEQ_PORT_CAP_SUBS_WRITE)
+#define PERM_READ (SNDRV_SEQ_PORT_CAP_READ|SNDRV_SEQ_PORT_CAP_SUBS_READ)
+/*
+ * register a new port if it doesn't exist yet
+ */
+int
+snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo)
+{
+ int i;
+ struct seq_oss_midi *mdev;
+ unsigned long flags;
+
+ debug_printk(("check for MIDI client %d port %d\n", pinfo->addr.client, pinfo->addr.port));
+ /* the port must include generic midi */
+ if (! (pinfo->type & SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC))
+ return 0;
+ /* either read or write subscribable */
+ if ((pinfo->capability & PERM_WRITE) != PERM_WRITE &&
+ (pinfo->capability & PERM_READ) != PERM_READ)
+ return 0;
+
+ /*
+ * look for the identical slot
+ */
+ if ((mdev = find_slot(pinfo->addr.client, pinfo->addr.port)) != NULL) {
+ /* already exists */
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+ }
+
+ /*
+ * allocate midi info record
+ */
+ if ((mdev = kzalloc(sizeof(*mdev), GFP_KERNEL)) == NULL) {
+ snd_printk(KERN_ERR "can't malloc midi info\n");
+ return -ENOMEM;
+ }
+
+ /* copy the port information */
+ mdev->client = pinfo->addr.client;
+ mdev->port = pinfo->addr.port;
+ mdev->flags = pinfo->capability;
+ mdev->opened = 0;
+ snd_use_lock_init(&mdev->use_lock);
+
+ /* copy and truncate the name of synth device */
+ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name));
+
+ /* create MIDI coder */
+ if (snd_midi_event_new(MAX_MIDI_EVENT_BUF, &mdev->coder) < 0) {
+ snd_printk(KERN_ERR "can't malloc midi coder\n");
+ kfree(mdev);
+ return -ENOMEM;
+ }
+ /* OSS sequencer adds running status to all sequences */
+ snd_midi_event_no_status(mdev->coder, 1);
+
+ /*
+ * look for en empty slot
+ */
+ spin_lock_irqsave(&register_lock, flags);
+ for (i = 0; i < max_midi_devs; i++) {
+ if (midi_devs[i] == NULL)
+ break;
+ }
+ if (i >= max_midi_devs) {
+ if (max_midi_devs >= SNDRV_SEQ_OSS_MAX_MIDI_DEVS) {
+ spin_unlock_irqrestore(&register_lock, flags);
+ snd_midi_event_free(mdev->coder);
+ kfree(mdev);
+ return -ENOMEM;
+ }
+ max_midi_devs++;
+ }
+ mdev->seq_device = i;
+ midi_devs[mdev->seq_device] = mdev;
+ spin_unlock_irqrestore(&register_lock, flags);
+
+ return 0;
+}
+
+/*
+ * release the midi device if it was registered
+ */
+int
+snd_seq_oss_midi_check_exit_port(int client, int port)
+{
+ struct seq_oss_midi *mdev;
+ unsigned long flags;
+ int index;
+
+ if ((mdev = find_slot(client, port)) != NULL) {
+ spin_lock_irqsave(&register_lock, flags);
+ midi_devs[mdev->seq_device] = NULL;
+ spin_unlock_irqrestore(&register_lock, flags);
+ snd_use_lock_free(&mdev->use_lock);
+ snd_use_lock_sync(&mdev->use_lock);
+ if (mdev->coder)
+ snd_midi_event_free(mdev->coder);
+ kfree(mdev);
+ }
+ spin_lock_irqsave(&register_lock, flags);
+ for (index = max_midi_devs - 1; index >= 0; index--) {
+ if (midi_devs[index])
+ break;
+ }
+ max_midi_devs = index + 1;
+ spin_unlock_irqrestore(&register_lock, flags);
+ return 0;
+}
+
+
+/*
+ * release the midi device if it was registered
+ */
+void
+snd_seq_oss_midi_clear_all(void)
+{
+ int i;
+ struct seq_oss_midi *mdev;
+ unsigned long flags;
+
+ spin_lock_irqsave(&register_lock, flags);
+ for (i = 0; i < max_midi_devs; i++) {
+ if ((mdev = midi_devs[i]) != NULL) {
+ if (mdev->coder)
+ snd_midi_event_free(mdev->coder);
+ kfree(mdev);
+ midi_devs[i] = NULL;
+ }
+ }
+ max_midi_devs = 0;
+ spin_unlock_irqrestore(&register_lock, flags);
+}
+
+
+/*
+ * set up midi tables
+ */
+void
+snd_seq_oss_midi_setup(struct seq_oss_devinfo *dp)
+{
+ dp->max_mididev = max_midi_devs;
+}
+
+/*
+ * clean up midi tables
+ */
+void
+snd_seq_oss_midi_cleanup(struct seq_oss_devinfo *dp)
+{
+ int i;
+ for (i = 0; i < dp->max_mididev; i++)
+ snd_seq_oss_midi_close(dp, i);
+ dp->max_mididev = 0;
+}
+
+
+/*
+ * open all midi devices. ignore errors.
+ */
+void
+snd_seq_oss_midi_open_all(struct seq_oss_devinfo *dp, int file_mode)
+{
+ int i;
+ for (i = 0; i < dp->max_mididev; i++)
+ snd_seq_oss_midi_open(dp, i, file_mode);
+}
+
+
+/*
+ * get the midi device information
+ */
+static struct seq_oss_midi *
+get_mididev(struct seq_oss_devinfo *dp, int dev)
+{
+ if (dev < 0 || dev >= dp->max_mididev)
+ return NULL;
+ return get_mdev(dev);
+}
+
+
+/*
+ * open the midi device if not opened yet
+ */
+int
+snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode)
+{
+ int perm;
+ struct seq_oss_midi *mdev;
+ struct snd_seq_port_subscribe subs;
+
+ if ((mdev = get_mididev(dp, dev)) == NULL)
+ return -ENODEV;
+
+ /* already used? */
+ if (mdev->opened && mdev->devinfo != dp) {
+ snd_use_lock_free(&mdev->use_lock);
+ return -EBUSY;
+ }
+
+ perm = 0;
+ if (is_write_mode(fmode))
+ perm |= PERM_WRITE;
+ if (is_read_mode(fmode))
+ perm |= PERM_READ;
+ perm &= mdev->flags;
+ if (perm == 0) {
+ snd_use_lock_free(&mdev->use_lock);
+ return -ENXIO;
+ }
+
+ /* already opened? */
+ if ((mdev->opened & perm) == perm) {
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+ }
+
+ perm &= ~mdev->opened;
+
+ memset(&subs, 0, sizeof(subs));
+
+ if (perm & PERM_WRITE) {
+ subs.sender = dp->addr;
+ subs.dest.client = mdev->client;
+ subs.dest.port = mdev->port;
+ if (snd_seq_kernel_client_ctl(dp->cseq, SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &subs) >= 0)
+ mdev->opened |= PERM_WRITE;
+ }
+ if (perm & PERM_READ) {
+ subs.sender.client = mdev->client;
+ subs.sender.port = mdev->port;
+ subs.dest = dp->addr;
+ subs.flags = SNDRV_SEQ_PORT_SUBS_TIMESTAMP;
+ subs.queue = dp->queue; /* queue for timestamps */
+ if (snd_seq_kernel_client_ctl(dp->cseq, SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, &subs) >= 0)
+ mdev->opened |= PERM_READ;
+ }
+
+ if (! mdev->opened) {
+ snd_use_lock_free(&mdev->use_lock);
+ return -ENXIO;
+ }
+
+ mdev->devinfo = dp;
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+}
+
+/*
+ * close the midi device if already opened
+ */
+int
+snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev)
+{
+ struct seq_oss_midi *mdev;
+ struct snd_seq_port_subscribe subs;
+
+ if ((mdev = get_mididev(dp, dev)) == NULL)
+ return -ENODEV;
+ if (! mdev->opened || mdev->devinfo != dp) {
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+ }
+
+ debug_printk(("closing client %d port %d mode %d\n", mdev->client, mdev->port, mdev->opened));
+ memset(&subs, 0, sizeof(subs));
+ if (mdev->opened & PERM_WRITE) {
+ subs.sender = dp->addr;
+ subs.dest.client = mdev->client;
+ subs.dest.port = mdev->port;
+ snd_seq_kernel_client_ctl(dp->cseq, SNDRV_SEQ_IOCTL_UNSUBSCRIBE_PORT, &subs);
+ }
+ if (mdev->opened & PERM_READ) {
+ subs.sender.client = mdev->client;
+ subs.sender.port = mdev->port;
+ subs.dest = dp->addr;
+ snd_seq_kernel_client_ctl(dp->cseq, SNDRV_SEQ_IOCTL_UNSUBSCRIBE_PORT, &subs);
+ }
+
+ mdev->opened = 0;
+ mdev->devinfo = NULL;
+
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+}
+
+/*
+ * change seq capability flags to file mode flags
+ */
+int
+snd_seq_oss_midi_filemode(struct seq_oss_devinfo *dp, int dev)
+{
+ struct seq_oss_midi *mdev;
+ int mode;
+
+ if ((mdev = get_mididev(dp, dev)) == NULL)
+ return 0;
+
+ mode = 0;
+ if (mdev->opened & PERM_WRITE)
+ mode |= SNDRV_SEQ_OSS_FILE_WRITE;
+ if (mdev->opened & PERM_READ)
+ mode |= SNDRV_SEQ_OSS_FILE_READ;
+
+ snd_use_lock_free(&mdev->use_lock);
+ return mode;
+}
+
+/*
+ * reset the midi device and close it:
+ * so far, only close the device.
+ */
+void
+snd_seq_oss_midi_reset(struct seq_oss_devinfo *dp, int dev)
+{
+ struct seq_oss_midi *mdev;
+
+ if ((mdev = get_mididev(dp, dev)) == NULL)
+ return;
+ if (! mdev->opened) {
+ snd_use_lock_free(&mdev->use_lock);
+ return;
+ }
+
+ if (mdev->opened & PERM_WRITE) {
+ struct snd_seq_event ev;
+ int c;
+
+ debug_printk(("resetting client %d port %d\n", mdev->client, mdev->port));
+ memset(&ev, 0, sizeof(ev));
+ ev.dest.client = mdev->client;
+ ev.dest.port = mdev->port;
+ ev.queue = dp->queue;
+ ev.source.port = dp->port;
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_SYNTH) {
+ ev.type = SNDRV_SEQ_EVENT_SENSING;
+ snd_seq_oss_dispatch(dp, &ev, 0, 0); /* active sensing */
+ }
+ for (c = 0; c < 16; c++) {
+ ev.type = SNDRV_SEQ_EVENT_CONTROLLER;
+ ev.data.control.channel = c;
+ ev.data.control.param = 123;
+ snd_seq_oss_dispatch(dp, &ev, 0, 0); /* all notes off */
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC) {
+ ev.data.control.param = 121;
+ snd_seq_oss_dispatch(dp, &ev, 0, 0); /* reset all controllers */
+ ev.type = SNDRV_SEQ_EVENT_PITCHBEND;
+ ev.data.control.value = 0;
+ snd_seq_oss_dispatch(dp, &ev, 0, 0); /* bender off */
+ }
+ }
+ }
+ // snd_seq_oss_midi_close(dp, dev);
+ snd_use_lock_free(&mdev->use_lock);
+}
+
+
+/*
+ * get client/port of the specified MIDI device
+ */
+void
+snd_seq_oss_midi_get_addr(struct seq_oss_devinfo *dp, int dev, struct snd_seq_addr *addr)
+{
+ struct seq_oss_midi *mdev;
+
+ if ((mdev = get_mididev(dp, dev)) == NULL)
+ return;
+ addr->client = mdev->client;
+ addr->port = mdev->port;
+ snd_use_lock_free(&mdev->use_lock);
+}
+
+
+/*
+ * input callback - this can be atomic
+ */
+int
+snd_seq_oss_midi_input(struct snd_seq_event *ev, int direct, void *private_data)
+{
+ struct seq_oss_devinfo *dp = (struct seq_oss_devinfo *)private_data;
+ struct seq_oss_midi *mdev;
+ int rc;
+
+ if (dp->readq == NULL)
+ return 0;
+ if ((mdev = find_slot(ev->source.client, ev->source.port)) == NULL)
+ return 0;
+ if (! (mdev->opened & PERM_READ)) {
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+ }
+
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC)
+ rc = send_synth_event(dp, ev, mdev->seq_device);
+ else
+ rc = send_midi_event(dp, ev, mdev);
+
+ snd_use_lock_free(&mdev->use_lock);
+ return rc;
+}
+
+/*
+ * convert ALSA sequencer event to OSS synth event
+ */
+static int
+send_synth_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, int dev)
+{
+ union evrec ossev;
+
+ memset(&ossev, 0, sizeof(ossev));
+
+ switch (ev->type) {
+ case SNDRV_SEQ_EVENT_NOTEON:
+ ossev.v.cmd = MIDI_NOTEON; break;
+ case SNDRV_SEQ_EVENT_NOTEOFF:
+ ossev.v.cmd = MIDI_NOTEOFF; break;
+ case SNDRV_SEQ_EVENT_KEYPRESS:
+ ossev.v.cmd = MIDI_KEY_PRESSURE; break;
+ case SNDRV_SEQ_EVENT_CONTROLLER:
+ ossev.l.cmd = MIDI_CTL_CHANGE; break;
+ case SNDRV_SEQ_EVENT_PGMCHANGE:
+ ossev.l.cmd = MIDI_PGM_CHANGE; break;
+ case SNDRV_SEQ_EVENT_CHANPRESS:
+ ossev.l.cmd = MIDI_CHN_PRESSURE; break;
+ case SNDRV_SEQ_EVENT_PITCHBEND:
+ ossev.l.cmd = MIDI_PITCH_BEND; break;
+ default:
+ return 0; /* not supported */
+ }
+
+ ossev.v.dev = dev;
+
+ switch (ev->type) {
+ case SNDRV_SEQ_EVENT_NOTEON:
+ case SNDRV_SEQ_EVENT_NOTEOFF:
+ case SNDRV_SEQ_EVENT_KEYPRESS:
+ ossev.v.code = EV_CHN_VOICE;
+ ossev.v.note = ev->data.note.note;
+ ossev.v.parm = ev->data.note.velocity;
+ ossev.v.chn = ev->data.note.channel;
+ break;
+ case SNDRV_SEQ_EVENT_CONTROLLER:
+ case SNDRV_SEQ_EVENT_PGMCHANGE:
+ case SNDRV_SEQ_EVENT_CHANPRESS:
+ ossev.l.code = EV_CHN_COMMON;
+ ossev.l.p1 = ev->data.control.param;
+ ossev.l.val = ev->data.control.value;
+ ossev.l.chn = ev->data.control.channel;
+ break;
+ case SNDRV_SEQ_EVENT_PITCHBEND:
+ ossev.l.code = EV_CHN_COMMON;
+ ossev.l.val = ev->data.control.value + 8192;
+ ossev.l.chn = ev->data.control.channel;
+ break;
+ }
+
+ snd_seq_oss_readq_put_timestamp(dp->readq, ev->time.tick, dp->seq_mode);
+ snd_seq_oss_readq_put_event(dp->readq, &ossev);
+
+ return 0;
+}
+
+/*
+ * decode event and send MIDI bytes to read queue
+ */
+static int
+send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, struct seq_oss_midi *mdev)
+{
+ char msg[32];
+ int len;
+
+ snd_seq_oss_readq_put_timestamp(dp->readq, ev->time.tick, dp->seq_mode);
+ if (!dp->timer->running)
+ len = snd_seq_oss_timer_start(dp->timer);
+ if (ev->type == SNDRV_SEQ_EVENT_SYSEX) {
+ if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) == SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
+ snd_seq_oss_readq_puts(dp->readq, mdev->seq_device,
+ ev->data.ext.ptr, ev->data.ext.len);
+ } else {
+ len = snd_midi_event_decode(mdev->coder, msg, sizeof(msg), ev);
+ if (len > 0)
+ snd_seq_oss_readq_puts(dp->readq, mdev->seq_device, msg, len);
+ }
+
+ return 0;
+}
+
+
+/*
+ * dump midi data
+ * return 0 : enqueued
+ * non-zero : invalid - ignored
+ */
+int
+snd_seq_oss_midi_putc(struct seq_oss_devinfo *dp, int dev, unsigned char c, struct snd_seq_event *ev)
+{
+ struct seq_oss_midi *mdev;
+
+ if ((mdev = get_mididev(dp, dev)) == NULL)
+ return -ENODEV;
+ if (snd_midi_event_encode_byte(mdev->coder, c, ev) > 0) {
+ snd_seq_oss_fill_addr(dp, ev, mdev->client, mdev->port);
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+ }
+ snd_use_lock_free(&mdev->use_lock);
+ return -EINVAL;
+}
+
+/*
+ * create OSS compatible midi_info record
+ */
+int
+snd_seq_oss_midi_make_info(struct seq_oss_devinfo *dp, int dev, struct midi_info *inf)
+{
+ struct seq_oss_midi *mdev;
+
+ if ((mdev = get_mididev(dp, dev)) == NULL)
+ return -ENXIO;
+ inf->device = dev;
+ inf->dev_type = 0; /* FIXME: ?? */
+ inf->capabilities = 0; /* FIXME: ?? */
+ strlcpy(inf->name, mdev->name, sizeof(inf->name));
+ snd_use_lock_free(&mdev->use_lock);
+ return 0;
+}
+
+
+#ifdef CONFIG_PROC_FS
+/*
+ * proc interface
+ */
+static char *
+capmode_str(int val)
+{
+ val &= PERM_READ|PERM_WRITE;
+ if (val == (PERM_READ|PERM_WRITE))
+ return "read/write";
+ else if (val == PERM_READ)
+ return "read";
+ else if (val == PERM_WRITE)
+ return "write";
+ else
+ return "none";
+}
+
+void
+snd_seq_oss_midi_info_read(struct snd_info_buffer *buf)
+{
+ int i;
+ struct seq_oss_midi *mdev;
+
+ snd_iprintf(buf, "\nNumber of MIDI devices: %d\n", max_midi_devs);
+ for (i = 0; i < max_midi_devs; i++) {
+ snd_iprintf(buf, "\nmidi %d: ", i);
+ mdev = get_mdev(i);
+ if (mdev == NULL) {
+ snd_iprintf(buf, "*empty*\n");
+ continue;
+ }
+ snd_iprintf(buf, "[%s] ALSA port %d:%d\n", mdev->name,
+ mdev->client, mdev->port);
+ snd_iprintf(buf, " capability %s / opened %s\n",
+ capmode_str(mdev->flags),
+ capmode_str(mdev->opened));
+ snd_use_lock_free(&mdev->use_lock);
+ }
+}
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/core/seq/oss/seq_oss_midi.h b/sound/core/seq/oss/seq_oss_midi.h
new file mode 100644
index 0000000..84eb866
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_midi.h
@@ -0,0 +1,48 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * midi device information
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SEQ_OSS_MIDI_H
+#define __SEQ_OSS_MIDI_H
+
+#include "seq_oss_device.h"
+#include <sound/seq_oss_legacy.h>
+
+int snd_seq_oss_midi_lookup_ports(int client);
+int snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo);
+int snd_seq_oss_midi_check_exit_port(int client, int port);
+void snd_seq_oss_midi_clear_all(void);
+
+void snd_seq_oss_midi_setup(struct seq_oss_devinfo *dp);
+void snd_seq_oss_midi_cleanup(struct seq_oss_devinfo *dp);
+
+int snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int file_mode);
+void snd_seq_oss_midi_open_all(struct seq_oss_devinfo *dp, int file_mode);
+int snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev);
+void snd_seq_oss_midi_reset(struct seq_oss_devinfo *dp, int dev);
+int snd_seq_oss_midi_putc(struct seq_oss_devinfo *dp, int dev, unsigned char c,
+ struct snd_seq_event *ev);
+int snd_seq_oss_midi_input(struct snd_seq_event *ev, int direct, void *private);
+int snd_seq_oss_midi_filemode(struct seq_oss_devinfo *dp, int dev);
+int snd_seq_oss_midi_make_info(struct seq_oss_devinfo *dp, int dev, struct midi_info *inf);
+void snd_seq_oss_midi_get_addr(struct seq_oss_devinfo *dp, int dev, struct snd_seq_addr *addr);
+
+#endif
diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c
new file mode 100644
index 0000000..f5de79f
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_readq.c
@@ -0,0 +1,236 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * seq_oss_readq.c - MIDI input queue
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_readq.h"
+#include "seq_oss_event.h"
+#include <sound/seq_oss_legacy.h>
+#include "../seq_lock.h"
+#include <linux/wait.h>
+
+/*
+ * constants
+ */
+//#define SNDRV_SEQ_OSS_MAX_TIMEOUT (unsigned long)(-1)
+#define SNDRV_SEQ_OSS_MAX_TIMEOUT (HZ * 3600)
+
+
+/*
+ * prototypes
+ */
+
+
+/*
+ * create a read queue
+ */
+struct seq_oss_readq *
+snd_seq_oss_readq_new(struct seq_oss_devinfo *dp, int maxlen)
+{
+ struct seq_oss_readq *q;
+
+ if ((q = kzalloc(sizeof(*q), GFP_KERNEL)) == NULL) {
+ snd_printk(KERN_ERR "can't malloc read queue\n");
+ return NULL;
+ }
+
+ if ((q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL)) == NULL) {
+ snd_printk(KERN_ERR "can't malloc read queue buffer\n");
+ kfree(q);
+ return NULL;
+ }
+
+ q->maxlen = maxlen;
+ q->qlen = 0;
+ q->head = q->tail = 0;
+ init_waitqueue_head(&q->midi_sleep);
+ spin_lock_init(&q->lock);
+ q->pre_event_timeout = SNDRV_SEQ_OSS_MAX_TIMEOUT;
+ q->input_time = (unsigned long)-1;
+
+ return q;
+}
+
+/*
+ * delete the read queue
+ */
+void
+snd_seq_oss_readq_delete(struct seq_oss_readq *q)
+{
+ if (q) {
+ kfree(q->q);
+ kfree(q);
+ }
+}
+
+/*
+ * reset the read queue
+ */
+void
+snd_seq_oss_readq_clear(struct seq_oss_readq *q)
+{
+ if (q->qlen) {
+ q->qlen = 0;
+ q->head = q->tail = 0;
+ }
+ /* if someone sleeping, wake'em up */
+ if (waitqueue_active(&q->midi_sleep))
+ wake_up(&q->midi_sleep);
+ q->input_time = (unsigned long)-1;
+}
+
+/*
+ * put a midi byte
+ */
+int
+snd_seq_oss_readq_puts(struct seq_oss_readq *q, int dev, unsigned char *data, int len)
+{
+ union evrec rec;
+ int result;
+
+ memset(&rec, 0, sizeof(rec));
+ rec.c[0] = SEQ_MIDIPUTC;
+ rec.c[2] = dev;
+
+ while (len-- > 0) {
+ rec.c[1] = *data++;
+ result = snd_seq_oss_readq_put_event(q, &rec);
+ if (result < 0)
+ return result;
+ }
+ return 0;
+}
+
+/*
+ * copy an event to input queue:
+ * return zero if enqueued
+ */
+int
+snd_seq_oss_readq_put_event(struct seq_oss_readq *q, union evrec *ev)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&q->lock, flags);
+ if (q->qlen >= q->maxlen - 1) {
+ spin_unlock_irqrestore(&q->lock, flags);
+ return -ENOMEM;
+ }
+
+ memcpy(&q->q[q->tail], ev, sizeof(*ev));
+ q->tail = (q->tail + 1) % q->maxlen;
+ q->qlen++;
+
+ /* wake up sleeper */
+ if (waitqueue_active(&q->midi_sleep))
+ wake_up(&q->midi_sleep);
+
+ spin_unlock_irqrestore(&q->lock, flags);
+
+ return 0;
+}
+
+
+/*
+ * pop queue
+ * caller must hold lock
+ */
+int
+snd_seq_oss_readq_pick(struct seq_oss_readq *q, union evrec *rec)
+{
+ if (q->qlen == 0)
+ return -EAGAIN;
+ memcpy(rec, &q->q[q->head], sizeof(*rec));
+ return 0;
+}
+
+/*
+ * sleep until ready
+ */
+void
+snd_seq_oss_readq_wait(struct seq_oss_readq *q)
+{
+ wait_event_interruptible_timeout(q->midi_sleep,
+ (q->qlen > 0 || q->head == q->tail),
+ q->pre_event_timeout);
+}
+
+/*
+ * drain one record
+ * caller must hold lock
+ */
+void
+snd_seq_oss_readq_free(struct seq_oss_readq *q)
+{
+ if (q->qlen > 0) {
+ q->head = (q->head + 1) % q->maxlen;
+ q->qlen--;
+ }
+}
+
+/*
+ * polling/select:
+ * return non-zero if readq is not empty.
+ */
+unsigned int
+snd_seq_oss_readq_poll(struct seq_oss_readq *q, struct file *file, poll_table *wait)
+{
+ poll_wait(file, &q->midi_sleep, wait);
+ return q->qlen;
+}
+
+/*
+ * put a timestamp
+ */
+int
+snd_seq_oss_readq_put_timestamp(struct seq_oss_readq *q, unsigned long curt, int seq_mode)
+{
+ if (curt != q->input_time) {
+ union evrec rec;
+ memset(&rec, 0, sizeof(rec));
+ switch (seq_mode) {
+ case SNDRV_SEQ_OSS_MODE_SYNTH:
+ rec.echo = (curt << 8) | SEQ_WAIT;
+ snd_seq_oss_readq_put_event(q, &rec);
+ break;
+ case SNDRV_SEQ_OSS_MODE_MUSIC:
+ rec.t.code = EV_TIMING;
+ rec.t.cmd = TMR_WAIT_ABS;
+ rec.t.time = curt;
+ snd_seq_oss_readq_put_event(q, &rec);
+ break;
+ }
+ q->input_time = curt;
+ }
+ return 0;
+}
+
+
+#ifdef CONFIG_PROC_FS
+/*
+ * proc interface
+ */
+void
+snd_seq_oss_readq_info_read(struct seq_oss_readq *q, struct snd_info_buffer *buf)
+{
+ snd_iprintf(buf, " read queue [%s] length = %d : tick = %ld\n",
+ (waitqueue_active(&q->midi_sleep) ? "sleeping":"running"),
+ q->qlen, q->input_time);
+}
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/core/seq/oss/seq_oss_readq.h b/sound/core/seq/oss/seq_oss_readq.h
new file mode 100644
index 0000000..f1463f1
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_readq.h
@@ -0,0 +1,56 @@
+/*
+ * OSS compatible sequencer driver
+ * read fifo queue
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SEQ_OSS_READQ_H
+#define __SEQ_OSS_READQ_H
+
+#include "seq_oss_device.h"
+
+
+/*
+ * definition of read queue
+ */
+struct seq_oss_readq {
+ union evrec *q;
+ int qlen;
+ int maxlen;
+ int head, tail;
+ unsigned long pre_event_timeout;
+ unsigned long input_time;
+ wait_queue_head_t midi_sleep;
+ spinlock_t lock;
+};
+
+struct seq_oss_readq *snd_seq_oss_readq_new(struct seq_oss_devinfo *dp, int maxlen);
+void snd_seq_oss_readq_delete(struct seq_oss_readq *q);
+void snd_seq_oss_readq_clear(struct seq_oss_readq *readq);
+unsigned int snd_seq_oss_readq_poll(struct seq_oss_readq *readq, struct file *file, poll_table *wait);
+int snd_seq_oss_readq_puts(struct seq_oss_readq *readq, int dev, unsigned char *data, int len);
+int snd_seq_oss_readq_put_event(struct seq_oss_readq *readq, union evrec *ev);
+int snd_seq_oss_readq_put_timestamp(struct seq_oss_readq *readq, unsigned long curt, int seq_mode);
+int snd_seq_oss_readq_pick(struct seq_oss_readq *q, union evrec *rec);
+void snd_seq_oss_readq_wait(struct seq_oss_readq *q);
+void snd_seq_oss_readq_free(struct seq_oss_readq *q);
+
+#define snd_seq_oss_readq_lock(q, flags) spin_lock_irqsave(&(q)->lock, flags)
+#define snd_seq_oss_readq_unlock(q, flags) spin_unlock_irqrestore(&(q)->lock, flags)
+
+#endif
diff --git a/sound/core/seq/oss/seq_oss_rw.c b/sound/core/seq/oss/seq_oss_rw.c
new file mode 100644
index 0000000..6a7b6ac
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_rw.c
@@ -0,0 +1,216 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * read/write/select interface to device file
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_device.h"
+#include "seq_oss_readq.h"
+#include "seq_oss_writeq.h"
+#include "seq_oss_synth.h"
+#include <sound/seq_oss_legacy.h>
+#include "seq_oss_event.h"
+#include "seq_oss_timer.h"
+#include "../seq_clientmgr.h"
+
+
+/*
+ * protoypes
+ */
+static int insert_queue(struct seq_oss_devinfo *dp, union evrec *rec, struct file *opt);
+
+
+/*
+ * read interface
+ */
+
+int
+snd_seq_oss_read(struct seq_oss_devinfo *dp, char __user *buf, int count)
+{
+ struct seq_oss_readq *readq = dp->readq;
+ int result = 0, err = 0;
+ int ev_len;
+ union evrec rec;
+ unsigned long flags;
+
+ if (readq == NULL || ! is_read_mode(dp->file_mode))
+ return -ENXIO;
+
+ while (count >= SHORT_EVENT_SIZE) {
+ snd_seq_oss_readq_lock(readq, flags);
+ err = snd_seq_oss_readq_pick(readq, &rec);
+ if (err == -EAGAIN &&
+ !is_nonblock_mode(dp->file_mode) && result == 0) {
+ snd_seq_oss_readq_unlock(readq, flags);
+ snd_seq_oss_readq_wait(readq);
+ snd_seq_oss_readq_lock(readq, flags);
+ if (signal_pending(current))
+ err = -ERESTARTSYS;
+ else
+ err = snd_seq_oss_readq_pick(readq, &rec);
+ }
+ if (err < 0) {
+ snd_seq_oss_readq_unlock(readq, flags);
+ break;
+ }
+ ev_len = ev_length(&rec);
+ if (ev_len < count) {
+ snd_seq_oss_readq_unlock(readq, flags);
+ break;
+ }
+ snd_seq_oss_readq_free(readq);
+ snd_seq_oss_readq_unlock(readq, flags);
+ if (copy_to_user(buf, &rec, ev_len)) {
+ err = -EFAULT;
+ break;
+ }
+ result += ev_len;
+ buf += ev_len;
+ count -= ev_len;
+ }
+ return result > 0 ? result : err;
+}
+
+
+/*
+ * write interface
+ */
+
+int
+snd_seq_oss_write(struct seq_oss_devinfo *dp, const char __user *buf, int count, struct file *opt)
+{
+ int result = 0, err = 0;
+ int ev_size, fmt;
+ union evrec rec;
+
+ if (! is_write_mode(dp->file_mode) || dp->writeq == NULL)
+ return -ENXIO;
+
+ while (count >= SHORT_EVENT_SIZE) {
+ if (copy_from_user(&rec, buf, SHORT_EVENT_SIZE)) {
+ err = -EFAULT;
+ break;
+ }
+ if (rec.s.code == SEQ_FULLSIZE) {
+ /* load patch */
+ if (result > 0) {
+ err = -EINVAL;
+ break;
+ }
+ fmt = (*(unsigned short *)rec.c) & 0xffff;
+ /* FIXME the return value isn't correct */
+ return snd_seq_oss_synth_load_patch(dp, rec.s.dev,
+ fmt, buf, 0, count);
+ }
+ if (ev_is_long(&rec)) {
+ /* extended code */
+ if (rec.s.code == SEQ_EXTENDED &&
+ dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC) {
+ err = -EINVAL;
+ break;
+ }
+ ev_size = LONG_EVENT_SIZE;
+ if (count < ev_size)
+ break;
+ /* copy the reset 4 bytes */
+ if (copy_from_user(rec.c + SHORT_EVENT_SIZE,
+ buf + SHORT_EVENT_SIZE,
+ LONG_EVENT_SIZE - SHORT_EVENT_SIZE)) {
+ err = -EFAULT;
+ break;
+ }
+ } else {
+ /* old-type code */
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC) {
+ err = -EINVAL;
+ break;
+ }
+ ev_size = SHORT_EVENT_SIZE;
+ }
+
+ /* insert queue */
+ if ((err = insert_queue(dp, &rec, opt)) < 0)
+ break;
+
+ result += ev_size;
+ buf += ev_size;
+ count -= ev_size;
+ }
+ return result > 0 ? result : err;
+}
+
+
+/*
+ * insert event record to write queue
+ * return: 0 = OK, non-zero = NG
+ */
+static int
+insert_queue(struct seq_oss_devinfo *dp, union evrec *rec, struct file *opt)
+{
+ int rc = 0;
+ struct snd_seq_event event;
+
+ /* if this is a timing event, process the current time */
+ if (snd_seq_oss_process_timer_event(dp->timer, rec))
+ return 0; /* no need to insert queue */
+
+ /* parse this event */
+ memset(&event, 0, sizeof(event));
+ /* set dummy -- to be sure */
+ event.type = SNDRV_SEQ_EVENT_NOTEOFF;
+ snd_seq_oss_fill_addr(dp, &event, dp->addr.port, dp->addr.client);
+
+ if (snd_seq_oss_process_event(dp, rec, &event))
+ return 0; /* invalid event - no need to insert queue */
+
+ event.time.tick = snd_seq_oss_timer_cur_tick(dp->timer);
+ if (dp->timer->realtime || !dp->timer->running) {
+ snd_seq_oss_dispatch(dp, &event, 0, 0);
+ } else {
+ if (is_nonblock_mode(dp->file_mode))
+ rc = snd_seq_kernel_client_enqueue(dp->cseq, &event, 0, 0);
+ else
+ rc = snd_seq_kernel_client_enqueue_blocking(dp->cseq, &event, opt, 0, 0);
+ }
+ return rc;
+}
+
+
+/*
+ * select / poll
+ */
+
+unsigned int
+snd_seq_oss_poll(struct seq_oss_devinfo *dp, struct file *file, poll_table * wait)
+{
+ unsigned int mask = 0;
+
+ /* input */
+ if (dp->readq && is_read_mode(dp->file_mode)) {
+ if (snd_seq_oss_readq_poll(dp->readq, file, wait))
+ mask |= POLLIN | POLLRDNORM;
+ }
+
+ /* output */
+ if (dp->writeq && is_write_mode(dp->file_mode)) {
+ if (snd_seq_kernel_client_write_poll(dp->cseq, file, wait))
+ mask |= POLLOUT | POLLWRNORM;
+ }
+ return mask;
+}
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
new file mode 100644
index 0000000..945a27c
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -0,0 +1,662 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * synth device handlers
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_synth.h"
+#include "seq_oss_midi.h"
+#include "../seq_lock.h"
+#include <linux/init.h>
+
+/*
+ * constants
+ */
+#define SNDRV_SEQ_OSS_MAX_SYNTH_NAME 30
+#define MAX_SYSEX_BUFLEN 128
+
+
+/*
+ * definition of synth info records
+ */
+
+/* sysex buffer */
+struct seq_oss_synth_sysex {
+ int len;
+ int skip;
+ unsigned char buf[MAX_SYSEX_BUFLEN];
+};
+
+/* synth info */
+struct seq_oss_synth {
+ int seq_device;
+
+ /* for synth_info */
+ int synth_type;
+ int synth_subtype;
+ int nr_voices;
+
+ char name[SNDRV_SEQ_OSS_MAX_SYNTH_NAME];
+ struct snd_seq_oss_callback oper;
+
+ int opened;
+
+ void *private_data;
+ snd_use_lock_t use_lock;
+};
+
+
+/*
+ * device table
+ */
+static int max_synth_devs;
+static struct seq_oss_synth *synth_devs[SNDRV_SEQ_OSS_MAX_SYNTH_DEVS];
+static struct seq_oss_synth midi_synth_dev = {
+ -1, /* seq_device */
+ SYNTH_TYPE_MIDI, /* synth_type */
+ 0, /* synth_subtype */
+ 16, /* nr_voices */
+ "MIDI", /* name */
+};
+
+static DEFINE_SPINLOCK(register_lock);
+
+/*
+ * prototypes
+ */
+static struct seq_oss_synth *get_synthdev(struct seq_oss_devinfo *dp, int dev);
+static void reset_channels(struct seq_oss_synthinfo *info);
+
+/*
+ * global initialization
+ */
+void __init
+snd_seq_oss_synth_init(void)
+{
+ snd_use_lock_init(&midi_synth_dev.use_lock);
+}
+
+/*
+ * registration of the synth device
+ */
+int
+snd_seq_oss_synth_register(struct snd_seq_device *dev)
+{
+ int i;
+ struct seq_oss_synth *rec;
+ struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ unsigned long flags;
+
+ if ((rec = kzalloc(sizeof(*rec), GFP_KERNEL)) == NULL) {
+ snd_printk(KERN_ERR "can't malloc synth info\n");
+ return -ENOMEM;
+ }
+ rec->seq_device = -1;
+ rec->synth_type = reg->type;
+ rec->synth_subtype = reg->subtype;
+ rec->nr_voices = reg->nvoices;
+ rec->oper = reg->oper;
+ rec->private_data = reg->private_data;
+ rec->opened = 0;
+ snd_use_lock_init(&rec->use_lock);
+
+ /* copy and truncate the name of synth device */
+ strlcpy(rec->name, dev->name, sizeof(rec->name));
+
+ /* registration */
+ spin_lock_irqsave(&register_lock, flags);
+ for (i = 0; i < max_synth_devs; i++) {
+ if (synth_devs[i] == NULL)
+ break;
+ }
+ if (i >= max_synth_devs) {
+ if (max_synth_devs >= SNDRV_SEQ_OSS_MAX_SYNTH_DEVS) {
+ spin_unlock_irqrestore(&register_lock, flags);
+ snd_printk(KERN_ERR "no more synth slot\n");
+ kfree(rec);
+ return -ENOMEM;
+ }
+ max_synth_devs++;
+ }
+ rec->seq_device = i;
+ synth_devs[i] = rec;
+ debug_printk(("synth %s registered %d\n", rec->name, i));
+ spin_unlock_irqrestore(&register_lock, flags);
+ dev->driver_data = rec;
+#ifdef SNDRV_OSS_INFO_DEV_SYNTH
+ if (i < SNDRV_CARDS)
+ snd_oss_info_register(SNDRV_OSS_INFO_DEV_SYNTH, i, rec->name);
+#endif
+ return 0;
+}
+
+
+int
+snd_seq_oss_synth_unregister(struct snd_seq_device *dev)
+{
+ int index;
+ struct seq_oss_synth *rec = dev->driver_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&register_lock, flags);
+ for (index = 0; index < max_synth_devs; index++) {
+ if (synth_devs[index] == rec)
+ break;
+ }
+ if (index >= max_synth_devs) {
+ spin_unlock_irqrestore(&register_lock, flags);
+ snd_printk(KERN_ERR "can't unregister synth\n");
+ return -EINVAL;
+ }
+ synth_devs[index] = NULL;
+ if (index == max_synth_devs - 1) {
+ for (index--; index >= 0; index--) {
+ if (synth_devs[index])
+ break;
+ }
+ max_synth_devs = index + 1;
+ }
+ spin_unlock_irqrestore(&register_lock, flags);
+#ifdef SNDRV_OSS_INFO_DEV_SYNTH
+ if (rec->seq_device < SNDRV_CARDS)
+ snd_oss_info_unregister(SNDRV_OSS_INFO_DEV_SYNTH, rec->seq_device);
+#endif
+
+ snd_use_lock_sync(&rec->use_lock);
+ kfree(rec);
+
+ return 0;
+}
+
+
+/*
+ */
+static struct seq_oss_synth *
+get_sdev(int dev)
+{
+ struct seq_oss_synth *rec;
+ unsigned long flags;
+
+ spin_lock_irqsave(&register_lock, flags);
+ rec = synth_devs[dev];
+ if (rec)
+ snd_use_lock_use(&rec->use_lock);
+ spin_unlock_irqrestore(&register_lock, flags);
+ return rec;
+}
+
+
+/*
+ * set up synth tables
+ */
+
+void
+snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp)
+{
+ int i;
+ struct seq_oss_synth *rec;
+ struct seq_oss_synthinfo *info;
+
+ dp->max_synthdev = max_synth_devs;
+ dp->synth_opened = 0;
+ memset(dp->synths, 0, sizeof(dp->synths));
+ for (i = 0; i < dp->max_synthdev; i++) {
+ rec = get_sdev(i);
+ if (rec == NULL)
+ continue;
+ if (rec->oper.open == NULL || rec->oper.close == NULL) {
+ snd_use_lock_free(&rec->use_lock);
+ continue;
+ }
+ info = &dp->synths[i];
+ info->arg.app_index = dp->port;
+ info->arg.file_mode = dp->file_mode;
+ info->arg.seq_mode = dp->seq_mode;
+ if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_SYNTH)
+ info->arg.event_passing = SNDRV_SEQ_OSS_PROCESS_EVENTS;
+ else
+ info->arg.event_passing = SNDRV_SEQ_OSS_PASS_EVENTS;
+ info->opened = 0;
+ if (!try_module_get(rec->oper.owner)) {
+ snd_use_lock_free(&rec->use_lock);
+ continue;
+ }
+ if (rec->oper.open(&info->arg, rec->private_data) < 0) {
+ module_put(rec->oper.owner);
+ snd_use_lock_free(&rec->use_lock);
+ continue;
+ }
+ info->nr_voices = rec->nr_voices;
+ if (info->nr_voices > 0) {
+ info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL);
+ if (!info->ch) {
+ snd_printk(KERN_ERR "Cannot malloc\n");
+ rec->oper.close(&info->arg);
+ module_put(rec->oper.owner);
+ snd_use_lock_free(&rec->use_lock);
+ continue;
+ }
+ reset_channels(info);
+ }
+ debug_printk(("synth %d assigned\n", i));
+ info->opened++;
+ rec->opened++;
+ dp->synth_opened++;
+ snd_use_lock_free(&rec->use_lock);
+ }
+}
+
+
+/*
+ * set up synth tables for MIDI emulation - /dev/music mode only
+ */
+
+void
+snd_seq_oss_synth_setup_midi(struct seq_oss_devinfo *dp)
+{
+ int i;
+
+ if (dp->max_synthdev >= SNDRV_SEQ_OSS_MAX_SYNTH_DEVS)
+ return;
+
+ for (i = 0; i < dp->max_mididev; i++) {
+ struct seq_oss_synthinfo *info;
+ info = &dp->synths[dp->max_synthdev];
+ if (snd_seq_oss_midi_open(dp, i, dp->file_mode) < 0)
+ continue;
+ info->arg.app_index = dp->port;
+ info->arg.file_mode = dp->file_mode;
+ info->arg.seq_mode = dp->seq_mode;
+ info->arg.private_data = info;
+ info->is_midi = 1;
+ info->midi_mapped = i;
+ info->arg.event_passing = SNDRV_SEQ_OSS_PASS_EVENTS;
+ snd_seq_oss_midi_get_addr(dp, i, &info->arg.addr);
+ info->opened = 1;
+ midi_synth_dev.opened++;
+ dp->max_synthdev++;
+ if (dp->max_synthdev >= SNDRV_SEQ_OSS_MAX_SYNTH_DEVS)
+ break;
+ }
+}
+
+
+/*
+ * clean up synth tables
+ */
+
+void
+snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp)
+{
+ int i;
+ struct seq_oss_synth *rec;
+ struct seq_oss_synthinfo *info;
+
+ if (snd_BUG_ON(dp->max_synthdev >= SNDRV_SEQ_OSS_MAX_SYNTH_DEVS))
+ return;
+ for (i = 0; i < dp->max_synthdev; i++) {
+ info = &dp->synths[i];
+ if (! info->opened)
+ continue;
+ if (info->is_midi) {
+ if (midi_synth_dev.opened > 0) {
+ snd_seq_oss_midi_close(dp, info->midi_mapped);
+ midi_synth_dev.opened--;
+ }
+ } else {
+ rec = get_sdev(i);
+ if (rec == NULL)
+ continue;
+ if (rec->opened > 0) {
+ debug_printk(("synth %d closed\n", i));
+ rec->oper.close(&info->arg);
+ module_put(rec->oper.owner);
+ rec->opened = 0;
+ }
+ snd_use_lock_free(&rec->use_lock);
+ }
+ kfree(info->sysex);
+ info->sysex = NULL;
+ kfree(info->ch);
+ info->ch = NULL;
+ }
+ dp->synth_opened = 0;
+ dp->max_synthdev = 0;
+}
+
+/*
+ * check if the specified device is MIDI mapped device
+ */
+static int
+is_midi_dev(struct seq_oss_devinfo *dp, int dev)
+{
+ if (dev < 0 || dev >= dp->max_synthdev)
+ return 0;
+ if (dp->synths[dev].is_midi)
+ return 1;
+ return 0;
+}
+
+/*
+ * return synth device information pointer
+ */
+static struct seq_oss_synth *
+get_synthdev(struct seq_oss_devinfo *dp, int dev)
+{
+ struct seq_oss_synth *rec;
+ if (dev < 0 || dev >= dp->max_synthdev)
+ return NULL;
+ if (! dp->synths[dev].opened)
+ return NULL;
+ if (dp->synths[dev].is_midi)
+ return &midi_synth_dev;
+ if ((rec = get_sdev(dev)) == NULL)
+ return NULL;
+ if (! rec->opened) {
+ snd_use_lock_free(&rec->use_lock);
+ return NULL;
+ }
+ return rec;
+}
+
+
+/*
+ * reset note and velocity on each channel.
+ */
+static void
+reset_channels(struct seq_oss_synthinfo *info)
+{
+ int i;
+ if (info->ch == NULL || ! info->nr_voices)
+ return;
+ for (i = 0; i < info->nr_voices; i++) {
+ info->ch[i].note = -1;
+ info->ch[i].vel = 0;
+ }
+}
+
+
+/*
+ * reset synth device:
+ * call reset callback. if no callback is defined, send a heartbeat
+ * event to the corresponding port.
+ */
+void
+snd_seq_oss_synth_reset(struct seq_oss_devinfo *dp, int dev)
+{
+ struct seq_oss_synth *rec;
+ struct seq_oss_synthinfo *info;
+
+ if (snd_BUG_ON(dev < 0 || dev >= dp->max_synthdev))
+ return;
+ info = &dp->synths[dev];
+ if (! info->opened)
+ return;
+ if (info->sysex)
+ info->sysex->len = 0; /* reset sysex */
+ reset_channels(info);
+ if (info->is_midi) {
+ if (midi_synth_dev.opened <= 0)
+ return;
+ snd_seq_oss_midi_reset(dp, info->midi_mapped);
+ /* reopen the device */
+ snd_seq_oss_midi_close(dp, dev);
+ if (snd_seq_oss_midi_open(dp, info->midi_mapped,
+ dp->file_mode) < 0) {
+ midi_synth_dev.opened--;
+ info->opened = 0;
+ kfree(info->sysex);
+ info->sysex = NULL;
+ kfree(info->ch);
+ info->ch = NULL;
+ }
+ return;
+ }
+
+ rec = get_sdev(dev);
+ if (rec == NULL)
+ return;
+ if (rec->oper.reset) {
+ rec->oper.reset(&info->arg);
+ } else {
+ struct snd_seq_event ev;
+ memset(&ev, 0, sizeof(ev));
+ snd_seq_oss_fill_addr(dp, &ev, info->arg.addr.client,
+ info->arg.addr.port);
+ ev.type = SNDRV_SEQ_EVENT_RESET;
+ snd_seq_oss_dispatch(dp, &ev, 0, 0);
+ }
+ snd_use_lock_free(&rec->use_lock);
+}
+
+
+/*
+ * load a patch record:
+ * call load_patch callback function
+ */
+int
+snd_seq_oss_synth_load_patch(struct seq_oss_devinfo *dp, int dev, int fmt,
+ const char __user *buf, int p, int c)
+{
+ struct seq_oss_synth *rec;
+ int rc;
+
+ if (dev < 0 || dev >= dp->max_synthdev)
+ return -ENXIO;
+
+ if (is_midi_dev(dp, dev))
+ return 0;
+ if ((rec = get_synthdev(dp, dev)) == NULL)
+ return -ENXIO;
+
+ if (rec->oper.load_patch == NULL)
+ rc = -ENXIO;
+ else
+ rc = rec->oper.load_patch(&dp->synths[dev].arg, fmt, buf, p, c);
+ snd_use_lock_free(&rec->use_lock);
+ return rc;
+}
+
+/*
+ * check if the device is valid synth device
+ */
+int
+snd_seq_oss_synth_is_valid(struct seq_oss_devinfo *dp, int dev)
+{
+ struct seq_oss_synth *rec;
+ rec = get_synthdev(dp, dev);
+ if (rec) {
+ snd_use_lock_free(&rec->use_lock);
+ return 1;
+ }
+ return 0;
+}
+
+
+/*
+ * receive OSS 6 byte sysex packet:
+ * the full sysex message will be sent if it reaches to the end of data
+ * (0xff).
+ */
+int
+snd_seq_oss_synth_sysex(struct seq_oss_devinfo *dp, int dev, unsigned char *buf, struct snd_seq_event *ev)
+{
+ int i, send;
+ unsigned char *dest;
+ struct seq_oss_synth_sysex *sysex;
+
+ if (! snd_seq_oss_synth_is_valid(dp, dev))
+ return -ENXIO;
+
+ sysex = dp->synths[dev].sysex;
+ if (sysex == NULL) {
+ sysex = kzalloc(sizeof(*sysex), GFP_KERNEL);
+ if (sysex == NULL)
+ return -ENOMEM;
+ dp->synths[dev].sysex = sysex;
+ }
+
+ send = 0;
+ dest = sysex->buf + sysex->len;
+ /* copy 6 byte packet to the buffer */
+ for (i = 0; i < 6; i++) {
+ if (buf[i] == 0xff) {
+ send = 1;
+ break;
+ }
+ dest[i] = buf[i];
+ sysex->len++;
+ if (sysex->len >= MAX_SYSEX_BUFLEN) {
+ sysex->len = 0;
+ sysex->skip = 1;
+ break;
+ }
+ }
+
+ if (sysex->len && send) {
+ if (sysex->skip) {
+ sysex->skip = 0;
+ sysex->len = 0;
+ return -EINVAL; /* skip */
+ }
+ /* copy the data to event record and send it */
+ ev->flags = SNDRV_SEQ_EVENT_LENGTH_VARIABLE;
+ if (snd_seq_oss_synth_addr(dp, dev, ev))
+ return -EINVAL;
+ ev->data.ext.len = sysex->len;
+ ev->data.ext.ptr = sysex->buf;
+ sysex->len = 0;
+ return 0;
+ }
+
+ return -EINVAL; /* skip */
+}
+
+/*
+ * fill the event source/destination addresses
+ */
+int
+snd_seq_oss_synth_addr(struct seq_oss_devinfo *dp, int dev, struct snd_seq_event *ev)
+{
+ if (! snd_seq_oss_synth_is_valid(dp, dev))
+ return -EINVAL;
+ snd_seq_oss_fill_addr(dp, ev, dp->synths[dev].arg.addr.client,
+ dp->synths[dev].arg.addr.port);
+ return 0;
+}
+
+
+/*
+ * OSS compatible ioctl
+ */
+int
+snd_seq_oss_synth_ioctl(struct seq_oss_devinfo *dp, int dev, unsigned int cmd, unsigned long addr)
+{
+ struct seq_oss_synth *rec;
+ int rc;
+
+ if (is_midi_dev(dp, dev))
+ return -ENXIO;
+ if ((rec = get_synthdev(dp, dev)) == NULL)
+ return -ENXIO;
+ if (rec->oper.ioctl == NULL)
+ rc = -ENXIO;
+ else
+ rc = rec->oper.ioctl(&dp->synths[dev].arg, cmd, addr);
+ snd_use_lock_free(&rec->use_lock);
+ return rc;
+}
+
+
+/*
+ * send OSS raw events - SEQ_PRIVATE and SEQ_VOLUME
+ */
+int
+snd_seq_oss_synth_raw_event(struct seq_oss_devinfo *dp, int dev, unsigned char *data, struct snd_seq_event *ev)
+{
+ if (! snd_seq_oss_synth_is_valid(dp, dev) || is_midi_dev(dp, dev))
+ return -ENXIO;
+ ev->type = SNDRV_SEQ_EVENT_OSS;
+ memcpy(ev->data.raw8.d, data, 8);
+ return snd_seq_oss_synth_addr(dp, dev, ev);
+}
+
+
+/*
+ * create OSS compatible synth_info record
+ */
+int
+snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_info *inf)
+{
+ struct seq_oss_synth *rec;
+
+ if (dev < 0 || dev >= dp->max_synthdev)
+ return -ENXIO;
+
+ if (dp->synths[dev].is_midi) {
+ struct midi_info minf;
+ snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf);
+ inf->synth_type = SYNTH_TYPE_MIDI;
+ inf->synth_subtype = 0;
+ inf->nr_voices = 16;
+ inf->device = dev;
+ strlcpy(inf->name, minf.name, sizeof(inf->name));
+ } else {
+ if ((rec = get_synthdev(dp, dev)) == NULL)
+ return -ENXIO;
+ inf->synth_type = rec->synth_type;
+ inf->synth_subtype = rec->synth_subtype;
+ inf->nr_voices = rec->nr_voices;
+ inf->device = dev;
+ strlcpy(inf->name, rec->name, sizeof(inf->name));
+ snd_use_lock_free(&rec->use_lock);
+ }
+ return 0;
+}
+
+
+#ifdef CONFIG_PROC_FS
+/*
+ * proc interface
+ */
+void
+snd_seq_oss_synth_info_read(struct snd_info_buffer *buf)
+{
+ int i;
+ struct seq_oss_synth *rec;
+
+ snd_iprintf(buf, "\nNumber of synth devices: %d\n", max_synth_devs);
+ for (i = 0; i < max_synth_devs; i++) {
+ snd_iprintf(buf, "\nsynth %d: ", i);
+ rec = get_sdev(i);
+ if (rec == NULL) {
+ snd_iprintf(buf, "*empty*\n");
+ continue;
+ }
+ snd_iprintf(buf, "[%s]\n", rec->name);
+ snd_iprintf(buf, " type 0x%x : subtype 0x%x : voices %d\n",
+ rec->synth_type, rec->synth_subtype,
+ rec->nr_voices);
+ snd_iprintf(buf, " capabilities : ioctl %s / load_patch %s\n",
+ enabled_str((long)rec->oper.ioctl),
+ enabled_str((long)rec->oper.load_patch));
+ snd_use_lock_free(&rec->use_lock);
+ }
+}
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/core/seq/oss/seq_oss_synth.h b/sound/core/seq/oss/seq_oss_synth.h
new file mode 100644
index 0000000..dbdfcbb
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_synth.h
@@ -0,0 +1,51 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * synth device information
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SEQ_OSS_SYNTH_H
+#define __SEQ_OSS_SYNTH_H
+
+#include "seq_oss_device.h"
+#include <sound/seq_oss_legacy.h>
+#include <sound/seq_device.h>
+
+void snd_seq_oss_synth_init(void);
+int snd_seq_oss_synth_register(struct snd_seq_device *dev);
+int snd_seq_oss_synth_unregister(struct snd_seq_device *dev);
+void snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp);
+void snd_seq_oss_synth_setup_midi(struct seq_oss_devinfo *dp);
+void snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp);
+
+void snd_seq_oss_synth_reset(struct seq_oss_devinfo *dp, int dev);
+int snd_seq_oss_synth_load_patch(struct seq_oss_devinfo *dp, int dev, int fmt,
+ const char __user *buf, int p, int c);
+int snd_seq_oss_synth_is_valid(struct seq_oss_devinfo *dp, int dev);
+int snd_seq_oss_synth_sysex(struct seq_oss_devinfo *dp, int dev, unsigned char *buf,
+ struct snd_seq_event *ev);
+int snd_seq_oss_synth_addr(struct seq_oss_devinfo *dp, int dev, struct snd_seq_event *ev);
+int snd_seq_oss_synth_ioctl(struct seq_oss_devinfo *dp, int dev, unsigned int cmd,
+ unsigned long addr);
+int snd_seq_oss_synth_raw_event(struct seq_oss_devinfo *dp, int dev,
+ unsigned char *data, struct snd_seq_event *ev);
+
+int snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_info *inf);
+
+#endif
diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c
new file mode 100644
index 0000000..c440fda
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_timer.c
@@ -0,0 +1,283 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * Timer control routines
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_timer.h"
+#include "seq_oss_event.h"
+#include <sound/seq_oss_legacy.h>
+
+/*
+ */
+#define MIN_OSS_TEMPO 8
+#define MAX_OSS_TEMPO 360
+#define MIN_OSS_TIMEBASE 1
+#define MAX_OSS_TIMEBASE 1000
+
+/*
+ */
+static void calc_alsa_tempo(struct seq_oss_timer *timer);
+static int send_timer_event(struct seq_oss_devinfo *dp, int type, int value);
+
+
+/*
+ * create and register a new timer.
+ * if queue is not started yet, start it.
+ */
+struct seq_oss_timer *
+snd_seq_oss_timer_new(struct seq_oss_devinfo *dp)
+{
+ struct seq_oss_timer *rec;
+
+ rec = kzalloc(sizeof(*rec), GFP_KERNEL);
+ if (rec == NULL)
+ return NULL;
+
+ rec->dp = dp;
+ rec->cur_tick = 0;
+ rec->realtime = 0;
+ rec->running = 0;
+ rec->oss_tempo = 60;
+ rec->oss_timebase = 100;
+ calc_alsa_tempo(rec);
+
+ return rec;
+}
+
+
+/*
+ * delete timer.
+ * if no more timer exists, stop the queue.
+ */
+void
+snd_seq_oss_timer_delete(struct seq_oss_timer *rec)
+{
+ if (rec) {
+ snd_seq_oss_timer_stop(rec);
+ kfree(rec);
+ }
+}
+
+
+/*
+ * process one timing event
+ * return 1 : event proceseed -- skip this event
+ * 0 : not a timer event -- enqueue this event
+ */
+int
+snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev)
+{
+ abstime_t parm = ev->t.time;
+
+ if (ev->t.code == EV_TIMING) {
+ switch (ev->t.cmd) {
+ case TMR_WAIT_REL:
+ parm += rec->cur_tick;
+ rec->realtime = 0;
+ /* continue to next */
+ case TMR_WAIT_ABS:
+ if (parm == 0) {
+ rec->realtime = 1;
+ } else if (parm >= rec->cur_tick) {
+ rec->realtime = 0;
+ rec->cur_tick = parm;
+ }
+ return 1; /* skip this event */
+
+ case TMR_START:
+ snd_seq_oss_timer_start(rec);
+ return 1;
+
+ }
+ } else if (ev->s.code == SEQ_WAIT) {
+ /* time = from 1 to 3 bytes */
+ parm = (ev->echo >> 8) & 0xffffff;
+ if (parm > rec->cur_tick) {
+ /* set next event time */
+ rec->cur_tick = parm;
+ rec->realtime = 0;
+ }
+ return 1;
+ }
+
+ return 0;
+}
+
+
+/*
+ * convert tempo units
+ */
+static void
+calc_alsa_tempo(struct seq_oss_timer *timer)
+{
+ timer->tempo = (60 * 1000000) / timer->oss_tempo;
+ timer->ppq = timer->oss_timebase;
+}
+
+
+/*
+ * dispatch a timer event
+ */
+static int
+send_timer_event(struct seq_oss_devinfo *dp, int type, int value)
+{
+ struct snd_seq_event ev;
+
+ memset(&ev, 0, sizeof(ev));
+ ev.type = type;
+ ev.source.client = dp->cseq;
+ ev.source.port = 0;
+ ev.dest.client = SNDRV_SEQ_CLIENT_SYSTEM;
+ ev.dest.port = SNDRV_SEQ_PORT_SYSTEM_TIMER;
+ ev.queue = dp->queue;
+ ev.data.queue.queue = dp->queue;
+ ev.data.queue.param.value = value;
+ return snd_seq_kernel_client_dispatch(dp->cseq, &ev, 1, 0);
+}
+
+/*
+ * set queue tempo and start queue
+ */
+int
+snd_seq_oss_timer_start(struct seq_oss_timer *timer)
+{
+ struct seq_oss_devinfo *dp = timer->dp;
+ struct snd_seq_queue_tempo tmprec;
+
+ if (timer->running)
+ snd_seq_oss_timer_stop(timer);
+
+ memset(&tmprec, 0, sizeof(tmprec));
+ tmprec.queue = dp->queue;
+ tmprec.ppq = timer->ppq;
+ tmprec.tempo = timer->tempo;
+ snd_seq_set_queue_tempo(dp->cseq, &tmprec);
+
+ send_timer_event(dp, SNDRV_SEQ_EVENT_START, 0);
+ timer->running = 1;
+ timer->cur_tick = 0;
+ return 0;
+}
+
+
+/*
+ * stop queue
+ */
+int
+snd_seq_oss_timer_stop(struct seq_oss_timer *timer)
+{
+ if (! timer->running)
+ return 0;
+ send_timer_event(timer->dp, SNDRV_SEQ_EVENT_STOP, 0);
+ timer->running = 0;
+ return 0;
+}
+
+
+/*
+ * continue queue
+ */
+int
+snd_seq_oss_timer_continue(struct seq_oss_timer *timer)
+{
+ if (timer->running)
+ return 0;
+ send_timer_event(timer->dp, SNDRV_SEQ_EVENT_CONTINUE, 0);
+ timer->running = 1;
+ return 0;
+}
+
+
+/*
+ * change queue tempo
+ */
+int
+snd_seq_oss_timer_tempo(struct seq_oss_timer *timer, int value)
+{
+ if (value < MIN_OSS_TEMPO)
+ value = MIN_OSS_TEMPO;
+ else if (value > MAX_OSS_TEMPO)
+ value = MAX_OSS_TEMPO;
+ timer->oss_tempo = value;
+ calc_alsa_tempo(timer);
+ if (timer->running)
+ send_timer_event(timer->dp, SNDRV_SEQ_EVENT_TEMPO, timer->tempo);
+ return 0;
+}
+
+
+/*
+ * ioctls
+ */
+int
+snd_seq_oss_timer_ioctl(struct seq_oss_timer *timer, unsigned int cmd, int __user *arg)
+{
+ int value;
+
+ if (cmd == SNDCTL_SEQ_CTRLRATE) {
+ debug_printk(("ctrl rate\n"));
+ /* if *arg == 0, just return the current rate */
+ if (get_user(value, arg))
+ return -EFAULT;
+ if (value)
+ return -EINVAL;
+ value = ((timer->oss_tempo * timer->oss_timebase) + 30) / 60;
+ return put_user(value, arg) ? -EFAULT : 0;
+ }
+
+ if (timer->dp->seq_mode == SNDRV_SEQ_OSS_MODE_SYNTH)
+ return 0;
+
+ switch (cmd) {
+ case SNDCTL_TMR_START:
+ debug_printk(("timer start\n"));
+ return snd_seq_oss_timer_start(timer);
+ case SNDCTL_TMR_STOP:
+ debug_printk(("timer stop\n"));
+ return snd_seq_oss_timer_stop(timer);
+ case SNDCTL_TMR_CONTINUE:
+ debug_printk(("timer continue\n"));
+ return snd_seq_oss_timer_continue(timer);
+ case SNDCTL_TMR_TEMPO:
+ debug_printk(("timer tempo\n"));
+ if (get_user(value, arg))
+ return -EFAULT;
+ return snd_seq_oss_timer_tempo(timer, value);
+ case SNDCTL_TMR_TIMEBASE:
+ debug_printk(("timer timebase\n"));
+ if (get_user(value, arg))
+ return -EFAULT;
+ if (value < MIN_OSS_TIMEBASE)
+ value = MIN_OSS_TIMEBASE;
+ else if (value > MAX_OSS_TIMEBASE)
+ value = MAX_OSS_TIMEBASE;
+ timer->oss_timebase = value;
+ calc_alsa_tempo(timer);
+ return 0;
+
+ case SNDCTL_TMR_METRONOME:
+ case SNDCTL_TMR_SELECT:
+ case SNDCTL_TMR_SOURCE:
+ debug_printk(("timer XXX\n"));
+ /* not supported */
+ return 0;
+ }
+ return 0;
+}
diff --git a/sound/core/seq/oss/seq_oss_timer.h b/sound/core/seq/oss/seq_oss_timer.h
new file mode 100644
index 0000000..b995bd6
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_timer.h
@@ -0,0 +1,70 @@
+/*
+ * OSS compatible sequencer driver
+ * timer handling routines
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SEQ_OSS_TIMER_H
+#define __SEQ_OSS_TIMER_H
+
+#include "seq_oss_device.h"
+
+/*
+ * timer information definition
+ */
+struct seq_oss_timer {
+ struct seq_oss_devinfo *dp;
+ reltime_t cur_tick;
+ int realtime;
+ int running;
+ int tempo, ppq; /* ALSA queue */
+ int oss_tempo, oss_timebase;
+};
+
+
+struct seq_oss_timer *snd_seq_oss_timer_new(struct seq_oss_devinfo *dp);
+void snd_seq_oss_timer_delete(struct seq_oss_timer *dp);
+
+int snd_seq_oss_timer_start(struct seq_oss_timer *timer);
+int snd_seq_oss_timer_stop(struct seq_oss_timer *timer);
+int snd_seq_oss_timer_continue(struct seq_oss_timer *timer);
+int snd_seq_oss_timer_tempo(struct seq_oss_timer *timer, int value);
+#define snd_seq_oss_timer_reset snd_seq_oss_timer_start
+
+int snd_seq_oss_timer_ioctl(struct seq_oss_timer *timer, unsigned int cmd, int __user *arg);
+
+/*
+ * get current processed time
+ */
+static inline abstime_t
+snd_seq_oss_timer_cur_tick(struct seq_oss_timer *timer)
+{
+ return timer->cur_tick;
+}
+
+
+/*
+ * is realtime event?
+ */
+static inline int
+snd_seq_oss_timer_is_realtime(struct seq_oss_timer *timer)
+{
+ return timer->realtime;
+}
+
+#endif
diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c
new file mode 100644
index 0000000..2174248
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_writeq.c
@@ -0,0 +1,172 @@
+/*
+ * OSS compatible sequencer driver
+ *
+ * seq_oss_writeq.c - write queue and sync
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "seq_oss_writeq.h"
+#include "seq_oss_event.h"
+#include "seq_oss_timer.h"
+#include <sound/seq_oss_legacy.h>
+#include "../seq_lock.h"
+#include "../seq_clientmgr.h"
+#include <linux/wait.h>
+
+
+/*
+ * create a write queue record
+ */
+struct seq_oss_writeq *
+snd_seq_oss_writeq_new(struct seq_oss_devinfo *dp, int maxlen)
+{
+ struct seq_oss_writeq *q;
+ struct snd_seq_client_pool pool;
+
+ if ((q = kzalloc(sizeof(*q), GFP_KERNEL)) == NULL)
+ return NULL;
+ q->dp = dp;
+ q->maxlen = maxlen;
+ spin_lock_init(&q->sync_lock);
+ q->sync_event_put = 0;
+ q->sync_time = 0;
+ init_waitqueue_head(&q->sync_sleep);
+
+ memset(&pool, 0, sizeof(pool));
+ pool.client = dp->cseq;
+ pool.output_pool = maxlen;
+ pool.output_room = maxlen / 2;
+
+ snd_seq_oss_control(dp, SNDRV_SEQ_IOCTL_SET_CLIENT_POOL, &pool);
+
+ return q;
+}
+
+/*
+ * delete the write queue
+ */
+void
+snd_seq_oss_writeq_delete(struct seq_oss_writeq *q)
+{
+ if (q) {
+ snd_seq_oss_writeq_clear(q); /* to be sure */
+ kfree(q);
+ }
+}
+
+
+/*
+ * reset the write queue
+ */
+void
+snd_seq_oss_writeq_clear(struct seq_oss_writeq *q)
+{
+ struct snd_seq_remove_events reset;
+
+ memset(&reset, 0, sizeof(reset));
+ reset.remove_mode = SNDRV_SEQ_REMOVE_OUTPUT; /* remove all */
+ snd_seq_oss_control(q->dp, SNDRV_SEQ_IOCTL_REMOVE_EVENTS, &reset);
+
+ /* wake up sleepers if any */
+ snd_seq_oss_writeq_wakeup(q, 0);
+}
+
+/*
+ * wait until the write buffer has enough room
+ */
+int
+snd_seq_oss_writeq_sync(struct seq_oss_writeq *q)
+{
+ struct seq_oss_devinfo *dp = q->dp;
+ abstime_t time;
+
+ time = snd_seq_oss_timer_cur_tick(dp->timer);
+ if (q->sync_time >= time)
+ return 0; /* already finished */
+
+ if (! q->sync_event_put) {
+ struct snd_seq_event ev;
+ union evrec *rec;
+
+ /* put echoback event */
+ memset(&ev, 0, sizeof(ev));
+ ev.flags = 0;
+ ev.type = SNDRV_SEQ_EVENT_ECHO;
+ ev.time.tick = time;
+ /* echo back to itself */
+ snd_seq_oss_fill_addr(dp, &ev, dp->addr.client, dp->addr.port);
+ rec = (union evrec *)&ev.data;
+ rec->t.code = SEQ_SYNCTIMER;
+ rec->t.time = time;
+ q->sync_event_put = 1;
+ snd_seq_kernel_client_enqueue_blocking(dp->cseq, &ev, NULL, 0, 0);
+ }
+
+ wait_event_interruptible_timeout(q->sync_sleep, ! q->sync_event_put, HZ);
+ if (signal_pending(current))
+ /* interrupted - return 0 to finish sync */
+ q->sync_event_put = 0;
+ if (! q->sync_event_put || q->sync_time >= time)
+ return 0;
+ return 1;
+}
+
+/*
+ * wake up sync - echo event was catched
+ */
+void
+snd_seq_oss_writeq_wakeup(struct seq_oss_writeq *q, abstime_t time)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&q->sync_lock, flags);
+ q->sync_time = time;
+ q->sync_event_put = 0;
+ if (waitqueue_active(&q->sync_sleep)) {
+ wake_up(&q->sync_sleep);
+ }
+ spin_unlock_irqrestore(&q->sync_lock, flags);
+}
+
+
+/*
+ * return the unused pool size
+ */
+int
+snd_seq_oss_writeq_get_free_size(struct seq_oss_writeq *q)
+{
+ struct snd_seq_client_pool pool;
+ pool.client = q->dp->cseq;
+ snd_seq_oss_control(q->dp, SNDRV_SEQ_IOCTL_GET_CLIENT_POOL, &pool);
+ return pool.output_free;
+}
+
+
+/*
+ * set output threshold size from ioctl
+ */
+void
+snd_seq_oss_writeq_set_output(struct seq_oss_writeq *q, int val)
+{
+ struct snd_seq_client_pool pool;
+ pool.client = q->dp->cseq;
+ snd_seq_oss_control(q->dp, SNDRV_SEQ_IOCTL_GET_CLIENT_POOL, &pool);
+ pool.output_room = val;
+ snd_seq_oss_control(q->dp, SNDRV_SEQ_IOCTL_SET_CLIENT_POOL, &pool);
+}
+
diff --git a/sound/core/seq/oss/seq_oss_writeq.h b/sound/core/seq/oss/seq_oss_writeq.h
new file mode 100644
index 0000000..c469d29
--- /dev/null
+++ b/sound/core/seq/oss/seq_oss_writeq.h
@@ -0,0 +1,50 @@
+/*
+ * OSS compatible sequencer driver
+ * write priority queue
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SEQ_OSS_WRITEQ_H
+#define __SEQ_OSS_WRITEQ_H
+
+#include "seq_oss_device.h"
+
+
+struct seq_oss_writeq {
+ struct seq_oss_devinfo *dp;
+ int maxlen;
+ abstime_t sync_time;
+ int sync_event_put;
+ wait_queue_head_t sync_sleep;
+ spinlock_t sync_lock;
+};
+
+
+/*
+ * seq_oss_writeq.c
+ */
+struct seq_oss_writeq *snd_seq_oss_writeq_new(struct seq_oss_devinfo *dp, int maxlen);
+void snd_seq_oss_writeq_delete(struct seq_oss_writeq *q);
+void snd_seq_oss_writeq_clear(struct seq_oss_writeq *q);
+int snd_seq_oss_writeq_sync(struct seq_oss_writeq *q);
+void snd_seq_oss_writeq_wakeup(struct seq_oss_writeq *q, abstime_t time);
+int snd_seq_oss_writeq_get_free_size(struct seq_oss_writeq *q);
+void snd_seq_oss_writeq_set_output(struct seq_oss_writeq *q, int size);
+
+
+#endif
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
new file mode 100644
index 0000000..ee0f840
--- /dev/null
+++ b/sound/core/seq/seq.c
@@ -0,0 +1,130 @@
+/*
+ * ALSA sequencer main module
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+
+#include <sound/seq_kernel.h>
+#include "seq_clientmgr.h"
+#include "seq_memory.h"
+#include "seq_queue.h"
+#include "seq_lock.h"
+#include "seq_timer.h"
+#include "seq_system.h"
+#include "seq_info.h"
+#include <sound/seq_device.h>
+
+#if defined(CONFIG_SND_SEQ_DUMMY_MODULE)
+int seq_client_load[15] = {[0] = SNDRV_SEQ_CLIENT_DUMMY, [1 ... 14] = -1};
+#else
+int seq_client_load[15] = {[0 ... 14] = -1};
+#endif
+int seq_default_timer_class = SNDRV_TIMER_CLASS_GLOBAL;
+int seq_default_timer_sclass = SNDRV_TIMER_SCLASS_NONE;
+int seq_default_timer_card = -1;
+int seq_default_timer_device =
+#ifdef CONFIG_SND_SEQ_RTCTIMER_DEFAULT
+ SNDRV_TIMER_GLOBAL_RTC
+#else
+ SNDRV_TIMER_GLOBAL_SYSTEM
+#endif
+ ;
+int seq_default_timer_subdevice = 0;
+int seq_default_timer_resolution = 0; /* Hz */
+
+MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer.");
+MODULE_LICENSE("GPL");
+
+module_param_array(seq_client_load, int, NULL, 0444);
+MODULE_PARM_DESC(seq_client_load, "The numbers of global (system) clients to load through kmod.");
+module_param(seq_default_timer_class, int, 0644);
+MODULE_PARM_DESC(seq_default_timer_class, "The default timer class.");
+module_param(seq_default_timer_sclass, int, 0644);
+MODULE_PARM_DESC(seq_default_timer_sclass, "The default timer slave class.");
+module_param(seq_default_timer_card, int, 0644);
+MODULE_PARM_DESC(seq_default_timer_card, "The default timer card number.");
+module_param(seq_default_timer_device, int, 0644);
+MODULE_PARM_DESC(seq_default_timer_device, "The default timer device number.");
+module_param(seq_default_timer_subdevice, int, 0644);
+MODULE_PARM_DESC(seq_default_timer_subdevice, "The default timer subdevice number.");
+module_param(seq_default_timer_resolution, int, 0644);
+MODULE_PARM_DESC(seq_default_timer_resolution, "The default timer resolution in Hz.");
+
+/*
+ * INIT PART
+ */
+
+static int __init alsa_seq_init(void)
+{
+ int err;
+
+ snd_seq_autoload_lock();
+ if ((err = client_init_data()) < 0)
+ goto error;
+
+ /* init memory, room for selected events */
+ if ((err = snd_sequencer_memory_init()) < 0)
+ goto error;
+
+ /* init event queues */
+ if ((err = snd_seq_queues_init()) < 0)
+ goto error;
+
+ /* register sequencer device */
+ if ((err = snd_sequencer_device_init()) < 0)
+ goto error;
+
+ /* register proc interface */
+ if ((err = snd_seq_info_init()) < 0)
+ goto error;
+
+ /* register our internal client */
+ if ((err = snd_seq_system_client_init()) < 0)
+ goto error;
+
+ error:
+ snd_seq_autoload_unlock();
+ return err;
+}
+
+static void __exit alsa_seq_exit(void)
+{
+ /* unregister our internal client */
+ snd_seq_system_client_done();
+
+ /* unregister proc interface */
+ snd_seq_info_done();
+
+ /* delete timing queues */
+ snd_seq_queues_delete();
+
+ /* unregister sequencer device */
+ snd_sequencer_device_done();
+
+ /* release event memory */
+ snd_sequencer_memory_done();
+}
+
+module_init(alsa_seq_init)
+module_exit(alsa_seq_exit)
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
new file mode 100644
index 0000000..8ca2be3
--- /dev/null
+++ b/sound/core/seq/seq_clientmgr.c
@@ -0,0 +1,2587 @@
+/*
+ * ALSA sequencer Client Manager
+ * Copyright (c) 1998-2001 by Frank van de Pol <fvdpol@coil.demon.nl>
+ * Jaroslav Kysela <perex@perex.cz>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <linux/kmod.h>
+
+#include <sound/seq_kernel.h>
+#include "seq_clientmgr.h"
+#include "seq_memory.h"
+#include "seq_queue.h"
+#include "seq_timer.h"
+#include "seq_info.h"
+#include "seq_system.h"
+#include <sound/seq_device.h>
+#ifdef CONFIG_COMPAT
+#include <linux/compat.h>
+#endif
+
+/* Client Manager
+
+ * this module handles the connections of userland and kernel clients
+ *
+ */
+
+/*
+ * There are four ranges of client numbers (last two shared):
+ * 0..15: global clients
+ * 16..127: statically allocated client numbers for cards 0..27
+ * 128..191: dynamically allocated client numbers for cards 28..31
+ * 128..191: dynamically allocated client numbers for applications
+ */
+
+/* number of kernel non-card clients */
+#define SNDRV_SEQ_GLOBAL_CLIENTS 16
+/* clients per cards, for static clients */
+#define SNDRV_SEQ_CLIENTS_PER_CARD 4
+/* dynamically allocated client numbers (both kernel drivers and user space) */
+#define SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN 128
+
+#define SNDRV_SEQ_LFLG_INPUT 0x0001
+#define SNDRV_SEQ_LFLG_OUTPUT 0x0002
+#define SNDRV_SEQ_LFLG_OPEN (SNDRV_SEQ_LFLG_INPUT|SNDRV_SEQ_LFLG_OUTPUT)
+
+static DEFINE_SPINLOCK(clients_lock);
+static DEFINE_MUTEX(register_mutex);
+
+/*
+ * client table
+ */
+static char clienttablock[SNDRV_SEQ_MAX_CLIENTS];
+static struct snd_seq_client *clienttab[SNDRV_SEQ_MAX_CLIENTS];
+static struct snd_seq_usage client_usage;
+
+/*
+ * prototypes
+ */
+static int bounce_error_event(struct snd_seq_client *client,
+ struct snd_seq_event *event,
+ int err, int atomic, int hop);
+static int snd_seq_deliver_single_event(struct snd_seq_client *client,
+ struct snd_seq_event *event,
+ int filter, int atomic, int hop);
+
+/*
+ */
+
+static inline mm_segment_t snd_enter_user(void)
+{
+ mm_segment_t fs = get_fs();
+ set_fs(get_ds());
+ return fs;
+}
+
+static inline void snd_leave_user(mm_segment_t fs)
+{
+ set_fs(fs);
+}
+
+/*
+ */
+static inline unsigned short snd_seq_file_flags(struct file *file)
+{
+ switch (file->f_mode & (FMODE_READ | FMODE_WRITE)) {
+ case FMODE_WRITE:
+ return SNDRV_SEQ_LFLG_OUTPUT;
+ case FMODE_READ:
+ return SNDRV_SEQ_LFLG_INPUT;
+ default:
+ return SNDRV_SEQ_LFLG_OPEN;
+ }
+}
+
+static inline int snd_seq_write_pool_allocated(struct snd_seq_client *client)
+{
+ return snd_seq_total_cells(client->pool) > 0;
+}
+
+/* return pointer to client structure for specified id */
+static struct snd_seq_client *clientptr(int clientid)
+{
+ if (clientid < 0 || clientid >= SNDRV_SEQ_MAX_CLIENTS) {
+ snd_printd("Seq: oops. Trying to get pointer to client %d\n",
+ clientid);
+ return NULL;
+ }
+ return clienttab[clientid];
+}
+
+struct snd_seq_client *snd_seq_client_use_ptr(int clientid)
+{
+ unsigned long flags;
+ struct snd_seq_client *client;
+
+ if (clientid < 0 || clientid >= SNDRV_SEQ_MAX_CLIENTS) {
+ snd_printd("Seq: oops. Trying to get pointer to client %d\n",
+ clientid);
+ return NULL;
+ }
+ spin_lock_irqsave(&clients_lock, flags);
+ client = clientptr(clientid);
+ if (client)
+ goto __lock;
+ if (clienttablock[clientid]) {
+ spin_unlock_irqrestore(&clients_lock, flags);
+ return NULL;
+ }
+ spin_unlock_irqrestore(&clients_lock, flags);
+#ifdef CONFIG_MODULES
+ if (!in_interrupt()) {
+ static char client_requested[SNDRV_SEQ_GLOBAL_CLIENTS];
+ static char card_requested[SNDRV_CARDS];
+ if (clientid < SNDRV_SEQ_GLOBAL_CLIENTS) {
+ int idx;
+
+ if (!client_requested[clientid]) {
+ client_requested[clientid] = 1;
+ for (idx = 0; idx < 15; idx++) {
+ if (seq_client_load[idx] < 0)
+ break;
+ if (seq_client_load[idx] == clientid) {
+ request_module("snd-seq-client-%i",
+ clientid);
+ break;
+ }
+ }
+ }
+ } else if (clientid < SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN) {
+ int card = (clientid - SNDRV_SEQ_GLOBAL_CLIENTS) /
+ SNDRV_SEQ_CLIENTS_PER_CARD;
+ if (card < snd_ecards_limit) {
+ if (! card_requested[card]) {
+ card_requested[card] = 1;
+ snd_request_card(card);
+ }
+ snd_seq_device_load_drivers();
+ }
+ }
+ spin_lock_irqsave(&clients_lock, flags);
+ client = clientptr(clientid);
+ if (client)
+ goto __lock;
+ spin_unlock_irqrestore(&clients_lock, flags);
+ }
+#endif
+ return NULL;
+
+ __lock:
+ snd_use_lock_use(&client->use_lock);
+ spin_unlock_irqrestore(&clients_lock, flags);
+ return client;
+}
+
+static void usage_alloc(struct snd_seq_usage *res, int num)
+{
+ res->cur += num;
+ if (res->cur > res->peak)
+ res->peak = res->cur;
+}
+
+static void usage_free(struct snd_seq_usage *res, int num)
+{
+ res->cur -= num;
+}
+
+/* initialise data structures */
+int __init client_init_data(void)
+{
+ /* zap out the client table */
+ memset(&clienttablock, 0, sizeof(clienttablock));
+ memset(&clienttab, 0, sizeof(clienttab));
+ return 0;
+}
+
+
+static struct snd_seq_client *seq_create_client1(int client_index, int poolsize)
+{
+ unsigned long flags;
+ int c;
+ struct snd_seq_client *client;
+
+ /* init client data */
+ client = kzalloc(sizeof(*client), GFP_KERNEL);
+ if (client == NULL)
+ return NULL;
+ client->pool = snd_seq_pool_new(poolsize);
+ if (client->pool == NULL) {
+ kfree(client);
+ return NULL;
+ }
+ client->type = NO_CLIENT;
+ snd_use_lock_init(&client->use_lock);
+ rwlock_init(&client->ports_lock);
+ mutex_init(&client->ports_mutex);
+ INIT_LIST_HEAD(&client->ports_list_head);
+
+ /* find free slot in the client table */
+ spin_lock_irqsave(&clients_lock, flags);
+ if (client_index < 0) {
+ for (c = SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN;
+ c < SNDRV_SEQ_MAX_CLIENTS;
+ c++) {
+ if (clienttab[c] || clienttablock[c])
+ continue;
+ clienttab[client->number = c] = client;
+ spin_unlock_irqrestore(&clients_lock, flags);
+ return client;
+ }
+ } else {
+ if (clienttab[client_index] == NULL && !clienttablock[client_index]) {
+ clienttab[client->number = client_index] = client;
+ spin_unlock_irqrestore(&clients_lock, flags);
+ return client;
+ }
+ }
+ spin_unlock_irqrestore(&clients_lock, flags);
+ snd_seq_pool_delete(&client->pool);
+ kfree(client);
+ return NULL; /* no free slot found or busy, return failure code */
+}
+
+
+static int seq_free_client1(struct snd_seq_client *client)
+{
+ unsigned long flags;
+
+ if (!client)
+ return 0;
+ snd_seq_delete_all_ports(client);
+ snd_seq_queue_client_leave(client->number);
+ spin_lock_irqsave(&clients_lock, flags);
+ clienttablock[client->number] = 1;
+ clienttab[client->number] = NULL;
+ spin_unlock_irqrestore(&clients_lock, flags);
+ snd_use_lock_sync(&client->use_lock);
+ snd_seq_queue_client_termination(client->number);
+ if (client->pool)
+ snd_seq_pool_delete(&client->pool);
+ spin_lock_irqsave(&clients_lock, flags);
+ clienttablock[client->number] = 0;
+ spin_unlock_irqrestore(&clients_lock, flags);
+ return 0;
+}
+
+
+static void seq_free_client(struct snd_seq_client * client)
+{
+ mutex_lock(&register_mutex);
+ switch (client->type) {
+ case NO_CLIENT:
+ snd_printk(KERN_WARNING "Seq: Trying to free unused client %d\n",
+ client->number);
+ break;
+ case USER_CLIENT:
+ case KERNEL_CLIENT:
+ seq_free_client1(client);
+ usage_free(&client_usage, 1);
+ break;
+
+ default:
+ snd_printk(KERN_ERR "Seq: Trying to free client %d with undefined type = %d\n",
+ client->number, client->type);
+ }
+ mutex_unlock(&register_mutex);
+
+ snd_seq_system_client_ev_client_exit(client->number);
+}
+
+
+
+/* -------------------------------------------------------- */
+
+/* create a user client */
+static int snd_seq_open(struct inode *inode, struct file *file)
+{
+ int c, mode; /* client id */
+ struct snd_seq_client *client;
+ struct snd_seq_user_client *user;
+
+ if (mutex_lock_interruptible(&register_mutex))
+ return -ERESTARTSYS;
+ client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS);
+ if (client == NULL) {
+ mutex_unlock(&register_mutex);
+ return -ENOMEM; /* failure code */
+ }
+
+ mode = snd_seq_file_flags(file);
+ if (mode & SNDRV_SEQ_LFLG_INPUT)
+ client->accept_input = 1;
+ if (mode & SNDRV_SEQ_LFLG_OUTPUT)
+ client->accept_output = 1;
+
+ user = &client->data.user;
+ user->fifo = NULL;
+ user->fifo_pool_size = 0;
+
+ if (mode & SNDRV_SEQ_LFLG_INPUT) {
+ user->fifo_pool_size = SNDRV_SEQ_DEFAULT_CLIENT_EVENTS;
+ user->fifo = snd_seq_fifo_new(user->fifo_pool_size);
+ if (user->fifo == NULL) {
+ seq_free_client1(client);
+ kfree(client);
+ mutex_unlock(&register_mutex);
+ return -ENOMEM;
+ }
+ }
+
+ usage_alloc(&client_usage, 1);
+ client->type = USER_CLIENT;
+ mutex_unlock(&register_mutex);
+
+ c = client->number;
+ file->private_data = client;
+
+ /* fill client data */
+ user->file = file;
+ sprintf(client->name, "Client-%d", c);
+
+ /* make others aware this new client */
+ snd_seq_system_client_ev_client_start(c);
+
+ return 0;
+}
+
+/* delete a user client */
+static int snd_seq_release(struct inode *inode, struct file *file)
+{
+ struct snd_seq_client *client = file->private_data;
+
+ if (client) {
+ seq_free_client(client);
+ if (client->data.user.fifo)
+ snd_seq_fifo_delete(&client->data.user.fifo);
+ kfree(client);
+ }
+
+ return 0;
+}
+
+
+/* handle client read() */
+/* possible error values:
+ * -ENXIO invalid client or file open mode
+ * -ENOSPC FIFO overflow (the flag is cleared after this error report)
+ * -EINVAL no enough user-space buffer to write the whole event
+ * -EFAULT seg. fault during copy to user space
+ */
+static ssize_t snd_seq_read(struct file *file, char __user *buf, size_t count,
+ loff_t *offset)
+{
+ struct snd_seq_client *client = file->private_data;
+ struct snd_seq_fifo *fifo;
+ int err;
+ long result = 0;
+ struct snd_seq_event_cell *cell;
+
+ if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT))
+ return -ENXIO;
+
+ if (!access_ok(VERIFY_WRITE, buf, count))
+ return -EFAULT;
+
+ /* check client structures are in place */
+ if (snd_BUG_ON(!client))
+ return -ENXIO;
+
+ if (!client->accept_input || (fifo = client->data.user.fifo) == NULL)
+ return -ENXIO;
+
+ if (atomic_read(&fifo->overflow) > 0) {
+ /* buffer overflow is detected */
+ snd_seq_fifo_clear(fifo);
+ /* return error code */
+ return -ENOSPC;
+ }
+
+ cell = NULL;
+ err = 0;
+ snd_seq_fifo_lock(fifo);
+
+ /* while data available in queue */
+ while (count >= sizeof(struct snd_seq_event)) {
+ int nonblock;
+
+ nonblock = (file->f_flags & O_NONBLOCK) || result > 0;
+ if ((err = snd_seq_fifo_cell_out(fifo, &cell, nonblock)) < 0) {
+ break;
+ }
+ if (snd_seq_ev_is_variable(&cell->event)) {
+ struct snd_seq_event tmpev;
+ tmpev = cell->event;
+ tmpev.data.ext.len &= ~SNDRV_SEQ_EXT_MASK;
+ if (copy_to_user(buf, &tmpev, sizeof(struct snd_seq_event))) {
+ err = -EFAULT;
+ break;
+ }
+ count -= sizeof(struct snd_seq_event);
+ buf += sizeof(struct snd_seq_event);
+ err = snd_seq_expand_var_event(&cell->event, count,
+ (char __force *)buf, 0,
+ sizeof(struct snd_seq_event));
+ if (err < 0)
+ break;
+ result += err;
+ count -= err;
+ buf += err;
+ } else {
+ if (copy_to_user(buf, &cell->event, sizeof(struct snd_seq_event))) {
+ err = -EFAULT;
+ break;
+ }
+ count -= sizeof(struct snd_seq_event);
+ buf += sizeof(struct snd_seq_event);
+ }
+ snd_seq_cell_free(cell);
+ cell = NULL; /* to be sure */
+ result += sizeof(struct snd_seq_event);
+ }
+
+ if (err < 0) {
+ if (cell)
+ snd_seq_fifo_cell_putback(fifo, cell);
+ if (err == -EAGAIN && result > 0)
+ err = 0;
+ }
+ snd_seq_fifo_unlock(fifo);
+
+ return (err < 0) ? err : result;
+}
+
+
+/*
+ * check access permission to the port
+ */
+static int check_port_perm(struct snd_seq_client_port *port, unsigned int flags)
+{
+ if ((port->capability & flags) != flags)
+ return 0;
+ return flags;
+}
+
+/*
+ * check if the destination client is available, and return the pointer
+ * if filter is non-zero, client filter bitmap is tested.
+ */
+static struct snd_seq_client *get_event_dest_client(struct snd_seq_event *event,
+ int filter)
+{
+ struct snd_seq_client *dest;
+
+ dest = snd_seq_client_use_ptr(event->dest.client);
+ if (dest == NULL)
+ return NULL;
+ if (! dest->accept_input)
+ goto __not_avail;
+ if ((dest->filter & SNDRV_SEQ_FILTER_USE_EVENT) &&
+ ! test_bit(event->type, dest->event_filter))
+ goto __not_avail;
+ if (filter && !(dest->filter & filter))
+ goto __not_avail;
+
+ return dest; /* ok - accessible */
+__not_avail:
+ snd_seq_client_unlock(dest);
+ return NULL;
+}
+
+
+/*
+ * Return the error event.
+ *
+ * If the receiver client is a user client, the original event is
+ * encapsulated in SNDRV_SEQ_EVENT_BOUNCE as variable length event. If
+ * the original event is also variable length, the external data is
+ * copied after the event record.
+ * If the receiver client is a kernel client, the original event is
+ * quoted in SNDRV_SEQ_EVENT_KERNEL_ERROR, since this requires no extra
+ * kmalloc.
+ */
+static int bounce_error_event(struct snd_seq_client *client,
+ struct snd_seq_event *event,
+ int err, int atomic, int hop)
+{
+ struct snd_seq_event bounce_ev;
+ int result;
+
+ if (client == NULL ||
+ ! (client->filter & SNDRV_SEQ_FILTER_BOUNCE) ||
+ ! client->accept_input)
+ return 0; /* ignored */
+
+ /* set up quoted error */
+ memset(&bounce_ev, 0, sizeof(bounce_ev));
+ bounce_ev.type = SNDRV_SEQ_EVENT_KERNEL_ERROR;
+ bounce_ev.flags = SNDRV_SEQ_EVENT_LENGTH_FIXED;
+ bounce_ev.queue = SNDRV_SEQ_QUEUE_DIRECT;
+ bounce_ev.source.client = SNDRV_SEQ_CLIENT_SYSTEM;
+ bounce_ev.source.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE;
+ bounce_ev.dest.client = client->number;
+ bounce_ev.dest.port = event->source.port;
+ bounce_ev.data.quote.origin = event->dest;
+ bounce_ev.data.quote.event = event;
+ bounce_ev.data.quote.value = -err; /* use positive value */
+ result = snd_seq_deliver_single_event(NULL, &bounce_ev, 0, atomic, hop + 1);
+ if (result < 0) {
+ client->event_lost++;
+ return result;
+ }
+
+ return result;
+}
+
+
+/*
+ * rewrite the time-stamp of the event record with the curren time
+ * of the given queue.
+ * return non-zero if updated.
+ */
+static int update_timestamp_of_queue(struct snd_seq_event *event,
+ int queue, int real_time)
+{
+ struct snd_seq_queue *q;
+
+ q = queueptr(queue);
+ if (! q)
+ return 0;
+ event->queue = queue;
+ event->flags &= ~SNDRV_SEQ_TIME_STAMP_MASK;
+ if (real_time) {
+ event->time.time = snd_seq_timer_get_cur_time(q->timer);
+ event->flags |= SNDRV_SEQ_TIME_STAMP_REAL;
+ } else {
+ event->time.tick = snd_seq_timer_get_cur_tick(q->timer);
+ event->flags |= SNDRV_SEQ_TIME_STAMP_TICK;
+ }
+ queuefree(q);
+ return 1;
+}
+
+
+/*
+ * deliver an event to the specified destination.
+ * if filter is non-zero, client filter bitmap is tested.
+ *
+ * RETURN VALUE: 0 : if succeeded
+ * <0 : error
+ */
+static int snd_seq_deliver_single_event(struct snd_seq_client *client,
+ struct snd_seq_event *event,
+ int filter, int atomic, int hop)
+{
+ struct snd_seq_client *dest = NULL;
+ struct snd_seq_client_port *dest_port = NULL;
+ int result = -ENOENT;
+ int direct;
+
+ direct = snd_seq_ev_is_direct(event);
+
+ dest = get_event_dest_client(event, filter);
+ if (dest == NULL)
+ goto __skip;
+ dest_port = snd_seq_port_use_ptr(dest, event->dest.port);
+ if (dest_port == NULL)
+ goto __skip;
+
+ /* check permission */
+ if (! check_port_perm(dest_port, SNDRV_SEQ_PORT_CAP_WRITE)) {
+ result = -EPERM;
+ goto __skip;
+ }
+
+ if (dest_port->timestamping)
+ update_timestamp_of_queue(event, dest_port->time_queue,
+ dest_port->time_real);
+
+ switch (dest->type) {
+ case USER_CLIENT:
+ if (dest->data.user.fifo)
+ result = snd_seq_fifo_event_in(dest->data.user.fifo, event);
+ break;
+
+ case KERNEL_CLIENT:
+ if (dest_port->event_input == NULL)
+ break;
+ result = dest_port->event_input(event, direct,
+ dest_port->private_data,
+ atomic, hop);
+ break;
+ default:
+ break;
+ }
+
+ __skip:
+ if (dest_port)
+ snd_seq_port_unlock(dest_port);
+ if (dest)
+ snd_seq_client_unlock(dest);
+
+ if (result < 0 && !direct) {
+ result = bounce_error_event(client, event, result, atomic, hop);
+ }
+ return result;
+}
+
+
+/*
+ * send the event to all subscribers:
+ */
+static int deliver_to_subscribers(struct snd_seq_client *client,
+ struct snd_seq_event *event,
+ int atomic, int hop)
+{
+ struct snd_seq_subscribers *subs;
+ int err = 0, num_ev = 0;
+ struct snd_seq_event event_saved;
+ struct snd_seq_client_port *src_port;
+ struct snd_seq_port_subs_info *grp;
+
+ src_port = snd_seq_port_use_ptr(client, event->source.port);
+ if (src_port == NULL)
+ return -EINVAL; /* invalid source port */
+ /* save original event record */
+ event_saved = *event;
+ grp = &src_port->c_src;
+
+ /* lock list */
+ if (atomic)
+ read_lock(&grp->list_lock);
+ else
+ down_read(&grp->list_mutex);
+ list_for_each_entry(subs, &grp->list_head, src_list) {
+ event->dest = subs->info.dest;
+ if (subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIMESTAMP)
+ /* convert time according to flag with subscription */
+ update_timestamp_of_queue(event, subs->info.queue,
+ subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL);
+ err = snd_seq_deliver_single_event(client, event,
+ 0, atomic, hop);
+ if (err < 0)
+ break;
+ num_ev++;
+ /* restore original event record */
+ *event = event_saved;
+ }
+ if (atomic)
+ read_unlock(&grp->list_lock);
+ else
+ up_read(&grp->list_mutex);
+ *event = event_saved; /* restore */
+ snd_seq_port_unlock(src_port);
+ return (err < 0) ? err : num_ev;
+}
+
+
+#ifdef SUPPORT_BROADCAST
+/*
+ * broadcast to all ports:
+ */
+static int port_broadcast_event(struct snd_seq_client *client,
+ struct snd_seq_event *event,
+ int atomic, int hop)
+{
+ int num_ev = 0, err = 0;
+ struct snd_seq_client *dest_client;
+ struct snd_seq_client_port *port;
+
+ dest_client = get_event_dest_client(event, SNDRV_SEQ_FILTER_BROADCAST);
+ if (dest_client == NULL)
+ return 0; /* no matching destination */
+
+ read_lock(&dest_client->ports_lock);
+ list_for_each_entry(port, &dest_client->ports_list_head, list) {
+ event->dest.port = port->addr.port;
+ /* pass NULL as source client to avoid error bounce */
+ err = snd_seq_deliver_single_event(NULL, event,
+ SNDRV_SEQ_FILTER_BROADCAST,
+ atomic, hop);
+ if (err < 0)
+ break;
+ num_ev++;
+ }
+ read_unlock(&dest_client->ports_lock);
+ snd_seq_client_unlock(dest_client);
+ event->dest.port = SNDRV_SEQ_ADDRESS_BROADCAST; /* restore */
+ return (err < 0) ? err : num_ev;
+}
+
+/*
+ * send the event to all clients:
+ * if destination port is also ADDRESS_BROADCAST, deliver to all ports.
+ */
+static int broadcast_event(struct snd_seq_client *client,
+ struct snd_seq_event *event, int atomic, int hop)
+{
+ int err = 0, num_ev = 0;
+ int dest;
+ struct snd_seq_addr addr;
+
+ addr = event->dest; /* save */
+
+ for (dest = 0; dest < SNDRV_SEQ_MAX_CLIENTS; dest++) {
+ /* don't send to itself */
+ if (dest == client->number)
+ continue;
+ event->dest.client = dest;
+ event->dest.port = addr.port;
+ if (addr.port == SNDRV_SEQ_ADDRESS_BROADCAST)
+ err = port_broadcast_event(client, event, atomic, hop);
+ else
+ /* pass NULL as source client to avoid error bounce */
+ err = snd_seq_deliver_single_event(NULL, event,
+ SNDRV_SEQ_FILTER_BROADCAST,
+ atomic, hop);
+ if (err < 0)
+ break;
+ num_ev += err;
+ }
+ event->dest = addr; /* restore */
+ return (err < 0) ? err : num_ev;
+}
+
+
+/* multicast - not supported yet */
+static int multicast_event(struct snd_seq_client *client, struct snd_seq_event *event,
+ int atomic, int hop)
+{
+ snd_printd("seq: multicast not supported yet.\n");
+ return 0; /* ignored */
+}
+#endif /* SUPPORT_BROADCAST */
+
+
+/* deliver an event to the destination port(s).
+ * if the event is to subscribers or broadcast, the event is dispatched
+ * to multiple targets.
+ *
+ * RETURN VALUE: n > 0 : the number of delivered events.
+ * n == 0 : the event was not passed to any client.
+ * n < 0 : error - event was not processed.
+ */
+static int snd_seq_deliver_event(struct snd_seq_client *client, struct snd_seq_event *event,
+ int atomic, int hop)
+{
+ int result;
+
+ hop++;
+ if (hop >= SNDRV_SEQ_MAX_HOPS) {
+ snd_printd("too long delivery path (%d:%d->%d:%d)\n",
+ event->source.client, event->source.port,
+ event->dest.client, event->dest.port);
+ return -EMLINK;
+ }
+
+ if (event->queue == SNDRV_SEQ_ADDRESS_SUBSCRIBERS ||
+ event->dest.client == SNDRV_SEQ_ADDRESS_SUBSCRIBERS)
+ result = deliver_to_subscribers(client, event, atomic, hop);
+#ifdef SUPPORT_BROADCAST
+ else if (event->queue == SNDRV_SEQ_ADDRESS_BROADCAST ||
+ event->dest.client == SNDRV_SEQ_ADDRESS_BROADCAST)
+ result = broadcast_event(client, event, atomic, hop);
+ else if (event->dest.client >= SNDRV_SEQ_MAX_CLIENTS)
+ result = multicast_event(client, event, atomic, hop);
+ else if (event->dest.port == SNDRV_SEQ_ADDRESS_BROADCAST)
+ result = port_broadcast_event(client, event, atomic, hop);
+#endif
+ else
+ result = snd_seq_deliver_single_event(client, event, 0, atomic, hop);
+
+ return result;
+}
+
+/*
+ * dispatch an event cell:
+ * This function is called only from queue check routines in timer
+ * interrupts or after enqueued.
+ * The event cell shall be released or re-queued in this function.
+ *
+ * RETURN VALUE: n > 0 : the number of delivered events.
+ * n == 0 : the event was not passed to any client.
+ * n < 0 : error - event was not processed.
+ */
+int snd_seq_dispatch_event(struct snd_seq_event_cell *cell, int atomic, int hop)
+{
+ struct snd_seq_client *client;
+ int result;
+
+ if (snd_BUG_ON(!cell))
+ return -EINVAL;
+
+ client = snd_seq_client_use_ptr(cell->event.source.client);
+ if (client == NULL) {
+ snd_seq_cell_free(cell); /* release this cell */
+ return -EINVAL;
+ }
+
+ if (cell->event.type == SNDRV_SEQ_EVENT_NOTE) {
+ /* NOTE event:
+ * the event cell is re-used as a NOTE-OFF event and
+ * enqueued again.
+ */
+ struct snd_seq_event tmpev, *ev;
+
+ /* reserve this event to enqueue note-off later */
+ tmpev = cell->event;
+ tmpev.type = SNDRV_SEQ_EVENT_NOTEON;
+ result = snd_seq_deliver_event(client, &tmpev, atomic, hop);
+
+ /*
+ * This was originally a note event. We now re-use the
+ * cell for the note-off event.
+ */
+
+ ev = &cell->event;
+ ev->type = SNDRV_SEQ_EVENT_NOTEOFF;
+ ev->flags |= SNDRV_SEQ_PRIORITY_HIGH;
+
+ /* add the duration time */
+ switch (ev->flags & SNDRV_SEQ_TIME_STAMP_MASK) {
+ case SNDRV_SEQ_TIME_STAMP_TICK:
+ ev->time.tick += ev->data.note.duration;
+ break;
+ case SNDRV_SEQ_TIME_STAMP_REAL:
+ /* unit for duration is ms */
+ ev->time.time.tv_nsec += 1000000 * (ev->data.note.duration % 1000);
+ ev->time.time.tv_sec += ev->data.note.duration / 1000 +
+ ev->time.time.tv_nsec / 1000000000;
+ ev->time.time.tv_nsec %= 1000000000;
+ break;
+ }
+ ev->data.note.velocity = ev->data.note.off_velocity;
+
+ /* Now queue this cell as the note off event */
+ if (snd_seq_enqueue_event(cell, atomic, hop) < 0)
+ snd_seq_cell_free(cell); /* release this cell */
+
+ } else {
+ /* Normal events:
+ * event cell is freed after processing the event
+ */
+
+ result = snd_seq_deliver_event(client, &cell->event, atomic, hop);
+ snd_seq_cell_free(cell);
+ }
+
+ snd_seq_client_unlock(client);
+ return result;
+}
+
+
+/* Allocate a cell from client pool and enqueue it to queue:
+ * if pool is empty and blocking is TRUE, sleep until a new cell is
+ * available.
+ */
+static int snd_seq_client_enqueue_event(struct snd_seq_client *client,
+ struct snd_seq_event *event,
+ struct file *file, int blocking,
+ int atomic, int hop)
+{
+ struct snd_seq_event_cell *cell;
+ int err;
+
+ /* special queue values - force direct passing */
+ if (event->queue == SNDRV_SEQ_ADDRESS_SUBSCRIBERS) {
+ event->dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+ event->queue = SNDRV_SEQ_QUEUE_DIRECT;
+ } else
+#ifdef SUPPORT_BROADCAST
+ if (event->queue == SNDRV_SEQ_ADDRESS_BROADCAST) {
+ event->dest.client = SNDRV_SEQ_ADDRESS_BROADCAST;
+ event->queue = SNDRV_SEQ_QUEUE_DIRECT;
+ }
+#endif
+ if (event->dest.client == SNDRV_SEQ_ADDRESS_SUBSCRIBERS) {
+ /* check presence of source port */
+ struct snd_seq_client_port *src_port = snd_seq_port_use_ptr(client, event->source.port);
+ if (src_port == NULL)
+ return -EINVAL;
+ snd_seq_port_unlock(src_port);
+ }
+
+ /* direct event processing without enqueued */
+ if (snd_seq_ev_is_direct(event)) {
+ if (event->type == SNDRV_SEQ_EVENT_NOTE)
+ return -EINVAL; /* this event must be enqueued! */
+ return snd_seq_deliver_event(client, event, atomic, hop);
+ }
+
+ /* Not direct, normal queuing */
+ if (snd_seq_queue_is_used(event->queue, client->number) <= 0)
+ return -EINVAL; /* invalid queue */
+ if (! snd_seq_write_pool_allocated(client))
+ return -ENXIO; /* queue is not allocated */
+
+ /* allocate an event cell */
+ err = snd_seq_event_dup(client->pool, event, &cell, !blocking || atomic, file);
+ if (err < 0)
+ return err;
+
+ /* we got a cell. enqueue it. */
+ if ((err = snd_seq_enqueue_event(cell, atomic, hop)) < 0) {
+ snd_seq_cell_free(cell);
+ return err;
+ }
+
+ return 0;
+}
+
+
+/*
+ * check validity of event type and data length.
+ * return non-zero if invalid.
+ */
+static int check_event_type_and_length(struct snd_seq_event *ev)
+{
+ switch (snd_seq_ev_length_type(ev)) {
+ case SNDRV_SEQ_EVENT_LENGTH_FIXED:
+ if (snd_seq_ev_is_variable_type(ev))
+ return -EINVAL;
+ break;
+ case SNDRV_SEQ_EVENT_LENGTH_VARIABLE:
+ if (! snd_seq_ev_is_variable_type(ev) ||
+ (ev->data.ext.len & ~SNDRV_SEQ_EXT_MASK) >= SNDRV_SEQ_MAX_EVENT_LEN)
+ return -EINVAL;
+ break;
+ case SNDRV_SEQ_EVENT_LENGTH_VARUSR:
+ if (! snd_seq_ev_is_direct(ev))
+ return -EINVAL;
+ break;
+ }
+ return 0;
+}
+
+
+/* handle write() */
+/* possible error values:
+ * -ENXIO invalid client or file open mode
+ * -ENOMEM malloc failed
+ * -EFAULT seg. fault during copy from user space
+ * -EINVAL invalid event
+ * -EAGAIN no space in output pool
+ * -EINTR interrupts while sleep
+ * -EMLINK too many hops
+ * others depends on return value from driver callback
+ */
+static ssize_t snd_seq_write(struct file *file, const char __user *buf,
+ size_t count, loff_t *offset)
+{
+ struct snd_seq_client *client = file->private_data;
+ int written = 0, len;
+ int err = -EINVAL;
+ struct snd_seq_event event;
+
+ if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT))
+ return -ENXIO;
+
+ /* check client structures are in place */
+ if (snd_BUG_ON(!client))
+ return -ENXIO;
+
+ if (!client->accept_output || client->pool == NULL)
+ return -ENXIO;
+
+ /* allocate the pool now if the pool is not allocated yet */
+ if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) {
+ if (snd_seq_pool_init(client->pool) < 0)
+ return -ENOMEM;
+ }
+
+ /* only process whole events */
+ while (count >= sizeof(struct snd_seq_event)) {
+ /* Read in the event header from the user */
+ len = sizeof(event);
+ if (copy_from_user(&event, buf, len)) {
+ err = -EFAULT;
+ break;
+ }
+ event.source.client = client->number; /* fill in client number */
+ /* Check for extension data length */
+ if (check_event_type_and_length(&event)) {
+ err = -EINVAL;
+ break;
+ }
+
+ /* check for special events */
+ if (event.type == SNDRV_SEQ_EVENT_NONE)
+ goto __skip_event;
+ else if (snd_seq_ev_is_reserved(&event)) {
+ err = -EINVAL;
+ break;
+ }
+
+ if (snd_seq_ev_is_variable(&event)) {
+ int extlen = event.data.ext.len & ~SNDRV_SEQ_EXT_MASK;
+ if ((size_t)(extlen + len) > count) {
+ /* back out, will get an error this time or next */
+ err = -EINVAL;
+ break;
+ }
+ /* set user space pointer */
+ event.data.ext.len = extlen | SNDRV_SEQ_EXT_USRPTR;
+ event.data.ext.ptr = (char __force *)buf
+ + sizeof(struct snd_seq_event);
+ len += extlen; /* increment data length */
+ } else {
+#ifdef CONFIG_COMPAT
+ if (client->convert32 && snd_seq_ev_is_varusr(&event)) {
+ void *ptr = compat_ptr(event.data.raw32.d[1]);
+ event.data.ext.ptr = ptr;
+ }
+#endif
+ }
+
+ /* ok, enqueue it */
+ err = snd_seq_client_enqueue_event(client, &event, file,
+ !(file->f_flags & O_NONBLOCK),
+ 0, 0);
+ if (err < 0)
+ break;
+
+ __skip_event:
+ /* Update pointers and counts */
+ count -= len;
+ buf += len;
+ written += len;
+ }
+
+ return written ? written : err;
+}
+
+
+/*
+ * handle polling
+ */
+static unsigned int snd_seq_poll(struct file *file, poll_table * wait)
+{
+ struct snd_seq_client *client = file->private_data;
+ unsigned int mask = 0;
+
+ /* check client structures are in place */
+ if (snd_BUG_ON(!client))
+ return -ENXIO;
+
+ if ((snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT) &&
+ client->data.user.fifo) {
+
+ /* check if data is available in the outqueue */
+ if (snd_seq_fifo_poll_wait(client->data.user.fifo, file, wait))
+ mask |= POLLIN | POLLRDNORM;
+ }
+
+ if (snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT) {
+
+ /* check if data is available in the pool */
+ if (!snd_seq_write_pool_allocated(client) ||
+ snd_seq_pool_poll_wait(client->pool, file, wait))
+ mask |= POLLOUT | POLLWRNORM;
+ }
+
+ return mask;
+}
+
+
+/*-----------------------------------------------------*/
+
+
+/* SYSTEM_INFO ioctl() */
+static int snd_seq_ioctl_system_info(struct snd_seq_client *client, void __user *arg)
+{
+ struct snd_seq_system_info info;
+
+ memset(&info, 0, sizeof(info));
+ /* fill the info fields */
+ info.queues = SNDRV_SEQ_MAX_QUEUES;
+ info.clients = SNDRV_SEQ_MAX_CLIENTS;
+ info.ports = 256; /* fixed limit */
+ info.channels = 256; /* fixed limit */
+ info.cur_clients = client_usage.cur;
+ info.cur_queues = snd_seq_queue_get_cur_queues();
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+
+/* RUNNING_MODE ioctl() */
+static int snd_seq_ioctl_running_mode(struct snd_seq_client *client, void __user *arg)
+{
+ struct snd_seq_running_info info;
+ struct snd_seq_client *cptr;
+ int err = 0;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ /* requested client number */
+ cptr = snd_seq_client_use_ptr(info.client);
+ if (cptr == NULL)
+ return -ENOENT; /* don't change !!! */
+
+#ifdef SNDRV_BIG_ENDIAN
+ if (! info.big_endian) {
+ err = -EINVAL;
+ goto __err;
+ }
+#else
+ if (info.big_endian) {
+ err = -EINVAL;
+ goto __err;
+ }
+
+#endif
+ if (info.cpu_mode > sizeof(long)) {
+ err = -EINVAL;
+ goto __err;
+ }
+ cptr->convert32 = (info.cpu_mode < sizeof(long));
+ __err:
+ snd_seq_client_unlock(cptr);
+ return err;
+}
+
+/* CLIENT_INFO ioctl() */
+static void get_client_info(struct snd_seq_client *cptr,
+ struct snd_seq_client_info *info)
+{
+ info->client = cptr->number;
+
+ /* fill the info fields */
+ info->type = cptr->type;
+ strcpy(info->name, cptr->name);
+ info->filter = cptr->filter;
+ info->event_lost = cptr->event_lost;
+ memcpy(info->event_filter, cptr->event_filter, 32);
+ info->num_ports = cptr->num_ports;
+ memset(info->reserved, 0, sizeof(info->reserved));
+}
+
+static int snd_seq_ioctl_get_client_info(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client *cptr;
+ struct snd_seq_client_info client_info;
+
+ if (copy_from_user(&client_info, arg, sizeof(client_info)))
+ return -EFAULT;
+
+ /* requested client number */
+ cptr = snd_seq_client_use_ptr(client_info.client);
+ if (cptr == NULL)
+ return -ENOENT; /* don't change !!! */
+
+ get_client_info(cptr, &client_info);
+ snd_seq_client_unlock(cptr);
+
+ if (copy_to_user(arg, &client_info, sizeof(client_info)))
+ return -EFAULT;
+ return 0;
+}
+
+
+/* CLIENT_INFO ioctl() */
+static int snd_seq_ioctl_set_client_info(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client_info client_info;
+
+ if (copy_from_user(&client_info, arg, sizeof(client_info)))
+ return -EFAULT;
+
+ /* it is not allowed to set the info fields for an another client */
+ if (client->number != client_info.client)
+ return -EPERM;
+ /* also client type must be set now */
+ if (client->type != client_info.type)
+ return -EINVAL;
+
+ /* fill the info fields */
+ if (client_info.name[0])
+ strlcpy(client->name, client_info.name, sizeof(client->name));
+
+ client->filter = client_info.filter;
+ client->event_lost = client_info.event_lost;
+ memcpy(client->event_filter, client_info.event_filter, 32);
+
+ return 0;
+}
+
+
+/*
+ * CREATE PORT ioctl()
+ */
+static int snd_seq_ioctl_create_port(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client_port *port;
+ struct snd_seq_port_info info;
+ struct snd_seq_port_callback *callback;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ /* it is not allowed to create the port for an another client */
+ if (info.addr.client != client->number)
+ return -EPERM;
+
+ port = snd_seq_create_port(client, (info.flags & SNDRV_SEQ_PORT_FLG_GIVEN_PORT) ? info.addr.port : -1);
+ if (port == NULL)
+ return -ENOMEM;
+
+ if (client->type == USER_CLIENT && info.kernel) {
+ snd_seq_delete_port(client, port->addr.port);
+ return -EINVAL;
+ }
+ if (client->type == KERNEL_CLIENT) {
+ if ((callback = info.kernel) != NULL) {
+ if (callback->owner)
+ port->owner = callback->owner;
+ port->private_data = callback->private_data;
+ port->private_free = callback->private_free;
+ port->callback_all = callback->callback_all;
+ port->event_input = callback->event_input;
+ port->c_src.open = callback->subscribe;
+ port->c_src.close = callback->unsubscribe;
+ port->c_dest.open = callback->use;
+ port->c_dest.close = callback->unuse;
+ }
+ }
+
+ info.addr = port->addr;
+
+ snd_seq_set_port_info(port, &info);
+ snd_seq_system_client_ev_port_start(port->addr.client, port->addr.port);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+/*
+ * DELETE PORT ioctl()
+ */
+static int snd_seq_ioctl_delete_port(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_port_info info;
+ int err;
+
+ /* set passed parameters */
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ /* it is not allowed to remove the port for an another client */
+ if (info.addr.client != client->number)
+ return -EPERM;
+
+ err = snd_seq_delete_port(client, info.addr.port);
+ if (err >= 0)
+ snd_seq_system_client_ev_port_exit(client->number, info.addr.port);
+ return err;
+}
+
+
+/*
+ * GET_PORT_INFO ioctl() (on any client)
+ */
+static int snd_seq_ioctl_get_port_info(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client *cptr;
+ struct snd_seq_client_port *port;
+ struct snd_seq_port_info info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+ cptr = snd_seq_client_use_ptr(info.addr.client);
+ if (cptr == NULL)
+ return -ENXIO;
+
+ port = snd_seq_port_use_ptr(cptr, info.addr.port);
+ if (port == NULL) {
+ snd_seq_client_unlock(cptr);
+ return -ENOENT; /* don't change */
+ }
+
+ /* get port info */
+ snd_seq_get_port_info(port, &info);
+ snd_seq_port_unlock(port);
+ snd_seq_client_unlock(cptr);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+
+/*
+ * SET_PORT_INFO ioctl() (only ports on this/own client)
+ */
+static int snd_seq_ioctl_set_port_info(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client_port *port;
+ struct snd_seq_port_info info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ if (info.addr.client != client->number) /* only set our own ports ! */
+ return -EPERM;
+ port = snd_seq_port_use_ptr(client, info.addr.port);
+ if (port) {
+ snd_seq_set_port_info(port, &info);
+ snd_seq_port_unlock(port);
+ }
+ return 0;
+}
+
+
+/*
+ * port subscription (connection)
+ */
+#define PERM_RD (SNDRV_SEQ_PORT_CAP_READ|SNDRV_SEQ_PORT_CAP_SUBS_READ)
+#define PERM_WR (SNDRV_SEQ_PORT_CAP_WRITE|SNDRV_SEQ_PORT_CAP_SUBS_WRITE)
+
+static int check_subscription_permission(struct snd_seq_client *client,
+ struct snd_seq_client_port *sport,
+ struct snd_seq_client_port *dport,
+ struct snd_seq_port_subscribe *subs)
+{
+ if (client->number != subs->sender.client &&
+ client->number != subs->dest.client) {
+ /* connection by third client - check export permission */
+ if (check_port_perm(sport, SNDRV_SEQ_PORT_CAP_NO_EXPORT))
+ return -EPERM;
+ if (check_port_perm(dport, SNDRV_SEQ_PORT_CAP_NO_EXPORT))
+ return -EPERM;
+ }
+
+ /* check read permission */
+ /* if sender or receiver is the subscribing client itself,
+ * no permission check is necessary
+ */
+ if (client->number != subs->sender.client) {
+ if (! check_port_perm(sport, PERM_RD))
+ return -EPERM;
+ }
+ /* check write permission */
+ if (client->number != subs->dest.client) {
+ if (! check_port_perm(dport, PERM_WR))
+ return -EPERM;
+ }
+ return 0;
+}
+
+/*
+ * send an subscription notify event to user client:
+ * client must be user client.
+ */
+int snd_seq_client_notify_subscription(int client, int port,
+ struct snd_seq_port_subscribe *info,
+ int evtype)
+{
+ struct snd_seq_event event;
+
+ memset(&event, 0, sizeof(event));
+ event.type = evtype;
+ event.data.connect.dest = info->dest;
+ event.data.connect.sender = info->sender;
+
+ return snd_seq_system_notify(client, port, &event); /* non-atomic */
+}
+
+
+/*
+ * add to port's subscription list IOCTL interface
+ */
+static int snd_seq_ioctl_subscribe_port(struct snd_seq_client *client,
+ void __user *arg)
+{
+ int result = -EINVAL;
+ struct snd_seq_client *receiver = NULL, *sender = NULL;
+ struct snd_seq_client_port *sport = NULL, *dport = NULL;
+ struct snd_seq_port_subscribe subs;
+
+ if (copy_from_user(&subs, arg, sizeof(subs)))
+ return -EFAULT;
+
+ if ((receiver = snd_seq_client_use_ptr(subs.dest.client)) == NULL)
+ goto __end;
+ if ((sender = snd_seq_client_use_ptr(subs.sender.client)) == NULL)
+ goto __end;
+ if ((sport = snd_seq_port_use_ptr(sender, subs.sender.port)) == NULL)
+ goto __end;
+ if ((dport = snd_seq_port_use_ptr(receiver, subs.dest.port)) == NULL)
+ goto __end;
+
+ result = check_subscription_permission(client, sport, dport, &subs);
+ if (result < 0)
+ goto __end;
+
+ /* connect them */
+ result = snd_seq_port_connect(client, sender, sport, receiver, dport, &subs);
+ if (! result) /* broadcast announce */
+ snd_seq_client_notify_subscription(SNDRV_SEQ_ADDRESS_SUBSCRIBERS, 0,
+ &subs, SNDRV_SEQ_EVENT_PORT_SUBSCRIBED);
+ __end:
+ if (sport)
+ snd_seq_port_unlock(sport);
+ if (dport)
+ snd_seq_port_unlock(dport);
+ if (sender)
+ snd_seq_client_unlock(sender);
+ if (receiver)
+ snd_seq_client_unlock(receiver);
+ return result;
+}
+
+
+/*
+ * remove from port's subscription list
+ */
+static int snd_seq_ioctl_unsubscribe_port(struct snd_seq_client *client,
+ void __user *arg)
+{
+ int result = -ENXIO;
+ struct snd_seq_client *receiver = NULL, *sender = NULL;
+ struct snd_seq_client_port *sport = NULL, *dport = NULL;
+ struct snd_seq_port_subscribe subs;
+
+ if (copy_from_user(&subs, arg, sizeof(subs)))
+ return -EFAULT;
+
+ if ((receiver = snd_seq_client_use_ptr(subs.dest.client)) == NULL)
+ goto __end;
+ if ((sender = snd_seq_client_use_ptr(subs.sender.client)) == NULL)
+ goto __end;
+ if ((sport = snd_seq_port_use_ptr(sender, subs.sender.port)) == NULL)
+ goto __end;
+ if ((dport = snd_seq_port_use_ptr(receiver, subs.dest.port)) == NULL)
+ goto __end;
+
+ result = check_subscription_permission(client, sport, dport, &subs);
+ if (result < 0)
+ goto __end;
+
+ result = snd_seq_port_disconnect(client, sender, sport, receiver, dport, &subs);
+ if (! result) /* broadcast announce */
+ snd_seq_client_notify_subscription(SNDRV_SEQ_ADDRESS_SUBSCRIBERS, 0,
+ &subs, SNDRV_SEQ_EVENT_PORT_UNSUBSCRIBED);
+ __end:
+ if (sport)
+ snd_seq_port_unlock(sport);
+ if (dport)
+ snd_seq_port_unlock(dport);
+ if (sender)
+ snd_seq_client_unlock(sender);
+ if (receiver)
+ snd_seq_client_unlock(receiver);
+ return result;
+}
+
+
+/* CREATE_QUEUE ioctl() */
+static int snd_seq_ioctl_create_queue(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_info info;
+ int result;
+ struct snd_seq_queue *q;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ result = snd_seq_queue_alloc(client->number, info.locked, info.flags);
+ if (result < 0)
+ return result;
+
+ q = queueptr(result);
+ if (q == NULL)
+ return -EINVAL;
+
+ info.queue = q->queue;
+ info.locked = q->locked;
+ info.owner = q->owner;
+
+ /* set queue name */
+ if (! info.name[0])
+ snprintf(info.name, sizeof(info.name), "Queue-%d", q->queue);
+ strlcpy(q->name, info.name, sizeof(q->name));
+ queuefree(q);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+/* DELETE_QUEUE ioctl() */
+static int snd_seq_ioctl_delete_queue(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_info info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ return snd_seq_queue_delete(client->number, info.queue);
+}
+
+/* GET_QUEUE_INFO ioctl() */
+static int snd_seq_ioctl_get_queue_info(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_info info;
+ struct snd_seq_queue *q;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ q = queueptr(info.queue);
+ if (q == NULL)
+ return -EINVAL;
+
+ memset(&info, 0, sizeof(info));
+ info.queue = q->queue;
+ info.owner = q->owner;
+ info.locked = q->locked;
+ strlcpy(info.name, q->name, sizeof(info.name));
+ queuefree(q);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+/* SET_QUEUE_INFO ioctl() */
+static int snd_seq_ioctl_set_queue_info(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_info info;
+ struct snd_seq_queue *q;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ if (info.owner != client->number)
+ return -EINVAL;
+
+ /* change owner/locked permission */
+ if (snd_seq_queue_check_access(info.queue, client->number)) {
+ if (snd_seq_queue_set_owner(info.queue, client->number, info.locked) < 0)
+ return -EPERM;
+ if (info.locked)
+ snd_seq_queue_use(info.queue, client->number, 1);
+ } else {
+ return -EPERM;
+ }
+
+ q = queueptr(info.queue);
+ if (! q)
+ return -EINVAL;
+ if (q->owner != client->number) {
+ queuefree(q);
+ return -EPERM;
+ }
+ strlcpy(q->name, info.name, sizeof(q->name));
+ queuefree(q);
+
+ return 0;
+}
+
+/* GET_NAMED_QUEUE ioctl() */
+static int snd_seq_ioctl_get_named_queue(struct snd_seq_client *client, void __user *arg)
+{
+ struct snd_seq_queue_info info;
+ struct snd_seq_queue *q;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ q = snd_seq_queue_find_name(info.name);
+ if (q == NULL)
+ return -EINVAL;
+ info.queue = q->queue;
+ info.owner = q->owner;
+ info.locked = q->locked;
+ queuefree(q);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+/* GET_QUEUE_STATUS ioctl() */
+static int snd_seq_ioctl_get_queue_status(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_status status;
+ struct snd_seq_queue *queue;
+ struct snd_seq_timer *tmr;
+
+ if (copy_from_user(&status, arg, sizeof(status)))
+ return -EFAULT;
+
+ queue = queueptr(status.queue);
+ if (queue == NULL)
+ return -EINVAL;
+ memset(&status, 0, sizeof(status));
+ status.queue = queue->queue;
+
+ tmr = queue->timer;
+ status.events = queue->tickq->cells + queue->timeq->cells;
+
+ status.time = snd_seq_timer_get_cur_time(tmr);
+ status.tick = snd_seq_timer_get_cur_tick(tmr);
+
+ status.running = tmr->running;
+
+ status.flags = queue->flags;
+ queuefree(queue);
+
+ if (copy_to_user(arg, &status, sizeof(status)))
+ return -EFAULT;
+ return 0;
+}
+
+
+/* GET_QUEUE_TEMPO ioctl() */
+static int snd_seq_ioctl_get_queue_tempo(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_tempo tempo;
+ struct snd_seq_queue *queue;
+ struct snd_seq_timer *tmr;
+
+ if (copy_from_user(&tempo, arg, sizeof(tempo)))
+ return -EFAULT;
+
+ queue = queueptr(tempo.queue);
+ if (queue == NULL)
+ return -EINVAL;
+ memset(&tempo, 0, sizeof(tempo));
+ tempo.queue = queue->queue;
+
+ tmr = queue->timer;
+
+ tempo.tempo = tmr->tempo;
+ tempo.ppq = tmr->ppq;
+ tempo.skew_value = tmr->skew;
+ tempo.skew_base = tmr->skew_base;
+ queuefree(queue);
+
+ if (copy_to_user(arg, &tempo, sizeof(tempo)))
+ return -EFAULT;
+ return 0;
+}
+
+
+/* SET_QUEUE_TEMPO ioctl() */
+int snd_seq_set_queue_tempo(int client, struct snd_seq_queue_tempo *tempo)
+{
+ if (!snd_seq_queue_check_access(tempo->queue, client))
+ return -EPERM;
+ return snd_seq_queue_timer_set_tempo(tempo->queue, client, tempo);
+}
+
+EXPORT_SYMBOL(snd_seq_set_queue_tempo);
+
+static int snd_seq_ioctl_set_queue_tempo(struct snd_seq_client *client,
+ void __user *arg)
+{
+ int result;
+ struct snd_seq_queue_tempo tempo;
+
+ if (copy_from_user(&tempo, arg, sizeof(tempo)))
+ return -EFAULT;
+
+ result = snd_seq_set_queue_tempo(client->number, &tempo);
+ return result < 0 ? result : 0;
+}
+
+
+/* GET_QUEUE_TIMER ioctl() */
+static int snd_seq_ioctl_get_queue_timer(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_timer timer;
+ struct snd_seq_queue *queue;
+ struct snd_seq_timer *tmr;
+
+ if (copy_from_user(&timer, arg, sizeof(timer)))
+ return -EFAULT;
+
+ queue = queueptr(timer.queue);
+ if (queue == NULL)
+ return -EINVAL;
+
+ if (mutex_lock_interruptible(&queue->timer_mutex)) {
+ queuefree(queue);
+ return -ERESTARTSYS;
+ }
+ tmr = queue->timer;
+ memset(&timer, 0, sizeof(timer));
+ timer.queue = queue->queue;
+
+ timer.type = tmr->type;
+ if (tmr->type == SNDRV_SEQ_TIMER_ALSA) {
+ timer.u.alsa.id = tmr->alsa_id;
+ timer.u.alsa.resolution = tmr->preferred_resolution;
+ }
+ mutex_unlock(&queue->timer_mutex);
+ queuefree(queue);
+
+ if (copy_to_user(arg, &timer, sizeof(timer)))
+ return -EFAULT;
+ return 0;
+}
+
+
+/* SET_QUEUE_TIMER ioctl() */
+static int snd_seq_ioctl_set_queue_timer(struct snd_seq_client *client,
+ void __user *arg)
+{
+ int result = 0;
+ struct snd_seq_queue_timer timer;
+
+ if (copy_from_user(&timer, arg, sizeof(timer)))
+ return -EFAULT;
+
+ if (timer.type != SNDRV_SEQ_TIMER_ALSA)
+ return -EINVAL;
+
+ if (snd_seq_queue_check_access(timer.queue, client->number)) {
+ struct snd_seq_queue *q;
+ struct snd_seq_timer *tmr;
+
+ q = queueptr(timer.queue);
+ if (q == NULL)
+ return -ENXIO;
+ if (mutex_lock_interruptible(&q->timer_mutex)) {
+ queuefree(q);
+ return -ERESTARTSYS;
+ }
+ tmr = q->timer;
+ snd_seq_queue_timer_close(timer.queue);
+ tmr->type = timer.type;
+ if (tmr->type == SNDRV_SEQ_TIMER_ALSA) {
+ tmr->alsa_id = timer.u.alsa.id;
+ tmr->preferred_resolution = timer.u.alsa.resolution;
+ }
+ result = snd_seq_queue_timer_open(timer.queue);
+ mutex_unlock(&q->timer_mutex);
+ queuefree(q);
+ } else {
+ return -EPERM;
+ }
+
+ return result;
+}
+
+
+/* GET_QUEUE_CLIENT ioctl() */
+static int snd_seq_ioctl_get_queue_client(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_queue_client info;
+ int used;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ used = snd_seq_queue_is_used(info.queue, client->number);
+ if (used < 0)
+ return -EINVAL;
+ info.used = used;
+ info.client = client->number;
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+
+/* SET_QUEUE_CLIENT ioctl() */
+static int snd_seq_ioctl_set_queue_client(struct snd_seq_client *client,
+ void __user *arg)
+{
+ int err;
+ struct snd_seq_queue_client info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ if (info.used >= 0) {
+ err = snd_seq_queue_use(info.queue, client->number, info.used);
+ if (err < 0)
+ return err;
+ }
+
+ return snd_seq_ioctl_get_queue_client(client, arg);
+}
+
+
+/* GET_CLIENT_POOL ioctl() */
+static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client_pool info;
+ struct snd_seq_client *cptr;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ cptr = snd_seq_client_use_ptr(info.client);
+ if (cptr == NULL)
+ return -ENOENT;
+ memset(&info, 0, sizeof(info));
+ info.output_pool = cptr->pool->size;
+ info.output_room = cptr->pool->room;
+ info.output_free = info.output_pool;
+ info.output_free = snd_seq_unused_cells(cptr->pool);
+ if (cptr->type == USER_CLIENT) {
+ info.input_pool = cptr->data.user.fifo_pool_size;
+ info.input_free = info.input_pool;
+ if (cptr->data.user.fifo)
+ info.input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool);
+ } else {
+ info.input_pool = 0;
+ info.input_free = 0;
+ }
+ snd_seq_client_unlock(cptr);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+/* SET_CLIENT_POOL ioctl() */
+static int snd_seq_ioctl_set_client_pool(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client_pool info;
+ int rc;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ if (client->number != info.client)
+ return -EINVAL; /* can't change other clients */
+
+ if (info.output_pool >= 1 && info.output_pool <= SNDRV_SEQ_MAX_EVENTS &&
+ (! snd_seq_write_pool_allocated(client) ||
+ info.output_pool != client->pool->size)) {
+ if (snd_seq_write_pool_allocated(client)) {
+ /* remove all existing cells */
+ snd_seq_queue_client_leave_cells(client->number);
+ snd_seq_pool_done(client->pool);
+ }
+ client->pool->size = info.output_pool;
+ rc = snd_seq_pool_init(client->pool);
+ if (rc < 0)
+ return rc;
+ }
+ if (client->type == USER_CLIENT && client->data.user.fifo != NULL &&
+ info.input_pool >= 1 &&
+ info.input_pool <= SNDRV_SEQ_MAX_CLIENT_EVENTS &&
+ info.input_pool != client->data.user.fifo_pool_size) {
+ /* change pool size */
+ rc = snd_seq_fifo_resize(client->data.user.fifo, info.input_pool);
+ if (rc < 0)
+ return rc;
+ client->data.user.fifo_pool_size = info.input_pool;
+ }
+ if (info.output_room >= 1 &&
+ info.output_room <= client->pool->size) {
+ client->pool->room = info.output_room;
+ }
+
+ return snd_seq_ioctl_get_client_pool(client, arg);
+}
+
+
+/* REMOVE_EVENTS ioctl() */
+static int snd_seq_ioctl_remove_events(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_remove_events info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ /*
+ * Input mostly not implemented XXX.
+ */
+ if (info.remove_mode & SNDRV_SEQ_REMOVE_INPUT) {
+ /*
+ * No restrictions so for a user client we can clear
+ * the whole fifo
+ */
+ if (client->type == USER_CLIENT)
+ snd_seq_fifo_clear(client->data.user.fifo);
+ }
+
+ if (info.remove_mode & SNDRV_SEQ_REMOVE_OUTPUT)
+ snd_seq_queue_remove_cells(client->number, &info);
+
+ return 0;
+}
+
+
+/*
+ * get subscription info
+ */
+static int snd_seq_ioctl_get_subscription(struct snd_seq_client *client,
+ void __user *arg)
+{
+ int result;
+ struct snd_seq_client *sender = NULL;
+ struct snd_seq_client_port *sport = NULL;
+ struct snd_seq_port_subscribe subs;
+ struct snd_seq_subscribers *p;
+
+ if (copy_from_user(&subs, arg, sizeof(subs)))
+ return -EFAULT;
+
+ result = -EINVAL;
+ if ((sender = snd_seq_client_use_ptr(subs.sender.client)) == NULL)
+ goto __end;
+ if ((sport = snd_seq_port_use_ptr(sender, subs.sender.port)) == NULL)
+ goto __end;
+ p = snd_seq_port_get_subscription(&sport->c_src, &subs.dest);
+ if (p) {
+ result = 0;
+ subs = p->info;
+ } else
+ result = -ENOENT;
+
+ __end:
+ if (sport)
+ snd_seq_port_unlock(sport);
+ if (sender)
+ snd_seq_client_unlock(sender);
+ if (result >= 0) {
+ if (copy_to_user(arg, &subs, sizeof(subs)))
+ return -EFAULT;
+ }
+ return result;
+}
+
+
+/*
+ * get subscription info - check only its presence
+ */
+static int snd_seq_ioctl_query_subs(struct snd_seq_client *client,
+ void __user *arg)
+{
+ int result = -ENXIO;
+ struct snd_seq_client *cptr = NULL;
+ struct snd_seq_client_port *port = NULL;
+ struct snd_seq_query_subs subs;
+ struct snd_seq_port_subs_info *group;
+ struct list_head *p;
+ int i;
+
+ if (copy_from_user(&subs, arg, sizeof(subs)))
+ return -EFAULT;
+
+ if ((cptr = snd_seq_client_use_ptr(subs.root.client)) == NULL)
+ goto __end;
+ if ((port = snd_seq_port_use_ptr(cptr, subs.root.port)) == NULL)
+ goto __end;
+
+ switch (subs.type) {
+ case SNDRV_SEQ_QUERY_SUBS_READ:
+ group = &port->c_src;
+ break;
+ case SNDRV_SEQ_QUERY_SUBS_WRITE:
+ group = &port->c_dest;
+ break;
+ default:
+ goto __end;
+ }
+
+ down_read(&group->list_mutex);
+ /* search for the subscriber */
+ subs.num_subs = group->count;
+ i = 0;
+ result = -ENOENT;
+ list_for_each(p, &group->list_head) {
+ if (i++ == subs.index) {
+ /* found! */
+ struct snd_seq_subscribers *s;
+ if (subs.type == SNDRV_SEQ_QUERY_SUBS_READ) {
+ s = list_entry(p, struct snd_seq_subscribers, src_list);
+ subs.addr = s->info.dest;
+ } else {
+ s = list_entry(p, struct snd_seq_subscribers, dest_list);
+ subs.addr = s->info.sender;
+ }
+ subs.flags = s->info.flags;
+ subs.queue = s->info.queue;
+ result = 0;
+ break;
+ }
+ }
+ up_read(&group->list_mutex);
+
+ __end:
+ if (port)
+ snd_seq_port_unlock(port);
+ if (cptr)
+ snd_seq_client_unlock(cptr);
+ if (result >= 0) {
+ if (copy_to_user(arg, &subs, sizeof(subs)))
+ return -EFAULT;
+ }
+ return result;
+}
+
+
+/*
+ * query next client
+ */
+static int snd_seq_ioctl_query_next_client(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client *cptr = NULL;
+ struct snd_seq_client_info info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+
+ /* search for next client */
+ info.client++;
+ if (info.client < 0)
+ info.client = 0;
+ for (; info.client < SNDRV_SEQ_MAX_CLIENTS; info.client++) {
+ cptr = snd_seq_client_use_ptr(info.client);
+ if (cptr)
+ break; /* found */
+ }
+ if (cptr == NULL)
+ return -ENOENT;
+
+ get_client_info(cptr, &info);
+ snd_seq_client_unlock(cptr);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+/*
+ * query next port
+ */
+static int snd_seq_ioctl_query_next_port(struct snd_seq_client *client,
+ void __user *arg)
+{
+ struct snd_seq_client *cptr;
+ struct snd_seq_client_port *port = NULL;
+ struct snd_seq_port_info info;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+ cptr = snd_seq_client_use_ptr(info.addr.client);
+ if (cptr == NULL)
+ return -ENXIO;
+
+ /* search for next port */
+ info.addr.port++;
+ port = snd_seq_port_query_nearest(cptr, &info);
+ if (port == NULL) {
+ snd_seq_client_unlock(cptr);
+ return -ENOENT;
+ }
+
+ /* get port info */
+ info.addr = port->addr;
+ snd_seq_get_port_info(port, &info);
+ snd_seq_port_unlock(port);
+ snd_seq_client_unlock(cptr);
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+/* -------------------------------------------------------- */
+
+static struct seq_ioctl_table {
+ unsigned int cmd;
+ int (*func)(struct snd_seq_client *client, void __user * arg);
+} ioctl_tables[] = {
+ { SNDRV_SEQ_IOCTL_SYSTEM_INFO, snd_seq_ioctl_system_info },
+ { SNDRV_SEQ_IOCTL_RUNNING_MODE, snd_seq_ioctl_running_mode },
+ { SNDRV_SEQ_IOCTL_GET_CLIENT_INFO, snd_seq_ioctl_get_client_info },
+ { SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, snd_seq_ioctl_set_client_info },
+ { SNDRV_SEQ_IOCTL_CREATE_PORT, snd_seq_ioctl_create_port },
+ { SNDRV_SEQ_IOCTL_DELETE_PORT, snd_seq_ioctl_delete_port },
+ { SNDRV_SEQ_IOCTL_GET_PORT_INFO, snd_seq_ioctl_get_port_info },
+ { SNDRV_SEQ_IOCTL_SET_PORT_INFO, snd_seq_ioctl_set_port_info },
+ { SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT, snd_seq_ioctl_subscribe_port },
+ { SNDRV_SEQ_IOCTL_UNSUBSCRIBE_PORT, snd_seq_ioctl_unsubscribe_port },
+ { SNDRV_SEQ_IOCTL_CREATE_QUEUE, snd_seq_ioctl_create_queue },
+ { SNDRV_SEQ_IOCTL_DELETE_QUEUE, snd_seq_ioctl_delete_queue },
+ { SNDRV_SEQ_IOCTL_GET_QUEUE_INFO, snd_seq_ioctl_get_queue_info },
+ { SNDRV_SEQ_IOCTL_SET_QUEUE_INFO, snd_seq_ioctl_set_queue_info },
+ { SNDRV_SEQ_IOCTL_GET_NAMED_QUEUE, snd_seq_ioctl_get_named_queue },
+ { SNDRV_SEQ_IOCTL_GET_QUEUE_STATUS, snd_seq_ioctl_get_queue_status },
+ { SNDRV_SEQ_IOCTL_GET_QUEUE_TEMPO, snd_seq_ioctl_get_queue_tempo },
+ { SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO, snd_seq_ioctl_set_queue_tempo },
+ { SNDRV_SEQ_IOCTL_GET_QUEUE_TIMER, snd_seq_ioctl_get_queue_timer },
+ { SNDRV_SEQ_IOCTL_SET_QUEUE_TIMER, snd_seq_ioctl_set_queue_timer },
+ { SNDRV_SEQ_IOCTL_GET_QUEUE_CLIENT, snd_seq_ioctl_get_queue_client },
+ { SNDRV_SEQ_IOCTL_SET_QUEUE_CLIENT, snd_seq_ioctl_set_queue_client },
+ { SNDRV_SEQ_IOCTL_GET_CLIENT_POOL, snd_seq_ioctl_get_client_pool },
+ { SNDRV_SEQ_IOCTL_SET_CLIENT_POOL, snd_seq_ioctl_set_client_pool },
+ { SNDRV_SEQ_IOCTL_GET_SUBSCRIPTION, snd_seq_ioctl_get_subscription },
+ { SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT, snd_seq_ioctl_query_next_client },
+ { SNDRV_SEQ_IOCTL_QUERY_NEXT_PORT, snd_seq_ioctl_query_next_port },
+ { SNDRV_SEQ_IOCTL_REMOVE_EVENTS, snd_seq_ioctl_remove_events },
+ { SNDRV_SEQ_IOCTL_QUERY_SUBS, snd_seq_ioctl_query_subs },
+ { 0, NULL },
+};
+
+static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd,
+ void __user *arg)
+{
+ struct seq_ioctl_table *p;
+
+ switch (cmd) {
+ case SNDRV_SEQ_IOCTL_PVERSION:
+ /* return sequencer version number */
+ return put_user(SNDRV_SEQ_VERSION, (int __user *)arg) ? -EFAULT : 0;
+ case SNDRV_SEQ_IOCTL_CLIENT_ID:
+ /* return the id of this client */
+ return put_user(client->number, (int __user *)arg) ? -EFAULT : 0;
+ }
+
+ if (! arg)
+ return -EFAULT;
+ for (p = ioctl_tables; p->cmd; p++) {
+ if (p->cmd == cmd)
+ return p->func(client, arg);
+ }
+ snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%2x)\n",
+ cmd, _IOC_TYPE(cmd), _IOC_NR(cmd));
+ return -ENOTTY;
+}
+
+
+static long snd_seq_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_seq_client *client = file->private_data;
+
+ if (snd_BUG_ON(!client))
+ return -ENXIO;
+
+ return snd_seq_do_ioctl(client, cmd, (void __user *) arg);
+}
+
+#ifdef CONFIG_COMPAT
+#include "seq_compat.c"
+#else
+#define snd_seq_ioctl_compat NULL
+#endif
+
+/* -------------------------------------------------------- */
+
+
+/* exported to kernel modules */
+int snd_seq_create_kernel_client(struct snd_card *card, int client_index,
+ const char *name_fmt, ...)
+{
+ struct snd_seq_client *client;
+ va_list args;
+
+ if (snd_BUG_ON(in_interrupt()))
+ return -EBUSY;
+
+ if (card && client_index >= SNDRV_SEQ_CLIENTS_PER_CARD)
+ return -EINVAL;
+ if (card == NULL && client_index >= SNDRV_SEQ_GLOBAL_CLIENTS)
+ return -EINVAL;
+
+ if (mutex_lock_interruptible(&register_mutex))
+ return -ERESTARTSYS;
+
+ if (card) {
+ client_index += SNDRV_SEQ_GLOBAL_CLIENTS
+ + card->number * SNDRV_SEQ_CLIENTS_PER_CARD;
+ if (client_index >= SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN)
+ client_index = -1;
+ }
+
+ /* empty write queue as default */
+ client = seq_create_client1(client_index, 0);
+ if (client == NULL) {
+ mutex_unlock(&register_mutex);
+ return -EBUSY; /* failure code */
+ }
+ usage_alloc(&client_usage, 1);
+
+ client->accept_input = 1;
+ client->accept_output = 1;
+
+ va_start(args, name_fmt);
+ vsnprintf(client->name, sizeof(client->name), name_fmt, args);
+ va_end(args);
+
+ client->type = KERNEL_CLIENT;
+ mutex_unlock(&register_mutex);
+
+ /* make others aware this new client */
+ snd_seq_system_client_ev_client_start(client->number);
+
+ /* return client number to caller */
+ return client->number;
+}
+
+EXPORT_SYMBOL(snd_seq_create_kernel_client);
+
+/* exported to kernel modules */
+int snd_seq_delete_kernel_client(int client)
+{
+ struct snd_seq_client *ptr;
+
+ if (snd_BUG_ON(in_interrupt()))
+ return -EBUSY;
+
+ ptr = clientptr(client);
+ if (ptr == NULL)
+ return -EINVAL;
+
+ seq_free_client(ptr);
+ kfree(ptr);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_seq_delete_kernel_client);
+
+/* skeleton to enqueue event, called from snd_seq_kernel_client_enqueue
+ * and snd_seq_kernel_client_enqueue_blocking
+ */
+static int kernel_client_enqueue(int client, struct snd_seq_event *ev,
+ struct file *file, int blocking,
+ int atomic, int hop)
+{
+ struct snd_seq_client *cptr;
+ int result;
+
+ if (snd_BUG_ON(!ev))
+ return -EINVAL;
+
+ if (ev->type == SNDRV_SEQ_EVENT_NONE)
+ return 0; /* ignore this */
+ if (ev->type == SNDRV_SEQ_EVENT_KERNEL_ERROR)
+ return -EINVAL; /* quoted events can't be enqueued */
+
+ /* fill in client number */
+ ev->source.client = client;
+
+ if (check_event_type_and_length(ev))
+ return -EINVAL;
+
+ cptr = snd_seq_client_use_ptr(client);
+ if (cptr == NULL)
+ return -EINVAL;
+
+ if (! cptr->accept_output)
+ result = -EPERM;
+ else /* send it */
+ result = snd_seq_client_enqueue_event(cptr, ev, file, blocking, atomic, hop);
+
+ snd_seq_client_unlock(cptr);
+ return result;
+}
+
+/*
+ * exported, called by kernel clients to enqueue events (w/o blocking)
+ *
+ * RETURN VALUE: zero if succeed, negative if error
+ */
+int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event * ev,
+ int atomic, int hop)
+{
+ return kernel_client_enqueue(client, ev, NULL, 0, atomic, hop);
+}
+
+EXPORT_SYMBOL(snd_seq_kernel_client_enqueue);
+
+/*
+ * exported, called by kernel clients to enqueue events (with blocking)
+ *
+ * RETURN VALUE: zero if succeed, negative if error
+ */
+int snd_seq_kernel_client_enqueue_blocking(int client, struct snd_seq_event * ev,
+ struct file *file,
+ int atomic, int hop)
+{
+ return kernel_client_enqueue(client, ev, file, 1, atomic, hop);
+}
+
+EXPORT_SYMBOL(snd_seq_kernel_client_enqueue_blocking);
+
+/*
+ * exported, called by kernel clients to dispatch events directly to other
+ * clients, bypassing the queues. Event time-stamp will be updated.
+ *
+ * RETURN VALUE: negative = delivery failed,
+ * zero, or positive: the number of delivered events
+ */
+int snd_seq_kernel_client_dispatch(int client, struct snd_seq_event * ev,
+ int atomic, int hop)
+{
+ struct snd_seq_client *cptr;
+ int result;
+
+ if (snd_BUG_ON(!ev))
+ return -EINVAL;
+
+ /* fill in client number */
+ ev->queue = SNDRV_SEQ_QUEUE_DIRECT;
+ ev->source.client = client;
+
+ if (check_event_type_and_length(ev))
+ return -EINVAL;
+
+ cptr = snd_seq_client_use_ptr(client);
+ if (cptr == NULL)
+ return -EINVAL;
+
+ if (!cptr->accept_output)
+ result = -EPERM;
+ else
+ result = snd_seq_deliver_event(cptr, ev, atomic, hop);
+
+ snd_seq_client_unlock(cptr);
+ return result;
+}
+
+EXPORT_SYMBOL(snd_seq_kernel_client_dispatch);
+
+/*
+ * exported, called by kernel clients to perform same functions as with
+ * userland ioctl()
+ */
+int snd_seq_kernel_client_ctl(int clientid, unsigned int cmd, void *arg)
+{
+ struct snd_seq_client *client;
+ mm_segment_t fs;
+ int result;
+
+ client = clientptr(clientid);
+ if (client == NULL)
+ return -ENXIO;
+ fs = snd_enter_user();
+ result = snd_seq_do_ioctl(client, cmd, (void __user *)arg);
+ snd_leave_user(fs);
+ return result;
+}
+
+EXPORT_SYMBOL(snd_seq_kernel_client_ctl);
+
+/* exported (for OSS emulator) */
+int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table *wait)
+{
+ struct snd_seq_client *client;
+
+ client = clientptr(clientid);
+ if (client == NULL)
+ return -ENXIO;
+
+ if (! snd_seq_write_pool_allocated(client))
+ return 1;
+ if (snd_seq_pool_poll_wait(client->pool, file, wait))
+ return 1;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_seq_kernel_client_write_poll);
+
+/*---------------------------------------------------------------------------*/
+
+#ifdef CONFIG_PROC_FS
+/*
+ * /proc interface
+ */
+static void snd_seq_info_dump_subscribers(struct snd_info_buffer *buffer,
+ struct snd_seq_port_subs_info *group,
+ int is_src, char *msg)
+{
+ struct list_head *p;
+ struct snd_seq_subscribers *s;
+ int count = 0;
+
+ down_read(&group->list_mutex);
+ if (list_empty(&group->list_head)) {
+ up_read(&group->list_mutex);
+ return;
+ }
+ snd_iprintf(buffer, msg);
+ list_for_each(p, &group->list_head) {
+ if (is_src)
+ s = list_entry(p, struct snd_seq_subscribers, src_list);
+ else
+ s = list_entry(p, struct snd_seq_subscribers, dest_list);
+ if (count++)
+ snd_iprintf(buffer, ", ");
+ snd_iprintf(buffer, "%d:%d",
+ is_src ? s->info.dest.client : s->info.sender.client,
+ is_src ? s->info.dest.port : s->info.sender.port);
+ if (s->info.flags & SNDRV_SEQ_PORT_SUBS_TIMESTAMP)
+ snd_iprintf(buffer, "[%c:%d]", ((s->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL) ? 'r' : 't'), s->info.queue);
+ if (group->exclusive)
+ snd_iprintf(buffer, "[ex]");
+ }
+ up_read(&group->list_mutex);
+ snd_iprintf(buffer, "\n");
+}
+
+#define FLAG_PERM_RD(perm) ((perm) & SNDRV_SEQ_PORT_CAP_READ ? ((perm) & SNDRV_SEQ_PORT_CAP_SUBS_READ ? 'R' : 'r') : '-')
+#define FLAG_PERM_WR(perm) ((perm) & SNDRV_SEQ_PORT_CAP_WRITE ? ((perm) & SNDRV_SEQ_PORT_CAP_SUBS_WRITE ? 'W' : 'w') : '-')
+#define FLAG_PERM_EX(perm) ((perm) & SNDRV_SEQ_PORT_CAP_NO_EXPORT ? '-' : 'e')
+
+#define FLAG_PERM_DUPLEX(perm) ((perm) & SNDRV_SEQ_PORT_CAP_DUPLEX ? 'X' : '-')
+
+static void snd_seq_info_dump_ports(struct snd_info_buffer *buffer,
+ struct snd_seq_client *client)
+{
+ struct snd_seq_client_port *p;
+
+ mutex_lock(&client->ports_mutex);
+ list_for_each_entry(p, &client->ports_list_head, list) {
+ snd_iprintf(buffer, " Port %3d : \"%s\" (%c%c%c%c)\n",
+ p->addr.port, p->name,
+ FLAG_PERM_RD(p->capability),
+ FLAG_PERM_WR(p->capability),
+ FLAG_PERM_EX(p->capability),
+ FLAG_PERM_DUPLEX(p->capability));
+ snd_seq_info_dump_subscribers(buffer, &p->c_src, 1, " Connecting To: ");
+ snd_seq_info_dump_subscribers(buffer, &p->c_dest, 0, " Connected From: ");
+ }
+ mutex_unlock(&client->ports_mutex);
+}
+
+
+void snd_seq_info_pool(struct snd_info_buffer *buffer,
+ struct snd_seq_pool *pool, char *space);
+
+/* exported to seq_info.c */
+void snd_seq_info_clients_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int c;
+ struct snd_seq_client *client;
+
+ snd_iprintf(buffer, "Client info\n");
+ snd_iprintf(buffer, " cur clients : %d\n", client_usage.cur);
+ snd_iprintf(buffer, " peak clients : %d\n", client_usage.peak);
+ snd_iprintf(buffer, " max clients : %d\n", SNDRV_SEQ_MAX_CLIENTS);
+ snd_iprintf(buffer, "\n");
+
+ /* list the client table */
+ for (c = 0; c < SNDRV_SEQ_MAX_CLIENTS; c++) {
+ client = snd_seq_client_use_ptr(c);
+ if (client == NULL)
+ continue;
+ if (client->type == NO_CLIENT) {
+ snd_seq_client_unlock(client);
+ continue;
+ }
+
+ snd_iprintf(buffer, "Client %3d : \"%s\" [%s]\n",
+ c, client->name,
+ client->type == USER_CLIENT ? "User" : "Kernel");
+ snd_seq_info_dump_ports(buffer, client);
+ if (snd_seq_write_pool_allocated(client)) {
+ snd_iprintf(buffer, " Output pool :\n");
+ snd_seq_info_pool(buffer, client->pool, " ");
+ }
+ if (client->type == USER_CLIENT && client->data.user.fifo &&
+ client->data.user.fifo->pool) {
+ snd_iprintf(buffer, " Input pool :\n");
+ snd_seq_info_pool(buffer, client->data.user.fifo->pool, " ");
+ }
+ snd_seq_client_unlock(client);
+ }
+}
+#endif /* CONFIG_PROC_FS */
+
+/*---------------------------------------------------------------------------*/
+
+
+/*
+ * REGISTRATION PART
+ */
+
+static const struct file_operations snd_seq_f_ops =
+{
+ .owner = THIS_MODULE,
+ .read = snd_seq_read,
+ .write = snd_seq_write,
+ .open = snd_seq_open,
+ .release = snd_seq_release,
+ .poll = snd_seq_poll,
+ .unlocked_ioctl = snd_seq_ioctl,
+ .compat_ioctl = snd_seq_ioctl_compat,
+};
+
+/*
+ * register sequencer device
+ */
+int __init snd_sequencer_device_init(void)
+{
+ int err;
+
+ if (mutex_lock_interruptible(&register_mutex))
+ return -ERESTARTSYS;
+
+ if ((err = snd_register_device(SNDRV_DEVICE_TYPE_SEQUENCER, NULL, 0,
+ &snd_seq_f_ops, NULL, "seq")) < 0) {
+ mutex_unlock(&register_mutex);
+ return err;
+ }
+
+ mutex_unlock(&register_mutex);
+
+ return 0;
+}
+
+
+
+/*
+ * unregister sequencer device
+ */
+void __exit snd_sequencer_device_done(void)
+{
+ snd_unregister_device(SNDRV_DEVICE_TYPE_SEQUENCER, NULL, 0);
+}
diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h
new file mode 100644
index 0000000..20f0a72
--- /dev/null
+++ b/sound/core/seq/seq_clientmgr.h
@@ -0,0 +1,103 @@
+/*
+ * ALSA sequencer Client Manager
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_CLIENTMGR_H
+#define __SND_SEQ_CLIENTMGR_H
+
+#include <sound/seq_kernel.h>
+#include <linux/bitops.h>
+#include "seq_fifo.h"
+#include "seq_ports.h"
+#include "seq_lock.h"
+
+
+/* client manager */
+
+struct snd_seq_user_client {
+ struct file *file; /* file struct of client */
+ /* ... */
+
+ /* fifo */
+ struct snd_seq_fifo *fifo; /* queue for incoming events */
+ int fifo_pool_size;
+};
+
+struct snd_seq_kernel_client {
+ /* ... */
+};
+
+
+struct snd_seq_client {
+ snd_seq_client_type_t type;
+ unsigned int accept_input: 1,
+ accept_output: 1;
+ char name[64]; /* client name */
+ int number; /* client number */
+ unsigned int filter; /* filter flags */
+ DECLARE_BITMAP(event_filter, 256);
+ snd_use_lock_t use_lock;
+ int event_lost;
+ /* ports */
+ int num_ports; /* number of ports */
+ struct list_head ports_list_head;
+ rwlock_t ports_lock;
+ struct mutex ports_mutex;
+ int convert32; /* convert 32->64bit */
+
+ /* output pool */
+ struct snd_seq_pool *pool; /* memory pool for this client */
+
+ union {
+ struct snd_seq_user_client user;
+ struct snd_seq_kernel_client kernel;
+ } data;
+};
+
+/* usage statistics */
+struct snd_seq_usage {
+ int cur;
+ int peak;
+};
+
+
+int client_init_data(void);
+int snd_sequencer_device_init(void);
+void snd_sequencer_device_done(void);
+
+/* get locked pointer to client */
+struct snd_seq_client *snd_seq_client_use_ptr(int clientid);
+
+/* unlock pointer to client */
+#define snd_seq_client_unlock(client) snd_use_lock_free(&(client)->use_lock)
+
+/* dispatch event to client(s) */
+int snd_seq_dispatch_event(struct snd_seq_event_cell *cell, int atomic, int hop);
+
+/* exported to other modules */
+int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, int atomic, int hop);
+int snd_seq_kernel_client_enqueue_blocking(int client, struct snd_seq_event * ev,
+ struct file *file, int atomic, int hop);
+int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table *wait);
+int snd_seq_client_notify_subscription(int client, int port,
+ struct snd_seq_port_subscribe *info, int evtype);
+
+extern int seq_client_load[15];
+
+#endif
diff --git a/sound/core/seq/seq_compat.c b/sound/core/seq/seq_compat.c
new file mode 100644
index 0000000..38693f4
--- /dev/null
+++ b/sound/core/seq/seq_compat.c
@@ -0,0 +1,138 @@
+/*
+ * 32bit -> 64bit ioctl wrapper for sequencer API
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* This file included from seq.c */
+
+#include <linux/compat.h>
+
+struct snd_seq_port_info32 {
+ struct snd_seq_addr addr; /* client/port numbers */
+ char name[64]; /* port name */
+
+ u32 capability; /* port capability bits */
+ u32 type; /* port type bits */
+ s32 midi_channels; /* channels per MIDI port */
+ s32 midi_voices; /* voices per MIDI port */
+ s32 synth_voices; /* voices per SYNTH port */
+
+ s32 read_use; /* R/O: subscribers for output (from this port) */
+ s32 write_use; /* R/O: subscribers for input (to this port) */
+
+ u32 kernel; /* reserved for kernel use (must be NULL) */
+ u32 flags; /* misc. conditioning */
+ unsigned char time_queue; /* queue # for timestamping */
+ char reserved[59]; /* for future use */
+};
+
+static int snd_seq_call_port_info_ioctl(struct snd_seq_client *client, unsigned int cmd,
+ struct snd_seq_port_info32 __user *data32)
+{
+ int err = -EFAULT;
+ struct snd_seq_port_info *data;
+ mm_segment_t fs;
+
+ data = kmalloc(sizeof(*data), GFP_KERNEL);
+ if (! data)
+ return -ENOMEM;
+
+ if (copy_from_user(data, data32, sizeof(*data32)) ||
+ get_user(data->flags, &data32->flags) ||
+ get_user(data->time_queue, &data32->time_queue))
+ goto error;
+ data->kernel = NULL;
+
+ fs = snd_enter_user();
+ err = snd_seq_do_ioctl(client, cmd, data);
+ snd_leave_user(fs);
+ if (err < 0)
+ goto error;
+
+ if (copy_to_user(data32, data, sizeof(*data32)) ||
+ put_user(data->flags, &data32->flags) ||
+ put_user(data->time_queue, &data32->time_queue))
+ err = -EFAULT;
+
+ error:
+ kfree(data);
+ return err;
+}
+
+
+
+/*
+ */
+
+enum {
+ SNDRV_SEQ_IOCTL_CREATE_PORT32 = _IOWR('S', 0x20, struct snd_seq_port_info32),
+ SNDRV_SEQ_IOCTL_DELETE_PORT32 = _IOW ('S', 0x21, struct snd_seq_port_info32),
+ SNDRV_SEQ_IOCTL_GET_PORT_INFO32 = _IOWR('S', 0x22, struct snd_seq_port_info32),
+ SNDRV_SEQ_IOCTL_SET_PORT_INFO32 = _IOW ('S', 0x23, struct snd_seq_port_info32),
+ SNDRV_SEQ_IOCTL_QUERY_NEXT_PORT32 = _IOWR('S', 0x52, struct snd_seq_port_info32),
+};
+
+static long snd_seq_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_seq_client *client = file->private_data;
+ void __user *argp = compat_ptr(arg);
+
+ if (snd_BUG_ON(!client))
+ return -ENXIO;
+
+ switch (cmd) {
+ case SNDRV_SEQ_IOCTL_PVERSION:
+ case SNDRV_SEQ_IOCTL_CLIENT_ID:
+ case SNDRV_SEQ_IOCTL_SYSTEM_INFO:
+ case SNDRV_SEQ_IOCTL_GET_CLIENT_INFO:
+ case SNDRV_SEQ_IOCTL_SET_CLIENT_INFO:
+ case SNDRV_SEQ_IOCTL_SUBSCRIBE_PORT:
+ case SNDRV_SEQ_IOCTL_UNSUBSCRIBE_PORT:
+ case SNDRV_SEQ_IOCTL_CREATE_QUEUE:
+ case SNDRV_SEQ_IOCTL_DELETE_QUEUE:
+ case SNDRV_SEQ_IOCTL_GET_QUEUE_INFO:
+ case SNDRV_SEQ_IOCTL_SET_QUEUE_INFO:
+ case SNDRV_SEQ_IOCTL_GET_NAMED_QUEUE:
+ case SNDRV_SEQ_IOCTL_GET_QUEUE_STATUS:
+ case SNDRV_SEQ_IOCTL_GET_QUEUE_TEMPO:
+ case SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO:
+ case SNDRV_SEQ_IOCTL_GET_QUEUE_TIMER:
+ case SNDRV_SEQ_IOCTL_SET_QUEUE_TIMER:
+ case SNDRV_SEQ_IOCTL_GET_QUEUE_CLIENT:
+ case SNDRV_SEQ_IOCTL_SET_QUEUE_CLIENT:
+ case SNDRV_SEQ_IOCTL_GET_CLIENT_POOL:
+ case SNDRV_SEQ_IOCTL_SET_CLIENT_POOL:
+ case SNDRV_SEQ_IOCTL_REMOVE_EVENTS:
+ case SNDRV_SEQ_IOCTL_QUERY_SUBS:
+ case SNDRV_SEQ_IOCTL_GET_SUBSCRIPTION:
+ case SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT:
+ case SNDRV_SEQ_IOCTL_RUNNING_MODE:
+ return snd_seq_do_ioctl(client, cmd, argp);
+ case SNDRV_SEQ_IOCTL_CREATE_PORT32:
+ return snd_seq_call_port_info_ioctl(client, SNDRV_SEQ_IOCTL_CREATE_PORT, argp);
+ case SNDRV_SEQ_IOCTL_DELETE_PORT32:
+ return snd_seq_call_port_info_ioctl(client, SNDRV_SEQ_IOCTL_DELETE_PORT, argp);
+ case SNDRV_SEQ_IOCTL_GET_PORT_INFO32:
+ return snd_seq_call_port_info_ioctl(client, SNDRV_SEQ_IOCTL_GET_PORT_INFO, argp);
+ case SNDRV_SEQ_IOCTL_SET_PORT_INFO32:
+ return snd_seq_call_port_info_ioctl(client, SNDRV_SEQ_IOCTL_SET_PORT_INFO, argp);
+ case SNDRV_SEQ_IOCTL_QUERY_NEXT_PORT32:
+ return snd_seq_call_port_info_ioctl(client, SNDRV_SEQ_IOCTL_QUERY_NEXT_PORT, argp);
+ }
+ return -ENOIOCTLCMD;
+}
diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c
new file mode 100644
index 0000000..1f99767
--- /dev/null
+++ b/sound/core/seq/seq_device.c
@@ -0,0 +1,572 @@
+/*
+ * ALSA sequencer device management
+ * Copyright (c) 1999 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ *----------------------------------------------------------------
+ *
+ * This device handler separates the card driver module from sequencer
+ * stuff (sequencer core, synth drivers, etc), so that user can avoid
+ * to spend unnecessary resources e.g. if he needs only listening to
+ * MP3s.
+ *
+ * The card (or lowlevel) driver creates a sequencer device entry
+ * via snd_seq_device_new(). This is an entry pointer to communicate
+ * with the sequencer device "driver", which is involved with the
+ * actual part to communicate with the sequencer core.
+ * Each sequencer device entry has an id string and the corresponding
+ * driver with the same id is loaded when required. For example,
+ * lowlevel codes to access emu8000 chip on sbawe card are included in
+ * emu8000-synth module. To activate this module, the hardware
+ * resources like i/o port are passed via snd_seq_device argument.
+ *
+ */
+
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/seq_device.h>
+#include <sound/seq_kernel.h>
+#include <sound/initval.h>
+#include <linux/kmod.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("ALSA sequencer device management");
+MODULE_LICENSE("GPL");
+
+/* driver state */
+#define DRIVER_EMPTY 0
+#define DRIVER_LOADED (1<<0)
+#define DRIVER_REQUESTED (1<<1)
+#define DRIVER_LOCKED (1<<2)
+
+struct ops_list {
+ char id[ID_LEN]; /* driver id */
+ int driver; /* driver state */
+ int used; /* reference counter */
+ int argsize; /* argument size */
+
+ /* operators */
+ struct snd_seq_dev_ops ops;
+
+ /* registred devices */
+ struct list_head dev_list; /* list of devices */
+ int num_devices; /* number of associated devices */
+ int num_init_devices; /* number of initialized devices */
+ struct mutex reg_mutex;
+
+ struct list_head list; /* next driver */
+};
+
+
+static LIST_HEAD(opslist);
+static int num_ops;
+static DEFINE_MUTEX(ops_mutex);
+#ifdef CONFIG_PROC_FS
+static struct snd_info_entry *info_entry;
+#endif
+
+/*
+ * prototypes
+ */
+static int snd_seq_device_free(struct snd_seq_device *dev);
+static int snd_seq_device_dev_free(struct snd_device *device);
+static int snd_seq_device_dev_register(struct snd_device *device);
+static int snd_seq_device_dev_disconnect(struct snd_device *device);
+
+static int init_device(struct snd_seq_device *dev, struct ops_list *ops);
+static int free_device(struct snd_seq_device *dev, struct ops_list *ops);
+static struct ops_list *find_driver(char *id, int create_if_empty);
+static struct ops_list *create_driver(char *id);
+static void unlock_driver(struct ops_list *ops);
+static void remove_drivers(void);
+
+/*
+ * show all drivers and their status
+ */
+
+#ifdef CONFIG_PROC_FS
+static void snd_seq_device_info(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct ops_list *ops;
+
+ mutex_lock(&ops_mutex);
+ list_for_each_entry(ops, &opslist, list) {
+ snd_iprintf(buffer, "snd-%s%s%s%s,%d\n",
+ ops->id,
+ ops->driver & DRIVER_LOADED ? ",loaded" : (ops->driver == DRIVER_EMPTY ? ",empty" : ""),
+ ops->driver & DRIVER_REQUESTED ? ",requested" : "",
+ ops->driver & DRIVER_LOCKED ? ",locked" : "",
+ ops->num_devices);
+ }
+ mutex_unlock(&ops_mutex);
+}
+#endif
+
+/*
+ * load all registered drivers (called from seq_clientmgr.c)
+ */
+
+#ifdef CONFIG_MODULES
+/* avoid auto-loading during module_init() */
+static int snd_seq_in_init;
+void snd_seq_autoload_lock(void)
+{
+ snd_seq_in_init++;
+}
+
+void snd_seq_autoload_unlock(void)
+{
+ snd_seq_in_init--;
+}
+#endif
+
+void snd_seq_device_load_drivers(void)
+{
+#ifdef CONFIG_MODULES
+ struct ops_list *ops;
+
+ /* Calling request_module during module_init()
+ * may cause blocking.
+ */
+ if (snd_seq_in_init)
+ return;
+
+ mutex_lock(&ops_mutex);
+ list_for_each_entry(ops, &opslist, list) {
+ if (! (ops->driver & DRIVER_LOADED) &&
+ ! (ops->driver & DRIVER_REQUESTED)) {
+ ops->used++;
+ mutex_unlock(&ops_mutex);
+ ops->driver |= DRIVER_REQUESTED;
+ request_module("snd-%s", ops->id);
+ mutex_lock(&ops_mutex);
+ ops->used--;
+ }
+ }
+ mutex_unlock(&ops_mutex);
+#endif
+}
+
+/*
+ * register a sequencer device
+ * card = card info (NULL allowed)
+ * device = device number (if any)
+ * id = id of driver
+ * result = return pointer (NULL allowed if unnecessary)
+ */
+int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize,
+ struct snd_seq_device **result)
+{
+ struct snd_seq_device *dev;
+ struct ops_list *ops;
+ int err;
+ static struct snd_device_ops dops = {
+ .dev_free = snd_seq_device_dev_free,
+ .dev_register = snd_seq_device_dev_register,
+ .dev_disconnect = snd_seq_device_dev_disconnect,
+ };
+
+ if (result)
+ *result = NULL;
+
+ if (snd_BUG_ON(!id))
+ return -EINVAL;
+
+ ops = find_driver(id, 1);
+ if (ops == NULL)
+ return -ENOMEM;
+
+ dev = kzalloc(sizeof(*dev)*2 + argsize, GFP_KERNEL);
+ if (dev == NULL) {
+ unlock_driver(ops);
+ return -ENOMEM;
+ }
+
+ /* set up device info */
+ dev->card = card;
+ dev->device = device;
+ strlcpy(dev->id, id, sizeof(dev->id));
+ dev->argsize = argsize;
+ dev->status = SNDRV_SEQ_DEVICE_FREE;
+
+ /* add this device to the list */
+ mutex_lock(&ops->reg_mutex);
+ list_add_tail(&dev->list, &ops->dev_list);
+ ops->num_devices++;
+ mutex_unlock(&ops->reg_mutex);
+
+ unlock_driver(ops);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops)) < 0) {
+ snd_seq_device_free(dev);
+ return err;
+ }
+
+ if (result)
+ *result = dev;
+
+ return 0;
+}
+
+/*
+ * free the existing device
+ */
+static int snd_seq_device_free(struct snd_seq_device *dev)
+{
+ struct ops_list *ops;
+
+ if (snd_BUG_ON(!dev))
+ return -EINVAL;
+
+ ops = find_driver(dev->id, 0);
+ if (ops == NULL)
+ return -ENXIO;
+
+ /* remove the device from the list */
+ mutex_lock(&ops->reg_mutex);
+ list_del(&dev->list);
+ ops->num_devices--;
+ mutex_unlock(&ops->reg_mutex);
+
+ free_device(dev, ops);
+ if (dev->private_free)
+ dev->private_free(dev);
+ kfree(dev);
+
+ unlock_driver(ops);
+
+ return 0;
+}
+
+static int snd_seq_device_dev_free(struct snd_device *device)
+{
+ struct snd_seq_device *dev = device->device_data;
+ return snd_seq_device_free(dev);
+}
+
+/*
+ * register the device
+ */
+static int snd_seq_device_dev_register(struct snd_device *device)
+{
+ struct snd_seq_device *dev = device->device_data;
+ struct ops_list *ops;
+
+ ops = find_driver(dev->id, 0);
+ if (ops == NULL)
+ return -ENOENT;
+
+ /* initialize this device if the corresponding driver was
+ * already loaded
+ */
+ if (ops->driver & DRIVER_LOADED)
+ init_device(dev, ops);
+
+ unlock_driver(ops);
+ return 0;
+}
+
+/*
+ * disconnect the device
+ */
+static int snd_seq_device_dev_disconnect(struct snd_device *device)
+{
+ struct snd_seq_device *dev = device->device_data;
+ struct ops_list *ops;
+
+ ops = find_driver(dev->id, 0);
+ if (ops == NULL)
+ return -ENOENT;
+
+ free_device(dev, ops);
+
+ unlock_driver(ops);
+ return 0;
+}
+
+/*
+ * register device driver
+ * id = driver id
+ * entry = driver operators - duplicated to each instance
+ */
+int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry,
+ int argsize)
+{
+ struct ops_list *ops;
+ struct snd_seq_device *dev;
+
+ if (id == NULL || entry == NULL ||
+ entry->init_device == NULL || entry->free_device == NULL)
+ return -EINVAL;
+
+ snd_seq_autoload_lock();
+ ops = find_driver(id, 1);
+ if (ops == NULL) {
+ snd_seq_autoload_unlock();
+ return -ENOMEM;
+ }
+ if (ops->driver & DRIVER_LOADED) {
+ snd_printk(KERN_WARNING "driver_register: driver '%s' already exists\n", id);
+ unlock_driver(ops);
+ snd_seq_autoload_unlock();
+ return -EBUSY;
+ }
+
+ mutex_lock(&ops->reg_mutex);
+ /* copy driver operators */
+ ops->ops = *entry;
+ ops->driver |= DRIVER_LOADED;
+ ops->argsize = argsize;
+
+ /* initialize existing devices if necessary */
+ list_for_each_entry(dev, &ops->dev_list, list) {
+ init_device(dev, ops);
+ }
+ mutex_unlock(&ops->reg_mutex);
+
+ unlock_driver(ops);
+ snd_seq_autoload_unlock();
+
+ return 0;
+}
+
+
+/*
+ * create driver record
+ */
+static struct ops_list * create_driver(char *id)
+{
+ struct ops_list *ops;
+
+ ops = kzalloc(sizeof(*ops), GFP_KERNEL);
+ if (ops == NULL)
+ return ops;
+
+ /* set up driver entry */
+ strlcpy(ops->id, id, sizeof(ops->id));
+ mutex_init(&ops->reg_mutex);
+ /*
+ * The ->reg_mutex locking rules are per-driver, so we create
+ * separate per-driver lock classes:
+ */
+ lockdep_set_class(&ops->reg_mutex, (struct lock_class_key *)id);
+
+ ops->driver = DRIVER_EMPTY;
+ INIT_LIST_HEAD(&ops->dev_list);
+ /* lock this instance */
+ ops->used = 1;
+
+ /* register driver entry */
+ mutex_lock(&ops_mutex);
+ list_add_tail(&ops->list, &opslist);
+ num_ops++;
+ mutex_unlock(&ops_mutex);
+
+ return ops;
+}
+
+
+/*
+ * unregister the specified driver
+ */
+int snd_seq_device_unregister_driver(char *id)
+{
+ struct ops_list *ops;
+ struct snd_seq_device *dev;
+
+ ops = find_driver(id, 0);
+ if (ops == NULL)
+ return -ENXIO;
+ if (! (ops->driver & DRIVER_LOADED) ||
+ (ops->driver & DRIVER_LOCKED)) {
+ snd_printk(KERN_ERR "driver_unregister: cannot unload driver '%s': status=%x\n",
+ id, ops->driver);
+ unlock_driver(ops);
+ return -EBUSY;
+ }
+
+ /* close and release all devices associated with this driver */
+ mutex_lock(&ops->reg_mutex);
+ ops->driver |= DRIVER_LOCKED; /* do not remove this driver recursively */
+ list_for_each_entry(dev, &ops->dev_list, list) {
+ free_device(dev, ops);
+ }
+
+ ops->driver = 0;
+ if (ops->num_init_devices > 0)
+ snd_printk(KERN_ERR "free_driver: init_devices > 0!! (%d)\n",
+ ops->num_init_devices);
+ mutex_unlock(&ops->reg_mutex);
+
+ unlock_driver(ops);
+
+ /* remove empty driver entries */
+ remove_drivers();
+
+ return 0;
+}
+
+
+/*
+ * remove empty driver entries
+ */
+static void remove_drivers(void)
+{
+ struct list_head *head;
+
+ mutex_lock(&ops_mutex);
+ head = opslist.next;
+ while (head != &opslist) {
+ struct ops_list *ops = list_entry(head, struct ops_list, list);
+ if (! (ops->driver & DRIVER_LOADED) &&
+ ops->used == 0 && ops->num_devices == 0) {
+ head = head->next;
+ list_del(&ops->list);
+ kfree(ops);
+ num_ops--;
+ } else
+ head = head->next;
+ }
+ mutex_unlock(&ops_mutex);
+}
+
+/*
+ * initialize the device - call init_device operator
+ */
+static int init_device(struct snd_seq_device *dev, struct ops_list *ops)
+{
+ if (! (ops->driver & DRIVER_LOADED))
+ return 0; /* driver is not loaded yet */
+ if (dev->status != SNDRV_SEQ_DEVICE_FREE)
+ return 0; /* already initialized */
+ if (ops->argsize != dev->argsize) {
+ snd_printk(KERN_ERR "incompatible device '%s' for plug-in '%s' (%d %d)\n",
+ dev->name, ops->id, ops->argsize, dev->argsize);
+ return -EINVAL;
+ }
+ if (ops->ops.init_device(dev) >= 0) {
+ dev->status = SNDRV_SEQ_DEVICE_REGISTERED;
+ ops->num_init_devices++;
+ } else {
+ snd_printk(KERN_ERR "init_device failed: %s: %s\n",
+ dev->name, dev->id);
+ }
+
+ return 0;
+}
+
+/*
+ * release the device - call free_device operator
+ */
+static int free_device(struct snd_seq_device *dev, struct ops_list *ops)
+{
+ int result;
+
+ if (! (ops->driver & DRIVER_LOADED))
+ return 0; /* driver is not loaded yet */
+ if (dev->status != SNDRV_SEQ_DEVICE_REGISTERED)
+ return 0; /* not registered */
+ if (ops->argsize != dev->argsize) {
+ snd_printk(KERN_ERR "incompatible device '%s' for plug-in '%s' (%d %d)\n",
+ dev->name, ops->id, ops->argsize, dev->argsize);
+ return -EINVAL;
+ }
+ if ((result = ops->ops.free_device(dev)) >= 0 || result == -ENXIO) {
+ dev->status = SNDRV_SEQ_DEVICE_FREE;
+ dev->driver_data = NULL;
+ ops->num_init_devices--;
+ } else {
+ snd_printk(KERN_ERR "free_device failed: %s: %s\n",
+ dev->name, dev->id);
+ }
+
+ return 0;
+}
+
+/*
+ * find the matching driver with given id
+ */
+static struct ops_list * find_driver(char *id, int create_if_empty)
+{
+ struct ops_list *ops;
+
+ mutex_lock(&ops_mutex);
+ list_for_each_entry(ops, &opslist, list) {
+ if (strcmp(ops->id, id) == 0) {
+ ops->used++;
+ mutex_unlock(&ops_mutex);
+ return ops;
+ }
+ }
+ mutex_unlock(&ops_mutex);
+ if (create_if_empty)
+ return create_driver(id);
+ return NULL;
+}
+
+static void unlock_driver(struct ops_list *ops)
+{
+ mutex_lock(&ops_mutex);
+ ops->used--;
+ mutex_unlock(&ops_mutex);
+}
+
+
+/*
+ * module part
+ */
+
+static int __init alsa_seq_device_init(void)
+{
+#ifdef CONFIG_PROC_FS
+ info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers",
+ snd_seq_root);
+ if (info_entry == NULL)
+ return -ENOMEM;
+ info_entry->content = SNDRV_INFO_CONTENT_TEXT;
+ info_entry->c.text.read = snd_seq_device_info;
+ if (snd_info_register(info_entry) < 0) {
+ snd_info_free_entry(info_entry);
+ return -ENOMEM;
+ }
+#endif
+ return 0;
+}
+
+static void __exit alsa_seq_device_exit(void)
+{
+ remove_drivers();
+#ifdef CONFIG_PROC_FS
+ snd_info_free_entry(info_entry);
+#endif
+ if (num_ops)
+ snd_printk(KERN_ERR "drivers not released (%d)\n", num_ops);
+}
+
+module_init(alsa_seq_device_init)
+module_exit(alsa_seq_device_exit)
+
+EXPORT_SYMBOL(snd_seq_device_load_drivers);
+EXPORT_SYMBOL(snd_seq_device_new);
+EXPORT_SYMBOL(snd_seq_device_register_driver);
+EXPORT_SYMBOL(snd_seq_device_unregister_driver);
+EXPORT_SYMBOL(snd_seq_autoload_lock);
+EXPORT_SYMBOL(snd_seq_autoload_unlock);
diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
new file mode 100644
index 0000000..f3bdc54
--- /dev/null
+++ b/sound/core/seq/seq_dummy.c
@@ -0,0 +1,261 @@
+/*
+ * ALSA sequencer MIDI-through client
+ * Copyright (c) 1999-2000 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include "seq_clientmgr.h"
+#include <sound/initval.h>
+#include <sound/asoundef.h>
+
+/*
+
+ Sequencer MIDI-through client
+
+ This gives a simple midi-through client. All the normal input events
+ are redirected to output port immediately.
+ The routing can be done via aconnect program in alsa-utils.
+
+ Each client has a static client number 62 (= SNDRV_SEQ_CLIENT_DUMMY).
+ If you want to auto-load this module, you may add the following alias
+ in your /etc/conf.modules file.
+
+ alias snd-seq-client-62 snd-seq-dummy
+
+ The module is loaded on demand for client 62, or /proc/asound/seq/
+ is accessed. If you don't need this module to be loaded, alias
+ snd-seq-client-62 as "off". This will help modprobe.
+
+ The number of ports to be created can be specified via the module
+ parameter "ports". For example, to create four ports, add the
+ following option in /etc/modprobe.conf:
+
+ option snd-seq-dummy ports=4
+
+ The modle option "duplex=1" enables duplex operation to the port.
+ In duplex mode, a pair of ports are created instead of single port,
+ and events are tunneled between pair-ports. For example, input to
+ port A is sent to output port of another port B and vice versa.
+ In duplex mode, each port has DUPLEX capability.
+
+ */
+
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("ALSA sequencer MIDI-through client");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("snd-seq-client-" __stringify(SNDRV_SEQ_CLIENT_DUMMY));
+
+static int ports = 1;
+static int duplex;
+
+module_param(ports, int, 0444);
+MODULE_PARM_DESC(ports, "number of ports to be created");
+module_param(duplex, bool, 0444);
+MODULE_PARM_DESC(duplex, "create DUPLEX ports");
+
+struct snd_seq_dummy_port {
+ int client;
+ int port;
+ int duplex;
+ int connect;
+};
+
+static int my_client = -1;
+
+/*
+ * unuse callback - send ALL_SOUNDS_OFF and RESET_CONTROLLERS events
+ * to subscribers.
+ * Note: this callback is called only after all subscribers are removed.
+ */
+static int
+dummy_unuse(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ struct snd_seq_dummy_port *p;
+ int i;
+ struct snd_seq_event ev;
+
+ p = private_data;
+ memset(&ev, 0, sizeof(ev));
+ if (p->duplex)
+ ev.source.port = p->connect;
+ else
+ ev.source.port = p->port;
+ ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+ ev.type = SNDRV_SEQ_EVENT_CONTROLLER;
+ for (i = 0; i < 16; i++) {
+ ev.data.control.channel = i;
+ ev.data.control.param = MIDI_CTL_ALL_SOUNDS_OFF;
+ snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
+ ev.data.control.param = MIDI_CTL_RESET_CONTROLLERS;
+ snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
+ }
+ return 0;
+}
+
+/*
+ * event input callback - just redirect events to subscribers
+ */
+static int
+dummy_input(struct snd_seq_event *ev, int direct, void *private_data,
+ int atomic, int hop)
+{
+ struct snd_seq_dummy_port *p;
+ struct snd_seq_event tmpev;
+
+ p = private_data;
+ if (ev->source.client == SNDRV_SEQ_CLIENT_SYSTEM ||
+ ev->type == SNDRV_SEQ_EVENT_KERNEL_ERROR)
+ return 0; /* ignore system messages */
+ tmpev = *ev;
+ if (p->duplex)
+ tmpev.source.port = p->connect;
+ else
+ tmpev.source.port = p->port;
+ tmpev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+ return snd_seq_kernel_client_dispatch(p->client, &tmpev, atomic, hop);
+}
+
+/*
+ * free_private callback
+ */
+static void
+dummy_free(void *private_data)
+{
+ kfree(private_data);
+}
+
+/*
+ * create a port
+ */
+static struct snd_seq_dummy_port __init *
+create_port(int idx, int type)
+{
+ struct snd_seq_port_info pinfo;
+ struct snd_seq_port_callback pcb;
+ struct snd_seq_dummy_port *rec;
+
+ if ((rec = kzalloc(sizeof(*rec), GFP_KERNEL)) == NULL)
+ return NULL;
+
+ rec->client = my_client;
+ rec->duplex = duplex;
+ rec->connect = 0;
+ memset(&pinfo, 0, sizeof(pinfo));
+ pinfo.addr.client = my_client;
+ if (duplex)
+ sprintf(pinfo.name, "Midi Through Port-%d:%c", idx,
+ (type ? 'B' : 'A'));
+ else
+ sprintf(pinfo.name, "Midi Through Port-%d", idx);
+ pinfo.capability = SNDRV_SEQ_PORT_CAP_READ | SNDRV_SEQ_PORT_CAP_SUBS_READ;
+ pinfo.capability |= SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE;
+ if (duplex)
+ pinfo.capability |= SNDRV_SEQ_PORT_CAP_DUPLEX;
+ pinfo.type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC
+ | SNDRV_SEQ_PORT_TYPE_SOFTWARE
+ | SNDRV_SEQ_PORT_TYPE_PORT;
+ memset(&pcb, 0, sizeof(pcb));
+ pcb.owner = THIS_MODULE;
+ pcb.unuse = dummy_unuse;
+ pcb.event_input = dummy_input;
+ pcb.private_free = dummy_free;
+ pcb.private_data = rec;
+ pinfo.kernel = &pcb;
+ if (snd_seq_kernel_client_ctl(my_client, SNDRV_SEQ_IOCTL_CREATE_PORT, &pinfo) < 0) {
+ kfree(rec);
+ return NULL;
+ }
+ rec->port = pinfo.addr.port;
+ return rec;
+}
+
+/*
+ * register client and create ports
+ */
+static int __init
+register_client(void)
+{
+ struct snd_seq_dummy_port *rec1, *rec2;
+ int i;
+
+ if (ports < 1) {
+ snd_printk(KERN_ERR "invalid number of ports %d\n", ports);
+ return -EINVAL;
+ }
+
+ /* create client */
+ my_client = snd_seq_create_kernel_client(NULL, SNDRV_SEQ_CLIENT_DUMMY,
+ "Midi Through");
+ if (my_client < 0)
+ return my_client;
+
+ /* create ports */
+ for (i = 0; i < ports; i++) {
+ rec1 = create_port(i, 0);
+ if (rec1 == NULL) {
+ snd_seq_delete_kernel_client(my_client);
+ return -ENOMEM;
+ }
+ if (duplex) {
+ rec2 = create_port(i, 1);
+ if (rec2 == NULL) {
+ snd_seq_delete_kernel_client(my_client);
+ return -ENOMEM;
+ }
+ rec1->connect = rec2->port;
+ rec2->connect = rec1->port;
+ }
+ }
+
+ return 0;
+}
+
+/*
+ * delete client if exists
+ */
+static void __exit
+delete_client(void)
+{
+ if (my_client >= 0)
+ snd_seq_delete_kernel_client(my_client);
+}
+
+/*
+ * Init part
+ */
+
+static int __init alsa_seq_dummy_init(void)
+{
+ int err;
+ snd_seq_autoload_lock();
+ err = register_client();
+ snd_seq_autoload_unlock();
+ return err;
+}
+
+static void __exit alsa_seq_dummy_exit(void)
+{
+ delete_client();
+}
+
+module_init(alsa_seq_dummy_init)
+module_exit(alsa_seq_dummy_exit)
diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c
new file mode 100644
index 0000000..0d75afa
--- /dev/null
+++ b/sound/core/seq/seq_fifo.c
@@ -0,0 +1,272 @@
+/*
+ * ALSA sequencer FIFO
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <linux/slab.h>
+#include "seq_fifo.h"
+#include "seq_lock.h"
+
+
+/* FIFO */
+
+/* create new fifo */
+struct snd_seq_fifo *snd_seq_fifo_new(int poolsize)
+{
+ struct snd_seq_fifo *f;
+
+ f = kzalloc(sizeof(*f), GFP_KERNEL);
+ if (f == NULL) {
+ snd_printd("malloc failed for snd_seq_fifo_new() \n");
+ return NULL;
+ }
+
+ f->pool = snd_seq_pool_new(poolsize);
+ if (f->pool == NULL) {
+ kfree(f);
+ return NULL;
+ }
+ if (snd_seq_pool_init(f->pool) < 0) {
+ snd_seq_pool_delete(&f->pool);
+ kfree(f);
+ return NULL;
+ }
+
+ spin_lock_init(&f->lock);
+ snd_use_lock_init(&f->use_lock);
+ init_waitqueue_head(&f->input_sleep);
+ atomic_set(&f->overflow, 0);
+
+ f->head = NULL;
+ f->tail = NULL;
+ f->cells = 0;
+
+ return f;
+}
+
+void snd_seq_fifo_delete(struct snd_seq_fifo **fifo)
+{
+ struct snd_seq_fifo *f;
+
+ if (snd_BUG_ON(!fifo))
+ return;
+ f = *fifo;
+ if (snd_BUG_ON(!f))
+ return;
+ *fifo = NULL;
+
+ snd_seq_fifo_clear(f);
+
+ /* wake up clients if any */
+ if (waitqueue_active(&f->input_sleep))
+ wake_up(&f->input_sleep);
+
+ /* release resources...*/
+ /*....................*/
+
+ if (f->pool) {
+ snd_seq_pool_done(f->pool);
+ snd_seq_pool_delete(&f->pool);
+ }
+
+ kfree(f);
+}
+
+static struct snd_seq_event_cell *fifo_cell_out(struct snd_seq_fifo *f);
+
+/* clear queue */
+void snd_seq_fifo_clear(struct snd_seq_fifo *f)
+{
+ struct snd_seq_event_cell *cell;
+ unsigned long flags;
+
+ /* clear overflow flag */
+ atomic_set(&f->overflow, 0);
+
+ snd_use_lock_sync(&f->use_lock);
+ spin_lock_irqsave(&f->lock, flags);
+ /* drain the fifo */
+ while ((cell = fifo_cell_out(f)) != NULL) {
+ snd_seq_cell_free(cell);
+ }
+ spin_unlock_irqrestore(&f->lock, flags);
+}
+
+
+/* enqueue event to fifo */
+int snd_seq_fifo_event_in(struct snd_seq_fifo *f,
+ struct snd_seq_event *event)
+{
+ struct snd_seq_event_cell *cell;
+ unsigned long flags;
+ int err;
+
+ if (snd_BUG_ON(!f))
+ return -EINVAL;
+
+ snd_use_lock_use(&f->use_lock);
+ err = snd_seq_event_dup(f->pool, event, &cell, 1, NULL); /* always non-blocking */
+ if (err < 0) {
+ if (err == -ENOMEM)
+ atomic_inc(&f->overflow);
+ snd_use_lock_free(&f->use_lock);
+ return err;
+ }
+
+ /* append new cells to fifo */
+ spin_lock_irqsave(&f->lock, flags);
+ if (f->tail != NULL)
+ f->tail->next = cell;
+ f->tail = cell;
+ if (f->head == NULL)
+ f->head = cell;
+ f->cells++;
+ spin_unlock_irqrestore(&f->lock, flags);
+
+ /* wakeup client */
+ if (waitqueue_active(&f->input_sleep))
+ wake_up(&f->input_sleep);
+
+ snd_use_lock_free(&f->use_lock);
+
+ return 0; /* success */
+
+}
+
+/* dequeue cell from fifo */
+static struct snd_seq_event_cell *fifo_cell_out(struct snd_seq_fifo *f)
+{
+ struct snd_seq_event_cell *cell;
+
+ if ((cell = f->head) != NULL) {
+ f->head = cell->next;
+
+ /* reset tail if this was the last element */
+ if (f->tail == cell)
+ f->tail = NULL;
+
+ cell->next = NULL;
+ f->cells--;
+ }
+
+ return cell;
+}
+
+/* dequeue cell from fifo and copy on user space */
+int snd_seq_fifo_cell_out(struct snd_seq_fifo *f,
+ struct snd_seq_event_cell **cellp, int nonblock)
+{
+ struct snd_seq_event_cell *cell;
+ unsigned long flags;
+ wait_queue_t wait;
+
+ if (snd_BUG_ON(!f))
+ return -EINVAL;
+
+ *cellp = NULL;
+ init_waitqueue_entry(&wait, current);
+ spin_lock_irqsave(&f->lock, flags);
+ while ((cell = fifo_cell_out(f)) == NULL) {
+ if (nonblock) {
+ /* non-blocking - return immediately */
+ spin_unlock_irqrestore(&f->lock, flags);
+ return -EAGAIN;
+ }
+ set_current_state(TASK_INTERRUPTIBLE);
+ add_wait_queue(&f->input_sleep, &wait);
+ spin_unlock_irq(&f->lock);
+ schedule();
+ spin_lock_irq(&f->lock);
+ remove_wait_queue(&f->input_sleep, &wait);
+ if (signal_pending(current)) {
+ spin_unlock_irqrestore(&f->lock, flags);
+ return -ERESTARTSYS;
+ }
+ }
+ spin_unlock_irqrestore(&f->lock, flags);
+ *cellp = cell;
+
+ return 0;
+}
+
+
+void snd_seq_fifo_cell_putback(struct snd_seq_fifo *f,
+ struct snd_seq_event_cell *cell)
+{
+ unsigned long flags;
+
+ if (cell) {
+ spin_lock_irqsave(&f->lock, flags);
+ cell->next = f->head;
+ f->head = cell;
+ f->cells++;
+ spin_unlock_irqrestore(&f->lock, flags);
+ }
+}
+
+
+/* polling; return non-zero if queue is available */
+int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file,
+ poll_table *wait)
+{
+ poll_wait(file, &f->input_sleep, wait);
+ return (f->cells > 0);
+}
+
+/* change the size of pool; all old events are removed */
+int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize)
+{
+ unsigned long flags;
+ struct snd_seq_pool *newpool, *oldpool;
+ struct snd_seq_event_cell *cell, *next, *oldhead;
+
+ if (snd_BUG_ON(!f || !f->pool))
+ return -EINVAL;
+
+ /* allocate new pool */
+ newpool = snd_seq_pool_new(poolsize);
+ if (newpool == NULL)
+ return -ENOMEM;
+ if (snd_seq_pool_init(newpool) < 0) {
+ snd_seq_pool_delete(&newpool);
+ return -ENOMEM;
+ }
+
+ spin_lock_irqsave(&f->lock, flags);
+ /* remember old pool */
+ oldpool = f->pool;
+ oldhead = f->head;
+ /* exchange pools */
+ f->pool = newpool;
+ f->head = NULL;
+ f->tail = NULL;
+ f->cells = 0;
+ /* NOTE: overflow flag is not cleared */
+ spin_unlock_irqrestore(&f->lock, flags);
+
+ /* release cells in old pool */
+ for (cell = oldhead; cell; cell = next) {
+ next = cell->next;
+ snd_seq_cell_free(cell);
+ }
+ snd_seq_pool_delete(&oldpool);
+
+ return 0;
+}
diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h
new file mode 100644
index 0000000..062c446
--- /dev/null
+++ b/sound/core/seq/seq_fifo.h
@@ -0,0 +1,72 @@
+/*
+ * ALSA sequencer FIFO
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_FIFO_H
+#define __SND_SEQ_FIFO_H
+
+#include "seq_memory.h"
+#include "seq_lock.h"
+
+
+/* === FIFO === */
+
+struct snd_seq_fifo {
+ struct snd_seq_pool *pool; /* FIFO pool */
+ struct snd_seq_event_cell *head; /* pointer to head of fifo */
+ struct snd_seq_event_cell *tail; /* pointer to tail of fifo */
+ int cells;
+ spinlock_t lock;
+ snd_use_lock_t use_lock;
+ wait_queue_head_t input_sleep;
+ atomic_t overflow;
+
+};
+
+/* create new fifo (constructor) */
+struct snd_seq_fifo *snd_seq_fifo_new(int poolsize);
+
+/* delete fifo (destructor) */
+void snd_seq_fifo_delete(struct snd_seq_fifo **f);
+
+
+/* enqueue event to fifo */
+int snd_seq_fifo_event_in(struct snd_seq_fifo *f, struct snd_seq_event *event);
+
+/* lock fifo from release */
+#define snd_seq_fifo_lock(fifo) snd_use_lock_use(&(fifo)->use_lock)
+#define snd_seq_fifo_unlock(fifo) snd_use_lock_free(&(fifo)->use_lock)
+
+/* get a cell from fifo - fifo should be locked */
+int snd_seq_fifo_cell_out(struct snd_seq_fifo *f, struct snd_seq_event_cell **cellp, int nonblock);
+
+/* free dequeued cell - fifo should be locked */
+void snd_seq_fifo_cell_putback(struct snd_seq_fifo *f, struct snd_seq_event_cell *cell);
+
+/* clean up queue */
+void snd_seq_fifo_clear(struct snd_seq_fifo *f);
+
+/* polling */
+int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table *wait);
+
+/* resize pool in fifo */
+int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize);
+
+
+#endif
diff --git a/sound/core/seq/seq_info.c b/sound/core/seq/seq_info.c
new file mode 100644
index 0000000..201f810
--- /dev/null
+++ b/sound/core/seq/seq_info.c
@@ -0,0 +1,71 @@
+/*
+ * ALSA sequencer /proc interface
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <sound/core.h>
+
+#include "seq_info.h"
+#include "seq_clientmgr.h"
+#include "seq_timer.h"
+
+#ifdef CONFIG_PROC_FS
+static struct snd_info_entry *queues_entry;
+static struct snd_info_entry *clients_entry;
+static struct snd_info_entry *timer_entry;
+
+
+static struct snd_info_entry * __init
+create_info_entry(char *name, void (*read)(struct snd_info_entry *,
+ struct snd_info_buffer *))
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_module_entry(THIS_MODULE, name, snd_seq_root);
+ if (entry == NULL)
+ return NULL;
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->c.text.read = read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ return NULL;
+ }
+ return entry;
+}
+
+/* create all our /proc entries */
+int __init snd_seq_info_init(void)
+{
+ queues_entry = create_info_entry("queues",
+ snd_seq_info_queues_read);
+ clients_entry = create_info_entry("clients",
+ snd_seq_info_clients_read);
+ timer_entry = create_info_entry("timer", snd_seq_info_timer_read);
+ return 0;
+}
+
+int __exit snd_seq_info_done(void)
+{
+ snd_info_free_entry(queues_entry);
+ snd_info_free_entry(clients_entry);
+ snd_info_free_entry(timer_entry);
+ return 0;
+}
+#endif
diff --git a/sound/core/seq/seq_info.h b/sound/core/seq/seq_info.h
new file mode 100644
index 0000000..4892a7f
--- /dev/null
+++ b/sound/core/seq/seq_info.h
@@ -0,0 +1,40 @@
+/*
+ * ALSA sequencer /proc info
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_INFO_H
+#define __SND_SEQ_INFO_H
+
+#include <sound/info.h>
+#include <sound/seq_kernel.h>
+
+void snd_seq_info_clients_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer);
+void snd_seq_info_timer_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer);
+void snd_seq_info_queues_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer);
+
+
+#ifdef CONFIG_PROC_FS
+int snd_seq_info_init( void );
+int snd_seq_info_done( void );
+#else
+static inline int snd_seq_info_init(void) { return 0; }
+static inline int snd_seq_info_done(void) { return 0; }
+#endif
+
+#endif
diff --git a/sound/core/seq/seq_lock.c b/sound/core/seq/seq_lock.c
new file mode 100644
index 0000000..54f921e
--- /dev/null
+++ b/sound/core/seq/seq_lock.c
@@ -0,0 +1,48 @@
+/*
+ * Do sleep inside a spin-lock
+ * Copyright (c) 1999 by Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include "seq_lock.h"
+
+#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG)
+
+/* wait until all locks are released */
+void snd_use_lock_sync_helper(snd_use_lock_t *lockp, const char *file, int line)
+{
+ int max_count = 5 * HZ;
+
+ if (atomic_read(lockp) < 0) {
+ printk(KERN_WARNING "seq_lock: lock trouble [counter = %d] in %s:%d\n", atomic_read(lockp), file, line);
+ return;
+ }
+ while (atomic_read(lockp) > 0) {
+ if (max_count == 0) {
+ snd_printk(KERN_WARNING "seq_lock: timeout [%d left] in %s:%d\n", atomic_read(lockp), file, line);
+ break;
+ }
+ schedule_timeout_uninterruptible(1);
+ max_count--;
+ }
+}
+
+EXPORT_SYMBOL(snd_use_lock_sync_helper);
+
+#endif
diff --git a/sound/core/seq/seq_lock.h b/sound/core/seq/seq_lock.h
new file mode 100644
index 0000000..54044bc
--- /dev/null
+++ b/sound/core/seq/seq_lock.h
@@ -0,0 +1,33 @@
+#ifndef __SND_SEQ_LOCK_H
+#define __SND_SEQ_LOCK_H
+
+#include <linux/sched.h>
+
+#if defined(CONFIG_SMP) || defined(CONFIG_SND_DEBUG)
+
+typedef atomic_t snd_use_lock_t;
+
+/* initialize lock */
+#define snd_use_lock_init(lockp) atomic_set(lockp, 0)
+
+/* increment lock */
+#define snd_use_lock_use(lockp) atomic_inc(lockp)
+
+/* release lock */
+#define snd_use_lock_free(lockp) atomic_dec(lockp)
+
+/* wait until all locks are released */
+void snd_use_lock_sync_helper(snd_use_lock_t *lock, const char *file, int line);
+#define snd_use_lock_sync(lockp) snd_use_lock_sync_helper(lockp, __BASE_FILE__, __LINE__)
+
+#else /* SMP || CONFIG_SND_DEBUG */
+
+typedef spinlock_t snd_use_lock_t; /* dummy */
+#define snd_use_lock_init(lockp) /**/
+#define snd_use_lock_use(lockp) /**/
+#define snd_use_lock_free(lockp) /**/
+#define snd_use_lock_sync(lockp) /**/
+
+#endif /* SMP || CONFIG_SND_DEBUG */
+
+#endif /* __SND_SEQ_LOCK_H */
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
new file mode 100644
index 0000000..7fb5543
--- /dev/null
+++ b/sound/core/seq/seq_memory.c
@@ -0,0 +1,520 @@
+/*
+ * ALSA sequencer Memory Manager
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ * Jaroslav Kysela <perex@perex.cz>
+ * 2000 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <sound/core.h>
+
+#include <sound/seq_kernel.h>
+#include "seq_memory.h"
+#include "seq_queue.h"
+#include "seq_info.h"
+#include "seq_lock.h"
+
+static inline int snd_seq_pool_available(struct snd_seq_pool *pool)
+{
+ return pool->total_elements - atomic_read(&pool->counter);
+}
+
+static inline int snd_seq_output_ok(struct snd_seq_pool *pool)
+{
+ return snd_seq_pool_available(pool) >= pool->room;
+}
+
+/*
+ * Variable length event:
+ * The event like sysex uses variable length type.
+ * The external data may be stored in three different formats.
+ * 1) kernel space
+ * This is the normal case.
+ * ext.data.len = length
+ * ext.data.ptr = buffer pointer
+ * 2) user space
+ * When an event is generated via read(), the external data is
+ * kept in user space until expanded.
+ * ext.data.len = length | SNDRV_SEQ_EXT_USRPTR
+ * ext.data.ptr = userspace pointer
+ * 3) chained cells
+ * When the variable length event is enqueued (in prioq or fifo),
+ * the external data is decomposed to several cells.
+ * ext.data.len = length | SNDRV_SEQ_EXT_CHAINED
+ * ext.data.ptr = the additiona cell head
+ * -> cell.next -> cell.next -> ..
+ */
+
+/*
+ * exported:
+ * call dump function to expand external data.
+ */
+
+static int get_var_len(const struct snd_seq_event *event)
+{
+ if ((event->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
+ return -EINVAL;
+
+ return event->data.ext.len & ~SNDRV_SEQ_EXT_MASK;
+}
+
+int snd_seq_dump_var_event(const struct snd_seq_event *event,
+ snd_seq_dump_func_t func, void *private_data)
+{
+ int len, err;
+ struct snd_seq_event_cell *cell;
+
+ if ((len = get_var_len(event)) <= 0)
+ return len;
+
+ if (event->data.ext.len & SNDRV_SEQ_EXT_USRPTR) {
+ char buf[32];
+ char __user *curptr = (char __user *)event->data.ext.ptr;
+ while (len > 0) {
+ int size = sizeof(buf);
+ if (len < size)
+ size = len;
+ if (copy_from_user(buf, curptr, size))
+ return -EFAULT;
+ err = func(private_data, buf, size);
+ if (err < 0)
+ return err;
+ curptr += size;
+ len -= size;
+ }
+ return 0;
+ } if (! (event->data.ext.len & SNDRV_SEQ_EXT_CHAINED)) {
+ return func(private_data, event->data.ext.ptr, len);
+ }
+
+ cell = (struct snd_seq_event_cell *)event->data.ext.ptr;
+ for (; len > 0 && cell; cell = cell->next) {
+ int size = sizeof(struct snd_seq_event);
+ if (len < size)
+ size = len;
+ err = func(private_data, &cell->event, size);
+ if (err < 0)
+ return err;
+ len -= size;
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_seq_dump_var_event);
+
+
+/*
+ * exported:
+ * expand the variable length event to linear buffer space.
+ */
+
+static int seq_copy_in_kernel(char **bufptr, const void *src, int size)
+{
+ memcpy(*bufptr, src, size);
+ *bufptr += size;
+ return 0;
+}
+
+static int seq_copy_in_user(char __user **bufptr, const void *src, int size)
+{
+ if (copy_to_user(*bufptr, src, size))
+ return -EFAULT;
+ *bufptr += size;
+ return 0;
+}
+
+int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char *buf,
+ int in_kernel, int size_aligned)
+{
+ int len, newlen;
+ int err;
+
+ if ((len = get_var_len(event)) < 0)
+ return len;
+ newlen = len;
+ if (size_aligned > 0)
+ newlen = roundup(len, size_aligned);
+ if (count < newlen)
+ return -EAGAIN;
+
+ if (event->data.ext.len & SNDRV_SEQ_EXT_USRPTR) {
+ if (! in_kernel)
+ return -EINVAL;
+ if (copy_from_user(buf, (void __user *)event->data.ext.ptr, len))
+ return -EFAULT;
+ return newlen;
+ }
+ err = snd_seq_dump_var_event(event,
+ in_kernel ? (snd_seq_dump_func_t)seq_copy_in_kernel :
+ (snd_seq_dump_func_t)seq_copy_in_user,
+ &buf);
+ return err < 0 ? err : newlen;
+}
+
+EXPORT_SYMBOL(snd_seq_expand_var_event);
+
+/*
+ * release this cell, free extended data if available
+ */
+
+static inline void free_cell(struct snd_seq_pool *pool,
+ struct snd_seq_event_cell *cell)
+{
+ cell->next = pool->free;
+ pool->free = cell;
+ atomic_dec(&pool->counter);
+}
+
+void snd_seq_cell_free(struct snd_seq_event_cell * cell)
+{
+ unsigned long flags;
+ struct snd_seq_pool *pool;
+
+ if (snd_BUG_ON(!cell))
+ return;
+ pool = cell->pool;
+ if (snd_BUG_ON(!pool))
+ return;
+
+ spin_lock_irqsave(&pool->lock, flags);
+ free_cell(pool, cell);
+ if (snd_seq_ev_is_variable(&cell->event)) {
+ if (cell->event.data.ext.len & SNDRV_SEQ_EXT_CHAINED) {
+ struct snd_seq_event_cell *curp, *nextptr;
+ curp = cell->event.data.ext.ptr;
+ for (; curp; curp = nextptr) {
+ nextptr = curp->next;
+ curp->next = pool->free;
+ free_cell(pool, curp);
+ }
+ }
+ }
+ if (waitqueue_active(&pool->output_sleep)) {
+ /* has enough space now? */
+ if (snd_seq_output_ok(pool))
+ wake_up(&pool->output_sleep);
+ }
+ spin_unlock_irqrestore(&pool->lock, flags);
+}
+
+
+/*
+ * allocate an event cell.
+ */
+static int snd_seq_cell_alloc(struct snd_seq_pool *pool,
+ struct snd_seq_event_cell **cellp,
+ int nonblock, struct file *file)
+{
+ struct snd_seq_event_cell *cell;
+ unsigned long flags;
+ int err = -EAGAIN;
+ wait_queue_t wait;
+
+ if (pool == NULL)
+ return -EINVAL;
+
+ *cellp = NULL;
+
+ init_waitqueue_entry(&wait, current);
+ spin_lock_irqsave(&pool->lock, flags);
+ if (pool->ptr == NULL) { /* not initialized */
+ snd_printd("seq: pool is not initialized\n");
+ err = -EINVAL;
+ goto __error;
+ }
+ while (pool->free == NULL && ! nonblock && ! pool->closing) {
+
+ set_current_state(TASK_INTERRUPTIBLE);
+ add_wait_queue(&pool->output_sleep, &wait);
+ spin_unlock_irq(&pool->lock);
+ schedule();
+ spin_lock_irq(&pool->lock);
+ remove_wait_queue(&pool->output_sleep, &wait);
+ /* interrupted? */
+ if (signal_pending(current)) {
+ err = -ERESTARTSYS;
+ goto __error;
+ }
+ }
+ if (pool->closing) { /* closing.. */
+ err = -ENOMEM;
+ goto __error;
+ }
+
+ cell = pool->free;
+ if (cell) {
+ int used;
+ pool->free = cell->next;
+ atomic_inc(&pool->counter);
+ used = atomic_read(&pool->counter);
+ if (pool->max_used < used)
+ pool->max_used = used;
+ pool->event_alloc_success++;
+ /* clear cell pointers */
+ cell->next = NULL;
+ err = 0;
+ } else
+ pool->event_alloc_failures++;
+ *cellp = cell;
+
+__error:
+ spin_unlock_irqrestore(&pool->lock, flags);
+ return err;
+}
+
+
+/*
+ * duplicate the event to a cell.
+ * if the event has external data, the data is decomposed to additional
+ * cells.
+ */
+int snd_seq_event_dup(struct snd_seq_pool *pool, struct snd_seq_event *event,
+ struct snd_seq_event_cell **cellp, int nonblock,
+ struct file *file)
+{
+ int ncells, err;
+ unsigned int extlen;
+ struct snd_seq_event_cell *cell;
+
+ *cellp = NULL;
+
+ ncells = 0;
+ extlen = 0;
+ if (snd_seq_ev_is_variable(event)) {
+ extlen = event->data.ext.len & ~SNDRV_SEQ_EXT_MASK;
+ ncells = (extlen + sizeof(struct snd_seq_event) - 1) / sizeof(struct snd_seq_event);
+ }
+ if (ncells >= pool->total_elements)
+ return -ENOMEM;
+
+ err = snd_seq_cell_alloc(pool, &cell, nonblock, file);
+ if (err < 0)
+ return err;
+
+ /* copy the event */
+ cell->event = *event;
+
+ /* decompose */
+ if (snd_seq_ev_is_variable(event)) {
+ int len = extlen;
+ int is_chained = event->data.ext.len & SNDRV_SEQ_EXT_CHAINED;
+ int is_usrptr = event->data.ext.len & SNDRV_SEQ_EXT_USRPTR;
+ struct snd_seq_event_cell *src, *tmp, *tail;
+ char *buf;
+
+ cell->event.data.ext.len = extlen | SNDRV_SEQ_EXT_CHAINED;
+ cell->event.data.ext.ptr = NULL;
+
+ src = (struct snd_seq_event_cell *)event->data.ext.ptr;
+ buf = (char *)event->data.ext.ptr;
+ tail = NULL;
+
+ while (ncells-- > 0) {
+ int size = sizeof(struct snd_seq_event);
+ if (len < size)
+ size = len;
+ err = snd_seq_cell_alloc(pool, &tmp, nonblock, file);
+ if (err < 0)
+ goto __error;
+ if (cell->event.data.ext.ptr == NULL)
+ cell->event.data.ext.ptr = tmp;
+ if (tail)
+ tail->next = tmp;
+ tail = tmp;
+ /* copy chunk */
+ if (is_chained && src) {
+ tmp->event = src->event;
+ src = src->next;
+ } else if (is_usrptr) {
+ if (copy_from_user(&tmp->event, (char __user *)buf, size)) {
+ err = -EFAULT;
+ goto __error;
+ }
+ } else {
+ memcpy(&tmp->event, buf, size);
+ }
+ buf += size;
+ len -= size;
+ }
+ }
+
+ *cellp = cell;
+ return 0;
+
+__error:
+ snd_seq_cell_free(cell);
+ return err;
+}
+
+
+/* poll wait */
+int snd_seq_pool_poll_wait(struct snd_seq_pool *pool, struct file *file,
+ poll_table *wait)
+{
+ poll_wait(file, &pool->output_sleep, wait);
+ return snd_seq_output_ok(pool);
+}
+
+
+/* allocate room specified number of events */
+int snd_seq_pool_init(struct snd_seq_pool *pool)
+{
+ int cell;
+ struct snd_seq_event_cell *cellptr;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!pool))
+ return -EINVAL;
+ if (pool->ptr) /* should be atomic? */
+ return 0;
+
+ pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size);
+ if (pool->ptr == NULL) {
+ snd_printd("seq: malloc for sequencer events failed\n");
+ return -ENOMEM;
+ }
+
+ /* add new cells to the free cell list */
+ spin_lock_irqsave(&pool->lock, flags);
+ pool->free = NULL;
+
+ for (cell = 0; cell < pool->size; cell++) {
+ cellptr = pool->ptr + cell;
+ cellptr->pool = pool;
+ cellptr->next = pool->free;
+ pool->free = cellptr;
+ }
+ pool->room = (pool->size + 1) / 2;
+
+ /* init statistics */
+ pool->max_used = 0;
+ pool->total_elements = pool->size;
+ spin_unlock_irqrestore(&pool->lock, flags);
+ return 0;
+}
+
+/* remove events */
+int snd_seq_pool_done(struct snd_seq_pool *pool)
+{
+ unsigned long flags;
+ struct snd_seq_event_cell *ptr;
+ int max_count = 5 * HZ;
+
+ if (snd_BUG_ON(!pool))
+ return -EINVAL;
+
+ /* wait for closing all threads */
+ spin_lock_irqsave(&pool->lock, flags);
+ pool->closing = 1;
+ spin_unlock_irqrestore(&pool->lock, flags);
+
+ if (waitqueue_active(&pool->output_sleep))
+ wake_up(&pool->output_sleep);
+
+ while (atomic_read(&pool->counter) > 0) {
+ if (max_count == 0) {
+ snd_printk(KERN_WARNING "snd_seq_pool_done timeout: %d cells remain\n", atomic_read(&pool->counter));
+ break;
+ }
+ schedule_timeout_uninterruptible(1);
+ max_count--;
+ }
+
+ /* release all resources */
+ spin_lock_irqsave(&pool->lock, flags);
+ ptr = pool->ptr;
+ pool->ptr = NULL;
+ pool->free = NULL;
+ pool->total_elements = 0;
+ spin_unlock_irqrestore(&pool->lock, flags);
+
+ vfree(ptr);
+
+ spin_lock_irqsave(&pool->lock, flags);
+ pool->closing = 0;
+ spin_unlock_irqrestore(&pool->lock, flags);
+
+ return 0;
+}
+
+
+/* init new memory pool */
+struct snd_seq_pool *snd_seq_pool_new(int poolsize)
+{
+ struct snd_seq_pool *pool;
+
+ /* create pool block */
+ pool = kzalloc(sizeof(*pool), GFP_KERNEL);
+ if (pool == NULL) {
+ snd_printd("seq: malloc failed for pool\n");
+ return NULL;
+ }
+ spin_lock_init(&pool->lock);
+ pool->ptr = NULL;
+ pool->free = NULL;
+ pool->total_elements = 0;
+ atomic_set(&pool->counter, 0);
+ pool->closing = 0;
+ init_waitqueue_head(&pool->output_sleep);
+
+ pool->size = poolsize;
+
+ /* init statistics */
+ pool->max_used = 0;
+ return pool;
+}
+
+/* remove memory pool */
+int snd_seq_pool_delete(struct snd_seq_pool **ppool)
+{
+ struct snd_seq_pool *pool = *ppool;
+
+ *ppool = NULL;
+ if (pool == NULL)
+ return 0;
+ snd_seq_pool_done(pool);
+ kfree(pool);
+ return 0;
+}
+
+/* initialize sequencer memory */
+int __init snd_sequencer_memory_init(void)
+{
+ return 0;
+}
+
+/* release sequencer memory */
+void __exit snd_sequencer_memory_done(void)
+{
+}
+
+
+/* exported to seq_clientmgr.c */
+void snd_seq_info_pool(struct snd_info_buffer *buffer,
+ struct snd_seq_pool *pool, char *space)
+{
+ if (pool == NULL)
+ return;
+ snd_iprintf(buffer, "%sPool size : %d\n", space, pool->total_elements);
+ snd_iprintf(buffer, "%sCells in use : %d\n", space, atomic_read(&pool->counter));
+ snd_iprintf(buffer, "%sPeak cells in use : %d\n", space, pool->max_used);
+ snd_iprintf(buffer, "%sAlloc success : %d\n", space, pool->event_alloc_success);
+ snd_iprintf(buffer, "%sAlloc failures : %d\n", space, pool->event_alloc_failures);
+}
diff --git a/sound/core/seq/seq_memory.h b/sound/core/seq/seq_memory.h
new file mode 100644
index 0000000..63e9143
--- /dev/null
+++ b/sound/core/seq/seq_memory.h
@@ -0,0 +1,103 @@
+/*
+ * ALSA sequencer Memory Manager
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_MEMORYMGR_H
+#define __SND_SEQ_MEMORYMGR_H
+
+#include <sound/seq_kernel.h>
+#include <linux/poll.h>
+
+/* container for sequencer event (internal use) */
+struct snd_seq_event_cell {
+ struct snd_seq_event event;
+ struct snd_seq_pool *pool; /* used pool */
+ struct snd_seq_event_cell *next; /* next cell */
+};
+
+/* design note: the pool is a contiguous block of memory, if we dynamicly
+ want to add additional cells to the pool be better store this in another
+ pool as we need to know the base address of the pool when releasing
+ memory. */
+
+struct snd_seq_pool {
+ struct snd_seq_event_cell *ptr; /* pointer to first event chunk */
+ struct snd_seq_event_cell *free; /* pointer to the head of the free list */
+
+ int total_elements; /* pool size actually allocated */
+ atomic_t counter; /* cells free */
+
+ int size; /* pool size to be allocated */
+ int room; /* watermark for sleep/wakeup */
+
+ int closing;
+
+ /* statistics */
+ int max_used;
+ int event_alloc_nopool;
+ int event_alloc_failures;
+ int event_alloc_success;
+
+ /* Write locking */
+ wait_queue_head_t output_sleep;
+
+ /* Pool lock */
+ spinlock_t lock;
+};
+
+void snd_seq_cell_free(struct snd_seq_event_cell *cell);
+
+int snd_seq_event_dup(struct snd_seq_pool *pool, struct snd_seq_event *event,
+ struct snd_seq_event_cell **cellp, int nonblock, struct file *file);
+
+/* return number of unused (free) cells */
+static inline int snd_seq_unused_cells(struct snd_seq_pool *pool)
+{
+ return pool ? pool->total_elements - atomic_read(&pool->counter) : 0;
+}
+
+/* return total number of allocated cells */
+static inline int snd_seq_total_cells(struct snd_seq_pool *pool)
+{
+ return pool ? pool->total_elements : 0;
+}
+
+/* init pool - allocate events */
+int snd_seq_pool_init(struct snd_seq_pool *pool);
+
+/* done pool - free events */
+int snd_seq_pool_done(struct snd_seq_pool *pool);
+
+/* create pool */
+struct snd_seq_pool *snd_seq_pool_new(int poolsize);
+
+/* remove pool */
+int snd_seq_pool_delete(struct snd_seq_pool **pool);
+
+/* init memory */
+int snd_sequencer_memory_init(void);
+
+/* release event memory */
+void snd_sequencer_memory_done(void);
+
+/* polling */
+int snd_seq_pool_poll_wait(struct snd_seq_pool *pool, struct file *file, poll_table *wait);
+
+
+#endif
diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c
new file mode 100644
index 0000000..4d26146
--- /dev/null
+++ b/sound/core/seq/seq_midi.c
@@ -0,0 +1,482 @@
+/*
+ * Generic MIDI synth driver for ALSA sequencer
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+Possible options for midisynth module:
+ - automatic opening of midi ports on first received event or subscription
+ (close will be performed when client leaves)
+*/
+
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/string.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+#include <sound/seq_kernel.h>
+#include <sound/seq_device.h>
+#include <sound/seq_midi_event.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Frank van de Pol <fvdpol@coil.demon.nl>, Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI synth.");
+MODULE_LICENSE("GPL");
+static int output_buffer_size = PAGE_SIZE;
+module_param(output_buffer_size, int, 0644);
+MODULE_PARM_DESC(output_buffer_size, "Output buffer size in bytes.");
+static int input_buffer_size = PAGE_SIZE;
+module_param(input_buffer_size, int, 0644);
+MODULE_PARM_DESC(input_buffer_size, "Input buffer size in bytes.");
+
+/* data for this midi synth driver */
+struct seq_midisynth {
+ struct snd_card *card;
+ int device;
+ int subdevice;
+ struct snd_rawmidi_file input_rfile;
+ struct snd_rawmidi_file output_rfile;
+ int seq_client;
+ int seq_port;
+ struct snd_midi_event *parser;
+};
+
+struct seq_midisynth_client {
+ int seq_client;
+ int num_ports;
+ int ports_per_device[SNDRV_RAWMIDI_DEVICES];
+ struct seq_midisynth *ports[SNDRV_RAWMIDI_DEVICES];
+};
+
+static struct seq_midisynth_client *synths[SNDRV_CARDS];
+static DEFINE_MUTEX(register_mutex);
+
+/* handle rawmidi input event (MIDI v1.0 stream) */
+static void snd_midi_input_event(struct snd_rawmidi_substream *substream)
+{
+ struct snd_rawmidi_runtime *runtime;
+ struct seq_midisynth *msynth;
+ struct snd_seq_event ev;
+ char buf[16], *pbuf;
+ long res, count;
+
+ if (substream == NULL)
+ return;
+ runtime = substream->runtime;
+ msynth = runtime->private_data;
+ if (msynth == NULL)
+ return;
+ memset(&ev, 0, sizeof(ev));
+ while (runtime->avail > 0) {
+ res = snd_rawmidi_kernel_read(substream, buf, sizeof(buf));
+ if (res <= 0)
+ continue;
+ if (msynth->parser == NULL)
+ continue;
+ pbuf = buf;
+ while (res > 0) {
+ count = snd_midi_event_encode(msynth->parser, pbuf, res, &ev);
+ if (count < 0)
+ break;
+ pbuf += count;
+ res -= count;
+ if (ev.type != SNDRV_SEQ_EVENT_NONE) {
+ ev.source.port = msynth->seq_port;
+ ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+ snd_seq_kernel_client_dispatch(msynth->seq_client, &ev, 1, 0);
+ /* clear event and reset header */
+ memset(&ev, 0, sizeof(ev));
+ }
+ }
+ }
+}
+
+static int dump_midi(struct snd_rawmidi_substream *substream, const char *buf, int count)
+{
+ struct snd_rawmidi_runtime *runtime;
+ int tmp;
+
+ if (snd_BUG_ON(!substream || !buf))
+ return -EINVAL;
+ runtime = substream->runtime;
+ if ((tmp = runtime->avail) < count) {
+ snd_printd("warning, output event was lost (count = %i, available = %i)\n", count, tmp);
+ return -ENOMEM;
+ }
+ if (snd_rawmidi_kernel_write(substream, buf, count) < count)
+ return -EINVAL;
+ return 0;
+}
+
+static int event_process_midi(struct snd_seq_event *ev, int direct,
+ void *private_data, int atomic, int hop)
+{
+ struct seq_midisynth *msynth = private_data;
+ unsigned char msg[10]; /* buffer for constructing midi messages */
+ struct snd_rawmidi_substream *substream;
+ int len;
+
+ if (snd_BUG_ON(!msynth))
+ return -EINVAL;
+ substream = msynth->output_rfile.output;
+ if (substream == NULL)
+ return -ENODEV;
+ if (ev->type == SNDRV_SEQ_EVENT_SYSEX) { /* special case, to save space */
+ if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE) {
+ /* invalid event */
+ snd_printd("seq_midi: invalid sysex event flags = 0x%x\n", ev->flags);
+ return 0;
+ }
+ snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream);
+ snd_midi_event_reset_decode(msynth->parser);
+ } else {
+ if (msynth->parser == NULL)
+ return -EIO;
+ len = snd_midi_event_decode(msynth->parser, msg, sizeof(msg), ev);
+ if (len < 0)
+ return 0;
+ if (dump_midi(substream, msg, len) < 0)
+ snd_midi_event_reset_decode(msynth->parser);
+ }
+ return 0;
+}
+
+
+static int snd_seq_midisynth_new(struct seq_midisynth *msynth,
+ struct snd_card *card,
+ int device,
+ int subdevice)
+{
+ if (snd_midi_event_new(MAX_MIDI_EVENT_BUF, &msynth->parser) < 0)
+ return -ENOMEM;
+ msynth->card = card;
+ msynth->device = device;
+ msynth->subdevice = subdevice;
+ return 0;
+}
+
+/* open associated midi device for input */
+static int midisynth_subscribe(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ int err;
+ struct seq_midisynth *msynth = private_data;
+ struct snd_rawmidi_runtime *runtime;
+ struct snd_rawmidi_params params;
+
+ /* open midi port */
+ if ((err = snd_rawmidi_kernel_open(msynth->card, msynth->device,
+ msynth->subdevice,
+ SNDRV_RAWMIDI_LFLG_INPUT,
+ &msynth->input_rfile)) < 0) {
+ snd_printd("midi input open failed!!!\n");
+ return err;
+ }
+ runtime = msynth->input_rfile.input->runtime;
+ memset(&params, 0, sizeof(params));
+ params.avail_min = 1;
+ params.buffer_size = input_buffer_size;
+ if ((err = snd_rawmidi_input_params(msynth->input_rfile.input, &params)) < 0) {
+ snd_rawmidi_kernel_release(&msynth->input_rfile);
+ return err;
+ }
+ snd_midi_event_reset_encode(msynth->parser);
+ runtime->event = snd_midi_input_event;
+ runtime->private_data = msynth;
+ snd_rawmidi_kernel_read(msynth->input_rfile.input, NULL, 0);
+ return 0;
+}
+
+/* close associated midi device for input */
+static int midisynth_unsubscribe(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ int err;
+ struct seq_midisynth *msynth = private_data;
+
+ if (snd_BUG_ON(!msynth->input_rfile.input))
+ return -EINVAL;
+ err = snd_rawmidi_kernel_release(&msynth->input_rfile);
+ return err;
+}
+
+/* open associated midi device for output */
+static int midisynth_use(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ int err;
+ struct seq_midisynth *msynth = private_data;
+ struct snd_rawmidi_params params;
+
+ /* open midi port */
+ if ((err = snd_rawmidi_kernel_open(msynth->card, msynth->device,
+ msynth->subdevice,
+ SNDRV_RAWMIDI_LFLG_OUTPUT,
+ &msynth->output_rfile)) < 0) {
+ snd_printd("midi output open failed!!!\n");
+ return err;
+ }
+ memset(&params, 0, sizeof(params));
+ params.avail_min = 1;
+ params.buffer_size = output_buffer_size;
+ if ((err = snd_rawmidi_output_params(msynth->output_rfile.output, &params)) < 0) {
+ snd_rawmidi_kernel_release(&msynth->output_rfile);
+ return err;
+ }
+ snd_midi_event_reset_decode(msynth->parser);
+ return 0;
+}
+
+/* close associated midi device for output */
+static int midisynth_unuse(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ struct seq_midisynth *msynth = private_data;
+ unsigned char buf = 0xff; /* MIDI reset */
+
+ if (snd_BUG_ON(!msynth->output_rfile.output))
+ return -EINVAL;
+ /* sending single MIDI reset message to shut the device up */
+ snd_rawmidi_kernel_write(msynth->output_rfile.output, &buf, 1);
+ snd_rawmidi_drain_output(msynth->output_rfile.output);
+ return snd_rawmidi_kernel_release(&msynth->output_rfile);
+}
+
+/* delete given midi synth port */
+static void snd_seq_midisynth_delete(struct seq_midisynth *msynth)
+{
+ if (msynth == NULL)
+ return;
+
+ if (msynth->seq_client > 0) {
+ /* delete port */
+ snd_seq_event_port_detach(msynth->seq_client, msynth->seq_port);
+ }
+
+ if (msynth->parser)
+ snd_midi_event_free(msynth->parser);
+}
+
+/* register new midi synth port */
+static int
+snd_seq_midisynth_register_port(struct snd_seq_device *dev)
+{
+ struct seq_midisynth_client *client;
+ struct seq_midisynth *msynth, *ms;
+ struct snd_seq_port_info *port;
+ struct snd_rawmidi_info *info;
+ struct snd_rawmidi *rmidi = dev->private_data;
+ int newclient = 0;
+ unsigned int p, ports;
+ struct snd_seq_port_callback pcallbacks;
+ struct snd_card *card = dev->card;
+ int device = dev->device;
+ unsigned int input_count = 0, output_count = 0;
+
+ if (snd_BUG_ON(!card || device < 0 || device >= SNDRV_RAWMIDI_DEVICES))
+ return -EINVAL;
+ info = kmalloc(sizeof(*info), GFP_KERNEL);
+ if (! info)
+ return -ENOMEM;
+ info->device = device;
+ info->stream = SNDRV_RAWMIDI_STREAM_OUTPUT;
+ info->subdevice = 0;
+ if (snd_rawmidi_info_select(card, info) >= 0)
+ output_count = info->subdevices_count;
+ info->stream = SNDRV_RAWMIDI_STREAM_INPUT;
+ if (snd_rawmidi_info_select(card, info) >= 0) {
+ input_count = info->subdevices_count;
+ }
+ ports = output_count;
+ if (ports < input_count)
+ ports = input_count;
+ if (ports == 0) {
+ kfree(info);
+ return -ENODEV;
+ }
+ if (ports > (256 / SNDRV_RAWMIDI_DEVICES))
+ ports = 256 / SNDRV_RAWMIDI_DEVICES;
+
+ mutex_lock(&register_mutex);
+ client = synths[card->number];
+ if (client == NULL) {
+ newclient = 1;
+ client = kzalloc(sizeof(*client), GFP_KERNEL);
+ if (client == NULL) {
+ mutex_unlock(&register_mutex);
+ kfree(info);
+ return -ENOMEM;
+ }
+ client->seq_client =
+ snd_seq_create_kernel_client(
+ card, 0, "%s", card->shortname[0] ?
+ (const char *)card->shortname : "External MIDI");
+ if (client->seq_client < 0) {
+ kfree(client);
+ mutex_unlock(&register_mutex);
+ kfree(info);
+ return -ENOMEM;
+ }
+ }
+
+ msynth = kcalloc(ports, sizeof(struct seq_midisynth), GFP_KERNEL);
+ port = kmalloc(sizeof(*port), GFP_KERNEL);
+ if (msynth == NULL || port == NULL)
+ goto __nomem;
+
+ for (p = 0; p < ports; p++) {
+ ms = &msynth[p];
+
+ if (snd_seq_midisynth_new(ms, card, device, p) < 0)
+ goto __nomem;
+
+ /* declare port */
+ memset(port, 0, sizeof(*port));
+ port->addr.client = client->seq_client;
+ port->addr.port = device * (256 / SNDRV_RAWMIDI_DEVICES) + p;
+ port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
+ memset(info, 0, sizeof(*info));
+ info->device = device;
+ if (p < output_count)
+ info->stream = SNDRV_RAWMIDI_STREAM_OUTPUT;
+ else
+ info->stream = SNDRV_RAWMIDI_STREAM_INPUT;
+ info->subdevice = p;
+ if (snd_rawmidi_info_select(card, info) >= 0)
+ strcpy(port->name, info->subname);
+ if (! port->name[0]) {
+ if (info->name[0]) {
+ if (ports > 1)
+ snprintf(port->name, sizeof(port->name), "%s-%d", info->name, p);
+ else
+ snprintf(port->name, sizeof(port->name), "%s", info->name);
+ } else {
+ /* last resort */
+ if (ports > 1)
+ sprintf(port->name, "MIDI %d-%d-%d", card->number, device, p);
+ else
+ sprintf(port->name, "MIDI %d-%d", card->number, device);
+ }
+ }
+ if ((info->flags & SNDRV_RAWMIDI_INFO_OUTPUT) && p < output_count)
+ port->capability |= SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SYNC_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE;
+ if ((info->flags & SNDRV_RAWMIDI_INFO_INPUT) && p < input_count)
+ port->capability |= SNDRV_SEQ_PORT_CAP_READ | SNDRV_SEQ_PORT_CAP_SYNC_READ | SNDRV_SEQ_PORT_CAP_SUBS_READ;
+ if ((port->capability & (SNDRV_SEQ_PORT_CAP_WRITE|SNDRV_SEQ_PORT_CAP_READ)) == (SNDRV_SEQ_PORT_CAP_WRITE|SNDRV_SEQ_PORT_CAP_READ) &&
+ info->flags & SNDRV_RAWMIDI_INFO_DUPLEX)
+ port->capability |= SNDRV_SEQ_PORT_CAP_DUPLEX;
+ port->type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC
+ | SNDRV_SEQ_PORT_TYPE_HARDWARE
+ | SNDRV_SEQ_PORT_TYPE_PORT;
+ port->midi_channels = 16;
+ memset(&pcallbacks, 0, sizeof(pcallbacks));
+ pcallbacks.owner = THIS_MODULE;
+ pcallbacks.private_data = ms;
+ pcallbacks.subscribe = midisynth_subscribe;
+ pcallbacks.unsubscribe = midisynth_unsubscribe;
+ pcallbacks.use = midisynth_use;
+ pcallbacks.unuse = midisynth_unuse;
+ pcallbacks.event_input = event_process_midi;
+ port->kernel = &pcallbacks;
+ if (rmidi->ops && rmidi->ops->get_port_info)
+ rmidi->ops->get_port_info(rmidi, p, port);
+ if (snd_seq_kernel_client_ctl(client->seq_client, SNDRV_SEQ_IOCTL_CREATE_PORT, port)<0)
+ goto __nomem;
+ ms->seq_client = client->seq_client;
+ ms->seq_port = port->addr.port;
+ }
+ client->ports_per_device[device] = ports;
+ client->ports[device] = msynth;
+ client->num_ports++;
+ if (newclient)
+ synths[card->number] = client;
+ mutex_unlock(&register_mutex);
+ kfree(info);
+ kfree(port);
+ return 0; /* success */
+
+ __nomem:
+ if (msynth != NULL) {
+ for (p = 0; p < ports; p++)
+ snd_seq_midisynth_delete(&msynth[p]);
+ kfree(msynth);
+ }
+ if (newclient) {
+ snd_seq_delete_kernel_client(client->seq_client);
+ kfree(client);
+ }
+ kfree(info);
+ kfree(port);
+ mutex_unlock(&register_mutex);
+ return -ENOMEM;
+}
+
+/* release midi synth port */
+static int
+snd_seq_midisynth_unregister_port(struct snd_seq_device *dev)
+{
+ struct seq_midisynth_client *client;
+ struct seq_midisynth *msynth;
+ struct snd_card *card = dev->card;
+ int device = dev->device, p, ports;
+
+ mutex_lock(&register_mutex);
+ client = synths[card->number];
+ if (client == NULL || client->ports[device] == NULL) {
+ mutex_unlock(&register_mutex);
+ return -ENODEV;
+ }
+ ports = client->ports_per_device[device];
+ client->ports_per_device[device] = 0;
+ msynth = client->ports[device];
+ client->ports[device] = NULL;
+ for (p = 0; p < ports; p++)
+ snd_seq_midisynth_delete(&msynth[p]);
+ kfree(msynth);
+ client->num_ports--;
+ if (client->num_ports <= 0) {
+ snd_seq_delete_kernel_client(client->seq_client);
+ synths[card->number] = NULL;
+ kfree(client);
+ }
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+
+static int __init alsa_seq_midi_init(void)
+{
+ static struct snd_seq_dev_ops ops = {
+ snd_seq_midisynth_register_port,
+ snd_seq_midisynth_unregister_port,
+ };
+ memset(&synths, 0, sizeof(synths));
+ snd_seq_autoload_lock();
+ snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH, &ops, 0);
+ snd_seq_autoload_unlock();
+ return 0;
+}
+
+static void __exit alsa_seq_midi_exit(void)
+{
+ snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH);
+}
+
+module_init(alsa_seq_midi_init)
+module_exit(alsa_seq_midi_exit)
diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c
new file mode 100644
index 0000000..07c6631
--- /dev/null
+++ b/sound/core/seq/seq_midi_emul.c
@@ -0,0 +1,739 @@
+/*
+ * GM/GS/XG midi module.
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ *
+ * Based on awe_wave.c by Takashi Iwai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+/*
+ * This module is used to keep track of the current midi state.
+ * It can be used for drivers that are required to emulate midi when
+ * the hardware doesn't.
+ *
+ * It was written for a AWE64 driver, but there should be no AWE specific
+ * code in here. If there is it should be reported as a bug.
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/seq_kernel.h>
+#include <sound/seq_midi_emul.h>
+#include <sound/initval.h>
+#include <sound/asoundef.h>
+
+MODULE_AUTHOR("Takashi Iwai / Steve Ratcliffe");
+MODULE_DESCRIPTION("Advanced Linux Sound Architecture sequencer MIDI emulation.");
+MODULE_LICENSE("GPL");
+
+/* Prototypes for static functions */
+static void note_off(struct snd_midi_op *ops, void *drv,
+ struct snd_midi_channel *chan,
+ int note, int vel);
+static void do_control(struct snd_midi_op *ops, void *private,
+ struct snd_midi_channel_set *chset,
+ struct snd_midi_channel *chan,
+ int control, int value);
+static void rpn(struct snd_midi_op *ops, void *drv, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset);
+static void nrpn(struct snd_midi_op *ops, void *drv, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset);
+static void sysex(struct snd_midi_op *ops, void *private, unsigned char *sysex,
+ int len, struct snd_midi_channel_set *chset);
+static void all_sounds_off(struct snd_midi_op *ops, void *private,
+ struct snd_midi_channel *chan);
+static void all_notes_off(struct snd_midi_op *ops, void *private,
+ struct snd_midi_channel *chan);
+static void snd_midi_reset_controllers(struct snd_midi_channel *chan);
+static void reset_all_channels(struct snd_midi_channel_set *chset);
+
+
+/*
+ * Process an event in a driver independent way. This means dealing
+ * with RPN, NRPN, SysEx etc that are defined for common midi applications
+ * such as GM, GS and XG.
+ * There modes that this module will run in are:
+ * Generic MIDI - no interpretation at all, it will just save current values
+ * of controllers etc.
+ * GM - You can use all gm_ prefixed elements of chan. Controls, RPN, NRPN,
+ * SysEx will be interpreded as defined in General Midi.
+ * GS - You can use all gs_ prefixed elements of chan. Codes for GS will be
+ * interpreted.
+ * XG - You can use all xg_ prefixed elements of chan. Codes for XG will
+ * be interpreted.
+ */
+void
+snd_midi_process_event(struct snd_midi_op *ops,
+ struct snd_seq_event *ev,
+ struct snd_midi_channel_set *chanset)
+{
+ struct snd_midi_channel *chan;
+ void *drv;
+ int dest_channel = 0;
+
+ if (ev == NULL || chanset == NULL) {
+ snd_printd("ev or chanbase NULL (snd_midi_process_event)\n");
+ return;
+ }
+ if (chanset->channels == NULL)
+ return;
+
+ if (snd_seq_ev_is_channel_type(ev)) {
+ dest_channel = ev->data.note.channel;
+ if (dest_channel >= chanset->max_channels) {
+ snd_printd("dest channel is %d, max is %d\n",
+ dest_channel, chanset->max_channels);
+ return;
+ }
+ }
+
+ chan = chanset->channels + dest_channel;
+ drv = chanset->private_data;
+
+ /* EVENT_NOTE should be processed before queued */
+ if (ev->type == SNDRV_SEQ_EVENT_NOTE)
+ return;
+
+ /* Make sure that we don't have a note on that should really be
+ * a note off */
+ if (ev->type == SNDRV_SEQ_EVENT_NOTEON && ev->data.note.velocity == 0)
+ ev->type = SNDRV_SEQ_EVENT_NOTEOFF;
+
+ /* Make sure the note is within array range */
+ if (ev->type == SNDRV_SEQ_EVENT_NOTEON ||
+ ev->type == SNDRV_SEQ_EVENT_NOTEOFF ||
+ ev->type == SNDRV_SEQ_EVENT_KEYPRESS) {
+ if (ev->data.note.note >= 128)
+ return;
+ }
+
+ switch (ev->type) {
+ case SNDRV_SEQ_EVENT_NOTEON:
+ if (chan->note[ev->data.note.note] & SNDRV_MIDI_NOTE_ON) {
+ if (ops->note_off)
+ ops->note_off(drv, ev->data.note.note, 0, chan);
+ }
+ chan->note[ev->data.note.note] = SNDRV_MIDI_NOTE_ON;
+ if (ops->note_on)
+ ops->note_on(drv, ev->data.note.note, ev->data.note.velocity, chan);
+ break;
+ case SNDRV_SEQ_EVENT_NOTEOFF:
+ if (! (chan->note[ev->data.note.note] & SNDRV_MIDI_NOTE_ON))
+ break;
+ if (ops->note_off)
+ note_off(ops, drv, chan, ev->data.note.note, ev->data.note.velocity);
+ break;
+ case SNDRV_SEQ_EVENT_KEYPRESS:
+ if (ops->key_press)
+ ops->key_press(drv, ev->data.note.note, ev->data.note.velocity, chan);
+ break;
+ case SNDRV_SEQ_EVENT_CONTROLLER:
+ do_control(ops, drv, chanset, chan,
+ ev->data.control.param, ev->data.control.value);
+ break;
+ case SNDRV_SEQ_EVENT_PGMCHANGE:
+ chan->midi_program = ev->data.control.value;
+ break;
+ case SNDRV_SEQ_EVENT_PITCHBEND:
+ chan->midi_pitchbend = ev->data.control.value;
+ if (ops->control)
+ ops->control(drv, MIDI_CTL_PITCHBEND, chan);
+ break;
+ case SNDRV_SEQ_EVENT_CHANPRESS:
+ chan->midi_pressure = ev->data.control.value;
+ if (ops->control)
+ ops->control(drv, MIDI_CTL_CHAN_PRESSURE, chan);
+ break;
+ case SNDRV_SEQ_EVENT_CONTROL14:
+ /* Best guess is that this is any of the 14 bit controller values */
+ if (ev->data.control.param < 32) {
+ /* set low part first */
+ chan->control[ev->data.control.param + 32] =
+ ev->data.control.value & 0x7f;
+ do_control(ops, drv, chanset, chan,
+ ev->data.control.param,
+ ((ev->data.control.value>>7) & 0x7f));
+ } else
+ do_control(ops, drv, chanset, chan,
+ ev->data.control.param,
+ ev->data.control.value);
+ break;
+ case SNDRV_SEQ_EVENT_NONREGPARAM:
+ /* Break it back into its controller values */
+ chan->param_type = SNDRV_MIDI_PARAM_TYPE_NONREGISTERED;
+ chan->control[MIDI_CTL_MSB_DATA_ENTRY]
+ = (ev->data.control.value >> 7) & 0x7f;
+ chan->control[MIDI_CTL_LSB_DATA_ENTRY]
+ = ev->data.control.value & 0x7f;
+ chan->control[MIDI_CTL_NONREG_PARM_NUM_MSB]
+ = (ev->data.control.param >> 7) & 0x7f;
+ chan->control[MIDI_CTL_NONREG_PARM_NUM_LSB]
+ = ev->data.control.param & 0x7f;
+ nrpn(ops, drv, chan, chanset);
+ break;
+ case SNDRV_SEQ_EVENT_REGPARAM:
+ /* Break it back into its controller values */
+ chan->param_type = SNDRV_MIDI_PARAM_TYPE_REGISTERED;
+ chan->control[MIDI_CTL_MSB_DATA_ENTRY]
+ = (ev->data.control.value >> 7) & 0x7f;
+ chan->control[MIDI_CTL_LSB_DATA_ENTRY]
+ = ev->data.control.value & 0x7f;
+ chan->control[MIDI_CTL_REGIST_PARM_NUM_MSB]
+ = (ev->data.control.param >> 7) & 0x7f;
+ chan->control[MIDI_CTL_REGIST_PARM_NUM_LSB]
+ = ev->data.control.param & 0x7f;
+ rpn(ops, drv, chan, chanset);
+ break;
+ case SNDRV_SEQ_EVENT_SYSEX:
+ if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) == SNDRV_SEQ_EVENT_LENGTH_VARIABLE) {
+ unsigned char sysexbuf[64];
+ int len;
+ len = snd_seq_expand_var_event(ev, sizeof(sysexbuf), sysexbuf, 1, 0);
+ if (len > 0)
+ sysex(ops, drv, sysexbuf, len, chanset);
+ }
+ break;
+ case SNDRV_SEQ_EVENT_SONGPOS:
+ case SNDRV_SEQ_EVENT_SONGSEL:
+ case SNDRV_SEQ_EVENT_CLOCK:
+ case SNDRV_SEQ_EVENT_START:
+ case SNDRV_SEQ_EVENT_CONTINUE:
+ case SNDRV_SEQ_EVENT_STOP:
+ case SNDRV_SEQ_EVENT_QFRAME:
+ case SNDRV_SEQ_EVENT_TEMPO:
+ case SNDRV_SEQ_EVENT_TIMESIGN:
+ case SNDRV_SEQ_EVENT_KEYSIGN:
+ goto not_yet;
+ case SNDRV_SEQ_EVENT_SENSING:
+ break;
+ case SNDRV_SEQ_EVENT_CLIENT_START:
+ case SNDRV_SEQ_EVENT_CLIENT_EXIT:
+ case SNDRV_SEQ_EVENT_CLIENT_CHANGE:
+ case SNDRV_SEQ_EVENT_PORT_START:
+ case SNDRV_SEQ_EVENT_PORT_EXIT:
+ case SNDRV_SEQ_EVENT_PORT_CHANGE:
+ case SNDRV_SEQ_EVENT_ECHO:
+ not_yet:
+ default:
+ /*snd_printd("Unimplemented event %d\n", ev->type);*/
+ break;
+ }
+}
+
+
+/*
+ * release note
+ */
+static void
+note_off(struct snd_midi_op *ops, void *drv, struct snd_midi_channel *chan,
+ int note, int vel)
+{
+ if (chan->gm_hold) {
+ /* Hold this note until pedal is turned off */
+ chan->note[note] |= SNDRV_MIDI_NOTE_RELEASED;
+ } else if (chan->note[note] & SNDRV_MIDI_NOTE_SOSTENUTO) {
+ /* Mark this note as release; it will be turned off when sostenuto
+ * is turned off */
+ chan->note[note] |= SNDRV_MIDI_NOTE_RELEASED;
+ } else {
+ chan->note[note] = 0;
+ if (ops->note_off)
+ ops->note_off(drv, note, vel, chan);
+ }
+}
+
+/*
+ * Do all driver independent operations for this controller and pass
+ * events that need to take place immediately to the driver.
+ */
+static void
+do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chset,
+ struct snd_midi_channel *chan, int control, int value)
+{
+ int i;
+
+ /* Switches */
+ if ((control >=64 && control <=69) || (control >= 80 && control <= 83)) {
+ /* These are all switches; either off or on so set to 0 or 127 */
+ value = (value >= 64)? 127: 0;
+ }
+ chan->control[control] = value;
+
+ switch (control) {
+ case MIDI_CTL_SUSTAIN:
+ if (value == 0) {
+ /* Sustain has been released, turn off held notes */
+ for (i = 0; i < 128; i++) {
+ if (chan->note[i] & SNDRV_MIDI_NOTE_RELEASED) {
+ chan->note[i] = SNDRV_MIDI_NOTE_OFF;
+ if (ops->note_off)
+ ops->note_off(drv, i, 0, chan);
+ }
+ }
+ }
+ break;
+ case MIDI_CTL_PORTAMENTO:
+ break;
+ case MIDI_CTL_SOSTENUTO:
+ if (value) {
+ /* Mark each note that is currently held down */
+ for (i = 0; i < 128; i++) {
+ if (chan->note[i] & SNDRV_MIDI_NOTE_ON)
+ chan->note[i] |= SNDRV_MIDI_NOTE_SOSTENUTO;
+ }
+ } else {
+ /* release all notes that were held */
+ for (i = 0; i < 128; i++) {
+ if (chan->note[i] & SNDRV_MIDI_NOTE_SOSTENUTO) {
+ chan->note[i] &= ~SNDRV_MIDI_NOTE_SOSTENUTO;
+ if (chan->note[i] & SNDRV_MIDI_NOTE_RELEASED) {
+ chan->note[i] = SNDRV_MIDI_NOTE_OFF;
+ if (ops->note_off)
+ ops->note_off(drv, i, 0, chan);
+ }
+ }
+ }
+ }
+ break;
+ case MIDI_CTL_MSB_DATA_ENTRY:
+ chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0;
+ /* go through here */
+ case MIDI_CTL_LSB_DATA_ENTRY:
+ if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED)
+ rpn(ops, drv, chan, chset);
+ else
+ nrpn(ops, drv, chan, chset);
+ break;
+ case MIDI_CTL_REGIST_PARM_NUM_LSB:
+ case MIDI_CTL_REGIST_PARM_NUM_MSB:
+ chan->param_type = SNDRV_MIDI_PARAM_TYPE_REGISTERED;
+ break;
+ case MIDI_CTL_NONREG_PARM_NUM_LSB:
+ case MIDI_CTL_NONREG_PARM_NUM_MSB:
+ chan->param_type = SNDRV_MIDI_PARAM_TYPE_NONREGISTERED;
+ break;
+
+ case MIDI_CTL_ALL_SOUNDS_OFF:
+ all_sounds_off(ops, drv, chan);
+ break;
+
+ case MIDI_CTL_ALL_NOTES_OFF:
+ all_notes_off(ops, drv, chan);
+ break;
+
+ case MIDI_CTL_MSB_BANK:
+ if (chset->midi_mode == SNDRV_MIDI_MODE_XG) {
+ if (value == 127)
+ chan->drum_channel = 1;
+ else
+ chan->drum_channel = 0;
+ }
+ break;
+ case MIDI_CTL_LSB_BANK:
+ break;
+
+ case MIDI_CTL_RESET_CONTROLLERS:
+ snd_midi_reset_controllers(chan);
+ break;
+
+ case MIDI_CTL_SOFT_PEDAL:
+ case MIDI_CTL_LEGATO_FOOTSWITCH:
+ case MIDI_CTL_HOLD2:
+ case MIDI_CTL_SC1_SOUND_VARIATION:
+ case MIDI_CTL_SC2_TIMBRE:
+ case MIDI_CTL_SC3_RELEASE_TIME:
+ case MIDI_CTL_SC4_ATTACK_TIME:
+ case MIDI_CTL_SC5_BRIGHTNESS:
+ case MIDI_CTL_E1_REVERB_DEPTH:
+ case MIDI_CTL_E2_TREMOLO_DEPTH:
+ case MIDI_CTL_E3_CHORUS_DEPTH:
+ case MIDI_CTL_E4_DETUNE_DEPTH:
+ case MIDI_CTL_E5_PHASER_DEPTH:
+ goto notyet;
+ notyet:
+ default:
+ if (ops->control)
+ ops->control(drv, control, chan);
+ break;
+ }
+}
+
+
+/*
+ * initialize the MIDI status
+ */
+void
+snd_midi_channel_set_clear(struct snd_midi_channel_set *chset)
+{
+ int i;
+
+ chset->midi_mode = SNDRV_MIDI_MODE_GM;
+ chset->gs_master_volume = 127;
+
+ for (i = 0; i < chset->max_channels; i++) {
+ struct snd_midi_channel *chan = chset->channels + i;
+ memset(chan->note, 0, sizeof(chan->note));
+
+ chan->midi_aftertouch = 0;
+ chan->midi_pressure = 0;
+ chan->midi_program = 0;
+ chan->midi_pitchbend = 0;
+ snd_midi_reset_controllers(chan);
+ chan->gm_rpn_pitch_bend_range = 256; /* 2 semitones */
+ chan->gm_rpn_fine_tuning = 0;
+ chan->gm_rpn_coarse_tuning = 0;
+
+ if (i == 9)
+ chan->drum_channel = 1;
+ else
+ chan->drum_channel = 0;
+ }
+}
+
+/*
+ * Process a rpn message.
+ */
+static void
+rpn(struct snd_midi_op *ops, void *drv, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset)
+{
+ int type;
+ int val;
+
+ if (chset->midi_mode != SNDRV_MIDI_MODE_NONE) {
+ type = (chan->control[MIDI_CTL_REGIST_PARM_NUM_MSB] << 8) |
+ chan->control[MIDI_CTL_REGIST_PARM_NUM_LSB];
+ val = (chan->control[MIDI_CTL_MSB_DATA_ENTRY] << 7) |
+ chan->control[MIDI_CTL_LSB_DATA_ENTRY];
+
+ switch (type) {
+ case 0x0000: /* Pitch bend sensitivity */
+ /* MSB only / 1 semitone per 128 */
+ chan->gm_rpn_pitch_bend_range = val;
+ break;
+
+ case 0x0001: /* fine tuning: */
+ /* MSB/LSB, 8192=center, 100/8192 cent step */
+ chan->gm_rpn_fine_tuning = val - 8192;
+ break;
+
+ case 0x0002: /* coarse tuning */
+ /* MSB only / 8192=center, 1 semitone per 128 */
+ chan->gm_rpn_coarse_tuning = val - 8192;
+ break;
+
+ case 0x7F7F: /* "lock-in" RPN */
+ /* ignored */
+ break;
+ }
+ }
+ /* should call nrpn or rpn callback here.. */
+}
+
+/*
+ * Process an nrpn message.
+ */
+static void
+nrpn(struct snd_midi_op *ops, void *drv, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset)
+{
+ /* parse XG NRPNs here if possible */
+ if (ops->nrpn)
+ ops->nrpn(drv, chan, chset);
+}
+
+
+/*
+ * convert channel parameter in GS sysex
+ */
+static int
+get_channel(unsigned char cmd)
+{
+ int p = cmd & 0x0f;
+ if (p == 0)
+ p = 9;
+ else if (p < 10)
+ p--;
+ return p;
+}
+
+
+/*
+ * Process a sysex message.
+ */
+static void
+sysex(struct snd_midi_op *ops, void *private, unsigned char *buf, int len,
+ struct snd_midi_channel_set *chset)
+{
+ /* GM on */
+ static unsigned char gm_on_macro[] = {
+ 0x7e,0x7f,0x09,0x01,
+ };
+ /* XG on */
+ static unsigned char xg_on_macro[] = {
+ 0x43,0x10,0x4c,0x00,0x00,0x7e,0x00,
+ };
+ /* GS prefix
+ * drum channel: XX=0x1?(channel), YY=0x15, ZZ=on/off
+ * reverb mode: XX=0x01, YY=0x30, ZZ=0-7
+ * chorus mode: XX=0x01, YY=0x38, ZZ=0-7
+ * master vol: XX=0x00, YY=0x04, ZZ=0-127
+ */
+ static unsigned char gs_pfx_macro[] = {
+ 0x41,0x10,0x42,0x12,0x40,/*XX,YY,ZZ*/
+ };
+
+ int parsed = SNDRV_MIDI_SYSEX_NOT_PARSED;
+
+ if (len <= 0 || buf[0] != 0xf0)
+ return;
+ /* skip first byte */
+ buf++;
+ len--;
+
+ /* GM on */
+ if (len >= (int)sizeof(gm_on_macro) &&
+ memcmp(buf, gm_on_macro, sizeof(gm_on_macro)) == 0) {
+ if (chset->midi_mode != SNDRV_MIDI_MODE_GS &&
+ chset->midi_mode != SNDRV_MIDI_MODE_XG) {
+ chset->midi_mode = SNDRV_MIDI_MODE_GM;
+ reset_all_channels(chset);
+ parsed = SNDRV_MIDI_SYSEX_GM_ON;
+ }
+ }
+
+ /* GS macros */
+ else if (len >= 8 &&
+ memcmp(buf, gs_pfx_macro, sizeof(gs_pfx_macro)) == 0) {
+ if (chset->midi_mode != SNDRV_MIDI_MODE_GS &&
+ chset->midi_mode != SNDRV_MIDI_MODE_XG)
+ chset->midi_mode = SNDRV_MIDI_MODE_GS;
+
+ if (buf[5] == 0x00 && buf[6] == 0x7f && buf[7] == 0x00) {
+ /* GS reset */
+ parsed = SNDRV_MIDI_SYSEX_GS_RESET;
+ reset_all_channels(chset);
+ }
+
+ else if ((buf[5] & 0xf0) == 0x10 && buf[6] == 0x15) {
+ /* drum pattern */
+ int p = get_channel(buf[5]);
+ if (p < chset->max_channels) {
+ parsed = SNDRV_MIDI_SYSEX_GS_DRUM_CHANNEL;
+ if (buf[7])
+ chset->channels[p].drum_channel = 1;
+ else
+ chset->channels[p].drum_channel = 0;
+ }
+
+ } else if ((buf[5] & 0xf0) == 0x10 && buf[6] == 0x21) {
+ /* program */
+ int p = get_channel(buf[5]);
+ if (p < chset->max_channels &&
+ ! chset->channels[p].drum_channel) {
+ parsed = SNDRV_MIDI_SYSEX_GS_DRUM_CHANNEL;
+ chset->channels[p].midi_program = buf[7];
+ }
+
+ } else if (buf[5] == 0x01 && buf[6] == 0x30) {
+ /* reverb mode */
+ parsed = SNDRV_MIDI_SYSEX_GS_REVERB_MODE;
+ chset->gs_reverb_mode = buf[7];
+
+ } else if (buf[5] == 0x01 && buf[6] == 0x38) {
+ /* chorus mode */
+ parsed = SNDRV_MIDI_SYSEX_GS_CHORUS_MODE;
+ chset->gs_chorus_mode = buf[7];
+
+ } else if (buf[5] == 0x00 && buf[6] == 0x04) {
+ /* master volume */
+ parsed = SNDRV_MIDI_SYSEX_GS_MASTER_VOLUME;
+ chset->gs_master_volume = buf[7];
+
+ }
+ }
+
+ /* XG on */
+ else if (len >= (int)sizeof(xg_on_macro) &&
+ memcmp(buf, xg_on_macro, sizeof(xg_on_macro)) == 0) {
+ int i;
+ chset->midi_mode = SNDRV_MIDI_MODE_XG;
+ parsed = SNDRV_MIDI_SYSEX_XG_ON;
+ /* reset CC#0 for drums */
+ for (i = 0; i < chset->max_channels; i++) {
+ if (chset->channels[i].drum_channel)
+ chset->channels[i].control[MIDI_CTL_MSB_BANK] = 127;
+ else
+ chset->channels[i].control[MIDI_CTL_MSB_BANK] = 0;
+ }
+ }
+
+ if (ops->sysex)
+ ops->sysex(private, buf - 1, len + 1, parsed, chset);
+}
+
+/*
+ * all sound off
+ */
+static void
+all_sounds_off(struct snd_midi_op *ops, void *drv, struct snd_midi_channel *chan)
+{
+ int n;
+
+ if (! ops->note_terminate)
+ return;
+ for (n = 0; n < 128; n++) {
+ if (chan->note[n]) {
+ ops->note_terminate(drv, n, chan);
+ chan->note[n] = 0;
+ }
+ }
+}
+
+/*
+ * all notes off
+ */
+static void
+all_notes_off(struct snd_midi_op *ops, void *drv, struct snd_midi_channel *chan)
+{
+ int n;
+
+ if (! ops->note_off)
+ return;
+ for (n = 0; n < 128; n++) {
+ if (chan->note[n] == SNDRV_MIDI_NOTE_ON)
+ note_off(ops, drv, chan, n, 0);
+ }
+}
+
+/*
+ * Initialise a single midi channel control block.
+ */
+static void snd_midi_channel_init(struct snd_midi_channel *p, int n)
+{
+ if (p == NULL)
+ return;
+
+ memset(p, 0, sizeof(struct snd_midi_channel));
+ p->private = NULL;
+ p->number = n;
+
+ snd_midi_reset_controllers(p);
+ p->gm_rpn_pitch_bend_range = 256; /* 2 semitones */
+ p->gm_rpn_fine_tuning = 0;
+ p->gm_rpn_coarse_tuning = 0;
+
+ if (n == 9)
+ p->drum_channel = 1; /* Default ch 10 as drums */
+}
+
+/*
+ * Allocate and initialise a set of midi channel control blocks.
+ */
+static struct snd_midi_channel *snd_midi_channel_init_set(int n)
+{
+ struct snd_midi_channel *chan;
+ int i;
+
+ chan = kmalloc(n * sizeof(struct snd_midi_channel), GFP_KERNEL);
+ if (chan) {
+ for (i = 0; i < n; i++)
+ snd_midi_channel_init(chan+i, i);
+ }
+
+ return chan;
+}
+
+/*
+ * reset all midi channels
+ */
+static void
+reset_all_channels(struct snd_midi_channel_set *chset)
+{
+ int ch;
+ for (ch = 0; ch < chset->max_channels; ch++) {
+ struct snd_midi_channel *chan = chset->channels + ch;
+ snd_midi_reset_controllers(chan);
+ chan->gm_rpn_pitch_bend_range = 256; /* 2 semitones */
+ chan->gm_rpn_fine_tuning = 0;
+ chan->gm_rpn_coarse_tuning = 0;
+
+ if (ch == 9)
+ chan->drum_channel = 1;
+ else
+ chan->drum_channel = 0;
+ }
+}
+
+
+/*
+ * Allocate and initialise a midi channel set.
+ */
+struct snd_midi_channel_set *snd_midi_channel_alloc_set(int n)
+{
+ struct snd_midi_channel_set *chset;
+
+ chset = kmalloc(sizeof(*chset), GFP_KERNEL);
+ if (chset) {
+ chset->channels = snd_midi_channel_init_set(n);
+ chset->private_data = NULL;
+ chset->max_channels = n;
+ }
+ return chset;
+}
+
+/*
+ * Reset the midi controllers on a particular channel to default values.
+ */
+static void snd_midi_reset_controllers(struct snd_midi_channel *chan)
+{
+ memset(chan->control, 0, sizeof(chan->control));
+ chan->gm_volume = 127;
+ chan->gm_expression = 127;
+ chan->gm_pan = 64;
+}
+
+
+/*
+ * Free a midi channel set.
+ */
+void snd_midi_channel_free_set(struct snd_midi_channel_set *chset)
+{
+ if (chset == NULL)
+ return;
+ kfree(chset->channels);
+ kfree(chset);
+}
+
+static int __init alsa_seq_midi_emul_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_seq_midi_emul_exit(void)
+{
+}
+
+module_init(alsa_seq_midi_emul_init)
+module_exit(alsa_seq_midi_emul_exit)
+
+EXPORT_SYMBOL(snd_midi_process_event);
+EXPORT_SYMBOL(snd_midi_channel_set_clear);
+EXPORT_SYMBOL(snd_midi_channel_alloc_set);
+EXPORT_SYMBOL(snd_midi_channel_free_set);
diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c
new file mode 100644
index 0000000..8284f17
--- /dev/null
+++ b/sound/core/seq/seq_midi_event.c
@@ -0,0 +1,549 @@
+/*
+ * MIDI byte <-> sequencer event coder
+ *
+ * Copyright (C) 1998,99 Takashi Iwai <tiwai@suse.de>,
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/seq_kernel.h>
+#include <sound/seq_midi_event.h>
+#include <sound/asoundef.h>
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("MIDI byte <-> sequencer event coder");
+MODULE_LICENSE("GPL");
+
+/* event type, index into status_event[] */
+/* from 0 to 6 are normal commands (note off, on, etc.) for 0x9?-0xe? */
+#define ST_INVALID 7
+#define ST_SPECIAL 8
+#define ST_SYSEX ST_SPECIAL
+/* from 8 to 15 are events for 0xf0-0xf7 */
+
+
+/*
+ * prototypes
+ */
+static void note_event(struct snd_midi_event *dev, struct snd_seq_event *ev);
+static void one_param_ctrl_event(struct snd_midi_event *dev, struct snd_seq_event *ev);
+static void pitchbend_ctrl_event(struct snd_midi_event *dev, struct snd_seq_event *ev);
+static void two_param_ctrl_event(struct snd_midi_event *dev, struct snd_seq_event *ev);
+static void one_param_event(struct snd_midi_event *dev, struct snd_seq_event *ev);
+static void songpos_event(struct snd_midi_event *dev, struct snd_seq_event *ev);
+static void note_decode(struct snd_seq_event *ev, unsigned char *buf);
+static void one_param_decode(struct snd_seq_event *ev, unsigned char *buf);
+static void pitchbend_decode(struct snd_seq_event *ev, unsigned char *buf);
+static void two_param_decode(struct snd_seq_event *ev, unsigned char *buf);
+static void songpos_decode(struct snd_seq_event *ev, unsigned char *buf);
+
+/*
+ * event list
+ */
+static struct status_event_list {
+ int event;
+ int qlen;
+ void (*encode)(struct snd_midi_event *dev, struct snd_seq_event *ev);
+ void (*decode)(struct snd_seq_event *ev, unsigned char *buf);
+} status_event[] = {
+ /* 0x80 - 0xef */
+ {SNDRV_SEQ_EVENT_NOTEOFF, 2, note_event, note_decode},
+ {SNDRV_SEQ_EVENT_NOTEON, 2, note_event, note_decode},
+ {SNDRV_SEQ_EVENT_KEYPRESS, 2, note_event, note_decode},
+ {SNDRV_SEQ_EVENT_CONTROLLER, 2, two_param_ctrl_event, two_param_decode},
+ {SNDRV_SEQ_EVENT_PGMCHANGE, 1, one_param_ctrl_event, one_param_decode},
+ {SNDRV_SEQ_EVENT_CHANPRESS, 1, one_param_ctrl_event, one_param_decode},
+ {SNDRV_SEQ_EVENT_PITCHBEND, 2, pitchbend_ctrl_event, pitchbend_decode},
+ /* invalid */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL},
+ /* 0xf0 - 0xff */
+ {SNDRV_SEQ_EVENT_SYSEX, 1, NULL, NULL}, /* sysex: 0xf0 */
+ {SNDRV_SEQ_EVENT_QFRAME, 1, one_param_event, one_param_decode}, /* 0xf1 */
+ {SNDRV_SEQ_EVENT_SONGPOS, 2, songpos_event, songpos_decode}, /* 0xf2 */
+ {SNDRV_SEQ_EVENT_SONGSEL, 1, one_param_event, one_param_decode}, /* 0xf3 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf4 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf5 */
+ {SNDRV_SEQ_EVENT_TUNE_REQUEST, 0, NULL, NULL}, /* 0xf6 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf7 */
+ {SNDRV_SEQ_EVENT_CLOCK, 0, NULL, NULL}, /* 0xf8 */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xf9 */
+ {SNDRV_SEQ_EVENT_START, 0, NULL, NULL}, /* 0xfa */
+ {SNDRV_SEQ_EVENT_CONTINUE, 0, NULL, NULL}, /* 0xfb */
+ {SNDRV_SEQ_EVENT_STOP, 0, NULL, NULL}, /* 0xfc */
+ {SNDRV_SEQ_EVENT_NONE, -1, NULL, NULL}, /* 0xfd */
+ {SNDRV_SEQ_EVENT_SENSING, 0, NULL, NULL}, /* 0xfe */
+ {SNDRV_SEQ_EVENT_RESET, 0, NULL, NULL}, /* 0xff */
+};
+
+static int extra_decode_ctrl14(struct snd_midi_event *dev, unsigned char *buf, int len,
+ struct snd_seq_event *ev);
+static int extra_decode_xrpn(struct snd_midi_event *dev, unsigned char *buf, int count,
+ struct snd_seq_event *ev);
+
+static struct extra_event_list {
+ int event;
+ int (*decode)(struct snd_midi_event *dev, unsigned char *buf, int len,
+ struct snd_seq_event *ev);
+} extra_event[] = {
+ {SNDRV_SEQ_EVENT_CONTROL14, extra_decode_ctrl14},
+ {SNDRV_SEQ_EVENT_NONREGPARAM, extra_decode_xrpn},
+ {SNDRV_SEQ_EVENT_REGPARAM, extra_decode_xrpn},
+};
+
+/*
+ * new/delete record
+ */
+
+int snd_midi_event_new(int bufsize, struct snd_midi_event **rdev)
+{
+ struct snd_midi_event *dev;
+
+ *rdev = NULL;
+ dev = kzalloc(sizeof(*dev), GFP_KERNEL);
+ if (dev == NULL)
+ return -ENOMEM;
+ if (bufsize > 0) {
+ dev->buf = kmalloc(bufsize, GFP_KERNEL);
+ if (dev->buf == NULL) {
+ kfree(dev);
+ return -ENOMEM;
+ }
+ }
+ dev->bufsize = bufsize;
+ dev->lastcmd = 0xff;
+ dev->type = ST_INVALID;
+ spin_lock_init(&dev->lock);
+ *rdev = dev;
+ return 0;
+}
+
+void snd_midi_event_free(struct snd_midi_event *dev)
+{
+ if (dev != NULL) {
+ kfree(dev->buf);
+ kfree(dev);
+ }
+}
+
+/*
+ * initialize record
+ */
+static inline void reset_encode(struct snd_midi_event *dev)
+{
+ dev->read = 0;
+ dev->qlen = 0;
+ dev->type = ST_INVALID;
+}
+
+void snd_midi_event_reset_encode(struct snd_midi_event *dev)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&dev->lock, flags);
+ reset_encode(dev);
+ spin_unlock_irqrestore(&dev->lock, flags);
+}
+
+void snd_midi_event_reset_decode(struct snd_midi_event *dev)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&dev->lock, flags);
+ dev->lastcmd = 0xff;
+ spin_unlock_irqrestore(&dev->lock, flags);
+}
+
+#if 0
+void snd_midi_event_init(struct snd_midi_event *dev)
+{
+ snd_midi_event_reset_encode(dev);
+ snd_midi_event_reset_decode(dev);
+}
+#endif /* 0 */
+
+void snd_midi_event_no_status(struct snd_midi_event *dev, int on)
+{
+ dev->nostat = on ? 1 : 0;
+}
+
+/*
+ * resize buffer
+ */
+#if 0
+int snd_midi_event_resize_buffer(struct snd_midi_event *dev, int bufsize)
+{
+ unsigned char *new_buf, *old_buf;
+ unsigned long flags;
+
+ if (bufsize == dev->bufsize)
+ return 0;
+ new_buf = kmalloc(bufsize, GFP_KERNEL);
+ if (new_buf == NULL)
+ return -ENOMEM;
+ spin_lock_irqsave(&dev->lock, flags);
+ old_buf = dev->buf;
+ dev->buf = new_buf;
+ dev->bufsize = bufsize;
+ reset_encode(dev);
+ spin_unlock_irqrestore(&dev->lock, flags);
+ kfree(old_buf);
+ return 0;
+}
+#endif /* 0 */
+
+/*
+ * read bytes and encode to sequencer event if finished
+ * return the size of encoded bytes
+ */
+long snd_midi_event_encode(struct snd_midi_event *dev, unsigned char *buf, long count,
+ struct snd_seq_event *ev)
+{
+ long result = 0;
+ int rc;
+
+ ev->type = SNDRV_SEQ_EVENT_NONE;
+
+ while (count-- > 0) {
+ rc = snd_midi_event_encode_byte(dev, *buf++, ev);
+ result++;
+ if (rc < 0)
+ return rc;
+ else if (rc > 0)
+ return result;
+ }
+
+ return result;
+}
+
+/*
+ * read one byte and encode to sequencer event:
+ * return 1 if MIDI bytes are encoded to an event
+ * 0 data is not finished
+ * negative for error
+ */
+int snd_midi_event_encode_byte(struct snd_midi_event *dev, int c,
+ struct snd_seq_event *ev)
+{
+ int rc = 0;
+ unsigned long flags;
+
+ c &= 0xff;
+
+ if (c >= MIDI_CMD_COMMON_CLOCK) {
+ /* real-time event */
+ ev->type = status_event[ST_SPECIAL + c - 0xf0].event;
+ ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK;
+ ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED;
+ return ev->type != SNDRV_SEQ_EVENT_NONE;
+ }
+
+ spin_lock_irqsave(&dev->lock, flags);
+ if ((c & 0x80) &&
+ (c != MIDI_CMD_COMMON_SYSEX_END || dev->type != ST_SYSEX)) {
+ /* new command */
+ dev->buf[0] = c;
+ if ((c & 0xf0) == 0xf0) /* system messages */
+ dev->type = (c & 0x0f) + ST_SPECIAL;
+ else
+ dev->type = (c >> 4) & 0x07;
+ dev->read = 1;
+ dev->qlen = status_event[dev->type].qlen;
+ } else {
+ if (dev->qlen > 0) {
+ /* rest of command */
+ dev->buf[dev->read++] = c;
+ if (dev->type != ST_SYSEX)
+ dev->qlen--;
+ } else {
+ /* running status */
+ dev->buf[1] = c;
+ dev->qlen = status_event[dev->type].qlen - 1;
+ dev->read = 2;
+ }
+ }
+ if (dev->qlen == 0) {
+ ev->type = status_event[dev->type].event;
+ ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK;
+ ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED;
+ if (status_event[dev->type].encode) /* set data values */
+ status_event[dev->type].encode(dev, ev);
+ if (dev->type >= ST_SPECIAL)
+ dev->type = ST_INVALID;
+ rc = 1;
+ } else if (dev->type == ST_SYSEX) {
+ if (c == MIDI_CMD_COMMON_SYSEX_END ||
+ dev->read >= dev->bufsize) {
+ ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK;
+ ev->flags |= SNDRV_SEQ_EVENT_LENGTH_VARIABLE;
+ ev->type = SNDRV_SEQ_EVENT_SYSEX;
+ ev->data.ext.len = dev->read;
+ ev->data.ext.ptr = dev->buf;
+ if (c != MIDI_CMD_COMMON_SYSEX_END)
+ dev->read = 0; /* continue to parse */
+ else
+ reset_encode(dev); /* all parsed */
+ rc = 1;
+ }
+ }
+
+ spin_unlock_irqrestore(&dev->lock, flags);
+ return rc;
+}
+
+/* encode note event */
+static void note_event(struct snd_midi_event *dev, struct snd_seq_event *ev)
+{
+ ev->data.note.channel = dev->buf[0] & 0x0f;
+ ev->data.note.note = dev->buf[1];
+ ev->data.note.velocity = dev->buf[2];
+}
+
+/* encode one parameter controls */
+static void one_param_ctrl_event(struct snd_midi_event *dev, struct snd_seq_event *ev)
+{
+ ev->data.control.channel = dev->buf[0] & 0x0f;
+ ev->data.control.value = dev->buf[1];
+}
+
+/* encode pitch wheel change */
+static void pitchbend_ctrl_event(struct snd_midi_event *dev, struct snd_seq_event *ev)
+{
+ ev->data.control.channel = dev->buf[0] & 0x0f;
+ ev->data.control.value = (int)dev->buf[2] * 128 + (int)dev->buf[1] - 8192;
+}
+
+/* encode midi control change */
+static void two_param_ctrl_event(struct snd_midi_event *dev, struct snd_seq_event *ev)
+{
+ ev->data.control.channel = dev->buf[0] & 0x0f;
+ ev->data.control.param = dev->buf[1];
+ ev->data.control.value = dev->buf[2];
+}
+
+/* encode one parameter value*/
+static void one_param_event(struct snd_midi_event *dev, struct snd_seq_event *ev)
+{
+ ev->data.control.value = dev->buf[1];
+}
+
+/* encode song position */
+static void songpos_event(struct snd_midi_event *dev, struct snd_seq_event *ev)
+{
+ ev->data.control.value = (int)dev->buf[2] * 128 + (int)dev->buf[1];
+}
+
+/*
+ * decode from a sequencer event to midi bytes
+ * return the size of decoded midi events
+ */
+long snd_midi_event_decode(struct snd_midi_event *dev, unsigned char *buf, long count,
+ struct snd_seq_event *ev)
+{
+ unsigned int cmd, type;
+
+ if (ev->type == SNDRV_SEQ_EVENT_NONE)
+ return -ENOENT;
+
+ for (type = 0; type < ARRAY_SIZE(status_event); type++) {
+ if (ev->type == status_event[type].event)
+ goto __found;
+ }
+ for (type = 0; type < ARRAY_SIZE(extra_event); type++) {
+ if (ev->type == extra_event[type].event)
+ return extra_event[type].decode(dev, buf, count, ev);
+ }
+ return -ENOENT;
+
+ __found:
+ if (type >= ST_SPECIAL)
+ cmd = 0xf0 + (type - ST_SPECIAL);
+ else
+ /* data.note.channel and data.control.channel is identical */
+ cmd = 0x80 | (type << 4) | (ev->data.note.channel & 0x0f);
+
+
+ if (cmd == MIDI_CMD_COMMON_SYSEX) {
+ snd_midi_event_reset_decode(dev);
+ return snd_seq_expand_var_event(ev, count, buf, 1, 0);
+ } else {
+ int qlen;
+ unsigned char xbuf[4];
+ unsigned long flags;
+
+ spin_lock_irqsave(&dev->lock, flags);
+ if ((cmd & 0xf0) == 0xf0 || dev->lastcmd != cmd || dev->nostat) {
+ dev->lastcmd = cmd;
+ spin_unlock_irqrestore(&dev->lock, flags);
+ xbuf[0] = cmd;
+ if (status_event[type].decode)
+ status_event[type].decode(ev, xbuf + 1);
+ qlen = status_event[type].qlen + 1;
+ } else {
+ spin_unlock_irqrestore(&dev->lock, flags);
+ if (status_event[type].decode)
+ status_event[type].decode(ev, xbuf + 0);
+ qlen = status_event[type].qlen;
+ }
+ if (count < qlen)
+ return -ENOMEM;
+ memcpy(buf, xbuf, qlen);
+ return qlen;
+ }
+}
+
+
+/* decode note event */
+static void note_decode(struct snd_seq_event *ev, unsigned char *buf)
+{
+ buf[0] = ev->data.note.note & 0x7f;
+ buf[1] = ev->data.note.velocity & 0x7f;
+}
+
+/* decode one parameter controls */
+static void one_param_decode(struct snd_seq_event *ev, unsigned char *buf)
+{
+ buf[0] = ev->data.control.value & 0x7f;
+}
+
+/* decode pitch wheel change */
+static void pitchbend_decode(struct snd_seq_event *ev, unsigned char *buf)
+{
+ int value = ev->data.control.value + 8192;
+ buf[0] = value & 0x7f;
+ buf[1] = (value >> 7) & 0x7f;
+}
+
+/* decode midi control change */
+static void two_param_decode(struct snd_seq_event *ev, unsigned char *buf)
+{
+ buf[0] = ev->data.control.param & 0x7f;
+ buf[1] = ev->data.control.value & 0x7f;
+}
+
+/* decode song position */
+static void songpos_decode(struct snd_seq_event *ev, unsigned char *buf)
+{
+ buf[0] = ev->data.control.value & 0x7f;
+ buf[1] = (ev->data.control.value >> 7) & 0x7f;
+}
+
+/* decode 14bit control */
+static int extra_decode_ctrl14(struct snd_midi_event *dev, unsigned char *buf,
+ int count, struct snd_seq_event *ev)
+{
+ unsigned char cmd;
+ int idx = 0;
+
+ cmd = MIDI_CMD_CONTROL|(ev->data.control.channel & 0x0f);
+ if (ev->data.control.param < 0x20) {
+ if (count < 4)
+ return -ENOMEM;
+ if (dev->nostat && count < 6)
+ return -ENOMEM;
+ if (cmd != dev->lastcmd || dev->nostat) {
+ if (count < 5)
+ return -ENOMEM;
+ buf[idx++] = dev->lastcmd = cmd;
+ }
+ buf[idx++] = ev->data.control.param;
+ buf[idx++] = (ev->data.control.value >> 7) & 0x7f;
+ if (dev->nostat)
+ buf[idx++] = cmd;
+ buf[idx++] = ev->data.control.param + 0x20;
+ buf[idx++] = ev->data.control.value & 0x7f;
+ } else {
+ if (count < 2)
+ return -ENOMEM;
+ if (cmd != dev->lastcmd || dev->nostat) {
+ if (count < 3)
+ return -ENOMEM;
+ buf[idx++] = dev->lastcmd = cmd;
+ }
+ buf[idx++] = ev->data.control.param & 0x7f;
+ buf[idx++] = ev->data.control.value & 0x7f;
+ }
+ return idx;
+}
+
+/* decode reg/nonreg param */
+static int extra_decode_xrpn(struct snd_midi_event *dev, unsigned char *buf,
+ int count, struct snd_seq_event *ev)
+{
+ unsigned char cmd;
+ char *cbytes;
+ static char cbytes_nrpn[4] = { MIDI_CTL_NONREG_PARM_NUM_MSB,
+ MIDI_CTL_NONREG_PARM_NUM_LSB,
+ MIDI_CTL_MSB_DATA_ENTRY,
+ MIDI_CTL_LSB_DATA_ENTRY };
+ static char cbytes_rpn[4] = { MIDI_CTL_REGIST_PARM_NUM_MSB,
+ MIDI_CTL_REGIST_PARM_NUM_LSB,
+ MIDI_CTL_MSB_DATA_ENTRY,
+ MIDI_CTL_LSB_DATA_ENTRY };
+ unsigned char bytes[4];
+ int idx = 0, i;
+
+ if (count < 8)
+ return -ENOMEM;
+ if (dev->nostat && count < 12)
+ return -ENOMEM;
+ cmd = MIDI_CMD_CONTROL|(ev->data.control.channel & 0x0f);
+ bytes[0] = ev->data.control.param & 0x007f;
+ bytes[1] = (ev->data.control.param & 0x3f80) >> 7;
+ bytes[2] = ev->data.control.value & 0x007f;
+ bytes[3] = (ev->data.control.value & 0x3f80) >> 7;
+ if (cmd != dev->lastcmd && !dev->nostat) {
+ if (count < 9)
+ return -ENOMEM;
+ buf[idx++] = dev->lastcmd = cmd;
+ }
+ cbytes = ev->type == SNDRV_SEQ_EVENT_NONREGPARAM ? cbytes_nrpn : cbytes_rpn;
+ for (i = 0; i < 4; i++) {
+ if (dev->nostat)
+ buf[idx++] = dev->lastcmd = cmd;
+ buf[idx++] = cbytes[i];
+ buf[idx++] = bytes[i];
+ }
+ return idx;
+}
+
+/*
+ * exports
+ */
+
+EXPORT_SYMBOL(snd_midi_event_new);
+EXPORT_SYMBOL(snd_midi_event_free);
+EXPORT_SYMBOL(snd_midi_event_reset_encode);
+EXPORT_SYMBOL(snd_midi_event_reset_decode);
+EXPORT_SYMBOL(snd_midi_event_no_status);
+EXPORT_SYMBOL(snd_midi_event_encode);
+EXPORT_SYMBOL(snd_midi_event_encode_byte);
+EXPORT_SYMBOL(snd_midi_event_decode);
+
+static int __init alsa_seq_midi_event_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_seq_midi_event_exit(void)
+{
+}
+
+module_init(alsa_seq_midi_event_init)
+module_exit(alsa_seq_midi_event_exit)
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
new file mode 100644
index 0000000..3bf7d73
--- /dev/null
+++ b/sound/core/seq/seq_ports.c
@@ -0,0 +1,684 @@
+/*
+ * ALSA sequencer Ports
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <linux/slab.h>
+#include "seq_system.h"
+#include "seq_ports.h"
+#include "seq_clientmgr.h"
+
+/*
+
+ registration of client ports
+
+ */
+
+
+/*
+
+NOTE: the current implementation of the port structure as a linked list is
+not optimal for clients that have many ports. For sending messages to all
+subscribers of a port we first need to find the address of the port
+structure, which means we have to traverse the list. A direct access table
+(array) would be better, but big preallocated arrays waste memory.
+
+Possible actions:
+
+1) leave it this way, a client does normaly does not have more than a few
+ports
+
+2) replace the linked list of ports by a array of pointers which is
+dynamicly kmalloced. When a port is added or deleted we can simply allocate
+a new array, copy the corresponding pointers, and delete the old one. We
+then only need a pointer to this array, and an integer that tells us how
+much elements are in array.
+
+*/
+
+/* return pointer to port structure - port is locked if found */
+struct snd_seq_client_port *snd_seq_port_use_ptr(struct snd_seq_client *client,
+ int num)
+{
+ struct snd_seq_client_port *port;
+
+ if (client == NULL)
+ return NULL;
+ read_lock(&client->ports_lock);
+ list_for_each_entry(port, &client->ports_list_head, list) {
+ if (port->addr.port == num) {
+ if (port->closing)
+ break; /* deleting now */
+ snd_use_lock_use(&port->use_lock);
+ read_unlock(&client->ports_lock);
+ return port;
+ }
+ }
+ read_unlock(&client->ports_lock);
+ return NULL; /* not found */
+}
+
+
+/* search for the next port - port is locked if found */
+struct snd_seq_client_port *snd_seq_port_query_nearest(struct snd_seq_client *client,
+ struct snd_seq_port_info *pinfo)
+{
+ int num;
+ struct snd_seq_client_port *port, *found;
+
+ num = pinfo->addr.port;
+ found = NULL;
+ read_lock(&client->ports_lock);
+ list_for_each_entry(port, &client->ports_list_head, list) {
+ if (port->addr.port < num)
+ continue;
+ if (port->addr.port == num) {
+ found = port;
+ break;
+ }
+ if (found == NULL || port->addr.port < found->addr.port)
+ found = port;
+ }
+ if (found) {
+ if (found->closing)
+ found = NULL;
+ else
+ snd_use_lock_use(&found->use_lock);
+ }
+ read_unlock(&client->ports_lock);
+ return found;
+}
+
+
+/* initialize snd_seq_port_subs_info */
+static void port_subs_info_init(struct snd_seq_port_subs_info *grp)
+{
+ INIT_LIST_HEAD(&grp->list_head);
+ grp->count = 0;
+ grp->exclusive = 0;
+ rwlock_init(&grp->list_lock);
+ init_rwsem(&grp->list_mutex);
+ grp->open = NULL;
+ grp->close = NULL;
+}
+
+
+/* create a port, port number is returned (-1 on failure) */
+struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client,
+ int port)
+{
+ unsigned long flags;
+ struct snd_seq_client_port *new_port, *p;
+ int num = -1;
+
+ /* sanity check */
+ if (snd_BUG_ON(!client))
+ return NULL;
+
+ if (client->num_ports >= SNDRV_SEQ_MAX_PORTS - 1) {
+ snd_printk(KERN_WARNING "too many ports for client %d\n", client->number);
+ return NULL;
+ }
+
+ /* create a new port */
+ new_port = kzalloc(sizeof(*new_port), GFP_KERNEL);
+ if (! new_port) {
+ snd_printd("malloc failed for registering client port\n");
+ return NULL; /* failure, out of memory */
+ }
+ /* init port data */
+ new_port->addr.client = client->number;
+ new_port->addr.port = -1;
+ new_port->owner = THIS_MODULE;
+ sprintf(new_port->name, "port-%d", num);
+ snd_use_lock_init(&new_port->use_lock);
+ port_subs_info_init(&new_port->c_src);
+ port_subs_info_init(&new_port->c_dest);
+
+ num = port >= 0 ? port : 0;
+ mutex_lock(&client->ports_mutex);
+ write_lock_irqsave(&client->ports_lock, flags);
+ list_for_each_entry(p, &client->ports_list_head, list) {
+ if (p->addr.port > num)
+ break;
+ if (port < 0) /* auto-probe mode */
+ num = p->addr.port + 1;
+ }
+ /* insert the new port */
+ list_add_tail(&new_port->list, &p->list);
+ client->num_ports++;
+ new_port->addr.port = num; /* store the port number in the port */
+ write_unlock_irqrestore(&client->ports_lock, flags);
+ mutex_unlock(&client->ports_mutex);
+ sprintf(new_port->name, "port-%d", num);
+
+ return new_port;
+}
+
+/* */
+enum group_type {
+ SRC_LIST, DEST_LIST
+};
+
+static int subscribe_port(struct snd_seq_client *client,
+ struct snd_seq_client_port *port,
+ struct snd_seq_port_subs_info *grp,
+ struct snd_seq_port_subscribe *info, int send_ack);
+static int unsubscribe_port(struct snd_seq_client *client,
+ struct snd_seq_client_port *port,
+ struct snd_seq_port_subs_info *grp,
+ struct snd_seq_port_subscribe *info, int send_ack);
+
+
+static struct snd_seq_client_port *get_client_port(struct snd_seq_addr *addr,
+ struct snd_seq_client **cp)
+{
+ struct snd_seq_client_port *p;
+ *cp = snd_seq_client_use_ptr(addr->client);
+ if (*cp) {
+ p = snd_seq_port_use_ptr(*cp, addr->port);
+ if (! p) {
+ snd_seq_client_unlock(*cp);
+ *cp = NULL;
+ }
+ return p;
+ }
+ return NULL;
+}
+
+/*
+ * remove all subscribers on the list
+ * this is called from port_delete, for each src and dest list.
+ */
+static void clear_subscriber_list(struct snd_seq_client *client,
+ struct snd_seq_client_port *port,
+ struct snd_seq_port_subs_info *grp,
+ int grptype)
+{
+ struct list_head *p, *n;
+
+ list_for_each_safe(p, n, &grp->list_head) {
+ struct snd_seq_subscribers *subs;
+ struct snd_seq_client *c;
+ struct snd_seq_client_port *aport;
+
+ if (grptype == SRC_LIST) {
+ subs = list_entry(p, struct snd_seq_subscribers, src_list);
+ aport = get_client_port(&subs->info.dest, &c);
+ } else {
+ subs = list_entry(p, struct snd_seq_subscribers, dest_list);
+ aport = get_client_port(&subs->info.sender, &c);
+ }
+ list_del(p);
+ unsubscribe_port(client, port, grp, &subs->info, 0);
+ if (!aport) {
+ /* looks like the connected port is being deleted.
+ * we decrease the counter, and when both ports are deleted
+ * remove the subscriber info
+ */
+ if (atomic_dec_and_test(&subs->ref_count))
+ kfree(subs);
+ } else {
+ /* ok we got the connected port */
+ struct snd_seq_port_subs_info *agrp;
+ agrp = (grptype == SRC_LIST) ? &aport->c_dest : &aport->c_src;
+ down_write(&agrp->list_mutex);
+ if (grptype == SRC_LIST)
+ list_del(&subs->dest_list);
+ else
+ list_del(&subs->src_list);
+ up_write(&agrp->list_mutex);
+ unsubscribe_port(c, aport, agrp, &subs->info, 1);
+ kfree(subs);
+ snd_seq_port_unlock(aport);
+ snd_seq_client_unlock(c);
+ }
+ }
+}
+
+/* delete port data */
+static int port_delete(struct snd_seq_client *client,
+ struct snd_seq_client_port *port)
+{
+ /* set closing flag and wait for all port access are gone */
+ port->closing = 1;
+ snd_use_lock_sync(&port->use_lock);
+
+ /* clear subscribers info */
+ clear_subscriber_list(client, port, &port->c_src, SRC_LIST);
+ clear_subscriber_list(client, port, &port->c_dest, DEST_LIST);
+
+ if (port->private_free)
+ port->private_free(port->private_data);
+
+ snd_BUG_ON(port->c_src.count != 0);
+ snd_BUG_ON(port->c_dest.count != 0);
+
+ kfree(port);
+ return 0;
+}
+
+
+/* delete a port with the given port id */
+int snd_seq_delete_port(struct snd_seq_client *client, int port)
+{
+ unsigned long flags;
+ struct snd_seq_client_port *found = NULL, *p;
+
+ mutex_lock(&client->ports_mutex);
+ write_lock_irqsave(&client->ports_lock, flags);
+ list_for_each_entry(p, &client->ports_list_head, list) {
+ if (p->addr.port == port) {
+ /* ok found. delete from the list at first */
+ list_del(&p->list);
+ client->num_ports--;
+ found = p;
+ break;
+ }
+ }
+ write_unlock_irqrestore(&client->ports_lock, flags);
+ mutex_unlock(&client->ports_mutex);
+ if (found)
+ return port_delete(client, found);
+ else
+ return -ENOENT;
+}
+
+/* delete the all ports belonging to the given client */
+int snd_seq_delete_all_ports(struct snd_seq_client *client)
+{
+ unsigned long flags;
+ struct list_head deleted_list;
+ struct snd_seq_client_port *port, *tmp;
+
+ /* move the port list to deleted_list, and
+ * clear the port list in the client data.
+ */
+ mutex_lock(&client->ports_mutex);
+ write_lock_irqsave(&client->ports_lock, flags);
+ if (! list_empty(&client->ports_list_head)) {
+ list_add(&deleted_list, &client->ports_list_head);
+ list_del_init(&client->ports_list_head);
+ } else {
+ INIT_LIST_HEAD(&deleted_list);
+ }
+ client->num_ports = 0;
+ write_unlock_irqrestore(&client->ports_lock, flags);
+
+ /* remove each port in deleted_list */
+ list_for_each_entry_safe(port, tmp, &deleted_list, list) {
+ list_del(&port->list);
+ snd_seq_system_client_ev_port_exit(port->addr.client, port->addr.port);
+ port_delete(client, port);
+ }
+ mutex_unlock(&client->ports_mutex);
+ return 0;
+}
+
+/* set port info fields */
+int snd_seq_set_port_info(struct snd_seq_client_port * port,
+ struct snd_seq_port_info * info)
+{
+ if (snd_BUG_ON(!port || !info))
+ return -EINVAL;
+
+ /* set port name */
+ if (info->name[0])
+ strlcpy(port->name, info->name, sizeof(port->name));
+
+ /* set capabilities */
+ port->capability = info->capability;
+
+ /* get port type */
+ port->type = info->type;
+
+ /* information about supported channels/voices */
+ port->midi_channels = info->midi_channels;
+ port->midi_voices = info->midi_voices;
+ port->synth_voices = info->synth_voices;
+
+ /* timestamping */
+ port->timestamping = (info->flags & SNDRV_SEQ_PORT_FLG_TIMESTAMP) ? 1 : 0;
+ port->time_real = (info->flags & SNDRV_SEQ_PORT_FLG_TIME_REAL) ? 1 : 0;
+ port->time_queue = info->time_queue;
+
+ return 0;
+}
+
+/* get port info fields */
+int snd_seq_get_port_info(struct snd_seq_client_port * port,
+ struct snd_seq_port_info * info)
+{
+ if (snd_BUG_ON(!port || !info))
+ return -EINVAL;
+
+ /* get port name */
+ strlcpy(info->name, port->name, sizeof(info->name));
+
+ /* get capabilities */
+ info->capability = port->capability;
+
+ /* get port type */
+ info->type = port->type;
+
+ /* information about supported channels/voices */
+ info->midi_channels = port->midi_channels;
+ info->midi_voices = port->midi_voices;
+ info->synth_voices = port->synth_voices;
+
+ /* get subscriber counts */
+ info->read_use = port->c_src.count;
+ info->write_use = port->c_dest.count;
+
+ /* timestamping */
+ info->flags = 0;
+ if (port->timestamping) {
+ info->flags |= SNDRV_SEQ_PORT_FLG_TIMESTAMP;
+ if (port->time_real)
+ info->flags |= SNDRV_SEQ_PORT_FLG_TIME_REAL;
+ info->time_queue = port->time_queue;
+ }
+
+ return 0;
+}
+
+
+
+/*
+ * call callback functions (if any):
+ * the callbacks are invoked only when the first (for connection) or
+ * the last subscription (for disconnection) is done. Second or later
+ * subscription results in increment of counter, but no callback is
+ * invoked.
+ * This feature is useful if these callbacks are associated with
+ * initialization or termination of devices (see seq_midi.c).
+ *
+ * If callback_all option is set, the callback function is invoked
+ * at each connnection/disconnection.
+ */
+
+static int subscribe_port(struct snd_seq_client *client,
+ struct snd_seq_client_port *port,
+ struct snd_seq_port_subs_info *grp,
+ struct snd_seq_port_subscribe *info,
+ int send_ack)
+{
+ int err = 0;
+
+ if (!try_module_get(port->owner))
+ return -EFAULT;
+ grp->count++;
+ if (grp->open && (port->callback_all || grp->count == 1)) {
+ err = grp->open(port->private_data, info);
+ if (err < 0) {
+ module_put(port->owner);
+ grp->count--;
+ }
+ }
+ if (err >= 0 && send_ack && client->type == USER_CLIENT)
+ snd_seq_client_notify_subscription(port->addr.client, port->addr.port,
+ info, SNDRV_SEQ_EVENT_PORT_SUBSCRIBED);
+
+ return err;
+}
+
+static int unsubscribe_port(struct snd_seq_client *client,
+ struct snd_seq_client_port *port,
+ struct snd_seq_port_subs_info *grp,
+ struct snd_seq_port_subscribe *info,
+ int send_ack)
+{
+ int err = 0;
+
+ if (! grp->count)
+ return -EINVAL;
+ grp->count--;
+ if (grp->close && (port->callback_all || grp->count == 0))
+ err = grp->close(port->private_data, info);
+ if (send_ack && client->type == USER_CLIENT)
+ snd_seq_client_notify_subscription(port->addr.client, port->addr.port,
+ info, SNDRV_SEQ_EVENT_PORT_UNSUBSCRIBED);
+ module_put(port->owner);
+ return err;
+}
+
+
+
+/* check if both addresses are identical */
+static inline int addr_match(struct snd_seq_addr *r, struct snd_seq_addr *s)
+{
+ return (r->client == s->client) && (r->port == s->port);
+}
+
+/* check the two subscribe info match */
+/* if flags is zero, checks only sender and destination addresses */
+static int match_subs_info(struct snd_seq_port_subscribe *r,
+ struct snd_seq_port_subscribe *s)
+{
+ if (addr_match(&r->sender, &s->sender) &&
+ addr_match(&r->dest, &s->dest)) {
+ if (r->flags && r->flags == s->flags)
+ return r->queue == s->queue;
+ else if (! r->flags)
+ return 1;
+ }
+ return 0;
+}
+
+
+/* connect two ports */
+int snd_seq_port_connect(struct snd_seq_client *connector,
+ struct snd_seq_client *src_client,
+ struct snd_seq_client_port *src_port,
+ struct snd_seq_client *dest_client,
+ struct snd_seq_client_port *dest_port,
+ struct snd_seq_port_subscribe *info)
+{
+ struct snd_seq_port_subs_info *src = &src_port->c_src;
+ struct snd_seq_port_subs_info *dest = &dest_port->c_dest;
+ struct snd_seq_subscribers *subs, *s;
+ int err, src_called = 0;
+ unsigned long flags;
+ int exclusive;
+
+ subs = kzalloc(sizeof(*subs), GFP_KERNEL);
+ if (! subs)
+ return -ENOMEM;
+
+ subs->info = *info;
+ atomic_set(&subs->ref_count, 2);
+
+ down_write(&src->list_mutex);
+ down_write_nested(&dest->list_mutex, SINGLE_DEPTH_NESTING);
+
+ exclusive = info->flags & SNDRV_SEQ_PORT_SUBS_EXCLUSIVE ? 1 : 0;
+ err = -EBUSY;
+ if (exclusive) {
+ if (! list_empty(&src->list_head) || ! list_empty(&dest->list_head))
+ goto __error;
+ } else {
+ if (src->exclusive || dest->exclusive)
+ goto __error;
+ /* check whether already exists */
+ list_for_each_entry(s, &src->list_head, src_list) {
+ if (match_subs_info(info, &s->info))
+ goto __error;
+ }
+ list_for_each_entry(s, &dest->list_head, dest_list) {
+ if (match_subs_info(info, &s->info))
+ goto __error;
+ }
+ }
+
+ if ((err = subscribe_port(src_client, src_port, src, info,
+ connector->number != src_client->number)) < 0)
+ goto __error;
+ src_called = 1;
+
+ if ((err = subscribe_port(dest_client, dest_port, dest, info,
+ connector->number != dest_client->number)) < 0)
+ goto __error;
+
+ /* add to list */
+ write_lock_irqsave(&src->list_lock, flags);
+ // write_lock(&dest->list_lock); // no other lock yet
+ list_add_tail(&subs->src_list, &src->list_head);
+ list_add_tail(&subs->dest_list, &dest->list_head);
+ // write_unlock(&dest->list_lock); // no other lock yet
+ write_unlock_irqrestore(&src->list_lock, flags);
+
+ src->exclusive = dest->exclusive = exclusive;
+
+ up_write(&dest->list_mutex);
+ up_write(&src->list_mutex);
+ return 0;
+
+ __error:
+ if (src_called)
+ unsubscribe_port(src_client, src_port, src, info,
+ connector->number != src_client->number);
+ kfree(subs);
+ up_write(&dest->list_mutex);
+ up_write(&src->list_mutex);
+ return err;
+}
+
+
+/* remove the connection */
+int snd_seq_port_disconnect(struct snd_seq_client *connector,
+ struct snd_seq_client *src_client,
+ struct snd_seq_client_port *src_port,
+ struct snd_seq_client *dest_client,
+ struct snd_seq_client_port *dest_port,
+ struct snd_seq_port_subscribe *info)
+{
+ struct snd_seq_port_subs_info *src = &src_port->c_src;
+ struct snd_seq_port_subs_info *dest = &dest_port->c_dest;
+ struct snd_seq_subscribers *subs;
+ int err = -ENOENT;
+ unsigned long flags;
+
+ down_write(&src->list_mutex);
+ down_write_nested(&dest->list_mutex, SINGLE_DEPTH_NESTING);
+
+ /* look for the connection */
+ list_for_each_entry(subs, &src->list_head, src_list) {
+ if (match_subs_info(info, &subs->info)) {
+ write_lock_irqsave(&src->list_lock, flags);
+ // write_lock(&dest->list_lock); // no lock yet
+ list_del(&subs->src_list);
+ list_del(&subs->dest_list);
+ // write_unlock(&dest->list_lock);
+ write_unlock_irqrestore(&src->list_lock, flags);
+ src->exclusive = dest->exclusive = 0;
+ unsubscribe_port(src_client, src_port, src, info,
+ connector->number != src_client->number);
+ unsubscribe_port(dest_client, dest_port, dest, info,
+ connector->number != dest_client->number);
+ kfree(subs);
+ err = 0;
+ break;
+ }
+ }
+
+ up_write(&dest->list_mutex);
+ up_write(&src->list_mutex);
+ return err;
+}
+
+
+/* get matched subscriber */
+struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp,
+ struct snd_seq_addr *dest_addr)
+{
+ struct snd_seq_subscribers *s, *found = NULL;
+
+ down_read(&src_grp->list_mutex);
+ list_for_each_entry(s, &src_grp->list_head, src_list) {
+ if (addr_match(dest_addr, &s->info.dest)) {
+ found = s;
+ break;
+ }
+ }
+ up_read(&src_grp->list_mutex);
+ return found;
+}
+
+/*
+ * Attach a device driver that wants to receive events from the
+ * sequencer. Returns the new port number on success.
+ * A driver that wants to receive the events converted to midi, will
+ * use snd_seq_midisynth_register_port().
+ */
+/* exported */
+int snd_seq_event_port_attach(int client,
+ struct snd_seq_port_callback *pcbp,
+ int cap, int type, int midi_channels,
+ int midi_voices, char *portname)
+{
+ struct snd_seq_port_info portinfo;
+ int ret;
+
+ /* Set up the port */
+ memset(&portinfo, 0, sizeof(portinfo));
+ portinfo.addr.client = client;
+ strlcpy(portinfo.name, portname ? portname : "Unamed port",
+ sizeof(portinfo.name));
+
+ portinfo.capability = cap;
+ portinfo.type = type;
+ portinfo.kernel = pcbp;
+ portinfo.midi_channels = midi_channels;
+ portinfo.midi_voices = midi_voices;
+
+ /* Create it */
+ ret = snd_seq_kernel_client_ctl(client,
+ SNDRV_SEQ_IOCTL_CREATE_PORT,
+ &portinfo);
+
+ if (ret >= 0)
+ ret = portinfo.addr.port;
+
+ return ret;
+}
+
+EXPORT_SYMBOL(snd_seq_event_port_attach);
+
+/*
+ * Detach the driver from a port.
+ */
+/* exported */
+int snd_seq_event_port_detach(int client, int port)
+{
+ struct snd_seq_port_info portinfo;
+ int err;
+
+ memset(&portinfo, 0, sizeof(portinfo));
+ portinfo.addr.client = client;
+ portinfo.addr.port = port;
+ err = snd_seq_kernel_client_ctl(client,
+ SNDRV_SEQ_IOCTL_DELETE_PORT,
+ &portinfo);
+
+ return err;
+}
+
+EXPORT_SYMBOL(snd_seq_event_port_detach);
diff --git a/sound/core/seq/seq_ports.h b/sound/core/seq/seq_ports.h
new file mode 100644
index 0000000..9d71171
--- /dev/null
+++ b/sound/core/seq/seq_ports.h
@@ -0,0 +1,142 @@
+/*
+ * ALSA sequencer Ports
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_PORTS_H
+#define __SND_SEQ_PORTS_H
+
+#include <sound/seq_kernel.h>
+#include "seq_lock.h"
+
+/* list of 'exported' ports */
+
+/* Client ports that are not exported are still accessible, but are
+ anonymous ports.
+
+ If a port supports SUBSCRIPTION, that port can send events to all
+ subscribersto a special address, with address
+ (queue==SNDRV_SEQ_ADDRESS_SUBSCRIBERS). The message is then send to all
+ recipients that are registered in the subscription list. A typical
+ application for these SUBSCRIPTION events is handling of incoming MIDI
+ data. The port doesn't 'know' what other clients are interested in this
+ message. If for instance a MIDI recording application would like to receive
+ the events from that port, it will first have to subscribe with that port.
+
+*/
+
+struct snd_seq_subscribers {
+ struct snd_seq_port_subscribe info; /* additional info */
+ struct list_head src_list; /* link of sources */
+ struct list_head dest_list; /* link of destinations */
+ atomic_t ref_count;
+};
+
+struct snd_seq_port_subs_info {
+ struct list_head list_head; /* list of subscribed ports */
+ unsigned int count; /* count of subscribers */
+ unsigned int exclusive: 1; /* exclusive mode */
+ struct rw_semaphore list_mutex;
+ rwlock_t list_lock;
+ int (*open)(void *private_data, struct snd_seq_port_subscribe *info);
+ int (*close)(void *private_data, struct snd_seq_port_subscribe *info);
+};
+
+struct snd_seq_client_port {
+
+ struct snd_seq_addr addr; /* client/port number */
+ struct module *owner; /* owner of this port */
+ char name[64]; /* port name */
+ struct list_head list; /* port list */
+ snd_use_lock_t use_lock;
+
+ /* subscribers */
+ struct snd_seq_port_subs_info c_src; /* read (sender) list */
+ struct snd_seq_port_subs_info c_dest; /* write (dest) list */
+
+ int (*event_input)(struct snd_seq_event *ev, int direct, void *private_data,
+ int atomic, int hop);
+ void (*private_free)(void *private_data);
+ void *private_data;
+ unsigned int callback_all : 1;
+ unsigned int closing : 1;
+ unsigned int timestamping: 1;
+ unsigned int time_real: 1;
+ int time_queue;
+
+ /* capability, inport, output, sync */
+ unsigned int capability; /* port capability bits */
+ unsigned int type; /* port type bits */
+
+ /* supported channels */
+ int midi_channels;
+ int midi_voices;
+ int synth_voices;
+
+};
+
+struct snd_seq_client;
+
+/* return pointer to port structure and lock port */
+struct snd_seq_client_port *snd_seq_port_use_ptr(struct snd_seq_client *client, int num);
+
+/* search for next port - port is locked if found */
+struct snd_seq_client_port *snd_seq_port_query_nearest(struct snd_seq_client *client,
+ struct snd_seq_port_info *pinfo);
+
+/* unlock the port */
+#define snd_seq_port_unlock(port) snd_use_lock_free(&(port)->use_lock)
+
+/* create a port, port number is returned (-1 on failure) */
+struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, int port_index);
+
+/* delete a port */
+int snd_seq_delete_port(struct snd_seq_client *client, int port);
+
+/* delete all ports */
+int snd_seq_delete_all_ports(struct snd_seq_client *client);
+
+/* set port info fields */
+int snd_seq_set_port_info(struct snd_seq_client_port *port,
+ struct snd_seq_port_info *info);
+
+/* get port info fields */
+int snd_seq_get_port_info(struct snd_seq_client_port *port,
+ struct snd_seq_port_info *info);
+
+/* add subscriber to subscription list */
+int snd_seq_port_connect(struct snd_seq_client *caller,
+ struct snd_seq_client *s, struct snd_seq_client_port *sp,
+ struct snd_seq_client *d, struct snd_seq_client_port *dp,
+ struct snd_seq_port_subscribe *info);
+
+/* remove subscriber from subscription list */
+int snd_seq_port_disconnect(struct snd_seq_client *caller,
+ struct snd_seq_client *s, struct snd_seq_client_port *sp,
+ struct snd_seq_client *d, struct snd_seq_client_port *dp,
+ struct snd_seq_port_subscribe *info);
+
+/* subscribe port */
+int snd_seq_port_subscribe(struct snd_seq_client_port *port,
+ struct snd_seq_port_subscribe *info);
+
+/* get matched subscriber */
+struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp,
+ struct snd_seq_addr *dest_addr);
+
+#endif
diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c
new file mode 100644
index 0000000..0101a8b
--- /dev/null
+++ b/sound/core/seq/seq_prioq.c
@@ -0,0 +1,452 @@
+/*
+ * ALSA sequencer Priority Queue
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "seq_timer.h"
+#include "seq_prioq.h"
+
+
+/* Implementation is a simple linked list for now...
+
+ This priority queue orders the events on timestamp. For events with an
+ equeal timestamp the queue behaves as a FIFO.
+
+ *
+ * +-------+
+ * Head --> | first |
+ * +-------+
+ * |next
+ * +-----v-+
+ * | |
+ * +-------+
+ * |
+ * +-----v-+
+ * | |
+ * +-------+
+ * |
+ * +-----v-+
+ * Tail --> | last |
+ * +-------+
+ *
+
+ */
+
+
+
+/* create new prioq (constructor) */
+struct snd_seq_prioq *snd_seq_prioq_new(void)
+{
+ struct snd_seq_prioq *f;
+
+ f = kzalloc(sizeof(*f), GFP_KERNEL);
+ if (f == NULL) {
+ snd_printd("oops: malloc failed for snd_seq_prioq_new()\n");
+ return NULL;
+ }
+
+ spin_lock_init(&f->lock);
+ f->head = NULL;
+ f->tail = NULL;
+ f->cells = 0;
+
+ return f;
+}
+
+/* delete prioq (destructor) */
+void snd_seq_prioq_delete(struct snd_seq_prioq **fifo)
+{
+ struct snd_seq_prioq *f = *fifo;
+ *fifo = NULL;
+
+ if (f == NULL) {
+ snd_printd("oops: snd_seq_prioq_delete() called with NULL prioq\n");
+ return;
+ }
+
+ /* release resources...*/
+ /*....................*/
+
+ if (f->cells > 0) {
+ /* drain prioQ */
+ while (f->cells > 0)
+ snd_seq_cell_free(snd_seq_prioq_cell_out(f));
+ }
+
+ kfree(f);
+}
+
+
+
+
+/* compare timestamp between events */
+/* return 1 if a >= b; 0 */
+static inline int compare_timestamp(struct snd_seq_event *a,
+ struct snd_seq_event *b)
+{
+ if ((a->flags & SNDRV_SEQ_TIME_STAMP_MASK) == SNDRV_SEQ_TIME_STAMP_TICK) {
+ /* compare ticks */
+ return (snd_seq_compare_tick_time(&a->time.tick, &b->time.tick));
+ } else {
+ /* compare real time */
+ return (snd_seq_compare_real_time(&a->time.time, &b->time.time));
+ }
+}
+
+/* compare timestamp between events */
+/* return negative if a < b;
+ * zero if a = b;
+ * positive if a > b;
+ */
+static inline int compare_timestamp_rel(struct snd_seq_event *a,
+ struct snd_seq_event *b)
+{
+ if ((a->flags & SNDRV_SEQ_TIME_STAMP_MASK) == SNDRV_SEQ_TIME_STAMP_TICK) {
+ /* compare ticks */
+ if (a->time.tick > b->time.tick)
+ return 1;
+ else if (a->time.tick == b->time.tick)
+ return 0;
+ else
+ return -1;
+ } else {
+ /* compare real time */
+ if (a->time.time.tv_sec > b->time.time.tv_sec)
+ return 1;
+ else if (a->time.time.tv_sec == b->time.time.tv_sec) {
+ if (a->time.time.tv_nsec > b->time.time.tv_nsec)
+ return 1;
+ else if (a->time.time.tv_nsec == b->time.time.tv_nsec)
+ return 0;
+ else
+ return -1;
+ } else
+ return -1;
+ }
+}
+
+/* enqueue cell to prioq */
+int snd_seq_prioq_cell_in(struct snd_seq_prioq * f,
+ struct snd_seq_event_cell * cell)
+{
+ struct snd_seq_event_cell *cur, *prev;
+ unsigned long flags;
+ int count;
+ int prior;
+
+ if (snd_BUG_ON(!f || !cell))
+ return -EINVAL;
+
+ /* check flags */
+ prior = (cell->event.flags & SNDRV_SEQ_PRIORITY_MASK);
+
+ spin_lock_irqsave(&f->lock, flags);
+
+ /* check if this element needs to inserted at the end (ie. ordered
+ data is inserted) This will be very likeley if a sequencer
+ application or midi file player is feeding us (sequential) data */
+ if (f->tail && !prior) {
+ if (compare_timestamp(&cell->event, &f->tail->event)) {
+ /* add new cell to tail of the fifo */
+ f->tail->next = cell;
+ f->tail = cell;
+ cell->next = NULL;
+ f->cells++;
+ spin_unlock_irqrestore(&f->lock, flags);
+ return 0;
+ }
+ }
+ /* traverse list of elements to find the place where the new cell is
+ to be inserted... Note that this is a order n process ! */
+
+ prev = NULL; /* previous cell */
+ cur = f->head; /* cursor */
+
+ count = 10000; /* FIXME: enough big, isn't it? */
+ while (cur != NULL) {
+ /* compare timestamps */
+ int rel = compare_timestamp_rel(&cell->event, &cur->event);
+ if (rel < 0)
+ /* new cell has earlier schedule time, */
+ break;
+ else if (rel == 0 && prior)
+ /* equal schedule time and prior to others */
+ break;
+ /* new cell has equal or larger schedule time, */
+ /* move cursor to next cell */
+ prev = cur;
+ cur = cur->next;
+ if (! --count) {
+ spin_unlock_irqrestore(&f->lock, flags);
+ snd_printk(KERN_ERR "cannot find a pointer.. infinite loop?\n");
+ return -EINVAL;
+ }
+ }
+
+ /* insert it before cursor */
+ if (prev != NULL)
+ prev->next = cell;
+ cell->next = cur;
+
+ if (f->head == cur) /* this is the first cell, set head to it */
+ f->head = cell;
+ if (cur == NULL) /* reached end of the list */
+ f->tail = cell;
+ f->cells++;
+ spin_unlock_irqrestore(&f->lock, flags);
+ return 0;
+}
+
+/* dequeue cell from prioq */
+struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f)
+{
+ struct snd_seq_event_cell *cell;
+ unsigned long flags;
+
+ if (f == NULL) {
+ snd_printd("oops: snd_seq_prioq_cell_in() called with NULL prioq\n");
+ return NULL;
+ }
+ spin_lock_irqsave(&f->lock, flags);
+
+ cell = f->head;
+ if (cell) {
+ f->head = cell->next;
+
+ /* reset tail if this was the last element */
+ if (f->tail == cell)
+ f->tail = NULL;
+
+ cell->next = NULL;
+ f->cells--;
+ }
+
+ spin_unlock_irqrestore(&f->lock, flags);
+ return cell;
+}
+
+/* return number of events available in prioq */
+int snd_seq_prioq_avail(struct snd_seq_prioq * f)
+{
+ if (f == NULL) {
+ snd_printd("oops: snd_seq_prioq_cell_in() called with NULL prioq\n");
+ return 0;
+ }
+ return f->cells;
+}
+
+
+/* peek at cell at the head of the prioq */
+struct snd_seq_event_cell *snd_seq_prioq_cell_peek(struct snd_seq_prioq * f)
+{
+ if (f == NULL) {
+ snd_printd("oops: snd_seq_prioq_cell_in() called with NULL prioq\n");
+ return NULL;
+ }
+ return f->head;
+}
+
+
+static inline int prioq_match(struct snd_seq_event_cell *cell,
+ int client, int timestamp)
+{
+ if (cell->event.source.client == client ||
+ cell->event.dest.client == client)
+ return 1;
+ if (!timestamp)
+ return 0;
+ switch (cell->event.flags & SNDRV_SEQ_TIME_STAMP_MASK) {
+ case SNDRV_SEQ_TIME_STAMP_TICK:
+ if (cell->event.time.tick)
+ return 1;
+ break;
+ case SNDRV_SEQ_TIME_STAMP_REAL:
+ if (cell->event.time.time.tv_sec ||
+ cell->event.time.time.tv_nsec)
+ return 1;
+ break;
+ }
+ return 0;
+}
+
+/* remove cells for left client */
+void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp)
+{
+ register struct snd_seq_event_cell *cell, *next;
+ unsigned long flags;
+ struct snd_seq_event_cell *prev = NULL;
+ struct snd_seq_event_cell *freefirst = NULL, *freeprev = NULL, *freenext;
+
+ /* collect all removed cells */
+ spin_lock_irqsave(&f->lock, flags);
+ cell = f->head;
+ while (cell) {
+ next = cell->next;
+ if (prioq_match(cell, client, timestamp)) {
+ /* remove cell from prioq */
+ if (cell == f->head) {
+ f->head = cell->next;
+ } else {
+ prev->next = cell->next;
+ }
+ if (cell == f->tail)
+ f->tail = cell->next;
+ f->cells--;
+ /* add cell to free list */
+ cell->next = NULL;
+ if (freefirst == NULL) {
+ freefirst = cell;
+ } else {
+ freeprev->next = cell;
+ }
+ freeprev = cell;
+ } else {
+#if 0
+ printk("type = %i, source = %i, dest = %i, client = %i\n",
+ cell->event.type,
+ cell->event.source.client,
+ cell->event.dest.client,
+ client);
+#endif
+ prev = cell;
+ }
+ cell = next;
+ }
+ spin_unlock_irqrestore(&f->lock, flags);
+
+ /* remove selected cells */
+ while (freefirst) {
+ freenext = freefirst->next;
+ snd_seq_cell_free(freefirst);
+ freefirst = freenext;
+ }
+}
+
+static int prioq_remove_match(struct snd_seq_remove_events *info,
+ struct snd_seq_event *ev)
+{
+ int res;
+
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_DEST) {
+ if (ev->dest.client != info->dest.client ||
+ ev->dest.port != info->dest.port)
+ return 0;
+ }
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_DEST_CHANNEL) {
+ if (! snd_seq_ev_is_channel_type(ev))
+ return 0;
+ /* data.note.channel and data.control.channel are identical */
+ if (ev->data.note.channel != info->channel)
+ return 0;
+ }
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_AFTER) {
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_TICK)
+ res = snd_seq_compare_tick_time(&ev->time.tick, &info->time.tick);
+ else
+ res = snd_seq_compare_real_time(&ev->time.time, &info->time.time);
+ if (!res)
+ return 0;
+ }
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_BEFORE) {
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_TIME_TICK)
+ res = snd_seq_compare_tick_time(&ev->time.tick, &info->time.tick);
+ else
+ res = snd_seq_compare_real_time(&ev->time.time, &info->time.time);
+ if (res)
+ return 0;
+ }
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_EVENT_TYPE) {
+ if (ev->type != info->type)
+ return 0;
+ }
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_IGNORE_OFF) {
+ /* Do not remove off events */
+ switch (ev->type) {
+ case SNDRV_SEQ_EVENT_NOTEOFF:
+ /* case SNDRV_SEQ_EVENT_SAMPLE_STOP: */
+ return 0;
+ default:
+ break;
+ }
+ }
+ if (info->remove_mode & SNDRV_SEQ_REMOVE_TAG_MATCH) {
+ if (info->tag != ev->tag)
+ return 0;
+ }
+
+ return 1;
+}
+
+/* remove cells matching remove criteria */
+void snd_seq_prioq_remove_events(struct snd_seq_prioq * f, int client,
+ struct snd_seq_remove_events *info)
+{
+ struct snd_seq_event_cell *cell, *next;
+ unsigned long flags;
+ struct snd_seq_event_cell *prev = NULL;
+ struct snd_seq_event_cell *freefirst = NULL, *freeprev = NULL, *freenext;
+
+ /* collect all removed cells */
+ spin_lock_irqsave(&f->lock, flags);
+ cell = f->head;
+
+ while (cell) {
+ next = cell->next;
+ if (cell->event.source.client == client &&
+ prioq_remove_match(info, &cell->event)) {
+
+ /* remove cell from prioq */
+ if (cell == f->head) {
+ f->head = cell->next;
+ } else {
+ prev->next = cell->next;
+ }
+
+ if (cell == f->tail)
+ f->tail = cell->next;
+ f->cells--;
+
+ /* add cell to free list */
+ cell->next = NULL;
+ if (freefirst == NULL) {
+ freefirst = cell;
+ } else {
+ freeprev->next = cell;
+ }
+
+ freeprev = cell;
+ } else {
+ prev = cell;
+ }
+ cell = next;
+ }
+ spin_unlock_irqrestore(&f->lock, flags);
+
+ /* remove selected cells */
+ while (freefirst) {
+ freenext = freefirst->next;
+ snd_seq_cell_free(freefirst);
+ freefirst = freenext;
+ }
+}
+
+
diff --git a/sound/core/seq/seq_prioq.h b/sound/core/seq/seq_prioq.h
new file mode 100644
index 0000000..d38bb78
--- /dev/null
+++ b/sound/core/seq/seq_prioq.h
@@ -0,0 +1,62 @@
+/*
+ * ALSA sequencer Priority Queue
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_PRIOQ_H
+#define __SND_SEQ_PRIOQ_H
+
+#include "seq_memory.h"
+
+
+/* === PRIOQ === */
+
+struct snd_seq_prioq {
+ struct snd_seq_event_cell *head; /* pointer to head of prioq */
+ struct snd_seq_event_cell *tail; /* pointer to tail of prioq */
+ int cells;
+ spinlock_t lock;
+};
+
+
+/* create new prioq (constructor) */
+struct snd_seq_prioq *snd_seq_prioq_new(void);
+
+/* delete prioq (destructor) */
+void snd_seq_prioq_delete(struct snd_seq_prioq **fifo);
+
+/* enqueue cell to prioq */
+int snd_seq_prioq_cell_in(struct snd_seq_prioq *f, struct snd_seq_event_cell *cell);
+
+/* dequeue cell from prioq */
+struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f);
+
+/* return number of events available in prioq */
+int snd_seq_prioq_avail(struct snd_seq_prioq *f);
+
+/* peek at cell at the head of the prioq */
+struct snd_seq_event_cell *snd_seq_prioq_cell_peek(struct snd_seq_prioq *f);
+
+/* client left queue */
+void snd_seq_prioq_leave(struct snd_seq_prioq *f, int client, int timestamp);
+
+/* Remove events */
+void snd_seq_prioq_remove_events(struct snd_seq_prioq *f, int client,
+ struct snd_seq_remove_events *info);
+
+#endif
diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c
new file mode 100644
index 0000000..e7a8e9e
--- /dev/null
+++ b/sound/core/seq/seq_queue.c
@@ -0,0 +1,795 @@
+/*
+ * ALSA sequencer Timing queue handling
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * MAJOR CHANGES
+ * Nov. 13, 1999 Takashi Iwai <iwai@ww.uni-erlangen.de>
+ * - Queues are allocated dynamically via ioctl.
+ * - When owner client is deleted, all owned queues are deleted, too.
+ * - Owner of unlocked queue is kept unmodified even if it is
+ * manipulated by other clients.
+ * - Owner field in SET_QUEUE_OWNER ioctl must be identical with the
+ * caller client. i.e. Changing owner to a third client is not
+ * allowed.
+ *
+ * Aug. 30, 2000 Takashi Iwai
+ * - Queues are managed in static array again, but with better way.
+ * The API itself is identical.
+ * - The queue is locked when struct snd_seq_queue pointer is returned via
+ * queueptr(). This pointer *MUST* be released afterward by
+ * queuefree(ptr).
+ * - Addition of experimental sync support.
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+
+#include "seq_memory.h"
+#include "seq_queue.h"
+#include "seq_clientmgr.h"
+#include "seq_fifo.h"
+#include "seq_timer.h"
+#include "seq_info.h"
+
+/* list of allocated queues */
+static struct snd_seq_queue *queue_list[SNDRV_SEQ_MAX_QUEUES];
+static DEFINE_SPINLOCK(queue_list_lock);
+/* number of queues allocated */
+static int num_queues;
+
+int snd_seq_queue_get_cur_queues(void)
+{
+ return num_queues;
+}
+
+/*----------------------------------------------------------------*/
+
+/* assign queue id and insert to list */
+static int queue_list_add(struct snd_seq_queue *q)
+{
+ int i;
+ unsigned long flags;
+
+ spin_lock_irqsave(&queue_list_lock, flags);
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if (! queue_list[i]) {
+ queue_list[i] = q;
+ q->queue = i;
+ num_queues++;
+ spin_unlock_irqrestore(&queue_list_lock, flags);
+ return i;
+ }
+ }
+ spin_unlock_irqrestore(&queue_list_lock, flags);
+ return -1;
+}
+
+static struct snd_seq_queue *queue_list_remove(int id, int client)
+{
+ struct snd_seq_queue *q;
+ unsigned long flags;
+
+ spin_lock_irqsave(&queue_list_lock, flags);
+ q = queue_list[id];
+ if (q) {
+ spin_lock(&q->owner_lock);
+ if (q->owner == client) {
+ /* found */
+ q->klocked = 1;
+ spin_unlock(&q->owner_lock);
+ queue_list[id] = NULL;
+ num_queues--;
+ spin_unlock_irqrestore(&queue_list_lock, flags);
+ return q;
+ }
+ spin_unlock(&q->owner_lock);
+ }
+ spin_unlock_irqrestore(&queue_list_lock, flags);
+ return NULL;
+}
+
+/*----------------------------------------------------------------*/
+
+/* create new queue (constructor) */
+static struct snd_seq_queue *queue_new(int owner, int locked)
+{
+ struct snd_seq_queue *q;
+
+ q = kzalloc(sizeof(*q), GFP_KERNEL);
+ if (q == NULL) {
+ snd_printd("malloc failed for snd_seq_queue_new()\n");
+ return NULL;
+ }
+
+ spin_lock_init(&q->owner_lock);
+ spin_lock_init(&q->check_lock);
+ mutex_init(&q->timer_mutex);
+ snd_use_lock_init(&q->use_lock);
+ q->queue = -1;
+
+ q->tickq = snd_seq_prioq_new();
+ q->timeq = snd_seq_prioq_new();
+ q->timer = snd_seq_timer_new();
+ if (q->tickq == NULL || q->timeq == NULL || q->timer == NULL) {
+ snd_seq_prioq_delete(&q->tickq);
+ snd_seq_prioq_delete(&q->timeq);
+ snd_seq_timer_delete(&q->timer);
+ kfree(q);
+ return NULL;
+ }
+
+ q->owner = owner;
+ q->locked = locked;
+ q->klocked = 0;
+
+ return q;
+}
+
+/* delete queue (destructor) */
+static void queue_delete(struct snd_seq_queue *q)
+{
+ /* stop and release the timer */
+ snd_seq_timer_stop(q->timer);
+ snd_seq_timer_close(q);
+ /* wait until access free */
+ snd_use_lock_sync(&q->use_lock);
+ /* release resources... */
+ snd_seq_prioq_delete(&q->tickq);
+ snd_seq_prioq_delete(&q->timeq);
+ snd_seq_timer_delete(&q->timer);
+
+ kfree(q);
+}
+
+
+/*----------------------------------------------------------------*/
+
+/* setup queues */
+int __init snd_seq_queues_init(void)
+{
+ /*
+ memset(queue_list, 0, sizeof(queue_list));
+ num_queues = 0;
+ */
+ return 0;
+}
+
+/* delete all existing queues */
+void __exit snd_seq_queues_delete(void)
+{
+ int i;
+
+ /* clear list */
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if (queue_list[i])
+ queue_delete(queue_list[i]);
+ }
+}
+
+/* allocate a new queue -
+ * return queue index value or negative value for error
+ */
+int snd_seq_queue_alloc(int client, int locked, unsigned int info_flags)
+{
+ struct snd_seq_queue *q;
+
+ q = queue_new(client, locked);
+ if (q == NULL)
+ return -ENOMEM;
+ q->info_flags = info_flags;
+ if (queue_list_add(q) < 0) {
+ queue_delete(q);
+ return -ENOMEM;
+ }
+ snd_seq_queue_use(q->queue, client, 1); /* use this queue */
+ return q->queue;
+}
+
+/* delete a queue - queue must be owned by the client */
+int snd_seq_queue_delete(int client, int queueid)
+{
+ struct snd_seq_queue *q;
+
+ if (queueid < 0 || queueid >= SNDRV_SEQ_MAX_QUEUES)
+ return -EINVAL;
+ q = queue_list_remove(queueid, client);
+ if (q == NULL)
+ return -EINVAL;
+ queue_delete(q);
+
+ return 0;
+}
+
+
+/* return pointer to queue structure for specified id */
+struct snd_seq_queue *queueptr(int queueid)
+{
+ struct snd_seq_queue *q;
+ unsigned long flags;
+
+ if (queueid < 0 || queueid >= SNDRV_SEQ_MAX_QUEUES)
+ return NULL;
+ spin_lock_irqsave(&queue_list_lock, flags);
+ q = queue_list[queueid];
+ if (q)
+ snd_use_lock_use(&q->use_lock);
+ spin_unlock_irqrestore(&queue_list_lock, flags);
+ return q;
+}
+
+/* return the (first) queue matching with the specified name */
+struct snd_seq_queue *snd_seq_queue_find_name(char *name)
+{
+ int i;
+ struct snd_seq_queue *q;
+
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if ((q = queueptr(i)) != NULL) {
+ if (strncmp(q->name, name, sizeof(q->name)) == 0)
+ return q;
+ queuefree(q);
+ }
+ }
+ return NULL;
+}
+
+
+/* -------------------------------------------------------- */
+
+void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop)
+{
+ unsigned long flags;
+ struct snd_seq_event_cell *cell;
+
+ if (q == NULL)
+ return;
+
+ /* make this function non-reentrant */
+ spin_lock_irqsave(&q->check_lock, flags);
+ if (q->check_blocked) {
+ q->check_again = 1;
+ spin_unlock_irqrestore(&q->check_lock, flags);
+ return; /* other thread is already checking queues */
+ }
+ q->check_blocked = 1;
+ spin_unlock_irqrestore(&q->check_lock, flags);
+
+ __again:
+ /* Process tick queue... */
+ while ((cell = snd_seq_prioq_cell_peek(q->tickq)) != NULL) {
+ if (snd_seq_compare_tick_time(&q->timer->tick.cur_tick,
+ &cell->event.time.tick)) {
+ cell = snd_seq_prioq_cell_out(q->tickq);
+ if (cell)
+ snd_seq_dispatch_event(cell, atomic, hop);
+ } else {
+ /* event remains in the queue */
+ break;
+ }
+ }
+
+
+ /* Process time queue... */
+ while ((cell = snd_seq_prioq_cell_peek(q->timeq)) != NULL) {
+ if (snd_seq_compare_real_time(&q->timer->cur_time,
+ &cell->event.time.time)) {
+ cell = snd_seq_prioq_cell_out(q->timeq);
+ if (cell)
+ snd_seq_dispatch_event(cell, atomic, hop);
+ } else {
+ /* event remains in the queue */
+ break;
+ }
+ }
+
+ /* free lock */
+ spin_lock_irqsave(&q->check_lock, flags);
+ if (q->check_again) {
+ q->check_again = 0;
+ spin_unlock_irqrestore(&q->check_lock, flags);
+ goto __again;
+ }
+ q->check_blocked = 0;
+ spin_unlock_irqrestore(&q->check_lock, flags);
+}
+
+
+/* enqueue a event to singe queue */
+int snd_seq_enqueue_event(struct snd_seq_event_cell *cell, int atomic, int hop)
+{
+ int dest, err;
+ struct snd_seq_queue *q;
+
+ if (snd_BUG_ON(!cell))
+ return -EINVAL;
+ dest = cell->event.queue; /* destination queue */
+ q = queueptr(dest);
+ if (q == NULL)
+ return -EINVAL;
+ /* handle relative time stamps, convert them into absolute */
+ if ((cell->event.flags & SNDRV_SEQ_TIME_MODE_MASK) == SNDRV_SEQ_TIME_MODE_REL) {
+ switch (cell->event.flags & SNDRV_SEQ_TIME_STAMP_MASK) {
+ case SNDRV_SEQ_TIME_STAMP_TICK:
+ cell->event.time.tick += q->timer->tick.cur_tick;
+ break;
+
+ case SNDRV_SEQ_TIME_STAMP_REAL:
+ snd_seq_inc_real_time(&cell->event.time.time,
+ &q->timer->cur_time);
+ break;
+ }
+ cell->event.flags &= ~SNDRV_SEQ_TIME_MODE_MASK;
+ cell->event.flags |= SNDRV_SEQ_TIME_MODE_ABS;
+ }
+ /* enqueue event in the real-time or midi queue */
+ switch (cell->event.flags & SNDRV_SEQ_TIME_STAMP_MASK) {
+ case SNDRV_SEQ_TIME_STAMP_TICK:
+ err = snd_seq_prioq_cell_in(q->tickq, cell);
+ break;
+
+ case SNDRV_SEQ_TIME_STAMP_REAL:
+ default:
+ err = snd_seq_prioq_cell_in(q->timeq, cell);
+ break;
+ }
+
+ if (err < 0) {
+ queuefree(q); /* unlock */
+ return err;
+ }
+
+ /* trigger dispatching */
+ snd_seq_check_queue(q, atomic, hop);
+
+ queuefree(q); /* unlock */
+
+ return 0;
+}
+
+
+/*----------------------------------------------------------------*/
+
+static inline int check_access(struct snd_seq_queue *q, int client)
+{
+ return (q->owner == client) || (!q->locked && !q->klocked);
+}
+
+/* check if the client has permission to modify queue parameters.
+ * if it does, lock the queue
+ */
+static int queue_access_lock(struct snd_seq_queue *q, int client)
+{
+ unsigned long flags;
+ int access_ok;
+
+ spin_lock_irqsave(&q->owner_lock, flags);
+ access_ok = check_access(q, client);
+ if (access_ok)
+ q->klocked = 1;
+ spin_unlock_irqrestore(&q->owner_lock, flags);
+ return access_ok;
+}
+
+/* unlock the queue */
+static inline void queue_access_unlock(struct snd_seq_queue *q)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&q->owner_lock, flags);
+ q->klocked = 0;
+ spin_unlock_irqrestore(&q->owner_lock, flags);
+}
+
+/* exported - only checking permission */
+int snd_seq_queue_check_access(int queueid, int client)
+{
+ struct snd_seq_queue *q = queueptr(queueid);
+ int access_ok;
+ unsigned long flags;
+
+ if (! q)
+ return 0;
+ spin_lock_irqsave(&q->owner_lock, flags);
+ access_ok = check_access(q, client);
+ spin_unlock_irqrestore(&q->owner_lock, flags);
+ queuefree(q);
+ return access_ok;
+}
+
+/*----------------------------------------------------------------*/
+
+/*
+ * change queue's owner and permission
+ */
+int snd_seq_queue_set_owner(int queueid, int client, int locked)
+{
+ struct snd_seq_queue *q = queueptr(queueid);
+
+ if (q == NULL)
+ return -EINVAL;
+
+ if (! queue_access_lock(q, client)) {
+ queuefree(q);
+ return -EPERM;
+ }
+
+ q->locked = locked ? 1 : 0;
+ q->owner = client;
+ queue_access_unlock(q);
+ queuefree(q);
+
+ return 0;
+}
+
+
+/*----------------------------------------------------------------*/
+
+/* open timer -
+ * q->use mutex should be down before calling this function to avoid
+ * confliction with snd_seq_queue_use()
+ */
+int snd_seq_queue_timer_open(int queueid)
+{
+ int result = 0;
+ struct snd_seq_queue *queue;
+ struct snd_seq_timer *tmr;
+
+ queue = queueptr(queueid);
+ if (queue == NULL)
+ return -EINVAL;
+ tmr = queue->timer;
+ if ((result = snd_seq_timer_open(queue)) < 0) {
+ snd_seq_timer_defaults(tmr);
+ result = snd_seq_timer_open(queue);
+ }
+ queuefree(queue);
+ return result;
+}
+
+/* close timer -
+ * q->use mutex should be down before calling this function
+ */
+int snd_seq_queue_timer_close(int queueid)
+{
+ struct snd_seq_queue *queue;
+ struct snd_seq_timer *tmr;
+ int result = 0;
+
+ queue = queueptr(queueid);
+ if (queue == NULL)
+ return -EINVAL;
+ tmr = queue->timer;
+ snd_seq_timer_close(queue);
+ queuefree(queue);
+ return result;
+}
+
+/* change queue tempo and ppq */
+int snd_seq_queue_timer_set_tempo(int queueid, int client,
+ struct snd_seq_queue_tempo *info)
+{
+ struct snd_seq_queue *q = queueptr(queueid);
+ int result;
+
+ if (q == NULL)
+ return -EINVAL;
+ if (! queue_access_lock(q, client)) {
+ queuefree(q);
+ return -EPERM;
+ }
+
+ result = snd_seq_timer_set_tempo(q->timer, info->tempo);
+ if (result >= 0)
+ result = snd_seq_timer_set_ppq(q->timer, info->ppq);
+ if (result >= 0 && info->skew_base > 0)
+ result = snd_seq_timer_set_skew(q->timer, info->skew_value,
+ info->skew_base);
+ queue_access_unlock(q);
+ queuefree(q);
+ return result;
+}
+
+
+/* use or unuse this queue -
+ * if it is the first client, starts the timer.
+ * if it is not longer used by any clients, stop the timer.
+ */
+int snd_seq_queue_use(int queueid, int client, int use)
+{
+ struct snd_seq_queue *queue;
+
+ queue = queueptr(queueid);
+ if (queue == NULL)
+ return -EINVAL;
+ mutex_lock(&queue->timer_mutex);
+ if (use) {
+ if (!test_and_set_bit(client, queue->clients_bitmap))
+ queue->clients++;
+ } else {
+ if (test_and_clear_bit(client, queue->clients_bitmap))
+ queue->clients--;
+ }
+ if (queue->clients) {
+ if (use && queue->clients == 1)
+ snd_seq_timer_defaults(queue->timer);
+ snd_seq_timer_open(queue);
+ } else {
+ snd_seq_timer_close(queue);
+ }
+ mutex_unlock(&queue->timer_mutex);
+ queuefree(queue);
+ return 0;
+}
+
+/*
+ * check if queue is used by the client
+ * return negative value if the queue is invalid.
+ * return 0 if not used, 1 if used.
+ */
+int snd_seq_queue_is_used(int queueid, int client)
+{
+ struct snd_seq_queue *q;
+ int result;
+
+ q = queueptr(queueid);
+ if (q == NULL)
+ return -EINVAL; /* invalid queue */
+ result = test_bit(client, q->clients_bitmap) ? 1 : 0;
+ queuefree(q);
+ return result;
+}
+
+
+/*----------------------------------------------------------------*/
+
+/* notification that client has left the system -
+ * stop the timer on all queues owned by this client
+ */
+void snd_seq_queue_client_termination(int client)
+{
+ unsigned long flags;
+ int i;
+ struct snd_seq_queue *q;
+
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if ((q = queueptr(i)) == NULL)
+ continue;
+ spin_lock_irqsave(&q->owner_lock, flags);
+ if (q->owner == client)
+ q->klocked = 1;
+ spin_unlock_irqrestore(&q->owner_lock, flags);
+ if (q->owner == client) {
+ if (q->timer->running)
+ snd_seq_timer_stop(q->timer);
+ snd_seq_timer_reset(q->timer);
+ }
+ queuefree(q);
+ }
+}
+
+/* final stage notification -
+ * remove cells for no longer exist client (for non-owned queue)
+ * or delete this queue (for owned queue)
+ */
+void snd_seq_queue_client_leave(int client)
+{
+ int i;
+ struct snd_seq_queue *q;
+
+ /* delete own queues from queue list */
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if ((q = queue_list_remove(i, client)) != NULL)
+ queue_delete(q);
+ }
+
+ /* remove cells from existing queues -
+ * they are not owned by this client
+ */
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if ((q = queueptr(i)) == NULL)
+ continue;
+ if (test_bit(client, q->clients_bitmap)) {
+ snd_seq_prioq_leave(q->tickq, client, 0);
+ snd_seq_prioq_leave(q->timeq, client, 0);
+ snd_seq_queue_use(q->queue, client, 0);
+ }
+ queuefree(q);
+ }
+}
+
+
+
+/*----------------------------------------------------------------*/
+
+/* remove cells from all queues */
+void snd_seq_queue_client_leave_cells(int client)
+{
+ int i;
+ struct snd_seq_queue *q;
+
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if ((q = queueptr(i)) == NULL)
+ continue;
+ snd_seq_prioq_leave(q->tickq, client, 0);
+ snd_seq_prioq_leave(q->timeq, client, 0);
+ queuefree(q);
+ }
+}
+
+/* remove cells based on flush criteria */
+void snd_seq_queue_remove_cells(int client, struct snd_seq_remove_events *info)
+{
+ int i;
+ struct snd_seq_queue *q;
+
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if ((q = queueptr(i)) == NULL)
+ continue;
+ if (test_bit(client, q->clients_bitmap) &&
+ (! (info->remove_mode & SNDRV_SEQ_REMOVE_DEST) ||
+ q->queue == info->queue)) {
+ snd_seq_prioq_remove_events(q->tickq, client, info);
+ snd_seq_prioq_remove_events(q->timeq, client, info);
+ }
+ queuefree(q);
+ }
+}
+
+/*----------------------------------------------------------------*/
+
+/*
+ * send events to all subscribed ports
+ */
+static void queue_broadcast_event(struct snd_seq_queue *q, struct snd_seq_event *ev,
+ int atomic, int hop)
+{
+ struct snd_seq_event sev;
+
+ sev = *ev;
+
+ sev.flags = SNDRV_SEQ_TIME_STAMP_TICK|SNDRV_SEQ_TIME_MODE_ABS;
+ sev.time.tick = q->timer->tick.cur_tick;
+ sev.queue = q->queue;
+ sev.data.queue.queue = q->queue;
+
+ /* broadcast events from Timer port */
+ sev.source.client = SNDRV_SEQ_CLIENT_SYSTEM;
+ sev.source.port = SNDRV_SEQ_PORT_SYSTEM_TIMER;
+ sev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+ snd_seq_kernel_client_dispatch(SNDRV_SEQ_CLIENT_SYSTEM, &sev, atomic, hop);
+}
+
+/*
+ * process a received queue-control event.
+ * this function is exported for seq_sync.c.
+ */
+static void snd_seq_queue_process_event(struct snd_seq_queue *q,
+ struct snd_seq_event *ev,
+ int atomic, int hop)
+{
+ switch (ev->type) {
+ case SNDRV_SEQ_EVENT_START:
+ snd_seq_prioq_leave(q->tickq, ev->source.client, 1);
+ snd_seq_prioq_leave(q->timeq, ev->source.client, 1);
+ if (! snd_seq_timer_start(q->timer))
+ queue_broadcast_event(q, ev, atomic, hop);
+ break;
+
+ case SNDRV_SEQ_EVENT_CONTINUE:
+ if (! snd_seq_timer_continue(q->timer))
+ queue_broadcast_event(q, ev, atomic, hop);
+ break;
+
+ case SNDRV_SEQ_EVENT_STOP:
+ snd_seq_timer_stop(q->timer);
+ queue_broadcast_event(q, ev, atomic, hop);
+ break;
+
+ case SNDRV_SEQ_EVENT_TEMPO:
+ snd_seq_timer_set_tempo(q->timer, ev->data.queue.param.value);
+ queue_broadcast_event(q, ev, atomic, hop);
+ break;
+
+ case SNDRV_SEQ_EVENT_SETPOS_TICK:
+ if (snd_seq_timer_set_position_tick(q->timer, ev->data.queue.param.time.tick) == 0) {
+ queue_broadcast_event(q, ev, atomic, hop);
+ }
+ break;
+
+ case SNDRV_SEQ_EVENT_SETPOS_TIME:
+ if (snd_seq_timer_set_position_time(q->timer, ev->data.queue.param.time.time) == 0) {
+ queue_broadcast_event(q, ev, atomic, hop);
+ }
+ break;
+ case SNDRV_SEQ_EVENT_QUEUE_SKEW:
+ if (snd_seq_timer_set_skew(q->timer,
+ ev->data.queue.param.skew.value,
+ ev->data.queue.param.skew.base) == 0) {
+ queue_broadcast_event(q, ev, atomic, hop);
+ }
+ break;
+ }
+}
+
+
+/*
+ * Queue control via timer control port:
+ * this function is exported as a callback of timer port.
+ */
+int snd_seq_control_queue(struct snd_seq_event *ev, int atomic, int hop)
+{
+ struct snd_seq_queue *q;
+
+ if (snd_BUG_ON(!ev))
+ return -EINVAL;
+ q = queueptr(ev->data.queue.queue);
+
+ if (q == NULL)
+ return -EINVAL;
+
+ if (! queue_access_lock(q, ev->source.client)) {
+ queuefree(q);
+ return -EPERM;
+ }
+
+ snd_seq_queue_process_event(q, ev, atomic, hop);
+
+ queue_access_unlock(q);
+ queuefree(q);
+ return 0;
+}
+
+
+/*----------------------------------------------------------------*/
+
+#ifdef CONFIG_PROC_FS
+/* exported to seq_info.c */
+void snd_seq_info_queues_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int i, bpm;
+ struct snd_seq_queue *q;
+ struct snd_seq_timer *tmr;
+
+ for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
+ if ((q = queueptr(i)) == NULL)
+ continue;
+
+ tmr = q->timer;
+ if (tmr->tempo)
+ bpm = 60000000 / tmr->tempo;
+ else
+ bpm = 0;
+
+ snd_iprintf(buffer, "queue %d: [%s]\n", q->queue, q->name);
+ snd_iprintf(buffer, "owned by client : %d\n", q->owner);
+ snd_iprintf(buffer, "lock status : %s\n", q->locked ? "Locked" : "Free");
+ snd_iprintf(buffer, "queued time events : %d\n", snd_seq_prioq_avail(q->timeq));
+ snd_iprintf(buffer, "queued tick events : %d\n", snd_seq_prioq_avail(q->tickq));
+ snd_iprintf(buffer, "timer state : %s\n", tmr->running ? "Running" : "Stopped");
+ snd_iprintf(buffer, "timer PPQ : %d\n", tmr->ppq);
+ snd_iprintf(buffer, "current tempo : %d\n", tmr->tempo);
+ snd_iprintf(buffer, "current BPM : %d\n", bpm);
+ snd_iprintf(buffer, "current time : %d.%09d s\n", tmr->cur_time.tv_sec, tmr->cur_time.tv_nsec);
+ snd_iprintf(buffer, "current tick : %d\n", tmr->tick.cur_tick);
+ snd_iprintf(buffer, "\n");
+ queuefree(q);
+ }
+}
+#endif /* CONFIG_PROC_FS */
+
diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h
new file mode 100644
index 0000000..30c8111
--- /dev/null
+++ b/sound/core/seq/seq_queue.h
@@ -0,0 +1,139 @@
+/*
+ * ALSA sequencer Queue handling
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_QUEUE_H
+#define __SND_SEQ_QUEUE_H
+
+#include "seq_memory.h"
+#include "seq_prioq.h"
+#include "seq_timer.h"
+#include "seq_lock.h"
+#include <linux/interrupt.h>
+#include <linux/list.h>
+#include <linux/bitops.h>
+
+#define SEQ_QUEUE_NO_OWNER (-1)
+
+struct snd_seq_queue {
+ int queue; /* queue number */
+
+ char name[64]; /* name of this queue */
+
+ struct snd_seq_prioq *tickq; /* midi tick event queue */
+ struct snd_seq_prioq *timeq; /* real-time event queue */
+
+ struct snd_seq_timer *timer; /* time keeper for this queue */
+ int owner; /* client that 'owns' the timer */
+ unsigned int locked:1, /* timer is only accesibble by owner if set */
+ klocked:1, /* kernel lock (after START) */
+ check_again:1,
+ check_blocked:1;
+
+ unsigned int flags; /* status flags */
+ unsigned int info_flags; /* info for sync */
+
+ spinlock_t owner_lock;
+ spinlock_t check_lock;
+
+ /* clients which uses this queue (bitmap) */
+ DECLARE_BITMAP(clients_bitmap, SNDRV_SEQ_MAX_CLIENTS);
+ unsigned int clients; /* users of this queue */
+ struct mutex timer_mutex;
+
+ snd_use_lock_t use_lock;
+};
+
+
+/* get the number of current queues */
+int snd_seq_queue_get_cur_queues(void);
+
+/* init queues structure */
+int snd_seq_queues_init(void);
+
+/* delete queues */
+void snd_seq_queues_delete(void);
+
+
+/* create new queue (constructor) */
+int snd_seq_queue_alloc(int client, int locked, unsigned int flags);
+
+/* delete queue (destructor) */
+int snd_seq_queue_delete(int client, int queueid);
+
+/* notification that client has left the system */
+void snd_seq_queue_client_termination(int client);
+
+/* final stage */
+void snd_seq_queue_client_leave(int client);
+
+/* enqueue a event received from one the clients */
+int snd_seq_enqueue_event(struct snd_seq_event_cell *cell, int atomic, int hop);
+
+/* Remove events */
+void snd_seq_queue_client_leave_cells(int client);
+void snd_seq_queue_remove_cells(int client, struct snd_seq_remove_events *info);
+
+/* return pointer to queue structure for specified id */
+struct snd_seq_queue *queueptr(int queueid);
+/* unlock */
+#define queuefree(q) snd_use_lock_free(&(q)->use_lock)
+
+/* return the (first) queue matching with the specified name */
+struct snd_seq_queue *snd_seq_queue_find_name(char *name);
+
+/* check single queue and dispatch events */
+void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop);
+
+/* access to queue's parameters */
+int snd_seq_queue_check_access(int queueid, int client);
+int snd_seq_queue_timer_set_tempo(int queueid, int client, struct snd_seq_queue_tempo *info);
+int snd_seq_queue_set_owner(int queueid, int client, int locked);
+int snd_seq_queue_set_locked(int queueid, int client, int locked);
+int snd_seq_queue_timer_open(int queueid);
+int snd_seq_queue_timer_close(int queueid);
+int snd_seq_queue_use(int queueid, int client, int use);
+int snd_seq_queue_is_used(int queueid, int client);
+
+int snd_seq_control_queue(struct snd_seq_event *ev, int atomic, int hop);
+
+/*
+ * 64bit division - for sync stuff..
+ */
+#if defined(i386) || defined(i486)
+
+#define udiv_qrnnd(q, r, n1, n0, d) \
+ __asm__ ("divl %4" \
+ : "=a" ((u32)(q)), \
+ "=d" ((u32)(r)) \
+ : "0" ((u32)(n0)), \
+ "1" ((u32)(n1)), \
+ "rm" ((u32)(d)))
+
+#define u64_div(x,y,q) do {u32 __tmp; udiv_qrnnd(q, __tmp, (x)>>32, x, y);} while (0)
+#define u64_mod(x,y,r) do {u32 __tmp; udiv_qrnnd(__tmp, q, (x)>>32, x, y);} while (0)
+#define u64_divmod(x,y,q,r) udiv_qrnnd(q, r, (x)>>32, x, y)
+
+#else
+#define u64_div(x,y,q) ((q) = (u32)((u64)(x) / (u64)(y)))
+#define u64_mod(x,y,r) ((r) = (u32)((u64)(x) % (u64)(y)))
+#define u64_divmod(x,y,q,r) (u64_div(x,y,q), u64_mod(x,y,r))
+#endif
+
+
+#endif
diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c
new file mode 100644
index 0000000..77884e6
--- /dev/null
+++ b/sound/core/seq/seq_system.c
@@ -0,0 +1,173 @@
+/*
+ * ALSA sequencer System services Client
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <sound/core.h>
+#include "seq_system.h"
+#include "seq_timer.h"
+#include "seq_queue.h"
+
+/* internal client that provide system services, access to timer etc. */
+
+/*
+ * Port "Timer"
+ * - send tempo /start/stop etc. events to this port to manipulate the
+ * queue's timer. The queue address is specified in
+ * data.queue.queue.
+ * - this port supports subscription. The received timer events are
+ * broadcasted to all subscribed clients. The modified tempo
+ * value is stored on data.queue.value.
+ * The modifier client/port is not send.
+ *
+ * Port "Announce"
+ * - does not receive message
+ * - supports supscription. For each client or port attaching to or
+ * detaching from the system an announcement is send to the subscribed
+ * clients.
+ *
+ * Idea: the subscription mechanism might also work handy for distributing
+ * synchronisation and timing information. In this case we would ideally have
+ * a list of subscribers for each type of sync (time, tick), for each timing
+ * queue.
+ *
+ * NOTE: the queue to be started, stopped, etc. must be specified
+ * in data.queue.addr.queue field. queue is used only for
+ * scheduling, and no longer referred as affected queue.
+ * They are used only for timer broadcast (see above).
+ * -- iwai
+ */
+
+
+/* client id of our system client */
+static int sysclient = -1;
+
+/* port id numbers for this client */
+static int announce_port = -1;
+
+
+
+/* fill standard header data, source port & channel are filled in */
+static int setheader(struct snd_seq_event * ev, int client, int port)
+{
+ if (announce_port < 0)
+ return -ENODEV;
+
+ memset(ev, 0, sizeof(struct snd_seq_event));
+
+ ev->flags &= ~SNDRV_SEQ_EVENT_LENGTH_MASK;
+ ev->flags |= SNDRV_SEQ_EVENT_LENGTH_FIXED;
+
+ ev->source.client = sysclient;
+ ev->source.port = announce_port;
+ ev->dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+
+ /* fill data */
+ /*ev->data.addr.queue = SNDRV_SEQ_ADDRESS_UNKNOWN;*/
+ ev->data.addr.client = client;
+ ev->data.addr.port = port;
+
+ return 0;
+}
+
+
+/* entry points for broadcasting system events */
+void snd_seq_system_broadcast(int client, int port, int type)
+{
+ struct snd_seq_event ev;
+
+ if (setheader(&ev, client, port) < 0)
+ return;
+ ev.type = type;
+ snd_seq_kernel_client_dispatch(sysclient, &ev, 0, 0);
+}
+
+/* entry points for broadcasting system events */
+int snd_seq_system_notify(int client, int port, struct snd_seq_event *ev)
+{
+ ev->flags = SNDRV_SEQ_EVENT_LENGTH_FIXED;
+ ev->source.client = sysclient;
+ ev->source.port = announce_port;
+ ev->dest.client = client;
+ ev->dest.port = port;
+ return snd_seq_kernel_client_dispatch(sysclient, ev, 0, 0);
+}
+
+/* call-back handler for timer events */
+static int event_input_timer(struct snd_seq_event * ev, int direct, void *private_data, int atomic, int hop)
+{
+ return snd_seq_control_queue(ev, atomic, hop);
+}
+
+/* register our internal client */
+int __init snd_seq_system_client_init(void)
+{
+ struct snd_seq_port_callback pcallbacks;
+ struct snd_seq_port_info *port;
+
+ port = kzalloc(sizeof(*port), GFP_KERNEL);
+ if (!port)
+ return -ENOMEM;
+
+ memset(&pcallbacks, 0, sizeof(pcallbacks));
+ pcallbacks.owner = THIS_MODULE;
+ pcallbacks.event_input = event_input_timer;
+
+ /* register client */
+ sysclient = snd_seq_create_kernel_client(NULL, 0, "System");
+
+ /* register timer */
+ strcpy(port->name, "Timer");
+ port->capability = SNDRV_SEQ_PORT_CAP_WRITE; /* accept queue control */
+ port->capability |= SNDRV_SEQ_PORT_CAP_READ|SNDRV_SEQ_PORT_CAP_SUBS_READ; /* for broadcast */
+ port->kernel = &pcallbacks;
+ port->type = 0;
+ port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
+ port->addr.client = sysclient;
+ port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER;
+ snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+
+ /* register announcement port */
+ strcpy(port->name, "Announce");
+ port->capability = SNDRV_SEQ_PORT_CAP_READ|SNDRV_SEQ_PORT_CAP_SUBS_READ; /* for broadcast only */
+ port->kernel = NULL;
+ port->type = 0;
+ port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT;
+ port->addr.client = sysclient;
+ port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE;
+ snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port);
+ announce_port = port->addr.port;
+
+ kfree(port);
+ return 0;
+}
+
+
+/* unregister our internal client */
+void __exit snd_seq_system_client_done(void)
+{
+ int oldsysclient = sysclient;
+
+ if (oldsysclient >= 0) {
+ sysclient = -1;
+ announce_port = -1;
+ snd_seq_delete_kernel_client(oldsysclient);
+ }
+}
diff --git a/sound/core/seq/seq_system.h b/sound/core/seq/seq_system.h
new file mode 100644
index 0000000..cf2cfa2
--- /dev/null
+++ b/sound/core/seq/seq_system.h
@@ -0,0 +1,46 @@
+/*
+ * ALSA sequencer System Client
+ * Copyright (c) 1998 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_SYSTEM_H
+#define __SND_SEQ_SYSTEM_H
+
+#include <sound/seq_kernel.h>
+
+
+/* entry points for broadcasting system events */
+void snd_seq_system_broadcast(int client, int port, int type);
+
+#define snd_seq_system_client_ev_client_start(client) snd_seq_system_broadcast(client, 0, SNDRV_SEQ_EVENT_CLIENT_START)
+#define snd_seq_system_client_ev_client_exit(client) snd_seq_system_broadcast(client, 0, SNDRV_SEQ_EVENT_CLIENT_EXIT)
+#define snd_seq_system_client_ev_client_change(client) snd_seq_system_broadcast(client, 0, SNDRV_SEQ_EVENT_CLIENT_CHANGE)
+#define snd_seq_system_client_ev_port_start(client, port) snd_seq_system_broadcast(client, port, SNDRV_SEQ_EVENT_PORT_START)
+#define snd_seq_system_client_ev_port_exit(client, port) snd_seq_system_broadcast(client, port, SNDRV_SEQ_EVENT_PORT_EXIT)
+#define snd_seq_system_client_ev_port_change(client, port) snd_seq_system_broadcast(client, port, SNDRV_SEQ_EVENT_PORT_CHANGE)
+
+int snd_seq_system_notify(int client, int port, struct snd_seq_event *ev);
+
+/* register our internal client */
+int snd_seq_system_client_init(void);
+
+/* unregister our internal client */
+void snd_seq_system_client_done(void);
+
+
+#endif
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
new file mode 100644
index 0000000..f745c31
--- /dev/null
+++ b/sound/core/seq/seq_timer.c
@@ -0,0 +1,456 @@
+/*
+ * ALSA sequencer Timer
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <linux/slab.h>
+#include "seq_timer.h"
+#include "seq_queue.h"
+#include "seq_info.h"
+
+/* allowed sequencer timer frequencies, in Hz */
+#define MIN_FREQUENCY 10
+#define MAX_FREQUENCY 6250
+#define DEFAULT_FREQUENCY 1000
+
+#define SKEW_BASE 0x10000 /* 16bit shift */
+
+static void snd_seq_timer_set_tick_resolution(struct snd_seq_timer_tick *tick,
+ int tempo, int ppq)
+{
+ if (tempo < 1000000)
+ tick->resolution = (tempo * 1000) / ppq;
+ else {
+ /* might overflow.. */
+ unsigned int s;
+ s = tempo % ppq;
+ s = (s * 1000) / ppq;
+ tick->resolution = (tempo / ppq) * 1000;
+ tick->resolution += s;
+ }
+ if (tick->resolution <= 0)
+ tick->resolution = 1;
+ snd_seq_timer_update_tick(tick, 0);
+}
+
+/* create new timer (constructor) */
+struct snd_seq_timer *snd_seq_timer_new(void)
+{
+ struct snd_seq_timer *tmr;
+
+ tmr = kzalloc(sizeof(*tmr), GFP_KERNEL);
+ if (tmr == NULL) {
+ snd_printd("malloc failed for snd_seq_timer_new() \n");
+ return NULL;
+ }
+ spin_lock_init(&tmr->lock);
+
+ /* reset setup to defaults */
+ snd_seq_timer_defaults(tmr);
+
+ /* reset time */
+ snd_seq_timer_reset(tmr);
+
+ return tmr;
+}
+
+/* delete timer (destructor) */
+void snd_seq_timer_delete(struct snd_seq_timer **tmr)
+{
+ struct snd_seq_timer *t = *tmr;
+ *tmr = NULL;
+
+ if (t == NULL) {
+ snd_printd("oops: snd_seq_timer_delete() called with NULL timer\n");
+ return;
+ }
+ t->running = 0;
+
+ /* reset time */
+ snd_seq_timer_stop(t);
+ snd_seq_timer_reset(t);
+
+ kfree(t);
+}
+
+void snd_seq_timer_defaults(struct snd_seq_timer * tmr)
+{
+ /* setup defaults */
+ tmr->ppq = 96; /* 96 PPQ */
+ tmr->tempo = 500000; /* 120 BPM */
+ snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq);
+ tmr->running = 0;
+
+ tmr->type = SNDRV_SEQ_TIMER_ALSA;
+ tmr->alsa_id.dev_class = seq_default_timer_class;
+ tmr->alsa_id.dev_sclass = seq_default_timer_sclass;
+ tmr->alsa_id.card = seq_default_timer_card;
+ tmr->alsa_id.device = seq_default_timer_device;
+ tmr->alsa_id.subdevice = seq_default_timer_subdevice;
+ tmr->preferred_resolution = seq_default_timer_resolution;
+
+ tmr->skew = tmr->skew_base = SKEW_BASE;
+}
+
+void snd_seq_timer_reset(struct snd_seq_timer * tmr)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&tmr->lock, flags);
+
+ /* reset time & songposition */
+ tmr->cur_time.tv_sec = 0;
+ tmr->cur_time.tv_nsec = 0;
+
+ tmr->tick.cur_tick = 0;
+ tmr->tick.fraction = 0;
+
+ spin_unlock_irqrestore(&tmr->lock, flags);
+}
+
+
+/* called by timer interrupt routine. the period time since previous invocation is passed */
+static void snd_seq_timer_interrupt(struct snd_timer_instance *timeri,
+ unsigned long resolution,
+ unsigned long ticks)
+{
+ unsigned long flags;
+ struct snd_seq_queue *q = timeri->callback_data;
+ struct snd_seq_timer *tmr;
+
+ if (q == NULL)
+ return;
+ tmr = q->timer;
+ if (tmr == NULL)
+ return;
+ if (!tmr->running)
+ return;
+
+ resolution *= ticks;
+ if (tmr->skew != tmr->skew_base) {
+ /* FIXME: assuming skew_base = 0x10000 */
+ resolution = (resolution >> 16) * tmr->skew +
+ (((resolution & 0xffff) * tmr->skew) >> 16);
+ }
+
+ spin_lock_irqsave(&tmr->lock, flags);
+
+ /* update timer */
+ snd_seq_inc_time_nsec(&tmr->cur_time, resolution);
+
+ /* calculate current tick */
+ snd_seq_timer_update_tick(&tmr->tick, resolution);
+
+ /* register actual time of this timer update */
+ do_gettimeofday(&tmr->last_update);
+
+ spin_unlock_irqrestore(&tmr->lock, flags);
+
+ /* check queues and dispatch events */
+ snd_seq_check_queue(q, 1, 0);
+}
+
+/* set current tempo */
+int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!tmr))
+ return -EINVAL;
+ if (tempo <= 0)
+ return -EINVAL;
+ spin_lock_irqsave(&tmr->lock, flags);
+ if ((unsigned int)tempo != tmr->tempo) {
+ tmr->tempo = tempo;
+ snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq);
+ }
+ spin_unlock_irqrestore(&tmr->lock, flags);
+ return 0;
+}
+
+/* set current ppq */
+int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!tmr))
+ return -EINVAL;
+ if (ppq <= 0)
+ return -EINVAL;
+ spin_lock_irqsave(&tmr->lock, flags);
+ if (tmr->running && (ppq != tmr->ppq)) {
+ /* refuse to change ppq on running timers */
+ /* because it will upset the song position (ticks) */
+ spin_unlock_irqrestore(&tmr->lock, flags);
+ snd_printd("seq: cannot change ppq of a running timer\n");
+ return -EBUSY;
+ }
+
+ tmr->ppq = ppq;
+ snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq);
+ spin_unlock_irqrestore(&tmr->lock, flags);
+ return 0;
+}
+
+/* set current tick position */
+int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr,
+ snd_seq_tick_time_t position)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!tmr))
+ return -EINVAL;
+
+ spin_lock_irqsave(&tmr->lock, flags);
+ tmr->tick.cur_tick = position;
+ tmr->tick.fraction = 0;
+ spin_unlock_irqrestore(&tmr->lock, flags);
+ return 0;
+}
+
+/* set current real-time position */
+int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr,
+ snd_seq_real_time_t position)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!tmr))
+ return -EINVAL;
+
+ snd_seq_sanity_real_time(&position);
+ spin_lock_irqsave(&tmr->lock, flags);
+ tmr->cur_time = position;
+ spin_unlock_irqrestore(&tmr->lock, flags);
+ return 0;
+}
+
+/* set timer skew */
+int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew,
+ unsigned int base)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!tmr))
+ return -EINVAL;
+
+ /* FIXME */
+ if (base != SKEW_BASE) {
+ snd_printd("invalid skew base 0x%x\n", base);
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&tmr->lock, flags);
+ tmr->skew = skew;
+ spin_unlock_irqrestore(&tmr->lock, flags);
+ return 0;
+}
+
+int snd_seq_timer_open(struct snd_seq_queue *q)
+{
+ struct snd_timer_instance *t;
+ struct snd_seq_timer *tmr;
+ char str[32];
+ int err;
+
+ tmr = q->timer;
+ if (snd_BUG_ON(!tmr))
+ return -EINVAL;
+ if (tmr->timeri)
+ return -EBUSY;
+ sprintf(str, "sequencer queue %i", q->queue);
+ if (tmr->type != SNDRV_SEQ_TIMER_ALSA) /* standard ALSA timer */
+ return -EINVAL;
+ if (tmr->alsa_id.dev_class != SNDRV_TIMER_CLASS_SLAVE)
+ tmr->alsa_id.dev_sclass = SNDRV_TIMER_SCLASS_SEQUENCER;
+ err = snd_timer_open(&t, str, &tmr->alsa_id, q->queue);
+ if (err < 0 && tmr->alsa_id.dev_class != SNDRV_TIMER_CLASS_SLAVE) {
+ if (tmr->alsa_id.dev_class != SNDRV_TIMER_CLASS_GLOBAL ||
+ tmr->alsa_id.device != SNDRV_TIMER_GLOBAL_SYSTEM) {
+ struct snd_timer_id tid;
+ memset(&tid, 0, sizeof(tid));
+ tid.dev_class = SNDRV_TIMER_CLASS_GLOBAL;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_SEQUENCER;
+ tid.card = -1;
+ tid.device = SNDRV_TIMER_GLOBAL_SYSTEM;
+ err = snd_timer_open(&t, str, &tid, q->queue);
+ }
+ if (err < 0) {
+ snd_printk(KERN_ERR "seq fatal error: cannot create timer (%i)\n", err);
+ return err;
+ }
+ }
+ t->callback = snd_seq_timer_interrupt;
+ t->callback_data = q;
+ t->flags |= SNDRV_TIMER_IFLG_AUTO;
+ tmr->timeri = t;
+ return 0;
+}
+
+int snd_seq_timer_close(struct snd_seq_queue *q)
+{
+ struct snd_seq_timer *tmr;
+
+ tmr = q->timer;
+ if (snd_BUG_ON(!tmr))
+ return -EINVAL;
+ if (tmr->timeri) {
+ snd_timer_stop(tmr->timeri);
+ snd_timer_close(tmr->timeri);
+ tmr->timeri = NULL;
+ }
+ return 0;
+}
+
+int snd_seq_timer_stop(struct snd_seq_timer * tmr)
+{
+ if (! tmr->timeri)
+ return -EINVAL;
+ if (!tmr->running)
+ return 0;
+ tmr->running = 0;
+ snd_timer_pause(tmr->timeri);
+ return 0;
+}
+
+static int initialize_timer(struct snd_seq_timer *tmr)
+{
+ struct snd_timer *t;
+ unsigned long freq;
+
+ t = tmr->timeri->timer;
+ if (snd_BUG_ON(!t))
+ return -EINVAL;
+
+ freq = tmr->preferred_resolution;
+ if (!freq)
+ freq = DEFAULT_FREQUENCY;
+ else if (freq < MIN_FREQUENCY)
+ freq = MIN_FREQUENCY;
+ else if (freq > MAX_FREQUENCY)
+ freq = MAX_FREQUENCY;
+
+ tmr->ticks = 1;
+ if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE)) {
+ unsigned long r = t->hw.resolution;
+ if (! r && t->hw.c_resolution)
+ r = t->hw.c_resolution(t);
+ if (r) {
+ tmr->ticks = (unsigned int)(1000000000uL / (r * freq));
+ if (! tmr->ticks)
+ tmr->ticks = 1;
+ }
+ }
+ tmr->initialized = 1;
+ return 0;
+}
+
+int snd_seq_timer_start(struct snd_seq_timer * tmr)
+{
+ if (! tmr->timeri)
+ return -EINVAL;
+ if (tmr->running)
+ snd_seq_timer_stop(tmr);
+ snd_seq_timer_reset(tmr);
+ if (initialize_timer(tmr) < 0)
+ return -EINVAL;
+ snd_timer_start(tmr->timeri, tmr->ticks);
+ tmr->running = 1;
+ do_gettimeofday(&tmr->last_update);
+ return 0;
+}
+
+int snd_seq_timer_continue(struct snd_seq_timer * tmr)
+{
+ if (! tmr->timeri)
+ return -EINVAL;
+ if (tmr->running)
+ return -EBUSY;
+ if (! tmr->initialized) {
+ snd_seq_timer_reset(tmr);
+ if (initialize_timer(tmr) < 0)
+ return -EINVAL;
+ }
+ snd_timer_start(tmr->timeri, tmr->ticks);
+ tmr->running = 1;
+ do_gettimeofday(&tmr->last_update);
+ return 0;
+}
+
+/* return current 'real' time. use timeofday() to get better granularity. */
+snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr)
+{
+ snd_seq_real_time_t cur_time;
+
+ cur_time = tmr->cur_time;
+ if (tmr->running) {
+ struct timeval tm;
+ int usec;
+ do_gettimeofday(&tm);
+ usec = (int)(tm.tv_usec - tmr->last_update.tv_usec);
+ if (usec < 0) {
+ cur_time.tv_nsec += (1000000 + usec) * 1000;
+ cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec - 1;
+ } else {
+ cur_time.tv_nsec += usec * 1000;
+ cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec;
+ }
+ snd_seq_sanity_real_time(&cur_time);
+ }
+
+ return cur_time;
+}
+
+/* TODO: use interpolation on tick queue (will only be useful for very
+ high PPQ values) */
+snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr)
+{
+ return tmr->tick.cur_tick;
+}
+
+
+#ifdef CONFIG_PROC_FS
+/* exported to seq_info.c */
+void snd_seq_info_timer_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int idx;
+ struct snd_seq_queue *q;
+ struct snd_seq_timer *tmr;
+ struct snd_timer_instance *ti;
+ unsigned long resolution;
+
+ for (idx = 0; idx < SNDRV_SEQ_MAX_QUEUES; idx++) {
+ q = queueptr(idx);
+ if (q == NULL)
+ continue;
+ if ((tmr = q->timer) == NULL ||
+ (ti = tmr->timeri) == NULL) {
+ queuefree(q);
+ continue;
+ }
+ snd_iprintf(buffer, "Timer for queue %i : %s\n", q->queue, ti->timer->name);
+ resolution = snd_timer_resolution(ti) * tmr->ticks;
+ snd_iprintf(buffer, " Period time : %lu.%09lu\n", resolution / 1000000000, resolution % 1000000000);
+ snd_iprintf(buffer, " Skew : %u / %u\n", tmr->skew, tmr->skew_base);
+ queuefree(q);
+ }
+}
+#endif /* CONFIG_PROC_FS */
+
diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h
new file mode 100644
index 0000000..88dfb71
--- /dev/null
+++ b/sound/core/seq/seq_timer.h
@@ -0,0 +1,148 @@
+/*
+ * ALSA sequencer Timer
+ * Copyright (c) 1998-1999 by Frank van de Pol <fvdpol@coil.demon.nl>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __SND_SEQ_TIMER_H
+#define __SND_SEQ_TIMER_H
+
+#include <sound/timer.h>
+#include <sound/seq_kernel.h>
+
+struct snd_seq_timer_tick {
+ snd_seq_tick_time_t cur_tick; /* current tick */
+ unsigned long resolution; /* time per tick in nsec */
+ unsigned long fraction; /* current time per tick in nsec */
+};
+
+struct snd_seq_timer {
+ /* ... tempo / offset / running state */
+
+ unsigned int running:1, /* running state of queue */
+ initialized:1; /* timer is initialized */
+
+ unsigned int tempo; /* current tempo, us/tick */
+ int ppq; /* time resolution, ticks/quarter */
+
+ snd_seq_real_time_t cur_time; /* current time */
+ struct snd_seq_timer_tick tick; /* current tick */
+ int tick_updated;
+
+ int type; /* timer type */
+ struct snd_timer_id alsa_id; /* ALSA's timer ID */
+ struct snd_timer_instance *timeri; /* timer instance */
+ unsigned int ticks;
+ unsigned long preferred_resolution; /* timer resolution, ticks/sec */
+
+ unsigned int skew;
+ unsigned int skew_base;
+
+ struct timeval last_update; /* time of last clock update, used for interpolation */
+
+ spinlock_t lock;
+};
+
+
+/* create new timer (constructor) */
+struct snd_seq_timer *snd_seq_timer_new(void);
+
+/* delete timer (destructor) */
+void snd_seq_timer_delete(struct snd_seq_timer **tmr);
+
+/* */
+static inline void snd_seq_timer_update_tick(struct snd_seq_timer_tick *tick,
+ unsigned long resolution)
+{
+ if (tick->resolution > 0) {
+ tick->fraction += resolution;
+ tick->cur_tick += (unsigned int)(tick->fraction / tick->resolution);
+ tick->fraction %= tick->resolution;
+ }
+}
+
+
+/* compare timestamp between events */
+/* return 1 if a >= b; otherwise return 0 */
+static inline int snd_seq_compare_tick_time(snd_seq_tick_time_t *a, snd_seq_tick_time_t *b)
+{
+ /* compare ticks */
+ return (*a >= *b);
+}
+
+static inline int snd_seq_compare_real_time(snd_seq_real_time_t *a, snd_seq_real_time_t *b)
+{
+ /* compare real time */
+ if (a->tv_sec > b->tv_sec)
+ return 1;
+ if ((a->tv_sec == b->tv_sec) && (a->tv_nsec >= b->tv_nsec))
+ return 1;
+ return 0;
+}
+
+
+static inline void snd_seq_sanity_real_time(snd_seq_real_time_t *tm)
+{
+ while (tm->tv_nsec >= 1000000000) {
+ /* roll-over */
+ tm->tv_nsec -= 1000000000;
+ tm->tv_sec++;
+ }
+}
+
+
+/* increment timestamp */
+static inline void snd_seq_inc_real_time(snd_seq_real_time_t *tm, snd_seq_real_time_t *inc)
+{
+ tm->tv_sec += inc->tv_sec;
+ tm->tv_nsec += inc->tv_nsec;
+ snd_seq_sanity_real_time(tm);
+}
+
+static inline void snd_seq_inc_time_nsec(snd_seq_real_time_t *tm, unsigned long nsec)
+{
+ tm->tv_nsec += nsec;
+ snd_seq_sanity_real_time(tm);
+}
+
+/* called by timer isr */
+struct snd_seq_queue;
+int snd_seq_timer_open(struct snd_seq_queue *q);
+int snd_seq_timer_close(struct snd_seq_queue *q);
+int snd_seq_timer_midi_open(struct snd_seq_queue *q);
+int snd_seq_timer_midi_close(struct snd_seq_queue *q);
+void snd_seq_timer_defaults(struct snd_seq_timer *tmr);
+void snd_seq_timer_reset(struct snd_seq_timer *tmr);
+int snd_seq_timer_stop(struct snd_seq_timer *tmr);
+int snd_seq_timer_start(struct snd_seq_timer *tmr);
+int snd_seq_timer_continue(struct snd_seq_timer *tmr);
+int snd_seq_timer_set_tempo(struct snd_seq_timer *tmr, int tempo);
+int snd_seq_timer_set_ppq(struct snd_seq_timer *tmr, int ppq);
+int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr, snd_seq_tick_time_t position);
+int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr, snd_seq_real_time_t position);
+int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, unsigned int base);
+snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr);
+snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr);
+
+extern int seq_default_timer_class;
+extern int seq_default_timer_sclass;
+extern int seq_default_timer_card;
+extern int seq_default_timer_device;
+extern int seq_default_timer_subdevice;
+extern int seq_default_timer_resolution;
+
+#endif
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
new file mode 100644
index 0000000..86e7739
--- /dev/null
+++ b/sound/core/seq/seq_virmidi.c
@@ -0,0 +1,542 @@
+/*
+ * Virtual Raw MIDI client on Sequencer
+ *
+ * Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>,
+ * Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * Virtual Raw MIDI client
+ *
+ * The virtual rawmidi client is a sequencer client which associate
+ * a rawmidi device file. The created rawmidi device file can be
+ * accessed as a normal raw midi, but its MIDI source and destination
+ * are arbitrary. For example, a user-client software synth connected
+ * to this port can be used as a normal midi device as well.
+ *
+ * The virtual rawmidi device accepts also multiple opens. Each file
+ * has its own input buffer, so that no conflict would occur. The drain
+ * of input/output buffer acts only to the local buffer.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/wait.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/minors.h>
+#include <sound/seq_kernel.h>
+#include <sound/seq_midi_event.h>
+#include <sound/seq_virmidi.h>
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("Virtual Raw MIDI client on Sequencer");
+MODULE_LICENSE("GPL");
+
+/*
+ * initialize an event record
+ */
+static void snd_virmidi_init_event(struct snd_virmidi *vmidi,
+ struct snd_seq_event *ev)
+{
+ memset(ev, 0, sizeof(*ev));
+ ev->source.port = vmidi->port;
+ switch (vmidi->seq_mode) {
+ case SNDRV_VIRMIDI_SEQ_DISPATCH:
+ ev->dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
+ break;
+ case SNDRV_VIRMIDI_SEQ_ATTACH:
+ /* FIXME: source and destination are same - not good.. */
+ ev->dest.client = vmidi->client;
+ ev->dest.port = vmidi->port;
+ break;
+ }
+ ev->type = SNDRV_SEQ_EVENT_NONE;
+}
+
+/*
+ * decode input event and put to read buffer of each opened file
+ */
+static int snd_virmidi_dev_receive_event(struct snd_virmidi_dev *rdev,
+ struct snd_seq_event *ev)
+{
+ struct snd_virmidi *vmidi;
+ unsigned char msg[4];
+ int len;
+
+ read_lock(&rdev->filelist_lock);
+ list_for_each_entry(vmidi, &rdev->filelist, list) {
+ if (!vmidi->trigger)
+ continue;
+ if (ev->type == SNDRV_SEQ_EVENT_SYSEX) {
+ if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
+ continue;
+ snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream);
+ } else {
+ len = snd_midi_event_decode(vmidi->parser, msg, sizeof(msg), ev);
+ if (len > 0)
+ snd_rawmidi_receive(vmidi->substream, msg, len);
+ }
+ }
+ read_unlock(&rdev->filelist_lock);
+
+ return 0;
+}
+
+/*
+ * receive an event from the remote virmidi port
+ *
+ * for rawmidi inputs, you can call this function from the event
+ * handler of a remote port which is attached to the virmidi via
+ * SNDRV_VIRMIDI_SEQ_ATTACH.
+ */
+#if 0
+int snd_virmidi_receive(struct snd_rawmidi *rmidi, struct snd_seq_event *ev)
+{
+ struct snd_virmidi_dev *rdev;
+
+ rdev = rmidi->private_data;
+ return snd_virmidi_dev_receive_event(rdev, ev);
+}
+#endif /* 0 */
+
+/*
+ * event handler of virmidi port
+ */
+static int snd_virmidi_event_input(struct snd_seq_event *ev, int direct,
+ void *private_data, int atomic, int hop)
+{
+ struct snd_virmidi_dev *rdev;
+
+ rdev = private_data;
+ if (!(rdev->flags & SNDRV_VIRMIDI_USE))
+ return 0; /* ignored */
+ return snd_virmidi_dev_receive_event(rdev, ev);
+}
+
+/*
+ * trigger rawmidi stream for input
+ */
+static void snd_virmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_virmidi *vmidi = substream->runtime->private_data;
+
+ if (up) {
+ vmidi->trigger = 1;
+ } else {
+ vmidi->trigger = 0;
+ }
+}
+
+/*
+ * trigger rawmidi stream for output
+ */
+static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_virmidi *vmidi = substream->runtime->private_data;
+ int count, res;
+ unsigned char buf[32], *pbuf;
+
+ if (up) {
+ vmidi->trigger = 1;
+ if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH &&
+ !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) {
+ snd_rawmidi_transmit_ack(substream, substream->runtime->buffer_size - substream->runtime->avail);
+ return; /* ignored */
+ }
+ if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) {
+ if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0)
+ return;
+ vmidi->event.type = SNDRV_SEQ_EVENT_NONE;
+ }
+ while (1) {
+ count = snd_rawmidi_transmit_peek(substream, buf, sizeof(buf));
+ if (count <= 0)
+ break;
+ pbuf = buf;
+ while (count > 0) {
+ res = snd_midi_event_encode(vmidi->parser, pbuf, count, &vmidi->event);
+ if (res < 0) {
+ snd_midi_event_reset_encode(vmidi->parser);
+ continue;
+ }
+ snd_rawmidi_transmit_ack(substream, res);
+ pbuf += res;
+ count -= res;
+ if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) {
+ if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0)
+ return;
+ vmidi->event.type = SNDRV_SEQ_EVENT_NONE;
+ }
+ }
+ }
+ } else {
+ vmidi->trigger = 0;
+ }
+}
+
+/*
+ * open rawmidi handle for input
+ */
+static int snd_virmidi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_virmidi_dev *rdev = substream->rmidi->private_data;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+ struct snd_virmidi *vmidi;
+ unsigned long flags;
+
+ vmidi = kzalloc(sizeof(*vmidi), GFP_KERNEL);
+ if (vmidi == NULL)
+ return -ENOMEM;
+ vmidi->substream = substream;
+ if (snd_midi_event_new(0, &vmidi->parser) < 0) {
+ kfree(vmidi);
+ return -ENOMEM;
+ }
+ vmidi->seq_mode = rdev->seq_mode;
+ vmidi->client = rdev->client;
+ vmidi->port = rdev->port;
+ runtime->private_data = vmidi;
+ write_lock_irqsave(&rdev->filelist_lock, flags);
+ list_add_tail(&vmidi->list, &rdev->filelist);
+ write_unlock_irqrestore(&rdev->filelist_lock, flags);
+ vmidi->rdev = rdev;
+ return 0;
+}
+
+/*
+ * open rawmidi handle for output
+ */
+static int snd_virmidi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_virmidi_dev *rdev = substream->rmidi->private_data;
+ struct snd_rawmidi_runtime *runtime = substream->runtime;
+ struct snd_virmidi *vmidi;
+
+ vmidi = kzalloc(sizeof(*vmidi), GFP_KERNEL);
+ if (vmidi == NULL)
+ return -ENOMEM;
+ vmidi->substream = substream;
+ if (snd_midi_event_new(MAX_MIDI_EVENT_BUF, &vmidi->parser) < 0) {
+ kfree(vmidi);
+ return -ENOMEM;
+ }
+ vmidi->seq_mode = rdev->seq_mode;
+ vmidi->client = rdev->client;
+ vmidi->port = rdev->port;
+ snd_virmidi_init_event(vmidi, &vmidi->event);
+ vmidi->rdev = rdev;
+ runtime->private_data = vmidi;
+ return 0;
+}
+
+/*
+ * close rawmidi handle for input
+ */
+static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_virmidi *vmidi = substream->runtime->private_data;
+ snd_midi_event_free(vmidi->parser);
+ list_del(&vmidi->list);
+ substream->runtime->private_data = NULL;
+ kfree(vmidi);
+ return 0;
+}
+
+/*
+ * close rawmidi handle for output
+ */
+static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_virmidi *vmidi = substream->runtime->private_data;
+ snd_midi_event_free(vmidi->parser);
+ substream->runtime->private_data = NULL;
+ kfree(vmidi);
+ return 0;
+}
+
+/*
+ * subscribe callback - allow output to rawmidi device
+ */
+static int snd_virmidi_subscribe(void *private_data,
+ struct snd_seq_port_subscribe *info)
+{
+ struct snd_virmidi_dev *rdev;
+
+ rdev = private_data;
+ if (!try_module_get(rdev->card->module))
+ return -EFAULT;
+ rdev->flags |= SNDRV_VIRMIDI_SUBSCRIBE;
+ return 0;
+}
+
+/*
+ * unsubscribe callback - disallow output to rawmidi device
+ */
+static int snd_virmidi_unsubscribe(void *private_data,
+ struct snd_seq_port_subscribe *info)
+{
+ struct snd_virmidi_dev *rdev;
+
+ rdev = private_data;
+ rdev->flags &= ~SNDRV_VIRMIDI_SUBSCRIBE;
+ module_put(rdev->card->module);
+ return 0;
+}
+
+
+/*
+ * use callback - allow input to rawmidi device
+ */
+static int snd_virmidi_use(void *private_data,
+ struct snd_seq_port_subscribe *info)
+{
+ struct snd_virmidi_dev *rdev;
+
+ rdev = private_data;
+ if (!try_module_get(rdev->card->module))
+ return -EFAULT;
+ rdev->flags |= SNDRV_VIRMIDI_USE;
+ return 0;
+}
+
+/*
+ * unuse callback - disallow input to rawmidi device
+ */
+static int snd_virmidi_unuse(void *private_data,
+ struct snd_seq_port_subscribe *info)
+{
+ struct snd_virmidi_dev *rdev;
+
+ rdev = private_data;
+ rdev->flags &= ~SNDRV_VIRMIDI_USE;
+ module_put(rdev->card->module);
+ return 0;
+}
+
+
+/*
+ * Register functions
+ */
+
+static struct snd_rawmidi_ops snd_virmidi_input_ops = {
+ .open = snd_virmidi_input_open,
+ .close = snd_virmidi_input_close,
+ .trigger = snd_virmidi_input_trigger,
+};
+
+static struct snd_rawmidi_ops snd_virmidi_output_ops = {
+ .open = snd_virmidi_output_open,
+ .close = snd_virmidi_output_close,
+ .trigger = snd_virmidi_output_trigger,
+};
+
+/*
+ * create a sequencer client and a port
+ */
+static int snd_virmidi_dev_attach_seq(struct snd_virmidi_dev *rdev)
+{
+ int client;
+ struct snd_seq_port_callback pcallbacks;
+ struct snd_seq_port_info *pinfo;
+ int err;
+
+ if (rdev->client >= 0)
+ return 0;
+
+ pinfo = kzalloc(sizeof(*pinfo), GFP_KERNEL);
+ if (!pinfo) {
+ err = -ENOMEM;
+ goto __error;
+ }
+
+ client = snd_seq_create_kernel_client(rdev->card, rdev->device,
+ "%s %d-%d", rdev->rmidi->name,
+ rdev->card->number,
+ rdev->device);
+ if (client < 0) {
+ err = client;
+ goto __error;
+ }
+ rdev->client = client;
+
+ /* create a port */
+ pinfo->addr.client = client;
+ sprintf(pinfo->name, "VirMIDI %d-%d", rdev->card->number, rdev->device);
+ /* set all capabilities */
+ pinfo->capability |= SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SYNC_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE;
+ pinfo->capability |= SNDRV_SEQ_PORT_CAP_READ | SNDRV_SEQ_PORT_CAP_SYNC_READ | SNDRV_SEQ_PORT_CAP_SUBS_READ;
+ pinfo->capability |= SNDRV_SEQ_PORT_CAP_DUPLEX;
+ pinfo->type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC
+ | SNDRV_SEQ_PORT_TYPE_SOFTWARE
+ | SNDRV_SEQ_PORT_TYPE_PORT;
+ pinfo->midi_channels = 16;
+ memset(&pcallbacks, 0, sizeof(pcallbacks));
+ pcallbacks.owner = THIS_MODULE;
+ pcallbacks.private_data = rdev;
+ pcallbacks.subscribe = snd_virmidi_subscribe;
+ pcallbacks.unsubscribe = snd_virmidi_unsubscribe;
+ pcallbacks.use = snd_virmidi_use;
+ pcallbacks.unuse = snd_virmidi_unuse;
+ pcallbacks.event_input = snd_virmidi_event_input;
+ pinfo->kernel = &pcallbacks;
+ err = snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_CREATE_PORT, pinfo);
+ if (err < 0) {
+ snd_seq_delete_kernel_client(client);
+ rdev->client = -1;
+ goto __error;
+ }
+
+ rdev->port = pinfo->addr.port;
+ err = 0; /* success */
+
+ __error:
+ kfree(pinfo);
+ return err;
+}
+
+
+/*
+ * release the sequencer client
+ */
+static void snd_virmidi_dev_detach_seq(struct snd_virmidi_dev *rdev)
+{
+ if (rdev->client >= 0) {
+ snd_seq_delete_kernel_client(rdev->client);
+ rdev->client = -1;
+ }
+}
+
+/*
+ * register the device
+ */
+static int snd_virmidi_dev_register(struct snd_rawmidi *rmidi)
+{
+ struct snd_virmidi_dev *rdev = rmidi->private_data;
+ int err;
+
+ switch (rdev->seq_mode) {
+ case SNDRV_VIRMIDI_SEQ_DISPATCH:
+ err = snd_virmidi_dev_attach_seq(rdev);
+ if (err < 0)
+ return err;
+ break;
+ case SNDRV_VIRMIDI_SEQ_ATTACH:
+ if (rdev->client == 0)
+ return -EINVAL;
+ /* should check presence of port more strictly.. */
+ break;
+ default:
+ snd_printk(KERN_ERR "seq_mode is not set: %d\n", rdev->seq_mode);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/*
+ * unregister the device
+ */
+static int snd_virmidi_dev_unregister(struct snd_rawmidi *rmidi)
+{
+ struct snd_virmidi_dev *rdev = rmidi->private_data;
+
+ if (rdev->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH)
+ snd_virmidi_dev_detach_seq(rdev);
+ return 0;
+}
+
+/*
+ *
+ */
+static struct snd_rawmidi_global_ops snd_virmidi_global_ops = {
+ .dev_register = snd_virmidi_dev_register,
+ .dev_unregister = snd_virmidi_dev_unregister,
+};
+
+/*
+ * free device
+ */
+static void snd_virmidi_free(struct snd_rawmidi *rmidi)
+{
+ struct snd_virmidi_dev *rdev = rmidi->private_data;
+ kfree(rdev);
+}
+
+/*
+ * create a new device
+ *
+ */
+/* exported */
+int snd_virmidi_new(struct snd_card *card, int device, struct snd_rawmidi **rrmidi)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_virmidi_dev *rdev;
+ int err;
+
+ *rrmidi = NULL;
+ if ((err = snd_rawmidi_new(card, "VirMidi", device,
+ 16, /* may be configurable */
+ 16, /* may be configurable */
+ &rmidi)) < 0)
+ return err;
+ strcpy(rmidi->name, rmidi->id);
+ rdev = kzalloc(sizeof(*rdev), GFP_KERNEL);
+ if (rdev == NULL) {
+ snd_device_free(card, rmidi);
+ return -ENOMEM;
+ }
+ rdev->card = card;
+ rdev->rmidi = rmidi;
+ rdev->device = device;
+ rdev->client = -1;
+ rwlock_init(&rdev->filelist_lock);
+ INIT_LIST_HEAD(&rdev->filelist);
+ rdev->seq_mode = SNDRV_VIRMIDI_SEQ_DISPATCH;
+ rmidi->private_data = rdev;
+ rmidi->private_free = snd_virmidi_free;
+ rmidi->ops = &snd_virmidi_global_ops;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_virmidi_input_ops);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_virmidi_output_ops);
+ rmidi->info_flags = SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ *rrmidi = rmidi;
+ return 0;
+}
+
+/*
+ * ENTRY functions
+ */
+
+static int __init alsa_virmidi_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_virmidi_exit(void)
+{
+}
+
+module_init(alsa_virmidi_init)
+module_exit(alsa_virmidi_exit)
+
+EXPORT_SYMBOL(snd_virmidi_new);
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
new file mode 100644
index 0000000..4e7ec2b
--- /dev/null
+++ b/sound/core/sgbuf.c
@@ -0,0 +1,138 @@
+/*
+ * Scatter-Gather buffer
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/mm.h>
+#include <linux/vmalloc.h>
+#include <sound/memalloc.h>
+
+
+/* table entries are align to 32 */
+#define SGBUF_TBL_ALIGN 32
+#define sgbuf_align_table(tbl) ALIGN((tbl), SGBUF_TBL_ALIGN)
+
+int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
+{
+ struct snd_sg_buf *sgbuf = dmab->private_data;
+ struct snd_dma_buffer tmpb;
+ int i;
+
+ if (! sgbuf)
+ return -EINVAL;
+
+ if (dmab->area)
+ vunmap(dmab->area);
+ dmab->area = NULL;
+
+ tmpb.dev.type = SNDRV_DMA_TYPE_DEV;
+ tmpb.dev.dev = sgbuf->dev;
+ for (i = 0; i < sgbuf->pages; i++) {
+ if (!(sgbuf->table[i].addr & ~PAGE_MASK))
+ continue; /* continuous pages */
+ tmpb.area = sgbuf->table[i].buf;
+ tmpb.addr = sgbuf->table[i].addr & PAGE_MASK;
+ tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT;
+ snd_dma_free_pages(&tmpb);
+ }
+
+ kfree(sgbuf->table);
+ kfree(sgbuf->page_table);
+ kfree(sgbuf);
+ dmab->private_data = NULL;
+
+ return 0;
+}
+
+#define MAX_ALLOC_PAGES 32
+
+void *snd_malloc_sgbuf_pages(struct device *device,
+ size_t size, struct snd_dma_buffer *dmab,
+ size_t *res_size)
+{
+ struct snd_sg_buf *sgbuf;
+ unsigned int i, pages, chunk, maxpages;
+ struct snd_dma_buffer tmpb;
+ struct snd_sg_page *table;
+ struct page **pgtable;
+
+ dmab->area = NULL;
+ dmab->addr = 0;
+ dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL);
+ if (! sgbuf)
+ return NULL;
+ sgbuf->dev = device;
+ pages = snd_sgbuf_aligned_pages(size);
+ sgbuf->tblsize = sgbuf_align_table(pages);
+ table = kcalloc(sgbuf->tblsize, sizeof(*table), GFP_KERNEL);
+ if (!table)
+ goto _failed;
+ sgbuf->table = table;
+ pgtable = kcalloc(sgbuf->tblsize, sizeof(*pgtable), GFP_KERNEL);
+ if (!pgtable)
+ goto _failed;
+ sgbuf->page_table = pgtable;
+
+ /* allocate pages */
+ maxpages = MAX_ALLOC_PAGES;
+ while (pages > 0) {
+ chunk = pages;
+ /* don't be too eager to take a huge chunk */
+ if (chunk > maxpages)
+ chunk = maxpages;
+ chunk <<= PAGE_SHIFT;
+ if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device,
+ chunk, &tmpb) < 0) {
+ if (!sgbuf->pages)
+ return NULL;
+ if (!res_size)
+ goto _failed;
+ size = sgbuf->pages * PAGE_SIZE;
+ break;
+ }
+ chunk = tmpb.bytes >> PAGE_SHIFT;
+ for (i = 0; i < chunk; i++) {
+ table->buf = tmpb.area;
+ table->addr = tmpb.addr;
+ if (!i)
+ table->addr |= chunk; /* mark head */
+ table++;
+ *pgtable++ = virt_to_page(tmpb.area);
+ tmpb.area += PAGE_SIZE;
+ tmpb.addr += PAGE_SIZE;
+ }
+ sgbuf->pages += chunk;
+ pages -= chunk;
+ if (chunk < maxpages)
+ maxpages = chunk;
+ }
+
+ sgbuf->size = size;
+ dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL);
+ if (! dmab->area)
+ goto _failed;
+ if (res_size)
+ *res_size = sgbuf->size;
+ return dmab->area;
+
+ _failed:
+ snd_free_sgbuf_pages(dmab); /* free the table */
+ return NULL;
+}
diff --git a/sound/core/sound.c b/sound/core/sound.c
new file mode 100644
index 0000000..44a69bb
--- /dev/null
+++ b/sound/core/sound.c
@@ -0,0 +1,468 @@
+/*
+ * Advanced Linux Sound Architecture
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/smp_lock.h>
+#include <linux/time.h>
+#include <linux/device.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/info.h>
+#include <sound/version.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <linux/kmod.h>
+#include <linux/mutex.h>
+
+static int major = CONFIG_SND_MAJOR;
+int snd_major;
+EXPORT_SYMBOL(snd_major);
+
+static int cards_limit = 1;
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Advanced Linux Sound Architecture driver for soundcards.");
+MODULE_LICENSE("GPL");
+module_param(major, int, 0444);
+MODULE_PARM_DESC(major, "Major # for sound driver.");
+module_param(cards_limit, int, 0444);
+MODULE_PARM_DESC(cards_limit, "Count of auto-loadable soundcards.");
+MODULE_ALIAS_CHARDEV_MAJOR(CONFIG_SND_MAJOR);
+
+/* this one holds the actual max. card number currently available.
+ * as default, it's identical with cards_limit option. when more
+ * modules are loaded manually, this limit number increases, too.
+ */
+int snd_ecards_limit;
+EXPORT_SYMBOL(snd_ecards_limit);
+
+static struct snd_minor *snd_minors[SNDRV_OS_MINORS];
+static DEFINE_MUTEX(sound_mutex);
+
+#ifdef CONFIG_MODULES
+
+/**
+ * snd_request_card - try to load the card module
+ * @card: the card number
+ *
+ * Tries to load the module "snd-card-X" for the given card number
+ * via request_module. Returns immediately if already loaded.
+ */
+void snd_request_card(int card)
+{
+ if (snd_card_locked(card))
+ return;
+ if (card < 0 || card >= cards_limit)
+ return;
+ request_module("snd-card-%i", card);
+}
+
+EXPORT_SYMBOL(snd_request_card);
+
+static void snd_request_other(int minor)
+{
+ char *str;
+
+ switch (minor) {
+ case SNDRV_MINOR_SEQUENCER: str = "snd-seq"; break;
+ case SNDRV_MINOR_TIMER: str = "snd-timer"; break;
+ default: return;
+ }
+ request_module(str);
+}
+
+#endif /* modular kernel */
+
+/**
+ * snd_lookup_minor_data - get user data of a registered device
+ * @minor: the minor number
+ * @type: device type (SNDRV_DEVICE_TYPE_XXX)
+ *
+ * Checks that a minor device with the specified type is registered, and returns
+ * its user data pointer.
+ */
+void *snd_lookup_minor_data(unsigned int minor, int type)
+{
+ struct snd_minor *mreg;
+ void *private_data;
+
+ if (minor >= ARRAY_SIZE(snd_minors))
+ return NULL;
+ mutex_lock(&sound_mutex);
+ mreg = snd_minors[minor];
+ if (mreg && mreg->type == type)
+ private_data = mreg->private_data;
+ else
+ private_data = NULL;
+ mutex_unlock(&sound_mutex);
+ return private_data;
+}
+
+EXPORT_SYMBOL(snd_lookup_minor_data);
+
+static int __snd_open(struct inode *inode, struct file *file)
+{
+ unsigned int minor = iminor(inode);
+ struct snd_minor *mptr = NULL;
+ const struct file_operations *old_fops;
+ int err = 0;
+
+ if (minor >= ARRAY_SIZE(snd_minors))
+ return -ENODEV;
+ mptr = snd_minors[minor];
+ if (mptr == NULL) {
+#ifdef CONFIG_MODULES
+ int dev = SNDRV_MINOR_DEVICE(minor);
+ if (dev == SNDRV_MINOR_CONTROL) {
+ /* /dev/aloadC? */
+ int card = SNDRV_MINOR_CARD(minor);
+ if (snd_cards[card] == NULL)
+ snd_request_card(card);
+ } else if (dev == SNDRV_MINOR_GLOBAL) {
+ /* /dev/aloadSEQ */
+ snd_request_other(minor);
+ }
+#ifndef CONFIG_SND_DYNAMIC_MINORS
+ /* /dev/snd/{controlC?,seq} */
+ mptr = snd_minors[minor];
+ if (mptr == NULL)
+#endif
+#endif
+ return -ENODEV;
+ }
+ old_fops = file->f_op;
+ file->f_op = fops_get(mptr->f_ops);
+ if (file->f_op->open)
+ err = file->f_op->open(inode, file);
+ if (err) {
+ fops_put(file->f_op);
+ file->f_op = fops_get(old_fops);
+ }
+ fops_put(old_fops);
+ return err;
+}
+
+
+/* BKL pushdown: nasty #ifdef avoidance wrapper */
+static int snd_open(struct inode *inode, struct file *file)
+{
+ int ret;
+
+ lock_kernel();
+ ret = __snd_open(inode, file);
+ unlock_kernel();
+ return ret;
+}
+
+static const struct file_operations snd_fops =
+{
+ .owner = THIS_MODULE,
+ .open = snd_open
+};
+
+#ifdef CONFIG_SND_DYNAMIC_MINORS
+static int snd_find_free_minor(void)
+{
+ int minor;
+
+ for (minor = 0; minor < ARRAY_SIZE(snd_minors); ++minor) {
+ /* skip minors still used statically for autoloading devices */
+ if (SNDRV_MINOR_DEVICE(minor) == SNDRV_MINOR_CONTROL ||
+ minor == SNDRV_MINOR_SEQUENCER)
+ continue;
+ if (!snd_minors[minor])
+ return minor;
+ }
+ return -EBUSY;
+}
+#else
+static int snd_kernel_minor(int type, struct snd_card *card, int dev)
+{
+ int minor;
+
+ switch (type) {
+ case SNDRV_DEVICE_TYPE_SEQUENCER:
+ case SNDRV_DEVICE_TYPE_TIMER:
+ minor = type;
+ break;
+ case SNDRV_DEVICE_TYPE_CONTROL:
+ if (snd_BUG_ON(!card))
+ return -EINVAL;
+ minor = SNDRV_MINOR(card->number, type);
+ break;
+ case SNDRV_DEVICE_TYPE_HWDEP:
+ case SNDRV_DEVICE_TYPE_RAWMIDI:
+ case SNDRV_DEVICE_TYPE_PCM_PLAYBACK:
+ case SNDRV_DEVICE_TYPE_PCM_CAPTURE:
+ if (snd_BUG_ON(!card))
+ return -EINVAL;
+ minor = SNDRV_MINOR(card->number, type + dev);
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OS_MINORS))
+ return -EINVAL;
+ return minor;
+}
+#endif
+
+/**
+ * snd_register_device_for_dev - Register the ALSA device file for the card
+ * @type: the device type, SNDRV_DEVICE_TYPE_XXX
+ * @card: the card instance
+ * @dev: the device index
+ * @f_ops: the file operations
+ * @private_data: user pointer for f_ops->open()
+ * @name: the device file name
+ * @device: the &struct device to link this new device to
+ *
+ * Registers an ALSA device file for the given card.
+ * The operators have to be set in reg parameter.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_register_device_for_dev(int type, struct snd_card *card, int dev,
+ const struct file_operations *f_ops,
+ void *private_data,
+ const char *name, struct device *device)
+{
+ int minor;
+ struct snd_minor *preg;
+
+ if (snd_BUG_ON(!name))
+ return -EINVAL;
+ preg = kmalloc(sizeof *preg, GFP_KERNEL);
+ if (preg == NULL)
+ return -ENOMEM;
+ preg->type = type;
+ preg->card = card ? card->number : -1;
+ preg->device = dev;
+ preg->f_ops = f_ops;
+ preg->private_data = private_data;
+ mutex_lock(&sound_mutex);
+#ifdef CONFIG_SND_DYNAMIC_MINORS
+ minor = snd_find_free_minor();
+#else
+ minor = snd_kernel_minor(type, card, dev);
+ if (minor >= 0 && snd_minors[minor])
+ minor = -EBUSY;
+#endif
+ if (minor < 0) {
+ mutex_unlock(&sound_mutex);
+ kfree(preg);
+ return minor;
+ }
+ snd_minors[minor] = preg;
+ preg->dev = device_create(sound_class, device, MKDEV(major, minor),
+ private_data, "%s", name);
+ if (IS_ERR(preg->dev)) {
+ snd_minors[minor] = NULL;
+ mutex_unlock(&sound_mutex);
+ minor = PTR_ERR(preg->dev);
+ kfree(preg);
+ return minor;
+ }
+
+ mutex_unlock(&sound_mutex);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_register_device_for_dev);
+
+/* find the matching minor record
+ * return the index of snd_minor, or -1 if not found
+ */
+static int find_snd_minor(int type, struct snd_card *card, int dev)
+{
+ int cardnum, minor;
+ struct snd_minor *mptr;
+
+ cardnum = card ? card->number : -1;
+ for (minor = 0; minor < ARRAY_SIZE(snd_minors); ++minor)
+ if ((mptr = snd_minors[minor]) != NULL &&
+ mptr->type == type &&
+ mptr->card == cardnum &&
+ mptr->device == dev)
+ return minor;
+ return -1;
+}
+
+/**
+ * snd_unregister_device - unregister the device on the given card
+ * @type: the device type, SNDRV_DEVICE_TYPE_XXX
+ * @card: the card instance
+ * @dev: the device index
+ *
+ * Unregisters the device file already registered via
+ * snd_register_device().
+ *
+ * Returns zero if sucecessful, or a negative error code on failure
+ */
+int snd_unregister_device(int type, struct snd_card *card, int dev)
+{
+ int minor;
+
+ mutex_lock(&sound_mutex);
+ minor = find_snd_minor(type, card, dev);
+ if (minor < 0) {
+ mutex_unlock(&sound_mutex);
+ return -EINVAL;
+ }
+
+ device_destroy(sound_class, MKDEV(major, minor));
+
+ kfree(snd_minors[minor]);
+ snd_minors[minor] = NULL;
+ mutex_unlock(&sound_mutex);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_unregister_device);
+
+int snd_add_device_sysfs_file(int type, struct snd_card *card, int dev,
+ struct device_attribute *attr)
+{
+ int minor, ret = -EINVAL;
+ struct device *d;
+
+ mutex_lock(&sound_mutex);
+ minor = find_snd_minor(type, card, dev);
+ if (minor >= 0 && (d = snd_minors[minor]->dev) != NULL)
+ ret = device_create_file(d, attr);
+ mutex_unlock(&sound_mutex);
+ return ret;
+
+}
+
+EXPORT_SYMBOL(snd_add_device_sysfs_file);
+
+#ifdef CONFIG_PROC_FS
+/*
+ * INFO PART
+ */
+
+static struct snd_info_entry *snd_minor_info_entry;
+
+static const char *snd_device_type_name(int type)
+{
+ switch (type) {
+ case SNDRV_DEVICE_TYPE_CONTROL:
+ return "control";
+ case SNDRV_DEVICE_TYPE_HWDEP:
+ return "hardware dependent";
+ case SNDRV_DEVICE_TYPE_RAWMIDI:
+ return "raw midi";
+ case SNDRV_DEVICE_TYPE_PCM_PLAYBACK:
+ return "digital audio playback";
+ case SNDRV_DEVICE_TYPE_PCM_CAPTURE:
+ return "digital audio capture";
+ case SNDRV_DEVICE_TYPE_SEQUENCER:
+ return "sequencer";
+ case SNDRV_DEVICE_TYPE_TIMER:
+ return "timer";
+ default:
+ return "?";
+ }
+}
+
+static void snd_minor_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ int minor;
+ struct snd_minor *mptr;
+
+ mutex_lock(&sound_mutex);
+ for (minor = 0; minor < SNDRV_OS_MINORS; ++minor) {
+ if (!(mptr = snd_minors[minor]))
+ continue;
+ if (mptr->card >= 0) {
+ if (mptr->device >= 0)
+ snd_iprintf(buffer, "%3i: [%2i-%2i]: %s\n",
+ minor, mptr->card, mptr->device,
+ snd_device_type_name(mptr->type));
+ else
+ snd_iprintf(buffer, "%3i: [%2i] : %s\n",
+ minor, mptr->card,
+ snd_device_type_name(mptr->type));
+ } else
+ snd_iprintf(buffer, "%3i: : %s\n", minor,
+ snd_device_type_name(mptr->type));
+ }
+ mutex_unlock(&sound_mutex);
+}
+
+int __init snd_minor_info_init(void)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_module_entry(THIS_MODULE, "devices", NULL);
+ if (entry) {
+ entry->c.text.read = snd_minor_info_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ snd_minor_info_entry = entry;
+ return 0;
+}
+
+int __exit snd_minor_info_done(void)
+{
+ snd_info_free_entry(snd_minor_info_entry);
+ return 0;
+}
+#endif /* CONFIG_PROC_FS */
+
+/*
+ * INIT PART
+ */
+
+static int __init alsa_sound_init(void)
+{
+ snd_major = major;
+ snd_ecards_limit = cards_limit;
+ if (register_chrdev(major, "alsa", &snd_fops)) {
+ snd_printk(KERN_ERR "unable to register native major device number %d\n", major);
+ return -EIO;
+ }
+ if (snd_info_init() < 0) {
+ unregister_chrdev(major, "alsa");
+ return -ENOMEM;
+ }
+ snd_info_minor_register();
+#ifndef MODULE
+ printk(KERN_INFO "Advanced Linux Sound Architecture Driver Version " CONFIG_SND_VERSION CONFIG_SND_DATE ".\n");
+#endif
+ return 0;
+}
+
+static void __exit alsa_sound_exit(void)
+{
+ snd_info_minor_unregister();
+ snd_info_done();
+ unregister_chrdev(major, "alsa");
+}
+
+module_init(alsa_sound_init)
+module_exit(alsa_sound_exit)
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
new file mode 100644
index 0000000..7fe1226
--- /dev/null
+++ b/sound/core/sound_oss.c
@@ -0,0 +1,278 @@
+/*
+ * Advanced Linux Sound Architecture
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifdef CONFIG_SND_OSSEMUL
+
+#if !defined(CONFIG_SOUND) && !(defined(MODULE) && defined(CONFIG_SOUND_MODULE))
+#error "Enable the OSS soundcore multiplexer (CONFIG_SOUND) in the kernel."
+#endif
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/minors.h>
+#include <sound/info.h>
+#include <linux/sound.h>
+#include <linux/mutex.h>
+
+#define SNDRV_OSS_MINORS 128
+
+static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS];
+static DEFINE_MUTEX(sound_oss_mutex);
+
+void *snd_lookup_oss_minor_data(unsigned int minor, int type)
+{
+ struct snd_minor *mreg;
+ void *private_data;
+
+ if (minor >= ARRAY_SIZE(snd_oss_minors))
+ return NULL;
+ mutex_lock(&sound_oss_mutex);
+ mreg = snd_oss_minors[minor];
+ if (mreg && mreg->type == type)
+ private_data = mreg->private_data;
+ else
+ private_data = NULL;
+ mutex_unlock(&sound_oss_mutex);
+ return private_data;
+}
+
+EXPORT_SYMBOL(snd_lookup_oss_minor_data);
+
+static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
+{
+ int minor;
+
+ switch (type) {
+ case SNDRV_OSS_DEVICE_TYPE_MIXER:
+ if (snd_BUG_ON(!card || dev < 0 || dev > 1))
+ return -EINVAL;
+ minor = SNDRV_MINOR_OSS(card->number, (dev ? SNDRV_MINOR_OSS_MIXER1 : SNDRV_MINOR_OSS_MIXER));
+ break;
+ case SNDRV_OSS_DEVICE_TYPE_SEQUENCER:
+ minor = SNDRV_MINOR_OSS_SEQUENCER;
+ break;
+ case SNDRV_OSS_DEVICE_TYPE_MUSIC:
+ minor = SNDRV_MINOR_OSS_MUSIC;
+ break;
+ case SNDRV_OSS_DEVICE_TYPE_PCM:
+ if (snd_BUG_ON(!card || dev < 0 || dev > 1))
+ return -EINVAL;
+ minor = SNDRV_MINOR_OSS(card->number, (dev ? SNDRV_MINOR_OSS_PCM1 : SNDRV_MINOR_OSS_PCM));
+ break;
+ case SNDRV_OSS_DEVICE_TYPE_MIDI:
+ if (snd_BUG_ON(!card || dev < 0 || dev > 1))
+ return -EINVAL;
+ minor = SNDRV_MINOR_OSS(card->number, (dev ? SNDRV_MINOR_OSS_MIDI1 : SNDRV_MINOR_OSS_MIDI));
+ break;
+ case SNDRV_OSS_DEVICE_TYPE_DMFM:
+ minor = SNDRV_MINOR_OSS(card->number, SNDRV_MINOR_OSS_DMFM);
+ break;
+ case SNDRV_OSS_DEVICE_TYPE_SNDSTAT:
+ minor = SNDRV_MINOR_OSS_SNDSTAT;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+ return -EINVAL;
+ return minor;
+}
+
+int snd_register_oss_device(int type, struct snd_card *card, int dev,
+ const struct file_operations *f_ops, void *private_data,
+ const char *name)
+{
+ int minor = snd_oss_kernel_minor(type, card, dev);
+ int minor_unit;
+ struct snd_minor *preg;
+ int cidx = SNDRV_MINOR_OSS_CARD(minor);
+ int track2 = -1;
+ int register1 = -1, register2 = -1;
+ struct device *carddev = snd_card_get_device_link(card);
+
+ if (card && card->number >= 8)
+ return 0; /* ignore silently */
+ if (minor < 0)
+ return minor;
+ preg = kmalloc(sizeof(struct snd_minor), GFP_KERNEL);
+ if (preg == NULL)
+ return -ENOMEM;
+ preg->type = type;
+ preg->card = card ? card->number : -1;
+ preg->device = dev;
+ preg->f_ops = f_ops;
+ preg->private_data = private_data;
+ mutex_lock(&sound_oss_mutex);
+ snd_oss_minors[minor] = preg;
+ minor_unit = SNDRV_MINOR_OSS_DEVICE(minor);
+ switch (minor_unit) {
+ case SNDRV_MINOR_OSS_PCM:
+ track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_AUDIO);
+ break;
+ case SNDRV_MINOR_OSS_MIDI:
+ track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_DMMIDI);
+ break;
+ case SNDRV_MINOR_OSS_MIDI1:
+ track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_DMMIDI1);
+ break;
+ }
+ register1 = register_sound_special_device(f_ops, minor, carddev);
+ if (register1 != minor)
+ goto __end;
+ if (track2 >= 0) {
+ register2 = register_sound_special_device(f_ops, track2,
+ carddev);
+ if (register2 != track2)
+ goto __end;
+ snd_oss_minors[track2] = preg;
+ }
+ mutex_unlock(&sound_oss_mutex);
+ return 0;
+
+ __end:
+ if (register2 >= 0)
+ unregister_sound_special(register2);
+ if (register1 >= 0)
+ unregister_sound_special(register1);
+ snd_oss_minors[minor] = NULL;
+ mutex_unlock(&sound_oss_mutex);
+ kfree(preg);
+ return -EBUSY;
+}
+
+EXPORT_SYMBOL(snd_register_oss_device);
+
+int snd_unregister_oss_device(int type, struct snd_card *card, int dev)
+{
+ int minor = snd_oss_kernel_minor(type, card, dev);
+ int cidx = SNDRV_MINOR_OSS_CARD(minor);
+ int track2 = -1;
+ struct snd_minor *mptr;
+
+ if (card && card->number >= 8)
+ return 0;
+ if (minor < 0)
+ return minor;
+ mutex_lock(&sound_oss_mutex);
+ mptr = snd_oss_minors[minor];
+ if (mptr == NULL) {
+ mutex_unlock(&sound_oss_mutex);
+ return -ENOENT;
+ }
+ unregister_sound_special(minor);
+ switch (SNDRV_MINOR_OSS_DEVICE(minor)) {
+ case SNDRV_MINOR_OSS_PCM:
+ track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_AUDIO);
+ break;
+ case SNDRV_MINOR_OSS_MIDI:
+ track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_DMMIDI);
+ break;
+ case SNDRV_MINOR_OSS_MIDI1:
+ track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_DMMIDI1);
+ break;
+ }
+ if (track2 >= 0) {
+ unregister_sound_special(track2);
+ snd_oss_minors[track2] = NULL;
+ }
+ snd_oss_minors[minor] = NULL;
+ mutex_unlock(&sound_oss_mutex);
+ kfree(mptr);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_unregister_oss_device);
+
+/*
+ * INFO PART
+ */
+
+#ifdef CONFIG_PROC_FS
+
+static struct snd_info_entry *snd_minor_info_oss_entry;
+
+static const char *snd_oss_device_type_name(int type)
+{
+ switch (type) {
+ case SNDRV_OSS_DEVICE_TYPE_MIXER:
+ return "mixer";
+ case SNDRV_OSS_DEVICE_TYPE_SEQUENCER:
+ case SNDRV_OSS_DEVICE_TYPE_MUSIC:
+ return "sequencer";
+ case SNDRV_OSS_DEVICE_TYPE_PCM:
+ return "digital audio";
+ case SNDRV_OSS_DEVICE_TYPE_MIDI:
+ return "raw midi";
+ case SNDRV_OSS_DEVICE_TYPE_DMFM:
+ return "hardware dependent";
+ default:
+ return "?";
+ }
+}
+
+static void snd_minor_info_oss_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int minor;
+ struct snd_minor *mptr;
+
+ mutex_lock(&sound_oss_mutex);
+ for (minor = 0; minor < SNDRV_OSS_MINORS; ++minor) {
+ if (!(mptr = snd_oss_minors[minor]))
+ continue;
+ if (mptr->card >= 0)
+ snd_iprintf(buffer, "%3i: [%i-%2i]: %s\n", minor,
+ mptr->card, mptr->device,
+ snd_oss_device_type_name(mptr->type));
+ else
+ snd_iprintf(buffer, "%3i: : %s\n", minor,
+ snd_oss_device_type_name(mptr->type));
+ }
+ mutex_unlock(&sound_oss_mutex);
+}
+
+
+int __init snd_minor_info_oss_init(void)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_module_entry(THIS_MODULE, "devices", snd_oss_root);
+ if (entry) {
+ entry->c.text.read = snd_minor_info_oss_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ snd_minor_info_oss_entry = entry;
+ return 0;
+}
+
+int __exit snd_minor_info_oss_done(void)
+{
+ snd_info_free_entry(snd_minor_info_oss_entry);
+ return 0;
+}
+#endif /* CONFIG_PROC_FS */
+
+#endif /* CONFIG_SND_OSSEMUL */
diff --git a/sound/core/timer.c b/sound/core/timer.c
new file mode 100644
index 0000000..c584408
--- /dev/null
+++ b/sound/core/timer.c
@@ -0,0 +1,1992 @@
+/*
+ * Timers abstract layer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/mutex.h>
+#include <linux/moduleparam.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/timer.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/minors.h>
+#include <sound/initval.h>
+#include <linux/kmod.h>
+
+#if defined(CONFIG_SND_HPET) || defined(CONFIG_SND_HPET_MODULE)
+#define DEFAULT_TIMER_LIMIT 3
+#elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE)
+#define DEFAULT_TIMER_LIMIT 2
+#else
+#define DEFAULT_TIMER_LIMIT 1
+#endif
+
+static int timer_limit = DEFAULT_TIMER_LIMIT;
+static int timer_tstamp_monotonic = 1;
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("ALSA timer interface");
+MODULE_LICENSE("GPL");
+module_param(timer_limit, int, 0444);
+MODULE_PARM_DESC(timer_limit, "Maximum global timers in system.");
+module_param(timer_tstamp_monotonic, int, 0444);
+MODULE_PARM_DESC(timer_tstamp_monotonic, "Use posix monotonic clock source for timestamps (default).");
+
+struct snd_timer_user {
+ struct snd_timer_instance *timeri;
+ int tread; /* enhanced read with timestamps and events */
+ unsigned long ticks;
+ unsigned long overrun;
+ int qhead;
+ int qtail;
+ int qused;
+ int queue_size;
+ struct snd_timer_read *queue;
+ struct snd_timer_tread *tqueue;
+ spinlock_t qlock;
+ unsigned long last_resolution;
+ unsigned int filter;
+ struct timespec tstamp; /* trigger tstamp */
+ wait_queue_head_t qchange_sleep;
+ struct fasync_struct *fasync;
+ struct mutex tread_sem;
+};
+
+/* list of timers */
+static LIST_HEAD(snd_timer_list);
+
+/* list of slave instances */
+static LIST_HEAD(snd_timer_slave_list);
+
+/* lock for slave active lists */
+static DEFINE_SPINLOCK(slave_active_lock);
+
+static DEFINE_MUTEX(register_mutex);
+
+static int snd_timer_free(struct snd_timer *timer);
+static int snd_timer_dev_free(struct snd_device *device);
+static int snd_timer_dev_register(struct snd_device *device);
+static int snd_timer_dev_disconnect(struct snd_device *device);
+
+static void snd_timer_reschedule(struct snd_timer * timer, unsigned long ticks_left);
+
+/*
+ * create a timer instance with the given owner string.
+ * when timer is not NULL, increments the module counter
+ */
+static struct snd_timer_instance *snd_timer_instance_new(char *owner,
+ struct snd_timer *timer)
+{
+ struct snd_timer_instance *timeri;
+ timeri = kzalloc(sizeof(*timeri), GFP_KERNEL);
+ if (timeri == NULL)
+ return NULL;
+ timeri->owner = kstrdup(owner, GFP_KERNEL);
+ if (! timeri->owner) {
+ kfree(timeri);
+ return NULL;
+ }
+ INIT_LIST_HEAD(&timeri->open_list);
+ INIT_LIST_HEAD(&timeri->active_list);
+ INIT_LIST_HEAD(&timeri->ack_list);
+ INIT_LIST_HEAD(&timeri->slave_list_head);
+ INIT_LIST_HEAD(&timeri->slave_active_head);
+
+ timeri->timer = timer;
+ if (timer && !try_module_get(timer->module)) {
+ kfree(timeri->owner);
+ kfree(timeri);
+ return NULL;
+ }
+
+ return timeri;
+}
+
+/*
+ * find a timer instance from the given timer id
+ */
+static struct snd_timer *snd_timer_find(struct snd_timer_id *tid)
+{
+ struct snd_timer *timer = NULL;
+
+ list_for_each_entry(timer, &snd_timer_list, device_list) {
+ if (timer->tmr_class != tid->dev_class)
+ continue;
+ if ((timer->tmr_class == SNDRV_TIMER_CLASS_CARD ||
+ timer->tmr_class == SNDRV_TIMER_CLASS_PCM) &&
+ (timer->card == NULL ||
+ timer->card->number != tid->card))
+ continue;
+ if (timer->tmr_device != tid->device)
+ continue;
+ if (timer->tmr_subdevice != tid->subdevice)
+ continue;
+ return timer;
+ }
+ return NULL;
+}
+
+#ifdef CONFIG_MODULES
+
+static void snd_timer_request(struct snd_timer_id *tid)
+{
+ switch (tid->dev_class) {
+ case SNDRV_TIMER_CLASS_GLOBAL:
+ if (tid->device < timer_limit)
+ request_module("snd-timer-%i", tid->device);
+ break;
+ case SNDRV_TIMER_CLASS_CARD:
+ case SNDRV_TIMER_CLASS_PCM:
+ if (tid->card < snd_ecards_limit)
+ request_module("snd-card-%i", tid->card);
+ break;
+ default:
+ break;
+ }
+}
+
+#endif
+
+/*
+ * look for a master instance matching with the slave id of the given slave.
+ * when found, relink the open_link of the slave.
+ *
+ * call this with register_mutex down.
+ */
+static void snd_timer_check_slave(struct snd_timer_instance *slave)
+{
+ struct snd_timer *timer;
+ struct snd_timer_instance *master;
+
+ /* FIXME: it's really dumb to look up all entries.. */
+ list_for_each_entry(timer, &snd_timer_list, device_list) {
+ list_for_each_entry(master, &timer->open_list_head, open_list) {
+ if (slave->slave_class == master->slave_class &&
+ slave->slave_id == master->slave_id) {
+ list_del(&slave->open_list);
+ list_add_tail(&slave->open_list,
+ &master->slave_list_head);
+ spin_lock_irq(&slave_active_lock);
+ slave->master = master;
+ slave->timer = master->timer;
+ spin_unlock_irq(&slave_active_lock);
+ return;
+ }
+ }
+ }
+}
+
+/*
+ * look for slave instances matching with the slave id of the given master.
+ * when found, relink the open_link of slaves.
+ *
+ * call this with register_mutex down.
+ */
+static void snd_timer_check_master(struct snd_timer_instance *master)
+{
+ struct snd_timer_instance *slave, *tmp;
+
+ /* check all pending slaves */
+ list_for_each_entry_safe(slave, tmp, &snd_timer_slave_list, open_list) {
+ if (slave->slave_class == master->slave_class &&
+ slave->slave_id == master->slave_id) {
+ list_move_tail(&slave->open_list, &master->slave_list_head);
+ spin_lock_irq(&slave_active_lock);
+ slave->master = master;
+ slave->timer = master->timer;
+ if (slave->flags & SNDRV_TIMER_IFLG_RUNNING)
+ list_add_tail(&slave->active_list,
+ &master->slave_active_head);
+ spin_unlock_irq(&slave_active_lock);
+ }
+ }
+}
+
+/*
+ * open a timer instance
+ * when opening a master, the slave id must be here given.
+ */
+int snd_timer_open(struct snd_timer_instance **ti,
+ char *owner, struct snd_timer_id *tid,
+ unsigned int slave_id)
+{
+ struct snd_timer *timer;
+ struct snd_timer_instance *timeri = NULL;
+
+ if (tid->dev_class == SNDRV_TIMER_CLASS_SLAVE) {
+ /* open a slave instance */
+ if (tid->dev_sclass <= SNDRV_TIMER_SCLASS_NONE ||
+ tid->dev_sclass > SNDRV_TIMER_SCLASS_OSS_SEQUENCER) {
+ snd_printd("invalid slave class %i\n", tid->dev_sclass);
+ return -EINVAL;
+ }
+ mutex_lock(&register_mutex);
+ timeri = snd_timer_instance_new(owner, NULL);
+ if (!timeri) {
+ mutex_unlock(&register_mutex);
+ return -ENOMEM;
+ }
+ timeri->slave_class = tid->dev_sclass;
+ timeri->slave_id = tid->device;
+ timeri->flags |= SNDRV_TIMER_IFLG_SLAVE;
+ list_add_tail(&timeri->open_list, &snd_timer_slave_list);
+ snd_timer_check_slave(timeri);
+ mutex_unlock(&register_mutex);
+ *ti = timeri;
+ return 0;
+ }
+
+ /* open a master instance */
+ mutex_lock(&register_mutex);
+ timer = snd_timer_find(tid);
+#ifdef CONFIG_MODULES
+ if (!timer) {
+ mutex_unlock(&register_mutex);
+ snd_timer_request(tid);
+ mutex_lock(&register_mutex);
+ timer = snd_timer_find(tid);
+ }
+#endif
+ if (!timer) {
+ mutex_unlock(&register_mutex);
+ return -ENODEV;
+ }
+ if (!list_empty(&timer->open_list_head)) {
+ timeri = list_entry(timer->open_list_head.next,
+ struct snd_timer_instance, open_list);
+ if (timeri->flags & SNDRV_TIMER_IFLG_EXCLUSIVE) {
+ mutex_unlock(&register_mutex);
+ return -EBUSY;
+ }
+ }
+ timeri = snd_timer_instance_new(owner, timer);
+ if (!timeri) {
+ mutex_unlock(&register_mutex);
+ return -ENOMEM;
+ }
+ timeri->slave_class = tid->dev_sclass;
+ timeri->slave_id = slave_id;
+ if (list_empty(&timer->open_list_head) && timer->hw.open)
+ timer->hw.open(timer);
+ list_add_tail(&timeri->open_list, &timer->open_list_head);
+ snd_timer_check_master(timeri);
+ mutex_unlock(&register_mutex);
+ *ti = timeri;
+ return 0;
+}
+
+static int _snd_timer_stop(struct snd_timer_instance *timeri,
+ int keep_flag, int event);
+
+/*
+ * close a timer instance
+ */
+int snd_timer_close(struct snd_timer_instance *timeri)
+{
+ struct snd_timer *timer = NULL;
+ struct snd_timer_instance *slave, *tmp;
+
+ if (snd_BUG_ON(!timeri))
+ return -ENXIO;
+
+ /* force to stop the timer */
+ snd_timer_stop(timeri);
+
+ if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) {
+ /* wait, until the active callback is finished */
+ spin_lock_irq(&slave_active_lock);
+ while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
+ spin_unlock_irq(&slave_active_lock);
+ udelay(10);
+ spin_lock_irq(&slave_active_lock);
+ }
+ spin_unlock_irq(&slave_active_lock);
+ mutex_lock(&register_mutex);
+ list_del(&timeri->open_list);
+ mutex_unlock(&register_mutex);
+ } else {
+ timer = timeri->timer;
+ /* wait, until the active callback is finished */
+ spin_lock_irq(&timer->lock);
+ while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) {
+ spin_unlock_irq(&timer->lock);
+ udelay(10);
+ spin_lock_irq(&timer->lock);
+ }
+ spin_unlock_irq(&timer->lock);
+ mutex_lock(&register_mutex);
+ list_del(&timeri->open_list);
+ if (timer && list_empty(&timer->open_list_head) &&
+ timer->hw.close)
+ timer->hw.close(timer);
+ /* remove slave links */
+ list_for_each_entry_safe(slave, tmp, &timeri->slave_list_head,
+ open_list) {
+ spin_lock_irq(&slave_active_lock);
+ _snd_timer_stop(slave, 1, SNDRV_TIMER_EVENT_RESOLUTION);
+ list_move_tail(&slave->open_list, &snd_timer_slave_list);
+ slave->master = NULL;
+ slave->timer = NULL;
+ spin_unlock_irq(&slave_active_lock);
+ }
+ mutex_unlock(&register_mutex);
+ }
+ if (timeri->private_free)
+ timeri->private_free(timeri);
+ kfree(timeri->owner);
+ kfree(timeri);
+ if (timer)
+ module_put(timer->module);
+ return 0;
+}
+
+unsigned long snd_timer_resolution(struct snd_timer_instance *timeri)
+{
+ struct snd_timer * timer;
+
+ if (timeri == NULL)
+ return 0;
+ if ((timer = timeri->timer) != NULL) {
+ if (timer->hw.c_resolution)
+ return timer->hw.c_resolution(timer);
+ return timer->hw.resolution;
+ }
+ return 0;
+}
+
+static void snd_timer_notify1(struct snd_timer_instance *ti, int event)
+{
+ struct snd_timer *timer;
+ unsigned long flags;
+ unsigned long resolution = 0;
+ struct snd_timer_instance *ts;
+ struct timespec tstamp;
+
+ if (timer_tstamp_monotonic)
+ do_posix_clock_monotonic_gettime(&tstamp);
+ else
+ getnstimeofday(&tstamp);
+ if (snd_BUG_ON(event < SNDRV_TIMER_EVENT_START ||
+ event > SNDRV_TIMER_EVENT_PAUSE))
+ return;
+ if (event == SNDRV_TIMER_EVENT_START ||
+ event == SNDRV_TIMER_EVENT_CONTINUE)
+ resolution = snd_timer_resolution(ti);
+ if (ti->ccallback)
+ ti->ccallback(ti, SNDRV_TIMER_EVENT_START, &tstamp, resolution);
+ if (ti->flags & SNDRV_TIMER_IFLG_SLAVE)
+ return;
+ timer = ti->timer;
+ if (timer == NULL)
+ return;
+ if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE)
+ return;
+ spin_lock_irqsave(&timer->lock, flags);
+ list_for_each_entry(ts, &ti->slave_active_head, active_list)
+ if (ts->ccallback)
+ ts->ccallback(ti, event + 100, &tstamp, resolution);
+ spin_unlock_irqrestore(&timer->lock, flags);
+}
+
+static int snd_timer_start1(struct snd_timer *timer, struct snd_timer_instance *timeri,
+ unsigned long sticks)
+{
+ list_del(&timeri->active_list);
+ list_add_tail(&timeri->active_list, &timer->active_list_head);
+ if (timer->running) {
+ if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE)
+ goto __start_now;
+ timer->flags |= SNDRV_TIMER_FLG_RESCHED;
+ timeri->flags |= SNDRV_TIMER_IFLG_START;
+ return 1; /* delayed start */
+ } else {
+ timer->sticks = sticks;
+ timer->hw.start(timer);
+ __start_now:
+ timer->running++;
+ timeri->flags |= SNDRV_TIMER_IFLG_RUNNING;
+ return 0;
+ }
+}
+
+static int snd_timer_start_slave(struct snd_timer_instance *timeri)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&slave_active_lock, flags);
+ timeri->flags |= SNDRV_TIMER_IFLG_RUNNING;
+ if (timeri->master)
+ list_add_tail(&timeri->active_list,
+ &timeri->master->slave_active_head);
+ spin_unlock_irqrestore(&slave_active_lock, flags);
+ return 1; /* delayed start */
+}
+
+/*
+ * start the timer instance
+ */
+int snd_timer_start(struct snd_timer_instance *timeri, unsigned int ticks)
+{
+ struct snd_timer *timer;
+ int result = -EINVAL;
+ unsigned long flags;
+
+ if (timeri == NULL || ticks < 1)
+ return -EINVAL;
+ if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) {
+ result = snd_timer_start_slave(timeri);
+ snd_timer_notify1(timeri, SNDRV_TIMER_EVENT_START);
+ return result;
+ }
+ timer = timeri->timer;
+ if (timer == NULL)
+ return -EINVAL;
+ spin_lock_irqsave(&timer->lock, flags);
+ timeri->ticks = timeri->cticks = ticks;
+ timeri->pticks = 0;
+ result = snd_timer_start1(timer, timeri, ticks);
+ spin_unlock_irqrestore(&timer->lock, flags);
+ snd_timer_notify1(timeri, SNDRV_TIMER_EVENT_START);
+ return result;
+}
+
+static int _snd_timer_stop(struct snd_timer_instance * timeri,
+ int keep_flag, int event)
+{
+ struct snd_timer *timer;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!timeri))
+ return -ENXIO;
+
+ if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) {
+ if (!keep_flag) {
+ spin_lock_irqsave(&slave_active_lock, flags);
+ timeri->flags &= ~SNDRV_TIMER_IFLG_RUNNING;
+ spin_unlock_irqrestore(&slave_active_lock, flags);
+ }
+ goto __end;
+ }
+ timer = timeri->timer;
+ if (!timer)
+ return -EINVAL;
+ spin_lock_irqsave(&timer->lock, flags);
+ list_del_init(&timeri->ack_list);
+ list_del_init(&timeri->active_list);
+ if ((timeri->flags & SNDRV_TIMER_IFLG_RUNNING) &&
+ !(--timer->running)) {
+ timer->hw.stop(timer);
+ if (timer->flags & SNDRV_TIMER_FLG_RESCHED) {
+ timer->flags &= ~SNDRV_TIMER_FLG_RESCHED;
+ snd_timer_reschedule(timer, 0);
+ if (timer->flags & SNDRV_TIMER_FLG_CHANGE) {
+ timer->flags &= ~SNDRV_TIMER_FLG_CHANGE;
+ timer->hw.start(timer);
+ }
+ }
+ }
+ if (!keep_flag)
+ timeri->flags &=
+ ~(SNDRV_TIMER_IFLG_RUNNING | SNDRV_TIMER_IFLG_START);
+ spin_unlock_irqrestore(&timer->lock, flags);
+ __end:
+ if (event != SNDRV_TIMER_EVENT_RESOLUTION)
+ snd_timer_notify1(timeri, event);
+ return 0;
+}
+
+/*
+ * stop the timer instance.
+ *
+ * do not call this from the timer callback!
+ */
+int snd_timer_stop(struct snd_timer_instance *timeri)
+{
+ struct snd_timer *timer;
+ unsigned long flags;
+ int err;
+
+ err = _snd_timer_stop(timeri, 0, SNDRV_TIMER_EVENT_STOP);
+ if (err < 0)
+ return err;
+ timer = timeri->timer;
+ spin_lock_irqsave(&timer->lock, flags);
+ timeri->cticks = timeri->ticks;
+ timeri->pticks = 0;
+ spin_unlock_irqrestore(&timer->lock, flags);
+ return 0;
+}
+
+/*
+ * start again.. the tick is kept.
+ */
+int snd_timer_continue(struct snd_timer_instance *timeri)
+{
+ struct snd_timer *timer;
+ int result = -EINVAL;
+ unsigned long flags;
+
+ if (timeri == NULL)
+ return result;
+ if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE)
+ return snd_timer_start_slave(timeri);
+ timer = timeri->timer;
+ if (! timer)
+ return -EINVAL;
+ spin_lock_irqsave(&timer->lock, flags);
+ if (!timeri->cticks)
+ timeri->cticks = 1;
+ timeri->pticks = 0;
+ result = snd_timer_start1(timer, timeri, timer->sticks);
+ spin_unlock_irqrestore(&timer->lock, flags);
+ snd_timer_notify1(timeri, SNDRV_TIMER_EVENT_CONTINUE);
+ return result;
+}
+
+/*
+ * pause.. remember the ticks left
+ */
+int snd_timer_pause(struct snd_timer_instance * timeri)
+{
+ return _snd_timer_stop(timeri, 0, SNDRV_TIMER_EVENT_PAUSE);
+}
+
+/*
+ * reschedule the timer
+ *
+ * start pending instances and check the scheduling ticks.
+ * when the scheduling ticks is changed set CHANGE flag to reprogram the timer.
+ */
+static void snd_timer_reschedule(struct snd_timer * timer, unsigned long ticks_left)
+{
+ struct snd_timer_instance *ti;
+ unsigned long ticks = ~0UL;
+
+ list_for_each_entry(ti, &timer->active_list_head, active_list) {
+ if (ti->flags & SNDRV_TIMER_IFLG_START) {
+ ti->flags &= ~SNDRV_TIMER_IFLG_START;
+ ti->flags |= SNDRV_TIMER_IFLG_RUNNING;
+ timer->running++;
+ }
+ if (ti->flags & SNDRV_TIMER_IFLG_RUNNING) {
+ if (ticks > ti->cticks)
+ ticks = ti->cticks;
+ }
+ }
+ if (ticks == ~0UL) {
+ timer->flags &= ~SNDRV_TIMER_FLG_RESCHED;
+ return;
+ }
+ if (ticks > timer->hw.ticks)
+ ticks = timer->hw.ticks;
+ if (ticks_left != ticks)
+ timer->flags |= SNDRV_TIMER_FLG_CHANGE;
+ timer->sticks = ticks;
+}
+
+/*
+ * timer tasklet
+ *
+ */
+static void snd_timer_tasklet(unsigned long arg)
+{
+ struct snd_timer *timer = (struct snd_timer *) arg;
+ struct snd_timer_instance *ti;
+ struct list_head *p;
+ unsigned long resolution, ticks;
+ unsigned long flags;
+
+ spin_lock_irqsave(&timer->lock, flags);
+ /* now process all callbacks */
+ while (!list_empty(&timer->sack_list_head)) {
+ p = timer->sack_list_head.next; /* get first item */
+ ti = list_entry(p, struct snd_timer_instance, ack_list);
+
+ /* remove from ack_list and make empty */
+ list_del_init(p);
+
+ ticks = ti->pticks;
+ ti->pticks = 0;
+ resolution = ti->resolution;
+
+ ti->flags |= SNDRV_TIMER_IFLG_CALLBACK;
+ spin_unlock(&timer->lock);
+ if (ti->callback)
+ ti->callback(ti, resolution, ticks);
+ spin_lock(&timer->lock);
+ ti->flags &= ~SNDRV_TIMER_IFLG_CALLBACK;
+ }
+ spin_unlock_irqrestore(&timer->lock, flags);
+}
+
+/*
+ * timer interrupt
+ *
+ * ticks_left is usually equal to timer->sticks.
+ *
+ */
+void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
+{
+ struct snd_timer_instance *ti, *ts, *tmp;
+ unsigned long resolution, ticks;
+ struct list_head *p, *ack_list_head;
+ unsigned long flags;
+ int use_tasklet = 0;
+
+ if (timer == NULL)
+ return;
+
+ spin_lock_irqsave(&timer->lock, flags);
+
+ /* remember the current resolution */
+ if (timer->hw.c_resolution)
+ resolution = timer->hw.c_resolution(timer);
+ else
+ resolution = timer->hw.resolution;
+
+ /* loop for all active instances
+ * Here we cannot use list_for_each_entry because the active_list of a
+ * processed instance is relinked to done_list_head before the callback
+ * is called.
+ */
+ list_for_each_entry_safe(ti, tmp, &timer->active_list_head,
+ active_list) {
+ if (!(ti->flags & SNDRV_TIMER_IFLG_RUNNING))
+ continue;
+ ti->pticks += ticks_left;
+ ti->resolution = resolution;
+ if (ti->cticks < ticks_left)
+ ti->cticks = 0;
+ else
+ ti->cticks -= ticks_left;
+ if (ti->cticks) /* not expired */
+ continue;
+ if (ti->flags & SNDRV_TIMER_IFLG_AUTO) {
+ ti->cticks = ti->ticks;
+ } else {
+ ti->flags &= ~SNDRV_TIMER_IFLG_RUNNING;
+ if (--timer->running)
+ list_del(&ti->active_list);
+ }
+ if ((timer->hw.flags & SNDRV_TIMER_HW_TASKLET) ||
+ (ti->flags & SNDRV_TIMER_IFLG_FAST))
+ ack_list_head = &timer->ack_list_head;
+ else
+ ack_list_head = &timer->sack_list_head;
+ if (list_empty(&ti->ack_list))
+ list_add_tail(&ti->ack_list, ack_list_head);
+ list_for_each_entry(ts, &ti->slave_active_head, active_list) {
+ ts->pticks = ti->pticks;
+ ts->resolution = resolution;
+ if (list_empty(&ts->ack_list))
+ list_add_tail(&ts->ack_list, ack_list_head);
+ }
+ }
+ if (timer->flags & SNDRV_TIMER_FLG_RESCHED)
+ snd_timer_reschedule(timer, timer->sticks);
+ if (timer->running) {
+ if (timer->hw.flags & SNDRV_TIMER_HW_STOP) {
+ timer->hw.stop(timer);
+ timer->flags |= SNDRV_TIMER_FLG_CHANGE;
+ }
+ if (!(timer->hw.flags & SNDRV_TIMER_HW_AUTO) ||
+ (timer->flags & SNDRV_TIMER_FLG_CHANGE)) {
+ /* restart timer */
+ timer->flags &= ~SNDRV_TIMER_FLG_CHANGE;
+ timer->hw.start(timer);
+ }
+ } else {
+ timer->hw.stop(timer);
+ }
+
+ /* now process all fast callbacks */
+ while (!list_empty(&timer->ack_list_head)) {
+ p = timer->ack_list_head.next; /* get first item */
+ ti = list_entry(p, struct snd_timer_instance, ack_list);
+
+ /* remove from ack_list and make empty */
+ list_del_init(p);
+
+ ticks = ti->pticks;
+ ti->pticks = 0;
+
+ ti->flags |= SNDRV_TIMER_IFLG_CALLBACK;
+ spin_unlock(&timer->lock);
+ if (ti->callback)
+ ti->callback(ti, resolution, ticks);
+ spin_lock(&timer->lock);
+ ti->flags &= ~SNDRV_TIMER_IFLG_CALLBACK;
+ }
+
+ /* do we have any slow callbacks? */
+ use_tasklet = !list_empty(&timer->sack_list_head);
+ spin_unlock_irqrestore(&timer->lock, flags);
+
+ if (use_tasklet)
+ tasklet_hi_schedule(&timer->task_queue);
+}
+
+/*
+
+ */
+
+int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid,
+ struct snd_timer **rtimer)
+{
+ struct snd_timer *timer;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_timer_dev_free,
+ .dev_register = snd_timer_dev_register,
+ .dev_disconnect = snd_timer_dev_disconnect,
+ };
+
+ if (snd_BUG_ON(!tid))
+ return -EINVAL;
+ if (rtimer)
+ *rtimer = NULL;
+ timer = kzalloc(sizeof(*timer), GFP_KERNEL);
+ if (timer == NULL) {
+ snd_printk(KERN_ERR "timer: cannot allocate\n");
+ return -ENOMEM;
+ }
+ timer->tmr_class = tid->dev_class;
+ timer->card = card;
+ timer->tmr_device = tid->device;
+ timer->tmr_subdevice = tid->subdevice;
+ if (id)
+ strlcpy(timer->id, id, sizeof(timer->id));
+ INIT_LIST_HEAD(&timer->device_list);
+ INIT_LIST_HEAD(&timer->open_list_head);
+ INIT_LIST_HEAD(&timer->active_list_head);
+ INIT_LIST_HEAD(&timer->ack_list_head);
+ INIT_LIST_HEAD(&timer->sack_list_head);
+ spin_lock_init(&timer->lock);
+ tasklet_init(&timer->task_queue, snd_timer_tasklet,
+ (unsigned long)timer);
+ if (card != NULL) {
+ timer->module = card->module;
+ err = snd_device_new(card, SNDRV_DEV_TIMER, timer, &ops);
+ if (err < 0) {
+ snd_timer_free(timer);
+ return err;
+ }
+ }
+ if (rtimer)
+ *rtimer = timer;
+ return 0;
+}
+
+static int snd_timer_free(struct snd_timer *timer)
+{
+ if (!timer)
+ return 0;
+
+ mutex_lock(&register_mutex);
+ if (! list_empty(&timer->open_list_head)) {
+ struct list_head *p, *n;
+ struct snd_timer_instance *ti;
+ snd_printk(KERN_WARNING "timer %p is busy?\n", timer);
+ list_for_each_safe(p, n, &timer->open_list_head) {
+ list_del_init(p);
+ ti = list_entry(p, struct snd_timer_instance, open_list);
+ ti->timer = NULL;
+ }
+ }
+ list_del(&timer->device_list);
+ mutex_unlock(&register_mutex);
+
+ if (timer->private_free)
+ timer->private_free(timer);
+ kfree(timer);
+ return 0;
+}
+
+static int snd_timer_dev_free(struct snd_device *device)
+{
+ struct snd_timer *timer = device->device_data;
+ return snd_timer_free(timer);
+}
+
+static int snd_timer_dev_register(struct snd_device *dev)
+{
+ struct snd_timer *timer = dev->device_data;
+ struct snd_timer *timer1;
+
+ if (snd_BUG_ON(!timer || !timer->hw.start || !timer->hw.stop))
+ return -ENXIO;
+ if (!(timer->hw.flags & SNDRV_TIMER_HW_SLAVE) &&
+ !timer->hw.resolution && timer->hw.c_resolution == NULL)
+ return -EINVAL;
+
+ mutex_lock(&register_mutex);
+ list_for_each_entry(timer1, &snd_timer_list, device_list) {
+ if (timer1->tmr_class > timer->tmr_class)
+ break;
+ if (timer1->tmr_class < timer->tmr_class)
+ continue;
+ if (timer1->card && timer->card) {
+ if (timer1->card->number > timer->card->number)
+ break;
+ if (timer1->card->number < timer->card->number)
+ continue;
+ }
+ if (timer1->tmr_device > timer->tmr_device)
+ break;
+ if (timer1->tmr_device < timer->tmr_device)
+ continue;
+ if (timer1->tmr_subdevice > timer->tmr_subdevice)
+ break;
+ if (timer1->tmr_subdevice < timer->tmr_subdevice)
+ continue;
+ /* conflicts.. */
+ mutex_unlock(&register_mutex);
+ return -EBUSY;
+ }
+ list_add_tail(&timer->device_list, &timer1->device_list);
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+static int snd_timer_dev_disconnect(struct snd_device *device)
+{
+ struct snd_timer *timer = device->device_data;
+ mutex_lock(&register_mutex);
+ list_del_init(&timer->device_list);
+ mutex_unlock(&register_mutex);
+ return 0;
+}
+
+void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstamp)
+{
+ unsigned long flags;
+ unsigned long resolution = 0;
+ struct snd_timer_instance *ti, *ts;
+
+ if (! (timer->hw.flags & SNDRV_TIMER_HW_SLAVE))
+ return;
+ if (snd_BUG_ON(event < SNDRV_TIMER_EVENT_MSTART ||
+ event > SNDRV_TIMER_EVENT_MRESUME))
+ return;
+ spin_lock_irqsave(&timer->lock, flags);
+ if (event == SNDRV_TIMER_EVENT_MSTART ||
+ event == SNDRV_TIMER_EVENT_MCONTINUE ||
+ event == SNDRV_TIMER_EVENT_MRESUME) {
+ if (timer->hw.c_resolution)
+ resolution = timer->hw.c_resolution(timer);
+ else
+ resolution = timer->hw.resolution;
+ }
+ list_for_each_entry(ti, &timer->active_list_head, active_list) {
+ if (ti->ccallback)
+ ti->ccallback(ti, event, tstamp, resolution);
+ list_for_each_entry(ts, &ti->slave_active_head, active_list)
+ if (ts->ccallback)
+ ts->ccallback(ts, event, tstamp, resolution);
+ }
+ spin_unlock_irqrestore(&timer->lock, flags);
+}
+
+/*
+ * exported functions for global timers
+ */
+int snd_timer_global_new(char *id, int device, struct snd_timer **rtimer)
+{
+ struct snd_timer_id tid;
+
+ tid.dev_class = SNDRV_TIMER_CLASS_GLOBAL;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = -1;
+ tid.device = device;
+ tid.subdevice = 0;
+ return snd_timer_new(NULL, id, &tid, rtimer);
+}
+
+int snd_timer_global_free(struct snd_timer *timer)
+{
+ return snd_timer_free(timer);
+}
+
+int snd_timer_global_register(struct snd_timer *timer)
+{
+ struct snd_device dev;
+
+ memset(&dev, 0, sizeof(dev));
+ dev.device_data = timer;
+ return snd_timer_dev_register(&dev);
+}
+
+/*
+ * System timer
+ */
+
+struct snd_timer_system_private {
+ struct timer_list tlist;
+ unsigned long last_expires;
+ unsigned long last_jiffies;
+ unsigned long correction;
+};
+
+static void snd_timer_s_function(unsigned long data)
+{
+ struct snd_timer *timer = (struct snd_timer *)data;
+ struct snd_timer_system_private *priv = timer->private_data;
+ unsigned long jiff = jiffies;
+ if (time_after(jiff, priv->last_expires))
+ priv->correction += (long)jiff - (long)priv->last_expires;
+ snd_timer_interrupt(timer, (long)jiff - (long)priv->last_jiffies);
+}
+
+static int snd_timer_s_start(struct snd_timer * timer)
+{
+ struct snd_timer_system_private *priv;
+ unsigned long njiff;
+
+ priv = (struct snd_timer_system_private *) timer->private_data;
+ njiff = (priv->last_jiffies = jiffies);
+ if (priv->correction > timer->sticks - 1) {
+ priv->correction -= timer->sticks - 1;
+ njiff++;
+ } else {
+ njiff += timer->sticks - priv->correction;
+ priv->correction = 0;
+ }
+ priv->last_expires = priv->tlist.expires = njiff;
+ add_timer(&priv->tlist);
+ return 0;
+}
+
+static int snd_timer_s_stop(struct snd_timer * timer)
+{
+ struct snd_timer_system_private *priv;
+ unsigned long jiff;
+
+ priv = (struct snd_timer_system_private *) timer->private_data;
+ del_timer(&priv->tlist);
+ jiff = jiffies;
+ if (time_before(jiff, priv->last_expires))
+ timer->sticks = priv->last_expires - jiff;
+ else
+ timer->sticks = 1;
+ priv->correction = 0;
+ return 0;
+}
+
+static struct snd_timer_hardware snd_timer_system =
+{
+ .flags = SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_TASKLET,
+ .resolution = 1000000000L / HZ,
+ .ticks = 10000000L,
+ .start = snd_timer_s_start,
+ .stop = snd_timer_s_stop
+};
+
+static void snd_timer_free_system(struct snd_timer *timer)
+{
+ kfree(timer->private_data);
+}
+
+static int snd_timer_register_system(void)
+{
+ struct snd_timer *timer;
+ struct snd_timer_system_private *priv;
+ int err;
+
+ err = snd_timer_global_new("system", SNDRV_TIMER_GLOBAL_SYSTEM, &timer);
+ if (err < 0)
+ return err;
+ strcpy(timer->name, "system timer");
+ timer->hw = snd_timer_system;
+ priv = kzalloc(sizeof(*priv), GFP_KERNEL);
+ if (priv == NULL) {
+ snd_timer_free(timer);
+ return -ENOMEM;
+ }
+ init_timer(&priv->tlist);
+ priv->tlist.function = snd_timer_s_function;
+ priv->tlist.data = (unsigned long) timer;
+ timer->private_data = priv;
+ timer->private_free = snd_timer_free_system;
+ return snd_timer_global_register(timer);
+}
+
+#ifdef CONFIG_PROC_FS
+/*
+ * Info interface
+ */
+
+static void snd_timer_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_timer *timer;
+ struct snd_timer_instance *ti;
+
+ mutex_lock(&register_mutex);
+ list_for_each_entry(timer, &snd_timer_list, device_list) {
+ switch (timer->tmr_class) {
+ case SNDRV_TIMER_CLASS_GLOBAL:
+ snd_iprintf(buffer, "G%i: ", timer->tmr_device);
+ break;
+ case SNDRV_TIMER_CLASS_CARD:
+ snd_iprintf(buffer, "C%i-%i: ",
+ timer->card->number, timer->tmr_device);
+ break;
+ case SNDRV_TIMER_CLASS_PCM:
+ snd_iprintf(buffer, "P%i-%i-%i: ", timer->card->number,
+ timer->tmr_device, timer->tmr_subdevice);
+ break;
+ default:
+ snd_iprintf(buffer, "?%i-%i-%i-%i: ", timer->tmr_class,
+ timer->card ? timer->card->number : -1,
+ timer->tmr_device, timer->tmr_subdevice);
+ }
+ snd_iprintf(buffer, "%s :", timer->name);
+ if (timer->hw.resolution)
+ snd_iprintf(buffer, " %lu.%03luus (%lu ticks)",
+ timer->hw.resolution / 1000,
+ timer->hw.resolution % 1000,
+ timer->hw.ticks);
+ if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE)
+ snd_iprintf(buffer, " SLAVE");
+ snd_iprintf(buffer, "\n");
+ list_for_each_entry(ti, &timer->open_list_head, open_list)
+ snd_iprintf(buffer, " Client %s : %s\n",
+ ti->owner ? ti->owner : "unknown",
+ ti->flags & (SNDRV_TIMER_IFLG_START |
+ SNDRV_TIMER_IFLG_RUNNING)
+ ? "running" : "stopped");
+ }
+ mutex_unlock(&register_mutex);
+}
+
+static struct snd_info_entry *snd_timer_proc_entry;
+
+static void __init snd_timer_proc_init(void)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_module_entry(THIS_MODULE, "timers", NULL);
+ if (entry != NULL) {
+ entry->c.text.read = snd_timer_proc_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ snd_timer_proc_entry = entry;
+}
+
+static void __exit snd_timer_proc_done(void)
+{
+ snd_info_free_entry(snd_timer_proc_entry);
+}
+#else /* !CONFIG_PROC_FS */
+#define snd_timer_proc_init()
+#define snd_timer_proc_done()
+#endif
+
+/*
+ * USER SPACE interface
+ */
+
+static void snd_timer_user_interrupt(struct snd_timer_instance *timeri,
+ unsigned long resolution,
+ unsigned long ticks)
+{
+ struct snd_timer_user *tu = timeri->callback_data;
+ struct snd_timer_read *r;
+ int prev;
+
+ spin_lock(&tu->qlock);
+ if (tu->qused > 0) {
+ prev = tu->qtail == 0 ? tu->queue_size - 1 : tu->qtail - 1;
+ r = &tu->queue[prev];
+ if (r->resolution == resolution) {
+ r->ticks += ticks;
+ goto __wake;
+ }
+ }
+ if (tu->qused >= tu->queue_size) {
+ tu->overrun++;
+ } else {
+ r = &tu->queue[tu->qtail++];
+ tu->qtail %= tu->queue_size;
+ r->resolution = resolution;
+ r->ticks = ticks;
+ tu->qused++;
+ }
+ __wake:
+ spin_unlock(&tu->qlock);
+ kill_fasync(&tu->fasync, SIGIO, POLL_IN);
+ wake_up(&tu->qchange_sleep);
+}
+
+static void snd_timer_user_append_to_tqueue(struct snd_timer_user *tu,
+ struct snd_timer_tread *tread)
+{
+ if (tu->qused >= tu->queue_size) {
+ tu->overrun++;
+ } else {
+ memcpy(&tu->tqueue[tu->qtail++], tread, sizeof(*tread));
+ tu->qtail %= tu->queue_size;
+ tu->qused++;
+ }
+}
+
+static void snd_timer_user_ccallback(struct snd_timer_instance *timeri,
+ int event,
+ struct timespec *tstamp,
+ unsigned long resolution)
+{
+ struct snd_timer_user *tu = timeri->callback_data;
+ struct snd_timer_tread r1;
+
+ if (event >= SNDRV_TIMER_EVENT_START &&
+ event <= SNDRV_TIMER_EVENT_PAUSE)
+ tu->tstamp = *tstamp;
+ if ((tu->filter & (1 << event)) == 0 || !tu->tread)
+ return;
+ r1.event = event;
+ r1.tstamp = *tstamp;
+ r1.val = resolution;
+ spin_lock(&tu->qlock);
+ snd_timer_user_append_to_tqueue(tu, &r1);
+ spin_unlock(&tu->qlock);
+ kill_fasync(&tu->fasync, SIGIO, POLL_IN);
+ wake_up(&tu->qchange_sleep);
+}
+
+static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri,
+ unsigned long resolution,
+ unsigned long ticks)
+{
+ struct snd_timer_user *tu = timeri->callback_data;
+ struct snd_timer_tread *r, r1;
+ struct timespec tstamp;
+ int prev, append = 0;
+
+ memset(&tstamp, 0, sizeof(tstamp));
+ spin_lock(&tu->qlock);
+ if ((tu->filter & ((1 << SNDRV_TIMER_EVENT_RESOLUTION) |
+ (1 << SNDRV_TIMER_EVENT_TICK))) == 0) {
+ spin_unlock(&tu->qlock);
+ return;
+ }
+ if (tu->last_resolution != resolution || ticks > 0) {
+ if (timer_tstamp_monotonic)
+ do_posix_clock_monotonic_gettime(&tstamp);
+ else
+ getnstimeofday(&tstamp);
+ }
+ if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) &&
+ tu->last_resolution != resolution) {
+ r1.event = SNDRV_TIMER_EVENT_RESOLUTION;
+ r1.tstamp = tstamp;
+ r1.val = resolution;
+ snd_timer_user_append_to_tqueue(tu, &r1);
+ tu->last_resolution = resolution;
+ append++;
+ }
+ if ((tu->filter & (1 << SNDRV_TIMER_EVENT_TICK)) == 0)
+ goto __wake;
+ if (ticks == 0)
+ goto __wake;
+ if (tu->qused > 0) {
+ prev = tu->qtail == 0 ? tu->queue_size - 1 : tu->qtail - 1;
+ r = &tu->tqueue[prev];
+ if (r->event == SNDRV_TIMER_EVENT_TICK) {
+ r->tstamp = tstamp;
+ r->val += ticks;
+ append++;
+ goto __wake;
+ }
+ }
+ r1.event = SNDRV_TIMER_EVENT_TICK;
+ r1.tstamp = tstamp;
+ r1.val = ticks;
+ snd_timer_user_append_to_tqueue(tu, &r1);
+ append++;
+ __wake:
+ spin_unlock(&tu->qlock);
+ if (append == 0)
+ return;
+ kill_fasync(&tu->fasync, SIGIO, POLL_IN);
+ wake_up(&tu->qchange_sleep);
+}
+
+static int snd_timer_user_open(struct inode *inode, struct file *file)
+{
+ struct snd_timer_user *tu;
+
+ tu = kzalloc(sizeof(*tu), GFP_KERNEL);
+ if (tu == NULL)
+ return -ENOMEM;
+ spin_lock_init(&tu->qlock);
+ init_waitqueue_head(&tu->qchange_sleep);
+ mutex_init(&tu->tread_sem);
+ tu->ticks = 1;
+ tu->queue_size = 128;
+ tu->queue = kmalloc(tu->queue_size * sizeof(struct snd_timer_read),
+ GFP_KERNEL);
+ if (tu->queue == NULL) {
+ kfree(tu);
+ return -ENOMEM;
+ }
+ file->private_data = tu;
+ return 0;
+}
+
+static int snd_timer_user_release(struct inode *inode, struct file *file)
+{
+ struct snd_timer_user *tu;
+
+ if (file->private_data) {
+ tu = file->private_data;
+ file->private_data = NULL;
+ if (tu->timeri)
+ snd_timer_close(tu->timeri);
+ kfree(tu->queue);
+ kfree(tu->tqueue);
+ kfree(tu);
+ }
+ return 0;
+}
+
+static void snd_timer_user_zero_id(struct snd_timer_id *id)
+{
+ id->dev_class = SNDRV_TIMER_CLASS_NONE;
+ id->dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ id->card = -1;
+ id->device = -1;
+ id->subdevice = -1;
+}
+
+static void snd_timer_user_copy_id(struct snd_timer_id *id, struct snd_timer *timer)
+{
+ id->dev_class = timer->tmr_class;
+ id->dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ id->card = timer->card ? timer->card->number : -1;
+ id->device = timer->tmr_device;
+ id->subdevice = timer->tmr_subdevice;
+}
+
+static int snd_timer_user_next_device(struct snd_timer_id __user *_tid)
+{
+ struct snd_timer_id id;
+ struct snd_timer *timer;
+ struct list_head *p;
+
+ if (copy_from_user(&id, _tid, sizeof(id)))
+ return -EFAULT;
+ mutex_lock(&register_mutex);
+ if (id.dev_class < 0) { /* first item */
+ if (list_empty(&snd_timer_list))
+ snd_timer_user_zero_id(&id);
+ else {
+ timer = list_entry(snd_timer_list.next,
+ struct snd_timer, device_list);
+ snd_timer_user_copy_id(&id, timer);
+ }
+ } else {
+ switch (id.dev_class) {
+ case SNDRV_TIMER_CLASS_GLOBAL:
+ id.device = id.device < 0 ? 0 : id.device + 1;
+ list_for_each(p, &snd_timer_list) {
+ timer = list_entry(p, struct snd_timer, device_list);
+ if (timer->tmr_class > SNDRV_TIMER_CLASS_GLOBAL) {
+ snd_timer_user_copy_id(&id, timer);
+ break;
+ }
+ if (timer->tmr_device >= id.device) {
+ snd_timer_user_copy_id(&id, timer);
+ break;
+ }
+ }
+ if (p == &snd_timer_list)
+ snd_timer_user_zero_id(&id);
+ break;
+ case SNDRV_TIMER_CLASS_CARD:
+ case SNDRV_TIMER_CLASS_PCM:
+ if (id.card < 0) {
+ id.card = 0;
+ } else {
+ if (id.card < 0) {
+ id.card = 0;
+ } else {
+ if (id.device < 0) {
+ id.device = 0;
+ } else {
+ if (id.subdevice < 0) {
+ id.subdevice = 0;
+ } else {
+ id.subdevice++;
+ }
+ }
+ }
+ }
+ list_for_each(p, &snd_timer_list) {
+ timer = list_entry(p, struct snd_timer, device_list);
+ if (timer->tmr_class > id.dev_class) {
+ snd_timer_user_copy_id(&id, timer);
+ break;
+ }
+ if (timer->tmr_class < id.dev_class)
+ continue;
+ if (timer->card->number > id.card) {
+ snd_timer_user_copy_id(&id, timer);
+ break;
+ }
+ if (timer->card->number < id.card)
+ continue;
+ if (timer->tmr_device > id.device) {
+ snd_timer_user_copy_id(&id, timer);
+ break;
+ }
+ if (timer->tmr_device < id.device)
+ continue;
+ if (timer->tmr_subdevice > id.subdevice) {
+ snd_timer_user_copy_id(&id, timer);
+ break;
+ }
+ if (timer->tmr_subdevice < id.subdevice)
+ continue;
+ snd_timer_user_copy_id(&id, timer);
+ break;
+ }
+ if (p == &snd_timer_list)
+ snd_timer_user_zero_id(&id);
+ break;
+ default:
+ snd_timer_user_zero_id(&id);
+ }
+ }
+ mutex_unlock(&register_mutex);
+ if (copy_to_user(_tid, &id, sizeof(*_tid)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_timer_user_ginfo(struct file *file,
+ struct snd_timer_ginfo __user *_ginfo)
+{
+ struct snd_timer_ginfo *ginfo;
+ struct snd_timer_id tid;
+ struct snd_timer *t;
+ struct list_head *p;
+ int err = 0;
+
+ ginfo = kmalloc(sizeof(*ginfo), GFP_KERNEL);
+ if (! ginfo)
+ return -ENOMEM;
+ if (copy_from_user(ginfo, _ginfo, sizeof(*ginfo))) {
+ kfree(ginfo);
+ return -EFAULT;
+ }
+ tid = ginfo->tid;
+ memset(ginfo, 0, sizeof(*ginfo));
+ ginfo->tid = tid;
+ mutex_lock(&register_mutex);
+ t = snd_timer_find(&tid);
+ if (t != NULL) {
+ ginfo->card = t->card ? t->card->number : -1;
+ if (t->hw.flags & SNDRV_TIMER_HW_SLAVE)
+ ginfo->flags |= SNDRV_TIMER_FLG_SLAVE;
+ strlcpy(ginfo->id, t->id, sizeof(ginfo->id));
+ strlcpy(ginfo->name, t->name, sizeof(ginfo->name));
+ ginfo->resolution = t->hw.resolution;
+ if (t->hw.resolution_min > 0) {
+ ginfo->resolution_min = t->hw.resolution_min;
+ ginfo->resolution_max = t->hw.resolution_max;
+ }
+ list_for_each(p, &t->open_list_head) {
+ ginfo->clients++;
+ }
+ } else {
+ err = -ENODEV;
+ }
+ mutex_unlock(&register_mutex);
+ if (err >= 0 && copy_to_user(_ginfo, ginfo, sizeof(*ginfo)))
+ err = -EFAULT;
+ kfree(ginfo);
+ return err;
+}
+
+static int snd_timer_user_gparams(struct file *file,
+ struct snd_timer_gparams __user *_gparams)
+{
+ struct snd_timer_gparams gparams;
+ struct snd_timer *t;
+ int err;
+
+ if (copy_from_user(&gparams, _gparams, sizeof(gparams)))
+ return -EFAULT;
+ mutex_lock(&register_mutex);
+ t = snd_timer_find(&gparams.tid);
+ if (!t) {
+ err = -ENODEV;
+ goto _error;
+ }
+ if (!list_empty(&t->open_list_head)) {
+ err = -EBUSY;
+ goto _error;
+ }
+ if (!t->hw.set_period) {
+ err = -ENOSYS;
+ goto _error;
+ }
+ err = t->hw.set_period(t, gparams.period_num, gparams.period_den);
+_error:
+ mutex_unlock(&register_mutex);
+ return err;
+}
+
+static int snd_timer_user_gstatus(struct file *file,
+ struct snd_timer_gstatus __user *_gstatus)
+{
+ struct snd_timer_gstatus gstatus;
+ struct snd_timer_id tid;
+ struct snd_timer *t;
+ int err = 0;
+
+ if (copy_from_user(&gstatus, _gstatus, sizeof(gstatus)))
+ return -EFAULT;
+ tid = gstatus.tid;
+ memset(&gstatus, 0, sizeof(gstatus));
+ gstatus.tid = tid;
+ mutex_lock(&register_mutex);
+ t = snd_timer_find(&tid);
+ if (t != NULL) {
+ if (t->hw.c_resolution)
+ gstatus.resolution = t->hw.c_resolution(t);
+ else
+ gstatus.resolution = t->hw.resolution;
+ if (t->hw.precise_resolution) {
+ t->hw.precise_resolution(t, &gstatus.resolution_num,
+ &gstatus.resolution_den);
+ } else {
+ gstatus.resolution_num = gstatus.resolution;
+ gstatus.resolution_den = 1000000000uL;
+ }
+ } else {
+ err = -ENODEV;
+ }
+ mutex_unlock(&register_mutex);
+ if (err >= 0 && copy_to_user(_gstatus, &gstatus, sizeof(gstatus)))
+ err = -EFAULT;
+ return err;
+}
+
+static int snd_timer_user_tselect(struct file *file,
+ struct snd_timer_select __user *_tselect)
+{
+ struct snd_timer_user *tu;
+ struct snd_timer_select tselect;
+ char str[32];
+ int err = 0;
+
+ tu = file->private_data;
+ mutex_lock(&tu->tread_sem);
+ if (tu->timeri) {
+ snd_timer_close(tu->timeri);
+ tu->timeri = NULL;
+ }
+ if (copy_from_user(&tselect, _tselect, sizeof(tselect))) {
+ err = -EFAULT;
+ goto __err;
+ }
+ sprintf(str, "application %i", current->pid);
+ if (tselect.id.dev_class != SNDRV_TIMER_CLASS_SLAVE)
+ tselect.id.dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION;
+ err = snd_timer_open(&tu->timeri, str, &tselect.id, current->pid);
+ if (err < 0)
+ goto __err;
+
+ kfree(tu->queue);
+ tu->queue = NULL;
+ kfree(tu->tqueue);
+ tu->tqueue = NULL;
+ if (tu->tread) {
+ tu->tqueue = kmalloc(tu->queue_size * sizeof(struct snd_timer_tread),
+ GFP_KERNEL);
+ if (tu->tqueue == NULL)
+ err = -ENOMEM;
+ } else {
+ tu->queue = kmalloc(tu->queue_size * sizeof(struct snd_timer_read),
+ GFP_KERNEL);
+ if (tu->queue == NULL)
+ err = -ENOMEM;
+ }
+
+ if (err < 0) {
+ snd_timer_close(tu->timeri);
+ tu->timeri = NULL;
+ } else {
+ tu->timeri->flags |= SNDRV_TIMER_IFLG_FAST;
+ tu->timeri->callback = tu->tread
+ ? snd_timer_user_tinterrupt : snd_timer_user_interrupt;
+ tu->timeri->ccallback = snd_timer_user_ccallback;
+ tu->timeri->callback_data = (void *)tu;
+ }
+
+ __err:
+ mutex_unlock(&tu->tread_sem);
+ return err;
+}
+
+static int snd_timer_user_info(struct file *file,
+ struct snd_timer_info __user *_info)
+{
+ struct snd_timer_user *tu;
+ struct snd_timer_info *info;
+ struct snd_timer *t;
+ int err = 0;
+
+ tu = file->private_data;
+ if (!tu->timeri)
+ return -EBADFD;
+ t = tu->timeri->timer;
+ if (!t)
+ return -EBADFD;
+
+ info = kzalloc(sizeof(*info), GFP_KERNEL);
+ if (! info)
+ return -ENOMEM;
+ info->card = t->card ? t->card->number : -1;
+ if (t->hw.flags & SNDRV_TIMER_HW_SLAVE)
+ info->flags |= SNDRV_TIMER_FLG_SLAVE;
+ strlcpy(info->id, t->id, sizeof(info->id));
+ strlcpy(info->name, t->name, sizeof(info->name));
+ info->resolution = t->hw.resolution;
+ if (copy_to_user(_info, info, sizeof(*_info)))
+ err = -EFAULT;
+ kfree(info);
+ return err;
+}
+
+static int snd_timer_user_params(struct file *file,
+ struct snd_timer_params __user *_params)
+{
+ struct snd_timer_user *tu;
+ struct snd_timer_params params;
+ struct snd_timer *t;
+ struct snd_timer_read *tr;
+ struct snd_timer_tread *ttr;
+ int err;
+
+ tu = file->private_data;
+ if (!tu->timeri)
+ return -EBADFD;
+ t = tu->timeri->timer;
+ if (!t)
+ return -EBADFD;
+ if (copy_from_user(&params, _params, sizeof(params)))
+ return -EFAULT;
+ if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE) && params.ticks < 1) {
+ err = -EINVAL;
+ goto _end;
+ }
+ if (params.queue_size > 0 &&
+ (params.queue_size < 32 || params.queue_size > 1024)) {
+ err = -EINVAL;
+ goto _end;
+ }
+ if (params.filter & ~((1<<SNDRV_TIMER_EVENT_RESOLUTION)|
+ (1<<SNDRV_TIMER_EVENT_TICK)|
+ (1<<SNDRV_TIMER_EVENT_START)|
+ (1<<SNDRV_TIMER_EVENT_STOP)|
+ (1<<SNDRV_TIMER_EVENT_CONTINUE)|
+ (1<<SNDRV_TIMER_EVENT_PAUSE)|
+ (1<<SNDRV_TIMER_EVENT_SUSPEND)|
+ (1<<SNDRV_TIMER_EVENT_RESUME)|
+ (1<<SNDRV_TIMER_EVENT_MSTART)|
+ (1<<SNDRV_TIMER_EVENT_MSTOP)|
+ (1<<SNDRV_TIMER_EVENT_MCONTINUE)|
+ (1<<SNDRV_TIMER_EVENT_MPAUSE)|
+ (1<<SNDRV_TIMER_EVENT_MSUSPEND)|
+ (1<<SNDRV_TIMER_EVENT_MRESUME))) {
+ err = -EINVAL;
+ goto _end;
+ }
+ snd_timer_stop(tu->timeri);
+ spin_lock_irq(&t->lock);
+ tu->timeri->flags &= ~(SNDRV_TIMER_IFLG_AUTO|
+ SNDRV_TIMER_IFLG_EXCLUSIVE|
+ SNDRV_TIMER_IFLG_EARLY_EVENT);
+ if (params.flags & SNDRV_TIMER_PSFLG_AUTO)
+ tu->timeri->flags |= SNDRV_TIMER_IFLG_AUTO;
+ if (params.flags & SNDRV_TIMER_PSFLG_EXCLUSIVE)
+ tu->timeri->flags |= SNDRV_TIMER_IFLG_EXCLUSIVE;
+ if (params.flags & SNDRV_TIMER_PSFLG_EARLY_EVENT)
+ tu->timeri->flags |= SNDRV_TIMER_IFLG_EARLY_EVENT;
+ spin_unlock_irq(&t->lock);
+ if (params.queue_size > 0 &&
+ (unsigned int)tu->queue_size != params.queue_size) {
+ if (tu->tread) {
+ ttr = kmalloc(params.queue_size * sizeof(*ttr),
+ GFP_KERNEL);
+ if (ttr) {
+ kfree(tu->tqueue);
+ tu->queue_size = params.queue_size;
+ tu->tqueue = ttr;
+ }
+ } else {
+ tr = kmalloc(params.queue_size * sizeof(*tr),
+ GFP_KERNEL);
+ if (tr) {
+ kfree(tu->queue);
+ tu->queue_size = params.queue_size;
+ tu->queue = tr;
+ }
+ }
+ }
+ tu->qhead = tu->qtail = tu->qused = 0;
+ if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) {
+ if (tu->tread) {
+ struct snd_timer_tread tread;
+ tread.event = SNDRV_TIMER_EVENT_EARLY;
+ tread.tstamp.tv_sec = 0;
+ tread.tstamp.tv_nsec = 0;
+ tread.val = 0;
+ snd_timer_user_append_to_tqueue(tu, &tread);
+ } else {
+ struct snd_timer_read *r = &tu->queue[0];
+ r->resolution = 0;
+ r->ticks = 0;
+ tu->qused++;
+ tu->qtail++;
+ }
+ }
+ tu->filter = params.filter;
+ tu->ticks = params.ticks;
+ err = 0;
+ _end:
+ if (copy_to_user(_params, &params, sizeof(params)))
+ return -EFAULT;
+ return err;
+}
+
+static int snd_timer_user_status(struct file *file,
+ struct snd_timer_status __user *_status)
+{
+ struct snd_timer_user *tu;
+ struct snd_timer_status status;
+
+ tu = file->private_data;
+ if (!tu->timeri)
+ return -EBADFD;
+ memset(&status, 0, sizeof(status));
+ status.tstamp = tu->tstamp;
+ status.resolution = snd_timer_resolution(tu->timeri);
+ status.lost = tu->timeri->lost;
+ status.overrun = tu->overrun;
+ spin_lock_irq(&tu->qlock);
+ status.queue = tu->qused;
+ spin_unlock_irq(&tu->qlock);
+ if (copy_to_user(_status, &status, sizeof(status)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_timer_user_start(struct file *file)
+{
+ int err;
+ struct snd_timer_user *tu;
+
+ tu = file->private_data;
+ if (!tu->timeri)
+ return -EBADFD;
+ snd_timer_stop(tu->timeri);
+ tu->timeri->lost = 0;
+ tu->last_resolution = 0;
+ return (err = snd_timer_start(tu->timeri, tu->ticks)) < 0 ? err : 0;
+}
+
+static int snd_timer_user_stop(struct file *file)
+{
+ int err;
+ struct snd_timer_user *tu;
+
+ tu = file->private_data;
+ if (!tu->timeri)
+ return -EBADFD;
+ return (err = snd_timer_stop(tu->timeri)) < 0 ? err : 0;
+}
+
+static int snd_timer_user_continue(struct file *file)
+{
+ int err;
+ struct snd_timer_user *tu;
+
+ tu = file->private_data;
+ if (!tu->timeri)
+ return -EBADFD;
+ tu->timeri->lost = 0;
+ return (err = snd_timer_continue(tu->timeri)) < 0 ? err : 0;
+}
+
+static int snd_timer_user_pause(struct file *file)
+{
+ int err;
+ struct snd_timer_user *tu;
+
+ tu = file->private_data;
+ if (!tu->timeri)
+ return -EBADFD;
+ return (err = snd_timer_pause(tu->timeri)) < 0 ? err : 0;
+}
+
+enum {
+ SNDRV_TIMER_IOCTL_START_OLD = _IO('T', 0x20),
+ SNDRV_TIMER_IOCTL_STOP_OLD = _IO('T', 0x21),
+ SNDRV_TIMER_IOCTL_CONTINUE_OLD = _IO('T', 0x22),
+ SNDRV_TIMER_IOCTL_PAUSE_OLD = _IO('T', 0x23),
+};
+
+static long snd_timer_user_ioctl(struct file *file, unsigned int cmd,
+ unsigned long arg)
+{
+ struct snd_timer_user *tu;
+ void __user *argp = (void __user *)arg;
+ int __user *p = argp;
+
+ tu = file->private_data;
+ switch (cmd) {
+ case SNDRV_TIMER_IOCTL_PVERSION:
+ return put_user(SNDRV_TIMER_VERSION, p) ? -EFAULT : 0;
+ case SNDRV_TIMER_IOCTL_NEXT_DEVICE:
+ return snd_timer_user_next_device(argp);
+ case SNDRV_TIMER_IOCTL_TREAD:
+ {
+ int xarg;
+
+ mutex_lock(&tu->tread_sem);
+ if (tu->timeri) { /* too late */
+ mutex_unlock(&tu->tread_sem);
+ return -EBUSY;
+ }
+ if (get_user(xarg, p)) {
+ mutex_unlock(&tu->tread_sem);
+ return -EFAULT;
+ }
+ tu->tread = xarg ? 1 : 0;
+ mutex_unlock(&tu->tread_sem);
+ return 0;
+ }
+ case SNDRV_TIMER_IOCTL_GINFO:
+ return snd_timer_user_ginfo(file, argp);
+ case SNDRV_TIMER_IOCTL_GPARAMS:
+ return snd_timer_user_gparams(file, argp);
+ case SNDRV_TIMER_IOCTL_GSTATUS:
+ return snd_timer_user_gstatus(file, argp);
+ case SNDRV_TIMER_IOCTL_SELECT:
+ return snd_timer_user_tselect(file, argp);
+ case SNDRV_TIMER_IOCTL_INFO:
+ return snd_timer_user_info(file, argp);
+ case SNDRV_TIMER_IOCTL_PARAMS:
+ return snd_timer_user_params(file, argp);
+ case SNDRV_TIMER_IOCTL_STATUS:
+ return snd_timer_user_status(file, argp);
+ case SNDRV_TIMER_IOCTL_START:
+ case SNDRV_TIMER_IOCTL_START_OLD:
+ return snd_timer_user_start(file);
+ case SNDRV_TIMER_IOCTL_STOP:
+ case SNDRV_TIMER_IOCTL_STOP_OLD:
+ return snd_timer_user_stop(file);
+ case SNDRV_TIMER_IOCTL_CONTINUE:
+ case SNDRV_TIMER_IOCTL_CONTINUE_OLD:
+ return snd_timer_user_continue(file);
+ case SNDRV_TIMER_IOCTL_PAUSE:
+ case SNDRV_TIMER_IOCTL_PAUSE_OLD:
+ return snd_timer_user_pause(file);
+ }
+ return -ENOTTY;
+}
+
+static int snd_timer_user_fasync(int fd, struct file * file, int on)
+{
+ struct snd_timer_user *tu;
+ int err;
+
+ tu = file->private_data;
+ err = fasync_helper(fd, file, on, &tu->fasync);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static ssize_t snd_timer_user_read(struct file *file, char __user *buffer,
+ size_t count, loff_t *offset)
+{
+ struct snd_timer_user *tu;
+ long result = 0, unit;
+ int err = 0;
+
+ tu = file->private_data;
+ unit = tu->tread ? sizeof(struct snd_timer_tread) : sizeof(struct snd_timer_read);
+ spin_lock_irq(&tu->qlock);
+ while ((long)count - result >= unit) {
+ while (!tu->qused) {
+ wait_queue_t wait;
+
+ if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) {
+ err = -EAGAIN;
+ break;
+ }
+
+ set_current_state(TASK_INTERRUPTIBLE);
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&tu->qchange_sleep, &wait);
+
+ spin_unlock_irq(&tu->qlock);
+ schedule();
+ spin_lock_irq(&tu->qlock);
+
+ remove_wait_queue(&tu->qchange_sleep, &wait);
+
+ if (signal_pending(current)) {
+ err = -ERESTARTSYS;
+ break;
+ }
+ }
+
+ spin_unlock_irq(&tu->qlock);
+ if (err < 0)
+ goto _error;
+
+ if (tu->tread) {
+ if (copy_to_user(buffer, &tu->tqueue[tu->qhead++],
+ sizeof(struct snd_timer_tread))) {
+ err = -EFAULT;
+ goto _error;
+ }
+ } else {
+ if (copy_to_user(buffer, &tu->queue[tu->qhead++],
+ sizeof(struct snd_timer_read))) {
+ err = -EFAULT;
+ goto _error;
+ }
+ }
+
+ tu->qhead %= tu->queue_size;
+
+ result += unit;
+ buffer += unit;
+
+ spin_lock_irq(&tu->qlock);
+ tu->qused--;
+ }
+ spin_unlock_irq(&tu->qlock);
+ _error:
+ return result > 0 ? result : err;
+}
+
+static unsigned int snd_timer_user_poll(struct file *file, poll_table * wait)
+{
+ unsigned int mask;
+ struct snd_timer_user *tu;
+
+ tu = file->private_data;
+
+ poll_wait(file, &tu->qchange_sleep, wait);
+
+ mask = 0;
+ if (tu->qused)
+ mask |= POLLIN | POLLRDNORM;
+
+ return mask;
+}
+
+#ifdef CONFIG_COMPAT
+#include "timer_compat.c"
+#else
+#define snd_timer_user_ioctl_compat NULL
+#endif
+
+static const struct file_operations snd_timer_f_ops =
+{
+ .owner = THIS_MODULE,
+ .read = snd_timer_user_read,
+ .open = snd_timer_user_open,
+ .release = snd_timer_user_release,
+ .poll = snd_timer_user_poll,
+ .unlocked_ioctl = snd_timer_user_ioctl,
+ .compat_ioctl = snd_timer_user_ioctl_compat,
+ .fasync = snd_timer_user_fasync,
+};
+
+/*
+ * ENTRY functions
+ */
+
+static int __init alsa_timer_init(void)
+{
+ int err;
+
+#ifdef SNDRV_OSS_INFO_DEV_TIMERS
+ snd_oss_info_register(SNDRV_OSS_INFO_DEV_TIMERS, SNDRV_CARDS - 1,
+ "system timer");
+#endif
+
+ if ((err = snd_timer_register_system()) < 0)
+ snd_printk(KERN_ERR "unable to register system timer (%i)\n",
+ err);
+ if ((err = snd_register_device(SNDRV_DEVICE_TYPE_TIMER, NULL, 0,
+ &snd_timer_f_ops, NULL, "timer")) < 0)
+ snd_printk(KERN_ERR "unable to register timer device (%i)\n",
+ err);
+ snd_timer_proc_init();
+ return 0;
+}
+
+static void __exit alsa_timer_exit(void)
+{
+ struct list_head *p, *n;
+
+ snd_unregister_device(SNDRV_DEVICE_TYPE_TIMER, NULL, 0);
+ /* unregister the system timer */
+ list_for_each_safe(p, n, &snd_timer_list) {
+ struct snd_timer *timer = list_entry(p, struct snd_timer, device_list);
+ snd_timer_free(timer);
+ }
+ snd_timer_proc_done();
+#ifdef SNDRV_OSS_INFO_DEV_TIMERS
+ snd_oss_info_unregister(SNDRV_OSS_INFO_DEV_TIMERS, SNDRV_CARDS - 1);
+#endif
+}
+
+module_init(alsa_timer_init)
+module_exit(alsa_timer_exit)
+
+EXPORT_SYMBOL(snd_timer_open);
+EXPORT_SYMBOL(snd_timer_close);
+EXPORT_SYMBOL(snd_timer_resolution);
+EXPORT_SYMBOL(snd_timer_start);
+EXPORT_SYMBOL(snd_timer_stop);
+EXPORT_SYMBOL(snd_timer_continue);
+EXPORT_SYMBOL(snd_timer_pause);
+EXPORT_SYMBOL(snd_timer_new);
+EXPORT_SYMBOL(snd_timer_notify);
+EXPORT_SYMBOL(snd_timer_global_new);
+EXPORT_SYMBOL(snd_timer_global_free);
+EXPORT_SYMBOL(snd_timer_global_register);
+EXPORT_SYMBOL(snd_timer_interrupt);
diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c
new file mode 100644
index 0000000..e05802a
--- /dev/null
+++ b/sound/core/timer_compat.c
@@ -0,0 +1,127 @@
+/*
+ * 32bit -> 64bit ioctl wrapper for timer API
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* This file included from timer.c */
+
+#include <linux/compat.h>
+
+struct snd_timer_info32 {
+ u32 flags;
+ s32 card;
+ unsigned char id[64];
+ unsigned char name[80];
+ u32 reserved0;
+ u32 resolution;
+ unsigned char reserved[64];
+};
+
+static int snd_timer_user_info_compat(struct file *file,
+ struct snd_timer_info32 __user *_info)
+{
+ struct snd_timer_user *tu;
+ struct snd_timer_info32 info;
+ struct snd_timer *t;
+
+ tu = file->private_data;
+ if (snd_BUG_ON(!tu->timeri))
+ return -ENXIO;
+ t = tu->timeri->timer;
+ if (snd_BUG_ON(!t))
+ return -ENXIO;
+ memset(&info, 0, sizeof(info));
+ info.card = t->card ? t->card->number : -1;
+ if (t->hw.flags & SNDRV_TIMER_HW_SLAVE)
+ info.flags |= SNDRV_TIMER_FLG_SLAVE;
+ strlcpy(info.id, t->id, sizeof(info.id));
+ strlcpy(info.name, t->name, sizeof(info.name));
+ info.resolution = t->hw.resolution;
+ if (copy_to_user(_info, &info, sizeof(*_info)))
+ return -EFAULT;
+ return 0;
+}
+
+struct snd_timer_status32 {
+ struct compat_timespec tstamp;
+ u32 resolution;
+ u32 lost;
+ u32 overrun;
+ u32 queue;
+ unsigned char reserved[64];
+};
+
+static int snd_timer_user_status_compat(struct file *file,
+ struct snd_timer_status32 __user *_status)
+{
+ struct snd_timer_user *tu;
+ struct snd_timer_status status;
+
+ tu = file->private_data;
+ if (snd_BUG_ON(!tu->timeri))
+ return -ENXIO;
+ memset(&status, 0, sizeof(status));
+ status.tstamp = tu->tstamp;
+ status.resolution = snd_timer_resolution(tu->timeri);
+ status.lost = tu->timeri->lost;
+ status.overrun = tu->overrun;
+ spin_lock_irq(&tu->qlock);
+ status.queue = tu->qused;
+ spin_unlock_irq(&tu->qlock);
+ if (copy_to_user(_status, &status, sizeof(status)))
+ return -EFAULT;
+ return 0;
+}
+
+/*
+ */
+
+enum {
+ SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32),
+ SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32),
+};
+
+static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ void __user *argp = compat_ptr(arg);
+
+ switch (cmd) {
+ case SNDRV_TIMER_IOCTL_PVERSION:
+ case SNDRV_TIMER_IOCTL_TREAD:
+ case SNDRV_TIMER_IOCTL_GINFO:
+ case SNDRV_TIMER_IOCTL_GPARAMS:
+ case SNDRV_TIMER_IOCTL_GSTATUS:
+ case SNDRV_TIMER_IOCTL_SELECT:
+ case SNDRV_TIMER_IOCTL_PARAMS:
+ case SNDRV_TIMER_IOCTL_START:
+ case SNDRV_TIMER_IOCTL_START_OLD:
+ case SNDRV_TIMER_IOCTL_STOP:
+ case SNDRV_TIMER_IOCTL_STOP_OLD:
+ case SNDRV_TIMER_IOCTL_CONTINUE:
+ case SNDRV_TIMER_IOCTL_CONTINUE_OLD:
+ case SNDRV_TIMER_IOCTL_PAUSE:
+ case SNDRV_TIMER_IOCTL_PAUSE_OLD:
+ case SNDRV_TIMER_IOCTL_NEXT_DEVICE:
+ return snd_timer_user_ioctl(file, cmd, (unsigned long)argp);
+ case SNDRV_TIMER_IOCTL_INFO32:
+ return snd_timer_user_info_compat(file, argp);
+ case SNDRV_TIMER_IOCTL_STATUS32:
+ return snd_timer_user_status_compat(file, argp);
+ }
+ return -ENOIOCTLCMD;
+}
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
new file mode 100644
index 0000000..4cc57f9
--- /dev/null
+++ b/sound/core/vmaster.c
@@ -0,0 +1,371 @@
+/*
+ * Virtual master and slave controls
+ *
+ * Copyright (c) 2008 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation, version 2.
+ *
+ */
+
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+
+/*
+ * a subset of information returned via ctl info callback
+ */
+struct link_ctl_info {
+ int type; /* value type */
+ int count; /* item count */
+ int min_val, max_val; /* min, max values */
+};
+
+/*
+ * link master - this contains a list of slave controls that are
+ * identical types, i.e. info returns the same value type and value
+ * ranges, but may have different number of counts.
+ *
+ * The master control is so far only mono volume/switch for simplicity.
+ * The same value will be applied to all slaves.
+ */
+struct link_master {
+ struct list_head slaves;
+ struct link_ctl_info info;
+ int val; /* the master value */
+ unsigned int tlv[4];
+};
+
+/*
+ * link slave - this contains a slave control element
+ *
+ * It fakes the control callbacsk with additional attenuation by the
+ * master control. A slave may have either one or two channels.
+ */
+
+struct link_slave {
+ struct list_head list;
+ struct link_master *master;
+ struct link_ctl_info info;
+ int vals[2]; /* current values */
+ struct snd_kcontrol slave; /* the copy of original control entry */
+};
+
+/* get the slave ctl info and save the initial values */
+static int slave_init(struct link_slave *slave)
+{
+ struct snd_ctl_elem_info *uinfo;
+ struct snd_ctl_elem_value *uctl;
+ int err, ch;
+
+ if (slave->info.count)
+ return 0; /* already initialized */
+
+ uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL);
+ if (!uinfo)
+ return -ENOMEM;
+ uinfo->id = slave->slave.id;
+ err = slave->slave.info(&slave->slave, uinfo);
+ if (err < 0) {
+ kfree(uinfo);
+ return err;
+ }
+ slave->info.type = uinfo->type;
+ slave->info.count = uinfo->count;
+ if (slave->info.count > 2 ||
+ (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
+ slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
+ snd_printk(KERN_ERR "invalid slave element\n");
+ kfree(uinfo);
+ return -EINVAL;
+ }
+ slave->info.min_val = uinfo->value.integer.min;
+ slave->info.max_val = uinfo->value.integer.max;
+ kfree(uinfo);
+
+ uctl = kmalloc(sizeof(*uctl), GFP_KERNEL);
+ if (!uctl)
+ return -ENOMEM;
+ uctl->id = slave->slave.id;
+ err = slave->slave.get(&slave->slave, uctl);
+ for (ch = 0; ch < slave->info.count; ch++)
+ slave->vals[ch] = uctl->value.integer.value[ch];
+ kfree(uctl);
+ return 0;
+}
+
+/* initialize master volume */
+static int master_init(struct link_master *master)
+{
+ struct link_slave *slave;
+
+ if (master->info.count)
+ return 0; /* already initialized */
+
+ list_for_each_entry(slave, &master->slaves, list) {
+ int err = slave_init(slave);
+ if (err < 0)
+ return err;
+ master->info = slave->info;
+ master->info.count = 1; /* always mono */
+ /* set full volume as default (= no attenuation) */
+ master->val = master->info.max_val;
+ return 0;
+ }
+ return -ENOENT;
+}
+
+static int slave_get_val(struct link_slave *slave,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int err, ch;
+
+ err = slave_init(slave);
+ if (err < 0)
+ return err;
+ for (ch = 0; ch < slave->info.count; ch++)
+ ucontrol->value.integer.value[ch] = slave->vals[ch];
+ return 0;
+}
+
+static int slave_put_val(struct link_slave *slave,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int err, ch, vol;
+
+ err = master_init(slave->master);
+ if (err < 0)
+ return err;
+
+ switch (slave->info.type) {
+ case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
+ for (ch = 0; ch < slave->info.count; ch++)
+ ucontrol->value.integer.value[ch] &=
+ !!slave->master->val;
+ break;
+ case SNDRV_CTL_ELEM_TYPE_INTEGER:
+ for (ch = 0; ch < slave->info.count; ch++) {
+ /* max master volume is supposed to be 0 dB */
+ vol = ucontrol->value.integer.value[ch];
+ vol += slave->master->val - slave->master->info.max_val;
+ if (vol < slave->info.min_val)
+ vol = slave->info.min_val;
+ else if (vol > slave->info.max_val)
+ vol = slave->info.max_val;
+ ucontrol->value.integer.value[ch] = vol;
+ }
+ break;
+ }
+ return slave->slave.put(&slave->slave, ucontrol);
+}
+
+/*
+ * ctl callbacks for slaves
+ */
+static int slave_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+ return slave->slave.info(&slave->slave, uinfo);
+}
+
+static int slave_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+ return slave_get_val(slave, ucontrol);
+}
+
+static int slave_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+ int err, ch, changed = 0;
+
+ err = slave_init(slave);
+ if (err < 0)
+ return err;
+ for (ch = 0; ch < slave->info.count; ch++) {
+ if (slave->vals[ch] != ucontrol->value.integer.value[ch]) {
+ changed = 1;
+ slave->vals[ch] = ucontrol->value.integer.value[ch];
+ }
+ }
+ if (!changed)
+ return 0;
+ return slave_put_val(slave, ucontrol);
+}
+
+static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
+ int op_flag, unsigned int size,
+ unsigned int __user *tlv)
+{
+ struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+ /* FIXME: this assumes that the max volume is 0 dB */
+ return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv);
+}
+
+static void slave_free(struct snd_kcontrol *kcontrol)
+{
+ struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+ if (slave->slave.private_free)
+ slave->slave.private_free(&slave->slave);
+ if (slave->master)
+ list_del(&slave->list);
+ kfree(slave);
+}
+
+/*
+ * Add a slave control to the group with the given master control
+ *
+ * All slaves must be the same type (returning the same information
+ * via info callback). The fucntion doesn't check it, so it's your
+ * responsibility.
+ *
+ * Also, some additional limitations:
+ * - at most two channels
+ * - logarithmic volume control (dB level), no linear volume
+ * - master can only attenuate the volume, no gain
+ */
+int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
+{
+ struct link_master *master_link = snd_kcontrol_chip(master);
+ struct link_slave *srec;
+
+ srec = kzalloc(sizeof(*srec) +
+ slave->count * sizeof(*slave->vd), GFP_KERNEL);
+ if (!srec)
+ return -ENOMEM;
+ srec->slave = *slave;
+ memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd));
+ srec->master = master_link;
+
+ /* override callbacks */
+ slave->info = slave_info;
+ slave->get = slave_get;
+ slave->put = slave_put;
+ if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
+ slave->tlv.c = slave_tlv_cmd;
+ slave->private_data = srec;
+ slave->private_free = slave_free;
+
+ list_add_tail(&srec->list, &master_link->slaves);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ctl_add_slave);
+
+/*
+ * ctl callbacks for master controls
+ */
+static int master_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct link_master *master = snd_kcontrol_chip(kcontrol);
+ int ret;
+
+ ret = master_init(master);
+ if (ret < 0)
+ return ret;
+ uinfo->type = master->info.type;
+ uinfo->count = master->info.count;
+ uinfo->value.integer.min = master->info.min_val;
+ uinfo->value.integer.max = master->info.max_val;
+ return 0;
+}
+
+static int master_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct link_master *master = snd_kcontrol_chip(kcontrol);
+ int err = master_init(master);
+ if (err < 0)
+ return err;
+ ucontrol->value.integer.value[0] = master->val;
+ return 0;
+}
+
+static int master_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct link_master *master = snd_kcontrol_chip(kcontrol);
+ struct link_slave *slave;
+ struct snd_ctl_elem_value *uval;
+ int err, old_val;
+
+ err = master_init(master);
+ if (err < 0)
+ return err;
+ old_val = master->val;
+ if (ucontrol->value.integer.value[0] == old_val)
+ return 0;
+
+ uval = kmalloc(sizeof(*uval), GFP_KERNEL);
+ if (!uval)
+ return -ENOMEM;
+ list_for_each_entry(slave, &master->slaves, list) {
+ master->val = old_val;
+ uval->id = slave->slave.id;
+ slave_get_val(slave, uval);
+ master->val = ucontrol->value.integer.value[0];
+ slave_put_val(slave, uval);
+ }
+ kfree(uval);
+ return 1;
+}
+
+static void master_free(struct snd_kcontrol *kcontrol)
+{
+ struct link_master *master = snd_kcontrol_chip(kcontrol);
+ struct link_slave *slave;
+
+ list_for_each_entry(slave, &master->slaves, list)
+ slave->master = NULL;
+ kfree(master);
+}
+
+
+/*
+ * Create a virtual master control with the given name
+ */
+struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
+ const unsigned int *tlv)
+{
+ struct link_master *master;
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_new knew;
+
+ memset(&knew, 0, sizeof(knew));
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.name = name;
+ knew.info = master_info;
+
+ master = kzalloc(sizeof(*master), GFP_KERNEL);
+ if (!master)
+ return NULL;
+ INIT_LIST_HEAD(&master->slaves);
+
+ kctl = snd_ctl_new1(&knew, master);
+ if (!kctl) {
+ kfree(master);
+ return NULL;
+ }
+ /* override some callbacks */
+ kctl->info = master_info;
+ kctl->get = master_get;
+ kctl->put = master_put;
+ kctl->private_free = master_free;
+
+ /* additional (constant) TLV read */
+ if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) {
+ kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ memcpy(master->tlv, tlv, sizeof(master->tlv));
+ kctl->tlv.p = master->tlv;
+ }
+
+ return kctl;
+}
+
+EXPORT_SYMBOL(snd_ctl_make_virtual_master);
diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
new file mode 100644
index 0000000..255fd18
--- /dev/null
+++ b/sound/drivers/Kconfig
@@ -0,0 +1,185 @@
+config SND_MPU401_UART
+ tristate
+ select SND_RAWMIDI
+
+config SND_OPL3_LIB
+ tristate
+ select SND_TIMER
+ select SND_HWDEP
+
+config SND_OPL4_LIB
+ tristate
+ select SND_TIMER
+ select SND_HWDEP
+
+config SND_VX_LIB
+ tristate
+ select SND_HWDEP
+ select SND_PCM
+
+config SND_AC97_CODEC
+ tristate
+ select SND_PCM
+ select AC97_BUS
+ select SND_VMASTER
+
+menuconfig SND_DRIVERS
+ bool "Generic sound devices"
+ default y
+ help
+ Support for generic sound devices.
+
+if SND_DRIVERS
+
+config SND_PCSP
+ tristate "PC-Speaker support (READ HELP!)"
+ depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS
+ depends on INPUT
+ depends on EXPERIMENTAL
+ select SND_PCM
+ help
+ If you don't have a sound card in your computer, you can include a
+ driver for the PC speaker which allows it to act like a primitive
+ sound card.
+ This driver also replaces the pcspkr driver for beeps.
+
+ You can compile this as a module which will be called snd-pcsp.
+
+ WARNING: if you already have a soundcard, enabling this
+ driver may lead to a problem. Namely, it may get loaded
+ before the other sound driver of yours, making the
+ pc-speaker a default sound device. Which is likely not
+ what you want. To make this driver play nicely with other
+ sound driver, you can add this into your /etc/modprobe.conf:
+ options snd-pcsp index=2
+
+ You don't need this driver if you only want your pc-speaker to beep.
+ You don't need this driver if you have a tablet piezo beeper
+ in your PC instead of the real speaker.
+
+ Say N if you have a sound card.
+ Say M if you don't.
+ Say Y only if you really know what you do.
+
+config SND_DUMMY
+ tristate "Dummy (/dev/null) soundcard"
+ select SND_PCM
+ help
+ Say Y here to include the dummy driver. This driver does
+ nothing, but emulates various mixer controls and PCM devices.
+
+ You don't need this unless you're testing the hardware support
+ of programs using the ALSA API.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-dummy.
+
+config SND_VIRMIDI
+ tristate "Virtual MIDI soundcard"
+ depends on SND_SEQUENCER
+ select SND_TIMER
+ select SND_RAWMIDI
+ help
+ Say Y here to include the virtual MIDI driver. This driver
+ allows to connect applications using raw MIDI devices to
+ sequencer clients.
+
+ If you don't know what MIDI is, say N here.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-virmidi.
+
+config SND_MTPAV
+ tristate "MOTU MidiTimePiece AV multiport MIDI"
+ select SND_RAWMIDI
+ help
+ To use a MOTU MidiTimePiece AV multiport MIDI adapter
+ connected to the parallel port, say Y here and make sure that
+ the standard parallel port driver isn't used for the port.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-mtpav.
+
+config SND_MTS64
+ tristate "ESI Miditerminal 4140 driver"
+ depends on PARPORT
+ select SND_RAWMIDI
+ help
+ The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with
+ additional SMPTE Timecode capabilities for the parallel port.
+
+ Say 'Y' to include support for this device.
+
+ To compile this driver as a module, chose 'M' here: the module
+ will be called snd-mts64.
+
+config SND_SERIAL_U16550
+ tristate "UART16550 serial MIDI driver"
+ select SND_RAWMIDI
+ help
+ To include support for MIDI serial port interfaces, say Y here
+ and read <file:Documentation/sound/alsa/serial-u16550.txt>.
+ This driver works with serial UARTs 16550 and better.
+
+ This driver accesses the serial port hardware directly, so
+ make sure that the standard serial driver isn't used or
+ deactivated with setserial before loading this driver.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-serial-u16550.
+
+config SND_MPU401
+ tristate "Generic MPU-401 UART driver"
+ select SND_MPU401_UART
+ help
+ Say Y here to include support for MIDI ports compatible with
+ the Roland MPU-401 interface in UART mode.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-mpu401.
+
+config SND_PORTMAN2X4
+ tristate "Portman 2x4 driver"
+ depends on PARPORT
+ select SND_RAWMIDI
+ help
+ Say Y here to include support for Midiman Portman 2x4 parallel
+ port MIDI device.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-portman2x4.
+
+config SND_ML403_AC97CR
+ tristate "Xilinx ML403 AC97 Controller Reference"
+ depends on XILINX_VIRTEX
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the
+ opb_ac97_controller_ref_v1_00_a ip core found in Xilinx' ML403
+ reference design.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ml403_ac97cr.
+
+config SND_AC97_POWER_SAVE
+ bool "AC97 Power-Saving Mode"
+ depends on SND_AC97_CODEC && EXPERIMENTAL
+ default n
+ help
+ Say Y here to enable the aggressive power-saving support of
+ AC97 codecs. In this mode, the power-mode is dynamically
+ controlled at each open/close.
+
+ The mode is activated by passing power_save=1 option to
+ snd-ac97-codec driver. You can toggle it dynamically over
+ sysfs, too.
+
+config SND_AC97_POWER_SAVE_DEFAULT
+ int "Default time-out for AC97 power-save mode"
+ depends on SND_AC97_POWER_SAVE
+ default 0
+ help
+ The default time-out value in seconds for AC97 automatic
+ power-save mode. 0 means to disable the power-save mode.
+
+endif # SND_DRIVERS
diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile
new file mode 100644
index 0000000..d4a07f9
--- /dev/null
+++ b/sound/drivers/Makefile
@@ -0,0 +1,23 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-dummy-objs := dummy.o
+snd-mtpav-objs := mtpav.o
+snd-mts64-objs := mts64.o
+snd-portman2x4-objs := portman2x4.o
+snd-serial-u16550-objs := serial-u16550.o
+snd-virmidi-objs := virmidi.o
+snd-ml403-ac97cr-objs := ml403-ac97cr.o pcm-indirect2.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_DUMMY) += snd-dummy.o
+obj-$(CONFIG_SND_VIRMIDI) += snd-virmidi.o
+obj-$(CONFIG_SND_SERIAL_U16550) += snd-serial-u16550.o
+obj-$(CONFIG_SND_MTPAV) += snd-mtpav.o
+obj-$(CONFIG_SND_MTS64) += snd-mts64.o
+obj-$(CONFIG_SND_PORTMAN2X4) += snd-portman2x4.o
+obj-$(CONFIG_SND_ML403_AC97CR) += snd-ml403-ac97cr.o
+
+obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ pcsp/
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
new file mode 100644
index 0000000..73be7e1
--- /dev/null
+++ b/sound/drivers/dummy.c
@@ -0,0 +1,715 @@
+/*
+ * Dummy soundcard
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/jiffies.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Dummy soundcard (/dev/null)");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}");
+
+#define MAX_PCM_DEVICES 4
+#define MAX_PCM_SUBSTREAMS 16
+#define MAX_MIDI_DEVICES 2
+
+#if 0 /* emu10k1 emulation */
+#define MAX_BUFFER_SIZE (128 * 1024)
+static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime)
+{
+ int err;
+ err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX);
+ if (err < 0)
+ return err;
+ return 0;
+}
+#define add_playback_constraints emu10k1_playback_constraints
+#endif
+
+#if 0 /* RME9652 emulation */
+#define MAX_BUFFER_SIZE (26 * 64 * 1024)
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE
+#define USE_CHANNELS_MIN 26
+#define USE_CHANNELS_MAX 26
+#define USE_PERIODS_MIN 2
+#define USE_PERIODS_MAX 2
+#endif
+
+#if 0 /* ICE1712 emulation */
+#define MAX_BUFFER_SIZE (256 * 1024)
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE
+#define USE_CHANNELS_MIN 10
+#define USE_CHANNELS_MAX 10
+#define USE_PERIODS_MIN 1
+#define USE_PERIODS_MAX 1024
+#endif
+
+#if 0 /* UDA1341 emulation */
+#define MAX_BUFFER_SIZE (16380)
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define USE_CHANNELS_MIN 2
+#define USE_CHANNELS_MAX 2
+#define USE_PERIODS_MIN 2
+#define USE_PERIODS_MAX 255
+#endif
+
+#if 0 /* simple AC97 bridge (intel8x0) with 48kHz AC97 only codec */
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define USE_CHANNELS_MIN 2
+#define USE_CHANNELS_MAX 2
+#define USE_RATE SNDRV_PCM_RATE_48000
+#define USE_RATE_MIN 48000
+#define USE_RATE_MAX 48000
+#endif
+
+#if 0 /* CA0106 */
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define USE_CHANNELS_MIN 2
+#define USE_CHANNELS_MAX 2
+#define USE_RATE (SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_192000)
+#define USE_RATE_MIN 48000
+#define USE_RATE_MAX 192000
+#define MAX_BUFFER_SIZE ((65536-64)*8)
+#define MAX_PERIOD_SIZE (65536-64)
+#define USE_PERIODS_MIN 2
+#define USE_PERIODS_MAX 8
+#endif
+
+
+/* defaults */
+#ifndef MAX_BUFFER_SIZE
+#define MAX_BUFFER_SIZE (64*1024)
+#endif
+#ifndef MAX_PERIOD_SIZE
+#define MAX_PERIOD_SIZE MAX_BUFFER_SIZE
+#endif
+#ifndef USE_FORMATS
+#define USE_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
+#endif
+#ifndef USE_RATE
+#define USE_RATE SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000
+#define USE_RATE_MIN 5500
+#define USE_RATE_MAX 48000
+#endif
+#ifndef USE_CHANNELS_MIN
+#define USE_CHANNELS_MIN 1
+#endif
+#ifndef USE_CHANNELS_MAX
+#define USE_CHANNELS_MAX 2
+#endif
+#ifndef USE_PERIODS_MIN
+#define USE_PERIODS_MIN 1
+#endif
+#ifndef USE_PERIODS_MAX
+#define USE_PERIODS_MAX 1024
+#endif
+#ifndef add_playback_constraints
+#define add_playback_constraints(x) 0
+#endif
+#ifndef add_capture_constraints
+#define add_capture_constraints(x) 0
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0};
+static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8};
+//static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for dummy soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for dummy soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable this dummy soundcard.");
+module_param_array(pcm_devs, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver.");
+module_param_array(pcm_substreams, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver.");
+//module_param_array(midi_devs, int, NULL, 0444);
+//MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver.");
+
+static struct platform_device *devices[SNDRV_CARDS];
+
+#define MIXER_ADDR_MASTER 0
+#define MIXER_ADDR_LINE 1
+#define MIXER_ADDR_MIC 2
+#define MIXER_ADDR_SYNTH 3
+#define MIXER_ADDR_CD 4
+#define MIXER_ADDR_LAST 4
+
+struct snd_dummy {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ spinlock_t mixer_lock;
+ int mixer_volume[MIXER_ADDR_LAST+1][2];
+ int capture_source[MIXER_ADDR_LAST+1][2];
+};
+
+struct snd_dummy_pcm {
+ struct snd_dummy *dummy;
+ spinlock_t lock;
+ struct timer_list timer;
+ unsigned int pcm_buffer_size;
+ unsigned int pcm_period_size;
+ unsigned int pcm_bps; /* bytes per second */
+ unsigned int pcm_hz; /* HZ */
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int pcm_buf_pos; /* position in buffer */
+ struct snd_pcm_substream *substream;
+};
+
+
+static inline void snd_card_dummy_pcm_timer_start(struct snd_dummy_pcm *dpcm)
+{
+ dpcm->timer.expires = 1 + jiffies;
+ add_timer(&dpcm->timer);
+}
+
+static inline void snd_card_dummy_pcm_timer_stop(struct snd_dummy_pcm *dpcm)
+{
+ del_timer(&dpcm->timer);
+}
+
+static int snd_card_dummy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dummy_pcm *dpcm = runtime->private_data;
+ int err = 0;
+
+ spin_lock(&dpcm->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ snd_card_dummy_pcm_timer_start(dpcm);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ snd_card_dummy_pcm_timer_stop(dpcm);
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ spin_unlock(&dpcm->lock);
+ return 0;
+}
+
+static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dummy_pcm *dpcm = runtime->private_data;
+ int bps;
+
+ bps = snd_pcm_format_width(runtime->format) * runtime->rate *
+ runtime->channels / 8;
+
+ if (bps <= 0)
+ return -EINVAL;
+
+ dpcm->pcm_bps = bps;
+ dpcm->pcm_hz = HZ;
+ dpcm->pcm_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ dpcm->pcm_period_size = snd_pcm_lib_period_bytes(substream);
+ dpcm->pcm_irq_pos = 0;
+ dpcm->pcm_buf_pos = 0;
+
+ snd_pcm_format_set_silence(runtime->format, runtime->dma_area,
+ bytes_to_samples(runtime, runtime->dma_bytes));
+
+ return 0;
+}
+
+static void snd_card_dummy_pcm_timer_function(unsigned long data)
+{
+ struct snd_dummy_pcm *dpcm = (struct snd_dummy_pcm *)data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dpcm->lock, flags);
+ dpcm->timer.expires = 1 + jiffies;
+ add_timer(&dpcm->timer);
+ dpcm->pcm_irq_pos += dpcm->pcm_bps;
+ dpcm->pcm_buf_pos += dpcm->pcm_bps;
+ dpcm->pcm_buf_pos %= dpcm->pcm_buffer_size * dpcm->pcm_hz;
+ if (dpcm->pcm_irq_pos >= dpcm->pcm_period_size * dpcm->pcm_hz) {
+ dpcm->pcm_irq_pos %= dpcm->pcm_period_size * dpcm->pcm_hz;
+ spin_unlock_irqrestore(&dpcm->lock, flags);
+ snd_pcm_period_elapsed(dpcm->substream);
+ } else
+ spin_unlock_irqrestore(&dpcm->lock, flags);
+}
+
+static snd_pcm_uframes_t snd_card_dummy_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dummy_pcm *dpcm = runtime->private_data;
+
+ return bytes_to_frames(runtime, dpcm->pcm_buf_pos / dpcm->pcm_hz);
+}
+
+static struct snd_pcm_hardware snd_card_dummy_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = USE_FORMATS,
+ .rates = USE_RATE,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .channels_min = USE_CHANNELS_MIN,
+ .channels_max = USE_CHANNELS_MAX,
+ .buffer_bytes_max = MAX_BUFFER_SIZE,
+ .period_bytes_min = 64,
+ .period_bytes_max = MAX_PERIOD_SIZE,
+ .periods_min = USE_PERIODS_MIN,
+ .periods_max = USE_PERIODS_MAX,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_card_dummy_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = USE_FORMATS,
+ .rates = USE_RATE,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .channels_min = USE_CHANNELS_MIN,
+ .channels_max = USE_CHANNELS_MAX,
+ .buffer_bytes_max = MAX_BUFFER_SIZE,
+ .period_bytes_min = 64,
+ .period_bytes_max = MAX_PERIOD_SIZE,
+ .periods_min = USE_PERIODS_MIN,
+ .periods_max = USE_PERIODS_MAX,
+ .fifo_size = 0,
+};
+
+static void snd_card_dummy_runtime_free(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static int snd_card_dummy_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_card_dummy_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_dummy_pcm *new_pcm_stream(struct snd_pcm_substream *substream)
+{
+ struct snd_dummy_pcm *dpcm;
+
+ dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL);
+ if (! dpcm)
+ return dpcm;
+ init_timer(&dpcm->timer);
+ dpcm->timer.data = (unsigned long) dpcm;
+ dpcm->timer.function = snd_card_dummy_pcm_timer_function;
+ spin_lock_init(&dpcm->lock);
+ dpcm->substream = substream;
+ return dpcm;
+}
+
+static int snd_card_dummy_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dummy_pcm *dpcm;
+ int err;
+
+ if ((dpcm = new_pcm_stream(substream)) == NULL)
+ return -ENOMEM;
+ runtime->private_data = dpcm;
+ /* makes the infrastructure responsible for freeing dpcm */
+ runtime->private_free = snd_card_dummy_runtime_free;
+ runtime->hw = snd_card_dummy_playback;
+ if (substream->pcm->device & 1) {
+ runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED;
+ runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED;
+ }
+ if (substream->pcm->device & 2)
+ runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID);
+ err = add_playback_constraints(runtime);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_card_dummy_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dummy_pcm *dpcm;
+ int err;
+
+ if ((dpcm = new_pcm_stream(substream)) == NULL)
+ return -ENOMEM;
+ runtime->private_data = dpcm;
+ /* makes the infrastructure responsible for freeing dpcm */
+ runtime->private_free = snd_card_dummy_runtime_free;
+ runtime->hw = snd_card_dummy_capture;
+ if (substream->pcm->device == 1) {
+ runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED;
+ runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED;
+ }
+ if (substream->pcm->device & 2)
+ runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID);
+ err = add_capture_constraints(runtime);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_card_dummy_playback_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static int snd_card_dummy_capture_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static struct snd_pcm_ops snd_card_dummy_playback_ops = {
+ .open = snd_card_dummy_playback_open,
+ .close = snd_card_dummy_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_card_dummy_hw_params,
+ .hw_free = snd_card_dummy_hw_free,
+ .prepare = snd_card_dummy_pcm_prepare,
+ .trigger = snd_card_dummy_pcm_trigger,
+ .pointer = snd_card_dummy_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_card_dummy_capture_ops = {
+ .open = snd_card_dummy_capture_open,
+ .close = snd_card_dummy_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_card_dummy_hw_params,
+ .hw_free = snd_card_dummy_hw_free,
+ .prepare = snd_card_dummy_pcm_prepare,
+ .trigger = snd_card_dummy_pcm_trigger,
+ .pointer = snd_card_dummy_pcm_pointer,
+};
+
+static int __devinit snd_card_dummy_pcm(struct snd_dummy *dummy, int device,
+ int substreams)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(dummy->card, "Dummy PCM", device,
+ substreams, substreams, &pcm);
+ if (err < 0)
+ return err;
+ dummy->pcm = pcm;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_dummy_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_dummy_capture_ops);
+ pcm->private_data = dummy;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "Dummy PCM");
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 0, 64*1024);
+ return 0;
+}
+
+#define DUMMY_VOLUME(xname, xindex, addr) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .index = xindex, \
+ .info = snd_dummy_volume_info, \
+ .get = snd_dummy_volume_get, .put = snd_dummy_volume_put, \
+ .private_value = addr, \
+ .tlv = { .p = db_scale_dummy } }
+
+static int snd_dummy_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = -50;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int snd_dummy_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dummy *dummy = snd_kcontrol_chip(kcontrol);
+ int addr = kcontrol->private_value;
+
+ spin_lock_irq(&dummy->mixer_lock);
+ ucontrol->value.integer.value[0] = dummy->mixer_volume[addr][0];
+ ucontrol->value.integer.value[1] = dummy->mixer_volume[addr][1];
+ spin_unlock_irq(&dummy->mixer_lock);
+ return 0;
+}
+
+static int snd_dummy_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dummy *dummy = snd_kcontrol_chip(kcontrol);
+ int change, addr = kcontrol->private_value;
+ int left, right;
+
+ left = ucontrol->value.integer.value[0];
+ if (left < -50)
+ left = -50;
+ if (left > 100)
+ left = 100;
+ right = ucontrol->value.integer.value[1];
+ if (right < -50)
+ right = -50;
+ if (right > 100)
+ right = 100;
+ spin_lock_irq(&dummy->mixer_lock);
+ change = dummy->mixer_volume[addr][0] != left ||
+ dummy->mixer_volume[addr][1] != right;
+ dummy->mixer_volume[addr][0] = left;
+ dummy->mixer_volume[addr][1] = right;
+ spin_unlock_irq(&dummy->mixer_lock);
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_dummy, -4500, 30, 0);
+
+#define DUMMY_CAPSRC(xname, xindex, addr) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_dummy_capsrc_info, \
+ .get = snd_dummy_capsrc_get, .put = snd_dummy_capsrc_put, \
+ .private_value = addr }
+
+#define snd_dummy_capsrc_info snd_ctl_boolean_stereo_info
+
+static int snd_dummy_capsrc_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dummy *dummy = snd_kcontrol_chip(kcontrol);
+ int addr = kcontrol->private_value;
+
+ spin_lock_irq(&dummy->mixer_lock);
+ ucontrol->value.integer.value[0] = dummy->capture_source[addr][0];
+ ucontrol->value.integer.value[1] = dummy->capture_source[addr][1];
+ spin_unlock_irq(&dummy->mixer_lock);
+ return 0;
+}
+
+static int snd_dummy_capsrc_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dummy *dummy = snd_kcontrol_chip(kcontrol);
+ int change, addr = kcontrol->private_value;
+ int left, right;
+
+ left = ucontrol->value.integer.value[0] & 1;
+ right = ucontrol->value.integer.value[1] & 1;
+ spin_lock_irq(&dummy->mixer_lock);
+ change = dummy->capture_source[addr][0] != left &&
+ dummy->capture_source[addr][1] != right;
+ dummy->capture_source[addr][0] = left;
+ dummy->capture_source[addr][1] = right;
+ spin_unlock_irq(&dummy->mixer_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_dummy_controls[] = {
+DUMMY_VOLUME("Master Volume", 0, MIXER_ADDR_MASTER),
+DUMMY_CAPSRC("Master Capture Switch", 0, MIXER_ADDR_MASTER),
+DUMMY_VOLUME("Synth Volume", 0, MIXER_ADDR_SYNTH),
+DUMMY_CAPSRC("Synth Capture Switch", 0, MIXER_ADDR_SYNTH),
+DUMMY_VOLUME("Line Volume", 0, MIXER_ADDR_LINE),
+DUMMY_CAPSRC("Line Capture Switch", 0, MIXER_ADDR_LINE),
+DUMMY_VOLUME("Mic Volume", 0, MIXER_ADDR_MIC),
+DUMMY_CAPSRC("Mic Capture Switch", 0, MIXER_ADDR_MIC),
+DUMMY_VOLUME("CD Volume", 0, MIXER_ADDR_CD),
+DUMMY_CAPSRC("CD Capture Switch", 0, MIXER_ADDR_CD)
+};
+
+static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy)
+{
+ struct snd_card *card = dummy->card;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!dummy))
+ return -EINVAL;
+ spin_lock_init(&dummy->mixer_lock);
+ strcpy(card->mixername, "Dummy Mixer");
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_dummy_controls); idx++) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_dummy_controls[idx], dummy));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int __devinit snd_dummy_probe(struct platform_device *devptr)
+{
+ struct snd_card *card;
+ struct snd_dummy *dummy;
+ int idx, err;
+ int dev = devptr->id;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_dummy));
+ if (card == NULL)
+ return -ENOMEM;
+ dummy = card->private_data;
+ dummy->card = card;
+ for (idx = 0; idx < MAX_PCM_DEVICES && idx < pcm_devs[dev]; idx++) {
+ if (pcm_substreams[dev] < 1)
+ pcm_substreams[dev] = 1;
+ if (pcm_substreams[dev] > MAX_PCM_SUBSTREAMS)
+ pcm_substreams[dev] = MAX_PCM_SUBSTREAMS;
+ err = snd_card_dummy_pcm(dummy, idx, pcm_substreams[dev]);
+ if (err < 0)
+ goto __nodev;
+ }
+ err = snd_card_dummy_new_mixer(dummy);
+ if (err < 0)
+ goto __nodev;
+ strcpy(card->driver, "Dummy");
+ strcpy(card->shortname, "Dummy");
+ sprintf(card->longname, "Dummy %i", dev + 1);
+
+ snd_card_set_dev(card, &devptr->dev);
+
+ err = snd_card_register(card);
+ if (err == 0) {
+ platform_set_drvdata(devptr, card);
+ return 0;
+ }
+ __nodev:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_dummy_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+ struct snd_dummy *dummy = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(dummy->pcm);
+ return 0;
+}
+
+static int snd_dummy_resume(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+#define SND_DUMMY_DRIVER "snd_dummy"
+
+static struct platform_driver snd_dummy_driver = {
+ .probe = snd_dummy_probe,
+ .remove = __devexit_p(snd_dummy_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_dummy_suspend,
+ .resume = snd_dummy_resume,
+#endif
+ .driver = {
+ .name = SND_DUMMY_DRIVER
+ },
+};
+
+static void snd_dummy_unregister_all(void)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(devices); ++i)
+ platform_device_unregister(devices[i]);
+ platform_driver_unregister(&snd_dummy_driver);
+}
+
+static int __init alsa_card_dummy_init(void)
+{
+ int i, cards, err;
+
+ err = platform_driver_register(&snd_dummy_driver);
+ if (err < 0)
+ return err;
+
+ cards = 0;
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ struct platform_device *device;
+ if (! enable[i])
+ continue;
+ device = platform_device_register_simple(SND_DUMMY_DRIVER,
+ i, NULL, 0);
+ if (IS_ERR(device))
+ continue;
+ if (!platform_get_drvdata(device)) {
+ platform_device_unregister(device);
+ continue;
+ }
+ devices[i] = device;
+ cards++;
+ }
+ if (!cards) {
+#ifdef MODULE
+ printk(KERN_ERR "Dummy soundcard not found or device busy\n");
+#endif
+ snd_dummy_unregister_all();
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_dummy_exit(void)
+{
+ snd_dummy_unregister_all();
+}
+
+module_init(alsa_card_dummy_init)
+module_exit(alsa_card_dummy_exit)
diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
new file mode 100644
index 0000000..7783843
--- /dev/null
+++ b/sound/drivers/ml403-ac97cr.c
@@ -0,0 +1,1354 @@
+/*
+ * ALSA driver for Xilinx ML403 AC97 Controller Reference
+ * IP: opb_ac97_controller_ref_v1_00_a (EDK 8.1i)
+ * IP: opb_ac97_controller_ref_v1_00_a (EDK 9.1i)
+ *
+ * Copyright (c) by 2007 Joachim Foerster <JOFT@gmx.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* Some notes / status of this driver:
+ *
+ * - Don't wonder about some strange implementations of things - especially the
+ * (heavy) shadowing of codec registers, with which I tried to reduce read
+ * accesses to a minimum, because after a variable amount of accesses, the AC97
+ * controller doesn't raise the register access finished bit anymore ...
+ *
+ * - Playback support seems to be pretty stable - no issues here.
+ * - Capture support "works" now, too. Overruns don't happen any longer so often.
+ * But there might still be some ...
+ */
+
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+
+#include <linux/platform_device.h>
+
+#include <linux/ioport.h>
+#include <linux/io.h>
+#include <linux/interrupt.h>
+
+/* HZ */
+#include <linux/param.h>
+/* jiffies, time_*() */
+#include <linux/jiffies.h>
+/* schedule_timeout*() */
+#include <linux/sched.h>
+/* spin_lock*() */
+#include <linux/spinlock.h>
+/* struct mutex, mutex_init(), mutex_*lock() */
+#include <linux/mutex.h>
+
+/* snd_printk(), snd_printd() */
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include "pcm-indirect2.h"
+
+
+#define SND_ML403_AC97CR_DRIVER "ml403-ac97cr"
+
+MODULE_AUTHOR("Joachim Foerster <JOFT@gmx.de>");
+MODULE_DESCRIPTION("Xilinx ML403 AC97 Controller Reference");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Xilinx,ML403 AC97 Controller Reference}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for ML403 AC97 Controller Reference.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for ML403 AC97 Controller Reference.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable this ML403 AC97 Controller Reference.");
+
+/* Special feature options */
+/*#define CODEC_WRITE_CHECK_RAF*/ /* don't return after a write to a codec
+ * register, while RAF bit is not set
+ */
+/* Debug options for code which may be removed completely in a final version */
+#ifdef CONFIG_SND_DEBUG
+/*#define CODEC_STAT*/ /* turn on some minimal "statistics"
+ * about codec register usage
+ */
+#define SND_PCM_INDIRECT2_STAT /* turn on some "statistics" about the
+ * process of copying bytes from the
+ * intermediate buffer to the hardware
+ * fifo and the other way round
+ */
+#endif
+
+/* Definition of a "level/facility dependent" printk(); may be removed
+ * completely in a final version
+ */
+#undef PDEBUG
+#ifdef CONFIG_SND_DEBUG
+/* "facilities" for PDEBUG */
+#define UNKNOWN (1<<0)
+#define CODEC_SUCCESS (1<<1)
+#define CODEC_FAKE (1<<2)
+#define INIT_INFO (1<<3)
+#define INIT_FAILURE (1<<4)
+#define WORK_INFO (1<<5)
+#define WORK_FAILURE (1<<6)
+
+#define PDEBUG_FACILITIES (UNKNOWN | INIT_FAILURE | WORK_FAILURE)
+
+#define PDEBUG(fac, fmt, args...) do { \
+ if (fac & PDEBUG_FACILITIES) \
+ snd_printd(KERN_DEBUG SND_ML403_AC97CR_DRIVER ": " \
+ fmt, ##args); \
+ } while (0)
+#else
+#define PDEBUG(fac, fmt, args...) /* nothing */
+#endif
+
+
+
+/* Defines for "waits"/timeouts (portions of HZ=250 on arch/ppc by default) */
+#define CODEC_TIMEOUT_ON_INIT 5 /* timeout for checking for codec
+ * readiness (after insmod)
+ */
+#ifndef CODEC_WRITE_CHECK_RAF
+#define CODEC_WAIT_AFTER_WRITE 100 /* general, static wait after a write
+ * access to a codec register, may be
+ * 0 to completely remove wait
+ */
+#else
+#define CODEC_TIMEOUT_AFTER_WRITE 5 /* timeout after a write access to a
+ * codec register, if RAF bit is used
+ */
+#endif
+#define CODEC_TIMEOUT_AFTER_READ 5 /* timeout after a read access to a
+ * codec register (checking RAF bit)
+ */
+
+/* Infrastructure for codec register shadowing */
+#define LM4550_REG_OK (1<<0) /* register exists */
+#define LM4550_REG_DONEREAD (1<<1) /* read register once, value should be
+ * the same currently in the register
+ */
+#define LM4550_REG_NOSAVE (1<<2) /* values written to this register will
+ * not be saved in the register
+ */
+#define LM4550_REG_NOSHADOW (1<<3) /* don't do register shadowing, use plain
+ * hardware access
+ */
+#define LM4550_REG_READONLY (1<<4) /* register is read only */
+#define LM4550_REG_FAKEPROBE (1<<5) /* fake write _and_ read actions during
+ * probe() correctly
+ */
+#define LM4550_REG_FAKEREAD (1<<6) /* fake read access, always return
+ * default value
+ */
+#define LM4550_REG_ALLFAKE (LM4550_REG_FAKEREAD | LM4550_REG_FAKEPROBE)
+
+struct lm4550_reg {
+ u16 value;
+ u16 flag;
+ u16 wmask;
+ u16 def;
+};
+
+struct lm4550_reg lm4550_regfile[64] = {
+ [AC97_RESET / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_NOSAVE \
+ | LM4550_REG_FAKEREAD,
+ .def = 0x0D50},
+ [AC97_MASTER / 2] = {.flag = LM4550_REG_OK
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x9F1F,
+ .def = 0x8000},
+ [AC97_HEADPHONE / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x9F1F,
+ .def = 0x8000},
+ [AC97_MASTER_MONO / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x801F,
+ .def = 0x8000},
+ [AC97_PC_BEEP / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x801E,
+ .def = 0x0},
+ [AC97_PHONE / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x801F,
+ .def = 0x8008},
+ [AC97_MIC / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x805F,
+ .def = 0x8008},
+ [AC97_LINE / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x9F1F,
+ .def = 0x8808},
+ [AC97_CD / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x9F1F,
+ .def = 0x8808},
+ [AC97_VIDEO / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x9F1F,
+ .def = 0x8808},
+ [AC97_AUX / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x9F1F,
+ .def = 0x8808},
+ [AC97_PCM / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x9F1F,
+ .def = 0x8008},
+ [AC97_REC_SEL / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x707,
+ .def = 0x0},
+ [AC97_REC_GAIN / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .wmask = 0x8F0F,
+ .def = 0x8000},
+ [AC97_GENERAL_PURPOSE / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .def = 0x0,
+ .wmask = 0xA380},
+ [AC97_3D_CONTROL / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEREAD \
+ | LM4550_REG_READONLY,
+ .def = 0x0101},
+ [AC97_POWERDOWN / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_NOSHADOW \
+ | LM4550_REG_NOSAVE,
+ .wmask = 0xFF00},
+ /* may not write ones to
+ * REF/ANL/DAC/ADC bits
+ * FIXME: Is this ok?
+ */
+ [AC97_EXTENDED_ID / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEREAD \
+ | LM4550_REG_READONLY,
+ .def = 0x0201}, /* primary codec */
+ [AC97_EXTENDED_STATUS / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_NOSHADOW \
+ | LM4550_REG_NOSAVE,
+ .wmask = 0x1},
+ [AC97_PCM_FRONT_DAC_RATE / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .def = 0xBB80,
+ .wmask = 0xFFFF},
+ [AC97_PCM_LR_ADC_RATE / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_FAKEPROBE,
+ .def = 0xBB80,
+ .wmask = 0xFFFF},
+ [AC97_VENDOR_ID1 / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_READONLY \
+ | LM4550_REG_FAKEREAD,
+ .def = 0x4E53},
+ [AC97_VENDOR_ID2 / 2] = {.flag = LM4550_REG_OK \
+ | LM4550_REG_READONLY \
+ | LM4550_REG_FAKEREAD,
+ .def = 0x4350}
+};
+
+#define LM4550_RF_OK(reg) (lm4550_regfile[reg / 2].flag & LM4550_REG_OK)
+
+static void lm4550_regfile_init(void)
+{
+ int i;
+ for (i = 0; i < 64; i++)
+ if (lm4550_regfile[i].flag & LM4550_REG_FAKEPROBE)
+ lm4550_regfile[i].value = lm4550_regfile[i].def;
+}
+
+static void lm4550_regfile_write_values_after_init(struct snd_ac97 *ac97)
+{
+ int i;
+ for (i = 0; i < 64; i++)
+ if ((lm4550_regfile[i].flag & LM4550_REG_FAKEPROBE) &&
+ (lm4550_regfile[i].value != lm4550_regfile[i].def)) {
+ PDEBUG(CODEC_FAKE, "lm4550_regfile_write_values_after_"
+ "init(): reg=0x%x value=0x%x / %d is different "
+ "from def=0x%x / %d\n",
+ i, lm4550_regfile[i].value,
+ lm4550_regfile[i].value, lm4550_regfile[i].def,
+ lm4550_regfile[i].def);
+ snd_ac97_write(ac97, i * 2, lm4550_regfile[i].value);
+ lm4550_regfile[i].flag |= LM4550_REG_DONEREAD;
+ }
+}
+
+
+/* direct registers */
+#define CR_REG(ml403_ac97cr, x) ((ml403_ac97cr)->port + CR_REG_##x)
+
+#define CR_REG_PLAYFIFO 0x00
+#define CR_PLAYDATA(a) ((a) & 0xFFFF)
+
+#define CR_REG_RECFIFO 0x04
+#define CR_RECDATA(a) ((a) & 0xFFFF)
+
+#define CR_REG_STATUS 0x08
+#define CR_RECOVER (1<<7)
+#define CR_PLAYUNDER (1<<6)
+#define CR_CODECREADY (1<<5)
+#define CR_RAF (1<<4)
+#define CR_RECEMPTY (1<<3)
+#define CR_RECFULL (1<<2)
+#define CR_PLAYHALF (1<<1)
+#define CR_PLAYFULL (1<<0)
+
+#define CR_REG_RESETFIFO 0x0C
+#define CR_RECRESET (1<<1)
+#define CR_PLAYRESET (1<<0)
+
+#define CR_REG_CODEC_ADDR 0x10
+/* UG082 says:
+ * #define CR_CODEC_ADDR(a) ((a) << 1)
+ * #define CR_CODEC_READ (1<<0)
+ * #define CR_CODEC_WRITE (0<<0)
+ */
+/* RefDesign example says: */
+#define CR_CODEC_ADDR(a) ((a) << 0)
+#define CR_CODEC_READ (1<<7)
+#define CR_CODEC_WRITE (0<<7)
+
+#define CR_REG_CODEC_DATAREAD 0x14
+#define CR_CODEC_DATAREAD(v) ((v) & 0xFFFF)
+
+#define CR_REG_CODEC_DATAWRITE 0x18
+#define CR_CODEC_DATAWRITE(v) ((v) & 0xFFFF)
+
+#define CR_FIFO_SIZE 32
+
+struct snd_ml403_ac97cr {
+ /* lock for access to (controller) registers */
+ spinlock_t reg_lock;
+ /* mutex for the whole sequence of accesses to (controller) registers
+ * which affect codec registers
+ */
+ struct mutex cdc_mutex;
+
+ int irq; /* for playback */
+ int enable_irq; /* for playback */
+
+ int capture_irq;
+ int enable_capture_irq;
+
+ struct resource *res_port;
+ void *port;
+
+ struct snd_ac97 *ac97;
+ int ac97_fake;
+#ifdef CODEC_STAT
+ int ac97_read;
+ int ac97_write;
+#endif
+
+ struct platform_device *pfdev;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ struct snd_pcm_indirect2 ind_rec; /* for playback */
+ struct snd_pcm_indirect2 capture_ind2_rec;
+};
+
+static struct snd_pcm_hardware snd_ml403_ac97cr_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_48000),
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = CR_FIFO_SIZE/2,
+ .period_bytes_max = (64*1024),
+ .periods_min = 2,
+ .periods_max = (128*1024)/(CR_FIFO_SIZE/2),
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_ml403_ac97cr_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_48000),
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = CR_FIFO_SIZE/2,
+ .period_bytes_max = (64*1024),
+ .periods_min = 2,
+ .periods_max = (128*1024)/(CR_FIFO_SIZE/2),
+ .fifo_size = 0,
+};
+
+static size_t
+snd_ml403_ac97cr_playback_ind2_zero(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ int copied_words = 0;
+ u32 full = 0;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+
+ spin_lock(&ml403_ac97cr->reg_lock);
+ while ((full = (in_be32(CR_REG(ml403_ac97cr, STATUS)) &
+ CR_PLAYFULL)) != CR_PLAYFULL) {
+ out_be32(CR_REG(ml403_ac97cr, PLAYFIFO), 0);
+ copied_words++;
+ }
+ rec->hw_ready = 0;
+ spin_unlock(&ml403_ac97cr->reg_lock);
+
+ return (size_t) (copied_words * 2);
+}
+
+static size_t
+snd_ml403_ac97cr_playback_ind2_copy(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ size_t bytes)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ u16 *src;
+ int copied_words = 0;
+ u32 full = 0;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+ src = (u16 *)(substream->runtime->dma_area + rec->sw_data);
+
+ spin_lock(&ml403_ac97cr->reg_lock);
+ while (((full = (in_be32(CR_REG(ml403_ac97cr, STATUS)) &
+ CR_PLAYFULL)) != CR_PLAYFULL) && (bytes > 1)) {
+ out_be32(CR_REG(ml403_ac97cr, PLAYFIFO),
+ CR_PLAYDATA(src[copied_words]));
+ copied_words++;
+ bytes = bytes - 2;
+ }
+ if (full != CR_PLAYFULL)
+ rec->hw_ready = 1;
+ else
+ rec->hw_ready = 0;
+ spin_unlock(&ml403_ac97cr->reg_lock);
+
+ return (size_t) (copied_words * 2);
+}
+
+static size_t
+snd_ml403_ac97cr_capture_ind2_null(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ int copied_words = 0;
+ u32 empty = 0;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+
+ spin_lock(&ml403_ac97cr->reg_lock);
+ while ((empty = (in_be32(CR_REG(ml403_ac97cr, STATUS)) &
+ CR_RECEMPTY)) != CR_RECEMPTY) {
+ volatile u32 trash;
+
+ trash = CR_RECDATA(in_be32(CR_REG(ml403_ac97cr, RECFIFO)));
+ /* Hmmmm, really necessary? Don't want call to in_be32()
+ * to be optimised away!
+ */
+ trash++;
+ copied_words++;
+ }
+ rec->hw_ready = 0;
+ spin_unlock(&ml403_ac97cr->reg_lock);
+
+ return (size_t) (copied_words * 2);
+}
+
+static size_t
+snd_ml403_ac97cr_capture_ind2_copy(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec, size_t bytes)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ u16 *dst;
+ int copied_words = 0;
+ u32 empty = 0;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+ dst = (u16 *)(substream->runtime->dma_area + rec->sw_data);
+
+ spin_lock(&ml403_ac97cr->reg_lock);
+ while (((empty = (in_be32(CR_REG(ml403_ac97cr, STATUS)) &
+ CR_RECEMPTY)) != CR_RECEMPTY) && (bytes > 1)) {
+ dst[copied_words] = CR_RECDATA(in_be32(CR_REG(ml403_ac97cr,
+ RECFIFO)));
+ copied_words++;
+ bytes = bytes - 2;
+ }
+ if (empty != CR_RECEMPTY)
+ rec->hw_ready = 1;
+ else
+ rec->hw_ready = 0;
+ spin_unlock(&ml403_ac97cr->reg_lock);
+
+ return (size_t) (copied_words * 2);
+}
+
+static snd_pcm_uframes_t
+snd_ml403_ac97cr_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ struct snd_pcm_indirect2 *ind2_rec = NULL;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+
+ if (substream == ml403_ac97cr->playback_substream)
+ ind2_rec = &ml403_ac97cr->ind_rec;
+ if (substream == ml403_ac97cr->capture_substream)
+ ind2_rec = &ml403_ac97cr->capture_ind2_rec;
+
+ if (ind2_rec != NULL)
+ return snd_pcm_indirect2_pointer(substream, ind2_rec);
+ return (snd_pcm_uframes_t) 0;
+}
+
+static int
+snd_ml403_ac97cr_pcm_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ int err = 0;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ PDEBUG(WORK_INFO, "trigger(playback): START\n");
+ ml403_ac97cr->ind_rec.hw_ready = 1;
+
+ /* clear play FIFO */
+ out_be32(CR_REG(ml403_ac97cr, RESETFIFO), CR_PLAYRESET);
+
+ /* enable play irq */
+ ml403_ac97cr->enable_irq = 1;
+ enable_irq(ml403_ac97cr->irq);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ PDEBUG(WORK_INFO, "trigger(playback): STOP\n");
+ ml403_ac97cr->ind_rec.hw_ready = 0;
+#ifdef SND_PCM_INDIRECT2_STAT
+ snd_pcm_indirect2_stat(substream, &ml403_ac97cr->ind_rec);
+#endif
+ /* disable play irq */
+ disable_irq_nosync(ml403_ac97cr->irq);
+ ml403_ac97cr->enable_irq = 0;
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ PDEBUG(WORK_INFO, "trigger(playback): (done)\n");
+ return err;
+}
+
+static int
+snd_ml403_ac97cr_pcm_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ int err = 0;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ PDEBUG(WORK_INFO, "trigger(capture): START\n");
+ ml403_ac97cr->capture_ind2_rec.hw_ready = 0;
+
+ /* clear record FIFO */
+ out_be32(CR_REG(ml403_ac97cr, RESETFIFO), CR_RECRESET);
+
+ /* enable record irq */
+ ml403_ac97cr->enable_capture_irq = 1;
+ enable_irq(ml403_ac97cr->capture_irq);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ PDEBUG(WORK_INFO, "trigger(capture): STOP\n");
+ ml403_ac97cr->capture_ind2_rec.hw_ready = 0;
+#ifdef SND_PCM_INDIRECT2_STAT
+ snd_pcm_indirect2_stat(substream,
+ &ml403_ac97cr->capture_ind2_rec);
+#endif
+ /* disable capture irq */
+ disable_irq_nosync(ml403_ac97cr->capture_irq);
+ ml403_ac97cr->enable_capture_irq = 0;
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ PDEBUG(WORK_INFO, "trigger(capture): (done)\n");
+ return err;
+}
+
+static int
+snd_ml403_ac97cr_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ struct snd_pcm_runtime *runtime;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+ runtime = substream->runtime;
+
+ PDEBUG(WORK_INFO,
+ "prepare(): period_bytes=%d, minperiod_bytes=%d\n",
+ snd_pcm_lib_period_bytes(substream), CR_FIFO_SIZE / 2);
+
+ /* set sampling rate */
+ snd_ac97_set_rate(ml403_ac97cr->ac97, AC97_PCM_FRONT_DAC_RATE,
+ runtime->rate);
+ PDEBUG(WORK_INFO, "prepare(): rate=%d\n", runtime->rate);
+
+ /* init struct for intermediate buffer */
+ memset(&ml403_ac97cr->ind_rec, 0,
+ sizeof(struct snd_pcm_indirect2));
+ ml403_ac97cr->ind_rec.hw_buffer_size = CR_FIFO_SIZE;
+ ml403_ac97cr->ind_rec.sw_buffer_size =
+ snd_pcm_lib_buffer_bytes(substream);
+ ml403_ac97cr->ind_rec.min_periods = -1;
+ ml403_ac97cr->ind_rec.min_multiple =
+ snd_pcm_lib_period_bytes(substream) / (CR_FIFO_SIZE / 2);
+ PDEBUG(WORK_INFO, "prepare(): hw_buffer_size=%d, "
+ "sw_buffer_size=%d, min_multiple=%d\n",
+ CR_FIFO_SIZE, ml403_ac97cr->ind_rec.sw_buffer_size,
+ ml403_ac97cr->ind_rec.min_multiple);
+ return 0;
+}
+
+static int
+snd_ml403_ac97cr_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ struct snd_pcm_runtime *runtime;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+ runtime = substream->runtime;
+
+ PDEBUG(WORK_INFO,
+ "prepare(capture): period_bytes=%d, minperiod_bytes=%d\n",
+ snd_pcm_lib_period_bytes(substream), CR_FIFO_SIZE / 2);
+
+ /* set sampling rate */
+ snd_ac97_set_rate(ml403_ac97cr->ac97, AC97_PCM_LR_ADC_RATE,
+ runtime->rate);
+ PDEBUG(WORK_INFO, "prepare(capture): rate=%d\n", runtime->rate);
+
+ /* init struct for intermediate buffer */
+ memset(&ml403_ac97cr->capture_ind2_rec, 0,
+ sizeof(struct snd_pcm_indirect2));
+ ml403_ac97cr->capture_ind2_rec.hw_buffer_size = CR_FIFO_SIZE;
+ ml403_ac97cr->capture_ind2_rec.sw_buffer_size =
+ snd_pcm_lib_buffer_bytes(substream);
+ ml403_ac97cr->capture_ind2_rec.min_multiple =
+ snd_pcm_lib_period_bytes(substream) / (CR_FIFO_SIZE / 2);
+ PDEBUG(WORK_INFO, "prepare(capture): hw_buffer_size=%d, "
+ "sw_buffer_size=%d, min_multiple=%d\n", CR_FIFO_SIZE,
+ ml403_ac97cr->capture_ind2_rec.sw_buffer_size,
+ ml403_ac97cr->capture_ind2_rec.min_multiple);
+ return 0;
+}
+
+static int snd_ml403_ac97cr_hw_free(struct snd_pcm_substream *substream)
+{
+ PDEBUG(WORK_INFO, "hw_free()\n");
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int
+snd_ml403_ac97cr_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ PDEBUG(WORK_INFO, "hw_params(): desired buffer bytes=%d, desired "
+ "period bytes=%d\n",
+ params_buffer_bytes(hw_params), params_period_bytes(hw_params));
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int snd_ml403_ac97cr_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ struct snd_pcm_runtime *runtime;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+ runtime = substream->runtime;
+
+ PDEBUG(WORK_INFO, "open(playback)\n");
+ ml403_ac97cr->playback_substream = substream;
+ runtime->hw = snd_ml403_ac97cr_playback;
+
+ snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ CR_FIFO_SIZE / 2);
+ return 0;
+}
+
+static int snd_ml403_ac97cr_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ struct snd_pcm_runtime *runtime;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+ runtime = substream->runtime;
+
+ PDEBUG(WORK_INFO, "open(capture)\n");
+ ml403_ac97cr->capture_substream = substream;
+ runtime->hw = snd_ml403_ac97cr_capture;
+
+ snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ CR_FIFO_SIZE / 2);
+ return 0;
+}
+
+static int snd_ml403_ac97cr_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+
+ PDEBUG(WORK_INFO, "close(playback)\n");
+ ml403_ac97cr->playback_substream = NULL;
+ return 0;
+}
+
+static int snd_ml403_ac97cr_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+
+ ml403_ac97cr = snd_pcm_substream_chip(substream);
+
+ PDEBUG(WORK_INFO, "close(capture)\n");
+ ml403_ac97cr->capture_substream = NULL;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ml403_ac97cr_playback_ops = {
+ .open = snd_ml403_ac97cr_playback_open,
+ .close = snd_ml403_ac97cr_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ml403_ac97cr_hw_params,
+ .hw_free = snd_ml403_ac97cr_hw_free,
+ .prepare = snd_ml403_ac97cr_pcm_playback_prepare,
+ .trigger = snd_ml403_ac97cr_pcm_playback_trigger,
+ .pointer = snd_ml403_ac97cr_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_ml403_ac97cr_capture_ops = {
+ .open = snd_ml403_ac97cr_capture_open,
+ .close = snd_ml403_ac97cr_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ml403_ac97cr_hw_params,
+ .hw_free = snd_ml403_ac97cr_hw_free,
+ .prepare = snd_ml403_ac97cr_pcm_capture_prepare,
+ .trigger = snd_ml403_ac97cr_pcm_capture_trigger,
+ .pointer = snd_ml403_ac97cr_pcm_pointer,
+};
+
+static irqreturn_t snd_ml403_ac97cr_irq(int irq, void *dev_id)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ struct platform_device *pfdev;
+ int cmp_irq;
+
+ ml403_ac97cr = (struct snd_ml403_ac97cr *)dev_id;
+ if (ml403_ac97cr == NULL)
+ return IRQ_NONE;
+
+ pfdev = ml403_ac97cr->pfdev;
+
+ /* playback interrupt */
+ cmp_irq = platform_get_irq(pfdev, 0);
+ if (irq == cmp_irq) {
+ if (ml403_ac97cr->enable_irq)
+ snd_pcm_indirect2_playback_interrupt(
+ ml403_ac97cr->playback_substream,
+ &ml403_ac97cr->ind_rec,
+ snd_ml403_ac97cr_playback_ind2_copy,
+ snd_ml403_ac97cr_playback_ind2_zero);
+ else
+ goto __disable_irq;
+ } else {
+ /* record interrupt */
+ cmp_irq = platform_get_irq(pfdev, 1);
+ if (irq == cmp_irq) {
+ if (ml403_ac97cr->enable_capture_irq)
+ snd_pcm_indirect2_capture_interrupt(
+ ml403_ac97cr->capture_substream,
+ &ml403_ac97cr->capture_ind2_rec,
+ snd_ml403_ac97cr_capture_ind2_copy,
+ snd_ml403_ac97cr_capture_ind2_null);
+ else
+ goto __disable_irq;
+ } else
+ return IRQ_NONE;
+ }
+ return IRQ_HANDLED;
+
+__disable_irq:
+ PDEBUG(INIT_INFO, "irq(): irq %d is meant to be disabled! So, now try "
+ "to disable it _really_!\n", irq);
+ disable_irq_nosync(irq);
+ return IRQ_HANDLED;
+}
+
+static unsigned short
+snd_ml403_ac97cr_codec_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data;
+#ifdef CODEC_STAT
+ u32 stat;
+ u32 rafaccess = 0;
+#endif
+ unsigned long end_time;
+ u16 value = 0;
+
+ if (!LM4550_RF_OK(reg)) {
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "access to unknown/unused codec register 0x%x "
+ "ignored!\n", reg);
+ return 0;
+ }
+ /* check if we can fake/answer this access from our shadow register */
+ if ((lm4550_regfile[reg / 2].flag &
+ (LM4550_REG_DONEREAD | LM4550_REG_ALLFAKE)) &&
+ !(lm4550_regfile[reg / 2].flag & LM4550_REG_NOSHADOW)) {
+ if (lm4550_regfile[reg / 2].flag & LM4550_REG_FAKEREAD) {
+ PDEBUG(CODEC_FAKE, "codec_read(): faking read from "
+ "reg=0x%x, val=0x%x / %d\n",
+ reg, lm4550_regfile[reg / 2].def,
+ lm4550_regfile[reg / 2].def);
+ return lm4550_regfile[reg / 2].def;
+ } else if ((lm4550_regfile[reg / 2].flag &
+ LM4550_REG_FAKEPROBE) &&
+ ml403_ac97cr->ac97_fake) {
+ PDEBUG(CODEC_FAKE, "codec_read(): faking read from "
+ "reg=0x%x, val=0x%x / %d (probe)\n",
+ reg, lm4550_regfile[reg / 2].value,
+ lm4550_regfile[reg / 2].value);
+ return lm4550_regfile[reg / 2].value;
+ } else {
+#ifdef CODEC_STAT
+ PDEBUG(CODEC_FAKE, "codec_read(): read access "
+ "answered by shadow register 0x%x (value=0x%x "
+ "/ %d) (cw=%d cr=%d)\n",
+ reg, lm4550_regfile[reg / 2].value,
+ lm4550_regfile[reg / 2].value,
+ ml403_ac97cr->ac97_write,
+ ml403_ac97cr->ac97_read);
+#else
+ PDEBUG(CODEC_FAKE, "codec_read(): read access "
+ "answered by shadow register 0x%x (value=0x%x "
+ "/ %d)\n",
+ reg, lm4550_regfile[reg / 2].value,
+ lm4550_regfile[reg / 2].value);
+#endif
+ return lm4550_regfile[reg / 2].value;
+ }
+ }
+ /* if we are here, we _have_ to access the codec really, no faking */
+ if (mutex_lock_interruptible(&ml403_ac97cr->cdc_mutex) != 0)
+ return 0;
+#ifdef CODEC_STAT
+ ml403_ac97cr->ac97_read++;
+#endif
+ spin_lock(&ml403_ac97cr->reg_lock);
+ out_be32(CR_REG(ml403_ac97cr, CODEC_ADDR),
+ CR_CODEC_ADDR(reg) | CR_CODEC_READ);
+ spin_unlock(&ml403_ac97cr->reg_lock);
+ end_time = jiffies + (HZ / CODEC_TIMEOUT_AFTER_READ);
+ do {
+ spin_lock(&ml403_ac97cr->reg_lock);
+#ifdef CODEC_STAT
+ rafaccess++;
+ stat = in_be32(CR_REG(ml403_ac97cr, STATUS));
+ if ((stat & CR_RAF) == CR_RAF) {
+ value = CR_CODEC_DATAREAD(
+ in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD)));
+ PDEBUG(CODEC_SUCCESS, "codec_read(): (done) reg=0x%x, "
+ "value=0x%x / %d (STATUS=0x%x)\n",
+ reg, value, value, stat);
+#else
+ if ((in_be32(CR_REG(ml403_ac97cr, STATUS)) &
+ CR_RAF) == CR_RAF) {
+ value = CR_CODEC_DATAREAD(
+ in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD)));
+ PDEBUG(CODEC_SUCCESS, "codec_read(): (done) "
+ "reg=0x%x, value=0x%x / %d\n",
+ reg, value, value);
+#endif
+ lm4550_regfile[reg / 2].value = value;
+ lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD;
+ spin_unlock(&ml403_ac97cr->reg_lock);
+ mutex_unlock(&ml403_ac97cr->cdc_mutex);
+ return value;
+ }
+ spin_unlock(&ml403_ac97cr->reg_lock);
+ schedule_timeout_uninterruptible(1);
+ } while (time_after(end_time, jiffies));
+ /* read the DATAREAD register anyway, see comment below */
+ spin_lock(&ml403_ac97cr->reg_lock);
+ value =
+ CR_CODEC_DATAREAD(in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD)));
+ spin_unlock(&ml403_ac97cr->reg_lock);
+#ifdef CODEC_STAT
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "timeout while codec read! "
+ "(reg=0x%x, last STATUS=0x%x, DATAREAD=0x%x / %d, %d) "
+ "(cw=%d, cr=%d)\n",
+ reg, stat, value, value, rafaccess,
+ ml403_ac97cr->ac97_write, ml403_ac97cr->ac97_read);
+#else
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "timeout while codec read! "
+ "(reg=0x%x, DATAREAD=0x%x / %d)\n",
+ reg, value, value);
+#endif
+ /* BUG: This is PURE speculation! But after _most_ read timeouts the
+ * value in the register is ok!
+ */
+ lm4550_regfile[reg / 2].value = value;
+ lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD;
+ mutex_unlock(&ml403_ac97cr->cdc_mutex);
+ return value;
+}
+
+static void
+snd_ml403_ac97cr_codec_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data;
+
+#ifdef CODEC_STAT
+ u32 stat;
+ u32 rafaccess = 0;
+#endif
+#ifdef CODEC_WRITE_CHECK_RAF
+ unsigned long end_time;
+#endif
+
+ if (!LM4550_RF_OK(reg)) {
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "access to unknown/unused codec register 0x%x "
+ "ignored!\n", reg);
+ return;
+ }
+ if (lm4550_regfile[reg / 2].flag & LM4550_REG_READONLY) {
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "write access to read only codec register 0x%x "
+ "ignored!\n", reg);
+ return;
+ }
+ if ((val & lm4550_regfile[reg / 2].wmask) != val) {
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "write access to codec register 0x%x "
+ "with bad value 0x%x / %d!\n",
+ reg, val, val);
+ val = val & lm4550_regfile[reg / 2].wmask;
+ }
+ if (((lm4550_regfile[reg / 2].flag & LM4550_REG_FAKEPROBE) &&
+ ml403_ac97cr->ac97_fake) &&
+ !(lm4550_regfile[reg / 2].flag & LM4550_REG_NOSHADOW)) {
+ PDEBUG(CODEC_FAKE, "codec_write(): faking write to reg=0x%x, "
+ "val=0x%x / %d\n", reg, val, val);
+ lm4550_regfile[reg / 2].value = (val &
+ lm4550_regfile[reg / 2].wmask);
+ return;
+ }
+ if (mutex_lock_interruptible(&ml403_ac97cr->cdc_mutex) != 0)
+ return;
+#ifdef CODEC_STAT
+ ml403_ac97cr->ac97_write++;
+#endif
+ spin_lock(&ml403_ac97cr->reg_lock);
+ out_be32(CR_REG(ml403_ac97cr, CODEC_DATAWRITE),
+ CR_CODEC_DATAWRITE(val));
+ out_be32(CR_REG(ml403_ac97cr, CODEC_ADDR),
+ CR_CODEC_ADDR(reg) | CR_CODEC_WRITE);
+ spin_unlock(&ml403_ac97cr->reg_lock);
+#ifdef CODEC_WRITE_CHECK_RAF
+ /* check CR_CODEC_RAF bit to see if write access to register is done;
+ * loop until bit is set or timeout happens
+ */
+ end_time = jiffies + HZ / CODEC_TIMEOUT_AFTER_WRITE;
+ do {
+ spin_lock(&ml403_ac97cr->reg_lock);
+#ifdef CODEC_STAT
+ rafaccess++;
+ stat = in_be32(CR_REG(ml403_ac97cr, STATUS))
+ if ((stat & CR_RAF) == CR_RAF) {
+#else
+ if ((in_be32(CR_REG(ml403_ac97cr, STATUS)) &
+ CR_RAF) == CR_RAF) {
+#endif
+ PDEBUG(CODEC_SUCCESS, "codec_write(): (done) "
+ "reg=0x%x, value=%d / 0x%x\n",
+ reg, val, val);
+ if (!(lm4550_regfile[reg / 2].flag &
+ LM4550_REG_NOSHADOW) &&
+ !(lm4550_regfile[reg / 2].flag &
+ LM4550_REG_NOSAVE))
+ lm4550_regfile[reg / 2].value = val;
+ lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD;
+ spin_unlock(&ml403_ac97cr->reg_lock);
+ mutex_unlock(&ml403_ac97cr->cdc_mutex);
+ return;
+ }
+ spin_unlock(&ml403_ac97cr->reg_lock);
+ schedule_timeout_uninterruptible(1);
+ } while (time_after(end_time, jiffies));
+#ifdef CODEC_STAT
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "timeout while codec write "
+ "(reg=0x%x, val=0x%x / %d, last STATUS=0x%x, %d) "
+ "(cw=%d, cr=%d)\n",
+ reg, val, val, stat, rafaccess, ml403_ac97cr->ac97_write,
+ ml403_ac97cr->ac97_read);
+#else
+ snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": "
+ "timeout while codec write (reg=0x%x, val=0x%x / %d)\n",
+ reg, val, val);
+#endif
+#else /* CODEC_WRITE_CHECK_RAF */
+#if CODEC_WAIT_AFTER_WRITE > 0
+ /* officially, in AC97 spec there is no possibility for a AC97
+ * controller to determine, if write access is done or not - so: How
+ * is Xilinx able to provide a RAF bit for write access?
+ * => very strange, thus just don't check RAF bit (compare with
+ * Xilinx's example app in EDK 8.1i) and wait
+ */
+ schedule_timeout_uninterruptible(HZ / CODEC_WAIT_AFTER_WRITE);
+#endif
+ PDEBUG(CODEC_SUCCESS, "codec_write(): (done) "
+ "reg=0x%x, value=%d / 0x%x (no RAF check)\n",
+ reg, val, val);
+#endif
+ mutex_unlock(&ml403_ac97cr->cdc_mutex);
+ return;
+}
+
+static int __devinit
+snd_ml403_ac97cr_chip_init(struct snd_ml403_ac97cr *ml403_ac97cr)
+{
+ unsigned long end_time;
+ PDEBUG(INIT_INFO, "chip_init():\n");
+ end_time = jiffies + HZ / CODEC_TIMEOUT_ON_INIT;
+ do {
+ if (in_be32(CR_REG(ml403_ac97cr, STATUS)) & CR_CODECREADY) {
+ /* clear both hardware FIFOs */
+ out_be32(CR_REG(ml403_ac97cr, RESETFIFO),
+ CR_RECRESET | CR_PLAYRESET);
+ PDEBUG(INIT_INFO, "chip_init(): (done)\n");
+ return 0;
+ }
+ schedule_timeout_uninterruptible(1);
+ } while (time_after(end_time, jiffies));
+ snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
+ "timeout while waiting for codec, "
+ "not ready!\n");
+ return -EBUSY;
+}
+
+static int snd_ml403_ac97cr_free(struct snd_ml403_ac97cr *ml403_ac97cr)
+{
+ PDEBUG(INIT_INFO, "free():\n");
+ /* irq release */
+ if (ml403_ac97cr->irq >= 0)
+ free_irq(ml403_ac97cr->irq, ml403_ac97cr);
+ if (ml403_ac97cr->capture_irq >= 0)
+ free_irq(ml403_ac97cr->capture_irq, ml403_ac97cr);
+ /* give back "port" */
+ if (ml403_ac97cr->port != NULL)
+ iounmap(ml403_ac97cr->port);
+ kfree(ml403_ac97cr);
+ PDEBUG(INIT_INFO, "free(): (done)\n");
+ return 0;
+}
+
+static int snd_ml403_ac97cr_dev_free(struct snd_device *snddev)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr = snddev->device_data;
+ PDEBUG(INIT_INFO, "dev_free():\n");
+ return snd_ml403_ac97cr_free(ml403_ac97cr);
+}
+
+static int __devinit
+snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
+ struct snd_ml403_ac97cr **rml403_ac97cr)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ml403_ac97cr_dev_free,
+ };
+ struct resource *resource;
+ int irq;
+
+ *rml403_ac97cr = NULL;
+ ml403_ac97cr = kzalloc(sizeof(*ml403_ac97cr), GFP_KERNEL);
+ if (ml403_ac97cr == NULL)
+ return -ENOMEM;
+ spin_lock_init(&ml403_ac97cr->reg_lock);
+ mutex_init(&ml403_ac97cr->cdc_mutex);
+ ml403_ac97cr->card = card;
+ ml403_ac97cr->pfdev = pfdev;
+ ml403_ac97cr->irq = -1;
+ ml403_ac97cr->enable_irq = 0;
+ ml403_ac97cr->capture_irq = -1;
+ ml403_ac97cr->enable_capture_irq = 0;
+ ml403_ac97cr->port = NULL;
+ ml403_ac97cr->res_port = NULL;
+
+ PDEBUG(INIT_INFO, "Trying to reserve resources now ...\n");
+ resource = platform_get_resource(pfdev, IORESOURCE_MEM, 0);
+ /* get "port" */
+ ml403_ac97cr->port = ioremap_nocache(resource->start,
+ (resource->end) -
+ (resource->start) + 1);
+ if (ml403_ac97cr->port == NULL) {
+ snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
+ "unable to remap memory region (%x to %x)\n",
+ resource->start, resource->end);
+ snd_ml403_ac97cr_free(ml403_ac97cr);
+ return -EBUSY;
+ }
+ snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": "
+ "remap controller memory region to "
+ "0x%x done\n", (unsigned int)ml403_ac97cr->port);
+ /* get irq */
+ irq = platform_get_irq(pfdev, 0);
+ if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
+ dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
+ snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
+ "unable to grab IRQ %d\n",
+ irq);
+ snd_ml403_ac97cr_free(ml403_ac97cr);
+ return -EBUSY;
+ }
+ ml403_ac97cr->irq = irq;
+ snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": "
+ "request (playback) irq %d done\n",
+ ml403_ac97cr->irq);
+ irq = platform_get_irq(pfdev, 1);
+ if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
+ dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
+ snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
+ "unable to grab IRQ %d\n",
+ irq);
+ snd_ml403_ac97cr_free(ml403_ac97cr);
+ return -EBUSY;
+ }
+ ml403_ac97cr->capture_irq = irq;
+ snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": "
+ "request (capture) irq %d done\n",
+ ml403_ac97cr->capture_irq);
+
+ err = snd_ml403_ac97cr_chip_init(ml403_ac97cr);
+ if (err < 0) {
+ snd_ml403_ac97cr_free(ml403_ac97cr);
+ return err;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ml403_ac97cr, &ops);
+ if (err < 0) {
+ PDEBUG(INIT_FAILURE, "probe(): snd_device_new() failed!\n");
+ snd_ml403_ac97cr_free(ml403_ac97cr);
+ return err;
+ }
+
+ *rml403_ac97cr = ml403_ac97cr;
+ return 0;
+}
+
+static void snd_ml403_ac97cr_mixer_free(struct snd_ac97 *ac97)
+{
+ struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data;
+ PDEBUG(INIT_INFO, "mixer_free():\n");
+ ml403_ac97cr->ac97 = NULL;
+ PDEBUG(INIT_INFO, "mixer_free(): (done)\n");
+}
+
+static int __devinit
+snd_ml403_ac97cr_mixer(struct snd_ml403_ac97cr *ml403_ac97cr)
+{
+ struct snd_ac97_bus *bus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_ml403_ac97cr_codec_write,
+ .read = snd_ml403_ac97cr_codec_read,
+ };
+ PDEBUG(INIT_INFO, "mixer():\n");
+ err = snd_ac97_bus(ml403_ac97cr->card, 0, &ops, NULL, &bus);
+ if (err < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ml403_ac97cr->ac97_fake = 1;
+ lm4550_regfile_init();
+#ifdef CODEC_STAT
+ ml403_ac97cr->ac97_read = 0;
+ ml403_ac97cr->ac97_write = 0;
+#endif
+ ac97.private_data = ml403_ac97cr;
+ ac97.private_free = snd_ml403_ac97cr_mixer_free;
+ ac97.scaps = AC97_SCAP_AUDIO | AC97_SCAP_SKIP_MODEM |
+ AC97_SCAP_NO_SPDIF;
+ err = snd_ac97_mixer(bus, &ac97, &ml403_ac97cr->ac97);
+ ml403_ac97cr->ac97_fake = 0;
+ lm4550_regfile_write_values_after_init(ml403_ac97cr->ac97);
+ PDEBUG(INIT_INFO, "mixer(): (done) snd_ac97_mixer()=%d\n", err);
+ return err;
+}
+
+static int __devinit
+snd_ml403_ac97cr_pcm(struct snd_ml403_ac97cr *ml403_ac97cr, int device,
+ struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ err = snd_pcm_new(ml403_ac97cr->card, "ML403AC97CR/1", device, 1, 1,
+ &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_ml403_ac97cr_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_ml403_ac97cr_capture_ops);
+ pcm->private_data = ml403_ac97cr;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "ML403AC97CR DAC/ADC");
+ ml403_ac97cr->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 64 * 1024,
+ 128 * 1024);
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static int __devinit snd_ml403_ac97cr_probe(struct platform_device *pfdev)
+{
+ struct snd_card *card;
+ struct snd_ml403_ac97cr *ml403_ac97cr = NULL;
+ int err;
+ int dev = pfdev->id;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev])
+ return -ENOENT;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ err = snd_ml403_ac97cr_create(card, pfdev, &ml403_ac97cr);
+ if (err < 0) {
+ PDEBUG(INIT_FAILURE, "probe(): create failed!\n");
+ snd_card_free(card);
+ return err;
+ }
+ PDEBUG(INIT_INFO, "probe(): create done\n");
+ card->private_data = ml403_ac97cr;
+ err = snd_ml403_ac97cr_mixer(ml403_ac97cr);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ PDEBUG(INIT_INFO, "probe(): mixer done\n");
+ err = snd_ml403_ac97cr_pcm(ml403_ac97cr, 0, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ PDEBUG(INIT_INFO, "probe(): PCM done\n");
+ strcpy(card->driver, SND_ML403_AC97CR_DRIVER);
+ strcpy(card->shortname, "ML403 AC97 Controller Reference");
+ sprintf(card->longname, "%s %s at 0x%lx, irq %i & %i, device %i",
+ card->shortname, card->driver,
+ (unsigned long)ml403_ac97cr->port, ml403_ac97cr->irq,
+ ml403_ac97cr->capture_irq, dev + 1);
+
+ snd_card_set_dev(card, &pfdev->dev);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ platform_set_drvdata(pfdev, card);
+ PDEBUG(INIT_INFO, "probe(): (done)\n");
+ return 0;
+}
+
+static int snd_ml403_ac97cr_remove(struct platform_device *pfdev)
+{
+ snd_card_free(platform_get_drvdata(pfdev));
+ platform_set_drvdata(pfdev, NULL);
+ return 0;
+}
+
+/* work with hotplug and coldplug */
+MODULE_ALIAS("platform:" SND_ML403_AC97CR_DRIVER);
+
+static struct platform_driver snd_ml403_ac97cr_driver = {
+ .probe = snd_ml403_ac97cr_probe,
+ .remove = snd_ml403_ac97cr_remove,
+ .driver = {
+ .name = SND_ML403_AC97CR_DRIVER,
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init alsa_card_ml403_ac97cr_init(void)
+{
+ return platform_driver_register(&snd_ml403_ac97cr_driver);
+}
+
+static void __exit alsa_card_ml403_ac97cr_exit(void)
+{
+ platform_driver_unregister(&snd_ml403_ac97cr_driver);
+}
+
+module_init(alsa_card_ml403_ac97cr_init)
+module_exit(alsa_card_ml403_ac97cr_exit)
diff --git a/sound/drivers/mpu401/Makefile b/sound/drivers/mpu401/Makefile
new file mode 100644
index 0000000..918f83f
--- /dev/null
+++ b/sound/drivers/mpu401/Makefile
@@ -0,0 +1,12 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-mpu401-objs := mpu401.o
+snd-mpu401-uart-objs := mpu401_uart.o
+
+obj-$(CONFIG_SND_MPU401_UART) += snd-mpu401-uart.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_MPU401) += snd-mpu401.o
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
new file mode 100644
index 0000000..5b996f3
--- /dev/null
+++ b/sound/drivers/mpu401/mpu401.c
@@ -0,0 +1,289 @@
+/*
+ * Driver for generic MPU-401 boards (UART mode only)
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Copyright (c) 2004 by Castet Matthieu <castet.matthieu@free.fr>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/pnp.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/mpu401.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("MPU-401 UART");
+MODULE_LICENSE("GPL");
+
+static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* exclude the first card */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+#ifdef CONFIG_PNP
+static int pnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* MPU-401 port number */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* MPU-401 IRQ */
+static int uart_enter[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for MPU-401 device.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for MPU-401 device.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable MPU-401 device.");
+#ifdef CONFIG_PNP
+module_param_array(pnp, bool, NULL, 0444);
+MODULE_PARM_DESC(pnp, "PnP detection for MPU-401 device.");
+#endif
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for MPU-401 device.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for MPU-401 device.");
+module_param_array(uart_enter, bool, NULL, 0444);
+MODULE_PARM_DESC(uart_enter, "Issue UART_ENTER command at open.");
+
+static struct platform_device *platform_devices[SNDRV_CARDS];
+static int pnp_registered;
+static unsigned int snd_mpu401_devices;
+
+static int snd_mpu401_create(int dev, struct snd_card **rcard)
+{
+ struct snd_card *card;
+ int err;
+
+ if (!uart_enter[dev])
+ snd_printk(KERN_ERR "the uart_enter option is obsolete; remove it\n");
+
+ *rcard = NULL;
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ strcpy(card->driver, "MPU-401 UART");
+ strcpy(card->shortname, card->driver);
+ sprintf(card->longname, "%s at %#lx, ", card->shortname, port[dev]);
+ if (irq[dev] >= 0) {
+ sprintf(card->longname + strlen(card->longname), "irq %d", irq[dev]);
+ } else {
+ strcat(card->longname, "polled");
+ }
+
+ err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0,
+ irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0,
+ NULL);
+ if (err < 0) {
+ printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]);
+ goto _err;
+ }
+
+ *rcard = card;
+ return 0;
+
+ _err:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devinit snd_mpu401_probe(struct platform_device *devptr)
+{
+ int dev = devptr->id;
+ int err;
+ struct snd_card *card;
+
+ if (port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR "specify port\n");
+ return -EINVAL;
+ }
+ if (irq[dev] == SNDRV_AUTO_IRQ) {
+ snd_printk(KERN_ERR "specify or disable IRQ\n");
+ return -EINVAL;
+ }
+ err = snd_mpu401_create(dev, &card);
+ if (err < 0)
+ return err;
+ snd_card_set_dev(card, &devptr->dev);
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ platform_set_drvdata(devptr, card);
+ return 0;
+}
+
+static int __devexit snd_mpu401_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define SND_MPU401_DRIVER "snd_mpu401"
+
+static struct platform_driver snd_mpu401_driver = {
+ .probe = snd_mpu401_probe,
+ .remove = __devexit_p(snd_mpu401_remove),
+ .driver = {
+ .name = SND_MPU401_DRIVER
+ },
+};
+
+
+#ifdef CONFIG_PNP
+
+#define IO_EXTENT 2
+
+static struct pnp_device_id snd_mpu401_pnpids[] = {
+ { .id = "PNPb006" },
+ { .id = "" }
+};
+
+MODULE_DEVICE_TABLE(pnp, snd_mpu401_pnpids);
+
+static int __devinit snd_mpu401_pnp(int dev, struct pnp_dev *device,
+ const struct pnp_device_id *id)
+{
+ if (!pnp_port_valid(device, 0) ||
+ pnp_port_flags(device, 0) & IORESOURCE_DISABLED) {
+ snd_printk(KERN_ERR "no PnP port\n");
+ return -ENODEV;
+ }
+ if (pnp_port_len(device, 0) < IO_EXTENT) {
+ snd_printk(KERN_ERR "PnP port length is %llu, expected %d\n",
+ (unsigned long long)pnp_port_len(device, 0),
+ IO_EXTENT);
+ return -ENODEV;
+ }
+ port[dev] = pnp_port_start(device, 0);
+
+ if (!pnp_irq_valid(device, 0) ||
+ pnp_irq_flags(device, 0) & IORESOURCE_DISABLED) {
+ snd_printk(KERN_WARNING "no PnP irq, using polling\n");
+ irq[dev] = -1;
+ } else {
+ irq[dev] = pnp_irq(device, 0);
+ }
+ return 0;
+}
+
+static int __devinit snd_mpu401_pnp_probe(struct pnp_dev *pnp_dev,
+ const struct pnp_device_id *id)
+{
+ static int dev;
+ struct snd_card *card;
+ int err;
+
+ for ( ; dev < SNDRV_CARDS; ++dev) {
+ if (!enable[dev] || !pnp[dev])
+ continue;
+ err = snd_mpu401_pnp(dev, pnp_dev, id);
+ if (err < 0)
+ return err;
+ err = snd_mpu401_create(dev, &card);
+ if (err < 0)
+ return err;
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_card_set_dev(card, &pnp_dev->dev);
+ pnp_set_drvdata(pnp_dev, card);
+ snd_mpu401_devices++;
+ ++dev;
+ return 0;
+ }
+ return -ENODEV;
+}
+
+static void __devexit snd_mpu401_pnp_remove(struct pnp_dev *dev)
+{
+ struct snd_card *card = (struct snd_card *) pnp_get_drvdata(dev);
+
+ snd_card_disconnect(card);
+ snd_card_free_when_closed(card);
+}
+
+static struct pnp_driver snd_mpu401_pnp_driver = {
+ .name = "mpu401",
+ .id_table = snd_mpu401_pnpids,
+ .probe = snd_mpu401_pnp_probe,
+ .remove = __devexit_p(snd_mpu401_pnp_remove),
+};
+#else
+static struct pnp_driver snd_mpu401_pnp_driver;
+#endif
+
+static void snd_mpu401_unregister_all(void)
+{
+ int i;
+
+ if (pnp_registered)
+ pnp_unregister_driver(&snd_mpu401_pnp_driver);
+ for (i = 0; i < ARRAY_SIZE(platform_devices); ++i)
+ platform_device_unregister(platform_devices[i]);
+ platform_driver_unregister(&snd_mpu401_driver);
+}
+
+static int __init alsa_card_mpu401_init(void)
+{
+ int i, err;
+
+ if ((err = platform_driver_register(&snd_mpu401_driver)) < 0)
+ return err;
+
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ struct platform_device *device;
+ if (! enable[i])
+ continue;
+#ifdef CONFIG_PNP
+ if (pnp[i])
+ continue;
+#endif
+ device = platform_device_register_simple(SND_MPU401_DRIVER,
+ i, NULL, 0);
+ if (IS_ERR(device))
+ continue;
+ if (!platform_get_drvdata(device)) {
+ platform_device_unregister(device);
+ continue;
+ }
+ platform_devices[i] = device;
+ snd_mpu401_devices++;
+ }
+ err = pnp_register_driver(&snd_mpu401_pnp_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ if (!snd_mpu401_devices) {
+#ifdef MODULE
+ printk(KERN_ERR "MPU-401 device not found or device busy\n");
+#endif
+ snd_mpu401_unregister_all();
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_mpu401_exit(void)
+{
+ snd_mpu401_unregister_all();
+}
+
+module_init(alsa_card_mpu401_init)
+module_exit(alsa_card_mpu401_exit)
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
new file mode 100644
index 0000000..2af0999
--- /dev/null
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -0,0 +1,631 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of MPU-401 in UART mode
+ *
+ * MPU-401 supports UART mode which is not capable generate transmit
+ * interrupts thus output is done via polling. Also, if irq < 0, then
+ * input is done also via polling. Do not expect good performance.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * 13-03-2003:
+ * Added support for different kind of hardware I/O. Build in choices
+ * are port and mmio. For other kind of I/O, set mpu->read and
+ * mpu->write to your own I/O functions.
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/interrupt.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/mpu401.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Routines for control of MPU-401 in UART mode");
+MODULE_LICENSE("GPL");
+
+static void snd_mpu401_uart_input_read(struct snd_mpu401 * mpu);
+static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu);
+
+/*
+
+ */
+
+#define snd_mpu401_input_avail(mpu) \
+ (!(mpu->read(mpu, MPU401C(mpu)) & MPU401_RX_EMPTY))
+#define snd_mpu401_output_ready(mpu) \
+ (!(mpu->read(mpu, MPU401C(mpu)) & MPU401_TX_FULL))
+
+/* Build in lowlevel io */
+static void mpu401_write_port(struct snd_mpu401 *mpu, unsigned char data,
+ unsigned long addr)
+{
+ outb(data, addr);
+}
+
+static unsigned char mpu401_read_port(struct snd_mpu401 *mpu,
+ unsigned long addr)
+{
+ return inb(addr);
+}
+
+static void mpu401_write_mmio(struct snd_mpu401 *mpu, unsigned char data,
+ unsigned long addr)
+{
+ writeb(data, (void __iomem *)addr);
+}
+
+static unsigned char mpu401_read_mmio(struct snd_mpu401 *mpu,
+ unsigned long addr)
+{
+ return readb((void __iomem *)addr);
+}
+/* */
+
+static void snd_mpu401_uart_clear_rx(struct snd_mpu401 *mpu)
+{
+ int timeout = 100000;
+ for (; timeout > 0 && snd_mpu401_input_avail(mpu); timeout--)
+ mpu->read(mpu, MPU401D(mpu));
+#ifdef CONFIG_SND_DEBUG
+ if (timeout <= 0)
+ snd_printk(KERN_ERR "cmd: clear rx timeout (status = 0x%x)\n",
+ mpu->read(mpu, MPU401C(mpu)));
+#endif
+}
+
+static void uart_interrupt_tx(struct snd_mpu401 *mpu)
+{
+ unsigned long flags;
+
+ if (test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode) &&
+ test_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode)) {
+ spin_lock_irqsave(&mpu->output_lock, flags);
+ snd_mpu401_uart_output_write(mpu);
+ spin_unlock_irqrestore(&mpu->output_lock, flags);
+ }
+}
+
+static void _snd_mpu401_uart_interrupt(struct snd_mpu401 *mpu)
+{
+ unsigned long flags;
+
+ if (mpu->info_flags & MPU401_INFO_INPUT) {
+ spin_lock_irqsave(&mpu->input_lock, flags);
+ if (test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode))
+ snd_mpu401_uart_input_read(mpu);
+ else
+ snd_mpu401_uart_clear_rx(mpu);
+ spin_unlock_irqrestore(&mpu->input_lock, flags);
+ }
+ if (! (mpu->info_flags & MPU401_INFO_TX_IRQ))
+ /* ok. for better Tx performance try do some output
+ when input is done */
+ uart_interrupt_tx(mpu);
+}
+
+/**
+ * snd_mpu401_uart_interrupt - generic MPU401-UART interrupt handler
+ * @irq: the irq number
+ * @dev_id: mpu401 instance
+ *
+ * Processes the interrupt for MPU401-UART i/o.
+ */
+irqreturn_t snd_mpu401_uart_interrupt(int irq, void *dev_id)
+{
+ struct snd_mpu401 *mpu = dev_id;
+
+ if (mpu == NULL)
+ return IRQ_NONE;
+ _snd_mpu401_uart_interrupt(mpu);
+ return IRQ_HANDLED;
+}
+
+EXPORT_SYMBOL(snd_mpu401_uart_interrupt);
+
+/**
+ * snd_mpu401_uart_interrupt_tx - generic MPU401-UART transmit irq handler
+ * @irq: the irq number
+ * @dev_id: mpu401 instance
+ *
+ * Processes the interrupt for MPU401-UART output.
+ */
+irqreturn_t snd_mpu401_uart_interrupt_tx(int irq, void *dev_id)
+{
+ struct snd_mpu401 *mpu = dev_id;
+
+ if (mpu == NULL)
+ return IRQ_NONE;
+ uart_interrupt_tx(mpu);
+ return IRQ_HANDLED;
+}
+
+EXPORT_SYMBOL(snd_mpu401_uart_interrupt_tx);
+
+/*
+ * timer callback
+ * reprogram the timer and call the interrupt job
+ */
+static void snd_mpu401_uart_timer(unsigned long data)
+{
+ struct snd_mpu401 *mpu = (struct snd_mpu401 *)data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&mpu->timer_lock, flags);
+ /*mpu->mode |= MPU401_MODE_TIMER;*/
+ mpu->timer.expires = 1 + jiffies;
+ add_timer(&mpu->timer);
+ spin_unlock_irqrestore(&mpu->timer_lock, flags);
+ if (mpu->rmidi)
+ _snd_mpu401_uart_interrupt(mpu);
+}
+
+/*
+ * initialize the timer callback if not programmed yet
+ */
+static void snd_mpu401_uart_add_timer (struct snd_mpu401 *mpu, int input)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave (&mpu->timer_lock, flags);
+ if (mpu->timer_invoked == 0) {
+ init_timer(&mpu->timer);
+ mpu->timer.data = (unsigned long)mpu;
+ mpu->timer.function = snd_mpu401_uart_timer;
+ mpu->timer.expires = 1 + jiffies;
+ add_timer(&mpu->timer);
+ }
+ mpu->timer_invoked |= input ? MPU401_MODE_INPUT_TIMER :
+ MPU401_MODE_OUTPUT_TIMER;
+ spin_unlock_irqrestore (&mpu->timer_lock, flags);
+}
+
+/*
+ * remove the timer callback if still active
+ */
+static void snd_mpu401_uart_remove_timer (struct snd_mpu401 *mpu, int input)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave (&mpu->timer_lock, flags);
+ if (mpu->timer_invoked) {
+ mpu->timer_invoked &= input ? ~MPU401_MODE_INPUT_TIMER :
+ ~MPU401_MODE_OUTPUT_TIMER;
+ if (! mpu->timer_invoked)
+ del_timer(&mpu->timer);
+ }
+ spin_unlock_irqrestore (&mpu->timer_lock, flags);
+}
+
+/*
+ * send a UART command
+ * return zero if successful, non-zero for some errors
+ */
+
+static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd,
+ int ack)
+{
+ unsigned long flags;
+ int timeout, ok;
+
+ spin_lock_irqsave(&mpu->input_lock, flags);
+ if (mpu->hardware != MPU401_HW_TRID4DWAVE) {
+ mpu->write(mpu, 0x00, MPU401D(mpu));
+ /*snd_mpu401_uart_clear_rx(mpu);*/
+ }
+ /* ok. standard MPU-401 initialization */
+ if (mpu->hardware != MPU401_HW_SB) {
+ for (timeout = 1000; timeout > 0 &&
+ !snd_mpu401_output_ready(mpu); timeout--)
+ udelay(10);
+#ifdef CONFIG_SND_DEBUG
+ if (!timeout)
+ snd_printk(KERN_ERR "cmd: tx timeout (status = 0x%x)\n",
+ mpu->read(mpu, MPU401C(mpu)));
+#endif
+ }
+ mpu->write(mpu, cmd, MPU401C(mpu));
+ if (ack && !(mpu->info_flags & MPU401_INFO_NO_ACK)) {
+ ok = 0;
+ timeout = 10000;
+ while (!ok && timeout-- > 0) {
+ if (snd_mpu401_input_avail(mpu)) {
+ if (mpu->read(mpu, MPU401D(mpu)) == MPU401_ACK)
+ ok = 1;
+ }
+ }
+ if (!ok && mpu->read(mpu, MPU401D(mpu)) == MPU401_ACK)
+ ok = 1;
+ } else
+ ok = 1;
+ spin_unlock_irqrestore(&mpu->input_lock, flags);
+ if (!ok) {
+ snd_printk(KERN_ERR "cmd: 0x%x failed at 0x%lx "
+ "(status = 0x%x, data = 0x%x)\n", cmd, mpu->port,
+ mpu->read(mpu, MPU401C(mpu)),
+ mpu->read(mpu, MPU401D(mpu)));
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_mpu401_do_reset(struct snd_mpu401 *mpu)
+{
+ if (snd_mpu401_uart_cmd(mpu, MPU401_RESET, 1))
+ return -EIO;
+ if (snd_mpu401_uart_cmd(mpu, MPU401_ENTER_UART, 0))
+ return -EIO;
+ return 0;
+}
+
+/*
+ * input/output open/close - protected by open_mutex in rawmidi.c
+ */
+static int snd_mpu401_uart_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_mpu401 *mpu;
+ int err;
+
+ mpu = substream->rmidi->private_data;
+ if (mpu->open_input && (err = mpu->open_input(mpu)) < 0)
+ return err;
+ if (! test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode)) {
+ if (snd_mpu401_do_reset(mpu) < 0)
+ goto error_out;
+ }
+ mpu->substream_input = substream;
+ set_bit(MPU401_MODE_BIT_INPUT, &mpu->mode);
+ return 0;
+
+error_out:
+ if (mpu->open_input && mpu->close_input)
+ mpu->close_input(mpu);
+ return -EIO;
+}
+
+static int snd_mpu401_uart_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_mpu401 *mpu;
+ int err;
+
+ mpu = substream->rmidi->private_data;
+ if (mpu->open_output && (err = mpu->open_output(mpu)) < 0)
+ return err;
+ if (! test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode)) {
+ if (snd_mpu401_do_reset(mpu) < 0)
+ goto error_out;
+ }
+ mpu->substream_output = substream;
+ set_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode);
+ return 0;
+
+error_out:
+ if (mpu->open_output && mpu->close_output)
+ mpu->close_output(mpu);
+ return -EIO;
+}
+
+static int snd_mpu401_uart_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_mpu401 *mpu;
+ int err = 0;
+
+ mpu = substream->rmidi->private_data;
+ clear_bit(MPU401_MODE_BIT_INPUT, &mpu->mode);
+ mpu->substream_input = NULL;
+ if (! test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode))
+ err = snd_mpu401_uart_cmd(mpu, MPU401_RESET, 0);
+ if (mpu->close_input)
+ mpu->close_input(mpu);
+ if (err)
+ return -EIO;
+ return 0;
+}
+
+static int snd_mpu401_uart_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_mpu401 *mpu;
+ int err = 0;
+
+ mpu = substream->rmidi->private_data;
+ clear_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode);
+ mpu->substream_output = NULL;
+ if (! test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode))
+ err = snd_mpu401_uart_cmd(mpu, MPU401_RESET, 0);
+ if (mpu->close_output)
+ mpu->close_output(mpu);
+ if (err)
+ return -EIO;
+ return 0;
+}
+
+/*
+ * trigger input callback
+ */
+static void
+snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct snd_mpu401 *mpu;
+ int max = 64;
+
+ mpu = substream->rmidi->private_data;
+ if (up) {
+ if (! test_and_set_bit(MPU401_MODE_BIT_INPUT_TRIGGER,
+ &mpu->mode)) {
+ /* first time - flush FIFO */
+ while (max-- > 0)
+ mpu->read(mpu, MPU401D(mpu));
+ if (mpu->irq < 0)
+ snd_mpu401_uart_add_timer(mpu, 1);
+ }
+
+ /* read data in advance */
+ spin_lock_irqsave(&mpu->input_lock, flags);
+ snd_mpu401_uart_input_read(mpu);
+ spin_unlock_irqrestore(&mpu->input_lock, flags);
+ } else {
+ if (mpu->irq < 0)
+ snd_mpu401_uart_remove_timer(mpu, 1);
+ clear_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode);
+ }
+
+}
+
+/*
+ * transfer input pending data
+ * call with input_lock spinlock held
+ */
+static void snd_mpu401_uart_input_read(struct snd_mpu401 * mpu)
+{
+ int max = 128;
+ unsigned char byte;
+
+ while (max-- > 0) {
+ if (! snd_mpu401_input_avail(mpu))
+ break; /* input not available */
+ byte = mpu->read(mpu, MPU401D(mpu));
+ if (test_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode))
+ snd_rawmidi_receive(mpu->substream_input, &byte, 1);
+ }
+}
+
+/*
+ * Tx FIFO sizes:
+ * CS4237B - 16 bytes
+ * AudioDrive ES1688 - 12 bytes
+ * S3 SonicVibes - 8 bytes
+ * SoundBlaster AWE 64 - 2 bytes (ugly hardware)
+ */
+
+/*
+ * write output pending bytes
+ * call with output_lock spinlock held
+ */
+static void snd_mpu401_uart_output_write(struct snd_mpu401 * mpu)
+{
+ unsigned char byte;
+ int max = 256;
+
+ do {
+ if (snd_rawmidi_transmit_peek(mpu->substream_output,
+ &byte, 1) == 1) {
+ /*
+ * Try twice because there is hardware that insists on
+ * setting the output busy bit after each write.
+ */
+ if (!snd_mpu401_output_ready(mpu) &&
+ !snd_mpu401_output_ready(mpu))
+ break; /* Tx FIFO full - try again later */
+ mpu->write(mpu, byte, MPU401D(mpu));
+ snd_rawmidi_transmit_ack(mpu->substream_output, 1);
+ } else {
+ snd_mpu401_uart_remove_timer (mpu, 0);
+ break; /* no other data - leave the tx loop */
+ }
+ } while (--max > 0);
+}
+
+/*
+ * output trigger callback
+ */
+static void
+snd_mpu401_uart_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct snd_mpu401 *mpu;
+
+ mpu = substream->rmidi->private_data;
+ if (up) {
+ set_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode);
+
+ /* try to add the timer at each output trigger,
+ * since the output timer might have been removed in
+ * snd_mpu401_uart_output_write().
+ */
+ if (! (mpu->info_flags & MPU401_INFO_TX_IRQ))
+ snd_mpu401_uart_add_timer(mpu, 0);
+
+ /* output pending data */
+ spin_lock_irqsave(&mpu->output_lock, flags);
+ snd_mpu401_uart_output_write(mpu);
+ spin_unlock_irqrestore(&mpu->output_lock, flags);
+ } else {
+ if (! (mpu->info_flags & MPU401_INFO_TX_IRQ))
+ snd_mpu401_uart_remove_timer(mpu, 0);
+ clear_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode);
+ }
+}
+
+/*
+
+ */
+
+static struct snd_rawmidi_ops snd_mpu401_uart_output =
+{
+ .open = snd_mpu401_uart_output_open,
+ .close = snd_mpu401_uart_output_close,
+ .trigger = snd_mpu401_uart_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_mpu401_uart_input =
+{
+ .open = snd_mpu401_uart_input_open,
+ .close = snd_mpu401_uart_input_close,
+ .trigger = snd_mpu401_uart_input_trigger,
+};
+
+static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi)
+{
+ struct snd_mpu401 *mpu = rmidi->private_data;
+ if (mpu->irq_flags && mpu->irq >= 0)
+ free_irq(mpu->irq, (void *) mpu);
+ release_and_free_resource(mpu->res);
+ kfree(mpu);
+}
+
+/**
+ * snd_mpu401_uart_new - create an MPU401-UART instance
+ * @card: the card instance
+ * @device: the device index, zero-based
+ * @hardware: the hardware type, MPU401_HW_XXXX
+ * @port: the base address of MPU401 port
+ * @info_flags: bitflags MPU401_INFO_XXX
+ * @irq: the irq number, -1 if no interrupt for mpu
+ * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved.
+ * @rrawmidi: the pointer to store the new rawmidi instance
+ *
+ * Creates a new MPU-401 instance.
+ *
+ * Note that the rawmidi instance is returned on the rrawmidi argument,
+ * not the mpu401 instance itself. To access to the mpu401 instance,
+ * cast from rawmidi->private_data (with struct snd_mpu401 magic-cast).
+ *
+ * Returns zero if successful, or a negative error code.
+ */
+int snd_mpu401_uart_new(struct snd_card *card, int device,
+ unsigned short hardware,
+ unsigned long port,
+ unsigned int info_flags,
+ int irq, int irq_flags,
+ struct snd_rawmidi ** rrawmidi)
+{
+ struct snd_mpu401 *mpu;
+ struct snd_rawmidi *rmidi;
+ int in_enable, out_enable;
+ int err;
+
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ if (! (info_flags & (MPU401_INFO_INPUT | MPU401_INFO_OUTPUT)))
+ info_flags |= MPU401_INFO_INPUT | MPU401_INFO_OUTPUT;
+ in_enable = (info_flags & MPU401_INFO_INPUT) ? 1 : 0;
+ out_enable = (info_flags & MPU401_INFO_OUTPUT) ? 1 : 0;
+ if ((err = snd_rawmidi_new(card, "MPU-401U", device,
+ out_enable, in_enable, &rmidi)) < 0)
+ return err;
+ mpu = kzalloc(sizeof(*mpu), GFP_KERNEL);
+ if (mpu == NULL) {
+ snd_printk(KERN_ERR "mpu401_uart: cannot allocate\n");
+ snd_device_free(card, rmidi);
+ return -ENOMEM;
+ }
+ rmidi->private_data = mpu;
+ rmidi->private_free = snd_mpu401_uart_free;
+ spin_lock_init(&mpu->input_lock);
+ spin_lock_init(&mpu->output_lock);
+ spin_lock_init(&mpu->timer_lock);
+ mpu->hardware = hardware;
+ if (! (info_flags & MPU401_INFO_INTEGRATED)) {
+ int res_size = hardware == MPU401_HW_PC98II ? 4 : 2;
+ mpu->res = request_region(port, res_size, "MPU401 UART");
+ if (mpu->res == NULL) {
+ snd_printk(KERN_ERR "mpu401_uart: "
+ "unable to grab port 0x%lx size %d\n",
+ port, res_size);
+ snd_device_free(card, rmidi);
+ return -EBUSY;
+ }
+ }
+ if (info_flags & MPU401_INFO_MMIO) {
+ mpu->write = mpu401_write_mmio;
+ mpu->read = mpu401_read_mmio;
+ } else {
+ mpu->write = mpu401_write_port;
+ mpu->read = mpu401_read_port;
+ }
+ mpu->port = port;
+ if (hardware == MPU401_HW_PC98II)
+ mpu->cport = port + 2;
+ else
+ mpu->cport = port + 1;
+ if (irq >= 0 && irq_flags) {
+ if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags,
+ "MPU401 UART", (void *) mpu)) {
+ snd_printk(KERN_ERR "mpu401_uart: "
+ "unable to grab IRQ %d\n", irq);
+ snd_device_free(card, rmidi);
+ return -EBUSY;
+ }
+ }
+ mpu->info_flags = info_flags;
+ mpu->irq = irq;
+ mpu->irq_flags = irq_flags;
+ if (card->shortname[0])
+ snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI",
+ card->shortname);
+ else
+ sprintf(rmidi->name, "MPU-401 MIDI %d-%d",card->number, device);
+ if (out_enable) {
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_mpu401_uart_output);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ }
+ if (in_enable) {
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_mpu401_uart_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ if (out_enable)
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+ }
+ mpu->rmidi = rmidi;
+ if (rrawmidi)
+ *rrawmidi = rmidi;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_mpu401_uart_new);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_mpu401_uart_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_mpu401_uart_exit(void)
+{
+}
+
+module_init(alsa_mpu401_uart_init)
+module_exit(alsa_mpu401_uart_exit)
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
new file mode 100644
index 0000000..48b64e6
--- /dev/null
+++ b/sound/drivers/mtpav.c
@@ -0,0 +1,791 @@
+/*
+ * MOTU Midi Timepiece ALSA Main routines
+ * Copyright by Michael T. Mayers (c) Jan 09, 2000
+ * mail: michael@tweakoz.com
+ * Thanks to John Galbraith
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ * This driver is for the 'Mark Of The Unicorn' (MOTU)
+ * MidiTimePiece AV multiport MIDI interface
+ *
+ * IOPORTS
+ * -------
+ * 8 MIDI Ins and 8 MIDI outs
+ * Video Sync In (BNC), Word Sync Out (BNC),
+ * ADAT Sync Out (DB9)
+ * SMPTE in/out (1/4")
+ * 2 programmable pedal/footswitch inputs and 4 programmable MIDI controller knobs.
+ * Macintosh RS422 serial port
+ * RS422 "network" port for ganging multiple MTP's
+ * PC Parallel Port ( which this driver currently uses )
+ *
+ * MISC FEATURES
+ * -------------
+ * Hardware MIDI routing, merging, and filtering
+ * MIDI Synchronization to Video, ADAT, SMPTE and other Clock sources
+ * 128 'scene' memories, recallable from MIDI program change
+ *
+ *
+ * ChangeLog
+ * Jun 11 2001 Takashi Iwai <tiwai@suse.de>
+ * - Recoded & debugged
+ * - Added timer interrupt for midi outputs
+ * - hwports is between 1 and 8, which specifies the number of hardware ports.
+ * The three global ports, computer, adat and broadcast ports, are created
+ * always after h/w and remote ports.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <linux/delay.h>
+
+#include <asm/io.h>
+
+/*
+ * globals
+ */
+MODULE_AUTHOR("Michael T. Mayers");
+MODULE_DESCRIPTION("MOTU MidiTimePiece AV multiport MIDI");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{MOTU,MidiTimePiece AV multiport MIDI}}");
+
+// io resources
+#define MTPAV_IOBASE 0x378
+#define MTPAV_IRQ 7
+#define MTPAV_MAX_PORTS 8
+
+static int index = SNDRV_DEFAULT_IDX1;
+static char *id = SNDRV_DEFAULT_STR1;
+static long port = MTPAV_IOBASE; /* 0x378, 0x278 */
+static int irq = MTPAV_IRQ; /* 7, 5 */
+static int hwports = MTPAV_MAX_PORTS; /* use hardware ports 1-8 */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for MotuMTPAV MIDI.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for MotuMTPAV MIDI.");
+module_param(port, long, 0444);
+MODULE_PARM_DESC(port, "Parallel port # for MotuMTPAV MIDI.");
+module_param(irq, int, 0444);
+MODULE_PARM_DESC(irq, "Parallel IRQ # for MotuMTPAV MIDI.");
+module_param(hwports, int, 0444);
+MODULE_PARM_DESC(hwports, "Hardware ports # for MotuMTPAV MIDI.");
+
+static struct platform_device *device;
+
+/*
+ * defines
+ */
+//#define USE_FAKE_MTP // don't actually read/write to MTP device (for debugging without an actual unit) (does not work yet)
+
+// parallel port usage masks
+#define SIGS_BYTE 0x08
+#define SIGS_RFD 0x80
+#define SIGS_IRQ 0x40
+#define SIGS_IN0 0x10
+#define SIGS_IN1 0x20
+
+#define SIGC_WRITE 0x04
+#define SIGC_READ 0x08
+#define SIGC_INTEN 0x10
+
+#define DREG 0
+#define SREG 1
+#define CREG 2
+
+//
+#define MTPAV_MODE_INPUT_OPENED 0x01
+#define MTPAV_MODE_OUTPUT_OPENED 0x02
+#define MTPAV_MODE_INPUT_TRIGGERED 0x04
+#define MTPAV_MODE_OUTPUT_TRIGGERED 0x08
+
+#define NUMPORTS (0x12+1)
+
+
+/*
+ */
+
+struct mtpav_port {
+ u8 number;
+ u8 hwport;
+ u8 mode;
+ u8 running_status;
+ struct snd_rawmidi_substream *input;
+ struct snd_rawmidi_substream *output;
+};
+
+struct mtpav {
+ struct snd_card *card;
+ unsigned long port;
+ struct resource *res_port;
+ int irq; /* interrupt (for inputs) */
+ spinlock_t spinlock;
+ int share_irq; /* number of accesses to input interrupts */
+ int istimer; /* number of accesses to timer interrupts */
+ struct timer_list timer; /* timer interrupts for outputs */
+ struct snd_rawmidi *rmidi;
+ int num_ports; /* number of hw ports (1-8) */
+ struct mtpav_port ports[NUMPORTS]; /* all ports including computer, adat and bc */
+
+ u32 inmidiport; /* selected input midi port */
+ u32 inmidistate; /* during midi command 0xf5 */
+
+ u32 outmidihwport; /* selected output midi hw port */
+};
+
+
+/*
+ * possible hardware ports (selected by 0xf5 port message)
+ * 0x00 all ports
+ * 0x01 .. 0x08 this MTP's ports 1..8
+ * 0x09 .. 0x10 networked MTP's ports (9..16)
+ * 0x11 networked MTP's computer port
+ * 0x63 to ADAT
+ *
+ * mappig:
+ * subdevice 0 - (X-1) ports
+ * X - (2*X-1) networked ports
+ * X computer
+ * X+1 ADAT
+ * X+2 all ports
+ *
+ * where X = chip->num_ports
+ */
+
+#define MTPAV_PIDX_COMPUTER 0
+#define MTPAV_PIDX_ADAT 1
+#define MTPAV_PIDX_BROADCAST 2
+
+
+static int translate_subdevice_to_hwport(struct mtpav *chip, int subdev)
+{
+ if (subdev < 0)
+ return 0x01; /* invalid - use port 0 as default */
+ else if (subdev < chip->num_ports)
+ return subdev + 1; /* single mtp port */
+ else if (subdev < chip->num_ports * 2)
+ return subdev - chip->num_ports + 0x09; /* remote port */
+ else if (subdev == chip->num_ports * 2 + MTPAV_PIDX_COMPUTER)
+ return 0x11; /* computer port */
+ else if (subdev == chip->num_ports + MTPAV_PIDX_ADAT)
+ return 0x63; /* ADAT */
+ return 0; /* all ports */
+}
+
+static int translate_hwport_to_subdevice(struct mtpav *chip, int hwport)
+{
+ int p;
+ if (hwport <= 0x00) /* all ports */
+ return chip->num_ports + MTPAV_PIDX_BROADCAST;
+ else if (hwport <= 0x08) { /* single port */
+ p = hwport - 1;
+ if (p >= chip->num_ports)
+ p = 0;
+ return p;
+ } else if (hwport <= 0x10) { /* remote port */
+ p = hwport - 0x09 + chip->num_ports;
+ if (p >= chip->num_ports * 2)
+ p = chip->num_ports;
+ return p;
+ } else if (hwport == 0x11) /* computer port */
+ return chip->num_ports + MTPAV_PIDX_COMPUTER;
+ else /* ADAT */
+ return chip->num_ports + MTPAV_PIDX_ADAT;
+}
+
+
+/*
+ */
+
+static u8 snd_mtpav_getreg(struct mtpav *chip, u16 reg)
+{
+ u8 rval = 0;
+
+ if (reg == SREG) {
+ rval = inb(chip->port + SREG);
+ rval = (rval & 0xf8);
+ } else if (reg == CREG) {
+ rval = inb(chip->port + CREG);
+ rval = (rval & 0x1c);
+ }
+
+ return rval;
+}
+
+/*
+ */
+
+static inline void snd_mtpav_mputreg(struct mtpav *chip, u16 reg, u8 val)
+{
+ if (reg == DREG || reg == CREG)
+ outb(val, chip->port + reg);
+}
+
+/*
+ */
+
+static void snd_mtpav_wait_rfdhi(struct mtpav *chip)
+{
+ int counts = 10000;
+ u8 sbyte;
+
+ sbyte = snd_mtpav_getreg(chip, SREG);
+ while (!(sbyte & SIGS_RFD) && counts--) {
+ sbyte = snd_mtpav_getreg(chip, SREG);
+ udelay(10);
+ }
+}
+
+static void snd_mtpav_send_byte(struct mtpav *chip, u8 byte)
+{
+ u8 tcbyt;
+ u8 clrwrite;
+ u8 setwrite;
+
+ snd_mtpav_wait_rfdhi(chip);
+
+ /////////////////
+
+ tcbyt = snd_mtpav_getreg(chip, CREG);
+ clrwrite = tcbyt & (SIGC_WRITE ^ 0xff);
+ setwrite = tcbyt | SIGC_WRITE;
+
+ snd_mtpav_mputreg(chip, DREG, byte);
+ snd_mtpav_mputreg(chip, CREG, clrwrite); // clear write bit
+
+ snd_mtpav_mputreg(chip, CREG, setwrite); // set write bit
+
+}
+
+
+/*
+ */
+
+/* call this with spin lock held */
+static void snd_mtpav_output_port_write(struct mtpav *mtp_card,
+ struct mtpav_port *portp,
+ struct snd_rawmidi_substream *substream)
+{
+ u8 outbyte;
+
+ // Get the outbyte first, so we can emulate running status if
+ // necessary
+ if (snd_rawmidi_transmit(substream, &outbyte, 1) != 1)
+ return;
+
+ // send port change command if necessary
+
+ if (portp->hwport != mtp_card->outmidihwport) {
+ mtp_card->outmidihwport = portp->hwport;
+
+ snd_mtpav_send_byte(mtp_card, 0xf5);
+ snd_mtpav_send_byte(mtp_card, portp->hwport);
+ //snd_printk("new outport: 0x%x\n", (unsigned int) portp->hwport);
+
+ if (!(outbyte & 0x80) && portp->running_status)
+ snd_mtpav_send_byte(mtp_card, portp->running_status);
+ }
+
+ // send data
+
+ do {
+ if (outbyte & 0x80)
+ portp->running_status = outbyte;
+
+ snd_mtpav_send_byte(mtp_card, outbyte);
+ } while (snd_rawmidi_transmit(substream, &outbyte, 1) == 1);
+}
+
+static void snd_mtpav_output_write(struct snd_rawmidi_substream *substream)
+{
+ struct mtpav *mtp_card = substream->rmidi->private_data;
+ struct mtpav_port *portp = &mtp_card->ports[substream->number];
+ unsigned long flags;
+
+ spin_lock_irqsave(&mtp_card->spinlock, flags);
+ snd_mtpav_output_port_write(mtp_card, portp, substream);
+ spin_unlock_irqrestore(&mtp_card->spinlock, flags);
+}
+
+
+/*
+ * mtpav control
+ */
+
+static void snd_mtpav_portscan(struct mtpav *chip) // put mtp into smart routing mode
+{
+ u8 p;
+
+ for (p = 0; p < 8; p++) {
+ snd_mtpav_send_byte(chip, 0xf5);
+ snd_mtpav_send_byte(chip, p);
+ snd_mtpav_send_byte(chip, 0xfe);
+ }
+}
+
+/*
+ */
+
+static int snd_mtpav_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct mtpav *mtp_card = substream->rmidi->private_data;
+ struct mtpav_port *portp = &mtp_card->ports[substream->number];
+ unsigned long flags;
+
+ spin_lock_irqsave(&mtp_card->spinlock, flags);
+ portp->mode |= MTPAV_MODE_INPUT_OPENED;
+ portp->input = substream;
+ if (mtp_card->share_irq++ == 0)
+ snd_mtpav_mputreg(mtp_card, CREG, (SIGC_INTEN | SIGC_WRITE)); // enable pport interrupts
+ spin_unlock_irqrestore(&mtp_card->spinlock, flags);
+ return 0;
+}
+
+/*
+ */
+
+static int snd_mtpav_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct mtpav *mtp_card = substream->rmidi->private_data;
+ struct mtpav_port *portp = &mtp_card->ports[substream->number];
+ unsigned long flags;
+
+ spin_lock_irqsave(&mtp_card->spinlock, flags);
+ portp->mode &= ~MTPAV_MODE_INPUT_OPENED;
+ portp->input = NULL;
+ if (--mtp_card->share_irq == 0)
+ snd_mtpav_mputreg(mtp_card, CREG, 0); // disable pport interrupts
+ spin_unlock_irqrestore(&mtp_card->spinlock, flags);
+ return 0;
+}
+
+/*
+ */
+
+static void snd_mtpav_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct mtpav *mtp_card = substream->rmidi->private_data;
+ struct mtpav_port *portp = &mtp_card->ports[substream->number];
+ unsigned long flags;
+
+ spin_lock_irqsave(&mtp_card->spinlock, flags);
+ if (up)
+ portp->mode |= MTPAV_MODE_INPUT_TRIGGERED;
+ else
+ portp->mode &= ~MTPAV_MODE_INPUT_TRIGGERED;
+ spin_unlock_irqrestore(&mtp_card->spinlock, flags);
+
+}
+
+
+/*
+ * timer interrupt for outputs
+ */
+
+static void snd_mtpav_output_timer(unsigned long data)
+{
+ unsigned long flags;
+ struct mtpav *chip = (struct mtpav *)data;
+ int p;
+
+ spin_lock_irqsave(&chip->spinlock, flags);
+ /* reprogram timer */
+ chip->timer.expires = 1 + jiffies;
+ add_timer(&chip->timer);
+ /* process each port */
+ for (p = 0; p <= chip->num_ports * 2 + MTPAV_PIDX_BROADCAST; p++) {
+ struct mtpav_port *portp = &chip->ports[p];
+ if ((portp->mode & MTPAV_MODE_OUTPUT_TRIGGERED) && portp->output)
+ snd_mtpav_output_port_write(chip, portp, portp->output);
+ }
+ spin_unlock_irqrestore(&chip->spinlock, flags);
+}
+
+/* spinlock held! */
+static void snd_mtpav_add_output_timer(struct mtpav *chip)
+{
+ chip->timer.expires = 1 + jiffies;
+ add_timer(&chip->timer);
+}
+
+/* spinlock held! */
+static void snd_mtpav_remove_output_timer(struct mtpav *chip)
+{
+ del_timer(&chip->timer);
+}
+
+/*
+ */
+
+static int snd_mtpav_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct mtpav *mtp_card = substream->rmidi->private_data;
+ struct mtpav_port *portp = &mtp_card->ports[substream->number];
+ unsigned long flags;
+
+ spin_lock_irqsave(&mtp_card->spinlock, flags);
+ portp->mode |= MTPAV_MODE_OUTPUT_OPENED;
+ portp->output = substream;
+ spin_unlock_irqrestore(&mtp_card->spinlock, flags);
+ return 0;
+};
+
+/*
+ */
+
+static int snd_mtpav_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct mtpav *mtp_card = substream->rmidi->private_data;
+ struct mtpav_port *portp = &mtp_card->ports[substream->number];
+ unsigned long flags;
+
+ spin_lock_irqsave(&mtp_card->spinlock, flags);
+ portp->mode &= ~MTPAV_MODE_OUTPUT_OPENED;
+ portp->output = NULL;
+ spin_unlock_irqrestore(&mtp_card->spinlock, flags);
+ return 0;
+};
+
+/*
+ */
+
+static void snd_mtpav_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct mtpav *mtp_card = substream->rmidi->private_data;
+ struct mtpav_port *portp = &mtp_card->ports[substream->number];
+ unsigned long flags;
+
+ spin_lock_irqsave(&mtp_card->spinlock, flags);
+ if (up) {
+ if (! (portp->mode & MTPAV_MODE_OUTPUT_TRIGGERED)) {
+ if (mtp_card->istimer++ == 0)
+ snd_mtpav_add_output_timer(mtp_card);
+ portp->mode |= MTPAV_MODE_OUTPUT_TRIGGERED;
+ }
+ } else {
+ portp->mode &= ~MTPAV_MODE_OUTPUT_TRIGGERED;
+ if (--mtp_card->istimer == 0)
+ snd_mtpav_remove_output_timer(mtp_card);
+ }
+ spin_unlock_irqrestore(&mtp_card->spinlock, flags);
+
+ if (up)
+ snd_mtpav_output_write(substream);
+}
+
+/*
+ * midi interrupt for inputs
+ */
+
+static void snd_mtpav_inmidi_process(struct mtpav *mcrd, u8 inbyte)
+{
+ struct mtpav_port *portp;
+
+ if ((int)mcrd->inmidiport > mcrd->num_ports * 2 + MTPAV_PIDX_BROADCAST)
+ return;
+
+ portp = &mcrd->ports[mcrd->inmidiport];
+ if (portp->mode & MTPAV_MODE_INPUT_TRIGGERED)
+ snd_rawmidi_receive(portp->input, &inbyte, 1);
+}
+
+static void snd_mtpav_inmidi_h(struct mtpav *mcrd, u8 inbyte)
+{
+ if (inbyte >= 0xf8) {
+ /* real-time midi code */
+ snd_mtpav_inmidi_process(mcrd, inbyte);
+ return;
+ }
+
+ if (mcrd->inmidistate == 0) { // awaiting command
+ if (inbyte == 0xf5) // MTP port #
+ mcrd->inmidistate = 1;
+ else
+ snd_mtpav_inmidi_process(mcrd, inbyte);
+ } else if (mcrd->inmidistate) {
+ mcrd->inmidiport = translate_hwport_to_subdevice(mcrd, inbyte);
+ mcrd->inmidistate = 0;
+ }
+}
+
+static void snd_mtpav_read_bytes(struct mtpav *mcrd)
+{
+ u8 clrread, setread;
+ u8 mtp_read_byte;
+ u8 sr, cbyt;
+ int i;
+
+ u8 sbyt = snd_mtpav_getreg(mcrd, SREG);
+
+ //printk("snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt);
+
+ if (!(sbyt & SIGS_BYTE))
+ return;
+
+ cbyt = snd_mtpav_getreg(mcrd, CREG);
+ clrread = cbyt & (SIGC_READ ^ 0xff);
+ setread = cbyt | SIGC_READ;
+
+ do {
+
+ mtp_read_byte = 0;
+ for (i = 0; i < 4; i++) {
+ snd_mtpav_mputreg(mcrd, CREG, setread);
+ sr = snd_mtpav_getreg(mcrd, SREG);
+ snd_mtpav_mputreg(mcrd, CREG, clrread);
+
+ sr &= SIGS_IN0 | SIGS_IN1;
+ sr >>= 4;
+ mtp_read_byte |= sr << (i * 2);
+ }
+
+ snd_mtpav_inmidi_h(mcrd, mtp_read_byte);
+
+ sbyt = snd_mtpav_getreg(mcrd, SREG);
+
+ } while (sbyt & SIGS_BYTE);
+}
+
+static irqreturn_t snd_mtpav_irqh(int irq, void *dev_id)
+{
+ struct mtpav *mcard = dev_id;
+
+ spin_lock(&mcard->spinlock);
+ snd_mtpav_read_bytes(mcard);
+ spin_unlock(&mcard->spinlock);
+ return IRQ_HANDLED;
+}
+
+/*
+ * get ISA resources
+ */
+static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard)
+{
+ if ((mcard->res_port = request_region(port, 3, "MotuMTPAV MIDI")) == NULL) {
+ snd_printk("MTVAP port 0x%lx is busy\n", port);
+ return -EBUSY;
+ }
+ mcard->port = port;
+ if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) {
+ snd_printk("MTVAP IRQ %d busy\n", irq);
+ return -EBUSY;
+ }
+ mcard->irq = irq;
+ return 0;
+}
+
+
+/*
+ */
+
+static struct snd_rawmidi_ops snd_mtpav_output = {
+ .open = snd_mtpav_output_open,
+ .close = snd_mtpav_output_close,
+ .trigger = snd_mtpav_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_mtpav_input = {
+ .open = snd_mtpav_input_open,
+ .close = snd_mtpav_input_close,
+ .trigger = snd_mtpav_input_trigger,
+};
+
+
+/*
+ * get RAWMIDI resources
+ */
+
+static void __devinit snd_mtpav_set_name(struct mtpav *chip,
+ struct snd_rawmidi_substream *substream)
+{
+ if (substream->number >= 0 && substream->number < chip->num_ports)
+ sprintf(substream->name, "MTP direct %d", (substream->number % chip->num_ports) + 1);
+ else if (substream->number >= 8 && substream->number < chip->num_ports * 2)
+ sprintf(substream->name, "MTP remote %d", (substream->number % chip->num_ports) + 1);
+ else if (substream->number == chip->num_ports * 2)
+ strcpy(substream->name, "MTP computer");
+ else if (substream->number == chip->num_ports * 2 + 1)
+ strcpy(substream->name, "MTP ADAT");
+ else
+ strcpy(substream->name, "MTP broadcast");
+}
+
+static int __devinit snd_mtpav_get_RAWMIDI(struct mtpav *mcard)
+{
+ int rval;
+ struct snd_rawmidi *rawmidi;
+ struct snd_rawmidi_substream *substream;
+ struct list_head *list;
+
+ if (hwports < 1)
+ hwports = 1;
+ else if (hwports > 8)
+ hwports = 8;
+ mcard->num_ports = hwports;
+
+ if ((rval = snd_rawmidi_new(mcard->card, "MotuMIDI", 0,
+ mcard->num_ports * 2 + MTPAV_PIDX_BROADCAST + 1,
+ mcard->num_ports * 2 + MTPAV_PIDX_BROADCAST + 1,
+ &mcard->rmidi)) < 0)
+ return rval;
+ rawmidi = mcard->rmidi;
+ rawmidi->private_data = mcard;
+
+ list_for_each(list, &rawmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) {
+ substream = list_entry(list, struct snd_rawmidi_substream, list);
+ snd_mtpav_set_name(mcard, substream);
+ substream->ops = &snd_mtpav_input;
+ }
+ list_for_each(list, &rawmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) {
+ substream = list_entry(list, struct snd_rawmidi_substream, list);
+ snd_mtpav_set_name(mcard, substream);
+ substream->ops = &snd_mtpav_output;
+ mcard->ports[substream->number].hwport = translate_subdevice_to_hwport(mcard, substream->number);
+ }
+ rawmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ sprintf(rawmidi->name, "MTP AV MIDI");
+ return 0;
+}
+
+/*
+ */
+
+static void snd_mtpav_free(struct snd_card *card)
+{
+ struct mtpav *crd = card->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&crd->spinlock, flags);
+ if (crd->istimer > 0)
+ snd_mtpav_remove_output_timer(crd);
+ spin_unlock_irqrestore(&crd->spinlock, flags);
+ if (crd->irq >= 0)
+ free_irq(crd->irq, (void *)crd);
+ release_and_free_resource(crd->res_port);
+}
+
+/*
+ */
+static int __devinit snd_mtpav_probe(struct platform_device *dev)
+{
+ struct snd_card *card;
+ int err;
+ struct mtpav *mtp_card;
+
+ card = snd_card_new(index, id, THIS_MODULE, sizeof(*mtp_card));
+ if (! card)
+ return -ENOMEM;
+
+ mtp_card = card->private_data;
+ spin_lock_init(&mtp_card->spinlock);
+ init_timer(&mtp_card->timer);
+ mtp_card->card = card;
+ mtp_card->irq = -1;
+ mtp_card->share_irq = 0;
+ mtp_card->inmidistate = 0;
+ mtp_card->outmidihwport = 0xffffffff;
+ init_timer(&mtp_card->timer);
+ mtp_card->timer.function = snd_mtpav_output_timer;
+ mtp_card->timer.data = (unsigned long) mtp_card;
+
+ card->private_free = snd_mtpav_free;
+
+ err = snd_mtpav_get_RAWMIDI(mtp_card);
+ if (err < 0)
+ goto __error;
+
+ mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST;
+
+ err = snd_mtpav_get_ISA(mtp_card);
+ if (err < 0)
+ goto __error;
+
+ strcpy(card->driver, "MTPAV");
+ strcpy(card->shortname, "MTPAV on parallel port");
+ snprintf(card->longname, sizeof(card->longname),
+ "MTPAV on parallel port at 0x%lx", port);
+
+ snd_mtpav_portscan(mtp_card);
+
+ snd_card_set_dev(card, &dev->dev);
+ err = snd_card_register(mtp_card->card);
+ if (err < 0)
+ goto __error;
+
+ platform_set_drvdata(dev, card);
+ printk(KERN_INFO "Motu MidiTimePiece on parallel port irq: %d ioport: 0x%lx\n", irq, port);
+ return 0;
+
+ __error:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_mtpav_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define SND_MTPAV_DRIVER "snd_mtpav"
+
+static struct platform_driver snd_mtpav_driver = {
+ .probe = snd_mtpav_probe,
+ .remove = __devexit_p(snd_mtpav_remove),
+ .driver = {
+ .name = SND_MTPAV_DRIVER
+ },
+};
+
+static int __init alsa_card_mtpav_init(void)
+{
+ int err;
+
+ if ((err = platform_driver_register(&snd_mtpav_driver)) < 0)
+ return err;
+
+ device = platform_device_register_simple(SND_MTPAV_DRIVER, -1, NULL, 0);
+ if (!IS_ERR(device)) {
+ if (platform_get_drvdata(device))
+ return 0;
+ platform_device_unregister(device);
+ err = -ENODEV;
+ } else
+ err = PTR_ERR(device);
+ platform_driver_unregister(&snd_mtpav_driver);
+ return err;
+}
+
+static void __exit alsa_card_mtpav_exit(void)
+{
+ platform_device_unregister(device);
+ platform_driver_unregister(&snd_mtpav_driver);
+}
+
+module_init(alsa_card_mtpav_init)
+module_exit(alsa_card_mtpav_exit)
diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c
new file mode 100644
index 0000000..87ba1dd
--- /dev/null
+++ b/sound/drivers/mts64.c
@@ -0,0 +1,1088 @@
+/*
+ * ALSA Driver for Ego Systems Inc. (ESI) Miditerminal 4140
+ * Copyright (c) 2006 by Matthias König <mk@phasorlab.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/parport.h>
+#include <linux/spinlock.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <sound/control.h>
+
+#define CARD_NAME "Miditerminal 4140"
+#define DRIVER_NAME "MTS64"
+#define PLATFORM_DRIVER "snd_mts64"
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+static struct platform_device *platform_devices[SNDRV_CARDS];
+static int device_count;
+
+module_param_array(index, int, NULL, S_IRUGO);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, S_IRUGO);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, S_IRUGO);
+MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
+
+MODULE_AUTHOR("Matthias Koenig <mk@phasorlab.de>");
+MODULE_DESCRIPTION("ESI Miditerminal 4140");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ESI,Miditerminal 4140}}");
+
+/*********************************************************************
+ * Chip specific
+ *********************************************************************/
+#define MTS64_NUM_INPUT_PORTS 5
+#define MTS64_NUM_OUTPUT_PORTS 4
+#define MTS64_SMPTE_SUBSTREAM 4
+
+struct mts64 {
+ spinlock_t lock;
+ struct snd_card *card;
+ struct snd_rawmidi *rmidi;
+ struct pardevice *pardev;
+ int pardev_claimed;
+
+ int open_count;
+ int current_midi_output_port;
+ int current_midi_input_port;
+ u8 mode[MTS64_NUM_INPUT_PORTS];
+ struct snd_rawmidi_substream *midi_input_substream[MTS64_NUM_INPUT_PORTS];
+ int smpte_switch;
+ u8 time[4]; /* [0]=hh, [1]=mm, [2]=ss, [3]=ff */
+ u8 fps;
+};
+
+static int snd_mts64_free(struct mts64 *mts)
+{
+ kfree(mts);
+ return 0;
+}
+
+static int __devinit snd_mts64_create(struct snd_card *card,
+ struct pardevice *pardev,
+ struct mts64 **rchip)
+{
+ struct mts64 *mts;
+
+ *rchip = NULL;
+
+ mts = kzalloc(sizeof(struct mts64), GFP_KERNEL);
+ if (mts == NULL)
+ return -ENOMEM;
+
+ /* Init chip specific data */
+ spin_lock_init(&mts->lock);
+ mts->card = card;
+ mts->pardev = pardev;
+ mts->current_midi_output_port = -1;
+ mts->current_midi_input_port = -1;
+
+ *rchip = mts;
+
+ return 0;
+}
+
+/*********************************************************************
+ * HW register related constants
+ *********************************************************************/
+
+/* Status Bits */
+#define MTS64_STAT_BSY 0x80
+#define MTS64_STAT_BIT_SET 0x20 /* readout process, bit is set */
+#define MTS64_STAT_PORT 0x10 /* read byte is a port number */
+
+/* Control Bits */
+#define MTS64_CTL_READOUT 0x08 /* enable readout */
+#define MTS64_CTL_WRITE_CMD 0x06
+#define MTS64_CTL_WRITE_DATA 0x02
+#define MTS64_CTL_STROBE 0x01
+
+/* Command */
+#define MTS64_CMD_RESET 0xfe
+#define MTS64_CMD_PROBE 0x8f /* Used in probing procedure */
+#define MTS64_CMD_SMPTE_SET_TIME 0xe8
+#define MTS64_CMD_SMPTE_SET_FPS 0xee
+#define MTS64_CMD_SMPTE_STOP 0xef
+#define MTS64_CMD_SMPTE_FPS_24 0xe3
+#define MTS64_CMD_SMPTE_FPS_25 0xe2
+#define MTS64_CMD_SMPTE_FPS_2997 0xe4
+#define MTS64_CMD_SMPTE_FPS_30D 0xe1
+#define MTS64_CMD_SMPTE_FPS_30 0xe0
+#define MTS64_CMD_COM_OPEN 0xf8 /* setting the communication mode */
+#define MTS64_CMD_COM_CLOSE1 0xff /* clearing communication mode */
+#define MTS64_CMD_COM_CLOSE2 0xf5
+
+/*********************************************************************
+ * Hardware specific functions
+ *********************************************************************/
+static void mts64_enable_readout(struct parport *p);
+static void mts64_disable_readout(struct parport *p);
+static int mts64_device_ready(struct parport *p);
+static int mts64_device_init(struct parport *p);
+static int mts64_device_open(struct mts64 *mts);
+static int mts64_device_close(struct mts64 *mts);
+static u8 mts64_map_midi_input(u8 c);
+static int mts64_probe(struct parport *p);
+static u16 mts64_read(struct parport *p);
+static u8 mts64_read_char(struct parport *p);
+static void mts64_smpte_start(struct parport *p,
+ u8 hours, u8 minutes,
+ u8 seconds, u8 frames,
+ u8 idx);
+static void mts64_smpte_stop(struct parport *p);
+static void mts64_write_command(struct parport *p, u8 c);
+static void mts64_write_data(struct parport *p, u8 c);
+static void mts64_write_midi(struct mts64 *mts, u8 c, int midiport);
+
+
+/* Enables the readout procedure
+ *
+ * Before we can read a midi byte from the device, we have to set
+ * bit 3 of control port.
+ */
+static void mts64_enable_readout(struct parport *p)
+{
+ u8 c;
+
+ c = parport_read_control(p);
+ c |= MTS64_CTL_READOUT;
+ parport_write_control(p, c);
+}
+
+/* Disables readout
+ *
+ * Readout is disabled by clearing bit 3 of control
+ */
+static void mts64_disable_readout(struct parport *p)
+{
+ u8 c;
+
+ c = parport_read_control(p);
+ c &= ~MTS64_CTL_READOUT;
+ parport_write_control(p, c);
+}
+
+/* waits for device ready
+ *
+ * Checks if BUSY (Bit 7 of status) is clear
+ * 1 device ready
+ * 0 failure
+ */
+static int mts64_device_ready(struct parport *p)
+{
+ int i;
+ u8 c;
+
+ for (i = 0; i < 0xffff; ++i) {
+ c = parport_read_status(p);
+ c &= MTS64_STAT_BSY;
+ if (c != 0)
+ return 1;
+ }
+
+ return 0;
+}
+
+/* Init device (LED blinking startup magic)
+ *
+ * Returns:
+ * 0 init ok
+ * -EIO failure
+ */
+static int __devinit mts64_device_init(struct parport *p)
+{
+ int i;
+
+ mts64_write_command(p, MTS64_CMD_RESET);
+
+ for (i = 0; i < 64; ++i) {
+ msleep(100);
+
+ if (mts64_probe(p) == 0) {
+ /* success */
+ mts64_disable_readout(p);
+ return 0;
+ }
+ }
+ mts64_disable_readout(p);
+
+ return -EIO;
+}
+
+/*
+ * Opens the device (set communication mode)
+ */
+static int mts64_device_open(struct mts64 *mts)
+{
+ int i;
+ struct parport *p = mts->pardev->port;
+
+ for (i = 0; i < 5; ++i)
+ mts64_write_command(p, MTS64_CMD_COM_OPEN);
+
+ return 0;
+}
+
+/*
+ * Close device (clear communication mode)
+ */
+static int mts64_device_close(struct mts64 *mts)
+{
+ int i;
+ struct parport *p = mts->pardev->port;
+
+ for (i = 0; i < 5; ++i) {
+ mts64_write_command(p, MTS64_CMD_COM_CLOSE1);
+ mts64_write_command(p, MTS64_CMD_COM_CLOSE2);
+ }
+
+ return 0;
+}
+
+/* map hardware port to substream number
+ *
+ * When reading a byte from the device, the device tells us
+ * on what port the byte is. This HW port has to be mapped to
+ * the midiport (substream number).
+ * substream 0-3 are Midiports 1-4
+ * substream 4 is SMPTE Timecode
+ * The mapping is done by the table:
+ * HW | 0 | 1 | 2 | 3 | 4
+ * SW | 0 | 1 | 4 | 2 | 3
+ */
+static u8 mts64_map_midi_input(u8 c)
+{
+ static u8 map[] = { 0, 1, 4, 2, 3 };
+
+ return map[c];
+}
+
+
+/* Probe parport for device
+ *
+ * Do we have a Miditerminal 4140 on parport?
+ * Returns:
+ * 0 device found
+ * -ENODEV no device
+ */
+static int __devinit mts64_probe(struct parport *p)
+{
+ u8 c;
+
+ mts64_smpte_stop(p);
+ mts64_write_command(p, MTS64_CMD_PROBE);
+
+ msleep(50);
+
+ c = mts64_read(p);
+
+ c &= 0x00ff;
+ if (c != MTS64_CMD_PROBE)
+ return -ENODEV;
+ else
+ return 0;
+
+}
+
+/* Read byte incl. status from device
+ *
+ * Returns:
+ * data in lower 8 bits and status in upper 8 bits
+ */
+static u16 mts64_read(struct parport *p)
+{
+ u8 data, status;
+
+ mts64_device_ready(p);
+ mts64_enable_readout(p);
+ status = parport_read_status(p);
+ data = mts64_read_char(p);
+ mts64_disable_readout(p);
+
+ return (status << 8) | data;
+}
+
+/* Read a byte from device
+ *
+ * Note, that readout mode has to be enabled.
+ * readout procedure is as follows:
+ * - Write number of the Bit to read to DATA
+ * - Read STATUS
+ * - Bit 5 of STATUS indicates if Bit is set
+ *
+ * Returns:
+ * Byte read from device
+ */
+static u8 mts64_read_char(struct parport *p)
+{
+ u8 c = 0;
+ u8 status;
+ u8 i;
+
+ for (i = 0; i < 8; ++i) {
+ parport_write_data(p, i);
+ c >>= 1;
+ status = parport_read_status(p);
+ if (status & MTS64_STAT_BIT_SET)
+ c |= 0x80;
+ }
+
+ return c;
+}
+
+/* Starts SMPTE Timecode generation
+ *
+ * The device creates SMPTE Timecode by hardware.
+ * 0 24 fps
+ * 1 25 fps
+ * 2 29.97 fps
+ * 3 30 fps (Drop-frame)
+ * 4 30 fps
+ */
+static void mts64_smpte_start(struct parport *p,
+ u8 hours, u8 minutes,
+ u8 seconds, u8 frames,
+ u8 idx)
+{
+ static u8 fps[5] = { MTS64_CMD_SMPTE_FPS_24,
+ MTS64_CMD_SMPTE_FPS_25,
+ MTS64_CMD_SMPTE_FPS_2997,
+ MTS64_CMD_SMPTE_FPS_30D,
+ MTS64_CMD_SMPTE_FPS_30 };
+
+ mts64_write_command(p, MTS64_CMD_SMPTE_SET_TIME);
+ mts64_write_command(p, frames);
+ mts64_write_command(p, seconds);
+ mts64_write_command(p, minutes);
+ mts64_write_command(p, hours);
+
+ mts64_write_command(p, MTS64_CMD_SMPTE_SET_FPS);
+ mts64_write_command(p, fps[idx]);
+}
+
+/* Stops SMPTE Timecode generation
+ */
+static void mts64_smpte_stop(struct parport *p)
+{
+ mts64_write_command(p, MTS64_CMD_SMPTE_STOP);
+}
+
+/* Write a command byte to device
+ */
+static void mts64_write_command(struct parport *p, u8 c)
+{
+ mts64_device_ready(p);
+
+ parport_write_data(p, c);
+
+ parport_write_control(p, MTS64_CTL_WRITE_CMD);
+ parport_write_control(p, MTS64_CTL_WRITE_CMD | MTS64_CTL_STROBE);
+ parport_write_control(p, MTS64_CTL_WRITE_CMD);
+}
+
+/* Write a data byte to device
+ */
+static void mts64_write_data(struct parport *p, u8 c)
+{
+ mts64_device_ready(p);
+
+ parport_write_data(p, c);
+
+ parport_write_control(p, MTS64_CTL_WRITE_DATA);
+ parport_write_control(p, MTS64_CTL_WRITE_DATA | MTS64_CTL_STROBE);
+ parport_write_control(p, MTS64_CTL_WRITE_DATA);
+}
+
+/* Write a MIDI byte to midiport
+ *
+ * midiport ranges from 0-3 and maps to Ports 1-4
+ * assumptions: communication mode is on
+ */
+static void mts64_write_midi(struct mts64 *mts, u8 c,
+ int midiport)
+{
+ struct parport *p = mts->pardev->port;
+
+ /* check current midiport */
+ if (mts->current_midi_output_port != midiport)
+ mts64_write_command(p, midiport);
+
+ /* write midi byte */
+ mts64_write_data(p, c);
+}
+
+/*********************************************************************
+ * Control elements
+ *********************************************************************/
+
+/* SMPTE Switch */
+#define snd_mts64_ctl_smpte_switch_info snd_ctl_boolean_mono_info
+
+static int snd_mts64_ctl_smpte_switch_get(struct snd_kcontrol* kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct mts64 *mts = snd_kcontrol_chip(kctl);
+
+ spin_lock_irq(&mts->lock);
+ uctl->value.integer.value[0] = mts->smpte_switch;
+ spin_unlock_irq(&mts->lock);
+
+ return 0;
+}
+
+/* smpte_switch is not accessed from IRQ handler, so we just need
+ to protect the HW access */
+static int snd_mts64_ctl_smpte_switch_put(struct snd_kcontrol* kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct mts64 *mts = snd_kcontrol_chip(kctl);
+ int changed = 0;
+ int val = !!uctl->value.integer.value[0];
+
+ spin_lock_irq(&mts->lock);
+ if (mts->smpte_switch == val)
+ goto __out;
+
+ changed = 1;
+ mts->smpte_switch = val;
+ if (mts->smpte_switch) {
+ mts64_smpte_start(mts->pardev->port,
+ mts->time[0], mts->time[1],
+ mts->time[2], mts->time[3],
+ mts->fps);
+ } else {
+ mts64_smpte_stop(mts->pardev->port);
+ }
+__out:
+ spin_unlock_irq(&mts->lock);
+ return changed;
+}
+
+static struct snd_kcontrol_new mts64_ctl_smpte_switch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_RAWMIDI,
+ .name = "SMPTE Playback Switch",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0,
+ .info = snd_mts64_ctl_smpte_switch_info,
+ .get = snd_mts64_ctl_smpte_switch_get,
+ .put = snd_mts64_ctl_smpte_switch_put
+};
+
+/* Time */
+static int snd_mts64_ctl_smpte_time_h_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 23;
+ return 0;
+}
+
+static int snd_mts64_ctl_smpte_time_f_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 99;
+ return 0;
+}
+
+static int snd_mts64_ctl_smpte_time_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 59;
+ return 0;
+}
+
+static int snd_mts64_ctl_smpte_time_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct mts64 *mts = snd_kcontrol_chip(kctl);
+ int idx = kctl->private_value;
+
+ spin_lock_irq(&mts->lock);
+ uctl->value.integer.value[0] = mts->time[idx];
+ spin_unlock_irq(&mts->lock);
+
+ return 0;
+}
+
+static int snd_mts64_ctl_smpte_time_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct mts64 *mts = snd_kcontrol_chip(kctl);
+ int idx = kctl->private_value;
+ unsigned int time = uctl->value.integer.value[0] % 60;
+ int changed = 0;
+
+ spin_lock_irq(&mts->lock);
+ if (mts->time[idx] != time) {
+ changed = 1;
+ mts->time[idx] = time;
+ }
+ spin_unlock_irq(&mts->lock);
+
+ return changed;
+}
+
+static struct snd_kcontrol_new mts64_ctl_smpte_time_hours __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_RAWMIDI,
+ .name = "SMPTE Time Hours",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0,
+ .info = snd_mts64_ctl_smpte_time_h_info,
+ .get = snd_mts64_ctl_smpte_time_get,
+ .put = snd_mts64_ctl_smpte_time_put
+};
+
+static struct snd_kcontrol_new mts64_ctl_smpte_time_minutes __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_RAWMIDI,
+ .name = "SMPTE Time Minutes",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 1,
+ .info = snd_mts64_ctl_smpte_time_info,
+ .get = snd_mts64_ctl_smpte_time_get,
+ .put = snd_mts64_ctl_smpte_time_put
+};
+
+static struct snd_kcontrol_new mts64_ctl_smpte_time_seconds __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_RAWMIDI,
+ .name = "SMPTE Time Seconds",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 2,
+ .info = snd_mts64_ctl_smpte_time_info,
+ .get = snd_mts64_ctl_smpte_time_get,
+ .put = snd_mts64_ctl_smpte_time_put
+};
+
+static struct snd_kcontrol_new mts64_ctl_smpte_time_frames __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_RAWMIDI,
+ .name = "SMPTE Time Frames",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 3,
+ .info = snd_mts64_ctl_smpte_time_f_info,
+ .get = snd_mts64_ctl_smpte_time_get,
+ .put = snd_mts64_ctl_smpte_time_put
+};
+
+/* FPS */
+static int snd_mts64_ctl_smpte_fps_info(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[5] = { "24",
+ "25",
+ "29.97",
+ "30D",
+ "30" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 5;
+ if (uinfo->value.enumerated.item > 4)
+ uinfo->value.enumerated.item = 4;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int snd_mts64_ctl_smpte_fps_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct mts64 *mts = snd_kcontrol_chip(kctl);
+
+ spin_lock_irq(&mts->lock);
+ uctl->value.enumerated.item[0] = mts->fps;
+ spin_unlock_irq(&mts->lock);
+
+ return 0;
+}
+
+static int snd_mts64_ctl_smpte_fps_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct mts64 *mts = snd_kcontrol_chip(kctl);
+ int changed = 0;
+
+ if (uctl->value.enumerated.item[0] >= 5)
+ return -EINVAL;
+ spin_lock_irq(&mts->lock);
+ if (mts->fps != uctl->value.enumerated.item[0]) {
+ changed = 1;
+ mts->fps = uctl->value.enumerated.item[0];
+ }
+ spin_unlock_irq(&mts->lock);
+
+ return changed;
+}
+
+static struct snd_kcontrol_new mts64_ctl_smpte_fps __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_RAWMIDI,
+ .name = "SMPTE Fps",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0,
+ .info = snd_mts64_ctl_smpte_fps_info,
+ .get = snd_mts64_ctl_smpte_fps_get,
+ .put = snd_mts64_ctl_smpte_fps_put
+};
+
+
+static int __devinit snd_mts64_ctl_create(struct snd_card *card,
+ struct mts64 *mts)
+{
+ int err, i;
+ static struct snd_kcontrol_new *control[] __devinitdata = {
+ &mts64_ctl_smpte_switch,
+ &mts64_ctl_smpte_time_hours,
+ &mts64_ctl_smpte_time_minutes,
+ &mts64_ctl_smpte_time_seconds,
+ &mts64_ctl_smpte_time_frames,
+ &mts64_ctl_smpte_fps,
+ NULL };
+
+ for (i = 0; control[i]; ++i) {
+ err = snd_ctl_add(card, snd_ctl_new1(control[i], mts));
+ if (err < 0) {
+ snd_printd("Cannot create control: %s\n",
+ control[i]->name);
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+/*********************************************************************
+ * Rawmidi
+ *********************************************************************/
+#define MTS64_MODE_INPUT_TRIGGERED 0x01
+
+static int snd_mts64_rawmidi_open(struct snd_rawmidi_substream *substream)
+{
+ struct mts64 *mts = substream->rmidi->private_data;
+
+ if (mts->open_count == 0) {
+ /* We don't need a spinlock here, because this is just called
+ if the device has not been opened before.
+ So there aren't any IRQs from the device */
+ mts64_device_open(mts);
+
+ msleep(50);
+ }
+ ++(mts->open_count);
+
+ return 0;
+}
+
+static int snd_mts64_rawmidi_close(struct snd_rawmidi_substream *substream)
+{
+ struct mts64 *mts = substream->rmidi->private_data;
+ unsigned long flags;
+
+ --(mts->open_count);
+ if (mts->open_count == 0) {
+ /* We need the spinlock_irqsave here because we can still
+ have IRQs at this point */
+ spin_lock_irqsave(&mts->lock, flags);
+ mts64_device_close(mts);
+ spin_unlock_irqrestore(&mts->lock, flags);
+
+ msleep(500);
+
+ } else if (mts->open_count < 0)
+ mts->open_count = 0;
+
+ return 0;
+}
+
+static void snd_mts64_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct mts64 *mts = substream->rmidi->private_data;
+ u8 data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&mts->lock, flags);
+ while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) {
+ mts64_write_midi(mts, data, substream->number+1);
+ snd_rawmidi_transmit_ack(substream, 1);
+ }
+ spin_unlock_irqrestore(&mts->lock, flags);
+}
+
+static void snd_mts64_rawmidi_input_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct mts64 *mts = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&mts->lock, flags);
+ if (up)
+ mts->mode[substream->number] |= MTS64_MODE_INPUT_TRIGGERED;
+ else
+ mts->mode[substream->number] &= ~MTS64_MODE_INPUT_TRIGGERED;
+
+ spin_unlock_irqrestore(&mts->lock, flags);
+}
+
+static struct snd_rawmidi_ops snd_mts64_rawmidi_output_ops = {
+ .open = snd_mts64_rawmidi_open,
+ .close = snd_mts64_rawmidi_close,
+ .trigger = snd_mts64_rawmidi_output_trigger
+};
+
+static struct snd_rawmidi_ops snd_mts64_rawmidi_input_ops = {
+ .open = snd_mts64_rawmidi_open,
+ .close = snd_mts64_rawmidi_close,
+ .trigger = snd_mts64_rawmidi_input_trigger
+};
+
+/* Create and initialize the rawmidi component */
+static int __devinit snd_mts64_rawmidi_create(struct snd_card *card)
+{
+ struct mts64 *mts = card->private_data;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream;
+ struct list_head *list;
+ int err;
+
+ err = snd_rawmidi_new(card, CARD_NAME, 0,
+ MTS64_NUM_OUTPUT_PORTS,
+ MTS64_NUM_INPUT_PORTS,
+ &rmidi);
+ if (err < 0)
+ return err;
+
+ rmidi->private_data = mts;
+ strcpy(rmidi->name, CARD_NAME);
+ rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ mts->rmidi = rmidi;
+
+ /* register rawmidi ops */
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_mts64_rawmidi_output_ops);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_mts64_rawmidi_input_ops);
+
+ /* name substreams */
+ /* output */
+ list_for_each(list,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) {
+ substream = list_entry(list, struct snd_rawmidi_substream, list);
+ sprintf(substream->name,
+ "Miditerminal %d", substream->number+1);
+ }
+ /* input */
+ list_for_each(list,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) {
+ substream = list_entry(list, struct snd_rawmidi_substream, list);
+ mts->midi_input_substream[substream->number] = substream;
+ switch(substream->number) {
+ case MTS64_SMPTE_SUBSTREAM:
+ strcpy(substream->name, "Miditerminal SMPTE");
+ break;
+ default:
+ sprintf(substream->name,
+ "Miditerminal %d", substream->number+1);
+ }
+ }
+
+ /* controls */
+ err = snd_mts64_ctl_create(card, mts);
+
+ return err;
+}
+
+/*********************************************************************
+ * parport stuff
+ *********************************************************************/
+static void snd_mts64_interrupt(void *private)
+{
+ struct mts64 *mts = ((struct snd_card*)private)->private_data;
+ u16 ret;
+ u8 status, data;
+ struct snd_rawmidi_substream *substream;
+
+ spin_lock(&mts->lock);
+ ret = mts64_read(mts->pardev->port);
+ data = ret & 0x00ff;
+ status = ret >> 8;
+
+ if (status & MTS64_STAT_PORT) {
+ mts->current_midi_input_port = mts64_map_midi_input(data);
+ } else {
+ if (mts->current_midi_input_port == -1)
+ goto __out;
+ substream = mts->midi_input_substream[mts->current_midi_input_port];
+ if (mts->mode[substream->number] & MTS64_MODE_INPUT_TRIGGERED)
+ snd_rawmidi_receive(substream, &data, 1);
+ }
+__out:
+ spin_unlock(&mts->lock);
+}
+
+static int __devinit snd_mts64_probe_port(struct parport *p)
+{
+ struct pardevice *pardev;
+ int res;
+
+ pardev = parport_register_device(p, DRIVER_NAME,
+ NULL, NULL, NULL,
+ 0, NULL);
+ if (!pardev)
+ return -EIO;
+
+ if (parport_claim(pardev)) {
+ parport_unregister_device(pardev);
+ return -EIO;
+ }
+
+ res = mts64_probe(p);
+
+ parport_release(pardev);
+ parport_unregister_device(pardev);
+
+ return res;
+}
+
+static void __devinit snd_mts64_attach(struct parport *p)
+{
+ struct platform_device *device;
+
+ device = platform_device_alloc(PLATFORM_DRIVER, device_count);
+ if (!device)
+ return;
+
+ /* Temporary assignment to forward the parport */
+ platform_set_drvdata(device, p);
+
+ if (platform_device_add(device) < 0) {
+ platform_device_put(device);
+ return;
+ }
+
+ /* Since we dont get the return value of probe
+ * We need to check if device probing succeeded or not */
+ if (!platform_get_drvdata(device)) {
+ platform_device_unregister(device);
+ return;
+ }
+
+ /* register device in global table */
+ platform_devices[device_count] = device;
+ device_count++;
+}
+
+static void snd_mts64_detach(struct parport *p)
+{
+ /* nothing to do here */
+}
+
+static struct parport_driver mts64_parport_driver = {
+ .name = "mts64",
+ .attach = snd_mts64_attach,
+ .detach = snd_mts64_detach
+};
+
+/*********************************************************************
+ * platform stuff
+ *********************************************************************/
+static void snd_mts64_card_private_free(struct snd_card *card)
+{
+ struct mts64 *mts = card->private_data;
+ struct pardevice *pardev = mts->pardev;
+
+ if (pardev) {
+ if (mts->pardev_claimed)
+ parport_release(pardev);
+ parport_unregister_device(pardev);
+ }
+
+ snd_mts64_free(mts);
+}
+
+static int __devinit snd_mts64_probe(struct platform_device *pdev)
+{
+ struct pardevice *pardev;
+ struct parport *p;
+ int dev = pdev->id;
+ struct snd_card *card = NULL;
+ struct mts64 *mts = NULL;
+ int err;
+
+ p = platform_get_drvdata(pdev);
+ platform_set_drvdata(pdev, NULL);
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev])
+ return -ENOENT;
+ if ((err = snd_mts64_probe_port(p)) < 0)
+ return err;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL) {
+ snd_printd("Cannot create card\n");
+ return -ENOMEM;
+ }
+ strcpy(card->driver, DRIVER_NAME);
+ strcpy(card->shortname, "ESI " CARD_NAME);
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, p->base, p->irq);
+
+ pardev = parport_register_device(p, /* port */
+ DRIVER_NAME, /* name */
+ NULL, /* preempt */
+ NULL, /* wakeup */
+ snd_mts64_interrupt, /* ISR */
+ PARPORT_DEV_EXCL, /* flags */
+ (void *)card); /* private */
+ if (pardev == NULL) {
+ snd_printd("Cannot register pardevice\n");
+ err = -EIO;
+ goto __err;
+ }
+
+ if ((err = snd_mts64_create(card, pardev, &mts)) < 0) {
+ snd_printd("Cannot create main component\n");
+ parport_unregister_device(pardev);
+ goto __err;
+ }
+ card->private_data = mts;
+ card->private_free = snd_mts64_card_private_free;
+
+ if ((err = snd_mts64_rawmidi_create(card)) < 0) {
+ snd_printd("Creating Rawmidi component failed\n");
+ goto __err;
+ }
+
+ /* claim parport */
+ if (parport_claim(pardev)) {
+ snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base);
+ err = -EIO;
+ goto __err;
+ }
+ mts->pardev_claimed = 1;
+
+ /* init device */
+ if ((err = mts64_device_init(p)) < 0)
+ goto __err;
+
+ platform_set_drvdata(pdev, card);
+
+ snd_card_set_dev(card, &pdev->dev);
+
+ /* At this point card will be usable */
+ if ((err = snd_card_register(card)) < 0) {
+ snd_printd("Cannot register card\n");
+ goto __err;
+ }
+
+ snd_printk("ESI Miditerminal 4140 on 0x%lx\n", p->base);
+ return 0;
+
+__err:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_mts64_remove(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ if (card)
+ snd_card_free(card);
+
+ return 0;
+}
+
+
+static struct platform_driver snd_mts64_driver = {
+ .probe = snd_mts64_probe,
+ .remove = __devexit_p(snd_mts64_remove),
+ .driver = {
+ .name = PLATFORM_DRIVER
+ }
+};
+
+/*********************************************************************
+ * module init stuff
+ *********************************************************************/
+static void snd_mts64_unregister_all(void)
+{
+ int i;
+
+ for (i = 0; i < SNDRV_CARDS; ++i) {
+ if (platform_devices[i]) {
+ platform_device_unregister(platform_devices[i]);
+ platform_devices[i] = NULL;
+ }
+ }
+ platform_driver_unregister(&snd_mts64_driver);
+ parport_unregister_driver(&mts64_parport_driver);
+}
+
+static int __init snd_mts64_module_init(void)
+{
+ int err;
+
+ if ((err = platform_driver_register(&snd_mts64_driver)) < 0)
+ return err;
+
+ if (parport_register_driver(&mts64_parport_driver) != 0) {
+ platform_driver_unregister(&snd_mts64_driver);
+ return -EIO;
+ }
+
+ if (device_count == 0) {
+ snd_mts64_unregister_all();
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static void __exit snd_mts64_module_exit(void)
+{
+ snd_mts64_unregister_all();
+}
+
+module_init(snd_mts64_module_init);
+module_exit(snd_mts64_module_exit);
diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile
new file mode 100644
index 0000000..19767a6
--- /dev/null
+++ b/sound/drivers/opl3/Makefile
@@ -0,0 +1,20 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-opl3-lib-objs := opl3_lib.o opl3_synth.o
+snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o
+snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o
+
+#
+# this function returns:
+# "m" - CONFIG_SND_SEQUENCER is m
+# <empty string> - CONFIG_SND_SEQUENCER is undefined
+# otherwise parameter #1 value
+#
+sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+
+obj-$(CONFIG_SND_OPL3_LIB) += snd-opl3-lib.o
+obj-$(CONFIG_SND_OPL4_LIB) += snd-opl3-lib.o
+obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-opl3-synth.o
diff --git a/sound/drivers/opl3/opl3_drums.c b/sound/drivers/opl3/opl3_drums.c
new file mode 100644
index 0000000..7369438
--- /dev/null
+++ b/sound/drivers/opl3/opl3_drums.c
@@ -0,0 +1,226 @@
+/*
+ * Copyright (c) by Uros Bizjak <uros@kss-loka.si>
+ *
+ * OPL2/OPL3/OPL4 FM routines for internal percussion channels
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include "opl3_voice.h"
+
+extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
+
+static char snd_opl3_drum_table[47] =
+{
+ OPL3_BASSDRUM_ON, OPL3_BASSDRUM_ON, OPL3_HIHAT_ON, /* 35 - 37 */
+ OPL3_SNAREDRUM_ON, OPL3_HIHAT_ON, OPL3_SNAREDRUM_ON, /* 38 - 40 */
+ OPL3_BASSDRUM_ON, OPL3_HIHAT_ON, OPL3_BASSDRUM_ON, /* 41 - 43 */
+ OPL3_HIHAT_ON, OPL3_TOMTOM_ON, OPL3_HIHAT_ON, /* 44 - 46 */
+ OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, OPL3_CYMBAL_ON, /* 47 - 49 */
+
+ OPL3_TOMTOM_ON, OPL3_CYMBAL_ON, OPL3_CYMBAL_ON, /* 50 - 52 */
+ OPL3_CYMBAL_ON, OPL3_CYMBAL_ON, OPL3_CYMBAL_ON, /* 53 - 55 */
+ OPL3_HIHAT_ON, OPL3_CYMBAL_ON, OPL3_TOMTOM_ON, /* 56 - 58 */
+ OPL3_CYMBAL_ON, OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, /* 59 - 61 */
+ OPL3_HIHAT_ON, OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, /* 62 - 64 */
+
+ OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, /* 65 - 67 */
+ OPL3_TOMTOM_ON, OPL3_HIHAT_ON, OPL3_HIHAT_ON, /* 68 - 70 */
+ OPL3_HIHAT_ON, OPL3_HIHAT_ON, OPL3_TOMTOM_ON, /* 71 - 73 */
+ OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, /* 74 - 76 */
+ OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, OPL3_TOMTOM_ON, /* 77 - 79 */
+ OPL3_CYMBAL_ON, OPL3_CYMBAL_ON /* 80 - 81 */
+};
+
+struct snd_opl3_drum_voice {
+ int voice;
+ int op;
+ unsigned char am_vib;
+ unsigned char ksl_level;
+ unsigned char attack_decay;
+ unsigned char sustain_release;
+ unsigned char feedback_connection;
+ unsigned char wave_select;
+};
+
+struct snd_opl3_drum_note {
+ int voice;
+ unsigned char fnum;
+ unsigned char octave_f;
+ unsigned char feedback_connection;
+};
+
+static struct snd_opl3_drum_voice bass_op0 = {6, 0, 0x00, 0x32, 0xf8, 0x66, 0x30, 0x00};
+static struct snd_opl3_drum_voice bass_op1 = {6, 1, 0x00, 0x03, 0xf6, 0x57, 0x30, 0x00};
+static struct snd_opl3_drum_note bass_note = {6, 0x90, 0x09};
+
+static struct snd_opl3_drum_voice hihat = {7, 0, 0x00, 0x03, 0xf0, 0x06, 0x20, 0x00};
+
+static struct snd_opl3_drum_voice snare = {7, 1, 0x00, 0x03, 0xf0, 0x07, 0x20, 0x02};
+static struct snd_opl3_drum_note snare_note = {7, 0xf4, 0x0d};
+
+static struct snd_opl3_drum_voice tomtom = {8, 0, 0x02, 0x03, 0xf0, 0x06, 0x10, 0x00};
+static struct snd_opl3_drum_note tomtom_note = {8, 0xf4, 0x09};
+
+static struct snd_opl3_drum_voice cymbal = {8, 1, 0x04, 0x03, 0xf0, 0x06, 0x10, 0x00};
+
+/*
+ * set drum voice characteristics
+ */
+static void snd_opl3_drum_voice_set(struct snd_opl3 *opl3,
+ struct snd_opl3_drum_voice *data)
+{
+ unsigned char op_offset = snd_opl3_regmap[data->voice][data->op];
+ unsigned char voice_offset = data->voice;
+ unsigned short opl3_reg;
+
+ /* Set OPL3 AM_VIB register */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_AM_VIB + op_offset);
+ opl3->command(opl3, opl3_reg, data->am_vib);
+
+ /* Set OPL3 KSL_LEVEL register */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_KSL_LEVEL + op_offset);
+ opl3->command(opl3, opl3_reg, data->ksl_level);
+
+ /* Set OPL3 ATTACK_DECAY register */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_ATTACK_DECAY + op_offset);
+ opl3->command(opl3, opl3_reg, data->attack_decay);
+
+ /* Set OPL3 SUSTAIN_RELEASE register */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_SUSTAIN_RELEASE + op_offset);
+ opl3->command(opl3, opl3_reg, data->sustain_release);
+
+ /* Set OPL3 FEEDBACK_CONNECTION register */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_FEEDBACK_CONNECTION + voice_offset);
+ opl3->command(opl3, opl3_reg, data->feedback_connection);
+
+ /* Select waveform */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_WAVE_SELECT + op_offset);
+ opl3->command(opl3, opl3_reg, data->wave_select);
+}
+
+/*
+ * Set drum voice pitch
+ */
+static void snd_opl3_drum_note_set(struct snd_opl3 *opl3,
+ struct snd_opl3_drum_note *data)
+{
+ unsigned char voice_offset = data->voice;
+ unsigned short opl3_reg;
+
+ /* Set OPL3 FNUM_LOW register */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_FNUM_LOW + voice_offset);
+ opl3->command(opl3, opl3_reg, data->fnum);
+
+ /* Set OPL3 KEYON_BLOCK register */
+ opl3_reg = OPL3_LEFT | (OPL3_REG_KEYON_BLOCK + voice_offset);
+ opl3->command(opl3, opl3_reg, data->octave_f);
+}
+
+/*
+ * Set drum voice volume and position
+ */
+static void snd_opl3_drum_vol_set(struct snd_opl3 *opl3,
+ struct snd_opl3_drum_voice *data,
+ int vel, struct snd_midi_channel *chan)
+{
+ unsigned char op_offset = snd_opl3_regmap[data->voice][data->op];
+ unsigned char voice_offset = data->voice;
+ unsigned char reg_val;
+ unsigned short opl3_reg;
+
+ /* Set OPL3 KSL_LEVEL register */
+ reg_val = data->ksl_level;
+ snd_opl3_calc_volume(&reg_val, vel, chan);
+ opl3_reg = OPL3_LEFT | (OPL3_REG_KSL_LEVEL + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set OPL3 FEEDBACK_CONNECTION register */
+ /* Set output voice connection */
+ reg_val = data->feedback_connection | OPL3_STEREO_BITS;
+ if (chan->gm_pan < 43)
+ reg_val &= ~OPL3_VOICE_TO_RIGHT;
+ if (chan->gm_pan > 85)
+ reg_val &= ~OPL3_VOICE_TO_LEFT;
+ opl3_reg = OPL3_LEFT | (OPL3_REG_FEEDBACK_CONNECTION + voice_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+}
+
+/*
+ * Loads drum voices at init time
+ */
+void snd_opl3_load_drums(struct snd_opl3 *opl3)
+{
+ snd_opl3_drum_voice_set(opl3, &bass_op0);
+ snd_opl3_drum_voice_set(opl3, &bass_op1);
+ snd_opl3_drum_note_set(opl3, &bass_note);
+
+ snd_opl3_drum_voice_set(opl3, &hihat);
+
+ snd_opl3_drum_voice_set(opl3, &snare);
+ snd_opl3_drum_note_set(opl3, &snare_note);
+
+ snd_opl3_drum_voice_set(opl3, &tomtom);
+ snd_opl3_drum_note_set(opl3, &tomtom_note);
+
+ snd_opl3_drum_voice_set(opl3, &cymbal);
+}
+
+/*
+ * Switch drum voice on or off
+ */
+void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int vel, int on_off,
+ struct snd_midi_channel *chan)
+{
+ unsigned char drum_mask;
+ struct snd_opl3_drum_voice *drum_voice;
+
+ if (!(opl3->drum_reg & OPL3_PERCUSSION_ENABLE))
+ return;
+
+ if ((note < 35) || (note > 81))
+ return;
+ drum_mask = snd_opl3_drum_table[note - 35];
+
+ if (on_off) {
+ switch (drum_mask) {
+ case OPL3_BASSDRUM_ON:
+ drum_voice = &bass_op1;
+ break;
+ case OPL3_HIHAT_ON:
+ drum_voice = &hihat;
+ break;
+ case OPL3_SNAREDRUM_ON:
+ drum_voice = &snare;
+ break;
+ case OPL3_TOMTOM_ON:
+ drum_voice = &tomtom;
+ break;
+ case OPL3_CYMBAL_ON:
+ drum_voice = &cymbal;
+ break;
+ default:
+ drum_voice = &tomtom;
+ }
+
+ snd_opl3_drum_vol_set(opl3, drum_voice, vel, chan);
+ opl3->drum_reg |= drum_mask;
+ } else {
+ opl3->drum_reg &= ~drum_mask;
+ }
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_PERCUSSION,
+ opl3->drum_reg);
+}
diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c
new file mode 100644
index 0000000..7805823
--- /dev/null
+++ b/sound/drivers/opl3/opl3_lib.c
@@ -0,0 +1,560 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Hannu Savolainen 1993-1996,
+ * Rob Hooft
+ *
+ * Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)
+ *
+ * Most if code is ported from OSS/Lite.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/opl3.h>
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <sound/minors.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Hannu Savolainen 1993-1996, Rob Hooft");
+MODULE_DESCRIPTION("Routines for control of AdLib FM cards (OPL2/OPL3/OPL4 chips)");
+MODULE_LICENSE("GPL");
+
+extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
+
+static void snd_opl2_command(struct snd_opl3 * opl3, unsigned short cmd, unsigned char val)
+{
+ unsigned long flags;
+ unsigned long port;
+
+ /*
+ * The original 2-OP synth requires a quite long delay
+ * after writing to a register.
+ */
+
+ port = (cmd & OPL3_RIGHT) ? opl3->r_port : opl3->l_port;
+
+ spin_lock_irqsave(&opl3->reg_lock, flags);
+
+ outb((unsigned char) cmd, port);
+ udelay(10);
+
+ outb((unsigned char) val, port + 1);
+ udelay(30);
+
+ spin_unlock_irqrestore(&opl3->reg_lock, flags);
+}
+
+static void snd_opl3_command(struct snd_opl3 * opl3, unsigned short cmd, unsigned char val)
+{
+ unsigned long flags;
+ unsigned long port;
+
+ /*
+ * The OPL-3 survives with just two INBs
+ * after writing to a register.
+ */
+
+ port = (cmd & OPL3_RIGHT) ? opl3->r_port : opl3->l_port;
+
+ spin_lock_irqsave(&opl3->reg_lock, flags);
+
+ outb((unsigned char) cmd, port);
+ inb(opl3->l_port);
+ inb(opl3->l_port);
+
+ outb((unsigned char) val, port + 1);
+ inb(opl3->l_port);
+ inb(opl3->l_port);
+
+ spin_unlock_irqrestore(&opl3->reg_lock, flags);
+}
+
+static int snd_opl3_detect(struct snd_opl3 * opl3)
+{
+ /*
+ * This function returns 1 if the FM chip is present at the given I/O port
+ * The detection algorithm plays with the timer built in the FM chip and
+ * looks for a change in the status register.
+ *
+ * Note! The timers of the FM chip are not connected to AdLib (and compatible)
+ * boards.
+ *
+ * Note2! The chip is initialized if detected.
+ */
+
+ unsigned char stat1, stat2, signature;
+
+ /* Reset timers 1 and 2 */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, OPL3_TIMER1_MASK | OPL3_TIMER2_MASK);
+ /* Reset the IRQ of the FM chip */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, OPL3_IRQ_RESET);
+ signature = stat1 = inb(opl3->l_port); /* Status register */
+ if ((stat1 & 0xe0) != 0x00) { /* Should be 0x00 */
+ snd_printd("OPL3: stat1 = 0x%x\n", stat1);
+ return -ENODEV;
+ }
+ /* Set timer1 to 0xff */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER1, 0xff);
+ /* Unmask and start timer 1 */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, OPL3_TIMER2_MASK | OPL3_TIMER1_START);
+ /* Now we have to delay at least 80us */
+ udelay(200);
+ /* Read status after timers have expired */
+ stat2 = inb(opl3->l_port);
+ /* Stop the timers */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, OPL3_TIMER1_MASK | OPL3_TIMER2_MASK);
+ /* Reset the IRQ of the FM chip */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, OPL3_IRQ_RESET);
+ if ((stat2 & 0xe0) != 0xc0) { /* There is no YM3812 */
+ snd_printd("OPL3: stat2 = 0x%x\n", stat2);
+ return -ENODEV;
+ }
+
+ /* If the toplevel code knows exactly the type of chip, don't try
+ to detect it. */
+ if (opl3->hardware != OPL3_HW_AUTO)
+ return 0;
+
+ /* There is a FM chip on this address. Detect the type (OPL2 to OPL4) */
+ if (signature == 0x06) { /* OPL2 */
+ opl3->hardware = OPL3_HW_OPL2;
+ } else {
+ /*
+ * If we had an OPL4 chip, opl3->hardware would have been set
+ * by the OPL4 driver; so we can assume OPL3 here.
+ */
+ if (snd_BUG_ON(!opl3->r_port))
+ return -ENODEV;
+ opl3->hardware = OPL3_HW_OPL3;
+ }
+ return 0;
+}
+
+/*
+ * AdLib timers
+ */
+
+/*
+ * Timer 1 - 80us
+ */
+
+static int snd_opl3_timer1_start(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ unsigned int ticks;
+ struct snd_opl3 *opl3;
+
+ opl3 = snd_timer_chip(timer);
+ spin_lock_irqsave(&opl3->timer_lock, flags);
+ ticks = timer->sticks;
+ tmp = (opl3->timer_enable | OPL3_TIMER1_START) & ~OPL3_TIMER1_MASK;
+ opl3->timer_enable = tmp;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER1, 256 - ticks); /* timer 1 count */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, tmp); /* enable timer 1 IRQ */
+ spin_unlock_irqrestore(&opl3->timer_lock, flags);
+ return 0;
+}
+
+static int snd_opl3_timer1_stop(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ struct snd_opl3 *opl3;
+
+ opl3 = snd_timer_chip(timer);
+ spin_lock_irqsave(&opl3->timer_lock, flags);
+ tmp = (opl3->timer_enable | OPL3_TIMER1_MASK) & ~OPL3_TIMER1_START;
+ opl3->timer_enable = tmp;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, tmp); /* disable timer #1 */
+ spin_unlock_irqrestore(&opl3->timer_lock, flags);
+ return 0;
+}
+
+/*
+ * Timer 2 - 320us
+ */
+
+static int snd_opl3_timer2_start(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ unsigned int ticks;
+ struct snd_opl3 *opl3;
+
+ opl3 = snd_timer_chip(timer);
+ spin_lock_irqsave(&opl3->timer_lock, flags);
+ ticks = timer->sticks;
+ tmp = (opl3->timer_enable | OPL3_TIMER2_START) & ~OPL3_TIMER2_MASK;
+ opl3->timer_enable = tmp;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER2, 256 - ticks); /* timer 1 count */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, tmp); /* enable timer 1 IRQ */
+ spin_unlock_irqrestore(&opl3->timer_lock, flags);
+ return 0;
+}
+
+static int snd_opl3_timer2_stop(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ struct snd_opl3 *opl3;
+
+ opl3 = snd_timer_chip(timer);
+ spin_lock_irqsave(&opl3->timer_lock, flags);
+ tmp = (opl3->timer_enable | OPL3_TIMER2_MASK) & ~OPL3_TIMER2_START;
+ opl3->timer_enable = tmp;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, tmp); /* disable timer #1 */
+ spin_unlock_irqrestore(&opl3->timer_lock, flags);
+ return 0;
+}
+
+/*
+
+ */
+
+static struct snd_timer_hardware snd_opl3_timer1 =
+{
+ .flags = SNDRV_TIMER_HW_STOP,
+ .resolution = 80000,
+ .ticks = 256,
+ .start = snd_opl3_timer1_start,
+ .stop = snd_opl3_timer1_stop,
+};
+
+static struct snd_timer_hardware snd_opl3_timer2 =
+{
+ .flags = SNDRV_TIMER_HW_STOP,
+ .resolution = 320000,
+ .ticks = 256,
+ .start = snd_opl3_timer2_start,
+ .stop = snd_opl3_timer2_stop,
+};
+
+static int snd_opl3_timer1_init(struct snd_opl3 * opl3, int timer_no)
+{
+ struct snd_timer *timer = NULL;
+ struct snd_timer_id tid;
+ int err;
+
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = opl3->card->number;
+ tid.device = timer_no;
+ tid.subdevice = 0;
+ if ((err = snd_timer_new(opl3->card, "AdLib timer #1", &tid, &timer)) >= 0) {
+ strcpy(timer->name, "AdLib timer #1");
+ timer->private_data = opl3;
+ timer->hw = snd_opl3_timer1;
+ }
+ opl3->timer1 = timer;
+ return err;
+}
+
+static int snd_opl3_timer2_init(struct snd_opl3 * opl3, int timer_no)
+{
+ struct snd_timer *timer = NULL;
+ struct snd_timer_id tid;
+ int err;
+
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = opl3->card->number;
+ tid.device = timer_no;
+ tid.subdevice = 0;
+ if ((err = snd_timer_new(opl3->card, "AdLib timer #2", &tid, &timer)) >= 0) {
+ strcpy(timer->name, "AdLib timer #2");
+ timer->private_data = opl3;
+ timer->hw = snd_opl3_timer2;
+ }
+ opl3->timer2 = timer;
+ return err;
+}
+
+/*
+
+ */
+
+void snd_opl3_interrupt(struct snd_hwdep * hw)
+{
+ unsigned char status;
+ struct snd_opl3 *opl3;
+ struct snd_timer *timer;
+
+ if (hw == NULL)
+ return;
+
+ opl3 = hw->private_data;
+ status = inb(opl3->l_port);
+#if 0
+ snd_printk("AdLib IRQ status = 0x%x\n", status);
+#endif
+ if (!(status & 0x80))
+ return;
+
+ if (status & 0x40) {
+ timer = opl3->timer1;
+ snd_timer_interrupt(timer, timer->sticks);
+ }
+ if (status & 0x20) {
+ timer = opl3->timer2;
+ snd_timer_interrupt(timer, timer->sticks);
+ }
+}
+
+EXPORT_SYMBOL(snd_opl3_interrupt);
+
+/*
+
+ */
+
+static int snd_opl3_free(struct snd_opl3 *opl3)
+{
+ if (snd_BUG_ON(!opl3))
+ return -ENXIO;
+ if (opl3->private_free)
+ opl3->private_free(opl3);
+ snd_opl3_clear_patches(opl3);
+ release_and_free_resource(opl3->res_l_port);
+ release_and_free_resource(opl3->res_r_port);
+ kfree(opl3);
+ return 0;
+}
+
+static int snd_opl3_dev_free(struct snd_device *device)
+{
+ struct snd_opl3 *opl3 = device->device_data;
+ return snd_opl3_free(opl3);
+}
+
+int snd_opl3_new(struct snd_card *card,
+ unsigned short hardware,
+ struct snd_opl3 **ropl3)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_opl3_dev_free,
+ };
+ struct snd_opl3 *opl3;
+ int err;
+
+ *ropl3 = NULL;
+ opl3 = kzalloc(sizeof(*opl3), GFP_KERNEL);
+ if (opl3 == NULL) {
+ snd_printk(KERN_ERR "opl3: cannot allocate\n");
+ return -ENOMEM;
+ }
+
+ opl3->card = card;
+ opl3->hardware = hardware;
+ spin_lock_init(&opl3->reg_lock);
+ spin_lock_init(&opl3->timer_lock);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_CODEC, opl3, &ops)) < 0) {
+ snd_opl3_free(opl3);
+ return err;
+ }
+
+ *ropl3 = opl3;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_opl3_new);
+
+int snd_opl3_init(struct snd_opl3 *opl3)
+{
+ if (! opl3->command) {
+ printk(KERN_ERR "snd_opl3_init: command not defined!\n");
+ return -EINVAL;
+ }
+
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TEST, OPL3_ENABLE_WAVE_SELECT);
+ /* Melodic mode */
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_PERCUSSION, 0x00);
+
+ switch (opl3->hardware & OPL3_HW_MASK) {
+ case OPL3_HW_OPL2:
+ opl3->max_voices = MAX_OPL2_VOICES;
+ break;
+ case OPL3_HW_OPL3:
+ case OPL3_HW_OPL4:
+ opl3->max_voices = MAX_OPL3_VOICES;
+ /* Enter OPL3 mode */
+ opl3->command(opl3, OPL3_RIGHT | OPL3_REG_MODE, OPL3_OPL3_ENABLE);
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_opl3_init);
+
+int snd_opl3_create(struct snd_card *card,
+ unsigned long l_port,
+ unsigned long r_port,
+ unsigned short hardware,
+ int integrated,
+ struct snd_opl3 ** ropl3)
+{
+ struct snd_opl3 *opl3;
+ int err;
+
+ *ropl3 = NULL;
+ if ((err = snd_opl3_new(card, hardware, &opl3)) < 0)
+ return err;
+ if (! integrated) {
+ if ((opl3->res_l_port = request_region(l_port, 2, "OPL2/3 (left)")) == NULL) {
+ snd_printk(KERN_ERR "opl3: can't grab left port 0x%lx\n", l_port);
+ snd_device_free(card, opl3);
+ return -EBUSY;
+ }
+ if (r_port != 0 &&
+ (opl3->res_r_port = request_region(r_port, 2, "OPL2/3 (right)")) == NULL) {
+ snd_printk(KERN_ERR "opl3: can't grab right port 0x%lx\n", r_port);
+ snd_device_free(card, opl3);
+ return -EBUSY;
+ }
+ }
+ opl3->l_port = l_port;
+ opl3->r_port = r_port;
+
+ switch (opl3->hardware) {
+ /* some hardware doesn't support timers */
+ case OPL3_HW_OPL3_SV:
+ case OPL3_HW_OPL3_CS:
+ case OPL3_HW_OPL3_FM801:
+ opl3->command = &snd_opl3_command;
+ break;
+ default:
+ opl3->command = &snd_opl2_command;
+ if ((err = snd_opl3_detect(opl3)) < 0) {
+ snd_printd("OPL2/3 chip not detected at 0x%lx/0x%lx\n",
+ opl3->l_port, opl3->r_port);
+ snd_device_free(card, opl3);
+ return err;
+ }
+ /* detect routine returns correct hardware type */
+ switch (opl3->hardware & OPL3_HW_MASK) {
+ case OPL3_HW_OPL3:
+ case OPL3_HW_OPL4:
+ opl3->command = &snd_opl3_command;
+ }
+ }
+
+ snd_opl3_init(opl3);
+
+ *ropl3 = opl3;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_opl3_create);
+
+int snd_opl3_timer_new(struct snd_opl3 * opl3, int timer1_dev, int timer2_dev)
+{
+ int err;
+
+ if (timer1_dev >= 0)
+ if ((err = snd_opl3_timer1_init(opl3, timer1_dev)) < 0)
+ return err;
+ if (timer2_dev >= 0) {
+ if ((err = snd_opl3_timer2_init(opl3, timer2_dev)) < 0) {
+ snd_device_free(opl3->card, opl3->timer1);
+ opl3->timer1 = NULL;
+ return err;
+ }
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_opl3_timer_new);
+
+int snd_opl3_hwdep_new(struct snd_opl3 * opl3,
+ int device, int seq_device,
+ struct snd_hwdep ** rhwdep)
+{
+ struct snd_hwdep *hw;
+ struct snd_card *card = opl3->card;
+ int err;
+
+ if (rhwdep)
+ *rhwdep = NULL;
+
+ /* create hardware dependent device (direct FM) */
+
+ if ((err = snd_hwdep_new(card, "OPL2/OPL3", device, &hw)) < 0) {
+ snd_device_free(card, opl3);
+ return err;
+ }
+ hw->private_data = opl3;
+ hw->exclusive = 1;
+#ifdef CONFIG_SND_OSSEMUL
+ if (device == 0) {
+ hw->oss_type = SNDRV_OSS_DEVICE_TYPE_DMFM;
+ sprintf(hw->oss_dev, "dmfm%i", card->number);
+ }
+#endif
+ strcpy(hw->name, hw->id);
+ switch (opl3->hardware & OPL3_HW_MASK) {
+ case OPL3_HW_OPL2:
+ strcpy(hw->name, "OPL2 FM");
+ hw->iface = SNDRV_HWDEP_IFACE_OPL2;
+ break;
+ case OPL3_HW_OPL3:
+ strcpy(hw->name, "OPL3 FM");
+ hw->iface = SNDRV_HWDEP_IFACE_OPL3;
+ break;
+ case OPL3_HW_OPL4:
+ strcpy(hw->name, "OPL4 FM");
+ hw->iface = SNDRV_HWDEP_IFACE_OPL4;
+ break;
+ }
+
+ /* operators - only ioctl */
+ hw->ops.open = snd_opl3_open;
+ hw->ops.ioctl = snd_opl3_ioctl;
+ hw->ops.write = snd_opl3_write;
+ hw->ops.release = snd_opl3_release;
+
+ opl3->hwdep = hw;
+ opl3->seq_dev_num = seq_device;
+#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))
+ if (snd_seq_device_new(card, seq_device, SNDRV_SEQ_DEV_ID_OPL3,
+ sizeof(struct snd_opl3 *), &opl3->seq_dev) >= 0) {
+ strcpy(opl3->seq_dev->name, hw->name);
+ *(struct snd_opl3 **)SNDRV_SEQ_DEVICE_ARGPTR(opl3->seq_dev) = opl3;
+ }
+#endif
+ if (rhwdep)
+ *rhwdep = hw;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_opl3_hwdep_new);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_opl3_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_opl3_exit(void)
+{
+}
+
+module_init(alsa_opl3_init)
+module_exit(alsa_opl3_exit)
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
new file mode 100644
index 0000000..16feafa
--- /dev/null
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -0,0 +1,869 @@
+/*
+ * Copyright (c) by Uros Bizjak <uros@kss-loka.si>
+ *
+ * Midi synth routines for OPL2/OPL3/OPL4 FM
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#undef DEBUG_ALLOC
+#undef DEBUG_MIDI
+
+#include "opl3_voice.h"
+#include <sound/asoundef.h>
+
+extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
+
+extern int use_internal_drums;
+
+/*
+ * The next table looks magical, but it certainly is not. Its values have
+ * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception
+ * for i=0. This log-table converts a linear volume-scaling (0..127) to a
+ * logarithmic scaling as present in the FM-synthesizer chips. so : Volume
+ * 64 = 0 db = relative volume 0 and: Volume 32 = -6 db = relative
+ * volume -8 it was implemented as a table because it is only 128 bytes and
+ * it saves a lot of log() calculations. (Rob Hooft <hooft@chem.ruu.nl>)
+ */
+
+static char opl3_volume_table[128] =
+{
+ -63, -48, -40, -35, -32, -29, -27, -26,
+ -24, -23, -21, -20, -19, -18, -18, -17,
+ -16, -15, -15, -14, -13, -13, -12, -12,
+ -11, -11, -10, -10, -10, -9, -9, -8,
+ -8, -8, -7, -7, -7, -6, -6, -6,
+ -5, -5, -5, -5, -4, -4, -4, -4,
+ -3, -3, -3, -3, -2, -2, -2, -2,
+ -2, -1, -1, -1, -1, 0, 0, 0,
+ 0, 0, 0, 1, 1, 1, 1, 1,
+ 1, 2, 2, 2, 2, 2, 2, 2,
+ 3, 3, 3, 3, 3, 3, 3, 4,
+ 4, 4, 4, 4, 4, 4, 4, 5,
+ 5, 5, 5, 5, 5, 5, 5, 5,
+ 6, 6, 6, 6, 6, 6, 6, 6,
+ 6, 7, 7, 7, 7, 7, 7, 7,
+ 7, 7, 7, 8, 8, 8, 8, 8
+};
+
+void snd_opl3_calc_volume(unsigned char *volbyte, int vel,
+ struct snd_midi_channel *chan)
+{
+ int oldvol, newvol, n;
+ int volume;
+
+ volume = (vel * chan->gm_volume * chan->gm_expression) / (127*127);
+ if (volume > 127)
+ volume = 127;
+
+ oldvol = OPL3_TOTAL_LEVEL_MASK - (*volbyte & OPL3_TOTAL_LEVEL_MASK);
+
+ newvol = opl3_volume_table[volume] + oldvol;
+ if (newvol > OPL3_TOTAL_LEVEL_MASK)
+ newvol = OPL3_TOTAL_LEVEL_MASK;
+ else if (newvol < 0)
+ newvol = 0;
+
+ n = OPL3_TOTAL_LEVEL_MASK - (newvol & OPL3_TOTAL_LEVEL_MASK);
+
+ *volbyte = (*volbyte & OPL3_KSL_MASK) | (n & OPL3_TOTAL_LEVEL_MASK);
+}
+
+/*
+ * Converts the note frequency to block and fnum values for the FM chip
+ */
+static short opl3_note_table[16] =
+{
+ 305, 323, /* for pitch bending, -2 semitones */
+ 343, 363, 385, 408, 432, 458, 485, 514, 544, 577, 611, 647,
+ 686, 726 /* for pitch bending, +2 semitones */
+};
+
+static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum,
+ int note, struct snd_midi_channel *chan)
+{
+ int block = ((note / 12) & 0x07) - 1;
+ int idx = (note % 12) + 2;
+ int freq;
+
+ if (chan->midi_pitchbend) {
+ int pitchbend = chan->midi_pitchbend;
+ int segment;
+
+ if (pitchbend > 0x1FFF)
+ pitchbend = 0x1FFF;
+
+ segment = pitchbend / 0x1000;
+ freq = opl3_note_table[idx+segment];
+ freq += ((opl3_note_table[idx+segment+1] - freq) *
+ (pitchbend % 0x1000)) / 0x1000;
+ } else {
+ freq = opl3_note_table[idx];
+ }
+
+ *fnum = (unsigned char) freq;
+ *blocknum = ((freq >> 8) & OPL3_FNUM_HIGH_MASK) |
+ ((block << 2) & OPL3_BLOCKNUM_MASK);
+}
+
+
+#ifdef DEBUG_ALLOC
+static void debug_alloc(struct snd_opl3 *opl3, char *s, int voice) {
+ int i;
+ char *str = "x.24";
+
+ printk("time %.5i: %s [%.2i]: ", opl3->use_time, s, voice);
+ for (i = 0; i < opl3->max_voices; i++)
+ printk("%c", *(str + opl3->voices[i].state + 1));
+ printk("\n");
+}
+#endif
+
+/*
+ * Get a FM voice (channel) to play a note on.
+ */
+static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op,
+ struct snd_midi_channel *chan) {
+ int chan_4op_1; /* first voice for 4op instrument */
+ int chan_4op_2; /* second voice for 4op instrument */
+
+ struct snd_opl3_voice *vp, *vp2;
+ unsigned int voice_time;
+ int i;
+
+#ifdef DEBUG_ALLOC
+ char *alloc_type[3] = { "FREE ", "CHEAP ", "EXPENSIVE" };
+#endif
+
+ /* This is our "allocation cost" table */
+ enum {
+ FREE = 0, CHEAP, EXPENSIVE, END
+ };
+
+ /* Keeps track of what we are finding */
+ struct best {
+ unsigned int time;
+ int voice;
+ } best[END];
+ struct best *bp;
+
+ for (i = 0; i < END; i++) {
+ best[i].time = (unsigned int)(-1); /* XXX MAX_?INT really */;
+ best[i].voice = -1;
+ }
+
+ /* Look through all the channels for the most suitable. */
+ for (i = 0; i < opl3->max_voices; i++) {
+ vp = &opl3->voices[i];
+
+ if (vp->state == SNDRV_OPL3_ST_NOT_AVAIL)
+ /* skip unavailable channels, allocated by
+ drum voices or by bounded 4op voices) */
+ continue;
+
+ voice_time = vp->time;
+ bp = best;
+
+ chan_4op_1 = ((i < 3) || (i > 8 && i < 12));
+ chan_4op_2 = ((i > 2 && i < 6) || (i > 11 && i < 15));
+ if (instr_4op) {
+ /* allocate 4op voice */
+ /* skip channels unavailable to 4op instrument */
+ if (!chan_4op_1)
+ continue;
+
+ if (vp->state)
+ /* kill one voice, CHEAP */
+ bp++;
+ /* get state of bounded 2op channel
+ to be allocated for 4op instrument */
+ vp2 = &opl3->voices[i + 3];
+ if (vp2->state == SNDRV_OPL3_ST_ON_2OP) {
+ /* kill two voices, EXPENSIVE */
+ bp++;
+ voice_time = (voice_time > vp->time) ?
+ voice_time : vp->time;
+ }
+ } else {
+ /* allocate 2op voice */
+ if ((chan_4op_1) || (chan_4op_2))
+ /* use bounded channels for 2op, CHEAP */
+ bp++;
+ else if (vp->state)
+ /* kill one voice on 2op channel, CHEAP */
+ bp++;
+ /* raise kill cost to EXPENSIVE for all channels */
+ if (vp->state)
+ bp++;
+ }
+ if (voice_time < bp->time) {
+ bp->time = voice_time;
+ bp->voice = i;
+ }
+ }
+
+ for (i = 0; i < END; i++) {
+ if (best[i].voice >= 0) {
+#ifdef DEBUG_ALLOC
+ printk("%s %iop allocation on voice %i\n",
+ alloc_type[i], instr_4op ? 4 : 2,
+ best[i].voice);
+#endif
+ return best[i].voice;
+ }
+ }
+ /* not found */
+ return -1;
+}
+
+/* ------------------------------ */
+
+/*
+ * System timer interrupt function
+ */
+void snd_opl3_timer_func(unsigned long data)
+{
+
+ struct snd_opl3 *opl3 = (struct snd_opl3 *)data;
+ unsigned long flags;
+ int again = 0;
+ int i;
+
+ spin_lock_irqsave(&opl3->sys_timer_lock, flags);
+ for (i = 0; i < opl3->max_voices; i++) {
+ struct snd_opl3_voice *vp = &opl3->voices[i];
+ if (vp->state > 0 && vp->note_off_check) {
+ if (vp->note_off == jiffies)
+ snd_opl3_note_off(opl3, vp->note, 0, vp->chan);
+ else
+ again++;
+ }
+ }
+ if (again) {
+ opl3->tlist.expires = jiffies + 1; /* invoke again */
+ add_timer(&opl3->tlist);
+ } else {
+ opl3->sys_timer_status = 0;
+ }
+ spin_unlock_irqrestore(&opl3->sys_timer_lock, flags);
+}
+
+/*
+ * Start system timer
+ */
+static void snd_opl3_start_timer(struct snd_opl3 *opl3)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&opl3->sys_timer_lock, flags);
+ if (! opl3->sys_timer_status) {
+ opl3->tlist.expires = jiffies + 1;
+ add_timer(&opl3->tlist);
+ opl3->sys_timer_status = 1;
+ }
+ spin_unlock_irqrestore(&opl3->sys_timer_lock, flags);
+}
+
+/* ------------------------------ */
+
+
+static int snd_opl3_oss_map[MAX_OPL3_VOICES] = {
+ 0, 1, 2, 9, 10, 11, 6, 7, 8, 15, 16, 17, 3, 4 ,5, 12, 13, 14
+};
+
+/*
+ * Start a note.
+ */
+void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
+{
+ struct snd_opl3 *opl3;
+ int instr_4op;
+
+ int voice;
+ struct snd_opl3_voice *vp, *vp2;
+ unsigned short connect_mask;
+ unsigned char connection;
+ unsigned char vol_op[4];
+
+ int extra_prg = 0;
+
+ unsigned short reg_side;
+ unsigned char op_offset;
+ unsigned char voice_offset;
+ unsigned short opl3_reg;
+ unsigned char reg_val;
+ unsigned char prg, bank;
+
+ int key = note;
+ unsigned char fnum, blocknum;
+ int i;
+
+ struct fm_patch *patch;
+ struct fm_instrument *fm;
+ unsigned long flags;
+
+ opl3 = p;
+
+#ifdef DEBUG_MIDI
+ snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n",
+ chan->number, chan->midi_program, note, vel);
+#endif
+
+ /* in SYNTH mode, application takes care of voices */
+ /* in SEQ mode, drum voice numbers are notes on drum channel */
+ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) {
+ if (chan->drum_channel) {
+ /* percussion instruments are located in bank 128 */
+ bank = 128;
+ prg = note;
+ } else {
+ bank = chan->gm_bank_select;
+ prg = chan->midi_program;
+ }
+ } else {
+ /* Prepare for OSS mode */
+ if (chan->number >= MAX_OPL3_VOICES)
+ return;
+
+ /* OSS instruments are located in bank 127 */
+ bank = 127;
+ prg = chan->midi_program;
+ }
+
+ spin_lock_irqsave(&opl3->voice_lock, flags);
+
+ if (use_internal_drums) {
+ snd_opl3_drum_switch(opl3, note, vel, 1, chan);
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+ return;
+ }
+
+ __extra_prg:
+ patch = snd_opl3_find_patch(opl3, prg, bank, 0);
+ if (!patch) {
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+ return;
+ }
+
+ fm = &patch->inst;
+ switch (patch->type) {
+ case FM_PATCH_OPL2:
+ instr_4op = 0;
+ break;
+ case FM_PATCH_OPL3:
+ if (opl3->hardware >= OPL3_HW_OPL3) {
+ instr_4op = 1;
+ break;
+ }
+ default:
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+ return;
+ }
+#ifdef DEBUG_MIDI
+ snd_printk(" --> OPL%i instrument: %s\n",
+ instr_4op ? 3 : 2, patch->name);
+#endif
+ /* in SYNTH mode, application takes care of voices */
+ /* in SEQ mode, allocate voice on free OPL3 channel */
+ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) {
+ voice = opl3_get_voice(opl3, instr_4op, chan);
+ } else {
+ /* remap OSS voice */
+ voice = snd_opl3_oss_map[chan->number];
+ }
+
+ if (voice < MAX_OPL2_VOICES) {
+ /* Left register block for voices 0 .. 8 */
+ reg_side = OPL3_LEFT;
+ voice_offset = voice;
+ connect_mask = (OPL3_LEFT_4OP_0 << voice_offset) & 0x07;
+ } else {
+ /* Right register block for voices 9 .. 17 */
+ reg_side = OPL3_RIGHT;
+ voice_offset = voice - MAX_OPL2_VOICES;
+ connect_mask = (OPL3_RIGHT_4OP_0 << voice_offset) & 0x38;
+ }
+
+ /* kill voice on channel */
+ vp = &opl3->voices[voice];
+ if (vp->state > 0) {
+ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
+ reg_val = vp->keyon_reg & ~OPL3_KEYON_BIT;
+ opl3->command(opl3, opl3_reg, reg_val);
+ }
+ if (instr_4op) {
+ vp2 = &opl3->voices[voice + 3];
+ if (vp->state > 0) {
+ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK +
+ voice_offset + 3);
+ reg_val = vp->keyon_reg & ~OPL3_KEYON_BIT;
+ opl3->command(opl3, opl3_reg, reg_val);
+ }
+ }
+
+ /* set connection register */
+ if (instr_4op) {
+ if ((opl3->connection_reg ^ connect_mask) & connect_mask) {
+ opl3->connection_reg |= connect_mask;
+ /* set connection bit */
+ opl3_reg = OPL3_RIGHT | OPL3_REG_CONNECTION_SELECT;
+ opl3->command(opl3, opl3_reg, opl3->connection_reg);
+ }
+ } else {
+ if ((opl3->connection_reg ^ ~connect_mask) & connect_mask) {
+ opl3->connection_reg &= ~connect_mask;
+ /* clear connection bit */
+ opl3_reg = OPL3_RIGHT | OPL3_REG_CONNECTION_SELECT;
+ opl3->command(opl3, opl3_reg, opl3->connection_reg);
+ }
+ }
+
+#ifdef DEBUG_MIDI
+ snd_printk(" --> setting OPL3 connection: 0x%x\n",
+ opl3->connection_reg);
+#endif
+ /*
+ * calculate volume depending on connection
+ * between FM operators (see include/opl3.h)
+ */
+ for (i = 0; i < (instr_4op ? 4 : 2); i++)
+ vol_op[i] = fm->op[i].ksl_level;
+
+ connection = fm->feedback_connection[0] & 0x01;
+ if (instr_4op) {
+ connection <<= 1;
+ connection |= fm->feedback_connection[1] & 0x01;
+
+ snd_opl3_calc_volume(&vol_op[3], vel, chan);
+ switch (connection) {
+ case 0x03:
+ snd_opl3_calc_volume(&vol_op[2], vel, chan);
+ /* fallthru */
+ case 0x02:
+ snd_opl3_calc_volume(&vol_op[0], vel, chan);
+ break;
+ case 0x01:
+ snd_opl3_calc_volume(&vol_op[1], vel, chan);
+ }
+ } else {
+ snd_opl3_calc_volume(&vol_op[1], vel, chan);
+ if (connection)
+ snd_opl3_calc_volume(&vol_op[0], vel, chan);
+ }
+
+ /* Program the FM voice characteristics */
+ for (i = 0; i < (instr_4op ? 4 : 2); i++) {
+#ifdef DEBUG_MIDI
+ snd_printk(" --> programming operator %i\n", i);
+#endif
+ op_offset = snd_opl3_regmap[voice_offset][i];
+
+ /* Set OPL3 AM_VIB register of requested voice/operator */
+ reg_val = fm->op[i].am_vib;
+ opl3_reg = reg_side | (OPL3_REG_AM_VIB + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set OPL3 KSL_LEVEL register of requested voice/operator */
+ reg_val = vol_op[i];
+ opl3_reg = reg_side | (OPL3_REG_KSL_LEVEL + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set OPL3 ATTACK_DECAY register of requested voice/operator */
+ reg_val = fm->op[i].attack_decay;
+ opl3_reg = reg_side | (OPL3_REG_ATTACK_DECAY + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set OPL3 SUSTAIN_RELEASE register of requested voice/operator */
+ reg_val = fm->op[i].sustain_release;
+ opl3_reg = reg_side | (OPL3_REG_SUSTAIN_RELEASE + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Select waveform */
+ reg_val = fm->op[i].wave_select;
+ opl3_reg = reg_side | (OPL3_REG_WAVE_SELECT + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+ }
+
+ /* Set operator feedback and 2op inter-operator connection */
+ reg_val = fm->feedback_connection[0];
+ /* Set output voice connection */
+ reg_val |= OPL3_STEREO_BITS;
+ if (chan->gm_pan < 43)
+ reg_val &= ~OPL3_VOICE_TO_RIGHT;
+ if (chan->gm_pan > 85)
+ reg_val &= ~OPL3_VOICE_TO_LEFT;
+ opl3_reg = reg_side | (OPL3_REG_FEEDBACK_CONNECTION + voice_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ if (instr_4op) {
+ /* Set 4op inter-operator connection */
+ reg_val = fm->feedback_connection[1] & OPL3_CONNECTION_BIT;
+ /* Set output voice connection */
+ reg_val |= OPL3_STEREO_BITS;
+ if (chan->gm_pan < 43)
+ reg_val &= ~OPL3_VOICE_TO_RIGHT;
+ if (chan->gm_pan > 85)
+ reg_val &= ~OPL3_VOICE_TO_LEFT;
+ opl3_reg = reg_side | (OPL3_REG_FEEDBACK_CONNECTION +
+ voice_offset + 3);
+ opl3->command(opl3, opl3_reg, reg_val);
+ }
+
+ /*
+ * Special treatment of percussion notes for fm:
+ * Requested pitch is really program, and pitch for
+ * device is whatever was specified in the patch library.
+ */
+ if (fm->fix_key)
+ note = fm->fix_key;
+ /*
+ * use transpose if defined in patch library
+ */
+ if (fm->trnsps)
+ note += (fm->trnsps - 64);
+
+ snd_opl3_calc_pitch(&fnum, &blocknum, note, chan);
+
+ /* Set OPL3 FNUM_LOW register of requested voice */
+ opl3_reg = reg_side | (OPL3_REG_FNUM_LOW + voice_offset);
+ opl3->command(opl3, opl3_reg, fnum);
+
+ opl3->voices[voice].keyon_reg = blocknum;
+
+ /* Set output sound flag */
+ blocknum |= OPL3_KEYON_BIT;
+
+#ifdef DEBUG_MIDI
+ snd_printk(" --> trigger voice %i\n", voice);
+#endif
+ /* Set OPL3 KEYON_BLOCK register of requested voice */
+ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
+ opl3->command(opl3, opl3_reg, blocknum);
+
+ /* kill note after fixed duration (in centiseconds) */
+ if (fm->fix_dur) {
+ opl3->voices[voice].note_off = jiffies +
+ (fm->fix_dur * HZ) / 100;
+ snd_opl3_start_timer(opl3);
+ opl3->voices[voice].note_off_check = 1;
+ } else
+ opl3->voices[voice].note_off_check = 0;
+
+ /* get extra pgm, but avoid possible loops */
+ extra_prg = (extra_prg) ? 0 : fm->modes;
+
+ /* do the bookkeeping */
+ vp->time = opl3->use_time++;
+ vp->note = key;
+ vp->chan = chan;
+
+ if (instr_4op) {
+ vp->state = SNDRV_OPL3_ST_ON_4OP;
+
+ vp2 = &opl3->voices[voice + 3];
+ vp2->time = opl3->use_time++;
+ vp2->note = key;
+ vp2->chan = chan;
+ vp2->state = SNDRV_OPL3_ST_NOT_AVAIL;
+ } else {
+ if (vp->state == SNDRV_OPL3_ST_ON_4OP) {
+ /* 4op killed by 2op, release bounded voice */
+ vp2 = &opl3->voices[voice + 3];
+ vp2->time = opl3->use_time++;
+ vp2->state = SNDRV_OPL3_ST_OFF;
+ }
+ vp->state = SNDRV_OPL3_ST_ON_2OP;
+ }
+
+#ifdef DEBUG_ALLOC
+ debug_alloc(opl3, "note on ", voice);
+#endif
+
+ /* allocate extra program if specified in patch library */
+ if (extra_prg) {
+ if (extra_prg > 128) {
+ bank = 128;
+ /* percussions start at 35 */
+ prg = extra_prg - 128 + 35 - 1;
+ } else {
+ bank = 0;
+ prg = extra_prg - 1;
+ }
+#ifdef DEBUG_MIDI
+ snd_printk(" *** allocating extra program\n");
+#endif
+ goto __extra_prg;
+ }
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+}
+
+static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice)
+{
+ unsigned short reg_side;
+ unsigned char voice_offset;
+ unsigned short opl3_reg;
+
+ struct snd_opl3_voice *vp, *vp2;
+
+ if (snd_BUG_ON(voice >= MAX_OPL3_VOICES))
+ return;
+
+ vp = &opl3->voices[voice];
+ if (voice < MAX_OPL2_VOICES) {
+ /* Left register block for voices 0 .. 8 */
+ reg_side = OPL3_LEFT;
+ voice_offset = voice;
+ } else {
+ /* Right register block for voices 9 .. 17 */
+ reg_side = OPL3_RIGHT;
+ voice_offset = voice - MAX_OPL2_VOICES;
+ }
+
+ /* kill voice */
+#ifdef DEBUG_MIDI
+ snd_printk(" --> kill voice %i\n", voice);
+#endif
+ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
+ /* clear Key ON bit */
+ opl3->command(opl3, opl3_reg, vp->keyon_reg);
+
+ /* do the bookkeeping */
+ vp->time = opl3->use_time++;
+
+ if (vp->state == SNDRV_OPL3_ST_ON_4OP) {
+ vp2 = &opl3->voices[voice + 3];
+
+ vp2->time = opl3->use_time++;
+ vp2->state = SNDRV_OPL3_ST_OFF;
+ }
+ vp->state = SNDRV_OPL3_ST_OFF;
+#ifdef DEBUG_ALLOC
+ debug_alloc(opl3, "note off", voice);
+#endif
+
+}
+
+/*
+ * Release a note in response to a midi note off.
+ */
+void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan)
+{
+ struct snd_opl3 *opl3;
+
+ int voice;
+ struct snd_opl3_voice *vp;
+
+ unsigned long flags;
+
+ opl3 = p;
+
+#ifdef DEBUG_MIDI
+ snd_printk("Note off, ch %i, inst %i, note %i\n",
+ chan->number, chan->midi_program, note);
+#endif
+
+ spin_lock_irqsave(&opl3->voice_lock, flags);
+
+ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) {
+ if (chan->drum_channel && use_internal_drums) {
+ snd_opl3_drum_switch(opl3, note, vel, 0, chan);
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+ return;
+ }
+ /* this loop will hopefully kill all extra voices, because
+ they are grouped by the same channel and note values */
+ for (voice = 0; voice < opl3->max_voices; voice++) {
+ vp = &opl3->voices[voice];
+ if (vp->state > 0 && vp->chan == chan && vp->note == note) {
+ snd_opl3_kill_voice(opl3, voice);
+ }
+ }
+ } else {
+ /* remap OSS voices */
+ if (chan->number < MAX_OPL3_VOICES) {
+ voice = snd_opl3_oss_map[chan->number];
+ snd_opl3_kill_voice(opl3, voice);
+ }
+ }
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+}
+
+/*
+ * key pressure change
+ */
+void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan)
+{
+ struct snd_opl3 *opl3;
+
+ opl3 = p;
+#ifdef DEBUG_MIDI
+ snd_printk("Key pressure, ch#: %i, inst#: %i\n",
+ chan->number, chan->midi_program);
+#endif
+}
+
+/*
+ * terminate note
+ */
+void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan)
+{
+ struct snd_opl3 *opl3;
+
+ opl3 = p;
+#ifdef DEBUG_MIDI
+ snd_printk("Terminate note, ch#: %i, inst#: %i\n",
+ chan->number, chan->midi_program);
+#endif
+}
+
+static void snd_opl3_update_pitch(struct snd_opl3 *opl3, int voice)
+{
+ unsigned short reg_side;
+ unsigned char voice_offset;
+ unsigned short opl3_reg;
+
+ unsigned char fnum, blocknum;
+
+ struct snd_opl3_voice *vp;
+
+ if (snd_BUG_ON(voice >= MAX_OPL3_VOICES))
+ return;
+
+ vp = &opl3->voices[voice];
+ if (vp->chan == NULL)
+ return; /* not allocated? */
+
+ if (voice < MAX_OPL2_VOICES) {
+ /* Left register block for voices 0 .. 8 */
+ reg_side = OPL3_LEFT;
+ voice_offset = voice;
+ } else {
+ /* Right register block for voices 9 .. 17 */
+ reg_side = OPL3_RIGHT;
+ voice_offset = voice - MAX_OPL2_VOICES;
+ }
+
+ snd_opl3_calc_pitch(&fnum, &blocknum, vp->note, vp->chan);
+
+ /* Set OPL3 FNUM_LOW register of requested voice */
+ opl3_reg = reg_side | (OPL3_REG_FNUM_LOW + voice_offset);
+ opl3->command(opl3, opl3_reg, fnum);
+
+ vp->keyon_reg = blocknum;
+
+ /* Set output sound flag */
+ blocknum |= OPL3_KEYON_BIT;
+
+ /* Set OPL3 KEYON_BLOCK register of requested voice */
+ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
+ opl3->command(opl3, opl3_reg, blocknum);
+
+ vp->time = opl3->use_time++;
+}
+
+/*
+ * Update voice pitch controller
+ */
+static void snd_opl3_pitch_ctrl(struct snd_opl3 *opl3, struct snd_midi_channel *chan)
+{
+ int voice;
+ struct snd_opl3_voice *vp;
+
+ unsigned long flags;
+
+ spin_lock_irqsave(&opl3->voice_lock, flags);
+
+ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) {
+ for (voice = 0; voice < opl3->max_voices; voice++) {
+ vp = &opl3->voices[voice];
+ if (vp->state > 0 && vp->chan == chan) {
+ snd_opl3_update_pitch(opl3, voice);
+ }
+ }
+ } else {
+ /* remap OSS voices */
+ if (chan->number < MAX_OPL3_VOICES) {
+ voice = snd_opl3_oss_map[chan->number];
+ snd_opl3_update_pitch(opl3, voice);
+ }
+ }
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+}
+
+/*
+ * Deal with a controller type event. This includes all types of
+ * control events, not just the midi controllers
+ */
+void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan)
+{
+ struct snd_opl3 *opl3;
+
+ opl3 = p;
+#ifdef DEBUG_MIDI
+ snd_printk("Controller, TYPE = %i, ch#: %i, inst#: %i\n",
+ type, chan->number, chan->midi_program);
+#endif
+
+ switch (type) {
+ case MIDI_CTL_MSB_MODWHEEL:
+ if (chan->control[MIDI_CTL_MSB_MODWHEEL] > 63)
+ opl3->drum_reg |= OPL3_VIBRATO_DEPTH;
+ else
+ opl3->drum_reg &= ~OPL3_VIBRATO_DEPTH;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_PERCUSSION,
+ opl3->drum_reg);
+ break;
+ case MIDI_CTL_E2_TREMOLO_DEPTH:
+ if (chan->control[MIDI_CTL_E2_TREMOLO_DEPTH] > 63)
+ opl3->drum_reg |= OPL3_TREMOLO_DEPTH;
+ else
+ opl3->drum_reg &= ~OPL3_TREMOLO_DEPTH;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_PERCUSSION,
+ opl3->drum_reg);
+ break;
+ case MIDI_CTL_PITCHBEND:
+ snd_opl3_pitch_ctrl(opl3, chan);
+ break;
+ }
+}
+
+/*
+ * NRPN events
+ */
+void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset)
+{
+ struct snd_opl3 *opl3;
+
+ opl3 = p;
+#ifdef DEBUG_MIDI
+ snd_printk("NRPN, ch#: %i, inst#: %i\n",
+ chan->number, chan->midi_program);
+#endif
+}
+
+/*
+ * receive sysex
+ */
+void snd_opl3_sysex(void *p, unsigned char *buf, int len,
+ int parsed, struct snd_midi_channel_set *chset)
+{
+ struct snd_opl3 *opl3;
+
+ opl3 = p;
+#ifdef DEBUG_MIDI
+ snd_printk("SYSEX\n");
+#endif
+}
diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c
new file mode 100644
index 0000000..9a2271d
--- /dev/null
+++ b/sound/drivers/opl3/opl3_oss.c
@@ -0,0 +1,283 @@
+/*
+ * Interface for OSS sequencer emulation
+ *
+ * Copyright (C) 2000 Uros Bizjak <uros@kss-loka.si>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "opl3_voice.h"
+#include <linux/slab.h>
+
+static int snd_opl3_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure);
+static int snd_opl3_close_seq_oss(struct snd_seq_oss_arg *arg);
+static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, unsigned long ioarg);
+static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, const char __user *buf, int offs, int count);
+static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg);
+
+/* */
+
+static inline mm_segment_t snd_enter_user(void)
+{
+ mm_segment_t fs = get_fs();
+ set_fs(get_ds());
+ return fs;
+}
+
+static inline void snd_leave_user(mm_segment_t fs)
+{
+ set_fs(fs);
+}
+
+/* operators */
+
+extern struct snd_midi_op opl3_ops;
+
+static struct snd_seq_oss_callback oss_callback = {
+ .owner = THIS_MODULE,
+ .open = snd_opl3_open_seq_oss,
+ .close = snd_opl3_close_seq_oss,
+ .ioctl = snd_opl3_ioctl_seq_oss,
+ .load_patch = snd_opl3_load_patch_seq_oss,
+ .reset = snd_opl3_reset_seq_oss,
+};
+
+static int snd_opl3_oss_event_input(struct snd_seq_event *ev, int direct,
+ void *private_data, int atomic, int hop)
+{
+ struct snd_opl3 *opl3 = private_data;
+
+ if (ev->type != SNDRV_SEQ_EVENT_OSS)
+ snd_midi_process_event(&opl3_ops, ev, opl3->oss_chset);
+ return 0;
+}
+
+/* ------------------------------ */
+
+static void snd_opl3_oss_free_port(void *private_data)
+{
+ struct snd_opl3 *opl3 = private_data;
+
+ snd_midi_channel_free_set(opl3->oss_chset);
+}
+
+static int snd_opl3_oss_create_port(struct snd_opl3 * opl3)
+{
+ struct snd_seq_port_callback callbacks;
+ char name[32];
+ int voices, opl_ver;
+
+ voices = (opl3->hardware < OPL3_HW_OPL3) ?
+ MAX_OPL2_VOICES : MAX_OPL3_VOICES;
+ opl3->oss_chset = snd_midi_channel_alloc_set(voices);
+ if (opl3->oss_chset == NULL)
+ return -ENOMEM;
+ opl3->oss_chset->private_data = opl3;
+
+ memset(&callbacks, 0, sizeof(callbacks));
+ callbacks.owner = THIS_MODULE;
+ callbacks.event_input = snd_opl3_oss_event_input;
+ callbacks.private_free = snd_opl3_oss_free_port;
+ callbacks.private_data = opl3;
+
+ opl_ver = (opl3->hardware & OPL3_HW_MASK) >> 8;
+ sprintf(name, "OPL%i OSS Port", opl_ver);
+
+ opl3->oss_chset->client = opl3->seq_client;
+ opl3->oss_chset->port = snd_seq_event_port_attach(opl3->seq_client, &callbacks,
+ SNDRV_SEQ_PORT_CAP_WRITE,
+ SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM |
+ SNDRV_SEQ_PORT_TYPE_HARDWARE |
+ SNDRV_SEQ_PORT_TYPE_SYNTHESIZER,
+ voices, voices,
+ name);
+ if (opl3->oss_chset->port < 0) {
+ int port;
+ port = opl3->oss_chset->port;
+ snd_midi_channel_free_set(opl3->oss_chset);
+ return port;
+ }
+ return 0;
+}
+
+/* ------------------------------ */
+
+/* register OSS synth */
+void snd_opl3_init_seq_oss(struct snd_opl3 *opl3, char *name)
+{
+ struct snd_seq_oss_reg *arg;
+ struct snd_seq_device *dev;
+
+ if (snd_seq_device_new(opl3->card, 0, SNDRV_SEQ_DEV_ID_OSS,
+ sizeof(struct snd_seq_oss_reg), &dev) < 0)
+ return;
+
+ opl3->oss_seq_dev = dev;
+ strlcpy(dev->name, name, sizeof(dev->name));
+ arg = SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ arg->type = SYNTH_TYPE_FM;
+ if (opl3->hardware < OPL3_HW_OPL3) {
+ arg->subtype = FM_TYPE_ADLIB;
+ arg->nvoices = MAX_OPL2_VOICES;
+ } else {
+ arg->subtype = FM_TYPE_OPL3;
+ arg->nvoices = MAX_OPL3_VOICES;
+ }
+ arg->oper = oss_callback;
+ arg->private_data = opl3;
+
+ if (snd_opl3_oss_create_port(opl3)) {
+ /* register to OSS synth table */
+ snd_device_register(opl3->card, dev);
+ }
+}
+
+/* unregister */
+void snd_opl3_free_seq_oss(struct snd_opl3 *opl3)
+{
+ if (opl3->oss_seq_dev) {
+ /* The instance should have been released in prior */
+ opl3->oss_seq_dev = NULL;
+ }
+}
+
+/* ------------------------------ */
+
+/* open OSS sequencer */
+static int snd_opl3_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure)
+{
+ struct snd_opl3 *opl3 = closure;
+ int err;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+
+ if ((err = snd_opl3_synth_setup(opl3)) < 0)
+ return err;
+
+ /* fill the argument data */
+ arg->private_data = opl3;
+ arg->addr.client = opl3->oss_chset->client;
+ arg->addr.port = opl3->oss_chset->port;
+
+ if ((err = snd_opl3_synth_use_inc(opl3)) < 0)
+ return err;
+
+ opl3->synth_mode = SNDRV_OPL3_MODE_SYNTH;
+ return 0;
+}
+
+/* close OSS sequencer */
+static int snd_opl3_close_seq_oss(struct snd_seq_oss_arg *arg)
+{
+ struct snd_opl3 *opl3;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ opl3 = arg->private_data;
+
+ snd_opl3_synth_cleanup(opl3);
+
+ snd_opl3_synth_use_dec(opl3);
+ return 0;
+}
+
+/* load patch */
+
+/* from sound_config.h */
+#define SBFM_MAXINSTR 256
+
+static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format,
+ const char __user *buf, int offs, int count)
+{
+ struct snd_opl3 *opl3;
+ struct sbi_instrument sbi;
+ char name[32];
+ int err, type;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ opl3 = arg->private_data;
+
+ if (format == FM_PATCH)
+ type = FM_PATCH_OPL2;
+ else if (format == OPL3_PATCH)
+ type = FM_PATCH_OPL3;
+ else
+ return -EINVAL;
+
+ if (count < (int)sizeof(sbi)) {
+ snd_printk("FM Error: Patch record too short\n");
+ return -EINVAL;
+ }
+ if (copy_from_user(&sbi, buf, sizeof(sbi)))
+ return -EFAULT;
+
+ if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) {
+ snd_printk("FM Error: Invalid instrument number %d\n",
+ sbi.channel);
+ return -EINVAL;
+ }
+
+ memset(name, 0, sizeof(name));
+ sprintf(name, "Chan%d", sbi.channel);
+
+ err = snd_opl3_load_patch(opl3, sbi.channel, 127, type, name, NULL,
+ sbi.operators);
+ if (err < 0)
+ return err;
+
+ return sizeof(sbi);
+}
+
+/* ioctl */
+static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd,
+ unsigned long ioarg)
+{
+ struct snd_opl3 *opl3;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ opl3 = arg->private_data;
+ switch (cmd) {
+ case SNDCTL_FM_LOAD_INSTR:
+ snd_printk("OPL3: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n");
+ return -EINVAL;
+
+ case SNDCTL_SYNTH_MEMAVL:
+ return 0x7fffffff;
+
+ case SNDCTL_FM_4OP_ENABLE:
+ // handled automatically by OPL instrument type
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/* reset device */
+static int snd_opl3_reset_seq_oss(struct snd_seq_oss_arg *arg)
+{
+ struct snd_opl3 *opl3;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ opl3 = arg->private_data;
+
+ return 0;
+}
diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c
new file mode 100644
index 0000000..2d33f53
--- /dev/null
+++ b/sound/drivers/opl3/opl3_seq.c
@@ -0,0 +1,297 @@
+/*
+ * Copyright (c) by Uros Bizjak <uros@kss-loka.si>
+ *
+ * Midi Sequencer interface routines for OPL2/OPL3/OPL4 FM
+ *
+ * OPL2/3 FM instrument loader:
+ * alsa-tools/seq/sbiload/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include "opl3_voice.h"
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Uros Bizjak <uros@kss-loka.si>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ALSA driver for OPL3 FM synth");
+
+int use_internal_drums = 0;
+module_param(use_internal_drums, bool, 0444);
+MODULE_PARM_DESC(use_internal_drums, "Enable internal OPL2/3 drums.");
+
+int snd_opl3_synth_use_inc(struct snd_opl3 * opl3)
+{
+ if (!try_module_get(opl3->card->module))
+ return -EFAULT;
+ return 0;
+
+}
+
+void snd_opl3_synth_use_dec(struct snd_opl3 * opl3)
+{
+ module_put(opl3->card->module);
+}
+
+int snd_opl3_synth_setup(struct snd_opl3 * opl3)
+{
+ int idx;
+ struct snd_hwdep *hwdep = opl3->hwdep;
+
+ mutex_lock(&hwdep->open_mutex);
+ if (hwdep->used) {
+ mutex_unlock(&hwdep->open_mutex);
+ return -EBUSY;
+ }
+ hwdep->used++;
+ mutex_unlock(&hwdep->open_mutex);
+
+ snd_opl3_reset(opl3);
+
+ for (idx = 0; idx < MAX_OPL3_VOICES; idx++) {
+ opl3->voices[idx].state = SNDRV_OPL3_ST_OFF;
+ opl3->voices[idx].time = 0;
+ opl3->voices[idx].keyon_reg = 0x00;
+ }
+ opl3->use_time = 0;
+ opl3->connection_reg = 0x00;
+ if (opl3->hardware >= OPL3_HW_OPL3) {
+ /* Clear 4-op connections */
+ opl3->command(opl3, OPL3_RIGHT | OPL3_REG_CONNECTION_SELECT,
+ opl3->connection_reg);
+ opl3->max_voices = MAX_OPL3_VOICES;
+ }
+ return 0;
+}
+
+void snd_opl3_synth_cleanup(struct snd_opl3 * opl3)
+{
+ unsigned long flags;
+ struct snd_hwdep *hwdep;
+
+ /* Stop system timer */
+ spin_lock_irqsave(&opl3->sys_timer_lock, flags);
+ if (opl3->sys_timer_status) {
+ del_timer(&opl3->tlist);
+ opl3->sys_timer_status = 0;
+ }
+ spin_unlock_irqrestore(&opl3->sys_timer_lock, flags);
+
+ snd_opl3_reset(opl3);
+ hwdep = opl3->hwdep;
+ mutex_lock(&hwdep->open_mutex);
+ hwdep->used--;
+ mutex_unlock(&hwdep->open_mutex);
+ wake_up(&hwdep->open_wait);
+}
+
+static int snd_opl3_synth_use(void *private_data, struct snd_seq_port_subscribe * info)
+{
+ struct snd_opl3 *opl3 = private_data;
+ int err;
+
+ if ((err = snd_opl3_synth_setup(opl3)) < 0)
+ return err;
+
+ if (use_internal_drums) {
+ /* Percussion mode */
+ opl3->voices[6].state = opl3->voices[7].state =
+ opl3->voices[8].state = SNDRV_OPL3_ST_NOT_AVAIL;
+ snd_opl3_load_drums(opl3);
+ opl3->drum_reg = OPL3_PERCUSSION_ENABLE;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_PERCUSSION, opl3->drum_reg);
+ } else {
+ opl3->drum_reg = 0x00;
+ }
+
+ if (info->sender.client != SNDRV_SEQ_CLIENT_SYSTEM) {
+ if ((err = snd_opl3_synth_use_inc(opl3)) < 0)
+ return err;
+ }
+ opl3->synth_mode = SNDRV_OPL3_MODE_SEQ;
+ return 0;
+}
+
+static int snd_opl3_synth_unuse(void *private_data, struct snd_seq_port_subscribe * info)
+{
+ struct snd_opl3 *opl3 = private_data;
+
+ snd_opl3_synth_cleanup(opl3);
+
+ if (info->sender.client != SNDRV_SEQ_CLIENT_SYSTEM)
+ snd_opl3_synth_use_dec(opl3);
+ return 0;
+}
+
+/*
+ * MIDI emulation operators
+ */
+struct snd_midi_op opl3_ops = {
+ .note_on = snd_opl3_note_on,
+ .note_off = snd_opl3_note_off,
+ .key_press = snd_opl3_key_press,
+ .note_terminate = snd_opl3_terminate_note,
+ .control = snd_opl3_control,
+ .nrpn = snd_opl3_nrpn,
+ .sysex = snd_opl3_sysex,
+};
+
+static int snd_opl3_synth_event_input(struct snd_seq_event * ev, int direct,
+ void *private_data, int atomic, int hop)
+{
+ struct snd_opl3 *opl3 = private_data;
+
+ snd_midi_process_event(&opl3_ops, ev, opl3->chset);
+ return 0;
+}
+
+/* ------------------------------ */
+
+static void snd_opl3_synth_free_port(void *private_data)
+{
+ struct snd_opl3 *opl3 = private_data;
+
+ snd_midi_channel_free_set(opl3->chset);
+}
+
+static int snd_opl3_synth_create_port(struct snd_opl3 * opl3)
+{
+ struct snd_seq_port_callback callbacks;
+ char name[32];
+ int voices, opl_ver;
+
+ voices = (opl3->hardware < OPL3_HW_OPL3) ?
+ MAX_OPL2_VOICES : MAX_OPL3_VOICES;
+ opl3->chset = snd_midi_channel_alloc_set(16);
+ if (opl3->chset == NULL)
+ return -ENOMEM;
+ opl3->chset->private_data = opl3;
+
+ memset(&callbacks, 0, sizeof(callbacks));
+ callbacks.owner = THIS_MODULE;
+ callbacks.use = snd_opl3_synth_use;
+ callbacks.unuse = snd_opl3_synth_unuse;
+ callbacks.event_input = snd_opl3_synth_event_input;
+ callbacks.private_free = snd_opl3_synth_free_port;
+ callbacks.private_data = opl3;
+
+ opl_ver = (opl3->hardware & OPL3_HW_MASK) >> 8;
+ sprintf(name, "OPL%i FM Port", opl_ver);
+
+ opl3->chset->client = opl3->seq_client;
+ opl3->chset->port = snd_seq_event_port_attach(opl3->seq_client, &callbacks,
+ SNDRV_SEQ_PORT_CAP_WRITE |
+ SNDRV_SEQ_PORT_CAP_SUBS_WRITE,
+ SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM |
+ SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE |
+ SNDRV_SEQ_PORT_TYPE_HARDWARE |
+ SNDRV_SEQ_PORT_TYPE_SYNTHESIZER,
+ 16, voices,
+ name);
+ if (opl3->chset->port < 0) {
+ int port;
+ port = opl3->chset->port;
+ snd_midi_channel_free_set(opl3->chset);
+ return port;
+ }
+ return 0;
+}
+
+/* ------------------------------ */
+
+static int snd_opl3_seq_new_device(struct snd_seq_device *dev)
+{
+ struct snd_opl3 *opl3;
+ int client, err;
+ char name[32];
+ int opl_ver;
+
+ opl3 = *(struct snd_opl3 **)SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ if (opl3 == NULL)
+ return -EINVAL;
+
+ spin_lock_init(&opl3->voice_lock);
+
+ opl3->seq_client = -1;
+
+ /* allocate new client */
+ opl_ver = (opl3->hardware & OPL3_HW_MASK) >> 8;
+ sprintf(name, "OPL%i FM synth", opl_ver);
+ client = opl3->seq_client =
+ snd_seq_create_kernel_client(opl3->card, opl3->seq_dev_num,
+ name);
+ if (client < 0)
+ return client;
+
+ if ((err = snd_opl3_synth_create_port(opl3)) < 0) {
+ snd_seq_delete_kernel_client(client);
+ opl3->seq_client = -1;
+ return err;
+ }
+
+ /* setup system timer */
+ init_timer(&opl3->tlist);
+ opl3->tlist.function = snd_opl3_timer_func;
+ opl3->tlist.data = (unsigned long) opl3;
+ spin_lock_init(&opl3->sys_timer_lock);
+ opl3->sys_timer_status = 0;
+
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ snd_opl3_init_seq_oss(opl3, name);
+#endif
+ return 0;
+}
+
+static int snd_opl3_seq_delete_device(struct snd_seq_device *dev)
+{
+ struct snd_opl3 *opl3;
+
+ opl3 = *(struct snd_opl3 **)SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ if (opl3 == NULL)
+ return -EINVAL;
+
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ snd_opl3_free_seq_oss(opl3);
+#endif
+ if (opl3->seq_client >= 0) {
+ snd_seq_delete_kernel_client(opl3->seq_client);
+ opl3->seq_client = -1;
+ }
+ return 0;
+}
+
+static int __init alsa_opl3_seq_init(void)
+{
+ static struct snd_seq_dev_ops ops =
+ {
+ snd_opl3_seq_new_device,
+ snd_opl3_seq_delete_device
+ };
+
+ return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL3, &ops,
+ sizeof(struct snd_opl3 *));
+}
+
+static void __exit alsa_opl3_seq_exit(void)
+{
+ snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL3);
+}
+
+module_init(alsa_opl3_seq_init)
+module_exit(alsa_opl3_seq_exit)
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
new file mode 100644
index 0000000..962bb9c
--- /dev/null
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -0,0 +1,614 @@
+/*
+ * Copyright (c) by Uros Bizjak <uros@kss-loka.si>
+ *
+ * Routines for OPL2/OPL3/OPL4 control
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/opl3.h>
+#include <sound/asound_fm.h>
+
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+#define OPL3_SUPPORT_SYNTH
+#endif
+
+/*
+ * There is 18 possible 2 OP voices
+ * (9 in the left and 9 in the right).
+ * The first OP is the modulator and 2nd is the carrier.
+ *
+ * The first three voices in the both sides may be connected
+ * with another voice to a 4 OP voice. For example voice 0
+ * can be connected with voice 3. The operators of voice 3 are
+ * used as operators 3 and 4 of the new 4 OP voice.
+ * In this case the 2 OP voice number 0 is the 'first half' and
+ * voice 3 is the second.
+ */
+
+
+/*
+ * Register offset table for OPL2/3 voices,
+ * OPL2 / one OPL3 register array side only
+ */
+
+char snd_opl3_regmap[MAX_OPL2_VOICES][4] =
+{
+/* OP1 OP2 OP3 OP4 */
+/* ------------------------ */
+ { 0x00, 0x03, 0x08, 0x0b },
+ { 0x01, 0x04, 0x09, 0x0c },
+ { 0x02, 0x05, 0x0a, 0x0d },
+
+ { 0x08, 0x0b, 0x00, 0x00 },
+ { 0x09, 0x0c, 0x00, 0x00 },
+ { 0x0a, 0x0d, 0x00, 0x00 },
+
+ { 0x10, 0x13, 0x00, 0x00 }, /* used by percussive voices */
+ { 0x11, 0x14, 0x00, 0x00 }, /* if the percussive mode */
+ { 0x12, 0x15, 0x00, 0x00 } /* is selected (only left reg block) */
+};
+
+EXPORT_SYMBOL(snd_opl3_regmap);
+
+/*
+ * prototypes
+ */
+static int snd_opl3_play_note(struct snd_opl3 * opl3, struct snd_dm_fm_note * note);
+static int snd_opl3_set_voice(struct snd_opl3 * opl3, struct snd_dm_fm_voice * voice);
+static int snd_opl3_set_params(struct snd_opl3 * opl3, struct snd_dm_fm_params * params);
+static int snd_opl3_set_mode(struct snd_opl3 * opl3, int mode);
+static int snd_opl3_set_connection(struct snd_opl3 * opl3, int connection);
+
+/* ------------------------------ */
+
+/*
+ * open the device exclusively
+ */
+int snd_opl3_open(struct snd_hwdep * hw, struct file *file)
+{
+ return 0;
+}
+
+/*
+ * ioctl for hwdep device:
+ */
+int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_opl3 *opl3 = hw->private_data;
+ void __user *argp = (void __user *)arg;
+
+ if (snd_BUG_ON(!opl3))
+ return -EINVAL;
+
+ switch (cmd) {
+ /* get information */
+ case SNDRV_DM_FM_IOCTL_INFO:
+ {
+ struct snd_dm_fm_info info;
+
+ info.fm_mode = opl3->fm_mode;
+ info.rhythm = opl3->rhythm;
+ if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info)))
+ return -EFAULT;
+ return 0;
+ }
+
+ case SNDRV_DM_FM_IOCTL_RESET:
+#ifdef CONFIG_SND_OSSEMUL
+ case SNDRV_DM_FM_OSS_IOCTL_RESET:
+#endif
+ snd_opl3_reset(opl3);
+ return 0;
+
+ case SNDRV_DM_FM_IOCTL_PLAY_NOTE:
+#ifdef CONFIG_SND_OSSEMUL
+ case SNDRV_DM_FM_OSS_IOCTL_PLAY_NOTE:
+#endif
+ {
+ struct snd_dm_fm_note note;
+ if (copy_from_user(&note, argp, sizeof(struct snd_dm_fm_note)))
+ return -EFAULT;
+ return snd_opl3_play_note(opl3, &note);
+ }
+
+ case SNDRV_DM_FM_IOCTL_SET_VOICE:
+#ifdef CONFIG_SND_OSSEMUL
+ case SNDRV_DM_FM_OSS_IOCTL_SET_VOICE:
+#endif
+ {
+ struct snd_dm_fm_voice voice;
+ if (copy_from_user(&voice, argp, sizeof(struct snd_dm_fm_voice)))
+ return -EFAULT;
+ return snd_opl3_set_voice(opl3, &voice);
+ }
+
+ case SNDRV_DM_FM_IOCTL_SET_PARAMS:
+#ifdef CONFIG_SND_OSSEMUL
+ case SNDRV_DM_FM_OSS_IOCTL_SET_PARAMS:
+#endif
+ {
+ struct snd_dm_fm_params params;
+ if (copy_from_user(&params, argp, sizeof(struct snd_dm_fm_params)))
+ return -EFAULT;
+ return snd_opl3_set_params(opl3, &params);
+ }
+
+ case SNDRV_DM_FM_IOCTL_SET_MODE:
+#ifdef CONFIG_SND_OSSEMUL
+ case SNDRV_DM_FM_OSS_IOCTL_SET_MODE:
+#endif
+ return snd_opl3_set_mode(opl3, (int) arg);
+
+ case SNDRV_DM_FM_IOCTL_SET_CONNECTION:
+#ifdef CONFIG_SND_OSSEMUL
+ case SNDRV_DM_FM_OSS_IOCTL_SET_OPL:
+#endif
+ return snd_opl3_set_connection(opl3, (int) arg);
+
+#ifdef OPL3_SUPPORT_SYNTH
+ case SNDRV_DM_FM_IOCTL_CLEAR_PATCHES:
+ snd_opl3_clear_patches(opl3);
+ return 0;
+#endif
+
+#ifdef CONFIG_SND_DEBUG
+ default:
+ snd_printk("unknown IOCTL: 0x%x\n", cmd);
+#endif
+ }
+ return -ENOTTY;
+}
+
+/*
+ * close the device
+ */
+int snd_opl3_release(struct snd_hwdep * hw, struct file *file)
+{
+ struct snd_opl3 *opl3 = hw->private_data;
+
+ snd_opl3_reset(opl3);
+ return 0;
+}
+
+#ifdef OPL3_SUPPORT_SYNTH
+/*
+ * write the device - load patches
+ */
+long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count,
+ loff_t *offset)
+{
+ struct snd_opl3 *opl3 = hw->private_data;
+ long result = 0;
+ int err = 0;
+ struct sbi_patch inst;
+
+ while (count >= sizeof(inst)) {
+ unsigned char type;
+ if (copy_from_user(&inst, buf, sizeof(inst)))
+ return -EFAULT;
+ if (!memcmp(inst.key, FM_KEY_SBI, 4) ||
+ !memcmp(inst.key, FM_KEY_2OP, 4))
+ type = FM_PATCH_OPL2;
+ else if (!memcmp(inst.key, FM_KEY_4OP, 4))
+ type = FM_PATCH_OPL3;
+ else /* invalid type */
+ break;
+ err = snd_opl3_load_patch(opl3, inst.prog, inst.bank, type,
+ inst.name, inst.extension,
+ inst.data);
+ if (err < 0)
+ break;
+ result += sizeof(inst);
+ count -= sizeof(inst);
+ }
+ return result > 0 ? result : err;
+}
+
+
+/*
+ * Patch management
+ */
+
+/* offsets for SBI params */
+#define AM_VIB 0
+#define KSL_LEVEL 2
+#define ATTACK_DECAY 4
+#define SUSTAIN_RELEASE 6
+#define WAVE_SELECT 8
+
+/* offset for SBI instrument */
+#define CONNECTION 10
+#define OFFSET_4OP 11
+
+/*
+ * load a patch, obviously.
+ *
+ * loaded on the given program and bank numbers with the given type
+ * (FM_PATCH_OPLx).
+ * data is the pointer of SBI record _without_ header (key and name).
+ * name is the name string of the patch.
+ * ext is the extension data of 7 bytes long (stored in name of SBI
+ * data up to offset 25), or NULL to skip.
+ * return 0 if successful or a negative error code.
+ */
+int snd_opl3_load_patch(struct snd_opl3 *opl3,
+ int prog, int bank, int type,
+ const char *name,
+ const unsigned char *ext,
+ const unsigned char *data)
+{
+ struct fm_patch *patch;
+ int i;
+
+ patch = snd_opl3_find_patch(opl3, prog, bank, 1);
+ if (!patch)
+ return -ENOMEM;
+
+ patch->type = type;
+
+ for (i = 0; i < 2; i++) {
+ patch->inst.op[i].am_vib = data[AM_VIB + i];
+ patch->inst.op[i].ksl_level = data[KSL_LEVEL + i];
+ patch->inst.op[i].attack_decay = data[ATTACK_DECAY + i];
+ patch->inst.op[i].sustain_release = data[SUSTAIN_RELEASE + i];
+ patch->inst.op[i].wave_select = data[WAVE_SELECT + i];
+ }
+ patch->inst.feedback_connection[0] = data[CONNECTION];
+
+ if (type == FM_PATCH_OPL3) {
+ for (i = 0; i < 2; i++) {
+ patch->inst.op[i+2].am_vib =
+ data[OFFSET_4OP + AM_VIB + i];
+ patch->inst.op[i+2].ksl_level =
+ data[OFFSET_4OP + KSL_LEVEL + i];
+ patch->inst.op[i+2].attack_decay =
+ data[OFFSET_4OP + ATTACK_DECAY + i];
+ patch->inst.op[i+2].sustain_release =
+ data[OFFSET_4OP + SUSTAIN_RELEASE + i];
+ patch->inst.op[i+2].wave_select =
+ data[OFFSET_4OP + WAVE_SELECT + i];
+ }
+ patch->inst.feedback_connection[1] =
+ data[OFFSET_4OP + CONNECTION];
+ }
+
+ if (ext) {
+ patch->inst.echo_delay = ext[0];
+ patch->inst.echo_atten = ext[1];
+ patch->inst.chorus_spread = ext[2];
+ patch->inst.trnsps = ext[3];
+ patch->inst.fix_dur = ext[4];
+ patch->inst.modes = ext[5];
+ patch->inst.fix_key = ext[6];
+ }
+
+ if (name)
+ strlcpy(patch->name, name, sizeof(patch->name));
+
+ return 0;
+}
+EXPORT_SYMBOL(snd_opl3_load_patch);
+
+/*
+ * find a patch with the given program and bank numbers, returns its pointer
+ * if no matching patch is found and create_patch is set, it creates a
+ * new patch object.
+ */
+struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank,
+ int create_patch)
+{
+ /* pretty dumb hash key */
+ unsigned int key = (prog + bank) % OPL3_PATCH_HASH_SIZE;
+ struct fm_patch *patch;
+
+ for (patch = opl3->patch_table[key]; patch; patch = patch->next) {
+ if (patch->prog == prog && patch->bank == bank)
+ return patch;
+ }
+ if (!create_patch)
+ return NULL;
+
+ patch = kzalloc(sizeof(*patch), GFP_KERNEL);
+ if (!patch)
+ return NULL;
+ patch->prog = prog;
+ patch->bank = bank;
+ patch->next = opl3->patch_table[key];
+ opl3->patch_table[key] = patch;
+ return patch;
+}
+EXPORT_SYMBOL(snd_opl3_find_patch);
+
+/*
+ * Clear all patches of the given OPL3 instance
+ */
+void snd_opl3_clear_patches(struct snd_opl3 *opl3)
+{
+ int i;
+ for (i = 0; i < OPL3_PATCH_HASH_SIZE; i++) {
+ struct fm_patch *patch, *next;
+ for (patch = opl3->patch_table[i]; patch; patch = next) {
+ next = patch->next;
+ kfree(patch);
+ }
+ }
+ memset(opl3->patch_table, 0, sizeof(opl3->patch_table));
+}
+#endif /* OPL3_SUPPORT_SYNTH */
+
+/* ------------------------------ */
+
+void snd_opl3_reset(struct snd_opl3 * opl3)
+{
+ unsigned short opl3_reg;
+
+ unsigned short reg_side;
+ unsigned char voice_offset;
+
+ int max_voices, i;
+
+ max_voices = (opl3->hardware < OPL3_HW_OPL3) ?
+ MAX_OPL2_VOICES : MAX_OPL3_VOICES;
+
+ for (i = 0; i < max_voices; i++) {
+ /* Get register array side and offset of voice */
+ if (i < MAX_OPL2_VOICES) {
+ /* Left register block for voices 0 .. 8 */
+ reg_side = OPL3_LEFT;
+ voice_offset = i;
+ } else {
+ /* Right register block for voices 9 .. 17 */
+ reg_side = OPL3_RIGHT;
+ voice_offset = i - MAX_OPL2_VOICES;
+ }
+ opl3_reg = reg_side | (OPL3_REG_KSL_LEVEL + snd_opl3_regmap[voice_offset][0]);
+ opl3->command(opl3, opl3_reg, OPL3_TOTAL_LEVEL_MASK); /* Operator 1 volume */
+ opl3_reg = reg_side | (OPL3_REG_KSL_LEVEL + snd_opl3_regmap[voice_offset][1]);
+ opl3->command(opl3, opl3_reg, OPL3_TOTAL_LEVEL_MASK); /* Operator 2 volume */
+
+ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
+ opl3->command(opl3, opl3_reg, 0x00); /* Note off */
+ }
+
+ opl3->max_voices = MAX_OPL2_VOICES;
+ opl3->fm_mode = SNDRV_DM_FM_MODE_OPL2;
+
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TEST, OPL3_ENABLE_WAVE_SELECT);
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_PERCUSSION, 0x00); /* Melodic mode */
+ opl3->rhythm = 0;
+}
+
+EXPORT_SYMBOL(snd_opl3_reset);
+
+static int snd_opl3_play_note(struct snd_opl3 * opl3, struct snd_dm_fm_note * note)
+{
+ unsigned short reg_side;
+ unsigned char voice_offset;
+
+ unsigned short opl3_reg;
+ unsigned char reg_val;
+
+ /* Voices 0 - 8 in OPL2 mode */
+ /* Voices 0 - 17 in OPL3 mode */
+ if (note->voice >= ((opl3->fm_mode == SNDRV_DM_FM_MODE_OPL3) ?
+ MAX_OPL3_VOICES : MAX_OPL2_VOICES))
+ return -EINVAL;
+
+ /* Get register array side and offset of voice */
+ if (note->voice < MAX_OPL2_VOICES) {
+ /* Left register block for voices 0 .. 8 */
+ reg_side = OPL3_LEFT;
+ voice_offset = note->voice;
+ } else {
+ /* Right register block for voices 9 .. 17 */
+ reg_side = OPL3_RIGHT;
+ voice_offset = note->voice - MAX_OPL2_VOICES;
+ }
+
+ /* Set lower 8 bits of note frequency */
+ reg_val = (unsigned char) note->fnum;
+ opl3_reg = reg_side | (OPL3_REG_FNUM_LOW + voice_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ reg_val = 0x00;
+ /* Set output sound flag */
+ if (note->key_on)
+ reg_val |= OPL3_KEYON_BIT;
+ /* Set octave */
+ reg_val |= (note->octave << 2) & OPL3_BLOCKNUM_MASK;
+ /* Set higher 2 bits of note frequency */
+ reg_val |= (unsigned char) (note->fnum >> 8) & OPL3_FNUM_HIGH_MASK;
+
+ /* Set OPL3 KEYON_BLOCK register of requested voice */
+ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ return 0;
+}
+
+
+static int snd_opl3_set_voice(struct snd_opl3 * opl3, struct snd_dm_fm_voice * voice)
+{
+ unsigned short reg_side;
+ unsigned char op_offset;
+ unsigned char voice_offset;
+
+ unsigned short opl3_reg;
+ unsigned char reg_val;
+
+ /* Only operators 1 and 2 */
+ if (voice->op > 1)
+ return -EINVAL;
+ /* Voices 0 - 8 in OPL2 mode */
+ /* Voices 0 - 17 in OPL3 mode */
+ if (voice->voice >= ((opl3->fm_mode == SNDRV_DM_FM_MODE_OPL3) ?
+ MAX_OPL3_VOICES : MAX_OPL2_VOICES))
+ return -EINVAL;
+
+ /* Get register array side and offset of voice */
+ if (voice->voice < MAX_OPL2_VOICES) {
+ /* Left register block for voices 0 .. 8 */
+ reg_side = OPL3_LEFT;
+ voice_offset = voice->voice;
+ } else {
+ /* Right register block for voices 9 .. 17 */
+ reg_side = OPL3_RIGHT;
+ voice_offset = voice->voice - MAX_OPL2_VOICES;
+ }
+ /* Get register offset of operator */
+ op_offset = snd_opl3_regmap[voice_offset][voice->op];
+
+ reg_val = 0x00;
+ /* Set amplitude modulation (tremolo) effect */
+ if (voice->am)
+ reg_val |= OPL3_TREMOLO_ON;
+ /* Set vibrato effect */
+ if (voice->vibrato)
+ reg_val |= OPL3_VIBRATO_ON;
+ /* Set sustaining sound phase */
+ if (voice->do_sustain)
+ reg_val |= OPL3_SUSTAIN_ON;
+ /* Set keyboard scaling bit */
+ if (voice->kbd_scale)
+ reg_val |= OPL3_KSR;
+ /* Set harmonic or frequency multiplier */
+ reg_val |= voice->harmonic & OPL3_MULTIPLE_MASK;
+
+ /* Set OPL3 AM_VIB register of requested voice/operator */
+ opl3_reg = reg_side | (OPL3_REG_AM_VIB + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set decreasing volume of higher notes */
+ reg_val = (voice->scale_level << 6) & OPL3_KSL_MASK;
+ /* Set output volume */
+ reg_val |= ~voice->volume & OPL3_TOTAL_LEVEL_MASK;
+
+ /* Set OPL3 KSL_LEVEL register of requested voice/operator */
+ opl3_reg = reg_side | (OPL3_REG_KSL_LEVEL + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set attack phase level */
+ reg_val = (voice->attack << 4) & OPL3_ATTACK_MASK;
+ /* Set decay phase level */
+ reg_val |= voice->decay & OPL3_DECAY_MASK;
+
+ /* Set OPL3 ATTACK_DECAY register of requested voice/operator */
+ opl3_reg = reg_side | (OPL3_REG_ATTACK_DECAY + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set sustain phase level */
+ reg_val = (voice->sustain << 4) & OPL3_SUSTAIN_MASK;
+ /* Set release phase level */
+ reg_val |= voice->release & OPL3_RELEASE_MASK;
+
+ /* Set OPL3 SUSTAIN_RELEASE register of requested voice/operator */
+ opl3_reg = reg_side | (OPL3_REG_SUSTAIN_RELEASE + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Set inter-operator feedback */
+ reg_val = (voice->feedback << 1) & OPL3_FEEDBACK_MASK;
+ /* Set inter-operator connection */
+ if (voice->connection)
+ reg_val |= OPL3_CONNECTION_BIT;
+ /* OPL-3 only */
+ if (opl3->fm_mode == SNDRV_DM_FM_MODE_OPL3) {
+ if (voice->left)
+ reg_val |= OPL3_VOICE_TO_LEFT;
+ if (voice->right)
+ reg_val |= OPL3_VOICE_TO_RIGHT;
+ }
+ /* Feedback/connection bits are applicable to voice */
+ opl3_reg = reg_side | (OPL3_REG_FEEDBACK_CONNECTION + voice_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ /* Select waveform */
+ reg_val = voice->waveform & OPL3_WAVE_SELECT_MASK;
+ opl3_reg = reg_side | (OPL3_REG_WAVE_SELECT + op_offset);
+ opl3->command(opl3, opl3_reg, reg_val);
+
+ return 0;
+}
+
+static int snd_opl3_set_params(struct snd_opl3 * opl3, struct snd_dm_fm_params * params)
+{
+ unsigned char reg_val;
+
+ reg_val = 0x00;
+ /* Set keyboard split method */
+ if (params->kbd_split)
+ reg_val |= OPL3_KEYBOARD_SPLIT;
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_KBD_SPLIT, reg_val);
+
+ reg_val = 0x00;
+ /* Set amplitude modulation (tremolo) depth */
+ if (params->am_depth)
+ reg_val |= OPL3_TREMOLO_DEPTH;
+ /* Set vibrato depth */
+ if (params->vib_depth)
+ reg_val |= OPL3_VIBRATO_DEPTH;
+ /* Set percussion mode */
+ if (params->rhythm) {
+ reg_val |= OPL3_PERCUSSION_ENABLE;
+ opl3->rhythm = 1;
+ } else {
+ opl3->rhythm = 0;
+ }
+ /* Play percussion instruments */
+ if (params->bass)
+ reg_val |= OPL3_BASSDRUM_ON;
+ if (params->snare)
+ reg_val |= OPL3_SNAREDRUM_ON;
+ if (params->tomtom)
+ reg_val |= OPL3_TOMTOM_ON;
+ if (params->cymbal)
+ reg_val |= OPL3_CYMBAL_ON;
+ if (params->hihat)
+ reg_val |= OPL3_HIHAT_ON;
+
+ opl3->command(opl3, OPL3_LEFT | OPL3_REG_PERCUSSION, reg_val);
+ return 0;
+}
+
+static int snd_opl3_set_mode(struct snd_opl3 * opl3, int mode)
+{
+ if ((mode == SNDRV_DM_FM_MODE_OPL3) && (opl3->hardware < OPL3_HW_OPL3))
+ return -EINVAL;
+
+ opl3->fm_mode = mode;
+ if (opl3->hardware >= OPL3_HW_OPL3)
+ opl3->command(opl3, OPL3_RIGHT | OPL3_REG_CONNECTION_SELECT, 0x00); /* Clear 4-op connections */
+
+ return 0;
+}
+
+static int snd_opl3_set_connection(struct snd_opl3 * opl3, int connection)
+{
+ unsigned char reg_val;
+
+ /* OPL-3 only */
+ if (opl3->fm_mode != SNDRV_DM_FM_MODE_OPL3)
+ return -EINVAL;
+
+ reg_val = connection & (OPL3_RIGHT_4OP_0 | OPL3_RIGHT_4OP_1 | OPL3_RIGHT_4OP_2 |
+ OPL3_LEFT_4OP_0 | OPL3_LEFT_4OP_1 | OPL3_LEFT_4OP_2);
+ /* Set 4-op connections */
+ opl3->command(opl3, OPL3_RIGHT | OPL3_REG_CONNECTION_SELECT, reg_val);
+
+ return 0;
+}
+
diff --git a/sound/drivers/opl3/opl3_voice.h b/sound/drivers/opl3/opl3_voice.h
new file mode 100644
index 0000000..a371c07
--- /dev/null
+++ b/sound/drivers/opl3/opl3_voice.h
@@ -0,0 +1,52 @@
+#ifndef __OPL3_VOICE_H
+#define __OPL3_VOICE_H
+
+/*
+ * Copyright (c) 2000 Uros Bizjak <uros@kss-loka.si>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/opl3.h>
+
+/* Prototypes for opl3_seq.c */
+int snd_opl3_synth_use_inc(struct snd_opl3 * opl3);
+void snd_opl3_synth_use_dec(struct snd_opl3 * opl3);
+int snd_opl3_synth_setup(struct snd_opl3 * opl3);
+void snd_opl3_synth_cleanup(struct snd_opl3 * opl3);
+
+/* Prototypes for opl3_midi.c */
+void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan);
+void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan);
+void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_channel_set *chset);
+void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset);
+
+void snd_opl3_calc_volume(unsigned char *reg, int vel, struct snd_midi_channel *chan);
+void snd_opl3_timer_func(unsigned long data);
+
+/* Prototypes for opl3_drums.c */
+void snd_opl3_load_drums(struct snd_opl3 *opl3);
+void snd_opl3_drum_switch(struct snd_opl3 *opl3, int note, int on_off, int vel, struct snd_midi_channel *chan);
+
+/* Prototypes for opl3_oss.c */
+#ifdef CONFIG_SND_SEQUENCER_OSS
+void snd_opl3_init_seq_oss(struct snd_opl3 *opl3, char *name);
+void snd_opl3_free_seq_oss(struct snd_opl3 *opl3);
+#endif
+
+#endif
diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile
new file mode 100644
index 0000000..d178b39
--- /dev/null
+++ b/sound/drivers/opl4/Makefile
@@ -0,0 +1,18 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o opl4_proc.o
+snd-opl4-synth-objs := opl4_seq.o opl4_synth.o yrw801.o
+
+#
+# this function returns:
+# "m" - CONFIG_SND_SEQUENCER is m
+# <empty string> - CONFIG_SND_SEQUENCER is undefined
+# otherwise parameter #1 value
+#
+sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+
+obj-$(CONFIG_SND_OPL4_LIB) += snd-opl4-lib.o
+obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-opl4-synth.o
diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c
new file mode 100644
index 0000000..01997f2
--- /dev/null
+++ b/sound/drivers/opl4/opl4_lib.c
@@ -0,0 +1,279 @@
+/*
+ * Functions for accessing OPL4 devices
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "opl4_local.h"
+#include <sound/initval.h>
+#include <linux/ioport.h>
+#include <linux/init.h>
+#include <asm/io.h>
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("OPL4 driver");
+MODULE_LICENSE("GPL");
+
+static void inline snd_opl4_wait(struct snd_opl4 *opl4)
+{
+ int timeout = 10;
+ while ((inb(opl4->fm_port) & OPL4_STATUS_BUSY) && --timeout > 0)
+ ;
+}
+
+void snd_opl4_write(struct snd_opl4 *opl4, u8 reg, u8 value)
+{
+ snd_opl4_wait(opl4);
+ outb(reg, opl4->pcm_port);
+
+ snd_opl4_wait(opl4);
+ outb(value, opl4->pcm_port + 1);
+}
+
+EXPORT_SYMBOL(snd_opl4_write);
+
+u8 snd_opl4_read(struct snd_opl4 *opl4, u8 reg)
+{
+ snd_opl4_wait(opl4);
+ outb(reg, opl4->pcm_port);
+
+ snd_opl4_wait(opl4);
+ return inb(opl4->pcm_port + 1);
+}
+
+EXPORT_SYMBOL(snd_opl4_read);
+
+void snd_opl4_read_memory(struct snd_opl4 *opl4, char *buf, int offset, int size)
+{
+ unsigned long flags;
+ u8 memcfg;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+
+ memcfg = snd_opl4_read(opl4, OPL4_REG_MEMORY_CONFIGURATION);
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_CONFIGURATION, memcfg | OPL4_MODE_BIT);
+
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_ADDRESS_HIGH, offset >> 16);
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_ADDRESS_MID, offset >> 8);
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_ADDRESS_LOW, offset);
+
+ snd_opl4_wait(opl4);
+ outb(OPL4_REG_MEMORY_DATA, opl4->pcm_port);
+ snd_opl4_wait(opl4);
+ insb(opl4->pcm_port + 1, buf, size);
+
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_CONFIGURATION, memcfg);
+
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+}
+
+EXPORT_SYMBOL(snd_opl4_read_memory);
+
+void snd_opl4_write_memory(struct snd_opl4 *opl4, const char *buf, int offset, int size)
+{
+ unsigned long flags;
+ u8 memcfg;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+
+ memcfg = snd_opl4_read(opl4, OPL4_REG_MEMORY_CONFIGURATION);
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_CONFIGURATION, memcfg | OPL4_MODE_BIT);
+
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_ADDRESS_HIGH, offset >> 16);
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_ADDRESS_MID, offset >> 8);
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_ADDRESS_LOW, offset);
+
+ snd_opl4_wait(opl4);
+ outb(OPL4_REG_MEMORY_DATA, opl4->pcm_port);
+ snd_opl4_wait(opl4);
+ outsb(opl4->pcm_port + 1, buf, size);
+
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_CONFIGURATION, memcfg);
+
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+}
+
+EXPORT_SYMBOL(snd_opl4_write_memory);
+
+static void snd_opl4_enable_opl4(struct snd_opl4 *opl4)
+{
+ outb(OPL3_REG_MODE, opl4->fm_port + 2);
+ inb(opl4->fm_port);
+ inb(opl4->fm_port);
+ outb(OPL3_OPL3_ENABLE | OPL3_OPL4_ENABLE, opl4->fm_port + 3);
+ inb(opl4->fm_port);
+ inb(opl4->fm_port);
+}
+
+static int snd_opl4_detect(struct snd_opl4 *opl4)
+{
+ u8 id1, id2;
+
+ snd_opl4_enable_opl4(opl4);
+
+ id1 = snd_opl4_read(opl4, OPL4_REG_MEMORY_CONFIGURATION);
+ snd_printdd("OPL4[02]=%02x\n", id1);
+ switch (id1 & OPL4_DEVICE_ID_MASK) {
+ case 0x20:
+ opl4->hardware = OPL3_HW_OPL4;
+ break;
+ case 0x40:
+ opl4->hardware = OPL3_HW_OPL4_ML;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ snd_opl4_write(opl4, OPL4_REG_MIX_CONTROL_FM, 0x00);
+ snd_opl4_write(opl4, OPL4_REG_MIX_CONTROL_PCM, 0xff);
+ id1 = snd_opl4_read(opl4, OPL4_REG_MIX_CONTROL_FM);
+ id2 = snd_opl4_read(opl4, OPL4_REG_MIX_CONTROL_PCM);
+ snd_printdd("OPL4 id1=%02x id2=%02x\n", id1, id2);
+ if (id1 != 0x00 || id2 != 0xff)
+ return -ENODEV;
+
+ snd_opl4_write(opl4, OPL4_REG_MIX_CONTROL_FM, 0x3f);
+ snd_opl4_write(opl4, OPL4_REG_MIX_CONTROL_PCM, 0x3f);
+ snd_opl4_write(opl4, OPL4_REG_MEMORY_CONFIGURATION, 0x00);
+ return 0;
+}
+
+#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))
+static void snd_opl4_seq_dev_free(struct snd_seq_device *seq_dev)
+{
+ struct snd_opl4 *opl4 = seq_dev->private_data;
+ opl4->seq_dev = NULL;
+}
+
+static int snd_opl4_create_seq_dev(struct snd_opl4 *opl4, int seq_device)
+{
+ opl4->seq_dev_num = seq_device;
+ if (snd_seq_device_new(opl4->card, seq_device, SNDRV_SEQ_DEV_ID_OPL4,
+ sizeof(struct snd_opl4 *), &opl4->seq_dev) >= 0) {
+ strcpy(opl4->seq_dev->name, "OPL4 Wavetable");
+ *(struct snd_opl4 **)SNDRV_SEQ_DEVICE_ARGPTR(opl4->seq_dev) = opl4;
+ opl4->seq_dev->private_data = opl4;
+ opl4->seq_dev->private_free = snd_opl4_seq_dev_free;
+ }
+ return 0;
+}
+#endif
+
+static void snd_opl4_free(struct snd_opl4 *opl4)
+{
+#ifdef CONFIG_PROC_FS
+ snd_opl4_free_proc(opl4);
+#endif
+ release_and_free_resource(opl4->res_fm_port);
+ release_and_free_resource(opl4->res_pcm_port);
+ kfree(opl4);
+}
+
+static int snd_opl4_dev_free(struct snd_device *device)
+{
+ struct snd_opl4 *opl4 = device->device_data;
+ snd_opl4_free(opl4);
+ return 0;
+}
+
+int snd_opl4_create(struct snd_card *card,
+ unsigned long fm_port, unsigned long pcm_port,
+ int seq_device,
+ struct snd_opl3 **ropl3, struct snd_opl4 **ropl4)
+{
+ struct snd_opl4 *opl4;
+ struct snd_opl3 *opl3;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_opl4_dev_free
+ };
+
+ if (ropl3)
+ *ropl3 = NULL;
+ if (ropl4)
+ *ropl4 = NULL;
+
+ opl4 = kzalloc(sizeof(*opl4), GFP_KERNEL);
+ if (!opl4)
+ return -ENOMEM;
+
+ opl4->res_fm_port = request_region(fm_port, 8, "OPL4 FM");
+ opl4->res_pcm_port = request_region(pcm_port, 8, "OPL4 PCM/MIX");
+ if (!opl4->res_fm_port || !opl4->res_pcm_port) {
+ snd_printk(KERN_ERR "opl4: can't grab ports 0x%lx, 0x%lx\n", fm_port, pcm_port);
+ snd_opl4_free(opl4);
+ return -EBUSY;
+ }
+
+ opl4->card = card;
+ opl4->fm_port = fm_port;
+ opl4->pcm_port = pcm_port;
+ spin_lock_init(&opl4->reg_lock);
+ mutex_init(&opl4->access_mutex);
+
+ err = snd_opl4_detect(opl4);
+ if (err < 0) {
+ snd_opl4_free(opl4);
+ snd_printd("OPL4 chip not detected at %#lx/%#lx\n", fm_port, pcm_port);
+ return err;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_CODEC, opl4, &ops);
+ if (err < 0) {
+ snd_opl4_free(opl4);
+ return err;
+ }
+
+ err = snd_opl3_create(card, fm_port, fm_port + 2, opl4->hardware, 1, &opl3);
+ if (err < 0) {
+ snd_device_free(card, opl4);
+ return err;
+ }
+
+ /* opl3 initialization disabled opl4, so reenable */
+ snd_opl4_enable_opl4(opl4);
+
+ snd_opl4_create_mixer(opl4);
+#ifdef CONFIG_PROC_FS
+ snd_opl4_create_proc(opl4);
+#endif
+
+#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))
+ opl4->seq_client = -1;
+ if (opl4->hardware < OPL3_HW_OPL4_ML)
+ snd_opl4_create_seq_dev(opl4, seq_device);
+#endif
+
+ if (ropl3)
+ *ropl3 = opl3;
+ if (ropl4)
+ *ropl4 = opl4;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_opl4_create);
+
+static int __init alsa_opl4_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_opl4_exit(void)
+{
+}
+
+module_init(alsa_opl4_init)
+module_exit(alsa_opl4_exit)
diff --git a/sound/drivers/opl4/opl4_local.h b/sound/drivers/opl4/opl4_local.h
new file mode 100644
index 0000000..470e5a7
--- /dev/null
+++ b/sound/drivers/opl4/opl4_local.h
@@ -0,0 +1,232 @@
+/*
+ * Local definitions for the OPL4 driver
+ *
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions, and the following disclaimer,
+ * without modification.
+ * 2. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * Alternatively, this software may be distributed and/or modified under the
+ * terms of the GNU General Public License as published by the Free Software
+ * Foundation; either version 2 of the License, or (at your option) any later
+ * version.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ */
+
+#ifndef __OPL4_LOCAL_H
+#define __OPL4_LOCAL_H
+
+#include <sound/opl4.h>
+
+/*
+ * Register numbers
+ */
+
+#define OPL4_REG_TEST0 0x00
+#define OPL4_REG_TEST1 0x01
+
+#define OPL4_REG_MEMORY_CONFIGURATION 0x02
+#define OPL4_MODE_BIT 0x01
+#define OPL4_MTYPE_BIT 0x02
+#define OPL4_TONE_HEADER_MASK 0x1c
+#define OPL4_DEVICE_ID_MASK 0xe0
+
+#define OPL4_REG_MEMORY_ADDRESS_HIGH 0x03
+#define OPL4_REG_MEMORY_ADDRESS_MID 0x04
+#define OPL4_REG_MEMORY_ADDRESS_LOW 0x05
+#define OPL4_REG_MEMORY_DATA 0x06
+
+/*
+ * Offsets to the register banks for voices. To get the
+ * register number just add the voice number to the bank offset.
+ *
+ * Wave Table Number low bits (0x08 to 0x1F)
+ */
+#define OPL4_REG_TONE_NUMBER 0x08
+
+/* Wave Table Number high bit, F-Number low bits (0x20 to 0x37) */
+#define OPL4_REG_F_NUMBER 0x20
+#define OPL4_TONE_NUMBER_BIT8 0x01
+#define OPL4_F_NUMBER_LOW_MASK 0xfe
+
+/* F-Number high bits, Octave, Pseudo-Reverb (0x38 to 0x4F) */
+#define OPL4_REG_OCTAVE 0x38
+#define OPL4_F_NUMBER_HIGH_MASK 0x07
+#define OPL4_BLOCK_MASK 0xf0
+#define OPL4_PSEUDO_REVERB_BIT 0x08
+
+/* Total Level, Level Direct (0x50 to 0x67) */
+#define OPL4_REG_LEVEL 0x50
+#define OPL4_TOTAL_LEVEL_MASK 0xfe
+#define OPL4_LEVEL_DIRECT_BIT 0x01
+
+/* Key On, Damp, LFO RST, CH, Panpot (0x68 to 0x7F) */
+#define OPL4_REG_MISC 0x68
+#define OPL4_KEY_ON_BIT 0x80
+#define OPL4_DAMP_BIT 0x40
+#define OPL4_LFO_RESET_BIT 0x20
+#define OPL4_OUTPUT_CHANNEL_BIT 0x10
+#define OPL4_PAN_POT_MASK 0x0f
+
+/* LFO, VIB (0x80 to 0x97) */
+#define OPL4_REG_LFO_VIBRATO 0x80
+#define OPL4_LFO_FREQUENCY_MASK 0x38
+#define OPL4_VIBRATO_DEPTH_MASK 0x07
+#define OPL4_CHORUS_SEND_MASK 0xc0 /* ML only */
+
+/* Attack / Decay 1 rate (0x98 to 0xAF) */
+#define OPL4_REG_ATTACK_DECAY1 0x98
+#define OPL4_ATTACK_RATE_MASK 0xf0
+#define OPL4_DECAY1_RATE_MASK 0x0f
+
+/* Decay level / 2 rate (0xB0 to 0xC7) */
+#define OPL4_REG_LEVEL_DECAY2 0xb0
+#define OPL4_DECAY_LEVEL_MASK 0xf0
+#define OPL4_DECAY2_RATE_MASK 0x0f
+
+/* Release rate / Rate correction (0xC8 to 0xDF) */
+#define OPL4_REG_RELEASE_CORRECTION 0xc8
+#define OPL4_RELEASE_RATE_MASK 0x0f
+#define OPL4_RATE_INTERPOLATION_MASK 0xf0
+
+/* AM (0xE0 to 0xF7) */
+#define OPL4_REG_TREMOLO 0xe0
+#define OPL4_TREMOLO_DEPTH_MASK 0x07
+#define OPL4_REVERB_SEND_MASK 0xe0 /* ML only */
+
+/* Mixer */
+#define OPL4_REG_MIX_CONTROL_FM 0xf8
+#define OPL4_REG_MIX_CONTROL_PCM 0xf9
+#define OPL4_MIX_LEFT_MASK 0x07
+#define OPL4_MIX_RIGHT_MASK 0x38
+
+#define OPL4_REG_ATC 0xfa
+#define OPL4_ATC_BIT 0x01 /* ???, ML only */
+
+/* bits in the OPL3 Status register */
+#define OPL4_STATUS_BUSY 0x01
+#define OPL4_STATUS_LOAD 0x02
+
+
+#define OPL4_MAX_VOICES 24
+
+#define SNDRV_SEQ_DEV_ID_OPL4 "opl4-synth"
+
+
+struct opl4_sound {
+ u16 tone;
+ s16 pitch_offset;
+ u8 key_scaling;
+ s8 panpot;
+ u8 vibrato;
+ u8 tone_attenuate;
+ u8 volume_factor;
+ u8 reg_lfo_vibrato;
+ u8 reg_attack_decay1;
+ u8 reg_level_decay2;
+ u8 reg_release_correction;
+ u8 reg_tremolo;
+};
+
+struct opl4_region {
+ u8 key_min, key_max;
+ struct opl4_sound sound;
+};
+
+struct opl4_region_ptr {
+ int count;
+ const struct opl4_region *regions;
+};
+
+struct opl4_voice {
+ struct list_head list;
+ int number;
+ struct snd_midi_channel *chan;
+ int note;
+ int velocity;
+ const struct opl4_sound *sound;
+ u8 level_direct;
+ u8 reg_f_number;
+ u8 reg_misc;
+ u8 reg_lfo_vibrato;
+};
+
+struct snd_opl4 {
+ unsigned long fm_port;
+ unsigned long pcm_port;
+ struct resource *res_fm_port;
+ struct resource *res_pcm_port;
+ unsigned short hardware;
+ spinlock_t reg_lock;
+ struct snd_card *card;
+
+#ifdef CONFIG_PROC_FS
+ struct snd_info_entry *proc_entry;
+ int memory_access;
+#endif
+ struct mutex access_mutex;
+
+#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+ int used;
+
+ int seq_dev_num;
+ int seq_client;
+ struct snd_seq_device *seq_dev;
+
+ struct snd_midi_channel_set *chset;
+ struct opl4_voice voices[OPL4_MAX_VOICES];
+ struct list_head off_voices;
+ struct list_head on_voices;
+#endif
+};
+
+/* opl4_lib.c */
+void snd_opl4_write(struct snd_opl4 *opl4, u8 reg, u8 value);
+u8 snd_opl4_read(struct snd_opl4 *opl4, u8 reg);
+void snd_opl4_read_memory(struct snd_opl4 *opl4, char *buf, int offset, int size);
+void snd_opl4_write_memory(struct snd_opl4 *opl4, const char *buf, int offset, int size);
+
+/* opl4_mixer.c */
+int snd_opl4_create_mixer(struct snd_opl4 *opl4);
+
+#ifdef CONFIG_PROC_FS
+/* opl4_proc.c */
+int snd_opl4_create_proc(struct snd_opl4 *opl4);
+void snd_opl4_free_proc(struct snd_opl4 *opl4);
+#endif
+
+/* opl4_seq.c */
+extern int volume_boost;
+
+/* opl4_synth.c */
+void snd_opl4_synth_reset(struct snd_opl4 *opl4);
+void snd_opl4_synth_shutdown(struct snd_opl4 *opl4);
+void snd_opl4_note_on(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_opl4_note_off(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_opl4_terminate_note(void *p, int note, struct snd_midi_channel *chan);
+void snd_opl4_control(void *p, int type, struct snd_midi_channel *chan);
+void snd_opl4_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset);
+
+/* yrw801.c */
+int snd_yrw801_detect(struct snd_opl4 *opl4);
+extern const struct opl4_region_ptr snd_yrw801_regions[];
+
+#endif /* __OPL4_LOCAL_H */
diff --git a/sound/drivers/opl4/opl4_mixer.c b/sound/drivers/opl4/opl4_mixer.c
new file mode 100644
index 0000000..04079de
--- /dev/null
+++ b/sound/drivers/opl4/opl4_mixer.c
@@ -0,0 +1,95 @@
+/*
+ * OPL4 mixer functions
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "opl4_local.h"
+#include <sound/control.h>
+
+static int snd_opl4_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 7;
+ return 0;
+}
+
+static int snd_opl4_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_opl4 *opl4 = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ u8 reg = kcontrol->private_value;
+ u8 value;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ value = snd_opl4_read(opl4, reg);
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+ ucontrol->value.integer.value[0] = 7 - (value & 7);
+ ucontrol->value.integer.value[1] = 7 - ((value >> 3) & 7);
+ return 0;
+}
+
+static int snd_opl4_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_opl4 *opl4 = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ u8 reg = kcontrol->private_value;
+ u8 value, old_value;
+
+ value = (7 - (ucontrol->value.integer.value[0] & 7)) |
+ ((7 - (ucontrol->value.integer.value[1] & 7)) << 3);
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ old_value = snd_opl4_read(opl4, reg);
+ snd_opl4_write(opl4, reg, value);
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+ return value != old_value;
+}
+
+static struct snd_kcontrol_new snd_opl4_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "FM Playback Volume",
+ .info = snd_opl4_ctl_info,
+ .get = snd_opl4_ctl_get,
+ .put = snd_opl4_ctl_put,
+ .private_value = OPL4_REG_MIX_CONTROL_FM
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Wavetable Playback Volume",
+ .info = snd_opl4_ctl_info,
+ .get = snd_opl4_ctl_get,
+ .put = snd_opl4_ctl_put,
+ .private_value = OPL4_REG_MIX_CONTROL_PCM
+ }
+};
+
+int snd_opl4_create_mixer(struct snd_opl4 *opl4)
+{
+ struct snd_card *card = opl4->card;
+ int i, err;
+
+ strcat(card->mixername, ",OPL4");
+
+ for (i = 0; i < 2; ++i) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_opl4_controls[i], opl4));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
diff --git a/sound/drivers/opl4/opl4_proc.c b/sound/drivers/opl4/opl4_proc.c
new file mode 100644
index 0000000..1679300
--- /dev/null
+++ b/sound/drivers/opl4/opl4_proc.c
@@ -0,0 +1,165 @@
+/*
+ * Functions for the OPL4 proc file
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "opl4_local.h"
+#include <linux/vmalloc.h>
+#include <sound/info.h>
+
+#ifdef CONFIG_PROC_FS
+
+static int snd_opl4_mem_proc_open(struct snd_info_entry *entry,
+ unsigned short mode, void **file_private_data)
+{
+ struct snd_opl4 *opl4 = entry->private_data;
+
+ mutex_lock(&opl4->access_mutex);
+ if (opl4->memory_access) {
+ mutex_unlock(&opl4->access_mutex);
+ return -EBUSY;
+ }
+ opl4->memory_access++;
+ mutex_unlock(&opl4->access_mutex);
+ return 0;
+}
+
+static int snd_opl4_mem_proc_release(struct snd_info_entry *entry,
+ unsigned short mode, void *file_private_data)
+{
+ struct snd_opl4 *opl4 = entry->private_data;
+
+ mutex_lock(&opl4->access_mutex);
+ opl4->memory_access--;
+ mutex_unlock(&opl4->access_mutex);
+ return 0;
+}
+
+static long snd_opl4_mem_proc_read(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, char __user *_buf,
+ unsigned long count, unsigned long pos)
+{
+ struct snd_opl4 *opl4 = entry->private_data;
+ long size;
+ char* buf;
+
+ size = count;
+ if (pos + size > entry->size)
+ size = entry->size - pos;
+ if (size > 0) {
+ buf = vmalloc(size);
+ if (!buf)
+ return -ENOMEM;
+ snd_opl4_read_memory(opl4, buf, pos, size);
+ if (copy_to_user(_buf, buf, size)) {
+ vfree(buf);
+ return -EFAULT;
+ }
+ vfree(buf);
+ return size;
+ }
+ return 0;
+}
+
+static long snd_opl4_mem_proc_write(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, const char __user *_buf,
+ unsigned long count, unsigned long pos)
+{
+ struct snd_opl4 *opl4 = entry->private_data;
+ long size;
+ char *buf;
+
+ size = count;
+ if (pos + size > entry->size)
+ size = entry->size - pos;
+ if (size > 0) {
+ buf = vmalloc(size);
+ if (!buf)
+ return -ENOMEM;
+ if (copy_from_user(buf, _buf, size)) {
+ vfree(buf);
+ return -EFAULT;
+ }
+ snd_opl4_write_memory(opl4, buf, pos, size);
+ vfree(buf);
+ return size;
+ }
+ return 0;
+}
+
+static long long snd_opl4_mem_proc_llseek(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, long long offset, int orig)
+{
+ switch (orig) {
+ case SEEK_SET:
+ file->f_pos = offset;
+ break;
+ case SEEK_CUR:
+ file->f_pos += offset;
+ break;
+ case SEEK_END: /* offset is negative */
+ file->f_pos = entry->size + offset;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (file->f_pos > entry->size)
+ file->f_pos = entry->size;
+ return file->f_pos;
+}
+
+static struct snd_info_entry_ops snd_opl4_mem_proc_ops = {
+ .open = snd_opl4_mem_proc_open,
+ .release = snd_opl4_mem_proc_release,
+ .read = snd_opl4_mem_proc_read,
+ .write = snd_opl4_mem_proc_write,
+ .llseek = snd_opl4_mem_proc_llseek,
+};
+
+int snd_opl4_create_proc(struct snd_opl4 *opl4)
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_card_entry(opl4->card, "opl4-mem", opl4->card->proc_root);
+ if (entry) {
+ if (opl4->hardware < OPL3_HW_OPL4_ML) {
+ /* OPL4 can access 4 MB external ROM/SRAM */
+ entry->mode |= S_IWUSR;
+ entry->size = 4 * 1024 * 1024;
+ } else {
+ /* OPL4-ML has 1 MB internal ROM */
+ entry->size = 1 * 1024 * 1024;
+ }
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->c.ops = &snd_opl4_mem_proc_ops;
+ entry->module = THIS_MODULE;
+ entry->private_data = opl4;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ opl4->proc_entry = entry;
+ return 0;
+}
+
+void snd_opl4_free_proc(struct snd_opl4 *opl4)
+{
+ snd_info_free_entry(opl4->proc_entry);
+}
+
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c
new file mode 100644
index 0000000..43d8a2b
--- /dev/null
+++ b/sound/drivers/opl4/opl4_seq.c
@@ -0,0 +1,214 @@
+/*
+ * OPL4 sequencer functions
+ *
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions, and the following disclaimer,
+ * without modification.
+ * 2. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * Alternatively, this software may be distributed and/or modified under the
+ * terms of the GNU General Public License as published by the Free Software
+ * Foundation; either version 2 of the License, or (at your option) any later
+ * version.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ */
+
+#include "opl4_local.h"
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("OPL4 wavetable synth driver");
+MODULE_LICENSE("Dual BSD/GPL");
+
+int volume_boost = 8;
+
+module_param(volume_boost, int, 0644);
+MODULE_PARM_DESC(volume_boost, "Additional volume for OPL4 wavetable sounds.");
+
+static int snd_opl4_seq_use_inc(struct snd_opl4 *opl4)
+{
+ if (!try_module_get(opl4->card->module))
+ return -EFAULT;
+ return 0;
+}
+
+static void snd_opl4_seq_use_dec(struct snd_opl4 *opl4)
+{
+ module_put(opl4->card->module);
+}
+
+static int snd_opl4_seq_use(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ struct snd_opl4 *opl4 = private_data;
+ int err;
+
+ mutex_lock(&opl4->access_mutex);
+
+ if (opl4->used) {
+ mutex_unlock(&opl4->access_mutex);
+ return -EBUSY;
+ }
+ opl4->used++;
+
+ if (info->sender.client != SNDRV_SEQ_CLIENT_SYSTEM) {
+ err = snd_opl4_seq_use_inc(opl4);
+ if (err < 0) {
+ mutex_unlock(&opl4->access_mutex);
+ return err;
+ }
+ }
+
+ mutex_unlock(&opl4->access_mutex);
+
+ snd_opl4_synth_reset(opl4);
+ return 0;
+}
+
+static int snd_opl4_seq_unuse(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ struct snd_opl4 *opl4 = private_data;
+
+ snd_opl4_synth_shutdown(opl4);
+
+ mutex_lock(&opl4->access_mutex);
+ opl4->used--;
+ mutex_unlock(&opl4->access_mutex);
+
+ if (info->sender.client != SNDRV_SEQ_CLIENT_SYSTEM)
+ snd_opl4_seq_use_dec(opl4);
+ return 0;
+}
+
+static struct snd_midi_op opl4_ops = {
+ .note_on = snd_opl4_note_on,
+ .note_off = snd_opl4_note_off,
+ .note_terminate = snd_opl4_terminate_note,
+ .control = snd_opl4_control,
+ .sysex = snd_opl4_sysex,
+};
+
+static int snd_opl4_seq_event_input(struct snd_seq_event *ev, int direct,
+ void *private_data, int atomic, int hop)
+{
+ struct snd_opl4 *opl4 = private_data;
+
+ snd_midi_process_event(&opl4_ops, ev, opl4->chset);
+ return 0;
+}
+
+static void snd_opl4_seq_free_port(void *private_data)
+{
+ struct snd_opl4 *opl4 = private_data;
+
+ snd_midi_channel_free_set(opl4->chset);
+}
+
+static int snd_opl4_seq_new_device(struct snd_seq_device *dev)
+{
+ struct snd_opl4 *opl4;
+ int client;
+ struct snd_seq_port_callback pcallbacks;
+
+ opl4 = *(struct snd_opl4 **)SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ if (!opl4)
+ return -EINVAL;
+
+ if (snd_yrw801_detect(opl4) < 0)
+ return -ENODEV;
+
+ opl4->chset = snd_midi_channel_alloc_set(16);
+ if (!opl4->chset)
+ return -ENOMEM;
+ opl4->chset->private_data = opl4;
+
+ /* allocate new client */
+ client = snd_seq_create_kernel_client(opl4->card, opl4->seq_dev_num,
+ "OPL4 Wavetable");
+ if (client < 0) {
+ snd_midi_channel_free_set(opl4->chset);
+ return client;
+ }
+ opl4->seq_client = client;
+ opl4->chset->client = client;
+
+ /* create new port */
+ memset(&pcallbacks, 0, sizeof(pcallbacks));
+ pcallbacks.owner = THIS_MODULE;
+ pcallbacks.use = snd_opl4_seq_use;
+ pcallbacks.unuse = snd_opl4_seq_unuse;
+ pcallbacks.event_input = snd_opl4_seq_event_input;
+ pcallbacks.private_free = snd_opl4_seq_free_port;
+ pcallbacks.private_data = opl4;
+
+ opl4->chset->port = snd_seq_event_port_attach(client, &pcallbacks,
+ SNDRV_SEQ_PORT_CAP_WRITE |
+ SNDRV_SEQ_PORT_CAP_SUBS_WRITE,
+ SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM |
+ SNDRV_SEQ_PORT_TYPE_HARDWARE |
+ SNDRV_SEQ_PORT_TYPE_SYNTHESIZER,
+ 16, 24,
+ "OPL4 Wavetable Port");
+ if (opl4->chset->port < 0) {
+ int err = opl4->chset->port;
+ snd_midi_channel_free_set(opl4->chset);
+ snd_seq_delete_kernel_client(client);
+ opl4->seq_client = -1;
+ return err;
+ }
+ return 0;
+}
+
+static int snd_opl4_seq_delete_device(struct snd_seq_device *dev)
+{
+ struct snd_opl4 *opl4;
+
+ opl4 = *(struct snd_opl4 **)SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ if (!opl4)
+ return -EINVAL;
+
+ if (opl4->seq_client >= 0) {
+ snd_seq_delete_kernel_client(opl4->seq_client);
+ opl4->seq_client = -1;
+ }
+ return 0;
+}
+
+static int __init alsa_opl4_synth_init(void)
+{
+ static struct snd_seq_dev_ops ops = {
+ snd_opl4_seq_new_device,
+ snd_opl4_seq_delete_device
+ };
+
+ return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL4, &ops,
+ sizeof(struct snd_opl4 *));
+}
+
+static void __exit alsa_opl4_synth_exit(void)
+{
+ snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL4);
+}
+
+module_init(alsa_opl4_synth_init)
+module_exit(alsa_opl4_synth_exit)
diff --git a/sound/drivers/opl4/opl4_synth.c b/sound/drivers/opl4/opl4_synth.c
new file mode 100644
index 0000000..49b9e24
--- /dev/null
+++ b/sound/drivers/opl4/opl4_synth.c
@@ -0,0 +1,634 @@
+/*
+ * OPL4 MIDI synthesizer functions
+ *
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions, and the following disclaimer,
+ * without modification.
+ * 2. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * Alternatively, this software may be distributed and/or modified under the
+ * terms of the GNU General Public License as published by the Free Software
+ * Foundation; either version 2 of the License, or (at your option) any later
+ * version.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ */
+
+#include "opl4_local.h"
+#include <linux/delay.h>
+#include <asm/io.h>
+#include <sound/asoundef.h>
+
+/* GM2 controllers */
+#ifndef MIDI_CTL_RELEASE_TIME
+#define MIDI_CTL_RELEASE_TIME 0x48
+#define MIDI_CTL_ATTACK_TIME 0x49
+#define MIDI_CTL_DECAY_TIME 0x4b
+#define MIDI_CTL_VIBRATO_RATE 0x4c
+#define MIDI_CTL_VIBRATO_DEPTH 0x4d
+#define MIDI_CTL_VIBRATO_DELAY 0x4e
+#endif
+
+/*
+ * This table maps 100/128 cents to F_NUMBER.
+ */
+static const s16 snd_opl4_pitch_map[0x600] = {
+ 0x000,0x000,0x001,0x001,0x002,0x002,0x003,0x003,
+ 0x004,0x004,0x005,0x005,0x006,0x006,0x006,0x007,
+ 0x007,0x008,0x008,0x009,0x009,0x00a,0x00a,0x00b,
+ 0x00b,0x00c,0x00c,0x00d,0x00d,0x00d,0x00e,0x00e,
+ 0x00f,0x00f,0x010,0x010,0x011,0x011,0x012,0x012,
+ 0x013,0x013,0x014,0x014,0x015,0x015,0x015,0x016,
+ 0x016,0x017,0x017,0x018,0x018,0x019,0x019,0x01a,
+ 0x01a,0x01b,0x01b,0x01c,0x01c,0x01d,0x01d,0x01e,
+ 0x01e,0x01e,0x01f,0x01f,0x020,0x020,0x021,0x021,
+ 0x022,0x022,0x023,0x023,0x024,0x024,0x025,0x025,
+ 0x026,0x026,0x027,0x027,0x028,0x028,0x029,0x029,
+ 0x029,0x02a,0x02a,0x02b,0x02b,0x02c,0x02c,0x02d,
+ 0x02d,0x02e,0x02e,0x02f,0x02f,0x030,0x030,0x031,
+ 0x031,0x032,0x032,0x033,0x033,0x034,0x034,0x035,
+ 0x035,0x036,0x036,0x037,0x037,0x038,0x038,0x038,
+ 0x039,0x039,0x03a,0x03a,0x03b,0x03b,0x03c,0x03c,
+ 0x03d,0x03d,0x03e,0x03e,0x03f,0x03f,0x040,0x040,
+ 0x041,0x041,0x042,0x042,0x043,0x043,0x044,0x044,
+ 0x045,0x045,0x046,0x046,0x047,0x047,0x048,0x048,
+ 0x049,0x049,0x04a,0x04a,0x04b,0x04b,0x04c,0x04c,
+ 0x04d,0x04d,0x04e,0x04e,0x04f,0x04f,0x050,0x050,
+ 0x051,0x051,0x052,0x052,0x053,0x053,0x054,0x054,
+ 0x055,0x055,0x056,0x056,0x057,0x057,0x058,0x058,
+ 0x059,0x059,0x05a,0x05a,0x05b,0x05b,0x05c,0x05c,
+ 0x05d,0x05d,0x05e,0x05e,0x05f,0x05f,0x060,0x060,
+ 0x061,0x061,0x062,0x062,0x063,0x063,0x064,0x064,
+ 0x065,0x065,0x066,0x066,0x067,0x067,0x068,0x068,
+ 0x069,0x069,0x06a,0x06a,0x06b,0x06b,0x06c,0x06c,
+ 0x06d,0x06d,0x06e,0x06e,0x06f,0x06f,0x070,0x071,
+ 0x071,0x072,0x072,0x073,0x073,0x074,0x074,0x075,
+ 0x075,0x076,0x076,0x077,0x077,0x078,0x078,0x079,
+ 0x079,0x07a,0x07a,0x07b,0x07b,0x07c,0x07c,0x07d,
+ 0x07d,0x07e,0x07e,0x07f,0x07f,0x080,0x081,0x081,
+ 0x082,0x082,0x083,0x083,0x084,0x084,0x085,0x085,
+ 0x086,0x086,0x087,0x087,0x088,0x088,0x089,0x089,
+ 0x08a,0x08a,0x08b,0x08b,0x08c,0x08d,0x08d,0x08e,
+ 0x08e,0x08f,0x08f,0x090,0x090,0x091,0x091,0x092,
+ 0x092,0x093,0x093,0x094,0x094,0x095,0x096,0x096,
+ 0x097,0x097,0x098,0x098,0x099,0x099,0x09a,0x09a,
+ 0x09b,0x09b,0x09c,0x09c,0x09d,0x09d,0x09e,0x09f,
+ 0x09f,0x0a0,0x0a0,0x0a1,0x0a1,0x0a2,0x0a2,0x0a3,
+ 0x0a3,0x0a4,0x0a4,0x0a5,0x0a6,0x0a6,0x0a7,0x0a7,
+ 0x0a8,0x0a8,0x0a9,0x0a9,0x0aa,0x0aa,0x0ab,0x0ab,
+ 0x0ac,0x0ad,0x0ad,0x0ae,0x0ae,0x0af,0x0af,0x0b0,
+ 0x0b0,0x0b1,0x0b1,0x0b2,0x0b2,0x0b3,0x0b4,0x0b4,
+ 0x0b5,0x0b5,0x0b6,0x0b6,0x0b7,0x0b7,0x0b8,0x0b8,
+ 0x0b9,0x0ba,0x0ba,0x0bb,0x0bb,0x0bc,0x0bc,0x0bd,
+ 0x0bd,0x0be,0x0be,0x0bf,0x0c0,0x0c0,0x0c1,0x0c1,
+ 0x0c2,0x0c2,0x0c3,0x0c3,0x0c4,0x0c4,0x0c5,0x0c6,
+ 0x0c6,0x0c7,0x0c7,0x0c8,0x0c8,0x0c9,0x0c9,0x0ca,
+ 0x0cb,0x0cb,0x0cc,0x0cc,0x0cd,0x0cd,0x0ce,0x0ce,
+ 0x0cf,0x0d0,0x0d0,0x0d1,0x0d1,0x0d2,0x0d2,0x0d3,
+ 0x0d3,0x0d4,0x0d5,0x0d5,0x0d6,0x0d6,0x0d7,0x0d7,
+ 0x0d8,0x0d8,0x0d9,0x0da,0x0da,0x0db,0x0db,0x0dc,
+ 0x0dc,0x0dd,0x0de,0x0de,0x0df,0x0df,0x0e0,0x0e0,
+ 0x0e1,0x0e1,0x0e2,0x0e3,0x0e3,0x0e4,0x0e4,0x0e5,
+ 0x0e5,0x0e6,0x0e7,0x0e7,0x0e8,0x0e8,0x0e9,0x0e9,
+ 0x0ea,0x0eb,0x0eb,0x0ec,0x0ec,0x0ed,0x0ed,0x0ee,
+ 0x0ef,0x0ef,0x0f0,0x0f0,0x0f1,0x0f1,0x0f2,0x0f3,
+ 0x0f3,0x0f4,0x0f4,0x0f5,0x0f5,0x0f6,0x0f7,0x0f7,
+ 0x0f8,0x0f8,0x0f9,0x0f9,0x0fa,0x0fb,0x0fb,0x0fc,
+ 0x0fc,0x0fd,0x0fd,0x0fe,0x0ff,0x0ff,0x100,0x100,
+ 0x101,0x101,0x102,0x103,0x103,0x104,0x104,0x105,
+ 0x106,0x106,0x107,0x107,0x108,0x108,0x109,0x10a,
+ 0x10a,0x10b,0x10b,0x10c,0x10c,0x10d,0x10e,0x10e,
+ 0x10f,0x10f,0x110,0x111,0x111,0x112,0x112,0x113,
+ 0x114,0x114,0x115,0x115,0x116,0x116,0x117,0x118,
+ 0x118,0x119,0x119,0x11a,0x11b,0x11b,0x11c,0x11c,
+ 0x11d,0x11e,0x11e,0x11f,0x11f,0x120,0x120,0x121,
+ 0x122,0x122,0x123,0x123,0x124,0x125,0x125,0x126,
+ 0x126,0x127,0x128,0x128,0x129,0x129,0x12a,0x12b,
+ 0x12b,0x12c,0x12c,0x12d,0x12e,0x12e,0x12f,0x12f,
+ 0x130,0x131,0x131,0x132,0x132,0x133,0x134,0x134,
+ 0x135,0x135,0x136,0x137,0x137,0x138,0x138,0x139,
+ 0x13a,0x13a,0x13b,0x13b,0x13c,0x13d,0x13d,0x13e,
+ 0x13e,0x13f,0x140,0x140,0x141,0x141,0x142,0x143,
+ 0x143,0x144,0x144,0x145,0x146,0x146,0x147,0x148,
+ 0x148,0x149,0x149,0x14a,0x14b,0x14b,0x14c,0x14c,
+ 0x14d,0x14e,0x14e,0x14f,0x14f,0x150,0x151,0x151,
+ 0x152,0x153,0x153,0x154,0x154,0x155,0x156,0x156,
+ 0x157,0x157,0x158,0x159,0x159,0x15a,0x15b,0x15b,
+ 0x15c,0x15c,0x15d,0x15e,0x15e,0x15f,0x160,0x160,
+ 0x161,0x161,0x162,0x163,0x163,0x164,0x165,0x165,
+ 0x166,0x166,0x167,0x168,0x168,0x169,0x16a,0x16a,
+ 0x16b,0x16b,0x16c,0x16d,0x16d,0x16e,0x16f,0x16f,
+ 0x170,0x170,0x171,0x172,0x172,0x173,0x174,0x174,
+ 0x175,0x175,0x176,0x177,0x177,0x178,0x179,0x179,
+ 0x17a,0x17a,0x17b,0x17c,0x17c,0x17d,0x17e,0x17e,
+ 0x17f,0x180,0x180,0x181,0x181,0x182,0x183,0x183,
+ 0x184,0x185,0x185,0x186,0x187,0x187,0x188,0x188,
+ 0x189,0x18a,0x18a,0x18b,0x18c,0x18c,0x18d,0x18e,
+ 0x18e,0x18f,0x190,0x190,0x191,0x191,0x192,0x193,
+ 0x193,0x194,0x195,0x195,0x196,0x197,0x197,0x198,
+ 0x199,0x199,0x19a,0x19a,0x19b,0x19c,0x19c,0x19d,
+ 0x19e,0x19e,0x19f,0x1a0,0x1a0,0x1a1,0x1a2,0x1a2,
+ 0x1a3,0x1a4,0x1a4,0x1a5,0x1a6,0x1a6,0x1a7,0x1a8,
+ 0x1a8,0x1a9,0x1a9,0x1aa,0x1ab,0x1ab,0x1ac,0x1ad,
+ 0x1ad,0x1ae,0x1af,0x1af,0x1b0,0x1b1,0x1b1,0x1b2,
+ 0x1b3,0x1b3,0x1b4,0x1b5,0x1b5,0x1b6,0x1b7,0x1b7,
+ 0x1b8,0x1b9,0x1b9,0x1ba,0x1bb,0x1bb,0x1bc,0x1bd,
+ 0x1bd,0x1be,0x1bf,0x1bf,0x1c0,0x1c1,0x1c1,0x1c2,
+ 0x1c3,0x1c3,0x1c4,0x1c5,0x1c5,0x1c6,0x1c7,0x1c7,
+ 0x1c8,0x1c9,0x1c9,0x1ca,0x1cb,0x1cb,0x1cc,0x1cd,
+ 0x1cd,0x1ce,0x1cf,0x1cf,0x1d0,0x1d1,0x1d1,0x1d2,
+ 0x1d3,0x1d3,0x1d4,0x1d5,0x1d5,0x1d6,0x1d7,0x1d7,
+ 0x1d8,0x1d9,0x1d9,0x1da,0x1db,0x1db,0x1dc,0x1dd,
+ 0x1dd,0x1de,0x1df,0x1df,0x1e0,0x1e1,0x1e1,0x1e2,
+ 0x1e3,0x1e4,0x1e4,0x1e5,0x1e6,0x1e6,0x1e7,0x1e8,
+ 0x1e8,0x1e9,0x1ea,0x1ea,0x1eb,0x1ec,0x1ec,0x1ed,
+ 0x1ee,0x1ee,0x1ef,0x1f0,0x1f0,0x1f1,0x1f2,0x1f3,
+ 0x1f3,0x1f4,0x1f5,0x1f5,0x1f6,0x1f7,0x1f7,0x1f8,
+ 0x1f9,0x1f9,0x1fa,0x1fb,0x1fb,0x1fc,0x1fd,0x1fe,
+ 0x1fe,0x1ff,0x200,0x200,0x201,0x202,0x202,0x203,
+ 0x204,0x205,0x205,0x206,0x207,0x207,0x208,0x209,
+ 0x209,0x20a,0x20b,0x20b,0x20c,0x20d,0x20e,0x20e,
+ 0x20f,0x210,0x210,0x211,0x212,0x212,0x213,0x214,
+ 0x215,0x215,0x216,0x217,0x217,0x218,0x219,0x21a,
+ 0x21a,0x21b,0x21c,0x21c,0x21d,0x21e,0x21e,0x21f,
+ 0x220,0x221,0x221,0x222,0x223,0x223,0x224,0x225,
+ 0x226,0x226,0x227,0x228,0x228,0x229,0x22a,0x22b,
+ 0x22b,0x22c,0x22d,0x22d,0x22e,0x22f,0x230,0x230,
+ 0x231,0x232,0x232,0x233,0x234,0x235,0x235,0x236,
+ 0x237,0x237,0x238,0x239,0x23a,0x23a,0x23b,0x23c,
+ 0x23c,0x23d,0x23e,0x23f,0x23f,0x240,0x241,0x241,
+ 0x242,0x243,0x244,0x244,0x245,0x246,0x247,0x247,
+ 0x248,0x249,0x249,0x24a,0x24b,0x24c,0x24c,0x24d,
+ 0x24e,0x24f,0x24f,0x250,0x251,0x251,0x252,0x253,
+ 0x254,0x254,0x255,0x256,0x257,0x257,0x258,0x259,
+ 0x259,0x25a,0x25b,0x25c,0x25c,0x25d,0x25e,0x25f,
+ 0x25f,0x260,0x261,0x262,0x262,0x263,0x264,0x265,
+ 0x265,0x266,0x267,0x267,0x268,0x269,0x26a,0x26a,
+ 0x26b,0x26c,0x26d,0x26d,0x26e,0x26f,0x270,0x270,
+ 0x271,0x272,0x273,0x273,0x274,0x275,0x276,0x276,
+ 0x277,0x278,0x279,0x279,0x27a,0x27b,0x27c,0x27c,
+ 0x27d,0x27e,0x27f,0x27f,0x280,0x281,0x282,0x282,
+ 0x283,0x284,0x285,0x285,0x286,0x287,0x288,0x288,
+ 0x289,0x28a,0x28b,0x28b,0x28c,0x28d,0x28e,0x28e,
+ 0x28f,0x290,0x291,0x291,0x292,0x293,0x294,0x294,
+ 0x295,0x296,0x297,0x298,0x298,0x299,0x29a,0x29b,
+ 0x29b,0x29c,0x29d,0x29e,0x29e,0x29f,0x2a0,0x2a1,
+ 0x2a1,0x2a2,0x2a3,0x2a4,0x2a5,0x2a5,0x2a6,0x2a7,
+ 0x2a8,0x2a8,0x2a9,0x2aa,0x2ab,0x2ab,0x2ac,0x2ad,
+ 0x2ae,0x2af,0x2af,0x2b0,0x2b1,0x2b2,0x2b2,0x2b3,
+ 0x2b4,0x2b5,0x2b5,0x2b6,0x2b7,0x2b8,0x2b9,0x2b9,
+ 0x2ba,0x2bb,0x2bc,0x2bc,0x2bd,0x2be,0x2bf,0x2c0,
+ 0x2c0,0x2c1,0x2c2,0x2c3,0x2c4,0x2c4,0x2c5,0x2c6,
+ 0x2c7,0x2c7,0x2c8,0x2c9,0x2ca,0x2cb,0x2cb,0x2cc,
+ 0x2cd,0x2ce,0x2ce,0x2cf,0x2d0,0x2d1,0x2d2,0x2d2,
+ 0x2d3,0x2d4,0x2d5,0x2d6,0x2d6,0x2d7,0x2d8,0x2d9,
+ 0x2da,0x2da,0x2db,0x2dc,0x2dd,0x2dd,0x2de,0x2df,
+ 0x2e0,0x2e1,0x2e1,0x2e2,0x2e3,0x2e4,0x2e5,0x2e5,
+ 0x2e6,0x2e7,0x2e8,0x2e9,0x2e9,0x2ea,0x2eb,0x2ec,
+ 0x2ed,0x2ed,0x2ee,0x2ef,0x2f0,0x2f1,0x2f1,0x2f2,
+ 0x2f3,0x2f4,0x2f5,0x2f5,0x2f6,0x2f7,0x2f8,0x2f9,
+ 0x2f9,0x2fa,0x2fb,0x2fc,0x2fd,0x2fd,0x2fe,0x2ff,
+ 0x300,0x301,0x302,0x302,0x303,0x304,0x305,0x306,
+ 0x306,0x307,0x308,0x309,0x30a,0x30a,0x30b,0x30c,
+ 0x30d,0x30e,0x30f,0x30f,0x310,0x311,0x312,0x313,
+ 0x313,0x314,0x315,0x316,0x317,0x318,0x318,0x319,
+ 0x31a,0x31b,0x31c,0x31c,0x31d,0x31e,0x31f,0x320,
+ 0x321,0x321,0x322,0x323,0x324,0x325,0x326,0x326,
+ 0x327,0x328,0x329,0x32a,0x32a,0x32b,0x32c,0x32d,
+ 0x32e,0x32f,0x32f,0x330,0x331,0x332,0x333,0x334,
+ 0x334,0x335,0x336,0x337,0x338,0x339,0x339,0x33a,
+ 0x33b,0x33c,0x33d,0x33e,0x33e,0x33f,0x340,0x341,
+ 0x342,0x343,0x343,0x344,0x345,0x346,0x347,0x348,
+ 0x349,0x349,0x34a,0x34b,0x34c,0x34d,0x34e,0x34e,
+ 0x34f,0x350,0x351,0x352,0x353,0x353,0x354,0x355,
+ 0x356,0x357,0x358,0x359,0x359,0x35a,0x35b,0x35c,
+ 0x35d,0x35e,0x35f,0x35f,0x360,0x361,0x362,0x363,
+ 0x364,0x364,0x365,0x366,0x367,0x368,0x369,0x36a,
+ 0x36a,0x36b,0x36c,0x36d,0x36e,0x36f,0x370,0x370,
+ 0x371,0x372,0x373,0x374,0x375,0x376,0x377,0x377,
+ 0x378,0x379,0x37a,0x37b,0x37c,0x37d,0x37d,0x37e,
+ 0x37f,0x380,0x381,0x382,0x383,0x383,0x384,0x385,
+ 0x386,0x387,0x388,0x389,0x38a,0x38a,0x38b,0x38c,
+ 0x38d,0x38e,0x38f,0x390,0x391,0x391,0x392,0x393,
+ 0x394,0x395,0x396,0x397,0x398,0x398,0x399,0x39a,
+ 0x39b,0x39c,0x39d,0x39e,0x39f,0x39f,0x3a0,0x3a1,
+ 0x3a2,0x3a3,0x3a4,0x3a5,0x3a6,0x3a7,0x3a7,0x3a8,
+ 0x3a9,0x3aa,0x3ab,0x3ac,0x3ad,0x3ae,0x3ae,0x3af,
+ 0x3b0,0x3b1,0x3b2,0x3b3,0x3b4,0x3b5,0x3b6,0x3b6,
+ 0x3b7,0x3b8,0x3b9,0x3ba,0x3bb,0x3bc,0x3bd,0x3be,
+ 0x3bf,0x3bf,0x3c0,0x3c1,0x3c2,0x3c3,0x3c4,0x3c5,
+ 0x3c6,0x3c7,0x3c7,0x3c8,0x3c9,0x3ca,0x3cb,0x3cc,
+ 0x3cd,0x3ce,0x3cf,0x3d0,0x3d1,0x3d1,0x3d2,0x3d3,
+ 0x3d4,0x3d5,0x3d6,0x3d7,0x3d8,0x3d9,0x3da,0x3da,
+ 0x3db,0x3dc,0x3dd,0x3de,0x3df,0x3e0,0x3e1,0x3e2,
+ 0x3e3,0x3e4,0x3e4,0x3e5,0x3e6,0x3e7,0x3e8,0x3e9,
+ 0x3ea,0x3eb,0x3ec,0x3ed,0x3ee,0x3ef,0x3ef,0x3f0,
+ 0x3f1,0x3f2,0x3f3,0x3f4,0x3f5,0x3f6,0x3f7,0x3f8,
+ 0x3f9,0x3fa,0x3fa,0x3fb,0x3fc,0x3fd,0x3fe,0x3ff
+};
+
+/*
+ * Attenuation according to GM recommendations, in -0.375 dB units.
+ * table[v] = 40 * log(v / 127) / -0.375
+ */
+static unsigned char snd_opl4_volume_table[128] = {
+ 255,224,192,173,160,150,141,134,
+ 128,122,117,113,109,105,102, 99,
+ 96, 93, 90, 88, 85, 83, 81, 79,
+ 77, 75, 73, 71, 70, 68, 67, 65,
+ 64, 62, 61, 59, 58, 57, 56, 54,
+ 53, 52, 51, 50, 49, 48, 47, 46,
+ 45, 44, 43, 42, 41, 40, 39, 39,
+ 38, 37, 36, 35, 34, 34, 33, 32,
+ 31, 31, 30, 29, 29, 28, 27, 27,
+ 26, 25, 25, 24, 24, 23, 22, 22,
+ 21, 21, 20, 19, 19, 18, 18, 17,
+ 17, 16, 16, 15, 15, 14, 14, 13,
+ 13, 12, 12, 11, 11, 10, 10, 9,
+ 9, 9, 8, 8, 7, 7, 6, 6,
+ 6, 5, 5, 4, 4, 4, 3, 3,
+ 2, 2, 2, 1, 1, 0, 0, 0
+};
+
+/*
+ * Initializes all voices.
+ */
+void snd_opl4_synth_reset(struct snd_opl4 *opl4)
+{
+ unsigned long flags;
+ int i;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ for (i = 0; i < OPL4_MAX_VOICES; i++)
+ snd_opl4_write(opl4, OPL4_REG_MISC + i, OPL4_DAMP_BIT);
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+
+ INIT_LIST_HEAD(&opl4->off_voices);
+ INIT_LIST_HEAD(&opl4->on_voices);
+ memset(opl4->voices, 0, sizeof(opl4->voices));
+ for (i = 0; i < OPL4_MAX_VOICES; i++) {
+ opl4->voices[i].number = i;
+ list_add_tail(&opl4->voices[i].list, &opl4->off_voices);
+ }
+
+ snd_midi_channel_set_clear(opl4->chset);
+}
+
+/*
+ * Shuts down all voices.
+ */
+void snd_opl4_synth_shutdown(struct snd_opl4 *opl4)
+{
+ unsigned long flags;
+ int i;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ for (i = 0; i < OPL4_MAX_VOICES; i++)
+ snd_opl4_write(opl4, OPL4_REG_MISC + i,
+ opl4->voices[i].reg_misc & ~OPL4_KEY_ON_BIT);
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+}
+
+/*
+ * Executes the callback for all voices playing the specified note.
+ */
+static void snd_opl4_do_for_note(struct snd_opl4 *opl4, int note, struct snd_midi_channel *chan,
+ void (*func)(struct snd_opl4 *opl4, struct opl4_voice *voice))
+{
+ int i;
+ unsigned long flags;
+ struct opl4_voice *voice;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ for (i = 0; i < OPL4_MAX_VOICES; i++) {
+ voice = &opl4->voices[i];
+ if (voice->chan == chan && voice->note == note) {
+ func(opl4, voice);
+ }
+ }
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+}
+
+/*
+ * Executes the callback for all voices of to the specified channel.
+ */
+static void snd_opl4_do_for_channel(struct snd_opl4 *opl4,
+ struct snd_midi_channel *chan,
+ void (*func)(struct snd_opl4 *opl4, struct opl4_voice *voice))
+{
+ int i;
+ unsigned long flags;
+ struct opl4_voice *voice;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ for (i = 0; i < OPL4_MAX_VOICES; i++) {
+ voice = &opl4->voices[i];
+ if (voice->chan == chan) {
+ func(opl4, voice);
+ }
+ }
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+}
+
+/*
+ * Executes the callback for all active voices.
+ */
+static void snd_opl4_do_for_all(struct snd_opl4 *opl4,
+ void (*func)(struct snd_opl4 *opl4, struct opl4_voice *voice))
+{
+ int i;
+ unsigned long flags;
+ struct opl4_voice *voice;
+
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ for (i = 0; i < OPL4_MAX_VOICES; i++) {
+ voice = &opl4->voices[i];
+ if (voice->chan)
+ func(opl4, voice);
+ }
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+}
+
+static void snd_opl4_update_volume(struct snd_opl4 *opl4, struct opl4_voice *voice)
+{
+ int att;
+
+ att = voice->sound->tone_attenuate;
+ att += snd_opl4_volume_table[opl4->chset->gs_master_volume & 0x7f];
+ att += snd_opl4_volume_table[voice->chan->gm_volume & 0x7f];
+ att += snd_opl4_volume_table[voice->chan->gm_expression & 0x7f];
+ att += snd_opl4_volume_table[voice->velocity];
+ att = 0x7f - (0x7f - att) * (voice->sound->volume_factor) / 0xfe - volume_boost;
+ if (att < 0)
+ att = 0;
+ else if (att > 0x7e)
+ att = 0x7e;
+ snd_opl4_write(opl4, OPL4_REG_LEVEL + voice->number,
+ (att << 1) | voice->level_direct);
+ voice->level_direct = 0;
+}
+
+static void snd_opl4_update_pan(struct snd_opl4 *opl4, struct opl4_voice *voice)
+{
+ int pan = voice->sound->panpot;
+
+ if (!voice->chan->drum_channel)
+ pan += (voice->chan->control[MIDI_CTL_MSB_PAN] - 0x40) >> 3;
+ if (pan < -7)
+ pan = -7;
+ else if (pan > 7)
+ pan = 7;
+ voice->reg_misc = (voice->reg_misc & ~OPL4_PAN_POT_MASK)
+ | (pan & OPL4_PAN_POT_MASK);
+ snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc);
+}
+
+static void snd_opl4_update_vibrato_depth(struct snd_opl4 *opl4,
+ struct opl4_voice *voice)
+{
+ int depth;
+
+ if (voice->chan->drum_channel)
+ return;
+ depth = (7 - voice->sound->vibrato)
+ * (voice->chan->control[MIDI_CTL_VIBRATO_DEPTH] & 0x7f);
+ depth = (depth >> 7) + voice->sound->vibrato;
+ voice->reg_lfo_vibrato &= ~OPL4_VIBRATO_DEPTH_MASK;
+ voice->reg_lfo_vibrato |= depth & OPL4_VIBRATO_DEPTH_MASK;
+ snd_opl4_write(opl4, OPL4_REG_LFO_VIBRATO + voice->number,
+ voice->reg_lfo_vibrato);
+}
+
+static void snd_opl4_update_pitch(struct snd_opl4 *opl4,
+ struct opl4_voice *voice)
+{
+ struct snd_midi_channel *chan = voice->chan;
+ int note, pitch, octave;
+
+ note = chan->drum_channel ? 60 : voice->note;
+ /*
+ * pitch is in 100/128 cents, so 0x80 is one semitone and
+ * 0x600 is one octave.
+ */
+ pitch = ((note - 60) << 7) * voice->sound->key_scaling / 100 + (60 << 7);
+ pitch += voice->sound->pitch_offset;
+ if (!chan->drum_channel)
+ pitch += chan->gm_rpn_coarse_tuning;
+ pitch += chan->gm_rpn_fine_tuning >> 7;
+ pitch += chan->midi_pitchbend * chan->gm_rpn_pitch_bend_range / 0x2000;
+ if (pitch < 0)
+ pitch = 0;
+ else if (pitch >= 0x6000)
+ pitch = 0x5fff;
+ octave = pitch / 0x600 - 8;
+ pitch = snd_opl4_pitch_map[pitch % 0x600];
+
+ snd_opl4_write(opl4, OPL4_REG_OCTAVE + voice->number,
+ (octave << 4) | ((pitch >> 7) & OPL4_F_NUMBER_HIGH_MASK));
+ voice->reg_f_number = (voice->reg_f_number & OPL4_TONE_NUMBER_BIT8)
+ | ((pitch << 1) & OPL4_F_NUMBER_LOW_MASK);
+ snd_opl4_write(opl4, OPL4_REG_F_NUMBER + voice->number, voice->reg_f_number);
+}
+
+static void snd_opl4_update_tone_parameters(struct snd_opl4 *opl4,
+ struct opl4_voice *voice)
+{
+ snd_opl4_write(opl4, OPL4_REG_ATTACK_DECAY1 + voice->number,
+ voice->sound->reg_attack_decay1);
+ snd_opl4_write(opl4, OPL4_REG_LEVEL_DECAY2 + voice->number,
+ voice->sound->reg_level_decay2);
+ snd_opl4_write(opl4, OPL4_REG_RELEASE_CORRECTION + voice->number,
+ voice->sound->reg_release_correction);
+ snd_opl4_write(opl4, OPL4_REG_TREMOLO + voice->number,
+ voice->sound->reg_tremolo);
+}
+
+/* allocate one voice */
+static struct opl4_voice *snd_opl4_get_voice(struct snd_opl4 *opl4)
+{
+ /* first, try to get the oldest key-off voice */
+ if (!list_empty(&opl4->off_voices))
+ return list_entry(opl4->off_voices.next, struct opl4_voice, list);
+ /* then get the oldest key-on voice */
+ snd_BUG_ON(list_empty(&opl4->on_voices));
+ return list_entry(opl4->on_voices.next, struct opl4_voice, list);
+}
+
+static void snd_opl4_wait_for_wave_headers(struct snd_opl4 *opl4)
+{
+ int timeout = 200;
+
+ while ((inb(opl4->fm_port) & OPL4_STATUS_LOAD) && --timeout > 0)
+ udelay(10);
+}
+
+void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_channel *chan)
+{
+ struct snd_opl4 *opl4 = private_data;
+ const struct opl4_region_ptr *regions;
+ struct opl4_voice *voice[2];
+ const struct opl4_sound *sound[2];
+ int voices = 0, i;
+ unsigned long flags;
+
+ /* determine the number of voices and voice parameters */
+ i = chan->drum_channel ? 0x80 : (chan->midi_program & 0x7f);
+ regions = &snd_yrw801_regions[i];
+ for (i = 0; i < regions->count; i++) {
+ if (note >= regions->regions[i].key_min &&
+ note <= regions->regions[i].key_max) {
+ sound[voices] = &regions->regions[i].sound;
+ if (++voices >= 2)
+ break;
+ }
+ }
+
+ /* allocate and initialize the needed voices */
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ for (i = 0; i < voices; i++) {
+ voice[i] = snd_opl4_get_voice(opl4);
+ list_del(&voice[i]->list);
+ list_add_tail(&voice[i]->list, &opl4->on_voices);
+ voice[i]->chan = chan;
+ voice[i]->note = note;
+ voice[i]->velocity = vel & 0x7f;
+ voice[i]->sound = sound[i];
+ }
+
+ /* set tone number (triggers header loading) */
+ for (i = 0; i < voices; i++) {
+ voice[i]->reg_f_number =
+ (sound[i]->tone >> 8) & OPL4_TONE_NUMBER_BIT8;
+ snd_opl4_write(opl4, OPL4_REG_F_NUMBER + voice[i]->number,
+ voice[i]->reg_f_number);
+ snd_opl4_write(opl4, OPL4_REG_TONE_NUMBER + voice[i]->number,
+ sound[i]->tone & 0xff);
+ }
+
+ /* set parameters which can be set while loading */
+ for (i = 0; i < voices; i++) {
+ voice[i]->reg_misc = OPL4_LFO_RESET_BIT;
+ snd_opl4_update_pan(opl4, voice[i]);
+ snd_opl4_update_pitch(opl4, voice[i]);
+ voice[i]->level_direct = OPL4_LEVEL_DIRECT_BIT;
+ snd_opl4_update_volume(opl4, voice[i]);
+ }
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+
+ /* wait for completion of loading */
+ snd_opl4_wait_for_wave_headers(opl4);
+
+ /* set remaining parameters */
+ spin_lock_irqsave(&opl4->reg_lock, flags);
+ for (i = 0; i < voices; i++) {
+ snd_opl4_update_tone_parameters(opl4, voice[i]);
+ voice[i]->reg_lfo_vibrato = voice[i]->sound->reg_lfo_vibrato;
+ snd_opl4_update_vibrato_depth(opl4, voice[i]);
+ }
+
+ /* finally, switch on all voices */
+ for (i = 0; i < voices; i++) {
+ voice[i]->reg_misc =
+ (voice[i]->reg_misc & 0x1f) | OPL4_KEY_ON_BIT;
+ snd_opl4_write(opl4, OPL4_REG_MISC + voice[i]->number,
+ voice[i]->reg_misc);
+ }
+ spin_unlock_irqrestore(&opl4->reg_lock, flags);
+}
+
+static void snd_opl4_voice_off(struct snd_opl4 *opl4, struct opl4_voice *voice)
+{
+ list_del(&voice->list);
+ list_add_tail(&voice->list, &opl4->off_voices);
+
+ voice->reg_misc &= ~OPL4_KEY_ON_BIT;
+ snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc);
+}
+
+void snd_opl4_note_off(void *private_data, int note, int vel, struct snd_midi_channel *chan)
+{
+ struct snd_opl4 *opl4 = private_data;
+
+ snd_opl4_do_for_note(opl4, note, chan, snd_opl4_voice_off);
+}
+
+static void snd_opl4_terminate_voice(struct snd_opl4 *opl4, struct opl4_voice *voice)
+{
+ list_del(&voice->list);
+ list_add_tail(&voice->list, &opl4->off_voices);
+
+ voice->reg_misc = (voice->reg_misc & ~OPL4_KEY_ON_BIT) | OPL4_DAMP_BIT;
+ snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc);
+}
+
+void snd_opl4_terminate_note(void *private_data, int note, struct snd_midi_channel *chan)
+{
+ struct snd_opl4 *opl4 = private_data;
+
+ snd_opl4_do_for_note(opl4, note, chan, snd_opl4_terminate_voice);
+}
+
+void snd_opl4_control(void *private_data, int type, struct snd_midi_channel *chan)
+{
+ struct snd_opl4 *opl4 = private_data;
+
+ switch (type) {
+ case MIDI_CTL_MSB_MODWHEEL:
+ chan->control[MIDI_CTL_VIBRATO_DEPTH] = chan->control[MIDI_CTL_MSB_MODWHEEL];
+ snd_opl4_do_for_channel(opl4, chan, snd_opl4_update_vibrato_depth);
+ break;
+ case MIDI_CTL_MSB_MAIN_VOLUME:
+ snd_opl4_do_for_channel(opl4, chan, snd_opl4_update_volume);
+ break;
+ case MIDI_CTL_MSB_PAN:
+ snd_opl4_do_for_channel(opl4, chan, snd_opl4_update_pan);
+ break;
+ case MIDI_CTL_MSB_EXPRESSION:
+ snd_opl4_do_for_channel(opl4, chan, snd_opl4_update_volume);
+ break;
+ case MIDI_CTL_VIBRATO_RATE:
+ /* not yet supported */
+ break;
+ case MIDI_CTL_VIBRATO_DEPTH:
+ snd_opl4_do_for_channel(opl4, chan, snd_opl4_update_vibrato_depth);
+ break;
+ case MIDI_CTL_VIBRATO_DELAY:
+ /* not yet supported */
+ break;
+ case MIDI_CTL_E1_REVERB_DEPTH:
+ /*
+ * Each OPL4 voice has a bit called "Pseudo-Reverb", but
+ * IMHO _not_ using it enhances the listening experience.
+ */
+ break;
+ case MIDI_CTL_PITCHBEND:
+ snd_opl4_do_for_channel(opl4, chan, snd_opl4_update_pitch);
+ break;
+ }
+}
+
+void snd_opl4_sysex(void *private_data, unsigned char *buf, int len,
+ int parsed, struct snd_midi_channel_set *chset)
+{
+ struct snd_opl4 *opl4 = private_data;
+
+ if (parsed == SNDRV_MIDI_SYSEX_GS_MASTER_VOLUME)
+ snd_opl4_do_for_all(opl4, snd_opl4_update_volume);
+}
diff --git a/sound/drivers/opl4/yrw801.c b/sound/drivers/opl4/yrw801.c
new file mode 100644
index 0000000..6c33549
--- /dev/null
+++ b/sound/drivers/opl4/yrw801.c
@@ -0,0 +1,961 @@
+/*
+ * Information about the Yamaha YRW801 wavetable ROM chip
+ *
+ * Copyright (c) 2003 by Clemens Ladisch <clemens@ladisch.de>
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions, and the following disclaimer,
+ * without modification.
+ * 2. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * Alternatively, this software may be distributed and/or modified under the
+ * terms of the GNU General Public License as published by the Free Software
+ * Foundation; either version 2 of the License, or (at your option) any later
+ * version.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ */
+
+#include "opl4_local.h"
+
+int snd_yrw801_detect(struct snd_opl4 *opl4)
+{
+ char buf[15];
+
+ snd_opl4_read_memory(opl4, buf, 0x001200, 15);
+ if (memcmp(buf, "CopyrightYAMAHA", 15))
+ return -ENODEV;
+ snd_opl4_read_memory(opl4, buf, 0x1ffffe, 2);
+ if (buf[0] != 0x01)
+ return -ENODEV;
+ snd_printdd("YRW801 ROM version %02x.%02x\n", buf[0], buf[1]);
+ return 0;
+}
+
+/*
+ * The instrument definitions are stored statically because, in practice, the
+ * OPL4 is always coupled with a YRW801. Dynamic instrument loading would be
+ * required if downloading sample data to external SRAM was actually supported
+ * by this driver.
+ */
+
+static const struct opl4_region regions_00[] = { /* Acoustic Grand Piano */
+ {0x14, 0x27, {0x12c,7474,100, 0,0,0x00,0xc8,0x20,0xf2,0x13,0x08,0x0}},
+ {0x28, 0x2d, {0x12d,6816,100, 0,0,0x00,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x2e, 0x33, {0x12e,5899,100, 0,0,0x00,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x34, 0x39, {0x12f,5290,100, 0,0,0x00,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x3a, 0x3f, {0x130,4260,100, 0,0,0x0a,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x40, 0x45, {0x131,3625,100, 0,0,0x0a,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x46, 0x4b, {0x132,3116,100, 0,0,0x04,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x4c, 0x52, {0x133,2081,100, 0,0,0x03,0xc8,0x20,0xf2,0x14,0x18,0x0}},
+ {0x53, 0x58, {0x134,1444,100, 0,0,0x07,0xc8,0x20,0xf3,0x14,0x18,0x0}},
+ {0x59, 0x6d, {0x135,1915,100, 0,0,0x00,0xc8,0x20,0xf4,0x15,0x08,0x0}}
+};
+static const struct opl4_region regions_01[] = { /* Bright Acoustic Piano */
+ {0x14, 0x2d, {0x12c,7474,100, 0,0,0x00,0xc8,0x20,0xf2,0x13,0x08,0x0}},
+ {0x2e, 0x33, {0x12d,6816,100, 0,0,0x00,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x34, 0x39, {0x12e,5899,100, 0,0,0x00,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x3a, 0x3f, {0x12f,5290,100, 0,0,0x00,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x40, 0x45, {0x130,4260,100, 0,0,0x0a,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x46, 0x4b, {0x131,3625,100, 0,0,0x0a,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x4c, 0x52, {0x132,3116,100, 0,0,0x04,0xc8,0x20,0xf2,0x14,0x08,0x0}},
+ {0x53, 0x58, {0x133,2081,100, 0,0,0x07,0xc8,0x20,0xf2,0x14,0x18,0x0}},
+ {0x59, 0x5e, {0x134,1444,100, 0,0,0x0a,0xc8,0x20,0xf3,0x14,0x18,0x0}},
+ {0x5f, 0x6d, {0x135,1915,100, 0,0,0x00,0xc8,0x20,0xf4,0x15,0x08,0x0}}
+};
+static const struct opl4_region regions_02[] = { /* Electric Grand Piano */
+ {0x14, 0x2d, {0x12c,7476,100, 1,0,0x00,0xae,0x20,0xf2,0x13,0x07,0x0}},
+ {0x2e, 0x33, {0x12d,6818,100, 1,0,0x00,0xae,0x20,0xf2,0x14,0x07,0x0}},
+ {0x34, 0x39, {0x12e,5901,100, 1,0,0x00,0xae,0x20,0xf2,0x14,0x07,0x0}},
+ {0x3a, 0x3f, {0x12f,5292,100, 1,0,0x00,0xae,0x20,0xf2,0x14,0x07,0x0}},
+ {0x40, 0x45, {0x130,4262,100, 1,0,0x00,0xae,0x20,0xf2,0x14,0x07,0x0}},
+ {0x46, 0x4b, {0x131,3627,100, 1,0,0x00,0xae,0x20,0xf2,0x14,0x07,0x0}},
+ {0x4c, 0x52, {0x132,3118,100, 1,0,0x00,0xae,0x20,0xf2,0x14,0x07,0x0}},
+ {0x53, 0x58, {0x133,2083,100, 1,0,0x00,0xae,0x20,0xf2,0x14,0x17,0x0}},
+ {0x59, 0x5e, {0x134,1446,100, 1,0,0x00,0xae,0x20,0xf3,0x14,0x17,0x0}},
+ {0x5f, 0x6d, {0x135,1917,100, 1,0,0x00,0xae,0x20,0xf4,0x15,0x07,0x0}},
+ {0x00, 0x7f, {0x06c,6375,100,-1,0,0x00,0xc2,0x28,0xf4,0x23,0x18,0x0}}
+};
+static const struct opl4_region regions_03[] = { /* Honky-Tonk Piano */
+ {0x14, 0x27, {0x12c,7474,100, 0,0,0x00,0xb4,0x20,0xf2,0x13,0x08,0x0}},
+ {0x28, 0x2d, {0x12d,6816,100, 0,0,0x00,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x2e, 0x33, {0x12e,5899,100, 0,0,0x00,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x34, 0x39, {0x12f,5290,100, 0,0,0x00,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x3a, 0x3f, {0x130,4260,100, 0,0,0x0a,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x40, 0x45, {0x131,3625,100, 0,0,0x0a,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x46, 0x4b, {0x132,3116,100, 0,0,0x04,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x4c, 0x52, {0x133,2081,100, 0,0,0x03,0xb4,0x20,0xf2,0x14,0x18,0x0}},
+ {0x53, 0x58, {0x134,1444,100, 0,0,0x07,0xb4,0x20,0xf3,0x14,0x18,0x0}},
+ {0x59, 0x6d, {0x135,1915,100, 0,0,0x00,0xb4,0x20,0xf4,0x15,0x08,0x0}},
+ {0x14, 0x27, {0x12c,7486,100, 0,0,0x00,0xb4,0x20,0xf2,0x13,0x08,0x0}},
+ {0x28, 0x2d, {0x12d,6803,100, 0,0,0x00,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x2e, 0x33, {0x12e,5912,100, 0,0,0x00,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x34, 0x39, {0x12f,5275,100, 0,0,0x00,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x3a, 0x3f, {0x130,4274,100, 0,0,0x0a,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x40, 0x45, {0x131,3611,100, 0,0,0x0a,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x46, 0x4b, {0x132,3129,100, 0,0,0x04,0xb4,0x20,0xf2,0x14,0x08,0x0}},
+ {0x4c, 0x52, {0x133,2074,100, 0,0,0x07,0xb4,0x20,0xf2,0x14,0x18,0x0}},
+ {0x53, 0x58, {0x134,1457,100, 0,0,0x01,0xb4,0x20,0xf3,0x14,0x18,0x0}},
+ {0x59, 0x6d, {0x135,1903,100, 0,0,0x00,0xb4,0x20,0xf4,0x15,0x08,0x0}}
+};
+static const struct opl4_region regions_04[] = { /* Electric Piano 1 */
+ {0x15, 0x6c, {0x00b,6570,100, 0,0,0x00,0x28,0x38,0xf0,0x00,0x0c,0x0}},
+ {0x00, 0x7f, {0x06c,6375,100, 0,2,0x00,0xb0,0x22,0xf4,0x23,0x19,0x0}}
+};
+static const struct opl4_region regions_05[] = { /* Electric Piano 2 */
+ {0x14, 0x27, {0x12c,7476,100, 0,3,0x00,0xa2,0x1b,0xf2,0x13,0x08,0x0}},
+ {0x28, 0x2d, {0x12d,6818,100, 0,3,0x00,0xa2,0x1b,0xf2,0x14,0x08,0x0}},
+ {0x2e, 0x33, {0x12e,5901,100, 0,3,0x00,0xa2,0x1b,0xf2,0x14,0x08,0x0}},
+ {0x34, 0x39, {0x12f,5292,100, 0,3,0x00,0xa2,0x1b,0xf2,0x14,0x08,0x0}},
+ {0x3a, 0x3f, {0x130,4262,100, 0,3,0x0a,0xa2,0x1b,0xf2,0x14,0x08,0x0}},
+ {0x40, 0x45, {0x131,3627,100, 0,3,0x0a,0xa2,0x1b,0xf2,0x14,0x08,0x0}},
+ {0x46, 0x4b, {0x132,3118,100, 0,3,0x04,0xa2,0x1b,0xf2,0x14,0x08,0x0}},
+ {0x4c, 0x52, {0x133,2083,100, 0,3,0x03,0xa2,0x1b,0xf2,0x14,0x18,0x0}},
+ {0x53, 0x58, {0x134,1446,100, 0,3,0x07,0xa2,0x1b,0xf3,0x14,0x18,0x0}},
+ {0x59, 0x6d, {0x135,1917,100, 0,3,0x00,0xa2,0x1b,0xf4,0x15,0x08,0x0}},
+ {0x14, 0x2d, {0x12c,7472,100, 0,0,0x00,0xa2,0x18,0xf2,0x13,0x08,0x0}},
+ {0x2e, 0x33, {0x12d,6814,100, 0,0,0x00,0xa2,0x18,0xf2,0x14,0x08,0x0}},
+ {0x34, 0x39, {0x12e,5897,100, 0,0,0x00,0xa2,0x18,0xf2,0x14,0x08,0x0}},
+ {0x3a, 0x3f, {0x12f,5288,100, 0,0,0x00,0xa2,0x18,0xf2,0x14,0x08,0x0}},
+ {0x40, 0x45, {0x130,4258,100, 0,0,0x0a,0xa2,0x18,0xf2,0x14,0x08,0x0}},
+ {0x46, 0x4b, {0x131,3623,100, 0,0,0x0a,0xa2,0x18,0xf2,0x14,0x08,0x0}},
+ {0x4c, 0x52, {0x132,3114,100, 0,0,0x04,0xa2,0x18,0xf2,0x14,0x08,0x0}},
+ {0x53, 0x58, {0x133,2079,100, 0,0,0x07,0xa2,0x18,0xf2,0x14,0x18,0x0}},
+ {0x59, 0x5e, {0x134,1442,100, 0,0,0x0a,0xa2,0x18,0xf3,0x14,0x18,0x0}},
+ {0x5f, 0x6d, {0x135,1913,100, 0,0,0x00,0xa2,0x18,0xf4,0x15,0x08,0x0}}
+};
+static const struct opl4_region regions_06[] = { /* Harpsichord */
+ {0x15, 0x39, {0x080,5158,100, 0,0,0x00,0xb2,0x20,0xf5,0x24,0x19,0x0}},
+ {0x3a, 0x3f, {0x081,4408,100, 0,0,0x00,0xb2,0x20,0xf5,0x25,0x09,0x0}},
+ {0x40, 0x45, {0x082,3622,100, 0,0,0x00,0xb2,0x20,0xf5,0x25,0x09,0x0}},
+ {0x46, 0x4d, {0x083,2843,100, 0,0,0x00,0xb2,0x20,0xf5,0x25,0x19,0x0}},
+ {0x4e, 0x6c, {0x084,1307,100, 0,0,0x00,0xb2,0x20,0xf5,0x25,0x29,0x0}}
+};
+static const struct opl4_region regions_07[] = { /* Clavinet */
+ {0x15, 0x51, {0x027,5009,100, 0,0,0x00,0xd2,0x28,0xf5,0x13,0x2b,0x0}},
+ {0x52, 0x6c, {0x028,3495,100, 0,0,0x00,0xd2,0x28,0xf5,0x13,0x3b,0x0}}
+};
+static const struct opl4_region regions_08[] = { /* Celesta */
+ {0x15, 0x6c, {0x02b,3267,100, 0,0,0x00,0xdc,0x20,0xf4,0x15,0x07,0x3}}
+};
+static const struct opl4_region regions_09[] = { /* Glockenspiel */
+ {0x15, 0x78, {0x0f3, 285,100, 0,0,0x00,0xc2,0x28,0xf6,0x25,0x25,0x0}}
+};
+static const struct opl4_region regions_0a[] = { /* Music Box */
+ {0x15, 0x6c, {0x0f3,3362,100, 0,0,0x00,0xb6,0x20,0xa6,0x25,0x25,0x0}},
+ {0x15, 0x6c, {0x101,4773,100, 0,0,0x00,0xaa,0x20,0xd4,0x14,0x16,0x0}}
+};
+static const struct opl4_region regions_0b[] = { /* Vibraphone */
+ {0x15, 0x6c, {0x101,4778,100, 0,0,0x00,0xc0,0x28,0xf4,0x14,0x16,0x4}}
+};
+static const struct opl4_region regions_0c[] = { /* Marimba */
+ {0x15, 0x3f, {0x0f4,4778,100, 0,0,0x00,0xc4,0x38,0xf7,0x47,0x08,0x0}},
+ {0x40, 0x4c, {0x0f5,3217,100, 0,0,0x00,0xc4,0x38,0xf7,0x47,0x08,0x0}},
+ {0x4d, 0x5a, {0x0f5,3217,100, 0,0,0x00,0xc4,0x38,0xf7,0x48,0x08,0x0}},
+ {0x5b, 0x7f, {0x0f5,3218,100, 0,0,0x00,0xc4,0x38,0xf7,0x48,0x18,0x0}}
+};
+static const struct opl4_region regions_0d[] = { /* Xylophone */
+ {0x00, 0x7f, {0x136,1729,100, 0,0,0x00,0xd2,0x38,0xf0,0x06,0x36,0x0}}
+};
+static const struct opl4_region regions_0e[] = { /* Tubular Bell */
+ {0x01, 0x7f, {0x0ff,3999,100, 0,1,0x00,0x90,0x21,0xf4,0xa3,0x25,0x1}}
+};
+static const struct opl4_region regions_0f[] = { /* Dulcimer */
+ {0x00, 0x7f, {0x03f,4236,100, 0,1,0x00,0xbc,0x29,0xf5,0x16,0x07,0x0}},
+ {0x00, 0x7f, {0x040,4236,100, 0,2,0x0e,0x94,0x2a,0xf5,0x16,0x07,0x0}}
+};
+static const struct opl4_region regions_10[] = { /* Drawbar Organ */
+ {0x01, 0x7f, {0x08e,4394,100, 0,2,0x14,0xc2,0x3a,0xf0,0x00,0x0a,0x0}}
+};
+static const struct opl4_region regions_11[] = { /* Percussive Organ */
+ {0x15, 0x3b, {0x08c,6062,100, 0,3,0x00,0xbe,0x3b,0xf0,0x00,0x09,0x0}},
+ {0x3c, 0x6c, {0x08d,2984,100, 0,3,0x00,0xbe,0x3b,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_12[] = { /* Rock Organ */
+ {0x15, 0x30, {0x128,6574,100, 0,1,0x00,0xcc,0x39,0xf0,0x00,0x0a,0x0}},
+ {0x31, 0x3c, {0x129,5040,100, 0,1,0x00,0xcc,0x39,0xf0,0x00,0x0a,0x0}},
+ {0x3d, 0x48, {0x12a,3498,100, 0,1,0x00,0xcc,0x39,0xf0,0x00,0x0a,0x0}},
+ {0x49, 0x54, {0x12b,1957,100, 0,1,0x00,0xcc,0x39,0xf0,0x00,0x0a,0x0}},
+ {0x55, 0x6c, {0x127, 423,100, 0,1,0x00,0xcc,0x39,0xf0,0x00,0x0a,0x0}}
+};
+static const struct opl4_region regions_13[] = { /* Church Organ */
+ {0x15, 0x29, {0x087,7466,100, 0,1,0x00,0xc4,0x11,0xf0,0x00,0x09,0x0}},
+ {0x2a, 0x30, {0x088,6456,100, 0,1,0x00,0xc4,0x11,0xf0,0x00,0x09,0x0}},
+ {0x31, 0x38, {0x089,5428,100, 0,1,0x00,0xc4,0x11,0xf0,0x00,0x09,0x0}},
+ {0x39, 0x41, {0x08a,4408,100, 0,1,0x00,0xc4,0x11,0xf0,0x00,0x09,0x0}},
+ {0x42, 0x6c, {0x08b,3406,100, 0,1,0x00,0xc4,0x11,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_14[] = { /* Reed Organ */
+ {0x00, 0x53, {0x0ac,5570,100, 0,0,0x06,0xc0,0x38,0xf0,0x00,0x09,0x1}},
+ {0x54, 0x7f, {0x0ad,2497,100, 0,0,0x00,0xc0,0x38,0xf0,0x00,0x09,0x1}}
+};
+static const struct opl4_region regions_15[] = { /* Accordion */
+ {0x15, 0x4c, {0x006,4261,100, 0,2,0x00,0xa4,0x22,0x90,0x00,0x09,0x0}},
+ {0x4d, 0x6c, {0x007,1530,100, 0,2,0x00,0xa4,0x22,0x90,0x00,0x09,0x0}},
+ {0x15, 0x6c, {0x070,4391,100, 0,3,0x00,0x8a,0x23,0xa0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_16[] = { /* Harmonica */
+ {0x15, 0x6c, {0x070,4408,100, 0,0,0x00,0xae,0x30,0xa0,0x00,0x09,0x2}}
+};
+static const struct opl4_region regions_17[] = { /* Tango Accordion */
+ {0x00, 0x53, {0x0ac,5573,100, 0,0,0x00,0xae,0x38,0xf0,0x00,0x09,0x0}},
+ {0x54, 0x7f, {0x0ad,2500,100, 0,0,0x00,0xae,0x38,0xf0,0x00,0x09,0x0}},
+ {0x15, 0x6c, {0x041,8479,100, 0,2,0x00,0x6a,0x3a,0x75,0x20,0x0a,0x0}}
+};
+static const struct opl4_region regions_18[] = { /* Nylon Guitar */
+ {0x15, 0x2f, {0x0b3,6964,100, 0,0,0x05,0xca,0x28,0xf5,0x34,0x09,0x0}},
+ {0x30, 0x36, {0x0b7,5567,100, 0,0,0x0c,0xca,0x28,0xf5,0x34,0x09,0x0}},
+ {0x37, 0x3c, {0x0b5,4653,100, 0,0,0x00,0xca,0x28,0xf6,0x34,0x09,0x0}},
+ {0x3d, 0x43, {0x0b4,3892,100, 0,0,0x00,0xca,0x28,0xf6,0x35,0x09,0x0}},
+ {0x44, 0x60, {0x0b6,2723,100, 0,0,0x00,0xca,0x28,0xf6,0x35,0x19,0x0}}
+};
+static const struct opl4_region regions_19[] = { /* Steel Guitar */
+ {0x15, 0x31, {0x00c,6937,100, 0,0,0x00,0xbc,0x28,0xf0,0x04,0x19,0x0}},
+ {0x32, 0x38, {0x00d,5410,100, 0,0,0x00,0xbc,0x28,0xf0,0x05,0x09,0x0}},
+ {0x39, 0x47, {0x00e,4379,100, 0,0,0x00,0xbc,0x28,0xf5,0x94,0x09,0x0}},
+ {0x48, 0x6c, {0x00f,2843,100, 0,0,0x00,0xbc,0x28,0xf6,0x95,0x09,0x0}}
+};
+static const struct opl4_region regions_1a[] = { /* Jazz Guitar */
+ {0x15, 0x31, {0x05a,6832,100, 0,0,0x00,0xca,0x28,0xf6,0x34,0x09,0x0}},
+ {0x32, 0x3f, {0x05b,4897,100, 0,0,0x00,0xca,0x28,0xf6,0x34,0x09,0x0}},
+ {0x40, 0x6c, {0x05c,3218,100, 0,0,0x00,0xca,0x28,0xf6,0x34,0x09,0x0}}
+};
+static const struct opl4_region regions_1b[] = { /* Clean Guitar */
+ {0x15, 0x2c, {0x061,7053,100, 0,1,0x00,0xb4,0x29,0xf5,0x54,0x0a,0x0}},
+ {0x2d, 0x31, {0x060,6434,100, 0,1,0x00,0xb4,0x29,0xf5,0x54,0x0a,0x0}},
+ {0x32, 0x38, {0x063,5764,100, 0,1,0x00,0xbe,0x29,0xf5,0x55,0x0a,0x0}},
+ {0x39, 0x3f, {0x062,4627,100, 0,1,0x00,0xb4,0x29,0xf5,0x55,0x0a,0x0}},
+ {0x40, 0x44, {0x065,3963,100, 0,1,0x00,0xb4,0x29,0xf5,0x55,0x1a,0x0}},
+ {0x45, 0x4b, {0x064,3313,100, 0,1,0x00,0xb4,0x29,0xf5,0x55,0x1a,0x0}},
+ {0x4c, 0x54, {0x066,2462,100, 0,1,0x00,0xb4,0x29,0xf5,0x55,0x2a,0x0}},
+ {0x55, 0x6c, {0x067,1307,100, 0,1,0x00,0xb4,0x29,0xf6,0x56,0x0a,0x0}}
+};
+static const struct opl4_region regions_1c[] = { /* Muted Guitar */
+ {0x01, 0x7f, {0x068,4408,100, 0,0,0x00,0xcc,0x28,0xf6,0x15,0x09,0x0}}
+};
+static const struct opl4_region regions_1d[] = { /* Overdriven Guitar */
+ {0x00, 0x40, {0x0a5,6589,100, 0,1,0x00,0xc0,0x29,0xf2,0x11,0x09,0x0}},
+ {0x41, 0x7f, {0x0a6,5428,100, 0,1,0x00,0xc0,0x29,0xf2,0x11,0x09,0x0}}
+};
+static const struct opl4_region regions_1e[] = { /* Distortion Guitar */
+ {0x15, 0x2a, {0x051,6928,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x2b, 0x2e, {0x052,6433,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x2f, 0x32, {0x053,5944,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x33, 0x36, {0x054,5391,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x37, 0x3a, {0x055,4897,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x3b, 0x3e, {0x056,4408,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x3f, 0x42, {0x057,3892,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x43, 0x46, {0x058,3361,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}},
+ {0x47, 0x6c, {0x059,2784,100, 0,1,0x00,0xbc,0x21,0xa2,0x12,0x0a,0x0}}
+};
+static const struct opl4_region regions_1f[] = { /* Guitar Harmonics */
+ {0x15, 0x44, {0x05e,5499,100, 0,0,0x00,0xce,0x28,0xf4,0x24,0x09,0x0}},
+ {0x45, 0x49, {0x05d,4850,100, 0,0,0x00,0xe2,0x28,0xf4,0x24,0x09,0x0}},
+ {0x4a, 0x6c, {0x05f,4259,100, 0,0,0x00,0xce,0x28,0xf4,0x24,0x09,0x0}}
+};
+static const struct opl4_region regions_20[] = { /* Acoustic Bass */
+ {0x15, 0x30, {0x004,8053,100, 0,0,0x00,0xe2,0x18,0xf5,0x15,0x09,0x0}},
+ {0x31, 0x6c, {0x005,4754,100, 0,0,0x00,0xe2,0x18,0xf5,0x15,0x09,0x0}}
+};
+static const struct opl4_region regions_21[] = { /* Fingered Bass */
+ {0x01, 0x20, {0x04a,8762,100, 0,0,0x00,0xde,0x18,0xf6,0x14,0x09,0x0}},
+ {0x21, 0x25, {0x04b,8114,100, 0,0,0x00,0xde,0x18,0xf6,0x14,0x09,0x0}},
+ {0x26, 0x2a, {0x04c,7475,100, 0,0,0x00,0xde,0x18,0xf6,0x14,0x09,0x0}},
+ {0x2b, 0x7f, {0x04d,6841,100, 0,0,0x00,0xde,0x18,0xf6,0x14,0x09,0x0}}
+};
+static const struct opl4_region regions_22[] = { /* Picked Bass */
+ {0x15, 0x23, {0x04f,7954,100, 0,0,0x00,0xcc,0x18,0xf3,0x90,0x0a,0x0}},
+ {0x24, 0x2a, {0x050,7318,100, 0,0,0x05,0xcc,0x18,0xf3,0x90,0x1a,0x0}},
+ {0x2b, 0x2f, {0x06b,6654,100, 0,0,0x00,0xcc,0x18,0xf3,0x90,0x2a,0x0}},
+ {0x30, 0x47, {0x069,6031,100, 0,0,0x00,0xcc,0x18,0xf5,0xb0,0x0a,0x0}},
+ {0x48, 0x6c, {0x06a,5393,100, 0,0,0x00,0xcc,0x18,0xf5,0xb0,0x0a,0x0}}
+};
+static const struct opl4_region regions_23[] = { /* Fretless Bass */
+ {0x01, 0x7f, {0x04e,5297,100, 0,0,0x00,0xd2,0x10,0xf3,0x63,0x19,0x0}}
+};
+static const struct opl4_region regions_24[] = { /* Slap Bass 1 */
+ {0x15, 0x6c, {0x0a3,7606,100, 0,1,0x00,0xde,0x19,0xf5,0x32,0x1a,0x0}}
+};
+static const struct opl4_region regions_25[] = { /* Slap Bass 2 */
+ {0x01, 0x7f, {0x0a2,6694,100, 0,0,0x00,0xda,0x20,0xb0,0x02,0x09,0x0}}
+};
+static const struct opl4_region regions_26[] = { /* Synth Bass 1 */
+ {0x15, 0x6c, {0x0be,7466,100, 0,1,0x00,0xb8,0x39,0xf4,0x14,0x09,0x0}}
+};
+static const struct opl4_region regions_27[] = { /* Synth Bass 2 */
+ {0x00, 0x7f, {0x117,8103,100, 0,1,0x00,0xca,0x39,0xf3,0x50,0x08,0x0}}
+};
+static const struct opl4_region regions_28[] = { /* Violin */
+ {0x15, 0x3a, {0x105,5158,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x3b, 0x3f, {0x102,4754,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x40, 0x41, {0x106,4132,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x42, 0x44, {0x107,4033,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x45, 0x47, {0x108,3580,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x48, 0x4a, {0x10a,2957,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x4b, 0x4c, {0x10b,2724,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x4d, 0x4e, {0x10c,2530,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x4f, 0x51, {0x10d,2166,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}},
+ {0x52, 0x6c, {0x109,1825,100, 0,3,0x00,0xcc,0x3b,0xf3,0x20,0x09,0x0}}
+};
+static const struct opl4_region regions_29[] = { /* Viola */
+ {0x15, 0x32, {0x103,5780,100, 0,3,0x00,0xc4,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x33, 0x35, {0x104,5534,100, 0,3,0x00,0xc4,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x36, 0x38, {0x105,5158,100, 0,3,0x00,0xc4,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x39, 0x3d, {0x102,4754,100, 0,3,0x00,0xca,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x3e, 0x3f, {0x106,4132,100, 0,3,0x00,0xc4,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x40, 0x42, {0x107,4033,100, 0,3,0x00,0xc4,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x43, 0x45, {0x108,3580,100, 0,3,0x00,0xd0,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x46, 0x48, {0x10a,2957,100, 0,3,0x00,0xca,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x49, 0x4a, {0x10b,2724,100, 0,3,0x00,0xd0,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x4b, 0x4c, {0x10c,2530,100, 0,3,0x00,0xca,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x4d, 0x4f, {0x10d,2166,100, 0,3,0x00,0xd0,0x3b,0xa3,0x20,0x09,0x0}},
+ {0x50, 0x6c, {0x109,1825,100, 0,3,0x00,0xd0,0x3b,0xa3,0x20,0x09,0x0}}
+};
+static const struct opl4_region regions_2a[] = { /* Cello */
+ {0x15, 0x2d, {0x112,6545,100, 0,3,0x00,0xc0,0x33,0xa0,0x00,0x08,0x0}},
+ {0x2e, 0x37, {0x113,5764,100, 0,3,0x00,0xc0,0x33,0xa0,0x00,0x08,0x0}},
+ {0x38, 0x3e, {0x115,4378,100, 0,3,0x00,0xc0,0x33,0xa0,0x00,0x18,0x0}},
+ {0x3f, 0x44, {0x116,3998,100, 0,3,0x00,0xc0,0x33,0xa0,0x00,0x18,0x0}},
+ {0x45, 0x6c, {0x114,3218,100, 0,3,0x00,0xc0,0x33,0xa0,0x00,0x18,0x0}}
+};
+static const struct opl4_region regions_2b[] = { /* Contrabass */
+ {0x15, 0x29, {0x110,7713,100, 0,1,0x00,0xc2,0x19,0x90,0x00,0x09,0x0}},
+ {0x2a, 0x6c, {0x111,6162,100, 0,1,0x00,0xc2,0x19,0x90,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_2c[] = { /* Tremolo Strings */
+ {0x15, 0x3b, {0x0b0,4810,100, 0,0,0x0a,0xde,0x38,0xf0,0x00,0x07,0x6}},
+ {0x3c, 0x41, {0x035,4035,100, 0,0,0x05,0xde,0x38,0xf0,0x00,0x07,0x6}},
+ {0x42, 0x47, {0x033,3129,100, 0,0,0x05,0xde,0x38,0xf0,0x00,0x07,0x6}},
+ {0x48, 0x52, {0x034,2625,100, 0,0,0x05,0xde,0x38,0xf0,0x00,0x07,0x6}},
+ {0x53, 0x6c, {0x0af, 936,100, 0,0,0x00,0xde,0x38,0xf0,0x00,0x07,0x6}}
+};
+static const struct opl4_region regions_2d[] = { /* Pizzicato Strings */
+ {0x15, 0x32, {0x0b8,6186,100, 0,0,0x00,0xbc,0x28,0xf0,0x00,0x05,0x0}},
+ {0x33, 0x3b, {0x0b9,5031,100, 0,0,0x00,0xbc,0x28,0xf0,0x00,0x05,0x0}},
+ {0x3c, 0x42, {0x0bb,4146,100, 0,0,0x00,0xbc,0x28,0xf0,0x00,0x05,0x0}},
+ {0x43, 0x48, {0x0ba,3245,100, 0,0,0x00,0xc2,0x28,0xf0,0x00,0x05,0x0}},
+ {0x49, 0x6c, {0x0bc,2352,100, 0,0,0x00,0xbc,0x28,0xf0,0x00,0x05,0x0}}
+};
+static const struct opl4_region regions_2e[] = { /* Harp */
+ {0x15, 0x46, {0x07e,3740,100, 0,1,0x00,0xd2,0x29,0xf5,0x25,0x07,0x0}},
+ {0x47, 0x6c, {0x07f,2319,100, 0,1,0x00,0xd2,0x29,0xf5,0x25,0x07,0x0}}
+};
+static const struct opl4_region regions_2f[] = { /* Timpani */
+ {0x15, 0x6c, {0x100,6570,100, 0,0,0x00,0xf8,0x28,0xf0,0x05,0x16,0x0}}
+};
+static const struct opl4_region regions_30[] = { /* Strings */
+ {0x15, 0x3b, {0x13c,4806,100, 0,0,0x00,0xc8,0x20,0x80,0x00,0x07,0x0}},
+ {0x3c, 0x41, {0x13e,4035,100, 0,0,0x00,0xc8,0x20,0x80,0x00,0x07,0x0}},
+ {0x42, 0x47, {0x13d,3122,100, 0,0,0x00,0xc8,0x20,0x80,0x00,0x07,0x0}},
+ {0x48, 0x52, {0x13f,2629,100, 0,0,0x00,0xbe,0x20,0x80,0x00,0x07,0x0}},
+ {0x53, 0x6c, {0x140, 950,100, 0,0,0x00,0xbe,0x20,0x80,0x00,0x07,0x0}}
+};
+static const struct opl4_region regions_31[] = { /* Slow Strings */
+ {0x15, 0x3b, {0x0b0,4810,100, 0,1,0x0a,0xbe,0x19,0xf0,0x00,0x07,0x0}},
+ {0x3c, 0x41, {0x035,4035,100, 0,1,0x05,0xbe,0x19,0xf0,0x00,0x07,0x0}},
+ {0x42, 0x47, {0x033,3129,100, 0,1,0x05,0xbe,0x19,0xf0,0x00,0x07,0x0}},
+ {0x48, 0x52, {0x034,2625,100, 0,1,0x05,0xbe,0x19,0xf0,0x00,0x07,0x0}},
+ {0x53, 0x6c, {0x0af, 936,100, 0,1,0x00,0xbe,0x19,0xf0,0x00,0x07,0x0}}
+};
+static const struct opl4_region regions_32[] = { /* Synth Strings 1 */
+ {0x05, 0x71, {0x002,6045,100,-2,0,0x00,0xa6,0x20,0x93,0x22,0x06,0x0}},
+ {0x15, 0x6c, {0x0ae,3261,100, 2,0,0x00,0xc6,0x20,0x70,0x01,0x06,0x0}}
+};
+static const struct opl4_region regions_33[] = { /* Synth Strings 2 */
+ {0x15, 0x6c, {0x002,4513,100, 5,1,0x00,0xb4,0x19,0x70,0x00,0x06,0x0}},
+ {0x15, 0x6c, {0x002,4501,100,-5,1,0x00,0xb4,0x19,0x70,0x00,0x06,0x0}}
+};
+static const struct opl4_region regions_34[] = { /* Choir Aahs */
+ {0x15, 0x3a, {0x018,5010,100, 0,2,0x00,0xc2,0x1a,0x70,0x00,0x08,0x0}},
+ {0x3b, 0x40, {0x019,4370,100, 0,2,0x00,0xc2,0x1a,0x70,0x00,0x08,0x0}},
+ {0x41, 0x47, {0x01a,3478,100, 0,2,0x00,0xc2,0x1a,0x70,0x00,0x08,0x0}},
+ {0x48, 0x6c, {0x01b,2197,100, 0,2,0x00,0xc2,0x1a,0x70,0x00,0x08,0x0}}
+};
+static const struct opl4_region regions_35[] = { /* Voice Oohs */
+ {0x15, 0x6c, {0x029,3596,100, 0,0,0x00,0xe6,0x20,0xf7,0x20,0x08,0x0}}
+};
+static const struct opl4_region regions_36[] = { /* Synth Voice */
+ {0x15, 0x6c, {0x02a,3482,100, 0,1,0x00,0xc2,0x19,0x85,0x21,0x07,0x0}}
+};
+static const struct opl4_region regions_37[] = { /* Orchestra Hit */
+ {0x15, 0x6c, {0x049,4394,100, 0,0,0x00,0xfe,0x30,0x80,0x05,0x05,0x0}}
+};
+static const struct opl4_region regions_38[] = { /* Trumpet */
+ {0x15, 0x3c, {0x0f6,4706,100, 0,2,0x00,0xd6,0x32,0xf3,0x20,0x0a,0x0}},
+ {0x3d, 0x43, {0x0f8,3894,100, 0,2,0x00,0xd6,0x32,0xf3,0x20,0x0a,0x0}},
+ {0x44, 0x48, {0x0f7,3118,100, 0,2,0x00,0xd6,0x32,0xf3,0x20,0x0a,0x0}},
+ {0x49, 0x4e, {0x0fa,2322,100, 0,2,0x00,0xd6,0x32,0xf3,0x20,0x0a,0x0}},
+ {0x4f, 0x55, {0x0f9,1634,100, 0,2,0x00,0xd6,0x32,0xf3,0x20,0x0a,0x0}},
+ {0x56, 0x6c, {0x0fb, 786,100, 0,2,0x00,0xd6,0x32,0xf3,0x20,0x0a,0x0}}
+};
+static const struct opl4_region regions_39[] = { /* Trombone */
+ {0x15, 0x3a, {0x0f0,5053,100, 0,1,0x00,0xd6,0x21,0xf0,0x00,0x09,0x0}},
+ {0x3b, 0x3f, {0x0f1,4290,100, 0,1,0x00,0xd6,0x21,0xf0,0x00,0x09,0x0}},
+ {0x40, 0x6c, {0x0f2,3580,100, 0,1,0x00,0xd6,0x21,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_3a[] = { /* Tuba */
+ {0x15, 0x2d, {0x085,7096,100, 0,1,0x00,0xde,0x21,0xf5,0x10,0x09,0x0}},
+ {0x2e, 0x6c, {0x086,6014,100, 0,1,0x00,0xde,0x21,0xf5,0x10,0x09,0x0}}
+};
+static const struct opl4_region regions_3b[] = { /* Muted Trumpet */
+ {0x15, 0x45, {0x0b1,4135,100, 0,0,0x00,0xcc,0x28,0xf3,0x10,0x0a,0x1}},
+ {0x46, 0x6c, {0x0b2,2599,100, 0,0,0x00,0xcc,0x28,0x83,0x10,0x0a,0x1}}
+};
+static const struct opl4_region regions_3c[] = { /* French Horns */
+ {0x15, 0x49, {0x07c,3624,100, 0,2,0x00,0xd0,0x1a,0xf0,0x00,0x09,0x0}},
+ {0x4a, 0x6c, {0x07d,2664,100, 0,2,0x00,0xd0,0x1a,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_3d[] = { /* Brass Section */
+ {0x15, 0x42, {0x0fc,4375,100, 0,0,0x00,0xd6,0x28,0xf0,0x00,0x0a,0x0}},
+ {0x43, 0x6c, {0x0fd,2854,100, 0,0,0x00,0xd6,0x28,0xf0,0x00,0x0a,0x0}}
+};
+static const struct opl4_region regions_3e[] = { /* Synth Brass 1 */
+ {0x01, 0x27, {0x0d3,9094,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x28, 0x2d, {0x0da,8335,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x2e, 0x33, {0x0d4,7558,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x34, 0x39, {0x0db,6785,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x3a, 0x3f, {0x0d5,6042,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x40, 0x45, {0x0dc,5257,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x46, 0x4b, {0x0d6,4493,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x4c, 0x51, {0x0dd,3741,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x52, 0x57, {0x0d7,3012,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x58, 0x5d, {0x0de,2167,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x5e, 0x63, {0x0d8,1421,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x64, 0x7f, {0x0d9,-115,100,-1,0,0x00,0xbe,0x18,0xa5,0x11,0x08,0x0}},
+ {0x01, 0x27, {0x118,9103,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x28, 0x2d, {0x119,8340,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x2e, 0x33, {0x11a,7565,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x34, 0x39, {0x11b,6804,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x3a, 0x3f, {0x11c,6042,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x40, 0x45, {0x11d,5277,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x46, 0x4b, {0x11e,4520,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x4c, 0x51, {0x11f,3741,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x52, 0x57, {0x120,3012,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x58, 0x5d, {0x121,2166,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x5e, 0x64, {0x122,1421,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}},
+ {0x65, 0x7f, {0x123,-115,100, 1,1,0x00,0xbe,0x19,0x85,0x23,0x08,0x0}}
+};
+static const struct opl4_region regions_3f[] = { /* Synth Brass 2 */
+ {0x01, 0x27, {0x118,9113,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x28, 0x2d, {0x119,8350,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x2e, 0x33, {0x11a,7575,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x34, 0x39, {0x11b,6814,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x3a, 0x3f, {0x11c,6052,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x40, 0x45, {0x11d,5287,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x46, 0x4b, {0x11e,4530,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x4c, 0x51, {0x11f,3751,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x52, 0x57, {0x120,3022,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x58, 0x5d, {0x121,2176,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x5e, 0x64, {0x122,1431,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x65, 0x7f, {0x123,-105,100, 3,6,0x00,0xae,0x26,0x85,0x23,0x08,0x0}},
+ {0x00, 0x7f, {0x124,4034,100,-3,2,0x00,0xea,0x22,0x85,0x23,0x08,0x0}}
+};
+static const struct opl4_region regions_40[] = { /* Soprano Sax */
+ {0x15, 0x3f, {0x0e3,4228,100, 0,1,0x00,0xc8,0x21,0xf5,0x20,0x0a,0x0}},
+ {0x40, 0x45, {0x0e4,3495,100, 0,1,0x00,0xc8,0x21,0xf5,0x20,0x0a,0x0}},
+ {0x46, 0x4b, {0x0e5,2660,100, 0,1,0x00,0xd6,0x21,0xf5,0x20,0x0a,0x0}},
+ {0x4c, 0x51, {0x0e6,2002,100, 0,1,0x00,0xd6,0x21,0xf5,0x20,0x0a,0x0}},
+ {0x52, 0x59, {0x0e7,1186,100, 0,1,0x00,0xd6,0x21,0xf5,0x20,0x0a,0x0}},
+ {0x59, 0x6c, {0x0e8,1730,100, 0,1,0x00,0xc8,0x21,0xf5,0x20,0x0a,0x0}}
+};
+static const struct opl4_region regions_41[] = { /* Alto Sax */
+ {0x15, 0x32, {0x092,6204,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x33, 0x35, {0x096,5812,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x36, 0x3a, {0x099,5318,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x3b, 0x3b, {0x08f,5076,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x3c, 0x3e, {0x093,4706,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x3f, 0x41, {0x097,4321,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x42, 0x44, {0x09a,3893,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x45, 0x47, {0x090,3497,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x48, 0x4a, {0x094,3119,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x4b, 0x4d, {0x098,2726,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x4e, 0x50, {0x09b,2393,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x51, 0x53, {0x091,2088,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}},
+ {0x54, 0x6c, {0x095,1732,100, 0,1,0x00,0xbe,0x19,0xf5,0x20,0x0b,0x0}}
+};
+static const struct opl4_region regions_42[] = { /* Tenor Sax */
+ {0x24, 0x30, {0x0e9,6301,100, 0,1,0x00,0xbc,0x19,0xf4,0x10,0x0b,0x0}},
+ {0x31, 0x34, {0x0ea,5781,100, 0,1,0x00,0xbc,0x19,0xf4,0x10,0x0b,0x0}},
+ {0x35, 0x3a, {0x0eb,5053,100, 0,1,0x00,0xbc,0x19,0xf4,0x10,0x0b,0x0}},
+ {0x3b, 0x41, {0x0ed,4165,100, 0,1,0x00,0xbc,0x19,0xf4,0x10,0x0b,0x0}},
+ {0x42, 0x47, {0x0ec,3218,100, 0,1,0x00,0xbc,0x19,0xf4,0x10,0x0b,0x0}},
+ {0x48, 0x51, {0x0ee,2462,100, 0,1,0x00,0xbc,0x19,0xf4,0x10,0x0b,0x0}},
+ {0x52, 0x6c, {0x0ef,1421,100, 0,1,0x00,0xbc,0x19,0xf4,0x10,0x0b,0x0}}
+};
+static const struct opl4_region regions_43[] = { /* Baritone Sax */
+ {0x15, 0x2d, {0x0df,6714,100, 0,1,0x00,0xce,0x19,0xf0,0x00,0x0a,0x0}},
+ {0x2e, 0x34, {0x0e1,5552,100, 0,1,0x00,0xce,0x19,0xf0,0x00,0x0a,0x0}},
+ {0x35, 0x39, {0x0e2,5178,100, 0,1,0x00,0xce,0x19,0xf0,0x00,0x0a,0x0}},
+ {0x3a, 0x6c, {0x0e0,4437,100, 0,1,0x00,0xce,0x19,0xf0,0x00,0x0a,0x0}}
+};
+static const struct opl4_region regions_44[] = { /* Oboe */
+ {0x15, 0x3c, {0x042,4493,100, 0,1,0x00,0xe6,0x39,0xf4,0x10,0x0a,0x0}},
+ {0x3d, 0x43, {0x044,3702,100, 0,1,0x00,0xdc,0x39,0xf4,0x10,0x0a,0x0}},
+ {0x44, 0x49, {0x043,2956,100, 0,1,0x00,0xdc,0x39,0xf4,0x10,0x0a,0x0}},
+ {0x4a, 0x4f, {0x046,2166,100, 0,1,0x00,0xdc,0x39,0xf4,0x10,0x0a,0x0}},
+ {0x50, 0x55, {0x045,1420,100, 0,1,0x00,0xdc,0x39,0xf4,0x10,0x0a,0x0}},
+ {0x56, 0x6c, {0x047, 630,100, 0,1,0x00,0xe6,0x39,0xf4,0x10,0x0a,0x0}}
+};
+static const struct opl4_region regions_45[] = { /* English Horn */
+ {0x15, 0x38, {0x03c,5098,100, 0,1,0x00,0xc4,0x31,0xf0,0x00,0x09,0x0}},
+ {0x39, 0x3e, {0x03b,4291,100, 0,1,0x00,0xc4,0x31,0xf0,0x00,0x09,0x0}},
+ {0x3f, 0x6c, {0x03d,3540,100, 0,1,0x00,0xc4,0x31,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_46[] = { /* Bassoon */
+ {0x15, 0x22, {0x038,7833,100, 0,1,0x00,0xc6,0x31,0xf0,0x00,0x0b,0x0}},
+ {0x23, 0x2e, {0x03a,7070,100, 0,1,0x00,0xc6,0x31,0xf0,0x00,0x0b,0x0}},
+ {0x2f, 0x6c, {0x039,6302,100, 0,1,0x00,0xc6,0x31,0xf0,0x00,0x0b,0x0}}
+};
+static const struct opl4_region regions_47[] = { /* Clarinet */
+ {0x15, 0x3b, {0x09e,5900,100, 0,1,0x00,0xc8,0x29,0xf3,0x20,0x0a,0x0}},
+ {0x3c, 0x41, {0x0a0,5158,100, 0,1,0x00,0xc8,0x29,0xf3,0x20,0x0a,0x0}},
+ {0x42, 0x4a, {0x09f,4260,100, 0,1,0x00,0xc8,0x29,0xf3,0x20,0x0a,0x0}},
+ {0x4b, 0x6c, {0x0a1,2957,100, 0,1,0x00,0xc8,0x29,0xf3,0x20,0x0a,0x0}}
+};
+static const struct opl4_region regions_48[] = { /* Piccolo */
+ {0x15, 0x40, {0x071,4803,100, 0,0,0x00,0xe6,0x38,0xf0,0x00,0x0a,0x2}},
+ {0x41, 0x4d, {0x072,3314,100, 0,0,0x00,0xe6,0x38,0xf0,0x00,0x0a,0x2}},
+ {0x4e, 0x53, {0x073,1731,100, 0,0,0x00,0xe6,0x38,0xf0,0x00,0x0a,0x2}},
+ {0x54, 0x5f, {0x074,2085,100, 0,0,0x00,0xe6,0x38,0xf0,0x00,0x0a,0x2}},
+ {0x60, 0x6c, {0x075,1421,100, 0,0,0x00,0xe6,0x38,0xf0,0x00,0x0a,0x2}}
+};
+static const struct opl4_region regions_49[] = { /* Flute */
+ {0x15, 0x40, {0x071,4803,100, 0,0,0x00,0xdc,0x38,0xf0,0x00,0x0a,0x2}},
+ {0x41, 0x4d, {0x072,3314,100, 0,0,0x00,0xdc,0x38,0xf0,0x00,0x0a,0x2}},
+ {0x4e, 0x6c, {0x073,1731,100, 0,0,0x00,0xe6,0x38,0xf0,0x00,0x0a,0x2}}
+};
+static const struct opl4_region regions_4a[] = { /* Recorder */
+ {0x15, 0x6f, {0x0bd,4897,100, 0,0,0x00,0xec,0x30,0x70,0x00,0x09,0x1}}
+};
+static const struct opl4_region regions_4b[] = { /* Pan Flute */
+ {0x15, 0x6c, {0x077,2359,100, 0,0,0x00,0xde,0x38,0xf0,0x00,0x09,0x3}}
+};
+static const struct opl4_region regions_4c[] = { /* Bottle Blow */
+ {0x15, 0x6c, {0x077,2359,100, 0,0,0x00,0xc8,0x38,0xf0,0x00,0x09,0x1}},
+ {0x01, 0x7f, {0x125,7372,100, 0,0,0x1e,0x80,0x00,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_4d[] = { /* Shakuhachi */
+ {0x00, 0x7f, {0x0ab,4548,100, 0,0,0x00,0xd6,0x30,0xf0,0x00,0x0a,0x3}},
+ {0x15, 0x6c, {0x076,3716,100, 0,0,0x00,0xa2,0x28,0x70,0x00,0x09,0x2}}
+};
+static const struct opl4_region regions_4e[] = { /* Whistle */
+ {0x00, 0x7f, {0x0aa,1731,100, 0,4,0x00,0xd2,0x2c,0x70,0x00,0x0a,0x0}}
+};
+static const struct opl4_region regions_4f[] = { /* Ocarina */
+ {0x00, 0x7f, {0x0aa,1731,100, 0,1,0x00,0xce,0x29,0x90,0x00,0x0a,0x1}}
+};
+static const struct opl4_region regions_50[] = { /* Square Lead */
+ {0x01, 0x2a, {0x0cc,9853,100, 3,0,0x00,0xac,0x38,0xc6,0x21,0x09,0x0}},
+ {0x2b, 0x36, {0x0cd,6785,100, 3,0,0x00,0xac,0x38,0xc6,0x21,0x09,0x0}},
+ {0x37, 0x42, {0x0ca,5248,100, 3,0,0x00,0xac,0x38,0xc6,0x21,0x09,0x0}},
+ {0x43, 0x4e, {0x0cf,3713,100, 3,0,0x00,0xac,0x38,0xc6,0x21,0x09,0x0}},
+ {0x4f, 0x5a, {0x0ce,2176,100, 3,0,0x00,0xac,0x38,0xc6,0x21,0x09,0x0}},
+ {0x5b, 0x7f, {0x0cb, 640,100, 3,0,0x00,0xac,0x38,0xc6,0x21,0x09,0x0}},
+ {0x01, 0x2a, {0x0cc,9844,100,-3,0,0x00,0xac,0x08,0xc6,0x21,0x09,0x0}},
+ {0x2b, 0x36, {0x0cd,6776,100,-3,0,0x00,0xac,0x08,0xc6,0x21,0x09,0x0}},
+ {0x37, 0x42, {0x0ca,5239,100,-3,0,0x00,0xac,0x08,0xc6,0x21,0x09,0x0}},
+ {0x43, 0x4e, {0x0cf,3704,100,-3,0,0x00,0xac,0x08,0xc6,0x21,0x09,0x0}},
+ {0x4f, 0x5a, {0x0ce,2167,100,-3,0,0x00,0xac,0x08,0xc6,0x21,0x09,0x0}},
+ {0x5b, 0x7f, {0x0cb, 631,100,-3,0,0x00,0xac,0x08,0xc6,0x21,0x09,0x0}}
+};
+static const struct opl4_region regions_51[] = { /* Sawtooth Lead */
+ {0x01, 0x27, {0x118,9108,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x28, 0x2d, {0x119,8345,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x2e, 0x33, {0x11a,7570,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x34, 0x39, {0x11b,6809,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x3a, 0x3f, {0x11c,6047,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x40, 0x45, {0x11d,5282,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x46, 0x4b, {0x11e,4525,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x4c, 0x51, {0x11f,3746,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x52, 0x57, {0x120,3017,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x58, 0x5d, {0x121,2171,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x5e, 0x66, {0x122,1426,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x67, 0x7f, {0x123,-110,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x01, 0x27, {0x118,9098,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x28, 0x2d, {0x119,8335,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x2e, 0x33, {0x11a,7560,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x34, 0x39, {0x11b,6799,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x3a, 0x3f, {0x11c,6037,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x40, 0x45, {0x11d,5272,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x46, 0x4b, {0x11e,4515,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x4c, 0x51, {0x11f,3736,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x52, 0x57, {0x120,3007,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x58, 0x5d, {0x121,2161,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x5e, 0x66, {0x122,1416,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}},
+ {0x67, 0x7f, {0x123,-120,100, 0,0,0x00,0xc8,0x30,0xf2,0x22,0x0a,0x0}}
+};
+static const struct opl4_region regions_52[] = { /* Calliope Lead */
+ {0x00, 0x7f, {0x0aa,1731,100, 0,0,0x00,0xc2,0x28,0x90,0x00,0x0a,0x2}},
+ {0x15, 0x6c, {0x076,3716,100, 0,0,0x00,0xb6,0x28,0xb0,0x00,0x09,0x2}}
+};
+static const struct opl4_region regions_53[] = { /* Chiffer Lead */
+ {0x00, 0x7f, {0x13a,3665,100, 0,2,0x00,0xcc,0x2a,0xf0,0x10,0x09,0x1}},
+ {0x01, 0x7f, {0x0fe,3660,100, 0,0,0x00,0xbe,0x28,0xf3,0x10,0x17,0x0}}
+};
+static const struct opl4_region regions_54[] = { /* Charang Lead */
+ {0x00, 0x40, {0x0a5,6594,100, 0,3,0x00,0xba,0x33,0xf2,0x11,0x09,0x0}},
+ {0x41, 0x7f, {0x0a6,5433,100, 0,3,0x00,0xba,0x33,0xf2,0x11,0x09,0x0}},
+ {0x01, 0x27, {0x118,9098,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x28, 0x2d, {0x119,8335,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x2e, 0x33, {0x11a,7560,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x34, 0x39, {0x11b,6799,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x3a, 0x3f, {0x11c,6037,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x40, 0x45, {0x11d,5272,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x46, 0x4b, {0x11e,4515,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x4c, 0x51, {0x11f,3736,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x52, 0x57, {0x120,3007,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x58, 0x5d, {0x121,2161,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x5e, 0x66, {0x122,1416,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}},
+ {0x67, 0x7f, {0x123,-120,100, 0,2,0x00,0xa4,0x2a,0xf2,0x22,0x0e,0x0}}
+};
+static const struct opl4_region regions_55[] = { /* Voice Lead */
+ {0x00, 0x7f, {0x0aa,1739,100, 0,6,0x00,0x8c,0x2e,0x90,0x00,0x0a,0x0}},
+ {0x15, 0x6c, {0x02a,3474,100, 0,1,0x00,0xd8,0x29,0xf0,0x05,0x0a,0x0}}
+};
+static const struct opl4_region regions_56[] = { /* 5ths Lead */
+ {0x01, 0x27, {0x118,8468,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x28, 0x2d, {0x119,7705,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x2e, 0x33, {0x11a,6930,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x34, 0x39, {0x11b,6169,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x3a, 0x3f, {0x11c,5407,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x40, 0x45, {0x11d,4642,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x46, 0x4b, {0x11e,3885,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x4c, 0x51, {0x11f,3106,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x52, 0x57, {0x120,2377,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x58, 0x5d, {0x121,1531,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x5e, 0x64, {0x122, 786,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x65, 0x7f, {0x123,-750,100, 0,2,0x00,0xd0,0x32,0xf5,0x20,0x08,0x0}},
+ {0x05, 0x71, {0x002,4503,100, 0,1,0x00,0xb8,0x31,0xb3,0x20,0x0b,0x0}}
+};
+static const struct opl4_region regions_57[] = { /* Bass & Lead */
+ {0x00, 0x7f, {0x117,8109,100, 0,1,0x00,0xbc,0x29,0xf3,0x50,0x08,0x0}},
+ {0x01, 0x27, {0x118,9097,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x28, 0x2d, {0x119,8334,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x2e, 0x33, {0x11a,7559,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x34, 0x39, {0x11b,6798,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x3a, 0x3f, {0x11c,6036,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x40, 0x45, {0x11d,5271,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x46, 0x4b, {0x11e,4514,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x4c, 0x51, {0x11f,3735,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x52, 0x57, {0x120,3006,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x58, 0x5d, {0x121,2160,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x5e, 0x66, {0x122,1415,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}},
+ {0x67, 0x7f, {0x123,-121,100, 0,2,0x00,0xbc,0x2a,0xf2,0x20,0x0a,0x0}}
+};
+static const struct opl4_region regions_58[] = { /* New Age Pad */
+ {0x15, 0x6c, {0x002,4501,100, 0,4,0x00,0xa4,0x24,0x80,0x01,0x05,0x0}},
+ {0x15, 0x6c, {0x0f3,4253,100, 0,3,0x00,0x8c,0x23,0xa2,0x14,0x06,0x1}}
+};
+static const struct opl4_region regions_59[] = { /* Warm Pad */
+ {0x15, 0x6c, {0x04e,5306,100, 2,2,0x00,0x92,0x2a,0x34,0x23,0x05,0x2}},
+ {0x15, 0x6c, {0x029,3575,100,-2,2,0x00,0xbe,0x22,0x31,0x23,0x06,0x0}}
+};
+static const struct opl4_region regions_5a[] = { /* Polysynth Pad */
+ {0x01, 0x27, {0x118,9111,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x28, 0x2d, {0x119,8348,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x2e, 0x33, {0x11a,7573,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x34, 0x39, {0x11b,6812,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x3a, 0x3f, {0x11c,6050,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x40, 0x45, {0x11d,5285,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x46, 0x4b, {0x11e,4528,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x4c, 0x51, {0x11f,3749,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x52, 0x57, {0x120,3020,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x58, 0x5d, {0x121,2174,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x5e, 0x66, {0x122,1429,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x67, 0x7f, {0x123,-107,100, 0,3,0x00,0xae,0x23,0xf2,0x20,0x07,0x1}},
+ {0x00, 0x7f, {0x124,4024,100, 0,2,0x00,0xae,0x22,0xe5,0x20,0x08,0x0}}
+};
+static const struct opl4_region regions_5b[] = { /* Choir Pad */
+ {0x15, 0x3a, {0x018,5010,100, 0,5,0x00,0xb0,0x25,0x70,0x00,0x06,0x0}},
+ {0x3b, 0x40, {0x019,4370,100, 0,5,0x00,0xb0,0x25,0x70,0x00,0x06,0x0}},
+ {0x41, 0x47, {0x01a,3478,100, 0,5,0x00,0xb0,0x25,0x70,0x00,0x06,0x0}},
+ {0x48, 0x6c, {0x01b,2197,100, 0,5,0x00,0xb0,0x25,0x70,0x00,0x06,0x0}},
+ {0x15, 0x6c, {0x02a,3482,100, 0,4,0x00,0x98,0x24,0x65,0x21,0x06,0x0}}
+};
+static const struct opl4_region regions_5c[] = { /* Bowed Pad */
+ {0x15, 0x6c, {0x101,4790,100,-1,1,0x00,0xbe,0x19,0x44,0x14,0x16,0x0}},
+ {0x00, 0x7f, {0x0aa,1720,100, 1,1,0x00,0x94,0x19,0x40,0x00,0x06,0x0}}
+};
+static const struct opl4_region regions_5d[] = { /* Metallic Pad */
+ {0x15, 0x31, {0x00c,6943,100, 0,2,0x00,0xa0,0x0a,0x60,0x03,0x06,0x0}},
+ {0x32, 0x38, {0x00d,5416,100, 0,2,0x00,0xa0,0x0a,0x60,0x03,0x06,0x0}},
+ {0x39, 0x47, {0x00e,4385,100, 0,2,0x00,0xa0,0x0a,0x60,0x03,0x06,0x0}},
+ {0x48, 0x6c, {0x00f,2849,100, 0,2,0x00,0xa0,0x0a,0x60,0x03,0x06,0x0}},
+ {0x00, 0x7f, {0x03f,4224,100, 0,1,0x00,0x9c,0x31,0x65,0x16,0x07,0x0}}
+};
+static const struct opl4_region regions_5e[] = { /* Halo Pad */
+ {0x00, 0x7f, {0x124,4038,100, 0,2,0x00,0xa6,0x1a,0x85,0x23,0x08,0x0}},
+ {0x15, 0x6c, {0x02a,3471,100, 0,3,0x00,0xc0,0x1b,0xc0,0x05,0x06,0x0}}
+};
+static const struct opl4_region regions_5f[] = { /* Sweep Pad */
+ {0x01, 0x27, {0x0d3,9100,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x28, 0x2d, {0x0da,8341,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x2e, 0x33, {0x0d4,7564,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x34, 0x39, {0x0db,6791,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x3a, 0x3f, {0x0d5,6048,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x40, 0x45, {0x0dc,5263,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x46, 0x4b, {0x0d6,4499,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x4c, 0x51, {0x0dd,3747,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x52, 0x57, {0x0d7,3018,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x58, 0x5d, {0x0de,2173,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x5e, 0x63, {0x0d8,1427,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x64, 0x7f, {0x0d9,-109,100, 0,1,0x00,0xce,0x19,0x13,0x11,0x06,0x0}},
+ {0x01, 0x27, {0x0d3,9088,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x28, 0x2d, {0x0da,8329,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x2e, 0x33, {0x0d4,7552,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x34, 0x39, {0x0db,6779,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x3a, 0x3f, {0x0d5,6036,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x40, 0x45, {0x0dc,5251,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x46, 0x4b, {0x0d6,4487,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x4c, 0x51, {0x0dd,3735,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x52, 0x57, {0x0d7,3006,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x58, 0x5d, {0x0de,2161,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x5e, 0x63, {0x0d8,1415,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}},
+ {0x64, 0x7f, {0x0d9,-121,100, 0,0,0x00,0xce,0x18,0x13,0x11,0x06,0x0}}
+};
+static const struct opl4_region regions_60[] = { /* Ice Rain */
+ {0x01, 0x7f, {0x04e,9345,100, 0,2,0x00,0xcc,0x22,0xa3,0x63,0x17,0x0}},
+ {0x00, 0x7f, {0x143,5586, 20, 0,2,0x00,0x6e,0x2a,0xf0,0x05,0x05,0x0}}
+};
+static const struct opl4_region regions_61[] = { /* Soundtrack */
+ {0x15, 0x6c, {0x002,4501,100, 0,2,0x00,0xb6,0x2a,0x60,0x01,0x05,0x0}},
+ {0x15, 0x6c, {0x0f3,1160,100, 0,5,0x00,0xa8,0x2d,0x52,0x14,0x06,0x2}}
+};
+static const struct opl4_region regions_62[] = { /* Crystal */
+ {0x15, 0x6c, {0x0f3,1826,100, 0,3,0x00,0xb8,0x33,0xf6,0x25,0x25,0x0}},
+ {0x15, 0x2c, {0x06d,7454,100, 0,3,0x00,0xac,0x3b,0x85,0x24,0x06,0x0}},
+ {0x2d, 0x36, {0x06e,5925,100, 0,3,0x00,0xac,0x3b,0x85,0x24,0x06,0x0}},
+ {0x37, 0x6c, {0x06f,4403,100, 0,3,0x09,0xac,0x3b,0x85,0x24,0x06,0x0}}
+};
+static const struct opl4_region regions_63[] = { /* Atmosphere */
+ {0x05, 0x71, {0x002,4509,100, 0,2,0x00,0xc8,0x32,0x73,0x22,0x06,0x1}},
+ {0x15, 0x2f, {0x0b3,6964,100, 0,2,0x05,0xc2,0x32,0xf5,0x34,0x07,0x2}},
+ {0x30, 0x36, {0x0b7,5567,100, 0,2,0x0c,0xc2,0x32,0xf5,0x34,0x07,0x2}},
+ {0x37, 0x3c, {0x0b5,4653,100, 0,2,0x00,0xc2,0x32,0xf6,0x34,0x07,0x2}},
+ {0x3d, 0x43, {0x0b4,3892,100, 0,2,0x00,0xc2,0x32,0xf6,0x35,0x07,0x2}},
+ {0x44, 0x60, {0x0b6,2723,100, 0,2,0x00,0xc2,0x32,0xf6,0x35,0x17,0x2}}
+};
+static const struct opl4_region regions_64[] = { /* Brightness */
+ {0x00, 0x7f, {0x137,5285,100, 0,2,0x00,0xbe,0x2a,0xa5,0x18,0x08,0x0}},
+ {0x15, 0x6c, {0x02a,3481,100, 0,1,0x00,0xc8,0x29,0x80,0x05,0x05,0x0}}
+};
+static const struct opl4_region regions_65[] = { /* Goblins */
+ {0x15, 0x6c, {0x002,4501,100,-1,2,0x00,0xca,0x2a,0x40,0x01,0x05,0x0}},
+ {0x15, 0x6c, {0x009,9679, 20, 1,4,0x00,0x3c,0x0c,0x22,0x11,0x06,0x0}}
+};
+static const struct opl4_region regions_66[] = { /* Echoes */
+ {0x15, 0x6c, {0x02a,3487,100, 0,3,0x00,0xae,0x2b,0xf5,0x21,0x06,0x0}},
+ {0x00, 0x7f, {0x124,4027,100, 0,3,0x00,0xae,0x2b,0x85,0x23,0x07,0x0}}
+};
+static const struct opl4_region regions_67[] = { /* Sci-Fi */
+ {0x15, 0x31, {0x00c,6940,100, 0,3,0x00,0xc8,0x2b,0x90,0x05,0x06,0x3}},
+ {0x32, 0x38, {0x00d,5413,100, 0,3,0x00,0xc8,0x2b,0x90,0x05,0x06,0x3}},
+ {0x39, 0x47, {0x00e,4382,100, 0,3,0x00,0xc8,0x2b,0x90,0x05,0x06,0x3}},
+ {0x48, 0x6c, {0x00f,2846,100, 0,3,0x00,0xc8,0x2b,0x90,0x05,0x06,0x3}},
+ {0x15, 0x6c, {0x002,4498,100, 0,2,0x00,0xd4,0x22,0x80,0x01,0x05,0x0}}
+};
+static const struct opl4_region regions_68[] = { /* Sitar */
+ {0x00, 0x7f, {0x10f,4408,100, 0,2,0x00,0xc4,0x32,0xf4,0x15,0x16,0x1}}
+};
+static const struct opl4_region regions_69[] = { /* Banjo */
+ {0x15, 0x34, {0x013,5685,100, 0,0,0x00,0xdc,0x38,0xf6,0x15,0x09,0x0}},
+ {0x35, 0x38, {0x014,5009,100, 0,0,0x00,0xdc,0x38,0xf6,0x15,0x09,0x0}},
+ {0x39, 0x3c, {0x012,4520,100, 0,0,0x00,0xdc,0x38,0xf6,0x15,0x09,0x0}},
+ {0x3d, 0x44, {0x015,3622,100, 0,0,0x00,0xdc,0x38,0xf6,0x15,0x09,0x0}},
+ {0x45, 0x4c, {0x017,2661,100, 0,0,0x00,0xdc,0x38,0xf6,0x15,0x09,0x0}},
+ {0x4d, 0x6d, {0x016,1632,100, 0,0,0x00,0xdc,0x38,0xf6,0x15,0x09,0x0}}
+};
+static const struct opl4_region regions_6a[] = { /* Shamisen */
+ {0x15, 0x6c, {0x10e,3273,100, 0,0,0x00,0xc0,0x28,0xf7,0x76,0x08,0x0}}
+};
+static const struct opl4_region regions_6b[] = { /* Koto */
+ {0x00, 0x7f, {0x0a9,4033,100, 0,0,0x00,0xc6,0x20,0xf0,0x06,0x07,0x0}}
+};
+static const struct opl4_region regions_6c[] = { /* Kalimba */
+ {0x00, 0x7f, {0x137,3749,100, 0,0,0x00,0xce,0x38,0xf5,0x18,0x08,0x0}}
+};
+static const struct opl4_region regions_6d[] = { /* Bagpipe */
+ {0x15, 0x39, {0x0a4,7683,100, 0,4,0x00,0xc0,0x1c,0xf0,0x00,0x09,0x0}},
+ {0x15, 0x39, {0x0a7,7680,100, 0,1,0x00,0xaa,0x19,0xf0,0x00,0x09,0x0}},
+ {0x3a, 0x6c, {0x0a8,3697,100, 0,1,0x00,0xaa,0x19,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_6e[] = { /* Fiddle */
+ {0x15, 0x3a, {0x105,5158,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x3b, 0x3f, {0x102,4754,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x40, 0x41, {0x106,4132,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x42, 0x44, {0x107,4033,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x45, 0x47, {0x108,3580,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x48, 0x4a, {0x10a,2957,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x4b, 0x4c, {0x10b,2724,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x4d, 0x4e, {0x10c,2530,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x4f, 0x51, {0x10d,2166,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}},
+ {0x52, 0x6c, {0x109,1825,100, 0,1,0x00,0xca,0x31,0xf3,0x20,0x09,0x0}}
+};
+static const struct opl4_region regions_6f[] = { /* Shanai */
+ {0x15, 0x6c, {0x041,6946,100, 0,1,0x00,0xc4,0x31,0x95,0x20,0x09,0x0}}
+};
+static const struct opl4_region regions_70[] = { /* Tinkle Bell */
+ {0x15, 0x73, {0x0f3,1821,100, 0,3,0x00,0xc8,0x3b,0xd6,0x25,0x25,0x0}},
+ {0x00, 0x7f, {0x137,5669,100, 0,3,0x00,0x66,0x3b,0xf5,0x18,0x08,0x0}}
+};
+static const struct opl4_region regions_71[] = { /* Agogo */
+ {0x15, 0x74, {0x00b,2474,100, 0,0,0x00,0xd2,0x38,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_72[] = { /* Steel Drums */
+ {0x01, 0x7f, {0x0fe,3670,100, 0,0,0x00,0xca,0x38,0xf3,0x06,0x17,0x1}},
+ {0x15, 0x6c, {0x100,9602,100, 0,0,0x00,0x54,0x38,0xb0,0x05,0x16,0x1}}
+};
+static const struct opl4_region regions_73[] = { /* Woodblock */
+ {0x15, 0x6c, {0x02c,2963, 50, 0,0,0x07,0xd4,0x00,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_74[] = { /* Taiko Drum */
+ {0x13, 0x6c, {0x03e,1194, 50, 0,0,0x00,0xaa,0x38,0xf0,0x04,0x04,0x0}}
+};
+static const struct opl4_region regions_75[] = { /* Melodic Tom */
+ {0x15, 0x6c, {0x0c7,6418, 50, 0,0,0x00,0xe4,0x38,0xf0,0x05,0x01,0x0}}
+};
+static const struct opl4_region regions_76[] = { /* Synth Drum */
+ {0x15, 0x6c, {0x026,3898, 50, 0,0,0x00,0xd0,0x38,0xf0,0x04,0x04,0x0}}
+};
+static const struct opl4_region regions_77[] = { /* Reverse Cymbal */
+ {0x15, 0x6c, {0x031,4138, 50, 0,0,0x00,0xfe,0x38,0x3a,0xf0,0x09,0x0}}
+};
+static const struct opl4_region regions_78[] = { /* Guitar Fret Noise */
+ {0x15, 0x6c, {0x138,5266,100, 0,0,0x00,0xa0,0x38,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_79[] = { /* Breath Noise */
+ {0x01, 0x7f, {0x125,4269,100, 0,0,0x1e,0xd0,0x38,0xf0,0x00,0x09,0x0}}
+};
+static const struct opl4_region regions_7a[] = { /* Seashore */
+ {0x15, 0x6c, {0x008,2965, 20,-2,0,0x00,0xfe,0x00,0x20,0x03,0x04,0x0}},
+ {0x01, 0x7f, {0x037,4394, 20, 2,0,0x14,0xfe,0x00,0x20,0x04,0x05,0x0}}
+};
+static const struct opl4_region regions_7b[] = { /* Bird Tweet */
+ {0x15, 0x6c, {0x009,8078, 5,-4,7,0x00,0xc2,0x0f,0x22,0x12,0x07,0x0}},
+ {0x15, 0x6c, {0x009,3583, 5, 4,5,0x00,0xae,0x15,0x72,0x12,0x07,0x0}}
+};
+static const struct opl4_region regions_7c[] = { /* Telephone Ring */
+ {0x15, 0x6c, {0x003,3602, 10, 0,0,0x00,0xce,0x00,0xf0,0x00,0x0f,0x0}}
+};
+static const struct opl4_region regions_7d[] = { /* Helicopter */
+ {0x0c, 0x7f, {0x001,2965, 10,-2,0,0x00,0xe0,0x08,0x30,0x01,0x07,0x0}},
+ {0x01, 0x7f, {0x037,4394, 10, 2,0,0x44,0x76,0x00,0x30,0x01,0x07,0x0}}
+};
+static const struct opl4_region regions_7e[] = { /* Applause */
+ {0x15, 0x6c, {0x036,8273, 20,-6,7,0x00,0xc4,0x0f,0x70,0x01,0x05,0x0}},
+ {0x15, 0x6c, {0x036,8115, 5, 6,7,0x00,0xc6,0x07,0x70,0x01,0x05,0x0}}
+};
+static const struct opl4_region regions_7f[] = { /* Gun Shot */
+ {0x15, 0x6c, {0x139,2858, 20, 0,0,0x00,0xbe,0x38,0xf0,0x03,0x00,0x0}}
+};
+static const struct opl4_region regions_drums[] = {
+ {0x18, 0x18, {0x0cb,6397,100, 3,0,0x00,0xf4,0x38,0xc9,0x1c,0x0c,0x0}},
+ {0x19, 0x19, {0x0c4,3714,100, 0,0,0x00,0xe0,0x00,0x97,0x19,0x09,0x0}},
+ {0x1a, 0x1a, {0x0c4,3519,100, 0,0,0x00,0xea,0x00,0x61,0x01,0x07,0x0}},
+ {0x1b, 0x1b, {0x0c4,3586,100, 0,0,0x00,0xea,0x00,0xf7,0x19,0x09,0x0}},
+ {0x1c, 0x1c, {0x0c4,3586,100, 0,0,0x00,0xea,0x00,0x81,0x01,0x07,0x0}},
+ {0x1e, 0x1e, {0x0c3,4783,100, 0,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x1f, 0x1f, {0x0d1,4042,100, 0,0,0x00,0xd6,0x00,0xf0,0x05,0x05,0x0}},
+ {0x20, 0x20, {0x0d2,5943,100, 0,0,0x00,0xcc,0x00,0xf0,0x00,0x09,0x0}},
+ {0x21, 0x21, {0x011,3842,100, 0,0,0x00,0xea,0x00,0xf0,0x16,0x06,0x0}},
+ {0x23, 0x23, {0x011,4098,100, 0,0,0x00,0xea,0x00,0xf0,0x16,0x06,0x0}},
+ {0x24, 0x24, {0x011,4370,100, 0,0,0x00,0xea,0x00,0xf0,0x00,0x06,0x0}},
+ {0x25, 0x25, {0x0d2,4404,100, 0,0,0x00,0xd6,0x00,0xf0,0x00,0x06,0x0}},
+ {0x26, 0x26, {0x0d1,4298,100, 0,0,0x00,0xd6,0x00,0xf0,0x05,0x05,0x0}},
+ {0x27, 0x27, {0x00a,4403,100,-1,0,0x00,0xd6,0x00,0xf0,0x00,0x09,0x0}},
+ {0x28, 0x28, {0x0d1,4554,100, 0,0,0x00,0xdc,0x00,0xf0,0x07,0x07,0x0}},
+ {0x29, 0x29, {0x0c8,4242,100,-4,0,0x00,0xd6,0x00,0xf6,0x16,0x06,0x0}},
+ {0x2a, 0x2a, {0x079,6160,100, 2,0,0x00,0xe0,0x00,0xf5,0x19,0x09,0x0}},
+ {0x2b, 0x2b, {0x0c8,4626,100,-3,0,0x00,0xd6,0x00,0xf6,0x16,0x06,0x0}},
+ {0x2c, 0x2c, {0x07b,6039,100, 2,0,0x00,0xd6,0x00,0xf0,0x00,0x09,0x0}},
+ {0x2d, 0x2d, {0x0c8,5394,100,-2,0,0x00,0xd6,0x00,0xf6,0x16,0x06,0x0}},
+ {0x2e, 0x2e, {0x07a,5690,100, 2,0,0x00,0xd6,0x00,0xf0,0x00,0x05,0x0}},
+ {0x2f, 0x2f, {0x0c7,5185,100, 2,0,0x00,0xe0,0x00,0xf6,0x17,0x07,0x0}},
+ {0x30, 0x30, {0x0c7,5650,100, 3,0,0x00,0xe0,0x00,0xf6,0x17,0x07,0x0}},
+ {0x31, 0x31, {0x031,4395,100, 2,0,0x00,0xea,0x00,0xf0,0x05,0x05,0x0}},
+ {0x32, 0x32, {0x0c7,6162,100, 4,0,0x00,0xe0,0x00,0xf6,0x17,0x07,0x0}},
+ {0x33, 0x33, {0x02e,4391,100,-2,0,0x00,0xea,0x00,0xf0,0x05,0x05,0x0}},
+ {0x34, 0x34, {0x07a,3009,100,-2,0,0x00,0xea,0x00,0xf2,0x15,0x05,0x0}},
+ {0x35, 0x35, {0x021,4522,100,-3,0,0x00,0xd6,0x00,0xf0,0x05,0x05,0x0}},
+ {0x36, 0x36, {0x025,5163,100, 1,0,0x00,0xe0,0x00,0xf0,0x00,0x09,0x0}},
+ {0x37, 0x37, {0x031,5287,100,-1,0,0x00,0xea,0x00,0xf5,0x16,0x06,0x0}},
+ {0x38, 0x38, {0x01d,4395,100, 2,0,0x00,0xe0,0x00,0xf0,0x00,0x09,0x0}},
+ {0x39, 0x39, {0x031,4647,100,-2,0,0x00,0xea,0x00,0xf4,0x16,0x06,0x0}},
+ {0x3a, 0x3a, {0x09d,4426,100,-4,0,0x00,0xe0,0x00,0xf4,0x17,0x07,0x0}},
+ {0x3b, 0x3b, {0x02e,4659,100,-2,0,0x00,0xea,0x00,0xf0,0x06,0x06,0x0}},
+ {0x3c, 0x3c, {0x01c,4769,100, 4,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x3d, 0x3d, {0x01c,4611,100, 4,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x3e, 0x3e, {0x01e,4402,100,-3,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x3f, 0x3f, {0x01f,4387,100,-3,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x40, 0x40, {0x01f,3983,100,-2,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x41, 0x41, {0x09c,4526,100, 2,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x42, 0x42, {0x09c,4016,100, 2,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x43, 0x43, {0x00b,4739,100,-4,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x44, 0x44, {0x00b,4179,100,-4,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x45, 0x45, {0x02f,4787,100,-4,0,0x00,0xd6,0x00,0xf0,0x00,0x09,0x0}},
+ {0x46, 0x46, {0x030,4665,100,-4,0,0x00,0xd6,0x00,0xf0,0x00,0x09,0x0}},
+ {0x47, 0x47, {0x144,4519,100, 4,0,0x00,0xea,0x00,0xf0,0x00,0x0b,0x0}},
+ {0x48, 0x48, {0x144,4111,100, 4,0,0x00,0xea,0x00,0xf0,0x00,0x0b,0x0}},
+ {0x49, 0x49, {0x024,6408,100, 3,0,0x00,0xe0,0x00,0xf0,0x00,0x09,0x0}},
+ {0x4a, 0x4a, {0x024,4144,100, 3,0,0x00,0xcc,0x00,0xf0,0x00,0x09,0x0}},
+ {0x4b, 0x4b, {0x020,4001,100, 2,0,0x00,0xe0,0x00,0xf0,0x00,0x09,0x0}},
+ {0x4c, 0x4c, {0x02c,4402,100, 4,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x4d, 0x4d, {0x02c,3612,100, 4,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x4e, 0x4e, {0x022,4129,100,-2,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x4f, 0x4f, {0x023,4147,100,-2,0,0x00,0xea,0x00,0xf0,0x00,0x09,0x0}},
+ {0x50, 0x50, {0x032,4412,100,-4,0,0x00,0xd6,0x00,0xf0,0x08,0x09,0x0}},
+ {0x51, 0x51, {0x032,4385,100,-4,0,0x00,0xd6,0x00,0xf0,0x00,0x09,0x0}},
+ {0x52, 0x52, {0x02f,5935,100,-1,0,0x00,0xd6,0x00,0xf0,0x00,0x09,0x0}}
+};
+
+#define REGION(num) { ARRAY_SIZE(regions ## num), regions ## num }
+const struct opl4_region_ptr snd_yrw801_regions[0x81] = {
+ REGION(_00), REGION(_01), REGION(_02), REGION(_03),
+ REGION(_04), REGION(_05), REGION(_06), REGION(_07),
+ REGION(_08), REGION(_09), REGION(_0a), REGION(_0b),
+ REGION(_0c), REGION(_0d), REGION(_0e), REGION(_0f),
+ REGION(_10), REGION(_11), REGION(_12), REGION(_13),
+ REGION(_14), REGION(_15), REGION(_16), REGION(_17),
+ REGION(_18), REGION(_19), REGION(_1a), REGION(_1b),
+ REGION(_1c), REGION(_1d), REGION(_1e), REGION(_1f),
+ REGION(_20), REGION(_21), REGION(_22), REGION(_23),
+ REGION(_24), REGION(_25), REGION(_26), REGION(_27),
+ REGION(_28), REGION(_29), REGION(_2a), REGION(_2b),
+ REGION(_2c), REGION(_2d), REGION(_2e), REGION(_2f),
+ REGION(_30), REGION(_31), REGION(_32), REGION(_33),
+ REGION(_34), REGION(_35), REGION(_36), REGION(_37),
+ REGION(_38), REGION(_39), REGION(_3a), REGION(_3b),
+ REGION(_3c), REGION(_3d), REGION(_3e), REGION(_3f),
+ REGION(_40), REGION(_41), REGION(_42), REGION(_43),
+ REGION(_44), REGION(_45), REGION(_46), REGION(_47),
+ REGION(_48), REGION(_49), REGION(_4a), REGION(_4b),
+ REGION(_4c), REGION(_4d), REGION(_4e), REGION(_4f),
+ REGION(_50), REGION(_51), REGION(_52), REGION(_53),
+ REGION(_54), REGION(_55), REGION(_56), REGION(_57),
+ REGION(_58), REGION(_59), REGION(_5a), REGION(_5b),
+ REGION(_5c), REGION(_5d), REGION(_5e), REGION(_5f),
+ REGION(_60), REGION(_61), REGION(_62), REGION(_63),
+ REGION(_64), REGION(_65), REGION(_66), REGION(_67),
+ REGION(_68), REGION(_69), REGION(_6a), REGION(_6b),
+ REGION(_6c), REGION(_6d), REGION(_6e), REGION(_6f),
+ REGION(_70), REGION(_71), REGION(_72), REGION(_73),
+ REGION(_74), REGION(_75), REGION(_76), REGION(_77),
+ REGION(_78), REGION(_79), REGION(_7a), REGION(_7b),
+ REGION(_7c), REGION(_7d), REGION(_7e), REGION(_7f),
+ REGION(_drums)
+};
diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c
new file mode 100644
index 0000000..3c93c23
--- /dev/null
+++ b/sound/drivers/pcm-indirect2.c
@@ -0,0 +1,573 @@
+/*
+ * Helper functions for indirect PCM data transfer to a simple FIFO in
+ * hardware (small, no possibility to read "hardware io position",
+ * updating position done by interrupt, ...)
+ *
+ * Copyright (c) by 2007 Joachim Foerster <JOFT@gmx.de>
+ *
+ * Based on "pcm-indirect.h" (alsa-driver-1.0.13) by
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ * Jaroslav Kysela <perex@suse.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+/* snd_printk/d() */
+#include <sound/core.h>
+/* struct snd_pcm_substream, struct snd_pcm_runtime, snd_pcm_uframes_t
+ * snd_pcm_period_elapsed() */
+#include <sound/pcm.h>
+
+#include "pcm-indirect2.h"
+
+#ifdef SND_PCM_INDIRECT2_STAT
+/* jiffies */
+#include <linux/jiffies.h>
+
+void snd_pcm_indirect2_stat(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int i;
+ int j;
+ int k;
+ int seconds = (rec->lastbytetime - rec->firstbytetime) / HZ;
+
+ snd_printk(KERN_DEBUG "STAT: mul_elapsed: %u, mul_elapsed_real: %d, "
+ "irq_occured: %d\n",
+ rec->mul_elapsed, rec->mul_elapsed_real, rec->irq_occured);
+ snd_printk(KERN_DEBUG "STAT: min_multiple: %d (irqs/period)\n",
+ rec->min_multiple);
+ snd_printk(KERN_DEBUG "STAT: firstbytetime: %lu, lastbytetime: %lu, "
+ "firstzerotime: %lu\n",
+ rec->firstbytetime, rec->lastbytetime, rec->firstzerotime);
+ snd_printk(KERN_DEBUG "STAT: bytes2hw: %u Bytes => (by runtime->rate) "
+ "length: %d s\n",
+ rec->bytes2hw, rec->bytes2hw / 2 / 2 / runtime->rate);
+ snd_printk(KERN_DEBUG "STAT: (by measurement) length: %d => "
+ "rate: %d Bytes/s = %d Frames/s|Hz\n",
+ seconds, rec->bytes2hw / seconds,
+ rec->bytes2hw / 2 / 2 / seconds);
+ snd_printk(KERN_DEBUG
+ "STAT: zeros2hw: %u = %d ms ~ %d * %d zero copies\n",
+ rec->zeros2hw, ((rec->zeros2hw / 2 / 2) * 1000) /
+ runtime->rate,
+ rec->zeros2hw / (rec->hw_buffer_size / 2),
+ (rec->hw_buffer_size / 2));
+ snd_printk(KERN_DEBUG "STAT: pointer_calls: %u, lastdifftime: %u\n",
+ rec->pointer_calls, rec->lastdifftime);
+ snd_printk(KERN_DEBUG "STAT: sw_io: %d, sw_data: %d\n", rec->sw_io,
+ rec->sw_data);
+ snd_printk(KERN_DEBUG "STAT: byte_sizes[]:\n");
+ k = 0;
+ for (j = 0; j < 8; j++) {
+ for (i = j * 8; i < (j + 1) * 8; i++)
+ if (rec->byte_sizes[i] != 0) {
+ snd_printk(KERN_DEBUG "%u: %u",
+ i, rec->byte_sizes[i]);
+ k++;
+ }
+ if (((k % 8) == 0) && (k != 0)) {
+ snd_printk(KERN_DEBUG "\n");
+ k = 0;
+ }
+ }
+ snd_printk(KERN_DEBUG "\n");
+ snd_printk(KERN_DEBUG "STAT: zero_sizes[]:\n");
+ for (j = 0; j < 8; j++) {
+ k = 0;
+ for (i = j * 8; i < (j + 1) * 8; i++)
+ if (rec->zero_sizes[i] != 0)
+ snd_printk(KERN_DEBUG "%u: %u",
+ i, rec->zero_sizes[i]);
+ else
+ k++;
+ if (!k)
+ snd_printk(KERN_DEBUG "\n");
+ }
+ snd_printk(KERN_DEBUG "\n");
+ snd_printk(KERN_DEBUG "STAT: min_adds[]:\n");
+ for (j = 0; j < 8; j++) {
+ if (rec->min_adds[j] != 0)
+ snd_printk(KERN_DEBUG "%u: %u", j, rec->min_adds[j]);
+ }
+ snd_printk(KERN_DEBUG "\n");
+ snd_printk(KERN_DEBUG "STAT: mul_adds[]:\n");
+ for (j = 0; j < 8; j++) {
+ if (rec->mul_adds[j] != 0)
+ snd_printk(KERN_DEBUG "%u: %u", j, rec->mul_adds[j]);
+ }
+ snd_printk(KERN_DEBUG "\n");
+ snd_printk(KERN_DEBUG
+ "STAT: zero_times_saved: %d, zero_times_notsaved: %d\n",
+ rec->zero_times_saved, rec->zero_times_notsaved);
+ /* snd_printk(KERN_DEBUG "STAT: zero_times[]\n");
+ i = 0;
+ for (j = 0; j < 3750; j++) {
+ if (rec->zero_times[j] != 0) {
+ snd_printk(KERN_DEBUG "%u: %u", j, rec->zero_times[j]);
+ i++;
+ }
+ if (((i % 8) == 0) && (i != 0))
+ snd_printk(KERN_DEBUG "\n");
+ }
+ snd_printk(KERN_DEBUG "\n"); */
+ return;
+}
+#endif
+
+/*
+ * _internal_ helper function for playback/capture transfer function
+ */
+static void
+snd_pcm_indirect2_increase_min_periods(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ int isplay, int iscopy,
+ unsigned int bytes)
+{
+ if (rec->min_periods >= 0) {
+ if (iscopy) {
+ rec->sw_io += bytes;
+ if (rec->sw_io >= rec->sw_buffer_size)
+ rec->sw_io -= rec->sw_buffer_size;
+ } else if (isplay) {
+ /* If application does not write data in multiples of
+ * a period, move sw_data to the next correctly aligned
+ * position, so that sw_io can converge to it (in the
+ * next step).
+ */
+ if (!rec->check_alignment) {
+ if (rec->bytes2hw %
+ snd_pcm_lib_period_bytes(substream)) {
+ unsigned bytes2hw_aligned =
+ (1 +
+ (rec->bytes2hw /
+ snd_pcm_lib_period_bytes
+ (substream))) *
+ snd_pcm_lib_period_bytes
+ (substream);
+ rec->sw_data =
+ bytes2hw_aligned %
+ rec->sw_buffer_size;
+#ifdef SND_PCM_INDIRECT2_STAT
+ snd_printk(KERN_DEBUG
+ "STAT: @re-align: aligned "
+ "bytes2hw to next period "
+ "size boundary: %d "
+ "(instead of %d)\n",
+ bytes2hw_aligned,
+ rec->bytes2hw);
+ snd_printk(KERN_DEBUG
+ "STAT: @re-align: sw_data "
+ "moves to: %d\n",
+ rec->sw_data);
+#endif
+ }
+ rec->check_alignment = 1;
+ }
+ /* We are at the end and are copying zeros into the
+ * fifo.
+ * Now, we have to make sure that sw_io is increased
+ * until the position of sw_data: Filling the fifo with
+ * the first zeros means, the last bytes were played.
+ */
+ if (rec->sw_io != rec->sw_data) {
+ unsigned int diff;
+ if (rec->sw_data > rec->sw_io)
+ diff = rec->sw_data - rec->sw_io;
+ else
+ diff = (rec->sw_buffer_size -
+ rec->sw_io) +
+ rec->sw_data;
+ if (bytes >= diff)
+ rec->sw_io = rec->sw_data;
+ else {
+ rec->sw_io += bytes;
+ if (rec->sw_io >= rec->sw_buffer_size)
+ rec->sw_io -=
+ rec->sw_buffer_size;
+ }
+ }
+ }
+ rec->min_period_count += bytes;
+ if (rec->min_period_count >= (rec->hw_buffer_size / 2)) {
+ rec->min_periods += (rec->min_period_count /
+ (rec->hw_buffer_size / 2));
+#ifdef SND_PCM_INDIRECT2_STAT
+ if ((rec->min_period_count /
+ (rec->hw_buffer_size / 2)) > 7)
+ snd_printk(KERN_DEBUG
+ "STAT: more than 7 (%d) min_adds "
+ "at once - too big to save!\n",
+ (rec->min_period_count /
+ (rec->hw_buffer_size / 2)));
+ else
+ rec->min_adds[(rec->min_period_count /
+ (rec->hw_buffer_size / 2))]++;
+#endif
+ rec->min_period_count = (rec->min_period_count %
+ (rec->hw_buffer_size / 2));
+ }
+ } else if (isplay && iscopy)
+ rec->min_periods = 0;
+}
+
+/*
+ * helper function for playback/capture pointer callback
+ */
+snd_pcm_uframes_t
+snd_pcm_indirect2_pointer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec)
+{
+#ifdef SND_PCM_INDIRECT2_STAT
+ rec->pointer_calls++;
+#endif
+ return bytes_to_frames(substream->runtime, rec->sw_io);
+}
+
+/*
+ * _internal_ helper function for playback interrupt callback
+ */
+static void
+snd_pcm_indirect2_playback_transfer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ snd_pcm_indirect2_copy_t copy,
+ snd_pcm_indirect2_zero_t zero)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr;
+
+ /* runtime->control->appl_ptr: position where ALSA will write next time
+ * rec->appl_ptr: position where ALSA was last time
+ * diff: obviously ALSA wrote that much bytes into the intermediate
+ * buffer since we checked last time
+ */
+ snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr;
+
+ if (diff) {
+#ifdef SND_PCM_INDIRECT2_STAT
+ rec->lastdifftime = jiffies;
+#endif
+ if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2))
+ diff += runtime->boundary;
+ /* number of bytes "added" by ALSA increases the number of
+ * bytes which are ready to "be transfered to HW"/"played"
+ * Then, set rec->appl_ptr to not count bytes twice next time.
+ */
+ rec->sw_ready += (int)frames_to_bytes(runtime, diff);
+ rec->appl_ptr = appl_ptr;
+ }
+ if (rec->hw_ready && (rec->sw_ready <= 0)) {
+ unsigned int bytes;
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ if (rec->firstzerotime == 0) {
+ rec->firstzerotime = jiffies;
+ snd_printk(KERN_DEBUG
+ "STAT: @firstzerotime: mul_elapsed: %d, "
+ "min_period_count: %d\n",
+ rec->mul_elapsed, rec->min_period_count);
+ snd_printk(KERN_DEBUG
+ "STAT: @firstzerotime: sw_io: %d, "
+ "sw_data: %d, appl_ptr: %u\n",
+ rec->sw_io, rec->sw_data,
+ (unsigned int)appl_ptr);
+ }
+ if ((jiffies - rec->firstzerotime) < 3750) {
+ rec->zero_times[(jiffies - rec->firstzerotime)]++;
+ rec->zero_times_saved++;
+ } else
+ rec->zero_times_notsaved++;
+#endif
+ bytes = zero(substream, rec);
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ rec->zeros2hw += bytes;
+ if (bytes < 64)
+ rec->zero_sizes[bytes]++;
+ else
+ snd_printk(KERN_DEBUG
+ "STAT: %d zero Bytes copied to hardware at "
+ "once - too big to save!\n",
+ bytes);
+#endif
+ snd_pcm_indirect2_increase_min_periods(substream, rec, 1, 0,
+ bytes);
+ return;
+ }
+ while (rec->hw_ready && (rec->sw_ready > 0)) {
+ /* sw_to_end: max. number of bytes that can be read/take from
+ * the current position (sw_data) in _one_ step
+ */
+ unsigned int sw_to_end = rec->sw_buffer_size - rec->sw_data;
+
+ /* bytes: number of bytes we have available (for reading) */
+ unsigned int bytes = rec->sw_ready;
+
+ if (sw_to_end < bytes)
+ bytes = sw_to_end;
+ if (!bytes)
+ break;
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ if (rec->firstbytetime == 0)
+ rec->firstbytetime = jiffies;
+ rec->lastbytetime = jiffies;
+#endif
+ /* copy bytes from intermediate buffer position sw_data to the
+ * HW and return number of bytes actually written
+ * Furthermore, set hw_ready to 0, if the fifo isn't empty
+ * now => more could be transfered to fifo
+ */
+ bytes = copy(substream, rec, bytes);
+ rec->bytes2hw += bytes;
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ if (bytes < 64)
+ rec->byte_sizes[bytes]++;
+ else
+ snd_printk(KERN_DEBUG
+ "STAT: %d Bytes copied to hardware at once "
+ "- too big to save!\n",
+ bytes);
+#endif
+ /* increase sw_data by the number of actually written bytes
+ * (= number of taken bytes from intermediate buffer)
+ */
+ rec->sw_data += bytes;
+ if (rec->sw_data == rec->sw_buffer_size)
+ rec->sw_data = 0;
+ /* now sw_data is the position where ALSA is going to write
+ * in the intermediate buffer next time = position we are going
+ * to read from next time
+ */
+
+ snd_pcm_indirect2_increase_min_periods(substream, rec, 1, 1,
+ bytes);
+
+ /* we read bytes from intermediate buffer, so we need to say
+ * that the number of bytes ready for transfer are decreased
+ * now
+ */
+ rec->sw_ready -= bytes;
+ }
+ return;
+}
+
+/*
+ * helper function for playback interrupt routine
+ */
+void
+snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ snd_pcm_indirect2_copy_t copy,
+ snd_pcm_indirect2_zero_t zero)
+{
+#ifdef SND_PCM_INDIRECT2_STAT
+ rec->irq_occured++;
+#endif
+ /* hardware played some bytes, so there is room again (in fifo) */
+ rec->hw_ready = 1;
+
+ /* don't call ack() now, instead call transfer() function directly
+ * (normally called by ack() )
+ */
+ snd_pcm_indirect2_playback_transfer(substream, rec, copy, zero);
+
+ if (rec->min_periods >= rec->min_multiple) {
+#ifdef SND_PCM_INDIRECT2_STAT
+ if ((rec->min_periods / rec->min_multiple) > 7)
+ snd_printk(KERN_DEBUG
+ "STAT: more than 7 (%d) mul_adds - too big "
+ "to save!\n",
+ (rec->min_periods / rec->min_multiple));
+ else
+ rec->mul_adds[(rec->min_periods /
+ rec->min_multiple)]++;
+ rec->mul_elapsed_real += (rec->min_periods /
+ rec->min_multiple);
+ rec->mul_elapsed++;
+#endif
+ rec->min_periods = (rec->min_periods % rec->min_multiple);
+ snd_pcm_period_elapsed(substream);
+ }
+}
+
+/*
+ * _internal_ helper function for capture interrupt callback
+ */
+static void
+snd_pcm_indirect2_capture_transfer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ snd_pcm_indirect2_copy_t copy,
+ snd_pcm_indirect2_zero_t null)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr;
+ snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr;
+
+ if (diff) {
+#ifdef SND_PCM_INDIRECT2_STAT
+ rec->lastdifftime = jiffies;
+#endif
+ if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2))
+ diff += runtime->boundary;
+ rec->sw_ready -= frames_to_bytes(runtime, diff);
+ rec->appl_ptr = appl_ptr;
+ }
+ /* if hardware has something, but the intermediate buffer is full
+ * => skip contents of buffer
+ */
+ if (rec->hw_ready && (rec->sw_ready >= (int)rec->sw_buffer_size)) {
+ unsigned int bytes;
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ if (rec->firstzerotime == 0) {
+ rec->firstzerotime = jiffies;
+ snd_printk(KERN_DEBUG "STAT: (capture) "
+ "@firstzerotime: mul_elapsed: %d, "
+ "min_period_count: %d\n",
+ rec->mul_elapsed, rec->min_period_count);
+ snd_printk(KERN_DEBUG "STAT: (capture) "
+ "@firstzerotime: sw_io: %d, sw_data: %d, "
+ "appl_ptr: %u\n",
+ rec->sw_io, rec->sw_data,
+ (unsigned int)appl_ptr);
+ }
+ if ((jiffies - rec->firstzerotime) < 3750) {
+ rec->zero_times[(jiffies - rec->firstzerotime)]++;
+ rec->zero_times_saved++;
+ } else
+ rec->zero_times_notsaved++;
+#endif
+ bytes = null(substream, rec);
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ rec->zeros2hw += bytes;
+ if (bytes < 64)
+ rec->zero_sizes[bytes]++;
+ else
+ snd_printk(KERN_DEBUG
+ "STAT: (capture) %d zero Bytes copied to "
+ "hardware at once - too big to save!\n",
+ bytes);
+#endif
+ snd_pcm_indirect2_increase_min_periods(substream, rec, 0, 0,
+ bytes);
+ /* report an overrun */
+ rec->sw_io = SNDRV_PCM_POS_XRUN;
+ return;
+ }
+ while (rec->hw_ready && (rec->sw_ready < (int)rec->sw_buffer_size)) {
+ /* sw_to_end: max. number of bytes that we can write to the
+ * intermediate buffer (until it's end)
+ */
+ size_t sw_to_end = rec->sw_buffer_size - rec->sw_data;
+
+ /* bytes: max. number of bytes, which may be copied to the
+ * intermediate buffer without overflow (in _one_ step)
+ */
+ size_t bytes = rec->sw_buffer_size - rec->sw_ready;
+
+ /* limit number of bytes (for transfer) by available room in
+ * the intermediate buffer
+ */
+ if (sw_to_end < bytes)
+ bytes = sw_to_end;
+ if (!bytes)
+ break;
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ if (rec->firstbytetime == 0)
+ rec->firstbytetime = jiffies;
+ rec->lastbytetime = jiffies;
+#endif
+ /* copy bytes from the intermediate buffer (position sw_data)
+ * to the HW at most and return number of bytes actually copied
+ * from HW
+ * Furthermore, set hw_ready to 0, if the fifo is empty now.
+ */
+ bytes = copy(substream, rec, bytes);
+ rec->bytes2hw += bytes;
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ if (bytes < 64)
+ rec->byte_sizes[bytes]++;
+ else
+ snd_printk(KERN_DEBUG
+ "STAT: (capture) %d Bytes copied to "
+ "hardware at once - too big to save!\n",
+ bytes);
+#endif
+ /* increase sw_data by the number of actually copied bytes from
+ * HW
+ */
+ rec->sw_data += bytes;
+ if (rec->sw_data == rec->sw_buffer_size)
+ rec->sw_data = 0;
+
+ snd_pcm_indirect2_increase_min_periods(substream, rec, 0, 1,
+ bytes);
+
+ /* number of bytes in the intermediate buffer, which haven't
+ * been fetched by ALSA yet.
+ */
+ rec->sw_ready += bytes;
+ }
+ return;
+}
+
+/*
+ * helper function for capture interrupt routine
+ */
+void
+snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ snd_pcm_indirect2_copy_t copy,
+ snd_pcm_indirect2_zero_t null)
+{
+#ifdef SND_PCM_INDIRECT2_STAT
+ rec->irq_occured++;
+#endif
+ /* hardware recorded some bytes, so there is something to read from the
+ * record fifo:
+ */
+ rec->hw_ready = 1;
+
+ /* don't call ack() now, instead call transfer() function directly
+ * (normally called by ack() )
+ */
+ snd_pcm_indirect2_capture_transfer(substream, rec, copy, null);
+
+ if (rec->min_periods >= rec->min_multiple) {
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ if ((rec->min_periods / rec->min_multiple) > 7)
+ snd_printk(KERN_DEBUG
+ "STAT: more than 7 (%d) mul_adds - "
+ "too big to save!\n",
+ (rec->min_periods / rec->min_multiple));
+ else
+ rec->mul_adds[(rec->min_periods /
+ rec->min_multiple)]++;
+ rec->mul_elapsed_real += (rec->min_periods /
+ rec->min_multiple);
+ rec->mul_elapsed++;
+#endif
+ rec->min_periods = (rec->min_periods % rec->min_multiple);
+ snd_pcm_period_elapsed(substream);
+ }
+}
diff --git a/sound/drivers/pcm-indirect2.h b/sound/drivers/pcm-indirect2.h
new file mode 100644
index 0000000..2ea6e46
--- /dev/null
+++ b/sound/drivers/pcm-indirect2.h
@@ -0,0 +1,140 @@
+/*
+ * Helper functions for indirect PCM data transfer to a simple FIFO in
+ * hardware (small, no possibility to read "hardware io position",
+ * updating position done by interrupt, ...)
+ *
+ * Copyright (c) by 2007 Joachim Foerster <JOFT@gmx.de>
+ *
+ * Based on "pcm-indirect.h" (alsa-driver-1.0.13) by
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ * Jaroslav Kysela <perex@suse.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef __SOUND_PCM_INDIRECT2_H
+#define __SOUND_PCM_INDIRECT2_H
+
+/* struct snd_pcm_substream, struct snd_pcm_runtime, snd_pcm_uframes_t */
+#include <sound/pcm.h>
+
+/* Debug options for code which may be removed completely in a final version */
+#ifdef CONFIG_SND_DEBUG
+#define SND_PCM_INDIRECT2_STAT /* turn on some "statistics" about the
+ * process of copying bytes from the
+ * intermediate buffer to the hardware
+ * fifo and the other way round
+ */
+#endif
+
+struct snd_pcm_indirect2 {
+ unsigned int hw_buffer_size; /* Byte size of hardware buffer */
+ int hw_ready; /* playback: 1 = hw fifo has room left,
+ * 0 = hw fifo is full
+ */
+ unsigned int min_multiple;
+ int min_periods; /* counts number of min. periods until
+ * min_multiple is reached
+ */
+ int min_period_count; /* counts bytes to count number of
+ * min. periods
+ */
+
+ unsigned int sw_buffer_size; /* Byte size of software buffer */
+
+ /* sw_data: position in intermediate buffer, where we will read (or
+ * write) from/to next time (to transfer data to/from HW)
+ */
+ unsigned int sw_data; /* Offset to next dst (or src) in sw
+ * ring buffer
+ */
+ /* easiest case (playback):
+ * sw_data is nearly the same as ~ runtime->control->appl_ptr, with the
+ * exception that sw_data is "behind" by the number if bytes ALSA wrote
+ * to the intermediate buffer last time.
+ * A call to ack() callback synchronizes both indirectly.
+ */
+
+ /* We have no real sw_io pointer here. Usually sw_io is pointing to the
+ * current playback/capture position _inside_ the hardware. Devices
+ * with plain FIFOs often have no possibility to publish this position.
+ * So we say: if sw_data is updated, that means bytes were copied to
+ * the hardware, we increase sw_io by that amount, because there have
+ * to be as much bytes which were played. So sw_io will stay behind
+ * sw_data all the time and has to converge to sw_data at the end of
+ * playback.
+ */
+ unsigned int sw_io; /* Current software pointer in bytes */
+
+ /* sw_ready: number of bytes ALSA copied to the intermediate buffer, so
+ * it represents the number of bytes which wait for transfer to the HW
+ */
+ int sw_ready; /* Bytes ready to be transferred to/from hw */
+
+ /* appl_ptr: last known position of ALSA (where ALSA is going to write
+ * next time into the intermediate buffer
+ */
+ snd_pcm_uframes_t appl_ptr; /* Last seen appl_ptr */
+
+ unsigned int bytes2hw;
+ int check_alignment;
+
+#ifdef SND_PCM_INDIRECT2_STAT
+ unsigned int zeros2hw;
+ unsigned int mul_elapsed;
+ unsigned int mul_elapsed_real;
+ unsigned long firstbytetime;
+ unsigned long lastbytetime;
+ unsigned long firstzerotime;
+ unsigned int byte_sizes[64];
+ unsigned int zero_sizes[64];
+ unsigned int min_adds[8];
+ unsigned int mul_adds[8];
+ unsigned int zero_times[3750]; /* = 15s */
+ unsigned int zero_times_saved;
+ unsigned int zero_times_notsaved;
+ unsigned int irq_occured;
+ unsigned int pointer_calls;
+ unsigned int lastdifftime;
+#endif
+};
+
+typedef size_t (*snd_pcm_indirect2_copy_t) (struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ size_t bytes);
+typedef size_t (*snd_pcm_indirect2_zero_t) (struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec);
+
+#ifdef SND_PCM_INDIRECT2_STAT
+void snd_pcm_indirect2_stat(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec);
+#endif
+
+snd_pcm_uframes_t
+snd_pcm_indirect2_pointer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec);
+void
+snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ snd_pcm_indirect2_copy_t copy,
+ snd_pcm_indirect2_zero_t zero);
+void
+snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect2 *rec,
+ snd_pcm_indirect2_copy_t copy,
+ snd_pcm_indirect2_zero_t null);
+
+#endif /* __SOUND_PCM_INDIRECT2_H */
diff --git a/sound/drivers/pcsp/Makefile b/sound/drivers/pcsp/Makefile
new file mode 100644
index 0000000..b19555b
--- /dev/null
+++ b/sound/drivers/pcsp/Makefile
@@ -0,0 +1,2 @@
+snd-pcsp-objs := pcsp.o pcsp_lib.o pcsp_mixer.o pcsp_input.o
+obj-$(CONFIG_SND_PCSP) += snd-pcsp.o
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
new file mode 100644
index 0000000..1899cf0
--- /dev/null
+++ b/sound/drivers/pcsp/pcsp.c
@@ -0,0 +1,239 @@
+/*
+ * PC-Speaker driver for Linux
+ *
+ * Copyright (C) 1997-2001 David Woodhouse
+ * Copyright (C) 2001-2008 Stas Sergeev
+ */
+
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <linux/input.h>
+#include <linux/delay.h>
+#include <asm/bitops.h>
+#include "pcsp_input.h"
+#include "pcsp.h"
+
+MODULE_AUTHOR("Stas Sergeev <stsp@users.sourceforge.net>");
+MODULE_DESCRIPTION("PC-Speaker driver");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{PC-Speaker, pcsp}}");
+MODULE_ALIAS("platform:pcspkr");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for pcsp soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for pcsp soundcard.");
+module_param(enable, bool, 0444);
+MODULE_PARM_DESC(enable, "Enable PC-Speaker sound.");
+
+struct snd_pcsp pcsp_chip;
+
+static int __devinit snd_pcsp_create(struct snd_card *card)
+{
+ static struct snd_device_ops ops = { };
+ struct timespec tp;
+ int err;
+ int div, min_div, order;
+
+ hrtimer_get_res(CLOCK_MONOTONIC, &tp);
+ if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) {
+ printk(KERN_ERR "PCSP: Timer resolution is not sufficient "
+ "(%linS)\n", tp.tv_nsec);
+ printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI "
+ "enabled.\n");
+ return -EIO;
+ }
+
+ if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS)
+ min_div = MIN_DIV;
+ else
+ min_div = MAX_DIV;
+#if PCSP_DEBUG
+ printk("PCSP: lpj=%li, min_div=%i, res=%li\n",
+ loops_per_jiffy, min_div, tp.tv_nsec);
+#endif
+
+ div = MAX_DIV / min_div;
+ order = fls(div) - 1;
+
+ pcsp_chip.max_treble = min(order, PCSP_MAX_TREBLE);
+ pcsp_chip.treble = min(pcsp_chip.max_treble, PCSP_DEFAULT_TREBLE);
+ pcsp_chip.playback_ptr = 0;
+ pcsp_chip.period_ptr = 0;
+ atomic_set(&pcsp_chip.timer_active, 0);
+ pcsp_chip.enable = 1;
+ pcsp_chip.pcspkr = 1;
+
+ spin_lock_init(&pcsp_chip.substream_lock);
+
+ pcsp_chip.card = card;
+ pcsp_chip.port = 0x61;
+ pcsp_chip.irq = -1;
+ pcsp_chip.dma = -1;
+
+ /* Register device */
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, &pcsp_chip, &ops);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev)
+{
+ struct snd_card *card;
+ int err;
+
+ if (devnum != 0)
+ return -EINVAL;
+
+ hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ pcsp_chip.timer.cb_mode = HRTIMER_CB_SOFTIRQ;
+ pcsp_chip.timer.function = pcsp_do_timer;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (!card)
+ return -ENOMEM;
+
+ err = snd_pcsp_create(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = snd_pcsp_new_pcm(&pcsp_chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = snd_pcsp_new_mixer(&pcsp_chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_card_set_dev(pcsp_chip.card, dev);
+
+ strcpy(card->driver, "PC-Speaker");
+ strcpy(card->shortname, "pcsp");
+ sprintf(card->longname, "Internal PC-Speaker at port 0x%x",
+ pcsp_chip.port);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ return 0;
+}
+
+static int __devinit alsa_card_pcsp_init(struct device *dev)
+{
+ int err;
+
+ err = snd_card_pcsp_probe(0, dev);
+ if (err) {
+ printk(KERN_ERR "PC-Speaker initialization failed.\n");
+ return err;
+ }
+
+#ifdef CONFIG_DEBUG_PAGEALLOC
+ /* Well, CONFIG_DEBUG_PAGEALLOC makes the sound horrible. Lets alert */
+ printk(KERN_WARNING "PCSP: CONFIG_DEBUG_PAGEALLOC is enabled, "
+ "which may make the sound noisy.\n");
+#endif
+
+ return 0;
+}
+
+static void __devexit alsa_card_pcsp_exit(struct snd_pcsp *chip)
+{
+ snd_card_free(chip->card);
+}
+
+static int __devinit pcsp_probe(struct platform_device *dev)
+{
+ int err;
+
+ err = pcspkr_input_init(&pcsp_chip.input_dev, &dev->dev);
+ if (err < 0)
+ return err;
+
+ err = alsa_card_pcsp_init(&dev->dev);
+ if (err < 0) {
+ pcspkr_input_remove(pcsp_chip.input_dev);
+ return err;
+ }
+
+ platform_set_drvdata(dev, &pcsp_chip);
+ return 0;
+}
+
+static int __devexit pcsp_remove(struct platform_device *dev)
+{
+ struct snd_pcsp *chip = platform_get_drvdata(dev);
+ alsa_card_pcsp_exit(chip);
+ pcspkr_input_remove(chip->input_dev);
+ platform_set_drvdata(dev, NULL);
+ return 0;
+}
+
+static void pcsp_stop_beep(struct snd_pcsp *chip)
+{
+ spin_lock_irq(&chip->substream_lock);
+ if (!chip->playback_substream)
+ pcspkr_stop_sound();
+ spin_unlock_irq(&chip->substream_lock);
+}
+
+#ifdef CONFIG_PM
+static int pcsp_suspend(struct platform_device *dev, pm_message_t state)
+{
+ struct snd_pcsp *chip = platform_get_drvdata(dev);
+ pcsp_stop_beep(chip);
+ snd_pcm_suspend_all(chip->pcm);
+ return 0;
+}
+#else
+#define pcsp_suspend NULL
+#endif /* CONFIG_PM */
+
+static void pcsp_shutdown(struct platform_device *dev)
+{
+ struct snd_pcsp *chip = platform_get_drvdata(dev);
+ pcsp_stop_beep(chip);
+}
+
+static struct platform_driver pcsp_platform_driver = {
+ .driver = {
+ .name = "pcspkr",
+ .owner = THIS_MODULE,
+ },
+ .probe = pcsp_probe,
+ .remove = __devexit_p(pcsp_remove),
+ .suspend = pcsp_suspend,
+ .shutdown = pcsp_shutdown,
+};
+
+static int __init pcsp_init(void)
+{
+ if (!enable)
+ return -ENODEV;
+ return platform_driver_register(&pcsp_platform_driver);
+}
+
+static void __exit pcsp_exit(void)
+{
+ platform_driver_unregister(&pcsp_platform_driver);
+}
+
+module_init(pcsp_init);
+module_exit(pcsp_exit);
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
new file mode 100644
index 0000000..1d661f7
--- /dev/null
+++ b/sound/drivers/pcsp/pcsp.h
@@ -0,0 +1,84 @@
+/*
+ * PC-Speaker driver for Linux
+ *
+ * Copyright (C) 1993-1997 Michael Beck
+ * Copyright (C) 1997-2001 David Woodhouse
+ * Copyright (C) 2001-2008 Stas Sergeev
+ */
+
+#ifndef __PCSP_H__
+#define __PCSP_H__
+
+#include <linux/hrtimer.h>
+#if defined(CONFIG_MIPS) || defined(CONFIG_X86)
+/* Use the global PIT lock ! */
+#include <asm/i8253.h>
+#else
+#include <asm/8253pit.h>
+static DEFINE_SPINLOCK(i8253_lock);
+#endif
+
+#define PCSP_SOUND_VERSION 0x400 /* read 4.00 */
+#define PCSP_DEBUG 0
+
+/* default timer freq for PC-Speaker: 18643 Hz */
+#define DIV_18KHZ 64
+#define MAX_DIV DIV_18KHZ
+#define CALC_DIV(d) (MAX_DIV >> (d))
+#define CUR_DIV() CALC_DIV(chip->treble)
+#define PCSP_MAX_TREBLE 1
+
+/* unfortunately, with hrtimers 37KHz does not work very well :( */
+#define PCSP_DEFAULT_TREBLE 0
+#define MIN_DIV (MAX_DIV >> PCSP_MAX_TREBLE)
+
+/* wild guess */
+#define PCSP_MIN_LPJ 1000000
+#define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1)
+#define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV)
+#define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble))
+#define PCSP_CALC_RATE(i) (PIT_TICK_RATE / CALC_DIV(i))
+#define PCSP_RATE() PCSP_CALC_RATE(chip->treble)
+#define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE
+#define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE
+#define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1)
+#define PCSP_MIN_PERIOD_NS (1000000000ULL * PCSP_MAX_RATE__1)
+#define PCSP_CALC_NS(div) ({ \
+ u64 __val = 1000000000ULL * (div); \
+ do_div(__val, PIT_TICK_RATE); \
+ __val; \
+})
+#define PCSP_PERIOD_NS() PCSP_CALC_NS(CUR_DIV())
+
+#define PCSP_MAX_PERIOD_SIZE (64*1024)
+#define PCSP_MAX_PERIODS 512
+#define PCSP_BUFFER_SIZE (128*1024)
+
+struct snd_pcsp {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct input_dev *input_dev;
+ struct hrtimer timer;
+ unsigned short port, irq, dma;
+ spinlock_t substream_lock;
+ struct snd_pcm_substream *playback_substream;
+ size_t playback_ptr;
+ size_t period_ptr;
+ atomic_t timer_active;
+ int thalf;
+ u64 ns_rem;
+ unsigned char val61;
+ int enable;
+ int max_treble;
+ int treble;
+ int pcspkr;
+};
+
+extern struct snd_pcsp pcsp_chip;
+
+extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle);
+
+extern int snd_pcsp_new_pcm(struct snd_pcsp *chip);
+extern int snd_pcsp_new_mixer(struct snd_pcsp *chip);
+
+#endif
diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c
new file mode 100644
index 0000000..0444cde
--- /dev/null
+++ b/sound/drivers/pcsp/pcsp_input.c
@@ -0,0 +1,116 @@
+/*
+ * PC Speaker beeper driver for Linux
+ *
+ * Copyright (c) 2002 Vojtech Pavlik
+ * Copyright (c) 1992 Orest Zborowski
+ *
+ */
+
+/*
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published by
+ * the Free Software Foundation
+ */
+
+#include <linux/init.h>
+#include <linux/input.h>
+#include <asm/io.h>
+#include "pcsp.h"
+
+static void pcspkr_do_sound(unsigned int count)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&i8253_lock, flags);
+
+ if (count) {
+ /* set command for counter 2, 2 byte write */
+ outb_p(0xB6, 0x43);
+ /* select desired HZ */
+ outb_p(count & 0xff, 0x42);
+ outb((count >> 8) & 0xff, 0x42);
+ /* enable counter 2 */
+ outb_p(inb_p(0x61) | 3, 0x61);
+ } else {
+ /* disable counter 2 */
+ outb(inb_p(0x61) & 0xFC, 0x61);
+ }
+
+ spin_unlock_irqrestore(&i8253_lock, flags);
+}
+
+void pcspkr_stop_sound(void)
+{
+ pcspkr_do_sound(0);
+}
+
+static int pcspkr_input_event(struct input_dev *dev, unsigned int type,
+ unsigned int code, int value)
+{
+ unsigned int count = 0;
+
+ if (atomic_read(&pcsp_chip.timer_active) || !pcsp_chip.pcspkr)
+ return 0;
+
+ switch (type) {
+ case EV_SND:
+ switch (code) {
+ case SND_BELL:
+ if (value)
+ value = 1000;
+ case SND_TONE:
+ break;
+ default:
+ return -1;
+ }
+ break;
+
+ default:
+ return -1;
+ }
+
+ if (value > 20 && value < 32767)
+ count = PIT_TICK_RATE / value;
+
+ pcspkr_do_sound(count);
+
+ return 0;
+}
+
+int __devinit pcspkr_input_init(struct input_dev **rdev, struct device *dev)
+{
+ int err;
+
+ struct input_dev *input_dev = input_allocate_device();
+ if (!input_dev)
+ return -ENOMEM;
+
+ input_dev->name = "PC Speaker";
+ input_dev->phys = "isa0061/input0";
+ input_dev->id.bustype = BUS_ISA;
+ input_dev->id.vendor = 0x001f;
+ input_dev->id.product = 0x0001;
+ input_dev->id.version = 0x0100;
+ input_dev->dev.parent = dev;
+
+ input_dev->evbit[0] = BIT(EV_SND);
+ input_dev->sndbit[0] = BIT(SND_BELL) | BIT(SND_TONE);
+ input_dev->event = pcspkr_input_event;
+
+ err = input_register_device(input_dev);
+ if (err) {
+ input_free_device(input_dev);
+ return err;
+ }
+
+ *rdev = input_dev;
+ return 0;
+}
+
+int pcspkr_input_remove(struct input_dev *dev)
+{
+ pcspkr_stop_sound();
+ input_unregister_device(dev); /* this also does kfree() */
+
+ return 0;
+}
diff --git a/sound/drivers/pcsp/pcsp_input.h b/sound/drivers/pcsp/pcsp_input.h
new file mode 100644
index 0000000..e66738c
--- /dev/null
+++ b/sound/drivers/pcsp/pcsp_input.h
@@ -0,0 +1,14 @@
+/*
+ * PC-Speaker driver for Linux
+ *
+ * Copyright (C) 2001-2008 Stas Sergeev
+ */
+
+#ifndef __PCSP_INPUT_H__
+#define __PCSP_INPUT_H__
+
+int __devinit pcspkr_input_init(struct input_dev **rdev, struct device *dev);
+int pcspkr_input_remove(struct input_dev *dev);
+void pcspkr_stop_sound(void);
+
+#endif
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
new file mode 100644
index 0000000..1f42e40
--- /dev/null
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -0,0 +1,321 @@
+/*
+ * PC-Speaker driver for Linux
+ *
+ * Copyright (C) 1993-1997 Michael Beck
+ * Copyright (C) 1997-2001 David Woodhouse
+ * Copyright (C) 2001-2008 Stas Sergeev
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/pcm.h>
+#include <asm/io.h>
+#include "pcsp.h"
+
+static int nforce_wa;
+module_param(nforce_wa, bool, 0444);
+MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround "
+ "(expect bad sound)");
+
+#define DMIX_WANTS_S16 1
+
+enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+{
+ unsigned char timer_cnt, val;
+ int fmt_size, periods_elapsed;
+ u64 ns;
+ size_t period_bytes, buffer_bytes;
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
+
+ if (chip->thalf) {
+ outb(chip->val61, 0x61);
+ chip->thalf = 0;
+ if (!atomic_read(&chip->timer_active))
+ return HRTIMER_NORESTART;
+ hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
+ ktime_set(0, chip->ns_rem));
+ return HRTIMER_RESTART;
+ }
+
+ spin_lock_irq(&chip->substream_lock);
+ /* Takashi Iwai says regarding this extra lock:
+
+ If the irq handler handles some data on the DMA buffer, it should
+ do snd_pcm_stream_lock().
+ That protects basically against all races among PCM callbacks, yes.
+ However, there are two remaining issues:
+ 1. The substream pointer you try to lock isn't protected _before_
+ this lock yet.
+ 2. snd_pcm_period_elapsed() itself acquires the lock.
+ The requirement of another lock is because of 1. When you get
+ chip->playback_substream, it's not protected.
+ Keeping this lock while snd_pcm_period_elapsed() assures the substream
+ is still protected (at least, not released). And the other status is
+ handled properly inside snd_pcm_stream_lock() in
+ snd_pcm_period_elapsed().
+
+ */
+ if (!chip->playback_substream)
+ goto exit_nr_unlock1;
+ substream = chip->playback_substream;
+ snd_pcm_stream_lock(substream);
+ if (!atomic_read(&chip->timer_active))
+ goto exit_nr_unlock2;
+
+ runtime = substream->runtime;
+ fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3;
+ /* assume it is mono! */
+ val = runtime->dma_area[chip->playback_ptr + fmt_size - 1];
+ if (snd_pcm_format_signed(runtime->format))
+ val ^= 0x80;
+ timer_cnt = val * CUR_DIV() / 256;
+
+ if (timer_cnt && chip->enable) {
+ spin_lock(&i8253_lock);
+ if (!nforce_wa) {
+ outb_p(chip->val61, 0x61);
+ outb_p(timer_cnt, 0x42);
+ outb(chip->val61 ^ 1, 0x61);
+ } else {
+ outb(chip->val61 ^ 2, 0x61);
+ chip->thalf = 1;
+ }
+ spin_unlock(&i8253_lock);
+ }
+
+ period_bytes = snd_pcm_lib_period_bytes(substream);
+ buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+ chip->playback_ptr += PCSP_INDEX_INC() * fmt_size;
+ periods_elapsed = chip->playback_ptr - chip->period_ptr;
+ if (periods_elapsed < 0) {
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: buffer_bytes mod period_bytes != 0 ? "
+ "(%zi %zi %zi)\n",
+ chip->playback_ptr, period_bytes, buffer_bytes);
+#endif
+ periods_elapsed += buffer_bytes;
+ }
+ periods_elapsed /= period_bytes;
+ /* wrap the pointer _before_ calling snd_pcm_period_elapsed(),
+ * or ALSA will BUG on us. */
+ chip->playback_ptr %= buffer_bytes;
+
+ snd_pcm_stream_unlock(substream);
+
+ if (periods_elapsed) {
+ snd_pcm_period_elapsed(substream);
+ chip->period_ptr += periods_elapsed * period_bytes;
+ chip->period_ptr %= buffer_bytes;
+ }
+
+ spin_unlock_irq(&chip->substream_lock);
+
+ if (!atomic_read(&chip->timer_active))
+ return HRTIMER_NORESTART;
+
+ chip->ns_rem = PCSP_PERIOD_NS();
+ ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem);
+ chip->ns_rem -= ns;
+ hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer),
+ ktime_set(0, ns));
+ return HRTIMER_RESTART;
+
+exit_nr_unlock2:
+ snd_pcm_stream_unlock(substream);
+exit_nr_unlock1:
+ spin_unlock_irq(&chip->substream_lock);
+ return HRTIMER_NORESTART;
+}
+
+static void pcsp_start_playing(struct snd_pcsp *chip)
+{
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: start_playing called\n");
+#endif
+ if (atomic_read(&chip->timer_active)) {
+ printk(KERN_ERR "PCSP: Timer already active\n");
+ return;
+ }
+
+ spin_lock(&i8253_lock);
+ chip->val61 = inb(0x61) | 0x03;
+ outb_p(0x92, 0x43); /* binary, mode 1, LSB only, ch 2 */
+ spin_unlock(&i8253_lock);
+ atomic_set(&chip->timer_active, 1);
+ chip->thalf = 0;
+
+ hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
+}
+
+static void pcsp_stop_playing(struct snd_pcsp *chip)
+{
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: stop_playing called\n");
+#endif
+ if (!atomic_read(&chip->timer_active))
+ return;
+
+ atomic_set(&chip->timer_active, 0);
+ spin_lock(&i8253_lock);
+ /* restore the timer */
+ outb_p(0xb6, 0x43); /* binary, mode 3, LSB/MSB, ch 2 */
+ outb(chip->val61 & 0xFC, 0x61);
+ spin_unlock(&i8253_lock);
+}
+
+static int snd_pcsp_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: close called\n");
+#endif
+ if (atomic_read(&chip->timer_active)) {
+ printk(KERN_ERR "PCSP: timer still active\n");
+ pcsp_stop_playing(chip);
+ }
+ spin_lock_irq(&chip->substream_lock);
+ chip->playback_substream = NULL;
+ spin_unlock_irq(&chip->substream_lock);
+ return 0;
+}
+
+static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ int err;
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream)
+{
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: hw_free called\n");
+#endif
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: prepare called, "
+ "size=%zi psize=%zi f=%zi f1=%i\n",
+ snd_pcm_lib_buffer_bytes(substream),
+ snd_pcm_lib_period_bytes(substream),
+ snd_pcm_lib_buffer_bytes(substream) /
+ snd_pcm_lib_period_bytes(substream),
+ substream->runtime->periods);
+#endif
+ chip->playback_ptr = 0;
+ chip->period_ptr = 0;
+ return 0;
+}
+
+static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: trigger called\n");
+#endif
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ pcsp_start_playing(chip);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ pcsp_stop_playing(chip);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_pcsp_playback_pointer(struct snd_pcm_substream
+ *substream)
+{
+ struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+ return bytes_to_frames(substream->runtime, chip->playback_ptr);
+}
+
+static struct snd_pcm_hardware snd_pcsp_playback = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_HALF_DUPLEX |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_U8
+#if DMIX_WANTS_S16
+ | SNDRV_PCM_FMTBIT_S16_LE
+#endif
+ ),
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .rate_min = PCSP_DEFAULT_SRATE,
+ .rate_max = PCSP_DEFAULT_SRATE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = PCSP_BUFFER_SIZE,
+ .period_bytes_min = 64,
+ .period_bytes_max = PCSP_MAX_PERIOD_SIZE,
+ .periods_min = 2,
+ .periods_max = PCSP_MAX_PERIODS,
+ .fifo_size = 0,
+};
+
+static int snd_pcsp_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: open called\n");
+#endif
+ if (atomic_read(&chip->timer_active)) {
+ printk(KERN_ERR "PCSP: still active!!\n");
+ return -EBUSY;
+ }
+ runtime->hw = snd_pcsp_playback;
+ spin_lock_irq(&chip->substream_lock);
+ chip->playback_substream = substream;
+ spin_unlock_irq(&chip->substream_lock);
+ return 0;
+}
+
+static struct snd_pcm_ops snd_pcsp_playback_ops = {
+ .open = snd_pcsp_playback_open,
+ .close = snd_pcsp_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_pcsp_playback_hw_params,
+ .hw_free = snd_pcsp_playback_hw_free,
+ .prepare = snd_pcsp_playback_prepare,
+ .trigger = snd_pcsp_trigger,
+ .pointer = snd_pcsp_playback_pointer,
+};
+
+int __devinit snd_pcsp_new_pcm(struct snd_pcsp *chip)
+{
+ int err;
+
+ err = snd_pcm_new(chip->card, "pcspeaker", 0, 1, 0, &chip->pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(chip->pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_pcsp_playback_ops);
+
+ chip->pcm->private_data = chip;
+ chip->pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
+ strcpy(chip->pcm->name, "pcsp");
+
+ snd_pcm_lib_preallocate_pages_for_all(chip->pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data
+ (GFP_KERNEL), PCSP_BUFFER_SIZE,
+ PCSP_BUFFER_SIZE);
+
+ return 0;
+}
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
new file mode 100644
index 0000000..caeb0f5
--- /dev/null
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -0,0 +1,144 @@
+/*
+ * PC-Speaker driver for Linux
+ *
+ * Mixer implementation.
+ * Copyright (C) 2001-2008 Stas Sergeev
+ */
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include "pcsp.h"
+
+
+static int pcsp_enable_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int pcsp_enable_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = chip->enable;
+ return 0;
+}
+
+static int pcsp_enable_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int enab = ucontrol->value.integer.value[0];
+ if (enab != chip->enable) {
+ chip->enable = enab;
+ changed = 1;
+ }
+ return changed;
+}
+
+static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = chip->max_treble + 1;
+ if (uinfo->value.enumerated.item > chip->max_treble)
+ uinfo->value.enumerated.item = chip->max_treble;
+ sprintf(uinfo->value.enumerated.name, "%d",
+ PCSP_CALC_RATE(uinfo->value.enumerated.item));
+ return 0;
+}
+
+static int pcsp_treble_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = chip->treble;
+ return 0;
+}
+
+static int pcsp_treble_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int treble = ucontrol->value.enumerated.item[0];
+ if (treble != chip->treble) {
+ chip->treble = treble;
+#if PCSP_DEBUG
+ printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE());
+#endif
+ changed = 1;
+ }
+ return changed;
+}
+
+static int pcsp_pcspkr_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int pcsp_pcspkr_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = chip->pcspkr;
+ return 0;
+}
+
+static int pcsp_pcspkr_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcsp *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int spkr = ucontrol->value.integer.value[0];
+ if (spkr != chip->pcspkr) {
+ chip->pcspkr = spkr;
+ changed = 1;
+ }
+ return changed;
+}
+
+#define PCSP_MIXER_CONTROL(ctl_type, ctl_name) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = ctl_name, \
+ .info = pcsp_##ctl_type##_info, \
+ .get = pcsp_##ctl_type##_get, \
+ .put = pcsp_##ctl_type##_put, \
+}
+
+static struct snd_kcontrol_new __devinitdata snd_pcsp_controls[] = {
+ PCSP_MIXER_CONTROL(enable, "Master Playback Switch"),
+ PCSP_MIXER_CONTROL(treble, "BaseFRQ Playback Volume"),
+ PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"),
+};
+
+int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip)
+{
+ struct snd_card *card = chip->card;
+ int i, err;
+
+ for (i = 0; i < ARRAY_SIZE(snd_pcsp_controls); i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(snd_pcsp_controls + i,
+ chip));
+ if (err < 0)
+ return err;
+ }
+
+ strcpy(card->mixername, "PC-Speaker");
+
+ return 0;
+}
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
new file mode 100644
index 0000000..b1c047e
--- /dev/null
+++ b/sound/drivers/portman2x4.c
@@ -0,0 +1,877 @@
+/*
+ * Driver for Midiman Portman2x4 parallel port midi interface
+ *
+ * Copyright (c) by Levent Guendogdu <levon@feature-it.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * ChangeLog
+ * Jan 24 2007 Matthias Koenig <mkoenig@suse.de>
+ * - cleanup and rewrite
+ * Sep 30 2004 Tobias Gehrig <tobias@gehrig.tk>
+ * - source code cleanup
+ * Sep 03 2004 Tobias Gehrig <tobias@gehrig.tk>
+ * - fixed compilation problem with alsa 1.0.6a (removed MODULE_CLASSES,
+ * MODULE_PARM_SYNTAX and changed MODULE_DEVICES to
+ * MODULE_SUPPORTED_DEVICE)
+ * Mar 24 2004 Tobias Gehrig <tobias@gehrig.tk>
+ * - added 2.6 kernel support
+ * Mar 18 2004 Tobias Gehrig <tobias@gehrig.tk>
+ * - added parport_unregister_driver to the startup routine if the driver fails to detect a portman
+ * - added support for all 4 output ports in portman_putmidi
+ * Mar 17 2004 Tobias Gehrig <tobias@gehrig.tk>
+ * - added checks for opened input device in interrupt handler
+ * Feb 20 2004 Tobias Gehrig <tobias@gehrig.tk>
+ * - ported from alsa 0.5 to 1.0
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/parport.h>
+#include <linux/spinlock.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <sound/control.h>
+
+#define CARD_NAME "Portman 2x4"
+#define DRIVER_NAME "portman"
+#define PLATFORM_DRIVER "snd_portman2x4"
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+static struct platform_device *platform_devices[SNDRV_CARDS];
+static int device_count;
+
+module_param_array(index, int, NULL, S_IRUGO);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, S_IRUGO);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, S_IRUGO);
+MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
+
+MODULE_AUTHOR("Levent Guendogdu, Tobias Gehrig, Matthias Koenig");
+MODULE_DESCRIPTION("Midiman Portman2x4");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Midiman,Portman2x4}}");
+
+/*********************************************************************
+ * Chip specific
+ *********************************************************************/
+#define PORTMAN_NUM_INPUT_PORTS 2
+#define PORTMAN_NUM_OUTPUT_PORTS 4
+
+struct portman {
+ spinlock_t reg_lock;
+ struct snd_card *card;
+ struct snd_rawmidi *rmidi;
+ struct pardevice *pardev;
+ int pardev_claimed;
+
+ int open_count;
+ int mode[PORTMAN_NUM_INPUT_PORTS];
+ struct snd_rawmidi_substream *midi_input[PORTMAN_NUM_INPUT_PORTS];
+};
+
+static int portman_free(struct portman *pm)
+{
+ kfree(pm);
+ return 0;
+}
+
+static int __devinit portman_create(struct snd_card *card,
+ struct pardevice *pardev,
+ struct portman **rchip)
+{
+ struct portman *pm;
+
+ *rchip = NULL;
+
+ pm = kzalloc(sizeof(struct portman), GFP_KERNEL);
+ if (pm == NULL)
+ return -ENOMEM;
+
+ /* Init chip specific data */
+ spin_lock_init(&pm->reg_lock);
+ pm->card = card;
+ pm->pardev = pardev;
+
+ *rchip = pm;
+
+ return 0;
+}
+
+/*********************************************************************
+ * HW related constants
+ *********************************************************************/
+
+/* Standard PC parallel port status register equates. */
+#define PP_STAT_BSY 0x80 /* Busy status. Inverted. */
+#define PP_STAT_ACK 0x40 /* Acknowledge. Non-Inverted. */
+#define PP_STAT_POUT 0x20 /* Paper Out. Non-Inverted. */
+#define PP_STAT_SEL 0x10 /* Select. Non-Inverted. */
+#define PP_STAT_ERR 0x08 /* Error. Non-Inverted. */
+
+/* Standard PC parallel port command register equates. */
+#define PP_CMD_IEN 0x10 /* IRQ Enable. Non-Inverted. */
+#define PP_CMD_SELI 0x08 /* Select Input. Inverted. */
+#define PP_CMD_INIT 0x04 /* Init Printer. Non-Inverted. */
+#define PP_CMD_FEED 0x02 /* Auto Feed. Inverted. */
+#define PP_CMD_STB 0x01 /* Strobe. Inverted. */
+
+/* Parallel Port Command Register as implemented by PCP2x4. */
+#define INT_EN PP_CMD_IEN /* Interrupt enable. */
+#define STROBE PP_CMD_STB /* Command strobe. */
+
+/* The parallel port command register field (b1..b3) selects the
+ * various "registers" within the PC/P 2x4. These are the internal
+ * address of these "registers" that must be written to the parallel
+ * port command register.
+ */
+#define RXDATA0 (0 << 1) /* PCP RxData channel 0. */
+#define RXDATA1 (1 << 1) /* PCP RxData channel 1. */
+#define GEN_CTL (2 << 1) /* PCP General Control Register. */
+#define SYNC_CTL (3 << 1) /* PCP Sync Control Register. */
+#define TXDATA0 (4 << 1) /* PCP TxData channel 0. */
+#define TXDATA1 (5 << 1) /* PCP TxData channel 1. */
+#define TXDATA2 (6 << 1) /* PCP TxData channel 2. */
+#define TXDATA3 (7 << 1) /* PCP TxData channel 3. */
+
+/* Parallel Port Status Register as implemented by PCP2x4. */
+#define ESTB PP_STAT_POUT /* Echoed strobe. */
+#define INT_REQ PP_STAT_ACK /* Input data int request. */
+#define BUSY PP_STAT_ERR /* Interface Busy. */
+
+/* Parallel Port Status Register BUSY and SELECT lines are multiplexed
+ * between several functions. Depending on which 2x4 "register" is
+ * currently selected (b1..b3), the BUSY and SELECT lines are
+ * assigned as follows:
+ *
+ * SELECT LINE: A3 A2 A1
+ * --------
+ */
+#define RXAVAIL PP_STAT_SEL /* Rx Available, channel 0. 0 0 0 */
+// RXAVAIL1 PP_STAT_SEL /* Rx Available, channel 1. 0 0 1 */
+#define SYNC_STAT PP_STAT_SEL /* Reserved - Sync Status. 0 1 0 */
+// /* Reserved. 0 1 1 */
+#define TXEMPTY PP_STAT_SEL /* Tx Empty, channel 0. 1 0 0 */
+// TXEMPTY1 PP_STAT_SEL /* Tx Empty, channel 1. 1 0 1 */
+// TXEMPTY2 PP_STAT_SEL /* Tx Empty, channel 2. 1 1 0 */
+// TXEMPTY3 PP_STAT_SEL /* Tx Empty, channel 3. 1 1 1 */
+
+/* BUSY LINE: A3 A2 A1
+ * --------
+ */
+#define RXDATA PP_STAT_BSY /* Rx Input Data, channel 0. 0 0 0 */
+// RXDATA1 PP_STAT_BSY /* Rx Input Data, channel 1. 0 0 1 */
+#define SYNC_DATA PP_STAT_BSY /* Reserved - Sync Data. 0 1 0 */
+ /* Reserved. 0 1 1 */
+#define DATA_ECHO PP_STAT_BSY /* Parallel Port Data Echo. 1 0 0 */
+#define A0_ECHO PP_STAT_BSY /* Address 0 Echo. 1 0 1 */
+#define A1_ECHO PP_STAT_BSY /* Address 1 Echo. 1 1 0 */
+#define A2_ECHO PP_STAT_BSY /* Address 2 Echo. 1 1 1 */
+
+#define PORTMAN2X4_MODE_INPUT_TRIGGERED 0x01
+
+/*********************************************************************
+ * Hardware specific functions
+ *********************************************************************/
+static inline void portman_write_command(struct portman *pm, u8 value)
+{
+ parport_write_control(pm->pardev->port, value);
+}
+
+static inline u8 portman_read_command(struct portman *pm)
+{
+ return parport_read_control(pm->pardev->port);
+}
+
+static inline u8 portman_read_status(struct portman *pm)
+{
+ return parport_read_status(pm->pardev->port);
+}
+
+static inline u8 portman_read_data(struct portman *pm)
+{
+ return parport_read_data(pm->pardev->port);
+}
+
+static inline void portman_write_data(struct portman *pm, u8 value)
+{
+ parport_write_data(pm->pardev->port, value);
+}
+
+static void portman_write_midi(struct portman *pm,
+ int port, u8 mididata)
+{
+ int command = ((port + 4) << 1);
+
+ /* Get entering data byte and port number in BL and BH respectively.
+ * Set up Tx Channel address field for use with PP Cmd Register.
+ * Store address field in BH register.
+ * Inputs: AH = Output port number (0..3).
+ * AL = Data byte.
+ * command = TXDATA0 | INT_EN;
+ * Align port num with address field (b1...b3),
+ * set address for TXDatax, Strobe=0
+ */
+ command |= INT_EN;
+
+ /* Disable interrupts so that the process is not interrupted, then
+ * write the address associated with the current Tx channel to the
+ * PP Command Reg. Do not set the Strobe signal yet.
+ */
+
+ do {
+ portman_write_command(pm, command);
+
+ /* While the address lines settle, write parallel output data to
+ * PP Data Reg. This has no effect until Strobe signal is asserted.
+ */
+
+ portman_write_data(pm, mididata);
+
+ /* If PCP channel's TxEmpty is set (TxEmpty is read through the PP
+ * Status Register), then go write data. Else go back and wait.
+ */
+ } while ((portman_read_status(pm) & TXEMPTY) != TXEMPTY);
+
+ /* TxEmpty is set. Maintain PC/P destination address and assert
+ * Strobe through the PP Command Reg. This will Strobe data into
+ * the PC/P transmitter and set the PC/P BUSY signal.
+ */
+
+ portman_write_command(pm, command | STROBE);
+
+ /* Wait for strobe line to settle and echo back through hardware.
+ * Once it has echoed back, assume that the address and data lines
+ * have settled!
+ */
+
+ while ((portman_read_status(pm) & ESTB) == 0)
+ cpu_relax();
+
+ /* Release strobe and immediately re-allow interrupts. */
+ portman_write_command(pm, command);
+
+ while ((portman_read_status(pm) & ESTB) == ESTB)
+ cpu_relax();
+
+ /* PC/P BUSY is now set. We must wait until BUSY resets itself.
+ * We'll reenable ints while we're waiting.
+ */
+
+ while ((portman_read_status(pm) & BUSY) == BUSY)
+ cpu_relax();
+
+ /* Data sent. */
+}
+
+
+/*
+ * Read MIDI byte from port
+ * Attempt to read input byte from specified hardware input port (0..).
+ * Return -1 if no data
+ */
+static int portman_read_midi(struct portman *pm, int port)
+{
+ unsigned char midi_data = 0;
+ unsigned char cmdout; /* Saved address+IE bit. */
+
+ /* Make sure clocking edge is down before starting... */
+ portman_write_data(pm, 0); /* Make sure edge is down. */
+
+ /* Set destination address to PCP. */
+ cmdout = (port << 1) | INT_EN; /* Address + IE + No Strobe. */
+ portman_write_command(pm, cmdout);
+
+ while ((portman_read_status(pm) & ESTB) == ESTB)
+ cpu_relax(); /* Wait for strobe echo. */
+
+ /* After the address lines settle, check multiplexed RxAvail signal.
+ * If data is available, read it.
+ */
+ if ((portman_read_status(pm) & RXAVAIL) == 0)
+ return -1; /* No data. */
+
+ /* Set the Strobe signal to enable the Rx clocking circuitry. */
+ portman_write_command(pm, cmdout | STROBE); /* Write address+IE+Strobe. */
+
+ while ((portman_read_status(pm) & ESTB) == 0)
+ cpu_relax(); /* Wait for strobe echo. */
+
+ /* The first data bit (msb) is already sitting on the input line. */
+ midi_data = (portman_read_status(pm) & 128);
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+
+ /* Data bit 6. */
+ portman_write_data(pm, 0); /* Cause falling edge while data settles. */
+ midi_data |= (portman_read_status(pm) >> 1) & 64;
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+
+ /* Data bit 5. */
+ portman_write_data(pm, 0); /* Cause falling edge while data settles. */
+ midi_data |= (portman_read_status(pm) >> 2) & 32;
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+
+ /* Data bit 4. */
+ portman_write_data(pm, 0); /* Cause falling edge while data settles. */
+ midi_data |= (portman_read_status(pm) >> 3) & 16;
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+
+ /* Data bit 3. */
+ portman_write_data(pm, 0); /* Cause falling edge while data settles. */
+ midi_data |= (portman_read_status(pm) >> 4) & 8;
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+
+ /* Data bit 2. */
+ portman_write_data(pm, 0); /* Cause falling edge while data settles. */
+ midi_data |= (portman_read_status(pm) >> 5) & 4;
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+
+ /* Data bit 1. */
+ portman_write_data(pm, 0); /* Cause falling edge while data settles. */
+ midi_data |= (portman_read_status(pm) >> 6) & 2;
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+
+ /* Data bit 0. */
+ portman_write_data(pm, 0); /* Cause falling edge while data settles. */
+ midi_data |= (portman_read_status(pm) >> 7) & 1;
+ portman_write_data(pm, 1); /* Cause rising edge, which shifts data. */
+ portman_write_data(pm, 0); /* Return data clock low. */
+
+
+ /* De-assert Strobe and return data. */
+ portman_write_command(pm, cmdout); /* Output saved address+IE. */
+
+ /* Wait for strobe echo. */
+ while ((portman_read_status(pm) & ESTB) == ESTB)
+ cpu_relax();
+
+ return (midi_data & 255); /* Shift back and return value. */
+}
+
+/*
+ * Checks if any input data on the given channel is available
+ * Checks RxAvail
+ */
+static int portman_data_avail(struct portman *pm, int channel)
+{
+ int command = INT_EN;
+ switch (channel) {
+ case 0:
+ command |= RXDATA0;
+ break;
+ case 1:
+ command |= RXDATA1;
+ break;
+ }
+ /* Write hardware (assumme STROBE=0) */
+ portman_write_command(pm, command);
+ /* Check multiplexed RxAvail signal */
+ if ((portman_read_status(pm) & RXAVAIL) == RXAVAIL)
+ return 1; /* Data available */
+
+ /* No Data available */
+ return 0;
+}
+
+
+/*
+ * Flushes any input
+ */
+static void portman_flush_input(struct portman *pm, unsigned char port)
+{
+ /* Local variable for counting things */
+ unsigned int i = 0;
+ unsigned char command = 0;
+
+ switch (port) {
+ case 0:
+ command = RXDATA0;
+ break;
+ case 1:
+ command = RXDATA1;
+ break;
+ default:
+ snd_printk(KERN_WARNING
+ "portman_flush_input() Won't flush port %i\n",
+ port);
+ return;
+ }
+
+ /* Set address for specified channel in port and allow to settle. */
+ portman_write_command(pm, command);
+
+ /* Assert the Strobe and wait for echo back. */
+ portman_write_command(pm, command | STROBE);
+
+ /* Wait for ESTB */
+ while ((portman_read_status(pm) & ESTB) == 0)
+ cpu_relax();
+
+ /* Output clock cycles to the Rx circuitry. */
+ portman_write_data(pm, 0);
+
+ /* Flush 250 bits... */
+ for (i = 0; i < 250; i++) {
+ portman_write_data(pm, 1);
+ portman_write_data(pm, 0);
+ }
+
+ /* Deassert the Strobe signal of the port and wait for it to settle. */
+ portman_write_command(pm, command | INT_EN);
+
+ /* Wait for settling */
+ while ((portman_read_status(pm) & ESTB) == ESTB)
+ cpu_relax();
+}
+
+static int portman_probe(struct parport *p)
+{
+ /* Initialize the parallel port data register. Will set Rx clocks
+ * low in case we happen to be addressing the Rx ports at this time.
+ */
+ /* 1 */
+ parport_write_data(p, 0);
+
+ /* Initialize the parallel port command register, thus initializing
+ * hardware handshake lines to midi box:
+ *
+ * Strobe = 0
+ * Interrupt Enable = 0
+ */
+ /* 2 */
+ parport_write_control(p, 0);
+
+ /* Check if Portman PC/P 2x4 is out there. */
+ /* 3 */
+ parport_write_control(p, RXDATA0); /* Write Strobe=0 to command reg. */
+
+ /* Check for ESTB to be clear */
+ /* 4 */
+ if ((parport_read_status(p) & ESTB) == ESTB)
+ return 1; /* CODE 1 - Strobe Failure. */
+
+ /* Set for RXDATA0 where no damage will be done. */
+ /* 5 */
+ parport_write_control(p, RXDATA0 + STROBE); /* Write Strobe=1 to command reg. */
+
+ /* 6 */
+ if ((parport_read_status(p) & ESTB) != ESTB)
+ return 1; /* CODE 1 - Strobe Failure. */
+
+ /* 7 */
+ parport_write_control(p, 0); /* Reset Strobe=0. */
+
+ /* Check if Tx circuitry is functioning properly. If initialized
+ * unit TxEmpty is false, send out char and see if if goes true.
+ */
+ /* 8 */
+ parport_write_control(p, TXDATA0); /* Tx channel 0, strobe off. */
+
+ /* If PCP channel's TxEmpty is set (TxEmpty is read through the PP
+ * Status Register), then go write data. Else go back and wait.
+ */
+ /* 9 */
+ if ((parport_read_status(p) & TXEMPTY) == 0)
+ return 2;
+
+ /* Return OK status. */
+ return 0;
+}
+
+static int portman_device_init(struct portman *pm)
+{
+ portman_flush_input(pm, 0);
+ portman_flush_input(pm, 1);
+
+ return 0;
+}
+
+/*********************************************************************
+ * Rawmidi
+ *********************************************************************/
+static int snd_portman_midi_open(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static int snd_portman_midi_close(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static void snd_portman_midi_input_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct portman *pm = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&pm->reg_lock, flags);
+ if (up)
+ pm->mode[substream->number] |= PORTMAN2X4_MODE_INPUT_TRIGGERED;
+ else
+ pm->mode[substream->number] &= ~PORTMAN2X4_MODE_INPUT_TRIGGERED;
+ spin_unlock_irqrestore(&pm->reg_lock, flags);
+}
+
+static void snd_portman_midi_output_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct portman *pm = substream->rmidi->private_data;
+ unsigned long flags;
+ unsigned char byte;
+
+ spin_lock_irqsave(&pm->reg_lock, flags);
+ if (up) {
+ while ((snd_rawmidi_transmit(substream, &byte, 1) == 1))
+ portman_write_midi(pm, substream->number, byte);
+ }
+ spin_unlock_irqrestore(&pm->reg_lock, flags);
+}
+
+static struct snd_rawmidi_ops snd_portman_midi_output = {
+ .open = snd_portman_midi_open,
+ .close = snd_portman_midi_close,
+ .trigger = snd_portman_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_portman_midi_input = {
+ .open = snd_portman_midi_open,
+ .close = snd_portman_midi_close,
+ .trigger = snd_portman_midi_input_trigger,
+};
+
+/* Create and initialize the rawmidi component */
+static int __devinit snd_portman_rawmidi_create(struct snd_card *card)
+{
+ struct portman *pm = card->private_data;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream;
+ int err;
+
+ err = snd_rawmidi_new(card, CARD_NAME, 0,
+ PORTMAN_NUM_OUTPUT_PORTS,
+ PORTMAN_NUM_INPUT_PORTS,
+ &rmidi);
+ if (err < 0)
+ return err;
+
+ rmidi->private_data = pm;
+ strcpy(rmidi->name, CARD_NAME);
+ rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ pm->rmidi = rmidi;
+
+ /* register rawmidi ops */
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_portman_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_portman_midi_input);
+
+ /* name substreams */
+ /* output */
+ list_for_each_entry(substream,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams,
+ list) {
+ sprintf(substream->name,
+ "Portman2x4 %d", substream->number+1);
+ }
+ /* input */
+ list_for_each_entry(substream,
+ &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams,
+ list) {
+ pm->midi_input[substream->number] = substream;
+ sprintf(substream->name,
+ "Portman2x4 %d", substream->number+1);
+ }
+
+ return err;
+}
+
+/*********************************************************************
+ * parport stuff
+ *********************************************************************/
+static void snd_portman_interrupt(void *userdata)
+{
+ unsigned char midivalue = 0;
+ struct portman *pm = ((struct snd_card*)userdata)->private_data;
+
+ spin_lock(&pm->reg_lock);
+
+ /* While any input data is waiting */
+ while ((portman_read_status(pm) & INT_REQ) == INT_REQ) {
+ /* If data available on channel 0,
+ read it and stuff it into the queue. */
+ if (portman_data_avail(pm, 0)) {
+ /* Read Midi */
+ midivalue = portman_read_midi(pm, 0);
+ /* put midi into queue... */
+ if (pm->mode[0] & PORTMAN2X4_MODE_INPUT_TRIGGERED)
+ snd_rawmidi_receive(pm->midi_input[0],
+ &midivalue, 1);
+
+ }
+ /* If data available on channel 1,
+ read it and stuff it into the queue. */
+ if (portman_data_avail(pm, 1)) {
+ /* Read Midi */
+ midivalue = portman_read_midi(pm, 1);
+ /* put midi into queue... */
+ if (pm->mode[1] & PORTMAN2X4_MODE_INPUT_TRIGGERED)
+ snd_rawmidi_receive(pm->midi_input[1],
+ &midivalue, 1);
+ }
+
+ }
+
+ spin_unlock(&pm->reg_lock);
+}
+
+static int __devinit snd_portman_probe_port(struct parport *p)
+{
+ struct pardevice *pardev;
+ int res;
+
+ pardev = parport_register_device(p, DRIVER_NAME,
+ NULL, NULL, NULL,
+ 0, NULL);
+ if (!pardev)
+ return -EIO;
+
+ if (parport_claim(pardev)) {
+ parport_unregister_device(pardev);
+ return -EIO;
+ }
+
+ res = portman_probe(p);
+
+ parport_release(pardev);
+ parport_unregister_device(pardev);
+
+ return res ? -EIO : 0;
+}
+
+static void __devinit snd_portman_attach(struct parport *p)
+{
+ struct platform_device *device;
+
+ device = platform_device_alloc(PLATFORM_DRIVER, device_count);
+ if (!device)
+ return;
+
+ /* Temporary assignment to forward the parport */
+ platform_set_drvdata(device, p);
+
+ if (platform_device_add(device) < 0) {
+ platform_device_put(device);
+ return;
+ }
+
+ /* Since we dont get the return value of probe
+ * We need to check if device probing succeeded or not */
+ if (!platform_get_drvdata(device)) {
+ platform_device_unregister(device);
+ return;
+ }
+
+ /* register device in global table */
+ platform_devices[device_count] = device;
+ device_count++;
+}
+
+static void snd_portman_detach(struct parport *p)
+{
+ /* nothing to do here */
+}
+
+static struct parport_driver portman_parport_driver = {
+ .name = "portman2x4",
+ .attach = snd_portman_attach,
+ .detach = snd_portman_detach
+};
+
+/*********************************************************************
+ * platform stuff
+ *********************************************************************/
+static void snd_portman_card_private_free(struct snd_card *card)
+{
+ struct portman *pm = card->private_data;
+ struct pardevice *pardev = pm->pardev;
+
+ if (pardev) {
+ if (pm->pardev_claimed)
+ parport_release(pardev);
+ parport_unregister_device(pardev);
+ }
+
+ portman_free(pm);
+}
+
+static int __devinit snd_portman_probe(struct platform_device *pdev)
+{
+ struct pardevice *pardev;
+ struct parport *p;
+ int dev = pdev->id;
+ struct snd_card *card = NULL;
+ struct portman *pm = NULL;
+ int err;
+
+ p = platform_get_drvdata(pdev);
+ platform_set_drvdata(pdev, NULL);
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev])
+ return -ENOENT;
+
+ if ((err = snd_portman_probe_port(p)) < 0)
+ return err;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL) {
+ snd_printd("Cannot create card\n");
+ return -ENOMEM;
+ }
+ strcpy(card->driver, DRIVER_NAME);
+ strcpy(card->shortname, CARD_NAME);
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, p->base, p->irq);
+
+ pardev = parport_register_device(p, /* port */
+ DRIVER_NAME, /* name */
+ NULL, /* preempt */
+ NULL, /* wakeup */
+ snd_portman_interrupt, /* ISR */
+ PARPORT_DEV_EXCL, /* flags */
+ (void *)card); /* private */
+ if (pardev == NULL) {
+ snd_printd("Cannot register pardevice\n");
+ err = -EIO;
+ goto __err;
+ }
+
+ if ((err = portman_create(card, pardev, &pm)) < 0) {
+ snd_printd("Cannot create main component\n");
+ parport_unregister_device(pardev);
+ goto __err;
+ }
+ card->private_data = pm;
+ card->private_free = snd_portman_card_private_free;
+
+ if ((err = snd_portman_rawmidi_create(card)) < 0) {
+ snd_printd("Creating Rawmidi component failed\n");
+ goto __err;
+ }
+
+ /* claim parport */
+ if (parport_claim(pardev)) {
+ snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base);
+ err = -EIO;
+ goto __err;
+ }
+ pm->pardev_claimed = 1;
+
+ /* init device */
+ if ((err = portman_device_init(pm)) < 0)
+ goto __err;
+
+ platform_set_drvdata(pdev, card);
+
+ snd_card_set_dev(card, &pdev->dev);
+
+ /* At this point card will be usable */
+ if ((err = snd_card_register(card)) < 0) {
+ snd_printd("Cannot register card\n");
+ goto __err;
+ }
+
+ snd_printk(KERN_INFO "Portman 2x4 on 0x%lx\n", p->base);
+ return 0;
+
+__err:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_portman_remove(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ if (card)
+ snd_card_free(card);
+
+ return 0;
+}
+
+
+static struct platform_driver snd_portman_driver = {
+ .probe = snd_portman_probe,
+ .remove = __devexit_p(snd_portman_remove),
+ .driver = {
+ .name = PLATFORM_DRIVER
+ }
+};
+
+/*********************************************************************
+ * module init stuff
+ *********************************************************************/
+static void snd_portman_unregister_all(void)
+{
+ int i;
+
+ for (i = 0; i < SNDRV_CARDS; ++i) {
+ if (platform_devices[i]) {
+ platform_device_unregister(platform_devices[i]);
+ platform_devices[i] = NULL;
+ }
+ }
+ platform_driver_unregister(&snd_portman_driver);
+ parport_unregister_driver(&portman_parport_driver);
+}
+
+static int __init snd_portman_module_init(void)
+{
+ int err;
+
+ if ((err = platform_driver_register(&snd_portman_driver)) < 0)
+ return err;
+
+ if (parport_register_driver(&portman_parport_driver) != 0) {
+ platform_driver_unregister(&snd_portman_driver);
+ return -EIO;
+ }
+
+ if (device_count == 0) {
+ snd_portman_unregister_all();
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static void __exit snd_portman_module_exit(void)
+{
+ snd_portman_unregister_all();
+}
+
+module_init(snd_portman_module_init);
+module_exit(snd_portman_module_exit);
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
new file mode 100644
index 0000000..d8aab9d
--- /dev/null
+++ b/sound/drivers/serial-u16550.c
@@ -0,0 +1,1049 @@
+/*
+ * serial.c
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Isaku Yamahata <yamahata@private.email.ne.jp>,
+ * George Hansper <ghansper@apana.org.au>,
+ * Hannu Savolainen
+ *
+ * This code is based on the code from ALSA 0.5.9, but heavily rewritten.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * Sat Mar 31 17:27:57 PST 2001 tim.mann@compaq.com
+ * Added support for the Midiator MS-124T and for the MS-124W in
+ * Single Addressed (S/A) or Multiple Burst (M/B) mode, with
+ * power derived either parasitically from the serial port or
+ * from a separate power supply.
+ *
+ * More documentation can be found in serial-u16550.txt.
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+#include <sound/initval.h>
+
+#include <linux/serial_reg.h>
+#include <linux/jiffies.h>
+
+#include <asm/io.h>
+
+MODULE_DESCRIPTION("MIDI serial u16550");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ALSA, MIDI serial u16550}}");
+
+#define SNDRV_SERIAL_SOUNDCANVAS 0 /* Roland Soundcanvas; F5 NN selects part */
+#define SNDRV_SERIAL_MS124T 1 /* Midiator MS-124T */
+#define SNDRV_SERIAL_MS124W_SA 2 /* Midiator MS-124W in S/A mode */
+#define SNDRV_SERIAL_MS124W_MB 3 /* Midiator MS-124W in M/B mode */
+#define SNDRV_SERIAL_GENERIC 4 /* Generic Interface */
+#define SNDRV_SERIAL_MAX_ADAPTOR SNDRV_SERIAL_GENERIC
+static char *adaptor_names[] = {
+ "Soundcanvas",
+ "MS-124T",
+ "MS-124W S/A",
+ "MS-124W M/B",
+ "Generic"
+};
+
+#define SNDRV_SERIAL_NORMALBUFF 0 /* Normal blocking buffer operation */
+#define SNDRV_SERIAL_DROPBUFF 1 /* Non-blocking discard operation */
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x3f8,0x2f8,0x3e8,0x2e8 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 3,4,5,7,9,10,11,14,15 */
+static int speed[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 38400}; /* 9600,19200,38400,57600,115200 */
+static int base[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 115200}; /* baud base */
+static int outs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; /* 1 to 16 */
+static int ins[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; /* 1 to 16 */
+static int adaptor[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = SNDRV_SERIAL_SOUNDCANVAS};
+static int droponfull[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS -1)] = SNDRV_SERIAL_NORMALBUFF };
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Serial MIDI.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Serial MIDI.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable UART16550A chip.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for UART16550A chip.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for UART16550A chip.");
+module_param_array(speed, int, NULL, 0444);
+MODULE_PARM_DESC(speed, "Speed in bauds.");
+module_param_array(base, int, NULL, 0444);
+MODULE_PARM_DESC(base, "Base for divisor in bauds.");
+module_param_array(outs, int, NULL, 0444);
+MODULE_PARM_DESC(outs, "Number of MIDI outputs.");
+module_param_array(ins, int, NULL, 0444);
+MODULE_PARM_DESC(ins, "Number of MIDI inputs.");
+module_param_array(droponfull, bool, NULL, 0444);
+MODULE_PARM_DESC(droponfull, "Flag to enable drop-on-full buffer mode");
+
+module_param_array(adaptor, int, NULL, 0444);
+MODULE_PARM_DESC(adaptor, "Type of adaptor.");
+
+/*#define SNDRV_SERIAL_MS124W_MB_NOCOMBO 1*/ /* Address outs as 0-3 instead of bitmap */
+
+#define SNDRV_SERIAL_MAX_OUTS 16 /* max 64, min 16 */
+#define SNDRV_SERIAL_MAX_INS 16 /* max 64, min 16 */
+
+#define TX_BUFF_SIZE (1<<15) /* Must be 2^n */
+#define TX_BUFF_MASK (TX_BUFF_SIZE - 1)
+
+#define SERIAL_MODE_NOT_OPENED (0)
+#define SERIAL_MODE_INPUT_OPEN (1 << 0)
+#define SERIAL_MODE_OUTPUT_OPEN (1 << 1)
+#define SERIAL_MODE_INPUT_TRIGGERED (1 << 2)
+#define SERIAL_MODE_OUTPUT_TRIGGERED (1 << 3)
+
+struct snd_uart16550 {
+ struct snd_card *card;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *midi_output[SNDRV_SERIAL_MAX_OUTS];
+ struct snd_rawmidi_substream *midi_input[SNDRV_SERIAL_MAX_INS];
+
+ int filemode; /* open status of file */
+
+ spinlock_t open_lock;
+
+ int irq;
+
+ unsigned long base;
+ struct resource *res_base;
+
+ unsigned int speed;
+ unsigned int speed_base;
+ unsigned char divisor;
+
+ unsigned char old_divisor_lsb;
+ unsigned char old_divisor_msb;
+ unsigned char old_line_ctrl_reg;
+
+ /* parameter for using of write loop */
+ short int fifo_limit; /* used in uart16550 */
+ short int fifo_count; /* used in uart16550 */
+
+ /* type of adaptor */
+ int adaptor;
+
+ /* inputs */
+ int prev_in;
+ unsigned char rstatus;
+
+ /* outputs */
+ int prev_out;
+ unsigned char prev_status[SNDRV_SERIAL_MAX_OUTS];
+
+ /* write buffer and its writing/reading position */
+ unsigned char tx_buff[TX_BUFF_SIZE];
+ int buff_in_count;
+ int buff_in;
+ int buff_out;
+ int drop_on_full;
+
+ /* wait timer */
+ unsigned int timer_running:1;
+ struct timer_list buffer_timer;
+
+};
+
+static struct platform_device *devices[SNDRV_CARDS];
+
+static inline void snd_uart16550_add_timer(struct snd_uart16550 *uart)
+{
+ if (!uart->timer_running) {
+ /* timer 38600bps * 10bit * 16byte */
+ uart->buffer_timer.expires = jiffies + (HZ+255)/256;
+ uart->timer_running = 1;
+ add_timer(&uart->buffer_timer);
+ }
+}
+
+static inline void snd_uart16550_del_timer(struct snd_uart16550 *uart)
+{
+ if (uart->timer_running) {
+ del_timer(&uart->buffer_timer);
+ uart->timer_running = 0;
+ }
+}
+
+/* This macro is only used in snd_uart16550_io_loop */
+static inline void snd_uart16550_buffer_output(struct snd_uart16550 *uart)
+{
+ unsigned short buff_out = uart->buff_out;
+ if (uart->buff_in_count > 0) {
+ outb(uart->tx_buff[buff_out], uart->base + UART_TX);
+ uart->fifo_count++;
+ buff_out++;
+ buff_out &= TX_BUFF_MASK;
+ uart->buff_out = buff_out;
+ uart->buff_in_count--;
+ }
+}
+
+/* This loop should be called with interrupts disabled
+ * We don't want to interrupt this,
+ * as we're already handling an interrupt
+ */
+static void snd_uart16550_io_loop(struct snd_uart16550 * uart)
+{
+ unsigned char c, status;
+ int substream;
+
+ /* recall previous stream */
+ substream = uart->prev_in;
+
+ /* Read Loop */
+ while ((status = inb(uart->base + UART_LSR)) & UART_LSR_DR) {
+ /* while receive data ready */
+ c = inb(uart->base + UART_RX);
+
+ /* keep track of last status byte */
+ if (c & 0x80)
+ uart->rstatus = c;
+
+ /* handle stream switch */
+ if (uart->adaptor == SNDRV_SERIAL_GENERIC) {
+ if (uart->rstatus == 0xf5) {
+ if (c <= SNDRV_SERIAL_MAX_INS && c > 0)
+ substream = c - 1;
+ if (c != 0xf5)
+ /* prevent future bytes from being
+ interpreted as streams */
+ uart->rstatus = 0;
+ } else if ((uart->filemode & SERIAL_MODE_INPUT_OPEN)
+ && uart->midi_input[substream])
+ snd_rawmidi_receive(uart->midi_input[substream],
+ &c, 1);
+ } else if ((uart->filemode & SERIAL_MODE_INPUT_OPEN) &&
+ uart->midi_input[substream])
+ snd_rawmidi_receive(uart->midi_input[substream], &c, 1);
+
+ if (status & UART_LSR_OE)
+ snd_printk("%s: Overrun on device at 0x%lx\n",
+ uart->rmidi->name, uart->base);
+ }
+
+ /* remember the last stream */
+ uart->prev_in = substream;
+
+ /* no need of check SERIAL_MODE_OUTPUT_OPEN because if not,
+ buffer is never filled. */
+ /* Check write status */
+ if (status & UART_LSR_THRE)
+ uart->fifo_count = 0;
+ if (uart->adaptor == SNDRV_SERIAL_MS124W_SA
+ || uart->adaptor == SNDRV_SERIAL_GENERIC) {
+ /* Can't use FIFO, must send only when CTS is true */
+ status = inb(uart->base + UART_MSR);
+ while (uart->fifo_count == 0 && (status & UART_MSR_CTS) &&
+ uart->buff_in_count > 0) {
+ snd_uart16550_buffer_output(uart);
+ status = inb(uart->base + UART_MSR);
+ }
+ } else {
+ /* Write loop */
+ while (uart->fifo_count < uart->fifo_limit /* Can we write ? */
+ && uart->buff_in_count > 0) /* Do we want to? */
+ snd_uart16550_buffer_output(uart);
+ }
+ if (uart->irq < 0 && uart->buff_in_count > 0)
+ snd_uart16550_add_timer(uart);
+}
+
+/* NOTES ON SERVICING INTERUPTS
+ * ---------------------------
+ * After receiving a interrupt, it is important to indicate to the UART that
+ * this has been done.
+ * For a Rx interrupt, this is done by reading the received byte.
+ * For a Tx interrupt this is done by either:
+ * a) Writing a byte
+ * b) Reading the IIR
+ * It is particularly important to read the IIR if a Tx interrupt is received
+ * when there is no data in tx_buff[], as in this case there no other
+ * indication that the interrupt has been serviced, and it remains outstanding
+ * indefinitely. This has the curious side effect that and no further interrupts
+ * will be generated from this device AT ALL!!.
+ * It is also desirable to clear outstanding interrupts when the device is
+ * opened/closed.
+ *
+ *
+ * Note that some devices need OUT2 to be set before they will generate
+ * interrupts at all. (Possibly tied to an internal pull-up on CTS?)
+ */
+static irqreturn_t snd_uart16550_interrupt(int irq, void *dev_id)
+{
+ struct snd_uart16550 *uart;
+
+ uart = dev_id;
+ spin_lock(&uart->open_lock);
+ if (uart->filemode == SERIAL_MODE_NOT_OPENED) {
+ spin_unlock(&uart->open_lock);
+ return IRQ_NONE;
+ }
+ /* indicate to the UART that the interrupt has been serviced */
+ inb(uart->base + UART_IIR);
+ snd_uart16550_io_loop(uart);
+ spin_unlock(&uart->open_lock);
+ return IRQ_HANDLED;
+}
+
+/* When the polling mode, this function calls snd_uart16550_io_loop. */
+static void snd_uart16550_buffer_timer(unsigned long data)
+{
+ unsigned long flags;
+ struct snd_uart16550 *uart;
+
+ uart = (struct snd_uart16550 *)data;
+ spin_lock_irqsave(&uart->open_lock, flags);
+ snd_uart16550_del_timer(uart);
+ snd_uart16550_io_loop(uart);
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+}
+
+/*
+ * this method probes, if an uart sits on given port
+ * return 0 if found
+ * return negative error if not found
+ */
+static int __devinit snd_uart16550_detect(struct snd_uart16550 *uart)
+{
+ unsigned long io_base = uart->base;
+ int ok;
+ unsigned char c;
+
+ /* Do some vague tests for the presence of the uart */
+ if (io_base == 0 || io_base == SNDRV_AUTO_PORT) {
+ return -ENODEV; /* Not configured */
+ }
+
+ uart->res_base = request_region(io_base, 8, "Serial MIDI");
+ if (uart->res_base == NULL) {
+ snd_printk(KERN_ERR "u16550: can't grab port 0x%lx\n", io_base);
+ return -EBUSY;
+ }
+
+ /* uart detected unless one of the following tests should fail */
+ ok = 1;
+ /* 8 data-bits, 1 stop-bit, parity off, DLAB = 0 */
+ outb(UART_LCR_WLEN8, io_base + UART_LCR); /* Line Control Register */
+ c = inb(io_base + UART_IER);
+ /* The top four bits of the IER should always == 0 */
+ if ((c & 0xf0) != 0)
+ ok = 0; /* failed */
+
+ outb(0xaa, io_base + UART_SCR);
+ /* Write arbitrary data into the scratch reg */
+ c = inb(io_base + UART_SCR);
+ /* If it comes back, it's OK */
+ if (c != 0xaa)
+ ok = 0; /* failed */
+
+ outb(0x55, io_base + UART_SCR);
+ /* Write arbitrary data into the scratch reg */
+ c = inb(io_base + UART_SCR);
+ /* If it comes back, it's OK */
+ if (c != 0x55)
+ ok = 0; /* failed */
+
+ return ok;
+}
+
+static void snd_uart16550_do_open(struct snd_uart16550 * uart)
+{
+ char byte;
+
+ /* Initialize basic variables */
+ uart->buff_in_count = 0;
+ uart->buff_in = 0;
+ uart->buff_out = 0;
+ uart->fifo_limit = 1;
+ uart->fifo_count = 0;
+ uart->timer_running = 0;
+
+ outb(UART_FCR_ENABLE_FIFO /* Enable FIFO's (if available) */
+ | UART_FCR_CLEAR_RCVR /* Clear receiver FIFO */
+ | UART_FCR_CLEAR_XMIT /* Clear transmitter FIFO */
+ | UART_FCR_TRIGGER_4 /* Set FIFO trigger at 4-bytes */
+ /* NOTE: interrupt generated after T=(time)4-bytes
+ * if less than UART_FCR_TRIGGER bytes received
+ */
+ ,uart->base + UART_FCR); /* FIFO Control Register */
+
+ if ((inb(uart->base + UART_IIR) & 0xf0) == 0xc0)
+ uart->fifo_limit = 16;
+ if (uart->divisor != 0) {
+ uart->old_line_ctrl_reg = inb(uart->base + UART_LCR);
+ outb(UART_LCR_DLAB /* Divisor latch access bit */
+ ,uart->base + UART_LCR); /* Line Control Register */
+ uart->old_divisor_lsb = inb(uart->base + UART_DLL);
+ uart->old_divisor_msb = inb(uart->base + UART_DLM);
+
+ outb(uart->divisor
+ ,uart->base + UART_DLL); /* Divisor Latch Low */
+ outb(0
+ ,uart->base + UART_DLM); /* Divisor Latch High */
+ /* DLAB is reset to 0 in next outb() */
+ }
+ /* Set serial parameters (parity off, etc) */
+ outb(UART_LCR_WLEN8 /* 8 data-bits */
+ | 0 /* 1 stop-bit */
+ | 0 /* parity off */
+ | 0 /* DLAB = 0 */
+ ,uart->base + UART_LCR); /* Line Control Register */
+
+ switch (uart->adaptor) {
+ default:
+ outb(UART_MCR_RTS /* Set Request-To-Send line active */
+ | UART_MCR_DTR /* Set Data-Terminal-Ready line active */
+ | UART_MCR_OUT2 /* Set OUT2 - not always required, but when
+ * it is, it is ESSENTIAL for enabling interrupts
+ */
+ ,uart->base + UART_MCR); /* Modem Control Register */
+ break;
+ case SNDRV_SERIAL_MS124W_SA:
+ case SNDRV_SERIAL_MS124W_MB:
+ /* MS-124W can draw power from RTS and DTR if they
+ are in opposite states. */
+ outb(UART_MCR_RTS | (0&UART_MCR_DTR) | UART_MCR_OUT2,
+ uart->base + UART_MCR);
+ break;
+ case SNDRV_SERIAL_MS124T:
+ /* MS-124T can draw power from RTS and/or DTR (preferably
+ both) if they are both asserted. */
+ outb(UART_MCR_RTS | UART_MCR_DTR | UART_MCR_OUT2,
+ uart->base + UART_MCR);
+ break;
+ }
+
+ if (uart->irq < 0) {
+ byte = (0 & UART_IER_RDI) /* Disable Receiver data interrupt */
+ |(0 & UART_IER_THRI) /* Disable Transmitter holding register empty interrupt */
+ ;
+ } else if (uart->adaptor == SNDRV_SERIAL_MS124W_SA) {
+ byte = UART_IER_RDI /* Enable Receiver data interrupt */
+ | UART_IER_MSI /* Enable Modem status interrupt */
+ ;
+ } else if (uart->adaptor == SNDRV_SERIAL_GENERIC) {
+ byte = UART_IER_RDI /* Enable Receiver data interrupt */
+ | UART_IER_MSI /* Enable Modem status interrupt */
+ | UART_IER_THRI /* Enable Transmitter holding register empty interrupt */
+ ;
+ } else {
+ byte = UART_IER_RDI /* Enable Receiver data interrupt */
+ | UART_IER_THRI /* Enable Transmitter holding register empty interrupt */
+ ;
+ }
+ outb(byte, uart->base + UART_IER); /* Interrupt enable Register */
+
+ inb(uart->base + UART_LSR); /* Clear any pre-existing overrun indication */
+ inb(uart->base + UART_IIR); /* Clear any pre-existing transmit interrupt */
+ inb(uart->base + UART_RX); /* Clear any pre-existing receive interrupt */
+}
+
+static void snd_uart16550_do_close(struct snd_uart16550 * uart)
+{
+ if (uart->irq < 0)
+ snd_uart16550_del_timer(uart);
+
+ /* NOTE: may need to disable interrupts before de-registering out handler.
+ * For now, the consequences are harmless.
+ */
+
+ outb((0 & UART_IER_RDI) /* Disable Receiver data interrupt */
+ |(0 & UART_IER_THRI) /* Disable Transmitter holding register empty interrupt */
+ ,uart->base + UART_IER); /* Interrupt enable Register */
+
+ switch (uart->adaptor) {
+ default:
+ outb((0 & UART_MCR_RTS) /* Deactivate Request-To-Send line */
+ |(0 & UART_MCR_DTR) /* Deactivate Data-Terminal-Ready line */
+ |(0 & UART_MCR_OUT2) /* Deactivate OUT2 */
+ ,uart->base + UART_MCR); /* Modem Control Register */
+ break;
+ case SNDRV_SERIAL_MS124W_SA:
+ case SNDRV_SERIAL_MS124W_MB:
+ /* MS-124W can draw power from RTS and DTR if they
+ are in opposite states; leave it powered. */
+ outb(UART_MCR_RTS | (0&UART_MCR_DTR) | (0&UART_MCR_OUT2),
+ uart->base + UART_MCR);
+ break;
+ case SNDRV_SERIAL_MS124T:
+ /* MS-124T can draw power from RTS and/or DTR (preferably
+ both) if they are both asserted; leave it powered. */
+ outb(UART_MCR_RTS | UART_MCR_DTR | (0&UART_MCR_OUT2),
+ uart->base + UART_MCR);
+ break;
+ }
+
+ inb(uart->base + UART_IIR); /* Clear any outstanding interrupts */
+
+ /* Restore old divisor */
+ if (uart->divisor != 0) {
+ outb(UART_LCR_DLAB /* Divisor latch access bit */
+ ,uart->base + UART_LCR); /* Line Control Register */
+ outb(uart->old_divisor_lsb
+ ,uart->base + UART_DLL); /* Divisor Latch Low */
+ outb(uart->old_divisor_msb
+ ,uart->base + UART_DLM); /* Divisor Latch High */
+ /* Restore old LCR (data bits, stop bits, parity, DLAB) */
+ outb(uart->old_line_ctrl_reg
+ ,uart->base + UART_LCR); /* Line Control Register */
+ }
+}
+
+static int snd_uart16550_input_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_uart16550 *uart = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&uart->open_lock, flags);
+ if (uart->filemode == SERIAL_MODE_NOT_OPENED)
+ snd_uart16550_do_open(uart);
+ uart->filemode |= SERIAL_MODE_INPUT_OPEN;
+ uart->midi_input[substream->number] = substream;
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+ return 0;
+}
+
+static int snd_uart16550_input_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_uart16550 *uart = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&uart->open_lock, flags);
+ uart->filemode &= ~SERIAL_MODE_INPUT_OPEN;
+ uart->midi_input[substream->number] = NULL;
+ if (uart->filemode == SERIAL_MODE_NOT_OPENED)
+ snd_uart16550_do_close(uart);
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+ return 0;
+}
+
+static void snd_uart16550_input_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ unsigned long flags;
+ struct snd_uart16550 *uart = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&uart->open_lock, flags);
+ if (up)
+ uart->filemode |= SERIAL_MODE_INPUT_TRIGGERED;
+ else
+ uart->filemode &= ~SERIAL_MODE_INPUT_TRIGGERED;
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+}
+
+static int snd_uart16550_output_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_uart16550 *uart = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&uart->open_lock, flags);
+ if (uart->filemode == SERIAL_MODE_NOT_OPENED)
+ snd_uart16550_do_open(uart);
+ uart->filemode |= SERIAL_MODE_OUTPUT_OPEN;
+ uart->midi_output[substream->number] = substream;
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+ return 0;
+};
+
+static int snd_uart16550_output_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_uart16550 *uart = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&uart->open_lock, flags);
+ uart->filemode &= ~SERIAL_MODE_OUTPUT_OPEN;
+ uart->midi_output[substream->number] = NULL;
+ if (uart->filemode == SERIAL_MODE_NOT_OPENED)
+ snd_uart16550_do_close(uart);
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+ return 0;
+};
+
+static inline int snd_uart16550_buffer_can_write(struct snd_uart16550 *uart,
+ int Num)
+{
+ if (uart->buff_in_count + Num < TX_BUFF_SIZE)
+ return 1;
+ else
+ return 0;
+}
+
+static inline int snd_uart16550_write_buffer(struct snd_uart16550 *uart,
+ unsigned char byte)
+{
+ unsigned short buff_in = uart->buff_in;
+ if (uart->buff_in_count < TX_BUFF_SIZE) {
+ uart->tx_buff[buff_in] = byte;
+ buff_in++;
+ buff_in &= TX_BUFF_MASK;
+ uart->buff_in = buff_in;
+ uart->buff_in_count++;
+ if (uart->irq < 0) /* polling mode */
+ snd_uart16550_add_timer(uart);
+ return 1;
+ } else
+ return 0;
+}
+
+static int snd_uart16550_output_byte(struct snd_uart16550 *uart,
+ struct snd_rawmidi_substream *substream,
+ unsigned char midi_byte)
+{
+ if (uart->buff_in_count == 0 /* Buffer empty? */
+ && ((uart->adaptor != SNDRV_SERIAL_MS124W_SA &&
+ uart->adaptor != SNDRV_SERIAL_GENERIC) ||
+ (uart->fifo_count == 0 /* FIFO empty? */
+ && (inb(uart->base + UART_MSR) & UART_MSR_CTS)))) { /* CTS? */
+
+ /* Tx Buffer Empty - try to write immediately */
+ if ((inb(uart->base + UART_LSR) & UART_LSR_THRE) != 0) {
+ /* Transmitter holding register (and Tx FIFO) empty */
+ uart->fifo_count = 1;
+ outb(midi_byte, uart->base + UART_TX);
+ } else {
+ if (uart->fifo_count < uart->fifo_limit) {
+ uart->fifo_count++;
+ outb(midi_byte, uart->base + UART_TX);
+ } else {
+ /* Cannot write (buffer empty) -
+ * put char in buffer */
+ snd_uart16550_write_buffer(uart, midi_byte);
+ }
+ }
+ } else {
+ if (!snd_uart16550_write_buffer(uart, midi_byte)) {
+ snd_printk("%s: Buffer overrun on device at 0x%lx\n",
+ uart->rmidi->name, uart->base);
+ return 0;
+ }
+ }
+
+ return 1;
+}
+
+static void snd_uart16550_output_write(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ unsigned char midi_byte, addr_byte;
+ struct snd_uart16550 *uart = substream->rmidi->private_data;
+ char first;
+ static unsigned long lasttime = 0;
+
+ /* Interrupts are disabled during the updating of the tx_buff,
+ * since it is 'bad' to have two processes updating the same
+ * variables (ie buff_in & buff_out)
+ */
+
+ spin_lock_irqsave(&uart->open_lock, flags);
+
+ if (uart->irq < 0) /* polling */
+ snd_uart16550_io_loop(uart);
+
+ if (uart->adaptor == SNDRV_SERIAL_MS124W_MB) {
+ while (1) {
+ /* buffer full? */
+ /* in this mode we need two bytes of space */
+ if (uart->buff_in_count > TX_BUFF_SIZE - 2)
+ break;
+ if (snd_rawmidi_transmit(substream, &midi_byte, 1) != 1)
+ break;
+#ifdef SNDRV_SERIAL_MS124W_MB_NOCOMBO
+ /* select exactly one of the four ports */
+ addr_byte = (1 << (substream->number + 4)) | 0x08;
+#else
+ /* select any combination of the four ports */
+ addr_byte = (substream->number << 4) | 0x08;
+ /* ...except none */
+ if (addr_byte == 0x08)
+ addr_byte = 0xf8;
+#endif
+ snd_uart16550_output_byte(uart, substream, addr_byte);
+ /* send midi byte */
+ snd_uart16550_output_byte(uart, substream, midi_byte);
+ }
+ } else {
+ first = 0;
+ while (snd_rawmidi_transmit_peek(substream, &midi_byte, 1) == 1) {
+ /* Also send F5 after 3 seconds with no data
+ * to handle device disconnect */
+ if (first == 0 &&
+ (uart->adaptor == SNDRV_SERIAL_SOUNDCANVAS ||
+ uart->adaptor == SNDRV_SERIAL_GENERIC) &&
+ (uart->prev_out != substream->number ||
+ time_after(jiffies, lasttime + 3*HZ))) {
+
+ if (snd_uart16550_buffer_can_write(uart, 3)) {
+ /* Roland Soundcanvas part selection */
+ /* If this substream of the data is
+ * different previous substream
+ * in this uart, send the change part
+ * event
+ */
+ uart->prev_out = substream->number;
+ /* change part */
+ snd_uart16550_output_byte(uart, substream,
+ 0xf5);
+ /* data */
+ snd_uart16550_output_byte(uart, substream,
+ uart->prev_out + 1);
+ /* If midi_byte is a data byte,
+ * send the previous status byte */
+ if (midi_byte < 0x80 &&
+ uart->adaptor == SNDRV_SERIAL_SOUNDCANVAS)
+ snd_uart16550_output_byte(uart, substream, uart->prev_status[uart->prev_out]);
+ } else if (!uart->drop_on_full)
+ break;
+
+ }
+
+ /* send midi byte */
+ if (!snd_uart16550_output_byte(uart, substream, midi_byte) &&
+ !uart->drop_on_full )
+ break;
+
+ if (midi_byte >= 0x80 && midi_byte < 0xf0)
+ uart->prev_status[uart->prev_out] = midi_byte;
+ first = 1;
+
+ snd_rawmidi_transmit_ack( substream, 1 );
+ }
+ lasttime = jiffies;
+ }
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+}
+
+static void snd_uart16550_output_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ unsigned long flags;
+ struct snd_uart16550 *uart = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&uart->open_lock, flags);
+ if (up)
+ uart->filemode |= SERIAL_MODE_OUTPUT_TRIGGERED;
+ else
+ uart->filemode &= ~SERIAL_MODE_OUTPUT_TRIGGERED;
+ spin_unlock_irqrestore(&uart->open_lock, flags);
+ if (up)
+ snd_uart16550_output_write(substream);
+}
+
+static struct snd_rawmidi_ops snd_uart16550_output =
+{
+ .open = snd_uart16550_output_open,
+ .close = snd_uart16550_output_close,
+ .trigger = snd_uart16550_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_uart16550_input =
+{
+ .open = snd_uart16550_input_open,
+ .close = snd_uart16550_input_close,
+ .trigger = snd_uart16550_input_trigger,
+};
+
+static int snd_uart16550_free(struct snd_uart16550 *uart)
+{
+ if (uart->irq >= 0)
+ free_irq(uart->irq, uart);
+ release_and_free_resource(uart->res_base);
+ kfree(uart);
+ return 0;
+};
+
+static int snd_uart16550_dev_free(struct snd_device *device)
+{
+ struct snd_uart16550 *uart = device->device_data;
+ return snd_uart16550_free(uart);
+}
+
+static int __devinit snd_uart16550_create(struct snd_card *card,
+ unsigned long iobase,
+ int irq,
+ unsigned int speed,
+ unsigned int base,
+ int adaptor,
+ int droponfull,
+ struct snd_uart16550 **ruart)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_uart16550_dev_free,
+ };
+ struct snd_uart16550 *uart;
+ int err;
+
+
+ if ((uart = kzalloc(sizeof(*uart), GFP_KERNEL)) == NULL)
+ return -ENOMEM;
+ uart->adaptor = adaptor;
+ uart->card = card;
+ spin_lock_init(&uart->open_lock);
+ uart->irq = -1;
+ uart->base = iobase;
+ uart->drop_on_full = droponfull;
+
+ if ((err = snd_uart16550_detect(uart)) <= 0) {
+ printk(KERN_ERR "no UART detected at 0x%lx\n", iobase);
+ snd_uart16550_free(uart);
+ return -ENODEV;
+ }
+
+ if (irq >= 0 && irq != SNDRV_AUTO_IRQ) {
+ if (request_irq(irq, snd_uart16550_interrupt,
+ IRQF_DISABLED, "Serial MIDI", uart)) {
+ snd_printk("irq %d busy. Using Polling.\n", irq);
+ } else {
+ uart->irq = irq;
+ }
+ }
+ uart->divisor = base / speed;
+ uart->speed = base / (unsigned int)uart->divisor;
+ uart->speed_base = base;
+ uart->prev_out = -1;
+ uart->prev_in = 0;
+ uart->rstatus = 0;
+ memset(uart->prev_status, 0x80, sizeof(unsigned char) * SNDRV_SERIAL_MAX_OUTS);
+ init_timer(&uart->buffer_timer);
+ uart->buffer_timer.function = snd_uart16550_buffer_timer;
+ uart->buffer_timer.data = (unsigned long)uart;
+ uart->timer_running = 0;
+
+ /* Register device */
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, uart, &ops)) < 0) {
+ snd_uart16550_free(uart);
+ return err;
+ }
+
+ switch (uart->adaptor) {
+ case SNDRV_SERIAL_MS124W_SA:
+ case SNDRV_SERIAL_MS124W_MB:
+ /* MS-124W can draw power from RTS and DTR if they
+ are in opposite states. */
+ outb(UART_MCR_RTS | (0&UART_MCR_DTR), uart->base + UART_MCR);
+ break;
+ case SNDRV_SERIAL_MS124T:
+ /* MS-124T can draw power from RTS and/or DTR (preferably
+ both) if they are asserted. */
+ outb(UART_MCR_RTS | UART_MCR_DTR, uart->base + UART_MCR);
+ break;
+ default:
+ break;
+ }
+
+ if (ruart)
+ *ruart = uart;
+
+ return 0;
+}
+
+static void __devinit snd_uart16550_substreams(struct snd_rawmidi_str *stream)
+{
+ struct snd_rawmidi_substream *substream;
+
+ list_for_each_entry(substream, &stream->substreams, list) {
+ sprintf(substream->name, "Serial MIDI %d", substream->number + 1);
+ }
+}
+
+static int __devinit snd_uart16550_rmidi(struct snd_uart16550 *uart, int device,
+ int outs, int ins,
+ struct snd_rawmidi **rmidi)
+{
+ struct snd_rawmidi *rrawmidi;
+ int err;
+
+ err = snd_rawmidi_new(uart->card, "UART Serial MIDI", device,
+ outs, ins, &rrawmidi);
+ if (err < 0)
+ return err;
+ snd_rawmidi_set_ops(rrawmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_uart16550_input);
+ snd_rawmidi_set_ops(rrawmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_uart16550_output);
+ strcpy(rrawmidi->name, "Serial MIDI");
+ snd_uart16550_substreams(&rrawmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]);
+ snd_uart16550_substreams(&rrawmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]);
+ rrawmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ rrawmidi->private_data = uart;
+ if (rmidi)
+ *rmidi = rrawmidi;
+ return 0;
+}
+
+static int __devinit snd_serial_probe(struct platform_device *devptr)
+{
+ struct snd_card *card;
+ struct snd_uart16550 *uart;
+ int err;
+ int dev = devptr->id;
+
+ switch (adaptor[dev]) {
+ case SNDRV_SERIAL_SOUNDCANVAS:
+ ins[dev] = 1;
+ break;
+ case SNDRV_SERIAL_MS124T:
+ case SNDRV_SERIAL_MS124W_SA:
+ outs[dev] = 1;
+ ins[dev] = 1;
+ break;
+ case SNDRV_SERIAL_MS124W_MB:
+ outs[dev] = 16;
+ ins[dev] = 1;
+ break;
+ case SNDRV_SERIAL_GENERIC:
+ break;
+ default:
+ snd_printk("Adaptor type is out of range 0-%d (%d)\n",
+ SNDRV_SERIAL_MAX_ADAPTOR, adaptor[dev]);
+ return -ENODEV;
+ }
+
+ if (outs[dev] < 1 || outs[dev] > SNDRV_SERIAL_MAX_OUTS) {
+ snd_printk("Count of outputs is out of range 1-%d (%d)\n",
+ SNDRV_SERIAL_MAX_OUTS, outs[dev]);
+ return -ENODEV;
+ }
+
+ if (ins[dev] < 1 || ins[dev] > SNDRV_SERIAL_MAX_INS) {
+ snd_printk("Count of inputs is out of range 1-%d (%d)\n",
+ SNDRV_SERIAL_MAX_INS, ins[dev]);
+ return -ENODEV;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "Serial");
+ strcpy(card->shortname, "Serial MIDI (UART16550A)");
+
+ if ((err = snd_uart16550_create(card,
+ port[dev],
+ irq[dev],
+ speed[dev],
+ base[dev],
+ adaptor[dev],
+ droponfull[dev],
+ &uart)) < 0)
+ goto _err;
+
+ err = snd_uart16550_rmidi(uart, 0, outs[dev], ins[dev], &uart->rmidi);
+ if (err < 0)
+ goto _err;
+
+ sprintf(card->longname, "%s at 0x%lx, irq %d speed %d div %d outs %d ins %d adaptor %s droponfull %d",
+ card->shortname,
+ uart->base,
+ uart->irq,
+ uart->speed,
+ (int)uart->divisor,
+ outs[dev],
+ ins[dev],
+ adaptor_names[uart->adaptor],
+ uart->drop_on_full);
+
+ snd_card_set_dev(card, &devptr->dev);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto _err;
+
+ platform_set_drvdata(devptr, card);
+ return 0;
+
+ _err:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_serial_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define SND_SERIAL_DRIVER "snd_serial_u16550"
+
+static struct platform_driver snd_serial_driver = {
+ .probe = snd_serial_probe,
+ .remove = __devexit_p( snd_serial_remove),
+ .driver = {
+ .name = SND_SERIAL_DRIVER
+ },
+};
+
+static void snd_serial_unregister_all(void)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(devices); ++i)
+ platform_device_unregister(devices[i]);
+ platform_driver_unregister(&snd_serial_driver);
+}
+
+static int __init alsa_card_serial_init(void)
+{
+ int i, cards, err;
+
+ if ((err = platform_driver_register(&snd_serial_driver)) < 0)
+ return err;
+
+ cards = 0;
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ struct platform_device *device;
+ if (! enable[i])
+ continue;
+ device = platform_device_register_simple(SND_SERIAL_DRIVER,
+ i, NULL, 0);
+ if (IS_ERR(device))
+ continue;
+ if (!platform_get_drvdata(device)) {
+ platform_device_unregister(device);
+ continue;
+ }
+ devices[i] = device;
+ cards++;
+ }
+ if (! cards) {
+#ifdef MODULE
+ printk(KERN_ERR "serial midi soundcard not found or device busy\n");
+#endif
+ snd_serial_unregister_all();
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_serial_exit(void)
+{
+ snd_serial_unregister_all();
+}
+
+module_init(alsa_card_serial_init)
+module_exit(alsa_card_serial_exit)
diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c
new file mode 100644
index 0000000..f79e361
--- /dev/null
+++ b/sound/drivers/virmidi.c
@@ -0,0 +1,195 @@
+/*
+ * Dummy soundcard for virtual rawmidi devices
+ *
+ * Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * VIRTUAL RAW MIDI DEVICE CARDS
+ *
+ * This dummy card contains up to 4 virtual rawmidi devices.
+ * They are not real rawmidi devices but just associated with sequencer
+ * clients, so that any input/output sources can be connected as a raw
+ * MIDI device arbitrary.
+ * Also, multiple access is allowed to a single rawmidi device.
+ *
+ * Typical usage is like following:
+ * - Load snd-virmidi module.
+ * # modprobe snd-virmidi index=2
+ * Then, sequencer clients 72:0 to 75:0 will be created, which are
+ * mapped from /dev/snd/midiC1D0 to /dev/snd/midiC1D3, respectively.
+ *
+ * - Connect input/output via aconnect.
+ * % aconnect 64:0 72:0 # keyboard input redirection 64:0 -> 72:0
+ * % aconnect 72:0 65:0 # output device redirection 72:0 -> 65:0
+ *
+ * - Run application using a midi device (eg. /dev/snd/midiC1D0)
+ */
+
+#include <linux/init.h>
+#include <linux/wait.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/seq_kernel.h>
+#include <sound/seq_virmidi.h>
+#include <sound/initval.h>
+
+/* hack: OSS defines midi_devs, so undefine it (versioned symbols) */
+#undef midi_devs
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("Dummy soundcard for virtual rawmidi devices");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual rawmidi device}}");
+
+#define MAX_MIDI_DEVICES 4
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0};
+static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for virmidi soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for virmidi soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable this soundcard.");
+module_param_array(midi_devs, int, NULL, 0444);
+MODULE_PARM_DESC(midi_devs, "MIDI devices # (1-4)");
+
+struct snd_card_virmidi {
+ struct snd_card *card;
+ struct snd_rawmidi *midi[MAX_MIDI_DEVICES];
+};
+
+static struct platform_device *devices[SNDRV_CARDS];
+
+
+static int __devinit snd_virmidi_probe(struct platform_device *devptr)
+{
+ struct snd_card *card;
+ struct snd_card_virmidi *vmidi;
+ int idx, err;
+ int dev = devptr->id;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_virmidi));
+ if (card == NULL)
+ return -ENOMEM;
+ vmidi = (struct snd_card_virmidi *)card->private_data;
+ vmidi->card = card;
+
+ if (midi_devs[dev] > MAX_MIDI_DEVICES) {
+ snd_printk("too much midi devices for virmidi %d: force to use %d\n", dev, MAX_MIDI_DEVICES);
+ midi_devs[dev] = MAX_MIDI_DEVICES;
+ }
+ for (idx = 0; idx < midi_devs[dev]; idx++) {
+ struct snd_rawmidi *rmidi;
+ struct snd_virmidi_dev *rdev;
+ if ((err = snd_virmidi_new(card, idx, &rmidi)) < 0)
+ goto __nodev;
+ rdev = rmidi->private_data;
+ vmidi->midi[idx] = rmidi;
+ strcpy(rmidi->name, "Virtual Raw MIDI");
+ rdev->seq_mode = SNDRV_VIRMIDI_SEQ_DISPATCH;
+ }
+
+ strcpy(card->driver, "VirMIDI");
+ strcpy(card->shortname, "VirMIDI");
+ sprintf(card->longname, "Virtual MIDI Card %i", dev + 1);
+
+ snd_card_set_dev(card, &devptr->dev);
+
+ if ((err = snd_card_register(card)) == 0) {
+ platform_set_drvdata(devptr, card);
+ return 0;
+ }
+ __nodev:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_virmidi_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define SND_VIRMIDI_DRIVER "snd_virmidi"
+
+static struct platform_driver snd_virmidi_driver = {
+ .probe = snd_virmidi_probe,
+ .remove = __devexit_p(snd_virmidi_remove),
+ .driver = {
+ .name = SND_VIRMIDI_DRIVER
+ },
+};
+
+static void snd_virmidi_unregister_all(void)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(devices); ++i)
+ platform_device_unregister(devices[i]);
+ platform_driver_unregister(&snd_virmidi_driver);
+}
+
+static int __init alsa_card_virmidi_init(void)
+{
+ int i, cards, err;
+
+ if ((err = platform_driver_register(&snd_virmidi_driver)) < 0)
+ return err;
+
+ cards = 0;
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ struct platform_device *device;
+ if (! enable[i])
+ continue;
+ device = platform_device_register_simple(SND_VIRMIDI_DRIVER,
+ i, NULL, 0);
+ if (IS_ERR(device))
+ continue;
+ if (!platform_get_drvdata(device)) {
+ platform_device_unregister(device);
+ continue;
+ }
+ devices[i] = device;
+ cards++;
+ }
+ if (!cards) {
+#ifdef MODULE
+ printk(KERN_ERR "Card-VirMIDI soundcard not found or device busy\n");
+#endif
+ snd_virmidi_unregister_all();
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_virmidi_exit(void)
+{
+ snd_virmidi_unregister_all();
+}
+
+module_init(alsa_card_virmidi_init)
+module_exit(alsa_card_virmidi_exit)
diff --git a/sound/drivers/vx/Makefile b/sound/drivers/vx/Makefile
new file mode 100644
index 0000000..9a168a3
--- /dev/null
+++ b/sound/drivers/vx/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-vx-lib-objs := vx_core.o vx_hwdep.o vx_pcm.o vx_mixer.o vx_cmd.o vx_uer.o
+
+obj-$(CONFIG_SND_VX_LIB) += snd-vx-lib.o
diff --git a/sound/drivers/vx/vx_cmd.c b/sound/drivers/vx/vx_cmd.c
new file mode 100644
index 0000000..23f4857
--- /dev/null
+++ b/sound/drivers/vx/vx_cmd.c
@@ -0,0 +1,109 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * DSP commands
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/vx_core.h>
+#include "vx_cmd.h"
+
+/*
+ * Array of DSP commands
+ */
+static struct vx_cmd_info vx_dsp_cmds[] = {
+[CMD_VERSION] = { 0x010000, 2, RMH_SSIZE_FIXED, 1 },
+[CMD_SUPPORTED] = { 0x020000, 1, RMH_SSIZE_FIXED, 2 },
+[CMD_TEST_IT] = { 0x040000, 1, RMH_SSIZE_FIXED, 1 },
+[CMD_SEND_IRQA] = { 0x070001, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_IBL] = { 0x080000, 1, RMH_SSIZE_FIXED, 4 },
+[CMD_ASYNC] = { 0x0A0000, 1, RMH_SSIZE_ARG, 0 },
+[CMD_RES_PIPE] = { 0x400000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_FREE_PIPE] = { 0x410000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_CONF_PIPE] = { 0x42A101, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_ABORT_CONF_PIPE] = { 0x42A100, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_PARAM_OUTPUT_PIPE] = { 0x43A000, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_STOP_PIPE] = { 0x470004, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_PIPE_STATE] = { 0x480000, 1, RMH_SSIZE_FIXED, 1 },
+[CMD_PIPE_SPL_COUNT] = { 0x49A000, 2, RMH_SSIZE_FIXED, 2 },
+[CMD_CAN_START_PIPE] = { 0x4b0000, 1, RMH_SSIZE_FIXED, 1 },
+[CMD_SIZE_HBUFFER] = { 0x4C0000, 1, RMH_SSIZE_FIXED, 1 },
+[CMD_START_STREAM] = { 0x80A000, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_START_ONE_STREAM] = { 0x800000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_PAUSE_STREAM] = { 0x81A000, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_PAUSE_ONE_STREAM] = { 0x810000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_STREAM_OUT_LEVEL_ADJUST] = { 0x828000, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_STOP_STREAM] = { 0x830000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_FORMAT_STREAM_OUT] = { 0x868000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_FORMAT_STREAM_IN] = { 0x878800, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_GET_STREAM_STATE] = { 0x890001, 2, RMH_SSIZE_FIXED, 1 },
+[CMD_DROP_BYTES_AWAY] = { 0x8A8000, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_GET_REMAINING_BYTES] = { 0x8D0800, 1, RMH_SSIZE_FIXED, 2 },
+[CMD_CONNECT_AUDIO] = { 0xC10000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_AUDIO_LEVEL_ADJUST] = { 0xC2A000, 3, RMH_SSIZE_FIXED, 0 },
+[CMD_AUDIO_VU_PIC_METER] = { 0xC3A003, 2, RMH_SSIZE_FIXED, 1 },
+[CMD_GET_AUDIO_LEVELS] = { 0xC4A000, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_GET_NOTIFY_EVENT] = { 0x4D0000, 1, RMH_SSIZE_ARG, 0 },
+[CMD_INFO_NOTIFIED] = { 0x0B0000, 1, RMH_SSIZE_FIXED, 2 },
+[CMD_ACCESS_IO_FCT] = { 0x098000, 1, RMH_SSIZE_ARG, 0 },
+[CMD_STATUS_R_BUFFERS] = { 0x440000, 1, RMH_SSIZE_ARG, 0 },
+[CMD_UPDATE_R_BUFFERS] = { 0x848000, 4, RMH_SSIZE_FIXED, 0 },
+[CMD_LOAD_EFFECT_CONTEXT] = { 0x0c8000, 3, RMH_SSIZE_FIXED, 1 },
+[CMD_EFFECT_ONE_PIPE] = { 0x458000, 0, RMH_SSIZE_FIXED, 0 },
+[CMD_MODIFY_CLOCK] = { 0x0d0000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_STREAM1_OUT_SET_N_LEVELS] ={ 0x858000, 3, RMH_SSIZE_FIXED, 0 },
+[CMD_PURGE_STREAM_DCMDS] = { 0x8b8000, 3, RMH_SSIZE_FIXED, 0 },
+[CMD_NOTIFY_PIPE_TIME] = { 0x4e0000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_LOAD_EFFECT_CONTEXT_PACKET] = { 0x0c8000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_RELIC_R_BUFFER] = { 0x8e0800, 1, RMH_SSIZE_FIXED, 1 },
+[CMD_RESYNC_AUDIO_INPUTS] = { 0x0e0000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_NOTIFY_STREAM_TIME] = { 0x8f0000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_STREAM_SAMPLE_COUNT] = { 0x900000, 1, RMH_SSIZE_FIXED, 2 },
+[CMD_CONFIG_TIME_CODE] = { 0x050000, 2, RMH_SSIZE_FIXED, 0 },
+[CMD_GET_TIME_CODE] = { 0x060000, 1, RMH_SSIZE_FIXED, 5 },
+[CMD_MANAGE_SIGNAL] = { 0x0f0000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_PARAMETER_STREAM_OUT] = { 0x91A000, 3, RMH_SSIZE_FIXED, 0 },
+[CMD_READ_BOARD_FREQ] = { 0x030000, 1, RMH_SSIZE_FIXED, 2 },
+[CMD_GET_STREAM_LEVELS] = { 0x8c0000, 1, RMH_SSIZE_FIXED, 3 },
+[CMD_PURGE_PIPE_DCMDS] = { 0x4f8000, 3, RMH_SSIZE_FIXED, 0 },
+// [CMD_SET_STREAM_OUT_EFFECTS] = { 0x888000, 34, RMH_SSIZE_FIXED, 0 },
+// [CMD_GET_STREAM_OUT_EFFECTS] = { 0x928000, 2, RMH_SSIZE_FIXED, 32 },
+[CMD_CONNECT_MONITORING] = { 0xC00000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_STREAM2_OUT_SET_N_LEVELS] = { 0x938000, 3, RMH_SSIZE_FIXED, 0 },
+[CMD_CANCEL_R_BUFFERS] = { 0x948000, 4, RMH_SSIZE_FIXED, 0 },
+[CMD_NOTIFY_END_OF_BUFFER] = { 0x950000, 1, RMH_SSIZE_FIXED, 0 },
+[CMD_GET_STREAM_VU_METER] = { 0x95A000, 2, RMH_SSIZE_ARG, 0 },
+};
+
+/**
+ * vx_init_rmh - initialize the RMH instance
+ * @rmh: the rmh pointer to be initialized
+ * @cmd: the rmh command to be set
+ */
+void vx_init_rmh(struct vx_rmh *rmh, unsigned int cmd)
+{
+ if (snd_BUG_ON(cmd >= CMD_LAST_INDEX))
+ return;
+ rmh->LgCmd = vx_dsp_cmds[cmd].length;
+ rmh->LgStat = vx_dsp_cmds[cmd].st_length;
+ rmh->DspStat = vx_dsp_cmds[cmd].st_type;
+ rmh->Cmd[0] = vx_dsp_cmds[cmd].opcode;
+}
+
diff --git a/sound/drivers/vx/vx_cmd.h b/sound/drivers/vx/vx_cmd.h
new file mode 100644
index 0000000..a85248b
--- /dev/null
+++ b/sound/drivers/vx/vx_cmd.h
@@ -0,0 +1,246 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * Definitions of DSP commands
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __VX_CMD_H
+#define __VX_CMD_H
+
+enum {
+ CMD_VERSION,
+ CMD_SUPPORTED,
+ CMD_TEST_IT,
+ CMD_SEND_IRQA,
+ CMD_IBL,
+ CMD_ASYNC,
+ CMD_RES_PIPE,
+ CMD_FREE_PIPE,
+ CMD_CONF_PIPE,
+ CMD_ABORT_CONF_PIPE,
+ CMD_PARAM_OUTPUT_PIPE,
+ CMD_STOP_PIPE,
+ CMD_PIPE_STATE,
+ CMD_PIPE_SPL_COUNT,
+ CMD_CAN_START_PIPE,
+ CMD_SIZE_HBUFFER,
+ CMD_START_STREAM,
+ CMD_START_ONE_STREAM,
+ CMD_PAUSE_STREAM,
+ CMD_PAUSE_ONE_STREAM,
+ CMD_STREAM_OUT_LEVEL_ADJUST,
+ CMD_STOP_STREAM,
+ CMD_FORMAT_STREAM_OUT,
+ CMD_FORMAT_STREAM_IN,
+ CMD_GET_STREAM_STATE,
+ CMD_DROP_BYTES_AWAY,
+ CMD_GET_REMAINING_BYTES,
+ CMD_CONNECT_AUDIO,
+ CMD_AUDIO_LEVEL_ADJUST,
+ CMD_AUDIO_VU_PIC_METER,
+ CMD_GET_AUDIO_LEVELS,
+ CMD_GET_NOTIFY_EVENT,
+ CMD_INFO_NOTIFIED,
+ CMD_ACCESS_IO_FCT,
+ CMD_STATUS_R_BUFFERS,
+ CMD_UPDATE_R_BUFFERS,
+ CMD_LOAD_EFFECT_CONTEXT,
+ CMD_EFFECT_ONE_PIPE,
+ CMD_MODIFY_CLOCK,
+ CMD_STREAM1_OUT_SET_N_LEVELS,
+ CMD_PURGE_STREAM_DCMDS,
+ CMD_NOTIFY_PIPE_TIME,
+ CMD_LOAD_EFFECT_CONTEXT_PACKET,
+ CMD_RELIC_R_BUFFER,
+ CMD_RESYNC_AUDIO_INPUTS,
+ CMD_NOTIFY_STREAM_TIME,
+ CMD_STREAM_SAMPLE_COUNT,
+ CMD_CONFIG_TIME_CODE,
+ CMD_GET_TIME_CODE,
+ CMD_MANAGE_SIGNAL,
+ CMD_PARAMETER_STREAM_OUT,
+ CMD_READ_BOARD_FREQ,
+ CMD_GET_STREAM_LEVELS,
+ CMD_PURGE_PIPE_DCMDS,
+ // CMD_SET_STREAM_OUT_EFFECTS,
+ // CMD_GET_STREAM_OUT_EFFECTS,
+ CMD_CONNECT_MONITORING,
+ CMD_STREAM2_OUT_SET_N_LEVELS,
+ CMD_CANCEL_R_BUFFERS,
+ CMD_NOTIFY_END_OF_BUFFER,
+ CMD_GET_STREAM_VU_METER,
+ CMD_LAST_INDEX
+};
+
+struct vx_cmd_info {
+ unsigned int opcode; /* command word */
+ int length; /* command length (in words) */
+ int st_type; /* status type (RMH_SSIZE_XXX) */
+ int st_length; /* fixed length */
+};
+
+/* Family and code op of some DSP requests. */
+#define CODE_OP_PIPE_TIME 0x004e0000
+#define CODE_OP_START_STREAM 0x00800000
+#define CODE_OP_PAUSE_STREAM 0x00810000
+#define CODE_OP_OUT_STREAM_LEVEL 0x00820000
+#define CODE_OP_UPDATE_R_BUFFERS 0x00840000
+#define CODE_OP_OUT_STREAM1_LEVEL_CURVE 0x00850000
+#define CODE_OP_OUT_STREAM2_LEVEL_CURVE 0x00930000
+#define CODE_OP_OUT_STREAM_FORMAT 0x00860000
+#define CODE_OP_STREAM_TIME 0x008f0000
+#define CODE_OP_OUT_STREAM_EXTRAPARAMETER 0x00910000
+#define CODE_OP_OUT_AUDIO_LEVEL 0x00c20000
+
+#define NOTIFY_LAST_COMMAND 0x00400000
+
+/* Values for a user delay */
+#define DC_DIFFERED_DELAY (1<<BIT_DIFFERED_COMMAND)
+#define DC_NOTIFY_DELAY (1<<BIT_NOTIFIED_COMMAND)
+#define DC_HBUFFER_DELAY (1<<BIT_TIME_RELATIVE_TO_BUFFER)
+#define DC_MULTIPLE_DELAY (1<<BIT_RESERVED)
+#define DC_STREAM_TIME_DELAY (1<<BIT_STREAM_TIME)
+#define DC_CANCELLED_DELAY (1<<BIT_CANCELLED_COMMAND)
+
+/* Values for tiDelayed field in TIME_INFO structure,
+ * and for pbPause field in PLAY_BUFFER_INFO structure
+ */
+#define BIT_DIFFERED_COMMAND 0
+#define BIT_NOTIFIED_COMMAND 1
+#define BIT_TIME_RELATIVE_TO_BUFFER 2
+#define BIT_RESERVED 3
+#define BIT_STREAM_TIME 4
+#define BIT_CANCELLED_COMMAND 5
+
+/* Access to the "Size" field of the response of the CMD_GET_NOTIFY_EVENT request. */
+#define GET_NOTIFY_EVENT_SIZE_FIELD_MASK 0x000000ff
+
+/* DSP commands general masks */
+#define OPCODE_MASK 0x00ff0000
+#define DSP_DIFFERED_COMMAND_MASK 0x0000C000
+
+/* Notifications (NOTIFY_INFO) */
+#define ALL_CMDS_NOTIFIED 0x0000 // reserved
+#define START_STREAM_NOTIFIED 0x0001
+#define PAUSE_STREAM_NOTIFIED 0x0002
+#define OUT_STREAM_LEVEL_NOTIFIED 0x0003
+#define OUT_STREAM_PARAMETER_NOTIFIED 0x0004 // left for backward compatibility
+#define OUT_STREAM_FORMAT_NOTIFIED 0x0004
+#define PIPE_TIME_NOTIFIED 0x0005
+#define OUT_AUDIO_LEVEL_NOTIFIED 0x0006
+#define OUT_STREAM_LEVEL_CURVE_NOTIFIED 0x0007
+#define STREAM_TIME_NOTIFIED 0x0008
+#define OUT_STREAM_EXTRAPARAMETER_NOTIFIED 0x0009
+#define UNKNOWN_COMMAND_NOTIFIED 0xffff
+
+/* Output pipe parameters setting */
+#define MASK_VALID_PIPE_MPEG_PARAM 0x000040
+#define MASK_VALID_PIPE_BACKWARD_PARAM 0x000020
+#define MASK_SET_PIPE_MPEG_PARAM 0x000002
+#define MASK_SET_PIPE_BACKWARD_PARAM 0x000001
+
+#define MASK_DSP_WORD 0x00FFFFFF
+#define MASK_ALL_STREAM 0x00FFFFFF
+#define MASK_DSP_WORD_LEVEL 0x000001FF
+#define MASK_FIRST_FIELD 0x0000001F
+#define FIELD_SIZE 5
+
+#define COMMAND_RECORD_MASK 0x000800
+
+/* PipeManagement definition bits (PIPE_DECL_INFO) */
+#define P_UNDERRUN_SKIP_SOUND_MASK 0x01
+#define P_PREPARE_FOR_MPEG3_MASK 0x02
+#define P_DO_NOT_RESET_ANALOG_LEVELS 0x04
+#define P_ALLOW_UNDER_ALLOCATION_MASK 0x08
+#define P_DATA_MODE_MASK 0x10
+#define P_ASIO_BUFFER_MANAGEMENT_MASK 0x20
+
+#define BIT_SKIP_SOUND 0x08 // bit 3
+#define BIT_DATA_MODE 0x10 // bit 4
+
+/* Bits in the CMD_MODIFY_CLOCK request. */
+#define CMD_MODIFY_CLOCK_FD_BIT 0x00000001
+#define CMD_MODIFY_CLOCK_T_BIT 0x00000002
+#define CMD_MODIFY_CLOCK_S_BIT 0x00000004
+
+/* Access to the results of the CMD_GET_TIME_CODE RMH. */
+#define TIME_CODE_V_MASK 0x00800000
+#define TIME_CODE_N_MASK 0x00400000
+#define TIME_CODE_B_MASK 0x00200000
+#define TIME_CODE_W_MASK 0x00100000
+
+/* Values for the CMD_MANAGE_SIGNAL RMH. */
+#define MANAGE_SIGNAL_TIME_CODE 0x01
+#define MANAGE_SIGNAL_MIDI 0x02
+
+/* Values for the CMD_CONFIG_TIME_CODE RMH. */
+#define CONFIG_TIME_CODE_CANCEL 0x00001000
+
+/* Mask to get only the effective time from the
+ * high word out of the 2 returned by the DSP
+ */
+#define PCX_TIME_HI_MASK 0x000fffff
+
+/* Values for setting a H-Buffer time */
+#define HBUFFER_TIME_HIGH 0x00200000
+#define HBUFFER_TIME_LOW 0x00000000
+
+#define NOTIFY_MASK_TIME_HIGH 0x00400000
+#define MULTIPLE_MASK_TIME_HIGH 0x00100000
+#define STREAM_MASK_TIME_HIGH 0x00800000
+
+
+/*
+ *
+ */
+void vx_init_rmh(struct vx_rmh *rmh, unsigned int cmd);
+
+/**
+ * vx_send_pipe_cmd_params - fill first command word for pipe commands
+ * @rmh: the rmh to be modified
+ * @is_capture: 0 = playback, 1 = capture operation
+ * @param1: first pipe-parameter
+ * @param2: second pipe-parameter
+ */
+static inline void vx_set_pipe_cmd_params(struct vx_rmh *rmh, int is_capture,
+ int param1, int param2)
+{
+ if (is_capture)
+ rmh->Cmd[0] |= COMMAND_RECORD_MASK;
+ rmh->Cmd[0] |= (((u32)param1 & MASK_FIRST_FIELD) << FIELD_SIZE) & MASK_DSP_WORD;
+
+ if (param2)
+ rmh->Cmd[0] |= ((u32)param2 & MASK_FIRST_FIELD) & MASK_DSP_WORD;
+
+}
+
+/**
+ * vx_set_stream_cmd_params - fill first command word for stream commands
+ * @rmh: the rmh to be modified
+ * @is_capture: 0 = playback, 1 = capture operation
+ * @pipe: the pipe index (zero-based)
+ */
+static inline void vx_set_stream_cmd_params(struct vx_rmh *rmh, int is_capture, int pipe)
+{
+ if (is_capture)
+ rmh->Cmd[0] |= COMMAND_RECORD_MASK;
+ rmh->Cmd[0] |= (((u32)pipe & MASK_FIRST_FIELD) << FIELD_SIZE) & MASK_DSP_WORD;
+}
+
+#endif /* __VX_CMD_H */
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
new file mode 100644
index 0000000..473b07f
--- /dev/null
+++ b/sound/drivers/vx/vx_core.c
@@ -0,0 +1,826 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * Hardware core part
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/asoundef.h>
+#include <sound/info.h>
+#include <asm/io.h>
+#include <sound/vx_core.h>
+#include "vx_cmd.h"
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("Common routines for Digigram VX drivers");
+MODULE_LICENSE("GPL");
+
+
+/*
+ * vx_check_reg_bit - wait for the specified bit is set/reset on a register
+ * @reg: register to check
+ * @mask: bit mask
+ * @bit: resultant bit to be checked
+ * @time: time-out of loop in msec
+ *
+ * returns zero if a bit matches, or a negative error code.
+ */
+int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int time)
+{
+ unsigned long end_time = jiffies + (time * HZ + 999) / 1000;
+#ifdef CONFIG_SND_DEBUG
+ static char *reg_names[VX_REG_MAX] = {
+ "ICR", "CVR", "ISR", "IVR", "RXH", "RXM", "RXL",
+ "DMA", "CDSP", "RFREQ", "RUER/V2", "DATA", "MEMIRQ",
+ "ACQ", "BIT0", "BIT1", "MIC0", "MIC1", "MIC2",
+ "MIC3", "INTCSR", "CNTRL", "GPIOC",
+ "LOFREQ", "HIFREQ", "CSUER", "RUER"
+ };
+#endif
+ do {
+ if ((snd_vx_inb(chip, reg) & mask) == bit)
+ return 0;
+ //msleep(10);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printd(KERN_DEBUG "vx_check_reg_bit: timeout, reg=%s, mask=0x%x, val=0x%x\n", reg_names[reg], mask, snd_vx_inb(chip, reg));
+ return -EIO;
+}
+
+EXPORT_SYMBOL(snd_vx_check_reg_bit);
+
+/*
+ * vx_send_irq_dsp - set command irq bit
+ * @num: the requested IRQ type, IRQ_XXX
+ *
+ * this triggers the specified IRQ request
+ * returns 0 if successful, or a negative error code.
+ *
+ */
+static int vx_send_irq_dsp(struct vx_core *chip, int num)
+{
+ int nirq;
+
+ /* wait for Hc = 0 */
+ if (snd_vx_check_reg_bit(chip, VX_CVR, CVR_HC, 0, 200) < 0)
+ return -EIO;
+
+ nirq = num;
+ if (vx_has_new_dsp(chip))
+ nirq += VXP_IRQ_OFFSET;
+ vx_outb(chip, CVR, (nirq >> 1) | CVR_HC);
+ return 0;
+}
+
+
+/*
+ * vx_reset_chk - reset CHK bit on ISR
+ *
+ * returns 0 if successful, or a negative error code.
+ */
+static int vx_reset_chk(struct vx_core *chip)
+{
+ /* Reset irq CHK */
+ if (vx_send_irq_dsp(chip, IRQ_RESET_CHK) < 0)
+ return -EIO;
+ /* Wait until CHK = 0 */
+ if (vx_check_isr(chip, ISR_CHK, 0, 200) < 0)
+ return -EIO;
+ return 0;
+}
+
+/*
+ * vx_transfer_end - terminate message transfer
+ * @cmd: IRQ message to send (IRQ_MESS_XXX_END)
+ *
+ * returns 0 if successful, or a negative error code.
+ * the error code can be VX-specific, retrieved via vx_get_error().
+ * NB: call with spinlock held!
+ */
+static int vx_transfer_end(struct vx_core *chip, int cmd)
+{
+ int err;
+
+ if ((err = vx_reset_chk(chip)) < 0)
+ return err;
+
+ /* irq MESS_READ/WRITE_END */
+ if ((err = vx_send_irq_dsp(chip, cmd)) < 0)
+ return err;
+
+ /* Wait CHK = 1 */
+ if ((err = vx_wait_isr_bit(chip, ISR_CHK)) < 0)
+ return err;
+
+ /* If error, Read RX */
+ if ((err = vx_inb(chip, ISR)) & ISR_ERR) {
+ if ((err = vx_wait_for_rx_full(chip)) < 0) {
+ snd_printd(KERN_DEBUG "transfer_end: error in rx_full\n");
+ return err;
+ }
+ err = vx_inb(chip, RXH) << 16;
+ err |= vx_inb(chip, RXM) << 8;
+ err |= vx_inb(chip, RXL);
+ snd_printd(KERN_DEBUG "transfer_end: error = 0x%x\n", err);
+ return -(VX_ERR_MASK | err);
+ }
+ return 0;
+}
+
+/*
+ * vx_read_status - return the status rmh
+ * @rmh: rmh record to store the status
+ *
+ * returns 0 if successful, or a negative error code.
+ * the error code can be VX-specific, retrieved via vx_get_error().
+ * NB: call with spinlock held!
+ */
+static int vx_read_status(struct vx_core *chip, struct vx_rmh *rmh)
+{
+ int i, err, val, size;
+
+ /* no read necessary? */
+ if (rmh->DspStat == RMH_SSIZE_FIXED && rmh->LgStat == 0)
+ return 0;
+
+ /* Wait for RX full (with timeout protection)
+ * The first word of status is in RX
+ */
+ err = vx_wait_for_rx_full(chip);
+ if (err < 0)
+ return err;
+
+ /* Read RX */
+ val = vx_inb(chip, RXH) << 16;
+ val |= vx_inb(chip, RXM) << 8;
+ val |= vx_inb(chip, RXL);
+
+ /* If status given by DSP, let's decode its size */
+ switch (rmh->DspStat) {
+ case RMH_SSIZE_ARG:
+ size = val & 0xff;
+ rmh->Stat[0] = val & 0xffff00;
+ rmh->LgStat = size + 1;
+ break;
+ case RMH_SSIZE_MASK:
+ /* Let's count the arg numbers from a mask */
+ rmh->Stat[0] = val;
+ size = 0;
+ while (val) {
+ if (val & 0x01)
+ size++;
+ val >>= 1;
+ }
+ rmh->LgStat = size + 1;
+ break;
+ default:
+ /* else retrieve the status length given by the driver */
+ size = rmh->LgStat;
+ rmh->Stat[0] = val; /* Val is the status 1st word */
+ size--; /* hence adjust remaining length */
+ break;
+ }
+
+ if (size < 1)
+ return 0;
+ if (snd_BUG_ON(size > SIZE_MAX_STATUS))
+ return -EINVAL;
+
+ for (i = 1; i <= size; i++) {
+ /* trigger an irq MESS_WRITE_NEXT */
+ err = vx_send_irq_dsp(chip, IRQ_MESS_WRITE_NEXT);
+ if (err < 0)
+ return err;
+ /* Wait for RX full (with timeout protection) */
+ err = vx_wait_for_rx_full(chip);
+ if (err < 0)
+ return err;
+ rmh->Stat[i] = vx_inb(chip, RXH) << 16;
+ rmh->Stat[i] |= vx_inb(chip, RXM) << 8;
+ rmh->Stat[i] |= vx_inb(chip, RXL);
+ }
+
+ return vx_transfer_end(chip, IRQ_MESS_WRITE_END);
+}
+
+
+#define MASK_MORE_THAN_1_WORD_COMMAND 0x00008000
+#define MASK_1_WORD_COMMAND 0x00ff7fff
+
+/*
+ * vx_send_msg_nolock - send a DSP message and read back the status
+ * @rmh: the rmh record to send and receive
+ *
+ * returns 0 if successful, or a negative error code.
+ * the error code can be VX-specific, retrieved via vx_get_error().
+ *
+ * this function doesn't call spinlock at all.
+ */
+int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh)
+{
+ int i, err;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+ if ((err = vx_reset_chk(chip)) < 0) {
+ snd_printd(KERN_DEBUG "vx_send_msg: vx_reset_chk error\n");
+ return err;
+ }
+
+#if 0
+ printk(KERN_DEBUG "rmh: cmd = 0x%06x, length = %d, stype = %d\n",
+ rmh->Cmd[0], rmh->LgCmd, rmh->DspStat);
+ if (rmh->LgCmd > 1) {
+ printk(KERN_DEBUG " ");
+ for (i = 1; i < rmh->LgCmd; i++)
+ printk("0x%06x ", rmh->Cmd[i]);
+ printk("\n");
+ }
+#endif
+ /* Check bit M is set according to length of the command */
+ if (rmh->LgCmd > 1)
+ rmh->Cmd[0] |= MASK_MORE_THAN_1_WORD_COMMAND;
+ else
+ rmh->Cmd[0] &= MASK_1_WORD_COMMAND;
+
+ /* Wait for TX empty */
+ if ((err = vx_wait_isr_bit(chip, ISR_TX_EMPTY)) < 0) {
+ snd_printd(KERN_DEBUG "vx_send_msg: wait tx empty error\n");
+ return err;
+ }
+
+ /* Write Cmd[0] */
+ vx_outb(chip, TXH, (rmh->Cmd[0] >> 16) & 0xff);
+ vx_outb(chip, TXM, (rmh->Cmd[0] >> 8) & 0xff);
+ vx_outb(chip, TXL, rmh->Cmd[0] & 0xff);
+
+ /* Trigger irq MESSAGE */
+ if ((err = vx_send_irq_dsp(chip, IRQ_MESSAGE)) < 0) {
+ snd_printd(KERN_DEBUG "vx_send_msg: send IRQ_MESSAGE error\n");
+ return err;
+ }
+
+ /* Wait for CHK = 1 */
+ if ((err = vx_wait_isr_bit(chip, ISR_CHK)) < 0)
+ return err;
+
+ /* If error, get error value from RX */
+ if (vx_inb(chip, ISR) & ISR_ERR) {
+ if ((err = vx_wait_for_rx_full(chip)) < 0) {
+ snd_printd(KERN_DEBUG "vx_send_msg: rx_full read error\n");
+ return err;
+ }
+ err = vx_inb(chip, RXH) << 16;
+ err |= vx_inb(chip, RXM) << 8;
+ err |= vx_inb(chip, RXL);
+ snd_printd(KERN_DEBUG "msg got error = 0x%x at cmd[0]\n", err);
+ err = -(VX_ERR_MASK | err);
+ return err;
+ }
+
+ /* Send the other words */
+ if (rmh->LgCmd > 1) {
+ for (i = 1; i < rmh->LgCmd; i++) {
+ /* Wait for TX ready */
+ if ((err = vx_wait_isr_bit(chip, ISR_TX_READY)) < 0) {
+ snd_printd(KERN_DEBUG "vx_send_msg: tx_ready error\n");
+ return err;
+ }
+
+ /* Write Cmd[i] */
+ vx_outb(chip, TXH, (rmh->Cmd[i] >> 16) & 0xff);
+ vx_outb(chip, TXM, (rmh->Cmd[i] >> 8) & 0xff);
+ vx_outb(chip, TXL, rmh->Cmd[i] & 0xff);
+
+ /* Trigger irq MESS_READ_NEXT */
+ if ((err = vx_send_irq_dsp(chip, IRQ_MESS_READ_NEXT)) < 0) {
+ snd_printd(KERN_DEBUG "vx_send_msg: IRQ_READ_NEXT error\n");
+ return err;
+ }
+ }
+ /* Wait for TX empty */
+ if ((err = vx_wait_isr_bit(chip, ISR_TX_READY)) < 0) {
+ snd_printd(KERN_DEBUG "vx_send_msg: TX_READY error\n");
+ return err;
+ }
+ /* End of transfer */
+ err = vx_transfer_end(chip, IRQ_MESS_READ_END);
+ if (err < 0)
+ return err;
+ }
+
+ return vx_read_status(chip, rmh);
+}
+
+
+/*
+ * vx_send_msg - send a DSP message with spinlock
+ * @rmh: the rmh record to send and receive
+ *
+ * returns 0 if successful, or a negative error code.
+ * see vx_send_msg_nolock().
+ */
+int vx_send_msg(struct vx_core *chip, struct vx_rmh *rmh)
+{
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ err = vx_send_msg_nolock(chip, rmh);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return err;
+}
+
+
+/*
+ * vx_send_rih_nolock - send an RIH to xilinx
+ * @cmd: the command to send
+ *
+ * returns 0 if successful, or a negative error code.
+ * the error code can be VX-specific, retrieved via vx_get_error().
+ *
+ * this function doesn't call spinlock at all.
+ *
+ * unlike RMH, no command is sent to DSP.
+ */
+int vx_send_rih_nolock(struct vx_core *chip, int cmd)
+{
+ int err;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+#if 0
+ printk(KERN_DEBUG "send_rih: cmd = 0x%x\n", cmd);
+#endif
+ if ((err = vx_reset_chk(chip)) < 0)
+ return err;
+ /* send the IRQ */
+ if ((err = vx_send_irq_dsp(chip, cmd)) < 0)
+ return err;
+ /* Wait CHK = 1 */
+ if ((err = vx_wait_isr_bit(chip, ISR_CHK)) < 0)
+ return err;
+ /* If error, read RX */
+ if (vx_inb(chip, ISR) & ISR_ERR) {
+ if ((err = vx_wait_for_rx_full(chip)) < 0)
+ return err;
+ err = vx_inb(chip, RXH) << 16;
+ err |= vx_inb(chip, RXM) << 8;
+ err |= vx_inb(chip, RXL);
+ return -(VX_ERR_MASK | err);
+ }
+ return 0;
+}
+
+
+/*
+ * vx_send_rih - send an RIH with spinlock
+ * @cmd: the command to send
+ *
+ * see vx_send_rih_nolock().
+ */
+int vx_send_rih(struct vx_core *chip, int cmd)
+{
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ err = vx_send_rih_nolock(chip, cmd);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return err;
+}
+
+#define END_OF_RESET_WAIT_TIME 500 /* us */
+
+/**
+ * snd_vx_boot_xilinx - boot up the xilinx interface
+ * @boot: the boot record to load
+ */
+int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *boot)
+{
+ unsigned int i;
+ int no_fillup = vx_has_new_dsp(chip);
+
+ /* check the length of boot image */
+ if (boot->size <= 0)
+ return -EINVAL;
+ if (boot->size % 3)
+ return -EINVAL;
+#if 0
+ {
+ /* more strict check */
+ unsigned int c = ((u32)boot->data[0] << 16) | ((u32)boot->data[1] << 8) | boot->data[2];
+ if (boot->size != (c + 2) * 3)
+ return -EINVAL;
+ }
+#endif
+
+ /* reset dsp */
+ vx_reset_dsp(chip);
+
+ udelay(END_OF_RESET_WAIT_TIME); /* another wait? */
+
+ /* download boot strap */
+ for (i = 0; i < 0x600; i += 3) {
+ if (i >= boot->size) {
+ if (no_fillup)
+ break;
+ if (vx_wait_isr_bit(chip, ISR_TX_EMPTY) < 0) {
+ snd_printk(KERN_ERR "dsp boot failed at %d\n", i);
+ return -EIO;
+ }
+ vx_outb(chip, TXH, 0);
+ vx_outb(chip, TXM, 0);
+ vx_outb(chip, TXL, 0);
+ } else {
+ const unsigned char *image = boot->data + i;
+ if (vx_wait_isr_bit(chip, ISR_TX_EMPTY) < 0) {
+ snd_printk(KERN_ERR "dsp boot failed at %d\n", i);
+ return -EIO;
+ }
+ vx_outb(chip, TXH, image[0]);
+ vx_outb(chip, TXM, image[1]);
+ vx_outb(chip, TXL, image[2]);
+ }
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_vx_load_boot_image);
+
+/*
+ * vx_test_irq_src - query the source of interrupts
+ *
+ * called from irq handler only
+ */
+static int vx_test_irq_src(struct vx_core *chip, unsigned int *ret)
+{
+ int err;
+
+ vx_init_rmh(&chip->irq_rmh, CMD_TEST_IT);
+ spin_lock(&chip->lock);
+ err = vx_send_msg_nolock(chip, &chip->irq_rmh);
+ if (err < 0)
+ *ret = 0;
+ else
+ *ret = chip->irq_rmh.Stat[0];
+ spin_unlock(&chip->lock);
+ return err;
+}
+
+
+/*
+ * vx_interrupt - soft irq handler
+ */
+static void vx_interrupt(unsigned long private_data)
+{
+ struct vx_core *chip = (struct vx_core *) private_data;
+ unsigned int events;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return;
+
+ if (vx_test_irq_src(chip, &events) < 0)
+ return;
+
+#if 0
+ if (events & 0x000800)
+ printk(KERN_ERR "DSP Stream underrun ! IRQ events = 0x%x\n", events);
+#endif
+ // printk(KERN_DEBUG "IRQ events = 0x%x\n", events);
+
+ /* We must prevent any application using this DSP
+ * and block any further request until the application
+ * either unregisters or reloads the DSP
+ */
+ if (events & FATAL_DSP_ERROR) {
+ snd_printk(KERN_ERR "vx_core: fatal DSP error!!\n");
+ return;
+ }
+
+ /* The start on time code conditions are filled (ie the time code
+ * received by the board is equal to one of those given to it).
+ */
+ if (events & TIME_CODE_EVENT_PENDING)
+ ; /* so far, nothing to do yet */
+
+ /* The frequency has changed on the board (UER mode). */
+ if (events & FREQUENCY_CHANGE_EVENT_PENDING)
+ vx_change_frequency(chip);
+
+ /* update the pcm streams */
+ vx_pcm_update_intr(chip, events);
+}
+
+
+/**
+ * snd_vx_irq_handler - interrupt handler
+ */
+irqreturn_t snd_vx_irq_handler(int irq, void *dev)
+{
+ struct vx_core *chip = dev;
+
+ if (! (chip->chip_status & VX_STAT_CHIP_INIT) ||
+ (chip->chip_status & VX_STAT_IS_STALE))
+ return IRQ_NONE;
+ if (! vx_test_and_ack(chip))
+ tasklet_hi_schedule(&chip->tq);
+ return IRQ_HANDLED;
+}
+
+EXPORT_SYMBOL(snd_vx_irq_handler);
+
+/*
+ */
+static void vx_reset_board(struct vx_core *chip, int cold_reset)
+{
+ if (snd_BUG_ON(!chip->ops->reset_board))
+ return;
+
+ /* current source, later sync'ed with target */
+ chip->audio_source = VX_AUDIO_SRC_LINE;
+ if (cold_reset) {
+ chip->audio_source_target = chip->audio_source;
+ chip->clock_source = INTERNAL_QUARTZ;
+ chip->clock_mode = VX_CLOCK_MODE_AUTO;
+ chip->freq = 48000;
+ chip->uer_detected = VX_UER_MODE_NOT_PRESENT;
+ chip->uer_bits = SNDRV_PCM_DEFAULT_CON_SPDIF;
+ }
+
+ chip->ops->reset_board(chip, cold_reset);
+
+ vx_reset_codec(chip, cold_reset);
+
+ vx_set_internal_clock(chip, chip->freq);
+
+ /* Reset the DSP */
+ vx_reset_dsp(chip);
+
+ if (vx_is_pcmcia(chip)) {
+ /* Acknowledge any pending IRQ and reset the MEMIRQ flag. */
+ vx_test_and_ack(chip);
+ vx_validate_irq(chip, 1);
+ }
+
+ /* init CBits */
+ vx_set_iec958_status(chip, chip->uer_bits);
+}
+
+
+/*
+ * proc interface
+ */
+
+static void vx_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct vx_core *chip = entry->private_data;
+ static char *audio_src_vxp[] = { "Line", "Mic", "Digital" };
+ static char *audio_src_vx2[] = { "Analog", "Analog", "Digital" };
+ static char *clock_mode[] = { "Auto", "Internal", "External" };
+ static char *clock_src[] = { "Internal", "External" };
+ static char *uer_type[] = { "Consumer", "Professional", "Not Present" };
+
+ snd_iprintf(buffer, "%s\n", chip->card->longname);
+ snd_iprintf(buffer, "Xilinx Firmware: %s\n",
+ chip->chip_status & VX_STAT_XILINX_LOADED ? "Loaded" : "No");
+ snd_iprintf(buffer, "Device Initialized: %s\n",
+ chip->chip_status & VX_STAT_DEVICE_INIT ? "Yes" : "No");
+ snd_iprintf(buffer, "DSP audio info:");
+ if (chip->audio_info & VX_AUDIO_INFO_REAL_TIME)
+ snd_iprintf(buffer, " realtime");
+ if (chip->audio_info & VX_AUDIO_INFO_OFFLINE)
+ snd_iprintf(buffer, " offline");
+ if (chip->audio_info & VX_AUDIO_INFO_MPEG1)
+ snd_iprintf(buffer, " mpeg1");
+ if (chip->audio_info & VX_AUDIO_INFO_MPEG2)
+ snd_iprintf(buffer, " mpeg2");
+ if (chip->audio_info & VX_AUDIO_INFO_LINEAR_8)
+ snd_iprintf(buffer, " linear8");
+ if (chip->audio_info & VX_AUDIO_INFO_LINEAR_16)
+ snd_iprintf(buffer, " linear16");
+ if (chip->audio_info & VX_AUDIO_INFO_LINEAR_24)
+ snd_iprintf(buffer, " linear24");
+ snd_iprintf(buffer, "\n");
+ snd_iprintf(buffer, "Input Source: %s\n", vx_is_pcmcia(chip) ?
+ audio_src_vxp[chip->audio_source] :
+ audio_src_vx2[chip->audio_source]);
+ snd_iprintf(buffer, "Clock Mode: %s\n", clock_mode[chip->clock_mode]);
+ snd_iprintf(buffer, "Clock Source: %s\n", clock_src[chip->clock_source]);
+ snd_iprintf(buffer, "Frequency: %d\n", chip->freq);
+ snd_iprintf(buffer, "Detected Frequency: %d\n", chip->freq_detected);
+ snd_iprintf(buffer, "Detected UER type: %s\n", uer_type[chip->uer_detected]);
+ snd_iprintf(buffer, "Min/Max/Cur IBL: %d/%d/%d (granularity=%d)\n",
+ chip->ibl.min_size, chip->ibl.max_size, chip->ibl.size,
+ chip->ibl.granularity);
+}
+
+static void vx_proc_init(struct vx_core *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "vx-status", &entry))
+ snd_info_set_text_ops(entry, chip, vx_proc_read);
+}
+
+
+/**
+ * snd_vx_dsp_boot - load the DSP boot
+ */
+int snd_vx_dsp_boot(struct vx_core *chip, const struct firmware *boot)
+{
+ int err;
+ int cold_reset = !(chip->chip_status & VX_STAT_DEVICE_INIT);
+
+ vx_reset_board(chip, cold_reset);
+ vx_validate_irq(chip, 0);
+
+ if ((err = snd_vx_load_boot_image(chip, boot)) < 0)
+ return err;
+ msleep(10);
+
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_vx_dsp_boot);
+
+/**
+ * snd_vx_dsp_load - load the DSP image
+ */
+int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp)
+{
+ unsigned int i;
+ int err;
+ unsigned int csum = 0;
+ const unsigned char *image, *cptr;
+
+ if (dsp->size % 3)
+ return -EINVAL;
+
+ vx_toggle_dac_mute(chip, 1);
+
+ /* Transfert data buffer from PC to DSP */
+ for (i = 0; i < dsp->size; i += 3) {
+ image = dsp->data + i;
+ /* Wait DSP ready for a new read */
+ if ((err = vx_wait_isr_bit(chip, ISR_TX_EMPTY)) < 0) {
+ printk("dsp loading error at position %d\n", i);
+ return err;
+ }
+ cptr = image;
+ csum ^= *cptr;
+ csum = (csum >> 24) | (csum << 8);
+ vx_outb(chip, TXH, *cptr++);
+ csum ^= *cptr;
+ csum = (csum >> 24) | (csum << 8);
+ vx_outb(chip, TXM, *cptr++);
+ csum ^= *cptr;
+ csum = (csum >> 24) | (csum << 8);
+ vx_outb(chip, TXL, *cptr++);
+ }
+ snd_printdd(KERN_DEBUG "checksum = 0x%08x\n", csum);
+
+ msleep(200);
+
+ if ((err = vx_wait_isr_bit(chip, ISR_CHK)) < 0)
+ return err;
+
+ vx_toggle_dac_mute(chip, 0);
+
+ vx_test_and_ack(chip);
+ vx_validate_irq(chip, 1);
+
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_vx_dsp_load);
+
+#ifdef CONFIG_PM
+/*
+ * suspend
+ */
+int snd_vx_suspend(struct vx_core *chip, pm_message_t state)
+{
+ unsigned int i;
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ chip->chip_status |= VX_STAT_IN_SUSPEND;
+ for (i = 0; i < chip->hw->num_codecs; i++)
+ snd_pcm_suspend_all(chip->pcm[i]);
+
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_vx_suspend);
+
+/*
+ * resume
+ */
+int snd_vx_resume(struct vx_core *chip)
+{
+ int i, err;
+
+ chip->chip_status &= ~VX_STAT_CHIP_INIT;
+
+ for (i = 0; i < 4; i++) {
+ if (! chip->firmware[i])
+ continue;
+ err = chip->ops->load_dsp(chip, i, chip->firmware[i]);
+ if (err < 0) {
+ snd_printk(KERN_ERR "vx: firmware resume error at DSP %d\n", i);
+ return -EIO;
+ }
+ }
+
+ chip->chip_status |= VX_STAT_CHIP_INIT;
+ chip->chip_status &= ~VX_STAT_IN_SUSPEND;
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_vx_resume);
+#endif
+
+/**
+ * snd_vx_create - constructor for struct vx_core
+ * @hw: hardware specific record
+ *
+ * this function allocates the instance and prepare for the hardware
+ * initialization.
+ *
+ * return the instance pointer if successful, NULL in error.
+ */
+struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw,
+ struct snd_vx_ops *ops,
+ int extra_size)
+{
+ struct vx_core *chip;
+
+ if (snd_BUG_ON(!card || !hw || !ops))
+ return NULL;
+
+ chip = kzalloc(sizeof(*chip) + extra_size, GFP_KERNEL);
+ if (! chip) {
+ snd_printk(KERN_ERR "vx_core: no memory\n");
+ return NULL;
+ }
+ spin_lock_init(&chip->lock);
+ spin_lock_init(&chip->irq_lock);
+ chip->irq = -1;
+ chip->hw = hw;
+ chip->type = hw->type;
+ chip->ops = ops;
+ tasklet_init(&chip->tq, vx_interrupt, (unsigned long)chip);
+ mutex_init(&chip->mixer_mutex);
+
+ chip->card = card;
+ card->private_data = chip;
+ strcpy(card->driver, hw->name);
+ sprintf(card->shortname, "Digigram %s", hw->name);
+
+ vx_proc_init(chip);
+
+ return chip;
+}
+
+EXPORT_SYMBOL(snd_vx_create);
+
+/*
+ * module entries
+ */
+static int __init alsa_vx_core_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_vx_core_exit(void)
+{
+}
+
+module_init(alsa_vx_core_init)
+module_exit(alsa_vx_core_exit)
diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c
new file mode 100644
index 0000000..8d6362e
--- /dev/null
+++ b/sound/drivers/vx/vx_hwdep.c
@@ -0,0 +1,270 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * DSP firmware management
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/device.h>
+#include <linux/firmware.h>
+#include <linux/vmalloc.h>
+#include <sound/core.h>
+#include <sound/hwdep.h>
+#include <sound/vx_core.h>
+
+#ifdef SND_VX_FW_LOADER
+
+MODULE_FIRMWARE("vx/bx_1_vxp.b56");
+MODULE_FIRMWARE("vx/bx_1_vp4.b56");
+MODULE_FIRMWARE("vx/x1_1_vx2.xlx");
+MODULE_FIRMWARE("vx/x1_2_v22.xlx");
+MODULE_FIRMWARE("vx/x1_1_vxp.xlx");
+MODULE_FIRMWARE("vx/x1_1_vp4.xlx");
+MODULE_FIRMWARE("vx/bd56002.boot");
+MODULE_FIRMWARE("vx/bd563v2.boot");
+MODULE_FIRMWARE("vx/bd563s3.boot");
+MODULE_FIRMWARE("vx/l_1_vx2.d56");
+MODULE_FIRMWARE("vx/l_1_v22.d56");
+MODULE_FIRMWARE("vx/l_1_vxp.d56");
+MODULE_FIRMWARE("vx/l_1_vp4.d56");
+
+int snd_vx_setup_firmware(struct vx_core *chip)
+{
+ static char *fw_files[VX_TYPE_NUMS][4] = {
+ [VX_TYPE_BOARD] = {
+ NULL, "x1_1_vx2.xlx", "bd56002.boot", "l_1_vx2.d56",
+ },
+ [VX_TYPE_V2] = {
+ NULL, "x1_2_v22.xlx", "bd563v2.boot", "l_1_v22.d56",
+ },
+ [VX_TYPE_MIC] = {
+ NULL, "x1_2_v22.xlx", "bd563v2.boot", "l_1_v22.d56",
+ },
+ [VX_TYPE_VXPOCKET] = {
+ "bx_1_vxp.b56", "x1_1_vxp.xlx", "bd563s3.boot", "l_1_vxp.d56"
+ },
+ [VX_TYPE_VXP440] = {
+ "bx_1_vp4.b56", "x1_1_vp4.xlx", "bd563s3.boot", "l_1_vp4.d56"
+ },
+ };
+
+ int i, err;
+
+ for (i = 0; i < 4; i++) {
+ char path[32];
+ const struct firmware *fw;
+ if (! fw_files[chip->type][i])
+ continue;
+ sprintf(path, "vx/%s", fw_files[chip->type][i]);
+ if (request_firmware(&fw, path, chip->dev)) {
+ snd_printk(KERN_ERR "vx: can't load firmware %s\n", path);
+ return -ENOENT;
+ }
+ err = chip->ops->load_dsp(chip, i, fw);
+ if (err < 0) {
+ release_firmware(fw);
+ return err;
+ }
+ if (i == 1)
+ chip->chip_status |= VX_STAT_XILINX_LOADED;
+#ifdef CONFIG_PM
+ chip->firmware[i] = fw;
+#else
+ release_firmware(fw);
+#endif
+ }
+
+ /* ok, we reached to the last one */
+ /* create the devices if not built yet */
+ if ((err = snd_vx_pcm_new(chip)) < 0)
+ return err;
+
+ if ((err = snd_vx_mixer_new(chip)) < 0)
+ return err;
+
+ if (chip->ops->add_controls)
+ if ((err = chip->ops->add_controls(chip)) < 0)
+ return err;
+
+ chip->chip_status |= VX_STAT_DEVICE_INIT;
+ chip->chip_status |= VX_STAT_CHIP_INIT;
+
+ return snd_card_register(chip->card);
+}
+
+/* exported */
+void snd_vx_free_firmware(struct vx_core *chip)
+{
+#ifdef CONFIG_PM
+ int i;
+ for (i = 0; i < 4; i++)
+ release_firmware(chip->firmware[i]);
+#endif
+}
+
+#else /* old style firmware loading */
+
+static int vx_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+static int vx_hwdep_release(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+static int vx_hwdep_dsp_status(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_status *info)
+{
+ static char *type_ids[VX_TYPE_NUMS] = {
+ [VX_TYPE_BOARD] = "vxboard",
+ [VX_TYPE_V2] = "vx222",
+ [VX_TYPE_MIC] = "vx222",
+ [VX_TYPE_VXPOCKET] = "vxpocket",
+ [VX_TYPE_VXP440] = "vxp440",
+ };
+ struct vx_core *vx = hw->private_data;
+
+ if (snd_BUG_ON(!type_ids[vx->type]))
+ return -EINVAL;
+ strcpy(info->id, type_ids[vx->type]);
+ if (vx_is_pcmcia(vx))
+ info->num_dsps = 4;
+ else
+ info->num_dsps = 3;
+ if (vx->chip_status & VX_STAT_CHIP_INIT)
+ info->chip_ready = 1;
+ info->version = VX_DRIVER_VERSION;
+ return 0;
+}
+
+static void free_fw(const struct firmware *fw)
+{
+ if (fw) {
+ vfree(fw->data);
+ kfree(fw);
+ }
+}
+
+static int vx_hwdep_dsp_load(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_image *dsp)
+{
+ struct vx_core *vx = hw->private_data;
+ int index, err;
+ struct firmware *fw;
+
+ if (snd_BUG_ON(!vx->ops->load_dsp))
+ return -ENXIO;
+
+ fw = kmalloc(sizeof(*fw), GFP_KERNEL);
+ if (! fw) {
+ snd_printk(KERN_ERR "cannot allocate firmware\n");
+ return -ENOMEM;
+ }
+ fw->size = dsp->length;
+ fw->data = vmalloc(fw->size);
+ if (! fw->data) {
+ snd_printk(KERN_ERR "cannot allocate firmware image (length=%d)\n",
+ (int)fw->size);
+ kfree(fw);
+ return -ENOMEM;
+ }
+ if (copy_from_user((void *)fw->data, dsp->image, dsp->length)) {
+ free_fw(fw);
+ return -EFAULT;
+ }
+
+ index = dsp->index;
+ if (! vx_is_pcmcia(vx))
+ index++;
+ err = vx->ops->load_dsp(vx, index, fw);
+ if (err < 0) {
+ free_fw(fw);
+ return err;
+ }
+#ifdef CONFIG_PM
+ vx->firmware[index] = fw;
+#else
+ free_fw(fw);
+#endif
+
+ if (index == 1)
+ vx->chip_status |= VX_STAT_XILINX_LOADED;
+ if (index < 3)
+ return 0;
+
+ /* ok, we reached to the last one */
+ /* create the devices if not built yet */
+ if (! (vx->chip_status & VX_STAT_DEVICE_INIT)) {
+ if ((err = snd_vx_pcm_new(vx)) < 0)
+ return err;
+
+ if ((err = snd_vx_mixer_new(vx)) < 0)
+ return err;
+
+ if (vx->ops->add_controls)
+ if ((err = vx->ops->add_controls(vx)) < 0)
+ return err;
+
+ if ((err = snd_card_register(vx->card)) < 0)
+ return err;
+
+ vx->chip_status |= VX_STAT_DEVICE_INIT;
+ }
+ vx->chip_status |= VX_STAT_CHIP_INIT;
+ return 0;
+}
+
+
+/* exported */
+int snd_vx_setup_firmware(struct vx_core *chip)
+{
+ int err;
+ struct snd_hwdep *hw;
+
+ if ((err = snd_hwdep_new(chip->card, SND_VX_HWDEP_ID, 0, &hw)) < 0)
+ return err;
+
+ hw->iface = SNDRV_HWDEP_IFACE_VX;
+ hw->private_data = chip;
+ hw->ops.open = vx_hwdep_open;
+ hw->ops.release = vx_hwdep_release;
+ hw->ops.dsp_status = vx_hwdep_dsp_status;
+ hw->ops.dsp_load = vx_hwdep_dsp_load;
+ hw->exclusive = 1;
+ sprintf(hw->name, "VX Loader (%s)", chip->card->driver);
+ chip->hwdep = hw;
+
+ return snd_card_register(chip->card);
+}
+
+/* exported */
+void snd_vx_free_firmware(struct vx_core *chip)
+{
+#ifdef CONFIG_PM
+ int i;
+ for (i = 0; i < 4; i++)
+ free_fw(chip->firmware[i]);
+#endif
+}
+
+#endif /* SND_VX_FW_LOADER */
+
+EXPORT_SYMBOL(snd_vx_setup_firmware);
+EXPORT_SYMBOL(snd_vx_free_firmware);
diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c
new file mode 100644
index 0000000..c71b8d1
--- /dev/null
+++ b/sound/drivers/vx/vx_mixer.c
@@ -0,0 +1,1028 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * Common mixer part
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/vx_core.h>
+#include "vx_cmd.h"
+
+
+/*
+ * write a codec data (24bit)
+ */
+static void vx_write_codec_reg(struct vx_core *chip, int codec, unsigned int data)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!chip->ops->write_codec))
+ return;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->ops->write_codec(chip, codec, data);
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+/*
+ * Data type used to access the Codec
+ */
+union vx_codec_data {
+ u32 l;
+#ifdef SNDRV_BIG_ENDIAN
+ struct w {
+ u16 h;
+ u16 l;
+ } w;
+ struct b {
+ u8 hh;
+ u8 mh;
+ u8 ml;
+ u8 ll;
+ } b;
+#else /* LITTLE_ENDIAN */
+ struct w {
+ u16 l;
+ u16 h;
+ } w;
+ struct b {
+ u8 ll;
+ u8 ml;
+ u8 mh;
+ u8 hh;
+ } b;
+#endif
+};
+
+#define SET_CDC_DATA_SEL(di,s) ((di).b.mh = (u8) (s))
+#define SET_CDC_DATA_REG(di,r) ((di).b.ml = (u8) (r))
+#define SET_CDC_DATA_VAL(di,d) ((di).b.ll = (u8) (d))
+#define SET_CDC_DATA_INIT(di) ((di).l = 0L, SET_CDC_DATA_SEL(di,XX_CODEC_SELECTOR))
+
+/*
+ * set up codec register and write the value
+ * @codec: the codec id, 0 or 1
+ * @reg: register index
+ * @val: data value
+ */
+static void vx_set_codec_reg(struct vx_core *chip, int codec, int reg, int val)
+{
+ union vx_codec_data data;
+ /* DAC control register */
+ SET_CDC_DATA_INIT(data);
+ SET_CDC_DATA_REG(data, reg);
+ SET_CDC_DATA_VAL(data, val);
+ vx_write_codec_reg(chip, codec, data.l);
+}
+
+
+/*
+ * vx_set_analog_output_level - set the output attenuation level
+ * @codec: the output codec, 0 or 1. (1 for VXP440 only)
+ * @left: left output level, 0 = mute
+ * @right: right output level
+ */
+static void vx_set_analog_output_level(struct vx_core *chip, int codec, int left, int right)
+{
+ left = chip->hw->output_level_max - left;
+ right = chip->hw->output_level_max - right;
+
+ if (chip->ops->akm_write) {
+ chip->ops->akm_write(chip, XX_CODEC_LEVEL_LEFT_REGISTER, left);
+ chip->ops->akm_write(chip, XX_CODEC_LEVEL_RIGHT_REGISTER, right);
+ } else {
+ /* convert to attenuation level: 0 = 0dB (max), 0xe3 = -113.5 dB (min) */
+ vx_set_codec_reg(chip, codec, XX_CODEC_LEVEL_LEFT_REGISTER, left);
+ vx_set_codec_reg(chip, codec, XX_CODEC_LEVEL_RIGHT_REGISTER, right);
+ }
+}
+
+
+/*
+ * vx_toggle_dac_mute - mute/unmute DAC
+ * @mute: 0 = unmute, 1 = mute
+ */
+
+#define DAC_ATTEN_MIN 0x08
+#define DAC_ATTEN_MAX 0x38
+
+void vx_toggle_dac_mute(struct vx_core *chip, int mute)
+{
+ unsigned int i;
+ for (i = 0; i < chip->hw->num_codecs; i++) {
+ if (chip->ops->akm_write)
+ chip->ops->akm_write(chip, XX_CODEC_DAC_CONTROL_REGISTER, mute); /* XXX */
+ else
+ vx_set_codec_reg(chip, i, XX_CODEC_DAC_CONTROL_REGISTER,
+ mute ? DAC_ATTEN_MAX : DAC_ATTEN_MIN);
+ }
+}
+
+/*
+ * vx_reset_codec - reset and initialize the codecs
+ */
+void vx_reset_codec(struct vx_core *chip, int cold_reset)
+{
+ unsigned int i;
+ int port = chip->type >= VX_TYPE_VXPOCKET ? 0x75 : 0x65;
+
+ chip->ops->reset_codec(chip);
+
+ /* AKM codecs should be initialized in reset_codec callback */
+ if (! chip->ops->akm_write) {
+ /* initialize old codecs */
+ for (i = 0; i < chip->hw->num_codecs; i++) {
+ /* DAC control register (change level when zero crossing + mute) */
+ vx_set_codec_reg(chip, i, XX_CODEC_DAC_CONTROL_REGISTER, DAC_ATTEN_MAX);
+ /* ADC control register */
+ vx_set_codec_reg(chip, i, XX_CODEC_ADC_CONTROL_REGISTER, 0x00);
+ /* Port mode register */
+ vx_set_codec_reg(chip, i, XX_CODEC_PORT_MODE_REGISTER, port);
+ /* Clock control register */
+ vx_set_codec_reg(chip, i, XX_CODEC_CLOCK_CONTROL_REGISTER, 0x00);
+ }
+ }
+
+ /* mute analog output */
+ for (i = 0; i < chip->hw->num_codecs; i++) {
+ chip->output_level[i][0] = 0;
+ chip->output_level[i][1] = 0;
+ vx_set_analog_output_level(chip, i, 0, 0);
+ }
+}
+
+/*
+ * change the audio input source
+ * @src: the target source (VX_AUDIO_SRC_XXX)
+ */
+static void vx_change_audio_source(struct vx_core *chip, int src)
+{
+ unsigned long flags;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->ops->change_audio_source(chip, src);
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+/*
+ * change the audio source if necessary and possible
+ * returns 1 if the source is actually changed.
+ */
+int vx_sync_audio_source(struct vx_core *chip)
+{
+ if (chip->audio_source_target == chip->audio_source ||
+ chip->pcm_running)
+ return 0;
+ vx_change_audio_source(chip, chip->audio_source_target);
+ chip->audio_source = chip->audio_source_target;
+ return 1;
+}
+
+
+/*
+ * audio level, mute, monitoring
+ */
+struct vx_audio_level {
+ unsigned int has_level: 1;
+ unsigned int has_monitor_level: 1;
+ unsigned int has_mute: 1;
+ unsigned int has_monitor_mute: 1;
+ unsigned int mute;
+ unsigned int monitor_mute;
+ short level;
+ short monitor_level;
+};
+
+static int vx_adjust_audio_level(struct vx_core *chip, int audio, int capture,
+ struct vx_audio_level *info)
+{
+ struct vx_rmh rmh;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+ vx_init_rmh(&rmh, CMD_AUDIO_LEVEL_ADJUST);
+ if (capture)
+ rmh.Cmd[0] |= COMMAND_RECORD_MASK;
+ /* Add Audio IO mask */
+ rmh.Cmd[1] = 1 << audio;
+ rmh.Cmd[2] = 0;
+ if (info->has_level) {
+ rmh.Cmd[0] |= VALID_AUDIO_IO_DIGITAL_LEVEL;
+ rmh.Cmd[2] |= info->level;
+ }
+ if (info->has_monitor_level) {
+ rmh.Cmd[0] |= VALID_AUDIO_IO_MONITORING_LEVEL;
+ rmh.Cmd[2] |= ((unsigned int)info->monitor_level << 10);
+ }
+ if (info->has_mute) {
+ rmh.Cmd[0] |= VALID_AUDIO_IO_MUTE_LEVEL;
+ if (info->mute)
+ rmh.Cmd[2] |= AUDIO_IO_HAS_MUTE_LEVEL;
+ }
+ if (info->has_monitor_mute) {
+ /* validate flag for M2 at least to unmute it */
+ rmh.Cmd[0] |= VALID_AUDIO_IO_MUTE_MONITORING_1 | VALID_AUDIO_IO_MUTE_MONITORING_2;
+ if (info->monitor_mute)
+ rmh.Cmd[2] |= AUDIO_IO_HAS_MUTE_MONITORING_1;
+ }
+
+ return vx_send_msg(chip, &rmh);
+}
+
+
+#if 0 // not used
+static int vx_read_audio_level(struct vx_core *chip, int audio, int capture,
+ struct vx_audio_level *info)
+{
+ int err;
+ struct vx_rmh rmh;
+
+ memset(info, 0, sizeof(*info));
+ vx_init_rmh(&rmh, CMD_GET_AUDIO_LEVELS);
+ if (capture)
+ rmh.Cmd[0] |= COMMAND_RECORD_MASK;
+ /* Add Audio IO mask */
+ rmh.Cmd[1] = 1 << audio;
+ err = vx_send_msg(chip, &rmh);
+ if (err < 0)
+ return err;
+ info.level = rmh.Stat[0] & MASK_DSP_WORD_LEVEL;
+ info.monitor_level = (rmh.Stat[0] >> 10) & MASK_DSP_WORD_LEVEL;
+ info.mute = (rmh.Stat[i] & AUDIO_IO_HAS_MUTE_LEVEL) ? 1 : 0;
+ info.monitor_mute = (rmh.Stat[i] & AUDIO_IO_HAS_MUTE_MONITORING_1) ? 1 : 0;
+ return 0;
+}
+#endif // not used
+
+/*
+ * set the monitoring level and mute state of the given audio
+ * no more static, because must be called from vx_pcm to demute monitoring
+ */
+int vx_set_monitor_level(struct vx_core *chip, int audio, int level, int active)
+{
+ struct vx_audio_level info;
+
+ memset(&info, 0, sizeof(info));
+ info.has_monitor_level = 1;
+ info.monitor_level = level;
+ info.has_monitor_mute = 1;
+ info.monitor_mute = !active;
+ chip->audio_monitor[audio] = level;
+ chip->audio_monitor_active[audio] = active;
+ return vx_adjust_audio_level(chip, audio, 0, &info); /* playback only */
+}
+
+
+/*
+ * set the mute status of the given audio
+ */
+static int vx_set_audio_switch(struct vx_core *chip, int audio, int active)
+{
+ struct vx_audio_level info;
+
+ memset(&info, 0, sizeof(info));
+ info.has_mute = 1;
+ info.mute = !active;
+ chip->audio_active[audio] = active;
+ return vx_adjust_audio_level(chip, audio, 0, &info); /* playback only */
+}
+
+/*
+ * set the mute status of the given audio
+ */
+static int vx_set_audio_gain(struct vx_core *chip, int audio, int capture, int level)
+{
+ struct vx_audio_level info;
+
+ memset(&info, 0, sizeof(info));
+ info.has_level = 1;
+ info.level = level;
+ chip->audio_gain[capture][audio] = level;
+ return vx_adjust_audio_level(chip, audio, capture, &info);
+}
+
+/*
+ * reset all audio levels
+ */
+static void vx_reset_audio_levels(struct vx_core *chip)
+{
+ unsigned int i, c;
+ struct vx_audio_level info;
+
+ memset(chip->audio_gain, 0, sizeof(chip->audio_gain));
+ memset(chip->audio_active, 0, sizeof(chip->audio_active));
+ memset(chip->audio_monitor, 0, sizeof(chip->audio_monitor));
+ memset(chip->audio_monitor_active, 0, sizeof(chip->audio_monitor_active));
+
+ for (c = 0; c < 2; c++) {
+ for (i = 0; i < chip->hw->num_ins * 2; i++) {
+ memset(&info, 0, sizeof(info));
+ if (c == 0) {
+ info.has_monitor_level = 1;
+ info.has_mute = 1;
+ info.has_monitor_mute = 1;
+ }
+ info.has_level = 1;
+ info.level = CVAL_0DB; /* default: 0dB */
+ vx_adjust_audio_level(chip, i, c, &info);
+ chip->audio_gain[c][i] = CVAL_0DB;
+ chip->audio_monitor[i] = CVAL_0DB;
+ }
+ }
+}
+
+
+/*
+ * VU, peak meter record
+ */
+
+#define VU_METER_CHANNELS 2
+
+struct vx_vu_meter {
+ int saturated;
+ int vu_level;
+ int peak_level;
+};
+
+/*
+ * get the VU and peak meter values
+ * @audio: the audio index
+ * @capture: 0 = playback, 1 = capture operation
+ * @info: the array of vx_vu_meter records (size = 2).
+ */
+static int vx_get_audio_vu_meter(struct vx_core *chip, int audio, int capture, struct vx_vu_meter *info)
+{
+ struct vx_rmh rmh;
+ int i, err;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+ vx_init_rmh(&rmh, CMD_AUDIO_VU_PIC_METER);
+ rmh.LgStat += 2 * VU_METER_CHANNELS;
+ if (capture)
+ rmh.Cmd[0] |= COMMAND_RECORD_MASK;
+
+ /* Add Audio IO mask */
+ rmh.Cmd[1] = 0;
+ for (i = 0; i < VU_METER_CHANNELS; i++)
+ rmh.Cmd[1] |= 1 << (audio + i);
+ err = vx_send_msg(chip, &rmh);
+ if (err < 0)
+ return err;
+ /* Read response */
+ for (i = 0; i < 2 * VU_METER_CHANNELS; i +=2) {
+ info->saturated = (rmh.Stat[0] & (1 << (audio + i))) ? 1 : 0;
+ info->vu_level = rmh.Stat[i + 1];
+ info->peak_level = rmh.Stat[i + 2];
+ info++;
+ }
+ return 0;
+}
+
+
+/*
+ * control API entries
+ */
+
+/*
+ * output level control
+ */
+static int vx_output_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = chip->hw->output_level_max;
+ return 0;
+}
+
+static int vx_output_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int codec = kcontrol->id.index;
+ mutex_lock(&chip->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->output_level[codec][0];
+ ucontrol->value.integer.value[1] = chip->output_level[codec][1];
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_output_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int codec = kcontrol->id.index;
+ unsigned int val[2], vmax;
+
+ vmax = chip->hw->output_level_max;
+ val[0] = ucontrol->value.integer.value[0];
+ val[1] = ucontrol->value.integer.value[1];
+ if (val[0] > vmax || val[1] > vmax)
+ return -EINVAL;
+ mutex_lock(&chip->mixer_mutex);
+ if (val[0] != chip->output_level[codec][0] ||
+ val[1] != chip->output_level[codec][1]) {
+ vx_set_analog_output_level(chip, codec, val[0], val[1]);
+ chip->output_level[codec][0] = val[0];
+ chip->output_level[codec][1] = val[1];
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static struct snd_kcontrol_new vx_control_output_level = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Master Playback Volume",
+ .info = vx_output_level_info,
+ .get = vx_output_level_get,
+ .put = vx_output_level_put,
+ /* tlv will be filled later */
+};
+
+/*
+ * audio source select
+ */
+static int vx_audio_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts_mic[3] = {
+ "Digital", "Line", "Mic"
+ };
+ static char *texts_vx2[2] = {
+ "Digital", "Analog"
+ };
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ if (chip->type >= VX_TYPE_VXPOCKET) {
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name,
+ texts_mic[uinfo->value.enumerated.item]);
+ } else {
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts_vx2[uinfo->value.enumerated.item]);
+ }
+ return 0;
+}
+
+static int vx_audio_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = chip->audio_source_target;
+ return 0;
+}
+
+static int vx_audio_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+
+ if (chip->type >= VX_TYPE_VXPOCKET) {
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ } else {
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ }
+ mutex_lock(&chip->mixer_mutex);
+ if (chip->audio_source_target != ucontrol->value.enumerated.item[0]) {
+ chip->audio_source_target = ucontrol->value.enumerated.item[0];
+ vx_sync_audio_source(chip);
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static struct snd_kcontrol_new vx_control_audio_src = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = vx_audio_src_info,
+ .get = vx_audio_src_get,
+ .put = vx_audio_src_put,
+};
+
+/*
+ * clock mode selection
+ */
+static int vx_clock_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {
+ "Auto", "Internal", "External"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int vx_clock_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = chip->clock_mode;
+ return 0;
+}
+
+static int vx_clock_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ mutex_lock(&chip->mixer_mutex);
+ if (chip->clock_mode != ucontrol->value.enumerated.item[0]) {
+ chip->clock_mode = ucontrol->value.enumerated.item[0];
+ vx_set_clock(chip, chip->freq);
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static struct snd_kcontrol_new vx_control_clock_mode = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Clock Mode",
+ .info = vx_clock_mode_info,
+ .get = vx_clock_mode_get,
+ .put = vx_clock_mode_put,
+};
+
+/*
+ * Audio Gain
+ */
+static int vx_audio_gain_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = CVAL_MAX;
+ return 0;
+}
+
+static int vx_audio_gain_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+ int capture = (kcontrol->private_value >> 8) & 1;
+
+ mutex_lock(&chip->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->audio_gain[capture][audio];
+ ucontrol->value.integer.value[1] = chip->audio_gain[capture][audio+1];
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_audio_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+ int capture = (kcontrol->private_value >> 8) & 1;
+ unsigned int val[2];
+
+ val[0] = ucontrol->value.integer.value[0];
+ val[1] = ucontrol->value.integer.value[1];
+ if (val[0] > CVAL_MAX || val[1] > CVAL_MAX)
+ return -EINVAL;
+ mutex_lock(&chip->mixer_mutex);
+ if (val[0] != chip->audio_gain[capture][audio] ||
+ val[1] != chip->audio_gain[capture][audio+1]) {
+ vx_set_audio_gain(chip, audio, capture, val[0]);
+ vx_set_audio_gain(chip, audio+1, capture, val[1]);
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_audio_monitor_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+
+ mutex_lock(&chip->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->audio_monitor[audio];
+ ucontrol->value.integer.value[1] = chip->audio_monitor[audio+1];
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_audio_monitor_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+ unsigned int val[2];
+
+ val[0] = ucontrol->value.integer.value[0];
+ val[1] = ucontrol->value.integer.value[1];
+ if (val[0] > CVAL_MAX || val[1] > CVAL_MAX)
+ return -EINVAL;
+
+ mutex_lock(&chip->mixer_mutex);
+ if (val[0] != chip->audio_monitor[audio] ||
+ val[1] != chip->audio_monitor[audio+1]) {
+ vx_set_monitor_level(chip, audio, val[0],
+ chip->audio_monitor_active[audio]);
+ vx_set_monitor_level(chip, audio+1, val[1],
+ chip->audio_monitor_active[audio+1]);
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+#define vx_audio_sw_info snd_ctl_boolean_stereo_info
+
+static int vx_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+
+ mutex_lock(&chip->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->audio_active[audio];
+ ucontrol->value.integer.value[1] = chip->audio_active[audio+1];
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+
+ mutex_lock(&chip->mixer_mutex);
+ if (ucontrol->value.integer.value[0] != chip->audio_active[audio] ||
+ ucontrol->value.integer.value[1] != chip->audio_active[audio+1]) {
+ vx_set_audio_switch(chip, audio,
+ !!ucontrol->value.integer.value[0]);
+ vx_set_audio_switch(chip, audio+1,
+ !!ucontrol->value.integer.value[1]);
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_monitor_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+
+ mutex_lock(&chip->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->audio_monitor_active[audio];
+ ucontrol->value.integer.value[1] = chip->audio_monitor_active[audio+1];
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ int audio = kcontrol->private_value & 0xff;
+
+ mutex_lock(&chip->mixer_mutex);
+ if (ucontrol->value.integer.value[0] != chip->audio_monitor_active[audio] ||
+ ucontrol->value.integer.value[1] != chip->audio_monitor_active[audio+1]) {
+ vx_set_monitor_level(chip, audio, chip->audio_monitor[audio],
+ !!ucontrol->value.integer.value[0]);
+ vx_set_monitor_level(chip, audio+1, chip->audio_monitor[audio+1],
+ !!ucontrol->value.integer.value[1]);
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_audio_gain, -10975, 25, 0);
+
+static struct snd_kcontrol_new vx_control_audio_gain = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ /* name will be filled later */
+ .info = vx_audio_gain_info,
+ .get = vx_audio_gain_get,
+ .put = vx_audio_gain_put,
+ .tlv = { .p = db_scale_audio_gain },
+};
+static struct snd_kcontrol_new vx_control_output_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .info = vx_audio_sw_info,
+ .get = vx_audio_sw_get,
+ .put = vx_audio_sw_put
+};
+static struct snd_kcontrol_new vx_control_monitor_gain = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitoring Volume",
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .info = vx_audio_gain_info, /* shared */
+ .get = vx_audio_monitor_get,
+ .put = vx_audio_monitor_put,
+ .tlv = { .p = db_scale_audio_gain },
+};
+static struct snd_kcontrol_new vx_control_monitor_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitoring Switch",
+ .info = vx_audio_sw_info, /* shared */
+ .get = vx_monitor_sw_get,
+ .put = vx_monitor_sw_put
+};
+
+
+/*
+ * IEC958 status bits
+ */
+static int vx_iec958_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int vx_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&chip->mixer_mutex);
+ ucontrol->value.iec958.status[0] = (chip->uer_bits >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (chip->uer_bits >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (chip->uer_bits >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (chip->uer_bits >> 24) & 0xff;
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_iec958_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static int vx_iec958_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+
+ val = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+ mutex_lock(&chip->mixer_mutex);
+ if (chip->uer_bits != val) {
+ chip->uer_bits = val;
+ vx_set_iec958_status(chip, val);
+ mutex_unlock(&chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&chip->mixer_mutex);
+ return 0;
+}
+
+static struct snd_kcontrol_new vx_control_iec958_mask = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .info = vx_iec958_info, /* shared */
+ .get = vx_iec958_mask_get,
+};
+
+static struct snd_kcontrol_new vx_control_iec958 = {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = vx_iec958_info,
+ .get = vx_iec958_get,
+ .put = vx_iec958_put
+};
+
+
+/*
+ * VU meter
+ */
+
+#define METER_MAX 0xff
+#define METER_SHIFT 16
+
+static int vx_vu_meter_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = METER_MAX;
+ return 0;
+}
+
+static int vx_vu_meter_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ struct vx_vu_meter meter[2];
+ int audio = kcontrol->private_value & 0xff;
+ int capture = (kcontrol->private_value >> 8) & 1;
+
+ vx_get_audio_vu_meter(chip, audio, capture, meter);
+ ucontrol->value.integer.value[0] = meter[0].vu_level >> METER_SHIFT;
+ ucontrol->value.integer.value[1] = meter[1].vu_level >> METER_SHIFT;
+ return 0;
+}
+
+static int vx_peak_meter_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ struct vx_vu_meter meter[2];
+ int audio = kcontrol->private_value & 0xff;
+ int capture = (kcontrol->private_value >> 8) & 1;
+
+ vx_get_audio_vu_meter(chip, audio, capture, meter);
+ ucontrol->value.integer.value[0] = meter[0].peak_level >> METER_SHIFT;
+ ucontrol->value.integer.value[1] = meter[1].peak_level >> METER_SHIFT;
+ return 0;
+}
+
+#define vx_saturation_info snd_ctl_boolean_stereo_info
+
+static int vx_saturation_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *chip = snd_kcontrol_chip(kcontrol);
+ struct vx_vu_meter meter[2];
+ int audio = kcontrol->private_value & 0xff;
+
+ vx_get_audio_vu_meter(chip, audio, 1, meter); /* capture only */
+ ucontrol->value.integer.value[0] = meter[0].saturated;
+ ucontrol->value.integer.value[1] = meter[1].saturated;
+ return 0;
+}
+
+static struct snd_kcontrol_new vx_control_vu_meter = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ /* name will be filled later */
+ .info = vx_vu_meter_info,
+ .get = vx_vu_meter_get,
+};
+
+static struct snd_kcontrol_new vx_control_peak_meter = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ /* name will be filled later */
+ .info = vx_vu_meter_info, /* shared */
+ .get = vx_peak_meter_get,
+};
+
+static struct snd_kcontrol_new vx_control_saturation = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Saturation",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = vx_saturation_info,
+ .get = vx_saturation_get,
+};
+
+
+
+/*
+ *
+ */
+
+int snd_vx_mixer_new(struct vx_core *chip)
+{
+ unsigned int i, c;
+ int err;
+ struct snd_kcontrol_new temp;
+ struct snd_card *card = chip->card;
+ char name[32];
+
+ strcpy(card->mixername, card->driver);
+
+ /* output level controls */
+ for (i = 0; i < chip->hw->num_outs; i++) {
+ temp = vx_control_output_level;
+ temp.index = i;
+ temp.tlv.p = chip->hw->output_level_db_scale;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+
+ /* PCM volumes, switches, monitoring */
+ for (i = 0; i < chip->hw->num_outs; i++) {
+ int val = i * 2;
+ temp = vx_control_audio_gain;
+ temp.index = i;
+ temp.name = "PCM Playback Volume";
+ temp.private_value = val;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ temp = vx_control_output_switch;
+ temp.index = i;
+ temp.private_value = val;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ temp = vx_control_monitor_gain;
+ temp.index = i;
+ temp.private_value = val;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ temp = vx_control_monitor_switch;
+ temp.index = i;
+ temp.private_value = val;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+ for (i = 0; i < chip->hw->num_outs; i++) {
+ temp = vx_control_audio_gain;
+ temp.index = i;
+ temp.name = "PCM Capture Volume";
+ temp.private_value = (i * 2) | (1 << 8);
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+
+ /* Audio source */
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&vx_control_audio_src, chip))) < 0)
+ return err;
+ /* clock mode */
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&vx_control_clock_mode, chip))) < 0)
+ return err;
+ /* IEC958 controls */
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&vx_control_iec958_mask, chip))) < 0)
+ return err;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&vx_control_iec958, chip))) < 0)
+ return err;
+ /* VU, peak, saturation meters */
+ for (c = 0; c < 2; c++) {
+ static char *dir[2] = { "Output", "Input" };
+ for (i = 0; i < chip->hw->num_ins; i++) {
+ int val = (i * 2) | (c << 8);
+ if (c == 1) {
+ temp = vx_control_saturation;
+ temp.index = i;
+ temp.private_value = val;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+ sprintf(name, "%s VU Meter", dir[c]);
+ temp = vx_control_vu_meter;
+ temp.index = i;
+ temp.name = name;
+ temp.private_value = val;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ sprintf(name, "%s Peak Meter", dir[c]);
+ temp = vx_control_peak_meter;
+ temp.index = i;
+ temp.name = name;
+ temp.private_value = val;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+ }
+ vx_reset_audio_levels(chip);
+ return 0;
+}
diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
new file mode 100644
index 0000000..27de574
--- /dev/null
+++ b/sound/drivers/vx/vx_pcm.c
@@ -0,0 +1,1330 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * PCM part
+ *
+ * Copyright (c) 2002,2003 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ * STRATEGY
+ * for playback, we send series of "chunks", which size is equal with the
+ * IBL size, typically 126 samples. at each end of chunk, the end-of-buffer
+ * interrupt is notified, and the interrupt handler will feed the next chunk.
+ *
+ * the current position is calculated from the sample count RMH.
+ * pipe->transferred is the counter of data which has been already transferred.
+ * if this counter reaches to the period size, snd_pcm_period_elapsed() will
+ * be issued.
+ *
+ * for capture, the situation is much easier.
+ * to get a low latency response, we'll check the capture streams at each
+ * interrupt (capture stream has no EOB notification). if the pending
+ * data is accumulated to the period size, snd_pcm_period_elapsed() is
+ * called and the pointer is updated.
+ *
+ * the current point of read buffer is kept in pipe->hw_ptr. note that
+ * this is in bytes.
+ *
+ *
+ * TODO
+ * - linked trigger for full-duplex mode.
+ * - scheduled action on the stream.
+ */
+
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/asoundef.h>
+#include <sound/pcm.h>
+#include <sound/vx_core.h>
+#include "vx_cmd.h"
+
+
+/*
+ * we use a vmalloc'ed (sg-)buffer
+ */
+
+/* get the physical page pointer on the given offset */
+static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs,
+ unsigned long offset)
+{
+ void *pageptr = subs->runtime->dma_area + offset;
+ return vmalloc_to_page(pageptr);
+}
+
+/*
+ * allocate a buffer via vmalloc_32().
+ * called from hw_params
+ * NOTE: this may be called not only once per pcm open!
+ */
+static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ if (runtime->dma_area) {
+ /* already allocated */
+ if (runtime->dma_bytes >= size)
+ return 0; /* already enough large */
+ vfree(runtime->dma_area);
+ }
+ runtime->dma_area = vmalloc_32(size);
+ if (! runtime->dma_area)
+ return -ENOMEM;
+ memset(runtime->dma_area, 0, size);
+ runtime->dma_bytes = size;
+ return 1; /* changed */
+}
+
+/*
+ * free the buffer.
+ * called from hw_free callback
+ * NOTE: this may be called not only once per pcm open!
+ */
+static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ vfree(runtime->dma_area);
+ runtime->dma_area = NULL;
+ return 0;
+}
+
+
+/*
+ * read three pending pcm bytes via inb()
+ */
+static void vx_pcm_read_per_bytes(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe)
+{
+ int offset = pipe->hw_ptr;
+ unsigned char *buf = (unsigned char *)(runtime->dma_area + offset);
+ *buf++ = vx_inb(chip, RXH);
+ if (++offset >= pipe->buffer_bytes) {
+ offset = 0;
+ buf = (unsigned char *)runtime->dma_area;
+ }
+ *buf++ = vx_inb(chip, RXM);
+ if (++offset >= pipe->buffer_bytes) {
+ offset = 0;
+ buf = (unsigned char *)runtime->dma_area;
+ }
+ *buf++ = vx_inb(chip, RXL);
+ if (++offset >= pipe->buffer_bytes) {
+ offset = 0;
+ buf = (unsigned char *)runtime->dma_area;
+ }
+ pipe->hw_ptr = offset;
+}
+
+/*
+ * vx_set_pcx_time - convert from the PC time to the RMH status time.
+ * @pc_time: the pointer for the PC-time to set
+ * @dsp_time: the pointer for RMH status time array
+ */
+static void vx_set_pcx_time(struct vx_core *chip, pcx_time_t *pc_time,
+ unsigned int *dsp_time)
+{
+ dsp_time[0] = (unsigned int)((*pc_time) >> 24) & PCX_TIME_HI_MASK;
+ dsp_time[1] = (unsigned int)(*pc_time) & MASK_DSP_WORD;
+}
+
+/*
+ * vx_set_differed_time - set the differed time if specified
+ * @rmh: the rmh record to modify
+ * @pipe: the pipe to be checked
+ *
+ * if the pipe is programmed with the differed time, set the DSP time
+ * on the rmh and changes its command length.
+ *
+ * returns the increase of the command length.
+ */
+static int vx_set_differed_time(struct vx_core *chip, struct vx_rmh *rmh,
+ struct vx_pipe *pipe)
+{
+ /* Update The length added to the RMH command by the timestamp */
+ if (! (pipe->differed_type & DC_DIFFERED_DELAY))
+ return 0;
+
+ /* Set the T bit */
+ rmh->Cmd[0] |= DSP_DIFFERED_COMMAND_MASK;
+
+ /* Time stamp is the 1st following parameter */
+ vx_set_pcx_time(chip, &pipe->pcx_time, &rmh->Cmd[1]);
+
+ /* Add the flags to a notified differed command */
+ if (pipe->differed_type & DC_NOTIFY_DELAY)
+ rmh->Cmd[1] |= NOTIFY_MASK_TIME_HIGH ;
+
+ /* Add the flags to a multiple differed command */
+ if (pipe->differed_type & DC_MULTIPLE_DELAY)
+ rmh->Cmd[1] |= MULTIPLE_MASK_TIME_HIGH;
+
+ /* Add the flags to a stream-time differed command */
+ if (pipe->differed_type & DC_STREAM_TIME_DELAY)
+ rmh->Cmd[1] |= STREAM_MASK_TIME_HIGH;
+
+ rmh->LgCmd += 2;
+ return 2;
+}
+
+/*
+ * vx_set_stream_format - send the stream format command
+ * @pipe: the affected pipe
+ * @data: format bitmask
+ */
+static int vx_set_stream_format(struct vx_core *chip, struct vx_pipe *pipe,
+ unsigned int data)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, pipe->is_capture ?
+ CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT);
+ rmh.Cmd[0] |= pipe->number << FIELD_SIZE;
+
+ /* Command might be longer since we may have to add a timestamp */
+ vx_set_differed_time(chip, &rmh, pipe);
+
+ rmh.Cmd[rmh.LgCmd] = (data & 0xFFFFFF00) >> 8;
+ rmh.Cmd[rmh.LgCmd + 1] = (data & 0xFF) << 16 /*| (datal & 0xFFFF00) >> 8*/;
+ rmh.LgCmd += 2;
+
+ return vx_send_msg(chip, &rmh);
+}
+
+
+/*
+ * vx_set_format - set the format of a pipe
+ * @pipe: the affected pipe
+ * @runtime: pcm runtime instance to be referred
+ *
+ * returns 0 if successful, or a negative error code.
+ */
+static int vx_set_format(struct vx_core *chip, struct vx_pipe *pipe,
+ struct snd_pcm_runtime *runtime)
+{
+ unsigned int header = HEADER_FMT_BASE;
+
+ if (runtime->channels == 1)
+ header |= HEADER_FMT_MONO;
+ if (snd_pcm_format_little_endian(runtime->format))
+ header |= HEADER_FMT_INTEL;
+ if (runtime->rate < 32000 && runtime->rate > 11025)
+ header |= HEADER_FMT_UPTO32;
+ else if (runtime->rate <= 11025)
+ header |= HEADER_FMT_UPTO11;
+
+ switch (snd_pcm_format_physical_width(runtime->format)) {
+ // case 8: break;
+ case 16: header |= HEADER_FMT_16BITS; break;
+ case 24: header |= HEADER_FMT_24BITS; break;
+ default :
+ snd_BUG();
+ return -EINVAL;
+ };
+
+ return vx_set_stream_format(chip, pipe, header);
+}
+
+/*
+ * set / query the IBL size
+ */
+static int vx_set_ibl(struct vx_core *chip, struct vx_ibl_info *info)
+{
+ int err;
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_IBL);
+ rmh.Cmd[0] |= info->size & 0x03ffff;
+ err = vx_send_msg(chip, &rmh);
+ if (err < 0)
+ return err;
+ info->size = rmh.Stat[0];
+ info->max_size = rmh.Stat[1];
+ info->min_size = rmh.Stat[2];
+ info->granularity = rmh.Stat[3];
+ snd_printdd(KERN_DEBUG "vx_set_ibl: size = %d, max = %d, min = %d, gran = %d\n",
+ info->size, info->max_size, info->min_size, info->granularity);
+ return 0;
+}
+
+
+/*
+ * vx_get_pipe_state - get the state of a pipe
+ * @pipe: the pipe to be checked
+ * @state: the pointer for the returned state
+ *
+ * checks the state of a given pipe, and stores the state (1 = running,
+ * 0 = paused) on the given pointer.
+ *
+ * called from trigger callback only
+ */
+static int vx_get_pipe_state(struct vx_core *chip, struct vx_pipe *pipe, int *state)
+{
+ int err;
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_PIPE_STATE);
+ vx_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->number, 0);
+ err = vx_send_msg_nolock(chip, &rmh); /* no lock needed for trigger */
+ if (! err)
+ *state = (rmh.Stat[0] & (1 << pipe->number)) ? 1 : 0;
+ return err;
+}
+
+
+/*
+ * vx_query_hbuffer_size - query available h-buffer size in bytes
+ * @pipe: the pipe to be checked
+ *
+ * return the available size on h-buffer in bytes,
+ * or a negative error code.
+ *
+ * NOTE: calling this function always switches to the stream mode.
+ * you'll need to disconnect the host to get back to the
+ * normal mode.
+ */
+static int vx_query_hbuffer_size(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ int result;
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_SIZE_HBUFFER);
+ vx_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->number, 0);
+ if (pipe->is_capture)
+ rmh.Cmd[0] |= 0x00000001;
+ result = vx_send_msg(chip, &rmh);
+ if (! result)
+ result = rmh.Stat[0] & 0xffff;
+ return result;
+}
+
+
+/*
+ * vx_pipe_can_start - query whether a pipe is ready for start
+ * @pipe: the pipe to be checked
+ *
+ * return 1 if ready, 0 if not ready, and negative value on error.
+ *
+ * called from trigger callback only
+ */
+static int vx_pipe_can_start(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ int err;
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_CAN_START_PIPE);
+ vx_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->number, 0);
+ rmh.Cmd[0] |= 1;
+
+ err = vx_send_msg_nolock(chip, &rmh); /* no lock needed for trigger */
+ if (! err) {
+ if (rmh.Stat[0])
+ err = 1;
+ }
+ return err;
+}
+
+/*
+ * vx_conf_pipe - tell the pipe to stand by and wait for IRQA.
+ * @pipe: the pipe to be configured
+ */
+static int vx_conf_pipe(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_CONF_PIPE);
+ if (pipe->is_capture)
+ rmh.Cmd[0] |= COMMAND_RECORD_MASK;
+ rmh.Cmd[1] = 1 << pipe->number;
+ return vx_send_msg_nolock(chip, &rmh); /* no lock needed for trigger */
+}
+
+/*
+ * vx_send_irqa - trigger IRQA
+ */
+static int vx_send_irqa(struct vx_core *chip)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_SEND_IRQA);
+ return vx_send_msg_nolock(chip, &rmh); /* no lock needed for trigger */
+}
+
+
+#define MAX_WAIT_FOR_DSP 250
+/*
+ * vx boards do not support inter-card sync, besides
+ * only 126 samples require to be prepared before a pipe can start
+ */
+#define CAN_START_DELAY 2 /* wait 2ms only before asking if the pipe is ready*/
+#define WAIT_STATE_DELAY 2 /* wait 2ms after irqA was requested and check if the pipe state toggled*/
+
+/*
+ * vx_toggle_pipe - start / pause a pipe
+ * @pipe: the pipe to be triggered
+ * @state: start = 1, pause = 0
+ *
+ * called from trigger callback only
+ *
+ */
+static int vx_toggle_pipe(struct vx_core *chip, struct vx_pipe *pipe, int state)
+{
+ int err, i, cur_state;
+
+ /* Check the pipe is not already in the requested state */
+ if (vx_get_pipe_state(chip, pipe, &cur_state) < 0)
+ return -EBADFD;
+ if (state == cur_state)
+ return 0;
+
+ /* If a start is requested, ask the DSP to get prepared
+ * and wait for a positive acknowledge (when there are
+ * enough sound buffer for this pipe)
+ */
+ if (state) {
+ for (i = 0 ; i < MAX_WAIT_FOR_DSP; i++) {
+ err = vx_pipe_can_start(chip, pipe);
+ if (err > 0)
+ break;
+ /* Wait for a few, before asking again
+ * to avoid flooding the DSP with our requests
+ */
+ mdelay(1);
+ }
+ }
+
+ if ((err = vx_conf_pipe(chip, pipe)) < 0)
+ return err;
+
+ if ((err = vx_send_irqa(chip)) < 0)
+ return err;
+
+ /* If it completes successfully, wait for the pipes
+ * reaching the expected state before returning
+ * Check one pipe only (since they are synchronous)
+ */
+ for (i = 0; i < MAX_WAIT_FOR_DSP; i++) {
+ err = vx_get_pipe_state(chip, pipe, &cur_state);
+ if (err < 0 || cur_state == state)
+ break;
+ err = -EIO;
+ mdelay(1);
+ }
+ return err < 0 ? -EIO : 0;
+}
+
+
+/*
+ * vx_stop_pipe - stop a pipe
+ * @pipe: the pipe to be stopped
+ *
+ * called from trigger callback only
+ */
+static int vx_stop_pipe(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ struct vx_rmh rmh;
+ vx_init_rmh(&rmh, CMD_STOP_PIPE);
+ vx_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->number, 0);
+ return vx_send_msg_nolock(chip, &rmh); /* no lock needed for trigger */
+}
+
+
+/*
+ * vx_alloc_pipe - allocate a pipe and initialize the pipe instance
+ * @capture: 0 = playback, 1 = capture operation
+ * @audioid: the audio id to be assigned
+ * @num_audio: number of audio channels
+ * @pipep: the returned pipe instance
+ *
+ * return 0 on success, or a negative error code.
+ */
+static int vx_alloc_pipe(struct vx_core *chip, int capture,
+ int audioid, int num_audio,
+ struct vx_pipe **pipep)
+{
+ int err;
+ struct vx_pipe *pipe;
+ struct vx_rmh rmh;
+ int data_mode;
+
+ *pipep = NULL;
+ vx_init_rmh(&rmh, CMD_RES_PIPE);
+ vx_set_pipe_cmd_params(&rmh, capture, audioid, num_audio);
+#if 0 // NYI
+ if (underrun_skip_sound)
+ rmh.Cmd[0] |= BIT_SKIP_SOUND;
+#endif // NYI
+ data_mode = (chip->uer_bits & IEC958_AES0_NONAUDIO) != 0;
+ if (! capture && data_mode)
+ rmh.Cmd[0] |= BIT_DATA_MODE;
+ err = vx_send_msg(chip, &rmh);
+ if (err < 0)
+ return err;
+
+ /* initialize the pipe record */
+ pipe = kzalloc(sizeof(*pipe), GFP_KERNEL);
+ if (! pipe) {
+ /* release the pipe */
+ vx_init_rmh(&rmh, CMD_FREE_PIPE);
+ vx_set_pipe_cmd_params(&rmh, capture, audioid, 0);
+ vx_send_msg(chip, &rmh);
+ return -ENOMEM;
+ }
+
+ /* the pipe index should be identical with the audio index */
+ pipe->number = audioid;
+ pipe->is_capture = capture;
+ pipe->channels = num_audio;
+ pipe->differed_type = 0;
+ pipe->pcx_time = 0;
+ pipe->data_mode = data_mode;
+ *pipep = pipe;
+
+ return 0;
+}
+
+
+/*
+ * vx_free_pipe - release a pipe
+ * @pipe: pipe to be released
+ */
+static int vx_free_pipe(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_FREE_PIPE);
+ vx_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->number, 0);
+ vx_send_msg(chip, &rmh);
+
+ kfree(pipe);
+ return 0;
+}
+
+
+/*
+ * vx_start_stream - start the stream
+ *
+ * called from trigger callback only
+ */
+static int vx_start_stream(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_START_ONE_STREAM);
+ vx_set_stream_cmd_params(&rmh, pipe->is_capture, pipe->number);
+ vx_set_differed_time(chip, &rmh, pipe);
+ return vx_send_msg_nolock(chip, &rmh); /* no lock needed for trigger */
+}
+
+
+/*
+ * vx_stop_stream - stop the stream
+ *
+ * called from trigger callback only
+ */
+static int vx_stop_stream(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_STOP_STREAM);
+ vx_set_stream_cmd_params(&rmh, pipe->is_capture, pipe->number);
+ return vx_send_msg_nolock(chip, &rmh); /* no lock needed for trigger */
+}
+
+
+/*
+ * playback hw information
+ */
+
+static struct snd_pcm_hardware vx_pcm_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP_VALID /*|*/
+ /*SNDRV_PCM_INFO_RESUME*/),
+ .formats = (/*SNDRV_PCM_FMTBIT_U8 |*/
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 126,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = VX_MAX_PERIODS,
+ .fifo_size = 126,
+};
+
+
+static void vx_pcm_delayed_start(unsigned long arg);
+
+/*
+ * vx_pcm_playback_open - open callback for playback
+ */
+static int vx_pcm_playback_open(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct vx_core *chip = snd_pcm_substream_chip(subs);
+ struct vx_pipe *pipe = NULL;
+ unsigned int audio;
+ int err;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+ audio = subs->pcm->device * 2;
+ if (snd_BUG_ON(audio >= chip->audio_outs))
+ return -EINVAL;
+
+ /* playback pipe may have been already allocated for monitoring */
+ pipe = chip->playback_pipes[audio];
+ if (! pipe) {
+ /* not allocated yet */
+ err = vx_alloc_pipe(chip, 0, audio, 2, &pipe); /* stereo playback */
+ if (err < 0)
+ return err;
+ chip->playback_pipes[audio] = pipe;
+ }
+ /* open for playback */
+ pipe->references++;
+
+ pipe->substream = subs;
+ tasklet_init(&pipe->start_tq, vx_pcm_delayed_start, (unsigned long)subs);
+ chip->playback_pipes[audio] = pipe;
+
+ runtime->hw = vx_pcm_playback_hw;
+ runtime->hw.period_bytes_min = chip->ibl.size;
+ runtime->private_data = pipe;
+
+ /* align to 4 bytes (otherwise will be problematic when 24bit is used) */
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
+
+ return 0;
+}
+
+/*
+ * vx_pcm_playback_close - close callback for playback
+ */
+static int vx_pcm_playback_close(struct snd_pcm_substream *subs)
+{
+ struct vx_core *chip = snd_pcm_substream_chip(subs);
+ struct vx_pipe *pipe;
+
+ if (! subs->runtime->private_data)
+ return -EINVAL;
+
+ pipe = subs->runtime->private_data;
+
+ if (--pipe->references == 0) {
+ chip->playback_pipes[pipe->number] = NULL;
+ vx_free_pipe(chip, pipe);
+ }
+
+ return 0;
+
+}
+
+
+/*
+ * vx_notify_end_of_buffer - send "end-of-buffer" notifier at the given pipe
+ * @pipe: the pipe to notify
+ *
+ * NB: call with a certain lock.
+ */
+static int vx_notify_end_of_buffer(struct vx_core *chip, struct vx_pipe *pipe)
+{
+ int err;
+ struct vx_rmh rmh; /* use a temporary rmh here */
+
+ /* Toggle Dsp Host Interface into Message mode */
+ vx_send_rih_nolock(chip, IRQ_PAUSE_START_CONNECT);
+ vx_init_rmh(&rmh, CMD_NOTIFY_END_OF_BUFFER);
+ vx_set_stream_cmd_params(&rmh, 0, pipe->number);
+ err = vx_send_msg_nolock(chip, &rmh);
+ if (err < 0)
+ return err;
+ /* Toggle Dsp Host Interface back to sound transfer mode */
+ vx_send_rih_nolock(chip, IRQ_PAUSE_START_CONNECT);
+ return 0;
+}
+
+/*
+ * vx_pcm_playback_transfer_chunk - transfer a single chunk
+ * @subs: substream
+ * @pipe: the pipe to transfer
+ * @size: chunk size in bytes
+ *
+ * transfer a single buffer chunk. EOB notificaton is added after that.
+ * called from the interrupt handler, too.
+ *
+ * return 0 if ok.
+ */
+static int vx_pcm_playback_transfer_chunk(struct vx_core *chip,
+ struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int size)
+{
+ int space, err = 0;
+
+ space = vx_query_hbuffer_size(chip, pipe);
+ if (space < 0) {
+ /* disconnect the host, SIZE_HBUF command always switches to the stream mode */
+ vx_send_rih(chip, IRQ_CONNECT_STREAM_NEXT);
+ snd_printd("error hbuffer\n");
+ return space;
+ }
+ if (space < size) {
+ vx_send_rih(chip, IRQ_CONNECT_STREAM_NEXT);
+ snd_printd("no enough hbuffer space %d\n", space);
+ return -EIO; /* XRUN */
+ }
+
+ /* we don't need irqsave here, because this function
+ * is called from either trigger callback or irq handler
+ */
+ spin_lock(&chip->lock);
+ vx_pseudo_dma_write(chip, runtime, pipe, size);
+ err = vx_notify_end_of_buffer(chip, pipe);
+ /* disconnect the host, SIZE_HBUF command always switches to the stream mode */
+ vx_send_rih_nolock(chip, IRQ_CONNECT_STREAM_NEXT);
+ spin_unlock(&chip->lock);
+ return err;
+}
+
+/*
+ * update the position of the given pipe.
+ * pipe->position is updated and wrapped within the buffer size.
+ * pipe->transferred is updated, too, but the size is not wrapped,
+ * so that the caller can check the total transferred size later
+ * (to call snd_pcm_period_elapsed).
+ */
+static int vx_update_pipe_position(struct vx_core *chip,
+ struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe)
+{
+ struct vx_rmh rmh;
+ int err, update;
+ u64 count;
+
+ vx_init_rmh(&rmh, CMD_STREAM_SAMPLE_COUNT);
+ vx_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->number, 0);
+ err = vx_send_msg(chip, &rmh);
+ if (err < 0)
+ return err;
+
+ count = ((u64)(rmh.Stat[0] & 0xfffff) << 24) | (u64)rmh.Stat[1];
+ update = (int)(count - pipe->cur_count);
+ pipe->cur_count = count;
+ pipe->position += update;
+ if (pipe->position >= (int)runtime->buffer_size)
+ pipe->position %= runtime->buffer_size;
+ pipe->transferred += update;
+ return 0;
+}
+
+/*
+ * transfer the pending playback buffer data to DSP
+ * called from interrupt handler
+ */
+static void vx_pcm_playback_transfer(struct vx_core *chip,
+ struct snd_pcm_substream *subs,
+ struct vx_pipe *pipe, int nchunks)
+{
+ int i, err;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ if (! pipe->prepared || (chip->chip_status & VX_STAT_IS_STALE))
+ return;
+ for (i = 0; i < nchunks; i++) {
+ if ((err = vx_pcm_playback_transfer_chunk(chip, runtime, pipe,
+ chip->ibl.size)) < 0)
+ return;
+ }
+}
+
+/*
+ * update the playback position and call snd_pcm_period_elapsed() if necessary
+ * called from interrupt handler
+ */
+static void vx_pcm_playback_update(struct vx_core *chip,
+ struct snd_pcm_substream *subs,
+ struct vx_pipe *pipe)
+{
+ int err;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ if (pipe->running && ! (chip->chip_status & VX_STAT_IS_STALE)) {
+ if ((err = vx_update_pipe_position(chip, runtime, pipe)) < 0)
+ return;
+ if (pipe->transferred >= (int)runtime->period_size) {
+ pipe->transferred %= runtime->period_size;
+ snd_pcm_period_elapsed(subs);
+ }
+ }
+}
+
+/*
+ * start the stream and pipe.
+ * this function is called from tasklet, which is invoked by the trigger
+ * START callback.
+ */
+static void vx_pcm_delayed_start(unsigned long arg)
+{
+ struct snd_pcm_substream *subs = (struct snd_pcm_substream *)arg;
+ struct vx_core *chip = subs->pcm->private_data;
+ struct vx_pipe *pipe = subs->runtime->private_data;
+ int err;
+
+ /* printk( KERN_DEBUG "DDDD tasklet delayed start jiffies = %ld\n", jiffies);*/
+
+ if ((err = vx_start_stream(chip, pipe)) < 0) {
+ snd_printk(KERN_ERR "vx: cannot start stream\n");
+ return;
+ }
+ if ((err = vx_toggle_pipe(chip, pipe, 1)) < 0) {
+ snd_printk(KERN_ERR "vx: cannot start pipe\n");
+ return;
+ }
+ /* printk( KERN_DEBUG "dddd tasklet delayed start jiffies = %ld \n", jiffies);*/
+}
+
+/*
+ * vx_pcm_playback_trigger - trigger callback for playback
+ */
+static int vx_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
+{
+ struct vx_core *chip = snd_pcm_substream_chip(subs);
+ struct vx_pipe *pipe = subs->runtime->private_data;
+ int err;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (! pipe->is_capture)
+ vx_pcm_playback_transfer(chip, subs, pipe, 2);
+ /* FIXME:
+ * we trigger the pipe using tasklet, so that the interrupts are
+ * issued surely after the trigger is completed.
+ */
+ tasklet_hi_schedule(&pipe->start_tq);
+ chip->pcm_running++;
+ pipe->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ vx_toggle_pipe(chip, pipe, 0);
+ vx_stop_pipe(chip, pipe);
+ vx_stop_stream(chip, pipe);
+ chip->pcm_running--;
+ pipe->running = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if ((err = vx_toggle_pipe(chip, pipe, 0)) < 0)
+ return err;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if ((err = vx_toggle_pipe(chip, pipe, 1)) < 0)
+ return err;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * vx_pcm_playback_pointer - pointer callback for playback
+ */
+static snd_pcm_uframes_t vx_pcm_playback_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct vx_pipe *pipe = runtime->private_data;
+ return pipe->position;
+}
+
+/*
+ * vx_pcm_hw_params - hw_params callback for playback and capture
+ */
+static int vx_pcm_hw_params(struct snd_pcm_substream *subs,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params));
+}
+
+/*
+ * vx_pcm_hw_free - hw_free callback for playback and capture
+ */
+static int vx_pcm_hw_free(struct snd_pcm_substream *subs)
+{
+ return snd_pcm_free_vmalloc_buffer(subs);
+}
+
+/*
+ * vx_pcm_prepare - prepare callback for playback and capture
+ */
+static int vx_pcm_prepare(struct snd_pcm_substream *subs)
+{
+ struct vx_core *chip = snd_pcm_substream_chip(subs);
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct vx_pipe *pipe = runtime->private_data;
+ int err, data_mode;
+ // int max_size, nchunks;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+ data_mode = (chip->uer_bits & IEC958_AES0_NONAUDIO) != 0;
+ if (data_mode != pipe->data_mode && ! pipe->is_capture) {
+ /* IEC958 status (raw-mode) was changed */
+ /* we reopen the pipe */
+ struct vx_rmh rmh;
+ snd_printdd(KERN_DEBUG "reopen the pipe with data_mode = %d\n", data_mode);
+ vx_init_rmh(&rmh, CMD_FREE_PIPE);
+ vx_set_pipe_cmd_params(&rmh, 0, pipe->number, 0);
+ if ((err = vx_send_msg(chip, &rmh)) < 0)
+ return err;
+ vx_init_rmh(&rmh, CMD_RES_PIPE);
+ vx_set_pipe_cmd_params(&rmh, 0, pipe->number, pipe->channels);
+ if (data_mode)
+ rmh.Cmd[0] |= BIT_DATA_MODE;
+ if ((err = vx_send_msg(chip, &rmh)) < 0)
+ return err;
+ pipe->data_mode = data_mode;
+ }
+
+ if (chip->pcm_running && chip->freq != runtime->rate) {
+ snd_printk(KERN_ERR "vx: cannot set different clock %d "
+ "from the current %d\n", runtime->rate, chip->freq);
+ return -EINVAL;
+ }
+ vx_set_clock(chip, runtime->rate);
+
+ if ((err = vx_set_format(chip, pipe, runtime)) < 0)
+ return err;
+
+ if (vx_is_pcmcia(chip)) {
+ pipe->align = 2; /* 16bit word */
+ } else {
+ pipe->align = 4; /* 32bit word */
+ }
+
+ pipe->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size);
+ pipe->period_bytes = frames_to_bytes(runtime, runtime->period_size);
+ pipe->hw_ptr = 0;
+
+ /* set the timestamp */
+ vx_update_pipe_position(chip, runtime, pipe);
+ /* clear again */
+ pipe->transferred = 0;
+ pipe->position = 0;
+
+ pipe->prepared = 1;
+
+ return 0;
+}
+
+
+/*
+ * operators for PCM playback
+ */
+static struct snd_pcm_ops vx_pcm_playback_ops = {
+ .open = vx_pcm_playback_open,
+ .close = vx_pcm_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = vx_pcm_hw_params,
+ .hw_free = vx_pcm_hw_free,
+ .prepare = vx_pcm_prepare,
+ .trigger = vx_pcm_trigger,
+ .pointer = vx_pcm_playback_pointer,
+ .page = snd_pcm_get_vmalloc_page,
+};
+
+
+/*
+ * playback hw information
+ */
+
+static struct snd_pcm_hardware vx_pcm_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP_VALID /*|*/
+ /*SNDRV_PCM_INFO_RESUME*/),
+ .formats = (/*SNDRV_PCM_FMTBIT_U8 |*/
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 126,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = VX_MAX_PERIODS,
+ .fifo_size = 126,
+};
+
+
+/*
+ * vx_pcm_capture_open - open callback for capture
+ */
+static int vx_pcm_capture_open(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct vx_core *chip = snd_pcm_substream_chip(subs);
+ struct vx_pipe *pipe;
+ struct vx_pipe *pipe_out_monitoring = NULL;
+ unsigned int audio;
+ int err;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return -EBUSY;
+
+ audio = subs->pcm->device * 2;
+ if (snd_BUG_ON(audio >= chip->audio_ins))
+ return -EINVAL;
+ err = vx_alloc_pipe(chip, 1, audio, 2, &pipe);
+ if (err < 0)
+ return err;
+ pipe->substream = subs;
+ tasklet_init(&pipe->start_tq, vx_pcm_delayed_start, (unsigned long)subs);
+ chip->capture_pipes[audio] = pipe;
+
+ /* check if monitoring is needed */
+ if (chip->audio_monitor_active[audio]) {
+ pipe_out_monitoring = chip->playback_pipes[audio];
+ if (! pipe_out_monitoring) {
+ /* allocate a pipe */
+ err = vx_alloc_pipe(chip, 0, audio, 2, &pipe_out_monitoring);
+ if (err < 0)
+ return err;
+ chip->playback_pipes[audio] = pipe_out_monitoring;
+ }
+ pipe_out_monitoring->references++;
+ /*
+ if an output pipe is available, it's audios still may need to be
+ unmuted. hence we'll have to call a mixer entry point.
+ */
+ vx_set_monitor_level(chip, audio, chip->audio_monitor[audio],
+ chip->audio_monitor_active[audio]);
+ /* assuming stereo */
+ vx_set_monitor_level(chip, audio+1, chip->audio_monitor[audio+1],
+ chip->audio_monitor_active[audio+1]);
+ }
+
+ pipe->monitoring_pipe = pipe_out_monitoring; /* default value NULL */
+
+ runtime->hw = vx_pcm_capture_hw;
+ runtime->hw.period_bytes_min = chip->ibl.size;
+ runtime->private_data = pipe;
+
+ /* align to 4 bytes (otherwise will be problematic when 24bit is used) */
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
+
+ return 0;
+}
+
+/*
+ * vx_pcm_capture_close - close callback for capture
+ */
+static int vx_pcm_capture_close(struct snd_pcm_substream *subs)
+{
+ struct vx_core *chip = snd_pcm_substream_chip(subs);
+ struct vx_pipe *pipe;
+ struct vx_pipe *pipe_out_monitoring;
+
+ if (! subs->runtime->private_data)
+ return -EINVAL;
+ pipe = subs->runtime->private_data;
+ chip->capture_pipes[pipe->number] = NULL;
+
+ pipe_out_monitoring = pipe->monitoring_pipe;
+
+ /*
+ if an output pipe is attached to this input,
+ check if it needs to be released.
+ */
+ if (pipe_out_monitoring) {
+ if (--pipe_out_monitoring->references == 0) {
+ vx_free_pipe(chip, pipe_out_monitoring);
+ chip->playback_pipes[pipe->number] = NULL;
+ pipe->monitoring_pipe = NULL;
+ }
+ }
+
+ vx_free_pipe(chip, pipe);
+ return 0;
+}
+
+
+
+#define DMA_READ_ALIGN 6 /* hardware alignment for read */
+
+/*
+ * vx_pcm_capture_update - update the capture buffer
+ */
+static void vx_pcm_capture_update(struct vx_core *chip, struct snd_pcm_substream *subs,
+ struct vx_pipe *pipe)
+{
+ int size, space, count;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ if (! pipe->prepared || (chip->chip_status & VX_STAT_IS_STALE))
+ return;
+
+ size = runtime->buffer_size - snd_pcm_capture_avail(runtime);
+ if (! size)
+ return;
+ size = frames_to_bytes(runtime, size);
+ space = vx_query_hbuffer_size(chip, pipe);
+ if (space < 0)
+ goto _error;
+ if (size > space)
+ size = space;
+ size = (size / 3) * 3; /* align to 3 bytes */
+ if (size < DMA_READ_ALIGN)
+ goto _error;
+
+ /* keep the last 6 bytes, they will be read after disconnection */
+ count = size - DMA_READ_ALIGN;
+ /* read bytes until the current pointer reaches to the aligned position
+ * for word-transfer
+ */
+ while (count > 0) {
+ if ((pipe->hw_ptr % pipe->align) == 0)
+ break;
+ if (vx_wait_for_rx_full(chip) < 0)
+ goto _error;
+ vx_pcm_read_per_bytes(chip, runtime, pipe);
+ count -= 3;
+ }
+ if (count > 0) {
+ /* ok, let's accelerate! */
+ int align = pipe->align * 3;
+ space = (count / align) * align;
+ vx_pseudo_dma_read(chip, runtime, pipe, space);
+ count -= space;
+ }
+ /* read the rest of bytes */
+ while (count > 0) {
+ if (vx_wait_for_rx_full(chip) < 0)
+ goto _error;
+ vx_pcm_read_per_bytes(chip, runtime, pipe);
+ count -= 3;
+ }
+ /* disconnect the host, SIZE_HBUF command always switches to the stream mode */
+ vx_send_rih_nolock(chip, IRQ_CONNECT_STREAM_NEXT);
+ /* read the last pending 6 bytes */
+ count = DMA_READ_ALIGN;
+ while (count > 0) {
+ vx_pcm_read_per_bytes(chip, runtime, pipe);
+ count -= 3;
+ }
+ /* update the position */
+ pipe->transferred += size;
+ if (pipe->transferred >= pipe->period_bytes) {
+ pipe->transferred %= pipe->period_bytes;
+ snd_pcm_period_elapsed(subs);
+ }
+ return;
+
+ _error:
+ /* disconnect the host, SIZE_HBUF command always switches to the stream mode */
+ vx_send_rih_nolock(chip, IRQ_CONNECT_STREAM_NEXT);
+ return;
+}
+
+/*
+ * vx_pcm_capture_pointer - pointer callback for capture
+ */
+static snd_pcm_uframes_t vx_pcm_capture_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct vx_pipe *pipe = runtime->private_data;
+ return bytes_to_frames(runtime, pipe->hw_ptr);
+}
+
+/*
+ * operators for PCM capture
+ */
+static struct snd_pcm_ops vx_pcm_capture_ops = {
+ .open = vx_pcm_capture_open,
+ .close = vx_pcm_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = vx_pcm_hw_params,
+ .hw_free = vx_pcm_hw_free,
+ .prepare = vx_pcm_prepare,
+ .trigger = vx_pcm_trigger,
+ .pointer = vx_pcm_capture_pointer,
+ .page = snd_pcm_get_vmalloc_page,
+};
+
+
+/*
+ * interrupt handler for pcm streams
+ */
+void vx_pcm_update_intr(struct vx_core *chip, unsigned int events)
+{
+ unsigned int i;
+ struct vx_pipe *pipe;
+
+#define EVENT_MASK (END_OF_BUFFER_EVENTS_PENDING|ASYNC_EVENTS_PENDING)
+
+ if (events & EVENT_MASK) {
+ vx_init_rmh(&chip->irq_rmh, CMD_ASYNC);
+ if (events & ASYNC_EVENTS_PENDING)
+ chip->irq_rmh.Cmd[0] |= 0x00000001; /* SEL_ASYNC_EVENTS */
+ if (events & END_OF_BUFFER_EVENTS_PENDING)
+ chip->irq_rmh.Cmd[0] |= 0x00000002; /* SEL_END_OF_BUF_EVENTS */
+
+ if (vx_send_msg(chip, &chip->irq_rmh) < 0) {
+ snd_printdd(KERN_ERR "msg send error!!\n");
+ return;
+ }
+
+ i = 1;
+ while (i < chip->irq_rmh.LgStat) {
+ int p, buf, capture, eob;
+ p = chip->irq_rmh.Stat[i] & MASK_FIRST_FIELD;
+ capture = (chip->irq_rmh.Stat[i] & 0x400000) ? 1 : 0;
+ eob = (chip->irq_rmh.Stat[i] & 0x800000) ? 1 : 0;
+ i++;
+ if (events & ASYNC_EVENTS_PENDING)
+ i++;
+ buf = 1; /* force to transfer */
+ if (events & END_OF_BUFFER_EVENTS_PENDING) {
+ if (eob)
+ buf = chip->irq_rmh.Stat[i];
+ i++;
+ }
+ if (capture)
+ continue;
+ if (snd_BUG_ON(p < 0 || p >= chip->audio_outs))
+ continue;
+ pipe = chip->playback_pipes[p];
+ if (pipe && pipe->substream) {
+ vx_pcm_playback_update(chip, pipe->substream, pipe);
+ vx_pcm_playback_transfer(chip, pipe->substream, pipe, buf);
+ }
+ }
+ }
+
+ /* update the capture pcm pointers as frequently as possible */
+ for (i = 0; i < chip->audio_ins; i++) {
+ pipe = chip->capture_pipes[i];
+ if (pipe && pipe->substream)
+ vx_pcm_capture_update(chip, pipe->substream, pipe);
+ }
+}
+
+
+/*
+ * vx_init_audio_io - check the availabe audio i/o and allocate pipe arrays
+ */
+static int vx_init_audio_io(struct vx_core *chip)
+{
+ struct vx_rmh rmh;
+ int preferred;
+
+ vx_init_rmh(&rmh, CMD_SUPPORTED);
+ if (vx_send_msg(chip, &rmh) < 0) {
+ snd_printk(KERN_ERR "vx: cannot get the supported audio data\n");
+ return -ENXIO;
+ }
+
+ chip->audio_outs = rmh.Stat[0] & MASK_FIRST_FIELD;
+ chip->audio_ins = (rmh.Stat[0] >> (FIELD_SIZE*2)) & MASK_FIRST_FIELD;
+ chip->audio_info = rmh.Stat[1];
+
+ /* allocate pipes */
+ chip->playback_pipes = kcalloc(chip->audio_outs, sizeof(struct vx_pipe *), GFP_KERNEL);
+ if (!chip->playback_pipes)
+ return -ENOMEM;
+ chip->capture_pipes = kcalloc(chip->audio_ins, sizeof(struct vx_pipe *), GFP_KERNEL);
+ if (!chip->capture_pipes) {
+ kfree(chip->playback_pipes);
+ return -ENOMEM;
+ }
+
+ preferred = chip->ibl.size;
+ chip->ibl.size = 0;
+ vx_set_ibl(chip, &chip->ibl); /* query the info */
+ if (preferred > 0) {
+ chip->ibl.size = ((preferred + chip->ibl.granularity - 1) /
+ chip->ibl.granularity) * chip->ibl.granularity;
+ if (chip->ibl.size > chip->ibl.max_size)
+ chip->ibl.size = chip->ibl.max_size;
+ } else
+ chip->ibl.size = chip->ibl.min_size; /* set to the minimum */
+ vx_set_ibl(chip, &chip->ibl);
+
+ return 0;
+}
+
+
+/*
+ * free callback for pcm
+ */
+static void snd_vx_pcm_free(struct snd_pcm *pcm)
+{
+ struct vx_core *chip = pcm->private_data;
+ chip->pcm[pcm->device] = NULL;
+ kfree(chip->playback_pipes);
+ chip->playback_pipes = NULL;
+ kfree(chip->capture_pipes);
+ chip->capture_pipes = NULL;
+}
+
+/*
+ * snd_vx_pcm_new - create and initialize a pcm
+ */
+int snd_vx_pcm_new(struct vx_core *chip)
+{
+ struct snd_pcm *pcm;
+ unsigned int i;
+ int err;
+
+ if ((err = vx_init_audio_io(chip)) < 0)
+ return err;
+
+ for (i = 0; i < chip->hw->num_codecs; i++) {
+ unsigned int outs, ins;
+ outs = chip->audio_outs > i * 2 ? 1 : 0;
+ ins = chip->audio_ins > i * 2 ? 1 : 0;
+ if (! outs && ! ins)
+ break;
+ err = snd_pcm_new(chip->card, "VX PCM", i,
+ outs, ins, &pcm);
+ if (err < 0)
+ return err;
+ if (outs)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &vx_pcm_playback_ops);
+ if (ins)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &vx_pcm_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->private_free = snd_vx_pcm_free;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcm[i] = pcm;
+ }
+
+ return 0;
+}
diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c
new file mode 100644
index 0000000..0e1ba9b
--- /dev/null
+++ b/sound/drivers/vx/vx_uer.c
@@ -0,0 +1,312 @@
+/*
+ * Driver for Digigram VX soundcards
+ *
+ * IEC958 stuff
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/vx_core.h>
+#include "vx_cmd.h"
+
+
+/*
+ * vx_modify_board_clock - tell the board that its clock has been modified
+ * @sync: DSP needs to resynchronize its FIFO
+ */
+static int vx_modify_board_clock(struct vx_core *chip, int sync)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_MODIFY_CLOCK);
+ /* Ask the DSP to resynchronize its FIFO. */
+ if (sync)
+ rmh.Cmd[0] |= CMD_MODIFY_CLOCK_S_BIT;
+ return vx_send_msg(chip, &rmh);
+}
+
+/*
+ * vx_modify_board_inputs - resync audio inputs
+ */
+static int vx_modify_board_inputs(struct vx_core *chip)
+{
+ struct vx_rmh rmh;
+
+ vx_init_rmh(&rmh, CMD_RESYNC_AUDIO_INPUTS);
+ rmh.Cmd[0] |= 1 << 0; /* reference: AUDIO 0 */
+ return vx_send_msg(chip, &rmh);
+}
+
+/*
+ * vx_read_one_cbit - read one bit from UER config
+ * @index: the bit index
+ * returns 0 or 1.
+ */
+static int vx_read_one_cbit(struct vx_core *chip, int index)
+{
+ unsigned long flags;
+ int val;
+ spin_lock_irqsave(&chip->lock, flags);
+ if (chip->type >= VX_TYPE_VXPOCKET) {
+ vx_outb(chip, CSUER, 1); /* read */
+ vx_outb(chip, RUER, index & XX_UER_CBITS_OFFSET_MASK);
+ val = (vx_inb(chip, RUER) >> 7) & 0x01;
+ } else {
+ vx_outl(chip, CSUER, 1); /* read */
+ vx_outl(chip, RUER, index & XX_UER_CBITS_OFFSET_MASK);
+ val = (vx_inl(chip, RUER) >> 7) & 0x01;
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return val;
+}
+
+/*
+ * vx_write_one_cbit - write one bit to UER config
+ * @index: the bit index
+ * @val: bit value, 0 or 1
+ */
+static void vx_write_one_cbit(struct vx_core *chip, int index, int val)
+{
+ unsigned long flags;
+ val = !!val; /* 0 or 1 */
+ spin_lock_irqsave(&chip->lock, flags);
+ if (vx_is_pcmcia(chip)) {
+ vx_outb(chip, CSUER, 0); /* write */
+ vx_outb(chip, RUER, (val << 7) | (index & XX_UER_CBITS_OFFSET_MASK));
+ } else {
+ vx_outl(chip, CSUER, 0); /* write */
+ vx_outl(chip, RUER, (val << 7) | (index & XX_UER_CBITS_OFFSET_MASK));
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+/*
+ * vx_read_uer_status - read the current UER status
+ * @mode: pointer to store the UER mode, VX_UER_MODE_XXX
+ *
+ * returns the frequency of UER, or 0 if not sync,
+ * or a negative error code.
+ */
+static int vx_read_uer_status(struct vx_core *chip, int *mode)
+{
+ int val, freq;
+
+ /* Default values */
+ freq = 0;
+
+ /* Read UER status */
+ if (vx_is_pcmcia(chip))
+ val = vx_inb(chip, CSUER);
+ else
+ val = vx_inl(chip, CSUER);
+ if (val < 0)
+ return val;
+ /* If clock is present, read frequency */
+ if (val & VX_SUER_CLOCK_PRESENT_MASK) {
+ switch (val & VX_SUER_FREQ_MASK) {
+ case VX_SUER_FREQ_32KHz_MASK:
+ freq = 32000;
+ break;
+ case VX_SUER_FREQ_44KHz_MASK:
+ freq = 44100;
+ break;
+ case VX_SUER_FREQ_48KHz_MASK:
+ freq = 48000;
+ break;
+ }
+ }
+ if (val & VX_SUER_DATA_PRESENT_MASK)
+ /* bit 0 corresponds to consumer/professional bit */
+ *mode = vx_read_one_cbit(chip, 0) ?
+ VX_UER_MODE_PROFESSIONAL : VX_UER_MODE_CONSUMER;
+ else
+ *mode = VX_UER_MODE_NOT_PRESENT;
+
+ return freq;
+}
+
+
+/*
+ * compute the sample clock value from frequency
+ *
+ * The formula is as follows:
+ *
+ * HexFreq = (dword) ((double) ((double) 28224000 / (double) Frequency))
+ * switch ( HexFreq & 0x00000F00 )
+ * case 0x00000100: ;
+ * case 0x00000200:
+ * case 0x00000300: HexFreq -= 0x00000201 ;
+ * case 0x00000400:
+ * case 0x00000500:
+ * case 0x00000600:
+ * case 0x00000700: HexFreq = (dword) (((double) 28224000 / (double) (Frequency*2)) - 1)
+ * default : HexFreq = (dword) ((double) 28224000 / (double) (Frequency*4)) - 0x000001FF
+ */
+
+static int vx_calc_clock_from_freq(struct vx_core *chip, int freq)
+{
+ int hexfreq;
+
+ if (snd_BUG_ON(freq <= 0))
+ return 0;
+
+ hexfreq = (28224000 * 10) / freq;
+ hexfreq = (hexfreq + 5) / 10;
+
+ /* max freq = 55125 Hz */
+ if (snd_BUG_ON(hexfreq <= 0x00000200))
+ return 0;
+
+ if (hexfreq <= 0x03ff)
+ return hexfreq - 0x00000201;
+ if (hexfreq <= 0x07ff)
+ return (hexfreq / 2) - 1;
+ if (hexfreq <= 0x0fff)
+ return (hexfreq / 4) + 0x000001ff;
+
+ return 0x5fe; /* min freq = 6893 Hz */
+}
+
+
+/*
+ * vx_change_clock_source - change the clock source
+ * @source: the new source
+ */
+static void vx_change_clock_source(struct vx_core *chip, int source)
+{
+ unsigned long flags;
+
+ /* we mute DAC to prevent clicks */
+ vx_toggle_dac_mute(chip, 1);
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->ops->set_clock_source(chip, source);
+ chip->clock_source = source;
+ spin_unlock_irqrestore(&chip->lock, flags);
+ /* unmute */
+ vx_toggle_dac_mute(chip, 0);
+}
+
+
+/*
+ * set the internal clock
+ */
+void vx_set_internal_clock(struct vx_core *chip, unsigned int freq)
+{
+ int clock;
+ unsigned long flags;
+ /* Get real clock value */
+ clock = vx_calc_clock_from_freq(chip, freq);
+ snd_printdd(KERN_DEBUG "set internal clock to 0x%x from freq %d\n", clock, freq);
+ spin_lock_irqsave(&chip->lock, flags);
+ if (vx_is_pcmcia(chip)) {
+ vx_outb(chip, HIFREQ, (clock >> 8) & 0x0f);
+ vx_outb(chip, LOFREQ, clock & 0xff);
+ } else {
+ vx_outl(chip, HIFREQ, (clock >> 8) & 0x0f);
+ vx_outl(chip, LOFREQ, clock & 0xff);
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+/*
+ * set the iec958 status bits
+ * @bits: 32-bit status bits
+ */
+void vx_set_iec958_status(struct vx_core *chip, unsigned int bits)
+{
+ int i;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return;
+
+ for (i = 0; i < 32; i++)
+ vx_write_one_cbit(chip, i, bits & (1 << i));
+}
+
+
+/*
+ * vx_set_clock - change the clock and audio source if necessary
+ */
+int vx_set_clock(struct vx_core *chip, unsigned int freq)
+{
+ int src_changed = 0;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return 0;
+
+ /* change the audio source if possible */
+ vx_sync_audio_source(chip);
+
+ if (chip->clock_mode == VX_CLOCK_MODE_EXTERNAL ||
+ (chip->clock_mode == VX_CLOCK_MODE_AUTO &&
+ chip->audio_source == VX_AUDIO_SRC_DIGITAL)) {
+ if (chip->clock_source != UER_SYNC) {
+ vx_change_clock_source(chip, UER_SYNC);
+ mdelay(6);
+ src_changed = 1;
+ }
+ } else if (chip->clock_mode == VX_CLOCK_MODE_INTERNAL ||
+ (chip->clock_mode == VX_CLOCK_MODE_AUTO &&
+ chip->audio_source != VX_AUDIO_SRC_DIGITAL)) {
+ if (chip->clock_source != INTERNAL_QUARTZ) {
+ vx_change_clock_source(chip, INTERNAL_QUARTZ);
+ src_changed = 1;
+ }
+ if (chip->freq == freq)
+ return 0;
+ vx_set_internal_clock(chip, freq);
+ if (src_changed)
+ vx_modify_board_inputs(chip);
+ }
+ if (chip->freq == freq)
+ return 0;
+ chip->freq = freq;
+ vx_modify_board_clock(chip, 1);
+ return 0;
+}
+
+
+/*
+ * vx_change_frequency - called from interrupt handler
+ */
+int vx_change_frequency(struct vx_core *chip)
+{
+ int freq;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return 0;
+
+ if (chip->clock_source == INTERNAL_QUARTZ)
+ return 0;
+ /*
+ * Read the real UER board frequency
+ */
+ freq = vx_read_uer_status(chip, &chip->uer_detected);
+ if (freq < 0)
+ return freq;
+ /*
+ * The frequency computed by the DSP is good and
+ * is different from the previous computed.
+ */
+ if (freq == 48000 || freq == 44100 || freq == 32000)
+ chip->freq_detected = freq;
+
+ return 0;
+}
diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile
new file mode 100644
index 0000000..3797066
--- /dev/null
+++ b/sound/i2c/Makefile
@@ -0,0 +1,17 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-i2c-objs := i2c.o
+snd-cs8427-objs := cs8427.o
+snd-tea6330t-objs := tea6330t.o
+
+obj-$(CONFIG_L3) += l3/
+
+obj-$(CONFIG_SND) += other/
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_INTERWAVE_STB) += snd-tea6330t.o snd-i2c.o
+obj-$(CONFIG_SND_ICE1712) += snd-cs8427.o snd-i2c.o
+obj-$(CONFIG_SND_ICE1724) += snd-i2c.o
diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c
new file mode 100644
index 0000000..020a5d5
--- /dev/null
+++ b/sound/i2c/cs8427.c
@@ -0,0 +1,626 @@
+/*
+ * Routines for control of the CS8427 via i2c bus
+ * IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <asm/unaligned.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/cs8427.h>
+#include <sound/asoundef.h>
+
+static void snd_cs8427_reset(struct snd_i2c_device *cs8427);
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic");
+MODULE_LICENSE("GPL");
+
+#define CS8427_ADDR (0x20>>1) /* fixed address */
+
+struct cs8427_stream {
+ struct snd_pcm_substream *substream;
+ char hw_status[24]; /* hardware status */
+ char def_status[24]; /* default status */
+ char pcm_status[24]; /* PCM private status */
+ char hw_udata[32];
+ struct snd_kcontrol *pcm_ctl;
+};
+
+struct cs8427 {
+ unsigned char regmap[0x14]; /* map of first 1 + 13 registers */
+ unsigned int rate;
+ unsigned int reset_timeout;
+ struct cs8427_stream playback;
+ struct cs8427_stream capture;
+};
+
+static unsigned char swapbits(unsigned char val)
+{
+ int bit;
+ unsigned char res = 0;
+ for (bit = 0; bit < 8; bit++) {
+ res <<= 1;
+ res |= val & 1;
+ val >>= 1;
+ }
+ return res;
+}
+
+int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg,
+ unsigned char val)
+{
+ int err;
+ unsigned char buf[2];
+
+ buf[0] = reg & 0x7f;
+ buf[1] = val;
+ if ((err = snd_i2c_sendbytes(device, buf, 2)) != 2) {
+ snd_printk(KERN_ERR "unable to send bytes 0x%02x:0x%02x "
+ "to CS8427 (%i)\n", buf[0], buf[1], err);
+ return err < 0 ? err : -EIO;
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_cs8427_reg_write);
+
+static int snd_cs8427_reg_read(struct snd_i2c_device *device, unsigned char reg)
+{
+ int err;
+ unsigned char buf;
+
+ if ((err = snd_i2c_sendbytes(device, &reg, 1)) != 1) {
+ snd_printk(KERN_ERR "unable to send register 0x%x byte "
+ "to CS8427\n", reg);
+ return err < 0 ? err : -EIO;
+ }
+ if ((err = snd_i2c_readbytes(device, &buf, 1)) != 1) {
+ snd_printk(KERN_ERR "unable to read register 0x%x byte "
+ "from CS8427\n", reg);
+ return err < 0 ? err : -EIO;
+ }
+ return buf;
+}
+
+static int snd_cs8427_select_corudata(struct snd_i2c_device *device, int udata)
+{
+ struct cs8427 *chip = device->private_data;
+ int err;
+
+ udata = udata ? CS8427_BSEL : 0;
+ if (udata != (chip->regmap[CS8427_REG_CSDATABUF] & udata)) {
+ chip->regmap[CS8427_REG_CSDATABUF] &= ~CS8427_BSEL;
+ chip->regmap[CS8427_REG_CSDATABUF] |= udata;
+ err = snd_cs8427_reg_write(device, CS8427_REG_CSDATABUF,
+ chip->regmap[CS8427_REG_CSDATABUF]);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int snd_cs8427_send_corudata(struct snd_i2c_device *device,
+ int udata,
+ unsigned char *ndata,
+ int count)
+{
+ struct cs8427 *chip = device->private_data;
+ char *hw_data = udata ?
+ chip->playback.hw_udata : chip->playback.hw_status;
+ char data[32];
+ int err, idx;
+
+ if (!memcmp(hw_data, ndata, count))
+ return 0;
+ if ((err = snd_cs8427_select_corudata(device, udata)) < 0)
+ return err;
+ memcpy(hw_data, ndata, count);
+ if (udata) {
+ memset(data, 0, sizeof(data));
+ if (memcmp(hw_data, data, count) == 0) {
+ chip->regmap[CS8427_REG_UDATABUF] &= ~CS8427_UBMMASK;
+ chip->regmap[CS8427_REG_UDATABUF] |= CS8427_UBMZEROS |
+ CS8427_EFTUI;
+ err = snd_cs8427_reg_write(device, CS8427_REG_UDATABUF,
+ chip->regmap[CS8427_REG_UDATABUF]);
+ return err < 0 ? err : 0;
+ }
+ }
+ data[0] = CS8427_REG_AUTOINC | CS8427_REG_CORU_DATABUF;
+ for (idx = 0; idx < count; idx++)
+ data[idx + 1] = swapbits(ndata[idx]);
+ if (snd_i2c_sendbytes(device, data, count + 1) != count + 1)
+ return -EIO;
+ return 1;
+}
+
+static void snd_cs8427_free(struct snd_i2c_device *device)
+{
+ kfree(device->private_data);
+}
+
+int snd_cs8427_create(struct snd_i2c_bus *bus,
+ unsigned char addr,
+ unsigned int reset_timeout,
+ struct snd_i2c_device **r_cs8427)
+{
+ static unsigned char initvals1[] = {
+ CS8427_REG_CONTROL1 | CS8427_REG_AUTOINC,
+ /* CS8427_REG_CONTROL1: RMCK to OMCK, valid PCM audio, disable mutes,
+ TCBL=output */
+ CS8427_SWCLK | CS8427_TCBLDIR,
+ /* CS8427_REG_CONTROL2: hold last valid audio sample, RMCK=256*Fs,
+ normal stereo operation */
+ 0x00,
+ /* CS8427_REG_DATAFLOW: output drivers normal operation, Tx<=serial,
+ Rx=>serial */
+ CS8427_TXDSERIAL | CS8427_SPDAES3RECEIVER,
+ /* CS8427_REG_CLOCKSOURCE: Run off, CMCK=256*Fs,
+ output time base = OMCK, input time base = recovered input clock,
+ recovered input clock source is ILRCK changed to AES3INPUT
+ (workaround, see snd_cs8427_reset) */
+ CS8427_RXDILRCK,
+ /* CS8427_REG_SERIALINPUT: Serial audio input port data format = I2S,
+ 24-bit, 64*Fsi */
+ CS8427_SIDEL | CS8427_SILRPOL,
+ /* CS8427_REG_SERIALOUTPUT: Serial audio output port data format
+ = I2S, 24-bit, 64*Fsi */
+ CS8427_SODEL | CS8427_SOLRPOL,
+ };
+ static unsigned char initvals2[] = {
+ CS8427_REG_RECVERRMASK | CS8427_REG_AUTOINC,
+ /* CS8427_REG_RECVERRMASK: unmask the input PLL clock, V, confidence,
+ biphase, parity status bits */
+ /* CS8427_UNLOCK | CS8427_V | CS8427_CONF | CS8427_BIP | CS8427_PAR,*/
+ 0xff, /* set everything */
+ /* CS8427_REG_CSDATABUF:
+ Registers 32-55 window to CS buffer
+ Inhibit D->E transfers from overwriting first 5 bytes of CS data.
+ Inhibit D->E transfers (all) of CS data.
+ Allow E->F transfer of CS data.
+ One byte mode; both A/B channels get same written CB data.
+ A channel info is output to chip's EMPH* pin. */
+ CS8427_CBMR | CS8427_DETCI,
+ /* CS8427_REG_UDATABUF:
+ Use internal buffer to transmit User (U) data.
+ Chip's U pin is an output.
+ Transmit all O's for user data.
+ Inhibit D->E transfers.
+ Inhibit E->F transfers. */
+ CS8427_UD | CS8427_EFTUI | CS8427_DETUI,
+ };
+ int err;
+ struct cs8427 *chip;
+ struct snd_i2c_device *device;
+ unsigned char buf[24];
+
+ if ((err = snd_i2c_device_create(bus, "CS8427",
+ CS8427_ADDR | (addr & 7),
+ &device)) < 0)
+ return err;
+ chip = device->private_data = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ snd_i2c_device_free(device);
+ return -ENOMEM;
+ }
+ device->private_free = snd_cs8427_free;
+
+ snd_i2c_lock(bus);
+ err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER);
+ if (err != CS8427_VER8427A) {
+ /* give second chance */
+ snd_printk(KERN_WARNING "invalid CS8427 signature 0x%x: "
+ "let me try again...\n", err);
+ err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER);
+ }
+ if (err != CS8427_VER8427A) {
+ snd_i2c_unlock(bus);
+ snd_printk(KERN_ERR "unable to find CS8427 signature "
+ "(expected 0x%x, read 0x%x),\n",
+ CS8427_VER8427A, err);
+ snd_printk(KERN_ERR " initialization is not completed\n");
+ return -EFAULT;
+ }
+ /* turn off run bit while making changes to configuration */
+ err = snd_cs8427_reg_write(device, CS8427_REG_CLOCKSOURCE, 0x00);
+ if (err < 0)
+ goto __fail;
+ /* send initial values */
+ memcpy(chip->regmap + (initvals1[0] & 0x7f), initvals1 + 1, 6);
+ if ((err = snd_i2c_sendbytes(device, initvals1, 7)) != 7) {
+ err = err < 0 ? err : -EIO;
+ goto __fail;
+ }
+ /* Turn off CS8427 interrupt stuff that is not used in hardware */
+ memset(buf, 0, 7);
+ /* from address 9 to 15 */
+ buf[0] = 9; /* register */
+ if ((err = snd_i2c_sendbytes(device, buf, 7)) != 7)
+ goto __fail;
+ /* send transfer initialization sequence */
+ memcpy(chip->regmap + (initvals2[0] & 0x7f), initvals2 + 1, 3);
+ if ((err = snd_i2c_sendbytes(device, initvals2, 4)) != 4) {
+ err = err < 0 ? err : -EIO;
+ goto __fail;
+ }
+ /* write default channel status bytes */
+ put_unaligned_le32(SNDRV_PCM_DEFAULT_CON_SPDIF, buf);
+ memset(buf + 4, 0, 24 - 4);
+ if (snd_cs8427_send_corudata(device, 0, buf, 24) < 0)
+ goto __fail;
+ memcpy(chip->playback.def_status, buf, 24);
+ memcpy(chip->playback.pcm_status, buf, 24);
+ snd_i2c_unlock(bus);
+
+ /* turn on run bit and rock'n'roll */
+ if (reset_timeout < 1)
+ reset_timeout = 1;
+ chip->reset_timeout = reset_timeout;
+ snd_cs8427_reset(device);
+
+#if 0 // it's nice for read tests
+ {
+ char buf[128];
+ int xx;
+ buf[0] = 0x81;
+ snd_i2c_sendbytes(device, buf, 1);
+ snd_i2c_readbytes(device, buf, 127);
+ for (xx = 0; xx < 127; xx++)
+ printk(KERN_DEBUG "reg[0x%x] = 0x%x\n", xx+1, buf[xx]);
+ }
+#endif
+
+ if (r_cs8427)
+ *r_cs8427 = device;
+ return 0;
+
+ __fail:
+ snd_i2c_unlock(bus);
+ snd_i2c_device_free(device);
+ return err < 0 ? err : -EIO;
+}
+
+EXPORT_SYMBOL(snd_cs8427_create);
+
+/*
+ * Reset the chip using run bit, also lock PLL using ILRCK and
+ * put back AES3INPUT. This workaround is described in latest
+ * CS8427 datasheet, otherwise TXDSERIAL will not work.
+ */
+static void snd_cs8427_reset(struct snd_i2c_device *cs8427)
+{
+ struct cs8427 *chip;
+ unsigned long end_time;
+ int data, aes3input = 0;
+
+ if (snd_BUG_ON(!cs8427))
+ return;
+ chip = cs8427->private_data;
+ snd_i2c_lock(cs8427->bus);
+ if ((chip->regmap[CS8427_REG_CLOCKSOURCE] & CS8427_RXDAES3INPUT) ==
+ CS8427_RXDAES3INPUT) /* AES3 bit is set */
+ aes3input = 1;
+ chip->regmap[CS8427_REG_CLOCKSOURCE] &= ~(CS8427_RUN | CS8427_RXDMASK);
+ snd_cs8427_reg_write(cs8427, CS8427_REG_CLOCKSOURCE,
+ chip->regmap[CS8427_REG_CLOCKSOURCE]);
+ udelay(200);
+ chip->regmap[CS8427_REG_CLOCKSOURCE] |= CS8427_RUN | CS8427_RXDILRCK;
+ snd_cs8427_reg_write(cs8427, CS8427_REG_CLOCKSOURCE,
+ chip->regmap[CS8427_REG_CLOCKSOURCE]);
+ udelay(200);
+ snd_i2c_unlock(cs8427->bus);
+ end_time = jiffies + chip->reset_timeout;
+ while (time_after_eq(end_time, jiffies)) {
+ snd_i2c_lock(cs8427->bus);
+ data = snd_cs8427_reg_read(cs8427, CS8427_REG_RECVERRORS);
+ snd_i2c_unlock(cs8427->bus);
+ if (!(data & CS8427_UNLOCK))
+ break;
+ schedule_timeout_uninterruptible(1);
+ }
+ snd_i2c_lock(cs8427->bus);
+ chip->regmap[CS8427_REG_CLOCKSOURCE] &= ~CS8427_RXDMASK;
+ if (aes3input)
+ chip->regmap[CS8427_REG_CLOCKSOURCE] |= CS8427_RXDAES3INPUT;
+ snd_cs8427_reg_write(cs8427, CS8427_REG_CLOCKSOURCE,
+ chip->regmap[CS8427_REG_CLOCKSOURCE]);
+ snd_i2c_unlock(cs8427->bus);
+}
+
+static int snd_cs8427_in_status_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_cs8427_in_status_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_i2c_device *device = snd_kcontrol_chip(kcontrol);
+ int data;
+
+ snd_i2c_lock(device->bus);
+ data = snd_cs8427_reg_read(device, kcontrol->private_value);
+ snd_i2c_unlock(device->bus);
+ if (data < 0)
+ return data;
+ ucontrol->value.integer.value[0] = data;
+ return 0;
+}
+
+static int snd_cs8427_qsubcode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = 10;
+ return 0;
+}
+
+static int snd_cs8427_qsubcode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_i2c_device *device = snd_kcontrol_chip(kcontrol);
+ unsigned char reg = CS8427_REG_QSUBCODE;
+ int err;
+
+ snd_i2c_lock(device->bus);
+ if ((err = snd_i2c_sendbytes(device, &reg, 1)) != 1) {
+ snd_printk(KERN_ERR "unable to send register 0x%x byte "
+ "to CS8427\n", reg);
+ snd_i2c_unlock(device->bus);
+ return err < 0 ? err : -EIO;
+ }
+ err = snd_i2c_readbytes(device, ucontrol->value.bytes.data, 10);
+ if (err != 10) {
+ snd_printk(KERN_ERR "unable to read Q-subcode bytes "
+ "from CS8427\n");
+ snd_i2c_unlock(device->bus);
+ return err < 0 ? err : -EIO;
+ }
+ snd_i2c_unlock(device->bus);
+ return 0;
+}
+
+static int snd_cs8427_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cs8427_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_i2c_device *device = snd_kcontrol_chip(kcontrol);
+ struct cs8427 *chip = device->private_data;
+
+ snd_i2c_lock(device->bus);
+ memcpy(ucontrol->value.iec958.status, chip->playback.def_status, 24);
+ snd_i2c_unlock(device->bus);
+ return 0;
+}
+
+static int snd_cs8427_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_i2c_device *device = snd_kcontrol_chip(kcontrol);
+ struct cs8427 *chip = device->private_data;
+ unsigned char *status = kcontrol->private_value ?
+ chip->playback.pcm_status : chip->playback.def_status;
+ struct snd_pcm_runtime *runtime = chip->playback.substream ?
+ chip->playback.substream->runtime : NULL;
+ int err, change;
+
+ snd_i2c_lock(device->bus);
+ change = memcmp(ucontrol->value.iec958.status, status, 24) != 0;
+ memcpy(status, ucontrol->value.iec958.status, 24);
+ if (change && (kcontrol->private_value ?
+ runtime != NULL : runtime == NULL)) {
+ err = snd_cs8427_send_corudata(device, 0, status, 24);
+ if (err < 0)
+ change = err;
+ }
+ snd_i2c_unlock(device->bus);
+ return change;
+}
+
+static int snd_cs8427_spdif_mask_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cs8427_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ memset(ucontrol->value.iec958.status, 0xff, 24);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_cs8427_iec958_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .info = snd_cs8427_in_status_info,
+ .name = "IEC958 CS8427 Input Status",
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+ .get = snd_cs8427_in_status_get,
+ .private_value = 15,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .info = snd_cs8427_in_status_info,
+ .name = "IEC958 CS8427 Error Status",
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+ .get = snd_cs8427_in_status_get,
+ .private_value = 16,
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .info = snd_cs8427_spdif_mask_info,
+ .get = snd_cs8427_spdif_mask_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_cs8427_spdif_info,
+ .get = snd_cs8427_spdif_get,
+ .put = snd_cs8427_spdif_put,
+ .private_value = 0
+},
+{
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE),
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_cs8427_spdif_info,
+ .get = snd_cs8427_spdif_get,
+ .put = snd_cs8427_spdif_put,
+ .private_value = 1
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .info = snd_cs8427_qsubcode_info,
+ .name = "IEC958 Q-subcode Capture Default",
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+ .get = snd_cs8427_qsubcode_get
+}};
+
+int snd_cs8427_iec958_build(struct snd_i2c_device *cs8427,
+ struct snd_pcm_substream *play_substream,
+ struct snd_pcm_substream *cap_substream)
+{
+ struct cs8427 *chip = cs8427->private_data;
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!play_substream || !cap_substream))
+ return -EINVAL;
+ for (idx = 0; idx < ARRAY_SIZE(snd_cs8427_iec958_controls); idx++) {
+ kctl = snd_ctl_new1(&snd_cs8427_iec958_controls[idx], cs8427);
+ if (kctl == NULL)
+ return -ENOMEM;
+ kctl->id.device = play_substream->pcm->device;
+ kctl->id.subdevice = play_substream->number;
+ err = snd_ctl_add(cs8427->bus->card, kctl);
+ if (err < 0)
+ return err;
+ if (! strcmp(kctl->id.name,
+ SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM)))
+ chip->playback.pcm_ctl = kctl;
+ }
+
+ chip->playback.substream = play_substream;
+ chip->capture.substream = cap_substream;
+ if (snd_BUG_ON(!chip->playback.pcm_ctl))
+ return -EIO;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_cs8427_iec958_build);
+
+int snd_cs8427_iec958_active(struct snd_i2c_device *cs8427, int active)
+{
+ struct cs8427 *chip;
+
+ if (snd_BUG_ON(!cs8427))
+ return -ENXIO;
+ chip = cs8427->private_data;
+ if (active)
+ memcpy(chip->playback.pcm_status,
+ chip->playback.def_status, 24);
+ chip->playback.pcm_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(cs8427->bus->card,
+ SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO,
+ &chip->playback.pcm_ctl->id);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_cs8427_iec958_active);
+
+int snd_cs8427_iec958_pcm(struct snd_i2c_device *cs8427, unsigned int rate)
+{
+ struct cs8427 *chip;
+ char *status;
+ int err, reset;
+
+ if (snd_BUG_ON(!cs8427))
+ return -ENXIO;
+ chip = cs8427->private_data;
+ status = chip->playback.pcm_status;
+ snd_i2c_lock(cs8427->bus);
+ if (status[0] & IEC958_AES0_PROFESSIONAL) {
+ status[0] &= ~IEC958_AES0_PRO_FS;
+ switch (rate) {
+ case 32000: status[0] |= IEC958_AES0_PRO_FS_32000; break;
+ case 44100: status[0] |= IEC958_AES0_PRO_FS_44100; break;
+ case 48000: status[0] |= IEC958_AES0_PRO_FS_48000; break;
+ default: status[0] |= IEC958_AES0_PRO_FS_NOTID; break;
+ }
+ } else {
+ status[3] &= ~IEC958_AES3_CON_FS;
+ switch (rate) {
+ case 32000: status[3] |= IEC958_AES3_CON_FS_32000; break;
+ case 44100: status[3] |= IEC958_AES3_CON_FS_44100; break;
+ case 48000: status[3] |= IEC958_AES3_CON_FS_48000; break;
+ }
+ }
+ err = snd_cs8427_send_corudata(cs8427, 0, status, 24);
+ if (err > 0)
+ snd_ctl_notify(cs8427->bus->card,
+ SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->playback.pcm_ctl->id);
+ reset = chip->rate != rate;
+ chip->rate = rate;
+ snd_i2c_unlock(cs8427->bus);
+ if (reset)
+ snd_cs8427_reset(cs8427);
+ return err < 0 ? err : 0;
+}
+
+EXPORT_SYMBOL(snd_cs8427_iec958_pcm);
+
+static int __init alsa_cs8427_module_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_cs8427_module_exit(void)
+{
+}
+
+module_init(alsa_cs8427_module_init)
+module_exit(alsa_cs8427_module_exit)
diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c
new file mode 100644
index 0000000..5c0c77d
--- /dev/null
+++ b/sound/i2c/i2c.c
@@ -0,0 +1,346 @@
+/*
+ * Generic i2c interface for ALSA
+ *
+ * (c) 1998 Gerd Knorr <kraxel@cs.tu-berlin.de>
+ * Modified for the ALSA driver by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/i2c.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Generic i2c interface for ALSA");
+MODULE_LICENSE("GPL");
+
+static int snd_i2c_bit_sendbytes(struct snd_i2c_device *device,
+ unsigned char *bytes, int count);
+static int snd_i2c_bit_readbytes(struct snd_i2c_device *device,
+ unsigned char *bytes, int count);
+static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus,
+ unsigned short addr);
+
+static struct snd_i2c_ops snd_i2c_bit_ops = {
+ .sendbytes = snd_i2c_bit_sendbytes,
+ .readbytes = snd_i2c_bit_readbytes,
+ .probeaddr = snd_i2c_bit_probeaddr,
+};
+
+static int snd_i2c_bus_free(struct snd_i2c_bus *bus)
+{
+ struct snd_i2c_bus *slave;
+ struct snd_i2c_device *device;
+
+ if (snd_BUG_ON(!bus))
+ return -EINVAL;
+ while (!list_empty(&bus->devices)) {
+ device = snd_i2c_device(bus->devices.next);
+ snd_i2c_device_free(device);
+ }
+ if (bus->master)
+ list_del(&bus->buses);
+ else {
+ while (!list_empty(&bus->buses)) {
+ slave = snd_i2c_slave_bus(bus->buses.next);
+ snd_device_free(bus->card, slave);
+ }
+ }
+ if (bus->private_free)
+ bus->private_free(bus);
+ kfree(bus);
+ return 0;
+}
+
+static int snd_i2c_bus_dev_free(struct snd_device *device)
+{
+ struct snd_i2c_bus *bus = device->device_data;
+ return snd_i2c_bus_free(bus);
+}
+
+int snd_i2c_bus_create(struct snd_card *card, const char *name,
+ struct snd_i2c_bus *master, struct snd_i2c_bus **ri2c)
+{
+ struct snd_i2c_bus *bus;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_i2c_bus_dev_free,
+ };
+
+ *ri2c = NULL;
+ bus = kzalloc(sizeof(*bus), GFP_KERNEL);
+ if (bus == NULL)
+ return -ENOMEM;
+ mutex_init(&bus->lock_mutex);
+ INIT_LIST_HEAD(&bus->devices);
+ INIT_LIST_HEAD(&bus->buses);
+ bus->card = card;
+ bus->ops = &snd_i2c_bit_ops;
+ if (master) {
+ list_add_tail(&bus->buses, &master->buses);
+ bus->master = master;
+ }
+ strlcpy(bus->name, name, sizeof(bus->name));
+ if ((err = snd_device_new(card, SNDRV_DEV_BUS, bus, &ops)) < 0) {
+ snd_i2c_bus_free(bus);
+ return err;
+ }
+ *ri2c = bus;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_i2c_bus_create);
+
+int snd_i2c_device_create(struct snd_i2c_bus *bus, const char *name,
+ unsigned char addr, struct snd_i2c_device **rdevice)
+{
+ struct snd_i2c_device *device;
+
+ *rdevice = NULL;
+ if (snd_BUG_ON(!bus))
+ return -EINVAL;
+ device = kzalloc(sizeof(*device), GFP_KERNEL);
+ if (device == NULL)
+ return -ENOMEM;
+ device->addr = addr;
+ strlcpy(device->name, name, sizeof(device->name));
+ list_add_tail(&device->list, &bus->devices);
+ device->bus = bus;
+ *rdevice = device;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_i2c_device_create);
+
+int snd_i2c_device_free(struct snd_i2c_device *device)
+{
+ if (device->bus)
+ list_del(&device->list);
+ if (device->private_free)
+ device->private_free(device);
+ kfree(device);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_i2c_device_free);
+
+int snd_i2c_sendbytes(struct snd_i2c_device *device, unsigned char *bytes, int count)
+{
+ return device->bus->ops->sendbytes(device, bytes, count);
+}
+
+EXPORT_SYMBOL(snd_i2c_sendbytes);
+
+int snd_i2c_readbytes(struct snd_i2c_device *device, unsigned char *bytes, int count)
+{
+ return device->bus->ops->readbytes(device, bytes, count);
+}
+
+EXPORT_SYMBOL(snd_i2c_readbytes);
+
+int snd_i2c_probeaddr(struct snd_i2c_bus *bus, unsigned short addr)
+{
+ return bus->ops->probeaddr(bus, addr);
+}
+
+EXPORT_SYMBOL(snd_i2c_probeaddr);
+
+/*
+ * bit-operations
+ */
+
+static inline void snd_i2c_bit_hw_start(struct snd_i2c_bus *bus)
+{
+ if (bus->hw_ops.bit->start)
+ bus->hw_ops.bit->start(bus);
+}
+
+static inline void snd_i2c_bit_hw_stop(struct snd_i2c_bus *bus)
+{
+ if (bus->hw_ops.bit->stop)
+ bus->hw_ops.bit->stop(bus);
+}
+
+static void snd_i2c_bit_direction(struct snd_i2c_bus *bus, int clock, int data)
+{
+ if (bus->hw_ops.bit->direction)
+ bus->hw_ops.bit->direction(bus, clock, data);
+}
+
+static void snd_i2c_bit_set(struct snd_i2c_bus *bus, int clock, int data)
+{
+ bus->hw_ops.bit->setlines(bus, clock, data);
+}
+
+#if 0
+static int snd_i2c_bit_clock(struct snd_i2c_bus *bus)
+{
+ if (bus->hw_ops.bit->getclock)
+ return bus->hw_ops.bit->getclock(bus);
+ return -ENXIO;
+}
+#endif
+
+static int snd_i2c_bit_data(struct snd_i2c_bus *bus, int ack)
+{
+ return bus->hw_ops.bit->getdata(bus, ack);
+}
+
+static void snd_i2c_bit_start(struct snd_i2c_bus *bus)
+{
+ snd_i2c_bit_hw_start(bus);
+ snd_i2c_bit_direction(bus, 1, 1); /* SCL - wr, SDA - wr */
+ snd_i2c_bit_set(bus, 1, 1);
+ snd_i2c_bit_set(bus, 1, 0);
+ snd_i2c_bit_set(bus, 0, 0);
+}
+
+static void snd_i2c_bit_stop(struct snd_i2c_bus *bus)
+{
+ snd_i2c_bit_set(bus, 0, 0);
+ snd_i2c_bit_set(bus, 1, 0);
+ snd_i2c_bit_set(bus, 1, 1);
+ snd_i2c_bit_hw_stop(bus);
+}
+
+static void snd_i2c_bit_send(struct snd_i2c_bus *bus, int data)
+{
+ snd_i2c_bit_set(bus, 0, data);
+ snd_i2c_bit_set(bus, 1, data);
+ snd_i2c_bit_set(bus, 0, data);
+}
+
+static int snd_i2c_bit_ack(struct snd_i2c_bus *bus)
+{
+ int ack;
+
+ snd_i2c_bit_set(bus, 0, 1);
+ snd_i2c_bit_set(bus, 1, 1);
+ snd_i2c_bit_direction(bus, 1, 0); /* SCL - wr, SDA - rd */
+ ack = snd_i2c_bit_data(bus, 1);
+ snd_i2c_bit_direction(bus, 1, 1); /* SCL - wr, SDA - wr */
+ snd_i2c_bit_set(bus, 0, 1);
+ return ack ? -EIO : 0;
+}
+
+static int snd_i2c_bit_sendbyte(struct snd_i2c_bus *bus, unsigned char data)
+{
+ int i, err;
+
+ for (i = 7; i >= 0; i--)
+ snd_i2c_bit_send(bus, !!(data & (1 << i)));
+ if ((err = snd_i2c_bit_ack(bus)) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_i2c_bit_readbyte(struct snd_i2c_bus *bus, int last)
+{
+ int i;
+ unsigned char data = 0;
+
+ snd_i2c_bit_set(bus, 0, 1);
+ snd_i2c_bit_direction(bus, 1, 0); /* SCL - wr, SDA - rd */
+ for (i = 7; i >= 0; i--) {
+ snd_i2c_bit_set(bus, 1, 1);
+ if (snd_i2c_bit_data(bus, 0))
+ data |= (1 << i);
+ snd_i2c_bit_set(bus, 0, 1);
+ }
+ snd_i2c_bit_direction(bus, 1, 1); /* SCL - wr, SDA - wr */
+ snd_i2c_bit_send(bus, !!last);
+ return data;
+}
+
+static int snd_i2c_bit_sendbytes(struct snd_i2c_device *device,
+ unsigned char *bytes, int count)
+{
+ struct snd_i2c_bus *bus = device->bus;
+ int err, res = 0;
+
+ if (device->flags & SND_I2C_DEVICE_ADDRTEN)
+ return -EIO; /* not yet implemented */
+ snd_i2c_bit_start(bus);
+ if ((err = snd_i2c_bit_sendbyte(bus, device->addr << 1)) < 0) {
+ snd_i2c_bit_hw_stop(bus);
+ return err;
+ }
+ while (count-- > 0) {
+ if ((err = snd_i2c_bit_sendbyte(bus, *bytes++)) < 0) {
+ snd_i2c_bit_hw_stop(bus);
+ return err;
+ }
+ res++;
+ }
+ snd_i2c_bit_stop(bus);
+ return res;
+}
+
+static int snd_i2c_bit_readbytes(struct snd_i2c_device *device,
+ unsigned char *bytes, int count)
+{
+ struct snd_i2c_bus *bus = device->bus;
+ int err, res = 0;
+
+ if (device->flags & SND_I2C_DEVICE_ADDRTEN)
+ return -EIO; /* not yet implemented */
+ snd_i2c_bit_start(bus);
+ if ((err = snd_i2c_bit_sendbyte(bus, (device->addr << 1) | 1)) < 0) {
+ snd_i2c_bit_hw_stop(bus);
+ return err;
+ }
+ while (count-- > 0) {
+ if ((err = snd_i2c_bit_readbyte(bus, count == 0)) < 0) {
+ snd_i2c_bit_hw_stop(bus);
+ return err;
+ }
+ *bytes++ = (unsigned char)err;
+ res++;
+ }
+ snd_i2c_bit_stop(bus);
+ return res;
+}
+
+static int snd_i2c_bit_probeaddr(struct snd_i2c_bus *bus, unsigned short addr)
+{
+ int err;
+
+ if (addr & 0x8000) /* 10-bit address */
+ return -EIO; /* not yet implemented */
+ if (addr & 0x7f80) /* invalid address */
+ return -EINVAL;
+ snd_i2c_bit_start(bus);
+ err = snd_i2c_bit_sendbyte(bus, addr << 1);
+ snd_i2c_bit_stop(bus);
+ return err;
+}
+
+
+static int __init alsa_i2c_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_i2c_exit(void)
+{
+}
+
+module_init(alsa_i2c_init)
+module_exit(alsa_i2c_exit)
diff --git a/sound/i2c/l3/Makefile b/sound/i2c/l3/Makefile
new file mode 100644
index 0000000..49455b8
--- /dev/null
+++ b/sound/i2c/l3/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+#
+
+snd-uda1341-objs := uda1341.o
+
+# Module Dependency
+obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o
diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c
new file mode 100644
index 0000000..9840eb4
--- /dev/null
+++ b/sound/i2c/l3/uda1341.c
@@ -0,0 +1,935 @@
+/*
+ * Philips UDA1341 mixer device driver
+ * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
+ *
+ * Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License.
+ *
+ * History:
+ *
+ * 2002-03-13 Tomas Kasparek initial release - based on uda1341.c from OSS
+ * 2002-03-28 Tomas Kasparek basic mixer is working (volume, bass, treble)
+ * 2002-03-30 Tomas Kasparek proc filesystem support, complete mixer and DSP
+ * features support
+ * 2002-04-12 Tomas Kasparek proc interface update, code cleanup
+ * 2002-05-12 Tomas Kasparek another code cleanup
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/types.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/ioctl.h>
+
+#include <asm/uaccess.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+
+#include <linux/l3/l3.h>
+
+#include <sound/uda1341.h>
+
+/* {{{ HW regs definition */
+
+#define STAT0 0x00
+#define STAT1 0x80
+#define STAT_MASK 0x80
+
+#define DATA0_0 0x00
+#define DATA0_1 0x40
+#define DATA0_2 0x80
+#define DATA_MASK 0xc0
+
+#define IS_DATA0(x) ((x) >= data0_0 && (x) <= data0_2)
+#define IS_DATA1(x) ((x) == data1)
+#define IS_STATUS(x) ((x) == stat0 || (x) == stat1)
+#define IS_EXTEND(x) ((x) >= ext0 && (x) <= ext6)
+
+/* }}} */
+
+
+static const char *peak_names[] = {
+ "before",
+ "after",
+};
+
+static const char *filter_names[] = {
+ "flat",
+ "min",
+ "min",
+ "max",
+};
+
+static const char *mixer_names[] = {
+ "double differential",
+ "input channel 1 (line in)",
+ "input channel 2 (microphone)",
+ "digital mixer",
+};
+
+static const char *deemp_names[] = {
+ "none",
+ "32 kHz",
+ "44.1 kHz",
+ "48 kHz",
+};
+
+enum uda1341_regs_names {
+ stat0,
+ stat1,
+ data0_0,
+ data0_1,
+ data0_2,
+ data1,
+ ext0,
+ ext1,
+ ext2,
+ empty,
+ ext4,
+ ext5,
+ ext6,
+ uda1341_reg_last,
+};
+
+static const char *uda1341_reg_names[] = {
+ "stat 0 ",
+ "stat 1 ",
+ "data 00",
+ "data 01",
+ "data 02",
+ "data 1 ",
+ "ext 0",
+ "ext 1",
+ "ext 2",
+ "empty",
+ "ext 4",
+ "ext 5",
+ "ext 6",
+};
+
+static const int uda1341_enum_items[] = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 2, //peak - before/after
+ 4, //deemp - none/32/44.1/48
+ 0,
+ 4, //filter - flat/min/min/max
+ 0, 0, 0,
+ 4, //mixer - differ/line/mic/mixer
+ 0, 0, 0, 0, 0,
+};
+
+static const char ** uda1341_enum_names[] = {
+ NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL,
+ peak_names, //peak - before/after
+ deemp_names, //deemp - none/32/44.1/48
+ NULL,
+ filter_names, //filter - flat/min/min/max
+ NULL, NULL, NULL,
+ mixer_names, //mixer - differ/line/mic/mixer
+ NULL, NULL, NULL, NULL, NULL,
+};
+
+typedef int uda1341_cfg[CMD_LAST];
+
+struct uda1341 {
+ int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val);
+ int (*read) (struct l3_client *uda1341, unsigned short reg);
+ unsigned char regs[uda1341_reg_last];
+ int active;
+ spinlock_t reg_lock;
+ struct snd_card *card;
+ uda1341_cfg cfg;
+#ifdef CONFIG_PM
+ unsigned char suspend_regs[uda1341_reg_last];
+ uda1341_cfg suspend_cfg;
+#endif
+};
+
+/* transfer 8bit integer into string with binary representation */
+static void int2str_bin8(uint8_t val, char *buf)
+{
+ const int size = sizeof(val) * 8;
+ int i;
+
+ for (i= 0; i < size; i++){
+ *(buf++) = (val >> (size - 1)) ? '1' : '0';
+ val <<= 1;
+ }
+ *buf = '\0'; //end the string with zero
+}
+
+/* {{{ HW manipulation routines */
+
+static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val)
+{
+ struct uda1341 *uda = clnt->driver_data;
+ unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing
+ int err = 0;
+
+ uda->regs[reg] = val;
+
+ if (uda->active) {
+ if (IS_DATA0(reg)) {
+ err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1);
+ } else if (IS_DATA1(reg)) {
+ err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1);
+ } else if (IS_STATUS(reg)) {
+ err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1);
+ } else if (IS_EXTEND(reg)) {
+ buf[0] |= (reg - ext0) & 0x7; //EXT address
+ buf[1] |= val; //EXT data
+ err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2);
+ }
+ } else
+ printk(KERN_ERR "UDA1341 codec not active!\n");
+ return err;
+}
+
+static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg)
+{
+ unsigned char val;
+ int err;
+
+ err = l3_read(clnt, reg, &val, 1);
+ if (err == 1)
+ // use just 6bits - the rest is address of the reg
+ return val & 63;
+ return err < 0 ? err : -EIO;
+}
+
+static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg)
+{
+ return reg < uda1341_reg_last;
+}
+
+static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg,
+ unsigned short mask, unsigned short shift,
+ unsigned short value, int flush)
+{
+ int change;
+ unsigned short old, new;
+ struct uda1341 *uda = clnt->driver_data;
+
+#if 0
+ printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n",
+ uda1341_reg_names[reg], mask, shift, value);
+#endif
+
+ if (!snd_uda1341_valid_reg(clnt, reg))
+ return -EINVAL;
+ spin_lock(&uda->reg_lock);
+ old = uda->regs[reg];
+ new = (old & ~(mask << shift)) | (value << shift);
+ change = old != new;
+ if (change) {
+ if (flush) uda->write(clnt, reg, new);
+ uda->regs[reg] = new;
+ }
+ spin_unlock(&uda->reg_lock);
+ return change;
+}
+
+static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what,
+ unsigned short value, int flush)
+{
+ struct uda1341 *uda = clnt->driver_data;
+ int ret = 0;
+#ifdef CONFIG_PM
+ int reg;
+#endif
+
+#if 0
+ printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value);
+#endif
+
+ uda->cfg[what] = value;
+
+ switch(what) {
+ case CMD_RESET:
+ ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush); // MUTE
+ ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush); // RESET
+ ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush); // RESTORE
+ uda->cfg[CMD_RESET]=0;
+ break;
+ case CMD_FS:
+ ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush);
+ break;
+ case CMD_FORMAT:
+ ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush);
+ break;
+ case CMD_OGAIN:
+ ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush);
+ break;
+ case CMD_IGAIN:
+ ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush);
+ break;
+ case CMD_DAC:
+ ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush);
+ break;
+ case CMD_ADC:
+ ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush);
+ break;
+ case CMD_VOLUME:
+ ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush);
+ break;
+ case CMD_BASS:
+ ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush);
+ break;
+ case CMD_TREBBLE:
+ ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush);
+ break;
+ case CMD_PEAK:
+ ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush);
+ break;
+ case CMD_DEEMP:
+ ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush);
+ break;
+ case CMD_MUTE:
+ ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush);
+ break;
+ case CMD_FILTER:
+ ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush);
+ break;
+ case CMD_CH1:
+ ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush);
+ break;
+ case CMD_CH2:
+ ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush);
+ break;
+ case CMD_MIC:
+ ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush);
+ break;
+ case CMD_MIXER:
+ ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush);
+ break;
+ case CMD_AGC:
+ ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush);
+ break;
+ case CMD_IG:
+ ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush);
+ ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush);
+ break;
+ case CMD_AGC_TIME:
+ ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush);
+ break;
+ case CMD_AGC_LEVEL:
+ ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush);
+ break;
+#ifdef CONFIG_PM
+ case CMD_SUSPEND:
+ for (reg = stat0; reg < uda1341_reg_last; reg++)
+ uda->suspend_regs[reg] = uda->regs[reg];
+ for (reg = 0; reg < CMD_LAST; reg++)
+ uda->suspend_cfg[reg] = uda->cfg[reg];
+ break;
+ case CMD_RESUME:
+ for (reg = stat0; reg < uda1341_reg_last; reg++)
+ snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]);
+ for (reg = 0; reg < CMD_LAST; reg++)
+ uda->cfg[reg] = uda->suspend_cfg[reg];
+ break;
+#endif
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ if (!uda->active)
+ printk(KERN_ERR "UDA1341 codec not active!\n");
+ return ret;
+}
+
+/* }}} */
+
+/* {{{ Proc interface */
+#ifdef CONFIG_PROC_FS
+
+static const char *format_names[] = {
+ "I2S-bus",
+ "LSB 16bits",
+ "LSB 18bits",
+ "LSB 20bits",
+ "MSB",
+ "in LSB 16bits/out MSB",
+ "in LSB 18bits/out MSB",
+ "in LSB 20bits/out MSB",
+};
+
+static const char *fs_names[] = {
+ "512*fs",
+ "384*fs",
+ "256*fs",
+ "Unused - bad value!",
+};
+
+static const char* bass_values[][16] = {
+ {"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB",
+ "0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat
+ {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB",
+ "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min
+ {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB",
+ "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min
+ {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB",
+ "22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max
+};
+
+static const char *mic_sens_value[] = {
+ "-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used",
+};
+
+static const unsigned short AGC_atime[] = {
+ 11, 16, 11, 16, 21, 11, 16, 21,
+};
+
+static const unsigned short AGC_dtime[] = {
+ 100, 100, 200, 200, 200, 400, 400, 400,
+};
+
+static const char *AGC_level[] = {
+ "-9.0", "-11.5", "-15.0", "-17.5",
+};
+
+static const char *ig_small_value[] = {
+ "-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5",
+};
+
+/*
+ * this was computed as peak_value[i] = pow((63-i)*1.42,1.013)
+ *
+ * UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2
+ * There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29
+ * [61]=-2.78, [62] = -1.48, [63] = 0.0
+ * I tried to compute it, but using but even using logarithm with base either 10 or 2
+ * i was'n able to get values in the table from the formula. So I constructed another
+ * formula (see above) to interpolate the values as good as possible. If there is some
+ * mistake, please contact me on tomas.kasparek@seznam.cz. Thanks.
+ * UDA1341TS datasheet is available at:
+ * http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf
+ */
+static const char *peak_value[] = {
+ "-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB",
+ "-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB",
+ "-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB",
+ "-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB",
+ "-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB",
+ "-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB",
+ "-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB",
+ "-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB",
+ "-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB",
+};
+
+static void snd_uda1341_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct l3_client *clnt = entry->private_data;
+ struct uda1341 *uda = clnt->driver_data;
+ int peak;
+
+ peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1);
+ if (peak < 0)
+ peak = 0;
+
+ snd_iprintf(buffer, "%s\n\n", uda->card->longname);
+
+ // for information about computed values see UDA1341TS datasheet pages 15 - 21
+ snd_iprintf(buffer, "DAC power : %s\n", uda->cfg[CMD_DAC] ? "on" : "off");
+ snd_iprintf(buffer, "ADC power : %s\n", uda->cfg[CMD_ADC] ? "on" : "off");
+ snd_iprintf(buffer, "Clock frequency : %s\n", fs_names[uda->cfg[CMD_FS]]);
+ snd_iprintf(buffer, "Data format : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]);
+
+ snd_iprintf(buffer, "Filter mode : %s\n", filter_names[uda->cfg[CMD_FILTER]]);
+ snd_iprintf(buffer, "Mixer mode : %s\n", mixer_names[uda->cfg[CMD_MIXER]]);
+ snd_iprintf(buffer, "De-emphasis : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]);
+ snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before");
+ snd_iprintf(buffer, "Peak value : %s\n\n", peak_value[peak]);
+
+ snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off");
+ snd_iprintf(buffer, "AGC attack time : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]);
+ snd_iprintf(buffer, "AGC decay time : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]);
+ snd_iprintf(buffer, "AGC output level : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]);
+
+ snd_iprintf(buffer, "Mute : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off");
+
+ if (uda->cfg[CMD_VOLUME] == 0)
+ snd_iprintf(buffer, "Volume : 0 dB\n");
+ else if (uda->cfg[CMD_VOLUME] < 62)
+ snd_iprintf(buffer, "Volume : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1);
+ else
+ snd_iprintf(buffer, "Volume : -INF dB\n");
+ snd_iprintf(buffer, "Bass : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]);
+ snd_iprintf(buffer, "Trebble : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0);
+ snd_iprintf(buffer, "Input Gain (6dB) : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off");
+ snd_iprintf(buffer, "Output Gain (6dB) : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off");
+ snd_iprintf(buffer, "Mic sensitivity : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]);
+
+
+ if(uda->cfg[CMD_CH1] < 31)
+ snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n",
+ ((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1),
+ uda->cfg[CMD_CH1] & 1 ? '5' : '0');
+ else
+ snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n");
+ if(uda->cfg[CMD_CH2] < 31)
+ snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n",
+ ((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1),
+ uda->cfg[CMD_CH2] & 1 ? '5' : '0');
+ else
+ snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n");
+
+ if(uda->cfg[CMD_IG] > 5)
+ snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n",
+ (uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0');
+ else
+ snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n", ig_small_value[uda->cfg[CMD_IG]]);
+}
+
+static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct l3_client *clnt = entry->private_data;
+ struct uda1341 *uda = clnt->driver_data;
+ int reg;
+ char buf[12];
+
+ for (reg = 0; reg < uda1341_reg_last; reg ++) {
+ if (reg == empty)
+ continue;
+ int2str_bin8(uda->regs[reg], buf);
+ snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf);
+ }
+
+ int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf);
+ snd_iprintf(buffer, "DATA1 = %s\n", buf);
+}
+#endif /* CONFIG_PROC_FS */
+
+static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(card, "uda1341", &entry))
+ snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read);
+ if (! snd_card_proc_new(card, "uda1341-regs", &entry))
+ snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read);
+}
+
+/* }}} */
+
+/* {{{ Mixer controls setting */
+
+/* {{{ UDA1341 single functions */
+
+#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \
+ .get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \
+ .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \
+}
+
+static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 12) & 63;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
+ struct uda1341 *uda = clnt->driver_data;
+ int where = kcontrol->private_value & 31;
+ int mask = (kcontrol->private_value >> 12) & 63;
+ int invert = (kcontrol->private_value >> 18) & 1;
+
+ ucontrol->value.integer.value[0] = uda->cfg[where];
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
+ struct uda1341 *uda = clnt->driver_data;
+ int where = kcontrol->private_value & 31;
+ int reg = (kcontrol->private_value >> 5) & 15;
+ int shift = (kcontrol->private_value >> 9) & 7;
+ int mask = (kcontrol->private_value >> 12) & 63;
+ int invert = (kcontrol->private_value >> 18) & 1;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+
+ uda->cfg[where] = val;
+ return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH);
+}
+
+/* }}} */
+
+/* {{{ UDA1341 enum functions */
+
+#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \
+ .get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \
+ .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \
+}
+
+static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int where = kcontrol->private_value & 31;
+ const char **texts;
+
+ // this register we don't handle this way
+ if (!uda1341_enum_items[where])
+ return -EINVAL;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = uda1341_enum_items[where];
+
+ if (uinfo->value.enumerated.item >= uda1341_enum_items[where])
+ uinfo->value.enumerated.item = uda1341_enum_items[where] - 1;
+
+ texts = uda1341_enum_names[where];
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
+ struct uda1341 *uda = clnt->driver_data;
+ int where = kcontrol->private_value & 31;
+
+ ucontrol->value.enumerated.item[0] = uda->cfg[where];
+ return 0;
+}
+
+static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
+ struct uda1341 *uda = clnt->driver_data;
+ int where = kcontrol->private_value & 31;
+ int reg = (kcontrol->private_value >> 5) & 15;
+ int shift = (kcontrol->private_value >> 9) & 7;
+ int mask = (kcontrol->private_value >> 12) & 63;
+
+ uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask);
+
+ return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH);
+}
+
+/* }}} */
+
+/* {{{ UDA1341 2regs functions */
+
+#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \
+ .get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \
+ .private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \
+ (mask_1 << 19) | (mask_2 << 25) | (invert << 31) \
+}
+
+
+static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask_1 = (kcontrol->private_value >> 19) & 63;
+ int mask_2 = (kcontrol->private_value >> 25) & 63;
+ int mask;
+
+ mask = (mask_2 + 1) * (mask_1 + 1) - 1;
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
+ struct uda1341 *uda = clnt->driver_data;
+ int where = kcontrol->private_value & 31;
+ int mask_1 = (kcontrol->private_value >> 19) & 63;
+ int mask_2 = (kcontrol->private_value >> 25) & 63;
+ int invert = (kcontrol->private_value >> 31) & 1;
+ int mask;
+
+ mask = (mask_2 + 1) * (mask_1 + 1) - 1;
+
+ ucontrol->value.integer.value[0] = uda->cfg[where];
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
+ struct uda1341 *uda = clnt->driver_data;
+ int where = kcontrol->private_value & 31;
+ int reg_1 = (kcontrol->private_value >> 5) & 15;
+ int reg_2 = (kcontrol->private_value >> 9) & 15;
+ int shift_1 = (kcontrol->private_value >> 13) & 7;
+ int shift_2 = (kcontrol->private_value >> 16) & 7;
+ int mask_1 = (kcontrol->private_value >> 19) & 63;
+ int mask_2 = (kcontrol->private_value >> 25) & 63;
+ int invert = (kcontrol->private_value >> 31) & 1;
+ int mask;
+ unsigned short val1, val2, val;
+
+ val = ucontrol->value.integer.value[0];
+
+ mask = (mask_2 + 1) * (mask_1 + 1) - 1;
+
+ val1 = val & mask_1;
+ val2 = (val / (mask_1 + 1)) & mask_2;
+
+ if (invert) {
+ val1 = mask_1 - val1;
+ val2 = mask_2 - val2;
+ }
+
+ uda->cfg[where] = invert ? mask - val : val;
+
+ //FIXME - return value
+ snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH);
+ return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH);
+}
+
+/* }}} */
+
+static struct snd_kcontrol_new snd_uda1341_controls[] = {
+ UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1),
+ UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1),
+
+ UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0),
+ UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0),
+
+ UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0),
+ UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0),
+
+ UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1),
+ UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1),
+
+ UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0),
+
+ UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0),
+ UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0),
+ UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0),
+
+ UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0),
+ UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0),
+
+ UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0),
+ UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0),
+ UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0),
+ UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0),
+
+ UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0),
+};
+
+static void uda1341_free(struct l3_client *clnt)
+{
+ l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341)
+ kfree(clnt);
+}
+
+static int uda1341_dev_free(struct snd_device *device)
+{
+ struct l3_client *clnt = device->device_data;
+ uda1341_free(clnt);
+ return 0;
+}
+
+int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = uda1341_dev_free,
+ };
+ struct l3_client *clnt;
+ int idx, err;
+
+ if (snd_BUG_ON(!card))
+ return -EINVAL;
+
+ clnt = kzalloc(sizeof(*clnt), GFP_KERNEL);
+ if (clnt == NULL)
+ return -ENOMEM;
+
+ if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) {
+ kfree(clnt);
+ return err;
+ }
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) {
+ uda1341_free(clnt);
+ return err;
+ }
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) {
+ uda1341_free(clnt);
+ return err;
+ }
+
+ *clntp = clnt;
+ strcpy(card->mixername, "UDA1341TS Mixer");
+ ((struct uda1341 *)clnt->driver_data)->card = card;
+
+ snd_uda1341_proc_init(card, clnt);
+
+ return 0;
+}
+
+/* }}} */
+
+/* {{{ L3 operations */
+
+static int uda1341_attach(struct l3_client *clnt)
+{
+ struct uda1341 *uda;
+
+ uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL);
+ if (!uda)
+ return -ENOMEM;
+
+ /* init fixed parts of my copy of registers */
+ uda->regs[stat0] = STAT0;
+ uda->regs[stat1] = STAT1;
+
+ uda->regs[data0_0] = DATA0_0;
+ uda->regs[data0_1] = DATA0_1;
+ uda->regs[data0_2] = DATA0_2;
+
+ uda->write = snd_uda1341_codec_write;
+ uda->read = snd_uda1341_codec_read;
+
+ spin_lock_init(&uda->reg_lock);
+
+ clnt->driver_data = uda;
+ return 0;
+}
+
+static void uda1341_detach(struct l3_client *clnt)
+{
+ kfree(clnt->driver_data);
+}
+
+static int
+uda1341_command(struct l3_client *clnt, int cmd, void *arg)
+{
+ if (cmd != CMD_READ_REG)
+ return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH);
+
+ return snd_uda1341_codec_read(clnt, (int) arg);
+}
+
+static int uda1341_open(struct l3_client *clnt)
+{
+ struct uda1341 *uda = clnt->driver_data;
+
+ uda->active = 1;
+
+ /* init default configuration */
+ snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY);
+ snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH); // unknown state after reset
+ snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH); // unknown state after reset
+ snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH); // default off after reset
+ snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH); // default off after reset
+ snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH); // ??? default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH); // ??? default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH); // default 0dB after reset
+ snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY); // default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset
+ //at this moment should be QMUTED by h3600_audio_init
+ snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY); // default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH); // defaul flat after reset
+ snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH); // default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH); // default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH); // default 0dB after reset
+ snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH); // default doub.dif.mode
+ snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH); // default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH); // unknown state after reset
+ snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH); // default value after reset
+ snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH); // default value after reset
+
+ return 0;
+}
+
+static void uda1341_close(struct l3_client *clnt)
+{
+ struct uda1341 *uda = clnt->driver_data;
+
+ uda->active = 0;
+}
+
+/* }}} */
+
+/* {{{ Module and L3 initialization */
+
+static struct l3_ops uda1341_ops = {
+ .open = uda1341_open,
+ .command = uda1341_command,
+ .close = uda1341_close,
+};
+
+static struct l3_driver uda1341_driver = {
+ .name = UDA1341_ALSA_NAME,
+ .attach_client = uda1341_attach,
+ .detach_client = uda1341_detach,
+ .ops = &uda1341_ops,
+ .owner = THIS_MODULE,
+};
+
+static int __init uda1341_init(void)
+{
+ return l3_add_driver(&uda1341_driver);
+}
+
+static void __exit uda1341_exit(void)
+{
+ l3_del_driver(&uda1341_driver);
+}
+
+module_init(uda1341_init);
+module_exit(uda1341_exit);
+
+MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA");
+MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}");
+
+EXPORT_SYMBOL(snd_chip_uda1341_mixer_new);
+
+/* }}} */
+
+/*
+ * Local variables:
+ * indent-tabs-mode: t
+ * End:
+ */
diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile
new file mode 100644
index 0000000..703d954
--- /dev/null
+++ b/sound/i2c/other/Makefile
@@ -0,0 +1,16 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ak4114-objs := ak4114.o
+snd-ak4117-objs := ak4117.o
+snd-ak4xxx-adda-objs := ak4xxx-adda.o
+snd-pt2258-objs := pt2258.o
+snd-tea575x-tuner-objs := tea575x-tuner.o
+
+# Module Dependency
+obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o
+obj-$(CONFIG_SND_ICE1712) += snd-ak4xxx-adda.o
+obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4xxx-adda.o snd-pt2258.o
+obj-$(CONFIG_SND_FM801_TEA575X) += snd-tea575x-tuner.o
diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c
new file mode 100644
index 0000000..0341451
--- /dev/null
+++ b/sound/i2c/other/ak4114.c
@@ -0,0 +1,626 @@
+/*
+ * Routines for control of the AK4114 via I2C and 4-wire serial interface
+ * IEC958 (S/PDIF) receiver by Asahi Kasei
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/ak4114.h>
+#include <sound/asoundef.h>
+#include <sound/info.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("AK4114 IEC958 (S/PDIF) receiver by Asahi Kasei");
+MODULE_LICENSE("GPL");
+
+#define AK4114_ADDR 0x00 /* fixed address */
+
+static void ak4114_stats(struct work_struct *work);
+static void ak4114_init_regs(struct ak4114 *chip);
+
+static void reg_write(struct ak4114 *ak4114, unsigned char reg, unsigned char val)
+{
+ ak4114->write(ak4114->private_data, reg, val);
+ if (reg <= AK4114_REG_INT1_MASK)
+ ak4114->regmap[reg] = val;
+ else if (reg >= AK4114_REG_TXCSB0 && reg <= AK4114_REG_TXCSB4)
+ ak4114->txcsb[reg-AK4114_REG_TXCSB0] = val;
+}
+
+static inline unsigned char reg_read(struct ak4114 *ak4114, unsigned char reg)
+{
+ return ak4114->read(ak4114->private_data, reg);
+}
+
+#if 0
+static void reg_dump(struct ak4114 *ak4114)
+{
+ int i;
+
+ printk(KERN_DEBUG "AK4114 REG DUMP:\n");
+ for (i = 0; i < 0x20; i++)
+ printk(KERN_DEBUG "reg[%02x] = %02x (%02x)\n", i, reg_read(ak4114, i), i < sizeof(ak4114->regmap) ? ak4114->regmap[i] : 0);
+}
+#endif
+
+static void snd_ak4114_free(struct ak4114 *chip)
+{
+ chip->init = 1; /* don't schedule new work */
+ mb();
+ cancel_delayed_work(&chip->work);
+ flush_scheduled_work();
+ kfree(chip);
+}
+
+static int snd_ak4114_dev_free(struct snd_device *device)
+{
+ struct ak4114 *chip = device->device_data;
+ snd_ak4114_free(chip);
+ return 0;
+}
+
+int snd_ak4114_create(struct snd_card *card,
+ ak4114_read_t *read, ak4114_write_t *write,
+ const unsigned char pgm[7], const unsigned char txcsb[5],
+ void *private_data, struct ak4114 **r_ak4114)
+{
+ struct ak4114 *chip;
+ int err = 0;
+ unsigned char reg;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ak4114_dev_free,
+ };
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ spin_lock_init(&chip->lock);
+ chip->card = card;
+ chip->read = read;
+ chip->write = write;
+ chip->private_data = private_data;
+ INIT_DELAYED_WORK(&chip->work, ak4114_stats);
+
+ for (reg = 0; reg < 7; reg++)
+ chip->regmap[reg] = pgm[reg];
+ for (reg = 0; reg < 5; reg++)
+ chip->txcsb[reg] = txcsb[reg];
+
+ ak4114_init_regs(chip);
+
+ chip->rcs0 = reg_read(chip, AK4114_REG_RCS0) & ~(AK4114_QINT | AK4114_CINT);
+ chip->rcs1 = reg_read(chip, AK4114_REG_RCS1);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0)
+ goto __fail;
+
+ if (r_ak4114)
+ *r_ak4114 = chip;
+ return 0;
+
+ __fail:
+ snd_ak4114_free(chip);
+ return err < 0 ? err : -EIO;
+}
+
+void snd_ak4114_reg_write(struct ak4114 *chip, unsigned char reg, unsigned char mask, unsigned char val)
+{
+ if (reg <= AK4114_REG_INT1_MASK)
+ reg_write(chip, reg, (chip->regmap[reg] & ~mask) | val);
+ else if (reg >= AK4114_REG_TXCSB0 && reg <= AK4114_REG_TXCSB4)
+ reg_write(chip, reg,
+ (chip->txcsb[reg-AK4114_REG_TXCSB0] & ~mask) | val);
+}
+
+static void ak4114_init_regs(struct ak4114 *chip)
+{
+ unsigned char old = chip->regmap[AK4114_REG_PWRDN], reg;
+
+ /* bring the chip to reset state and powerdown state */
+ reg_write(chip, AK4114_REG_PWRDN, old & ~(AK4114_RST|AK4114_PWN));
+ udelay(200);
+ /* release reset, but leave powerdown */
+ reg_write(chip, AK4114_REG_PWRDN, (old | AK4114_RST) & ~AK4114_PWN);
+ udelay(200);
+ for (reg = 1; reg < 7; reg++)
+ reg_write(chip, reg, chip->regmap[reg]);
+ for (reg = 0; reg < 5; reg++)
+ reg_write(chip, reg + AK4114_REG_TXCSB0, chip->txcsb[reg]);
+ /* release powerdown, everything is initialized now */
+ reg_write(chip, AK4114_REG_PWRDN, old | AK4114_RST | AK4114_PWN);
+}
+
+void snd_ak4114_reinit(struct ak4114 *chip)
+{
+ chip->init = 1;
+ mb();
+ flush_scheduled_work();
+ ak4114_init_regs(chip);
+ /* bring up statistics / event queing */
+ chip->init = 0;
+ if (chip->kctls[0])
+ schedule_delayed_work(&chip->work, HZ / 10);
+}
+
+static unsigned int external_rate(unsigned char rcs1)
+{
+ switch (rcs1 & (AK4114_FS0|AK4114_FS1|AK4114_FS2|AK4114_FS3)) {
+ case AK4114_FS_32000HZ: return 32000;
+ case AK4114_FS_44100HZ: return 44100;
+ case AK4114_FS_48000HZ: return 48000;
+ case AK4114_FS_88200HZ: return 88200;
+ case AK4114_FS_96000HZ: return 96000;
+ case AK4114_FS_176400HZ: return 176400;
+ case AK4114_FS_192000HZ: return 192000;
+ default: return 0;
+ }
+}
+
+static int snd_ak4114_in_error_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = LONG_MAX;
+ return 0;
+}
+
+static int snd_ak4114_in_error_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+ long *ptr;
+
+ spin_lock_irq(&chip->lock);
+ ptr = (long *)(((char *)chip) + kcontrol->private_value);
+ ucontrol->value.integer.value[0] = *ptr;
+ *ptr = 0;
+ spin_unlock_irq(&chip->lock);
+ return 0;
+}
+
+#define snd_ak4114_in_bit_info snd_ctl_boolean_mono_info
+
+static int snd_ak4114_in_bit_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char reg = kcontrol->private_value & 0xff;
+ unsigned char bit = (kcontrol->private_value >> 8) & 0xff;
+ unsigned char inv = (kcontrol->private_value >> 31) & 1;
+
+ ucontrol->value.integer.value[0] = ((reg_read(chip, reg) & (1 << bit)) ? 1 : 0) ^ inv;
+ return 0;
+}
+
+static int snd_ak4114_rate_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 192000;
+ return 0;
+}
+
+static int snd_ak4114_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = external_rate(reg_read(chip, AK4114_REG_RCS1));
+ return 0;
+}
+
+static int snd_ak4114_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ak4114_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned i;
+
+ for (i = 0; i < AK4114_REG_RXCSB_SIZE; i++)
+ ucontrol->value.iec958.status[i] = reg_read(chip, AK4114_REG_RXCSB0 + i);
+ return 0;
+}
+
+static int snd_ak4114_spdif_playback_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned i;
+
+ for (i = 0; i < AK4114_REG_TXCSB_SIZE; i++)
+ ucontrol->value.iec958.status[i] = chip->txcsb[i];
+ return 0;
+}
+
+static int snd_ak4114_spdif_playback_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned i;
+
+ for (i = 0; i < AK4114_REG_TXCSB_SIZE; i++)
+ reg_write(chip, AK4114_REG_TXCSB0 + i, ucontrol->value.iec958.status[i]);
+ return 0;
+}
+
+static int snd_ak4114_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ak4114_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ memset(ucontrol->value.iec958.status, 0xff, AK4114_REG_RXCSB_SIZE);
+ return 0;
+}
+
+static int snd_ak4114_spdif_pinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xffff;
+ uinfo->count = 4;
+ return 0;
+}
+
+static int snd_ak4114_spdif_pget(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned short tmp;
+
+ ucontrol->value.integer.value[0] = 0xf8f2;
+ ucontrol->value.integer.value[1] = 0x4e1f;
+ tmp = reg_read(chip, AK4114_REG_Pc0) | (reg_read(chip, AK4114_REG_Pc1) << 8);
+ ucontrol->value.integer.value[2] = tmp;
+ tmp = reg_read(chip, AK4114_REG_Pd0) | (reg_read(chip, AK4114_REG_Pd1) << 8);
+ ucontrol->value.integer.value[3] = tmp;
+ return 0;
+}
+
+static int snd_ak4114_spdif_qinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = AK4114_REG_QSUB_SIZE;
+ return 0;
+}
+
+static int snd_ak4114_spdif_qget(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4114 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned i;
+
+ for (i = 0; i < AK4114_REG_QSUB_SIZE; i++)
+ ucontrol->value.bytes.data[i] = reg_read(chip, AK4114_REG_QSUB_ADDR + i);
+ return 0;
+}
+
+/* Don't forget to change AK4114_CONTROLS define!!! */
+static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Parity Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_error_info,
+ .get = snd_ak4114_in_error_get,
+ .private_value = offsetof(struct ak4114, parity_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 V-Bit Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_error_info,
+ .get = snd_ak4114_in_error_get,
+ .private_value = offsetof(struct ak4114, v_bit_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 C-CRC Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_error_info,
+ .get = snd_ak4114_in_error_get,
+ .private_value = offsetof(struct ak4114, ccrc_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Q-CRC Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_error_info,
+ .get = snd_ak4114_in_error_get,
+ .private_value = offsetof(struct ak4114, qcrc_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 External Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_rate_info,
+ .get = snd_ak4114_rate_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_ak4114_spdif_mask_info,
+ .get = snd_ak4114_spdif_mask_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_spdif_info,
+ .get = snd_ak4114_spdif_playback_get,
+ .put = snd_ak4114_spdif_playback_put,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,MASK),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_ak4114_spdif_mask_info,
+ .get = snd_ak4114_spdif_mask_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_spdif_info,
+ .get = snd_ak4114_spdif_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Preample Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_spdif_pinfo,
+ .get = snd_ak4114_spdif_pget,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Q-subcode Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_spdif_qinfo,
+ .get = snd_ak4114_spdif_qget,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Audio",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_bit_info,
+ .get = snd_ak4114_in_bit_get,
+ .private_value = (1<<31) | (1<<8) | AK4114_REG_RCS0,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Non-PCM Bitstream",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_bit_info,
+ .get = snd_ak4114_in_bit_get,
+ .private_value = (6<<8) | AK4114_REG_RCS0,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 DTS Bitstream",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_bit_info,
+ .get = snd_ak4114_in_bit_get,
+ .private_value = (3<<8) | AK4114_REG_RCS0,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 PPL Lock Status",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4114_in_bit_info,
+ .get = snd_ak4114_in_bit_get,
+ .private_value = (1<<31) | (4<<8) | AK4114_REG_RCS0,
+}
+};
+
+
+static void snd_ak4114_proc_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct ak4114 *ak4114 = entry->private_data;
+ int reg, val;
+ /* all ak4114 registers 0x00 - 0x1f */
+ for (reg = 0; reg < 0x20; reg++) {
+ val = reg_read(ak4114, reg);
+ snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val);
+ }
+}
+
+static void snd_ak4114_proc_init(struct ak4114 *ak4114)
+{
+ struct snd_info_entry *entry;
+ if (!snd_card_proc_new(ak4114->card, "ak4114", &entry))
+ snd_info_set_text_ops(entry, ak4114, snd_ak4114_proc_regs_read);
+}
+
+int snd_ak4114_build(struct ak4114 *ak4114,
+ struct snd_pcm_substream *ply_substream,
+ struct snd_pcm_substream *cap_substream)
+{
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!cap_substream))
+ return -EINVAL;
+ ak4114->playback_substream = ply_substream;
+ ak4114->capture_substream = cap_substream;
+ for (idx = 0; idx < AK4114_CONTROLS; idx++) {
+ kctl = snd_ctl_new1(&snd_ak4114_iec958_controls[idx], ak4114);
+ if (kctl == NULL)
+ return -ENOMEM;
+ if (strstr(kctl->id.name, "Playback")) {
+ if (ply_substream == NULL) {
+ snd_ctl_free_one(kctl);
+ ak4114->kctls[idx] = NULL;
+ continue;
+ }
+ kctl->id.device = ply_substream->pcm->device;
+ kctl->id.subdevice = ply_substream->number;
+ } else {
+ kctl->id.device = cap_substream->pcm->device;
+ kctl->id.subdevice = cap_substream->number;
+ }
+ err = snd_ctl_add(ak4114->card, kctl);
+ if (err < 0)
+ return err;
+ ak4114->kctls[idx] = kctl;
+ }
+ snd_ak4114_proc_init(ak4114);
+ /* trigger workq */
+ schedule_delayed_work(&ak4114->work, HZ / 10);
+ return 0;
+}
+
+/* notify kcontrols if any parameters are changed */
+static void ak4114_notify(struct ak4114 *ak4114,
+ unsigned char rcs0, unsigned char rcs1,
+ unsigned char c0, unsigned char c1)
+{
+ if (!ak4114->kctls[0])
+ return;
+
+ if (rcs0 & AK4114_PAR)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[0]->id);
+ if (rcs0 & AK4114_V)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[1]->id);
+ if (rcs1 & AK4114_CCRC)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[2]->id);
+ if (rcs1 & AK4114_QCRC)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[3]->id);
+
+ /* rate change */
+ if (c1 & 0xf0)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[4]->id);
+
+ if ((c0 & AK4114_PEM) | (c0 & AK4114_CINT))
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[9]->id);
+ if (c0 & AK4114_QINT)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[10]->id);
+
+ if (c0 & AK4114_AUDION)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[11]->id);
+ if (c0 & AK4114_AUTO)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[12]->id);
+ if (c0 & AK4114_DTSCD)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[13]->id);
+ if (c0 & AK4114_UNLCK)
+ snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &ak4114->kctls[14]->id);
+}
+
+int snd_ak4114_external_rate(struct ak4114 *ak4114)
+{
+ unsigned char rcs1;
+
+ rcs1 = reg_read(ak4114, AK4114_REG_RCS1);
+ return external_rate(rcs1);
+}
+
+int snd_ak4114_check_rate_and_errors(struct ak4114 *ak4114, unsigned int flags)
+{
+ struct snd_pcm_runtime *runtime = ak4114->capture_substream ? ak4114->capture_substream->runtime : NULL;
+ unsigned long _flags;
+ int res = 0;
+ unsigned char rcs0, rcs1;
+ unsigned char c0, c1;
+
+ rcs1 = reg_read(ak4114, AK4114_REG_RCS1);
+ if (flags & AK4114_CHECK_NO_STAT)
+ goto __rate;
+ rcs0 = reg_read(ak4114, AK4114_REG_RCS0);
+ spin_lock_irqsave(&ak4114->lock, _flags);
+ if (rcs0 & AK4114_PAR)
+ ak4114->parity_errors++;
+ if (rcs1 & AK4114_V)
+ ak4114->v_bit_errors++;
+ if (rcs1 & AK4114_CCRC)
+ ak4114->ccrc_errors++;
+ if (rcs1 & AK4114_QCRC)
+ ak4114->qcrc_errors++;
+ c0 = (ak4114->rcs0 & (AK4114_QINT | AK4114_CINT | AK4114_PEM | AK4114_AUDION | AK4114_AUTO | AK4114_UNLCK)) ^
+ (rcs0 & (AK4114_QINT | AK4114_CINT | AK4114_PEM | AK4114_AUDION | AK4114_AUTO | AK4114_UNLCK));
+ c1 = (ak4114->rcs1 & 0xf0) ^ (rcs1 & 0xf0);
+ ak4114->rcs0 = rcs0 & ~(AK4114_QINT | AK4114_CINT);
+ ak4114->rcs1 = rcs1;
+ spin_unlock_irqrestore(&ak4114->lock, _flags);
+
+ ak4114_notify(ak4114, rcs0, rcs1, c0, c1);
+ if (ak4114->change_callback && (c0 | c1) != 0)
+ ak4114->change_callback(ak4114, c0, c1);
+
+ __rate:
+ /* compare rate */
+ res = external_rate(rcs1);
+ if (!(flags & AK4114_CHECK_NO_RATE) && runtime && runtime->rate != res) {
+ snd_pcm_stream_lock_irqsave(ak4114->capture_substream, _flags);
+ if (snd_pcm_running(ak4114->capture_substream)) {
+ // printk(KERN_DEBUG "rate changed (%i <- %i)\n", runtime->rate, res);
+ snd_pcm_stop(ak4114->capture_substream, SNDRV_PCM_STATE_DRAINING);
+ res = 1;
+ }
+ snd_pcm_stream_unlock_irqrestore(ak4114->capture_substream, _flags);
+ }
+ return res;
+}
+
+static void ak4114_stats(struct work_struct *work)
+{
+ struct ak4114 *chip = container_of(work, struct ak4114, work.work);
+
+ if (!chip->init)
+ snd_ak4114_check_rate_and_errors(chip, chip->check_flags);
+
+ schedule_delayed_work(&chip->work, HZ / 10);
+}
+
+EXPORT_SYMBOL(snd_ak4114_create);
+EXPORT_SYMBOL(snd_ak4114_reg_write);
+EXPORT_SYMBOL(snd_ak4114_reinit);
+EXPORT_SYMBOL(snd_ak4114_build);
+EXPORT_SYMBOL(snd_ak4114_external_rate);
+EXPORT_SYMBOL(snd_ak4114_check_rate_and_errors);
diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c
new file mode 100644
index 0000000..2cad2d6
--- /dev/null
+++ b/sound/i2c/other/ak4117.c
@@ -0,0 +1,551 @@
+/*
+ * Routines for control of the AK4117 via 4-wire serial interface
+ * IEC958 (S/PDIF) receiver by Asahi Kasei
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/ak4117.h>
+#include <sound/asoundef.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("AK4117 IEC958 (S/PDIF) receiver by Asahi Kasei");
+MODULE_LICENSE("GPL");
+
+#define AK4117_ADDR 0x00 /* fixed address */
+
+static void snd_ak4117_timer(unsigned long data);
+
+static void reg_write(struct ak4117 *ak4117, unsigned char reg, unsigned char val)
+{
+ ak4117->write(ak4117->private_data, reg, val);
+ if (reg < sizeof(ak4117->regmap))
+ ak4117->regmap[reg] = val;
+}
+
+static inline unsigned char reg_read(struct ak4117 *ak4117, unsigned char reg)
+{
+ return ak4117->read(ak4117->private_data, reg);
+}
+
+#if 0
+static void reg_dump(struct ak4117 *ak4117)
+{
+ int i;
+
+ printk(KERN_DEBUG "AK4117 REG DUMP:\n");
+ for (i = 0; i < 0x1b; i++)
+ printk(KERN_DEBUG "reg[%02x] = %02x (%02x)\n", i, reg_read(ak4117, i), i < sizeof(ak4117->regmap) ? ak4117->regmap[i] : 0);
+}
+#endif
+
+static void snd_ak4117_free(struct ak4117 *chip)
+{
+ del_timer(&chip->timer);
+ kfree(chip);
+}
+
+static int snd_ak4117_dev_free(struct snd_device *device)
+{
+ struct ak4117 *chip = device->device_data;
+ snd_ak4117_free(chip);
+ return 0;
+}
+
+int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t *write,
+ const unsigned char pgm[5], void *private_data, struct ak4117 **r_ak4117)
+{
+ struct ak4117 *chip;
+ int err = 0;
+ unsigned char reg;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ak4117_dev_free,
+ };
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ spin_lock_init(&chip->lock);
+ chip->card = card;
+ chip->read = read;
+ chip->write = write;
+ chip->private_data = private_data;
+ init_timer(&chip->timer);
+ chip->timer.data = (unsigned long)chip;
+ chip->timer.function = snd_ak4117_timer;
+
+ for (reg = 0; reg < 5; reg++)
+ chip->regmap[reg] = pgm[reg];
+ snd_ak4117_reinit(chip);
+
+ chip->rcs0 = reg_read(chip, AK4117_REG_RCS0) & ~(AK4117_QINT | AK4117_CINT | AK4117_STC);
+ chip->rcs1 = reg_read(chip, AK4117_REG_RCS1);
+ chip->rcs2 = reg_read(chip, AK4117_REG_RCS2);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_CODEC, chip, &ops)) < 0)
+ goto __fail;
+
+ if (r_ak4117)
+ *r_ak4117 = chip;
+ return 0;
+
+ __fail:
+ snd_ak4117_free(chip);
+ return err < 0 ? err : -EIO;
+}
+
+void snd_ak4117_reg_write(struct ak4117 *chip, unsigned char reg, unsigned char mask, unsigned char val)
+{
+ if (reg >= 5)
+ return;
+ reg_write(chip, reg, (chip->regmap[reg] & ~mask) | val);
+}
+
+void snd_ak4117_reinit(struct ak4117 *chip)
+{
+ unsigned char old = chip->regmap[AK4117_REG_PWRDN], reg;
+
+ del_timer(&chip->timer);
+ chip->init = 1;
+ /* bring the chip to reset state and powerdown state */
+ reg_write(chip, AK4117_REG_PWRDN, 0);
+ udelay(200);
+ /* release reset, but leave powerdown */
+ reg_write(chip, AK4117_REG_PWRDN, (old | AK4117_RST) & ~AK4117_PWN);
+ udelay(200);
+ for (reg = 1; reg < 5; reg++)
+ reg_write(chip, reg, chip->regmap[reg]);
+ /* release powerdown, everything is initialized now */
+ reg_write(chip, AK4117_REG_PWRDN, old | AK4117_RST | AK4117_PWN);
+ chip->init = 0;
+ chip->timer.expires = 1 + jiffies;
+ add_timer(&chip->timer);
+}
+
+static unsigned int external_rate(unsigned char rcs1)
+{
+ switch (rcs1 & (AK4117_FS0|AK4117_FS1|AK4117_FS2|AK4117_FS3)) {
+ case AK4117_FS_32000HZ: return 32000;
+ case AK4117_FS_44100HZ: return 44100;
+ case AK4117_FS_48000HZ: return 48000;
+ case AK4117_FS_88200HZ: return 88200;
+ case AK4117_FS_96000HZ: return 96000;
+ case AK4117_FS_176400HZ: return 176400;
+ case AK4117_FS_192000HZ: return 192000;
+ default: return 0;
+ }
+}
+
+static int snd_ak4117_in_error_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = LONG_MAX;
+ return 0;
+}
+
+static int snd_ak4117_in_error_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+ long *ptr;
+
+ spin_lock_irq(&chip->lock);
+ ptr = (long *)(((char *)chip) + kcontrol->private_value);
+ ucontrol->value.integer.value[0] = *ptr;
+ *ptr = 0;
+ spin_unlock_irq(&chip->lock);
+ return 0;
+}
+
+#define snd_ak4117_in_bit_info snd_ctl_boolean_mono_info
+
+static int snd_ak4117_in_bit_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char reg = kcontrol->private_value & 0xff;
+ unsigned char bit = (kcontrol->private_value >> 8) & 0xff;
+ unsigned char inv = (kcontrol->private_value >> 31) & 1;
+
+ ucontrol->value.integer.value[0] = ((reg_read(chip, reg) & (1 << bit)) ? 1 : 0) ^ inv;
+ return 0;
+}
+
+static int snd_ak4117_rx_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_ak4117_rx_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = (chip->regmap[AK4117_REG_IO] & AK4117_IPS) ? 1 : 0;
+ return 0;
+}
+
+static int snd_ak4117_rx_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+ int change;
+ u8 old_val;
+
+ spin_lock_irq(&chip->lock);
+ old_val = chip->regmap[AK4117_REG_IO];
+ change = !!ucontrol->value.integer.value[0] != ((old_val & AK4117_IPS) ? 1 : 0);
+ if (change)
+ reg_write(chip, AK4117_REG_IO, (old_val & ~AK4117_IPS) | (ucontrol->value.integer.value[0] ? AK4117_IPS : 0));
+ spin_unlock_irq(&chip->lock);
+ return change;
+}
+
+static int snd_ak4117_rate_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 192000;
+ return 0;
+}
+
+static int snd_ak4117_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = external_rate(reg_read(chip, AK4117_REG_RCS1));
+ return 0;
+}
+
+static int snd_ak4117_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ak4117_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned i;
+
+ for (i = 0; i < AK4117_REG_RXCSB_SIZE; i++)
+ ucontrol->value.iec958.status[i] = reg_read(chip, AK4117_REG_RXCSB0 + i);
+ return 0;
+}
+
+static int snd_ak4117_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ak4117_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ memset(ucontrol->value.iec958.status, 0xff, AK4117_REG_RXCSB_SIZE);
+ return 0;
+}
+
+static int snd_ak4117_spdif_pinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xffff;
+ uinfo->count = 4;
+ return 0;
+}
+
+static int snd_ak4117_spdif_pget(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned short tmp;
+
+ ucontrol->value.integer.value[0] = 0xf8f2;
+ ucontrol->value.integer.value[1] = 0x4e1f;
+ tmp = reg_read(chip, AK4117_REG_Pc0) | (reg_read(chip, AK4117_REG_Pc1) << 8);
+ ucontrol->value.integer.value[2] = tmp;
+ tmp = reg_read(chip, AK4117_REG_Pd0) | (reg_read(chip, AK4117_REG_Pd1) << 8);
+ ucontrol->value.integer.value[3] = tmp;
+ return 0;
+}
+
+static int snd_ak4117_spdif_qinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = AK4117_REG_QSUB_SIZE;
+ return 0;
+}
+
+static int snd_ak4117_spdif_qget(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ak4117 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned i;
+
+ for (i = 0; i < AK4117_REG_QSUB_SIZE; i++)
+ ucontrol->value.bytes.data[i] = reg_read(chip, AK4117_REG_QSUB_ADDR + i);
+ return 0;
+}
+
+/* Don't forget to change AK4117_CONTROLS define!!! */
+static struct snd_kcontrol_new snd_ak4117_iec958_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Parity Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_in_error_info,
+ .get = snd_ak4117_in_error_get,
+ .private_value = offsetof(struct ak4117, parity_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 V-Bit Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_in_error_info,
+ .get = snd_ak4117_in_error_get,
+ .private_value = offsetof(struct ak4117, v_bit_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 C-CRC Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_in_error_info,
+ .get = snd_ak4117_in_error_get,
+ .private_value = offsetof(struct ak4117, ccrc_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Q-CRC Errors",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_in_error_info,
+ .get = snd_ak4117_in_error_get,
+ .private_value = offsetof(struct ak4117, qcrc_errors),
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 External Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_rate_info,
+ .get = snd_ak4117_rate_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,MASK),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_ak4117_spdif_mask_info,
+ .get = snd_ak4117_spdif_mask_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_spdif_info,
+ .get = snd_ak4117_spdif_get,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Preample Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_spdif_pinfo,
+ .get = snd_ak4117_spdif_pget,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Q-subcode Capture Default",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_spdif_qinfo,
+ .get = snd_ak4117_spdif_qget,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Audio",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_in_bit_info,
+ .get = snd_ak4117_in_bit_get,
+ .private_value = (1<<31) | (3<<8) | AK4117_REG_RCS0,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 Non-PCM Bitstream",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_in_bit_info,
+ .get = snd_ak4117_in_bit_get,
+ .private_value = (5<<8) | AK4117_REG_RCS1,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "IEC958 DTS Bitstream",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ak4117_in_bit_info,
+ .get = snd_ak4117_in_bit_get,
+ .private_value = (6<<8) | AK4117_REG_RCS1,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "AK4117 Input Select",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE,
+ .info = snd_ak4117_rx_info,
+ .get = snd_ak4117_rx_get,
+ .put = snd_ak4117_rx_put,
+}
+};
+
+int snd_ak4117_build(struct ak4117 *ak4117, struct snd_pcm_substream *cap_substream)
+{
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!cap_substream))
+ return -EINVAL;
+ ak4117->substream = cap_substream;
+ for (idx = 0; idx < AK4117_CONTROLS; idx++) {
+ kctl = snd_ctl_new1(&snd_ak4117_iec958_controls[idx], ak4117);
+ if (kctl == NULL)
+ return -ENOMEM;
+ kctl->id.device = cap_substream->pcm->device;
+ kctl->id.subdevice = cap_substream->number;
+ err = snd_ctl_add(ak4117->card, kctl);
+ if (err < 0)
+ return err;
+ ak4117->kctls[idx] = kctl;
+ }
+ return 0;
+}
+
+int snd_ak4117_external_rate(struct ak4117 *ak4117)
+{
+ unsigned char rcs1;
+
+ rcs1 = reg_read(ak4117, AK4117_REG_RCS1);
+ return external_rate(rcs1);
+}
+
+int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags)
+{
+ struct snd_pcm_runtime *runtime = ak4117->substream ? ak4117->substream->runtime : NULL;
+ unsigned long _flags;
+ int res = 0;
+ unsigned char rcs0, rcs1, rcs2;
+ unsigned char c0, c1;
+
+ rcs1 = reg_read(ak4117, AK4117_REG_RCS1);
+ if (flags & AK4117_CHECK_NO_STAT)
+ goto __rate;
+ rcs0 = reg_read(ak4117, AK4117_REG_RCS0);
+ rcs2 = reg_read(ak4117, AK4117_REG_RCS2);
+ // printk(KERN_DEBUG "AK IRQ: rcs0 = 0x%x, rcs1 = 0x%x, rcs2 = 0x%x\n", rcs0, rcs1, rcs2);
+ spin_lock_irqsave(&ak4117->lock, _flags);
+ if (rcs0 & AK4117_PAR)
+ ak4117->parity_errors++;
+ if (rcs0 & AK4117_V)
+ ak4117->v_bit_errors++;
+ if (rcs2 & AK4117_CCRC)
+ ak4117->ccrc_errors++;
+ if (rcs2 & AK4117_QCRC)
+ ak4117->qcrc_errors++;
+ c0 = (ak4117->rcs0 & (AK4117_QINT | AK4117_CINT | AK4117_STC | AK4117_AUDION | AK4117_AUTO | AK4117_UNLCK)) ^
+ (rcs0 & (AK4117_QINT | AK4117_CINT | AK4117_STC | AK4117_AUDION | AK4117_AUTO | AK4117_UNLCK));
+ c1 = (ak4117->rcs1 & (AK4117_DTSCD | AK4117_NPCM | AK4117_PEM | 0x0f)) ^
+ (rcs1 & (AK4117_DTSCD | AK4117_NPCM | AK4117_PEM | 0x0f));
+ ak4117->rcs0 = rcs0 & ~(AK4117_QINT | AK4117_CINT | AK4117_STC);
+ ak4117->rcs1 = rcs1;
+ ak4117->rcs2 = rcs2;
+ spin_unlock_irqrestore(&ak4117->lock, _flags);
+
+ if (rcs0 & AK4117_PAR)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[0]->id);
+ if (rcs0 & AK4117_V)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[1]->id);
+ if (rcs2 & AK4117_CCRC)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[2]->id);
+ if (rcs2 & AK4117_QCRC)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[3]->id);
+
+ /* rate change */
+ if (c1 & 0x0f)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[4]->id);
+
+ if ((c1 & AK4117_PEM) | (c0 & AK4117_CINT))
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[6]->id);
+ if (c0 & AK4117_QINT)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[8]->id);
+
+ if (c0 & AK4117_AUDION)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[9]->id);
+ if (c1 & AK4117_NPCM)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[10]->id);
+ if (c1 & AK4117_DTSCD)
+ snd_ctl_notify(ak4117->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4117->kctls[11]->id);
+
+ if (ak4117->change_callback && (c0 | c1) != 0)
+ ak4117->change_callback(ak4117, c0, c1);
+
+ __rate:
+ /* compare rate */
+ res = external_rate(rcs1);
+ if (!(flags & AK4117_CHECK_NO_RATE) && runtime && runtime->rate != res) {
+ snd_pcm_stream_lock_irqsave(ak4117->substream, _flags);
+ if (snd_pcm_running(ak4117->substream)) {
+ // printk(KERN_DEBUG "rate changed (%i <- %i)\n", runtime->rate, res);
+ snd_pcm_stop(ak4117->substream, SNDRV_PCM_STATE_DRAINING);
+ wake_up(&runtime->sleep);
+ res = 1;
+ }
+ snd_pcm_stream_unlock_irqrestore(ak4117->substream, _flags);
+ }
+ return res;
+}
+
+static void snd_ak4117_timer(unsigned long data)
+{
+ struct ak4117 *chip = (struct ak4117 *)data;
+
+ if (chip->init)
+ return;
+ snd_ak4117_check_rate_and_errors(chip, 0);
+ chip->timer.expires = 1 + jiffies;
+ add_timer(&chip->timer);
+}
+
+EXPORT_SYMBOL(snd_ak4117_create);
+EXPORT_SYMBOL(snd_ak4117_reg_write);
+EXPORT_SYMBOL(snd_ak4117_reinit);
+EXPORT_SYMBOL(snd_ak4117_build);
+EXPORT_SYMBOL(snd_ak4117_external_rate);
+EXPORT_SYMBOL(snd_ak4117_check_rate_and_errors);
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
new file mode 100644
index 0000000..ee47aba
--- /dev/null
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -0,0 +1,872 @@
+/*
+ * ALSA driver for AK4524 / AK4528 / AK4529 / AK4355 / AK4358 / AK4381
+ * AD and DA converters
+ *
+ * Copyright (c) 2000-2004 Jaroslav Kysela <perex@perex.cz>,
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/ak4xxx-adda.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters");
+MODULE_LICENSE("GPL");
+
+/* write the given register and save the data to the cache */
+void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+ unsigned char val)
+{
+ ak->ops.lock(ak, chip);
+ ak->ops.write(ak, chip, reg, val);
+
+ /* save the data */
+ snd_akm4xxx_set(ak, chip, reg, val);
+ ak->ops.unlock(ak, chip);
+}
+
+EXPORT_SYMBOL(snd_akm4xxx_write);
+
+/* reset procedure for AK4524 and AK4528 */
+static void ak4524_reset(struct snd_akm4xxx *ak, int state)
+{
+ unsigned int chip;
+ unsigned char reg, maxreg;
+
+ if (ak->type == SND_AK4528)
+ maxreg = 0x06;
+ else
+ maxreg = 0x08;
+ for (chip = 0; chip < ak->num_dacs/2; chip++) {
+ snd_akm4xxx_write(ak, chip, 0x01, state ? 0x00 : 0x03);
+ if (state)
+ continue;
+ /* DAC volumes */
+ for (reg = 0x04; reg < maxreg; reg++)
+ snd_akm4xxx_write(ak, chip, reg,
+ snd_akm4xxx_get(ak, chip, reg));
+ }
+}
+
+/* reset procedure for AK4355 and AK4358 */
+static void ak435X_reset(struct snd_akm4xxx *ak, int state,
+ unsigned char total_regs)
+{
+ unsigned char reg;
+
+ if (state) {
+ snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */
+ return;
+ }
+ for (reg = 0x00; reg < total_regs; reg++)
+ if (reg != 0x01)
+ snd_akm4xxx_write(ak, 0, reg,
+ snd_akm4xxx_get(ak, 0, reg));
+ snd_akm4xxx_write(ak, 0, 0x01, 0x01); /* un-reset, unmute */
+}
+
+/* reset procedure for AK4381 */
+static void ak4381_reset(struct snd_akm4xxx *ak, int state)
+{
+ unsigned int chip;
+ unsigned char reg;
+
+ for (chip = 0; chip < ak->num_dacs/2; chip++) {
+ snd_akm4xxx_write(ak, chip, 0x00, state ? 0x0c : 0x0f);
+ if (state)
+ continue;
+ for (reg = 0x01; reg < 0x05; reg++)
+ snd_akm4xxx_write(ak, chip, reg,
+ snd_akm4xxx_get(ak, chip, reg));
+ }
+}
+
+/*
+ * reset the AKM codecs
+ * @state: 1 = reset codec, 0 = restore the registers
+ *
+ * assert the reset operation and restores the register values to the chips.
+ */
+void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state)
+{
+ switch (ak->type) {
+ case SND_AK4524:
+ case SND_AK4528:
+ ak4524_reset(ak, state);
+ break;
+ case SND_AK4529:
+ /* FIXME: needed for ak4529? */
+ break;
+ case SND_AK4355:
+ ak435X_reset(ak, state, 0x0b);
+ break;
+ case SND_AK4358:
+ ak435X_reset(ak, state, 0x10);
+ break;
+ case SND_AK4381:
+ ak4381_reset(ak, state);
+ break;
+ default:
+ break;
+ }
+}
+
+EXPORT_SYMBOL(snd_akm4xxx_reset);
+
+
+/*
+ * Volume conversion table for non-linear volumes
+ * from -63.5dB (mute) to 0dB step 0.5dB
+ *
+ * Used for AK4524 input/ouput attenuation, AK4528, and
+ * AK5365 input attenuation
+ */
+static const unsigned char vol_cvt_datt[128] = {
+ 0x00, 0x01, 0x01, 0x02, 0x02, 0x03, 0x03, 0x04,
+ 0x04, 0x04, 0x04, 0x05, 0x05, 0x05, 0x06, 0x06,
+ 0x06, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x0a,
+ 0x0a, 0x0b, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x0f,
+ 0x10, 0x10, 0x11, 0x12, 0x12, 0x13, 0x13, 0x14,
+ 0x15, 0x16, 0x17, 0x17, 0x18, 0x19, 0x1a, 0x1c,
+ 0x1d, 0x1e, 0x1f, 0x20, 0x21, 0x22, 0x23, 0x23,
+ 0x24, 0x25, 0x26, 0x28, 0x29, 0x2a, 0x2b, 0x2d,
+ 0x2e, 0x30, 0x30, 0x31, 0x32, 0x33, 0x34, 0x35,
+ 0x37, 0x38, 0x39, 0x3b, 0x3c, 0x3e, 0x3f, 0x40,
+ 0x41, 0x42, 0x43, 0x44, 0x46, 0x47, 0x48, 0x4a,
+ 0x4b, 0x4d, 0x4e, 0x50, 0x51, 0x52, 0x53, 0x54,
+ 0x55, 0x56, 0x58, 0x59, 0x5b, 0x5c, 0x5e, 0x5f,
+ 0x60, 0x61, 0x62, 0x64, 0x65, 0x66, 0x67, 0x69,
+ 0x6a, 0x6c, 0x6d, 0x6f, 0x70, 0x71, 0x72, 0x73,
+ 0x75, 0x76, 0x77, 0x79, 0x7a, 0x7c, 0x7d, 0x7f,
+};
+
+/*
+ * dB tables
+ */
+static const DECLARE_TLV_DB_SCALE(db_scale_vol_datt, -6350, 50, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_8bit, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_7bit, -6350, 50, 1);
+static const DECLARE_TLV_DB_LINEAR(db_scale_linear, TLV_DB_GAIN_MUTE, 0);
+
+/*
+ * initialize all the ak4xxx chips
+ */
+void snd_akm4xxx_init(struct snd_akm4xxx *ak)
+{
+ static const unsigned char inits_ak4524[] = {
+ 0x00, 0x07, /* 0: all power up */
+ 0x01, 0x00, /* 1: ADC/DAC reset */
+ 0x02, 0x60, /* 2: 24bit I2S */
+ 0x03, 0x19, /* 3: deemphasis off */
+ 0x01, 0x03, /* 1: ADC/DAC enable */
+ 0x04, 0x00, /* 4: ADC left muted */
+ 0x05, 0x00, /* 5: ADC right muted */
+ 0x06, 0x00, /* 6: DAC left muted */
+ 0x07, 0x00, /* 7: DAC right muted */
+ 0xff, 0xff
+ };
+ static const unsigned char inits_ak4528[] = {
+ 0x00, 0x07, /* 0: all power up */
+ 0x01, 0x00, /* 1: ADC/DAC reset */
+ 0x02, 0x60, /* 2: 24bit I2S */
+ 0x03, 0x0d, /* 3: deemphasis off, turn LR highpass filters on */
+ 0x01, 0x03, /* 1: ADC/DAC enable */
+ 0x04, 0x00, /* 4: ADC left muted */
+ 0x05, 0x00, /* 5: ADC right muted */
+ 0xff, 0xff
+ };
+ static const unsigned char inits_ak4529[] = {
+ 0x09, 0x01, /* 9: ATS=0, RSTN=1 */
+ 0x0a, 0x3f, /* A: all power up, no zero/overflow detection */
+ 0x00, 0x0c, /* 0: TDM=0, 24bit I2S, SMUTE=0 */
+ 0x01, 0x00, /* 1: ACKS=0, ADC, loop off */
+ 0x02, 0xff, /* 2: LOUT1 muted */
+ 0x03, 0xff, /* 3: ROUT1 muted */
+ 0x04, 0xff, /* 4: LOUT2 muted */
+ 0x05, 0xff, /* 5: ROUT2 muted */
+ 0x06, 0xff, /* 6: LOUT3 muted */
+ 0x07, 0xff, /* 7: ROUT3 muted */
+ 0x0b, 0xff, /* B: LOUT4 muted */
+ 0x0c, 0xff, /* C: ROUT4 muted */
+ 0x08, 0x55, /* 8: deemphasis all off */
+ 0xff, 0xff
+ };
+ static const unsigned char inits_ak4355[] = {
+ 0x01, 0x02, /* 1: reset and soft-mute */
+ 0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect,
+ * disable DZF, sharp roll-off, RSTN#=0 */
+ 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */
+ // 0x02, 0x2e, /* quad speed */
+ 0x03, 0x01, /* 3: de-emphasis off */
+ 0x04, 0x00, /* 4: LOUT1 volume muted */
+ 0x05, 0x00, /* 5: ROUT1 volume muted */
+ 0x06, 0x00, /* 6: LOUT2 volume muted */
+ 0x07, 0x00, /* 7: ROUT2 volume muted */
+ 0x08, 0x00, /* 8: LOUT3 volume muted */
+ 0x09, 0x00, /* 9: ROUT3 volume muted */
+ 0x0a, 0x00, /* a: DATT speed=0, ignore DZF */
+ 0x01, 0x01, /* 1: un-reset, unmute */
+ 0xff, 0xff
+ };
+ static const unsigned char inits_ak4358[] = {
+ 0x01, 0x02, /* 1: reset and soft-mute */
+ 0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect,
+ * disable DZF, sharp roll-off, RSTN#=0 */
+ 0x02, 0x4e, /* 2: DA's power up, normal speed, RSTN#=0 */
+ /* 0x02, 0x6e,*/ /* quad speed */
+ 0x03, 0x01, /* 3: de-emphasis off */
+ 0x04, 0x00, /* 4: LOUT1 volume muted */
+ 0x05, 0x00, /* 5: ROUT1 volume muted */
+ 0x06, 0x00, /* 6: LOUT2 volume muted */
+ 0x07, 0x00, /* 7: ROUT2 volume muted */
+ 0x08, 0x00, /* 8: LOUT3 volume muted */
+ 0x09, 0x00, /* 9: ROUT3 volume muted */
+ 0x0b, 0x00, /* b: LOUT4 volume muted */
+ 0x0c, 0x00, /* c: ROUT4 volume muted */
+ 0x0a, 0x00, /* a: DATT speed=0, ignore DZF */
+ 0x01, 0x01, /* 1: un-reset, unmute */
+ 0xff, 0xff
+ };
+ static const unsigned char inits_ak4381[] = {
+ 0x00, 0x0c, /* 0: mode3(i2s), disable auto-clock detect */
+ 0x01, 0x02, /* 1: de-emphasis off, normal speed,
+ * sharp roll-off, DZF off */
+ // 0x01, 0x12, /* quad speed */
+ 0x02, 0x00, /* 2: DZF disabled */
+ 0x03, 0x00, /* 3: LATT 0 */
+ 0x04, 0x00, /* 4: RATT 0 */
+ 0x00, 0x0f, /* 0: power-up, un-reset */
+ 0xff, 0xff
+ };
+
+ int chip, num_chips;
+ const unsigned char *ptr, *inits;
+ unsigned char reg, data;
+
+ memset(ak->images, 0, sizeof(ak->images));
+ memset(ak->volumes, 0, sizeof(ak->volumes));
+
+ switch (ak->type) {
+ case SND_AK4524:
+ inits = inits_ak4524;
+ num_chips = ak->num_dacs / 2;
+ break;
+ case SND_AK4528:
+ inits = inits_ak4528;
+ num_chips = ak->num_dacs / 2;
+ break;
+ case SND_AK4529:
+ inits = inits_ak4529;
+ num_chips = 1;
+ break;
+ case SND_AK4355:
+ inits = inits_ak4355;
+ num_chips = 1;
+ break;
+ case SND_AK4358:
+ inits = inits_ak4358;
+ num_chips = 1;
+ break;
+ case SND_AK4381:
+ inits = inits_ak4381;
+ num_chips = ak->num_dacs / 2;
+ break;
+ case SND_AK5365:
+ /* FIXME: any init sequence? */
+ return;
+ default:
+ snd_BUG();
+ return;
+ }
+
+ for (chip = 0; chip < num_chips; chip++) {
+ ptr = inits;
+ while (*ptr != 0xff) {
+ reg = *ptr++;
+ data = *ptr++;
+ snd_akm4xxx_write(ak, chip, reg, data);
+ }
+ }
+}
+
+EXPORT_SYMBOL(snd_akm4xxx_init);
+
+/*
+ * Mixer callbacks
+ */
+#define AK_IPGA (1<<20) /* including IPGA */
+#define AK_VOL_CVT (1<<21) /* need dB conversion */
+#define AK_NEEDSMSB (1<<22) /* need MSB update bit */
+#define AK_INVERT (1<<23) /* data is inverted */
+#define AK_GET_CHIP(val) (((val) >> 8) & 0xff)
+#define AK_GET_ADDR(val) ((val) & 0xff)
+#define AK_GET_SHIFT(val) (((val) >> 16) & 0x0f)
+#define AK_GET_VOL_CVT(val) (((val) >> 21) & 1)
+#define AK_GET_IPGA(val) (((val) >> 20) & 1)
+#define AK_GET_NEEDSMSB(val) (((val) >> 22) & 1)
+#define AK_GET_INVERT(val) (((val) >> 23) & 1)
+#define AK_GET_MASK(val) (((val) >> 24) & 0xff)
+#define AK_COMPOSE(chip,addr,shift,mask) \
+ (((chip) << 8) | (addr) | ((shift) << 16) | ((mask) << 24))
+
+static int snd_akm4xxx_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_akm4xxx_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+
+ ucontrol->value.integer.value[0] = snd_akm4xxx_get_vol(ak, chip, addr);
+ return 0;
+}
+
+static int put_ak_reg(struct snd_kcontrol *kcontrol, int addr,
+ unsigned char nval)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+
+ if (snd_akm4xxx_get_vol(ak, chip, addr) == nval)
+ return 0;
+
+ snd_akm4xxx_set_vol(ak, chip, addr, nval);
+ if (AK_GET_VOL_CVT(kcontrol->private_value) && nval < 128)
+ nval = vol_cvt_datt[nval];
+ if (AK_GET_IPGA(kcontrol->private_value) && nval >= 128)
+ nval++; /* need to correct + 1 since both 127 and 128 are 0dB */
+ if (AK_GET_INVERT(kcontrol->private_value))
+ nval = mask - nval;
+ if (AK_GET_NEEDSMSB(kcontrol->private_value))
+ nval |= 0x80;
+ /* printk(KERN_DEBUG "DEBUG - AK writing reg: chip %x addr %x,
+ nval %x\n", chip, addr, nval); */
+ snd_akm4xxx_write(ak, chip, addr, nval);
+ return 1;
+}
+
+static int snd_akm4xxx_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+ unsigned int val = ucontrol->value.integer.value[0];
+ if (val > mask)
+ return -EINVAL;
+ return put_ak_reg(kcontrol, AK_GET_ADDR(kcontrol->private_value), val);
+}
+
+static int snd_akm4xxx_stereo_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_akm4xxx_stereo_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+
+ ucontrol->value.integer.value[0] = snd_akm4xxx_get_vol(ak, chip, addr);
+ ucontrol->value.integer.value[1] = snd_akm4xxx_get_vol(ak, chip, addr+1);
+ return 0;
+}
+
+static int snd_akm4xxx_stereo_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+ unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+ unsigned int val[2];
+ int change;
+
+ val[0] = ucontrol->value.integer.value[0];
+ val[1] = ucontrol->value.integer.value[1];
+ if (val[0] > mask || val[1] > mask)
+ return -EINVAL;
+ change = put_ak_reg(kcontrol, addr, val[0]);
+ change |= put_ak_reg(kcontrol, addr + 1, val[1]);
+ return change;
+}
+
+static int snd_akm4xxx_deemphasis_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {
+ "44.1kHz", "Off", "48kHz", "32kHz",
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= 4)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_akm4xxx_deemphasis_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+ int shift = AK_GET_SHIFT(kcontrol->private_value);
+ ucontrol->value.enumerated.item[0] =
+ (snd_akm4xxx_get(ak, chip, addr) >> shift) & 3;
+ return 0;
+}
+
+static int snd_akm4xxx_deemphasis_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+ int shift = AK_GET_SHIFT(kcontrol->private_value);
+ unsigned char nval = ucontrol->value.enumerated.item[0] & 3;
+ int change;
+
+ nval = (nval << shift) |
+ (snd_akm4xxx_get(ak, chip, addr) & ~(3 << shift));
+ change = snd_akm4xxx_get(ak, chip, addr) != nval;
+ if (change)
+ snd_akm4xxx_write(ak, chip, addr, nval);
+ return change;
+}
+
+#define ak4xxx_switch_info snd_ctl_boolean_mono_info
+
+static int ak4xxx_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+ int shift = AK_GET_SHIFT(kcontrol->private_value);
+ int invert = AK_GET_INVERT(kcontrol->private_value);
+ /* we observe the (1<<shift) bit only */
+ unsigned char val = snd_akm4xxx_get(ak, chip, addr) & (1<<shift);
+ if (invert)
+ val = ! val;
+ ucontrol->value.integer.value[0] = (val & (1<<shift)) != 0;
+ return 0;
+}
+
+static int ak4xxx_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+ int shift = AK_GET_SHIFT(kcontrol->private_value);
+ int invert = AK_GET_INVERT(kcontrol->private_value);
+ long flag = ucontrol->value.integer.value[0];
+ unsigned char val, oval;
+ int change;
+
+ if (invert)
+ flag = ! flag;
+ oval = snd_akm4xxx_get(ak, chip, addr);
+ if (flag)
+ val = oval | (1<<shift);
+ else
+ val = oval & ~(1<<shift);
+ change = (oval != val);
+ if (change)
+ snd_akm4xxx_write(ak, chip, addr, val);
+ return change;
+}
+
+#define AK5365_NUM_INPUTS 5
+
+static int ak4xxx_capture_num_inputs(struct snd_akm4xxx *ak, int mixer_ch)
+{
+ int num_names;
+ const char **input_names;
+
+ input_names = ak->adc_info[mixer_ch].input_names;
+ num_names = 0;
+ while (num_names < AK5365_NUM_INPUTS && input_names[num_names])
+ ++num_names;
+ return num_names;
+}
+
+static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int mixer_ch = AK_GET_SHIFT(kcontrol->private_value);
+ const char **input_names;
+ int num_names, idx;
+
+ num_names = ak4xxx_capture_num_inputs(ak, mixer_ch);
+ if (!num_names)
+ return -EINVAL;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = num_names;
+ idx = uinfo->value.enumerated.item;
+ if (idx >= num_names)
+ return -EINVAL;
+ input_names = ak->adc_info[mixer_ch].input_names;
+ strncpy(uinfo->value.enumerated.name, input_names[idx],
+ sizeof(uinfo->value.enumerated.name));
+ return 0;
+}
+
+static int ak4xxx_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+ int mask = AK_GET_MASK(kcontrol->private_value);
+ unsigned char val;
+
+ val = snd_akm4xxx_get(ak, chip, addr) & mask;
+ ucontrol->value.enumerated.item[0] = val;
+ return 0;
+}
+
+static int ak4xxx_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+ int mixer_ch = AK_GET_SHIFT(kcontrol->private_value);
+ int chip = AK_GET_CHIP(kcontrol->private_value);
+ int addr = AK_GET_ADDR(kcontrol->private_value);
+ int mask = AK_GET_MASK(kcontrol->private_value);
+ unsigned char oval, val;
+ int num_names = ak4xxx_capture_num_inputs(ak, mixer_ch);
+
+ if (ucontrol->value.enumerated.item[0] >= num_names)
+ return -EINVAL;
+
+ oval = snd_akm4xxx_get(ak, chip, addr);
+ val = oval & ~mask;
+ val |= ucontrol->value.enumerated.item[0] & mask;
+ if (val != oval) {
+ snd_akm4xxx_write(ak, chip, addr, val);
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * build AK4xxx controls
+ */
+
+static int build_dac_controls(struct snd_akm4xxx *ak)
+{
+ int idx, err, mixer_ch, num_stereo;
+ struct snd_kcontrol_new knew;
+
+ mixer_ch = 0;
+ for (idx = 0; idx < ak->num_dacs; ) {
+ /* mute control for Revolution 7.1 - AK4381 */
+ if (ak->type == SND_AK4381
+ && ak->dac_info[mixer_ch].switch_name) {
+ memset(&knew, 0, sizeof(knew));
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.count = 1;
+ knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ knew.name = ak->dac_info[mixer_ch].switch_name;
+ knew.info = ak4xxx_switch_info;
+ knew.get = ak4xxx_switch_get;
+ knew.put = ak4xxx_switch_put;
+ knew.access = 0;
+ /* register 1, bit 0 (SMUTE): 0 = normal operation,
+ 1 = mute */
+ knew.private_value =
+ AK_COMPOSE(idx/2, 1, 0, 0) | AK_INVERT;
+ err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak));
+ if (err < 0)
+ return err;
+ }
+ memset(&knew, 0, sizeof(knew));
+ if (! ak->dac_info || ! ak->dac_info[mixer_ch].name) {
+ knew.name = "DAC Volume";
+ knew.index = mixer_ch + ak->idx_offset * 2;
+ num_stereo = 1;
+ } else {
+ knew.name = ak->dac_info[mixer_ch].name;
+ num_stereo = ak->dac_info[mixer_ch].num_channels;
+ }
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.count = 1;
+ knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ if (num_stereo == 2) {
+ knew.info = snd_akm4xxx_stereo_volume_info;
+ knew.get = snd_akm4xxx_stereo_volume_get;
+ knew.put = snd_akm4xxx_stereo_volume_put;
+ } else {
+ knew.info = snd_akm4xxx_volume_info;
+ knew.get = snd_akm4xxx_volume_get;
+ knew.put = snd_akm4xxx_volume_put;
+ }
+ switch (ak->type) {
+ case SND_AK4524:
+ /* register 6 & 7 */
+ knew.private_value =
+ AK_COMPOSE(idx/2, (idx%2) + 6, 0, 127) |
+ AK_VOL_CVT;
+ knew.tlv.p = db_scale_vol_datt;
+ break;
+ case SND_AK4528:
+ /* register 4 & 5 */
+ knew.private_value =
+ AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127) |
+ AK_VOL_CVT;
+ knew.tlv.p = db_scale_vol_datt;
+ break;
+ case SND_AK4529: {
+ /* registers 2-7 and b,c */
+ int val = idx < 6 ? idx + 2 : (idx - 6) + 0xb;
+ knew.private_value =
+ AK_COMPOSE(0, val, 0, 255) | AK_INVERT;
+ knew.tlv.p = db_scale_8bit;
+ break;
+ }
+ case SND_AK4355:
+ /* register 4-9, chip #0 only */
+ knew.private_value = AK_COMPOSE(0, idx + 4, 0, 255);
+ knew.tlv.p = db_scale_8bit;
+ break;
+ case SND_AK4358: {
+ /* register 4-9 and 11-12, chip #0 only */
+ int addr = idx < 6 ? idx + 4 : idx + 5;
+ knew.private_value =
+ AK_COMPOSE(0, addr, 0, 127) | AK_NEEDSMSB;
+ knew.tlv.p = db_scale_7bit;
+ break;
+ }
+ case SND_AK4381:
+ /* register 3 & 4 */
+ knew.private_value =
+ AK_COMPOSE(idx/2, (idx%2) + 3, 0, 255);
+ knew.tlv.p = db_scale_linear;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak));
+ if (err < 0)
+ return err;
+
+ idx += num_stereo;
+ mixer_ch++;
+ }
+ return 0;
+}
+
+static int build_adc_controls(struct snd_akm4xxx *ak)
+{
+ int idx, err, mixer_ch, num_stereo;
+ struct snd_kcontrol_new knew;
+
+ mixer_ch = 0;
+ for (idx = 0; idx < ak->num_adcs;) {
+ memset(&knew, 0, sizeof(knew));
+ if (! ak->adc_info || ! ak->adc_info[mixer_ch].name) {
+ knew.name = "ADC Volume";
+ knew.index = mixer_ch + ak->idx_offset * 2;
+ num_stereo = 1;
+ } else {
+ knew.name = ak->adc_info[mixer_ch].name;
+ num_stereo = ak->adc_info[mixer_ch].num_channels;
+ }
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.count = 1;
+ knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ if (num_stereo == 2) {
+ knew.info = snd_akm4xxx_stereo_volume_info;
+ knew.get = snd_akm4xxx_stereo_volume_get;
+ knew.put = snd_akm4xxx_stereo_volume_put;
+ } else {
+ knew.info = snd_akm4xxx_volume_info;
+ knew.get = snd_akm4xxx_volume_get;
+ knew.put = snd_akm4xxx_volume_put;
+ }
+ /* register 4 & 5 */
+ if (ak->type == SND_AK5365)
+ knew.private_value =
+ AK_COMPOSE(idx/2, (idx%2) + 4, 0, 151) |
+ AK_VOL_CVT | AK_IPGA;
+ else
+ knew.private_value =
+ AK_COMPOSE(idx/2, (idx%2) + 4, 0, 163) |
+ AK_VOL_CVT | AK_IPGA;
+ knew.tlv.p = db_scale_vol_datt;
+ err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak));
+ if (err < 0)
+ return err;
+
+ if (ak->type == SND_AK5365 && (idx % 2) == 0) {
+ if (! ak->adc_info ||
+ ! ak->adc_info[mixer_ch].switch_name) {
+ knew.name = "Capture Switch";
+ knew.index = mixer_ch + ak->idx_offset * 2;
+ } else
+ knew.name = ak->adc_info[mixer_ch].switch_name;
+ knew.info = ak4xxx_switch_info;
+ knew.get = ak4xxx_switch_get;
+ knew.put = ak4xxx_switch_put;
+ knew.access = 0;
+ /* register 2, bit 0 (SMUTE): 0 = normal operation,
+ 1 = mute */
+ knew.private_value =
+ AK_COMPOSE(idx/2, 2, 0, 0) | AK_INVERT;
+ err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak));
+ if (err < 0)
+ return err;
+
+ memset(&knew, 0, sizeof(knew));
+ knew.name = ak->adc_info[mixer_ch].selector_name;
+ if (!knew.name) {
+ knew.name = "Capture Channel";
+ knew.index = mixer_ch + ak->idx_offset * 2;
+ }
+
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.info = ak4xxx_capture_source_info;
+ knew.get = ak4xxx_capture_source_get;
+ knew.put = ak4xxx_capture_source_put;
+ knew.access = 0;
+ /* input selector control: reg. 1, bits 0-2.
+ * mis-use 'shift' to pass mixer_ch */
+ knew.private_value
+ = AK_COMPOSE(idx/2, 1, mixer_ch, 0x07);
+ err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak));
+ if (err < 0)
+ return err;
+ }
+
+ idx += num_stereo;
+ mixer_ch++;
+ }
+ return 0;
+}
+
+static int build_deemphasis(struct snd_akm4xxx *ak, int num_emphs)
+{
+ int idx, err;
+ struct snd_kcontrol_new knew;
+
+ for (idx = 0; idx < num_emphs; idx++) {
+ memset(&knew, 0, sizeof(knew));
+ knew.name = "Deemphasis";
+ knew.index = idx + ak->idx_offset;
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.count = 1;
+ knew.info = snd_akm4xxx_deemphasis_info;
+ knew.get = snd_akm4xxx_deemphasis_get;
+ knew.put = snd_akm4xxx_deemphasis_put;
+ switch (ak->type) {
+ case SND_AK4524:
+ case SND_AK4528:
+ /* register 3 */
+ knew.private_value = AK_COMPOSE(idx, 3, 0, 0);
+ break;
+ case SND_AK4529: {
+ int shift = idx == 3 ? 6 : (2 - idx) * 2;
+ /* register 8 with shift */
+ knew.private_value = AK_COMPOSE(0, 8, shift, 0);
+ break;
+ }
+ case SND_AK4355:
+ case SND_AK4358:
+ knew.private_value = AK_COMPOSE(idx, 3, 0, 0);
+ break;
+ case SND_AK4381:
+ knew.private_value = AK_COMPOSE(idx, 1, 1, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+ err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
+{
+ int err, num_emphs;
+
+ err = build_dac_controls(ak);
+ if (err < 0)
+ return err;
+
+ err = build_adc_controls(ak);
+ if (err < 0)
+ return err;
+
+ if (ak->type == SND_AK4355 || ak->type == SND_AK4358)
+ num_emphs = 1;
+ else
+ num_emphs = ak->num_dacs / 2;
+ err = build_deemphasis(ak, num_emphs);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_akm4xxx_build_controls);
+
+static int __init alsa_akm4xxx_module_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_akm4xxx_module_exit(void)
+{
+}
+
+module_init(alsa_akm4xxx_module_init)
+module_exit(alsa_akm4xxx_module_exit)
diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c
new file mode 100644
index 0000000..797d3a6
--- /dev/null
+++ b/sound/i2c/other/pt2258.c
@@ -0,0 +1,226 @@
+/*
+ * ALSA Driver for the PT2258 volume controller.
+ *
+ * Copyright (c) 2006 Jochen Voss <voss@seehuhn.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/i2c.h>
+#include <sound/pt2258.h>
+
+MODULE_AUTHOR("Jochen Voss <voss@seehuhn.de>");
+MODULE_DESCRIPTION("PT2258 volume controller (Princeton Technology Corp.)");
+MODULE_LICENSE("GPL");
+
+#define PT2258_CMD_RESET 0xc0
+#define PT2258_CMD_UNMUTE 0xf8
+#define PT2258_CMD_MUTE 0xf9
+
+static const unsigned char pt2258_channel_code[12] = {
+ 0x80, 0x90, /* channel 1: -10dB, -1dB */
+ 0x40, 0x50, /* channel 2: -10dB, -1dB */
+ 0x00, 0x10, /* channel 3: -10dB, -1dB */
+ 0x20, 0x30, /* channel 4: -10dB, -1dB */
+ 0x60, 0x70, /* channel 5: -10dB, -1dB */
+ 0xa0, 0xb0 /* channel 6: -10dB, -1dB */
+};
+
+int snd_pt2258_reset(struct snd_pt2258 *pt)
+{
+ unsigned char bytes[2];
+ int i;
+
+ /* reset chip */
+ bytes[0] = PT2258_CMD_RESET;
+ snd_i2c_lock(pt->i2c_bus);
+ if (snd_i2c_sendbytes(pt->i2c_dev, bytes, 1) != 1)
+ goto __error;
+ snd_i2c_unlock(pt->i2c_bus);
+
+ /* mute all channels */
+ pt->mute = 1;
+ bytes[0] = PT2258_CMD_MUTE;
+ snd_i2c_lock(pt->i2c_bus);
+ if (snd_i2c_sendbytes(pt->i2c_dev, bytes, 1) != 1)
+ goto __error;
+ snd_i2c_unlock(pt->i2c_bus);
+
+ /* set all channels to 0dB */
+ for (i = 0; i < 6; ++i)
+ pt->volume[i] = 0;
+ bytes[0] = 0xd0;
+ bytes[1] = 0xe0;
+ snd_i2c_lock(pt->i2c_bus);
+ if (snd_i2c_sendbytes(pt->i2c_dev, bytes, 2) != 2)
+ goto __error;
+ snd_i2c_unlock(pt->i2c_bus);
+
+ return 0;
+
+ __error:
+ snd_i2c_unlock(pt->i2c_bus);
+ snd_printk(KERN_ERR "PT2258 reset failed\n");
+ return -EIO;
+}
+
+static int pt2258_stereo_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 79;
+ return 0;
+}
+
+static int pt2258_stereo_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pt2258 *pt = kcontrol->private_data;
+ int base = kcontrol->private_value;
+
+ /* chip does not support register reads */
+ ucontrol->value.integer.value[0] = 79 - pt->volume[base];
+ ucontrol->value.integer.value[1] = 79 - pt->volume[base + 1];
+ return 0;
+}
+
+static int pt2258_stereo_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pt2258 *pt = kcontrol->private_data;
+ int base = kcontrol->private_value;
+ unsigned char bytes[2];
+ int val0, val1;
+
+ val0 = 79 - ucontrol->value.integer.value[0];
+ val1 = 79 - ucontrol->value.integer.value[1];
+ if (val0 < 0 || val0 > 79 || val1 < 0 || val1 > 79)
+ return -EINVAL;
+ if (val0 == pt->volume[base] && val1 == pt->volume[base + 1])
+ return 0;
+
+ pt->volume[base] = val0;
+ bytes[0] = pt2258_channel_code[2 * base] | (val0 / 10);
+ bytes[1] = pt2258_channel_code[2 * base + 1] | (val0 % 10);
+ snd_i2c_lock(pt->i2c_bus);
+ if (snd_i2c_sendbytes(pt->i2c_dev, bytes, 2) != 2)
+ goto __error;
+ snd_i2c_unlock(pt->i2c_bus);
+
+ pt->volume[base + 1] = val1;
+ bytes[0] = pt2258_channel_code[2 * base + 2] | (val1 / 10);
+ bytes[1] = pt2258_channel_code[2 * base + 3] | (val1 % 10);
+ snd_i2c_lock(pt->i2c_bus);
+ if (snd_i2c_sendbytes(pt->i2c_dev, bytes, 2) != 2)
+ goto __error;
+ snd_i2c_unlock(pt->i2c_bus);
+
+ return 1;
+
+ __error:
+ snd_i2c_unlock(pt->i2c_bus);
+ snd_printk(KERN_ERR "PT2258 access failed\n");
+ return -EIO;
+}
+
+#define pt2258_switch_info snd_ctl_boolean_mono_info
+
+static int pt2258_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pt2258 *pt = kcontrol->private_data;
+
+ ucontrol->value.integer.value[0] = !pt->mute;
+ return 0;
+}
+
+static int pt2258_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pt2258 *pt = kcontrol->private_data;
+ unsigned char bytes[2];
+ int val;
+
+ val = !ucontrol->value.integer.value[0];
+ if (pt->mute == val)
+ return 0;
+
+ pt->mute = val;
+ bytes[0] = val ? PT2258_CMD_MUTE : PT2258_CMD_UNMUTE;
+ snd_i2c_lock(pt->i2c_bus);
+ if (snd_i2c_sendbytes(pt->i2c_dev, bytes, 1) != 1)
+ goto __error;
+ snd_i2c_unlock(pt->i2c_bus);
+
+ return 1;
+
+ __error:
+ snd_i2c_unlock(pt->i2c_bus);
+ snd_printk(KERN_ERR "PT2258 access failed 2\n");
+ return -EIO;
+}
+
+static const DECLARE_TLV_DB_SCALE(pt2258_db_scale, -7900, 100, 0);
+
+int snd_pt2258_build_controls(struct snd_pt2258 *pt)
+{
+ struct snd_kcontrol_new knew;
+ char *names[3] = {
+ "Mic Loopback Playback Volume",
+ "Line Loopback Playback Volume",
+ "CD Loopback Playback Volume"
+ };
+ int i, err;
+
+ for (i = 0; i < 3; ++i) {
+ memset(&knew, 0, sizeof(knew));
+ knew.name = names[i];
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.count = 1;
+ knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ knew.private_value = 2 * i;
+ knew.info = pt2258_stereo_volume_info;
+ knew.get = pt2258_stereo_volume_get;
+ knew.put = pt2258_stereo_volume_put;
+ knew.tlv.p = pt2258_db_scale;
+
+ err = snd_ctl_add(pt->card, snd_ctl_new1(&knew, pt));
+ if (err < 0)
+ return err;
+ }
+
+ memset(&knew, 0, sizeof(knew));
+ knew.name = "Loopback Switch";
+ knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ knew.info = pt2258_switch_info;
+ knew.get = pt2258_switch_get;
+ knew.put = pt2258_switch_put;
+ knew.access = 0;
+ err = snd_ctl_add(pt->card, snd_ctl_new1(&knew, pt));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_pt2258_reset);
+EXPORT_SYMBOL(snd_pt2258_build_controls);
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
new file mode 100644
index 0000000..c13a178
--- /dev/null
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -0,0 +1,252 @@
+/*
+ * ALSA driver for TEA5757/5759 Philips AM/FM radio tuner chips
+ *
+ * Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/tea575x-tuner.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
+MODULE_LICENSE("GPL");
+
+/*
+ * definitions
+ */
+
+#define TEA575X_BIT_SEARCH (1<<24) /* 1 = search action, 0 = tuned */
+#define TEA575X_BIT_UPDOWN (1<<23) /* 0 = search down, 1 = search up */
+#define TEA575X_BIT_MONO (1<<22) /* 0 = stereo, 1 = mono */
+#define TEA575X_BIT_BAND_MASK (3<<20)
+#define TEA575X_BIT_BAND_FM (0<<20)
+#define TEA575X_BIT_BAND_MW (1<<20)
+#define TEA575X_BIT_BAND_LW (1<<21)
+#define TEA575X_BIT_BAND_SW (1<<22)
+#define TEA575X_BIT_PORT_0 (1<<19) /* user bit */
+#define TEA575X_BIT_PORT_1 (1<<18) /* user bit */
+#define TEA575X_BIT_SEARCH_MASK (3<<16) /* search level */
+#define TEA575X_BIT_SEARCH_5_28 (0<<16) /* FM >5uV, AM >28uV */
+#define TEA575X_BIT_SEARCH_10_40 (1<<16) /* FM >10uV, AM > 40uV */
+#define TEA575X_BIT_SEARCH_30_63 (2<<16) /* FM >30uV, AM > 63uV */
+#define TEA575X_BIT_SEARCH_150_1000 (3<<16) /* FM > 150uV, AM > 1000uV */
+#define TEA575X_BIT_DUMMY (1<<15) /* buffer */
+#define TEA575X_BIT_FREQ_MASK 0x7fff
+
+/*
+ * lowlevel part
+ */
+
+static void snd_tea575x_set_freq(struct snd_tea575x *tea)
+{
+ unsigned long freq;
+
+ freq = tea->freq / 16; /* to kHz */
+ if (freq > 108000)
+ freq = 108000;
+ if (freq < 87000)
+ freq = 87000;
+ /* crystal fixup */
+ if (tea->tea5759)
+ freq -= tea->freq_fixup;
+ else
+ freq += tea->freq_fixup;
+ /* freq /= 12.5 */
+ freq *= 10;
+ freq /= 125;
+
+ tea->val &= ~TEA575X_BIT_FREQ_MASK;
+ tea->val |= freq & TEA575X_BIT_FREQ_MASK;
+ tea->ops->write(tea, tea->val);
+}
+
+/*
+ * Linux Video interface
+ */
+
+static int snd_tea575x_ioctl(struct inode *inode, struct file *file,
+ unsigned int cmd, unsigned long data)
+{
+ struct snd_tea575x *tea = video_drvdata(file);
+ void __user *arg = (void __user *)data;
+
+ switch(cmd) {
+ case VIDIOCGCAP:
+ {
+ struct video_capability v;
+ v.type = VID_TYPE_TUNER;
+ v.channels = 1;
+ v.audios = 1;
+ /* No we don't do pictures */
+ v.maxwidth = 0;
+ v.maxheight = 0;
+ v.minwidth = 0;
+ v.minheight = 0;
+ strcpy(v.name, tea->tea5759 ? "TEA5759" : "TEA5757");
+ if (copy_to_user(arg,&v,sizeof(v)))
+ return -EFAULT;
+ return 0;
+ }
+ case VIDIOCGTUNER:
+ {
+ struct video_tuner v;
+ if (copy_from_user(&v, arg,sizeof(v))!=0)
+ return -EFAULT;
+ if (v.tuner) /* Only 1 tuner */
+ return -EINVAL;
+ v.rangelow = (87*16000);
+ v.rangehigh = (108*16000);
+ v.flags = VIDEO_TUNER_LOW;
+ v.mode = VIDEO_MODE_AUTO;
+ strcpy(v.name, "FM");
+ v.signal = 0xFFFF;
+ if (copy_to_user(arg, &v, sizeof(v)))
+ return -EFAULT;
+ return 0;
+ }
+ case VIDIOCSTUNER:
+ {
+ struct video_tuner v;
+ if(copy_from_user(&v, arg, sizeof(v)))
+ return -EFAULT;
+ if(v.tuner!=0)
+ return -EINVAL;
+ /* Only 1 tuner so no setting needed ! */
+ return 0;
+ }
+ case VIDIOCGFREQ:
+ if(copy_to_user(arg, &tea->freq, sizeof(tea->freq)))
+ return -EFAULT;
+ return 0;
+ case VIDIOCSFREQ:
+ if(copy_from_user(&tea->freq, arg, sizeof(tea->freq)))
+ return -EFAULT;
+ snd_tea575x_set_freq(tea);
+ return 0;
+ case VIDIOCGAUDIO:
+ {
+ struct video_audio v;
+ memset(&v, 0, sizeof(v));
+ strcpy(v.name, "Radio");
+ if(copy_to_user(arg,&v, sizeof(v)))
+ return -EFAULT;
+ return 0;
+ }
+ case VIDIOCSAUDIO:
+ {
+ struct video_audio v;
+ if(copy_from_user(&v, arg, sizeof(v)))
+ return -EFAULT;
+ if (tea->ops->mute)
+ tea->ops->mute(tea,
+ (v.flags &
+ VIDEO_AUDIO_MUTE) ? 1 : 0);
+ if(v.audio)
+ return -EINVAL;
+ return 0;
+ }
+ default:
+ return -ENOIOCTLCMD;
+ }
+}
+
+static void snd_tea575x_release(struct video_device *vfd)
+{
+}
+
+static int snd_tea575x_exclusive_open(struct inode *inode, struct file *file)
+{
+ struct snd_tea575x *tea = video_drvdata(file);
+
+ return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0;
+}
+
+static int snd_tea575x_exclusive_release(struct inode *inode, struct file *file)
+{
+ struct snd_tea575x *tea = video_drvdata(file);
+
+ clear_bit(0, &tea->in_use);
+ return 0;
+}
+
+/*
+ * initialize all the tea575x chips
+ */
+void snd_tea575x_init(struct snd_tea575x *tea)
+{
+ unsigned int val;
+
+ val = tea->ops->read(tea);
+ if (val == 0x1ffffff || val == 0) {
+ snd_printk(KERN_ERR "Cannot find TEA575x chip\n");
+ return;
+ }
+
+ memset(&tea->vd, 0, sizeof(tea->vd));
+ strcpy(tea->vd.name, tea->tea5759 ? "TEA5759 radio" : "TEA5757 radio");
+ tea->vd.release = snd_tea575x_release;
+ video_set_drvdata(&tea->vd, tea);
+ tea->vd.fops = &tea->fops;
+ tea->in_use = 0;
+ tea->fops.owner = tea->card->module;
+ tea->fops.open = snd_tea575x_exclusive_open;
+ tea->fops.release = snd_tea575x_exclusive_release;
+ tea->fops.ioctl = snd_tea575x_ioctl;
+ if (video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->dev_nr - 1) < 0) {
+ snd_printk(KERN_ERR "unable to register tea575x tuner\n");
+ return;
+ }
+ tea->vd_registered = 1;
+
+ tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40;
+ tea->freq = 90500 * 16; /* 90.5Mhz default */
+
+ snd_tea575x_set_freq(tea);
+
+ /* mute on init */
+ if (tea->ops->mute)
+ tea->ops->mute(tea, 1);
+}
+
+void snd_tea575x_exit(struct snd_tea575x *tea)
+{
+ if (tea->vd_registered) {
+ video_unregister_device(&tea->vd);
+ tea->vd_registered = 0;
+ }
+}
+
+static int __init alsa_tea575x_module_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_tea575x_module_exit(void)
+{
+}
+
+module_init(alsa_tea575x_module_init)
+module_exit(alsa_tea575x_module_exit)
+
+EXPORT_SYMBOL(snd_tea575x_init);
+EXPORT_SYMBOL(snd_tea575x_exit);
diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c
new file mode 100644
index 0000000..0e3a9f2
--- /dev/null
+++ b/sound/i2c/tea6330t.c
@@ -0,0 +1,385 @@
+/*
+ * Routines for control of the TEA6330T circuit via i2c bus
+ * Sound fader control circuit for car radios by Philips Semiconductors
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tea6330t.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Routines for control of the TEA6330T circuit via i2c bus");
+MODULE_LICENSE("GPL");
+
+#define TEA6330T_ADDR (0x80>>1) /* fixed address */
+
+#define TEA6330T_SADDR_VOLUME_LEFT 0x00 /* volume left */
+#define TEA6330T_SADDR_VOLUME_RIGHT 0x01 /* volume right */
+#define TEA6330T_SADDR_BASS 0x02 /* bass control */
+#define TEA6330T_SADDR_TREBLE 0x03 /* treble control */
+#define TEA6330T_SADDR_FADER 0x04 /* fader control */
+#define TEA6330T_MFN 0x20 /* mute control for selected channels */
+#define TEA6330T_FCH 0x10 /* select fader channels - front or rear */
+#define TEA6330T_SADDR_AUDIO_SWITCH 0x05 /* audio switch */
+#define TEA6330T_GMU 0x80 /* mute control, general mute */
+#define TEA6330T_EQN 0x40 /* equalizer switchover (0=equalizer-on) */
+
+
+struct tea6330t {
+ struct snd_i2c_device *device;
+ struct snd_i2c_bus *bus;
+ int equalizer;
+ int fader;
+ unsigned char regs[8];
+ unsigned char mleft, mright;
+ unsigned char bass, treble;
+ unsigned char max_bass, max_treble;
+};
+
+
+int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer)
+{
+ int res;
+
+ snd_i2c_lock(bus);
+ res = snd_i2c_probeaddr(bus, TEA6330T_ADDR);
+ snd_i2c_unlock(bus);
+ return res;
+}
+
+#if 0
+static void snd_tea6330t_set(struct tea6330t *tea,
+ unsigned char addr, unsigned char value)
+{
+#if 0
+ printk(KERN_DEBUG "set - 0x%x/0x%x\n", addr, value);
+#endif
+ snd_i2c_write(tea->bus, TEA6330T_ADDR, addr, value, 1);
+}
+#endif
+
+#define TEA6330T_MASTER_VOLUME(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_tea6330t_info_master_volume, \
+ .get = snd_tea6330t_get_master_volume, .put = snd_tea6330t_put_master_volume }
+
+static int snd_tea6330t_info_master_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 43;
+ return 0;
+}
+
+static int snd_tea6330t_get_master_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+
+ snd_i2c_lock(tea->bus);
+ ucontrol->value.integer.value[0] = tea->mleft - 0x14;
+ ucontrol->value.integer.value[1] = tea->mright - 0x14;
+ snd_i2c_unlock(tea->bus);
+ return 0;
+}
+
+static int snd_tea6330t_put_master_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+ int change, count, err;
+ unsigned char bytes[3];
+ unsigned char val1, val2;
+
+ val1 = (ucontrol->value.integer.value[0] % 44) + 0x14;
+ val2 = (ucontrol->value.integer.value[1] % 44) + 0x14;
+ snd_i2c_lock(tea->bus);
+ change = val1 != tea->mleft || val2 != tea->mright;
+ tea->mleft = val1;
+ tea->mright = val2;
+ count = 0;
+ if (tea->regs[TEA6330T_SADDR_VOLUME_LEFT] != 0) {
+ bytes[count++] = TEA6330T_SADDR_VOLUME_LEFT;
+ bytes[count++] = tea->regs[TEA6330T_SADDR_VOLUME_LEFT] = tea->mleft;
+ }
+ if (tea->regs[TEA6330T_SADDR_VOLUME_RIGHT] != 0) {
+ if (count == 0)
+ bytes[count++] = TEA6330T_SADDR_VOLUME_RIGHT;
+ bytes[count++] = tea->regs[TEA6330T_SADDR_VOLUME_RIGHT] = tea->mright;
+ }
+ if (count > 0) {
+ if ((err = snd_i2c_sendbytes(tea->device, bytes, count)) < 0)
+ change = err;
+ }
+ snd_i2c_unlock(tea->bus);
+ return change;
+}
+
+#define TEA6330T_MASTER_SWITCH(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_tea6330t_info_master_switch, \
+ .get = snd_tea6330t_get_master_switch, .put = snd_tea6330t_put_master_switch }
+
+#define snd_tea6330t_info_master_switch snd_ctl_boolean_stereo_info
+
+static int snd_tea6330t_get_master_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+
+ snd_i2c_lock(tea->bus);
+ ucontrol->value.integer.value[0] = tea->regs[TEA6330T_SADDR_VOLUME_LEFT] == 0 ? 0 : 1;
+ ucontrol->value.integer.value[1] = tea->regs[TEA6330T_SADDR_VOLUME_RIGHT] == 0 ? 0 : 1;
+ snd_i2c_unlock(tea->bus);
+ return 0;
+}
+
+static int snd_tea6330t_put_master_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+ int change, err;
+ unsigned char bytes[3];
+ unsigned char oval1, oval2, val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & 1;
+ val2 = ucontrol->value.integer.value[1] & 1;
+ snd_i2c_lock(tea->bus);
+ oval1 = tea->regs[TEA6330T_SADDR_VOLUME_LEFT] == 0 ? 0 : 1;
+ oval2 = tea->regs[TEA6330T_SADDR_VOLUME_RIGHT] == 0 ? 0 : 1;
+ change = val1 != oval1 || val2 != oval2;
+ tea->regs[TEA6330T_SADDR_VOLUME_LEFT] = val1 ? tea->mleft : 0;
+ tea->regs[TEA6330T_SADDR_VOLUME_RIGHT] = val2 ? tea->mright : 0;
+ bytes[0] = TEA6330T_SADDR_VOLUME_LEFT;
+ bytes[1] = tea->regs[TEA6330T_SADDR_VOLUME_LEFT];
+ bytes[2] = tea->regs[TEA6330T_SADDR_VOLUME_RIGHT];
+ if ((err = snd_i2c_sendbytes(tea->device, bytes, 3)) < 0)
+ change = err;
+ snd_i2c_unlock(tea->bus);
+ return change;
+}
+
+#define TEA6330T_BASS(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_tea6330t_info_bass, \
+ .get = snd_tea6330t_get_bass, .put = snd_tea6330t_put_bass }
+
+static int snd_tea6330t_info_bass(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = tea->max_bass;
+ return 0;
+}
+
+static int snd_tea6330t_get_bass(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = tea->bass;
+ return 0;
+}
+
+static int snd_tea6330t_put_bass(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+ int change, err;
+ unsigned char bytes[2];
+ unsigned char val1;
+
+ val1 = ucontrol->value.integer.value[0] % (tea->max_bass + 1);
+ snd_i2c_lock(tea->bus);
+ tea->bass = val1;
+ val1 += tea->equalizer ? 7 : 3;
+ change = tea->regs[TEA6330T_SADDR_BASS] != val1;
+ bytes[0] = TEA6330T_SADDR_BASS;
+ bytes[1] = tea->regs[TEA6330T_SADDR_BASS] = val1;
+ if ((err = snd_i2c_sendbytes(tea->device, bytes, 2)) < 0)
+ change = err;
+ snd_i2c_unlock(tea->bus);
+ return change;
+}
+
+#define TEA6330T_TREBLE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_tea6330t_info_treble, \
+ .get = snd_tea6330t_get_treble, .put = snd_tea6330t_put_treble }
+
+static int snd_tea6330t_info_treble(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = tea->max_treble;
+ return 0;
+}
+
+static int snd_tea6330t_get_treble(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = tea->treble;
+ return 0;
+}
+
+static int snd_tea6330t_put_treble(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tea6330t *tea = snd_kcontrol_chip(kcontrol);
+ int change, err;
+ unsigned char bytes[2];
+ unsigned char val1;
+
+ val1 = ucontrol->value.integer.value[0] % (tea->max_treble + 1);
+ snd_i2c_lock(tea->bus);
+ tea->treble = val1;
+ val1 += 3;
+ change = tea->regs[TEA6330T_SADDR_TREBLE] != val1;
+ bytes[0] = TEA6330T_SADDR_TREBLE;
+ bytes[1] = tea->regs[TEA6330T_SADDR_TREBLE] = val1;
+ if ((err = snd_i2c_sendbytes(tea->device, bytes, 2)) < 0)
+ change = err;
+ snd_i2c_unlock(tea->bus);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_tea6330t_controls[] = {
+TEA6330T_MASTER_SWITCH("Master Playback Switch", 0),
+TEA6330T_MASTER_VOLUME("Master Playback Volume", 0),
+TEA6330T_BASS("Tone Control - Bass", 0),
+TEA6330T_TREBLE("Tone Control - Treble", 0)
+};
+
+static void snd_tea6330_free(struct snd_i2c_device *device)
+{
+ kfree(device->private_data);
+}
+
+int snd_tea6330t_update_mixer(struct snd_card *card,
+ struct snd_i2c_bus *bus,
+ int equalizer, int fader)
+{
+ struct snd_i2c_device *device;
+ struct tea6330t *tea;
+ struct snd_kcontrol_new *knew;
+ unsigned int idx;
+ int err = -ENOMEM;
+ u8 default_treble, default_bass;
+ unsigned char bytes[7];
+
+ tea = kzalloc(sizeof(*tea), GFP_KERNEL);
+ if (tea == NULL)
+ return -ENOMEM;
+ if ((err = snd_i2c_device_create(bus, "TEA6330T", TEA6330T_ADDR, &device)) < 0) {
+ kfree(tea);
+ return err;
+ }
+ tea->device = device;
+ tea->bus = bus;
+ tea->equalizer = equalizer;
+ tea->fader = fader;
+ device->private_data = tea;
+ device->private_free = snd_tea6330_free;
+
+ snd_i2c_lock(bus);
+
+ /* turn fader off and handle equalizer */
+ tea->regs[TEA6330T_SADDR_FADER] = 0x3f;
+ tea->regs[TEA6330T_SADDR_AUDIO_SWITCH] = equalizer ? 0 : TEA6330T_EQN;
+ /* initialize mixer */
+ if (!tea->equalizer) {
+ tea->max_bass = 9;
+ tea->max_treble = 8;
+ default_bass = 3 + 4;
+ tea->bass = 4;
+ default_treble = 3 + 4;
+ tea->treble = 4;
+ } else {
+ tea->max_bass = 5;
+ tea->max_treble = 0;
+ default_bass = 7 + 4;
+ tea->bass = 4;
+ default_treble = 3;
+ tea->treble = 0;
+ }
+ tea->mleft = tea->mright = 0x14;
+ tea->regs[TEA6330T_SADDR_BASS] = default_bass;
+ tea->regs[TEA6330T_SADDR_TREBLE] = default_treble;
+
+ /* compose I2C message and put the hardware to initial state */
+ bytes[0] = TEA6330T_SADDR_VOLUME_LEFT;
+ for (idx = 0; idx < 6; idx++)
+ bytes[idx+1] = tea->regs[idx];
+ if ((err = snd_i2c_sendbytes(device, bytes, 7)) < 0)
+ goto __error;
+
+ strcat(card->mixername, ",TEA6330T");
+ if ((err = snd_component_add(card, "TEA6330T")) < 0)
+ goto __error;
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_tea6330t_controls); idx++) {
+ knew = &snd_tea6330t_controls[idx];
+ if (tea->treble == 0 && !strcmp(knew->name, "Tone Control - Treble"))
+ continue;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(knew, tea))) < 0)
+ goto __error;
+ }
+
+ snd_i2c_unlock(bus);
+ return 0;
+
+ __error:
+ snd_i2c_unlock(bus);
+ snd_i2c_device_free(device);
+ return err;
+}
+
+EXPORT_SYMBOL(snd_tea6330t_detect);
+EXPORT_SYMBOL(snd_tea6330t_update_mixer);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_tea6330t_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_tea6330t_exit(void)
+{
+}
+
+module_init(alsa_tea6330t_init)
+module_exit(alsa_tea6330t_exit)
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
new file mode 100644
index 0000000..ce0aa04
--- /dev/null
+++ b/sound/isa/Kconfig
@@ -0,0 +1,415 @@
+# ALSA ISA drivers
+
+config SND_WSS_LIB
+ tristate
+ select SND_PCM
+
+config SND_SB_COMMON
+ tristate
+
+config SND_SB8_DSP
+ tristate
+ select SND_PCM
+ select SND_SB_COMMON
+
+config SND_SB16_DSP
+ tristate
+ select SND_PCM
+ select SND_SB_COMMON
+
+menuconfig SND_ISA
+ bool "ISA sound devices"
+ depends on ISA && ISA_DMA_API
+ default y
+ help
+ Support for sound devices connected via the ISA bus.
+
+if SND_ISA
+
+config SND_ADLIB
+ tristate "AdLib FM card"
+ select SND_OPL3_LIB
+ help
+ Say Y here to include support for AdLib FM cards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-adlib.
+
+config SND_AD1816A
+ tristate "Analog Devices SoundPort AD1816A"
+ depends on PNP
+ select ISAPNP
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ help
+ Say Y here to include support for Analog Devices SoundPort
+ AD1816A or compatible sound chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ad1816a.
+
+config SND_AD1848
+ tristate "Generic AD1848/CS4248 driver"
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for AD1848 (Analog Devices) or
+ CS4248 (Cirrus Logic - Crystal Semiconductors) chips.
+
+ For newer chips from Cirrus Logic, use the CS4231, CS4232 or
+ CS4236+ drivers.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ad1848.
+
+config SND_ALS100
+ tristate "Avance Logic ALS100/ALS120"
+ depends on PNP
+ select ISAPNP
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_SB16_DSP
+ help
+ Say Y here to include support for soundcards based on Avance
+ Logic ALS100, ALS110, ALS120 and ALS200 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-als100.
+
+config SND_AZT2320
+ tristate "Aztech Systems AZT2320"
+ depends on PNP
+ select ISAPNP
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for soundcards based on the
+ Aztech Systems AZT2320 chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-azt2320.
+
+config SND_CMI8330
+ tristate "C-Media CMI8330"
+ select SND_WSS_LIB
+ select SND_SB16_DSP
+ help
+ Say Y here to include support for soundcards based on the
+ C-Media CMI8330 chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cmi8330.
+
+config SND_CS4231
+ tristate "Generic Cirrus Logic CS4231 driver"
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for CS4231 chips from Cirrus
+ Logic - Crystal Semiconductors.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cs4231.
+
+config SND_CS4232
+ tristate "Generic Cirrus Logic CS4232 driver"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for CS4232 chips from Cirrus
+ Logic - Crystal Semiconductors.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cs4232.
+
+config SND_CS4236
+ tristate "Generic Cirrus Logic CS4236+ driver"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y to include support for CS4235,CS4236,CS4237B,CS4238B,
+ CS4239 chips from Cirrus Logic - Crystal Semiconductors.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cs4236.
+
+config SND_DT019X
+ tristate "Diamond Technologies DT-019X, Avance Logic ALS-007"
+ depends on PNP
+ select ISAPNP
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_SB16_DSP
+ help
+ Say Y here to include support for soundcards based on the
+ Diamond Technologies DT-019X or Avance Logic ALS-007 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-dt019x.
+
+config SND_ES968
+ tristate "Generic ESS ES968 driver"
+ depends on PNP
+ select ISAPNP
+ select SND_MPU401_UART
+ select SND_SB8_DSP
+ help
+ Say Y here to include support for ESS AudioDrive ES968 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-es968.
+
+config SND_ES1688
+ tristate "Generic ESS ES688/ES1688 driver"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ help
+ Say Y here to include support for ESS AudioDrive ES688 or
+ ES1688 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-es1688.
+
+config SND_ES18XX
+ tristate "Generic ESS ES18xx driver"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ help
+ Say Y here to include support for ESS AudioDrive ES18xx chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-es18xx.
+
+config SND_SC6000
+ tristate "Gallant SC-6000, Audio Excel DSP 16"
+ depends on HAS_IOPORT
+ select SND_WSS_LIB
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ help
+ Say Y here to include support for Gallant SC-6000 card and clones:
+ Audio Excel DSP 16 and Zoltrix AV302.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sc6000.
+
+config SND_GUSCLASSIC
+ tristate "Gravis UltraSound Classic"
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for Gravis UltraSound Classic
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-gusclassic.
+
+config SND_GUSEXTREME
+ tristate "Gravis UltraSound Extreme"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ help
+ Say Y here to include support for Gravis UltraSound Extreme
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-gusextreme.
+
+config SND_GUSMAX
+ tristate "Gravis UltraSound MAX"
+ select SND_RAWMIDI
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for Gravis UltraSound MAX
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-gusmax.
+
+config SND_INTERWAVE
+ tristate "AMD InterWave, Gravis UltraSound PnP"
+ depends on PNP
+ select SND_RAWMIDI
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for AMD InterWave based
+ soundcards (Gravis UltraSound Plug & Play, STB SoundRage32,
+ MED3210, Dynasonic Pro, Panasonic PCA761AW).
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-interwave.
+
+config SND_INTERWAVE_STB
+ tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)"
+ depends on PNP
+ select SND_RAWMIDI
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for AMD InterWave based
+ soundcards with a TEA6330T bass and treble regulator
+ (UltraSound 32-Pro).
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-interwave-stb.
+
+config SND_OPL3SA2
+ tristate "Yamaha OPL3-SA2/SA3"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for Yamaha OPL3-SA2 and OPL3-SA3
+ chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-opl3sa2.
+
+config SND_OPTI92X_AD1848
+ tristate "OPTi 82C92x - AD1848"
+ select SND_OPL3_LIB
+ select SND_OPL4_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for soundcards based on Opti
+ 82C92x or OTI-601 chips and using an AD1848 codec.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-opti92x-ad1848.
+
+config SND_OPTI92X_CS4231
+ tristate "OPTi 82C92x - CS4231"
+ select SND_OPL3_LIB
+ select SND_OPL4_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for soundcards based on Opti
+ 82C92x chips and using a CS4231 codec.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-opti92x-cs4231.
+
+config SND_OPTI93X
+ tristate "OPTi 82C93x"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for soundcards based on Opti
+ 82C93x chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-opti93x.
+
+config SND_MIRO
+ tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver"
+ select SND_OPL4_LIB
+ select SND_WSS_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Miro miroSOUND PCM1 pro,
+ miroSOUND PCM12 and miroSOUND PCM20 Radio soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-miro.
+
+config SND_SB8
+ tristate "Sound Blaster 1.0/2.0/Pro (8-bit)"
+ select SND_OPL3_LIB
+ select SND_RAWMIDI
+ select SND_SB8_DSP
+ help
+ Say Y here to include support for Creative Sound Blaster 1.0/
+ 2.0/Pro (8-bit) or 100% compatible soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sb8.
+
+config SND_SB16
+ tristate "Sound Blaster 16 (PnP)"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_SB16_DSP
+ help
+ Say Y here to include support for Sound Blaster 16 soundcards
+ (including the Plug and Play version).
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sb16.
+
+config SND_SBAWE
+ tristate "Sound Blaster AWE (32,64) (PnP)"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_SB16_DSP
+ help
+ Say Y here to include support for Sound Blaster AWE soundcards
+ (including the Plug and Play version).
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sbawe.
+
+config SND_SB16_CSP
+ bool "Sound Blaster 16/AWE CSP support"
+ depends on (SND_SB16 || SND_SBAWE) && (BROKEN || !PPC)
+ select FW_LOADER
+ help
+ Say Y here to include support for the CSP core. This special
+ coprocessor can do variable tasks like various compression and
+ decompression algorithms.
+
+config SND_SGALAXY
+ tristate "Aztech Sound Galaxy"
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for Aztech Sound Galaxy
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sgalaxy.
+
+config SND_SSCAPE
+ tristate "Ensoniq SoundScape PnP driver"
+ select SND_HWDEP
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for Ensoniq SoundScape PnP
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sscape.
+
+config SND_WAVEFRONT
+ tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)"
+ select FW_LOADER
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_WSS_LIB
+ help
+ Say Y here to include support for Turtle Beach Maui, Tropez
+ and Tropez+ soundcards based on the Wavefront chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-wavefront.
+
+config SND_WAVEFRONT_FIRMWARE_IN_KERNEL
+ bool "In-kernel firmware for Wavefront"
+ depends on SND_WAVEFRONT
+ default y
+ help
+ Say Y here to include the static firmware for FX DSP built in
+ the kernel for the Wavefront driver. If you choose N here,
+ you need to install the firmware files from the
+ alsa-firmware package.
+
+endif # SND_ISA
+
diff --git a/sound/isa/Makefile b/sound/isa/Makefile
new file mode 100644
index 0000000..63af13d
--- /dev/null
+++ b/sound/isa/Makefile
@@ -0,0 +1,30 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-adlib-objs := adlib.o
+snd-als100-objs := als100.o
+snd-azt2320-objs := azt2320.o
+snd-cmi8330-objs := cmi8330.o
+snd-dt019x-objs := dt019x.o
+snd-es18xx-objs := es18xx.o
+snd-opl3sa2-objs := opl3sa2.o
+snd-sc6000-objs := sc6000.o
+snd-sgalaxy-objs := sgalaxy.o
+snd-sscape-objs := sscape.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_ADLIB) += snd-adlib.o
+obj-$(CONFIG_SND_ALS100) += snd-als100.o
+obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o
+obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o
+obj-$(CONFIG_SND_DT019X) += snd-dt019x.o
+obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o
+obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o
+obj-$(CONFIG_SND_SC6000) += snd-sc6000.o
+obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o
+obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o
+
+obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \
+ sb/ wavefront/ wss/
diff --git a/sound/isa/ad1816a/Makefile b/sound/isa/ad1816a/Makefile
new file mode 100644
index 0000000..487ab23
--- /dev/null
+++ b/sound/isa/ad1816a/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ad1816a-objs := ad1816a.o ad1816a_lib.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AD1816A) += snd-ad1816a.o
diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c
new file mode 100644
index 0000000..7752424
--- /dev/null
+++ b/sound/isa/ad1816a/ad1816a.c
@@ -0,0 +1,289 @@
+
+/*
+ card-ad1816a.c - driver for ADI SoundPort AD1816A based soundcards.
+ Copyright (C) 2000 by Massimo Piccioni <dafastidio@libero.it>
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/ad1816a.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+
+#define PFX "ad1816a: "
+
+MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
+MODULE_DESCRIPTION("AD1816A, AD1815");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Highscreen,Sound-Boostar 16 3D},"
+ "{Analog Devices,AD1815},"
+ "{Analog Devices,AD1816A},"
+ "{TerraTec,Base 64},"
+ "{TerraTec,AudioSystem EWS64S},"
+ "{Aztech/Newcom SC-16 3D},"
+ "{Shark Predator ISA}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 1-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* Pnp setup */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* Pnp setup */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+static int clockfreq[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for ad1816a based soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for ad1816a based soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable ad1816a based soundcard.");
+module_param_array(clockfreq, int, NULL, 0444);
+MODULE_PARM_DESC(clockfreq, "Clock frequency for ad1816a driver (default = 0).");
+
+struct snd_card_ad1816a {
+ struct pnp_dev *dev;
+ struct pnp_dev *devmpu;
+};
+
+static struct pnp_card_device_id snd_ad1816a_pnpids[] = {
+ /* Analog Devices AD1815 */
+ { .id = "ADS7150", .devs = { { .id = "ADS7150" }, { .id = "ADS7151" } } },
+ /* Analog Device AD1816? */
+ { .id = "ADS7180", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
+ /* Analog Devices AD1816A - added by Kenneth Platz <kxp@atl.hp.com> */
+ { .id = "ADS7181", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
+ /* Analog Devices AD1816A - Aztech/Newcom SC-16 3D */
+ { .id = "AZT1022", .devs = { { .id = "AZT1018" }, { .id = "AZT2002" } } },
+ /* Highscreen Sound-Boostar 16 3D - added by Stefan Behnel */
+ { .id = "LWC1061", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
+ /* Highscreen Sound-Boostar 16 3D */
+ { .id = "MDK1605", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
+ /* Shark Predator ISA - added by Ken Arromdee */
+ { .id = "SMM7180", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
+ /* Analog Devices AD1816A - Terratec AudioSystem EWS64 S */
+ { .id = "TER1112", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
+ /* Analog Devices AD1816A - Terratec AudioSystem EWS64 S */
+ { .id = "TER1112", .devs = { { .id = "TER1100" }, { .id = "TER1101" } } },
+ /* Analog Devices AD1816A - Terratec Base 64 */
+ { .id = "TER1411", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
+ /* end */
+ { .id = "" }
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_ad1816a_pnpids);
+
+
+#define DRIVER_NAME "snd-card-ad1816a"
+
+
+static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->dev == NULL)
+ return -EBUSY;
+
+ acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL);
+ if (acard->devmpu == NULL) {
+ mpu_port[dev] = -1;
+ snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n");
+ }
+
+ pdev = acard->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ printk(KERN_ERR PFX "AUDIO PnP configure failure\n");
+ return -EBUSY;
+ }
+
+ port[dev] = pnp_port_start(pdev, 2);
+ fm_port[dev] = pnp_port_start(pdev, 1);
+ dma1[dev] = pnp_dma(pdev, 0);
+ dma2[dev] = pnp_dma(pdev, 1);
+ irq[dev] = pnp_irq(pdev, 0);
+
+ if (acard->devmpu == NULL)
+ return 0;
+
+ pdev = acard->devmpu;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ printk(KERN_ERR PFX "MPU401 PnP configure failure\n");
+ mpu_port[dev] = -1;
+ acard->devmpu = NULL;
+ } else {
+ mpu_port[dev] = pnp_port_start(pdev, 0);
+ mpu_irq[dev] = pnp_irq(pdev, 0);
+ }
+
+ return 0;
+}
+
+static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ int error;
+ struct snd_card *card;
+ struct snd_card_ad1816a *acard;
+ struct snd_ad1816a *chip;
+ struct snd_opl3 *opl3;
+
+ if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_ad1816a))) == NULL)
+ return -ENOMEM;
+ acard = (struct snd_card_ad1816a *)card->private_data;
+
+ if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) {
+ snd_card_free(card);
+ return error;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+
+ if ((error = snd_ad1816a_create(card, port[dev],
+ irq[dev],
+ dma1[dev],
+ dma2[dev],
+ &chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ if (clockfreq[dev] >= 5000 && clockfreq[dev] <= 100000)
+ chip->clock_freq = clockfreq[dev];
+
+ strcpy(card->driver, "AD1816A");
+ strcpy(card->shortname, "ADI SoundPort AD1816A");
+ sprintf(card->longname, "%s, SS at 0x%lx, irq %d, dma %d&%d",
+ card->shortname, chip->port, irq[dev], dma1[dev], dma2[dev]);
+
+ if ((error = snd_ad1816a_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if ((error = snd_ad1816a_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if (mpu_port[dev] > 0) {
+ if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
+ mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED,
+ NULL) < 0)
+ printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]);
+ }
+
+ if (fm_port[dev] > 0) {
+ if (snd_opl3_create(card,
+ fm_port[dev], fm_port[dev] + 2,
+ OPL3_HW_AUTO, 0, &opl3) < 0) {
+ printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx.\n", fm_port[dev], fm_port[dev] + 2);
+ } else {
+ if ((error = snd_opl3_timer_new(opl3, 1, 2)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ }
+ }
+
+ if ((error = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ return 0;
+}
+
+static unsigned int __devinitdata ad1816a_devices;
+
+static int __devinit snd_ad1816a_pnp_detect(struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ static int dev;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (!enable[dev])
+ continue;
+ res = snd_card_ad1816a_probe(dev, card, id);
+ if (res < 0)
+ return res;
+ dev++;
+ ad1816a_devices++;
+ return 0;
+ }
+ return -ENODEV;
+}
+
+static void __devexit snd_ad1816a_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+static struct pnp_card_driver ad1816a_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "ad1816a",
+ .id_table = snd_ad1816a_pnpids,
+ .probe = snd_ad1816a_pnp_detect,
+ .remove = __devexit_p(snd_ad1816a_pnp_remove),
+ /* FIXME: suspend/resume */
+};
+
+static int __init alsa_card_ad1816a_init(void)
+{
+ int err;
+
+ err = pnp_register_card_driver(&ad1816a_pnpc_driver);
+ if (err)
+ return err;
+
+ if (!ad1816a_devices) {
+ pnp_unregister_card_driver(&ad1816a_pnpc_driver);
+#ifdef MODULE
+ printk(KERN_ERR "no AD1816A based soundcards found.\n");
+#endif /* MODULE */
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_ad1816a_exit(void)
+{
+ pnp_unregister_card_driver(&ad1816a_pnpc_driver);
+}
+
+module_init(alsa_card_ad1816a_init)
+module_exit(alsa_card_ad1816a_exit)
diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c
new file mode 100644
index 0000000..3bfca7c
--- /dev/null
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -0,0 +1,977 @@
+/*
+ ad1816a.c - lowlevel code for Analog Devices AD1816A chip.
+ Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <sound/core.h>
+#include <sound/tlv.h>
+#include <sound/ad1816a.h>
+
+#include <asm/io.h>
+#include <asm/dma.h>
+
+static inline int snd_ad1816a_busy_wait(struct snd_ad1816a *chip)
+{
+ int timeout;
+
+ for (timeout = 1000; timeout-- > 0; udelay(10))
+ if (inb(AD1816A_REG(AD1816A_CHIP_STATUS)) & AD1816A_READY)
+ return 0;
+
+ snd_printk("chip busy.\n");
+ return -EBUSY;
+}
+
+static inline unsigned char snd_ad1816a_in(struct snd_ad1816a *chip, unsigned char reg)
+{
+ snd_ad1816a_busy_wait(chip);
+ return inb(AD1816A_REG(reg));
+}
+
+static inline void snd_ad1816a_out(struct snd_ad1816a *chip, unsigned char reg,
+ unsigned char value)
+{
+ snd_ad1816a_busy_wait(chip);
+ outb(value, AD1816A_REG(reg));
+}
+
+static inline void snd_ad1816a_out_mask(struct snd_ad1816a *chip, unsigned char reg,
+ unsigned char mask, unsigned char value)
+{
+ snd_ad1816a_out(chip, reg,
+ (value & mask) | (snd_ad1816a_in(chip, reg) & ~mask));
+}
+
+static unsigned short snd_ad1816a_read(struct snd_ad1816a *chip, unsigned char reg)
+{
+ snd_ad1816a_out(chip, AD1816A_INDIR_ADDR, reg & 0x3f);
+ return snd_ad1816a_in(chip, AD1816A_INDIR_DATA_LOW) |
+ (snd_ad1816a_in(chip, AD1816A_INDIR_DATA_HIGH) << 8);
+}
+
+static void snd_ad1816a_write(struct snd_ad1816a *chip, unsigned char reg,
+ unsigned short value)
+{
+ snd_ad1816a_out(chip, AD1816A_INDIR_ADDR, reg & 0x3f);
+ snd_ad1816a_out(chip, AD1816A_INDIR_DATA_LOW, value & 0xff);
+ snd_ad1816a_out(chip, AD1816A_INDIR_DATA_HIGH, (value >> 8) & 0xff);
+}
+
+static void snd_ad1816a_write_mask(struct snd_ad1816a *chip, unsigned char reg,
+ unsigned short mask, unsigned short value)
+{
+ snd_ad1816a_write(chip, reg,
+ (value & mask) | (snd_ad1816a_read(chip, reg) & ~mask));
+}
+
+
+static unsigned char snd_ad1816a_get_format(struct snd_ad1816a *chip,
+ unsigned int format, int channels)
+{
+ unsigned char retval = AD1816A_FMT_LINEAR_8;
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ retval = AD1816A_FMT_ULAW_8;
+ break;
+ case SNDRV_PCM_FORMAT_A_LAW:
+ retval = AD1816A_FMT_ALAW_8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ retval = AD1816A_FMT_LINEAR_16_LIT;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ retval = AD1816A_FMT_LINEAR_16_BIG;
+ }
+ return (channels > 1) ? (retval | AD1816A_FMT_STEREO) : retval;
+}
+
+static int snd_ad1816a_open(struct snd_ad1816a *chip, unsigned int mode)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ if (chip->mode & mode) {
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return -EAGAIN;
+ }
+
+ switch ((mode &= AD1816A_MODE_OPEN)) {
+ case AD1816A_MODE_PLAYBACK:
+ snd_ad1816a_out_mask(chip, AD1816A_INTERRUPT_STATUS,
+ AD1816A_PLAYBACK_IRQ_PENDING, 0x00);
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_PLAYBACK_IRQ_ENABLE, 0xffff);
+ break;
+ case AD1816A_MODE_CAPTURE:
+ snd_ad1816a_out_mask(chip, AD1816A_INTERRUPT_STATUS,
+ AD1816A_CAPTURE_IRQ_PENDING, 0x00);
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_CAPTURE_IRQ_ENABLE, 0xffff);
+ break;
+ case AD1816A_MODE_TIMER:
+ snd_ad1816a_out_mask(chip, AD1816A_INTERRUPT_STATUS,
+ AD1816A_TIMER_IRQ_PENDING, 0x00);
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_TIMER_IRQ_ENABLE, 0xffff);
+ }
+ chip->mode |= mode;
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return 0;
+}
+
+static void snd_ad1816a_close(struct snd_ad1816a *chip, unsigned int mode)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ switch ((mode &= AD1816A_MODE_OPEN)) {
+ case AD1816A_MODE_PLAYBACK:
+ snd_ad1816a_out_mask(chip, AD1816A_INTERRUPT_STATUS,
+ AD1816A_PLAYBACK_IRQ_PENDING, 0x00);
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_PLAYBACK_IRQ_ENABLE, 0x0000);
+ break;
+ case AD1816A_MODE_CAPTURE:
+ snd_ad1816a_out_mask(chip, AD1816A_INTERRUPT_STATUS,
+ AD1816A_CAPTURE_IRQ_PENDING, 0x00);
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_CAPTURE_IRQ_ENABLE, 0x0000);
+ break;
+ case AD1816A_MODE_TIMER:
+ snd_ad1816a_out_mask(chip, AD1816A_INTERRUPT_STATUS,
+ AD1816A_TIMER_IRQ_PENDING, 0x00);
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_TIMER_IRQ_ENABLE, 0x0000);
+ }
+ if (!((chip->mode &= ~mode) & AD1816A_MODE_OPEN))
+ chip->mode = 0;
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what,
+ int channel, int cmd, int iscapture)
+{
+ int error = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_STOP:
+ spin_lock(&chip->lock);
+ cmd = (cmd == SNDRV_PCM_TRIGGER_START) ? 0xff: 0x00;
+ /* if (what & AD1816A_PLAYBACK_ENABLE) */
+ /* That is not valid, because playback and capture enable
+ * are the same bit pattern, just to different addresses
+ */
+ if (! iscapture)
+ snd_ad1816a_out_mask(chip, AD1816A_PLAYBACK_CONFIG,
+ AD1816A_PLAYBACK_ENABLE, cmd);
+ else
+ snd_ad1816a_out_mask(chip, AD1816A_CAPTURE_CONFIG,
+ AD1816A_CAPTURE_ENABLE, cmd);
+ spin_unlock(&chip->lock);
+ break;
+ default:
+ snd_printk("invalid trigger mode 0x%x.\n", what);
+ error = -EINVAL;
+ }
+
+ return error;
+}
+
+static int snd_ad1816a_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ return snd_ad1816a_trigger(chip, AD1816A_PLAYBACK_ENABLE,
+ SNDRV_PCM_STREAM_PLAYBACK, cmd, 0);
+}
+
+static int snd_ad1816a_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ return snd_ad1816a_trigger(chip, AD1816A_CAPTURE_ENABLE,
+ SNDRV_PCM_STREAM_CAPTURE, cmd, 1);
+}
+
+static int snd_ad1816a_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_ad1816a_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_ad1816a_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ unsigned long flags;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size, rate;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ chip->p_dma_size = size = snd_pcm_lib_buffer_bytes(substream);
+ snd_ad1816a_out_mask(chip, AD1816A_PLAYBACK_CONFIG,
+ AD1816A_PLAYBACK_ENABLE | AD1816A_PLAYBACK_PIO, 0x00);
+
+ snd_dma_program(chip->dma1, runtime->dma_addr, size,
+ DMA_MODE_WRITE | DMA_AUTOINIT);
+
+ rate = runtime->rate;
+ if (chip->clock_freq)
+ rate = (rate * 33000) / chip->clock_freq;
+ snd_ad1816a_write(chip, AD1816A_PLAYBACK_SAMPLE_RATE, rate);
+ snd_ad1816a_out_mask(chip, AD1816A_PLAYBACK_CONFIG,
+ AD1816A_FMT_ALL | AD1816A_FMT_STEREO,
+ snd_ad1816a_get_format(chip, runtime->format,
+ runtime->channels));
+
+ snd_ad1816a_write(chip, AD1816A_PLAYBACK_BASE_COUNT,
+ snd_pcm_lib_period_bytes(substream) / 4 - 1);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return 0;
+}
+
+static int snd_ad1816a_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ unsigned long flags;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size, rate;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ chip->c_dma_size = size = snd_pcm_lib_buffer_bytes(substream);
+ snd_ad1816a_out_mask(chip, AD1816A_CAPTURE_CONFIG,
+ AD1816A_CAPTURE_ENABLE | AD1816A_CAPTURE_PIO, 0x00);
+
+ snd_dma_program(chip->dma2, runtime->dma_addr, size,
+ DMA_MODE_READ | DMA_AUTOINIT);
+
+ rate = runtime->rate;
+ if (chip->clock_freq)
+ rate = (rate * 33000) / chip->clock_freq;
+ snd_ad1816a_write(chip, AD1816A_CAPTURE_SAMPLE_RATE, rate);
+ snd_ad1816a_out_mask(chip, AD1816A_CAPTURE_CONFIG,
+ AD1816A_FMT_ALL | AD1816A_FMT_STEREO,
+ snd_ad1816a_get_format(chip, runtime->format,
+ runtime->channels));
+
+ snd_ad1816a_write(chip, AD1816A_CAPTURE_BASE_COUNT,
+ snd_pcm_lib_period_bytes(substream) / 4 - 1);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return 0;
+}
+
+
+static snd_pcm_uframes_t snd_ad1816a_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+ if (!(chip->mode & AD1816A_MODE_PLAYBACK))
+ return 0;
+ ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_ad1816a_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+ if (!(chip->mode & AD1816A_MODE_CAPTURE))
+ return 0;
+ ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+
+static irqreturn_t snd_ad1816a_interrupt(int irq, void *dev_id)
+{
+ struct snd_ad1816a *chip = dev_id;
+ unsigned char status;
+
+ spin_lock(&chip->lock);
+ status = snd_ad1816a_in(chip, AD1816A_INTERRUPT_STATUS);
+ spin_unlock(&chip->lock);
+
+ if ((status & AD1816A_PLAYBACK_IRQ_PENDING) && chip->playback_substream)
+ snd_pcm_period_elapsed(chip->playback_substream);
+
+ if ((status & AD1816A_CAPTURE_IRQ_PENDING) && chip->capture_substream)
+ snd_pcm_period_elapsed(chip->capture_substream);
+
+ if ((status & AD1816A_TIMER_IRQ_PENDING) && chip->timer)
+ snd_timer_interrupt(chip->timer, chip->timer->sticks);
+
+ spin_lock(&chip->lock);
+ snd_ad1816a_out(chip, AD1816A_INTERRUPT_STATUS, 0x00);
+ spin_unlock(&chip->lock);
+ return IRQ_HANDLED;
+}
+
+
+static struct snd_pcm_hardware snd_ad1816a_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW |
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 55200,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_ad1816a_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW |
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 55200,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+#if 0 /* not used now */
+static int snd_ad1816a_timer_close(struct snd_timer *timer)
+{
+ struct snd_ad1816a *chip = snd_timer_chip(timer);
+ snd_ad1816a_close(chip, AD1816A_MODE_TIMER);
+ return 0;
+}
+
+static int snd_ad1816a_timer_open(struct snd_timer *timer)
+{
+ struct snd_ad1816a *chip = snd_timer_chip(timer);
+ snd_ad1816a_open(chip, AD1816A_MODE_TIMER);
+ return 0;
+}
+
+static unsigned long snd_ad1816a_timer_resolution(struct snd_timer *timer)
+{
+ if (snd_BUG_ON(!timer))
+ return 0;
+
+ return 10000;
+}
+
+static int snd_ad1816a_timer_start(struct snd_timer *timer)
+{
+ unsigned short bits;
+ unsigned long flags;
+ struct snd_ad1816a *chip = snd_timer_chip(timer);
+ spin_lock_irqsave(&chip->lock, flags);
+ bits = snd_ad1816a_read(chip, AD1816A_INTERRUPT_ENABLE);
+
+ if (!(bits & AD1816A_TIMER_ENABLE)) {
+ snd_ad1816a_write(chip, AD1816A_TIMER_BASE_COUNT,
+ timer->sticks & 0xffff);
+
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_TIMER_ENABLE, 0xffff);
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return 0;
+}
+
+static int snd_ad1816a_timer_stop(struct snd_timer *timer)
+{
+ unsigned long flags;
+ struct snd_ad1816a *chip = snd_timer_chip(timer);
+ spin_lock_irqsave(&chip->lock, flags);
+
+ snd_ad1816a_write_mask(chip, AD1816A_INTERRUPT_ENABLE,
+ AD1816A_TIMER_ENABLE, 0x0000);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return 0;
+}
+
+static struct snd_timer_hardware snd_ad1816a_timer_table = {
+ .flags = SNDRV_TIMER_HW_AUTO,
+ .resolution = 10000,
+ .ticks = 65535,
+ .open = snd_ad1816a_timer_open,
+ .close = snd_ad1816a_timer_close,
+ .c_resolution = snd_ad1816a_timer_resolution,
+ .start = snd_ad1816a_timer_start,
+ .stop = snd_ad1816a_timer_stop,
+};
+#endif /* not used now */
+
+
+static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int error;
+
+ if ((error = snd_ad1816a_open(chip, AD1816A_MODE_PLAYBACK)) < 0)
+ return error;
+ runtime->hw = snd_ad1816a_playback;
+ snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max);
+ snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max);
+ chip->playback_substream = substream;
+ return 0;
+}
+
+static int snd_ad1816a_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int error;
+
+ if ((error = snd_ad1816a_open(chip, AD1816A_MODE_CAPTURE)) < 0)
+ return error;
+ runtime->hw = snd_ad1816a_capture;
+ snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max);
+ snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max);
+ chip->capture_substream = substream;
+ return 0;
+}
+
+static int snd_ad1816a_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+
+ chip->playback_substream = NULL;
+ snd_ad1816a_close(chip, AD1816A_MODE_PLAYBACK);
+ return 0;
+}
+
+static int snd_ad1816a_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ad1816a *chip = snd_pcm_substream_chip(substream);
+
+ chip->capture_substream = NULL;
+ snd_ad1816a_close(chip, AD1816A_MODE_CAPTURE);
+ return 0;
+}
+
+
+static void __devinit snd_ad1816a_init(struct snd_ad1816a *chip)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ snd_ad1816a_out(chip, AD1816A_INTERRUPT_STATUS, 0x00);
+ snd_ad1816a_out_mask(chip, AD1816A_PLAYBACK_CONFIG,
+ AD1816A_PLAYBACK_ENABLE | AD1816A_PLAYBACK_PIO, 0x00);
+ snd_ad1816a_out_mask(chip, AD1816A_CAPTURE_CONFIG,
+ AD1816A_CAPTURE_ENABLE | AD1816A_CAPTURE_PIO, 0x00);
+ snd_ad1816a_write(chip, AD1816A_INTERRUPT_ENABLE, 0x0000);
+ snd_ad1816a_write_mask(chip, AD1816A_CHIP_CONFIG,
+ AD1816A_CAPTURE_NOT_EQUAL | AD1816A_WSS_ENABLE, 0xffff);
+ snd_ad1816a_write(chip, AD1816A_DSP_CONFIG, 0x0000);
+ snd_ad1816a_write(chip, AD1816A_POWERDOWN_CTRL, 0x0000);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+static int __devinit snd_ad1816a_probe(struct snd_ad1816a *chip)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ switch (chip->version = snd_ad1816a_read(chip, AD1816A_VERSION_ID)) {
+ case 0:
+ chip->hardware = AD1816A_HW_AD1815;
+ break;
+ case 1:
+ chip->hardware = AD1816A_HW_AD18MAX10;
+ break;
+ case 3:
+ chip->hardware = AD1816A_HW_AD1816A;
+ break;
+ default:
+ chip->hardware = AD1816A_HW_AUTO;
+ }
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return 0;
+}
+
+static int snd_ad1816a_free(struct snd_ad1816a *chip)
+{
+ release_and_free_resource(chip->res_port);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *) chip);
+ if (chip->dma1 >= 0) {
+ snd_dma_disable(chip->dma1);
+ free_dma(chip->dma1);
+ }
+ if (chip->dma2 >= 0) {
+ snd_dma_disable(chip->dma2);
+ free_dma(chip->dma2);
+ }
+ kfree(chip);
+ return 0;
+}
+
+static int snd_ad1816a_dev_free(struct snd_device *device)
+{
+ struct snd_ad1816a *chip = device->device_data;
+ return snd_ad1816a_free(chip);
+}
+
+static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip)
+{
+ switch (chip->hardware) {
+ case AD1816A_HW_AD1816A: return "AD1816A";
+ case AD1816A_HW_AD1815: return "AD1815";
+ case AD1816A_HW_AD18MAX10: return "AD18max10";
+ default:
+ snd_printk("Unknown chip version %d:%d.\n",
+ chip->version, chip->hardware);
+ return "AD1816A - unknown";
+ }
+}
+
+int __devinit snd_ad1816a_create(struct snd_card *card,
+ unsigned long port, int irq, int dma1, int dma2,
+ struct snd_ad1816a **rchip)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ad1816a_dev_free,
+ };
+ int error;
+ struct snd_ad1816a *chip;
+
+ *rchip = NULL;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ chip->irq = -1;
+ chip->dma1 = -1;
+ chip->dma2 = -1;
+
+ if ((chip->res_port = request_region(port, 16, "AD1816A")) == NULL) {
+ snd_printk(KERN_ERR "ad1816a: can't grab port 0x%lx\n", port);
+ snd_ad1816a_free(chip);
+ return -EBUSY;
+ }
+ if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) {
+ snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq);
+ snd_ad1816a_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = irq;
+ if (request_dma(dma1, "AD1816A - 1")) {
+ snd_printk(KERN_ERR "ad1816a: can't grab DMA1 %d\n", dma1);
+ snd_ad1816a_free(chip);
+ return -EBUSY;
+ }
+ chip->dma1 = dma1;
+ if (request_dma(dma2, "AD1816A - 2")) {
+ snd_printk(KERN_ERR "ad1816a: can't grab DMA2 %d\n", dma2);
+ snd_ad1816a_free(chip);
+ return -EBUSY;
+ }
+ chip->dma2 = dma2;
+
+ chip->card = card;
+ chip->port = port;
+ spin_lock_init(&chip->lock);
+
+ if ((error = snd_ad1816a_probe(chip))) {
+ snd_ad1816a_free(chip);
+ return error;
+ }
+
+ snd_ad1816a_init(chip);
+
+ /* Register device */
+ if ((error = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_ad1816a_free(chip);
+ return error;
+ }
+
+ *rchip = chip;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ad1816a_playback_ops = {
+ .open = snd_ad1816a_playback_open,
+ .close = snd_ad1816a_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ad1816a_hw_params,
+ .hw_free = snd_ad1816a_hw_free,
+ .prepare = snd_ad1816a_playback_prepare,
+ .trigger = snd_ad1816a_playback_trigger,
+ .pointer = snd_ad1816a_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_ad1816a_capture_ops = {
+ .open = snd_ad1816a_capture_open,
+ .close = snd_ad1816a_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ad1816a_hw_params,
+ .hw_free = snd_ad1816a_hw_free,
+ .prepare = snd_ad1816a_capture_prepare,
+ .trigger = snd_ad1816a_capture_trigger,
+ .pointer = snd_ad1816a_capture_pointer,
+};
+
+int __devinit snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm)
+{
+ int error;
+ struct snd_pcm *pcm;
+
+ if ((error = snd_pcm_new(chip->card, "AD1816A", device, 1, 1, &pcm)))
+ return error;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ad1816a_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ad1816a_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = (chip->dma1 == chip->dma2 ) ? SNDRV_PCM_INFO_JOINT_DUPLEX : 0;
+
+ strcpy(pcm->name, snd_ad1816a_chip_id(chip));
+ snd_ad1816a_init(chip);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, chip->dma1 > 3 || chip->dma2 > 3 ? 128*1024 : 64*1024);
+
+ chip->pcm = pcm;
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+#if 0 /* not used now */
+int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer)
+{
+ struct snd_timer *timer;
+ struct snd_timer_id tid;
+ int error;
+
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = chip->card->number;
+ tid.device = device;
+ tid.subdevice = 0;
+ if ((error = snd_timer_new(chip->card, "AD1816A", &tid, &timer)) < 0)
+ return error;
+ strcpy(timer->name, snd_ad1816a_chip_id(chip));
+ timer->private_data = chip;
+ chip->timer = timer;
+ timer->hw = snd_ad1816a_timer_table;
+ if (rtimer)
+ *rtimer = timer;
+ return 0;
+}
+#endif /* not used now */
+
+/*
+ *
+ */
+
+static int snd_ad1816a_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[8] = {
+ "Line", "Mix", "CD", "Synth", "Video",
+ "Mic", "Phone",
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 2;
+ uinfo->value.enumerated.items = 7;
+ if (uinfo->value.enumerated.item > 6)
+ uinfo->value.enumerated.item = 6;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ad1816a_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ad1816a *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned short val;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ val = snd_ad1816a_read(chip, AD1816A_ADC_SOURCE_SEL);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ ucontrol->value.enumerated.item[0] = (val >> 12) & 7;
+ ucontrol->value.enumerated.item[1] = (val >> 4) & 7;
+ return 0;
+}
+
+static int snd_ad1816a_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ad1816a *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned short val;
+ int change;
+
+ if (ucontrol->value.enumerated.item[0] > 6 ||
+ ucontrol->value.enumerated.item[1] > 6)
+ return -EINVAL;
+ val = (ucontrol->value.enumerated.item[0] << 12) |
+ (ucontrol->value.enumerated.item[1] << 4);
+ spin_lock_irqsave(&chip->lock, flags);
+ change = snd_ad1816a_read(chip, AD1816A_ADC_SOURCE_SEL) != val;
+ snd_ad1816a_write(chip, AD1816A_ADC_SOURCE_SEL, val);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return change;
+}
+
+#define AD1816A_SINGLE_TLV(xname, reg, shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .info = snd_ad1816a_info_single, \
+ .get = snd_ad1816a_get_single, .put = snd_ad1816a_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = (xtlv) } }
+#define AD1816A_SINGLE(xname, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_ad1816a_info_single, \
+ .get = snd_ad1816a_get_single, .put = snd_ad1816a_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_ad1816a_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_ad1816a_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ad1816a *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ ucontrol->value.integer.value[0] = (snd_ad1816a_read(chip, reg) >> shift) & mask;
+ spin_unlock_irqrestore(&chip->lock, flags);
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_ad1816a_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ad1816a *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short old_val, val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+ spin_lock_irqsave(&chip->lock, flags);
+ old_val = snd_ad1816a_read(chip, reg);
+ val = (old_val & ~(mask << shift)) | val;
+ change = val != old_val;
+ snd_ad1816a_write(chip, reg, val);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return change;
+}
+
+#define AD1816A_DOUBLE_TLV(xname, reg, shift_left, shift_right, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .info = snd_ad1816a_info_double, \
+ .get = snd_ad1816a_get_double, .put = snd_ad1816a_put_double, \
+ .private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = (xtlv) } }
+
+#define AD1816A_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_ad1816a_info_double, \
+ .get = snd_ad1816a_get_double, .put = snd_ad1816a_put_double, \
+ .private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24) }
+
+static int snd_ad1816a_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_ad1816a_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ad1816a *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift_left = (kcontrol->private_value >> 8) & 0x0f;
+ int shift_right = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ unsigned short val;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ val = snd_ad1816a_read(chip, reg);
+ ucontrol->value.integer.value[0] = (val >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (val >> shift_right) & mask;
+ spin_unlock_irqrestore(&chip->lock, flags);
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_ad1816a_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ad1816a *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift_left = (kcontrol->private_value >> 8) & 0x0f;
+ int shift_right = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short old_val, val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irqsave(&chip->lock, flags);
+ old_val = snd_ad1816a_read(chip, reg);
+ val1 = (old_val & ~((mask << shift_left) | (mask << shift_right))) | val1 | val2;
+ change = val1 != old_val;
+ snd_ad1816a_write(chip, reg, val1);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0);
+
+static struct snd_kcontrol_new snd_ad1816a_controls[] __devinitdata = {
+AD1816A_DOUBLE("Master Playback Switch", AD1816A_MASTER_ATT, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("Master Playback Volume", AD1816A_MASTER_ATT, 8, 0, 31, 1,
+ db_scale_5bit),
+AD1816A_DOUBLE("PCM Playback Switch", AD1816A_VOICE_ATT, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("PCM Playback Volume", AD1816A_VOICE_ATT, 8, 0, 63, 1,
+ db_scale_6bit),
+AD1816A_DOUBLE("Line Playback Switch", AD1816A_LINE_GAIN_ATT, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("Line Playback Volume", AD1816A_LINE_GAIN_ATT, 8, 0, 31, 1,
+ db_scale_5bit_12db_max),
+AD1816A_DOUBLE("CD Playback Switch", AD1816A_CD_GAIN_ATT, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("CD Playback Volume", AD1816A_CD_GAIN_ATT, 8, 0, 31, 1,
+ db_scale_5bit_12db_max),
+AD1816A_DOUBLE("Synth Playback Switch", AD1816A_SYNTH_GAIN_ATT, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("Synth Playback Volume", AD1816A_SYNTH_GAIN_ATT, 8, 0, 31, 1,
+ db_scale_5bit_12db_max),
+AD1816A_DOUBLE("FM Playback Switch", AD1816A_FM_ATT, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("FM Playback Volume", AD1816A_FM_ATT, 8, 0, 63, 1,
+ db_scale_6bit),
+AD1816A_SINGLE("Mic Playback Switch", AD1816A_MIC_GAIN_ATT, 15, 1, 1),
+AD1816A_SINGLE_TLV("Mic Playback Volume", AD1816A_MIC_GAIN_ATT, 8, 31, 1,
+ db_scale_5bit_12db_max),
+AD1816A_SINGLE("Mic Boost", AD1816A_MIC_GAIN_ATT, 14, 1, 0),
+AD1816A_DOUBLE("Video Playback Switch", AD1816A_VID_GAIN_ATT, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("Video Playback Volume", AD1816A_VID_GAIN_ATT, 8, 0, 31, 1,
+ db_scale_5bit_12db_max),
+AD1816A_SINGLE("Phone Capture Switch", AD1816A_PHONE_IN_GAIN_ATT, 15, 1, 1),
+AD1816A_SINGLE_TLV("Phone Capture Volume", AD1816A_PHONE_IN_GAIN_ATT, 0, 15, 1,
+ db_scale_4bit),
+AD1816A_SINGLE("Phone Playback Switch", AD1816A_PHONE_OUT_ATT, 7, 1, 1),
+AD1816A_SINGLE_TLV("Phone Playback Volume", AD1816A_PHONE_OUT_ATT, 0, 31, 1,
+ db_scale_5bit),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_ad1816a_info_mux,
+ .get = snd_ad1816a_get_mux,
+ .put = snd_ad1816a_put_mux,
+},
+AD1816A_DOUBLE("Capture Switch", AD1816A_ADC_PGA, 15, 7, 1, 1),
+AD1816A_DOUBLE_TLV("Capture Volume", AD1816A_ADC_PGA, 8, 0, 15, 0,
+ db_scale_rec_gain),
+AD1816A_SINGLE("3D Control - Switch", AD1816A_3D_PHAT_CTRL, 15, 1, 1),
+AD1816A_SINGLE("3D Control - Level", AD1816A_3D_PHAT_CTRL, 0, 15, 0),
+};
+
+int __devinit snd_ad1816a_mixer(struct snd_ad1816a *chip)
+{
+ struct snd_card *card;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!chip || !chip->card))
+ return -EINVAL;
+
+ card = chip->card;
+
+ strcpy(card->mixername, snd_ad1816a_chip_id(chip));
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_ad1816a_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_ad1816a_controls[idx], chip))) < 0)
+ return err;
+ }
+ return 0;
+}
diff --git a/sound/isa/ad1848/Makefile b/sound/isa/ad1848/Makefile
new file mode 100644
index 0000000..3d6dea3
--- /dev/null
+++ b/sound/isa/ad1848/Makefile
@@ -0,0 +1,10 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ad1848-objs := ad1848.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AD1848) += snd-ad1848.o
+
diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c
new file mode 100644
index 0000000..223a6c0
--- /dev/null
+++ b/sound/isa/ad1848/ad1848.c
@@ -0,0 +1,188 @@
+/*
+ * Generic driver for AD1848/AD1847/CS4248 chips (0.1 Alpha)
+ * Copyright (c) by Tugrul Galatali <galatalt@stuy.edu>,
+ * Jaroslav Kysela <perex@perex.cz>
+ * Based on card-4232.c by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/initval.h>
+
+#define CRD_NAME "Generic AD1848/AD1847/CS4248"
+#define DEV_NAME "ad1848"
+
+MODULE_DESCRIPTION(CRD_NAME);
+MODULE_AUTHOR("Tugrul Galatali <galatalt@stuy.edu>, Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1848},"
+ "{Analog Devices,AD1847},"
+ "{Crystal Semiconductors,CS4248}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,11,12,15 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+static int thinkpad[SNDRV_CARDS]; /* Thinkpad special case */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver.");
+module_param_array(thinkpad, bool, NULL, 0444);
+MODULE_PARM_DESC(thinkpad, "Enable only for the onboard CS4248 of IBM Thinkpad 360/750/755 series.");
+
+static int __devinit snd_ad1848_match(struct device *dev, unsigned int n)
+{
+ if (!enable[n])
+ return 0;
+
+ if (port[n] == SNDRV_AUTO_PORT) {
+ dev_err(dev, "please specify port\n");
+ return 0;
+ }
+ if (irq[n] == SNDRV_AUTO_IRQ) {
+ dev_err(dev, "please specify irq\n");
+ return 0;
+ }
+ if (dma1[n] == SNDRV_AUTO_DMA) {
+ dev_err(dev, "please specify dma1\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_ad1848_probe(struct device *dev, unsigned int n)
+{
+ struct snd_card *card;
+ struct snd_wss *chip;
+ struct snd_pcm *pcm;
+ int error;
+
+ card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
+ if (!card)
+ return -EINVAL;
+
+ error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], -1,
+ thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT,
+ 0, &chip);
+ if (error < 0)
+ goto out;
+
+ card->private_data = chip;
+
+ error = snd_wss_pcm(chip, 0, &pcm);
+ if (error < 0)
+ goto out;
+
+ error = snd_wss_mixer(chip);
+ if (error < 0)
+ goto out;
+
+ strcpy(card->driver, "AD1848");
+ strcpy(card->shortname, pcm->name);
+
+ sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
+ pcm->name, chip->port, irq[n], dma1[n]);
+ if (thinkpad[n])
+ strcat(card->longname, " [Thinkpad]");
+
+ snd_card_set_dev(card, dev);
+
+ error = snd_card_register(card);
+ if (error < 0)
+ goto out;
+
+ dev_set_drvdata(dev, card);
+ return 0;
+
+out: snd_card_free(card);
+ return error;
+}
+
+static int __devexit snd_ad1848_remove(struct device *dev, unsigned int n)
+{
+ snd_card_free(dev_get_drvdata(dev));
+ dev_set_drvdata(dev, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_ad1848_suspend(struct device *dev, unsigned int n, pm_message_t state)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_wss *chip = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ chip->suspend(chip);
+ return 0;
+}
+
+static int snd_ad1848_resume(struct device *dev, unsigned int n)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_wss *chip = card->private_data;
+
+ chip->resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct isa_driver snd_ad1848_driver = {
+ .match = snd_ad1848_match,
+ .probe = snd_ad1848_probe,
+ .remove = __devexit_p(snd_ad1848_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_ad1848_suspend,
+ .resume = snd_ad1848_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ }
+};
+
+static int __init alsa_card_ad1848_init(void)
+{
+ return isa_register_driver(&snd_ad1848_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_ad1848_exit(void)
+{
+ isa_unregister_driver(&snd_ad1848_driver);
+}
+
+module_init(alsa_card_ad1848_init);
+module_exit(alsa_card_ad1848_exit);
diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c
new file mode 100644
index 0000000..374b717
--- /dev/null
+++ b/sound/isa/adlib.c
@@ -0,0 +1,129 @@
+/*
+ * AdLib FM card driver.
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/isa.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/opl3.h>
+
+#define CRD_NAME "AdLib FM"
+#define DEV_NAME "adlib"
+
+MODULE_DESCRIPTION(CRD_NAME);
+MODULE_AUTHOR("Rene Herman");
+MODULE_LICENSE("GPL");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE;
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver.");
+
+static int __devinit snd_adlib_match(struct device *dev, unsigned int n)
+{
+ if (!enable[n])
+ return 0;
+
+ if (port[n] == SNDRV_AUTO_PORT) {
+ dev_err(dev, "please specify port\n");
+ return 0;
+ }
+ return 1;
+}
+
+static void snd_adlib_free(struct snd_card *card)
+{
+ release_and_free_resource(card->private_data);
+}
+
+static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
+{
+ struct snd_card *card;
+ struct snd_opl3 *opl3;
+ int error;
+
+ card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
+ if (!card) {
+ dev_err(dev, "could not create card\n");
+ return -EINVAL;
+ }
+
+ card->private_data = request_region(port[n], 4, CRD_NAME);
+ if (!card->private_data) {
+ dev_err(dev, "could not grab ports\n");
+ error = -EBUSY;
+ goto out;
+ }
+ card->private_free = snd_adlib_free;
+
+ strcpy(card->driver, DEV_NAME);
+ strcpy(card->shortname, CRD_NAME);
+ sprintf(card->longname, CRD_NAME " at %#lx", port[n]);
+
+ error = snd_opl3_create(card, port[n], port[n] + 2, OPL3_HW_AUTO, 1, &opl3);
+ if (error < 0) {
+ dev_err(dev, "could not create OPL\n");
+ goto out;
+ }
+
+ error = snd_opl3_hwdep_new(opl3, 0, 0, NULL);
+ if (error < 0) {
+ dev_err(dev, "could not create FM\n");
+ goto out;
+ }
+
+ snd_card_set_dev(card, dev);
+
+ error = snd_card_register(card);
+ if (error < 0) {
+ dev_err(dev, "could not register card\n");
+ goto out;
+ }
+
+ dev_set_drvdata(dev, card);
+ return 0;
+
+out: snd_card_free(card);
+ return error;
+}
+
+static int __devexit snd_adlib_remove(struct device *dev, unsigned int n)
+{
+ snd_card_free(dev_get_drvdata(dev));
+ dev_set_drvdata(dev, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_adlib_driver = {
+ .match = snd_adlib_match,
+ .probe = snd_adlib_probe,
+ .remove = __devexit_p(snd_adlib_remove),
+
+ .driver = {
+ .name = DEV_NAME
+ }
+};
+
+static int __init alsa_card_adlib_init(void)
+{
+ return isa_register_driver(&snd_adlib_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_adlib_exit(void)
+{
+ isa_unregister_driver(&snd_adlib_driver);
+}
+
+module_init(alsa_card_adlib_init);
+module_exit(alsa_card_adlib_exit);
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
new file mode 100644
index 0000000..f1ce30f
--- /dev/null
+++ b/sound/isa/als100.c
@@ -0,0 +1,327 @@
+
+/*
+ card-als100.c - driver for Avance Logic ALS100 based soundcards.
+ Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
+
+ Thanks to Pierfrancesco 'qM2' Passerini.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/wait.h>
+#include <linux/time.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/sb.h>
+
+#define PFX "als100: "
+
+MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
+MODULE_DESCRIPTION("Avance Logic ALS1X0");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP},"
+ "{Avance Logic,ALS110},"
+ "{Avance Logic,ALS120},"
+ "{Avance Logic,ALS200},"
+ "{3D Melody,MF1000},"
+ "{Digimate,3D Sound},"
+ "{Avance Logic,ALS120},"
+ "{RTL,RTL3000}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */
+static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for als100 based soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for als100 based soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable als100 based soundcard.");
+
+struct snd_card_als100 {
+ int dev_no;
+ struct pnp_dev *dev;
+ struct pnp_dev *devmpu;
+ struct pnp_dev *devopl;
+ struct snd_sb *chip;
+};
+
+static struct pnp_card_device_id snd_als100_pnpids[] = {
+ /* ALS100 - PRO16PNP */
+ { .id = "ALS0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } },
+ /* ALS110 - MF1000 - Digimate 3D Sound */
+ { .id = "ALS0110", .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } } },
+ /* ALS120 */
+ { .id = "ALS0120", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } },
+ /* ALS200 */
+ { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } } },
+ /* ALS200 OEM */
+ { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } } },
+ /* RTL3000 */
+ { .id = "RTL3000", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } },
+ { .id = "", } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_als100_pnpids);
+
+#define DRIVER_NAME "snd-card-als100"
+
+static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->dev == NULL)
+ return -ENODEV;
+
+ acard->devmpu = pnp_request_card_device(card, id->devs[1].id, acard->dev);
+ acard->devopl = pnp_request_card_device(card, id->devs[2].id, acard->dev);
+
+ pdev = acard->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n");
+ return err;
+ }
+ port[dev] = pnp_port_start(pdev, 0);
+ dma8[dev] = pnp_dma(pdev, 1);
+ dma16[dev] = pnp_dma(pdev, 0);
+ irq[dev] = pnp_irq(pdev, 0);
+
+ pdev = acard->devmpu;
+ if (pdev != NULL) {
+ err = pnp_activate_dev(pdev);
+ if (err < 0)
+ goto __mpu_error;
+ mpu_port[dev] = pnp_port_start(pdev, 0);
+ mpu_irq[dev] = pnp_irq(pdev, 0);
+ } else {
+ __mpu_error:
+ if (pdev) {
+ pnp_release_card_device(pdev);
+ snd_printk(KERN_ERR PFX "MPU401 pnp configure failure, skipping\n");
+ }
+ acard->devmpu = NULL;
+ mpu_port[dev] = -1;
+ }
+
+ pdev = acard->devopl;
+ if (pdev != NULL) {
+ err = pnp_activate_dev(pdev);
+ if (err < 0)
+ goto __fm_error;
+ fm_port[dev] = pnp_port_start(pdev, 0);
+ } else {
+ __fm_error:
+ if (pdev) {
+ pnp_release_card_device(pdev);
+ snd_printk(KERN_ERR PFX "OPL3 pnp configure failure, skipping\n");
+ }
+ acard->devopl = NULL;
+ fm_port[dev] = -1;
+ }
+
+ return 0;
+}
+
+static int __devinit snd_card_als100_probe(int dev,
+ struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ int error;
+ struct snd_sb *chip;
+ struct snd_card *card;
+ struct snd_card_als100 *acard;
+ struct snd_opl3 *opl3;
+
+ if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_als100))) == NULL)
+ return -ENOMEM;
+ acard = card->private_data;
+
+ if ((error = snd_card_als100_pnp(dev, acard, pcard, pid))) {
+ snd_card_free(card);
+ return error;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+
+ if ((error = snd_sbdsp_create(card, port[dev],
+ irq[dev],
+ snd_sb16dsp_interrupt,
+ dma8[dev],
+ dma16[dev],
+ SB_HW_ALS100, &chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ acard->chip = chip;
+
+ strcpy(card->driver, "ALS100");
+ strcpy(card->shortname, "Avance Logic ALS100");
+ sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d",
+ card->shortname, chip->name, chip->port,
+ irq[dev], dma8[dev], dma16[dev]);
+
+ if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if ((error = snd_sbmixer_new(chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_mpu401_uart_new(card, 0, MPU401_HW_ALS100,
+ mpu_port[dev], 0,
+ mpu_irq[dev], IRQF_DISABLED,
+ NULL) < 0)
+ snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
+ }
+
+ if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_opl3_create(card,
+ fm_port[dev], fm_port[dev] + 2,
+ OPL3_HW_AUTO, 0, &opl3) < 0) {
+ snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx\n",
+ fm_port[dev], fm_port[dev] + 2);
+ } else {
+ if ((error = snd_opl3_timer_new(opl3, 0, 1)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ }
+ }
+
+ if ((error = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ return 0;
+}
+
+static unsigned int __devinitdata als100_devices;
+
+static int __devinit snd_als100_pnp_detect(struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ static int dev;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (!enable[dev])
+ continue;
+ res = snd_card_als100_probe(dev, card, id);
+ if (res < 0)
+ return res;
+ dev++;
+ als100_devices++;
+ return 0;
+ }
+ return -ENODEV;
+}
+
+static void __devexit snd_als100_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_als100_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_als100 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_sbmixer_suspend(chip);
+ return 0;
+}
+
+static int snd_als100_pnp_resume(struct pnp_card_link *pcard)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_als100 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_sbdsp_reset(chip);
+ snd_sbmixer_resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct pnp_card_driver als100_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "als100",
+ .id_table = snd_als100_pnpids,
+ .probe = snd_als100_pnp_detect,
+ .remove = __devexit_p(snd_als100_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_als100_pnp_suspend,
+ .resume = snd_als100_pnp_resume,
+#endif
+};
+
+static int __init alsa_card_als100_init(void)
+{
+ int err;
+
+ err = pnp_register_card_driver(&als100_pnpc_driver);
+ if (err)
+ return err;
+
+ if (!als100_devices) {
+ pnp_unregister_card_driver(&als100_pnpc_driver);
+#ifdef MODULE
+ snd_printk(KERN_ERR "no ALS100 based soundcards found\n");
+#endif
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_als100_exit(void)
+{
+ pnp_unregister_card_driver(&als100_pnpc_driver);
+}
+
+module_init(alsa_card_als100_init)
+module_exit(alsa_card_als100_exit)
diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c
new file mode 100644
index 0000000..3e74d1a
--- /dev/null
+++ b/sound/isa/azt2320.c
@@ -0,0 +1,354 @@
+/*
+ card-azt2320.c - driver for Aztech Systems AZT2320 based soundcards.
+ Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+/*
+ This driver should provide support for most Aztech AZT2320 based cards.
+ Several AZT2316 chips are also supported/tested, but autoprobe doesn't
+ work: all module option have to be set.
+
+ No docs available for us at Aztech headquarters !!! Unbelievable ...
+ No other help obtained.
+
+ Thanks to Rainer Wiesner <rainer.wiesner@01019freenet.de> for the WSS
+ activation method (full-duplex audio!).
+*/
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/wss.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+
+#define PFX "azt2320: "
+
+MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
+MODULE_DESCRIPTION("Aztech Systems AZT2320");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Aztech Systems,PRO16V},"
+ "{Aztech Systems,AZT2320},"
+ "{Aztech Systems,AZT3300},"
+ "{Aztech Systems,AZT2320},"
+ "{Aztech Systems,AZT3000}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* Pnp setup */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* Pnp setup */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for azt2320 based soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for azt2320 based soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable azt2320 based soundcard.");
+
+struct snd_card_azt2320 {
+ int dev_no;
+ struct pnp_dev *dev;
+ struct pnp_dev *devmpu;
+ struct snd_wss *chip;
+};
+
+static struct pnp_card_device_id snd_azt2320_pnpids[] = {
+ /* PRO16V */
+ { .id = "AZT1008", .devs = { { "AZT1008" }, { "AZT2001" }, } },
+ /* Aztech Sound Galaxy 16 */
+ { .id = "AZT2320", .devs = { { "AZT0001" }, { "AZT0002" }, } },
+ /* Packard Bell Sound III 336 AM/SP */
+ { .id = "AZT3000", .devs = { { "AZT1003" }, { "AZT2001" }, } },
+ /* AT3300 */
+ { .id = "AZT3002", .devs = { { "AZT1004" }, { "AZT2001" }, } },
+ /* --- */
+ { .id = "AZT3005", .devs = { { "AZT1003" }, { "AZT2001" }, } },
+ /* --- */
+ { .id = "AZT3011", .devs = { { "AZT1003" }, { "AZT2001" }, } },
+ { .id = "" } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_azt2320_pnpids);
+
+#define DRIVER_NAME "snd-card-azt2320"
+
+static int __devinit snd_card_azt2320_pnp(int dev, struct snd_card_azt2320 *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->dev == NULL)
+ return -ENODEV;
+
+ acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL);
+
+ pdev = acard->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n");
+ return err;
+ }
+ port[dev] = pnp_port_start(pdev, 0);
+ fm_port[dev] = pnp_port_start(pdev, 1);
+ wss_port[dev] = pnp_port_start(pdev, 2);
+ dma1[dev] = pnp_dma(pdev, 0);
+ dma2[dev] = pnp_dma(pdev, 1);
+ irq[dev] = pnp_irq(pdev, 0);
+
+ pdev = acard->devmpu;
+ if (pdev != NULL) {
+ err = pnp_activate_dev(pdev);
+ if (err < 0)
+ goto __mpu_error;
+ mpu_port[dev] = pnp_port_start(pdev, 0);
+ mpu_irq[dev] = pnp_irq(pdev, 0);
+ } else {
+ __mpu_error:
+ if (pdev) {
+ pnp_release_card_device(pdev);
+ snd_printk(KERN_ERR PFX "MPU401 pnp configure failure, skipping\n");
+ }
+ acard->devmpu = NULL;
+ mpu_port[dev] = -1;
+ }
+
+ return 0;
+}
+
+/* same of snd_sbdsp_command by Jaroslav Kysela */
+static int __devinit snd_card_azt2320_command(unsigned long port, unsigned char val)
+{
+ int i;
+ unsigned long limit;
+
+ limit = jiffies + HZ / 10;
+ for (i = 50000; i && time_after(limit, jiffies); i--)
+ if (!(inb(port + 0x0c) & 0x80)) {
+ outb(val, port + 0x0c);
+ return 0;
+ }
+ return -EBUSY;
+}
+
+static int __devinit snd_card_azt2320_enable_wss(unsigned long port)
+{
+ int error;
+
+ if ((error = snd_card_azt2320_command(port, 0x09)))
+ return error;
+ if ((error = snd_card_azt2320_command(port, 0x00)))
+ return error;
+
+ mdelay(5);
+ return 0;
+}
+
+static int __devinit snd_card_azt2320_probe(int dev,
+ struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ int error;
+ struct snd_card *card;
+ struct snd_card_azt2320 *acard;
+ struct snd_wss *chip;
+ struct snd_opl3 *opl3;
+
+ if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_azt2320))) == NULL)
+ return -ENOMEM;
+ acard = (struct snd_card_azt2320 *)card->private_data;
+
+ if ((error = snd_card_azt2320_pnp(dev, acard, pcard, pid))) {
+ snd_card_free(card);
+ return error;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+
+ if ((error = snd_card_azt2320_enable_wss(port[dev]))) {
+ snd_card_free(card);
+ return error;
+ }
+
+ error = snd_wss_create(card, wss_port[dev], -1,
+ irq[dev],
+ dma1[dev], dma2[dev],
+ WSS_HW_DETECT, 0, &chip);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ strcpy(card->driver, "AZT2320");
+ strcpy(card->shortname, "Aztech AZT2320");
+ sprintf(card->longname, "%s, WSS at 0x%lx, irq %i, dma %i&%i",
+ card->shortname, chip->port, irq[dev], dma1[dev], dma2[dev]);
+
+ error = snd_wss_pcm(chip, 0, NULL);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ error = snd_wss_mixer(chip);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ error = snd_wss_timer(chip, 0, NULL);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_mpu401_uart_new(card, 0, MPU401_HW_AZT2320,
+ mpu_port[dev], 0,
+ mpu_irq[dev], IRQF_DISABLED,
+ NULL) < 0)
+ snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
+ }
+
+ if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_opl3_create(card,
+ fm_port[dev], fm_port[dev] + 2,
+ OPL3_HW_AUTO, 0, &opl3) < 0) {
+ snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx\n",
+ fm_port[dev], fm_port[dev] + 2);
+ } else {
+ if ((error = snd_opl3_timer_new(opl3, 1, 2)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ }
+ }
+
+ if ((error = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ return 0;
+}
+
+static unsigned int __devinitdata azt2320_devices;
+
+static int __devinit snd_azt2320_pnp_detect(struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ static int dev;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (!enable[dev])
+ continue;
+ res = snd_card_azt2320_probe(dev, card, id);
+ if (res < 0)
+ return res;
+ dev++;
+ azt2320_devices++;
+ return 0;
+ }
+ return -ENODEV;
+}
+
+static void __devexit snd_azt2320_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_azt2320_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_azt2320 *acard = card->private_data;
+ struct snd_wss *chip = acard->chip;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ chip->suspend(chip);
+ return 0;
+}
+
+static int snd_azt2320_pnp_resume(struct pnp_card_link *pcard)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_azt2320 *acard = card->private_data;
+ struct snd_wss *chip = acard->chip;
+
+ chip->resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct pnp_card_driver azt2320_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "azt2320",
+ .id_table = snd_azt2320_pnpids,
+ .probe = snd_azt2320_pnp_detect,
+ .remove = __devexit_p(snd_azt2320_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_azt2320_pnp_suspend,
+ .resume = snd_azt2320_pnp_resume,
+#endif
+};
+
+static int __init alsa_card_azt2320_init(void)
+{
+ int err;
+
+ err = pnp_register_card_driver(&azt2320_pnpc_driver);
+ if (err)
+ return err;
+
+ if (!azt2320_devices) {
+ pnp_unregister_card_driver(&azt2320_pnpc_driver);
+#ifdef MODULE
+ snd_printk(KERN_ERR "no AZT2320 based soundcards found\n");
+#endif
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_azt2320_exit(void)
+{
+ pnp_unregister_card_driver(&azt2320_pnpc_driver);
+}
+
+module_init(alsa_card_azt2320_init)
+module_exit(alsa_card_azt2320_exit)
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
new file mode 100644
index 0000000..e49aec7
--- /dev/null
+++ b/sound/isa/cmi8330.c
@@ -0,0 +1,711 @@
+/*
+ * Driver for C-Media's CMI8330 soundcards.
+ * Copyright (c) by George Talusan <gstalusan@uwaterloo.ca>
+ * http://www.undergrad.math.uwaterloo.ca/~gstalusa
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * NOTES
+ *
+ * The extended registers contain mixer settings which are largely
+ * untapped for the time being.
+ *
+ * MPU401 and SPDIF are not supported yet. I don't have the hardware
+ * to aid in coding and testing, so I won't bother.
+ *
+ * To quickly load the module,
+ *
+ * modprobe -a snd-cmi8330 sbport=0x220 sbirq=5 sbdma8=1
+ * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0
+ *
+ * This card has two mixers and two PCM devices. I've cheesed it such
+ * that recording and playback can be done through the same device.
+ * The driver "magically" routes the capturing to the AD1848 codec,
+ * and playback to the SB16 codec. This allows for full-duplex mode
+ * to some extent.
+ * The utilities in alsa-utils are aware of both devices, so passing
+ * the appropriate parameters to amixer and alsactl will give you
+ * full control over both mixers.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/slab.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/sb.h>
+#include <sound/initval.h>
+
+/*
+ */
+/* #define ENABLE_SB_MIXER */
+#define PLAYBACK_ON_SB
+
+/*
+ */
+MODULE_AUTHOR("George Talusan <gstalusan@uwaterloo.ca>");
+MODULE_DESCRIPTION("C-Media CMI8330");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8330,isapnp:{CMI0001,@@@0001,@X@0001}}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP;
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long sbport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int sbirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
+static int sbdma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
+static int sbdma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
+static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
+static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for CMI8330 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable CMI8330 soundcard.");
+#ifdef CONFIG_PNP
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard.");
+#endif
+
+module_param_array(sbport, long, NULL, 0444);
+MODULE_PARM_DESC(sbport, "Port # for CMI8330 SB driver.");
+module_param_array(sbirq, int, NULL, 0444);
+MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330 SB driver.");
+module_param_array(sbdma8, int, NULL, 0444);
+MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330 SB driver.");
+module_param_array(sbdma16, int, NULL, 0444);
+MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330 SB driver.");
+
+module_param_array(wssport, long, NULL, 0444);
+MODULE_PARM_DESC(wssport, "Port # for CMI8330 WSS driver.");
+module_param_array(wssirq, int, NULL, 0444);
+MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver.");
+module_param_array(wssdma, int, NULL, 0444);
+MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver.");
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnp_registered;
+#endif
+
+#define CMI8330_RMUX3D 16
+#define CMI8330_MUTEMUX 17
+#define CMI8330_OUTPUTVOL 18
+#define CMI8330_MASTVOL 19
+#define CMI8330_LINVOL 20
+#define CMI8330_CDINVOL 21
+#define CMI8330_WAVVOL 22
+#define CMI8330_RECMUX 23
+#define CMI8330_WAVGAIN 24
+#define CMI8330_LINGAIN 25
+#define CMI8330_CDINGAIN 26
+
+static unsigned char snd_cmi8330_image[((CMI8330_CDINGAIN)-16) + 1] =
+{
+ 0x40, /* 16 - recording mux (SB-mixer-enabled) */
+#ifdef ENABLE_SB_MIXER
+ 0x40, /* 17 - mute mux (Mode2) */
+#else
+ 0x0, /* 17 - mute mux */
+#endif
+ 0x0, /* 18 - vol */
+ 0x0, /* 19 - master volume */
+ 0x0, /* 20 - line-in volume */
+ 0x0, /* 21 - cd-in volume */
+ 0x0, /* 22 - wave volume */
+ 0x0, /* 23 - mute/rec mux */
+ 0x0, /* 24 - wave rec gain */
+ 0x0, /* 25 - line-in rec gain */
+ 0x0 /* 26 - cd-in rec gain */
+};
+
+typedef int (*snd_pcm_open_callback_t)(struct snd_pcm_substream *);
+
+struct snd_cmi8330 {
+#ifdef CONFIG_PNP
+ struct pnp_dev *cap;
+ struct pnp_dev *play;
+#endif
+ struct snd_card *card;
+ struct snd_wss *wss;
+ struct snd_sb *sb;
+
+ struct snd_pcm *pcm;
+ struct snd_cmi8330_stream {
+ struct snd_pcm_ops ops;
+ snd_pcm_open_callback_t open;
+ void *private_data; /* sb or wss */
+ } streams[2];
+};
+
+#ifdef CONFIG_PNP
+
+static struct pnp_card_device_id snd_cmi8330_pnpids[] = {
+ { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" } } },
+ { .id = "" }
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_cmi8330_pnpids);
+
+#endif
+
+
+static struct snd_kcontrol_new snd_cmi8330_controls[] __devinitdata = {
+WSS_DOUBLE("Master Playback Volume", 0,
+ CMI8330_MASTVOL, CMI8330_MASTVOL, 4, 0, 15, 0),
+WSS_SINGLE("Loud Playback Switch", 0,
+ CMI8330_MUTEMUX, 6, 1, 1),
+WSS_DOUBLE("PCM Playback Switch", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
+WSS_DOUBLE("PCM Playback Volume", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1),
+WSS_DOUBLE("Line Playback Switch", 0,
+ CMI8330_MUTEMUX, CMI8330_MUTEMUX, 4, 3, 1, 0),
+WSS_DOUBLE("Line Playback Volume", 0,
+ CMI8330_LINVOL, CMI8330_LINVOL, 4, 0, 15, 0),
+WSS_DOUBLE("Line Capture Switch", 0,
+ CMI8330_RMUX3D, CMI8330_RMUX3D, 2, 1, 1, 0),
+WSS_DOUBLE("Line Capture Volume", 0,
+ CMI8330_LINGAIN, CMI8330_LINGAIN, 4, 0, 15, 0),
+WSS_DOUBLE("CD Playback Switch", 0,
+ CMI8330_MUTEMUX, CMI8330_MUTEMUX, 2, 1, 1, 0),
+WSS_DOUBLE("CD Capture Switch", 0,
+ CMI8330_RMUX3D, CMI8330_RMUX3D, 4, 3, 1, 0),
+WSS_DOUBLE("CD Playback Volume", 0,
+ CMI8330_CDINVOL, CMI8330_CDINVOL, 4, 0, 15, 0),
+WSS_DOUBLE("CD Capture Volume", 0,
+ CMI8330_CDINGAIN, CMI8330_CDINGAIN, 4, 0, 15, 0),
+WSS_SINGLE("Mic Playback Switch", 0,
+ CMI8330_MUTEMUX, 0, 1, 0),
+WSS_SINGLE("Mic Playback Volume", 0,
+ CMI8330_OUTPUTVOL, 0, 7, 0),
+WSS_SINGLE("Mic Capture Switch", 0,
+ CMI8330_RMUX3D, 0, 1, 0),
+WSS_SINGLE("Mic Capture Volume", 0,
+ CMI8330_OUTPUTVOL, 5, 7, 0),
+WSS_DOUBLE("Wavetable Playback Switch", 0,
+ CMI8330_RECMUX, CMI8330_RECMUX, 1, 0, 1, 0),
+WSS_DOUBLE("Wavetable Playback Volume", 0,
+ CMI8330_WAVVOL, CMI8330_WAVVOL, 4, 0, 15, 0),
+WSS_DOUBLE("Wavetable Capture Switch", 0,
+ CMI8330_RECMUX, CMI8330_RECMUX, 5, 4, 1, 0),
+WSS_DOUBLE("Wavetable Capture Volume", 0,
+ CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0),
+WSS_SINGLE("3D Control - Switch", 0,
+ CMI8330_RMUX3D, 5, 1, 1),
+WSS_SINGLE("PC Speaker Playback Volume", 0,
+ CMI8330_OUTPUTVOL, 3, 3, 0),
+WSS_SINGLE("FM Playback Switch", 0,
+ CMI8330_RECMUX, 3, 1, 1),
+WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0,
+ CMI8330_RMUX3D, 7, 1, 1),
+WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0,
+ CMI8330_MUTEMUX, 7, 1, 1),
+};
+
+#ifdef ENABLE_SB_MIXER
+static struct sbmix_elem cmi8330_sb_mixers[] __devinitdata = {
+SB_DOUBLE("SB Master Playback Volume", SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31),
+SB_DOUBLE("Tone Control - Bass", SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15),
+SB_DOUBLE("Tone Control - Treble", SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15),
+SB_DOUBLE("SB PCM Playback Volume", SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31),
+SB_DOUBLE("SB Synth Playback Volume", SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31),
+SB_DOUBLE("SB CD Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1),
+SB_DOUBLE("SB CD Playback Volume", SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31),
+SB_DOUBLE("SB Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1),
+SB_DOUBLE("SB Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31),
+SB_SINGLE("SB Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1),
+SB_SINGLE("SB Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31),
+SB_SINGLE("SB PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
+SB_DOUBLE("SB Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3),
+SB_DOUBLE("SB Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3),
+SB_SINGLE("SB Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1),
+};
+
+static unsigned char cmi8330_sb_init_values[][2] __devinitdata = {
+ { SB_DSP4_MASTER_DEV + 0, 0 },
+ { SB_DSP4_MASTER_DEV + 1, 0 },
+ { SB_DSP4_PCM_DEV + 0, 0 },
+ { SB_DSP4_PCM_DEV + 1, 0 },
+ { SB_DSP4_SYNTH_DEV + 0, 0 },
+ { SB_DSP4_SYNTH_DEV + 1, 0 },
+ { SB_DSP4_INPUT_LEFT, 0 },
+ { SB_DSP4_INPUT_RIGHT, 0 },
+ { SB_DSP4_OUTPUT_SW, 0 },
+ { SB_DSP4_SPEAKER_DEV, 0 },
+};
+
+
+static int __devinit cmi8330_add_sb_mixers(struct snd_sb *chip)
+{
+ int idx, err;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ snd_sbmixer_write(chip, 0x00, 0x00); /* mixer reset */
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+
+ /* mute and zero volume channels */
+ for (idx = 0; idx < ARRAY_SIZE(cmi8330_sb_init_values); idx++) {
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ snd_sbmixer_write(chip, cmi8330_sb_init_values[idx][0],
+ cmi8330_sb_init_values[idx][1]);
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+ }
+
+ for (idx = 0; idx < ARRAY_SIZE(cmi8330_sb_mixers); idx++) {
+ if ((err = snd_sbmixer_add_ctl_elem(chip, &cmi8330_sb_mixers[idx])) < 0)
+ return err;
+ }
+ return 0;
+}
+#endif
+
+static int __devinit snd_cmi8330_mixer(struct snd_card *card, struct snd_cmi8330 *acard)
+{
+ unsigned int idx;
+ int err;
+
+ strcpy(card->mixername, "CMI8330/C3D");
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_cmi8330_controls); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_cmi8330_controls[idx],
+ acard->wss));
+ if (err < 0)
+ return err;
+ }
+
+#ifdef ENABLE_SB_MIXER
+ if ((err = cmi8330_add_sb_mixers(acard->sb)) < 0)
+ return err;
+#endif
+ return 0;
+}
+
+#ifdef CONFIG_PNP
+static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->cap == NULL)
+ return -EBUSY;
+
+ acard->play = pnp_request_card_device(card, id->devs[1].id, NULL);
+ if (acard->play == NULL)
+ return -EBUSY;
+
+ pdev = acard->cap;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n");
+ return -EBUSY;
+ }
+ wssport[dev] = pnp_port_start(pdev, 0);
+ wssdma[dev] = pnp_dma(pdev, 0);
+ wssirq[dev] = pnp_irq(pdev, 0);
+
+ /* allocate SB16 resources */
+ pdev = acard->play;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP configure failure\n");
+ return -EBUSY;
+ }
+ sbport[dev] = pnp_port_start(pdev, 0);
+ sbdma8[dev] = pnp_dma(pdev, 0);
+ sbdma16[dev] = pnp_dma(pdev, 1);
+ sbirq[dev] = pnp_irq(pdev, 0);
+
+ return 0;
+}
+#endif
+
+/*
+ * PCM interface
+ *
+ * since we call the different chip interfaces for playback and capture
+ * directions, we need a trick.
+ *
+ * - copy the ops for each direction into a local record.
+ * - replace the open callback with the new one, which replaces the
+ * substream->private_data with the corresponding chip instance
+ * and calls again the original open callback of the chip.
+ *
+ */
+
+#ifdef PLAYBACK_ON_SB
+#define CMI_SB_STREAM SNDRV_PCM_STREAM_PLAYBACK
+#define CMI_AD_STREAM SNDRV_PCM_STREAM_CAPTURE
+#else
+#define CMI_SB_STREAM SNDRV_PCM_STREAM_CAPTURE
+#define CMI_AD_STREAM SNDRV_PCM_STREAM_PLAYBACK
+#endif
+
+static int snd_cmi8330_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_cmi8330 *chip = snd_pcm_substream_chip(substream);
+
+ /* replace the private_data and call the original open callback */
+ substream->private_data = chip->streams[SNDRV_PCM_STREAM_PLAYBACK].private_data;
+ return chip->streams[SNDRV_PCM_STREAM_PLAYBACK].open(substream);
+}
+
+static int snd_cmi8330_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_cmi8330 *chip = snd_pcm_substream_chip(substream);
+
+ /* replace the private_data and call the original open callback */
+ substream->private_data = chip->streams[SNDRV_PCM_STREAM_CAPTURE].private_data;
+ return chip->streams[SNDRV_PCM_STREAM_CAPTURE].open(substream);
+}
+
+static int __devinit snd_cmi8330_pcm(struct snd_card *card, struct snd_cmi8330 *chip)
+{
+ struct snd_pcm *pcm;
+ const struct snd_pcm_ops *ops;
+ int err;
+ static snd_pcm_open_callback_t cmi_open_callbacks[2] = {
+ snd_cmi8330_playback_open,
+ snd_cmi8330_capture_open
+ };
+
+ if ((err = snd_pcm_new(card, "CMI8330", 0, 1, 1, &pcm)) < 0)
+ return err;
+ strcpy(pcm->name, "CMI8330");
+ pcm->private_data = chip;
+
+ /* SB16 */
+ ops = snd_sb16dsp_get_pcm_ops(CMI_SB_STREAM);
+ chip->streams[CMI_SB_STREAM].ops = *ops;
+ chip->streams[CMI_SB_STREAM].open = ops->open;
+ chip->streams[CMI_SB_STREAM].ops.open = cmi_open_callbacks[CMI_SB_STREAM];
+ chip->streams[CMI_SB_STREAM].private_data = chip->sb;
+
+ /* AD1848 */
+ ops = snd_wss_get_pcm_ops(CMI_AD_STREAM);
+ chip->streams[CMI_AD_STREAM].ops = *ops;
+ chip->streams[CMI_AD_STREAM].open = ops->open;
+ chip->streams[CMI_AD_STREAM].ops.open = cmi_open_callbacks[CMI_AD_STREAM];
+ chip->streams[CMI_AD_STREAM].private_data = chip->wss;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &chip->streams[SNDRV_PCM_STREAM_PLAYBACK].ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &chip->streams[SNDRV_PCM_STREAM_CAPTURE].ops);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, 128*1024);
+ chip->pcm = pcm;
+
+ return 0;
+}
+
+
+#ifdef CONFIG_PM
+static int snd_cmi8330_suspend(struct snd_card *card)
+{
+ struct snd_cmi8330 *acard = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(acard->pcm);
+ acard->wss->suspend(acard->wss);
+ snd_sbmixer_suspend(acard->sb);
+ return 0;
+}
+
+static int snd_cmi8330_resume(struct snd_card *card)
+{
+ struct snd_cmi8330 *acard = card->private_data;
+
+ snd_sbdsp_reset(acard->sb);
+ snd_sbmixer_suspend(acard->sb);
+ acard->wss->resume(acard->wss);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+
+/*
+ */
+
+#ifdef CONFIG_PNP
+#define is_isapnp_selected(dev) isapnp[dev]
+#else
+#define is_isapnp_selected(dev) 0
+#endif
+
+#define PFX "cmi8330: "
+
+static struct snd_card *snd_cmi8330_card_new(int dev)
+{
+ struct snd_card *card;
+ struct snd_cmi8330 *acard;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_cmi8330));
+ if (card == NULL) {
+ snd_printk(KERN_ERR PFX "could not get a new card\n");
+ return NULL;
+ }
+ acard = card->private_data;
+ acard->card = card;
+ return card;
+}
+
+static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
+{
+ struct snd_cmi8330 *acard;
+ int i, err;
+
+ acard = card->private_data;
+ err = snd_wss_create(card, wssport[dev] + 4, -1,
+ wssirq[dev],
+ wssdma[dev], -1,
+ WSS_HW_DETECT, 0, &acard->wss);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "(AD1848) device busy??\n");
+ return err;
+ }
+ if (acard->wss->hardware != WSS_HW_CMI8330) {
+ snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n");
+ return -ENODEV;
+ }
+
+ if ((err = snd_sbdsp_create(card, sbport[dev],
+ sbirq[dev],
+ snd_sb16dsp_interrupt,
+ sbdma8[dev],
+ sbdma16[dev],
+ SB_HW_AUTO, &acard->sb)) < 0) {
+ snd_printk(KERN_ERR PFX "(SB16) device busy??\n");
+ return err;
+ }
+ if (acard->sb->hardware != SB_HW_16) {
+ snd_printk(KERN_ERR PFX "(SB16) not found during probe\n");
+ return err;
+ }
+
+ snd_wss_out(acard->wss, CS4231_MISC_INFO, 0x40); /* switch on MODE2 */
+ for (i = CMI8330_RMUX3D; i <= CMI8330_CDINGAIN; i++)
+ snd_wss_out(acard->wss, i,
+ snd_cmi8330_image[i - CMI8330_RMUX3D]);
+
+ if ((err = snd_cmi8330_mixer(card, acard)) < 0) {
+ snd_printk(KERN_ERR PFX "failed to create mixers\n");
+ return err;
+ }
+
+ if ((err = snd_cmi8330_pcm(card, acard)) < 0) {
+ snd_printk(KERN_ERR PFX "failed to create pcms\n");
+ return err;
+ }
+
+ strcpy(card->driver, "CMI8330/C3D");
+ strcpy(card->shortname, "C-Media CMI8330/C3D");
+ sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
+ card->shortname,
+ acard->wss->port,
+ wssirq[dev],
+ wssdma[dev]);
+
+ return snd_card_register(card);
+}
+
+static int __devinit snd_cmi8330_isa_match(struct device *pdev,
+ unsigned int dev)
+{
+ if (!enable[dev] || is_isapnp_selected(dev))
+ return 0;
+ if (wssport[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify wssport\n");
+ return 0;
+ }
+ if (sbport[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify sbport\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_cmi8330_isa_probe(struct device *pdev,
+ unsigned int dev)
+{
+ struct snd_card *card;
+ int err;
+
+ card = snd_cmi8330_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ snd_card_set_dev(card, pdev);
+ if ((err = snd_cmi8330_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ dev_set_drvdata(pdev, card);
+ return 0;
+}
+
+static int __devexit snd_cmi8330_isa_remove(struct device *devptr,
+ unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_cmi8330_isa_suspend(struct device *dev, unsigned int n,
+ pm_message_t state)
+{
+ return snd_cmi8330_suspend(dev_get_drvdata(dev));
+}
+
+static int snd_cmi8330_isa_resume(struct device *dev, unsigned int n)
+{
+ return snd_cmi8330_resume(dev_get_drvdata(dev));
+}
+#endif
+
+#define DEV_NAME "cmi8330"
+
+static struct isa_driver snd_cmi8330_driver = {
+ .match = snd_cmi8330_isa_match,
+ .probe = snd_cmi8330_isa_probe,
+ .remove = __devexit_p(snd_cmi8330_isa_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cmi8330_isa_suspend,
+ .resume = snd_cmi8330_isa_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+
+#ifdef CONFIG_PNP
+static int __devinit snd_cmi8330_pnp_detect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ static int dev;
+ struct snd_card *card;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_cmi8330_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ if ((res = snd_cmi8330_pnp(dev, card->private_data, pcard, pid)) < 0) {
+ snd_printk(KERN_ERR PFX "PnP detection failed\n");
+ snd_card_free(card);
+ return res;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((res = snd_cmi8330_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return res;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_cmi8330_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_cmi8330_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ return snd_cmi8330_suspend(pnp_get_card_drvdata(pcard));
+}
+
+static int snd_cmi8330_pnp_resume(struct pnp_card_link *pcard)
+{
+ return snd_cmi8330_resume(pnp_get_card_drvdata(pcard));
+}
+#endif
+
+static struct pnp_card_driver cmi8330_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "cmi8330",
+ .id_table = snd_cmi8330_pnpids,
+ .probe = snd_cmi8330_pnp_detect,
+ .remove = __devexit_p(snd_cmi8330_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cmi8330_pnp_suspend,
+ .resume = snd_cmi8330_pnp_resume,
+#endif
+};
+#endif /* CONFIG_PNP */
+
+static int __init alsa_card_cmi8330_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&snd_cmi8330_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+
+ err = pnp_register_card_driver(&cmi8330_pnpc_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ if (isa_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit alsa_card_cmi8330_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnp_registered)
+ pnp_unregister_card_driver(&cmi8330_pnpc_driver);
+
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&snd_cmi8330_driver);
+}
+
+module_init(alsa_card_cmi8330_init)
+module_exit(alsa_card_cmi8330_exit)
diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile
new file mode 100644
index 0000000..5870ca2
--- /dev/null
+++ b/sound/isa/cs423x/Makefile
@@ -0,0 +1,15 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-cs4236-lib-objs := cs4236_lib.o
+snd-cs4231-objs := cs4231.o
+snd-cs4232-objs := cs4232.o
+snd-cs4236-objs := cs4236.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_CS4231) += snd-cs4231.o
+obj-$(CONFIG_SND_CS4232) += snd-cs4232.o
+obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o
+
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
new file mode 100644
index 0000000..f019d44
--- /dev/null
+++ b/sound/isa/cs423x/cs4231.c
@@ -0,0 +1,205 @@
+/*
+ * Generic driver for CS4231 chips
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Originally the CS4232/CS4232A driver, modified for use on CS4231 by
+ * Tugrul Galatali <galatalt@stuy.edu>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/mpu401.h>
+#include <sound/initval.h>
+
+#define CRD_NAME "Generic CS4231"
+#define DEV_NAME "cs4231"
+
+MODULE_DESCRIPTION(CRD_NAME);
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4231}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,11,12,15 */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 9,11,12,15 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver.");
+module_param_array(mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for " CRD_NAME " driver.");
+
+static int __devinit snd_cs4231_match(struct device *dev, unsigned int n)
+{
+ if (!enable[n])
+ return 0;
+
+ if (port[n] == SNDRV_AUTO_PORT) {
+ dev_err(dev, "please specify port\n");
+ return 0;
+ }
+ if (irq[n] == SNDRV_AUTO_IRQ) {
+ dev_err(dev, "please specify irq\n");
+ return 0;
+ }
+ if (dma1[n] == SNDRV_AUTO_DMA) {
+ dev_err(dev, "please specify dma1\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n)
+{
+ struct snd_card *card;
+ struct snd_wss *chip;
+ struct snd_pcm *pcm;
+ int error;
+
+ card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
+ if (!card)
+ return -EINVAL;
+
+ error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], dma2[n],
+ WSS_HW_DETECT, 0, &chip);
+ if (error < 0)
+ goto out;
+
+ card->private_data = chip;
+
+ error = snd_wss_pcm(chip, 0, &pcm);
+ if (error < 0)
+ goto out;
+
+ strcpy(card->driver, "CS4231");
+ strcpy(card->shortname, pcm->name);
+
+ sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
+ pcm->name, chip->port, irq[n], dma1[n]);
+ if (dma2[n] >= 0)
+ sprintf(card->longname + strlen(card->longname), "&%d", dma2[n]);
+
+ error = snd_wss_mixer(chip);
+ if (error < 0)
+ goto out;
+
+ error = snd_wss_timer(chip, 0, NULL);
+ if (error < 0)
+ goto out;
+
+ if (mpu_port[n] > 0 && mpu_port[n] != SNDRV_AUTO_PORT) {
+ if (mpu_irq[n] == SNDRV_AUTO_IRQ)
+ mpu_irq[n] = -1;
+ if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232,
+ mpu_port[n], 0, mpu_irq[n],
+ mpu_irq[n] >= 0 ? IRQF_DISABLED : 0,
+ NULL) < 0)
+ dev_warn(dev, "MPU401 not detected\n");
+ }
+
+ snd_card_set_dev(card, dev);
+
+ error = snd_card_register(card);
+ if (error < 0)
+ goto out;
+
+ dev_set_drvdata(dev, card);
+ return 0;
+
+out: snd_card_free(card);
+ return error;
+}
+
+static int __devexit snd_cs4231_remove(struct device *dev, unsigned int n)
+{
+ snd_card_free(dev_get_drvdata(dev));
+ dev_set_drvdata(dev, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_cs4231_suspend(struct device *dev, unsigned int n, pm_message_t state)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_wss *chip = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ chip->suspend(chip);
+ return 0;
+}
+
+static int snd_cs4231_resume(struct device *dev, unsigned int n)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_wss *chip = card->private_data;
+
+ chip->resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct isa_driver snd_cs4231_driver = {
+ .match = snd_cs4231_match,
+ .probe = snd_cs4231_probe,
+ .remove = __devexit_p(snd_cs4231_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cs4231_suspend,
+ .resume = snd_cs4231_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ }
+};
+
+static int __init alsa_card_cs4231_init(void)
+{
+ return isa_register_driver(&snd_cs4231_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_cs4231_exit(void)
+{
+ isa_unregister_driver(&snd_cs4231_driver);
+}
+
+module_init(alsa_card_cs4231_init);
+module_exit(alsa_card_cs4231_exit);
diff --git a/sound/isa/cs423x/cs4232.c b/sound/isa/cs423x/cs4232.c
new file mode 100644
index 0000000..9fad2e6
--- /dev/null
+++ b/sound/isa/cs423x/cs4232.c
@@ -0,0 +1,2 @@
+#define CS4232
+#include "cs4236.c"
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
new file mode 100644
index 0000000..019c940
--- /dev/null
+++ b/sound/isa/cs423x/cs4236.c
@@ -0,0 +1,750 @@
+/*
+ * Driver for generic CS4232/CS4235/CS4236/CS4236B/CS4237B/CS4238B/CS4239 chips
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/slab.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+#ifdef CS4232
+MODULE_DESCRIPTION("Cirrus Logic CS4232");
+MODULE_SUPPORTED_DEVICE("{{Turtle Beach,TBS-2000},"
+ "{Turtle Beach,Tropez Plus},"
+ "{SIC CrystalWave 32},"
+ "{Hewlett Packard,Omnibook 5500},"
+ "{TerraTec,Maestro 32/96},"
+ "{Philips,PCA70PS}}");
+#else
+MODULE_DESCRIPTION("Cirrus Logic CS4235-9");
+MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235},"
+ "{Crystal Semiconductors,CS4236},"
+ "{Crystal Semiconductors,CS4237},"
+ "{Crystal Semiconductors,CS4238},"
+ "{Crystal Semiconductors,CS4239},"
+ "{Acer,AW37},"
+ "{Acer,AW35/Pro},"
+ "{Crystal,3D},"
+ "{Crystal Computer,TidalWave128},"
+ "{Dell,Optiplex GX1},"
+ "{Dell,Workstation 400 sound},"
+ "{EliteGroup,P5TX-LA sound},"
+ "{Gallant,SC-70P},"
+ "{Gateway,E1000 Onboard CS4236B},"
+ "{Genius,Sound Maker 3DJ},"
+ "{Hewlett Packard,HP6330 sound},"
+ "{IBM,PC 300PL sound},"
+ "{IBM,Aptiva 2137 E24},"
+ "{IBM,IntelliStation M Pro},"
+ "{Intel,Marlin Spike Mobo CS4235},"
+ "{Intel PR440FX Onboard},"
+ "{Guillemot,MaxiSound 16 PnP},"
+ "{NewClear,3D},"
+ "{TerraTec,AudioSystem EWS64L/XL},"
+ "{Typhoon Soundsystem,CS4236B},"
+ "{Turtle Beach,Malibu},"
+ "{Unknown,Digital PC 5000 Onboard}}");
+#endif
+
+#ifdef CS4232
+#define IDENT "CS4232"
+#define DEV_NAME "cs4232"
+#else
+#define IDENT "CS4236+"
+#define DEV_NAME "cs4236"
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long cport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* PnP setup */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long sb_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,11,12,15 */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 9,11,12,15 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " IDENT " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " IDENT " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " IDENT " soundcard.");
+#ifdef CONFIG_PNP
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard.");
+#endif
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for " IDENT " driver.");
+module_param_array(cport, long, NULL, 0444);
+MODULE_PARM_DESC(cport, "Control port # for " IDENT " driver.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " IDENT " driver.");
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM port # for " IDENT " driver.");
+module_param_array(sb_port, long, NULL, 0444);
+MODULE_PARM_DESC(sb_port, "SB port # for " IDENT " driver (optional).");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for " IDENT " driver.");
+module_param_array(mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " IDENT " driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for " IDENT " driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver.");
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnpc_registered;
+#ifdef CS4232
+static int pnp_registered;
+#endif
+#endif /* CONFIG_PNP */
+
+struct snd_card_cs4236 {
+ struct snd_wss *chip;
+ struct resource *res_sb_port;
+#ifdef CONFIG_PNP
+ struct pnp_dev *wss;
+ struct pnp_dev *ctrl;
+ struct pnp_dev *mpu;
+#endif
+};
+
+#ifdef CONFIG_PNP
+
+#ifdef CS4232
+/*
+ * PNP BIOS
+ */
+static const struct pnp_device_id snd_cs4232_pnpbiosids[] = {
+ { .id = "CSC0100" },
+ { .id = "CSC0000" },
+ /* Guillemot Turtlebeach something appears to be cs4232 compatible
+ * (untested) */
+ { .id = "GIM0100" },
+ { .id = "" }
+};
+MODULE_DEVICE_TABLE(pnp, snd_cs4232_pnpbiosids);
+#endif /* CS4232 */
+
+#ifdef CS4232
+#define CS423X_ISAPNP_DRIVER "cs4232_isapnp"
+static struct pnp_card_device_id snd_cs423x_pnpids[] = {
+ /* Philips PCA70PS */
+ { .id = "CSC0d32", .devs = { { "CSC0000" }, { "CSC0010" }, { "PNPb006" } } },
+ /* TerraTec Maestro 32/96 (CS4232) */
+ { .id = "CSC1a32", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* HP Omnibook 5500 onboard */
+ { .id = "CSC4232", .devs = { { "CSC0000" }, { "CSC0002" }, { "CSC0003" } } },
+ /* Unnamed CS4236 card (Made in Taiwan) */
+ { .id = "CSC4236", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Turtle Beach TBS-2000 (CS4232) */
+ { .id = "CSC7532", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSCb006" } } },
+ /* Turtle Beach Tropez Plus (CS4232) */
+ { .id = "CSC7632", .devs = { { "CSC0000" }, { "CSC0010" }, { "PNPb006" } } },
+ /* SIC CrystalWave 32 (CS4232) */
+ { .id = "CSCf032", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Netfinity 3000 on-board soundcard */
+ { .id = "CSCe825", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC010f" } } },
+ /* --- */
+ { .id = "" } /* end */
+};
+#else /* CS4236 */
+#define CS423X_ISAPNP_DRIVER "cs4236_isapnp"
+static struct pnp_card_device_id snd_cs423x_pnpids[] = {
+ /* Intel Marlin Spike Motherboard - CS4235 */
+ { .id = "CSC0225", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Intel Marlin Spike Motherboard (#2) - CS4235 */
+ { .id = "CSC0225", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC0103" } } },
+ /* Unknown Intel mainboard - CS4235 */
+ { .id = "CSC0225", .devs = { { "CSC0100" }, { "CSC0110" } } },
+ /* Genius Sound Maker 3DJ - CS4237B */
+ { .id = "CSC0437", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Digital PC 5000 Onboard - CS4236B */
+ { .id = "CSC0735", .devs = { { "CSC0000" }, { "CSC0010" } } },
+ /* some uknown CS4236B */
+ { .id = "CSC0b35", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Intel PR440FX Onboard sound */
+ { .id = "CSC0b36", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* CS4235 on mainboard without MPU */
+ { .id = "CSC1425", .devs = { { "CSC0100" }, { "CSC0110" } } },
+ /* Gateway E1000 Onboard CS4236B */
+ { .id = "CSC1335", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* HP 6330 Onboard sound */
+ { .id = "CSC1525", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC0103" } } },
+ /* Crystal Computer TidalWave128 */
+ { .id = "CSC1e37", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* ACER AW37 - CS4235 */
+ { .id = "CSC4236", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* build-in soundcard in EliteGroup P5TX-LA motherboard - CS4237B */
+ { .id = "CSC4237", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Crystal 3D - CS4237B */
+ { .id = "CSC4336", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Typhoon Soundsystem PnP - CS4236B */
+ { .id = "CSC4536", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Crystal CX4235-XQ3 EP - CS4235 */
+ { .id = "CSC4625", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC0103" } } },
+ /* Crystal Semiconductors CS4237B */
+ { .id = "CSC4637", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* NewClear 3D - CX4237B-XQ3 */
+ { .id = "CSC4837", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Dell Optiplex GX1 - CS4236B */
+ { .id = "CSC6835", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Dell P410 motherboard - CS4236B */
+ { .id = "CSC6835", .devs = { { "CSC0000" }, { "CSC0010" } } },
+ /* Dell Workstation 400 Onboard - CS4236B */
+ { .id = "CSC6836", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Turtle Beach Malibu - CS4237B */
+ { .id = "CSC7537", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* CS4235 - onboard */
+ { .id = "CSC8025", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC0103" } } },
+ /* IBM Aptiva 2137 E24 Onboard - CS4237B */
+ { .id = "CSC8037", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* IBM IntelliStation M Pro motherboard */
+ { .id = "CSCc835", .devs = { { "CSC0000" }, { "CSC0010" } } },
+ /* Guillemot MaxiSound 16 PnP - CS4236B */
+ { .id = "CSC9836", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Gallant SC-70P */
+ { .id = "CSC9837", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* Techmakers MF-4236PW */
+ { .id = "CSCa736", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* TerraTec AudioSystem EWS64XL - CS4236B */
+ { .id = "CSCa836", .devs = { { "CSCa800" }, { "CSCa810" }, { "CSCa803" } } },
+ /* TerraTec AudioSystem EWS64XL - CS4236B */
+ { .id = "CSCa836", .devs = { { "CSCa800" }, { "CSCa810" } } },
+ /* ACER AW37/Pro - CS4235 */
+ { .id = "CSCd925", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* ACER AW35/Pro - CS4237B */
+ { .id = "CSCd937", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* CS4235 without MPU401 */
+ { .id = "CSCe825", .devs = { { "CSC0100" }, { "CSC0110" } } },
+ /* Unknown SiS530 - CS4235 */
+ { .id = "CSC4825", .devs = { { "CSC0100" }, { "CSC0110" } } },
+ /* IBM IntelliStation M Pro 6898 11U - CS4236B */
+ { .id = "CSCe835", .devs = { { "CSC0000" }, { "CSC0010" } } },
+ /* IBM PC 300PL Onboard - CS4236B */
+ { .id = "CSCe836", .devs = { { "CSC0000" }, { "CSC0010" } } },
+ /* Some noname CS4236 based card */
+ { .id = "CSCe936", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* CS4236B */
+ { .id = "CSCf235", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* CS4236B */
+ { .id = "CSCf238", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
+ /* --- */
+ { .id = "" } /* end */
+};
+#endif
+
+MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids);
+
+/* WSS initialization */
+static int __devinit snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev)
+{
+ if (pnp_activate_dev(pdev) < 0) {
+ printk(KERN_ERR IDENT " WSS PnP configure failed for WSS (out of resources?)\n");
+ return -EBUSY;
+ }
+ port[dev] = pnp_port_start(pdev, 0);
+ if (fm_port[dev] > 0)
+ fm_port[dev] = pnp_port_start(pdev, 1);
+ sb_port[dev] = pnp_port_start(pdev, 2);
+ irq[dev] = pnp_irq(pdev, 0);
+ dma1[dev] = pnp_dma(pdev, 0);
+ dma2[dev] = pnp_dma(pdev, 1) == 4 ? -1 : (int)pnp_dma(pdev, 1);
+ snd_printdd("isapnp WSS: wss port=0x%lx, fm port=0x%lx, sb port=0x%lx\n",
+ port[dev], fm_port[dev], sb_port[dev]);
+ snd_printdd("isapnp WSS: irq=%i, dma1=%i, dma2=%i\n",
+ irq[dev], dma1[dev], dma2[dev]);
+ return 0;
+}
+
+/* CTRL initialization */
+static int __devinit snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev)
+{
+ if (pnp_activate_dev(pdev) < 0) {
+ printk(KERN_ERR IDENT " CTRL PnP configure failed for WSS (out of resources?)\n");
+ return -EBUSY;
+ }
+ cport[dev] = pnp_port_start(pdev, 0);
+ snd_printdd("isapnp CTRL: control port=0x%lx\n", cport[dev]);
+ return 0;
+}
+
+/* MPU initialization */
+static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev)
+{
+ if (pnp_activate_dev(pdev) < 0) {
+ printk(KERN_ERR IDENT " MPU401 PnP configure failed for WSS (out of resources?)\n");
+ mpu_port[dev] = SNDRV_AUTO_PORT;
+ mpu_irq[dev] = SNDRV_AUTO_IRQ;
+ } else {
+ mpu_port[dev] = pnp_port_start(pdev, 0);
+ if (mpu_irq[dev] >= 0 &&
+ pnp_irq_valid(pdev, 0) && pnp_irq(pdev, 0) >= 0) {
+ mpu_irq[dev] = pnp_irq(pdev, 0);
+ } else {
+ mpu_irq[dev] = -1; /* disable interrupt */
+ }
+ }
+ snd_printdd("isapnp MPU: port=0x%lx, irq=%i\n", mpu_port[dev], mpu_irq[dev]);
+ return 0;
+}
+
+#ifdef CS4232
+static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard,
+ struct pnp_dev *pdev)
+{
+ acard->wss = pdev;
+ if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0)
+ return -EBUSY;
+ cport[dev] = -1;
+ return 0;
+}
+#endif
+
+static int __devinit snd_card_cs423x_pnpc(int dev, struct snd_card_cs4236 *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ acard->wss = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->wss == NULL)
+ return -EBUSY;
+ acard->ctrl = pnp_request_card_device(card, id->devs[1].id, NULL);
+ if (acard->ctrl == NULL)
+ return -EBUSY;
+ if (id->devs[2].id[0]) {
+ acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL);
+ if (acard->mpu == NULL)
+ return -EBUSY;
+ }
+
+ /* WSS initialization */
+ if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0)
+ return -EBUSY;
+
+ /* CTRL initialization */
+ if (acard->ctrl && cport[dev] > 0) {
+ if (snd_cs423x_pnp_init_ctrl(dev, acard->ctrl) < 0)
+ return -EBUSY;
+ }
+ /* MPU initialization */
+ if (acard->mpu && mpu_port[dev] > 0) {
+ if (snd_cs423x_pnp_init_mpu(dev, acard->mpu) < 0)
+ return -EBUSY;
+ }
+ return 0;
+}
+#endif /* CONFIG_PNP */
+
+#ifdef CONFIG_PNP
+#define is_isapnp_selected(dev) isapnp[dev]
+#else
+#define is_isapnp_selected(dev) 0
+#endif
+
+static void snd_card_cs4236_free(struct snd_card *card)
+{
+ struct snd_card_cs4236 *acard = card->private_data;
+
+ release_and_free_resource(acard->res_sb_port);
+}
+
+static struct snd_card *snd_cs423x_card_new(int dev)
+{
+ struct snd_card *card;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_cs4236));
+ if (card == NULL)
+ return NULL;
+ card->private_free = snd_card_cs4236_free;
+ return card;
+}
+
+static int __devinit snd_cs423x_probe(struct snd_card *card, int dev)
+{
+ struct snd_card_cs4236 *acard;
+ struct snd_pcm *pcm;
+ struct snd_wss *chip;
+ struct snd_opl3 *opl3;
+ int err;
+
+ acard = card->private_data;
+ if (sb_port[dev] > 0 && sb_port[dev] != SNDRV_AUTO_PORT)
+ if ((acard->res_sb_port = request_region(sb_port[dev], 16, IDENT " SB")) == NULL) {
+ printk(KERN_ERR IDENT ": unable to register SB port at 0x%lx\n", sb_port[dev]);
+ return -EBUSY;
+ }
+
+#ifdef CS4232
+ err = snd_wss_create(card, port[dev], cport[dev],
+ irq[dev],
+ dma1[dev], dma2[dev],
+ WSS_HW_DETECT, 0, &chip);
+ if (err < 0)
+ return err;
+ acard->chip = chip;
+
+ err = snd_wss_pcm(chip, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ err = snd_wss_mixer(chip);
+ if (err < 0)
+ return err;
+
+#else /* CS4236 */
+ err = snd_cs4236_create(card,
+ port[dev], cport[dev],
+ irq[dev], dma1[dev], dma2[dev],
+ WSS_HW_DETECT, 0, &chip);
+ if (err < 0)
+ return err;
+ acard->chip = chip;
+
+ err = snd_cs4236_pcm(chip, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ err = snd_cs4236_mixer(chip);
+ if (err < 0)
+ return err;
+#endif
+ strcpy(card->driver, pcm->name);
+ strcpy(card->shortname, pcm->name);
+ sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i",
+ pcm->name,
+ chip->port,
+ irq[dev],
+ dma1[dev]);
+ if (dma2[dev] >= 0)
+ sprintf(card->longname + strlen(card->longname), "&%d", dma2[dev]);
+
+ err = snd_wss_timer(chip, 0, NULL);
+ if (err < 0)
+ return err;
+
+ if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_opl3_create(card,
+ fm_port[dev], fm_port[dev] + 2,
+ OPL3_HW_OPL3_CS, 0, &opl3) < 0) {
+ printk(KERN_WARNING IDENT ": OPL3 not detected\n");
+ } else {
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0)
+ return err;
+ }
+ }
+
+ if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
+ if (mpu_irq[dev] == SNDRV_AUTO_IRQ)
+ mpu_irq[dev] = -1;
+ if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232,
+ mpu_port[dev], 0,
+ mpu_irq[dev],
+ mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0)
+ printk(KERN_WARNING IDENT ": MPU401 not detected\n");
+ }
+
+ return snd_card_register(card);
+}
+
+static int __devinit snd_cs423x_isa_match(struct device *pdev,
+ unsigned int dev)
+{
+ if (!enable[dev] || is_isapnp_selected(dev))
+ return 0;
+
+ if (port[dev] == SNDRV_AUTO_PORT) {
+ dev_err(pdev, "please specify port\n");
+ return 0;
+ }
+ if (cport[dev] == SNDRV_AUTO_PORT) {
+ dev_err(pdev, "please specify cport\n");
+ return 0;
+ }
+ if (irq[dev] == SNDRV_AUTO_IRQ) {
+ dev_err(pdev, "please specify irq\n");
+ return 0;
+ }
+ if (dma1[dev] == SNDRV_AUTO_DMA) {
+ dev_err(pdev, "please specify dma1\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_cs423x_isa_probe(struct device *pdev,
+ unsigned int dev)
+{
+ struct snd_card *card;
+ int err;
+
+ card = snd_cs423x_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ snd_card_set_dev(card, pdev);
+ if ((err = snd_cs423x_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ dev_set_drvdata(pdev, card);
+ return 0;
+}
+
+static int __devexit snd_cs423x_isa_remove(struct device *pdev,
+ unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(pdev));
+ dev_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_cs423x_suspend(struct snd_card *card)
+{
+ struct snd_card_cs4236 *acard = card->private_data;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ acard->chip->suspend(acard->chip);
+ return 0;
+}
+
+static int snd_cs423x_resume(struct snd_card *card)
+{
+ struct snd_card_cs4236 *acard = card->private_data;
+ acard->chip->resume(acard->chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+
+static int snd_cs423x_isa_suspend(struct device *dev, unsigned int n,
+ pm_message_t state)
+{
+ return snd_cs423x_suspend(dev_get_drvdata(dev));
+}
+
+static int snd_cs423x_isa_resume(struct device *dev, unsigned int n)
+{
+ return snd_cs423x_resume(dev_get_drvdata(dev));
+}
+#endif
+
+static struct isa_driver cs423x_isa_driver = {
+ .match = snd_cs423x_isa_match,
+ .probe = snd_cs423x_isa_probe,
+ .remove = __devexit_p(snd_cs423x_isa_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cs423x_isa_suspend,
+ .resume = snd_cs423x_isa_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+
+#ifdef CONFIG_PNP
+#ifdef CS4232
+static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev,
+ const struct pnp_device_id *id)
+{
+ static int dev;
+ int err;
+ struct snd_card *card;
+
+ if (pnp_device_is_isapnp(pdev))
+ return -ENOENT; /* we have another procedure - card */
+ for (; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_cs423x_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ if ((err = snd_card_cs4232_pnp(dev, card->private_data, pdev)) < 0) {
+ printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n");
+ snd_card_free(card);
+ return err;
+ }
+ snd_card_set_dev(card, &pdev->dev);
+ if ((err = snd_cs423x_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pnp_set_drvdata(pdev, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_cs4232_pnp_remove(struct pnp_dev * pdev)
+{
+ snd_card_free(pnp_get_drvdata(pdev));
+ pnp_set_drvdata(pdev, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_cs4232_pnp_suspend(struct pnp_dev *pdev, pm_message_t state)
+{
+ return snd_cs423x_suspend(pnp_get_drvdata(pdev));
+}
+
+static int snd_cs4232_pnp_resume(struct pnp_dev *pdev)
+{
+ return snd_cs423x_resume(pnp_get_drvdata(pdev));
+}
+#endif
+
+static struct pnp_driver cs4232_pnp_driver = {
+ .name = "cs4232-pnpbios",
+ .id_table = snd_cs4232_pnpbiosids,
+ .probe = snd_cs4232_pnpbios_detect,
+ .remove = __devexit_p(snd_cs4232_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cs4232_pnp_suspend,
+ .resume = snd_cs4232_pnp_resume,
+#endif
+};
+#endif /* CS4232 */
+
+static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ static int dev;
+ struct snd_card *card;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_cs423x_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ if ((res = snd_card_cs423x_pnpc(dev, card->private_data, pcard, pid)) < 0) {
+ printk(KERN_ERR "isapnp detection failed and probing for " IDENT
+ " is not supported\n");
+ snd_card_free(card);
+ return res;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((res = snd_cs423x_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return res;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_cs423x_pnpc_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_cs423x_pnpc_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ return snd_cs423x_suspend(pnp_get_card_drvdata(pcard));
+}
+
+static int snd_cs423x_pnpc_resume(struct pnp_card_link *pcard)
+{
+ return snd_cs423x_resume(pnp_get_card_drvdata(pcard));
+}
+#endif
+
+static struct pnp_card_driver cs423x_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = CS423X_ISAPNP_DRIVER,
+ .id_table = snd_cs423x_pnpids,
+ .probe = snd_cs423x_pnpc_detect,
+ .remove = __devexit_p(snd_cs423x_pnpc_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cs423x_pnpc_suspend,
+ .resume = snd_cs423x_pnpc_resume,
+#endif
+};
+#endif /* CONFIG_PNP */
+
+static int __init alsa_card_cs423x_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&cs423x_isa_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+#ifdef CS4232
+ err = pnp_register_driver(&cs4232_pnp_driver);
+ if (!err)
+ pnp_registered = 1;
+#endif
+ err = pnp_register_card_driver(&cs423x_pnpc_driver);
+ if (!err)
+ pnpc_registered = 1;
+#ifdef CS4232
+ if (pnp_registered)
+ err = 0;
+#endif
+ if (isa_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit alsa_card_cs423x_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnpc_registered)
+ pnp_unregister_card_driver(&cs423x_pnpc_driver);
+#ifdef CS4232
+ if (pnp_registered)
+ pnp_unregister_driver(&cs4232_pnp_driver);
+#endif
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&cs423x_isa_driver);
+}
+
+module_init(alsa_card_cs423x_init)
+module_exit(alsa_card_cs423x_exit)
diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c
new file mode 100644
index 0000000..6a85fdc
--- /dev/null
+++ b/sound/isa/cs423x/cs4236_lib.c
@@ -0,0 +1,1037 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of CS4235/4236B/4237B/4238B/4239 chips
+ *
+ * Note:
+ * -----
+ *
+ * Bugs:
+ * -----
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * Indirect control registers (CS4236B+)
+ *
+ * C0
+ * D8: WSS reset (all chips)
+ *
+ * C1 (all chips except CS4236)
+ * D7-D5: version
+ * D4-D0: chip id
+ * 11101 - CS4235
+ * 01011 - CS4236B
+ * 01000 - CS4237B
+ * 01001 - CS4238B
+ * 11110 - CS4239
+ *
+ * C2
+ * D7-D4: 3D Space (CS4235,CS4237B,CS4238B,CS4239)
+ * D3-D0: 3D Center (CS4237B); 3D Volume (CS4238B)
+ *
+ * C3
+ * D7: 3D Enable (CS4237B)
+ * D6: 3D Mono Enable (CS4237B)
+ * D5: 3D Serial Output (CS4237B,CS4238B)
+ * D4: 3D Enable (CS4235,CS4238B,CS4239)
+ *
+ * C4
+ * D7: consumer serial port enable (CS4237B,CS4238B)
+ * D6: channels status block reset (CS4237B,CS4238B)
+ * D5: user bit in sub-frame of digital audio data (CS4237B,CS4238B)
+ * D4: validity bit bit in sub-frame of digital audio data (CS4237B,CS4238B)
+ *
+ * C5 lower channel status (digital serial data description) (CS4237B,CS4238B)
+ * D7-D6: first two bits of category code
+ * D5: lock
+ * D4-D3: pre-emphasis (0 = none, 1 = 50/15us)
+ * D2: copy/copyright (0 = copy inhibited)
+ * D1: 0 = digital audio / 1 = non-digital audio
+ *
+ * C6 upper channel status (digital serial data description) (CS4237B,CS4238B)
+ * D7-D6: sample frequency (0 = 44.1kHz)
+ * D5: generation status (0 = no indication, 1 = original/commercially precaptureed data)
+ * D4-D0: category code (upper bits)
+ *
+ * C7 reserved (must write 0)
+ *
+ * C8 wavetable control
+ * D7: volume control interrupt enable (CS4235,CS4239)
+ * D6: hardware volume control format (CS4235,CS4239)
+ * D3: wavetable serial port enable (all chips)
+ * D2: DSP serial port switch (all chips)
+ * D1: disable MCLK (all chips)
+ * D0: force BRESET low (all chips)
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/asoundef.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips");
+MODULE_LICENSE("GPL");
+
+/*
+ *
+ */
+
+static unsigned char snd_cs4236_ext_map[18] = {
+ /* CS4236_LEFT_LINE */ 0xff,
+ /* CS4236_RIGHT_LINE */ 0xff,
+ /* CS4236_LEFT_MIC */ 0xdf,
+ /* CS4236_RIGHT_MIC */ 0xdf,
+ /* CS4236_LEFT_MIX_CTRL */ 0xe0 | 0x18,
+ /* CS4236_RIGHT_MIX_CTRL */ 0xe0,
+ /* CS4236_LEFT_FM */ 0xbf,
+ /* CS4236_RIGHT_FM */ 0xbf,
+ /* CS4236_LEFT_DSP */ 0xbf,
+ /* CS4236_RIGHT_DSP */ 0xbf,
+ /* CS4236_RIGHT_LOOPBACK */ 0xbf,
+ /* CS4236_DAC_MUTE */ 0xe0,
+ /* CS4236_ADC_RATE */ 0x01, /* 48kHz */
+ /* CS4236_DAC_RATE */ 0x01, /* 48kHz */
+ /* CS4236_LEFT_MASTER */ 0xbf,
+ /* CS4236_RIGHT_MASTER */ 0xbf,
+ /* CS4236_LEFT_WAVE */ 0xbf,
+ /* CS4236_RIGHT_WAVE */ 0xbf
+};
+
+/*
+ *
+ */
+
+static void snd_cs4236_ctrl_out(struct snd_wss *chip,
+ unsigned char reg, unsigned char val)
+{
+ outb(reg, chip->cport + 3);
+ outb(chip->cimage[reg] = val, chip->cport + 4);
+}
+
+static unsigned char snd_cs4236_ctrl_in(struct snd_wss *chip, unsigned char reg)
+{
+ outb(reg, chip->cport + 3);
+ return inb(chip->cport + 4);
+}
+
+/*
+ * PCM
+ */
+
+#define CLOCKS 8
+
+static struct snd_ratnum clocks[CLOCKS] = {
+ { .num = 16934400, .den_min = 353, .den_max = 353, .den_step = 1 },
+ { .num = 16934400, .den_min = 529, .den_max = 529, .den_step = 1 },
+ { .num = 16934400, .den_min = 617, .den_max = 617, .den_step = 1 },
+ { .num = 16934400, .den_min = 1058, .den_max = 1058, .den_step = 1 },
+ { .num = 16934400, .den_min = 1764, .den_max = 1764, .den_step = 1 },
+ { .num = 16934400, .den_min = 2117, .den_max = 2117, .den_step = 1 },
+ { .num = 16934400, .den_min = 2558, .den_max = 2558, .den_step = 1 },
+ { .num = 16934400/16, .den_min = 21, .den_max = 192, .den_step = 1 }
+};
+
+static struct snd_pcm_hw_constraint_ratnums hw_constraints_clocks = {
+ .nrats = CLOCKS,
+ .rats = clocks,
+};
+
+static int snd_cs4236_xrate(struct snd_pcm_runtime *runtime)
+{
+ return snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_clocks);
+}
+
+static unsigned char divisor_to_rate_register(unsigned int divisor)
+{
+ switch (divisor) {
+ case 353: return 1;
+ case 529: return 2;
+ case 617: return 3;
+ case 1058: return 4;
+ case 1764: return 5;
+ case 2117: return 6;
+ case 2558: return 7;
+ default:
+ if (divisor < 21 || divisor > 192) {
+ snd_BUG();
+ return 192;
+ }
+ return divisor;
+ }
+}
+
+static void snd_cs4236_playback_format(struct snd_wss *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char pdfr)
+{
+ unsigned long flags;
+ unsigned char rate = divisor_to_rate_register(params->rate_den);
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ /* set fast playback format change and clean playback FIFO */
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] | 0x10);
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT, pdfr & 0xf0);
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] & ~0x10);
+ snd_cs4236_ext_out(chip, CS4236_DAC_RATE, rate);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static void snd_cs4236_capture_format(struct snd_wss *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char cdfr)
+{
+ unsigned long flags;
+ unsigned char rate = divisor_to_rate_register(params->rate_den);
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ /* set fast capture format change and clean capture FIFO */
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] | 0x20);
+ snd_wss_out(chip, CS4231_REC_FORMAT, cdfr & 0xf0);
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] & ~0x20);
+ snd_cs4236_ext_out(chip, CS4236_ADC_RATE, rate);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+#ifdef CONFIG_PM
+
+static void snd_cs4236_suspend(struct snd_wss *chip)
+{
+ int reg;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ for (reg = 0; reg < 32; reg++)
+ chip->image[reg] = snd_wss_in(chip, reg);
+ for (reg = 0; reg < 18; reg++)
+ chip->eimage[reg] = snd_cs4236_ext_in(chip, CS4236_I23VAL(reg));
+ for (reg = 2; reg < 9; reg++)
+ chip->cimage[reg] = snd_cs4236_ctrl_in(chip, reg);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static void snd_cs4236_resume(struct snd_wss *chip)
+{
+ int reg;
+ unsigned long flags;
+
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ for (reg = 0; reg < 32; reg++) {
+ switch (reg) {
+ case CS4236_EXT_REG:
+ case CS4231_VERSION:
+ case 27: /* why? CS4235 - master left */
+ case 29: /* why? CS4235 - master right */
+ break;
+ default:
+ snd_wss_out(chip, reg, chip->image[reg]);
+ break;
+ }
+ }
+ for (reg = 0; reg < 18; reg++)
+ snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), chip->eimage[reg]);
+ for (reg = 2; reg < 9; reg++) {
+ switch (reg) {
+ case 7:
+ break;
+ default:
+ snd_cs4236_ctrl_out(chip, reg, chip->cimage[reg]);
+ }
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+}
+
+#endif /* CONFIG_PM */
+
+int snd_cs4236_create(struct snd_card *card,
+ unsigned long port,
+ unsigned long cport,
+ int irq, int dma1, int dma2,
+ unsigned short hardware,
+ unsigned short hwshare,
+ struct snd_wss **rchip)
+{
+ struct snd_wss *chip;
+ unsigned char ver1, ver2;
+ unsigned int reg;
+ int err;
+
+ *rchip = NULL;
+ if (hardware == WSS_HW_DETECT)
+ hardware = WSS_HW_DETECT3;
+ if (cport < 0x100) {
+ snd_printk("please, specify control port for CS4236+ chips\n");
+ return -ENODEV;
+ }
+ err = snd_wss_create(card, port, cport,
+ irq, dma1, dma2, hardware, hwshare, &chip);
+ if (err < 0)
+ return err;
+
+ if (!(chip->hardware & WSS_HW_CS4236B_MASK)) {
+ snd_printk("CS4236+: MODE3 and extended registers not available, hardware=0x%x\n",chip->hardware);
+ snd_device_free(card, chip);
+ return -ENODEV;
+ }
+#if 0
+ {
+ int idx;
+ for (idx = 0; idx < 8; idx++)
+ snd_printk("CD%i = 0x%x\n", idx, inb(chip->cport + idx));
+ for (idx = 0; idx < 9; idx++)
+ snd_printk("C%i = 0x%x\n", idx, snd_cs4236_ctrl_in(chip, idx));
+ }
+#endif
+ ver1 = snd_cs4236_ctrl_in(chip, 1);
+ ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION);
+ snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2);
+ if (ver1 != ver2) {
+ snd_printk("CS4236+ chip detected, but control port 0x%lx is not valid\n", cport);
+ snd_device_free(card, chip);
+ return -ENODEV;
+ }
+ snd_cs4236_ctrl_out(chip, 0, 0x00);
+ snd_cs4236_ctrl_out(chip, 2, 0xff);
+ snd_cs4236_ctrl_out(chip, 3, 0x00);
+ snd_cs4236_ctrl_out(chip, 4, 0x80);
+ snd_cs4236_ctrl_out(chip, 5, ((IEC958_AES1_CON_PCM_CODER & 3) << 6) | IEC958_AES0_CON_EMPHASIS_NONE);
+ snd_cs4236_ctrl_out(chip, 6, IEC958_AES1_CON_PCM_CODER >> 2);
+ snd_cs4236_ctrl_out(chip, 7, 0x00);
+ /* 0x8c for C8 is valid for Turtle Beach Malibu - the IEC-958 output */
+ /* is working with this setup, other hardware should have */
+ /* different signal paths and this value should be selectable */
+ /* in the future */
+ snd_cs4236_ctrl_out(chip, 8, 0x8c);
+ chip->rate_constraint = snd_cs4236_xrate;
+ chip->set_playback_format = snd_cs4236_playback_format;
+ chip->set_capture_format = snd_cs4236_capture_format;
+#ifdef CONFIG_PM
+ chip->suspend = snd_cs4236_suspend;
+ chip->resume = snd_cs4236_resume;
+#endif
+
+ /* initialize extended registers */
+ for (reg = 0; reg < sizeof(snd_cs4236_ext_map); reg++)
+ snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), snd_cs4236_ext_map[reg]);
+
+ /* initialize compatible but more featured registers */
+ snd_wss_out(chip, CS4231_LEFT_INPUT, 0x40);
+ snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x40);
+ snd_wss_out(chip, CS4231_AUX1_LEFT_INPUT, 0xff);
+ snd_wss_out(chip, CS4231_AUX1_RIGHT_INPUT, 0xff);
+ snd_wss_out(chip, CS4231_AUX2_LEFT_INPUT, 0xdf);
+ snd_wss_out(chip, CS4231_AUX2_RIGHT_INPUT, 0xdf);
+ snd_wss_out(chip, CS4231_RIGHT_LINE_IN, 0xff);
+ snd_wss_out(chip, CS4231_LEFT_LINE_IN, 0xff);
+ snd_wss_out(chip, CS4231_RIGHT_LINE_IN, 0xff);
+ switch (chip->hardware) {
+ case WSS_HW_CS4235:
+ case WSS_HW_CS4239:
+ snd_wss_out(chip, CS4235_LEFT_MASTER, 0xff);
+ snd_wss_out(chip, CS4235_RIGHT_MASTER, 0xff);
+ break;
+ }
+
+ *rchip = chip;
+ return 0;
+}
+
+int snd_cs4236_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_wss_pcm(chip, device, &pcm);
+ if (err < 0)
+ return err;
+ pcm->info_flags &= ~SNDRV_PCM_INFO_JOINT_DUPLEX;
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/*
+ * MIXER
+ */
+
+#define CS4236_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_cs4236_info_single, \
+ .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_cs4236_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_cs4236_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->eimage[CS4236_REG(reg)] >> shift) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_cs4236_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = (chip->eimage[CS4236_REG(reg)] & ~(mask << shift)) | val;
+ change = val != chip->eimage[CS4236_REG(reg)];
+ snd_cs4236_ext_out(chip, reg, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define CS4236_SINGLEC(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_cs4236_info_single, \
+ .get = snd_cs4236_get_singlec, .put = snd_cs4236_put_singlec, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_cs4236_get_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->cimage[reg] >> shift) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_cs4236_put_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = (chip->cimage[reg] & ~(mask << shift)) | val;
+ change = val != chip->cimage[reg];
+ snd_cs4236_ctrl_out(chip, reg, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define CS4236_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_cs4236_info_double, \
+ .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) }
+
+static int snd_cs4236_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_cs4236_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->eimage[CS4236_REG(left_reg)] >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (chip->eimage[CS4236_REG(right_reg)] >> shift_right) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_cs4236_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned short val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (left_reg != right_reg) {
+ val1 = (chip->eimage[CS4236_REG(left_reg)] & ~(mask << shift_left)) | val1;
+ val2 = (chip->eimage[CS4236_REG(right_reg)] & ~(mask << shift_right)) | val2;
+ change = val1 != chip->eimage[CS4236_REG(left_reg)] || val2 != chip->eimage[CS4236_REG(right_reg)];
+ snd_cs4236_ext_out(chip, left_reg, val1);
+ snd_cs4236_ext_out(chip, right_reg, val2);
+ } else {
+ val1 = (chip->eimage[CS4236_REG(left_reg)] & ~((mask << shift_left) | (mask << shift_right))) | val1 | val2;
+ change = val1 != chip->eimage[CS4236_REG(left_reg)];
+ snd_cs4236_ext_out(chip, left_reg, val1);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_cs4236_info_double, \
+ .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) }
+
+static int snd_cs4236_get_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (chip->eimage[CS4236_REG(right_reg)] >> shift_right) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_cs4236_put_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned short val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1;
+ val2 = (chip->eimage[CS4236_REG(right_reg)] & ~(mask << shift_right)) | val2;
+ change = val1 != chip->image[left_reg] || val2 != chip->eimage[CS4236_REG(right_reg)];
+ snd_wss_out(chip, left_reg, val1);
+ snd_cs4236_ext_out(chip, right_reg, val2);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define CS4236_MASTER_DIGITAL(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_cs4236_info_double, \
+ .get = snd_cs4236_get_master_digital, .put = snd_cs4236_put_master_digital, \
+ .private_value = 71 << 24 }
+
+static inline int snd_cs4236_mixer_master_digital_invert_volume(int vol)
+{
+ return (vol < 64) ? 63 - vol : 64 + (71 - vol);
+}
+
+static int snd_cs4236_get_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = snd_cs4236_mixer_master_digital_invert_volume(chip->eimage[CS4236_REG(CS4236_LEFT_MASTER)] & 0x7f);
+ ucontrol->value.integer.value[1] = snd_cs4236_mixer_master_digital_invert_volume(chip->eimage[CS4236_REG(CS4236_RIGHT_MASTER)] & 0x7f);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_cs4236_put_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned short val1, val2;
+
+ val1 = snd_cs4236_mixer_master_digital_invert_volume(ucontrol->value.integer.value[0] & 0x7f);
+ val2 = snd_cs4236_mixer_master_digital_invert_volume(ucontrol->value.integer.value[1] & 0x7f);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val1 = (chip->eimage[CS4236_REG(CS4236_LEFT_MASTER)] & ~0x7f) | val1;
+ val2 = (chip->eimage[CS4236_REG(CS4236_RIGHT_MASTER)] & ~0x7f) | val2;
+ change = val1 != chip->eimage[CS4236_REG(CS4236_LEFT_MASTER)] || val2 != chip->eimage[CS4236_REG(CS4236_RIGHT_MASTER)];
+ snd_cs4236_ext_out(chip, CS4236_LEFT_MASTER, val1);
+ snd_cs4236_ext_out(chip, CS4236_RIGHT_MASTER, val2);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define CS4235_OUTPUT_ACCU(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_cs4236_info_double, \
+ .get = snd_cs4235_get_output_accu, .put = snd_cs4235_put_output_accu, \
+ .private_value = 3 << 24 }
+
+static inline int snd_cs4235_mixer_output_accu_get_volume(int vol)
+{
+ switch ((vol >> 5) & 3) {
+ case 0: return 1;
+ case 1: return 3;
+ case 2: return 2;
+ case 3: return 0;
+ }
+ return 3;
+}
+
+static inline int snd_cs4235_mixer_output_accu_set_volume(int vol)
+{
+ switch (vol & 3) {
+ case 0: return 3 << 5;
+ case 1: return 0 << 5;
+ case 2: return 2 << 5;
+ case 3: return 1 << 5;
+ }
+ return 1 << 5;
+}
+
+static int snd_cs4235_get_output_accu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = snd_cs4235_mixer_output_accu_get_volume(chip->image[CS4235_LEFT_MASTER]);
+ ucontrol->value.integer.value[1] = snd_cs4235_mixer_output_accu_get_volume(chip->image[CS4235_RIGHT_MASTER]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_cs4235_put_output_accu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned short val1, val2;
+
+ val1 = snd_cs4235_mixer_output_accu_set_volume(ucontrol->value.integer.value[0]);
+ val2 = snd_cs4235_mixer_output_accu_set_volume(ucontrol->value.integer.value[1]);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val1 = (chip->image[CS4235_LEFT_MASTER] & ~(3 << 5)) | val1;
+ val2 = (chip->image[CS4235_RIGHT_MASTER] & ~(3 << 5)) | val2;
+ change = val1 != chip->image[CS4235_LEFT_MASTER] || val2 != chip->image[CS4235_RIGHT_MASTER];
+ snd_wss_out(chip, CS4235_LEFT_MASTER, val1);
+ snd_wss_out(chip, CS4235_RIGHT_MASTER, val2);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_cs4236_controls[] = {
+
+CS4236_DOUBLE("Master Digital Playback Switch", 0,
+ CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1),
+CS4236_DOUBLE("Master Digital Capture Switch", 0,
+ CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1),
+CS4236_MASTER_DIGITAL("Master Digital Volume", 0),
+
+CS4236_DOUBLE("Capture Boost Volume", 0,
+ CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1),
+
+WSS_DOUBLE("PCM Playback Switch", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
+WSS_DOUBLE("PCM Playback Volume", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1),
+
+CS4236_DOUBLE("DSP Playback Switch", 0,
+ CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1),
+CS4236_DOUBLE("DSP Playback Volume", 0,
+ CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1),
+
+CS4236_DOUBLE("FM Playback Switch", 0,
+ CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1),
+CS4236_DOUBLE("FM Playback Volume", 0,
+ CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1),
+
+CS4236_DOUBLE("Wavetable Playback Switch", 0,
+ CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1),
+CS4236_DOUBLE("Wavetable Playback Volume", 0,
+ CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1),
+
+WSS_DOUBLE("Synth Playback Switch", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
+WSS_DOUBLE("Synth Volume", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1),
+WSS_DOUBLE("Synth Capture Switch", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1),
+WSS_DOUBLE("Synth Capture Bypass", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 5, 5, 1, 1),
+
+CS4236_DOUBLE("Mic Playback Switch", 0,
+ CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1),
+CS4236_DOUBLE("Mic Capture Switch", 0,
+ CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1),
+CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1),
+CS4236_DOUBLE("Mic Playback Boost", 0,
+ CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0),
+
+WSS_DOUBLE("Line Playback Switch", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Line Volume", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1),
+WSS_DOUBLE("Line Capture Switch", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1),
+WSS_DOUBLE("Line Capture Bypass", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 5, 5, 1, 1),
+
+WSS_DOUBLE("CD Playback Switch", 0,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("CD Volume", 0,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1),
+WSS_DOUBLE("CD Capture Switch", 0,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1),
+
+CS4236_DOUBLE1("Mono Output Playback Switch", 0,
+ CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1),
+CS4236_DOUBLE1("Mono Playback Switch", 0,
+ CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1),
+WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1),
+WSS_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0),
+
+WSS_DOUBLE("Capture Volume", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0),
+WSS_DOUBLE("Analog Loopback Capture Switch", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0),
+
+WSS_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0),
+CS4236_DOUBLE1("Digital Loopback Playback Volume", 0,
+ CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1)
+};
+
+static struct snd_kcontrol_new snd_cs4235_controls[] = {
+
+WSS_DOUBLE("Master Switch", 0,
+ CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1),
+WSS_DOUBLE("Master Volume", 0,
+ CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1),
+
+CS4235_OUTPUT_ACCU("Playback Volume", 0),
+
+CS4236_DOUBLE("Master Digital Playback Switch", 0,
+ CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1),
+CS4236_DOUBLE("Master Digital Capture Switch", 0,
+ CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1),
+CS4236_MASTER_DIGITAL("Master Digital Volume", 0),
+
+WSS_DOUBLE("Master Digital Playback Switch", 1,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
+WSS_DOUBLE("Master Digital Capture Switch", 1,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1),
+WSS_DOUBLE("Master Digital Volume", 1,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1),
+
+CS4236_DOUBLE("Capture Volume", 0,
+ CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1),
+
+WSS_DOUBLE("PCM Switch", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
+WSS_DOUBLE("PCM Volume", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1),
+
+CS4236_DOUBLE("DSP Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1),
+
+CS4236_DOUBLE("FM Switch", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1),
+
+CS4236_DOUBLE("Wavetable Switch", 0,
+ CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1),
+
+CS4236_DOUBLE("Mic Capture Switch", 0,
+ CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1),
+CS4236_DOUBLE("Mic Playback Switch", 0,
+ CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1),
+CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1),
+CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0),
+
+WSS_DOUBLE("Aux Playback Switch", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Aux Capture Switch", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1),
+WSS_DOUBLE("Aux Volume", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1),
+
+WSS_DOUBLE("Aux Playback Switch", 1,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Aux Capture Switch", 1,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1),
+WSS_DOUBLE("Aux Volume", 1,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1),
+
+CS4236_DOUBLE1("Master Mono Switch", 0,
+ CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1),
+
+CS4236_DOUBLE1("Mono Switch", 0,
+ CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1),
+WSS_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1),
+
+WSS_DOUBLE("Analog Loopback Switch", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0),
+};
+
+#define CS4236_IEC958_ENABLE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_cs4236_info_single, \
+ .get = snd_cs4236_get_iec958_switch, .put = snd_cs4236_put_iec958_switch, \
+ .private_value = 1 << 16 }
+
+static int snd_cs4236_get_iec958_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = chip->image[CS4231_ALT_FEATURE_1] & 0x02 ? 1 : 0;
+#if 0
+ printk("get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n",
+ snd_wss_in(chip, CS4231_ALT_FEATURE_1),
+ snd_cs4236_ctrl_in(chip, 3),
+ snd_cs4236_ctrl_in(chip, 4),
+ snd_cs4236_ctrl_in(chip, 5),
+ snd_cs4236_ctrl_in(chip, 6),
+ snd_cs4236_ctrl_in(chip, 8));
+#endif
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned short enable, val;
+
+ enable = ucontrol->value.integer.value[0] & 1;
+
+ mutex_lock(&chip->mce_mutex);
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = (chip->image[CS4231_ALT_FEATURE_1] & ~0x0e) | (0<<2) | (enable << 1);
+ change = val != chip->image[CS4231_ALT_FEATURE_1];
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1, val);
+ val = snd_cs4236_ctrl_in(chip, 4) | 0xc0;
+ snd_cs4236_ctrl_out(chip, 4, val);
+ udelay(100);
+ val &= ~0x40;
+ snd_cs4236_ctrl_out(chip, 4, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+ mutex_unlock(&chip->mce_mutex);
+
+#if 0
+ printk("set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n",
+ snd_wss_in(chip, CS4231_ALT_FEATURE_1),
+ snd_cs4236_ctrl_in(chip, 3),
+ snd_cs4236_ctrl_in(chip, 4),
+ snd_cs4236_ctrl_in(chip, 5),
+ snd_cs4236_ctrl_in(chip, 6),
+ snd_cs4236_ctrl_in(chip, 8));
+#endif
+ return change;
+}
+
+static struct snd_kcontrol_new snd_cs4236_iec958_controls[] = {
+CS4236_IEC958_ENABLE("IEC958 Output Enable", 0),
+CS4236_SINGLEC("IEC958 Output Validity", 0, 4, 4, 1, 0),
+CS4236_SINGLEC("IEC958 Output User", 0, 4, 5, 1, 0),
+CS4236_SINGLEC("IEC958 Output CSBR", 0, 4, 6, 1, 0),
+CS4236_SINGLEC("IEC958 Output Channel Status Low", 0, 5, 1, 127, 0),
+CS4236_SINGLEC("IEC958 Output Channel Status High", 0, 6, 0, 255, 0)
+};
+
+static struct snd_kcontrol_new snd_cs4236_3d_controls_cs4235[] = {
+CS4236_SINGLEC("3D Control - Switch", 0, 3, 4, 1, 0),
+CS4236_SINGLEC("3D Control - Space", 0, 2, 4, 15, 1)
+};
+
+static struct snd_kcontrol_new snd_cs4236_3d_controls_cs4237[] = {
+CS4236_SINGLEC("3D Control - Switch", 0, 3, 7, 1, 0),
+CS4236_SINGLEC("3D Control - Space", 0, 2, 4, 15, 1),
+CS4236_SINGLEC("3D Control - Center", 0, 2, 0, 15, 1),
+CS4236_SINGLEC("3D Control - Mono", 0, 3, 6, 1, 0),
+CS4236_SINGLEC("3D Control - IEC958", 0, 3, 5, 1, 0)
+};
+
+static struct snd_kcontrol_new snd_cs4236_3d_controls_cs4238[] = {
+CS4236_SINGLEC("3D Control - Switch", 0, 3, 4, 1, 0),
+CS4236_SINGLEC("3D Control - Space", 0, 2, 4, 15, 1),
+CS4236_SINGLEC("3D Control - Volume", 0, 2, 0, 15, 1),
+CS4236_SINGLEC("3D Control - IEC958", 0, 3, 5, 1, 0)
+};
+
+int snd_cs4236_mixer(struct snd_wss *chip)
+{
+ struct snd_card *card;
+ unsigned int idx, count;
+ int err;
+ struct snd_kcontrol_new *kcontrol;
+
+ if (snd_BUG_ON(!chip || !chip->card))
+ return -EINVAL;
+ card = chip->card;
+ strcpy(card->mixername, snd_wss_chip_id(chip));
+
+ if (chip->hardware == WSS_HW_CS4235 ||
+ chip->hardware == WSS_HW_CS4239) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_cs4235_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4235_controls[idx], chip))) < 0)
+ return err;
+ }
+ } else {
+ for (idx = 0; idx < ARRAY_SIZE(snd_cs4236_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4236_controls[idx], chip))) < 0)
+ return err;
+ }
+ }
+ switch (chip->hardware) {
+ case WSS_HW_CS4235:
+ case WSS_HW_CS4239:
+ count = ARRAY_SIZE(snd_cs4236_3d_controls_cs4235);
+ kcontrol = snd_cs4236_3d_controls_cs4235;
+ break;
+ case WSS_HW_CS4237B:
+ count = ARRAY_SIZE(snd_cs4236_3d_controls_cs4237);
+ kcontrol = snd_cs4236_3d_controls_cs4237;
+ break;
+ case WSS_HW_CS4238B:
+ count = ARRAY_SIZE(snd_cs4236_3d_controls_cs4238);
+ kcontrol = snd_cs4236_3d_controls_cs4238;
+ break;
+ default:
+ count = 0;
+ kcontrol = NULL;
+ }
+ for (idx = 0; idx < count; idx++, kcontrol++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(kcontrol, chip))) < 0)
+ return err;
+ }
+ if (chip->hardware == WSS_HW_CS4237B ||
+ chip->hardware == WSS_HW_CS4238B) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_cs4236_iec958_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4236_iec958_controls[idx], chip))) < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_cs4236_create);
+EXPORT_SYMBOL(snd_cs4236_pcm);
+EXPORT_SYMBOL(snd_cs4236_mixer);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_cs4236_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_cs4236_exit(void)
+{
+}
+
+module_init(alsa_cs4236_init)
+module_exit(alsa_cs4236_exit)
diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c
new file mode 100644
index 0000000..a0242c3
--- /dev/null
+++ b/sound/isa/dt019x.c
@@ -0,0 +1,320 @@
+
+/*
+ dt019x.c - driver for Diamond Technologies DT-0197H based soundcards.
+ Copyright (C) 1999, 2002 by Massimo Piccioni <dafastidio@libero.it>
+
+ Generalised for soundcards based on DT-0196 and ALS-007 chips
+ by Jonathan Woithe <jwoithe@physics.adelaide.edu.au>: June 2002.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/wait.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/sb.h>
+
+#define PFX "dt019x: "
+
+MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
+MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X},"
+ "{Avance Logic ALS-007}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */
+static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for DT-019X based soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard.");
+
+struct snd_card_dt019x {
+ struct pnp_dev *dev;
+ struct pnp_dev *devmpu;
+ struct pnp_dev *devopl;
+ struct snd_sb *chip;
+};
+
+static struct pnp_card_device_id snd_dt019x_pnpids[] = {
+ /* DT197A30 */
+ { .id = "RWB1688", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } },
+ /* DT0196 / ALS-007 */
+ { .id = "ALS0007", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } },
+ { .id = "", }
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_dt019x_pnpids);
+
+
+#define DRIVER_NAME "snd-card-dt019x"
+
+
+static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *pid)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL);
+ if (acard->dev == NULL)
+ return -ENODEV;
+
+ acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL);
+ acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL);
+
+ pdev = acard->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "DT-019X AUDIO pnp configure failure\n");
+ return err;
+ }
+
+ port[dev] = pnp_port_start(pdev, 0);
+ dma8[dev] = pnp_dma(pdev, 0);
+ irq[dev] = pnp_irq(pdev, 0);
+ snd_printdd("dt019x: found audio interface: port=0x%lx, irq=0x%x, dma=0x%x\n",
+ port[dev],irq[dev],dma8[dev]);
+
+ pdev = acard->devmpu;
+ if (pdev != NULL) {
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ pnp_release_card_device(pdev);
+ snd_printk(KERN_ERR PFX "DT-019X MPU401 pnp configure failure, skipping\n");
+ goto __mpu_error;
+ }
+ mpu_port[dev] = pnp_port_start(pdev, 0);
+ mpu_irq[dev] = pnp_irq(pdev, 0);
+ snd_printdd("dt019x: found MPU-401: port=0x%lx, irq=0x%x\n",
+ mpu_port[dev],mpu_irq[dev]);
+ } else {
+ __mpu_error:
+ acard->devmpu = NULL;
+ mpu_port[dev] = -1;
+ }
+
+ pdev = acard->devopl;
+ if (pdev != NULL) {
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ pnp_release_card_device(pdev);
+ snd_printk(KERN_ERR PFX "DT-019X OPL3 pnp configure failure, skipping\n");
+ goto __fm_error;
+ }
+ fm_port[dev] = pnp_port_start(pdev, 0);
+ snd_printdd("dt019x: found OPL3 synth: port=0x%lx\n",fm_port[dev]);
+ } else {
+ __fm_error:
+ acard->devopl = NULL;
+ fm_port[dev] = -1;
+ }
+
+ return 0;
+}
+
+static int __devinit snd_card_dt019x_probe(int dev, struct pnp_card_link *pcard, const struct pnp_card_device_id *pid)
+{
+ int error;
+ struct snd_sb *chip;
+ struct snd_card *card;
+ struct snd_card_dt019x *acard;
+ struct snd_opl3 *opl3;
+
+ if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_dt019x))) == NULL)
+ return -ENOMEM;
+ acard = card->private_data;
+
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((error = snd_card_dt019x_pnp(dev, acard, pcard, pid))) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if ((error = snd_sbdsp_create(card, port[dev],
+ irq[dev],
+ snd_sb16dsp_interrupt,
+ dma8[dev],
+ -1,
+ SB_HW_DT019X,
+ &chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ acard->chip = chip;
+
+ strcpy(card->driver, "DT-019X");
+ strcpy(card->shortname, "Diamond Tech. DT-019X");
+ sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d",
+ card->shortname, chip->name, chip->port,
+ irq[dev], dma8[dev]);
+
+ if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ if ((error = snd_sbmixer_new(chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
+ if (mpu_irq[dev] == SNDRV_AUTO_IRQ)
+ mpu_irq[dev] = -1;
+ if (snd_mpu401_uart_new(card, 0,
+/* MPU401_HW_SB,*/
+ MPU401_HW_MPU401,
+ mpu_port[dev], 0,
+ mpu_irq[dev],
+ mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0,
+ NULL) < 0)
+ snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]);
+ }
+
+ if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_opl3_create(card,
+ fm_port[dev],
+ fm_port[dev] + 2,
+ OPL3_HW_AUTO, 0, &opl3) < 0) {
+ snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx ?\n",
+ fm_port[dev], fm_port[dev] + 2);
+ } else {
+ if ((error = snd_opl3_timer_new(opl3, 0, 1)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ }
+ }
+
+ if ((error = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ return 0;
+}
+
+static unsigned int __devinitdata dt019x_devices;
+
+static int __devinit snd_dt019x_pnp_probe(struct pnp_card_link *card,
+ const struct pnp_card_device_id *pid)
+{
+ static int dev;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (!enable[dev])
+ continue;
+ res = snd_card_dt019x_probe(dev, card, pid);
+ if (res < 0)
+ return res;
+ dev++;
+ dt019x_devices++;
+ return 0;
+ }
+ return -ENODEV;
+}
+
+static void __devexit snd_dt019x_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_dt019x_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_dt019x *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_sbmixer_suspend(chip);
+ return 0;
+}
+
+static int snd_dt019x_pnp_resume(struct pnp_card_link *pcard)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_dt019x *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_sbdsp_reset(chip);
+ snd_sbmixer_resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct pnp_card_driver dt019x_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "dt019x",
+ .id_table = snd_dt019x_pnpids,
+ .probe = snd_dt019x_pnp_probe,
+ .remove = __devexit_p(snd_dt019x_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_dt019x_pnp_suspend,
+ .resume = snd_dt019x_pnp_resume,
+#endif
+};
+
+static int __init alsa_card_dt019x_init(void)
+{
+ int err;
+
+ err = pnp_register_card_driver(&dt019x_pnpc_driver);
+ if (err)
+ return err;
+
+ if (!dt019x_devices) {
+ pnp_unregister_card_driver(&dt019x_pnpc_driver);
+#ifdef MODULE
+ snd_printk(KERN_ERR "no DT-019X / ALS-007 based soundcards found\n");
+#endif
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_dt019x_exit(void)
+{
+ pnp_unregister_card_driver(&dt019x_pnpc_driver);
+}
+
+module_init(alsa_card_dt019x_init)
+module_exit(alsa_card_dt019x_exit)
diff --git a/sound/isa/es1688/Makefile b/sound/isa/es1688/Makefile
new file mode 100644
index 0000000..aee1e4d
--- /dev/null
+++ b/sound/isa/es1688/Makefile
@@ -0,0 +1,11 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-es1688-lib-objs := es1688_lib.o
+snd-es1688-objs := es1688.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_ES1688) += snd-es1688.o snd-es1688-lib.o
+obj-$(CONFIG_SND_GUSEXTREME) += snd-es1688-lib.o
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
new file mode 100644
index 0000000..b463771
--- /dev/null
+++ b/sound/isa/es1688/es1688.c
@@ -0,0 +1,208 @@
+/*
+ * Driver for generic ESS AudioDrive ESx688 soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/moduleparam.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/es1688.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+#define CRD_NAME "Generic ESS ES1688/ES688 AudioDrive"
+#define DEV_NAME "es1688"
+
+MODULE_DESCRIPTION(CRD_NAME);
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ESS,ES688 PnP AudioDrive,pnp:ESS0100},"
+ "{ESS,ES1688 PnP AudioDrive,pnp:ESS0102},"
+ "{ESS,ES688 AudioDrive,pnp:ESS6881},"
+ "{ESS,ES1688 AudioDrive,pnp:ESS1681}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */
+static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1};
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */
+static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver.");
+module_param_array(mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver.");
+module_param_array(dma8, int, NULL, 0444);
+MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver.");
+
+static int __devinit snd_es1688_match(struct device *dev, unsigned int n)
+{
+ return enable[n];
+}
+
+static int __devinit snd_es1688_legacy_create(struct snd_card *card,
+ struct device *dev, unsigned int n, struct snd_es1688 **rchip)
+{
+ static long possible_ports[] = {0x220, 0x240, 0x260};
+ static int possible_irqs[] = {5, 9, 10, 7, -1};
+ static int possible_dmas[] = {1, 3, 0, -1};
+
+ int i, error;
+
+ if (irq[n] == SNDRV_AUTO_IRQ) {
+ irq[n] = snd_legacy_find_free_irq(possible_irqs);
+ if (irq[n] < 0) {
+ dev_err(dev, "unable to find a free IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (dma8[n] == SNDRV_AUTO_DMA) {
+ dma8[n] = snd_legacy_find_free_dma(possible_dmas);
+ if (dma8[n] < 0) {
+ dev_err(dev, "unable to find a free DMA\n");
+ return -EBUSY;
+ }
+ }
+
+ if (port[n] != SNDRV_AUTO_PORT)
+ return snd_es1688_create(card, port[n], mpu_port[n], irq[n],
+ mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip);
+
+ i = 0;
+ do {
+ port[n] = possible_ports[i];
+ error = snd_es1688_create(card, port[n], mpu_port[n], irq[n],
+ mpu_irq[n], dma8[n], ES1688_HW_AUTO, rchip);
+ } while (error < 0 && ++i < ARRAY_SIZE(possible_ports));
+
+ return error;
+}
+
+static int __devinit snd_es1688_probe(struct device *dev, unsigned int n)
+{
+ struct snd_card *card;
+ struct snd_es1688 *chip;
+ struct snd_opl3 *opl3;
+ struct snd_pcm *pcm;
+ int error;
+
+ card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
+ if (!card)
+ return -EINVAL;
+
+ error = snd_es1688_legacy_create(card, dev, n, &chip);
+ if (error < 0)
+ goto out;
+
+ error = snd_es1688_pcm(chip, 0, &pcm);
+ if (error < 0)
+ goto out;
+
+ error = snd_es1688_mixer(chip);
+ if (error < 0)
+ goto out;
+
+ strcpy(card->driver, "ES1688");
+ strcpy(card->shortname, pcm->name);
+ sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", pcm->name,
+ chip->port, chip->irq, chip->dma8);
+
+ if (snd_opl3_create(card, chip->port, chip->port + 2,
+ OPL3_HW_OPL3, 0, &opl3) < 0)
+ dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port);
+ else {
+ error = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+ if (error < 0)
+ goto out;
+ }
+
+ if (mpu_irq[n] >= 0 && mpu_irq[n] != SNDRV_AUTO_IRQ &&
+ chip->mpu_port > 0) {
+ error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
+ chip->mpu_port, 0,
+ mpu_irq[n], IRQF_DISABLED, NULL);
+ if (error < 0)
+ goto out;
+ }
+
+ snd_card_set_dev(card, dev);
+
+ error = snd_card_register(card);
+ if (error < 0)
+ goto out;
+
+ dev_set_drvdata(dev, card);
+ return 0;
+
+out: snd_card_free(card);
+ return error;
+}
+
+static int __devexit snd_es1688_remove(struct device *dev, unsigned int n)
+{
+ snd_card_free(dev_get_drvdata(dev));
+ dev_set_drvdata(dev, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_es1688_driver = {
+ .match = snd_es1688_match,
+ .probe = snd_es1688_probe,
+ .remove = snd_es1688_remove,
+#if 0 /* FIXME */
+ .suspend = snd_es1688_suspend,
+ .resume = snd_es1688_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ }
+};
+
+static int __init alsa_card_es1688_init(void)
+{
+ return isa_register_driver(&snd_es1688_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_es1688_exit(void)
+{
+ isa_unregister_driver(&snd_es1688_driver);
+}
+
+module_init(alsa_card_es1688_init);
+module_exit(alsa_card_es1688_exit);
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
new file mode 100644
index 0000000..4fbb508
--- /dev/null
+++ b/sound/isa/es1688/es1688_lib.c
@@ -0,0 +1,1053 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of ESS ES1688/688/488 chip
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <sound/core.h>
+#include <sound/es1688.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+#include <asm/dma.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("ESS ESx688 lowlevel module");
+MODULE_LICENSE("GPL");
+
+static int snd_es1688_dsp_command(struct snd_es1688 *chip, unsigned char val)
+{
+ int i;
+
+ for (i = 10000; i; i--)
+ if ((inb(ES1688P(chip, STATUS)) & 0x80) == 0) {
+ outb(val, ES1688P(chip, COMMAND));
+ return 1;
+ }
+#ifdef CONFIG_SND_DEBUG
+ printk("snd_es1688_dsp_command: timeout (0x%x)\n", val);
+#endif
+ return 0;
+}
+
+static int snd_es1688_dsp_get_byte(struct snd_es1688 *chip)
+{
+ int i;
+
+ for (i = 1000; i; i--)
+ if (inb(ES1688P(chip, DATA_AVAIL)) & 0x80)
+ return inb(ES1688P(chip, READ));
+ snd_printd("es1688 get byte failed: 0x%lx = 0x%x!!!\n", ES1688P(chip, DATA_AVAIL), inb(ES1688P(chip, DATA_AVAIL)));
+ return -ENODEV;
+}
+
+static int snd_es1688_write(struct snd_es1688 *chip,
+ unsigned char reg, unsigned char data)
+{
+ if (!snd_es1688_dsp_command(chip, reg))
+ return 0;
+ return snd_es1688_dsp_command(chip, data);
+}
+
+static int snd_es1688_read(struct snd_es1688 *chip, unsigned char reg)
+{
+ /* Read a byte from an extended mode register of ES1688 */
+ if (!snd_es1688_dsp_command(chip, 0xc0))
+ return -1;
+ if (!snd_es1688_dsp_command(chip, reg))
+ return -1;
+ return snd_es1688_dsp_get_byte(chip);
+}
+
+void snd_es1688_mixer_write(struct snd_es1688 *chip,
+ unsigned char reg, unsigned char data)
+{
+ outb(reg, ES1688P(chip, MIXER_ADDR));
+ udelay(10);
+ outb(data, ES1688P(chip, MIXER_DATA));
+ udelay(10);
+}
+
+static unsigned char snd_es1688_mixer_read(struct snd_es1688 *chip, unsigned char reg)
+{
+ unsigned char result;
+
+ outb(reg, ES1688P(chip, MIXER_ADDR));
+ udelay(10);
+ result = inb(ES1688P(chip, MIXER_DATA));
+ udelay(10);
+ return result;
+}
+
+static int snd_es1688_reset(struct snd_es1688 *chip)
+{
+ int i;
+
+ outb(3, ES1688P(chip, RESET)); /* valid only for ESS chips, SB -> 1 */
+ udelay(10);
+ outb(0, ES1688P(chip, RESET));
+ udelay(30);
+ for (i = 0; i < 1000 && !(inb(ES1688P(chip, DATA_AVAIL)) & 0x80); i++);
+ if (inb(ES1688P(chip, READ)) != 0xaa) {
+ snd_printd("ess_reset at 0x%lx: failed!!!\n", chip->port);
+ return -ENODEV;
+ }
+ snd_es1688_dsp_command(chip, 0xc6); /* enable extended mode */
+ return 0;
+}
+
+static int snd_es1688_probe(struct snd_es1688 *chip)
+{
+ unsigned long flags;
+ unsigned short major, minor, hw;
+ int i;
+
+ /*
+ * initialization sequence
+ */
+
+ spin_lock_irqsave(&chip->reg_lock, flags); /* Some ESS1688 cards need this */
+ inb(ES1688P(chip, ENABLE1)); /* ENABLE1 */
+ inb(ES1688P(chip, ENABLE1)); /* ENABLE1 */
+ inb(ES1688P(chip, ENABLE1)); /* ENABLE1 */
+ inb(ES1688P(chip, ENABLE2)); /* ENABLE2 */
+ inb(ES1688P(chip, ENABLE1)); /* ENABLE1 */
+ inb(ES1688P(chip, ENABLE2)); /* ENABLE2 */
+ inb(ES1688P(chip, ENABLE1)); /* ENABLE1 */
+ inb(ES1688P(chip, ENABLE1)); /* ENABLE1 */
+ inb(ES1688P(chip, ENABLE2)); /* ENABLE2 */
+ inb(ES1688P(chip, ENABLE1)); /* ENABLE1 */
+ inb(ES1688P(chip, ENABLE0)); /* ENABLE0 */
+
+ if (snd_es1688_reset(chip) < 0) {
+ snd_printdd("ESS: [0x%lx] reset failed... 0x%x\n", chip->port, inb(ES1688P(chip, READ)));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return -ENODEV;
+ }
+ snd_es1688_dsp_command(chip, 0xe7); /* return identification */
+
+ for (i = 1000, major = minor = 0; i; i--) {
+ if (inb(ES1688P(chip, DATA_AVAIL)) & 0x80) {
+ if (major == 0) {
+ major = inb(ES1688P(chip, READ));
+ } else {
+ minor = inb(ES1688P(chip, READ));
+ }
+ }
+ }
+
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ snd_printdd("ESS: [0x%lx] found.. major = 0x%x, minor = 0x%x\n", chip->port, major, minor);
+
+ chip->version = (major << 8) | minor;
+ if (!chip->version)
+ return -ENODEV; /* probably SB */
+
+ hw = ES1688_HW_AUTO;
+ switch (chip->version & 0xfff0) {
+ case 0x4880:
+ snd_printk("[0x%lx] ESS: AudioDrive ES488 detected, but driver is in another place\n", chip->port);
+ return -ENODEV;
+ case 0x6880:
+ hw = (chip->version & 0x0f) >= 8 ? ES1688_HW_1688 : ES1688_HW_688;
+ break;
+ default:
+ snd_printk("[0x%lx] ESS: unknown AudioDrive chip with version 0x%x (Jazz16 soundcard?)\n", chip->port, chip->version);
+ return -ENODEV;
+ }
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_write(chip, 0xb1, 0x10); /* disable IRQ */
+ snd_es1688_write(chip, 0xb2, 0x00); /* disable DMA */
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ /* enable joystick, but disable OPL3 */
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ snd_es1688_mixer_write(chip, 0x40, 0x01);
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+
+ return 0;
+}
+
+static int snd_es1688_init(struct snd_es1688 * chip, int enable)
+{
+ static int irqs[16] = {-1, -1, 0, -1, -1, 1, -1, 2, -1, 0, 3, -1, -1, -1, -1, -1};
+ unsigned long flags;
+ int cfg, irq_bits, dma, dma_bits, tmp, tmp1;
+
+ /* ok.. setup MPU-401 port and joystick and OPL3 */
+ cfg = 0x01; /* enable joystick, but disable OPL3 */
+ if (enable && chip->mpu_port >= 0x300 && chip->mpu_irq > 0 && chip->hardware != ES1688_HW_688) {
+ tmp = (chip->mpu_port & 0x0f0) >> 4;
+ if (tmp <= 3) {
+ switch (chip->mpu_irq) {
+ case 9:
+ tmp1 = 4;
+ break;
+ case 5:
+ tmp1 = 5;
+ break;
+ case 7:
+ tmp1 = 6;
+ break;
+ case 10:
+ tmp1 = 7;
+ break;
+ default:
+ tmp1 = 0;
+ }
+ if (tmp1) {
+ cfg |= (tmp << 3) | (tmp1 << 5);
+ }
+ }
+ }
+#if 0
+ snd_printk("mpu cfg = 0x%x\n", cfg);
+#endif
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_mixer_write(chip, 0x40, cfg);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ /* --- */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_read(chip, 0xb1);
+ snd_es1688_read(chip, 0xb2);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (enable) {
+ cfg = 0xf0; /* enable only DMA counter interrupt */
+ irq_bits = irqs[chip->irq & 0x0f];
+ if (irq_bits < 0) {
+ snd_printk("[0x%lx] ESS: bad IRQ %d for ES1688 chip!!\n", chip->port, chip->irq);
+#if 0
+ irq_bits = 0;
+ cfg = 0x10;
+#endif
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_write(chip, 0xb1, cfg | (irq_bits << 2));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ cfg = 0xf0; /* extended mode DMA enable */
+ dma = chip->dma8;
+ if (dma > 3 || dma == 2) {
+ snd_printk("[0x%lx] ESS: bad DMA channel %d for ES1688 chip!!\n", chip->port, dma);
+#if 0
+ dma_bits = 0;
+ cfg = 0x00; /* disable all DMA */
+#endif
+ return -EINVAL;
+ } else {
+ dma_bits = dma;
+ if (dma != 3)
+ dma_bits++;
+ }
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_write(chip, 0xb2, cfg | (dma_bits << 2));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ } else {
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_write(chip, 0xb1, 0x10); /* disable IRQ */
+ snd_es1688_write(chip, 0xb2, 0x00); /* disable DMA */
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_read(chip, 0xb1);
+ snd_es1688_read(chip, 0xb2);
+ snd_es1688_reset(chip);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+/*
+
+ */
+
+static struct snd_ratnum clocks[2] = {
+ {
+ .num = 795444,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ },
+ {
+ .num = 397722,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ }
+};
+
+static struct snd_pcm_hw_constraint_ratnums hw_constraints_clocks = {
+ .nrats = 2,
+ .rats = clocks,
+};
+
+static void snd_es1688_set_rate(struct snd_es1688 *chip, struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int bits, divider;
+
+ if (runtime->rate_num == clocks[0].num)
+ bits = 256 - runtime->rate_den;
+ else
+ bits = 128 - runtime->rate_den;
+ /* set filter register */
+ divider = 256 - 7160000*20/(8*82*runtime->rate);
+ /* write result to hardware */
+ snd_es1688_write(chip, 0xa1, bits);
+ snd_es1688_write(chip, 0xa2, divider);
+}
+
+static int snd_es1688_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static int snd_es1688_trigger(struct snd_es1688 *chip, int cmd, unsigned char value)
+{
+ int val;
+
+ if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ value = 0x00;
+ } else if (cmd != SNDRV_PCM_TRIGGER_START) {
+ return -EINVAL;
+ }
+ spin_lock(&chip->reg_lock);
+ chip->trigger_value = value;
+ val = snd_es1688_read(chip, 0xb8);
+ if ((val < 0) || (val & 0x0f) == value) {
+ spin_unlock(&chip->reg_lock);
+ return -EINVAL; /* something is wrong */
+ }
+#if 0
+ printk("trigger: val = 0x%x, value = 0x%x\n", val, value);
+ printk("trigger: pointer = 0x%x\n", snd_dma_pointer(chip->dma8, chip->dma_size));
+#endif
+ snd_es1688_write(chip, 0xb8, (val & 0xf0) | value);
+ spin_unlock(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_es1688_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_es1688_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_es1688_playback_prepare(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma_size = size;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_reset(chip);
+ snd_es1688_set_rate(chip, substream);
+ snd_es1688_write(chip, 0xb8, 4); /* auto init DMA mode */
+ snd_es1688_write(chip, 0xa8, (snd_es1688_read(chip, 0xa8) & ~0x03) | (3 - runtime->channels));
+ snd_es1688_write(chip, 0xb9, 2); /* demand mode (4 bytes/request) */
+ if (runtime->channels == 1) {
+ if (snd_pcm_format_width(runtime->format) == 8) {
+ /* 8. bit mono */
+ snd_es1688_write(chip, 0xb6, 0x80);
+ snd_es1688_write(chip, 0xb7, 0x51);
+ snd_es1688_write(chip, 0xb7, 0xd0);
+ } else {
+ /* 16. bit mono */
+ snd_es1688_write(chip, 0xb6, 0x00);
+ snd_es1688_write(chip, 0xb7, 0x71);
+ snd_es1688_write(chip, 0xb7, 0xf4);
+ }
+ } else {
+ if (snd_pcm_format_width(runtime->format) == 8) {
+ /* 8. bit stereo */
+ snd_es1688_write(chip, 0xb6, 0x80);
+ snd_es1688_write(chip, 0xb7, 0x51);
+ snd_es1688_write(chip, 0xb7, 0x98);
+ } else {
+ /* 16. bit stereo */
+ snd_es1688_write(chip, 0xb6, 0x00);
+ snd_es1688_write(chip, 0xb7, 0x71);
+ snd_es1688_write(chip, 0xb7, 0xbc);
+ }
+ }
+ snd_es1688_write(chip, 0xb1, (snd_es1688_read(chip, 0xb1) & 0x0f) | 0x50);
+ snd_es1688_write(chip, 0xb2, (snd_es1688_read(chip, 0xb2) & 0x0f) | 0x50);
+ snd_es1688_dsp_command(chip, ES1688_DSP_CMD_SPKON);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ /* --- */
+ count = -count;
+ snd_dma_program(chip->dma8, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_write(chip, 0xa4, (unsigned char) count);
+ snd_es1688_write(chip, 0xa5, (unsigned char) (count >> 8));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_es1688_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ return snd_es1688_trigger(chip, cmd, 0x05);
+}
+
+static int snd_es1688_capture_prepare(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma_size = size;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_reset(chip);
+ snd_es1688_set_rate(chip, substream);
+ snd_es1688_dsp_command(chip, ES1688_DSP_CMD_SPKOFF);
+ snd_es1688_write(chip, 0xb8, 0x0e); /* auto init DMA mode */
+ snd_es1688_write(chip, 0xa8, (snd_es1688_read(chip, 0xa8) & ~0x03) | (3 - runtime->channels));
+ snd_es1688_write(chip, 0xb9, 2); /* demand mode (4 bytes/request) */
+ if (runtime->channels == 1) {
+ if (snd_pcm_format_width(runtime->format) == 8) {
+ /* 8. bit mono */
+ snd_es1688_write(chip, 0xb7, 0x51);
+ snd_es1688_write(chip, 0xb7, 0xd0);
+ } else {
+ /* 16. bit mono */
+ snd_es1688_write(chip, 0xb7, 0x71);
+ snd_es1688_write(chip, 0xb7, 0xf4);
+ }
+ } else {
+ if (snd_pcm_format_width(runtime->format) == 8) {
+ /* 8. bit stereo */
+ snd_es1688_write(chip, 0xb7, 0x51);
+ snd_es1688_write(chip, 0xb7, 0x98);
+ } else {
+ /* 16. bit stereo */
+ snd_es1688_write(chip, 0xb7, 0x71);
+ snd_es1688_write(chip, 0xb7, 0xbc);
+ }
+ }
+ snd_es1688_write(chip, 0xb1, (snd_es1688_read(chip, 0xb1) & 0x0f) | 0x50);
+ snd_es1688_write(chip, 0xb2, (snd_es1688_read(chip, 0xb2) & 0x0f) | 0x50);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ /* --- */
+ count = -count;
+ snd_dma_program(chip->dma8, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1688_write(chip, 0xa4, (unsigned char) count);
+ snd_es1688_write(chip, 0xa5, (unsigned char) (count >> 8));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_es1688_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ return snd_es1688_trigger(chip, cmd, 0x0f);
+}
+
+static irqreturn_t snd_es1688_interrupt(int irq, void *dev_id)
+{
+ struct snd_es1688 *chip = dev_id;
+
+ if (chip->trigger_value == 0x05) /* ok.. playback is active */
+ snd_pcm_period_elapsed(chip->playback_substream);
+ if (chip->trigger_value == 0x0f) /* ok.. capture is active */
+ snd_pcm_period_elapsed(chip->capture_substream);
+
+ inb(ES1688P(chip, DATA_AVAIL)); /* ack interrupt */
+ return IRQ_HANDLED;
+}
+
+static snd_pcm_uframes_t snd_es1688_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (chip->trigger_value != 0x05)
+ return 0;
+ ptr = snd_dma_pointer(chip->dma8, chip->dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_es1688_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (chip->trigger_value != 0x0f)
+ return 0;
+ ptr = snd_dma_pointer(chip->dma8, chip->dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+/*
+
+ */
+
+static struct snd_pcm_hardware snd_es1688_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_es1688_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+
+ */
+
+static int snd_es1688_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (chip->capture_substream != NULL)
+ return -EAGAIN;
+ chip->playback_substream = substream;
+ runtime->hw = snd_es1688_playback;
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_clocks);
+ return 0;
+}
+
+static int snd_es1688_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (chip->playback_substream != NULL)
+ return -EAGAIN;
+ chip->capture_substream = substream;
+ runtime->hw = snd_es1688_capture;
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_clocks);
+ return 0;
+}
+
+static int snd_es1688_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+
+ chip->playback_substream = NULL;
+ return 0;
+}
+
+static int snd_es1688_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_es1688 *chip = snd_pcm_substream_chip(substream);
+
+ chip->capture_substream = NULL;
+ return 0;
+}
+
+static int snd_es1688_free(struct snd_es1688 *chip)
+{
+ if (chip->res_port) {
+ snd_es1688_init(chip, 0);
+ release_and_free_resource(chip->res_port);
+ }
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *) chip);
+ if (chip->dma8 >= 0) {
+ disable_dma(chip->dma8);
+ free_dma(chip->dma8);
+ }
+ kfree(chip);
+ return 0;
+}
+
+static int snd_es1688_dev_free(struct snd_device *device)
+{
+ struct snd_es1688 *chip = device->device_data;
+ return snd_es1688_free(chip);
+}
+
+static const char *snd_es1688_chip_id(struct snd_es1688 *chip)
+{
+ static char tmp[16];
+ sprintf(tmp, "ES%s688 rev %i", chip->hardware == ES1688_HW_688 ? "" : "1", chip->version & 0x0f);
+ return tmp;
+}
+
+int snd_es1688_create(struct snd_card *card,
+ unsigned long port,
+ unsigned long mpu_port,
+ int irq,
+ int mpu_irq,
+ int dma8,
+ unsigned short hardware,
+ struct snd_es1688 **rchip)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_es1688_dev_free,
+ };
+
+ struct snd_es1688 *chip;
+ int err;
+
+ *rchip = NULL;
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ chip->irq = -1;
+ chip->dma8 = -1;
+
+ if ((chip->res_port = request_region(port + 4, 12, "ES1688")) == NULL) {
+ snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4);
+ snd_es1688_free(chip);
+ return -EBUSY;
+ }
+ if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) {
+ snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq);
+ snd_es1688_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = irq;
+ if (request_dma(dma8, "ES1688")) {
+ snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8);
+ snd_es1688_free(chip);
+ return -EBUSY;
+ }
+ chip->dma8 = dma8;
+
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->mixer_lock);
+ chip->card = card;
+ chip->port = port;
+ mpu_port &= ~0x000f;
+ if (mpu_port < 0x300 || mpu_port > 0x330)
+ mpu_port = 0;
+ chip->mpu_port = mpu_port;
+ chip->mpu_irq = mpu_irq;
+ chip->hardware = hardware;
+
+ if ((err = snd_es1688_probe(chip)) < 0) {
+ snd_es1688_free(chip);
+ return err;
+ }
+ if ((err = snd_es1688_init(chip, 1)) < 0) {
+ snd_es1688_free(chip);
+ return err;
+ }
+
+ /* Register device */
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_es1688_free(chip);
+ return err;
+ }
+
+ *rchip = chip;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_es1688_playback_ops = {
+ .open = snd_es1688_playback_open,
+ .close = snd_es1688_playback_close,
+ .ioctl = snd_es1688_ioctl,
+ .hw_params = snd_es1688_hw_params,
+ .hw_free = snd_es1688_hw_free,
+ .prepare = snd_es1688_playback_prepare,
+ .trigger = snd_es1688_playback_trigger,
+ .pointer = snd_es1688_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_es1688_capture_ops = {
+ .open = snd_es1688_capture_open,
+ .close = snd_es1688_capture_close,
+ .ioctl = snd_es1688_ioctl,
+ .hw_params = snd_es1688_hw_params,
+ .hw_free = snd_es1688_hw_free,
+ .prepare = snd_es1688_capture_prepare,
+ .trigger = snd_es1688_capture_trigger,
+ .pointer = snd_es1688_capture_pointer,
+};
+
+int snd_es1688_pcm(struct snd_es1688 * chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(chip->card, "ESx688", device, 1, 1, &pcm)) < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1688_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_es1688_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
+ sprintf(pcm->name, snd_es1688_chip_id(chip));
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, 64*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/*
+ * MIXER part
+ */
+
+static int snd_es1688_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[9] = {
+ "Mic", "Mic Master", "CD", "AOUT",
+ "Mic1", "Mix", "Line", "Master"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item > 7)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_es1688_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es1688 *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = snd_es1688_mixer_read(chip, ES1688_REC_DEV) & 7;
+ return 0;
+}
+
+static int snd_es1688_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es1688 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned char oval, nval;
+ int change;
+
+ if (ucontrol->value.enumerated.item[0] > 8)
+ return -EINVAL;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ oval = snd_es1688_mixer_read(chip, ES1688_REC_DEV);
+ nval = (ucontrol->value.enumerated.item[0] & 7) | (oval & ~15);
+ change = nval != oval;
+ if (change)
+ snd_es1688_mixer_write(chip, ES1688_REC_DEV, nval);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define ES1688_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_es1688_info_single, \
+ .get = snd_es1688_get_single, .put = snd_es1688_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_es1688_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_es1688_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es1688 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (snd_es1688_mixer_read(chip, reg) >> shift) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_es1688_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es1688 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned char oval, nval;
+
+ nval = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ nval = mask - nval;
+ nval <<= shift;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ oval = snd_es1688_mixer_read(chip, reg);
+ nval = (oval & ~(mask << shift)) | nval;
+ change = nval != oval;
+ if (change)
+ snd_es1688_mixer_write(chip, reg, nval);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define ES1688_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_es1688_info_double, \
+ .get = snd_es1688_get_double, .put = snd_es1688_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) }
+
+static int snd_es1688_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_es1688_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es1688 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ unsigned char left, right;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (left_reg < 0xa0)
+ left = snd_es1688_mixer_read(chip, left_reg);
+ else
+ left = snd_es1688_read(chip, left_reg);
+ if (left_reg != right_reg) {
+ if (right_reg < 0xa0)
+ right = snd_es1688_mixer_read(chip, right_reg);
+ else
+ right = snd_es1688_read(chip, right_reg);
+ } else
+ right = left;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (left >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (right >> shift_right) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_es1688_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es1688 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned char val1, val2, oval1, oval2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (left_reg != right_reg) {
+ if (left_reg < 0xa0)
+ oval1 = snd_es1688_mixer_read(chip, left_reg);
+ else
+ oval1 = snd_es1688_read(chip, left_reg);
+ if (right_reg < 0xa0)
+ oval2 = snd_es1688_mixer_read(chip, right_reg);
+ else
+ oval2 = snd_es1688_read(chip, right_reg);
+ val1 = (oval1 & ~(mask << shift_left)) | val1;
+ val2 = (oval2 & ~(mask << shift_right)) | val2;
+ change = val1 != oval1 || val2 != oval2;
+ if (change) {
+ if (left_reg < 0xa0)
+ snd_es1688_mixer_write(chip, left_reg, val1);
+ else
+ snd_es1688_write(chip, left_reg, val1);
+ if (right_reg < 0xa0)
+ snd_es1688_mixer_write(chip, right_reg, val1);
+ else
+ snd_es1688_write(chip, right_reg, val1);
+ }
+ } else {
+ if (left_reg < 0xa0)
+ oval1 = snd_es1688_mixer_read(chip, left_reg);
+ else
+ oval1 = snd_es1688_read(chip, left_reg);
+ val1 = (oval1 & ~((mask << shift_left) | (mask << shift_right))) | val1 | val2;
+ change = val1 != oval1;
+ if (change) {
+ if (left_reg < 0xa0)
+ snd_es1688_mixer_write(chip, left_reg, val1);
+ else
+ snd_es1688_write(chip, left_reg, val1);
+ }
+
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_es1688_controls[] = {
+ES1688_DOUBLE("Master Playback Volume", 0, ES1688_MASTER_DEV, ES1688_MASTER_DEV, 4, 0, 15, 0),
+ES1688_DOUBLE("PCM Playback Volume", 0, ES1688_PCM_DEV, ES1688_PCM_DEV, 4, 0, 15, 0),
+ES1688_DOUBLE("Line Playback Volume", 0, ES1688_LINE_DEV, ES1688_LINE_DEV, 4, 0, 15, 0),
+ES1688_DOUBLE("CD Playback Volume", 0, ES1688_CD_DEV, ES1688_CD_DEV, 4, 0, 15, 0),
+ES1688_DOUBLE("FM Playback Volume", 0, ES1688_FM_DEV, ES1688_FM_DEV, 4, 0, 15, 0),
+ES1688_DOUBLE("Mic Playback Volume", 0, ES1688_MIC_DEV, ES1688_MIC_DEV, 4, 0, 15, 0),
+ES1688_DOUBLE("Aux Playback Volume", 0, ES1688_AUX_DEV, ES1688_AUX_DEV, 4, 0, 15, 0),
+ES1688_SINGLE("PC Speaker Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0),
+ES1688_DOUBLE("Capture Volume", 0, ES1688_RECLEV_DEV, ES1688_RECLEV_DEV, 4, 0, 15, 0),
+ES1688_SINGLE("Capture Switch", 0, ES1688_REC_DEV, 4, 1, 1),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_es1688_info_mux,
+ .get = snd_es1688_get_mux,
+ .put = snd_es1688_put_mux,
+},
+};
+
+#define ES1688_INIT_TABLE_SIZE (sizeof(snd_es1688_init_table)/2)
+
+static unsigned char snd_es1688_init_table[][2] = {
+ { ES1688_MASTER_DEV, 0 },
+ { ES1688_PCM_DEV, 0 },
+ { ES1688_LINE_DEV, 0 },
+ { ES1688_CD_DEV, 0 },
+ { ES1688_FM_DEV, 0 },
+ { ES1688_MIC_DEV, 0 },
+ { ES1688_AUX_DEV, 0 },
+ { ES1688_SPEAKER_DEV, 0 },
+ { ES1688_RECLEV_DEV, 0 },
+ { ES1688_REC_DEV, 0x17 }
+};
+
+int snd_es1688_mixer(struct snd_es1688 *chip)
+{
+ struct snd_card *card;
+ unsigned int idx;
+ int err;
+ unsigned char reg, val;
+
+ if (snd_BUG_ON(!chip || !chip->card))
+ return -EINVAL;
+
+ card = chip->card;
+
+ strcpy(card->mixername, snd_es1688_chip_id(chip));
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_es1688_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es1688_controls[idx], chip))) < 0)
+ return err;
+ }
+ for (idx = 0; idx < ES1688_INIT_TABLE_SIZE; idx++) {
+ reg = snd_es1688_init_table[idx][0];
+ val = snd_es1688_init_table[idx][1];
+ if (reg < 0xa0)
+ snd_es1688_mixer_write(chip, reg, val);
+ else
+ snd_es1688_write(chip, reg, val);
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_es1688_mixer_write);
+EXPORT_SYMBOL(snd_es1688_create);
+EXPORT_SYMBOL(snd_es1688_pcm);
+EXPORT_SYMBOL(snd_es1688_mixer);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_es1688_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_es1688_exit(void)
+{
+}
+
+module_init(alsa_es1688_init)
+module_exit(alsa_es1688_exit)
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
new file mode 100644
index 0000000..90498e4
--- /dev/null
+++ b/sound/isa/es18xx.c
@@ -0,0 +1,2452 @@
+/*
+ * Driver for generic ESS AudioDrive ES18xx soundcards
+ * Copyright (c) by Christian Fischbach <fishbach@pool.informatik.rwth-aachen.de>
+ * Copyright (c) by Abramo Bagnara <abramo@alsa-project.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+/* GENERAL NOTES:
+ *
+ * BUGS:
+ * - There are pops (we can't delay in trigger function, cause midlevel
+ * often need to trigger down and then up very quickly).
+ * Any ideas?
+ * - Support for 16 bit DMA seems to be broken. I've no hardware to tune it.
+ */
+
+/*
+ * ES1868 NOTES:
+ * - The chip has one half duplex pcm (with very limited full duplex support).
+ *
+ * - Duplex stereophonic sound is impossible.
+ * - Record and playback must share the same frequency rate.
+ *
+ * - The driver use dma2 for playback and dma1 for capture.
+ */
+
+/*
+ * ES1869 NOTES:
+ *
+ * - there are a first full duplex pcm and a second playback only pcm
+ * (incompatible with first pcm capture)
+ *
+ * - there is support for the capture volume and ESS Spatializer 3D effect.
+ *
+ * - contrarily to some pages in DS_1869.PDF the rates can be set
+ * independently.
+ *
+ * - Zoom Video is implemented by sharing the FM DAC, thus the user can
+ * have either FM playback or Video playback but not both simultaneously.
+ * The Video Playback Switch mixer control toggles this choice.
+ *
+ * BUGS:
+ *
+ * - There is a major trouble I noted:
+ *
+ * using both channel for playback stereo 16 bit samples at 44100 Hz
+ * the second pcm (Audio1) DMA slows down irregularly and sound is garbled.
+ *
+ * The same happens using Audio1 for captureing.
+ *
+ * The Windows driver does not suffer of this (although it use Audio1
+ * only for captureing). I'm unable to discover why.
+ *
+ */
+
+/*
+ * ES1879 NOTES:
+ * - When Zoom Video is enabled (reg 0x71 bit 6 toggled on) the PCM playback
+ * seems to be effected (speaker_test plays a lower frequency). Can't find
+ * anything in the datasheet to account for this, so a Video Playback Switch
+ * control has been included to allow ZV to be enabled only when necessary.
+ * Then again on at least one test system the 0x71 bit 6 enable bit is not
+ * needed for ZV, so maybe the datasheet is entirely wrong here.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/slab.h>
+#include <linux/pnp.h>
+#include <linux/isapnp.h>
+#include <linux/moduleparam.h>
+#include <linux/delay.h>
+
+#include <asm/io.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+#define PFX "es18xx: "
+
+struct snd_es18xx {
+ unsigned long port; /* port of ESS chip */
+ unsigned long mpu_port; /* MPU-401 port of ESS chip */
+ unsigned long fm_port; /* FM port */
+ unsigned long ctrl_port; /* Control port of ESS chip */
+ struct resource *res_port;
+ struct resource *res_mpu_port;
+ struct resource *res_ctrl_port;
+ int irq; /* IRQ number of ESS chip */
+ int dma1; /* DMA1 */
+ int dma2; /* DMA2 */
+ unsigned short version; /* version of ESS chip */
+ int caps; /* Chip capabilities */
+ unsigned short audio2_vol; /* volume level of audio2 */
+
+ unsigned short active; /* active channel mask */
+ unsigned int dma1_size;
+ unsigned int dma2_size;
+ unsigned int dma1_shift;
+ unsigned int dma2_shift;
+
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_a_substream;
+ struct snd_pcm_substream *capture_a_substream;
+ struct snd_pcm_substream *playback_b_substream;
+
+ struct snd_rawmidi *rmidi;
+
+ struct snd_kcontrol *hw_volume;
+ struct snd_kcontrol *hw_switch;
+ struct snd_kcontrol *master_volume;
+ struct snd_kcontrol *master_switch;
+
+ spinlock_t reg_lock;
+ spinlock_t mixer_lock;
+ spinlock_t ctrl_lock;
+#ifdef CONFIG_PM
+ unsigned char pm_reg;
+#endif
+};
+
+struct snd_audiodrive {
+ struct snd_es18xx *chip;
+#ifdef CONFIG_PNP
+ struct pnp_dev *dev;
+ struct pnp_dev *devc;
+#endif
+};
+
+#define AUDIO1_IRQ 0x01
+#define AUDIO2_IRQ 0x02
+#define HWV_IRQ 0x04
+#define MPU_IRQ 0x08
+
+#define ES18XX_PCM2 0x0001 /* Has two useable PCM */
+#define ES18XX_SPATIALIZER 0x0002 /* Has 3D Spatializer */
+#define ES18XX_RECMIX 0x0004 /* Has record mixer */
+#define ES18XX_DUPLEX_MONO 0x0008 /* Has mono duplex only */
+#define ES18XX_DUPLEX_SAME 0x0010 /* Playback and record must share the same rate */
+#define ES18XX_NEW_RATE 0x0020 /* More precise rate setting */
+#define ES18XX_AUXB 0x0040 /* AuxB mixer control */
+#define ES18XX_HWV 0x0080 /* Has separate hardware volume mixer controls*/
+#define ES18XX_MONO 0x0100 /* Mono_in mixer control */
+#define ES18XX_I2S 0x0200 /* I2S mixer control */
+#define ES18XX_MUTEREC 0x0400 /* Record source can be muted */
+#define ES18XX_CONTROL 0x0800 /* Has control ports */
+
+/* Power Management */
+#define ES18XX_PM 0x07
+#define ES18XX_PM_GPO0 0x01
+#define ES18XX_PM_GPO1 0x02
+#define ES18XX_PM_PDR 0x04
+#define ES18XX_PM_ANA 0x08
+#define ES18XX_PM_FM 0x020
+#define ES18XX_PM_SUS 0x080
+
+/* Lowlevel */
+
+#define DAC1 0x01
+#define ADC1 0x02
+#define DAC2 0x04
+#define MILLISECOND 10000
+
+static int snd_es18xx_dsp_command(struct snd_es18xx *chip, unsigned char val)
+{
+ int i;
+
+ for(i = MILLISECOND; i; i--)
+ if ((inb(chip->port + 0x0C) & 0x80) == 0) {
+ outb(val, chip->port + 0x0C);
+ return 0;
+ }
+ snd_printk(KERN_ERR "dsp_command: timeout (0x%x)\n", val);
+ return -EINVAL;
+}
+
+static int snd_es18xx_dsp_get_byte(struct snd_es18xx *chip)
+{
+ int i;
+
+ for(i = MILLISECOND/10; i; i--)
+ if (inb(chip->port + 0x0C) & 0x40)
+ return inb(chip->port + 0x0A);
+ snd_printk(KERN_ERR "dsp_get_byte failed: 0x%lx = 0x%x!!!\n",
+ chip->port + 0x0A, inb(chip->port + 0x0A));
+ return -ENODEV;
+}
+
+#undef REG_DEBUG
+
+static int snd_es18xx_write(struct snd_es18xx *chip,
+ unsigned char reg, unsigned char data)
+{
+ unsigned long flags;
+ int ret;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ret = snd_es18xx_dsp_command(chip, reg);
+ if (ret < 0)
+ goto end;
+ ret = snd_es18xx_dsp_command(chip, data);
+ end:
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Reg %02x set to %02x\n", reg, data);
+#endif
+ return ret;
+}
+
+static int snd_es18xx_read(struct snd_es18xx *chip, unsigned char reg)
+{
+ unsigned long flags;
+ int ret, data;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ret = snd_es18xx_dsp_command(chip, 0xC0);
+ if (ret < 0)
+ goto end;
+ ret = snd_es18xx_dsp_command(chip, reg);
+ if (ret < 0)
+ goto end;
+ data = snd_es18xx_dsp_get_byte(chip);
+ ret = data;
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Reg %02x now is %02x (%d)\n", reg, data, ret);
+#endif
+ end:
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return ret;
+}
+
+/* Return old value */
+static int snd_es18xx_bits(struct snd_es18xx *chip, unsigned char reg,
+ unsigned char mask, unsigned char val)
+{
+ int ret;
+ unsigned char old, new, oval;
+ unsigned long flags;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ret = snd_es18xx_dsp_command(chip, 0xC0);
+ if (ret < 0)
+ goto end;
+ ret = snd_es18xx_dsp_command(chip, reg);
+ if (ret < 0)
+ goto end;
+ ret = snd_es18xx_dsp_get_byte(chip);
+ if (ret < 0) {
+ goto end;
+ }
+ old = ret;
+ oval = old & mask;
+ if (val != oval) {
+ ret = snd_es18xx_dsp_command(chip, reg);
+ if (ret < 0)
+ goto end;
+ new = (old & ~mask) | (val & mask);
+ ret = snd_es18xx_dsp_command(chip, new);
+ if (ret < 0)
+ goto end;
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Reg %02x was %02x, set to %02x (%d)\n",
+ reg, old, new, ret);
+#endif
+ }
+ ret = oval;
+ end:
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return ret;
+}
+
+static inline void snd_es18xx_mixer_write(struct snd_es18xx *chip,
+ unsigned char reg, unsigned char data)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ outb(reg, chip->port + 0x04);
+ outb(data, chip->port + 0x05);
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Mixer reg %02x set to %02x\n", reg, data);
+#endif
+}
+
+static inline int snd_es18xx_mixer_read(struct snd_es18xx *chip, unsigned char reg)
+{
+ unsigned long flags;
+ int data;
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ outb(reg, chip->port + 0x04);
+ data = inb(chip->port + 0x05);
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Mixer reg %02x now is %02x\n", reg, data);
+#endif
+ return data;
+}
+
+/* Return old value */
+static inline int snd_es18xx_mixer_bits(struct snd_es18xx *chip, unsigned char reg,
+ unsigned char mask, unsigned char val)
+{
+ unsigned char old, new, oval;
+ unsigned long flags;
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ outb(reg, chip->port + 0x04);
+ old = inb(chip->port + 0x05);
+ oval = old & mask;
+ if (val != oval) {
+ new = (old & ~mask) | (val & mask);
+ outb(new, chip->port + 0x05);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Mixer reg %02x was %02x, set to %02x\n",
+ reg, old, new);
+#endif
+ }
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+ return oval;
+}
+
+static inline int snd_es18xx_mixer_writable(struct snd_es18xx *chip, unsigned char reg,
+ unsigned char mask)
+{
+ int old, expected, new;
+ unsigned long flags;
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ outb(reg, chip->port + 0x04);
+ old = inb(chip->port + 0x05);
+ expected = old ^ mask;
+ outb(expected, chip->port + 0x05);
+ new = inb(chip->port + 0x05);
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Mixer reg %02x was %02x, set to %02x, now is %02x\n",
+ reg, old, expected, new);
+#endif
+ return expected == new;
+}
+
+
+static int snd_es18xx_reset(struct snd_es18xx *chip)
+{
+ int i;
+ outb(0x03, chip->port + 0x06);
+ inb(chip->port + 0x06);
+ outb(0x00, chip->port + 0x06);
+ for(i = 0; i < MILLISECOND && !(inb(chip->port + 0x0E) & 0x80); i++);
+ if (inb(chip->port + 0x0A) != 0xAA)
+ return -1;
+ return 0;
+}
+
+static int snd_es18xx_reset_fifo(struct snd_es18xx *chip)
+{
+ outb(0x02, chip->port + 0x06);
+ inb(chip->port + 0x06);
+ outb(0x00, chip->port + 0x06);
+ return 0;
+}
+
+static struct snd_ratnum new_clocks[2] = {
+ {
+ .num = 793800,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ },
+ {
+ .num = 768000,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ }
+};
+
+static struct snd_pcm_hw_constraint_ratnums new_hw_constraints_clocks = {
+ .nrats = 2,
+ .rats = new_clocks,
+};
+
+static struct snd_ratnum old_clocks[2] = {
+ {
+ .num = 795444,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ },
+ {
+ .num = 397722,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ }
+};
+
+static struct snd_pcm_hw_constraint_ratnums old_hw_constraints_clocks = {
+ .nrats = 2,
+ .rats = old_clocks,
+};
+
+
+static void snd_es18xx_rate_set(struct snd_es18xx *chip,
+ struct snd_pcm_substream *substream,
+ int mode)
+{
+ unsigned int bits, div0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (chip->caps & ES18XX_NEW_RATE) {
+ if (runtime->rate_num == new_clocks[0].num)
+ bits = 128 - runtime->rate_den;
+ else
+ bits = 256 - runtime->rate_den;
+ } else {
+ if (runtime->rate_num == old_clocks[0].num)
+ bits = 256 - runtime->rate_den;
+ else
+ bits = 128 - runtime->rate_den;
+ }
+
+ /* set filter register */
+ div0 = 256 - 7160000*20/(8*82*runtime->rate);
+
+ if ((chip->caps & ES18XX_PCM2) && mode == DAC2) {
+ snd_es18xx_mixer_write(chip, 0x70, bits);
+ /*
+ * Comment from kernel oss driver:
+ * FKS: fascinating: 0x72 doesn't seem to work.
+ */
+ snd_es18xx_write(chip, 0xA2, div0);
+ snd_es18xx_mixer_write(chip, 0x72, div0);
+ } else {
+ snd_es18xx_write(chip, 0xA1, bits);
+ snd_es18xx_write(chip, 0xA2, div0);
+ }
+}
+
+static int snd_es18xx_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+ int shift, err;
+
+ shift = 0;
+ if (params_channels(hw_params) == 2)
+ shift++;
+ if (snd_pcm_format_width(params_format(hw_params)) == 16)
+ shift++;
+
+ if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) {
+ if ((chip->caps & ES18XX_DUPLEX_MONO) &&
+ (chip->capture_a_substream) &&
+ params_channels(hw_params) != 1) {
+ _snd_pcm_hw_param_setempty(hw_params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ return -EBUSY;
+ }
+ chip->dma2_shift = shift;
+ } else {
+ chip->dma1_shift = shift;
+ }
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_es18xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_es18xx_playback1_prepare(struct snd_es18xx *chip,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma2_size = size;
+
+ snd_es18xx_rate_set(chip, substream, DAC2);
+
+ /* Transfer Count Reload */
+ count = 0x10000 - count;
+ snd_es18xx_mixer_write(chip, 0x74, count & 0xff);
+ snd_es18xx_mixer_write(chip, 0x76, count >> 8);
+
+ /* Set format */
+ snd_es18xx_mixer_bits(chip, 0x7A, 0x07,
+ ((runtime->channels == 1) ? 0x00 : 0x02) |
+ (snd_pcm_format_width(runtime->format) == 16 ? 0x01 : 0x00) |
+ (snd_pcm_format_unsigned(runtime->format) ? 0x00 : 0x04));
+
+ /* Set DMA controller */
+ snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT);
+
+ return 0;
+}
+
+static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip,
+ struct snd_pcm_substream *substream,
+ int cmd)
+{
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (chip->active & DAC2)
+ return 0;
+ chip->active |= DAC2;
+ /* Start DMA */
+ if (chip->dma2 >= 4)
+ snd_es18xx_mixer_write(chip, 0x78, 0xb3);
+ else
+ snd_es18xx_mixer_write(chip, 0x78, 0x93);
+#ifdef AVOID_POPS
+ /* Avoid pops */
+ udelay(100000);
+ if (chip->caps & ES18XX_PCM2)
+ /* Restore Audio 2 volume */
+ snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol);
+ else
+ /* Enable PCM output */
+ snd_es18xx_dsp_command(chip, 0xD1);
+#endif
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (!(chip->active & DAC2))
+ return 0;
+ chip->active &= ~DAC2;
+ /* Stop DMA */
+ snd_es18xx_mixer_write(chip, 0x78, 0x00);
+#ifdef AVOID_POPS
+ udelay(25000);
+ if (chip->caps & ES18XX_PCM2)
+ /* Set Audio 2 volume to 0 */
+ snd_es18xx_mixer_write(chip, 0x7C, 0);
+ else
+ /* Disable PCM output */
+ snd_es18xx_dsp_command(chip, 0xD3);
+#endif
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int snd_es18xx_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+ int shift, err;
+
+ shift = 0;
+ if ((chip->caps & ES18XX_DUPLEX_MONO) &&
+ chip->playback_a_substream &&
+ params_channels(hw_params) != 1) {
+ _snd_pcm_hw_param_setempty(hw_params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ return -EBUSY;
+ }
+ if (params_channels(hw_params) == 2)
+ shift++;
+ if (snd_pcm_format_width(params_format(hw_params)) == 16)
+ shift++;
+ chip->dma1_shift = shift;
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma1_size = size;
+
+ snd_es18xx_reset_fifo(chip);
+
+ /* Set stereo/mono */
+ snd_es18xx_bits(chip, 0xA8, 0x03, runtime->channels == 1 ? 0x02 : 0x01);
+
+ snd_es18xx_rate_set(chip, substream, ADC1);
+
+ /* Transfer Count Reload */
+ count = 0x10000 - count;
+ snd_es18xx_write(chip, 0xA4, count & 0xff);
+ snd_es18xx_write(chip, 0xA5, count >> 8);
+
+#ifdef AVOID_POPS
+ udelay(100000);
+#endif
+
+ /* Set format */
+ snd_es18xx_write(chip, 0xB7,
+ snd_pcm_format_unsigned(runtime->format) ? 0x51 : 0x71);
+ snd_es18xx_write(chip, 0xB7, 0x90 |
+ ((runtime->channels == 1) ? 0x40 : 0x08) |
+ (snd_pcm_format_width(runtime->format) == 16 ? 0x04 : 0x00) |
+ (snd_pcm_format_unsigned(runtime->format) ? 0x00 : 0x20));
+
+ /* Set DMA controller */
+ snd_dma_program(chip->dma1, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT);
+
+ return 0;
+}
+
+static int snd_es18xx_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (chip->active & ADC1)
+ return 0;
+ chip->active |= ADC1;
+ /* Start DMA */
+ snd_es18xx_write(chip, 0xB8, 0x0f);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (!(chip->active & ADC1))
+ return 0;
+ chip->active &= ~ADC1;
+ /* Stop DMA */
+ snd_es18xx_write(chip, 0xB8, 0x00);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int snd_es18xx_playback2_prepare(struct snd_es18xx *chip,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma1_size = size;
+
+ snd_es18xx_reset_fifo(chip);
+
+ /* Set stereo/mono */
+ snd_es18xx_bits(chip, 0xA8, 0x03, runtime->channels == 1 ? 0x02 : 0x01);
+
+ snd_es18xx_rate_set(chip, substream, DAC1);
+
+ /* Transfer Count Reload */
+ count = 0x10000 - count;
+ snd_es18xx_write(chip, 0xA4, count & 0xff);
+ snd_es18xx_write(chip, 0xA5, count >> 8);
+
+ /* Set format */
+ snd_es18xx_write(chip, 0xB6,
+ snd_pcm_format_unsigned(runtime->format) ? 0x80 : 0x00);
+ snd_es18xx_write(chip, 0xB7,
+ snd_pcm_format_unsigned(runtime->format) ? 0x51 : 0x71);
+ snd_es18xx_write(chip, 0xB7, 0x90 |
+ (runtime->channels == 1 ? 0x40 : 0x08) |
+ (snd_pcm_format_width(runtime->format) == 16 ? 0x04 : 0x00) |
+ (snd_pcm_format_unsigned(runtime->format) ? 0x00 : 0x20));
+
+ /* Set DMA controller */
+ snd_dma_program(chip->dma1, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT);
+
+ return 0;
+}
+
+static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip,
+ struct snd_pcm_substream *substream,
+ int cmd)
+{
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (chip->active & DAC1)
+ return 0;
+ chip->active |= DAC1;
+ /* Start DMA */
+ snd_es18xx_write(chip, 0xB8, 0x05);
+#ifdef AVOID_POPS
+ /* Avoid pops */
+ udelay(100000);
+ /* Enable Audio 1 */
+ snd_es18xx_dsp_command(chip, 0xD1);
+#endif
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (!(chip->active & DAC1))
+ return 0;
+ chip->active &= ~DAC1;
+ /* Stop DMA */
+ snd_es18xx_write(chip, 0xB8, 0x00);
+#ifdef AVOID_POPS
+ /* Avoid pops */
+ udelay(25000);
+ /* Disable Audio 1 */
+ snd_es18xx_dsp_command(chip, 0xD3);
+#endif
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int snd_es18xx_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+ if (substream->number == 0 && (chip->caps & ES18XX_PCM2))
+ return snd_es18xx_playback1_prepare(chip, substream);
+ else
+ return snd_es18xx_playback2_prepare(chip, substream);
+}
+
+static int snd_es18xx_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+ if (substream->number == 0 && (chip->caps & ES18XX_PCM2))
+ return snd_es18xx_playback1_trigger(chip, substream, cmd);
+ else
+ return snd_es18xx_playback2_trigger(chip, substream, cmd);
+}
+
+static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id)
+{
+ struct snd_es18xx *chip = dev_id;
+ unsigned char status;
+
+ if (chip->caps & ES18XX_CONTROL) {
+ /* Read Interrupt status */
+ status = inb(chip->ctrl_port + 6);
+ } else {
+ /* Read Interrupt status */
+ status = snd_es18xx_mixer_read(chip, 0x7f) >> 4;
+ }
+#if 0
+ else {
+ status = 0;
+ if (inb(chip->port + 0x0C) & 0x01)
+ status |= AUDIO1_IRQ;
+ if (snd_es18xx_mixer_read(chip, 0x7A) & 0x80)
+ status |= AUDIO2_IRQ;
+ if ((chip->caps & ES18XX_HWV) &&
+ snd_es18xx_mixer_read(chip, 0x64) & 0x10)
+ status |= HWV_IRQ;
+ }
+#endif
+
+ /* Audio 1 & Audio 2 */
+ if (status & AUDIO2_IRQ) {
+ if (chip->active & DAC2)
+ snd_pcm_period_elapsed(chip->playback_a_substream);
+ /* ack interrupt */
+ snd_es18xx_mixer_bits(chip, 0x7A, 0x80, 0x00);
+ }
+ if (status & AUDIO1_IRQ) {
+ /* ok.. capture is active */
+ if (chip->active & ADC1)
+ snd_pcm_period_elapsed(chip->capture_a_substream);
+ /* ok.. playback2 is active */
+ else if (chip->active & DAC1)
+ snd_pcm_period_elapsed(chip->playback_b_substream);
+ /* ack interrupt */
+ inb(chip->port + 0x0E);
+ }
+
+ /* MPU */
+ if ((status & MPU_IRQ) && chip->rmidi)
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+
+ /* Hardware volume */
+ if (status & HWV_IRQ) {
+ int split = 0;
+ if (chip->caps & ES18XX_HWV) {
+ split = snd_es18xx_mixer_read(chip, 0x64) & 0x80;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id);
+ }
+ if (!split) {
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id);
+ }
+ /* ack interrupt */
+ snd_es18xx_mixer_write(chip, 0x66, 0x00);
+ }
+ return IRQ_HANDLED;
+}
+
+static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+ int pos;
+
+ if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) {
+ if (!(chip->active & DAC2))
+ return 0;
+ pos = snd_dma_pointer(chip->dma2, chip->dma2_size);
+ return pos >> chip->dma2_shift;
+ } else {
+ if (!(chip->active & DAC1))
+ return 0;
+ pos = snd_dma_pointer(chip->dma1, chip->dma1_size);
+ return pos >> chip->dma1_shift;
+ }
+}
+
+static snd_pcm_uframes_t snd_es18xx_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+ int pos;
+
+ if (!(chip->active & ADC1))
+ return 0;
+ pos = snd_dma_pointer(chip->dma1, chip->dma1_size);
+ return pos >> chip->dma1_shift;
+}
+
+static struct snd_pcm_hardware snd_es18xx_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_es18xx_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_es18xx_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+
+ if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) {
+ if ((chip->caps & ES18XX_DUPLEX_MONO) &&
+ chip->capture_a_substream &&
+ chip->capture_a_substream->runtime->channels != 1)
+ return -EAGAIN;
+ chip->playback_a_substream = substream;
+ } else if (substream->number <= 1) {
+ if (chip->capture_a_substream)
+ return -EAGAIN;
+ chip->playback_b_substream = substream;
+ } else {
+ snd_BUG();
+ return -EINVAL;
+ }
+ substream->runtime->hw = snd_es18xx_playback;
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ (chip->caps & ES18XX_NEW_RATE) ? &new_hw_constraints_clocks : &old_hw_constraints_clocks);
+ return 0;
+}
+
+static int snd_es18xx_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+
+ if (chip->playback_b_substream)
+ return -EAGAIN;
+ if ((chip->caps & ES18XX_DUPLEX_MONO) &&
+ chip->playback_a_substream &&
+ chip->playback_a_substream->runtime->channels != 1)
+ return -EAGAIN;
+ chip->capture_a_substream = substream;
+ substream->runtime->hw = snd_es18xx_capture;
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ (chip->caps & ES18XX_NEW_RATE) ? &new_hw_constraints_clocks : &old_hw_constraints_clocks);
+ return 0;
+}
+
+static int snd_es18xx_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+
+ if (substream->number == 0 && (chip->caps & ES18XX_PCM2))
+ chip->playback_a_substream = NULL;
+ else
+ chip->playback_b_substream = NULL;
+
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_es18xx_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_es18xx *chip = snd_pcm_substream_chip(substream);
+
+ chip->capture_a_substream = NULL;
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+/*
+ * MIXER part
+ */
+
+/* Record source mux routines:
+ * Depending on the chipset this mux switches between 4, 5, or 8 possible inputs.
+ * bit table for the 4/5 source mux:
+ * reg 1C:
+ * b2 b1 b0 muxSource
+ * x 0 x microphone
+ * 0 1 x CD
+ * 1 1 0 line
+ * 1 1 1 mixer
+ * if it's "mixer" and it's a 5 source mux chipset then reg 7A bit 3 determines
+ * either the play mixer or the capture mixer.
+ *
+ * "map4Source" translates from source number to reg bit pattern
+ * "invMap4Source" translates from reg bit pattern to source number
+ */
+
+static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts4Source[4] = {
+ "Mic", "CD", "Line", "Master"
+ };
+ static char *texts5Source[5] = {
+ "Mic", "CD", "Line", "Master", "Mix"
+ };
+ static char *texts8Source[8] = {
+ "Mic", "Mic Master", "CD", "AOUT",
+ "Mic1", "Mix", "Line", "Master"
+ };
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ switch (chip->version) {
+ case 0x1868:
+ case 0x1878:
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts4Source[uinfo->value.enumerated.item]);
+ break;
+ case 0x1887:
+ case 0x1888:
+ uinfo->value.enumerated.items = 5;
+ if (uinfo->value.enumerated.item > 4)
+ uinfo->value.enumerated.item = 4;
+ strcpy(uinfo->value.enumerated.name, texts5Source[uinfo->value.enumerated.item]);
+ break;
+ case 0x1869: /* DS somewhat contradictory for 1869: could be be 5 or 8 */
+ case 0x1879:
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item > 7)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts8Source[uinfo->value.enumerated.item]);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int snd_es18xx_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ static unsigned char invMap4Source[8] = {0, 0, 1, 1, 0, 0, 2, 3};
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ int muxSource = snd_es18xx_mixer_read(chip, 0x1c) & 0x07;
+ if (!(chip->version == 0x1869 || chip->version == 0x1879)) {
+ muxSource = invMap4Source[muxSource];
+ if (muxSource==3 &&
+ (chip->version == 0x1887 || chip->version == 0x1888) &&
+ (snd_es18xx_mixer_read(chip, 0x7a) & 0x08)
+ )
+ muxSource = 4;
+ }
+ ucontrol->value.enumerated.item[0] = muxSource;
+ return 0;
+}
+
+static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ static unsigned char map4Source[4] = {0, 2, 6, 7};
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char val = ucontrol->value.enumerated.item[0];
+ unsigned char retVal = 0;
+
+ switch (chip->version) {
+ /* 5 source chips */
+ case 0x1887:
+ case 0x1888:
+ if (val > 4)
+ return -EINVAL;
+ if (val == 4) {
+ retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x08) != 0x08;
+ val = 3;
+ } else
+ retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x00) != 0x00;
+ /* 4 source chips */
+ case 0x1868:
+ case 0x1878:
+ if (val > 3)
+ return -EINVAL;
+ val = map4Source[val];
+ break;
+ /* 8 source chips */
+ case 0x1869:
+ case 0x1879:
+ if (val > 7)
+ return -EINVAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return (snd_es18xx_mixer_bits(chip, 0x1c, 0x07, val) != val) || retVal;
+}
+
+#define snd_es18xx_info_spatializer_enable snd_ctl_boolean_mono_info
+
+static int snd_es18xx_get_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char val = snd_es18xx_mixer_read(chip, 0x50);
+ ucontrol->value.integer.value[0] = !!(val & 8);
+ return 0;
+}
+
+static int snd_es18xx_put_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char oval, nval;
+ int change;
+ nval = ucontrol->value.integer.value[0] ? 0x0c : 0x04;
+ oval = snd_es18xx_mixer_read(chip, 0x50) & 0x0c;
+ change = nval != oval;
+ if (change) {
+ snd_es18xx_mixer_write(chip, 0x50, nval & ~0x04);
+ snd_es18xx_mixer_write(chip, 0x50, nval);
+ }
+ return change;
+}
+
+static int snd_es18xx_info_hw_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 63;
+ return 0;
+}
+
+static int snd_es18xx_get_hw_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = snd_es18xx_mixer_read(chip, 0x61) & 0x3f;
+ ucontrol->value.integer.value[1] = snd_es18xx_mixer_read(chip, 0x63) & 0x3f;
+ return 0;
+}
+
+#define snd_es18xx_info_hw_switch snd_ctl_boolean_stereo_info
+
+static int snd_es18xx_get_hw_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = !(snd_es18xx_mixer_read(chip, 0x61) & 0x40);
+ ucontrol->value.integer.value[1] = !(snd_es18xx_mixer_read(chip, 0x63) & 0x40);
+ return 0;
+}
+
+static void snd_es18xx_hwv_free(struct snd_kcontrol *kcontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ chip->master_volume = NULL;
+ chip->master_switch = NULL;
+ chip->hw_volume = NULL;
+ chip->hw_switch = NULL;
+}
+
+static int snd_es18xx_reg_bits(struct snd_es18xx *chip, unsigned char reg,
+ unsigned char mask, unsigned char val)
+{
+ if (reg < 0xa0)
+ return snd_es18xx_mixer_bits(chip, reg, mask, val);
+ else
+ return snd_es18xx_bits(chip, reg, mask, val);
+}
+
+static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg)
+{
+ if (reg < 0xa0)
+ return snd_es18xx_mixer_read(chip, reg);
+ else
+ return snd_es18xx_read(chip, reg);
+}
+
+#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_es18xx_info_single, \
+ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int val;
+
+ val = snd_es18xx_reg_read(chip, reg);
+ ucontrol->value.integer.value[0] = (val >> shift) & mask;
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ unsigned char val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ mask <<= shift;
+ val <<= shift;
+ return snd_es18xx_reg_bits(chip, reg, mask, val) != val;
+}
+
+#define ES18XX_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_es18xx_info_double, \
+ .get = snd_es18xx_get_double, .put = snd_es18xx_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) }
+
+static int snd_es18xx_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_es18xx_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ unsigned char left, right;
+
+ left = snd_es18xx_reg_read(chip, left_reg);
+ if (left_reg != right_reg)
+ right = snd_es18xx_reg_read(chip, right_reg);
+ else
+ right = left;
+ ucontrol->value.integer.value[0] = (left >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (right >> shift_right) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_es18xx_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned char val1, val2, mask1, mask2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ mask1 = mask << shift_left;
+ mask2 = mask << shift_right;
+ if (left_reg != right_reg) {
+ change = 0;
+ if (snd_es18xx_reg_bits(chip, left_reg, mask1, val1) != val1)
+ change = 1;
+ if (snd_es18xx_reg_bits(chip, right_reg, mask2, val2) != val2)
+ change = 1;
+ } else {
+ change = (snd_es18xx_reg_bits(chip, left_reg, mask1 | mask2,
+ val1 | val2) != (val1 | val2));
+ }
+ return change;
+}
+
+/* Mixer controls
+ * These arrays contain setup data for mixer controls.
+ *
+ * The controls that are universal to all chipsets are fully initialized
+ * here.
+ */
+static struct snd_kcontrol_new snd_es18xx_base_controls[] = {
+ES18XX_DOUBLE("Master Playback Volume", 0, 0x60, 0x62, 0, 0, 63, 0),
+ES18XX_DOUBLE("Master Playback Switch", 0, 0x60, 0x62, 6, 6, 1, 1),
+ES18XX_DOUBLE("Line Playback Volume", 0, 0x3e, 0x3e, 4, 0, 15, 0),
+ES18XX_DOUBLE("CD Playback Volume", 0, 0x38, 0x38, 4, 0, 15, 0),
+ES18XX_DOUBLE("FM Playback Volume", 0, 0x36, 0x36, 4, 0, 15, 0),
+ES18XX_DOUBLE("Mic Playback Volume", 0, 0x1a, 0x1a, 4, 0, 15, 0),
+ES18XX_DOUBLE("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0),
+ES18XX_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0),
+ES18XX_DOUBLE("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_es18xx_info_mux,
+ .get = snd_es18xx_get_mux,
+ .put = snd_es18xx_put_mux,
+}
+};
+
+static struct snd_kcontrol_new snd_es18xx_recmix_controls[] = {
+ES18XX_DOUBLE("PCM Capture Volume", 0, 0x69, 0x69, 4, 0, 15, 0),
+ES18XX_DOUBLE("Mic Capture Volume", 0, 0x68, 0x68, 4, 0, 15, 0),
+ES18XX_DOUBLE("Line Capture Volume", 0, 0x6e, 0x6e, 4, 0, 15, 0),
+ES18XX_DOUBLE("FM Capture Volume", 0, 0x6b, 0x6b, 4, 0, 15, 0),
+ES18XX_DOUBLE("CD Capture Volume", 0, 0x6a, 0x6a, 4, 0, 15, 0),
+ES18XX_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0)
+};
+
+/*
+ * The chipset specific mixer controls
+ */
+static struct snd_kcontrol_new snd_es18xx_opt_speaker =
+ ES18XX_SINGLE("PC Speaker Playback Volume", 0, 0x3c, 0, 7, 0);
+
+static struct snd_kcontrol_new snd_es18xx_opt_1869[] = {
+ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1),
+ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0),
+ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0),
+ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0)
+};
+
+static struct snd_kcontrol_new snd_es18xx_opt_1878 =
+ ES18XX_DOUBLE("Video Playback Volume", 0, 0x68, 0x68, 4, 0, 15, 0);
+
+static struct snd_kcontrol_new snd_es18xx_opt_1879[] = {
+ES18XX_SINGLE("Video Playback Switch", 0, 0x71, 6, 1, 0),
+ES18XX_DOUBLE("Video Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0),
+ES18XX_DOUBLE("Video Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0)
+};
+
+static struct snd_kcontrol_new snd_es18xx_pcm1_controls[] = {
+ES18XX_DOUBLE("PCM Playback Volume", 0, 0x14, 0x14, 4, 0, 15, 0),
+};
+
+static struct snd_kcontrol_new snd_es18xx_pcm2_controls[] = {
+ES18XX_DOUBLE("PCM Playback Volume", 0, 0x7c, 0x7c, 4, 0, 15, 0),
+ES18XX_DOUBLE("PCM Playback Volume", 1, 0x14, 0x14, 4, 0, 15, 0)
+};
+
+static struct snd_kcontrol_new snd_es18xx_spatializer_controls[] = {
+ES18XX_SINGLE("3D Control - Level", 0, 0x52, 0, 63, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "3D Control - Switch",
+ .info = snd_es18xx_info_spatializer_enable,
+ .get = snd_es18xx_get_spatializer_enable,
+ .put = snd_es18xx_put_spatializer_enable,
+}
+};
+
+static struct snd_kcontrol_new snd_es18xx_micpre1_control =
+ES18XX_SINGLE("Mic Boost (+26dB)", 0, 0xa9, 2, 1, 0);
+
+static struct snd_kcontrol_new snd_es18xx_micpre2_control =
+ES18XX_SINGLE("Mic Boost (+26dB)", 0, 0x7d, 3, 1, 0);
+
+static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Hardware Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_es18xx_info_hw_volume,
+ .get = snd_es18xx_get_hw_volume,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Hardware Master Playback Switch",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_es18xx_info_hw_switch,
+ .get = snd_es18xx_get_hw_switch,
+},
+ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0),
+};
+
+static int __devinit snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg)
+{
+ int data;
+ unsigned long flags;
+ spin_lock_irqsave(&chip->ctrl_lock, flags);
+ outb(reg, chip->ctrl_port);
+ data = inb(chip->ctrl_port + 1);
+ spin_unlock_irqrestore(&chip->ctrl_lock, flags);
+ return data;
+}
+
+static void __devinit snd_es18xx_config_write(struct snd_es18xx *chip,
+ unsigned char reg, unsigned char data)
+{
+ /* No need for spinlocks, this function is used only in
+ otherwise protected init code */
+ outb(reg, chip->ctrl_port);
+ outb(data, chip->ctrl_port + 1);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Config reg %02x set to %02x\n", reg, data);
+#endif
+}
+
+static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip)
+{
+ int mask = 0;
+
+ /* enable extended mode */
+ snd_es18xx_dsp_command(chip, 0xC6);
+ /* Reset mixer registers */
+ snd_es18xx_mixer_write(chip, 0x00, 0x00);
+
+ /* Audio 1 DMA demand mode (4 bytes/request) */
+ snd_es18xx_write(chip, 0xB9, 2);
+ if (chip->caps & ES18XX_CONTROL) {
+ /* Hardware volume IRQ */
+ snd_es18xx_config_write(chip, 0x27, chip->irq);
+ if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) {
+ /* FM I/O */
+ snd_es18xx_config_write(chip, 0x62, chip->fm_port >> 8);
+ snd_es18xx_config_write(chip, 0x63, chip->fm_port & 0xff);
+ }
+ if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) {
+ /* MPU-401 I/O */
+ snd_es18xx_config_write(chip, 0x64, chip->mpu_port >> 8);
+ snd_es18xx_config_write(chip, 0x65, chip->mpu_port & 0xff);
+ /* MPU-401 IRQ */
+ snd_es18xx_config_write(chip, 0x28, chip->irq);
+ }
+ /* Audio1 IRQ */
+ snd_es18xx_config_write(chip, 0x70, chip->irq);
+ /* Audio2 IRQ */
+ snd_es18xx_config_write(chip, 0x72, chip->irq);
+ /* Audio1 DMA */
+ snd_es18xx_config_write(chip, 0x74, chip->dma1);
+ /* Audio2 DMA */
+ snd_es18xx_config_write(chip, 0x75, chip->dma2);
+
+ /* Enable Audio 1 IRQ */
+ snd_es18xx_write(chip, 0xB1, 0x50);
+ /* Enable Audio 2 IRQ */
+ snd_es18xx_mixer_write(chip, 0x7A, 0x40);
+ /* Enable Audio 1 DMA */
+ snd_es18xx_write(chip, 0xB2, 0x50);
+ /* Enable MPU and hardware volume interrupt */
+ snd_es18xx_mixer_write(chip, 0x64, 0x42);
+ /* Enable ESS wavetable input */
+ snd_es18xx_mixer_bits(chip, 0x48, 0x10, 0x10);
+ }
+ else {
+ int irqmask, dma1mask, dma2mask;
+ switch (chip->irq) {
+ case 2:
+ case 9:
+ irqmask = 0;
+ break;
+ case 5:
+ irqmask = 1;
+ break;
+ case 7:
+ irqmask = 2;
+ break;
+ case 10:
+ irqmask = 3;
+ break;
+ default:
+ snd_printk(KERN_ERR "invalid irq %d\n", chip->irq);
+ return -ENODEV;
+ }
+ switch (chip->dma1) {
+ case 0:
+ dma1mask = 1;
+ break;
+ case 1:
+ dma1mask = 2;
+ break;
+ case 3:
+ dma1mask = 3;
+ break;
+ default:
+ snd_printk(KERN_ERR "invalid dma1 %d\n", chip->dma1);
+ return -ENODEV;
+ }
+ switch (chip->dma2) {
+ case 0:
+ dma2mask = 0;
+ break;
+ case 1:
+ dma2mask = 1;
+ break;
+ case 3:
+ dma2mask = 2;
+ break;
+ case 5:
+ dma2mask = 3;
+ break;
+ default:
+ snd_printk(KERN_ERR "invalid dma2 %d\n", chip->dma2);
+ return -ENODEV;
+ }
+
+ /* Enable and set Audio 1 IRQ */
+ snd_es18xx_write(chip, 0xB1, 0x50 | (irqmask << 2));
+ /* Enable and set Audio 1 DMA */
+ snd_es18xx_write(chip, 0xB2, 0x50 | (dma1mask << 2));
+ /* Set Audio 2 DMA */
+ snd_es18xx_mixer_bits(chip, 0x7d, 0x07, 0x04 | dma2mask);
+ /* Enable Audio 2 IRQ and DMA
+ Set capture mixer input */
+ snd_es18xx_mixer_write(chip, 0x7A, 0x68);
+ /* Enable and set hardware volume interrupt */
+ snd_es18xx_mixer_write(chip, 0x64, 0x06);
+ if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) {
+ /* MPU401 share irq with audio
+ Joystick enabled
+ FM enabled */
+ snd_es18xx_mixer_write(chip, 0x40, 0x43 | (chip->mpu_port & 0xf0) >> 1);
+ }
+ snd_es18xx_mixer_write(chip, 0x7f, ((irqmask + 1) << 1) | 0x01);
+ }
+ if (chip->caps & ES18XX_NEW_RATE) {
+ /* Change behaviour of register A1
+ 4x oversampling
+ 2nd channel DAC asynchronous */
+ snd_es18xx_mixer_write(chip, 0x71, 0x32);
+ }
+ if (!(chip->caps & ES18XX_PCM2)) {
+ /* Enable DMA FIFO */
+ snd_es18xx_write(chip, 0xB7, 0x80);
+ }
+ if (chip->caps & ES18XX_SPATIALIZER) {
+ /* Set spatializer parameters to recommended values */
+ snd_es18xx_mixer_write(chip, 0x54, 0x8f);
+ snd_es18xx_mixer_write(chip, 0x56, 0x95);
+ snd_es18xx_mixer_write(chip, 0x58, 0x94);
+ snd_es18xx_mixer_write(chip, 0x5a, 0x80);
+ }
+ /* Flip the "enable I2S" bits for those chipsets that need it */
+ switch (chip->version) {
+ case 0x1879:
+ //Leaving I2S enabled on the 1879 screws up the PCM playback (rate effected somehow)
+ //so a Switch control has been added to toggle this 0x71 bit on/off:
+ //snd_es18xx_mixer_bits(chip, 0x71, 0x40, 0x40);
+ /* Note: we fall through on purpose here. */
+ case 0x1878:
+ snd_es18xx_config_write(chip, 0x29, snd_es18xx_config_read(chip, 0x29) | 0x40);
+ break;
+ }
+ /* Mute input source */
+ if (chip->caps & ES18XX_MUTEREC)
+ mask = 0x10;
+ if (chip->caps & ES18XX_RECMIX)
+ snd_es18xx_mixer_write(chip, 0x1c, 0x05 | mask);
+ else {
+ snd_es18xx_mixer_write(chip, 0x1c, 0x00 | mask);
+ snd_es18xx_write(chip, 0xb4, 0x00);
+ }
+#ifndef AVOID_POPS
+ /* Enable PCM output */
+ snd_es18xx_dsp_command(chip, 0xD1);
+#endif
+
+ return 0;
+}
+
+static int __devinit snd_es18xx_identify(struct snd_es18xx *chip)
+{
+ int hi,lo;
+
+ /* reset */
+ if (snd_es18xx_reset(chip) < 0) {
+ snd_printk(KERN_ERR "reset at 0x%lx failed!!!\n", chip->port);
+ return -ENODEV;
+ }
+
+ snd_es18xx_dsp_command(chip, 0xe7);
+ hi = snd_es18xx_dsp_get_byte(chip);
+ if (hi < 0) {
+ return hi;
+ }
+ lo = snd_es18xx_dsp_get_byte(chip);
+ if ((lo & 0xf0) != 0x80) {
+ return -ENODEV;
+ }
+ if (hi == 0x48) {
+ chip->version = 0x488;
+ return 0;
+ }
+ if (hi != 0x68) {
+ return -ENODEV;
+ }
+ if ((lo & 0x0f) < 8) {
+ chip->version = 0x688;
+ return 0;
+ }
+
+ outb(0x40, chip->port + 0x04);
+ udelay(10);
+ hi = inb(chip->port + 0x05);
+ udelay(10);
+ lo = inb(chip->port + 0x05);
+ if (hi != lo) {
+ chip->version = hi << 8 | lo;
+ chip->ctrl_port = inb(chip->port + 0x05) << 8;
+ udelay(10);
+ chip->ctrl_port += inb(chip->port + 0x05);
+
+ if ((chip->res_ctrl_port = request_region(chip->ctrl_port, 8, "ES18xx - CTRL")) == NULL) {
+ snd_printk(KERN_ERR PFX "unable go grab port 0x%lx\n", chip->ctrl_port);
+ return -EBUSY;
+ }
+
+ return 0;
+ }
+
+ /* If has Hardware volume */
+ if (snd_es18xx_mixer_writable(chip, 0x64, 0x04)) {
+ /* If has Audio2 */
+ if (snd_es18xx_mixer_writable(chip, 0x70, 0x7f)) {
+ /* If has volume count */
+ if (snd_es18xx_mixer_writable(chip, 0x64, 0x20)) {
+ chip->version = 0x1887;
+ } else {
+ chip->version = 0x1888;
+ }
+ } else {
+ chip->version = 0x1788;
+ }
+ }
+ else
+ chip->version = 0x1688;
+ return 0;
+}
+
+static int __devinit snd_es18xx_probe(struct snd_es18xx *chip)
+{
+ if (snd_es18xx_identify(chip) < 0) {
+ snd_printk(KERN_ERR PFX "[0x%lx] ESS chip not found\n", chip->port);
+ return -ENODEV;
+ }
+
+ switch (chip->version) {
+ case 0x1868:
+ chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL;
+ break;
+ case 0x1869:
+ chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV;
+ break;
+ case 0x1878:
+ chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL;
+ break;
+ case 0x1879:
+ chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_I2S | ES18XX_CONTROL | ES18XX_HWV;
+ break;
+ case 0x1887:
+ chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME;
+ break;
+ case 0x1888:
+ chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME;
+ break;
+ default:
+ snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n",
+ chip->port, chip->version);
+ return -ENODEV;
+ }
+
+ snd_printd("[0x%lx] ESS%x chip found\n", chip->port, chip->version);
+
+ if (chip->dma1 == chip->dma2)
+ chip->caps &= ~(ES18XX_PCM2 | ES18XX_DUPLEX_SAME);
+
+ return snd_es18xx_initialize(chip);
+}
+
+static struct snd_pcm_ops snd_es18xx_playback_ops = {
+ .open = snd_es18xx_playback_open,
+ .close = snd_es18xx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_es18xx_playback_hw_params,
+ .hw_free = snd_es18xx_pcm_hw_free,
+ .prepare = snd_es18xx_playback_prepare,
+ .trigger = snd_es18xx_playback_trigger,
+ .pointer = snd_es18xx_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_es18xx_capture_ops = {
+ .open = snd_es18xx_capture_open,
+ .close = snd_es18xx_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_es18xx_capture_hw_params,
+ .hw_free = snd_es18xx_pcm_hw_free,
+ .prepare = snd_es18xx_capture_prepare,
+ .trigger = snd_es18xx_capture_trigger,
+ .pointer = snd_es18xx_capture_pointer,
+};
+
+static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ char str[16];
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ sprintf(str, "ES%x", chip->version);
+ if (chip->caps & ES18XX_PCM2)
+ err = snd_pcm_new(chip->card, str, device, 2, 1, &pcm);
+ else
+ err = snd_pcm_new(chip->card, str, device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es18xx_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_es18xx_capture_ops);
+
+ /* global setup */
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ if (chip->caps & ES18XX_DUPLEX_SAME)
+ pcm->info_flags |= SNDRV_PCM_INFO_JOINT_DUPLEX;
+ if (! (chip->caps & ES18XX_PCM2))
+ pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX;
+ sprintf(pcm->name, "ESS AudioDrive ES%x", chip->version);
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024,
+ chip->dma1 > 3 || chip->dma2 > 3 ? 128*1024 : 64*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/* Power Management support functions */
+#ifdef CONFIG_PM
+static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state)
+{
+ struct snd_audiodrive *acard = card->private_data;
+ struct snd_es18xx *chip = acard->chip;
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+
+ snd_pcm_suspend_all(chip->pcm);
+
+ /* power down */
+ chip->pm_reg = (unsigned char)snd_es18xx_read(chip, ES18XX_PM);
+ chip->pm_reg |= (ES18XX_PM_FM | ES18XX_PM_SUS);
+ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg);
+ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_SUS);
+
+ return 0;
+}
+
+static int snd_es18xx_resume(struct snd_card *card)
+{
+ struct snd_audiodrive *acard = card->private_data;
+ struct snd_es18xx *chip = acard->chip;
+
+ /* restore PM register, we won't wake till (not 0x07) i/o activity though */
+ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_FM);
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static int snd_es18xx_free(struct snd_es18xx *chip)
+{
+ release_and_free_resource(chip->res_port);
+ release_and_free_resource(chip->res_ctrl_port);
+ release_and_free_resource(chip->res_mpu_port);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *) chip);
+ if (chip->dma1 >= 0) {
+ disable_dma(chip->dma1);
+ free_dma(chip->dma1);
+ }
+ if (chip->dma2 >= 0 && chip->dma1 != chip->dma2) {
+ disable_dma(chip->dma2);
+ free_dma(chip->dma2);
+ }
+ kfree(chip);
+ return 0;
+}
+
+static int snd_es18xx_dev_free(struct snd_device *device)
+{
+ struct snd_es18xx *chip = device->device_data;
+ return snd_es18xx_free(chip);
+}
+
+static int __devinit snd_es18xx_new_device(struct snd_card *card,
+ unsigned long port,
+ unsigned long mpu_port,
+ unsigned long fm_port,
+ int irq, int dma1, int dma2,
+ struct snd_es18xx ** rchip)
+{
+ struct snd_es18xx *chip;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_es18xx_dev_free,
+ };
+ int err;
+
+ *rchip = NULL;
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->mixer_lock);
+ spin_lock_init(&chip->ctrl_lock);
+ chip->card = card;
+ chip->port = port;
+ chip->mpu_port = mpu_port;
+ chip->fm_port = fm_port;
+ chip->irq = -1;
+ chip->dma1 = -1;
+ chip->dma2 = -1;
+ chip->audio2_vol = 0x00;
+ chip->active = 0;
+
+ if ((chip->res_port = request_region(port, 16, "ES18xx")) == NULL) {
+ snd_es18xx_free(chip);
+ snd_printk(KERN_ERR PFX "unable to grap ports 0x%lx-0x%lx\n", port, port + 16 - 1);
+ return -EBUSY;
+ }
+
+ if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", (void *) chip)) {
+ snd_es18xx_free(chip);
+ snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq);
+ return -EBUSY;
+ }
+ chip->irq = irq;
+
+ if (request_dma(dma1, "ES18xx DMA 1")) {
+ snd_es18xx_free(chip);
+ snd_printk(KERN_ERR PFX "unable to grap DMA1 %d\n", dma1);
+ return -EBUSY;
+ }
+ chip->dma1 = dma1;
+
+ if (dma2 != dma1 && request_dma(dma2, "ES18xx DMA 2")) {
+ snd_es18xx_free(chip);
+ snd_printk(KERN_ERR PFX "unable to grap DMA2 %d\n", dma2);
+ return -EBUSY;
+ }
+ chip->dma2 = dma2;
+
+ if (snd_es18xx_probe(chip) < 0) {
+ snd_es18xx_free(chip);
+ return -ENODEV;
+ }
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_es18xx_free(chip);
+ return err;
+ }
+ *rchip = chip;
+ return 0;
+}
+
+static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip)
+{
+ struct snd_card *card;
+ int err;
+ unsigned int idx;
+
+ card = chip->card;
+
+ strcpy(card->mixername, chip->pcm->name);
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_base_controls); idx++) {
+ struct snd_kcontrol *kctl;
+ kctl = snd_ctl_new1(&snd_es18xx_base_controls[idx], chip);
+ if (chip->caps & ES18XX_HWV) {
+ switch (idx) {
+ case 0:
+ chip->master_volume = kctl;
+ kctl->private_free = snd_es18xx_hwv_free;
+ break;
+ case 1:
+ chip->master_switch = kctl;
+ kctl->private_free = snd_es18xx_hwv_free;
+ break;
+ }
+ }
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ return err;
+ }
+ if (chip->caps & ES18XX_PCM2) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_pcm2_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_pcm2_controls[idx], chip))) < 0)
+ return err;
+ }
+ } else {
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_pcm1_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_pcm1_controls[idx], chip))) < 0)
+ return err;
+ }
+ }
+
+ if (chip->caps & ES18XX_RECMIX) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_recmix_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_recmix_controls[idx], chip))) < 0)
+ return err;
+ }
+ }
+ switch (chip->version) {
+ default:
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_micpre1_control, chip))) < 0)
+ return err;
+ break;
+ case 0x1869:
+ case 0x1879:
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_micpre2_control, chip))) < 0)
+ return err;
+ break;
+ }
+ if (chip->caps & ES18XX_SPATIALIZER) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_spatializer_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_spatializer_controls[idx], chip))) < 0)
+ return err;
+ }
+ }
+ if (chip->caps & ES18XX_HWV) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_hw_volume_controls); idx++) {
+ struct snd_kcontrol *kctl;
+ kctl = snd_ctl_new1(&snd_es18xx_hw_volume_controls[idx], chip);
+ if (idx == 0)
+ chip->hw_volume = kctl;
+ else
+ chip->hw_switch = kctl;
+ kctl->private_free = snd_es18xx_hwv_free;
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ return err;
+
+ }
+ }
+ /* finish initializing other chipset specific controls
+ */
+ if (chip->version != 0x1868) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_opt_speaker,
+ chip));
+ if (err < 0)
+ return err;
+ }
+ if (chip->version == 0x1869) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_1869); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_es18xx_opt_1869[idx],
+ chip));
+ if (err < 0)
+ return err;
+ }
+ } else if (chip->version == 0x1878) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_opt_1878,
+ chip));
+ if (err < 0)
+ return err;
+ } else if (chip->version == 0x1879) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_1879); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_es18xx_opt_1879[idx],
+ chip));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+
+/* Card level */
+
+MODULE_AUTHOR("Christian Fischbach <fishbach@pool.informatik.rwth-aachen.de>, Abramo Bagnara <abramo@alsa-project.org>");
+MODULE_DESCRIPTION("ESS ES18xx AudioDrive");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ESS,ES1868 PnP AudioDrive},"
+ "{ESS,ES1869 PnP AudioDrive},"
+ "{ESS,ES1878 PnP AudioDrive},"
+ "{ESS,ES1879 PnP AudioDrive},"
+ "{ESS,ES1887 PnP AudioDrive},"
+ "{ESS,ES1888 PnP AudioDrive},"
+ "{ESS,ES1887 AudioDrive},"
+ "{ESS,ES1888 AudioDrive}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */
+#ifndef CONFIG_PNP
+static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1};
+#else
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+#endif
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for ES18xx soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for ES18xx soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable ES18xx soundcard.");
+#ifdef CONFIG_PNP
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard.");
+#endif
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for ES18xx driver.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for ES18xx driver.");
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM port # for ES18xx driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for ES18xx driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA 1 # for ES18xx driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA 2 # for ES18xx driver.");
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnp_registered;
+static int pnpc_registered;
+
+static struct pnp_device_id snd_audiodrive_pnpbiosids[] = {
+ { .id = "ESS1869" },
+ { .id = "ESS1879" },
+ { .id = "" } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp, snd_audiodrive_pnpbiosids);
+
+/* PnP main device initialization */
+static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev)
+{
+ if (pnp_activate_dev(pdev) < 0) {
+ snd_printk(KERN_ERR PFX "PnP configure failure (out of resources?)\n");
+ return -EBUSY;
+ }
+ /* ok. hack using Vendor-Defined Card-Level registers */
+ /* skip csn and logdev initialization - already done in isapnp_configure */
+ if (pnp_device_is_isapnp(pdev)) {
+ isapnp_cfg_begin(isapnp_card_number(pdev), isapnp_csn_number(pdev));
+ isapnp_write_byte(0x27, pnp_irq(pdev, 0)); /* Hardware Volume IRQ Number */
+ if (mpu_port[dev] != SNDRV_AUTO_PORT)
+ isapnp_write_byte(0x28, pnp_irq(pdev, 0)); /* MPU-401 IRQ Number */
+ isapnp_write_byte(0x72, pnp_irq(pdev, 0)); /* second IRQ */
+ isapnp_cfg_end();
+ }
+ port[dev] = pnp_port_start(pdev, 0);
+ fm_port[dev] = pnp_port_start(pdev, 1);
+ mpu_port[dev] = pnp_port_start(pdev, 2);
+ dma1[dev] = pnp_dma(pdev, 0);
+ dma2[dev] = pnp_dma(pdev, 1);
+ irq[dev] = pnp_irq(pdev, 0);
+ snd_printdd("PnP ES18xx: port=0x%lx, fm port=0x%lx, mpu port=0x%lx\n", port[dev], fm_port[dev], mpu_port[dev]);
+ snd_printdd("PnP ES18xx: dma1=%i, dma2=%i, irq=%i\n", dma1[dev], dma2[dev], irq[dev]);
+ return 0;
+}
+
+static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard,
+ struct pnp_dev *pdev)
+{
+ acard->dev = pdev;
+ if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0)
+ return -EBUSY;
+ return 0;
+}
+
+static struct pnp_card_device_id snd_audiodrive_pnpids[] = {
+ /* ESS 1868 (integrated on Compaq dual P-Pro motherboard and Genius 18PnP 3D) */
+ { .id = "ESS1868", .devs = { { "ESS1868" }, { "ESS0000" } } },
+ /* ESS 1868 (integrated on Maxisound Cards) */
+ { .id = "ESS1868", .devs = { { "ESS8601" }, { "ESS8600" } } },
+ /* ESS 1868 (integrated on Maxisound Cards) */
+ { .id = "ESS1868", .devs = { { "ESS8611" }, { "ESS8610" } } },
+ /* ESS ES1869 Plug and Play AudioDrive */
+ { .id = "ESS0003", .devs = { { "ESS1869" }, { "ESS0006" } } },
+ /* ESS 1869 */
+ { .id = "ESS1869", .devs = { { "ESS1869" }, { "ESS0006" } } },
+ /* ESS 1878 */
+ { .id = "ESS1878", .devs = { { "ESS1878" }, { "ESS0004" } } },
+ /* ESS 1879 */
+ { .id = "ESS1879", .devs = { { "ESS1879" }, { "ESS0009" } } },
+ /* --- */
+ { .id = "" } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_audiodrive_pnpids);
+
+static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->dev == NULL)
+ return -EBUSY;
+
+ acard->devc = pnp_request_card_device(card, id->devs[1].id, NULL);
+ if (acard->devc == NULL)
+ return -EBUSY;
+
+ /* Control port initialization */
+ if (pnp_activate_dev(acard->devc) < 0) {
+ snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n");
+ return -EAGAIN;
+ }
+ snd_printdd("pnp: port=0x%llx\n",
+ (unsigned long long)pnp_port_start(acard->devc, 0));
+ if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0)
+ return -EBUSY;
+
+ return 0;
+}
+#endif /* CONFIG_PNP */
+
+#ifdef CONFIG_PNP
+#define is_isapnp_selected(dev) isapnp[dev]
+#else
+#define is_isapnp_selected(dev) 0
+#endif
+
+static struct snd_card *snd_es18xx_card_new(int dev)
+{
+ return snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_audiodrive));
+}
+
+static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev)
+{
+ struct snd_audiodrive *acard = card->private_data;
+ struct snd_es18xx *chip;
+ struct snd_opl3 *opl3;
+ int err;
+
+ if ((err = snd_es18xx_new_device(card,
+ port[dev],
+ mpu_port[dev],
+ fm_port[dev],
+ irq[dev], dma1[dev], dma2[dev],
+ &chip)) < 0)
+ return err;
+ acard->chip = chip;
+
+ sprintf(card->driver, "ES%x", chip->version);
+
+ sprintf(card->shortname, "ESS AudioDrive ES%x", chip->version);
+ if (dma1[dev] != dma2[dev])
+ sprintf(card->longname, "%s at 0x%lx, irq %d, dma1 %d, dma2 %d",
+ card->shortname,
+ chip->port,
+ irq[dev], dma1[dev], dma2[dev]);
+ else
+ sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
+ card->shortname,
+ chip->port,
+ irq[dev], dma1[dev]);
+
+ if ((err = snd_es18xx_pcm(chip, 0, NULL)) < 0)
+ return err;
+
+ if ((err = snd_es18xx_mixer(chip)) < 0)
+ return err;
+
+ if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_opl3_create(card, chip->fm_port, chip->fm_port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) {
+ snd_printk(KERN_WARNING PFX "opl3 not detected at 0x%lx\n", chip->fm_port);
+ } else {
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0)
+ return err;
+ }
+ }
+
+ if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX,
+ chip->mpu_port, 0,
+ irq[dev], 0,
+ &chip->rmidi)) < 0)
+ return err;
+ }
+
+ return snd_card_register(card);
+}
+
+static int __devinit snd_es18xx_isa_match(struct device *pdev, unsigned int dev)
+{
+ return enable[dev] && !is_isapnp_selected(dev);
+}
+
+static int __devinit snd_es18xx_isa_probe1(int dev, struct device *devptr)
+{
+ struct snd_card *card;
+ int err;
+
+ card = snd_es18xx_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ snd_card_set_dev(card, devptr);
+ if ((err = snd_audiodrive_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ dev_set_drvdata(devptr, card);
+ return 0;
+}
+
+static int __devinit snd_es18xx_isa_probe(struct device *pdev, unsigned int dev)
+{
+ int err;
+ static int possible_irqs[] = {5, 9, 10, 7, 11, 12, -1};
+ static int possible_dmas[] = {1, 0, 3, 5, -1};
+
+ if (irq[dev] == SNDRV_AUTO_IRQ) {
+ if ((irq[dev] = snd_legacy_find_free_irq(possible_irqs)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (dma1[dev] == SNDRV_AUTO_DMA) {
+ if ((dma1[dev] = snd_legacy_find_free_dma(possible_dmas)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA1\n");
+ return -EBUSY;
+ }
+ }
+ if (dma2[dev] == SNDRV_AUTO_DMA) {
+ if ((dma2[dev] = snd_legacy_find_free_dma(possible_dmas)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA2\n");
+ return -EBUSY;
+ }
+ }
+
+ if (port[dev] != SNDRV_AUTO_PORT) {
+ return snd_es18xx_isa_probe1(dev, pdev);
+ } else {
+ static unsigned long possible_ports[] = {0x220, 0x240, 0x260, 0x280};
+ int i;
+ for (i = 0; i < ARRAY_SIZE(possible_ports); i++) {
+ port[dev] = possible_ports[i];
+ err = snd_es18xx_isa_probe1(dev, pdev);
+ if (! err)
+ return 0;
+ }
+ return err;
+ }
+}
+
+static int __devexit snd_es18xx_isa_remove(struct device *devptr,
+ unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_es18xx_isa_suspend(struct device *dev, unsigned int n,
+ pm_message_t state)
+{
+ return snd_es18xx_suspend(dev_get_drvdata(dev), state);
+}
+
+static int snd_es18xx_isa_resume(struct device *dev, unsigned int n)
+{
+ return snd_es18xx_resume(dev_get_drvdata(dev));
+}
+#endif
+
+#define DEV_NAME "es18xx"
+
+static struct isa_driver snd_es18xx_isa_driver = {
+ .match = snd_es18xx_isa_match,
+ .probe = snd_es18xx_isa_probe,
+ .remove = __devexit_p(snd_es18xx_isa_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_es18xx_isa_suspend,
+ .resume = snd_es18xx_isa_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+
+#ifdef CONFIG_PNP
+static int __devinit snd_audiodrive_pnp_detect(struct pnp_dev *pdev,
+ const struct pnp_device_id *id)
+{
+ static int dev;
+ int err;
+ struct snd_card *card;
+
+ if (pnp_device_is_isapnp(pdev))
+ return -ENOENT; /* we have another procedure - card */
+ for (; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_es18xx_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ if ((err = snd_audiodrive_pnp(dev, card->private_data, pdev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_card_set_dev(card, &pdev->dev);
+ if ((err = snd_audiodrive_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pnp_set_drvdata(pdev, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_audiodrive_pnp_remove(struct pnp_dev * pdev)
+{
+ snd_card_free(pnp_get_drvdata(pdev));
+ pnp_set_drvdata(pdev, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_audiodrive_pnp_suspend(struct pnp_dev *pdev, pm_message_t state)
+{
+ return snd_es18xx_suspend(pnp_get_drvdata(pdev), state);
+}
+static int snd_audiodrive_pnp_resume(struct pnp_dev *pdev)
+{
+ return snd_es18xx_resume(pnp_get_drvdata(pdev));
+}
+#endif
+
+static struct pnp_driver es18xx_pnp_driver = {
+ .name = "es18xx-pnpbios",
+ .id_table = snd_audiodrive_pnpbiosids,
+ .probe = snd_audiodrive_pnp_detect,
+ .remove = __devexit_p(snd_audiodrive_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_audiodrive_pnp_suspend,
+ .resume = snd_audiodrive_pnp_resume,
+#endif
+};
+
+static int __devinit snd_audiodrive_pnpc_detect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ static int dev;
+ struct snd_card *card;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_es18xx_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+
+ if ((res = snd_audiodrive_pnpc(dev, card->private_data, pcard, pid)) < 0) {
+ snd_card_free(card);
+ return res;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((res = snd_audiodrive_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return res;
+ }
+
+ pnp_set_card_drvdata(pcard, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_audiodrive_pnpc_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_audiodrive_pnpc_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ return snd_es18xx_suspend(pnp_get_card_drvdata(pcard), state);
+}
+
+static int snd_audiodrive_pnpc_resume(struct pnp_card_link *pcard)
+{
+ return snd_es18xx_resume(pnp_get_card_drvdata(pcard));
+}
+
+#endif
+
+static struct pnp_card_driver es18xx_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "es18xx",
+ .id_table = snd_audiodrive_pnpids,
+ .probe = snd_audiodrive_pnpc_detect,
+ .remove = __devexit_p(snd_audiodrive_pnpc_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_audiodrive_pnpc_suspend,
+ .resume = snd_audiodrive_pnpc_resume,
+#endif
+};
+#endif /* CONFIG_PNP */
+
+static int __init alsa_card_es18xx_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&snd_es18xx_isa_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+
+ err = pnp_register_driver(&es18xx_pnp_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ err = pnp_register_card_driver(&es18xx_pnpc_driver);
+ if (!err)
+ pnpc_registered = 1;
+
+ if (isa_registered || pnp_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit alsa_card_es18xx_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnpc_registered)
+ pnp_unregister_card_driver(&es18xx_pnpc_driver);
+ if (pnp_registered)
+ pnp_unregister_driver(&es18xx_pnp_driver);
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&snd_es18xx_isa_driver);
+}
+
+module_init(alsa_card_es18xx_init)
+module_exit(alsa_card_es18xx_exit)
diff --git a/sound/isa/gus/Makefile b/sound/isa/gus/Makefile
new file mode 100644
index 0000000..6cd4ee0
--- /dev/null
+++ b/sound/isa/gus/Makefile
@@ -0,0 +1,24 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-gus-lib-objs := gus_main.o \
+ gus_io.o gus_irq.o gus_timer.o \
+ gus_mem.o gus_mem_proc.o gus_dram.o gus_dma.o gus_volume.o \
+ gus_pcm.o gus_mixer.o \
+ gus_uart.o \
+ gus_reset.o
+
+snd-gusclassic-objs := gusclassic.o
+snd-gusextreme-objs := gusextreme.o
+snd-gusmax-objs := gusmax.o
+snd-interwave-objs := interwave.o
+snd-interwave-stb-objs := interwave-stb.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_GUSCLASSIC) += snd-gusclassic.o snd-gus-lib.o
+obj-$(CONFIG_SND_GUSMAX) += snd-gusmax.o snd-gus-lib.o
+obj-$(CONFIG_SND_GUSEXTREME) += snd-gusextreme.o snd-gus-lib.o
+obj-$(CONFIG_SND_INTERWAVE) += snd-interwave.o snd-gus-lib.o
+obj-$(CONFIG_SND_INTERWAVE_STB) += snd-interwave-stb.o snd-gus-lib.o
diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c
new file mode 100644
index 0000000..f45f611
--- /dev/null
+++ b/sound/isa/gus/gus_dma.c
@@ -0,0 +1,243 @@
+/*
+ * Routines for GF1 DMA control
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/dma.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+
+static void snd_gf1_dma_ack(struct snd_gus_card * gus)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL, 0x00);
+ snd_gf1_look8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+static void snd_gf1_dma_program(struct snd_gus_card * gus,
+ unsigned int addr,
+ unsigned long buf_addr,
+ unsigned int count,
+ unsigned int cmd)
+{
+ unsigned long flags;
+ unsigned int address;
+ unsigned char dma_cmd;
+ unsigned int address_high;
+
+ // snd_printk("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", addr, (long) buf, count);
+
+ if (gus->gf1.dma1 > 3) {
+ if (gus->gf1.enh_mode) {
+ address = addr >> 1;
+ } else {
+ if (addr & 0x1f) {
+ snd_printd("snd_gf1_dma_transfer: unaligned address (0x%x)?\n", addr);
+ return;
+ }
+ address = (addr & 0x000c0000) | ((addr & 0x0003ffff) >> 1);
+ }
+ } else {
+ address = addr;
+ }
+
+ dma_cmd = SNDRV_GF1_DMA_ENABLE | (unsigned short) cmd;
+#if 0
+ dma_cmd |= 0x08;
+#endif
+ if (dma_cmd & SNDRV_GF1_DMA_16BIT) {
+ count++;
+ count &= ~1; /* align */
+ }
+ if (gus->gf1.dma1 > 3) {
+ dma_cmd |= SNDRV_GF1_DMA_WIDTH16;
+ count++;
+ count &= ~1; /* align */
+ }
+ snd_gf1_dma_ack(gus);
+ snd_dma_program(gus->gf1.dma1, buf_addr, count, dma_cmd & SNDRV_GF1_DMA_READ ? DMA_MODE_READ : DMA_MODE_WRITE);
+#if 0
+ snd_printk("address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", address << 1, count, dma_cmd);
+#endif
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ if (gus->gf1.enh_mode) {
+ address_high = ((address >> 16) & 0x000000f0) | (address & 0x0000000f);
+ snd_gf1_write16(gus, SNDRV_GF1_GW_DRAM_DMA_LOW, (unsigned short) (address >> 4));
+ snd_gf1_write8(gus, SNDRV_GF1_GB_DRAM_DMA_HIGH, (unsigned char) address_high);
+ } else
+ snd_gf1_write16(gus, SNDRV_GF1_GW_DRAM_DMA_LOW, (unsigned short) (address >> 4));
+ snd_gf1_write8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL, dma_cmd);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+static struct snd_gf1_dma_block *snd_gf1_dma_next_block(struct snd_gus_card * gus)
+{
+ struct snd_gf1_dma_block *block;
+
+ /* PCM block have bigger priority than synthesizer one */
+ if (gus->gf1.dma_data_pcm) {
+ block = gus->gf1.dma_data_pcm;
+ if (gus->gf1.dma_data_pcm_last == block) {
+ gus->gf1.dma_data_pcm =
+ gus->gf1.dma_data_pcm_last = NULL;
+ } else {
+ gus->gf1.dma_data_pcm = block->next;
+ }
+ } else if (gus->gf1.dma_data_synth) {
+ block = gus->gf1.dma_data_synth;
+ if (gus->gf1.dma_data_synth_last == block) {
+ gus->gf1.dma_data_synth =
+ gus->gf1.dma_data_synth_last = NULL;
+ } else {
+ gus->gf1.dma_data_synth = block->next;
+ }
+ } else {
+ block = NULL;
+ }
+ if (block) {
+ gus->gf1.dma_ack = block->ack;
+ gus->gf1.dma_private_data = block->private_data;
+ }
+ return block;
+}
+
+
+static void snd_gf1_dma_interrupt(struct snd_gus_card * gus)
+{
+ struct snd_gf1_dma_block *block;
+
+ snd_gf1_dma_ack(gus);
+ if (gus->gf1.dma_ack)
+ gus->gf1.dma_ack(gus, gus->gf1.dma_private_data);
+ spin_lock(&gus->dma_lock);
+ if (gus->gf1.dma_data_pcm == NULL &&
+ gus->gf1.dma_data_synth == NULL) {
+ gus->gf1.dma_ack = NULL;
+ gus->gf1.dma_flags &= ~SNDRV_GF1_DMA_TRIGGER;
+ spin_unlock(&gus->dma_lock);
+ return;
+ }
+ block = snd_gf1_dma_next_block(gus);
+ spin_unlock(&gus->dma_lock);
+ snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd);
+ kfree(block);
+#if 0
+ printk("program dma (IRQ) - addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", addr, (long) buffer, count, cmd);
+#endif
+}
+
+int snd_gf1_dma_init(struct snd_gus_card * gus)
+{
+ mutex_lock(&gus->dma_mutex);
+ gus->gf1.dma_shared++;
+ if (gus->gf1.dma_shared > 1) {
+ mutex_unlock(&gus->dma_mutex);
+ return 0;
+ }
+ gus->gf1.interrupt_handler_dma_write = snd_gf1_dma_interrupt;
+ gus->gf1.dma_data_pcm =
+ gus->gf1.dma_data_pcm_last =
+ gus->gf1.dma_data_synth =
+ gus->gf1.dma_data_synth_last = NULL;
+ mutex_unlock(&gus->dma_mutex);
+ return 0;
+}
+
+int snd_gf1_dma_done(struct snd_gus_card * gus)
+{
+ struct snd_gf1_dma_block *block;
+
+ mutex_lock(&gus->dma_mutex);
+ gus->gf1.dma_shared--;
+ if (!gus->gf1.dma_shared) {
+ snd_dma_disable(gus->gf1.dma1);
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_DMA_WRITE);
+ snd_gf1_dma_ack(gus);
+ while ((block = gus->gf1.dma_data_pcm)) {
+ gus->gf1.dma_data_pcm = block->next;
+ kfree(block);
+ }
+ while ((block = gus->gf1.dma_data_synth)) {
+ gus->gf1.dma_data_synth = block->next;
+ kfree(block);
+ }
+ gus->gf1.dma_data_pcm_last =
+ gus->gf1.dma_data_synth_last = NULL;
+ }
+ mutex_unlock(&gus->dma_mutex);
+ return 0;
+}
+
+int snd_gf1_dma_transfer_block(struct snd_gus_card * gus,
+ struct snd_gf1_dma_block * __block,
+ int atomic,
+ int synth)
+{
+ unsigned long flags;
+ struct snd_gf1_dma_block *block;
+
+ block = kmalloc(sizeof(*block), atomic ? GFP_ATOMIC : GFP_KERNEL);
+ if (block == NULL) {
+ snd_printk(KERN_ERR "gf1: DMA transfer failure; not enough memory\n");
+ return -ENOMEM;
+ }
+ *block = *__block;
+ block->next = NULL;
+#if 0
+ printk("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", block->addr, (long) block->buffer, block->count, block->cmd);
+#endif
+#if 0
+ printk("gus->gf1.dma_data_pcm_last = 0x%lx\n", (long)gus->gf1.dma_data_pcm_last);
+ printk("gus->gf1.dma_data_pcm = 0x%lx\n", (long)gus->gf1.dma_data_pcm);
+#endif
+ spin_lock_irqsave(&gus->dma_lock, flags);
+ if (synth) {
+ if (gus->gf1.dma_data_synth_last) {
+ gus->gf1.dma_data_synth_last->next = block;
+ gus->gf1.dma_data_synth_last = block;
+ } else {
+ gus->gf1.dma_data_synth =
+ gus->gf1.dma_data_synth_last = block;
+ }
+ } else {
+ if (gus->gf1.dma_data_pcm_last) {
+ gus->gf1.dma_data_pcm_last->next = block;
+ gus->gf1.dma_data_pcm_last = block;
+ } else {
+ gus->gf1.dma_data_pcm =
+ gus->gf1.dma_data_pcm_last = block;
+ }
+ }
+ if (!(gus->gf1.dma_flags & SNDRV_GF1_DMA_TRIGGER)) {
+ gus->gf1.dma_flags |= SNDRV_GF1_DMA_TRIGGER;
+ block = snd_gf1_dma_next_block(gus);
+ spin_unlock_irqrestore(&gus->dma_lock, flags);
+ if (block == NULL)
+ return 0;
+ snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd);
+ kfree(block);
+ return 0;
+ }
+ spin_unlock_irqrestore(&gus->dma_lock, flags);
+ return 0;
+}
diff --git a/sound/isa/gus/gus_dram.c b/sound/isa/gus/gus_dram.c
new file mode 100644
index 0000000..fd2e2e2
--- /dev/null
+++ b/sound/isa/gus/gus_dram.c
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * DRAM access routines
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#include <sound/info.h>
+
+
+static int snd_gus_dram_poke(struct snd_gus_card *gus, char __user *_buffer,
+ unsigned int address, unsigned int size)
+{
+ unsigned long flags;
+ unsigned int size1, size2;
+ char buffer[256], *pbuffer;
+
+ while (size > 0) {
+ size1 = size > sizeof(buffer) ? sizeof(buffer) : size;
+ if (copy_from_user(buffer, _buffer, size1))
+ return -EFAULT;
+ if (gus->interwave) {
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x01);
+ snd_gf1_dram_addr(gus, address);
+ outsb(GUSP(gus, DRAM), buffer, size1);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ address += size1;
+ } else {
+ pbuffer = buffer;
+ size2 = size1;
+ while (size2--)
+ snd_gf1_poke(gus, address++, *pbuffer++);
+ }
+ size -= size1;
+ _buffer += size1;
+ }
+ return 0;
+}
+
+
+int snd_gus_dram_write(struct snd_gus_card *gus, char __user *buffer,
+ unsigned int address, unsigned int size)
+{
+ return snd_gus_dram_poke(gus, buffer, address, size);
+}
+
+static int snd_gus_dram_peek(struct snd_gus_card *gus, char __user *_buffer,
+ unsigned int address, unsigned int size,
+ int rom)
+{
+ unsigned long flags;
+ unsigned int size1, size2;
+ char buffer[256], *pbuffer;
+
+ while (size > 0) {
+ size1 = size > sizeof(buffer) ? sizeof(buffer) : size;
+ if (gus->interwave) {
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, rom ? 0x03 : 0x01);
+ snd_gf1_dram_addr(gus, address);
+ insb(GUSP(gus, DRAM), buffer, size1);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x01);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ address += size1;
+ } else {
+ pbuffer = buffer;
+ size2 = size1;
+ while (size2--)
+ *pbuffer++ = snd_gf1_peek(gus, address++);
+ }
+ if (copy_to_user(_buffer, buffer, size1))
+ return -EFAULT;
+ size -= size1;
+ _buffer += size1;
+ }
+ return 0;
+}
+
+int snd_gus_dram_read(struct snd_gus_card *gus, char __user *buffer,
+ unsigned int address, unsigned int size,
+ int rom)
+{
+ return snd_gus_dram_peek(gus, buffer, address, size, rom);
+}
diff --git a/sound/isa/gus/gus_instr.c b/sound/isa/gus/gus_instr.c
new file mode 100644
index 0000000..4dc9caf
--- /dev/null
+++ b/sound/isa/gus/gus_instr.c
@@ -0,0 +1,172 @@
+/*
+ * Routines for Gravis UltraSound soundcards - Synthesizer
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+
+/*
+ *
+ */
+
+int snd_gus_iwffff_put_sample(void *private_data, struct iwffff_wave *wave,
+ char __user *data, long len, int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+ struct snd_gf1_mem_block *block;
+ int err;
+
+ if (wave->format & IWFFFF_WAVE_ROM)
+ return 0; /* it's probably ok - verify the address? */
+ if (wave->format & IWFFFF_WAVE_STEREO)
+ return -EINVAL; /* not supported */
+ block = snd_gf1_mem_alloc(&gus->gf1.mem_alloc,
+ SNDRV_GF1_MEM_OWNER_WAVE_IWFFFF,
+ NULL, wave->size,
+ wave->format & IWFFFF_WAVE_16BIT, 1,
+ wave->share_id);
+ if (block == NULL)
+ return -ENOMEM;
+ err = snd_gus_dram_write(gus, data,
+ block->ptr, wave->size);
+ if (err < 0) {
+ snd_gf1_mem_lock(&gus->gf1.mem_alloc, 0);
+ snd_gf1_mem_xfree(&gus->gf1.mem_alloc, block);
+ snd_gf1_mem_lock(&gus->gf1.mem_alloc, 1);
+ return err;
+ }
+ wave->address.memory = block->ptr;
+ return 0;
+}
+
+int snd_gus_iwffff_get_sample(void *private_data, struct iwffff_wave *wave,
+ char __user *data, long len, int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+
+ return snd_gus_dram_read(gus, data, wave->address.memory, wave->size,
+ wave->format & IWFFFF_WAVE_ROM ? 1 : 0);
+}
+
+int snd_gus_iwffff_remove_sample(void *private_data, struct iwffff_wave *wave,
+ int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+
+ if (wave->format & IWFFFF_WAVE_ROM)
+ return 0; /* it's probably ok - verify the address? */
+ return snd_gf1_mem_free(&gus->gf1.mem_alloc, wave->address.memory);
+}
+
+/*
+ *
+ */
+
+int snd_gus_gf1_put_sample(void *private_data, struct gf1_wave *wave,
+ char __user *data, long len, int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+ struct snd_gf1_mem_block *block;
+ int err;
+
+ if (wave->format & GF1_WAVE_STEREO)
+ return -EINVAL; /* not supported */
+ block = snd_gf1_mem_alloc(&gus->gf1.mem_alloc,
+ SNDRV_GF1_MEM_OWNER_WAVE_GF1,
+ NULL, wave->size,
+ wave->format & GF1_WAVE_16BIT, 1,
+ wave->share_id);
+ if (block == NULL)
+ return -ENOMEM;
+ err = snd_gus_dram_write(gus, data,
+ block->ptr, wave->size);
+ if (err < 0) {
+ snd_gf1_mem_lock(&gus->gf1.mem_alloc, 0);
+ snd_gf1_mem_xfree(&gus->gf1.mem_alloc, block);
+ snd_gf1_mem_lock(&gus->gf1.mem_alloc, 1);
+ return err;
+ }
+ wave->address.memory = block->ptr;
+ return 0;
+}
+
+int snd_gus_gf1_get_sample(void *private_data, struct gf1_wave *wave,
+ char __user *data, long len, int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+
+ return snd_gus_dram_read(gus, data, wave->address.memory, wave->size, 0);
+}
+
+int snd_gus_gf1_remove_sample(void *private_data, struct gf1_wave *wave,
+ int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+
+ return snd_gf1_mem_free(&gus->gf1.mem_alloc, wave->address.memory);
+}
+
+/*
+ *
+ */
+
+int snd_gus_simple_put_sample(void *private_data, struct simple_instrument *instr,
+ char __user *data, long len, int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+ struct snd_gf1_mem_block *block;
+ int err;
+
+ if (instr->format & SIMPLE_WAVE_STEREO)
+ return -EINVAL; /* not supported */
+ block = snd_gf1_mem_alloc(&gus->gf1.mem_alloc,
+ SNDRV_GF1_MEM_OWNER_WAVE_SIMPLE,
+ NULL, instr->size,
+ instr->format & SIMPLE_WAVE_16BIT, 1,
+ instr->share_id);
+ if (block == NULL)
+ return -ENOMEM;
+ err = snd_gus_dram_write(gus, data, block->ptr, instr->size);
+ if (err < 0) {
+ snd_gf1_mem_lock(&gus->gf1.mem_alloc, 0);
+ snd_gf1_mem_xfree(&gus->gf1.mem_alloc, block);
+ snd_gf1_mem_lock(&gus->gf1.mem_alloc, 1);
+ return err;
+ }
+ instr->address.memory = block->ptr;
+ return 0;
+}
+
+int snd_gus_simple_get_sample(void *private_data, struct simple_instrument *instr,
+ char __user *data, long len, int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+
+ return snd_gus_dram_read(gus, data, instr->address.memory, instr->size, 0);
+}
+
+int snd_gus_simple_remove_sample(void *private_data, struct simple_instrument *instr,
+ int atomic)
+{
+ struct snd_gus_card *gus = private_data;
+
+ return snd_gf1_mem_free(&gus->gf1.mem_alloc, instr->address.memory);
+}
diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c
new file mode 100644
index 0000000..ca79878
--- /dev/null
+++ b/sound/isa/gus/gus_io.c
@@ -0,0 +1,540 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * I/O routines for GF1/InterWave synthesizer chips
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+
+void snd_gf1_delay(struct snd_gus_card * gus)
+{
+ int i;
+
+ for (i = 0; i < 6; i++) {
+ mb();
+ inb(GUSP(gus, DRAM));
+ }
+}
+
+/*
+ * =======================================================================
+ */
+
+/*
+ * ok.. stop of control registers (wave & ramp) need some special things..
+ * big UltraClick (tm) elimination...
+ */
+
+static inline void __snd_gf1_ctrl_stop(struct snd_gus_card * gus, unsigned char reg)
+{
+ unsigned char value;
+
+ outb(reg | 0x80, gus->gf1.reg_regsel);
+ mb();
+ value = inb(gus->gf1.reg_data8);
+ mb();
+ outb(reg, gus->gf1.reg_regsel);
+ mb();
+ outb((value | 0x03) & ~(0x80 | 0x20), gus->gf1.reg_data8);
+ mb();
+}
+
+static inline void __snd_gf1_write8(struct snd_gus_card * gus,
+ unsigned char reg,
+ unsigned char data)
+{
+ outb(reg, gus->gf1.reg_regsel);
+ mb();
+ outb(data, gus->gf1.reg_data8);
+ mb();
+}
+
+static inline unsigned char __snd_gf1_look8(struct snd_gus_card * gus,
+ unsigned char reg)
+{
+ outb(reg, gus->gf1.reg_regsel);
+ mb();
+ return inb(gus->gf1.reg_data8);
+}
+
+static inline void __snd_gf1_write16(struct snd_gus_card * gus,
+ unsigned char reg, unsigned int data)
+{
+ outb(reg, gus->gf1.reg_regsel);
+ mb();
+ outw((unsigned short) data, gus->gf1.reg_data16);
+ mb();
+}
+
+static inline unsigned short __snd_gf1_look16(struct snd_gus_card * gus,
+ unsigned char reg)
+{
+ outb(reg, gus->gf1.reg_regsel);
+ mb();
+ return inw(gus->gf1.reg_data16);
+}
+
+static inline void __snd_gf1_adlib_write(struct snd_gus_card * gus,
+ unsigned char reg, unsigned char data)
+{
+ outb(reg, gus->gf1.reg_timerctrl);
+ inb(gus->gf1.reg_timerctrl);
+ inb(gus->gf1.reg_timerctrl);
+ outb(data, gus->gf1.reg_timerdata);
+ inb(gus->gf1.reg_timerctrl);
+ inb(gus->gf1.reg_timerctrl);
+}
+
+static inline void __snd_gf1_write_addr(struct snd_gus_card * gus, unsigned char reg,
+ unsigned int addr, int w_16bit)
+{
+ if (gus->gf1.enh_mode) {
+ if (w_16bit)
+ addr = ((addr >> 1) & ~0x0000000f) | (addr & 0x0000000f);
+ __snd_gf1_write8(gus, SNDRV_GF1_VB_UPPER_ADDRESS, (unsigned char) ((addr >> 26) & 0x03));
+ } else if (w_16bit)
+ addr = (addr & 0x00c0000f) | ((addr & 0x003ffff0) >> 1);
+ __snd_gf1_write16(gus, reg, (unsigned short) (addr >> 11));
+ __snd_gf1_write16(gus, reg + 1, (unsigned short) (addr << 5));
+}
+
+static inline unsigned int __snd_gf1_read_addr(struct snd_gus_card * gus,
+ unsigned char reg, short w_16bit)
+{
+ unsigned int res;
+
+ res = ((unsigned int) __snd_gf1_look16(gus, reg | 0x80) << 11) & 0xfff800;
+ res |= ((unsigned int) __snd_gf1_look16(gus, (reg + 1) | 0x80) >> 5) & 0x0007ff;
+ if (gus->gf1.enh_mode) {
+ res |= (unsigned int) __snd_gf1_look8(gus, SNDRV_GF1_VB_UPPER_ADDRESS | 0x80) << 26;
+ if (w_16bit)
+ res = ((res << 1) & 0xffffffe0) | (res & 0x0000000f);
+ } else if (w_16bit)
+ res = ((res & 0x001ffff0) << 1) | (res & 0x00c0000f);
+ return res;
+}
+
+
+/*
+ * =======================================================================
+ */
+
+void snd_gf1_ctrl_stop(struct snd_gus_card * gus, unsigned char reg)
+{
+ __snd_gf1_ctrl_stop(gus, reg);
+}
+
+void snd_gf1_write8(struct snd_gus_card * gus,
+ unsigned char reg,
+ unsigned char data)
+{
+ __snd_gf1_write8(gus, reg, data);
+}
+
+unsigned char snd_gf1_look8(struct snd_gus_card * gus, unsigned char reg)
+{
+ return __snd_gf1_look8(gus, reg);
+}
+
+void snd_gf1_write16(struct snd_gus_card * gus,
+ unsigned char reg,
+ unsigned int data)
+{
+ __snd_gf1_write16(gus, reg, data);
+}
+
+unsigned short snd_gf1_look16(struct snd_gus_card * gus, unsigned char reg)
+{
+ return __snd_gf1_look16(gus, reg);
+}
+
+void snd_gf1_adlib_write(struct snd_gus_card * gus,
+ unsigned char reg,
+ unsigned char data)
+{
+ __snd_gf1_adlib_write(gus, reg, data);
+}
+
+void snd_gf1_write_addr(struct snd_gus_card * gus, unsigned char reg,
+ unsigned int addr, short w_16bit)
+{
+ __snd_gf1_write_addr(gus, reg, addr, w_16bit);
+}
+
+unsigned int snd_gf1_read_addr(struct snd_gus_card * gus,
+ unsigned char reg,
+ short w_16bit)
+{
+ return __snd_gf1_read_addr(gus, reg, w_16bit);
+}
+
+/*
+
+ */
+
+void snd_gf1_i_ctrl_stop(struct snd_gus_card * gus, unsigned char reg)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ __snd_gf1_ctrl_stop(gus, reg);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+void snd_gf1_i_write8(struct snd_gus_card * gus,
+ unsigned char reg,
+ unsigned char data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ __snd_gf1_write8(gus, reg, data);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+unsigned char snd_gf1_i_look8(struct snd_gus_card * gus, unsigned char reg)
+{
+ unsigned long flags;
+ unsigned char res;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ res = __snd_gf1_look8(gus, reg);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return res;
+}
+
+void snd_gf1_i_write16(struct snd_gus_card * gus,
+ unsigned char reg,
+ unsigned int data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ __snd_gf1_write16(gus, reg, data);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+unsigned short snd_gf1_i_look16(struct snd_gus_card * gus, unsigned char reg)
+{
+ unsigned long flags;
+ unsigned short res;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ res = __snd_gf1_look16(gus, reg);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return res;
+}
+
+#if 0
+
+void snd_gf1_i_adlib_write(struct snd_gus_card * gus,
+ unsigned char reg,
+ unsigned char data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ __snd_gf1_adlib_write(gus, reg, data);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+void snd_gf1_i_write_addr(struct snd_gus_card * gus, unsigned char reg,
+ unsigned int addr, short w_16bit)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ __snd_gf1_write_addr(gus, reg, addr, w_16bit);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+#endif /* 0 */
+
+#ifdef CONFIG_SND_DEBUG
+static unsigned int snd_gf1_i_read_addr(struct snd_gus_card * gus,
+ unsigned char reg, short w_16bit)
+{
+ unsigned int res;
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ res = __snd_gf1_read_addr(gus, reg, w_16bit);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return res;
+}
+#endif
+
+/*
+
+ */
+
+void snd_gf1_dram_addr(struct snd_gus_card * gus, unsigned int addr)
+{
+ outb(0x43, gus->gf1.reg_regsel);
+ mb();
+ outw((unsigned short) addr, gus->gf1.reg_data16);
+ mb();
+ outb(0x44, gus->gf1.reg_regsel);
+ mb();
+ outb((unsigned char) (addr >> 16), gus->gf1.reg_data8);
+ mb();
+}
+
+void snd_gf1_poke(struct snd_gus_card * gus, unsigned int addr, unsigned char data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(SNDRV_GF1_GW_DRAM_IO_LOW, gus->gf1.reg_regsel);
+ mb();
+ outw((unsigned short) addr, gus->gf1.reg_data16);
+ mb();
+ outb(SNDRV_GF1_GB_DRAM_IO_HIGH, gus->gf1.reg_regsel);
+ mb();
+ outb((unsigned char) (addr >> 16), gus->gf1.reg_data8);
+ mb();
+ outb(data, gus->gf1.reg_dram);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+unsigned char snd_gf1_peek(struct snd_gus_card * gus, unsigned int addr)
+{
+ unsigned long flags;
+ unsigned char res;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(SNDRV_GF1_GW_DRAM_IO_LOW, gus->gf1.reg_regsel);
+ mb();
+ outw((unsigned short) addr, gus->gf1.reg_data16);
+ mb();
+ outb(SNDRV_GF1_GB_DRAM_IO_HIGH, gus->gf1.reg_regsel);
+ mb();
+ outb((unsigned char) (addr >> 16), gus->gf1.reg_data8);
+ mb();
+ res = inb(gus->gf1.reg_dram);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return res;
+}
+
+#if 0
+
+void snd_gf1_pokew(struct snd_gus_card * gus, unsigned int addr, unsigned short data)
+{
+ unsigned long flags;
+
+#ifdef CONFIG_SND_DEBUG
+ if (!gus->interwave)
+ snd_printk(KERN_DEBUG "snd_gf1_pokew - GF1!!!\n");
+#endif
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(SNDRV_GF1_GW_DRAM_IO_LOW, gus->gf1.reg_regsel);
+ mb();
+ outw((unsigned short) addr, gus->gf1.reg_data16);
+ mb();
+ outb(SNDRV_GF1_GB_DRAM_IO_HIGH, gus->gf1.reg_regsel);
+ mb();
+ outb((unsigned char) (addr >> 16), gus->gf1.reg_data8);
+ mb();
+ outb(SNDRV_GF1_GW_DRAM_IO16, gus->gf1.reg_regsel);
+ mb();
+ outw(data, gus->gf1.reg_data16);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+unsigned short snd_gf1_peekw(struct snd_gus_card * gus, unsigned int addr)
+{
+ unsigned long flags;
+ unsigned short res;
+
+#ifdef CONFIG_SND_DEBUG
+ if (!gus->interwave)
+ snd_printk(KERN_DEBUG "snd_gf1_peekw - GF1!!!\n");
+#endif
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(SNDRV_GF1_GW_DRAM_IO_LOW, gus->gf1.reg_regsel);
+ mb();
+ outw((unsigned short) addr, gus->gf1.reg_data16);
+ mb();
+ outb(SNDRV_GF1_GB_DRAM_IO_HIGH, gus->gf1.reg_regsel);
+ mb();
+ outb((unsigned char) (addr >> 16), gus->gf1.reg_data8);
+ mb();
+ outb(SNDRV_GF1_GW_DRAM_IO16, gus->gf1.reg_regsel);
+ mb();
+ res = inw(gus->gf1.reg_data16);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return res;
+}
+
+void snd_gf1_dram_setmem(struct snd_gus_card * gus, unsigned int addr,
+ unsigned short value, unsigned int count)
+{
+ unsigned long port;
+ unsigned long flags;
+
+#ifdef CONFIG_SND_DEBUG
+ if (!gus->interwave)
+ snd_printk(KERN_DEBUG "snd_gf1_dram_setmem - GF1!!!\n");
+#endif
+ addr &= ~1;
+ count >>= 1;
+ port = GUSP(gus, GF1DATALOW);
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(SNDRV_GF1_GW_DRAM_IO_LOW, gus->gf1.reg_regsel);
+ mb();
+ outw((unsigned short) addr, gus->gf1.reg_data16);
+ mb();
+ outb(SNDRV_GF1_GB_DRAM_IO_HIGH, gus->gf1.reg_regsel);
+ mb();
+ outb((unsigned char) (addr >> 16), gus->gf1.reg_data8);
+ mb();
+ outb(SNDRV_GF1_GW_DRAM_IO16, gus->gf1.reg_regsel);
+ while (count--)
+ outw(value, port);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+#endif /* 0 */
+
+void snd_gf1_select_active_voices(struct snd_gus_card * gus)
+{
+ unsigned short voices;
+
+ static unsigned short voices_tbl[32 - 14 + 1] =
+ {
+ 44100, 41160, 38587, 36317, 34300, 32494, 30870, 29400, 28063, 26843,
+ 25725, 24696, 23746, 22866, 22050, 21289, 20580, 19916, 19293
+ };
+
+ voices = gus->gf1.active_voices;
+ if (voices > 32)
+ voices = 32;
+ if (voices < 14)
+ voices = 14;
+ if (gus->gf1.enh_mode)
+ voices = 32;
+ gus->gf1.active_voices = voices;
+ gus->gf1.playback_freq =
+ gus->gf1.enh_mode ? 44100 : voices_tbl[voices - 14];
+ if (!gus->gf1.enh_mode) {
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_ACTIVE_VOICES, 0xc0 | (voices - 1));
+ udelay(100);
+ }
+}
+
+#ifdef CONFIG_SND_DEBUG
+
+void snd_gf1_print_voice_registers(struct snd_gus_card * gus)
+{
+ unsigned char mode;
+ int voice, ctrl;
+
+ voice = gus->gf1.active_voice;
+ printk(KERN_INFO " -%i- GF1 voice ctrl, ramp ctrl = 0x%x, 0x%x\n", voice, ctrl = snd_gf1_i_read8(gus, 0), snd_gf1_i_read8(gus, 0x0d));
+ printk(KERN_INFO " -%i- GF1 frequency = 0x%x\n", voice, snd_gf1_i_read16(gus, 1));
+ printk(KERN_INFO " -%i- GF1 loop start, end = 0x%x (0x%x), 0x%x (0x%x)\n", voice, snd_gf1_i_read_addr(gus, 2, ctrl & 4), snd_gf1_i_read_addr(gus, 2, (ctrl & 4) ^ 4), snd_gf1_i_read_addr(gus, 4, ctrl & 4), snd_gf1_i_read_addr(gus, 4, (ctrl & 4) ^ 4));
+ printk(KERN_INFO " -%i- GF1 ramp start, end, rate = 0x%x, 0x%x, 0x%x\n", voice, snd_gf1_i_read8(gus, 7), snd_gf1_i_read8(gus, 8), snd_gf1_i_read8(gus, 6));
+ printk(KERN_INFO" -%i- GF1 volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 9));
+ printk(KERN_INFO " -%i- GF1 position = 0x%x (0x%x)\n", voice, snd_gf1_i_read_addr(gus, 0x0a, ctrl & 4), snd_gf1_i_read_addr(gus, 0x0a, (ctrl & 4) ^ 4));
+ if (gus->interwave && snd_gf1_i_read8(gus, 0x19) & 0x01) { /* enhanced mode */
+ mode = snd_gf1_i_read8(gus, 0x15);
+ printk(KERN_INFO " -%i- GFA1 mode = 0x%x\n", voice, mode);
+ if (mode & 0x01) { /* Effect processor */
+ printk(KERN_INFO " -%i- GFA1 effect address = 0x%x\n", voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4));
+ printk(KERN_INFO " -%i- GFA1 effect volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x16));
+ printk(KERN_INFO " -%i- GFA1 effect volume final = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x1d));
+ printk(KERN_INFO " -%i- GFA1 effect acumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14));
+ }
+ if (mode & 0x20) {
+ printk(KERN_INFO " -%i- GFA1 left offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x13), snd_gf1_i_read16(gus, 0x13) >> 4);
+ printk(KERN_INFO " -%i- GFA1 left offset final = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x1c), snd_gf1_i_read16(gus, 0x1c) >> 4);
+ printk(KERN_INFO " -%i- GFA1 right offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x0c), snd_gf1_i_read16(gus, 0x0c) >> 4);
+ printk(KERN_INFO " -%i- GFA1 right offset final = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x1b), snd_gf1_i_read16(gus, 0x1b) >> 4);
+ } else
+ printk(KERN_INFO " -%i- GF1 pan = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x0c));
+ } else
+ printk(KERN_INFO " -%i- GF1 pan = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x0c));
+}
+
+#if 0
+
+void snd_gf1_print_global_registers(struct snd_gus_card * gus)
+{
+ unsigned char global_mode = 0x00;
+
+ printk(KERN_INFO " -G- GF1 active voices = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_ACTIVE_VOICES));
+ if (gus->interwave) {
+ global_mode = snd_gf1_i_read8(gus, SNDRV_GF1_GB_GLOBAL_MODE);
+ printk(KERN_INFO " -G- GF1 global mode = 0x%x\n", global_mode);
+ }
+ if (global_mode & 0x02) /* LFO enabled? */
+ printk(KERN_INFO " -G- GF1 LFO base = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_LFO_BASE));
+ printk(KERN_INFO " -G- GF1 voices IRQ read = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_VOICES_IRQ_READ));
+ printk(KERN_INFO " -G- GF1 DRAM DMA control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL));
+ printk(KERN_INFO " -G- GF1 DRAM DMA high/low = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_HIGH), snd_gf1_i_read16(gus, SNDRV_GF1_GW_DRAM_DMA_LOW));
+ printk(KERN_INFO " -G- GF1 DRAM IO high/low = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_IO_HIGH), snd_gf1_i_read16(gus, SNDRV_GF1_GW_DRAM_IO_LOW));
+ if (!gus->interwave)
+ printk(KERN_INFO " -G- GF1 record DMA control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL));
+ printk(KERN_INFO " -G- GF1 DRAM IO 16 = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_DRAM_IO16));
+ if (gus->gf1.enh_mode) {
+ printk(KERN_INFO " -G- GFA1 memory config = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG));
+ printk(KERN_INFO " -G- GFA1 memory control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_MEMORY_CONTROL));
+ printk(KERN_INFO " -G- GFA1 FIFO record base = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_FIFO_RECORD_BASE_ADDR));
+ printk(KERN_INFO " -G- GFA1 FIFO playback base = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_FIFO_PLAY_BASE_ADDR));
+ printk(KERN_INFO " -G- GFA1 interleave control = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_INTERLEAVE));
+ }
+}
+
+void snd_gf1_print_setup_registers(struct snd_gus_card * gus)
+{
+ printk(KERN_INFO " -S- mix control = 0x%x\n", inb(GUSP(gus, MIXCNTRLREG)));
+ printk(KERN_INFO " -S- IRQ status = 0x%x\n", inb(GUSP(gus, IRQSTAT)));
+ printk(KERN_INFO " -S- timer control = 0x%x\n", inb(GUSP(gus, TIMERCNTRL)));
+ printk(KERN_INFO " -S- timer data = 0x%x\n", inb(GUSP(gus, TIMERDATA)));
+ printk(KERN_INFO " -S- status read = 0x%x\n", inb(GUSP(gus, REGCNTRLS)));
+ printk(KERN_INFO " -S- Sound Blaster control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL));
+ printk(KERN_INFO " -S- AdLib timer 1/2 = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_ADLIB_TIMER_1), snd_gf1_i_look8(gus, SNDRV_GF1_GB_ADLIB_TIMER_2));
+ printk(KERN_INFO " -S- reset = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET));
+ if (gus->interwave) {
+ printk(KERN_INFO " -S- compatibility = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_COMPATIBILITY));
+ printk(KERN_INFO " -S- decode control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DECODE_CONTROL));
+ printk(KERN_INFO " -S- version number = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_VERSION_NUMBER));
+ printk(KERN_INFO " -S- MPU-401 emul. control A/B = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_MPU401_CONTROL_A), snd_gf1_i_look8(gus, SNDRV_GF1_GB_MPU401_CONTROL_B));
+ printk(KERN_INFO " -S- emulation IRQ = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_EMULATION_IRQ));
+ }
+}
+
+void snd_gf1_peek_print_block(struct snd_gus_card * gus, unsigned int addr, int count, int w_16bit)
+{
+ if (!w_16bit) {
+ while (count-- > 0)
+ printk(count > 0 ? "%02x:" : "%02x", snd_gf1_peek(gus, addr++));
+ } else {
+ while (count-- > 0) {
+ printk(count > 0 ? "%04x:" : "%04x", snd_gf1_peek(gus, addr) | (snd_gf1_peek(gus, addr + 1) << 8));
+ addr += 2;
+ }
+ }
+}
+
+#endif /* 0 */
+
+#endif
diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c
new file mode 100644
index 0000000..041894d
--- /dev/null
+++ b/sound/isa/gus/gus_irq.c
@@ -0,0 +1,147 @@
+/*
+ * Routine for IRQ handling from GF1/InterWave chip
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/gus.h>
+
+#ifdef CONFIG_SND_DEBUG
+#define STAT_ADD(x) ((x)++)
+#else
+#define STAT_ADD(x) while (0) { ; }
+#endif
+
+irqreturn_t snd_gus_interrupt(int irq, void *dev_id)
+{
+ struct snd_gus_card * gus = dev_id;
+ unsigned char status;
+ int loop = 100;
+ int handled = 0;
+
+__again:
+ status = inb(gus->gf1.reg_irqstat);
+ if (status == 0)
+ return IRQ_RETVAL(handled);
+ handled = 1;
+ // snd_printk("IRQ: status = 0x%x\n", status);
+ if (status & 0x02) {
+ STAT_ADD(gus->gf1.interrupt_stat_midi_in);
+ if (gus->gf1.interrupt_handler_midi_in)
+ gus->gf1.interrupt_handler_midi_in(gus);
+ }
+ if (status & 0x01) {
+ STAT_ADD(gus->gf1.interrupt_stat_midi_out);
+ if (gus->gf1.interrupt_handler_midi_out)
+ gus->gf1.interrupt_handler_midi_out(gus);
+ }
+ if (status & (0x20 | 0x40)) {
+ unsigned int already, _current_;
+ unsigned char voice_status, voice;
+ struct snd_gus_voice *pvoice;
+
+ already = 0;
+ while (((voice_status = snd_gf1_i_read8(gus, SNDRV_GF1_GB_VOICES_IRQ)) & 0xc0) != 0xc0) {
+ voice = voice_status & 0x1f;
+ _current_ = 1 << voice;
+ if (already & _current_)
+ continue; /* multi request */
+ already |= _current_; /* mark request */
+#if 0
+ printk("voice = %i, voice_status = 0x%x, voice_verify = %i\n", voice, voice_status, inb(GUSP(gus, GF1PAGE)));
+#endif
+ pvoice = &gus->gf1.voices[voice];
+ if (pvoice->use) {
+ if (!(voice_status & 0x80)) { /* voice position IRQ */
+ STAT_ADD(pvoice->interrupt_stat_wave);
+ pvoice->handler_wave(gus, pvoice);
+ }
+ if (!(voice_status & 0x40)) { /* volume ramp IRQ */
+ STAT_ADD(pvoice->interrupt_stat_volume);
+ pvoice->handler_volume(gus, pvoice);
+ }
+ } else {
+ STAT_ADD(gus->gf1.interrupt_stat_voice_lost);
+ snd_gf1_i_ctrl_stop(gus, SNDRV_GF1_VB_ADDRESS_CONTROL);
+ snd_gf1_i_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
+ }
+ }
+ }
+ if (status & 0x04) {
+ STAT_ADD(gus->gf1.interrupt_stat_timer1);
+ if (gus->gf1.interrupt_handler_timer1)
+ gus->gf1.interrupt_handler_timer1(gus);
+ }
+ if (status & 0x08) {
+ STAT_ADD(gus->gf1.interrupt_stat_timer2);
+ if (gus->gf1.interrupt_handler_timer2)
+ gus->gf1.interrupt_handler_timer2(gus);
+ }
+ if (status & 0x80) {
+ if (snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL) & 0x40) {
+ STAT_ADD(gus->gf1.interrupt_stat_dma_write);
+ if (gus->gf1.interrupt_handler_dma_write)
+ gus->gf1.interrupt_handler_dma_write(gus);
+ }
+ if (snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL) & 0x40) {
+ STAT_ADD(gus->gf1.interrupt_stat_dma_read);
+ if (gus->gf1.interrupt_handler_dma_read)
+ gus->gf1.interrupt_handler_dma_read(gus);
+ }
+ }
+ if (--loop > 0)
+ goto __again;
+ return IRQ_NONE;
+}
+
+#ifdef CONFIG_SND_DEBUG
+static void snd_gus_irq_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_gus_card *gus;
+ struct snd_gus_voice *pvoice;
+ int idx;
+
+ gus = entry->private_data;
+ snd_iprintf(buffer, "midi out = %u\n", gus->gf1.interrupt_stat_midi_out);
+ snd_iprintf(buffer, "midi in = %u\n", gus->gf1.interrupt_stat_midi_in);
+ snd_iprintf(buffer, "timer1 = %u\n", gus->gf1.interrupt_stat_timer1);
+ snd_iprintf(buffer, "timer2 = %u\n", gus->gf1.interrupt_stat_timer2);
+ snd_iprintf(buffer, "dma write = %u\n", gus->gf1.interrupt_stat_dma_write);
+ snd_iprintf(buffer, "dma read = %u\n", gus->gf1.interrupt_stat_dma_read);
+ snd_iprintf(buffer, "voice lost = %u\n", gus->gf1.interrupt_stat_voice_lost);
+ for (idx = 0; idx < 32; idx++) {
+ pvoice = &gus->gf1.voices[idx];
+ snd_iprintf(buffer, "voice %i: wave = %u, volume = %u\n",
+ idx,
+ pvoice->interrupt_stat_wave,
+ pvoice->interrupt_stat_volume);
+ }
+}
+
+void snd_gus_irq_profile_init(struct snd_gus_card *gus)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(gus->card, "gusirq", &entry))
+ snd_info_set_text_ops(entry, gus, snd_gus_irq_info_read);
+}
+
+#endif
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
new file mode 100644
index 0000000..12eb98f
--- /dev/null
+++ b/sound/isa/gus/gus_main.c
@@ -0,0 +1,482 @@
+/*
+ * Routines for Gravis UltraSound soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#include <sound/control.h>
+
+#include <asm/dma.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Routines for Gravis UltraSound soundcards");
+MODULE_LICENSE("GPL");
+
+static int snd_gus_init_dma_irq(struct snd_gus_card * gus, int latches);
+
+int snd_gus_use_inc(struct snd_gus_card * gus)
+{
+ if (!try_module_get(gus->card->module))
+ return 0;
+ return 1;
+}
+
+void snd_gus_use_dec(struct snd_gus_card * gus)
+{
+ module_put(gus->card->module);
+}
+
+static int snd_gus_joystick_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 31;
+ return 0;
+}
+
+static int snd_gus_joystick_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = gus->joystick_dac & 31;
+ return 0;
+}
+
+static int snd_gus_joystick_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned char nval;
+
+ nval = ucontrol->value.integer.value[0] & 31;
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ change = gus->joystick_dac != nval;
+ gus->joystick_dac = nval;
+ snd_gf1_write8(gus, SNDRV_GF1_GB_JOYSTICK_DAC_LEVEL, gus->joystick_dac);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_gus_joystick_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "Joystick Speed",
+ .info = snd_gus_joystick_info,
+ .get = snd_gus_joystick_get,
+ .put = snd_gus_joystick_put
+};
+
+static void snd_gus_init_control(struct snd_gus_card *gus)
+{
+ if (!gus->ace_flag)
+ snd_ctl_add(gus->card, snd_ctl_new1(&snd_gus_joystick_control, gus));
+}
+
+/*
+ *
+ */
+
+static int snd_gus_free(struct snd_gus_card *gus)
+{
+ if (gus->gf1.res_port2 == NULL)
+ goto __hw_end;
+ snd_gf1_stop(gus);
+ snd_gus_init_dma_irq(gus, 0);
+ __hw_end:
+ release_and_free_resource(gus->gf1.res_port1);
+ release_and_free_resource(gus->gf1.res_port2);
+ if (gus->gf1.irq >= 0)
+ free_irq(gus->gf1.irq, (void *) gus);
+ if (gus->gf1.dma1 >= 0) {
+ disable_dma(gus->gf1.dma1);
+ free_dma(gus->gf1.dma1);
+ }
+ if (!gus->equal_dma && gus->gf1.dma2 >= 0) {
+ disable_dma(gus->gf1.dma2);
+ free_dma(gus->gf1.dma2);
+ }
+ kfree(gus);
+ return 0;
+}
+
+static int snd_gus_dev_free(struct snd_device *device)
+{
+ struct snd_gus_card *gus = device->device_data;
+ return snd_gus_free(gus);
+}
+
+int snd_gus_create(struct snd_card *card,
+ unsigned long port,
+ int irq, int dma1, int dma2,
+ int timer_dev,
+ int voices,
+ int pcm_channels,
+ int effect,
+ struct snd_gus_card **rgus)
+{
+ struct snd_gus_card *gus;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_gus_dev_free,
+ };
+
+ *rgus = NULL;
+ gus = kzalloc(sizeof(*gus), GFP_KERNEL);
+ if (gus == NULL)
+ return -ENOMEM;
+ spin_lock_init(&gus->reg_lock);
+ spin_lock_init(&gus->voice_alloc);
+ spin_lock_init(&gus->active_voice_lock);
+ spin_lock_init(&gus->event_lock);
+ spin_lock_init(&gus->dma_lock);
+ spin_lock_init(&gus->pcm_volume_level_lock);
+ spin_lock_init(&gus->uart_cmd_lock);
+ mutex_init(&gus->dma_mutex);
+ gus->gf1.irq = -1;
+ gus->gf1.dma1 = -1;
+ gus->gf1.dma2 = -1;
+ gus->card = card;
+ gus->gf1.port = port;
+ /* fill register variables for speedup */
+ gus->gf1.reg_page = GUSP(gus, GF1PAGE);
+ gus->gf1.reg_regsel = GUSP(gus, GF1REGSEL);
+ gus->gf1.reg_data8 = GUSP(gus, GF1DATAHIGH);
+ gus->gf1.reg_data16 = GUSP(gus, GF1DATALOW);
+ gus->gf1.reg_irqstat = GUSP(gus, IRQSTAT);
+ gus->gf1.reg_dram = GUSP(gus, DRAM);
+ gus->gf1.reg_timerctrl = GUSP(gus, TIMERCNTRL);
+ gus->gf1.reg_timerdata = GUSP(gus, TIMERDATA);
+ /* allocate resources */
+ if ((gus->gf1.res_port1 = request_region(port, 16, "GUS GF1 (Adlib/SB)")) == NULL) {
+ snd_printk(KERN_ERR "gus: can't grab SB port 0x%lx\n", port);
+ snd_gus_free(gus);
+ return -EBUSY;
+ }
+ if ((gus->gf1.res_port2 = request_region(port + 0x100, 12, "GUS GF1 (Synth)")) == NULL) {
+ snd_printk(KERN_ERR "gus: can't grab synth port 0x%lx\n", port + 0x100);
+ snd_gus_free(gus);
+ return -EBUSY;
+ }
+ if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) {
+ snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq);
+ snd_gus_free(gus);
+ return -EBUSY;
+ }
+ gus->gf1.irq = irq;
+ if (request_dma(dma1, "GUS - 1")) {
+ snd_printk(KERN_ERR "gus: can't grab DMA1 %d\n", dma1);
+ snd_gus_free(gus);
+ return -EBUSY;
+ }
+ gus->gf1.dma1 = dma1;
+ if (dma2 >= 0 && dma1 != dma2) {
+ if (request_dma(dma2, "GUS - 2")) {
+ snd_printk(KERN_ERR "gus: can't grab DMA2 %d\n", dma2);
+ snd_gus_free(gus);
+ return -EBUSY;
+ }
+ gus->gf1.dma2 = dma2;
+ } else {
+ gus->gf1.dma2 = gus->gf1.dma1;
+ gus->equal_dma = 1;
+ }
+ gus->timer_dev = timer_dev;
+ if (voices < 14)
+ voices = 14;
+ if (voices > 32)
+ voices = 32;
+ if (pcm_channels < 0)
+ pcm_channels = 0;
+ if (pcm_channels > 8)
+ pcm_channels = 8;
+ pcm_channels++;
+ pcm_channels &= ~1;
+ gus->gf1.effect = effect ? 1 : 0;
+ gus->gf1.active_voices = voices;
+ gus->gf1.pcm_channels = pcm_channels;
+ gus->gf1.volume_ramp = 25;
+ gus->gf1.smooth_pan = 1;
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, gus, &ops)) < 0) {
+ snd_gus_free(gus);
+ return err;
+ }
+ *rgus = gus;
+ return 0;
+}
+
+/*
+ * Memory detection routine for plain GF1 soundcards
+ */
+
+static int snd_gus_detect_memory(struct snd_gus_card * gus)
+{
+ int l, idx, local;
+ unsigned char d;
+
+ snd_gf1_poke(gus, 0L, 0xaa);
+ snd_gf1_poke(gus, 1L, 0x55);
+ if (snd_gf1_peek(gus, 0L) != 0xaa || snd_gf1_peek(gus, 1L) != 0x55) {
+ snd_printk(KERN_ERR "plain GF1 card at 0x%lx without onboard DRAM?\n", gus->gf1.port);
+ return -ENOMEM;
+ }
+ for (idx = 1, d = 0xab; idx < 4; idx++, d++) {
+ local = idx << 18;
+ snd_gf1_poke(gus, local, d);
+ snd_gf1_poke(gus, local + 1, d + 1);
+ if (snd_gf1_peek(gus, local) != d ||
+ snd_gf1_peek(gus, local + 1) != d + 1 ||
+ snd_gf1_peek(gus, 0L) != 0xaa)
+ break;
+ }
+#if 1
+ gus->gf1.memory = idx << 18;
+#else
+ gus->gf1.memory = 256 * 1024;
+#endif
+ for (l = 0, local = gus->gf1.memory; l < 4; l++, local -= 256 * 1024) {
+ gus->gf1.mem_alloc.banks_8[l].address =
+ gus->gf1.mem_alloc.banks_8[l].size = 0;
+ gus->gf1.mem_alloc.banks_16[l].address = l << 18;
+ gus->gf1.mem_alloc.banks_16[l].size = local > 0 ? 256 * 1024 : 0;
+ }
+ gus->gf1.mem_alloc.banks_8[0].size = gus->gf1.memory;
+ return 0; /* some memory were detected */
+}
+
+static int snd_gus_init_dma_irq(struct snd_gus_card * gus, int latches)
+{
+ struct snd_card *card;
+ unsigned long flags;
+ int irq, dma1, dma2;
+ static unsigned char irqs[16] =
+ {0, 0, 1, 3, 0, 2, 0, 4, 0, 1, 0, 5, 6, 0, 0, 7};
+ static unsigned char dmas[8] =
+ {6, 1, 0, 2, 0, 3, 4, 5};
+
+ if (snd_BUG_ON(!gus))
+ return -EINVAL;
+ card = gus->card;
+ if (snd_BUG_ON(!card))
+ return -EINVAL;
+
+ gus->mix_cntrl_reg &= 0xf8;
+ gus->mix_cntrl_reg |= 0x01; /* disable MIC, LINE IN, enable LINE OUT */
+ if (gus->codec_flag || gus->ess_flag) {
+ gus->mix_cntrl_reg &= ~1; /* enable LINE IN */
+ gus->mix_cntrl_reg |= 4; /* enable MIC */
+ }
+ dma1 = gus->gf1.dma1;
+ dma1 = abs(dma1);
+ dma1 = dmas[dma1 & 7];
+ dma2 = gus->gf1.dma2;
+ dma2 = abs(dma2);
+ dma2 = dmas[dma2 & 7];
+ dma1 |= gus->equal_dma ? 0x40 : (dma2 << 3);
+
+ if ((dma1 & 7) == 0 || (dma2 & 7) == 0) {
+ snd_printk(KERN_ERR "Error! DMA isn't defined.\n");
+ return -EINVAL;
+ }
+ irq = gus->gf1.irq;
+ irq = abs(irq);
+ irq = irqs[irq & 0x0f];
+ if (irq == 0) {
+ snd_printk(KERN_ERR "Error! IRQ isn't defined.\n");
+ return -EINVAL;
+ }
+ irq |= 0x40;
+#if 0
+ card->mixer.mix_ctrl_reg |= 0x10;
+#endif
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(5, GUSP(gus, REGCNTRLS));
+ outb(gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG));
+ outb(0x00, GUSP(gus, IRQDMACNTRLREG));
+ outb(0, GUSP(gus, REGCNTRLS));
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+
+ udelay(100);
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(0x00 | gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG));
+ outb(dma1, GUSP(gus, IRQDMACNTRLREG));
+ if (latches) {
+ outb(0x40 | gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG));
+ outb(irq, GUSP(gus, IRQDMACNTRLREG));
+ }
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+
+ udelay(100);
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(0x00 | gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG));
+ outb(dma1, GUSP(gus, IRQDMACNTRLREG));
+ if (latches) {
+ outb(0x40 | gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG));
+ outb(irq, GUSP(gus, IRQDMACNTRLREG));
+ }
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+
+ snd_gf1_delay(gus);
+
+ if (latches)
+ gus->mix_cntrl_reg |= 0x08; /* enable latches */
+ else
+ gus->mix_cntrl_reg &= ~0x08; /* disable latches */
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG));
+ outb(0, GUSP(gus, GF1PAGE));
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+
+ return 0;
+}
+
+static int snd_gus_check_version(struct snd_gus_card * gus)
+{
+ unsigned long flags;
+ unsigned char val, rev;
+ struct snd_card *card;
+
+ card = gus->card;
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(0x20, GUSP(gus, REGCNTRLS));
+ val = inb(GUSP(gus, REGCNTRLS));
+ rev = inb(GUSP(gus, BOARDVERSION));
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ snd_printdd("GF1 [0x%lx] init - val = 0x%x, rev = 0x%x\n", gus->gf1.port, val, rev);
+ strcpy(card->driver, "GUS");
+ strcpy(card->longname, "Gravis UltraSound Classic (2.4)");
+ if ((val != 255 && (val & 0x06)) || (rev >= 5 && rev != 255)) {
+ if (rev >= 5 && rev <= 9) {
+ gus->ics_flag = 1;
+ if (rev == 5)
+ gus->ics_flipped = 1;
+ card->longname[27] = '3';
+ card->longname[29] = rev == 5 ? '5' : '7';
+ }
+ if (rev >= 10 && rev != 255) {
+ if (rev >= 10 && rev <= 11) {
+ strcpy(card->driver, "GUS MAX");
+ strcpy(card->longname, "Gravis UltraSound MAX");
+ gus->max_flag = 1;
+ } else if (rev == 0x30) {
+ strcpy(card->driver, "GUS ACE");
+ strcpy(card->longname, "Gravis UltraSound Ace");
+ gus->ace_flag = 1;
+ } else if (rev == 0x50) {
+ strcpy(card->driver, "GUS Extreme");
+ strcpy(card->longname, "Gravis UltraSound Extreme");
+ gus->ess_flag = 1;
+ } else {
+ snd_printk(KERN_ERR "unknown GF1 revision number at 0x%lx - 0x%x (0x%x)\n", gus->gf1.port, rev, val);
+ snd_printk(KERN_ERR " please - report to <perex@perex.cz>\n");
+ }
+ }
+ }
+ strcpy(card->shortname, card->longname);
+ gus->uart_enable = 1; /* standard GUSes doesn't have midi uart trouble */
+ snd_gus_init_control(gus);
+ return 0;
+}
+
+int snd_gus_initialize(struct snd_gus_card *gus)
+{
+ int err;
+
+ if (!gus->interwave) {
+ if ((err = snd_gus_check_version(gus)) < 0) {
+ snd_printk(KERN_ERR "version check failed\n");
+ return err;
+ }
+ if ((err = snd_gus_detect_memory(gus)) < 0)
+ return err;
+ }
+ if ((err = snd_gus_init_dma_irq(gus, 1)) < 0)
+ return err;
+ snd_gf1_start(gus);
+ gus->initialized = 1;
+ return 0;
+}
+
+ /* gus_io.c */
+EXPORT_SYMBOL(snd_gf1_delay);
+EXPORT_SYMBOL(snd_gf1_write8);
+EXPORT_SYMBOL(snd_gf1_look8);
+EXPORT_SYMBOL(snd_gf1_write16);
+EXPORT_SYMBOL(snd_gf1_look16);
+EXPORT_SYMBOL(snd_gf1_i_write8);
+EXPORT_SYMBOL(snd_gf1_i_look8);
+EXPORT_SYMBOL(snd_gf1_i_look16);
+EXPORT_SYMBOL(snd_gf1_dram_addr);
+EXPORT_SYMBOL(snd_gf1_write_addr);
+EXPORT_SYMBOL(snd_gf1_poke);
+EXPORT_SYMBOL(snd_gf1_peek);
+ /* gus_reset.c */
+EXPORT_SYMBOL(snd_gf1_alloc_voice);
+EXPORT_SYMBOL(snd_gf1_free_voice);
+EXPORT_SYMBOL(snd_gf1_ctrl_stop);
+EXPORT_SYMBOL(snd_gf1_stop_voice);
+ /* gus_mixer.c */
+EXPORT_SYMBOL(snd_gf1_new_mixer);
+ /* gus_pcm.c */
+EXPORT_SYMBOL(snd_gf1_pcm_new);
+ /* gus.c */
+EXPORT_SYMBOL(snd_gus_use_inc);
+EXPORT_SYMBOL(snd_gus_use_dec);
+EXPORT_SYMBOL(snd_gus_create);
+EXPORT_SYMBOL(snd_gus_initialize);
+ /* gus_irq.c */
+EXPORT_SYMBOL(snd_gus_interrupt);
+ /* gus_uart.c */
+EXPORT_SYMBOL(snd_gf1_rawmidi_new);
+ /* gus_dram.c */
+EXPORT_SYMBOL(snd_gus_dram_write);
+EXPORT_SYMBOL(snd_gus_dram_read);
+ /* gus_volume.c */
+EXPORT_SYMBOL(snd_gf1_lvol_to_gvol_raw);
+EXPORT_SYMBOL(snd_gf1_translate_freq);
+ /* gus_mem.c */
+EXPORT_SYMBOL(snd_gf1_mem_alloc);
+EXPORT_SYMBOL(snd_gf1_mem_xfree);
+EXPORT_SYMBOL(snd_gf1_mem_free);
+EXPORT_SYMBOL(snd_gf1_mem_lock);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_gus_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_gus_exit(void)
+{
+}
+
+module_init(alsa_gus_init)
+module_exit(alsa_gus_exit)
diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c
new file mode 100644
index 0000000..661205c
--- /dev/null
+++ b/sound/isa/gus/gus_mem.c
@@ -0,0 +1,350 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * GUS's memory allocation routines / bottom layer
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#include <sound/info.h>
+
+#ifdef CONFIG_SND_DEBUG
+static void snd_gf1_mem_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer);
+#endif
+
+void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup)
+{
+ if (!xup) {
+ mutex_lock(&alloc->memory_mutex);
+ } else {
+ mutex_unlock(&alloc->memory_mutex);
+ }
+}
+
+static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc,
+ struct snd_gf1_mem_block * block)
+{
+ struct snd_gf1_mem_block *pblock, *nblock;
+
+ nblock = kmalloc(sizeof(struct snd_gf1_mem_block), GFP_KERNEL);
+ if (nblock == NULL)
+ return NULL;
+ *nblock = *block;
+ pblock = alloc->first;
+ while (pblock) {
+ if (pblock->ptr > nblock->ptr) {
+ nblock->prev = pblock->prev;
+ nblock->next = pblock;
+ pblock->prev = nblock;
+ if (pblock == alloc->first)
+ alloc->first = nblock;
+ else
+ nblock->prev->next = nblock;
+ mutex_unlock(&alloc->memory_mutex);
+ return NULL;
+ }
+ pblock = pblock->next;
+ }
+ nblock->next = NULL;
+ if (alloc->last == NULL) {
+ nblock->prev = NULL;
+ alloc->first = alloc->last = nblock;
+ } else {
+ nblock->prev = alloc->last;
+ alloc->last->next = nblock;
+ alloc->last = nblock;
+ }
+ return nblock;
+}
+
+int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * block)
+{
+ if (block->share) { /* ok.. shared block */
+ block->share--;
+ mutex_unlock(&alloc->memory_mutex);
+ return 0;
+ }
+ if (alloc->first == block) {
+ alloc->first = block->next;
+ if (block->next)
+ block->next->prev = NULL;
+ } else {
+ block->prev->next = block->next;
+ if (block->next)
+ block->next->prev = block->prev;
+ }
+ if (alloc->last == block) {
+ alloc->last = block->prev;
+ if (block->prev)
+ block->prev->next = NULL;
+ } else {
+ block->next->prev = block->prev;
+ if (block->prev)
+ block->prev->next = block->next;
+ }
+ kfree(block->name);
+ kfree(block);
+ return 0;
+}
+
+static struct snd_gf1_mem_block *snd_gf1_mem_look(struct snd_gf1_mem * alloc,
+ unsigned int address)
+{
+ struct snd_gf1_mem_block *block;
+
+ for (block = alloc->first; block; block = block->next) {
+ if (block->ptr == address) {
+ return block;
+ }
+ }
+ return NULL;
+}
+
+static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc,
+ unsigned int *share_id)
+{
+ struct snd_gf1_mem_block *block;
+
+ if (!share_id[0] && !share_id[1] &&
+ !share_id[2] && !share_id[3])
+ return NULL;
+ for (block = alloc->first; block; block = block->next)
+ if (!memcmp(share_id, block->share_id, sizeof(share_id)))
+ return block;
+ return NULL;
+}
+
+static int snd_gf1_mem_find(struct snd_gf1_mem * alloc,
+ struct snd_gf1_mem_block * block,
+ unsigned int size, int w_16, int align)
+{
+ struct snd_gf1_bank_info *info = w_16 ? alloc->banks_16 : alloc->banks_8;
+ unsigned int idx, boundary;
+ int size1;
+ struct snd_gf1_mem_block *pblock;
+ unsigned int ptr1, ptr2;
+
+ if (w_16 && align < 2)
+ align = 2;
+ block->flags = w_16 ? SNDRV_GF1_MEM_BLOCK_16BIT : 0;
+ block->owner = SNDRV_GF1_MEM_OWNER_DRIVER;
+ block->share = 0;
+ block->share_id[0] = block->share_id[1] =
+ block->share_id[2] = block->share_id[3] = 0;
+ block->name = NULL;
+ block->prev = block->next = NULL;
+ for (pblock = alloc->first, idx = 0; pblock; pblock = pblock->next) {
+ while (pblock->ptr >= (boundary = info[idx].address + info[idx].size))
+ idx++;
+ while (pblock->ptr + pblock->size >= (boundary = info[idx].address + info[idx].size))
+ idx++;
+ ptr2 = boundary;
+ if (pblock->next) {
+ if (pblock->ptr + pblock->size == pblock->next->ptr)
+ continue;
+ if (pblock->next->ptr < boundary)
+ ptr2 = pblock->next->ptr;
+ }
+ ptr1 = ALIGN(pblock->ptr + pblock->size, align);
+ if (ptr1 >= ptr2)
+ continue;
+ size1 = ptr2 - ptr1;
+ if ((int)size <= size1) {
+ block->ptr = ptr1;
+ block->size = size;
+ return 0;
+ }
+ }
+ while (++idx < 4) {
+ if (size <= info[idx].size) {
+ /* I assume that bank address is already aligned.. */
+ block->ptr = info[idx].address;
+ block->size = size;
+ return 0;
+ }
+ }
+ return -ENOMEM;
+}
+
+struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owner,
+ char *name, int size, int w_16, int align,
+ unsigned int *share_id)
+{
+ struct snd_gf1_mem_block block, *nblock;
+
+ snd_gf1_mem_lock(alloc, 0);
+ if (share_id != NULL) {
+ nblock = snd_gf1_mem_share(alloc, share_id);
+ if (nblock != NULL) {
+ if (size != (int)nblock->size) {
+ /* TODO: remove in the future */
+ snd_printk(KERN_ERR "snd_gf1_mem_alloc - share: sizes differ\n");
+ goto __std;
+ }
+ nblock->share++;
+ snd_gf1_mem_lock(alloc, 1);
+ return NULL;
+ }
+ }
+ __std:
+ if (snd_gf1_mem_find(alloc, &block, size, w_16, align) < 0) {
+ snd_gf1_mem_lock(alloc, 1);
+ return NULL;
+ }
+ if (share_id != NULL)
+ memcpy(&block.share_id, share_id, sizeof(block.share_id));
+ block.owner = owner;
+ block.name = kstrdup(name, GFP_KERNEL);
+ nblock = snd_gf1_mem_xalloc(alloc, &block);
+ snd_gf1_mem_lock(alloc, 1);
+ return nblock;
+}
+
+int snd_gf1_mem_free(struct snd_gf1_mem * alloc, unsigned int address)
+{
+ int result;
+ struct snd_gf1_mem_block *block;
+
+ snd_gf1_mem_lock(alloc, 0);
+ if ((block = snd_gf1_mem_look(alloc, address)) != NULL) {
+ result = snd_gf1_mem_xfree(alloc, block);
+ snd_gf1_mem_lock(alloc, 1);
+ return result;
+ }
+ snd_gf1_mem_lock(alloc, 1);
+ return -EINVAL;
+}
+
+int snd_gf1_mem_init(struct snd_gus_card * gus)
+{
+ struct snd_gf1_mem *alloc;
+ struct snd_gf1_mem_block block;
+#ifdef CONFIG_SND_DEBUG
+ struct snd_info_entry *entry;
+#endif
+
+ alloc = &gus->gf1.mem_alloc;
+ mutex_init(&alloc->memory_mutex);
+ alloc->first = alloc->last = NULL;
+ if (!gus->gf1.memory)
+ return 0;
+
+ memset(&block, 0, sizeof(block));
+ block.owner = SNDRV_GF1_MEM_OWNER_DRIVER;
+ if (gus->gf1.enh_mode) {
+ block.ptr = 0;
+ block.size = 1024;
+ block.name = kstrdup("InterWave LFOs", GFP_KERNEL);
+ if (snd_gf1_mem_xalloc(alloc, &block) == NULL)
+ return -ENOMEM;
+ }
+ block.ptr = gus->gf1.default_voice_address;
+ block.size = 4;
+ block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL);
+ if (snd_gf1_mem_xalloc(alloc, &block) == NULL)
+ return -ENOMEM;
+#ifdef CONFIG_SND_DEBUG
+ if (! snd_card_proc_new(gus->card, "gusmem", &entry))
+ snd_info_set_text_ops(entry, gus, snd_gf1_mem_info_read);
+#endif
+ return 0;
+}
+
+int snd_gf1_mem_done(struct snd_gus_card * gus)
+{
+ struct snd_gf1_mem *alloc;
+ struct snd_gf1_mem_block *block, *nblock;
+
+ alloc = &gus->gf1.mem_alloc;
+ block = alloc->first;
+ while (block) {
+ nblock = block->next;
+ snd_gf1_mem_xfree(alloc, block);
+ block = nblock;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_SND_DEBUG
+static void snd_gf1_mem_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_gus_card *gus;
+ struct snd_gf1_mem *alloc;
+ struct snd_gf1_mem_block *block;
+ unsigned int total, used;
+ int i;
+
+ gus = entry->private_data;
+ alloc = &gus->gf1.mem_alloc;
+ mutex_lock(&alloc->memory_mutex);
+ snd_iprintf(buffer, "8-bit banks : \n ");
+ for (i = 0; i < 4; i++)
+ snd_iprintf(buffer, "0x%06x (%04ik)%s", alloc->banks_8[i].address, alloc->banks_8[i].size >> 10, i + 1 < 4 ? "," : "");
+ snd_iprintf(buffer, "\n"
+ "16-bit banks : \n ");
+ for (i = total = 0; i < 4; i++) {
+ snd_iprintf(buffer, "0x%06x (%04ik)%s", alloc->banks_16[i].address, alloc->banks_16[i].size >> 10, i + 1 < 4 ? "," : "");
+ total += alloc->banks_16[i].size;
+ }
+ snd_iprintf(buffer, "\n");
+ used = 0;
+ for (block = alloc->first, i = 0; block; block = block->next, i++) {
+ used += block->size;
+ snd_iprintf(buffer, "Block %i at 0x%lx onboard 0x%x size %i (0x%x):\n", i, (long) block, block->ptr, block->size, block->size);
+ if (block->share ||
+ block->share_id[0] || block->share_id[1] ||
+ block->share_id[2] || block->share_id[3])
+ snd_iprintf(buffer, " Share : %i [id0 0x%x] [id1 0x%x] [id2 0x%x] [id3 0x%x]\n",
+ block->share,
+ block->share_id[0], block->share_id[1],
+ block->share_id[2], block->share_id[3]);
+ snd_iprintf(buffer, " Flags :%s\n",
+ block->flags & SNDRV_GF1_MEM_BLOCK_16BIT ? " 16-bit" : "");
+ snd_iprintf(buffer, " Owner : ");
+ switch (block->owner) {
+ case SNDRV_GF1_MEM_OWNER_DRIVER:
+ snd_iprintf(buffer, "driver - %s\n", block->name);
+ break;
+ case SNDRV_GF1_MEM_OWNER_WAVE_SIMPLE:
+ snd_iprintf(buffer, "SIMPLE wave\n");
+ break;
+ case SNDRV_GF1_MEM_OWNER_WAVE_GF1:
+ snd_iprintf(buffer, "GF1 wave\n");
+ break;
+ case SNDRV_GF1_MEM_OWNER_WAVE_IWFFFF:
+ snd_iprintf(buffer, "IWFFFF wave\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ }
+ }
+ snd_iprintf(buffer, " Total: memory = %i, used = %i, free = %i\n",
+ total, used, total - used);
+ mutex_unlock(&alloc->memory_mutex);
+#if 0
+ ultra_iprintf(buffer, " Verify: free = %i, max 8-bit block = %i, max 16-bit block = %i\n",
+ ultra_memory_free_size(card, &card->gf1.mem_alloc),
+ ultra_memory_free_block(card, &card->gf1.mem_alloc, 0),
+ ultra_memory_free_block(card, &card->gf1.mem_alloc, 1));
+#endif
+}
+#endif
diff --git a/sound/isa/gus/gus_mem_proc.c b/sound/isa/gus/gus_mem_proc.c
new file mode 100644
index 0000000..2803e22
--- /dev/null
+++ b/sound/isa/gus/gus_mem_proc.c
@@ -0,0 +1,134 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * GUS's memory access via proc filesystem
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#include <sound/info.h>
+
+struct gus_proc_private {
+ int rom; /* data are in ROM */
+ unsigned int address;
+ unsigned int size;
+ struct snd_gus_card * gus;
+};
+
+static long snd_gf1_mem_proc_dump(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, char __user *buf,
+ unsigned long count, unsigned long pos)
+{
+ long size;
+ struct gus_proc_private *priv = entry->private_data;
+ struct snd_gus_card *gus = priv->gus;
+ int err;
+
+ size = count;
+ if (pos + size > priv->size)
+ size = (long)priv->size - pos;
+ if (size > 0) {
+ if ((err = snd_gus_dram_read(gus, buf, pos, size, priv->rom)) < 0)
+ return err;
+ return size;
+ }
+ return 0;
+}
+
+static long long snd_gf1_mem_proc_llseek(struct snd_info_entry *entry,
+ void *private_file_data,
+ struct file *file,
+ long long offset,
+ int orig)
+{
+ struct gus_proc_private *priv = entry->private_data;
+
+ switch (orig) {
+ case SEEK_SET:
+ file->f_pos = offset;
+ break;
+ case SEEK_CUR:
+ file->f_pos += offset;
+ break;
+ case SEEK_END: /* offset is negative */
+ file->f_pos = priv->size + offset;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (file->f_pos > priv->size)
+ file->f_pos = priv->size;
+ return file->f_pos;
+}
+
+static void snd_gf1_mem_proc_free(struct snd_info_entry *entry)
+{
+ struct gus_proc_private *priv = entry->private_data;
+ kfree(priv);
+}
+
+static struct snd_info_entry_ops snd_gf1_mem_proc_ops = {
+ .read = snd_gf1_mem_proc_dump,
+ .llseek = snd_gf1_mem_proc_llseek,
+};
+
+int snd_gf1_mem_proc_init(struct snd_gus_card * gus)
+{
+ int idx;
+ char name[16];
+ struct gus_proc_private *priv;
+ struct snd_info_entry *entry;
+
+ for (idx = 0; idx < 4; idx++) {
+ if (gus->gf1.mem_alloc.banks_8[idx].size > 0) {
+ priv = kzalloc(sizeof(*priv), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+ priv->gus = gus;
+ sprintf(name, "gus-ram-%i", idx);
+ if (! snd_card_proc_new(gus->card, name, &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = priv;
+ entry->private_free = snd_gf1_mem_proc_free;
+ entry->c.ops = &snd_gf1_mem_proc_ops;
+ priv->address = gus->gf1.mem_alloc.banks_8[idx].address;
+ priv->size = entry->size = gus->gf1.mem_alloc.banks_8[idx].size;
+ }
+ }
+ }
+ for (idx = 0; idx < 4; idx++) {
+ if (gus->gf1.rom_present & (1 << idx)) {
+ priv = kzalloc(sizeof(*priv), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+ priv->rom = 1;
+ priv->gus = gus;
+ sprintf(name, "gus-rom-%i", idx);
+ if (! snd_card_proc_new(gus->card, name, &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = priv;
+ entry->private_free = snd_gf1_mem_proc_free;
+ entry->c.ops = &snd_gf1_mem_proc_ops;
+ priv->address = idx * 4096 * 1024;
+ priv->size = entry->size = gus->gf1.rom_memory;
+ }
+ }
+ }
+ return 0;
+}
diff --git a/sound/isa/gus/gus_mixer.c b/sound/isa/gus/gus_mixer.c
new file mode 100644
index 0000000..0dd4341
--- /dev/null
+++ b/sound/isa/gus/gus_mixer.c
@@ -0,0 +1,193 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of ICS 2101 chip and "mixer" in GF1 chip
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/gus.h>
+
+/*
+ *
+ */
+
+#define GF1_SINGLE(xname, xindex, shift, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_gf1_info_single, \
+ .get = snd_gf1_get_single, .put = snd_gf1_put_single, \
+ .private_value = shift | (invert << 8) }
+
+#define snd_gf1_info_single snd_ctl_boolean_mono_info
+
+static int snd_gf1_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value >> 8) & 1;
+
+ ucontrol->value.integer.value[0] = (gus->mix_cntrl_reg >> shift) & 1;
+ if (invert)
+ ucontrol->value.integer.value[0] ^= 1;
+ return 0;
+}
+
+static int snd_gf1_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ int change;
+ unsigned char oval, nval;
+
+ nval = ucontrol->value.integer.value[0] & 1;
+ if (invert)
+ nval ^= 1;
+ nval <<= shift;
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ oval = gus->mix_cntrl_reg;
+ nval = (oval & ~(1 << shift)) | nval;
+ change = nval != oval;
+ outb(gus->mix_cntrl_reg = nval, GUSP(gus, MIXCNTRLREG));
+ outb(gus->gf1.active_voice = 0, GUSP(gus, GF1PAGE));
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return change;
+}
+
+#define ICS_DOUBLE(xname, xindex, addr) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_ics_info_double, \
+ .get = snd_ics_get_double, .put = snd_ics_put_double, \
+ .private_value = addr }
+
+static int snd_ics_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 127;
+ return 0;
+}
+
+static int snd_ics_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int addr = kcontrol->private_value & 0xff;
+ unsigned char left, right;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ left = gus->gf1.ics_regs[addr][0];
+ right = gus->gf1.ics_regs[addr][1];
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ ucontrol->value.integer.value[0] = left & 127;
+ ucontrol->value.integer.value[1] = right & 127;
+ return 0;
+}
+
+static int snd_ics_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int addr = kcontrol->private_value & 0xff;
+ int change;
+ unsigned char val1, val2, oval1, oval2, tmp;
+
+ val1 = ucontrol->value.integer.value[0] & 127;
+ val2 = ucontrol->value.integer.value[1] & 127;
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ oval1 = gus->gf1.ics_regs[addr][0];
+ oval2 = gus->gf1.ics_regs[addr][1];
+ change = val1 != oval1 || val2 != oval2;
+ gus->gf1.ics_regs[addr][0] = val1;
+ gus->gf1.ics_regs[addr][1] = val2;
+ if (gus->ics_flag && gus->ics_flipped &&
+ (addr == SNDRV_ICS_GF1_DEV || addr == SNDRV_ICS_MASTER_DEV)) {
+ tmp = val1;
+ val1 = val2;
+ val2 = tmp;
+ }
+ addr <<= 3;
+ outb(addr | 0, GUSP(gus, MIXCNTRLPORT));
+ outb(1, GUSP(gus, MIXDATAPORT));
+ outb(addr | 2, GUSP(gus, MIXCNTRLPORT));
+ outb((unsigned char) val1, GUSP(gus, MIXDATAPORT));
+ outb(addr | 1, GUSP(gus, MIXCNTRLPORT));
+ outb(2, GUSP(gus, MIXDATAPORT));
+ outb(addr | 3, GUSP(gus, MIXCNTRLPORT));
+ outb((unsigned char) val2, GUSP(gus, MIXDATAPORT));
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_gf1_controls[] = {
+GF1_SINGLE("Master Playback Switch", 0, 1, 1),
+GF1_SINGLE("Line Switch", 0, 0, 1),
+GF1_SINGLE("Mic Switch", 0, 2, 0)
+};
+
+static struct snd_kcontrol_new snd_ics_controls[] = {
+GF1_SINGLE("Master Playback Switch", 0, 1, 1),
+ICS_DOUBLE("Master Playback Volume", 0, SNDRV_ICS_MASTER_DEV),
+ICS_DOUBLE("Synth Playback Volume", 0, SNDRV_ICS_GF1_DEV),
+GF1_SINGLE("Line Switch", 0, 0, 1),
+ICS_DOUBLE("Line Playback Volume", 0, SNDRV_ICS_LINE_DEV),
+GF1_SINGLE("Mic Switch", 0, 2, 0),
+ICS_DOUBLE("Mic Playback Volume", 0, SNDRV_ICS_MIC_DEV),
+ICS_DOUBLE("CD Playback Volume", 0, SNDRV_ICS_CD_DEV)
+};
+
+int snd_gf1_new_mixer(struct snd_gus_card * gus)
+{
+ struct snd_card *card;
+ unsigned int idx, max;
+ int err;
+
+ if (snd_BUG_ON(!gus))
+ return -EINVAL;
+ card = gus->card;
+ if (snd_BUG_ON(!card))
+ return -EINVAL;
+
+ if (gus->ics_flag)
+ snd_component_add(card, "ICS2101");
+ if (card->mixername[0] == '\0') {
+ strcpy(card->mixername, gus->ics_flag ? "GF1,ICS2101" : "GF1");
+ } else {
+ if (gus->ics_flag)
+ strcat(card->mixername, ",ICS2101");
+ strcat(card->mixername, ",GF1");
+ }
+
+ if (!gus->ics_flag) {
+ max = gus->ess_flag ? 1 : ARRAY_SIZE(snd_gf1_controls);
+ for (idx = 0; idx < max; idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_gf1_controls[idx], gus))) < 0)
+ return err;
+ }
+ } else {
+ for (idx = 0; idx < ARRAY_SIZE(snd_ics_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_ics_controls[idx], gus))) < 0)
+ return err;
+ }
+ }
+ return 0;
+}
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
new file mode 100644
index 0000000..38510ae
--- /dev/null
+++ b/sound/isa/gus/gus_pcm.c
@@ -0,0 +1,896 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of GF1 chip (PCM things)
+ *
+ * InterWave chips supports interleaved DMA, but this feature isn't used in
+ * this code.
+ *
+ * This code emulates autoinit DMA transfer for playback, recording by GF1
+ * chip doesn't support autoinit DMA.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/dma.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/gus.h>
+#include <sound/pcm_params.h>
+#include "gus_tables.h"
+
+/* maximum rate */
+
+#define SNDRV_GF1_PCM_RATE 48000
+
+#define SNDRV_GF1_PCM_PFLG_NONE 0
+#define SNDRV_GF1_PCM_PFLG_ACTIVE (1<<0)
+#define SNDRV_GF1_PCM_PFLG_NEUTRAL (2<<0)
+
+struct gus_pcm_private {
+ struct snd_gus_card * gus;
+ struct snd_pcm_substream *substream;
+ spinlock_t lock;
+ unsigned int voices;
+ struct snd_gus_voice *pvoices[2];
+ unsigned int memory;
+ unsigned short flags;
+ unsigned char voice_ctrl, ramp_ctrl;
+ unsigned int bpos;
+ unsigned int blocks;
+ unsigned int block_size;
+ unsigned int dma_size;
+ wait_queue_head_t sleep;
+ atomic_t dma_count;
+ int final_volume;
+};
+
+static int snd_gf1_pcm_use_dma = 1;
+
+static void snd_gf1_pcm_block_change_ack(struct snd_gus_card * gus, void *private_data)
+{
+ struct gus_pcm_private *pcmp = private_data;
+
+ if (pcmp) {
+ atomic_dec(&pcmp->dma_count);
+ wake_up(&pcmp->sleep);
+ }
+}
+
+static int snd_gf1_pcm_block_change(struct snd_pcm_substream *substream,
+ unsigned int offset,
+ unsigned int addr,
+ unsigned int count)
+{
+ struct snd_gf1_dma_block block;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+
+ count += offset & 31;
+ offset &= ~31;
+ // snd_printk("block change - offset = 0x%x, count = 0x%x\n", offset, count);
+ memset(&block, 0, sizeof(block));
+ block.cmd = SNDRV_GF1_DMA_IRQ;
+ if (snd_pcm_format_unsigned(runtime->format))
+ block.cmd |= SNDRV_GF1_DMA_UNSIGNED;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ block.cmd |= SNDRV_GF1_DMA_16BIT;
+ block.addr = addr & ~31;
+ block.buffer = runtime->dma_area + offset;
+ block.buf_addr = runtime->dma_addr + offset;
+ block.count = count;
+ block.private_data = pcmp;
+ block.ack = snd_gf1_pcm_block_change_ack;
+ if (!snd_gf1_dma_transfer_block(pcmp->gus, &block, 0, 0))
+ atomic_inc(&pcmp->dma_count);
+ return 0;
+}
+
+static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+ struct snd_gus_card * gus = pcmp->gus;
+ unsigned long flags;
+ unsigned char voice_ctrl, ramp_ctrl;
+ unsigned short rate;
+ unsigned int curr, begin, end;
+ unsigned short vol;
+ unsigned char pan;
+ unsigned int voice;
+
+ spin_lock_irqsave(&pcmp->lock, flags);
+ if (pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE) {
+ spin_unlock_irqrestore(&pcmp->lock, flags);
+ return;
+ }
+ pcmp->flags |= SNDRV_GF1_PCM_PFLG_ACTIVE;
+ pcmp->final_volume = 0;
+ spin_unlock_irqrestore(&pcmp->lock, flags);
+ rate = snd_gf1_translate_freq(gus, runtime->rate << 4);
+ /* enable WAVE IRQ */
+ voice_ctrl = snd_pcm_format_width(runtime->format) == 16 ? 0x24 : 0x20;
+ /* enable RAMP IRQ + rollover */
+ ramp_ctrl = 0x24;
+ if (pcmp->blocks == 1) {
+ voice_ctrl |= 0x08; /* loop enable */
+ ramp_ctrl &= ~0x04; /* disable rollover */
+ }
+ for (voice = 0; voice < pcmp->voices; voice++) {
+ begin = pcmp->memory + voice * (pcmp->dma_size / runtime->channels);
+ curr = begin + (pcmp->bpos * pcmp->block_size) / runtime->channels;
+ end = curr + (pcmp->block_size / runtime->channels);
+ end -= snd_pcm_format_width(runtime->format) == 16 ? 2 : 1;
+ // snd_printk("init: curr=0x%x, begin=0x%x, end=0x%x, ctrl=0x%x, ramp=0x%x, rate=0x%x\n", curr, begin, end, voice_ctrl, ramp_ctrl, rate);
+ pan = runtime->channels == 2 ? (!voice ? 1 : 14) : 8;
+ vol = !voice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_select_voice(gus, pcmp->pvoices[voice]->number);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_PAN, pan);
+ snd_gf1_write16(gus, SNDRV_GF1_VW_FREQUENCY, rate);
+ snd_gf1_write_addr(gus, SNDRV_GF1_VA_START, begin << 4, voice_ctrl & 4);
+ snd_gf1_write_addr(gus, SNDRV_GF1_VA_END, end << 4, voice_ctrl & 4);
+ snd_gf1_write_addr(gus, SNDRV_GF1_VA_CURRENT, curr << 4, voice_ctrl & 4);
+ snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, SNDRV_GF1_MIN_VOLUME << 4);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_RATE, 0x2f);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_START, SNDRV_GF1_MIN_OFFSET);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_END, vol >> 8);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, ramp_ctrl);
+ if (!gus->gf1.enh_mode) {
+ snd_gf1_delay(gus);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, ramp_ctrl);
+ }
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ }
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ for (voice = 0; voice < pcmp->voices; voice++) {
+ snd_gf1_select_voice(gus, pcmp->pvoices[voice]->number);
+ if (gus->gf1.enh_mode)
+ snd_gf1_write8(gus, SNDRV_GF1_VB_MODE, 0x00); /* deactivate voice */
+ snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice_ctrl);
+ voice_ctrl &= ~0x20;
+ }
+ voice_ctrl |= 0x20;
+ if (!gus->gf1.enh_mode) {
+ snd_gf1_delay(gus);
+ for (voice = 0; voice < pcmp->voices; voice++) {
+ snd_gf1_select_voice(gus, pcmp->pvoices[voice]->number);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice_ctrl);
+ voice_ctrl &= ~0x20; /* disable IRQ for next voice */
+ }
+ }
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+static void snd_gf1_pcm_interrupt_wave(struct snd_gus_card * gus,
+ struct snd_gus_voice *pvoice)
+{
+ struct gus_pcm_private * pcmp;
+ struct snd_pcm_runtime *runtime;
+ unsigned char voice_ctrl, ramp_ctrl;
+ unsigned int idx;
+ unsigned int end, step;
+
+ if (!pvoice->private_data) {
+ snd_printd("snd_gf1_pcm: unknown wave irq?\n");
+ snd_gf1_smart_stop_voice(gus, pvoice->number);
+ return;
+ }
+ pcmp = pvoice->private_data;
+ if (pcmp == NULL) {
+ snd_printd("snd_gf1_pcm: unknown wave irq?\n");
+ snd_gf1_smart_stop_voice(gus, pvoice->number);
+ return;
+ }
+ gus = pcmp->gus;
+ runtime = pcmp->substream->runtime;
+
+ spin_lock(&gus->reg_lock);
+ snd_gf1_select_voice(gus, pvoice->number);
+ voice_ctrl = snd_gf1_read8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL) & ~0x8b;
+ ramp_ctrl = (snd_gf1_read8(gus, SNDRV_GF1_VB_VOLUME_CONTROL) & ~0xa4) | 0x03;
+#if 0
+ snd_gf1_select_voice(gus, pvoice->number);
+ printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4));
+ snd_gf1_select_voice(gus, pcmp->pvoices[1]->number);
+ printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4));
+ snd_gf1_select_voice(gus, pvoice->number);
+#endif
+ pcmp->bpos++;
+ pcmp->bpos %= pcmp->blocks;
+ if (pcmp->bpos + 1 >= pcmp->blocks) { /* last block? */
+ voice_ctrl |= 0x08; /* enable loop */
+ } else {
+ ramp_ctrl |= 0x04; /* enable rollover */
+ }
+ end = pcmp->memory + (((pcmp->bpos + 1) * pcmp->block_size) / runtime->channels);
+ end -= voice_ctrl & 4 ? 2 : 1;
+ step = pcmp->dma_size / runtime->channels;
+ voice_ctrl |= 0x20;
+ if (!pcmp->final_volume) {
+ ramp_ctrl |= 0x20;
+ ramp_ctrl &= ~0x03;
+ }
+ for (idx = 0; idx < pcmp->voices; idx++, end += step) {
+ snd_gf1_select_voice(gus, pcmp->pvoices[idx]->number);
+ snd_gf1_write_addr(gus, SNDRV_GF1_VA_END, end << 4, voice_ctrl & 4);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice_ctrl);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, ramp_ctrl);
+ voice_ctrl &= ~0x20;
+ }
+ if (!gus->gf1.enh_mode) {
+ snd_gf1_delay(gus);
+ voice_ctrl |= 0x20;
+ for (idx = 0; idx < pcmp->voices; idx++) {
+ snd_gf1_select_voice(gus, pcmp->pvoices[idx]->number);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice_ctrl);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, ramp_ctrl);
+ voice_ctrl &= ~0x20;
+ }
+ }
+ spin_unlock(&gus->reg_lock);
+
+ snd_pcm_period_elapsed(pcmp->substream);
+#if 0
+ if ((runtime->flags & SNDRV_PCM_FLG_MMAP) &&
+ *runtime->state == SNDRV_PCM_STATE_RUNNING) {
+ end = pcmp->bpos * pcmp->block_size;
+ if (runtime->channels > 1) {
+ snd_gf1_pcm_block_change(pcmp->substream, end, pcmp->memory + (end / 2), pcmp->block_size / 2);
+ snd_gf1_pcm_block_change(pcmp->substream, end + (pcmp->block_size / 2), pcmp->memory + (pcmp->dma_size / 2) + (end / 2), pcmp->block_size / 2);
+ } else {
+ snd_gf1_pcm_block_change(pcmp->substream, end, pcmp->memory + end, pcmp->block_size);
+ }
+ }
+#endif
+}
+
+static void snd_gf1_pcm_interrupt_volume(struct snd_gus_card * gus,
+ struct snd_gus_voice * pvoice)
+{
+ unsigned short vol;
+ int cvoice;
+ struct gus_pcm_private *pcmp = pvoice->private_data;
+
+ /* stop ramp, but leave rollover bit untouched */
+ spin_lock(&gus->reg_lock);
+ snd_gf1_select_voice(gus, pvoice->number);
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
+ spin_unlock(&gus->reg_lock);
+ if (pcmp == NULL)
+ return;
+ /* are we active? */
+ if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE))
+ return;
+ /* load real volume - better precision */
+ cvoice = pcmp->pvoices[0] == pvoice ? 0 : 1;
+ if (pcmp->substream == NULL)
+ return;
+ vol = !cvoice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
+ spin_lock(&gus->reg_lock);
+ snd_gf1_select_voice(gus, pvoice->number);
+ snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol);
+ pcmp->final_volume = 1;
+ spin_unlock(&gus->reg_lock);
+}
+
+static void snd_gf1_pcm_volume_change(struct snd_gus_card * gus)
+{
+}
+
+static int snd_gf1_pcm_poke_block(struct snd_gus_card *gus, unsigned char *buf,
+ unsigned int pos, unsigned int count,
+ int w16, int invert)
+{
+ unsigned int len;
+ unsigned long flags;
+
+ // printk("poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", (int)buf, pos, count, gus->gf1.port);
+ while (count > 0) {
+ len = count;
+ if (len > 512) /* limit, to allow IRQ */
+ len = 512;
+ count -= len;
+ if (gus->interwave) {
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x01 | (invert ? 0x08 : 0x00));
+ snd_gf1_dram_addr(gus, pos);
+ if (w16) {
+ outb(SNDRV_GF1_GW_DRAM_IO16, GUSP(gus, GF1REGSEL));
+ outsw(GUSP(gus, GF1DATALOW), buf, len >> 1);
+ } else {
+ outsb(GUSP(gus, DRAM), buf, len);
+ }
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ buf += 512;
+ pos += 512;
+ } else {
+ invert = invert ? 0x80 : 0x00;
+ if (w16) {
+ len >>= 1;
+ while (len--) {
+ snd_gf1_poke(gus, pos++, *buf++);
+ snd_gf1_poke(gus, pos++, *buf++ ^ invert);
+ }
+ } else {
+ while (len--)
+ snd_gf1_poke(gus, pos++, *buf++ ^ invert);
+ }
+ }
+ if (count > 0 && !in_interrupt()) {
+ schedule_timeout_interruptible(1);
+ if (signal_pending(current))
+ return -EAGAIN;
+ }
+ }
+ return 0;
+}
+
+static int snd_gf1_pcm_playback_copy(struct snd_pcm_substream *substream,
+ int voice,
+ snd_pcm_uframes_t pos,
+ void __user *src,
+ snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+ unsigned int bpos, len;
+
+ bpos = samples_to_bytes(runtime, pos) + (voice * (pcmp->dma_size / 2));
+ len = samples_to_bytes(runtime, count);
+ if (snd_BUG_ON(bpos > pcmp->dma_size))
+ return -EIO;
+ if (snd_BUG_ON(bpos + len > pcmp->dma_size))
+ return -EIO;
+ if (copy_from_user(runtime->dma_area + bpos, src, len))
+ return -EFAULT;
+ if (snd_gf1_pcm_use_dma && len > 32) {
+ return snd_gf1_pcm_block_change(substream, bpos, pcmp->memory + bpos, len);
+ } else {
+ struct snd_gus_card *gus = pcmp->gus;
+ int err, w16, invert;
+
+ w16 = (snd_pcm_format_width(runtime->format) == 16);
+ invert = snd_pcm_format_unsigned(runtime->format);
+ if ((err = snd_gf1_pcm_poke_block(gus, runtime->dma_area + bpos, pcmp->memory + bpos, len, w16, invert)) < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int snd_gf1_pcm_playback_silence(struct snd_pcm_substream *substream,
+ int voice,
+ snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+ unsigned int bpos, len;
+
+ bpos = samples_to_bytes(runtime, pos) + (voice * (pcmp->dma_size / 2));
+ len = samples_to_bytes(runtime, count);
+ if (snd_BUG_ON(bpos > pcmp->dma_size))
+ return -EIO;
+ if (snd_BUG_ON(bpos + len > pcmp->dma_size))
+ return -EIO;
+ snd_pcm_format_set_silence(runtime->format, runtime->dma_area + bpos, count);
+ if (snd_gf1_pcm_use_dma && len > 32) {
+ return snd_gf1_pcm_block_change(substream, bpos, pcmp->memory + bpos, len);
+ } else {
+ struct snd_gus_card *gus = pcmp->gus;
+ int err, w16, invert;
+
+ w16 = (snd_pcm_format_width(runtime->format) == 16);
+ invert = snd_pcm_format_unsigned(runtime->format);
+ if ((err = snd_gf1_pcm_poke_block(gus, runtime->dma_area + bpos, pcmp->memory + bpos, len, w16, invert)) < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int snd_gf1_pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ if (err > 0) { /* change */
+ struct snd_gf1_mem_block *block;
+ if (pcmp->memory > 0) {
+ snd_gf1_mem_free(&gus->gf1.mem_alloc, pcmp->memory);
+ pcmp->memory = 0;
+ }
+ if ((block = snd_gf1_mem_alloc(&gus->gf1.mem_alloc,
+ SNDRV_GF1_MEM_OWNER_DRIVER,
+ "GF1 PCM",
+ runtime->dma_bytes, 1, 32,
+ NULL)) == NULL)
+ return -ENOMEM;
+ pcmp->memory = block->ptr;
+ }
+ pcmp->voices = params_channels(hw_params);
+ if (pcmp->pvoices[0] == NULL) {
+ if ((pcmp->pvoices[0] = snd_gf1_alloc_voice(pcmp->gus, SNDRV_GF1_VOICE_TYPE_PCM, 0, 0)) == NULL)
+ return -ENOMEM;
+ pcmp->pvoices[0]->handler_wave = snd_gf1_pcm_interrupt_wave;
+ pcmp->pvoices[0]->handler_volume = snd_gf1_pcm_interrupt_volume;
+ pcmp->pvoices[0]->volume_change = snd_gf1_pcm_volume_change;
+ pcmp->pvoices[0]->private_data = pcmp;
+ }
+ if (pcmp->voices > 1 && pcmp->pvoices[1] == NULL) {
+ if ((pcmp->pvoices[1] = snd_gf1_alloc_voice(pcmp->gus, SNDRV_GF1_VOICE_TYPE_PCM, 0, 0)) == NULL)
+ return -ENOMEM;
+ pcmp->pvoices[1]->handler_wave = snd_gf1_pcm_interrupt_wave;
+ pcmp->pvoices[1]->handler_volume = snd_gf1_pcm_interrupt_volume;
+ pcmp->pvoices[1]->volume_change = snd_gf1_pcm_volume_change;
+ pcmp->pvoices[1]->private_data = pcmp;
+ } else if (pcmp->voices == 1) {
+ if (pcmp->pvoices[1]) {
+ snd_gf1_free_voice(pcmp->gus, pcmp->pvoices[1]);
+ pcmp->pvoices[1] = NULL;
+ }
+ }
+ return 0;
+}
+
+static int snd_gf1_pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+
+ snd_pcm_lib_free_pages(substream);
+ if (pcmp->pvoices[0]) {
+ snd_gf1_free_voice(pcmp->gus, pcmp->pvoices[0]);
+ pcmp->pvoices[0] = NULL;
+ }
+ if (pcmp->pvoices[1]) {
+ snd_gf1_free_voice(pcmp->gus, pcmp->pvoices[1]);
+ pcmp->pvoices[1] = NULL;
+ }
+ if (pcmp->memory > 0) {
+ snd_gf1_mem_free(&pcmp->gus->gf1.mem_alloc, pcmp->memory);
+ pcmp->memory = 0;
+ }
+ return 0;
+}
+
+static int snd_gf1_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+
+ pcmp->bpos = 0;
+ pcmp->dma_size = snd_pcm_lib_buffer_bytes(substream);
+ pcmp->block_size = snd_pcm_lib_period_bytes(substream);
+ pcmp->blocks = pcmp->dma_size / pcmp->block_size;
+ return 0;
+}
+
+static int snd_gf1_pcm_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+ int voice;
+
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ snd_gf1_pcm_trigger_up(substream);
+ } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ spin_lock(&pcmp->lock);
+ pcmp->flags &= ~SNDRV_GF1_PCM_PFLG_ACTIVE;
+ spin_unlock(&pcmp->lock);
+ voice = pcmp->pvoices[0]->number;
+ snd_gf1_stop_voices(gus, voice, voice);
+ if (pcmp->pvoices[1]) {
+ voice = pcmp->pvoices[1]->number;
+ snd_gf1_stop_voices(gus, voice, voice);
+ }
+ } else {
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_gf1_pcm_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+ unsigned int pos;
+ unsigned char voice_ctrl;
+
+ pos = 0;
+ spin_lock(&gus->reg_lock);
+ if (pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE) {
+ snd_gf1_select_voice(gus, pcmp->pvoices[0]->number);
+ voice_ctrl = snd_gf1_read8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL);
+ pos = (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4) - pcmp->memory;
+ if (substream->runtime->channels > 1)
+ pos <<= 1;
+ pos = bytes_to_frames(runtime, pos);
+ }
+ spin_unlock(&gus->reg_lock);
+ return pos;
+}
+
+static struct snd_ratnum clock = {
+ .num = 9878400/16,
+ .den_min = 2,
+ .den_max = 257,
+ .den_step = 1,
+};
+
+static struct snd_pcm_hw_constraint_ratnums hw_constraints_clocks = {
+ .nrats = 1,
+ .rats = &clock,
+};
+
+static int snd_gf1_pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+
+ gus->c_dma_size = params_buffer_bytes(hw_params);
+ gus->c_period_size = params_period_bytes(hw_params);
+ gus->c_pos = 0;
+ gus->gf1.pcm_rcntrl_reg = 0x21; /* IRQ at end, enable & start */
+ if (params_channels(hw_params) > 1)
+ gus->gf1.pcm_rcntrl_reg |= 2;
+ if (gus->gf1.dma2 > 3)
+ gus->gf1.pcm_rcntrl_reg |= 4;
+ if (snd_pcm_format_unsigned(params_format(hw_params)))
+ gus->gf1.pcm_rcntrl_reg |= 0x80;
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_gf1_pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_gf1_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RECORD_RATE, runtime->rate_den - 2);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL, 0); /* disable sampling */
+ snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL); /* Sampling Control Register */
+ snd_dma_program(gus->gf1.dma2, runtime->dma_addr, gus->c_period_size, DMA_MODE_READ);
+ return 0;
+}
+
+static int snd_gf1_pcm_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ int val;
+
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ val = gus->gf1.pcm_rcntrl_reg;
+ } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ val = 0;
+ } else {
+ return -EINVAL;
+ }
+
+ spin_lock(&gus->reg_lock);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL, val);
+ snd_gf1_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL);
+ spin_unlock(&gus->reg_lock);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_gf1_pcm_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ int pos = snd_dma_pointer(gus->gf1.dma2, gus->c_period_size);
+ pos = bytes_to_frames(substream->runtime, (gus->c_pos + pos) % gus->c_dma_size);
+ return pos;
+}
+
+static void snd_gf1_pcm_interrupt_dma_read(struct snd_gus_card * gus)
+{
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL, 0); /* disable sampling */
+ snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL); /* Sampling Control Register */
+ if (gus->pcm_cap_substream != NULL) {
+ snd_gf1_pcm_capture_prepare(gus->pcm_cap_substream);
+ snd_gf1_pcm_capture_trigger(gus->pcm_cap_substream, SNDRV_PCM_TRIGGER_START);
+ gus->c_pos += gus->c_period_size;
+ snd_pcm_period_elapsed(gus->pcm_cap_substream);
+ }
+}
+
+static struct snd_pcm_hardware snd_gf1_pcm_playback =
+{
+ .info = SNDRV_PCM_INFO_NONINTERLEAVED,
+ .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5510,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_gf1_pcm_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_44100,
+ .rate_min = 5510,
+ .rate_max = 44100,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static void snd_gf1_pcm_playback_free(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static int snd_gf1_pcm_playback_open(struct snd_pcm_substream *substream)
+{
+ struct gus_pcm_private *pcmp;
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ pcmp = kzalloc(sizeof(*pcmp), GFP_KERNEL);
+ if (pcmp == NULL)
+ return -ENOMEM;
+ pcmp->gus = gus;
+ spin_lock_init(&pcmp->lock);
+ init_waitqueue_head(&pcmp->sleep);
+ atomic_set(&pcmp->dma_count, 0);
+
+ runtime->private_data = pcmp;
+ runtime->private_free = snd_gf1_pcm_playback_free;
+
+#if 0
+ printk("playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer);
+#endif
+ if ((err = snd_gf1_dma_init(gus)) < 0)
+ return err;
+ pcmp->flags = SNDRV_GF1_PCM_PFLG_NONE;
+ pcmp->substream = substream;
+ runtime->hw = snd_gf1_pcm_playback;
+ snd_pcm_limit_isa_dma_size(gus->gf1.dma1, &runtime->hw.buffer_bytes_max);
+ snd_pcm_limit_isa_dma_size(gus->gf1.dma1, &runtime->hw.period_bytes_max);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64);
+ return 0;
+}
+
+static int snd_gf1_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct gus_pcm_private *pcmp = runtime->private_data;
+
+ if (!wait_event_timeout(pcmp->sleep, (atomic_read(&pcmp->dma_count) <= 0), 2*HZ))
+ snd_printk(KERN_ERR "gf1 pcm - serious DMA problem\n");
+
+ snd_gf1_dma_done(gus);
+ return 0;
+}
+
+static int snd_gf1_pcm_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+
+ gus->gf1.interrupt_handler_dma_read = snd_gf1_pcm_interrupt_dma_read;
+ gus->pcm_cap_substream = substream;
+ substream->runtime->hw = snd_gf1_pcm_capture;
+ snd_pcm_limit_isa_dma_size(gus->gf1.dma2, &runtime->hw.buffer_bytes_max);
+ snd_pcm_limit_isa_dma_size(gus->gf1.dma2, &runtime->hw.period_bytes_max);
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_clocks);
+ return 0;
+}
+
+static int snd_gf1_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_gus_card *gus = snd_pcm_substream_chip(substream);
+
+ gus->pcm_cap_substream = NULL;
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_DMA_READ);
+ return 0;
+}
+
+static int snd_gf1_pcm_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 127;
+ return 0;
+}
+
+static int snd_gf1_pcm_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->pcm_volume_level_lock, flags);
+ ucontrol->value.integer.value[0] = gus->gf1.pcm_volume_level_left1;
+ ucontrol->value.integer.value[1] = gus->gf1.pcm_volume_level_right1;
+ spin_unlock_irqrestore(&gus->pcm_volume_level_lock, flags);
+ return 0;
+}
+
+static int snd_gf1_pcm_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_gus_card *gus = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned int idx;
+ unsigned short val1, val2, vol;
+ struct gus_pcm_private *pcmp;
+ struct snd_gus_voice *pvoice;
+
+ val1 = ucontrol->value.integer.value[0] & 127;
+ val2 = ucontrol->value.integer.value[1] & 127;
+ spin_lock_irqsave(&gus->pcm_volume_level_lock, flags);
+ change = val1 != gus->gf1.pcm_volume_level_left1 ||
+ val2 != gus->gf1.pcm_volume_level_right1;
+ gus->gf1.pcm_volume_level_left1 = val1;
+ gus->gf1.pcm_volume_level_right1 = val2;
+ gus->gf1.pcm_volume_level_left = snd_gf1_lvol_to_gvol_raw(val1 << 9) << 4;
+ gus->gf1.pcm_volume_level_right = snd_gf1_lvol_to_gvol_raw(val2 << 9) << 4;
+ spin_unlock_irqrestore(&gus->pcm_volume_level_lock, flags);
+ /* are we active? */
+ spin_lock_irqsave(&gus->voice_alloc, flags);
+ for (idx = 0; idx < 32; idx++) {
+ pvoice = &gus->gf1.voices[idx];
+ if (!pvoice->pcm)
+ continue;
+ pcmp = pvoice->private_data;
+ if (!(pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE))
+ continue;
+ /* load real volume - better precision */
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_select_voice(gus, pvoice->number);
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
+ vol = pvoice == pcmp->pvoices[0] ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
+ snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, vol);
+ pcmp->final_volume = 1;
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ }
+ spin_unlock_irqrestore(&gus->voice_alloc, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_gf1_pcm_volume_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Volume",
+ .info = snd_gf1_pcm_volume_info,
+ .get = snd_gf1_pcm_volume_get,
+ .put = snd_gf1_pcm_volume_put
+};
+
+static struct snd_kcontrol_new snd_gf1_pcm_volume_control1 =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "GPCM Playback Volume",
+ .info = snd_gf1_pcm_volume_info,
+ .get = snd_gf1_pcm_volume_get,
+ .put = snd_gf1_pcm_volume_put
+};
+
+static struct snd_pcm_ops snd_gf1_pcm_playback_ops = {
+ .open = snd_gf1_pcm_playback_open,
+ .close = snd_gf1_pcm_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_gf1_pcm_playback_hw_params,
+ .hw_free = snd_gf1_pcm_playback_hw_free,
+ .prepare = snd_gf1_pcm_playback_prepare,
+ .trigger = snd_gf1_pcm_playback_trigger,
+ .pointer = snd_gf1_pcm_playback_pointer,
+ .copy = snd_gf1_pcm_playback_copy,
+ .silence = snd_gf1_pcm_playback_silence,
+};
+
+static struct snd_pcm_ops snd_gf1_pcm_capture_ops = {
+ .open = snd_gf1_pcm_capture_open,
+ .close = snd_gf1_pcm_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_gf1_pcm_capture_hw_params,
+ .hw_free = snd_gf1_pcm_capture_hw_free,
+ .prepare = snd_gf1_pcm_capture_prepare,
+ .trigger = snd_gf1_pcm_capture_trigger,
+ .pointer = snd_gf1_pcm_capture_pointer,
+};
+
+int snd_gf1_pcm_new(struct snd_gus_card * gus, int pcm_dev, int control_index, struct snd_pcm ** rpcm)
+{
+ struct snd_card *card;
+ struct snd_kcontrol *kctl;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ int capture, err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ card = gus->card;
+ capture = !gus->interwave && !gus->ess_flag && !gus->ace_flag ? 1 : 0;
+ err = snd_pcm_new(card,
+ gus->interwave ? "AMD InterWave" : "GF1",
+ pcm_dev,
+ gus->gf1.pcm_channels / 2,
+ capture,
+ &pcm);
+ if (err < 0)
+ return err;
+ pcm->private_data = gus;
+ /* playback setup */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_gf1_pcm_playback_ops);
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next)
+ snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, gus->gf1.dma1 > 3 ? 128*1024 : 64*1024);
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ if (capture) {
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_gf1_pcm_capture_ops);
+ if (gus->gf1.dma2 == gus->gf1.dma1)
+ pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX;
+ snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
+ SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(),
+ 64*1024, gus->gf1.dma2 > 3 ? 128*1024 : 64*1024);
+ }
+ strcpy(pcm->name, pcm->id);
+ if (gus->interwave) {
+ sprintf(pcm->name + strlen(pcm->name), " rev %c", gus->revision + 'A');
+ }
+ strcat(pcm->name, " (synth)");
+ gus->pcm = pcm;
+
+ if (gus->codec_flag)
+ kctl = snd_ctl_new1(&snd_gf1_pcm_volume_control1, gus);
+ else
+ kctl = snd_ctl_new1(&snd_gf1_pcm_volume_control, gus);
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ return err;
+ kctl->id.index = control_index;
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c
new file mode 100644
index 0000000..3d1fed0
--- /dev/null
+++ b/sound/isa/gus/gus_reset.c
@@ -0,0 +1,413 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+
+extern void snd_gf1_timers_init(struct snd_gus_card * gus);
+extern void snd_gf1_timers_done(struct snd_gus_card * gus);
+extern int snd_gf1_synth_init(struct snd_gus_card * gus);
+extern void snd_gf1_synth_done(struct snd_gus_card * gus);
+
+/*
+ * ok.. default interrupt handlers...
+ */
+
+static void snd_gf1_default_interrupt_handler_midi_out(struct snd_gus_card * gus)
+{
+ snd_gf1_uart_cmd(gus, gus->gf1.uart_cmd &= ~0x20);
+}
+
+static void snd_gf1_default_interrupt_handler_midi_in(struct snd_gus_card * gus)
+{
+ snd_gf1_uart_cmd(gus, gus->gf1.uart_cmd &= ~0x80);
+}
+
+static void snd_gf1_default_interrupt_handler_timer1(struct snd_gus_card * gus)
+{
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, gus->gf1.timer_enabled &= ~4);
+}
+
+static void snd_gf1_default_interrupt_handler_timer2(struct snd_gus_card * gus)
+{
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, gus->gf1.timer_enabled &= ~8);
+}
+
+static void snd_gf1_default_interrupt_handler_wave_and_volume(struct snd_gus_card * gus, struct snd_gus_voice * voice)
+{
+ snd_gf1_i_ctrl_stop(gus, 0x00);
+ snd_gf1_i_ctrl_stop(gus, 0x0d);
+}
+
+static void snd_gf1_default_interrupt_handler_dma_write(struct snd_gus_card * gus)
+{
+ snd_gf1_i_write8(gus, 0x41, 0x00);
+}
+
+static void snd_gf1_default_interrupt_handler_dma_read(struct snd_gus_card * gus)
+{
+ snd_gf1_i_write8(gus, 0x49, 0x00);
+}
+
+void snd_gf1_set_default_handlers(struct snd_gus_card * gus, unsigned int what)
+{
+ if (what & SNDRV_GF1_HANDLER_MIDI_OUT)
+ gus->gf1.interrupt_handler_midi_out = snd_gf1_default_interrupt_handler_midi_out;
+ if (what & SNDRV_GF1_HANDLER_MIDI_IN)
+ gus->gf1.interrupt_handler_midi_in = snd_gf1_default_interrupt_handler_midi_in;
+ if (what & SNDRV_GF1_HANDLER_TIMER1)
+ gus->gf1.interrupt_handler_timer1 = snd_gf1_default_interrupt_handler_timer1;
+ if (what & SNDRV_GF1_HANDLER_TIMER2)
+ gus->gf1.interrupt_handler_timer2 = snd_gf1_default_interrupt_handler_timer2;
+ if (what & SNDRV_GF1_HANDLER_VOICE) {
+ struct snd_gus_voice *voice;
+
+ voice = &gus->gf1.voices[what & 0xffff];
+ voice->handler_wave =
+ voice->handler_volume = snd_gf1_default_interrupt_handler_wave_and_volume;
+ voice->handler_effect = NULL;
+ voice->volume_change = NULL;
+ }
+ if (what & SNDRV_GF1_HANDLER_DMA_WRITE)
+ gus->gf1.interrupt_handler_dma_write = snd_gf1_default_interrupt_handler_dma_write;
+ if (what & SNDRV_GF1_HANDLER_DMA_READ)
+ gus->gf1.interrupt_handler_dma_read = snd_gf1_default_interrupt_handler_dma_read;
+}
+
+/*
+
+ */
+
+static void snd_gf1_clear_regs(struct snd_gus_card * gus)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ inb(GUSP(gus, IRQSTAT));
+ snd_gf1_write8(gus, 0x41, 0); /* DRAM DMA Control Register */
+ snd_gf1_write8(gus, 0x45, 0); /* Timer Control */
+ snd_gf1_write8(gus, 0x49, 0); /* Sampling Control Register */
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+static void snd_gf1_look_regs(struct snd_gus_card * gus)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_look8(gus, 0x41); /* DRAM DMA Control Register */
+ snd_gf1_look8(gus, 0x49); /* Sampling Control Register */
+ inb(GUSP(gus, IRQSTAT));
+ snd_gf1_read8(gus, 0x0f); /* IRQ Source Register */
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+/*
+ * put selected GF1 voices to initial stage...
+ */
+
+void snd_gf1_smart_stop_voice(struct snd_gus_card * gus, unsigned short voice)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_select_voice(gus, voice);
+#if 0
+ printk(KERN_DEBUG " -%i- smart stop voice - volume = 0x%x\n", voice, snd_gf1_i_read16(gus, SNDRV_GF1_VW_VOLUME));
+#endif
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_ADDRESS_CONTROL);
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+}
+
+void snd_gf1_stop_voice(struct snd_gus_card * gus, unsigned short voice)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_select_voice(gus, voice);
+#if 0
+ printk(KERN_DEBUG " -%i- stop voice - volume = 0x%x\n", voice, snd_gf1_i_read16(gus, SNDRV_GF1_VW_VOLUME));
+#endif
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_ADDRESS_CONTROL);
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL);
+ if (gus->gf1.enh_mode)
+ snd_gf1_write8(gus, SNDRV_GF1_VB_ACCUMULATOR, 0);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+#if 0
+ snd_gf1_lfo_shutdown(gus, voice, ULTRA_LFO_VIBRATO);
+ snd_gf1_lfo_shutdown(gus, voice, ULTRA_LFO_TREMOLO);
+#endif
+}
+
+static void snd_gf1_clear_voices(struct snd_gus_card * gus, unsigned short v_min,
+ unsigned short v_max)
+{
+ unsigned long flags;
+ unsigned int daddr;
+ unsigned short i, w_16;
+
+ daddr = gus->gf1.default_voice_address << 4;
+ for (i = v_min; i <= v_max; i++) {
+#if 0
+ if (gus->gf1.syn_voices)
+ gus->gf1.syn_voices[i].flags = ~VFLG_DYNAMIC;
+#endif
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_select_voice(gus, i);
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_ADDRESS_CONTROL); /* Voice Control Register = voice stop */
+ snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); /* Volume Ramp Control Register = ramp off */
+ if (gus->gf1.enh_mode)
+ snd_gf1_write8(gus, SNDRV_GF1_VB_MODE, gus->gf1.memory ? 0x02 : 0x82); /* Deactivate voice */
+ w_16 = snd_gf1_read8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL) & 0x04;
+ snd_gf1_write16(gus, SNDRV_GF1_VW_FREQUENCY, 0x400);
+ snd_gf1_write_addr(gus, SNDRV_GF1_VA_START, daddr, w_16);
+ snd_gf1_write_addr(gus, SNDRV_GF1_VA_END, daddr, w_16);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_START, 0);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_END, 0);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_RATE, 0);
+ snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, 0);
+ snd_gf1_write_addr(gus, SNDRV_GF1_VA_CURRENT, daddr, w_16);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_PAN, 7);
+ if (gus->gf1.enh_mode) {
+ snd_gf1_write8(gus, SNDRV_GF1_VB_ACCUMULATOR, 0);
+ snd_gf1_write16(gus, SNDRV_GF1_VW_EFFECT_VOLUME, 0);
+ snd_gf1_write16(gus, SNDRV_GF1_VW_EFFECT_VOLUME_FINAL, 0);
+ }
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+#if 0
+ snd_gf1_lfo_shutdown(gus, i, ULTRA_LFO_VIBRATO);
+ snd_gf1_lfo_shutdown(gus, i, ULTRA_LFO_TREMOLO);
+#endif
+ }
+}
+
+void snd_gf1_stop_voices(struct snd_gus_card * gus, unsigned short v_min, unsigned short v_max)
+{
+ unsigned long flags;
+ short i, ramp_ok;
+ unsigned short ramp_end;
+
+ if (!in_interrupt()) { /* this can't be done in interrupt */
+ for (i = v_min, ramp_ok = 0; i <= v_max; i++) {
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_select_voice(gus, i);
+ ramp_end = snd_gf1_read16(gus, 9) >> 8;
+ if (ramp_end > SNDRV_GF1_MIN_OFFSET) {
+ ramp_ok++;
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_RATE, 20); /* ramp rate */
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_START, SNDRV_GF1_MIN_OFFSET); /* ramp start */
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_END, ramp_end); /* ramp end */
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, 0x40); /* ramp down */
+ if (gus->gf1.enh_mode) {
+ snd_gf1_delay(gus);
+ snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, 0x40);
+ }
+ }
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ }
+ msleep_interruptible(50);
+ }
+ snd_gf1_clear_voices(gus, v_min, v_max);
+}
+
+static void snd_gf1_alloc_voice_use(struct snd_gus_card * gus,
+ struct snd_gus_voice * pvoice,
+ int type, int client, int port)
+{
+ pvoice->use = 1;
+ switch (type) {
+ case SNDRV_GF1_VOICE_TYPE_PCM:
+ gus->gf1.pcm_alloc_voices++;
+ pvoice->pcm = 1;
+ break;
+ case SNDRV_GF1_VOICE_TYPE_SYNTH:
+ pvoice->synth = 1;
+ pvoice->client = client;
+ pvoice->port = port;
+ break;
+ case SNDRV_GF1_VOICE_TYPE_MIDI:
+ pvoice->midi = 1;
+ pvoice->client = client;
+ pvoice->port = port;
+ break;
+ }
+}
+
+struct snd_gus_voice *snd_gf1_alloc_voice(struct snd_gus_card * gus, int type, int client, int port)
+{
+ struct snd_gus_voice *pvoice;
+ unsigned long flags;
+ int idx;
+
+ spin_lock_irqsave(&gus->voice_alloc, flags);
+ if (type == SNDRV_GF1_VOICE_TYPE_PCM) {
+ if (gus->gf1.pcm_alloc_voices >= gus->gf1.pcm_channels) {
+ spin_unlock_irqrestore(&gus->voice_alloc, flags);
+ return NULL;
+ }
+ }
+ for (idx = 0; idx < 32; idx++) {
+ pvoice = &gus->gf1.voices[idx];
+ if (!pvoice->use) {
+ snd_gf1_alloc_voice_use(gus, pvoice, type, client, port);
+ spin_unlock_irqrestore(&gus->voice_alloc, flags);
+ return pvoice;
+ }
+ }
+ for (idx = 0; idx < 32; idx++) {
+ pvoice = &gus->gf1.voices[idx];
+ if (pvoice->midi && !pvoice->client) {
+ snd_gf1_clear_voices(gus, pvoice->number, pvoice->number);
+ snd_gf1_alloc_voice_use(gus, pvoice, type, client, port);
+ spin_unlock_irqrestore(&gus->voice_alloc, flags);
+ return pvoice;
+ }
+ }
+ spin_unlock_irqrestore(&gus->voice_alloc, flags);
+ return NULL;
+}
+
+void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice)
+{
+ unsigned long flags;
+ void (*private_free)(struct snd_gus_voice *voice);
+ void *private_data;
+
+ if (voice == NULL || !voice->use)
+ return;
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_VOICE | voice->number);
+ snd_gf1_clear_voices(gus, voice->number, voice->number);
+ spin_lock_irqsave(&gus->voice_alloc, flags);
+ private_free = voice->private_free;
+ private_data = voice->private_data;
+ voice->private_free = NULL;
+ voice->private_data = NULL;
+ if (voice->pcm)
+ gus->gf1.pcm_alloc_voices--;
+ voice->use = voice->pcm = 0;
+ voice->sample_ops = NULL;
+ spin_unlock_irqrestore(&gus->voice_alloc, flags);
+ if (private_free)
+ private_free(voice);
+}
+
+/*
+ * call this function only by start of driver
+ */
+
+int snd_gf1_start(struct snd_gus_card * gus)
+{
+ unsigned long flags;
+ unsigned int i;
+
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 1); /* disable IRQ & DAC */
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_JOYSTICK_DAC_LEVEL, gus->joystick_dac);
+
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_ALL);
+ for (i = 0; i < 32; i++) {
+ gus->gf1.voices[i].number = i;
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_VOICE | i);
+ }
+
+ snd_gf1_uart_cmd(gus, 0x03); /* huh.. this cleanup took me some time... */
+
+ if (gus->gf1.enh_mode) { /* enhanced mode !!!! */
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_GLOBAL_MODE, snd_gf1_i_look8(gus, SNDRV_GF1_GB_GLOBAL_MODE) | 0x01);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x01);
+ }
+ snd_gf1_clear_regs(gus);
+ snd_gf1_select_active_voices(gus);
+ snd_gf1_delay(gus);
+ gus->gf1.default_voice_address = gus->gf1.memory > 0 ? 0 : 512 - 8;
+ /* initialize LFOs & clear LFOs memory */
+ if (gus->gf1.enh_mode && gus->gf1.memory) {
+ gus->gf1.hw_lfo = 1;
+ gus->gf1.default_voice_address += 1024;
+ } else {
+ gus->gf1.sw_lfo = 1;
+ }
+#if 0
+ snd_gf1_lfo_init(gus);
+#endif
+ if (gus->gf1.memory > 0)
+ for (i = 0; i < 4; i++)
+ snd_gf1_poke(gus, gus->gf1.default_voice_address + i, 0);
+ snd_gf1_clear_regs(gus);
+ snd_gf1_clear_voices(gus, 0, 31);
+ snd_gf1_look_regs(gus);
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 7); /* Reset Register = IRQ enable, DAC enable */
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 7); /* Reset Register = IRQ enable, DAC enable */
+ if (gus->gf1.enh_mode) { /* enhanced mode !!!! */
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_GLOBAL_MODE, snd_gf1_i_look8(gus, SNDRV_GF1_GB_GLOBAL_MODE) | 0x01);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x01);
+ }
+ while ((snd_gf1_i_read8(gus, SNDRV_GF1_GB_VOICES_IRQ) & 0xc0) != 0xc0);
+
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ outb(gus->gf1.active_voice = 0, GUSP(gus, GF1PAGE));
+ outb(gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG));
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+
+ snd_gf1_timers_init(gus);
+ snd_gf1_look_regs(gus);
+ snd_gf1_mem_init(gus);
+ snd_gf1_mem_proc_init(gus);
+#ifdef CONFIG_SND_DEBUG
+ snd_gus_irq_profile_init(gus);
+#endif
+
+#if 0
+ if (gus->pnp_flag) {
+ if (gus->chip.playback_fifo_size > 0)
+ snd_gf1_i_write16(gus, SNDRV_GF1_GW_FIFO_RECORD_BASE_ADDR, gus->chip.playback_fifo_block->ptr >> 8);
+ if (gus->chip.record_fifo_size > 0)
+ snd_gf1_i_write16(gus, SNDRV_GF1_GW_FIFO_PLAY_BASE_ADDR, gus->chip.record_fifo_block->ptr >> 8);
+ snd_gf1_i_write16(gus, SNDRV_GF1_GW_FIFO_SIZE, gus->chip.interwave_fifo_reg);
+ }
+#endif
+
+ return 0;
+}
+
+/*
+ * call this function only by shutdown of driver
+ */
+
+int snd_gf1_stop(struct snd_gus_card * gus)
+{
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, 0); /* stop all timers */
+ snd_gf1_stop_voices(gus, 0, 31); /* stop all voices */
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 1); /* disable IRQ & DAC */
+ snd_gf1_timers_done(gus);
+ snd_gf1_mem_done(gus);
+#if 0
+ snd_gf1_lfo_done(gus);
+#endif
+ return 0;
+}
diff --git a/sound/isa/gus/gus_tables.h b/sound/isa/gus/gus_tables.h
new file mode 100644
index 0000000..42a4ca0
--- /dev/null
+++ b/sound/isa/gus/gus_tables.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define SNDRV_GF1_SCALE_TABLE_SIZE 128
+#define SNDRV_GF1_ATTEN_TABLE_SIZE 128
+
+#ifdef __GUS_TABLES_ALLOC__
+
+#if 0
+
+unsigned int snd_gf1_scale_table[SNDRV_GF1_SCALE_TABLE_SIZE] =
+{
+ 8372, 8870, 9397, 9956, 10548, 11175,
+ 11840, 12544, 13290, 14080, 14917, 15804,
+ 16744, 17740, 18795, 19912, 21096, 22351,
+ 23680, 25088, 26580, 28160, 29834, 31609,
+ 33488, 35479, 37589, 39824, 42192, 44701,
+ 47359, 50175, 53159, 56320, 59669, 63217,
+ 66976, 70959, 75178, 79649, 84385, 89402,
+ 94719, 100351, 106318, 112640, 119338, 126434,
+ 133952, 141918, 150356, 159297, 168769, 178805,
+ 189437, 200702, 212636, 225280, 238676, 252868,
+ 267905, 283835, 300713, 318594, 337539, 357610,
+ 378874, 401403, 425272, 450560, 477352, 505737,
+ 535809, 567670, 601425, 637188, 675077, 715219,
+ 757749, 802807, 850544, 901120, 954703, 1011473,
+ 1071618, 1135340, 1202851, 1274376, 1350154, 1430439,
+ 1515497, 1605613, 1701088, 1802240, 1909407, 2022946,
+ 2143237, 2270680, 2405702, 2548752, 2700309, 2860878,
+ 3030994, 3211227, 3402176, 3604480, 3818814, 4045892,
+ 4286473, 4541360, 4811404, 5097505, 5400618, 5721755,
+ 6061989, 6422453, 6804352, 7208960, 7637627, 8091784,
+ 8572947, 9082720, 9622807, 10195009, 10801236, 11443511,
+ 12123977, 12844906
+};
+
+#endif /* 0 */
+
+unsigned short snd_gf1_atten_table[SNDRV_GF1_ATTEN_TABLE_SIZE] = {
+ 4095 /* 0 */,1789 /* 1 */,1533 /* 2 */,1383 /* 3 */,1277 /* 4 */,
+ 1195 /* 5 */,1127 /* 6 */,1070 /* 7 */,1021 /* 8 */,978 /* 9 */,
+ 939 /* 10 */,903 /* 11 */,871 /* 12 */,842 /* 13 */,814 /* 14 */,
+ 789 /* 15 */,765 /* 16 */,743 /* 17 */,722 /* 18 */,702 /* 19 */,
+ 683 /* 20 */,665 /* 21 */,647 /* 22 */,631 /* 23 */,615 /* 24 */,
+ 600 /* 25 */,586 /* 26 */,572 /* 27 */,558 /* 28 */,545 /* 29 */,
+ 533 /* 30 */,521 /* 31 */,509 /* 32 */,498 /* 33 */,487 /* 34 */,
+ 476 /* 35 */,466 /* 36 */,455 /* 37 */,446 /* 38 */,436 /* 39 */,
+ 427 /* 40 */,418 /* 41 */,409 /* 42 */,400 /* 43 */,391 /* 44 */,
+ 383 /* 45 */,375 /* 46 */,367 /* 47 */,359 /* 48 */,352 /* 49 */,
+ 344 /* 50 */,337 /* 51 */,330 /* 52 */,323 /* 53 */,316 /* 54 */,
+ 309 /* 55 */,302 /* 56 */,296 /* 57 */,289 /* 58 */,283 /* 59 */,
+ 277 /* 60 */,271 /* 61 */,265 /* 62 */,259 /* 63 */,253 /* 64 */,
+ 247 /* 65 */,242 /* 66 */,236 /* 67 */,231 /* 68 */,225 /* 69 */,
+ 220 /* 70 */,215 /* 71 */,210 /* 72 */,205 /* 73 */,199 /* 74 */,
+ 195 /* 75 */,190 /* 76 */,185 /* 77 */,180 /* 78 */,175 /* 79 */,
+ 171 /* 80 */,166 /* 81 */,162 /* 82 */,157 /* 83 */,153 /* 84 */,
+ 148 /* 85 */,144 /* 86 */,140 /* 87 */,135 /* 88 */,131 /* 89 */,
+ 127 /* 90 */,123 /* 91 */,119 /* 92 */,115 /* 93 */,111 /* 94 */,
+ 107 /* 95 */,103 /* 96 */,100 /* 97 */,96 /* 98 */,92 /* 99 */,
+ 88 /* 100 */,85 /* 101 */,81 /* 102 */,77 /* 103 */,74 /* 104 */,
+ 70 /* 105 */,67 /* 106 */,63 /* 107 */,60 /* 108 */,56 /* 109 */,
+ 53 /* 110 */,50 /* 111 */,46 /* 112 */,43 /* 113 */,40 /* 114 */,
+ 37 /* 115 */,33 /* 116 */,30 /* 117 */,27 /* 118 */,24 /* 119 */,
+ 21 /* 120 */,18 /* 121 */,15 /* 122 */,12 /* 123 */,9 /* 124 */,
+ 6 /* 125 */,3 /* 126 */,0 /* 127 */,
+};
+
+#else
+
+extern unsigned int snd_gf1_scale_table[SNDRV_GF1_SCALE_TABLE_SIZE];
+extern unsigned short snd_gf1_atten_table[SNDRV_GF1_ATTEN_TABLE_SIZE];
+
+#endif
diff --git a/sound/isa/gus/gus_timer.c b/sound/isa/gus/gus_timer.c
new file mode 100644
index 0000000..c537271
--- /dev/null
+++ b/sound/isa/gus/gus_timer.c
@@ -0,0 +1,203 @@
+/*
+ * Routines for Gravis UltraSound soundcards - Timers
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ * GUS have similar timers as AdLib (OPL2/OPL3 chips).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+
+/*
+ * Timer 1 - 80us
+ */
+
+static int snd_gf1_timer1_start(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ unsigned int ticks;
+ struct snd_gus_card *gus;
+
+ gus = snd_timer_chip(timer);
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ ticks = timer->sticks;
+ tmp = (gus->gf1.timer_enabled |= 4);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_ADLIB_TIMER_1, 256 - ticks); /* timer 1 count */
+ snd_gf1_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, tmp); /* enable timer 1 IRQ */
+ snd_gf1_adlib_write(gus, 0x04, tmp >> 2); /* timer 2 start */
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return 0;
+}
+
+static int snd_gf1_timer1_stop(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ struct snd_gus_card *gus;
+
+ gus = snd_timer_chip(timer);
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ tmp = (gus->gf1.timer_enabled &= ~4);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, tmp); /* disable timer #1 */
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return 0;
+}
+
+/*
+ * Timer 2 - 320us
+ */
+
+static int snd_gf1_timer2_start(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ unsigned int ticks;
+ struct snd_gus_card *gus;
+
+ gus = snd_timer_chip(timer);
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ ticks = timer->sticks;
+ tmp = (gus->gf1.timer_enabled |= 8);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_ADLIB_TIMER_2, 256 - ticks); /* timer 2 count */
+ snd_gf1_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, tmp); /* enable timer 2 IRQ */
+ snd_gf1_adlib_write(gus, 0x04, tmp >> 2); /* timer 2 start */
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return 0;
+}
+
+static int snd_gf1_timer2_stop(struct snd_timer * timer)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ struct snd_gus_card *gus;
+
+ gus = snd_timer_chip(timer);
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ tmp = (gus->gf1.timer_enabled &= ~8);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, tmp); /* disable timer #1 */
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ return 0;
+}
+
+/*
+
+ */
+
+static void snd_gf1_interrupt_timer1(struct snd_gus_card * gus)
+{
+ struct snd_timer *timer = gus->gf1.timer1;
+
+ if (timer == NULL)
+ return;
+ snd_timer_interrupt(timer, timer->sticks);
+}
+
+static void snd_gf1_interrupt_timer2(struct snd_gus_card * gus)
+{
+ struct snd_timer *timer = gus->gf1.timer2;
+
+ if (timer == NULL)
+ return;
+ snd_timer_interrupt(timer, timer->sticks);
+}
+
+/*
+
+ */
+
+static struct snd_timer_hardware snd_gf1_timer1 =
+{
+ .flags = SNDRV_TIMER_HW_STOP,
+ .resolution = 80000,
+ .ticks = 256,
+ .start = snd_gf1_timer1_start,
+ .stop = snd_gf1_timer1_stop,
+};
+
+static struct snd_timer_hardware snd_gf1_timer2 =
+{
+ .flags = SNDRV_TIMER_HW_STOP,
+ .resolution = 320000,
+ .ticks = 256,
+ .start = snd_gf1_timer2_start,
+ .stop = snd_gf1_timer2_stop,
+};
+
+static void snd_gf1_timer1_free(struct snd_timer *timer)
+{
+ struct snd_gus_card *gus = timer->private_data;
+ gus->gf1.timer1 = NULL;
+}
+
+static void snd_gf1_timer2_free(struct snd_timer *timer)
+{
+ struct snd_gus_card *gus = timer->private_data;
+ gus->gf1.timer2 = NULL;
+}
+
+void snd_gf1_timers_init(struct snd_gus_card * gus)
+{
+ struct snd_timer *timer;
+ struct snd_timer_id tid;
+
+ if (gus->gf1.timer1 != NULL || gus->gf1.timer2 != NULL)
+ return;
+
+ gus->gf1.interrupt_handler_timer1 = snd_gf1_interrupt_timer1;
+ gus->gf1.interrupt_handler_timer2 = snd_gf1_interrupt_timer2;
+
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = gus->card->number;
+ tid.device = gus->timer_dev;
+ tid.subdevice = 0;
+
+ if (snd_timer_new(gus->card, "GF1 timer", &tid, &timer) >= 0) {
+ strcpy(timer->name, "GF1 timer #1");
+ timer->private_data = gus;
+ timer->private_free = snd_gf1_timer1_free;
+ timer->hw = snd_gf1_timer1;
+ }
+ gus->gf1.timer1 = timer;
+
+ tid.device++;
+
+ if (snd_timer_new(gus->card, "GF1 timer", &tid, &timer) >= 0) {
+ strcpy(timer->name, "GF1 timer #2");
+ timer->private_data = gus;
+ timer->private_free = snd_gf1_timer2_free;
+ timer->hw = snd_gf1_timer2;
+ }
+ gus->gf1.timer2 = timer;
+}
+
+void snd_gf1_timers_done(struct snd_gus_card * gus)
+{
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_TIMER1 | SNDRV_GF1_HANDLER_TIMER2);
+ if (gus->gf1.timer1) {
+ snd_device_free(gus->card, gus->gf1.timer1);
+ gus->gf1.timer1 = NULL;
+ }
+ if (gus->gf1.timer2) {
+ snd_device_free(gus->card, gus->gf1.timer2);
+ gus->gf1.timer2 = NULL;
+ }
+}
diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c
new file mode 100644
index 0000000..f0af3f7
--- /dev/null
+++ b/sound/isa/gus/gus_uart.c
@@ -0,0 +1,256 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for the GF1 MIDI interface - like UART 6850
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+
+static void snd_gf1_interrupt_midi_in(struct snd_gus_card * gus)
+{
+ int count;
+ unsigned char stat, data, byte;
+ unsigned long flags;
+
+ count = 10;
+ while (count) {
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ stat = snd_gf1_uart_stat(gus);
+ if (!(stat & 0x01)) { /* data in Rx FIFO? */
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ count--;
+ continue;
+ }
+ count = 100; /* arm counter to new value */
+ data = snd_gf1_uart_get(gus);
+ if (!(gus->gf1.uart_cmd & 0x80)) {
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ continue;
+ }
+ if (stat & 0x10) { /* framing error */
+ gus->gf1.uart_framing++;
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ continue;
+ }
+ byte = snd_gf1_uart_get(gus);
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ snd_rawmidi_receive(gus->midi_substream_input, &byte, 1);
+ if (stat & 0x20) {
+ gus->gf1.uart_overrun++;
+ }
+ }
+}
+
+static void snd_gf1_interrupt_midi_out(struct snd_gus_card * gus)
+{
+ char byte;
+ unsigned long flags;
+
+ /* try unlock output */
+ if (snd_gf1_uart_stat(gus) & 0x01)
+ snd_gf1_interrupt_midi_in(gus);
+
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (snd_gf1_uart_stat(gus) & 0x02) { /* Tx FIFO free? */
+ if (snd_rawmidi_transmit(gus->midi_substream_output, &byte, 1) != 1) { /* no other bytes or error */
+ snd_gf1_uart_cmd(gus, gus->gf1.uart_cmd & ~0x20); /* disable Tx interrupt */
+ } else {
+ snd_gf1_uart_put(gus, byte);
+ }
+ }
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+}
+
+static void snd_gf1_uart_reset(struct snd_gus_card * gus, int close)
+{
+ snd_gf1_uart_cmd(gus, 0x03); /* reset */
+ if (!close && gus->uart_enable) {
+ udelay(160);
+ snd_gf1_uart_cmd(gus, 0x00); /* normal operations */
+ }
+}
+
+static int snd_gf1_uart_output_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_gus_card *gus;
+
+ gus = substream->rmidi->private_data;
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (!(gus->gf1.uart_cmd & 0x80)) { /* input active? */
+ snd_gf1_uart_reset(gus, 0);
+ }
+ gus->gf1.interrupt_handler_midi_out = snd_gf1_interrupt_midi_out;
+ gus->midi_substream_output = substream;
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+#if 0
+ snd_printk(KERN_DEBUG "write init - cmd = 0x%x, stat = 0x%x\n", gus->gf1.uart_cmd, snd_gf1_uart_stat(gus));
+#endif
+ return 0;
+}
+
+static int snd_gf1_uart_input_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_gus_card *gus;
+ int i;
+
+ gus = substream->rmidi->private_data;
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (gus->gf1.interrupt_handler_midi_out != snd_gf1_interrupt_midi_out) {
+ snd_gf1_uart_reset(gus, 0);
+ }
+ gus->gf1.interrupt_handler_midi_in = snd_gf1_interrupt_midi_in;
+ gus->midi_substream_input = substream;
+ if (gus->uart_enable) {
+ for (i = 0; i < 1000 && (snd_gf1_uart_stat(gus) & 0x01); i++)
+ snd_gf1_uart_get(gus); /* clean Rx */
+ if (i >= 1000)
+ snd_printk(KERN_ERR "gus midi uart init read - cleanup error\n");
+ }
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+#if 0
+ snd_printk("read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus));
+ snd_printk("[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x (page = 0x%x)\n", gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102));
+#endif
+ return 0;
+}
+
+static int snd_gf1_uart_output_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_gus_card *gus;
+
+ gus = substream->rmidi->private_data;
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (gus->gf1.interrupt_handler_midi_in != snd_gf1_interrupt_midi_in)
+ snd_gf1_uart_reset(gus, 1);
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_MIDI_OUT);
+ gus->midi_substream_output = NULL;
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ return 0;
+}
+
+static int snd_gf1_uart_input_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_gus_card *gus;
+
+ gus = substream->rmidi->private_data;
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (gus->gf1.interrupt_handler_midi_out != snd_gf1_interrupt_midi_out)
+ snd_gf1_uart_reset(gus, 1);
+ snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_MIDI_IN);
+ gus->midi_substream_input = NULL;
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ return 0;
+}
+
+static void snd_gf1_uart_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_gus_card *gus;
+ unsigned long flags;
+
+ gus = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (up) {
+ if ((gus->gf1.uart_cmd & 0x80) == 0)
+ snd_gf1_uart_cmd(gus, gus->gf1.uart_cmd | 0x80); /* enable Rx interrupts */
+ } else {
+ if (gus->gf1.uart_cmd & 0x80)
+ snd_gf1_uart_cmd(gus, gus->gf1.uart_cmd & ~0x80); /* disable Rx interrupts */
+ }
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+}
+
+static void snd_gf1_uart_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct snd_gus_card *gus;
+ char byte;
+ int timeout;
+
+ gus = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (up) {
+ if ((gus->gf1.uart_cmd & 0x20) == 0) {
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ /* wait for empty Rx - Tx is probably unlocked */
+ timeout = 10000;
+ while (timeout-- > 0 && snd_gf1_uart_stat(gus) & 0x01);
+ /* Tx FIFO free? */
+ spin_lock_irqsave(&gus->uart_cmd_lock, flags);
+ if (gus->gf1.uart_cmd & 0x20) {
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ return;
+ }
+ if (snd_gf1_uart_stat(gus) & 0x02) {
+ if (snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+ return;
+ }
+ snd_gf1_uart_put(gus, byte);
+ }
+ snd_gf1_uart_cmd(gus, gus->gf1.uart_cmd | 0x20); /* enable Tx interrupt */
+ }
+ } else {
+ if (gus->gf1.uart_cmd & 0x20)
+ snd_gf1_uart_cmd(gus, gus->gf1.uart_cmd & ~0x20);
+ }
+ spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
+}
+
+static struct snd_rawmidi_ops snd_gf1_uart_output =
+{
+ .open = snd_gf1_uart_output_open,
+ .close = snd_gf1_uart_output_close,
+ .trigger = snd_gf1_uart_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_gf1_uart_input =
+{
+ .open = snd_gf1_uart_input_open,
+ .close = snd_gf1_uart_input_close,
+ .trigger = snd_gf1_uart_input_trigger,
+};
+
+int snd_gf1_rawmidi_new(struct snd_gus_card * gus, int device, struct snd_rawmidi ** rrawmidi)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ if ((err = snd_rawmidi_new(gus->card, "GF1", device, 1, 1, &rmidi)) < 0)
+ return err;
+ strcpy(rmidi->name, gus->interwave ? "AMD InterWave" : "GF1");
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_gf1_uart_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_gf1_uart_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = gus;
+ gus->midi_uart = rmidi;
+ if (rrawmidi)
+ *rrawmidi = rmidi;
+ return err;
+}
diff --git a/sound/isa/gus/gus_volume.c b/sound/isa/gus/gus_volume.c
new file mode 100644
index 0000000..c3c028a
--- /dev/null
+++ b/sound/isa/gus/gus_volume.c
@@ -0,0 +1,217 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#define __GUS_TABLES_ALLOC__
+#include "gus_tables.h"
+
+EXPORT_SYMBOL(snd_gf1_atten_table); /* for snd-gus-synth module */
+
+unsigned short snd_gf1_lvol_to_gvol_raw(unsigned int vol)
+{
+ unsigned short e, m, tmp;
+
+ if (vol > 65535)
+ vol = 65535;
+ tmp = vol;
+ e = 7;
+ if (tmp < 128) {
+ while (e > 0 && tmp < (1 << e))
+ e--;
+ } else {
+ while (tmp > 255) {
+ tmp >>= 1;
+ e++;
+ }
+ }
+ m = vol - (1 << e);
+ if (m > 0) {
+ if (e > 8)
+ m >>= e - 8;
+ else if (e < 8)
+ m <<= 8 - e;
+ m &= 255;
+ }
+ return (e << 8) | m;
+}
+
+#if 0
+
+unsigned int snd_gf1_gvol_to_lvol_raw(unsigned short gf1_vol)
+{
+ unsigned int rvol;
+ unsigned short e, m;
+
+ if (!gf1_vol)
+ return 0;
+ e = gf1_vol >> 8;
+ m = (unsigned char) gf1_vol;
+ rvol = 1 << e;
+ if (e > 8)
+ return rvol | (m << (e - 8));
+ return rvol | (m >> (8 - e));
+}
+
+unsigned int snd_gf1_calc_ramp_rate(struct snd_gus_card * gus,
+ unsigned short start,
+ unsigned short end,
+ unsigned int us)
+{
+ static unsigned char vol_rates[19] =
+ {
+ 23, 24, 26, 28, 29, 31, 32, 34,
+ 36, 37, 39, 40, 42, 44, 45, 47,
+ 49, 50, 52
+ };
+ unsigned short range, increment, value, i;
+
+ start >>= 4;
+ end >>= 4;
+ if (start < end)
+ us /= end - start;
+ else
+ us /= start - end;
+ range = 4;
+ value = gus->gf1.enh_mode ?
+ vol_rates[0] :
+ vol_rates[gus->gf1.active_voices - 14];
+ for (i = 0; i < 3; i++) {
+ if (us < value) {
+ range = i;
+ break;
+ } else
+ value <<= 3;
+ }
+ if (range == 4) {
+ range = 3;
+ increment = 1;
+ } else
+ increment = (value + (value >> 1)) / us;
+ return (range << 6) | (increment & 0x3f);
+}
+
+#endif /* 0 */
+
+unsigned short snd_gf1_translate_freq(struct snd_gus_card * gus, unsigned int freq16)
+{
+ freq16 >>= 3;
+ if (freq16 < 50)
+ freq16 = 50;
+ if (freq16 & 0xf8000000) {
+ freq16 = ~0xf8000000;
+ snd_printk(KERN_ERR "snd_gf1_translate_freq: overflow - freq = 0x%x\n", freq16);
+ }
+ return ((freq16 << 9) + (gus->gf1.playback_freq >> 1)) / gus->gf1.playback_freq;
+}
+
+#if 0
+
+short snd_gf1_compute_vibrato(short cents, unsigned short fc_register)
+{
+ static short vibrato_table[] =
+ {
+ 0, 0, 32, 592, 61, 1175, 93, 1808,
+ 124, 2433, 152, 3007, 182, 3632, 213, 4290,
+ 241, 4834, 255, 5200
+ };
+
+ long depth;
+ short *vi1, *vi2, pcents, v1;
+
+ pcents = cents < 0 ? -cents : cents;
+ for (vi1 = vibrato_table, vi2 = vi1 + 2; pcents > *vi2; vi1 = vi2, vi2 += 2);
+ v1 = *(vi1 + 1);
+ /* The FC table above is a list of pairs. The first number in the pair */
+ /* is the cents index from 0-255 cents, and the second number in the */
+ /* pair is the FC adjustment needed to change the pitch by the indexed */
+ /* number of cents. The table was created for an FC of 32768. */
+ /* The following expression does a linear interpolation against the */
+ /* approximated log curve in the table above, and then scales the number */
+ /* by the FC before the LFO. This calculation also adjusts the output */
+ /* value to produce the appropriate depth for the hardware. The depth */
+ /* is 2 * desired FC + 1. */
+ depth = (((int) (*(vi2 + 1) - *vi1) * (pcents - *vi1) / (*vi2 - *vi1)) + v1) * fc_register >> 14;
+ if (depth)
+ depth++;
+ if (depth > 255)
+ depth = 255;
+ return cents < 0 ? -(short) depth : (short) depth;
+}
+
+unsigned short snd_gf1_compute_pitchbend(unsigned short pitchbend, unsigned short sens)
+{
+ static long log_table[] = {1024, 1085, 1149, 1218, 1290, 1367, 1448, 1534, 1625, 1722, 1825, 1933};
+ int wheel, sensitivity;
+ unsigned int mantissa, f1, f2;
+ unsigned short semitones, f1_index, f2_index, f1_power, f2_power;
+ char bend_down = 0;
+ int bend;
+
+ if (!sens)
+ return 1024;
+ wheel = (int) pitchbend - 8192;
+ sensitivity = ((int) sens * wheel) / 128;
+ if (sensitivity < 0) {
+ bend_down = 1;
+ sensitivity = -sensitivity;
+ }
+ semitones = (unsigned int) (sensitivity >> 13);
+ mantissa = sensitivity % 8192;
+ f1_index = semitones % 12;
+ f2_index = (semitones + 1) % 12;
+ f1_power = semitones / 12;
+ f2_power = (semitones + 1) / 12;
+ f1 = log_table[f1_index] << f1_power;
+ f2 = log_table[f2_index] << f2_power;
+ bend = (int) ((((f2 - f1) * mantissa) >> 13) + f1);
+ if (bend_down)
+ bend = 1048576L / bend;
+ return bend;
+}
+
+unsigned short snd_gf1_compute_freq(unsigned int freq,
+ unsigned int rate,
+ unsigned short mix_rate)
+{
+ unsigned int fc;
+ int scale = 0;
+
+ while (freq >= 4194304L) {
+ scale++;
+ freq >>= 1;
+ }
+ fc = (freq << 10) / rate;
+ if (fc > 97391L) {
+ fc = 97391;
+ snd_printk(KERN_ERR "patch: (1) fc frequency overflow - %u\n", fc);
+ }
+ fc = (fc * 44100UL) / mix_rate;
+ while (scale--)
+ fc <<= 1;
+ if (fc > 65535L) {
+ fc = 65535;
+ snd_printk(KERN_ERR "patch: (2) fc frequency overflow - %u\n", fc);
+ }
+ return (unsigned short) fc;
+}
+
+#endif /* 0 */
diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c
new file mode 100644
index 0000000..426532a
--- /dev/null
+++ b/sound/isa/gus/gusclassic.c
@@ -0,0 +1,245 @@
+/*
+ * Driver for Gravis UltraSound Classic soundcard
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <linux/moduleparam.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+#define CRD_NAME "Gravis UltraSound Classic"
+#define DEV_NAME "gusclassic"
+
+MODULE_DESCRIPTION(CRD_NAME);
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Classic}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x230,0x240,0x250,0x260 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 3,5,9,11,12,15 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */
+static int joystick_dac[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 29};
+ /* 0 to 31, (0.59V-4.52V or 0.389V-2.98V) */
+static int channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 24};
+static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for " CRD_NAME " driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for " CRD_NAME " driver.");
+module_param_array(joystick_dac, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for " CRD_NAME " driver.");
+module_param_array(channels, int, NULL, 0444);
+MODULE_PARM_DESC(channels, "GF1 channels for " CRD_NAME " driver.");
+module_param_array(pcm_channels, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_channels, "Reserved PCM channels for " CRD_NAME " driver.");
+
+static int __devinit snd_gusclassic_match(struct device *dev, unsigned int n)
+{
+ return enable[n];
+}
+
+static int __devinit snd_gusclassic_create(struct snd_card *card,
+ struct device *dev, unsigned int n, struct snd_gus_card **rgus)
+{
+ static long possible_ports[] = {0x220, 0x230, 0x240, 0x250, 0x260};
+ static int possible_irqs[] = {5, 11, 12, 9, 7, 15, 3, 4, -1};
+ static int possible_dmas[] = {5, 6, 7, 1, 3, -1};
+
+ int i, error;
+
+ if (irq[n] == SNDRV_AUTO_IRQ) {
+ irq[n] = snd_legacy_find_free_irq(possible_irqs);
+ if (irq[n] < 0) {
+ dev_err(dev, "unable to find a free IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (dma1[n] == SNDRV_AUTO_DMA) {
+ dma1[n] = snd_legacy_find_free_dma(possible_dmas);
+ if (dma1[n] < 0) {
+ dev_err(dev, "unable to find a free DMA1\n");
+ return -EBUSY;
+ }
+ }
+ if (dma2[n] == SNDRV_AUTO_DMA) {
+ dma2[n] = snd_legacy_find_free_dma(possible_dmas);
+ if (dma2[n] < 0) {
+ dev_err(dev, "unable to find a free DMA2\n");
+ return -EBUSY;
+ }
+ }
+
+ if (port[n] != SNDRV_AUTO_PORT)
+ return snd_gus_create(card, port[n], irq[n], dma1[n], dma2[n],
+ 0, channels[n], pcm_channels[n], 0, rgus);
+
+ i = 0;
+ do {
+ port[n] = possible_ports[i];
+ error = snd_gus_create(card, port[n], irq[n], dma1[n], dma2[n],
+ 0, channels[n], pcm_channels[n], 0, rgus);
+ } while (error < 0 && ++i < ARRAY_SIZE(possible_ports));
+
+ return error;
+}
+
+static int __devinit snd_gusclassic_detect(struct snd_gus_card *gus)
+{
+ unsigned char d;
+
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 0) {
+ snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d);
+ return -ENODEV;
+ }
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 1); /* release reset */
+ udelay(160);
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 1) {
+ snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d);
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n)
+{
+ struct snd_card *card;
+ struct snd_gus_card *gus;
+ int error;
+
+ card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
+ if (!card)
+ return -EINVAL;
+
+ if (pcm_channels[n] < 2)
+ pcm_channels[n] = 2;
+
+ error = snd_gusclassic_create(card, dev, n, &gus);
+ if (error < 0)
+ goto out;
+
+ error = snd_gusclassic_detect(gus);
+ if (error < 0)
+ goto out;
+
+ gus->joystick_dac = joystick_dac[n];
+
+ error = snd_gus_initialize(gus);
+ if (error < 0)
+ goto out;
+
+ error = -ENODEV;
+ if (gus->max_flag || gus->ess_flag) {
+ dev_err(dev, "GUS Classic or ACE soundcard was "
+ "not detected at 0x%lx\n", gus->gf1.port);
+ goto out;
+ }
+
+ error = snd_gf1_new_mixer(gus);
+ if (error < 0)
+ goto out;
+
+ error = snd_gf1_pcm_new(gus, 0, 0, NULL);
+ if (error < 0)
+ goto out;
+
+ if (!gus->ace_flag) {
+ error = snd_gf1_rawmidi_new(gus, 0, NULL);
+ if (error < 0)
+ goto out;
+ }
+
+ sprintf(card->longname + strlen(card->longname),
+ " at 0x%lx, irq %d, dma %d",
+ gus->gf1.port, gus->gf1.irq, gus->gf1.dma1);
+
+ if (gus->gf1.dma2 >= 0)
+ sprintf(card->longname + strlen(card->longname),
+ "&%d", gus->gf1.dma2);
+
+ snd_card_set_dev(card, dev);
+
+ error = snd_card_register(card);
+ if (error < 0)
+ goto out;
+
+ dev_set_drvdata(dev, card);
+ return 0;
+
+out: snd_card_free(card);
+ return error;
+}
+
+static int __devexit snd_gusclassic_remove(struct device *dev, unsigned int n)
+{
+ snd_card_free(dev_get_drvdata(dev));
+ dev_set_drvdata(dev, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_gusclassic_driver = {
+ .match = snd_gusclassic_match,
+ .probe = snd_gusclassic_probe,
+ .remove = __devexit_p(snd_gusclassic_remove),
+#if 0 /* FIXME */
+ .suspend = snd_gusclassic_suspend,
+ .remove = snd_gusclassic_remove,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ }
+};
+
+static int __init alsa_card_gusclassic_init(void)
+{
+ return isa_register_driver(&snd_gusclassic_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_gusclassic_exit(void)
+{
+ isa_unregister_driver(&snd_gusclassic_driver);
+}
+
+module_init(alsa_card_gusclassic_init);
+module_exit(alsa_card_gusclassic_exit);
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
new file mode 100644
index 0000000..7ad4c3b
--- /dev/null
+++ b/sound/isa/gus/gusextreme.c
@@ -0,0 +1,372 @@
+/*
+ * Driver for Gravis UltraSound Extreme soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <linux/moduleparam.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#include <sound/es1688.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#define SNDRV_LEGACY_AUTO_PROBE
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+#define CRD_NAME "Gravis UltraSound Extreme"
+#define DEV_NAME "gusextreme"
+
+MODULE_DESCRIPTION(CRD_NAME);
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Extreme}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */
+static long gf1_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS) - 1] = -1}; /* 0x210,0x220,0x230,0x240,0x250,0x260,0x270 */
+static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS) - 1] = -1}; /* 0x300,0x310,0x320 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */
+static int gf1_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 2,3,5,9,11,12,15 */
+static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
+static int joystick_dac[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 29};
+ /* 0 to 31, (0.59V-4.52V or 0.389V-2.98V) */
+static int channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 24};
+static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver.");
+module_param_array(gf1_port, long, NULL, 0444);
+MODULE_PARM_DESC(gf1_port, "GF1 port # for " CRD_NAME " driver (optional).");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver.");
+module_param_array(mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver.");
+module_param_array(gf1_irq, int, NULL, 0444);
+MODULE_PARM_DESC(gf1_irq, "GF1 IRQ # for " CRD_NAME " driver.");
+module_param_array(dma8, int, NULL, 0444);
+MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "GF1 DMA # for " CRD_NAME " driver.");
+module_param_array(joystick_dac, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for " CRD_NAME " driver.");
+module_param_array(channels, int, NULL, 0444);
+MODULE_PARM_DESC(channels, "GF1 channels for " CRD_NAME " driver.");
+module_param_array(pcm_channels, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_channels, "Reserved PCM channels for " CRD_NAME " driver.");
+
+static int __devinit snd_gusextreme_match(struct device *dev, unsigned int n)
+{
+ return enable[n];
+}
+
+static int __devinit snd_gusextreme_es1688_create(struct snd_card *card,
+ struct device *dev, unsigned int n, struct snd_es1688 **rchip)
+{
+ static long possible_ports[] = {0x220, 0x240, 0x260};
+ static int possible_irqs[] = {5, 9, 10, 7, -1};
+ static int possible_dmas[] = {1, 3, 0, -1};
+
+ int i, error;
+
+ if (irq[n] == SNDRV_AUTO_IRQ) {
+ irq[n] = snd_legacy_find_free_irq(possible_irqs);
+ if (irq[n] < 0) {
+ dev_err(dev, "unable to find a free IRQ for ES1688\n");
+ return -EBUSY;
+ }
+ }
+ if (dma8[n] == SNDRV_AUTO_DMA) {
+ dma8[n] = snd_legacy_find_free_dma(possible_dmas);
+ if (dma8[n] < 0) {
+ dev_err(dev, "unable to find a free DMA for ES1688\n");
+ return -EBUSY;
+ }
+ }
+
+ if (port[n] != SNDRV_AUTO_PORT)
+ return snd_es1688_create(card, port[n], mpu_port[n], irq[n],
+ mpu_irq[n], dma8[n], ES1688_HW_1688, rchip);
+
+ i = 0;
+ do {
+ port[n] = possible_ports[i];
+ error = snd_es1688_create(card, port[n], mpu_port[n], irq[n],
+ mpu_irq[n], dma8[n], ES1688_HW_1688, rchip);
+ } while (error < 0 && ++i < ARRAY_SIZE(possible_ports));
+
+ return error;
+}
+
+static int __devinit snd_gusextreme_gus_card_create(struct snd_card *card,
+ struct device *dev, unsigned int n, struct snd_gus_card **rgus)
+{
+ static int possible_irqs[] = {11, 12, 15, 9, 5, 7, 3, -1};
+ static int possible_dmas[] = {5, 6, 7, 3, 1, -1};
+
+ if (gf1_irq[n] == SNDRV_AUTO_IRQ) {
+ gf1_irq[n] = snd_legacy_find_free_irq(possible_irqs);
+ if (gf1_irq[n] < 0) {
+ dev_err(dev, "unable to find a free IRQ for GF1\n");
+ return -EBUSY;
+ }
+ }
+ if (dma1[n] == SNDRV_AUTO_DMA) {
+ dma1[n] = snd_legacy_find_free_dma(possible_dmas);
+ if (dma1[n] < 0) {
+ dev_err(dev, "unable to find a free DMA for GF1\n");
+ return -EBUSY;
+ }
+ }
+ return snd_gus_create(card, gf1_port[n], gf1_irq[n], dma1[n], -1,
+ 0, channels[n], pcm_channels[n], 0, rgus);
+}
+
+static int __devinit snd_gusextreme_detect(struct snd_gus_card *gus,
+ struct snd_es1688 *es1688)
+{
+ unsigned long flags;
+ unsigned char d;
+
+ /*
+ * This is main stuff - enable access to GF1 chip...
+ * I'm not sure, if this will work for card which have
+ * ES1688 chip in another place than 0x220.
+ *
+ * I used reverse-engineering in DOSEMU. [--jk]
+ *
+ * ULTRINIT.EXE:
+ * 0x230 = 0,2,3
+ * 0x240 = 2,0,1
+ * 0x250 = 2,0,3
+ * 0x260 = 2,2,1
+ */
+
+ spin_lock_irqsave(&es1688->mixer_lock, flags);
+ snd_es1688_mixer_write(es1688, 0x40, 0x0b); /* don't change!!! */
+ spin_unlock_irqrestore(&es1688->mixer_lock, flags);
+
+ spin_lock_irqsave(&es1688->reg_lock, flags);
+ outb(gus->gf1.port & 0x040 ? 2 : 0, ES1688P(es1688, INIT1));
+ outb(0, 0x201);
+ outb(gus->gf1.port & 0x020 ? 2 : 0, ES1688P(es1688, INIT1));
+ outb(0, 0x201);
+ outb(gus->gf1.port & 0x010 ? 3 : 1, ES1688P(es1688, INIT1));
+ spin_unlock_irqrestore(&es1688->reg_lock, flags);
+
+ udelay(100);
+
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 0) {
+ snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d);
+ return -EIO;
+ }
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 1); /* release reset */
+ udelay(160);
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 1) {
+ snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d);
+ return -EIO;
+ }
+
+ return 0;
+}
+
+static int __devinit snd_gusextreme_mixer(struct snd_es1688 *chip)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ctl_elem_id id1, id2;
+ int error;
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+
+ /* reassign AUX to SYNTHESIZER */
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "Synth Playback Volume");
+ error = snd_ctl_rename_id(card, &id1, &id2);
+ if (error < 0)
+ return error;
+
+ /* reassign Master Playback Switch to Synth Playback Switch */
+ strcpy(id1.name, "Master Playback Switch");
+ strcpy(id2.name, "Synth Playback Switch");
+ error = snd_ctl_rename_id(card, &id1, &id2);
+ if (error < 0)
+ return error;
+
+ return 0;
+}
+
+static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
+{
+ struct snd_card *card;
+ struct snd_gus_card *gus;
+ struct snd_es1688 *es1688;
+ struct snd_opl3 *opl3;
+ int error;
+
+ card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
+ if (!card)
+ return -EINVAL;
+
+ if (mpu_port[n] == SNDRV_AUTO_PORT)
+ mpu_port[n] = 0;
+
+ if (mpu_irq[n] > 15)
+ mpu_irq[n] = -1;
+
+ error = snd_gusextreme_es1688_create(card, dev, n, &es1688);
+ if (error < 0)
+ goto out;
+
+ if (gf1_port[n] < 0)
+ gf1_port[n] = es1688->port + 0x20;
+
+ error = snd_gusextreme_gus_card_create(card, dev, n, &gus);
+ if (error < 0)
+ goto out;
+
+ error = snd_gusextreme_detect(gus, es1688);
+ if (error < 0)
+ goto out;
+
+ gus->joystick_dac = joystick_dac[n];
+
+ error = snd_gus_initialize(gus);
+ if (error < 0)
+ goto out;
+
+ error = -ENODEV;
+ if (!gus->ess_flag) {
+ dev_err(dev, "GUS Extreme soundcard was not "
+ "detected at 0x%lx\n", gus->gf1.port);
+ goto out;
+ }
+ gus->codec_flag = 1;
+
+ error = snd_es1688_pcm(es1688, 0, NULL);
+ if (error < 0)
+ goto out;
+
+ error = snd_es1688_mixer(es1688);
+ if (error < 0)
+ goto out;
+
+ snd_component_add(card, "ES1688");
+
+ if (pcm_channels[n] > 0) {
+ error = snd_gf1_pcm_new(gus, 1, 1, NULL);
+ if (error < 0)
+ goto out;
+ }
+
+ error = snd_gf1_new_mixer(gus);
+ if (error < 0)
+ goto out;
+
+ error = snd_gusextreme_mixer(es1688);
+ if (error < 0)
+ goto out;
+
+ if (snd_opl3_create(card, es1688->port, es1688->port + 2,
+ OPL3_HW_OPL3, 0, &opl3) < 0)
+ dev_warn(dev, "opl3 not detected at 0x%lx\n", es1688->port);
+ else {
+ error = snd_opl3_hwdep_new(opl3, 0, 2, NULL);
+ if (error < 0)
+ goto out;
+ }
+
+ if (es1688->mpu_port >= 0x300) {
+ error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688,
+ es1688->mpu_port, 0,
+ mpu_irq[n], IRQF_DISABLED, NULL);
+ if (error < 0)
+ goto out;
+ }
+
+ sprintf(card->longname, "Gravis UltraSound Extreme at 0x%lx, "
+ "irq %i&%i, dma %i&%i", es1688->port,
+ gus->gf1.irq, es1688->irq, gus->gf1.dma1, es1688->dma8);
+
+ snd_card_set_dev(card, dev);
+
+ error = snd_card_register(card);
+ if (error < 0)
+ goto out;
+
+ dev_set_drvdata(dev, card);
+ return 0;
+
+out: snd_card_free(card);
+ return error;
+}
+
+static int __devexit snd_gusextreme_remove(struct device *dev, unsigned int n)
+{
+ snd_card_free(dev_get_drvdata(dev));
+ dev_set_drvdata(dev, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_gusextreme_driver = {
+ .match = snd_gusextreme_match,
+ .probe = snd_gusextreme_probe,
+ .remove = snd_gusextreme_remove,
+#if 0 /* FIXME */
+ .suspend = snd_gusextreme_suspend,
+ .resume = snd_gusextreme_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ }
+};
+
+static int __init alsa_card_gusextreme_init(void)
+{
+ return isa_register_driver(&snd_gusextreme_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_gusextreme_exit(void)
+{
+ isa_unregister_driver(&snd_gusextreme_driver);
+}
+
+module_init(alsa_card_gusextreme_init);
+module_exit(alsa_card_gusextreme_exit);
diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c
new file mode 100644
index 0000000..f94c197
--- /dev/null
+++ b/sound/isa/gus/gusmax.c
@@ -0,0 +1,387 @@
+/*
+ * Driver for Gravis UltraSound MAX soundcard
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <linux/moduleparam.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#include <sound/wss.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Gravis UltraSound MAX");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound MAX}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x230,0x240,0x250,0x260 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 2,3,5,9,11,12,15 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */
+static int joystick_dac[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 29};
+ /* 0 to 31, (0.59V-4.52V or 0.389V-2.98V) */
+static int channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 24};
+static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for GUS MAX soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for GUS MAX soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable GUS MAX soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for GUS MAX driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for GUS MAX driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for GUS MAX driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for GUS MAX driver.");
+module_param_array(joystick_dac, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for GUS MAX driver.");
+module_param_array(channels, int, NULL, 0444);
+MODULE_PARM_DESC(channels, "Used GF1 channels for GUS MAX driver.");
+module_param_array(pcm_channels, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_channels, "Reserved PCM channels for GUS MAX driver.");
+
+struct snd_gusmax {
+ int irq;
+ struct snd_card *card;
+ struct snd_gus_card *gus;
+ struct snd_wss *wss;
+ unsigned short gus_status_reg;
+ unsigned short pcm_status_reg;
+};
+
+#define PFX "gusmax: "
+
+static int __devinit snd_gusmax_detect(struct snd_gus_card * gus)
+{
+ unsigned char d;
+
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 0) {
+ snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d);
+ return -ENODEV;
+ }
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 1); /* release reset */
+ udelay(160);
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 1) {
+ snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d);
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static irqreturn_t snd_gusmax_interrupt(int irq, void *dev_id)
+{
+ struct snd_gusmax *maxcard = dev_id;
+ int loop, max = 5;
+ int handled = 0;
+
+ do {
+ loop = 0;
+ if (inb(maxcard->gus_status_reg)) {
+ handled = 1;
+ snd_gus_interrupt(irq, maxcard->gus);
+ loop++;
+ }
+ if (inb(maxcard->pcm_status_reg) & 0x01) { /* IRQ bit is set? */
+ handled = 1;
+ snd_wss_interrupt(irq, maxcard->wss);
+ loop++;
+ }
+ } while (loop && --max > 0);
+ return IRQ_RETVAL(handled);
+}
+
+static void __devinit snd_gusmax_init(int dev, struct snd_card *card,
+ struct snd_gus_card * gus)
+{
+ gus->equal_irq = 1;
+ gus->codec_flag = 1;
+ gus->joystick_dac = joystick_dac[dev];
+ /* init control register */
+ gus->max_cntrl_val = (gus->gf1.port >> 4) & 0x0f;
+ if (gus->gf1.dma1 > 3)
+ gus->max_cntrl_val |= 0x10;
+ if (gus->gf1.dma2 > 3)
+ gus->max_cntrl_val |= 0x20;
+ gus->max_cntrl_val |= 0x40;
+ outb(gus->max_cntrl_val, GUSP(gus, MAXCNTRLPORT));
+}
+
+static int __devinit snd_gusmax_mixer(struct snd_wss *chip)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ctl_elem_id id1, id2;
+ int err;
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ /* reassign AUXA to SYNTHESIZER */
+ strcpy(id1.name, "Aux Playback Switch");
+ strcpy(id2.name, "Synth Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "Synth Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ /* reassign AUXB to CD */
+ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1;
+ strcpy(id2.name, "CD Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "CD Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+#if 0
+ /* reassign Mono Input to MIC */
+ if (snd_mixer_group_rename(mixer,
+ SNDRV_MIXER_IN_MONO, 0,
+ SNDRV_MIXER_IN_MIC, 0) < 0)
+ goto __error;
+ if (snd_mixer_elem_rename(mixer,
+ SNDRV_MIXER_IN_MONO, 0, SNDRV_MIXER_ETYPE_INPUT,
+ SNDRV_MIXER_IN_MIC, 0) < 0)
+ goto __error;
+ if (snd_mixer_elem_rename(mixer,
+ "Mono Capture Volume", 0, SNDRV_MIXER_ETYPE_VOLUME1,
+ "Mic Capture Volume", 0) < 0)
+ goto __error;
+ if (snd_mixer_elem_rename(mixer,
+ "Mono Capture Switch", 0, SNDRV_MIXER_ETYPE_SWITCH1,
+ "Mic Capture Switch", 0) < 0)
+ goto __error;
+#endif
+ return 0;
+}
+
+static void snd_gusmax_free(struct snd_card *card)
+{
+ struct snd_gusmax *maxcard = (struct snd_gusmax *)card->private_data;
+
+ if (maxcard == NULL)
+ return;
+ if (maxcard->irq >= 0)
+ free_irq(maxcard->irq, (void *)maxcard);
+}
+
+static int __devinit snd_gusmax_match(struct device *pdev, unsigned int dev)
+{
+ return enable[dev];
+}
+
+static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev)
+{
+ static int possible_irqs[] = {5, 11, 12, 9, 7, 15, 3, -1};
+ static int possible_dmas[] = {5, 6, 7, 1, 3, -1};
+ int xirq, xdma1, xdma2, err;
+ struct snd_card *card;
+ struct snd_gus_card *gus = NULL;
+ struct snd_wss *wss;
+ struct snd_gusmax *maxcard;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_gusmax));
+ if (card == NULL)
+ return -ENOMEM;
+ card->private_free = snd_gusmax_free;
+ maxcard = (struct snd_gusmax *)card->private_data;
+ maxcard->card = card;
+ maxcard->irq = -1;
+
+ xirq = irq[dev];
+ if (xirq == SNDRV_AUTO_IRQ) {
+ if ((xirq = snd_legacy_find_free_irq(possible_irqs)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free IRQ\n");
+ err = -EBUSY;
+ goto _err;
+ }
+ }
+ xdma1 = dma1[dev];
+ if (xdma1 == SNDRV_AUTO_DMA) {
+ if ((xdma1 = snd_legacy_find_free_dma(possible_dmas)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA1\n");
+ err = -EBUSY;
+ goto _err;
+ }
+ }
+ xdma2 = dma2[dev];
+ if (xdma2 == SNDRV_AUTO_DMA) {
+ if ((xdma2 = snd_legacy_find_free_dma(possible_dmas)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA2\n");
+ err = -EBUSY;
+ goto _err;
+ }
+ }
+
+ if (port[dev] != SNDRV_AUTO_PORT) {
+ err = snd_gus_create(card,
+ port[dev],
+ -xirq, xdma1, xdma2,
+ 0, channels[dev],
+ pcm_channels[dev],
+ 0, &gus);
+ } else {
+ static unsigned long possible_ports[] = {
+ 0x220, 0x230, 0x240, 0x250, 0x260
+ };
+ int i;
+ for (i = 0; i < ARRAY_SIZE(possible_ports); i++) {
+ err = snd_gus_create(card,
+ possible_ports[i],
+ -xirq, xdma1, xdma2,
+ 0, channels[dev],
+ pcm_channels[dev],
+ 0, &gus);
+ if (err >= 0) {
+ port[dev] = possible_ports[i];
+ break;
+ }
+ }
+ }
+ if (err < 0)
+ goto _err;
+
+ if ((err = snd_gusmax_detect(gus)) < 0)
+ goto _err;
+
+ maxcard->gus_status_reg = gus->gf1.reg_irqstat;
+ maxcard->pcm_status_reg = gus->gf1.port + 0x10c + 2;
+ snd_gusmax_init(dev, card, gus);
+ if ((err = snd_gus_initialize(gus)) < 0)
+ goto _err;
+
+ if (!gus->max_flag) {
+ snd_printk(KERN_ERR PFX "GUS MAX soundcard was not detected at 0x%lx\n", gus->gf1.port);
+ err = -ENODEV;
+ goto _err;
+ }
+
+ if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) {
+ snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
+ err = -EBUSY;
+ goto _err;
+ }
+ maxcard->irq = xirq;
+
+ err = snd_wss_create(card,
+ gus->gf1.port + 0x10c, -1, xirq,
+ xdma2 < 0 ? xdma1 : xdma2, xdma1,
+ WSS_HW_DETECT,
+ WSS_HWSHARE_IRQ |
+ WSS_HWSHARE_DMA1 |
+ WSS_HWSHARE_DMA2,
+ &wss);
+ if (err < 0)
+ goto _err;
+
+ err = snd_wss_pcm(wss, 0, NULL);
+ if (err < 0)
+ goto _err;
+
+ err = snd_wss_mixer(wss);
+ if (err < 0)
+ goto _err;
+
+ err = snd_wss_timer(wss, 2, NULL);
+ if (err < 0)
+ goto _err;
+
+ if (pcm_channels[dev] > 0) {
+ if ((err = snd_gf1_pcm_new(gus, 1, 1, NULL)) < 0)
+ goto _err;
+ }
+ err = snd_gusmax_mixer(wss);
+ if (err < 0)
+ goto _err;
+
+ err = snd_gf1_rawmidi_new(gus, 0, NULL);
+ if (err < 0)
+ goto _err;
+
+ sprintf(card->longname + strlen(card->longname), " at 0x%lx, irq %i, dma %i", gus->gf1.port, xirq, xdma1);
+ if (xdma2 >= 0)
+ sprintf(card->longname + strlen(card->longname), "&%i", xdma2);
+
+ snd_card_set_dev(card, pdev);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto _err;
+
+ maxcard->gus = gus;
+ maxcard->wss = wss;
+
+ dev_set_drvdata(pdev, card);
+ return 0;
+
+ _err:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_gusmax_remove(struct device *devptr, unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define DEV_NAME "gusmax"
+
+static struct isa_driver snd_gusmax_driver = {
+ .match = snd_gusmax_match,
+ .probe = snd_gusmax_probe,
+ .remove = __devexit_p(snd_gusmax_remove),
+ /* FIXME: suspend/resume */
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+static int __init alsa_card_gusmax_init(void)
+{
+ return isa_register_driver(&snd_gusmax_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_gusmax_exit(void)
+{
+ isa_unregister_driver(&snd_gusmax_driver);
+}
+
+module_init(alsa_card_gusmax_init)
+module_exit(alsa_card_gusmax_exit)
diff --git a/sound/isa/gus/interwave-stb.c b/sound/isa/gus/interwave-stb.c
new file mode 100644
index 0000000..dbe4f48
--- /dev/null
+++ b/sound/isa/gus/interwave-stb.c
@@ -0,0 +1,2 @@
+#define SNDRV_STB
+#include "interwave.c"
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
new file mode 100644
index 0000000..5faecfb
--- /dev/null
+++ b/sound/isa/gus/interwave.c
@@ -0,0 +1,944 @@
+/*
+ * Driver for AMD InterWave soundcard
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * 1999/07/22 Erik Inge Bolso <knan@mo.himolde.no>
+ * * mixer group handlers
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/gus.h>
+#include <sound/wss.h>
+#ifdef SNDRV_STB
+#include <sound/tea6330t.h>
+#endif
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+#ifndef SNDRV_STB
+MODULE_DESCRIPTION("AMD InterWave");
+MODULE_SUPPORTED_DEVICE("{{Gravis,UltraSound Plug & Play},"
+ "{STB,SoundRage32},"
+ "{MED,MED3210},"
+ "{Dynasonix,Dynasonix Pro},"
+ "{Panasonic,PCA761AW}}");
+#else
+MODULE_DESCRIPTION("AMD InterWave STB with TEA6330T");
+MODULE_SUPPORTED_DEVICE("{{AMD,InterWave STB with TEA6330T}}");
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x210,0x220,0x230,0x240,0x250,0x260 */
+#ifdef SNDRV_STB
+static long port_tc[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x350,0x360,0x370,0x380 */
+#endif
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 2,3,5,9,11,12,15 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+static int joystick_dac[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 29};
+ /* 0 to 31, (0.59V-4.52V or 0.389V-2.98V) */
+static int midi[SNDRV_CARDS];
+static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+static int effect[SNDRV_CARDS];
+
+#ifdef SNDRV_STB
+#define PFX "interwave-stb: "
+#define INTERWAVE_DRIVER "snd_interwave_stb"
+#define INTERWAVE_PNP_DRIVER "interwave-stb"
+#else
+#define PFX "interwave: "
+#define INTERWAVE_DRIVER "snd_interwave"
+#define INTERWAVE_PNP_DRIVER "interwave"
+#endif
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for InterWave soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for InterWave soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable InterWave soundcard.");
+#ifdef CONFIG_PNP
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard.");
+#endif
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for InterWave driver.");
+#ifdef SNDRV_STB
+module_param_array(port_tc, long, NULL, 0444);
+MODULE_PARM_DESC(port_tc, "Tone control (TEA6330T - i2c bus) port # for InterWave driver.");
+#endif
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for InterWave driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for InterWave driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for InterWave driver.");
+module_param_array(joystick_dac, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_dac, "Joystick DAC level 0.59V-4.52V or 0.389V-2.98V for InterWave driver.");
+module_param_array(midi, int, NULL, 0444);
+MODULE_PARM_DESC(midi, "MIDI UART enable for InterWave driver.");
+module_param_array(pcm_channels, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_channels, "Reserved PCM channels for InterWave driver.");
+module_param_array(effect, int, NULL, 0444);
+MODULE_PARM_DESC(effect, "Effects enable for InterWave driver.");
+
+struct snd_interwave {
+ int irq;
+ struct snd_card *card;
+ struct snd_gus_card *gus;
+ struct snd_wss *wss;
+#ifdef SNDRV_STB
+ struct resource *i2c_res;
+#endif
+ unsigned short gus_status_reg;
+ unsigned short pcm_status_reg;
+#ifdef CONFIG_PNP
+ struct pnp_dev *dev;
+#ifdef SNDRV_STB
+ struct pnp_dev *devtc;
+#endif
+#endif
+};
+
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnp_registered;
+
+static struct pnp_card_device_id snd_interwave_pnpids[] = {
+#ifndef SNDRV_STB
+ /* Gravis UltraSound Plug & Play */
+ { .id = "GRV0001", .devs = { { .id = "GRV0000" } } },
+ /* STB SoundRage32 */
+ { .id = "STB011a", .devs = { { .id = "STB0010" } } },
+ /* MED3210 */
+ { .id = "DXP3201", .devs = { { .id = "DXP0010" } } },
+ /* Dynasonic Pro */
+ /* This device also have CDC1117:DynaSonix Pro Audio Effects Processor */
+ { .id = "CDC1111", .devs = { { .id = "CDC1112" } } },
+ /* Panasonic PCA761AW Audio Card */
+ { .id = "ADV55ff", .devs = { { .id = "ADV0010" } } },
+ /* InterWave STB without TEA6330T */
+ { .id = "ADV550a", .devs = { { .id = "ADV0010" } } },
+#else
+ /* InterWave STB with TEA6330T */
+ { .id = "ADV550a", .devs = { { .id = "ADV0010" }, { .id = "ADV0015" } } },
+#endif
+ { .id = "" }
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_interwave_pnpids);
+
+#endif /* CONFIG_PNP */
+
+
+#ifdef SNDRV_STB
+static void snd_interwave_i2c_setlines(struct snd_i2c_bus *bus, int ctrl, int data)
+{
+ unsigned long port = bus->private_value;
+
+#if 0
+ printk("i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data);
+#endif
+ outb((data << 1) | ctrl, port);
+ udelay(10);
+}
+
+static int snd_interwave_i2c_getclockline(struct snd_i2c_bus *bus)
+{
+ unsigned long port = bus->private_value;
+ unsigned char res;
+
+ res = inb(port) & 1;
+#if 0
+ printk("i2c_getclockline - 0x%lx -> %i\n", port, res);
+#endif
+ return res;
+}
+
+static int snd_interwave_i2c_getdataline(struct snd_i2c_bus *bus, int ack)
+{
+ unsigned long port = bus->private_value;
+ unsigned char res;
+
+ if (ack)
+ udelay(10);
+ res = (inb(port) & 2) >> 1;
+#if 0
+ printk("i2c_getdataline - 0x%lx -> %i\n", port, res);
+#endif
+ return res;
+}
+
+static struct snd_i2c_bit_ops snd_interwave_i2c_bit_ops = {
+ .setlines = snd_interwave_i2c_setlines,
+ .getclock = snd_interwave_i2c_getclockline,
+ .getdata = snd_interwave_i2c_getdataline,
+};
+
+static int __devinit snd_interwave_detect_stb(struct snd_interwave *iwcard,
+ struct snd_gus_card * gus, int dev,
+ struct snd_i2c_bus **rbus)
+{
+ unsigned long port;
+ struct snd_i2c_bus *bus;
+ struct snd_card *card = iwcard->card;
+ char name[32];
+ int err;
+
+ *rbus = NULL;
+ port = port_tc[dev];
+ if (port == SNDRV_AUTO_PORT) {
+ port = 0x350;
+ if (gus->gf1.port == 0x250) {
+ port = 0x360;
+ }
+ while (port <= 0x380) {
+ if ((iwcard->i2c_res = request_region(port, 1, "InterWave (I2C bus)")) != NULL)
+ break;
+ port += 0x10;
+ }
+ } else {
+ iwcard->i2c_res = request_region(port, 1, "InterWave (I2C bus)");
+ }
+ if (iwcard->i2c_res == NULL) {
+ snd_printk(KERN_ERR "interwave: can't grab i2c bus port\n");
+ return -ENODEV;
+ }
+
+ sprintf(name, "InterWave-%i", card->number);
+ if ((err = snd_i2c_bus_create(card, name, NULL, &bus)) < 0)
+ return err;
+ bus->private_value = port;
+ bus->hw_ops.bit = &snd_interwave_i2c_bit_ops;
+ if ((err = snd_tea6330t_detect(bus, 0)) < 0)
+ return err;
+ *rbus = bus;
+ return 0;
+}
+#endif
+
+static int __devinit snd_interwave_detect(struct snd_interwave *iwcard,
+ struct snd_gus_card * gus,
+ int dev
+#ifdef SNDRV_STB
+ , struct snd_i2c_bus **rbus
+#endif
+ )
+{
+ unsigned long flags;
+ unsigned char rev1, rev2;
+ int d;
+
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 0) {
+ snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d);
+ return -ENODEV;
+ }
+ udelay(160);
+ snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 1); /* release reset */
+ udelay(160);
+ if (((d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)) & 0x07) != 1) {
+ snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d);
+ return -ENODEV;
+ }
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ rev1 = snd_gf1_look8(gus, SNDRV_GF1_GB_VERSION_NUMBER);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_VERSION_NUMBER, ~rev1);
+ rev2 = snd_gf1_look8(gus, SNDRV_GF1_GB_VERSION_NUMBER);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_VERSION_NUMBER, rev1);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ snd_printdd("[0x%lx] InterWave check - rev1=0x%x, rev2=0x%x\n", gus->gf1.port, rev1, rev2);
+ if ((rev1 & 0xf0) == (rev2 & 0xf0) &&
+ (rev1 & 0x0f) != (rev2 & 0x0f)) {
+ snd_printdd("[0x%lx] InterWave check - passed\n", gus->gf1.port);
+ gus->interwave = 1;
+ strcpy(gus->card->shortname, "AMD InterWave");
+ gus->revision = rev1 >> 4;
+#ifndef SNDRV_STB
+ return 0; /* ok.. We have an InterWave board */
+#else
+ return snd_interwave_detect_stb(iwcard, gus, dev, rbus);
+#endif
+ }
+ snd_printdd("[0x%lx] InterWave check - failed\n", gus->gf1.port);
+ return -ENODEV;
+}
+
+static irqreturn_t snd_interwave_interrupt(int irq, void *dev_id)
+{
+ struct snd_interwave *iwcard = dev_id;
+ int loop, max = 5;
+ int handled = 0;
+
+ do {
+ loop = 0;
+ if (inb(iwcard->gus_status_reg)) {
+ handled = 1;
+ snd_gus_interrupt(irq, iwcard->gus);
+ loop++;
+ }
+ if (inb(iwcard->pcm_status_reg) & 0x01) { /* IRQ bit is set? */
+ handled = 1;
+ snd_wss_interrupt(irq, iwcard->wss);
+ loop++;
+ }
+ } while (loop && --max > 0);
+ return IRQ_RETVAL(handled);
+}
+
+static void __devinit snd_interwave_reset(struct snd_gus_card * gus)
+{
+ snd_gf1_write8(gus, SNDRV_GF1_GB_RESET, 0x00);
+ udelay(160);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_RESET, 0x01);
+ udelay(160);
+}
+
+static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *sizes)
+{
+ unsigned int idx;
+ unsigned int local;
+ unsigned char d;
+
+ for (idx = 0; idx < 4; idx++) {
+ sizes[idx] = 0;
+ d = 0x55;
+ for (local = idx << 22;
+ local < (idx << 22) + 0x400000;
+ local += 0x40000, d++) {
+ snd_gf1_poke(gus, local, d);
+ snd_gf1_poke(gus, local + 1, d + 1);
+#if 0
+ printk("d = 0x%x, local = 0x%x, local + 1 = 0x%x, idx << 22 = 0x%x\n",
+ d,
+ snd_gf1_peek(gus, local),
+ snd_gf1_peek(gus, local + 1),
+ snd_gf1_peek(gus, idx << 22));
+#endif
+ if (snd_gf1_peek(gus, local) != d ||
+ snd_gf1_peek(gus, local + 1) != d + 1 ||
+ snd_gf1_peek(gus, idx << 22) != 0x55)
+ break;
+ sizes[idx]++;
+ }
+ }
+#if 0
+ printk("sizes: %i %i %i %i\n", sizes[0], sizes[1], sizes[2], sizes[3]);
+#endif
+}
+
+struct rom_hdr {
+ /* 000 */ unsigned char iwave[8];
+ /* 008 */ unsigned char rom_hdr_revision;
+ /* 009 */ unsigned char series_number;
+ /* 010 */ unsigned char series_name[16];
+ /* 026 */ unsigned char date[10];
+ /* 036 */ unsigned short vendor_revision_major;
+ /* 038 */ unsigned short vendor_revision_minor;
+ /* 040 */ unsigned int rom_size;
+ /* 044 */ unsigned char copyright[128];
+ /* 172 */ unsigned char vendor_name[64];
+ /* 236 */ unsigned char rom_description[128];
+ /* 364 */ unsigned char pad[147];
+ /* 511 */ unsigned char csum;
+};
+
+static void __devinit snd_interwave_detect_memory(struct snd_gus_card * gus)
+{
+ static unsigned int lmc[13] =
+ {
+ 0x00000001, 0x00000101, 0x01010101, 0x00000401,
+ 0x04040401, 0x00040101, 0x04040101, 0x00000004,
+ 0x00000404, 0x04040404, 0x00000010, 0x00001010,
+ 0x10101010
+ };
+
+ int bank_pos, pages;
+ unsigned int i, lmct;
+ int psizes[4];
+ unsigned char iwave[8];
+ unsigned char csum;
+
+ snd_interwave_reset(gus);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_GLOBAL_MODE, snd_gf1_read8(gus, SNDRV_GF1_GB_GLOBAL_MODE) | 0x01); /* enhanced mode */
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x01); /* DRAM I/O cycles selected */
+ snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xff10) | 0x004c);
+ /* ok.. simple test of memory size */
+ pages = 0;
+ snd_gf1_poke(gus, 0, 0x55);
+ snd_gf1_poke(gus, 1, 0xaa);
+#if 1
+ if (snd_gf1_peek(gus, 0) == 0x55 && snd_gf1_peek(gus, 1) == 0xaa)
+#else
+ if (0) /* ok.. for testing of 0k RAM */
+#endif
+ {
+ snd_interwave_bank_sizes(gus, psizes);
+ lmct = (psizes[3] << 24) | (psizes[2] << 16) |
+ (psizes[1] << 8) | psizes[0];
+#if 0
+ printk("lmct = 0x%08x\n", lmct);
+#endif
+ for (i = 0; i < ARRAY_SIZE(lmc); i++)
+ if (lmct == lmc[i]) {
+#if 0
+ printk("found !!! %i\n", i);
+#endif
+ snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | i);
+ snd_interwave_bank_sizes(gus, psizes);
+ break;
+ }
+ if (i >= ARRAY_SIZE(lmc) && !gus->gf1.enh_mode)
+ snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | 2);
+ for (i = 0; i < 4; i++) {
+ gus->gf1.mem_alloc.banks_8[i].address =
+ gus->gf1.mem_alloc.banks_16[i].address = i << 22;
+ gus->gf1.mem_alloc.banks_8[i].size =
+ gus->gf1.mem_alloc.banks_16[i].size = psizes[i] << 18;
+ pages += psizes[i];
+ }
+ }
+ pages <<= 18;
+ gus->gf1.memory = pages;
+
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x03); /* select ROM */
+ snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xff1f) | (4 << 5));
+ gus->gf1.rom_banks = 0;
+ gus->gf1.rom_memory = 0;
+ for (bank_pos = 0; bank_pos < 16L * 1024L * 1024L; bank_pos += 4L * 1024L * 1024L) {
+ for (i = 0; i < 8; ++i)
+ iwave[i] = snd_gf1_peek(gus, bank_pos + i);
+#ifdef CONFIG_SND_DEBUG_ROM
+ printk(KERN_DEBUG "ROM at 0x%06x = %02x:%02x:%02x:%02x:%02x:%02x:%02x:%02x\n", bank_pos,
+ iwave[0], iwave[1], iwave[2], iwave[3],
+ iwave[4], iwave[5], iwave[6], iwave[7]);
+#endif
+ if (strncmp(iwave, "INTRWAVE", 8))
+ continue; /* first check */
+ csum = 0;
+ for (i = 0; i < sizeof(struct rom_hdr); i++)
+ csum += snd_gf1_peek(gus, bank_pos + i);
+#ifdef CONFIG_SND_DEBUG_ROM
+ printk(KERN_DEBUG "ROM checksum = 0x%x (computed)\n", csum);
+#endif
+ if (csum != 0)
+ continue; /* not valid rom */
+ gus->gf1.rom_banks++;
+ gus->gf1.rom_present |= 1 << (bank_pos >> 22);
+ gus->gf1.rom_memory = snd_gf1_peek(gus, bank_pos + 40) |
+ (snd_gf1_peek(gus, bank_pos + 41) << 8) |
+ (snd_gf1_peek(gus, bank_pos + 42) << 16) |
+ (snd_gf1_peek(gus, bank_pos + 43) << 24);
+ }
+#if 0
+ if (gus->gf1.rom_memory > 0) {
+ if (gus->gf1.rom_banks == 1 && gus->gf1.rom_present == 8)
+ gus->card->type = SNDRV_CARD_TYPE_IW_DYNASONIC;
+ }
+#endif
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x00); /* select RAM */
+
+ if (!gus->gf1.enh_mode)
+ snd_interwave_reset(gus);
+}
+
+static void __devinit snd_interwave_init(int dev, struct snd_gus_card * gus)
+{
+ unsigned long flags;
+
+ /* ok.. some InterWave specific initialization */
+ spin_lock_irqsave(&gus->reg_lock, flags);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, 0x00);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_COMPATIBILITY, 0x1f);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_DECODE_CONTROL, 0x49);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_VERSION_NUMBER, 0x11);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MPU401_CONTROL_A, 0x00);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_MPU401_CONTROL_B, 0x30);
+ snd_gf1_write8(gus, SNDRV_GF1_GB_EMULATION_IRQ, 0x00);
+ spin_unlock_irqrestore(&gus->reg_lock, flags);
+ gus->equal_irq = 1;
+ gus->codec_flag = 1;
+ gus->interwave = 1;
+ gus->max_flag = 1;
+ gus->joystick_dac = joystick_dac[dev];
+
+}
+
+static struct snd_kcontrol_new snd_interwave_controls[] = {
+WSS_DOUBLE("Master Playback Switch", 0,
+ CS4231_LINE_LEFT_OUTPUT, CS4231_LINE_RIGHT_OUTPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Master Playback Volume", 0,
+ CS4231_LINE_LEFT_OUTPUT, CS4231_LINE_RIGHT_OUTPUT, 0, 0, 31, 1),
+WSS_DOUBLE("Mic Playback Switch", 0,
+ CS4231_LEFT_MIC_INPUT, CS4231_RIGHT_MIC_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Mic Playback Volume", 0,
+ CS4231_LEFT_MIC_INPUT, CS4231_RIGHT_MIC_INPUT, 0, 0, 31, 1)
+};
+
+static int __devinit snd_interwave_mixer(struct snd_wss *chip)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ctl_elem_id id1, id2;
+ unsigned int idx;
+ int err;
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+#if 0
+ /* remove mono microphone controls */
+ strcpy(id1.name, "Mic Playback Switch");
+ if ((err = snd_ctl_remove_id(card, &id1)) < 0)
+ return err;
+ strcpy(id1.name, "Mic Playback Volume");
+ if ((err = snd_ctl_remove_id(card, &id1)) < 0)
+ return err;
+#endif
+ /* add new master and mic controls */
+ for (idx = 0; idx < ARRAY_SIZE(snd_interwave_controls); idx++)
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_interwave_controls[idx], chip))) < 0)
+ return err;
+ snd_wss_out(chip, CS4231_LINE_LEFT_OUTPUT, 0x9f);
+ snd_wss_out(chip, CS4231_LINE_RIGHT_OUTPUT, 0x9f);
+ snd_wss_out(chip, CS4231_LEFT_MIC_INPUT, 0x9f);
+ snd_wss_out(chip, CS4231_RIGHT_MIC_INPUT, 0x9f);
+ /* reassign AUXA to SYNTHESIZER */
+ strcpy(id1.name, "Aux Playback Switch");
+ strcpy(id2.name, "Synth Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "Synth Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ /* reassign AUXB to CD */
+ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1;
+ strcpy(id2.name, "CD Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "CD Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ return 0;
+}
+
+#ifdef CONFIG_PNP
+
+static int __devinit snd_interwave_pnp(int dev, struct snd_interwave *iwcard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ iwcard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (iwcard->dev == NULL)
+ return -EBUSY;
+
+#ifdef SNDRV_STB
+ iwcard->devtc = pnp_request_card_device(card, id->devs[1].id, NULL);
+ if (iwcard->devtc == NULL)
+ return -EBUSY;
+#endif
+ /* Synth & Codec initialization */
+ pdev = iwcard->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "InterWave PnP configure failure (out of resources?)\n");
+ return err;
+ }
+ if (pnp_port_start(pdev, 0) + 0x100 != pnp_port_start(pdev, 1) ||
+ pnp_port_start(pdev, 0) + 0x10c != pnp_port_start(pdev, 2)) {
+ snd_printk(KERN_ERR "PnP configure failure (wrong ports)\n");
+ return -ENOENT;
+ }
+ port[dev] = pnp_port_start(pdev, 0);
+ dma1[dev] = pnp_dma(pdev, 0);
+ if (dma2[dev] >= 0)
+ dma2[dev] = pnp_dma(pdev, 1);
+ irq[dev] = pnp_irq(pdev, 0);
+ snd_printdd("isapnp IW: sb port=0x%llx, gf1 port=0x%llx, codec port=0x%llx\n",
+ (unsigned long long)pnp_port_start(pdev, 0),
+ (unsigned long long)pnp_port_start(pdev, 1),
+ (unsigned long long)pnp_port_start(pdev, 2));
+ snd_printdd("isapnp IW: dma1=%i, dma2=%i, irq=%i\n", dma1[dev], dma2[dev], irq[dev]);
+#ifdef SNDRV_STB
+ /* Tone Control initialization */
+ pdev = iwcard->devtc;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "InterWave ToneControl PnP configure failure (out of resources?)\n");
+ return err;
+ }
+ port_tc[dev] = pnp_port_start(pdev, 0);
+ snd_printdd("isapnp IW: tone control port=0x%lx\n", port_tc[dev]);
+#endif
+ return 0;
+}
+#endif /* CONFIG_PNP */
+
+static void snd_interwave_free(struct snd_card *card)
+{
+ struct snd_interwave *iwcard = card->private_data;
+
+ if (iwcard == NULL)
+ return;
+#ifdef SNDRV_STB
+ release_and_free_resource(iwcard->i2c_res);
+#endif
+ if (iwcard->irq >= 0)
+ free_irq(iwcard->irq, (void *)iwcard);
+}
+
+static struct snd_card *snd_interwave_card_new(int dev)
+{
+ struct snd_card *card;
+ struct snd_interwave *iwcard;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_interwave));
+ if (card == NULL)
+ return NULL;
+ iwcard = card->private_data;
+ iwcard->card = card;
+ iwcard->irq = -1;
+ card->private_free = snd_interwave_free;
+ return card;
+}
+
+static int __devinit snd_interwave_probe(struct snd_card *card, int dev)
+{
+ int xirq, xdma1, xdma2;
+ struct snd_interwave *iwcard = card->private_data;
+ struct snd_wss *wss;
+ struct snd_gus_card *gus;
+#ifdef SNDRV_STB
+ struct snd_i2c_bus *i2c_bus;
+#endif
+ struct snd_pcm *pcm;
+ char *str;
+ int err;
+
+ xirq = irq[dev];
+ xdma1 = dma1[dev];
+ xdma2 = dma2[dev];
+
+ if ((err = snd_gus_create(card,
+ port[dev],
+ -xirq, xdma1, xdma2,
+ 0, 32,
+ pcm_channels[dev], effect[dev], &gus)) < 0)
+ return err;
+
+ if ((err = snd_interwave_detect(iwcard, gus, dev
+#ifdef SNDRV_STB
+ , &i2c_bus
+#endif
+ )) < 0)
+ return err;
+
+ iwcard->gus_status_reg = gus->gf1.reg_irqstat;
+ iwcard->pcm_status_reg = gus->gf1.port + 0x10c + 2;
+
+ snd_interwave_init(dev, gus);
+ snd_interwave_detect_memory(gus);
+ if ((err = snd_gus_initialize(gus)) < 0)
+ return err;
+
+ if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED,
+ "InterWave", iwcard)) {
+ snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
+ return -EBUSY;
+ }
+ iwcard->irq = xirq;
+
+ err = snd_wss_create(card,
+ gus->gf1.port + 0x10c, -1, xirq,
+ xdma2 < 0 ? xdma1 : xdma2, xdma1,
+ WSS_HW_INTERWAVE,
+ WSS_HWSHARE_IRQ |
+ WSS_HWSHARE_DMA1 |
+ WSS_HWSHARE_DMA2,
+ &wss);
+ if (err < 0)
+ return err;
+
+ err = snd_wss_pcm(wss, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ sprintf(pcm->name + strlen(pcm->name), " rev %c", gus->revision + 'A');
+ strcat(pcm->name, " (codec)");
+
+ err = snd_wss_timer(wss, 2, NULL);
+ if (err < 0)
+ return err;
+
+ err = snd_wss_mixer(wss);
+ if (err < 0)
+ return err;
+
+ if (pcm_channels[dev] > 0) {
+ err = snd_gf1_pcm_new(gus, 1, 1, NULL);
+ if (err < 0)
+ return err;
+ }
+ err = snd_interwave_mixer(wss);
+ if (err < 0)
+ return err;
+
+#ifdef SNDRV_STB
+ {
+ struct snd_ctl_elem_id id1, id2;
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id1.name, "Master Playback Switch");
+ strcpy(id2.name, id1.name);
+ id2.index = 1;
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ strcpy(id1.name, "Master Playback Volume");
+ strcpy(id2.name, id1.name);
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ if ((err = snd_tea6330t_update_mixer(card, i2c_bus, 0, 1)) < 0)
+ return err;
+ }
+#endif
+
+ gus->uart_enable = midi[dev];
+ if ((err = snd_gf1_rawmidi_new(gus, 0, NULL)) < 0)
+ return err;
+
+#ifndef SNDRV_STB
+ str = "AMD InterWave";
+ if (gus->gf1.rom_banks == 1 && gus->gf1.rom_present == 8)
+ str = "Dynasonic 3-D";
+#else
+ str = "InterWave STB";
+#endif
+ strcpy(card->driver, str);
+ strcpy(card->shortname, str);
+ sprintf(card->longname, "%s at 0x%lx, irq %i, dma %d",
+ str,
+ gus->gf1.port,
+ xirq,
+ xdma1);
+ if (xdma2 >= 0)
+ sprintf(card->longname + strlen(card->longname), "&%d", xdma2);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ return err;
+
+ iwcard->wss = wss;
+ iwcard->gus = gus;
+ return 0;
+}
+
+static int __devinit snd_interwave_isa_probe1(int dev, struct device *devptr)
+{
+ struct snd_card *card;
+ int err;
+
+ card = snd_interwave_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+
+ snd_card_set_dev(card, devptr);
+ if ((err = snd_interwave_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ dev_set_drvdata(devptr, card);
+ return 0;
+}
+
+static int __devinit snd_interwave_isa_match(struct device *pdev,
+ unsigned int dev)
+{
+ if (!enable[dev])
+ return 0;
+#ifdef CONFIG_PNP
+ if (isapnp[dev])
+ return 0;
+#endif
+ return 1;
+}
+
+static int __devinit snd_interwave_isa_probe(struct device *pdev,
+ unsigned int dev)
+{
+ int err;
+ static int possible_irqs[] = {5, 11, 12, 9, 7, 15, 3, -1};
+ static int possible_dmas[] = {0, 1, 3, 5, 6, 7, -1};
+
+ if (irq[dev] == SNDRV_AUTO_IRQ) {
+ if ((irq[dev] = snd_legacy_find_free_irq(possible_irqs)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (dma1[dev] == SNDRV_AUTO_DMA) {
+ if ((dma1[dev] = snd_legacy_find_free_dma(possible_dmas)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA1\n");
+ return -EBUSY;
+ }
+ }
+ if (dma2[dev] == SNDRV_AUTO_DMA) {
+ if ((dma2[dev] = snd_legacy_find_free_dma(possible_dmas)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA2\n");
+ return -EBUSY;
+ }
+ }
+
+ if (port[dev] != SNDRV_AUTO_PORT)
+ return snd_interwave_isa_probe1(dev, pdev);
+ else {
+ static long possible_ports[] = {0x210, 0x220, 0x230, 0x240, 0x250, 0x260};
+ int i;
+ for (i = 0; i < ARRAY_SIZE(possible_ports); i++) {
+ port[dev] = possible_ports[i];
+ err = snd_interwave_isa_probe1(dev, pdev);
+ if (! err)
+ return 0;
+ }
+ return err;
+ }
+}
+
+static int __devexit snd_interwave_isa_remove(struct device *devptr, unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_interwave_driver = {
+ .match = snd_interwave_isa_match,
+ .probe = snd_interwave_isa_probe,
+ .remove = __devexit_p(snd_interwave_isa_remove),
+ /* FIXME: suspend,resume */
+ .driver = {
+ .name = INTERWAVE_DRIVER
+ },
+};
+
+#ifdef CONFIG_PNP
+static int __devinit snd_interwave_pnp_detect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ static int dev;
+ struct snd_card *card;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_interwave_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+
+ if ((res = snd_interwave_pnp(dev, card->private_data, pcard, pid)) < 0) {
+ snd_card_free(card);
+ return res;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((res = snd_interwave_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return res;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_interwave_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+static struct pnp_card_driver interwave_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = INTERWAVE_PNP_DRIVER,
+ .id_table = snd_interwave_pnpids,
+ .probe = snd_interwave_pnp_detect,
+ .remove = __devexit_p(snd_interwave_pnp_remove),
+ /* FIXME: suspend,resume */
+};
+
+#endif /* CONFIG_PNP */
+
+static int __init alsa_card_interwave_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&snd_interwave_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+
+ err = pnp_register_card_driver(&interwave_pnpc_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ if (isa_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit alsa_card_interwave_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnp_registered)
+ pnp_unregister_card_driver(&interwave_pnpc_driver);
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&snd_interwave_driver);
+}
+
+module_init(alsa_card_interwave_init)
+module_exit(alsa_card_interwave_exit)
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
new file mode 100644
index 0000000..b848d10
--- /dev/null
+++ b/sound/isa/opl3sa2.c
@@ -0,0 +1,963 @@
+/*
+ * Driver for Yamaha OPL3-SA[2,3] soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/interrupt.h>
+#include <linux/pm.h>
+#include <linux/slab.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <asm/io.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Yamaha OPL3SA2+");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF719E-S},"
+ "{Genius,Sound Maker 3DX},"
+ "{Yamaha,OPL3SA3},"
+ "{Intel,AL440LX sound},"
+ "{NeoMagic,MagicWave 3DX}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0xf86,0x370,0x100 */
+static long sb_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */
+static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* 0x530,0xe80,0xf40,0x604 */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x388 */
+static long midi_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;/* 0x330,0x300 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 0,1,3,5,9,11,12,15 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3,5,6,7 */
+static int opl3sa3_ymode[SNDRV_CARDS]; /* 0,1,2,3 */ /*SL Added*/
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for OPL3-SA soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for OPL3-SA soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable OPL3-SA soundcard.");
+#ifdef CONFIG_PNP
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard.");
+#endif
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for OPL3-SA driver.");
+module_param_array(sb_port, long, NULL, 0444);
+MODULE_PARM_DESC(sb_port, "SB port # for OPL3-SA driver.");
+module_param_array(wss_port, long, NULL, 0444);
+MODULE_PARM_DESC(wss_port, "WSS port # for OPL3-SA driver.");
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM port # for OPL3-SA driver.");
+module_param_array(midi_port, long, NULL, 0444);
+MODULE_PARM_DESC(midi_port, "MIDI port # for OPL3-SA driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for OPL3-SA driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for OPL3-SA driver.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for OPL3-SA driver.");
+module_param_array(opl3sa3_ymode, int, NULL, 0444);
+MODULE_PARM_DESC(opl3sa3_ymode, "Speaker size selection for 3D Enhancement mode: Desktop/Large Notebook/Small Notebook/HiFi.");
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnp_registered;
+static int pnpc_registered;
+#endif
+
+/* control ports */
+#define OPL3SA2_PM_CTRL 0x01
+#define OPL3SA2_SYS_CTRL 0x02
+#define OPL3SA2_IRQ_CONFIG 0x03
+#define OPL3SA2_IRQ_STATUS 0x04
+#define OPL3SA2_DMA_CONFIG 0x06
+#define OPL3SA2_MASTER_LEFT 0x07
+#define OPL3SA2_MASTER_RIGHT 0x08
+#define OPL3SA2_MIC 0x09
+#define OPL3SA2_MISC 0x0A
+
+/* opl3sa3 only */
+#define OPL3SA3_DGTL_DOWN 0x12
+#define OPL3SA3_ANLG_DOWN 0x13
+#define OPL3SA3_WIDE 0x14
+#define OPL3SA3_BASS 0x15
+#define OPL3SA3_TREBLE 0x16
+
+/* power management bits */
+#define OPL3SA2_PM_ADOWN 0x20
+#define OPL3SA2_PM_PSV 0x04
+#define OPL3SA2_PM_PDN 0x02
+#define OPL3SA2_PM_PDX 0x01
+
+#define OPL3SA2_PM_D0 0x00
+#define OPL3SA2_PM_D3 (OPL3SA2_PM_ADOWN|OPL3SA2_PM_PSV|OPL3SA2_PM_PDN|OPL3SA2_PM_PDX)
+
+struct snd_opl3sa2 {
+ int version; /* 2 or 3 */
+ unsigned long port; /* control port */
+ struct resource *res_port; /* control port resource */
+ int irq;
+ int single_dma;
+ spinlock_t reg_lock;
+ struct snd_hwdep *synth;
+ struct snd_rawmidi *rmidi;
+ struct snd_wss *wss;
+ unsigned char ctlregs[0x20];
+ int ymode; /* SL added */
+ struct snd_kcontrol *master_switch;
+ struct snd_kcontrol *master_volume;
+};
+
+#define PFX "opl3sa2: "
+
+#ifdef CONFIG_PNP
+
+static struct pnp_device_id snd_opl3sa2_pnpbiosids[] = {
+ { .id = "YMH0021" },
+ { .id = "NMX2210" }, /* Gateway Solo 2500 */
+ { .id = "" } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp, snd_opl3sa2_pnpbiosids);
+
+static struct pnp_card_device_id snd_opl3sa2_pnpids[] = {
+ /* Yamaha YMF719E-S (Genius Sound Maker 3DX) */
+ { .id = "YMH0020", .devs = { { "YMH0021" } } },
+ /* Yamaha OPL3-SA3 (integrated on Intel's Pentium II AL440LX motherboard) */
+ { .id = "YMH0030", .devs = { { "YMH0021" } } },
+ /* Yamaha OPL3-SA2 */
+ { .id = "YMH0800", .devs = { { "YMH0021" } } },
+ /* Yamaha OPL3-SA2 */
+ { .id = "YMH0801", .devs = { { "YMH0021" } } },
+ /* NeoMagic MagicWave 3DX */
+ { .id = "NMX2200", .devs = { { "YMH2210" } } },
+ /* NeoMagic MagicWave 3D */
+ { .id = "NMX2200", .devs = { { "NMX2210" } } },
+ /* --- */
+ { .id = "" } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_opl3sa2_pnpids);
+
+#endif /* CONFIG_PNP */
+
+
+/* read control port (w/o spinlock) */
+static unsigned char __snd_opl3sa2_read(struct snd_opl3sa2 *chip, unsigned char reg)
+{
+ unsigned char result;
+#if 0
+ outb(0x1d, port); /* password */
+ printk("read [0x%lx] = 0x%x\n", port, inb(port));
+#endif
+ outb(reg, chip->port); /* register */
+ result = inb(chip->port + 1);
+#if 0
+ printk("read [0x%lx] = 0x%x [0x%x]\n", port, result, inb(port));
+#endif
+ return result;
+}
+
+/* read control port (with spinlock) */
+static unsigned char snd_opl3sa2_read(struct snd_opl3sa2 *chip, unsigned char reg)
+{
+ unsigned long flags;
+ unsigned char result;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ result = __snd_opl3sa2_read(chip, reg);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return result;
+}
+
+/* write control port (w/o spinlock) */
+static void __snd_opl3sa2_write(struct snd_opl3sa2 *chip, unsigned char reg, unsigned char value)
+{
+#if 0
+ outb(0x1d, port); /* password */
+#endif
+ outb(reg, chip->port); /* register */
+ outb(value, chip->port + 1);
+ chip->ctlregs[reg] = value;
+}
+
+/* write control port (with spinlock) */
+static void snd_opl3sa2_write(struct snd_opl3sa2 *chip, unsigned char reg, unsigned char value)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ __snd_opl3sa2_write(chip, reg, value);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static int __devinit snd_opl3sa2_detect(struct snd_card *card)
+{
+ struct snd_opl3sa2 *chip = card->private_data;
+ unsigned long port;
+ unsigned char tmp, tmp1;
+ char str[2];
+
+ port = chip->port;
+ if ((chip->res_port = request_region(port, 2, "OPL3-SA control")) == NULL) {
+ snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port);
+ return -EBUSY;
+ }
+ // snd_printk("REG 0A = 0x%x\n", snd_opl3sa2_read(chip, 0x0a));
+ chip->version = 0;
+ tmp = snd_opl3sa2_read(chip, OPL3SA2_MISC);
+ if (tmp == 0xff) {
+ snd_printd("OPL3-SA [0x%lx] detect = 0x%x\n", port, tmp);
+ return -ENODEV;
+ }
+ switch (tmp & 0x07) {
+ case 0x01:
+ chip->version = 2; /* YMF711 */
+ break;
+ default:
+ chip->version = 3;
+ /* 0x02 - standard */
+ /* 0x03 - YM715B */
+ /* 0x04 - YM719 - OPL-SA4? */
+ /* 0x05 - OPL3-SA3 - Libretto 100 */
+ /* 0x07 - unknown - Neomagic MagicWave 3D */
+ break;
+ }
+ str[0] = chip->version + '0';
+ str[1] = 0;
+ strcat(card->shortname, str);
+ snd_opl3sa2_write(chip, OPL3SA2_MISC, tmp ^ 7);
+ if ((tmp1 = snd_opl3sa2_read(chip, OPL3SA2_MISC)) != tmp) {
+ snd_printd("OPL3-SA [0x%lx] detect (1) = 0x%x (0x%x)\n", port, tmp, tmp1);
+ return -ENODEV;
+ }
+ /* try if the MIC register is accesible */
+ tmp = snd_opl3sa2_read(chip, OPL3SA2_MIC);
+ snd_opl3sa2_write(chip, OPL3SA2_MIC, 0x8a);
+ if (((tmp1 = snd_opl3sa2_read(chip, OPL3SA2_MIC)) & 0x9f) != 0x8a) {
+ snd_printd("OPL3-SA [0x%lx] detect (2) = 0x%x (0x%x)\n", port, tmp, tmp1);
+ return -ENODEV;
+ }
+ snd_opl3sa2_write(chip, OPL3SA2_MIC, 0x9f);
+ /* initialization */
+ /* Power Management - full on */
+ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0);
+ if (chip->version > 2) {
+ /* ymode is bits 4&5 (of 0 to 7) on all but opl3sa2 versions */
+ snd_opl3sa2_write(chip, OPL3SA2_SYS_CTRL, (chip->ymode << 4));
+ } else {
+ /* default for opl3sa2 versions */
+ snd_opl3sa2_write(chip, OPL3SA2_SYS_CTRL, 0x00);
+ }
+ snd_opl3sa2_write(chip, OPL3SA2_IRQ_CONFIG, 0x0d); /* Interrupt Channel Configuration - IRQ A = OPL3 + MPU + WSS */
+ if (chip->single_dma) {
+ snd_opl3sa2_write(chip, OPL3SA2_DMA_CONFIG, 0x03); /* DMA Configuration - DMA A = WSS-R + WSS-P */
+ } else {
+ snd_opl3sa2_write(chip, OPL3SA2_DMA_CONFIG, 0x21); /* DMA Configuration - DMA B = WSS-R, DMA A = WSS-P */
+ }
+ snd_opl3sa2_write(chip, OPL3SA2_MISC, 0x80 | (tmp & 7)); /* Miscellaneous - default */
+ if (chip->version > 2) {
+ snd_opl3sa2_write(chip, OPL3SA3_DGTL_DOWN, 0x00); /* Digital Block Partial Power Down - default */
+ snd_opl3sa2_write(chip, OPL3SA3_ANLG_DOWN, 0x00); /* Analog Block Partial Power Down - default */
+ }
+ return 0;
+}
+
+static irqreturn_t snd_opl3sa2_interrupt(int irq, void *dev_id)
+{
+ unsigned short status;
+ struct snd_card *card = dev_id;
+ struct snd_opl3sa2 *chip;
+ int handled = 0;
+
+ if (card == NULL)
+ return IRQ_NONE;
+
+ chip = card->private_data;
+ status = snd_opl3sa2_read(chip, OPL3SA2_IRQ_STATUS);
+
+ if (status & 0x20) {
+ handled = 1;
+ snd_opl3_interrupt(chip->synth);
+ }
+
+ if ((status & 0x10) && chip->rmidi != NULL) {
+ handled = 1;
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+ }
+
+ if (status & 0x07) { /* TI,CI,PI */
+ handled = 1;
+ snd_wss_interrupt(irq, chip->wss);
+ }
+
+ if (status & 0x40) { /* hardware volume change */
+ handled = 1;
+ /* reading from Master Lch register at 0x07 clears this bit */
+ snd_opl3sa2_read(chip, OPL3SA2_MASTER_RIGHT);
+ snd_opl3sa2_read(chip, OPL3SA2_MASTER_LEFT);
+ if (chip->master_switch && chip->master_volume) {
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_switch->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ }
+ }
+ return IRQ_RETVAL(handled);
+}
+
+#define OPL3SA2_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_wss_info_single, \
+ .get = snd_opl3sa2_get_single, .put = snd_opl3sa2_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+#define OPL3SA2_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .index = xindex, \
+ .info = snd_wss_info_single, \
+ .get = snd_opl3sa2_get_single, .put = snd_opl3sa2_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = (xtlv) } }
+
+static int snd_opl3sa2_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_opl3sa2 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->ctlregs[reg] >> shift) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_opl3sa2_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_opl3sa2 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short val, oval;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ oval = chip->ctlregs[reg];
+ val = (oval & ~(mask << shift)) | val;
+ change = val != oval;
+ __snd_opl3sa2_write(chip, reg, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+#define OPL3SA2_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_wss_info_double, \
+ .get = snd_opl3sa2_get_double, .put = snd_opl3sa2_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) }
+#define OPL3SA2_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .index = xindex, \
+ .info = snd_wss_info_double, \
+ .get = snd_opl3sa2_get_double, .put = snd_opl3sa2_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = (xtlv) } }
+
+static int snd_opl3sa2_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_opl3sa2 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->ctlregs[left_reg] >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (chip->ctlregs[right_reg] >> shift_right) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_opl3sa2_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_opl3sa2 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned short val1, val2, oval1, oval2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (left_reg != right_reg) {
+ oval1 = chip->ctlregs[left_reg];
+ oval2 = chip->ctlregs[right_reg];
+ val1 = (oval1 & ~(mask << shift_left)) | val1;
+ val2 = (oval2 & ~(mask << shift_right)) | val2;
+ change = val1 != oval1 || val2 != oval2;
+ __snd_opl3sa2_write(chip, left_reg, val1);
+ __snd_opl3sa2_write(chip, right_reg, val2);
+ } else {
+ oval1 = chip->ctlregs[left_reg];
+ val1 = (oval1 & ~((mask << shift_left) | (mask << shift_right))) | val1 | val2;
+ change = val1 != oval1;
+ __snd_opl3sa2_write(chip, left_reg, val1);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_master, -3000, 200, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0);
+
+static struct snd_kcontrol_new snd_opl3sa2_controls[] = {
+OPL3SA2_DOUBLE("Master Playback Switch", 0, 0x07, 0x08, 7, 7, 1, 1),
+OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1,
+ db_scale_master),
+OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1),
+OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1,
+ db_scale_5bit_12db_max),
+};
+
+static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = {
+OPL3SA2_DOUBLE("3D Control - Wide", 0, 0x14, 0x14, 4, 0, 7, 0),
+OPL3SA2_DOUBLE("Tone Control - Bass", 0, 0x15, 0x15, 4, 0, 7, 0),
+OPL3SA2_DOUBLE("Tone Control - Treble", 0, 0x16, 0x16, 4, 0, 7, 0)
+};
+
+static void snd_opl3sa2_master_free(struct snd_kcontrol *kcontrol)
+{
+ struct snd_opl3sa2 *chip = snd_kcontrol_chip(kcontrol);
+ chip->master_switch = NULL;
+ chip->master_volume = NULL;
+}
+
+static int __devinit snd_opl3sa2_mixer(struct snd_card *card)
+{
+ struct snd_opl3sa2 *chip = card->private_data;
+ struct snd_ctl_elem_id id1, id2;
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+ int err;
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ /* reassign AUX0 to CD */
+ strcpy(id1.name, "Aux Playback Switch");
+ strcpy(id2.name, "CD Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) {
+ snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n");
+ return err;
+ }
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "CD Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) {
+ snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n");
+ return err;
+ }
+ /* reassign AUX1 to FM */
+ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1;
+ strcpy(id2.name, "FM Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) {
+ snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n");
+ return err;
+ }
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "FM Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) {
+ snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n");
+ return err;
+ }
+ /* add OPL3SA2 controls */
+ for (idx = 0; idx < ARRAY_SIZE(snd_opl3sa2_controls); idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_opl3sa2_controls[idx], chip))) < 0)
+ return err;
+ switch (idx) {
+ case 0: chip->master_switch = kctl; kctl->private_free = snd_opl3sa2_master_free; break;
+ case 1: chip->master_volume = kctl; kctl->private_free = snd_opl3sa2_master_free; break;
+ }
+ }
+ if (chip->version > 2) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_opl3sa2_tone_controls); idx++)
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_opl3sa2_tone_controls[idx], chip))) < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* Power Management support functions */
+#ifdef CONFIG_PM
+static int snd_opl3sa2_suspend(struct snd_card *card, pm_message_t state)
+{
+ if (card) {
+ struct snd_opl3sa2 *chip = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ chip->wss->suspend(chip->wss);
+ /* power down */
+ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3);
+ }
+
+ return 0;
+}
+
+static int snd_opl3sa2_resume(struct snd_card *card)
+{
+ struct snd_opl3sa2 *chip;
+ int i;
+
+ if (!card)
+ return 0;
+
+ chip = card->private_data;
+ /* power up */
+ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0);
+
+ /* restore registers */
+ for (i = 2; i <= 0x0a; i++) {
+ if (i != OPL3SA2_IRQ_STATUS)
+ snd_opl3sa2_write(chip, i, chip->ctlregs[i]);
+ }
+ if (chip->version > 2) {
+ for (i = 0x12; i <= 0x16; i++)
+ snd_opl3sa2_write(chip, i, chip->ctlregs[i]);
+ }
+ /* restore wss */
+ chip->wss->resume(chip->wss);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+#ifdef CONFIG_PNP
+static int __devinit snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip,
+ struct pnp_dev *pdev)
+{
+ if (pnp_activate_dev(pdev) < 0) {
+ snd_printk(KERN_ERR "PnP configure failure (out of resources?)\n");
+ return -EBUSY;
+ }
+ sb_port[dev] = pnp_port_start(pdev, 0);
+ wss_port[dev] = pnp_port_start(pdev, 1);
+ fm_port[dev] = pnp_port_start(pdev, 2);
+ midi_port[dev] = pnp_port_start(pdev, 3);
+ port[dev] = pnp_port_start(pdev, 4);
+ dma1[dev] = pnp_dma(pdev, 0);
+ dma2[dev] = pnp_dma(pdev, 1);
+ irq[dev] = pnp_irq(pdev, 0);
+ snd_printdd("%sPnP OPL3-SA: sb port=0x%lx, wss port=0x%lx, fm port=0x%lx, midi port=0x%lx\n",
+ pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", sb_port[dev], wss_port[dev], fm_port[dev], midi_port[dev]);
+ snd_printdd("%sPnP OPL3-SA: control port=0x%lx, dma1=%i, dma2=%i, irq=%i\n",
+ pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", port[dev], dma1[dev], dma2[dev], irq[dev]);
+ return 0;
+}
+#endif /* CONFIG_PNP */
+
+static void snd_opl3sa2_free(struct snd_card *card)
+{
+ struct snd_opl3sa2 *chip = card->private_data;
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *)chip);
+ release_and_free_resource(chip->res_port);
+}
+
+static struct snd_card *snd_opl3sa2_card_new(int dev)
+{
+ struct snd_card *card;
+ struct snd_opl3sa2 *chip;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_opl3sa2));
+ if (card == NULL)
+ return NULL;
+ strcpy(card->driver, "OPL3SA2");
+ strcpy(card->shortname, "Yamaha OPL3-SA2");
+ chip = card->private_data;
+ spin_lock_init(&chip->reg_lock);
+ chip->irq = -1;
+ card->private_free = snd_opl3sa2_free;
+ return card;
+}
+
+static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev)
+{
+ int xirq, xdma1, xdma2;
+ struct snd_opl3sa2 *chip;
+ struct snd_wss *wss;
+ struct snd_opl3 *opl3;
+ int err;
+
+ /* initialise this card from supplied (or default) parameter*/
+ chip = card->private_data;
+ chip->ymode = opl3sa3_ymode[dev] & 0x03 ;
+ chip->port = port[dev];
+ xirq = irq[dev];
+ xdma1 = dma1[dev];
+ xdma2 = dma2[dev];
+ if (xdma2 < 0)
+ chip->single_dma = 1;
+ err = snd_opl3sa2_detect(card);
+ if (err < 0)
+ return err;
+ err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED,
+ "OPL3-SA2", card);
+ if (err) {
+ snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq);
+ return -ENODEV;
+ }
+ chip->irq = xirq;
+ err = snd_wss_create(card,
+ wss_port[dev] + 4, -1,
+ xirq, xdma1, xdma2,
+ WSS_HW_OPL3SA2, WSS_HWSHARE_IRQ, &wss);
+ if (err < 0) {
+ snd_printd("Oops, WSS not detected at 0x%lx\n", wss_port[dev] + 4);
+ return err;
+ }
+ chip->wss = wss;
+ err = snd_wss_pcm(wss, 0, NULL);
+ if (err < 0)
+ return err;
+ err = snd_wss_mixer(wss);
+ if (err < 0)
+ return err;
+ err = snd_opl3sa2_mixer(card);
+ if (err < 0)
+ return err;
+ err = snd_wss_timer(wss, 0, NULL);
+ if (err < 0)
+ return err;
+ if (fm_port[dev] >= 0x340 && fm_port[dev] < 0x400) {
+ if ((err = snd_opl3_create(card, fm_port[dev],
+ fm_port[dev] + 2,
+ OPL3_HW_OPL3, 0, &opl3)) < 0)
+ return err;
+ if ((err = snd_opl3_timer_new(opl3, 1, 2)) < 0)
+ return err;
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, &chip->synth)) < 0)
+ return err;
+ }
+ if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x340) {
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_OPL3SA2,
+ midi_port[dev], 0,
+ xirq, 0, &chip->rmidi)) < 0)
+ return err;
+ }
+ sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
+ card->shortname, chip->port, xirq, xdma1);
+ if (xdma2 >= 0)
+ sprintf(card->longname + strlen(card->longname), "&%d", xdma2);
+
+ return snd_card_register(card);
+}
+
+#ifdef CONFIG_PNP
+static int __devinit snd_opl3sa2_pnp_detect(struct pnp_dev *pdev,
+ const struct pnp_device_id *id)
+{
+ static int dev;
+ int err;
+ struct snd_card *card;
+
+ if (pnp_device_is_isapnp(pdev))
+ return -ENOENT; /* we have another procedure - card */
+ for (; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_opl3sa2_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ if ((err = snd_opl3sa2_pnp(dev, card->private_data, pdev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_card_set_dev(card, &pdev->dev);
+ if ((err = snd_opl3sa2_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pnp_set_drvdata(pdev, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_opl3sa2_pnp_remove(struct pnp_dev * pdev)
+{
+ snd_card_free(pnp_get_drvdata(pdev));
+ pnp_set_drvdata(pdev, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_opl3sa2_pnp_suspend(struct pnp_dev *pdev, pm_message_t state)
+{
+ return snd_opl3sa2_suspend(pnp_get_drvdata(pdev), state);
+}
+static int snd_opl3sa2_pnp_resume(struct pnp_dev *pdev)
+{
+ return snd_opl3sa2_resume(pnp_get_drvdata(pdev));
+}
+#endif
+
+static struct pnp_driver opl3sa2_pnp_driver = {
+ .name = "snd-opl3sa2-pnpbios",
+ .id_table = snd_opl3sa2_pnpbiosids,
+ .probe = snd_opl3sa2_pnp_detect,
+ .remove = __devexit_p(snd_opl3sa2_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_opl3sa2_pnp_suspend,
+ .resume = snd_opl3sa2_pnp_resume,
+#endif
+};
+
+static int __devinit snd_opl3sa2_pnp_cdetect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *id)
+{
+ static int dev;
+ struct pnp_dev *pdev;
+ int err;
+ struct snd_card *card;
+
+ pdev = pnp_request_card_device(pcard, id->devs[0].id, NULL);
+ if (pdev == NULL) {
+ snd_printk(KERN_ERR PFX "can't get pnp device from id '%s'\n",
+ id->devs[0].id);
+ return -EBUSY;
+ }
+ for (; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_opl3sa2_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ if ((err = snd_opl3sa2_pnp(dev, card->private_data, pdev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_card_set_dev(card, &pdev->dev);
+ if ((err = snd_opl3sa2_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_opl3sa2_pnp_cremove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_opl3sa2_pnp_csuspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ return snd_opl3sa2_suspend(pnp_get_card_drvdata(pcard), state);
+}
+static int snd_opl3sa2_pnp_cresume(struct pnp_card_link *pcard)
+{
+ return snd_opl3sa2_resume(pnp_get_card_drvdata(pcard));
+}
+#endif
+
+static struct pnp_card_driver opl3sa2_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "snd-opl3sa2-cpnp",
+ .id_table = snd_opl3sa2_pnpids,
+ .probe = snd_opl3sa2_pnp_cdetect,
+ .remove = __devexit_p(snd_opl3sa2_pnp_cremove),
+#ifdef CONFIG_PM
+ .suspend = snd_opl3sa2_pnp_csuspend,
+ .resume = snd_opl3sa2_pnp_cresume,
+#endif
+};
+#endif /* CONFIG_PNP */
+
+static int __devinit snd_opl3sa2_isa_match(struct device *pdev,
+ unsigned int dev)
+{
+ if (!enable[dev])
+ return 0;
+#ifdef CONFIG_PNP
+ if (isapnp[dev])
+ return 0;
+#endif
+ if (port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify port\n");
+ return 0;
+ }
+ if (wss_port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify wss_port\n");
+ return 0;
+ }
+ if (fm_port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify fm_port\n");
+ return 0;
+ }
+ if (midi_port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify midi_port\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_opl3sa2_isa_probe(struct device *pdev,
+ unsigned int dev)
+{
+ struct snd_card *card;
+ int err;
+
+ card = snd_opl3sa2_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ snd_card_set_dev(card, pdev);
+ if ((err = snd_opl3sa2_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ dev_set_drvdata(pdev, card);
+ return 0;
+}
+
+static int __devexit snd_opl3sa2_isa_remove(struct device *devptr,
+ unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_opl3sa2_isa_suspend(struct device *dev, unsigned int n,
+ pm_message_t state)
+{
+ return snd_opl3sa2_suspend(dev_get_drvdata(dev), state);
+}
+
+static int snd_opl3sa2_isa_resume(struct device *dev, unsigned int n)
+{
+ return snd_opl3sa2_resume(dev_get_drvdata(dev));
+}
+#endif
+
+#define DEV_NAME "opl3sa2"
+
+static struct isa_driver snd_opl3sa2_isa_driver = {
+ .match = snd_opl3sa2_isa_match,
+ .probe = snd_opl3sa2_isa_probe,
+ .remove = __devexit_p(snd_opl3sa2_isa_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_opl3sa2_isa_suspend,
+ .resume = snd_opl3sa2_isa_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+static int __init alsa_card_opl3sa2_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&snd_opl3sa2_isa_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+
+ err = pnp_register_driver(&opl3sa2_pnp_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ err = pnp_register_card_driver(&opl3sa2_pnpc_driver);
+ if (!err)
+ pnpc_registered = 1;
+
+ if (isa_registered || pnp_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit alsa_card_opl3sa2_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnpc_registered)
+ pnp_unregister_card_driver(&opl3sa2_pnpc_driver);
+ if (pnp_registered)
+ pnp_unregister_driver(&opl3sa2_pnp_driver);
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&snd_opl3sa2_isa_driver);
+}
+
+module_init(alsa_card_opl3sa2_init)
+module_exit(alsa_card_opl3sa2_exit)
diff --git a/sound/isa/opti9xx/Makefile b/sound/isa/opti9xx/Makefile
new file mode 100644
index 0000000..b4d894d
--- /dev/null
+++ b/sound/isa/opti9xx/Makefile
@@ -0,0 +1,15 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-opti92x-ad1848-objs := opti92x-ad1848.o
+snd-opti92x-cs4231-objs := opti92x-cs4231.o
+snd-opti93x-objs := opti93x.o
+snd-miro-objs := miro.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_OPTI92X_AD1848) += snd-opti92x-ad1848.o
+obj-$(CONFIG_SND_OPTI92X_CS4231) += snd-opti92x-cs4231.o
+obj-$(CONFIG_SND_OPTI93X) += snd-opti93x.o
+obj-$(CONFIG_SND_MIRO) += snd-miro.o
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
new file mode 100644
index 0000000..440755c
--- /dev/null
+++ b/sound/isa/opti9xx/miro.c
@@ -0,0 +1,1444 @@
+/*
+ * ALSA soundcard driver for Miro miroSOUND PCM1 pro
+ * miroSOUND PCM12
+ * miroSOUND PCM20 Radio
+ *
+ * Copyright (C) 2004-2005 Martin Langer <martin-langer@gmx.de>
+ *
+ * Based on OSS ACI and ALSA OPTi9xx drivers
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/moduleparam.h>
+#include <asm/io.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/mpu401.h>
+#include <sound/opl4.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+#include "miro.h"
+
+MODULE_AUTHOR("Martin Langer <martin-langer@gmx.de>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Miro miroSOUND PCM1 pro, PCM12, PCM20 Radio");
+MODULE_SUPPORTED_DEVICE("{{Miro,miroSOUND PCM1 pro}, "
+ "{Miro,miroSOUND PCM12}, "
+ "{Miro,miroSOUND PCM20 Radio}}");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */
+static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */
+static long fm_port = SNDRV_DEFAULT_PORT1; /* 0x388 */
+static int irq = SNDRV_DEFAULT_IRQ1; /* 5,7,9,10,11 */
+static int mpu_irq = SNDRV_DEFAULT_IRQ1; /* 5,7,9,10 */
+static int dma1 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */
+static int dma2 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */
+static int wss;
+static int ide;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for miro soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for miro soundcard.");
+module_param(port, long, 0444);
+MODULE_PARM_DESC(port, "WSS port # for miro driver.");
+module_param(mpu_port, long, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for miro driver.");
+module_param(fm_port, long, 0444);
+MODULE_PARM_DESC(fm_port, "FM Port # for miro driver.");
+module_param(irq, int, 0444);
+MODULE_PARM_DESC(irq, "WSS irq # for miro driver.");
+module_param(mpu_irq, int, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 irq # for miro driver.");
+module_param(dma1, int, 0444);
+MODULE_PARM_DESC(dma1, "1st dma # for miro driver.");
+module_param(dma2, int, 0444);
+MODULE_PARM_DESC(dma2, "2nd dma # for miro driver.");
+module_param(wss, int, 0444);
+MODULE_PARM_DESC(wss, "wss mode");
+module_param(ide, int, 0444);
+MODULE_PARM_DESC(ide, "enable ide port");
+
+#define OPTi9XX_HW_DETECT 0
+#define OPTi9XX_HW_82C928 1
+#define OPTi9XX_HW_82C929 2
+#define OPTi9XX_HW_82C924 3
+#define OPTi9XX_HW_82C925 4
+#define OPTi9XX_HW_82C930 5
+#define OPTi9XX_HW_82C931 6
+#define OPTi9XX_HW_82C933 7
+#define OPTi9XX_HW_LAST OPTi9XX_HW_82C933
+
+#define OPTi9XX_MC_REG(n) n
+
+
+struct snd_miro {
+ unsigned short hardware;
+ unsigned char password;
+ char name[7];
+
+ struct resource *res_mc_base;
+ struct resource *res_aci_port;
+
+ unsigned long mc_base;
+ unsigned long mc_base_size;
+ unsigned long pwd_reg;
+
+ spinlock_t lock;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+
+ long wss_base;
+ int irq;
+ int dma1;
+ int dma2;
+
+ long fm_port;
+
+ long mpu_port;
+ int mpu_irq;
+
+ unsigned long aci_port;
+ int aci_vendor;
+ int aci_product;
+ int aci_version;
+ int aci_amp;
+ int aci_preamp;
+ int aci_solomode;
+
+ struct mutex aci_mutex;
+};
+
+static void snd_miro_proc_init(struct snd_miro * miro);
+
+static char * snd_opti9xx_names[] = {
+ "unkown",
+ "82C928", "82C929",
+ "82C924", "82C925",
+ "82C930", "82C931", "82C933"
+};
+
+/*
+ * ACI control
+ */
+
+static int aci_busy_wait(struct snd_miro * miro)
+{
+ long timeout;
+ unsigned char byte;
+
+ for (timeout = 1; timeout <= ACI_MINTIME+30; timeout++) {
+ if (((byte=inb(miro->aci_port + ACI_REG_BUSY)) & 1) == 0) {
+ if (timeout >= ACI_MINTIME)
+ snd_printd("aci ready in round %ld.\n",
+ timeout-ACI_MINTIME);
+ return byte;
+ }
+ if (timeout >= ACI_MINTIME) {
+ long out=10*HZ;
+ switch (timeout-ACI_MINTIME) {
+ case 0 ... 9:
+ out /= 10;
+ case 10 ... 19:
+ out /= 10;
+ case 20 ... 30:
+ out /= 10;
+ default:
+ set_current_state(TASK_UNINTERRUPTIBLE);
+ schedule_timeout(out);
+ break;
+ }
+ }
+ }
+ snd_printk(KERN_ERR "aci_busy_wait() time out\n");
+ return -EBUSY;
+}
+
+static inline int aci_write(struct snd_miro * miro, unsigned char byte)
+{
+ if (aci_busy_wait(miro) >= 0) {
+ outb(byte, miro->aci_port + ACI_REG_COMMAND);
+ return 0;
+ } else {
+ snd_printk(KERN_ERR "aci busy, aci_write(0x%x) stopped.\n", byte);
+ return -EBUSY;
+ }
+}
+
+static inline int aci_read(struct snd_miro * miro)
+{
+ unsigned char byte;
+
+ if (aci_busy_wait(miro) >= 0) {
+ byte=inb(miro->aci_port + ACI_REG_STATUS);
+ return byte;
+ } else {
+ snd_printk(KERN_ERR "aci busy, aci_read() stopped.\n");
+ return -EBUSY;
+ }
+}
+
+static int aci_cmd(struct snd_miro * miro, int write1, int write2, int write3)
+{
+ int write[] = {write1, write2, write3};
+ int value, i;
+
+ if (mutex_lock_interruptible(&miro->aci_mutex))
+ return -EINTR;
+
+ for (i=0; i<3; i++) {
+ if (write[i]< 0 || write[i] > 255)
+ break;
+ else {
+ value = aci_write(miro, write[i]);
+ if (value < 0)
+ goto out;
+ }
+ }
+
+ value = aci_read(miro);
+
+out: mutex_unlock(&miro->aci_mutex);
+ return value;
+}
+
+static int aci_getvalue(struct snd_miro * miro, unsigned char index)
+{
+ return aci_cmd(miro, ACI_STATUS, index, -1);
+}
+
+static int aci_setvalue(struct snd_miro * miro, unsigned char index, int value)
+{
+ return aci_cmd(miro, index, value, -1);
+}
+
+/*
+ * MIXER part
+ */
+
+#define snd_miro_info_capture snd_ctl_boolean_mono_info
+
+static int snd_miro_get_capture(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ int value;
+
+ if ((value = aci_getvalue(miro, ACI_S_GENERAL)) < 0) {
+ snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", value);
+ return value;
+ }
+
+ ucontrol->value.integer.value[0] = value & 0x20;
+
+ return 0;
+}
+
+static int snd_miro_put_capture(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ int change, value, error;
+
+ value = !(ucontrol->value.integer.value[0]);
+
+ if ((error = aci_setvalue(miro, ACI_SET_SOLOMODE, value)) < 0) {
+ snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", error);
+ return error;
+ }
+
+ change = (value != miro->aci_solomode);
+ miro->aci_solomode = value;
+
+ return change;
+}
+
+static int snd_miro_info_preamp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 3;
+
+ return 0;
+}
+
+static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ int value;
+
+ if (miro->aci_version <= 176) {
+
+ /*
+ OSS says it's not readable with versions < 176.
+ But it doesn't work on my card,
+ which is a PCM12 with aci_version = 176.
+ */
+
+ ucontrol->value.integer.value[0] = miro->aci_preamp;
+ return 0;
+ }
+
+ if ((value = aci_getvalue(miro, ACI_GET_PREAMP)) < 0) {
+ snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", value);
+ return value;
+ }
+
+ ucontrol->value.integer.value[0] = value;
+
+ return 0;
+}
+
+static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ int error, value, change;
+
+ value = ucontrol->value.integer.value[0];
+
+ if ((error = aci_setvalue(miro, ACI_SET_PREAMP, value)) < 0) {
+ snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", error);
+ return error;
+ }
+
+ change = (value != miro->aci_preamp);
+ miro->aci_preamp = value;
+
+ return change;
+}
+
+#define snd_miro_info_amp snd_ctl_boolean_mono_info
+
+static int snd_miro_get_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = miro->aci_amp;
+
+ return 0;
+}
+
+static int snd_miro_put_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ int error, value, change;
+
+ value = ucontrol->value.integer.value[0];
+
+ if ((error = aci_setvalue(miro, ACI_SET_POWERAMP, value)) < 0) {
+ snd_printk(KERN_ERR "snd_miro_put_amp() to %d failed: %d\n", value, error);
+ return error;
+ }
+
+ change = (value != miro->aci_amp);
+ miro->aci_amp = value;
+
+ return change;
+}
+
+#define MIRO_DOUBLE(ctl_name, ctl_index, get_right_reg, set_right_reg) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = ctl_name, \
+ .index = ctl_index, \
+ .info = snd_miro_info_double, \
+ .get = snd_miro_get_double, \
+ .put = snd_miro_put_double, \
+ .private_value = get_right_reg | (set_right_reg << 8) \
+}
+
+static int snd_miro_info_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int reg = kcontrol->private_value & 0xff;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+
+ if ((reg >= ACI_GET_EQ1) && (reg <= ACI_GET_EQ7)) {
+
+ /* equalizer elements */
+
+ uinfo->value.integer.min = - 0x7f;
+ uinfo->value.integer.max = 0x7f;
+ } else {
+
+ /* non-equalizer elements */
+
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0x20;
+ }
+
+ return 0;
+}
+
+static int snd_miro_get_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uinfo)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ int left_val, right_val;
+
+ int right_reg = kcontrol->private_value & 0xff;
+ int left_reg = right_reg + 1;
+
+ if ((right_val = aci_getvalue(miro, right_reg)) < 0) {
+ snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", right_reg, right_val);
+ return right_val;
+ }
+
+ if ((left_val = aci_getvalue(miro, left_reg)) < 0) {
+ snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", left_reg, left_val);
+ return left_val;
+ }
+
+ if ((right_reg >= ACI_GET_EQ1) && (right_reg <= ACI_GET_EQ7)) {
+
+ /* equalizer elements */
+
+ if (left_val < 0x80) {
+ uinfo->value.integer.value[0] = left_val;
+ } else {
+ uinfo->value.integer.value[0] = 0x80 - left_val;
+ }
+
+ if (right_val < 0x80) {
+ uinfo->value.integer.value[1] = right_val;
+ } else {
+ uinfo->value.integer.value[1] = 0x80 - right_val;
+ }
+
+ } else {
+
+ /* non-equalizer elements */
+
+ uinfo->value.integer.value[0] = 0x20 - left_val;
+ uinfo->value.integer.value[1] = 0x20 - right_val;
+ }
+
+ return 0;
+}
+
+static int snd_miro_put_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_miro *miro = snd_kcontrol_chip(kcontrol);
+ int left, right, left_old, right_old;
+ int setreg_left, setreg_right, getreg_left, getreg_right;
+ int change, error;
+
+ left = ucontrol->value.integer.value[0];
+ right = ucontrol->value.integer.value[1];
+
+ setreg_right = (kcontrol->private_value >> 8) & 0xff;
+ if (setreg_right == ACI_SET_MASTER) {
+ setreg_left = setreg_right + 1;
+ } else {
+ setreg_left = setreg_right + 8;
+ }
+
+ getreg_right = kcontrol->private_value & 0xff;
+ getreg_left = getreg_right + 1;
+
+ if ((left_old = aci_getvalue(miro, getreg_left)) < 0) {
+ snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_left, left_old);
+ return left_old;
+ }
+
+ if ((right_old = aci_getvalue(miro, getreg_right)) < 0) {
+ snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_right, right_old);
+ return right_old;
+ }
+
+ if ((getreg_right >= ACI_GET_EQ1) && (getreg_right <= ACI_GET_EQ7)) {
+
+ /* equalizer elements */
+
+ if (left < -0x7f || left > 0x7f ||
+ right < -0x7f || right > 0x7f)
+ return -EINVAL;
+
+ if (left_old > 0x80)
+ left_old = 0x80 - left_old;
+ if (right_old > 0x80)
+ right_old = 0x80 - right_old;
+
+ if (left >= 0) {
+ if ((error = aci_setvalue(miro, setreg_left, left)) < 0) {
+ snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n",
+ left, error);
+ return error;
+ }
+ } else {
+ if ((error = aci_setvalue(miro, setreg_left, 0x80 - left)) < 0) {
+ snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n",
+ 0x80 - left, error);
+ return error;
+ }
+ }
+
+ if (right >= 0) {
+ if ((error = aci_setvalue(miro, setreg_right, right)) < 0) {
+ snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n",
+ right, error);
+ return error;
+ }
+ } else {
+ if ((error = aci_setvalue(miro, setreg_right, 0x80 - right)) < 0) {
+ snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n",
+ 0x80 - right, error);
+ return error;
+ }
+ }
+
+ } else {
+
+ /* non-equalizer elements */
+
+ if (left < 0 || left > 0x20 ||
+ right < 0 || right > 0x20)
+ return -EINVAL;
+
+ left_old = 0x20 - left_old;
+ right_old = 0x20 - right_old;
+
+ if ((error = aci_setvalue(miro, setreg_left, 0x20 - left)) < 0) {
+ snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n",
+ 0x20 - left, error);
+ return error;
+ }
+ if ((error = aci_setvalue(miro, setreg_right, 0x20 - right)) < 0) {
+ snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n",
+ 0x20 - right, error);
+ return error;
+ }
+ }
+
+ change = (left != left_old) || (right != right_old);
+
+ return change;
+}
+
+static struct snd_kcontrol_new snd_miro_controls[] __devinitdata = {
+MIRO_DOUBLE("Master Playback Volume", 0, ACI_GET_MASTER, ACI_SET_MASTER),
+MIRO_DOUBLE("Mic Playback Volume", 1, ACI_GET_MIC, ACI_SET_MIC),
+MIRO_DOUBLE("Line Playback Volume", 1, ACI_GET_LINE, ACI_SET_LINE),
+MIRO_DOUBLE("CD Playback Volume", 0, ACI_GET_CD, ACI_SET_CD),
+MIRO_DOUBLE("Synth Playback Volume", 0, ACI_GET_SYNTH, ACI_SET_SYNTH),
+MIRO_DOUBLE("PCM Playback Volume", 1, ACI_GET_PCM, ACI_SET_PCM),
+MIRO_DOUBLE("Aux Playback Volume", 2, ACI_GET_LINE2, ACI_SET_LINE2),
+};
+
+/* Equalizer with seven bands (only PCM20)
+ from -12dB up to +12dB on each band */
+static struct snd_kcontrol_new snd_miro_eq_controls[] __devinitdata = {
+MIRO_DOUBLE("Tone Control - 28 Hz", 0, ACI_GET_EQ1, ACI_SET_EQ1),
+MIRO_DOUBLE("Tone Control - 160 Hz", 0, ACI_GET_EQ2, ACI_SET_EQ2),
+MIRO_DOUBLE("Tone Control - 400 Hz", 0, ACI_GET_EQ3, ACI_SET_EQ3),
+MIRO_DOUBLE("Tone Control - 1 kHz", 0, ACI_GET_EQ4, ACI_SET_EQ4),
+MIRO_DOUBLE("Tone Control - 2.5 kHz", 0, ACI_GET_EQ5, ACI_SET_EQ5),
+MIRO_DOUBLE("Tone Control - 6.3 kHz", 0, ACI_GET_EQ6, ACI_SET_EQ6),
+MIRO_DOUBLE("Tone Control - 16 kHz", 0, ACI_GET_EQ7, ACI_SET_EQ7),
+};
+
+static struct snd_kcontrol_new snd_miro_radio_control[] __devinitdata = {
+MIRO_DOUBLE("Radio Playback Volume", 0, ACI_GET_LINE1, ACI_SET_LINE1),
+};
+
+static struct snd_kcontrol_new snd_miro_line_control[] __devinitdata = {
+MIRO_DOUBLE("Line Playback Volume", 2, ACI_GET_LINE1, ACI_SET_LINE1),
+};
+
+static struct snd_kcontrol_new snd_miro_preamp_control[] __devinitdata = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Boost",
+ .index = 1,
+ .info = snd_miro_info_preamp,
+ .get = snd_miro_get_preamp,
+ .put = snd_miro_put_preamp,
+}};
+
+static struct snd_kcontrol_new snd_miro_amp_control[] __devinitdata = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Boost",
+ .index = 0,
+ .info = snd_miro_info_amp,
+ .get = snd_miro_get_amp,
+ .put = snd_miro_put_amp,
+}};
+
+static struct snd_kcontrol_new snd_miro_capture_control[] __devinitdata = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Capture Switch",
+ .index = 0,
+ .info = snd_miro_info_capture,
+ .get = snd_miro_get_capture,
+ .put = snd_miro_put_capture,
+}};
+
+static unsigned char aci_init_values[][2] __devinitdata = {
+ { ACI_SET_MUTE, 0x00 },
+ { ACI_SET_POWERAMP, 0x00 },
+ { ACI_SET_PREAMP, 0x00 },
+ { ACI_SET_SOLOMODE, 0x00 },
+ { ACI_SET_MIC + 0, 0x20 },
+ { ACI_SET_MIC + 8, 0x20 },
+ { ACI_SET_LINE + 0, 0x20 },
+ { ACI_SET_LINE + 8, 0x20 },
+ { ACI_SET_CD + 0, 0x20 },
+ { ACI_SET_CD + 8, 0x20 },
+ { ACI_SET_PCM + 0, 0x20 },
+ { ACI_SET_PCM + 8, 0x20 },
+ { ACI_SET_LINE1 + 0, 0x20 },
+ { ACI_SET_LINE1 + 8, 0x20 },
+ { ACI_SET_LINE2 + 0, 0x20 },
+ { ACI_SET_LINE2 + 8, 0x20 },
+ { ACI_SET_SYNTH + 0, 0x20 },
+ { ACI_SET_SYNTH + 8, 0x20 },
+ { ACI_SET_MASTER + 0, 0x20 },
+ { ACI_SET_MASTER + 1, 0x20 },
+};
+
+static int __devinit snd_set_aci_init_values(struct snd_miro *miro)
+{
+ int idx, error;
+
+ /* enable WSS on PCM1 */
+
+ if ((miro->aci_product == 'A') && wss) {
+ if ((error = aci_setvalue(miro, ACI_SET_WSS, wss)) < 0) {
+ snd_printk(KERN_ERR "enabling WSS mode failed\n");
+ return error;
+ }
+ }
+
+ /* enable IDE port */
+
+ if (ide) {
+ if ((error = aci_setvalue(miro, ACI_SET_IDE, ide)) < 0) {
+ snd_printk(KERN_ERR "enabling IDE port failed\n");
+ return error;
+ }
+ }
+
+ /* set common aci values */
+
+ for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++)
+ if ((error = aci_setvalue(miro, aci_init_values[idx][0],
+ aci_init_values[idx][1])) < 0) {
+ snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n",
+ aci_init_values[idx][0], error);
+ return error;
+ }
+
+ miro->aci_amp = 0;
+ miro->aci_preamp = 0;
+ miro->aci_solomode = 1;
+
+ return 0;
+}
+
+static int __devinit snd_miro_mixer(struct snd_miro *miro)
+{
+ struct snd_card *card;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!miro || !miro->card))
+ return -EINVAL;
+
+ card = miro->card;
+
+ switch (miro->hardware) {
+ case OPTi9XX_HW_82C924:
+ strcpy(card->mixername, "ACI & OPTi924");
+ break;
+ case OPTi9XX_HW_82C929:
+ strcpy(card->mixername, "ACI & OPTi929");
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_miro_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_controls[idx], miro))) < 0)
+ return err;
+ }
+
+ if ((miro->aci_product == 'A') || (miro->aci_product == 'B')) {
+ /* PCM1/PCM12 with power-amp and Line 2 */
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_line_control[0], miro))) < 0)
+ return err;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_amp_control[0], miro))) < 0)
+ return err;
+ }
+
+ if ((miro->aci_product == 'B') || (miro->aci_product == 'C')) {
+ /* PCM12/PCM20 with mic-preamp */
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_preamp_control[0], miro))) < 0)
+ return err;
+ if (miro->aci_version >= 176)
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_capture_control[0], miro))) < 0)
+ return err;
+ }
+
+ if (miro->aci_product == 'C') {
+ /* PCM20 with radio and 7 band equalizer */
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_radio_control[0], miro))) < 0)
+ return err;
+ for (idx = 0; idx < ARRAY_SIZE(snd_miro_eq_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_eq_controls[idx], miro))) < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static long snd_legacy_find_free_ioport(long *port_table, long size)
+{
+ while (*port_table != -1) {
+ struct resource *res;
+ if ((res = request_region(*port_table, size,
+ "ALSA test")) != NULL) {
+ release_and_free_resource(res);
+ return *port_table;
+ }
+ port_table++;
+ }
+ return -1;
+}
+
+static int __devinit snd_miro_init(struct snd_miro *chip,
+ unsigned short hardware)
+{
+ static int opti9xx_mc_size[] = {7, 7, 10, 10, 2, 2, 2};
+
+ chip->hardware = hardware;
+ strcpy(chip->name, snd_opti9xx_names[hardware]);
+
+ chip->mc_base_size = opti9xx_mc_size[hardware];
+
+ spin_lock_init(&chip->lock);
+
+ chip->wss_base = -1;
+ chip->irq = -1;
+ chip->dma1 = -1;
+ chip->dma2 = -1;
+ chip->fm_port = -1;
+ chip->mpu_port = -1;
+ chip->mpu_irq = -1;
+
+ switch (hardware) {
+ case OPTi9XX_HW_82C929:
+ chip->mc_base = 0xf8c;
+ chip->password = 0xe3;
+ chip->pwd_reg = 3;
+ break;
+
+ case OPTi9XX_HW_82C924:
+ chip->mc_base = 0xf8c;
+ chip->password = 0xe5;
+ chip->pwd_reg = 3;
+ break;
+
+ default:
+ snd_printk(KERN_ERR "sorry, no support for %d\n", hardware);
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static unsigned char snd_miro_read(struct snd_miro *chip,
+ unsigned char reg)
+{
+ unsigned long flags;
+ unsigned char retval = 0xff;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+
+ switch (chip->hardware) {
+ case OPTi9XX_HW_82C924:
+ if (reg > 7) {
+ outb(reg, chip->mc_base + 8);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+ retval = inb(chip->mc_base + 9);
+ break;
+ }
+
+ case OPTi9XX_HW_82C929:
+ retval = inb(chip->mc_base + reg);
+ break;
+
+ default:
+ snd_printk(KERN_ERR "sorry, no support for %d\n", chip->hardware);
+ }
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return retval;
+}
+
+static void snd_miro_write(struct snd_miro *chip, unsigned char reg,
+ unsigned char value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+
+ switch (chip->hardware) {
+ case OPTi9XX_HW_82C924:
+ if (reg > 7) {
+ outb(reg, chip->mc_base + 8);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+ outb(value, chip->mc_base + 9);
+ break;
+ }
+
+ case OPTi9XX_HW_82C929:
+ outb(value, chip->mc_base + reg);
+ break;
+
+ default:
+ snd_printk(KERN_ERR "sorry, no support for %d\n", chip->hardware);
+ }
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+#define snd_miro_write_mask(chip, reg, value, mask) \
+ snd_miro_write(chip, reg, \
+ (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask)))
+
+/*
+ * Proc Interface
+ */
+
+static void snd_miro_proc_read(struct snd_info_entry * entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_miro *miro = (struct snd_miro *) entry->private_data;
+ char* model = "unknown";
+
+ /* miroSOUND PCM1 pro, early PCM12 */
+
+ if ((miro->hardware == OPTi9XX_HW_82C929) &&
+ (miro->aci_vendor == 'm') &&
+ (miro->aci_product == 'A')) {
+ switch(miro->aci_version) {
+ case 3:
+ model = "miroSOUND PCM1 pro";
+ break;
+ default:
+ model = "miroSOUND PCM1 pro / (early) PCM12";
+ break;
+ }
+ }
+
+ /* miroSOUND PCM12, PCM12 (Rev. E), PCM12 pnp */
+
+ if ((miro->hardware == OPTi9XX_HW_82C924) &&
+ (miro->aci_vendor == 'm') &&
+ (miro->aci_product == 'B')) {
+ switch(miro->aci_version) {
+ case 4:
+ model = "miroSOUND PCM12";
+ break;
+ case 176:
+ model = "miroSOUND PCM12 (Rev. E)";
+ break;
+ default:
+ model = "miroSOUND PCM12 / PCM12 pnp";
+ break;
+ }
+ }
+
+ /* miroSOUND PCM20 radio */
+
+ if ((miro->hardware == OPTi9XX_HW_82C924) &&
+ (miro->aci_vendor == 'm') &&
+ (miro->aci_product == 'C')) {
+ switch(miro->aci_version) {
+ case 7:
+ model = "miroSOUND PCM20 radio (Rev. E)";
+ break;
+ default:
+ model = "miroSOUND PCM20 radio";
+ break;
+ }
+ }
+
+ snd_iprintf(buffer, "\nGeneral information:\n");
+ snd_iprintf(buffer, " model : %s\n", model);
+ snd_iprintf(buffer, " opti : %s\n", miro->name);
+ snd_iprintf(buffer, " codec : %s\n", miro->pcm->name);
+ snd_iprintf(buffer, " port : 0x%lx\n", miro->wss_base);
+ snd_iprintf(buffer, " irq : %d\n", miro->irq);
+ snd_iprintf(buffer, " dma : %d,%d\n\n", miro->dma1, miro->dma2);
+
+ snd_iprintf(buffer, "MPU-401:\n");
+ snd_iprintf(buffer, " port : 0x%lx\n", miro->mpu_port);
+ snd_iprintf(buffer, " irq : %d\n\n", miro->mpu_irq);
+
+ snd_iprintf(buffer, "ACI information:\n");
+ snd_iprintf(buffer, " vendor : ");
+ switch(miro->aci_vendor) {
+ case 'm':
+ snd_iprintf(buffer, "Miro\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_vendor);
+ break;
+ }
+
+ snd_iprintf(buffer, " product : ");
+ switch(miro->aci_product) {
+ case 'A':
+ snd_iprintf(buffer, "miroSOUND PCM1 pro / (early) PCM12\n");
+ break;
+ case 'B':
+ snd_iprintf(buffer, "miroSOUND PCM12\n");
+ break;
+ case 'C':
+ snd_iprintf(buffer, "miroSOUND PCM20 radio\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_product);
+ break;
+ }
+
+ snd_iprintf(buffer, " firmware: %d (0x%x)\n",
+ miro->aci_version, miro->aci_version);
+ snd_iprintf(buffer, " port : 0x%lx-0x%lx\n",
+ miro->aci_port, miro->aci_port+2);
+ snd_iprintf(buffer, " wss : 0x%x\n", wss);
+ snd_iprintf(buffer, " ide : 0x%x\n", ide);
+ snd_iprintf(buffer, " solomode: 0x%x\n", miro->aci_solomode);
+ snd_iprintf(buffer, " amp : 0x%x\n", miro->aci_amp);
+ snd_iprintf(buffer, " preamp : 0x%x\n", miro->aci_preamp);
+}
+
+static void __devinit snd_miro_proc_init(struct snd_miro * miro)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(miro->card, "miro", &entry))
+ snd_info_set_text_ops(entry, miro, snd_miro_proc_read);
+}
+
+/*
+ * Init
+ */
+
+static int __devinit snd_miro_configure(struct snd_miro *chip)
+{
+ unsigned char wss_base_bits;
+ unsigned char irq_bits;
+ unsigned char dma_bits;
+ unsigned char mpu_port_bits = 0;
+ unsigned char mpu_irq_bits;
+ unsigned long flags;
+
+ switch (chip->hardware) {
+ case OPTi9XX_HW_82C924:
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02);
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80);
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02);
+ break;
+ case OPTi9XX_HW_82C929:
+ /* untested init commands for OPTi929 */
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80);
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(4), 0x00, 0x0c);
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02);
+ break;
+ default:
+ snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware);
+ return -EINVAL;
+ }
+
+ switch (chip->wss_base) {
+ case 0x530:
+ wss_base_bits = 0x00;
+ break;
+ case 0x604:
+ wss_base_bits = 0x03;
+ break;
+ case 0xe80:
+ wss_base_bits = 0x01;
+ break;
+ case 0xf40:
+ wss_base_bits = 0x02;
+ break;
+ default:
+ snd_printk(KERN_ERR "WSS port 0x%lx not valid\n", chip->wss_base);
+ goto __skip_base;
+ }
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30);
+
+__skip_base:
+ switch (chip->irq) {
+ case 5:
+ irq_bits = 0x05;
+ break;
+ case 7:
+ irq_bits = 0x01;
+ break;
+ case 9:
+ irq_bits = 0x02;
+ break;
+ case 10:
+ irq_bits = 0x03;
+ break;
+ case 11:
+ irq_bits = 0x04;
+ break;
+ default:
+ snd_printk(KERN_ERR "WSS irq # %d not valid\n", chip->irq);
+ goto __skip_resources;
+ }
+
+ switch (chip->dma1) {
+ case 0:
+ dma_bits = 0x01;
+ break;
+ case 1:
+ dma_bits = 0x02;
+ break;
+ case 3:
+ dma_bits = 0x03;
+ break;
+ default:
+ snd_printk(KERN_ERR "WSS dma1 # %d not valid\n", chip->dma1);
+ goto __skip_resources;
+ }
+
+ if (chip->dma1 == chip->dma2) {
+ snd_printk(KERN_ERR "don't want to share dmas\n");
+ return -EBUSY;
+ }
+
+ switch (chip->dma2) {
+ case 0:
+ case 1:
+ break;
+ default:
+ snd_printk(KERN_ERR "WSS dma2 # %d not valid\n", chip->dma2);
+ goto __skip_resources;
+ }
+ dma_bits |= 0x04;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ outb(irq_bits << 3 | dma_bits, chip->wss_base);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+__skip_resources:
+ if (chip->hardware > OPTi9XX_HW_82C928) {
+ switch (chip->mpu_port) {
+ case 0:
+ case -1:
+ break;
+ case 0x300:
+ mpu_port_bits = 0x03;
+ break;
+ case 0x310:
+ mpu_port_bits = 0x02;
+ break;
+ case 0x320:
+ mpu_port_bits = 0x01;
+ break;
+ case 0x330:
+ mpu_port_bits = 0x00;
+ break;
+ default:
+ snd_printk(KERN_ERR "MPU-401 port 0x%lx not valid\n",
+ chip->mpu_port);
+ goto __skip_mpu;
+ }
+
+ switch (chip->mpu_irq) {
+ case 5:
+ mpu_irq_bits = 0x02;
+ break;
+ case 7:
+ mpu_irq_bits = 0x03;
+ break;
+ case 9:
+ mpu_irq_bits = 0x00;
+ break;
+ case 10:
+ mpu_irq_bits = 0x01;
+ break;
+ default:
+ snd_printk(KERN_ERR "MPU-401 irq # %d not valid\n",
+ chip->mpu_irq);
+ goto __skip_mpu;
+ }
+
+ snd_miro_write_mask(chip, OPTi9XX_MC_REG(6),
+ (chip->mpu_port <= 0) ? 0x00 :
+ 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3,
+ 0xf8);
+ }
+__skip_mpu:
+
+ return 0;
+}
+
+static int __devinit snd_card_miro_detect(struct snd_card *card,
+ struct snd_miro *chip)
+{
+ int i, err;
+ unsigned char value;
+
+ for (i = OPTi9XX_HW_82C929; i <= OPTi9XX_HW_82C924; i++) {
+
+ if ((err = snd_miro_init(chip, i)) < 0)
+ return err;
+
+ if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL)
+ continue;
+
+ value = snd_miro_read(chip, OPTi9XX_MC_REG(1));
+ if ((value != 0xff) && (value != inb(chip->mc_base + 1)))
+ if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1)))
+ return 1;
+
+ release_and_free_resource(chip->res_mc_base);
+ chip->res_mc_base = NULL;
+
+ }
+
+ return -ENODEV;
+}
+
+static int __devinit snd_card_miro_aci_detect(struct snd_card *card,
+ struct snd_miro * miro)
+{
+ unsigned char regval;
+ int i;
+
+ mutex_init(&miro->aci_mutex);
+
+ /* get ACI port from OPTi9xx MC 4 */
+
+ miro->mc_base = 0xf8c;
+ regval=inb(miro->mc_base + 4);
+ miro->aci_port = (regval & 0x10) ? 0x344: 0x354;
+
+ if ((miro->res_aci_port = request_region(miro->aci_port, 3, "miro aci")) == NULL) {
+ snd_printk(KERN_ERR "aci i/o area 0x%lx-0x%lx already used.\n",
+ miro->aci_port, miro->aci_port+2);
+ return -ENOMEM;
+ }
+
+ /* force ACI into a known state */
+ for (i = 0; i < 3; i++)
+ if (aci_cmd(miro, ACI_ERROR_OP, -1, -1) < 0) {
+ snd_printk(KERN_ERR "can't force aci into known state.\n");
+ return -ENXIO;
+ }
+
+ if ((miro->aci_vendor=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0 ||
+ (miro->aci_product=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0) {
+ snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", miro->aci_port);
+ return -ENXIO;
+ }
+
+ if ((miro->aci_version=aci_cmd(miro, ACI_READ_VERSION, -1, -1)) < 0) {
+ snd_printk(KERN_ERR "can't read aci version on 0x%lx.\n",
+ miro->aci_port);
+ return -ENXIO;
+ }
+
+ if (aci_cmd(miro, ACI_INIT, -1, -1) < 0 ||
+ aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 ||
+ aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) {
+ snd_printk(KERN_ERR "can't initialize aci.\n");
+ return -ENXIO;
+ }
+
+ return 0;
+}
+
+static void snd_card_miro_free(struct snd_card *card)
+{
+ struct snd_miro *miro = card->private_data;
+
+ release_and_free_resource(miro->res_aci_port);
+ release_and_free_resource(miro->res_mc_base);
+}
+
+static int __devinit snd_miro_match(struct device *devptr, unsigned int n)
+{
+ return 1;
+}
+
+static int __devinit snd_miro_probe(struct device *devptr, unsigned int n)
+{
+ static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1};
+ static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1};
+ static int possible_irqs[] = {11, 9, 10, 7, -1};
+ static int possible_mpu_irqs[] = {10, 5, 9, 7, -1};
+ static int possible_dma1s[] = {3, 1, 0, -1};
+ static int possible_dma2s[][2] = {{1,-1}, {0,-1}, {-1,-1}, {0,-1}};
+
+ int error;
+ struct snd_miro *miro;
+ struct snd_wss *codec;
+ struct snd_timer *timer;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_rawmidi *rmidi;
+
+ if (!(card = snd_card_new(index, id, THIS_MODULE,
+ sizeof(struct snd_miro))))
+ return -ENOMEM;
+
+ card->private_free = snd_card_miro_free;
+ miro = card->private_data;
+ miro->card = card;
+
+ if ((error = snd_card_miro_aci_detect(card, miro)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to detect aci chip\n");
+ return -ENODEV;
+ }
+
+ /* init proc interface */
+ snd_miro_proc_init(miro);
+
+ if ((error = snd_card_miro_detect(card, miro)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n");
+ return -ENODEV;
+ }
+
+ if (! miro->res_mc_base &&
+ (miro->res_mc_base = request_region(miro->mc_base, miro->mc_base_size,
+ "miro (OPTi9xx MC)")) == NULL) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "request for OPTI9xx MC failed\n");
+ return -ENOMEM;
+ }
+
+ miro->wss_base = port;
+ miro->fm_port = fm_port;
+ miro->mpu_port = mpu_port;
+ miro->irq = irq;
+ miro->mpu_irq = mpu_irq;
+ miro->dma1 = dma1;
+ miro->dma2 = dma2;
+
+ if (miro->wss_base == SNDRV_AUTO_PORT) {
+ if ((miro->wss_base = snd_legacy_find_free_ioport(possible_ports, 4)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to find a free WSS port\n");
+ return -EBUSY;
+ }
+ }
+
+ if (miro->mpu_port == SNDRV_AUTO_PORT) {
+ if ((miro->mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to find a free MPU401 port\n");
+ return -EBUSY;
+ }
+ }
+ if (miro->irq == SNDRV_AUTO_IRQ) {
+ if ((miro->irq = snd_legacy_find_free_irq(possible_irqs)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to find a free IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (miro->mpu_irq == SNDRV_AUTO_IRQ) {
+ if ((miro->mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (miro->dma1 == SNDRV_AUTO_DMA) {
+ if ((miro->dma1 = snd_legacy_find_free_dma(possible_dma1s)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to find a free DMA1\n");
+ return -EBUSY;
+ }
+ }
+ if (miro->dma2 == SNDRV_AUTO_DMA) {
+ if ((miro->dma2 = snd_legacy_find_free_dma(possible_dma2s[miro->dma1 % 4])) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "unable to find a free DMA2\n");
+ return -EBUSY;
+ }
+ }
+
+ error = snd_miro_configure(miro);
+ if (error) {
+ snd_card_free(card);
+ return error;
+ }
+
+ error = snd_wss_create(card, miro->wss_base + 4, -1,
+ miro->irq, miro->dma1, miro->dma2,
+ WSS_HW_AD1845, 0, &codec);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ error = snd_wss_pcm(codec, 0, &pcm);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ error = snd_wss_mixer(codec);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ error = snd_wss_timer(codec, 0, &timer);
+ if (error < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ miro->pcm = pcm;
+
+ if ((error = snd_miro_mixer(miro)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if (miro->aci_vendor == 'm') {
+ /* It looks like a miro sound card. */
+ switch (miro->aci_product) {
+ case 'A':
+ sprintf(card->shortname,
+ "miroSOUND PCM1 pro / PCM12");
+ break;
+ case 'B':
+ sprintf(card->shortname,
+ "miroSOUND PCM12");
+ break;
+ case 'C':
+ sprintf(card->shortname,
+ "miroSOUND PCM20 radio");
+ break;
+ default:
+ sprintf(card->shortname,
+ "unknown miro");
+ snd_printk(KERN_INFO "unknown miro aci id\n");
+ break;
+ }
+ } else {
+ snd_printk(KERN_INFO "found unsupported aci card\n");
+ sprintf(card->shortname, "unknown Cardinal Technologies");
+ }
+
+ strcpy(card->driver, "miro");
+ sprintf(card->longname, "%s: OPTi%s, %s at 0x%lx, irq %d, dma %d&%d",
+ card->shortname, miro->name, pcm->name, miro->wss_base + 4,
+ miro->irq, miro->dma1, miro->dma2);
+
+ if (miro->mpu_port <= 0 || miro->mpu_port == SNDRV_AUTO_PORT)
+ rmidi = NULL;
+ else
+ if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
+ miro->mpu_port, 0, miro->mpu_irq, IRQF_DISABLED,
+ &rmidi)))
+ snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", miro->mpu_port);
+
+ if (miro->fm_port > 0 && miro->fm_port != SNDRV_AUTO_PORT) {
+ struct snd_opl3 *opl3 = NULL;
+ struct snd_opl4 *opl4;
+ if (snd_opl4_create(card, miro->fm_port, miro->fm_port - 8,
+ 2, &opl3, &opl4) < 0)
+ snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", miro->fm_port);
+ }
+
+ if ((error = snd_set_aci_init_values(miro)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ snd_card_set_dev(card, devptr);
+
+ if ((error = snd_card_register(card))) {
+ snd_card_free(card);
+ return error;
+ }
+
+ dev_set_drvdata(devptr, card);
+ return 0;
+}
+
+static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define DEV_NAME "miro"
+
+static struct isa_driver snd_miro_driver = {
+ .match = snd_miro_match,
+ .probe = snd_miro_probe,
+ .remove = __devexit_p(snd_miro_remove),
+ /* FIXME: suspend/resume */
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+static int __init alsa_card_miro_init(void)
+{
+ return isa_register_driver(&snd_miro_driver, 1);
+}
+
+static void __exit alsa_card_miro_exit(void)
+{
+ isa_unregister_driver(&snd_miro_driver);
+}
+
+module_init(alsa_card_miro_init)
+module_exit(alsa_card_miro_exit)
diff --git a/sound/isa/opti9xx/miro.h b/sound/isa/opti9xx/miro.h
new file mode 100644
index 0000000..6e1385b
--- /dev/null
+++ b/sound/isa/opti9xx/miro.h
@@ -0,0 +1,73 @@
+#ifndef _MIRO_H_
+#define _MIRO_H_
+
+#define ACI_REG_COMMAND 0 /* write register offset */
+#define ACI_REG_STATUS 1 /* read register offset */
+#define ACI_REG_BUSY 2 /* busy register offset */
+#define ACI_REG_RDS 2 /* PCM20: RDS register offset */
+#define ACI_MINTIME 500 /* ACI time out limit */
+
+#define ACI_SET_MUTE 0x0d
+#define ACI_SET_POWERAMP 0x0f
+#define ACI_SET_TUNERMUTE 0xa3
+#define ACI_SET_TUNERMONO 0xa4
+#define ACI_SET_IDE 0xd0
+#define ACI_SET_WSS 0xd1
+#define ACI_SET_SOLOMODE 0xd2
+#define ACI_SET_PREAMP 0x03
+#define ACI_GET_PREAMP 0x21
+#define ACI_WRITE_TUNE 0xa7
+#define ACI_READ_TUNERSTEREO 0xa8
+#define ACI_READ_TUNERSTATION 0xa9
+#define ACI_READ_VERSION 0xf1
+#define ACI_READ_IDCODE 0xf2
+#define ACI_INIT 0xff
+#define ACI_STATUS 0xf0
+#define ACI_S_GENERAL 0x00
+#define ACI_ERROR_OP 0xdf
+
+/* ACI Mixer */
+
+/* These are the values for the right channel GET registers.
+ Add an offset of 0x01 for the left channel register.
+ (left=right+0x01) */
+
+#define ACI_GET_MASTER 0x03
+#define ACI_GET_MIC 0x05
+#define ACI_GET_LINE 0x07
+#define ACI_GET_CD 0x09
+#define ACI_GET_SYNTH 0x0b
+#define ACI_GET_PCM 0x0d
+#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */
+#define ACI_GET_LINE2 0x12
+
+#define ACI_GET_EQ1 0x22 /* from Bass ... */
+#define ACI_GET_EQ2 0x24
+#define ACI_GET_EQ3 0x26
+#define ACI_GET_EQ4 0x28
+#define ACI_GET_EQ5 0x2a
+#define ACI_GET_EQ6 0x2c
+#define ACI_GET_EQ7 0x2e /* ... to Treble */
+
+/* And these are the values for the right channel SET registers.
+ For left channel access you have to add an offset of 0x08.
+ MASTER is an exception, which needs an offset of 0x01 */
+
+#define ACI_SET_MASTER 0x00
+#define ACI_SET_MIC 0x30
+#define ACI_SET_LINE 0x31
+#define ACI_SET_CD 0x34
+#define ACI_SET_SYNTH 0x33
+#define ACI_SET_PCM 0x32
+#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */
+#define ACI_SET_LINE2 0x36
+
+#define ACI_SET_EQ1 0x40 /* from Bass ... */
+#define ACI_SET_EQ2 0x41
+#define ACI_SET_EQ3 0x42
+#define ACI_SET_EQ4 0x43
+#define ACI_SET_EQ5 0x44
+#define ACI_SET_EQ6 0x45
+#define ACI_SET_EQ7 0x46 /* ... to Treble */
+
+#endif /* _MIRO_H_ */
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
new file mode 100644
index 0000000..19706b0
--- /dev/null
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -0,0 +1,1035 @@
+/*
+ card-opti92x-ad1848.c - driver for OPTi 82c92x based soundcards.
+ Copyright (C) 1998-2000 by Massimo Piccioni <dafastidio@libero.it>
+
+ Part of this code was developed at the Italian Ministry of Air Defence,
+ Sixth Division (oh, che pace ...), Rome.
+
+ Thanks to Maria Grazia Pollarini, Salvatore Vassallo.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <asm/io.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#ifndef OPTi93X
+#include <sound/opl4.h>
+#endif
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
+MODULE_LICENSE("GPL");
+#ifdef OPTi93X
+MODULE_DESCRIPTION("OPTi93X");
+MODULE_SUPPORTED_DEVICE("{{OPTi,82C931/3}}");
+#else /* OPTi93X */
+#ifdef CS4231
+MODULE_DESCRIPTION("OPTi92X - CS4231");
+MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (CS4231)},"
+ "{OPTi,82C925 (CS4231)}}");
+#else /* CS4231 */
+MODULE_DESCRIPTION("OPTi92X - AD1848");
+MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (AD1848)},"
+ "{OPTi,82C925 (AD1848)},"
+ "{OAK,Mozart}}");
+#endif /* CS4231 */
+#endif /* OPTi93X */
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+//static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */
+#ifdef CONFIG_PNP
+static int isapnp = 1; /* Enable ISA PnP detection */
+#endif
+static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */
+static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */
+static long fm_port = SNDRV_DEFAULT_PORT1; /* 0x388 */
+static int irq = SNDRV_DEFAULT_IRQ1; /* 5,7,9,10,11 */
+static int mpu_irq = SNDRV_DEFAULT_IRQ1; /* 5,7,9,10 */
+static int dma1 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */
+#if defined(CS4231) || defined(OPTi93X)
+static int dma2 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */
+#endif /* CS4231 || OPTi93X */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for opti9xx based soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for opti9xx based soundcard.");
+//module_param(enable, bool, 0444);
+//MODULE_PARM_DESC(enable, "Enable opti9xx soundcard.");
+#ifdef CONFIG_PNP
+module_param(isapnp, bool, 0444);
+MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard.");
+#endif
+module_param(port, long, 0444);
+MODULE_PARM_DESC(port, "WSS port # for opti9xx driver.");
+module_param(mpu_port, long, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for opti9xx driver.");
+module_param(fm_port, long, 0444);
+MODULE_PARM_DESC(fm_port, "FM port # for opti9xx driver.");
+module_param(irq, int, 0444);
+MODULE_PARM_DESC(irq, "WSS irq # for opti9xx driver.");
+module_param(mpu_irq, int, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 irq # for opti9xx driver.");
+module_param(dma1, int, 0444);
+MODULE_PARM_DESC(dma1, "1st dma # for opti9xx driver.");
+#if defined(CS4231) || defined(OPTi93X)
+module_param(dma2, int, 0444);
+MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver.");
+#endif /* CS4231 || OPTi93X */
+
+#define OPTi9XX_HW_82C928 1
+#define OPTi9XX_HW_82C929 2
+#define OPTi9XX_HW_82C924 3
+#define OPTi9XX_HW_82C925 4
+#define OPTi9XX_HW_82C930 5
+#define OPTi9XX_HW_82C931 6
+#define OPTi9XX_HW_82C933 7
+#define OPTi9XX_HW_LAST OPTi9XX_HW_82C933
+
+#define OPTi9XX_MC_REG(n) n
+
+#ifdef OPTi93X
+
+#define OPTi93X_STATUS 0x02
+#define OPTi93X_PORT(chip, r) ((chip)->port + OPTi93X_##r)
+
+#define OPTi93X_IRQ_PLAYBACK 0x04
+#define OPTi93X_IRQ_CAPTURE 0x08
+
+#endif /* OPTi93X */
+
+struct snd_opti9xx {
+ unsigned short hardware;
+ unsigned char password;
+ char name[7];
+
+ unsigned long mc_base;
+ struct resource *res_mc_base;
+ unsigned long mc_base_size;
+#ifdef OPTi93X
+ unsigned long mc_indir_index;
+ struct snd_wss *codec;
+#endif /* OPTi93X */
+ unsigned long pwd_reg;
+
+ spinlock_t lock;
+
+ long wss_base;
+ int irq;
+ int dma1;
+ int dma2;
+
+ long fm_port;
+
+ long mpu_port;
+ int mpu_irq;
+
+#ifdef CONFIG_PNP
+ struct pnp_dev *dev;
+ struct pnp_dev *devmpu;
+#endif /* CONFIG_PNP */
+};
+
+static int snd_opti9xx_pnp_is_probed;
+
+#ifdef CONFIG_PNP
+
+static struct pnp_card_device_id snd_opti9xx_pnpids[] = {
+#ifndef OPTi93X
+ /* OPTi 82C924 */
+ { .id = "OPT0924", .devs = { { "OPT0000" }, { "OPT0002" } }, .driver_data = 0x0924 },
+ /* OPTi 82C925 */
+ { .id = "OPT0925", .devs = { { "OPT9250" }, { "OPT0002" } }, .driver_data = 0x0925 },
+#else
+ /* OPTi 82C931/3 */
+ { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, .driver_data = 0x0931 },
+#endif /* OPTi93X */
+ { .id = "" }
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_opti9xx_pnpids);
+
+#endif /* CONFIG_PNP */
+
+#ifdef OPTi93X
+#define DEV_NAME "opti93x"
+#else
+#define DEV_NAME "opti92x"
+#endif
+
+static char * snd_opti9xx_names[] = {
+ "unkown",
+ "82C928", "82C929",
+ "82C924", "82C925",
+ "82C930", "82C931", "82C933"
+};
+
+
+static long __devinit snd_legacy_find_free_ioport(long *port_table, long size)
+{
+ while (*port_table != -1) {
+ if (request_region(*port_table, size, "ALSA test")) {
+ release_region(*port_table, size);
+ return *port_table;
+ }
+ port_table++;
+ }
+ return -1;
+}
+
+static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip,
+ unsigned short hardware)
+{
+ static int opti9xx_mc_size[] = {7, 7, 10, 10, 2, 2, 2};
+
+ chip->hardware = hardware;
+ strcpy(chip->name, snd_opti9xx_names[hardware]);
+
+ chip->mc_base_size = opti9xx_mc_size[hardware];
+
+ spin_lock_init(&chip->lock);
+
+ chip->wss_base = -1;
+ chip->irq = -1;
+ chip->dma1 = -1;
+ chip->dma2 = -1;
+ chip->fm_port = -1;
+ chip->mpu_port = -1;
+ chip->mpu_irq = -1;
+
+ switch (hardware) {
+#ifndef OPTi93X
+ case OPTi9XX_HW_82C928:
+ case OPTi9XX_HW_82C929:
+ chip->mc_base = 0xf8c;
+ chip->password = (hardware == OPTi9XX_HW_82C928) ? 0xe2 : 0xe3;
+ chip->pwd_reg = 3;
+ break;
+
+ case OPTi9XX_HW_82C924:
+ case OPTi9XX_HW_82C925:
+ chip->mc_base = 0xf8c;
+ chip->password = 0xe5;
+ chip->pwd_reg = 3;
+ break;
+#else /* OPTi93X */
+
+ case OPTi9XX_HW_82C930:
+ case OPTi9XX_HW_82C931:
+ case OPTi9XX_HW_82C933:
+ chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d;
+ chip->mc_indir_index = 0xe0e;
+ chip->password = 0xe4;
+ chip->pwd_reg = 0;
+ break;
+#endif /* OPTi93X */
+
+ default:
+ snd_printk("chip %d not supported\n", hardware);
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip,
+ unsigned char reg)
+{
+ unsigned long flags;
+ unsigned char retval = 0xff;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+
+ switch (chip->hardware) {
+#ifndef OPTi93X
+ case OPTi9XX_HW_82C924:
+ case OPTi9XX_HW_82C925:
+ if (reg > 7) {
+ outb(reg, chip->mc_base + 8);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+ retval = inb(chip->mc_base + 9);
+ break;
+ }
+
+ case OPTi9XX_HW_82C928:
+ case OPTi9XX_HW_82C929:
+ retval = inb(chip->mc_base + reg);
+ break;
+#else /* OPTi93X */
+
+ case OPTi9XX_HW_82C930:
+ case OPTi9XX_HW_82C931:
+ case OPTi9XX_HW_82C933:
+ outb(reg, chip->mc_indir_index);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+ retval = inb(chip->mc_indir_index + 1);
+ break;
+#endif /* OPTi93X */
+
+ default:
+ snd_printk("chip %d not supported\n", chip->hardware);
+ }
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return retval;
+}
+
+static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg,
+ unsigned char value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+
+ switch (chip->hardware) {
+#ifndef OPTi93X
+ case OPTi9XX_HW_82C924:
+ case OPTi9XX_HW_82C925:
+ if (reg > 7) {
+ outb(reg, chip->mc_base + 8);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+ outb(value, chip->mc_base + 9);
+ break;
+ }
+
+ case OPTi9XX_HW_82C928:
+ case OPTi9XX_HW_82C929:
+ outb(value, chip->mc_base + reg);
+ break;
+#else /* OPTi93X */
+
+ case OPTi9XX_HW_82C930:
+ case OPTi9XX_HW_82C931:
+ case OPTi9XX_HW_82C933:
+ outb(reg, chip->mc_indir_index);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+ outb(value, chip->mc_indir_index + 1);
+ break;
+#endif /* OPTi93X */
+
+ default:
+ snd_printk("chip %d not supported\n", chip->hardware);
+ }
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+#define snd_opti9xx_write_mask(chip, reg, value, mask) \
+ snd_opti9xx_write(chip, reg, \
+ (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
+
+
+static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip)
+{
+ unsigned char wss_base_bits;
+ unsigned char irq_bits;
+ unsigned char dma_bits;
+ unsigned char mpu_port_bits = 0;
+ unsigned char mpu_irq_bits;
+
+ switch (chip->hardware) {
+#ifndef OPTi93X
+ case OPTi9XX_HW_82C924:
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc);
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02);
+
+ case OPTi9XX_HW_82C925:
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80);
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x00, 0x20);
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff);
+#ifdef CS4231
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02);
+#else
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x00, 0x02);
+#endif /* CS4231 */
+ break;
+
+ case OPTi9XX_HW_82C928:
+ case OPTi9XX_HW_82C929:
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80);
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x00, 0x20);
+ /*
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xa2, 0xae);
+ */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x00, 0x0c);
+#ifdef CS4231
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02);
+#else
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x00, 0x02);
+#endif /* CS4231 */
+ break;
+
+#else /* OPTi93X */
+ case OPTi9XX_HW_82C931:
+ case OPTi9XX_HW_82C933:
+ /*
+ * The BTC 1817DW has QS1000 wavetable which is connected
+ * to the serial digital input of the OPTI931.
+ */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(21), 0x82, 0xff);
+ /*
+ * This bit sets OPTI931 to automaticaly select FM
+ * or digital input signal.
+ */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01);
+ case OPTi9XX_HW_82C930: /* FALL THROUGH */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03);
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff);
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 |
+ (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04),
+ 0x34);
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf);
+ break;
+#endif /* OPTi93X */
+
+ default:
+ snd_printk("chip %d not supported\n", chip->hardware);
+ return -EINVAL;
+ }
+
+ switch (chip->wss_base) {
+ case 0x530:
+ wss_base_bits = 0x00;
+ break;
+ case 0x604:
+ wss_base_bits = 0x03;
+ break;
+ case 0xe80:
+ wss_base_bits = 0x01;
+ break;
+ case 0xf40:
+ wss_base_bits = 0x02;
+ break;
+ default:
+ snd_printk("WSS port 0x%lx not valid\n", chip->wss_base);
+ goto __skip_base;
+ }
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30);
+
+__skip_base:
+ switch (chip->irq) {
+//#ifdef OPTi93X
+ case 5:
+ irq_bits = 0x05;
+ break;
+//#endif /* OPTi93X */
+ case 7:
+ irq_bits = 0x01;
+ break;
+ case 9:
+ irq_bits = 0x02;
+ break;
+ case 10:
+ irq_bits = 0x03;
+ break;
+ case 11:
+ irq_bits = 0x04;
+ break;
+ default:
+ snd_printk("WSS irq # %d not valid\n", chip->irq);
+ goto __skip_resources;
+ }
+
+ switch (chip->dma1) {
+ case 0:
+ dma_bits = 0x01;
+ break;
+ case 1:
+ dma_bits = 0x02;
+ break;
+ case 3:
+ dma_bits = 0x03;
+ break;
+ default:
+ snd_printk("WSS dma1 # %d not valid\n", chip->dma1);
+ goto __skip_resources;
+ }
+
+#if defined(CS4231) || defined(OPTi93X)
+ if (chip->dma1 == chip->dma2) {
+ snd_printk("don't want to share dmas\n");
+ return -EBUSY;
+ }
+
+ switch (chip->dma2) {
+ case 0:
+ case 1:
+ break;
+ default:
+ snd_printk("WSS dma2 # %d not valid\n", chip->dma2);
+ goto __skip_resources;
+ }
+ dma_bits |= 0x04;
+#endif /* CS4231 || OPTi93X */
+
+#ifndef OPTi93X
+ outb(irq_bits << 3 | dma_bits, chip->wss_base);
+#else /* OPTi93X */
+ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits));
+#endif /* OPTi93X */
+
+__skip_resources:
+ if (chip->hardware > OPTi9XX_HW_82C928) {
+ switch (chip->mpu_port) {
+ case 0:
+ case -1:
+ break;
+ case 0x300:
+ mpu_port_bits = 0x03;
+ break;
+ case 0x310:
+ mpu_port_bits = 0x02;
+ break;
+ case 0x320:
+ mpu_port_bits = 0x01;
+ break;
+ case 0x330:
+ mpu_port_bits = 0x00;
+ break;
+ default:
+ snd_printk("MPU-401 port 0x%lx not valid\n",
+ chip->mpu_port);
+ goto __skip_mpu;
+ }
+
+ switch (chip->mpu_irq) {
+ case 5:
+ mpu_irq_bits = 0x02;
+ break;
+ case 7:
+ mpu_irq_bits = 0x03;
+ break;
+ case 9:
+ mpu_irq_bits = 0x00;
+ break;
+ case 10:
+ mpu_irq_bits = 0x01;
+ break;
+ default:
+ snd_printk("MPU-401 irq # %d not valid\n",
+ chip->mpu_irq);
+ goto __skip_mpu;
+ }
+
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6),
+ (chip->mpu_port <= 0) ? 0x00 :
+ 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3,
+ 0xf8);
+ }
+__skip_mpu:
+
+ return 0;
+}
+
+#ifdef OPTi93X
+
+static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id)
+{
+ struct snd_wss *codec = dev_id;
+ struct snd_opti9xx *chip = codec->card->private_data;
+ unsigned char status;
+
+ status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11));
+ if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream)
+ snd_pcm_period_elapsed(codec->playback_substream);
+ if ((status & OPTi93X_IRQ_CAPTURE) && codec->capture_substream) {
+ snd_wss_overrange(codec);
+ snd_pcm_period_elapsed(codec->capture_substream);
+ }
+ outb(0x00, OPTi93X_PORT(codec, STATUS));
+ return IRQ_HANDLED;
+}
+
+#endif /* OPTi93X */
+
+static int __devinit snd_card_opti9xx_detect(struct snd_card *card,
+ struct snd_opti9xx *chip)
+{
+ int i, err;
+
+#ifndef OPTi93X
+ for (i = OPTi9XX_HW_82C928; i < OPTi9XX_HW_82C930; i++) {
+ unsigned char value;
+
+ if ((err = snd_opti9xx_init(chip, i)) < 0)
+ return err;
+
+ if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL)
+ continue;
+
+ value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(1));
+ if ((value != 0xff) && (value != inb(chip->mc_base + 1)))
+ if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)))
+ return 1;
+
+ release_and_free_resource(chip->res_mc_base);
+ chip->res_mc_base = NULL;
+
+ }
+#else /* OPTi93X */
+ for (i = OPTi9XX_HW_82C931; i >= OPTi9XX_HW_82C930; i--) {
+ unsigned long flags;
+ unsigned char value;
+
+ if ((err = snd_opti9xx_init(chip, i)) < 0)
+ return err;
+
+ if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL)
+ continue;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ outb(chip->password, chip->mc_base + chip->pwd_reg);
+ outb(((chip->mc_indir_index & (1 << 8)) >> 4) |
+ ((chip->mc_indir_index & 0xf0) >> 4), chip->mc_base);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ value = snd_opti9xx_read(chip, OPTi9XX_MC_REG(7));
+ snd_opti9xx_write(chip, OPTi9XX_MC_REG(7), 0xff - value);
+ if (snd_opti9xx_read(chip, OPTi9XX_MC_REG(7)) == 0xff - value)
+ return 1;
+
+ release_and_free_resource(chip->res_mc_base);
+ chip->res_mc_base = NULL;
+ }
+#endif /* OPTi93X */
+
+ return -ENODEV;
+}
+
+#ifdef CONFIG_PNP
+static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *pid)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL);
+ if (chip->dev == NULL)
+ return -EBUSY;
+
+ chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL);
+
+ pdev = chip->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err);
+ return err;
+ }
+
+#ifdef OPTi93X
+ port = pnp_port_start(pdev, 0) - 4;
+ fm_port = pnp_port_start(pdev, 1) + 8;
+#else
+ if (pid->driver_data != 0x0924)
+ port = pnp_port_start(pdev, 1);
+ fm_port = pnp_port_start(pdev, 2) + 8;
+#endif /* OPTi93X */
+ irq = pnp_irq(pdev, 0);
+ dma1 = pnp_dma(pdev, 0);
+#if defined(CS4231) || defined(OPTi93X)
+ dma2 = pnp_dma(pdev, 1);
+#endif /* CS4231 || OPTi93X */
+
+ pdev = chip->devmpu;
+ if (pdev && mpu_port > 0) {
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "AUDIO pnp configure failure\n");
+ mpu_port = -1;
+ chip->devmpu = NULL;
+ } else {
+ mpu_port = pnp_port_start(pdev, 0);
+ mpu_irq = pnp_irq(pdev, 0);
+ }
+ }
+ return pid->driver_data;
+}
+#endif /* CONFIG_PNP */
+
+static void snd_card_opti9xx_free(struct snd_card *card)
+{
+ struct snd_opti9xx *chip = card->private_data;
+
+ if (chip) {
+#ifdef OPTi93X
+ struct snd_wss *codec = chip->codec;
+ if (codec && codec->irq > 0) {
+ disable_irq(codec->irq);
+ free_irq(codec->irq, codec);
+ }
+#endif
+ release_and_free_resource(chip->res_mc_base);
+ }
+}
+
+static int __devinit snd_opti9xx_probe(struct snd_card *card)
+{
+ static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1};
+ int error;
+ struct snd_opti9xx *chip = card->private_data;
+ struct snd_wss *codec;
+#ifdef CS4231
+ struct snd_timer *timer;
+#endif
+ struct snd_pcm *pcm;
+ struct snd_rawmidi *rmidi;
+ struct snd_hwdep *synth;
+
+ if (! chip->res_mc_base &&
+ (chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size,
+ "OPTi9xx MC")) == NULL)
+ return -ENOMEM;
+
+ chip->wss_base = port;
+ chip->fm_port = fm_port;
+ chip->mpu_port = mpu_port;
+ chip->irq = irq;
+ chip->mpu_irq = mpu_irq;
+ chip->dma1 = dma1;
+#if defined(CS4231) || defined(OPTi93X)
+ chip->dma2 = dma2;
+#else
+ chip->dma2 = -1;
+#endif
+
+ if (chip->wss_base == SNDRV_AUTO_PORT) {
+ chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4);
+ if (chip->wss_base < 0) {
+ snd_printk("unable to find a free WSS port\n");
+ return -EBUSY;
+ }
+ }
+ error = snd_opti9xx_configure(chip);
+ if (error)
+ return error;
+
+ error = snd_wss_create(card, chip->wss_base + 4, -1,
+ chip->irq, chip->dma1, chip->dma2,
+#ifdef OPTi93X
+ WSS_HW_OPTI93X, WSS_HWSHARE_IRQ,
+#else
+ WSS_HW_DETECT, 0,
+#endif
+ &codec);
+ if (error < 0)
+ return error;
+#ifdef OPTi93X
+ chip->codec = codec;
+#endif
+ error = snd_wss_pcm(codec, 0, &pcm);
+ if (error < 0)
+ return error;
+ error = snd_wss_mixer(codec);
+ if (error < 0)
+ return error;
+#ifdef CS4231
+ error = snd_wss_timer(codec, 0, &timer);
+ if (error < 0)
+ return error;
+#endif
+#ifdef OPTi93X
+ error = request_irq(chip->irq, snd_opti93x_interrupt,
+ IRQF_DISABLED, DEV_NAME" - WSS", codec);
+ if (error < 0) {
+ snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq);
+ return error;
+ }
+#endif
+ strcpy(card->driver, chip->name);
+ sprintf(card->shortname, "OPTi %s", card->driver);
+#if defined(CS4231) || defined(OPTi93X)
+ sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d",
+ card->shortname, pcm->name, chip->wss_base + 4,
+ chip->irq, chip->dma1, chip->dma2);
+#else
+ sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d",
+ card->shortname, pcm->name, chip->wss_base + 4,
+ chip->irq, chip->dma1);
+#endif /* CS4231 || OPTi93X */
+
+ if (chip->mpu_port <= 0 || chip->mpu_port == SNDRV_AUTO_PORT)
+ rmidi = NULL;
+ else
+ if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
+ chip->mpu_port, 0, chip->mpu_irq, IRQF_DISABLED,
+ &rmidi)))
+ snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n",
+ chip->mpu_port);
+
+ if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) {
+ struct snd_opl3 *opl3 = NULL;
+#ifndef OPTi93X
+ if (chip->hardware == OPTi9XX_HW_82C928 ||
+ chip->hardware == OPTi9XX_HW_82C929 ||
+ chip->hardware == OPTi9XX_HW_82C924) {
+ struct snd_opl4 *opl4;
+ /* assume we have an OPL4 */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2),
+ 0x20, 0x20);
+ if (snd_opl4_create(card,
+ chip->fm_port,
+ chip->fm_port - 8,
+ 2, &opl3, &opl4) < 0) {
+ /* no luck, use OPL3 instead */
+ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2),
+ 0x00, 0x20);
+ }
+ }
+#endif /* !OPTi93X */
+ if (!opl3 && snd_opl3_create(card,
+ chip->fm_port,
+ chip->fm_port + 2,
+ OPL3_HW_AUTO, 0, &opl3) < 0) {
+ snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n",
+ chip->fm_port, chip->fm_port + 4 - 1);
+ }
+ if (opl3) {
+#ifdef CS4231
+ const int t1dev = 1;
+#else
+ const int t1dev = 0;
+#endif
+ if ((error = snd_opl3_timer_new(opl3, t1dev, t1dev+1)) < 0)
+ return error;
+ if ((error = snd_opl3_hwdep_new(opl3, 0, 1, &synth)) < 0)
+ return error;
+ }
+ }
+
+ return snd_card_register(card);
+}
+
+static struct snd_card *snd_opti9xx_card_new(void)
+{
+ struct snd_card *card;
+
+ card = snd_card_new(index, id, THIS_MODULE, sizeof(struct snd_opti9xx));
+ if (! card)
+ return NULL;
+ card->private_free = snd_card_opti9xx_free;
+ return card;
+}
+
+static int __devinit snd_opti9xx_isa_match(struct device *devptr,
+ unsigned int dev)
+{
+#ifdef CONFIG_PNP
+ if (snd_opti9xx_pnp_is_probed)
+ return 0;
+ if (isapnp)
+ return 0;
+#endif
+ return 1;
+}
+
+static int __devinit snd_opti9xx_isa_probe(struct device *devptr,
+ unsigned int dev)
+{
+ struct snd_card *card;
+ int error;
+ static long possible_mpu_ports[] = {0x300, 0x310, 0x320, 0x330, -1};
+#ifdef OPTi93X
+ static int possible_irqs[] = {5, 9, 10, 11, 7, -1};
+#else
+ static int possible_irqs[] = {9, 10, 11, 7, -1};
+#endif /* OPTi93X */
+ static int possible_mpu_irqs[] = {5, 9, 10, 7, -1};
+ static int possible_dma1s[] = {3, 1, 0, -1};
+#if defined(CS4231) || defined(OPTi93X)
+ static int possible_dma2s[][2] = {{1,-1}, {0,-1}, {-1,-1}, {0,-1}};
+#endif /* CS4231 || OPTi93X */
+
+ if (mpu_port == SNDRV_AUTO_PORT) {
+ if ((mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2)) < 0) {
+ snd_printk(KERN_ERR "unable to find a free MPU401 port\n");
+ return -EBUSY;
+ }
+ }
+ if (irq == SNDRV_AUTO_IRQ) {
+ if ((irq = snd_legacy_find_free_irq(possible_irqs)) < 0) {
+ snd_printk(KERN_ERR "unable to find a free IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (mpu_irq == SNDRV_AUTO_IRQ) {
+ if ((mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs)) < 0) {
+ snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (dma1 == SNDRV_AUTO_DMA) {
+ if ((dma1 = snd_legacy_find_free_dma(possible_dma1s)) < 0) {
+ snd_printk(KERN_ERR "unable to find a free DMA1\n");
+ return -EBUSY;
+ }
+ }
+#if defined(CS4231) || defined(OPTi93X)
+ if (dma2 == SNDRV_AUTO_DMA) {
+ if ((dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4])) < 0) {
+ snd_printk("unable to find a free DMA2\n");
+ return -EBUSY;
+ }
+ }
+#endif
+
+ card = snd_opti9xx_card_new();
+ if (! card)
+ return -ENOMEM;
+
+ if ((error = snd_card_opti9xx_detect(card, card->private_data)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ snd_card_set_dev(card, devptr);
+ if ((error = snd_opti9xx_probe(card)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ dev_set_drvdata(devptr, card);
+ return 0;
+}
+
+static int __devexit snd_opti9xx_isa_remove(struct device *devptr,
+ unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_opti9xx_driver = {
+ .match = snd_opti9xx_isa_match,
+ .probe = snd_opti9xx_isa_probe,
+ .remove = __devexit_p(snd_opti9xx_isa_remove),
+ /* FIXME: suspend/resume */
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+#ifdef CONFIG_PNP
+static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ struct snd_card *card;
+ int error, hw;
+ struct snd_opti9xx *chip;
+
+ if (snd_opti9xx_pnp_is_probed)
+ return -EBUSY;
+ if (! isapnp)
+ return -ENODEV;
+ card = snd_opti9xx_card_new();
+ if (! card)
+ return -ENOMEM;
+ chip = card->private_data;
+
+ hw = snd_card_opti9xx_pnp(chip, pcard, pid);
+ switch (hw) {
+ case 0x0924:
+ hw = OPTi9XX_HW_82C924;
+ break;
+ case 0x0925:
+ hw = OPTi9XX_HW_82C925;
+ break;
+ case 0x0931:
+ hw = OPTi9XX_HW_82C931;
+ break;
+ default:
+ snd_card_free(card);
+ return -ENODEV;
+ }
+
+ if ((error = snd_opti9xx_init(chip, hw))) {
+ snd_card_free(card);
+ return error;
+ }
+ if (hw <= OPTi9XX_HW_82C930)
+ chip->mc_base -= 0x80;
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((error = snd_opti9xx_probe(card)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ snd_opti9xx_pnp_is_probed = 1;
+ return 0;
+}
+
+static void __devexit snd_opti9xx_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+ snd_opti9xx_pnp_is_probed = 0;
+}
+
+static struct pnp_card_driver opti9xx_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "opti9xx",
+ .id_table = snd_opti9xx_pnpids,
+ .probe = snd_opti9xx_pnp_probe,
+ .remove = __devexit_p(snd_opti9xx_pnp_remove),
+};
+#endif
+
+#ifdef OPTi93X
+#define CHIP_NAME "82C93x"
+#else
+#define CHIP_NAME "82C92x"
+#endif
+
+static int __init alsa_card_opti9xx_init(void)
+{
+#ifdef CONFIG_PNP
+ pnp_register_card_driver(&opti9xx_pnpc_driver);
+ if (snd_opti9xx_pnp_is_probed)
+ return 0;
+ pnp_unregister_card_driver(&opti9xx_pnpc_driver);
+#endif
+ return isa_register_driver(&snd_opti9xx_driver, 1);
+}
+
+static void __exit alsa_card_opti9xx_exit(void)
+{
+ if (!snd_opti9xx_pnp_is_probed) {
+ isa_unregister_driver(&snd_opti9xx_driver);
+ return;
+ }
+#ifdef CONFIG_PNP
+ pnp_unregister_card_driver(&opti9xx_pnpc_driver);
+#endif
+}
+
+module_init(alsa_card_opti9xx_init)
+module_exit(alsa_card_opti9xx_exit)
diff --git a/sound/isa/opti9xx/opti92x-cs4231.c b/sound/isa/opti9xx/opti92x-cs4231.c
new file mode 100644
index 0000000..b17ab19
--- /dev/null
+++ b/sound/isa/opti9xx/opti92x-cs4231.c
@@ -0,0 +1,2 @@
+#define CS4231
+#include "opti92x-ad1848.c"
diff --git a/sound/isa/opti9xx/opti93x.c b/sound/isa/opti9xx/opti93x.c
new file mode 100644
index 0000000..bad9da5
--- /dev/null
+++ b/sound/isa/opti9xx/opti93x.c
@@ -0,0 +1,3 @@
+#define OPTi93X
+#include "opti92x-ad1848.c"
+
diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile
new file mode 100644
index 0000000..1098a56
--- /dev/null
+++ b/sound/isa/sb/Makefile
@@ -0,0 +1,36 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-sb-common-objs := sb_common.o sb_mixer.o
+snd-sb8-dsp-objs := sb8_main.o sb8_midi.o
+snd-sb16-dsp-objs := sb16_main.o
+snd-sb16-csp-objs := sb16_csp.o
+snd-sb8-objs := sb8.o
+snd-sb16-objs := sb16.o
+snd-sbawe-objs := sbawe.o emu8000.o
+snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o
+snd-es968-objs := es968.o
+
+#
+# this function returns:
+# "m" - CONFIG_SND_SEQUENCER is m
+# <empty string> - CONFIG_SND_SEQUENCER is undefined
+# otherwise parameter #1 value
+#
+sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o
+obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o
+obj-$(CONFIG_SND_SB8_DSP) += snd-sb8-dsp.o
+obj-$(CONFIG_SND_SB8) += snd-sb8.o
+obj-$(CONFIG_SND_SB16) += snd-sb16.o
+obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o
+obj-$(CONFIG_SND_ES968) += snd-es968.o
+ifeq ($(CONFIG_SND_SB16_CSP),y)
+ obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o
+ obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o
+endif
+obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
new file mode 100644
index 0000000..96678d5
--- /dev/null
+++ b/sound/isa/sb/emu8000.c
@@ -0,0 +1,1159 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * and (c) 1999 Steve Ratcliffe <steve@parabola.demon.co.uk>
+ * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Routines for control of EMU8000 chip
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/wait.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/emu8000.h>
+#include <sound/emu8000_reg.h>
+#include <asm/io.h>
+#include <asm/uaccess.h>
+#include <linux/init.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+
+/*
+ * emu8000 register controls
+ */
+
+/*
+ * The following routines read and write registers on the emu8000. They
+ * should always be called via the EMU8000*READ/WRITE macros and never
+ * directly. The macros handle the port number and command word.
+ */
+/* Write a word */
+void snd_emu8000_poke(struct snd_emu8000 *emu, unsigned int port, unsigned int reg, unsigned int val)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ if (reg != emu->last_reg) {
+ outw((unsigned short)reg, EMU8000_PTR(emu)); /* Set register */
+ emu->last_reg = reg;
+ }
+ outw((unsigned short)val, port); /* Send data */
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+}
+
+/* Read a word */
+unsigned short snd_emu8000_peek(struct snd_emu8000 *emu, unsigned int port, unsigned int reg)
+{
+ unsigned short res;
+ unsigned long flags;
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ if (reg != emu->last_reg) {
+ outw((unsigned short)reg, EMU8000_PTR(emu)); /* Set register */
+ emu->last_reg = reg;
+ }
+ res = inw(port); /* Read data */
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return res;
+}
+
+/* Write a double word */
+void snd_emu8000_poke_dw(struct snd_emu8000 *emu, unsigned int port, unsigned int reg, unsigned int val)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ if (reg != emu->last_reg) {
+ outw((unsigned short)reg, EMU8000_PTR(emu)); /* Set register */
+ emu->last_reg = reg;
+ }
+ outw((unsigned short)val, port); /* Send low word of data */
+ outw((unsigned short)(val>>16), port+2); /* Send high word of data */
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+}
+
+/* Read a double word */
+unsigned int snd_emu8000_peek_dw(struct snd_emu8000 *emu, unsigned int port, unsigned int reg)
+{
+ unsigned short low;
+ unsigned int res;
+ unsigned long flags;
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ if (reg != emu->last_reg) {
+ outw((unsigned short)reg, EMU8000_PTR(emu)); /* Set register */
+ emu->last_reg = reg;
+ }
+ low = inw(port); /* Read low word of data */
+ res = low + (inw(port+2) << 16);
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return res;
+}
+
+/*
+ * Set up / close a channel to be used for DMA.
+ */
+/*exported*/ void
+snd_emu8000_dma_chan(struct snd_emu8000 *emu, int ch, int mode)
+{
+ unsigned right_bit = (mode & EMU8000_RAM_RIGHT) ? 0x01000000 : 0;
+ mode &= EMU8000_RAM_MODE_MASK;
+ if (mode == EMU8000_RAM_CLOSE) {
+ EMU8000_CCCA_WRITE(emu, ch, 0);
+ EMU8000_DCYSUSV_WRITE(emu, ch, 0x807F);
+ return;
+ }
+ EMU8000_DCYSUSV_WRITE(emu, ch, 0x80);
+ EMU8000_VTFT_WRITE(emu, ch, 0);
+ EMU8000_CVCF_WRITE(emu, ch, 0);
+ EMU8000_PTRX_WRITE(emu, ch, 0x40000000);
+ EMU8000_CPF_WRITE(emu, ch, 0x40000000);
+ EMU8000_PSST_WRITE(emu, ch, 0);
+ EMU8000_CSL_WRITE(emu, ch, 0);
+ if (mode == EMU8000_RAM_WRITE) /* DMA write */
+ EMU8000_CCCA_WRITE(emu, ch, 0x06000000 | right_bit);
+ else /* DMA read */
+ EMU8000_CCCA_WRITE(emu, ch, 0x04000000 | right_bit);
+}
+
+/*
+ */
+static void __devinit
+snd_emu8000_read_wait(struct snd_emu8000 *emu)
+{
+ while ((EMU8000_SMALR_READ(emu) & 0x80000000) != 0) {
+ schedule_timeout_interruptible(1);
+ if (signal_pending(current))
+ break;
+ }
+}
+
+/*
+ */
+static void __devinit
+snd_emu8000_write_wait(struct snd_emu8000 *emu)
+{
+ while ((EMU8000_SMALW_READ(emu) & 0x80000000) != 0) {
+ schedule_timeout_interruptible(1);
+ if (signal_pending(current))
+ break;
+ }
+}
+
+/*
+ * detect a card at the given port
+ */
+static int __devinit
+snd_emu8000_detect(struct snd_emu8000 *emu)
+{
+ /* Initialise */
+ EMU8000_HWCF1_WRITE(emu, 0x0059);
+ EMU8000_HWCF2_WRITE(emu, 0x0020);
+ EMU8000_HWCF3_WRITE(emu, 0x0000);
+ /* Check for a recognisable emu8000 */
+ /*
+ if ((EMU8000_U1_READ(emu) & 0x000f) != 0x000c)
+ return -ENODEV;
+ */
+ if ((EMU8000_HWCF1_READ(emu) & 0x007e) != 0x0058)
+ return -ENODEV;
+ if ((EMU8000_HWCF2_READ(emu) & 0x0003) != 0x0003)
+ return -ENODEV;
+
+ snd_printdd("EMU8000 [0x%lx]: Synth chip found\n",
+ emu->port1);
+ return 0;
+}
+
+
+/*
+ * intiailize audio channels
+ */
+static void __devinit
+init_audio(struct snd_emu8000 *emu)
+{
+ int ch;
+
+ /* turn off envelope engines */
+ for (ch = 0; ch < EMU8000_CHANNELS; ch++)
+ EMU8000_DCYSUSV_WRITE(emu, ch, 0x80);
+
+ /* reset all other parameters to zero */
+ for (ch = 0; ch < EMU8000_CHANNELS; ch++) {
+ EMU8000_ENVVOL_WRITE(emu, ch, 0);
+ EMU8000_ENVVAL_WRITE(emu, ch, 0);
+ EMU8000_DCYSUS_WRITE(emu, ch, 0);
+ EMU8000_ATKHLDV_WRITE(emu, ch, 0);
+ EMU8000_LFO1VAL_WRITE(emu, ch, 0);
+ EMU8000_ATKHLD_WRITE(emu, ch, 0);
+ EMU8000_LFO2VAL_WRITE(emu, ch, 0);
+ EMU8000_IP_WRITE(emu, ch, 0);
+ EMU8000_IFATN_WRITE(emu, ch, 0);
+ EMU8000_PEFE_WRITE(emu, ch, 0);
+ EMU8000_FMMOD_WRITE(emu, ch, 0);
+ EMU8000_TREMFRQ_WRITE(emu, ch, 0);
+ EMU8000_FM2FRQ2_WRITE(emu, ch, 0);
+ EMU8000_PTRX_WRITE(emu, ch, 0);
+ EMU8000_VTFT_WRITE(emu, ch, 0);
+ EMU8000_PSST_WRITE(emu, ch, 0);
+ EMU8000_CSL_WRITE(emu, ch, 0);
+ EMU8000_CCCA_WRITE(emu, ch, 0);
+ }
+
+ for (ch = 0; ch < EMU8000_CHANNELS; ch++) {
+ EMU8000_CPF_WRITE(emu, ch, 0);
+ EMU8000_CVCF_WRITE(emu, ch, 0);
+ }
+}
+
+
+/*
+ * initialize DMA address
+ */
+static void __devinit
+init_dma(struct snd_emu8000 *emu)
+{
+ EMU8000_SMALR_WRITE(emu, 0);
+ EMU8000_SMARR_WRITE(emu, 0);
+ EMU8000_SMALW_WRITE(emu, 0);
+ EMU8000_SMARW_WRITE(emu, 0);
+}
+
+/*
+ * initialization arrays; from ADIP
+ */
+static unsigned short init1[128] /*__devinitdata*/ = {
+ 0x03ff, 0x0030, 0x07ff, 0x0130, 0x0bff, 0x0230, 0x0fff, 0x0330,
+ 0x13ff, 0x0430, 0x17ff, 0x0530, 0x1bff, 0x0630, 0x1fff, 0x0730,
+ 0x23ff, 0x0830, 0x27ff, 0x0930, 0x2bff, 0x0a30, 0x2fff, 0x0b30,
+ 0x33ff, 0x0c30, 0x37ff, 0x0d30, 0x3bff, 0x0e30, 0x3fff, 0x0f30,
+
+ 0x43ff, 0x0030, 0x47ff, 0x0130, 0x4bff, 0x0230, 0x4fff, 0x0330,
+ 0x53ff, 0x0430, 0x57ff, 0x0530, 0x5bff, 0x0630, 0x5fff, 0x0730,
+ 0x63ff, 0x0830, 0x67ff, 0x0930, 0x6bff, 0x0a30, 0x6fff, 0x0b30,
+ 0x73ff, 0x0c30, 0x77ff, 0x0d30, 0x7bff, 0x0e30, 0x7fff, 0x0f30,
+
+ 0x83ff, 0x0030, 0x87ff, 0x0130, 0x8bff, 0x0230, 0x8fff, 0x0330,
+ 0x93ff, 0x0430, 0x97ff, 0x0530, 0x9bff, 0x0630, 0x9fff, 0x0730,
+ 0xa3ff, 0x0830, 0xa7ff, 0x0930, 0xabff, 0x0a30, 0xafff, 0x0b30,
+ 0xb3ff, 0x0c30, 0xb7ff, 0x0d30, 0xbbff, 0x0e30, 0xbfff, 0x0f30,
+
+ 0xc3ff, 0x0030, 0xc7ff, 0x0130, 0xcbff, 0x0230, 0xcfff, 0x0330,
+ 0xd3ff, 0x0430, 0xd7ff, 0x0530, 0xdbff, 0x0630, 0xdfff, 0x0730,
+ 0xe3ff, 0x0830, 0xe7ff, 0x0930, 0xebff, 0x0a30, 0xefff, 0x0b30,
+ 0xf3ff, 0x0c30, 0xf7ff, 0x0d30, 0xfbff, 0x0e30, 0xffff, 0x0f30,
+};
+
+static unsigned short init2[128] /*__devinitdata*/ = {
+ 0x03ff, 0x8030, 0x07ff, 0x8130, 0x0bff, 0x8230, 0x0fff, 0x8330,
+ 0x13ff, 0x8430, 0x17ff, 0x8530, 0x1bff, 0x8630, 0x1fff, 0x8730,
+ 0x23ff, 0x8830, 0x27ff, 0x8930, 0x2bff, 0x8a30, 0x2fff, 0x8b30,
+ 0x33ff, 0x8c30, 0x37ff, 0x8d30, 0x3bff, 0x8e30, 0x3fff, 0x8f30,
+
+ 0x43ff, 0x8030, 0x47ff, 0x8130, 0x4bff, 0x8230, 0x4fff, 0x8330,
+ 0x53ff, 0x8430, 0x57ff, 0x8530, 0x5bff, 0x8630, 0x5fff, 0x8730,
+ 0x63ff, 0x8830, 0x67ff, 0x8930, 0x6bff, 0x8a30, 0x6fff, 0x8b30,
+ 0x73ff, 0x8c30, 0x77ff, 0x8d30, 0x7bff, 0x8e30, 0x7fff, 0x8f30,
+
+ 0x83ff, 0x8030, 0x87ff, 0x8130, 0x8bff, 0x8230, 0x8fff, 0x8330,
+ 0x93ff, 0x8430, 0x97ff, 0x8530, 0x9bff, 0x8630, 0x9fff, 0x8730,
+ 0xa3ff, 0x8830, 0xa7ff, 0x8930, 0xabff, 0x8a30, 0xafff, 0x8b30,
+ 0xb3ff, 0x8c30, 0xb7ff, 0x8d30, 0xbbff, 0x8e30, 0xbfff, 0x8f30,
+
+ 0xc3ff, 0x8030, 0xc7ff, 0x8130, 0xcbff, 0x8230, 0xcfff, 0x8330,
+ 0xd3ff, 0x8430, 0xd7ff, 0x8530, 0xdbff, 0x8630, 0xdfff, 0x8730,
+ 0xe3ff, 0x8830, 0xe7ff, 0x8930, 0xebff, 0x8a30, 0xefff, 0x8b30,
+ 0xf3ff, 0x8c30, 0xf7ff, 0x8d30, 0xfbff, 0x8e30, 0xffff, 0x8f30,
+};
+
+static unsigned short init3[128] /*__devinitdata*/ = {
+ 0x0C10, 0x8470, 0x14FE, 0xB488, 0x167F, 0xA470, 0x18E7, 0x84B5,
+ 0x1B6E, 0x842A, 0x1F1D, 0x852A, 0x0DA3, 0x8F7C, 0x167E, 0xF254,
+ 0x0000, 0x842A, 0x0001, 0x852A, 0x18E6, 0x8BAA, 0x1B6D, 0xF234,
+ 0x229F, 0x8429, 0x2746, 0x8529, 0x1F1C, 0x86E7, 0x229E, 0xF224,
+
+ 0x0DA4, 0x8429, 0x2C29, 0x8529, 0x2745, 0x87F6, 0x2C28, 0xF254,
+ 0x383B, 0x8428, 0x320F, 0x8528, 0x320E, 0x8F02, 0x1341, 0xF264,
+ 0x3EB6, 0x8428, 0x3EB9, 0x8528, 0x383A, 0x8FA9, 0x3EB5, 0xF294,
+ 0x3EB7, 0x8474, 0x3EBA, 0x8575, 0x3EB8, 0xC4C3, 0x3EBB, 0xC5C3,
+
+ 0x0000, 0xA404, 0x0001, 0xA504, 0x141F, 0x8671, 0x14FD, 0x8287,
+ 0x3EBC, 0xE610, 0x3EC8, 0x8C7B, 0x031A, 0x87E6, 0x3EC8, 0x86F7,
+ 0x3EC0, 0x821E, 0x3EBE, 0xD208, 0x3EBD, 0x821F, 0x3ECA, 0x8386,
+ 0x3EC1, 0x8C03, 0x3EC9, 0x831E, 0x3ECA, 0x8C4C, 0x3EBF, 0x8C55,
+
+ 0x3EC9, 0xC208, 0x3EC4, 0xBC84, 0x3EC8, 0x8EAD, 0x3EC8, 0xD308,
+ 0x3EC2, 0x8F7E, 0x3ECB, 0x8219, 0x3ECB, 0xD26E, 0x3EC5, 0x831F,
+ 0x3EC6, 0xC308, 0x3EC3, 0xB2FF, 0x3EC9, 0x8265, 0x3EC9, 0x8319,
+ 0x1342, 0xD36E, 0x3EC7, 0xB3FF, 0x0000, 0x8365, 0x1420, 0x9570,
+};
+
+static unsigned short init4[128] /*__devinitdata*/ = {
+ 0x0C10, 0x8470, 0x14FE, 0xB488, 0x167F, 0xA470, 0x18E7, 0x84B5,
+ 0x1B6E, 0x842A, 0x1F1D, 0x852A, 0x0DA3, 0x0F7C, 0x167E, 0x7254,
+ 0x0000, 0x842A, 0x0001, 0x852A, 0x18E6, 0x0BAA, 0x1B6D, 0x7234,
+ 0x229F, 0x8429, 0x2746, 0x8529, 0x1F1C, 0x06E7, 0x229E, 0x7224,
+
+ 0x0DA4, 0x8429, 0x2C29, 0x8529, 0x2745, 0x07F6, 0x2C28, 0x7254,
+ 0x383B, 0x8428, 0x320F, 0x8528, 0x320E, 0x0F02, 0x1341, 0x7264,
+ 0x3EB6, 0x8428, 0x3EB9, 0x8528, 0x383A, 0x0FA9, 0x3EB5, 0x7294,
+ 0x3EB7, 0x8474, 0x3EBA, 0x8575, 0x3EB8, 0x44C3, 0x3EBB, 0x45C3,
+
+ 0x0000, 0xA404, 0x0001, 0xA504, 0x141F, 0x0671, 0x14FD, 0x0287,
+ 0x3EBC, 0xE610, 0x3EC8, 0x0C7B, 0x031A, 0x07E6, 0x3EC8, 0x86F7,
+ 0x3EC0, 0x821E, 0x3EBE, 0xD208, 0x3EBD, 0x021F, 0x3ECA, 0x0386,
+ 0x3EC1, 0x0C03, 0x3EC9, 0x031E, 0x3ECA, 0x8C4C, 0x3EBF, 0x0C55,
+
+ 0x3EC9, 0xC208, 0x3EC4, 0xBC84, 0x3EC8, 0x0EAD, 0x3EC8, 0xD308,
+ 0x3EC2, 0x8F7E, 0x3ECB, 0x0219, 0x3ECB, 0xD26E, 0x3EC5, 0x031F,
+ 0x3EC6, 0xC308, 0x3EC3, 0x32FF, 0x3EC9, 0x0265, 0x3EC9, 0x8319,
+ 0x1342, 0xD36E, 0x3EC7, 0x33FF, 0x0000, 0x8365, 0x1420, 0x9570,
+};
+
+/* send an initialization array
+ * Taken from the oss driver, not obvious from the doc how this
+ * is meant to work
+ */
+static void __devinit
+send_array(struct snd_emu8000 *emu, unsigned short *data, int size)
+{
+ int i;
+ unsigned short *p;
+
+ p = data;
+ for (i = 0; i < size; i++, p++)
+ EMU8000_INIT1_WRITE(emu, i, *p);
+ for (i = 0; i < size; i++, p++)
+ EMU8000_INIT2_WRITE(emu, i, *p);
+ for (i = 0; i < size; i++, p++)
+ EMU8000_INIT3_WRITE(emu, i, *p);
+ for (i = 0; i < size; i++, p++)
+ EMU8000_INIT4_WRITE(emu, i, *p);
+}
+
+
+/*
+ * Send initialization arrays to start up, this just follows the
+ * initialisation sequence in the adip.
+ */
+static void __devinit
+init_arrays(struct snd_emu8000 *emu)
+{
+ send_array(emu, init1, ARRAY_SIZE(init1)/4);
+
+ msleep((1024 * 1000) / 44100); /* wait for 1024 clocks */
+ send_array(emu, init2, ARRAY_SIZE(init2)/4);
+ send_array(emu, init3, ARRAY_SIZE(init3)/4);
+
+ EMU8000_HWCF4_WRITE(emu, 0);
+ EMU8000_HWCF5_WRITE(emu, 0x83);
+ EMU8000_HWCF6_WRITE(emu, 0x8000);
+
+ send_array(emu, init4, ARRAY_SIZE(init4)/4);
+}
+
+
+#define UNIQUE_ID1 0xa5b9
+#define UNIQUE_ID2 0x9d53
+
+/*
+ * Size the onboard memory.
+ * This is written so as not to need arbitary delays after the write. It
+ * seems that the only way to do this is to use the one channel and keep
+ * reallocating between read and write.
+ */
+static void __devinit
+size_dram(struct snd_emu8000 *emu)
+{
+ int i, size;
+
+ if (emu->dram_checked)
+ return;
+
+ size = 0;
+
+ /* write out a magic number */
+ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE);
+ snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_READ);
+ EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET);
+ EMU8000_SMLD_WRITE(emu, UNIQUE_ID1);
+ snd_emu8000_init_fm(emu); /* This must really be here and not 2 lines back even */
+
+ while (size < EMU8000_MAX_DRAM) {
+
+ size += 512 * 1024; /* increment 512kbytes */
+
+ /* Write a unique data on the test address.
+ * if the address is out of range, the data is written on
+ * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is
+ * changed by this data.
+ */
+ /*snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE);*/
+ EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET + (size>>1));
+ EMU8000_SMLD_WRITE(emu, UNIQUE_ID2);
+ snd_emu8000_write_wait(emu);
+
+ /*
+ * read the data on the just written DRAM address
+ * if not the same then we have reached the end of ram.
+ */
+ /*snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_READ);*/
+ EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET + (size>>1));
+ /*snd_emu8000_read_wait(emu);*/
+ EMU8000_SMLD_READ(emu); /* discard stale data */
+ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2)
+ break; /* we must have wrapped around */
+
+ snd_emu8000_read_wait(emu);
+
+ /*
+ * If it is the same it could be that the address just
+ * wraps back to the beginning; so check to see if the
+ * initial value has been overwritten.
+ */
+ EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET);
+ EMU8000_SMLD_READ(emu); /* discard stale data */
+ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1)
+ break; /* we must have wrapped around */
+ snd_emu8000_read_wait(emu);
+ }
+
+ /* wait until FULL bit in SMAxW register is false */
+ for (i = 0; i < 10000; i++) {
+ if ((EMU8000_SMALW_READ(emu) & 0x80000000) == 0)
+ break;
+ schedule_timeout_interruptible(1);
+ if (signal_pending(current))
+ break;
+ }
+ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_CLOSE);
+ snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE);
+
+ snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n",
+ emu->port1, size/1024);
+
+ emu->mem_size = size;
+ emu->dram_checked = 1;
+}
+
+
+/*
+ * Initiailise the FM section. You have to do this to use sample RAM
+ * and therefore lose 2 voices.
+ */
+/*exported*/ void
+snd_emu8000_init_fm(struct snd_emu8000 *emu)
+{
+ unsigned long flags;
+
+ /* Initialize the last two channels for DRAM refresh and producing
+ the reverb and chorus effects for Yamaha OPL-3 synthesizer */
+
+ /* 31: FM left channel, 0xffffe0-0xffffe8 */
+ EMU8000_DCYSUSV_WRITE(emu, 30, 0x80);
+ EMU8000_PSST_WRITE(emu, 30, 0xFFFFFFE0); /* full left */
+ EMU8000_CSL_WRITE(emu, 30, 0x00FFFFE8 | (emu->fm_chorus_depth << 24));
+ EMU8000_PTRX_WRITE(emu, 30, (emu->fm_reverb_depth << 8));
+ EMU8000_CPF_WRITE(emu, 30, 0);
+ EMU8000_CCCA_WRITE(emu, 30, 0x00FFFFE3);
+
+ /* 32: FM right channel, 0xfffff0-0xfffff8 */
+ EMU8000_DCYSUSV_WRITE(emu, 31, 0x80);
+ EMU8000_PSST_WRITE(emu, 31, 0x00FFFFF0); /* full right */
+ EMU8000_CSL_WRITE(emu, 31, 0x00FFFFF8 | (emu->fm_chorus_depth << 24));
+ EMU8000_PTRX_WRITE(emu, 31, (emu->fm_reverb_depth << 8));
+ EMU8000_CPF_WRITE(emu, 31, 0x8000);
+ EMU8000_CCCA_WRITE(emu, 31, 0x00FFFFF3);
+
+ snd_emu8000_poke((emu), EMU8000_DATA0(emu), EMU8000_CMD(1, (30)), 0);
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ while (!(inw(EMU8000_PTR(emu)) & 0x1000))
+ ;
+ while ((inw(EMU8000_PTR(emu)) & 0x1000))
+ ;
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ snd_emu8000_poke((emu), EMU8000_DATA0(emu), EMU8000_CMD(1, (30)), 0x4828);
+ /* this is really odd part.. */
+ outb(0x3C, EMU8000_PTR(emu));
+ outb(0, EMU8000_DATA1(emu));
+
+ /* skew volume & cutoff */
+ EMU8000_VTFT_WRITE(emu, 30, 0x8000FFFF);
+ EMU8000_VTFT_WRITE(emu, 31, 0x8000FFFF);
+}
+
+
+/*
+ * The main initialization routine.
+ */
+static void __devinit
+snd_emu8000_init_hw(struct snd_emu8000 *emu)
+{
+ int i;
+
+ emu->last_reg = 0xffff; /* reset the last register index */
+
+ /* initialize hardware configuration */
+ EMU8000_HWCF1_WRITE(emu, 0x0059);
+ EMU8000_HWCF2_WRITE(emu, 0x0020);
+
+ /* disable audio; this seems to reduce a clicking noise a bit.. */
+ EMU8000_HWCF3_WRITE(emu, 0);
+
+ /* initialize audio channels */
+ init_audio(emu);
+
+ /* initialize DMA */
+ init_dma(emu);
+
+ /* initialize init arrays */
+ init_arrays(emu);
+
+ /*
+ * Initialize the FM section of the AWE32, this is needed
+ * for DRAM refresh as well
+ */
+ snd_emu8000_init_fm(emu);
+
+ /* terminate all voices */
+ for (i = 0; i < EMU8000_DRAM_VOICES; i++)
+ EMU8000_DCYSUSV_WRITE(emu, 0, 0x807F);
+
+ /* check DRAM memory size */
+ size_dram(emu);
+
+ /* enable audio */
+ EMU8000_HWCF3_WRITE(emu, 0x4);
+
+ /* set equzlier, chorus and reverb modes */
+ snd_emu8000_update_equalizer(emu);
+ snd_emu8000_update_chorus_mode(emu);
+ snd_emu8000_update_reverb_mode(emu);
+}
+
+
+/*----------------------------------------------------------------
+ * Bass/Treble Equalizer
+ *----------------------------------------------------------------*/
+
+static unsigned short bass_parm[12][3] = {
+ {0xD26A, 0xD36A, 0x0000}, /* -12 dB */
+ {0xD25B, 0xD35B, 0x0000}, /* -8 */
+ {0xD24C, 0xD34C, 0x0000}, /* -6 */
+ {0xD23D, 0xD33D, 0x0000}, /* -4 */
+ {0xD21F, 0xD31F, 0x0000}, /* -2 */
+ {0xC208, 0xC308, 0x0001}, /* 0 (HW default) */
+ {0xC219, 0xC319, 0x0001}, /* +2 */
+ {0xC22A, 0xC32A, 0x0001}, /* +4 */
+ {0xC24C, 0xC34C, 0x0001}, /* +6 */
+ {0xC26E, 0xC36E, 0x0001}, /* +8 */
+ {0xC248, 0xC384, 0x0002}, /* +10 */
+ {0xC26A, 0xC36A, 0x0002}, /* +12 dB */
+};
+
+static unsigned short treble_parm[12][9] = {
+ {0x821E, 0xC26A, 0x031E, 0xC36A, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001}, /* -12 dB */
+ {0x821E, 0xC25B, 0x031E, 0xC35B, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001},
+ {0x821E, 0xC24C, 0x031E, 0xC34C, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001},
+ {0x821E, 0xC23D, 0x031E, 0xC33D, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001},
+ {0x821E, 0xC21F, 0x031E, 0xC31F, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001},
+ {0x821E, 0xD208, 0x031E, 0xD308, 0x021E, 0xD208, 0x831E, 0xD308, 0x0002},
+ {0x821E, 0xD208, 0x031E, 0xD308, 0x021D, 0xD219, 0x831D, 0xD319, 0x0002},
+ {0x821E, 0xD208, 0x031E, 0xD308, 0x021C, 0xD22A, 0x831C, 0xD32A, 0x0002},
+ {0x821E, 0xD208, 0x031E, 0xD308, 0x021A, 0xD24C, 0x831A, 0xD34C, 0x0002},
+ {0x821E, 0xD208, 0x031E, 0xD308, 0x0219, 0xD26E, 0x8319, 0xD36E, 0x0002}, /* +8 (HW default) */
+ {0x821D, 0xD219, 0x031D, 0xD319, 0x0219, 0xD26E, 0x8319, 0xD36E, 0x0002},
+ {0x821C, 0xD22A, 0x031C, 0xD32A, 0x0219, 0xD26E, 0x8319, 0xD36E, 0x0002} /* +12 dB */
+};
+
+
+/*
+ * set Emu8000 digital equalizer; from 0 to 11 [-12dB - 12dB]
+ */
+/*exported*/ void
+snd_emu8000_update_equalizer(struct snd_emu8000 *emu)
+{
+ unsigned short w;
+ int bass = emu->bass_level;
+ int treble = emu->treble_level;
+
+ if (bass < 0 || bass > 11 || treble < 0 || treble > 11)
+ return;
+ EMU8000_INIT4_WRITE(emu, 0x01, bass_parm[bass][0]);
+ EMU8000_INIT4_WRITE(emu, 0x11, bass_parm[bass][1]);
+ EMU8000_INIT3_WRITE(emu, 0x11, treble_parm[treble][0]);
+ EMU8000_INIT3_WRITE(emu, 0x13, treble_parm[treble][1]);
+ EMU8000_INIT3_WRITE(emu, 0x1b, treble_parm[treble][2]);
+ EMU8000_INIT4_WRITE(emu, 0x07, treble_parm[treble][3]);
+ EMU8000_INIT4_WRITE(emu, 0x0b, treble_parm[treble][4]);
+ EMU8000_INIT4_WRITE(emu, 0x0d, treble_parm[treble][5]);
+ EMU8000_INIT4_WRITE(emu, 0x17, treble_parm[treble][6]);
+ EMU8000_INIT4_WRITE(emu, 0x19, treble_parm[treble][7]);
+ w = bass_parm[bass][2] + treble_parm[treble][8];
+ EMU8000_INIT4_WRITE(emu, 0x15, (unsigned short)(w + 0x0262));
+ EMU8000_INIT4_WRITE(emu, 0x1d, (unsigned short)(w + 0x8362));
+}
+
+
+/*----------------------------------------------------------------
+ * Chorus mode control
+ *----------------------------------------------------------------*/
+
+/*
+ * chorus mode parameters
+ */
+#define SNDRV_EMU8000_CHORUS_1 0
+#define SNDRV_EMU8000_CHORUS_2 1
+#define SNDRV_EMU8000_CHORUS_3 2
+#define SNDRV_EMU8000_CHORUS_4 3
+#define SNDRV_EMU8000_CHORUS_FEEDBACK 4
+#define SNDRV_EMU8000_CHORUS_FLANGER 5
+#define SNDRV_EMU8000_CHORUS_SHORTDELAY 6
+#define SNDRV_EMU8000_CHORUS_SHORTDELAY2 7
+#define SNDRV_EMU8000_CHORUS_PREDEFINED 8
+/* user can define chorus modes up to 32 */
+#define SNDRV_EMU8000_CHORUS_NUMBERS 32
+
+struct soundfont_chorus_fx {
+ unsigned short feedback; /* feedback level (0xE600-0xE6FF) */
+ unsigned short delay_offset; /* delay (0-0x0DA3) [1/44100 sec] */
+ unsigned short lfo_depth; /* LFO depth (0xBC00-0xBCFF) */
+ unsigned int delay; /* right delay (0-0xFFFFFFFF) [1/256/44100 sec] */
+ unsigned int lfo_freq; /* LFO freq LFO freq (0-0xFFFFFFFF) */
+};
+
+/* 5 parameters for each chorus mode; 3 x 16bit, 2 x 32bit */
+static char chorus_defined[SNDRV_EMU8000_CHORUS_NUMBERS];
+static struct soundfont_chorus_fx chorus_parm[SNDRV_EMU8000_CHORUS_NUMBERS] = {
+ {0xE600, 0x03F6, 0xBC2C ,0x00000000, 0x0000006D}, /* chorus 1 */
+ {0xE608, 0x031A, 0xBC6E, 0x00000000, 0x0000017C}, /* chorus 2 */
+ {0xE610, 0x031A, 0xBC84, 0x00000000, 0x00000083}, /* chorus 3 */
+ {0xE620, 0x0269, 0xBC6E, 0x00000000, 0x0000017C}, /* chorus 4 */
+ {0xE680, 0x04D3, 0xBCA6, 0x00000000, 0x0000005B}, /* feedback */
+ {0xE6E0, 0x044E, 0xBC37, 0x00000000, 0x00000026}, /* flanger */
+ {0xE600, 0x0B06, 0xBC00, 0x0006E000, 0x00000083}, /* short delay */
+ {0xE6C0, 0x0B06, 0xBC00, 0x0006E000, 0x00000083}, /* short delay + feedback */
+};
+
+/*exported*/ int
+snd_emu8000_load_chorus_fx(struct snd_emu8000 *emu, int mode, const void __user *buf, long len)
+{
+ struct soundfont_chorus_fx rec;
+ if (mode < SNDRV_EMU8000_CHORUS_PREDEFINED || mode >= SNDRV_EMU8000_CHORUS_NUMBERS) {
+ snd_printk(KERN_WARNING "invalid chorus mode %d for uploading\n", mode);
+ return -EINVAL;
+ }
+ if (len < (long)sizeof(rec) || copy_from_user(&rec, buf, sizeof(rec)))
+ return -EFAULT;
+ chorus_parm[mode] = rec;
+ chorus_defined[mode] = 1;
+ return 0;
+}
+
+/*exported*/ void
+snd_emu8000_update_chorus_mode(struct snd_emu8000 *emu)
+{
+ int effect = emu->chorus_mode;
+ if (effect < 0 || effect >= SNDRV_EMU8000_CHORUS_NUMBERS ||
+ (effect >= SNDRV_EMU8000_CHORUS_PREDEFINED && !chorus_defined[effect]))
+ return;
+ EMU8000_INIT3_WRITE(emu, 0x09, chorus_parm[effect].feedback);
+ EMU8000_INIT3_WRITE(emu, 0x0c, chorus_parm[effect].delay_offset);
+ EMU8000_INIT4_WRITE(emu, 0x03, chorus_parm[effect].lfo_depth);
+ EMU8000_HWCF4_WRITE(emu, chorus_parm[effect].delay);
+ EMU8000_HWCF5_WRITE(emu, chorus_parm[effect].lfo_freq);
+ EMU8000_HWCF6_WRITE(emu, 0x8000);
+ EMU8000_HWCF7_WRITE(emu, 0x0000);
+}
+
+/*----------------------------------------------------------------
+ * Reverb mode control
+ *----------------------------------------------------------------*/
+
+/*
+ * reverb mode parameters
+ */
+#define SNDRV_EMU8000_REVERB_ROOM1 0
+#define SNDRV_EMU8000_REVERB_ROOM2 1
+#define SNDRV_EMU8000_REVERB_ROOM3 2
+#define SNDRV_EMU8000_REVERB_HALL1 3
+#define SNDRV_EMU8000_REVERB_HALL2 4
+#define SNDRV_EMU8000_REVERB_PLATE 5
+#define SNDRV_EMU8000_REVERB_DELAY 6
+#define SNDRV_EMU8000_REVERB_PANNINGDELAY 7
+#define SNDRV_EMU8000_REVERB_PREDEFINED 8
+/* user can define reverb modes up to 32 */
+#define SNDRV_EMU8000_REVERB_NUMBERS 32
+
+struct soundfont_reverb_fx {
+ unsigned short parms[28];
+};
+
+/* reverb mode settings; write the following 28 data of 16 bit length
+ * on the corresponding ports in the reverb_cmds array
+ */
+static char reverb_defined[SNDRV_EMU8000_CHORUS_NUMBERS];
+static struct soundfont_reverb_fx reverb_parm[SNDRV_EMU8000_REVERB_NUMBERS] = {
+{{ /* room 1 */
+ 0xB488, 0xA450, 0x9550, 0x84B5, 0x383A, 0x3EB5, 0x72F4,
+ 0x72A4, 0x7254, 0x7204, 0x7204, 0x7204, 0x4416, 0x4516,
+ 0xA490, 0xA590, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429,
+ 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528,
+}},
+{{ /* room 2 */
+ 0xB488, 0xA458, 0x9558, 0x84B5, 0x383A, 0x3EB5, 0x7284,
+ 0x7254, 0x7224, 0x7224, 0x7254, 0x7284, 0x4448, 0x4548,
+ 0xA440, 0xA540, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429,
+ 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528,
+}},
+{{ /* room 3 */
+ 0xB488, 0xA460, 0x9560, 0x84B5, 0x383A, 0x3EB5, 0x7284,
+ 0x7254, 0x7224, 0x7224, 0x7254, 0x7284, 0x4416, 0x4516,
+ 0xA490, 0xA590, 0x842C, 0x852C, 0x842C, 0x852C, 0x842B,
+ 0x852B, 0x842B, 0x852B, 0x842A, 0x852A, 0x842A, 0x852A,
+}},
+{{ /* hall 1 */
+ 0xB488, 0xA470, 0x9570, 0x84B5, 0x383A, 0x3EB5, 0x7284,
+ 0x7254, 0x7224, 0x7224, 0x7254, 0x7284, 0x4448, 0x4548,
+ 0xA440, 0xA540, 0x842B, 0x852B, 0x842B, 0x852B, 0x842A,
+ 0x852A, 0x842A, 0x852A, 0x8429, 0x8529, 0x8429, 0x8529,
+}},
+{{ /* hall 2 */
+ 0xB488, 0xA470, 0x9570, 0x84B5, 0x383A, 0x3EB5, 0x7254,
+ 0x7234, 0x7224, 0x7254, 0x7264, 0x7294, 0x44C3, 0x45C3,
+ 0xA404, 0xA504, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429,
+ 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528,
+}},
+{{ /* plate */
+ 0xB4FF, 0xA470, 0x9570, 0x84B5, 0x383A, 0x3EB5, 0x7234,
+ 0x7234, 0x7234, 0x7234, 0x7234, 0x7234, 0x4448, 0x4548,
+ 0xA440, 0xA540, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429,
+ 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528,
+}},
+{{ /* delay */
+ 0xB4FF, 0xA470, 0x9500, 0x84B5, 0x333A, 0x39B5, 0x7204,
+ 0x7204, 0x7204, 0x7204, 0x7204, 0x72F4, 0x4400, 0x4500,
+ 0xA4FF, 0xA5FF, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420,
+ 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520,
+}},
+{{ /* panning delay */
+ 0xB4FF, 0xA490, 0x9590, 0x8474, 0x333A, 0x39B5, 0x7204,
+ 0x7204, 0x7204, 0x7204, 0x7204, 0x72F4, 0x4400, 0x4500,
+ 0xA4FF, 0xA5FF, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420,
+ 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520,
+}},
+};
+
+enum { DATA1, DATA2 };
+#define AWE_INIT1(c) EMU8000_CMD(2,c), DATA1
+#define AWE_INIT2(c) EMU8000_CMD(2,c), DATA2
+#define AWE_INIT3(c) EMU8000_CMD(3,c), DATA1
+#define AWE_INIT4(c) EMU8000_CMD(3,c), DATA2
+
+static struct reverb_cmd_pair {
+ unsigned short cmd, port;
+} reverb_cmds[28] = {
+ {AWE_INIT1(0x03)}, {AWE_INIT1(0x05)}, {AWE_INIT4(0x1F)}, {AWE_INIT1(0x07)},
+ {AWE_INIT2(0x14)}, {AWE_INIT2(0x16)}, {AWE_INIT1(0x0F)}, {AWE_INIT1(0x17)},
+ {AWE_INIT1(0x1F)}, {AWE_INIT2(0x07)}, {AWE_INIT2(0x0F)}, {AWE_INIT2(0x17)},
+ {AWE_INIT2(0x1D)}, {AWE_INIT2(0x1F)}, {AWE_INIT3(0x01)}, {AWE_INIT3(0x03)},
+ {AWE_INIT1(0x09)}, {AWE_INIT1(0x0B)}, {AWE_INIT1(0x11)}, {AWE_INIT1(0x13)},
+ {AWE_INIT1(0x19)}, {AWE_INIT1(0x1B)}, {AWE_INIT2(0x01)}, {AWE_INIT2(0x03)},
+ {AWE_INIT2(0x09)}, {AWE_INIT2(0x0B)}, {AWE_INIT2(0x11)}, {AWE_INIT2(0x13)},
+};
+
+/*exported*/ int
+snd_emu8000_load_reverb_fx(struct snd_emu8000 *emu, int mode, const void __user *buf, long len)
+{
+ struct soundfont_reverb_fx rec;
+
+ if (mode < SNDRV_EMU8000_REVERB_PREDEFINED || mode >= SNDRV_EMU8000_REVERB_NUMBERS) {
+ snd_printk(KERN_WARNING "invalid reverb mode %d for uploading\n", mode);
+ return -EINVAL;
+ }
+ if (len < (long)sizeof(rec) || copy_from_user(&rec, buf, sizeof(rec)))
+ return -EFAULT;
+ reverb_parm[mode] = rec;
+ reverb_defined[mode] = 1;
+ return 0;
+}
+
+/*exported*/ void
+snd_emu8000_update_reverb_mode(struct snd_emu8000 *emu)
+{
+ int effect = emu->reverb_mode;
+ int i;
+
+ if (effect < 0 || effect >= SNDRV_EMU8000_REVERB_NUMBERS ||
+ (effect >= SNDRV_EMU8000_REVERB_PREDEFINED && !reverb_defined[effect]))
+ return;
+ for (i = 0; i < 28; i++) {
+ int port;
+ if (reverb_cmds[i].port == DATA1)
+ port = EMU8000_DATA1(emu);
+ else
+ port = EMU8000_DATA2(emu);
+ snd_emu8000_poke(emu, port, reverb_cmds[i].cmd, reverb_parm[effect].parms[i]);
+ }
+}
+
+
+/*----------------------------------------------------------------
+ * mixer interface
+ *----------------------------------------------------------------*/
+
+/*
+ * bass/treble
+ */
+static int mixer_bass_treble_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 11;
+ return 0;
+}
+
+static int mixer_bass_treble_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu8000 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = kcontrol->private_value ? emu->treble_level : emu->bass_level;
+ return 0;
+}
+
+static int mixer_bass_treble_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu8000 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned short val1;
+
+ val1 = ucontrol->value.integer.value[0] % 12;
+ spin_lock_irqsave(&emu->control_lock, flags);
+ if (kcontrol->private_value) {
+ change = val1 != emu->treble_level;
+ emu->treble_level = val1;
+ } else {
+ change = val1 != emu->bass_level;
+ emu->bass_level = val1;
+ }
+ spin_unlock_irqrestore(&emu->control_lock, flags);
+ snd_emu8000_update_equalizer(emu);
+ return change;
+}
+
+static struct snd_kcontrol_new mixer_bass_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Synth Tone Control - Bass",
+ .info = mixer_bass_treble_info,
+ .get = mixer_bass_treble_get,
+ .put = mixer_bass_treble_put,
+ .private_value = 0,
+};
+
+static struct snd_kcontrol_new mixer_treble_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Synth Tone Control - Treble",
+ .info = mixer_bass_treble_info,
+ .get = mixer_bass_treble_get,
+ .put = mixer_bass_treble_put,
+ .private_value = 1,
+};
+
+/*
+ * chorus/reverb mode
+ */
+static int mixer_chorus_reverb_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = kcontrol->private_value ? (SNDRV_EMU8000_CHORUS_NUMBERS-1) : (SNDRV_EMU8000_REVERB_NUMBERS-1);
+ return 0;
+}
+
+static int mixer_chorus_reverb_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu8000 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = kcontrol->private_value ? emu->chorus_mode : emu->reverb_mode;
+ return 0;
+}
+
+static int mixer_chorus_reverb_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu8000 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned short val1;
+
+ spin_lock_irqsave(&emu->control_lock, flags);
+ if (kcontrol->private_value) {
+ val1 = ucontrol->value.integer.value[0] % SNDRV_EMU8000_CHORUS_NUMBERS;
+ change = val1 != emu->chorus_mode;
+ emu->chorus_mode = val1;
+ } else {
+ val1 = ucontrol->value.integer.value[0] % SNDRV_EMU8000_REVERB_NUMBERS;
+ change = val1 != emu->reverb_mode;
+ emu->reverb_mode = val1;
+ }
+ spin_unlock_irqrestore(&emu->control_lock, flags);
+ if (change) {
+ if (kcontrol->private_value)
+ snd_emu8000_update_chorus_mode(emu);
+ else
+ snd_emu8000_update_reverb_mode(emu);
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new mixer_chorus_mode_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Chorus Mode",
+ .info = mixer_chorus_reverb_info,
+ .get = mixer_chorus_reverb_get,
+ .put = mixer_chorus_reverb_put,
+ .private_value = 1,
+};
+
+static struct snd_kcontrol_new mixer_reverb_mode_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Reverb Mode",
+ .info = mixer_chorus_reverb_info,
+ .get = mixer_chorus_reverb_get,
+ .put = mixer_chorus_reverb_put,
+ .private_value = 0,
+};
+
+/*
+ * FM OPL3 chorus/reverb depth
+ */
+static int mixer_fm_depth_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int mixer_fm_depth_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu8000 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = kcontrol->private_value ? emu->fm_chorus_depth : emu->fm_reverb_depth;
+ return 0;
+}
+
+static int mixer_fm_depth_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu8000 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned short val1;
+
+ val1 = ucontrol->value.integer.value[0] % 256;
+ spin_lock_irqsave(&emu->control_lock, flags);
+ if (kcontrol->private_value) {
+ change = val1 != emu->fm_chorus_depth;
+ emu->fm_chorus_depth = val1;
+ } else {
+ change = val1 != emu->fm_reverb_depth;
+ emu->fm_reverb_depth = val1;
+ }
+ spin_unlock_irqrestore(&emu->control_lock, flags);
+ if (change)
+ snd_emu8000_init_fm(emu);
+ return change;
+}
+
+static struct snd_kcontrol_new mixer_fm_chorus_depth_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "FM Chorus Depth",
+ .info = mixer_fm_depth_info,
+ .get = mixer_fm_depth_get,
+ .put = mixer_fm_depth_put,
+ .private_value = 1,
+};
+
+static struct snd_kcontrol_new mixer_fm_reverb_depth_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "FM Reverb Depth",
+ .info = mixer_fm_depth_info,
+ .get = mixer_fm_depth_get,
+ .put = mixer_fm_depth_put,
+ .private_value = 0,
+};
+
+
+static struct snd_kcontrol_new *mixer_defs[EMU8000_NUM_CONTROLS] = {
+ &mixer_bass_control,
+ &mixer_treble_control,
+ &mixer_chorus_mode_control,
+ &mixer_reverb_mode_control,
+ &mixer_fm_chorus_depth_control,
+ &mixer_fm_reverb_depth_control,
+};
+
+/*
+ * create and attach mixer elements for WaveTable treble/bass controls
+ */
+static int __devinit
+snd_emu8000_create_mixer(struct snd_card *card, struct snd_emu8000 *emu)
+{
+ int i, err = 0;
+
+ if (snd_BUG_ON(!emu || !card))
+ return -EINVAL;
+
+ spin_lock_init(&emu->control_lock);
+
+ memset(emu->controls, 0, sizeof(emu->controls));
+ for (i = 0; i < EMU8000_NUM_CONTROLS; i++) {
+ if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0)
+ goto __error;
+ }
+ return 0;
+
+__error:
+ for (i = 0; i < EMU8000_NUM_CONTROLS; i++) {
+ down_write(&card->controls_rwsem);
+ if (emu->controls[i])
+ snd_ctl_remove(card, emu->controls[i]);
+ up_write(&card->controls_rwsem);
+ }
+ return err;
+}
+
+
+/*
+ * free resources
+ */
+static int snd_emu8000_free(struct snd_emu8000 *hw)
+{
+ release_and_free_resource(hw->res_port1);
+ release_and_free_resource(hw->res_port2);
+ release_and_free_resource(hw->res_port3);
+ kfree(hw);
+ return 0;
+}
+
+/*
+ */
+static int snd_emu8000_dev_free(struct snd_device *device)
+{
+ struct snd_emu8000 *hw = device->device_data;
+ return snd_emu8000_free(hw);
+}
+
+/*
+ * initialize and register emu8000 synth device.
+ */
+int __devinit
+snd_emu8000_new(struct snd_card *card, int index, long port, int seq_ports,
+ struct snd_seq_device **awe_ret)
+{
+ struct snd_seq_device *awe;
+ struct snd_emu8000 *hw;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_emu8000_dev_free,
+ };
+
+ if (awe_ret)
+ *awe_ret = NULL;
+
+ if (seq_ports <= 0)
+ return 0;
+
+ hw = kzalloc(sizeof(*hw), GFP_KERNEL);
+ if (hw == NULL)
+ return -ENOMEM;
+ spin_lock_init(&hw->reg_lock);
+ hw->index = index;
+ hw->port1 = port;
+ hw->port2 = port + 0x400;
+ hw->port3 = port + 0x800;
+ if (!(hw->res_port1 = request_region(hw->port1, 4, "Emu8000-1")) ||
+ !(hw->res_port2 = request_region(hw->port2, 4, "Emu8000-2")) ||
+ !(hw->res_port3 = request_region(hw->port3, 4, "Emu8000-3"))) {
+ snd_printk(KERN_ERR "sbawe: can't grab ports 0x%lx, 0x%lx, 0x%lx\n", hw->port1, hw->port2, hw->port3);
+ snd_emu8000_free(hw);
+ return -EBUSY;
+ }
+ hw->mem_size = 0;
+ hw->card = card;
+ hw->seq_ports = seq_ports;
+ hw->bass_level = 5;
+ hw->treble_level = 9;
+ hw->chorus_mode = 2;
+ hw->reverb_mode = 4;
+ hw->fm_chorus_depth = 0;
+ hw->fm_reverb_depth = 0;
+
+ if (snd_emu8000_detect(hw) < 0) {
+ snd_emu8000_free(hw);
+ return -ENODEV;
+ }
+
+ snd_emu8000_init_hw(hw);
+ if ((err = snd_emu8000_create_mixer(card, hw)) < 0) {
+ snd_emu8000_free(hw);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_CODEC, hw, &ops)) < 0) {
+ snd_emu8000_free(hw);
+ return err;
+ }
+#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))
+ if (snd_seq_device_new(card, index, SNDRV_SEQ_DEV_ID_EMU8000,
+ sizeof(struct snd_emu8000*), &awe) >= 0) {
+ strcpy(awe->name, "EMU-8000");
+ *(struct snd_emu8000 **)SNDRV_SEQ_DEVICE_ARGPTR(awe) = hw;
+ }
+#else
+ awe = NULL;
+#endif
+ if (awe_ret)
+ *awe_ret = awe;
+
+ return 0;
+}
+
+
+/*
+ * exported stuff
+ */
+
+EXPORT_SYMBOL(snd_emu8000_poke);
+EXPORT_SYMBOL(snd_emu8000_peek);
+EXPORT_SYMBOL(snd_emu8000_poke_dw);
+EXPORT_SYMBOL(snd_emu8000_peek_dw);
+EXPORT_SYMBOL(snd_emu8000_dma_chan);
+EXPORT_SYMBOL(snd_emu8000_init_fm);
+EXPORT_SYMBOL(snd_emu8000_load_chorus_fx);
+EXPORT_SYMBOL(snd_emu8000_load_reverb_fx);
+EXPORT_SYMBOL(snd_emu8000_update_chorus_mode);
+EXPORT_SYMBOL(snd_emu8000_update_reverb_mode);
+EXPORT_SYMBOL(snd_emu8000_update_equalizer);
diff --git a/sound/isa/sb/emu8000_callback.c b/sound/isa/sb/emu8000_callback.c
new file mode 100644
index 0000000..9a3c71c
--- /dev/null
+++ b/sound/isa/sb/emu8000_callback.c
@@ -0,0 +1,546 @@
+/*
+ * synth callback routines for the emu8000 (AWE32/64)
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "emu8000_local.h"
+#include <sound/asoundef.h>
+
+/*
+ * prototypes
+ */
+static struct snd_emux_voice *get_voice(struct snd_emux *emu,
+ struct snd_emux_port *port);
+static int start_voice(struct snd_emux_voice *vp);
+static void trigger_voice(struct snd_emux_voice *vp);
+static void release_voice(struct snd_emux_voice *vp);
+static void update_voice(struct snd_emux_voice *vp, int update);
+static void reset_voice(struct snd_emux *emu, int ch);
+static void terminate_voice(struct snd_emux_voice *vp);
+static void sysex(struct snd_emux *emu, char *buf, int len, int parsed,
+ struct snd_midi_channel_set *chset);
+#ifdef CONFIG_SND_SEQUENCER_OSS
+static int oss_ioctl(struct snd_emux *emu, int cmd, int p1, int p2);
+#endif
+static int load_fx(struct snd_emux *emu, int type, int mode,
+ const void __user *buf, long len);
+
+static void set_pitch(struct snd_emu8000 *hw, struct snd_emux_voice *vp);
+static void set_volume(struct snd_emu8000 *hw, struct snd_emux_voice *vp);
+static void set_pan(struct snd_emu8000 *hw, struct snd_emux_voice *vp);
+static void set_fmmod(struct snd_emu8000 *hw, struct snd_emux_voice *vp);
+static void set_tremfreq(struct snd_emu8000 *hw, struct snd_emux_voice *vp);
+static void set_fm2frq2(struct snd_emu8000 *hw, struct snd_emux_voice *vp);
+static void set_filterQ(struct snd_emu8000 *hw, struct snd_emux_voice *vp);
+static void snd_emu8000_tweak_voice(struct snd_emu8000 *emu, int ch);
+
+/*
+ * Ensure a value is between two points
+ * macro evaluates its args more than once, so changed to upper-case.
+ */
+#define LIMITVALUE(x, a, b) do { if ((x) < (a)) (x) = (a); else if ((x) > (b)) (x) = (b); } while (0)
+#define LIMITMAX(x, a) do {if ((x) > (a)) (x) = (a); } while (0)
+
+
+/*
+ * set up operators
+ */
+static struct snd_emux_operators emu8000_ops = {
+ .owner = THIS_MODULE,
+ .get_voice = get_voice,
+ .prepare = start_voice,
+ .trigger = trigger_voice,
+ .release = release_voice,
+ .update = update_voice,
+ .terminate = terminate_voice,
+ .reset = reset_voice,
+ .sample_new = snd_emu8000_sample_new,
+ .sample_free = snd_emu8000_sample_free,
+ .sample_reset = snd_emu8000_sample_reset,
+ .load_fx = load_fx,
+ .sysex = sysex,
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ .oss_ioctl = oss_ioctl,
+#endif
+};
+
+void
+snd_emu8000_ops_setup(struct snd_emu8000 *hw)
+{
+ hw->emu->ops = emu8000_ops;
+}
+
+
+
+/*
+ * Terminate a voice
+ */
+static void
+release_voice(struct snd_emux_voice *vp)
+{
+ int dcysusv;
+ struct snd_emu8000 *hw;
+
+ hw = vp->hw;
+ dcysusv = 0x8000 | (unsigned char)vp->reg.parm.modrelease;
+ EMU8000_DCYSUS_WRITE(hw, vp->ch, dcysusv);
+ dcysusv = 0x8000 | (unsigned char)vp->reg.parm.volrelease;
+ EMU8000_DCYSUSV_WRITE(hw, vp->ch, dcysusv);
+}
+
+
+/*
+ */
+static void
+terminate_voice(struct snd_emux_voice *vp)
+{
+ struct snd_emu8000 *hw;
+
+ hw = vp->hw;
+ EMU8000_DCYSUSV_WRITE(hw, vp->ch, 0x807F);
+}
+
+
+/*
+ */
+static void
+update_voice(struct snd_emux_voice *vp, int update)
+{
+ struct snd_emu8000 *hw;
+
+ hw = vp->hw;
+ if (update & SNDRV_EMUX_UPDATE_VOLUME)
+ set_volume(hw, vp);
+ if (update & SNDRV_EMUX_UPDATE_PITCH)
+ set_pitch(hw, vp);
+ if ((update & SNDRV_EMUX_UPDATE_PAN) &&
+ vp->port->ctrls[EMUX_MD_REALTIME_PAN])
+ set_pan(hw, vp);
+ if (update & SNDRV_EMUX_UPDATE_FMMOD)
+ set_fmmod(hw, vp);
+ if (update & SNDRV_EMUX_UPDATE_TREMFREQ)
+ set_tremfreq(hw, vp);
+ if (update & SNDRV_EMUX_UPDATE_FM2FRQ2)
+ set_fm2frq2(hw, vp);
+ if (update & SNDRV_EMUX_UPDATE_Q)
+ set_filterQ(hw, vp);
+}
+
+
+/*
+ * Find a channel (voice) within the EMU that is not in use or at least
+ * less in use than other channels. Always returns a valid pointer
+ * no matter what. If there is a real shortage of voices then one
+ * will be cut. Such is life.
+ *
+ * The channel index (vp->ch) must be initialized in this routine.
+ * In Emu8k, it is identical with the array index.
+ */
+static struct snd_emux_voice *
+get_voice(struct snd_emux *emu, struct snd_emux_port *port)
+{
+ int i;
+ struct snd_emux_voice *vp;
+ struct snd_emu8000 *hw;
+
+ /* what we are looking for, in order of preference */
+ enum {
+ OFF=0, RELEASED, PLAYING, END
+ };
+
+ /* Keeps track of what we are finding */
+ struct best {
+ unsigned int time;
+ int voice;
+ } best[END];
+ struct best *bp;
+
+ hw = emu->hw;
+
+ for (i = 0; i < END; i++) {
+ best[i].time = (unsigned int)(-1); /* XXX MAX_?INT really */;
+ best[i].voice = -1;
+ }
+
+ /*
+ * Go through them all and get a best one to use.
+ */
+ for (i = 0; i < emu->max_voices; i++) {
+ int state, val;
+
+ vp = &emu->voices[i];
+ state = vp->state;
+
+ if (state == SNDRV_EMUX_ST_OFF)
+ bp = best + OFF;
+ else if (state == SNDRV_EMUX_ST_RELEASED ||
+ state == SNDRV_EMUX_ST_PENDING) {
+ bp = best + RELEASED;
+ val = (EMU8000_CVCF_READ(hw, vp->ch) >> 16) & 0xffff;
+ if (! val)
+ bp = best + OFF;
+ }
+ else if (state & SNDRV_EMUX_ST_ON)
+ bp = best + PLAYING;
+ else
+ continue;
+
+ /* check if sample is finished playing (non-looping only) */
+ if (state != SNDRV_EMUX_ST_OFF &&
+ (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) {
+ val = EMU8000_CCCA_READ(hw, vp->ch) & 0xffffff;
+ if (val >= vp->reg.loopstart)
+ bp = best + OFF;
+ }
+
+ if (vp->time < bp->time) {
+ bp->time = vp->time;
+ bp->voice = i;
+ }
+ }
+
+ for (i = 0; i < END; i++) {
+ if (best[i].voice >= 0) {
+ vp = &emu->voices[best[i].voice];
+ vp->ch = best[i].voice;
+ return vp;
+ }
+ }
+
+ /* not found */
+ return NULL;
+}
+
+/*
+ */
+static int
+start_voice(struct snd_emux_voice *vp)
+{
+ unsigned int temp;
+ int ch;
+ int addr;
+ struct snd_midi_channel *chan;
+ struct snd_emu8000 *hw;
+
+ hw = vp->hw;
+ ch = vp->ch;
+ chan = vp->chan;
+
+ /* channel to be silent and idle */
+ EMU8000_DCYSUSV_WRITE(hw, ch, 0x0080);
+ EMU8000_VTFT_WRITE(hw, ch, 0x0000FFFF);
+ EMU8000_CVCF_WRITE(hw, ch, 0x0000FFFF);
+ EMU8000_PTRX_WRITE(hw, ch, 0);
+ EMU8000_CPF_WRITE(hw, ch, 0);
+
+ /* set pitch offset */
+ set_pitch(hw, vp);
+
+ /* set envelope parameters */
+ EMU8000_ENVVAL_WRITE(hw, ch, vp->reg.parm.moddelay);
+ EMU8000_ATKHLD_WRITE(hw, ch, vp->reg.parm.modatkhld);
+ EMU8000_DCYSUS_WRITE(hw, ch, vp->reg.parm.moddcysus);
+ EMU8000_ENVVOL_WRITE(hw, ch, vp->reg.parm.voldelay);
+ EMU8000_ATKHLDV_WRITE(hw, ch, vp->reg.parm.volatkhld);
+ /* decay/sustain parameter for volume envelope is used
+ for triggerg the voice */
+
+ /* cutoff and volume */
+ set_volume(hw, vp);
+
+ /* modulation envelope heights */
+ EMU8000_PEFE_WRITE(hw, ch, vp->reg.parm.pefe);
+
+ /* lfo1/2 delay */
+ EMU8000_LFO1VAL_WRITE(hw, ch, vp->reg.parm.lfo1delay);
+ EMU8000_LFO2VAL_WRITE(hw, ch, vp->reg.parm.lfo2delay);
+
+ /* lfo1 pitch & cutoff shift */
+ set_fmmod(hw, vp);
+ /* lfo1 volume & freq */
+ set_tremfreq(hw, vp);
+ /* lfo2 pitch & freq */
+ set_fm2frq2(hw, vp);
+ /* pan & loop start */
+ set_pan(hw, vp);
+
+ /* chorus & loop end (chorus 8bit, MSB) */
+ addr = vp->reg.loopend - 1;
+ temp = vp->reg.parm.chorus;
+ temp += (int)chan->control[MIDI_CTL_E3_CHORUS_DEPTH] * 9 / 10;
+ LIMITMAX(temp, 255);
+ temp = (temp <<24) | (unsigned int)addr;
+ EMU8000_CSL_WRITE(hw, ch, temp);
+
+ /* Q & current address (Q 4bit value, MSB) */
+ addr = vp->reg.start - 1;
+ temp = vp->reg.parm.filterQ;
+ temp = (temp<<28) | (unsigned int)addr;
+ EMU8000_CCCA_WRITE(hw, ch, temp);
+
+ /* clear unknown registers */
+ EMU8000_00A0_WRITE(hw, ch, 0);
+ EMU8000_0080_WRITE(hw, ch, 0);
+
+ /* reset volume */
+ temp = vp->vtarget << 16;
+ EMU8000_VTFT_WRITE(hw, ch, temp | vp->ftarget);
+ EMU8000_CVCF_WRITE(hw, ch, temp | 0xff00);
+
+ return 0;
+}
+
+/*
+ * Start envelope
+ */
+static void
+trigger_voice(struct snd_emux_voice *vp)
+{
+ int ch = vp->ch;
+ unsigned int temp;
+ struct snd_emu8000 *hw;
+
+ hw = vp->hw;
+
+ /* set reverb and pitch target */
+ temp = vp->reg.parm.reverb;
+ temp += (int)vp->chan->control[MIDI_CTL_E1_REVERB_DEPTH] * 9 / 10;
+ LIMITMAX(temp, 255);
+ temp = (temp << 8) | (vp->ptarget << 16) | vp->aaux;
+ EMU8000_PTRX_WRITE(hw, ch, temp);
+ EMU8000_CPF_WRITE(hw, ch, vp->ptarget << 16);
+ EMU8000_DCYSUSV_WRITE(hw, ch, vp->reg.parm.voldcysus);
+}
+
+/*
+ * reset voice parameters
+ */
+static void
+reset_voice(struct snd_emux *emu, int ch)
+{
+ struct snd_emu8000 *hw;
+
+ hw = emu->hw;
+ EMU8000_DCYSUSV_WRITE(hw, ch, 0x807F);
+ snd_emu8000_tweak_voice(hw, ch);
+}
+
+/*
+ * Set the pitch of a possibly playing note.
+ */
+static void
+set_pitch(struct snd_emu8000 *hw, struct snd_emux_voice *vp)
+{
+ EMU8000_IP_WRITE(hw, vp->ch, vp->apitch);
+}
+
+/*
+ * Set the volume of a possibly already playing note
+ */
+static void
+set_volume(struct snd_emu8000 *hw, struct snd_emux_voice *vp)
+{
+ int ifatn;
+
+ ifatn = (unsigned char)vp->acutoff;
+ ifatn = (ifatn << 8);
+ ifatn |= (unsigned char)vp->avol;
+ EMU8000_IFATN_WRITE(hw, vp->ch, ifatn);
+}
+
+/*
+ * Set pan and loop start address.
+ */
+static void
+set_pan(struct snd_emu8000 *hw, struct snd_emux_voice *vp)
+{
+ unsigned int temp;
+
+ temp = ((unsigned int)vp->apan<<24) | ((unsigned int)vp->reg.loopstart - 1);
+ EMU8000_PSST_WRITE(hw, vp->ch, temp);
+}
+
+#define MOD_SENSE 18
+
+static void
+set_fmmod(struct snd_emu8000 *hw, struct snd_emux_voice *vp)
+{
+ unsigned short fmmod;
+ short pitch;
+ unsigned char cutoff;
+ int modulation;
+
+ pitch = (char)(vp->reg.parm.fmmod>>8);
+ cutoff = (vp->reg.parm.fmmod & 0xff);
+ modulation = vp->chan->gm_modulation + vp->chan->midi_pressure;
+ pitch += (MOD_SENSE * modulation) / 1200;
+ LIMITVALUE(pitch, -128, 127);
+ fmmod = ((unsigned char)pitch<<8) | cutoff;
+ EMU8000_FMMOD_WRITE(hw, vp->ch, fmmod);
+}
+
+/* set tremolo (lfo1) volume & frequency */
+static void
+set_tremfreq(struct snd_emu8000 *hw, struct snd_emux_voice *vp)
+{
+ EMU8000_TREMFRQ_WRITE(hw, vp->ch, vp->reg.parm.tremfrq);
+}
+
+/* set lfo2 pitch & frequency */
+static void
+set_fm2frq2(struct snd_emu8000 *hw, struct snd_emux_voice *vp)
+{
+ unsigned short fm2frq2;
+ short pitch;
+ unsigned char freq;
+ int modulation;
+
+ pitch = (char)(vp->reg.parm.fm2frq2>>8);
+ freq = vp->reg.parm.fm2frq2 & 0xff;
+ modulation = vp->chan->gm_modulation + vp->chan->midi_pressure;
+ pitch += (MOD_SENSE * modulation) / 1200;
+ LIMITVALUE(pitch, -128, 127);
+ fm2frq2 = ((unsigned char)pitch<<8) | freq;
+ EMU8000_FM2FRQ2_WRITE(hw, vp->ch, fm2frq2);
+}
+
+/* set filterQ */
+static void
+set_filterQ(struct snd_emu8000 *hw, struct snd_emux_voice *vp)
+{
+ unsigned int addr;
+ addr = EMU8000_CCCA_READ(hw, vp->ch) & 0xffffff;
+ addr |= (vp->reg.parm.filterQ << 28);
+ EMU8000_CCCA_WRITE(hw, vp->ch, addr);
+}
+
+/*
+ * set the envelope & LFO parameters to the default values
+ */
+static void
+snd_emu8000_tweak_voice(struct snd_emu8000 *emu, int i)
+{
+ /* set all mod/vol envelope shape to minimum */
+ EMU8000_ENVVOL_WRITE(emu, i, 0x8000);
+ EMU8000_ENVVAL_WRITE(emu, i, 0x8000);
+ EMU8000_DCYSUS_WRITE(emu, i, 0x7F7F);
+ EMU8000_ATKHLDV_WRITE(emu, i, 0x7F7F);
+ EMU8000_ATKHLD_WRITE(emu, i, 0x7F7F);
+ EMU8000_PEFE_WRITE(emu, i, 0); /* mod envelope height to zero */
+ EMU8000_LFO1VAL_WRITE(emu, i, 0x8000); /* no delay for LFO1 */
+ EMU8000_LFO2VAL_WRITE(emu, i, 0x8000);
+ EMU8000_IP_WRITE(emu, i, 0xE000); /* no pitch shift */
+ EMU8000_IFATN_WRITE(emu, i, 0xFF00); /* volume to minimum */
+ EMU8000_FMMOD_WRITE(emu, i, 0);
+ EMU8000_TREMFRQ_WRITE(emu, i, 0);
+ EMU8000_FM2FRQ2_WRITE(emu, i, 0);
+}
+
+/*
+ * sysex callback
+ */
+static void
+sysex(struct snd_emux *emu, char *buf, int len, int parsed, struct snd_midi_channel_set *chset)
+{
+ struct snd_emu8000 *hw;
+
+ hw = emu->hw;
+
+ switch (parsed) {
+ case SNDRV_MIDI_SYSEX_GS_CHORUS_MODE:
+ hw->chorus_mode = chset->gs_chorus_mode;
+ snd_emu8000_update_chorus_mode(hw);
+ break;
+
+ case SNDRV_MIDI_SYSEX_GS_REVERB_MODE:
+ hw->reverb_mode = chset->gs_reverb_mode;
+ snd_emu8000_update_reverb_mode(hw);
+ break;
+ }
+}
+
+
+#ifdef CONFIG_SND_SEQUENCER_OSS
+/*
+ * OSS ioctl callback
+ */
+static int
+oss_ioctl(struct snd_emux *emu, int cmd, int p1, int p2)
+{
+ struct snd_emu8000 *hw;
+
+ hw = emu->hw;
+
+ switch (cmd) {
+ case _EMUX_OSS_REVERB_MODE:
+ hw->reverb_mode = p1;
+ snd_emu8000_update_reverb_mode(hw);
+ break;
+
+ case _EMUX_OSS_CHORUS_MODE:
+ hw->chorus_mode = p1;
+ snd_emu8000_update_chorus_mode(hw);
+ break;
+
+ case _EMUX_OSS_INITIALIZE_CHIP:
+ /* snd_emu8000_init(hw); */ /*ignored*/
+ break;
+
+ case _EMUX_OSS_EQUALIZER:
+ hw->bass_level = p1;
+ hw->treble_level = p2;
+ snd_emu8000_update_equalizer(hw);
+ break;
+ }
+ return 0;
+}
+#endif
+
+
+/*
+ * additional patch keys
+ */
+
+#define SNDRV_EMU8000_LOAD_CHORUS_FX 0x10 /* optarg=mode */
+#define SNDRV_EMU8000_LOAD_REVERB_FX 0x11 /* optarg=mode */
+
+
+/*
+ * callback routine
+ */
+
+static int
+load_fx(struct snd_emux *emu, int type, int mode, const void __user *buf, long len)
+{
+ struct snd_emu8000 *hw;
+ hw = emu->hw;
+
+ /* skip header */
+ buf += 16;
+ len -= 16;
+
+ switch (type) {
+ case SNDRV_EMU8000_LOAD_CHORUS_FX:
+ return snd_emu8000_load_chorus_fx(hw, mode, buf, len);
+ case SNDRV_EMU8000_LOAD_REVERB_FX:
+ return snd_emu8000_load_reverb_fx(hw, mode, buf, len);
+ }
+ return -EINVAL;
+}
+
diff --git a/sound/isa/sb/emu8000_local.h b/sound/isa/sb/emu8000_local.h
new file mode 100644
index 0000000..7e87c34
--- /dev/null
+++ b/sound/isa/sb/emu8000_local.h
@@ -0,0 +1,45 @@
+#ifndef __EMU8000_LOCAL_H
+#define __EMU8000_LOCAL_H
+/*
+ * Local defininitons for the emu8000 (AWE32/64)
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/wait.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/emu8000.h>
+#include <sound/emu8000_reg.h>
+
+/* emu8000_patch.c */
+int snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr,
+ const void __user *data, long count);
+int snd_emu8000_sample_free(struct snd_emux *rec, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr);
+void snd_emu8000_sample_reset(struct snd_emux *rec);
+
+/* emu8000_callback.c */
+void snd_emu8000_ops_setup(struct snd_emu8000 *emu);
+
+/* emu8000_pcm.c */
+int snd_emu8000_pcm_new(struct snd_card *card, struct snd_emu8000 *emu, int index);
+
+#endif /* __EMU8000_LOCAL_H */
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
new file mode 100644
index 0000000..c99c607
--- /dev/null
+++ b/sound/isa/sb/emu8000_patch.c
@@ -0,0 +1,305 @@
+/*
+ * Patch routines for the emu8000 (AWE32/64)
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "emu8000_local.h"
+#include <asm/uaccess.h>
+#include <linux/moduleparam.h>
+
+static int emu8000_reset_addr;
+module_param(emu8000_reset_addr, int, 0444);
+MODULE_PARM_DESC(emu8000_reset_addr, "reset write address at each time (makes slowdown)");
+
+
+/*
+ * Open up channels.
+ */
+static int
+snd_emu8000_open_dma(struct snd_emu8000 *emu, int write)
+{
+ int i;
+
+ /* reserve all 30 voices for loading */
+ for (i = 0; i < EMU8000_DRAM_VOICES; i++) {
+ snd_emux_lock_voice(emu->emu, i);
+ snd_emu8000_dma_chan(emu, i, write);
+ }
+
+ /* assign voice 31 and 32 to ROM */
+ EMU8000_VTFT_WRITE(emu, 30, 0);
+ EMU8000_PSST_WRITE(emu, 30, 0x1d8);
+ EMU8000_CSL_WRITE(emu, 30, 0x1e0);
+ EMU8000_CCCA_WRITE(emu, 30, 0x1d8);
+ EMU8000_VTFT_WRITE(emu, 31, 0);
+ EMU8000_PSST_WRITE(emu, 31, 0x1d8);
+ EMU8000_CSL_WRITE(emu, 31, 0x1e0);
+ EMU8000_CCCA_WRITE(emu, 31, 0x1d8);
+
+ return 0;
+}
+
+/*
+ * Close all dram channels.
+ */
+static void
+snd_emu8000_close_dma(struct snd_emu8000 *emu)
+{
+ int i;
+
+ for (i = 0; i < EMU8000_DRAM_VOICES; i++) {
+ snd_emu8000_dma_chan(emu, i, EMU8000_RAM_CLOSE);
+ snd_emux_unlock_voice(emu->emu, i);
+ }
+}
+
+/*
+ */
+
+#define BLANK_LOOP_START 4
+#define BLANK_LOOP_END 8
+#define BLANK_LOOP_SIZE 12
+#define BLANK_HEAD_SIZE 48
+
+/*
+ * Read a word from userland, taking care of conversions from
+ * 8bit samples etc.
+ */
+static unsigned short
+read_word(const void __user *buf, int offset, int mode)
+{
+ unsigned short c;
+ if (mode & SNDRV_SFNT_SAMPLE_8BITS) {
+ unsigned char cc;
+ get_user(cc, (unsigned char __user *)buf + offset);
+ c = cc << 8; /* convert 8bit -> 16bit */
+ } else {
+#ifdef SNDRV_LITTLE_ENDIAN
+ get_user(c, (unsigned short __user *)buf + offset);
+#else
+ unsigned short cc;
+ get_user(cc, (unsigned short __user *)buf + offset);
+ c = swab16(cc);
+#endif
+ }
+ if (mode & SNDRV_SFNT_SAMPLE_UNSIGNED)
+ c ^= 0x8000; /* unsigned -> signed */
+ return c;
+}
+
+/*
+ */
+static void
+snd_emu8000_write_wait(struct snd_emu8000 *emu)
+{
+ while ((EMU8000_SMALW_READ(emu) & 0x80000000) != 0) {
+ schedule_timeout_interruptible(1);
+ if (signal_pending(current))
+ break;
+ }
+}
+
+/*
+ * write sample word data
+ *
+ * You should not have to keep resetting the address each time
+ * as the chip is supposed to step on the next address automatically.
+ * It mostly does, but during writes of some samples at random it
+ * completely loses words (every one in 16 roughly but with no
+ * obvious pattern).
+ *
+ * This is therefore much slower than need be, but is at least
+ * working.
+ */
+static inline void
+write_word(struct snd_emu8000 *emu, int *offset, unsigned short data)
+{
+ if (emu8000_reset_addr) {
+ if (emu8000_reset_addr > 1)
+ snd_emu8000_write_wait(emu);
+ EMU8000_SMALW_WRITE(emu, *offset);
+ }
+ EMU8000_SMLD_WRITE(emu, data);
+ *offset += 1;
+}
+
+/*
+ * Write the sample to EMU800 memory. This routine is invoked out of
+ * the generic soundfont routines as a callback.
+ */
+int
+snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr,
+ const void __user *data, long count)
+{
+ int i;
+ int rc;
+ int offset;
+ int truesize;
+ int dram_offset, dram_start;
+ struct snd_emu8000 *emu;
+
+ emu = rec->hw;
+ if (snd_BUG_ON(!sp))
+ return -EINVAL;
+
+ if (sp->v.size == 0)
+ return 0;
+
+ /* be sure loop points start < end */
+ if (sp->v.loopstart > sp->v.loopend) {
+ int tmp = sp->v.loopstart;
+ sp->v.loopstart = sp->v.loopend;
+ sp->v.loopend = tmp;
+ }
+
+ /* compute true data size to be loaded */
+ truesize = sp->v.size;
+ if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP))
+ truesize += sp->v.loopend - sp->v.loopstart;
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK)
+ truesize += BLANK_LOOP_SIZE;
+
+ sp->block = snd_util_mem_alloc(hdr, truesize * 2);
+ if (sp->block == NULL) {
+ /*snd_printd("EMU8000: out of memory\n");*/
+ /* not ENOMEM (for compatibility) */
+ return -ENOSPC;
+ }
+
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) {
+ if (!access_ok(VERIFY_READ, data, sp->v.size))
+ return -EFAULT;
+ } else {
+ if (!access_ok(VERIFY_READ, data, sp->v.size * 2))
+ return -EFAULT;
+ }
+
+ /* recalculate address offset */
+ sp->v.end -= sp->v.start;
+ sp->v.loopstart -= sp->v.start;
+ sp->v.loopend -= sp->v.start;
+ sp->v.start = 0;
+
+ /* dram position (in word) -- mem_offset is byte */
+ dram_offset = EMU8000_DRAM_OFFSET + (sp->block->offset >> 1);
+ dram_start = dram_offset;
+
+ /* set the total size (store onto obsolete checksum value) */
+ sp->v.truesize = truesize * 2; /* in bytes */
+
+ snd_emux_terminate_all(emu->emu);
+ if ((rc = snd_emu8000_open_dma(emu, EMU8000_RAM_WRITE)) != 0)
+ return rc;
+
+ /* Set the address to start writing at */
+ snd_emu8000_write_wait(emu);
+ EMU8000_SMALW_WRITE(emu, dram_offset);
+
+ /*snd_emu8000_init_fm(emu);*/
+
+#if 0
+ /* first block - write 48 samples for silence */
+ if (! sp->block->offset) {
+ for (i = 0; i < BLANK_HEAD_SIZE; i++) {
+ write_word(emu, &dram_offset, 0);
+ }
+ }
+#endif
+
+ offset = 0;
+ for (i = 0; i < sp->v.size; i++) {
+ unsigned short s;
+
+ s = read_word(data, offset, sp->v.mode_flags);
+ offset++;
+ write_word(emu, &dram_offset, s);
+
+ /* we may take too long time in this loop.
+ * so give controls back to kernel if needed.
+ */
+ cond_resched();
+
+ if (i == sp->v.loopend &&
+ (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)))
+ {
+ int looplen = sp->v.loopend - sp->v.loopstart;
+ int k;
+
+ /* copy reverse loop */
+ for (k = 1; k <= looplen; k++) {
+ s = read_word(data, offset - k, sp->v.mode_flags);
+ write_word(emu, &dram_offset, s);
+ }
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_BIDIR_LOOP) {
+ sp->v.loopend += looplen;
+ } else {
+ sp->v.loopstart += looplen;
+ sp->v.loopend += looplen;
+ }
+ sp->v.end += looplen;
+ }
+ }
+
+ /* if no blank loop is attached in the sample, add it */
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) {
+ for (i = 0; i < BLANK_LOOP_SIZE; i++) {
+ write_word(emu, &dram_offset, 0);
+ }
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) {
+ sp->v.loopstart = sp->v.end + BLANK_LOOP_START;
+ sp->v.loopend = sp->v.end + BLANK_LOOP_END;
+ }
+ }
+
+ /* add dram offset */
+ sp->v.start += dram_start;
+ sp->v.end += dram_start;
+ sp->v.loopstart += dram_start;
+ sp->v.loopend += dram_start;
+
+ snd_emu8000_close_dma(emu);
+ snd_emu8000_init_fm(emu);
+
+ return 0;
+}
+
+/*
+ * free a sample block
+ */
+int
+snd_emu8000_sample_free(struct snd_emux *rec, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr)
+{
+ if (sp->block) {
+ snd_util_mem_free(hdr, sp->block);
+ sp->block = NULL;
+ }
+ return 0;
+}
+
+
+/*
+ * sample_reset callback - terminate voices
+ */
+void
+snd_emu8000_sample_reset(struct snd_emux *rec)
+{
+ snd_emux_terminate_all(rec);
+}
diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c
new file mode 100644
index 0000000..91dc3d8
--- /dev/null
+++ b/sound/isa/sb/emu8000_pcm.c
@@ -0,0 +1,701 @@
+/*
+ * pcm emulation on emu8000 wavetable
+ *
+ * Copyright (C) 2002 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "emu8000_local.h"
+#include <linux/init.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+
+/*
+ * define the following if you want to use this pcm with non-interleaved mode
+ */
+/* #define USE_NONINTERLEAVE */
+
+/* NOTE: for using the non-interleaved mode with alsa-lib, you have to set
+ * mmap_emulation flag to 1 in your .asoundrc, such like
+ *
+ * pcm.emu8k {
+ * type plug
+ * slave.pcm {
+ * type hw
+ * card 0
+ * device 1
+ * mmap_emulation 1
+ * }
+ * }
+ *
+ * besides, for the time being, the non-interleaved mode doesn't work well on
+ * alsa-lib...
+ */
+
+
+struct snd_emu8k_pcm {
+ struct snd_emu8000 *emu;
+ struct snd_pcm_substream *substream;
+
+ unsigned int allocated_bytes;
+ struct snd_util_memblk *block;
+ unsigned int offset;
+ unsigned int buf_size;
+ unsigned int period_size;
+ unsigned int loop_start[2];
+ unsigned int pitch;
+ int panning[2];
+ int last_ptr;
+ int period_pos;
+ int voices;
+ unsigned int dram_opened: 1;
+ unsigned int running: 1;
+ unsigned int timer_running: 1;
+ struct timer_list timer;
+ spinlock_t timer_lock;
+};
+
+#define LOOP_BLANK_SIZE 8
+
+
+/*
+ * open up channels for the simultaneous data transfer and playback
+ */
+static int
+emu8k_open_dram_for_pcm(struct snd_emu8000 *emu, int channels)
+{
+ int i;
+
+ /* reserve up to 2 voices for playback */
+ snd_emux_lock_voice(emu->emu, 0);
+ if (channels > 1)
+ snd_emux_lock_voice(emu->emu, 1);
+
+ /* reserve 28 voices for loading */
+ for (i = channels + 1; i < EMU8000_DRAM_VOICES; i++) {
+ unsigned int mode = EMU8000_RAM_WRITE;
+ snd_emux_lock_voice(emu->emu, i);
+#ifndef USE_NONINTERLEAVE
+ if (channels > 1 && (i & 1) != 0)
+ mode |= EMU8000_RAM_RIGHT;
+#endif
+ snd_emu8000_dma_chan(emu, i, mode);
+ }
+
+ /* assign voice 31 and 32 to ROM */
+ EMU8000_VTFT_WRITE(emu, 30, 0);
+ EMU8000_PSST_WRITE(emu, 30, 0x1d8);
+ EMU8000_CSL_WRITE(emu, 30, 0x1e0);
+ EMU8000_CCCA_WRITE(emu, 30, 0x1d8);
+ EMU8000_VTFT_WRITE(emu, 31, 0);
+ EMU8000_PSST_WRITE(emu, 31, 0x1d8);
+ EMU8000_CSL_WRITE(emu, 31, 0x1e0);
+ EMU8000_CCCA_WRITE(emu, 31, 0x1d8);
+
+ return 0;
+}
+
+/*
+ */
+static void
+snd_emu8000_write_wait(struct snd_emu8000 *emu, int can_schedule)
+{
+ while ((EMU8000_SMALW_READ(emu) & 0x80000000) != 0) {
+ if (can_schedule) {
+ schedule_timeout_interruptible(1);
+ if (signal_pending(current))
+ break;
+ }
+ }
+}
+
+/*
+ * close all channels
+ */
+static void
+emu8k_close_dram(struct snd_emu8000 *emu)
+{
+ int i;
+
+ for (i = 0; i < 2; i++)
+ snd_emux_unlock_voice(emu->emu, i);
+ for (; i < EMU8000_DRAM_VOICES; i++) {
+ snd_emu8000_dma_chan(emu, i, EMU8000_RAM_CLOSE);
+ snd_emux_unlock_voice(emu->emu, i);
+ }
+}
+
+/*
+ * convert Hz to AWE32 rate offset (see emux/soundfont.c)
+ */
+
+#define OFFSET_SAMPLERATE 1011119 /* base = 44100 */
+#define SAMPLERATE_RATIO 4096
+
+static int calc_rate_offset(int hz)
+{
+ return snd_sf_linear_to_log(hz, OFFSET_SAMPLERATE, SAMPLERATE_RATIO);
+}
+
+
+/*
+ */
+
+static struct snd_pcm_hardware emu8k_pcm_hw = {
+#ifdef USE_NONINTERLEAVE
+ .info = SNDRV_PCM_INFO_NONINTERLEAVED,
+#else
+ .info = SNDRV_PCM_INFO_INTERLEAVED,
+#endif
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 1024,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+
+};
+
+/*
+ * get the current position at the given channel from CCCA register
+ */
+static inline int emu8k_get_curpos(struct snd_emu8k_pcm *rec, int ch)
+{
+ int val = EMU8000_CCCA_READ(rec->emu, ch) & 0xfffffff;
+ val -= rec->loop_start[ch] - 1;
+ return val;
+}
+
+
+/*
+ * timer interrupt handler
+ * check the current position and update the period if necessary.
+ */
+static void emu8k_pcm_timer_func(unsigned long data)
+{
+ struct snd_emu8k_pcm *rec = (struct snd_emu8k_pcm *)data;
+ int ptr, delta;
+
+ spin_lock(&rec->timer_lock);
+ /* update the current pointer */
+ ptr = emu8k_get_curpos(rec, 0);
+ if (ptr < rec->last_ptr)
+ delta = ptr + rec->buf_size - rec->last_ptr;
+ else
+ delta = ptr - rec->last_ptr;
+ rec->period_pos += delta;
+ rec->last_ptr = ptr;
+
+ /* reprogram timer */
+ rec->timer.expires = jiffies + 1;
+ add_timer(&rec->timer);
+
+ /* update period */
+ if (rec->period_pos >= (int)rec->period_size) {
+ rec->period_pos %= rec->period_size;
+ spin_unlock(&rec->timer_lock);
+ snd_pcm_period_elapsed(rec->substream);
+ return;
+ }
+ spin_unlock(&rec->timer_lock);
+}
+
+
+/*
+ * open pcm
+ * creating an instance here
+ */
+static int emu8k_pcm_open(struct snd_pcm_substream *subs)
+{
+ struct snd_emu8000 *emu = snd_pcm_substream_chip(subs);
+ struct snd_emu8k_pcm *rec;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ rec = kzalloc(sizeof(*rec), GFP_KERNEL);
+ if (! rec)
+ return -ENOMEM;
+
+ rec->emu = emu;
+ rec->substream = subs;
+ runtime->private_data = rec;
+
+ spin_lock_init(&rec->timer_lock);
+ init_timer(&rec->timer);
+ rec->timer.function = emu8k_pcm_timer_func;
+ rec->timer.data = (unsigned long)rec;
+
+ runtime->hw = emu8k_pcm_hw;
+ runtime->hw.buffer_bytes_max = emu->mem_size - LOOP_BLANK_SIZE * 3;
+ runtime->hw.period_bytes_max = runtime->hw.buffer_bytes_max / 2;
+
+ /* use timer to update periods.. (specified in msec) */
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+ (1000000 + HZ - 1) / HZ, UINT_MAX);
+
+ return 0;
+}
+
+static int emu8k_pcm_close(struct snd_pcm_substream *subs)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+ kfree(rec);
+ subs->runtime->private_data = NULL;
+ return 0;
+}
+
+/*
+ * calculate pitch target
+ */
+static int calc_pitch_target(int pitch)
+{
+ int ptarget = 1 << (pitch >> 12);
+ if (pitch & 0x800) ptarget += (ptarget * 0x102e) / 0x2710;
+ if (pitch & 0x400) ptarget += (ptarget * 0x764) / 0x2710;
+ if (pitch & 0x200) ptarget += (ptarget * 0x389) / 0x2710;
+ ptarget += (ptarget >> 1);
+ if (ptarget > 0xffff) ptarget = 0xffff;
+ return ptarget;
+}
+
+/*
+ * set up the voice
+ */
+static void setup_voice(struct snd_emu8k_pcm *rec, int ch)
+{
+ struct snd_emu8000 *hw = rec->emu;
+ unsigned int temp;
+
+ /* channel to be silent and idle */
+ EMU8000_DCYSUSV_WRITE(hw, ch, 0x0080);
+ EMU8000_VTFT_WRITE(hw, ch, 0x0000FFFF);
+ EMU8000_CVCF_WRITE(hw, ch, 0x0000FFFF);
+ EMU8000_PTRX_WRITE(hw, ch, 0);
+ EMU8000_CPF_WRITE(hw, ch, 0);
+
+ /* pitch offset */
+ EMU8000_IP_WRITE(hw, ch, rec->pitch);
+ /* set envelope parameters */
+ EMU8000_ENVVAL_WRITE(hw, ch, 0x8000);
+ EMU8000_ATKHLD_WRITE(hw, ch, 0x7f7f);
+ EMU8000_DCYSUS_WRITE(hw, ch, 0x7f7f);
+ EMU8000_ENVVOL_WRITE(hw, ch, 0x8000);
+ EMU8000_ATKHLDV_WRITE(hw, ch, 0x7f7f);
+ /* decay/sustain parameter for volume envelope is used
+ for triggerg the voice */
+ /* modulation envelope heights */
+ EMU8000_PEFE_WRITE(hw, ch, 0x0);
+ /* lfo1/2 delay */
+ EMU8000_LFO1VAL_WRITE(hw, ch, 0x8000);
+ EMU8000_LFO2VAL_WRITE(hw, ch, 0x8000);
+ /* lfo1 pitch & cutoff shift */
+ EMU8000_FMMOD_WRITE(hw, ch, 0);
+ /* lfo1 volume & freq */
+ EMU8000_TREMFRQ_WRITE(hw, ch, 0);
+ /* lfo2 pitch & freq */
+ EMU8000_FM2FRQ2_WRITE(hw, ch, 0);
+ /* pan & loop start */
+ temp = rec->panning[ch];
+ temp = (temp <<24) | ((unsigned int)rec->loop_start[ch] - 1);
+ EMU8000_PSST_WRITE(hw, ch, temp);
+ /* chorus & loop end (chorus 8bit, MSB) */
+ temp = 0; // chorus
+ temp = (temp << 24) | ((unsigned int)rec->loop_start[ch] + rec->buf_size - 1);
+ EMU8000_CSL_WRITE(hw, ch, temp);
+ /* Q & current address (Q 4bit value, MSB) */
+ temp = 0; // filterQ
+ temp = (temp << 28) | ((unsigned int)rec->loop_start[ch] - 1);
+ EMU8000_CCCA_WRITE(hw, ch, temp);
+ /* clear unknown registers */
+ EMU8000_00A0_WRITE(hw, ch, 0);
+ EMU8000_0080_WRITE(hw, ch, 0);
+}
+
+/*
+ * trigger the voice
+ */
+static void start_voice(struct snd_emu8k_pcm *rec, int ch)
+{
+ unsigned long flags;
+ struct snd_emu8000 *hw = rec->emu;
+ unsigned int temp, aux;
+ int pt = calc_pitch_target(rec->pitch);
+
+ /* cutoff and volume */
+ EMU8000_IFATN_WRITE(hw, ch, 0xff00);
+ EMU8000_VTFT_WRITE(hw, ch, 0xffff);
+ EMU8000_CVCF_WRITE(hw, ch, 0xffff);
+ /* trigger envelope */
+ EMU8000_DCYSUSV_WRITE(hw, ch, 0x7f7f);
+ /* set reverb and pitch target */
+ temp = 0; // reverb
+ if (rec->panning[ch] == 0)
+ aux = 0xff;
+ else
+ aux = (-rec->panning[ch]) & 0xff;
+ temp = (temp << 8) | (pt << 16) | aux;
+ EMU8000_PTRX_WRITE(hw, ch, temp);
+ EMU8000_CPF_WRITE(hw, ch, pt << 16);
+
+ /* start timer */
+ spin_lock_irqsave(&rec->timer_lock, flags);
+ if (! rec->timer_running) {
+ rec->timer.expires = jiffies + 1;
+ add_timer(&rec->timer);
+ rec->timer_running = 1;
+ }
+ spin_unlock_irqrestore(&rec->timer_lock, flags);
+}
+
+/*
+ * stop the voice immediately
+ */
+static void stop_voice(struct snd_emu8k_pcm *rec, int ch)
+{
+ unsigned long flags;
+ struct snd_emu8000 *hw = rec->emu;
+
+ EMU8000_DCYSUSV_WRITE(hw, ch, 0x807F);
+
+ /* stop timer */
+ spin_lock_irqsave(&rec->timer_lock, flags);
+ if (rec->timer_running) {
+ del_timer(&rec->timer);
+ rec->timer_running = 0;
+ }
+ spin_unlock_irqrestore(&rec->timer_lock, flags);
+}
+
+static int emu8k_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+ int ch;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ for (ch = 0; ch < rec->voices; ch++)
+ start_voice(rec, ch);
+ rec->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ rec->running = 0;
+ for (ch = 0; ch < rec->voices; ch++)
+ stop_voice(rec, ch);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/*
+ * copy / silence ops
+ */
+
+/*
+ * this macro should be inserted in the copy/silence loops
+ * to reduce the latency. without this, the system will hang up
+ * during the whole loop.
+ */
+#define CHECK_SCHEDULER() \
+do { \
+ cond_resched();\
+ if (signal_pending(current))\
+ return -EAGAIN;\
+} while (0)
+
+
+#ifdef USE_NONINTERLEAVE
+/* copy one channel block */
+static int emu8k_transfer_block(struct snd_emu8000 *emu, int offset, unsigned short *buf, int count)
+{
+ EMU8000_SMALW_WRITE(emu, offset);
+ while (count > 0) {
+ unsigned short sval;
+ CHECK_SCHEDULER();
+ get_user(sval, buf);
+ EMU8000_SMLD_WRITE(emu, sval);
+ buf++;
+ count--;
+ }
+ return 0;
+}
+
+static int emu8k_pcm_copy(struct snd_pcm_substream *subs,
+ int voice,
+ snd_pcm_uframes_t pos,
+ void *src,
+ snd_pcm_uframes_t count)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+ struct snd_emu8000 *emu = rec->emu;
+
+ snd_emu8000_write_wait(emu, 1);
+ if (voice == -1) {
+ unsigned short *buf = src;
+ int i, err;
+ count /= rec->voices;
+ for (i = 0; i < rec->voices; i++) {
+ err = emu8k_transfer_block(emu, pos + rec->loop_start[i], buf, count);
+ if (err < 0)
+ return err;
+ buf += count;
+ }
+ return 0;
+ } else {
+ return emu8k_transfer_block(emu, pos + rec->loop_start[voice], src, count);
+ }
+}
+
+/* make a channel block silence */
+static int emu8k_silence_block(struct snd_emu8000 *emu, int offset, int count)
+{
+ EMU8000_SMALW_WRITE(emu, offset);
+ while (count > 0) {
+ CHECK_SCHEDULER();
+ EMU8000_SMLD_WRITE(emu, 0);
+ count--;
+ }
+ return 0;
+}
+
+static int emu8k_pcm_silence(struct snd_pcm_substream *subs,
+ int voice,
+ snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+ struct snd_emu8000 *emu = rec->emu;
+
+ snd_emu8000_write_wait(emu, 1);
+ if (voice == -1 && rec->voices == 1)
+ voice = 0;
+ if (voice == -1) {
+ int err;
+ err = emu8k_silence_block(emu, pos + rec->loop_start[0], count / 2);
+ if (err < 0)
+ return err;
+ return emu8k_silence_block(emu, pos + rec->loop_start[1], count / 2);
+ } else {
+ return emu8k_silence_block(emu, pos + rec->loop_start[voice], count);
+ }
+}
+
+#else /* interleave */
+
+/*
+ * copy the interleaved data can be done easily by using
+ * DMA "left" and "right" channels on emu8k engine.
+ */
+static int emu8k_pcm_copy(struct snd_pcm_substream *subs,
+ int voice,
+ snd_pcm_uframes_t pos,
+ void __user *src,
+ snd_pcm_uframes_t count)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+ struct snd_emu8000 *emu = rec->emu;
+ unsigned short __user *buf = src;
+
+ snd_emu8000_write_wait(emu, 1);
+ EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]);
+ if (rec->voices > 1)
+ EMU8000_SMARW_WRITE(emu, pos + rec->loop_start[1]);
+
+ while (count-- > 0) {
+ unsigned short sval;
+ CHECK_SCHEDULER();
+ get_user(sval, buf);
+ EMU8000_SMLD_WRITE(emu, sval);
+ buf++;
+ if (rec->voices > 1) {
+ CHECK_SCHEDULER();
+ get_user(sval, buf);
+ EMU8000_SMRD_WRITE(emu, sval);
+ buf++;
+ }
+ }
+ return 0;
+}
+
+static int emu8k_pcm_silence(struct snd_pcm_substream *subs,
+ int voice,
+ snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+ struct snd_emu8000 *emu = rec->emu;
+
+ snd_emu8000_write_wait(emu, 1);
+ EMU8000_SMALW_WRITE(emu, rec->loop_start[0] + pos);
+ if (rec->voices > 1)
+ EMU8000_SMARW_WRITE(emu, rec->loop_start[1] + pos);
+ while (count-- > 0) {
+ CHECK_SCHEDULER();
+ EMU8000_SMLD_WRITE(emu, 0);
+ if (rec->voices > 1) {
+ CHECK_SCHEDULER();
+ EMU8000_SMRD_WRITE(emu, 0);
+ }
+ }
+ return 0;
+}
+#endif
+
+
+/*
+ * allocate a memory block
+ */
+static int emu8k_pcm_hw_params(struct snd_pcm_substream *subs,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+
+ if (rec->block) {
+ /* reallocation - release the old block */
+ snd_util_mem_free(rec->emu->memhdr, rec->block);
+ rec->block = NULL;
+ }
+
+ rec->allocated_bytes = params_buffer_bytes(hw_params) + LOOP_BLANK_SIZE * 4;
+ rec->block = snd_util_mem_alloc(rec->emu->memhdr, rec->allocated_bytes);
+ if (! rec->block)
+ return -ENOMEM;
+ rec->offset = EMU8000_DRAM_OFFSET + (rec->block->offset >> 1); /* in word */
+ /* at least dma_bytes must be set for non-interleaved mode */
+ subs->dma_buffer.bytes = params_buffer_bytes(hw_params);
+
+ return 0;
+}
+
+/*
+ * free the memory block
+ */
+static int emu8k_pcm_hw_free(struct snd_pcm_substream *subs)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+
+ if (rec->block) {
+ int ch;
+ for (ch = 0; ch < rec->voices; ch++)
+ stop_voice(rec, ch); // to be sure
+ if (rec->dram_opened)
+ emu8k_close_dram(rec->emu);
+ snd_util_mem_free(rec->emu->memhdr, rec->block);
+ rec->block = NULL;
+ }
+ return 0;
+}
+
+/*
+ */
+static int emu8k_pcm_prepare(struct snd_pcm_substream *subs)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+
+ rec->pitch = 0xe000 + calc_rate_offset(subs->runtime->rate);
+ rec->last_ptr = 0;
+ rec->period_pos = 0;
+
+ rec->buf_size = subs->runtime->buffer_size;
+ rec->period_size = subs->runtime->period_size;
+ rec->voices = subs->runtime->channels;
+ rec->loop_start[0] = rec->offset + LOOP_BLANK_SIZE;
+ if (rec->voices > 1)
+ rec->loop_start[1] = rec->loop_start[0] + rec->buf_size + LOOP_BLANK_SIZE;
+ if (rec->voices > 1) {
+ rec->panning[0] = 0xff;
+ rec->panning[1] = 0x00;
+ } else
+ rec->panning[0] = 0x80;
+
+ if (! rec->dram_opened) {
+ int err, i, ch;
+
+ snd_emux_terminate_all(rec->emu->emu);
+ if ((err = emu8k_open_dram_for_pcm(rec->emu, rec->voices)) != 0)
+ return err;
+ rec->dram_opened = 1;
+
+ /* clear loop blanks */
+ snd_emu8000_write_wait(rec->emu, 0);
+ EMU8000_SMALW_WRITE(rec->emu, rec->offset);
+ for (i = 0; i < LOOP_BLANK_SIZE; i++)
+ EMU8000_SMLD_WRITE(rec->emu, 0);
+ for (ch = 0; ch < rec->voices; ch++) {
+ EMU8000_SMALW_WRITE(rec->emu, rec->loop_start[ch] + rec->buf_size);
+ for (i = 0; i < LOOP_BLANK_SIZE; i++)
+ EMU8000_SMLD_WRITE(rec->emu, 0);
+ }
+ }
+
+ setup_voice(rec, 0);
+ if (rec->voices > 1)
+ setup_voice(rec, 1);
+ return 0;
+}
+
+static snd_pcm_uframes_t emu8k_pcm_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_emu8k_pcm *rec = subs->runtime->private_data;
+ if (rec->running)
+ return emu8k_get_curpos(rec, 0);
+ return 0;
+}
+
+
+static struct snd_pcm_ops emu8k_pcm_ops = {
+ .open = emu8k_pcm_open,
+ .close = emu8k_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = emu8k_pcm_hw_params,
+ .hw_free = emu8k_pcm_hw_free,
+ .prepare = emu8k_pcm_prepare,
+ .trigger = emu8k_pcm_trigger,
+ .pointer = emu8k_pcm_pointer,
+ .copy = emu8k_pcm_copy,
+ .silence = emu8k_pcm_silence,
+};
+
+
+static void snd_emu8000_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_emu8000 *emu = pcm->private_data;
+ emu->pcm = NULL;
+}
+
+int snd_emu8000_pcm_new(struct snd_card *card, struct snd_emu8000 *emu, int index)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(card, "Emu8000 PCM", index, 1, 0, &pcm)) < 0)
+ return err;
+ pcm->private_data = emu;
+ pcm->private_free = snd_emu8000_pcm_free;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &emu8k_pcm_ops);
+ emu->pcm = pcm;
+
+ snd_device_register(card, pcm);
+
+ return 0;
+}
diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c
new file mode 100644
index 0000000..0c7905c
--- /dev/null
+++ b/sound/isa/sb/emu8000_synth.c
@@ -0,0 +1,135 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * and (c) 1999 Steve Ratcliffe <steve@parabola.demon.co.uk>
+ * Copyright (C) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Emu8000 synth plug-in routine
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "emu8000_local.h"
+#include <linux/init.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Takashi Iwai, Steve Ratcliffe");
+MODULE_DESCRIPTION("Emu8000 synth plug-in routine");
+MODULE_LICENSE("GPL");
+
+/*----------------------------------------------------------------*/
+
+/*
+ * create a new hardware dependent device for Emu8000
+ */
+static int snd_emu8000_new_device(struct snd_seq_device *dev)
+{
+ struct snd_emu8000 *hw;
+ struct snd_emux *emu;
+
+ hw = *(struct snd_emu8000**)SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ if (hw == NULL)
+ return -EINVAL;
+
+ if (hw->emu)
+ return -EBUSY; /* already exists..? */
+
+ if (snd_emux_new(&emu) < 0)
+ return -ENOMEM;
+
+ hw->emu = emu;
+ snd_emu8000_ops_setup(hw);
+
+ emu->hw = hw;
+ emu->max_voices = EMU8000_DRAM_VOICES;
+ emu->num_ports = hw->seq_ports;
+
+ if (hw->memhdr) {
+ snd_printk(KERN_ERR "memhdr is already initialized!?\n");
+ snd_util_memhdr_free(hw->memhdr);
+ }
+ hw->memhdr = snd_util_memhdr_new(hw->mem_size);
+ if (hw->memhdr == NULL) {
+ snd_emux_free(emu);
+ hw->emu = NULL;
+ return -ENOMEM;
+ }
+
+ emu->memhdr = hw->memhdr;
+ emu->midi_ports = hw->seq_ports < 2 ? hw->seq_ports : 2; /* number of virmidi ports */
+ emu->midi_devidx = 1;
+ emu->linear_panning = 1;
+ emu->hwdep_idx = 2; /* FIXED */
+
+ if (snd_emux_register(emu, dev->card, hw->index, "Emu8000") < 0) {
+ snd_emux_free(emu);
+ snd_util_memhdr_free(hw->memhdr);
+ hw->emu = NULL;
+ hw->memhdr = NULL;
+ return -ENOMEM;
+ }
+
+ if (hw->mem_size > 0)
+ snd_emu8000_pcm_new(dev->card, hw, 1);
+
+ dev->driver_data = hw;
+
+ return 0;
+}
+
+
+/*
+ * free all resources
+ */
+static int snd_emu8000_delete_device(struct snd_seq_device *dev)
+{
+ struct snd_emu8000 *hw;
+
+ if (dev->driver_data == NULL)
+ return 0; /* no synth was allocated actually */
+
+ hw = dev->driver_data;
+ if (hw->pcm)
+ snd_device_free(dev->card, hw->pcm);
+ if (hw->emu)
+ snd_emux_free(hw->emu);
+ if (hw->memhdr)
+ snd_util_memhdr_free(hw->memhdr);
+ hw->emu = NULL;
+ hw->memhdr = NULL;
+ return 0;
+}
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_emu8000_init(void)
+{
+
+ static struct snd_seq_dev_ops ops = {
+ snd_emu8000_new_device,
+ snd_emu8000_delete_device,
+ };
+ return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU8000, &ops,
+ sizeof(struct snd_emu8000*));
+}
+
+static void __exit alsa_emu8000_exit(void)
+{
+ snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU8000);
+}
+
+module_init(alsa_emu8000_init)
+module_exit(alsa_emu8000_exit)
diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c
new file mode 100644
index 0000000..c8c8e21
--- /dev/null
+++ b/sound/isa/sb/es968.c
@@ -0,0 +1,247 @@
+
+/*
+ card-es968.c - driver for ESS AudioDrive ES968 based soundcards.
+ Copyright (C) 1999 by Massimo Piccioni <dafastidio@libero.it>
+
+ Thanks to Pierfrancesco 'qM2' Passerini.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/sb.h>
+
+#define PFX "es968: "
+
+MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
+MODULE_DESCRIPTION("ESS AudioDrive ES968");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ESS,AudioDrive ES968}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* Pnp setup */
+static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for es968 based soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for es968 based soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable es968 based soundcard.");
+
+struct snd_card_es968 {
+ struct pnp_dev *dev;
+ struct snd_sb *chip;
+};
+
+static struct pnp_card_device_id snd_es968_pnpids[] = {
+ { .id = "ESS0968", .devs = { { "@@@0968" }, } },
+ { .id = "", } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_es968_pnpids);
+
+#define DRIVER_NAME "snd-card-es968"
+
+static irqreturn_t snd_card_es968_interrupt(int irq, void *dev_id)
+{
+ struct snd_sb *chip = dev_id;
+
+ if (chip->open & SB_OPEN_PCM) {
+ return snd_sb8dsp_interrupt(chip);
+ } else {
+ return snd_sb8dsp_midi_interrupt(chip);
+ }
+}
+
+static int __devinit snd_card_es968_pnp(int dev, struct snd_card_es968 *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->dev == NULL)
+ return -ENODEV;
+
+ pdev = acard->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n");
+ return err;
+ }
+ port[dev] = pnp_port_start(pdev, 0);
+ dma8[dev] = pnp_dma(pdev, 1);
+ irq[dev] = pnp_irq(pdev, 0);
+
+ return 0;
+}
+
+static int __devinit snd_card_es968_probe(int dev,
+ struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ int error;
+ struct snd_sb *chip;
+ struct snd_card *card;
+ struct snd_card_es968 *acard;
+
+ if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_es968))) == NULL)
+ return -ENOMEM;
+ acard = card->private_data;
+ if ((error = snd_card_es968_pnp(dev, acard, pcard, pid))) {
+ snd_card_free(card);
+ return error;
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+
+ if ((error = snd_sbdsp_create(card, port[dev],
+ irq[dev],
+ snd_card_es968_interrupt,
+ dma8[dev],
+ -1,
+ SB_HW_AUTO, &chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ acard->chip = chip;
+
+ if ((error = snd_sb8dsp_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if ((error = snd_sbmixer_new(chip)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ if ((error = snd_sb8dsp_midi(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+
+ strcpy(card->driver, "ES968");
+ strcpy(card->shortname, "ESS ES968");
+ sprintf(card->longname, "%s soundcard, %s at 0x%lx, irq %d, dma %d",
+ card->shortname, chip->name, chip->port, irq[dev], dma8[dev]);
+
+ if ((error = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return error;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ return 0;
+}
+
+static unsigned int __devinitdata es968_devices;
+
+static int __devinit snd_es968_pnp_detect(struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ static int dev;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (!enable[dev])
+ continue;
+ res = snd_card_es968_probe(dev, card, id);
+ if (res < 0)
+ return res;
+ dev++;
+ es968_devices++;
+ return 0;
+ }
+ return -ENODEV;
+}
+
+static void __devexit snd_es968_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_es968_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_es968 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_sbmixer_suspend(chip);
+ return 0;
+}
+
+static int snd_es968_pnp_resume(struct pnp_card_link *pcard)
+{
+ struct snd_card *card = pnp_get_card_drvdata(pcard);
+ struct snd_card_es968 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_sbdsp_reset(chip);
+ snd_sbmixer_resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct pnp_card_driver es968_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "es968",
+ .id_table = snd_es968_pnpids,
+ .probe = snd_es968_pnp_detect,
+ .remove = __devexit_p(snd_es968_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_es968_pnp_suspend,
+ .resume = snd_es968_pnp_resume,
+#endif
+};
+
+static int __init alsa_card_es968_init(void)
+{
+ int err = pnp_register_card_driver(&es968_pnpc_driver);
+ if (err)
+ return err;
+
+ if (!es968_devices) {
+ pnp_unregister_card_driver(&es968_pnpc_driver);
+#ifdef MODULE
+ snd_printk(KERN_ERR "no ES968 based soundcards found\n");
+#endif
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit alsa_card_es968_exit(void)
+{
+ pnp_unregister_card_driver(&es968_pnpc_driver);
+}
+
+module_init(alsa_card_es968_init)
+module_exit(alsa_card_es968_exit)
diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c
new file mode 100644
index 0000000..2c201f7
--- /dev/null
+++ b/sound/isa/sb/sb16.c
@@ -0,0 +1,695 @@
+/*
+ * Driver for SoundBlaster 16/AWE32/AWE64 soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/dma.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pnp.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+#include <sound/sb16_csp.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/emu8000.h>
+#include <sound/seq_device.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+#ifdef SNDRV_SBAWE
+#define PFX "sbawe: "
+#else
+#define PFX "sb16: "
+#endif
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_LICENSE("GPL");
+#ifndef SNDRV_SBAWE
+MODULE_DESCRIPTION("Sound Blaster 16");
+MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 16},"
+ "{Creative Labs,SB Vibra16S},"
+ "{Creative Labs,SB Vibra16C},"
+ "{Creative Labs,SB Vibra16CL},"
+ "{Creative Labs,SB Vibra16X}}");
+#else
+MODULE_DESCRIPTION("Sound Blaster AWE");
+MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB AWE 32},"
+ "{Creative Labs,SB AWE 64},"
+ "{Creative Labs,SB AWE 64 Gold}}");
+#endif
+
+#if 0
+#define SNDRV_DEBUG_IRQ
+#endif
+
+#if defined(SNDRV_SBAWE) && (defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)))
+#define SNDRV_SBAWE_EMU8000
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x330,0x300 */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+#ifdef SNDRV_SBAWE_EMU8000
+static long awe_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+#endif
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */
+static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */
+static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 5,6,7 */
+static int mic_agc[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#ifdef CONFIG_SND_SB16_CSP
+static int csp[SNDRV_CARDS];
+#endif
+#ifdef SNDRV_SBAWE_EMU8000
+static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4};
+#endif
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for SoundBlaster 16 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for SoundBlaster 16 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable SoundBlaster 16 soundcard.");
+#ifdef CONFIG_PNP
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard.");
+#endif
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for SB16 driver.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for SB16 driver.");
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM port # for SB16 PnP driver.");
+#ifdef SNDRV_SBAWE_EMU8000
+module_param_array(awe_port, long, NULL, 0444);
+MODULE_PARM_DESC(awe_port, "AWE port # for SB16 PnP driver.");
+#endif
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for SB16 driver.");
+module_param_array(dma8, int, NULL, 0444);
+MODULE_PARM_DESC(dma8, "8-bit DMA # for SB16 driver.");
+module_param_array(dma16, int, NULL, 0444);
+MODULE_PARM_DESC(dma16, "16-bit DMA # for SB16 driver.");
+module_param_array(mic_agc, int, NULL, 0444);
+MODULE_PARM_DESC(mic_agc, "Mic Auto-Gain-Control switch.");
+#ifdef CONFIG_SND_SB16_CSP
+module_param_array(csp, int, NULL, 0444);
+MODULE_PARM_DESC(csp, "ASP/CSP chip support.");
+#endif
+#ifdef SNDRV_SBAWE_EMU8000
+module_param_array(seq_ports, int, NULL, 0444);
+MODULE_PARM_DESC(seq_ports, "Number of sequencer ports for WaveTable synth.");
+#endif
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnp_registered;
+#endif
+
+struct snd_card_sb16 {
+ struct resource *fm_res; /* used to block FM i/o region for legacy cards */
+ struct snd_sb *chip;
+#ifdef CONFIG_PNP
+ int dev_no;
+ struct pnp_dev *dev;
+#ifdef SNDRV_SBAWE_EMU8000
+ struct pnp_dev *devwt;
+#endif
+#endif
+};
+
+#ifdef CONFIG_PNP
+
+static struct pnp_card_device_id snd_sb16_pnpids[] = {
+#ifndef SNDRV_SBAWE
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL0024", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL0025", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL0026", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL0027", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL0028", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL0029", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL002a", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ /* Note: This card has also a CTL0051:StereoEnhance device!!! */
+ { .id = "CTL002b", .devs = { { "CTL0031" } } },
+ /* Sound Blaster 16 PnP */
+ { .id = "CTL002c", .devs = { { "CTL0031" } } },
+ /* Sound Blaster Vibra16S */
+ { .id = "CTL0051", .devs = { { "CTL0001" } } },
+ /* Sound Blaster Vibra16C */
+ { .id = "CTL0070", .devs = { { "CTL0001" } } },
+ /* Sound Blaster Vibra16CL - added by ctm@ardi.com */
+ { .id = "CTL0080", .devs = { { "CTL0041" } } },
+ /* Sound Blaster 16 'value' PnP. It says model ct4130 on the pcb, */
+ /* but ct4131 on a sticker on the board.. */
+ { .id = "CTL0086", .devs = { { "CTL0041" } } },
+ /* Sound Blaster Vibra16X */
+ { .id = "CTL00f0", .devs = { { "CTL0043" } } },
+ /* Sound Blaster 16 (Virtual PC 2004) */
+ { .id = "tBA03b0", .devs = { {.id="PNPb003" } } },
+#else /* SNDRV_SBAWE defined */
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0035", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0039", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0042", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0043", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ /* Note: This card has also a CTL0051:StereoEnhance device!!! */
+ { .id = "CTL0044", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ /* Note: This card has also a CTL0051:StereoEnhance device!!! */
+ { .id = "CTL0045", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0046", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0047", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0048", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL0054", .devs = { { "CTL0031" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL009a", .devs = { { "CTL0041" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 32 PnP */
+ { .id = "CTL009c", .devs = { { "CTL0041" }, { "CTL0021" } } },
+ /* Sound Blaster 32 PnP */
+ { .id = "CTL009f", .devs = { { "CTL0041" }, { "CTL0021" } } },
+ /* Sound Blaster AWE 64 PnP */
+ { .id = "CTL009d", .devs = { { "CTL0042" }, { "CTL0022" } } },
+ /* Sound Blaster AWE 64 PnP Gold */
+ { .id = "CTL009e", .devs = { { "CTL0044" }, { "CTL0023" } } },
+ /* Sound Blaster AWE 64 PnP Gold */
+ { .id = "CTL00b2", .devs = { { "CTL0044" }, { "CTL0023" } } },
+ /* Sound Blaster AWE 64 PnP */
+ { .id = "CTL00c1", .devs = { { "CTL0042" }, { "CTL0022" } } },
+ /* Sound Blaster AWE 64 PnP */
+ { .id = "CTL00c3", .devs = { { "CTL0045" }, { "CTL0022" } } },
+ /* Sound Blaster AWE 64 PnP */
+ { .id = "CTL00c5", .devs = { { "CTL0045" }, { "CTL0022" } } },
+ /* Sound Blaster AWE 64 PnP */
+ { .id = "CTL00c7", .devs = { { "CTL0045" }, { "CTL0022" } } },
+ /* Sound Blaster AWE 64 PnP */
+ { .id = "CTL00e4", .devs = { { "CTL0045" }, { "CTL0022" } } },
+ /* Sound Blaster AWE 64 PnP */
+ { .id = "CTL00e9", .devs = { { "CTL0045" }, { "CTL0022" } } },
+ /* Sound Blaster 16 PnP (AWE) */
+ { .id = "CTL00ed", .devs = { { "CTL0041" }, { "CTL0070" } } },
+ /* Generic entries */
+ { .id = "CTLXXXX" , .devs = { { "CTL0031" }, { "CTL0021" } } },
+ { .id = "CTLXXXX" , .devs = { { "CTL0041" }, { "CTL0021" } } },
+ { .id = "CTLXXXX" , .devs = { { "CTL0042" }, { "CTL0022" } } },
+ { .id = "CTLXXXX" , .devs = { { "CTL0044" }, { "CTL0023" } } },
+ { .id = "CTLXXXX" , .devs = { { "CTL0045" }, { "CTL0022" } } },
+#endif /* SNDRV_SBAWE */
+ { .id = "", }
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_sb16_pnpids);
+
+#endif /* CONFIG_PNP */
+
+#ifdef SNDRV_SBAWE_EMU8000
+#define DRIVER_NAME "snd-card-sbawe"
+#else
+#define DRIVER_NAME "snd-card-sb16"
+#endif
+
+#ifdef CONFIG_PNP
+
+static int __devinit snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard,
+ struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->dev == NULL)
+ return -ENODEV;
+
+#ifdef SNDRV_SBAWE_EMU8000
+ acard->devwt = pnp_request_card_device(card, id->devs[1].id, acard->dev);
+#endif
+ /* Audio initialization */
+ pdev = acard->dev;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n");
+ return err;
+ }
+ port[dev] = pnp_port_start(pdev, 0);
+ mpu_port[dev] = pnp_port_start(pdev, 1);
+ fm_port[dev] = pnp_port_start(pdev, 2);
+ dma8[dev] = pnp_dma(pdev, 0);
+ dma16[dev] = pnp_dma(pdev, 1);
+ irq[dev] = pnp_irq(pdev, 0);
+ snd_printdd("pnp SB16: port=0x%lx, mpu port=0x%lx, fm port=0x%lx\n",
+ port[dev], mpu_port[dev], fm_port[dev]);
+ snd_printdd("pnp SB16: dma1=%i, dma2=%i, irq=%i\n",
+ dma8[dev], dma16[dev], irq[dev]);
+#ifdef SNDRV_SBAWE_EMU8000
+ /* WaveTable initialization */
+ pdev = acard->devwt;
+ if (pdev != NULL) {
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ goto __wt_error;
+ }
+ awe_port[dev] = pnp_port_start(pdev, 0);
+ snd_printdd("pnp SB16: wavetable port=0x%llx\n",
+ (unsigned long long)pnp_port_start(pdev, 0));
+ } else {
+__wt_error:
+ if (pdev) {
+ pnp_release_card_device(pdev);
+ snd_printk(KERN_ERR PFX "WaveTable pnp configure failure\n");
+ }
+ acard->devwt = NULL;
+ awe_port[dev] = -1;
+ }
+#endif
+ return 0;
+}
+
+#endif /* CONFIG_PNP */
+
+static void snd_sb16_free(struct snd_card *card)
+{
+ struct snd_card_sb16 *acard = card->private_data;
+
+ if (acard == NULL)
+ return;
+ release_and_free_resource(acard->fm_res);
+}
+
+#ifdef CONFIG_PNP
+#define is_isapnp_selected(dev) isapnp[dev]
+#else
+#define is_isapnp_selected(dev) 0
+#endif
+
+static struct snd_card *snd_sb16_card_new(int dev)
+{
+ struct snd_card *card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_card_sb16));
+ if (card == NULL)
+ return NULL;
+ card->private_free = snd_sb16_free;
+ return card;
+}
+
+static int __devinit snd_sb16_probe(struct snd_card *card, int dev)
+{
+ int xirq, xdma8, xdma16;
+ struct snd_sb *chip;
+ struct snd_card_sb16 *acard = card->private_data;
+ struct snd_opl3 *opl3;
+ struct snd_hwdep *synth = NULL;
+#ifdef CONFIG_SND_SB16_CSP
+ struct snd_hwdep *xcsp = NULL;
+#endif
+ unsigned long flags;
+ int err;
+
+ xirq = irq[dev];
+ xdma8 = dma8[dev];
+ xdma16 = dma16[dev];
+
+ if ((err = snd_sbdsp_create(card,
+ port[dev],
+ xirq,
+ snd_sb16dsp_interrupt,
+ xdma8,
+ xdma16,
+ SB_HW_AUTO,
+ &chip)) < 0)
+ return err;
+
+ acard->chip = chip;
+ if (chip->hardware != SB_HW_16) {
+ snd_printk(KERN_ERR PFX "SB 16 chip was not detected at 0x%lx\n", port[dev]);
+ return -ENODEV;
+ }
+ chip->mpu_port = mpu_port[dev];
+ if (! is_isapnp_selected(dev) && (err = snd_sb16dsp_configure(chip)) < 0)
+ return err;
+
+ if ((err = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0)
+ return err;
+
+ strcpy(card->driver,
+#ifdef SNDRV_SBAWE_EMU8000
+ awe_port[dev] > 0 ? "SB AWE" :
+#endif
+ "SB16");
+ strcpy(card->shortname, chip->name);
+ sprintf(card->longname, "%s at 0x%lx, irq %i, dma ",
+ chip->name,
+ chip->port,
+ xirq);
+ if (xdma8 >= 0)
+ sprintf(card->longname + strlen(card->longname), "%d", xdma8);
+ if (xdma16 >= 0)
+ sprintf(card->longname + strlen(card->longname), "%s%d",
+ xdma8 >= 0 ? "&" : "", xdma16);
+
+ if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) {
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SB,
+ chip->mpu_port, 0,
+ xirq, 0, &chip->rmidi)) < 0)
+ return err;
+ chip->rmidi_callback = snd_mpu401_uart_interrupt;
+ }
+
+#ifdef SNDRV_SBAWE_EMU8000
+ if (awe_port[dev] == SNDRV_AUTO_PORT)
+ awe_port[dev] = 0; /* disable */
+#endif
+
+ if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
+ if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2,
+ OPL3_HW_OPL3,
+ acard->fm_res != NULL || fm_port[dev] == port[dev],
+ &opl3) < 0) {
+ snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx\n",
+ fm_port[dev], fm_port[dev] + 2);
+ } else {
+#ifdef SNDRV_SBAWE_EMU8000
+ int seqdev = awe_port[dev] > 0 ? 2 : 1;
+#else
+ int seqdev = 1;
+#endif
+ if ((err = snd_opl3_hwdep_new(opl3, 0, seqdev, &synth)) < 0)
+ return err;
+ }
+ }
+
+ if ((err = snd_sbmixer_new(chip)) < 0)
+ return err;
+
+#ifdef CONFIG_SND_SB16_CSP
+ /* CSP chip on SB16ASP/AWE32 */
+ if ((chip->hardware == SB_HW_16) && csp[dev]) {
+ snd_sb_csp_new(chip, synth != NULL ? 1 : 0, &xcsp);
+ if (xcsp) {
+ chip->csp = xcsp->private_data;
+ chip->hardware = SB_HW_16CSP;
+ } else {
+ snd_printk(KERN_INFO PFX "warning - CSP chip not detected on soundcard #%i\n", dev + 1);
+ }
+ }
+#endif
+#ifdef SNDRV_SBAWE_EMU8000
+ if (awe_port[dev] > 0) {
+ if ((err = snd_emu8000_new(card, 1, awe_port[dev],
+ seq_ports[dev], NULL)) < 0) {
+ snd_printk(KERN_ERR PFX "fatal error - EMU-8000 synthesizer not detected at 0x%lx\n", awe_port[dev]);
+
+ return err;
+ }
+ }
+#endif
+
+ /* setup Mic AGC */
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ snd_sbmixer_write(chip, SB_DSP4_MIC_AGC,
+ (snd_sbmixer_read(chip, SB_DSP4_MIC_AGC) & 0x01) |
+ (mic_agc[dev] ? 0x00 : 0x01));
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+
+ if ((err = snd_card_register(card)) < 0)
+ return err;
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_sb16_suspend(struct snd_card *card, pm_message_t state)
+{
+ struct snd_card_sb16 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_sbmixer_suspend(chip);
+ return 0;
+}
+
+static int snd_sb16_resume(struct snd_card *card)
+{
+ struct snd_card_sb16 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_sbdsp_reset(chip);
+ snd_sbmixer_resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev)
+{
+ struct snd_card_sb16 *acard;
+ struct snd_card *card;
+ int err;
+
+ card = snd_sb16_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+
+ acard = card->private_data;
+ /* non-PnP FM port address is hardwired with base port address */
+ fm_port[dev] = port[dev];
+ /* block the 0x388 port to avoid PnP conflicts */
+ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM");
+#ifdef SNDRV_SBAWE_EMU8000
+ /* non-PnP AWE port address is hardwired with base port address */
+ awe_port[dev] = port[dev] + 0x400;
+#endif
+
+ snd_card_set_dev(card, pdev);
+ if ((err = snd_sb16_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ dev_set_drvdata(pdev, card);
+ return 0;
+}
+
+
+static int __devinit snd_sb16_isa_match(struct device *pdev, unsigned int dev)
+{
+ return enable[dev] && !is_isapnp_selected(dev);
+}
+
+static int __devinit snd_sb16_isa_probe(struct device *pdev, unsigned int dev)
+{
+ int err;
+ static int possible_irqs[] = {5, 9, 10, 7, -1};
+ static int possible_dmas8[] = {1, 3, 0, -1};
+ static int possible_dmas16[] = {5, 6, 7, -1};
+
+ if (irq[dev] == SNDRV_AUTO_IRQ) {
+ if ((irq[dev] = snd_legacy_find_free_irq(possible_irqs)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free IRQ\n");
+ return -EBUSY;
+ }
+ }
+ if (dma8[dev] == SNDRV_AUTO_DMA) {
+ if ((dma8[dev] = snd_legacy_find_free_dma(possible_dmas8)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free 8-bit DMA\n");
+ return -EBUSY;
+ }
+ }
+ if (dma16[dev] == SNDRV_AUTO_DMA) {
+ if ((dma16[dev] = snd_legacy_find_free_dma(possible_dmas16)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free 16-bit DMA\n");
+ return -EBUSY;
+ }
+ }
+
+ if (port[dev] != SNDRV_AUTO_PORT)
+ return snd_sb16_isa_probe1(dev, pdev);
+ else {
+ static int possible_ports[] = {0x220, 0x240, 0x260, 0x280};
+ int i;
+ for (i = 0; i < ARRAY_SIZE(possible_ports); i++) {
+ port[dev] = possible_ports[i];
+ err = snd_sb16_isa_probe1(dev, pdev);
+ if (! err)
+ return 0;
+ }
+ return err;
+ }
+}
+
+static int __devexit snd_sb16_isa_remove(struct device *pdev, unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(pdev));
+ dev_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_sb16_isa_suspend(struct device *dev, unsigned int n,
+ pm_message_t state)
+{
+ return snd_sb16_suspend(dev_get_drvdata(dev), state);
+}
+
+static int snd_sb16_isa_resume(struct device *dev, unsigned int n)
+{
+ return snd_sb16_resume(dev_get_drvdata(dev));
+}
+#endif
+
+#ifdef SNDRV_SBAWE
+#define DEV_NAME "sbawe"
+#else
+#define DEV_NAME "sb16"
+#endif
+
+static struct isa_driver snd_sb16_isa_driver = {
+ .match = snd_sb16_isa_match,
+ .probe = snd_sb16_isa_probe,
+ .remove = __devexit_p(snd_sb16_isa_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_sb16_isa_suspend,
+ .resume = snd_sb16_isa_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+
+#ifdef CONFIG_PNP
+static int __devinit snd_sb16_pnp_detect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ static int dev;
+ struct snd_card *card;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (!enable[dev] || !isapnp[dev])
+ continue;
+ card = snd_sb16_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((res = snd_card_sb16_pnp(dev, card->private_data, pcard, pid)) < 0 ||
+ (res = snd_sb16_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return res;
+ }
+ pnp_set_card_drvdata(pcard, card);
+ dev++;
+ return 0;
+ }
+
+ return -ENODEV;
+}
+
+static void __devexit snd_sb16_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_sb16_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state)
+{
+ return snd_sb16_suspend(pnp_get_card_drvdata(pcard), state);
+}
+static int snd_sb16_pnp_resume(struct pnp_card_link *pcard)
+{
+ return snd_sb16_resume(pnp_get_card_drvdata(pcard));
+}
+#endif
+
+static struct pnp_card_driver sb16_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+#ifdef SNDRV_SBAWE
+ .name = "sbawe",
+#else
+ .name = "sb16",
+#endif
+ .id_table = snd_sb16_pnpids,
+ .probe = snd_sb16_pnp_detect,
+ .remove = __devexit_p(snd_sb16_pnp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_sb16_pnp_suspend,
+ .resume = snd_sb16_pnp_resume,
+#endif
+};
+
+#endif /* CONFIG_PNP */
+
+static int __init alsa_card_sb16_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&snd_sb16_isa_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+
+ err = pnp_register_card_driver(&sb16_pnpc_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ if (isa_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit alsa_card_sb16_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnp_registered)
+ pnp_unregister_card_driver(&sb16_pnpc_driver);
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&snd_sb16_isa_driver);
+}
+
+module_init(alsa_card_sb16_init)
+module_exit(alsa_card_sb16_exit)
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
new file mode 100644
index 0000000..49037d0
--- /dev/null
+++ b/sound/isa/sb/sb16_csp.c
@@ -0,0 +1,1200 @@
+/*
+ * Copyright (c) 1999 by Uros Bizjak <uros@kss-loka.si>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * SB16ASP/AWE32 CSP control
+ *
+ * CSP microcode loader:
+ * alsa-tools/sb16_csp/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/sb16_csp.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Uros Bizjak <uros@kss-loka.si>");
+MODULE_DESCRIPTION("ALSA driver for SB16 Creative Signal Processor");
+MODULE_LICENSE("GPL");
+MODULE_FIRMWARE("sb16/mulaw_main.csp");
+MODULE_FIRMWARE("sb16/alaw_main.csp");
+MODULE_FIRMWARE("sb16/ima_adpcm_init.csp");
+MODULE_FIRMWARE("sb16/ima_adpcm_playback.csp");
+MODULE_FIRMWARE("sb16/ima_adpcm_capture.csp");
+
+#ifdef SNDRV_LITTLE_ENDIAN
+#define CSP_HDR_VALUE(a,b,c,d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24))
+#else
+#define CSP_HDR_VALUE(a,b,c,d) ((d) | ((c)<<8) | ((b)<<16) | ((a)<<24))
+#endif
+
+#define RIFF_HEADER CSP_HDR_VALUE('R', 'I', 'F', 'F')
+#define CSP__HEADER CSP_HDR_VALUE('C', 'S', 'P', ' ')
+#define LIST_HEADER CSP_HDR_VALUE('L', 'I', 'S', 'T')
+#define FUNC_HEADER CSP_HDR_VALUE('f', 'u', 'n', 'c')
+#define CODE_HEADER CSP_HDR_VALUE('c', 'o', 'd', 'e')
+#define INIT_HEADER CSP_HDR_VALUE('i', 'n', 'i', 't')
+#define MAIN_HEADER CSP_HDR_VALUE('m', 'a', 'i', 'n')
+
+/*
+ * RIFF data format
+ */
+struct riff_header {
+ __u32 name;
+ __u32 len;
+};
+
+struct desc_header {
+ struct riff_header info;
+ __u16 func_nr;
+ __u16 VOC_type;
+ __u16 flags_play_rec;
+ __u16 flags_16bit_8bit;
+ __u16 flags_stereo_mono;
+ __u16 flags_rates;
+};
+
+/*
+ * prototypes
+ */
+static void snd_sb_csp_free(struct snd_hwdep *hw);
+static int snd_sb_csp_open(struct snd_hwdep * hw, struct file *file);
+static int snd_sb_csp_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg);
+static int snd_sb_csp_release(struct snd_hwdep * hw, struct file *file);
+
+static int csp_detect(struct snd_sb *chip, int *version);
+static int set_codec_parameter(struct snd_sb *chip, unsigned char par, unsigned char val);
+static int set_register(struct snd_sb *chip, unsigned char reg, unsigned char val);
+static int read_register(struct snd_sb *chip, unsigned char reg);
+static int set_mode_register(struct snd_sb *chip, unsigned char mode);
+static int get_version(struct snd_sb *chip);
+
+static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
+ struct snd_sb_csp_microcode __user * code);
+static int snd_sb_csp_unload(struct snd_sb_csp * p);
+static int snd_sb_csp_load_user(struct snd_sb_csp * p, const unsigned char __user *buf, int size, int load_flags);
+static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode);
+static int snd_sb_csp_check_version(struct snd_sb_csp * p);
+
+static int snd_sb_csp_use(struct snd_sb_csp * p);
+static int snd_sb_csp_unuse(struct snd_sb_csp * p);
+static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channels);
+static int snd_sb_csp_stop(struct snd_sb_csp * p);
+static int snd_sb_csp_pause(struct snd_sb_csp * p);
+static int snd_sb_csp_restart(struct snd_sb_csp * p);
+
+static int snd_sb_qsound_build(struct snd_sb_csp * p);
+static void snd_sb_qsound_destroy(struct snd_sb_csp * p);
+static int snd_sb_csp_qsound_transfer(struct snd_sb_csp * p);
+
+static int init_proc_entry(struct snd_sb_csp * p, int device);
+static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer);
+
+/*
+ * Detect CSP chip and create a new instance
+ */
+int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep)
+{
+ struct snd_sb_csp *p;
+ int uninitialized_var(version);
+ int err;
+ struct snd_hwdep *hw;
+
+ if (rhwdep)
+ *rhwdep = NULL;
+
+ if (csp_detect(chip, &version))
+ return -ENODEV;
+
+ if ((err = snd_hwdep_new(chip->card, "SB16-CSP", device, &hw)) < 0)
+ return err;
+
+ if ((p = kzalloc(sizeof(*p), GFP_KERNEL)) == NULL) {
+ snd_device_free(chip->card, hw);
+ return -ENOMEM;
+ }
+ p->chip = chip;
+ p->version = version;
+
+ /* CSP operators */
+ p->ops.csp_use = snd_sb_csp_use;
+ p->ops.csp_unuse = snd_sb_csp_unuse;
+ p->ops.csp_autoload = snd_sb_csp_autoload;
+ p->ops.csp_start = snd_sb_csp_start;
+ p->ops.csp_stop = snd_sb_csp_stop;
+ p->ops.csp_qsound_transfer = snd_sb_csp_qsound_transfer;
+
+ mutex_init(&p->access_mutex);
+ sprintf(hw->name, "CSP v%d.%d", (version >> 4), (version & 0x0f));
+ hw->iface = SNDRV_HWDEP_IFACE_SB16CSP;
+ hw->private_data = p;
+ hw->private_free = snd_sb_csp_free;
+
+ /* operators - only write/ioctl */
+ hw->ops.open = snd_sb_csp_open;
+ hw->ops.ioctl = snd_sb_csp_ioctl;
+ hw->ops.release = snd_sb_csp_release;
+
+ /* create a proc entry */
+ init_proc_entry(p, device);
+ if (rhwdep)
+ *rhwdep = hw;
+ return 0;
+}
+
+/*
+ * free_private for hwdep instance
+ */
+static void snd_sb_csp_free(struct snd_hwdep *hwdep)
+{
+ int i;
+ struct snd_sb_csp *p = hwdep->private_data;
+ if (p) {
+ if (p->running & SNDRV_SB_CSP_ST_RUNNING)
+ snd_sb_csp_stop(p);
+ for (i = 0; i < ARRAY_SIZE(p->csp_programs); ++i)
+ release_firmware(p->csp_programs[i]);
+ kfree(p);
+ }
+}
+
+/* ------------------------------ */
+
+/*
+ * open the device exclusively
+ */
+static int snd_sb_csp_open(struct snd_hwdep * hw, struct file *file)
+{
+ struct snd_sb_csp *p = hw->private_data;
+ return (snd_sb_csp_use(p));
+}
+
+/*
+ * ioctl for hwdep device:
+ */
+static int snd_sb_csp_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_sb_csp *p = hw->private_data;
+ struct snd_sb_csp_info info;
+ struct snd_sb_csp_start start_info;
+ int err;
+
+ if (snd_BUG_ON(!p))
+ return -EINVAL;
+
+ if (snd_sb_csp_check_version(p))
+ return -ENODEV;
+
+ switch (cmd) {
+ /* get information */
+ case SNDRV_SB_CSP_IOCTL_INFO:
+ *info.codec_name = *p->codec_name;
+ info.func_nr = p->func_nr;
+ info.acc_format = p->acc_format;
+ info.acc_channels = p->acc_channels;
+ info.acc_width = p->acc_width;
+ info.acc_rates = p->acc_rates;
+ info.csp_mode = p->mode;
+ info.run_channels = p->run_channels;
+ info.run_width = p->run_width;
+ info.version = p->version;
+ info.state = p->running;
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ err = -EFAULT;
+ else
+ err = 0;
+ break;
+
+ /* load CSP microcode */
+ case SNDRV_SB_CSP_IOCTL_LOAD_CODE:
+ err = (p->running & SNDRV_SB_CSP_ST_RUNNING ?
+ -EBUSY : snd_sb_csp_riff_load(p, (struct snd_sb_csp_microcode __user *) arg));
+ break;
+ case SNDRV_SB_CSP_IOCTL_UNLOAD_CODE:
+ err = (p->running & SNDRV_SB_CSP_ST_RUNNING ?
+ -EBUSY : snd_sb_csp_unload(p));
+ break;
+
+ /* change CSP running state */
+ case SNDRV_SB_CSP_IOCTL_START:
+ if (copy_from_user(&start_info, (void __user *) arg, sizeof(start_info)))
+ err = -EFAULT;
+ else
+ err = snd_sb_csp_start(p, start_info.sample_width, start_info.channels);
+ break;
+ case SNDRV_SB_CSP_IOCTL_STOP:
+ err = snd_sb_csp_stop(p);
+ break;
+ case SNDRV_SB_CSP_IOCTL_PAUSE:
+ err = snd_sb_csp_pause(p);
+ break;
+ case SNDRV_SB_CSP_IOCTL_RESTART:
+ err = snd_sb_csp_restart(p);
+ break;
+ default:
+ err = -ENOTTY;
+ break;
+ }
+
+ return err;
+}
+
+/*
+ * close the device
+ */
+static int snd_sb_csp_release(struct snd_hwdep * hw, struct file *file)
+{
+ struct snd_sb_csp *p = hw->private_data;
+ return (snd_sb_csp_unuse(p));
+}
+
+/* ------------------------------ */
+
+/*
+ * acquire device
+ */
+static int snd_sb_csp_use(struct snd_sb_csp * p)
+{
+ mutex_lock(&p->access_mutex);
+ if (p->used) {
+ mutex_unlock(&p->access_mutex);
+ return -EAGAIN;
+ }
+ p->used++;
+ mutex_unlock(&p->access_mutex);
+
+ return 0;
+
+}
+
+/*
+ * release device
+ */
+static int snd_sb_csp_unuse(struct snd_sb_csp * p)
+{
+ mutex_lock(&p->access_mutex);
+ p->used--;
+ mutex_unlock(&p->access_mutex);
+
+ return 0;
+}
+
+/*
+ * load microcode via ioctl:
+ * code is user-space pointer
+ */
+static int snd_sb_csp_riff_load(struct snd_sb_csp * p,
+ struct snd_sb_csp_microcode __user * mcode)
+{
+ struct snd_sb_csp_mc_header info;
+
+ unsigned char __user *data_ptr;
+ unsigned char __user *data_end;
+ unsigned short func_nr = 0;
+
+ struct riff_header file_h, item_h, code_h;
+ __u32 item_type;
+ struct desc_header funcdesc_h;
+
+ unsigned long flags;
+ int err;
+
+ if (copy_from_user(&info, mcode, sizeof(info)))
+ return -EFAULT;
+ data_ptr = mcode->data;
+
+ if (copy_from_user(&file_h, data_ptr, sizeof(file_h)))
+ return -EFAULT;
+ if ((file_h.name != RIFF_HEADER) ||
+ (le32_to_cpu(file_h.len) >= SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE - sizeof(file_h))) {
+ snd_printd("%s: Invalid RIFF header\n", __func__);
+ return -EINVAL;
+ }
+ data_ptr += sizeof(file_h);
+ data_end = data_ptr + le32_to_cpu(file_h.len);
+
+ if (copy_from_user(&item_type, data_ptr, sizeof(item_type)))
+ return -EFAULT;
+ if (item_type != CSP__HEADER) {
+ snd_printd("%s: Invalid RIFF file type\n", __func__);
+ return -EINVAL;
+ }
+ data_ptr += sizeof (item_type);
+
+ for (; data_ptr < data_end; data_ptr += le32_to_cpu(item_h.len)) {
+ if (copy_from_user(&item_h, data_ptr, sizeof(item_h)))
+ return -EFAULT;
+ data_ptr += sizeof(item_h);
+ if (item_h.name != LIST_HEADER)
+ continue;
+
+ if (copy_from_user(&item_type, data_ptr, sizeof(item_type)))
+ return -EFAULT;
+ switch (item_type) {
+ case FUNC_HEADER:
+ if (copy_from_user(&funcdesc_h, data_ptr + sizeof(item_type), sizeof(funcdesc_h)))
+ return -EFAULT;
+ func_nr = le16_to_cpu(funcdesc_h.func_nr);
+ break;
+ case CODE_HEADER:
+ if (func_nr != info.func_req)
+ break; /* not required function, try next */
+ data_ptr += sizeof(item_type);
+
+ /* destroy QSound mixer element */
+ if (p->mode == SNDRV_SB_CSP_MODE_QSOUND) {
+ snd_sb_qsound_destroy(p);
+ }
+ /* Clear all flags */
+ p->running = 0;
+ p->mode = 0;
+
+ /* load microcode blocks */
+ for (;;) {
+ if (data_ptr >= data_end)
+ return -EINVAL;
+ if (copy_from_user(&code_h, data_ptr, sizeof(code_h)))
+ return -EFAULT;
+
+ /* init microcode blocks */
+ if (code_h.name != INIT_HEADER)
+ break;
+ data_ptr += sizeof(code_h);
+ err = snd_sb_csp_load_user(p, data_ptr, le32_to_cpu(code_h.len),
+ SNDRV_SB_CSP_LOAD_INITBLOCK);
+ if (err)
+ return err;
+ data_ptr += le32_to_cpu(code_h.len);
+ }
+ /* main microcode block */
+ if (copy_from_user(&code_h, data_ptr, sizeof(code_h)))
+ return -EFAULT;
+
+ if (code_h.name != MAIN_HEADER) {
+ snd_printd("%s: Missing 'main' microcode\n", __func__);
+ return -EINVAL;
+ }
+ data_ptr += sizeof(code_h);
+ err = snd_sb_csp_load_user(p, data_ptr,
+ le32_to_cpu(code_h.len), 0);
+ if (err)
+ return err;
+
+ /* fill in codec header */
+ strlcpy(p->codec_name, info.codec_name, sizeof(p->codec_name));
+ p->func_nr = func_nr;
+ p->mode = le16_to_cpu(funcdesc_h.flags_play_rec);
+ switch (le16_to_cpu(funcdesc_h.VOC_type)) {
+ case 0x0001: /* QSound decoder */
+ if (le16_to_cpu(funcdesc_h.flags_play_rec) == SNDRV_SB_CSP_MODE_DSP_WRITE) {
+ if (snd_sb_qsound_build(p) == 0)
+ /* set QSound flag and clear all other mode flags */
+ p->mode = SNDRV_SB_CSP_MODE_QSOUND;
+ }
+ p->acc_format = 0;
+ break;
+ case 0x0006: /* A Law codec */
+ p->acc_format = SNDRV_PCM_FMTBIT_A_LAW;
+ break;
+ case 0x0007: /* Mu Law codec */
+ p->acc_format = SNDRV_PCM_FMTBIT_MU_LAW;
+ break;
+ case 0x0011: /* what Creative thinks is IMA ADPCM codec */
+ case 0x0200: /* Creative ADPCM codec */
+ p->acc_format = SNDRV_PCM_FMTBIT_IMA_ADPCM;
+ break;
+ case 201: /* Text 2 Speech decoder */
+ /* TODO: Text2Speech handling routines */
+ p->acc_format = 0;
+ break;
+ case 0x0202: /* Fast Speech 8 codec */
+ case 0x0203: /* Fast Speech 10 codec */
+ p->acc_format = SNDRV_PCM_FMTBIT_SPECIAL;
+ break;
+ default: /* other codecs are unsupported */
+ p->acc_format = p->acc_width = p->acc_rates = 0;
+ p->mode = 0;
+ snd_printd("%s: Unsupported CSP codec type: 0x%04x\n",
+ __func__,
+ le16_to_cpu(funcdesc_h.VOC_type));
+ return -EINVAL;
+ }
+ p->acc_channels = le16_to_cpu(funcdesc_h.flags_stereo_mono);
+ p->acc_width = le16_to_cpu(funcdesc_h.flags_16bit_8bit);
+ p->acc_rates = le16_to_cpu(funcdesc_h.flags_rates);
+
+ /* Decouple CSP from IRQ and DMAREQ lines */
+ spin_lock_irqsave(&p->chip->reg_lock, flags);
+ set_mode_register(p->chip, 0xfc);
+ set_mode_register(p->chip, 0x00);
+ spin_unlock_irqrestore(&p->chip->reg_lock, flags);
+
+ /* finished loading successfully */
+ p->running = SNDRV_SB_CSP_ST_LOADED; /* set LOADED flag */
+ return 0;
+ }
+ }
+ snd_printd("%s: Function #%d not found\n", __func__, info.func_req);
+ return -EINVAL;
+}
+
+/*
+ * unload CSP microcode
+ */
+static int snd_sb_csp_unload(struct snd_sb_csp * p)
+{
+ if (p->running & SNDRV_SB_CSP_ST_RUNNING)
+ return -EBUSY;
+ if (!(p->running & SNDRV_SB_CSP_ST_LOADED))
+ return -ENXIO;
+
+ /* clear supported formats */
+ p->acc_format = 0;
+ p->acc_channels = p->acc_width = p->acc_rates = 0;
+ /* destroy QSound mixer element */
+ if (p->mode == SNDRV_SB_CSP_MODE_QSOUND) {
+ snd_sb_qsound_destroy(p);
+ }
+ /* clear all flags */
+ p->running = 0;
+ p->mode = 0;
+ return 0;
+}
+
+/*
+ * send command sequence to DSP
+ */
+static inline int command_seq(struct snd_sb *chip, const unsigned char *seq, int size)
+{
+ int i;
+ for (i = 0; i < size; i++) {
+ if (!snd_sbdsp_command(chip, seq[i]))
+ return -EIO;
+ }
+ return 0;
+}
+
+/*
+ * set CSP codec parameter
+ */
+static int set_codec_parameter(struct snd_sb *chip, unsigned char par, unsigned char val)
+{
+ unsigned char dsp_cmd[3];
+
+ dsp_cmd[0] = 0x05; /* CSP set codec parameter */
+ dsp_cmd[1] = val; /* Parameter value */
+ dsp_cmd[2] = par; /* Parameter */
+ command_seq(chip, dsp_cmd, 3);
+ snd_sbdsp_command(chip, 0x03); /* DSP read? */
+ if (snd_sbdsp_get_byte(chip) != par)
+ return -EIO;
+ return 0;
+}
+
+/*
+ * set CSP register
+ */
+static int set_register(struct snd_sb *chip, unsigned char reg, unsigned char val)
+{
+ unsigned char dsp_cmd[3];
+
+ dsp_cmd[0] = 0x0e; /* CSP set register */
+ dsp_cmd[1] = reg; /* CSP Register */
+ dsp_cmd[2] = val; /* value */
+ return command_seq(chip, dsp_cmd, 3);
+}
+
+/*
+ * read CSP register
+ * return < 0 -> error
+ */
+static int read_register(struct snd_sb *chip, unsigned char reg)
+{
+ unsigned char dsp_cmd[2];
+
+ dsp_cmd[0] = 0x0f; /* CSP read register */
+ dsp_cmd[1] = reg; /* CSP Register */
+ command_seq(chip, dsp_cmd, 2);
+ return snd_sbdsp_get_byte(chip); /* Read DSP value */
+}
+
+/*
+ * set CSP mode register
+ */
+static int set_mode_register(struct snd_sb *chip, unsigned char mode)
+{
+ unsigned char dsp_cmd[2];
+
+ dsp_cmd[0] = 0x04; /* CSP set mode register */
+ dsp_cmd[1] = mode; /* mode */
+ return command_seq(chip, dsp_cmd, 2);
+}
+
+/*
+ * Detect CSP
+ * return 0 if CSP exists.
+ */
+static int csp_detect(struct snd_sb *chip, int *version)
+{
+ unsigned char csp_test1, csp_test2;
+ unsigned long flags;
+ int result = -ENODEV;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ set_codec_parameter(chip, 0x00, 0x00);
+ set_mode_register(chip, 0xfc); /* 0xfc = ?? */
+
+ csp_test1 = read_register(chip, 0x83);
+ set_register(chip, 0x83, ~csp_test1);
+ csp_test2 = read_register(chip, 0x83);
+ if (csp_test2 != (csp_test1 ^ 0xff))
+ goto __fail;
+
+ set_register(chip, 0x83, csp_test1);
+ csp_test2 = read_register(chip, 0x83);
+ if (csp_test2 != csp_test1)
+ goto __fail;
+
+ set_mode_register(chip, 0x00); /* 0x00 = ? */
+
+ *version = get_version(chip);
+ snd_sbdsp_reset(chip); /* reset DSP after getversion! */
+ if (*version >= 0x10 && *version <= 0x1f)
+ result = 0; /* valid version id */
+
+ __fail:
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return result;
+}
+
+/*
+ * get CSP version number
+ */
+static int get_version(struct snd_sb *chip)
+{
+ unsigned char dsp_cmd[2];
+
+ dsp_cmd[0] = 0x08; /* SB_DSP_!something! */
+ dsp_cmd[1] = 0x03; /* get chip version id? */
+ command_seq(chip, dsp_cmd, 2);
+
+ return (snd_sbdsp_get_byte(chip));
+}
+
+/*
+ * check if the CSP version is valid
+ */
+static int snd_sb_csp_check_version(struct snd_sb_csp * p)
+{
+ if (p->version < 0x10 || p->version > 0x1f) {
+ snd_printd("%s: Invalid CSP version: 0x%x\n", __func__, p->version);
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * download microcode to CSP (microcode should have one "main" block).
+ */
+static int snd_sb_csp_load(struct snd_sb_csp * p, const unsigned char *buf, int size, int load_flags)
+{
+ int status, i;
+ int err;
+ int result = -EIO;
+ unsigned long flags;
+
+ spin_lock_irqsave(&p->chip->reg_lock, flags);
+ snd_sbdsp_command(p->chip, 0x01); /* CSP download command */
+ if (snd_sbdsp_get_byte(p->chip)) {
+ snd_printd("%s: Download command failed\n", __func__);
+ goto __fail;
+ }
+ /* Send CSP low byte (size - 1) */
+ snd_sbdsp_command(p->chip, (unsigned char)(size - 1));
+ /* Send high byte */
+ snd_sbdsp_command(p->chip, (unsigned char)((size - 1) >> 8));
+ /* send microcode sequence */
+ /* load from kernel space */
+ while (size--) {
+ if (!snd_sbdsp_command(p->chip, *buf++))
+ goto __fail;
+ }
+ if (snd_sbdsp_get_byte(p->chip))
+ goto __fail;
+
+ if (load_flags & SNDRV_SB_CSP_LOAD_INITBLOCK) {
+ i = 0;
+ /* some codecs (FastSpeech) take some time to initialize */
+ while (1) {
+ snd_sbdsp_command(p->chip, 0x03);
+ status = snd_sbdsp_get_byte(p->chip);
+ if (status == 0x55 || ++i >= 10)
+ break;
+ udelay (10);
+ }
+ if (status != 0x55) {
+ snd_printd("%s: Microcode initialization failed\n", __func__);
+ goto __fail;
+ }
+ } else {
+ /*
+ * Read mixer register SB_DSP4_DMASETUP after loading 'main' code.
+ * Start CSP chip if no 16bit DMA channel is set - some kind
+ * of autorun or perhaps a bugfix?
+ */
+ spin_lock(&p->chip->mixer_lock);
+ status = snd_sbmixer_read(p->chip, SB_DSP4_DMASETUP);
+ spin_unlock(&p->chip->mixer_lock);
+ if (!(status & (SB_DMASETUP_DMA7 | SB_DMASETUP_DMA6 | SB_DMASETUP_DMA5))) {
+ err = (set_codec_parameter(p->chip, 0xaa, 0x00) ||
+ set_codec_parameter(p->chip, 0xff, 0x00));
+ snd_sbdsp_reset(p->chip); /* really! */
+ if (err)
+ goto __fail;
+ set_mode_register(p->chip, 0xc0); /* c0 = STOP */
+ set_mode_register(p->chip, 0x70); /* 70 = RUN */
+ }
+ }
+ result = 0;
+
+ __fail:
+ spin_unlock_irqrestore(&p->chip->reg_lock, flags);
+ return result;
+}
+
+static int snd_sb_csp_load_user(struct snd_sb_csp * p, const unsigned char __user *buf, int size, int load_flags)
+{
+ int err = -ENOMEM;
+ unsigned char *kbuf = kmalloc(size, GFP_KERNEL);
+ if (kbuf) {
+ if (copy_from_user(kbuf, buf, size))
+ err = -EFAULT;
+ else
+ err = snd_sb_csp_load(p, kbuf, size, load_flags);
+ kfree(kbuf);
+ }
+ return err;
+}
+
+static int snd_sb_csp_firmware_load(struct snd_sb_csp *p, int index, int flags)
+{
+ static const char *const names[] = {
+ "sb16/mulaw_main.csp",
+ "sb16/alaw_main.csp",
+ "sb16/ima_adpcm_init.csp",
+ "sb16/ima_adpcm_playback.csp",
+ "sb16/ima_adpcm_capture.csp",
+ };
+ const struct firmware *program;
+
+ BUILD_BUG_ON(ARRAY_SIZE(names) != CSP_PROGRAM_COUNT);
+ program = p->csp_programs[index];
+ if (!program) {
+ int err = request_firmware(&program, names[index],
+ p->chip->card->dev);
+ if (err < 0)
+ return err;
+ p->csp_programs[index] = program;
+ }
+ return snd_sb_csp_load(p, program->data, program->size, flags);
+}
+
+/*
+ * autoload hardware codec if necessary
+ * return 0 if CSP is loaded and ready to run (p->running != 0)
+ */
+static int snd_sb_csp_autoload(struct snd_sb_csp * p, int pcm_sfmt, int play_rec_mode)
+{
+ unsigned long flags;
+ int err = 0;
+
+ /* if CSP is running or manually loaded then exit */
+ if (p->running & (SNDRV_SB_CSP_ST_RUNNING | SNDRV_SB_CSP_ST_LOADED))
+ return -EBUSY;
+
+ /* autoload microcode only if requested hardware codec is not already loaded */
+ if (((1 << pcm_sfmt) & p->acc_format) && (play_rec_mode & p->mode)) {
+ p->running = SNDRV_SB_CSP_ST_AUTO;
+ } else {
+ switch (pcm_sfmt) {
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ err = snd_sb_csp_firmware_load(p, CSP_PROGRAM_MULAW, 0);
+ p->acc_format = SNDRV_PCM_FMTBIT_MU_LAW;
+ p->mode = SNDRV_SB_CSP_MODE_DSP_READ | SNDRV_SB_CSP_MODE_DSP_WRITE;
+ break;
+ case SNDRV_PCM_FORMAT_A_LAW:
+ err = snd_sb_csp_firmware_load(p, CSP_PROGRAM_ALAW, 0);
+ p->acc_format = SNDRV_PCM_FMTBIT_A_LAW;
+ p->mode = SNDRV_SB_CSP_MODE_DSP_READ | SNDRV_SB_CSP_MODE_DSP_WRITE;
+ break;
+ case SNDRV_PCM_FORMAT_IMA_ADPCM:
+ err = snd_sb_csp_firmware_load(p, CSP_PROGRAM_ADPCM_INIT,
+ SNDRV_SB_CSP_LOAD_INITBLOCK);
+ if (err)
+ break;
+ if (play_rec_mode == SNDRV_SB_CSP_MODE_DSP_WRITE) {
+ err = snd_sb_csp_firmware_load
+ (p, CSP_PROGRAM_ADPCM_PLAYBACK, 0);
+ p->mode = SNDRV_SB_CSP_MODE_DSP_WRITE;
+ } else {
+ err = snd_sb_csp_firmware_load
+ (p, CSP_PROGRAM_ADPCM_CAPTURE, 0);
+ p->mode = SNDRV_SB_CSP_MODE_DSP_READ;
+ }
+ p->acc_format = SNDRV_PCM_FMTBIT_IMA_ADPCM;
+ break;
+ default:
+ /* Decouple CSP from IRQ and DMAREQ lines */
+ if (p->running & SNDRV_SB_CSP_ST_AUTO) {
+ spin_lock_irqsave(&p->chip->reg_lock, flags);
+ set_mode_register(p->chip, 0xfc);
+ set_mode_register(p->chip, 0x00);
+ spin_unlock_irqrestore(&p->chip->reg_lock, flags);
+ p->running = 0; /* clear autoloaded flag */
+ }
+ return -EINVAL;
+ }
+ if (err) {
+ p->acc_format = 0;
+ p->acc_channels = p->acc_width = p->acc_rates = 0;
+
+ p->running = 0; /* clear autoloaded flag */
+ p->mode = 0;
+ return (err);
+ } else {
+ p->running = SNDRV_SB_CSP_ST_AUTO; /* set autoloaded flag */
+ p->acc_width = SNDRV_SB_CSP_SAMPLE_16BIT; /* only 16 bit data */
+ p->acc_channels = SNDRV_SB_CSP_MONO | SNDRV_SB_CSP_STEREO;
+ p->acc_rates = SNDRV_SB_CSP_RATE_ALL; /* HW codecs accept all rates */
+ }
+
+ }
+ return (p->running & SNDRV_SB_CSP_ST_AUTO) ? 0 : -ENXIO;
+}
+
+/*
+ * start CSP
+ */
+static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channels)
+{
+ unsigned char s_type; /* sample type */
+ unsigned char mixL, mixR;
+ int result = -EIO;
+ unsigned long flags;
+
+ if (!(p->running & (SNDRV_SB_CSP_ST_LOADED | SNDRV_SB_CSP_ST_AUTO))) {
+ snd_printd("%s: Microcode not loaded\n", __func__);
+ return -ENXIO;
+ }
+ if (p->running & SNDRV_SB_CSP_ST_RUNNING) {
+ snd_printd("%s: CSP already running\n", __func__);
+ return -EBUSY;
+ }
+ if (!(sample_width & p->acc_width)) {
+ snd_printd("%s: Unsupported PCM sample width\n", __func__);
+ return -EINVAL;
+ }
+ if (!(channels & p->acc_channels)) {
+ snd_printd("%s: Invalid number of channels\n", __func__);
+ return -EINVAL;
+ }
+
+ /* Mute PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
+ mixL = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV);
+ mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+
+ spin_lock(&p->chip->reg_lock);
+ set_mode_register(p->chip, 0xc0); /* c0 = STOP */
+ set_mode_register(p->chip, 0x70); /* 70 = RUN */
+
+ s_type = 0x00;
+ if (channels == SNDRV_SB_CSP_MONO)
+ s_type = 0x11; /* 000n 000n (n = 1 if mono) */
+ if (sample_width == SNDRV_SB_CSP_SAMPLE_8BIT)
+ s_type |= 0x22; /* 00dX 00dX (d = 1 if 8 bit samples) */
+
+ if (set_codec_parameter(p->chip, 0x81, s_type)) {
+ snd_printd("%s: Set sample type command failed\n", __func__);
+ goto __fail;
+ }
+ if (set_codec_parameter(p->chip, 0x80, 0x00)) {
+ snd_printd("%s: Codec start command failed\n", __func__);
+ goto __fail;
+ }
+ p->run_width = sample_width;
+ p->run_channels = channels;
+
+ p->running |= SNDRV_SB_CSP_ST_RUNNING;
+
+ if (p->mode & SNDRV_SB_CSP_MODE_QSOUND) {
+ set_codec_parameter(p->chip, 0xe0, 0x01);
+ /* enable QSound decoder */
+ set_codec_parameter(p->chip, 0x00, 0xff);
+ set_codec_parameter(p->chip, 0x01, 0xff);
+ p->running |= SNDRV_SB_CSP_ST_QSOUND;
+ /* set QSound startup value */
+ snd_sb_csp_qsound_transfer(p);
+ }
+ result = 0;
+
+ __fail:
+ spin_unlock(&p->chip->reg_lock);
+
+ /* restore PCM volume */
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
+
+ return result;
+}
+
+/*
+ * stop CSP
+ */
+static int snd_sb_csp_stop(struct snd_sb_csp * p)
+{
+ int result;
+ unsigned char mixL, mixR;
+ unsigned long flags;
+
+ if (!(p->running & SNDRV_SB_CSP_ST_RUNNING))
+ return 0;
+
+ /* Mute PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
+ mixL = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV);
+ mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+
+ spin_lock(&p->chip->reg_lock);
+ if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
+ set_codec_parameter(p->chip, 0xe0, 0x01);
+ /* disable QSound decoder */
+ set_codec_parameter(p->chip, 0x00, 0x00);
+ set_codec_parameter(p->chip, 0x01, 0x00);
+
+ p->running &= ~SNDRV_SB_CSP_ST_QSOUND;
+ }
+ result = set_mode_register(p->chip, 0xc0); /* c0 = STOP */
+ spin_unlock(&p->chip->reg_lock);
+
+ /* restore PCM volume */
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
+ snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
+
+ if (!(result))
+ p->running &= ~(SNDRV_SB_CSP_ST_PAUSED | SNDRV_SB_CSP_ST_RUNNING);
+ return result;
+}
+
+/*
+ * pause CSP codec and hold DMA transfer
+ */
+static int snd_sb_csp_pause(struct snd_sb_csp * p)
+{
+ int result;
+ unsigned long flags;
+
+ if (!(p->running & SNDRV_SB_CSP_ST_RUNNING))
+ return -EBUSY;
+
+ spin_lock_irqsave(&p->chip->reg_lock, flags);
+ result = set_codec_parameter(p->chip, 0x80, 0xff);
+ spin_unlock_irqrestore(&p->chip->reg_lock, flags);
+ if (!(result))
+ p->running |= SNDRV_SB_CSP_ST_PAUSED;
+
+ return result;
+}
+
+/*
+ * restart CSP codec and resume DMA transfer
+ */
+static int snd_sb_csp_restart(struct snd_sb_csp * p)
+{
+ int result;
+ unsigned long flags;
+
+ if (!(p->running & SNDRV_SB_CSP_ST_PAUSED))
+ return -EBUSY;
+
+ spin_lock_irqsave(&p->chip->reg_lock, flags);
+ result = set_codec_parameter(p->chip, 0x80, 0x00);
+ spin_unlock_irqrestore(&p->chip->reg_lock, flags);
+ if (!(result))
+ p->running &= ~SNDRV_SB_CSP_ST_PAUSED;
+
+ return result;
+}
+
+/* ------------------------------ */
+
+/*
+ * QSound mixer control for PCM
+ */
+
+#define snd_sb_qsound_switch_info snd_ctl_boolean_mono_info
+
+static int snd_sb_qsound_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb_csp *p = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = p->q_enabled ? 1 : 0;
+ return 0;
+}
+
+static int snd_sb_qsound_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb_csp *p = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned char nval;
+
+ nval = ucontrol->value.integer.value[0] & 0x01;
+ spin_lock_irqsave(&p->q_lock, flags);
+ change = p->q_enabled != nval;
+ p->q_enabled = nval;
+ spin_unlock_irqrestore(&p->q_lock, flags);
+ return change;
+}
+
+static int snd_sb_qsound_space_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = SNDRV_SB_CSP_QSOUND_MAX_RIGHT;
+ return 0;
+}
+
+static int snd_sb_qsound_space_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb_csp *p = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&p->q_lock, flags);
+ ucontrol->value.integer.value[0] = p->qpos_left;
+ ucontrol->value.integer.value[1] = p->qpos_right;
+ spin_unlock_irqrestore(&p->q_lock, flags);
+ return 0;
+}
+
+static int snd_sb_qsound_space_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb_csp *p = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned char nval1, nval2;
+
+ nval1 = ucontrol->value.integer.value[0];
+ if (nval1 > SNDRV_SB_CSP_QSOUND_MAX_RIGHT)
+ nval1 = SNDRV_SB_CSP_QSOUND_MAX_RIGHT;
+ nval2 = ucontrol->value.integer.value[1];
+ if (nval2 > SNDRV_SB_CSP_QSOUND_MAX_RIGHT)
+ nval2 = SNDRV_SB_CSP_QSOUND_MAX_RIGHT;
+ spin_lock_irqsave(&p->q_lock, flags);
+ change = p->qpos_left != nval1 || p->qpos_right != nval2;
+ p->qpos_left = nval1;
+ p->qpos_right = nval2;
+ p->qpos_changed = change;
+ spin_unlock_irqrestore(&p->q_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_sb_qsound_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "3D Control - Switch",
+ .info = snd_sb_qsound_switch_info,
+ .get = snd_sb_qsound_switch_get,
+ .put = snd_sb_qsound_switch_put
+};
+
+static struct snd_kcontrol_new snd_sb_qsound_space = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "3D Control - Space",
+ .info = snd_sb_qsound_space_info,
+ .get = snd_sb_qsound_space_get,
+ .put = snd_sb_qsound_space_put
+};
+
+static int snd_sb_qsound_build(struct snd_sb_csp * p)
+{
+ struct snd_card *card;
+ int err;
+
+ if (snd_BUG_ON(!p))
+ return -EINVAL;
+
+ card = p->chip->card;
+ p->qpos_left = p->qpos_right = SNDRV_SB_CSP_QSOUND_MAX_RIGHT / 2;
+ p->qpos_changed = 0;
+
+ spin_lock_init(&p->q_lock);
+
+ if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0)
+ goto __error;
+ if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0)
+ goto __error;
+
+ return 0;
+
+ __error:
+ snd_sb_qsound_destroy(p);
+ return err;
+}
+
+static void snd_sb_qsound_destroy(struct snd_sb_csp * p)
+{
+ struct snd_card *card;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!p))
+ return;
+
+ card = p->chip->card;
+
+ down_write(&card->controls_rwsem);
+ if (p->qsound_switch)
+ snd_ctl_remove(card, p->qsound_switch);
+ if (p->qsound_space)
+ snd_ctl_remove(card, p->qsound_space);
+ up_write(&card->controls_rwsem);
+
+ /* cancel pending transfer of QSound parameters */
+ spin_lock_irqsave (&p->q_lock, flags);
+ p->qpos_changed = 0;
+ spin_unlock_irqrestore (&p->q_lock, flags);
+}
+
+/*
+ * Transfer qsound parameters to CSP,
+ * function should be called from interrupt routine
+ */
+static int snd_sb_csp_qsound_transfer(struct snd_sb_csp * p)
+{
+ int err = -ENXIO;
+
+ spin_lock(&p->q_lock);
+ if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
+ set_codec_parameter(p->chip, 0xe0, 0x01);
+ /* left channel */
+ set_codec_parameter(p->chip, 0x00, p->qpos_left);
+ set_codec_parameter(p->chip, 0x02, 0x00);
+ /* right channel */
+ set_codec_parameter(p->chip, 0x00, p->qpos_right);
+ set_codec_parameter(p->chip, 0x03, 0x00);
+ err = 0;
+ }
+ p->qpos_changed = 0;
+ spin_unlock(&p->q_lock);
+ return err;
+}
+
+/* ------------------------------ */
+
+/*
+ * proc interface
+ */
+static int init_proc_entry(struct snd_sb_csp * p, int device)
+{
+ char name[16];
+ struct snd_info_entry *entry;
+ sprintf(name, "cspD%d", device);
+ if (! snd_card_proc_new(p->chip->card, name, &entry))
+ snd_info_set_text_ops(entry, p, info_read);
+ return 0;
+}
+
+static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_sb_csp *p = entry->private_data;
+
+ snd_iprintf(buffer, "Creative Signal Processor [v%d.%d]\n", (p->version >> 4), (p->version & 0x0f));
+ snd_iprintf(buffer, "State: %cx%c%c%c\n", ((p->running & SNDRV_SB_CSP_ST_QSOUND) ? 'Q' : '-'),
+ ((p->running & SNDRV_SB_CSP_ST_PAUSED) ? 'P' : '-'),
+ ((p->running & SNDRV_SB_CSP_ST_RUNNING) ? 'R' : '-'),
+ ((p->running & SNDRV_SB_CSP_ST_LOADED) ? 'L' : '-'));
+ if (p->running & SNDRV_SB_CSP_ST_LOADED) {
+ snd_iprintf(buffer, "Codec: %s [func #%d]\n", p->codec_name, p->func_nr);
+ snd_iprintf(buffer, "Sample rates: ");
+ if (p->acc_rates == SNDRV_SB_CSP_RATE_ALL) {
+ snd_iprintf(buffer, "All\n");
+ } else {
+ snd_iprintf(buffer, "%s%s%s%s\n",
+ ((p->acc_rates & SNDRV_SB_CSP_RATE_8000) ? "8000Hz " : ""),
+ ((p->acc_rates & SNDRV_SB_CSP_RATE_11025) ? "11025Hz " : ""),
+ ((p->acc_rates & SNDRV_SB_CSP_RATE_22050) ? "22050Hz " : ""),
+ ((p->acc_rates & SNDRV_SB_CSP_RATE_44100) ? "44100Hz" : ""));
+ }
+ if (p->mode == SNDRV_SB_CSP_MODE_QSOUND) {
+ snd_iprintf(buffer, "QSound decoder %sabled\n",
+ p->q_enabled ? "en" : "dis");
+ } else {
+ snd_iprintf(buffer, "PCM format ID: 0x%x (%s/%s) [%s/%s] [%s/%s]\n",
+ p->acc_format,
+ ((p->acc_width & SNDRV_SB_CSP_SAMPLE_16BIT) ? "16bit" : "-"),
+ ((p->acc_width & SNDRV_SB_CSP_SAMPLE_8BIT) ? "8bit" : "-"),
+ ((p->acc_channels & SNDRV_SB_CSP_MONO) ? "mono" : "-"),
+ ((p->acc_channels & SNDRV_SB_CSP_STEREO) ? "stereo" : "-"),
+ ((p->mode & SNDRV_SB_CSP_MODE_DSP_WRITE) ? "playback" : "-"),
+ ((p->mode & SNDRV_SB_CSP_MODE_DSP_READ) ? "capture" : "-"));
+ }
+ }
+ if (p->running & SNDRV_SB_CSP_ST_AUTO) {
+ snd_iprintf(buffer, "Autoloaded Mu-Law, A-Law or Ima-ADPCM hardware codec\n");
+ }
+ if (p->running & SNDRV_SB_CSP_ST_RUNNING) {
+ snd_iprintf(buffer, "Processing %dbit %s PCM samples\n",
+ ((p->run_width & SNDRV_SB_CSP_SAMPLE_16BIT) ? 16 : 8),
+ ((p->run_channels & SNDRV_SB_CSP_MONO) ? "mono" : "stereo"));
+ }
+ if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
+ snd_iprintf(buffer, "Qsound position: left = 0x%x, right = 0x%x\n",
+ p->qpos_left, p->qpos_right);
+ }
+}
+
+/* */
+
+EXPORT_SYMBOL(snd_sb_csp_new);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_sb_csp_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_sb_csp_exit(void)
+{
+}
+
+module_init(alsa_sb_csp_init)
+module_exit(alsa_sb_csp_exit)
diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c
new file mode 100644
index 0000000..2a6cc1c
--- /dev/null
+++ b/sound/isa/sb/sb16_main.c
@@ -0,0 +1,924 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of 16-bit SoundBlaster cards and clones
+ * Note: This is very ugly hardware which uses one 8-bit DMA channel and
+ * second 16-bit DMA channel. Unfortunately 8-bit DMA channel can't
+ * transfer 16-bit samples and 16-bit DMA channels can't transfer
+ * 8-bit samples. This make full duplex more complicated than
+ * can be... People, don't buy these soundcards for full 16-bit
+ * duplex!!!
+ * Note: 16-bit wide is assigned to first direction which made request.
+ * With full duplex - playback is preferred with abstract layer.
+ *
+ * Note: Some chip revisions have hardware bug. Changing capture
+ * channel from full-duplex 8bit DMA to 16bit DMA will block
+ * 16bit DMA transfers from DSP chip (capture) until 8bit transfer
+ * to DSP chip (playback) starts. This bug can be avoided with
+ * "16bit DMA Allocation" setting set to Playback or Capture.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <asm/dma.h>
+#include <linux/init.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+#include <sound/sb16_csp.h>
+#include <sound/mpu401.h>
+#include <sound/control.h>
+#include <sound/info.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Routines for control of 16-bit SoundBlaster cards and clones");
+MODULE_LICENSE("GPL");
+
+#ifdef CONFIG_SND_SB16_CSP
+static void snd_sb16_csp_playback_prepare(struct snd_sb *chip, struct snd_pcm_runtime *runtime)
+{
+ if (chip->hardware == SB_HW_16CSP) {
+ struct snd_sb_csp *csp = chip->csp;
+
+ if (csp->running & SNDRV_SB_CSP_ST_LOADED) {
+ /* manually loaded codec */
+ if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_WRITE) &&
+ ((1U << runtime->format) == csp->acc_format)) {
+ /* Supported runtime PCM format for playback */
+ if (csp->ops.csp_use(csp) == 0) {
+ /* If CSP was successfully acquired */
+ goto __start_CSP;
+ }
+ } else if ((csp->mode & SNDRV_SB_CSP_MODE_QSOUND) && (csp->q_enabled)) {
+ /* QSound decoder is loaded and enabled */
+ if ((1 << runtime->format) & (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE)) {
+ /* Only for simple PCM formats */
+ if (csp->ops.csp_use(csp) == 0) {
+ /* If CSP was successfully acquired */
+ goto __start_CSP;
+ }
+ }
+ }
+ } else if (csp->ops.csp_use(csp) == 0) {
+ /* Acquire CSP and try to autoload hardware codec */
+ if (csp->ops.csp_autoload(csp, runtime->format, SNDRV_SB_CSP_MODE_DSP_WRITE)) {
+ /* Unsupported format, release CSP */
+ csp->ops.csp_unuse(csp);
+ } else {
+ __start_CSP:
+ /* Try to start CSP */
+ if (csp->ops.csp_start(csp, (chip->mode & SB_MODE_PLAYBACK_16) ?
+ SNDRV_SB_CSP_SAMPLE_16BIT : SNDRV_SB_CSP_SAMPLE_8BIT,
+ (runtime->channels > 1) ?
+ SNDRV_SB_CSP_STEREO : SNDRV_SB_CSP_MONO)) {
+ /* Failed, release CSP */
+ csp->ops.csp_unuse(csp);
+ } else {
+ /* Success, CSP acquired and running */
+ chip->open = SNDRV_SB_CSP_MODE_DSP_WRITE;
+ }
+ }
+ }
+ }
+}
+
+static void snd_sb16_csp_capture_prepare(struct snd_sb *chip, struct snd_pcm_runtime *runtime)
+{
+ if (chip->hardware == SB_HW_16CSP) {
+ struct snd_sb_csp *csp = chip->csp;
+
+ if (csp->running & SNDRV_SB_CSP_ST_LOADED) {
+ /* manually loaded codec */
+ if ((csp->mode & SNDRV_SB_CSP_MODE_DSP_READ) &&
+ ((1U << runtime->format) == csp->acc_format)) {
+ /* Supported runtime PCM format for capture */
+ if (csp->ops.csp_use(csp) == 0) {
+ /* If CSP was successfully acquired */
+ goto __start_CSP;
+ }
+ }
+ } else if (csp->ops.csp_use(csp) == 0) {
+ /* Acquire CSP and try to autoload hardware codec */
+ if (csp->ops.csp_autoload(csp, runtime->format, SNDRV_SB_CSP_MODE_DSP_READ)) {
+ /* Unsupported format, release CSP */
+ csp->ops.csp_unuse(csp);
+ } else {
+ __start_CSP:
+ /* Try to start CSP */
+ if (csp->ops.csp_start(csp, (chip->mode & SB_MODE_CAPTURE_16) ?
+ SNDRV_SB_CSP_SAMPLE_16BIT : SNDRV_SB_CSP_SAMPLE_8BIT,
+ (runtime->channels > 1) ?
+ SNDRV_SB_CSP_STEREO : SNDRV_SB_CSP_MONO)) {
+ /* Failed, release CSP */
+ csp->ops.csp_unuse(csp);
+ } else {
+ /* Success, CSP acquired and running */
+ chip->open = SNDRV_SB_CSP_MODE_DSP_READ;
+ }
+ }
+ }
+ }
+}
+
+static void snd_sb16_csp_update(struct snd_sb *chip)
+{
+ if (chip->hardware == SB_HW_16CSP) {
+ struct snd_sb_csp *csp = chip->csp;
+
+ if (csp->qpos_changed) {
+ spin_lock(&chip->reg_lock);
+ csp->ops.csp_qsound_transfer (csp);
+ spin_unlock(&chip->reg_lock);
+ }
+ }
+}
+
+static void snd_sb16_csp_playback_open(struct snd_sb *chip, struct snd_pcm_runtime *runtime)
+{
+ /* CSP decoders (QSound excluded) support only 16bit transfers */
+ if (chip->hardware == SB_HW_16CSP) {
+ struct snd_sb_csp *csp = chip->csp;
+
+ if (csp->running & SNDRV_SB_CSP_ST_LOADED) {
+ /* manually loaded codec */
+ if (csp->mode & SNDRV_SB_CSP_MODE_DSP_WRITE) {
+ runtime->hw.formats |= csp->acc_format;
+ }
+ } else {
+ /* autoloaded codecs */
+ runtime->hw.formats |= SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW |
+ SNDRV_PCM_FMTBIT_IMA_ADPCM;
+ }
+ }
+}
+
+static void snd_sb16_csp_playback_close(struct snd_sb *chip)
+{
+ if ((chip->hardware == SB_HW_16CSP) && (chip->open == SNDRV_SB_CSP_MODE_DSP_WRITE)) {
+ struct snd_sb_csp *csp = chip->csp;
+
+ if (csp->ops.csp_stop(csp) == 0) {
+ csp->ops.csp_unuse(csp);
+ chip->open = 0;
+ }
+ }
+}
+
+static void snd_sb16_csp_capture_open(struct snd_sb *chip, struct snd_pcm_runtime *runtime)
+{
+ /* CSP coders support only 16bit transfers */
+ if (chip->hardware == SB_HW_16CSP) {
+ struct snd_sb_csp *csp = chip->csp;
+
+ if (csp->running & SNDRV_SB_CSP_ST_LOADED) {
+ /* manually loaded codec */
+ if (csp->mode & SNDRV_SB_CSP_MODE_DSP_READ) {
+ runtime->hw.formats |= csp->acc_format;
+ }
+ } else {
+ /* autoloaded codecs */
+ runtime->hw.formats |= SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW |
+ SNDRV_PCM_FMTBIT_IMA_ADPCM;
+ }
+ }
+}
+
+static void snd_sb16_csp_capture_close(struct snd_sb *chip)
+{
+ if ((chip->hardware == SB_HW_16CSP) && (chip->open == SNDRV_SB_CSP_MODE_DSP_READ)) {
+ struct snd_sb_csp *csp = chip->csp;
+
+ if (csp->ops.csp_stop(csp) == 0) {
+ csp->ops.csp_unuse(csp);
+ chip->open = 0;
+ }
+ }
+}
+#else
+#define snd_sb16_csp_playback_prepare(chip, runtime) /*nop*/
+#define snd_sb16_csp_capture_prepare(chip, runtime) /*nop*/
+#define snd_sb16_csp_update(chip) /*nop*/
+#define snd_sb16_csp_playback_open(chip, runtime) /*nop*/
+#define snd_sb16_csp_playback_close(chip) /*nop*/
+#define snd_sb16_csp_capture_open(chip, runtime) /*nop*/
+#define snd_sb16_csp_capture_close(chip) /*nop*/
+#endif
+
+
+static void snd_sb16_setup_rate(struct snd_sb *chip,
+ unsigned short rate,
+ int channel)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->mode & (channel == SNDRV_PCM_STREAM_PLAYBACK ? SB_MODE_PLAYBACK_16 : SB_MODE_CAPTURE_16))
+ snd_sb_ack_16bit(chip);
+ else
+ snd_sb_ack_8bit(chip);
+ if (!(chip->mode & SB_RATE_LOCK)) {
+ chip->locked_rate = rate;
+ snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE_IN);
+ snd_sbdsp_command(chip, rate >> 8);
+ snd_sbdsp_command(chip, rate & 0xff);
+ snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE_OUT);
+ snd_sbdsp_command(chip, rate >> 8);
+ snd_sbdsp_command(chip, rate & 0xff);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static int snd_sb16_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_sb16_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_sb16_playback_prepare(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned char format;
+ unsigned int size, count, dma;
+
+ snd_sb16_csp_playback_prepare(chip, runtime);
+ if (snd_pcm_format_unsigned(runtime->format) > 0) {
+ format = runtime->channels > 1 ? SB_DSP4_MODE_UNS_STEREO : SB_DSP4_MODE_UNS_MONO;
+ } else {
+ format = runtime->channels > 1 ? SB_DSP4_MODE_SIGN_STEREO : SB_DSP4_MODE_SIGN_MONO;
+ }
+
+ snd_sb16_setup_rate(chip, runtime->rate, SNDRV_PCM_STREAM_PLAYBACK);
+ size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream);
+ dma = (chip->mode & SB_MODE_PLAYBACK_8) ? chip->dma8 : chip->dma16;
+ snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT);
+
+ count = snd_pcm_lib_period_bytes(substream);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->mode & SB_MODE_PLAYBACK_16) {
+ count >>= 1;
+ count--;
+ snd_sbdsp_command(chip, SB_DSP4_OUT16_AI);
+ snd_sbdsp_command(chip, format);
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ snd_sbdsp_command(chip, SB_DSP_DMA16_OFF);
+ } else {
+ count--;
+ snd_sbdsp_command(chip, SB_DSP4_OUT8_AI);
+ snd_sbdsp_command(chip, format);
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ snd_sbdsp_command(chip, SB_DSP_DMA8_OFF);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_sb16_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ int result = 0;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ chip->mode |= SB_RATE_LOCK_PLAYBACK;
+ snd_sbdsp_command(chip, chip->mode & SB_MODE_PLAYBACK_16 ? SB_DSP_DMA16_ON : SB_DSP_DMA8_ON);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ snd_sbdsp_command(chip, chip->mode & SB_MODE_PLAYBACK_16 ? SB_DSP_DMA16_OFF : SB_DSP_DMA8_OFF);
+ /* next two lines are needed for some types of DSP4 (SB AWE 32 - 4.13) */
+ if (chip->mode & SB_RATE_LOCK_CAPTURE)
+ snd_sbdsp_command(chip, chip->mode & SB_MODE_CAPTURE_16 ? SB_DSP_DMA16_ON : SB_DSP_DMA8_ON);
+ chip->mode &= ~SB_RATE_LOCK_PLAYBACK;
+ break;
+ default:
+ result = -EINVAL;
+ }
+ spin_unlock(&chip->reg_lock);
+ return result;
+}
+
+static int snd_sb16_capture_prepare(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned char format;
+ unsigned int size, count, dma;
+
+ snd_sb16_csp_capture_prepare(chip, runtime);
+ if (snd_pcm_format_unsigned(runtime->format) > 0) {
+ format = runtime->channels > 1 ? SB_DSP4_MODE_UNS_STEREO : SB_DSP4_MODE_UNS_MONO;
+ } else {
+ format = runtime->channels > 1 ? SB_DSP4_MODE_SIGN_STEREO : SB_DSP4_MODE_SIGN_MONO;
+ }
+ snd_sb16_setup_rate(chip, runtime->rate, SNDRV_PCM_STREAM_CAPTURE);
+ size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream);
+ dma = (chip->mode & SB_MODE_CAPTURE_8) ? chip->dma8 : chip->dma16;
+ snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT);
+
+ count = snd_pcm_lib_period_bytes(substream);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->mode & SB_MODE_CAPTURE_16) {
+ count >>= 1;
+ count--;
+ snd_sbdsp_command(chip, SB_DSP4_IN16_AI);
+ snd_sbdsp_command(chip, format);
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ snd_sbdsp_command(chip, SB_DSP_DMA16_OFF);
+ } else {
+ count--;
+ snd_sbdsp_command(chip, SB_DSP4_IN8_AI);
+ snd_sbdsp_command(chip, format);
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ snd_sbdsp_command(chip, SB_DSP_DMA8_OFF);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_sb16_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ int result = 0;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ chip->mode |= SB_RATE_LOCK_CAPTURE;
+ snd_sbdsp_command(chip, chip->mode & SB_MODE_CAPTURE_16 ? SB_DSP_DMA16_ON : SB_DSP_DMA8_ON);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ snd_sbdsp_command(chip, chip->mode & SB_MODE_CAPTURE_16 ? SB_DSP_DMA16_OFF : SB_DSP_DMA8_OFF);
+ /* next two lines are needed for some types of DSP4 (SB AWE 32 - 4.13) */
+ if (chip->mode & SB_RATE_LOCK_PLAYBACK)
+ snd_sbdsp_command(chip, chip->mode & SB_MODE_PLAYBACK_16 ? SB_DSP_DMA16_ON : SB_DSP_DMA8_ON);
+ chip->mode &= ~SB_RATE_LOCK_CAPTURE;
+ break;
+ default:
+ result = -EINVAL;
+ }
+ spin_unlock(&chip->reg_lock);
+ return result;
+}
+
+irqreturn_t snd_sb16dsp_interrupt(int irq, void *dev_id)
+{
+ struct snd_sb *chip = dev_id;
+ unsigned char status;
+ int ok;
+
+ spin_lock(&chip->mixer_lock);
+ status = snd_sbmixer_read(chip, SB_DSP4_IRQSTATUS);
+ spin_unlock(&chip->mixer_lock);
+ if ((status & SB_IRQTYPE_MPUIN) && chip->rmidi_callback)
+ chip->rmidi_callback(irq, chip->rmidi->private_data);
+ if (status & SB_IRQTYPE_8BIT) {
+ ok = 0;
+ if (chip->mode & SB_MODE_PLAYBACK_8) {
+ snd_pcm_period_elapsed(chip->playback_substream);
+ snd_sb16_csp_update(chip);
+ ok++;
+ }
+ if (chip->mode & SB_MODE_CAPTURE_8) {
+ snd_pcm_period_elapsed(chip->capture_substream);
+ ok++;
+ }
+ spin_lock(&chip->reg_lock);
+ if (!ok)
+ snd_sbdsp_command(chip, SB_DSP_DMA8_OFF);
+ snd_sb_ack_8bit(chip);
+ spin_unlock(&chip->reg_lock);
+ }
+ if (status & SB_IRQTYPE_16BIT) {
+ ok = 0;
+ if (chip->mode & SB_MODE_PLAYBACK_16) {
+ snd_pcm_period_elapsed(chip->playback_substream);
+ snd_sb16_csp_update(chip);
+ ok++;
+ }
+ if (chip->mode & SB_MODE_CAPTURE_16) {
+ snd_pcm_period_elapsed(chip->capture_substream);
+ ok++;
+ }
+ spin_lock(&chip->reg_lock);
+ if (!ok)
+ snd_sbdsp_command(chip, SB_DSP_DMA16_OFF);
+ snd_sb_ack_16bit(chip);
+ spin_unlock(&chip->reg_lock);
+ }
+ return IRQ_HANDLED;
+}
+
+/*
+
+ */
+
+static snd_pcm_uframes_t snd_sb16_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ unsigned int dma;
+ size_t ptr;
+
+ dma = (chip->mode & SB_MODE_PLAYBACK_8) ? chip->dma8 : chip->dma16;
+ ptr = snd_dma_pointer(dma, chip->p_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_sb16_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ unsigned int dma;
+ size_t ptr;
+
+ dma = (chip->mode & SB_MODE_CAPTURE_8) ? chip->dma8 : chip->dma16;
+ ptr = snd_dma_pointer(dma, chip->c_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+/*
+
+ */
+
+static struct snd_pcm_hardware snd_sb16_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = 0,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_44100,
+ .rate_min = 4000,
+ .rate_max = 44100,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_sb16_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = 0,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_44100,
+ .rate_min = 4000,
+ .rate_max = 44100,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+ * open/close
+ */
+
+static int snd_sb16_playback_open(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (chip->mode & SB_MODE_PLAYBACK) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return -EAGAIN;
+ }
+ runtime->hw = snd_sb16_playback;
+
+ /* skip if 16 bit DMA was reserved for capture */
+ if (chip->force_mode16 & SB_MODE_CAPTURE_16)
+ goto __skip_16bit;
+
+ if (chip->dma16 >= 0 && !(chip->mode & SB_MODE_CAPTURE_16)) {
+ chip->mode |= SB_MODE_PLAYBACK_16;
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE;
+ /* Vibra16X hack */
+ if (chip->dma16 <= 3) {
+ runtime->hw.buffer_bytes_max =
+ runtime->hw.period_bytes_max = 64 * 1024;
+ } else {
+ snd_sb16_csp_playback_open(chip, runtime);
+ }
+ goto __open_ok;
+ }
+
+ __skip_16bit:
+ if (chip->dma8 >= 0 && !(chip->mode & SB_MODE_CAPTURE_8)) {
+ chip->mode |= SB_MODE_PLAYBACK_8;
+ /* DSP v 4.xx can transfer 16bit data through 8bit DMA channel, SBHWPG 2-7 */
+ if (chip->dma16 < 0) {
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE;
+ chip->mode |= SB_MODE_PLAYBACK_16;
+ } else {
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8;
+ }
+ runtime->hw.buffer_bytes_max =
+ runtime->hw.period_bytes_max = 64 * 1024;
+ goto __open_ok;
+ }
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return -EAGAIN;
+
+ __open_ok:
+ if (chip->hardware == SB_HW_ALS100)
+ runtime->hw.rate_max = 48000;
+ if (chip->hardware == SB_HW_CS5530) {
+ runtime->hw.buffer_bytes_max = 32 * 1024;
+ runtime->hw.periods_min = 2;
+ runtime->hw.rate_min = 44100;
+ }
+ if (chip->mode & SB_RATE_LOCK)
+ runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate;
+ chip->playback_substream = substream;
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return 0;
+}
+
+static int snd_sb16_playback_close(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+
+ snd_sb16_csp_playback_close(chip);
+ spin_lock_irqsave(&chip->open_lock, flags);
+ chip->playback_substream = NULL;
+ chip->mode &= ~SB_MODE_PLAYBACK;
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return 0;
+}
+
+static int snd_sb16_capture_open(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (chip->mode & SB_MODE_CAPTURE) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return -EAGAIN;
+ }
+ runtime->hw = snd_sb16_capture;
+
+ /* skip if 16 bit DMA was reserved for playback */
+ if (chip->force_mode16 & SB_MODE_PLAYBACK_16)
+ goto __skip_16bit;
+
+ if (chip->dma16 >= 0 && !(chip->mode & SB_MODE_PLAYBACK_16)) {
+ chip->mode |= SB_MODE_CAPTURE_16;
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE;
+ /* Vibra16X hack */
+ if (chip->dma16 <= 3) {
+ runtime->hw.buffer_bytes_max =
+ runtime->hw.period_bytes_max = 64 * 1024;
+ } else {
+ snd_sb16_csp_capture_open(chip, runtime);
+ }
+ goto __open_ok;
+ }
+
+ __skip_16bit:
+ if (chip->dma8 >= 0 && !(chip->mode & SB_MODE_PLAYBACK_8)) {
+ chip->mode |= SB_MODE_CAPTURE_8;
+ /* DSP v 4.xx can transfer 16bit data through 8bit DMA channel, SBHWPG 2-7 */
+ if (chip->dma16 < 0) {
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE;
+ chip->mode |= SB_MODE_CAPTURE_16;
+ } else {
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8;
+ }
+ runtime->hw.buffer_bytes_max =
+ runtime->hw.period_bytes_max = 64 * 1024;
+ goto __open_ok;
+ }
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return -EAGAIN;
+
+ __open_ok:
+ if (chip->hardware == SB_HW_ALS100)
+ runtime->hw.rate_max = 48000;
+ if (chip->hardware == SB_HW_CS5530) {
+ runtime->hw.buffer_bytes_max = 32 * 1024;
+ runtime->hw.periods_min = 2;
+ runtime->hw.rate_min = 44100;
+ }
+ if (chip->mode & SB_RATE_LOCK)
+ runtime->hw.rate_min = runtime->hw.rate_max = chip->locked_rate;
+ chip->capture_substream = substream;
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return 0;
+}
+
+static int snd_sb16_capture_close(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+
+ snd_sb16_csp_capture_close(chip);
+ spin_lock_irqsave(&chip->open_lock, flags);
+ chip->capture_substream = NULL;
+ chip->mode &= ~SB_MODE_CAPTURE;
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return 0;
+}
+
+/*
+ * DMA control interface
+ */
+
+static int snd_sb16_set_dma_mode(struct snd_sb *chip, int what)
+{
+ if (chip->dma8 < 0 || chip->dma16 < 0) {
+ if (snd_BUG_ON(what))
+ return -EINVAL;
+ return 0;
+ }
+ if (what == 0) {
+ chip->force_mode16 = 0;
+ } else if (what == 1) {
+ chip->force_mode16 = SB_MODE_PLAYBACK_16;
+ } else if (what == 2) {
+ chip->force_mode16 = SB_MODE_CAPTURE_16;
+ } else {
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int snd_sb16_get_dma_mode(struct snd_sb *chip)
+{
+ if (chip->dma8 < 0 || chip->dma16 < 0)
+ return 0;
+ switch (chip->force_mode16) {
+ case SB_MODE_PLAYBACK_16:
+ return 1;
+ case SB_MODE_CAPTURE_16:
+ return 2;
+ default:
+ return 0;
+ }
+}
+
+static int snd_sb16_dma_control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {
+ "Auto", "Playback", "Capture"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_sb16_dma_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.enumerated.item[0] = snd_sb16_get_dma_mode(chip);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_sb16_dma_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned char nval, oval;
+ int change;
+
+ if ((nval = ucontrol->value.enumerated.item[0]) > 2)
+ return -EINVAL;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ oval = snd_sb16_get_dma_mode(chip);
+ change = nval != oval;
+ snd_sb16_set_dma_mode(chip, nval);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_sb16_dma_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "16-bit DMA Allocation",
+ .info = snd_sb16_dma_control_info,
+ .get = snd_sb16_dma_control_get,
+ .put = snd_sb16_dma_control_put
+};
+
+/*
+ * Initialization part
+ */
+
+int snd_sb16dsp_configure(struct snd_sb * chip)
+{
+ unsigned long flags;
+ unsigned char irqreg = 0, dmareg = 0, mpureg;
+ unsigned char realirq, realdma, realmpureg;
+ /* note: mpu register should be present only on SB16 Vibra soundcards */
+
+ // printk(KERN_DEBUG "codec->irq=%i, codec->dma8=%i, codec->dma16=%i\n", chip->irq, chip->dma8, chip->dma16);
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ mpureg = snd_sbmixer_read(chip, SB_DSP4_MPUSETUP) & ~0x06;
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+ switch (chip->irq) {
+ case 2:
+ case 9:
+ irqreg |= SB_IRQSETUP_IRQ9;
+ break;
+ case 5:
+ irqreg |= SB_IRQSETUP_IRQ5;
+ break;
+ case 7:
+ irqreg |= SB_IRQSETUP_IRQ7;
+ break;
+ case 10:
+ irqreg |= SB_IRQSETUP_IRQ10;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (chip->dma8 >= 0) {
+ switch (chip->dma8) {
+ case 0:
+ dmareg |= SB_DMASETUP_DMA0;
+ break;
+ case 1:
+ dmareg |= SB_DMASETUP_DMA1;
+ break;
+ case 3:
+ dmareg |= SB_DMASETUP_DMA3;
+ break;
+ default:
+ return -EINVAL;
+ }
+ }
+ if (chip->dma16 >= 0 && chip->dma16 != chip->dma8) {
+ switch (chip->dma16) {
+ case 5:
+ dmareg |= SB_DMASETUP_DMA5;
+ break;
+ case 6:
+ dmareg |= SB_DMASETUP_DMA6;
+ break;
+ case 7:
+ dmareg |= SB_DMASETUP_DMA7;
+ break;
+ default:
+ return -EINVAL;
+ }
+ }
+ switch (chip->mpu_port) {
+ case 0x300:
+ mpureg |= 0x04;
+ break;
+ case 0x330:
+ mpureg |= 0x00;
+ break;
+ default:
+ mpureg |= 0x02; /* disable MPU */
+ }
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+
+ snd_sbmixer_write(chip, SB_DSP4_IRQSETUP, irqreg);
+ realirq = snd_sbmixer_read(chip, SB_DSP4_IRQSETUP);
+
+ snd_sbmixer_write(chip, SB_DSP4_DMASETUP, dmareg);
+ realdma = snd_sbmixer_read(chip, SB_DSP4_DMASETUP);
+
+ snd_sbmixer_write(chip, SB_DSP4_MPUSETUP, mpureg);
+ realmpureg = snd_sbmixer_read(chip, SB_DSP4_MPUSETUP);
+
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+ if ((~realirq) & irqreg || (~realdma) & dmareg) {
+ snd_printk(KERN_ERR "SB16 [0x%lx]: unable to set DMA & IRQ (PnP device?)\n", chip->port);
+ snd_printk(KERN_ERR "SB16 [0x%lx]: wanted: irqreg=0x%x, dmareg=0x%x, mpureg = 0x%x\n", chip->port, realirq, realdma, realmpureg);
+ snd_printk(KERN_ERR "SB16 [0x%lx]: got: irqreg=0x%x, dmareg=0x%x, mpureg = 0x%x\n", chip->port, irqreg, dmareg, mpureg);
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static struct snd_pcm_ops snd_sb16_playback_ops = {
+ .open = snd_sb16_playback_open,
+ .close = snd_sb16_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sb16_hw_params,
+ .hw_free = snd_sb16_hw_free,
+ .prepare = snd_sb16_playback_prepare,
+ .trigger = snd_sb16_playback_trigger,
+ .pointer = snd_sb16_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_sb16_capture_ops = {
+ .open = snd_sb16_capture_open,
+ .close = snd_sb16_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sb16_hw_params,
+ .hw_free = snd_sb16_hw_free,
+ .prepare = snd_sb16_capture_prepare,
+ .trigger = snd_sb16_capture_trigger,
+ .pointer = snd_sb16_capture_pointer,
+};
+
+int snd_sb16dsp_pcm(struct snd_sb * chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_card *card = chip->card;
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(card, "SB16 DSP", device, 1, 1, &pcm)) < 0)
+ return err;
+ sprintf(pcm->name, "DSP v%i.%i", chip->version >> 8, chip->version & 0xff);
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops);
+
+ if (chip->dma16 >= 0 && chip->dma8 != chip->dma16)
+ snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip));
+ else
+ pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+const struct snd_pcm_ops *snd_sb16dsp_get_pcm_ops(int direction)
+{
+ return direction == SNDRV_PCM_STREAM_PLAYBACK ?
+ &snd_sb16_playback_ops : &snd_sb16_capture_ops;
+}
+
+EXPORT_SYMBOL(snd_sb16dsp_pcm);
+EXPORT_SYMBOL(snd_sb16dsp_get_pcm_ops);
+EXPORT_SYMBOL(snd_sb16dsp_configure);
+EXPORT_SYMBOL(snd_sb16dsp_interrupt);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_sb16_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_sb16_exit(void)
+{
+}
+
+module_init(alsa_sb16_init)
+module_exit(alsa_sb16_exit)
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
new file mode 100644
index 0000000..667eccc
--- /dev/null
+++ b/sound/isa/sb/sb8.c
@@ -0,0 +1,267 @@
+/*
+ * Driver for SoundBlaster 1.0/2.0/Pro soundcards and compatible
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Sound Blaster 1.0/2.0/Pro");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB 1.0/SB 2.0/SB Pro}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */
+static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 1,3 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Sound Blaster soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Sound Blaster soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Sound Blaster soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for SB8 driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for SB8 driver.");
+module_param_array(dma8, int, NULL, 0444);
+MODULE_PARM_DESC(dma8, "8-bit DMA # for SB8 driver.");
+
+struct snd_sb8 {
+ struct resource *fm_res; /* used to block FM i/o region for legacy cards */
+ struct snd_sb *chip;
+};
+
+static irqreturn_t snd_sb8_interrupt(int irq, void *dev_id)
+{
+ struct snd_sb *chip = dev_id;
+
+ if (chip->open & SB_OPEN_PCM) {
+ return snd_sb8dsp_interrupt(chip);
+ } else {
+ return snd_sb8dsp_midi_interrupt(chip);
+ }
+}
+
+static void snd_sb8_free(struct snd_card *card)
+{
+ struct snd_sb8 *acard = (struct snd_sb8 *)card->private_data;
+
+ if (acard == NULL)
+ return;
+ release_and_free_resource(acard->fm_res);
+}
+
+static int __devinit snd_sb8_match(struct device *pdev, unsigned int dev)
+{
+ if (!enable[dev])
+ return 0;
+ if (irq[dev] == SNDRV_AUTO_IRQ) {
+ dev_err(pdev, "please specify irq\n");
+ return 0;
+ }
+ if (dma8[dev] == SNDRV_AUTO_DMA) {
+ dev_err(pdev, "please specify dma8\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_sb8_probe(struct device *pdev, unsigned int dev)
+{
+ struct snd_sb *chip;
+ struct snd_card *card;
+ struct snd_sb8 *acard;
+ struct snd_opl3 *opl3;
+ int err;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_sb8));
+ if (card == NULL)
+ return -ENOMEM;
+ acard = card->private_data;
+ card->private_free = snd_sb8_free;
+
+ /* block the 0x388 port to avoid PnP conflicts */
+ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM");
+
+ if (port[dev] != SNDRV_AUTO_PORT) {
+ if ((err = snd_sbdsp_create(card, port[dev], irq[dev],
+ snd_sb8_interrupt,
+ dma8[dev],
+ -1,
+ SB_HW_AUTO,
+ &chip)) < 0)
+ goto _err;
+ } else {
+ /* auto-probe legacy ports */
+ static unsigned long possible_ports[] = {
+ 0x220, 0x240, 0x260,
+ };
+ int i;
+ for (i = 0; i < ARRAY_SIZE(possible_ports); i++) {
+ err = snd_sbdsp_create(card, possible_ports[i],
+ irq[dev],
+ snd_sb8_interrupt,
+ dma8[dev],
+ -1,
+ SB_HW_AUTO,
+ &chip);
+ if (err >= 0) {
+ port[dev] = possible_ports[i];
+ break;
+ }
+ }
+ if (i >= ARRAY_SIZE(possible_ports))
+ goto _err;
+ }
+ acard->chip = chip;
+
+ if (chip->hardware >= SB_HW_16) {
+ if (chip->hardware == SB_HW_ALS100)
+ snd_printk(KERN_WARNING "ALS100 chip detected at 0x%lx, try snd-als100 module\n",
+ port[dev]);
+ else
+ snd_printk(KERN_WARNING "SB 16 chip detected at 0x%lx, try snd-sb16 module\n",
+ port[dev]);
+ err = -ENODEV;
+ goto _err;
+ }
+
+ if ((err = snd_sb8dsp_pcm(chip, 0, NULL)) < 0)
+ goto _err;
+
+ if ((err = snd_sbmixer_new(chip)) < 0)
+ goto _err;
+
+ if (chip->hardware == SB_HW_10 || chip->hardware == SB_HW_20) {
+ if ((err = snd_opl3_create(card, chip->port + 8, 0,
+ OPL3_HW_AUTO, 1,
+ &opl3)) < 0) {
+ snd_printk(KERN_WARNING "sb8: no OPL device at 0x%lx\n", chip->port + 8);
+ }
+ } else {
+ if ((err = snd_opl3_create(card, chip->port, chip->port + 2,
+ OPL3_HW_AUTO, 1,
+ &opl3)) < 0) {
+ snd_printk(KERN_WARNING "sb8: no OPL device at 0x%lx-0x%lx\n",
+ chip->port, chip->port + 2);
+ }
+ }
+ if (err >= 0) {
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0)
+ goto _err;
+ }
+
+ if ((err = snd_sb8dsp_midi(chip, 0, NULL)) < 0)
+ goto _err;
+
+ strcpy(card->driver, chip->hardware == SB_HW_PRO ? "SB Pro" : "SB8");
+ strcpy(card->shortname, chip->name);
+ sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
+ chip->name,
+ chip->port,
+ irq[dev], dma8[dev]);
+
+ snd_card_set_dev(card, pdev);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto _err;
+
+ dev_set_drvdata(pdev, card);
+ return 0;
+
+ _err:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_sb8_remove(struct device *pdev, unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(pdev));
+ dev_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_sb8_suspend(struct device *dev, unsigned int n,
+ pm_message_t state)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_sb8 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_sbmixer_suspend(chip);
+ return 0;
+}
+
+static int snd_sb8_resume(struct device *dev, unsigned int n)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct snd_sb8 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_sbdsp_reset(chip);
+ snd_sbmixer_resume(chip);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+#define DEV_NAME "sb8"
+
+static struct isa_driver snd_sb8_driver = {
+ .match = snd_sb8_match,
+ .probe = snd_sb8_probe,
+ .remove = __devexit_p(snd_sb8_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_sb8_suspend,
+ .resume = snd_sb8_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+static int __init alsa_card_sb8_init(void)
+{
+ return isa_register_driver(&snd_sb8_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_sb8_exit(void)
+{
+ isa_unregister_driver(&snd_sb8_driver);
+}
+
+module_init(alsa_card_sb8_init)
+module_exit(alsa_card_sb8_exit)
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
new file mode 100644
index 0000000..658d557
--- /dev/null
+++ b/sound/isa/sb/sb8_main.c
@@ -0,0 +1,559 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Uros Bizjak <uros@kss-loka.si>
+ *
+ * Routines for control of 8-bit SoundBlaster cards and clones
+ * Please note: I don't have access to old SB8 soundcards.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * --
+ *
+ * Thu Apr 29 20:36:17 BST 1999 George David Morrison <gdm@gedamo.demon.co.uk>
+ * DSP can't respond to commands whilst in "high speed" mode. Caused
+ * glitching during playback. Fixed.
+ *
+ * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros@kss-loka.si>
+ * Cleaned up and rewrote lowlevel routines.
+ */
+
+#include <asm/io.h>
+#include <asm/dma.h>
+#include <linux/init.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Uros Bizjak <uros@kss-loka.si>");
+MODULE_DESCRIPTION("Routines for control of 8-bit SoundBlaster cards and clones");
+MODULE_LICENSE("GPL");
+
+#define SB8_CLOCK 1000000
+#define SB8_DEN(v) ((SB8_CLOCK + (v) / 2) / (v))
+#define SB8_RATE(v) (SB8_CLOCK / SB8_DEN(v))
+
+static struct snd_ratnum clock = {
+ .num = SB8_CLOCK,
+ .den_min = 1,
+ .den_max = 256,
+ .den_step = 1,
+};
+
+static struct snd_pcm_hw_constraint_ratnums hw_constraints_clock = {
+ .nrats = 1,
+ .rats = &clock,
+};
+
+static struct snd_ratnum stereo_clocks[] = {
+ {
+ .num = SB8_CLOCK,
+ .den_min = SB8_DEN(22050),
+ .den_max = SB8_DEN(22050),
+ .den_step = 1,
+ },
+ {
+ .num = SB8_CLOCK,
+ .den_min = SB8_DEN(11025),
+ .den_max = SB8_DEN(11025),
+ .den_step = 1,
+ }
+};
+
+static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ if (c->min > 1) {
+ unsigned int num = 0, den = 0;
+ int err = snd_interval_ratnum(hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE),
+ 2, stereo_clocks, &num, &den);
+ if (err >= 0 && den) {
+ params->rate_num = num;
+ params->rate_den = den;
+ }
+ return err;
+ }
+ return 0;
+}
+
+static int snd_sb8_hw_constraint_channels_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (r->min > SB8_RATE(22050) || r->max <= SB8_RATE(11025)) {
+ struct snd_interval t = { .min = 1, .max = 1 };
+ return snd_interval_refine(hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS), &t);
+ }
+ return 0;
+}
+
+static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int mixreg, rate, size, count;
+
+ rate = runtime->rate;
+ switch (chip->hardware) {
+ case SB_HW_PRO:
+ if (runtime->channels > 1) {
+ if (snd_BUG_ON(rate != SB8_RATE(11025) &&
+ rate != SB8_RATE(22050)))
+ return -EINVAL;
+ chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
+ break;
+ }
+ /* fallthru */
+ case SB_HW_201:
+ if (rate > 23000) {
+ chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
+ break;
+ }
+ /* fallthru */
+ case SB_HW_20:
+ chip->playback_format = SB_DSP_LO_OUTPUT_AUTO;
+ break;
+ case SB_HW_10:
+ chip->playback_format = SB_DSP_OUTPUT;
+ break;
+ default:
+ return -EINVAL;
+ }
+ size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream);
+ count = chip->p_period_size = snd_pcm_lib_period_bytes(substream);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON);
+ if (runtime->channels > 1) {
+ /* set playback stereo mode */
+ spin_lock(&chip->mixer_lock);
+ mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW);
+ snd_sbmixer_write(chip, SB_DSP_STEREO_SW, mixreg | 0x02);
+ spin_unlock(&chip->mixer_lock);
+
+ /* Soundblaster hardware programming reference guide, 3-23 */
+ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT);
+ runtime->dma_area[0] = 0x80;
+ snd_dma_program(chip->dma8, runtime->dma_addr, 1, DMA_MODE_WRITE);
+ /* force interrupt */
+ chip->mode = SB_MODE_HALT;
+ snd_sbdsp_command(chip, SB_DSP_OUTPUT);
+ snd_sbdsp_command(chip, 0);
+ snd_sbdsp_command(chip, 0);
+ }
+ snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE);
+ if (runtime->channels > 1) {
+ snd_sbdsp_command(chip, 256 - runtime->rate_den / 2);
+ spin_lock(&chip->mixer_lock);
+ /* save output filter status and turn it off */
+ mixreg = snd_sbmixer_read(chip, SB_DSP_PLAYBACK_FILT);
+ snd_sbmixer_write(chip, SB_DSP_PLAYBACK_FILT, mixreg | 0x20);
+ spin_unlock(&chip->mixer_lock);
+ /* just use force_mode16 for temporary storate... */
+ chip->force_mode16 = mixreg;
+ } else {
+ snd_sbdsp_command(chip, 256 - runtime->rate_den);
+ }
+ if (chip->playback_format != SB_DSP_OUTPUT) {
+ count--;
+ snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE);
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_dma_program(chip->dma8, runtime->dma_addr,
+ size, DMA_MODE_WRITE | DMA_AUTOINIT);
+ return 0;
+}
+
+static int snd_sb8_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ unsigned int count;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_sbdsp_command(chip, chip->playback_format);
+ if (chip->playback_format == SB_DSP_OUTPUT) {
+ count = chip->p_period_size - 1;
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (chip->playback_format == SB_DSP_HI_OUTPUT_AUTO) {
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_sbdsp_reset(chip);
+ if (runtime->channels > 1) {
+ spin_lock(&chip->mixer_lock);
+ /* restore output filter and set hardware to mono mode */
+ snd_sbmixer_write(chip, SB_DSP_STEREO_SW, chip->force_mode16 & ~0x02);
+ spin_unlock(&chip->mixer_lock);
+ }
+ } else {
+ snd_sbdsp_command(chip, SB_DSP_DMA8_OFF);
+ }
+ snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_PLAYBACK_8 : SB_MODE_HALT;
+ return 0;
+}
+
+static int snd_sb8_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_sb8_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int mixreg, rate, size, count;
+
+ rate = runtime->rate;
+ switch (chip->hardware) {
+ case SB_HW_PRO:
+ if (runtime->channels > 1) {
+ if (snd_BUG_ON(rate != SB8_RATE(11025) &&
+ rate != SB8_RATE(22050)))
+ return -EINVAL;
+ chip->capture_format = SB_DSP_HI_INPUT_AUTO;
+ break;
+ }
+ chip->capture_format = (rate > 23000) ? SB_DSP_HI_INPUT_AUTO : SB_DSP_LO_INPUT_AUTO;
+ break;
+ case SB_HW_201:
+ if (rate > 13000) {
+ chip->capture_format = SB_DSP_HI_INPUT_AUTO;
+ break;
+ }
+ /* fallthru */
+ case SB_HW_20:
+ chip->capture_format = SB_DSP_LO_INPUT_AUTO;
+ break;
+ case SB_HW_10:
+ chip->capture_format = SB_DSP_INPUT;
+ break;
+ default:
+ return -EINVAL;
+ }
+ size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream);
+ count = chip->c_period_size = snd_pcm_lib_period_bytes(substream);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF);
+ if (runtime->channels > 1)
+ snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT);
+ snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE);
+ if (runtime->channels > 1) {
+ snd_sbdsp_command(chip, 256 - runtime->rate_den / 2);
+ spin_lock(&chip->mixer_lock);
+ /* save input filter status and turn it off */
+ mixreg = snd_sbmixer_read(chip, SB_DSP_CAPTURE_FILT);
+ snd_sbmixer_write(chip, SB_DSP_CAPTURE_FILT, mixreg | 0x20);
+ spin_unlock(&chip->mixer_lock);
+ /* just use force_mode16 for temporary storate... */
+ chip->force_mode16 = mixreg;
+ } else {
+ snd_sbdsp_command(chip, 256 - runtime->rate_den);
+ }
+ if (chip->capture_format != SB_DSP_INPUT) {
+ count--;
+ snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE);
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_dma_program(chip->dma8, runtime->dma_addr,
+ size, DMA_MODE_READ | DMA_AUTOINIT);
+ return 0;
+}
+
+static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ unsigned int count;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_sbdsp_command(chip, chip->capture_format);
+ if (chip->capture_format == SB_DSP_INPUT) {
+ count = chip->c_period_size - 1;
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (chip->capture_format == SB_DSP_HI_INPUT_AUTO) {
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_sbdsp_reset(chip);
+ if (runtime->channels > 1) {
+ /* restore input filter status */
+ spin_lock(&chip->mixer_lock);
+ snd_sbmixer_write(chip, SB_DSP_CAPTURE_FILT, chip->force_mode16);
+ spin_unlock(&chip->mixer_lock);
+ /* set hardware to mono mode */
+ snd_sbdsp_command(chip, SB_DSP_MONO_8BIT);
+ }
+ } else {
+ snd_sbdsp_command(chip, SB_DSP_DMA8_OFF);
+ }
+ snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_CAPTURE_8 : SB_MODE_HALT;
+ return 0;
+}
+
+irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+
+ snd_sb_ack_8bit(chip);
+ switch (chip->mode) {
+ case SB_MODE_PLAYBACK_8: /* ok.. playback is active */
+ substream = chip->playback_substream;
+ runtime = substream->runtime;
+ if (chip->playback_format == SB_DSP_OUTPUT)
+ snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START);
+ snd_pcm_period_elapsed(substream);
+ break;
+ case SB_MODE_CAPTURE_8:
+ substream = chip->capture_substream;
+ runtime = substream->runtime;
+ if (chip->capture_format == SB_DSP_INPUT)
+ snd_sb8_capture_trigger(substream, SNDRV_PCM_TRIGGER_START);
+ snd_pcm_period_elapsed(substream);
+ break;
+ }
+ return IRQ_HANDLED;
+}
+
+static snd_pcm_uframes_t snd_sb8_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (chip->mode != SB_MODE_PLAYBACK_8)
+ return 0;
+ ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_sb8_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (chip->mode != SB_MODE_CAPTURE_8)
+ return 0;
+ ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+/*
+
+ */
+
+static struct snd_pcm_hardware snd_sb8_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8,
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_22050),
+ .rate_min = 4000,
+ .rate_max = 23000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_sb8_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8,
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_11025),
+ .rate_min = 4000,
+ .rate_max = 13000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+ *
+ */
+
+static int snd_sb8_open(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (chip->open) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return -EAGAIN;
+ }
+ chip->open |= SB_OPEN_PCM;
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ chip->playback_substream = substream;
+ runtime->hw = snd_sb8_playback;
+ } else {
+ chip->capture_substream = substream;
+ runtime->hw = snd_sb8_capture;
+ }
+ switch (chip->hardware) {
+ case SB_HW_PRO:
+ runtime->hw.rate_max = 44100;
+ runtime->hw.channels_max = 2;
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_sb8_hw_constraint_rate_channels, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_sb8_hw_constraint_channels_rate, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ break;
+ case SB_HW_201:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw.rate_max = 44100;
+ } else {
+ runtime->hw.rate_max = 15000;
+ }
+ default:
+ break;
+ }
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_clock);
+ return 0;
+}
+
+static int snd_sb8_close(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+
+ chip->playback_substream = NULL;
+ chip->capture_substream = NULL;
+ spin_lock_irqsave(&chip->open_lock, flags);
+ chip->open &= ~SB_OPEN_PCM;
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return 0;
+}
+
+/*
+ * Initialization part
+ */
+
+static struct snd_pcm_ops snd_sb8_playback_ops = {
+ .open = snd_sb8_open,
+ .close = snd_sb8_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sb8_hw_params,
+ .hw_free = snd_sb8_hw_free,
+ .prepare = snd_sb8_playback_prepare,
+ .trigger = snd_sb8_playback_trigger,
+ .pointer = snd_sb8_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_sb8_capture_ops = {
+ .open = snd_sb8_open,
+ .close = snd_sb8_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sb8_hw_params,
+ .hw_free = snd_sb8_hw_free,
+ .prepare = snd_sb8_capture_prepare,
+ .trigger = snd_sb8_capture_trigger,
+ .pointer = snd_sb8_capture_pointer,
+};
+
+int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_card *card = chip->card;
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(card, "SB8 DSP", device, 1, 1, &pcm)) < 0)
+ return err;
+ sprintf(pcm->name, "DSP v%i.%i", chip->version >> 8, chip->version & 0xff);
+ pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, 64*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_sb8dsp_pcm);
+EXPORT_SYMBOL(snd_sb8dsp_interrupt);
+ /* sb8_midi.c */
+EXPORT_SYMBOL(snd_sb8dsp_midi_interrupt);
+EXPORT_SYMBOL(snd_sb8dsp_midi);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_sb8_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_sb8_exit(void)
+{
+}
+
+module_init(alsa_sb8_init)
+module_exit(alsa_sb8_exit)
diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c
new file mode 100644
index 0000000..988a8b7
--- /dev/null
+++ b/sound/isa/sb/sb8_midi.c
@@ -0,0 +1,286 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of SoundBlaster cards - MIDI interface
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * --
+ *
+ * Sun May 9 22:54:38 BST 1999 George David Morrison <gdm@gedamo.demon.co.uk>
+ * Fixed typo in snd_sb8dsp_midi_new_device which prevented midi from
+ * working.
+ *
+ * Sun May 11 12:34:56 UTC 2003 Clemens Ladisch <clemens@ladisch.de>
+ * Added full duplex UART mode for DSP version 2.0 and later.
+ */
+
+#include <asm/io.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+
+
+irqreturn_t snd_sb8dsp_midi_interrupt(struct snd_sb *chip)
+{
+ struct snd_rawmidi *rmidi;
+ int max = 64;
+ char byte;
+
+ if (!chip)
+ return IRQ_NONE;
+
+ rmidi = chip->rmidi;
+ if (!rmidi) {
+ inb(SBP(chip, DATA_AVAIL)); /* ack interrupt */
+ return IRQ_NONE;
+ }
+
+ spin_lock(&chip->midi_input_lock);
+ while (max-- > 0) {
+ if (inb(SBP(chip, DATA_AVAIL)) & 0x80) {
+ byte = inb(SBP(chip, READ));
+ if (chip->open & SB_OPEN_MIDI_INPUT_TRIGGER) {
+ snd_rawmidi_receive(chip->midi_substream_input, &byte, 1);
+ }
+ }
+ }
+ spin_unlock(&chip->midi_input_lock);
+ return IRQ_HANDLED;
+}
+
+static int snd_sb8dsp_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip;
+ unsigned int valid_open_flags;
+
+ chip = substream->rmidi->private_data;
+ valid_open_flags = chip->hardware >= SB_HW_20
+ ? SB_OPEN_MIDI_OUTPUT | SB_OPEN_MIDI_OUTPUT_TRIGGER : 0;
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (chip->open & ~valid_open_flags) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return -EAGAIN;
+ }
+ chip->open |= SB_OPEN_MIDI_INPUT;
+ chip->midi_substream_input = substream;
+ if (!(chip->open & SB_OPEN_MIDI_OUTPUT)) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ snd_sbdsp_reset(chip); /* reset DSP */
+ if (chip->hardware >= SB_HW_20)
+ snd_sbdsp_command(chip, SB_DSP_MIDI_UART_IRQ);
+ } else {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ }
+ return 0;
+}
+
+static int snd_sb8dsp_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip;
+ unsigned int valid_open_flags;
+
+ chip = substream->rmidi->private_data;
+ valid_open_flags = chip->hardware >= SB_HW_20
+ ? SB_OPEN_MIDI_INPUT | SB_OPEN_MIDI_INPUT_TRIGGER : 0;
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (chip->open & ~valid_open_flags) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ return -EAGAIN;
+ }
+ chip->open |= SB_OPEN_MIDI_OUTPUT;
+ chip->midi_substream_output = substream;
+ if (!(chip->open & SB_OPEN_MIDI_INPUT)) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ snd_sbdsp_reset(chip); /* reset DSP */
+ if (chip->hardware >= SB_HW_20)
+ snd_sbdsp_command(chip, SB_DSP_MIDI_UART_IRQ);
+ } else {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ }
+ return 0;
+}
+
+static int snd_sb8dsp_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip;
+
+ chip = substream->rmidi->private_data;
+ spin_lock_irqsave(&chip->open_lock, flags);
+ chip->open &= ~(SB_OPEN_MIDI_INPUT | SB_OPEN_MIDI_INPUT_TRIGGER);
+ chip->midi_substream_input = NULL;
+ if (!(chip->open & SB_OPEN_MIDI_OUTPUT)) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ snd_sbdsp_reset(chip); /* reset DSP */
+ } else {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ }
+ return 0;
+}
+
+static int snd_sb8dsp_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip;
+
+ chip = substream->rmidi->private_data;
+ spin_lock_irqsave(&chip->open_lock, flags);
+ chip->open &= ~(SB_OPEN_MIDI_OUTPUT | SB_OPEN_MIDI_OUTPUT_TRIGGER);
+ chip->midi_substream_output = NULL;
+ if (!(chip->open & SB_OPEN_MIDI_INPUT)) {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ snd_sbdsp_reset(chip); /* reset DSP */
+ } else {
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ }
+ return 0;
+}
+
+static void snd_sb8dsp_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct snd_sb *chip;
+
+ chip = substream->rmidi->private_data;
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (up) {
+ if (!(chip->open & SB_OPEN_MIDI_INPUT_TRIGGER)) {
+ if (chip->hardware < SB_HW_20)
+ snd_sbdsp_command(chip, SB_DSP_MIDI_INPUT_IRQ);
+ chip->open |= SB_OPEN_MIDI_INPUT_TRIGGER;
+ }
+ } else {
+ if (chip->open & SB_OPEN_MIDI_INPUT_TRIGGER) {
+ if (chip->hardware < SB_HW_20)
+ snd_sbdsp_command(chip, SB_DSP_MIDI_INPUT_IRQ);
+ chip->open &= ~SB_OPEN_MIDI_INPUT_TRIGGER;
+ }
+ }
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+}
+
+static void snd_sb8dsp_midi_output_write(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ struct snd_sb *chip;
+ char byte;
+ int max = 32;
+
+ /* how big is Tx FIFO? */
+ chip = substream->rmidi->private_data;
+ while (max-- > 0) {
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (snd_rawmidi_transmit_peek(substream, &byte, 1) != 1) {
+ chip->open &= ~SB_OPEN_MIDI_OUTPUT_TRIGGER;
+ del_timer(&chip->midi_timer);
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ break;
+ }
+ if (chip->hardware >= SB_HW_20) {
+ int timeout = 8;
+ while ((inb(SBP(chip, STATUS)) & 0x80) != 0 && --timeout > 0)
+ ;
+ if (timeout == 0) {
+ /* Tx FIFO full - try again later */
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ break;
+ }
+ outb(byte, SBP(chip, WRITE));
+ } else {
+ snd_sbdsp_command(chip, SB_DSP_MIDI_OUTPUT);
+ snd_sbdsp_command(chip, byte);
+ }
+ snd_rawmidi_transmit_ack(substream, 1);
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ }
+}
+
+static void snd_sb8dsp_midi_output_timer(unsigned long data)
+{
+ struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *) data;
+ struct snd_sb * chip = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->open_lock, flags);
+ chip->midi_timer.expires = 1 + jiffies;
+ add_timer(&chip->midi_timer);
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+ snd_sb8dsp_midi_output_write(substream);
+}
+
+static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct snd_sb *chip;
+
+ chip = substream->rmidi->private_data;
+ spin_lock_irqsave(&chip->open_lock, flags);
+ if (up) {
+ if (!(chip->open & SB_OPEN_MIDI_OUTPUT_TRIGGER)) {
+ init_timer(&chip->midi_timer);
+ chip->midi_timer.function = snd_sb8dsp_midi_output_timer;
+ chip->midi_timer.data = (unsigned long) substream;
+ chip->midi_timer.expires = 1 + jiffies;
+ add_timer(&chip->midi_timer);
+ chip->open |= SB_OPEN_MIDI_OUTPUT_TRIGGER;
+ }
+ } else {
+ if (chip->open & SB_OPEN_MIDI_OUTPUT_TRIGGER) {
+ chip->open &= ~SB_OPEN_MIDI_OUTPUT_TRIGGER;
+ }
+ }
+ spin_unlock_irqrestore(&chip->open_lock, flags);
+
+ if (up)
+ snd_sb8dsp_midi_output_write(substream);
+}
+
+static struct snd_rawmidi_ops snd_sb8dsp_midi_output =
+{
+ .open = snd_sb8dsp_midi_output_open,
+ .close = snd_sb8dsp_midi_output_close,
+ .trigger = snd_sb8dsp_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_sb8dsp_midi_input =
+{
+ .open = snd_sb8dsp_midi_input_open,
+ .close = snd_sb8dsp_midi_input_close,
+ .trigger = snd_sb8dsp_midi_input_trigger,
+};
+
+int snd_sb8dsp_midi(struct snd_sb *chip, int device, struct snd_rawmidi ** rrawmidi)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ if ((err = snd_rawmidi_new(chip->card, "SB8 MIDI", device, 1, 1, &rmidi)) < 0)
+ return err;
+ strcpy(rmidi->name, "SB8 MIDI");
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_sb8dsp_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_sb8dsp_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT;
+ if (chip->hardware >= SB_HW_20)
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = chip;
+ chip->rmidi = rmidi;
+ if (rrawmidi)
+ *rrawmidi = rmidi;
+ return 0;
+}
diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
new file mode 100644
index 0000000..27a6515
--- /dev/null
+++ b/sound/isa/sb/sb_common.c
@@ -0,0 +1,320 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Uros Bizjak <uros@kss-loka.si>
+ *
+ * Lowlevel routines for control of Sound Blaster cards
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+#include <asm/dma.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("ALSA lowlevel driver for Sound Blaster cards");
+MODULE_LICENSE("GPL");
+
+#define BUSY_LOOPS 100000
+
+#undef IO_DEBUG
+
+int snd_sbdsp_command(struct snd_sb *chip, unsigned char val)
+{
+ int i;
+#ifdef IO_DEBUG
+ snd_printk(KERN_DEBUG "command 0x%x\n", val);
+#endif
+ for (i = BUSY_LOOPS; i; i--)
+ if ((inb(SBP(chip, STATUS)) & 0x80) == 0) {
+ outb(val, SBP(chip, COMMAND));
+ return 1;
+ }
+ snd_printd("%s [0x%lx]: timeout (0x%x)\n", __func__, chip->port, val);
+ return 0;
+}
+
+int snd_sbdsp_get_byte(struct snd_sb *chip)
+{
+ int val;
+ int i;
+ for (i = BUSY_LOOPS; i; i--) {
+ if (inb(SBP(chip, DATA_AVAIL)) & 0x80) {
+ val = inb(SBP(chip, READ));
+#ifdef IO_DEBUG
+ snd_printk(KERN_DEBUG "get_byte 0x%x\n", val);
+#endif
+ return val;
+ }
+ }
+ snd_printd("%s [0x%lx]: timeout\n", __func__, chip->port);
+ return -ENODEV;
+}
+
+int snd_sbdsp_reset(struct snd_sb *chip)
+{
+ int i;
+
+ outb(1, SBP(chip, RESET));
+ udelay(10);
+ outb(0, SBP(chip, RESET));
+ udelay(30);
+ for (i = BUSY_LOOPS; i; i--)
+ if (inb(SBP(chip, DATA_AVAIL)) & 0x80) {
+ if (inb(SBP(chip, READ)) == 0xaa)
+ return 0;
+ else
+ break;
+ }
+ snd_printdd("%s [0x%lx] failed...\n", __func__, chip->port);
+ return -ENODEV;
+}
+
+static int snd_sbdsp_version(struct snd_sb * chip)
+{
+ unsigned int result = -ENODEV;
+
+ snd_sbdsp_command(chip, SB_DSP_GET_VERSION);
+ result = (short) snd_sbdsp_get_byte(chip) << 8;
+ result |= (short) snd_sbdsp_get_byte(chip);
+ return result;
+}
+
+static int snd_sbdsp_probe(struct snd_sb * chip)
+{
+ int version;
+ int major, minor;
+ char *str;
+ unsigned long flags;
+
+ /*
+ * initialization sequence
+ */
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (snd_sbdsp_reset(chip) < 0) {
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return -ENODEV;
+ }
+ version = snd_sbdsp_version(chip);
+ if (version < 0) {
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return -ENODEV;
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ major = version >> 8;
+ minor = version & 0xff;
+ snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n",
+ chip->port, major, minor);
+
+ switch (chip->hardware) {
+ case SB_HW_AUTO:
+ switch (major) {
+ case 1:
+ chip->hardware = SB_HW_10;
+ str = "1.0";
+ break;
+ case 2:
+ if (minor) {
+ chip->hardware = SB_HW_201;
+ str = "2.01+";
+ } else {
+ chip->hardware = SB_HW_20;
+ str = "2.0";
+ }
+ break;
+ case 3:
+ chip->hardware = SB_HW_PRO;
+ str = "Pro";
+ break;
+ case 4:
+ chip->hardware = SB_HW_16;
+ str = "16";
+ break;
+ default:
+ snd_printk(KERN_INFO "SB [0x%lx]: unknown DSP chip version %i.%i\n",
+ chip->port, major, minor);
+ return -ENODEV;
+ }
+ break;
+ case SB_HW_ALS100:
+ str = "16 (ALS-100)";
+ break;
+ case SB_HW_ALS4000:
+ str = "16 (ALS-4000)";
+ break;
+ case SB_HW_DT019X:
+ str = "(DT019X/ALS007)";
+ break;
+ case SB_HW_CS5530:
+ str = "16 (CS5530)";
+ break;
+ default:
+ return -ENODEV;
+ }
+ sprintf(chip->name, "Sound Blaster %s", str);
+ chip->version = (major << 8) | minor;
+ return 0;
+}
+
+static int snd_sbdsp_free(struct snd_sb *chip)
+{
+ if (chip->res_port)
+ release_and_free_resource(chip->res_port);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *) chip);
+#ifdef CONFIG_ISA
+ if (chip->dma8 >= 0) {
+ disable_dma(chip->dma8);
+ free_dma(chip->dma8);
+ }
+ if (chip->dma16 >= 0 && chip->dma16 != chip->dma8) {
+ disable_dma(chip->dma16);
+ free_dma(chip->dma16);
+ }
+#endif
+ kfree(chip);
+ return 0;
+}
+
+static int snd_sbdsp_dev_free(struct snd_device *device)
+{
+ struct snd_sb *chip = device->device_data;
+ return snd_sbdsp_free(chip);
+}
+
+int snd_sbdsp_create(struct snd_card *card,
+ unsigned long port,
+ int irq,
+ irq_handler_t irq_handler,
+ int dma8,
+ int dma16,
+ unsigned short hardware,
+ struct snd_sb **r_chip)
+{
+ struct snd_sb *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_sbdsp_dev_free,
+ };
+
+ if (snd_BUG_ON(!r_chip))
+ return -EINVAL;
+ *r_chip = NULL;
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->open_lock);
+ spin_lock_init(&chip->midi_input_lock);
+ spin_lock_init(&chip->mixer_lock);
+ chip->irq = -1;
+ chip->dma8 = -1;
+ chip->dma16 = -1;
+ chip->port = port;
+
+ if (request_irq(irq, irq_handler,
+ (hardware == SB_HW_ALS4000 ||
+ hardware == SB_HW_CS5530) ?
+ IRQF_SHARED : IRQF_DISABLED,
+ "SoundBlaster", (void *) chip)) {
+ snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq);
+ snd_sbdsp_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = irq;
+
+ if (hardware == SB_HW_ALS4000)
+ goto __skip_allocation;
+
+ if ((chip->res_port = request_region(port, 16, "SoundBlaster")) == NULL) {
+ snd_printk(KERN_ERR "sb: can't grab port 0x%lx\n", port);
+ snd_sbdsp_free(chip);
+ return -EBUSY;
+ }
+
+#ifdef CONFIG_ISA
+ if (dma8 >= 0 && request_dma(dma8, "SoundBlaster - 8bit")) {
+ snd_printk(KERN_ERR "sb: can't grab DMA8 %d\n", dma8);
+ snd_sbdsp_free(chip);
+ return -EBUSY;
+ }
+ chip->dma8 = dma8;
+ if (dma16 >= 0) {
+ if (hardware != SB_HW_ALS100 && (dma16 < 5 || dma16 > 7)) {
+ /* no duplex */
+ dma16 = -1;
+ } else if (request_dma(dma16, "SoundBlaster - 16bit")) {
+ snd_printk(KERN_ERR "sb: can't grab DMA16 %d\n", dma16);
+ snd_sbdsp_free(chip);
+ return -EBUSY;
+ }
+ }
+ chip->dma16 = dma16;
+#endif
+
+ __skip_allocation:
+ chip->card = card;
+ chip->hardware = hardware;
+ if ((err = snd_sbdsp_probe(chip)) < 0) {
+ snd_sbdsp_free(chip);
+ return err;
+ }
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_sbdsp_free(chip);
+ return err;
+ }
+ *r_chip = chip;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_sbdsp_command);
+EXPORT_SYMBOL(snd_sbdsp_get_byte);
+EXPORT_SYMBOL(snd_sbdsp_reset);
+EXPORT_SYMBOL(snd_sbdsp_create);
+/* sb_mixer.c */
+EXPORT_SYMBOL(snd_sbmixer_write);
+EXPORT_SYMBOL(snd_sbmixer_read);
+EXPORT_SYMBOL(snd_sbmixer_new);
+EXPORT_SYMBOL(snd_sbmixer_add_ctl);
+#ifdef CONFIG_PM
+EXPORT_SYMBOL(snd_sbmixer_suspend);
+EXPORT_SYMBOL(snd_sbmixer_resume);
+#endif
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_sb_common_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_sb_common_exit(void)
+{
+}
+
+module_init(alsa_sb_common_init)
+module_exit(alsa_sb_common_exit)
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
new file mode 100644
index 0000000..406a431
--- /dev/null
+++ b/sound/isa/sb/sb_mixer.c
@@ -0,0 +1,995 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for Sound Blaster mixer control
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+#include <sound/control.h>
+
+#undef IO_DEBUG
+
+void snd_sbmixer_write(struct snd_sb *chip, unsigned char reg, unsigned char data)
+{
+ outb(reg, SBP(chip, MIXER_ADDR));
+ udelay(10);
+ outb(data, SBP(chip, MIXER_DATA));
+ udelay(10);
+#ifdef IO_DEBUG
+ snd_printk(KERN_DEBUG "mixer_write 0x%x 0x%x\n", reg, data);
+#endif
+}
+
+unsigned char snd_sbmixer_read(struct snd_sb *chip, unsigned char reg)
+{
+ unsigned char result;
+
+ outb(reg, SBP(chip, MIXER_ADDR));
+ udelay(10);
+ result = inb(SBP(chip, MIXER_DATA));
+ udelay(10);
+#ifdef IO_DEBUG
+ snd_printk(KERN_DEBUG "mixer_read 0x%x 0x%x\n", reg, result);
+#endif
+ return result;
+}
+
+/*
+ * Single channel mixer element
+ */
+
+static int snd_sbmixer_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 16) & 0xff;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ unsigned char val;
+
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ val = (snd_sbmixer_read(sb, reg) >> shift) & mask;
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+}
+
+static int snd_sbmixer_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 16) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned char val, oval;
+
+ val = (ucontrol->value.integer.value[0] & mask) << shift;
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ oval = snd_sbmixer_read(sb, reg);
+ val = (oval & ~(mask << shift)) | val;
+ change = val != oval;
+ if (change)
+ snd_sbmixer_write(sb, reg, val);
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ return change;
+}
+
+/*
+ * Double channel mixer element
+ */
+
+static int snd_sbmixer_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_sbmixer_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x07;
+ int right_shift = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ unsigned char left, right;
+
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ left = (snd_sbmixer_read(sb, left_reg) >> left_shift) & mask;
+ right = (snd_sbmixer_read(sb, right_reg) >> right_shift) & mask;
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ ucontrol->value.integer.value[0] = left;
+ ucontrol->value.integer.value[1] = right;
+ return 0;
+}
+
+static int snd_sbmixer_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x07;
+ int right_shift = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned char left, right, oleft, oright;
+
+ left = (ucontrol->value.integer.value[0] & mask) << left_shift;
+ right = (ucontrol->value.integer.value[1] & mask) << right_shift;
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ if (left_reg == right_reg) {
+ oleft = snd_sbmixer_read(sb, left_reg);
+ left = (oleft & ~((mask << left_shift) | (mask << right_shift))) | left | right;
+ change = left != oleft;
+ if (change)
+ snd_sbmixer_write(sb, left_reg, left);
+ } else {
+ oleft = snd_sbmixer_read(sb, left_reg);
+ oright = snd_sbmixer_read(sb, right_reg);
+ left = (oleft & ~(mask << left_shift)) | left;
+ right = (oright & ~(mask << right_shift)) | right;
+ change = left != oleft || right != oright;
+ if (change) {
+ snd_sbmixer_write(sb, left_reg, left);
+ snd_sbmixer_write(sb, right_reg, right);
+ }
+ }
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ return change;
+}
+
+/*
+ * DT-019x / ALS-007 capture/input switch
+ */
+
+static int snd_dt019x_input_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[5] = {
+ "CD", "Mic", "Line", "Synth", "Master"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 5;
+ if (uinfo->value.enumerated.item > 4)
+ uinfo->value.enumerated.item = 4;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_dt019x_input_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned char oval;
+
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ oval = snd_sbmixer_read(sb, SB_DT019X_CAPTURE_SW);
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ switch (oval & 0x07) {
+ case SB_DT019X_CAP_CD:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ case SB_DT019X_CAP_MIC:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ case SB_DT019X_CAP_LINE:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case SB_DT019X_CAP_MAIN:
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ /* To record the synth on these cards you must record the main. */
+ /* Thus SB_DT019X_CAP_SYNTH == SB_DT019X_CAP_MAIN and would cause */
+ /* duplicate case labels if left uncommented. */
+ /* case SB_DT019X_CAP_SYNTH:
+ * ucontrol->value.enumerated.item[0] = 3;
+ * break;
+ */
+ default:
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ }
+ return 0;
+}
+
+static int snd_dt019x_input_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned char nval, oval;
+
+ if (ucontrol->value.enumerated.item[0] > 4)
+ return -EINVAL;
+ switch (ucontrol->value.enumerated.item[0]) {
+ case 0:
+ nval = SB_DT019X_CAP_CD;
+ break;
+ case 1:
+ nval = SB_DT019X_CAP_MIC;
+ break;
+ case 2:
+ nval = SB_DT019X_CAP_LINE;
+ break;
+ case 3:
+ nval = SB_DT019X_CAP_SYNTH;
+ break;
+ case 4:
+ nval = SB_DT019X_CAP_MAIN;
+ break;
+ default:
+ nval = SB_DT019X_CAP_MAIN;
+ }
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ oval = snd_sbmixer_read(sb, SB_DT019X_CAPTURE_SW);
+ change = nval != oval;
+ if (change)
+ snd_sbmixer_write(sb, SB_DT019X_CAPTURE_SW, nval);
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ return change;
+}
+
+/*
+ * SBPRO input multiplexer
+ */
+
+static int snd_sb8mixer_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {
+ "Mic", "CD", "Line"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+
+static int snd_sb8mixer_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned char oval;
+
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ oval = snd_sbmixer_read(sb, SB_DSP_CAPTURE_SOURCE);
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ switch ((oval >> 0x01) & 0x03) {
+ case SB_DSP_MIXS_CD:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ case SB_DSP_MIXS_LINE:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ }
+ return 0;
+}
+
+static int snd_sb8mixer_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int change;
+ unsigned char nval, oval;
+
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ switch (ucontrol->value.enumerated.item[0]) {
+ case 1:
+ nval = SB_DSP_MIXS_CD;
+ break;
+ case 2:
+ nval = SB_DSP_MIXS_LINE;
+ break;
+ default:
+ nval = SB_DSP_MIXS_MIC;
+ }
+ nval <<= 1;
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ oval = snd_sbmixer_read(sb, SB_DSP_CAPTURE_SOURCE);
+ nval |= oval & ~0x06;
+ change = nval != oval;
+ if (change)
+ snd_sbmixer_write(sb, SB_DSP_CAPTURE_SOURCE, nval);
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ return change;
+}
+
+/*
+ * SB16 input switch
+ */
+
+static int snd_sb16mixer_info_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 4;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_sb16mixer_get_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg1 = kcontrol->private_value & 0xff;
+ int reg2 = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x0f;
+ int right_shift = (kcontrol->private_value >> 24) & 0x0f;
+ unsigned char val1, val2;
+
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ val1 = snd_sbmixer_read(sb, reg1);
+ val2 = snd_sbmixer_read(sb, reg2);
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ ucontrol->value.integer.value[0] = (val1 >> left_shift) & 0x01;
+ ucontrol->value.integer.value[1] = (val2 >> left_shift) & 0x01;
+ ucontrol->value.integer.value[2] = (val1 >> right_shift) & 0x01;
+ ucontrol->value.integer.value[3] = (val2 >> right_shift) & 0x01;
+ return 0;
+}
+
+static int snd_sb16mixer_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg1 = kcontrol->private_value & 0xff;
+ int reg2 = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x0f;
+ int right_shift = (kcontrol->private_value >> 24) & 0x0f;
+ int change;
+ unsigned char val1, val2, oval1, oval2;
+
+ spin_lock_irqsave(&sb->mixer_lock, flags);
+ oval1 = snd_sbmixer_read(sb, reg1);
+ oval2 = snd_sbmixer_read(sb, reg2);
+ val1 = oval1 & ~((1 << left_shift) | (1 << right_shift));
+ val2 = oval2 & ~((1 << left_shift) | (1 << right_shift));
+ val1 |= (ucontrol->value.integer.value[0] & 1) << left_shift;
+ val2 |= (ucontrol->value.integer.value[1] & 1) << left_shift;
+ val1 |= (ucontrol->value.integer.value[2] & 1) << right_shift;
+ val2 |= (ucontrol->value.integer.value[3] & 1) << right_shift;
+ change = val1 != oval1 || val2 != oval2;
+ if (change) {
+ snd_sbmixer_write(sb, reg1, val1);
+ snd_sbmixer_write(sb, reg2, val2);
+ }
+ spin_unlock_irqrestore(&sb->mixer_lock, flags);
+ return change;
+}
+
+
+/*
+ */
+/*
+ */
+int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int type, unsigned long value)
+{
+ static struct snd_kcontrol_new newctls[] = {
+ [SB_MIX_SINGLE] = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_sbmixer_info_single,
+ .get = snd_sbmixer_get_single,
+ .put = snd_sbmixer_put_single,
+ },
+ [SB_MIX_DOUBLE] = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_sbmixer_info_double,
+ .get = snd_sbmixer_get_double,
+ .put = snd_sbmixer_put_double,
+ },
+ [SB_MIX_INPUT_SW] = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_sb16mixer_info_input_sw,
+ .get = snd_sb16mixer_get_input_sw,
+ .put = snd_sb16mixer_put_input_sw,
+ },
+ [SB_MIX_CAPTURE_PRO] = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_sb8mixer_info_mux,
+ .get = snd_sb8mixer_get_mux,
+ .put = snd_sb8mixer_put_mux,
+ },
+ [SB_MIX_CAPTURE_DT019X] = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_dt019x_input_sw_info,
+ .get = snd_dt019x_input_sw_get,
+ .put = snd_dt019x_input_sw_put,
+ },
+ };
+ struct snd_kcontrol *ctl;
+ int err;
+
+ ctl = snd_ctl_new1(&newctls[type], chip);
+ if (! ctl)
+ return -ENOMEM;
+ strlcpy(ctl->id.name, name, sizeof(ctl->id.name));
+ ctl->id.index = index;
+ ctl->private_value = value;
+ if ((err = snd_ctl_add(chip->card, ctl)) < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * SB 2.0 specific mixer elements
+ */
+
+static struct sbmix_elem snd_sb20_ctl_master_play_vol =
+ SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7);
+static struct sbmix_elem snd_sb20_ctl_pcm_play_vol =
+ SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3);
+static struct sbmix_elem snd_sb20_ctl_synth_play_vol =
+ SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7);
+static struct sbmix_elem snd_sb20_ctl_cd_play_vol =
+ SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7);
+
+static struct sbmix_elem *snd_sb20_controls[] = {
+ &snd_sb20_ctl_master_play_vol,
+ &snd_sb20_ctl_pcm_play_vol,
+ &snd_sb20_ctl_synth_play_vol,
+ &snd_sb20_ctl_cd_play_vol
+};
+
+static unsigned char snd_sb20_init_values[][2] = {
+ { SB_DSP20_MASTER_DEV, 0 },
+ { SB_DSP20_FM_DEV, 0 },
+};
+
+/*
+ * SB Pro specific mixer elements
+ */
+static struct sbmix_elem snd_sbpro_ctl_master_play_vol =
+ SB_DOUBLE("Master Playback Volume", SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7);
+static struct sbmix_elem snd_sbpro_ctl_pcm_play_vol =
+ SB_DOUBLE("PCM Playback Volume", SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7);
+static struct sbmix_elem snd_sbpro_ctl_pcm_play_filter =
+ SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1);
+static struct sbmix_elem snd_sbpro_ctl_synth_play_vol =
+ SB_DOUBLE("Synth Playback Volume", SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7);
+static struct sbmix_elem snd_sbpro_ctl_cd_play_vol =
+ SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7);
+static struct sbmix_elem snd_sbpro_ctl_line_play_vol =
+ SB_DOUBLE("Line Playback Volume", SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7);
+static struct sbmix_elem snd_sbpro_ctl_mic_play_vol =
+ SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3);
+static struct sbmix_elem snd_sbpro_ctl_capture_source =
+ {
+ .name = "Capture Source",
+ .type = SB_MIX_CAPTURE_PRO
+ };
+static struct sbmix_elem snd_sbpro_ctl_capture_filter =
+ SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1);
+static struct sbmix_elem snd_sbpro_ctl_capture_low_filter =
+ SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1);
+
+static struct sbmix_elem *snd_sbpro_controls[] = {
+ &snd_sbpro_ctl_master_play_vol,
+ &snd_sbpro_ctl_pcm_play_vol,
+ &snd_sbpro_ctl_pcm_play_filter,
+ &snd_sbpro_ctl_synth_play_vol,
+ &snd_sbpro_ctl_cd_play_vol,
+ &snd_sbpro_ctl_line_play_vol,
+ &snd_sbpro_ctl_mic_play_vol,
+ &snd_sbpro_ctl_capture_source,
+ &snd_sbpro_ctl_capture_filter,
+ &snd_sbpro_ctl_capture_low_filter
+};
+
+static unsigned char snd_sbpro_init_values[][2] = {
+ { SB_DSP_MASTER_DEV, 0 },
+ { SB_DSP_PCM_DEV, 0 },
+ { SB_DSP_FM_DEV, 0 },
+};
+
+/*
+ * SB16 specific mixer elements
+ */
+static struct sbmix_elem snd_sb16_ctl_master_play_vol =
+ SB_DOUBLE("Master Playback Volume", SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31);
+static struct sbmix_elem snd_sb16_ctl_3d_enhance_switch =
+ SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1);
+static struct sbmix_elem snd_sb16_ctl_tone_bass =
+ SB_DOUBLE("Tone Control - Bass", SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15);
+static struct sbmix_elem snd_sb16_ctl_tone_treble =
+ SB_DOUBLE("Tone Control - Treble", SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15);
+static struct sbmix_elem snd_sb16_ctl_pcm_play_vol =
+ SB_DOUBLE("PCM Playback Volume", SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31);
+static struct sbmix_elem snd_sb16_ctl_synth_capture_route =
+ SB16_INPUT_SW("Synth Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5);
+static struct sbmix_elem snd_sb16_ctl_synth_play_vol =
+ SB_DOUBLE("Synth Playback Volume", SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31);
+static struct sbmix_elem snd_sb16_ctl_cd_capture_route =
+ SB16_INPUT_SW("CD Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1);
+static struct sbmix_elem snd_sb16_ctl_cd_play_switch =
+ SB_DOUBLE("CD Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1);
+static struct sbmix_elem snd_sb16_ctl_cd_play_vol =
+ SB_DOUBLE("CD Playback Volume", SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31);
+static struct sbmix_elem snd_sb16_ctl_line_capture_route =
+ SB16_INPUT_SW("Line Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3);
+static struct sbmix_elem snd_sb16_ctl_line_play_switch =
+ SB_DOUBLE("Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1);
+static struct sbmix_elem snd_sb16_ctl_line_play_vol =
+ SB_DOUBLE("Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31);
+static struct sbmix_elem snd_sb16_ctl_mic_capture_route =
+ SB16_INPUT_SW("Mic Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0);
+static struct sbmix_elem snd_sb16_ctl_mic_play_switch =
+ SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1);
+static struct sbmix_elem snd_sb16_ctl_mic_play_vol =
+ SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31);
+static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol =
+ SB_SINGLE("PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3);
+static struct sbmix_elem snd_sb16_ctl_capture_vol =
+ SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3);
+static struct sbmix_elem snd_sb16_ctl_play_vol =
+ SB_DOUBLE("Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3);
+static struct sbmix_elem snd_sb16_ctl_auto_mic_gain =
+ SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1);
+
+static struct sbmix_elem *snd_sb16_controls[] = {
+ &snd_sb16_ctl_master_play_vol,
+ &snd_sb16_ctl_3d_enhance_switch,
+ &snd_sb16_ctl_tone_bass,
+ &snd_sb16_ctl_tone_treble,
+ &snd_sb16_ctl_pcm_play_vol,
+ &snd_sb16_ctl_synth_capture_route,
+ &snd_sb16_ctl_synth_play_vol,
+ &snd_sb16_ctl_cd_capture_route,
+ &snd_sb16_ctl_cd_play_switch,
+ &snd_sb16_ctl_cd_play_vol,
+ &snd_sb16_ctl_line_capture_route,
+ &snd_sb16_ctl_line_play_switch,
+ &snd_sb16_ctl_line_play_vol,
+ &snd_sb16_ctl_mic_capture_route,
+ &snd_sb16_ctl_mic_play_switch,
+ &snd_sb16_ctl_mic_play_vol,
+ &snd_sb16_ctl_pc_speaker_vol,
+ &snd_sb16_ctl_capture_vol,
+ &snd_sb16_ctl_play_vol,
+ &snd_sb16_ctl_auto_mic_gain
+};
+
+static unsigned char snd_sb16_init_values[][2] = {
+ { SB_DSP4_MASTER_DEV + 0, 0 },
+ { SB_DSP4_MASTER_DEV + 1, 0 },
+ { SB_DSP4_PCM_DEV + 0, 0 },
+ { SB_DSP4_PCM_DEV + 1, 0 },
+ { SB_DSP4_SYNTH_DEV + 0, 0 },
+ { SB_DSP4_SYNTH_DEV + 1, 0 },
+ { SB_DSP4_INPUT_LEFT, 0 },
+ { SB_DSP4_INPUT_RIGHT, 0 },
+ { SB_DSP4_OUTPUT_SW, 0 },
+ { SB_DSP4_SPEAKER_DEV, 0 },
+};
+
+/*
+ * DT019x specific mixer elements
+ */
+static struct sbmix_elem snd_dt019x_ctl_master_play_vol =
+ SB_DOUBLE("Master Playback Volume", SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4,0, 15);
+static struct sbmix_elem snd_dt019x_ctl_pcm_play_vol =
+ SB_DOUBLE("PCM Playback Volume", SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4,0, 15);
+static struct sbmix_elem snd_dt019x_ctl_synth_play_vol =
+ SB_DOUBLE("Synth Playback Volume", SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4,0, 15);
+static struct sbmix_elem snd_dt019x_ctl_cd_play_vol =
+ SB_DOUBLE("CD Playback Volume", SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4,0, 15);
+static struct sbmix_elem snd_dt019x_ctl_mic_play_vol =
+ SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7);
+static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol =
+ SB_SINGLE("PC Speaker Volume", SB_DT019X_SPKR_DEV, 0, 7);
+static struct sbmix_elem snd_dt019x_ctl_line_play_vol =
+ SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15);
+static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch =
+ SB_DOUBLE("PCM Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2,1, 1);
+static struct sbmix_elem snd_dt019x_ctl_synth_play_switch =
+ SB_DOUBLE("Synth Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4,3, 1);
+static struct sbmix_elem snd_dt019x_ctl_capture_source =
+ {
+ .name = "Capture Source",
+ .type = SB_MIX_CAPTURE_DT019X
+ };
+
+static struct sbmix_elem *snd_dt019x_controls[] = {
+ &snd_dt019x_ctl_master_play_vol,
+ &snd_dt019x_ctl_pcm_play_vol,
+ &snd_dt019x_ctl_synth_play_vol,
+ &snd_dt019x_ctl_cd_play_vol,
+ &snd_dt019x_ctl_mic_play_vol,
+ &snd_dt019x_ctl_pc_speaker_vol,
+ &snd_dt019x_ctl_line_play_vol,
+ &snd_sb16_ctl_mic_play_switch,
+ &snd_sb16_ctl_cd_play_switch,
+ &snd_sb16_ctl_line_play_switch,
+ &snd_dt019x_ctl_pcm_play_switch,
+ &snd_dt019x_ctl_synth_play_switch,
+ &snd_dt019x_ctl_capture_source
+};
+
+static unsigned char snd_dt019x_init_values[][2] = {
+ { SB_DT019X_MASTER_DEV, 0 },
+ { SB_DT019X_PCM_DEV, 0 },
+ { SB_DT019X_SYNTH_DEV, 0 },
+ { SB_DT019X_CD_DEV, 0 },
+ { SB_DT019X_MIC_DEV, 0 }, /* Includes PC-speaker in high nibble */
+ { SB_DT019X_LINE_DEV, 0 },
+ { SB_DSP4_OUTPUT_SW, 0 },
+ { SB_DT019X_OUTPUT_SW2, 0 },
+ { SB_DT019X_CAPTURE_SW, 0x06 },
+};
+
+/*
+ * ALS4000 specific mixer elements
+ */
+/* FIXME: SB_ALS4000_MONO_IO_CTRL needs output select ctrl! */
+static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch =
+ SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1);
+static struct sbmix_elem snd_als4000_ctl_master_mono_capture_route =
+ SB_SINGLE("Master Mono Capture Route", SB_ALS4000_MONO_IO_CTRL, 6, 0x03);
+/* FIXME: mono playback switch also available on DT019X? */
+static struct sbmix_elem snd_als4000_ctl_mono_playback_switch =
+ SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1);
+static struct sbmix_elem snd_als4000_ctl_mic_20db_boost =
+ SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03);
+static struct sbmix_elem snd_als4000_ctl_mixer_loopback =
+ SB_SINGLE("Analog Loopback", SB_ALS4000_MIC_IN_GAIN, 7, 0x01);
+/* FIXME: functionality of 3D controls might be swapped, I didn't find
+ * a description of how to identify what is supposed to be what */
+static struct sbmix_elem snd_als4000_3d_control_switch =
+ SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01);
+static struct sbmix_elem snd_als4000_3d_control_ratio =
+ SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07);
+static struct sbmix_elem snd_als4000_3d_control_freq =
+ /* FIXME: maybe there's actually some standard 3D ctrl name for it?? */
+ SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03);
+static struct sbmix_elem snd_als4000_3d_control_delay =
+ /* FIXME: ALS4000a.pdf mentions BBD (Bucket Brigade Device) time delay,
+ * but what ALSA 3D attribute is that actually? "Center", "Depth",
+ * "Wide" or "Space" or even "Level"? Assuming "Wide" for now... */
+ SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f);
+static struct sbmix_elem snd_als4000_3d_control_poweroff_switch =
+ SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01);
+#ifdef NOT_AVAILABLE
+static struct sbmix_elem snd_als4000_ctl_fmdac =
+ SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01);
+static struct sbmix_elem snd_als4000_ctl_qsound =
+ SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f);
+#endif
+
+static struct sbmix_elem *snd_als4000_controls[] = {
+ &snd_sb16_ctl_master_play_vol,
+ &snd_dt019x_ctl_pcm_play_switch,
+ &snd_sb16_ctl_pcm_play_vol,
+ &snd_sb16_ctl_synth_capture_route,
+ &snd_dt019x_ctl_synth_play_switch,
+ &snd_sb16_ctl_synth_play_vol,
+ &snd_sb16_ctl_cd_capture_route,
+ &snd_sb16_ctl_cd_play_switch,
+ &snd_sb16_ctl_cd_play_vol,
+ &snd_sb16_ctl_line_capture_route,
+ &snd_sb16_ctl_line_play_switch,
+ &snd_sb16_ctl_line_play_vol,
+ &snd_sb16_ctl_mic_capture_route,
+ &snd_als4000_ctl_mic_20db_boost,
+ &snd_sb16_ctl_auto_mic_gain,
+ &snd_sb16_ctl_mic_play_switch,
+ &snd_sb16_ctl_mic_play_vol,
+ &snd_sb16_ctl_pc_speaker_vol,
+ &snd_sb16_ctl_capture_vol,
+ &snd_sb16_ctl_play_vol,
+ &snd_als4000_ctl_master_mono_playback_switch,
+ &snd_als4000_ctl_master_mono_capture_route,
+ &snd_als4000_ctl_mono_playback_switch,
+ &snd_als4000_ctl_mixer_loopback,
+ &snd_als4000_3d_control_switch,
+ &snd_als4000_3d_control_ratio,
+ &snd_als4000_3d_control_freq,
+ &snd_als4000_3d_control_delay,
+ &snd_als4000_3d_control_poweroff_switch,
+#ifdef NOT_AVAILABLE
+ &snd_als4000_ctl_fmdac,
+ &snd_als4000_ctl_qsound,
+#endif
+};
+
+static unsigned char snd_als4000_init_values[][2] = {
+ { SB_DSP4_MASTER_DEV + 0, 0 },
+ { SB_DSP4_MASTER_DEV + 1, 0 },
+ { SB_DSP4_PCM_DEV + 0, 0 },
+ { SB_DSP4_PCM_DEV + 1, 0 },
+ { SB_DSP4_SYNTH_DEV + 0, 0 },
+ { SB_DSP4_SYNTH_DEV + 1, 0 },
+ { SB_DSP4_SPEAKER_DEV, 0 },
+ { SB_DSP4_OUTPUT_SW, 0 },
+ { SB_DSP4_INPUT_LEFT, 0 },
+ { SB_DSP4_INPUT_RIGHT, 0 },
+ { SB_DT019X_OUTPUT_SW2, 0 },
+ { SB_ALS4000_MIC_IN_GAIN, 0 },
+};
+
+
+/*
+ */
+static int snd_sbmixer_init(struct snd_sb *chip,
+ struct sbmix_elem **controls,
+ int controls_count,
+ unsigned char map[][2],
+ int map_count,
+ char *name)
+{
+ unsigned long flags;
+ struct snd_card *card = chip->card;
+ int idx, err;
+
+ /* mixer reset */
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ snd_sbmixer_write(chip, 0x00, 0x00);
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+
+ /* mute and zero volume channels */
+ for (idx = 0; idx < map_count; idx++) {
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ snd_sbmixer_write(chip, map[idx][0], map[idx][1]);
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+ }
+
+ for (idx = 0; idx < controls_count; idx++) {
+ if ((err = snd_sbmixer_add_ctl_elem(chip, controls[idx])) < 0)
+ return err;
+ }
+ snd_component_add(card, name);
+ strcpy(card->mixername, name);
+ return 0;
+}
+
+int snd_sbmixer_new(struct snd_sb *chip)
+{
+ struct snd_card *card;
+ int err;
+
+ if (snd_BUG_ON(!chip || !chip->card))
+ return -EINVAL;
+
+ card = chip->card;
+
+ switch (chip->hardware) {
+ case SB_HW_10:
+ return 0; /* no mixer chip on SB1.x */
+ case SB_HW_20:
+ case SB_HW_201:
+ if ((err = snd_sbmixer_init(chip,
+ snd_sb20_controls,
+ ARRAY_SIZE(snd_sb20_controls),
+ snd_sb20_init_values,
+ ARRAY_SIZE(snd_sb20_init_values),
+ "CTL1335")) < 0)
+ return err;
+ break;
+ case SB_HW_PRO:
+ if ((err = snd_sbmixer_init(chip,
+ snd_sbpro_controls,
+ ARRAY_SIZE(snd_sbpro_controls),
+ snd_sbpro_init_values,
+ ARRAY_SIZE(snd_sbpro_init_values),
+ "CTL1345")) < 0)
+ return err;
+ break;
+ case SB_HW_16:
+ case SB_HW_ALS100:
+ case SB_HW_CS5530:
+ if ((err = snd_sbmixer_init(chip,
+ snd_sb16_controls,
+ ARRAY_SIZE(snd_sb16_controls),
+ snd_sb16_init_values,
+ ARRAY_SIZE(snd_sb16_init_values),
+ "CTL1745")) < 0)
+ return err;
+ break;
+ case SB_HW_ALS4000:
+ if ((err = snd_sbmixer_init(chip,
+ snd_als4000_controls,
+ ARRAY_SIZE(snd_als4000_controls),
+ snd_als4000_init_values,
+ ARRAY_SIZE(snd_als4000_init_values),
+ "ALS4000")) < 0)
+ return err;
+ break;
+ case SB_HW_DT019X:
+ if ((err = snd_sbmixer_init(chip,
+ snd_dt019x_controls,
+ ARRAY_SIZE(snd_dt019x_controls),
+ snd_dt019x_init_values,
+ ARRAY_SIZE(snd_dt019x_init_values),
+ "DT019X")) < 0)
+ break;
+ default:
+ strcpy(card->mixername, "???");
+ }
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static unsigned char sb20_saved_regs[] = {
+ SB_DSP20_MASTER_DEV,
+ SB_DSP20_PCM_DEV,
+ SB_DSP20_FM_DEV,
+ SB_DSP20_CD_DEV,
+};
+
+static unsigned char sbpro_saved_regs[] = {
+ SB_DSP_MASTER_DEV,
+ SB_DSP_PCM_DEV,
+ SB_DSP_PLAYBACK_FILT,
+ SB_DSP_FM_DEV,
+ SB_DSP_CD_DEV,
+ SB_DSP_LINE_DEV,
+ SB_DSP_MIC_DEV,
+ SB_DSP_CAPTURE_SOURCE,
+ SB_DSP_CAPTURE_FILT,
+};
+
+static unsigned char sb16_saved_regs[] = {
+ SB_DSP4_MASTER_DEV, SB_DSP4_MASTER_DEV + 1,
+ SB_DSP4_3DSE,
+ SB_DSP4_BASS_DEV, SB_DSP4_BASS_DEV + 1,
+ SB_DSP4_TREBLE_DEV, SB_DSP4_TREBLE_DEV + 1,
+ SB_DSP4_PCM_DEV, SB_DSP4_PCM_DEV + 1,
+ SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT,
+ SB_DSP4_SYNTH_DEV, SB_DSP4_SYNTH_DEV + 1,
+ SB_DSP4_OUTPUT_SW,
+ SB_DSP4_CD_DEV, SB_DSP4_CD_DEV + 1,
+ SB_DSP4_LINE_DEV, SB_DSP4_LINE_DEV + 1,
+ SB_DSP4_MIC_DEV,
+ SB_DSP4_SPEAKER_DEV,
+ SB_DSP4_IGAIN_DEV, SB_DSP4_IGAIN_DEV + 1,
+ SB_DSP4_OGAIN_DEV, SB_DSP4_OGAIN_DEV + 1,
+ SB_DSP4_MIC_AGC
+};
+
+static unsigned char dt019x_saved_regs[] = {
+ SB_DT019X_MASTER_DEV,
+ SB_DT019X_PCM_DEV,
+ SB_DT019X_SYNTH_DEV,
+ SB_DT019X_CD_DEV,
+ SB_DT019X_MIC_DEV,
+ SB_DT019X_SPKR_DEV,
+ SB_DT019X_LINE_DEV,
+ SB_DSP4_OUTPUT_SW,
+ SB_DT019X_OUTPUT_SW2,
+ SB_DT019X_CAPTURE_SW,
+};
+
+static unsigned char als4000_saved_regs[] = {
+ SB_DSP4_MASTER_DEV, SB_DSP4_MASTER_DEV + 1,
+ SB_DSP4_OUTPUT_SW,
+ SB_DSP4_PCM_DEV, SB_DSP4_PCM_DEV + 1,
+ SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT,
+ SB_DSP4_SYNTH_DEV, SB_DSP4_SYNTH_DEV + 1,
+ SB_DSP4_CD_DEV, SB_DSP4_CD_DEV + 1,
+ SB_DSP4_MIC_AGC,
+ SB_DSP4_MIC_DEV,
+ SB_DSP4_SPEAKER_DEV,
+ SB_DSP4_IGAIN_DEV, SB_DSP4_IGAIN_DEV + 1,
+ SB_DSP4_OGAIN_DEV, SB_DSP4_OGAIN_DEV + 1,
+ SB_DT019X_OUTPUT_SW2,
+ SB_ALS4000_MONO_IO_CTRL,
+ SB_ALS4000_MIC_IN_GAIN,
+ SB_ALS4000_3D_SND_FX,
+ SB_ALS4000_3D_TIME_DELAY,
+};
+
+static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
+{
+ unsigned char *val = chip->saved_regs;
+ if (snd_BUG_ON(num_regs > ARRAY_SIZE(chip->saved_regs)))
+ return;
+ for (; num_regs; num_regs--)
+ *val++ = snd_sbmixer_read(chip, *regs++);
+}
+
+static void restore_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
+{
+ unsigned char *val = chip->saved_regs;
+ if (snd_BUG_ON(num_regs > ARRAY_SIZE(chip->saved_regs)))
+ return;
+ for (; num_regs; num_regs--)
+ snd_sbmixer_write(chip, *regs++, *val++);
+}
+
+void snd_sbmixer_suspend(struct snd_sb *chip)
+{
+ switch (chip->hardware) {
+ case SB_HW_20:
+ case SB_HW_201:
+ save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs));
+ break;
+ case SB_HW_PRO:
+ save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs));
+ break;
+ case SB_HW_16:
+ case SB_HW_ALS100:
+ case SB_HW_CS5530:
+ save_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs));
+ break;
+ case SB_HW_ALS4000:
+ save_mixer(chip, als4000_saved_regs, ARRAY_SIZE(als4000_saved_regs));
+ break;
+ case SB_HW_DT019X:
+ save_mixer(chip, dt019x_saved_regs, ARRAY_SIZE(dt019x_saved_regs));
+ break;
+ default:
+ break;
+ }
+}
+
+void snd_sbmixer_resume(struct snd_sb *chip)
+{
+ switch (chip->hardware) {
+ case SB_HW_20:
+ case SB_HW_201:
+ restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs));
+ break;
+ case SB_HW_PRO:
+ restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs));
+ break;
+ case SB_HW_16:
+ case SB_HW_ALS100:
+ case SB_HW_CS5530:
+ restore_mixer(chip, sb16_saved_regs, ARRAY_SIZE(sb16_saved_regs));
+ break;
+ case SB_HW_ALS4000:
+ restore_mixer(chip, als4000_saved_regs, ARRAY_SIZE(als4000_saved_regs));
+ break;
+ case SB_HW_DT019X:
+ restore_mixer(chip, dt019x_saved_regs, ARRAY_SIZE(dt019x_saved_regs));
+ break;
+ default:
+ break;
+ }
+}
+#endif
diff --git a/sound/isa/sb/sbawe.c b/sound/isa/sb/sbawe.c
new file mode 100644
index 0000000..2ec52a3
--- /dev/null
+++ b/sound/isa/sb/sbawe.c
@@ -0,0 +1,2 @@
+#define SNDRV_SBAWE
+#include "sb16.c"
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
new file mode 100644
index 0000000..ca35924
--- /dev/null
+++ b/sound/isa/sc6000.c
@@ -0,0 +1,655 @@
+/*
+ * Driver for Gallant SC-6000 soundcard. This card is also known as
+ * Audio Excel DSP 16 or Zoltrix AV302.
+ * These cards use CompuMedia ASC-9308 chip + AD1848 codec.
+ *
+ * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1@wp.pl>
+ *
+ * I don't have documentation for this card. I used the driver
+ * for OSS/Free included in the kernel source as reference.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/isa.h>
+#include <linux/io.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/opl3.h>
+#include <sound/mpu401.h>
+#include <sound/control.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Krzysztof Helt");
+MODULE_DESCRIPTION("Gallant SC-6000");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000},"
+ "{AudioExcel, Audio Excel DSP 16},"
+ "{Zoltrix, AV302}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */
+static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */
+static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+ /* 0x300, 0x310, 0x320, 0x330 */
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */
+static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard.");
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for sc-6000 driver.");
+module_param_array(mss_port, long, NULL, 0444);
+MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver.");
+module_param_array(mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver.");
+module_param_array(dma, int, NULL, 0444);
+MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver.");
+
+/*
+ * Commands of SC6000's DSP (SBPRO+special).
+ * Some of them are COMMAND_xx, in the future they may change.
+ */
+#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */
+#define COMMAND_52 0x52 /* */
+#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */
+#define COMMAND_5C 0x5c /* */
+#define COMMAND_60 0x60 /* */
+#define COMMAND_66 0x66 /* */
+#define COMMAND_6C 0x6c /* */
+#define COMMAND_6E 0x6e /* */
+#define COMMAND_88 0x88 /* Unknown command */
+#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */
+#define COMMAND_C5 0xc5 /* */
+#define GET_DSP_VERSION 0xe1 /* Get DSP Version */
+#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */
+
+/*
+ * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port
+ * to have the actual I/O port.
+ * Register permissions are:
+ * (wo) == Write Only
+ * (ro) == Read Only
+ * (w-) == Write
+ * (r-) == Read
+ */
+#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */
+#define DSP_READ 0x0a /* offset of DSP READ (ro) */
+#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */
+#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */
+#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */
+#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */
+
+#define PFX "sc6000: "
+#define DRV_NAME "SC-6000"
+
+/* hardware dependent functions */
+
+/*
+ * sc6000_irq_to_softcfg - Decode irq number into cfg code.
+ */
+static __devinit unsigned char sc6000_irq_to_softcfg(int irq)
+{
+ unsigned char val = 0;
+
+ switch (irq) {
+ case 5:
+ val = 0x28;
+ break;
+ case 7:
+ val = 0x8;
+ break;
+ case 9:
+ val = 0x10;
+ break;
+ case 10:
+ val = 0x18;
+ break;
+ case 11:
+ val = 0x20;
+ break;
+ default:
+ break;
+ }
+ return val;
+}
+
+/*
+ * sc6000_dma_to_softcfg - Decode dma number into cfg code.
+ */
+static __devinit unsigned char sc6000_dma_to_softcfg(int dma)
+{
+ unsigned char val = 0;
+
+ switch (dma) {
+ case 0:
+ val = 1;
+ break;
+ case 1:
+ val = 2;
+ break;
+ case 3:
+ val = 3;
+ break;
+ default:
+ break;
+ }
+ return val;
+}
+
+/*
+ * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code.
+ */
+static __devinit unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq)
+{
+ unsigned char val = 0;
+
+ switch (mpu_irq) {
+ case 5:
+ val = 4;
+ break;
+ case 7:
+ val = 0x44;
+ break;
+ case 9:
+ val = 0x84;
+ break;
+ case 10:
+ val = 0xc4;
+ break;
+ default:
+ break;
+ }
+ return val;
+}
+
+static __devinit int sc6000_wait_data(char __iomem *vport)
+{
+ int loop = 1000;
+ unsigned char val = 0;
+
+ do {
+ val = ioread8(vport + DSP_DATAVAIL);
+ if (val & 0x80)
+ return 0;
+ cpu_relax();
+ } while (loop--);
+
+ return -EAGAIN;
+}
+
+static __devinit int sc6000_read(char __iomem *vport)
+{
+ if (sc6000_wait_data(vport))
+ return -EBUSY;
+
+ return ioread8(vport + DSP_READ);
+
+}
+
+static __devinit int sc6000_write(char __iomem *vport, int cmd)
+{
+ unsigned char val;
+ int loop = 500000;
+
+ do {
+ val = ioread8(vport + DSP_STATUS);
+ /*
+ * DSP ready to receive data if bit 7 of val == 0
+ */
+ if (!(val & 0x80)) {
+ iowrite8(cmd, vport + DSP_COMMAND);
+ return 0;
+ }
+ cpu_relax();
+ } while (loop--);
+
+ snd_printk(KERN_ERR "DSP Command (0x%x) timeout.\n", cmd);
+
+ return -EIO;
+}
+
+static int __devinit sc6000_dsp_get_answer(char __iomem *vport, int command,
+ char *data, int data_len)
+{
+ int len = 0;
+
+ if (sc6000_write(vport, command)) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n", command);
+ return -EIO;
+ }
+
+ do {
+ int val = sc6000_read(vport);
+
+ if (val < 0)
+ break;
+
+ data[len++] = val;
+
+ } while (len < data_len);
+
+ /*
+ * If no more data available, return to the caller, no error if len>0.
+ * We have no other way to know when the string is finished.
+ */
+ return len ? len : -EIO;
+}
+
+static int __devinit sc6000_dsp_reset(char __iomem *vport)
+{
+ iowrite8(1, vport + DSP_RESET);
+ udelay(10);
+ iowrite8(0, vport + DSP_RESET);
+ udelay(20);
+ if (sc6000_read(vport) == 0xaa)
+ return 0;
+ return -ENODEV;
+}
+
+/* detection and initialization */
+static int __devinit sc6000_cfg_write(char __iomem *vport,
+ unsigned char softcfg)
+{
+
+ if (sc6000_write(vport, WRITE_MDIRQ_CFG)) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG);
+ return -EIO;
+ }
+ if (sc6000_write(vport, softcfg)) {
+ snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n");
+ return -EIO;
+ }
+ return 0;
+}
+
+static int __devinit sc6000_setup_board(char __iomem *vport, int config)
+{
+ int loop = 10;
+
+ do {
+ if (sc6000_write(vport, COMMAND_88)) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n",
+ COMMAND_88);
+ return -EIO;
+ }
+ } while ((sc6000_wait_data(vport) < 0) && loop--);
+
+ if (sc6000_read(vport) < 0) {
+ snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n",
+ COMMAND_88);
+ return -EIO;
+ }
+
+ if (sc6000_cfg_write(vport, config))
+ return -ENODEV;
+
+ return 0;
+}
+
+static int __devinit sc6000_init_mss(char __iomem *vport, int config,
+ char __iomem *vmss_port, int mss_config)
+{
+ if (sc6000_write(vport, DSP_INIT_MSS)) {
+ snd_printk(KERN_ERR "sc6000_init_mss [0x%x]: failed!\n",
+ DSP_INIT_MSS);
+ return -EIO;
+ }
+
+ msleep(10);
+
+ if (sc6000_cfg_write(vport, config))
+ return -EIO;
+
+ iowrite8(mss_config, vmss_port);
+
+ return 0;
+}
+
+static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma,
+ char __iomem *vmss_port, int mpu_irq)
+{
+ char answer[15];
+ char version[2];
+ int mss_config = sc6000_irq_to_softcfg(irq) |
+ sc6000_dma_to_softcfg(dma);
+ int config = mss_config |
+ sc6000_mpu_irq_to_softcfg(mpu_irq);
+ int err;
+
+ err = sc6000_dsp_reset(vport);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sc6000_dsp_reset: failed!\n");
+ return err;
+ }
+
+ memset(answer, 0, sizeof(answer));
+ err = sc6000_dsp_get_answer(vport, GET_DSP_COPYRIGHT, answer, 15);
+ if (err <= 0) {
+ snd_printk(KERN_ERR "sc6000_dsp_copyright: failed!\n");
+ return -ENODEV;
+ }
+ /*
+ * My SC-6000 card return "SC-6000" in DSPCopyright, so
+ * if we have something different, we have to be warned.
+ * Mine returns "SC-6000A " - KH
+ */
+ if (strncmp("SC-6000", answer, 7))
+ snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n");
+
+ if (sc6000_dsp_get_answer(vport, GET_DSP_VERSION, version, 2) < 2) {
+ snd_printk(KERN_ERR "sc6000_dsp_version: failed!\n");
+ return -ENODEV;
+ }
+ printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n",
+ answer, version[0], version[1]);
+
+ /*
+ * 0x0A == (IRQ 7, DMA 1, MIRQ 0)
+ */
+ err = sc6000_cfg_write(vport, 0x0a);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n");
+ return -EFAULT;
+ }
+
+ err = sc6000_setup_board(vport, config);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sc6000_setup_board: failed!\n");
+ return -ENODEV;
+ }
+
+ err = sc6000_init_mss(vport, config, vmss_port, mss_config);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Can not initialize "
+ "Microsoft Sound System mode.\n");
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static int __devinit snd_sc6000_mixer(struct snd_wss *chip)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ctl_elem_id id1, id2;
+ int err;
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ /* reassign AUX0 to FM */
+ strcpy(id1.name, "Aux Playback Switch");
+ strcpy(id2.name, "FM Playback Switch");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "FM Playback Volume");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ /* reassign AUX1 to CD */
+ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1;
+ strcpy(id2.name, "CD Playback Switch");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "CD Playback Volume");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev)
+{
+ if (!enable[dev])
+ return 0;
+ if (port[dev] == SNDRV_AUTO_PORT) {
+ printk(KERN_ERR PFX "specify IO port\n");
+ return 0;
+ }
+ if (mss_port[dev] == SNDRV_AUTO_PORT) {
+ printk(KERN_ERR PFX "specify MSS port\n");
+ return 0;
+ }
+ if (port[dev] != 0x220 && port[dev] != 0x240) {
+ printk(KERN_ERR PFX "Port must be 0x220 or 0x240\n");
+ return 0;
+ }
+ if (mss_port[dev] != 0x530 && mss_port[dev] != 0xe80) {
+ printk(KERN_ERR PFX "MSS port must be 0x530 or 0xe80\n");
+ return 0;
+ }
+ if (irq[dev] != SNDRV_AUTO_IRQ && !sc6000_irq_to_softcfg(irq[dev])) {
+ printk(KERN_ERR PFX "invalid IRQ %d\n", irq[dev]);
+ return 0;
+ }
+ if (dma[dev] != SNDRV_AUTO_DMA && !sc6000_dma_to_softcfg(dma[dev])) {
+ printk(KERN_ERR PFX "invalid DMA %d\n", dma[dev]);
+ return 0;
+ }
+ if (mpu_port[dev] != SNDRV_AUTO_PORT &&
+ (mpu_port[dev] & ~0x30L) != 0x300) {
+ printk(KERN_ERR PFX "invalid MPU-401 port %lx\n",
+ mpu_port[dev]);
+ return 0;
+ }
+ if (mpu_port[dev] != SNDRV_AUTO_PORT &&
+ mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] != 0 &&
+ !sc6000_mpu_irq_to_softcfg(mpu_irq[dev])) {
+ printk(KERN_ERR PFX "invalid MPU-401 IRQ %d\n", mpu_irq[dev]);
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev)
+{
+ static int possible_irqs[] = { 5, 7, 9, 10, 11, -1 };
+ static int possible_dmas[] = { 1, 3, 0, -1 };
+ int err;
+ int xirq = irq[dev];
+ int xdma = dma[dev];
+ struct snd_card *card;
+ struct snd_wss *chip;
+ struct snd_opl3 *opl3;
+ char __iomem *vport;
+ char __iomem *vmss_port;
+
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (!card)
+ return -ENOMEM;
+
+ if (xirq == SNDRV_AUTO_IRQ) {
+ xirq = snd_legacy_find_free_irq(possible_irqs);
+ if (xirq < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free IRQ\n");
+ err = -EBUSY;
+ goto err_exit;
+ }
+ }
+
+ if (xdma == SNDRV_AUTO_DMA) {
+ xdma = snd_legacy_find_free_dma(possible_dmas);
+ if (xdma < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA\n");
+ err = -EBUSY;
+ goto err_exit;
+ }
+ }
+
+ if (!request_region(port[dev], 0x10, DRV_NAME)) {
+ snd_printk(KERN_ERR PFX
+ "I/O port region is already in use.\n");
+ err = -EBUSY;
+ goto err_exit;
+ }
+ vport = devm_ioport_map(devptr, port[dev], 0x10);
+ if (!vport) {
+ snd_printk(KERN_ERR PFX
+ "I/O port cannot be iomaped.\n");
+ err = -EBUSY;
+ goto err_unmap1;
+ }
+
+ /* to make it marked as used */
+ if (!request_region(mss_port[dev], 4, DRV_NAME)) {
+ snd_printk(KERN_ERR PFX
+ "SC-6000 port I/O port region is already in use.\n");
+ err = -EBUSY;
+ goto err_unmap1;
+ }
+ vmss_port = devm_ioport_map(devptr, mss_port[dev], 4);
+ if (!vport) {
+ snd_printk(KERN_ERR PFX
+ "MSS port I/O cannot be iomaped.\n");
+ err = -EBUSY;
+ goto err_unmap2;
+ }
+
+ snd_printd("Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n",
+ port[dev], xirq, xdma,
+ mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]);
+
+ err = sc6000_init_board(vport, xirq, xdma, vmss_port, mpu_irq[dev]);
+ if (err < 0)
+ goto err_unmap2;
+
+ err = snd_wss_create(card, mss_port[dev] + 4, -1, xirq, xdma, -1,
+ WSS_HW_DETECT, 0, &chip);
+ if (err < 0)
+ goto err_unmap2;
+ card->private_data = chip;
+
+ err = snd_wss_pcm(chip, 0, NULL);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX
+ "error creating new WSS PCM device\n");
+ goto err_unmap2;
+ }
+ err = snd_wss_mixer(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "error creating new WSS mixer\n");
+ goto err_unmap2;
+ }
+ err = snd_sc6000_mixer(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "the mixer rewrite failed\n");
+ goto err_unmap2;
+ }
+ if (snd_opl3_create(card,
+ 0x388, 0x388 + 2,
+ OPL3_HW_AUTO, 0, &opl3) < 0) {
+ snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n",
+ 0x388, 0x388 + 2);
+ } else {
+ err = snd_opl3_timer_new(opl3, 0, 1);
+ if (err < 0)
+ goto err_unmap2;
+
+ err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+ if (err < 0)
+ goto err_unmap2;
+ }
+
+ if (mpu_port[dev] != SNDRV_AUTO_PORT) {
+ if (mpu_irq[dev] == SNDRV_AUTO_IRQ)
+ mpu_irq[dev] = -1;
+ if (snd_mpu401_uart_new(card, 0,
+ MPU401_HW_MPU401,
+ mpu_port[dev], 0,
+ mpu_irq[dev], IRQF_DISABLED,
+ NULL) < 0)
+ snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n",
+ mpu_port[dev]);
+ }
+
+ strcpy(card->driver, DRV_NAME);
+ strcpy(card->shortname, "SC-6000");
+ sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d",
+ mss_port[dev], xirq, xdma);
+
+ snd_card_set_dev(card, devptr);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto err_unmap2;
+
+ dev_set_drvdata(devptr, card);
+ return 0;
+
+err_unmap2:
+ release_region(mss_port[dev], 4);
+err_unmap1:
+ release_region(port[dev], 0x10);
+err_exit:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev)
+{
+ release_region(port[dev], 0x10);
+ release_region(mss_port[dev], 4);
+
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+static struct isa_driver snd_sc6000_driver = {
+ .match = snd_sc6000_match,
+ .probe = snd_sc6000_probe,
+ .remove = __devexit_p(snd_sc6000_remove),
+ /* FIXME: suspend/resume */
+ .driver = {
+ .name = DRV_NAME,
+ },
+};
+
+
+static int __init alsa_card_sc6000_init(void)
+{
+ return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_sc6000_exit(void)
+{
+ isa_unregister_driver(&snd_sc6000_driver);
+}
+
+module_init(alsa_card_sc6000_init)
+module_exit(alsa_card_sc6000_exit)
diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c
new file mode 100644
index 0000000..2c7503b
--- /dev/null
+++ b/sound/isa/sgalaxy.c
@@ -0,0 +1,369 @@
+/*
+ * Driver for Aztech Sound Galaxy cards
+ * Copyright (c) by Christopher Butler <chrisb@sandy.force9.co.uk.
+ *
+ * I don't have documentation for this card, I based this driver on the
+ * driver for OSS/Free included in the kernel source (drivers/sound/sgalaxy.c)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+#include <sound/wss.h>
+#include <sound/control.h>
+#define SNDRV_LEGACY_FIND_FREE_IRQ
+#define SNDRV_LEGACY_FIND_FREE_DMA
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Christopher Butler <chrisb@sandy.force9.co.uk>");
+MODULE_DESCRIPTION("Aztech Sound Galaxy");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Aztech Systems,Sound Galaxy}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+static long sbport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240 */
+static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530,0xe80,0xf40,0x604 */
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 7,9,10,11 */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Sound Galaxy soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Sound Galaxy soundcard.");
+module_param_array(sbport, long, NULL, 0444);
+MODULE_PARM_DESC(sbport, "Port # for Sound Galaxy SB driver.");
+module_param_array(wssport, long, NULL, 0444);
+MODULE_PARM_DESC(wssport, "Port # for Sound Galaxy WSS driver.");
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for Sound Galaxy driver.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for Sound Galaxy driver.");
+
+#define SGALAXY_AUXC_LEFT 18
+#define SGALAXY_AUXC_RIGHT 19
+
+#define PFX "sgalaxy: "
+
+/*
+
+ */
+
+#define AD1848P1( port, x ) ( port + c_d_c_AD1848##x )
+
+/* from lowlevel/sb/sb.c - to avoid having to allocate a struct snd_sb for the */
+/* short time we actually need it.. */
+
+static int snd_sgalaxy_sbdsp_reset(unsigned long port)
+{
+ int i;
+
+ outb(1, SBP1(port, RESET));
+ udelay(10);
+ outb(0, SBP1(port, RESET));
+ udelay(30);
+ for (i = 0; i < 1000 && !(inb(SBP1(port, DATA_AVAIL)) & 0x80); i++);
+ if (inb(SBP1(port, READ)) != 0xaa) {
+ snd_printd("sb_reset: failed at 0x%lx!!!\n", port);
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static int __devinit snd_sgalaxy_sbdsp_command(unsigned long port,
+ unsigned char val)
+{
+ int i;
+
+ for (i = 10000; i; i--)
+ if ((inb(SBP1(port, STATUS)) & 0x80) == 0) {
+ outb(val, SBP1(port, COMMAND));
+ return 1;
+ }
+
+ return 0;
+}
+
+static irqreturn_t snd_sgalaxy_dummy_interrupt(int irq, void *dev_id)
+{
+ return IRQ_NONE;
+}
+
+static int __devinit snd_sgalaxy_setup_wss(unsigned long port, int irq, int dma)
+{
+ static int interrupt_bits[] = {-1, -1, -1, -1, -1, -1, -1, 0x08, -1,
+ 0x10, 0x18, 0x20, -1, -1, -1, -1};
+ static int dma_bits[] = {1, 2, 0, 3};
+ int tmp, tmp1;
+
+ if ((tmp = inb(port + 3)) == 0xff)
+ {
+ snd_printdd("I/O address dead (0x%lx)\n", port);
+ return 0;
+ }
+#if 0
+ snd_printdd("WSS signature = 0x%x\n", tmp);
+#endif
+
+ if ((tmp & 0x3f) != 0x04 &&
+ (tmp & 0x3f) != 0x0f &&
+ (tmp & 0x3f) != 0x00) {
+ snd_printdd("No WSS signature detected on port 0x%lx\n",
+ port + 3);
+ return 0;
+ }
+
+#if 0
+ snd_printdd(PFX "setting up IRQ/DMA for WSS\n");
+#endif
+
+ /* initialize IRQ for WSS codec */
+ tmp = interrupt_bits[irq % 16];
+ if (tmp < 0)
+ return -EINVAL;
+
+ if (request_irq(irq, snd_sgalaxy_dummy_interrupt, IRQF_DISABLED, "sgalaxy", NULL)) {
+ snd_printk(KERN_ERR "sgalaxy: can't grab irq %d\n", irq);
+ return -EIO;
+ }
+
+ outb(tmp | 0x40, port);
+ tmp1 = dma_bits[dma % 4];
+ outb(tmp | tmp1, port);
+
+ free_irq(irq, NULL);
+
+ return 0;
+}
+
+static int __devinit snd_sgalaxy_detect(int dev, int irq, int dma)
+{
+#if 0
+ snd_printdd(PFX "switching to WSS mode\n");
+#endif
+
+ /* switch to WSS mode */
+ snd_sgalaxy_sbdsp_reset(sbport[dev]);
+
+ snd_sgalaxy_sbdsp_command(sbport[dev], 9);
+ snd_sgalaxy_sbdsp_command(sbport[dev], 0);
+
+ udelay(400);
+ return snd_sgalaxy_setup_wss(wssport[dev], irq, dma);
+}
+
+static struct snd_kcontrol_new snd_sgalaxy_controls[] = {
+WSS_DOUBLE("Aux Playback Switch", 0,
+ SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1),
+WSS_DOUBLE("Aux Playback Volume", 0,
+ SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0)
+};
+
+static int __devinit snd_sgalaxy_mixer(struct snd_wss *chip)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ctl_elem_id id1, id2;
+ unsigned int idx;
+ int err;
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ /* reassign AUX0 to LINE */
+ strcpy(id1.name, "Aux Playback Switch");
+ strcpy(id2.name, "Line Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "Line Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ /* reassign AUX1 to FM */
+ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1;
+ strcpy(id2.name, "FM Playback Switch");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ strcpy(id1.name, "Aux Playback Volume");
+ strcpy(id2.name, "FM Playback Volume");
+ if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0)
+ return err;
+ /* build AUX2 input */
+ for (idx = 0; idx < ARRAY_SIZE(snd_sgalaxy_controls); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_sgalaxy_controls[idx], chip));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int __devinit snd_sgalaxy_match(struct device *devptr, unsigned int dev)
+{
+ if (!enable[dev])
+ return 0;
+ if (sbport[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify SB port\n");
+ return 0;
+ }
+ if (wssport[dev] == SNDRV_AUTO_PORT) {
+ snd_printk(KERN_ERR PFX "specify WSS port\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_sgalaxy_probe(struct device *devptr, unsigned int dev)
+{
+ static int possible_irqs[] = {7, 9, 10, 11, -1};
+ static int possible_dmas[] = {1, 3, 0, -1};
+ int err, xirq, xdma1;
+ struct snd_card *card;
+ struct snd_wss *chip;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ xirq = irq[dev];
+ if (xirq == SNDRV_AUTO_IRQ) {
+ if ((xirq = snd_legacy_find_free_irq(possible_irqs)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free IRQ\n");
+ err = -EBUSY;
+ goto _err;
+ }
+ }
+ xdma1 = dma1[dev];
+ if (xdma1 == SNDRV_AUTO_DMA) {
+ if ((xdma1 = snd_legacy_find_free_dma(possible_dmas)) < 0) {
+ snd_printk(KERN_ERR PFX "unable to find a free DMA\n");
+ err = -EBUSY;
+ goto _err;
+ }
+ }
+
+ if ((err = snd_sgalaxy_detect(dev, xirq, xdma1)) < 0)
+ goto _err;
+
+ err = snd_wss_create(card, wssport[dev] + 4, -1,
+ xirq, xdma1, -1,
+ WSS_HW_DETECT, 0, &chip);
+ if (err < 0)
+ goto _err;
+ card->private_data = chip;
+
+ err = snd_wss_pcm(chip, 0, NULL);
+ if (err < 0) {
+ snd_printdd(PFX "error creating new WSS PCM device\n");
+ goto _err;
+ }
+ err = snd_wss_mixer(chip);
+ if (err < 0) {
+ snd_printdd(PFX "error creating new WSS mixer\n");
+ goto _err;
+ }
+ if ((err = snd_sgalaxy_mixer(chip)) < 0) {
+ snd_printdd(PFX "the mixer rewrite failed\n");
+ goto _err;
+ }
+
+ strcpy(card->driver, "Sound Galaxy");
+ strcpy(card->shortname, "Sound Galaxy");
+ sprintf(card->longname, "Sound Galaxy at 0x%lx, irq %d, dma %d",
+ wssport[dev], xirq, xdma1);
+
+ snd_card_set_dev(card, devptr);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto _err;
+
+ dev_set_drvdata(devptr, card);
+ return 0;
+
+ _err:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit snd_sgalaxy_remove(struct device *devptr, unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_sgalaxy_suspend(struct device *pdev, unsigned int n,
+ pm_message_t state)
+{
+ struct snd_card *card = dev_get_drvdata(pdev);
+ struct snd_wss *chip = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ chip->suspend(chip);
+ return 0;
+}
+
+static int snd_sgalaxy_resume(struct device *pdev, unsigned int n)
+{
+ struct snd_card *card = dev_get_drvdata(pdev);
+ struct snd_wss *chip = card->private_data;
+
+ chip->resume(chip);
+ snd_wss_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]);
+ snd_wss_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+#define DEV_NAME "sgalaxy"
+
+static struct isa_driver snd_sgalaxy_driver = {
+ .match = snd_sgalaxy_match,
+ .probe = snd_sgalaxy_probe,
+ .remove = __devexit_p(snd_sgalaxy_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_sgalaxy_suspend,
+ .resume = snd_sgalaxy_resume,
+#endif
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+static int __init alsa_card_sgalaxy_init(void)
+{
+ return isa_register_driver(&snd_sgalaxy_driver, SNDRV_CARDS);
+}
+
+static void __exit alsa_card_sgalaxy_exit(void)
+{
+ isa_unregister_driver(&snd_sgalaxy_driver);
+}
+
+module_init(alsa_card_sgalaxy_init)
+module_exit(alsa_card_sgalaxy_exit)
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
new file mode 100644
index 0000000..48a16d8
--- /dev/null
+++ b/sound/isa/sscape.c
@@ -0,0 +1,1560 @@
+/*
+ * Low-level ALSA driver for the ENSONIQ SoundScape PnP
+ * Copyright (c) by Chris Rankin
+ *
+ * This driver was written in part using information obtained from
+ * the OSS/Free SoundScape driver, written by Hannu Savolainen.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/delay.h>
+#include <linux/pnp.h>
+#include <linux/spinlock.h>
+#include <linux/moduleparam.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/hwdep.h>
+#include <sound/wss.h>
+#include <sound/mpu401.h>
+#include <sound/initval.h>
+
+#include <sound/sscape_ioctl.h>
+
+
+MODULE_AUTHOR("Chris Rankin");
+MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver");
+MODULE_LICENSE("GPL");
+
+static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX;
+static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR;
+static long port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT;
+static long wss_port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT;
+static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ;
+static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ;
+static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA;
+static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index number for SoundScape soundcard");
+
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "Description for SoundScape card");
+
+module_param_array(port, long, NULL, 0444);
+MODULE_PARM_DESC(port, "Port # for SoundScape driver.");
+
+module_param_array(wss_port, long, NULL, 0444);
+MODULE_PARM_DESC(wss_port, "WSS Port # for SoundScape driver.");
+
+module_param_array(irq, int, NULL, 0444);
+MODULE_PARM_DESC(irq, "IRQ # for SoundScape driver.");
+
+module_param_array(mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_irq, "MPU401 IRQ # for SoundScape driver.");
+
+module_param_array(dma, int, NULL, 0444);
+MODULE_PARM_DESC(dma, "DMA # for SoundScape driver.");
+
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver.");
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnp_registered;
+
+static struct pnp_card_device_id sscape_pnpids[] = {
+ { .id = "ENS3081", .devs = { { "ENS0000" } } }, /* Soundscape PnP */
+ { .id = "ENS4081", .devs = { { "ENS1011" } } }, /* VIVO90 */
+ { .id = "" } /* end */
+};
+
+MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids);
+#endif
+
+
+#define MPU401_IO(i) ((i) + 0)
+#define MIDI_DATA_IO(i) ((i) + 0)
+#define MIDI_CTRL_IO(i) ((i) + 1)
+#define HOST_CTRL_IO(i) ((i) + 2)
+#define HOST_DATA_IO(i) ((i) + 3)
+#define ODIE_ADDR_IO(i) ((i) + 4)
+#define ODIE_DATA_IO(i) ((i) + 5)
+#define CODEC_IO(i) ((i) + 8)
+
+#define IC_ODIE 1
+#define IC_OPUS 2
+
+#define RX_READY 0x01
+#define TX_READY 0x02
+
+#define CMD_ACK 0x80
+#define CMD_SET_MIDI_VOL 0x84
+#define CMD_GET_MIDI_VOL 0x85
+#define CMD_XXX_MIDI_VOL 0x86
+#define CMD_SET_EXTMIDI 0x8a
+#define CMD_GET_EXTMIDI 0x8b
+#define CMD_SET_MT32 0x8c
+#define CMD_GET_MT32 0x8d
+
+enum GA_REG {
+ GA_INTSTAT_REG = 0,
+ GA_INTENA_REG,
+ GA_DMAA_REG,
+ GA_DMAB_REG,
+ GA_INTCFG_REG,
+ GA_DMACFG_REG,
+ GA_CDCFG_REG,
+ GA_SMCFGA_REG,
+ GA_SMCFGB_REG,
+ GA_HMCTL_REG
+};
+
+#define DMA_8BIT 0x80
+
+
+#define AD1845_FREQ_SEL_MSB 0x16
+#define AD1845_FREQ_SEL_LSB 0x17
+
+enum card_type {
+ SSCAPE,
+ SSCAPE_PNP,
+ SSCAPE_VIVO,
+};
+
+struct soundscape {
+ spinlock_t lock;
+ unsigned io_base;
+ unsigned wss_base;
+ int codec_type;
+ int ic_type;
+ enum card_type type;
+ struct resource *io_res;
+ struct resource *wss_res;
+ struct snd_wss *chip;
+ struct snd_mpu401 *mpu;
+ struct snd_hwdep *hw;
+
+ /*
+ * The MIDI device won't work until we've loaded
+ * its firmware via a hardware-dependent device IOCTL
+ */
+ spinlock_t fwlock;
+ int hw_in_use;
+ unsigned long midi_usage;
+ unsigned char midi_vol;
+};
+
+#define INVALID_IRQ ((unsigned)-1)
+
+
+static inline struct soundscape *get_card_soundscape(struct snd_card *c)
+{
+ return (struct soundscape *) (c->private_data);
+}
+
+static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu)
+{
+ return (struct soundscape *) (mpu->private_data);
+}
+
+static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw)
+{
+ return (struct soundscape *) (hw->private_data);
+}
+
+
+/*
+ * Allocates some kernel memory that we can use for DMA.
+ * I think this means that the memory has to map to
+ * contiguous pages of physical memory.
+ */
+static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size)
+{
+ if (buf) {
+ if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(),
+ size, buf) < 0) {
+ snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size);
+ return NULL;
+ }
+ }
+
+ return buf;
+}
+
+/*
+ * Release the DMA-able kernel memory ...
+ */
+static void free_dmabuf(struct snd_dma_buffer *buf)
+{
+ if (buf && buf->area)
+ snd_dma_free_pages(buf);
+}
+
+
+/*
+ * This function writes to the SoundScape's control registers,
+ * but doesn't do any locking. It's up to the caller to do that.
+ * This is why this function is "unsafe" ...
+ */
+static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val)
+{
+ outb(reg, ODIE_ADDR_IO(io_base));
+ outb(val, ODIE_DATA_IO(io_base));
+}
+
+/*
+ * Write to the SoundScape's control registers, and do the
+ * necessary locking ...
+ */
+static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&s->lock, flags);
+ sscape_write_unsafe(s->io_base, reg, val);
+ spin_unlock_irqrestore(&s->lock, flags);
+}
+
+/*
+ * Read from the SoundScape's control registers, but leave any
+ * locking to the caller. This is why the function is "unsafe" ...
+ */
+static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg)
+{
+ outb(reg, ODIE_ADDR_IO(io_base));
+ return inb(ODIE_DATA_IO(io_base));
+}
+
+/*
+ * Puts the SoundScape into "host" mode, as compared to "MIDI" mode
+ */
+static inline void set_host_mode_unsafe(unsigned io_base)
+{
+ outb(0x0, HOST_CTRL_IO(io_base));
+}
+
+/*
+ * Puts the SoundScape into "MIDI" mode, as compared to "host" mode
+ */
+static inline void set_midi_mode_unsafe(unsigned io_base)
+{
+ outb(0x3, HOST_CTRL_IO(io_base));
+}
+
+/*
+ * Read the SoundScape's host-mode control register, but leave
+ * any locking issues to the caller ...
+ */
+static inline int host_read_unsafe(unsigned io_base)
+{
+ int data = -1;
+ if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) {
+ data = inb(HOST_DATA_IO(io_base));
+ }
+
+ return data;
+}
+
+/*
+ * Read the SoundScape's host-mode control register, performing
+ * a limited amount of busy-waiting if the register isn't ready.
+ * Also leaves all locking-issues to the caller ...
+ */
+static int host_read_ctrl_unsafe(unsigned io_base, unsigned timeout)
+{
+ int data;
+
+ while (((data = host_read_unsafe(io_base)) < 0) && (timeout != 0)) {
+ udelay(100);
+ --timeout;
+ } /* while */
+
+ return data;
+}
+
+/*
+ * Write to the SoundScape's host-mode control registers, but
+ * leave any locking issues to the caller ...
+ */
+static inline int host_write_unsafe(unsigned io_base, unsigned char data)
+{
+ if ((inb(HOST_CTRL_IO(io_base)) & TX_READY) != 0) {
+ outb(data, HOST_DATA_IO(io_base));
+ return 1;
+ }
+
+ return 0;
+}
+
+/*
+ * Write to the SoundScape's host-mode control registers, performing
+ * a limited amount of busy-waiting if the register isn't ready.
+ * Also leaves all locking-issues to the caller ...
+ */
+static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data,
+ unsigned timeout)
+{
+ int err;
+
+ while (!(err = host_write_unsafe(io_base, data)) && (timeout != 0)) {
+ udelay(100);
+ --timeout;
+ } /* while */
+
+ return err;
+}
+
+
+/*
+ * Check that the MIDI subsystem is operational. If it isn't,
+ * then we will hang the computer if we try to use it ...
+ *
+ * NOTE: This check is based upon observation, not documentation.
+ */
+static inline int verify_mpu401(const struct snd_mpu401 * mpu)
+{
+ return ((inb(MIDI_CTRL_IO(mpu->port)) & 0xc0) == 0x80);
+}
+
+/*
+ * This is apparently the standard way to initailise an MPU-401
+ */
+static inline void initialise_mpu401(const struct snd_mpu401 * mpu)
+{
+ outb(0, MIDI_DATA_IO(mpu->port));
+}
+
+/*
+ * Tell the SoundScape to activate the AD1845 chip (I think).
+ * The AD1845 detection fails if we *don't* do this, so I
+ * think that this is a good idea ...
+ */
+static inline void activate_ad1845_unsafe(unsigned io_base)
+{
+ sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10);
+ sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80);
+}
+
+/*
+ * Do the necessary ALSA-level cleanup to deallocate our driver ...
+ */
+static void soundscape_free(struct snd_card *c)
+{
+ struct soundscape *sscape = get_card_soundscape(c);
+ release_and_free_resource(sscape->io_res);
+ release_and_free_resource(sscape->wss_res);
+ free_dma(sscape->chip->dma1);
+}
+
+/*
+ * Tell the SoundScape to begin a DMA tranfer using the given channel.
+ * All locking issues are left to the caller.
+ */
+static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
+{
+ sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01);
+ sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe);
+}
+
+/*
+ * Wait for a DMA transfer to complete. This is a "limited busy-wait",
+ * and all locking issues are left to the caller.
+ */
+static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout)
+{
+ while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) {
+ udelay(100);
+ --timeout;
+ } /* while */
+
+ return (sscape_read_unsafe(io_base, reg) & 0x01);
+}
+
+/*
+ * Wait for the On-Board Processor to return its start-up
+ * acknowledgement sequence. This wait is too long for
+ * us to perform "busy-waiting", and so we must sleep.
+ * This in turn means that we must not be holding any
+ * spinlocks when we call this function.
+ */
+static int obp_startup_ack(struct soundscape *s, unsigned timeout)
+{
+ while (timeout != 0) {
+ unsigned long flags;
+ unsigned char x;
+
+ schedule_timeout_uninterruptible(1);
+
+ spin_lock_irqsave(&s->lock, flags);
+ x = inb(HOST_DATA_IO(s->io_base));
+ spin_unlock_irqrestore(&s->lock, flags);
+ if ((x & 0xfe) == 0xfe)
+ return 1;
+
+ --timeout;
+ } /* while */
+
+ return 0;
+}
+
+/*
+ * Wait for the host to return its start-up acknowledgement
+ * sequence. This wait is too long for us to perform
+ * "busy-waiting", and so we must sleep. This in turn means
+ * that we must not be holding any spinlocks when we call
+ * this function.
+ */
+static int host_startup_ack(struct soundscape *s, unsigned timeout)
+{
+ while (timeout != 0) {
+ unsigned long flags;
+ unsigned char x;
+
+ schedule_timeout_uninterruptible(1);
+
+ spin_lock_irqsave(&s->lock, flags);
+ x = inb(HOST_DATA_IO(s->io_base));
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (x == 0xfe)
+ return 1;
+
+ --timeout;
+ } /* while */
+
+ return 0;
+}
+
+/*
+ * Upload a byte-stream into the SoundScape using DMA channel A.
+ */
+static int upload_dma_data(struct soundscape *s,
+ const unsigned char __user *data,
+ size_t size)
+{
+ unsigned long flags;
+ struct snd_dma_buffer dma;
+ int ret;
+
+ if (!get_dmabuf(&dma, PAGE_ALIGN(size)))
+ return -ENOMEM;
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ /*
+ * Reset the board ...
+ */
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f);
+
+ /*
+ * Enable the DMA channels and configure them ...
+ */
+ sscape_write_unsafe(s->io_base, GA_DMACFG_REG, 0x50);
+ sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT);
+ sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20);
+
+ /*
+ * Take the board out of reset ...
+ */
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80);
+
+ /*
+ * Upload the user's data (firmware?) to the SoundScape
+ * board through the DMA channel ...
+ */
+ while (size != 0) {
+ unsigned long len;
+
+ /*
+ * Apparently, copying to/from userspace can sleep.
+ * We are therefore forbidden from holding any
+ * spinlocks while we copy ...
+ */
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ /*
+ * Remember that the data that we want to DMA
+ * comes from USERSPACE. We have already verified
+ * the userspace pointer ...
+ */
+ len = min(size, dma.bytes);
+ len -= __copy_from_user(dma.area, data, len);
+ data += len;
+ size -= len;
+
+ /*
+ * Grab that spinlock again, now that we've
+ * finished copying!
+ */
+ spin_lock_irqsave(&s->lock, flags);
+
+ snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE);
+ sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG);
+ if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) {
+ /*
+ * Don't forget to release this spinlock we're holding ...
+ */
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ snd_printk(KERN_ERR "sscape: DMA upload has timed out\n");
+ ret = -EAGAIN;
+ goto _release_dma;
+ }
+ } /* while */
+
+ set_host_mode_unsafe(s->io_base);
+
+ /*
+ * Boot the board ... (I think)
+ */
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40);
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ /*
+ * If all has gone well, then the board should acknowledge
+ * the new upload and tell us that it has rebooted OK. We
+ * give it 5 seconds (max) ...
+ */
+ ret = 0;
+ if (!obp_startup_ack(s, 5)) {
+ snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n");
+ ret = -EAGAIN;
+ } else if (!host_startup_ack(s, 5)) {
+ snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n");
+ ret = -EAGAIN;
+ }
+
+_release_dma:
+ /*
+ * NOTE!!! We are NOT holding any spinlocks at this point !!!
+ */
+ sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40));
+ free_dmabuf(&dma);
+
+ return ret;
+}
+
+/*
+ * Upload the bootblock(?) into the SoundScape. The only
+ * purpose of this block of code seems to be to tell
+ * us which version of the microcode we should be using.
+ *
+ * NOTE: The boot-block data resides in USER-SPACE!!!
+ * However, we have already verified its memory
+ * addresses by the time we get here.
+ */
+static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb)
+{
+ unsigned long flags;
+ int data = 0;
+ int ret;
+
+ ret = upload_dma_data(sscape, bb->code, sizeof(bb->code));
+
+ spin_lock_irqsave(&sscape->lock, flags);
+ if (ret == 0) {
+ data = host_read_ctrl_unsafe(sscape->io_base, 100);
+ }
+ set_midi_mode_unsafe(sscape->io_base);
+ spin_unlock_irqrestore(&sscape->lock, flags);
+
+ if (ret == 0) {
+ if (data < 0) {
+ snd_printk(KERN_ERR "sscape: timeout reading firmware version\n");
+ ret = -EAGAIN;
+ }
+ else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) {
+ ret = -EFAULT;
+ }
+ }
+
+ return ret;
+}
+
+/*
+ * Upload the microcode into the SoundScape. The
+ * microcode is 64K of data, and if we try to copy
+ * it into a local variable then we will SMASH THE
+ * KERNEL'S STACK! We therefore leave it in USER
+ * SPACE, and save ourselves from copying it at all.
+ */
+static int sscape_upload_microcode(struct soundscape *sscape,
+ const struct sscape_microcode __user *mc)
+{
+ unsigned long flags;
+ char __user *code;
+ int err;
+
+ /*
+ * We are going to have to copy this data into a special
+ * DMA-able buffer before we can upload it. We shall therefore
+ * just check that the data pointer is valid for now.
+ *
+ * NOTE: This buffer is 64K long! That's WAY too big to
+ * copy into a stack-temporary anyway.
+ */
+ if ( get_user(code, &mc->code) ||
+ !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) )
+ return -EFAULT;
+
+ if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) {
+ snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n");
+ }
+
+ spin_lock_irqsave(&sscape->lock, flags);
+ set_midi_mode_unsafe(sscape->io_base);
+ spin_unlock_irqrestore(&sscape->lock, flags);
+
+ initialise_mpu401(sscape->mpu);
+
+ return err;
+}
+
+/*
+ * Hardware-specific device functions, to implement special
+ * IOCTLs for the SoundScape card. This is how we upload
+ * the microcode into the card, for example, and so we
+ * must ensure that no two processes can open this device
+ * simultaneously, and that we can't open it at all if
+ * someone is using the MIDI device.
+ */
+static int sscape_hw_open(struct snd_hwdep * hw, struct file *file)
+{
+ register struct soundscape *sscape = get_hwdep_soundscape(hw);
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&sscape->fwlock, flags);
+
+ if ((sscape->midi_usage != 0) || sscape->hw_in_use) {
+ err = -EBUSY;
+ } else {
+ sscape->hw_in_use = 1;
+ err = 0;
+ }
+
+ spin_unlock_irqrestore(&sscape->fwlock, flags);
+ return err;
+}
+
+static int sscape_hw_release(struct snd_hwdep * hw, struct file *file)
+{
+ register struct soundscape *sscape = get_hwdep_soundscape(hw);
+ unsigned long flags;
+
+ spin_lock_irqsave(&sscape->fwlock, flags);
+ sscape->hw_in_use = 0;
+ spin_unlock_irqrestore(&sscape->fwlock, flags);
+ return 0;
+}
+
+static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct soundscape *sscape = get_hwdep_soundscape(hw);
+ int err = -EBUSY;
+
+ switch (cmd) {
+ case SND_SSCAPE_LOAD_BOOTB:
+ {
+ register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg;
+
+ /*
+ * We are going to have to copy this data into a special
+ * DMA-able buffer before we can upload it. We shall therefore
+ * just check that the data pointer is valid for now ...
+ */
+ if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) )
+ return -EFAULT;
+
+ /*
+ * Now check that we can write the firmware version number too...
+ */
+ if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) )
+ return -EFAULT;
+
+ err = sscape_upload_bootblock(sscape, bb);
+ }
+ break;
+
+ case SND_SSCAPE_LOAD_MCODE:
+ {
+ register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg;
+
+ err = sscape_upload_microcode(sscape, mc);
+ }
+ break;
+
+ default:
+ err = -EINVAL;
+ break;
+ } /* switch */
+
+ return err;
+}
+
+
+/*
+ * Mixer control for the SoundScape's MIDI device.
+ */
+static int sscape_midi_info(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 127;
+ return 0;
+}
+
+static int sscape_midi_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kctl);
+ struct snd_card *card = chip->card;
+ register struct soundscape *s = get_card_soundscape(card);
+ unsigned long flags;
+
+ spin_lock_irqsave(&s->lock, flags);
+ set_host_mode_unsafe(s->io_base);
+
+ if (host_write_ctrl_unsafe(s->io_base, CMD_GET_MIDI_VOL, 100)) {
+ uctl->value.integer.value[0] = host_read_ctrl_unsafe(s->io_base, 100);
+ }
+
+ set_midi_mode_unsafe(s->io_base);
+ spin_unlock_irqrestore(&s->lock, flags);
+ return 0;
+}
+
+static int sscape_midi_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *uctl)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kctl);
+ struct snd_card *card = chip->card;
+ register struct soundscape *s = get_card_soundscape(card);
+ unsigned long flags;
+ int change;
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ /*
+ * We need to put the board into HOST mode before we
+ * can send any volume-changing HOST commands ...
+ */
+ set_host_mode_unsafe(s->io_base);
+
+ /*
+ * To successfully change the MIDI volume setting, you seem to
+ * have to write a volume command, write the new volume value,
+ * and then perform another volume-related command. Perhaps the
+ * first command is an "open" and the second command is a "close"?
+ */
+ if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) {
+ change = 0;
+ goto __skip_change;
+ }
+ change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100)
+ && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100)
+ && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100));
+ __skip_change:
+
+ /*
+ * Take the board out of HOST mode and back into MIDI mode ...
+ */
+ set_midi_mode_unsafe(s->io_base);
+
+ spin_unlock_irqrestore(&s->lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new midi_mixer_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "MIDI",
+ .info = sscape_midi_info,
+ .get = sscape_midi_get,
+ .put = sscape_midi_put
+};
+
+/*
+ * The SoundScape can use two IRQs from a possible set of four.
+ * These IRQs are encoded as bit patterns so that they can be
+ * written to the control registers.
+ */
+static unsigned __devinit get_irq_config(int irq)
+{
+ static const int valid_irq[] = { 9, 5, 7, 10 };
+ unsigned cfg;
+
+ for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) {
+ if (irq == valid_irq[cfg])
+ return cfg;
+ } /* for */
+
+ return INVALID_IRQ;
+}
+
+
+/*
+ * Perform certain arcane port-checks to see whether there
+ * is a SoundScape board lurking behind the given ports.
+ */
+static int __devinit detect_sscape(struct soundscape *s)
+{
+ unsigned long flags;
+ unsigned d;
+ int retval = 0;
+ int codec = s->wss_base;
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ /*
+ * The following code is lifted from the original OSS driver,
+ * and as I don't have a datasheet I cannot really comment
+ * on what it is doing...
+ */
+ if ((inb(HOST_CTRL_IO(s->io_base)) & 0x78) != 0)
+ goto _done;
+
+ d = inb(ODIE_ADDR_IO(s->io_base)) & 0xf0;
+ if ((d & 0x80) != 0)
+ goto _done;
+
+ if (d == 0) {
+ s->codec_type = 1;
+ s->ic_type = IC_ODIE;
+ } else if ((d & 0x60) != 0) {
+ s->codec_type = 2;
+ s->ic_type = IC_OPUS;
+ } else
+ goto _done;
+
+ outb(0xfa, ODIE_ADDR_IO(s->io_base));
+ if ((inb(ODIE_ADDR_IO(s->io_base)) & 0x9f) != 0x0a)
+ goto _done;
+
+ outb(0xfe, ODIE_ADDR_IO(s->io_base));
+ if ((inb(ODIE_ADDR_IO(s->io_base)) & 0x9f) != 0x0e)
+ goto _done;
+
+ outb(0xfe, ODIE_ADDR_IO(s->io_base));
+ d = inb(ODIE_DATA_IO(s->io_base));
+ if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e)
+ goto _done;
+
+ d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f;
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0);
+
+ if (s->type == SSCAPE_VIVO)
+ codec += 4;
+ /* wait for WSS codec */
+ for (d = 0; d < 500; d++) {
+ if ((inb(codec) & 0x80) == 0)
+ break;
+ spin_unlock_irqrestore(&s->lock, flags);
+ msleep(1);
+ spin_lock_irqsave(&s->lock, flags);
+ }
+ snd_printd(KERN_INFO "init delay = %d ms\n", d);
+
+ /*
+ * SoundScape successfully detected!
+ */
+ retval = 1;
+
+ _done:
+ spin_unlock_irqrestore(&s->lock, flags);
+ return retval;
+}
+
+/*
+ * ALSA callback function, called when attempting to open the MIDI device.
+ * Check that the MIDI firmware has been loaded, because we don't want
+ * to crash the machine. Also check that someone isn't using the hardware
+ * IOCTL device.
+ */
+static int mpu401_open(struct snd_mpu401 * mpu)
+{
+ int err;
+
+ if (!verify_mpu401(mpu)) {
+ snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n");
+ err = -ENODEV;
+ } else {
+ register struct soundscape *sscape = get_mpu401_soundscape(mpu);
+ unsigned long flags;
+
+ spin_lock_irqsave(&sscape->fwlock, flags);
+
+ if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) {
+ err = -EBUSY;
+ } else {
+ ++(sscape->midi_usage);
+ err = 0;
+ }
+
+ spin_unlock_irqrestore(&sscape->fwlock, flags);
+ }
+
+ return err;
+}
+
+static void mpu401_close(struct snd_mpu401 * mpu)
+{
+ register struct soundscape *sscape = get_mpu401_soundscape(mpu);
+ unsigned long flags;
+
+ spin_lock_irqsave(&sscape->fwlock, flags);
+ --(sscape->midi_usage);
+ spin_unlock_irqrestore(&sscape->fwlock, flags);
+}
+
+/*
+ * Initialse an MPU-401 subdevice for MIDI support on the SoundScape.
+ */
+static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq)
+{
+ struct soundscape *sscape = get_card_soundscape(card);
+ struct snd_rawmidi *rawmidi;
+ int err;
+
+ if ((err = snd_mpu401_uart_new(card, devnum,
+ MPU401_HW_MPU401,
+ port, MPU401_INFO_INTEGRATED,
+ irq, IRQF_DISABLED,
+ &rawmidi)) == 0) {
+ struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data;
+ mpu->open_input = mpu401_open;
+ mpu->open_output = mpu401_open;
+ mpu->close_input = mpu401_close;
+ mpu->close_output = mpu401_close;
+ mpu->private_data = sscape;
+ sscape->mpu = mpu;
+
+ initialise_mpu401(mpu);
+ }
+
+ return err;
+}
+
+
+/*
+ * Override for the CS4231 playback format function.
+ * The AD1845 has much simpler format and rate selection.
+ */
+static void ad1845_playback_format(struct snd_wss *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char format)
+{
+ unsigned long flags;
+ unsigned rate = params_rate(params);
+
+ /*
+ * The AD1845 can't handle sample frequencies
+ * outside of 4 kHZ to 50 kHZ
+ */
+ if (rate > 50000)
+ rate = 50000;
+ else if (rate < 4000)
+ rate = 4000;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ /*
+ * Program the AD1845 correctly for the playback stream.
+ * Note that we do NOT need to toggle the MCE bit because
+ * the PLAYBACK_ENABLE bit of the Interface Configuration
+ * register is set.
+ *
+ * NOTE: We seem to need to write to the MSB before the LSB
+ * to get the correct sample frequency.
+ */
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0));
+ snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8));
+ snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate);
+
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+/*
+ * Override for the CS4231 capture format function.
+ * The AD1845 has much simpler format and rate selection.
+ */
+static void ad1845_capture_format(struct snd_wss *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char format)
+{
+ unsigned long flags;
+ unsigned rate = params_rate(params);
+
+ /*
+ * The AD1845 can't handle sample frequencies
+ * outside of 4 kHZ to 50 kHZ
+ */
+ if (rate > 50000)
+ rate = 50000;
+ else if (rate < 4000)
+ rate = 4000;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ /*
+ * Program the AD1845 correctly for the playback stream.
+ * Note that we do NOT need to toggle the MCE bit because
+ * the CAPTURE_ENABLE bit of the Interface Configuration
+ * register is set.
+ *
+ * NOTE: We seem to need to write to the MSB before the LSB
+ * to get the correct sample frequency.
+ */
+ snd_wss_out(chip, CS4231_REC_FORMAT, (format & 0xf0));
+ snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8));
+ snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate);
+
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+/*
+ * Create an AD1845 PCM subdevice on the SoundScape. The AD1845
+ * is very much like a CS4231, with a few extra bits. We will
+ * try to support at least some of the extra bits by overriding
+ * some of the CS4231 callback.
+ */
+static int __devinit create_ad1845(struct snd_card *card, unsigned port,
+ int irq, int dma1, int dma2)
+{
+ register struct soundscape *sscape = get_card_soundscape(card);
+ struct snd_wss *chip;
+ int err;
+
+ if (sscape->type == SSCAPE_VIVO)
+ port += 4;
+
+ if (dma1 == dma2)
+ dma2 = -1;
+
+ err = snd_wss_create(card, port, -1, irq, dma1, dma2,
+ WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip);
+ if (!err) {
+ unsigned long flags;
+ struct snd_pcm *pcm;
+
+#define AD1845_FREQ_SEL_ENABLE 0x08
+
+#define AD1845_PWR_DOWN_CTRL 0x1b
+#define AD1845_CRYS_CLOCK_SEL 0x1d
+
+/*
+ * It turns out that the PLAYBACK_ENABLE bit is set
+ * by the lowlevel driver ...
+ *
+#define AD1845_IFACE_CONFIG \
+ (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE)
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+ */
+
+ if (sscape->type != SSCAPE_VIVO) {
+ int val;
+ /*
+ * The input clock frequency on the SoundScape must
+ * be 14.31818 MHz, because we must set this register
+ * to get the playback to sound correct ...
+ */
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_wss_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+
+ /*
+ * More custom configuration:
+ * a) select "mode 2" and provide a current drive of 8mA
+ * b) enable frequency selection (for capture/playback)
+ */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_wss_out(chip, CS4231_MISC_INFO,
+ CS4231_MODE2 | 0x10);
+ val = snd_wss_in(chip, AD1845_PWR_DOWN_CTRL);
+ snd_wss_out(chip, AD1845_PWR_DOWN_CTRL,
+ val | AD1845_FREQ_SEL_ENABLE);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
+
+ err = snd_wss_pcm(chip, 0, &pcm);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: No PCM device "
+ "for AD1845 chip\n");
+ goto _error;
+ }
+
+ err = snd_wss_mixer(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: No mixer device "
+ "for AD1845 chip\n");
+ goto _error;
+ }
+ err = snd_wss_timer(chip, 0, NULL);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: No timer device "
+ "for AD1845 chip\n");
+ goto _error;
+ }
+
+ if (sscape->type != SSCAPE_VIVO) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&midi_mixer_ctl, chip));
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: Could not create "
+ "MIDI mixer control\n");
+ goto _error;
+ }
+ chip->set_playback_format = ad1845_playback_format;
+ chip->set_capture_format = ad1845_capture_format;
+ }
+
+ strcpy(card->driver, "SoundScape");
+ strcpy(card->shortname, pcm->name);
+ snprintf(card->longname, sizeof(card->longname),
+ "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n",
+ pcm->name, chip->port, chip->irq,
+ chip->dma1, chip->dma2);
+
+ sscape->chip = chip;
+ }
+
+ _error:
+ return err;
+}
+
+
+/*
+ * Create an ALSA soundcard entry for the SoundScape, using
+ * the given list of port, IRQ and DMA resources.
+ */
+static int __devinit create_sscape(int dev, struct snd_card *card)
+{
+ struct soundscape *sscape = get_card_soundscape(card);
+ unsigned dma_cfg;
+ unsigned irq_cfg;
+ unsigned mpu_irq_cfg;
+ unsigned xport;
+ struct resource *io_res;
+ struct resource *wss_res;
+ unsigned long flags;
+ int err;
+
+ /*
+ * Check that the user didn't pass us garbage data ...
+ */
+ irq_cfg = get_irq_config(irq[dev]);
+ if (irq_cfg == INVALID_IRQ) {
+ snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]);
+ return -ENXIO;
+ }
+
+ mpu_irq_cfg = get_irq_config(mpu_irq[dev]);
+ if (mpu_irq_cfg == INVALID_IRQ) {
+ printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]);
+ return -ENXIO;
+ }
+ xport = port[dev];
+
+ /*
+ * Grab IO ports that we will need to probe so that we
+ * can detect and control this hardware ...
+ */
+ io_res = request_region(xport, 8, "SoundScape");
+ if (!io_res) {
+ snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport);
+ return -EBUSY;
+ }
+ wss_res = NULL;
+ if (sscape->type == SSCAPE_VIVO) {
+ wss_res = request_region(wss_port[dev], 4, "SoundScape");
+ if (!wss_res) {
+ snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n",
+ wss_port[dev]);
+ err = -EBUSY;
+ goto _release_region;
+ }
+ }
+
+ /*
+ * Grab one DMA channel ...
+ */
+ err = request_dma(dma[dev], "SoundScape");
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]);
+ goto _release_region;
+ }
+
+ spin_lock_init(&sscape->lock);
+ spin_lock_init(&sscape->fwlock);
+ sscape->io_res = io_res;
+ sscape->wss_res = wss_res;
+ sscape->io_base = xport;
+ sscape->wss_base = wss_port[dev];
+
+ if (!detect_sscape(sscape)) {
+ printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base);
+ err = -ENODEV;
+ goto _release_dma;
+ }
+
+ printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n",
+ sscape->io_base, irq[dev], dma[dev]);
+
+ if (sscape->type != SSCAPE_VIVO) {
+ /*
+ * Now create the hardware-specific device so that we can
+ * load the microcode into the on-board processor.
+ * We cannot use the MPU-401 MIDI system until this firmware
+ * has been loaded into the card.
+ */
+ err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw));
+ if (err < 0) {
+ printk(KERN_ERR "sscape: Failed to create "
+ "firmware device\n");
+ goto _release_dma;
+ }
+ strlcpy(sscape->hw->name, "SoundScape M68K",
+ sizeof(sscape->hw->name));
+ sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0';
+ sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE;
+ sscape->hw->ops.open = sscape_hw_open;
+ sscape->hw->ops.release = sscape_hw_release;
+ sscape->hw->ops.ioctl = sscape_hw_ioctl;
+ sscape->hw->private_data = sscape;
+ }
+
+ /*
+ * Tell the on-board devices where their resources are (I think -
+ * I can't be sure without a datasheet ... So many magic values!)
+ */
+ spin_lock_irqsave(&sscape->lock, flags);
+
+ activate_ad1845_unsafe(sscape->io_base);
+
+ sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */
+ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e);
+ sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00);
+
+ /*
+ * Enable and configure the DMA channels ...
+ */
+ sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50);
+ dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40);
+ sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg);
+ sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20);
+
+ sscape_write_unsafe(sscape->io_base,
+ GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg);
+ sscape_write_unsafe(sscape->io_base,
+ GA_CDCFG_REG, 0x09 | DMA_8BIT
+ | (dma[dev] << 4) | (irq_cfg << 1));
+
+ spin_unlock_irqrestore(&sscape->lock, flags);
+
+ /*
+ * We have now enabled the codec chip, and so we should
+ * detect the AD1845 device ...
+ */
+ err = create_ad1845(card, wss_port[dev], irq[dev],
+ dma[dev], dma2[dev]);
+ if (err < 0) {
+ printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n",
+ wss_port[dev], irq[dev]);
+ goto _release_dma;
+ }
+#define MIDI_DEVNUM 0
+ if (sscape->type != SSCAPE_VIVO) {
+ err = create_mpu401(card, MIDI_DEVNUM,
+ MPU401_IO(xport), mpu_irq[dev]);
+ if (err < 0) {
+ printk(KERN_ERR "sscape: Failed to create "
+ "MPU-401 device at 0x%x\n",
+ MPU401_IO(xport));
+ goto _release_dma;
+ }
+
+ /*
+ * Enable the master IRQ ...
+ */
+ sscape_write(sscape, GA_INTENA_REG, 0x80);
+
+ /*
+ * Initialize mixer
+ */
+ sscape->midi_vol = 0;
+ host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100);
+ host_write_ctrl_unsafe(sscape->io_base, 0, 100);
+ host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100);
+ }
+
+ /*
+ * Now that we have successfully created this sound card,
+ * it is safe to store the pointer.
+ * NOTE: we only register the sound card's "destructor"
+ * function now that our "constructor" has completed.
+ */
+ card->private_free = soundscape_free;
+
+ return 0;
+
+_release_dma:
+ free_dma(dma[dev]);
+
+_release_region:
+ release_and_free_resource(wss_res);
+ release_and_free_resource(io_res);
+
+ return err;
+}
+
+
+static int __devinit snd_sscape_match(struct device *pdev, unsigned int i)
+{
+ /*
+ * Make sure we were given ALL of the other parameters.
+ */
+ if (port[i] == SNDRV_AUTO_PORT)
+ return 0;
+
+ if (irq[i] == SNDRV_AUTO_IRQ ||
+ mpu_irq[i] == SNDRV_AUTO_IRQ ||
+ dma[i] == SNDRV_AUTO_DMA) {
+ printk(KERN_INFO
+ "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n");
+ return 0;
+ }
+
+ return 1;
+}
+
+static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev)
+{
+ struct snd_card *card;
+ struct soundscape *sscape;
+ int ret;
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct soundscape));
+ if (!card)
+ return -ENOMEM;
+
+ sscape = get_card_soundscape(card);
+ sscape->type = SSCAPE;
+
+ dma[dev] &= 0x03;
+ ret = create_sscape(dev, card);
+ if (ret < 0)
+ goto _release_card;
+
+ snd_card_set_dev(card, pdev);
+ if ((ret = snd_card_register(card)) < 0) {
+ printk(KERN_ERR "sscape: Failed to register sound card\n");
+ goto _release_card;
+ }
+ dev_set_drvdata(pdev, card);
+ return 0;
+
+_release_card:
+ snd_card_free(card);
+ return ret;
+}
+
+static int __devexit snd_sscape_remove(struct device *devptr, unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define DEV_NAME "sscape"
+
+static struct isa_driver snd_sscape_driver = {
+ .match = snd_sscape_match,
+ .probe = snd_sscape_probe,
+ .remove = __devexit_p(snd_sscape_remove),
+ /* FIXME: suspend/resume */
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+#ifdef CONFIG_PNP
+static inline int __devinit get_next_autoindex(int i)
+{
+ while (i < SNDRV_CARDS && port[i] != SNDRV_AUTO_PORT)
+ ++i;
+ return i;
+}
+
+
+static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ static int idx = 0;
+ struct pnp_dev *dev;
+ struct snd_card *card;
+ struct soundscape *sscape;
+ int ret;
+
+ /*
+ * Allow this function to fail *quietly* if all the ISA PnP
+ * devices were configured using module parameters instead.
+ */
+ if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS)
+ return -ENOSPC;
+
+ /*
+ * We have found a candidate ISA PnP card. Now we
+ * have to check that it has the devices that we
+ * expect it to have.
+ *
+ * We will NOT try and autoconfigure all of the resources
+ * needed and then activate the card as we are assuming that
+ * has already been done at boot-time using /proc/isapnp.
+ * We shall simply try to give each active card the resources
+ * that it wants. This is a sensible strategy for a modular
+ * system where unused modules are unloaded regularly.
+ *
+ * This strategy is utterly useless if we compile the driver
+ * into the kernel, of course.
+ */
+ // printk(KERN_INFO "sscape: %s\n", card->name);
+
+ /*
+ * Check that we still have room for another sound card ...
+ */
+ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL);
+ if (! dev)
+ return -ENODEV;
+
+ if (!pnp_is_active(dev)) {
+ if (pnp_activate_dev(dev) < 0) {
+ printk(KERN_INFO "sscape: device is inactive\n");
+ return -EBUSY;
+ }
+ }
+
+ /*
+ * Create a new ALSA sound card entry, in anticipation
+ * of detecting our hardware ...
+ */
+ card = snd_card_new(index[idx], id[idx], THIS_MODULE,
+ sizeof(struct soundscape));
+ if (!card)
+ return -ENOMEM;
+
+ sscape = get_card_soundscape(card);
+
+ /*
+ * Identify card model ...
+ */
+ if (!strncmp("ENS4081", pid->id, 7))
+ sscape->type = SSCAPE_VIVO;
+ else
+ sscape->type = SSCAPE_PNP;
+
+ /*
+ * Read the correct parameters off the ISA PnP bus ...
+ */
+ port[idx] = pnp_port_start(dev, 0);
+ irq[idx] = pnp_irq(dev, 0);
+ mpu_irq[idx] = pnp_irq(dev, 1);
+ dma[idx] = pnp_dma(dev, 0) & 0x03;
+ if (sscape->type == SSCAPE_PNP) {
+ dma2[idx] = dma[idx];
+ wss_port[idx] = CODEC_IO(port[idx]);
+ } else {
+ wss_port[idx] = pnp_port_start(dev, 1);
+ dma2[idx] = pnp_dma(dev, 1);
+ }
+
+ ret = create_sscape(idx, card);
+ if (ret < 0)
+ goto _release_card;
+
+ snd_card_set_dev(card, &pcard->card->dev);
+ if ((ret = snd_card_register(card)) < 0) {
+ printk(KERN_ERR "sscape: Failed to register sound card\n");
+ goto _release_card;
+ }
+
+ pnp_set_card_drvdata(pcard, card);
+ ++idx;
+ return 0;
+
+_release_card:
+ snd_card_free(card);
+ return ret;
+}
+
+static void __devexit sscape_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+static struct pnp_card_driver sscape_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DO_NOT_CHANGE,
+ .name = "sscape",
+ .id_table = sscape_pnpids,
+ .probe = sscape_pnp_detect,
+ .remove = __devexit_p(sscape_pnp_remove),
+};
+
+#endif /* CONFIG_PNP */
+
+static int __init sscape_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&snd_sscape_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+
+ err = pnp_register_card_driver(&sscape_pnpc_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ if (isa_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit sscape_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnp_registered)
+ pnp_unregister_card_driver(&sscape_pnpc_driver);
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&snd_sscape_driver);
+}
+
+module_init(sscape_init);
+module_exit(sscape_exit);
diff --git a/sound/isa/wavefront/Makefile b/sound/isa/wavefront/Makefile
new file mode 100644
index 0000000..601bddd
--- /dev/null
+++ b/sound/isa/wavefront/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-wavefront-objs := wavefront.o wavefront_fx.o wavefront_synth.o wavefront_midi.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_WAVEFRONT) += snd-wavefront.o
diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c
new file mode 100644
index 0000000..4c095bc
--- /dev/null
+++ b/sound/isa/wavefront/wavefront.c
@@ -0,0 +1,687 @@
+/*
+ * ALSA card-level driver for Turtle Beach Wavefront cards
+ * (Maui,Tropez,Tropez+)
+ *
+ * Copyright (c) 1997-1999 by Paul Barton-Davis <pbd@op.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/err.h>
+#include <linux/isa.h>
+#include <linux/pnp.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/opl3.h>
+#include <sound/wss.h>
+#include <sound/snd_wavefront.h>
+
+MODULE_AUTHOR("Paul Barton-Davis <pbd@op.net>");
+MODULE_DESCRIPTION("Turtle Beach Wavefront");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Turtle Beach,Maui/Tropez/Tropez+}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+#endif
+static long cs4232_pcm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int cs4232_pcm_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,11,12,15 */
+static long cs4232_mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int cs4232_mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 9,11,12,15 */
+static long ics2115_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int ics2115_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 2,9,11,12,15 */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
+static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3,5,6,7 */
+static int use_cs4232_midi[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for WaveFront soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for WaveFront soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable WaveFront soundcard.");
+#ifdef CONFIG_PNP
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "ISA PnP detection for WaveFront soundcards.");
+#endif
+module_param_array(cs4232_pcm_port, long, NULL, 0444);
+MODULE_PARM_DESC(cs4232_pcm_port, "Port # for CS4232 PCM interface.");
+module_param_array(cs4232_pcm_irq, int, NULL, 0444);
+MODULE_PARM_DESC(cs4232_pcm_irq, "IRQ # for CS4232 PCM interface.");
+module_param_array(dma1, int, NULL, 0444);
+MODULE_PARM_DESC(dma1, "DMA1 # for CS4232 PCM interface.");
+module_param_array(dma2, int, NULL, 0444);
+MODULE_PARM_DESC(dma2, "DMA2 # for CS4232 PCM interface.");
+module_param_array(cs4232_mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(cs4232_mpu_port, "port # for CS4232 MPU-401 interface.");
+module_param_array(cs4232_mpu_irq, int, NULL, 0444);
+MODULE_PARM_DESC(cs4232_mpu_irq, "IRQ # for CS4232 MPU-401 interface.");
+module_param_array(ics2115_irq, int, NULL, 0444);
+MODULE_PARM_DESC(ics2115_irq, "IRQ # for ICS2115.");
+module_param_array(ics2115_port, long, NULL, 0444);
+MODULE_PARM_DESC(ics2115_port, "Port # for ICS2115.");
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM port #.");
+module_param_array(use_cs4232_midi, bool, NULL, 0444);
+MODULE_PARM_DESC(use_cs4232_midi, "Use CS4232 MPU-401 interface (inaccessibly located inside your computer)");
+
+#ifdef CONFIG_PNP
+static int isa_registered;
+static int pnp_registered;
+
+static struct pnp_card_device_id snd_wavefront_pnpids[] = {
+ /* Tropez */
+ { .id = "CSC7532", .devs = { { "CSC0000" }, { "CSC0010" }, { "PnPb006" }, { "CSC0004" } } },
+ /* Tropez+ */
+ { .id = "CSC7632", .devs = { { "CSC0000" }, { "CSC0010" }, { "PnPb006" }, { "CSC0004" } } },
+ { .id = "" }
+};
+
+MODULE_DEVICE_TABLE(pnp_card, snd_wavefront_pnpids);
+
+static int __devinit
+snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *card,
+ const struct pnp_card_device_id *id)
+{
+ struct pnp_dev *pdev;
+ int err;
+
+ /* Check for each logical device. */
+
+ /* CS4232 chip (aka "windows sound system") is logical device 0 */
+
+ acard->wss = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (acard->wss == NULL)
+ return -EBUSY;
+
+ /* there is a game port at logical device 1, but we ignore it completely */
+
+ /* the control interface is logical device 2, but we ignore it
+ completely. in fact, nobody even seems to know what it
+ does.
+ */
+
+ /* Only configure the CS4232 MIDI interface if its been
+ specifically requested. It is logical device 3.
+ */
+
+ if (use_cs4232_midi[dev]) {
+ acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL);
+ if (acard->mpu == NULL)
+ return -EBUSY;
+ }
+
+ /* The ICS2115 synth is logical device 4 */
+
+ acard->synth = pnp_request_card_device(card, id->devs[3].id, NULL);
+ if (acard->synth == NULL)
+ return -EBUSY;
+
+ /* PCM/FM initialization */
+
+ pdev = acard->wss;
+
+ /* An interesting note from the Tropez+ FAQ:
+
+ Q. [Ports] Why is the base address of the WSS I/O ports off by 4?
+
+ A. WSS I/O requires a block of 8 I/O addresses ("ports"). Of these, the first
+ 4 are used to identify and configure the board. With the advent of PnP,
+ these first 4 addresses have become obsolete, and software applications
+ only use the last 4 addresses to control the codec chip. Therefore, the
+ base address setting "skips past" the 4 unused addresses.
+
+ */
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "PnP WSS pnp configure failure\n");
+ return err;
+ }
+
+ cs4232_pcm_port[dev] = pnp_port_start(pdev, 0);
+ fm_port[dev] = pnp_port_start(pdev, 1);
+ dma1[dev] = pnp_dma(pdev, 0);
+ dma2[dev] = pnp_dma(pdev, 1);
+ cs4232_pcm_irq[dev] = pnp_irq(pdev, 0);
+
+ /* Synth initialization */
+
+ pdev = acard->synth;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "PnP ICS2115 pnp configure failure\n");
+ return err;
+ }
+
+ ics2115_port[dev] = pnp_port_start(pdev, 0);
+ ics2115_irq[dev] = pnp_irq(pdev, 0);
+
+ /* CS4232 MPU initialization. Configure this only if
+ explicitly requested, since its physically inaccessible and
+ consumes another IRQ.
+ */
+
+ if (use_cs4232_midi[dev]) {
+
+ pdev = acard->mpu;
+
+ err = pnp_activate_dev(pdev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "PnP MPU401 pnp configure failure\n");
+ cs4232_mpu_port[dev] = SNDRV_AUTO_PORT;
+ } else {
+ cs4232_mpu_port[dev] = pnp_port_start(pdev, 0);
+ cs4232_mpu_irq[dev] = pnp_irq(pdev, 0);
+ }
+
+ snd_printk (KERN_INFO "CS4232 MPU: port=0x%lx, irq=%i\n",
+ cs4232_mpu_port[dev],
+ cs4232_mpu_irq[dev]);
+ }
+
+ snd_printdd ("CS4232: pcm port=0x%lx, fm port=0x%lx, dma1=%i, dma2=%i, irq=%i\nICS2115: port=0x%lx, irq=%i\n",
+ cs4232_pcm_port[dev],
+ fm_port[dev],
+ dma1[dev],
+ dma2[dev],
+ cs4232_pcm_irq[dev],
+ ics2115_port[dev],
+ ics2115_irq[dev]);
+
+ return 0;
+}
+
+#endif /* CONFIG_PNP */
+
+static irqreturn_t snd_wavefront_ics2115_interrupt(int irq, void *dev_id)
+{
+ snd_wavefront_card_t *acard;
+
+ acard = (snd_wavefront_card_t *) dev_id;
+
+ if (acard == NULL)
+ return IRQ_NONE;
+
+ if (acard->wavefront.interrupts_are_midi) {
+ snd_wavefront_midi_interrupt (acard);
+ } else {
+ snd_wavefront_internal_interrupt (acard);
+ }
+ return IRQ_HANDLED;
+}
+
+static struct snd_hwdep * __devinit
+snd_wavefront_new_synth (struct snd_card *card,
+ int hw_dev,
+ snd_wavefront_card_t *acard)
+{
+ struct snd_hwdep *wavefront_synth;
+
+ if (snd_wavefront_detect (acard) < 0) {
+ return NULL;
+ }
+
+ if (snd_wavefront_start (&acard->wavefront) < 0) {
+ return NULL;
+ }
+
+ if (snd_hwdep_new(card, "WaveFront", hw_dev, &wavefront_synth) < 0)
+ return NULL;
+ strcpy (wavefront_synth->name,
+ "WaveFront (ICS2115) wavetable synthesizer");
+ wavefront_synth->ops.open = snd_wavefront_synth_open;
+ wavefront_synth->ops.release = snd_wavefront_synth_release;
+ wavefront_synth->ops.ioctl = snd_wavefront_synth_ioctl;
+
+ return wavefront_synth;
+}
+
+static struct snd_hwdep * __devinit
+snd_wavefront_new_fx (struct snd_card *card,
+ int hw_dev,
+ snd_wavefront_card_t *acard,
+ unsigned long port)
+
+{
+ struct snd_hwdep *fx_processor;
+
+ if (snd_wavefront_fx_start (&acard->wavefront)) {
+ snd_printk (KERN_ERR "cannot initialize YSS225 FX processor");
+ return NULL;
+ }
+
+ if (snd_hwdep_new (card, "YSS225", hw_dev, &fx_processor) < 0)
+ return NULL;
+ sprintf (fx_processor->name, "YSS225 FX Processor at 0x%lx", port);
+ fx_processor->ops.open = snd_wavefront_fx_open;
+ fx_processor->ops.release = snd_wavefront_fx_release;
+ fx_processor->ops.ioctl = snd_wavefront_fx_ioctl;
+
+ return fx_processor;
+}
+
+static snd_wavefront_mpu_id internal_id = internal_mpu;
+static snd_wavefront_mpu_id external_id = external_mpu;
+
+static struct snd_rawmidi *__devinit
+snd_wavefront_new_midi (struct snd_card *card,
+ int midi_dev,
+ snd_wavefront_card_t *acard,
+ unsigned long port,
+ snd_wavefront_mpu_id mpu)
+
+{
+ struct snd_rawmidi *rmidi;
+ static int first = 1;
+
+ if (first) {
+ first = 0;
+ acard->wavefront.midi.base = port;
+ if (snd_wavefront_midi_start (acard)) {
+ snd_printk (KERN_ERR "cannot initialize MIDI interface\n");
+ return NULL;
+ }
+ }
+
+ if (snd_rawmidi_new (card, "WaveFront MIDI", midi_dev, 1, 1, &rmidi) < 0)
+ return NULL;
+
+ if (mpu == internal_mpu) {
+ strcpy(rmidi->name, "WaveFront MIDI (Internal)");
+ rmidi->private_data = &internal_id;
+ } else {
+ strcpy(rmidi->name, "WaveFront MIDI (External)");
+ rmidi->private_data = &external_id;
+ }
+
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_wavefront_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_wavefront_midi_input);
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return rmidi;
+}
+
+static void
+snd_wavefront_free(struct snd_card *card)
+{
+ snd_wavefront_card_t *acard = (snd_wavefront_card_t *)card->private_data;
+
+ if (acard) {
+ release_and_free_resource(acard->wavefront.res_base);
+ if (acard->wavefront.irq > 0)
+ free_irq(acard->wavefront.irq, (void *)acard);
+ }
+}
+
+static struct snd_card *snd_wavefront_card_new(int dev)
+{
+ struct snd_card *card;
+ snd_wavefront_card_t *acard;
+
+ card = snd_card_new (index[dev], id[dev], THIS_MODULE,
+ sizeof(snd_wavefront_card_t));
+ if (card == NULL)
+ return NULL;
+
+ acard = card->private_data;
+ acard->wavefront.irq = -1;
+ spin_lock_init(&acard->wavefront.irq_lock);
+ init_waitqueue_head(&acard->wavefront.interrupt_sleeper);
+ spin_lock_init(&acard->wavefront.midi.open);
+ spin_lock_init(&acard->wavefront.midi.virtual);
+ acard->wavefront.card = card;
+ card->private_free = snd_wavefront_free;
+
+ return card;
+}
+
+static int __devinit
+snd_wavefront_probe (struct snd_card *card, int dev)
+{
+ snd_wavefront_card_t *acard = card->private_data;
+ struct snd_wss *chip;
+ struct snd_hwdep *wavefront_synth;
+ struct snd_rawmidi *ics2115_internal_rmidi = NULL;
+ struct snd_rawmidi *ics2115_external_rmidi = NULL;
+ struct snd_hwdep *fx_processor;
+ int hw_dev = 0, midi_dev = 0, err;
+
+ /* --------- PCM --------------- */
+
+ err = snd_wss_create(card, cs4232_pcm_port[dev], -1,
+ cs4232_pcm_irq[dev], dma1[dev], dma2[dev],
+ WSS_HW_DETECT, 0, &chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "can't allocate WSS device\n");
+ return err;
+ }
+
+ err = snd_wss_pcm(chip, 0, NULL);
+ if (err < 0)
+ return err;
+
+ err = snd_wss_timer(chip, 0, NULL);
+ if (err < 0)
+ return err;
+
+ /* ---------- OPL3 synth --------- */
+
+ if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
+ struct snd_opl3 *opl3;
+
+ err = snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2,
+ OPL3_HW_OPL3_CS, 0, &opl3);
+ if (err < 0) {
+ snd_printk (KERN_ERR "can't allocate or detect OPL3 synth\n");
+ return err;
+ }
+
+ err = snd_opl3_hwdep_new(opl3, hw_dev, 1, NULL);
+ if (err < 0)
+ return err;
+ hw_dev++;
+ }
+
+ /* ------- ICS2115 Wavetable synth ------- */
+
+ acard->wavefront.res_base = request_region(ics2115_port[dev], 16,
+ "ICS2115");
+ if (acard->wavefront.res_base == NULL) {
+ snd_printk(KERN_ERR "unable to grab ICS2115 i/o region 0x%lx-0x%lx\n",
+ ics2115_port[dev], ics2115_port[dev] + 16 - 1);
+ return -EBUSY;
+ }
+ if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt,
+ IRQF_DISABLED, "ICS2115", acard)) {
+ snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]);
+ return -EBUSY;
+ }
+
+ acard->wavefront.irq = ics2115_irq[dev];
+ acard->wavefront.base = ics2115_port[dev];
+
+ wavefront_synth = snd_wavefront_new_synth(card, hw_dev, acard);
+ if (wavefront_synth == NULL) {
+ snd_printk (KERN_ERR "can't create WaveFront synth device\n");
+ return -ENOMEM;
+ }
+
+ strcpy (wavefront_synth->name, "ICS2115 Wavetable MIDI Synthesizer");
+ wavefront_synth->iface = SNDRV_HWDEP_IFACE_ICS2115;
+ hw_dev++;
+
+ /* --------- Mixer ------------ */
+
+ err = snd_wss_mixer(chip);
+ if (err < 0) {
+ snd_printk (KERN_ERR "can't allocate mixer device\n");
+ return err;
+ }
+
+ /* -------- CS4232 MPU-401 interface -------- */
+
+ if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) {
+ err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232,
+ cs4232_mpu_port[dev], 0,
+ cs4232_mpu_irq[dev], IRQF_DISABLED,
+ NULL);
+ if (err < 0) {
+ snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n");
+ return err;
+ }
+ midi_dev++;
+ }
+
+ /* ------ ICS2115 internal MIDI ------------ */
+
+ if (ics2115_port[dev] > 0 && ics2115_port[dev] != SNDRV_AUTO_PORT) {
+ ics2115_internal_rmidi =
+ snd_wavefront_new_midi (card,
+ midi_dev,
+ acard,
+ ics2115_port[dev],
+ internal_mpu);
+ if (ics2115_internal_rmidi == NULL) {
+ snd_printk (KERN_ERR "can't setup ICS2115 internal MIDI device\n");
+ return -ENOMEM;
+ }
+ midi_dev++;
+ }
+
+ /* ------ ICS2115 external MIDI ------------ */
+
+ if (ics2115_port[dev] > 0 && ics2115_port[dev] != SNDRV_AUTO_PORT) {
+ ics2115_external_rmidi =
+ snd_wavefront_new_midi (card,
+ midi_dev,
+ acard,
+ ics2115_port[dev],
+ external_mpu);
+ if (ics2115_external_rmidi == NULL) {
+ snd_printk (KERN_ERR "can't setup ICS2115 external MIDI device\n");
+ return -ENOMEM;
+ }
+ midi_dev++;
+ }
+
+ /* FX processor for Tropez+ */
+
+ if (acard->wavefront.has_fx) {
+ fx_processor = snd_wavefront_new_fx (card,
+ hw_dev,
+ acard,
+ ics2115_port[dev]);
+ if (fx_processor == NULL) {
+ snd_printk (KERN_ERR "can't setup FX device\n");
+ return -ENOMEM;
+ }
+
+ hw_dev++;
+
+ strcpy(card->driver, "Tropez+");
+ strcpy(card->shortname, "Turtle Beach Tropez+");
+ } else {
+ /* Need a way to distinguish between Maui and Tropez */
+ strcpy(card->driver, "WaveFront");
+ strcpy(card->shortname, "Turtle Beach WaveFront");
+ }
+
+ /* ----- Register the card --------- */
+
+ /* Not safe to include "Turtle Beach" in longname, due to
+ length restrictions
+ */
+
+ sprintf(card->longname, "%s PCM 0x%lx irq %d dma %d",
+ card->driver,
+ chip->port,
+ cs4232_pcm_irq[dev],
+ dma1[dev]);
+
+ if (dma2[dev] >= 0 && dma2[dev] < 8)
+ sprintf(card->longname + strlen(card->longname), "&%d", dma2[dev]);
+
+ if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) {
+ sprintf (card->longname + strlen (card->longname),
+ " MPU-401 0x%lx irq %d",
+ cs4232_mpu_port[dev],
+ cs4232_mpu_irq[dev]);
+ }
+
+ sprintf (card->longname + strlen (card->longname),
+ " SYNTH 0x%lx irq %d",
+ ics2115_port[dev],
+ ics2115_irq[dev]);
+
+ return snd_card_register(card);
+}
+
+static int __devinit snd_wavefront_isa_match(struct device *pdev,
+ unsigned int dev)
+{
+ if (!enable[dev])
+ return 0;
+#ifdef CONFIG_PNP
+ if (isapnp[dev])
+ return 0;
+#endif
+ if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk("specify CS4232 port\n");
+ return 0;
+ }
+ if (ics2115_port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk("specify ICS2115 port\n");
+ return 0;
+ }
+ return 1;
+}
+
+static int __devinit snd_wavefront_isa_probe(struct device *pdev,
+ unsigned int dev)
+{
+ struct snd_card *card;
+ int err;
+
+ card = snd_wavefront_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+ snd_card_set_dev(card, pdev);
+ if ((err = snd_wavefront_probe(card, dev)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ dev_set_drvdata(pdev, card);
+ return 0;
+}
+
+static int __devexit snd_wavefront_isa_remove(struct device *devptr,
+ unsigned int dev)
+{
+ snd_card_free(dev_get_drvdata(devptr));
+ dev_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#define DEV_NAME "wavefront"
+
+static struct isa_driver snd_wavefront_driver = {
+ .match = snd_wavefront_isa_match,
+ .probe = snd_wavefront_isa_probe,
+ .remove = __devexit_p(snd_wavefront_isa_remove),
+ /* FIXME: suspend, resume */
+ .driver = {
+ .name = DEV_NAME
+ },
+};
+
+
+#ifdef CONFIG_PNP
+static int __devinit snd_wavefront_pnp_detect(struct pnp_card_link *pcard,
+ const struct pnp_card_device_id *pid)
+{
+ static int dev;
+ struct snd_card *card;
+ int res;
+
+ for ( ; dev < SNDRV_CARDS; dev++) {
+ if (enable[dev] && isapnp[dev])
+ break;
+ }
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ card = snd_wavefront_card_new(dev);
+ if (! card)
+ return -ENOMEM;
+
+ if (snd_wavefront_pnp (dev, card->private_data, pcard, pid) < 0) {
+ if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) {
+ snd_printk (KERN_ERR "isapnp detection failed\n");
+ snd_card_free (card);
+ return -ENODEV;
+ }
+ }
+ snd_card_set_dev(card, &pcard->card->dev);
+
+ if ((res = snd_wavefront_probe(card, dev)) < 0)
+ return res;
+
+ pnp_set_card_drvdata(pcard, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_wavefront_pnp_remove(struct pnp_card_link * pcard)
+{
+ snd_card_free(pnp_get_card_drvdata(pcard));
+ pnp_set_card_drvdata(pcard, NULL);
+}
+
+static struct pnp_card_driver wavefront_pnpc_driver = {
+ .flags = PNP_DRIVER_RES_DISABLE,
+ .name = "wavefront",
+ .id_table = snd_wavefront_pnpids,
+ .probe = snd_wavefront_pnp_detect,
+ .remove = __devexit_p(snd_wavefront_pnp_remove),
+ /* FIXME: suspend,resume */
+};
+
+#endif /* CONFIG_PNP */
+
+static int __init alsa_card_wavefront_init(void)
+{
+ int err;
+
+ err = isa_register_driver(&snd_wavefront_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+ if (!err)
+ isa_registered = 1;
+
+ err = pnp_register_card_driver(&wavefront_pnpc_driver);
+ if (!err)
+ pnp_registered = 1;
+
+ if (isa_registered)
+ err = 0;
+#endif
+ return err;
+}
+
+static void __exit alsa_card_wavefront_exit(void)
+{
+#ifdef CONFIG_PNP
+ if (pnp_registered)
+ pnp_unregister_card_driver(&wavefront_pnpc_driver);
+ if (isa_registered)
+#endif
+ isa_unregister_driver(&snd_wavefront_driver);
+}
+
+module_init(alsa_card_wavefront_init)
+module_exit(alsa_card_wavefront_exit)
diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c
new file mode 100644
index 0000000..dfc449a
--- /dev/null
+++ b/sound/isa/wavefront/wavefront_fx.c
@@ -0,0 +1,303 @@
+/*
+ * Copyright (c) 1998-2002 by Paul Davis <pbd@op.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <asm/io.h>
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/snd_wavefront.h>
+#include <sound/initval.h>
+
+/* Control bits for the Load Control Register
+ */
+
+#define FX_LSB_TRANSFER 0x01 /* transfer after DSP LSB byte written */
+#define FX_MSB_TRANSFER 0x02 /* transfer after DSP MSB byte written */
+#define FX_AUTO_INCR 0x04 /* auto-increment DSP address after transfer */
+
+#define WAIT_IDLE 0xff
+
+#ifdef CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL
+#include "yss225.c"
+static const struct firmware yss225_registers_firmware = {
+ .data = (u8 *)yss225_registers,
+ .size = sizeof yss225_registers
+};
+#endif
+
+static int
+wavefront_fx_idle (snd_wavefront_t *dev)
+
+{
+ int i;
+ unsigned int x = 0x80;
+
+ for (i = 0; i < 1000; i++) {
+ x = inb (dev->fx_status);
+ if ((x & 0x80) == 0) {
+ break;
+ }
+ }
+
+ if (x & 0x80) {
+ snd_printk ("FX device never idle.\n");
+ return 0;
+ }
+
+ return (1);
+}
+
+static void
+wavefront_fx_mute (snd_wavefront_t *dev, int onoff)
+
+{
+ if (!wavefront_fx_idle(dev)) {
+ return;
+ }
+
+ outb (onoff ? 0x02 : 0x00, dev->fx_op);
+}
+
+static int
+wavefront_fx_memset (snd_wavefront_t *dev,
+ int page,
+ int addr,
+ int cnt,
+ unsigned short *data)
+{
+ if (page < 0 || page > 7) {
+ snd_printk ("FX memset: "
+ "page must be >= 0 and <= 7\n");
+ return -(EINVAL);
+ }
+
+ if (addr < 0 || addr > 0x7f) {
+ snd_printk ("FX memset: "
+ "addr must be >= 0 and <= 7f\n");
+ return -(EINVAL);
+ }
+
+ if (cnt == 1) {
+
+ outb (FX_LSB_TRANSFER, dev->fx_lcr);
+ outb (page, dev->fx_dsp_page);
+ outb (addr, dev->fx_dsp_addr);
+ outb ((data[0] >> 8), dev->fx_dsp_msb);
+ outb ((data[0] & 0xff), dev->fx_dsp_lsb);
+
+ snd_printk ("FX: addr %d:%x set to 0x%x\n",
+ page, addr, data[0]);
+
+ } else {
+ int i;
+
+ outb (FX_AUTO_INCR|FX_LSB_TRANSFER, dev->fx_lcr);
+ outb (page, dev->fx_dsp_page);
+ outb (addr, dev->fx_dsp_addr);
+
+ for (i = 0; i < cnt; i++) {
+ outb ((data[i] >> 8), dev->fx_dsp_msb);
+ outb ((data[i] & 0xff), dev->fx_dsp_lsb);
+ if (!wavefront_fx_idle (dev)) {
+ break;
+ }
+ }
+
+ if (i != cnt) {
+ snd_printk ("FX memset "
+ "(0x%x, 0x%x, 0x%lx, %d) incomplete\n",
+ page, addr, (unsigned long) data, cnt);
+ return -(EIO);
+ }
+ }
+
+ return 0;
+}
+
+int
+snd_wavefront_fx_detect (snd_wavefront_t *dev)
+
+{
+ /* This is a crude check, but its the best one I have for now.
+ Certainly on the Maui and the Tropez, wavefront_fx_idle() will
+ report "never idle", which suggests that this test should
+ work OK.
+ */
+
+ if (inb (dev->fx_status) & 0x80) {
+ snd_printk ("Hmm, probably a Maui or Tropez.\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+int
+snd_wavefront_fx_open (struct snd_hwdep *hw, struct file *file)
+
+{
+ if (!try_module_get(hw->card->module))
+ return -EFAULT;
+ file->private_data = hw;
+ return 0;
+}
+
+int
+snd_wavefront_fx_release (struct snd_hwdep *hw, struct file *file)
+
+{
+ module_put(hw->card->module);
+ return 0;
+}
+
+int
+snd_wavefront_fx_ioctl (struct snd_hwdep *sdev, struct file *file,
+ unsigned int cmd, unsigned long arg)
+
+{
+ struct snd_card *card;
+ snd_wavefront_card_t *acard;
+ snd_wavefront_t *dev;
+ wavefront_fx_info r;
+ unsigned short *page_data = NULL;
+ unsigned short *pd;
+ int err = 0;
+
+ card = sdev->card;
+ if (snd_BUG_ON(!card))
+ return -ENODEV;
+ if (snd_BUG_ON(!card->private_data))
+ return -ENODEV;
+
+ acard = card->private_data;
+ dev = &acard->wavefront;
+
+ if (copy_from_user (&r, (void __user *)arg, sizeof (wavefront_fx_info)))
+ return -EFAULT;
+
+ switch (r.request) {
+ case WFFX_MUTE:
+ wavefront_fx_mute (dev, r.data[0]);
+ return -EIO;
+
+ case WFFX_MEMSET:
+ if (r.data[2] <= 0) {
+ snd_printk ("cannot write "
+ "<= 0 bytes to FX\n");
+ return -EIO;
+ } else if (r.data[2] == 1) {
+ pd = (unsigned short *) &r.data[3];
+ } else {
+ if (r.data[2] > 256) {
+ snd_printk ("cannot write "
+ "> 512 bytes to FX\n");
+ return -EIO;
+ }
+ page_data = kmalloc(r.data[2] * sizeof(short), GFP_KERNEL);
+ if (!page_data)
+ return -ENOMEM;
+ if (copy_from_user (page_data,
+ (unsigned char __user *) r.data[3],
+ r.data[2] * sizeof(short))) {
+ kfree(page_data);
+ return -EFAULT;
+ }
+ pd = page_data;
+ }
+
+ err = wavefront_fx_memset (dev,
+ r.data[0], /* page */
+ r.data[1], /* addr */
+ r.data[2], /* cnt */
+ pd);
+ kfree(page_data);
+ break;
+
+ default:
+ snd_printk ("FX: ioctl %d not yet supported\n",
+ r.request);
+ return -ENOTTY;
+ }
+ return err;
+}
+
+/* YSS225 initialization.
+
+ This code was developed using DOSEMU. The Turtle Beach SETUPSND
+ utility was run with I/O tracing in DOSEMU enabled, and a reconstruction
+ of the port I/O done, using the Yamaha faxback document as a guide
+ to add more logic to the code. Its really pretty weird.
+
+ This is the approach of just dumping the whole I/O
+ sequence as a series of port/value pairs and a simple loop
+ that outputs it.
+*/
+
+int __devinit
+snd_wavefront_fx_start (snd_wavefront_t *dev)
+{
+ unsigned int i;
+ int err;
+ const struct firmware *firmware = NULL;
+
+ if (dev->fx_initialized)
+ return 0;
+
+#ifdef CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL
+ firmware = &yss225_registers_firmware;
+#else
+ err = request_firmware(&firmware, "yamaha/yss225_registers.bin",
+ dev->card->dev);
+ if (err < 0) {
+ err = -1;
+ goto out;
+ }
+#endif
+
+ for (i = 0; i + 1 < firmware->size; i += 2) {
+ if (firmware->data[i] >= 8 && firmware->data[i] < 16) {
+ outb(firmware->data[i + 1],
+ dev->base + firmware->data[i]);
+ } else if (firmware->data[i] == WAIT_IDLE) {
+ if (!wavefront_fx_idle(dev)) {
+ err = -1;
+ goto out;
+ }
+ } else {
+ snd_printk(KERN_ERR "invalid address"
+ " in register data\n");
+ err = -1;
+ goto out;
+ }
+ }
+
+ dev->fx_initialized = 1;
+ err = 0;
+
+out:
+#ifndef CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL
+ release_firmware(firmware);
+#endif
+ return err;
+}
+
+#ifndef CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL
+MODULE_FIRMWARE("yamaha/yss225_registers.bin");
+#endif
diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c
new file mode 100644
index 0000000..f14a7c0
--- /dev/null
+++ b/sound/isa/wavefront/wavefront_midi.c
@@ -0,0 +1,577 @@
+/*
+ * Copyright (C) by Paul Barton-Davis 1998-1999
+ *
+ * This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this
+ * software for more info.
+ */
+
+/* The low level driver for the WaveFront ICS2115 MIDI interface(s)
+ *
+ * Note that there is also an MPU-401 emulation (actually, a UART-401
+ * emulation) on the CS4232 on the Tropez and Tropez Plus. This code
+ * has nothing to do with that interface at all.
+ *
+ * The interface is essentially just a UART-401, but is has the
+ * interesting property of supporting what Turtle Beach called
+ * "Virtual MIDI" mode. In this mode, there are effectively *two*
+ * MIDI buses accessible via the interface, one that is routed
+ * solely to/from the external WaveFront synthesizer and the other
+ * corresponding to the pin/socket connector used to link external
+ * MIDI devices to the board.
+ *
+ * This driver fully supports this mode, allowing two distinct MIDI
+ * busses to be used completely independently, giving 32 channels of
+ * MIDI routing, 16 to the WaveFront synth and 16 to the external MIDI
+ * bus. The devices are named /dev/snd/midiCnD0 and /dev/snd/midiCnD1,
+ * where `n' is the card number. Note that the device numbers may be
+ * something other than 0 and 1 if the CS4232 UART/MPU-401 interface
+ * is enabled.
+ *
+ * Switching between the two is accomplished externally by the driver
+ * using the two otherwise unused MIDI bytes. See the code for more details.
+ *
+ * NOTE: VIRTUAL MIDI MODE IS ON BY DEFAULT (see lowlevel/isa/wavefront.c)
+ *
+ * The main reason to turn off Virtual MIDI mode is when you want to
+ * tightly couple the WaveFront synth with an external MIDI
+ * device. You won't be able to distinguish the source of any MIDI
+ * data except via SysEx ID, but thats probably OK, since for the most
+ * part, the WaveFront won't be sending any MIDI data at all.
+ *
+ * The main reason to turn on Virtual MIDI Mode is to provide two
+ * completely independent 16-channel MIDI buses, one to the
+ * WaveFront and one to any external MIDI devices. Given the 32
+ * voice nature of the WaveFront, its pretty easy to find a use
+ * for all 16 channels driving just that synth.
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/init.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <sound/core.h>
+#include <sound/snd_wavefront.h>
+
+static inline int
+wf_mpu_status (snd_wavefront_midi_t *midi)
+
+{
+ return inb (midi->mpu_status_port);
+}
+
+static inline int
+input_avail (snd_wavefront_midi_t *midi)
+
+{
+ return !(wf_mpu_status(midi) & INPUT_AVAIL);
+}
+
+static inline int
+output_ready (snd_wavefront_midi_t *midi)
+
+{
+ return !(wf_mpu_status(midi) & OUTPUT_READY);
+}
+
+static inline int
+read_data (snd_wavefront_midi_t *midi)
+
+{
+ return inb (midi->mpu_data_port);
+}
+
+static inline void
+write_data (snd_wavefront_midi_t *midi, unsigned char byte)
+
+{
+ outb (byte, midi->mpu_data_port);
+}
+
+static snd_wavefront_midi_t *
+get_wavefront_midi (struct snd_rawmidi_substream *substream)
+
+{
+ struct snd_card *card;
+ snd_wavefront_card_t *acard;
+
+ if (substream == NULL || substream->rmidi == NULL)
+ return NULL;
+
+ card = substream->rmidi->card;
+
+ if (card == NULL)
+ return NULL;
+
+ if (card->private_data == NULL)
+ return NULL;
+
+ acard = card->private_data;
+
+ return &acard->wavefront.midi;
+}
+
+static void snd_wavefront_midi_output_write(snd_wavefront_card_t *card)
+{
+ snd_wavefront_midi_t *midi = &card->wavefront.midi;
+ snd_wavefront_mpu_id mpu;
+ unsigned long flags;
+ unsigned char midi_byte;
+ int max = 256, mask = 1;
+ int timeout;
+
+ /* Its not OK to try to change the status of "virtuality" of
+ the MIDI interface while we're outputting stuff. See
+ snd_wavefront_midi_{enable,disable}_virtual () for the
+ other half of this.
+
+ The first loop attempts to flush any data from the
+ current output device, and then the second
+ emits the switch byte (if necessary), and starts
+ outputting data for the output device currently in use.
+ */
+
+ if (midi->substream_output[midi->output_mpu] == NULL) {
+ goto __second;
+ }
+
+ while (max > 0) {
+
+ /* XXX fix me - no hard timing loops allowed! */
+
+ for (timeout = 30000; timeout > 0; timeout--) {
+ if (output_ready (midi))
+ break;
+ }
+
+ spin_lock_irqsave (&midi->virtual, flags);
+ if ((midi->mode[midi->output_mpu] & MPU401_MODE_OUTPUT) == 0) {
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ goto __second;
+ }
+ if (output_ready (midi)) {
+ if (snd_rawmidi_transmit(midi->substream_output[midi->output_mpu], &midi_byte, 1) == 1) {
+ if (!midi->isvirtual ||
+ (midi_byte != WF_INTERNAL_SWITCH &&
+ midi_byte != WF_EXTERNAL_SWITCH))
+ write_data(midi, midi_byte);
+ max--;
+ } else {
+ if (midi->istimer) {
+ if (--midi->istimer <= 0)
+ del_timer(&midi->timer);
+ }
+ midi->mode[midi->output_mpu] &= ~MPU401_MODE_OUTPUT_TRIGGER;
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ goto __second;
+ }
+ } else {
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ return;
+ }
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ }
+
+ __second:
+
+ if (midi->substream_output[!midi->output_mpu] == NULL) {
+ return;
+ }
+
+ while (max > 0) {
+
+ /* XXX fix me - no hard timing loops allowed! */
+
+ for (timeout = 30000; timeout > 0; timeout--) {
+ if (output_ready (midi))
+ break;
+ }
+
+ spin_lock_irqsave (&midi->virtual, flags);
+ if (!midi->isvirtual)
+ mask = 0;
+ mpu = midi->output_mpu ^ mask;
+ mask = 0; /* don't invert the value from now */
+ if ((midi->mode[mpu] & MPU401_MODE_OUTPUT) == 0) {
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ return;
+ }
+ if (snd_rawmidi_transmit_empty(midi->substream_output[mpu]))
+ goto __timer;
+ if (output_ready (midi)) {
+ if (mpu != midi->output_mpu) {
+ write_data(midi, mpu == internal_mpu ?
+ WF_INTERNAL_SWITCH :
+ WF_EXTERNAL_SWITCH);
+ midi->output_mpu = mpu;
+ } else if (snd_rawmidi_transmit(midi->substream_output[mpu], &midi_byte, 1) == 1) {
+ if (!midi->isvirtual ||
+ (midi_byte != WF_INTERNAL_SWITCH &&
+ midi_byte != WF_EXTERNAL_SWITCH))
+ write_data(midi, midi_byte);
+ max--;
+ } else {
+ __timer:
+ if (midi->istimer) {
+ if (--midi->istimer <= 0)
+ del_timer(&midi->timer);
+ }
+ midi->mode[mpu] &= ~MPU401_MODE_OUTPUT_TRIGGER;
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ return;
+ }
+ } else {
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ return;
+ }
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ }
+}
+
+static int snd_wavefront_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ snd_wavefront_midi_t *midi;
+ snd_wavefront_mpu_id mpu;
+
+ if (snd_BUG_ON(!substream || !substream->rmidi))
+ return -ENXIO;
+ if (snd_BUG_ON(!substream->rmidi->private_data))
+ return -ENXIO;
+
+ mpu = *((snd_wavefront_mpu_id *) substream->rmidi->private_data);
+
+ if ((midi = get_wavefront_midi (substream)) == NULL)
+ return -EIO;
+
+ spin_lock_irqsave (&midi->open, flags);
+ midi->mode[mpu] |= MPU401_MODE_INPUT;
+ midi->substream_input[mpu] = substream;
+ spin_unlock_irqrestore (&midi->open, flags);
+
+ return 0;
+}
+
+static int snd_wavefront_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ snd_wavefront_midi_t *midi;
+ snd_wavefront_mpu_id mpu;
+
+ if (snd_BUG_ON(!substream || !substream->rmidi))
+ return -ENXIO;
+ if (snd_BUG_ON(!substream->rmidi->private_data))
+ return -ENXIO;
+
+ mpu = *((snd_wavefront_mpu_id *) substream->rmidi->private_data);
+
+ if ((midi = get_wavefront_midi (substream)) == NULL)
+ return -EIO;
+
+ spin_lock_irqsave (&midi->open, flags);
+ midi->mode[mpu] |= MPU401_MODE_OUTPUT;
+ midi->substream_output[mpu] = substream;
+ spin_unlock_irqrestore (&midi->open, flags);
+
+ return 0;
+}
+
+static int snd_wavefront_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ snd_wavefront_midi_t *midi;
+ snd_wavefront_mpu_id mpu;
+
+ if (snd_BUG_ON(!substream || !substream->rmidi))
+ return -ENXIO;
+ if (snd_BUG_ON(!substream->rmidi->private_data))
+ return -ENXIO;
+
+ mpu = *((snd_wavefront_mpu_id *) substream->rmidi->private_data);
+
+ if ((midi = get_wavefront_midi (substream)) == NULL)
+ return -EIO;
+
+ spin_lock_irqsave (&midi->open, flags);
+ midi->mode[mpu] &= ~MPU401_MODE_INPUT;
+ spin_unlock_irqrestore (&midi->open, flags);
+
+ return 0;
+}
+
+static int snd_wavefront_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ unsigned long flags;
+ snd_wavefront_midi_t *midi;
+ snd_wavefront_mpu_id mpu;
+
+ if (snd_BUG_ON(!substream || !substream->rmidi))
+ return -ENXIO;
+ if (snd_BUG_ON(!substream->rmidi->private_data))
+ return -ENXIO;
+
+ mpu = *((snd_wavefront_mpu_id *) substream->rmidi->private_data);
+
+ if ((midi = get_wavefront_midi (substream)) == NULL)
+ return -EIO;
+
+ spin_lock_irqsave (&midi->open, flags);
+ midi->mode[mpu] &= ~MPU401_MODE_OUTPUT;
+ spin_unlock_irqrestore (&midi->open, flags);
+ return 0;
+}
+
+static void snd_wavefront_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ snd_wavefront_midi_t *midi;
+ snd_wavefront_mpu_id mpu;
+
+ if (substream == NULL || substream->rmidi == NULL)
+ return;
+
+ if (substream->rmidi->private_data == NULL)
+ return;
+
+ mpu = *((snd_wavefront_mpu_id *) substream->rmidi->private_data);
+
+ if ((midi = get_wavefront_midi (substream)) == NULL) {
+ return;
+ }
+
+ spin_lock_irqsave (&midi->virtual, flags);
+ if (up) {
+ midi->mode[mpu] |= MPU401_MODE_INPUT_TRIGGER;
+ } else {
+ midi->mode[mpu] &= ~MPU401_MODE_INPUT_TRIGGER;
+ }
+ spin_unlock_irqrestore (&midi->virtual, flags);
+}
+
+static void snd_wavefront_midi_output_timer(unsigned long data)
+{
+ snd_wavefront_card_t *card = (snd_wavefront_card_t *)data;
+ snd_wavefront_midi_t *midi = &card->wavefront.midi;
+ unsigned long flags;
+
+ spin_lock_irqsave (&midi->virtual, flags);
+ midi->timer.expires = 1 + jiffies;
+ add_timer(&midi->timer);
+ spin_unlock_irqrestore (&midi->virtual, flags);
+ snd_wavefront_midi_output_write(card);
+}
+
+static void snd_wavefront_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ snd_wavefront_midi_t *midi;
+ snd_wavefront_mpu_id mpu;
+
+ if (substream == NULL || substream->rmidi == NULL)
+ return;
+
+ if (substream->rmidi->private_data == NULL)
+ return;
+
+ mpu = *((snd_wavefront_mpu_id *) substream->rmidi->private_data);
+
+ if ((midi = get_wavefront_midi (substream)) == NULL) {
+ return;
+ }
+
+ spin_lock_irqsave (&midi->virtual, flags);
+ if (up) {
+ if ((midi->mode[mpu] & MPU401_MODE_OUTPUT_TRIGGER) == 0) {
+ if (!midi->istimer) {
+ init_timer(&midi->timer);
+ midi->timer.function = snd_wavefront_midi_output_timer;
+ midi->timer.data = (unsigned long) substream->rmidi->card->private_data;
+ midi->timer.expires = 1 + jiffies;
+ add_timer(&midi->timer);
+ }
+ midi->istimer++;
+ midi->mode[mpu] |= MPU401_MODE_OUTPUT_TRIGGER;
+ }
+ } else {
+ midi->mode[mpu] &= ~MPU401_MODE_OUTPUT_TRIGGER;
+ }
+ spin_unlock_irqrestore (&midi->virtual, flags);
+
+ if (up)
+ snd_wavefront_midi_output_write((snd_wavefront_card_t *)substream->rmidi->card->private_data);
+}
+
+void
+snd_wavefront_midi_interrupt (snd_wavefront_card_t *card)
+
+{
+ unsigned long flags;
+ snd_wavefront_midi_t *midi;
+ static struct snd_rawmidi_substream *substream = NULL;
+ static int mpu = external_mpu;
+ int max = 128;
+ unsigned char byte;
+
+ midi = &card->wavefront.midi;
+
+ if (!input_avail (midi)) { /* not for us */
+ snd_wavefront_midi_output_write(card);
+ return;
+ }
+
+ spin_lock_irqsave (&midi->virtual, flags);
+ while (--max) {
+
+ if (input_avail (midi)) {
+ byte = read_data (midi);
+
+ if (midi->isvirtual) {
+ if (byte == WF_EXTERNAL_SWITCH) {
+ substream = midi->substream_input[external_mpu];
+ mpu = external_mpu;
+ } else if (byte == WF_INTERNAL_SWITCH) {
+ substream = midi->substream_output[internal_mpu];
+ mpu = internal_mpu;
+ } /* else just leave it as it is */
+ } else {
+ substream = midi->substream_input[internal_mpu];
+ mpu = internal_mpu;
+ }
+
+ if (substream == NULL) {
+ continue;
+ }
+
+ if (midi->mode[mpu] & MPU401_MODE_INPUT_TRIGGER) {
+ snd_rawmidi_receive(substream, &byte, 1);
+ }
+ } else {
+ break;
+ }
+ }
+ spin_unlock_irqrestore (&midi->virtual, flags);
+
+ snd_wavefront_midi_output_write(card);
+}
+
+void
+snd_wavefront_midi_enable_virtual (snd_wavefront_card_t *card)
+
+{
+ unsigned long flags;
+
+ spin_lock_irqsave (&card->wavefront.midi.virtual, flags);
+ card->wavefront.midi.isvirtual = 1;
+ card->wavefront.midi.output_mpu = internal_mpu;
+ card->wavefront.midi.input_mpu = internal_mpu;
+ spin_unlock_irqrestore (&card->wavefront.midi.virtual, flags);
+}
+
+void
+snd_wavefront_midi_disable_virtual (snd_wavefront_card_t *card)
+
+{
+ unsigned long flags;
+
+ spin_lock_irqsave (&card->wavefront.midi.virtual, flags);
+ // snd_wavefront_midi_input_close (card->ics2115_external_rmidi);
+ // snd_wavefront_midi_output_close (card->ics2115_external_rmidi);
+ card->wavefront.midi.isvirtual = 0;
+ spin_unlock_irqrestore (&card->wavefront.midi.virtual, flags);
+}
+
+int __devinit
+snd_wavefront_midi_start (snd_wavefront_card_t *card)
+
+{
+ int ok, i;
+ unsigned char rbuf[4], wbuf[4];
+ snd_wavefront_t *dev;
+ snd_wavefront_midi_t *midi;
+
+ dev = &card->wavefront;
+ midi = &dev->midi;
+
+ /* The ICS2115 MPU-401 interface doesn't do anything
+ until its set into UART mode.
+ */
+
+ /* XXX fix me - no hard timing loops allowed! */
+
+ for (i = 0; i < 30000 && !output_ready (midi); i++);
+
+ if (!output_ready (midi)) {
+ snd_printk ("MIDI interface not ready for command\n");
+ return -1;
+ }
+
+ /* Any interrupts received from now on
+ are owned by the MIDI side of things.
+ */
+
+ dev->interrupts_are_midi = 1;
+
+ outb (UART_MODE_ON, midi->mpu_command_port);
+
+ for (ok = 0, i = 50000; i > 0 && !ok; i--) {
+ if (input_avail (midi)) {
+ if (read_data (midi) == MPU_ACK) {
+ ok = 1;
+ break;
+ }
+ }
+ }
+
+ if (!ok) {
+ snd_printk ("cannot set UART mode for MIDI interface");
+ dev->interrupts_are_midi = 0;
+ return -1;
+ }
+
+ /* Route external MIDI to WaveFront synth (by default) */
+
+ if (snd_wavefront_cmd (dev, WFC_MISYNTH_ON, rbuf, wbuf)) {
+ snd_printk ("can't enable MIDI-IN-2-synth routing.\n");
+ /* XXX error ? */
+ }
+
+ /* Turn on Virtual MIDI, but first *always* turn it off,
+ since otherwise consectutive reloads of the driver will
+ never cause the hardware to generate the initial "internal" or
+ "external" source bytes in the MIDI data stream. This
+ is pretty important, since the internal hardware generally will
+ be used to generate none or very little MIDI output, and
+ thus the only source of MIDI data is actually external. Without
+ the switch bytes, the driver will think it all comes from
+ the internal interface. Duh.
+ */
+
+ if (snd_wavefront_cmd (dev, WFC_VMIDI_OFF, rbuf, wbuf)) {
+ snd_printk ("virtual MIDI mode not disabled\n");
+ return 0; /* We're OK, but missing the external MIDI dev */
+ }
+
+ snd_wavefront_midi_enable_virtual (card);
+
+ if (snd_wavefront_cmd (dev, WFC_VMIDI_ON, rbuf, wbuf)) {
+ snd_printk ("cannot enable virtual MIDI mode.\n");
+ snd_wavefront_midi_disable_virtual (card);
+ }
+ return 0;
+}
+
+struct snd_rawmidi_ops snd_wavefront_midi_output =
+{
+ .open = snd_wavefront_midi_output_open,
+ .close = snd_wavefront_midi_output_close,
+ .trigger = snd_wavefront_midi_output_trigger,
+};
+
+struct snd_rawmidi_ops snd_wavefront_midi_input =
+{
+ .open = snd_wavefront_midi_input_open,
+ .close = snd_wavefront_midi_input_close,
+ .trigger = snd_wavefront_midi_input_trigger,
+};
+
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
new file mode 100644
index 0000000..4c41082
--- /dev/null
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -0,0 +1,2198 @@
+/* Copyright (C) by Paul Barton-Davis 1998-1999
+ *
+ * Some portions of this file are taken from work that is
+ * copyright (C) by Hannu Savolainen 1993-1996
+ *
+ * This program is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+
+/*
+ * An ALSA lowlevel driver for Turtle Beach ICS2115 wavetable synth
+ * (Maui, Tropez, Tropez Plus)
+ *
+ * This driver supports the onboard wavetable synthesizer (an ICS2115),
+ * including patch, sample and program loading and unloading, conversion
+ * of GUS patches during loading, and full user-level access to all
+ * WaveFront commands. It tries to provide semi-intelligent patch and
+ * sample management as well.
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/firmware.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/snd_wavefront.h>
+#include <sound/initval.h>
+
+static int wf_raw = 0; /* we normally check for "raw state" to firmware
+ loading. if non-zero, then during driver loading, the
+ state of the board is ignored, and we reset the
+ board and load the firmware anyway.
+ */
+
+static int fx_raw = 1; /* if this is zero, we'll leave the FX processor in
+ whatever state it is when the driver is loaded.
+ The default is to download the microprogram and
+ associated coefficients to set it up for "default"
+ operation, whatever that means.
+ */
+
+static int debug_default = 0; /* you can set this to control debugging
+ during driver loading. it takes any combination
+ of the WF_DEBUG_* flags defined in
+ wavefront.h
+ */
+
+/* XXX this needs to be made firmware and hardware version dependent */
+
+#define DEFAULT_OSPATH "wavefront.os"
+static char *ospath = DEFAULT_OSPATH; /* the firmware file name */
+
+static int wait_usecs = 150; /* This magic number seems to give pretty optimal
+ throughput based on my limited experimentation.
+ If you want to play around with it and find a better
+ value, be my guest. Remember, the idea is to
+ get a number that causes us to just busy wait
+ for as many WaveFront commands as possible, without
+ coming up with a number so large that we hog the
+ whole CPU.
+
+ Specifically, with this number, out of about 134,000
+ status waits, only about 250 result in a sleep.
+ */
+
+static int sleep_interval = 100; /* HZ/sleep_interval seconds per sleep */
+static int sleep_tries = 50; /* number of times we'll try to sleep */
+
+static int reset_time = 2; /* hundreths of a second we wait after a HW
+ reset for the expected interrupt.
+ */
+
+static int ramcheck_time = 20; /* time in seconds to wait while ROM code
+ checks on-board RAM.
+ */
+
+static int osrun_time = 10; /* time in seconds we wait for the OS to
+ start running.
+ */
+module_param(wf_raw, int, 0444);
+MODULE_PARM_DESC(wf_raw, "if non-zero, assume that we need to boot the OS");
+module_param(fx_raw, int, 0444);
+MODULE_PARM_DESC(fx_raw, "if non-zero, assume that the FX process needs help");
+module_param(debug_default, int, 0444);
+MODULE_PARM_DESC(debug_default, "debug parameters for card initialization");
+module_param(wait_usecs, int, 0444);
+MODULE_PARM_DESC(wait_usecs, "how long to wait without sleeping, usecs");
+module_param(sleep_interval, int, 0444);
+MODULE_PARM_DESC(sleep_interval, "how long to sleep when waiting for reply");
+module_param(sleep_tries, int, 0444);
+MODULE_PARM_DESC(sleep_tries, "how many times to try sleeping during a wait");
+module_param(ospath, charp, 0444);
+MODULE_PARM_DESC(ospath, "pathname to processed ICS2115 OS firmware");
+module_param(reset_time, int, 0444);
+MODULE_PARM_DESC(reset_time, "how long to wait for a reset to take effect");
+module_param(ramcheck_time, int, 0444);
+MODULE_PARM_DESC(ramcheck_time, "how many seconds to wait for the RAM test");
+module_param(osrun_time, int, 0444);
+MODULE_PARM_DESC(osrun_time, "how many seconds to wait for the ICS2115 OS");
+
+/* if WF_DEBUG not defined, no run-time debugging messages will
+ be available via the debug flag setting. Given the current
+ beta state of the driver, this will remain set until a future
+ version.
+*/
+
+#define WF_DEBUG 1
+
+#ifdef WF_DEBUG
+
+#define DPRINT(cond, ...) \
+ if ((dev->debug & (cond)) == (cond)) { \
+ snd_printk (__VA_ARGS__); \
+ }
+#else
+#define DPRINT(cond, args...)
+#endif /* WF_DEBUG */
+
+#define LOGNAME "WaveFront: "
+
+/* bitmasks for WaveFront status port value */
+
+#define STAT_RINTR_ENABLED 0x01
+#define STAT_CAN_READ 0x02
+#define STAT_INTR_READ 0x04
+#define STAT_WINTR_ENABLED 0x10
+#define STAT_CAN_WRITE 0x20
+#define STAT_INTR_WRITE 0x40
+
+static int wavefront_delete_sample (snd_wavefront_t *, int sampnum);
+static int wavefront_find_free_sample (snd_wavefront_t *);
+
+struct wavefront_command {
+ int cmd;
+ char *action;
+ unsigned int read_cnt;
+ unsigned int write_cnt;
+ int need_ack;
+};
+
+static struct {
+ int errno;
+ const char *errstr;
+} wavefront_errors[] = {
+ { 0x01, "Bad sample number" },
+ { 0x02, "Out of sample memory" },
+ { 0x03, "Bad patch number" },
+ { 0x04, "Error in number of voices" },
+ { 0x06, "Sample load already in progress" },
+ { 0x0B, "No sample load request pending" },
+ { 0x0E, "Bad MIDI channel number" },
+ { 0x10, "Download Record Error" },
+ { 0x80, "Success" },
+ { 0x0 }
+};
+
+#define NEEDS_ACK 1
+
+static struct wavefront_command wavefront_commands[] = {
+ { WFC_SET_SYNTHVOL, "set synthesizer volume", 0, 1, NEEDS_ACK },
+ { WFC_GET_SYNTHVOL, "get synthesizer volume", 1, 0, 0},
+ { WFC_SET_NVOICES, "set number of voices", 0, 1, NEEDS_ACK },
+ { WFC_GET_NVOICES, "get number of voices", 1, 0, 0 },
+ { WFC_SET_TUNING, "set synthesizer tuning", 0, 2, NEEDS_ACK },
+ { WFC_GET_TUNING, "get synthesizer tuning", 2, 0, 0 },
+ { WFC_DISABLE_CHANNEL, "disable synth channel", 0, 1, NEEDS_ACK },
+ { WFC_ENABLE_CHANNEL, "enable synth channel", 0, 1, NEEDS_ACK },
+ { WFC_GET_CHANNEL_STATUS, "get synth channel status", 3, 0, 0 },
+ { WFC_MISYNTH_OFF, "disable midi-in to synth", 0, 0, NEEDS_ACK },
+ { WFC_MISYNTH_ON, "enable midi-in to synth", 0, 0, NEEDS_ACK },
+ { WFC_VMIDI_ON, "enable virtual midi mode", 0, 0, NEEDS_ACK },
+ { WFC_VMIDI_OFF, "disable virtual midi mode", 0, 0, NEEDS_ACK },
+ { WFC_MIDI_STATUS, "report midi status", 1, 0, 0 },
+ { WFC_FIRMWARE_VERSION, "report firmware version", 2, 0, 0 },
+ { WFC_HARDWARE_VERSION, "report hardware version", 2, 0, 0 },
+ { WFC_GET_NSAMPLES, "report number of samples", 2, 0, 0 },
+ { WFC_INSTOUT_LEVELS, "report instantaneous output levels", 7, 0, 0 },
+ { WFC_PEAKOUT_LEVELS, "report peak output levels", 7, 0, 0 },
+ { WFC_DOWNLOAD_SAMPLE, "download sample",
+ 0, WF_SAMPLE_BYTES, NEEDS_ACK },
+ { WFC_DOWNLOAD_BLOCK, "download block", 0, 0, NEEDS_ACK},
+ { WFC_DOWNLOAD_SAMPLE_HEADER, "download sample header",
+ 0, WF_SAMPLE_HDR_BYTES, NEEDS_ACK },
+ { WFC_UPLOAD_SAMPLE_HEADER, "upload sample header", 13, 2, 0 },
+
+ /* This command requires a variable number of bytes to be written.
+ There is a hack in snd_wavefront_cmd() to support this. The actual
+ count is passed in as the read buffer ptr, cast appropriately.
+ Ugh.
+ */
+
+ { WFC_DOWNLOAD_MULTISAMPLE, "download multisample", 0, 0, NEEDS_ACK },
+
+ /* This one is a hack as well. We just read the first byte of the
+ response, don't fetch an ACK, and leave the rest to the
+ calling function. Ugly, ugly, ugly.
+ */
+
+ { WFC_UPLOAD_MULTISAMPLE, "upload multisample", 2, 1, 0 },
+ { WFC_DOWNLOAD_SAMPLE_ALIAS, "download sample alias",
+ 0, WF_ALIAS_BYTES, NEEDS_ACK },
+ { WFC_UPLOAD_SAMPLE_ALIAS, "upload sample alias", WF_ALIAS_BYTES, 2, 0},
+ { WFC_DELETE_SAMPLE, "delete sample", 0, 2, NEEDS_ACK },
+ { WFC_IDENTIFY_SAMPLE_TYPE, "identify sample type", 5, 2, 0 },
+ { WFC_UPLOAD_SAMPLE_PARAMS, "upload sample parameters" },
+ { WFC_REPORT_FREE_MEMORY, "report free memory", 4, 0, 0 },
+ { WFC_DOWNLOAD_PATCH, "download patch", 0, 134, NEEDS_ACK },
+ { WFC_UPLOAD_PATCH, "upload patch", 132, 2, 0 },
+ { WFC_DOWNLOAD_PROGRAM, "download program", 0, 33, NEEDS_ACK },
+ { WFC_UPLOAD_PROGRAM, "upload program", 32, 1, 0 },
+ { WFC_DOWNLOAD_EDRUM_PROGRAM, "download enhanced drum program", 0, 9,
+ NEEDS_ACK},
+ { WFC_UPLOAD_EDRUM_PROGRAM, "upload enhanced drum program", 8, 1, 0},
+ { WFC_SET_EDRUM_CHANNEL, "set enhanced drum program channel",
+ 0, 1, NEEDS_ACK },
+ { WFC_DISABLE_DRUM_PROGRAM, "disable drum program", 0, 1, NEEDS_ACK },
+ { WFC_REPORT_CHANNEL_PROGRAMS, "report channel program numbers",
+ 32, 0, 0 },
+ { WFC_NOOP, "the no-op command", 0, 0, NEEDS_ACK },
+ { 0x00 }
+};
+
+static const char *
+wavefront_errorstr (int errnum)
+
+{
+ int i;
+
+ for (i = 0; wavefront_errors[i].errstr; i++) {
+ if (wavefront_errors[i].errno == errnum) {
+ return wavefront_errors[i].errstr;
+ }
+ }
+
+ return "Unknown WaveFront error";
+}
+
+static struct wavefront_command *
+wavefront_get_command (int cmd)
+
+{
+ int i;
+
+ for (i = 0; wavefront_commands[i].cmd != 0; i++) {
+ if (cmd == wavefront_commands[i].cmd) {
+ return &wavefront_commands[i];
+ }
+ }
+
+ return NULL;
+}
+
+static inline int
+wavefront_status (snd_wavefront_t *dev)
+
+{
+ return inb (dev->status_port);
+}
+
+static int
+wavefront_sleep (int limit)
+
+{
+ schedule_timeout_interruptible(limit);
+
+ return signal_pending(current);
+}
+
+static int
+wavefront_wait (snd_wavefront_t *dev, int mask)
+
+{
+ int i;
+
+ /* Spin for a short period of time, because >99% of all
+ requests to the WaveFront can be serviced inline like this.
+ */
+
+ for (i = 0; i < wait_usecs; i += 5) {
+ if (wavefront_status (dev) & mask) {
+ return 1;
+ }
+ udelay(5);
+ }
+
+ for (i = 0; i < sleep_tries; i++) {
+
+ if (wavefront_status (dev) & mask) {
+ return 1;
+ }
+
+ if (wavefront_sleep (HZ/sleep_interval)) {
+ return (0);
+ }
+ }
+
+ return (0);
+}
+
+static int
+wavefront_read (snd_wavefront_t *dev)
+
+{
+ if (wavefront_wait (dev, STAT_CAN_READ))
+ return inb (dev->data_port);
+
+ DPRINT (WF_DEBUG_DATA, "read timeout.\n");
+
+ return -1;
+}
+
+static int
+wavefront_write (snd_wavefront_t *dev, unsigned char data)
+
+{
+ if (wavefront_wait (dev, STAT_CAN_WRITE)) {
+ outb (data, dev->data_port);
+ return 0;
+ }
+
+ DPRINT (WF_DEBUG_DATA, "write timeout.\n");
+
+ return -1;
+}
+
+int
+snd_wavefront_cmd (snd_wavefront_t *dev,
+ int cmd, unsigned char *rbuf, unsigned char *wbuf)
+
+{
+ int ack;
+ unsigned int i;
+ int c;
+ struct wavefront_command *wfcmd;
+
+ if ((wfcmd = wavefront_get_command (cmd)) == NULL) {
+ snd_printk ("command 0x%x not supported.\n",
+ cmd);
+ return 1;
+ }
+
+ /* Hack to handle the one variable-size write command. See
+ wavefront_send_multisample() for the other half of this
+ gross and ugly strategy.
+ */
+
+ if (cmd == WFC_DOWNLOAD_MULTISAMPLE) {
+ wfcmd->write_cnt = (unsigned long) rbuf;
+ rbuf = NULL;
+ }
+
+ DPRINT (WF_DEBUG_CMD, "0x%x [%s] (%d,%d,%d)\n",
+ cmd, wfcmd->action, wfcmd->read_cnt,
+ wfcmd->write_cnt, wfcmd->need_ack);
+
+ if (wavefront_write (dev, cmd)) {
+ DPRINT ((WF_DEBUG_IO|WF_DEBUG_CMD), "cannot request "
+ "0x%x [%s].\n",
+ cmd, wfcmd->action);
+ return 1;
+ }
+
+ if (wfcmd->write_cnt > 0) {
+ DPRINT (WF_DEBUG_DATA, "writing %d bytes "
+ "for 0x%x\n",
+ wfcmd->write_cnt, cmd);
+
+ for (i = 0; i < wfcmd->write_cnt; i++) {
+ if (wavefront_write (dev, wbuf[i])) {
+ DPRINT (WF_DEBUG_IO, "bad write for byte "
+ "%d of 0x%x [%s].\n",
+ i, cmd, wfcmd->action);
+ return 1;
+ }
+
+ DPRINT (WF_DEBUG_DATA, "write[%d] = 0x%x\n",
+ i, wbuf[i]);
+ }
+ }
+
+ if (wfcmd->read_cnt > 0) {
+ DPRINT (WF_DEBUG_DATA, "reading %d ints "
+ "for 0x%x\n",
+ wfcmd->read_cnt, cmd);
+
+ for (i = 0; i < wfcmd->read_cnt; i++) {
+
+ if ((c = wavefront_read (dev)) == -1) {
+ DPRINT (WF_DEBUG_IO, "bad read for byte "
+ "%d of 0x%x [%s].\n",
+ i, cmd, wfcmd->action);
+ return 1;
+ }
+
+ /* Now handle errors. Lots of special cases here */
+
+ if (c == 0xff) {
+ if ((c = wavefront_read (dev)) == -1) {
+ DPRINT (WF_DEBUG_IO, "bad read for "
+ "error byte at "
+ "read byte %d "
+ "of 0x%x [%s].\n",
+ i, cmd,
+ wfcmd->action);
+ return 1;
+ }
+
+ /* Can you believe this madness ? */
+
+ if (c == 1 &&
+ wfcmd->cmd == WFC_IDENTIFY_SAMPLE_TYPE) {
+ rbuf[0] = WF_ST_EMPTY;
+ return (0);
+
+ } else if (c == 3 &&
+ wfcmd->cmd == WFC_UPLOAD_PATCH) {
+
+ return 3;
+
+ } else if (c == 1 &&
+ wfcmd->cmd == WFC_UPLOAD_PROGRAM) {
+
+ return 1;
+
+ } else {
+
+ DPRINT (WF_DEBUG_IO, "error %d (%s) "
+ "during "
+ "read for byte "
+ "%d of 0x%x "
+ "[%s].\n",
+ c,
+ wavefront_errorstr (c),
+ i, cmd,
+ wfcmd->action);
+ return 1;
+
+ }
+
+ } else {
+ rbuf[i] = c;
+ }
+
+ DPRINT (WF_DEBUG_DATA, "read[%d] = 0x%x\n",i, rbuf[i]);
+ }
+ }
+
+ if ((wfcmd->read_cnt == 0 && wfcmd->write_cnt == 0) || wfcmd->need_ack) {
+
+ DPRINT (WF_DEBUG_CMD, "reading ACK for 0x%x\n", cmd);
+
+ /* Some commands need an ACK, but return zero instead
+ of the standard value.
+ */
+
+ if ((ack = wavefront_read (dev)) == 0) {
+ ack = WF_ACK;
+ }
+
+ if (ack != WF_ACK) {
+ if (ack == -1) {
+ DPRINT (WF_DEBUG_IO, "cannot read ack for "
+ "0x%x [%s].\n",
+ cmd, wfcmd->action);
+ return 1;
+
+ } else {
+ int err = -1; /* something unknown */
+
+ if (ack == 0xff) { /* explicit error */
+
+ if ((err = wavefront_read (dev)) == -1) {
+ DPRINT (WF_DEBUG_DATA,
+ "cannot read err "
+ "for 0x%x [%s].\n",
+ cmd, wfcmd->action);
+ }
+ }
+
+ DPRINT (WF_DEBUG_IO, "0x%x [%s] "
+ "failed (0x%x, 0x%x, %s)\n",
+ cmd, wfcmd->action, ack, err,
+ wavefront_errorstr (err));
+
+ return -err;
+ }
+ }
+
+ DPRINT (WF_DEBUG_DATA, "ack received "
+ "for 0x%x [%s]\n",
+ cmd, wfcmd->action);
+ } else {
+
+ DPRINT (WF_DEBUG_CMD, "0x%x [%s] does not need "
+ "ACK (%d,%d,%d)\n",
+ cmd, wfcmd->action, wfcmd->read_cnt,
+ wfcmd->write_cnt, wfcmd->need_ack);
+ }
+
+ return 0;
+
+}
+
+/***********************************************************************
+WaveFront data munging
+
+Things here are weird. All data written to the board cannot
+have its most significant bit set. Any data item with values
+potentially > 0x7F (127) must be split across multiple bytes.
+
+Sometimes, we need to munge numeric values that are represented on
+the x86 side as 8-32 bit values. Sometimes, we need to munge data
+that is represented on the x86 side as an array of bytes. The most
+efficient approach to handling both cases seems to be to use 2
+different functions for munging and 2 for de-munging. This avoids
+weird casting and worrying about bit-level offsets.
+
+**********************************************************************/
+
+static unsigned char *
+munge_int32 (unsigned int src,
+ unsigned char *dst,
+ unsigned int dst_size)
+{
+ unsigned int i;
+
+ for (i = 0; i < dst_size; i++) {
+ *dst = src & 0x7F; /* Mask high bit of LSB */
+ src = src >> 7; /* Rotate Right 7 bits */
+ /* Note: we leave the upper bits in place */
+
+ dst++;
+ };
+ return dst;
+};
+
+static int
+demunge_int32 (unsigned char* src, int src_size)
+
+{
+ int i;
+ int outval = 0;
+
+ for (i = src_size - 1; i >= 0; i--) {
+ outval=(outval<<7)+src[i];
+ }
+
+ return outval;
+};
+
+static
+unsigned char *
+munge_buf (unsigned char *src, unsigned char *dst, unsigned int dst_size)
+
+{
+ unsigned int i;
+ unsigned int last = dst_size / 2;
+
+ for (i = 0; i < last; i++) {
+ *dst++ = src[i] & 0x7f;
+ *dst++ = src[i] >> 7;
+ }
+ return dst;
+}
+
+static
+unsigned char *
+demunge_buf (unsigned char *src, unsigned char *dst, unsigned int src_bytes)
+
+{
+ int i;
+ unsigned char *end = src + src_bytes;
+
+ end = src + src_bytes;
+
+ /* NOTE: src and dst *CAN* point to the same address */
+
+ for (i = 0; src != end; i++) {
+ dst[i] = *src++;
+ dst[i] |= (*src++)<<7;
+ }
+
+ return dst;
+}
+
+/***********************************************************************
+WaveFront: sample, patch and program management.
+***********************************************************************/
+
+static int
+wavefront_delete_sample (snd_wavefront_t *dev, int sample_num)
+
+{
+ unsigned char wbuf[2];
+ int x;
+
+ wbuf[0] = sample_num & 0x7f;
+ wbuf[1] = sample_num >> 7;
+
+ if ((x = snd_wavefront_cmd (dev, WFC_DELETE_SAMPLE, NULL, wbuf)) == 0) {
+ dev->sample_status[sample_num] = WF_ST_EMPTY;
+ }
+
+ return x;
+}
+
+static int
+wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom)
+
+{
+ int i;
+ unsigned char rbuf[32], wbuf[32];
+ unsigned int sc_real, sc_alias, sc_multi;
+
+ /* check sample status */
+
+ if (snd_wavefront_cmd (dev, WFC_GET_NSAMPLES, rbuf, wbuf)) {
+ snd_printk ("cannot request sample count.\n");
+ return -1;
+ }
+
+ sc_real = sc_alias = sc_multi = dev->samples_used = 0;
+
+ for (i = 0; i < WF_MAX_SAMPLE; i++) {
+
+ wbuf[0] = i & 0x7f;
+ wbuf[1] = i >> 7;
+
+ if (snd_wavefront_cmd (dev, WFC_IDENTIFY_SAMPLE_TYPE, rbuf, wbuf)) {
+ snd_printk("cannot identify sample "
+ "type of slot %d\n", i);
+ dev->sample_status[i] = WF_ST_EMPTY;
+ continue;
+ }
+
+ dev->sample_status[i] = (WF_SLOT_FILLED|rbuf[0]);
+
+ if (assume_rom) {
+ dev->sample_status[i] |= WF_SLOT_ROM;
+ }
+
+ switch (rbuf[0] & WF_ST_MASK) {
+ case WF_ST_SAMPLE:
+ sc_real++;
+ break;
+ case WF_ST_MULTISAMPLE:
+ sc_multi++;
+ break;
+ case WF_ST_ALIAS:
+ sc_alias++;
+ break;
+ case WF_ST_EMPTY:
+ break;
+
+ default:
+ snd_printk ("unknown sample type for "
+ "slot %d (0x%x)\n",
+ i, rbuf[0]);
+ }
+
+ if (rbuf[0] != WF_ST_EMPTY) {
+ dev->samples_used++;
+ }
+ }
+
+ snd_printk ("%d samples used (%d real, %d aliases, %d multi), "
+ "%d empty\n", dev->samples_used, sc_real, sc_alias, sc_multi,
+ WF_MAX_SAMPLE - dev->samples_used);
+
+
+ return (0);
+
+}
+
+static int
+wavefront_get_patch_status (snd_wavefront_t *dev)
+
+{
+ unsigned char patchbuf[WF_PATCH_BYTES];
+ unsigned char patchnum[2];
+ wavefront_patch *p;
+ int i, x, cnt, cnt2;
+
+ for (i = 0; i < WF_MAX_PATCH; i++) {
+ patchnum[0] = i & 0x7f;
+ patchnum[1] = i >> 7;
+
+ if ((x = snd_wavefront_cmd (dev, WFC_UPLOAD_PATCH, patchbuf,
+ patchnum)) == 0) {
+
+ dev->patch_status[i] |= WF_SLOT_FILLED;
+ p = (wavefront_patch *) patchbuf;
+ dev->sample_status
+ [p->sample_number|(p->sample_msb<<7)] |=
+ WF_SLOT_USED;
+
+ } else if (x == 3) { /* Bad patch number */
+ dev->patch_status[i] = 0;
+ } else {
+ snd_printk ("upload patch "
+ "error 0x%x\n", x);
+ dev->patch_status[i] = 0;
+ return 1;
+ }
+ }
+
+ /* program status has already filled in slot_used bits */
+
+ for (i = 0, cnt = 0, cnt2 = 0; i < WF_MAX_PATCH; i++) {
+ if (dev->patch_status[i] & WF_SLOT_FILLED) {
+ cnt++;
+ }
+ if (dev->patch_status[i] & WF_SLOT_USED) {
+ cnt2++;
+ }
+
+ }
+ snd_printk ("%d patch slots filled, %d in use\n", cnt, cnt2);
+
+ return (0);
+}
+
+static int
+wavefront_get_program_status (snd_wavefront_t *dev)
+
+{
+ unsigned char progbuf[WF_PROGRAM_BYTES];
+ wavefront_program prog;
+ unsigned char prognum;
+ int i, x, l, cnt;
+
+ for (i = 0; i < WF_MAX_PROGRAM; i++) {
+ prognum = i;
+
+ if ((x = snd_wavefront_cmd (dev, WFC_UPLOAD_PROGRAM, progbuf,
+ &prognum)) == 0) {
+
+ dev->prog_status[i] |= WF_SLOT_USED;
+
+ demunge_buf (progbuf, (unsigned char *) &prog,
+ WF_PROGRAM_BYTES);
+
+ for (l = 0; l < WF_NUM_LAYERS; l++) {
+ if (prog.layer[l].mute) {
+ dev->patch_status
+ [prog.layer[l].patch_number] |=
+ WF_SLOT_USED;
+ }
+ }
+ } else if (x == 1) { /* Bad program number */
+ dev->prog_status[i] = 0;
+ } else {
+ snd_printk ("upload program "
+ "error 0x%x\n", x);
+ dev->prog_status[i] = 0;
+ }
+ }
+
+ for (i = 0, cnt = 0; i < WF_MAX_PROGRAM; i++) {
+ if (dev->prog_status[i]) {
+ cnt++;
+ }
+ }
+
+ snd_printk ("%d programs slots in use\n", cnt);
+
+ return (0);
+}
+
+static int
+wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header)
+
+{
+ unsigned char buf[WF_PATCH_BYTES+2];
+ unsigned char *bptr;
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n",
+ header->number);
+
+ dev->patch_status[header->number] |= WF_SLOT_FILLED;
+
+ bptr = buf;
+ bptr = munge_int32 (header->number, buf, 2);
+ munge_buf ((unsigned char *)&header->hdr.p, bptr, WF_PATCH_BYTES);
+
+ if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PATCH, NULL, buf)) {
+ snd_printk ("download patch failed\n");
+ return -(EIO);
+ }
+
+ return (0);
+}
+
+static int
+wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header)
+
+{
+ unsigned char buf[WF_PROGRAM_BYTES+1];
+ int i;
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n",
+ header->number);
+
+ dev->prog_status[header->number] = WF_SLOT_USED;
+
+ /* XXX need to zero existing SLOT_USED bit for program_status[i]
+ where `i' is the program that's being (potentially) overwritten.
+ */
+
+ for (i = 0; i < WF_NUM_LAYERS; i++) {
+ if (header->hdr.pr.layer[i].mute) {
+ dev->patch_status[header->hdr.pr.layer[i].patch_number] |=
+ WF_SLOT_USED;
+
+ /* XXX need to mark SLOT_USED for sample used by
+ patch_number, but this means we have to load it. Ick.
+ */
+ }
+ }
+
+ buf[0] = header->number;
+ munge_buf ((unsigned char *)&header->hdr.pr, &buf[1], WF_PROGRAM_BYTES);
+
+ if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PROGRAM, NULL, buf)) {
+ snd_printk ("download patch failed\n");
+ return -(EIO);
+ }
+
+ return (0);
+}
+
+static int
+wavefront_freemem (snd_wavefront_t *dev)
+
+{
+ char rbuf[8];
+
+ if (snd_wavefront_cmd (dev, WFC_REPORT_FREE_MEMORY, rbuf, NULL)) {
+ snd_printk ("can't get memory stats.\n");
+ return -1;
+ } else {
+ return demunge_int32 (rbuf, 4);
+ }
+}
+
+static int
+wavefront_send_sample (snd_wavefront_t *dev,
+ wavefront_patch_info *header,
+ u16 __user *dataptr,
+ int data_is_unsigned)
+
+{
+ /* samples are downloaded via a 16-bit wide i/o port
+ (you could think of it as 2 adjacent 8-bit wide ports
+ but its less efficient that way). therefore, all
+ the blocksizes and so forth listed in the documentation,
+ and used conventionally to refer to sample sizes,
+ which are given in 8-bit units (bytes), need to be
+ divided by 2.
+ */
+
+ u16 sample_short = 0;
+ u32 length;
+ u16 __user *data_end = NULL;
+ unsigned int i;
+ const unsigned int max_blksize = 4096/2;
+ unsigned int written;
+ unsigned int blocksize;
+ int dma_ack;
+ int blocknum;
+ unsigned char sample_hdr[WF_SAMPLE_HDR_BYTES];
+ unsigned char *shptr;
+ int skip = 0;
+ int initial_skip = 0;
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "sample %sdownload for slot %d, "
+ "type %d, %d bytes from 0x%lx\n",
+ header->size ? "" : "header ",
+ header->number, header->subkey,
+ header->size,
+ (unsigned long) header->dataptr);
+
+ if (header->number == WAVEFRONT_FIND_FREE_SAMPLE_SLOT) {
+ int x;
+
+ if ((x = wavefront_find_free_sample (dev)) < 0) {
+ return -ENOMEM;
+ }
+ snd_printk ("unspecified sample => %d\n", x);
+ header->number = x;
+ }
+
+ if (header->size) {
+
+ /* XXX it's a debatable point whether or not RDONLY semantics
+ on the ROM samples should cover just the sample data or
+ the sample header. For now, it only covers the sample data,
+ so anyone is free at all times to rewrite sample headers.
+
+ My reason for this is that we have the sample headers
+ available in the WFB file for General MIDI, and so these
+ can always be reset if needed. The sample data, however,
+ cannot be recovered without a complete reset and firmware
+ reload of the ICS2115, which is a very expensive operation.
+
+ So, doing things this way allows us to honor the notion of
+ "RESETSAMPLES" reasonably cheaply. Note however, that this
+ is done purely at user level: there is no WFB parser in
+ this driver, and so a complete reset (back to General MIDI,
+ or theoretically some other configuration) is the
+ responsibility of the user level library.
+
+ To try to do this in the kernel would be a little
+ crazy: we'd need 158K of kernel space just to hold
+ a copy of the patch/program/sample header data.
+ */
+
+ if (dev->rom_samples_rdonly) {
+ if (dev->sample_status[header->number] & WF_SLOT_ROM) {
+ snd_printk ("sample slot %d "
+ "write protected\n",
+ header->number);
+ return -EACCES;
+ }
+ }
+
+ wavefront_delete_sample (dev, header->number);
+ }
+
+ if (header->size) {
+ dev->freemem = wavefront_freemem (dev);
+
+ if (dev->freemem < (int)header->size) {
+ snd_printk ("insufficient memory to "
+ "load %d byte sample.\n",
+ header->size);
+ return -ENOMEM;
+ }
+
+ }
+
+ skip = WF_GET_CHANNEL(&header->hdr.s);
+
+ if (skip > 0 && header->hdr.s.SampleResolution != LINEAR_16BIT) {
+ snd_printk ("channel selection only "
+ "possible on 16-bit samples");
+ return -(EINVAL);
+ }
+
+ switch (skip) {
+ case 0:
+ initial_skip = 0;
+ skip = 1;
+ break;
+ case 1:
+ initial_skip = 0;
+ skip = 2;
+ break;
+ case 2:
+ initial_skip = 1;
+ skip = 2;
+ break;
+ case 3:
+ initial_skip = 2;
+ skip = 3;
+ break;
+ case 4:
+ initial_skip = 3;
+ skip = 4;
+ break;
+ case 5:
+ initial_skip = 4;
+ skip = 5;
+ break;
+ case 6:
+ initial_skip = 5;
+ skip = 6;
+ break;
+ }
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "channel selection: %d => "
+ "initial skip = %d, skip = %d\n",
+ WF_GET_CHANNEL (&header->hdr.s),
+ initial_skip, skip);
+
+ /* Be safe, and zero the "Unused" bits ... */
+
+ WF_SET_CHANNEL(&header->hdr.s, 0);
+
+ /* adjust size for 16 bit samples by dividing by two. We always
+ send 16 bits per write, even for 8 bit samples, so the length
+ is always half the size of the sample data in bytes.
+ */
+
+ length = header->size / 2;
+
+ /* the data we're sent has not been munged, and in fact, the
+ header we have to send isn't just a munged copy either.
+ so, build the sample header right here.
+ */
+
+ shptr = &sample_hdr[0];
+
+ shptr = munge_int32 (header->number, shptr, 2);
+
+ if (header->size) {
+ shptr = munge_int32 (length, shptr, 4);
+ }
+
+ /* Yes, a 4 byte result doesn't contain all of the offset bits,
+ but the offset only uses 24 bits.
+ */
+
+ shptr = munge_int32 (*((u32 *) &header->hdr.s.sampleStartOffset),
+ shptr, 4);
+ shptr = munge_int32 (*((u32 *) &header->hdr.s.loopStartOffset),
+ shptr, 4);
+ shptr = munge_int32 (*((u32 *) &header->hdr.s.loopEndOffset),
+ shptr, 4);
+ shptr = munge_int32 (*((u32 *) &header->hdr.s.sampleEndOffset),
+ shptr, 4);
+
+ /* This one is truly weird. What kind of weirdo decided that in
+ a system dominated by 16 and 32 bit integers, they would use
+ a just 12 bits ?
+ */
+
+ shptr = munge_int32 (header->hdr.s.FrequencyBias, shptr, 3);
+
+ /* Why is this nybblified, when the MSB is *always* zero ?
+ Anyway, we can't take address of bitfield, so make a
+ good-faith guess at where it starts.
+ */
+
+ shptr = munge_int32 (*(&header->hdr.s.FrequencyBias+1),
+ shptr, 2);
+
+ if (snd_wavefront_cmd (dev,
+ header->size ?
+ WFC_DOWNLOAD_SAMPLE : WFC_DOWNLOAD_SAMPLE_HEADER,
+ NULL, sample_hdr)) {
+ snd_printk ("sample %sdownload refused.\n",
+ header->size ? "" : "header ");
+ return -(EIO);
+ }
+
+ if (header->size == 0) {
+ goto sent; /* Sorry. Just had to have one somewhere */
+ }
+
+ data_end = dataptr + length;
+
+ /* Do any initial skip over an unused channel's data */
+
+ dataptr += initial_skip;
+
+ for (written = 0, blocknum = 0;
+ written < length; written += max_blksize, blocknum++) {
+
+ if ((length - written) > max_blksize) {
+ blocksize = max_blksize;
+ } else {
+ /* round to nearest 16-byte value */
+ blocksize = ALIGN(length - written, 8);
+ }
+
+ if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_BLOCK, NULL, NULL)) {
+ snd_printk ("download block "
+ "request refused.\n");
+ return -(EIO);
+ }
+
+ for (i = 0; i < blocksize; i++) {
+
+ if (dataptr < data_end) {
+
+ __get_user (sample_short, dataptr);
+ dataptr += skip;
+
+ if (data_is_unsigned) { /* GUS ? */
+
+ if (WF_SAMPLE_IS_8BIT(&header->hdr.s)) {
+
+ /* 8 bit sample
+ resolution, sign
+ extend both bytes.
+ */
+
+ ((unsigned char*)
+ &sample_short)[0] += 0x7f;
+ ((unsigned char*)
+ &sample_short)[1] += 0x7f;
+
+ } else {
+
+ /* 16 bit sample
+ resolution, sign
+ extend the MSB.
+ */
+
+ sample_short += 0x7fff;
+ }
+ }
+
+ } else {
+
+ /* In padding section of final block:
+
+ Don't fetch unsupplied data from
+ user space, just continue with
+ whatever the final value was.
+ */
+ }
+
+ if (i < blocksize - 1) {
+ outw (sample_short, dev->block_port);
+ } else {
+ outw (sample_short, dev->last_block_port);
+ }
+ }
+
+ /* Get "DMA page acknowledge", even though its really
+ nothing to do with DMA at all.
+ */
+
+ if ((dma_ack = wavefront_read (dev)) != WF_DMA_ACK) {
+ if (dma_ack == -1) {
+ snd_printk ("upload sample "
+ "DMA ack timeout\n");
+ return -(EIO);
+ } else {
+ snd_printk ("upload sample "
+ "DMA ack error 0x%x\n",
+ dma_ack);
+ return -(EIO);
+ }
+ }
+ }
+
+ dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_SAMPLE);
+
+ /* Note, label is here because sending the sample header shouldn't
+ alter the sample_status info at all.
+ */
+
+ sent:
+ return (0);
+}
+
+static int
+wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header)
+
+{
+ unsigned char alias_hdr[WF_ALIAS_BYTES];
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "download alias, %d is "
+ "alias for %d\n",
+ header->number,
+ header->hdr.a.OriginalSample);
+
+ munge_int32 (header->number, &alias_hdr[0], 2);
+ munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2);
+ munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset),
+ &alias_hdr[4], 4);
+ munge_int32 (*((unsigned int *)&header->hdr.a.loopStartOffset),
+ &alias_hdr[8], 4);
+ munge_int32 (*((unsigned int *)&header->hdr.a.loopEndOffset),
+ &alias_hdr[12], 4);
+ munge_int32 (*((unsigned int *)&header->hdr.a.sampleEndOffset),
+ &alias_hdr[16], 4);
+ munge_int32 (header->hdr.a.FrequencyBias, &alias_hdr[20], 3);
+ munge_int32 (*(&header->hdr.a.FrequencyBias+1), &alias_hdr[23], 2);
+
+ if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_SAMPLE_ALIAS, NULL, alias_hdr)) {
+ snd_printk ("download alias failed.\n");
+ return -(EIO);
+ }
+
+ dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_ALIAS);
+
+ return (0);
+}
+
+static int
+wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header)
+{
+ int i;
+ int num_samples;
+ unsigned char *msample_hdr;
+
+ msample_hdr = kmalloc(sizeof(WF_MSAMPLE_BYTES), GFP_KERNEL);
+ if (! msample_hdr)
+ return -ENOMEM;
+
+ munge_int32 (header->number, &msample_hdr[0], 2);
+
+ /* You'll recall at this point that the "number of samples" value
+ in a wavefront_multisample struct is actually the log2 of the
+ real number of samples.
+ */
+
+ num_samples = (1<<(header->hdr.ms.NumberOfSamples&7));
+ msample_hdr[2] = (unsigned char) header->hdr.ms.NumberOfSamples;
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "multi %d with %d=%d samples\n",
+ header->number,
+ header->hdr.ms.NumberOfSamples,
+ num_samples);
+
+ for (i = 0; i < num_samples; i++) {
+ DPRINT(WF_DEBUG_LOAD_PATCH|WF_DEBUG_DATA, "sample[%d] = %d\n",
+ i, header->hdr.ms.SampleNumber[i]);
+ munge_int32 (header->hdr.ms.SampleNumber[i],
+ &msample_hdr[3+(i*2)], 2);
+ }
+
+ /* Need a hack here to pass in the number of bytes
+ to be written to the synth. This is ugly, and perhaps
+ one day, I'll fix it.
+ */
+
+ if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_MULTISAMPLE,
+ (unsigned char *) (long) ((num_samples*2)+3),
+ msample_hdr)) {
+ snd_printk ("download of multisample failed.\n");
+ kfree(msample_hdr);
+ return -(EIO);
+ }
+
+ dev->sample_status[header->number] = (WF_SLOT_FILLED|WF_ST_MULTISAMPLE);
+
+ kfree(msample_hdr);
+ return (0);
+}
+
+static int
+wavefront_fetch_multisample (snd_wavefront_t *dev,
+ wavefront_patch_info *header)
+{
+ int i;
+ unsigned char log_ns[1];
+ unsigned char number[2];
+ int num_samples;
+
+ munge_int32 (header->number, number, 2);
+
+ if (snd_wavefront_cmd (dev, WFC_UPLOAD_MULTISAMPLE, log_ns, number)) {
+ snd_printk ("upload multisample failed.\n");
+ return -(EIO);
+ }
+
+ DPRINT (WF_DEBUG_DATA, "msample %d has %d samples\n",
+ header->number, log_ns[0]);
+
+ header->hdr.ms.NumberOfSamples = log_ns[0];
+
+ /* get the number of samples ... */
+
+ num_samples = (1 << log_ns[0]);
+
+ for (i = 0; i < num_samples; i++) {
+ char d[2];
+ int val;
+
+ if ((val = wavefront_read (dev)) == -1) {
+ snd_printk ("upload multisample failed "
+ "during sample loop.\n");
+ return -(EIO);
+ }
+ d[0] = val;
+
+ if ((val = wavefront_read (dev)) == -1) {
+ snd_printk ("upload multisample failed "
+ "during sample loop.\n");
+ return -(EIO);
+ }
+ d[1] = val;
+
+ header->hdr.ms.SampleNumber[i] =
+ demunge_int32 ((unsigned char *) d, 2);
+
+ DPRINT (WF_DEBUG_DATA, "msample sample[%d] = %d\n",
+ i, header->hdr.ms.SampleNumber[i]);
+ }
+
+ return (0);
+}
+
+
+static int
+wavefront_send_drum (snd_wavefront_t *dev, wavefront_patch_info *header)
+
+{
+ unsigned char drumbuf[WF_DRUM_BYTES];
+ wavefront_drum *drum = &header->hdr.d;
+ int i;
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "downloading edrum for MIDI "
+ "note %d, patch = %d\n",
+ header->number, drum->PatchNumber);
+
+ drumbuf[0] = header->number & 0x7f;
+
+ for (i = 0; i < 4; i++) {
+ munge_int32 (((unsigned char *)drum)[i], &drumbuf[1+(i*2)], 2);
+ }
+
+ if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_EDRUM_PROGRAM, NULL, drumbuf)) {
+ snd_printk ("download drum failed.\n");
+ return -(EIO);
+ }
+
+ return (0);
+}
+
+static int
+wavefront_find_free_sample (snd_wavefront_t *dev)
+
+{
+ int i;
+
+ for (i = 0; i < WF_MAX_SAMPLE; i++) {
+ if (!(dev->sample_status[i] & WF_SLOT_FILLED)) {
+ return i;
+ }
+ }
+ snd_printk ("no free sample slots!\n");
+ return -1;
+}
+
+#if 0
+static int
+wavefront_find_free_patch (snd_wavefront_t *dev)
+
+{
+ int i;
+
+ for (i = 0; i < WF_MAX_PATCH; i++) {
+ if (!(dev->patch_status[i] & WF_SLOT_FILLED)) {
+ return i;
+ }
+ }
+ snd_printk ("no free patch slots!\n");
+ return -1;
+}
+#endif
+
+static int
+wavefront_load_patch (snd_wavefront_t *dev, const char __user *addr)
+{
+ wavefront_patch_info *header;
+ int err;
+
+ header = kmalloc(sizeof(*header), GFP_KERNEL);
+ if (! header)
+ return -ENOMEM;
+
+ if (copy_from_user (header, addr, sizeof(wavefront_patch_info) -
+ sizeof(wavefront_any))) {
+ snd_printk ("bad address for load patch.\n");
+ err = -EFAULT;
+ goto __error;
+ }
+
+ DPRINT (WF_DEBUG_LOAD_PATCH, "download "
+ "Sample type: %d "
+ "Sample number: %d "
+ "Sample size: %d\n",
+ header->subkey,
+ header->number,
+ header->size);
+
+ switch (header->subkey) {
+ case WF_ST_SAMPLE: /* sample or sample_header, based on patch->size */
+
+ if (copy_from_user (&header->hdr.s, header->hdrptr,
+ sizeof (wavefront_sample))) {
+ err = -EFAULT;
+ break;
+ }
+
+ err = wavefront_send_sample (dev, header, header->dataptr, 0);
+ break;
+
+ case WF_ST_MULTISAMPLE:
+
+ if (copy_from_user (&header->hdr.s, header->hdrptr,
+ sizeof (wavefront_multisample))) {
+ err = -EFAULT;
+ break;
+ }
+
+ err = wavefront_send_multisample (dev, header);
+ break;
+
+ case WF_ST_ALIAS:
+
+ if (copy_from_user (&header->hdr.a, header->hdrptr,
+ sizeof (wavefront_alias))) {
+ err = -EFAULT;
+ break;
+ }
+
+ err = wavefront_send_alias (dev, header);
+ break;
+
+ case WF_ST_DRUM:
+ if (copy_from_user (&header->hdr.d, header->hdrptr,
+ sizeof (wavefront_drum))) {
+ err = -EFAULT;
+ break;
+ }
+
+ err = wavefront_send_drum (dev, header);
+ break;
+
+ case WF_ST_PATCH:
+ if (copy_from_user (&header->hdr.p, header->hdrptr,
+ sizeof (wavefront_patch))) {
+ err = -EFAULT;
+ break;
+ }
+
+ err = wavefront_send_patch (dev, header);
+ break;
+
+ case WF_ST_PROGRAM:
+ if (copy_from_user (&header->hdr.pr, header->hdrptr,
+ sizeof (wavefront_program))) {
+ err = -EFAULT;
+ break;
+ }
+
+ err = wavefront_send_program (dev, header);
+ break;
+
+ default:
+ snd_printk ("unknown patch type %d.\n",
+ header->subkey);
+ err = -EINVAL;
+ break;
+ }
+
+ __error:
+ kfree(header);
+ return err;
+}
+
+/***********************************************************************
+WaveFront: hardware-dependent interface
+***********************************************************************/
+
+static void
+process_sample_hdr (u8 *buf)
+
+{
+ wavefront_sample s;
+ u8 *ptr;
+
+ ptr = buf;
+
+ /* The board doesn't send us an exact copy of a "wavefront_sample"
+ in response to an Upload Sample Header command. Instead, we
+ have to convert the data format back into our data structure,
+ just as in the Download Sample command, where we have to do
+ something very similar in the reverse direction.
+ */
+
+ *((u32 *) &s.sampleStartOffset) = demunge_int32 (ptr, 4); ptr += 4;
+ *((u32 *) &s.loopStartOffset) = demunge_int32 (ptr, 4); ptr += 4;
+ *((u32 *) &s.loopEndOffset) = demunge_int32 (ptr, 4); ptr += 4;
+ *((u32 *) &s.sampleEndOffset) = demunge_int32 (ptr, 4); ptr += 4;
+ *((u32 *) &s.FrequencyBias) = demunge_int32 (ptr, 3); ptr += 3;
+
+ s.SampleResolution = *ptr & 0x3;
+ s.Loop = *ptr & 0x8;
+ s.Bidirectional = *ptr & 0x10;
+ s.Reverse = *ptr & 0x40;
+
+ /* Now copy it back to where it came from */
+
+ memcpy (buf, (unsigned char *) &s, sizeof (wavefront_sample));
+}
+
+static int
+wavefront_synth_control (snd_wavefront_card_t *acard,
+ wavefront_control *wc)
+
+{
+ snd_wavefront_t *dev = &acard->wavefront;
+ unsigned char patchnumbuf[2];
+ int i;
+
+ DPRINT (WF_DEBUG_CMD, "synth control with "
+ "cmd 0x%x\n", wc->cmd);
+
+ /* Pre-handling of or for various commands */
+
+ switch (wc->cmd) {
+
+ case WFC_DISABLE_INTERRUPTS:
+ snd_printk ("interrupts disabled.\n");
+ outb (0x80|0x20, dev->control_port);
+ dev->interrupts_are_midi = 1;
+ return 0;
+
+ case WFC_ENABLE_INTERRUPTS:
+ snd_printk ("interrupts enabled.\n");
+ outb (0x80|0x40|0x20, dev->control_port);
+ dev->interrupts_are_midi = 1;
+ return 0;
+
+ case WFC_INTERRUPT_STATUS:
+ wc->rbuf[0] = dev->interrupts_are_midi;
+ return 0;
+
+ case WFC_ROMSAMPLES_RDONLY:
+ dev->rom_samples_rdonly = wc->wbuf[0];
+ wc->status = 0;
+ return 0;
+
+ case WFC_IDENTIFY_SLOT_TYPE:
+ i = wc->wbuf[0] | (wc->wbuf[1] << 7);
+ if (i <0 || i >= WF_MAX_SAMPLE) {
+ snd_printk ("invalid slot ID %d\n",
+ i);
+ wc->status = EINVAL;
+ return -EINVAL;
+ }
+ wc->rbuf[0] = dev->sample_status[i];
+ wc->status = 0;
+ return 0;
+
+ case WFC_DEBUG_DRIVER:
+ dev->debug = wc->wbuf[0];
+ snd_printk ("debug = 0x%x\n", dev->debug);
+ return 0;
+
+ case WFC_UPLOAD_PATCH:
+ munge_int32 (*((u32 *) wc->wbuf), patchnumbuf, 2);
+ memcpy (wc->wbuf, patchnumbuf, 2);
+ break;
+
+ case WFC_UPLOAD_MULTISAMPLE:
+ /* multisamples have to be handled differently, and
+ cannot be dealt with properly by snd_wavefront_cmd() alone.
+ */
+ wc->status = wavefront_fetch_multisample
+ (dev, (wavefront_patch_info *) wc->rbuf);
+ return 0;
+
+ case WFC_UPLOAD_SAMPLE_ALIAS:
+ snd_printk ("support for sample alias upload "
+ "being considered.\n");
+ wc->status = EINVAL;
+ return -EINVAL;
+ }
+
+ wc->status = snd_wavefront_cmd (dev, wc->cmd, wc->rbuf, wc->wbuf);
+
+ /* Post-handling of certain commands.
+
+ In particular, if the command was an upload, demunge the data
+ so that the user-level doesn't have to think about it.
+ */
+
+ if (wc->status == 0) {
+ switch (wc->cmd) {
+ /* intercept any freemem requests so that we know
+ we are always current with the user-level view
+ of things.
+ */
+
+ case WFC_REPORT_FREE_MEMORY:
+ dev->freemem = demunge_int32 (wc->rbuf, 4);
+ break;
+
+ case WFC_UPLOAD_PATCH:
+ demunge_buf (wc->rbuf, wc->rbuf, WF_PATCH_BYTES);
+ break;
+
+ case WFC_UPLOAD_PROGRAM:
+ demunge_buf (wc->rbuf, wc->rbuf, WF_PROGRAM_BYTES);
+ break;
+
+ case WFC_UPLOAD_EDRUM_PROGRAM:
+ demunge_buf (wc->rbuf, wc->rbuf, WF_DRUM_BYTES - 1);
+ break;
+
+ case WFC_UPLOAD_SAMPLE_HEADER:
+ process_sample_hdr (wc->rbuf);
+ break;
+
+ case WFC_UPLOAD_SAMPLE_ALIAS:
+ snd_printk ("support for "
+ "sample aliases still "
+ "being considered.\n");
+ break;
+
+ case WFC_VMIDI_OFF:
+ snd_wavefront_midi_disable_virtual (acard);
+ break;
+
+ case WFC_VMIDI_ON:
+ snd_wavefront_midi_enable_virtual (acard);
+ break;
+ }
+ }
+
+ return 0;
+}
+
+int
+snd_wavefront_synth_open (struct snd_hwdep *hw, struct file *file)
+
+{
+ if (!try_module_get(hw->card->module))
+ return -EFAULT;
+ file->private_data = hw;
+ return 0;
+}
+
+int
+snd_wavefront_synth_release (struct snd_hwdep *hw, struct file *file)
+
+{
+ module_put(hw->card->module);
+ return 0;
+}
+
+int
+snd_wavefront_synth_ioctl (struct snd_hwdep *hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+
+{
+ struct snd_card *card;
+ snd_wavefront_t *dev;
+ snd_wavefront_card_t *acard;
+ wavefront_control *wc;
+ void __user *argp = (void __user *)arg;
+ int err;
+
+ card = (struct snd_card *) hw->card;
+
+ if (snd_BUG_ON(!card))
+ return -ENODEV;
+ if (snd_BUG_ON(!card->private_data))
+ return -ENODEV;
+
+ acard = card->private_data;
+ dev = &acard->wavefront;
+
+ switch (cmd) {
+ case WFCTL_LOAD_SPP:
+ if (wavefront_load_patch (dev, argp) != 0) {
+ return -EIO;
+ }
+ break;
+
+ case WFCTL_WFCMD:
+ wc = kmalloc(sizeof(*wc), GFP_KERNEL);
+ if (! wc)
+ return -ENOMEM;
+ if (copy_from_user (wc, argp, sizeof (*wc)))
+ err = -EFAULT;
+ else if (wavefront_synth_control (acard, wc) < 0)
+ err = -EIO;
+ else if (copy_to_user (argp, wc, sizeof (*wc)))
+ err = -EFAULT;
+ else
+ err = 0;
+ kfree(wc);
+ return err;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+/***********************************************************************/
+/* WaveFront: interface for card-level wavefront module */
+/***********************************************************************/
+
+void
+snd_wavefront_internal_interrupt (snd_wavefront_card_t *card)
+{
+ snd_wavefront_t *dev = &card->wavefront;
+
+ /*
+ Some comments on interrupts. I attempted a version of this
+ driver that used interrupts throughout the code instead of
+ doing busy and/or sleep-waiting. Alas, it appears that once
+ the Motorola firmware is downloaded, the card *never*
+ generates an RX interrupt. These are successfully generated
+ during firmware loading, and after that wavefront_status()
+ reports that an interrupt is pending on the card from time
+ to time, but it never seems to be delivered to this
+ driver. Note also that wavefront_status() continues to
+ report that RX interrupts are enabled, suggesting that I
+ didn't goof up and disable them by mistake.
+
+ Thus, I stepped back to a prior version of
+ wavefront_wait(), the only place where this really
+ matters. Its sad, but I've looked through the code to check
+ on things, and I really feel certain that the Motorola
+ firmware prevents RX-ready interrupts.
+ */
+
+ if ((wavefront_status(dev) & (STAT_INTR_READ|STAT_INTR_WRITE)) == 0) {
+ return;
+ }
+
+ spin_lock(&dev->irq_lock);
+ dev->irq_ok = 1;
+ dev->irq_cnt++;
+ spin_unlock(&dev->irq_lock);
+ wake_up(&dev->interrupt_sleeper);
+}
+
+/* STATUS REGISTER
+
+0 Host Rx Interrupt Enable (1=Enabled)
+1 Host Rx Register Full (1=Full)
+2 Host Rx Interrupt Pending (1=Interrupt)
+3 Unused
+4 Host Tx Interrupt (1=Enabled)
+5 Host Tx Register empty (1=Empty)
+6 Host Tx Interrupt Pending (1=Interrupt)
+7 Unused
+*/
+
+static int __devinit
+snd_wavefront_interrupt_bits (int irq)
+
+{
+ int bits;
+
+ switch (irq) {
+ case 9:
+ bits = 0x00;
+ break;
+ case 5:
+ bits = 0x08;
+ break;
+ case 12:
+ bits = 0x10;
+ break;
+ case 15:
+ bits = 0x18;
+ break;
+
+ default:
+ snd_printk ("invalid IRQ %d\n", irq);
+ bits = -1;
+ }
+
+ return bits;
+}
+
+static void __devinit
+wavefront_should_cause_interrupt (snd_wavefront_t *dev,
+ int val, int port, unsigned long timeout)
+
+{
+ wait_queue_t wait;
+
+ init_waitqueue_entry(&wait, current);
+ spin_lock_irq(&dev->irq_lock);
+ add_wait_queue(&dev->interrupt_sleeper, &wait);
+ dev->irq_ok = 0;
+ outb (val,port);
+ spin_unlock_irq(&dev->irq_lock);
+ while (!dev->irq_ok && time_before(jiffies, timeout)) {
+ schedule_timeout_uninterruptible(1);
+ barrier();
+ }
+}
+
+static int __devinit
+wavefront_reset_to_cleanliness (snd_wavefront_t *dev)
+
+{
+ int bits;
+ int hwv[2];
+
+ /* IRQ already checked */
+
+ bits = snd_wavefront_interrupt_bits (dev->irq);
+
+ /* try reset of port */
+
+ outb (0x0, dev->control_port);
+
+ /* At this point, the board is in reset, and the H/W initialization
+ register is accessed at the same address as the data port.
+
+ Bit 7 - Enable IRQ Driver
+ 0 - Tri-state the Wave-Board drivers for the PC Bus IRQs
+ 1 - Enable IRQ selected by bits 5:3 to be driven onto the PC Bus.
+
+ Bit 6 - MIDI Interface Select
+
+ 0 - Use the MIDI Input from the 26-pin WaveBlaster
+ compatible header as the serial MIDI source
+ 1 - Use the MIDI Input from the 9-pin D connector as the
+ serial MIDI source.
+
+ Bits 5:3 - IRQ Selection
+ 0 0 0 - IRQ 2/9
+ 0 0 1 - IRQ 5
+ 0 1 0 - IRQ 12
+ 0 1 1 - IRQ 15
+ 1 0 0 - Reserved
+ 1 0 1 - Reserved
+ 1 1 0 - Reserved
+ 1 1 1 - Reserved
+
+ Bits 2:1 - Reserved
+ Bit 0 - Disable Boot ROM
+ 0 - memory accesses to 03FC30-03FFFFH utilize the internal Boot ROM
+ 1 - memory accesses to 03FC30-03FFFFH are directed to external
+ storage.
+
+ */
+
+ /* configure hardware: IRQ, enable interrupts,
+ plus external 9-pin MIDI interface selected
+ */
+
+ outb (0x80 | 0x40 | bits, dev->data_port);
+
+ /* CONTROL REGISTER
+
+ 0 Host Rx Interrupt Enable (1=Enabled) 0x1
+ 1 Unused 0x2
+ 2 Unused 0x4
+ 3 Unused 0x8
+ 4 Host Tx Interrupt Enable 0x10
+ 5 Mute (0=Mute; 1=Play) 0x20
+ 6 Master Interrupt Enable (1=Enabled) 0x40
+ 7 Master Reset (0=Reset; 1=Run) 0x80
+
+ Take us out of reset, mute output, master + TX + RX interrupts on.
+
+ We'll get an interrupt presumably to tell us that the TX
+ register is clear.
+ */
+
+ wavefront_should_cause_interrupt(dev, 0x80|0x40|0x10|0x1,
+ dev->control_port,
+ (reset_time*HZ)/100);
+
+ /* Note: data port is now the data port, not the h/w initialization
+ port.
+ */
+
+ if (!dev->irq_ok) {
+ snd_printk ("intr not received after h/w un-reset.\n");
+ goto gone_bad;
+ }
+
+ /* Note: data port is now the data port, not the h/w initialization
+ port.
+
+ At this point, only "HW VERSION" or "DOWNLOAD OS" commands
+ will work. So, issue one of them, and wait for TX
+ interrupt. This can take a *long* time after a cold boot,
+ while the ISC ROM does its RAM test. The SDK says up to 4
+ seconds - with 12MB of RAM on a Tropez+, it takes a lot
+ longer than that (~16secs). Note that the card understands
+ the difference between a warm and a cold boot, so
+ subsequent ISC2115 reboots (say, caused by module
+ reloading) will get through this much faster.
+
+ XXX Interesting question: why is no RX interrupt received first ?
+ */
+
+ wavefront_should_cause_interrupt(dev, WFC_HARDWARE_VERSION,
+ dev->data_port, ramcheck_time*HZ);
+
+ if (!dev->irq_ok) {
+ snd_printk ("post-RAM-check interrupt not received.\n");
+ goto gone_bad;
+ }
+
+ if (!wavefront_wait (dev, STAT_CAN_READ)) {
+ snd_printk ("no response to HW version cmd.\n");
+ goto gone_bad;
+ }
+
+ if ((hwv[0] = wavefront_read (dev)) == -1) {
+ snd_printk ("board not responding correctly.\n");
+ goto gone_bad;
+ }
+
+ if (hwv[0] == 0xFF) { /* NAK */
+
+ /* Board's RAM test failed. Try to read error code,
+ and tell us about it either way.
+ */
+
+ if ((hwv[0] = wavefront_read (dev)) == -1) {
+ snd_printk ("on-board RAM test failed "
+ "(bad error code).\n");
+ } else {
+ snd_printk ("on-board RAM test failed "
+ "(error code: 0x%x).\n",
+ hwv[0]);
+ }
+ goto gone_bad;
+ }
+
+ /* We're OK, just get the next byte of the HW version response */
+
+ if ((hwv[1] = wavefront_read (dev)) == -1) {
+ snd_printk ("incorrect h/w response.\n");
+ goto gone_bad;
+ }
+
+ snd_printk ("hardware version %d.%d\n",
+ hwv[0], hwv[1]);
+
+ return 0;
+
+
+ gone_bad:
+ return (1);
+}
+
+static int __devinit
+wavefront_download_firmware (snd_wavefront_t *dev, char *path)
+
+{
+ const unsigned char *buf;
+ int len, err;
+ int section_cnt_downloaded = 0;
+ const struct firmware *firmware;
+
+ err = request_firmware(&firmware, path, dev->card->dev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "firmware (%s) download failed!!!\n", path);
+ return 1;
+ }
+
+ len = 0;
+ buf = firmware->data;
+ for (;;) {
+ int section_length = *(signed char *)buf;
+ if (section_length == 0)
+ break;
+ if (section_length < 0 || section_length > WF_SECTION_MAX) {
+ snd_printk(KERN_ERR
+ "invalid firmware section length %d\n",
+ section_length);
+ goto failure;
+ }
+ buf++;
+ len++;
+
+ if (firmware->size < len + section_length) {
+ snd_printk(KERN_ERR "firmware section read error.\n");
+ goto failure;
+ }
+
+ /* Send command */
+ if (wavefront_write(dev, WFC_DOWNLOAD_OS))
+ goto failure;
+
+ for (; section_length; section_length--) {
+ if (wavefront_write(dev, *buf))
+ goto failure;
+ buf++;
+ len++;
+ }
+
+ /* get ACK */
+ if (!wavefront_wait(dev, STAT_CAN_READ)) {
+ snd_printk(KERN_ERR "time out for firmware ACK.\n");
+ goto failure;
+ }
+ err = inb(dev->data_port);
+ if (err != WF_ACK) {
+ snd_printk(KERN_ERR
+ "download of section #%d not "
+ "acknowledged, ack = 0x%x\n",
+ section_cnt_downloaded + 1, err);
+ goto failure;
+ }
+
+ section_cnt_downloaded++;
+ }
+
+ release_firmware(firmware);
+ return 0;
+
+ failure:
+ release_firmware(firmware);
+ snd_printk(KERN_ERR "firmware download failed!!!\n");
+ return 1;
+}
+
+
+static int __devinit
+wavefront_do_reset (snd_wavefront_t *dev)
+
+{
+ char voices[1];
+
+ if (wavefront_reset_to_cleanliness (dev)) {
+ snd_printk ("hw reset failed.\n");
+ goto gone_bad;
+ }
+
+ if (dev->israw) {
+ if (wavefront_download_firmware (dev, ospath)) {
+ goto gone_bad;
+ }
+
+ dev->israw = 0;
+
+ /* Wait for the OS to get running. The protocol for
+ this is non-obvious, and was determined by
+ using port-IO tracing in DOSemu and some
+ experimentation here.
+
+ Rather than using timed waits, use interrupts creatively.
+ */
+
+ wavefront_should_cause_interrupt (dev, WFC_NOOP,
+ dev->data_port,
+ (osrun_time*HZ));
+
+ if (!dev->irq_ok) {
+ snd_printk ("no post-OS interrupt.\n");
+ goto gone_bad;
+ }
+
+ /* Now, do it again ! */
+
+ wavefront_should_cause_interrupt (dev, WFC_NOOP,
+ dev->data_port, (10*HZ));
+
+ if (!dev->irq_ok) {
+ snd_printk ("no post-OS interrupt(2).\n");
+ goto gone_bad;
+ }
+
+ /* OK, no (RX/TX) interrupts any more, but leave mute
+ in effect.
+ */
+
+ outb (0x80|0x40, dev->control_port);
+ }
+
+ /* SETUPSND.EXE asks for sample memory config here, but since i
+ have no idea how to interpret the result, we'll forget
+ about it.
+ */
+
+ if ((dev->freemem = wavefront_freemem (dev)) < 0) {
+ goto gone_bad;
+ }
+
+ snd_printk ("available DRAM %dk\n", dev->freemem / 1024);
+
+ if (wavefront_write (dev, 0xf0) ||
+ wavefront_write (dev, 1) ||
+ (wavefront_read (dev) < 0)) {
+ dev->debug = 0;
+ snd_printk ("MPU emulation mode not set.\n");
+ goto gone_bad;
+ }
+
+ voices[0] = 32;
+
+ if (snd_wavefront_cmd (dev, WFC_SET_NVOICES, NULL, voices)) {
+ snd_printk ("cannot set number of voices to 32.\n");
+ goto gone_bad;
+ }
+
+
+ return 0;
+
+ gone_bad:
+ /* reset that sucker so that it doesn't bother us. */
+
+ outb (0x0, dev->control_port);
+ dev->interrupts_are_midi = 0;
+ return 1;
+}
+
+int __devinit
+snd_wavefront_start (snd_wavefront_t *dev)
+
+{
+ int samples_are_from_rom;
+
+ /* IMPORTANT: assumes that snd_wavefront_detect() and/or
+ wavefront_reset_to_cleanliness() has already been called
+ */
+
+ if (dev->israw) {
+ samples_are_from_rom = 1;
+ } else {
+ /* XXX is this always true ? */
+ samples_are_from_rom = 0;
+ }
+
+ if (dev->israw || fx_raw) {
+ if (wavefront_do_reset (dev)) {
+ return -1;
+ }
+ }
+ /* Check for FX device, present only on Tropez+ */
+
+ dev->has_fx = (snd_wavefront_fx_detect (dev) == 0);
+
+ if (dev->has_fx && fx_raw) {
+ snd_wavefront_fx_start (dev);
+ }
+
+ wavefront_get_sample_status (dev, samples_are_from_rom);
+ wavefront_get_program_status (dev);
+ wavefront_get_patch_status (dev);
+
+ /* Start normal operation: unreset, master interrupt enabled, no mute
+ */
+
+ outb (0x80|0x40|0x20, dev->control_port);
+
+ return (0);
+}
+
+int __devinit
+snd_wavefront_detect (snd_wavefront_card_t *card)
+
+{
+ unsigned char rbuf[4], wbuf[4];
+ snd_wavefront_t *dev = &card->wavefront;
+
+ /* returns zero if a WaveFront card is successfully detected.
+ negative otherwise.
+ */
+
+ dev->israw = 0;
+ dev->has_fx = 0;
+ dev->debug = debug_default;
+ dev->interrupts_are_midi = 0;
+ dev->irq_cnt = 0;
+ dev->rom_samples_rdonly = 1;
+
+ if (snd_wavefront_cmd (dev, WFC_FIRMWARE_VERSION, rbuf, wbuf) == 0) {
+
+ dev->fw_version[0] = rbuf[0];
+ dev->fw_version[1] = rbuf[1];
+
+ snd_printk ("firmware %d.%d already loaded.\n",
+ rbuf[0], rbuf[1]);
+
+ /* check that a command actually works */
+
+ if (snd_wavefront_cmd (dev, WFC_HARDWARE_VERSION,
+ rbuf, wbuf) == 0) {
+ dev->hw_version[0] = rbuf[0];
+ dev->hw_version[1] = rbuf[1];
+ } else {
+ snd_printk ("not raw, but no "
+ "hardware version!\n");
+ return -1;
+ }
+
+ if (!wf_raw) {
+ return 0;
+ } else {
+ snd_printk ("reloading firmware as you requested.\n");
+ dev->israw = 1;
+ }
+
+ } else {
+
+ dev->israw = 1;
+ snd_printk ("no response to firmware probe, assume raw.\n");
+
+ }
+
+ return 0;
+}
+
+MODULE_FIRMWARE(DEFAULT_OSPATH);
diff --git a/sound/isa/wavefront/yss225.c b/sound/isa/wavefront/yss225.c
new file mode 100644
index 0000000..9f6be3f
--- /dev/null
+++ b/sound/isa/wavefront/yss225.c
@@ -0,0 +1,2739 @@
+/*
+ * Copyright (c) 1998-2002 by Paul Davis <pbd@op.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/* weird stuff, derived from port I/O tracing with dosemu */
+
+static const struct {
+ unsigned char addr;
+ unsigned char data;
+} yss225_registers[] __devinitdata = {
+/* Set all bits for all channels on the MOD unit to zero */
+{ WAIT_IDLE }, { 0xe, 0x10 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x11 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x12 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x13 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x14 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x15 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x16 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x17 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x18 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x19 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x20 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x21 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x22 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x23 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x24 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x25 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x26 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x27 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x28 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x29 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x30 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x31 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x32 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x33 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x34 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x35 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x36 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x37 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x38 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x39 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x40 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x41 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x42 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x43 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x44 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x45 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x46 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x47 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x48 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x49 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x50 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x51 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x52 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x53 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x54 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x55 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x56 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x57 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x58 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x59 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x60 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x61 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x62 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x63 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x64 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x65 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x66 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x67 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x68 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x69 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x70 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x71 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x72 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x73 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x74 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x75 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x76 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x77 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x78 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x79 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x80 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x81 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x82 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x83 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x84 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x85 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x86 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x87 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x88 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x89 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x90 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x91 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x92 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x93 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x94 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x95 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x96 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x97 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x98 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x99 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xaa }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xab }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xac }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xad }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xae }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xaf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xba }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbe }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xca }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xce }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xda }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xde }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xea }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xeb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xec }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xed }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xee }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xef }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfa }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfe }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xff }, { 0xf, 0x00 },
+
+/* XXX But why do this twice? */
+{ WAIT_IDLE }, { 0xe, 0x10 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x11 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x12 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x13 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x14 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x15 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x16 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x17 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x18 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x19 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x20 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x21 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x22 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x23 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x24 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x25 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x26 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x27 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x28 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x29 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x30 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x31 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x32 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x33 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x34 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x35 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x36 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x37 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x38 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x39 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x40 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x41 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x42 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x43 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x44 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x45 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x46 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x47 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x48 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x49 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x50 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x51 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x52 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x53 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x54 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x55 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x56 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x57 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x58 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x59 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x60 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x61 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x62 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x63 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x64 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x65 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x66 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x67 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x68 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x69 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x70 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x71 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x72 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x73 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x74 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x75 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x76 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x77 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x78 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x79 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x80 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x81 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x82 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x83 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x84 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x85 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x86 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x87 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x88 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x89 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x90 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x91 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x92 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x93 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x94 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x95 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x96 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x97 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x98 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x99 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xaa }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xab }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xac }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xad }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xae }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xaf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xba }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbe }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xca }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xce }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xda }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xde }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xea }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xeb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xec }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xed }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xee }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xef }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfa }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfe }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xff }, { 0xf, 0x00 },
+
+/* mute on */
+{ WAIT_IDLE }, { 0x8, 0x02 },
+
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x44 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x42 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x43 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x7c }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x7e }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x46 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x49 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x47 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x4a }, { 0xd, 0x00 }, { 0xc, 0x00 },
+
+/* either because of stupidity by TB's programmers, or because it
+ actually does something, rezero the MOD page. */
+{ WAIT_IDLE }, { 0xe, 0x10 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x11 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x12 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x13 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x14 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x15 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x16 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x17 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x18 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x19 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x1f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x20 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x21 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x22 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x23 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x24 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x25 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x26 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x27 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x28 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x29 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x2f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x30 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x31 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x32 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x33 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x34 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x35 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x36 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x37 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x38 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x39 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x3f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x40 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x41 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x42 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x43 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x44 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x45 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x46 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x47 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x48 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x49 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x4f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x50 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x51 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x52 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x53 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x54 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x55 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x56 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x57 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x58 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x59 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x5f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x60 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x61 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x62 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x63 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x64 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x65 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x66 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x67 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x68 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x69 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x6f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x70 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x71 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x72 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x73 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x74 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x75 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x76 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x77 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x78 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x79 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x7f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x80 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x81 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x82 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x83 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x84 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x85 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x86 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x87 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x88 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x89 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x8f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x90 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x91 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x92 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x93 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x94 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x95 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x96 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x97 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x98 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x99 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9a }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9b }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9c }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9d }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9e }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0x9f }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xa9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xaa }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xab }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xac }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xad }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xae }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xaf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xb9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xba }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbe }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xbf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xc9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xca }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xce }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xcf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xd9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xda }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xde }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xdf }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xe9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xea }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xeb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xec }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xed }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xee }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xef }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf0 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf1 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf2 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf3 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf4 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf5 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf6 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf7 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf8 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xf9 }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfa }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfb }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfc }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfd }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xfe }, { 0xf, 0x00 },
+{ WAIT_IDLE }, { 0xe, 0xff }, { 0xf, 0x00 },
+
+/* load page zero */
+{ 0x9, 0x05 }, { 0xb, 0x00 }, { 0xa, 0x00 },
+
+{ 0xd, 0x01 }, { 0xc, 0x7c }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1e }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0xf5 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x11 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x32 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x13 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x14 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x76 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x18 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x19 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0x1a }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0x17 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x10 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0xa0 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xd1 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0xf2 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x13 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0xf4 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xe0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x15 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x16 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x17 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x50 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x71 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x92 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0xb3 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0xa0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0xd4 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0xf5 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x70 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0xa0 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x11 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x16 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x10 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x17 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1b }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1d }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xdf }, { WAIT_IDLE },
+
+/* Now load page one */
+{ 0x9, 0x05 }, { 0xb, 0x01 }, { 0xa, 0x00 },
+
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x19 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1f }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xd8 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x19 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x18 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0xfa }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1a }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xfb }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xa0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1b }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xd7 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xf7 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1c }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0x3c }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0x3f }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xdf }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0x5d }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0x7d }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0x9e }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xbe }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x03 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1b }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xdb }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xdb }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xe0 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xfb }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xfb }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1b }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x02 }, { 0xa, 0x00 },
+
+{ 0xc, 0xc4 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x25 }, { WAIT_IDLE },
+{ 0xc, 0x01 }, { WAIT_IDLE },
+{ 0xc, 0x06 }, { WAIT_IDLE },
+{ 0xc, 0xc4 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x25 }, { WAIT_IDLE },
+{ 0xc, 0x01 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x04 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x04 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x05 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x44 }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x03 }, { 0xa, 0x00 },
+
+{ 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x47 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x06 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x70 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x42 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x42 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x42 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x42 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x40 }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x04 }, { 0xa, 0x00 },
+
+{ 0xc, 0x63 }, { WAIT_IDLE },
+{ 0xc, 0x03 }, { WAIT_IDLE },
+{ 0xc, 0x26 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x2c }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x24 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x2e }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x01 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x22 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x22 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x22 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x62 }, { WAIT_IDLE },
+{ 0xc, 0x02 }, { WAIT_IDLE },
+{ 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xc, 0x01 }, { WAIT_IDLE },
+{ 0xc, 0x21 }, { WAIT_IDLE },
+{ 0xc, 0x01 }, { WAIT_IDLE },
+
+/* Load memory area (page six) */
+{ 0x9, 0x01 }, { 0xb, 0x06 },
+
+{ 0xa, 0x00 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x02 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x04 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x06 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x08 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x0a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x0c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x0e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x10 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x12 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x14 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x16 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x18 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x1a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x1c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x1e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x20 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x22 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x24 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x26 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x28 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x2a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x2c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x2e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x30 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x32 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x34 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x36 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x38 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x3a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x3c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x3e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x40 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x42 }, { 0xd, 0x03 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x44 }, { 0xd, 0x01 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x46 }, { 0xd, 0x0a }, { 0xc, 0x21 }, { WAIT_IDLE },
+{ 0xa, 0x48 }, { 0xd, 0x0d }, { 0xc, 0x23 }, { WAIT_IDLE },
+{ 0xa, 0x4a }, { 0xd, 0x23 }, { 0xc, 0x1b }, { WAIT_IDLE },
+{ 0xa, 0x4c }, { 0xd, 0x37 }, { 0xc, 0x8f }, { WAIT_IDLE },
+{ 0xa, 0x4e }, { 0xd, 0x45 }, { 0xc, 0x77 }, { WAIT_IDLE },
+{ 0xa, 0x50 }, { 0xd, 0x52 }, { 0xc, 0xe2 }, { WAIT_IDLE },
+{ 0xa, 0x52 }, { 0xd, 0x1c }, { 0xc, 0x92 }, { WAIT_IDLE },
+{ 0xa, 0x54 }, { 0xd, 0x1c }, { 0xc, 0x52 }, { WAIT_IDLE },
+{ 0xa, 0x56 }, { 0xd, 0x07 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x58 }, { 0xd, 0x2f }, { 0xc, 0xc6 }, { WAIT_IDLE },
+{ 0xa, 0x5a }, { 0xd, 0x0b }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x5c }, { 0xd, 0x30 }, { 0xc, 0x06 }, { WAIT_IDLE },
+{ 0xa, 0x5e }, { 0xd, 0x17 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x60 }, { 0xd, 0x3d }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xa, 0x62 }, { 0xd, 0x29 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x64 }, { 0xd, 0x3e }, { 0xc, 0x41 }, { WAIT_IDLE },
+{ 0xa, 0x66 }, { 0xd, 0x39 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x68 }, { 0xd, 0x4c }, { 0xc, 0x48 }, { WAIT_IDLE },
+{ 0xa, 0x6a }, { 0xd, 0x49 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x6c }, { 0xd, 0x4c }, { 0xc, 0x6c }, { WAIT_IDLE },
+{ 0xa, 0x6e }, { 0xd, 0x11 }, { 0xc, 0xd2 }, { WAIT_IDLE },
+{ 0xa, 0x70 }, { 0xd, 0x16 }, { 0xc, 0x0c }, { WAIT_IDLE },
+{ 0xa, 0x72 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x74 }, { 0xd, 0x00 }, { 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xa, 0x76 }, { 0xd, 0x0f }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x78 }, { 0xd, 0x00 }, { 0xc, 0x80 }, { WAIT_IDLE },
+{ 0xa, 0x7a }, { 0xd, 0x13 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x7c }, { 0xd, 0x80 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x7e }, { 0xd, 0x80 }, { 0xc, 0x80 }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x07 }, { 0xa, 0x00 },
+
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x08 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x08 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x08 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x08 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0xe9 }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x8c }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x8c }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x1a }, { 0xc, 0x75 }, { WAIT_IDLE },
+{ 0xd, 0x0d }, { 0xc, 0x8b }, { WAIT_IDLE },
+{ 0xd, 0x04 }, { 0xc, 0xe9 }, { WAIT_IDLE },
+{ 0xd, 0x0b }, { 0xc, 0x16 }, { WAIT_IDLE },
+{ 0xd, 0x1a }, { 0xc, 0x38 }, { WAIT_IDLE },
+{ 0xd, 0x0d }, { 0xc, 0xc8 }, { WAIT_IDLE },
+{ 0xd, 0x04 }, { 0xc, 0x6f }, { WAIT_IDLE },
+{ 0xd, 0x0b }, { 0xc, 0x91 }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x8f }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x62 }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x62 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x7b }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x97 }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0x97 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x52 }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0xf6 }, { WAIT_IDLE },
+{ 0xd, 0x06 }, { 0xc, 0xf6 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x19 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x55 }, { WAIT_IDLE },
+{ 0xd, 0x14 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xd, 0x0d }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xd, 0x04 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xd, 0x14 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xd, 0x0d }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xd, 0x04 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xd, 0x05 }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+
+/* Now setup the MOD area. */
+{ 0xe, 0x01 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x02 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x03 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x04 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x05 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x06 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x07 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x08 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x09 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0a }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0b }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0c }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0d }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0e }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0f }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+
+{ 0xe, 0xb0 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb1 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb2 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb3 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb4 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb5 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb6 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb7 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb8 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb9 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xba }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xbb }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xbc }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xbd }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xbe }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xbf }, { 0xf, 0x20 }, { WAIT_IDLE },
+
+{ 0xe, 0xf0 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf1 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf2 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf3 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf4 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf5 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf6 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf7 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf8 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf9 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xfa }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xfb }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xfc }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xfd }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xfe }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xff }, { 0xf, 0x20 }, { WAIT_IDLE },
+
+{ 0xe, 0x10 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x11 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x12 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x13 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x14 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x15 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x16 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x17 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x18 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x19 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x1a }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x1b }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x1c }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x1d }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x1e }, { 0xf, 0x40 }, { WAIT_IDLE },
+{ 0xe, 0x1f }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x20 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x21 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x22 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x23 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x24 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x25 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x26 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x27 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x28 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x29 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x2a }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x2b }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x2c }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x2d }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x2e }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x2f }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x30 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x31 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x32 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x33 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x34 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x35 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x36 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x37 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x38 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x39 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x3a }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x3b }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x3c }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x3d }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x3e }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x3f }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0x40 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x41 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x42 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x43 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x44 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x45 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x46 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x47 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x48 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x49 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x4a }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x4b }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x4c }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x4d }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x4e }, { 0xf, 0x0e }, { WAIT_IDLE },
+{ 0xe, 0x4f }, { 0xf, 0x0e }, { WAIT_IDLE },
+{ 0xe, 0x50 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x51 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x52 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x53 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x54 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x55 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x56 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x57 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x58 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x59 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x5a }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x5b }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x5c }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x5d }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x5e }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x5f }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x60 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x61 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x62 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x63 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x64 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x65 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x66 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x67 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x68 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x69 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x6a }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x6b }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x6c }, { 0xf, 0x40 }, { WAIT_IDLE },
+{ 0xe, 0x6d }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x6e }, { 0xf, 0x40 }, { WAIT_IDLE },
+{ 0xe, 0x6f }, { 0xf, 0x40 }, { WAIT_IDLE },
+{ 0xe, 0x70 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x71 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x72 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x73 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x74 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x75 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x76 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x77 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x78 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x79 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x7a }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x7b }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x7c }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x7d }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x7e }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x7f }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x80 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x81 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x82 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x83 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x84 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x85 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x86 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x87 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x88 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x89 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x8a }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x8b }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x8c }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x8d }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x8e }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x8f }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x90 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x91 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x92 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x93 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x94 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x95 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x96 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x97 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x98 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x99 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x9a }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x9b }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x9c }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x9d }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x9e }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x9f }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa8 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa9 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xaa }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xab }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xac }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xad }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xae }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xaf }, { 0xf, 0x00 }, { WAIT_IDLE },
+
+{ 0xe, 0xc0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc8 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc9 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xca }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xcb }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xcc }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xcd }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xce }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xcf }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd8 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd9 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xda }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xdb }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xdc }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xdd }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xde }, { 0xf, 0x10 }, { WAIT_IDLE },
+{ 0xe, 0xdf }, { 0xf, 0x10 }, { WAIT_IDLE },
+{ 0xe, 0xe0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe8 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe9 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xea }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xeb }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xec }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xed }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xee }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xef }, { 0xf, 0x00 }, { WAIT_IDLE },
+
+{ 0xe, 0x01 }, { 0xf, 0x00 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x01 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x02 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x03 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x04 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x05 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x06 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x07 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x08 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x09 }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0a }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0b }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0c }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0d }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0e }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x0f }, { 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+
+/* mute on */
+{ 0x8, 0x02 },
+
+/* Now set the coefficients and so forth for the programs above */
+{ 0xb, 0x07 }, { 0xa, 0x46 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x49 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x4b }, { 0xd, 0x03 }, { 0xc, 0x11 }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x4d }, { 0xd, 0x01 }, { 0xc, 0x32 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x46 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x49 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x40 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x41 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x01 }, { 0xa, 0x40 }, { 0xd, 0x02 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xb, 0x01 }, { 0xa, 0x41 }, { 0xd, 0x02 }, { 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x40 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x41 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x47 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x4a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x47 }, { 0xd, 0x01 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x4a }, { 0xd, 0x01 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x47 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x4a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x00 }, { 0xd, 0x01 }, { 0xc, 0x1c }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x44 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x44 }, { 0xd, 0x01 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x44 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x42 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x43 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x42 }, { 0xd, 0x01 }, { 0xc, 0x1a }, { WAIT_IDLE },
+{ 0xb, 0x00 }, { 0xa, 0x43 }, { 0xd, 0x01 }, { 0xc, 0x20 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x42 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x43 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x40 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x41 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x01 }, { 0xa, 0x40 }, { 0xd, 0x02 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xb, 0x01 }, { 0xa, 0x41 }, { 0xd, 0x02 }, { 0xc, 0x60 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x40 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x41 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x44 }, { 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x42 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x43 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x40 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x41 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x51 }, { 0xd, 0x06 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x50 }, { 0xd, 0x06 }, { 0xc, 0x40 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x4f }, { 0xd, 0x03 }, { 0xc, 0x81 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x53 }, { 0xd, 0x1a }, { 0xc, 0x76 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x54 }, { 0xd, 0x0d }, { 0xc, 0x8b }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x55 }, { 0xd, 0x04 }, { 0xc, 0xe9 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x56 }, { 0xd, 0x0b }, { 0xc, 0x17 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x57 }, { 0xd, 0x1a }, { 0xc, 0x38 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x58 }, { 0xd, 0x0d }, { 0xc, 0xc9 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x59 }, { 0xd, 0x04 }, { 0xc, 0x6f }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x5a }, { 0xd, 0x0b }, { 0xc, 0x91 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x73 }, { 0xd, 0x14 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x74 }, { 0xd, 0x0d }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x75 }, { 0xd, 0x04 }, { 0xc, 0xd9 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x76 }, { 0xd, 0x05 }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x77 }, { 0xd, 0x14 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x78 }, { 0xd, 0x0d }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x79 }, { 0xd, 0x04 }, { 0xc, 0xd9 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7a }, { 0xd, 0x05 }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x5e }, { 0xd, 0x03 }, { 0xc, 0x68 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x5c }, { 0xd, 0x04 }, { 0xc, 0x31 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x5d }, { 0xd, 0x04 }, { 0xc, 0x31 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x62 }, { 0xd, 0x03 }, { 0xc, 0x52 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x60 }, { 0xd, 0x04 }, { 0xc, 0x76 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x61 }, { 0xd, 0x04 }, { 0xc, 0x76 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x66 }, { 0xd, 0x03 }, { 0xc, 0x2e }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x64 }, { 0xd, 0x04 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x65 }, { 0xd, 0x04 }, { 0xc, 0xda }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x6a }, { 0xd, 0x02 }, { 0xc, 0xf6 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x68 }, { 0xd, 0x05 }, { 0xc, 0x62 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x69 }, { 0xd, 0x05 }, { 0xc, 0x62 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x46 }, { 0xd, 0x0a }, { 0xc, 0x22 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x48 }, { 0xd, 0x0d }, { 0xc, 0x24 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x6e }, { 0xd, 0x11 }, { 0xc, 0xd3 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x70 }, { 0xd, 0x15 }, { 0xc, 0xcb }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x52 }, { 0xd, 0x20 }, { 0xc, 0x93 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x54 }, { 0xd, 0x20 }, { 0xc, 0x54 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x4a }, { 0xd, 0x27 }, { 0xc, 0x1d }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x58 }, { 0xd, 0x2f }, { 0xc, 0xc8 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x5c }, { 0xd, 0x30 }, { 0xc, 0x07 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x4c }, { 0xd, 0x37 }, { 0xc, 0x90 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x60 }, { 0xd, 0x3d }, { 0xc, 0xdb }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x64 }, { 0xd, 0x3e }, { 0xc, 0x42 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x4e }, { 0xd, 0x45 }, { 0xc, 0x78 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x68 }, { 0xd, 0x4c }, { 0xc, 0x48 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x6c }, { 0xd, 0x4c }, { 0xc, 0x6c }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x50 }, { 0xd, 0x52 }, { 0xc, 0xe2 }, { WAIT_IDLE },
+{ 0xb, 0x06 }, { 0xa, 0x42 }, { 0xd, 0x02 }, { 0xc, 0xba }, { WAIT_IDLE },
+
+/* Some settings (?) */
+{ WAIT_IDLE }, { 0xe, 0x1e }, { 0xf, 0x14 },
+{ WAIT_IDLE }, { 0xe, 0xde }, { 0xf, 0x20 },
+{ WAIT_IDLE }, { 0xe, 0xdf }, { 0xf, 0x20 },
+
+/* some more coefficients */
+{ WAIT_IDLE }, { 0xb, 0x06 }, { 0xa, 0x78 }, { 0xd, 0x00 }, { 0xc, 0x40 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x03 }, { 0xd, 0x0f }, { 0xc, 0xff },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x0b }, { 0xd, 0x0f }, { 0xc, 0xff },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x02 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x0a }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x46 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ WAIT_IDLE }, { 0xb, 0x07 }, { 0xa, 0x49 }, { 0xd, 0x00 }, { 0xc, 0x00 },
+
+/* Now, for some strange reason, lets reload every page
+ and all the coefficients over again. I have *NO* idea
+ why this is done. I do know that no sound is produced
+ is this phase is omitted. */
+{ 0x9, 0x05 }, { 0xb, 0x00 }, { 0xa, 0x10 },
+
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x02 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x01 }, { 0xa, 0x10 },
+
+{ 0xd, 0x01 }, { 0xc, 0xc0 }, { WAIT_IDLE },
+{ 0xd, 0x01 }, { 0xc, 0xfa }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x1a }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+
+{ WAIT_IDLE }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x02 }, { 0xa, 0x10 },
+
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x46 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x03 }, { 0xa, 0x10 },
+
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x04 }, { 0xa, 0x10 },
+
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xc, 0x00 }, { WAIT_IDLE },
+
+/* Page six v.2 */
+{ 0x9, 0x01 }, { 0xb, 0x06 },
+
+{ 0xa, 0x10 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x12 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x14 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x16 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x18 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x1a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x1c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x1e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x20 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x22 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x24 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x26 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x28 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x2a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x2c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x2e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x30 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x32 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x34 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x36 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x38 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x3a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x3c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xa, 0x3e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+
+{ 0x9, 0x05 }, { 0xb, 0x07 }, { 0xa, 0x10 },
+
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+
+{ 0xe, 0x01 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x02 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x03 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x04 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x05 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x06 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x07 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xb0 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb1 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb2 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb3 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb4 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb5 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb6 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xb7 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf0 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf1 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf2 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf3 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf4 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf5 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf6 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0xf7 }, { 0xf, 0x20 }, { WAIT_IDLE },
+{ 0xe, 0x10 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x11 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x12 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x13 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x14 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x15 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x16 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x17 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x20 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x21 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x22 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x23 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x24 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x25 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x26 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x27 }, { 0xf, 0xff }, { WAIT_IDLE },
+{ 0xe, 0x30 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x31 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x32 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x33 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x34 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x35 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x36 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x37 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x40 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x41 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x42 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x43 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x44 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x45 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x46 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x47 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x50 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x51 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x52 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x53 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x54 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x55 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x56 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x57 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x60 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x61 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x62 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x63 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x64 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x65 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x66 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x67 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x70 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x71 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x72 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x73 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x74 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x75 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x76 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x77 }, { 0xf, 0xc0 }, { WAIT_IDLE },
+{ 0xe, 0x80 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x81 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x82 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x83 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x84 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x85 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x86 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x87 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x90 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x91 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x92 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x93 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x94 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x95 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x96 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x97 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xa7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xc7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xd7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe0 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe1 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe2 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe3 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe4 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe5 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe6 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0xe7 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x00 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x02 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x03 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x04 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x05 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x06 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+{ 0xe, 0x01 }, { 0xf, 0x07 }, { WAIT_IDLE },
+{ 0xe, 0x02 }, { 0xf, 0x01 }, { WAIT_IDLE },
+
+{ 0xb, 0x07 }, { 0xa, 0x46 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x49 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x45 }, { 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x48 }, { 0xd, 0x0f }, { 0xc, 0xff }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7b }, { 0xd, 0x04 }, { 0xc, 0xcc }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7d }, { 0xd, 0x04 }, { 0xc, 0xcc }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x7e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x46 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x49 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x47 }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x4a }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0x00 }, { WAIT_IDLE },
+
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x00 }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x00 }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x01 }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x01 }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x01 }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x01 }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x01 }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x01 }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x01 }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x01 }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x01 }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x01 }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x01 }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x01 }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x02 }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x02 }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x02 }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x02 }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x02 }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x02 }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x02 }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x02 }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x02 }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x02 }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x02 }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x02 }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x03 }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x03 }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x03 }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x03 }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x03 }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x03 }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x03 }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x03 }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x03 }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x03 }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x03 }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x03 }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x04 }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x04 }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x04 }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x04 }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x04 }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x04 }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x04 }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x04 }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x04 }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x04 }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x04 }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x04 }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x04 }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x04 }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x05 }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x05 }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x05 }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x05 }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x05 }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x05 }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x05 }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x05 }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x05 }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x05 }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x05 }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x05 }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x06 }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x06 }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x06 }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x06 }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x06 }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x06 }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x06 }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x06 }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x06 }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x06 }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x06 }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x06 }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x07 }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x07 }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x07 }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x07 }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x07 }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x07 }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x07 }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x07 }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x07 }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x07 }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x07 }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x07 }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x08 }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x08 }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x08 }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x08 }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x08 }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x08 }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x08 }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x08 }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x08 }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x08 }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x08 }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x08 }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x08 }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x08 }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x09 }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x09 }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x09 }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x09 }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x09 }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x09 }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x09 }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x09 }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x09 }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x09 }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x09 }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x09 }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0a }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0a }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0a }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0a }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0a }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0a }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0a }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0a }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0a }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0a }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0a }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0a }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0b }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0b }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0b }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0b }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0b }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0b }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0b }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0b }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0b }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0b }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0b }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0b }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0c }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0c }, { 0xc, 0x00 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0c }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0c }, { 0xc, 0x28 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0c }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0c }, { 0xc, 0x51 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0c }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0c }, { 0xc, 0x7a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0c }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0c }, { 0xc, 0xa3 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0c }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0c }, { 0xc, 0xcc },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0c }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0c }, { 0xc, 0xf5 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0d }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0d }, { 0xc, 0x1e },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0d }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0d }, { 0xc, 0x47 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0d }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0d }, { 0xc, 0x70 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0d }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0d }, { 0xc, 0x99 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0d }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0d }, { 0xc, 0xc2 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0d }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0d }, { 0xc, 0xeb },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0e }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0e }, { 0xc, 0x14 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0e }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0e }, { 0xc, 0x3d },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0e }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0e }, { 0xc, 0x66 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0e }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0e }, { 0xc, 0x8f },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0e }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0e }, { 0xc, 0xb8 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0e }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0e }, { 0xc, 0xe1 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0f }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0f }, { 0xc, 0x0a },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0f }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0f }, { 0xc, 0x33 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0f }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0f }, { 0xc, 0x5c },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0f }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0f }, { 0xc, 0x85 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0f }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0f }, { 0xc, 0xae },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0f }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0f }, { 0xc, 0xd7 },
+{ 0xb, 0x07 }, { 0xa, 0x4c }, { 0xd, 0x0f }, { 0xc, 0xff },
+{ 0xb, 0x07 }, { 0xa, 0x4e }, { 0xd, 0x0f }, { 0xc, 0xff },
+
+/* mute off */
+{ 0x8, 0x00 }, { WAIT_IDLE }
+};
diff --git a/sound/isa/wss/Makefile b/sound/isa/wss/Makefile
new file mode 100644
index 0000000..454fee7
--- /dev/null
+++ b/sound/isa/wss/Makefile
@@ -0,0 +1,10 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2008 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-wss-lib-objs := wss_lib.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_WSS_LIB) += snd-wss-lib.o
+
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
new file mode 100644
index 0000000..3d6c5f2
--- /dev/null
+++ b/sound/isa/wss/wss_lib.c
@@ -0,0 +1,2322 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of CS4231(A)/CS4232/InterWave & compatible chips
+ *
+ * Bugs:
+ * - sometimes record brokes playback with WSS portion of
+ * Yamaha OPL3-SA3 chip
+ * - CS4231 (GUS MAX) - still trouble with occasional noises
+ * - broken initialization?
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <sound/core.h>
+#include <sound/wss.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#include <asm/io.h>
+#include <asm/dma.h>
+#include <asm/irq.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Routines for control of CS4231(A)/CS4232/InterWave & compatible chips");
+MODULE_LICENSE("GPL");
+
+#if 0
+#define SNDRV_DEBUG_MCE
+#endif
+
+/*
+ * Some variables
+ */
+
+static unsigned char freq_bits[14] = {
+ /* 5510 */ 0x00 | CS4231_XTAL2,
+ /* 6620 */ 0x0E | CS4231_XTAL2,
+ /* 8000 */ 0x00 | CS4231_XTAL1,
+ /* 9600 */ 0x0E | CS4231_XTAL1,
+ /* 11025 */ 0x02 | CS4231_XTAL2,
+ /* 16000 */ 0x02 | CS4231_XTAL1,
+ /* 18900 */ 0x04 | CS4231_XTAL2,
+ /* 22050 */ 0x06 | CS4231_XTAL2,
+ /* 27042 */ 0x04 | CS4231_XTAL1,
+ /* 32000 */ 0x06 | CS4231_XTAL1,
+ /* 33075 */ 0x0C | CS4231_XTAL2,
+ /* 37800 */ 0x08 | CS4231_XTAL2,
+ /* 44100 */ 0x0A | CS4231_XTAL2,
+ /* 48000 */ 0x0C | CS4231_XTAL1
+};
+
+static unsigned int rates[14] = {
+ 5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050,
+ 27042, 32000, 33075, 37800, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static int snd_wss_xrate(struct snd_pcm_runtime *runtime)
+{
+ return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+}
+
+static unsigned char snd_wss_original_image[32] =
+{
+ 0x00, /* 00/00 - lic */
+ 0x00, /* 01/01 - ric */
+ 0x9f, /* 02/02 - la1ic */
+ 0x9f, /* 03/03 - ra1ic */
+ 0x9f, /* 04/04 - la2ic */
+ 0x9f, /* 05/05 - ra2ic */
+ 0xbf, /* 06/06 - loc */
+ 0xbf, /* 07/07 - roc */
+ 0x20, /* 08/08 - pdfr */
+ CS4231_AUTOCALIB, /* 09/09 - ic */
+ 0x00, /* 0a/10 - pc */
+ 0x00, /* 0b/11 - ti */
+ CS4231_MODE2, /* 0c/12 - mi */
+ 0xfc, /* 0d/13 - lbc */
+ 0x00, /* 0e/14 - pbru */
+ 0x00, /* 0f/15 - pbrl */
+ 0x80, /* 10/16 - afei */
+ 0x01, /* 11/17 - afeii */
+ 0x9f, /* 12/18 - llic */
+ 0x9f, /* 13/19 - rlic */
+ 0x00, /* 14/20 - tlb */
+ 0x00, /* 15/21 - thb */
+ 0x00, /* 16/22 - la3mic/reserved */
+ 0x00, /* 17/23 - ra3mic/reserved */
+ 0x00, /* 18/24 - afs */
+ 0x00, /* 19/25 - lamoc/version */
+ 0xcf, /* 1a/26 - mioc */
+ 0x00, /* 1b/27 - ramoc/reserved */
+ 0x20, /* 1c/28 - cdfr */
+ 0x00, /* 1d/29 - res4 */
+ 0x00, /* 1e/30 - cbru */
+ 0x00, /* 1f/31 - cbrl */
+};
+
+static unsigned char snd_opti93x_original_image[32] =
+{
+ 0x00, /* 00/00 - l_mixout_outctrl */
+ 0x00, /* 01/01 - r_mixout_outctrl */
+ 0x88, /* 02/02 - l_cd_inctrl */
+ 0x88, /* 03/03 - r_cd_inctrl */
+ 0x88, /* 04/04 - l_a1/fm_inctrl */
+ 0x88, /* 05/05 - r_a1/fm_inctrl */
+ 0x80, /* 06/06 - l_dac_inctrl */
+ 0x80, /* 07/07 - r_dac_inctrl */
+ 0x00, /* 08/08 - ply_dataform_reg */
+ 0x00, /* 09/09 - if_conf */
+ 0x00, /* 0a/10 - pin_ctrl */
+ 0x00, /* 0b/11 - err_init_reg */
+ 0x0a, /* 0c/12 - id_reg */
+ 0x00, /* 0d/13 - reserved */
+ 0x00, /* 0e/14 - ply_upcount_reg */
+ 0x00, /* 0f/15 - ply_lowcount_reg */
+ 0x88, /* 10/16 - reserved/l_a1_inctrl */
+ 0x88, /* 11/17 - reserved/r_a1_inctrl */
+ 0x88, /* 12/18 - l_line_inctrl */
+ 0x88, /* 13/19 - r_line_inctrl */
+ 0x88, /* 14/20 - l_mic_inctrl */
+ 0x88, /* 15/21 - r_mic_inctrl */
+ 0x80, /* 16/22 - l_out_outctrl */
+ 0x80, /* 17/23 - r_out_outctrl */
+ 0x00, /* 18/24 - reserved */
+ 0x00, /* 19/25 - reserved */
+ 0x00, /* 1a/26 - reserved */
+ 0x00, /* 1b/27 - reserved */
+ 0x00, /* 1c/28 - cap_dataform_reg */
+ 0x00, /* 1d/29 - reserved */
+ 0x00, /* 1e/30 - cap_upcount_reg */
+ 0x00 /* 1f/31 - cap_lowcount_reg */
+};
+
+/*
+ * Basic I/O functions
+ */
+
+static inline void wss_outb(struct snd_wss *chip, u8 offset, u8 val)
+{
+ outb(val, chip->port + offset);
+}
+
+static inline u8 wss_inb(struct snd_wss *chip, u8 offset)
+{
+ return inb(chip->port + offset);
+}
+
+static void snd_wss_wait(struct snd_wss *chip)
+{
+ int timeout;
+
+ for (timeout = 250;
+ timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT);
+ timeout--)
+ udelay(100);
+}
+
+static void snd_wss_outm(struct snd_wss *chip, unsigned char reg,
+ unsigned char mask, unsigned char value)
+{
+ unsigned char tmp = (chip->image[reg] & mask) | value;
+
+ snd_wss_wait(chip);
+#ifdef CONFIG_SND_DEBUG
+ if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
+ snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value);
+#endif
+ chip->image[reg] = tmp;
+ if (!chip->calibrate_mute) {
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
+ wmb();
+ wss_outb(chip, CS4231P(REG), tmp);
+ mb();
+ }
+}
+
+static void snd_wss_dout(struct snd_wss *chip, unsigned char reg,
+ unsigned char value)
+{
+ int timeout;
+
+ for (timeout = 250;
+ timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT);
+ timeout--)
+ udelay(10);
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
+ wss_outb(chip, CS4231P(REG), value);
+ mb();
+}
+
+void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char value)
+{
+ snd_wss_wait(chip);
+#ifdef CONFIG_SND_DEBUG
+ if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
+ snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value);
+#endif
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
+ wss_outb(chip, CS4231P(REG), value);
+ chip->image[reg] = value;
+ mb();
+ snd_printdd("codec out - reg 0x%x = 0x%x\n",
+ chip->mce_bit | reg, value);
+}
+EXPORT_SYMBOL(snd_wss_out);
+
+unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg)
+{
+ snd_wss_wait(chip);
+#ifdef CONFIG_SND_DEBUG
+ if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
+ snd_printk("in: auto calibration time out - reg = 0x%x\n", reg);
+#endif
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
+ mb();
+ return wss_inb(chip, CS4231P(REG));
+}
+EXPORT_SYMBOL(snd_wss_in);
+
+void snd_cs4236_ext_out(struct snd_wss *chip, unsigned char reg,
+ unsigned char val)
+{
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17);
+ wss_outb(chip, CS4231P(REG),
+ reg | (chip->image[CS4236_EXT_REG] & 0x01));
+ wss_outb(chip, CS4231P(REG), val);
+ chip->eimage[CS4236_REG(reg)] = val;
+#if 0
+ printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val);
+#endif
+}
+EXPORT_SYMBOL(snd_cs4236_ext_out);
+
+unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg)
+{
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17);
+ wss_outb(chip, CS4231P(REG),
+ reg | (chip->image[CS4236_EXT_REG] & 0x01));
+#if 1
+ return wss_inb(chip, CS4231P(REG));
+#else
+ {
+ unsigned char res;
+ res = wss_inb(chip, CS4231P(REG));
+ printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res);
+ return res;
+ }
+#endif
+}
+EXPORT_SYMBOL(snd_cs4236_ext_in);
+
+#if 0
+
+static void snd_wss_debug(struct snd_wss *chip)
+{
+ printk(KERN_DEBUG
+ "CS4231 REGS: INDEX = 0x%02x "
+ " STATUS = 0x%02x\n",
+ wss_inb(chip, CS4231P(REGSEL)),
+ wss_inb(chip, CS4231P(STATUS)));
+ printk(KERN_DEBUG
+ " 0x00: left input = 0x%02x "
+ " 0x10: alt 1 (CFIG 2) = 0x%02x\n",
+ snd_wss_in(chip, 0x00),
+ snd_wss_in(chip, 0x10));
+ printk(KERN_DEBUG
+ " 0x01: right input = 0x%02x "
+ " 0x11: alt 2 (CFIG 3) = 0x%02x\n",
+ snd_wss_in(chip, 0x01),
+ snd_wss_in(chip, 0x11));
+ printk(KERN_DEBUG
+ " 0x02: GF1 left input = 0x%02x "
+ " 0x12: left line in = 0x%02x\n",
+ snd_wss_in(chip, 0x02),
+ snd_wss_in(chip, 0x12));
+ printk(KERN_DEBUG
+ " 0x03: GF1 right input = 0x%02x "
+ " 0x13: right line in = 0x%02x\n",
+ snd_wss_in(chip, 0x03),
+ snd_wss_in(chip, 0x13));
+ printk(KERN_DEBUG
+ " 0x04: CD left input = 0x%02x "
+ " 0x14: timer low = 0x%02x\n",
+ snd_wss_in(chip, 0x04),
+ snd_wss_in(chip, 0x14));
+ printk(KERN_DEBUG
+ " 0x05: CD right input = 0x%02x "
+ " 0x15: timer high = 0x%02x\n",
+ snd_wss_in(chip, 0x05),
+ snd_wss_in(chip, 0x15));
+ printk(KERN_DEBUG
+ " 0x06: left output = 0x%02x "
+ " 0x16: left MIC (PnP) = 0x%02x\n",
+ snd_wss_in(chip, 0x06),
+ snd_wss_in(chip, 0x16));
+ printk(KERN_DEBUG
+ " 0x07: right output = 0x%02x "
+ " 0x17: right MIC (PnP) = 0x%02x\n",
+ snd_wss_in(chip, 0x07),
+ snd_wss_in(chip, 0x17));
+ printk(KERN_DEBUG
+ " 0x08: playback format = 0x%02x "
+ " 0x18: IRQ status = 0x%02x\n",
+ snd_wss_in(chip, 0x08),
+ snd_wss_in(chip, 0x18));
+ printk(KERN_DEBUG
+ " 0x09: iface (CFIG 1) = 0x%02x "
+ " 0x19: left line out = 0x%02x\n",
+ snd_wss_in(chip, 0x09),
+ snd_wss_in(chip, 0x19));
+ printk(KERN_DEBUG
+ " 0x0a: pin control = 0x%02x "
+ " 0x1a: mono control = 0x%02x\n",
+ snd_wss_in(chip, 0x0a),
+ snd_wss_in(chip, 0x1a));
+ printk(KERN_DEBUG
+ " 0x0b: init & status = 0x%02x "
+ " 0x1b: right line out = 0x%02x\n",
+ snd_wss_in(chip, 0x0b),
+ snd_wss_in(chip, 0x1b));
+ printk(KERN_DEBUG
+ " 0x0c: revision & mode = 0x%02x "
+ " 0x1c: record format = 0x%02x\n",
+ snd_wss_in(chip, 0x0c),
+ snd_wss_in(chip, 0x1c));
+ printk(KERN_DEBUG
+ " 0x0d: loopback = 0x%02x "
+ " 0x1d: var freq (PnP) = 0x%02x\n",
+ snd_wss_in(chip, 0x0d),
+ snd_wss_in(chip, 0x1d));
+ printk(KERN_DEBUG
+ " 0x0e: ply upr count = 0x%02x "
+ " 0x1e: ply lwr count = 0x%02x\n",
+ snd_wss_in(chip, 0x0e),
+ snd_wss_in(chip, 0x1e));
+ printk(KERN_DEBUG
+ " 0x0f: rec upr count = 0x%02x "
+ " 0x1f: rec lwr count = 0x%02x\n",
+ snd_wss_in(chip, 0x0f),
+ snd_wss_in(chip, 0x1f));
+}
+
+#endif
+
+/*
+ * CS4231 detection / MCE routines
+ */
+
+static void snd_wss_busy_wait(struct snd_wss *chip)
+{
+ int timeout;
+
+ /* huh.. looks like this sequence is proper for CS4231A chip (GUS MAX) */
+ for (timeout = 5; timeout > 0; timeout--)
+ wss_inb(chip, CS4231P(REGSEL));
+ /* end of cleanup sequence */
+ for (timeout = 25000;
+ timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT);
+ timeout--)
+ udelay(10);
+}
+
+void snd_wss_mce_up(struct snd_wss *chip)
+{
+ unsigned long flags;
+ int timeout;
+
+ snd_wss_wait(chip);
+#ifdef CONFIG_SND_DEBUG
+ if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
+ snd_printk("mce_up - auto calibration time out (0)\n");
+#endif
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->mce_bit |= CS4231_MCE;
+ timeout = wss_inb(chip, CS4231P(REGSEL));
+ if (timeout == 0x80)
+ snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port);
+ if (!(timeout & CS4231_MCE))
+ wss_outb(chip, CS4231P(REGSEL),
+ chip->mce_bit | (timeout & 0x1f));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+EXPORT_SYMBOL(snd_wss_mce_up);
+
+void snd_wss_mce_down(struct snd_wss *chip)
+{
+ unsigned long flags;
+ unsigned long end_time;
+ int timeout;
+ int hw_mask = WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK | WSS_HW_AD1848;
+
+ snd_wss_busy_wait(chip);
+
+#ifdef CONFIG_SND_DEBUG
+ if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
+ snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL));
+#endif
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->mce_bit &= ~CS4231_MCE;
+ timeout = wss_inb(chip, CS4231P(REGSEL));
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (timeout == 0x80)
+ snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port);
+ if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask))
+ return;
+
+ /*
+ * Wait for (possible -- during init auto-calibration may not be set)
+ * calibration process to start. Needs upto 5 sample periods on AD1848
+ * which at the slowest possible rate of 5.5125 kHz means 907 us.
+ */
+ msleep(1);
+
+ snd_printdd("(1) jiffies = %lu\n", jiffies);
+
+ /* check condition up to 250 ms */
+ end_time = jiffies + msecs_to_jiffies(250);
+ while (snd_wss_in(chip, CS4231_TEST_INIT) &
+ CS4231_CALIB_IN_PROGRESS) {
+
+ if (time_after(jiffies, end_time)) {
+ snd_printk(KERN_ERR "mce_down - "
+ "auto calibration time out (2)\n");
+ return;
+ }
+ msleep(1);
+ }
+
+ snd_printdd("(2) jiffies = %lu\n", jiffies);
+
+ /* check condition up to 100 ms */
+ end_time = jiffies + msecs_to_jiffies(100);
+ while (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) {
+ if (time_after(jiffies, end_time)) {
+ snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n");
+ return;
+ }
+ msleep(1);
+ }
+
+ snd_printdd("(3) jiffies = %lu\n", jiffies);
+ snd_printd("mce_down - exit = 0x%x\n", wss_inb(chip, CS4231P(REGSEL)));
+}
+EXPORT_SYMBOL(snd_wss_mce_down);
+
+static unsigned int snd_wss_get_count(unsigned char format, unsigned int size)
+{
+ switch (format & 0xe0) {
+ case CS4231_LINEAR_16:
+ case CS4231_LINEAR_16_BIG:
+ size >>= 1;
+ break;
+ case CS4231_ADPCM_16:
+ return size >> 2;
+ }
+ if (format & CS4231_STEREO)
+ size >>= 1;
+ return size;
+}
+
+static int snd_wss_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ int result = 0;
+ unsigned int what;
+ struct snd_pcm_substream *s;
+ int do_start;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ do_start = 1; break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ do_start = 0; break;
+ default:
+ return -EINVAL;
+ }
+
+ what = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == chip->playback_substream) {
+ what |= CS4231_PLAYBACK_ENABLE;
+ snd_pcm_trigger_done(s, substream);
+ } else if (s == chip->capture_substream) {
+ what |= CS4231_RECORD_ENABLE;
+ snd_pcm_trigger_done(s, substream);
+ }
+ }
+ spin_lock(&chip->reg_lock);
+ if (do_start) {
+ chip->image[CS4231_IFACE_CTRL] |= what;
+ if (chip->trigger)
+ chip->trigger(chip, what, 1);
+ } else {
+ chip->image[CS4231_IFACE_CTRL] &= ~what;
+ if (chip->trigger)
+ chip->trigger(chip, what, 0);
+ }
+ snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]);
+ spin_unlock(&chip->reg_lock);
+#if 0
+ snd_wss_debug(chip);
+#endif
+ return result;
+}
+
+/*
+ * CODEC I/O
+ */
+
+static unsigned char snd_wss_get_rate(unsigned int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(rates); i++)
+ if (rate == rates[i])
+ return freq_bits[i];
+ // snd_BUG();
+ return freq_bits[ARRAY_SIZE(rates) - 1];
+}
+
+static unsigned char snd_wss_get_format(struct snd_wss *chip,
+ int format,
+ int channels)
+{
+ unsigned char rformat;
+
+ rformat = CS4231_LINEAR_8;
+ switch (format) {
+ case SNDRV_PCM_FORMAT_MU_LAW: rformat = CS4231_ULAW_8; break;
+ case SNDRV_PCM_FORMAT_A_LAW: rformat = CS4231_ALAW_8; break;
+ case SNDRV_PCM_FORMAT_S16_LE: rformat = CS4231_LINEAR_16; break;
+ case SNDRV_PCM_FORMAT_S16_BE: rformat = CS4231_LINEAR_16_BIG; break;
+ case SNDRV_PCM_FORMAT_IMA_ADPCM: rformat = CS4231_ADPCM_16; break;
+ }
+ if (channels > 1)
+ rformat |= CS4231_STEREO;
+#if 0
+ snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode);
+#endif
+ return rformat;
+}
+
+static void snd_wss_calibrate_mute(struct snd_wss *chip, int mute)
+{
+ unsigned long flags;
+
+ mute = mute ? 0x80 : 0;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->calibrate_mute == mute) {
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return;
+ }
+ if (!mute) {
+ snd_wss_dout(chip, CS4231_LEFT_INPUT,
+ chip->image[CS4231_LEFT_INPUT]);
+ snd_wss_dout(chip, CS4231_RIGHT_INPUT,
+ chip->image[CS4231_RIGHT_INPUT]);
+ snd_wss_dout(chip, CS4231_LOOPBACK,
+ chip->image[CS4231_LOOPBACK]);
+ }
+ snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT,
+ mute | chip->image[CS4231_AUX1_LEFT_INPUT]);
+ snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT,
+ mute | chip->image[CS4231_AUX1_RIGHT_INPUT]);
+ snd_wss_dout(chip, CS4231_AUX2_LEFT_INPUT,
+ mute | chip->image[CS4231_AUX2_LEFT_INPUT]);
+ snd_wss_dout(chip, CS4231_AUX2_RIGHT_INPUT,
+ mute | chip->image[CS4231_AUX2_RIGHT_INPUT]);
+ snd_wss_dout(chip, CS4231_LEFT_OUTPUT,
+ mute | chip->image[CS4231_LEFT_OUTPUT]);
+ snd_wss_dout(chip, CS4231_RIGHT_OUTPUT,
+ mute | chip->image[CS4231_RIGHT_OUTPUT]);
+ if (!(chip->hardware & WSS_HW_AD1848_MASK)) {
+ snd_wss_dout(chip, CS4231_LEFT_LINE_IN,
+ mute | chip->image[CS4231_LEFT_LINE_IN]);
+ snd_wss_dout(chip, CS4231_RIGHT_LINE_IN,
+ mute | chip->image[CS4231_RIGHT_LINE_IN]);
+ snd_wss_dout(chip, CS4231_MONO_CTRL,
+ mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]);
+ }
+ if (chip->hardware == WSS_HW_INTERWAVE) {
+ snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT,
+ mute | chip->image[CS4231_LEFT_MIC_INPUT]);
+ snd_wss_dout(chip, CS4231_RIGHT_MIC_INPUT,
+ mute | chip->image[CS4231_RIGHT_MIC_INPUT]);
+ snd_wss_dout(chip, CS4231_LINE_LEFT_OUTPUT,
+ mute | chip->image[CS4231_LINE_LEFT_OUTPUT]);
+ snd_wss_dout(chip, CS4231_LINE_RIGHT_OUTPUT,
+ mute | chip->image[CS4231_LINE_RIGHT_OUTPUT]);
+ }
+ chip->calibrate_mute = mute;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static void snd_wss_playback_format(struct snd_wss *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char pdfr)
+{
+ unsigned long flags;
+ int full_calib = 1;
+
+ mutex_lock(&chip->mce_mutex);
+ snd_wss_calibrate_mute(chip, 1);
+ if (chip->hardware == WSS_HW_CS4231A ||
+ (chip->hardware & WSS_HW_CS4232_MASK)) {
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (pdfr & 0x0f)) { /* rate is same? */
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] | 0x10);
+ chip->image[CS4231_PLAYBK_FORMAT] = pdfr;
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT,
+ chip->image[CS4231_PLAYBK_FORMAT]);
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] &= ~0x10);
+ udelay(100); /* Fixes audible clicks at least on GUS MAX */
+ full_calib = 0;
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
+ if (full_calib) {
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->hardware != WSS_HW_INTERWAVE && !chip->single_dma) {
+ if (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE)
+ pdfr = (pdfr & 0xf0) |
+ (chip->image[CS4231_REC_FORMAT] & 0x0f);
+ } else {
+ chip->image[CS4231_PLAYBK_FORMAT] = pdfr;
+ }
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT, pdfr);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (chip->hardware == WSS_HW_OPL3SA2)
+ udelay(100); /* this seems to help */
+ snd_wss_mce_down(chip);
+ }
+ snd_wss_calibrate_mute(chip, 0);
+ mutex_unlock(&chip->mce_mutex);
+}
+
+static void snd_wss_capture_format(struct snd_wss *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char cdfr)
+{
+ unsigned long flags;
+ int full_calib = 1;
+
+ mutex_lock(&chip->mce_mutex);
+ snd_wss_calibrate_mute(chip, 1);
+ if (chip->hardware == WSS_HW_CS4231A ||
+ (chip->hardware & WSS_HW_CS4232_MASK)) {
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (cdfr & 0x0f) || /* rate is same? */
+ (chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) {
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] | 0x20);
+ snd_wss_out(chip, CS4231_REC_FORMAT,
+ chip->image[CS4231_REC_FORMAT] = cdfr);
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] &= ~0x20);
+ full_calib = 0;
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
+ if (full_calib) {
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->hardware != WSS_HW_INTERWAVE &&
+ !(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) {
+ if (chip->single_dma)
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT, cdfr);
+ else
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT,
+ (chip->image[CS4231_PLAYBK_FORMAT] & 0xf0) |
+ (cdfr & 0x0f));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ }
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT, cdfr);
+ else
+ snd_wss_out(chip, CS4231_REC_FORMAT, cdfr);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+ }
+ snd_wss_calibrate_mute(chip, 0);
+ mutex_unlock(&chip->mce_mutex);
+}
+
+/*
+ * Timer interface
+ */
+
+static unsigned long snd_wss_timer_resolution(struct snd_timer *timer)
+{
+ struct snd_wss *chip = snd_timer_chip(timer);
+ if (chip->hardware & WSS_HW_CS4236B_MASK)
+ return 14467;
+ else
+ return chip->image[CS4231_PLAYBK_FORMAT] & 1 ? 9969 : 9920;
+}
+
+static int snd_wss_timer_start(struct snd_timer *timer)
+{
+ unsigned long flags;
+ unsigned int ticks;
+ struct snd_wss *chip = snd_timer_chip(timer);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ticks = timer->sticks;
+ if ((chip->image[CS4231_ALT_FEATURE_1] & CS4231_TIMER_ENABLE) == 0 ||
+ (unsigned char)(ticks >> 8) != chip->image[CS4231_TIMER_HIGH] ||
+ (unsigned char)ticks != chip->image[CS4231_TIMER_LOW]) {
+ chip->image[CS4231_TIMER_HIGH] = (unsigned char) (ticks >> 8);
+ snd_wss_out(chip, CS4231_TIMER_HIGH,
+ chip->image[CS4231_TIMER_HIGH]);
+ chip->image[CS4231_TIMER_LOW] = (unsigned char) ticks;
+ snd_wss_out(chip, CS4231_TIMER_LOW,
+ chip->image[CS4231_TIMER_LOW]);
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] |
+ CS4231_TIMER_ENABLE);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_wss_timer_stop(struct snd_timer *timer)
+{
+ unsigned long flags;
+ struct snd_wss *chip = snd_timer_chip(timer);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE;
+ snd_wss_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static void snd_wss_init(struct snd_wss *chip)
+{
+ unsigned long flags;
+
+ snd_wss_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printk("init: (1)\n");
+#endif
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE |
+ CS4231_PLAYBACK_PIO |
+ CS4231_RECORD_ENABLE |
+ CS4231_RECORD_PIO |
+ CS4231_CALIB_MODE);
+ chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB;
+ snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printk("init: (2)\n");
+#endif
+
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_wss_out(chip,
+ CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printk("init: (3) - afei = 0x%x\n",
+ chip->image[CS4231_ALT_FEATURE_1]);
+#endif
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_wss_out(chip, CS4231_ALT_FEATURE_2,
+ chip->image[CS4231_ALT_FEATURE_2]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_wss_out(chip, CS4231_PLAYBK_FORMAT,
+ chip->image[CS4231_PLAYBK_FORMAT]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printk("init: (4)\n");
+#endif
+
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (!(chip->hardware & WSS_HW_AD1848_MASK))
+ snd_wss_out(chip, CS4231_REC_FORMAT,
+ chip->image[CS4231_REC_FORMAT]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printk("init: (5)\n");
+#endif
+}
+
+static int snd_wss_open(struct snd_wss *chip, unsigned int mode)
+{
+ unsigned long flags;
+
+ mutex_lock(&chip->open_mutex);
+ if ((chip->mode & mode) ||
+ ((chip->mode & WSS_MODE_OPEN) && chip->single_dma)) {
+ mutex_unlock(&chip->open_mutex);
+ return -EAGAIN;
+ }
+ if (chip->mode & WSS_MODE_OPEN) {
+ chip->mode |= mode;
+ mutex_unlock(&chip->open_mutex);
+ return 0;
+ }
+ /* ok. now enable and ack CODEC IRQ */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (!(chip->hardware & WSS_HW_AD1848_MASK)) {
+ snd_wss_out(chip, CS4231_IRQ_STATUS,
+ CS4231_PLAYBACK_IRQ |
+ CS4231_RECORD_IRQ |
+ CS4231_TIMER_IRQ);
+ snd_wss_out(chip, CS4231_IRQ_STATUS, 0);
+ }
+ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */
+ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */
+ chip->image[CS4231_PIN_CTRL] |= CS4231_IRQ_ENABLE;
+ snd_wss_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]);
+ if (!(chip->hardware & WSS_HW_AD1848_MASK)) {
+ snd_wss_out(chip, CS4231_IRQ_STATUS,
+ CS4231_PLAYBACK_IRQ |
+ CS4231_RECORD_IRQ |
+ CS4231_TIMER_IRQ);
+ snd_wss_out(chip, CS4231_IRQ_STATUS, 0);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ chip->mode = mode;
+ mutex_unlock(&chip->open_mutex);
+ return 0;
+}
+
+static void snd_wss_close(struct snd_wss *chip, unsigned int mode)
+{
+ unsigned long flags;
+
+ mutex_lock(&chip->open_mutex);
+ chip->mode &= ~mode;
+ if (chip->mode & WSS_MODE_OPEN) {
+ mutex_unlock(&chip->open_mutex);
+ return;
+ }
+ snd_wss_calibrate_mute(chip, 1);
+
+ /* disable IRQ */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (!(chip->hardware & WSS_HW_AD1848_MASK))
+ snd_wss_out(chip, CS4231_IRQ_STATUS, 0);
+ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */
+ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */
+ chip->image[CS4231_PIN_CTRL] &= ~CS4231_IRQ_ENABLE;
+ snd_wss_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]);
+
+ /* now disable record & playback */
+
+ if (chip->image[CS4231_IFACE_CTRL] & (CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO |
+ CS4231_RECORD_ENABLE | CS4231_RECORD_PIO)) {
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO |
+ CS4231_RECORD_ENABLE | CS4231_RECORD_PIO);
+ snd_wss_out(chip, CS4231_IFACE_CTRL,
+ chip->image[CS4231_IFACE_CTRL]);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_down(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ }
+
+ /* clear IRQ again */
+ if (!(chip->hardware & WSS_HW_AD1848_MASK))
+ snd_wss_out(chip, CS4231_IRQ_STATUS, 0);
+ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */
+ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ snd_wss_calibrate_mute(chip, 0);
+
+ chip->mode = 0;
+ mutex_unlock(&chip->open_mutex);
+}
+
+/*
+ * timer open/close
+ */
+
+static int snd_wss_timer_open(struct snd_timer *timer)
+{
+ struct snd_wss *chip = snd_timer_chip(timer);
+ snd_wss_open(chip, WSS_MODE_TIMER);
+ return 0;
+}
+
+static int snd_wss_timer_close(struct snd_timer *timer)
+{
+ struct snd_wss *chip = snd_timer_chip(timer);
+ snd_wss_close(chip, WSS_MODE_TIMER);
+ return 0;
+}
+
+static struct snd_timer_hardware snd_wss_timer_table =
+{
+ .flags = SNDRV_TIMER_HW_AUTO,
+ .resolution = 9945,
+ .ticks = 65535,
+ .open = snd_wss_timer_open,
+ .close = snd_wss_timer_close,
+ .c_resolution = snd_wss_timer_resolution,
+ .start = snd_wss_timer_start,
+ .stop = snd_wss_timer_stop,
+};
+
+/*
+ * ok.. exported functions..
+ */
+
+static int snd_wss_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ unsigned char new_pdfr;
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ new_pdfr = snd_wss_get_format(chip, params_format(hw_params),
+ params_channels(hw_params)) |
+ snd_wss_get_rate(params_rate(hw_params));
+ chip->set_playback_format(chip, hw_params, new_pdfr);
+ return 0;
+}
+
+static int snd_wss_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_wss_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long flags;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->p_dma_size = size;
+ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO);
+ snd_dma_program(chip->dma1, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT);
+ count = snd_wss_get_count(chip->image[CS4231_PLAYBK_FORMAT], count) - 1;
+ snd_wss_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count);
+ snd_wss_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+#if 0
+ snd_wss_debug(chip);
+#endif
+ return 0;
+}
+
+static int snd_wss_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ unsigned char new_cdfr;
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ new_cdfr = snd_wss_get_format(chip, params_format(hw_params),
+ params_channels(hw_params)) |
+ snd_wss_get_rate(params_rate(hw_params));
+ chip->set_capture_format(chip, hw_params, new_cdfr);
+ return 0;
+}
+
+static int snd_wss_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_wss_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long flags;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->c_dma_size = size;
+ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | CS4231_RECORD_PIO);
+ snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT);
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ count = snd_wss_get_count(chip->image[CS4231_PLAYBK_FORMAT],
+ count);
+ else
+ count = snd_wss_get_count(chip->image[CS4231_REC_FORMAT],
+ count);
+ count--;
+ if (chip->single_dma && chip->hardware != WSS_HW_INTERWAVE) {
+ snd_wss_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count);
+ snd_wss_out(chip, CS4231_PLY_UPR_CNT,
+ (unsigned char) (count >> 8));
+ } else {
+ snd_wss_out(chip, CS4231_REC_LWR_CNT, (unsigned char) count);
+ snd_wss_out(chip, CS4231_REC_UPR_CNT,
+ (unsigned char) (count >> 8));
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+void snd_wss_overrange(struct snd_wss *chip)
+{
+ unsigned long flags;
+ unsigned char res;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ res = snd_wss_in(chip, CS4231_TEST_INIT);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (res & (0x08 | 0x02)) /* detect overrange only above 0dB; may be user selectable? */
+ chip->capture_substream->runtime->overrange++;
+}
+EXPORT_SYMBOL(snd_wss_overrange);
+
+irqreturn_t snd_wss_interrupt(int irq, void *dev_id)
+{
+ struct snd_wss *chip = dev_id;
+ unsigned char status;
+
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ /* pretend it was the only possible irq for AD1848 */
+ status = CS4231_PLAYBACK_IRQ;
+ else
+ status = snd_wss_in(chip, CS4231_IRQ_STATUS);
+ if (status & CS4231_TIMER_IRQ) {
+ if (chip->timer)
+ snd_timer_interrupt(chip->timer, chip->timer->sticks);
+ }
+ if (chip->single_dma && chip->hardware != WSS_HW_INTERWAVE) {
+ if (status & CS4231_PLAYBACK_IRQ) {
+ if (chip->mode & WSS_MODE_PLAY) {
+ if (chip->playback_substream)
+ snd_pcm_period_elapsed(chip->playback_substream);
+ }
+ if (chip->mode & WSS_MODE_RECORD) {
+ if (chip->capture_substream) {
+ snd_wss_overrange(chip);
+ snd_pcm_period_elapsed(chip->capture_substream);
+ }
+ }
+ }
+ } else {
+ if (status & CS4231_PLAYBACK_IRQ) {
+ if (chip->playback_substream)
+ snd_pcm_period_elapsed(chip->playback_substream);
+ }
+ if (status & CS4231_RECORD_IRQ) {
+ if (chip->capture_substream) {
+ snd_wss_overrange(chip);
+ snd_pcm_period_elapsed(chip->capture_substream);
+ }
+ }
+ }
+
+ spin_lock(&chip->reg_lock);
+ status = ~CS4231_ALL_IRQS | ~status;
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ wss_outb(chip, CS4231P(STATUS), 0);
+ else
+ snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0);
+ spin_unlock(&chip->reg_lock);
+ return IRQ_HANDLED;
+}
+EXPORT_SYMBOL(snd_wss_interrupt);
+
+static snd_pcm_uframes_t snd_wss_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE))
+ return 0;
+ ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_wss_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE))
+ return 0;
+ ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+/*
+
+ */
+
+static int snd_ad1848_probe(struct snd_wss *chip)
+{
+ unsigned long timeout = jiffies + msecs_to_jiffies(1000);
+ unsigned long flags;
+ unsigned char r;
+ unsigned short hardware = 0;
+ int err = 0;
+ int i;
+
+ while (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) {
+ if (time_after(jiffies, timeout))
+ return -ENODEV;
+ cond_resched();
+ }
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ /* set CS423x MODE 1 */
+ snd_wss_dout(chip, CS4231_MISC_INFO, 0);
+
+ snd_wss_dout(chip, CS4231_RIGHT_INPUT, 0x45); /* 0x55 & ~0x10 */
+ r = snd_wss_in(chip, CS4231_RIGHT_INPUT);
+ if (r != 0x45) {
+ /* RMGE always high on AD1847 */
+ if ((r & ~CS4231_ENABLE_MIC_GAIN) != 0x45) {
+ err = -ENODEV;
+ goto out;
+ }
+ hardware = WSS_HW_AD1847;
+ } else {
+ snd_wss_dout(chip, CS4231_LEFT_INPUT, 0xaa);
+ r = snd_wss_in(chip, CS4231_LEFT_INPUT);
+ /* L/RMGE always low on AT2320 */
+ if ((r | CS4231_ENABLE_MIC_GAIN) != 0xaa) {
+ err = -ENODEV;
+ goto out;
+ }
+ }
+
+ /* clear pending IRQ */
+ wss_inb(chip, CS4231P(STATUS));
+ wss_outb(chip, CS4231P(STATUS), 0);
+ mb();
+
+ if ((chip->hardware & WSS_HW_TYPE_MASK) != WSS_HW_DETECT)
+ goto out;
+
+ if (hardware) {
+ chip->hardware = hardware;
+ goto out;
+ }
+
+ r = snd_wss_in(chip, CS4231_MISC_INFO);
+
+ /* set CS423x MODE 2 */
+ snd_wss_dout(chip, CS4231_MISC_INFO, CS4231_MODE2);
+ for (i = 0; i < 16; i++) {
+ if (snd_wss_in(chip, i) != snd_wss_in(chip, 16 + i)) {
+ /* we have more than 16 registers: check ID */
+ if ((r & 0xf) != 0xa)
+ goto out_mode;
+ /*
+ * on CMI8330, CS4231_VERSION is volume control and
+ * can be set to 0
+ */
+ snd_wss_dout(chip, CS4231_VERSION, 0);
+ r = snd_wss_in(chip, CS4231_VERSION) & 0xe7;
+ if (!r)
+ chip->hardware = WSS_HW_CMI8330;
+ goto out_mode;
+ }
+ }
+ if (r & 0x80)
+ chip->hardware = WSS_HW_CS4248;
+ else
+ chip->hardware = WSS_HW_AD1848;
+out_mode:
+ snd_wss_dout(chip, CS4231_MISC_INFO, 0);
+out:
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return err;
+}
+
+static int snd_wss_probe(struct snd_wss *chip)
+{
+ unsigned long flags;
+ int i, id, rev, regnum;
+ unsigned char *ptr;
+ unsigned int hw;
+
+ id = snd_ad1848_probe(chip);
+ if (id < 0)
+ return id;
+
+ hw = chip->hardware;
+ if ((hw & WSS_HW_TYPE_MASK) == WSS_HW_DETECT) {
+ for (i = 0; i < 50; i++) {
+ mb();
+ if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
+ msleep(2);
+ else {
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_wss_out(chip, CS4231_MISC_INFO,
+ CS4231_MODE2);
+ id = snd_wss_in(chip, CS4231_MISC_INFO) & 0x0f;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (id == 0x0a)
+ break; /* this is valid value */
+ }
+ }
+ snd_printdd("wss: port = 0x%lx, id = 0x%x\n", chip->port, id);
+ if (id != 0x0a)
+ return -ENODEV; /* no valid device found */
+
+ rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7;
+ snd_printdd("CS4231: VERSION (I25) = 0x%x\n", rev);
+ if (rev == 0x80) {
+ unsigned char tmp = snd_wss_in(chip, 23);
+ snd_wss_out(chip, 23, ~tmp);
+ if (snd_wss_in(chip, 23) != tmp)
+ chip->hardware = WSS_HW_AD1845;
+ else
+ chip->hardware = WSS_HW_CS4231;
+ } else if (rev == 0xa0) {
+ chip->hardware = WSS_HW_CS4231A;
+ } else if (rev == 0xa2) {
+ chip->hardware = WSS_HW_CS4232;
+ } else if (rev == 0xb2) {
+ chip->hardware = WSS_HW_CS4232A;
+ } else if (rev == 0x83) {
+ chip->hardware = WSS_HW_CS4236;
+ } else if (rev == 0x03) {
+ chip->hardware = WSS_HW_CS4236B;
+ } else {
+ snd_printk("unknown CS chip with version 0x%x\n", rev);
+ return -ENODEV; /* unknown CS4231 chip? */
+ }
+ }
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ wss_inb(chip, CS4231P(STATUS)); /* clear any pendings IRQ */
+ wss_outb(chip, CS4231P(STATUS), 0);
+ mb();
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ if (!(chip->hardware & WSS_HW_AD1848_MASK))
+ chip->image[CS4231_MISC_INFO] = CS4231_MODE2;
+ switch (chip->hardware) {
+ case WSS_HW_INTERWAVE:
+ chip->image[CS4231_MISC_INFO] = CS4231_IW_MODE3;
+ break;
+ case WSS_HW_CS4235:
+ case WSS_HW_CS4236B:
+ case WSS_HW_CS4237B:
+ case WSS_HW_CS4238B:
+ case WSS_HW_CS4239:
+ if (hw == WSS_HW_DETECT3)
+ chip->image[CS4231_MISC_INFO] = CS4231_4236_MODE3;
+ else
+ chip->hardware = WSS_HW_CS4236;
+ break;
+ }
+
+ chip->image[CS4231_IFACE_CTRL] =
+ (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) |
+ (chip->single_dma ? CS4231_SINGLE_DMA : 0);
+ if (chip->hardware != WSS_HW_OPTI93X) {
+ chip->image[CS4231_ALT_FEATURE_1] = 0x80;
+ chip->image[CS4231_ALT_FEATURE_2] =
+ chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01;
+ }
+ ptr = (unsigned char *) &chip->image;
+ regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32;
+ snd_wss_mce_down(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ for (i = 0; i < regnum; i++) /* ok.. fill all registers */
+ snd_wss_out(chip, i, *ptr++);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_wss_mce_up(chip);
+ snd_wss_mce_down(chip);
+
+ mdelay(2);
+
+ /* ok.. try check hardware version for CS4236+ chips */
+ if ((hw & WSS_HW_TYPE_MASK) == WSS_HW_DETECT) {
+ if (chip->hardware == WSS_HW_CS4236B) {
+ rev = snd_cs4236_ext_in(chip, CS4236_VERSION);
+ snd_cs4236_ext_out(chip, CS4236_VERSION, 0xff);
+ id = snd_cs4236_ext_in(chip, CS4236_VERSION);
+ snd_cs4236_ext_out(chip, CS4236_VERSION, rev);
+ snd_printdd("CS4231: ext version; rev = 0x%x, id = 0x%x\n", rev, id);
+ if ((id & 0x1f) == 0x1d) { /* CS4235 */
+ chip->hardware = WSS_HW_CS4235;
+ switch (id >> 5) {
+ case 4:
+ case 5:
+ case 6:
+ break;
+ default:
+ snd_printk("unknown CS4235 chip (enhanced version = 0x%x)\n", id);
+ }
+ } else if ((id & 0x1f) == 0x0b) { /* CS4236/B */
+ switch (id >> 5) {
+ case 4:
+ case 5:
+ case 6:
+ case 7:
+ chip->hardware = WSS_HW_CS4236B;
+ break;
+ default:
+ snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id);
+ }
+ } else if ((id & 0x1f) == 0x08) { /* CS4237B */
+ chip->hardware = WSS_HW_CS4237B;
+ switch (id >> 5) {
+ case 4:
+ case 5:
+ case 6:
+ case 7:
+ break;
+ default:
+ snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id);
+ }
+ } else if ((id & 0x1f) == 0x09) { /* CS4238B */
+ chip->hardware = WSS_HW_CS4238B;
+ switch (id >> 5) {
+ case 5:
+ case 6:
+ case 7:
+ break;
+ default:
+ snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id);
+ }
+ } else if ((id & 0x1f) == 0x1e) { /* CS4239 */
+ chip->hardware = WSS_HW_CS4239;
+ switch (id >> 5) {
+ case 4:
+ case 5:
+ case 6:
+ break;
+ default:
+ snd_printk("unknown CS4239 chip (enhanced version = 0x%x)\n", id);
+ }
+ } else {
+ snd_printk("unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", id);
+ }
+ }
+ }
+ return 0; /* all things are ok.. */
+}
+
+/*
+
+ */
+
+static struct snd_pcm_hardware snd_wss_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM |
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE),
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5510,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_wss_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM |
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE),
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5510,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+
+ */
+
+static int snd_wss_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ runtime->hw = snd_wss_playback;
+
+ /* hardware limitation of older chipsets */
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ runtime->hw.formats &= ~(SNDRV_PCM_FMTBIT_IMA_ADPCM |
+ SNDRV_PCM_FMTBIT_S16_BE);
+
+ /* hardware bug in InterWave chipset */
+ if (chip->hardware == WSS_HW_INTERWAVE && chip->dma1 > 3)
+ runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_MU_LAW;
+
+ /* hardware limitation of cheap chips */
+ if (chip->hardware == WSS_HW_CS4235 ||
+ chip->hardware == WSS_HW_CS4239)
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE;
+
+ snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max);
+ snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max);
+
+ if (chip->claim_dma) {
+ if ((err = chip->claim_dma(chip, chip->dma_private_data, chip->dma1)) < 0)
+ return err;
+ }
+
+ err = snd_wss_open(chip, WSS_MODE_PLAY);
+ if (err < 0) {
+ if (chip->release_dma)
+ chip->release_dma(chip, chip->dma_private_data, chip->dma1);
+ snd_free_pages(runtime->dma_area, runtime->dma_bytes);
+ return err;
+ }
+ chip->playback_substream = substream;
+ snd_pcm_set_sync(substream);
+ chip->rate_constraint(runtime);
+ return 0;
+}
+
+static int snd_wss_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ runtime->hw = snd_wss_capture;
+
+ /* hardware limitation of older chipsets */
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ runtime->hw.formats &= ~(SNDRV_PCM_FMTBIT_IMA_ADPCM |
+ SNDRV_PCM_FMTBIT_S16_BE);
+
+ /* hardware limitation of cheap chips */
+ if (chip->hardware == WSS_HW_CS4235 ||
+ chip->hardware == WSS_HW_CS4239 ||
+ chip->hardware == WSS_HW_OPTI93X)
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE;
+
+ snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max);
+ snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max);
+
+ if (chip->claim_dma) {
+ if ((err = chip->claim_dma(chip, chip->dma_private_data, chip->dma2)) < 0)
+ return err;
+ }
+
+ err = snd_wss_open(chip, WSS_MODE_RECORD);
+ if (err < 0) {
+ if (chip->release_dma)
+ chip->release_dma(chip, chip->dma_private_data, chip->dma2);
+ snd_free_pages(runtime->dma_area, runtime->dma_bytes);
+ return err;
+ }
+ chip->capture_substream = substream;
+ snd_pcm_set_sync(substream);
+ chip->rate_constraint(runtime);
+ return 0;
+}
+
+static int snd_wss_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+
+ chip->playback_substream = NULL;
+ snd_wss_close(chip, WSS_MODE_PLAY);
+ return 0;
+}
+
+static int snd_wss_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_wss *chip = snd_pcm_substream_chip(substream);
+
+ chip->capture_substream = NULL;
+ snd_wss_close(chip, WSS_MODE_RECORD);
+ return 0;
+}
+
+static void snd_wss_thinkpad_twiddle(struct snd_wss *chip, int on)
+{
+ int tmp;
+
+ if (!chip->thinkpad_flag)
+ return;
+
+ outb(0x1c, AD1848_THINKPAD_CTL_PORT1);
+ tmp = inb(AD1848_THINKPAD_CTL_PORT2);
+
+ if (on)
+ /* turn it on */
+ tmp |= AD1848_THINKPAD_CS4248_ENABLE_BIT;
+ else
+ /* turn it off */
+ tmp &= ~AD1848_THINKPAD_CS4248_ENABLE_BIT;
+
+ outb(tmp, AD1848_THINKPAD_CTL_PORT2);
+}
+
+#ifdef CONFIG_PM
+
+/* lowlevel suspend callback for CS4231 */
+static void snd_wss_suspend(struct snd_wss *chip)
+{
+ int reg;
+ unsigned long flags;
+
+ snd_pcm_suspend_all(chip->pcm);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ for (reg = 0; reg < 32; reg++)
+ chip->image[reg] = snd_wss_in(chip, reg);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (chip->thinkpad_flag)
+ snd_wss_thinkpad_twiddle(chip, 0);
+}
+
+/* lowlevel resume callback for CS4231 */
+static void snd_wss_resume(struct snd_wss *chip)
+{
+ int reg;
+ unsigned long flags;
+ /* int timeout; */
+
+ if (chip->thinkpad_flag)
+ snd_wss_thinkpad_twiddle(chip, 1);
+ snd_wss_mce_up(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ for (reg = 0; reg < 32; reg++) {
+ switch (reg) {
+ case CS4231_VERSION:
+ break;
+ default:
+ snd_wss_out(chip, reg, chip->image[reg]);
+ break;
+ }
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+#if 1
+ snd_wss_mce_down(chip);
+#else
+ /* The following is a workaround to avoid freeze after resume on TP600E.
+ This is the first half of copy of snd_wss_mce_down(), but doesn't
+ include rescheduling. -- iwai
+ */
+ snd_wss_busy_wait(chip);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->mce_bit &= ~CS4231_MCE;
+ timeout = wss_inb(chip, CS4231P(REGSEL));
+ wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (timeout == 0x80)
+ snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port);
+ if ((timeout & CS4231_MCE) == 0 ||
+ !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) {
+ return;
+ }
+ snd_wss_busy_wait(chip);
+#endif
+}
+#endif /* CONFIG_PM */
+
+static int snd_wss_free(struct snd_wss *chip)
+{
+ release_and_free_resource(chip->res_port);
+ release_and_free_resource(chip->res_cport);
+ if (chip->irq >= 0) {
+ disable_irq(chip->irq);
+ if (!(chip->hwshare & WSS_HWSHARE_IRQ))
+ free_irq(chip->irq, (void *) chip);
+ }
+ if (!(chip->hwshare & WSS_HWSHARE_DMA1) && chip->dma1 >= 0) {
+ snd_dma_disable(chip->dma1);
+ free_dma(chip->dma1);
+ }
+ if (!(chip->hwshare & WSS_HWSHARE_DMA2) &&
+ chip->dma2 >= 0 && chip->dma2 != chip->dma1) {
+ snd_dma_disable(chip->dma2);
+ free_dma(chip->dma2);
+ }
+ if (chip->timer)
+ snd_device_free(chip->card, chip->timer);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_wss_dev_free(struct snd_device *device)
+{
+ struct snd_wss *chip = device->device_data;
+ return snd_wss_free(chip);
+}
+
+const char *snd_wss_chip_id(struct snd_wss *chip)
+{
+ switch (chip->hardware) {
+ case WSS_HW_CS4231:
+ return "CS4231";
+ case WSS_HW_CS4231A:
+ return "CS4231A";
+ case WSS_HW_CS4232:
+ return "CS4232";
+ case WSS_HW_CS4232A:
+ return "CS4232A";
+ case WSS_HW_CS4235:
+ return "CS4235";
+ case WSS_HW_CS4236:
+ return "CS4236";
+ case WSS_HW_CS4236B:
+ return "CS4236B";
+ case WSS_HW_CS4237B:
+ return "CS4237B";
+ case WSS_HW_CS4238B:
+ return "CS4238B";
+ case WSS_HW_CS4239:
+ return "CS4239";
+ case WSS_HW_INTERWAVE:
+ return "AMD InterWave";
+ case WSS_HW_OPL3SA2:
+ return chip->card->shortname;
+ case WSS_HW_AD1845:
+ return "AD1845";
+ case WSS_HW_OPTI93X:
+ return "OPTi 93x";
+ case WSS_HW_AD1847:
+ return "AD1847";
+ case WSS_HW_AD1848:
+ return "AD1848";
+ case WSS_HW_CS4248:
+ return "CS4248";
+ case WSS_HW_CMI8330:
+ return "CMI8330/C3D";
+ default:
+ return "???";
+ }
+}
+EXPORT_SYMBOL(snd_wss_chip_id);
+
+static int snd_wss_new(struct snd_card *card,
+ unsigned short hardware,
+ unsigned short hwshare,
+ struct snd_wss **rchip)
+{
+ struct snd_wss *chip;
+
+ *rchip = NULL;
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ chip->hardware = hardware;
+ chip->hwshare = hwshare;
+
+ spin_lock_init(&chip->reg_lock);
+ mutex_init(&chip->mce_mutex);
+ mutex_init(&chip->open_mutex);
+ chip->card = card;
+ chip->rate_constraint = snd_wss_xrate;
+ chip->set_playback_format = snd_wss_playback_format;
+ chip->set_capture_format = snd_wss_capture_format;
+ if (chip->hardware == WSS_HW_OPTI93X)
+ memcpy(&chip->image, &snd_opti93x_original_image,
+ sizeof(snd_opti93x_original_image));
+ else
+ memcpy(&chip->image, &snd_wss_original_image,
+ sizeof(snd_wss_original_image));
+ if (chip->hardware & WSS_HW_AD1848_MASK) {
+ chip->image[CS4231_PIN_CTRL] = 0;
+ chip->image[CS4231_TEST_INIT] = 0;
+ }
+
+ *rchip = chip;
+ return 0;
+}
+
+int snd_wss_create(struct snd_card *card,
+ unsigned long port,
+ unsigned long cport,
+ int irq, int dma1, int dma2,
+ unsigned short hardware,
+ unsigned short hwshare,
+ struct snd_wss **rchip)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_wss_dev_free,
+ };
+ struct snd_wss *chip;
+ int err;
+
+ err = snd_wss_new(card, hardware, hwshare, &chip);
+ if (err < 0)
+ return err;
+
+ chip->irq = -1;
+ chip->dma1 = -1;
+ chip->dma2 = -1;
+
+ chip->res_port = request_region(port, 4, "WSS");
+ if (!chip->res_port) {
+ snd_printk(KERN_ERR "wss: can't grab port 0x%lx\n", port);
+ snd_wss_free(chip);
+ return -EBUSY;
+ }
+ chip->port = port;
+ if ((long)cport >= 0) {
+ chip->res_cport = request_region(cport, 8, "CS4232 Control");
+ if (!chip->res_cport) {
+ snd_printk(KERN_ERR
+ "wss: can't grab control port 0x%lx\n", cport);
+ snd_wss_free(chip);
+ return -ENODEV;
+ }
+ }
+ chip->cport = cport;
+ if (!(hwshare & WSS_HWSHARE_IRQ))
+ if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED,
+ "WSS", (void *) chip)) {
+ snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq);
+ snd_wss_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = irq;
+ if (!(hwshare & WSS_HWSHARE_DMA1) && request_dma(dma1, "WSS - 1")) {
+ snd_printk(KERN_ERR "wss: can't grab DMA1 %d\n", dma1);
+ snd_wss_free(chip);
+ return -EBUSY;
+ }
+ chip->dma1 = dma1;
+ if (!(hwshare & WSS_HWSHARE_DMA2) && dma1 != dma2 &&
+ dma2 >= 0 && request_dma(dma2, "WSS - 2")) {
+ snd_printk(KERN_ERR "wss: can't grab DMA2 %d\n", dma2);
+ snd_wss_free(chip);
+ return -EBUSY;
+ }
+ if (dma1 == dma2 || dma2 < 0) {
+ chip->single_dma = 1;
+ chip->dma2 = chip->dma1;
+ } else
+ chip->dma2 = dma2;
+
+ if (hardware == WSS_HW_THINKPAD) {
+ chip->thinkpad_flag = 1;
+ chip->hardware = WSS_HW_DETECT; /* reset */
+ snd_wss_thinkpad_twiddle(chip, 1);
+ }
+
+ /* global setup */
+ if (snd_wss_probe(chip) < 0) {
+ snd_wss_free(chip);
+ return -ENODEV;
+ }
+ snd_wss_init(chip);
+
+#if 0
+ if (chip->hardware & WSS_HW_CS4232_MASK) {
+ if (chip->res_cport == NULL)
+ snd_printk("CS4232 control port features are not accessible\n");
+ }
+#endif
+
+ /* Register device */
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ snd_wss_free(chip);
+ return err;
+ }
+
+#ifdef CONFIG_PM
+ /* Power Management */
+ chip->suspend = snd_wss_suspend;
+ chip->resume = snd_wss_resume;
+#endif
+
+ *rchip = chip;
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_create);
+
+static struct snd_pcm_ops snd_wss_playback_ops = {
+ .open = snd_wss_playback_open,
+ .close = snd_wss_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_wss_playback_hw_params,
+ .hw_free = snd_wss_playback_hw_free,
+ .prepare = snd_wss_playback_prepare,
+ .trigger = snd_wss_trigger,
+ .pointer = snd_wss_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_wss_capture_ops = {
+ .open = snd_wss_capture_open,
+ .close = snd_wss_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_wss_capture_hw_params,
+ .hw_free = snd_wss_capture_hw_free,
+ .prepare = snd_wss_capture_prepare,
+ .trigger = snd_wss_trigger,
+ .pointer = snd_wss_capture_pointer,
+};
+
+int snd_wss_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(chip->card, "WSS", device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops);
+
+ /* global setup */
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ if (chip->single_dma)
+ pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX;
+ if (chip->hardware != WSS_HW_INTERWAVE)
+ pcm->info_flags |= SNDRV_PCM_INFO_JOINT_DUPLEX;
+ strcpy(pcm->name, snd_wss_chip_id(chip));
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
+ 64*1024, chip->dma1 > 3 || chip->dma2 > 3 ? 128*1024 : 64*1024);
+
+ chip->pcm = pcm;
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_pcm);
+
+static void snd_wss_timer_free(struct snd_timer *timer)
+{
+ struct snd_wss *chip = timer->private_data;
+ chip->timer = NULL;
+}
+
+int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer)
+{
+ struct snd_timer *timer;
+ struct snd_timer_id tid;
+ int err;
+
+ /* Timer initialization */
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = chip->card->number;
+ tid.device = device;
+ tid.subdevice = 0;
+ if ((err = snd_timer_new(chip->card, "CS4231", &tid, &timer)) < 0)
+ return err;
+ strcpy(timer->name, snd_wss_chip_id(chip));
+ timer->private_data = chip;
+ timer->private_free = snd_wss_timer_free;
+ timer->hw = snd_wss_timer_table;
+ chip->timer = timer;
+ if (rtimer)
+ *rtimer = timer;
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_timer);
+
+/*
+ * MIXER part
+ */
+
+static int snd_wss_info_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {
+ "Line", "Aux", "Mic", "Mix"
+ };
+ static char *opl3sa_texts[4] = {
+ "Line", "CD", "Mic", "Mix"
+ };
+ static char *gusmax_texts[4] = {
+ "Line", "Synth", "Mic", "Mix"
+ };
+ char **ptexts = texts;
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+
+ if (snd_BUG_ON(!chip->card))
+ return -EINVAL;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 2;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ if (!strcmp(chip->card->driver, "GUS MAX"))
+ ptexts = gusmax_texts;
+ switch (chip->hardware) {
+ case WSS_HW_INTERWAVE:
+ ptexts = gusmax_texts;
+ break;
+ case WSS_HW_OPL3SA2:
+ ptexts = opl3sa_texts;
+ break;
+ }
+ strcpy(uinfo->value.enumerated.name, ptexts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_wss_get_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.enumerated.item[0] = (chip->image[CS4231_LEFT_INPUT] & CS4231_MIXS_ALL) >> 6;
+ ucontrol->value.enumerated.item[1] = (chip->image[CS4231_RIGHT_INPUT] & CS4231_MIXS_ALL) >> 6;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_wss_put_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned short left, right;
+ int change;
+
+ if (ucontrol->value.enumerated.item[0] > 3 ||
+ ucontrol->value.enumerated.item[1] > 3)
+ return -EINVAL;
+ left = ucontrol->value.enumerated.item[0] << 6;
+ right = ucontrol->value.enumerated.item[1] << 6;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left;
+ right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right;
+ change = left != chip->image[CS4231_LEFT_INPUT] ||
+ right != chip->image[CS4231_RIGHT_INPUT];
+ snd_wss_out(chip, CS4231_LEFT_INPUT, left);
+ snd_wss_out(chip, CS4231_RIGHT_INPUT, right);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+int snd_wss_info_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_info_single);
+
+int snd_wss_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_get_single);
+
+int snd_wss_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = (chip->image[reg] & ~(mask << shift)) | val;
+ change = val != chip->image[reg];
+ snd_wss_out(chip, reg, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+EXPORT_SYMBOL(snd_wss_put_single);
+
+int snd_wss_info_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_info_double);
+
+int snd_wss_get_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_get_double);
+
+int snd_wss_put_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_wss *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned short val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (left_reg != right_reg) {
+ val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1;
+ val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2;
+ change = val1 != chip->image[left_reg] ||
+ val2 != chip->image[right_reg];
+ snd_wss_out(chip, left_reg, val1);
+ snd_wss_out(chip, right_reg, val2);
+ } else {
+ mask = (mask << shift_left) | (mask << shift_right);
+ val1 = (chip->image[left_reg] & ~mask) | val1 | val2;
+ change = val1 != chip->image[left_reg];
+ snd_wss_out(chip, left_reg, val1);
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+EXPORT_SYMBOL(snd_wss_put_double);
+
+static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0);
+
+static struct snd_kcontrol_new snd_ad1848_controls[] = {
+WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT,
+ 7, 7, 1, 1),
+WSS_DOUBLE_TLV("PCM Playback Volume", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1,
+ db_scale_6bit),
+WSS_DOUBLE("Aux Playback Switch", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE_TLV("Aux Playback Volume", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1,
+ db_scale_5bit_12db_max),
+WSS_DOUBLE("Aux Playback Switch", 1,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE_TLV("Aux Playback Volume", 1,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1,
+ db_scale_5bit_12db_max),
+WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT,
+ 0, 0, 15, 0, db_scale_rec_gain),
+{
+ .name = "Capture Source",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_wss_info_mux,
+ .get = snd_wss_get_mux,
+ .put = snd_wss_put_mux,
+},
+WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0),
+WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0,
+ db_scale_6bit),
+};
+
+static struct snd_kcontrol_new snd_wss_controls[] = {
+WSS_DOUBLE("PCM Playback Switch", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
+WSS_DOUBLE("PCM Playback Volume", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1),
+WSS_DOUBLE("Line Playback Switch", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
+WSS_DOUBLE("Line Playback Volume", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1),
+WSS_DOUBLE("Aux Playback Switch", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Aux Playback Volume", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1),
+WSS_DOUBLE("Aux Playback Switch", 1,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Aux Playback Volume", 1,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1),
+WSS_SINGLE("Mono Playback Switch", 0,
+ CS4231_MONO_CTRL, 7, 1, 1),
+WSS_SINGLE("Mono Playback Volume", 0,
+ CS4231_MONO_CTRL, 0, 15, 1),
+WSS_SINGLE("Mono Output Playback Switch", 0,
+ CS4231_MONO_CTRL, 6, 1, 1),
+WSS_SINGLE("Mono Output Playback Bypass", 0,
+ CS4231_MONO_CTRL, 5, 1, 0),
+WSS_DOUBLE("Capture Volume", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_wss_info_mux,
+ .get = snd_wss_get_mux,
+ .put = snd_wss_put_mux,
+},
+WSS_DOUBLE("Mic Boost", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0),
+WSS_SINGLE("Loopback Capture Switch", 0,
+ CS4231_LOOPBACK, 0, 1, 0),
+WSS_SINGLE("Loopback Capture Volume", 0,
+ CS4231_LOOPBACK, 2, 63, 1)
+};
+
+static struct snd_kcontrol_new snd_opti93x_controls[] = {
+WSS_DOUBLE("Master Playback Switch", 0,
+ OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1),
+WSS_DOUBLE("Master Playback Volume", 0,
+ OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1),
+WSS_DOUBLE("PCM Playback Switch", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
+WSS_DOUBLE("PCM Playback Volume", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1),
+WSS_DOUBLE("FM Playback Switch", 0,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("FM Playback Volume", 0,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1),
+WSS_DOUBLE("Line Playback Switch", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
+WSS_DOUBLE("Line Playback Volume", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1),
+WSS_DOUBLE("Mic Playback Switch", 0,
+ OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Mic Playback Volume", 0,
+ OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1),
+WSS_DOUBLE("Mic Boost", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0),
+WSS_DOUBLE("CD Playback Switch", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("CD Playback Volume", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1),
+WSS_DOUBLE("Aux Playback Switch", 0,
+ OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("Aux Playback Volume", 0,
+ OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1),
+WSS_DOUBLE("Capture Volume", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_wss_info_mux,
+ .get = snd_wss_get_mux,
+ .put = snd_wss_put_mux,
+}
+};
+
+int snd_wss_mixer(struct snd_wss *chip)
+{
+ struct snd_card *card;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!chip || !chip->pcm))
+ return -EINVAL;
+
+ card = chip->card;
+
+ strcpy(card->mixername, chip->pcm->name);
+
+ if (chip->hardware == WSS_HW_OPTI93X)
+ for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_opti93x_controls[idx],
+ chip));
+ if (err < 0)
+ return err;
+ }
+ else if (chip->hardware & WSS_HW_AD1848_MASK)
+ for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_ad1848_controls[idx],
+ chip));
+ if (err < 0)
+ return err;
+ }
+ else
+ for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_wss_controls[idx],
+ chip));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+EXPORT_SYMBOL(snd_wss_mixer);
+
+const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction)
+{
+ return direction == SNDRV_PCM_STREAM_PLAYBACK ?
+ &snd_wss_playback_ops : &snd_wss_capture_ops;
+}
+EXPORT_SYMBOL(snd_wss_get_pcm_ops);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_wss_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_wss_exit(void)
+{
+}
+
+module_init(alsa_wss_init);
+module_exit(alsa_wss_exit);
diff --git a/sound/last.c b/sound/last.c
new file mode 100644
index 0000000..bdd0857
--- /dev/null
+++ b/sound/last.c
@@ -0,0 +1,41 @@
+/*
+ * Advanced Linux Sound Architecture
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define SNDRV_MAIN_OBJECT_FILE
+#include <linux/init.h>
+#include <sound/core.h>
+
+static int __init alsa_sound_last_init(void)
+{
+ int idx, ok = 0;
+
+ printk(KERN_INFO "ALSA device list:\n");
+ for (idx = 0; idx < SNDRV_CARDS; idx++)
+ if (snd_cards[idx] != NULL) {
+ printk(KERN_INFO " #%i: %s\n", idx, snd_cards[idx]->longname);
+ ok++;
+ }
+ if (ok == 0)
+ printk(KERN_INFO " No soundcards found.\n");
+ return 0;
+}
+
+__initcall(alsa_sound_last_init);
diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig
new file mode 100644
index 0000000..a9823fa
--- /dev/null
+++ b/sound/mips/Kconfig
@@ -0,0 +1,34 @@
+# ALSA MIPS drivers
+
+menuconfig SND_MIPS
+ bool "MIPS sound devices"
+ depends on MIPS
+ default y
+ help
+ Support for sound devices of MIPS architectures.
+
+if SND_MIPS
+
+config SND_SGI_O2
+ tristate "SGI O2 Audio"
+ depends on SGI_IP32
+ help
+ Sound support for the SGI O2 Workstation.
+
+config SND_SGI_HAL2
+ tristate "SGI HAL2 Audio"
+ depends on SGI_HAS_HAL2
+ help
+ Sound support for the SGI Indy and Indigo2 Workstation.
+
+
+config SND_AU1X00
+ tristate "Au1x00 AC97 Port Driver"
+ depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500
+ select SND_PCM
+ select SND_AC97_CODEC
+ help
+ ALSA Sound driver for the Au1x00's AC97 port.
+
+endif # SND_MIPS
+
diff --git a/sound/mips/Makefile b/sound/mips/Makefile
new file mode 100644
index 0000000..861ec0a
--- /dev/null
+++ b/sound/mips/Makefile
@@ -0,0 +1,12 @@
+#
+# Makefile for ALSA
+#
+
+snd-au1x00-objs := au1x00.o
+snd-sgi-o2-objs := sgio2audio.o ad1843.o
+snd-sgi-hal2-objs := hal2.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o
+obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o
+obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o
diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c
new file mode 100644
index 0000000..c624510
--- /dev/null
+++ b/sound/mips/ad1843.c
@@ -0,0 +1,561 @@
+/*
+ * AD1843 low level driver
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ * Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
+ *
+ * inspired from vwsnd.c (SGI VW audio driver)
+ * Copyright 1999 Silicon Graphics, Inc. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/sched.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ad1843.h>
+
+/*
+ * AD1843 bitfield definitions. All are named as in the AD1843 data
+ * sheet, with ad1843_ prepended and individual bit numbers removed.
+ *
+ * E.g., bits LSS0 through LSS2 become ad1843_LSS.
+ *
+ * Only the bitfields we need are defined.
+ */
+
+struct ad1843_bitfield {
+ char reg;
+ char lo_bit;
+ char nbits;
+};
+
+static const struct ad1843_bitfield
+ ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */
+ ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */
+ ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */
+ ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */
+ ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */
+ ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */
+ ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */
+ ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */
+ ad1843_RD2M = { 3, 0, 5 }, /* Right DAC 2 Mix Gain/Atten */
+ ad1843_RD2MM = { 3, 7, 1 }, /* Right DAC 2 Mix Mute */
+ ad1843_LD2M = { 3, 8, 5 }, /* Left DAC 2 Mix Gain/Atten */
+ ad1843_LD2MM = { 3, 15, 1 }, /* Left DAC 2 Mix Mute */
+ ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */
+ ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */
+ ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */
+ ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */
+ ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */
+ ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */
+ ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */
+ ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */
+ ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */
+ ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */
+ ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */
+ ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */
+ ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */
+ ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */
+ ad1843_MPOM = { 8, 6, 1 }, /* Mono Output Mute */
+ ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */
+ ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */
+ ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */
+ ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */
+ ad1843_RDA2G = { 10, 0, 6 }, /* Right DAC2 Analog/Digital Gain */
+ ad1843_RDA2GM = { 10, 7, 1 }, /* Right DAC2 Analog Mute */
+ ad1843_LDA2G = { 10, 8, 6 }, /* Left DAC2 Analog/Digital Gain */
+ ad1843_LDA2GM = { 10, 15, 1 }, /* Left DAC2 Analog Mute */
+ ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */
+ ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */
+ ad1843_RDA2AM = { 12, 7, 1 }, /* Right DAC2 Digital Mute */
+ ad1843_LDA2AM = { 12, 15, 1 }, /* Left DAC2 Digital Mute */
+ ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */
+ ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */
+ ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */
+ ad1843_DA2C = { 15, 10, 2 }, /* DAC2 Sample Rate Source */
+ ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */
+ ad1843_C2C = { 20, 0, 16 }, /* Clock 2 Sample Rate Select */
+ ad1843_C3C = { 23, 0, 16 }, /* Clock 3 Sample Rate Select */
+ ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */
+ ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */
+ ad1843_DAMIX = { 25, 14, 1 }, /* DAC Digital Mix Enable */
+ ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */
+ ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */
+ ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */
+ ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */
+ ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */
+ ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */
+ ad1843_DA2F = { 26, 10, 2 }, /* DAC2 Data Format Select */
+ ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */
+ ad1843_DA2SM = { 26, 15, 1 }, /* DAC2 Stereo/Mono Mode Select */
+ ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */
+ ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */
+ ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */
+ ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */
+ ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */
+ ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */
+ ad1843_DDMEN = { 27, 12, 1 }, /* DAC2 to DAC1 Mix Enable */
+ ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */
+ ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */
+ ad1843_C3EN = { 28, 13, 1 }, /* Clock Generator 3 Enable */
+ ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */
+
+/*
+ * The various registers of the AD1843 use three different formats for
+ * specifying gain. The ad1843_gain structure parameterizes the
+ * formats.
+ */
+
+struct ad1843_gain {
+ int negative; /* nonzero if gain is negative. */
+ const struct ad1843_bitfield *lfield;
+ const struct ad1843_bitfield *rfield;
+ const struct ad1843_bitfield *lmute;
+ const struct ad1843_bitfield *rmute;
+};
+
+static const struct ad1843_gain ad1843_gain_RECLEV = {
+ .negative = 0,
+ .lfield = &ad1843_LIG,
+ .rfield = &ad1843_RIG
+};
+static const struct ad1843_gain ad1843_gain_LINE = {
+ .negative = 1,
+ .lfield = &ad1843_LX1M,
+ .rfield = &ad1843_RX1M,
+ .lmute = &ad1843_LX1MM,
+ .rmute = &ad1843_RX1MM
+};
+static const struct ad1843_gain ad1843_gain_LINE_2 = {
+ .negative = 1,
+ .lfield = &ad1843_LDA2G,
+ .rfield = &ad1843_RDA2G,
+ .lmute = &ad1843_LDA2GM,
+ .rmute = &ad1843_RDA2GM
+};
+static const struct ad1843_gain ad1843_gain_MIC = {
+ .negative = 1,
+ .lfield = &ad1843_LMCM,
+ .rfield = &ad1843_RMCM,
+ .lmute = &ad1843_LMCMM,
+ .rmute = &ad1843_RMCMM
+};
+static const struct ad1843_gain ad1843_gain_PCM_0 = {
+ .negative = 1,
+ .lfield = &ad1843_LDA1G,
+ .rfield = &ad1843_RDA1G,
+ .lmute = &ad1843_LDA1GM,
+ .rmute = &ad1843_RDA1GM
+};
+static const struct ad1843_gain ad1843_gain_PCM_1 = {
+ .negative = 1,
+ .lfield = &ad1843_LD2M,
+ .rfield = &ad1843_RD2M,
+ .lmute = &ad1843_LD2MM,
+ .rmute = &ad1843_RD2MM
+};
+
+static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] =
+{
+ &ad1843_gain_RECLEV,
+ &ad1843_gain_LINE,
+ &ad1843_gain_LINE_2,
+ &ad1843_gain_MIC,
+ &ad1843_gain_PCM_0,
+ &ad1843_gain_PCM_1,
+};
+
+/* read the current value of an AD1843 bitfield. */
+
+static int ad1843_read_bits(struct snd_ad1843 *ad1843,
+ const struct ad1843_bitfield *field)
+{
+ int w;
+
+ w = ad1843->read(ad1843->chip, field->reg);
+ return w >> field->lo_bit & ((1 << field->nbits) - 1);
+}
+
+/*
+ * write a new value to an AD1843 bitfield and return the old value.
+ */
+
+static int ad1843_write_bits(struct snd_ad1843 *ad1843,
+ const struct ad1843_bitfield *field,
+ int newval)
+{
+ int w, mask, oldval, newbits;
+
+ w = ad1843->read(ad1843->chip, field->reg);
+ mask = ((1 << field->nbits) - 1) << field->lo_bit;
+ oldval = (w & mask) >> field->lo_bit;
+ newbits = (newval << field->lo_bit) & mask;
+ w = (w & ~mask) | newbits;
+ ad1843->write(ad1843->chip, field->reg, w);
+
+ return oldval;
+}
+
+/*
+ * ad1843_read_multi reads multiple bitfields from the same AD1843
+ * register. It uses a single read cycle to do it. (Reading the
+ * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20
+ * microseconds.)
+ *
+ * Called like this.
+ *
+ * ad1843_read_multi(ad1843, nfields,
+ * &ad1843_FIELD1, &val1,
+ * &ad1843_FIELD2, &val2, ...);
+ */
+
+static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+ va_list ap;
+ const struct ad1843_bitfield *fp;
+ int w = 0, mask, *value, reg = -1;
+
+ va_start(ap, argcount);
+ while (--argcount >= 0) {
+ fp = va_arg(ap, const struct ad1843_bitfield *);
+ value = va_arg(ap, int *);
+ if (reg == -1) {
+ reg = fp->reg;
+ w = ad1843->read(ad1843->chip, reg);
+ }
+
+ mask = (1 << fp->nbits) - 1;
+ *value = w >> fp->lo_bit & mask;
+ }
+ va_end(ap);
+}
+
+/*
+ * ad1843_write_multi stores multiple bitfields into the same AD1843
+ * register. It uses one read and one write cycle to do it.
+ *
+ * Called like this.
+ *
+ * ad1843_write_multi(ad1843, nfields,
+ * &ad1843_FIELD1, val1,
+ * &ad1843_FIELF2, val2, ...);
+ */
+
+static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+ va_list ap;
+ int reg;
+ const struct ad1843_bitfield *fp;
+ int value;
+ int w, m, mask, bits;
+
+ mask = 0;
+ bits = 0;
+ reg = -1;
+
+ va_start(ap, argcount);
+ while (--argcount >= 0) {
+ fp = va_arg(ap, const struct ad1843_bitfield *);
+ value = va_arg(ap, int);
+ if (reg == -1)
+ reg = fp->reg;
+ else
+ BUG_ON(reg != fp->reg);
+ m = ((1 << fp->nbits) - 1) << fp->lo_bit;
+ mask |= m;
+ bits |= (value << fp->lo_bit) & m;
+ }
+ va_end(ap);
+
+ if (~mask & 0xFFFF)
+ w = ad1843->read(ad1843->chip, reg);
+ else
+ w = 0;
+ w = (w & ~mask) | bits;
+ ad1843->write(ad1843->chip, reg, w);
+}
+
+int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id)
+{
+ const struct ad1843_gain *gp = ad1843_gain[id];
+ int ret;
+
+ ret = (1 << gp->lfield->nbits);
+ if (!gp->lmute)
+ ret -= 1;
+ return ret;
+}
+
+/*
+ * ad1843_get_gain reads the specified register and extracts the gain value
+ * using the supplied gain type.
+ */
+
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id)
+{
+ int lg, rg, lm, rm;
+ const struct ad1843_gain *gp = ad1843_gain[id];
+ unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+ ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg);
+ if (gp->negative) {
+ lg = mask - lg;
+ rg = mask - rg;
+ }
+ if (gp->lmute) {
+ ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm);
+ if (lm)
+ lg = 0;
+ if (rm)
+ rg = 0;
+ }
+ return lg << 0 | rg << 8;
+}
+
+/*
+ * Set an audio channel's gain.
+ *
+ * Returns the new gain, which may be lower than the old gain.
+ */
+
+int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval)
+{
+ const struct ad1843_gain *gp = ad1843_gain[id];
+ unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+ int lg = (newval >> 0) & mask;
+ int rg = (newval >> 8) & mask;
+ int lm = (lg == 0) ? 1 : 0;
+ int rm = (rg == 0) ? 1 : 0;
+
+ if (gp->negative) {
+ lg = mask - lg;
+ rg = mask - rg;
+ }
+ if (gp->lmute)
+ ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm);
+ ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg);
+ return ad1843_get_gain(ad1843, id);
+}
+
+/* Returns the current recording source */
+
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843)
+{
+ int val = ad1843_read_bits(ad1843, &ad1843_LSS);
+
+ if (val < 0 || val > 2) {
+ val = 2;
+ ad1843_write_multi(ad1843, 2,
+ &ad1843_LSS, val, &ad1843_RSS, val);
+ }
+ return val;
+}
+
+/*
+ * Set recording source.
+ *
+ * Returns newsrc on success, -errno on failure.
+ */
+
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc)
+{
+ if (newsrc < 0 || newsrc > 2)
+ return -EINVAL;
+
+ ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc);
+ return newsrc;
+}
+
+/* Setup ad1843 for D/A conversion. */
+
+void ad1843_setup_dac(struct snd_ad1843 *ad1843,
+ unsigned int id,
+ unsigned int framerate,
+ snd_pcm_format_t fmt,
+ unsigned int channels)
+{
+ int ad_fmt = 0, ad_mode = 0;
+
+ switch (fmt) {
+ case SNDRV_PCM_FORMAT_S8:
+ ad_fmt = 0;
+ break;
+ case SNDRV_PCM_FORMAT_U8:
+ ad_fmt = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ ad_fmt = 1;
+ break;
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ ad_fmt = 2;
+ break;
+ case SNDRV_PCM_FORMAT_A_LAW:
+ ad_fmt = 3;
+ break;
+ default:
+ break;
+ }
+
+ switch (channels) {
+ case 2:
+ ad_mode = 0;
+ break;
+ case 1:
+ ad_mode = 1;
+ break;
+ default:
+ break;
+ }
+
+ if (id) {
+ ad1843_write_bits(ad1843, &ad1843_C2C, framerate);
+ ad1843_write_multi(ad1843, 2,
+ &ad1843_DA2SM, ad_mode,
+ &ad1843_DA2F, ad_fmt);
+ } else {
+ ad1843_write_bits(ad1843, &ad1843_C1C, framerate);
+ ad1843_write_multi(ad1843, 2,
+ &ad1843_DA1SM, ad_mode,
+ &ad1843_DA1F, ad_fmt);
+ }
+}
+
+void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id)
+{
+ if (id)
+ ad1843_write_bits(ad1843, &ad1843_DA2F, 1);
+ else
+ ad1843_write_bits(ad1843, &ad1843_DA1F, 1);
+}
+
+void ad1843_setup_adc(struct snd_ad1843 *ad1843,
+ unsigned int framerate,
+ snd_pcm_format_t fmt,
+ unsigned int channels)
+{
+ int da_fmt = 0;
+
+ switch (fmt) {
+ case SNDRV_PCM_FORMAT_S8: da_fmt = 0; break;
+ case SNDRV_PCM_FORMAT_U8: da_fmt = 0; break;
+ case SNDRV_PCM_FORMAT_S16_LE: da_fmt = 1; break;
+ case SNDRV_PCM_FORMAT_MU_LAW: da_fmt = 2; break;
+ case SNDRV_PCM_FORMAT_A_LAW: da_fmt = 3; break;
+ default: break;
+ }
+
+ ad1843_write_bits(ad1843, &ad1843_C3C, framerate);
+ ad1843_write_multi(ad1843, 2,
+ &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt);
+}
+
+void ad1843_shutdown_adc(struct snd_ad1843 *ad1843)
+{
+ /* nothing to do */
+}
+
+/*
+ * Fully initialize the ad1843. As described in the AD1843 data
+ * sheet, section "START-UP SEQUENCE". The numbered comments are
+ * subsection headings from the data sheet. See the data sheet, pages
+ * 52-54, for more info.
+ *
+ * return 0 on success, -errno on failure. */
+
+int ad1843_init(struct snd_ad1843 *ad1843)
+{
+ unsigned long later;
+
+ if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) {
+ printk(KERN_ERR "ad1843: AD1843 won't initialize\n");
+ return -EIO;
+ }
+
+ ad1843_write_bits(ad1843, &ad1843_SCF, 1);
+
+ /* 4. Put the conversion resources into standby. */
+ ad1843_write_bits(ad1843, &ad1843_PDNI, 0);
+ later = jiffies + msecs_to_jiffies(500);
+
+ while (ad1843_read_bits(ad1843, &ad1843_PDNO)) {
+ if (time_after(jiffies, later)) {
+ printk(KERN_ERR
+ "ad1843: AD1843 won't power up\n");
+ return -EIO;
+ }
+ schedule_timeout_interruptible(5);
+ }
+
+ /* 5. Power up the clock generators and enable clock output pins. */
+ ad1843_write_multi(ad1843, 3,
+ &ad1843_C1EN, 1,
+ &ad1843_C2EN, 1,
+ &ad1843_C3EN, 1);
+
+ /* 6. Configure conversion resources while they are in standby. */
+
+ /* DAC1/2 use clock 1/2 as source, ADC uses clock 3. Always. */
+ ad1843_write_multi(ad1843, 4,
+ &ad1843_DA1C, 1,
+ &ad1843_DA2C, 2,
+ &ad1843_ADLC, 3,
+ &ad1843_ADRC, 3);
+
+ /* 7. Enable conversion resources. */
+ ad1843_write_bits(ad1843, &ad1843_ADTLK, 1);
+ ad1843_write_multi(ad1843, 7,
+ &ad1843_ANAEN, 1,
+ &ad1843_AAMEN, 1,
+ &ad1843_DA1EN, 1,
+ &ad1843_DA2EN, 1,
+ &ad1843_DDMEN, 1,
+ &ad1843_ADLEN, 1,
+ &ad1843_ADREN, 1);
+
+ /* 8. Configure conversion resources while they are enabled. */
+
+ /* set gain to 0 for all channels */
+ ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0);
+ ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0);
+ ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0);
+ ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0);
+ ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0);
+ ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0);
+
+ /* Unmute all channels. */
+ /* DAC1 */
+ ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0);
+ /* DAC2 */
+ ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0);
+
+ /* Set default recording source to Line In and set
+ * mic gain to +20 dB.
+ */
+ ad1843_set_recsrc(ad1843, 2);
+ ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1);
+
+ /* Set Speaker Out level to +/- 4V and unmute it. */
+ ad1843_write_multi(ad1843, 3,
+ &ad1843_HPOS, 1,
+ &ad1843_HPOM, 0,
+ &ad1843_MPOM, 0);
+
+ return 0;
+}
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
new file mode 100644
index 0000000..1881cec
--- /dev/null
+++ b/sound/mips/au1x00.c
@@ -0,0 +1,693 @@
+/*
+ * BRIEF MODULE DESCRIPTION
+ * Driver for AMD Au1000 MIPS Processor, AC'97 Sound Port
+ *
+ * Copyright 2004 Cooper Street Innovations Inc.
+ * Author: Charles Eidsness <charles@cooper-street.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
+ * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
+ * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
+ * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * History:
+ *
+ * 2004-09-09 Charles Eidsness -- Original verion -- based on
+ * sa11xx-uda1341.c ALSA driver and the
+ * au1000.c OSS driver.
+ * 2004-09-09 Matt Porter -- Added support for ALSA 1.0.6
+ *
+ */
+
+#include <linux/ioport.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/ac97_codec.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
+
+MODULE_AUTHOR("Charles Eidsness <charles@cooper-street.com>");
+MODULE_DESCRIPTION("Au1000 AC'97 ALSA Driver");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{AMD,Au1000 AC'97}}");
+
+#define PLAYBACK 0
+#define CAPTURE 1
+#define AC97_SLOT_3 0x01
+#define AC97_SLOT_4 0x02
+#define AC97_SLOT_6 0x08
+#define AC97_CMD_IRQ 31
+#define READ 0
+#define WRITE 1
+#define READ_WAIT 2
+#define RW_DONE 3
+
+struct au1000_period
+{
+ u32 start;
+ u32 relative_end; /*realtive to start of buffer*/
+ struct au1000_period * next;
+};
+
+/*Au1000 AC97 Port Control Reisters*/
+struct au1000_ac97_reg {
+ u32 volatile config;
+ u32 volatile status;
+ u32 volatile data;
+ u32 volatile cmd;
+ u32 volatile cntrl;
+};
+
+struct audio_stream {
+ struct snd_pcm_substream *substream;
+ int dma;
+ spinlock_t dma_lock;
+ struct au1000_period * buffer;
+ unsigned int period_size;
+ unsigned int periods;
+};
+
+struct snd_au1000 {
+ struct snd_card *card;
+ struct au1000_ac97_reg volatile *ac97_ioport;
+
+ struct resource *ac97_res_port;
+ spinlock_t ac97_lock;
+ struct snd_ac97 *ac97;
+
+ struct snd_pcm *pcm;
+ struct audio_stream *stream[2]; /* playback & capture */
+};
+
+/*--------------------------- Local Functions --------------------------------*/
+static void
+au1000_set_ac97_xmit_slots(struct snd_au1000 *au1000, long xmit_slots)
+{
+ u32 volatile ac97_config;
+
+ spin_lock(&au1000->ac97_lock);
+ ac97_config = au1000->ac97_ioport->config;
+ ac97_config = ac97_config & ~AC97C_XMIT_SLOTS_MASK;
+ ac97_config |= (xmit_slots << AC97C_XMIT_SLOTS_BIT);
+ au1000->ac97_ioport->config = ac97_config;
+ spin_unlock(&au1000->ac97_lock);
+}
+
+static void
+au1000_set_ac97_recv_slots(struct snd_au1000 *au1000, long recv_slots)
+{
+ u32 volatile ac97_config;
+
+ spin_lock(&au1000->ac97_lock);
+ ac97_config = au1000->ac97_ioport->config;
+ ac97_config = ac97_config & ~AC97C_RECV_SLOTS_MASK;
+ ac97_config |= (recv_slots << AC97C_RECV_SLOTS_BIT);
+ au1000->ac97_ioport->config = ac97_config;
+ spin_unlock(&au1000->ac97_lock);
+}
+
+
+static void
+au1000_release_dma_link(struct audio_stream *stream)
+{
+ struct au1000_period * pointer;
+ struct au1000_period * pointer_next;
+
+ stream->period_size = 0;
+ stream->periods = 0;
+ pointer = stream->buffer;
+ if (! pointer)
+ return;
+ do {
+ pointer_next = pointer->next;
+ kfree(pointer);
+ pointer = pointer_next;
+ } while (pointer != stream->buffer);
+ stream->buffer = NULL;
+}
+
+static int
+au1000_setup_dma_link(struct audio_stream *stream, unsigned int period_bytes,
+ unsigned int periods)
+{
+ struct snd_pcm_substream *substream = stream->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct au1000_period *pointer;
+ unsigned long dma_start;
+ int i;
+
+ dma_start = virt_to_phys(runtime->dma_area);
+
+ if (stream->period_size == period_bytes &&
+ stream->periods == periods)
+ return 0; /* not changed */
+
+ au1000_release_dma_link(stream);
+
+ stream->period_size = period_bytes;
+ stream->periods = periods;
+
+ stream->buffer = kmalloc(sizeof(struct au1000_period), GFP_KERNEL);
+ if (! stream->buffer)
+ return -ENOMEM;
+ pointer = stream->buffer;
+ for (i = 0; i < periods; i++) {
+ pointer->start = (u32)(dma_start + (i * period_bytes));
+ pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1);
+ if (i < periods - 1) {
+ pointer->next = kmalloc(sizeof(struct au1000_period), GFP_KERNEL);
+ if (! pointer->next) {
+ au1000_release_dma_link(stream);
+ return -ENOMEM;
+ }
+ pointer = pointer->next;
+ }
+ }
+ pointer->next = stream->buffer;
+ return 0;
+}
+
+static void
+au1000_dma_stop(struct audio_stream *stream)
+{
+ if (snd_BUG_ON(!stream->buffer))
+ return;
+ disable_dma(stream->dma);
+}
+
+static void
+au1000_dma_start(struct audio_stream *stream)
+{
+ if (snd_BUG_ON(!stream->buffer))
+ return;
+
+ init_dma(stream->dma);
+ if (get_dma_active_buffer(stream->dma) == 0) {
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ } else {
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ }
+ enable_dma_buffers(stream->dma);
+ start_dma(stream->dma);
+}
+
+static irqreturn_t
+au1000_dma_interrupt(int irq, void *dev_id)
+{
+ struct audio_stream *stream = (struct audio_stream *) dev_id;
+ struct snd_pcm_substream *substream = stream->substream;
+
+ spin_lock(&stream->dma_lock);
+ switch (get_dma_buffer_done(stream->dma)) {
+ case DMA_D0:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done0(stream->dma);
+ set_dma_addr0(stream->dma, stream->buffer->next->start);
+ set_dma_count0(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer0(stream->dma);
+ break;
+ case DMA_D1:
+ stream->buffer = stream->buffer->next;
+ clear_dma_done1(stream->dma);
+ set_dma_addr1(stream->dma, stream->buffer->next->start);
+ set_dma_count1(stream->dma, stream->period_size >> 1);
+ enable_dma_buffer1(stream->dma);
+ break;
+ case (DMA_D0 | DMA_D1):
+ printk(KERN_ERR "DMA %d missed interrupt.\n",stream->dma);
+ au1000_dma_stop(stream);
+ au1000_dma_start(stream);
+ break;
+ case (~DMA_D0 & ~DMA_D1):
+ printk(KERN_ERR "DMA %d empty irq.\n",stream->dma);
+ }
+ spin_unlock(&stream->dma_lock);
+ snd_pcm_period_elapsed(substream);
+ return IRQ_HANDLED;
+}
+
+/*-------------------------- PCM Audio Streams -------------------------------*/
+
+static unsigned int rates[] = {8000, 11025, 16000, 22050};
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static struct snd_pcm_hardware snd_au1000_hw =
+{
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | \
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050),
+ .rate_min = 8000,
+ .rate_max = 22050,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 128*1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 16*1024,
+ .periods_min = 8,
+ .periods_max = 255,
+ .fifo_size = 16,
+};
+
+static int
+snd_au1000_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+ au1000->stream[PLAYBACK]->substream = substream;
+ au1000->stream[PLAYBACK]->buffer = NULL;
+ substream->private_data = au1000->stream[PLAYBACK];
+ substream->runtime->hw = snd_au1000_hw;
+ return (snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0);
+}
+
+static int
+snd_au1000_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+ au1000->stream[CAPTURE]->substream = substream;
+ au1000->stream[CAPTURE]->buffer = NULL;
+ substream->private_data = au1000->stream[CAPTURE];
+ substream->runtime->hw = snd_au1000_hw;
+ return (snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0);
+}
+
+static int
+snd_au1000_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+ au1000->stream[PLAYBACK]->substream = NULL;
+ return 0;
+}
+
+static int
+snd_au1000_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+ au1000->stream[CAPTURE]->substream = NULL;
+ return 0;
+}
+
+static int
+snd_au1000_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct audio_stream *stream = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ return au1000_setup_dma_link(stream,
+ params_period_bytes(hw_params),
+ params_periods(hw_params));
+}
+
+static int
+snd_au1000_hw_free(struct snd_pcm_substream *substream)
+{
+ struct audio_stream *stream = substream->private_data;
+ au1000_release_dma_link(stream);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int
+snd_au1000_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_au1000 *au1000 = substream->pcm->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (runtime->channels == 1)
+ au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_4);
+ else
+ au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4);
+ snd_ac97_set_rate(au1000->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate);
+ return 0;
+}
+
+static int
+snd_au1000_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_au1000 *au1000 = substream->pcm->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (runtime->channels == 1)
+ au1000_set_ac97_recv_slots(au1000, AC97_SLOT_4);
+ else
+ au1000_set_ac97_recv_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4);
+ snd_ac97_set_rate(au1000->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate);
+ return 0;
+}
+
+static int
+snd_au1000_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct audio_stream *stream = substream->private_data;
+ int err = 0;
+
+ spin_lock(&stream->dma_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ au1000_dma_start(stream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ au1000_dma_stop(stream);
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ spin_unlock(&stream->dma_lock);
+ return err;
+}
+
+static snd_pcm_uframes_t
+snd_au1000_pointer(struct snd_pcm_substream *substream)
+{
+ struct audio_stream *stream = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ long location;
+
+ spin_lock(&stream->dma_lock);
+ location = get_dma_residue(stream->dma);
+ spin_unlock(&stream->dma_lock);
+ location = stream->buffer->relative_end - location;
+ if (location == -1)
+ location = 0;
+ return bytes_to_frames(runtime,location);
+}
+
+static struct snd_pcm_ops snd_card_au1000_playback_ops = {
+ .open = snd_au1000_playback_open,
+ .close = snd_au1000_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_au1000_hw_params,
+ .hw_free = snd_au1000_hw_free,
+ .prepare = snd_au1000_playback_prepare,
+ .trigger = snd_au1000_trigger,
+ .pointer = snd_au1000_pointer,
+};
+
+static struct snd_pcm_ops snd_card_au1000_capture_ops = {
+ .open = snd_au1000_capture_open,
+ .close = snd_au1000_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_au1000_hw_params,
+ .hw_free = snd_au1000_hw_free,
+ .prepare = snd_au1000_capture_prepare,
+ .trigger = snd_au1000_trigger,
+ .pointer = snd_au1000_pointer,
+};
+
+static int __devinit
+snd_au1000_pcm_new(struct snd_au1000 *au1000)
+{
+ struct snd_pcm *pcm;
+ int err;
+ unsigned long flags;
+
+ if ((err = snd_pcm_new(au1000->card, "AU1000 AC97 PCM", 0, 1, 1, &pcm)) < 0)
+ return err;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL), 128*1024, 128*1024);
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_card_au1000_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_card_au1000_capture_ops);
+
+ pcm->private_data = au1000;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "Au1000 AC97 PCM");
+
+ spin_lock_init(&au1000->stream[PLAYBACK]->dma_lock);
+ spin_lock_init(&au1000->stream[CAPTURE]->dma_lock);
+
+ flags = claim_dma_lock();
+ if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX,
+ "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED,
+ au1000->stream[PLAYBACK])) < 0) {
+ release_dma_lock(flags);
+ return -EBUSY;
+ }
+ if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX,
+ "AC97 RX", au1000_dma_interrupt, IRQF_DISABLED,
+ au1000->stream[CAPTURE])) < 0){
+ release_dma_lock(flags);
+ return -EBUSY;
+ }
+ /* enable DMA coherency in read/write DMA channels */
+ set_dma_mode(au1000->stream[PLAYBACK]->dma,
+ get_dma_mode(au1000->stream[PLAYBACK]->dma) & ~DMA_NC);
+ set_dma_mode(au1000->stream[CAPTURE]->dma,
+ get_dma_mode(au1000->stream[CAPTURE]->dma) & ~DMA_NC);
+ release_dma_lock(flags);
+ au1000->pcm = pcm;
+ return 0;
+}
+
+
+/*-------------------------- AC97 CODEC Control ------------------------------*/
+
+static unsigned short
+snd_au1000_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct snd_au1000 *au1000 = ac97->private_data;
+ u32 volatile cmd;
+ u16 volatile data;
+ int i;
+
+ spin_lock(&au1000->ac97_lock);
+/* would rather use the interrupt than this polling but it works and I can't
+get the interrupt driven case to work efficiently */
+ for (i = 0; i < 0x5000; i++)
+ if (!(au1000->ac97_ioport->status & AC97C_CP))
+ break;
+ if (i == 0x5000)
+ printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n");
+
+ cmd = (u32) reg & AC97C_INDEX_MASK;
+ cmd |= AC97C_READ;
+ au1000->ac97_ioport->cmd = cmd;
+
+ /* now wait for the data */
+ for (i = 0; i < 0x5000; i++)
+ if (!(au1000->ac97_ioport->status & AC97C_CP))
+ break;
+ if (i == 0x5000) {
+ printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n");
+ return 0;
+ }
+
+ data = au1000->ac97_ioport->cmd & 0xffff;
+ spin_unlock(&au1000->ac97_lock);
+
+ return data;
+
+}
+
+
+static void
+snd_au1000_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
+{
+ struct snd_au1000 *au1000 = ac97->private_data;
+ u32 cmd;
+ int i;
+
+ spin_lock(&au1000->ac97_lock);
+/* would rather use the interrupt than this polling but it works and I can't
+get the interrupt driven case to work efficiently */
+ for (i = 0; i < 0x5000; i++)
+ if (!(au1000->ac97_ioport->status & AC97C_CP))
+ break;
+ if (i == 0x5000)
+ printk(KERN_ERR "au1000 AC97: AC97 command write timeout\n");
+
+ cmd = (u32) reg & AC97C_INDEX_MASK;
+ cmd &= ~AC97C_READ;
+ cmd |= ((u32) val << AC97C_WD_BIT);
+ au1000->ac97_ioport->cmd = cmd;
+ spin_unlock(&au1000->ac97_lock);
+}
+
+static int __devinit
+snd_au1000_ac97_new(struct snd_au1000 *au1000)
+{
+ int err;
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_au1000_ac97_write,
+ .read = snd_au1000_ac97_read,
+ };
+
+ if ((au1000->ac97_res_port = request_mem_region(CPHYSADDR(AC97C_CONFIG),
+ 0x100000, "Au1x00 AC97")) == NULL) {
+ snd_printk(KERN_ERR "ALSA AC97: can't grap AC97 port\n");
+ return -EBUSY;
+ }
+ au1000->ac97_ioport = (struct au1000_ac97_reg *)
+ KSEG1ADDR(au1000->ac97_res_port->start);
+
+ spin_lock_init(&au1000->ac97_lock);
+
+ /* configure pins for AC'97
+ TODO: move to board_setup.c */
+ au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC);
+
+ /* Initialise Au1000's AC'97 Control Block */
+ au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE;
+ udelay(10);
+ au1000->ac97_ioport->cntrl = AC97C_CE;
+ udelay(10);
+
+ /* Initialise External CODEC -- cold reset */
+ au1000->ac97_ioport->config = AC97C_RESET;
+ udelay(10);
+ au1000->ac97_ioport->config = 0x0;
+ mdelay(5);
+
+ /* Initialise AC97 middle-layer */
+ if ((err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus)) < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = au1000;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97)) < 0)
+ return err;
+
+ return 0;
+}
+
+/*------------------------------ Setup / Destroy ----------------------------*/
+
+void
+snd_au1000_free(struct snd_card *card)
+{
+ struct snd_au1000 *au1000 = card->private_data;
+
+ if (au1000->ac97_res_port) {
+ /* put internal AC97 block into reset */
+ au1000->ac97_ioport->cntrl = AC97C_RS;
+ au1000->ac97_ioport = NULL;
+ release_and_free_resource(au1000->ac97_res_port);
+ }
+
+ if (au1000->stream[PLAYBACK]) {
+ if (au1000->stream[PLAYBACK]->dma >= 0)
+ free_au1000_dma(au1000->stream[PLAYBACK]->dma);
+ kfree(au1000->stream[PLAYBACK]);
+ }
+
+ if (au1000->stream[CAPTURE]) {
+ if (au1000->stream[CAPTURE]->dma >= 0)
+ free_au1000_dma(au1000->stream[CAPTURE]->dma);
+ kfree(au1000->stream[CAPTURE]);
+ }
+}
+
+
+static struct snd_card *au1000_card;
+
+static int __init
+au1000_init(void)
+{
+ int err;
+ struct snd_card *card;
+ struct snd_au1000 *au1000;
+
+ card = snd_card_new(-1, "AC97", THIS_MODULE, sizeof(struct snd_au1000));
+ if (card == NULL)
+ return -ENOMEM;
+
+ card->private_free = snd_au1000_free;
+ au1000 = card->private_data;
+ au1000->card = card;
+
+ au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL);
+ au1000->stream[CAPTURE ] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL);
+ /* so that snd_au1000_free will work as intended */
+ au1000->ac97_res_port = NULL;
+ if (au1000->stream[PLAYBACK])
+ au1000->stream[PLAYBACK]->dma = -1;
+ if (au1000->stream[CAPTURE ])
+ au1000->stream[CAPTURE ]->dma = -1;
+
+ if (au1000->stream[PLAYBACK] == NULL ||
+ au1000->stream[CAPTURE ] == NULL) {
+ snd_card_free(card);
+ return -ENOMEM;
+ }
+
+ if ((err = snd_au1000_ac97_new(au1000)) < 0 ) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_au1000_pcm_new(au1000)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "Au1000-AC97");
+ strcpy(card->shortname, "AMD Au1000-AC97");
+ sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver");
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ printk( KERN_INFO "ALSA AC97: Driver Initialized\n" );
+ au1000_card = card;
+ return 0;
+}
+
+static void __exit au1000_exit(void)
+{
+ snd_card_free(au1000_card);
+}
+
+module_init(au1000_init);
+module_exit(au1000_exit);
+
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
new file mode 100644
index 0000000..db495be
--- /dev/null
+++ b/sound/mips/hal2.c
@@ -0,0 +1,947 @@
+/*
+ * Driver for A2 audio system used in SGI machines
+ * Copyright (c) 2008 Thomas Bogendoerfer <tsbogend@alpha.fanken.de>
+ *
+ * Based on OSS code from Ladislav Michl <ladis@linux-mips.org>, which
+ * was based on code from Ulf Carlsson
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include <asm/sgi/hpc3.h>
+#include <asm/sgi/ip22.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm-indirect.h>
+#include <sound/initval.h>
+
+#include "hal2.h"
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SGI HAL2 soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SGI HAL2 soundcard.");
+MODULE_DESCRIPTION("ALSA driver for SGI HAL2 audio");
+MODULE_AUTHOR("Thomas Bogendoerfer");
+MODULE_LICENSE("GPL");
+
+
+#define H2_BLOCK_SIZE 1024
+#define H2_BUF_SIZE 16384
+
+struct hal2_pbus {
+ struct hpc3_pbus_dmacregs *pbus;
+ int pbusnr;
+ unsigned int ctrl; /* Current state of pbus->pbdma_ctrl */
+};
+
+struct hal2_desc {
+ struct hpc_dma_desc desc;
+ u32 pad; /* padding */
+};
+
+struct hal2_codec {
+ struct snd_pcm_indirect pcm_indirect;
+ struct snd_pcm_substream *substream;
+
+ unsigned char *buffer;
+ dma_addr_t buffer_dma;
+ struct hal2_desc *desc;
+ dma_addr_t desc_dma;
+ int desc_count;
+ struct hal2_pbus pbus;
+ int voices; /* mono/stereo */
+ unsigned int sample_rate;
+ unsigned int master; /* Master frequency */
+ unsigned short mod; /* MOD value */
+ unsigned short inc; /* INC value */
+};
+
+#define H2_MIX_OUTPUT_ATT 0
+#define H2_MIX_INPUT_GAIN 1
+
+struct snd_hal2 {
+ struct snd_card *card;
+
+ struct hal2_ctl_regs *ctl_regs; /* HAL2 ctl registers */
+ struct hal2_aes_regs *aes_regs; /* HAL2 aes registers */
+ struct hal2_vol_regs *vol_regs; /* HAL2 vol registers */
+ struct hal2_syn_regs *syn_regs; /* HAL2 syn registers */
+
+ struct hal2_codec dac;
+ struct hal2_codec adc;
+};
+
+#define H2_INDIRECT_WAIT(regs) while (hal2_read(&regs->isr) & H2_ISR_TSTATUS);
+
+#define H2_READ_ADDR(addr) (addr | (1<<7))
+#define H2_WRITE_ADDR(addr) (addr)
+
+static inline u32 hal2_read(u32 *reg)
+{
+ return __raw_readl(reg);
+}
+
+static inline void hal2_write(u32 val, u32 *reg)
+{
+ __raw_writel(val, reg);
+}
+
+
+static u32 hal2_i_read32(struct snd_hal2 *hal2, u16 addr)
+{
+ u32 ret;
+ struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+ hal2_write(H2_READ_ADDR(addr), &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+ ret = hal2_read(&regs->idr0) & 0xffff;
+ hal2_write(H2_READ_ADDR(addr) | 0x1, &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+ ret |= (hal2_read(&regs->idr0) & 0xffff) << 16;
+ return ret;
+}
+
+static void hal2_i_write16(struct snd_hal2 *hal2, u16 addr, u16 val)
+{
+ struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+ hal2_write(val, &regs->idr0);
+ hal2_write(0, &regs->idr1);
+ hal2_write(0, &regs->idr2);
+ hal2_write(0, &regs->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_write32(struct snd_hal2 *hal2, u16 addr, u32 val)
+{
+ struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+ hal2_write(val & 0xffff, &regs->idr0);
+ hal2_write(val >> 16, &regs->idr1);
+ hal2_write(0, &regs->idr2);
+ hal2_write(0, &regs->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_setbit16(struct snd_hal2 *hal2, u16 addr, u16 bit)
+{
+ struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+ hal2_write(H2_READ_ADDR(addr), &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+ hal2_write((hal2_read(&regs->idr0) & 0xffff) | bit, &regs->idr0);
+ hal2_write(0, &regs->idr1);
+ hal2_write(0, &regs->idr2);
+ hal2_write(0, &regs->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_clearbit16(struct snd_hal2 *hal2, u16 addr, u16 bit)
+{
+ struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+ hal2_write(H2_READ_ADDR(addr), &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+ hal2_write((hal2_read(&regs->idr0) & 0xffff) & ~bit, &regs->idr0);
+ hal2_write(0, &regs->idr1);
+ hal2_write(0, &regs->idr2);
+ hal2_write(0, &regs->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+ H2_INDIRECT_WAIT(regs);
+}
+
+static int hal2_gain_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ switch ((int)kcontrol->private_value) {
+ case H2_MIX_OUTPUT_ATT:
+ uinfo->value.integer.max = 31;
+ break;
+ case H2_MIX_INPUT_GAIN:
+ uinfo->value.integer.max = 15;
+ break;
+ }
+ return 0;
+}
+
+static int hal2_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol);
+ u32 tmp;
+ int l, r;
+
+ switch ((int)kcontrol->private_value) {
+ case H2_MIX_OUTPUT_ATT:
+ tmp = hal2_i_read32(hal2, H2I_DAC_C2);
+ if (tmp & H2I_C2_MUTE) {
+ l = 0;
+ r = 0;
+ } else {
+ l = 31 - ((tmp >> H2I_C2_L_ATT_SHIFT) & 31);
+ r = 31 - ((tmp >> H2I_C2_R_ATT_SHIFT) & 31);
+ }
+ break;
+ case H2_MIX_INPUT_GAIN:
+ tmp = hal2_i_read32(hal2, H2I_ADC_C2);
+ l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15;
+ r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15;
+ break;
+ }
+ ucontrol->value.integer.value[0] = l;
+ ucontrol->value.integer.value[1] = r;
+
+ return 0;
+}
+
+static int hal2_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol);
+ u32 old, new;
+ int l, r;
+
+ l = ucontrol->value.integer.value[0];
+ r = ucontrol->value.integer.value[1];
+
+ switch ((int)kcontrol->private_value) {
+ case H2_MIX_OUTPUT_ATT:
+ old = hal2_i_read32(hal2, H2I_DAC_C2);
+ new = old & ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE);
+ if (l | r) {
+ l = 31 - l;
+ r = 31 - r;
+ new |= (l << H2I_C2_L_ATT_SHIFT);
+ new |= (r << H2I_C2_R_ATT_SHIFT);
+ } else
+ new |= H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE;
+ hal2_i_write32(hal2, H2I_DAC_C2, new);
+ break;
+ case H2_MIX_INPUT_GAIN:
+ old = hal2_i_read32(hal2, H2I_ADC_C2);
+ new = old & ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M);
+ new |= (l << H2I_C2_L_GAIN_SHIFT);
+ new |= (r << H2I_C2_R_GAIN_SHIFT);
+ hal2_i_write32(hal2, H2I_ADC_C2, new);
+ break;
+ }
+ return old != new;
+}
+
+static struct snd_kcontrol_new hal2_ctrl_headphone __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = H2_MIX_OUTPUT_ATT,
+ .info = hal2_gain_info,
+ .get = hal2_gain_get,
+ .put = hal2_gain_put,
+};
+
+static struct snd_kcontrol_new hal2_ctrl_mic __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Capture Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = H2_MIX_INPUT_GAIN,
+ .info = hal2_gain_info,
+ .get = hal2_gain_get,
+ .put = hal2_gain_put,
+};
+
+static int __devinit hal2_mixer_create(struct snd_hal2 *hal2)
+{
+ int err;
+
+ /* mute DAC */
+ hal2_i_write32(hal2, H2I_DAC_C2,
+ H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE);
+ /* mute ADC */
+ hal2_i_write32(hal2, H2I_ADC_C2, 0);
+
+ err = snd_ctl_add(hal2->card,
+ snd_ctl_new1(&hal2_ctrl_headphone, hal2));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(hal2->card,
+ snd_ctl_new1(&hal2_ctrl_mic, hal2));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static irqreturn_t hal2_interrupt(int irq, void *dev_id)
+{
+ struct snd_hal2 *hal2 = dev_id;
+ irqreturn_t ret = IRQ_NONE;
+
+ /* decide what caused this interrupt */
+ if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) {
+ snd_pcm_period_elapsed(hal2->dac.substream);
+ ret = IRQ_HANDLED;
+ }
+ if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) {
+ snd_pcm_period_elapsed(hal2->adc.substream);
+ ret = IRQ_HANDLED;
+ }
+ return ret;
+}
+
+static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate)
+{
+ unsigned short mod;
+
+ if (44100 % rate < 48000 % rate) {
+ mod = 4 * 44100 / rate;
+ codec->master = 44100;
+ } else {
+ mod = 4 * 48000 / rate;
+ codec->master = 48000;
+ }
+
+ codec->inc = 4;
+ codec->mod = mod;
+ rate = 4 * codec->master / mod;
+
+ return rate;
+}
+
+static void hal2_set_dac_rate(struct snd_hal2 *hal2)
+{
+ unsigned int master = hal2->dac.master;
+ int inc = hal2->dac.inc;
+ int mod = hal2->dac.mod;
+
+ hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0);
+ hal2_i_write32(hal2, H2I_BRES1_C2,
+ ((0xffff & (inc - mod - 1)) << 16) | inc);
+}
+
+static void hal2_set_adc_rate(struct snd_hal2 *hal2)
+{
+ unsigned int master = hal2->adc.master;
+ int inc = hal2->adc.inc;
+ int mod = hal2->adc.mod;
+
+ hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0);
+ hal2_i_write32(hal2, H2I_BRES2_C2,
+ ((0xffff & (inc - mod - 1)) << 16) | inc);
+}
+
+static void hal2_setup_dac(struct snd_hal2 *hal2)
+{
+ unsigned int fifobeg, fifoend, highwater, sample_size;
+ struct hal2_pbus *pbus = &hal2->dac.pbus;
+
+ /* Now we set up some PBUS information. The PBUS needs information about
+ * what portion of the fifo it will use. If it's receiving or
+ * transmitting, and finally whether the stream is little endian or big
+ * endian. The information is written later, on the start call.
+ */
+ sample_size = 2 * hal2->dac.voices;
+ /* Fifo should be set to hold exactly four samples. Highwater mark
+ * should be set to two samples. */
+ highwater = (sample_size * 2) >> 1; /* halfwords */
+ fifobeg = 0; /* playback is first */
+ fifoend = (sample_size * 4) >> 3; /* doublewords */
+ pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD |
+ (highwater << 8) | (fifobeg << 16) | (fifoend << 24);
+ /* We disable everything before we do anything at all */
+ pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+ hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX);
+ /* Setup the HAL2 for playback */
+ hal2_set_dac_rate(hal2);
+ /* Set endianess */
+ hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX);
+ /* Set DMA bus */
+ hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr));
+ /* We are using 1st Bresenham clock generator for playback */
+ hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT)
+ | (1 << H2I_C1_CLKID_SHIFT)
+ | (hal2->dac.voices << H2I_C1_DATAT_SHIFT));
+}
+
+static void hal2_setup_adc(struct snd_hal2 *hal2)
+{
+ unsigned int fifobeg, fifoend, highwater, sample_size;
+ struct hal2_pbus *pbus = &hal2->adc.pbus;
+
+ sample_size = 2 * hal2->adc.voices;
+ highwater = (sample_size * 2) >> 1; /* halfwords */
+ fifobeg = (4 * 4) >> 3; /* record is second */
+ fifoend = (4 * 4 + sample_size * 4) >> 3; /* doublewords */
+ pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD |
+ (highwater << 8) | (fifobeg << 16) | (fifoend << 24);
+ pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+ hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR);
+ /* Setup the HAL2 for record */
+ hal2_set_adc_rate(hal2);
+ /* Set endianess */
+ hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR);
+ /* Set DMA bus */
+ hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr));
+ /* We are using 2nd Bresenham clock generator for record */
+ hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT)
+ | (2 << H2I_C1_CLKID_SHIFT)
+ | (hal2->adc.voices << H2I_C1_DATAT_SHIFT));
+}
+
+static void hal2_start_dac(struct snd_hal2 *hal2)
+{
+ struct hal2_pbus *pbus = &hal2->dac.pbus;
+
+ pbus->pbus->pbdma_dptr = hal2->dac.desc_dma;
+ pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT;
+ /* enable DAC */
+ hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX);
+}
+
+static void hal2_start_adc(struct snd_hal2 *hal2)
+{
+ struct hal2_pbus *pbus = &hal2->adc.pbus;
+
+ pbus->pbus->pbdma_dptr = hal2->adc.desc_dma;
+ pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT;
+ /* enable ADC */
+ hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR);
+}
+
+static inline void hal2_stop_dac(struct snd_hal2 *hal2)
+{
+ hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+ /* The HAL2 itself may remain enabled safely */
+}
+
+static inline void hal2_stop_adc(struct snd_hal2 *hal2)
+{
+ hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+}
+
+static int hal2_alloc_dmabuf(struct hal2_codec *codec)
+{
+ struct hal2_desc *desc;
+ dma_addr_t desc_dma, buffer_dma;
+ int count = H2_BUF_SIZE / H2_BLOCK_SIZE;
+ int i;
+
+ codec->buffer = dma_alloc_noncoherent(NULL, H2_BUF_SIZE,
+ &buffer_dma, GFP_KERNEL);
+ if (!codec->buffer)
+ return -ENOMEM;
+ desc = dma_alloc_noncoherent(NULL, count * sizeof(struct hal2_desc),
+ &desc_dma, GFP_KERNEL);
+ if (!desc) {
+ dma_free_noncoherent(NULL, H2_BUF_SIZE,
+ codec->buffer, buffer_dma);
+ return -ENOMEM;
+ }
+ codec->buffer_dma = buffer_dma;
+ codec->desc_dma = desc_dma;
+ codec->desc = desc;
+ for (i = 0; i < count; i++) {
+ desc->desc.pbuf = buffer_dma + i * H2_BLOCK_SIZE;
+ desc->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE;
+ desc->desc.pnext = (i == count - 1) ?
+ desc_dma : desc_dma + (i + 1) * sizeof(struct hal2_desc);
+ desc++;
+ }
+ dma_cache_sync(NULL, codec->desc, count * sizeof(struct hal2_desc),
+ DMA_TO_DEVICE);
+ codec->desc_count = count;
+ return 0;
+}
+
+static void hal2_free_dmabuf(struct hal2_codec *codec)
+{
+ dma_free_noncoherent(NULL, codec->desc_count * sizeof(struct hal2_desc),
+ codec->desc, codec->desc_dma);
+ dma_free_noncoherent(NULL, H2_BUF_SIZE, codec->buffer,
+ codec->buffer_dma);
+}
+
+static struct snd_pcm_hardware hal2_pcm_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER),
+ .formats = SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 65536,
+ .periods_min = 2,
+ .periods_max = 1024,
+};
+
+static int hal2_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int hal2_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int hal2_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ int err;
+
+ runtime->hw = hal2_pcm_hw;
+
+ err = hal2_alloc_dmabuf(&hal2->dac);
+ if (err)
+ return err;
+ return 0;
+}
+
+static int hal2_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+ hal2_free_dmabuf(&hal2->dac);
+ return 0;
+}
+
+static int hal2_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hal2_codec *dac = &hal2->dac;
+
+ dac->voices = runtime->channels;
+ dac->sample_rate = hal2_compute_rate(dac, runtime->rate);
+ memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect));
+ dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+ dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ dac->substream = substream;
+ hal2_setup_dac(hal2);
+ return 0;
+}
+
+static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma;
+ hal2->dac.pcm_indirect.hw_data = 0;
+ substream->ops->ack(substream);
+ hal2_start_dac(hal2);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ hal2_stop_dac(hal2);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t
+hal2_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ struct hal2_codec *dac = &hal2->dac;
+
+ return snd_pcm_indirect_playback_pointer(substream, &dac->pcm_indirect,
+ dac->pbus.pbus->pbdma_bptr);
+}
+
+static void hal2_playback_transfer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ unsigned char *buf = hal2->dac.buffer + rec->hw_data;
+
+ memcpy(buf, substream->runtime->dma_area + rec->sw_data, bytes);
+ dma_cache_sync(NULL, buf, bytes, DMA_TO_DEVICE);
+
+}
+
+static int hal2_playback_ack(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ struct hal2_codec *dac = &hal2->dac;
+
+ dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+ snd_pcm_indirect_playback_transfer(substream,
+ &dac->pcm_indirect,
+ hal2_playback_transfer);
+ return 0;
+}
+
+static int hal2_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ struct hal2_codec *adc = &hal2->adc;
+ int err;
+
+ runtime->hw = hal2_pcm_hw;
+
+ err = hal2_alloc_dmabuf(adc);
+ if (err)
+ return err;
+ return 0;
+}
+
+static int hal2_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+ hal2_free_dmabuf(&hal2->adc);
+ return 0;
+}
+
+static int hal2_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hal2_codec *adc = &hal2->adc;
+
+ adc->voices = runtime->channels;
+ adc->sample_rate = hal2_compute_rate(adc, runtime->rate);
+ memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect));
+ adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+ adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+ adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ adc->substream = substream;
+ hal2_setup_adc(hal2);
+ return 0;
+}
+
+static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma;
+ hal2->adc.pcm_indirect.hw_data = 0;
+ printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma);
+ hal2_start_adc(hal2);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ hal2_stop_adc(hal2);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t
+hal2_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ struct hal2_codec *adc = &hal2->adc;
+
+ return snd_pcm_indirect_capture_pointer(substream, &adc->pcm_indirect,
+ adc->pbus.pbus->pbdma_bptr);
+}
+
+static void hal2_capture_transfer(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ unsigned char *buf = hal2->adc.buffer + rec->hw_data;
+
+ dma_cache_sync(NULL, buf, bytes, DMA_FROM_DEVICE);
+ memcpy(substream->runtime->dma_area + rec->sw_data, buf, bytes);
+}
+
+static int hal2_capture_ack(struct snd_pcm_substream *substream)
+{
+ struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+ struct hal2_codec *adc = &hal2->adc;
+
+ snd_pcm_indirect_capture_transfer(substream,
+ &adc->pcm_indirect,
+ hal2_capture_transfer);
+ return 0;
+}
+
+static struct snd_pcm_ops hal2_playback_ops = {
+ .open = hal2_playback_open,
+ .close = hal2_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = hal2_pcm_hw_params,
+ .hw_free = hal2_pcm_hw_free,
+ .prepare = hal2_playback_prepare,
+ .trigger = hal2_playback_trigger,
+ .pointer = hal2_playback_pointer,
+ .ack = hal2_playback_ack,
+};
+
+static struct snd_pcm_ops hal2_capture_ops = {
+ .open = hal2_capture_open,
+ .close = hal2_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = hal2_pcm_hw_params,
+ .hw_free = hal2_pcm_hw_free,
+ .prepare = hal2_capture_prepare,
+ .trigger = hal2_capture_trigger,
+ .pointer = hal2_capture_pointer,
+ .ack = hal2_capture_ack,
+};
+
+static int __devinit hal2_pcm_create(struct snd_hal2 *hal2)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ /* create first pcm device with one outputs and one input */
+ err = snd_pcm_new(hal2->card, "SGI HAL2 Audio", 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = hal2;
+ strcpy(pcm->name, "SGI HAL2");
+
+ /* set operators */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &hal2_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &hal2_capture_ops);
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 0, 1024 * 1024);
+
+ return 0;
+}
+
+static int hal2_dev_free(struct snd_device *device)
+{
+ struct snd_hal2 *hal2 = device->device_data;
+
+ free_irq(SGI_HPCDMA_IRQ, hal2);
+ kfree(hal2);
+ return 0;
+}
+
+static struct snd_device_ops hal2_ops = {
+ .dev_free = hal2_dev_free,
+};
+
+static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3,
+ int index)
+{
+ codec->pbus.pbusnr = index;
+ codec->pbus.pbus = &hpc3->pbdma[index];
+}
+
+static int hal2_detect(struct snd_hal2 *hal2)
+{
+ unsigned short board, major, minor;
+ unsigned short rev;
+
+ /* reset HAL2 */
+ hal2_write(0, &hal2->ctl_regs->isr);
+
+ /* release reset */
+ hal2_write(H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N,
+ &hal2->ctl_regs->isr);
+
+
+ hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE);
+ rev = hal2_read(&hal2->ctl_regs->rev);
+ if (rev & H2_REV_AUDIO_PRESENT)
+ return -ENODEV;
+
+ board = (rev & H2_REV_BOARD_M) >> 12;
+ major = (rev & H2_REV_MAJOR_CHIP_M) >> 4;
+ minor = (rev & H2_REV_MINOR_CHIP_M);
+
+ printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n",
+ board, major, minor);
+
+ return 0;
+}
+
+static int hal2_create(struct snd_card *card, struct snd_hal2 **rchip)
+{
+ struct snd_hal2 *hal2;
+ struct hpc3_regs *hpc3 = hpc3c0;
+ int err;
+
+ hal2 = kzalloc(sizeof(struct snd_hal2), GFP_KERNEL);
+ if (!hal2)
+ return -ENOMEM;
+
+ hal2->card = card;
+
+ if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED,
+ "SGI HAL2", hal2)) {
+ printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ);
+ kfree(hal2);
+ return -EAGAIN;
+ }
+
+ hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0];
+ hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1];
+ hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2];
+ hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3];
+
+ if (hal2_detect(hal2) < 0) {
+ kfree(hal2);
+ return -ENODEV;
+ }
+
+ hal2_init_codec(&hal2->dac, hpc3, 0);
+ hal2_init_codec(&hal2->adc, hpc3, 1);
+
+ /*
+ * All DMA channel interfaces in HAL2 are designed to operate with
+ * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles
+ * in D5. HAL2 is a 16-bit device which can accept both big and little
+ * endian format. It assumes that even address bytes are on high
+ * portion of PBUS (15:8) and assumes that HPC3 is programmed to
+ * accept a live (unsynchronized) version of P_DREQ_N from HAL2.
+ */
+#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \
+ (2 << HPC3_DMACFG_D4R_SHIFT) | \
+ (2 << HPC3_DMACFG_D5R_SHIFT) | \
+ (0 << HPC3_DMACFG_D3W_SHIFT) | \
+ (2 << HPC3_DMACFG_D4W_SHIFT) | \
+ (2 << HPC3_DMACFG_D5W_SHIFT) | \
+ HPC3_DMACFG_DS16 | \
+ HPC3_DMACFG_EVENHI | \
+ HPC3_DMACFG_RTIME | \
+ (8 << HPC3_DMACFG_BURST_SHIFT) | \
+ HPC3_DMACFG_DRQLIVE)
+ /*
+ * Ignore what's mentioned in the specification and write value which
+ * works in The Real World (TM)
+ */
+ hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844;
+ hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844;
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, hal2, &hal2_ops);
+ if (err < 0) {
+ free_irq(SGI_HPCDMA_IRQ, hal2);
+ kfree(hal2);
+ return err;
+ }
+ *rchip = hal2;
+ return 0;
+}
+
+static int __devinit hal2_probe(struct platform_device *pdev)
+{
+ struct snd_card *card;
+ struct snd_hal2 *chip;
+ int err;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ err = hal2_create(card, &chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_card_set_dev(card, &pdev->dev);
+
+ err = hal2_pcm_create(chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = hal2_mixer_create(chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "SGI HAL2 Audio");
+ strcpy(card->shortname, "SGI HAL2 Audio");
+ sprintf(card->longname, "%s irq %i",
+ card->shortname,
+ SGI_HPCDMA_IRQ);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ platform_set_drvdata(pdev, card);
+ return 0;
+}
+
+static int __exit hal2_remove(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ snd_card_free(card);
+ platform_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+static struct platform_driver hal2_driver = {
+ .probe = hal2_probe,
+ .remove = __devexit_p(hal2_remove),
+ .driver = {
+ .name = "sgihal2",
+ .owner = THIS_MODULE,
+ }
+};
+
+static int __init alsa_card_hal2_init(void)
+{
+ return platform_driver_register(&hal2_driver);
+}
+
+static void __exit alsa_card_hal2_exit(void)
+{
+ platform_driver_unregister(&hal2_driver);
+}
+
+module_init(alsa_card_hal2_init);
+module_exit(alsa_card_hal2_exit);
diff --git a/sound/mips/hal2.h b/sound/mips/hal2.h
new file mode 100644
index 0000000..f19828b
--- /dev/null
+++ b/sound/mips/hal2.h
@@ -0,0 +1,245 @@
+#ifndef __HAL2_H
+#define __HAL2_H
+
+/*
+ * Driver for HAL2 sound processors
+ * Copyright (c) 1999 Ulf Carlsson <ulfc@bun.falkenberg.se>
+ * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis@linux-mips.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/types.h>
+
+/* Indirect status register */
+
+#define H2_ISR_TSTATUS 0x01 /* RO: transaction status 1=busy */
+#define H2_ISR_USTATUS 0x02 /* RO: utime status bit 1=armed */
+#define H2_ISR_QUAD_MODE 0x04 /* codec mode 0=indigo 1=quad */
+#define H2_ISR_GLOBAL_RESET_N 0x08 /* chip global reset 0=reset */
+#define H2_ISR_CODEC_RESET_N 0x10 /* codec/synth reset 0=reset */
+
+/* Revision register */
+
+#define H2_REV_AUDIO_PRESENT 0x8000 /* RO: audio present 0=present */
+#define H2_REV_BOARD_M 0x7000 /* RO: bits 14:12, board revision */
+#define H2_REV_MAJOR_CHIP_M 0x00F0 /* RO: bits 7:4, major chip revision */
+#define H2_REV_MINOR_CHIP_M 0x000F /* RO: bits 3:0, minor chip revision */
+
+/* Indirect address register */
+
+/*
+ * Address of indirect internal register to be accessed. A write to this
+ * register initiates read or write access to the indirect registers in the
+ * HAL2. Note that there af four indirect data registers for write access to
+ * registers larger than 16 byte.
+ */
+
+#define H2_IAR_TYPE_M 0xF000 /* bits 15:12, type of functional */
+ /* block the register resides in */
+ /* 1=DMA Port */
+ /* 9=Global DMA Control */
+ /* 2=Bresenham */
+ /* 3=Unix Timer */
+#define H2_IAR_NUM_M 0x0F00 /* bits 11:8 instance of the */
+ /* blockin which the indirect */
+ /* register resides */
+ /* If IAR_TYPE_M=DMA Port: */
+ /* 1=Synth In */
+ /* 2=AES In */
+ /* 3=AES Out */
+ /* 4=DAC Out */
+ /* 5=ADC Out */
+ /* 6=Synth Control */
+ /* If IAR_TYPE_M=Global DMA Control: */
+ /* 1=Control */
+ /* If IAR_TYPE_M=Bresenham: */
+ /* 1=Bresenham Clock Gen 1 */
+ /* 2=Bresenham Clock Gen 2 */
+ /* 3=Bresenham Clock Gen 3 */
+ /* If IAR_TYPE_M=Unix Timer: */
+ /* 1=Unix Timer */
+#define H2_IAR_ACCESS_SELECT 0x0080 /* 1=read 0=write */
+#define H2_IAR_PARAM 0x000C /* Parameter Select */
+#define H2_IAR_RB_INDEX_M 0x0003 /* Read Back Index */
+ /* 00:word0 */
+ /* 01:word1 */
+ /* 10:word2 */
+ /* 11:word3 */
+/*
+ * HAL2 internal addressing
+ *
+ * The HAL2 has "indirect registers" (idr) which are accessed by writing to the
+ * Indirect Data registers. Write the address to the Indirect Address register
+ * to transfer the data.
+ *
+ * We define the H2IR_* to the read address and H2IW_* to the write address and
+ * H2I_* to be fields in whatever register is referred to.
+ *
+ * When we write to indirect registers which are larger than one word (16 bit)
+ * we have to fill more than one indirect register before writing. When we read
+ * back however we have to read several times, each time with different Read
+ * Back Indexes (there are defs for doing this easily).
+ */
+
+/*
+ * Relay Control
+ */
+#define H2I_RELAY_C 0x9100
+#define H2I_RELAY_C_STATE 0x01 /* state of RELAY pin signal */
+
+/* DMA port enable */
+
+#define H2I_DMA_PORT_EN 0x9104
+#define H2I_DMA_PORT_EN_SY_IN 0x01 /* Synth_in DMA port */
+#define H2I_DMA_PORT_EN_AESRX 0x02 /* AES receiver DMA port */
+#define H2I_DMA_PORT_EN_AESTX 0x04 /* AES transmitter DMA port */
+#define H2I_DMA_PORT_EN_CODECTX 0x08 /* CODEC transmit DMA port */
+#define H2I_DMA_PORT_EN_CODECR 0x10 /* CODEC receive DMA port */
+
+#define H2I_DMA_END 0x9108 /* global dma endian select */
+#define H2I_DMA_END_SY_IN 0x01 /* Synth_in DMA port */
+#define H2I_DMA_END_AESRX 0x02 /* AES receiver DMA port */
+#define H2I_DMA_END_AESTX 0x04 /* AES transmitter DMA port */
+#define H2I_DMA_END_CODECTX 0x08 /* CODEC transmit DMA port */
+#define H2I_DMA_END_CODECR 0x10 /* CODEC receive DMA port */
+ /* 0=b_end 1=l_end */
+
+#define H2I_DMA_DRV 0x910C /* global PBUS DMA enable */
+
+#define H2I_SYNTH_C 0x1104 /* Synth DMA control */
+
+#define H2I_AESRX_C 0x1204 /* AES RX dma control */
+
+#define H2I_C_TS_EN 0x20 /* Timestamp enable */
+#define H2I_C_TS_FRMT 0x40 /* Timestamp format */
+#define H2I_C_NAUDIO 0x80 /* Sign extend */
+
+/* AESRX CTL, 16 bit */
+
+#define H2I_AESTX_C 0x1304 /* AES TX DMA control */
+#define H2I_AESTX_C_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */
+#define H2I_AESTX_C_CLKID_M 0x18
+#define H2I_AESTX_C_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */
+#define H2I_AESTX_C_DATAT_M 0x300
+
+/* CODEC registers */
+
+#define H2I_DAC_C1 0x1404 /* DAC DMA control, 16 bit */
+#define H2I_DAC_C2 0x1408 /* DAC DMA control, 32 bit */
+#define H2I_ADC_C1 0x1504 /* ADC DMA control, 16 bit */
+#define H2I_ADC_C2 0x1508 /* ADC DMA control, 32 bit */
+
+/* Bits in CTL1 register */
+
+#define H2I_C1_DMA_SHIFT 0 /* DMA channel */
+#define H2I_C1_DMA_M 0x7
+#define H2I_C1_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */
+#define H2I_C1_CLKID_M 0x18
+#define H2I_C1_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */
+#define H2I_C1_DATAT_M 0x300
+
+/* Bits in CTL2 register */
+
+#define H2I_C2_R_GAIN_SHIFT 0 /* right a/d input gain */
+#define H2I_C2_R_GAIN_M 0xf
+#define H2I_C2_L_GAIN_SHIFT 4 /* left a/d input gain */
+#define H2I_C2_L_GAIN_M 0xf0
+#define H2I_C2_R_SEL 0x100 /* right input select */
+#define H2I_C2_L_SEL 0x200 /* left input select */
+#define H2I_C2_MUTE 0x400 /* mute */
+#define H2I_C2_DO1 0x00010000 /* digital output port bit 0 */
+#define H2I_C2_DO2 0x00020000 /* digital output port bit 1 */
+#define H2I_C2_R_ATT_SHIFT 18 /* right d/a output - */
+#define H2I_C2_R_ATT_M 0x007c0000 /* attenuation */
+#define H2I_C2_L_ATT_SHIFT 23 /* left d/a output - */
+#define H2I_C2_L_ATT_M 0x0f800000 /* attenuation */
+
+#define H2I_SYNTH_MAP_C 0x1104 /* synth dma handshake ctrl */
+
+/* Clock generator CTL 1, 16 bit */
+
+#define H2I_BRES1_C1 0x2104
+#define H2I_BRES2_C1 0x2204
+#define H2I_BRES3_C1 0x2304
+
+#define H2I_BRES_C1_SHIFT 0 /* 0=48.0 1=44.1 2=aes_rx */
+#define H2I_BRES_C1_M 0x03
+
+/* Clock generator CTL 2, 32 bit */
+
+#define H2I_BRES1_C2 0x2108
+#define H2I_BRES2_C2 0x2208
+#define H2I_BRES3_C2 0x2308
+
+#define H2I_BRES_C2_INC_SHIFT 0 /* increment value */
+#define H2I_BRES_C2_INC_M 0xffff
+#define H2I_BRES_C2_MOD_SHIFT 16 /* modcontrol value */
+#define H2I_BRES_C2_MOD_M 0xffff0000 /* modctrl=0xffff&(modinc-1) */
+
+/* Unix timer, 64 bit */
+
+#define H2I_UTIME 0x3104
+#define H2I_UTIME_0_LD 0xffff /* microseconds, LSB's */
+#define H2I_UTIME_1_LD0 0x0f /* microseconds, MSB's */
+#define H2I_UTIME_1_LD1 0xf0 /* tenths of microseconds */
+#define H2I_UTIME_2_LD 0xffff /* seconds, LSB's */
+#define H2I_UTIME_3_LD 0xffff /* seconds, MSB's */
+
+struct hal2_ctl_regs {
+ u32 _unused0[4];
+ u32 isr; /* 0x10 Status Register */
+ u32 _unused1[3];
+ u32 rev; /* 0x20 Revision Register */
+ u32 _unused2[3];
+ u32 iar; /* 0x30 Indirect Address Register */
+ u32 _unused3[3];
+ u32 idr0; /* 0x40 Indirect Data Register 0 */
+ u32 _unused4[3];
+ u32 idr1; /* 0x50 Indirect Data Register 1 */
+ u32 _unused5[3];
+ u32 idr2; /* 0x60 Indirect Data Register 2 */
+ u32 _unused6[3];
+ u32 idr3; /* 0x70 Indirect Data Register 3 */
+};
+
+struct hal2_aes_regs {
+ u32 rx_stat[2]; /* Status registers */
+ u32 rx_cr[2]; /* Control registers */
+ u32 rx_ud[4]; /* User data window */
+ u32 rx_st[24]; /* Channel status data */
+
+ u32 tx_stat[1]; /* Status register */
+ u32 tx_cr[3]; /* Control registers */
+ u32 tx_ud[4]; /* User data window */
+ u32 tx_st[24]; /* Channel status data */
+};
+
+struct hal2_vol_regs {
+ u32 right; /* Right volume */
+ u32 left; /* Left volume */
+};
+
+struct hal2_syn_regs {
+ u32 _unused0[2];
+ u32 page; /* DOC Page register */
+ u32 regsel; /* DOC Register selection */
+ u32 dlow; /* DOC Data low */
+ u32 dhigh; /* DOC Data high */
+ u32 irq; /* IRQ Status */
+ u32 dram; /* DRAM Access */
+};
+
+#endif /* __HAL2_H */
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
new file mode 100644
index 0000000..4c63504
--- /dev/null
+++ b/sound/mips/sgio2audio.c
@@ -0,0 +1,1006 @@
+/*
+ * Sound driver for Silicon Graphics O2 Workstations A/V board audio.
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ * Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
+ * Mxier part taken from mace_audio.c:
+ * Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/spinlock.h>
+#include <linux/gfp.h>
+#include <linux/vmalloc.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include <asm/ip32/ip32_ints.h>
+#include <asm/ip32/mace.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#define SNDRV_GET_ID
+#include <sound/initval.h>
+#include <sound/ad1843.h>
+
+
+MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
+MODULE_DESCRIPTION("SGI O2 Audio");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
+
+
+#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */
+#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */
+
+#define CODEC_CONTROL_WORD_SHIFT 0
+#define CODEC_CONTROL_READ BIT(16)
+#define CODEC_CONTROL_ADDRESS_SHIFT 17
+
+#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */
+#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */
+#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */
+#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */
+#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */
+#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */
+#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */
+#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
+#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */
+#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */
+
+#define CHANNEL_RING_SHIFT 12
+#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT)
+#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1)
+
+#define CHANNEL_LEFT_SHIFT 40
+#define CHANNEL_RIGHT_SHIFT 8
+
+struct snd_sgio2audio_chan {
+ int idx;
+ struct snd_pcm_substream *substream;
+ int pos;
+ snd_pcm_uframes_t size;
+ spinlock_t lock;
+};
+
+/* definition of the chip-specific record */
+struct snd_sgio2audio {
+ struct snd_card *card;
+
+ /* codec */
+ struct snd_ad1843 ad1843;
+ spinlock_t ad1843_lock;
+
+ /* channels */
+ struct snd_sgio2audio_chan channel[3];
+
+ /* resources */
+ void *ring_base;
+ dma_addr_t ring_base_dma;
+};
+
+/* AD1843 access */
+
+/*
+ * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
+ *
+ * Returns unsigned register value on success, -errno on failure.
+ */
+static int read_ad1843_reg(void *priv, int reg)
+{
+ struct snd_sgio2audio *chip = priv;
+ int val;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->ad1843_lock, flags);
+
+ writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
+ CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
+ wmb();
+ val = readq(&mace->perif.audio.codec_control); /* flush bus */
+ udelay(200);
+
+ val = readq(&mace->perif.audio.codec_read);
+
+ spin_unlock_irqrestore(&chip->ad1843_lock, flags);
+ return val;
+}
+
+/*
+ * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
+ */
+static int write_ad1843_reg(void *priv, int reg, int word)
+{
+ struct snd_sgio2audio *chip = priv;
+ int val;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->ad1843_lock, flags);
+
+ writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
+ (word << CODEC_CONTROL_WORD_SHIFT),
+ &mace->perif.audio.codec_control);
+ wmb();
+ val = readq(&mace->perif.audio.codec_control); /* flush bus */
+ udelay(200);
+
+ spin_unlock_irqrestore(&chip->ad1843_lock, flags);
+ return 0;
+}
+
+static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
+ (int)kcontrol->private_value);
+ return 0;
+}
+
+static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+ int vol;
+
+ vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
+
+ ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
+ ucontrol->value.integer.value[1] = vol & 0xFF;
+
+ return 0;
+}
+
+static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+ int newvol, oldvol;
+
+ oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
+ newvol = (ucontrol->value.integer.value[0] << 8) |
+ ucontrol->value.integer.value[1];
+
+ newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
+ newvol);
+
+ return newvol != oldvol;
+}
+
+static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *texts[3] = {
+ "Cam Mic", "Mic", "Line"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= 3)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
+ return 0;
+}
+
+static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+ int newsrc, oldsrc;
+
+ oldsrc = ad1843_get_recsrc(&chip->ad1843);
+ newsrc = ad1843_set_recsrc(&chip->ad1843,
+ ucontrol->value.enumerated.item[0]);
+
+ return newsrc != oldsrc;
+}
+
+/* dac1/pcm0 mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Volume",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = AD1843_GAIN_PCM_0,
+ .info = sgio2audio_gain_info,
+ .get = sgio2audio_gain_get,
+ .put = sgio2audio_gain_put,
+};
+
+/* dac2/pcm1 mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Volume",
+ .index = 1,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = AD1843_GAIN_PCM_1,
+ .info = sgio2audio_gain_info,
+ .get = sgio2audio_gain_get,
+ .put = sgio2audio_gain_put,
+};
+
+/* record level mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = AD1843_GAIN_RECLEV,
+ .info = sgio2audio_gain_info,
+ .get = sgio2audio_gain_get,
+ .put = sgio2audio_gain_put,
+};
+
+/* record level source control */
+static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = sgio2audio_source_info,
+ .get = sgio2audio_source_get,
+ .put = sgio2audio_source_put,
+};
+
+/* line mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Playback Volume",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = AD1843_GAIN_LINE,
+ .info = sgio2audio_gain_info,
+ .get = sgio2audio_gain_get,
+ .put = sgio2audio_gain_put,
+};
+
+/* cd mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Playback Volume",
+ .index = 1,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = AD1843_GAIN_LINE_2,
+ .info = sgio2audio_gain_info,
+ .get = sgio2audio_gain_get,
+ .put = sgio2audio_gain_put,
+};
+
+/* mic mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = AD1843_GAIN_MIC,
+ .info = sgio2audio_gain_info,
+ .get = sgio2audio_gain_get,
+ .put = sgio2audio_gain_put,
+};
+
+
+static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
+{
+ int err;
+
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&sgio2audio_ctrl_line, chip));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+/* low-level audio interface DMA */
+
+/* get data out of bounce buffer, count must be a multiple of 32 */
+/* returns 1 if a period has elapsed */
+static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
+ unsigned int ch, unsigned int count)
+{
+ int ret;
+ unsigned long src_base, src_pos, dst_mask;
+ unsigned char *dst_base;
+ int dst_pos;
+ u64 *src;
+ s16 *dst;
+ u64 x;
+ unsigned long flags;
+ struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
+
+ spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+ src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
+ src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
+ dst_base = runtime->dma_area;
+ dst_pos = chip->channel[ch].pos;
+ dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
+
+ /* check if a period has elapsed */
+ chip->channel[ch].size += (count >> 3); /* in frames */
+ ret = chip->channel[ch].size >= runtime->period_size;
+ chip->channel[ch].size %= runtime->period_size;
+
+ while (count) {
+ src = (u64 *)(src_base + src_pos);
+ dst = (s16 *)(dst_base + dst_pos);
+
+ x = *src;
+ dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
+ dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
+
+ src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
+ dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
+ count -= sizeof(u64);
+ }
+
+ writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
+ chip->channel[ch].pos = dst_pos;
+
+ spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+ return ret;
+}
+
+/* put some DMA data in bounce buffer, count must be a multiple of 32 */
+/* returns 1 if a period has elapsed */
+static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
+ unsigned int ch, unsigned int count)
+{
+ int ret;
+ s64 l, r;
+ unsigned long dst_base, dst_pos, src_mask;
+ unsigned char *src_base;
+ int src_pos;
+ u64 *dst;
+ s16 *src;
+ unsigned long flags;
+ struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
+
+ spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+ dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
+ dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
+ src_base = runtime->dma_area;
+ src_pos = chip->channel[ch].pos;
+ src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
+
+ /* check if a period has elapsed */
+ chip->channel[ch].size += (count >> 3); /* in frames */
+ ret = chip->channel[ch].size >= runtime->period_size;
+ chip->channel[ch].size %= runtime->period_size;
+
+ while (count) {
+ src = (s16 *)(src_base + src_pos);
+ dst = (u64 *)(dst_base + dst_pos);
+
+ l = src[0]; /* sign extend */
+ r = src[1]; /* sign extend */
+
+ *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
+ ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
+
+ dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
+ src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
+ count -= sizeof(u64);
+ }
+
+ writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
+ chip->channel[ch].pos = src_pos;
+
+ spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+ return ret;
+}
+
+static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
+{
+ struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+ struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+ int ch = chan->idx;
+
+ /* reset DMA channel */
+ writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
+ udelay(10);
+ writeq(0, &mace->perif.audio.chan[ch].control);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* push a full buffer */
+ snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
+ }
+ /* set DMA to wake on 50% empty and enable interrupt */
+ writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
+ &mace->perif.audio.chan[ch].control);
+ return 0;
+}
+
+static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
+{
+ struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+
+ writeq(0, &mace->perif.audio.chan[chan->idx].control);
+ return 0;
+}
+
+static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
+{
+ struct snd_sgio2audio_chan *chan = dev_id;
+ struct snd_pcm_substream *substream;
+ struct snd_sgio2audio *chip;
+ int count, ch;
+
+ substream = chan->substream;
+ chip = snd_pcm_substream_chip(substream);
+ ch = chan->idx;
+
+ /* empty the ring */
+ count = CHANNEL_RING_SIZE -
+ readq(&mace->perif.audio.chan[ch].depth) - 32;
+ if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
+ snd_pcm_period_elapsed(substream);
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
+{
+ struct snd_sgio2audio_chan *chan = dev_id;
+ struct snd_pcm_substream *substream;
+ struct snd_sgio2audio *chip;
+ int count, ch;
+
+ substream = chan->substream;
+ chip = snd_pcm_substream_chip(substream);
+ ch = chan->idx;
+ /* fill the ring */
+ count = CHANNEL_RING_SIZE -
+ readq(&mace->perif.audio.chan[ch].depth) - 32;
+ if (snd_sgio2audio_dma_push_frag(chip, ch, count))
+ snd_pcm_period_elapsed(substream);
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
+{
+ struct snd_sgio2audio_chan *chan = dev_id;
+ struct snd_pcm_substream *substream;
+
+ substream = chan->substream;
+ snd_sgio2audio_dma_stop(substream);
+ snd_sgio2audio_dma_start(substream);
+ return IRQ_HANDLED;
+}
+
+/* PCM part */
+/* PCM hardware definition */
+static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER),
+ .formats = SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 32768,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+/* PCM playback open callback */
+static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
+{
+ struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_sgio2audio_pcm_hw;
+ runtime->private_data = &chip->channel[1];
+ return 0;
+}
+
+static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
+{
+ struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_sgio2audio_pcm_hw;
+ runtime->private_data = &chip->channel[2];
+ return 0;
+}
+
+/* PCM capture open callback */
+static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_sgio2audio_pcm_hw;
+ runtime->private_data = &chip->channel[0];
+ return 0;
+}
+
+/* PCM close callback */
+static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->private_data = NULL;
+ return 0;
+}
+
+
+/* hw_params callback */
+static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int size = params_buffer_bytes(hw_params);
+
+ /* alloc virtual 'dma' area */
+ if (runtime->dma_area)
+ vfree(runtime->dma_area);
+ runtime->dma_area = vmalloc(size);
+ if (runtime->dma_area == NULL)
+ return -ENOMEM;
+ runtime->dma_bytes = size;
+ return 0;
+}
+
+/* hw_free callback */
+static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ if (substream->runtime->dma_area)
+ vfree(substream->runtime->dma_area);
+ substream->runtime->dma_area = NULL;
+ return 0;
+}
+
+/* prepare callback */
+static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+ int ch = chan->idx;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+ /* Setup the pseudo-dma transfer pointers. */
+ chip->channel[ch].pos = 0;
+ chip->channel[ch].size = 0;
+ chip->channel[ch].substream = substream;
+
+ /* set AD1843 format */
+ /* hardware format is always S16_LE */
+ switch (substream->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ ad1843_setup_dac(&chip->ad1843,
+ ch - 1,
+ runtime->rate,
+ SNDRV_PCM_FORMAT_S16_LE,
+ runtime->channels);
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ ad1843_setup_adc(&chip->ad1843,
+ runtime->rate,
+ SNDRV_PCM_FORMAT_S16_LE,
+ runtime->channels);
+ break;
+ }
+ spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+ return 0;
+}
+
+/* trigger callback */
+static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* start the PCM engine */
+ snd_sgio2audio_dma_start(substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ /* stop the PCM engine */
+ snd_sgio2audio_dma_stop(substream);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/* pointer callback */
+static snd_pcm_uframes_t
+snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+ struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+
+ /* get the current hardware pointer */
+ return bytes_to_frames(substream->runtime,
+ chip->channel[chan->idx].pos);
+}
+
+/* get the physical page pointer on the given offset */
+static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream,
+ unsigned long offset)
+{
+ return vmalloc_to_page(substream->runtime->dma_area + offset);
+}
+
+/* operators */
+static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
+ .open = snd_sgio2audio_playback1_open,
+ .close = snd_sgio2audio_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sgio2audio_pcm_hw_params,
+ .hw_free = snd_sgio2audio_pcm_hw_free,
+ .prepare = snd_sgio2audio_pcm_prepare,
+ .trigger = snd_sgio2audio_pcm_trigger,
+ .pointer = snd_sgio2audio_pcm_pointer,
+ .page = snd_sgio2audio_page,
+};
+
+static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
+ .open = snd_sgio2audio_playback2_open,
+ .close = snd_sgio2audio_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sgio2audio_pcm_hw_params,
+ .hw_free = snd_sgio2audio_pcm_hw_free,
+ .prepare = snd_sgio2audio_pcm_prepare,
+ .trigger = snd_sgio2audio_pcm_trigger,
+ .pointer = snd_sgio2audio_pcm_pointer,
+ .page = snd_sgio2audio_page,
+};
+
+static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
+ .open = snd_sgio2audio_capture_open,
+ .close = snd_sgio2audio_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sgio2audio_pcm_hw_params,
+ .hw_free = snd_sgio2audio_pcm_hw_free,
+ .prepare = snd_sgio2audio_pcm_prepare,
+ .trigger = snd_sgio2audio_pcm_trigger,
+ .pointer = snd_sgio2audio_pcm_pointer,
+ .page = snd_sgio2audio_page,
+};
+
+/*
+ * definitions of capture are omitted here...
+ */
+
+/* create a pcm device */
+static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ /* create first pcm device with one outputs and one input */
+ err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = chip;
+ strcpy(pcm->name, "SGI O2 DAC1");
+
+ /* set operators */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_sgio2audio_playback1_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_sgio2audio_capture_ops);
+
+ /* create second pcm device with one outputs and no input */
+ err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = chip;
+ strcpy(pcm->name, "SGI O2 DAC2");
+
+ /* set operators */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_sgio2audio_playback2_ops);
+
+ return 0;
+}
+
+static struct {
+ int idx;
+ int irq;
+ irqreturn_t (*isr)(int, void *);
+ const char *desc;
+} snd_sgio2_isr_table[] = {
+ {
+ .idx = 0,
+ .irq = MACEISA_AUDIO1_DMAT_IRQ,
+ .isr = snd_sgio2audio_dma_in_isr,
+ .desc = "Capture DMA Channel 0"
+ }, {
+ .idx = 0,
+ .irq = MACEISA_AUDIO1_OF_IRQ,
+ .isr = snd_sgio2audio_error_isr,
+ .desc = "Capture Overflow"
+ }, {
+ .idx = 1,
+ .irq = MACEISA_AUDIO2_DMAT_IRQ,
+ .isr = snd_sgio2audio_dma_out_isr,
+ .desc = "Playback DMA Channel 1"
+ }, {
+ .idx = 1,
+ .irq = MACEISA_AUDIO2_MERR_IRQ,
+ .isr = snd_sgio2audio_error_isr,
+ .desc = "Memory Error Channel 1"
+ }, {
+ .idx = 2,
+ .irq = MACEISA_AUDIO3_DMAT_IRQ,
+ .isr = snd_sgio2audio_dma_out_isr,
+ .desc = "Playback DMA Channel 2"
+ }, {
+ .idx = 2,
+ .irq = MACEISA_AUDIO3_MERR_IRQ,
+ .isr = snd_sgio2audio_error_isr,
+ .desc = "Memory Error Channel 2"
+ }
+};
+
+/* ALSA driver */
+
+static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
+{
+ int i;
+
+ /* reset interface */
+ writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+ udelay(1);
+ writeq(0, &mace->perif.audio.control);
+
+ /* release IRQ's */
+ for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
+ free_irq(snd_sgio2_isr_table[i].irq,
+ &chip->channel[snd_sgio2_isr_table[i].idx]);
+
+ dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
+ chip->ring_base, chip->ring_base_dma);
+
+ /* release card data */
+ kfree(chip);
+ return 0;
+}
+
+static int snd_sgio2audio_dev_free(struct snd_device *device)
+{
+ struct snd_sgio2audio *chip = device->device_data;
+
+ return snd_sgio2audio_free(chip);
+}
+
+static struct snd_device_ops ops = {
+ .dev_free = snd_sgio2audio_dev_free,
+};
+
+static int __devinit snd_sgio2audio_create(struct snd_card *card,
+ struct snd_sgio2audio **rchip)
+{
+ struct snd_sgio2audio *chip;
+ int i, err;
+
+ *rchip = NULL;
+
+ /* check if a codec is attached to the interface */
+ /* (Audio or Audio/Video board present) */
+ if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
+ return -ENOENT;
+
+ chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+
+ chip->card = card;
+
+ chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
+ &chip->ring_base_dma, GFP_USER);
+ if (chip->ring_base == NULL) {
+ printk(KERN_ERR
+ "sgio2audio: could not allocate ring buffers\n");
+ kfree(chip);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&chip->ad1843_lock);
+
+ /* initialize channels */
+ for (i = 0; i < 3; i++) {
+ spin_lock_init(&chip->channel[i].lock);
+ chip->channel[i].idx = i;
+ }
+
+ /* allocate IRQs */
+ for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
+ if (request_irq(snd_sgio2_isr_table[i].irq,
+ snd_sgio2_isr_table[i].isr,
+ 0,
+ snd_sgio2_isr_table[i].desc,
+ &chip->channel[snd_sgio2_isr_table[i].idx])) {
+ snd_sgio2audio_free(chip);
+ printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
+ snd_sgio2_isr_table[i].irq);
+ return -EBUSY;
+ }
+ }
+
+ /* reset the interface */
+ writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+ udelay(1);
+ writeq(0, &mace->perif.audio.control);
+ msleep_interruptible(1); /* give time to recover */
+
+ /* set ring base */
+ writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
+
+ /* attach the AD1843 codec */
+ chip->ad1843.read = read_ad1843_reg;
+ chip->ad1843.write = write_ad1843_reg;
+ chip->ad1843.chip = chip;
+
+ /* initialize the AD1843 codec */
+ err = ad1843_init(&chip->ad1843);
+ if (err < 0) {
+ snd_sgio2audio_free(chip);
+ return err;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ snd_sgio2audio_free(chip);
+ return err;
+ }
+ *rchip = chip;
+ return 0;
+}
+
+static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
+{
+ struct snd_card *card;
+ struct snd_sgio2audio *chip;
+ int err;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ err = snd_sgio2audio_create(card, &chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_card_set_dev(card, &pdev->dev);
+
+ err = snd_sgio2audio_new_pcm(chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = snd_sgio2audio_new_mixer(chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "SGI O2 Audio");
+ strcpy(card->shortname, "SGI O2 Audio");
+ sprintf(card->longname, "%s irq %i-%i",
+ card->shortname,
+ MACEISA_AUDIO1_DMAT_IRQ,
+ MACEISA_AUDIO3_MERR_IRQ);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ platform_set_drvdata(pdev, card);
+ return 0;
+}
+
+static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ snd_card_free(card);
+ platform_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+static struct platform_driver sgio2audio_driver = {
+ .probe = snd_sgio2audio_probe,
+ .remove = __devexit_p(snd_sgio2audio_remove),
+ .driver = {
+ .name = "sgio2audio",
+ .owner = THIS_MODULE,
+ }
+};
+
+static int __init alsa_card_sgio2audio_init(void)
+{
+ return platform_driver_register(&sgio2audio_driver);
+}
+
+static void __exit alsa_card_sgio2audio_exit(void)
+{
+ platform_driver_unregister(&sgio2audio_driver);
+}
+
+module_init(alsa_card_sgio2audio_init)
+module_exit(alsa_card_sgio2audio_exit)
diff --git a/sound/oss/.gitignore b/sound/oss/.gitignore
new file mode 100644
index 0000000..7efb12b
--- /dev/null
+++ b/sound/oss/.gitignore
@@ -0,0 +1,4 @@
+#Ignore generated files
+maui_boot.h
+pss_boot.h
+trix_boot.h
diff --git a/sound/oss/CHANGELOG b/sound/oss/CHANGELOG
new file mode 100644
index 0000000..8706cd6
--- /dev/null
+++ b/sound/oss/CHANGELOG
@@ -0,0 +1,369 @@
+Note these changes relate to Hannu's code and don't include the changes
+made outside of this for modularising the sound
+
+Changelog for version 3.8o
+--------------------------
+
+Since 3.8h
+- Included support for OPL3-SA1 and SoftOSS
+
+Since 3.8
+- Fixed SNDCTL_DSP_GETOSPACE
+- Compatibility fixes for Linux 2.1.47
+
+Since 3.8-beta21
+- Fixed all known bugs (I think).
+
+Since 3.8-beta8
+- Lot of fixes to audio playback code in dmabuf.c
+
+Since 3.8-beta6
+- Fixed the famous Quake delay bug.
+
+Since 3.8-beta5
+- Fixed many bugs in audio playback.
+
+Since 3.8-beta4
+- Just minor changes.
+
+Since 3.8-beta1
+- Major rewrite of audio playback handling.
+- Added AWE32 support by Takashi Iwai (in ./lowlevel/).
+
+Since 3.7-beta#
+- Passing of ioctl() parameters between soundcard.c and other modules has been
+changed so that arg always points to kernel space.
+- Some bugfixes.
+
+Since 3.7-beta5
+- Disabled MIDI input with GUS PnP (Interwave). There seems to be constant
+stream of received 0x00 bytes when the MIDI receiver is enabled.
+
+Since 3.5
+- Changes almost everywhere.
+- Support for OPTi 82C924-based sound cards.
+
+Since 3.5.4-beta8
+- Fixed a bug in handling of non-fragment sized writes in 16 bit/stereo mode
+ with GUS.
+- Limited minimum fragment size with some audio devices (GUS=512 and
+ SB=32). These devices require more time to "recover" from processing
+ of each fragment.
+
+Since 3.5.4-beta6/7
+- There seems to be problems in the OPTi 82C930 so cards based on this
+ chip don't necessarily work yet. There are problems in detecting the
+ MIDI interface. Also mixer volumes may be seriously wrong on some systems.
+ You can safely use this driver version with C930 if it looks to work.
+ However please don't complain if you have problems with it. C930 support
+ should be fixed in future releases.
+- Got initialization of GUS PnP to work. With this version GUS PnP should
+ work in GUS compatible mode after initialization using isapnptools.
+- Fixed a bug in handling of full duplex cards in write only mode. This has
+ been causing "audio device opening" errors with RealAudio player.
+
+Since 3.5.4.beta5
+- Changes to OPTi 82C930 driver.
+- Major changes to the Soundscape driver. The driver requires now just one
+ DMA channel. The extra audio/dsp device (the "Not functional" one) used
+ for code download in the earlier versions has been eliminated. There is now
+ just one /dev/dsp# device which is used both for code download and audio.
+
+Since 3.5.4.beta4
+- Minor changes.
+
+Since 3.5.4-beta2
+- Fixed silent playback with ESS 688/1688.
+- Got SB16 to work without the 16 bit DMA channel (only the 8 bit one
+ is required for 8 and 16 bit modes).
+- Added the "lowlevel" subdirectory for additional low level drivers that
+ are not part of USS core. See lowlevel/README for more info.
+- Included support for ACI mixer (by Markus Kuhn). ACI is a mixer used in
+ miroPCM sound cards. See lowlevel/aci.readme for more info.
+- Support for Aztech Washington chipset (AZT2316 ASIC).
+
+Since 3.5.4-beta1
+- Reduced clicking with AD1848.
+- Support for OPTi 82C930. Only half duplex at this time. 16 bit playback
+ is sometimes just white noise (occurs randomly).
+
+Since 3.5.2
+- Major changes to the SB/Jazz16/ESS driver (most parts rewritten).
+ The most noticeable new feature is support for multiple SB cards at the same
+ time.
+- Renamed sb16_midi.c to uart401.c. Also modified it to work also with
+ other MPU401 UART compatible cards than SB16/ESS/Jazz.
+- Some changes which reduce clicking in audio playback.
+- Copying policy is now GPL.
+
+Since 3.5.1
+- TB Maui initialization support
+Since 3.5
+- Improved handling of playback underrun situations.
+
+Since 3.5-beta10
+- Bug fixing
+
+Since 3.5-beta9
+- Fixed for compatibility with Linux 1.3.70 and later.
+- Changed boot time passing of 16 bit DMA channel number to SB driver.
+
+Since 3.5-beta8
+- Minor changes
+
+Since 3.5-beta7
+- enhancements to configure program (by Jeff Tranter):
+ - prompts are in same format as 1.3.x Linux kernel config program
+ - on-line help for each question
+ - fixed some compile warnings detected by gcc/g++ -Wall
+ - minor grammatical changes to prompts
+
+Since 3.5-beta6
+- Fixed bugs in mmap() support.
+- Minor changes to Maui driver.
+
+Since 3.5-beta5
+- Fixed crash after recording with ESS688. It's generally a good
+ idea to stop inbound DMA transfers before freeing the memory
+ buffer.
+- Fixed handling of AD1845 codec (for example Shuttle Sound System).
+- Few other fixes.
+
+Since 3.5-beta4
+- Fixed bug in handling of uninitialized instruments with GUS.
+
+Since 3.5-beta3
+- Few changes which decrease popping at end/beginning of audio playback.
+
+Since 3.5-beta2
+- Removed MAD16+CS4231 hack made in previous version since it didn't
+ help.
+- Fixed the above bug in proper way and in proper place. Many thanks
+ to James Hightower.
+
+Since 3.5-beta1
+- Bug fixes.
+- Full duplex audio with MAD16+CS4231 may work now. The driver configures
+ SB DMA of MAD16 so that it doesn't conflict with codec's DMA channels.
+ The side effect is that all 8 bit DMA channels (0,1,3) are populated in
+ duplex mode.
+
+Since 3.5-alpha9
+- Bug fixes (mostly in Jazz16 and ESS1688/688 supports).
+- Temporarily disabled recording with ESS1688/688 since it causes crash.
+- Changed audio buffer partitioning algorithm so that it selects
+ smaller fragment size than earlier. This improves real time capabilities
+ of the driver and makes recording to disk to work better. Unfortunately
+ this change breaks some programs which assume that fragments cannot be
+ shorter than 4096 bytes.
+
+Since 3.5-alpha8
+- Bug fixes
+
+Since 3.5-alpha7
+- Linux kernel compatible configuration (_EXPERIMENTAL_). Enable
+ using command "cd /linux/drivers/sound;make script" and then
+ just run kernel's make config normally.
+- Minor fixes to the SB support. Hopefully the driver works with
+ all SB models now.
+- Added support for ESS ES1688 "AudioDrive" based cards.
+
+Since 3.5-alpha6
+- SB Pro and SB16 supports are no longer separately selectable options.
+ Enabling SB enables them too.
+- Changed all #ifndef EXCLUDE_xx stuff to #ifdef CONFIG_xx. Modified
+configure to handle this.
+- Removed initialization messages from the
+modularized version. They can be enabled by using init_trace=1 in
+the insmod command line (insmod sound init_trace=1).
+- More AIX stuff.
+- Added support for synchronizing dsp/audio devices with /dev/sequencer.
+- mmap() support for dsp/audio devices.
+
+Since 3.5-alpha5
+- AIX port.
+- Changed some xxx_PATCH macros in soundcard.h to work with
+ big endian machines.
+
+Since 3.5-alpha4
+- Removed the 'setfx' stuff from the version distributed with kernel
+ sources. Running 'setfx' is required again.
+
+Since 3.5-alpha3
+- Moved stuff from the 'setfx' program to the AudioTrix Pro driver.
+
+Since 3.5-alpha2
+- Modifications to makefile and configure.c. Unnecessary sources
+ are no longer compiled. Newly created local.h is also copied to
+ /etc/soundconf. "make oldconfig" reads /etc/soundconf and produces
+ new local.h which is compatible with current version of the driver.
+- Some fixes to the SB16 support.
+- Fixed random protection fault in gus_wave.c
+
+Since 3.5-alpha1
+- Modified to work with Linux-1.3.33 and later
+- Some minor changes
+
+Since 3.0.2
+- Support for CS4232 based PnP cards (AcerMagic S23 etc).
+- Full duplex support for some CS4231, CS4232 and AD1845 based cards
+(GUS MAX, AudioTrix Pro, AcerMagic S23 and many MAD16/Mozart cards
+having a codec mentioned above).
+- Almost fully rewritten loadable modules support.
+- Fixed some bugs.
+- Huge amount of testing (more testing is still required).
+- mmap() support (works with some cards). Requires much more testing.
+- Sample/patch/program loading for TB Maui/Tropez. No initialization
+since TB doesn't allow me to release that code.
+- Using CS4231 compatible codecs as timer for /dev/music.
+
+Since 3.0.1
+- Added allocation of I/O ports, DMA channels and interrupts
+to the initialization code. This may break modules support since
+the driver may not free some resources on unload. Should be fixed soon.
+
+Since 3.0
+- Some important bug fixes.
+- select() for /dev/dsp and /dev/audio (Linux only).
+(To use select() with read, you have to call read() to start
+the recording. Calling write() kills recording immediately so
+use select() carefully when you are writing a half duplex app.
+Full duplex mode is not implemented yet.) Select works also with
+/dev/sequencer and /dev/music. Maybe with /dev/midi## too.
+
+Since 3.0-beta2
+- Minor fixes.
+- Added Readme.cards
+
+Since 3.0-beta1
+- Minor fixes to the modules support.
+- Eliminated call to sb_free_irq() in ad1848.c
+- Rewritten MAD16&Mozart support (not tested with MAD16 Pro).
+- Fix to DMA initialization of PSS cards.
+- Some fixes to ad1848/cs42xx mixer support (GUS MAX, MSS, etc.)
+- Fixed some bugs in the PSS driver which caused I/O errors with
+ the MSS mode (/dev/dsp).
+
+Since 3.0-950506
+- Recording with GUS MAX fixed. It works when the driver is configured
+ to use two DMA channels with GUS MAX (16 bit ones recommended).
+
+Since 3.0-94xxxx
+- Too many changes
+
+Since 3.0-940818
+- Fixes for Linux 1.1.4x.
+- Disables Disney Sound System with SG NX Pro 16 (less noise).
+
+Since 2.90-2
+- Fixes to soundcard.h
+- Non blocking mode to /dev/sequencer
+- Experimental detection code for Ensoniq Soundscape.
+
+Since 2.90
+- Minor and major bug fixes
+
+Since pre-3.0-940712
+- GUS MAX support
+- Partially working MSS/WSS support (could work with some cards).
+- Hardware u-Law and A-Law support with AD1848/CS4248 and CS4231 codecs
+ (GUS MAX, GUS16, WSS etc). Hardware ADPCM is possible with GUS16 and
+ GUS MAX, but it doesn't work yet.
+Since pre-3.0-940426
+- AD1848/CS4248/CS4231 codec support (MSS, GUS MAX, Aztec, Orchid etc).
+This codec chip is used in various sound cards. This version is developed
+for the 16 bit daughtercard of GUS. It should work with other cards also
+if the following requirements are met:
+ - The I/O, IRQ and DMA settings are jumper selectable or
+ the card is initialized by booting DOS before booting Linux (etc.).
+ - You add the IO, IRQ and DMA settings manually to the local.h.
+ (Just define GUS16_BASE, GUS16_IRQ and GUS16_DMA). Note that
+ the base address bust be the base address of the codec chip not the
+ card itself. For the GUS16 these are the same but most MSS compatible
+ cards have the codec located at card_base+4.
+- Some minor changes
+
+Since 2.5 (******* MAJOR REWRITE ***********)
+
+This version is based on v2.3. I have tried to maintain two versions
+together so that this one should have the same features than v2.5.
+Something may still be missing. If you notice such things, please let me
+know.
+
+The Readme.v30 contains more details.
+
+- /dev/midi## devices.
+- /dev/sequencer2
+
+Since 2.5-beta2
+- Some fine tuning to the GUS v3.7 mixer code.
+- Fixed speed limits for the plain SB (1.0 to 2.0).
+
+Since 2.5-beta
+- Fixed OPL-3 detection with SB. Caused problems with PAS16.
+- GUS v3.7 mixer support.
+
+Since 2.4
+- Mixer support for Sound Galaxy NX Pro (define __SGNXPRO__ on your local.h).
+- Fixed truncated sound on /dev/dsp when the device is closed.
+- Linear volume mode for GUS
+- Pitch bends larger than +/- 2 octaves.
+- MIDI recording for SB and SB Pro. (Untested).
+- Some other fixes.
+- SB16 MIDI and DSP drivers only initialized if SB16 actually installed.
+- Implemented better detection for OPL-3. This should be useful if you
+ have an old SB Pro (the non-OPL-3 one) or a SB 2.0 clone which has a OPL-3.
+- SVR4.2 support by Ian Hartas. Initial ALPHA TEST version (untested).
+
+Since 2.3b
+- Fixed bug which made it impossible to make long recordings to disk.
+ Recording was not restarted after a buffer overflow situation.
+- Limited mixer support for GUS.
+- Numerous improvements to the GUS driver by Andrew Robinson. Including
+ some click removal etc.
+
+Since 2.3
+- Fixed some minor bugs in the SB16 driver.
+
+Since 2.2b
+- Full SB16 DSP support. 8/16 bit, mono/stereo
+- The SCO and FreeBSD versions should be in sync now. There are some
+ problems with SB16 and GUS in the FreeBSD versions.
+ The DMA buffer allocation of the SCO version has been polished but
+ there could still be some problems. At least it hogs memory.
+ The DMA channel
+ configuration method used in the SCO/System is a hack.
+- Support for the MPU emulation of the SB16.
+- Some big arrays are now allocated boot time. This makes the BSS segment
+ smaller which makes it possible to use the full driver with
+ NetBSD. These arrays are not allocated if no suitable sound card is available.
+- Fixed a bug in the compute_and_set_volume in gus_wave.c
+- Fixed the too fast mono playback problem of SB Pro and PAS16.
+
+Since 2.2
+- Stereo recording for SB Pro. Somehow it was missing and nobody
+ had noticed it earlier.
+- Minor polishing.
+- Interpreting of boot time arguments (sound=) for Linux.
+- Breakup of sb_dsp.c. Parts of the code has been moved to
+ sb_mixer.c and sb_midi.c
+
+Since 2.1
+- Preliminary support for SB16.
+ - The SB16 mixer is supported in its native mode.
+ - Digitized voice capability up to 44.1 kHz/8 bit/mono
+ (16 bit and stereo support coming in the next release).
+- Fixed some bugs in the digitized voice driver for PAS16.
+- Proper initialization of the SB emulation of latest PAS16 models.
+
+- Significantly improved /dev/dsp and /dev/audio support.
+ - Now supports half duplex mode. It's now possible to record and
+ playback without closing and reopening the device.
+ - It's possible to use smaller buffers than earlier. There is a new
+ ioctl(fd, SNDCTL_DSP_SUBDIVIDE, &n) where n should be 1, 2 or 4.
+ This call instructs the driver to use smaller buffers. The default
+ buffer size (0.5 to 1.0 seconds) is divided by n. Should be called
+ immediately after opening the device.
+
+Since 2.0
+Just cosmetic changes.
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
new file mode 100644
index 0000000..1ca7427
--- /dev/null
+++ b/sound/oss/Kconfig
@@ -0,0 +1,569 @@
+# drivers/sound/Config.in
+#
+# 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net>
+# More hacking for modularisation.
+#
+# Prompt user for primary drivers.
+
+config SOUND_BCM_CS4297A
+ tristate "Crystal Sound CS4297a (for Swarm)"
+ depends on SIBYTE_SWARM
+ help
+ The BCM91250A has a Crystal CS4297a on synchronous serial
+ port B (in addition to the DB-9 serial port). Say Y or M
+ here to enable the sound chip instead of the UART. Also
+ note that CONFIG_KGDB should not be enabled at the same
+ time, since it also attempts to use this UART port.
+
+config SOUND_VWSND
+ tristate "SGI Visual Workstation Sound"
+ depends on X86_VISWS
+ help
+ Say Y or M if you have an SGI Visual Workstation and you want to be
+ able to use its on-board audio. Read
+ <file:Documentation/sound/oss/vwsnd> for more info on this driver's
+ capabilities.
+
+config SOUND_AU1550_AC97
+ tristate "Au1550/Au1200 AC97 Sound"
+ depends on SOC_AU1550 || SOC_AU1200
+
+config SOUND_MSNDCLAS
+ tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey"
+ depends on (m || !STANDALONE) && ISA
+ help
+ Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or
+ Monterey (not for the Pinnacle or Fiji).
+
+ See <file:Documentation/sound/oss/MultiSound> for important information
+ about this driver. Note that it has been discontinued, but the
+ Voyetra Turtle Beach knowledge base entry for it is still available
+ at <http://www.turtlebeach.com/site/kb_ftp/790.asp>.
+
+comment "Compiled-in MSND Classic support requires firmware during compilation."
+ depends on SOUND_PRIME && SOUND_MSNDCLAS=y
+
+config MSNDCLAS_HAVE_BOOT
+ bool
+ depends on SOUND_MSNDCLAS=y && !STANDALONE
+ default y
+
+config MSNDCLAS_INIT_FILE
+ string "Full pathname of MSNDINIT.BIN firmware file"
+ depends on SOUND_MSNDCLAS
+ default "/etc/sound/msndinit.bin"
+ help
+ The MultiSound cards have two firmware files which are required for
+ operation, and are not currently included. These files can be
+ obtained from Turtle Beach. See
+ <file:Documentation/sound/oss/MultiSound> for information on how to
+ obtain this.
+
+config MSNDCLAS_PERM_FILE
+ string "Full pathname of MSNDPERM.BIN firmware file"
+ depends on SOUND_MSNDCLAS
+ default "/etc/sound/msndperm.bin"
+ help
+ The MultiSound cards have two firmware files which are required for
+ operation, and are not currently included. These files can be
+ obtained from Turtle Beach. See
+ <file:Documentation/sound/oss/MultiSound> for information on how to
+ obtain this.
+
+config MSNDCLAS_IRQ
+ int "MSND Classic IRQ 5, 7, 9, 10, 11, 12"
+ depends on SOUND_MSNDCLAS=y
+ default "5"
+ help
+ Interrupt Request line for the MultiSound Classic and related cards.
+
+config MSNDCLAS_MEM
+ hex "MSND Classic memory B0000, C8000, D0000, D8000, E0000, E8000"
+ depends on SOUND_MSNDCLAS=y
+ default "D0000"
+ help
+ Memory-mapped I/O base address for the MultiSound Classic and
+ related cards.
+
+config MSNDCLAS_IO
+ hex "MSND Classic I/O 210, 220, 230, 240, 250, 260, 290, 3E0"
+ depends on SOUND_MSNDCLAS=y
+ default "290"
+ help
+ I/O port address for the MultiSound Classic and related cards.
+
+config SOUND_MSNDPIN
+ tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji"
+ depends on (m || !STANDALONE) && ISA
+ help
+ Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji.
+ See <file:Documentation/sound/oss/MultiSound> for important information
+ about this driver. Note that it has been discontinued, but the
+ Voyetra Turtle Beach knowledge base entry for it is still available
+ at <http://www.turtlebeach.com/site/kb_ftp/600.asp>.
+
+comment "Compiled-in MSND Pinnacle support requires firmware during compilation."
+ depends on SOUND_PRIME && SOUND_MSNDPIN=y
+
+config MSNDPIN_HAVE_BOOT
+ bool
+ depends on SOUND_MSNDPIN=y
+ default y
+
+config MSNDPIN_INIT_FILE
+ string "Full pathname of PNDSPINI.BIN firmware file"
+ depends on SOUND_MSNDPIN
+ default "/etc/sound/pndspini.bin"
+ help
+ The MultiSound cards have two firmware files which are required
+ for operation, and are not currently included. These files can be
+ obtained from Turtle Beach. See
+ <file:Documentation/sound/oss/MultiSound> for information on how to
+ obtain this.
+
+config MSNDPIN_PERM_FILE
+ string "Full pathname of PNDSPERM.BIN firmware file"
+ depends on SOUND_MSNDPIN
+ default "/etc/sound/pndsperm.bin"
+ help
+ The MultiSound cards have two firmware files which are required for
+ operation, and are not currently included. These files can be
+ obtained from Turtle Beach. See
+ <file:Documentation/sound/oss/MultiSound> for information on how to
+ obtain this.
+
+config MSNDPIN_IRQ
+ int "MSND Pinnacle IRQ 5, 7, 9, 10, 11, 12"
+ depends on SOUND_MSNDPIN=y
+ default "5"
+ help
+ Interrupt request line for the primary synthesizer on MultiSound
+ Pinnacle and Fiji sound cards.
+
+config MSNDPIN_MEM
+ hex "MSND Pinnacle memory B0000, C8000, D0000, D8000, E0000, E8000"
+ depends on SOUND_MSNDPIN=y
+ default "D0000"
+ help
+ Memory-mapped I/O base address for the primary synthesizer on
+ MultiSound Pinnacle and Fiji sound cards.
+
+config MSNDPIN_IO
+ hex "MSND Pinnacle I/O 210, 220, 230, 240, 250, 260, 290, 3E0"
+ depends on SOUND_MSNDPIN=y
+ default "290"
+ help
+ Memory-mapped I/O base address for the primary synthesizer on
+ MultiSound Pinnacle and Fiji sound cards.
+
+config MSNDPIN_DIGITAL
+ bool "MSND Pinnacle has S/PDIF I/O"
+ depends on SOUND_MSNDPIN=y
+ help
+ If you have the S/PDIF daughter board for the Pinnacle or Fiji,
+ answer Y here; otherwise, say N. If you have this, you will be able
+ to play and record from the S/PDIF port (digital signal). See
+ <file:Documentation/sound/oss/MultiSound> for information on how to make
+ use of this capability.
+
+config MSNDPIN_NONPNP
+ bool "MSND Pinnacle non-PnP Mode"
+ depends on SOUND_MSNDPIN=y
+ help
+ The Pinnacle and Fiji card resources can be configured either with
+ PnP, or through a configuration port. Say Y here if your card is NOT
+ in PnP mode. For the Pinnacle, configuration in non-PnP mode allows
+ use of the IDE and joystick peripherals on the card as well; these
+ do not show up when the card is in PnP mode. Specifying zero for any
+ resource of a device will disable the device. If you are running the
+ card in PnP mode, you must say N here and use isapnptools to
+ configure the card's resources.
+
+comment "MSND Pinnacle DSP section will be configured to above parameters."
+ depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP
+
+config MSNDPIN_CFG
+ hex "MSND Pinnacle config port 250,260,270"
+ depends on MSNDPIN_NONPNP
+ default "250"
+ help
+ This is the port which the Pinnacle and Fiji uses to configure the
+ card's resources when not in PnP mode. If your card is in PnP mode,
+ then be sure to say N to the previous option, "MSND Pinnacle Non-PnP
+ Mode".
+
+comment "Pinnacle-specific Device Configuration (0 disables)"
+ depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP
+
+config MSNDPIN_MPU_IO
+ hex "MSND Pinnacle MPU I/O (e.g. 330)"
+ depends on MSNDPIN_NONPNP
+ default "0"
+ help
+ Memory-mapped I/O base address for the Kurzweil daughterboard
+ synthesizer on MultiSound Pinnacle and Fiji sound cards.
+
+config MSNDPIN_MPU_IRQ
+ int "MSND Pinnacle MPU IRQ (e.g. 9)"
+ depends on MSNDPIN_NONPNP
+ default "0"
+ help
+ Interrupt request number for the Kurzweil daughterboard
+ synthesizer on MultiSound Pinnacle and Fiji sound cards.
+
+config MSNDPIN_IDE_IO0
+ hex "MSND Pinnacle IDE I/O 0 (e.g. 170)"
+ depends on MSNDPIN_NONPNP
+ default "0"
+ help
+ CD-ROM drive 0 memory-mapped I/O base address for the MultiSound
+ Pinnacle and Fiji sound cards.
+
+config MSNDPIN_IDE_IO1
+ hex "MSND Pinnacle IDE I/O 1 (e.g. 376)"
+ depends on MSNDPIN_NONPNP
+ default "0"
+ help
+ CD-ROM drive 1 memory-mapped I/O base address for the MultiSound
+ Pinnacle and Fiji sound cards.
+
+config MSNDPIN_IDE_IRQ
+ int "MSND Pinnacle IDE IRQ (e.g. 15)"
+ depends on MSNDPIN_NONPNP
+ default "0"
+ help
+ Interrupt request number for the IDE CD-ROM interface on the
+ MultiSound Pinnacle and Fiji sound cards.
+
+config MSNDPIN_JOYSTICK_IO
+ hex "MSND Pinnacle joystick I/O (e.g. 200)"
+ depends on MSNDPIN_NONPNP
+ default "0"
+ help
+ Memory-mapped I/O base address for the joystick port on MultiSound
+ Pinnacle and Fiji sound cards.
+
+config MSND_FIFOSIZE
+ int "MSND buffer size (kB)"
+ depends on SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y
+ default "128"
+ help
+ Configures the size of each audio buffer, in kilobytes, for
+ recording and playing in the MultiSound drivers (both the Classic
+ and Pinnacle). Larger values reduce the chance of data overruns at
+ the expense of overall latency. If unsure, use the default.
+
+menuconfig SOUND_OSS
+ tristate "OSS sound modules"
+ depends on ISA_DMA_API && VIRT_TO_BUS
+ help
+ OSS is the Open Sound System suite of sound card drivers. They make
+ sound programming easier since they provide a common API. Say Y or
+ M here (the module will be called sound) if you haven't found a
+ driver for your sound card above, then pick your driver from the
+ list below.
+
+if SOUND_OSS
+
+config SOUND_TRACEINIT
+ bool "Verbose initialisation"
+ help
+ Verbose soundcard initialization -- affects the format of autoprobe
+ and initialization messages at boot time.
+
+config SOUND_DMAP
+ bool "Persistent DMA buffers"
+ ---help---
+ Linux can often have problems allocating DMA buffers for ISA sound
+ cards on machines with more than 16MB of RAM. This is because ISA
+ DMA buffers must exist below the 16MB boundary and it is quite
+ possible that a large enough free block in this region cannot be
+ found after the machine has been running for a while. If you say Y
+ here the DMA buffers (64Kb) will be allocated at boot time and kept
+ until the shutdown. This option is only useful if you said Y to
+ "OSS sound modules", above. If you said M to "OSS sound modules"
+ then you can get the persistent DMA buffer functionality by passing
+ the command-line argument "dmabuf=1" to the sound module.
+
+ Say Y unless you have 16MB or more RAM or a PCI sound card.
+
+config SOUND_SSCAPE
+ tristate "Ensoniq SoundScape support"
+ help
+ Answer Y if you have a sound card based on the Ensoniq SoundScape
+ chipset. Such cards are being manufactured at least by Ensoniq, Spea
+ and Reveal (Reveal makes also other cards).
+
+ If you compile the driver into the kernel, you have to add
+ "sscape=<io>,<irq>,<dma>,<mpuio>,<mpuirq>" to the kernel command
+ line.
+
+
+config SOUND_VMIDI
+ tristate "Loopback MIDI device support"
+ help
+ Support for MIDI loopback on port 1 or 2.
+
+config SOUND_TRIX
+ tristate "MediaTrix AudioTrix Pro support"
+ help
+ Answer Y if you have the AudioTriX Pro sound card manufactured
+ by MediaTrix.
+
+config TRIX_HAVE_BOOT
+ bool "Have TRXPRO.HEX firmware file"
+ depends on SOUND_TRIX=y && !STANDALONE
+ help
+ The MediaTrix AudioTrix Pro has an on-board microcontroller which
+ needs to be initialized by downloading the code from the file
+ TRXPRO.HEX in the DOS driver directory. If you don't have the
+ TRXPRO.HEX file handy you may skip this step. However, the SB and
+ MPU-401 modes of AudioTrix Pro will not work without this file!
+
+config TRIX_BOOT_FILE
+ string "Full pathname of TRXPRO.HEX firmware file"
+ depends on TRIX_HAVE_BOOT
+ default "/etc/sound/trxpro.hex"
+ help
+ Enter the full pathname of your TRXPRO.HEX file, starting from /.
+
+config SOUND_MSS
+ tristate "Microsoft Sound System support"
+ ---help---
+ Again think carefully before answering Y to this question. It's
+ safe to answer Y if you have the original Windows Sound System card
+ made by Microsoft or Aztech SG 16 Pro (or NX16 Pro). Also you may
+ say Y in case your card is NOT among these:
+
+ ATI Stereo F/X, AdLib, Audio Excell DSP16, Cardinal DSP16,
+ Ensoniq SoundScape (and compatibles made by Reveal and Spea),
+ Gravis Ultrasound, Gravis Ultrasound ACE, Gravis Ultrasound Max,
+ Gravis Ultrasound with 16 bit option, Logitech Sound Man 16,
+ Logitech SoundMan Games, Logitech SoundMan Wave, MAD16 Pro (OPTi
+ 82C929), Media Vision Jazz16, MediaTriX AudioTriX Pro, Microsoft
+ Windows Sound System (MSS/WSS), Mozart (OAK OTI-601), Orchid
+ SW32, Personal Sound System (PSS), Pro Audio Spectrum 16, Pro
+ Audio Studio 16, Pro Sonic 16, Roland MPU-401 MIDI interface,
+ Sound Blaster 1.0, Sound Blaster 16, Sound Blaster 16ASP, Sound
+ Blaster 2.0, Sound Blaster AWE32, Sound Blaster Pro, TI TM4000M
+ notebook, ThunderBoard, Turtle Beach Tropez, Yamaha FM
+ synthesizers (OPL2, OPL3 and OPL4), 6850 UART MIDI Interface.
+
+ For cards having native support in VoxWare, consult the card
+ specific instructions in <file:Documentation/sound/oss/README.OSS>.
+ Some drivers have their own MSS support and saying Y to this option
+ will cause a conflict.
+
+ If you compile the driver into the kernel, you have to add
+ "ad1848=<io>,<irq>,<dma>,<dma2>[,<type>]" to the kernel command
+ line.
+
+config SOUND_MPU401
+ tristate "MPU-401 support (NOT for SB16)"
+ ---help---
+ Be careful with this question. The MPU401 interface is supported by
+ all sound cards. However, some natively supported cards have their
+ own driver for MPU401. Enabling this MPU401 option with these cards
+ will cause a conflict. Also, enabling MPU401 on a system that
+ doesn't really have a MPU401 could cause some trouble. If your card
+ was in the list of supported cards, look at the card specific
+ instructions in the <file:Documentation/sound/oss/README.OSS> file. It
+ is safe to answer Y if you have a true MPU401 MIDI interface card.
+
+ If you compile the driver into the kernel, you have to add
+ "mpu401=<io>,<irq>" to the kernel command line.
+
+config SOUND_PAS
+ tristate "ProAudioSpectrum 16 support"
+ ---help---
+ Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio
+ 16 or Logitech SoundMan 16 sound card. Answer N if you have some
+ other card made by Media Vision or Logitech since those are not
+ PAS16 compatible. Please read <file:Documentation/sound/oss/PAS16>.
+ It is not necessary to add Sound Blaster support separately; it
+ is included in PAS support.
+
+ If you compile the driver into the kernel, you have to add
+ "pas2=<io>,<irq>,<dma>,<dma2>,<sbio>,<sbirq>,<sbdma>,<sbdma2>
+ to the kernel command line.
+
+config PAS_JOYSTICK
+ bool "Enable PAS16 joystick port"
+ depends on SOUND_PAS=y
+ help
+ Say Y here to enable the Pro Audio Spectrum 16's auxiliary joystick
+ port.
+
+config SOUND_PSS
+ tristate "PSS (AD1848, ADSP-2115, ESC614) support"
+ help
+ Answer Y or M if you have an Orchid SW32, Cardinal DSP16, Beethoven
+ ADSP-16 or some other card based on the PSS chipset (AD1848 codec +
+ ADSP-2115 DSP chip + Echo ESC614 ASIC CHIP). For more information on
+ how to compile it into the kernel or as a module see the file
+ <file:Documentation/sound/oss/PSS>.
+
+ If you compile the driver into the kernel, you have to add
+ "pss=<io>,<mssio>,<mssirq>,<mssdma>,<mpuio>,<mpuirq>" to the kernel
+ command line.
+
+config PSS_MIXER
+ bool "Enable PSS mixer (Beethoven ADSP-16 and other compatible)"
+ depends on SOUND_PSS
+ help
+ Answer Y for Beethoven ADSP-16. You may try to say Y also for other
+ cards if they have master volume, bass, treble, and you can't
+ control it under Linux. If you answer N for Beethoven ADSP-16, you
+ can't control master volume, bass, treble and synth volume.
+
+ If you said M to "PSS support" above, you may enable or disable this
+ PSS mixer with the module parameter pss_mixer. For more information
+ see the file <file:Documentation/sound/oss/PSS>.
+
+config PSS_HAVE_BOOT
+ bool "Have DSPxxx.LD firmware file"
+ depends on SOUND_PSS && !STANDALONE
+ help
+ If you have the DSPxxx.LD file or SYNTH.LD file for you card, say Y
+ to include this file. Without this file the synth device (OPL) may
+ not work.
+
+config PSS_BOOT_FILE
+ string "Full pathname of DSPxxx.LD firmware file"
+ depends on PSS_HAVE_BOOT
+ default "/etc/sound/dsp001.ld"
+ help
+ Enter the full pathname of your DSPxxx.LD file or SYNTH.LD file,
+ starting from /.
+
+config SOUND_SB
+ tristate "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support"
+ ---help---
+ Answer Y if you have an original Sound Blaster card made by Creative
+ Labs or a 100% hardware compatible clone (like the Thunderboard or
+ SM Games). For an unknown card you may answer Y if the card claims
+ to be Sound Blaster-compatible.
+
+ Please read the file <file:Documentation/sound/oss/Soundblaster>.
+
+ You should also say Y here for cards based on the Avance Logic
+ ALS-007 and ALS-1X0 chips (read <file:Documentation/sound/oss/ALS>) and
+ for cards based on ESS chips (read
+ <file:Documentation/sound/oss/ESS1868> and
+ <file:Documentation/sound/oss/ESS>). If you have an SB AWE 32 or SB AWE
+ 64, say Y here and also to "AWE32 synth" below and read
+ <file:Documentation/sound/oss/INSTALL.awe>. If you have an IBM Mwave
+ card, say Y here and read <file:Documentation/sound/oss/mwave>.
+
+ If you compile the driver into the kernel and don't want to use
+ isapnp, you have to add "sb=<io>,<irq>,<dma>,<dma2>" to the kernel
+ command line.
+
+ You can say M here to compile this driver as a module; the module is
+ called sb.
+
+config SOUND_YM3812
+ tristate "Yamaha FM synthesizer (YM3812/OPL-3) support"
+ ---help---
+ Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
+ Answering Y is usually a safe and recommended choice, however some
+ cards may have software (TSR) FM emulation. Enabling FM support with
+ these cards may cause trouble (I don't currently know of any such
+ cards, however). Please read the file
+ <file:Documentation/sound/oss/OPL3> if your card has an OPL3 chip.
+
+ If you compile the driver into the kernel, you have to add
+ "opl3=<io>" to the kernel command line.
+
+ If unsure, say Y.
+
+config SOUND_UART6850
+ tristate "6850 UART support"
+ help
+ This option enables support for MIDI interfaces based on the 6850
+ UART chip. This interface is rarely found on sound cards. It's safe
+ to answer N to this question.
+
+ If you compile the driver into the kernel, you have to add
+ "uart6850=<io>,<irq>" to the kernel command line.
+
+config SOUND_AEDSP16
+ tristate "Gallant Audio Cards (SC-6000 and SC-6600 based)"
+ ---help---
+ Answer Y if you have a Gallant's Audio Excel DSP 16 card. This
+ driver supports Audio Excel DSP 16 but not the III nor PnP versions
+ of this card.
+
+ The Gallant's Audio Excel DSP 16 card can emulate either an SBPro or
+ a Microsoft Sound System card, so you should have said Y to either
+ "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support"
+ or "Microsoft Sound System support", above, and you need to answer
+ the "MSS emulation" and "SBPro emulation" questions below
+ accordingly. You should say Y to one and only one of these two
+ questions.
+
+ Read the <file:Documentation/sound/oss/README.OSS> file and the head of
+ <file:sound/oss/aedsp16.c> as well as
+ <file:Documentation/sound/oss/AudioExcelDSP16> to get more information
+ about this driver and its configuration.
+
+config SC6600
+ bool "SC-6600 based audio cards (new Audio Excel DSP 16)"
+ depends on SOUND_AEDSP16
+ help
+ The SC6600 is the new version of DSP mounted on the Audio Excel DSP
+ 16 cards. Find in the manual the FCC ID of your audio card and
+ answer Y if you have an SC6600 DSP.
+
+config SC6600_JOY
+ bool "Activate SC-6600 Joystick Interface"
+ depends on SC6600
+ help
+ Say Y here in order to use the joystick interface of the Audio Excel
+ DSP 16 card.
+
+config SC6600_CDROM
+ int "SC-6600 CDROM Interface (4=None, 3=IDE, 1=Panasonic, 0=?Sony?)"
+ depends on SC6600
+ default "4"
+ help
+ This is used to activate the CD-ROM interface of the Audio Excel
+ DSP 16 card. Enter: 0 for Sony, 1 for Panasonic, 2 for IDE, 4 for no
+ CD-ROM present.
+
+config SC6600_CDROMBASE
+ hex "SC-6600 CDROM Interface I/O Address"
+ depends on SC6600
+ default "0"
+ help
+ Base I/O port address for the CD-ROM interface of the Audio Excel
+ DSP 16 card.
+
+config SOUND_VIDC
+ tristate "VIDC 16-bit sound"
+ depends on ARM && (ARCH_ACORN || ARCH_CLPS7500)
+ help
+ 16-bit support for the VIDC onboard sound hardware found on Acorn
+ machines.
+
+config SOUND_WAVEARTIST
+ tristate "Netwinder WaveArtist"
+ depends on ARM && ARCH_NETWINDER
+ help
+ Say Y here to include support for the Rockwell WaveArtist sound
+ system. This driver is mainly for the NetWinder.
+
+config SOUND_KAHLUA
+ tristate "XpressAudio Sound Blaster emulation"
+ depends on SOUND_SB
+
+endif # SOUND_OSS
+
+config SOUND_SH_DAC_AUDIO
+ tristate "SuperH DAC audio support"
+ depends on CPU_SH3
+
+config SOUND_SH_DAC_AUDIO_CHANNEL
+ int "DAC channel"
+ default "1"
+ depends on SOUND_SH_DAC_AUDIO
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
new file mode 100644
index 0000000..e0ae4d4
--- /dev/null
+++ b/sound/oss/Makefile
@@ -0,0 +1,111 @@
+# Makefile for the Linux sound card driver
+#
+# 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net>
+# Rewritten to use lists instead of if-statements.
+
+# Each configuration option enables a list of files.
+
+obj-$(CONFIG_SOUND_OSS) += sound.o
+
+# Please leave it as is, cause the link order is significant !
+
+obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o
+obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o
+obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o
+obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o
+obj-$(CONFIG_SOUND_SSCAPE) += sscape.o ad1848.o mpu401.o
+obj-$(CONFIG_SOUND_MSS) += ad1848.o
+obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o
+obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o
+obj-$(CONFIG_SOUND_KAHLUA) += kahlua.o
+obj-$(CONFIG_SOUND_MPU401) += mpu401.o
+obj-$(CONFIG_SOUND_UART6850) += uart6850.o
+obj-$(CONFIG_SOUND_YM3812) += opl3.o
+obj-$(CONFIG_SOUND_VMIDI) += v_midi.o
+obj-$(CONFIG_SOUND_VIDC) += vidc_mod.o
+obj-$(CONFIG_SOUND_WAVEARTIST) += waveartist.o
+obj-$(CONFIG_SOUND_MSNDCLAS) += msnd.o msnd_classic.o
+obj-$(CONFIG_SOUND_MSNDPIN) += msnd.o msnd_pinnacle.o
+obj-$(CONFIG_SOUND_VWSND) += vwsnd.o
+obj-$(CONFIG_SOUND_AU1550_AC97) += au1550_ac97.o ac97_codec.o
+obj-$(CONFIG_SOUND_BCM_CS4297A) += swarm_cs4297a.o
+
+obj-$(CONFIG_DMASOUND) += dmasound/
+
+# Declare multi-part drivers.
+
+sound-objs := \
+ dev_table.o soundcard.o \
+ audio.o dmabuf.o \
+ midi_synth.o midibuf.o \
+ sequencer.o sound_timer.o sys_timer.o
+
+pas2-objs := pas2_card.o pas2_midi.o pas2_mixer.o pas2_pcm.o
+sb-objs := sb_card.o
+sb_lib-objs := sb_common.o sb_audio.o sb_midi.o sb_mixer.o sb_ess.o
+vidc_mod-objs := vidc.o vidc_fill.o
+
+hostprogs-y := bin2hex hex2hex
+
+# Files generated that shall be removed upon make clean
+clean-files := msndperm.c msndinit.c pndsperm.c pndspini.c \
+ pss_boot.h trix_boot.h
+
+# Firmware files that need translation
+#
+# The translated files are protected by a file that keeps track
+# of what name was used to build them. If the name changes, they
+# will be forced to be remade.
+#
+
+# Turtle Beach MultiSound
+
+ifeq ($(CONFIG_MSNDCLAS_HAVE_BOOT),y)
+ $(obj)/msnd_classic.o: $(obj)/msndperm.c $(obj)/msndinit.c
+
+ $(obj)/msndperm.c: $(patsubst "%", %, $(CONFIG_MSNDCLAS_PERM_FILE)) $(obj)/bin2hex
+ $(obj)/bin2hex msndperm < $< > $@
+
+ $(obj)/msndinit.c: $(patsubst "%", %, $(CONFIG_MSNDCLAS_INIT_FILE)) $(obj)/bin2hex
+ $(obj)/bin2hex msndinit < $< > $@
+endif
+
+ifeq ($(CONFIG_MSNDPIN_HAVE_BOOT),y)
+ $(obj)/msnd_pinnacle.o: $(obj)/pndsperm.c $(obj)/pndspini.c
+
+ $(obj)/pndsperm.c: $(patsubst "%", %, $(CONFIG_MSNDPIN_PERM_FILE)) $(obj)/bin2hex
+ $(obj)/bin2hex pndsperm < $< > $@
+
+ $(obj)/pndspini.c: $(patsubst "%", %, $(CONFIG_MSNDPIN_INIT_FILE)) $(obj)/bin2hex
+ $(obj)/bin2hex pndspini < $< > $@
+endif
+
+# PSS (ECHO-ADI2111)
+
+$(obj)/pss.o: $(obj)/pss_boot.h
+
+ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
+ $(obj)/pss_boot.h: $(patsubst "%", %, $(CONFIG_PSS_BOOT_FILE)) $(obj)/bin2hex
+ $(obj)/bin2hex pss_synth < $< > $@
+else
+ $(obj)/pss_boot.h:
+ ( \
+ echo 'static unsigned char * pss_synth = NULL;'; \
+ echo 'static int pss_synthLen = 0;'; \
+ ) > $@
+endif
+
+# MediaTrix AudioTrix Pro
+
+$(obj)/trix.o: $(obj)/trix_boot.h
+
+ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
+ $(obj)/trix_boot.h: $(patsubst "%", %, $(CONFIG_TRIX_BOOT_FILE)) $(obj)/hex2hex
+ $(obj)/hex2hex -i trix_boot < $< > $@
+else
+ $(obj)/trix_boot.h:
+ ( \
+ echo 'static unsigned char * trix_boot = NULL;'; \
+ echo 'static int trix_boot_len = 0;'; \
+ ) > $@
+endif
diff --git a/sound/oss/README.FIRST b/sound/oss/README.FIRST
new file mode 100644
index 0000000..90fdcf0
--- /dev/null
+++ b/sound/oss/README.FIRST
@@ -0,0 +1,6 @@
+The modular sound driver patches were funded by Red Hat Software
+(www.redhat.com). The sound driver here is thus a modified version of
+Hannu's code. Please bear that in mind when considering the appropriate
+forums for bug reporting.
+
+Alan Cox
diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c
new file mode 100644
index 0000000..456a1b4
--- /dev/null
+++ b/sound/oss/ac97_codec.c
@@ -0,0 +1,1206 @@
+/*
+ * ac97_codec.c: Generic AC97 mixer/modem module
+ *
+ * Derived from ac97 mixer in maestro and trident driver.
+ *
+ * Copyright 2000 Silicon Integrated System Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ **************************************************************************
+ *
+ * The Intel Audio Codec '97 specification is available at the Intel
+ * audio homepage: http://developer.intel.com/ial/scalableplatforms/audio/
+ *
+ * The specification itself is currently available at:
+ * ftp://download.intel.com/ial/scalableplatforms/ac97r22.pdf
+ *
+ **************************************************************************
+ *
+ * History
+ * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
+ * Removed non existant WM9700
+ * Added support for WM9705, WM9708, WM9709, WM9710, WM9711
+ * WM9712 and WM9717
+ * Mar 28, 2002 Randolph Bentson <bentson@holmsjoen.com>
+ * corrections to support WM9707 in ViewPad 1000
+ * v0.4 Mar 15 2000 Ollie Lho
+ * dual codecs support verified with 4 channels output
+ * v0.3 Feb 22 2000 Ollie Lho
+ * bug fix for record mask setting
+ * v0.2 Feb 10 2000 Ollie Lho
+ * add ac97_read_proc for /proc/driver/{vendor}/ac97
+ * v0.1 Jan 14 2000 Ollie Lho <ollie@sis.com.tw>
+ * Isolated from trident.c to support multiple ac97 codec
+ */
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <linux/errno.h>
+#include <linux/bitops.h>
+#include <linux/delay.h>
+#include <linux/pci.h>
+#include <linux/ac97_codec.h>
+#include <asm/uaccess.h>
+#include <linux/mutex.h>
+
+#define CODEC_ID_BUFSZ 14
+
+static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel);
+static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel,
+ unsigned int left, unsigned int right);
+static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val );
+static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask);
+static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg);
+
+static int ac97_init_mixer(struct ac97_codec *codec);
+
+static int wolfson_init03(struct ac97_codec * codec);
+static int wolfson_init04(struct ac97_codec * codec);
+static int wolfson_init05(struct ac97_codec * codec);
+static int wolfson_init11(struct ac97_codec * codec);
+static int wolfson_init13(struct ac97_codec * codec);
+static int tritech_init(struct ac97_codec * codec);
+static int tritech_maestro_init(struct ac97_codec * codec);
+static int sigmatel_9708_init(struct ac97_codec *codec);
+static int sigmatel_9721_init(struct ac97_codec *codec);
+static int sigmatel_9744_init(struct ac97_codec *codec);
+static int ad1886_init(struct ac97_codec *codec);
+static int eapd_control(struct ac97_codec *codec, int);
+static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
+static int cmedia_init(struct ac97_codec * codec);
+static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
+static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode);
+
+
+/*
+ * AC97 operations.
+ *
+ * If you are adding a codec then you should be able to use
+ * eapd_ops - any codec that supports EAPD amp control (most)
+ * null_ops - any ancient codec that supports nothing
+ *
+ * The three functions are
+ * init - used for non AC97 standard initialisation
+ * amplifier - used to do amplifier control (1=on 0=off)
+ * digital - switch to digital modes (0 = analog)
+ *
+ * Not all codecs support all features, not all drivers use all the
+ * operations yet
+ */
+
+static struct ac97_ops null_ops = { NULL, NULL, NULL };
+static struct ac97_ops default_ops = { NULL, eapd_control, NULL };
+static struct ac97_ops default_digital_ops = { NULL, eapd_control, generic_digital_control};
+static struct ac97_ops wolfson_ops03 = { wolfson_init03, NULL, NULL };
+static struct ac97_ops wolfson_ops04 = { wolfson_init04, NULL, NULL };
+static struct ac97_ops wolfson_ops05 = { wolfson_init05, NULL, NULL };
+static struct ac97_ops wolfson_ops11 = { wolfson_init11, NULL, NULL };
+static struct ac97_ops wolfson_ops13 = { wolfson_init13, NULL, NULL };
+static struct ac97_ops tritech_ops = { tritech_init, NULL, NULL };
+static struct ac97_ops tritech_m_ops = { tritech_maestro_init, NULL, NULL };
+static struct ac97_ops sigmatel_9708_ops = { sigmatel_9708_init, NULL, NULL };
+static struct ac97_ops sigmatel_9721_ops = { sigmatel_9721_init, NULL, NULL };
+static struct ac97_ops sigmatel_9744_ops = { sigmatel_9744_init, NULL, NULL };
+static struct ac97_ops crystal_digital_ops = { NULL, eapd_control, crystal_digital_control };
+static struct ac97_ops ad1886_ops = { ad1886_init, eapd_control, NULL };
+static struct ac97_ops cmedia_ops = { NULL, eapd_control, NULL};
+static struct ac97_ops cmedia_digital_ops = { cmedia_init, eapd_control, cmedia_digital_control};
+
+/* sorted by vendor/device id */
+static const struct {
+ u32 id;
+ char *name;
+ struct ac97_ops *ops;
+ int flags;
+} ac97_codec_ids[] = {
+ {0x41445303, "Analog Devices AD1819", &null_ops},
+ {0x41445340, "Analog Devices AD1881", &null_ops},
+ {0x41445348, "Analog Devices AD1881A", &null_ops},
+ {0x41445360, "Analog Devices AD1885", &default_ops},
+ {0x41445361, "Analog Devices AD1886", &ad1886_ops},
+ {0x41445370, "Analog Devices AD1981", &null_ops},
+ {0x41445372, "Analog Devices AD1981A", &null_ops},
+ {0x41445374, "Analog Devices AD1981B", &null_ops},
+ {0x41445460, "Analog Devices AD1885", &default_ops},
+ {0x41445461, "Analog Devices AD1886", &ad1886_ops},
+ {0x414B4D00, "Asahi Kasei AK4540", &null_ops},
+ {0x414B4D01, "Asahi Kasei AK4542", &null_ops},
+ {0x414B4D02, "Asahi Kasei AK4543", &null_ops},
+ {0x414C4326, "ALC100P", &null_ops},
+ {0x414C4710, "ALC200/200P", &null_ops},
+ {0x414C4720, "ALC650", &default_digital_ops},
+ {0x434D4941, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME },
+ {0x434D4942, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME },
+ {0x434D4961, "CMedia", &cmedia_digital_ops, AC97_NO_PCM_VOLUME },
+ {0x43525900, "Cirrus Logic CS4297", &default_ops},
+ {0x43525903, "Cirrus Logic CS4297", &default_ops},
+ {0x43525913, "Cirrus Logic CS4297A rev A", &default_ops},
+ {0x43525914, "Cirrus Logic CS4297A rev B", &default_ops},
+ {0x43525923, "Cirrus Logic CS4298", &null_ops},
+ {0x4352592B, "Cirrus Logic CS4294", &null_ops},
+ {0x4352592D, "Cirrus Logic CS4294", &null_ops},
+ {0x43525931, "Cirrus Logic CS4299 rev A", &crystal_digital_ops},
+ {0x43525933, "Cirrus Logic CS4299 rev C", &crystal_digital_ops},
+ {0x43525934, "Cirrus Logic CS4299 rev D", &crystal_digital_ops},
+ {0x43585430, "CXT48", &default_ops, AC97_DELUDED_MODEM },
+ {0x43585442, "CXT66", &default_ops, AC97_DELUDED_MODEM },
+ {0x44543031, "Diamond Technology DT0893", &default_ops},
+ {0x45838308, "ESS Allegro ES1988", &null_ops},
+ {0x49434511, "ICE1232", &null_ops}, /* I hope --jk */
+ {0x4e534331, "National Semiconductor LM4549", &null_ops},
+ {0x53494c22, "Silicon Laboratory Si3036", &null_ops},
+ {0x53494c23, "Silicon Laboratory Si3038", &null_ops},
+ {0x545200FF, "TriTech TR?????", &tritech_m_ops},
+ {0x54524102, "TriTech TR28022", &null_ops},
+ {0x54524103, "TriTech TR28023", &null_ops},
+ {0x54524106, "TriTech TR28026", &null_ops},
+ {0x54524108, "TriTech TR28028", &tritech_ops},
+ {0x54524123, "TriTech TR A5", &null_ops},
+ {0x574D4C03, "Wolfson WM9703/07/08/17", &wolfson_ops03},
+ {0x574D4C04, "Wolfson WM9704M/WM9704Q", &wolfson_ops04},
+ {0x574D4C05, "Wolfson WM9705/WM9710", &wolfson_ops05},
+ {0x574D4C09, "Wolfson WM9709", &null_ops},
+ {0x574D4C12, "Wolfson WM9711/9712", &wolfson_ops11},
+ {0x574D4C13, "Wolfson WM9713", &wolfson_ops13, AC97_DEFAULT_POWER_OFF},
+ {0x83847600, "SigmaTel STAC????", &null_ops},
+ {0x83847604, "SigmaTel STAC9701/3/4/5", &null_ops},
+ {0x83847605, "SigmaTel STAC9704", &null_ops},
+ {0x83847608, "SigmaTel STAC9708", &sigmatel_9708_ops},
+ {0x83847609, "SigmaTel STAC9721/23", &sigmatel_9721_ops},
+ {0x83847644, "SigmaTel STAC9744/45", &sigmatel_9744_ops},
+ {0x83847652, "SigmaTel STAC9752/53", &default_ops},
+ {0x83847656, "SigmaTel STAC9756/57", &sigmatel_9744_ops},
+ {0x83847666, "SigmaTel STAC9750T", &sigmatel_9744_ops},
+ {0x83847684, "SigmaTel STAC9783/84?", &null_ops},
+ {0x57454301, "Winbond 83971D", &null_ops},
+};
+
+/* this table has default mixer values for all OSS mixers. */
+static struct mixer_defaults {
+ int mixer;
+ unsigned int value;
+} mixer_defaults[SOUND_MIXER_NRDEVICES] = {
+ /* all values 0 -> 100 in bytes */
+ {SOUND_MIXER_VOLUME, 0x4343},
+ {SOUND_MIXER_BASS, 0x4343},
+ {SOUND_MIXER_TREBLE, 0x4343},
+ {SOUND_MIXER_PCM, 0x4343},
+ {SOUND_MIXER_SPEAKER, 0x4343},
+ {SOUND_MIXER_LINE, 0x4343},
+ {SOUND_MIXER_MIC, 0x0000},
+ {SOUND_MIXER_CD, 0x4343},
+ {SOUND_MIXER_ALTPCM, 0x4343},
+ {SOUND_MIXER_IGAIN, 0x4343},
+ {SOUND_MIXER_LINE1, 0x4343},
+ {SOUND_MIXER_PHONEIN, 0x4343},
+ {SOUND_MIXER_PHONEOUT, 0x4343},
+ {SOUND_MIXER_VIDEO, 0x4343},
+ {-1,0}
+};
+
+/* table to scale scale from OSS mixer value to AC97 mixer register value */
+static struct ac97_mixer_hw {
+ unsigned char offset;
+ int scale;
+} ac97_hw[SOUND_MIXER_NRDEVICES]= {
+ [SOUND_MIXER_VOLUME] = {AC97_MASTER_VOL_STEREO,64},
+ [SOUND_MIXER_BASS] = {AC97_MASTER_TONE, 16},
+ [SOUND_MIXER_TREBLE] = {AC97_MASTER_TONE, 16},
+ [SOUND_MIXER_PCM] = {AC97_PCMOUT_VOL, 32},
+ [SOUND_MIXER_SPEAKER] = {AC97_PCBEEP_VOL, 16},
+ [SOUND_MIXER_LINE] = {AC97_LINEIN_VOL, 32},
+ [SOUND_MIXER_MIC] = {AC97_MIC_VOL, 32},
+ [SOUND_MIXER_CD] = {AC97_CD_VOL, 32},
+ [SOUND_MIXER_ALTPCM] = {AC97_HEADPHONE_VOL, 64},
+ [SOUND_MIXER_IGAIN] = {AC97_RECORD_GAIN, 16},
+ [SOUND_MIXER_LINE1] = {AC97_AUX_VOL, 32},
+ [SOUND_MIXER_PHONEIN] = {AC97_PHONE_VOL, 32},
+ [SOUND_MIXER_PHONEOUT] = {AC97_MASTER_VOL_MONO, 64},
+ [SOUND_MIXER_VIDEO] = {AC97_VIDEO_VOL, 32},
+};
+
+/* the following tables allow us to go from OSS <-> ac97 quickly. */
+enum ac97_recsettings {
+ AC97_REC_MIC=0,
+ AC97_REC_CD,
+ AC97_REC_VIDEO,
+ AC97_REC_AUX,
+ AC97_REC_LINE,
+ AC97_REC_STEREO, /* combination of all enabled outputs.. */
+ AC97_REC_MONO, /*.. or the mono equivalent */
+ AC97_REC_PHONE
+};
+
+static const unsigned int ac97_rm2oss[] = {
+ [AC97_REC_MIC] = SOUND_MIXER_MIC,
+ [AC97_REC_CD] = SOUND_MIXER_CD,
+ [AC97_REC_VIDEO] = SOUND_MIXER_VIDEO,
+ [AC97_REC_AUX] = SOUND_MIXER_LINE1,
+ [AC97_REC_LINE] = SOUND_MIXER_LINE,
+ [AC97_REC_STEREO]= SOUND_MIXER_IGAIN,
+ [AC97_REC_PHONE] = SOUND_MIXER_PHONEIN
+};
+
+/* indexed by bit position */
+static const unsigned int ac97_oss_rm[] = {
+ [SOUND_MIXER_MIC] = AC97_REC_MIC,
+ [SOUND_MIXER_CD] = AC97_REC_CD,
+ [SOUND_MIXER_VIDEO] = AC97_REC_VIDEO,
+ [SOUND_MIXER_LINE1] = AC97_REC_AUX,
+ [SOUND_MIXER_LINE] = AC97_REC_LINE,
+ [SOUND_MIXER_IGAIN] = AC97_REC_STEREO,
+ [SOUND_MIXER_PHONEIN] = AC97_REC_PHONE
+};
+
+static LIST_HEAD(codecs);
+static LIST_HEAD(codec_drivers);
+static DEFINE_MUTEX(codec_mutex);
+
+/* reads the given OSS mixer from the ac97 the caller must have insured that the ac97 knows
+ about that given mixer, and should be holding a spinlock for the card */
+static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel)
+{
+ u16 val;
+ int ret = 0;
+ int scale;
+ struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];
+
+ val = codec->codec_read(codec , mh->offset);
+
+ if (val & AC97_MUTE) {
+ ret = 0;
+ } else if (AC97_STEREO_MASK & (1 << oss_channel)) {
+ /* nice stereo mixers .. */
+ int left,right;
+
+ left = (val >> 8) & 0x7f;
+ right = val & 0x7f;
+
+ if (oss_channel == SOUND_MIXER_IGAIN) {
+ right = (right * 100) / mh->scale;
+ left = (left * 100) / mh->scale;
+ } else {
+ /* these may have 5 or 6 bit resolution */
+ if(oss_channel == SOUND_MIXER_VOLUME || oss_channel == SOUND_MIXER_ALTPCM)
+ scale = (1 << codec->bit_resolution);
+ else
+ scale = mh->scale;
+
+ right = 100 - ((right * 100) / scale);
+ left = 100 - ((left * 100) / scale);
+ }
+ ret = left | (right << 8);
+ } else if (oss_channel == SOUND_MIXER_SPEAKER) {
+ ret = 100 - ((((val & 0x1e)>>1) * 100) / mh->scale);
+ } else if (oss_channel == SOUND_MIXER_PHONEIN) {
+ ret = 100 - (((val & 0x1f) * 100) / mh->scale);
+ } else if (oss_channel == SOUND_MIXER_PHONEOUT) {
+ scale = (1 << codec->bit_resolution);
+ ret = 100 - (((val & 0x1f) * 100) / scale);
+ } else if (oss_channel == SOUND_MIXER_MIC) {
+ ret = 100 - (((val & 0x1f) * 100) / mh->scale);
+ /* the low bit is optional in the tone sliders and masking
+ it lets us avoid the 0xf 'bypass'.. */
+ } else if (oss_channel == SOUND_MIXER_BASS) {
+ ret = 100 - ((((val >> 8) & 0xe) * 100) / mh->scale);
+ } else if (oss_channel == SOUND_MIXER_TREBLE) {
+ ret = 100 - (((val & 0xe) * 100) / mh->scale);
+ }
+
+#ifdef DEBUG
+ printk("ac97_codec: read OSS mixer %2d (%s ac97 register 0x%02x), "
+ "0x%04x -> 0x%04x\n",
+ oss_channel, codec->id ? "Secondary" : "Primary",
+ mh->offset, val, ret);
+#endif
+
+ return ret;
+}
+
+/* write the OSS encoded volume to the given OSS encoded mixer, again caller's job to
+ make sure all is well in arg land, call with spinlock held */
+static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel,
+ unsigned int left, unsigned int right)
+{
+ u16 val = 0;
+ int scale;
+ struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];
+
+#ifdef DEBUG
+ printk("ac97_codec: wrote OSS mixer %2d (%s ac97 register 0x%02x), "
+ "left vol:%2d, right vol:%2d:",
+ oss_channel, codec->id ? "Secondary" : "Primary",
+ mh->offset, left, right);
+#endif
+
+ if (AC97_STEREO_MASK & (1 << oss_channel)) {
+ /* stereo mixers */
+ if (left == 0 && right == 0) {
+ val = AC97_MUTE;
+ } else {
+ if (oss_channel == SOUND_MIXER_IGAIN) {
+ right = (right * mh->scale) / 100;
+ left = (left * mh->scale) / 100;
+ if (right >= mh->scale)
+ right = mh->scale-1;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ } else {
+ /* these may have 5 or 6 bit resolution */
+ if (oss_channel == SOUND_MIXER_VOLUME ||
+ oss_channel == SOUND_MIXER_ALTPCM)
+ scale = (1 << codec->bit_resolution);
+ else
+ scale = mh->scale;
+
+ right = ((100 - right) * scale) / 100;
+ left = ((100 - left) * scale) / 100;
+ if (right >= scale)
+ right = scale-1;
+ if (left >= scale)
+ left = scale-1;
+ }
+ val = (left << 8) | right;
+ }
+ } else if (oss_channel == SOUND_MIXER_BASS) {
+ val = codec->codec_read(codec , mh->offset) & ~0x0f00;
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val |= (left << 8) & 0x0e00;
+ } else if (oss_channel == SOUND_MIXER_TREBLE) {
+ val = codec->codec_read(codec , mh->offset) & ~0x000f;
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val |= left & 0x000e;
+ } else if(left == 0) {
+ val = AC97_MUTE;
+ } else if (oss_channel == SOUND_MIXER_SPEAKER) {
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val = left << 1;
+ } else if (oss_channel == SOUND_MIXER_PHONEIN) {
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val = left;
+ } else if (oss_channel == SOUND_MIXER_PHONEOUT) {
+ scale = (1 << codec->bit_resolution);
+ left = ((100 - left) * scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val = left;
+ } else if (oss_channel == SOUND_MIXER_MIC) {
+ val = codec->codec_read(codec , mh->offset) & ~0x801f;
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val |= left;
+ /* the low bit is optional in the tone sliders and masking
+ it lets us avoid the 0xf 'bypass'.. */
+ }
+#ifdef DEBUG
+ printk(" 0x%04x", val);
+#endif
+
+ codec->codec_write(codec, mh->offset, val);
+
+#ifdef DEBUG
+ val = codec->codec_read(codec, mh->offset);
+ printk(" -> 0x%04x\n", val);
+#endif
+}
+
+/* a thin wrapper for write_mixer */
+static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val )
+{
+ unsigned int left,right;
+
+ /* cleanse input a little */
+ right = ((val >> 8) & 0xff) ;
+ left = (val & 0xff) ;
+
+ if (right > 100) right = 100;
+ if (left > 100) left = 100;
+
+ codec->mixer_state[oss_mixer] = (right << 8) | left;
+ codec->write_mixer(codec, oss_mixer, left, right);
+}
+
+/* read or write the recmask, the ac97 can really have left and right recording
+ inputs independantly set, but OSS doesn't seem to want us to express that to
+ the user. the caller guarantees that we have a supported bit set, and they
+ must be holding the card's spinlock */
+static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask)
+{
+ unsigned int val;
+
+ if (rw) {
+ /* read it from the card */
+ val = codec->codec_read(codec, AC97_RECORD_SELECT);
+#ifdef DEBUG
+ printk("ac97_codec: ac97 recmask to set to 0x%04x\n", val);
+#endif
+ return (1 << ac97_rm2oss[val & 0x07]);
+ }
+
+ /* else, write the first set in the mask as the
+ output */
+ /* clear out current set value first (AC97 supports only 1 input!) */
+ val = (1 << ac97_rm2oss[codec->codec_read(codec, AC97_RECORD_SELECT) & 0x07]);
+ if (mask != val)
+ mask &= ~val;
+
+ val = ffs(mask);
+ val = ac97_oss_rm[val-1];
+ val |= val << 8; /* set both channels */
+
+#ifdef DEBUG
+ printk("ac97_codec: setting ac97 recmask to 0x%04x\n", val);
+#endif
+
+ codec->codec_write(codec, AC97_RECORD_SELECT, val);
+
+ return 0;
+};
+
+static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg)
+{
+ int i, val = 0;
+
+ if (cmd == SOUND_MIXER_INFO) {
+ mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, codec->name, sizeof(info.id));
+ strlcpy(info.name, codec->name, sizeof(info.name));
+ info.modify_counter = codec->modcnt;
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ }
+ if (cmd == SOUND_OLD_MIXER_INFO) {
+ _old_mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, codec->name, sizeof(info.id));
+ strlcpy(info.name, codec->name, sizeof(info.name));
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ }
+
+ if (_IOC_TYPE(cmd) != 'M' || _SIOC_SIZE(cmd) != sizeof(int))
+ return -EINVAL;
+
+ if (cmd == OSS_GETVERSION)
+ return put_user(SOUND_VERSION, (int __user *)arg);
+
+ if (_SIOC_DIR(cmd) == _SIOC_READ) {
+ switch (_IOC_NR(cmd)) {
+ case SOUND_MIXER_RECSRC: /* give them the current record source */
+ if (!codec->recmask_io) {
+ val = 0;
+ } else {
+ val = codec->recmask_io(codec, 1, 0);
+ }
+ break;
+
+ case SOUND_MIXER_DEVMASK: /* give them the supported mixers */
+ val = codec->supported_mixers;
+ break;
+
+ case SOUND_MIXER_RECMASK: /* Arg contains a bit for each supported recording source */
+ val = codec->record_sources;
+ break;
+
+ case SOUND_MIXER_STEREODEVS: /* Mixer channels supporting stereo */
+ val = codec->stereo_mixers;
+ break;
+
+ case SOUND_MIXER_CAPS:
+ val = SOUND_CAP_EXCL_INPUT;
+ break;
+
+ default: /* read a specific mixer */
+ i = _IOC_NR(cmd);
+
+ if (!supported_mixer(codec, i))
+ return -EINVAL;
+
+ /* do we ever want to touch the hardware? */
+ /* val = codec->read_mixer(codec, i); */
+ val = codec->mixer_state[i];
+ break;
+ }
+ return put_user(val, (int __user *)arg);
+ }
+
+ if (_SIOC_DIR(cmd) == (_SIOC_WRITE|_SIOC_READ)) {
+ codec->modcnt++;
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+
+ switch (_IOC_NR(cmd)) {
+ case SOUND_MIXER_RECSRC: /* Arg contains a bit for each recording source */
+ if (!codec->recmask_io) return -EINVAL;
+ if (!val) return 0;
+ if (!(val &= codec->record_sources)) return -EINVAL;
+
+ codec->recmask_io(codec, 0, val);
+
+ return 0;
+ default: /* write a specific mixer */
+ i = _IOC_NR(cmd);
+
+ if (!supported_mixer(codec, i))
+ return -EINVAL;
+
+ ac97_set_mixer(codec, i, val);
+
+ return 0;
+ }
+ }
+ return -EINVAL;
+}
+
+/**
+ * codec_id - Turn id1/id2 into a PnP string
+ * @id1: Vendor ID1
+ * @id2: Vendor ID2
+ * @buf: CODEC_ID_BUFSZ byte buffer
+ *
+ * Fills buf with a zero terminated PnP ident string for the id1/id2
+ * pair. For convenience the return is the passed in buffer pointer.
+ */
+
+static char *codec_id(u16 id1, u16 id2, char *buf)
+{
+ if(id1&0x8080) {
+ snprintf(buf, CODEC_ID_BUFSZ, "0x%04x:0x%04x", id1, id2);
+ } else {
+ buf[0] = (id1 >> 8);
+ buf[1] = (id1 & 0xFF);
+ buf[2] = (id2 >> 8);
+ snprintf(buf+3, CODEC_ID_BUFSZ - 3, "%d", id2&0xFF);
+ }
+ return buf;
+}
+
+/**
+ * ac97_check_modem - Check if the Codec is a modem
+ * @codec: codec to check
+ *
+ * Return true if the device is an AC97 1.0 or AC97 2.0 modem
+ */
+
+static int ac97_check_modem(struct ac97_codec *codec)
+{
+ /* Check for an AC97 1.0 soft modem (ID1) */
+ if(codec->codec_read(codec, AC97_RESET) & 2)
+ return 1;
+ /* Check for an AC97 2.x soft modem */
+ codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0L);
+ if(codec->codec_read(codec, AC97_EXTENDED_MODEM_ID) & 1)
+ return 1;
+ return 0;
+}
+
+
+/**
+ * ac97_alloc_codec - Allocate an AC97 codec
+ *
+ * Returns a new AC97 codec structure. AC97 codecs may become
+ * refcounted soon so this interface is needed. Returns with
+ * one reference taken.
+ */
+
+struct ac97_codec *ac97_alloc_codec(void)
+{
+ struct ac97_codec *codec = kzalloc(sizeof(struct ac97_codec), GFP_KERNEL);
+ if(!codec)
+ return NULL;
+
+ spin_lock_init(&codec->lock);
+ INIT_LIST_HEAD(&codec->list);
+ return codec;
+}
+
+EXPORT_SYMBOL(ac97_alloc_codec);
+
+/**
+ * ac97_release_codec - Release an AC97 codec
+ * @codec: codec to release
+ *
+ * Release an allocated AC97 codec. This will be refcounted in
+ * time but for the moment is trivial. Calls the unregister
+ * handler if the codec is now defunct.
+ */
+
+void ac97_release_codec(struct ac97_codec *codec)
+{
+ /* Remove from the list first, we don't want to be
+ "rediscovered" */
+ mutex_lock(&codec_mutex);
+ list_del(&codec->list);
+ mutex_unlock(&codec_mutex);
+ /*
+ * The driver needs to deal with internal
+ * locking to avoid accidents here.
+ */
+ if(codec->driver)
+ codec->driver->remove(codec, codec->driver);
+ kfree(codec);
+}
+
+EXPORT_SYMBOL(ac97_release_codec);
+
+/**
+ * ac97_probe_codec - Initialize and setup AC97-compatible codec
+ * @codec: (in/out) Kernel info for a single AC97 codec
+ *
+ * Reset the AC97 codec, then initialize the mixer and
+ * the rest of the @codec structure.
+ *
+ * The codec_read and codec_write fields of @codec are
+ * required to be setup and working when this function
+ * is called. All other fields are set by this function.
+ *
+ * codec_wait field of @codec can optionally be provided
+ * when calling this function. If codec_wait is not %NULL,
+ * this function will call codec_wait any time it is
+ * necessary to wait for the audio chip to reach the
+ * codec-ready state. If codec_wait is %NULL, then
+ * the default behavior is to call schedule_timeout.
+ * Currently codec_wait is used to wait for AC97 codec
+ * reset to complete.
+ *
+ * Some codecs will power down when a register reset is
+ * performed. We now check for such codecs.
+ *
+ * Returns 1 (true) on success, or 0 (false) on failure.
+ */
+
+int ac97_probe_codec(struct ac97_codec *codec)
+{
+ u16 id1, id2;
+ u16 audio;
+ int i;
+ char cidbuf[CODEC_ID_BUFSZ];
+ u16 f;
+ struct list_head *l;
+ struct ac97_driver *d;
+
+ /* wait for codec-ready state */
+ if (codec->codec_wait)
+ codec->codec_wait(codec);
+ else
+ udelay(10);
+
+ /* will the codec power down if register reset ? */
+ id1 = codec->codec_read(codec, AC97_VENDOR_ID1);
+ id2 = codec->codec_read(codec, AC97_VENDOR_ID2);
+ codec->name = NULL;
+ codec->codec_ops = &null_ops;
+ for (i = 0; i < ARRAY_SIZE(ac97_codec_ids); i++) {
+ if (ac97_codec_ids[i].id == ((id1 << 16) | id2)) {
+ codec->type = ac97_codec_ids[i].id;
+ codec->name = ac97_codec_ids[i].name;
+ codec->codec_ops = ac97_codec_ids[i].ops;
+ codec->flags = ac97_codec_ids[i].flags;
+ break;
+ }
+ }
+
+ codec->model = (id1 << 16) | id2;
+ if ((codec->flags & AC97_DEFAULT_POWER_OFF) == 0) {
+ /* reset codec and wait for the ready bit before we continue */
+ codec->codec_write(codec, AC97_RESET, 0L);
+ if (codec->codec_wait)
+ codec->codec_wait(codec);
+ else
+ udelay(10);
+ }
+
+ /* probing AC97 codec, AC97 2.0 says that bit 15 of register 0x00 (reset) should
+ * be read zero.
+ *
+ * FIXME: is the following comment outdated? -jgarzik
+ * Probing of AC97 in this way is not reliable, it is not even SAFE !!
+ */
+ if ((audio = codec->codec_read(codec, AC97_RESET)) & 0x8000) {
+ printk(KERN_ERR "ac97_codec: %s ac97 codec not present\n",
+ (codec->id & 0x2) ? (codec->id&1 ? "4th" : "Tertiary")
+ : (codec->id&1 ? "Secondary": "Primary"));
+ return 0;
+ }
+
+ /* probe for Modem Codec */
+ codec->modem = ac97_check_modem(codec);
+
+ /* enable SPDIF */
+ f = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ if((codec->codec_ops == &null_ops) && (f & 4))
+ codec->codec_ops = &default_digital_ops;
+
+ /* A device which thinks its a modem but isnt */
+ if(codec->flags & AC97_DELUDED_MODEM)
+ codec->modem = 0;
+
+ if (codec->name == NULL)
+ codec->name = "Unknown";
+ printk(KERN_INFO "ac97_codec: AC97 %s codec, id: %s (%s)\n",
+ codec->modem ? "Modem" : (audio ? "Audio" : ""),
+ codec_id(id1, id2, cidbuf), codec->name);
+
+ if(!ac97_init_mixer(codec))
+ return 0;
+
+ /*
+ * Attach last so the caller can override the mixer
+ * callbacks.
+ */
+
+ mutex_lock(&codec_mutex);
+ list_add(&codec->list, &codecs);
+
+ list_for_each(l, &codec_drivers) {
+ d = list_entry(l, struct ac97_driver, list);
+ if ((codec->model ^ d->codec_id) & d->codec_mask)
+ continue;
+ if(d->probe(codec, d) == 0)
+ {
+ codec->driver = d;
+ break;
+ }
+ }
+
+ mutex_unlock(&codec_mutex);
+ return 1;
+}
+
+static int ac97_init_mixer(struct ac97_codec *codec)
+{
+ u16 cap;
+ int i;
+
+ cap = codec->codec_read(codec, AC97_RESET);
+
+ /* mixer masks */
+ codec->supported_mixers = AC97_SUPPORTED_MASK;
+ codec->stereo_mixers = AC97_STEREO_MASK;
+ codec->record_sources = AC97_RECORD_MASK;
+ if (!(cap & 0x04))
+ codec->supported_mixers &= ~(SOUND_MASK_BASS|SOUND_MASK_TREBLE);
+ if (!(cap & 0x10))
+ codec->supported_mixers &= ~SOUND_MASK_ALTPCM;
+
+
+ /* detect bit resolution */
+ codec->codec_write(codec, AC97_MASTER_VOL_STEREO, 0x2020);
+ if(codec->codec_read(codec, AC97_MASTER_VOL_STEREO) == 0x2020)
+ codec->bit_resolution = 6;
+ else
+ codec->bit_resolution = 5;
+
+ /* generic OSS to AC97 wrapper */
+ codec->read_mixer = ac97_read_mixer;
+ codec->write_mixer = ac97_write_mixer;
+ codec->recmask_io = ac97_recmask_io;
+ codec->mixer_ioctl = ac97_mixer_ioctl;
+
+ /* initialize mixer channel volumes */
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
+ struct mixer_defaults *md = &mixer_defaults[i];
+ if (md->mixer == -1)
+ break;
+ if (!supported_mixer(codec, md->mixer))
+ continue;
+ ac97_set_mixer(codec, md->mixer, md->value);
+ }
+
+ /* codec specific initialization for 4-6 channel output or secondary codec stuff */
+ if (codec->codec_ops->init != NULL) {
+ codec->codec_ops->init(codec);
+ }
+
+ /*
+ * Volume is MUTE only on this device. We have to initialise
+ * it but its useless beyond that.
+ */
+ if(codec->flags & AC97_NO_PCM_VOLUME)
+ {
+ codec->supported_mixers &= ~SOUND_MASK_PCM;
+ printk(KERN_WARNING "AC97 codec does not have proper volume support.\n");
+ }
+ return 1;
+}
+
+#define AC97_SIGMATEL_ANALOG 0x6c /* Analog Special */
+#define AC97_SIGMATEL_DAC2INVERT 0x6e
+#define AC97_SIGMATEL_BIAS1 0x70
+#define AC97_SIGMATEL_BIAS2 0x72
+#define AC97_SIGMATEL_MULTICHN 0x74 /* Multi-Channel programming */
+#define AC97_SIGMATEL_CIC1 0x76
+#define AC97_SIGMATEL_CIC2 0x78
+
+
+static int sigmatel_9708_init(struct ac97_codec * codec)
+{
+ u16 codec72, codec6c;
+
+ codec72 = codec->codec_read(codec, AC97_SIGMATEL_BIAS2) & 0x8000;
+ codec6c = codec->codec_read(codec, AC97_SIGMATEL_ANALOG);
+
+ if ((codec72==0) && (codec6c==0)) {
+ codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
+ codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1000);
+ codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba);
+ codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0007);
+ } else if ((codec72==0x8000) && (codec6c==0)) {
+ codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
+ codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1001);
+ codec->codec_write(codec, AC97_SIGMATEL_DAC2INVERT, 0x0008);
+ } else if ((codec72==0x8000) && (codec6c==0x0080)) {
+ /* nothing */
+ }
+ codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000);
+ return 0;
+}
+
+
+static int sigmatel_9721_init(struct ac97_codec * codec)
+{
+ /* Only set up secondary codec */
+ if (codec->id == 0)
+ return 0;
+
+ codec->codec_write(codec, AC97_SURROUND_MASTER, 0L);
+
+ /* initialize SigmaTel STAC9721/23 as secondary codec, decoding AC link
+ sloc 3,4 = 0x01, slot 7,8 = 0x00, */
+ codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x00);
+
+ /* we don't have the crystal when we are on an AMR card, so use
+ BIT_CLK as our clock source. Write the magic word ABBA and read
+ back to enable register 0x78 */
+ codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
+ codec->codec_read(codec, AC97_SIGMATEL_CIC1);
+
+ /* sync all the clocks*/
+ codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x3802);
+
+ return 0;
+}
+
+
+static int sigmatel_9744_init(struct ac97_codec * codec)
+{
+ // patch for SigmaTel
+ codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba);
+ codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x0000); // is this correct? --jk
+ codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba);
+ codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0002);
+ codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000);
+ return 0;
+}
+
+static int cmedia_init(struct ac97_codec *codec)
+{
+ /* Initialise the CMedia 9739 */
+ /*
+ We could set various options here
+ Register 0x20 bit 0x100 sets mic as center bass
+ Also do multi_channel_ctrl &=~0x3000 |=0x1000
+
+ For now we set up the GPIO and PC beep
+ */
+
+ u16 v;
+
+ /* MIC */
+ codec->codec_write(codec, 0x64, 0x3000);
+ v = codec->codec_read(codec, 0x64);
+ v &= ~0x8000;
+ codec->codec_write(codec, 0x64, v);
+ codec->codec_write(codec, 0x70, 0x0100);
+ codec->codec_write(codec, 0x72, 0x0020);
+ return 0;
+}
+
+#define AC97_WM97XX_FMIXER_VOL 0x72
+#define AC97_WM97XX_RMIXER_VOL 0x74
+#define AC97_WM97XX_TEST 0x5a
+#define AC97_WM9704_RPCM_VOL 0x70
+#define AC97_WM9711_OUT3VOL 0x16
+
+static int wolfson_init03(struct ac97_codec * codec)
+{
+ /* this is known to work for the ViewSonic ViewPad 1000 */
+ codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
+ codec->codec_write(codec, AC97_GENERAL_PURPOSE, 0x8000);
+ return 0;
+}
+
+static int wolfson_init04(struct ac97_codec * codec)
+{
+ codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
+ codec->codec_write(codec, AC97_WM97XX_RMIXER_VOL, 0x0808);
+
+ // patch for DVD noise
+ codec->codec_write(codec, AC97_WM97XX_TEST, 0x0200);
+
+ // init vol as PCM vol
+ codec->codec_write(codec, AC97_WM9704_RPCM_VOL,
+ codec->codec_read(codec, AC97_PCMOUT_VOL));
+
+ /* set rear surround volume */
+ codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000);
+ return 0;
+}
+
+/* WM9705, WM9710 */
+static int wolfson_init05(struct ac97_codec * codec)
+{
+ /* set front mixer volume */
+ codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808);
+ return 0;
+}
+
+/* WM9711, WM9712 */
+static int wolfson_init11(struct ac97_codec * codec)
+{
+ /* stop pop's during suspend/resume */
+ codec->codec_write(codec, AC97_WM97XX_TEST,
+ codec->codec_read(codec, AC97_WM97XX_TEST) & 0xffbf);
+
+ /* set out3 volume */
+ codec->codec_write(codec, AC97_WM9711_OUT3VOL, 0x0808);
+ return 0;
+}
+
+/* WM9713 */
+static int wolfson_init13(struct ac97_codec * codec)
+{
+ codec->codec_write(codec, AC97_RECORD_GAIN, 0x00a0);
+ codec->codec_write(codec, AC97_POWER_CONTROL, 0x0000);
+ codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0xDA00);
+ codec->codec_write(codec, AC97_EXTEND_MODEM_STAT, 0x3810);
+ codec->codec_write(codec, AC97_PHONE_VOL, 0x0808);
+ codec->codec_write(codec, AC97_PCBEEP_VOL, 0x0808);
+
+ return 0;
+}
+
+static int tritech_init(struct ac97_codec * codec)
+{
+ codec->codec_write(codec, 0x26, 0x0300);
+ codec->codec_write(codec, 0x26, 0x0000);
+ codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000);
+ codec->codec_write(codec, AC97_RESERVED_3A, 0x0000);
+ return 0;
+}
+
+
+/* copied from drivers/sound/maestro.c */
+static int tritech_maestro_init(struct ac97_codec * codec)
+{
+ /* no idea what this does */
+ codec->codec_write(codec, 0x2A, 0x0001);
+ codec->codec_write(codec, 0x2C, 0x0000);
+ codec->codec_write(codec, 0x2C, 0XFFFF);
+ return 0;
+}
+
+
+
+/*
+ * Presario700 workaround
+ * for Jack Sense/SPDIF Register mis-setting causing
+ * no audible output
+ * by Santiago Nullo 04/05/2002
+ */
+
+#define AC97_AD1886_JACK_SENSE 0x72
+
+static int ad1886_init(struct ac97_codec * codec)
+{
+ /* from AD1886 Specs */
+ codec->codec_write(codec, AC97_AD1886_JACK_SENSE, 0x0010);
+ return 0;
+}
+
+
+
+
+/*
+ * This is basically standard AC97. It should work as a default for
+ * almost all modern codecs. Note that some cards wire EAPD *backwards*
+ * That side of it is up to the card driver not us to cope with.
+ *
+ */
+
+static int eapd_control(struct ac97_codec * codec, int on)
+{
+ if(on)
+ codec->codec_write(codec, AC97_POWER_CONTROL,
+ codec->codec_read(codec, AC97_POWER_CONTROL)|0x8000);
+ else
+ codec->codec_write(codec, AC97_POWER_CONTROL,
+ codec->codec_read(codec, AC97_POWER_CONTROL)&~0x8000);
+ return 0;
+}
+
+static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
+{
+ u16 reg;
+
+ reg = codec->codec_read(codec, AC97_SPDIF_CONTROL);
+
+ switch(rate)
+ {
+ /* Off by default */
+ default:
+ case 0:
+ reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ codec->codec_write(codec, AC97_EXTENDED_STATUS, (reg & ~AC97_EA_SPDIF));
+ if(rate == 0)
+ return 0;
+ return -EINVAL;
+ case 1:
+ reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_48K;
+ break;
+ case 2:
+ reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_44K;
+ break;
+ case 3:
+ reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_32K;
+ break;
+ }
+
+ reg &= ~AC97_SC_CC_MASK;
+ reg |= (mode & AUDIO_CCMASK) << 6;
+
+ if(mode & AUDIO_DIGITAL)
+ reg |= 2;
+ if(mode & AUDIO_PRO)
+ reg |= 1;
+ if(mode & AUDIO_DRS)
+ reg |= 0x4000;
+
+ codec->codec_write(codec, AC97_SPDIF_CONTROL, reg);
+
+ reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ reg &= (AC97_EA_SLOT_MASK);
+ reg |= AC97_EA_VRA | AC97_EA_SPDIF | slots;
+ codec->codec_write(codec, AC97_EXTENDED_STATUS, reg);
+
+ reg = codec->codec_read(codec, AC97_EXTENDED_STATUS);
+ if(!(reg & 0x0400))
+ {
+ codec->codec_write(codec, AC97_EXTENDED_STATUS, reg & ~ AC97_EA_SPDIF);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * Crystal digital audio control (CS4299)
+ */
+
+static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
+{
+ u16 cv;
+
+ if(mode & AUDIO_DIGITAL)
+ return -EINVAL;
+
+ switch(rate)
+ {
+ case 0: cv = 0x0; break; /* SPEN off */
+ case 48000: cv = 0x8004; break; /* 48KHz digital */
+ case 44100: cv = 0x8104; break; /* 44.1KHz digital */
+ case 32768: /* 32Khz */
+ default:
+ return -EINVAL;
+ }
+ codec->codec_write(codec, 0x68, cv);
+ return 0;
+}
+
+/*
+ * CMedia digital audio control
+ * Needs more work.
+ */
+
+static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode)
+{
+ u16 cv;
+
+ if(mode & AUDIO_DIGITAL)
+ return -EINVAL;
+
+ switch(rate)
+ {
+ case 0: cv = 0x0001; break; /* SPEN off */
+ case 48000: cv = 0x0009; break; /* 48KHz digital */
+ default:
+ return -EINVAL;
+ }
+ codec->codec_write(codec, 0x2A, 0x05c4);
+ codec->codec_write(codec, 0x6C, cv);
+
+ /* Switch on mix to surround */
+ cv = codec->codec_read(codec, 0x64);
+ cv &= ~0x0200;
+ if(mode)
+ cv |= 0x0200;
+ codec->codec_write(codec, 0x64, cv);
+ return 0;
+}
+
+
+/* copied from drivers/sound/maestro.c */
+#if 0 /* there has been 1 person on the planet with a pt101 that we
+ know of. If they care, they can put this back in :) */
+static int pt101_init(struct ac97_codec * codec)
+{
+ printk(KERN_INFO "ac97_codec: PT101 Codec detected, initializing but _not_ installing mixer device.\n");
+ /* who knows.. */
+ codec->codec_write(codec, 0x2A, 0x0001);
+ codec->codec_write(codec, 0x2C, 0x0000);
+ codec->codec_write(codec, 0x2C, 0xFFFF);
+ codec->codec_write(codec, 0x10, 0x9F1F);
+ codec->codec_write(codec, 0x12, 0x0808);
+ codec->codec_write(codec, 0x14, 0x9F1F);
+ codec->codec_write(codec, 0x16, 0x9F1F);
+ codec->codec_write(codec, 0x18, 0x0404);
+ codec->codec_write(codec, 0x1A, 0x0000);
+ codec->codec_write(codec, 0x1C, 0x0000);
+ codec->codec_write(codec, 0x02, 0x0404);
+ codec->codec_write(codec, 0x04, 0x0808);
+ codec->codec_write(codec, 0x0C, 0x801F);
+ codec->codec_write(codec, 0x0E, 0x801F);
+ return 0;
+}
+#endif
+
+
+EXPORT_SYMBOL(ac97_probe_codec);
+
+MODULE_LICENSE("GPL");
+
diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c
new file mode 100644
index 0000000..7cf9913
--- /dev/null
+++ b/sound/oss/ad1848.c
@@ -0,0 +1,3068 @@
+/*
+ * sound/oss/ad1848.c
+ *
+ * The low level driver for the AD1848/CS4248 codec chip which
+ * is used for example in the MS Sound System.
+ *
+ * The CS4231 which is used in the GUS MAX and some other cards is
+ * upwards compatible with AD1848 and this driver is able to drive it.
+ *
+ * CS4231A and AD1845 are upward compatible with CS4231. However
+ * the new features of these chips are different.
+ *
+ * CS4232 is a PnP audio chip which contains a CS4231A (and SB, MPU).
+ * CS4232A is an improved version of CS4232.
+ *
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * general sleep/wakeup clean up.
+ * Alan Cox : reformatted. Fixed SMP bugs. Moved to kernel alloc/free
+ * of irqs. Use dev_id.
+ * Christoph Hellwig : adapted to module_init/module_exit
+ * Aki Laukkanen : added power management support
+ * Arnaldo C. de Melo : added missing restore_flags in ad1848_resume
+ * Miguel Freitas : added ISA PnP support
+ * Alan Cox : Added CS4236->4239 identification
+ * Daniel T. Cobra : Alernate config/mixer for later chips
+ * Alan Cox : Merged chip idents and config code
+ *
+ * TODO
+ * APM save restore assist code on IBM thinkpad
+ *
+ * Status:
+ * Tested. Believed fully functional.
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/stddef.h>
+#include <linux/isapnp.h>
+#include <linux/pnp.h>
+#include <linux/spinlock.h>
+
+#define DEB(x)
+#define DEB1(x)
+#include "sound_config.h"
+
+#include "ad1848.h"
+#include "ad1848_mixer.h"
+
+typedef struct
+{
+ spinlock_t lock;
+ int base;
+ int irq;
+ int dma1, dma2;
+ int dual_dma; /* 1, when two DMA channels allocated */
+ int subtype;
+ unsigned char MCE_bit;
+ unsigned char saved_regs[64]; /* Includes extended register space */
+ int debug_flag;
+
+ int audio_flags;
+ int record_dev, playback_dev;
+
+ int xfer_count;
+ int audio_mode;
+ int open_mode;
+ int intr_active;
+ char *chip_name, *name;
+ int model;
+#define MD_1848 1
+#define MD_4231 2
+#define MD_4231A 3
+#define MD_1845 4
+#define MD_4232 5
+#define MD_C930 6
+#define MD_IWAVE 7
+#define MD_4235 8 /* Crystal Audio CS4235 */
+#define MD_1845_SSCAPE 9 /* Ensoniq Soundscape PNP*/
+#define MD_4236 10 /* 4236 and higher */
+#define MD_42xB 11 /* CS 42xB */
+#define MD_4239 12 /* CS4239 */
+
+ /* Mixer parameters */
+ int recmask;
+ int supported_devices, orig_devices;
+ int supported_rec_devices, orig_rec_devices;
+ int *levels;
+ short mixer_reroute[32];
+ int dev_no;
+ volatile unsigned long timer_ticks;
+ int timer_running;
+ int irq_ok;
+ mixer_ents *mix_devices;
+ int mixer_output_port;
+} ad1848_info;
+
+typedef struct ad1848_port_info
+{
+ int open_mode;
+ int speed;
+ unsigned char speed_bits;
+ int channels;
+ int audio_format;
+ unsigned char format_bits;
+}
+ad1848_port_info;
+
+static struct address_info cfg;
+static int nr_ad1848_devs;
+
+static int deskpro_xl;
+static int deskpro_m;
+static int soundpro;
+
+static volatile signed char irq2dev[17] = {
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+#ifndef EXCLUDE_TIMERS
+static int timer_installed = -1;
+#endif
+
+static int loaded;
+
+static int ad_format_mask[13 /*devc->model */ ] =
+{
+ 0,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW, /* AD1845 */
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM,
+ AFMT_U8 | AFMT_S16_LE /* CS4235 */,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW /* Ensoniq Soundscape*/,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM,
+ AFMT_U8 | AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW | AFMT_S16_BE | AFMT_IMA_ADPCM
+};
+
+static ad1848_info adev_info[MAX_AUDIO_DEV];
+
+#define io_Index_Addr(d) ((d)->base)
+#define io_Indexed_Data(d) ((d)->base+1)
+#define io_Status(d) ((d)->base+2)
+#define io_Polled_IO(d) ((d)->base+3)
+
+static struct {
+ unsigned char flags;
+#define CAP_F_TIMER 0x01
+} capabilities [10 /*devc->model */ ] = {
+ {0}
+ ,{0} /* MD_1848 */
+ ,{CAP_F_TIMER} /* MD_4231 */
+ ,{CAP_F_TIMER} /* MD_4231A */
+ ,{CAP_F_TIMER} /* MD_1845 */
+ ,{CAP_F_TIMER} /* MD_4232 */
+ ,{0} /* MD_C930 */
+ ,{CAP_F_TIMER} /* MD_IWAVE */
+ ,{0} /* MD_4235 */
+ ,{CAP_F_TIMER} /* MD_1845_SSCAPE */
+};
+
+#ifdef CONFIG_PNP
+static int isapnp = 1;
+static int isapnpjump;
+static int reverse;
+
+static int audio_activated;
+#else
+static int isapnp;
+#endif
+
+
+
+static int ad1848_open(int dev, int mode);
+static void ad1848_close(int dev);
+static void ad1848_output_block(int dev, unsigned long buf, int count, int intrflag);
+static void ad1848_start_input(int dev, unsigned long buf, int count, int intrflag);
+static int ad1848_prepare_for_output(int dev, int bsize, int bcount);
+static int ad1848_prepare_for_input(int dev, int bsize, int bcount);
+static void ad1848_halt(int dev);
+static void ad1848_halt_input(int dev);
+static void ad1848_halt_output(int dev);
+static void ad1848_trigger(int dev, int bits);
+static irqreturn_t adintr(int irq, void *dev_id);
+
+#ifndef EXCLUDE_TIMERS
+static int ad1848_tmr_install(int dev);
+static void ad1848_tmr_reprogram(int dev);
+#endif
+
+static int ad_read(ad1848_info * devc, int reg)
+{
+ int x;
+ int timeout = 900000;
+
+ while (timeout > 0 && inb(devc->base) == 0x80) /*Are we initializing */
+ timeout--;
+
+ if(reg < 32)
+ {
+ outb(((unsigned char) (reg & 0xff) | devc->MCE_bit), io_Index_Addr(devc));
+ x = inb(io_Indexed_Data(devc));
+ }
+ else
+ {
+ int xreg, xra;
+
+ xreg = (reg & 0xff) - 32;
+ xra = (((xreg & 0x0f) << 4) & 0xf0) | 0x08 | ((xreg & 0x10) >> 2);
+ outb(((unsigned char) (23 & 0xff) | devc->MCE_bit), io_Index_Addr(devc));
+ outb(((unsigned char) (xra & 0xff)), io_Indexed_Data(devc));
+ x = inb(io_Indexed_Data(devc));
+ }
+
+ return x;
+}
+
+static void ad_write(ad1848_info * devc, int reg, int data)
+{
+ int timeout = 900000;
+
+ while (timeout > 0 && inb(devc->base) == 0x80) /* Are we initializing */
+ timeout--;
+
+ if(reg < 32)
+ {
+ outb(((unsigned char) (reg & 0xff) | devc->MCE_bit), io_Index_Addr(devc));
+ outb(((unsigned char) (data & 0xff)), io_Indexed_Data(devc));
+ }
+ else
+ {
+ int xreg, xra;
+
+ xreg = (reg & 0xff) - 32;
+ xra = (((xreg & 0x0f) << 4) & 0xf0) | 0x08 | ((xreg & 0x10) >> 2);
+ outb(((unsigned char) (23 & 0xff) | devc->MCE_bit), io_Index_Addr(devc));
+ outb(((unsigned char) (xra & 0xff)), io_Indexed_Data(devc));
+ outb((unsigned char) (data & 0xff), io_Indexed_Data(devc));
+ }
+}
+
+static void wait_for_calibration(ad1848_info * devc)
+{
+ int timeout = 0;
+
+ /*
+ * Wait until the auto calibration process has finished.
+ *
+ * 1) Wait until the chip becomes ready (reads don't return 0x80).
+ * 2) Wait until the ACI bit of I11 gets on and then off.
+ */
+
+ timeout = 100000;
+ while (timeout > 0 && inb(devc->base) == 0x80)
+ timeout--;
+ if (inb(devc->base) & 0x80)
+ printk(KERN_WARNING "ad1848: Auto calibration timed out(1).\n");
+
+ timeout = 100;
+ while (timeout > 0 && !(ad_read(devc, 11) & 0x20))
+ timeout--;
+ if (!(ad_read(devc, 11) & 0x20))
+ return;
+
+ timeout = 80000;
+ while (timeout > 0 && (ad_read(devc, 11) & 0x20))
+ timeout--;
+ if (ad_read(devc, 11) & 0x20)
+ if ( (devc->model != MD_1845) || (devc->model != MD_1845_SSCAPE))
+ printk(KERN_WARNING "ad1848: Auto calibration timed out(3).\n");
+}
+
+static void ad_mute(ad1848_info * devc)
+{
+ int i;
+ unsigned char prev;
+
+ /*
+ * Save old register settings and mute output channels
+ */
+
+ for (i = 6; i < 8; i++)
+ {
+ prev = devc->saved_regs[i] = ad_read(devc, i);
+ }
+
+}
+
+static void ad_unmute(ad1848_info * devc)
+{
+}
+
+static void ad_enter_MCE(ad1848_info * devc)
+{
+ int timeout = 1000;
+ unsigned short prev;
+
+ while (timeout > 0 && inb(devc->base) == 0x80) /*Are we initializing */
+ timeout--;
+
+ devc->MCE_bit = 0x40;
+ prev = inb(io_Index_Addr(devc));
+ if (prev & 0x40)
+ {
+ return;
+ }
+ outb((devc->MCE_bit), io_Index_Addr(devc));
+}
+
+static void ad_leave_MCE(ad1848_info * devc)
+{
+ unsigned char prev, acal;
+ int timeout = 1000;
+
+ while (timeout > 0 && inb(devc->base) == 0x80) /*Are we initializing */
+ timeout--;
+
+ acal = ad_read(devc, 9);
+
+ devc->MCE_bit = 0x00;
+ prev = inb(io_Index_Addr(devc));
+ outb((0x00), io_Index_Addr(devc)); /* Clear the MCE bit */
+
+ if ((prev & 0x40) == 0) /* Not in MCE mode */
+ {
+ return;
+ }
+ outb((0x00), io_Index_Addr(devc)); /* Clear the MCE bit */
+ if (acal & 0x08) /* Auto calibration is enabled */
+ wait_for_calibration(devc);
+}
+
+static int ad1848_set_recmask(ad1848_info * devc, int mask)
+{
+ unsigned char recdev;
+ int i, n;
+ unsigned long flags;
+
+ mask &= devc->supported_rec_devices;
+
+ /* Rename the mixer bits if necessary */
+ for (i = 0; i < 32; i++)
+ {
+ if (devc->mixer_reroute[i] != i)
+ {
+ if (mask & (1 << i))
+ {
+ mask &= ~(1 << i);
+ mask |= (1 << devc->mixer_reroute[i]);
+ }
+ }
+ }
+
+ n = 0;
+ for (i = 0; i < 32; i++) /* Count selected device bits */
+ if (mask & (1 << i))
+ n++;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ if (!soundpro) {
+ if (n == 0)
+ mask = SOUND_MASK_MIC;
+ else if (n != 1) { /* Too many devices selected */
+ mask &= ~devc->recmask; /* Filter out active settings */
+
+ n = 0;
+ for (i = 0; i < 32; i++) /* Count selected device bits */
+ if (mask & (1 << i))
+ n++;
+
+ if (n != 1)
+ mask = SOUND_MASK_MIC;
+ }
+ switch (mask) {
+ case SOUND_MASK_MIC:
+ recdev = 2;
+ break;
+
+ case SOUND_MASK_LINE:
+ case SOUND_MASK_LINE3:
+ recdev = 0;
+ break;
+
+ case SOUND_MASK_CD:
+ case SOUND_MASK_LINE1:
+ recdev = 1;
+ break;
+
+ case SOUND_MASK_IMIX:
+ recdev = 3;
+ break;
+
+ default:
+ mask = SOUND_MASK_MIC;
+ recdev = 2;
+ }
+
+ recdev <<= 6;
+ ad_write(devc, 0, (ad_read(devc, 0) & 0x3f) | recdev);
+ ad_write(devc, 1, (ad_read(devc, 1) & 0x3f) | recdev);
+ } else { /* soundpro */
+ unsigned char val;
+ int set_rec_bit;
+ int j;
+
+ for (i = 0; i < 32; i++) { /* For each bit */
+ if ((devc->supported_rec_devices & (1 << i)) == 0)
+ continue; /* Device not supported */
+
+ for (j = LEFT_CHN; j <= RIGHT_CHN; j++) {
+ if (devc->mix_devices[i][j].nbits == 0) /* Inexistent channel */
+ continue;
+
+ /*
+ * This is tricky:
+ * set_rec_bit becomes 1 if the corresponding bit in mask is set
+ * then it gets flipped if the polarity is inverse
+ */
+ set_rec_bit = ((mask & (1 << i)) != 0) ^ devc->mix_devices[i][j].recpol;
+
+ val = ad_read(devc, devc->mix_devices[i][j].recreg);
+ val &= ~(1 << devc->mix_devices[i][j].recpos);
+ val |= (set_rec_bit << devc->mix_devices[i][j].recpos);
+ ad_write(devc, devc->mix_devices[i][j].recreg, val);
+ }
+ }
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ /* Rename the mixer bits back if necessary */
+ for (i = 0; i < 32; i++)
+ {
+ if (devc->mixer_reroute[i] != i)
+ {
+ if (mask & (1 << devc->mixer_reroute[i]))
+ {
+ mask &= ~(1 << devc->mixer_reroute[i]);
+ mask |= (1 << i);
+ }
+ }
+ }
+ devc->recmask = mask;
+ return mask;
+}
+
+static void change_bits(ad1848_info * devc, unsigned char *regval,
+ unsigned char *muteval, int dev, int chn, int newval)
+{
+ unsigned char mask;
+ int shift;
+ int mute;
+ int mutemask;
+ int set_mute_bit;
+
+ set_mute_bit = (newval == 0) ^ devc->mix_devices[dev][chn].mutepol;
+
+ if (devc->mix_devices[dev][chn].polarity == 1) /* Reverse */
+ newval = 100 - newval;
+
+ mask = (1 << devc->mix_devices[dev][chn].nbits) - 1;
+ shift = devc->mix_devices[dev][chn].bitpos;
+
+ if (devc->mix_devices[dev][chn].mutepos == 8)
+ { /* if there is no mute bit */
+ mute = 0; /* No mute bit; do nothing special */
+ mutemask = ~0; /* No mute bit; do nothing special */
+ }
+ else
+ {
+ mute = (set_mute_bit << devc->mix_devices[dev][chn].mutepos);
+ mutemask = ~(1 << devc->mix_devices[dev][chn].mutepos);
+ }
+
+ newval = (int) ((newval * mask) + 50) / 100; /* Scale it */
+ *regval &= ~(mask << shift); /* Clear bits */
+ *regval |= (newval & mask) << shift; /* Set new value */
+
+ *muteval &= mutemask;
+ *muteval |= mute;
+}
+
+static int ad1848_mixer_get(ad1848_info * devc, int dev)
+{
+ if (!((1 << dev) & devc->supported_devices))
+ return -EINVAL;
+
+ dev = devc->mixer_reroute[dev];
+
+ return devc->levels[dev];
+}
+
+static void ad1848_mixer_set_channel(ad1848_info *devc, int dev, int value, int channel)
+{
+ int regoffs, muteregoffs;
+ unsigned char val, muteval;
+ unsigned long flags;
+
+ regoffs = devc->mix_devices[dev][channel].regno;
+ muteregoffs = devc->mix_devices[dev][channel].mutereg;
+ val = ad_read(devc, regoffs);
+
+ if (muteregoffs != regoffs) {
+ muteval = ad_read(devc, muteregoffs);
+ change_bits(devc, &val, &muteval, dev, channel, value);
+ }
+ else
+ change_bits(devc, &val, &val, dev, channel, value);
+
+ spin_lock_irqsave(&devc->lock,flags);
+ ad_write(devc, regoffs, val);
+ devc->saved_regs[regoffs] = val;
+ if (muteregoffs != regoffs) {
+ ad_write(devc, muteregoffs, muteval);
+ devc->saved_regs[muteregoffs] = muteval;
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static int ad1848_mixer_set(ad1848_info * devc, int dev, int value)
+{
+ int left = value & 0x000000ff;
+ int right = (value & 0x0000ff00) >> 8;
+ int retvol;
+
+ if (dev > 31)
+ return -EINVAL;
+
+ if (!(devc->supported_devices & (1 << dev)))
+ return -EINVAL;
+
+ dev = devc->mixer_reroute[dev];
+
+ if (devc->mix_devices[dev][LEFT_CHN].nbits == 0)
+ return -EINVAL;
+
+ if (left > 100)
+ left = 100;
+ if (right > 100)
+ right = 100;
+
+ if (devc->mix_devices[dev][RIGHT_CHN].nbits == 0) /* Mono control */
+ right = left;
+
+ retvol = left | (right << 8);
+
+ /* Scale volumes */
+ left = mix_cvt[left];
+ right = mix_cvt[right];
+
+ devc->levels[dev] = retvol;
+
+ /*
+ * Set the left channel
+ */
+ ad1848_mixer_set_channel(devc, dev, left, LEFT_CHN);
+
+ /*
+ * Set the right channel
+ */
+ if (devc->mix_devices[dev][RIGHT_CHN].nbits == 0)
+ goto out;
+ ad1848_mixer_set_channel(devc, dev, right, RIGHT_CHN);
+
+ out:
+ return retvol;
+}
+
+static void ad1848_mixer_reset(ad1848_info * devc)
+{
+ int i;
+ char name[32];
+ unsigned long flags;
+
+ devc->mix_devices = &(ad1848_mix_devices[0]);
+
+ sprintf(name, "%s_%d", devc->chip_name, nr_ad1848_devs);
+
+ for (i = 0; i < 32; i++)
+ devc->mixer_reroute[i] = i;
+
+ devc->supported_rec_devices = MODE1_REC_DEVICES;
+
+ switch (devc->model)
+ {
+ case MD_4231:
+ case MD_4231A:
+ case MD_1845:
+ case MD_1845_SSCAPE:
+ devc->supported_devices = MODE2_MIXER_DEVICES;
+ break;
+
+ case MD_C930:
+ devc->supported_devices = C930_MIXER_DEVICES;
+ devc->mix_devices = &(c930_mix_devices[0]);
+ break;
+
+ case MD_IWAVE:
+ devc->supported_devices = MODE3_MIXER_DEVICES;
+ devc->mix_devices = &(iwave_mix_devices[0]);
+ break;
+
+ case MD_42xB:
+ case MD_4239:
+ devc->mix_devices = &(cs42xb_mix_devices[0]);
+ devc->supported_devices = MODE3_MIXER_DEVICES;
+ break;
+ case MD_4232:
+ case MD_4235:
+ case MD_4236:
+ devc->supported_devices = MODE3_MIXER_DEVICES;
+ break;
+
+ case MD_1848:
+ if (soundpro) {
+ devc->supported_devices = SPRO_MIXER_DEVICES;
+ devc->supported_rec_devices = SPRO_REC_DEVICES;
+ devc->mix_devices = &(spro_mix_devices[0]);
+ break;
+ }
+
+ default:
+ devc->supported_devices = MODE1_MIXER_DEVICES;
+ }
+
+ devc->orig_devices = devc->supported_devices;
+ devc->orig_rec_devices = devc->supported_rec_devices;
+
+ devc->levels = load_mixer_volumes(name, default_mixer_levels, 1);
+
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++)
+ {
+ if (devc->supported_devices & (1 << i))
+ ad1848_mixer_set(devc, i, devc->levels[i]);
+ }
+
+ ad1848_set_recmask(devc, SOUND_MASK_MIC);
+
+ devc->mixer_output_port = devc->levels[31] | AUDIO_HEADPHONE | AUDIO_LINE_OUT;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ if (!soundpro) {
+ if (devc->mixer_output_port & AUDIO_SPEAKER)
+ ad_write(devc, 26, ad_read(devc, 26) & ~0x40); /* Unmute mono out */
+ else
+ ad_write(devc, 26, ad_read(devc, 26) | 0x40); /* Mute mono out */
+ } else {
+ /*
+ * From the "wouldn't it be nice if the mixer API had (better)
+ * support for custom stuff" category
+ */
+ /* Enable surround mode and SB16 mixer */
+ ad_write(devc, 16, 0x60);
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static int ad1848_mixer_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ ad1848_info *devc = mixer_devs[dev]->devc;
+ int val;
+
+ if (cmd == SOUND_MIXER_PRIVATE1)
+ {
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+
+ if (val != 0xffff)
+ {
+ unsigned long flags;
+ val &= (AUDIO_SPEAKER | AUDIO_HEADPHONE | AUDIO_LINE_OUT);
+ devc->mixer_output_port = val;
+ val |= AUDIO_HEADPHONE | AUDIO_LINE_OUT; /* Always on */
+ devc->mixer_output_port = val;
+ spin_lock_irqsave(&devc->lock,flags);
+ if (val & AUDIO_SPEAKER)
+ ad_write(devc, 26, ad_read(devc, 26) & ~0x40); /* Unmute mono out */
+ else
+ ad_write(devc, 26, ad_read(devc, 26) | 0x40); /* Mute mono out */
+ spin_unlock_irqrestore(&devc->lock,flags);
+ }
+ val = devc->mixer_output_port;
+ return put_user(val, (int __user *)arg);
+ }
+ if (cmd == SOUND_MIXER_PRIVATE2)
+ {
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ return(ad1848_control(AD1848_MIXER_REROUTE, val));
+ }
+ if (((cmd >> 8) & 0xff) == 'M')
+ {
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE)
+ {
+ switch (cmd & 0xff)
+ {
+ case SOUND_MIXER_RECSRC:
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ val = ad1848_set_recmask(devc, val);
+ break;
+
+ default:
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ val = ad1848_mixer_set(devc, cmd & 0xff, val);
+ break;
+ }
+ return put_user(val, (int __user *)arg);
+ }
+ else
+ {
+ switch (cmd & 0xff)
+ {
+ /*
+ * Return parameters
+ */
+
+ case SOUND_MIXER_RECSRC:
+ val = devc->recmask;
+ break;
+
+ case SOUND_MIXER_DEVMASK:
+ val = devc->supported_devices;
+ break;
+
+ case SOUND_MIXER_STEREODEVS:
+ val = devc->supported_devices;
+ if (devc->model != MD_C930)
+ val &= ~(SOUND_MASK_SPEAKER | SOUND_MASK_IMIX);
+ break;
+
+ case SOUND_MIXER_RECMASK:
+ val = devc->supported_rec_devices;
+ break;
+
+ case SOUND_MIXER_CAPS:
+ val=SOUND_CAP_EXCL_INPUT;
+ break;
+
+ default:
+ val = ad1848_mixer_get(devc, cmd & 0xff);
+ break;
+ }
+ return put_user(val, (int __user *)arg);
+ }
+ }
+ else
+ return -EINVAL;
+}
+
+static int ad1848_set_speed(int dev, int arg)
+{
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ /*
+ * The sampling speed is encoded in the least significant nibble of I8. The
+ * LSB selects the clock source (0=24.576 MHz, 1=16.9344 MHz) and other
+ * three bits select the divisor (indirectly):
+ *
+ * The available speeds are in the following table. Keep the speeds in
+ * the increasing order.
+ */
+ typedef struct
+ {
+ int speed;
+ unsigned char bits;
+ }
+ speed_struct;
+
+ static speed_struct speed_table[] =
+ {
+ {5510, (0 << 1) | 1},
+ {5510, (0 << 1) | 1},
+ {6620, (7 << 1) | 1},
+ {8000, (0 << 1) | 0},
+ {9600, (7 << 1) | 0},
+ {11025, (1 << 1) | 1},
+ {16000, (1 << 1) | 0},
+ {18900, (2 << 1) | 1},
+ {22050, (3 << 1) | 1},
+ {27420, (2 << 1) | 0},
+ {32000, (3 << 1) | 0},
+ {33075, (6 << 1) | 1},
+ {37800, (4 << 1) | 1},
+ {44100, (5 << 1) | 1},
+ {48000, (6 << 1) | 0}
+ };
+
+ int i, n, selected = -1;
+
+ n = sizeof(speed_table) / sizeof(speed_struct);
+
+ if (arg <= 0)
+ return portc->speed;
+
+ if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE) /* AD1845 has different timer than others */
+ {
+ if (arg < 4000)
+ arg = 4000;
+ if (arg > 50000)
+ arg = 50000;
+
+ portc->speed = arg;
+ portc->speed_bits = speed_table[3].bits;
+ return portc->speed;
+ }
+ if (arg < speed_table[0].speed)
+ selected = 0;
+ if (arg > speed_table[n - 1].speed)
+ selected = n - 1;
+
+ for (i = 1 /*really */ ; selected == -1 && i < n; i++)
+ {
+ if (speed_table[i].speed == arg)
+ selected = i;
+ else if (speed_table[i].speed > arg)
+ {
+ int diff1, diff2;
+
+ diff1 = arg - speed_table[i - 1].speed;
+ diff2 = speed_table[i].speed - arg;
+
+ if (diff1 < diff2)
+ selected = i - 1;
+ else
+ selected = i;
+ }
+ }
+ if (selected == -1)
+ {
+ printk(KERN_WARNING "ad1848: Can't find speed???\n");
+ selected = 3;
+ }
+ portc->speed = speed_table[selected].speed;
+ portc->speed_bits = speed_table[selected].bits;
+ return portc->speed;
+}
+
+static short ad1848_set_channels(int dev, short arg)
+{
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ if (arg != 1 && arg != 2)
+ return portc->channels;
+
+ portc->channels = arg;
+ return arg;
+}
+
+static unsigned int ad1848_set_bits(int dev, unsigned int arg)
+{
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ static struct format_tbl
+ {
+ int format;
+ unsigned char bits;
+ }
+ format2bits[] =
+ {
+ {
+ 0, 0
+ }
+ ,
+ {
+ AFMT_MU_LAW, 1
+ }
+ ,
+ {
+ AFMT_A_LAW, 3
+ }
+ ,
+ {
+ AFMT_IMA_ADPCM, 5
+ }
+ ,
+ {
+ AFMT_U8, 0
+ }
+ ,
+ {
+ AFMT_S16_LE, 2
+ }
+ ,
+ {
+ AFMT_S16_BE, 6
+ }
+ ,
+ {
+ AFMT_S8, 0
+ }
+ ,
+ {
+ AFMT_U16_LE, 0
+ }
+ ,
+ {
+ AFMT_U16_BE, 0
+ }
+ };
+ int i, n = sizeof(format2bits) / sizeof(struct format_tbl);
+
+ if (arg == 0)
+ return portc->audio_format;
+
+ if (!(arg & ad_format_mask[devc->model]))
+ arg = AFMT_U8;
+
+ portc->audio_format = arg;
+
+ for (i = 0; i < n; i++)
+ if (format2bits[i].format == arg)
+ {
+ if ((portc->format_bits = format2bits[i].bits) == 0)
+ return portc->audio_format = AFMT_U8; /* Was not supported */
+
+ return arg;
+ }
+ /* Still hanging here. Something must be terribly wrong */
+ portc->format_bits = 0;
+ return portc->audio_format = AFMT_U8;
+}
+
+static struct audio_driver ad1848_audio_driver =
+{
+ .owner = THIS_MODULE,
+ .open = ad1848_open,
+ .close = ad1848_close,
+ .output_block = ad1848_output_block,
+ .start_input = ad1848_start_input,
+ .prepare_for_input = ad1848_prepare_for_input,
+ .prepare_for_output = ad1848_prepare_for_output,
+ .halt_io = ad1848_halt,
+ .halt_input = ad1848_halt_input,
+ .halt_output = ad1848_halt_output,
+ .trigger = ad1848_trigger,
+ .set_speed = ad1848_set_speed,
+ .set_bits = ad1848_set_bits,
+ .set_channels = ad1848_set_channels
+};
+
+static struct mixer_operations ad1848_mixer_operations =
+{
+ .owner = THIS_MODULE,
+ .id = "SOUNDPORT",
+ .name = "AD1848/CS4248/CS4231",
+ .ioctl = ad1848_mixer_ioctl
+};
+
+static int ad1848_open(int dev, int mode)
+{
+ ad1848_info *devc;
+ ad1848_port_info *portc;
+ unsigned long flags;
+
+ if (dev < 0 || dev >= num_audiodevs)
+ return -ENXIO;
+
+ devc = (ad1848_info *) audio_devs[dev]->devc;
+ portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ /* here we don't have to protect against intr */
+ spin_lock(&devc->lock);
+ if (portc->open_mode || (devc->open_mode & mode))
+ {
+ spin_unlock(&devc->lock);
+ return -EBUSY;
+ }
+ devc->dual_dma = 0;
+
+ if (audio_devs[dev]->flags & DMA_DUPLEX)
+ {
+ devc->dual_dma = 1;
+ }
+ devc->intr_active = 0;
+ devc->audio_mode = 0;
+ devc->open_mode |= mode;
+ portc->open_mode = mode;
+ spin_unlock(&devc->lock);
+ ad1848_trigger(dev, 0);
+
+ if (mode & OPEN_READ)
+ devc->record_dev = dev;
+ if (mode & OPEN_WRITE)
+ devc->playback_dev = dev;
+/*
+ * Mute output until the playback really starts. This decreases clicking (hope so).
+ */
+ spin_lock_irqsave(&devc->lock,flags);
+ ad_mute(devc);
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ return 0;
+}
+
+static void ad1848_close(int dev)
+{
+ unsigned long flags;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ DEB(printk("ad1848_close(void)\n"));
+
+ devc->intr_active = 0;
+ ad1848_halt(dev);
+
+ spin_lock_irqsave(&devc->lock,flags);
+
+ devc->audio_mode = 0;
+ devc->open_mode &= ~portc->open_mode;
+ portc->open_mode = 0;
+
+ ad_unmute(devc);
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void ad1848_output_block(int dev, unsigned long buf, int count, int intrflag)
+{
+ unsigned long flags, cnt;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ cnt = count;
+
+ if (portc->audio_format == AFMT_IMA_ADPCM)
+ {
+ cnt /= 4;
+ }
+ else
+ {
+ if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE)) /* 16 bit data */
+ cnt >>= 1;
+ }
+ if (portc->channels > 1)
+ cnt >>= 1;
+ cnt--;
+
+ if ((devc->audio_mode & PCM_ENABLE_OUTPUT) && (audio_devs[dev]->flags & DMA_AUTOMODE) &&
+ intrflag &&
+ cnt == devc->xfer_count)
+ {
+ devc->audio_mode |= PCM_ENABLE_OUTPUT;
+ devc->intr_active = 1;
+ return; /*
+ * Auto DMA mode on. No need to react
+ */
+ }
+ spin_lock_irqsave(&devc->lock,flags);
+
+ ad_write(devc, 15, (unsigned char) (cnt & 0xff));
+ ad_write(devc, 14, (unsigned char) ((cnt >> 8) & 0xff));
+
+ devc->xfer_count = cnt;
+ devc->audio_mode |= PCM_ENABLE_OUTPUT;
+ devc->intr_active = 1;
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void ad1848_start_input(int dev, unsigned long buf, int count, int intrflag)
+{
+ unsigned long flags, cnt;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ cnt = count;
+ if (portc->audio_format == AFMT_IMA_ADPCM)
+ {
+ cnt /= 4;
+ }
+ else
+ {
+ if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE)) /* 16 bit data */
+ cnt >>= 1;
+ }
+ if (portc->channels > 1)
+ cnt >>= 1;
+ cnt--;
+
+ if ((devc->audio_mode & PCM_ENABLE_INPUT) && (audio_devs[dev]->flags & DMA_AUTOMODE) &&
+ intrflag &&
+ cnt == devc->xfer_count)
+ {
+ devc->audio_mode |= PCM_ENABLE_INPUT;
+ devc->intr_active = 1;
+ return; /*
+ * Auto DMA mode on. No need to react
+ */
+ }
+ spin_lock_irqsave(&devc->lock,flags);
+
+ if (devc->model == MD_1848)
+ {
+ ad_write(devc, 15, (unsigned char) (cnt & 0xff));
+ ad_write(devc, 14, (unsigned char) ((cnt >> 8) & 0xff));
+ }
+ else
+ {
+ ad_write(devc, 31, (unsigned char) (cnt & 0xff));
+ ad_write(devc, 30, (unsigned char) ((cnt >> 8) & 0xff));
+ }
+
+ ad_unmute(devc);
+
+ devc->xfer_count = cnt;
+ devc->audio_mode |= PCM_ENABLE_INPUT;
+ devc->intr_active = 1;
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static int ad1848_prepare_for_output(int dev, int bsize, int bcount)
+{
+ int timeout;
+ unsigned char fs, old_fs, tmp = 0;
+ unsigned long flags;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ ad_mute(devc);
+
+ spin_lock_irqsave(&devc->lock,flags);
+ fs = portc->speed_bits | (portc->format_bits << 5);
+
+ if (portc->channels > 1)
+ fs |= 0x10;
+
+ ad_enter_MCE(devc); /* Enables changes to the format select reg */
+
+ if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE) /* Use alternate speed select registers */
+ {
+ fs &= 0xf0; /* Mask off the rate select bits */
+
+ ad_write(devc, 22, (portc->speed >> 8) & 0xff); /* Speed MSB */
+ ad_write(devc, 23, portc->speed & 0xff); /* Speed LSB */
+ }
+ old_fs = ad_read(devc, 8);
+
+ if (devc->model == MD_4232 || devc->model >= MD_4236)
+ {
+ tmp = ad_read(devc, 16);
+ ad_write(devc, 16, tmp | 0x30);
+ }
+ if (devc->model == MD_IWAVE)
+ ad_write(devc, 17, 0xc2); /* Disable variable frequency select */
+
+ ad_write(devc, 8, fs);
+
+ /*
+ * Write to I8 starts resynchronization. Wait until it completes.
+ */
+
+ timeout = 0;
+ while (timeout < 100 && inb(devc->base) != 0x80)
+ timeout++;
+ timeout = 0;
+ while (timeout < 10000 && inb(devc->base) == 0x80)
+ timeout++;
+
+ if (devc->model >= MD_4232)
+ ad_write(devc, 16, tmp & ~0x30);
+
+ ad_leave_MCE(devc); /*
+ * Starts the calibration process.
+ */
+ spin_unlock_irqrestore(&devc->lock,flags);
+ devc->xfer_count = 0;
+
+#ifndef EXCLUDE_TIMERS
+ if (dev == timer_installed && devc->timer_running)
+ if ((fs & 0x01) != (old_fs & 0x01))
+ {
+ ad1848_tmr_reprogram(dev);
+ }
+#endif
+ ad1848_halt_output(dev);
+ return 0;
+}
+
+static int ad1848_prepare_for_input(int dev, int bsize, int bcount)
+{
+ int timeout;
+ unsigned char fs, old_fs, tmp = 0;
+ unsigned long flags;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ if (devc->audio_mode)
+ return 0;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ fs = portc->speed_bits | (portc->format_bits << 5);
+
+ if (portc->channels > 1)
+ fs |= 0x10;
+
+ ad_enter_MCE(devc); /* Enables changes to the format select reg */
+
+ if ((devc->model == MD_1845) || (devc->model == MD_1845_SSCAPE)) /* Use alternate speed select registers */
+ {
+ fs &= 0xf0; /* Mask off the rate select bits */
+
+ ad_write(devc, 22, (portc->speed >> 8) & 0xff); /* Speed MSB */
+ ad_write(devc, 23, portc->speed & 0xff); /* Speed LSB */
+ }
+ if (devc->model == MD_4232)
+ {
+ tmp = ad_read(devc, 16);
+ ad_write(devc, 16, tmp | 0x30);
+ }
+ if (devc->model == MD_IWAVE)
+ ad_write(devc, 17, 0xc2); /* Disable variable frequency select */
+
+ /*
+ * If mode >= 2 (CS4231), set I28. It's the capture format register.
+ */
+
+ if (devc->model != MD_1848)
+ {
+ old_fs = ad_read(devc, 28);
+ ad_write(devc, 28, fs);
+
+ /*
+ * Write to I28 starts resynchronization. Wait until it completes.
+ */
+
+ timeout = 0;
+ while (timeout < 100 && inb(devc->base) != 0x80)
+ timeout++;
+
+ timeout = 0;
+ while (timeout < 10000 && inb(devc->base) == 0x80)
+ timeout++;
+
+ if (devc->model != MD_1848 && devc->model != MD_1845 && devc->model != MD_1845_SSCAPE)
+ {
+ /*
+ * CS4231 compatible devices don't have separate sampling rate selection
+ * register for recording an playback. The I8 register is shared so we have to
+ * set the speed encoding bits of it too.
+ */
+ unsigned char tmp = portc->speed_bits | (ad_read(devc, 8) & 0xf0);
+
+ ad_write(devc, 8, tmp);
+ /*
+ * Write to I8 starts resynchronization. Wait until it completes.
+ */
+ timeout = 0;
+ while (timeout < 100 && inb(devc->base) != 0x80)
+ timeout++;
+
+ timeout = 0;
+ while (timeout < 10000 && inb(devc->base) == 0x80)
+ timeout++;
+ }
+ }
+ else
+ { /* For AD1848 set I8. */
+
+ old_fs = ad_read(devc, 8);
+ ad_write(devc, 8, fs);
+ /*
+ * Write to I8 starts resynchronization. Wait until it completes.
+ */
+ timeout = 0;
+ while (timeout < 100 && inb(devc->base) != 0x80)
+ timeout++;
+ timeout = 0;
+ while (timeout < 10000 && inb(devc->base) == 0x80)
+ timeout++;
+ }
+
+ if (devc->model == MD_4232)
+ ad_write(devc, 16, tmp & ~0x30);
+
+ ad_leave_MCE(devc); /*
+ * Starts the calibration process.
+ */
+ spin_unlock_irqrestore(&devc->lock,flags);
+ devc->xfer_count = 0;
+
+#ifndef EXCLUDE_TIMERS
+ if (dev == timer_installed && devc->timer_running)
+ {
+ if ((fs & 0x01) != (old_fs & 0x01))
+ {
+ ad1848_tmr_reprogram(dev);
+ }
+ }
+#endif
+ ad1848_halt_input(dev);
+ return 0;
+}
+
+static void ad1848_halt(int dev)
+{
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+
+ unsigned char bits = ad_read(devc, 9);
+
+ if (bits & 0x01 && (portc->open_mode & OPEN_WRITE))
+ ad1848_halt_output(dev);
+
+ if (bits & 0x02 && (portc->open_mode & OPEN_READ))
+ ad1848_halt_input(dev);
+ devc->audio_mode = 0;
+}
+
+static void ad1848_halt_input(int dev)
+{
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ unsigned long flags;
+
+ if (!(ad_read(devc, 9) & 0x02))
+ return; /* Capture not enabled */
+
+ spin_lock_irqsave(&devc->lock,flags);
+
+ ad_mute(devc);
+
+ {
+ int tmout;
+
+ if(!isa_dma_bridge_buggy)
+ disable_dma(audio_devs[dev]->dmap_in->dma);
+
+ for (tmout = 0; tmout < 100000; tmout++)
+ if (ad_read(devc, 11) & 0x10)
+ break;
+ ad_write(devc, 9, ad_read(devc, 9) & ~0x02); /* Stop capture */
+
+ if(!isa_dma_bridge_buggy)
+ enable_dma(audio_devs[dev]->dmap_in->dma);
+ devc->audio_mode &= ~PCM_ENABLE_INPUT;
+ }
+
+ outb(0, io_Status(devc)); /* Clear interrupt status */
+ outb(0, io_Status(devc)); /* Clear interrupt status */
+
+ devc->audio_mode &= ~PCM_ENABLE_INPUT;
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void ad1848_halt_output(int dev)
+{
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ unsigned long flags;
+
+ if (!(ad_read(devc, 9) & 0x01))
+ return; /* Playback not enabled */
+
+ spin_lock_irqsave(&devc->lock,flags);
+
+ ad_mute(devc);
+ {
+ int tmout;
+
+ if(!isa_dma_bridge_buggy)
+ disable_dma(audio_devs[dev]->dmap_out->dma);
+
+ for (tmout = 0; tmout < 100000; tmout++)
+ if (ad_read(devc, 11) & 0x10)
+ break;
+ ad_write(devc, 9, ad_read(devc, 9) & ~0x01); /* Stop playback */
+
+ if(!isa_dma_bridge_buggy)
+ enable_dma(audio_devs[dev]->dmap_out->dma);
+
+ devc->audio_mode &= ~PCM_ENABLE_OUTPUT;
+ }
+
+ outb((0), io_Status(devc)); /* Clear interrupt status */
+ outb((0), io_Status(devc)); /* Clear interrupt status */
+
+ devc->audio_mode &= ~PCM_ENABLE_OUTPUT;
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void ad1848_trigger(int dev, int state)
+{
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc;
+ unsigned long flags;
+ unsigned char tmp, old;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ state &= devc->audio_mode;
+
+ tmp = old = ad_read(devc, 9);
+
+ if (portc->open_mode & OPEN_READ)
+ {
+ if (state & PCM_ENABLE_INPUT)
+ tmp |= 0x02;
+ else
+ tmp &= ~0x02;
+ }
+ if (portc->open_mode & OPEN_WRITE)
+ {
+ if (state & PCM_ENABLE_OUTPUT)
+ tmp |= 0x01;
+ else
+ tmp &= ~0x01;
+ }
+ /* ad_mute(devc); */
+ if (tmp != old)
+ {
+ ad_write(devc, 9, tmp);
+ ad_unmute(devc);
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void ad1848_init_hw(ad1848_info * devc)
+{
+ int i;
+ int *init_values;
+
+ /*
+ * Initial values for the indirect registers of CS4248/AD1848.
+ */
+ static int init_values_a[] =
+ {
+ 0xa8, 0xa8, 0x08, 0x08, 0x08, 0x08, 0x00, 0x00,
+ 0x00, 0x0c, 0x02, 0x00, 0x8a, 0x01, 0x00, 0x00,
+
+ /* Positions 16 to 31 just for CS4231/2 and ad1845 */
+ 0x80, 0x00, 0x10, 0x10, 0x00, 0x00, 0x1f, 0x40,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+ };
+
+ static int init_values_b[] =
+ {
+ /*
+ Values for the newer chips
+ Some of the register initialization values were changed. In
+ order to get rid of the click that preceded PCM playback,
+ calibration was disabled on the 10th byte. On that same byte,
+ dual DMA was enabled; on the 11th byte, ADC dithering was
+ enabled, since that is theoretically desirable; on the 13th
+ byte, Mode 3 was selected, to enable access to extended
+ registers.
+ */
+ 0xa8, 0xa8, 0x08, 0x08, 0x08, 0x08, 0x00, 0x00,
+ 0x00, 0x00, 0x06, 0x00, 0xe0, 0x01, 0x00, 0x00,
+ 0x80, 0x00, 0x10, 0x10, 0x00, 0x00, 0x1f, 0x40,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+ };
+
+ /*
+ * Select initialisation data
+ */
+
+ init_values = init_values_a;
+ if(devc->model >= MD_4236)
+ init_values = init_values_b;
+
+ for (i = 0; i < 16; i++)
+ ad_write(devc, i, init_values[i]);
+
+
+ ad_mute(devc); /* Initialize some variables */
+ ad_unmute(devc); /* Leave it unmuted now */
+
+ if (devc->model > MD_1848)
+ {
+ if (devc->model == MD_1845_SSCAPE)
+ ad_write(devc, 12, ad_read(devc, 12) | 0x50);
+ else
+ ad_write(devc, 12, ad_read(devc, 12) | 0x40); /* Mode2 = enabled */
+
+ if (devc->model == MD_IWAVE)
+ ad_write(devc, 12, 0x6c); /* Select codec mode 3 */
+
+ if (devc->model != MD_1845_SSCAPE)
+ for (i = 16; i < 32; i++)
+ ad_write(devc, i, init_values[i]);
+
+ if (devc->model == MD_IWAVE)
+ ad_write(devc, 16, 0x30); /* Playback and capture counters enabled */
+ }
+ if (devc->model > MD_1848)
+ {
+ if (devc->audio_flags & DMA_DUPLEX)
+ ad_write(devc, 9, ad_read(devc, 9) & ~0x04); /* Dual DMA mode */
+ else
+ ad_write(devc, 9, ad_read(devc, 9) | 0x04); /* Single DMA mode */
+
+ if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE)
+ ad_write(devc, 27, ad_read(devc, 27) | 0x08); /* Alternate freq select enabled */
+
+ if (devc->model == MD_IWAVE)
+ { /* Some magic Interwave specific initialization */
+ ad_write(devc, 12, 0x6c); /* Select codec mode 3 */
+ ad_write(devc, 16, 0x30); /* Playback and capture counters enabled */
+ ad_write(devc, 17, 0xc2); /* Alternate feature enable */
+ }
+ }
+ else
+ {
+ devc->audio_flags &= ~DMA_DUPLEX;
+ ad_write(devc, 9, ad_read(devc, 9) | 0x04); /* Single DMA mode */
+ if (soundpro)
+ ad_write(devc, 12, ad_read(devc, 12) | 0x40); /* Mode2 = enabled */
+ }
+
+ outb((0), io_Status(devc)); /* Clear pending interrupts */
+
+ /*
+ * Toggle the MCE bit. It completes the initialization phase.
+ */
+
+ ad_enter_MCE(devc); /* In case the bit was off */
+ ad_leave_MCE(devc);
+
+ ad1848_mixer_reset(devc);
+}
+
+int ad1848_detect(struct resource *ports, int *ad_flags, int *osp)
+{
+ unsigned char tmp;
+ ad1848_info *devc = &adev_info[nr_ad1848_devs];
+ unsigned char tmp1 = 0xff, tmp2 = 0xff;
+ int optiC930 = 0; /* OPTi 82C930 flag */
+ int interwave = 0;
+ int ad1847_flag = 0;
+ int cs4248_flag = 0;
+ int sscape_flag = 0;
+ int io_base = ports->start;
+
+ int i;
+
+ DDB(printk("ad1848_detect(%x)\n", io_base));
+
+ if (ad_flags)
+ {
+ if (*ad_flags == 0x12345678)
+ {
+ interwave = 1;
+ *ad_flags = 0;
+ }
+
+ if (*ad_flags == 0x87654321)
+ {
+ sscape_flag = 1;
+ *ad_flags = 0;
+ }
+
+ if (*ad_flags == 0x12345677)
+ {
+ cs4248_flag = 1;
+ *ad_flags = 0;
+ }
+ }
+ if (nr_ad1848_devs >= MAX_AUDIO_DEV)
+ {
+ printk(KERN_ERR "ad1848 - Too many audio devices\n");
+ return 0;
+ }
+ spin_lock_init(&devc->lock);
+ devc->base = io_base;
+ devc->irq_ok = 0;
+ devc->timer_running = 0;
+ devc->MCE_bit = 0x40;
+ devc->irq = 0;
+ devc->open_mode = 0;
+ devc->chip_name = devc->name = "AD1848";
+ devc->model = MD_1848; /* AD1848 or CS4248 */
+ devc->levels = NULL;
+ devc->debug_flag = 0;
+
+ /*
+ * Check that the I/O address is in use.
+ *
+ * The bit 0x80 of the base I/O port is known to be 0 after the
+ * chip has performed its power on initialization. Just assume
+ * this has happened before the OS is starting.
+ *
+ * If the I/O address is unused, it typically returns 0xff.
+ */
+
+ if (inb(devc->base) == 0xff)
+ {
+ DDB(printk("ad1848_detect: The base I/O address appears to be dead\n"));
+ }
+
+ /*
+ * Wait for the device to stop initialization
+ */
+
+ DDB(printk("ad1848_detect() - step 0\n"));
+
+ for (i = 0; i < 10000000; i++)
+ {
+ unsigned char x = inb(devc->base);
+
+ if (x == 0xff || !(x & 0x80))
+ break;
+ }
+
+ DDB(printk("ad1848_detect() - step A\n"));
+
+ if (inb(devc->base) == 0x80) /* Not ready. Let's wait */
+ ad_leave_MCE(devc);
+
+ if ((inb(devc->base) & 0x80) != 0x00) /* Not a AD1848 */
+ {
+ DDB(printk("ad1848 detect error - step A (%02x)\n", (int) inb(devc->base)));
+ return 0;
+ }
+
+ /*
+ * Test if it's possible to change contents of the indirect registers.
+ * Registers 0 and 1 are ADC volume registers. The bit 0x10 is read only
+ * so try to avoid using it.
+ */
+
+ DDB(printk("ad1848_detect() - step B\n"));
+ ad_write(devc, 0, 0xaa);
+ ad_write(devc, 1, 0x45); /* 0x55 with bit 0x10 clear */
+
+ if ((tmp1 = ad_read(devc, 0)) != 0xaa || (tmp2 = ad_read(devc, 1)) != 0x45)
+ {
+ if (tmp2 == 0x65) /* AD1847 has couple of bits hardcoded to 1 */
+ ad1847_flag = 1;
+ else
+ {
+ DDB(printk("ad1848 detect error - step B (%x/%x)\n", tmp1, tmp2));
+ return 0;
+ }
+ }
+ DDB(printk("ad1848_detect() - step C\n"));
+ ad_write(devc, 0, 0x45);
+ ad_write(devc, 1, 0xaa);
+
+ if ((tmp1 = ad_read(devc, 0)) != 0x45 || (tmp2 = ad_read(devc, 1)) != 0xaa)
+ {
+ if (tmp2 == 0x8a) /* AD1847 has few bits hardcoded to 1 */
+ ad1847_flag = 1;
+ else
+ {
+ DDB(printk("ad1848 detect error - step C (%x/%x)\n", tmp1, tmp2));
+ return 0;
+ }
+ }
+
+ /*
+ * The indirect register I12 has some read only bits. Let's
+ * try to change them.
+ */
+
+ DDB(printk("ad1848_detect() - step D\n"));
+ tmp = ad_read(devc, 12);
+ ad_write(devc, 12, (~tmp) & 0x0f);
+
+ if ((tmp & 0x0f) != ((tmp1 = ad_read(devc, 12)) & 0x0f))
+ {
+ DDB(printk("ad1848 detect error - step D (%x)\n", tmp1));
+ return 0;
+ }
+
+ /*
+ * NOTE! Last 4 bits of the reg I12 tell the chip revision.
+ * 0x01=RevB and 0x0A=RevC.
+ */
+
+ /*
+ * The original AD1848/CS4248 has just 15 indirect registers. This means
+ * that I0 and I16 should return the same value (etc.).
+ * However this doesn't work with CS4248. Actually it seems to be impossible
+ * to detect if the chip is a CS4231 or CS4248.
+ * Ensure that the Mode2 enable bit of I12 is 0. Otherwise this test fails
+ * with CS4231.
+ */
+
+ /*
+ * OPTi 82C930 has mode2 control bit in another place. This test will fail
+ * with it. Accept this situation as a possible indication of this chip.
+ */
+
+ DDB(printk("ad1848_detect() - step F\n"));
+ ad_write(devc, 12, 0); /* Mode2=disabled */
+
+ for (i = 0; i < 16; i++)
+ {
+ if ((tmp1 = ad_read(devc, i)) != (tmp2 = ad_read(devc, i + 16)))
+ {
+ DDB(printk("ad1848 detect step F(%d/%x/%x) - OPTi chip???\n", i, tmp1, tmp2));
+ if (!ad1847_flag)
+ optiC930 = 1;
+ break;
+ }
+ }
+
+ /*
+ * Try to switch the chip to mode2 (CS4231) by setting the MODE2 bit (0x40).
+ * The bit 0x80 is always 1 in CS4248 and CS4231.
+ */
+
+ DDB(printk("ad1848_detect() - step G\n"));
+
+ if (ad_flags && *ad_flags == 400)
+ *ad_flags = 0;
+ else
+ ad_write(devc, 12, 0x40); /* Set mode2, clear 0x80 */
+
+
+ if (ad_flags)
+ *ad_flags = 0;
+
+ tmp1 = ad_read(devc, 12);
+ if (tmp1 & 0x80)
+ {
+ if (ad_flags)
+ *ad_flags |= AD_F_CS4248;
+
+ devc->chip_name = "CS4248"; /* Our best knowledge just now */
+ }
+ if (optiC930 || (tmp1 & 0xc0) == (0x80 | 0x40))
+ {
+ /*
+ * CS4231 detected - is it?
+ *
+ * Verify that setting I0 doesn't change I16.
+ */
+
+ DDB(printk("ad1848_detect() - step H\n"));
+ ad_write(devc, 16, 0); /* Set I16 to known value */
+
+ ad_write(devc, 0, 0x45);
+ if ((tmp1 = ad_read(devc, 16)) != 0x45) /* No change -> CS4231? */
+ {
+ ad_write(devc, 0, 0xaa);
+ if ((tmp1 = ad_read(devc, 16)) == 0xaa) /* Rotten bits? */
+ {
+ DDB(printk("ad1848 detect error - step H(%x)\n", tmp1));
+ return 0;
+ }
+
+ /*
+ * Verify that some bits of I25 are read only.
+ */
+
+ DDB(printk("ad1848_detect() - step I\n"));
+ tmp1 = ad_read(devc, 25); /* Original bits */
+ ad_write(devc, 25, ~tmp1); /* Invert all bits */
+ if ((ad_read(devc, 25) & 0xe7) == (tmp1 & 0xe7))
+ {
+ int id;
+
+ /*
+ * It's at least CS4231
+ */
+
+ devc->chip_name = "CS4231";
+ devc->model = MD_4231;
+
+ /*
+ * It could be an AD1845 or CS4231A as well.
+ * CS4231 and AD1845 report the same revision info in I25
+ * while the CS4231A reports different.
+ */
+
+ id = ad_read(devc, 25);
+ if ((id & 0xe7) == 0x80) /* Device busy??? */
+ id = ad_read(devc, 25);
+ if ((id & 0xe7) == 0x80) /* Device still busy??? */
+ id = ad_read(devc, 25);
+ DDB(printk("ad1848_detect() - step J (%02x/%02x)\n", id, ad_read(devc, 25)));
+
+ if ((id & 0xe7) == 0x80) {
+ /*
+ * It must be a CS4231 or AD1845. The register I23 of
+ * CS4231 is undefined and it appears to be read only.
+ * AD1845 uses I23 for setting sample rate. Assume
+ * the chip is AD1845 if I23 is changeable.
+ */
+
+ unsigned char tmp = ad_read(devc, 23);
+ ad_write(devc, 23, ~tmp);
+
+ if (interwave)
+ {
+ devc->model = MD_IWAVE;
+ devc->chip_name = "IWave";
+ }
+ else if (ad_read(devc, 23) != tmp) /* AD1845 ? */
+ {
+ devc->chip_name = "AD1845";
+ devc->model = MD_1845;
+ }
+ else if (cs4248_flag)
+ {
+ if (ad_flags)
+ *ad_flags |= AD_F_CS4248;
+ devc->chip_name = "CS4248";
+ devc->model = MD_1848;
+ ad_write(devc, 12, ad_read(devc, 12) & ~0x40); /* Mode2 off */
+ }
+ ad_write(devc, 23, tmp); /* Restore */
+ }
+ else
+ {
+ switch (id & 0x1f) {
+ case 3: /* CS4236/CS4235/CS42xB/CS4239 */
+ {
+ int xid;
+ ad_write(devc, 12, ad_read(devc, 12) | 0x60); /* switch to mode 3 */
+ ad_write(devc, 23, 0x9c); /* select extended register 25 */
+ xid = inb(io_Indexed_Data(devc));
+ ad_write(devc, 12, ad_read(devc, 12) & ~0x60); /* back to mode 0 */
+ switch (xid & 0x1f)
+ {
+ case 0x00:
+ devc->chip_name = "CS4237B(B)";
+ devc->model = MD_42xB;
+ break;
+ case 0x08:
+ /* Seems to be a 4238 ?? */
+ devc->chip_name = "CS4238";
+ devc->model = MD_42xB;
+ break;
+ case 0x09:
+ devc->chip_name = "CS4238B";
+ devc->model = MD_42xB;
+ break;
+ case 0x0b:
+ devc->chip_name = "CS4236B";
+ devc->model = MD_4236;
+ break;
+ case 0x10:
+ devc->chip_name = "CS4237B";
+ devc->model = MD_42xB;
+ break;
+ case 0x1d:
+ devc->chip_name = "CS4235";
+ devc->model = MD_4235;
+ break;
+ case 0x1e:
+ devc->chip_name = "CS4239";
+ devc->model = MD_4239;
+ break;
+ default:
+ printk("Chip ident is %X.\n", xid&0x1F);
+ devc->chip_name = "CS42xx";
+ devc->model = MD_4232;
+ break;
+ }
+ }
+ break;
+
+ case 2: /* CS4232/CS4232A */
+ devc->chip_name = "CS4232";
+ devc->model = MD_4232;
+ break;
+
+ case 0:
+ if ((id & 0xe0) == 0xa0)
+ {
+ devc->chip_name = "CS4231A";
+ devc->model = MD_4231A;
+ }
+ else
+ {
+ devc->chip_name = "CS4321";
+ devc->model = MD_4231;
+ }
+ break;
+
+ default: /* maybe */
+ DDB(printk("ad1848: I25 = %02x/%02x\n", ad_read(devc, 25), ad_read(devc, 25) & 0xe7));
+ if (optiC930)
+ {
+ devc->chip_name = "82C930";
+ devc->model = MD_C930;
+ }
+ else
+ {
+ devc->chip_name = "CS4231";
+ devc->model = MD_4231;
+ }
+ }
+ }
+ }
+ ad_write(devc, 25, tmp1); /* Restore bits */
+
+ DDB(printk("ad1848_detect() - step K\n"));
+ }
+ } else if (tmp1 == 0x0a) {
+ /*
+ * Is it perhaps a SoundPro CMI8330?
+ * If so, then we should be able to change indirect registers
+ * greater than I15 after activating MODE2, even though reading
+ * back I12 does not show it.
+ */
+
+ /*
+ * Let's try comparing register values
+ */
+ for (i = 0; i < 16; i++) {
+ if ((tmp1 = ad_read(devc, i)) != (tmp2 = ad_read(devc, i + 16))) {
+ DDB(printk("ad1848 detect step H(%d/%x/%x) - SoundPro chip?\n", i, tmp1, tmp2));
+ soundpro = 1;
+ devc->chip_name = "SoundPro CMI 8330";
+ break;
+ }
+ }
+ }
+
+ DDB(printk("ad1848_detect() - step L\n"));
+ if (ad_flags)
+ {
+ if (devc->model != MD_1848)
+ *ad_flags |= AD_F_CS4231;
+ }
+ DDB(printk("ad1848_detect() - Detected OK\n"));
+
+ if (devc->model == MD_1848 && ad1847_flag)
+ devc->chip_name = "AD1847";
+
+
+ if (sscape_flag == 1)
+ devc->model = MD_1845_SSCAPE;
+
+ return 1;
+}
+
+int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback,
+ int dma_capture, int share_dma, int *osp, struct module *owner)
+{
+ /*
+ * NOTE! If irq < 0, there is another driver which has allocated the IRQ
+ * so that this driver doesn't need to allocate/deallocate it.
+ * The actually used IRQ is ABS(irq).
+ */
+
+ int my_dev;
+ char dev_name[100];
+ int e;
+
+ ad1848_info *devc = &adev_info[nr_ad1848_devs];
+
+ ad1848_port_info *portc = NULL;
+
+ devc->irq = (irq > 0) ? irq : 0;
+ devc->open_mode = 0;
+ devc->timer_ticks = 0;
+ devc->dma1 = dma_playback;
+ devc->dma2 = dma_capture;
+ devc->subtype = cfg.card_subtype;
+ devc->audio_flags = DMA_AUTOMODE;
+ devc->playback_dev = devc->record_dev = 0;
+ if (name != NULL)
+ devc->name = name;
+
+ if (name != NULL && name[0] != 0)
+ sprintf(dev_name,
+ "%s (%s)", name, devc->chip_name);
+ else
+ sprintf(dev_name,
+ "Generic audio codec (%s)", devc->chip_name);
+
+ rename_region(ports, devc->name);
+
+ conf_printf2(dev_name, devc->base, devc->irq, dma_playback, dma_capture);
+
+ if (devc->model == MD_1848 || devc->model == MD_C930)
+ devc->audio_flags |= DMA_HARDSTOP;
+
+ if (devc->model > MD_1848)
+ {
+ if (devc->dma1 == devc->dma2 || devc->dma2 == -1 || devc->dma1 == -1)
+ devc->audio_flags &= ~DMA_DUPLEX;
+ else
+ devc->audio_flags |= DMA_DUPLEX;
+ }
+
+ portc = kmalloc(sizeof(ad1848_port_info), GFP_KERNEL);
+ if(portc==NULL) {
+ release_region(devc->base, 4);
+ return -1;
+ }
+
+ if ((my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION,
+ dev_name,
+ &ad1848_audio_driver,
+ sizeof(struct audio_driver),
+ devc->audio_flags,
+ ad_format_mask[devc->model],
+ devc,
+ dma_playback,
+ dma_capture)) < 0)
+ {
+ release_region(devc->base, 4);
+ kfree(portc);
+ return -1;
+ }
+
+ audio_devs[my_dev]->portc = portc;
+ audio_devs[my_dev]->mixer_dev = -1;
+ if (owner)
+ audio_devs[my_dev]->d->owner = owner;
+ memset((char *) portc, 0, sizeof(*portc));
+
+ nr_ad1848_devs++;
+
+ ad1848_init_hw(devc);
+
+ if (irq > 0)
+ {
+ devc->dev_no = my_dev;
+ if (request_irq(devc->irq, adintr, 0, devc->name,
+ (void *)(long)my_dev) < 0)
+ {
+ printk(KERN_WARNING "ad1848: Unable to allocate IRQ\n");
+ /* Don't free it either then.. */
+ devc->irq = 0;
+ }
+ if (capabilities[devc->model].flags & CAP_F_TIMER)
+ {
+#ifndef CONFIG_SMP
+ int x;
+ unsigned char tmp = ad_read(devc, 16);
+#endif
+
+ devc->timer_ticks = 0;
+
+ ad_write(devc, 21, 0x00); /* Timer MSB */
+ ad_write(devc, 20, 0x10); /* Timer LSB */
+#ifndef CONFIG_SMP
+ ad_write(devc, 16, tmp | 0x40); /* Enable timer */
+ for (x = 0; x < 100000 && devc->timer_ticks == 0; x++);
+ ad_write(devc, 16, tmp & ~0x40); /* Disable timer */
+
+ if (devc->timer_ticks == 0)
+ printk(KERN_WARNING "ad1848: Interrupt test failed (IRQ%d)\n", irq);
+ else
+ {
+ DDB(printk("Interrupt test OK\n"));
+ devc->irq_ok = 1;
+ }
+#else
+ devc->irq_ok = 1;
+#endif
+ }
+ else
+ devc->irq_ok = 1; /* Couldn't test. assume it's OK */
+ } else if (irq < 0)
+ irq2dev[-irq] = devc->dev_no = my_dev;
+
+#ifndef EXCLUDE_TIMERS
+ if ((capabilities[devc->model].flags & CAP_F_TIMER) &&
+ devc->irq_ok)
+ ad1848_tmr_install(my_dev);
+#endif
+
+ if (!share_dma)
+ {
+ if (sound_alloc_dma(dma_playback, devc->name))
+ printk(KERN_WARNING "ad1848.c: Can't allocate DMA%d\n", dma_playback);
+
+ if (dma_capture != dma_playback)
+ if (sound_alloc_dma(dma_capture, devc->name))
+ printk(KERN_WARNING "ad1848.c: Can't allocate DMA%d\n", dma_capture);
+ }
+
+ if ((e = sound_install_mixer(MIXER_DRIVER_VERSION,
+ dev_name,
+ &ad1848_mixer_operations,
+ sizeof(struct mixer_operations),
+ devc)) >= 0)
+ {
+ audio_devs[my_dev]->mixer_dev = e;
+ if (owner)
+ mixer_devs[e]->owner = owner;
+ }
+ return my_dev;
+}
+
+int ad1848_control(int cmd, int arg)
+{
+ ad1848_info *devc;
+ unsigned long flags;
+
+ if (nr_ad1848_devs < 1)
+ return -ENODEV;
+
+ devc = &adev_info[nr_ad1848_devs - 1];
+
+ switch (cmd)
+ {
+ case AD1848_SET_XTAL: /* Change clock frequency of AD1845 (only ) */
+ if (devc->model != MD_1845 || devc->model != MD_1845_SSCAPE)
+ return -EINVAL;
+ spin_lock_irqsave(&devc->lock,flags);
+ ad_enter_MCE(devc);
+ ad_write(devc, 29, (ad_read(devc, 29) & 0x1f) | (arg << 5));
+ ad_leave_MCE(devc);
+ spin_unlock_irqrestore(&devc->lock,flags);
+ break;
+
+ case AD1848_MIXER_REROUTE:
+ {
+ int o = (arg >> 8) & 0xff;
+ int n = arg & 0xff;
+
+ if (o < 0 || o >= SOUND_MIXER_NRDEVICES)
+ return -EINVAL;
+
+ if (!(devc->supported_devices & (1 << o)) &&
+ !(devc->supported_rec_devices & (1 << o)))
+ return -EINVAL;
+
+ if (n == SOUND_MIXER_NONE)
+ { /* Just hide this control */
+ ad1848_mixer_set(devc, o, 0); /* Shut up it */
+ devc->supported_devices &= ~(1 << o);
+ devc->supported_rec_devices &= ~(1 << o);
+ break;
+ }
+
+ /* Make the mixer control identified by o to appear as n */
+ if (n < 0 || n >= SOUND_MIXER_NRDEVICES)
+ return -EINVAL;
+
+ devc->mixer_reroute[n] = o; /* Rename the control */
+ if (devc->supported_devices & (1 << o))
+ devc->supported_devices |= (1 << n);
+ if (devc->supported_rec_devices & (1 << o))
+ devc->supported_rec_devices |= (1 << n);
+
+ devc->supported_devices &= ~(1 << o);
+ devc->supported_rec_devices &= ~(1 << o);
+ }
+ break;
+ }
+ return 0;
+}
+
+void ad1848_unload(int io_base, int irq, int dma_playback, int dma_capture, int share_dma)
+{
+ int i, mixer, dev = 0;
+ ad1848_info *devc = NULL;
+
+ for (i = 0; devc == NULL && i < nr_ad1848_devs; i++)
+ {
+ if (adev_info[i].base == io_base)
+ {
+ devc = &adev_info[i];
+ dev = devc->dev_no;
+ }
+ }
+
+ if (devc != NULL)
+ {
+ kfree(audio_devs[dev]->portc);
+ release_region(devc->base, 4);
+
+ if (!share_dma)
+ {
+ if (devc->irq > 0) /* There is no point in freeing irq, if it wasn't allocated */
+ free_irq(devc->irq, (void *)(long)devc->dev_no);
+
+ sound_free_dma(dma_playback);
+
+ if (dma_playback != dma_capture)
+ sound_free_dma(dma_capture);
+
+ }
+ mixer = audio_devs[devc->dev_no]->mixer_dev;
+ if(mixer>=0)
+ sound_unload_mixerdev(mixer);
+
+ nr_ad1848_devs--;
+ for ( ; i < nr_ad1848_devs ; i++)
+ adev_info[i] = adev_info[i+1];
+ }
+ else
+ printk(KERN_ERR "ad1848: Can't find device to be unloaded. Base=%x\n", io_base);
+}
+
+static irqreturn_t adintr(int irq, void *dev_id)
+{
+ unsigned char status;
+ ad1848_info *devc;
+ int dev;
+ int alt_stat = 0xff;
+ unsigned char c930_stat = 0;
+ int cnt = 0;
+
+ dev = (long)dev_id;
+ devc = (ad1848_info *) audio_devs[dev]->devc;
+
+interrupt_again: /* Jump back here if int status doesn't reset */
+
+ status = inb(io_Status(devc));
+
+ if (status == 0x80)
+ printk(KERN_DEBUG "adintr: Why?\n");
+ if (devc->model == MD_1848)
+ outb((0), io_Status(devc)); /* Clear interrupt status */
+
+ if (status & 0x01)
+ {
+ if (devc->model == MD_C930)
+ { /* 82C930 has interrupt status register in MAD16 register MC11 */
+
+ spin_lock(&devc->lock);
+
+ /* 0xe0e is C930 address port
+ * 0xe0f is C930 data port
+ */
+ outb(11, 0xe0e);
+ c930_stat = inb(0xe0f);
+ outb((~c930_stat), 0xe0f);
+
+ spin_unlock(&devc->lock);
+
+ alt_stat = (c930_stat << 2) & 0x30;
+ }
+ else if (devc->model != MD_1848)
+ {
+ spin_lock(&devc->lock);
+ alt_stat = ad_read(devc, 24);
+ ad_write(devc, 24, ad_read(devc, 24) & ~alt_stat); /* Selective ack */
+ spin_unlock(&devc->lock);
+ }
+
+ if ((devc->open_mode & OPEN_READ) && (devc->audio_mode & PCM_ENABLE_INPUT) && (alt_stat & 0x20))
+ {
+ DMAbuf_inputintr(devc->record_dev);
+ }
+ if ((devc->open_mode & OPEN_WRITE) && (devc->audio_mode & PCM_ENABLE_OUTPUT) &&
+ (alt_stat & 0x10))
+ {
+ DMAbuf_outputintr(devc->playback_dev, 1);
+ }
+ if (devc->model != MD_1848 && (alt_stat & 0x40)) /* Timer interrupt */
+ {
+ devc->timer_ticks++;
+#ifndef EXCLUDE_TIMERS
+ if (timer_installed == dev && devc->timer_running)
+ sound_timer_interrupt();
+#endif
+ }
+ }
+/*
+ * Sometimes playback or capture interrupts occur while a timer interrupt
+ * is being handled. The interrupt will not be retriggered if we don't
+ * handle it now. Check if an interrupt is still pending and restart
+ * the handler in this case.
+ */
+ if (inb(io_Status(devc)) & 0x01 && cnt++ < 4)
+ {
+ goto interrupt_again;
+ }
+ return IRQ_HANDLED;
+}
+
+/*
+ * Experimental initialization sequence for the integrated sound system
+ * of the Compaq Deskpro M.
+ */
+
+static int init_deskpro_m(struct address_info *hw_config)
+{
+ unsigned char tmp;
+
+ if ((tmp = inb(0xc44)) == 0xff)
+ {
+ DDB(printk("init_deskpro_m: Dead port 0xc44\n"));
+ return 0;
+ }
+
+ outb(0x10, 0xc44);
+ outb(0x40, 0xc45);
+ outb(0x00, 0xc46);
+ outb(0xe8, 0xc47);
+ outb(0x14, 0xc44);
+ outb(0x40, 0xc45);
+ outb(0x00, 0xc46);
+ outb(0xe8, 0xc47);
+ outb(0x10, 0xc44);
+
+ return 1;
+}
+
+/*
+ * Experimental initialization sequence for the integrated sound system
+ * of Compaq Deskpro XL.
+ */
+
+static int init_deskpro(struct address_info *hw_config)
+{
+ unsigned char tmp;
+
+ if ((tmp = inb(0xc44)) == 0xff)
+ {
+ DDB(printk("init_deskpro: Dead port 0xc44\n"));
+ return 0;
+ }
+ outb((tmp | 0x04), 0xc44); /* Select bank 1 */
+ if (inb(0xc44) != 0x04)
+ {
+ DDB(printk("init_deskpro: Invalid bank1 signature in port 0xc44\n"));
+ return 0;
+ }
+ /*
+ * OK. It looks like a Deskpro so let's proceed.
+ */
+
+ /*
+ * I/O port 0xc44 Audio configuration register.
+ *
+ * bits 0xc0: Audio revision bits
+ * 0x00 = Compaq Business Audio
+ * 0x40 = MS Sound System Compatible (reset default)
+ * 0x80 = Reserved
+ * 0xc0 = Reserved
+ * bit 0x20: No Wait State Enable
+ * 0x00 = Disabled (reset default, DMA mode)
+ * 0x20 = Enabled (programmed I/O mode)
+ * bit 0x10: MS Sound System Decode Enable
+ * 0x00 = Decoding disabled (reset default)
+ * 0x10 = Decoding enabled
+ * bit 0x08: FM Synthesis Decode Enable
+ * 0x00 = Decoding Disabled (reset default)
+ * 0x08 = Decoding enabled
+ * bit 0x04 Bank select
+ * 0x00 = Bank 0
+ * 0x04 = Bank 1
+ * bits 0x03 MSS Base address
+ * 0x00 = 0x530 (reset default)
+ * 0x01 = 0x604
+ * 0x02 = 0xf40
+ * 0x03 = 0xe80
+ */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc44 (before): ");
+ outb((tmp & ~0x04), 0xc44);
+ printk("%02x ", inb(0xc44));
+ outb((tmp | 0x04), 0xc44);
+ printk("%02x\n", inb(0xc44));
+#endif
+
+ /* Set bank 1 of the register */
+ tmp = 0x58; /* MSS Mode, MSS&FM decode enabled */
+
+ switch (hw_config->io_base)
+ {
+ case 0x530:
+ tmp |= 0x00;
+ break;
+ case 0x604:
+ tmp |= 0x01;
+ break;
+ case 0xf40:
+ tmp |= 0x02;
+ break;
+ case 0xe80:
+ tmp |= 0x03;
+ break;
+ default:
+ DDB(printk("init_deskpro: Invalid MSS port %x\n", hw_config->io_base));
+ return 0;
+ }
+ outb((tmp & ~0x04), 0xc44); /* Write to bank=0 */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc44 (after): ");
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ printk("%02x ", inb(0xc44));
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ printk("%02x\n", inb(0xc44));
+#endif
+
+ /*
+ * I/O port 0xc45 FM Address Decode/MSS ID Register.
+ *
+ * bank=0, bits 0xfe: FM synthesis Decode Compare bits 7:1 (default=0x88)
+ * bank=0, bit 0x01: SBIC Power Control Bit
+ * 0x00 = Powered up
+ * 0x01 = Powered down
+ * bank=1, bits 0xfc: MSS ID (default=0x40)
+ */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc45 (before): ");
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ printk("%02x ", inb(0xc45));
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ printk("%02x\n", inb(0xc45));
+#endif
+
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ outb((0x88), 0xc45); /* FM base 7:0 = 0x88 */
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ outb((0x10), 0xc45); /* MSS ID = 0x10 (MSS port returns 0x04) */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc45 (after): ");
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ printk("%02x ", inb(0xc45));
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ printk("%02x\n", inb(0xc45));
+#endif
+
+
+ /*
+ * I/O port 0xc46 FM Address Decode/Address ASIC Revision Register.
+ *
+ * bank=0, bits 0xff: FM synthesis Decode Compare bits 15:8 (default=0x03)
+ * bank=1, bits 0xff: Audio addressing ASIC id
+ */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc46 (before): ");
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ printk("%02x ", inb(0xc46));
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ printk("%02x\n", inb(0xc46));
+#endif
+
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ outb((0x03), 0xc46); /* FM base 15:8 = 0x03 */
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ outb((0x11), 0xc46); /* ASIC ID = 0x11 */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc46 (after): ");
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ printk("%02x ", inb(0xc46));
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ printk("%02x\n", inb(0xc46));
+#endif
+
+ /*
+ * I/O port 0xc47 FM Address Decode Register.
+ *
+ * bank=0, bits 0xff: Decode enable selection for various FM address bits
+ * bank=1, bits 0xff: Reserved
+ */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc47 (before): ");
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ printk("%02x ", inb(0xc47));
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ printk("%02x\n", inb(0xc47));
+#endif
+
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ outb((0x7c), 0xc47); /* FM decode enable bits = 0x7c */
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ outb((0x00), 0xc47); /* Reserved bank1 = 0x00 */
+
+#ifdef DEBUGXL
+ /* Debug printing */
+ printk("Port 0xc47 (after): ");
+ outb((tmp & ~0x04), 0xc44); /* Select bank=0 */
+ printk("%02x ", inb(0xc47));
+ outb((tmp | 0x04), 0xc44); /* Select bank=1 */
+ printk("%02x\n", inb(0xc47));
+#endif
+
+ /*
+ * I/O port 0xc6f = Audio Disable Function Register
+ */
+
+#ifdef DEBUGXL
+ printk("Port 0xc6f (before) = %02x\n", inb(0xc6f));
+#endif
+
+ outb((0x80), 0xc6f);
+
+#ifdef DEBUGXL
+ printk("Port 0xc6f (after) = %02x\n", inb(0xc6f));
+#endif
+
+ return 1;
+}
+
+int probe_ms_sound(struct address_info *hw_config, struct resource *ports)
+{
+ unsigned char tmp;
+
+ DDB(printk("Entered probe_ms_sound(%x, %d)\n", hw_config->io_base, hw_config->card_subtype));
+
+ if (hw_config->card_subtype == 1) /* Has no IRQ/DMA registers */
+ {
+ /* check_opl3(0x388, hw_config); */
+ return ad1848_detect(ports, NULL, hw_config->osp);
+ }
+
+ if (deskpro_xl && hw_config->card_subtype == 2) /* Compaq Deskpro XL */
+ {
+ if (!init_deskpro(hw_config))
+ return 0;
+ }
+
+ if (deskpro_m) /* Compaq Deskpro M */
+ {
+ if (!init_deskpro_m(hw_config))
+ return 0;
+ }
+
+ /*
+ * Check if the IO port returns valid signature. The original MS Sound
+ * system returns 0x04 while some cards (AudioTrix Pro for example)
+ * return 0x00 or 0x0f.
+ */
+
+ if ((tmp = inb(hw_config->io_base + 3)) == 0xff) /* Bus float */
+ {
+ int ret;
+
+ DDB(printk("I/O address is inactive (%x)\n", tmp));
+ if (!(ret = ad1848_detect(ports, NULL, hw_config->osp)))
+ return 0;
+ return 1;
+ }
+ DDB(printk("MSS signature = %x\n", tmp & 0x3f));
+ if ((tmp & 0x3f) != 0x04 &&
+ (tmp & 0x3f) != 0x0f &&
+ (tmp & 0x3f) != 0x00)
+ {
+ int ret;
+
+ MDB(printk(KERN_ERR "No MSS signature detected on port 0x%x (0x%x)\n", hw_config->io_base, (int) inb(hw_config->io_base + 3)));
+ DDB(printk("Trying to detect codec anyway but IRQ/DMA may not work\n"));
+ if (!(ret = ad1848_detect(ports, NULL, hw_config->osp)))
+ return 0;
+
+ hw_config->card_subtype = 1;
+ return 1;
+ }
+ if ((hw_config->irq != 5) &&
+ (hw_config->irq != 7) &&
+ (hw_config->irq != 9) &&
+ (hw_config->irq != 10) &&
+ (hw_config->irq != 11) &&
+ (hw_config->irq != 12))
+ {
+ printk(KERN_ERR "MSS: Bad IRQ %d\n", hw_config->irq);
+ return 0;
+ }
+ if (hw_config->dma != 0 && hw_config->dma != 1 && hw_config->dma != 3)
+ {
+ printk(KERN_ERR "MSS: Bad DMA %d\n", hw_config->dma);
+ return 0;
+ }
+ /*
+ * Check that DMA0 is not in use with a 8 bit board.
+ */
+
+ if (hw_config->dma == 0 && inb(hw_config->io_base + 3) & 0x80)
+ {
+ printk(KERN_ERR "MSS: Can't use DMA0 with a 8 bit card/slot\n");
+ return 0;
+ }
+ if (hw_config->irq > 7 && hw_config->irq != 9 && inb(hw_config->io_base + 3) & 0x80)
+ {
+ printk(KERN_ERR "MSS: Can't use IRQ%d with a 8 bit card/slot\n", hw_config->irq);
+ return 0;
+ }
+ return ad1848_detect(ports, NULL, hw_config->osp);
+}
+
+void attach_ms_sound(struct address_info *hw_config, struct resource *ports, struct module *owner)
+{
+ static signed char interrupt_bits[12] =
+ {
+ -1, -1, -1, -1, -1, 0x00, -1, 0x08, -1, 0x10, 0x18, 0x20
+ };
+ signed char bits;
+ char dma2_bit = 0;
+
+ static char dma_bits[4] =
+ {
+ 1, 2, 0, 3
+ };
+
+ int config_port = hw_config->io_base + 0;
+ int version_port = hw_config->io_base + 3;
+ int dma = hw_config->dma;
+ int dma2 = hw_config->dma2;
+
+ if (hw_config->card_subtype == 1) /* Has no IRQ/DMA registers */
+ {
+ hw_config->slots[0] = ad1848_init("MS Sound System", ports,
+ hw_config->irq,
+ hw_config->dma,
+ hw_config->dma2, 0,
+ hw_config->osp,
+ owner);
+ return;
+ }
+ /*
+ * Set the IRQ and DMA addresses.
+ */
+
+ bits = interrupt_bits[hw_config->irq];
+ if (bits == -1)
+ {
+ printk(KERN_ERR "MSS: Bad IRQ %d\n", hw_config->irq);
+ release_region(ports->start, 4);
+ release_region(ports->start - 4, 4);
+ return;
+ }
+ outb((bits | 0x40), config_port);
+ if ((inb(version_port) & 0x40) == 0)
+ printk(KERN_ERR "[MSS: IRQ Conflict?]\n");
+
+/*
+ * Handle the capture DMA channel
+ */
+
+ if (dma2 != -1 && dma2 != dma)
+ {
+ if (!((dma == 0 && dma2 == 1) ||
+ (dma == 1 && dma2 == 0) ||
+ (dma == 3 && dma2 == 0)))
+ { /* Unsupported combination. Try to swap channels */
+ int tmp = dma;
+
+ dma = dma2;
+ dma2 = tmp;
+ }
+ if ((dma == 0 && dma2 == 1) ||
+ (dma == 1 && dma2 == 0) ||
+ (dma == 3 && dma2 == 0))
+ {
+ dma2_bit = 0x04; /* Enable capture DMA */
+ }
+ else
+ {
+ printk(KERN_WARNING "MSS: Invalid capture DMA\n");
+ dma2 = dma;
+ }
+ }
+ else
+ {
+ dma2 = dma;
+ }
+
+ hw_config->dma = dma;
+ hw_config->dma2 = dma2;
+
+ outb((bits | dma_bits[dma] | dma2_bit), config_port); /* Write IRQ+DMA setup */
+
+ hw_config->slots[0] = ad1848_init("MS Sound System", ports,
+ hw_config->irq,
+ dma, dma2, 0,
+ hw_config->osp,
+ THIS_MODULE);
+}
+
+void unload_ms_sound(struct address_info *hw_config)
+{
+ ad1848_unload(hw_config->io_base + 4,
+ hw_config->irq,
+ hw_config->dma,
+ hw_config->dma2, 0);
+ sound_unload_audiodev(hw_config->slots[0]);
+ release_region(hw_config->io_base, 4);
+}
+
+#ifndef EXCLUDE_TIMERS
+
+/*
+ * Timer stuff (for /dev/music).
+ */
+
+static unsigned int current_interval;
+
+static unsigned int ad1848_tmr_start(int dev, unsigned int usecs)
+{
+ unsigned long flags;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+ unsigned long xtal_nsecs; /* nanoseconds per xtal oscillator tick */
+ unsigned long divider;
+
+ spin_lock_irqsave(&devc->lock,flags);
+
+ /*
+ * Length of the timer interval (in nanoseconds) depends on the
+ * selected crystal oscillator. Check this from bit 0x01 of I8.
+ *
+ * AD1845 has just one oscillator which has cycle time of 10.050 us
+ * (when a 24.576 MHz xtal oscillator is used).
+ *
+ * Convert requested interval to nanoseconds before computing
+ * the timer divider.
+ */
+
+ if (devc->model == MD_1845 || devc->model == MD_1845_SSCAPE)
+ xtal_nsecs = 10050;
+ else if (ad_read(devc, 8) & 0x01)
+ xtal_nsecs = 9920;
+ else
+ xtal_nsecs = 9969;
+
+ divider = (usecs * 1000 + xtal_nsecs / 2) / xtal_nsecs;
+
+ if (divider < 100) /* Don't allow shorter intervals than about 1ms */
+ divider = 100;
+
+ if (divider > 65535) /* Overflow check */
+ divider = 65535;
+
+ ad_write(devc, 21, (divider >> 8) & 0xff); /* Set upper bits */
+ ad_write(devc, 20, divider & 0xff); /* Set lower bits */
+ ad_write(devc, 16, ad_read(devc, 16) | 0x40); /* Start the timer */
+ devc->timer_running = 1;
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ return current_interval = (divider * xtal_nsecs + 500) / 1000;
+}
+
+static void ad1848_tmr_reprogram(int dev)
+{
+ /*
+ * Audio driver has changed sampling rate so that a different xtal
+ * oscillator was selected. We have to reprogram the timer rate.
+ */
+
+ ad1848_tmr_start(dev, current_interval);
+ sound_timer_syncinterval(current_interval);
+}
+
+static void ad1848_tmr_disable(int dev)
+{
+ unsigned long flags;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ ad_write(devc, 16, ad_read(devc, 16) & ~0x40);
+ devc->timer_running = 0;
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void ad1848_tmr_restart(int dev)
+{
+ unsigned long flags;
+ ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc;
+
+ if (current_interval == 0)
+ return;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ ad_write(devc, 16, ad_read(devc, 16) | 0x40);
+ devc->timer_running = 1;
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static struct sound_lowlev_timer ad1848_tmr =
+{
+ 0,
+ 2,
+ ad1848_tmr_start,
+ ad1848_tmr_disable,
+ ad1848_tmr_restart
+};
+
+static int ad1848_tmr_install(int dev)
+{
+ if (timer_installed != -1)
+ return 0; /* Don't install another timer */
+
+ timer_installed = ad1848_tmr.dev = dev;
+ sound_timer_init(&ad1848_tmr, audio_devs[dev]->name);
+
+ return 1;
+}
+#endif /* EXCLUDE_TIMERS */
+
+EXPORT_SYMBOL(ad1848_detect);
+EXPORT_SYMBOL(ad1848_init);
+EXPORT_SYMBOL(ad1848_unload);
+EXPORT_SYMBOL(ad1848_control);
+EXPORT_SYMBOL(probe_ms_sound);
+EXPORT_SYMBOL(attach_ms_sound);
+EXPORT_SYMBOL(unload_ms_sound);
+
+static int __initdata io = -1;
+static int __initdata irq = -1;
+static int __initdata dma = -1;
+static int __initdata dma2 = -1;
+static int __initdata type = 0;
+
+module_param(io, int, 0); /* I/O for a raw AD1848 card */
+module_param(irq, int, 0); /* IRQ to use */
+module_param(dma, int, 0); /* First DMA channel */
+module_param(dma2, int, 0); /* Second DMA channel */
+module_param(type, int, 0); /* Card type */
+module_param(deskpro_xl, bool, 0); /* Special magic for Deskpro XL boxen */
+module_param(deskpro_m, bool, 0); /* Special magic for Deskpro M box */
+module_param(soundpro, bool, 0); /* More special magic for SoundPro chips */
+
+#ifdef CONFIG_PNP
+module_param(isapnp, int, 0);
+module_param(isapnpjump, int, 0);
+module_param(reverse, bool, 0);
+MODULE_PARM_DESC(isapnp, "When set to 0, Plug & Play support will be disabled");
+MODULE_PARM_DESC(isapnpjump, "Jumps to a specific slot in the driver's PnP table. Use the source, Luke.");
+MODULE_PARM_DESC(reverse, "When set to 1, will reverse ISAPnP search order");
+
+static struct pnp_dev *ad1848_dev = NULL;
+
+/* Please add new entries at the end of the table */
+static struct {
+ char *name;
+ unsigned short card_vendor, card_device,
+ vendor, function;
+ short mss_io, irq, dma, dma2; /* index into isapnp table */
+ int type;
+} ad1848_isapnp_list[] __initdata = {
+ {"CMI 8330 SoundPRO",
+ ISAPNP_VENDOR('C','M','I'), ISAPNP_DEVICE(0x0001),
+ ISAPNP_VENDOR('@','@','@'), ISAPNP_FUNCTION(0x0001),
+ 0, 0, 0,-1, 0},
+ {"CS4232 based card",
+ ISAPNP_ANY_ID, ISAPNP_ANY_ID,
+ ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0000),
+ 0, 0, 0, 1, 0},
+ {"CS4232 based card",
+ ISAPNP_ANY_ID, ISAPNP_ANY_ID,
+ ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0100),
+ 0, 0, 0, 1, 0},
+ {"OPL3-SA2 WSS mode",
+ ISAPNP_ANY_ID, ISAPNP_ANY_ID,
+ ISAPNP_VENDOR('Y','M','H'), ISAPNP_FUNCTION(0x0021),
+ 1, 0, 0, 1, 1},
+ {"Advanced Gravis InterWave Audio",
+ ISAPNP_VENDOR('G','R','V'), ISAPNP_DEVICE(0x0001),
+ ISAPNP_VENDOR('G','R','V'), ISAPNP_FUNCTION(0x0000),
+ 0, 0, 0, 1, 0},
+ {NULL}
+};
+
+static struct isapnp_device_id id_table[] __devinitdata = {
+ { ISAPNP_VENDOR('C','M','I'), ISAPNP_DEVICE(0x0001),
+ ISAPNP_VENDOR('@','@','@'), ISAPNP_FUNCTION(0x0001), 0 },
+ { ISAPNP_ANY_ID, ISAPNP_ANY_ID,
+ ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0000), 0 },
+ { ISAPNP_ANY_ID, ISAPNP_ANY_ID,
+ ISAPNP_VENDOR('C','S','C'), ISAPNP_FUNCTION(0x0100), 0 },
+ /* The main driver for this card is opl3sa2
+ { ISAPNP_ANY_ID, ISAPNP_ANY_ID,
+ ISAPNP_VENDOR('Y','M','H'), ISAPNP_FUNCTION(0x0021), 0 },
+ */
+ { ISAPNP_VENDOR('G','R','V'), ISAPNP_DEVICE(0x0001),
+ ISAPNP_VENDOR('G','R','V'), ISAPNP_FUNCTION(0x0000), 0 },
+ {0}
+};
+
+MODULE_DEVICE_TABLE(isapnp, id_table);
+
+static struct pnp_dev *activate_dev(char *devname, char *resname, struct pnp_dev *dev)
+{
+ int err;
+
+ err = pnp_device_attach(dev);
+ if (err < 0)
+ return(NULL);
+
+ if((err = pnp_activate_dev(dev)) < 0) {
+ printk(KERN_ERR "ad1848: %s %s config failed (out of resources?)[%d]\n", devname, resname, err);
+
+ pnp_device_detach(dev);
+
+ return(NULL);
+ }
+ audio_activated = 1;
+ return(dev);
+}
+
+static struct pnp_dev __init *ad1848_init_generic(struct pnp_card *bus,
+ struct address_info *hw_config, int slot)
+{
+
+ /* Configure Audio device */
+ if((ad1848_dev = pnp_find_dev(bus, ad1848_isapnp_list[slot].vendor, ad1848_isapnp_list[slot].function, NULL)))
+ {
+ if((ad1848_dev = activate_dev(ad1848_isapnp_list[slot].name, "ad1848", ad1848_dev)))
+ {
+ hw_config->io_base = pnp_port_start(ad1848_dev, ad1848_isapnp_list[slot].mss_io);
+ hw_config->irq = pnp_irq(ad1848_dev, ad1848_isapnp_list[slot].irq);
+ hw_config->dma = pnp_dma(ad1848_dev, ad1848_isapnp_list[slot].dma);
+ if(ad1848_isapnp_list[slot].dma2 != -1)
+ hw_config->dma2 = pnp_dma(ad1848_dev, ad1848_isapnp_list[slot].dma2);
+ else
+ hw_config->dma2 = -1;
+ hw_config->card_subtype = ad1848_isapnp_list[slot].type;
+ } else
+ return(NULL);
+ } else
+ return(NULL);
+
+ return(ad1848_dev);
+}
+
+static int __init ad1848_isapnp_init(struct address_info *hw_config, struct pnp_card *bus, int slot)
+{
+ char *busname = bus->name[0] ? bus->name : ad1848_isapnp_list[slot].name;
+
+ /* Initialize this baby. */
+
+ if(ad1848_init_generic(bus, hw_config, slot)) {
+ /* We got it. */
+
+ printk(KERN_NOTICE "ad1848: PnP reports '%s' at i/o %#x, irq %d, dma %d, %d\n",
+ busname,
+ hw_config->io_base, hw_config->irq, hw_config->dma,
+ hw_config->dma2);
+ return 1;
+ }
+ return 0;
+}
+
+static int __init ad1848_isapnp_probe(struct address_info *hw_config)
+{
+ static int first = 1;
+ int i;
+
+ /* Count entries in sb_isapnp_list */
+ for (i = 0; ad1848_isapnp_list[i].card_vendor != 0; i++);
+ i--;
+
+ /* Check and adjust isapnpjump */
+ if( isapnpjump < 0 || isapnpjump > i) {
+ isapnpjump = reverse ? i : 0;
+ printk(KERN_ERR "ad1848: Valid range for isapnpjump is 0-%d. Adjusted to %d.\n", i, isapnpjump);
+ }
+
+ if(!first || !reverse)
+ i = isapnpjump;
+ first = 0;
+ while(ad1848_isapnp_list[i].card_vendor != 0) {
+ static struct pnp_card *bus = NULL;
+
+ while ((bus = pnp_find_card(
+ ad1848_isapnp_list[i].card_vendor,
+ ad1848_isapnp_list[i].card_device,
+ bus))) {
+
+ if(ad1848_isapnp_init(hw_config, bus, i)) {
+ isapnpjump = i; /* start next search from here */
+ return 0;
+ }
+ }
+ i += reverse ? -1 : 1;
+ }
+
+ return -ENODEV;
+}
+#endif
+
+
+static int __init init_ad1848(void)
+{
+ printk(KERN_INFO "ad1848/cs4248 codec driver Copyright (C) by Hannu Savolainen 1993-1996\n");
+
+#ifdef CONFIG_PNP
+ if(isapnp && (ad1848_isapnp_probe(&cfg) < 0) ) {
+ printk(KERN_NOTICE "ad1848: No ISAPnP cards found, trying standard ones...\n");
+ isapnp = 0;
+ }
+#endif
+
+ if(io != -1) {
+ struct resource *ports;
+ if( isapnp == 0 )
+ {
+ if(irq == -1 || dma == -1) {
+ printk(KERN_WARNING "ad1848: must give I/O , IRQ and DMA.\n");
+ return -EINVAL;
+ }
+
+ cfg.irq = irq;
+ cfg.io_base = io;
+ cfg.dma = dma;
+ cfg.dma2 = dma2;
+ cfg.card_subtype = type;
+ }
+
+ ports = request_region(io + 4, 4, "ad1848");
+
+ if (!ports)
+ return -EBUSY;
+
+ if (!request_region(io, 4, "WSS config")) {
+ release_region(io + 4, 4);
+ return -EBUSY;
+ }
+
+ if (!probe_ms_sound(&cfg, ports)) {
+ release_region(io + 4, 4);
+ release_region(io, 4);
+ return -ENODEV;
+ }
+ attach_ms_sound(&cfg, ports, THIS_MODULE);
+ loaded = 1;
+ }
+ return 0;
+}
+
+static void __exit cleanup_ad1848(void)
+{
+ if(loaded)
+ unload_ms_sound(&cfg);
+
+#ifdef CONFIG_PNP
+ if(ad1848_dev){
+ if(audio_activated)
+ pnp_device_detach(ad1848_dev);
+ }
+#endif
+}
+
+module_init(init_ad1848);
+module_exit(cleanup_ad1848);
+
+#ifndef MODULE
+static int __init setup_ad1848(char *str)
+{
+ /* io, irq, dma, dma2, type */
+ int ints[6];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+ dma = ints[3];
+ dma2 = ints[4];
+ type = ints[5];
+
+ return 1;
+}
+
+__setup("ad1848=", setup_ad1848);
+#endif
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/ad1848.h b/sound/oss/ad1848.h
new file mode 100644
index 0000000..b95ebe2
--- /dev/null
+++ b/sound/oss/ad1848.h
@@ -0,0 +1,24 @@
+
+#include <linux/interrupt.h>
+
+#define AD_F_CS4231 0x0001 /* Returned if a CS4232 (or compatible) detected */
+#define AD_F_CS4248 0x0001 /* Returned if a CS4248 (or compatible) detected */
+
+#define AD1848_SET_XTAL 1
+#define AD1848_MIXER_REROUTE 2
+
+#define AD1848_REROUTE(oldctl, newctl) \
+ ad1848_control(AD1848_MIXER_REROUTE, ((oldctl)<<8)|(newctl))
+
+
+int ad1848_init(char *name, struct resource *ports, int irq, int dma_playback,
+ int dma_capture, int share_dma, int *osp, struct module *owner);
+void ad1848_unload (int io_base, int irq, int dma_playback, int dma_capture, int share_dma);
+
+int ad1848_detect (struct resource *ports, int *flags, int *osp);
+int ad1848_control(int cmd, int arg);
+
+void attach_ms_sound(struct address_info * hw_config, struct resource *ports, struct module * owner);
+
+int probe_ms_sound(struct address_info *hw_config, struct resource *ports);
+void unload_ms_sound(struct address_info *hw_info);
diff --git a/sound/oss/ad1848_mixer.h b/sound/oss/ad1848_mixer.h
new file mode 100644
index 0000000..2cf719b
--- /dev/null
+++ b/sound/oss/ad1848_mixer.h
@@ -0,0 +1,253 @@
+/*
+ * sound/oss/ad1848_mixer.h
+ *
+ * Definitions for the mixer of AD1848 and compatible codecs.
+ */
+
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+
+
+/*
+ * The AD1848 codec has generic input lines called Line, Aux1 and Aux2.
+ * Sound card manufacturers have connected actual inputs (CD, synth, line,
+ * etc) to these inputs in different order. Therefore it's difficult
+ * to assign mixer channels to these inputs correctly. The following
+ * contains two alternative mappings. The first one is for GUS MAX and
+ * the second is just a generic one (line1, line2 and line3).
+ * (Actually this is not a mapping but rather some kind of interleaving
+ * solution).
+ */
+#define MODE1_REC_DEVICES (SOUND_MASK_LINE3 | SOUND_MASK_MIC | \
+ SOUND_MASK_LINE1 | SOUND_MASK_IMIX)
+
+#define SPRO_REC_DEVICES (SOUND_MASK_LINE | SOUND_MASK_MIC | \
+ SOUND_MASK_CD | SOUND_MASK_LINE1)
+
+#define MODE1_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_MIC | \
+ SOUND_MASK_LINE2 | \
+ SOUND_MASK_IGAIN | \
+ SOUND_MASK_PCM | SOUND_MASK_IMIX)
+
+#define MODE2_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | \
+ SOUND_MASK_MIC | \
+ SOUND_MASK_LINE3 | SOUND_MASK_SPEAKER | \
+ SOUND_MASK_IGAIN | \
+ SOUND_MASK_PCM | SOUND_MASK_IMIX)
+
+#define MODE3_MIXER_DEVICES (MODE2_MIXER_DEVICES | SOUND_MASK_VOLUME)
+
+/* OPTi 82C930 has no IMIX level control, but it can still be selected as an
+ * input
+ */
+#define C930_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | \
+ SOUND_MASK_MIC | SOUND_MASK_VOLUME | \
+ SOUND_MASK_LINE3 | \
+ SOUND_MASK_IGAIN | SOUND_MASK_PCM)
+
+#define SPRO_MIXER_DEVICES (SOUND_MASK_VOLUME | SOUND_MASK_PCM | \
+ SOUND_MASK_LINE | SOUND_MASK_SYNTH | \
+ SOUND_MASK_CD | SOUND_MASK_MIC | \
+ SOUND_MASK_SPEAKER | SOUND_MASK_LINE1 | \
+ SOUND_MASK_OGAIN)
+
+struct mixer_def {
+ unsigned int regno:6; /* register number for volume */
+ unsigned int polarity:1; /* volume polarity: 0=normal, 1=reversed */
+ unsigned int bitpos:3; /* position of bits in register for volume */
+ unsigned int nbits:3; /* number of bits in register for volume */
+ unsigned int mutereg:6; /* register number for mute bit */
+ unsigned int mutepol:1; /* mute polarity: 0=normal, 1=reversed */
+ unsigned int mutepos:4; /* position of mute bit in register */
+ unsigned int recreg:6; /* register number for recording bit */
+ unsigned int recpol:1; /* recording polarity: 0=normal, 1=reversed */
+ unsigned int recpos:4; /* position of recording bit in register */
+};
+
+static char mix_cvt[101] = {
+ 0, 0, 3, 7,10,13,16,19,21,23,26,28,30,32,34,35,37,39,40,42,
+ 43,45,46,47,49,50,51,52,53,55,56,57,58,59,60,61,62,63,64,65,
+ 65,66,67,68,69,70,70,71,72,73,73,74,75,75,76,77,77,78,79,79,
+ 80,81,81,82,82,83,84,84,85,85,86,86,87,87,88,88,89,89,90,90,
+ 91,91,92,92,93,93,94,94,95,95,96,96,96,97,97,98,98,98,99,99,
+ 100
+};
+
+typedef struct mixer_def mixer_ent;
+typedef mixer_ent mixer_ents[2];
+
+/*
+ * Most of the mixer entries work in backwards. Setting the polarity field
+ * makes them to work correctly.
+ *
+ * The channel numbering used by individual sound cards is not fixed. Some
+ * cards have assigned different meanings for the AUX1, AUX2 and LINE inputs.
+ * The current version doesn't try to compensate this.
+ */
+
+#define MIX_ENT(name, reg_l, pola_l, pos_l, len_l, reg_r, pola_r, pos_r, len_r, mute_bit) \
+ [name] = {{reg_l, pola_l, pos_l, len_l, reg_l, 0, mute_bit, 0, 0, 8}, \
+ {reg_r, pola_r, pos_r, len_r, reg_r, 0, mute_bit, 0, 0, 8}}
+
+#define MIX_ENT2(name, reg_l, pola_l, pos_l, len_l, mute_reg_l, mute_pola_l, mute_pos_l, \
+ rec_reg_l, rec_pola_l, rec_pos_l, \
+ reg_r, pola_r, pos_r, len_r, mute_reg_r, mute_pola_r, mute_pos_r, \
+ rec_reg_r, rec_pola_r, rec_pos_r) \
+ [name] = {{reg_l, pola_l, pos_l, len_l, mute_reg_l, mute_pola_l, mute_pos_l, \
+ rec_reg_l, rec_pola_l, rec_pos_l}, \
+ {reg_r, pola_r, pos_r, len_r, mute_reg_r, mute_pola_r, mute_pos_r, \
+ rec_reg_r, rec_pola_r, rec_pos_r}}
+
+static mixer_ents ad1848_mix_devices[32] = {
+ MIX_ENT(SOUND_MIXER_VOLUME, 27, 1, 0, 4, 29, 1, 0, 4, 8),
+ MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7),
+ MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_MIC, 0, 0, 5, 1, 1, 0, 5, 1, 8),
+ MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_IMIX, 13, 1, 2, 6, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
+ MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 0, 5, 19, 1, 0, 5, 7)
+};
+
+static mixer_ents iwave_mix_devices[32] = {
+ MIX_ENT(SOUND_MIXER_VOLUME, 25, 1, 0, 5, 27, 1, 0, 5, 8),
+ MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7),
+ MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_MIC, 0, 0, 5, 1, 1, 0, 5, 1, 8),
+ MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_IMIX, 16, 1, 0, 5, 17, 1, 0, 5, 8),
+ MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
+ MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 0, 5, 19, 1, 0, 5, 7)
+};
+
+static mixer_ents cs42xb_mix_devices[32] = {
+ /* Digital master volume actually has seven bits, but we only use
+ six to avoid the discontinuity when the analog gain kicks in. */
+ MIX_ENT(SOUND_MIXER_VOLUME, 46, 1, 0, 6, 47, 1, 0, 6, 7),
+ MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7),
+ MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_MIC, 34, 1, 0, 5, 35, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7),
+ /* For the IMIX entry, it was not possible to use the MIX_ENT macro
+ because the mute bit is in different positions for the two
+ channels and requires reverse polarity. */
+ [SOUND_MIXER_IMIX] = {{13, 1, 2, 6, 13, 1, 0, 0, 0, 8},
+ {42, 1, 0, 6, 42, 1, 7, 0, 0, 8}},
+ MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
+ MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_LINE3, 38, 1, 0, 6, 39, 1, 0, 6, 7)
+};
+
+/* OPTi 82C930 has somewhat different port addresses.
+ * Note: VOLUME == SPEAKER, SYNTH == LINE2, LINE == LINE3, CD == LINE1
+ * VOLUME, SYNTH, LINE, CD are not enabled above.
+ * MIC is level of mic monitoring direct to output. Same for CD, LINE, etc.
+ */
+static mixer_ents c930_mix_devices[32] = {
+ MIX_ENT(SOUND_MIXER_VOLUME, 22, 1, 1, 5, 23, 1, 1, 5, 7),
+ MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 1, 4, 5, 1, 1, 4, 7),
+ MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 5, 7, 1, 0, 5, 7),
+ MIX_ENT(SOUND_MIXER_SPEAKER, 22, 1, 1, 5, 23, 1, 1, 5, 7),
+ MIX_ENT(SOUND_MIXER_LINE, 18, 1, 1, 4, 19, 1, 1, 4, 7),
+ MIX_ENT(SOUND_MIXER_MIC, 20, 1, 1, 4, 21, 1, 1, 4, 7),
+ MIX_ENT(SOUND_MIXER_CD, 2, 1, 1, 4, 3, 1, 1, 4, 7),
+ MIX_ENT(SOUND_MIXER_IMIX, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
+ MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 1, 4, 3, 1, 1, 4, 7),
+ MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 1, 4, 5, 1, 1, 4, 7),
+ MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 1, 4, 19, 1, 1, 4, 7)
+};
+
+static mixer_ents spro_mix_devices[32] = {
+ MIX_ENT (SOUND_MIXER_VOLUME, 19, 0, 4, 4, 19, 0, 0, 4, 8),
+ MIX_ENT (SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT (SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT2(SOUND_MIXER_SYNTH, 4, 1, 1, 4, 23, 0, 3, 0, 0, 8,
+ 5, 1, 1, 4, 23, 0, 3, 0, 0, 8),
+ MIX_ENT (SOUND_MIXER_PCM, 6, 1, 1, 4, 7, 1, 1, 4, 8),
+ MIX_ENT (SOUND_MIXER_SPEAKER, 18, 0, 3, 2, 0, 0, 0, 0, 8),
+ MIX_ENT2(SOUND_MIXER_LINE, 20, 0, 4, 4, 17, 1, 4, 16, 0, 2,
+ 20, 0, 0, 4, 17, 1, 3, 16, 0, 1),
+ MIX_ENT2(SOUND_MIXER_MIC, 18, 0, 0, 3, 17, 1, 0, 16, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0),
+ MIX_ENT2(SOUND_MIXER_CD, 21, 0, 4, 4, 17, 1, 2, 16, 0, 4,
+ 21, 0, 0, 4, 17, 1, 1, 16, 0, 3),
+ MIX_ENT (SOUND_MIXER_IMIX, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT (SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT (SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT (SOUND_MIXER_IGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
+ MIX_ENT (SOUND_MIXER_OGAIN, 17, 1, 6, 1, 0, 0, 0, 0, 8),
+ /* This is external wavetable */
+ MIX_ENT2(SOUND_MIXER_LINE1, 22, 0, 4, 4, 23, 1, 1, 23, 0, 4,
+ 22, 0, 0, 4, 23, 1, 0, 23, 0, 5),
+};
+
+static int default_mixer_levels[32] =
+{
+ 0x3232, /* Master Volume */
+ 0x3232, /* Bass */
+ 0x3232, /* Treble */
+ 0x4b4b, /* FM */
+ 0x3232, /* PCM */
+ 0x1515, /* PC Speaker */
+ 0x2020, /* Ext Line */
+ 0x1010, /* Mic */
+ 0x4b4b, /* CD */
+ 0x0000, /* Recording monitor */
+ 0x4b4b, /* Second PCM */
+ 0x4b4b, /* Recording level */
+ 0x4b4b, /* Input gain */
+ 0x4b4b, /* Output gain */
+ 0x2020, /* Line1 */
+ 0x2020, /* Line2 */
+ 0x1515 /* Line3 (usually line in)*/
+};
+
+#define LEFT_CHN 0
+#define RIGHT_CHN 1
+
+/*
+ * Channel enable bits for ioctl(SOUND_MIXER_PRIVATE1)
+ */
+
+#ifndef AUDIO_SPEAKER
+#define AUDIO_SPEAKER 0x01 /* Enable mono output */
+#define AUDIO_HEADPHONE 0x02 /* Sparc only */
+#define AUDIO_LINE_OUT 0x04 /* Sparc only */
+#endif
diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c
new file mode 100644
index 0000000..a0274f3
--- /dev/null
+++ b/sound/oss/aedsp16.c
@@ -0,0 +1,1372 @@
+/*
+ sound/oss/aedsp16.c
+
+ Audio Excel DSP 16 software configuration routines
+ Copyright (C) 1995,1996,1997,1998 Riccardo Facchetti (fizban@tin.it)
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+ */
+/*
+ * Include the main OSS Lite header file. It include all the os, OSS Lite, etc
+ * headers needed by this source.
+ */
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/init.h>
+#include "sound_config.h"
+
+/*
+
+ READ THIS
+
+ This module started to configure the Audio Excel DSP 16 Sound Card.
+ Now works with the SC-6000 (old aedsp16) and new SC-6600 based cards.
+
+ NOTE: I have NO idea about Audio Excel DSP 16 III. If someone owns this
+ audio card and want to see the kernel support for it, please contact me.
+
+ Audio Excel DSP 16 is an SB pro II, Microsoft Sound System and MPU-401
+ compatible card.
+ It is software-only configurable (no jumpers to hard-set irq/dma/mpu-irq),
+ so before this module, the only way to configure the DSP under linux was
+ boot the MS-DOS loading the sound.sys device driver (this driver soft-
+ configure the sound board hardware by massaging someone of its registers),
+ and then ctrl-alt-del to boot linux with the DSP configured by the DOS
+ driver.
+
+ This module works configuring your Audio Excel DSP 16's irq, dma and
+ mpu-401-irq. The OSS Lite routines rely on the fact that if the
+ hardware is there, they can detect it. The problem with AEDSP16 is
+ that no hardware can be found by the probe routines if the sound card
+ is not configured properly. Sometimes the kernel probe routines can find
+ an SBPRO even when the card is not configured (this is the standard setup
+ of the card), but the SBPRO emulation don't work well if the card is not
+ properly initialized. For this reason
+
+ aedsp16_init_board()
+
+ routine is called before the OSS Lite probe routines try to detect the
+ hardware.
+
+ NOTE (READ THE NOTE TOO, IT CONTAIN USEFUL INFORMATIONS)
+
+ NOTE: Now it works with SC-6000 and SC-6600 based audio cards. The new cards
+ have no jumper switch at all. No more WSS or MPU-401 I/O port switches. They
+ have to be configured by software.
+
+ NOTE: The driver is merged with the new OSS Lite sound driver. It works
+ as a lowlevel driver.
+
+ The Audio Excel DSP 16 Sound Card emulates both SBPRO and MSS;
+ the OSS Lite sound driver can be configured for SBPRO and MSS cards
+ at the same time, but the aedsp16 can't be two cards!!
+ When we configure it, we have to choose the SBPRO or the MSS emulation
+ for AEDSP16. We also can install a *REAL* card of the other type (see [1]).
+
+ NOTE: If someone can test the combination AEDSP16+MSS or AEDSP16+SBPRO
+ please let me know if it works.
+
+ The MPU-401 support can be compiled in together with one of the other
+ two operating modes.
+
+ NOTE: This is something like plug-and-play: we have only to plug
+ the AEDSP16 board in the socket, and then configure and compile
+ a kernel that uses the AEDSP16 software configuration capability.
+ No jumper setting is needed!
+
+ For example, if you want AEDSP16 to be an SBPro, on irq 10, dma 3
+ you have just to make config the OSS Lite package, configuring
+ the AEDSP16 sound card, then activating the SBPro emulation mode
+ and at last configuring IRQ and DMA.
+ Compile the kernel and run it.
+
+ NOTE: This means for SC-6000 cards that you can choose irq and dma,
+ but not the I/O addresses. To change I/O addresses you have to set
+ them with jumpers. For SC-6600 cards you have no jumpers so you have
+ to set up your full card configuration in the make config.
+
+ You can change the irq/dma/mirq settings WITHOUT THE NEED to open
+ your computer and massage the jumpers (there are no irq/dma/mirq
+ jumpers to be configured anyway, only I/O BASE values have to be
+ configured with jumpers)
+
+ For some ununderstandable reason, the card default of irq 7, dma 1,
+ don't work for me. Seems to be an IRQ or DMA conflict. Under heavy
+ HDD work, the kernel start to erupt out a lot of messages like:
+
+ 'Sound: DMA timed out - IRQ/DRQ config error?'
+
+ For what I can say, I have NOT any conflict at irq 7 (under linux I'm
+ using the lp polling driver), and dma line 1 is unused as stated by
+ /proc/dma. I can suppose this is a bug of AEDSP16. I know my hardware so
+ I'm pretty sure I have not any conflict, but may be I'm wrong. Who knows!
+ Anyway a setting of irq 10, dma 3 works really fine.
+
+ NOTE: if someone can use AEDSP16 with irq 7, dma 1, please let me know
+ the emulation mode, all the installed hardware and the hardware
+ configuration (irq and dma settings of all the hardware).
+
+ This init module should work with SBPRO+MSS, when one of the two is
+ the AEDSP16 emulation and the other the real card. (see [1])
+ For example:
+
+ AEDSP16 (0x220) in SBPRO emu (0x220) + real MSS + other
+ AEDSP16 (0x220) in MSS emu + real SBPRO (0x240) + other
+
+ MPU401 should work. (see [2])
+
+ [1]
+ ---
+ Date: Mon, 29 Jul 1997 08:35:40 +0100
+ From: Mr S J Greenaway <sjg95@unixfe.rl.ac.uk>
+
+ [...]
+ Just to let you know got my Audio Excel (emulating a MSS) working
+ with my original SB16, thanks for the driver!
+ [...]
+ ---
+
+ [2] Not tested by me for lack of hardware.
+
+ TODO, WISHES AND TECH
+
+ - About I/O ports allocation -
+
+ Request the 2x0h region (port base) in any case if we are using this card.
+
+ NOTE: the "aedsp16 (base)" string with which we are requesting the aedsp16
+ port base region (see code) does not mean necessarily that we are emulating
+ sbpro. Even if this region is the sbpro I/O ports region, we use this
+ region to access the control registers of the card, and if emulating
+ sbpro, I/O sbpro registers too. If we are emulating MSS, the sbpro
+ registers are not used, in no way, to emulate an sbpro: they are
+ used only for configuration purposes.
+
+ Started Fri Mar 17 16:13:18 MET 1995
+
+ v0.1 (ALPHA, was an user-level program called AudioExcelDSP16.c)
+ - Initial code.
+ v0.2 (ALPHA)
+ - Cleanups.
+ - Integrated with Linux voxware v 2.90-2 kernel sound driver.
+ - SoundBlaster Pro mode configuration.
+ - Microsoft Sound System mode configuration.
+ - MPU-401 mode configuration.
+ v0.3 (ALPHA)
+ - Cleanups.
+ - Rearranged the code to let aedsp16_init_board be more general.
+ - Erased the REALLY_SLOW_IO. We don't need it. Erased the linux/io.h
+ inclusion too. We rely on os.h
+ - Used the to get a variable
+ len string (we are not sure about the len of Copyright string).
+ This works with any SB and compatible.
+ - Added the code to request_region at device init (should go in
+ the main body of voxware).
+ v0.4 (BETA)
+ - Better configure.c patch for aedsp16 configuration (better
+ logic of inclusion of AEDSP16 support)
+ - Modified the conditional compilation to better support more than
+ one sound card of the emulated type (read the NOTES above)
+ - Moved the sb init routine from the attach to the very first
+ probe in sb_card.c
+ - Rearrangements and cleanups
+ - Wiped out some unnecessary code and variables: this is kernel
+ code so it is better save some TEXT and DATA
+ - Fixed the request_region code. We must allocate the aedsp16 (sbpro)
+ I/O ports in any case because they are used to access the DSP
+ configuration registers and we can not allow anyone to get them.
+ v0.5
+ - cleanups on comments
+ - prep for diffs against v3.0-proto-950402
+ v0.6
+ - removed the request_region()s when compiling the MODULE sound.o
+ because we are not allowed (by the actual voxware structure) to
+ release_region()
+ v0.7 (pre ALPHA, not distributed)
+ - started porting this module to kernel 1.3.84. Dummy probe/attach
+ routines.
+ v0.8 (ALPHA)
+ - attached all the init routines.
+ v0.9 (BETA)
+ - Integrated with linux-pre2.0.7
+ - Integrated with configuration scripts.
+ - Cleaned up and beautyfied the code.
+ v0.9.9 (BETA)
+ - Thanks to Piercarlo Grandi: corrected the conditonal compilation code.
+ Now only the code configured is compiled in, with some memory saving.
+ v0.9.10
+ - Integration into the sound/lowlevel/ section of the sound driver.
+ - Re-organized the code.
+ v0.9.11 (not distributed)
+ - Rewritten the init interface-routines to initialize the AEDSP16 in
+ one shot.
+ - More cosmetics.
+ - SC-6600 support.
+ - More soft/hard configuration.
+ v0.9.12
+ - Refined the v0.9.11 code with conditional compilation to distinguish
+ between SC-6000 and SC-6600 code.
+ v1.0.0
+ - Prep for merging with OSS Lite and Linux kernel 2.1.13
+ - Corrected a bug in request/check/release region calls (thanks to the
+ new kernel exception handling).
+ v1.1
+ - Revamped for integration with new modularized sound drivers: to enhance
+ the flexibility of modular version, I have removed all the conditional
+ compilation for SBPRO, MPU and MSS code. Now it is all managed with
+ the ae_config structure.
+ v1.2
+ - Module informations added.
+ - Removed aedsp16_delay_10msec(), now using mdelay(10)
+ - All data and funcs moved to .*.init section.
+ v1.3
+ Arnaldo Carvalho de Melo <acme@conectiva.com.br> - 2000/09/27
+ - got rid of check_region
+
+ Known Problems:
+ - Audio Excel DSP 16 III don't work with this driver.
+
+ Credits:
+ Many thanks to Gerald Britton <gbritton@CapAccess.org>. He helped me a
+ lot in testing the 0.9.11 and 0.9.12 versions of this driver.
+
+ */
+
+
+#define VERSION "1.3" /* Version of Audio Excel DSP 16 driver */
+
+#undef AEDSP16_DEBUG /* Define this to 1 to enable debug code */
+#undef AEDSP16_DEBUG_MORE /* Define this to 1 to enable more debug */
+#undef AEDSP16_INFO /* Define this to 1 to enable info code */
+
+#if defined(AEDSP16_DEBUG)
+# define DBG(x) printk x
+# if defined(AEDSP16_DEBUG_MORE)
+# define DBG1(x) printk x
+# else
+# define DBG1(x)
+# endif
+#else
+# define DBG(x)
+# define DBG1(x)
+#endif
+
+/*
+ * Misc definitions
+ */
+#define TRUE 1
+#define FALSE 0
+
+/*
+ * Region Size for request/check/release region.
+ */
+#define IOBASE_REGION_SIZE 0x10
+
+/*
+ * Hardware related defaults
+ */
+#define DEF_AEDSP16_IOB 0x220 /* 0x220(default) 0x240 */
+#define DEF_AEDSP16_IRQ 7 /* 5 7(default) 9 10 11 */
+#define DEF_AEDSP16_MRQ 0 /* 5 7 9 10 0(default), 0 means disable */
+#define DEF_AEDSP16_DMA 1 /* 0 1(default) 3 */
+
+/*
+ * Commands of AEDSP16's DSP (SBPRO+special).
+ * Some of them are COMMAND_xx, in the future they may change.
+ */
+#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */
+#define COMMAND_52 0x52 /* */
+#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */
+#define COMMAND_5C 0x5c /* */
+#define COMMAND_60 0x60 /* */
+#define COMMAND_66 0x66 /* */
+#define COMMAND_6C 0x6c /* */
+#define COMMAND_6E 0x6e /* */
+#define COMMAND_88 0x88 /* */
+#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */
+#define COMMAND_C5 0xc5 /* */
+#define GET_DSP_VERSION 0xe1 /* Get DSP Version */
+#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */
+
+/*
+ * Offsets of AEDSP16 DSP I/O ports. The offset is added to base I/O port
+ * to have the actual I/O port.
+ * Register permissions are:
+ * (wo) == Write Only
+ * (ro) == Read Only
+ * (w-) == Write
+ * (r-) == Read
+ */
+#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */
+#define DSP_READ 0x0a /* offset of DSP READ (ro) */
+#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */
+#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */
+#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */
+#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */
+
+
+#define RETRY 10 /* Various retry values on I/O opera- */
+#define STATUSRETRY 1000 /* tions. Sometimes we have to */
+#define HARDRETRY 500000 /* wait for previous cmd to complete */
+
+/*
+ * Size of character arrays that store name and version of sound card
+ */
+#define CARDNAMELEN 15 /* Size of the card's name in chars */
+#define CARDVERLEN 2 /* Size of the card's version in chars */
+
+#if defined(CONFIG_SC6600)
+/*
+ * Bitmapped flags of hard configuration
+ */
+/*
+ * Decode macros (xl == low byte, xh = high byte)
+ */
+#define IOBASE(xl) ((xl & 0x01)?0x240:0x220)
+#define JOY(xl) (xl & 0x02)
+#define MPUADDR(xl) ( \
+ (xl & 0x0C)?0x330: \
+ (xl & 0x08)?0x320: \
+ (xl & 0x04)?0x310: \
+ 0x300)
+#define WSSADDR(xl) ((xl & 0x10)?0xE80:0x530)
+#define CDROM(xh) (xh & 0x20)
+#define CDROMADDR(xh) (((xh & 0x1F) << 4) + 0x200)
+/*
+ * Encode macros
+ */
+#define BLDIOBASE(xl, val) { \
+ xl &= ~0x01; \
+ if (val == 0x240) \
+ xl |= 0x01; \
+ }
+#define BLDJOY(xl, val) { \
+ xl &= ~0x02; \
+ if (val == 1) \
+ xl |= 0x02; \
+ }
+#define BLDMPUADDR(xl, val) { \
+ xl &= ~0x0C; \
+ switch (val) { \
+ case 0x330: \
+ xl |= 0x0C; \
+ break; \
+ case 0x320: \
+ xl |= 0x08; \
+ break; \
+ case 0x310: \
+ xl |= 0x04; \
+ break; \
+ case 0x300: \
+ xl |= 0x00; \
+ break; \
+ default: \
+ xl |= 0x00; \
+ break; \
+ } \
+ }
+#define BLDWSSADDR(xl, val) { \
+ xl &= ~0x10; \
+ if (val == 0xE80) \
+ xl |= 0x10; \
+ }
+#define BLDCDROM(xh, val) { \
+ xh &= ~0x20; \
+ if (val == 1) \
+ xh |= 0x20; \
+ }
+#define BLDCDROMADDR(xh, val) { \
+ int tmp = val; \
+ tmp -= 0x200; \
+ tmp >>= 4; \
+ tmp &= 0x1F; \
+ xh |= tmp; \
+ xh &= 0x7F; \
+ xh |= 0x40; \
+ }
+#endif /* CONFIG_SC6600 */
+
+/*
+ * Bit mapped flags for calling aedsp16_init_board(), and saving the current
+ * emulation mode.
+ */
+#define INIT_NONE (0 )
+#define INIT_SBPRO (1<<0)
+#define INIT_MSS (1<<1)
+#define INIT_MPU401 (1<<2)
+
+static int soft_cfg __initdata = 0; /* bitmapped config */
+static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */
+static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver:
+ hi->ver[0] lo->ver[1] */
+
+#if defined(CONFIG_SC6600)
+static int hard_cfg[2] /* lo<-hard_cfg[0] hi<-hard_cfg[1] */
+ __initdata = { 0, 0};
+#endif /* CONFIG_SC6600 */
+
+#if defined(CONFIG_SC6600)
+/* Decoded hard configuration */
+struct d_hcfg {
+ int iobase;
+ int joystick;
+ int mpubase;
+ int wssbase;
+ int cdrom;
+ int cdrombase;
+};
+
+static struct d_hcfg decoded_hcfg __initdata = {0, };
+
+#endif /* CONFIG_SC6600 */
+
+/* orVals contain the values to be or'ed */
+struct orVals {
+ int val; /* irq|mirq|dma */
+ int or; /* soft_cfg |= TheStruct.or */
+};
+
+/* aedsp16_info contain the audio card configuration */
+struct aedsp16_info {
+ int base_io; /* base I/O address for accessing card */
+ int irq; /* irq value for DSP I/O */
+ int mpu_irq; /* irq for mpu401 interface I/O */
+ int dma; /* dma value for DSP I/O */
+ int mss_base; /* base I/O for Microsoft Sound System */
+ int mpu_base; /* base I/O for MPU-401 emulation */
+ int init; /* Initialization status of the card */
+};
+
+/*
+ * Magic values that the DSP will eat when configuring irq/mirq/dma
+ */
+/* DSP IRQ conversion array */
+static struct orVals orIRQ[] __initdata = {
+ {0x05, 0x28},
+ {0x07, 0x08},
+ {0x09, 0x10},
+ {0x0a, 0x18},
+ {0x0b, 0x20},
+ {0x00, 0x00}
+};
+
+/* MPU-401 IRQ conversion array */
+static struct orVals orMIRQ[] __initdata = {
+ {0x05, 0x04},
+ {0x07, 0x44},
+ {0x09, 0x84},
+ {0x0a, 0xc4},
+ {0x00, 0x00}
+};
+
+/* DMA Channels conversion array */
+static struct orVals orDMA[] __initdata = {
+ {0x00, 0x01},
+ {0x01, 0x02},
+ {0x03, 0x03},
+ {0x00, 0x00}
+};
+
+static struct aedsp16_info ae_config = {
+ DEF_AEDSP16_IOB,
+ DEF_AEDSP16_IRQ,
+ DEF_AEDSP16_MRQ,
+ DEF_AEDSP16_DMA,
+ -1,
+ -1,
+ INIT_NONE
+};
+
+/*
+ * Buffers to store audio card informations
+ */
+static char DSPCopyright[CARDNAMELEN + 1] __initdata = {0, };
+static char DSPVersion[CARDVERLEN + 1] __initdata = {0, };
+
+static int __init aedsp16_wait_data(int port)
+{
+ int loop = STATUSRETRY;
+ unsigned char ret = 0;
+
+ DBG1(("aedsp16_wait_data (0x%x): ", port));
+
+ do {
+ ret = inb(port + DSP_DATAVAIL);
+ /*
+ * Wait for data available (bit 7 of ret == 1)
+ */
+ } while (!(ret & 0x80) && loop--);
+
+ if (ret & 0x80) {
+ DBG1(("success.\n"));
+ return TRUE;
+ }
+
+ DBG1(("failure.\n"));
+ return FALSE;
+}
+
+static int __init aedsp16_read(int port)
+{
+ int inbyte;
+
+ DBG((" Read DSP Byte (0x%x): ", port));
+
+ if (aedsp16_wait_data(port) == FALSE) {
+ DBG(("failure.\n"));
+ return -1;
+ }
+
+ inbyte = inb(port + DSP_READ);
+
+ DBG(("read [0x%x]/{%c}.\n", inbyte, inbyte));
+
+ return inbyte;
+}
+
+static int __init aedsp16_test_dsp(int port)
+{
+ return ((aedsp16_read(port) == 0xaa) ? TRUE : FALSE);
+}
+
+static int __init aedsp16_dsp_reset(int port)
+{
+ /*
+ * Reset DSP
+ */
+
+ DBG(("Reset DSP:\n"));
+
+ outb(1, (port + DSP_RESET));
+ udelay(10);
+ outb(0, (port + DSP_RESET));
+ udelay(10);
+ udelay(10);
+ if (aedsp16_test_dsp(port) == TRUE) {
+ DBG(("success.\n"));
+ return TRUE;
+ } else
+ DBG(("failure.\n"));
+ return FALSE;
+}
+
+static int __init aedsp16_write(int port, int cmd)
+{
+ unsigned char ret;
+ int loop = HARDRETRY;
+
+ DBG((" Write DSP Byte (0x%x) [0x%x]: ", port, cmd));
+
+ do {
+ ret = inb(port + DSP_STATUS);
+ /*
+ * DSP ready to receive data if bit 7 of ret == 0
+ */
+ if (!(ret & 0x80)) {
+ outb(cmd, port + DSP_COMMAND);
+ DBG(("success.\n"));
+ return 0;
+ }
+ } while (loop--);
+
+ DBG(("timeout.\n"));
+ printk("[AEDSP16] DSP Command (0x%x) timeout.\n", cmd);
+
+ return -1;
+}
+
+#if defined(CONFIG_SC6600)
+
+#if defined(AEDSP16_INFO) || defined(AEDSP16_DEBUG)
+void __init aedsp16_pinfo(void) {
+ DBG(("\n Base address: %x\n", decoded_hcfg.iobase));
+ DBG((" Joystick : %s present\n", decoded_hcfg.joystick?"":" not"));
+ DBG((" WSS addr : %x\n", decoded_hcfg.wssbase));
+ DBG((" MPU-401 addr: %x\n", decoded_hcfg.mpubase));
+ DBG((" CDROM : %s present\n", (decoded_hcfg.cdrom!=4)?"":" not"));
+ DBG((" CDROMADDR : %x\n\n", decoded_hcfg.cdrombase));
+}
+#endif
+
+static void __init aedsp16_hard_decode(void) {
+
+ DBG((" aedsp16_hard_decode: 0x%x, 0x%x\n", hard_cfg[0], hard_cfg[1]));
+
+/*
+ * Decode Cfg Bytes.
+ */
+ decoded_hcfg.iobase = IOBASE(hard_cfg[0]);
+ decoded_hcfg.joystick = JOY(hard_cfg[0]);
+ decoded_hcfg.wssbase = WSSADDR(hard_cfg[0]);
+ decoded_hcfg.mpubase = MPUADDR(hard_cfg[0]);
+ decoded_hcfg.cdrom = CDROM(hard_cfg[1]);
+ decoded_hcfg.cdrombase = CDROMADDR(hard_cfg[1]);
+
+#if defined(AEDSP16_INFO) || defined(AEDSP16_DEBUG)
+ printk(" Original sound card configuration:\n");
+ aedsp16_pinfo();
+#endif
+
+/*
+ * Now set up the real kernel configuration.
+ */
+ decoded_hcfg.iobase = ae_config.base_io;
+ decoded_hcfg.wssbase = ae_config.mss_base;
+ decoded_hcfg.mpubase = ae_config.mpu_base;
+
+#if defined(CONFIG_SC6600_JOY)
+ decoded_hcfg.joystick = CONFIG_SC6600_JOY; /* Enable */
+#endif
+#if defined(CONFIG_SC6600_CDROM)
+ decoded_hcfg.cdrom = CONFIG_SC6600_CDROM; /* 4:N-3:I-2:G-1:P-0:S */
+#endif
+#if defined(CONFIG_SC6600_CDROMBASE)
+ decoded_hcfg.cdrombase = CONFIG_SC6600_CDROMBASE; /* 0 Disable */
+#endif
+
+#if defined(AEDSP16_DEBUG)
+ DBG((" New Values:\n"));
+ aedsp16_pinfo();
+#endif
+
+ DBG(("success.\n"));
+}
+
+static void __init aedsp16_hard_encode(void) {
+
+ DBG((" aedsp16_hard_encode: 0x%x, 0x%x\n", hard_cfg[0], hard_cfg[1]));
+
+ hard_cfg[0] = 0;
+ hard_cfg[1] = 0;
+
+ hard_cfg[0] |= 0x20;
+
+ BLDIOBASE (hard_cfg[0], decoded_hcfg.iobase);
+ BLDWSSADDR(hard_cfg[0], decoded_hcfg.wssbase);
+ BLDMPUADDR(hard_cfg[0], decoded_hcfg.mpubase);
+ BLDJOY(hard_cfg[0], decoded_hcfg.joystick);
+ BLDCDROM(hard_cfg[1], decoded_hcfg.cdrom);
+ BLDCDROMADDR(hard_cfg[1], decoded_hcfg.cdrombase);
+
+#if defined(AEDSP16_DEBUG)
+ aedsp16_pinfo();
+#endif
+
+ DBG((" aedsp16_hard_encode: 0x%x, 0x%x\n", hard_cfg[0], hard_cfg[1]));
+ DBG(("success.\n"));
+
+}
+
+static int __init aedsp16_hard_write(int port) {
+
+ DBG(("aedsp16_hard_write:\n"));
+
+ if (aedsp16_write(port, COMMAND_6C)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_6C);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+ if (aedsp16_write(port, COMMAND_5C)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_5C);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+ if (aedsp16_write(port, hard_cfg[0])) {
+ printk("[AEDSP16] DATA 0x%x: failed!\n", hard_cfg[0]);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+ if (aedsp16_write(port, hard_cfg[1])) {
+ printk("[AEDSP16] DATA 0x%x: failed!\n", hard_cfg[1]);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+ if (aedsp16_write(port, COMMAND_C5)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_C5);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+
+ DBG(("success.\n"));
+
+ return TRUE;
+}
+
+static int __init aedsp16_hard_read(int port) {
+
+ DBG(("aedsp16_hard_read:\n"));
+
+ if (aedsp16_write(port, READ_HARD_CFG)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", READ_HARD_CFG);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+
+ if ((hard_cfg[0] = aedsp16_read(port)) == -1) {
+ printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n",
+ READ_HARD_CFG);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+ if ((hard_cfg[1] = aedsp16_read(port)) == -1) {
+ printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n",
+ READ_HARD_CFG);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+ if (aedsp16_read(port) == -1) {
+ printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n",
+ READ_HARD_CFG);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+
+ DBG(("success.\n"));
+
+ return TRUE;
+}
+
+static int __init aedsp16_ext_cfg_write(int port) {
+
+ int extcfg, val;
+
+ if (aedsp16_write(port, COMMAND_66)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_66);
+ return FALSE;
+ }
+
+ extcfg = 7;
+ if (decoded_hcfg.cdrom != 2)
+ extcfg = 0x0F;
+ if ((decoded_hcfg.cdrom == 4) ||
+ (decoded_hcfg.cdrom == 3))
+ extcfg &= ~2;
+ if (decoded_hcfg.cdrombase == 0)
+ extcfg &= ~2;
+ if (decoded_hcfg.mpubase == 0)
+ extcfg &= ~1;
+
+ if (aedsp16_write(port, extcfg)) {
+ printk("[AEDSP16] Write extcfg: failed!\n");
+ return FALSE;
+ }
+ if (aedsp16_write(port, 0)) {
+ printk("[AEDSP16] Write extcfg: failed!\n");
+ return FALSE;
+ }
+ if (decoded_hcfg.cdrom == 3) {
+ if (aedsp16_write(port, COMMAND_52)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_52);
+ return FALSE;
+ }
+ if ((val = aedsp16_read(port)) == -1) {
+ printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n"
+ , COMMAND_52);
+ return FALSE;
+ }
+ val &= 0x7F;
+ if (aedsp16_write(port, COMMAND_60)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_60);
+ return FALSE;
+ }
+ if (aedsp16_write(port, val)) {
+ printk("[AEDSP16] Write val: failed!\n");
+ return FALSE;
+ }
+ }
+
+ return TRUE;
+}
+
+#endif /* CONFIG_SC6600 */
+
+static int __init aedsp16_cfg_write(int port) {
+ if (aedsp16_write(port, WRITE_MDIRQ_CFG)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG);
+ return FALSE;
+ }
+ if (aedsp16_write(port, soft_cfg)) {
+ printk("[AEDSP16] Initialization of (M)IRQ and DMA: failed!\n");
+ return FALSE;
+ }
+ return TRUE;
+}
+
+static int __init aedsp16_init_mss(int port)
+{
+ DBG(("aedsp16_init_mss:\n"));
+
+ mdelay(10);
+
+ if (aedsp16_write(port, DSP_INIT_MSS)) {
+ printk("[AEDSP16] aedsp16_init_mss [0x%x]: failed!\n",
+ DSP_INIT_MSS);
+ DBG(("failure.\n"));
+ return FALSE;
+ }
+
+ mdelay(10);
+
+ if (aedsp16_cfg_write(port) == FALSE)
+ return FALSE;
+
+ outb(soft_cfg_mss, ae_config.mss_base);
+
+ DBG(("success.\n"));
+
+ return TRUE;
+}
+
+static int __init aedsp16_setup_board(int port) {
+ int loop = RETRY;
+
+#if defined(CONFIG_SC6600)
+ int val = 0;
+
+ if (aedsp16_hard_read(port) == FALSE) {
+ printk("[AEDSP16] aedsp16_hard_read: failed!\n");
+ return FALSE;
+ }
+
+ if (aedsp16_write(port, COMMAND_52)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_52);
+ return FALSE;
+ }
+
+ if ((val = aedsp16_read(port)) == -1) {
+ printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n",
+ COMMAND_52);
+ return FALSE;
+ }
+#endif
+
+ do {
+ if (aedsp16_write(port, COMMAND_88)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_88);
+ return FALSE;
+ }
+ mdelay(10);
+ } while ((aedsp16_wait_data(port) == FALSE) && loop--);
+
+ if (aedsp16_read(port) == -1) {
+ printk("[AEDSP16] aedsp16_read after CMD 0x%x: failed\n",
+ COMMAND_88);
+ return FALSE;
+ }
+
+#if !defined(CONFIG_SC6600)
+ if (aedsp16_write(port, COMMAND_5C)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_5C);
+ return FALSE;
+ }
+#endif
+
+ if (aedsp16_cfg_write(port) == FALSE)
+ return FALSE;
+
+#if defined(CONFIG_SC6600)
+ if (aedsp16_write(port, COMMAND_60)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_60);
+ return FALSE;
+ }
+ if (aedsp16_write(port, val)) {
+ printk("[AEDSP16] DATA 0x%x: failed!\n", val);
+ return FALSE;
+ }
+ if (aedsp16_write(port, COMMAND_6E)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_6E);
+ return FALSE;
+ }
+ if (aedsp16_write(port, ver[0])) {
+ printk("[AEDSP16] DATA 0x%x: failed!\n", ver[0]);
+ return FALSE;
+ }
+ if (aedsp16_write(port, ver[1])) {
+ printk("[AEDSP16] DATA 0x%x: failed!\n", ver[1]);
+ return FALSE;
+ }
+
+ if (aedsp16_hard_write(port) == FALSE) {
+ printk("[AEDSP16] aedsp16_hard_write: failed!\n");
+ return FALSE;
+ }
+
+ if (aedsp16_write(port, COMMAND_5C)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", COMMAND_5C);
+ return FALSE;
+ }
+
+#if defined(THIS_IS_A_THING_I_HAVE_NOT_TESTED_YET)
+ if (aedsp16_cfg_write(port) == FALSE)
+ return FALSE;
+#endif
+
+#endif
+
+ return TRUE;
+}
+
+static int __init aedsp16_stdcfg(int port) {
+ if (aedsp16_write(port, WRITE_MDIRQ_CFG)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG);
+ return FALSE;
+ }
+ /*
+ * 0x0A == (IRQ 7, DMA 1, MIRQ 0)
+ */
+ if (aedsp16_write(port, 0x0A)) {
+ printk("[AEDSP16] aedsp16_stdcfg: failed!\n");
+ return FALSE;
+ }
+ return TRUE;
+}
+
+static int __init aedsp16_dsp_version(int port)
+{
+ int len = 0;
+ int ret;
+
+ DBG(("Get DSP Version:\n"));
+
+ if (aedsp16_write(ae_config.base_io, GET_DSP_VERSION)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", GET_DSP_VERSION);
+ DBG(("failed.\n"));
+ return FALSE;
+ }
+
+ do {
+ if ((ret = aedsp16_read(port)) == -1) {
+ DBG(("failed.\n"));
+ return FALSE;
+ }
+ /*
+ * We already know how many int are stored (2), so we know when the
+ * string is finished.
+ */
+ ver[len++] = ret;
+ } while (len < CARDVERLEN);
+ sprintf(DSPVersion, "%d.%d", ver[0], ver[1]);
+
+ DBG(("success.\n"));
+
+ return TRUE;
+}
+
+static int __init aedsp16_dsp_copyright(int port)
+{
+ int len = 0;
+ int ret;
+
+ DBG(("Get DSP Copyright:\n"));
+
+ if (aedsp16_write(ae_config.base_io, GET_DSP_COPYRIGHT)) {
+ printk("[AEDSP16] CMD 0x%x: failed!\n", GET_DSP_COPYRIGHT);
+ DBG(("failed.\n"));
+ return FALSE;
+ }
+
+ do {
+ if ((ret = aedsp16_read(port)) == -1) {
+ /*
+ * If no more data available, return to the caller, no error if len>0.
+ * We have no other way to know when the string is finished.
+ */
+ if (len)
+ break;
+ else {
+ DBG(("failed.\n"));
+ return FALSE;
+ }
+ }
+
+ DSPCopyright[len++] = ret;
+
+ } while (len < CARDNAMELEN);
+
+ DBG(("success.\n"));
+
+ return TRUE;
+}
+
+static void __init aedsp16_init_tables(void)
+{
+ int i = 0;
+
+ memset(DSPCopyright, 0, CARDNAMELEN + 1);
+ memset(DSPVersion, 0, CARDVERLEN + 1);
+
+ for (i = 0; orIRQ[i].or; i++)
+ if (orIRQ[i].val == ae_config.irq) {
+ soft_cfg |= orIRQ[i].or;
+ soft_cfg_mss |= orIRQ[i].or;
+ }
+
+ for (i = 0; orMIRQ[i].or; i++)
+ if (orMIRQ[i].or == ae_config.mpu_irq)
+ soft_cfg |= orMIRQ[i].or;
+
+ for (i = 0; orDMA[i].or; i++)
+ if (orDMA[i].val == ae_config.dma) {
+ soft_cfg |= orDMA[i].or;
+ soft_cfg_mss |= orDMA[i].or;
+ }
+}
+
+static int __init aedsp16_init_board(void)
+{
+ aedsp16_init_tables();
+
+ if (aedsp16_dsp_reset(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_dsp_reset: failed!\n");
+ return FALSE;
+ }
+ if (aedsp16_dsp_copyright(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_dsp_copyright: failed!\n");
+ return FALSE;
+ }
+
+ /*
+ * My AEDSP16 card return SC-6000 in DSPCopyright, so
+ * if we have something different, we have to be warned.
+ */
+ if (strcmp("SC-6000", DSPCopyright))
+ printk("[AEDSP16] Warning: non SC-6000 audio card!\n");
+
+ if (aedsp16_dsp_version(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_dsp_version: failed!\n");
+ return FALSE;
+ }
+
+ if (aedsp16_stdcfg(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_stdcfg: failed!\n");
+ return FALSE;
+ }
+
+#if defined(CONFIG_SC6600)
+ if (aedsp16_hard_read(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_hard_read: failed!\n");
+ return FALSE;
+ }
+
+ aedsp16_hard_decode();
+
+ aedsp16_hard_encode();
+
+ if (aedsp16_hard_write(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_hard_write: failed!\n");
+ return FALSE;
+ }
+
+ if (aedsp16_ext_cfg_write(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_ext_cfg_write: failed!\n");
+ return FALSE;
+ }
+#endif /* CONFIG_SC6600 */
+
+ if (aedsp16_setup_board(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] aedsp16_setup_board: failed!\n");
+ return FALSE;
+ }
+
+ if (ae_config.mss_base != -1) {
+ if (ae_config.init & INIT_MSS) {
+ if (aedsp16_init_mss(ae_config.base_io) == FALSE) {
+ printk("[AEDSP16] Can not initialize"
+ "Microsoft Sound System mode.\n");
+ return FALSE;
+ }
+ }
+ }
+
+#if !defined(MODULE) || defined(AEDSP16_INFO) || defined(AEDSP16_DEBUG)
+
+ printk("Audio Excel DSP 16 init v%s (%s %s) [",
+ VERSION, DSPCopyright,
+ DSPVersion);
+
+ if (ae_config.mpu_base != -1) {
+ if (ae_config.init & INIT_MPU401) {
+ printk("MPU401");
+ if ((ae_config.init & INIT_MSS) ||
+ (ae_config.init & INIT_SBPRO))
+ printk(" ");
+ }
+ }
+
+ if (ae_config.mss_base == -1) {
+ if (ae_config.init & INIT_SBPRO) {
+ printk("SBPro");
+ if (ae_config.init & INIT_MSS)
+ printk(" ");
+ }
+ }
+
+ if (ae_config.mss_base != -1)
+ if (ae_config.init & INIT_MSS)
+ printk("MSS");
+
+ printk("]\n");
+#endif /* MODULE || AEDSP16_INFO || AEDSP16_DEBUG */
+
+ mdelay(10);
+
+ return TRUE;
+}
+
+static int __init init_aedsp16_sb(void)
+{
+ DBG(("init_aedsp16_sb: "));
+
+/*
+ * If the card is already init'ed MSS, we can not init it to SBPRO too
+ * because the board can not emulate simultaneously MSS and SBPRO.
+ */
+ if (ae_config.init & INIT_MSS)
+ return FALSE;
+ if (ae_config.init & INIT_SBPRO)
+ return FALSE;
+
+ ae_config.init |= INIT_SBPRO;
+
+ DBG(("done.\n"));
+
+ return TRUE;
+}
+
+static void uninit_aedsp16_sb(void)
+{
+ DBG(("uninit_aedsp16_sb: "));
+
+ ae_config.init &= ~INIT_SBPRO;
+
+ DBG(("done.\n"));
+}
+
+static int __init init_aedsp16_mss(void)
+{
+ DBG(("init_aedsp16_mss: "));
+
+/*
+ * If the card is already init'ed SBPRO, we can not init it to MSS too
+ * because the board can not emulate simultaneously MSS and SBPRO.
+ */
+ if (ae_config.init & INIT_SBPRO)
+ return FALSE;
+ if (ae_config.init & INIT_MSS)
+ return FALSE;
+/*
+ * We must allocate the CONFIG_AEDSP16_BASE region too because these are the
+ * I/O ports to access card's control registers.
+ */
+ if (!(ae_config.init & INIT_MPU401)) {
+ if (!request_region(ae_config.base_io, IOBASE_REGION_SIZE,
+ "aedsp16 (base)")) {
+ printk(
+ "AEDSP16 BASE I/O port region is already in use.\n");
+ return FALSE;
+ }
+ }
+
+ ae_config.init |= INIT_MSS;
+
+ DBG(("done.\n"));
+
+ return TRUE;
+}
+
+static void uninit_aedsp16_mss(void)
+{
+ DBG(("uninit_aedsp16_mss: "));
+
+ if ((!(ae_config.init & INIT_MPU401)) &&
+ (ae_config.init & INIT_MSS)) {
+ release_region(ae_config.base_io, IOBASE_REGION_SIZE);
+ DBG(("AEDSP16 base region released.\n"));
+ }
+
+ ae_config.init &= ~INIT_MSS;
+ DBG(("done.\n"));
+}
+
+static int __init init_aedsp16_mpu(void)
+{
+ DBG(("init_aedsp16_mpu: "));
+
+ if (ae_config.init & INIT_MPU401)
+ return FALSE;
+
+/*
+ * We must request the CONFIG_AEDSP16_BASE region too because these are the I/O
+ * ports to access card's control registers.
+ */
+ if (!(ae_config.init & (INIT_MSS | INIT_SBPRO))) {
+ if (!request_region(ae_config.base_io, IOBASE_REGION_SIZE,
+ "aedsp16 (base)")) {
+ printk(
+ "AEDSP16 BASE I/O port region is already in use.\n");
+ return FALSE;
+ }
+ }
+
+ ae_config.init |= INIT_MPU401;
+
+ DBG(("done.\n"));
+
+ return TRUE;
+}
+
+static void uninit_aedsp16_mpu(void)
+{
+ DBG(("uninit_aedsp16_mpu: "));
+
+ if ((!(ae_config.init & (INIT_MSS | INIT_SBPRO))) &&
+ (ae_config.init & INIT_MPU401)) {
+ release_region(ae_config.base_io, IOBASE_REGION_SIZE);
+ DBG(("AEDSP16 base region released.\n"));
+ }
+
+ ae_config.init &= ~INIT_MPU401;
+
+ DBG(("done.\n"));
+}
+
+static int __init init_aedsp16(void)
+{
+ int initialized = FALSE;
+
+ DBG(("Initializing BASE[0x%x] IRQ[%d] DMA[%d] MIRQ[%d]\n",
+ ae_config.base_io,ae_config.irq,ae_config.dma,ae_config.mpu_irq));
+
+ if (ae_config.mss_base == -1) {
+ if (init_aedsp16_sb() == FALSE) {
+ uninit_aedsp16_sb();
+ } else {
+ initialized = TRUE;
+ }
+ }
+
+ if (ae_config.mpu_base != -1) {
+ if (init_aedsp16_mpu() == FALSE) {
+ uninit_aedsp16_mpu();
+ } else {
+ initialized = TRUE;
+ }
+ }
+
+/*
+ * In the sequence of init routines, the MSS init MUST be the last!
+ * This because of the special register programming the MSS mode needs.
+ * A board reset would disable the MSS mode restoring the default SBPRO
+ * mode.
+ */
+ if (ae_config.mss_base != -1) {
+ if (init_aedsp16_mss() == FALSE) {
+ uninit_aedsp16_mss();
+ } else {
+ initialized = TRUE;
+ }
+ }
+
+ if (initialized)
+ initialized = aedsp16_init_board();
+ return initialized;
+}
+
+static void __exit uninit_aedsp16(void)
+{
+ if (ae_config.mss_base != -1)
+ uninit_aedsp16_mss();
+ else
+ uninit_aedsp16_sb();
+ if (ae_config.mpu_base != -1)
+ uninit_aedsp16_mpu();
+}
+
+static int __initdata io = -1;
+static int __initdata irq = -1;
+static int __initdata dma = -1;
+static int __initdata mpu_irq = -1;
+static int __initdata mss_base = -1;
+static int __initdata mpu_base = -1;
+
+module_param(io, int, 0);
+MODULE_PARM_DESC(io, "I/O base address (0x220 0x240)");
+module_param(irq, int, 0);
+MODULE_PARM_DESC(irq, "IRQ line (5 7 9 10 11)");
+module_param(dma, int, 0);
+MODULE_PARM_DESC(dma, "dma line (0 1 3)");
+module_param(mpu_irq, int, 0);
+MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ line (5 7 9 10 0)");
+module_param(mss_base, int, 0);
+MODULE_PARM_DESC(mss_base, "MSS emulation I/O base address (0x530 0xE80)");
+module_param(mpu_base, int, 0);
+MODULE_PARM_DESC(mpu_base,"MPU-401 I/O base address (0x300 0x310 0x320 0x330)");
+MODULE_AUTHOR("Riccardo Facchetti <fizban@tin.it>");
+MODULE_DESCRIPTION("Audio Excel DSP 16 Driver Version " VERSION);
+MODULE_LICENSE("GPL");
+
+static int __init do_init_aedsp16(void) {
+ printk("Audio Excel DSP 16 init driver Copyright (C) Riccardo Facchetti 1995-98\n");
+ if (io == -1 || dma == -1 || irq == -1) {
+ printk(KERN_INFO "aedsp16: I/O, IRQ and DMA are mandatory\n");
+ return -EINVAL;
+ }
+
+ ae_config.base_io = io;
+ ae_config.irq = irq;
+ ae_config.dma = dma;
+
+ ae_config.mss_base = mss_base;
+ ae_config.mpu_base = mpu_base;
+ ae_config.mpu_irq = mpu_irq;
+
+ if (init_aedsp16() == FALSE) {
+ printk(KERN_ERR "aedsp16: initialization failed\n");
+ /*
+ * XXX
+ * What error should we return here ?
+ */
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static void __exit cleanup_aedsp16(void) {
+ uninit_aedsp16();
+}
+
+module_init(do_init_aedsp16);
+module_exit(cleanup_aedsp16);
+
+#ifndef MODULE
+static int __init setup_aedsp16(char *str)
+{
+ /* io, irq, dma, mss_io, mpu_io, mpu_irq */
+ int ints[7];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+ dma = ints[3];
+ mss_base = ints[4];
+ mpu_base = ints[5];
+ mpu_irq = ints[6];
+ return 1;
+}
+
+__setup("aedsp16=", setup_aedsp16);
+#endif
diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c
new file mode 100644
index 0000000..81e1f44
--- /dev/null
+++ b/sound/oss/au1550_ac97.c
@@ -0,0 +1,2129 @@
+/*
+ * au1550_ac97.c -- Sound driver for Alchemy Au1550 MIPS Internet Edge
+ * Processor.
+ *
+ * Copyright 2004 Embedded Edge, LLC
+ * dan@embeddededge.com
+ *
+ * Mostly copied from the au1000.c driver and some from the
+ * PowerMac dbdma driver.
+ * We assume the processor can do memory coherent DMA.
+ *
+ * Ported to 2.6 by Matt Porter <mporter@kernel.crashing.org>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
+ * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
+ * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
+ * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#undef DEBUG
+
+#include <linux/module.h>
+#include <linux/string.h>
+#include <linux/ioport.h>
+#include <linux/sched.h>
+#include <linux/delay.h>
+#include <linux/sound.h>
+#include <linux/slab.h>
+#include <linux/soundcard.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/kernel.h>
+#include <linux/poll.h>
+#include <linux/bitops.h>
+#include <linux/spinlock.h>
+#include <linux/smp_lock.h>
+#include <linux/ac97_codec.h>
+#include <linux/mutex.h>
+
+#include <asm/io.h>
+#include <asm/uaccess.h>
+#include <asm/hardirq.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-au1x00/au1xxx.h>
+
+#undef OSS_DOCUMENTED_MIXER_SEMANTICS
+
+/* misc stuff */
+#define POLL_COUNT 0x50000
+#define AC97_EXT_DACS (AC97_EXTID_SDAC | AC97_EXTID_CDAC | AC97_EXTID_LDAC)
+
+/* The number of DBDMA ring descriptors to allocate. No sense making
+ * this too large....if you can't keep up with a few you aren't likely
+ * to be able to with lots of them, either.
+ */
+#define NUM_DBDMA_DESCRIPTORS 4
+
+#define err(format, arg...) printk(KERN_ERR format "\n" , ## arg)
+
+/* Boot options
+ * 0 = no VRA, 1 = use VRA if codec supports it
+ */
+static int vra = 1;
+module_param(vra, bool, 0);
+MODULE_PARM_DESC(vra, "if 1 use VRA if codec supports it");
+
+static struct au1550_state {
+ /* soundcore stuff */
+ int dev_audio;
+
+ struct ac97_codec *codec;
+ unsigned codec_base_caps; /* AC'97 reg 00h, "Reset Register" */
+ unsigned codec_ext_caps; /* AC'97 reg 28h, "Extended Audio ID" */
+ int no_vra; /* do not use VRA */
+
+ spinlock_t lock;
+ struct mutex open_mutex;
+ struct mutex sem;
+ fmode_t open_mode;
+ wait_queue_head_t open_wait;
+
+ struct dmabuf {
+ u32 dmanr;
+ unsigned sample_rate;
+ unsigned src_factor;
+ unsigned sample_size;
+ int num_channels;
+ int dma_bytes_per_sample;
+ int user_bytes_per_sample;
+ int cnt_factor;
+
+ void *rawbuf;
+ unsigned buforder;
+ unsigned numfrag;
+ unsigned fragshift;
+ void *nextIn;
+ void *nextOut;
+ int count;
+ unsigned total_bytes;
+ unsigned error;
+ wait_queue_head_t wait;
+
+ /* redundant, but makes calculations easier */
+ unsigned fragsize;
+ unsigned dma_fragsize;
+ unsigned dmasize;
+ unsigned dma_qcount;
+
+ /* OSS stuff */
+ unsigned mapped:1;
+ unsigned ready:1;
+ unsigned stopped:1;
+ unsigned ossfragshift;
+ int ossmaxfrags;
+ unsigned subdivision;
+ } dma_dac, dma_adc;
+} au1550_state;
+
+static unsigned
+ld2(unsigned int x)
+{
+ unsigned r = 0;
+
+ if (x >= 0x10000) {
+ x >>= 16;
+ r += 16;
+ }
+ if (x >= 0x100) {
+ x >>= 8;
+ r += 8;
+ }
+ if (x >= 0x10) {
+ x >>= 4;
+ r += 4;
+ }
+ if (x >= 4) {
+ x >>= 2;
+ r += 2;
+ }
+ if (x >= 2)
+ r++;
+ return r;
+}
+
+static void
+au1550_delay(int msec)
+{
+ unsigned long tmo;
+ signed long tmo2;
+
+ if (in_interrupt())
+ return;
+
+ tmo = jiffies + (msec * HZ) / 1000;
+ for (;;) {
+ tmo2 = tmo - jiffies;
+ if (tmo2 <= 0)
+ break;
+ schedule_timeout(tmo2);
+ }
+}
+
+static u16
+rdcodec(struct ac97_codec *codec, u8 addr)
+{
+ struct au1550_state *s = (struct au1550_state *)codec->private_data;
+ unsigned long flags;
+ u32 cmd, val;
+ u16 data;
+ int i;
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ for (i = 0; i < POLL_COUNT; i++) {
+ val = au_readl(PSC_AC97STAT);
+ au_sync();
+ if (!(val & PSC_AC97STAT_CP))
+ break;
+ }
+ if (i == POLL_COUNT)
+ err("rdcodec: codec cmd pending expired!");
+
+ cmd = (u32)PSC_AC97CDC_INDX(addr);
+ cmd |= PSC_AC97CDC_RD; /* read command */
+ au_writel(cmd, PSC_AC97CDC);
+ au_sync();
+
+ /* now wait for the data
+ */
+ for (i = 0; i < POLL_COUNT; i++) {
+ val = au_readl(PSC_AC97STAT);
+ au_sync();
+ if (!(val & PSC_AC97STAT_CP))
+ break;
+ }
+ if (i == POLL_COUNT) {
+ err("rdcodec: read poll expired!");
+ data = 0;
+ goto out;
+ }
+
+ /* wait for command done?
+ */
+ for (i = 0; i < POLL_COUNT; i++) {
+ val = au_readl(PSC_AC97EVNT);
+ au_sync();
+ if (val & PSC_AC97EVNT_CD)
+ break;
+ }
+ if (i == POLL_COUNT) {
+ err("rdcodec: read cmdwait expired!");
+ data = 0;
+ goto out;
+ }
+
+ data = au_readl(PSC_AC97CDC) & 0xffff;
+ au_sync();
+
+ /* Clear command done event.
+ */
+ au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT);
+ au_sync();
+
+ out:
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ return data;
+}
+
+
+static void
+wrcodec(struct ac97_codec *codec, u8 addr, u16 data)
+{
+ struct au1550_state *s = (struct au1550_state *)codec->private_data;
+ unsigned long flags;
+ u32 cmd, val;
+ int i;
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ for (i = 0; i < POLL_COUNT; i++) {
+ val = au_readl(PSC_AC97STAT);
+ au_sync();
+ if (!(val & PSC_AC97STAT_CP))
+ break;
+ }
+ if (i == POLL_COUNT)
+ err("wrcodec: codec cmd pending expired!");
+
+ cmd = (u32)PSC_AC97CDC_INDX(addr);
+ cmd |= (u32)data;
+ au_writel(cmd, PSC_AC97CDC);
+ au_sync();
+
+ for (i = 0; i < POLL_COUNT; i++) {
+ val = au_readl(PSC_AC97STAT);
+ au_sync();
+ if (!(val & PSC_AC97STAT_CP))
+ break;
+ }
+ if (i == POLL_COUNT)
+ err("wrcodec: codec cmd pending expired!");
+
+ for (i = 0; i < POLL_COUNT; i++) {
+ val = au_readl(PSC_AC97EVNT);
+ au_sync();
+ if (val & PSC_AC97EVNT_CD)
+ break;
+ }
+ if (i == POLL_COUNT)
+ err("wrcodec: read cmdwait expired!");
+
+ /* Clear command done event.
+ */
+ au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT);
+ au_sync();
+
+ spin_unlock_irqrestore(&s->lock, flags);
+}
+
+static void
+waitcodec(struct ac97_codec *codec)
+{
+ u16 temp;
+ u32 val;
+ int i;
+
+ /* codec_wait is used to wait for a ready state after
+ * an AC97C_RESET.
+ */
+ au1550_delay(10);
+
+ /* first poll the CODEC_READY tag bit
+ */
+ for (i = 0; i < POLL_COUNT; i++) {
+ val = au_readl(PSC_AC97STAT);
+ au_sync();
+ if (val & PSC_AC97STAT_CR)
+ break;
+ }
+ if (i == POLL_COUNT) {
+ err("waitcodec: CODEC_READY poll expired!");
+ return;
+ }
+
+ /* get AC'97 powerdown control/status register
+ */
+ temp = rdcodec(codec, AC97_POWER_CONTROL);
+
+ /* If anything is powered down, power'em up
+ */
+ if (temp & 0x7f00) {
+ /* Power on
+ */
+ wrcodec(codec, AC97_POWER_CONTROL, 0);
+ au1550_delay(100);
+
+ /* Reread
+ */
+ temp = rdcodec(codec, AC97_POWER_CONTROL);
+ }
+
+ /* Check if Codec REF,ANL,DAC,ADC ready
+ */
+ if ((temp & 0x7f0f) != 0x000f)
+ err("codec reg 26 status (0x%x) not ready!!", temp);
+}
+
+/* stop the ADC before calling */
+static void
+set_adc_rate(struct au1550_state *s, unsigned rate)
+{
+ struct dmabuf *adc = &s->dma_adc;
+ struct dmabuf *dac = &s->dma_dac;
+ unsigned adc_rate, dac_rate;
+ u16 ac97_extstat;
+
+ if (s->no_vra) {
+ /* calc SRC factor
+ */
+ adc->src_factor = ((96000 / rate) + 1) >> 1;
+ adc->sample_rate = 48000 / adc->src_factor;
+ return;
+ }
+
+ adc->src_factor = 1;
+
+ ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS);
+
+ rate = rate > 48000 ? 48000 : rate;
+
+ /* enable VRA
+ */
+ wrcodec(s->codec, AC97_EXTENDED_STATUS,
+ ac97_extstat | AC97_EXTSTAT_VRA);
+
+ /* now write the sample rate
+ */
+ wrcodec(s->codec, AC97_PCM_LR_ADC_RATE, (u16) rate);
+
+ /* read it back for actual supported rate
+ */
+ adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE);
+
+ pr_debug("set_adc_rate: set to %d Hz\n", adc_rate);
+
+ /* some codec's don't allow unequal DAC and ADC rates, in which case
+ * writing one rate reg actually changes both.
+ */
+ dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE);
+ if (dac->num_channels > 2)
+ wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, dac_rate);
+ if (dac->num_channels > 4)
+ wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, dac_rate);
+
+ adc->sample_rate = adc_rate;
+ dac->sample_rate = dac_rate;
+}
+
+/* stop the DAC before calling */
+static void
+set_dac_rate(struct au1550_state *s, unsigned rate)
+{
+ struct dmabuf *dac = &s->dma_dac;
+ struct dmabuf *adc = &s->dma_adc;
+ unsigned adc_rate, dac_rate;
+ u16 ac97_extstat;
+
+ if (s->no_vra) {
+ /* calc SRC factor
+ */
+ dac->src_factor = ((96000 / rate) + 1) >> 1;
+ dac->sample_rate = 48000 / dac->src_factor;
+ return;
+ }
+
+ dac->src_factor = 1;
+
+ ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS);
+
+ rate = rate > 48000 ? 48000 : rate;
+
+ /* enable VRA
+ */
+ wrcodec(s->codec, AC97_EXTENDED_STATUS,
+ ac97_extstat | AC97_EXTSTAT_VRA);
+
+ /* now write the sample rate
+ */
+ wrcodec(s->codec, AC97_PCM_FRONT_DAC_RATE, (u16) rate);
+
+ /* I don't support different sample rates for multichannel,
+ * so make these channels the same.
+ */
+ if (dac->num_channels > 2)
+ wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, (u16) rate);
+ if (dac->num_channels > 4)
+ wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, (u16) rate);
+ /* read it back for actual supported rate
+ */
+ dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE);
+
+ pr_debug("set_dac_rate: set to %d Hz\n", dac_rate);
+
+ /* some codec's don't allow unequal DAC and ADC rates, in which case
+ * writing one rate reg actually changes both.
+ */
+ adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE);
+
+ dac->sample_rate = dac_rate;
+ adc->sample_rate = adc_rate;
+}
+
+static void
+stop_dac(struct au1550_state *s)
+{
+ struct dmabuf *db = &s->dma_dac;
+ u32 stat;
+ unsigned long flags;
+
+ if (db->stopped)
+ return;
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ au_writel(PSC_AC97PCR_TP, PSC_AC97PCR);
+ au_sync();
+
+ /* Wait for Transmit Busy to show disabled.
+ */
+ do {
+ stat = au_readl(PSC_AC97STAT);
+ au_sync();
+ } while ((stat & PSC_AC97STAT_TB) != 0);
+
+ au1xxx_dbdma_reset(db->dmanr);
+
+ db->stopped = 1;
+
+ spin_unlock_irqrestore(&s->lock, flags);
+}
+
+static void
+stop_adc(struct au1550_state *s)
+{
+ struct dmabuf *db = &s->dma_adc;
+ unsigned long flags;
+ u32 stat;
+
+ if (db->stopped)
+ return;
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ au_writel(PSC_AC97PCR_RP, PSC_AC97PCR);
+ au_sync();
+
+ /* Wait for Receive Busy to show disabled.
+ */
+ do {
+ stat = au_readl(PSC_AC97STAT);
+ au_sync();
+ } while ((stat & PSC_AC97STAT_RB) != 0);
+
+ au1xxx_dbdma_reset(db->dmanr);
+
+ db->stopped = 1;
+
+ spin_unlock_irqrestore(&s->lock, flags);
+}
+
+
+static void
+set_xmit_slots(int num_channels)
+{
+ u32 ac97_config, stat;
+
+ ac97_config = au_readl(PSC_AC97CFG);
+ au_sync();
+ ac97_config &= ~(PSC_AC97CFG_TXSLOT_MASK | PSC_AC97CFG_DE_ENABLE);
+ au_writel(ac97_config, PSC_AC97CFG);
+ au_sync();
+
+ switch (num_channels) {
+ case 6: /* stereo with surround and center/LFE,
+ * slots 3,4,6,7,8,9
+ */
+ ac97_config |= PSC_AC97CFG_TXSLOT_ENA(6);
+ ac97_config |= PSC_AC97CFG_TXSLOT_ENA(9);
+
+ case 4: /* stereo with surround, slots 3,4,7,8 */
+ ac97_config |= PSC_AC97CFG_TXSLOT_ENA(7);
+ ac97_config |= PSC_AC97CFG_TXSLOT_ENA(8);
+
+ case 2: /* stereo, slots 3,4 */
+ case 1: /* mono */
+ ac97_config |= PSC_AC97CFG_TXSLOT_ENA(3);
+ ac97_config |= PSC_AC97CFG_TXSLOT_ENA(4);
+ }
+
+ au_writel(ac97_config, PSC_AC97CFG);
+ au_sync();
+
+ ac97_config |= PSC_AC97CFG_DE_ENABLE;
+ au_writel(ac97_config, PSC_AC97CFG);
+ au_sync();
+
+ /* Wait for Device ready.
+ */
+ do {
+ stat = au_readl(PSC_AC97STAT);
+ au_sync();
+ } while ((stat & PSC_AC97STAT_DR) == 0);
+}
+
+static void
+set_recv_slots(int num_channels)
+{
+ u32 ac97_config, stat;
+
+ ac97_config = au_readl(PSC_AC97CFG);
+ au_sync();
+ ac97_config &= ~(PSC_AC97CFG_RXSLOT_MASK | PSC_AC97CFG_DE_ENABLE);
+ au_writel(ac97_config, PSC_AC97CFG);
+ au_sync();
+
+ /* Always enable slots 3 and 4 (stereo). Slot 6 is
+ * optional Mic ADC, which we don't support yet.
+ */
+ ac97_config |= PSC_AC97CFG_RXSLOT_ENA(3);
+ ac97_config |= PSC_AC97CFG_RXSLOT_ENA(4);
+
+ au_writel(ac97_config, PSC_AC97CFG);
+ au_sync();
+
+ ac97_config |= PSC_AC97CFG_DE_ENABLE;
+ au_writel(ac97_config, PSC_AC97CFG);
+ au_sync();
+
+ /* Wait for Device ready.
+ */
+ do {
+ stat = au_readl(PSC_AC97STAT);
+ au_sync();
+ } while ((stat & PSC_AC97STAT_DR) == 0);
+}
+
+/* Hold spinlock for both start_dac() and start_adc() calls */
+static void
+start_dac(struct au1550_state *s)
+{
+ struct dmabuf *db = &s->dma_dac;
+
+ if (!db->stopped)
+ return;
+
+ set_xmit_slots(db->num_channels);
+ au_writel(PSC_AC97PCR_TC, PSC_AC97PCR);
+ au_sync();
+ au_writel(PSC_AC97PCR_TS, PSC_AC97PCR);
+ au_sync();
+
+ au1xxx_dbdma_start(db->dmanr);
+
+ db->stopped = 0;
+}
+
+static void
+start_adc(struct au1550_state *s)
+{
+ struct dmabuf *db = &s->dma_adc;
+ int i;
+
+ if (!db->stopped)
+ return;
+
+ /* Put two buffers on the ring to get things started.
+ */
+ for (i=0; i<2; i++) {
+ au1xxx_dbdma_put_dest(db->dmanr, db->nextIn, db->dma_fragsize);
+
+ db->nextIn += db->dma_fragsize;
+ if (db->nextIn >= db->rawbuf + db->dmasize)
+ db->nextIn -= db->dmasize;
+ }
+
+ set_recv_slots(db->num_channels);
+ au1xxx_dbdma_start(db->dmanr);
+ au_writel(PSC_AC97PCR_RC, PSC_AC97PCR);
+ au_sync();
+ au_writel(PSC_AC97PCR_RS, PSC_AC97PCR);
+ au_sync();
+
+ db->stopped = 0;
+}
+
+static int
+prog_dmabuf(struct au1550_state *s, struct dmabuf *db)
+{
+ unsigned user_bytes_per_sec;
+ unsigned bufs;
+ unsigned rate = db->sample_rate;
+
+ if (!db->rawbuf) {
+ db->ready = db->mapped = 0;
+ db->buforder = 5; /* 32 * PAGE_SIZE */
+ db->rawbuf = kmalloc((PAGE_SIZE << db->buforder), GFP_KERNEL);
+ if (!db->rawbuf)
+ return -ENOMEM;
+ }
+
+ db->cnt_factor = 1;
+ if (db->sample_size == 8)
+ db->cnt_factor *= 2;
+ if (db->num_channels == 1)
+ db->cnt_factor *= 2;
+ db->cnt_factor *= db->src_factor;
+
+ db->count = 0;
+ db->dma_qcount = 0;
+ db->nextIn = db->nextOut = db->rawbuf;
+
+ db->user_bytes_per_sample = (db->sample_size>>3) * db->num_channels;
+ db->dma_bytes_per_sample = 2 * ((db->num_channels == 1) ?
+ 2 : db->num_channels);
+
+ user_bytes_per_sec = rate * db->user_bytes_per_sample;
+ bufs = PAGE_SIZE << db->buforder;
+ if (db->ossfragshift) {
+ if ((1000 << db->ossfragshift) < user_bytes_per_sec)
+ db->fragshift = ld2(user_bytes_per_sec/1000);
+ else
+ db->fragshift = db->ossfragshift;
+ } else {
+ db->fragshift = ld2(user_bytes_per_sec / 100 /
+ (db->subdivision ? db->subdivision : 1));
+ if (db->fragshift < 3)
+ db->fragshift = 3;
+ }
+
+ db->fragsize = 1 << db->fragshift;
+ db->dma_fragsize = db->fragsize * db->cnt_factor;
+ db->numfrag = bufs / db->dma_fragsize;
+
+ while (db->numfrag < 4 && db->fragshift > 3) {
+ db->fragshift--;
+ db->fragsize = 1 << db->fragshift;
+ db->dma_fragsize = db->fragsize * db->cnt_factor;
+ db->numfrag = bufs / db->dma_fragsize;
+ }
+
+ if (db->ossmaxfrags >= 4 && db->ossmaxfrags < db->numfrag)
+ db->numfrag = db->ossmaxfrags;
+
+ db->dmasize = db->dma_fragsize * db->numfrag;
+ memset(db->rawbuf, 0, bufs);
+
+ pr_debug("prog_dmabuf: rate=%d, samplesize=%d, channels=%d\n",
+ rate, db->sample_size, db->num_channels);
+ pr_debug("prog_dmabuf: fragsize=%d, cnt_factor=%d, dma_fragsize=%d\n",
+ db->fragsize, db->cnt_factor, db->dma_fragsize);
+ pr_debug("prog_dmabuf: numfrag=%d, dmasize=%d\n", db->numfrag, db->dmasize);
+
+ db->ready = 1;
+ return 0;
+}
+
+static int
+prog_dmabuf_adc(struct au1550_state *s)
+{
+ stop_adc(s);
+ return prog_dmabuf(s, &s->dma_adc);
+
+}
+
+static int
+prog_dmabuf_dac(struct au1550_state *s)
+{
+ stop_dac(s);
+ return prog_dmabuf(s, &s->dma_dac);
+}
+
+
+static void dac_dma_interrupt(int irq, void *dev_id)
+{
+ struct au1550_state *s = (struct au1550_state *) dev_id;
+ struct dmabuf *db = &s->dma_dac;
+ u32 ac97c_stat;
+
+ spin_lock(&s->lock);
+
+ ac97c_stat = au_readl(PSC_AC97STAT);
+ if (ac97c_stat & (AC97C_XU | AC97C_XO | AC97C_TE))
+ pr_debug("AC97C status = 0x%08x\n", ac97c_stat);
+ db->dma_qcount--;
+
+ if (db->count >= db->fragsize) {
+ if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut,
+ db->fragsize) == 0) {
+ err("qcount < 2 and no ring room!");
+ }
+ db->nextOut += db->fragsize;
+ if (db->nextOut >= db->rawbuf + db->dmasize)
+ db->nextOut -= db->dmasize;
+ db->count -= db->fragsize;
+ db->total_bytes += db->dma_fragsize;
+ db->dma_qcount++;
+ }
+
+ /* wake up anybody listening */
+ if (waitqueue_active(&db->wait))
+ wake_up(&db->wait);
+
+ spin_unlock(&s->lock);
+}
+
+
+static void adc_dma_interrupt(int irq, void *dev_id)
+{
+ struct au1550_state *s = (struct au1550_state *)dev_id;
+ struct dmabuf *dp = &s->dma_adc;
+ u32 obytes;
+ char *obuf;
+
+ spin_lock(&s->lock);
+
+ /* Pull the buffer from the dma queue.
+ */
+ au1xxx_dbdma_get_dest(dp->dmanr, (void *)(&obuf), &obytes);
+
+ if ((dp->count + obytes) > dp->dmasize) {
+ /* Overrun. Stop ADC and log the error
+ */
+ spin_unlock(&s->lock);
+ stop_adc(s);
+ dp->error++;
+ err("adc overrun");
+ return;
+ }
+
+ /* Put a new empty buffer on the destination DMA.
+ */
+ au1xxx_dbdma_put_dest(dp->dmanr, dp->nextIn, dp->dma_fragsize);
+
+ dp->nextIn += dp->dma_fragsize;
+ if (dp->nextIn >= dp->rawbuf + dp->dmasize)
+ dp->nextIn -= dp->dmasize;
+
+ dp->count += obytes;
+ dp->total_bytes += obytes;
+
+ /* wake up anybody listening
+ */
+ if (waitqueue_active(&dp->wait))
+ wake_up(&dp->wait);
+
+ spin_unlock(&s->lock);
+}
+
+static loff_t
+au1550_llseek(struct file *file, loff_t offset, int origin)
+{
+ return -ESPIPE;
+}
+
+
+static int
+au1550_open_mixdev(struct inode *inode, struct file *file)
+{
+ file->private_data = &au1550_state;
+ return 0;
+}
+
+static int
+au1550_release_mixdev(struct inode *inode, struct file *file)
+{
+ return 0;
+}
+
+static int
+mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd,
+ unsigned long arg)
+{
+ return codec->mixer_ioctl(codec, cmd, arg);
+}
+
+static int
+au1550_ioctl_mixdev(struct inode *inode, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct au1550_state *s = (struct au1550_state *)file->private_data;
+ struct ac97_codec *codec = s->codec;
+
+ return mixdev_ioctl(codec, cmd, arg);
+}
+
+static /*const */ struct file_operations au1550_mixer_fops = {
+ owner:THIS_MODULE,
+ llseek:au1550_llseek,
+ ioctl:au1550_ioctl_mixdev,
+ open:au1550_open_mixdev,
+ release:au1550_release_mixdev,
+};
+
+static int
+drain_dac(struct au1550_state *s, int nonblock)
+{
+ unsigned long flags;
+ int count, tmo;
+
+ if (s->dma_dac.mapped || !s->dma_dac.ready || s->dma_dac.stopped)
+ return 0;
+
+ for (;;) {
+ spin_lock_irqsave(&s->lock, flags);
+ count = s->dma_dac.count;
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (count <= s->dma_dac.fragsize)
+ break;
+ if (signal_pending(current))
+ break;
+ if (nonblock)
+ return -EBUSY;
+ tmo = 1000 * count / (s->no_vra ?
+ 48000 : s->dma_dac.sample_rate);
+ tmo /= s->dma_dac.dma_bytes_per_sample;
+ au1550_delay(tmo);
+ }
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ return 0;
+}
+
+static inline u8 S16_TO_U8(s16 ch)
+{
+ return (u8) (ch >> 8) + 0x80;
+}
+static inline s16 U8_TO_S16(u8 ch)
+{
+ return (s16) (ch - 0x80) << 8;
+}
+
+/*
+ * Translates user samples to dma buffer suitable for AC'97 DAC data:
+ * If mono, copy left channel to right channel in dma buffer.
+ * If 8 bit samples, cvt to 16-bit before writing to dma buffer.
+ * If interpolating (no VRA), duplicate every audio frame src_factor times.
+ */
+static int
+translate_from_user(struct dmabuf *db, char* dmabuf, char* userbuf,
+ int dmacount)
+{
+ int sample, i;
+ int interp_bytes_per_sample;
+ int num_samples;
+ int mono = (db->num_channels == 1);
+ char usersample[12];
+ s16 ch, dmasample[6];
+
+ if (db->sample_size == 16 && !mono && db->src_factor == 1) {
+ /* no translation necessary, just copy
+ */
+ if (copy_from_user(dmabuf, userbuf, dmacount))
+ return -EFAULT;
+ return dmacount;
+ }
+
+ interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor;
+ num_samples = dmacount / interp_bytes_per_sample;
+
+ for (sample = 0; sample < num_samples; sample++) {
+ if (copy_from_user(usersample, userbuf,
+ db->user_bytes_per_sample)) {
+ return -EFAULT;
+ }
+
+ for (i = 0; i < db->num_channels; i++) {
+ if (db->sample_size == 8)
+ ch = U8_TO_S16(usersample[i]);
+ else
+ ch = *((s16 *) (&usersample[i * 2]));
+ dmasample[i] = ch;
+ if (mono)
+ dmasample[i + 1] = ch; /* right channel */
+ }
+
+ /* duplicate every audio frame src_factor times
+ */
+ for (i = 0; i < db->src_factor; i++)
+ memcpy(dmabuf, dmasample, db->dma_bytes_per_sample);
+
+ userbuf += db->user_bytes_per_sample;
+ dmabuf += interp_bytes_per_sample;
+ }
+
+ return num_samples * interp_bytes_per_sample;
+}
+
+/*
+ * Translates AC'97 ADC samples to user buffer:
+ * If mono, send only left channel to user buffer.
+ * If 8 bit samples, cvt from 16 to 8 bit before writing to user buffer.
+ * If decimating (no VRA), skip over src_factor audio frames.
+ */
+static int
+translate_to_user(struct dmabuf *db, char* userbuf, char* dmabuf,
+ int dmacount)
+{
+ int sample, i;
+ int interp_bytes_per_sample;
+ int num_samples;
+ int mono = (db->num_channels == 1);
+ char usersample[12];
+
+ if (db->sample_size == 16 && !mono && db->src_factor == 1) {
+ /* no translation necessary, just copy
+ */
+ if (copy_to_user(userbuf, dmabuf, dmacount))
+ return -EFAULT;
+ return dmacount;
+ }
+
+ interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor;
+ num_samples = dmacount / interp_bytes_per_sample;
+
+ for (sample = 0; sample < num_samples; sample++) {
+ for (i = 0; i < db->num_channels; i++) {
+ if (db->sample_size == 8)
+ usersample[i] =
+ S16_TO_U8(*((s16 *) (&dmabuf[i * 2])));
+ else
+ *((s16 *) (&usersample[i * 2])) =
+ *((s16 *) (&dmabuf[i * 2]));
+ }
+
+ if (copy_to_user(userbuf, usersample,
+ db->user_bytes_per_sample)) {
+ return -EFAULT;
+ }
+
+ userbuf += db->user_bytes_per_sample;
+ dmabuf += interp_bytes_per_sample;
+ }
+
+ return num_samples * interp_bytes_per_sample;
+}
+
+/*
+ * Copy audio data to/from user buffer from/to dma buffer, taking care
+ * that we wrap when reading/writing the dma buffer. Returns actual byte
+ * count written to or read from the dma buffer.
+ */
+static int
+copy_dmabuf_user(struct dmabuf *db, char* userbuf, int count, int to_user)
+{
+ char *bufptr = to_user ? db->nextOut : db->nextIn;
+ char *bufend = db->rawbuf + db->dmasize;
+ int cnt, ret;
+
+ if (bufptr + count > bufend) {
+ int partial = (int) (bufend - bufptr);
+ if (to_user) {
+ if ((cnt = translate_to_user(db, userbuf,
+ bufptr, partial)) < 0)
+ return cnt;
+ ret = cnt;
+ if ((cnt = translate_to_user(db, userbuf + partial,
+ db->rawbuf,
+ count - partial)) < 0)
+ return cnt;
+ ret += cnt;
+ } else {
+ if ((cnt = translate_from_user(db, bufptr, userbuf,
+ partial)) < 0)
+ return cnt;
+ ret = cnt;
+ if ((cnt = translate_from_user(db, db->rawbuf,
+ userbuf + partial,
+ count - partial)) < 0)
+ return cnt;
+ ret += cnt;
+ }
+ } else {
+ if (to_user)
+ ret = translate_to_user(db, userbuf, bufptr, count);
+ else
+ ret = translate_from_user(db, bufptr, userbuf, count);
+ }
+
+ return ret;
+}
+
+
+static ssize_t
+au1550_read(struct file *file, char *buffer, size_t count, loff_t *ppos)
+{
+ struct au1550_state *s = (struct au1550_state *)file->private_data;
+ struct dmabuf *db = &s->dma_adc;
+ DECLARE_WAITQUEUE(wait, current);
+ ssize_t ret;
+ unsigned long flags;
+ int cnt, usercnt, avail;
+
+ if (db->mapped)
+ return -ENXIO;
+ if (!access_ok(VERIFY_WRITE, buffer, count))
+ return -EFAULT;
+ ret = 0;
+
+ count *= db->cnt_factor;
+
+ mutex_lock(&s->sem);
+ add_wait_queue(&db->wait, &wait);
+
+ while (count > 0) {
+ /* wait for samples in ADC dma buffer
+ */
+ do {
+ spin_lock_irqsave(&s->lock, flags);
+ if (db->stopped)
+ start_adc(s);
+ avail = db->count;
+ if (avail <= 0)
+ __set_current_state(TASK_INTERRUPTIBLE);
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (avail <= 0) {
+ if (file->f_flags & O_NONBLOCK) {
+ if (!ret)
+ ret = -EAGAIN;
+ goto out;
+ }
+ mutex_unlock(&s->sem);
+ schedule();
+ if (signal_pending(current)) {
+ if (!ret)
+ ret = -ERESTARTSYS;
+ goto out2;
+ }
+ mutex_lock(&s->sem);
+ }
+ } while (avail <= 0);
+
+ /* copy from nextOut to user
+ */
+ if ((cnt = copy_dmabuf_user(db, buffer,
+ count > avail ?
+ avail : count, 1)) < 0) {
+ if (!ret)
+ ret = -EFAULT;
+ goto out;
+ }
+
+ spin_lock_irqsave(&s->lock, flags);
+ db->count -= cnt;
+ db->nextOut += cnt;
+ if (db->nextOut >= db->rawbuf + db->dmasize)
+ db->nextOut -= db->dmasize;
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ count -= cnt;
+ usercnt = cnt / db->cnt_factor;
+ buffer += usercnt;
+ ret += usercnt;
+ } /* while (count > 0) */
+
+out:
+ mutex_unlock(&s->sem);
+out2:
+ remove_wait_queue(&db->wait, &wait);
+ set_current_state(TASK_RUNNING);
+ return ret;
+}
+
+static ssize_t
+au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos)
+{
+ struct au1550_state *s = (struct au1550_state *)file->private_data;
+ struct dmabuf *db = &s->dma_dac;
+ DECLARE_WAITQUEUE(wait, current);
+ ssize_t ret = 0;
+ unsigned long flags;
+ int cnt, usercnt, avail;
+
+ pr_debug("write: count=%d\n", count);
+
+ if (db->mapped)
+ return -ENXIO;
+ if (!access_ok(VERIFY_READ, buffer, count))
+ return -EFAULT;
+
+ count *= db->cnt_factor;
+
+ mutex_lock(&s->sem);
+ add_wait_queue(&db->wait, &wait);
+
+ while (count > 0) {
+ /* wait for space in playback buffer
+ */
+ do {
+ spin_lock_irqsave(&s->lock, flags);
+ avail = (int) db->dmasize - db->count;
+ if (avail <= 0)
+ __set_current_state(TASK_INTERRUPTIBLE);
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (avail <= 0) {
+ if (file->f_flags & O_NONBLOCK) {
+ if (!ret)
+ ret = -EAGAIN;
+ goto out;
+ }
+ mutex_unlock(&s->sem);
+ schedule();
+ if (signal_pending(current)) {
+ if (!ret)
+ ret = -ERESTARTSYS;
+ goto out2;
+ }
+ mutex_lock(&s->sem);
+ }
+ } while (avail <= 0);
+
+ /* copy from user to nextIn
+ */
+ if ((cnt = copy_dmabuf_user(db, (char *) buffer,
+ count > avail ?
+ avail : count, 0)) < 0) {
+ if (!ret)
+ ret = -EFAULT;
+ goto out;
+ }
+
+ spin_lock_irqsave(&s->lock, flags);
+ db->count += cnt;
+ db->nextIn += cnt;
+ if (db->nextIn >= db->rawbuf + db->dmasize)
+ db->nextIn -= db->dmasize;
+
+ /* If the data is available, we want to keep two buffers
+ * on the dma queue. If the queue count reaches zero,
+ * we know the dma has stopped.
+ */
+ while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) {
+ if (au1xxx_dbdma_put_source(db->dmanr, db->nextOut,
+ db->fragsize) == 0) {
+ err("qcount < 2 and no ring room!");
+ }
+ db->nextOut += db->fragsize;
+ if (db->nextOut >= db->rawbuf + db->dmasize)
+ db->nextOut -= db->dmasize;
+ db->total_bytes += db->dma_fragsize;
+ if (db->dma_qcount == 0)
+ start_dac(s);
+ db->dma_qcount++;
+ }
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ count -= cnt;
+ usercnt = cnt / db->cnt_factor;
+ buffer += usercnt;
+ ret += usercnt;
+ } /* while (count > 0) */
+
+out:
+ mutex_unlock(&s->sem);
+out2:
+ remove_wait_queue(&db->wait, &wait);
+ set_current_state(TASK_RUNNING);
+ return ret;
+}
+
+
+/* No kernel lock - we have our own spinlock */
+static unsigned int
+au1550_poll(struct file *file, struct poll_table_struct *wait)
+{
+ struct au1550_state *s = (struct au1550_state *)file->private_data;
+ unsigned long flags;
+ unsigned int mask = 0;
+
+ if (file->f_mode & FMODE_WRITE) {
+ if (!s->dma_dac.ready)
+ return 0;
+ poll_wait(file, &s->dma_dac.wait, wait);
+ }
+ if (file->f_mode & FMODE_READ) {
+ if (!s->dma_adc.ready)
+ return 0;
+ poll_wait(file, &s->dma_adc.wait, wait);
+ }
+
+ spin_lock_irqsave(&s->lock, flags);
+
+ if (file->f_mode & FMODE_READ) {
+ if (s->dma_adc.count >= (signed)s->dma_adc.dma_fragsize)
+ mask |= POLLIN | POLLRDNORM;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ if (s->dma_dac.mapped) {
+ if (s->dma_dac.count >=
+ (signed)s->dma_dac.dma_fragsize)
+ mask |= POLLOUT | POLLWRNORM;
+ } else {
+ if ((signed) s->dma_dac.dmasize >=
+ s->dma_dac.count + (signed)s->dma_dac.dma_fragsize)
+ mask |= POLLOUT | POLLWRNORM;
+ }
+ }
+ spin_unlock_irqrestore(&s->lock, flags);
+ return mask;
+}
+
+static int
+au1550_mmap(struct file *file, struct vm_area_struct *vma)
+{
+ struct au1550_state *s = (struct au1550_state *)file->private_data;
+ struct dmabuf *db;
+ unsigned long size;
+ int ret = 0;
+
+ lock_kernel();
+ mutex_lock(&s->sem);
+ if (vma->vm_flags & VM_WRITE)
+ db = &s->dma_dac;
+ else if (vma->vm_flags & VM_READ)
+ db = &s->dma_adc;
+ else {
+ ret = -EINVAL;
+ goto out;
+ }
+ if (vma->vm_pgoff != 0) {
+ ret = -EINVAL;
+ goto out;
+ }
+ size = vma->vm_end - vma->vm_start;
+ if (size > (PAGE_SIZE << db->buforder)) {
+ ret = -EINVAL;
+ goto out;
+ }
+ if (remap_pfn_range(vma, vma->vm_start, page_to_pfn(virt_to_page(db->rawbuf)),
+ size, vma->vm_page_prot)) {
+ ret = -EAGAIN;
+ goto out;
+ }
+ vma->vm_flags &= ~VM_IO;
+ db->mapped = 1;
+out:
+ mutex_unlock(&s->sem);
+ unlock_kernel();
+ return ret;
+}
+
+#ifdef DEBUG
+static struct ioctl_str_t {
+ unsigned int cmd;
+ const char *str;
+} ioctl_str[] = {
+ {SNDCTL_DSP_RESET, "SNDCTL_DSP_RESET"},
+ {SNDCTL_DSP_SYNC, "SNDCTL_DSP_SYNC"},
+ {SNDCTL_DSP_SPEED, "SNDCTL_DSP_SPEED"},
+ {SNDCTL_DSP_STEREO, "SNDCTL_DSP_STEREO"},
+ {SNDCTL_DSP_GETBLKSIZE, "SNDCTL_DSP_GETBLKSIZE"},
+ {SNDCTL_DSP_SAMPLESIZE, "SNDCTL_DSP_SAMPLESIZE"},
+ {SNDCTL_DSP_CHANNELS, "SNDCTL_DSP_CHANNELS"},
+ {SOUND_PCM_WRITE_CHANNELS, "SOUND_PCM_WRITE_CHANNELS"},
+ {SOUND_PCM_WRITE_FILTER, "SOUND_PCM_WRITE_FILTER"},
+ {SNDCTL_DSP_POST, "SNDCTL_DSP_POST"},
+ {SNDCTL_DSP_SUBDIVIDE, "SNDCTL_DSP_SUBDIVIDE"},
+ {SNDCTL_DSP_SETFRAGMENT, "SNDCTL_DSP_SETFRAGMENT"},
+ {SNDCTL_DSP_GETFMTS, "SNDCTL_DSP_GETFMTS"},
+ {SNDCTL_DSP_SETFMT, "SNDCTL_DSP_SETFMT"},
+ {SNDCTL_DSP_GETOSPACE, "SNDCTL_DSP_GETOSPACE"},
+ {SNDCTL_DSP_GETISPACE, "SNDCTL_DSP_GETISPACE"},
+ {SNDCTL_DSP_NONBLOCK, "SNDCTL_DSP_NONBLOCK"},
+ {SNDCTL_DSP_GETCAPS, "SNDCTL_DSP_GETCAPS"},
+ {SNDCTL_DSP_GETTRIGGER, "SNDCTL_DSP_GETTRIGGER"},
+ {SNDCTL_DSP_SETTRIGGER, "SNDCTL_DSP_SETTRIGGER"},
+ {SNDCTL_DSP_GETIPTR, "SNDCTL_DSP_GETIPTR"},
+ {SNDCTL_DSP_GETOPTR, "SNDCTL_DSP_GETOPTR"},
+ {SNDCTL_DSP_MAPINBUF, "SNDCTL_DSP_MAPINBUF"},
+ {SNDCTL_DSP_MAPOUTBUF, "SNDCTL_DSP_MAPOUTBUF"},
+ {SNDCTL_DSP_SETSYNCRO, "SNDCTL_DSP_SETSYNCRO"},
+ {SNDCTL_DSP_SETDUPLEX, "SNDCTL_DSP_SETDUPLEX"},
+ {SNDCTL_DSP_GETODELAY, "SNDCTL_DSP_GETODELAY"},
+ {SNDCTL_DSP_GETCHANNELMASK, "SNDCTL_DSP_GETCHANNELMASK"},
+ {SNDCTL_DSP_BIND_CHANNEL, "SNDCTL_DSP_BIND_CHANNEL"},
+ {OSS_GETVERSION, "OSS_GETVERSION"},
+ {SOUND_PCM_READ_RATE, "SOUND_PCM_READ_RATE"},
+ {SOUND_PCM_READ_CHANNELS, "SOUND_PCM_READ_CHANNELS"},
+ {SOUND_PCM_READ_BITS, "SOUND_PCM_READ_BITS"},
+ {SOUND_PCM_READ_FILTER, "SOUND_PCM_READ_FILTER"}
+};
+#endif
+
+static int
+dma_count_done(struct dmabuf *db)
+{
+ if (db->stopped)
+ return 0;
+
+ return db->dma_fragsize - au1xxx_get_dma_residue(db->dmanr);
+}
+
+
+static int
+au1550_ioctl(struct inode *inode, struct file *file, unsigned int cmd,
+ unsigned long arg)
+{
+ struct au1550_state *s = (struct au1550_state *)file->private_data;
+ unsigned long flags;
+ audio_buf_info abinfo;
+ count_info cinfo;
+ int count;
+ int val, mapped, ret, diff;
+
+ mapped = ((file->f_mode & FMODE_WRITE) && s->dma_dac.mapped) ||
+ ((file->f_mode & FMODE_READ) && s->dma_adc.mapped);
+
+#ifdef DEBUG
+ for (count = 0; count < ARRAY_SIZE(ioctl_str); count++) {
+ if (ioctl_str[count].cmd == cmd)
+ break;
+ }
+ if (count < ARRAY_SIZE(ioctl_str))
+ pr_debug("ioctl %s, arg=0x%lxn", ioctl_str[count].str, arg);
+ else
+ pr_debug("ioctl 0x%x unknown, arg=0x%lx\n", cmd, arg);
+#endif
+
+ switch (cmd) {
+ case OSS_GETVERSION:
+ return put_user(SOUND_VERSION, (int *) arg);
+
+ case SNDCTL_DSP_SYNC:
+ if (file->f_mode & FMODE_WRITE)
+ return drain_dac(s, file->f_flags & O_NONBLOCK);
+ return 0;
+
+ case SNDCTL_DSP_SETDUPLEX:
+ return 0;
+
+ case SNDCTL_DSP_GETCAPS:
+ return put_user(DSP_CAP_DUPLEX | DSP_CAP_REALTIME |
+ DSP_CAP_TRIGGER | DSP_CAP_MMAP, (int *)arg);
+
+ case SNDCTL_DSP_RESET:
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ synchronize_irq();
+ s->dma_dac.count = s->dma_dac.total_bytes = 0;
+ s->dma_dac.nextIn = s->dma_dac.nextOut =
+ s->dma_dac.rawbuf;
+ }
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ synchronize_irq();
+ s->dma_adc.count = s->dma_adc.total_bytes = 0;
+ s->dma_adc.nextIn = s->dma_adc.nextOut =
+ s->dma_adc.rawbuf;
+ }
+ return 0;
+
+ case SNDCTL_DSP_SPEED:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (val >= 0) {
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ set_adc_rate(s, val);
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ set_dac_rate(s, val);
+ }
+ if (s->open_mode & FMODE_READ)
+ if ((ret = prog_dmabuf_adc(s)))
+ return ret;
+ if (s->open_mode & FMODE_WRITE)
+ if ((ret = prog_dmabuf_dac(s)))
+ return ret;
+ }
+ return put_user((file->f_mode & FMODE_READ) ?
+ s->dma_adc.sample_rate :
+ s->dma_dac.sample_rate,
+ (int *)arg);
+
+ case SNDCTL_DSP_STEREO:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ s->dma_adc.num_channels = val ? 2 : 1;
+ if ((ret = prog_dmabuf_adc(s)))
+ return ret;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ s->dma_dac.num_channels = val ? 2 : 1;
+ if (s->codec_ext_caps & AC97_EXT_DACS) {
+ /* disable surround and center/lfe in AC'97
+ */
+ u16 ext_stat = rdcodec(s->codec,
+ AC97_EXTENDED_STATUS);
+ wrcodec(s->codec, AC97_EXTENDED_STATUS,
+ ext_stat | (AC97_EXTSTAT_PRI |
+ AC97_EXTSTAT_PRJ |
+ AC97_EXTSTAT_PRK));
+ }
+ if ((ret = prog_dmabuf_dac(s)))
+ return ret;
+ }
+ return 0;
+
+ case SNDCTL_DSP_CHANNELS:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (val != 0) {
+ if (file->f_mode & FMODE_READ) {
+ if (val < 0 || val > 2)
+ return -EINVAL;
+ stop_adc(s);
+ s->dma_adc.num_channels = val;
+ if ((ret = prog_dmabuf_adc(s)))
+ return ret;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ switch (val) {
+ case 1:
+ case 2:
+ break;
+ case 3:
+ case 5:
+ return -EINVAL;
+ case 4:
+ if (!(s->codec_ext_caps &
+ AC97_EXTID_SDAC))
+ return -EINVAL;
+ break;
+ case 6:
+ if ((s->codec_ext_caps &
+ AC97_EXT_DACS) != AC97_EXT_DACS)
+ return -EINVAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ stop_dac(s);
+ if (val <= 2 &&
+ (s->codec_ext_caps & AC97_EXT_DACS)) {
+ /* disable surround and center/lfe
+ * channels in AC'97
+ */
+ u16 ext_stat =
+ rdcodec(s->codec,
+ AC97_EXTENDED_STATUS);
+ wrcodec(s->codec,
+ AC97_EXTENDED_STATUS,
+ ext_stat | (AC97_EXTSTAT_PRI |
+ AC97_EXTSTAT_PRJ |
+ AC97_EXTSTAT_PRK));
+ } else if (val >= 4) {
+ /* enable surround, center/lfe
+ * channels in AC'97
+ */
+ u16 ext_stat =
+ rdcodec(s->codec,
+ AC97_EXTENDED_STATUS);
+ ext_stat &= ~AC97_EXTSTAT_PRJ;
+ if (val == 6)
+ ext_stat &=
+ ~(AC97_EXTSTAT_PRI |
+ AC97_EXTSTAT_PRK);
+ wrcodec(s->codec,
+ AC97_EXTENDED_STATUS,
+ ext_stat);
+ }
+
+ s->dma_dac.num_channels = val;
+ if ((ret = prog_dmabuf_dac(s)))
+ return ret;
+ }
+ }
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_GETFMTS: /* Returns a mask */
+ return put_user(AFMT_S16_LE | AFMT_U8, (int *) arg);
+
+ case SNDCTL_DSP_SETFMT: /* Selects ONE fmt */
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (val != AFMT_QUERY) {
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ if (val == AFMT_S16_LE)
+ s->dma_adc.sample_size = 16;
+ else {
+ val = AFMT_U8;
+ s->dma_adc.sample_size = 8;
+ }
+ if ((ret = prog_dmabuf_adc(s)))
+ return ret;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ if (val == AFMT_S16_LE)
+ s->dma_dac.sample_size = 16;
+ else {
+ val = AFMT_U8;
+ s->dma_dac.sample_size = 8;
+ }
+ if ((ret = prog_dmabuf_dac(s)))
+ return ret;
+ }
+ } else {
+ if (file->f_mode & FMODE_READ)
+ val = (s->dma_adc.sample_size == 16) ?
+ AFMT_S16_LE : AFMT_U8;
+ else
+ val = (s->dma_dac.sample_size == 16) ?
+ AFMT_S16_LE : AFMT_U8;
+ }
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_POST:
+ return 0;
+
+ case SNDCTL_DSP_GETTRIGGER:
+ val = 0;
+ spin_lock_irqsave(&s->lock, flags);
+ if (file->f_mode & FMODE_READ && !s->dma_adc.stopped)
+ val |= PCM_ENABLE_INPUT;
+ if (file->f_mode & FMODE_WRITE && !s->dma_dac.stopped)
+ val |= PCM_ENABLE_OUTPUT;
+ spin_unlock_irqrestore(&s->lock, flags);
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_SETTRIGGER:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (file->f_mode & FMODE_READ) {
+ if (val & PCM_ENABLE_INPUT) {
+ spin_lock_irqsave(&s->lock, flags);
+ start_adc(s);
+ spin_unlock_irqrestore(&s->lock, flags);
+ } else
+ stop_adc(s);
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ if (val & PCM_ENABLE_OUTPUT) {
+ spin_lock_irqsave(&s->lock, flags);
+ start_dac(s);
+ spin_unlock_irqrestore(&s->lock, flags);
+ } else
+ stop_dac(s);
+ }
+ return 0;
+
+ case SNDCTL_DSP_GETOSPACE:
+ if (!(file->f_mode & FMODE_WRITE))
+ return -EINVAL;
+ abinfo.fragsize = s->dma_dac.fragsize;
+ spin_lock_irqsave(&s->lock, flags);
+ count = s->dma_dac.count;
+ count -= dma_count_done(&s->dma_dac);
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (count < 0)
+ count = 0;
+ abinfo.bytes = (s->dma_dac.dmasize - count) /
+ s->dma_dac.cnt_factor;
+ abinfo.fragstotal = s->dma_dac.numfrag;
+ abinfo.fragments = abinfo.bytes >> s->dma_dac.fragshift;
+ pr_debug("ioctl SNDCTL_DSP_GETOSPACE: bytes=%d, fragments=%d\n", abinfo.bytes, abinfo.fragments);
+ return copy_to_user((void *) arg, &abinfo,
+ sizeof(abinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_GETISPACE:
+ if (!(file->f_mode & FMODE_READ))
+ return -EINVAL;
+ abinfo.fragsize = s->dma_adc.fragsize;
+ spin_lock_irqsave(&s->lock, flags);
+ count = s->dma_adc.count;
+ count += dma_count_done(&s->dma_adc);
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (count < 0)
+ count = 0;
+ abinfo.bytes = count / s->dma_adc.cnt_factor;
+ abinfo.fragstotal = s->dma_adc.numfrag;
+ abinfo.fragments = abinfo.bytes >> s->dma_adc.fragshift;
+ return copy_to_user((void *) arg, &abinfo,
+ sizeof(abinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_NONBLOCK:
+ file->f_flags |= O_NONBLOCK;
+ return 0;
+
+ case SNDCTL_DSP_GETODELAY:
+ if (!(file->f_mode & FMODE_WRITE))
+ return -EINVAL;
+ spin_lock_irqsave(&s->lock, flags);
+ count = s->dma_dac.count;
+ count -= dma_count_done(&s->dma_dac);
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (count < 0)
+ count = 0;
+ count /= s->dma_dac.cnt_factor;
+ return put_user(count, (int *) arg);
+
+ case SNDCTL_DSP_GETIPTR:
+ if (!(file->f_mode & FMODE_READ))
+ return -EINVAL;
+ spin_lock_irqsave(&s->lock, flags);
+ cinfo.bytes = s->dma_adc.total_bytes;
+ count = s->dma_adc.count;
+ if (!s->dma_adc.stopped) {
+ diff = dma_count_done(&s->dma_adc);
+ count += diff;
+ cinfo.bytes += diff;
+ cinfo.ptr = virt_to_phys(s->dma_adc.nextIn) + diff -
+ virt_to_phys(s->dma_adc.rawbuf);
+ } else
+ cinfo.ptr = virt_to_phys(s->dma_adc.nextIn) -
+ virt_to_phys(s->dma_adc.rawbuf);
+ if (s->dma_adc.mapped)
+ s->dma_adc.count &= (s->dma_adc.dma_fragsize-1);
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (count < 0)
+ count = 0;
+ cinfo.blocks = count >> s->dma_adc.fragshift;
+ return copy_to_user((void *) arg, &cinfo, sizeof(cinfo));
+
+ case SNDCTL_DSP_GETOPTR:
+ if (!(file->f_mode & FMODE_READ))
+ return -EINVAL;
+ spin_lock_irqsave(&s->lock, flags);
+ cinfo.bytes = s->dma_dac.total_bytes;
+ count = s->dma_dac.count;
+ if (!s->dma_dac.stopped) {
+ diff = dma_count_done(&s->dma_dac);
+ count -= diff;
+ cinfo.bytes += diff;
+ cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) + diff -
+ virt_to_phys(s->dma_dac.rawbuf);
+ } else
+ cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) -
+ virt_to_phys(s->dma_dac.rawbuf);
+ if (s->dma_dac.mapped)
+ s->dma_dac.count &= (s->dma_dac.dma_fragsize-1);
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (count < 0)
+ count = 0;
+ cinfo.blocks = count >> s->dma_dac.fragshift;
+ return copy_to_user((void *) arg, &cinfo, sizeof(cinfo));
+
+ case SNDCTL_DSP_GETBLKSIZE:
+ if (file->f_mode & FMODE_WRITE)
+ return put_user(s->dma_dac.fragsize, (int *) arg);
+ else
+ return put_user(s->dma_adc.fragsize, (int *) arg);
+
+ case SNDCTL_DSP_SETFRAGMENT:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ s->dma_adc.ossfragshift = val & 0xffff;
+ s->dma_adc.ossmaxfrags = (val >> 16) & 0xffff;
+ if (s->dma_adc.ossfragshift < 4)
+ s->dma_adc.ossfragshift = 4;
+ if (s->dma_adc.ossfragshift > 15)
+ s->dma_adc.ossfragshift = 15;
+ if (s->dma_adc.ossmaxfrags < 4)
+ s->dma_adc.ossmaxfrags = 4;
+ if ((ret = prog_dmabuf_adc(s)))
+ return ret;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ s->dma_dac.ossfragshift = val & 0xffff;
+ s->dma_dac.ossmaxfrags = (val >> 16) & 0xffff;
+ if (s->dma_dac.ossfragshift < 4)
+ s->dma_dac.ossfragshift = 4;
+ if (s->dma_dac.ossfragshift > 15)
+ s->dma_dac.ossfragshift = 15;
+ if (s->dma_dac.ossmaxfrags < 4)
+ s->dma_dac.ossmaxfrags = 4;
+ if ((ret = prog_dmabuf_dac(s)))
+ return ret;
+ }
+ return 0;
+
+ case SNDCTL_DSP_SUBDIVIDE:
+ if ((file->f_mode & FMODE_READ && s->dma_adc.subdivision) ||
+ (file->f_mode & FMODE_WRITE && s->dma_dac.subdivision))
+ return -EINVAL;
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (val != 1 && val != 2 && val != 4)
+ return -EINVAL;
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ s->dma_adc.subdivision = val;
+ if ((ret = prog_dmabuf_adc(s)))
+ return ret;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ s->dma_dac.subdivision = val;
+ if ((ret = prog_dmabuf_dac(s)))
+ return ret;
+ }
+ return 0;
+
+ case SOUND_PCM_READ_RATE:
+ return put_user((file->f_mode & FMODE_READ) ?
+ s->dma_adc.sample_rate :
+ s->dma_dac.sample_rate,
+ (int *)arg);
+
+ case SOUND_PCM_READ_CHANNELS:
+ if (file->f_mode & FMODE_READ)
+ return put_user(s->dma_adc.num_channels, (int *)arg);
+ else
+ return put_user(s->dma_dac.num_channels, (int *)arg);
+
+ case SOUND_PCM_READ_BITS:
+ if (file->f_mode & FMODE_READ)
+ return put_user(s->dma_adc.sample_size, (int *)arg);
+ else
+ return put_user(s->dma_dac.sample_size, (int *)arg);
+
+ case SOUND_PCM_WRITE_FILTER:
+ case SNDCTL_DSP_SETSYNCRO:
+ case SOUND_PCM_READ_FILTER:
+ return -EINVAL;
+ }
+
+ return mixdev_ioctl(s->codec, cmd, arg);
+}
+
+
+static int
+au1550_open(struct inode *inode, struct file *file)
+{
+ int minor = MINOR(inode->i_rdev);
+ DECLARE_WAITQUEUE(wait, current);
+ struct au1550_state *s = &au1550_state;
+ int ret;
+
+#ifdef DEBUG
+ if (file->f_flags & O_NONBLOCK)
+ pr_debug("open: non-blocking\n");
+ else
+ pr_debug("open: blocking\n");
+#endif
+
+ file->private_data = s;
+ /* wait for device to become free */
+ mutex_lock(&s->open_mutex);
+ while (s->open_mode & file->f_mode) {
+ if (file->f_flags & O_NONBLOCK) {
+ mutex_unlock(&s->open_mutex);
+ return -EBUSY;
+ }
+ add_wait_queue(&s->open_wait, &wait);
+ __set_current_state(TASK_INTERRUPTIBLE);
+ mutex_unlock(&s->open_mutex);
+ schedule();
+ remove_wait_queue(&s->open_wait, &wait);
+ set_current_state(TASK_RUNNING);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ mutex_lock(&s->open_mutex);
+ }
+
+ stop_dac(s);
+ stop_adc(s);
+
+ if (file->f_mode & FMODE_READ) {
+ s->dma_adc.ossfragshift = s->dma_adc.ossmaxfrags =
+ s->dma_adc.subdivision = s->dma_adc.total_bytes = 0;
+ s->dma_adc.num_channels = 1;
+ s->dma_adc.sample_size = 8;
+ set_adc_rate(s, 8000);
+ if ((minor & 0xf) == SND_DEV_DSP16)
+ s->dma_adc.sample_size = 16;
+ }
+
+ if (file->f_mode & FMODE_WRITE) {
+ s->dma_dac.ossfragshift = s->dma_dac.ossmaxfrags =
+ s->dma_dac.subdivision = s->dma_dac.total_bytes = 0;
+ s->dma_dac.num_channels = 1;
+ s->dma_dac.sample_size = 8;
+ set_dac_rate(s, 8000);
+ if ((minor & 0xf) == SND_DEV_DSP16)
+ s->dma_dac.sample_size = 16;
+ }
+
+ if (file->f_mode & FMODE_READ) {
+ if ((ret = prog_dmabuf_adc(s)))
+ return ret;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ if ((ret = prog_dmabuf_dac(s)))
+ return ret;
+ }
+
+ s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE);
+ mutex_unlock(&s->open_mutex);
+ mutex_init(&s->sem);
+ return 0;
+}
+
+static int
+au1550_release(struct inode *inode, struct file *file)
+{
+ struct au1550_state *s = (struct au1550_state *)file->private_data;
+
+ lock_kernel();
+
+ if (file->f_mode & FMODE_WRITE) {
+ unlock_kernel();
+ drain_dac(s, file->f_flags & O_NONBLOCK);
+ lock_kernel();
+ }
+
+ mutex_lock(&s->open_mutex);
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ kfree(s->dma_dac.rawbuf);
+ s->dma_dac.rawbuf = NULL;
+ }
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ kfree(s->dma_adc.rawbuf);
+ s->dma_adc.rawbuf = NULL;
+ }
+ s->open_mode &= ((~file->f_mode) & (FMODE_READ|FMODE_WRITE));
+ mutex_unlock(&s->open_mutex);
+ wake_up(&s->open_wait);
+ unlock_kernel();
+ return 0;
+}
+
+static /*const */ struct file_operations au1550_audio_fops = {
+ owner: THIS_MODULE,
+ llseek: au1550_llseek,
+ read: au1550_read,
+ write: au1550_write,
+ poll: au1550_poll,
+ ioctl: au1550_ioctl,
+ mmap: au1550_mmap,
+ open: au1550_open,
+ release: au1550_release,
+};
+
+MODULE_AUTHOR("Advanced Micro Devices (AMD), dan@embeddededge.com");
+MODULE_DESCRIPTION("Au1550 AC97 Audio Driver");
+MODULE_LICENSE("GPL");
+
+
+static int __devinit
+au1550_probe(void)
+{
+ struct au1550_state *s = &au1550_state;
+ int val;
+
+ memset(s, 0, sizeof(struct au1550_state));
+
+ init_waitqueue_head(&s->dma_adc.wait);
+ init_waitqueue_head(&s->dma_dac.wait);
+ init_waitqueue_head(&s->open_wait);
+ mutex_init(&s->open_mutex);
+ spin_lock_init(&s->lock);
+
+ s->codec = ac97_alloc_codec();
+ if(s->codec == NULL) {
+ err("Out of memory");
+ return -1;
+ }
+ s->codec->private_data = s;
+ s->codec->id = 0;
+ s->codec->codec_read = rdcodec;
+ s->codec->codec_write = wrcodec;
+ s->codec->codec_wait = waitcodec;
+
+ if (!request_mem_region(CPHYSADDR(AC97_PSC_SEL),
+ 0x30, "Au1550 AC97")) {
+ err("AC'97 ports in use");
+ }
+
+ /* Allocate the DMA Channels
+ */
+ if ((s->dma_dac.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_MEM_CHAN,
+ DBDMA_AC97_TX_CHAN, dac_dma_interrupt, (void *)s)) == 0) {
+ err("Can't get DAC DMA");
+ goto err_dma1;
+ }
+ au1xxx_dbdma_set_devwidth(s->dma_dac.dmanr, 16);
+ if (au1xxx_dbdma_ring_alloc(s->dma_dac.dmanr,
+ NUM_DBDMA_DESCRIPTORS) == 0) {
+ err("Can't get DAC DMA descriptors");
+ goto err_dma1;
+ }
+
+ if ((s->dma_adc.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_AC97_RX_CHAN,
+ DBDMA_MEM_CHAN, adc_dma_interrupt, (void *)s)) == 0) {
+ err("Can't get ADC DMA");
+ goto err_dma2;
+ }
+ au1xxx_dbdma_set_devwidth(s->dma_adc.dmanr, 16);
+ if (au1xxx_dbdma_ring_alloc(s->dma_adc.dmanr,
+ NUM_DBDMA_DESCRIPTORS) == 0) {
+ err("Can't get ADC DMA descriptors");
+ goto err_dma2;
+ }
+
+ pr_info("DAC: DMA%d, ADC: DMA%d", DBDMA_AC97_TX_CHAN, DBDMA_AC97_RX_CHAN);
+
+ /* register devices */
+
+ if ((s->dev_audio = register_sound_dsp(&au1550_audio_fops, -1)) < 0)
+ goto err_dev1;
+ if ((s->codec->dev_mixer =
+ register_sound_mixer(&au1550_mixer_fops, -1)) < 0)
+ goto err_dev2;
+
+ /* The GPIO for the appropriate PSC was configured by the
+ * board specific start up.
+ *
+ * configure PSC for AC'97
+ */
+ au_writel(0, AC97_PSC_CTRL); /* Disable PSC */
+ au_sync();
+ au_writel((PSC_SEL_CLK_SERCLK | PSC_SEL_PS_AC97MODE), AC97_PSC_SEL);
+ au_sync();
+
+ /* cold reset the AC'97
+ */
+ au_writel(PSC_AC97RST_RST, PSC_AC97RST);
+ au_sync();
+ au1550_delay(10);
+ au_writel(0, PSC_AC97RST);
+ au_sync();
+
+ /* need to delay around 500msec(bleech) to give
+ some CODECs enough time to wakeup */
+ au1550_delay(500);
+
+ /* warm reset the AC'97 to start the bitclk
+ */
+ au_writel(PSC_AC97RST_SNC, PSC_AC97RST);
+ au_sync();
+ udelay(100);
+ au_writel(0, PSC_AC97RST);
+ au_sync();
+
+ /* Enable PSC
+ */
+ au_writel(PSC_CTRL_ENABLE, AC97_PSC_CTRL);
+ au_sync();
+
+ /* Wait for PSC ready.
+ */
+ do {
+ val = au_readl(PSC_AC97STAT);
+ au_sync();
+ } while ((val & PSC_AC97STAT_SR) == 0);
+
+ /* Configure AC97 controller.
+ * Deep FIFO, 16-bit sample, DMA, make sure DMA matches fifo size.
+ */
+ val = PSC_AC97CFG_SET_LEN(16);
+ val |= PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8;
+
+ /* Enable device so we can at least
+ * talk over the AC-link.
+ */
+ au_writel(val, PSC_AC97CFG);
+ au_writel(PSC_AC97MSK_ALLMASK, PSC_AC97MSK);
+ au_sync();
+ val |= PSC_AC97CFG_DE_ENABLE;
+ au_writel(val, PSC_AC97CFG);
+ au_sync();
+
+ /* Wait for Device ready.
+ */
+ do {
+ val = au_readl(PSC_AC97STAT);
+ au_sync();
+ } while ((val & PSC_AC97STAT_DR) == 0);
+
+ /* codec init */
+ if (!ac97_probe_codec(s->codec))
+ goto err_dev3;
+
+ s->codec_base_caps = rdcodec(s->codec, AC97_RESET);
+ s->codec_ext_caps = rdcodec(s->codec, AC97_EXTENDED_ID);
+ pr_info("AC'97 Base/Extended ID = %04x/%04x",
+ s->codec_base_caps, s->codec_ext_caps);
+
+ if (!(s->codec_ext_caps & AC97_EXTID_VRA)) {
+ /* codec does not support VRA
+ */
+ s->no_vra = 1;
+ } else if (!vra) {
+ /* Boot option says disable VRA
+ */
+ u16 ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS);
+ wrcodec(s->codec, AC97_EXTENDED_STATUS,
+ ac97_extstat & ~AC97_EXTSTAT_VRA);
+ s->no_vra = 1;
+ }
+ if (s->no_vra)
+ pr_info("no VRA, interpolating and decimating");
+
+ /* set mic to be the recording source */
+ val = SOUND_MASK_MIC;
+ mixdev_ioctl(s->codec, SOUND_MIXER_WRITE_RECSRC,
+ (unsigned long) &val);
+
+ return 0;
+
+ err_dev3:
+ unregister_sound_mixer(s->codec->dev_mixer);
+ err_dev2:
+ unregister_sound_dsp(s->dev_audio);
+ err_dev1:
+ au1xxx_dbdma_chan_free(s->dma_adc.dmanr);
+ err_dma2:
+ au1xxx_dbdma_chan_free(s->dma_dac.dmanr);
+ err_dma1:
+ release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30);
+
+ ac97_release_codec(s->codec);
+ return -1;
+}
+
+static void __devinit
+au1550_remove(void)
+{
+ struct au1550_state *s = &au1550_state;
+
+ if (!s)
+ return;
+ synchronize_irq();
+ au1xxx_dbdma_chan_free(s->dma_adc.dmanr);
+ au1xxx_dbdma_chan_free(s->dma_dac.dmanr);
+ release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30);
+ unregister_sound_dsp(s->dev_audio);
+ unregister_sound_mixer(s->codec->dev_mixer);
+ ac97_release_codec(s->codec);
+}
+
+static int __init
+init_au1550(void)
+{
+ return au1550_probe();
+}
+
+static void __exit
+cleanup_au1550(void)
+{
+ au1550_remove();
+}
+
+module_init(init_au1550);
+module_exit(cleanup_au1550);
+
+#ifndef MODULE
+
+static int __init
+au1550_setup(char *options)
+{
+ char *this_opt;
+
+ if (!options || !*options)
+ return 0;
+
+ while ((this_opt = strsep(&options, ","))) {
+ if (!*this_opt)
+ continue;
+ if (!strncmp(this_opt, "vra", 3)) {
+ vra = 1;
+ }
+ }
+
+ return 1;
+}
+
+__setup("au1550_audio=", au1550_setup);
+
+#endif /* MODULE */
diff --git a/sound/oss/audio.c b/sound/oss/audio.c
new file mode 100644
index 0000000..89bd27a
--- /dev/null
+++ b/sound/oss/audio.c
@@ -0,0 +1,983 @@
+/*
+ * sound/oss/audio.c
+ *
+ * Device file manager for /dev/audio
+ */
+
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+/*
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Thomas Sailer : moved several static variables into struct audio_operations
+ * (which is grossly misnamed btw.) because they have the same
+ * lifetime as the rest in there and dynamic allocation saves
+ * 12k or so
+ * Thomas Sailer : use more logical O_NONBLOCK semantics
+ * Daniel Rodriksson: reworked the use of the device specific copy_user
+ * still generic
+ * Horst von Brand: Add missing #include <linux/string.h>
+ * Chris Rankin : Update the module-usage counter for the coprocessor,
+ * and decrement the counters again if we cannot open
+ * the audio device.
+ */
+
+#include <linux/stddef.h>
+#include <linux/string.h>
+#include <linux/kmod.h>
+
+#include "sound_config.h"
+#include "ulaw.h"
+#include "coproc.h"
+
+#define NEUTRAL8 0x80
+#define NEUTRAL16 0x00
+
+
+static int dma_ioctl(int dev, unsigned int cmd, void __user *arg);
+
+static int set_format(int dev, int fmt)
+{
+ if (fmt != AFMT_QUERY)
+ {
+ audio_devs[dev]->local_conversion = 0;
+
+ if (!(audio_devs[dev]->format_mask & fmt)) /* Not supported */
+ {
+ if (fmt == AFMT_MU_LAW)
+ {
+ fmt = AFMT_U8;
+ audio_devs[dev]->local_conversion = CNV_MU_LAW;
+ }
+ else
+ fmt = AFMT_U8; /* This is always supported */
+ }
+ audio_devs[dev]->audio_format = audio_devs[dev]->d->set_bits(dev, fmt);
+ audio_devs[dev]->local_format = fmt;
+ }
+ else
+ return audio_devs[dev]->local_format;
+
+ if (audio_devs[dev]->local_conversion)
+ return audio_devs[dev]->local_conversion;
+ else
+ return audio_devs[dev]->local_format;
+}
+
+int audio_open(int dev, struct file *file)
+{
+ int ret;
+ int bits;
+ int dev_type = dev & 0x0f;
+ int mode = translate_mode(file);
+ const struct audio_driver *driver;
+ const struct coproc_operations *coprocessor;
+
+ dev = dev >> 4;
+
+ if (dev_type == SND_DEV_DSP16)
+ bits = 16;
+ else
+ bits = 8;
+
+ if (dev < 0 || dev >= num_audiodevs)
+ return -ENXIO;
+
+ driver = audio_devs[dev]->d;
+
+ if (!try_module_get(driver->owner))
+ return -ENODEV;
+
+ if ((ret = DMAbuf_open(dev, mode)) < 0)
+ goto error_1;
+
+ if ( (coprocessor = audio_devs[dev]->coproc) != NULL ) {
+ if (!try_module_get(coprocessor->owner))
+ goto error_2;
+
+ if ((ret = coprocessor->open(coprocessor->devc, COPR_PCM)) < 0) {
+ printk(KERN_WARNING "Sound: Can't access coprocessor device\n");
+ goto error_3;
+ }
+ }
+
+ audio_devs[dev]->local_conversion = 0;
+
+ if (dev_type == SND_DEV_AUDIO)
+ set_format(dev, AFMT_MU_LAW);
+ else
+ set_format(dev, bits);
+
+ audio_devs[dev]->audio_mode = AM_NONE;
+
+ return 0;
+
+ /*
+ * Clean-up stack: this is what needs (un)doing if
+ * we can't open the audio device ...
+ */
+ error_3:
+ module_put(coprocessor->owner);
+
+ error_2:
+ DMAbuf_release(dev, mode);
+
+ error_1:
+ module_put(driver->owner);
+
+ return ret;
+}
+
+static void sync_output(int dev)
+{
+ int p, i;
+ int l;
+ struct dma_buffparms *dmap = audio_devs[dev]->dmap_out;
+
+ if (dmap->fragment_size <= 0)
+ return;
+ dmap->flags |= DMA_POST;
+
+ /* Align the write pointer with fragment boundaries */
+
+ if ((l = dmap->user_counter % dmap->fragment_size) > 0)
+ {
+ int len;
+ unsigned long offs = dmap->user_counter % dmap->bytes_in_use;
+
+ len = dmap->fragment_size - l;
+ memset(dmap->raw_buf + offs, dmap->neutral_byte, len);
+ DMAbuf_move_wrpointer(dev, len);
+ }
+
+ /*
+ * Clean all unused buffer fragments.
+ */
+
+ p = dmap->qtail;
+ dmap->flags |= DMA_POST;
+
+ for (i = dmap->qlen + 1; i < dmap->nbufs; i++)
+ {
+ p = (p + 1) % dmap->nbufs;
+ if (((dmap->raw_buf + p * dmap->fragment_size) + dmap->fragment_size) >
+ (dmap->raw_buf + dmap->buffsize))
+ printk(KERN_ERR "audio: Buffer error 2\n");
+
+ memset(dmap->raw_buf + p * dmap->fragment_size,
+ dmap->neutral_byte,
+ dmap->fragment_size);
+ }
+
+ dmap->flags |= DMA_DIRTY;
+}
+
+void audio_release(int dev, struct file *file)
+{
+ const struct coproc_operations *coprocessor;
+ int mode = translate_mode(file);
+
+ dev = dev >> 4;
+
+ /*
+ * We do this in DMAbuf_release(). Why are we doing it
+ * here? Why don't we test the file mode before setting
+ * both flags? DMAbuf_release() does.
+ * ...pester...pester...pester...
+ */
+ audio_devs[dev]->dmap_out->closing = 1;
+ audio_devs[dev]->dmap_in->closing = 1;
+
+ /*
+ * We need to make sure we allocated the dmap_out buffer
+ * before we go mucking around with it in sync_output().
+ */
+ if (mode & OPEN_WRITE)
+ sync_output(dev);
+
+ if ( (coprocessor = audio_devs[dev]->coproc) != NULL ) {
+ coprocessor->close(coprocessor->devc, COPR_PCM);
+ module_put(coprocessor->owner);
+ }
+ DMAbuf_release(dev, mode);
+
+ module_put(audio_devs[dev]->d->owner);
+}
+
+static void translate_bytes(const unsigned char *table, unsigned char *buff, int n)
+{
+ unsigned long i;
+
+ if (n <= 0)
+ return;
+
+ for (i = 0; i < n; ++i)
+ buff[i] = table[buff[i]];
+}
+
+int audio_write(int dev, struct file *file, const char __user *buf, int count)
+{
+ int c, p, l, buf_size, used, returned;
+ int err;
+ char *dma_buf;
+
+ dev = dev >> 4;
+
+ p = 0;
+ c = count;
+
+ if(count < 0)
+ return -EINVAL;
+
+ if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return -EPERM;
+
+ if (audio_devs[dev]->flags & DMA_DUPLEX)
+ audio_devs[dev]->audio_mode |= AM_WRITE;
+ else
+ audio_devs[dev]->audio_mode = AM_WRITE;
+
+ if (!count) /* Flush output */
+ {
+ sync_output(dev);
+ return 0;
+ }
+
+ while (c)
+ {
+ if ((err = DMAbuf_getwrbuffer(dev, &dma_buf, &buf_size, !!(file->f_flags & O_NONBLOCK))) < 0)
+ {
+ /* Handle nonblocking mode */
+ if ((file->f_flags & O_NONBLOCK) && err == -EAGAIN)
+ return p? p : -EAGAIN; /* No more space. Return # of accepted bytes */
+ return err;
+ }
+ l = c;
+
+ if (l > buf_size)
+ l = buf_size;
+
+ returned = l;
+ used = l;
+ if (!audio_devs[dev]->d->copy_user)
+ {
+ if ((dma_buf + l) >
+ (audio_devs[dev]->dmap_out->raw_buf + audio_devs[dev]->dmap_out->buffsize))
+ {
+ printk(KERN_ERR "audio: Buffer error 3 (%lx,%d), (%lx, %d)\n", (long) dma_buf, l, (long) audio_devs[dev]->dmap_out->raw_buf, (int) audio_devs[dev]->dmap_out->buffsize);
+ return -EDOM;
+ }
+ if (dma_buf < audio_devs[dev]->dmap_out->raw_buf)
+ {
+ printk(KERN_ERR "audio: Buffer error 13 (%lx<%lx)\n", (long) dma_buf, (long) audio_devs[dev]->dmap_out->raw_buf);
+ return -EDOM;
+ }
+ if(copy_from_user(dma_buf, &(buf)[p], l))
+ return -EFAULT;
+ }
+ else audio_devs[dev]->d->copy_user (dev,
+ dma_buf, 0,
+ buf, p,
+ c, buf_size,
+ &used, &returned,
+ l);
+ l = returned;
+
+ if (audio_devs[dev]->local_conversion & CNV_MU_LAW)
+ {
+ translate_bytes(ulaw_dsp, (unsigned char *) dma_buf, l);
+ }
+ c -= used;
+ p += used;
+ DMAbuf_move_wrpointer(dev, l);
+
+ }
+
+ return count;
+}
+
+int audio_read(int dev, struct file *file, char __user *buf, int count)
+{
+ int c, p, l;
+ char *dmabuf;
+ int buf_no;
+
+ dev = dev >> 4;
+ p = 0;
+ c = count;
+
+ if (!(audio_devs[dev]->open_mode & OPEN_READ))
+ return -EPERM;
+
+ if ((audio_devs[dev]->audio_mode & AM_WRITE) && !(audio_devs[dev]->flags & DMA_DUPLEX))
+ sync_output(dev);
+
+ if (audio_devs[dev]->flags & DMA_DUPLEX)
+ audio_devs[dev]->audio_mode |= AM_READ;
+ else
+ audio_devs[dev]->audio_mode = AM_READ;
+
+ while(c)
+ {
+ if ((buf_no = DMAbuf_getrdbuffer(dev, &dmabuf, &l, !!(file->f_flags & O_NONBLOCK))) < 0)
+ {
+ /*
+ * Nonblocking mode handling. Return current # of bytes
+ */
+
+ if (p > 0) /* Avoid throwing away data */
+ return p; /* Return it instead */
+
+ if ((file->f_flags & O_NONBLOCK) && buf_no == -EAGAIN)
+ return -EAGAIN;
+
+ return buf_no;
+ }
+ if (l > c)
+ l = c;
+
+ /*
+ * Insert any local processing here.
+ */
+
+ if (audio_devs[dev]->local_conversion & CNV_MU_LAW)
+ {
+ translate_bytes(dsp_ulaw, (unsigned char *) dmabuf, l);
+ }
+
+ {
+ char *fixit = dmabuf;
+
+ if(copy_to_user(&(buf)[p], fixit, l))
+ return -EFAULT;
+ };
+
+ DMAbuf_rmchars(dev, buf_no, l);
+
+ p += l;
+ c -= l;
+ }
+
+ return count - c;
+}
+
+int audio_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg)
+{
+ int val, count;
+ unsigned long flags;
+ struct dma_buffparms *dmap;
+ int __user *p = arg;
+
+ dev = dev >> 4;
+
+ if (_IOC_TYPE(cmd) == 'C') {
+ if (audio_devs[dev]->coproc) /* Coprocessor ioctl */
+ return audio_devs[dev]->coproc->ioctl(audio_devs[dev]->coproc->devc, cmd, arg, 0);
+ /* else
+ printk(KERN_DEBUG"/dev/dsp%d: No coprocessor for this device\n", dev); */
+ return -ENXIO;
+ }
+ else switch (cmd)
+ {
+ case SNDCTL_DSP_SYNC:
+ if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return 0;
+ if (audio_devs[dev]->dmap_out->fragment_size == 0)
+ return 0;
+ sync_output(dev);
+ DMAbuf_sync(dev);
+ DMAbuf_reset(dev);
+ return 0;
+
+ case SNDCTL_DSP_POST:
+ if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return 0;
+ if (audio_devs[dev]->dmap_out->fragment_size == 0)
+ return 0;
+ audio_devs[dev]->dmap_out->flags |= DMA_POST | DMA_DIRTY;
+ sync_output(dev);
+ dma_ioctl(dev, SNDCTL_DSP_POST, NULL);
+ return 0;
+
+ case SNDCTL_DSP_RESET:
+ audio_devs[dev]->audio_mode = AM_NONE;
+ DMAbuf_reset(dev);
+ return 0;
+
+ case SNDCTL_DSP_GETFMTS:
+ val = audio_devs[dev]->format_mask | AFMT_MU_LAW;
+ break;
+
+ case SNDCTL_DSP_SETFMT:
+ if (get_user(val, p))
+ return -EFAULT;
+ val = set_format(dev, val);
+ break;
+
+ case SNDCTL_DSP_GETISPACE:
+ if (!(audio_devs[dev]->open_mode & OPEN_READ))
+ return 0;
+ if ((audio_devs[dev]->audio_mode & AM_WRITE) && !(audio_devs[dev]->flags & DMA_DUPLEX))
+ return -EBUSY;
+ return dma_ioctl(dev, cmd, arg);
+
+ case SNDCTL_DSP_GETOSPACE:
+ if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return -EPERM;
+ if ((audio_devs[dev]->audio_mode & AM_READ) && !(audio_devs[dev]->flags & DMA_DUPLEX))
+ return -EBUSY;
+ return dma_ioctl(dev, cmd, arg);
+
+ case SNDCTL_DSP_NONBLOCK:
+ file->f_flags |= O_NONBLOCK;
+ return 0;
+
+ case SNDCTL_DSP_GETCAPS:
+ val = 1 | DSP_CAP_MMAP; /* Revision level of this ioctl() */
+ if (audio_devs[dev]->flags & DMA_DUPLEX &&
+ audio_devs[dev]->open_mode == OPEN_READWRITE)
+ val |= DSP_CAP_DUPLEX;
+ if (audio_devs[dev]->coproc)
+ val |= DSP_CAP_COPROC;
+ if (audio_devs[dev]->d->local_qlen) /* Device has hidden buffers */
+ val |= DSP_CAP_BATCH;
+ if (audio_devs[dev]->d->trigger) /* Supports SETTRIGGER */
+ val |= DSP_CAP_TRIGGER;
+ break;
+
+ case SOUND_PCM_WRITE_RATE:
+ if (get_user(val, p))
+ return -EFAULT;
+ val = audio_devs[dev]->d->set_speed(dev, val);
+ break;
+
+ case SOUND_PCM_READ_RATE:
+ val = audio_devs[dev]->d->set_speed(dev, 0);
+ break;
+
+ case SNDCTL_DSP_STEREO:
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val > 1 || val < 0)
+ return -EINVAL;
+ val = audio_devs[dev]->d->set_channels(dev, val + 1) - 1;
+ break;
+
+ case SOUND_PCM_WRITE_CHANNELS:
+ if (get_user(val, p))
+ return -EFAULT;
+ val = audio_devs[dev]->d->set_channels(dev, val);
+ break;
+
+ case SOUND_PCM_READ_CHANNELS:
+ val = audio_devs[dev]->d->set_channels(dev, 0);
+ break;
+
+ case SOUND_PCM_READ_BITS:
+ val = audio_devs[dev]->d->set_bits(dev, 0);
+ break;
+
+ case SNDCTL_DSP_SETDUPLEX:
+ if (audio_devs[dev]->open_mode != OPEN_READWRITE)
+ return -EPERM;
+ return (audio_devs[dev]->flags & DMA_DUPLEX) ? 0 : -EIO;
+
+ case SNDCTL_DSP_PROFILE:
+ if (get_user(val, p))
+ return -EFAULT;
+ if (audio_devs[dev]->open_mode & OPEN_WRITE)
+ audio_devs[dev]->dmap_out->applic_profile = val;
+ if (audio_devs[dev]->open_mode & OPEN_READ)
+ audio_devs[dev]->dmap_in->applic_profile = val;
+ return 0;
+
+ case SNDCTL_DSP_GETODELAY:
+ dmap = audio_devs[dev]->dmap_out;
+ if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return -EINVAL;
+ if (!(dmap->flags & DMA_ALLOC_DONE))
+ {
+ val=0;
+ break;
+ }
+
+ spin_lock_irqsave(&dmap->lock,flags);
+ /* Compute number of bytes that have been played */
+ count = DMAbuf_get_buffer_pointer (dev, dmap, DMODE_OUTPUT);
+ if (count < dmap->fragment_size && dmap->qhead != 0)
+ count += dmap->bytes_in_use; /* Pointer wrap not handled yet */
+ count += dmap->byte_counter;
+
+ /* Substract current count from the number of bytes written by app */
+ count = dmap->user_counter - count;
+ if (count < 0)
+ count = 0;
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ val = count;
+ break;
+
+ default:
+ return dma_ioctl(dev, cmd, arg);
+ }
+ return put_user(val, p);
+}
+
+void audio_init_devices(void)
+{
+ /*
+ * NOTE! This routine could be called several times during boot.
+ */
+}
+
+void reorganize_buffers(int dev, struct dma_buffparms *dmap, int recording)
+{
+ /*
+ * This routine breaks the physical device buffers to logical ones.
+ */
+
+ struct audio_operations *dsp_dev = audio_devs[dev];
+
+ unsigned i, n;
+ unsigned sr, nc, sz, bsz;
+
+ sr = dsp_dev->d->set_speed(dev, 0);
+ nc = dsp_dev->d->set_channels(dev, 0);
+ sz = dsp_dev->d->set_bits(dev, 0);
+
+ if (sz == 8)
+ dmap->neutral_byte = NEUTRAL8;
+ else
+ dmap->neutral_byte = NEUTRAL16;
+
+ if (sr < 1 || nc < 1 || sz < 1)
+ {
+/* printk(KERN_DEBUG "Warning: Invalid PCM parameters[%d] sr=%d, nc=%d, sz=%d\n", dev, sr, nc, sz);*/
+ sr = DSP_DEFAULT_SPEED;
+ nc = 1;
+ sz = 8;
+ }
+
+ sz = sr * nc * sz;
+
+ sz /= 8; /* #bits -> #bytes */
+ dmap->data_rate = sz;
+
+ if (!dmap->needs_reorg)
+ return;
+ dmap->needs_reorg = 0;
+
+ if (dmap->fragment_size == 0)
+ {
+ /* Compute the fragment size using the default algorithm */
+
+ /*
+ * Compute a buffer size for time not exceeding 1 second.
+ * Usually this algorithm gives a buffer size for 0.5 to 1.0 seconds
+ * of sound (using the current speed, sample size and #channels).
+ */
+
+ bsz = dmap->buffsize;
+ while (bsz > sz)
+ bsz /= 2;
+
+ if (bsz == dmap->buffsize)
+ bsz /= 2; /* Needs at least 2 buffers */
+
+ /*
+ * Split the computed fragment to smaller parts. After 3.5a9
+ * the default subdivision is 4 which should give better
+ * results when recording.
+ */
+
+ if (dmap->subdivision == 0) /* Not already set */
+ {
+ dmap->subdivision = 4; /* Init to the default value */
+
+ if ((bsz / dmap->subdivision) > 4096)
+ dmap->subdivision *= 2;
+ if ((bsz / dmap->subdivision) < 4096)
+ dmap->subdivision = 1;
+ }
+ bsz /= dmap->subdivision;
+
+ if (bsz < 16)
+ bsz = 16; /* Just a sanity check */
+
+ dmap->fragment_size = bsz;
+ }
+ else
+ {
+ /*
+ * The process has specified the buffer size with SNDCTL_DSP_SETFRAGMENT or
+ * the buffer size computation has already been done.
+ */
+ if (dmap->fragment_size > (dmap->buffsize / 2))
+ dmap->fragment_size = (dmap->buffsize / 2);
+ bsz = dmap->fragment_size;
+ }
+
+ if (audio_devs[dev]->min_fragment)
+ if (bsz < (1 << audio_devs[dev]->min_fragment))
+ bsz = 1 << audio_devs[dev]->min_fragment;
+ if (audio_devs[dev]->max_fragment)
+ if (bsz > (1 << audio_devs[dev]->max_fragment))
+ bsz = 1 << audio_devs[dev]->max_fragment;
+ bsz &= ~0x07; /* Force size which is multiple of 8 bytes */
+#ifdef OS_DMA_ALIGN_CHECK
+ OS_DMA_ALIGN_CHECK(bsz);
+#endif
+
+ n = dmap->buffsize / bsz;
+ if (n > MAX_SUB_BUFFERS)
+ n = MAX_SUB_BUFFERS;
+ if (n > dmap->max_fragments)
+ n = dmap->max_fragments;
+
+ if (n < 2)
+ {
+ n = 2;
+ bsz /= 2;
+ }
+ dmap->nbufs = n;
+ dmap->bytes_in_use = n * bsz;
+ dmap->fragment_size = bsz;
+ dmap->max_byte_counter = (dmap->data_rate * 60 * 60) +
+ dmap->bytes_in_use; /* Approximately one hour */
+
+ if (dmap->raw_buf)
+ {
+ memset(dmap->raw_buf, dmap->neutral_byte, dmap->bytes_in_use);
+ }
+
+ for (i = 0; i < dmap->nbufs; i++)
+ {
+ dmap->counts[i] = 0;
+ }
+
+ dmap->flags |= DMA_ALLOC_DONE | DMA_EMPTY;
+}
+
+static int dma_subdivide(int dev, struct dma_buffparms *dmap, int fact)
+{
+ if (fact == 0)
+ {
+ fact = dmap->subdivision;
+ if (fact == 0)
+ fact = 1;
+ return fact;
+ }
+ if (dmap->subdivision != 0 || dmap->fragment_size) /* Too late to change */
+ return -EINVAL;
+
+ if (fact > MAX_REALTIME_FACTOR)
+ return -EINVAL;
+
+ if (fact != 1 && fact != 2 && fact != 4 && fact != 8 && fact != 16)
+ return -EINVAL;
+
+ dmap->subdivision = fact;
+ return fact;
+}
+
+static int dma_set_fragment(int dev, struct dma_buffparms *dmap, int fact)
+{
+ int bytes, count;
+
+ if (fact == 0)
+ return -EIO;
+
+ if (dmap->subdivision != 0 ||
+ dmap->fragment_size) /* Too late to change */
+ return -EINVAL;
+
+ bytes = fact & 0xffff;
+ count = (fact >> 16) & 0x7fff;
+
+ if (count == 0)
+ count = MAX_SUB_BUFFERS;
+ else if (count < MAX_SUB_BUFFERS)
+ count++;
+
+ if (bytes < 4 || bytes > 17) /* <16 || > 512k */
+ return -EINVAL;
+
+ if (count < 2)
+ return -EINVAL;
+
+ if (audio_devs[dev]->min_fragment > 0)
+ if (bytes < audio_devs[dev]->min_fragment)
+ bytes = audio_devs[dev]->min_fragment;
+
+ if (audio_devs[dev]->max_fragment > 0)
+ if (bytes > audio_devs[dev]->max_fragment)
+ bytes = audio_devs[dev]->max_fragment;
+
+#ifdef OS_DMA_MINBITS
+ if (bytes < OS_DMA_MINBITS)
+ bytes = OS_DMA_MINBITS;
+#endif
+
+ dmap->fragment_size = (1 << bytes);
+ dmap->max_fragments = count;
+
+ if (dmap->fragment_size > dmap->buffsize)
+ dmap->fragment_size = dmap->buffsize;
+
+ if (dmap->fragment_size == dmap->buffsize &&
+ audio_devs[dev]->flags & DMA_AUTOMODE)
+ dmap->fragment_size /= 2; /* Needs at least 2 buffers */
+
+ dmap->subdivision = 1; /* Disable SNDCTL_DSP_SUBDIVIDE */
+ return bytes | ((count - 1) << 16);
+}
+
+static int dma_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ struct dma_buffparms *dmap_out = audio_devs[dev]->dmap_out;
+ struct dma_buffparms *dmap_in = audio_devs[dev]->dmap_in;
+ struct dma_buffparms *dmap;
+ audio_buf_info info;
+ count_info cinfo;
+ int fact, ret, changed, bits, count, err;
+ unsigned long flags;
+
+ switch (cmd)
+ {
+ case SNDCTL_DSP_SUBDIVIDE:
+ ret = 0;
+ if (get_user(fact, (int __user *)arg))
+ return -EFAULT;
+ if (audio_devs[dev]->open_mode & OPEN_WRITE)
+ ret = dma_subdivide(dev, dmap_out, fact);
+ if (ret < 0)
+ return ret;
+ if (audio_devs[dev]->open_mode != OPEN_WRITE ||
+ (audio_devs[dev]->flags & DMA_DUPLEX &&
+ audio_devs[dev]->open_mode & OPEN_READ))
+ ret = dma_subdivide(dev, dmap_in, fact);
+ if (ret < 0)
+ return ret;
+ break;
+
+ case SNDCTL_DSP_GETISPACE:
+ case SNDCTL_DSP_GETOSPACE:
+ dmap = dmap_out;
+ if (cmd == SNDCTL_DSP_GETISPACE && !(audio_devs[dev]->open_mode & OPEN_READ))
+ return -EINVAL;
+ if (cmd == SNDCTL_DSP_GETOSPACE && !(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return -EINVAL;
+ if (cmd == SNDCTL_DSP_GETISPACE && audio_devs[dev]->flags & DMA_DUPLEX)
+ dmap = dmap_in;
+ if (dmap->mapping_flags & DMA_MAP_MAPPED)
+ return -EINVAL;
+ if (!(dmap->flags & DMA_ALLOC_DONE))
+ reorganize_buffers(dev, dmap, (cmd == SNDCTL_DSP_GETISPACE));
+ info.fragstotal = dmap->nbufs;
+ if (cmd == SNDCTL_DSP_GETISPACE)
+ info.fragments = dmap->qlen;
+ else
+ {
+ if (!DMAbuf_space_in_queue(dev))
+ info.fragments = 0;
+ else
+ {
+ info.fragments = DMAbuf_space_in_queue(dev);
+ if (audio_devs[dev]->d->local_qlen)
+ {
+ int tmp = audio_devs[dev]->d->local_qlen(dev);
+ if (tmp && info.fragments)
+ tmp--; /*
+ * This buffer has been counted twice
+ */
+ info.fragments -= tmp;
+ }
+ }
+ }
+ if (info.fragments < 0)
+ info.fragments = 0;
+ else if (info.fragments > dmap->nbufs)
+ info.fragments = dmap->nbufs;
+
+ info.fragsize = dmap->fragment_size;
+ info.bytes = info.fragments * dmap->fragment_size;
+
+ if (cmd == SNDCTL_DSP_GETISPACE && dmap->qlen)
+ info.bytes -= dmap->counts[dmap->qhead];
+ else
+ {
+ info.fragments = info.bytes / dmap->fragment_size;
+ info.bytes -= dmap->user_counter % dmap->fragment_size;
+ }
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_SETTRIGGER:
+ if (get_user(bits, (int __user *)arg))
+ return -EFAULT;
+ bits &= audio_devs[dev]->open_mode;
+ if (audio_devs[dev]->d->trigger == NULL)
+ return -EINVAL;
+ if (!(audio_devs[dev]->flags & DMA_DUPLEX) && (bits & PCM_ENABLE_INPUT) &&
+ (bits & PCM_ENABLE_OUTPUT))
+ return -EINVAL;
+
+ if (bits & PCM_ENABLE_INPUT)
+ {
+ spin_lock_irqsave(&dmap_in->lock,flags);
+ changed = (audio_devs[dev]->enable_bits ^ bits) & PCM_ENABLE_INPUT;
+ if (changed && audio_devs[dev]->go)
+ {
+ reorganize_buffers(dev, dmap_in, 1);
+ if ((err = audio_devs[dev]->d->prepare_for_input(dev,
+ dmap_in->fragment_size, dmap_in->nbufs)) < 0) {
+ spin_unlock_irqrestore(&dmap_in->lock,flags);
+ return -err;
+ }
+ dmap_in->dma_mode = DMODE_INPUT;
+ audio_devs[dev]->enable_bits |= PCM_ENABLE_INPUT;
+ DMAbuf_activate_recording(dev, dmap_in);
+ } else
+ audio_devs[dev]->enable_bits &= ~PCM_ENABLE_INPUT;
+ spin_unlock_irqrestore(&dmap_in->lock,flags);
+ }
+ if (bits & PCM_ENABLE_OUTPUT)
+ {
+ spin_lock_irqsave(&dmap_out->lock,flags);
+ changed = (audio_devs[dev]->enable_bits ^ bits) & PCM_ENABLE_OUTPUT;
+ if (changed &&
+ (dmap_out->mapping_flags & DMA_MAP_MAPPED || dmap_out->qlen > 0) &&
+ audio_devs[dev]->go)
+ {
+ if (!(dmap_out->flags & DMA_ALLOC_DONE))
+ reorganize_buffers(dev, dmap_out, 0);
+ dmap_out->dma_mode = DMODE_OUTPUT;
+ audio_devs[dev]->enable_bits |= PCM_ENABLE_OUTPUT;
+ dmap_out->counts[dmap_out->qhead] = dmap_out->fragment_size;
+ DMAbuf_launch_output(dev, dmap_out);
+ } else
+ audio_devs[dev]->enable_bits &= ~PCM_ENABLE_OUTPUT;
+ spin_unlock_irqrestore(&dmap_out->lock,flags);
+ }
+#if 0
+ if (changed && audio_devs[dev]->d->trigger)
+ audio_devs[dev]->d->trigger(dev, bits * audio_devs[dev]->go);
+#endif
+ /* Falls through... */
+
+ case SNDCTL_DSP_GETTRIGGER:
+ ret = audio_devs[dev]->enable_bits;
+ break;
+
+ case SNDCTL_DSP_SETSYNCRO:
+ if (!audio_devs[dev]->d->trigger)
+ return -EINVAL;
+ audio_devs[dev]->d->trigger(dev, 0);
+ audio_devs[dev]->go = 0;
+ return 0;
+
+ case SNDCTL_DSP_GETIPTR:
+ if (!(audio_devs[dev]->open_mode & OPEN_READ))
+ return -EINVAL;
+ spin_lock_irqsave(&dmap_in->lock,flags);
+ cinfo.bytes = dmap_in->byte_counter;
+ cinfo.ptr = DMAbuf_get_buffer_pointer(dev, dmap_in, DMODE_INPUT) & ~3;
+ if (cinfo.ptr < dmap_in->fragment_size && dmap_in->qtail != 0)
+ cinfo.bytes += dmap_in->bytes_in_use; /* Pointer wrap not handled yet */
+ cinfo.blocks = dmap_in->qlen;
+ cinfo.bytes += cinfo.ptr;
+ if (dmap_in->mapping_flags & DMA_MAP_MAPPED)
+ dmap_in->qlen = 0; /* Reset interrupt counter */
+ spin_unlock_irqrestore(&dmap_in->lock,flags);
+ if (copy_to_user(arg, &cinfo, sizeof(cinfo)))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_GETOPTR:
+ if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return -EINVAL;
+
+ spin_lock_irqsave(&dmap_out->lock,flags);
+ cinfo.bytes = dmap_out->byte_counter;
+ cinfo.ptr = DMAbuf_get_buffer_pointer(dev, dmap_out, DMODE_OUTPUT) & ~3;
+ if (cinfo.ptr < dmap_out->fragment_size && dmap_out->qhead != 0)
+ cinfo.bytes += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */
+ cinfo.blocks = dmap_out->qlen;
+ cinfo.bytes += cinfo.ptr;
+ if (dmap_out->mapping_flags & DMA_MAP_MAPPED)
+ dmap_out->qlen = 0; /* Reset interrupt counter */
+ spin_unlock_irqrestore(&dmap_out->lock,flags);
+ if (copy_to_user(arg, &cinfo, sizeof(cinfo)))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_GETODELAY:
+ if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
+ return -EINVAL;
+ if (!(dmap_out->flags & DMA_ALLOC_DONE))
+ {
+ ret=0;
+ break;
+ }
+ spin_lock_irqsave(&dmap_out->lock,flags);
+ /* Compute number of bytes that have been played */
+ count = DMAbuf_get_buffer_pointer (dev, dmap_out, DMODE_OUTPUT);
+ if (count < dmap_out->fragment_size && dmap_out->qhead != 0)
+ count += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */
+ count += dmap_out->byte_counter;
+ /* Substract current count from the number of bytes written by app */
+ count = dmap_out->user_counter - count;
+ if (count < 0)
+ count = 0;
+ spin_unlock_irqrestore(&dmap_out->lock,flags);
+ ret = count;
+ break;
+
+ case SNDCTL_DSP_POST:
+ if (audio_devs[dev]->dmap_out->qlen > 0)
+ if (!(audio_devs[dev]->dmap_out->flags & DMA_ACTIVE))
+ DMAbuf_launch_output(dev, audio_devs[dev]->dmap_out);
+ return 0;
+
+ case SNDCTL_DSP_GETBLKSIZE:
+ dmap = dmap_out;
+ if (audio_devs[dev]->open_mode & OPEN_WRITE)
+ reorganize_buffers(dev, dmap_out, (audio_devs[dev]->open_mode == OPEN_READ));
+ if (audio_devs[dev]->open_mode == OPEN_READ ||
+ (audio_devs[dev]->flags & DMA_DUPLEX &&
+ audio_devs[dev]->open_mode & OPEN_READ))
+ reorganize_buffers(dev, dmap_in, (audio_devs[dev]->open_mode == OPEN_READ));
+ if (audio_devs[dev]->open_mode == OPEN_READ)
+ dmap = dmap_in;
+ ret = dmap->fragment_size;
+ break;
+
+ case SNDCTL_DSP_SETFRAGMENT:
+ ret = 0;
+ if (get_user(fact, (int __user *)arg))
+ return -EFAULT;
+ if (audio_devs[dev]->open_mode & OPEN_WRITE)
+ ret = dma_set_fragment(dev, dmap_out, fact);
+ if (ret < 0)
+ return ret;
+ if (audio_devs[dev]->open_mode == OPEN_READ ||
+ (audio_devs[dev]->flags & DMA_DUPLEX &&
+ audio_devs[dev]->open_mode & OPEN_READ))
+ ret = dma_set_fragment(dev, dmap_in, fact);
+ if (ret < 0)
+ return ret;
+ if (!arg) /* don't know what this is good for, but preserve old semantics */
+ return 0;
+ break;
+
+ default:
+ if (!audio_devs[dev]->d->ioctl)
+ return -EINVAL;
+ return audio_devs[dev]->d->ioctl(dev, cmd, arg);
+ }
+ return put_user(ret, (int __user *)arg);
+}
diff --git a/sound/oss/bin2hex.c b/sound/oss/bin2hex.c
new file mode 100644
index 0000000..b59109e
--- /dev/null
+++ b/sound/oss/bin2hex.c
@@ -0,0 +1,39 @@
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+
+int main( int argc, const char * argv [] )
+{
+ const char * varname;
+ int i = 0;
+ int c;
+ int id = 0;
+
+ if(argv[1] && strcmp(argv[1],"-i")==0)
+ {
+ argv++;
+ argc--;
+ id=1;
+ }
+
+ if(argc==1)
+ {
+ fprintf(stderr, "bin2hex: [-i] firmware\n");
+ exit(1);
+ }
+
+ varname = argv[1];
+ printf( "/* automatically generated by bin2hex */\n" );
+ printf( "static unsigned char %s [] %s =\n{\n", varname , id?"__initdata":"");
+
+ while ( ( c = getchar( ) ) != EOF )
+ {
+ if ( i != 0 && i % 10 == 0 )
+ printf( "\n" );
+ printf( "0x%02lx,", c & 0xFFl );
+ i++;
+ }
+
+ printf( "};\nstatic int %sLen = %d;\n", varname, i );
+ return 0;
+}
diff --git a/sound/oss/coproc.h b/sound/oss/coproc.h
new file mode 100644
index 0000000..7306346
--- /dev/null
+++ b/sound/oss/coproc.h
@@ -0,0 +1,12 @@
+/*
+ * Definitions for various on board processors on the sound cards. For
+ * example DSP processors.
+ */
+
+/*
+ * Coprocessor access types
+ */
+#define COPR_CUSTOM 0x0001 /* Custom applications */
+#define COPR_MIDI 0x0002 /* MIDI (MPU-401) emulation */
+#define COPR_PCM 0x0004 /* Digitized voice applications */
+#define COPR_SYNTH 0x0008 /* Music synthesis */
diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c
new file mode 100644
index 0000000..08274c9
--- /dev/null
+++ b/sound/oss/dev_table.c
@@ -0,0 +1,256 @@
+/*
+ * sound/oss/dev_table.c
+ *
+ * Device call tables.
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+
+#include <linux/init.h>
+
+#include "sound_config.h"
+
+struct audio_operations *audio_devs[MAX_AUDIO_DEV];
+EXPORT_SYMBOL(audio_devs);
+
+int num_audiodevs;
+EXPORT_SYMBOL(num_audiodevs);
+
+struct mixer_operations *mixer_devs[MAX_MIXER_DEV];
+EXPORT_SYMBOL(mixer_devs);
+
+int num_mixers;
+EXPORT_SYMBOL(num_mixers);
+
+struct synth_operations *synth_devs[MAX_SYNTH_DEV+MAX_MIDI_DEV];
+EXPORT_SYMBOL(synth_devs);
+
+int num_synths;
+
+struct midi_operations *midi_devs[MAX_MIDI_DEV];
+EXPORT_SYMBOL(midi_devs);
+
+int num_midis;
+EXPORT_SYMBOL(num_midis);
+
+struct sound_timer_operations *sound_timer_devs[MAX_TIMER_DEV] = {
+ &default_sound_timer, NULL
+};
+EXPORT_SYMBOL(sound_timer_devs);
+
+int num_sound_timers = 1;
+
+
+static int sound_alloc_audiodev(void);
+
+int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
+ int driver_size, int flags, unsigned int format_mask,
+ void *devc, int dma1, int dma2)
+{
+ struct audio_driver *d;
+ struct audio_operations *op;
+ int num;
+
+ if (vers != AUDIO_DRIVER_VERSION || driver_size > sizeof(struct audio_driver)) {
+ printk(KERN_ERR "Sound: Incompatible audio driver for %s\n", name);
+ return -(EINVAL);
+ }
+ num = sound_alloc_audiodev();
+
+ if (num == -1) {
+ printk(KERN_ERR "sound: Too many audio drivers\n");
+ return -(EBUSY);
+ }
+ d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver)));
+
+ if (sound_nblocks < 1024)
+ sound_nblocks++;
+
+ op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_operations)));
+
+ if (sound_nblocks < 1024)
+ sound_nblocks++;
+ if (d == NULL || op == NULL) {
+ printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name);
+ sound_unload_audiodev(num);
+ return -(ENOMEM);
+ }
+ memset((char *) op, 0, sizeof(struct audio_operations));
+ init_waitqueue_head(&op->in_sleeper);
+ init_waitqueue_head(&op->out_sleeper);
+ init_waitqueue_head(&op->poll_sleeper);
+ if (driver_size < sizeof(struct audio_driver))
+ memset((char *) d, 0, sizeof(struct audio_driver));
+
+ memcpy((char *) d, (char *) driver, driver_size);
+
+ op->d = d;
+ strlcpy(op->name, name, sizeof(op->name));
+ op->flags = flags;
+ op->format_mask = format_mask;
+ op->devc = devc;
+
+ /*
+ * Hardcoded defaults
+ */
+ audio_devs[num] = op;
+
+ DMAbuf_init(num, dma1, dma2);
+
+ audio_init_devices();
+ return num;
+}
+EXPORT_SYMBOL(sound_install_audiodrv);
+
+int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
+ int driver_size, void *devc)
+{
+ struct mixer_operations *op;
+
+ int n = sound_alloc_mixerdev();
+
+ if (n == -1) {
+ printk(KERN_ERR "Sound: Too many mixer drivers\n");
+ return -EBUSY;
+ }
+ if (vers != MIXER_DRIVER_VERSION ||
+ driver_size > sizeof(struct mixer_operations)) {
+ printk(KERN_ERR "Sound: Incompatible mixer driver for %s\n", name);
+ return -EINVAL;
+ }
+
+ /* FIXME: This leaks a mixer_operations struct every time its called
+ until you unload sound! */
+
+ op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct mixer_operations)));
+
+ if (sound_nblocks < 1024)
+ sound_nblocks++;
+ if (op == NULL) {
+ printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
+ return -ENOMEM;
+ }
+ memset((char *) op, 0, sizeof(struct mixer_operations));
+ memcpy((char *) op, (char *) driver, driver_size);
+
+ strlcpy(op->name, name, sizeof(op->name));
+ op->devc = devc;
+
+ mixer_devs[n] = op;
+ return n;
+}
+EXPORT_SYMBOL(sound_install_mixer);
+
+void sound_unload_audiodev(int dev)
+{
+ if (dev != -1) {
+ DMAbuf_deinit(dev);
+ audio_devs[dev] = NULL;
+ unregister_sound_dsp((dev<<4)+3);
+ }
+}
+EXPORT_SYMBOL(sound_unload_audiodev);
+
+static int sound_alloc_audiodev(void)
+{
+ int i = register_sound_dsp(&oss_sound_fops, -1);
+ if(i==-1)
+ return i;
+ i>>=4;
+ if(i>=num_audiodevs)
+ num_audiodevs = i + 1;
+ return i;
+}
+
+int sound_alloc_mididev(void)
+{
+ int i = register_sound_midi(&oss_sound_fops, -1);
+ if(i==-1)
+ return i;
+ i>>=4;
+ if(i>=num_midis)
+ num_midis = i + 1;
+ return i;
+}
+EXPORT_SYMBOL(sound_alloc_mididev);
+
+int sound_alloc_synthdev(void)
+{
+ int i;
+
+ for (i = 0; i < MAX_SYNTH_DEV; i++) {
+ if (synth_devs[i] == NULL) {
+ if (i >= num_synths)
+ num_synths++;
+ return i;
+ }
+ }
+ return -1;
+}
+EXPORT_SYMBOL(sound_alloc_synthdev);
+
+int sound_alloc_mixerdev(void)
+{
+ int i = register_sound_mixer(&oss_sound_fops, -1);
+ if(i==-1)
+ return -1;
+ i>>=4;
+ if(i>=num_mixers)
+ num_mixers = i + 1;
+ return i;
+}
+EXPORT_SYMBOL(sound_alloc_mixerdev);
+
+int sound_alloc_timerdev(void)
+{
+ int i;
+
+ for (i = 0; i < MAX_TIMER_DEV; i++) {
+ if (sound_timer_devs[i] == NULL) {
+ if (i >= num_sound_timers)
+ num_sound_timers++;
+ return i;
+ }
+ }
+ return -1;
+}
+EXPORT_SYMBOL(sound_alloc_timerdev);
+
+void sound_unload_mixerdev(int dev)
+{
+ if (dev != -1) {
+ mixer_devs[dev] = NULL;
+ unregister_sound_mixer(dev<<4);
+ num_mixers--;
+ }
+}
+EXPORT_SYMBOL(sound_unload_mixerdev);
+
+void sound_unload_mididev(int dev)
+{
+ if (dev != -1) {
+ midi_devs[dev] = NULL;
+ unregister_sound_midi((dev<<4)+2);
+ }
+}
+EXPORT_SYMBOL(sound_unload_mididev);
+
+void sound_unload_synthdev(int dev)
+{
+ if (dev != -1)
+ synth_devs[dev] = NULL;
+}
+EXPORT_SYMBOL(sound_unload_synthdev);
+
+void sound_unload_timerdev(int dev)
+{
+ if (dev != -1)
+ sound_timer_devs[dev] = NULL;
+}
+EXPORT_SYMBOL(sound_unload_timerdev);
+
diff --git a/sound/oss/dev_table.h b/sound/oss/dev_table.h
new file mode 100644
index 0000000..b7617be
--- /dev/null
+++ b/sound/oss/dev_table.h
@@ -0,0 +1,390 @@
+/*
+ * dev_table.h
+ *
+ * Global definitions for device call tables
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+
+
+#ifndef _DEV_TABLE_H_
+#define _DEV_TABLE_H_
+
+#include <linux/spinlock.h>
+/*
+ * Sound card numbers 27 to 999. (1 to 26 are defined in soundcard.h)
+ * Numbers 1000 to N are reserved for driver's internal use.
+ */
+
+#define SNDCARD_DESKPROXL 27 /* Compaq Deskpro XL */
+#define SNDCARD_VIDC 28 /* ARMs VIDC */
+#define SNDCARD_SBPNP 29
+#define SNDCARD_SOFTOSS 36
+#define SNDCARD_VMIDI 37
+#define SNDCARD_OPL3SA1 38 /* Note: clash in msnd.h */
+#define SNDCARD_OPL3SA1_SB 39
+#define SNDCARD_OPL3SA1_MPU 40
+#define SNDCARD_WAVEFRONT 41
+#define SNDCARD_OPL3SA2 42
+#define SNDCARD_OPL3SA2_MPU 43
+#define SNDCARD_WAVEARTIST 44 /* Waveartist */
+#define SNDCARD_OPL3SA2_MSS 45 /* Originally missed */
+#define SNDCARD_AD1816 88
+
+/*
+ * NOTE! NOTE! NOTE! NOTE!
+ *
+ * If you modify this file, please check the dev_table.c also.
+ *
+ * NOTE! NOTE! NOTE! NOTE!
+ */
+
+struct driver_info
+{
+ char *driver_id;
+ int card_subtype; /* Driver specific. Usually 0 */
+ int card_type; /* From soundcard.h */
+ char *name;
+ void (*attach) (struct address_info *hw_config);
+ int (*probe) (struct address_info *hw_config);
+ void (*unload) (struct address_info *hw_config);
+};
+
+struct card_info
+{
+ int card_type; /* Link (search key) to the driver list */
+ struct address_info config;
+ int enabled;
+ void *for_driver_use;
+};
+
+
+/*
+ * Device specific parameters (used only by dmabuf.c)
+ */
+#define MAX_SUB_BUFFERS (32*MAX_REALTIME_FACTOR)
+
+#define DMODE_NONE 0
+#define DMODE_OUTPUT PCM_ENABLE_OUTPUT
+#define DMODE_INPUT PCM_ENABLE_INPUT
+
+struct dma_buffparms
+{
+ int dma_mode; /* DMODE_INPUT, DMODE_OUTPUT or DMODE_NONE */
+ int closing;
+
+ /*
+ * Pointers to raw buffers
+ */
+
+ char *raw_buf;
+ unsigned long raw_buf_phys;
+ int buffsize;
+
+ /*
+ * Device state tables
+ */
+
+ unsigned long flags;
+#define DMA_BUSY 0x00000001
+#define DMA_RESTART 0x00000002
+#define DMA_ACTIVE 0x00000004
+#define DMA_STARTED 0x00000008
+#define DMA_EMPTY 0x00000010
+#define DMA_ALLOC_DONE 0x00000020
+#define DMA_SYNCING 0x00000040
+#define DMA_DIRTY 0x00000080
+#define DMA_POST 0x00000100
+#define DMA_NODMA 0x00000200
+#define DMA_NOTIMEOUT 0x00000400
+
+ int open_mode;
+
+ /*
+ * Queue parameters.
+ */
+ int qlen;
+ int qhead;
+ int qtail;
+ spinlock_t lock;
+
+ int cfrag; /* Current incomplete fragment (write) */
+
+ int nbufs;
+ int counts[MAX_SUB_BUFFERS];
+ int subdivision;
+
+ int fragment_size;
+ int needs_reorg;
+ int max_fragments;
+
+ int bytes_in_use;
+
+ int underrun_count;
+ unsigned long byte_counter;
+ unsigned long user_counter;
+ unsigned long max_byte_counter;
+ int data_rate; /* Bytes/second */
+
+ int mapping_flags;
+#define DMA_MAP_MAPPED 0x00000001
+ char neutral_byte;
+ int dma; /* DMA channel */
+
+ int applic_profile; /* Application profile (APF_*) */
+ /* Interrupt callback stuff */
+ void (*audio_callback) (int dev, int parm);
+ int callback_parm;
+
+ int buf_flags[MAX_SUB_BUFFERS];
+#define BUFF_EOF 0x00000001 /* Increment eof count */
+#define BUFF_DIRTY 0x00000002 /* Buffer written */
+};
+
+/*
+ * Structure for use with various microcontrollers and DSP processors
+ * in the recent sound cards.
+ */
+typedef struct coproc_operations
+{
+ char name[64];
+ struct module *owner;
+ int (*open) (void *devc, int sub_device);
+ void (*close) (void *devc, int sub_device);
+ int (*ioctl) (void *devc, unsigned int cmd, void __user * arg, int local);
+ void (*reset) (void *devc);
+
+ void *devc; /* Driver specific info */
+} coproc_operations;
+
+struct audio_driver
+{
+ struct module *owner;
+ int (*open) (int dev, int mode);
+ void (*close) (int dev);
+ void (*output_block) (int dev, unsigned long buf,
+ int count, int intrflag);
+ void (*start_input) (int dev, unsigned long buf,
+ int count, int intrflag);
+ int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
+ int (*prepare_for_input) (int dev, int bufsize, int nbufs);
+ int (*prepare_for_output) (int dev, int bufsize, int nbufs);
+ void (*halt_io) (int dev);
+ int (*local_qlen)(int dev);
+ void (*copy_user) (int dev,
+ char *localbuf, int localoffs,
+ const char __user *userbuf, int useroffs,
+ int max_in, int max_out,
+ int *used, int *returned,
+ int len);
+ void (*halt_input) (int dev);
+ void (*halt_output) (int dev);
+ void (*trigger) (int dev, int bits);
+ int (*set_speed)(int dev, int speed);
+ unsigned int (*set_bits)(int dev, unsigned int bits);
+ short (*set_channels)(int dev, short channels);
+ void (*postprocess_write)(int dev); /* Device spesific postprocessing for written data */
+ void (*preprocess_read)(int dev); /* Device spesific preprocessing for read data */
+ void (*mmap)(int dev);
+};
+
+struct audio_operations
+{
+ char name[128];
+ int flags;
+#define NOTHING_SPECIAL 0x00
+#define NEEDS_RESTART 0x01
+#define DMA_AUTOMODE 0x02
+#define DMA_DUPLEX 0x04
+#define DMA_PSEUDO_AUTOMODE 0x08
+#define DMA_HARDSTOP 0x10
+#define DMA_EXACT 0x40
+#define DMA_NORESET 0x80
+ int format_mask; /* Bitmask for supported audio formats */
+ void *devc; /* Driver specific info */
+ struct audio_driver *d;
+ void *portc; /* Driver specific info */
+ struct dma_buffparms *dmap_in, *dmap_out;
+ struct coproc_operations *coproc;
+ int mixer_dev;
+ int enable_bits;
+ int open_mode;
+ int go;
+ int min_fragment; /* 0 == unlimited */
+ int max_fragment; /* 0 == unlimited */
+ int parent_dev; /* 0 -> no parent, 1 to n -> parent=parent_dev+1 */
+
+ /* fields formerly in dmabuf.c */
+ wait_queue_head_t in_sleeper;
+ wait_queue_head_t out_sleeper;
+ wait_queue_head_t poll_sleeper;
+
+ /* fields formerly in audio.c */
+ int audio_mode;
+
+#define AM_NONE 0
+#define AM_WRITE OPEN_WRITE
+#define AM_READ OPEN_READ
+
+ int local_format;
+ int audio_format;
+ int local_conversion;
+#define CNV_MU_LAW 0x00000001
+
+ /* large structures at the end to keep offsets small */
+ struct dma_buffparms dmaps[2];
+};
+
+int *load_mixer_volumes(char *name, int *levels, int present);
+
+struct mixer_operations
+{
+ struct module *owner;
+ char id[16];
+ char name[64];
+ int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
+
+ void *devc;
+ int modify_counter;
+};
+
+struct synth_operations
+{
+ struct module *owner;
+ char *id; /* Unique identifier (ASCII) max 29 char */
+ struct synth_info *info;
+ int midi_dev;
+ int synth_type;
+ int synth_subtype;
+
+ int (*open) (int dev, int mode);
+ void (*close) (int dev);
+ int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
+ int (*kill_note) (int dev, int voice, int note, int velocity);
+ int (*start_note) (int dev, int voice, int note, int velocity);
+ int (*set_instr) (int dev, int voice, int instr);
+ void (*reset) (int dev);
+ void (*hw_control) (int dev, unsigned char *event);
+ int (*load_patch) (int dev, int format, const char __user *addr,
+ int offs, int count, int pmgr_flag);
+ void (*aftertouch) (int dev, int voice, int pressure);
+ void (*controller) (int dev, int voice, int ctrl_num, int value);
+ void (*panning) (int dev, int voice, int value);
+ void (*volume_method) (int dev, int mode);
+ void (*bender) (int dev, int chn, int value);
+ int (*alloc_voice) (int dev, int chn, int note, struct voice_alloc_info *alloc);
+ void (*setup_voice) (int dev, int voice, int chn);
+ int (*send_sysex)(int dev, unsigned char *bytes, int len);
+
+ struct voice_alloc_info alloc;
+ struct channel_info chn_info[16];
+ int emulation;
+#define EMU_GM 1 /* General MIDI */
+#define EMU_XG 2 /* Yamaha XG */
+#define MAX_SYSEX_BUF 64
+ unsigned char sysex_buf[MAX_SYSEX_BUF];
+ int sysex_ptr;
+};
+
+struct midi_input_info
+{
+ /* MIDI input scanner variables */
+#define MI_MAX 10
+ volatile int m_busy;
+ unsigned char m_buf[MI_MAX];
+ unsigned char m_prev_status; /* For running status */
+ int m_ptr;
+#define MST_INIT 0
+#define MST_DATA 1
+#define MST_SYSEX 2
+ int m_state;
+ int m_left;
+};
+
+struct midi_operations
+{
+ struct module *owner;
+ struct midi_info info;
+ struct synth_operations *converter;
+ struct midi_input_info in_info;
+ int (*open) (int dev, int mode,
+ void (*inputintr)(int dev, unsigned char data),
+ void (*outputintr)(int dev)
+ );
+ void (*close) (int dev);
+ int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
+ int (*outputc) (int dev, unsigned char data);
+ int (*start_read) (int dev);
+ int (*end_read) (int dev);
+ void (*kick)(int dev);
+ int (*command) (int dev, unsigned char *data);
+ int (*buffer_status) (int dev);
+ int (*prefix_cmd) (int dev, unsigned char status);
+ struct coproc_operations *coproc;
+ void *devc;
+};
+
+struct sound_lowlev_timer
+{
+ int dev;
+ int priority;
+ unsigned int (*tmr_start)(int dev, unsigned int usecs);
+ void (*tmr_disable)(int dev);
+ void (*tmr_restart)(int dev);
+};
+
+struct sound_timer_operations
+{
+ struct module *owner;
+ struct sound_timer_info info;
+ int priority;
+ int devlink;
+ int (*open)(int dev, int mode);
+ void (*close)(int dev);
+ int (*event)(int dev, unsigned char *ev);
+ unsigned long (*get_time)(int dev);
+ int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
+ void (*arm_timer)(int dev, long time);
+};
+
+extern struct sound_timer_operations default_sound_timer;
+
+extern struct audio_operations *audio_devs[MAX_AUDIO_DEV];
+extern int num_audiodevs;
+extern struct mixer_operations *mixer_devs[MAX_MIXER_DEV];
+extern int num_mixers;
+extern struct synth_operations *synth_devs[MAX_SYNTH_DEV+MAX_MIDI_DEV];
+extern int num_synths;
+extern struct midi_operations *midi_devs[MAX_MIDI_DEV];
+extern int num_midis;
+extern struct sound_timer_operations * sound_timer_devs[MAX_TIMER_DEV];
+extern int num_sound_timers;
+
+extern int sound_map_buffer (int dev, struct dma_buffparms *dmap, buffmem_desc *info);
+void sound_timer_init (struct sound_lowlev_timer *t, char *name);
+void sound_dma_intr (int dev, struct dma_buffparms *dmap, int chan);
+
+#define AUDIO_DRIVER_VERSION 2
+#define MIXER_DRIVER_VERSION 2
+int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
+ int driver_size, int flags, unsigned int format_mask,
+ void *devc, int dma1, int dma2);
+int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
+ int driver_size, void *devc);
+
+void sound_unload_audiodev(int dev);
+void sound_unload_mixerdev(int dev);
+void sound_unload_mididev(int dev);
+void sound_unload_synthdev(int dev);
+void sound_unload_timerdev(int dev);
+int sound_alloc_mixerdev(void);
+int sound_alloc_timerdev(void);
+int sound_alloc_synthdev(void);
+int sound_alloc_mididev(void);
+#endif /* _DEV_TABLE_H_ */
+
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
new file mode 100644
index 0000000..1e90d76
--- /dev/null
+++ b/sound/oss/dmabuf.c
@@ -0,0 +1,1267 @@
+/*
+ * sound/oss/dmabuf.c
+ *
+ * The DMA buffer manager for digitized voice applications
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ * Thomas Sailer : moved several static variables into struct audio_operations
+ * (which is grossly misnamed btw.) because they have the same
+ * lifetime as the rest in there and dynamic allocation saves
+ * 12k or so
+ * Thomas Sailer : remove {in,out}_sleep_flag. It was used for the sleeper to
+ * determine if it was woken up by the expiring timeout or by
+ * an explicit wake_up. The return value from schedule_timeout
+ * can be used instead; if 0, the wakeup was due to the timeout.
+ *
+ * Rob Riggs Added persistent DMA buffers (1998/10/17)
+ */
+
+#define BE_CONSERVATIVE
+#define SAMPLE_ROUNDUP 0
+
+#include <linux/mm.h>
+#include "sound_config.h"
+
+#define DMAP_FREE_ON_CLOSE 0
+#define DMAP_KEEP_ON_CLOSE 1
+extern int sound_dmap_flag;
+
+static void dma_reset_output(int dev);
+static void dma_reset_input(int dev);
+static int local_start_dma(struct audio_operations *adev, unsigned long physaddr, int count, int dma_mode);
+
+
+
+static int debugmem; /* switched off by default */
+static int dma_buffsize = DSP_BUFFSIZE;
+
+static long dmabuf_timeout(struct dma_buffparms *dmap)
+{
+ long tmout;
+
+ tmout = (dmap->fragment_size * HZ) / dmap->data_rate;
+ tmout += HZ / 5; /* Some safety distance */
+ if (tmout < (HZ / 2))
+ tmout = HZ / 2;
+ if (tmout > 20 * HZ)
+ tmout = 20 * HZ;
+ return tmout;
+}
+
+static int sound_alloc_dmap(struct dma_buffparms *dmap)
+{
+ char *start_addr, *end_addr;
+ int dma_pagesize;
+ int sz, size;
+ struct page *page;
+
+ dmap->mapping_flags &= ~DMA_MAP_MAPPED;
+
+ if (dmap->raw_buf != NULL)
+ return 0; /* Already done */
+ if (dma_buffsize < 4096)
+ dma_buffsize = 4096;
+ dma_pagesize = (dmap->dma < 4) ? (64 * 1024) : (128 * 1024);
+
+ /*
+ * Now check for the Cyrix problem.
+ */
+
+ if(isa_dma_bridge_buggy==2)
+ dma_pagesize=32768;
+
+ dmap->raw_buf = NULL;
+ dmap->buffsize = dma_buffsize;
+ if (dmap->buffsize > dma_pagesize)
+ dmap->buffsize = dma_pagesize;
+ start_addr = NULL;
+ /*
+ * Now loop until we get a free buffer. Try to get smaller buffer if
+ * it fails. Don't accept smaller than 8k buffer for performance
+ * reasons.
+ */
+ while (start_addr == NULL && dmap->buffsize > PAGE_SIZE) {
+ for (sz = 0, size = PAGE_SIZE; size < dmap->buffsize; sz++, size <<= 1);
+ dmap->buffsize = PAGE_SIZE * (1 << sz);
+ start_addr = (char *) __get_free_pages(GFP_ATOMIC|GFP_DMA|__GFP_NOWARN, sz);
+ if (start_addr == NULL)
+ dmap->buffsize /= 2;
+ }
+
+ if (start_addr == NULL) {
+ printk(KERN_WARNING "Sound error: Couldn't allocate DMA buffer\n");
+ return -ENOMEM;
+ } else {
+ /* make some checks */
+ end_addr = start_addr + dmap->buffsize - 1;
+
+ if (debugmem)
+ printk(KERN_DEBUG "sound: start 0x%lx, end 0x%lx\n", (long) start_addr, (long) end_addr);
+
+ /* now check if it fits into the same dma-pagesize */
+
+ if (((long) start_addr & ~(dma_pagesize - 1)) != ((long) end_addr & ~(dma_pagesize - 1))
+ || end_addr >= (char *) (MAX_DMA_ADDRESS)) {
+ printk(KERN_ERR "sound: Got invalid address 0x%lx for %db DMA-buffer\n", (long) start_addr, dmap->buffsize);
+ return -EFAULT;
+ }
+ }
+ dmap->raw_buf = start_addr;
+ dmap->raw_buf_phys = virt_to_bus(start_addr);
+
+ for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
+ SetPageReserved(page);
+ return 0;
+}
+
+static void sound_free_dmap(struct dma_buffparms *dmap)
+{
+ int sz, size;
+ struct page *page;
+ unsigned long start_addr, end_addr;
+
+ if (dmap->raw_buf == NULL)
+ return;
+ if (dmap->mapping_flags & DMA_MAP_MAPPED)
+ return; /* Don't free mmapped buffer. Will use it next time */
+ for (sz = 0, size = PAGE_SIZE; size < dmap->buffsize; sz++, size <<= 1);
+
+ start_addr = (unsigned long) dmap->raw_buf;
+ end_addr = start_addr + dmap->buffsize;
+
+ for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
+ ClearPageReserved(page);
+
+ free_pages((unsigned long) dmap->raw_buf, sz);
+ dmap->raw_buf = NULL;
+}
+
+
+/* Intel version !!!!!!!!! */
+
+static int sound_start_dma(struct dma_buffparms *dmap, unsigned long physaddr, int count, int dma_mode)
+{
+ unsigned long flags;
+ int chan = dmap->dma;
+
+ /* printk( "Start DMA%d %d, %d\n", chan, (int)(physaddr-dmap->raw_buf_phys), count); */
+
+ flags = claim_dma_lock();
+ disable_dma(chan);
+ clear_dma_ff(chan);
+ set_dma_mode(chan, dma_mode);
+ set_dma_addr(chan, physaddr);
+ set_dma_count(chan, count);
+ enable_dma(chan);
+ release_dma_lock(flags);
+
+ return 0;
+}
+
+static void dma_init_buffers(struct dma_buffparms *dmap)
+{
+ dmap->qlen = dmap->qhead = dmap->qtail = dmap->user_counter = 0;
+ dmap->byte_counter = 0;
+ dmap->max_byte_counter = 8000 * 60 * 60;
+ dmap->bytes_in_use = dmap->buffsize;
+
+ dmap->dma_mode = DMODE_NONE;
+ dmap->mapping_flags = 0;
+ dmap->neutral_byte = 0x80;
+ dmap->data_rate = 8000;
+ dmap->cfrag = -1;
+ dmap->closing = 0;
+ dmap->nbufs = 1;
+ dmap->flags = DMA_BUSY; /* Other flags off */
+}
+
+static int open_dmap(struct audio_operations *adev, int mode, struct dma_buffparms *dmap)
+{
+ int err;
+
+ if (dmap->flags & DMA_BUSY)
+ return -EBUSY;
+ if ((err = sound_alloc_dmap(dmap)) < 0)
+ return err;
+
+ if (dmap->raw_buf == NULL) {
+ printk(KERN_WARNING "Sound: DMA buffers not available\n");
+ return -ENOSPC; /* Memory allocation failed during boot */
+ }
+ if (dmap->dma >= 0 && sound_open_dma(dmap->dma, adev->name)) {
+ printk(KERN_WARNING "Unable to grab(2) DMA%d for the audio driver\n", dmap->dma);
+ return -EBUSY;
+ }
+ dma_init_buffers(dmap);
+ spin_lock_init(&dmap->lock);
+ dmap->open_mode = mode;
+ dmap->subdivision = dmap->underrun_count = 0;
+ dmap->fragment_size = 0;
+ dmap->max_fragments = 65536; /* Just a large value */
+ dmap->byte_counter = 0;
+ dmap->max_byte_counter = 8000 * 60 * 60;
+ dmap->applic_profile = APF_NORMAL;
+ dmap->needs_reorg = 1;
+ dmap->audio_callback = NULL;
+ dmap->callback_parm = 0;
+ return 0;
+}
+
+static void close_dmap(struct audio_operations *adev, struct dma_buffparms *dmap)
+{
+ unsigned long flags;
+
+ if (dmap->dma >= 0) {
+ sound_close_dma(dmap->dma);
+ flags=claim_dma_lock();
+ disable_dma(dmap->dma);
+ release_dma_lock(flags);
+ }
+ if (dmap->flags & DMA_BUSY)
+ dmap->dma_mode = DMODE_NONE;
+ dmap->flags &= ~DMA_BUSY;
+
+ if (sound_dmap_flag == DMAP_FREE_ON_CLOSE)
+ sound_free_dmap(dmap);
+}
+
+
+static unsigned int default_set_bits(int dev, unsigned int bits)
+{
+ mm_segment_t fs = get_fs();
+
+ set_fs(get_ds());
+ audio_devs[dev]->d->ioctl(dev, SNDCTL_DSP_SETFMT, (void __user *)&bits);
+ set_fs(fs);
+ return bits;
+}
+
+static int default_set_speed(int dev, int speed)
+{
+ mm_segment_t fs = get_fs();
+
+ set_fs(get_ds());
+ audio_devs[dev]->d->ioctl(dev, SNDCTL_DSP_SPEED, (void __user *)&speed);
+ set_fs(fs);
+ return speed;
+}
+
+static short default_set_channels(int dev, short channels)
+{
+ int c = channels;
+ mm_segment_t fs = get_fs();
+
+ set_fs(get_ds());
+ audio_devs[dev]->d->ioctl(dev, SNDCTL_DSP_CHANNELS, (void __user *)&c);
+ set_fs(fs);
+ return c;
+}
+
+static void check_driver(struct audio_driver *d)
+{
+ if (d->set_speed == NULL)
+ d->set_speed = default_set_speed;
+ if (d->set_bits == NULL)
+ d->set_bits = default_set_bits;
+ if (d->set_channels == NULL)
+ d->set_channels = default_set_channels;
+}
+
+int DMAbuf_open(int dev, int mode)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ int retval;
+ struct dma_buffparms *dmap_in = NULL;
+ struct dma_buffparms *dmap_out = NULL;
+
+ if (!adev)
+ return -ENXIO;
+ if (!(adev->flags & DMA_DUPLEX))
+ adev->dmap_in = adev->dmap_out;
+ check_driver(adev->d);
+
+ if ((retval = adev->d->open(dev, mode)) < 0)
+ return retval;
+ dmap_out = adev->dmap_out;
+ dmap_in = adev->dmap_in;
+ if (dmap_in == dmap_out)
+ adev->flags &= ~DMA_DUPLEX;
+
+ if (mode & OPEN_WRITE) {
+ if ((retval = open_dmap(adev, mode, dmap_out)) < 0) {
+ adev->d->close(dev);
+ return retval;
+ }
+ }
+ adev->enable_bits = mode;
+
+ if (mode == OPEN_READ || (mode != OPEN_WRITE && (adev->flags & DMA_DUPLEX))) {
+ if ((retval = open_dmap(adev, mode, dmap_in)) < 0) {
+ adev->d->close(dev);
+ if (mode & OPEN_WRITE)
+ close_dmap(adev, dmap_out);
+ return retval;
+ }
+ }
+ adev->open_mode = mode;
+ adev->go = 1;
+
+ adev->d->set_bits(dev, 8);
+ adev->d->set_channels(dev, 1);
+ adev->d->set_speed(dev, DSP_DEFAULT_SPEED);
+ if (adev->dmap_out->dma_mode == DMODE_OUTPUT)
+ memset(adev->dmap_out->raw_buf, adev->dmap_out->neutral_byte,
+ adev->dmap_out->bytes_in_use);
+ return 0;
+}
+/* MUST not hold the spinlock */
+void DMAbuf_reset(int dev)
+{
+ if (audio_devs[dev]->open_mode & OPEN_WRITE)
+ dma_reset_output(dev);
+
+ if (audio_devs[dev]->open_mode & OPEN_READ)
+ dma_reset_input(dev);
+}
+
+static void dma_reset_output(int dev)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ unsigned long flags,f ;
+ struct dma_buffparms *dmap = adev->dmap_out;
+
+ if (!(dmap->flags & DMA_STARTED)) /* DMA is not active */
+ return;
+
+ /*
+ * First wait until the current fragment has been played completely
+ */
+ spin_lock_irqsave(&dmap->lock,flags);
+ adev->dmap_out->flags |= DMA_SYNCING;
+
+ adev->dmap_out->underrun_count = 0;
+ if (!signal_pending(current) && adev->dmap_out->qlen &&
+ adev->dmap_out->underrun_count == 0){
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ interruptible_sleep_on_timeout(&adev->out_sleeper,
+ dmabuf_timeout(dmap));
+ spin_lock_irqsave(&dmap->lock,flags);
+ }
+ adev->dmap_out->flags &= ~(DMA_SYNCING | DMA_ACTIVE);
+
+ /*
+ * Finally shut the device off
+ */
+ if (!(adev->flags & DMA_DUPLEX) || !adev->d->halt_output)
+ adev->d->halt_io(dev);
+ else
+ adev->d->halt_output(dev);
+ adev->dmap_out->flags &= ~DMA_STARTED;
+
+ f=claim_dma_lock();
+ clear_dma_ff(dmap->dma);
+ disable_dma(dmap->dma);
+ release_dma_lock(f);
+
+ dmap->byte_counter = 0;
+ reorganize_buffers(dev, adev->dmap_out, 0);
+ dmap->qlen = dmap->qhead = dmap->qtail = dmap->user_counter = 0;
+ spin_unlock_irqrestore(&dmap->lock,flags);
+}
+
+static void dma_reset_input(int dev)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ unsigned long flags;
+ struct dma_buffparms *dmap = adev->dmap_in;
+
+ spin_lock_irqsave(&dmap->lock,flags);
+ if (!(adev->flags & DMA_DUPLEX) || !adev->d->halt_input)
+ adev->d->halt_io(dev);
+ else
+ adev->d->halt_input(dev);
+ adev->dmap_in->flags &= ~DMA_STARTED;
+
+ dmap->qlen = dmap->qhead = dmap->qtail = dmap->user_counter = 0;
+ dmap->byte_counter = 0;
+ reorganize_buffers(dev, adev->dmap_in, 1);
+ spin_unlock_irqrestore(&dmap->lock,flags);
+}
+/* MUST be called with holding the dmap->lock */
+void DMAbuf_launch_output(int dev, struct dma_buffparms *dmap)
+{
+ struct audio_operations *adev = audio_devs[dev];
+
+ if (!((adev->enable_bits * adev->go) & PCM_ENABLE_OUTPUT))
+ return; /* Don't start DMA yet */
+ dmap->dma_mode = DMODE_OUTPUT;
+
+ if (!(dmap->flags & DMA_ACTIVE) || !(adev->flags & DMA_AUTOMODE) || (dmap->flags & DMA_NODMA)) {
+ if (!(dmap->flags & DMA_STARTED)) {
+ reorganize_buffers(dev, dmap, 0);
+ if (adev->d->prepare_for_output(dev, dmap->fragment_size, dmap->nbufs))
+ return;
+ if (!(dmap->flags & DMA_NODMA))
+ local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use,DMA_MODE_WRITE);
+ dmap->flags |= DMA_STARTED;
+ }
+ if (dmap->counts[dmap->qhead] == 0)
+ dmap->counts[dmap->qhead] = dmap->fragment_size;
+ dmap->dma_mode = DMODE_OUTPUT;
+ adev->d->output_block(dev, dmap->raw_buf_phys + dmap->qhead * dmap->fragment_size,
+ dmap->counts[dmap->qhead], 1);
+ if (adev->d->trigger)
+ adev->d->trigger(dev,adev->enable_bits * adev->go);
+ }
+ dmap->flags |= DMA_ACTIVE;
+}
+
+int DMAbuf_sync(int dev)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ unsigned long flags;
+ int n = 0;
+ struct dma_buffparms *dmap;
+
+ if (!adev->go && !(adev->enable_bits & PCM_ENABLE_OUTPUT))
+ return 0;
+
+ if (adev->dmap_out->dma_mode == DMODE_OUTPUT) {
+ dmap = adev->dmap_out;
+ spin_lock_irqsave(&dmap->lock,flags);
+ if (dmap->qlen > 0 && !(dmap->flags & DMA_ACTIVE))
+ DMAbuf_launch_output(dev, dmap);
+ adev->dmap_out->flags |= DMA_SYNCING;
+ adev->dmap_out->underrun_count = 0;
+ while (!signal_pending(current) && n++ <= adev->dmap_out->nbufs &&
+ adev->dmap_out->qlen && adev->dmap_out->underrun_count == 0) {
+ long t = dmabuf_timeout(dmap);
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ /* FIXME: not safe may miss events */
+ t = interruptible_sleep_on_timeout(&adev->out_sleeper, t);
+ spin_lock_irqsave(&dmap->lock,flags);
+ if (!t) {
+ adev->dmap_out->flags &= ~DMA_SYNCING;
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return adev->dmap_out->qlen;
+ }
+ }
+ adev->dmap_out->flags &= ~(DMA_SYNCING | DMA_ACTIVE);
+
+ /*
+ * Some devices such as GUS have huge amount of on board RAM for the
+ * audio data. We have to wait until the device has finished playing.
+ */
+
+ /* still holding the lock */
+ if (adev->d->local_qlen) { /* Device has hidden buffers */
+ while (!signal_pending(current) &&
+ adev->d->local_qlen(dev)){
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ interruptible_sleep_on_timeout(&adev->out_sleeper,
+ dmabuf_timeout(dmap));
+ spin_lock_irqsave(&dmap->lock,flags);
+ }
+ }
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ }
+ adev->dmap_out->dma_mode = DMODE_NONE;
+ return adev->dmap_out->qlen;
+}
+
+int DMAbuf_release(int dev, int mode)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap;
+ unsigned long flags;
+
+ dmap = adev->dmap_out;
+ if (adev->open_mode & OPEN_WRITE)
+ adev->dmap_out->closing = 1;
+
+ if (adev->open_mode & OPEN_READ){
+ adev->dmap_in->closing = 1;
+ dmap = adev->dmap_in;
+ }
+ if (adev->open_mode & OPEN_WRITE)
+ if (!(adev->dmap_out->mapping_flags & DMA_MAP_MAPPED))
+ if (!signal_pending(current) && (adev->dmap_out->dma_mode == DMODE_OUTPUT))
+ DMAbuf_sync(dev);
+ if (adev->dmap_out->dma_mode == DMODE_OUTPUT)
+ memset(adev->dmap_out->raw_buf, adev->dmap_out->neutral_byte, adev->dmap_out->bytes_in_use);
+
+ DMAbuf_reset(dev);
+ spin_lock_irqsave(&dmap->lock,flags);
+ adev->d->close(dev);
+
+ if (adev->open_mode & OPEN_WRITE)
+ close_dmap(adev, adev->dmap_out);
+
+ if (adev->open_mode == OPEN_READ ||
+ (adev->open_mode != OPEN_WRITE &&
+ (adev->flags & DMA_DUPLEX)))
+ close_dmap(adev, adev->dmap_in);
+ adev->open_mode = 0;
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return 0;
+}
+/* called with dmap->lock dold */
+int DMAbuf_activate_recording(int dev, struct dma_buffparms *dmap)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ int err;
+
+ if (!(adev->open_mode & OPEN_READ))
+ return 0;
+ if (!(adev->enable_bits & PCM_ENABLE_INPUT))
+ return 0;
+ if (dmap->dma_mode == DMODE_OUTPUT) { /* Direction change */
+ /* release lock - it's not recursive */
+ spin_unlock_irq(&dmap->lock);
+ DMAbuf_sync(dev);
+ DMAbuf_reset(dev);
+ spin_lock_irq(&dmap->lock);
+ dmap->dma_mode = DMODE_NONE;
+ }
+ if (!dmap->dma_mode) {
+ reorganize_buffers(dev, dmap, 1);
+ if ((err = adev->d->prepare_for_input(dev,
+ dmap->fragment_size, dmap->nbufs)) < 0)
+ return err;
+ dmap->dma_mode = DMODE_INPUT;
+ }
+ if (!(dmap->flags & DMA_ACTIVE)) {
+ if (dmap->needs_reorg)
+ reorganize_buffers(dev, dmap, 0);
+ local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use, DMA_MODE_READ);
+ adev->d->start_input(dev, dmap->raw_buf_phys + dmap->qtail * dmap->fragment_size,
+ dmap->fragment_size, 0);
+ dmap->flags |= DMA_ACTIVE;
+ if (adev->d->trigger)
+ adev->d->trigger(dev, adev->enable_bits * adev->go);
+ }
+ return 0;
+}
+/* acquires lock */
+int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ unsigned long flags;
+ int err = 0, n = 0;
+ struct dma_buffparms *dmap = adev->dmap_in;
+ int go;
+
+ if (!(adev->open_mode & OPEN_READ))
+ return -EIO;
+ spin_lock_irqsave(&dmap->lock,flags);
+ if (dmap->needs_reorg)
+ reorganize_buffers(dev, dmap, 0);
+ if (adev->dmap_in->mapping_flags & DMA_MAP_MAPPED) {
+/* printk(KERN_WARNING "Sound: Can't read from mmapped device (1)\n");*/
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return -EINVAL;
+ } else while (dmap->qlen <= 0 && n++ < 10) {
+ long timeout = MAX_SCHEDULE_TIMEOUT;
+ if (!(adev->enable_bits & PCM_ENABLE_INPUT) || !adev->go) {
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return -EAGAIN;
+ }
+ if ((err = DMAbuf_activate_recording(dev, dmap)) < 0) {
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return err;
+ }
+ /* Wait for the next block */
+
+ if (dontblock) {
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return -EAGAIN;
+ }
+ if ((go = adev->go))
+ timeout = dmabuf_timeout(dmap);
+
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ timeout = interruptible_sleep_on_timeout(&adev->in_sleeper,
+ timeout);
+ if (!timeout) {
+ /* FIXME: include device name */
+ err = -EIO;
+ printk(KERN_WARNING "Sound: DMA (input) timed out - IRQ/DRQ config error?\n");
+ dma_reset_input(dev);
+ } else
+ err = -EINTR;
+ spin_lock_irqsave(&dmap->lock,flags);
+ }
+ spin_unlock_irqrestore(&dmap->lock,flags);
+
+ if (dmap->qlen <= 0)
+ return err ? err : -EINTR;
+ *buf = &dmap->raw_buf[dmap->qhead * dmap->fragment_size + dmap->counts[dmap->qhead]];
+ *len = dmap->fragment_size - dmap->counts[dmap->qhead];
+
+ return dmap->qhead;
+}
+
+int DMAbuf_rmchars(int dev, int buff_no, int c)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_in;
+ int p = dmap->counts[dmap->qhead] + c;
+
+ if (dmap->mapping_flags & DMA_MAP_MAPPED)
+ {
+/* printk("Sound: Can't read from mmapped device (2)\n");*/
+ return -EINVAL;
+ }
+ else if (dmap->qlen <= 0)
+ return -EIO;
+ else if (p >= dmap->fragment_size) { /* This buffer is completely empty */
+ dmap->counts[dmap->qhead] = 0;
+ dmap->qlen--;
+ dmap->qhead = (dmap->qhead + 1) % dmap->nbufs;
+ }
+ else dmap->counts[dmap->qhead] = p;
+
+ return 0;
+}
+/* MUST be called with dmap->lock hold */
+int DMAbuf_get_buffer_pointer(int dev, struct dma_buffparms *dmap, int direction)
+{
+ /*
+ * Try to approximate the active byte position of the DMA pointer within the
+ * buffer area as well as possible.
+ */
+
+ int pos;
+ unsigned long f;
+
+ if (!(dmap->flags & DMA_ACTIVE))
+ pos = 0;
+ else {
+ int chan = dmap->dma;
+
+ f=claim_dma_lock();
+ clear_dma_ff(chan);
+
+ if(!isa_dma_bridge_buggy)
+ disable_dma(dmap->dma);
+
+ pos = get_dma_residue(chan);
+
+ pos = dmap->bytes_in_use - pos;
+
+ if (!(dmap->mapping_flags & DMA_MAP_MAPPED)) {
+ if (direction == DMODE_OUTPUT) {
+ if (dmap->qhead == 0)
+ if (pos > dmap->fragment_size)
+ pos = 0;
+ } else {
+ if (dmap->qtail == 0)
+ if (pos > dmap->fragment_size)
+ pos = 0;
+ }
+ }
+ if (pos < 0)
+ pos = 0;
+ if (pos >= dmap->bytes_in_use)
+ pos = 0;
+
+ if(!isa_dma_bridge_buggy)
+ enable_dma(dmap->dma);
+
+ release_dma_lock(f);
+ }
+ /* printk( "%04x ", pos); */
+
+ return pos;
+}
+
+/*
+ * DMAbuf_start_devices() is called by the /dev/music driver to start
+ * one or more audio devices at desired moment.
+ */
+
+void DMAbuf_start_devices(unsigned int devmask)
+{
+ struct audio_operations *adev;
+ int dev;
+
+ for (dev = 0; dev < num_audiodevs; dev++) {
+ if (!(devmask & (1 << dev)))
+ continue;
+ if (!(adev = audio_devs[dev]))
+ continue;
+ if (adev->open_mode == 0)
+ continue;
+ if (adev->go)
+ continue;
+ /* OK to start the device */
+ adev->go = 1;
+ if (adev->d->trigger)
+ adev->d->trigger(dev,adev->enable_bits * adev->go);
+ }
+}
+/* via poll called without a lock ?*/
+int DMAbuf_space_in_queue(int dev)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ int len, max, tmp;
+ struct dma_buffparms *dmap = adev->dmap_out;
+ int lim = dmap->nbufs;
+
+ if (lim < 2)
+ lim = 2;
+
+ if (dmap->qlen >= lim) /* No space at all */
+ return 0;
+
+ /*
+ * Verify that there are no more pending buffers than the limit
+ * defined by the process.
+ */
+
+ max = dmap->max_fragments;
+ if (max > lim)
+ max = lim;
+ len = dmap->qlen;
+
+ if (adev->d->local_qlen) {
+ tmp = adev->d->local_qlen(dev);
+ if (tmp && len)
+ tmp--; /* This buffer has been counted twice */
+ len += tmp;
+ }
+ if (dmap->byte_counter % dmap->fragment_size) /* There is a partial fragment */
+ len = len + 1;
+
+ if (len >= max)
+ return 0;
+ return max - len;
+}
+/* MUST not hold the spinlock - this function may sleep */
+static int output_sleep(int dev, int dontblock)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ int err = 0;
+ struct dma_buffparms *dmap = adev->dmap_out;
+ long timeout;
+ long timeout_value;
+
+ if (dontblock)
+ return -EAGAIN;
+ if (!(adev->enable_bits & PCM_ENABLE_OUTPUT))
+ return -EAGAIN;
+
+ /*
+ * Wait for free space
+ */
+ if (signal_pending(current))
+ return -EINTR;
+ timeout = (adev->go && !(dmap->flags & DMA_NOTIMEOUT));
+ if (timeout)
+ timeout_value = dmabuf_timeout(dmap);
+ else
+ timeout_value = MAX_SCHEDULE_TIMEOUT;
+ timeout_value = interruptible_sleep_on_timeout(&adev->out_sleeper,
+ timeout_value);
+ if (timeout != MAX_SCHEDULE_TIMEOUT && !timeout_value) {
+ printk(KERN_WARNING "Sound: DMA (output) timed out - IRQ/DRQ config error?\n");
+ dma_reset_output(dev);
+ } else {
+ if (signal_pending(current))
+ err = -EINTR;
+ }
+ return err;
+}
+/* called with the lock held */
+static int find_output_space(int dev, char **buf, int *size)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_out;
+ unsigned long active_offs;
+ long len, offs;
+ int maxfrags;
+ int occupied_bytes = (dmap->user_counter % dmap->fragment_size);
+
+ *buf = dmap->raw_buf;
+ if (!(maxfrags = DMAbuf_space_in_queue(dev)) && !occupied_bytes)
+ return 0;
+
+#ifdef BE_CONSERVATIVE
+ active_offs = dmap->byte_counter + dmap->qhead * dmap->fragment_size;
+#else
+ active_offs = max(DMAbuf_get_buffer_pointer(dev, dmap, DMODE_OUTPUT), 0);
+ /* Check for pointer wrapping situation */
+ if (active_offs >= dmap->bytes_in_use)
+ active_offs = 0;
+ active_offs += dmap->byte_counter;
+#endif
+
+ offs = (dmap->user_counter % dmap->bytes_in_use) & ~SAMPLE_ROUNDUP;
+ if (offs < 0 || offs >= dmap->bytes_in_use) {
+ printk(KERN_ERR "Sound: Got unexpected offs %ld. Giving up.\n", offs);
+ printk("Counter = %ld, bytes=%d\n", dmap->user_counter, dmap->bytes_in_use);
+ return 0;
+ }
+ *buf = dmap->raw_buf + offs;
+
+ len = active_offs + dmap->bytes_in_use - dmap->user_counter; /* Number of unused bytes in buffer */
+
+ if ((offs + len) > dmap->bytes_in_use)
+ len = dmap->bytes_in_use - offs;
+ if (len < 0) {
+ return 0;
+ }
+ if (len > ((maxfrags * dmap->fragment_size) - occupied_bytes))
+ len = (maxfrags * dmap->fragment_size) - occupied_bytes;
+ *size = len & ~SAMPLE_ROUNDUP;
+ return (*size > 0);
+}
+/* acquires lock */
+int DMAbuf_getwrbuffer(int dev, char **buf, int *size, int dontblock)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ unsigned long flags;
+ int err = -EIO;
+ struct dma_buffparms *dmap = adev->dmap_out;
+
+ if (dmap->mapping_flags & DMA_MAP_MAPPED) {
+/* printk(KERN_DEBUG "Sound: Can't write to mmapped device (3)\n");*/
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&dmap->lock,flags);
+ if (dmap->needs_reorg)
+ reorganize_buffers(dev, dmap, 0);
+
+ if (dmap->dma_mode == DMODE_INPUT) { /* Direction change */
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ DMAbuf_reset(dev);
+ spin_lock_irqsave(&dmap->lock,flags);
+ }
+ dmap->dma_mode = DMODE_OUTPUT;
+
+ while (find_output_space(dev, buf, size) <= 0) {
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ if ((err = output_sleep(dev, dontblock)) < 0) {
+ return err;
+ }
+ spin_lock_irqsave(&dmap->lock,flags);
+ }
+
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return 0;
+}
+/* has to acquire dmap->lock */
+int DMAbuf_move_wrpointer(int dev, int l)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_out;
+ unsigned long ptr;
+ unsigned long end_ptr, p;
+ int post;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dmap->lock,flags);
+ post= (dmap->flags & DMA_POST);
+ ptr = (dmap->user_counter / dmap->fragment_size) * dmap->fragment_size;
+
+ dmap->flags &= ~DMA_POST;
+ dmap->cfrag = -1;
+ dmap->user_counter += l;
+ dmap->flags |= DMA_DIRTY;
+
+ if (dmap->byte_counter >= dmap->max_byte_counter) {
+ /* Wrap the byte counters */
+ long decr = dmap->byte_counter;
+ dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use);
+ decr -= dmap->byte_counter;
+ dmap->user_counter -= decr;
+ }
+ end_ptr = (dmap->user_counter / dmap->fragment_size) * dmap->fragment_size;
+
+ p = (dmap->user_counter - 1) % dmap->bytes_in_use;
+ dmap->neutral_byte = dmap->raw_buf[p];
+
+ /* Update the fragment based bookkeeping too */
+ while (ptr < end_ptr) {
+ dmap->counts[dmap->qtail] = dmap->fragment_size;
+ dmap->qtail = (dmap->qtail + 1) % dmap->nbufs;
+ dmap->qlen++;
+ ptr += dmap->fragment_size;
+ }
+
+ dmap->counts[dmap->qtail] = dmap->user_counter - ptr;
+
+ /*
+ * Let the low level driver perform some postprocessing to
+ * the written data.
+ */
+ if (adev->d->postprocess_write)
+ adev->d->postprocess_write(dev);
+
+ if (!(dmap->flags & DMA_ACTIVE))
+ if (dmap->qlen > 1 || (dmap->qlen > 0 && (post || dmap->qlen >= dmap->nbufs - 1)))
+ DMAbuf_launch_output(dev, dmap);
+
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ return 0;
+}
+
+int DMAbuf_start_dma(int dev, unsigned long physaddr, int count, int dma_mode)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = (dma_mode == DMA_MODE_WRITE) ? adev->dmap_out : adev->dmap_in;
+
+ if (dmap->raw_buf == NULL) {
+ printk(KERN_ERR "sound: DMA buffer(1) == NULL\n");
+ printk("Device %d, chn=%s\n", dev, (dmap == adev->dmap_out) ? "out" : "in");
+ return 0;
+ }
+ if (dmap->dma < 0)
+ return 0;
+ sound_start_dma(dmap, physaddr, count, dma_mode);
+ return count;
+}
+EXPORT_SYMBOL(DMAbuf_start_dma);
+
+static int local_start_dma(struct audio_operations *adev, unsigned long physaddr, int count, int dma_mode)
+{
+ struct dma_buffparms *dmap = (dma_mode == DMA_MODE_WRITE) ? adev->dmap_out : adev->dmap_in;
+
+ if (dmap->raw_buf == NULL) {
+ printk(KERN_ERR "sound: DMA buffer(2) == NULL\n");
+ printk(KERN_ERR "Device %s, chn=%s\n", adev->name, (dmap == adev->dmap_out) ? "out" : "in");
+ return 0;
+ }
+ if (dmap->flags & DMA_NODMA)
+ return 1;
+ if (dmap->dma < 0)
+ return 0;
+ sound_start_dma(dmap, dmap->raw_buf_phys, dmap->bytes_in_use, dma_mode | DMA_AUTOINIT);
+ dmap->flags |= DMA_STARTED;
+ return count;
+}
+
+static void finish_output_interrupt(int dev, struct dma_buffparms *dmap)
+{
+ struct audio_operations *adev = audio_devs[dev];
+
+ if (dmap->audio_callback != NULL)
+ dmap->audio_callback(dev, dmap->callback_parm);
+ wake_up(&adev->out_sleeper);
+ wake_up(&adev->poll_sleeper);
+}
+/* called with dmap->lock held in irq context*/
+static void do_outputintr(int dev, int dummy)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_out;
+ int this_fragment;
+
+ if (dmap->raw_buf == NULL) {
+ printk(KERN_ERR "Sound: Error. Audio interrupt (%d) after freeing buffers.\n", dev);
+ return;
+ }
+ if (dmap->mapping_flags & DMA_MAP_MAPPED) { /* Virtual memory mapped access */
+ /* mmapped access */
+ dmap->qhead = (dmap->qhead + 1) % dmap->nbufs;
+ if (dmap->qhead == 0) { /* Wrapped */
+ dmap->byte_counter += dmap->bytes_in_use;
+ if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */
+ long decr = dmap->byte_counter;
+ dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use);
+ decr -= dmap->byte_counter;
+ dmap->user_counter -= decr;
+ }
+ }
+ dmap->qlen++; /* Yes increment it (don't decrement) */
+ if (!(adev->flags & DMA_AUTOMODE))
+ dmap->flags &= ~DMA_ACTIVE;
+ dmap->counts[dmap->qhead] = dmap->fragment_size;
+ DMAbuf_launch_output(dev, dmap);
+ finish_output_interrupt(dev, dmap);
+ return;
+ }
+
+ dmap->qlen--;
+ this_fragment = dmap->qhead;
+ dmap->qhead = (dmap->qhead + 1) % dmap->nbufs;
+
+ if (dmap->qhead == 0) { /* Wrapped */
+ dmap->byte_counter += dmap->bytes_in_use;
+ if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */
+ long decr = dmap->byte_counter;
+ dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use);
+ decr -= dmap->byte_counter;
+ dmap->user_counter -= decr;
+ }
+ }
+ if (!(adev->flags & DMA_AUTOMODE))
+ dmap->flags &= ~DMA_ACTIVE;
+
+ /*
+ * This is dmap->qlen <= 0 except when closing when
+ * dmap->qlen < 0
+ */
+
+ while (dmap->qlen <= -dmap->closing) {
+ dmap->underrun_count++;
+ dmap->qlen++;
+ if ((dmap->flags & DMA_DIRTY) && dmap->applic_profile != APF_CPUINTENS) {
+ dmap->flags &= ~DMA_DIRTY;
+ memset(adev->dmap_out->raw_buf, adev->dmap_out->neutral_byte,
+ adev->dmap_out->buffsize);
+ }
+ dmap->user_counter += dmap->fragment_size;
+ dmap->qtail = (dmap->qtail + 1) % dmap->nbufs;
+ }
+ if (dmap->qlen > 0)
+ DMAbuf_launch_output(dev, dmap);
+ finish_output_interrupt(dev, dmap);
+}
+/* called in irq context */
+void DMAbuf_outputintr(int dev, int notify_only)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ unsigned long flags;
+ struct dma_buffparms *dmap = adev->dmap_out;
+
+ spin_lock_irqsave(&dmap->lock,flags);
+ if (!(dmap->flags & DMA_NODMA)) {
+ int chan = dmap->dma, pos, n;
+ unsigned long f;
+
+ f=claim_dma_lock();
+
+ if(!isa_dma_bridge_buggy)
+ disable_dma(dmap->dma);
+ clear_dma_ff(chan);
+ pos = dmap->bytes_in_use - get_dma_residue(chan);
+ if(!isa_dma_bridge_buggy)
+ enable_dma(dmap->dma);
+ release_dma_lock(f);
+
+ pos = pos / dmap->fragment_size; /* Actual qhead */
+ if (pos < 0 || pos >= dmap->nbufs)
+ pos = 0;
+ n = 0;
+ while (dmap->qhead != pos && n++ < dmap->nbufs)
+ do_outputintr(dev, notify_only);
+ }
+ else
+ do_outputintr(dev, notify_only);
+ spin_unlock_irqrestore(&dmap->lock,flags);
+}
+EXPORT_SYMBOL(DMAbuf_outputintr);
+
+/* called with dmap->lock held in irq context */
+static void do_inputintr(int dev)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_in;
+
+ if (dmap->raw_buf == NULL) {
+ printk(KERN_ERR "Sound: Fatal error. Audio interrupt after freeing buffers.\n");
+ return;
+ }
+ if (dmap->mapping_flags & DMA_MAP_MAPPED) {
+ dmap->qtail = (dmap->qtail + 1) % dmap->nbufs;
+ if (dmap->qtail == 0) { /* Wrapped */
+ dmap->byte_counter += dmap->bytes_in_use;
+ if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */
+ long decr = dmap->byte_counter;
+ dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use) + dmap->bytes_in_use;
+ decr -= dmap->byte_counter;
+ dmap->user_counter -= decr;
+ }
+ }
+ dmap->qlen++;
+
+ if (!(adev->flags & DMA_AUTOMODE)) {
+ if (dmap->needs_reorg)
+ reorganize_buffers(dev, dmap, 0);
+ local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use,DMA_MODE_READ);
+ adev->d->start_input(dev, dmap->raw_buf_phys + dmap->qtail * dmap->fragment_size,
+ dmap->fragment_size, 1);
+ if (adev->d->trigger)
+ adev->d->trigger(dev, adev->enable_bits * adev->go);
+ }
+ dmap->flags |= DMA_ACTIVE;
+ } else if (dmap->qlen >= (dmap->nbufs - 1)) {
+ printk(KERN_WARNING "Sound: Recording overrun\n");
+ dmap->underrun_count++;
+
+ /* Just throw away the oldest fragment but keep the engine running */
+ dmap->qhead = (dmap->qhead + 1) % dmap->nbufs;
+ dmap->qtail = (dmap->qtail + 1) % dmap->nbufs;
+ } else if (dmap->qlen >= 0 && dmap->qlen < dmap->nbufs) {
+ dmap->qlen++;
+ dmap->qtail = (dmap->qtail + 1) % dmap->nbufs;
+ if (dmap->qtail == 0) { /* Wrapped */
+ dmap->byte_counter += dmap->bytes_in_use;
+ if (dmap->byte_counter >= dmap->max_byte_counter) { /* Overflow */
+ long decr = dmap->byte_counter;
+ dmap->byte_counter = (dmap->byte_counter % dmap->bytes_in_use) + dmap->bytes_in_use;
+ decr -= dmap->byte_counter;
+ dmap->user_counter -= decr;
+ }
+ }
+ }
+ if (!(adev->flags & DMA_AUTOMODE) || (dmap->flags & DMA_NODMA)) {
+ local_start_dma(adev, dmap->raw_buf_phys, dmap->bytes_in_use, DMA_MODE_READ);
+ adev->d->start_input(dev, dmap->raw_buf_phys + dmap->qtail * dmap->fragment_size, dmap->fragment_size, 1);
+ if (adev->d->trigger)
+ adev->d->trigger(dev,adev->enable_bits * adev->go);
+ }
+ dmap->flags |= DMA_ACTIVE;
+ if (dmap->qlen > 0)
+ {
+ wake_up(&adev->in_sleeper);
+ wake_up(&adev->poll_sleeper);
+ }
+}
+/* called in irq context */
+void DMAbuf_inputintr(int dev)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_in;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dmap->lock,flags);
+
+ if (!(dmap->flags & DMA_NODMA)) {
+ int chan = dmap->dma, pos, n;
+ unsigned long f;
+
+ f=claim_dma_lock();
+ if(!isa_dma_bridge_buggy)
+ disable_dma(dmap->dma);
+ clear_dma_ff(chan);
+ pos = dmap->bytes_in_use - get_dma_residue(chan);
+ if(!isa_dma_bridge_buggy)
+ enable_dma(dmap->dma);
+ release_dma_lock(f);
+
+ pos = pos / dmap->fragment_size; /* Actual qhead */
+ if (pos < 0 || pos >= dmap->nbufs)
+ pos = 0;
+
+ n = 0;
+ while (dmap->qtail != pos && ++n < dmap->nbufs)
+ do_inputintr(dev);
+ } else
+ do_inputintr(dev);
+ spin_unlock_irqrestore(&dmap->lock,flags);
+}
+EXPORT_SYMBOL(DMAbuf_inputintr);
+
+void DMAbuf_init(int dev, int dma1, int dma2)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ /*
+ * NOTE! This routine could be called several times.
+ */
+
+ if (adev && adev->dmap_out == NULL) {
+ if (adev->d == NULL)
+ panic("OSS: audio_devs[%d]->d == NULL\n", dev);
+
+ if (adev->parent_dev) { /* Use DMA map of the parent dev */
+ int parent = adev->parent_dev - 1;
+ adev->dmap_out = audio_devs[parent]->dmap_out;
+ adev->dmap_in = audio_devs[parent]->dmap_in;
+ } else {
+ adev->dmap_out = adev->dmap_in = &adev->dmaps[0];
+ adev->dmap_out->dma = dma1;
+ if (adev->flags & DMA_DUPLEX) {
+ adev->dmap_in = &adev->dmaps[1];
+ adev->dmap_in->dma = dma2;
+ }
+ }
+ /* Persistent DMA buffers allocated here */
+ if (sound_dmap_flag == DMAP_KEEP_ON_CLOSE) {
+ if (adev->dmap_in->raw_buf == NULL)
+ sound_alloc_dmap(adev->dmap_in);
+ if (adev->dmap_out->raw_buf == NULL)
+ sound_alloc_dmap(adev->dmap_out);
+ }
+ }
+}
+
+/* No kernel lock - DMAbuf_activate_recording protected by global cli/sti */
+static unsigned int poll_input(struct file * file, int dev, poll_table *wait)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_in;
+
+ if (!(adev->open_mode & OPEN_READ))
+ return 0;
+ if (dmap->mapping_flags & DMA_MAP_MAPPED) {
+ if (dmap->qlen)
+ return POLLIN | POLLRDNORM;
+ return 0;
+ }
+ if (dmap->dma_mode != DMODE_INPUT) {
+ if (dmap->dma_mode == DMODE_NONE &&
+ adev->enable_bits & PCM_ENABLE_INPUT &&
+ !dmap->qlen && adev->go) {
+ unsigned long flags;
+
+ spin_lock_irqsave(&dmap->lock,flags);
+ DMAbuf_activate_recording(dev, dmap);
+ spin_unlock_irqrestore(&dmap->lock,flags);
+ }
+ return 0;
+ }
+ if (!dmap->qlen)
+ return 0;
+ return POLLIN | POLLRDNORM;
+}
+
+static unsigned int poll_output(struct file * file, int dev, poll_table *wait)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ struct dma_buffparms *dmap = adev->dmap_out;
+
+ if (!(adev->open_mode & OPEN_WRITE))
+ return 0;
+ if (dmap->mapping_flags & DMA_MAP_MAPPED) {
+ if (dmap->qlen)
+ return POLLOUT | POLLWRNORM;
+ return 0;
+ }
+ if (dmap->dma_mode == DMODE_INPUT)
+ return 0;
+ if (dmap->dma_mode == DMODE_NONE)
+ return POLLOUT | POLLWRNORM;
+ if (!DMAbuf_space_in_queue(dev))
+ return 0;
+ return POLLOUT | POLLWRNORM;
+}
+
+unsigned int DMAbuf_poll(struct file * file, int dev, poll_table *wait)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ poll_wait(file, &adev->poll_sleeper, wait);
+ return poll_input(file, dev, wait) | poll_output(file, dev, wait);
+}
+
+void DMAbuf_deinit(int dev)
+{
+ struct audio_operations *adev = audio_devs[dev];
+ /* This routine is called when driver is being unloaded */
+ if (!adev)
+ return;
+
+ /* Persistent DMA buffers deallocated here */
+ if (sound_dmap_flag == DMAP_KEEP_ON_CLOSE) {
+ sound_free_dmap(adev->dmap_out);
+ if (adev->flags & DMA_DUPLEX)
+ sound_free_dmap(adev->dmap_in);
+ }
+}
diff --git a/sound/oss/dmasound/Kconfig b/sound/oss/dmasound/Kconfig
new file mode 100644
index 0000000..f456574
--- /dev/null
+++ b/sound/oss/dmasound/Kconfig
@@ -0,0 +1,45 @@
+config DMASOUND_ATARI
+ tristate "Atari DMA sound support"
+ depends on ATARI && SOUND
+ select DMASOUND
+ help
+ If you want to use the internal audio of your Atari in Linux, answer
+ Y to this question. This will provide a Sun-like /dev/audio,
+ compatible with the Linux/i386 sound system. Otherwise, say N.
+
+ This driver is also available as a module ( = code which can be
+ inserted in and removed from the running kernel whenever you
+ want). If you want to compile it as a module, say M here and read
+ <file:Documentation/kbuild/modules.txt>.
+
+config DMASOUND_PAULA
+ tristate "Amiga DMA sound support"
+ depends on AMIGA && SOUND
+ select DMASOUND
+ help
+ If you want to use the internal audio of your Amiga in Linux, answer
+ Y to this question. This will provide a Sun-like /dev/audio,
+ compatible with the Linux/i386 sound system. Otherwise, say N.
+
+ This driver is also available as a module ( = code which can be
+ inserted in and removed from the running kernel whenever you
+ want). If you want to compile it as a module, say M here and read
+ <file:Documentation/kbuild/modules.txt>.
+
+config DMASOUND_Q40
+ tristate "Q40 sound support"
+ depends on Q40 && SOUND
+ select DMASOUND
+ help
+ If you want to use the internal audio of your Q40 in Linux, answer
+ Y to this question. This will provide a Sun-like /dev/audio,
+ compatible with the Linux/i386 sound system. Otherwise, say N.
+
+ This driver is also available as a module ( = code which can be
+ inserted in and removed from the running kernel whenever you
+ want). If you want to compile it as a module, say M here and read
+ <file:Documentation/kbuild/modules.txt>.
+
+config DMASOUND
+ tristate
+ select SOUND_OSS_CORE
diff --git a/sound/oss/dmasound/Makefile b/sound/oss/dmasound/Makefile
new file mode 100644
index 0000000..3c15316
--- /dev/null
+++ b/sound/oss/dmasound/Makefile
@@ -0,0 +1,7 @@
+#
+# Makefile for the DMA sound driver
+#
+
+obj-$(CONFIG_DMASOUND_ATARI) += dmasound_core.o dmasound_atari.o
+obj-$(CONFIG_DMASOUND_PAULA) += dmasound_core.o dmasound_paula.o
+obj-$(CONFIG_DMASOUND_Q40) += dmasound_core.o dmasound_q40.o
diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h
new file mode 100644
index 0000000..1308d8d
--- /dev/null
+++ b/sound/oss/dmasound/dmasound.h
@@ -0,0 +1,262 @@
+#ifndef _dmasound_h_
+/*
+ * linux/sound/oss/dmasound/dmasound.h
+ *
+ *
+ * Minor numbers for the sound driver.
+ *
+ * Unfortunately Creative called the codec chip of SB as a DSP. For this
+ * reason the /dev/dsp is reserved for digitized audio use. There is a
+ * device for true DSP processors but it will be called something else.
+ * In v3.0 it's /dev/sndproc but this could be a temporary solution.
+ */
+#define _dmasound_h_
+
+#include <linux/types.h>
+
+#define SND_NDEVS 256 /* Number of supported devices */
+#define SND_DEV_CTL 0 /* Control port /dev/mixer */
+#define SND_DEV_SEQ 1 /* Sequencer output /dev/sequencer (FM
+ synthesizer and MIDI output) */
+#define SND_DEV_MIDIN 2 /* Raw midi access */
+#define SND_DEV_DSP 3 /* Digitized voice /dev/dsp */
+#define SND_DEV_AUDIO 4 /* Sparc compatible /dev/audio */
+#define SND_DEV_DSP16 5 /* Like /dev/dsp but 16 bits/sample */
+#define SND_DEV_STATUS 6 /* /dev/sndstat */
+/* #7 not in use now. Was in 2.4. Free for use after v3.0. */
+#define SND_DEV_SEQ2 8 /* /dev/sequencer, level 2 interface */
+#define SND_DEV_SNDPROC 9 /* /dev/sndproc for programmable devices */
+#define SND_DEV_PSS SND_DEV_SNDPROC
+
+/* switch on various prinks */
+#define DEBUG_DMASOUND 1
+
+#define MAX_AUDIO_DEV 5
+#define MAX_MIXER_DEV 4
+#define MAX_SYNTH_DEV 3
+#define MAX_MIDI_DEV 6
+#define MAX_TIMER_DEV 3
+
+#define MAX_CATCH_RADIUS 10
+
+#define le2be16(x) (((x)<<8 & 0xff00) | ((x)>>8 & 0x00ff))
+#define le2be16dbl(x) (((x)<<8 & 0xff00ff00) | ((x)>>8 & 0x00ff00ff))
+
+#define IOCTL_IN(arg, ret) \
+ do { int error = get_user(ret, (int __user *)(arg)); \
+ if (error) return error; \
+ } while (0)
+#define IOCTL_OUT(arg, ret) ioctl_return((int __user *)(arg), ret)
+
+static inline int ioctl_return(int __user *addr, int value)
+{
+ return value < 0 ? value : put_user(value, addr);
+}
+
+
+ /*
+ * Configuration
+ */
+
+#undef HAS_8BIT_TABLES
+
+#if defined(CONFIG_DMASOUND_ATARI) || defined(CONFIG_DMASOUND_ATARI_MODULE) ||\
+ defined(CONFIG_DMASOUND_PAULA) || defined(CONFIG_DMASOUND_PAULA_MODULE) ||\
+ defined(CONFIG_DMASOUND_Q40) || defined(CONFIG_DMASOUND_Q40_MODULE)
+#define HAS_8BIT_TABLES
+#define MIN_BUFFERS 4
+#define MIN_BUFSIZE (1<<12) /* in bytes (- where does this come from ?) */
+#define MIN_FRAG_SIZE 8 /* not 100% sure about this */
+#define MAX_BUFSIZE (1<<17) /* Limit for Amiga is 128 kb */
+#define MAX_FRAG_SIZE 15 /* allow *4 for mono-8 => stereo-16 (for multi) */
+
+#else /* is pmac and multi is off */
+
+#define MIN_BUFFERS 2
+#define MIN_BUFSIZE (1<<8) /* in bytes */
+#define MIN_FRAG_SIZE 8
+#define MAX_BUFSIZE (1<<18) /* this is somewhat arbitrary for pmac */
+#define MAX_FRAG_SIZE 16 /* need to allow *4 for mono-8 => stereo-16 */
+#endif
+
+#define DEFAULT_N_BUFFERS 4
+#define DEFAULT_BUFF_SIZE (1<<15)
+
+ /*
+ * Initialization
+ */
+
+extern int dmasound_init(void);
+#ifdef MODULE
+extern void dmasound_deinit(void);
+#else
+#define dmasound_deinit() do { } while (0)
+#endif
+
+/* description of the set-up applies to either hard or soft settings */
+
+typedef struct {
+ int format; /* AFMT_* */
+ int stereo; /* 0 = mono, 1 = stereo */
+ int size; /* 8/16 bit*/
+ int speed; /* speed */
+} SETTINGS;
+
+ /*
+ * Machine definitions
+ */
+
+typedef struct {
+ const char *name;
+ const char *name2;
+ struct module *owner;
+ void *(*dma_alloc)(unsigned int, gfp_t);
+ void (*dma_free)(void *, unsigned int);
+ int (*irqinit)(void);
+#ifdef MODULE
+ void (*irqcleanup)(void);
+#endif
+ void (*init)(void);
+ void (*silence)(void);
+ int (*setFormat)(int);
+ int (*setVolume)(int);
+ int (*setBass)(int);
+ int (*setTreble)(int);
+ int (*setGain)(int);
+ void (*play)(void);
+ void (*record)(void); /* optional */
+ void (*mixer_init)(void); /* optional */
+ int (*mixer_ioctl)(u_int, u_long); /* optional */
+ int (*write_sq_setup)(void); /* optional */
+ int (*read_sq_setup)(void); /* optional */
+ int (*sq_open)(fmode_t); /* optional */
+ int (*state_info)(char *, size_t); /* optional */
+ void (*abort_read)(void); /* optional */
+ int min_dsp_speed;
+ int max_dsp_speed;
+ int version ;
+ int hardware_afmts ; /* OSS says we only return h'ware info */
+ /* when queried via SNDCTL_DSP_GETFMTS */
+ int capabilities ; /* low-level reply to SNDCTL_DSP_GETCAPS */
+ SETTINGS default_hard ; /* open() or init() should set something valid */
+ SETTINGS default_soft ; /* you can make it look like old OSS, if you want to */
+} MACHINE;
+
+ /*
+ * Low level stuff
+ */
+
+typedef struct {
+ ssize_t (*ct_ulaw)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+ ssize_t (*ct_alaw)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+ ssize_t (*ct_s8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+ ssize_t (*ct_u8)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+ ssize_t (*ct_s16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+ ssize_t (*ct_u16be)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+ ssize_t (*ct_s16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+ ssize_t (*ct_u16le)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+} TRANS;
+
+struct sound_settings {
+ MACHINE mach; /* machine dependent things */
+ SETTINGS hard; /* hardware settings */
+ SETTINGS soft; /* software settings */
+ SETTINGS dsp; /* /dev/dsp default settings */
+ TRANS *trans_write; /* supported translations */
+ int volume_left; /* volume (range is machine dependent) */
+ int volume_right;
+ int bass; /* tone (range is machine dependent) */
+ int treble;
+ int gain;
+ int minDev; /* minor device number currently open */
+ spinlock_t lock;
+};
+
+extern struct sound_settings dmasound;
+
+#ifdef HAS_8BIT_TABLES
+extern char dmasound_ulaw2dma8[];
+extern char dmasound_alaw2dma8[];
+#endif
+
+ /*
+ * Mid level stuff
+ */
+
+static inline int dmasound_set_volume(int volume)
+{
+ return dmasound.mach.setVolume(volume);
+}
+
+static inline int dmasound_set_bass(int bass)
+{
+ return dmasound.mach.setBass ? dmasound.mach.setBass(bass) : 50;
+}
+
+static inline int dmasound_set_treble(int treble)
+{
+ return dmasound.mach.setTreble ? dmasound.mach.setTreble(treble) : 50;
+}
+
+static inline int dmasound_set_gain(int gain)
+{
+ return dmasound.mach.setGain ? dmasound.mach.setGain(gain) : 100;
+}
+
+
+ /*
+ * Sound queue stuff, the heart of the driver
+ */
+
+struct sound_queue {
+ /* buffers allocated for this queue */
+ int numBufs; /* real limits on what the user can have */
+ int bufSize; /* in bytes */
+ char **buffers;
+
+ /* current parameters */
+ int locked ; /* params cannot be modified when != 0 */
+ int user_frags ; /* user requests this many */
+ int user_frag_size ; /* of this size */
+ int max_count; /* actual # fragments <= numBufs */
+ int block_size; /* internal block size in bytes */
+ int max_active; /* in-use fragments <= max_count */
+
+ /* it shouldn't be necessary to declare any of these volatile */
+ int front, rear, count;
+ int rear_size;
+ /*
+ * The use of the playing field depends on the hardware
+ *
+ * Atari, PMac: The number of frames that are loaded/playing
+ *
+ * Amiga: Bit 0 is set: a frame is loaded
+ * Bit 1 is set: a frame is playing
+ */
+ int active;
+ wait_queue_head_t action_queue, open_queue, sync_queue;
+ int non_blocking;
+ int busy, syncing, xruns, died;
+};
+
+#define SLEEP(queue) interruptible_sleep_on_timeout(&queue, HZ)
+#define WAKE_UP(queue) (wake_up_interruptible(&queue))
+
+extern struct sound_queue dmasound_write_sq;
+#define write_sq dmasound_write_sq
+
+extern int dmasound_catchRadius;
+#define catchRadius dmasound_catchRadius
+
+/* define the value to be put in the byte-swap reg in mac-io
+ when we want it to swap for us.
+*/
+#define BS_VAL 1
+
+#define SW_INPUT_VOLUME_SCALE 4
+#define SW_INPUT_VOLUME_DEFAULT (128 / SW_INPUT_VOLUME_SCALE)
+
+extern int expand_read_bal; /* Balance factor for reading */
+extern uint software_input_volume; /* software implemented recording volume! */
+
+#endif /* _dmasound_h_ */
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
new file mode 100644
index 0000000..4d45bd6
--- /dev/null
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -0,0 +1,1618 @@
+/*
+ * linux/sound/oss/dmasound/dmasound_atari.c
+ *
+ * Atari TT and Falcon DMA Sound Driver
+ *
+ * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
+ * prior to 28/01/2001
+ *
+ * 28/01/2001 [0.1] Iain Sandoe
+ * - added versioning
+ * - put in and populated the hardware_afmts field.
+ * [0.2] - put in SNDCTL_DSP_GETCAPS value.
+ * 01/02/2001 [0.3] - put in default hard/soft settings.
+ */
+
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/soundcard.h>
+#include <linux/mm.h>
+#include <linux/spinlock.h>
+#include <linux/interrupt.h>
+
+#include <asm/uaccess.h>
+#include <asm/atariints.h>
+#include <asm/atari_stram.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_ATARI_REVISION 0
+#define DMASOUND_ATARI_EDITION 3
+
+extern void atari_microwire_cmd(int cmd);
+
+static int is_falcon;
+static int write_sq_ignore_int; /* ++TeSche: used for Falcon */
+
+static int expand_bal; /* Balance factor for expanding (not volume!) */
+static int expand_data; /* Data for expanding */
+
+
+/*** Translations ************************************************************/
+
+
+/* ++TeSche: radically changed for new expanding purposes...
+ *
+ * These two routines now deal with copying/expanding/translating the samples
+ * from user space into our buffer at the right frequency. They take care about
+ * how much data there's actually to read, how much buffer space there is and
+ * to convert samples into the right frequency/encoding. They will only work on
+ * complete samples so it may happen they leave some bytes in the input stream
+ * if the user didn't write a multiple of the current sample size. They both
+ * return the number of bytes they've used from both streams so you may detect
+ * such a situation. Luckily all programs should be able to cope with that.
+ *
+ * I think I've optimized anything as far as one can do in plain C, all
+ * variables should fit in registers and the loops are really short. There's
+ * one loop for every possible situation. Writing a more generalized and thus
+ * parameterized loop would only produce slower code. Feel free to optimize
+ * this in assembler if you like. :)
+ *
+ * I think these routines belong here because they're not yet really hardware
+ * independent, especially the fact that the Falcon can play 16bit samples
+ * only in stereo is hardcoded in both of them!
+ *
+ * ++geert: split in even more functions (one per format)
+ */
+
+static ssize_t ata_ct_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_s16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_u16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_s16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ct_u16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_s16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_u16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_s16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+static ssize_t ata_ctx_u16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft);
+
+
+/*** Low level stuff *********************************************************/
+
+
+static void *AtaAlloc(unsigned int size, gfp_t flags);
+static void AtaFree(void *, unsigned int size);
+static int AtaIrqInit(void);
+#ifdef MODULE
+static void AtaIrqCleanUp(void);
+#endif /* MODULE */
+static int AtaSetBass(int bass);
+static int AtaSetTreble(int treble);
+static void TTSilence(void);
+static void TTInit(void);
+static int TTSetFormat(int format);
+static int TTSetVolume(int volume);
+static int TTSetGain(int gain);
+static void FalconSilence(void);
+static void FalconInit(void);
+static int FalconSetFormat(int format);
+static int FalconSetVolume(int volume);
+static void AtaPlayNextFrame(int index);
+static void AtaPlay(void);
+static irqreturn_t AtaInterrupt(int irq, void *dummy);
+
+/*** Mid level stuff *********************************************************/
+
+static void TTMixerInit(void);
+static void FalconMixerInit(void);
+static int AtaMixerIoctl(u_int cmd, u_long arg);
+static int TTMixerIoctl(u_int cmd, u_long arg);
+static int FalconMixerIoctl(u_int cmd, u_long arg);
+static int AtaWriteSqSetup(void);
+static int AtaSqOpen(fmode_t mode);
+static int TTStateInfo(char *buffer, size_t space);
+static int FalconStateInfo(char *buffer, size_t space);
+
+
+/*** Translations ************************************************************/
+
+
+static ssize_t ata_ct_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8
+ : dmasound_alaw2dma8;
+ ssize_t count, used;
+ u_char *p = &frame[*frameUsed];
+
+ count = min_t(unsigned long, userCount, frameLeft);
+ if (dmasound.soft.stereo)
+ count &= ~1;
+ used = count;
+ while (count > 0) {
+ u_char data;
+ if (get_user(data, userPtr++))
+ return -EFAULT;
+ *p++ = table[data];
+ count--;
+ }
+ *frameUsed += used;
+ return used;
+}
+
+
+static ssize_t ata_ct_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+ void *p = &frame[*frameUsed];
+
+ count = min_t(unsigned long, userCount, frameLeft);
+ if (dmasound.soft.stereo)
+ count &= ~1;
+ used = count;
+ if (copy_from_user(p, userPtr, count))
+ return -EFAULT;
+ *frameUsed += used;
+ return used;
+}
+
+
+static ssize_t ata_ct_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+
+ if (!dmasound.soft.stereo) {
+ u_char *p = &frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft);
+ used = count;
+ while (count > 0) {
+ u_char data;
+ if (get_user(data, userPtr++))
+ return -EFAULT;
+ *p++ = data ^ 0x80;
+ count--;
+ }
+ } else {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>1;
+ used = count*2;
+ while (count > 0) {
+ u_short data;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ *p++ = data ^ 0x8080;
+ count--;
+ }
+ }
+ *frameUsed += used;
+ return used;
+}
+
+
+static ssize_t ata_ct_s16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>1;
+ used = count*2;
+ while (count > 0) {
+ u_short data;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ *p++ = data;
+ *p++ = data;
+ count--;
+ }
+ *frameUsed += used*2;
+ } else {
+ void *p = (u_short *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft) & ~3;
+ used = count;
+ if (copy_from_user(p, userPtr, count))
+ return -EFAULT;
+ *frameUsed += used;
+ }
+ return used;
+}
+
+
+static ssize_t ata_ct_u16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>1;
+ used = count*2;
+ while (count > 0) {
+ u_short data;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ data ^= 0x8000;
+ *p++ = data;
+ *p++ = data;
+ count--;
+ }
+ *frameUsed += used*2;
+ } else {
+ u_long *p = (u_long *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>2;
+ used = count*4;
+ while (count > 0) {
+ u_int data;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ *p++ = data ^ 0x80008000;
+ count--;
+ }
+ *frameUsed += used;
+ }
+ return used;
+}
+
+
+static ssize_t ata_ct_s16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+
+ count = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>1;
+ used = count*2;
+ while (count > 0) {
+ u_short data;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ data = le2be16(data);
+ *p++ = data;
+ *p++ = data;
+ count--;
+ }
+ *frameUsed += used*2;
+ } else {
+ u_long *p = (u_long *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>2;
+ used = count*4;
+ while (count > 0) {
+ u_long data;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ data = le2be16dbl(data);
+ *p++ = data;
+ count--;
+ }
+ *frameUsed += used;
+ }
+ return used;
+}
+
+
+static ssize_t ata_ct_u16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+
+ count = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>1;
+ used = count*2;
+ while (count > 0) {
+ u_short data;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ data = le2be16(data) ^ 0x8000;
+ *p++ = data;
+ *p++ = data;
+ }
+ *frameUsed += used*2;
+ } else {
+ u_long *p = (u_long *)&frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft)>>2;
+ used = count;
+ while (count > 0) {
+ u_long data;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ data = le2be16dbl(data) ^ 0x80008000;
+ *p++ = data;
+ count--;
+ }
+ *frameUsed += used;
+ }
+ return used;
+}
+
+
+static ssize_t ata_ctx_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8
+ : dmasound_alaw2dma8;
+ /* this should help gcc to stuff everything into registers */
+ long bal = expand_bal;
+ long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ ssize_t used, usedf;
+
+ used = userCount;
+ usedf = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_char *p = &frame[*frameUsed];
+ u_char data = expand_data;
+ while (frameLeft) {
+ u_char c;
+ if (bal < 0) {
+ if (!userCount)
+ break;
+ if (get_user(c, userPtr++))
+ return -EFAULT;
+ data = table[c];
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ } else {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ u_short data = expand_data;
+ while (frameLeft >= 2) {
+ u_char c;
+ if (bal < 0) {
+ if (userCount < 2)
+ break;
+ if (get_user(c, userPtr++))
+ return -EFAULT;
+ data = table[c] << 8;
+ if (get_user(c, userPtr++))
+ return -EFAULT;
+ data |= table[c];
+ userCount -= 2;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft -= 2;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ }
+ expand_bal = bal;
+ used -= userCount;
+ *frameUsed += usedf-frameLeft;
+ return used;
+}
+
+
+static ssize_t ata_ctx_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ /* this should help gcc to stuff everything into registers */
+ long bal = expand_bal;
+ long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ ssize_t used, usedf;
+
+ used = userCount;
+ usedf = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_char *p = &frame[*frameUsed];
+ u_char data = expand_data;
+ while (frameLeft) {
+ if (bal < 0) {
+ if (!userCount)
+ break;
+ if (get_user(data, userPtr++))
+ return -EFAULT;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ } else {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ u_short data = expand_data;
+ while (frameLeft >= 2) {
+ if (bal < 0) {
+ if (userCount < 2)
+ break;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ userCount -= 2;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft -= 2;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ }
+ expand_bal = bal;
+ used -= userCount;
+ *frameUsed += usedf-frameLeft;
+ return used;
+}
+
+
+static ssize_t ata_ctx_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ /* this should help gcc to stuff everything into registers */
+ long bal = expand_bal;
+ long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ ssize_t used, usedf;
+
+ used = userCount;
+ usedf = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_char *p = &frame[*frameUsed];
+ u_char data = expand_data;
+ while (frameLeft) {
+ if (bal < 0) {
+ if (!userCount)
+ break;
+ if (get_user(data, userPtr++))
+ return -EFAULT;
+ data ^= 0x80;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ } else {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ u_short data = expand_data;
+ while (frameLeft >= 2) {
+ if (bal < 0) {
+ if (userCount < 2)
+ break;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ data ^= 0x8080;
+ userCount -= 2;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft -= 2;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ }
+ expand_bal = bal;
+ used -= userCount;
+ *frameUsed += usedf-frameLeft;
+ return used;
+}
+
+
+static ssize_t ata_ctx_s16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ /* this should help gcc to stuff everything into registers */
+ long bal = expand_bal;
+ long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ ssize_t used, usedf;
+
+ used = userCount;
+ usedf = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ u_short data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 2)
+ break;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ userCount -= 2;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ } else {
+ u_long *p = (u_long *)&frame[*frameUsed];
+ u_long data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 4)
+ break;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ userCount -= 4;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ }
+ expand_bal = bal;
+ used -= userCount;
+ *frameUsed += usedf-frameLeft;
+ return used;
+}
+
+
+static ssize_t ata_ctx_u16be(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ /* this should help gcc to stuff everything into registers */
+ long bal = expand_bal;
+ long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ ssize_t used, usedf;
+
+ used = userCount;
+ usedf = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ u_short data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 2)
+ break;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ data ^= 0x8000;
+ userCount -= 2;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ } else {
+ u_long *p = (u_long *)&frame[*frameUsed];
+ u_long data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 4)
+ break;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ data ^= 0x80008000;
+ userCount -= 4;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ }
+ expand_bal = bal;
+ used -= userCount;
+ *frameUsed += usedf-frameLeft;
+ return used;
+}
+
+
+static ssize_t ata_ctx_s16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ /* this should help gcc to stuff everything into registers */
+ long bal = expand_bal;
+ long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ ssize_t used, usedf;
+
+ used = userCount;
+ usedf = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ u_short data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 2)
+ break;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ data = le2be16(data);
+ userCount -= 2;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ } else {
+ u_long *p = (u_long *)&frame[*frameUsed];
+ u_long data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 4)
+ break;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ data = le2be16dbl(data);
+ userCount -= 4;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ }
+ expand_bal = bal;
+ used -= userCount;
+ *frameUsed += usedf-frameLeft;
+ return used;
+}
+
+
+static ssize_t ata_ctx_u16le(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ /* this should help gcc to stuff everything into registers */
+ long bal = expand_bal;
+ long hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ ssize_t used, usedf;
+
+ used = userCount;
+ usedf = frameLeft;
+ if (!dmasound.soft.stereo) {
+ u_short *p = (u_short *)&frame[*frameUsed];
+ u_short data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 2)
+ break;
+ if (get_user(data, (u_short __user *)userPtr))
+ return -EFAULT;
+ userPtr += 2;
+ data = le2be16(data) ^ 0x8000;
+ userCount -= 2;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ } else {
+ u_long *p = (u_long *)&frame[*frameUsed];
+ u_long data = expand_data;
+ while (frameLeft >= 4) {
+ if (bal < 0) {
+ if (userCount < 4)
+ break;
+ if (get_user(data, (u_int __user *)userPtr))
+ return -EFAULT;
+ userPtr += 4;
+ data = le2be16dbl(data) ^ 0x80008000;
+ userCount -= 4;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft -= 4;
+ bal -= sSpeed;
+ }
+ expand_data = data;
+ }
+ expand_bal = bal;
+ used -= userCount;
+ *frameUsed += usedf-frameLeft;
+ return used;
+}
+
+
+static TRANS transTTNormal = {
+ .ct_ulaw = ata_ct_law,
+ .ct_alaw = ata_ct_law,
+ .ct_s8 = ata_ct_s8,
+ .ct_u8 = ata_ct_u8,
+};
+
+static TRANS transTTExpanding = {
+ .ct_ulaw = ata_ctx_law,
+ .ct_alaw = ata_ctx_law,
+ .ct_s8 = ata_ctx_s8,
+ .ct_u8 = ata_ctx_u8,
+};
+
+static TRANS transFalconNormal = {
+ .ct_ulaw = ata_ct_law,
+ .ct_alaw = ata_ct_law,
+ .ct_s8 = ata_ct_s8,
+ .ct_u8 = ata_ct_u8,
+ .ct_s16be = ata_ct_s16be,
+ .ct_u16be = ata_ct_u16be,
+ .ct_s16le = ata_ct_s16le,
+ .ct_u16le = ata_ct_u16le
+};
+
+static TRANS transFalconExpanding = {
+ .ct_ulaw = ata_ctx_law,
+ .ct_alaw = ata_ctx_law,
+ .ct_s8 = ata_ctx_s8,
+ .ct_u8 = ata_ctx_u8,
+ .ct_s16be = ata_ctx_s16be,
+ .ct_u16be = ata_ctx_u16be,
+ .ct_s16le = ata_ctx_s16le,
+ .ct_u16le = ata_ctx_u16le,
+};
+
+
+/*** Low level stuff *********************************************************/
+
+
+
+/*
+ * Atari (TT/Falcon)
+ */
+
+static void *AtaAlloc(unsigned int size, gfp_t flags)
+{
+ return atari_stram_alloc(size, "dmasound");
+}
+
+static void AtaFree(void *obj, unsigned int size)
+{
+ atari_stram_free( obj );
+}
+
+static int __init AtaIrqInit(void)
+{
+ /* Set up timer A. Timer A
+ will receive a signal upon end of playing from the sound
+ hardware. Furthermore Timer A is able to count events
+ and will cause an interrupt after a programmed number
+ of events. So all we need to keep the music playing is
+ to provide the sound hardware with new data upon
+ an interrupt from timer A. */
+ mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */
+ mfp.tim_dt_a = 1; /* Cause interrupt after first event. */
+ mfp.tim_ct_a = 8; /* Turn on event counting. */
+ /* Register interrupt handler. */
+ request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound",
+ AtaInterrupt);
+ mfp.int_en_a |= 0x20; /* Turn interrupt on. */
+ mfp.int_mk_a |= 0x20;
+ return 1;
+}
+
+#ifdef MODULE
+static void AtaIrqCleanUp(void)
+{
+ mfp.tim_ct_a = 0; /* stop timer */
+ mfp.int_en_a &= ~0x20; /* turn interrupt off */
+ free_irq(IRQ_MFP_TIMA, AtaInterrupt);
+}
+#endif /* MODULE */
+
+
+#define TONE_VOXWARE_TO_DB(v) \
+ (((v) < 0) ? -12 : ((v) > 100) ? 12 : ((v) - 50) * 6 / 25)
+#define TONE_DB_TO_VOXWARE(v) (((v) * 25 + ((v) > 0 ? 5 : -5)) / 6 + 50)
+
+
+static int AtaSetBass(int bass)
+{
+ dmasound.bass = TONE_VOXWARE_TO_DB(bass);
+ atari_microwire_cmd(MW_LM1992_BASS(dmasound.bass));
+ return TONE_DB_TO_VOXWARE(dmasound.bass);
+}
+
+
+static int AtaSetTreble(int treble)
+{
+ dmasound.treble = TONE_VOXWARE_TO_DB(treble);
+ atari_microwire_cmd(MW_LM1992_TREBLE(dmasound.treble));
+ return TONE_DB_TO_VOXWARE(dmasound.treble);
+}
+
+
+
+/*
+ * TT
+ */
+
+
+static void TTSilence(void)
+{
+ tt_dmasnd.ctrl = DMASND_CTRL_OFF;
+ atari_microwire_cmd(MW_LM1992_PSG_HIGH); /* mix in PSG signal 1:1 */
+}
+
+
+static void TTInit(void)
+{
+ int mode, i, idx;
+ const int freq[4] = {50066, 25033, 12517, 6258};
+
+ /* search a frequency that fits into the allowed error range */
+
+ idx = -1;
+ for (i = 0; i < ARRAY_SIZE(freq); i++)
+ /* this isn't as much useful for a TT than for a Falcon, but
+ * then it doesn't hurt very much to implement it for a TT too.
+ */
+ if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) < catchRadius)
+ idx = i;
+ if (idx > -1) {
+ dmasound.soft.speed = freq[idx];
+ dmasound.trans_write = &transTTNormal;
+ } else
+ dmasound.trans_write = &transTTExpanding;
+
+ TTSilence();
+ dmasound.hard = dmasound.soft;
+
+ if (dmasound.hard.speed > 50066) {
+ /* we would need to squeeze the sound, but we won't do that */
+ dmasound.hard.speed = 50066;
+ mode = DMASND_MODE_50KHZ;
+ dmasound.trans_write = &transTTNormal;
+ } else if (dmasound.hard.speed > 25033) {
+ dmasound.hard.speed = 50066;
+ mode = DMASND_MODE_50KHZ;
+ } else if (dmasound.hard.speed > 12517) {
+ dmasound.hard.speed = 25033;
+ mode = DMASND_MODE_25KHZ;
+ } else if (dmasound.hard.speed > 6258) {
+ dmasound.hard.speed = 12517;
+ mode = DMASND_MODE_12KHZ;
+ } else {
+ dmasound.hard.speed = 6258;
+ mode = DMASND_MODE_6KHZ;
+ }
+
+ tt_dmasnd.mode = (dmasound.hard.stereo ?
+ DMASND_MODE_STEREO : DMASND_MODE_MONO) |
+ DMASND_MODE_8BIT | mode;
+
+ expand_bal = -dmasound.soft.speed;
+}
+
+
+static int TTSetFormat(int format)
+{
+ /* TT sound DMA supports only 8bit modes */
+
+ switch (format) {
+ case AFMT_QUERY:
+ return dmasound.soft.format;
+ case AFMT_MU_LAW:
+ case AFMT_A_LAW:
+ case AFMT_S8:
+ case AFMT_U8:
+ break;
+ default:
+ format = AFMT_S8;
+ }
+
+ dmasound.soft.format = format;
+ dmasound.soft.size = 8;
+ if (dmasound.minDev == SND_DEV_DSP) {
+ dmasound.dsp.format = format;
+ dmasound.dsp.size = 8;
+ }
+ TTInit();
+
+ return format;
+}
+
+
+#define VOLUME_VOXWARE_TO_DB(v) \
+ (((v) < 0) ? -40 : ((v) > 100) ? 0 : ((v) * 2) / 5 - 40)
+#define VOLUME_DB_TO_VOXWARE(v) ((((v) + 40) * 5 + 1) / 2)
+
+
+static int TTSetVolume(int volume)
+{
+ dmasound.volume_left = VOLUME_VOXWARE_TO_DB(volume & 0xff);
+ atari_microwire_cmd(MW_LM1992_BALLEFT(dmasound.volume_left));
+ dmasound.volume_right = VOLUME_VOXWARE_TO_DB((volume & 0xff00) >> 8);
+ atari_microwire_cmd(MW_LM1992_BALRIGHT(dmasound.volume_right));
+ return VOLUME_DB_TO_VOXWARE(dmasound.volume_left) |
+ (VOLUME_DB_TO_VOXWARE(dmasound.volume_right) << 8);
+}
+
+
+#define GAIN_VOXWARE_TO_DB(v) \
+ (((v) < 0) ? -80 : ((v) > 100) ? 0 : ((v) * 4) / 5 - 80)
+#define GAIN_DB_TO_VOXWARE(v) ((((v) + 80) * 5 + 1) / 4)
+
+static int TTSetGain(int gain)
+{
+ dmasound.gain = GAIN_VOXWARE_TO_DB(gain);
+ atari_microwire_cmd(MW_LM1992_VOLUME(dmasound.gain));
+ return GAIN_DB_TO_VOXWARE(dmasound.gain);
+}
+
+
+
+/*
+ * Falcon
+ */
+
+
+static void FalconSilence(void)
+{
+ /* stop playback, set sample rate 50kHz for PSG sound */
+ tt_dmasnd.ctrl = DMASND_CTRL_OFF;
+ tt_dmasnd.mode = DMASND_MODE_50KHZ | DMASND_MODE_STEREO | DMASND_MODE_8BIT;
+ tt_dmasnd.int_div = 0; /* STE compatible divider */
+ tt_dmasnd.int_ctrl = 0x0;
+ tt_dmasnd.cbar_src = 0x0000; /* no matrix inputs */
+ tt_dmasnd.cbar_dst = 0x0000; /* no matrix outputs */
+ tt_dmasnd.dac_src = 1; /* connect ADC to DAC, disconnect matrix */
+ tt_dmasnd.adc_src = 3; /* ADC Input = PSG */
+}
+
+
+static void FalconInit(void)
+{
+ int divider, i, idx;
+ const int freq[8] = {49170, 32780, 24585, 19668, 16390, 12292, 9834, 8195};
+
+ /* search a frequency that fits into the allowed error range */
+
+ idx = -1;
+ for (i = 0; i < ARRAY_SIZE(freq); i++)
+ /* if we will tolerate 3% error 8000Hz->8195Hz (2.38%) would
+ * be playable without expanding, but that now a kernel runtime
+ * option
+ */
+ if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) < catchRadius)
+ idx = i;
+ if (idx > -1) {
+ dmasound.soft.speed = freq[idx];
+ dmasound.trans_write = &transFalconNormal;
+ } else
+ dmasound.trans_write = &transFalconExpanding;
+
+ FalconSilence();
+ dmasound.hard = dmasound.soft;
+
+ if (dmasound.hard.size == 16) {
+ /* the Falcon can play 16bit samples only in stereo */
+ dmasound.hard.stereo = 1;
+ }
+
+ if (dmasound.hard.speed > 49170) {
+ /* we would need to squeeze the sound, but we won't do that */
+ dmasound.hard.speed = 49170;
+ divider = 1;
+ dmasound.trans_write = &transFalconNormal;
+ } else if (dmasound.hard.speed > 32780) {
+ dmasound.hard.speed = 49170;
+ divider = 1;
+ } else if (dmasound.hard.speed > 24585) {
+ dmasound.hard.speed = 32780;
+ divider = 2;
+ } else if (dmasound.hard.speed > 19668) {
+ dmasound.hard.speed = 24585;
+ divider = 3;
+ } else if (dmasound.hard.speed > 16390) {
+ dmasound.hard.speed = 19668;
+ divider = 4;
+ } else if (dmasound.hard.speed > 12292) {
+ dmasound.hard.speed = 16390;
+ divider = 5;
+ } else if (dmasound.hard.speed > 9834) {
+ dmasound.hard.speed = 12292;
+ divider = 7;
+ } else if (dmasound.hard.speed > 8195) {
+ dmasound.hard.speed = 9834;
+ divider = 9;
+ } else {
+ dmasound.hard.speed = 8195;
+ divider = 11;
+ }
+ tt_dmasnd.int_div = divider;
+
+ /* Setup Falcon sound DMA for playback */
+ tt_dmasnd.int_ctrl = 0x4; /* Timer A int at play end */
+ tt_dmasnd.track_select = 0x0; /* play 1 track, track 1 */
+ tt_dmasnd.cbar_src = 0x0001; /* DMA(25MHz) --> DAC */
+ tt_dmasnd.cbar_dst = 0x0000;
+ tt_dmasnd.rec_track_select = 0;
+ tt_dmasnd.dac_src = 2; /* connect matrix to DAC */
+ tt_dmasnd.adc_src = 0; /* ADC Input = Mic */
+
+ tt_dmasnd.mode = (dmasound.hard.stereo ?
+ DMASND_MODE_STEREO : DMASND_MODE_MONO) |
+ ((dmasound.hard.size == 8) ?
+ DMASND_MODE_8BIT : DMASND_MODE_16BIT) |
+ DMASND_MODE_6KHZ;
+
+ expand_bal = -dmasound.soft.speed;
+}
+
+
+static int FalconSetFormat(int format)
+{
+ int size;
+ /* Falcon sound DMA supports 8bit and 16bit modes */
+
+ switch (format) {
+ case AFMT_QUERY:
+ return dmasound.soft.format;
+ case AFMT_MU_LAW:
+ case AFMT_A_LAW:
+ case AFMT_U8:
+ case AFMT_S8:
+ size = 8;
+ break;
+ case AFMT_S16_BE:
+ case AFMT_U16_BE:
+ case AFMT_S16_LE:
+ case AFMT_U16_LE:
+ size = 16;
+ break;
+ default: /* :-) */
+ size = 8;
+ format = AFMT_S8;
+ }
+
+ dmasound.soft.format = format;
+ dmasound.soft.size = size;
+ if (dmasound.minDev == SND_DEV_DSP) {
+ dmasound.dsp.format = format;
+ dmasound.dsp.size = dmasound.soft.size;
+ }
+
+ FalconInit();
+
+ return format;
+}
+
+
+/* This is for the Falcon output *attenuation* in 1.5dB steps,
+ * i.e. output level from 0 to -22.5dB in -1.5dB steps.
+ */
+#define VOLUME_VOXWARE_TO_ATT(v) \
+ ((v) < 0 ? 15 : (v) > 100 ? 0 : 15 - (v) * 3 / 20)
+#define VOLUME_ATT_TO_VOXWARE(v) (100 - (v) * 20 / 3)
+
+
+static int FalconSetVolume(int volume)
+{
+ dmasound.volume_left = VOLUME_VOXWARE_TO_ATT(volume & 0xff);
+ dmasound.volume_right = VOLUME_VOXWARE_TO_ATT((volume & 0xff00) >> 8);
+ tt_dmasnd.output_atten = dmasound.volume_left << 8 | dmasound.volume_right << 4;
+ return VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) |
+ VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8;
+}
+
+
+static void AtaPlayNextFrame(int index)
+{
+ char *start, *end;
+
+ /* used by AtaPlay() if all doubts whether there really is something
+ * to be played are already wiped out.
+ */
+ start = write_sq.buffers[write_sq.front];
+ end = start+((write_sq.count == index) ? write_sq.rear_size
+ : write_sq.block_size);
+ /* end might not be a legal virtual address. */
+ DMASNDSetEnd(virt_to_phys(end - 1) + 1);
+ DMASNDSetBase(virt_to_phys(start));
+ /* Since only an even number of samples per frame can
+ be played, we might lose one byte here. (TO DO) */
+ write_sq.front = (write_sq.front+1) % write_sq.max_count;
+ write_sq.active++;
+ tt_dmasnd.ctrl = DMASND_CTRL_ON | DMASND_CTRL_REPEAT;
+}
+
+
+static void AtaPlay(void)
+{
+ /* ++TeSche: Note that write_sq.active is no longer just a flag but
+ * holds the number of frames the DMA is currently programmed for
+ * instead, may be 0, 1 (currently being played) or 2 (pre-programmed).
+ *
+ * Changes done to write_sq.count and write_sq.active are a bit more
+ * subtle again so now I must admit I also prefer disabling the irq
+ * here rather than considering all possible situations. But the point
+ * is that disabling the irq doesn't have any bad influence on this
+ * version of the driver as we benefit from having pre-programmed the
+ * DMA wherever possible: There's no need to reload the DMA at the
+ * exact time of an interrupt but only at some time while the
+ * pre-programmed frame is playing!
+ */
+ atari_disable_irq(IRQ_MFP_TIMA);
+
+ if (write_sq.active == 2 || /* DMA is 'full' */
+ write_sq.count <= 0) { /* nothing to do */
+ atari_enable_irq(IRQ_MFP_TIMA);
+ return;
+ }
+
+ if (write_sq.active == 0) {
+ /* looks like there's nothing 'in' the DMA yet, so try
+ * to put two frames into it (at least one is available).
+ */
+ if (write_sq.count == 1 &&
+ write_sq.rear_size < write_sq.block_size &&
+ !write_sq.syncing) {
+ /* hmmm, the only existing frame is not
+ * yet filled and we're not syncing?
+ */
+ atari_enable_irq(IRQ_MFP_TIMA);
+ return;
+ }
+ AtaPlayNextFrame(1);
+ if (write_sq.count == 1) {
+ /* no more frames */
+ atari_enable_irq(IRQ_MFP_TIMA);
+ return;
+ }
+ if (write_sq.count == 2 &&
+ write_sq.rear_size < write_sq.block_size &&
+ !write_sq.syncing) {
+ /* hmmm, there were two frames, but the second
+ * one is not yet filled and we're not syncing?
+ */
+ atari_enable_irq(IRQ_MFP_TIMA);
+ return;
+ }
+ AtaPlayNextFrame(2);
+ } else {
+ /* there's already a frame being played so we may only stuff
+ * one new into the DMA, but even if this may be the last
+ * frame existing the previous one is still on write_sq.count.
+ */
+ if (write_sq.count == 2 &&
+ write_sq.rear_size < write_sq.block_size &&
+ !write_sq.syncing) {
+ /* hmmm, the only existing frame is not
+ * yet filled and we're not syncing?
+ */
+ atari_enable_irq(IRQ_MFP_TIMA);
+ return;
+ }
+ AtaPlayNextFrame(2);
+ }
+ atari_enable_irq(IRQ_MFP_TIMA);
+}
+
+
+static irqreturn_t AtaInterrupt(int irq, void *dummy)
+{
+#if 0
+ /* ++TeSche: if you should want to test this... */
+ static int cnt;
+ if (write_sq.active == 2)
+ if (++cnt == 10) {
+ /* simulate losing an interrupt */
+ cnt = 0;
+ return IRQ_HANDLED;
+ }
+#endif
+ spin_lock(&dmasound.lock);
+ if (write_sq_ignore_int && is_falcon) {
+ /* ++TeSche: Falcon only: ignore first irq because it comes
+ * immediately after starting a frame. after that, irqs come
+ * (almost) like on the TT.
+ */
+ write_sq_ignore_int = 0;
+ return IRQ_HANDLED;
+ }
+
+ if (!write_sq.active) {
+ /* playing was interrupted and sq_reset() has already cleared
+ * the sq variables, so better don't do anything here.
+ */
+ WAKE_UP(write_sq.sync_queue);
+ return IRQ_HANDLED;
+ }
+
+ /* Probably ;) one frame is finished. Well, in fact it may be that a
+ * pre-programmed one is also finished because there has been a long
+ * delay in interrupt delivery and we've completely lost one, but
+ * there's no way to detect such a situation. In such a case the last
+ * frame will be played more than once and the situation will recover
+ * as soon as the irq gets through.
+ */
+ write_sq.count--;
+ write_sq.active--;
+
+ if (!write_sq.active) {
+ tt_dmasnd.ctrl = DMASND_CTRL_OFF;
+ write_sq_ignore_int = 1;
+ }
+
+ WAKE_UP(write_sq.action_queue);
+ /* At least one block of the queue is free now
+ so wake up a writing process blocked because
+ of a full queue. */
+
+ if ((write_sq.active != 1) || (write_sq.count != 1))
+ /* We must be a bit carefully here: write_sq.count indicates the
+ * number of buffers used and not the number of frames to be
+ * played. If write_sq.count==1 and write_sq.active==1 that
+ * means the only remaining frame was already programmed
+ * earlier (and is currently running) so we mustn't call
+ * AtaPlay() here, otherwise we'll play one frame too much.
+ */
+ AtaPlay();
+
+ if (!write_sq.active) WAKE_UP(write_sq.sync_queue);
+ /* We are not playing after AtaPlay(), so there
+ is nothing to play any more. Wake up a process
+ waiting for audio output to drain. */
+ spin_unlock(&dmasound.lock);
+ return IRQ_HANDLED;
+}
+
+
+/*** Mid level stuff *********************************************************/
+
+
+/*
+ * /dev/mixer abstraction
+ */
+
+#define RECLEVEL_VOXWARE_TO_GAIN(v) \
+ ((v) < 0 ? 0 : (v) > 100 ? 15 : (v) * 3 / 20)
+#define RECLEVEL_GAIN_TO_VOXWARE(v) (((v) * 20 + 2) / 3)
+
+
+static void __init TTMixerInit(void)
+{
+ atari_microwire_cmd(MW_LM1992_VOLUME(0));
+ dmasound.volume_left = 0;
+ atari_microwire_cmd(MW_LM1992_BALLEFT(0));
+ dmasound.volume_right = 0;
+ atari_microwire_cmd(MW_LM1992_BALRIGHT(0));
+ atari_microwire_cmd(MW_LM1992_TREBLE(0));
+ atari_microwire_cmd(MW_LM1992_BASS(0));
+}
+
+static void __init FalconMixerInit(void)
+{
+ dmasound.volume_left = (tt_dmasnd.output_atten & 0xf00) >> 8;
+ dmasound.volume_right = (tt_dmasnd.output_atten & 0xf0) >> 4;
+}
+
+static int AtaMixerIoctl(u_int cmd, u_long arg)
+{
+ int data;
+ unsigned long flags;
+ switch (cmd) {
+ case SOUND_MIXER_READ_SPEAKER:
+ if (is_falcon || MACH_IS_TT) {
+ int porta;
+ spin_lock_irqsave(&dmasound.lock, flags);
+ sound_ym.rd_data_reg_sel = 14;
+ porta = sound_ym.rd_data_reg_sel;
+ spin_unlock_irqrestore(&dmasound.lock, flags);
+ return IOCTL_OUT(arg, porta & 0x40 ? 0 : 100);
+ }
+ break;
+ case SOUND_MIXER_WRITE_VOLUME:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_volume(data));
+ case SOUND_MIXER_WRITE_SPEAKER:
+ if (is_falcon || MACH_IS_TT) {
+ int porta;
+ IOCTL_IN(arg, data);
+ spin_lock_irqsave(&dmasound.lock, flags);
+ sound_ym.rd_data_reg_sel = 14;
+ porta = (sound_ym.rd_data_reg_sel & ~0x40) |
+ (data < 50 ? 0x40 : 0);
+ sound_ym.wd_data = porta;
+ spin_unlock_irqrestore(&dmasound.lock, flags);
+ return IOCTL_OUT(arg, porta & 0x40 ? 0 : 100);
+ }
+ }
+ return -EINVAL;
+}
+
+
+static int TTMixerIoctl(u_int cmd, u_long arg)
+{
+ int data;
+ switch (cmd) {
+ case SOUND_MIXER_READ_RECMASK:
+ return IOCTL_OUT(arg, 0);
+ case SOUND_MIXER_READ_DEVMASK:
+ return IOCTL_OUT(arg,
+ SOUND_MASK_VOLUME | SOUND_MASK_TREBLE | SOUND_MASK_BASS |
+ (MACH_IS_TT ? SOUND_MASK_SPEAKER : 0));
+ case SOUND_MIXER_READ_STEREODEVS:
+ return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
+ case SOUND_MIXER_READ_VOLUME:
+ return IOCTL_OUT(arg,
+ VOLUME_DB_TO_VOXWARE(dmasound.volume_left) |
+ (VOLUME_DB_TO_VOXWARE(dmasound.volume_right) << 8));
+ case SOUND_MIXER_READ_BASS:
+ return IOCTL_OUT(arg, TONE_DB_TO_VOXWARE(dmasound.bass));
+ case SOUND_MIXER_READ_TREBLE:
+ return IOCTL_OUT(arg, TONE_DB_TO_VOXWARE(dmasound.treble));
+ case SOUND_MIXER_READ_OGAIN:
+ return IOCTL_OUT(arg, GAIN_DB_TO_VOXWARE(dmasound.gain));
+ case SOUND_MIXER_WRITE_BASS:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_bass(data));
+ case SOUND_MIXER_WRITE_TREBLE:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_treble(data));
+ case SOUND_MIXER_WRITE_OGAIN:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_gain(data));
+ }
+ return AtaMixerIoctl(cmd, arg);
+}
+
+static int FalconMixerIoctl(u_int cmd, u_long arg)
+{
+ int data;
+ switch (cmd) {
+ case SOUND_MIXER_READ_RECMASK:
+ return IOCTL_OUT(arg, SOUND_MASK_MIC);
+ case SOUND_MIXER_READ_DEVMASK:
+ return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC | SOUND_MASK_SPEAKER);
+ case SOUND_MIXER_READ_STEREODEVS:
+ return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_MIC);
+ case SOUND_MIXER_READ_VOLUME:
+ return IOCTL_OUT(arg,
+ VOLUME_ATT_TO_VOXWARE(dmasound.volume_left) |
+ VOLUME_ATT_TO_VOXWARE(dmasound.volume_right) << 8);
+ case SOUND_MIXER_READ_CAPS:
+ return IOCTL_OUT(arg, SOUND_CAP_EXCL_INPUT);
+ case SOUND_MIXER_WRITE_MIC:
+ IOCTL_IN(arg, data);
+ tt_dmasnd.input_gain =
+ RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 |
+ RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff);
+ /* fall thru, return set value */
+ case SOUND_MIXER_READ_MIC:
+ return IOCTL_OUT(arg,
+ RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) |
+ RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain & 0xf) << 8);
+ }
+ return AtaMixerIoctl(cmd, arg);
+}
+
+static int AtaWriteSqSetup(void)
+{
+ write_sq_ignore_int = 0;
+ return 0 ;
+}
+
+static int AtaSqOpen(fmode_t mode)
+{
+ write_sq_ignore_int = 1;
+ return 0 ;
+}
+
+static int TTStateInfo(char *buffer, size_t space)
+{
+ int len = 0;
+ len += sprintf(buffer+len, "\tvol left %ddB [-40... 0]\n",
+ dmasound.volume_left);
+ len += sprintf(buffer+len, "\tvol right %ddB [-40... 0]\n",
+ dmasound.volume_right);
+ len += sprintf(buffer+len, "\tbass %ddB [-12...+12]\n",
+ dmasound.bass);
+ len += sprintf(buffer+len, "\ttreble %ddB [-12...+12]\n",
+ dmasound.treble);
+ if (len >= space) {
+ printk(KERN_ERR "dmasound_atari: overflowed state buffer alloc.\n") ;
+ len = space ;
+ }
+ return len;
+}
+
+static int FalconStateInfo(char *buffer, size_t space)
+{
+ int len = 0;
+ len += sprintf(buffer+len, "\tvol left %ddB [-22.5 ... 0]\n",
+ dmasound.volume_left);
+ len += sprintf(buffer+len, "\tvol right %ddB [-22.5 ... 0]\n",
+ dmasound.volume_right);
+ if (len >= space) {
+ printk(KERN_ERR "dmasound_atari: overflowed state buffer alloc.\n") ;
+ len = space ;
+ }
+ return len;
+}
+
+
+/*** Machine definitions *****************************************************/
+
+static SETTINGS def_hard_falcon = {
+ .format = AFMT_S8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 8195
+} ;
+
+static SETTINGS def_hard_tt = {
+ .format = AFMT_S8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 12517
+} ;
+
+static SETTINGS def_soft = {
+ .format = AFMT_U8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 8000
+} ;
+
+static MACHINE machTT = {
+ .name = "Atari",
+ .name2 = "TT",
+ .owner = THIS_MODULE,
+ .dma_alloc = AtaAlloc,
+ .dma_free = AtaFree,
+ .irqinit = AtaIrqInit,
+#ifdef MODULE
+ .irqcleanup = AtaIrqCleanUp,
+#endif /* MODULE */
+ .init = TTInit,
+ .silence = TTSilence,
+ .setFormat = TTSetFormat,
+ .setVolume = TTSetVolume,
+ .setBass = AtaSetBass,
+ .setTreble = AtaSetTreble,
+ .setGain = TTSetGain,
+ .play = AtaPlay,
+ .mixer_init = TTMixerInit,
+ .mixer_ioctl = TTMixerIoctl,
+ .write_sq_setup = AtaWriteSqSetup,
+ .sq_open = AtaSqOpen,
+ .state_info = TTStateInfo,
+ .min_dsp_speed = 6258,
+ .version = ((DMASOUND_ATARI_REVISION<<8) | DMASOUND_ATARI_EDITION),
+ .hardware_afmts = AFMT_S8, /* h'ware-supported formats *only* here */
+ .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
+};
+
+static MACHINE machFalcon = {
+ .name = "Atari",
+ .name2 = "FALCON",
+ .dma_alloc = AtaAlloc,
+ .dma_free = AtaFree,
+ .irqinit = AtaIrqInit,
+#ifdef MODULE
+ .irqcleanup = AtaIrqCleanUp,
+#endif /* MODULE */
+ .init = FalconInit,
+ .silence = FalconSilence,
+ .setFormat = FalconSetFormat,
+ .setVolume = FalconSetVolume,
+ .setBass = AtaSetBass,
+ .setTreble = AtaSetTreble,
+ .play = AtaPlay,
+ .mixer_init = FalconMixerInit,
+ .mixer_ioctl = FalconMixerIoctl,
+ .write_sq_setup = AtaWriteSqSetup,
+ .sq_open = AtaSqOpen,
+ .state_info = FalconStateInfo,
+ .min_dsp_speed = 8195,
+ .version = ((DMASOUND_ATARI_REVISION<<8) | DMASOUND_ATARI_EDITION),
+ .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
+ .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
+};
+
+
+/*** Config & Setup **********************************************************/
+
+
+static int __init dmasound_atari_init(void)
+{
+ if (MACH_IS_ATARI && ATARIHW_PRESENT(PCM_8BIT)) {
+ if (ATARIHW_PRESENT(CODEC)) {
+ dmasound.mach = machFalcon;
+ dmasound.mach.default_soft = def_soft ;
+ dmasound.mach.default_hard = def_hard_falcon ;
+ is_falcon = 1;
+ } else if (ATARIHW_PRESENT(MICROWIRE)) {
+ dmasound.mach = machTT;
+ dmasound.mach.default_soft = def_soft ;
+ dmasound.mach.default_hard = def_hard_tt ;
+ is_falcon = 0;
+ } else
+ return -ENODEV;
+ if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0)
+ return dmasound_init();
+ else {
+ printk("DMA sound driver: Timer A interrupt already in use\n");
+ return -EBUSY;
+ }
+ }
+ return -ENODEV;
+}
+
+static void __exit dmasound_atari_cleanup(void)
+{
+ dmasound_deinit();
+}
+
+module_init(dmasound_atari_init);
+module_exit(dmasound_atari_cleanup);
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
new file mode 100644
index 0000000..793b7f4
--- /dev/null
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -0,0 +1,1549 @@
+/*
+ * linux/sound/oss/dmasound/dmasound_core.c
+ *
+ *
+ * OSS/Free compatible Atari TT/Falcon and Amiga DMA sound driver for
+ * Linux/m68k
+ * Extended to support Power Macintosh for Linux/ppc by Paul Mackerras
+ *
+ * (c) 1995 by Michael Schlueter & Michael Marte
+ *
+ * Michael Schlueter (michael@duck.syd.de) did the basic structure of the VFS
+ * interface and the u-law to signed byte conversion.
+ *
+ * Michael Marte (marte@informatik.uni-muenchen.de) did the sound queue,
+ * /dev/mixer, /dev/sndstat and complemented the VFS interface. He would like
+ * to thank:
+ * - Michael Schlueter for initial ideas and documentation on the MFP and
+ * the DMA sound hardware.
+ * - Therapy? for their CD 'Troublegum' which really made me rock.
+ *
+ * /dev/sndstat is based on code by Hannu Savolainen, the author of the
+ * VoxWare family of drivers.
+ *
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License. See the file COPYING in the main directory of this archive
+ * for more details.
+ *
+ * History:
+ *
+ * 1995/8/25 First release
+ *
+ * 1995/9/02 Roman Hodek:
+ * - Fixed atari_stram_alloc() call, the timer
+ * programming and several race conditions
+ * 1995/9/14 Roman Hodek:
+ * - After some discussion with Michael Schlueter,
+ * revised the interrupt disabling
+ * - Slightly speeded up U8->S8 translation by using
+ * long operations where possible
+ * - Added 4:3 interpolation for /dev/audio
+ *
+ * 1995/9/20 Torsten Scherer:
+ * - Fixed a bug in sq_write and changed /dev/audio
+ * converting to play at 12517Hz instead of 6258Hz.
+ *
+ * 1995/9/23 Torsten Scherer:
+ * - Changed sq_interrupt() and sq_play() to pre-program
+ * the DMA for another frame while there's still one
+ * running. This allows the IRQ response to be
+ * arbitrarily delayed and playing will still continue.
+ *
+ * 1995/10/14 Guenther Kelleter, Torsten Scherer:
+ * - Better support for Falcon audio (the Falcon doesn't
+ * raise an IRQ at the end of a frame, but at the
+ * beginning instead!). uses 'if (codec_dma)' in lots
+ * of places to simply switch between Falcon and TT
+ * code.
+ *
+ * 1995/11/06 Torsten Scherer:
+ * - Started introducing a hardware abstraction scheme
+ * (may perhaps also serve for Amigas?)
+ * - Can now play samples at almost all frequencies by
+ * means of a more generalized expand routine
+ * - Takes a good deal of care to cut data only at
+ * sample sizes
+ * - Buffer size is now a kernel runtime option
+ * - Implemented fsync() & several minor improvements
+ * Guenther Kelleter:
+ * - Useful hints and bug fixes
+ * - Cross-checked it for Falcons
+ *
+ * 1996/3/9 Geert Uytterhoeven:
+ * - Support added for Amiga, A-law, 16-bit little
+ * endian.
+ * - Unification to drivers/sound/dmasound.c.
+ *
+ * 1996/4/6 Martin Mitchell:
+ * - Updated to 1.3 kernel.
+ *
+ * 1996/6/13 Topi Kanerva:
+ * - Fixed things that were broken (mainly the amiga
+ * 14-bit routines)
+ * - /dev/sndstat shows now the real hardware frequency
+ * - The lowpass filter is disabled by default now
+ *
+ * 1996/9/25 Geert Uytterhoeven:
+ * - Modularization
+ *
+ * 1998/6/10 Andreas Schwab:
+ * - Converted to use sound_core
+ *
+ * 1999/12/28 Richard Zidlicky:
+ * - Added support for Q40
+ *
+ * 2000/2/27 Geert Uytterhoeven:
+ * - Clean up and split the code into 4 parts:
+ * o dmasound_core: machine-independent code
+ * o dmasound_atari: Atari TT and Falcon support
+ * o dmasound_awacs: Apple PowerMac support
+ * o dmasound_paula: Amiga support
+ *
+ * 2000/3/25 Geert Uytterhoeven:
+ * - Integration of dmasound_q40
+ * - Small clean ups
+ *
+ * 2001/01/26 [1.0] Iain Sandoe
+ * - make /dev/sndstat show revision & edition info.
+ * - since dmasound.mach.sq_setup() can fail on pmac
+ * its type has been changed to int and the returns
+ * are checked.
+ * [1.1] - stop missing translations from being called.
+ * 2001/02/08 [1.2] - remove unused translation tables & move machine-
+ * specific tables to low-level.
+ * - return correct info. for SNDCTL_DSP_GETFMTS.
+ * [1.3] - implement SNDCTL_DSP_GETCAPS fully.
+ * [1.4] - make /dev/sndstat text length usage deterministic.
+ * - make /dev/sndstat call to low-level
+ * dmasound.mach.state_info() pass max space to ll driver.
+ * - tidy startup banners and output info.
+ * [1.5] - tidy up a little (removed some unused #defines in
+ * dmasound.h)
+ * - fix up HAS_RECORD conditionalisation.
+ * - add record code in places it is missing...
+ * - change buf-sizes to bytes to allow < 1kb for pmac
+ * if user param entry is < 256 the value is taken to
+ * be in kb > 256 is taken to be in bytes.
+ * - make default buff/frag params conditional on
+ * machine to allow smaller values for pmac.
+ * - made the ioctls, read & write comply with the OSS
+ * rules on setting params.
+ * - added parsing of _setup() params for record.
+ * 2001/04/04 [1.6] - fix bug where sample rates higher than maximum were
+ * being reported as OK.
+ * - fix open() to return -EBUSY as per OSS doc. when
+ * audio is in use - this is independent of O_NOBLOCK.
+ * - fix bug where SNDCTL_DSP_POST was blocking.
+ */
+
+ /* Record capability notes 30/01/2001:
+ * At present these observations apply only to pmac LL driver (the only one
+ * that can do record, at present). However, if other LL drivers for machines
+ * with record are added they may apply.
+ *
+ * The fragment parameters for the record and play channels are separate.
+ * However, if the driver is opened O_RDWR there is no way (in the current OSS
+ * API) to specify their values independently for the record and playback
+ * channels. Since the only common factor between the input & output is the
+ * sample rate (on pmac) it should be possible to open /dev/dspX O_WRONLY and
+ * /dev/dspY O_RDONLY. The input & output channels could then have different
+ * characteristics (other than the first that sets sample rate claiming the
+ * right to set it for ever). As it stands, the format, channels, number of
+ * bits & sample rate are assumed to be common. In the future perhaps these
+ * should be the responsibility of the LL driver - and then if a card really
+ * does not share items between record & playback they can be specified
+ * separately.
+*/
+
+/* Thread-safeness of shared_resources notes: 31/01/2001
+ * If the user opens O_RDWR and then splits record & play between two threads
+ * both of which inherit the fd - and then starts changing things from both
+ * - we will have difficulty telling.
+ *
+ * It's bad application coding - but ...
+ * TODO: think about how to sort this out... without bogging everything down in
+ * semaphores.
+ *
+ * Similarly, the OSS spec says "all changes to parameters must be between
+ * open() and the first read() or write(). - and a bit later on (by
+ * implication) "between SNDCTL_DSP_RESET and the first read() or write() after
+ * it". If the app is multi-threaded and this rule is broken between threads
+ * we will have trouble spotting it - and the fault will be rather obscure :-(
+ *
+ * We will try and put out at least a kmsg if we see it happen... but I think
+ * it will be quite hard to trap it with an -EXXX return... because we can't
+ * see the fault until after the damage is done.
+*/
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/sound.h>
+#include <linux/init.h>
+#include <linux/soundcard.h>
+#include <linux/poll.h>
+#include <linux/smp_lock.h>
+
+#include <asm/uaccess.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_CORE_REVISION 1
+#define DMASOUND_CORE_EDITION 6
+
+ /*
+ * Declarations
+ */
+
+int dmasound_catchRadius = 0;
+module_param(dmasound_catchRadius, int, 0);
+
+static unsigned int numWriteBufs = DEFAULT_N_BUFFERS;
+module_param(numWriteBufs, int, 0);
+static unsigned int writeBufSize = DEFAULT_BUFF_SIZE ; /* in bytes */
+module_param(writeBufSize, int, 0);
+
+MODULE_LICENSE("GPL");
+
+#ifdef MODULE
+static int sq_unit = -1;
+static int mixer_unit = -1;
+static int state_unit = -1;
+static int irq_installed;
+#endif /* MODULE */
+
+/* control over who can modify resources shared between play/record */
+static fmode_t shared_resource_owner;
+static int shared_resources_initialised;
+
+ /*
+ * Mid level stuff
+ */
+
+struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED };
+
+static inline void sound_silence(void)
+{
+ dmasound.mach.silence(); /* _MUST_ stop DMA */
+}
+
+static inline int sound_set_format(int format)
+{
+ return dmasound.mach.setFormat(format);
+}
+
+
+static int sound_set_speed(int speed)
+{
+ if (speed < 0)
+ return dmasound.soft.speed;
+
+ /* trap out-of-range speed settings.
+ at present we allow (arbitrarily) low rates - using soft
+ up-conversion - but we can't allow > max because there is
+ no soft down-conversion.
+ */
+ if (dmasound.mach.max_dsp_speed &&
+ (speed > dmasound.mach.max_dsp_speed))
+ speed = dmasound.mach.max_dsp_speed ;
+
+ dmasound.soft.speed = speed;
+
+ if (dmasound.minDev == SND_DEV_DSP)
+ dmasound.dsp.speed = dmasound.soft.speed;
+
+ return dmasound.soft.speed;
+}
+
+static int sound_set_stereo(int stereo)
+{
+ if (stereo < 0)
+ return dmasound.soft.stereo;
+
+ stereo = !!stereo; /* should be 0 or 1 now */
+
+ dmasound.soft.stereo = stereo;
+ if (dmasound.minDev == SND_DEV_DSP)
+ dmasound.dsp.stereo = stereo;
+
+ return stereo;
+}
+
+static ssize_t sound_copy_translate(TRANS *trans, const u_char __user *userPtr,
+ size_t userCount, u_char frame[],
+ ssize_t *frameUsed, ssize_t frameLeft)
+{
+ ssize_t (*ct_func)(const u_char __user *, size_t, u_char *, ssize_t *, ssize_t);
+
+ switch (dmasound.soft.format) {
+ case AFMT_MU_LAW:
+ ct_func = trans->ct_ulaw;
+ break;
+ case AFMT_A_LAW:
+ ct_func = trans->ct_alaw;
+ break;
+ case AFMT_S8:
+ ct_func = trans->ct_s8;
+ break;
+ case AFMT_U8:
+ ct_func = trans->ct_u8;
+ break;
+ case AFMT_S16_BE:
+ ct_func = trans->ct_s16be;
+ break;
+ case AFMT_U16_BE:
+ ct_func = trans->ct_u16be;
+ break;
+ case AFMT_S16_LE:
+ ct_func = trans->ct_s16le;
+ break;
+ case AFMT_U16_LE:
+ ct_func = trans->ct_u16le;
+ break;
+ default:
+ return 0;
+ }
+ /* if the user has requested a non-existent translation don't try
+ to call it but just return 0 bytes moved
+ */
+ if (ct_func)
+ return ct_func(userPtr, userCount, frame, frameUsed, frameLeft);
+ return 0;
+}
+
+ /*
+ * /dev/mixer abstraction
+ */
+
+static struct {
+ int busy;
+ int modify_counter;
+} mixer;
+
+static int mixer_open(struct inode *inode, struct file *file)
+{
+ if (!try_module_get(dmasound.mach.owner))
+ return -ENODEV;
+ mixer.busy = 1;
+ return 0;
+}
+
+static int mixer_release(struct inode *inode, struct file *file)
+{
+ lock_kernel();
+ mixer.busy = 0;
+ module_put(dmasound.mach.owner);
+ unlock_kernel();
+ return 0;
+}
+static int mixer_ioctl(struct inode *inode, struct file *file, u_int cmd,
+ u_long arg)
+{
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE)
+ mixer.modify_counter++;
+ switch (cmd) {
+ case OSS_GETVERSION:
+ return IOCTL_OUT(arg, SOUND_VERSION);
+ case SOUND_MIXER_INFO:
+ {
+ mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, dmasound.mach.name2, sizeof(info.id));
+ strlcpy(info.name, dmasound.mach.name2, sizeof(info.name));
+ info.modify_counter = mixer.modify_counter;
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ }
+ }
+ if (dmasound.mach.mixer_ioctl)
+ return dmasound.mach.mixer_ioctl(cmd, arg);
+ return -EINVAL;
+}
+
+static const struct file_operations mixer_fops =
+{
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .ioctl = mixer_ioctl,
+ .open = mixer_open,
+ .release = mixer_release,
+};
+
+static void mixer_init(void)
+{
+#ifndef MODULE
+ int mixer_unit;
+#endif
+ mixer_unit = register_sound_mixer(&mixer_fops, -1);
+ if (mixer_unit < 0)
+ return;
+
+ mixer.busy = 0;
+ dmasound.treble = 0;
+ dmasound.bass = 0;
+ if (dmasound.mach.mixer_init)
+ dmasound.mach.mixer_init();
+}
+
+
+ /*
+ * Sound queue stuff, the heart of the driver
+ */
+
+struct sound_queue dmasound_write_sq;
+static void sq_reset_output(void) ;
+
+static int sq_allocate_buffers(struct sound_queue *sq, int num, int size)
+{
+ int i;
+
+ if (sq->buffers)
+ return 0;
+ sq->numBufs = num;
+ sq->bufSize = size;
+ sq->buffers = kmalloc (num * sizeof(char *), GFP_KERNEL);
+ if (!sq->buffers)
+ return -ENOMEM;
+ for (i = 0; i < num; i++) {
+ sq->buffers[i] = dmasound.mach.dma_alloc(size, GFP_KERNEL);
+ if (!sq->buffers[i]) {
+ while (i--)
+ dmasound.mach.dma_free(sq->buffers[i], size);
+ kfree(sq->buffers);
+ sq->buffers = NULL;
+ return -ENOMEM;
+ }
+ }
+ return 0;
+}
+
+static void sq_release_buffers(struct sound_queue *sq)
+{
+ int i;
+
+ if (sq->buffers) {
+ for (i = 0; i < sq->numBufs; i++)
+ dmasound.mach.dma_free(sq->buffers[i], sq->bufSize);
+ kfree(sq->buffers);
+ sq->buffers = NULL;
+ }
+}
+
+
+static int sq_setup(struct sound_queue *sq)
+{
+ int (*setup_func)(void) = NULL;
+ int hard_frame ;
+
+ if (sq->locked) { /* are we already set? - and not changeable */
+#ifdef DEBUG_DMASOUND
+printk("dmasound_core: tried to sq_setup a locked queue\n") ;
+#endif
+ return -EINVAL ;
+ }
+ sq->locked = 1 ; /* don't think we have a race prob. here _check_ */
+
+ /* make sure that the parameters are set up
+ This should have been done already...
+ */
+
+ dmasound.mach.init();
+
+ /* OK. If the user has set fragment parameters explicitly, then we
+ should leave them alone... as long as they are valid.
+ Invalid user fragment params can occur if we allow the whole buffer
+ to be used when the user requests the fragments sizes (with no soft
+ x-lation) and then the user subsequently sets a soft x-lation that
+ requires increased internal buffering.
+
+ Othwerwise (if the user did not set them) OSS says that we should
+ select frag params on the basis of 0.5 s output & 0.1 s input
+ latency. (TODO. For now we will copy in the defaults.)
+ */
+
+ if (sq->user_frags <= 0) {
+ sq->max_count = sq->numBufs ;
+ sq->max_active = sq->numBufs ;
+ sq->block_size = sq->bufSize;
+ /* set up the user info */
+ sq->user_frags = sq->numBufs ;
+ sq->user_frag_size = sq->bufSize ;
+ sq->user_frag_size *=
+ (dmasound.soft.size * (dmasound.soft.stereo+1) ) ;
+ sq->user_frag_size /=
+ (dmasound.hard.size * (dmasound.hard.stereo+1) ) ;
+ } else {
+ /* work out requested block size */
+ sq->block_size = sq->user_frag_size ;
+ sq->block_size *=
+ (dmasound.hard.size * (dmasound.hard.stereo+1) ) ;
+ sq->block_size /=
+ (dmasound.soft.size * (dmasound.soft.stereo+1) ) ;
+ /* the user wants to write frag-size chunks */
+ sq->block_size *= dmasound.hard.speed ;
+ sq->block_size /= dmasound.soft.speed ;
+ /* this only works for size values which are powers of 2 */
+ hard_frame =
+ (dmasound.hard.size * (dmasound.hard.stereo+1))/8 ;
+ sq->block_size += (hard_frame - 1) ;
+ sq->block_size &= ~(hard_frame - 1) ; /* make sure we are aligned */
+ /* let's just check for obvious mistakes */
+ if ( sq->block_size <= 0 || sq->block_size > sq->bufSize) {
+#ifdef DEBUG_DMASOUND
+printk("dmasound_core: invalid frag size (user set %d)\n", sq->user_frag_size) ;
+#endif
+ sq->block_size = sq->bufSize ;
+ }
+ if ( sq->user_frags <= sq->numBufs ) {
+ sq->max_count = sq->user_frags ;
+ /* if user has set max_active - then use it */
+ sq->max_active = (sq->max_active <= sq->max_count) ?
+ sq->max_active : sq->max_count ;
+ } else {
+#ifdef DEBUG_DMASOUND
+printk("dmasound_core: invalid frag count (user set %d)\n", sq->user_frags) ;
+#endif
+ sq->max_count =
+ sq->max_active = sq->numBufs ;
+ }
+ }
+ sq->front = sq->count = sq->rear_size = 0;
+ sq->syncing = 0;
+ sq->active = 0;
+
+ if (sq == &write_sq) {
+ sq->rear = -1;
+ setup_func = dmasound.mach.write_sq_setup;
+ }
+ if (setup_func)
+ return setup_func();
+ return 0 ;
+}
+
+static inline void sq_play(void)
+{
+ dmasound.mach.play();
+}
+
+static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft,
+ loff_t *ppos)
+{
+ ssize_t uWritten = 0;
+ u_char *dest;
+ ssize_t uUsed = 0, bUsed, bLeft;
+ unsigned long flags ;
+
+ /* ++TeSche: Is something like this necessary?
+ * Hey, that's an honest question! Or does any other part of the
+ * filesystem already checks this situation? I really don't know.
+ */
+ if (uLeft == 0)
+ return 0;
+
+ /* implement any changes we have made to the soft/hard params.
+ this is not satisfactory really, all we have done up to now is to
+ say what we would like - there hasn't been any real checking of capability
+ */
+
+ if (shared_resources_initialised == 0) {
+ dmasound.mach.init() ;
+ shared_resources_initialised = 1 ;
+ }
+
+ /* set up the sq if it is not already done. This may seem a dumb place
+ to do it - but it is what OSS requires. It means that write() can
+ return memory allocation errors. To avoid this possibility use the
+ GETBLKSIZE or GETOSPACE ioctls (after you've fiddled with all the
+ params you want to change) - these ioctls also force the setup.
+ */
+
+ if (write_sq.locked == 0) {
+ if ((uWritten = sq_setup(&write_sq)) < 0) return uWritten ;
+ uWritten = 0 ;
+ }
+
+/* FIXME: I think that this may be the wrong behaviour when we get strapped
+ for time and the cpu is close to being (or actually) behind in sending data.
+ - because we've lost the time that the N samples, already in the buffer,
+ would have given us to get here with the next lot from the user.
+*/
+ /* The interrupt doesn't start to play the last, incomplete frame.
+ * Thus we can append to it without disabling the interrupts! (Note
+ * also that write_sq.rear isn't affected by the interrupt.)
+ */
+
+ /* as of 1.6 this behaviour changes if SNDCTL_DSP_POST has been issued:
+ this will mimic the behaviour of syncing and allow the sq_play() to
+ queue a partial fragment. Since sq_play() may/will be called from
+ the IRQ handler - at least on Pmac we have to deal with it.
+ The strategy - possibly not optimum - is to kill _POST status if we
+ get here. This seems, at least, reasonable - in the sense that POST
+ is supposed to indicate that we might not write before the queue
+ is drained - and if we get here in time then it does not apply.
+ */
+
+ spin_lock_irqsave(&dmasound.lock, flags);
+ write_sq.syncing &= ~2 ; /* take out POST status */
+ spin_unlock_irqrestore(&dmasound.lock, flags);
+
+ if (write_sq.count > 0 &&
+ (bLeft = write_sq.block_size-write_sq.rear_size) > 0) {
+ dest = write_sq.buffers[write_sq.rear];
+ bUsed = write_sq.rear_size;
+ uUsed = sound_copy_translate(dmasound.trans_write, src, uLeft,
+ dest, &bUsed, bLeft);
+ if (uUsed <= 0)
+ return uUsed;
+ src += uUsed;
+ uWritten += uUsed;
+ uLeft = (uUsed <= uLeft) ? (uLeft - uUsed) : 0 ; /* paranoia */
+ write_sq.rear_size = bUsed;
+ }
+
+ while (uLeft) {
+ while (write_sq.count >= write_sq.max_active) {
+ sq_play();
+ if (write_sq.non_blocking)
+ return uWritten > 0 ? uWritten : -EAGAIN;
+ SLEEP(write_sq.action_queue);
+ if (signal_pending(current))
+ return uWritten > 0 ? uWritten : -EINTR;
+ }
+
+ /* Here, we can avoid disabling the interrupt by first
+ * copying and translating the data, and then updating
+ * the write_sq variables. Until this is done, the interrupt
+ * won't see the new frame and we can work on it
+ * undisturbed.
+ */
+
+ dest = write_sq.buffers[(write_sq.rear+1) % write_sq.max_count];
+ bUsed = 0;
+ bLeft = write_sq.block_size;
+ uUsed = sound_copy_translate(dmasound.trans_write, src, uLeft,
+ dest, &bUsed, bLeft);
+ if (uUsed <= 0)
+ break;
+ src += uUsed;
+ uWritten += uUsed;
+ uLeft = (uUsed <= uLeft) ? (uLeft - uUsed) : 0 ; /* paranoia */
+ if (bUsed) {
+ write_sq.rear = (write_sq.rear+1) % write_sq.max_count;
+ write_sq.rear_size = bUsed;
+ write_sq.count++;
+ }
+ } /* uUsed may have been 0 */
+
+ sq_play();
+
+ return uUsed < 0? uUsed: uWritten;
+}
+
+static unsigned int sq_poll(struct file *file, struct poll_table_struct *wait)
+{
+ unsigned int mask = 0;
+ int retVal;
+
+ if (write_sq.locked == 0) {
+ if ((retVal = sq_setup(&write_sq)) < 0)
+ return retVal;
+ return 0;
+ }
+ if (file->f_mode & FMODE_WRITE )
+ poll_wait(file, &write_sq.action_queue, wait);
+ if (file->f_mode & FMODE_WRITE)
+ if (write_sq.count < write_sq.max_active || write_sq.block_size - write_sq.rear_size > 0)
+ mask |= POLLOUT | POLLWRNORM;
+ return mask;
+
+}
+
+static inline void sq_init_waitqueue(struct sound_queue *sq)
+{
+ init_waitqueue_head(&sq->action_queue);
+ init_waitqueue_head(&sq->open_queue);
+ init_waitqueue_head(&sq->sync_queue);
+ sq->busy = 0;
+}
+
+#if 0 /* blocking open() */
+static inline void sq_wake_up(struct sound_queue *sq, struct file *file,
+ fmode_t mode)
+{
+ if (file->f_mode & mode) {
+ sq->busy = 0; /* CHECK: IS THIS OK??? */
+ WAKE_UP(sq->open_queue);
+ }
+}
+#endif
+
+static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode,
+ int numbufs, int bufsize)
+{
+ int rc = 0;
+
+ if (file->f_mode & mode) {
+ if (sq->busy) {
+#if 0 /* blocking open() */
+ rc = -EBUSY;
+ if (file->f_flags & O_NONBLOCK)
+ return rc;
+ rc = -EINTR;
+ while (sq->busy) {
+ SLEEP(sq->open_queue);
+ if (signal_pending(current))
+ return rc;
+ }
+ rc = 0;
+#else
+ /* OSS manual says we will return EBUSY regardless
+ of O_NOBLOCK.
+ */
+ return -EBUSY ;
+#endif
+ }
+ sq->busy = 1; /* Let's play spot-the-race-condition */
+
+ /* allocate the default number & size of buffers.
+ (i.e. specified in _setup() or as module params)
+ can't be changed at the moment - but _could_ be perhaps
+ in the setfragments ioctl.
+ */
+ if (( rc = sq_allocate_buffers(sq, numbufs, bufsize))) {
+#if 0 /* blocking open() */
+ sq_wake_up(sq, file, mode);
+#else
+ sq->busy = 0 ;
+#endif
+ return rc;
+ }
+
+ sq->non_blocking = file->f_flags & O_NONBLOCK;
+ }
+ return rc;
+}
+
+#define write_sq_init_waitqueue() sq_init_waitqueue(&write_sq)
+#if 0 /* blocking open() */
+#define write_sq_wake_up(file) sq_wake_up(&write_sq, file, FMODE_WRITE)
+#endif
+#define write_sq_release_buffers() sq_release_buffers(&write_sq)
+#define write_sq_open(file) \
+ sq_open2(&write_sq, file, FMODE_WRITE, numWriteBufs, writeBufSize )
+
+static int sq_open(struct inode *inode, struct file *file)
+{
+ int rc;
+
+ if (!try_module_get(dmasound.mach.owner))
+ return -ENODEV;
+
+ rc = write_sq_open(file); /* checks the f_mode */
+ if (rc)
+ goto out;
+ if (file->f_mode & FMODE_READ) {
+ /* TODO: if O_RDWR, release any resources grabbed by write part */
+ rc = -ENXIO ; /* I think this is what is required by open(2) */
+ goto out;
+ }
+
+ if (dmasound.mach.sq_open)
+ dmasound.mach.sq_open(file->f_mode);
+
+ /* CHECK whether this is sensible - in the case that dsp0 could be opened
+ O_RDONLY and dsp1 could be opened O_WRONLY
+ */
+
+ dmasound.minDev = iminor(inode) & 0x0f;
+
+ /* OK. - we should make some attempt at consistency. At least the H'ware
+ options should be set with a valid mode. We will make it that the LL
+ driver must supply defaults for hard & soft params.
+ */
+
+ if (shared_resource_owner == 0) {
+ /* you can make this AFMT_U8/mono/8K if you want to mimic old
+ OSS behaviour - while we still have soft translations ;-) */
+ dmasound.soft = dmasound.mach.default_soft ;
+ dmasound.dsp = dmasound.mach.default_soft ;
+ dmasound.hard = dmasound.mach.default_hard ;
+ }
+
+#ifndef DMASOUND_STRICT_OSS_COMPLIANCE
+ /* none of the current LL drivers can actually do this "native" at the moment
+ OSS does not really require us to supply /dev/audio if we can't do it.
+ */
+ if (dmasound.minDev == SND_DEV_AUDIO) {
+ sound_set_speed(8000);
+ sound_set_stereo(0);
+ sound_set_format(AFMT_MU_LAW);
+ }
+#endif
+
+ return 0;
+ out:
+ module_put(dmasound.mach.owner);
+ return rc;
+}
+
+static void sq_reset_output(void)
+{
+ sound_silence(); /* this _must_ stop DMA, we might be about to lose the buffers */
+ write_sq.active = 0;
+ write_sq.count = 0;
+ write_sq.rear_size = 0;
+ /* write_sq.front = (write_sq.rear+1) % write_sq.max_count;*/
+ write_sq.front = 0 ;
+ write_sq.rear = -1 ; /* same as for set-up */
+
+ /* OK - we can unlock the parameters and fragment settings */
+ write_sq.locked = 0 ;
+ write_sq.user_frags = 0 ;
+ write_sq.user_frag_size = 0 ;
+}
+
+static void sq_reset(void)
+{
+ sq_reset_output() ;
+ /* we could consider resetting the shared_resources_owner here... but I
+ think it is probably still rather non-obvious to application writer
+ */
+
+ /* we release everything else though */
+ shared_resources_initialised = 0 ;
+}
+
+static int sq_fsync(struct file *filp, struct dentry *dentry)
+{
+ int rc = 0;
+ int timeout = 5;
+
+ write_sq.syncing |= 1;
+ sq_play(); /* there may be an incomplete frame waiting */
+
+ while (write_sq.active) {
+ SLEEP(write_sq.sync_queue);
+ if (signal_pending(current)) {
+ /* While waiting for audio output to drain, an
+ * interrupt occurred. Stop audio output immediately
+ * and clear the queue. */
+ sq_reset_output();
+ rc = -EINTR;
+ break;
+ }
+ if (!--timeout) {
+ printk(KERN_WARNING "dmasound: Timeout draining output\n");
+ sq_reset_output();
+ rc = -EIO;
+ break;
+ }
+ }
+
+ /* flag no sync regardless of whether we had a DSP_POST or not */
+ write_sq.syncing = 0 ;
+ return rc;
+}
+
+static int sq_release(struct inode *inode, struct file *file)
+{
+ int rc = 0;
+
+ lock_kernel();
+
+ if (file->f_mode & FMODE_WRITE) {
+ if (write_sq.busy)
+ rc = sq_fsync(file, file->f_path.dentry);
+
+ sq_reset_output() ; /* make sure dma is stopped and all is quiet */
+ write_sq_release_buffers();
+ write_sq.busy = 0;
+ }
+
+ if (file->f_mode & shared_resource_owner) { /* it's us that has them */
+ shared_resource_owner = 0 ;
+ shared_resources_initialised = 0 ;
+ dmasound.hard = dmasound.mach.default_hard ;
+ }
+
+ module_put(dmasound.mach.owner);
+
+#if 0 /* blocking open() */
+ /* Wake up a process waiting for the queue being released.
+ * Note: There may be several processes waiting for a call
+ * to open() returning. */
+
+ /* Iain: hmm I don't understand this next comment ... */
+ /* There is probably a DOS atack here. They change the mode flag. */
+ /* XXX add check here,*/
+ read_sq_wake_up(file); /* checks f_mode */
+ write_sq_wake_up(file); /* checks f_mode */
+#endif /* blocking open() */
+
+ unlock_kernel();
+
+ return rc;
+}
+
+/* here we see if we have a right to modify format, channels, size and so on
+ if no-one else has claimed it already then we do...
+
+ TODO: We might change this to mask O_RDWR such that only one or the other channel
+ is the owner - if we have problems.
+*/
+
+static int shared_resources_are_mine(fmode_t md)
+{
+ if (shared_resource_owner)
+ return (shared_resource_owner & md) != 0;
+ else {
+ shared_resource_owner = md ;
+ return 1 ;
+ }
+}
+
+/* if either queue is locked we must deny the right to change shared params
+*/
+
+static int queues_are_quiescent(void)
+{
+ if (write_sq.locked)
+ return 0 ;
+ return 1 ;
+}
+
+/* check and set a queue's fragments per user's wishes...
+ we will check against the pre-defined literals and the actual sizes.
+ This is a bit fraught - because soft translations can mess with our
+ buffer requirements *after* this call - OSS says "call setfrags first"
+*/
+
+/* It is possible to replace all the -EINVAL returns with an override that
+ just puts the allowable value in. This may be what many OSS apps require
+*/
+
+static int set_queue_frags(struct sound_queue *sq, int bufs, int size)
+{
+ if (sq->locked) {
+#ifdef DEBUG_DMASOUND
+printk("dmasound_core: tried to set_queue_frags on a locked queue\n") ;
+#endif
+ return -EINVAL ;
+ }
+
+ if ((size < MIN_FRAG_SIZE) || (size > MAX_FRAG_SIZE))
+ return -EINVAL ;
+ size = (1<<size) ; /* now in bytes */
+ if (size > sq->bufSize)
+ return -EINVAL ; /* this might still not work */
+
+ if (bufs <= 0)
+ return -EINVAL ;
+ if (bufs > sq->numBufs) /* the user is allowed say "don't care" with 0x7fff */
+ bufs = sq->numBufs ;
+
+ /* there is, currently, no way to specify max_active separately
+ from max_count. This could be a LL driver issue - I guess
+ if there is a requirement for these values to be different then
+ we will have to pass that info. up to this level.
+ */
+ sq->user_frags =
+ sq->max_active = bufs ;
+ sq->user_frag_size = size ;
+
+ return 0 ;
+}
+
+static int sq_ioctl(struct inode *inode, struct file *file, u_int cmd,
+ u_long arg)
+{
+ int val, result;
+ u_long fmt;
+ int data;
+ int size, nbufs;
+ audio_buf_info info;
+
+ switch (cmd) {
+ case SNDCTL_DSP_RESET:
+ sq_reset();
+ return 0;
+ break ;
+ case SNDCTL_DSP_GETFMTS:
+ fmt = dmasound.mach.hardware_afmts ; /* this is what OSS says.. */
+ return IOCTL_OUT(arg, fmt);
+ break ;
+ case SNDCTL_DSP_GETBLKSIZE:
+ /* this should tell the caller about bytes that the app can
+ read/write - the app doesn't care about our internal buffers.
+ We force sq_setup() here as per OSS 1.1 (which should
+ compute the values necessary).
+ Since there is no mechanism to specify read/write separately, for
+ fds opened O_RDWR, the write_sq values will, arbitrarily, overwrite
+ the read_sq ones.
+ */
+ size = 0 ;
+ if (file->f_mode & FMODE_WRITE) {
+ if ( !write_sq.locked )
+ sq_setup(&write_sq) ;
+ size = write_sq.user_frag_size ;
+ }
+ return IOCTL_OUT(arg, size);
+ break ;
+ case SNDCTL_DSP_POST:
+ /* all we are going to do is to tell the LL that any
+ partial frags can be queued for output.
+ The LL will have to clear this flag when last output
+ is queued.
+ */
+ write_sq.syncing |= 0x2 ;
+ sq_play() ;
+ return 0 ;
+ case SNDCTL_DSP_SYNC:
+ /* This call, effectively, has the same behaviour as SNDCTL_DSP_RESET
+ except that it waits for output to finish before resetting
+ everything - read, however, is killed imediately.
+ */
+ result = 0 ;
+ if (file->f_mode & FMODE_WRITE) {
+ result = sq_fsync(file, file->f_path.dentry);
+ sq_reset_output() ;
+ }
+ /* if we are the shared resource owner then release them */
+ if (file->f_mode & shared_resource_owner)
+ shared_resources_initialised = 0 ;
+ return result ;
+ break ;
+ case SOUND_PCM_READ_RATE:
+ return IOCTL_OUT(arg, dmasound.soft.speed);
+ case SNDCTL_DSP_SPEED:
+ /* changing this on the fly will have weird effects on the sound.
+ Where there are rate conversions implemented in soft form - it
+ will cause the _ctx_xxx() functions to be substituted.
+ However, there doesn't appear to be any reason to dis-allow it from
+ a driver pov.
+ */
+ if (shared_resources_are_mine(file->f_mode)) {
+ IOCTL_IN(arg, data);
+ data = sound_set_speed(data) ;
+ shared_resources_initialised = 0 ;
+ return IOCTL_OUT(arg, data);
+ } else
+ return -EINVAL ;
+ break ;
+ /* OSS says these next 4 actions are undefined when the device is
+ busy/active - we will just return -EINVAL.
+ To be allowed to change one - (a) you have to own the right
+ (b) the queue(s) must be quiescent
+ */
+ case SNDCTL_DSP_STEREO:
+ if (shared_resources_are_mine(file->f_mode) &&
+ queues_are_quiescent()) {
+ IOCTL_IN(arg, data);
+ shared_resources_initialised = 0 ;
+ return IOCTL_OUT(arg, sound_set_stereo(data));
+ } else
+ return -EINVAL ;
+ break ;
+ case SOUND_PCM_WRITE_CHANNELS:
+ if (shared_resources_are_mine(file->f_mode) &&
+ queues_are_quiescent()) {
+ IOCTL_IN(arg, data);
+ /* the user might ask for 20 channels, we will return 1 or 2 */
+ shared_resources_initialised = 0 ;
+ return IOCTL_OUT(arg, sound_set_stereo(data-1)+1);
+ } else
+ return -EINVAL ;
+ break ;
+ case SNDCTL_DSP_SETFMT:
+ if (shared_resources_are_mine(file->f_mode) &&
+ queues_are_quiescent()) {
+ int format;
+ IOCTL_IN(arg, data);
+ shared_resources_initialised = 0 ;
+ format = sound_set_format(data);
+ result = IOCTL_OUT(arg, format);
+ if (result < 0)
+ return result;
+ if (format != data && data != AFMT_QUERY)
+ return -EINVAL;
+ return 0;
+ } else
+ return -EINVAL ;
+ case SNDCTL_DSP_SUBDIVIDE:
+ return -EINVAL ;
+ case SNDCTL_DSP_SETFRAGMENT:
+ /* we can do this independently for the two queues - with the
+ proviso that for fds opened O_RDWR we cannot separate the
+ actions and both queues will be set per the last call.
+ NOTE: this does *NOT* actually set the queue up - merely
+ registers our intentions.
+ */
+ IOCTL_IN(arg, data);
+ result = 0 ;
+ nbufs = (data >> 16) & 0x7fff ; /* 0x7fff is 'use maximum' */
+ size = data & 0xffff;
+ if (file->f_mode & FMODE_WRITE) {
+ result = set_queue_frags(&write_sq, nbufs, size) ;
+ if (result)
+ return result ;
+ }
+ /* NOTE: this return value is irrelevant - OSS specifically says that
+ the value is 'random' and that the user _must_ check the actual
+ frags values using SNDCTL_DSP_GETBLKSIZE or similar */
+ return IOCTL_OUT(arg, data);
+ break ;
+ case SNDCTL_DSP_GETOSPACE:
+ /*
+ */
+ if (file->f_mode & FMODE_WRITE) {
+ if ( !write_sq.locked )
+ sq_setup(&write_sq) ;
+ info.fragments = write_sq.max_active - write_sq.count;
+ info.fragstotal = write_sq.max_active;
+ info.fragsize = write_sq.user_frag_size;
+ info.bytes = info.fragments * info.fragsize;
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ } else
+ return -EINVAL ;
+ break ;
+ case SNDCTL_DSP_GETCAPS:
+ val = dmasound.mach.capabilities & 0xffffff00;
+ return IOCTL_OUT(arg,val);
+
+ default:
+ return mixer_ioctl(inode, file, cmd, arg);
+ }
+ return -EINVAL;
+}
+
+static const struct file_operations sq_fops =
+{
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .write = sq_write,
+ .poll = sq_poll,
+ .ioctl = sq_ioctl,
+ .open = sq_open,
+ .release = sq_release,
+};
+
+static int sq_init(void)
+{
+ const struct file_operations *fops = &sq_fops;
+#ifndef MODULE
+ int sq_unit;
+#endif
+
+ sq_unit = register_sound_dsp(fops, -1);
+ if (sq_unit < 0) {
+ printk(KERN_ERR "dmasound_core: couldn't register fops\n") ;
+ return sq_unit ;
+ }
+
+ write_sq_init_waitqueue();
+
+ /* These parameters will be restored for every clean open()
+ * in the case of multiple open()s (e.g. dsp0 & dsp1) they
+ * will be set so long as the shared resources have no owner.
+ */
+
+ if (shared_resource_owner == 0) {
+ dmasound.soft = dmasound.mach.default_soft ;
+ dmasound.hard = dmasound.mach.default_hard ;
+ dmasound.dsp = dmasound.mach.default_soft ;
+ shared_resources_initialised = 0 ;
+ }
+ return 0 ;
+}
+
+
+ /*
+ * /dev/sndstat
+ */
+
+/* we allow more space for record-enabled because there are extra output lines.
+ the number here must include the amount we are prepared to give to the low-level
+ driver.
+*/
+
+#define STAT_BUFF_LEN 768
+
+/* this is how much space we will allow the low-level driver to use
+ in the stat buffer. Currently, 2 * (80 character line + <NL>).
+ We do not police this (it is up to the ll driver to be honest).
+*/
+
+#define LOW_LEVEL_STAT_ALLOC 162
+
+static struct {
+ int busy;
+ char buf[STAT_BUFF_LEN]; /* state.buf should not overflow! */
+ int len, ptr;
+} state;
+
+/* publish this function for use by low-level code, if required */
+
+static char *get_afmt_string(int afmt)
+{
+ switch(afmt) {
+ case AFMT_MU_LAW:
+ return "mu-law";
+ break;
+ case AFMT_A_LAW:
+ return "A-law";
+ break;
+ case AFMT_U8:
+ return "unsigned 8 bit";
+ break;
+ case AFMT_S8:
+ return "signed 8 bit";
+ break;
+ case AFMT_S16_BE:
+ return "signed 16 bit BE";
+ break;
+ case AFMT_U16_BE:
+ return "unsigned 16 bit BE";
+ break;
+ case AFMT_S16_LE:
+ return "signed 16 bit LE";
+ break;
+ case AFMT_U16_LE:
+ return "unsigned 16 bit LE";
+ break;
+ case 0:
+ return "format not set" ;
+ break ;
+ default:
+ break ;
+ }
+ return "ERROR: Unsupported AFMT_XXXX code" ;
+}
+
+static int state_open(struct inode *inode, struct file *file)
+{
+ char *buffer = state.buf;
+ int len = 0;
+
+ if (state.busy)
+ return -EBUSY;
+
+ if (!try_module_get(dmasound.mach.owner))
+ return -ENODEV;
+ state.ptr = 0;
+ state.busy = 1;
+
+ len += sprintf(buffer+len, "%sDMA sound driver rev %03d :\n",
+ dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) +
+ ((dmasound.mach.version>>8) & 0x0f));
+ len += sprintf(buffer+len,
+ "Core driver edition %02d.%02d : %s driver edition %02d.%02d\n",
+ DMASOUND_CORE_REVISION, DMASOUND_CORE_EDITION, dmasound.mach.name2,
+ (dmasound.mach.version >> 8), (dmasound.mach.version & 0xff)) ;
+
+ /* call the low-level module to fill in any stat info. that it has
+ if present. Maximum buffer usage is specified.
+ */
+
+ if (dmasound.mach.state_info)
+ len += dmasound.mach.state_info(buffer+len,
+ (size_t) LOW_LEVEL_STAT_ALLOC) ;
+
+ /* make usage of the state buffer as deterministic as poss.
+ exceptional conditions could cause overrun - and this is flagged as
+ a kernel error.
+ */
+
+ /* formats and settings */
+
+ len += sprintf(buffer+len,"\t\t === Formats & settings ===\n") ;
+ len += sprintf(buffer+len,"Parameter %20s%20s\n","soft","hard") ;
+ len += sprintf(buffer+len,"Format :%20s%20s\n",
+ get_afmt_string(dmasound.soft.format),
+ get_afmt_string(dmasound.hard.format));
+
+ len += sprintf(buffer+len,"Samp Rate:%14d s/sec%14d s/sec\n",
+ dmasound.soft.speed, dmasound.hard.speed);
+
+ len += sprintf(buffer+len,"Channels :%20s%20s\n",
+ dmasound.soft.stereo ? "stereo" : "mono",
+ dmasound.hard.stereo ? "stereo" : "mono" );
+
+ /* sound queue status */
+
+ len += sprintf(buffer+len,"\t\t === Sound Queue status ===\n");
+ len += sprintf(buffer+len,"Allocated:%8s%6s\n","Buffers","Size") ;
+ len += sprintf(buffer+len,"%9s:%8d%6d\n",
+ "write", write_sq.numBufs, write_sq.bufSize) ;
+ len += sprintf(buffer+len,
+ "Current : MaxFrg FragSiz MaxAct Frnt Rear "
+ "Cnt RrSize A B S L xruns\n") ;
+ len += sprintf(buffer+len,"%9s:%7d%8d%7d%5d%5d%4d%7d%2d%2d%2d%2d%7d\n",
+ "write", write_sq.max_count, write_sq.block_size,
+ write_sq.max_active, write_sq.front, write_sq.rear,
+ write_sq.count, write_sq.rear_size, write_sq.active,
+ write_sq.busy, write_sq.syncing, write_sq.locked, write_sq.xruns) ;
+#ifdef DEBUG_DMASOUND
+printk("dmasound: stat buffer used %d bytes\n", len) ;
+#endif
+
+ if (len >= STAT_BUFF_LEN)
+ printk(KERN_ERR "dmasound_core: stat buffer overflowed!\n");
+
+ state.len = len;
+ return 0;
+}
+
+static int state_release(struct inode *inode, struct file *file)
+{
+ lock_kernel();
+ state.busy = 0;
+ module_put(dmasound.mach.owner);
+ unlock_kernel();
+ return 0;
+}
+
+static ssize_t state_read(struct file *file, char __user *buf, size_t count,
+ loff_t *ppos)
+{
+ int n = state.len - state.ptr;
+ if (n > count)
+ n = count;
+ if (n <= 0)
+ return 0;
+ if (copy_to_user(buf, &state.buf[state.ptr], n))
+ return -EFAULT;
+ state.ptr += n;
+ return n;
+}
+
+static const struct file_operations state_fops = {
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .read = state_read,
+ .open = state_open,
+ .release = state_release,
+};
+
+static int state_init(void)
+{
+#ifndef MODULE
+ int state_unit;
+#endif
+ state_unit = register_sound_special(&state_fops, SND_DEV_STATUS);
+ if (state_unit < 0)
+ return state_unit ;
+ state.busy = 0;
+ return 0 ;
+}
+
+
+ /*
+ * Config & Setup
+ *
+ * This function is called by _one_ chipset-specific driver
+ */
+
+int dmasound_init(void)
+{
+ int res ;
+#ifdef MODULE
+ if (irq_installed)
+ return -EBUSY;
+#endif
+
+ /* Set up sound queue, /dev/audio and /dev/dsp. */
+
+ /* Set default settings. */
+ if ((res = sq_init()) < 0)
+ return res ;
+
+ /* Set up /dev/sndstat. */
+ if ((res = state_init()) < 0)
+ return res ;
+
+ /* Set up /dev/mixer. */
+ mixer_init();
+
+ if (!dmasound.mach.irqinit()) {
+ printk(KERN_ERR "DMA sound driver: Interrupt initialization failed\n");
+ return -ENODEV;
+ }
+#ifdef MODULE
+ irq_installed = 1;
+#endif
+
+ printk(KERN_INFO "%s DMA sound driver rev %03d installed\n",
+ dmasound.mach.name, (DMASOUND_CORE_REVISION<<4) +
+ ((dmasound.mach.version>>8) & 0x0f));
+ printk(KERN_INFO
+ "Core driver edition %02d.%02d : %s driver edition %02d.%02d\n",
+ DMASOUND_CORE_REVISION, DMASOUND_CORE_EDITION, dmasound.mach.name2,
+ (dmasound.mach.version >> 8), (dmasound.mach.version & 0xff)) ;
+ printk(KERN_INFO "Write will use %4d fragments of %7d bytes as default\n",
+ numWriteBufs, writeBufSize) ;
+ return 0;
+}
+
+#ifdef MODULE
+
+void dmasound_deinit(void)
+{
+ if (irq_installed) {
+ sound_silence();
+ dmasound.mach.irqcleanup();
+ irq_installed = 0;
+ }
+
+ write_sq_release_buffers();
+
+ if (mixer_unit >= 0)
+ unregister_sound_mixer(mixer_unit);
+ if (state_unit >= 0)
+ unregister_sound_special(state_unit);
+ if (sq_unit >= 0)
+ unregister_sound_dsp(sq_unit);
+}
+
+#else /* !MODULE */
+
+static int dmasound_setup(char *str)
+{
+ int ints[6], size;
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ /* check the bootstrap parameter for "dmasound=" */
+
+ /* FIXME: other than in the most naive of cases there is no sense in these
+ * buffers being other than powers of two. This is not checked yet.
+ */
+
+ switch (ints[0]) {
+ case 3:
+ if ((ints[3] < 0) || (ints[3] > MAX_CATCH_RADIUS))
+ printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius);
+ else
+ catchRadius = ints[3];
+ /* fall through */
+ case 2:
+ if (ints[1] < MIN_BUFFERS)
+ printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs);
+ else
+ numWriteBufs = ints[1];
+ /* fall through */
+ case 1:
+ if ((size = ints[2]) < 256) /* check for small buffer specs */
+ size <<= 10 ;
+ if (size < MIN_BUFSIZE || size > MAX_BUFSIZE)
+ printk("dmasound_setup: invalid write buffer size, using default = %d\n", writeBufSize);
+ else
+ writeBufSize = size;
+ case 0:
+ break;
+ default:
+ printk("dmasound_setup: invalid number of arguments\n");
+ return 0;
+ }
+ return 1;
+}
+
+__setup("dmasound=", dmasound_setup);
+
+#endif /* !MODULE */
+
+ /*
+ * Conversion tables
+ */
+
+#ifdef HAS_8BIT_TABLES
+/* 8 bit mu-law */
+
+char dmasound_ulaw2dma8[] = {
+ -126, -122, -118, -114, -110, -106, -102, -98,
+ -94, -90, -86, -82, -78, -74, -70, -66,
+ -63, -61, -59, -57, -55, -53, -51, -49,
+ -47, -45, -43, -41, -39, -37, -35, -33,
+ -31, -30, -29, -28, -27, -26, -25, -24,
+ -23, -22, -21, -20, -19, -18, -17, -16,
+ -16, -15, -15, -14, -14, -13, -13, -12,
+ -12, -11, -11, -10, -10, -9, -9, -8,
+ -8, -8, -7, -7, -7, -7, -6, -6,
+ -6, -6, -5, -5, -5, -5, -4, -4,
+ -4, -4, -4, -4, -3, -3, -3, -3,
+ -3, -3, -3, -3, -2, -2, -2, -2,
+ -2, -2, -2, -2, -2, -2, -2, -2,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, 0,
+ 125, 121, 117, 113, 109, 105, 101, 97,
+ 93, 89, 85, 81, 77, 73, 69, 65,
+ 62, 60, 58, 56, 54, 52, 50, 48,
+ 46, 44, 42, 40, 38, 36, 34, 32,
+ 30, 29, 28, 27, 26, 25, 24, 23,
+ 22, 21, 20, 19, 18, 17, 16, 15,
+ 15, 14, 14, 13, 13, 12, 12, 11,
+ 11, 10, 10, 9, 9, 8, 8, 7,
+ 7, 7, 6, 6, 6, 6, 5, 5,
+ 5, 5, 4, 4, 4, 4, 3, 3,
+ 3, 3, 3, 3, 2, 2, 2, 2,
+ 2, 2, 2, 2, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0
+};
+
+/* 8 bit A-law */
+
+char dmasound_alaw2dma8[] = {
+ -22, -21, -24, -23, -18, -17, -20, -19,
+ -30, -29, -32, -31, -26, -25, -28, -27,
+ -11, -11, -12, -12, -9, -9, -10, -10,
+ -15, -15, -16, -16, -13, -13, -14, -14,
+ -86, -82, -94, -90, -70, -66, -78, -74,
+ -118, -114, -126, -122, -102, -98, -110, -106,
+ -43, -41, -47, -45, -35, -33, -39, -37,
+ -59, -57, -63, -61, -51, -49, -55, -53,
+ -2, -2, -2, -2, -2, -2, -2, -2,
+ -2, -2, -2, -2, -2, -2, -2, -2,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -6, -6, -6, -6, -5, -5, -5, -5,
+ -8, -8, -8, -8, -7, -7, -7, -7,
+ -3, -3, -3, -3, -3, -3, -3, -3,
+ -4, -4, -4, -4, -4, -4, -4, -4,
+ 21, 20, 23, 22, 17, 16, 19, 18,
+ 29, 28, 31, 30, 25, 24, 27, 26,
+ 10, 10, 11, 11, 8, 8, 9, 9,
+ 14, 14, 15, 15, 12, 12, 13, 13,
+ 86, 82, 94, 90, 70, 66, 78, 74,
+ 118, 114, 126, 122, 102, 98, 110, 106,
+ 43, 41, 47, 45, 35, 33, 39, 37,
+ 59, 57, 63, 61, 51, 49, 55, 53,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ 5, 5, 5, 5, 4, 4, 4, 4,
+ 7, 7, 7, 7, 6, 6, 6, 6,
+ 2, 2, 2, 2, 2, 2, 2, 2,
+ 3, 3, 3, 3, 3, 3, 3, 3
+};
+#endif /* HAS_8BIT_TABLES */
+
+ /*
+ * Visible symbols for modules
+ */
+
+EXPORT_SYMBOL(dmasound);
+EXPORT_SYMBOL(dmasound_init);
+#ifdef MODULE
+EXPORT_SYMBOL(dmasound_deinit);
+#endif
+EXPORT_SYMBOL(dmasound_write_sq);
+EXPORT_SYMBOL(dmasound_catchRadius);
+#ifdef HAS_8BIT_TABLES
+EXPORT_SYMBOL(dmasound_ulaw2dma8);
+EXPORT_SYMBOL(dmasound_alaw2dma8);
+#endif
diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c
new file mode 100644
index 0000000..06e9e88
--- /dev/null
+++ b/sound/oss/dmasound/dmasound_paula.c
@@ -0,0 +1,740 @@
+/*
+ * linux/sound/oss/dmasound/dmasound_paula.c
+ *
+ * Amiga `Paula' DMA Sound Driver
+ *
+ * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
+ * prior to 28/01/2001
+ *
+ * 28/01/2001 [0.1] Iain Sandoe
+ * - added versioning
+ * - put in and populated the hardware_afmts field.
+ * [0.2] - put in SNDCTL_DSP_GETCAPS value.
+ * [0.3] - put in constraint on state buffer usage.
+ * [0.4] - put in default hard/soft settings
+*/
+
+
+#include <linux/module.h>
+#include <linux/mm.h>
+#include <linux/init.h>
+#include <linux/ioport.h>
+#include <linux/soundcard.h>
+#include <linux/interrupt.h>
+
+#include <asm/uaccess.h>
+#include <asm/setup.h>
+#include <asm/amigahw.h>
+#include <asm/amigaints.h>
+#include <asm/machdep.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_PAULA_REVISION 0
+#define DMASOUND_PAULA_EDITION 4
+
+#define custom amiga_custom
+ /*
+ * The minimum period for audio depends on htotal (for OCS/ECS/AGA)
+ * (Imported from arch/m68k/amiga/amisound.c)
+ */
+
+extern volatile u_short amiga_audio_min_period;
+
+
+ /*
+ * amiga_mksound() should be able to restore the period after beeping
+ * (Imported from arch/m68k/amiga/amisound.c)
+ */
+
+extern u_short amiga_audio_period;
+
+
+ /*
+ * Audio DMA masks
+ */
+
+#define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
+#define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
+#define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)
+
+
+ /*
+ * Helper pointers for 16(14)-bit sound
+ */
+
+static int write_sq_block_size_half, write_sq_block_size_quarter;
+
+
+/*** Low level stuff *********************************************************/
+
+
+static void *AmiAlloc(unsigned int size, gfp_t flags);
+static void AmiFree(void *obj, unsigned int size);
+static int AmiIrqInit(void);
+#ifdef MODULE
+static void AmiIrqCleanUp(void);
+#endif
+static void AmiSilence(void);
+static void AmiInit(void);
+static int AmiSetFormat(int format);
+static int AmiSetVolume(int volume);
+static int AmiSetTreble(int treble);
+static void AmiPlayNextFrame(int index);
+static void AmiPlay(void);
+static irqreturn_t AmiInterrupt(int irq, void *dummy);
+
+#ifdef CONFIG_HEARTBEAT
+
+ /*
+ * Heartbeat interferes with sound since the 7 kHz low-pass filter and the
+ * power LED are controlled by the same line.
+ */
+
+static void (*saved_heartbeat)(int) = NULL;
+
+static inline void disable_heartbeat(void)
+{
+ if (mach_heartbeat) {
+ saved_heartbeat = mach_heartbeat;
+ mach_heartbeat = NULL;
+ }
+ AmiSetTreble(dmasound.treble);
+}
+
+static inline void enable_heartbeat(void)
+{
+ if (saved_heartbeat)
+ mach_heartbeat = saved_heartbeat;
+}
+#else /* !CONFIG_HEARTBEAT */
+#define disable_heartbeat() do { } while (0)
+#define enable_heartbeat() do { } while (0)
+#endif /* !CONFIG_HEARTBEAT */
+
+
+/*** Mid level stuff *********************************************************/
+
+static void AmiMixerInit(void);
+static int AmiMixerIoctl(u_int cmd, u_long arg);
+static int AmiWriteSqSetup(void);
+static int AmiStateInfo(char *buffer, size_t space);
+
+
+/*** Translations ************************************************************/
+
+/* ++TeSche: radically changed for new expanding purposes...
+ *
+ * These two routines now deal with copying/expanding/translating the samples
+ * from user space into our buffer at the right frequency. They take care about
+ * how much data there's actually to read, how much buffer space there is and
+ * to convert samples into the right frequency/encoding. They will only work on
+ * complete samples so it may happen they leave some bytes in the input stream
+ * if the user didn't write a multiple of the current sample size. They both
+ * return the number of bytes they've used from both streams so you may detect
+ * such a situation. Luckily all programs should be able to cope with that.
+ *
+ * I think I've optimized anything as far as one can do in plain C, all
+ * variables should fit in registers and the loops are really short. There's
+ * one loop for every possible situation. Writing a more generalized and thus
+ * parameterized loop would only produce slower code. Feel free to optimize
+ * this in assembler if you like. :)
+ *
+ * I think these routines belong here because they're not yet really hardware
+ * independent, especially the fact that the Falcon can play 16bit samples
+ * only in stereo is hardcoded in both of them!
+ *
+ * ++geert: split in even more functions (one per format)
+ */
+
+
+ /*
+ * Native format
+ */
+
+static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
+{
+ ssize_t count, used;
+
+ if (!dmasound.soft.stereo) {
+ void *p = &frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft) & ~1;
+ used = count;
+ if (copy_from_user(p, userPtr, count))
+ return -EFAULT;
+ } else {
+ u_char *left = &frame[*frameUsed>>1];
+ u_char *right = left+write_sq_block_size_half;
+ count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
+ used = count*2;
+ while (count > 0) {
+ if (get_user(*left++, userPtr++)
+ || get_user(*right++, userPtr++))
+ return -EFAULT;
+ count--;
+ }
+ }
+ *frameUsed += used;
+ return used;
+}
+
+
+ /*
+ * Copy and convert 8 bit data
+ */
+
+#define GENERATE_AMI_CT8(funcname, convsample) \
+static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
+ u_char frame[], ssize_t *frameUsed, \
+ ssize_t frameLeft) \
+{ \
+ ssize_t count, used; \
+ \
+ if (!dmasound.soft.stereo) { \
+ u_char *p = &frame[*frameUsed]; \
+ count = min_t(size_t, userCount, frameLeft) & ~1; \
+ used = count; \
+ while (count > 0) { \
+ u_char data; \
+ if (get_user(data, userPtr++)) \
+ return -EFAULT; \
+ *p++ = convsample(data); \
+ count--; \
+ } \
+ } else { \
+ u_char *left = &frame[*frameUsed>>1]; \
+ u_char *right = left+write_sq_block_size_half; \
+ count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
+ used = count*2; \
+ while (count > 0) { \
+ u_char data; \
+ if (get_user(data, userPtr++)) \
+ return -EFAULT; \
+ *left++ = convsample(data); \
+ if (get_user(data, userPtr++)) \
+ return -EFAULT; \
+ *right++ = convsample(data); \
+ count--; \
+ } \
+ } \
+ *frameUsed += used; \
+ return used; \
+}
+
+#define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)])
+#define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)])
+#define AMI_CT_U8(x) ((x) ^ 0x80)
+
+GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
+GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
+GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)
+
+
+ /*
+ * Copy and convert 16 bit data
+ */
+
+#define GENERATE_AMI_CT_16(funcname, convsample) \
+static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
+ u_char frame[], ssize_t *frameUsed, \
+ ssize_t frameLeft) \
+{ \
+ const u_short __user *ptr = (const u_short __user *)userPtr; \
+ ssize_t count, used; \
+ u_short data; \
+ \
+ if (!dmasound.soft.stereo) { \
+ u_char *high = &frame[*frameUsed>>1]; \
+ u_char *low = high+write_sq_block_size_half; \
+ count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
+ used = count*2; \
+ while (count > 0) { \
+ if (get_user(data, ptr++)) \
+ return -EFAULT; \
+ data = convsample(data); \
+ *high++ = data>>8; \
+ *low++ = (data>>2) & 0x3f; \
+ count--; \
+ } \
+ } else { \
+ u_char *lefth = &frame[*frameUsed>>2]; \
+ u_char *leftl = lefth+write_sq_block_size_quarter; \
+ u_char *righth = lefth+write_sq_block_size_half; \
+ u_char *rightl = righth+write_sq_block_size_quarter; \
+ count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \
+ used = count*4; \
+ while (count > 0) { \
+ if (get_user(data, ptr++)) \
+ return -EFAULT; \
+ data = convsample(data); \
+ *lefth++ = data>>8; \
+ *leftl++ = (data>>2) & 0x3f; \
+ if (get_user(data, ptr++)) \
+ return -EFAULT; \
+ data = convsample(data); \
+ *righth++ = data>>8; \
+ *rightl++ = (data>>2) & 0x3f; \
+ count--; \
+ } \
+ } \
+ *frameUsed += used; \
+ return used; \
+}
+
+#define AMI_CT_S16BE(x) (x)
+#define AMI_CT_U16BE(x) ((x) ^ 0x8000)
+#define AMI_CT_S16LE(x) (le2be16((x)))
+#define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000)
+
+GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
+GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
+GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
+GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)
+
+
+static TRANS transAmiga = {
+ .ct_ulaw = ami_ct_ulaw,
+ .ct_alaw = ami_ct_alaw,
+ .ct_s8 = ami_ct_s8,
+ .ct_u8 = ami_ct_u8,
+ .ct_s16be = ami_ct_s16be,
+ .ct_u16be = ami_ct_u16be,
+ .ct_s16le = ami_ct_s16le,
+ .ct_u16le = ami_ct_u16le,
+};
+
+/*** Low level stuff *********************************************************/
+
+static inline void StopDMA(void)
+{
+ custom.aud[0].audvol = custom.aud[1].audvol = 0;
+ custom.aud[2].audvol = custom.aud[3].audvol = 0;
+ custom.dmacon = AMI_AUDIO_OFF;
+ enable_heartbeat();
+}
+
+static void *AmiAlloc(unsigned int size, gfp_t flags)
+{
+ return amiga_chip_alloc((long)size, "dmasound [Paula]");
+}
+
+static void AmiFree(void *obj, unsigned int size)
+{
+ amiga_chip_free (obj);
+}
+
+static int __init AmiIrqInit(void)
+{
+ /* turn off DMA for audio channels */
+ StopDMA();
+
+ /* Register interrupt handler. */
+ if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
+ AmiInterrupt))
+ return 0;
+ return 1;
+}
+
+#ifdef MODULE
+static void AmiIrqCleanUp(void)
+{
+ /* turn off DMA for audio channels */
+ StopDMA();
+ /* release the interrupt */
+ free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
+}
+#endif /* MODULE */
+
+static void AmiSilence(void)
+{
+ /* turn off DMA for audio channels */
+ StopDMA();
+}
+
+
+static void AmiInit(void)
+{
+ int period, i;
+
+ AmiSilence();
+
+ if (dmasound.soft.speed)
+ period = amiga_colorclock/dmasound.soft.speed-1;
+ else
+ period = amiga_audio_min_period;
+ dmasound.hard = dmasound.soft;
+ dmasound.trans_write = &transAmiga;
+
+ if (period < amiga_audio_min_period) {
+ /* we would need to squeeze the sound, but we won't do that */
+ period = amiga_audio_min_period;
+ } else if (period > 65535) {
+ period = 65535;
+ }
+ dmasound.hard.speed = amiga_colorclock/(period+1);
+
+ for (i = 0; i < 4; i++)
+ custom.aud[i].audper = period;
+ amiga_audio_period = period;
+}
+
+
+static int AmiSetFormat(int format)
+{
+ int size;
+
+ /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
+
+ switch (format) {
+ case AFMT_QUERY:
+ return dmasound.soft.format;
+ case AFMT_MU_LAW:
+ case AFMT_A_LAW:
+ case AFMT_U8:
+ case AFMT_S8:
+ size = 8;
+ break;
+ case AFMT_S16_BE:
+ case AFMT_U16_BE:
+ case AFMT_S16_LE:
+ case AFMT_U16_LE:
+ size = 16;
+ break;
+ default: /* :-) */
+ size = 8;
+ format = AFMT_S8;
+ }
+
+ dmasound.soft.format = format;
+ dmasound.soft.size = size;
+ if (dmasound.minDev == SND_DEV_DSP) {
+ dmasound.dsp.format = format;
+ dmasound.dsp.size = dmasound.soft.size;
+ }
+ AmiInit();
+
+ return format;
+}
+
+
+#define VOLUME_VOXWARE_TO_AMI(v) \
+ (((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
+#define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)
+
+static int AmiSetVolume(int volume)
+{
+ dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
+ custom.aud[0].audvol = dmasound.volume_left;
+ dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
+ custom.aud[1].audvol = dmasound.volume_right;
+ if (dmasound.hard.size == 16) {
+ if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
+ custom.aud[2].audvol = 1;
+ custom.aud[3].audvol = 1;
+ } else {
+ custom.aud[2].audvol = 0;
+ custom.aud[3].audvol = 0;
+ }
+ }
+ return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
+ (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
+}
+
+static int AmiSetTreble(int treble)
+{
+ dmasound.treble = treble;
+ if (treble < 50)
+ ciaa.pra &= ~0x02;
+ else
+ ciaa.pra |= 0x02;
+ return treble;
+}
+
+
+#define AMI_PLAY_LOADED 1
+#define AMI_PLAY_PLAYING 2
+#define AMI_PLAY_MASK 3
+
+
+static void AmiPlayNextFrame(int index)
+{
+ u_char *start, *ch0, *ch1, *ch2, *ch3;
+ u_long size;
+
+ /* used by AmiPlay() if all doubts whether there really is something
+ * to be played are already wiped out.
+ */
+ start = write_sq.buffers[write_sq.front];
+ size = (write_sq.count == index ? write_sq.rear_size
+ : write_sq.block_size)>>1;
+
+ if (dmasound.hard.stereo) {
+ ch0 = start;
+ ch1 = start+write_sq_block_size_half;
+ size >>= 1;
+ } else {
+ ch0 = start;
+ ch1 = start;
+ }
+
+ disable_heartbeat();
+ custom.aud[0].audvol = dmasound.volume_left;
+ custom.aud[1].audvol = dmasound.volume_right;
+ if (dmasound.hard.size == 8) {
+ custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
+ custom.aud[0].audlen = size;
+ custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
+ custom.aud[1].audlen = size;
+ custom.dmacon = AMI_AUDIO_8;
+ } else {
+ size >>= 1;
+ custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
+ custom.aud[0].audlen = size;
+ custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
+ custom.aud[1].audlen = size;
+ if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
+ /* We can play pseudo 14-bit only with the maximum volume */
+ ch3 = ch0+write_sq_block_size_quarter;
+ ch2 = ch1+write_sq_block_size_quarter;
+ custom.aud[2].audvol = 1; /* we are being affected by the beeps */
+ custom.aud[3].audvol = 1; /* restoring volume here helps a bit */
+ custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2);
+ custom.aud[2].audlen = size;
+ custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3);
+ custom.aud[3].audlen = size;
+ custom.dmacon = AMI_AUDIO_14;
+ } else {
+ custom.aud[2].audvol = 0;
+ custom.aud[3].audvol = 0;
+ custom.dmacon = AMI_AUDIO_8;
+ }
+ }
+ write_sq.front = (write_sq.front+1) % write_sq.max_count;
+ write_sq.active |= AMI_PLAY_LOADED;
+}
+
+
+static void AmiPlay(void)
+{
+ int minframes = 1;
+
+ custom.intena = IF_AUD0;
+
+ if (write_sq.active & AMI_PLAY_LOADED) {
+ /* There's already a frame loaded */
+ custom.intena = IF_SETCLR | IF_AUD0;
+ return;
+ }
+
+ if (write_sq.active & AMI_PLAY_PLAYING)
+ /* Increase threshold: frame 1 is already being played */
+ minframes = 2;
+
+ if (write_sq.count < minframes) {
+ /* Nothing to do */
+ custom.intena = IF_SETCLR | IF_AUD0;
+ return;
+ }
+
+ if (write_sq.count <= minframes &&
+ write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
+ /* hmmm, the only existing frame is not
+ * yet filled and we're not syncing?
+ */
+ custom.intena = IF_SETCLR | IF_AUD0;
+ return;
+ }
+
+ AmiPlayNextFrame(minframes);
+
+ custom.intena = IF_SETCLR | IF_AUD0;
+}
+
+
+static irqreturn_t AmiInterrupt(int irq, void *dummy)
+{
+ int minframes = 1;
+
+ custom.intena = IF_AUD0;
+
+ if (!write_sq.active) {
+ /* Playing was interrupted and sq_reset() has already cleared
+ * the sq variables, so better don't do anything here.
+ */
+ WAKE_UP(write_sq.sync_queue);
+ return IRQ_HANDLED;
+ }
+
+ if (write_sq.active & AMI_PLAY_PLAYING) {
+ /* We've just finished a frame */
+ write_sq.count--;
+ WAKE_UP(write_sq.action_queue);
+ }
+
+ if (write_sq.active & AMI_PLAY_LOADED)
+ /* Increase threshold: frame 1 is already being played */
+ minframes = 2;
+
+ /* Shift the flags */
+ write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;
+
+ if (!write_sq.active)
+ /* No frame is playing, disable audio DMA */
+ StopDMA();
+
+ custom.intena = IF_SETCLR | IF_AUD0;
+
+ if (write_sq.count >= minframes)
+ /* Try to play the next frame */
+ AmiPlay();
+
+ if (!write_sq.active)
+ /* Nothing to play anymore.
+ Wake up a process waiting for audio output to drain. */
+ WAKE_UP(write_sq.sync_queue);
+ return IRQ_HANDLED;
+}
+
+/*** Mid level stuff *********************************************************/
+
+
+/*
+ * /dev/mixer abstraction
+ */
+
+static void __init AmiMixerInit(void)
+{
+ dmasound.volume_left = 64;
+ dmasound.volume_right = 64;
+ custom.aud[0].audvol = dmasound.volume_left;
+ custom.aud[3].audvol = 1; /* For pseudo 14bit */
+ custom.aud[1].audvol = dmasound.volume_right;
+ custom.aud[2].audvol = 1; /* For pseudo 14bit */
+ dmasound.treble = 50;
+}
+
+static int AmiMixerIoctl(u_int cmd, u_long arg)
+{
+ int data;
+ switch (cmd) {
+ case SOUND_MIXER_READ_DEVMASK:
+ return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE);
+ case SOUND_MIXER_READ_RECMASK:
+ return IOCTL_OUT(arg, 0);
+ case SOUND_MIXER_READ_STEREODEVS:
+ return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
+ case SOUND_MIXER_READ_VOLUME:
+ return IOCTL_OUT(arg,
+ VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
+ VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
+ case SOUND_MIXER_WRITE_VOLUME:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_volume(data));
+ case SOUND_MIXER_READ_TREBLE:
+ return IOCTL_OUT(arg, dmasound.treble);
+ case SOUND_MIXER_WRITE_TREBLE:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_treble(data));
+ }
+ return -EINVAL;
+}
+
+
+static int AmiWriteSqSetup(void)
+{
+ write_sq_block_size_half = write_sq.block_size>>1;
+ write_sq_block_size_quarter = write_sq_block_size_half>>1;
+ return 0;
+}
+
+
+static int AmiStateInfo(char *buffer, size_t space)
+{
+ int len = 0;
+ len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n",
+ dmasound.volume_left);
+ len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
+ dmasound.volume_right);
+ if (len >= space) {
+ printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ;
+ len = space ;
+ }
+ return len;
+}
+
+
+/*** Machine definitions *****************************************************/
+
+static SETTINGS def_hard = {
+ .format = AFMT_S8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 8000
+} ;
+
+static SETTINGS def_soft = {
+ .format = AFMT_U8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 8000
+} ;
+
+static MACHINE machAmiga = {
+ .name = "Amiga",
+ .name2 = "AMIGA",
+ .owner = THIS_MODULE,
+ .dma_alloc = AmiAlloc,
+ .dma_free = AmiFree,
+ .irqinit = AmiIrqInit,
+#ifdef MODULE
+ .irqcleanup = AmiIrqCleanUp,
+#endif /* MODULE */
+ .init = AmiInit,
+ .silence = AmiSilence,
+ .setFormat = AmiSetFormat,
+ .setVolume = AmiSetVolume,
+ .setTreble = AmiSetTreble,
+ .play = AmiPlay,
+ .mixer_init = AmiMixerInit,
+ .mixer_ioctl = AmiMixerIoctl,
+ .write_sq_setup = AmiWriteSqSetup,
+ .state_info = AmiStateInfo,
+ .min_dsp_speed = 8000,
+ .version = ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION),
+ .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
+ .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
+};
+
+
+/*** Config & Setup **********************************************************/
+
+
+static int __init dmasound_paula_init(void)
+{
+ int err;
+
+ if (MACH_IS_AMIGA && AMIGAHW_PRESENT(AMI_AUDIO)) {
+ if (!request_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40,
+ "dmasound [Paula]"))
+ return -EBUSY;
+ dmasound.mach = machAmiga;
+ dmasound.mach.default_hard = def_hard ;
+ dmasound.mach.default_soft = def_soft ;
+ err = dmasound_init();
+ if (err)
+ release_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40);
+ return err;
+ } else
+ return -ENODEV;
+}
+
+static void __exit dmasound_paula_cleanup(void)
+{
+ dmasound_deinit();
+ release_mem_region(CUSTOM_PHYSADDR+0xa0, 0x40);
+}
+
+module_init(dmasound_paula_init);
+module_exit(dmasound_paula_cleanup);
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/dmasound/dmasound_q40.c b/sound/oss/dmasound/dmasound_q40.c
new file mode 100644
index 0000000..1855b14
--- /dev/null
+++ b/sound/oss/dmasound/dmasound_q40.c
@@ -0,0 +1,634 @@
+/*
+ * linux/sound/oss/dmasound/dmasound_q40.c
+ *
+ * Q40 DMA Sound Driver
+ *
+ * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
+ * prior to 28/01/2001
+ *
+ * 28/01/2001 [0.1] Iain Sandoe
+ * - added versioning
+ * - put in and populated the hardware_afmts field.
+ * [0.2] - put in SNDCTL_DSP_GETCAPS value.
+ * [0.3] - put in default hard/soft settings.
+ */
+
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/soundcard.h>
+#include <linux/interrupt.h>
+
+#include <asm/uaccess.h>
+#include <asm/q40ints.h>
+#include <asm/q40_master.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_Q40_REVISION 0
+#define DMASOUND_Q40_EDITION 3
+
+static int expand_bal; /* Balance factor for expanding (not volume!) */
+static int expand_data; /* Data for expanding */
+
+
+/*** Low level stuff *********************************************************/
+
+
+static void *Q40Alloc(unsigned int size, gfp_t flags);
+static void Q40Free(void *, unsigned int);
+static int Q40IrqInit(void);
+#ifdef MODULE
+static void Q40IrqCleanUp(void);
+#endif
+static void Q40Silence(void);
+static void Q40Init(void);
+static int Q40SetFormat(int format);
+static int Q40SetVolume(int volume);
+static void Q40PlayNextFrame(int index);
+static void Q40Play(void);
+static irqreturn_t Q40StereoInterrupt(int irq, void *dummy);
+static irqreturn_t Q40MonoInterrupt(int irq, void *dummy);
+static void Q40Interrupt(void);
+
+
+/*** Mid level stuff *********************************************************/
+
+
+
+/* userCount, frameUsed, frameLeft == byte counts */
+static ssize_t q40_ct_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ char *table = dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8;
+ ssize_t count, used;
+ u_char *p = (u_char *) &frame[*frameUsed];
+
+ used = count = min_t(size_t, userCount, frameLeft);
+ if (copy_from_user(p,userPtr,count))
+ return -EFAULT;
+ while (count > 0) {
+ *p = table[*p]+128;
+ p++;
+ count--;
+ }
+ *frameUsed += used ;
+ return used;
+}
+
+
+static ssize_t q40_ct_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+ u_char *p = (u_char *) &frame[*frameUsed];
+
+ used = count = min_t(size_t, userCount, frameLeft);
+ if (copy_from_user(p,userPtr,count))
+ return -EFAULT;
+ while (count > 0) {
+ *p = *p + 128;
+ p++;
+ count--;
+ }
+ *frameUsed += used;
+ return used;
+}
+
+static ssize_t q40_ct_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ ssize_t count, used;
+ u_char *p = (u_char *) &frame[*frameUsed];
+
+ used = count = min_t(size_t, userCount, frameLeft);
+ if (copy_from_user(p,userPtr,count))
+ return -EFAULT;
+ *frameUsed += used;
+ return used;
+}
+
+
+/* a bit too complicated to optimise right now ..*/
+static ssize_t q40_ctx_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ unsigned char *table = (unsigned char *)
+ (dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8);
+ unsigned int data = expand_data;
+ u_char *p = (u_char *) &frame[*frameUsed];
+ int bal = expand_bal;
+ int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ int utotal, ftotal;
+
+ ftotal = frameLeft;
+ utotal = userCount;
+ while (frameLeft) {
+ u_char c;
+ if (bal < 0) {
+ if (userCount == 0)
+ break;
+ if (get_user(c, userPtr++))
+ return -EFAULT;
+ data = table[c];
+ data += 0x80;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ expand_bal = bal;
+ expand_data = data;
+ *frameUsed += (ftotal - frameLeft);
+ utotal -= userCount;
+ return utotal;
+}
+
+
+static ssize_t q40_ctx_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ u_char *p = (u_char *) &frame[*frameUsed];
+ unsigned int data = expand_data;
+ int bal = expand_bal;
+ int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ int utotal, ftotal;
+
+
+ ftotal = frameLeft;
+ utotal = userCount;
+ while (frameLeft) {
+ u_char c;
+ if (bal < 0) {
+ if (userCount == 0)
+ break;
+ if (get_user(c, userPtr++))
+ return -EFAULT;
+ data = c ;
+ data += 0x80;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ expand_bal = bal;
+ expand_data = data;
+ *frameUsed += (ftotal - frameLeft);
+ utotal -= userCount;
+ return utotal;
+}
+
+
+static ssize_t q40_ctx_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ u_char *p = (u_char *) &frame[*frameUsed];
+ unsigned int data = expand_data;
+ int bal = expand_bal;
+ int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ int utotal, ftotal;
+
+ ftotal = frameLeft;
+ utotal = userCount;
+ while (frameLeft) {
+ u_char c;
+ if (bal < 0) {
+ if (userCount == 0)
+ break;
+ if (get_user(c, userPtr++))
+ return -EFAULT;
+ data = c ;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ expand_bal = bal;
+ expand_data = data;
+ *frameUsed += (ftotal - frameLeft) ;
+ utotal -= userCount;
+ return utotal;
+}
+
+/* compressing versions */
+static ssize_t q40_ctc_law(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ unsigned char *table = (unsigned char *)
+ (dmasound.soft.format == AFMT_MU_LAW ? dmasound_ulaw2dma8: dmasound_alaw2dma8);
+ unsigned int data = expand_data;
+ u_char *p = (u_char *) &frame[*frameUsed];
+ int bal = expand_bal;
+ int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ int utotal, ftotal;
+
+ ftotal = frameLeft;
+ utotal = userCount;
+ while (frameLeft) {
+ u_char c;
+ while(bal<0) {
+ if (userCount == 0)
+ goto lout;
+ if (!(bal<(-hSpeed))) {
+ if (get_user(c, userPtr))
+ return -EFAULT;
+ data = 0x80 + table[c];
+ }
+ userPtr++;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ lout:
+ expand_bal = bal;
+ expand_data = data;
+ *frameUsed += (ftotal - frameLeft);
+ utotal -= userCount;
+ return utotal;
+}
+
+
+static ssize_t q40_ctc_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ u_char *p = (u_char *) &frame[*frameUsed];
+ unsigned int data = expand_data;
+ int bal = expand_bal;
+ int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ int utotal, ftotal;
+
+ ftotal = frameLeft;
+ utotal = userCount;
+ while (frameLeft) {
+ u_char c;
+ while (bal < 0) {
+ if (userCount == 0)
+ goto lout;
+ if (!(bal<(-hSpeed))) {
+ if (get_user(c, userPtr))
+ return -EFAULT;
+ data = c + 0x80;
+ }
+ userPtr++;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ lout:
+ expand_bal = bal;
+ expand_data = data;
+ *frameUsed += (ftotal - frameLeft);
+ utotal -= userCount;
+ return utotal;
+}
+
+
+static ssize_t q40_ctc_u8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed,
+ ssize_t frameLeft)
+{
+ u_char *p = (u_char *) &frame[*frameUsed];
+ unsigned int data = expand_data;
+ int bal = expand_bal;
+ int hSpeed = dmasound.hard.speed, sSpeed = dmasound.soft.speed;
+ int utotal, ftotal;
+
+ ftotal = frameLeft;
+ utotal = userCount;
+ while (frameLeft) {
+ u_char c;
+ while (bal < 0) {
+ if (userCount == 0)
+ goto lout;
+ if (!(bal<(-hSpeed))) {
+ if (get_user(c, userPtr))
+ return -EFAULT;
+ data = c ;
+ }
+ userPtr++;
+ userCount--;
+ bal += hSpeed;
+ }
+ *p++ = data;
+ frameLeft--;
+ bal -= sSpeed;
+ }
+ lout:
+ expand_bal = bal;
+ expand_data = data;
+ *frameUsed += (ftotal - frameLeft) ;
+ utotal -= userCount;
+ return utotal;
+}
+
+
+static TRANS transQ40Normal = {
+ q40_ct_law, q40_ct_law, q40_ct_s8, q40_ct_u8, NULL, NULL, NULL, NULL
+};
+
+static TRANS transQ40Expanding = {
+ q40_ctx_law, q40_ctx_law, q40_ctx_s8, q40_ctx_u8, NULL, NULL, NULL, NULL
+};
+
+static TRANS transQ40Compressing = {
+ q40_ctc_law, q40_ctc_law, q40_ctc_s8, q40_ctc_u8, NULL, NULL, NULL, NULL
+};
+
+
+/*** Low level stuff *********************************************************/
+
+static void *Q40Alloc(unsigned int size, gfp_t flags)
+{
+ return kmalloc(size, flags); /* change to vmalloc */
+}
+
+static void Q40Free(void *ptr, unsigned int size)
+{
+ kfree(ptr);
+}
+
+static int __init Q40IrqInit(void)
+{
+ /* Register interrupt handler. */
+ request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0,
+ "DMA sound", Q40Interrupt);
+
+ return(1);
+}
+
+
+#ifdef MODULE
+static void Q40IrqCleanUp(void)
+{
+ master_outb(0,SAMPLE_ENABLE_REG);
+ free_irq(Q40_IRQ_SAMPLE, Q40Interrupt);
+}
+#endif /* MODULE */
+
+
+static void Q40Silence(void)
+{
+ master_outb(0,SAMPLE_ENABLE_REG);
+ *DAC_LEFT=*DAC_RIGHT=127;
+}
+
+static char *q40_pp;
+static unsigned int q40_sc;
+
+static void Q40PlayNextFrame(int index)
+{
+ u_char *start;
+ u_long size;
+ u_char speed;
+
+ /* used by Q40Play() if all doubts whether there really is something
+ * to be played are already wiped out.
+ */
+ start = write_sq.buffers[write_sq.front];
+ size = (write_sq.count == index ? write_sq.rear_size : write_sq.block_size);
+
+ q40_pp=start;
+ q40_sc=size;
+
+ write_sq.front = (write_sq.front+1) % write_sq.max_count;
+ write_sq.active++;
+
+ speed=(dmasound.hard.speed==10000 ? 0 : 1);
+
+ master_outb( 0,SAMPLE_ENABLE_REG);
+ free_irq(Q40_IRQ_SAMPLE, Q40Interrupt);
+ if (dmasound.soft.stereo)
+ request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0,
+ "Q40 sound", Q40Interrupt);
+ else
+ request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0,
+ "Q40 sound", Q40Interrupt);
+
+ master_outb( speed, SAMPLE_RATE_REG);
+ master_outb( 1,SAMPLE_CLEAR_REG);
+ master_outb( 1,SAMPLE_ENABLE_REG);
+}
+
+static void Q40Play(void)
+{
+ unsigned long flags;
+
+ if (write_sq.active || write_sq.count<=0 ) {
+ /* There's already a frame loaded */
+ return;
+ }
+
+ /* nothing in the queue */
+ if (write_sq.count <= 1 && write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
+ /* hmmm, the only existing frame is not
+ * yet filled and we're not syncing?
+ */
+ return;
+ }
+ spin_lock_irqsave(&dmasound.lock, flags);
+ Q40PlayNextFrame(1);
+ spin_unlock_irqrestore(&dmasound.lock, flags);
+}
+
+static irqreturn_t Q40StereoInterrupt(int irq, void *dummy)
+{
+ spin_lock(&dmasound.lock);
+ if (q40_sc>1){
+ *DAC_LEFT=*q40_pp++;
+ *DAC_RIGHT=*q40_pp++;
+ q40_sc -=2;
+ master_outb(1,SAMPLE_CLEAR_REG);
+ }else Q40Interrupt();
+ spin_unlock(&dmasound.lock);
+ return IRQ_HANDLED;
+}
+static irqreturn_t Q40MonoInterrupt(int irq, void *dummy)
+{
+ spin_lock(&dmasound.lock);
+ if (q40_sc>0){
+ *DAC_LEFT=*q40_pp;
+ *DAC_RIGHT=*q40_pp++;
+ q40_sc --;
+ master_outb(1,SAMPLE_CLEAR_REG);
+ }else Q40Interrupt();
+ spin_unlock(&dmasound.lock);
+ return IRQ_HANDLED;
+}
+static void Q40Interrupt(void)
+{
+ if (!write_sq.active) {
+ /* playing was interrupted and sq_reset() has already cleared
+ * the sq variables, so better don't do anything here.
+ */
+ WAKE_UP(write_sq.sync_queue);
+ master_outb(0,SAMPLE_ENABLE_REG); /* better safe */
+ goto exit;
+ } else write_sq.active=0;
+ write_sq.count--;
+ Q40Play();
+
+ if (q40_sc<2)
+ { /* there was nothing to play, disable irq */
+ master_outb(0,SAMPLE_ENABLE_REG);
+ *DAC_LEFT=*DAC_RIGHT=127;
+ }
+ WAKE_UP(write_sq.action_queue);
+
+ exit:
+ master_outb(1,SAMPLE_CLEAR_REG);
+}
+
+
+static void Q40Init(void)
+{
+ int i, idx;
+ const int freq[] = {10000, 20000};
+
+ /* search a frequency that fits into the allowed error range */
+
+ idx = -1;
+ for (i = 0; i < 2; i++)
+ if ((100 * abs(dmasound.soft.speed - freq[i]) / freq[i]) <= catchRadius)
+ idx = i;
+
+ dmasound.hard = dmasound.soft;
+ /*sound.hard.stereo=1;*/ /* no longer true */
+ dmasound.hard.size=8;
+
+ if (idx > -1) {
+ dmasound.soft.speed = freq[idx];
+ dmasound.trans_write = &transQ40Normal;
+ } else
+ dmasound.trans_write = &transQ40Expanding;
+
+ Q40Silence();
+
+ if (dmasound.hard.speed > 20200) {
+ /* squeeze the sound, we do that */
+ dmasound.hard.speed = 20000;
+ dmasound.trans_write = &transQ40Compressing;
+ } else if (dmasound.hard.speed > 10000) {
+ dmasound.hard.speed = 20000;
+ } else {
+ dmasound.hard.speed = 10000;
+ }
+ expand_bal = -dmasound.soft.speed;
+}
+
+
+static int Q40SetFormat(int format)
+{
+ /* Q40 sound supports only 8bit modes */
+
+ switch (format) {
+ case AFMT_QUERY:
+ return(dmasound.soft.format);
+ case AFMT_MU_LAW:
+ case AFMT_A_LAW:
+ case AFMT_S8:
+ case AFMT_U8:
+ break;
+ default:
+ format = AFMT_S8;
+ }
+
+ dmasound.soft.format = format;
+ dmasound.soft.size = 8;
+ if (dmasound.minDev == SND_DEV_DSP) {
+ dmasound.dsp.format = format;
+ dmasound.dsp.size = 8;
+ }
+ Q40Init();
+
+ return(format);
+}
+
+static int Q40SetVolume(int volume)
+{
+ return 0;
+}
+
+
+/*** Machine definitions *****************************************************/
+
+static SETTINGS def_hard = {
+ .format = AFMT_U8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 10000
+} ;
+
+static SETTINGS def_soft = {
+ .format = AFMT_U8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 8000
+} ;
+
+static MACHINE machQ40 = {
+ .name = "Q40",
+ .name2 = "Q40",
+ .owner = THIS_MODULE,
+ .dma_alloc = Q40Alloc,
+ .dma_free = Q40Free,
+ .irqinit = Q40IrqInit,
+#ifdef MODULE
+ .irqcleanup = Q40IrqCleanUp,
+#endif /* MODULE */
+ .init = Q40Init,
+ .silence = Q40Silence,
+ .setFormat = Q40SetFormat,
+ .setVolume = Q40SetVolume,
+ .play = Q40Play,
+ .min_dsp_speed = 10000,
+ .version = ((DMASOUND_Q40_REVISION<<8) | DMASOUND_Q40_EDITION),
+ .hardware_afmts = AFMT_U8, /* h'ware-supported formats *only* here */
+ .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
+};
+
+
+/*** Config & Setup **********************************************************/
+
+
+static int __init dmasound_q40_init(void)
+{
+ if (MACH_IS_Q40) {
+ dmasound.mach = machQ40;
+ dmasound.mach.default_hard = def_hard ;
+ dmasound.mach.default_soft = def_soft ;
+ return dmasound_init();
+ } else
+ return -ENODEV;
+}
+
+static void __exit dmasound_q40_cleanup(void)
+{
+ dmasound_deinit();
+}
+
+module_init(dmasound_q40_init);
+module_exit(dmasound_q40_cleanup);
+
+MODULE_DESCRIPTION("Q40/Q60 sound driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/hex2hex.c b/sound/oss/hex2hex.c
new file mode 100644
index 0000000..5460faa
--- /dev/null
+++ b/sound/oss/hex2hex.c
@@ -0,0 +1,101 @@
+/*
+ * hex2hex reads stdin in Intel HEX format and produces an
+ * (unsigned char) array which contains the bytes and writes it
+ * to stdout using C syntax
+ */
+
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+
+#define ABANDON(why) { fprintf(stderr, "%s\n", why); exit(1); }
+#define MAX_SIZE (256*1024)
+unsigned char buf[MAX_SIZE];
+
+int loadhex(FILE *inf, unsigned char *buf)
+{
+ int l=0, c, i;
+
+ while ((c=getc(inf))!=EOF)
+ {
+ if (c == ':') /* Sync with beginning of line */
+ {
+ int n, check;
+ unsigned char sum;
+ int addr;
+ int linetype;
+
+ if (fscanf(inf, "%02x", &n) != 1)
+ ABANDON("File format error");
+ sum = n;
+
+ if (fscanf(inf, "%04x", &addr) != 1)
+ ABANDON("File format error");
+ sum += addr/256;
+ sum += addr%256;
+
+ if (fscanf(inf, "%02x", &linetype) != 1)
+ ABANDON("File format error");
+ sum += linetype;
+
+ if (linetype != 0)
+ continue;
+
+ for (i=0;i<n;i++)
+ {
+ if (fscanf(inf, "%02x", &c) != 1)
+ ABANDON("File format error");
+ if (addr >= MAX_SIZE)
+ ABANDON("File too large");
+ buf[addr++] = c;
+ if (addr > l)
+ l = addr;
+ sum += c;
+ }
+
+ if (fscanf(inf, "%02x", &check) != 1)
+ ABANDON("File format error");
+
+ sum = ~sum + 1;
+ if (check != sum)
+ ABANDON("Line checksum error");
+ }
+ }
+
+ return l;
+}
+
+int main( int argc, const char * argv [] )
+{
+ const char * varline;
+ int i,l;
+ int id=0;
+
+ if(argv[1] && strcmp(argv[1], "-i")==0)
+ {
+ argv++;
+ argc--;
+ id=1;
+ }
+ if(argv[1]==NULL)
+ {
+ fprintf(stderr,"hex2hex: [-i] filename\n");
+ exit(1);
+ }
+ varline = argv[1];
+ l = loadhex(stdin, buf);
+
+ printf("/*\n *\t Computer generated file. Do not edit.\n */\n");
+ printf("static int %s_len = %d;\n", varline, l);
+ printf("static unsigned char %s[] %s = {\n", varline, id?"__initdata":"");
+
+ for (i=0;i<l;i++)
+ {
+ if (i) printf(",");
+ if (i && !(i % 16)) printf("\n");
+ printf("0x%02x", buf[i]);
+ }
+
+ printf("\n};\n\n");
+ return 0;
+}
diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c
new file mode 100644
index 0000000..c180598
--- /dev/null
+++ b/sound/oss/kahlua.c
@@ -0,0 +1,230 @@
+/*
+ * Initialisation code for Cyrix/NatSemi VSA1 softaudio
+ *
+ * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
+ *
+ * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
+ * The older version (VSA1) provides fairly good soundblaster emulation
+ * although there are a couple of bugs: large DMA buffers break record,
+ * and the MPU event handling seems suspect. VSA2 allows the native driver
+ * to control the AC97 audio engine directly and requires a different driver.
+ *
+ * Thanks to National Semiconductor for providing the needed information
+ * on the XpressAudio(tm) internals.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2, or (at your option) any
+ * later version.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * TO DO:
+ * Investigate whether we can portably support Cognac (5520) in the
+ * same manner.
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/pci.h>
+
+#include "sound_config.h"
+
+#include "sb.h"
+
+/*
+ * Read a soundblaster compatible mixer register.
+ * In this case we are actually reading an SMI trap
+ * not real hardware.
+ */
+
+static u8 __devinit mixer_read(unsigned long io, u8 reg)
+{
+ outb(reg, io + 4);
+ udelay(20);
+ reg = inb(io + 5);
+ udelay(20);
+ return reg;
+}
+
+static int __devinit probe_one(struct pci_dev *pdev, const struct pci_device_id *ent)
+{
+ struct address_info *hw_config;
+ unsigned long base;
+ void __iomem *mem;
+ unsigned long io;
+ u16 map;
+ u8 irq, dma8, dma16;
+ int oldquiet;
+ extern int sb_be_quiet;
+
+ base = pci_resource_start(pdev, 0);
+ if(base == 0UL)
+ return 1;
+
+ mem = ioremap(base, 128);
+ if (!mem)
+ return 1;
+ map = readw(mem + 0x18); /* Read the SMI enables */
+ iounmap(mem);
+
+ /* Map bits
+ 0:1 * 0x20 + 0x200 = sb base
+ 2 sb enable
+ 3 adlib enable
+ 5 MPU enable 0x330
+ 6 MPU enable 0x300
+
+ The other bits may be used internally so must be masked */
+
+ io = 0x220 + 0x20 * (map & 3);
+
+ if(map & (1<<2))
+ printk(KERN_INFO "kahlua: XpressAudio at 0x%lx\n", io);
+ else
+ return 1;
+
+ if(map & (1<<5))
+ printk(KERN_INFO "kahlua: MPU at 0x300\n");
+ else if(map & (1<<6))
+ printk(KERN_INFO "kahlua: MPU at 0x330\n");
+
+ irq = mixer_read(io, 0x80) & 0x0F;
+ dma8 = mixer_read(io, 0x81);
+
+ // printk("IRQ=%x MAP=%x DMA=%x\n", irq, map, dma8);
+
+ if(dma8 & 0x20)
+ dma16 = 5;
+ else if(dma8 & 0x40)
+ dma16 = 6;
+ else if(dma8 & 0x80)
+ dma16 = 7;
+ else
+ {
+ printk(KERN_ERR "kahlua: No 16bit DMA enabled.\n");
+ return 1;
+ }
+
+ if(dma8 & 0x01)
+ dma8 = 0;
+ else if(dma8 & 0x02)
+ dma8 = 1;
+ else if(dma8 & 0x08)
+ dma8 = 3;
+ else
+ {
+ printk(KERN_ERR "kahlua: No 8bit DMA enabled.\n");
+ return 1;
+ }
+
+ if(irq & 1)
+ irq = 9;
+ else if(irq & 2)
+ irq = 5;
+ else if(irq & 4)
+ irq = 7;
+ else if(irq & 8)
+ irq = 10;
+ else
+ {
+ printk(KERN_ERR "kahlua: SB IRQ not set.\n");
+ return 1;
+ }
+
+ printk(KERN_INFO "kahlua: XpressAudio on IRQ %d, DMA %d, %d\n",
+ irq, dma8, dma16);
+
+ hw_config = kzalloc(sizeof(struct address_info), GFP_KERNEL);
+ if(hw_config == NULL)
+ {
+ printk(KERN_ERR "kahlua: out of memory.\n");
+ return 1;
+ }
+
+ pci_set_drvdata(pdev, hw_config);
+
+ hw_config->io_base = io;
+ hw_config->irq = irq;
+ hw_config->dma = dma8;
+ hw_config->dma2 = dma16;
+ hw_config->name = "Cyrix XpressAudio";
+ hw_config->driver_use_1 = SB_NO_MIDI | SB_PCI_IRQ;
+
+ if (!request_region(io, 16, "soundblaster"))
+ goto err_out_free;
+
+ if(sb_dsp_detect(hw_config, 0, 0, NULL)==0)
+ {
+ printk(KERN_ERR "kahlua: audio not responding.\n");
+ release_region(io, 16);
+ goto err_out_free;
+ }
+
+ oldquiet = sb_be_quiet;
+ sb_be_quiet = 1;
+ if(sb_dsp_init(hw_config, THIS_MODULE))
+ {
+ sb_be_quiet = oldquiet;
+ goto err_out_free;
+ }
+ sb_be_quiet = oldquiet;
+
+ return 0;
+
+err_out_free:
+ pci_set_drvdata(pdev, NULL);
+ kfree(hw_config);
+ return 1;
+}
+
+static void __devexit remove_one(struct pci_dev *pdev)
+{
+ struct address_info *hw_config = pci_get_drvdata(pdev);
+ sb_dsp_unload(hw_config, 0);
+ pci_set_drvdata(pdev, NULL);
+ kfree(hw_config);
+}
+
+MODULE_AUTHOR("Alan Cox");
+MODULE_DESCRIPTION("Kahlua VSA1 PCI Audio");
+MODULE_LICENSE("GPL");
+
+/*
+ * 5530 only. The 5510/5520 decode is different.
+ */
+
+static struct pci_device_id id_tbl[] = {
+ { PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 },
+ { }
+};
+
+MODULE_DEVICE_TABLE(pci, id_tbl);
+
+static struct pci_driver kahlua_driver = {
+ .name = "kahlua",
+ .id_table = id_tbl,
+ .probe = probe_one,
+ .remove = __devexit_p(remove_one),
+};
+
+
+static int __init kahlua_init_module(void)
+{
+ printk(KERN_INFO "Cyrix Kahlua VSA1 XpressAudio support (c) Copyright 2003 Red Hat Inc\n");
+ return pci_register_driver(&kahlua_driver);
+}
+
+static void __devexit kahlua_cleanup_module(void)
+{
+ pci_unregister_driver(&kahlua_driver);
+}
+
+
+module_init(kahlua_init_module);
+module_exit(kahlua_cleanup_module);
+
diff --git a/sound/oss/midi_ctrl.h b/sound/oss/midi_ctrl.h
new file mode 100644
index 0000000..3353e5a
--- /dev/null
+++ b/sound/oss/midi_ctrl.h
@@ -0,0 +1,22 @@
+static unsigned char ctrl_def_values[128] =
+{
+ 0x40,0x00,0x40,0x40, 0x40,0x40,0x40,0x7f, /* 0 to 7 */
+ 0x40,0x40,0x40,0x7f, 0x40,0x40,0x40,0x40, /* 8 to 15 */
+ 0x40,0x40,0x40,0x40, 0x40,0x40,0x40,0x40, /* 16 to 23 */
+ 0x40,0x40,0x40,0x40, 0x40,0x40,0x40,0x40, /* 24 to 31 */
+
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 32 to 39 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 40 to 47 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 48 to 55 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 56 to 63 */
+
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 64 to 71 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 72 to 79 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 80 to 87 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 88 to 95 */
+
+ 0x00,0x00,0x7f,0x7f, 0x7f,0x7f,0x00,0x00, /* 96 to 103 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 104 to 111 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 112 to 119 */
+ 0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 120 to 127 */
+};
diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c
new file mode 100644
index 0000000..9e45098
--- /dev/null
+++ b/sound/oss/midi_synth.c
@@ -0,0 +1,714 @@
+/*
+ * sound/oss/midi_synth.c
+ *
+ * High level midi sequencer manager for dumb MIDI interfaces.
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+/*
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Andrew Veliath : fixed running status in MIDI input state machine
+ */
+#define USE_SEQ_MACROS
+#define USE_SIMPLE_MACROS
+
+#include "sound_config.h"
+
+#define _MIDI_SYNTH_C_
+
+#include "midi_synth.h"
+
+static int midi2synth[MAX_MIDI_DEV];
+static int sysex_state[MAX_MIDI_DEV] =
+{0};
+static unsigned char prev_out_status[MAX_MIDI_DEV];
+
+#define STORE(cmd) \
+{ \
+ int len; \
+ unsigned char obuf[8]; \
+ cmd; \
+ seq_input_event(obuf, len); \
+}
+
+#define _seqbuf obuf
+#define _seqbufptr 0
+#define _SEQ_ADVBUF(x) len=x
+
+void
+do_midi_msg(int synthno, unsigned char *msg, int mlen)
+{
+ switch (msg[0] & 0xf0)
+ {
+ case 0x90:
+ if (msg[2] != 0)
+ {
+ STORE(SEQ_START_NOTE(synthno, msg[0] & 0x0f, msg[1], msg[2]));
+ break;
+ }
+ msg[2] = 64;
+
+ case 0x80:
+ STORE(SEQ_STOP_NOTE(synthno, msg[0] & 0x0f, msg[1], msg[2]));
+ break;
+
+ case 0xA0:
+ STORE(SEQ_KEY_PRESSURE(synthno, msg[0] & 0x0f, msg[1], msg[2]));
+ break;
+
+ case 0xB0:
+ STORE(SEQ_CONTROL(synthno, msg[0] & 0x0f,
+ msg[1], msg[2]));
+ break;
+
+ case 0xC0:
+ STORE(SEQ_SET_PATCH(synthno, msg[0] & 0x0f, msg[1]));
+ break;
+
+ case 0xD0:
+ STORE(SEQ_CHN_PRESSURE(synthno, msg[0] & 0x0f, msg[1]));
+ break;
+
+ case 0xE0:
+ STORE(SEQ_BENDER(synthno, msg[0] & 0x0f,
+ (msg[1] & 0x7f) | ((msg[2] & 0x7f) << 7)));
+ break;
+
+ default:
+ /* printk( "MPU: Unknown midi channel message %02x\n", msg[0]); */
+ ;
+ }
+}
+EXPORT_SYMBOL(do_midi_msg);
+
+static void
+midi_outc(int midi_dev, int data)
+{
+ int timeout;
+
+ for (timeout = 0; timeout < 3200; timeout++)
+ if (midi_devs[midi_dev]->outputc(midi_dev, (unsigned char) (data & 0xff)))
+ {
+ if (data & 0x80) /*
+ * Status byte
+ */
+ prev_out_status[midi_dev] =
+ (unsigned char) (data & 0xff); /*
+ * Store for running status
+ */
+ return; /*
+ * Mission complete
+ */
+ }
+ /*
+ * Sorry! No space on buffers.
+ */
+ printk("Midi send timed out\n");
+}
+
+static int
+prefix_cmd(int midi_dev, unsigned char status)
+{
+ if ((char *) midi_devs[midi_dev]->prefix_cmd == NULL)
+ return 1;
+
+ return midi_devs[midi_dev]->prefix_cmd(midi_dev, status);
+}
+
+static void
+midi_synth_input(int orig_dev, unsigned char data)
+{
+ int dev;
+ struct midi_input_info *inc;
+
+ static unsigned char len_tab[] = /* # of data bytes following a status
+ */
+ {
+ 2, /* 8x */
+ 2, /* 9x */
+ 2, /* Ax */
+ 2, /* Bx */
+ 1, /* Cx */
+ 1, /* Dx */
+ 2, /* Ex */
+ 0 /* Fx */
+ };
+
+ if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL)
+ return;
+
+ if (data == 0xfe) /* Ignore active sensing */
+ return;
+
+ dev = midi2synth[orig_dev];
+ inc = &midi_devs[orig_dev]->in_info;
+
+ switch (inc->m_state)
+ {
+ case MST_INIT:
+ if (data & 0x80) /* MIDI status byte */
+ {
+ if ((data & 0xf0) == 0xf0) /* Common message */
+ {
+ switch (data)
+ {
+ case 0xf0: /* Sysex */
+ inc->m_state = MST_SYSEX;
+ break; /* Sysex */
+
+ case 0xf1: /* MTC quarter frame */
+ case 0xf3: /* Song select */
+ inc->m_state = MST_DATA;
+ inc->m_ptr = 1;
+ inc->m_left = 1;
+ inc->m_buf[0] = data;
+ break;
+
+ case 0xf2: /* Song position pointer */
+ inc->m_state = MST_DATA;
+ inc->m_ptr = 1;
+ inc->m_left = 2;
+ inc->m_buf[0] = data;
+ break;
+
+ default:
+ inc->m_buf[0] = data;
+ inc->m_ptr = 1;
+ do_midi_msg(dev, inc->m_buf, inc->m_ptr);
+ inc->m_ptr = 0;
+ inc->m_left = 0;
+ }
+ } else
+ {
+ inc->m_state = MST_DATA;
+ inc->m_ptr = 1;
+ inc->m_left = len_tab[(data >> 4) - 8];
+ inc->m_buf[0] = inc->m_prev_status = data;
+ }
+ } else if (inc->m_prev_status & 0x80) {
+ /* Data byte (use running status) */
+ inc->m_ptr = 2;
+ inc->m_buf[1] = data;
+ inc->m_buf[0] = inc->m_prev_status;
+ inc->m_left = len_tab[(inc->m_buf[0] >> 4) - 8] - 1;
+ if (inc->m_left > 0)
+ inc->m_state = MST_DATA; /* Not done yet */
+ else {
+ inc->m_state = MST_INIT;
+ do_midi_msg(dev, inc->m_buf, inc->m_ptr);
+ inc->m_ptr = 0;
+ }
+ }
+ break; /* MST_INIT */
+
+ case MST_DATA:
+ inc->m_buf[inc->m_ptr++] = data;
+ if (--inc->m_left <= 0)
+ {
+ inc->m_state = MST_INIT;
+ do_midi_msg(dev, inc->m_buf, inc->m_ptr);
+ inc->m_ptr = 0;
+ }
+ break; /* MST_DATA */
+
+ case MST_SYSEX:
+ if (data == 0xf7) /* Sysex end */
+ {
+ inc->m_state = MST_INIT;
+ inc->m_left = 0;
+ inc->m_ptr = 0;
+ }
+ break; /* MST_SYSEX */
+
+ default:
+ printk("MIDI%d: Unexpected state %d (%02x)\n", orig_dev, inc->m_state, (int) data);
+ inc->m_state = MST_INIT;
+ }
+}
+
+static void
+leave_sysex(int dev)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int timeout = 0;
+
+ if (!sysex_state[dev])
+ return;
+
+ sysex_state[dev] = 0;
+
+ while (!midi_devs[orig_dev]->outputc(orig_dev, 0xf7) &&
+ timeout < 1000)
+ timeout++;
+
+ sysex_state[dev] = 0;
+}
+
+static void
+midi_synth_output(int dev)
+{
+ /*
+ * Currently NOP
+ */
+}
+
+int midi_synth_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ /*
+ * int orig_dev = synth_devs[dev]->midi_dev;
+ */
+
+ switch (cmd) {
+
+ case SNDCTL_SYNTH_INFO:
+ if (__copy_to_user(arg, synth_devs[dev]->info, sizeof(struct synth_info)))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_SYNTH_MEMAVL:
+ return 0x7fffffff;
+
+ default:
+ return -EINVAL;
+ }
+}
+EXPORT_SYMBOL(midi_synth_ioctl);
+
+int
+midi_synth_kill_note(int dev, int channel, int note, int velocity)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int msg, chn;
+
+ if (note < 0 || note > 127)
+ return 0;
+ if (channel < 0 || channel > 15)
+ return 0;
+ if (velocity < 0)
+ velocity = 0;
+ if (velocity > 127)
+ velocity = 127;
+
+ leave_sysex(dev);
+
+ msg = prev_out_status[orig_dev] & 0xf0;
+ chn = prev_out_status[orig_dev] & 0x0f;
+
+ if (chn == channel && ((msg == 0x90 && velocity == 64) || msg == 0x80))
+ { /*
+ * Use running status
+ */
+ if (!prefix_cmd(orig_dev, note))
+ return 0;
+
+ midi_outc(orig_dev, note);
+
+ if (msg == 0x90) /*
+ * Running status = Note on
+ */
+ midi_outc(orig_dev, 0); /*
+ * Note on with velocity 0 == note
+ * off
+ */
+ else
+ midi_outc(orig_dev, velocity);
+ } else
+ {
+ if (velocity == 64)
+ {
+ if (!prefix_cmd(orig_dev, 0x90 | (channel & 0x0f)))
+ return 0;
+ midi_outc(orig_dev, 0x90 | (channel & 0x0f)); /*
+ * Note on
+ */
+ midi_outc(orig_dev, note);
+ midi_outc(orig_dev, 0); /*
+ * Zero G
+ */
+ } else
+ {
+ if (!prefix_cmd(orig_dev, 0x80 | (channel & 0x0f)))
+ return 0;
+ midi_outc(orig_dev, 0x80 | (channel & 0x0f)); /*
+ * Note off
+ */
+ midi_outc(orig_dev, note);
+ midi_outc(orig_dev, velocity);
+ }
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL(midi_synth_kill_note);
+
+int
+midi_synth_set_instr(int dev, int channel, int instr_no)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+
+ if (instr_no < 0 || instr_no > 127)
+ instr_no = 0;
+ if (channel < 0 || channel > 15)
+ return 0;
+
+ leave_sysex(dev);
+
+ if (!prefix_cmd(orig_dev, 0xc0 | (channel & 0x0f)))
+ return 0;
+ midi_outc(orig_dev, 0xc0 | (channel & 0x0f)); /*
+ * Program change
+ */
+ midi_outc(orig_dev, instr_no);
+
+ return 0;
+}
+EXPORT_SYMBOL(midi_synth_set_instr);
+
+int
+midi_synth_start_note(int dev, int channel, int note, int velocity)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int msg, chn;
+
+ if (note < 0 || note > 127)
+ return 0;
+ if (channel < 0 || channel > 15)
+ return 0;
+ if (velocity < 0)
+ velocity = 0;
+ if (velocity > 127)
+ velocity = 127;
+
+ leave_sysex(dev);
+
+ msg = prev_out_status[orig_dev] & 0xf0;
+ chn = prev_out_status[orig_dev] & 0x0f;
+
+ if (chn == channel && msg == 0x90)
+ { /*
+ * Use running status
+ */
+ if (!prefix_cmd(orig_dev, note))
+ return 0;
+ midi_outc(orig_dev, note);
+ midi_outc(orig_dev, velocity);
+ } else
+ {
+ if (!prefix_cmd(orig_dev, 0x90 | (channel & 0x0f)))
+ return 0;
+ midi_outc(orig_dev, 0x90 | (channel & 0x0f)); /*
+ * Note on
+ */
+ midi_outc(orig_dev, note);
+ midi_outc(orig_dev, velocity);
+ }
+ return 0;
+}
+EXPORT_SYMBOL(midi_synth_start_note);
+
+void
+midi_synth_reset(int dev)
+{
+
+ leave_sysex(dev);
+}
+EXPORT_SYMBOL(midi_synth_reset);
+
+int
+midi_synth_open(int dev, int mode)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int err;
+ struct midi_input_info *inc;
+
+ if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL)
+ return -ENXIO;
+
+ midi2synth[orig_dev] = dev;
+ sysex_state[dev] = 0;
+ prev_out_status[orig_dev] = 0;
+
+ if ((err = midi_devs[orig_dev]->open(orig_dev, mode,
+ midi_synth_input, midi_synth_output)) < 0)
+ return err;
+ inc = &midi_devs[orig_dev]->in_info;
+
+ /* save_flags(flags);
+ cli();
+ don't know against what irqhandler to protect*/
+ inc->m_busy = 0;
+ inc->m_state = MST_INIT;
+ inc->m_ptr = 0;
+ inc->m_left = 0;
+ inc->m_prev_status = 0x00;
+ /* restore_flags(flags); */
+
+ return 1;
+}
+EXPORT_SYMBOL(midi_synth_open);
+
+void
+midi_synth_close(int dev)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+
+ leave_sysex(dev);
+
+ /*
+ * Shut up the synths by sending just single active sensing message.
+ */
+ midi_devs[orig_dev]->outputc(orig_dev, 0xfe);
+
+ midi_devs[orig_dev]->close(orig_dev);
+}
+EXPORT_SYMBOL(midi_synth_close);
+
+void
+midi_synth_hw_control(int dev, unsigned char *event)
+{
+}
+EXPORT_SYMBOL(midi_synth_hw_control);
+
+int
+midi_synth_load_patch(int dev, int format, const char __user *addr,
+ int offs, int count, int pmgr_flag)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+
+ struct sysex_info sysex;
+ int i;
+ unsigned long left, src_offs, eox_seen = 0;
+ int first_byte = 1;
+ int hdr_size = (unsigned long) &sysex.data[0] - (unsigned long) &sysex;
+
+ leave_sysex(dev);
+
+ if (!prefix_cmd(orig_dev, 0xf0))
+ return 0;
+
+ if (format != SYSEX_PATCH)
+ {
+/* printk("MIDI Error: Invalid patch format (key) 0x%x\n", format);*/
+ return -EINVAL;
+ }
+ if (count < hdr_size)
+ {
+/* printk("MIDI Error: Patch header too short\n");*/
+ return -EINVAL;
+ }
+ count -= hdr_size;
+
+ /*
+ * Copy the header from user space but ignore the first bytes which have
+ * been transferred already.
+ */
+
+ if(copy_from_user(&((char *) &sysex)[offs], &(addr)[offs], hdr_size - offs))
+ return -EFAULT;
+
+ if (count < sysex.len)
+ {
+/* printk(KERN_WARNING "MIDI Warning: Sysex record too short (%d<%d)\n", count, (int) sysex.len);*/
+ sysex.len = count;
+ }
+ left = sysex.len;
+ src_offs = 0;
+
+ for (i = 0; i < left && !signal_pending(current); i++)
+ {
+ unsigned char data;
+
+ get_user(*(unsigned char *) &data, (unsigned char __user *) &((addr)[hdr_size + i]));
+
+ eox_seen = (i > 0 && data & 0x80); /* End of sysex */
+
+ if (eox_seen && data != 0xf7)
+ data = 0xf7;
+
+ if (i == 0)
+ {
+ if (data != 0xf0)
+ {
+ printk(KERN_WARNING "midi_synth: Sysex start missing\n");
+ return -EINVAL;
+ }
+ }
+ while (!midi_devs[orig_dev]->outputc(orig_dev, (unsigned char) (data & 0xff)) &&
+ !signal_pending(current))
+ schedule();
+
+ if (!first_byte && data & 0x80)
+ return 0;
+ first_byte = 0;
+ }
+
+ if (!eox_seen)
+ midi_outc(orig_dev, 0xf7);
+ return 0;
+}
+EXPORT_SYMBOL(midi_synth_load_patch);
+
+void midi_synth_panning(int dev, int channel, int pressure)
+{
+}
+EXPORT_SYMBOL(midi_synth_panning);
+
+void midi_synth_aftertouch(int dev, int channel, int pressure)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int msg, chn;
+
+ if (pressure < 0 || pressure > 127)
+ return;
+ if (channel < 0 || channel > 15)
+ return;
+
+ leave_sysex(dev);
+
+ msg = prev_out_status[orig_dev] & 0xf0;
+ chn = prev_out_status[orig_dev] & 0x0f;
+
+ if (msg != 0xd0 || chn != channel) /*
+ * Test for running status
+ */
+ {
+ if (!prefix_cmd(orig_dev, 0xd0 | (channel & 0x0f)))
+ return;
+ midi_outc(orig_dev, 0xd0 | (channel & 0x0f)); /*
+ * Channel pressure
+ */
+ } else if (!prefix_cmd(orig_dev, pressure))
+ return;
+
+ midi_outc(orig_dev, pressure);
+}
+EXPORT_SYMBOL(midi_synth_aftertouch);
+
+void
+midi_synth_controller(int dev, int channel, int ctrl_num, int value)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int chn, msg;
+
+ if (ctrl_num < 0 || ctrl_num > 127)
+ return;
+ if (channel < 0 || channel > 15)
+ return;
+
+ leave_sysex(dev);
+
+ msg = prev_out_status[orig_dev] & 0xf0;
+ chn = prev_out_status[orig_dev] & 0x0f;
+
+ if (msg != 0xb0 || chn != channel)
+ {
+ if (!prefix_cmd(orig_dev, 0xb0 | (channel & 0x0f)))
+ return;
+ midi_outc(orig_dev, 0xb0 | (channel & 0x0f));
+ } else if (!prefix_cmd(orig_dev, ctrl_num))
+ return;
+
+ midi_outc(orig_dev, ctrl_num);
+ midi_outc(orig_dev, value & 0x7f);
+}
+EXPORT_SYMBOL(midi_synth_controller);
+
+void
+midi_synth_bender(int dev, int channel, int value)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int msg, prev_chn;
+
+ if (channel < 0 || channel > 15)
+ return;
+
+ if (value < 0 || value > 16383)
+ return;
+
+ leave_sysex(dev);
+
+ msg = prev_out_status[orig_dev] & 0xf0;
+ prev_chn = prev_out_status[orig_dev] & 0x0f;
+
+ if (msg != 0xd0 || prev_chn != channel) /*
+ * Test for running status
+ */
+ {
+ if (!prefix_cmd(orig_dev, 0xe0 | (channel & 0x0f)))
+ return;
+ midi_outc(orig_dev, 0xe0 | (channel & 0x0f));
+ } else if (!prefix_cmd(orig_dev, value & 0x7f))
+ return;
+
+ midi_outc(orig_dev, value & 0x7f);
+ midi_outc(orig_dev, (value >> 7) & 0x7f);
+}
+EXPORT_SYMBOL(midi_synth_bender);
+
+void
+midi_synth_setup_voice(int dev, int voice, int channel)
+{
+}
+EXPORT_SYMBOL(midi_synth_setup_voice);
+
+int
+midi_synth_send_sysex(int dev, unsigned char *bytes, int len)
+{
+ int orig_dev = synth_devs[dev]->midi_dev;
+ int i;
+
+ for (i = 0; i < len; i++)
+ {
+ switch (bytes[i])
+ {
+ case 0xf0: /* Start sysex */
+ if (!prefix_cmd(orig_dev, 0xf0))
+ return 0;
+ sysex_state[dev] = 1;
+ break;
+
+ case 0xf7: /* End sysex */
+ if (!sysex_state[dev]) /* Orphan sysex end */
+ return 0;
+ sysex_state[dev] = 0;
+ break;
+
+ default:
+ if (!sysex_state[dev])
+ return 0;
+
+ if (bytes[i] & 0x80) /* Error. Another message before sysex end */
+ {
+ bytes[i] = 0xf7; /* Sysex end */
+ sysex_state[dev] = 0;
+ }
+ }
+
+ if (!midi_devs[orig_dev]->outputc(orig_dev, bytes[i]))
+ {
+/*
+ * Hardware level buffer is full. Abort the sysex message.
+ */
+
+ int timeout = 0;
+
+ bytes[i] = 0xf7;
+ sysex_state[dev] = 0;
+
+ while (!midi_devs[orig_dev]->outputc(orig_dev, bytes[i]) &&
+ timeout < 1000)
+ timeout++;
+ }
+ if (!sysex_state[dev])
+ return 0;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL(midi_synth_send_sysex);
+
diff --git a/sound/oss/midi_synth.h b/sound/oss/midi_synth.h
new file mode 100644
index 0000000..6bc9d00
--- /dev/null
+++ b/sound/oss/midi_synth.h
@@ -0,0 +1,47 @@
+int midi_synth_ioctl (int dev,
+ unsigned int cmd, void __user * arg);
+int midi_synth_kill_note (int dev, int channel, int note, int velocity);
+int midi_synth_set_instr (int dev, int channel, int instr_no);
+int midi_synth_start_note (int dev, int channel, int note, int volume);
+void midi_synth_reset (int dev);
+int midi_synth_open (int dev, int mode);
+void midi_synth_close (int dev);
+void midi_synth_hw_control (int dev, unsigned char *event);
+int midi_synth_load_patch (int dev, int format, const char __user * addr,
+ int offs, int count, int pmgr_flag);
+void midi_synth_panning (int dev, int channel, int pressure);
+void midi_synth_aftertouch (int dev, int channel, int pressure);
+void midi_synth_controller (int dev, int channel, int ctrl_num, int value);
+void midi_synth_bender (int dev, int chn, int value);
+void midi_synth_setup_voice (int dev, int voice, int chn);
+int midi_synth_send_sysex(int dev, unsigned char *bytes,int len);
+
+#ifndef _MIDI_SYNTH_C_
+static struct synth_info std_synth_info =
+{MIDI_SYNTH_NAME, 0, SYNTH_TYPE_MIDI, 0, 0, 128, 0, 128, MIDI_SYNTH_CAPS};
+
+static struct synth_operations std_midi_synth =
+{
+ .owner = THIS_MODULE,
+ .id = "MIDI",
+ .info = &std_synth_info,
+ .midi_dev = 0,
+ .synth_type = SYNTH_TYPE_MIDI,
+ .synth_subtype = 0,
+ .open = midi_synth_open,
+ .close = midi_synth_close,
+ .ioctl = midi_synth_ioctl,
+ .kill_note = midi_synth_kill_note,
+ .start_note = midi_synth_start_note,
+ .set_instr = midi_synth_set_instr,
+ .reset = midi_synth_reset,
+ .hw_control = midi_synth_hw_control,
+ .load_patch = midi_synth_load_patch,
+ .aftertouch = midi_synth_aftertouch,
+ .controller = midi_synth_controller,
+ .panning = midi_synth_panning,
+ .bender = midi_synth_bender,
+ .setup_voice = midi_synth_setup_voice,
+ .send_sysex = midi_synth_send_sysex
+};
+#endif
diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c
new file mode 100644
index 0000000..a40be0c
--- /dev/null
+++ b/sound/oss/midibuf.c
@@ -0,0 +1,424 @@
+/*
+ * sound/oss/midibuf.c
+ *
+ * Device file manager for /dev/midi#
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+/*
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ */
+#include <linux/stddef.h>
+#include <linux/kmod.h>
+#include <linux/spinlock.h>
+#define MIDIBUF_C
+
+#include "sound_config.h"
+
+
+/*
+ * Don't make MAX_QUEUE_SIZE larger than 4000
+ */
+
+#define MAX_QUEUE_SIZE 4000
+
+static wait_queue_head_t midi_sleeper[MAX_MIDI_DEV];
+static wait_queue_head_t input_sleeper[MAX_MIDI_DEV];
+
+struct midi_buf
+{
+ int len, head, tail;
+ unsigned char queue[MAX_QUEUE_SIZE];
+};
+
+struct midi_parms
+{
+ long prech_timeout; /*
+ * Timeout before the first ch
+ */
+};
+
+static struct midi_buf *midi_out_buf[MAX_MIDI_DEV] = {NULL};
+static struct midi_buf *midi_in_buf[MAX_MIDI_DEV] = {NULL};
+static struct midi_parms parms[MAX_MIDI_DEV];
+
+static void midi_poll(unsigned long dummy);
+
+
+static DEFINE_TIMER(poll_timer, midi_poll, 0, 0);
+
+static volatile int open_devs;
+static DEFINE_SPINLOCK(lock);
+
+#define DATA_AVAIL(q) (q->len)
+#define SPACE_AVAIL(q) (MAX_QUEUE_SIZE - q->len)
+
+#define QUEUE_BYTE(q, data) \
+ if (SPACE_AVAIL(q)) \
+ { \
+ unsigned long flags; \
+ spin_lock_irqsave(&lock, flags); \
+ q->queue[q->tail] = (data); \
+ q->len++; q->tail = (q->tail+1) % MAX_QUEUE_SIZE; \
+ spin_unlock_irqrestore(&lock, flags); \
+ }
+
+#define REMOVE_BYTE(q, data) \
+ if (DATA_AVAIL(q)) \
+ { \
+ unsigned long flags; \
+ spin_lock_irqsave(&lock, flags); \
+ data = q->queue[q->head]; \
+ q->len--; q->head = (q->head+1) % MAX_QUEUE_SIZE; \
+ spin_unlock_irqrestore(&lock, flags); \
+ }
+
+static void drain_midi_queue(int dev)
+{
+
+ /*
+ * Give the Midi driver time to drain its output queues
+ */
+
+ if (midi_devs[dev]->buffer_status != NULL)
+ while (!signal_pending(current) && midi_devs[dev]->buffer_status(dev))
+ interruptible_sleep_on_timeout(&midi_sleeper[dev],
+ HZ/10);
+}
+
+static void midi_input_intr(int dev, unsigned char data)
+{
+ if (midi_in_buf[dev] == NULL)
+ return;
+
+ if (data == 0xfe) /*
+ * Active sensing
+ */
+ return; /*
+ * Ignore
+ */
+
+ if (SPACE_AVAIL(midi_in_buf[dev])) {
+ QUEUE_BYTE(midi_in_buf[dev], data);
+ wake_up(&input_sleeper[dev]);
+ }
+}
+
+static void midi_output_intr(int dev)
+{
+ /*
+ * Currently NOP
+ */
+}
+
+static void midi_poll(unsigned long dummy)
+{
+ unsigned long flags;
+ int dev;
+
+ spin_lock_irqsave(&lock, flags);
+ if (open_devs)
+ {
+ for (dev = 0; dev < num_midis; dev++)
+ if (midi_devs[dev] != NULL && midi_out_buf[dev] != NULL)
+ {
+ int ok = 1;
+
+ while (DATA_AVAIL(midi_out_buf[dev]) && ok)
+ {
+ int c = midi_out_buf[dev]->queue[midi_out_buf[dev]->head];
+
+ spin_unlock_irqrestore(&lock,flags);/* Give some time to others */
+ ok = midi_devs[dev]->outputc(dev, c);
+ spin_lock_irqsave(&lock, flags);
+ midi_out_buf[dev]->head = (midi_out_buf[dev]->head + 1) % MAX_QUEUE_SIZE;
+ midi_out_buf[dev]->len--;
+ }
+
+ if (DATA_AVAIL(midi_out_buf[dev]) < 100)
+ wake_up(&midi_sleeper[dev]);
+ }
+ poll_timer.expires = (1) + jiffies;
+ add_timer(&poll_timer);
+ /*
+ * Come back later
+ */
+ }
+ spin_unlock_irqrestore(&lock, flags);
+}
+
+int MIDIbuf_open(int dev, struct file *file)
+{
+ int mode, err;
+
+ dev = dev >> 4;
+ mode = translate_mode(file);
+
+ if (num_midis > MAX_MIDI_DEV)
+ {
+ printk(KERN_ERR "midi: Too many midi interfaces\n");
+ num_midis = MAX_MIDI_DEV;
+ }
+ if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL)
+ return -ENXIO;
+ /*
+ * Interrupts disabled. Be careful
+ */
+
+ module_put(midi_devs[dev]->owner);
+
+ if ((err = midi_devs[dev]->open(dev, mode,
+ midi_input_intr, midi_output_intr)) < 0)
+ return err;
+
+ parms[dev].prech_timeout = MAX_SCHEDULE_TIMEOUT;
+ midi_in_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+
+ if (midi_in_buf[dev] == NULL)
+ {
+ printk(KERN_WARNING "midi: Can't allocate buffer\n");
+ midi_devs[dev]->close(dev);
+ return -EIO;
+ }
+ midi_in_buf[dev]->len = midi_in_buf[dev]->head = midi_in_buf[dev]->tail = 0;
+
+ midi_out_buf[dev] = (struct midi_buf *) vmalloc(sizeof(struct midi_buf));
+
+ if (midi_out_buf[dev] == NULL)
+ {
+ printk(KERN_WARNING "midi: Can't allocate buffer\n");
+ midi_devs[dev]->close(dev);
+ vfree(midi_in_buf[dev]);
+ midi_in_buf[dev] = NULL;
+ return -EIO;
+ }
+ midi_out_buf[dev]->len = midi_out_buf[dev]->head = midi_out_buf[dev]->tail = 0;
+ open_devs++;
+
+ init_waitqueue_head(&midi_sleeper[dev]);
+ init_waitqueue_head(&input_sleeper[dev]);
+
+ if (open_devs < 2) /* This was first open */
+ {
+ poll_timer.expires = 1 + jiffies;
+ add_timer(&poll_timer); /* Start polling */
+ }
+ return err;
+}
+
+void MIDIbuf_release(int dev, struct file *file)
+{
+ int mode;
+
+ dev = dev >> 4;
+ mode = translate_mode(file);
+
+ if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL)
+ return;
+
+ /*
+ * Wait until the queue is empty
+ */
+
+ if (mode != OPEN_READ)
+ {
+ midi_devs[dev]->outputc(dev, 0xfe); /*
+ * Active sensing to shut the
+ * devices
+ */
+
+ while (!signal_pending(current) && DATA_AVAIL(midi_out_buf[dev]))
+ interruptible_sleep_on(&midi_sleeper[dev]);
+ /*
+ * Sync
+ */
+
+ drain_midi_queue(dev); /*
+ * Ensure the output queues are empty
+ */
+ }
+
+ midi_devs[dev]->close(dev);
+
+ open_devs--;
+ if (open_devs == 0)
+ del_timer_sync(&poll_timer);
+ vfree(midi_in_buf[dev]);
+ vfree(midi_out_buf[dev]);
+ midi_in_buf[dev] = NULL;
+ midi_out_buf[dev] = NULL;
+
+ module_put(midi_devs[dev]->owner);
+}
+
+int MIDIbuf_write(int dev, struct file *file, const char __user *buf, int count)
+{
+ int c, n, i;
+ unsigned char tmp_data;
+
+ dev = dev >> 4;
+
+ if (!count)
+ return 0;
+
+ c = 0;
+
+ while (c < count)
+ {
+ n = SPACE_AVAIL(midi_out_buf[dev]);
+
+ if (n == 0) { /*
+ * No space just now.
+ */
+
+ if (file->f_flags & O_NONBLOCK) {
+ c = -EAGAIN;
+ goto out;
+ }
+
+ interruptible_sleep_on(&midi_sleeper[dev]);
+ if (signal_pending(current))
+ {
+ c = -EINTR;
+ goto out;
+ }
+ n = SPACE_AVAIL(midi_out_buf[dev]);
+ }
+ if (n > (count - c))
+ n = count - c;
+
+ for (i = 0; i < n; i++)
+ {
+ /* BROKE BROKE BROKE - CANT DO THIS WITH CLI !! */
+ /* yes, think the same, so I removed the cli() brackets
+ QUEUE_BYTE is protected against interrupts */
+ if (copy_from_user((char *) &tmp_data, &(buf)[c], 1)) {
+ c = -EFAULT;
+ goto out;
+ }
+ QUEUE_BYTE(midi_out_buf[dev], tmp_data);
+ c++;
+ }
+ }
+out:
+ return c;
+}
+
+
+int MIDIbuf_read(int dev, struct file *file, char __user *buf, int count)
+{
+ int n, c = 0;
+ unsigned char tmp_data;
+
+ dev = dev >> 4;
+
+ if (!DATA_AVAIL(midi_in_buf[dev])) { /*
+ * No data yet, wait
+ */
+ if (file->f_flags & O_NONBLOCK) {
+ c = -EAGAIN;
+ goto out;
+ }
+ interruptible_sleep_on_timeout(&input_sleeper[dev],
+ parms[dev].prech_timeout);
+
+ if (signal_pending(current))
+ c = -EINTR; /* The user is getting restless */
+ }
+ if (c == 0 && DATA_AVAIL(midi_in_buf[dev])) /*
+ * Got some bytes
+ */
+ {
+ n = DATA_AVAIL(midi_in_buf[dev]);
+ if (n > count)
+ n = count;
+ c = 0;
+
+ while (c < n)
+ {
+ char *fixit;
+ REMOVE_BYTE(midi_in_buf[dev], tmp_data);
+ fixit = (char *) &tmp_data;
+ /* BROKE BROKE BROKE */
+ /* yes removed the cli() brackets again
+ should q->len,tail&head be atomic_t? */
+ if (copy_to_user(&(buf)[c], fixit, 1)) {
+ c = -EFAULT;
+ goto out;
+ }
+ c++;
+ }
+ }
+out:
+ return c;
+}
+
+int MIDIbuf_ioctl(int dev, struct file *file,
+ unsigned int cmd, void __user *arg)
+{
+ int val;
+
+ dev = dev >> 4;
+
+ if (((cmd >> 8) & 0xff) == 'C')
+ {
+ if (midi_devs[dev]->coproc) /* Coprocessor ioctl */
+ return midi_devs[dev]->coproc->ioctl(midi_devs[dev]->coproc->devc, cmd, arg, 0);
+/* printk("/dev/midi%d: No coprocessor for this device\n", dev);*/
+ return -ENXIO;
+ }
+ else
+ {
+ switch (cmd)
+ {
+ case SNDCTL_MIDI_PRETIME:
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ if (val < 0)
+ val = 0;
+ val = (HZ * val) / 10;
+ parms[dev].prech_timeout = val;
+ return put_user(val, (int __user *)arg);
+
+ default:
+ if (!midi_devs[dev]->ioctl)
+ return -EINVAL;
+ return midi_devs[dev]->ioctl(dev, cmd, arg);
+ }
+ }
+}
+
+/* No kernel lock - fine */
+unsigned int MIDIbuf_poll(int dev, struct file *file, poll_table * wait)
+{
+ unsigned int mask = 0;
+
+ dev = dev >> 4;
+
+ /* input */
+ poll_wait(file, &input_sleeper[dev], wait);
+ if (DATA_AVAIL(midi_in_buf[dev]))
+ mask |= POLLIN | POLLRDNORM;
+
+ /* output */
+ poll_wait(file, &midi_sleeper[dev], wait);
+ if (!SPACE_AVAIL(midi_out_buf[dev]))
+ mask |= POLLOUT | POLLWRNORM;
+
+ return mask;
+}
+
+
+int MIDIbuf_avail(int dev)
+{
+ if (midi_in_buf[dev])
+ return DATA_AVAIL (midi_in_buf[dev]);
+ return 0;
+}
+EXPORT_SYMBOL(MIDIbuf_avail);
+
diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c
new file mode 100644
index 0000000..6c0a770
--- /dev/null
+++ b/sound/oss/mpu401.c
@@ -0,0 +1,1815 @@
+/*
+ * sound/oss/mpu401.c
+ *
+ * The low level driver for Roland MPU-401 compatible Midi cards.
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Thomas Sailer ioctl code reworked (vmalloc/vfree removed)
+ * Alan Cox modularisation, use normal request_irq, use dev_id
+ * Bartlomiej Zolnierkiewicz removed some __init to allow using many drivers
+ * Chris Rankin Update the module-usage counter for the coprocessor
+ * Zwane Mwaikambo Changed attach/unload resource freeing
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+#define USE_SEQ_MACROS
+#define USE_SIMPLE_MACROS
+
+#include "sound_config.h"
+
+#include "coproc.h"
+#include "mpu401.h"
+
+static int timer_mode = TMR_INTERNAL, timer_caps = TMR_INTERNAL;
+
+struct mpu_config
+{
+ int base; /*
+ * I/O base
+ */
+ int irq;
+ int opened; /*
+ * Open mode
+ */
+ int devno;
+ int synthno;
+ int uart_mode;
+ int initialized;
+ int mode;
+#define MODE_MIDI 1
+#define MODE_SYNTH 2
+ unsigned char version, revision;
+ unsigned int capabilities;
+#define MPU_CAP_INTLG 0x10000000
+#define MPU_CAP_SYNC 0x00000010
+#define MPU_CAP_FSK 0x00000020
+#define MPU_CAP_CLS 0x00000040
+#define MPU_CAP_SMPTE 0x00000080
+#define MPU_CAP_2PORT 0x00000001
+ int timer_flag;
+
+#define MBUF_MAX 10
+#define BUFTEST(dc) if (dc->m_ptr >= MBUF_MAX || dc->m_ptr < 0) \
+ {printk( "MPU: Invalid buffer pointer %d/%d, s=%d\n", dc->m_ptr, dc->m_left, dc->m_state);dc->m_ptr--;}
+ int m_busy;
+ unsigned char m_buf[MBUF_MAX];
+ int m_ptr;
+ int m_state;
+ int m_left;
+ unsigned char last_status;
+ void (*inputintr) (int dev, unsigned char data);
+ int shared_irq;
+ int *osp;
+ spinlock_t lock;
+ };
+
+#define DATAPORT(base) (base)
+#define COMDPORT(base) (base+1)
+#define STATPORT(base) (base+1)
+
+
+static void mpu401_close(int dev);
+
+static inline int mpu401_status(struct mpu_config *devc)
+{
+ return inb(STATPORT(devc->base));
+}
+
+#define input_avail(devc) (!(mpu401_status(devc)&INPUT_AVAIL))
+#define output_ready(devc) (!(mpu401_status(devc)&OUTPUT_READY))
+
+static inline void write_command(struct mpu_config *devc, unsigned char cmd)
+{
+ outb(cmd, COMDPORT(devc->base));
+}
+
+static inline int read_data(struct mpu_config *devc)
+{
+ return inb(DATAPORT(devc->base));
+}
+
+static inline void write_data(struct mpu_config *devc, unsigned char byte)
+{
+ outb(byte, DATAPORT(devc->base));
+}
+
+#define OUTPUT_READY 0x40
+#define INPUT_AVAIL 0x80
+#define MPU_ACK 0xFE
+#define MPU_RESET 0xFF
+#define UART_MODE_ON 0x3F
+
+static struct mpu_config dev_conf[MAX_MIDI_DEV];
+
+static int n_mpu_devs;
+
+static int reset_mpu401(struct mpu_config *devc);
+static void set_uart_mode(int dev, struct mpu_config *devc, int arg);
+
+static int mpu_timer_init(int midi_dev);
+static void mpu_timer_interrupt(void);
+static void timer_ext_event(struct mpu_config *devc, int event, int parm);
+
+static struct synth_info mpu_synth_info_proto = {
+ "MPU-401 MIDI interface",
+ 0,
+ SYNTH_TYPE_MIDI,
+ MIDI_TYPE_MPU401,
+ 0, 128,
+ 0, 128,
+ SYNTH_CAP_INPUT
+};
+
+static struct synth_info mpu_synth_info[MAX_MIDI_DEV];
+
+/*
+ * States for the input scanner
+ */
+
+#define ST_INIT 0 /* Ready for timing byte or msg */
+#define ST_TIMED 1 /* Leading timing byte rcvd */
+#define ST_DATABYTE 2 /* Waiting for (nr_left) data bytes */
+
+#define ST_SYSMSG 100 /* System message (sysx etc). */
+#define ST_SYSEX 101 /* System exclusive msg */
+#define ST_MTC 102 /* Midi Time Code (MTC) qframe msg */
+#define ST_SONGSEL 103 /* Song select */
+#define ST_SONGPOS 104 /* Song position pointer */
+
+static unsigned char len_tab[] = /* # of data bytes following a status
+ */
+{
+ 2, /* 8x */
+ 2, /* 9x */
+ 2, /* Ax */
+ 2, /* Bx */
+ 1, /* Cx */
+ 1, /* Dx */
+ 2, /* Ex */
+ 0 /* Fx */
+};
+
+#define STORE(cmd) \
+{ \
+ int len; \
+ unsigned char obuf[8]; \
+ cmd; \
+ seq_input_event(obuf, len); \
+}
+
+#define _seqbuf obuf
+#define _seqbufptr 0
+#define _SEQ_ADVBUF(x) len=x
+
+static int mpu_input_scanner(struct mpu_config *devc, unsigned char midic)
+{
+
+ switch (devc->m_state)
+ {
+ case ST_INIT:
+ switch (midic)
+ {
+ case 0xf8:
+ /* Timer overflow */
+ break;
+
+ case 0xfc:
+ printk("<all end>");
+ break;
+
+ case 0xfd:
+ if (devc->timer_flag)
+ mpu_timer_interrupt();
+ break;
+
+ case 0xfe:
+ return MPU_ACK;
+
+ case 0xf0:
+ case 0xf1:
+ case 0xf2:
+ case 0xf3:
+ case 0xf4:
+ case 0xf5:
+ case 0xf6:
+ case 0xf7:
+ printk("<Trk data rq #%d>", midic & 0x0f);
+ break;
+
+ case 0xf9:
+ printk("<conductor rq>");
+ break;
+
+ case 0xff:
+ devc->m_state = ST_SYSMSG;
+ break;
+
+ default:
+ if (midic <= 0xef)
+ {
+ /* printk( "mpu time: %d ", midic); */
+ devc->m_state = ST_TIMED;
+ }
+ else
+ printk("<MPU: Unknown event %02x> ", midic);
+ }
+ break;
+
+ case ST_TIMED:
+ {
+ int msg = ((int) (midic & 0xf0) >> 4);
+
+ devc->m_state = ST_DATABYTE;
+
+ if (msg < 8) /* Data byte */
+ {
+ /* printk( "midi msg (running status) "); */
+ msg = ((int) (devc->last_status & 0xf0) >> 4);
+ msg -= 8;
+ devc->m_left = len_tab[msg] - 1;
+
+ devc->m_ptr = 2;
+ devc->m_buf[0] = devc->last_status;
+ devc->m_buf[1] = midic;
+
+ if (devc->m_left <= 0)
+ {
+ devc->m_state = ST_INIT;
+ do_midi_msg(devc->synthno, devc->m_buf, devc->m_ptr);
+ devc->m_ptr = 0;
+ }
+ }
+ else if (msg == 0xf) /* MPU MARK */
+ {
+ devc->m_state = ST_INIT;
+
+ switch (midic)
+ {
+ case 0xf8:
+ /* printk( "NOP "); */
+ break;
+
+ case 0xf9:
+ /* printk( "meas end "); */
+ break;
+
+ case 0xfc:
+ /* printk( "data end "); */
+ break;
+
+ default:
+ printk("Unknown MPU mark %02x\n", midic);
+ }
+ }
+ else
+ {
+ devc->last_status = midic;
+ /* printk( "midi msg "); */
+ msg -= 8;
+ devc->m_left = len_tab[msg];
+
+ devc->m_ptr = 1;
+ devc->m_buf[0] = midic;
+
+ if (devc->m_left <= 0)
+ {
+ devc->m_state = ST_INIT;
+ do_midi_msg(devc->synthno, devc->m_buf, devc->m_ptr);
+ devc->m_ptr = 0;
+ }
+ }
+ }
+ break;
+
+ case ST_SYSMSG:
+ switch (midic)
+ {
+ case 0xf0:
+ printk("<SYX>");
+ devc->m_state = ST_SYSEX;
+ break;
+
+ case 0xf1:
+ devc->m_state = ST_MTC;
+ break;
+
+ case 0xf2:
+ devc->m_state = ST_SONGPOS;
+ devc->m_ptr = 0;
+ break;
+
+ case 0xf3:
+ devc->m_state = ST_SONGSEL;
+ break;
+
+ case 0xf6:
+ /* printk( "tune_request\n"); */
+ devc->m_state = ST_INIT;
+
+ /*
+ * Real time messages
+ */
+ case 0xf8:
+ /* midi clock */
+ devc->m_state = ST_INIT;
+ timer_ext_event(devc, TMR_CLOCK, 0);
+ break;
+
+ case 0xfA:
+ devc->m_state = ST_INIT;
+ timer_ext_event(devc, TMR_START, 0);
+ break;
+
+ case 0xFB:
+ devc->m_state = ST_INIT;
+ timer_ext_event(devc, TMR_CONTINUE, 0);
+ break;
+
+ case 0xFC:
+ devc->m_state = ST_INIT;
+ timer_ext_event(devc, TMR_STOP, 0);
+ break;
+
+ case 0xFE:
+ /* active sensing */
+ devc->m_state = ST_INIT;
+ break;
+
+ case 0xff:
+ /* printk( "midi hard reset"); */
+ devc->m_state = ST_INIT;
+ break;
+
+ default:
+ printk("unknown MIDI sysmsg %0x\n", midic);
+ devc->m_state = ST_INIT;
+ }
+ break;
+
+ case ST_MTC:
+ devc->m_state = ST_INIT;
+ printk("MTC frame %x02\n", midic);
+ break;
+
+ case ST_SYSEX:
+ if (midic == 0xf7)
+ {
+ printk("<EOX>");
+ devc->m_state = ST_INIT;
+ }
+ else
+ printk("%02x ", midic);
+ break;
+
+ case ST_SONGPOS:
+ BUFTEST(devc);
+ devc->m_buf[devc->m_ptr++] = midic;
+ if (devc->m_ptr == 2)
+ {
+ devc->m_state = ST_INIT;
+ devc->m_ptr = 0;
+ timer_ext_event(devc, TMR_SPP,
+ ((devc->m_buf[1] & 0x7f) << 7) |
+ (devc->m_buf[0] & 0x7f));
+ }
+ break;
+
+ case ST_DATABYTE:
+ BUFTEST(devc);
+ devc->m_buf[devc->m_ptr++] = midic;
+ if ((--devc->m_left) <= 0)
+ {
+ devc->m_state = ST_INIT;
+ do_midi_msg(devc->synthno, devc->m_buf, devc->m_ptr);
+ devc->m_ptr = 0;
+ }
+ break;
+
+ default:
+ printk("Bad state %d ", devc->m_state);
+ devc->m_state = ST_INIT;
+ }
+ return 1;
+}
+
+static void mpu401_input_loop(struct mpu_config *devc)
+{
+ unsigned long flags;
+ int busy;
+ int n;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ busy = devc->m_busy;
+ devc->m_busy = 1;
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ if (busy) /* Already inside the scanner */
+ return;
+
+ n = 50;
+
+ while (input_avail(devc) && n-- > 0)
+ {
+ unsigned char c = read_data(devc);
+
+ if (devc->mode == MODE_SYNTH)
+ {
+ mpu_input_scanner(devc, c);
+ }
+ else if (devc->opened & OPEN_READ && devc->inputintr != NULL)
+ devc->inputintr(devc->devno, c);
+ }
+ devc->m_busy = 0;
+}
+
+static irqreturn_t mpuintr(int irq, void *dev_id)
+{
+ struct mpu_config *devc;
+ int dev = (int)(unsigned long) dev_id;
+ int handled = 0;
+
+ devc = &dev_conf[dev];
+
+ if (input_avail(devc))
+ {
+ handled = 1;
+ if (devc->base != 0 && (devc->opened & OPEN_READ || devc->mode == MODE_SYNTH))
+ mpu401_input_loop(devc);
+ else
+ {
+ /* Dummy read (just to acknowledge the interrupt) */
+ read_data(devc);
+ }
+ }
+ return IRQ_RETVAL(handled);
+}
+
+static int mpu401_open(int dev, int mode,
+ void (*input) (int dev, unsigned char data),
+ void (*output) (int dev)
+)
+{
+ int err;
+ struct mpu_config *devc;
+ struct coproc_operations *coprocessor;
+
+ if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL)
+ return -ENXIO;
+
+ devc = &dev_conf[dev];
+
+ if (devc->opened)
+ return -EBUSY;
+ /*
+ * Verify that the device is really running.
+ * Some devices (such as Ensoniq SoundScape don't
+ * work before the on board processor (OBP) is initialized
+ * by downloading its microcode.
+ */
+
+ if (!devc->initialized)
+ {
+ if (mpu401_status(devc) == 0xff) /* Bus float */
+ {
+ printk(KERN_ERR "mpu401: Device not initialized properly\n");
+ return -EIO;
+ }
+ reset_mpu401(devc);
+ }
+
+ if ( (coprocessor = midi_devs[dev]->coproc) != NULL )
+ {
+ if (!try_module_get(coprocessor->owner)) {
+ mpu401_close(dev);
+ return -ENODEV;
+ }
+
+ if ((err = coprocessor->open(coprocessor->devc, COPR_MIDI)) < 0)
+ {
+ printk(KERN_WARNING "MPU-401: Can't access coprocessor device\n");
+ mpu401_close(dev);
+ return err;
+ }
+ }
+
+ set_uart_mode(dev, devc, 1);
+ devc->mode = MODE_MIDI;
+ devc->synthno = 0;
+
+ mpu401_input_loop(devc);
+
+ devc->inputintr = input;
+ devc->opened = mode;
+
+ return 0;
+}
+
+static void mpu401_close(int dev)
+{
+ struct mpu_config *devc;
+ struct coproc_operations *coprocessor;
+
+ devc = &dev_conf[dev];
+ if (devc->uart_mode)
+ reset_mpu401(devc); /*
+ * This disables the UART mode
+ */
+ devc->mode = 0;
+ devc->inputintr = NULL;
+
+ coprocessor = midi_devs[dev]->coproc;
+ if (coprocessor) {
+ coprocessor->close(coprocessor->devc, COPR_MIDI);
+ module_put(coprocessor->owner);
+ }
+ devc->opened = 0;
+}
+
+static int mpu401_out(int dev, unsigned char midi_byte)
+{
+ int timeout;
+ unsigned long flags;
+
+ struct mpu_config *devc;
+
+ devc = &dev_conf[dev];
+
+ /*
+ * Sometimes it takes about 30000 loops before the output becomes ready
+ * (After reset). Normally it takes just about 10 loops.
+ */
+
+ for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--);
+
+ spin_lock_irqsave(&devc->lock,flags);
+ if (!output_ready(devc))
+ {
+ printk(KERN_WARNING "mpu401: Send data timeout\n");
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return 0;
+ }
+ write_data(devc, midi_byte);
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return 1;
+}
+
+static int mpu401_command(int dev, mpu_command_rec * cmd)
+{
+ int i, timeout, ok;
+ int ret = 0;
+ unsigned long flags;
+ struct mpu_config *devc;
+
+ devc = &dev_conf[dev];
+
+ if (devc->uart_mode) /*
+ * Not possible in UART mode
+ */
+ {
+ printk(KERN_WARNING "mpu401: commands not possible in the UART mode\n");
+ return -EINVAL;
+ }
+ /*
+ * Test for input since pending input seems to block the output.
+ */
+ if (input_avail(devc))
+ mpu401_input_loop(devc);
+
+ /*
+ * Sometimes it takes about 50000 loops before the output becomes ready
+ * (After reset). Normally it takes just about 10 loops.
+ */
+
+ timeout = 50000;
+retry:
+ if (timeout-- <= 0)
+ {
+ printk(KERN_WARNING "mpu401: Command (0x%x) timeout\n", (int) cmd->cmd);
+ return -EIO;
+ }
+ spin_lock_irqsave(&devc->lock,flags);
+
+ if (!output_ready(devc))
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ goto retry;
+ }
+ write_command(devc, cmd->cmd);
+
+ ok = 0;
+ for (timeout = 50000; timeout > 0 && !ok; timeout--)
+ {
+ if (input_avail(devc))
+ {
+ if (devc->opened && devc->mode == MODE_SYNTH)
+ {
+ if (mpu_input_scanner(devc, read_data(devc)) == MPU_ACK)
+ ok = 1;
+ }
+ else
+ {
+ /* Device is not currently open. Use simpler method */
+ if (read_data(devc) == MPU_ACK)
+ ok = 1;
+ }
+ }
+ }
+ if (!ok)
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return -EIO;
+ }
+ if (cmd->nr_args)
+ {
+ for (i = 0; i < cmd->nr_args; i++)
+ {
+ for (timeout = 3000; timeout > 0 && !output_ready(devc); timeout--);
+
+ if (!mpu401_out(dev, cmd->data[i]))
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ printk(KERN_WARNING "mpu401: Command (0x%x), parm send failed.\n", (int) cmd->cmd);
+ return -EIO;
+ }
+ }
+ }
+ ret = 0;
+ cmd->data[0] = 0;
+
+ if (cmd->nr_returns)
+ {
+ for (i = 0; i < cmd->nr_returns; i++)
+ {
+ ok = 0;
+ for (timeout = 5000; timeout > 0 && !ok; timeout--)
+ if (input_avail(devc))
+ {
+ cmd->data[i] = read_data(devc);
+ ok = 1;
+ }
+ if (!ok)
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return -EIO;
+ }
+ }
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return ret;
+}
+
+static int mpu_cmd(int dev, int cmd, int data)
+{
+ int ret;
+
+ static mpu_command_rec rec;
+
+ rec.cmd = cmd & 0xff;
+ rec.nr_args = ((cmd & 0xf0) == 0xE0);
+ rec.nr_returns = ((cmd & 0xf0) == 0xA0);
+ rec.data[0] = data & 0xff;
+
+ if ((ret = mpu401_command(dev, &rec)) < 0)
+ return ret;
+ return (unsigned char) rec.data[0];
+}
+
+static int mpu401_prefix_cmd(int dev, unsigned char status)
+{
+ struct mpu_config *devc = &dev_conf[dev];
+
+ if (devc->uart_mode)
+ return 1;
+
+ if (status < 0xf0)
+ {
+ if (mpu_cmd(dev, 0xD0, 0) < 0)
+ return 0;
+ return 1;
+ }
+ switch (status)
+ {
+ case 0xF0:
+ if (mpu_cmd(dev, 0xDF, 0) < 0)
+ return 0;
+ return 1;
+
+ default:
+ return 0;
+ }
+}
+
+static int mpu401_start_read(int dev)
+{
+ return 0;
+}
+
+static int mpu401_end_read(int dev)
+{
+ return 0;
+}
+
+static int mpu401_ioctl(int dev, unsigned cmd, void __user *arg)
+{
+ struct mpu_config *devc;
+ mpu_command_rec rec;
+ int val, ret;
+
+ devc = &dev_conf[dev];
+ switch (cmd)
+ {
+ case SNDCTL_MIDI_MPUMODE:
+ if (!(devc->capabilities & MPU_CAP_INTLG)) { /* No intelligent mode */
+ printk(KERN_WARNING "mpu401: Intelligent mode not supported by the HW\n");
+ return -EINVAL;
+ }
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ set_uart_mode(dev, devc, !val);
+ return 0;
+
+ case SNDCTL_MIDI_MPUCMD:
+ if (copy_from_user(&rec, arg, sizeof(rec)))
+ return -EFAULT;
+ if ((ret = mpu401_command(dev, &rec)) < 0)
+ return ret;
+ if (copy_to_user(arg, &rec, sizeof(rec)))
+ return -EFAULT;
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static void mpu401_kick(int dev)
+{
+}
+
+static int mpu401_buffer_status(int dev)
+{
+ return 0; /*
+ * No data in buffers
+ */
+}
+
+static int mpu_synth_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ int midi_dev;
+ struct mpu_config *devc;
+
+ midi_dev = synth_devs[dev]->midi_dev;
+
+ if (midi_dev < 0 || midi_dev > num_midis || midi_devs[midi_dev] == NULL)
+ return -ENXIO;
+
+ devc = &dev_conf[midi_dev];
+
+ switch (cmd)
+ {
+
+ case SNDCTL_SYNTH_INFO:
+ if (copy_to_user(arg, &mpu_synth_info[midi_dev],
+ sizeof(struct synth_info)))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_SYNTH_MEMAVL:
+ return 0x7fffffff;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int mpu_synth_open(int dev, int mode)
+{
+ int midi_dev, err;
+ struct mpu_config *devc;
+ struct coproc_operations *coprocessor;
+
+ midi_dev = synth_devs[dev]->midi_dev;
+
+ if (midi_dev < 0 || midi_dev > num_midis || midi_devs[midi_dev] == NULL)
+ return -ENXIO;
+
+ devc = &dev_conf[midi_dev];
+
+ /*
+ * Verify that the device is really running.
+ * Some devices (such as Ensoniq SoundScape don't
+ * work before the on board processor (OBP) is initialized
+ * by downloading its microcode.
+ */
+
+ if (!devc->initialized)
+ {
+ if (mpu401_status(devc) == 0xff) /* Bus float */
+ {
+ printk(KERN_ERR "mpu401: Device not initialized properly\n");
+ return -EIO;
+ }
+ reset_mpu401(devc);
+ }
+ if (devc->opened)
+ return -EBUSY;
+ devc->mode = MODE_SYNTH;
+ devc->synthno = dev;
+
+ devc->inputintr = NULL;
+
+ coprocessor = midi_devs[midi_dev]->coproc;
+ if (coprocessor) {
+ if (!try_module_get(coprocessor->owner))
+ return -ENODEV;
+
+ if ((err = coprocessor->open(coprocessor->devc, COPR_MIDI)) < 0)
+ {
+ printk(KERN_WARNING "mpu401: Can't access coprocessor device\n");
+ return err;
+ }
+ }
+ devc->opened = mode;
+ reset_mpu401(devc);
+
+ if (mode & OPEN_READ)
+ {
+ mpu_cmd(midi_dev, 0x8B, 0); /* Enable data in stop mode */
+ mpu_cmd(midi_dev, 0x34, 0); /* Return timing bytes in stop mode */
+ mpu_cmd(midi_dev, 0x87, 0); /* Enable pitch & controller */
+ }
+ return 0;
+}
+
+static void mpu_synth_close(int dev)
+{
+ int midi_dev;
+ struct mpu_config *devc;
+ struct coproc_operations *coprocessor;
+
+ midi_dev = synth_devs[dev]->midi_dev;
+
+ devc = &dev_conf[midi_dev];
+ mpu_cmd(midi_dev, 0x15, 0); /* Stop recording, playback and MIDI */
+ mpu_cmd(midi_dev, 0x8a, 0); /* Disable data in stopped mode */
+
+ devc->inputintr = NULL;
+
+ coprocessor = midi_devs[midi_dev]->coproc;
+ if (coprocessor) {
+ coprocessor->close(coprocessor->devc, COPR_MIDI);
+ module_put(coprocessor->owner);
+ }
+ devc->opened = 0;
+ devc->mode = 0;
+}
+
+#define MIDI_SYNTH_NAME "MPU-401 UART Midi"
+#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT
+#include "midi_synth.h"
+
+static struct synth_operations mpu401_synth_proto =
+{
+ .owner = THIS_MODULE,
+ .id = "MPU401",
+ .info = NULL,
+ .midi_dev = 0,
+ .synth_type = SYNTH_TYPE_MIDI,
+ .synth_subtype = 0,
+ .open = mpu_synth_open,
+ .close = mpu_synth_close,
+ .ioctl = mpu_synth_ioctl,
+ .kill_note = midi_synth_kill_note,
+ .start_note = midi_synth_start_note,
+ .set_instr = midi_synth_set_instr,
+ .reset = midi_synth_reset,
+ .hw_control = midi_synth_hw_control,
+ .load_patch = midi_synth_load_patch,
+ .aftertouch = midi_synth_aftertouch,
+ .controller = midi_synth_controller,
+ .panning = midi_synth_panning,
+ .bender = midi_synth_bender,
+ .setup_voice = midi_synth_setup_voice,
+ .send_sysex = midi_synth_send_sysex
+};
+
+static struct synth_operations *mpu401_synth_operations[MAX_MIDI_DEV];
+
+static struct midi_operations mpu401_midi_proto =
+{
+ .owner = THIS_MODULE,
+ .info = {"MPU-401 Midi", 0, MIDI_CAP_MPU401, SNDCARD_MPU401},
+ .in_info = {0},
+ .open = mpu401_open,
+ .close = mpu401_close,
+ .ioctl = mpu401_ioctl,
+ .outputc = mpu401_out,
+ .start_read = mpu401_start_read,
+ .end_read = mpu401_end_read,
+ .kick = mpu401_kick,
+ .buffer_status = mpu401_buffer_status,
+ .prefix_cmd = mpu401_prefix_cmd
+};
+
+static struct midi_operations mpu401_midi_operations[MAX_MIDI_DEV];
+
+static void mpu401_chk_version(int n, struct mpu_config *devc)
+{
+ int tmp;
+ unsigned long flags;
+
+ devc->version = devc->revision = 0;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ if ((tmp = mpu_cmd(n, 0xAC, 0)) < 0)
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return;
+ }
+ if ((tmp & 0xf0) > 0x20) /* Why it's larger than 2.x ??? */
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return;
+ }
+ devc->version = tmp;
+
+ if ((tmp = mpu_cmd(n, 0xAD, 0)) < 0)
+ {
+ devc->version = 0;
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return;
+ }
+ devc->revision = tmp;
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+int attach_mpu401(struct address_info *hw_config, struct module *owner)
+{
+ unsigned long flags;
+ char revision_char;
+
+ int m, ret;
+ struct mpu_config *devc;
+
+ hw_config->slots[1] = -1;
+ m = sound_alloc_mididev();
+ if (m == -1)
+ {
+ printk(KERN_WARNING "MPU-401: Too many midi devices detected\n");
+ ret = -ENOMEM;
+ goto out_err;
+ }
+ devc = &dev_conf[m];
+ devc->base = hw_config->io_base;
+ devc->osp = hw_config->osp;
+ devc->irq = hw_config->irq;
+ devc->opened = 0;
+ devc->uart_mode = 0;
+ devc->initialized = 0;
+ devc->version = 0;
+ devc->revision = 0;
+ devc->capabilities = 0;
+ devc->timer_flag = 0;
+ devc->m_busy = 0;
+ devc->m_state = ST_INIT;
+ devc->shared_irq = hw_config->always_detect;
+ devc->irq = hw_config->irq;
+ spin_lock_init(&devc->lock);
+
+ if (devc->irq < 0)
+ {
+ devc->irq *= -1;
+ devc->shared_irq = 1;
+ }
+
+ if (!hw_config->always_detect)
+ {
+ /* Verify the hardware again */
+ if (!reset_mpu401(devc))
+ {
+ printk(KERN_WARNING "mpu401: Device didn't respond\n");
+ ret = -ENODEV;
+ goto out_mididev;
+ }
+ if (!devc->shared_irq)
+ {
+ if (request_irq(devc->irq, mpuintr, 0, "mpu401",
+ hw_config) < 0)
+ {
+ printk(KERN_WARNING "mpu401: Failed to allocate IRQ%d\n", devc->irq);
+ ret = -ENOMEM;
+ goto out_mididev;
+ }
+ }
+ spin_lock_irqsave(&devc->lock,flags);
+ mpu401_chk_version(m, devc);
+ if (devc->version == 0)
+ mpu401_chk_version(m, devc);
+ spin_unlock_irqrestore(&devc->lock, flags);
+ }
+
+ if (devc->version != 0)
+ if (mpu_cmd(m, 0xC5, 0) >= 0) /* Set timebase OK */
+ if (mpu_cmd(m, 0xE0, 120) >= 0) /* Set tempo OK */
+ devc->capabilities |= MPU_CAP_INTLG; /* Supports intelligent mode */
+
+
+ mpu401_synth_operations[m] = kmalloc(sizeof(struct synth_operations), GFP_KERNEL);
+
+ if (mpu401_synth_operations[m] == NULL)
+ {
+ printk(KERN_ERR "mpu401: Can't allocate memory\n");
+ ret = -ENOMEM;
+ goto out_irq;
+ }
+ if (!(devc->capabilities & MPU_CAP_INTLG)) /* No intelligent mode */
+ {
+ memcpy((char *) mpu401_synth_operations[m],
+ (char *) &std_midi_synth,
+ sizeof(struct synth_operations));
+ }
+ else
+ {
+ memcpy((char *) mpu401_synth_operations[m],
+ (char *) &mpu401_synth_proto,
+ sizeof(struct synth_operations));
+ }
+ if (owner)
+ mpu401_synth_operations[m]->owner = owner;
+
+ memcpy((char *) &mpu401_midi_operations[m],
+ (char *) &mpu401_midi_proto,
+ sizeof(struct midi_operations));
+
+ mpu401_midi_operations[m].converter = mpu401_synth_operations[m];
+
+ memcpy((char *) &mpu_synth_info[m],
+ (char *) &mpu_synth_info_proto,
+ sizeof(struct synth_info));
+
+ n_mpu_devs++;
+
+ if (devc->version == 0x20 && devc->revision >= 0x07) /* MusicQuest interface */
+ {
+ int ports = (devc->revision & 0x08) ? 32 : 16;
+
+ devc->capabilities |= MPU_CAP_SYNC | MPU_CAP_SMPTE |
+ MPU_CAP_CLS | MPU_CAP_2PORT;
+
+ revision_char = (devc->revision == 0x7f) ? 'M' : ' ';
+ sprintf(mpu_synth_info[m].name, "MQX-%d%c MIDI Interface #%d",
+ ports,
+ revision_char,
+ n_mpu_devs);
+ }
+ else
+ {
+ revision_char = devc->revision ? devc->revision + '@' : ' ';
+ if ((int) devc->revision > ('Z' - '@'))
+ revision_char = '+';
+
+ devc->capabilities |= MPU_CAP_SYNC | MPU_CAP_FSK;
+
+ if (hw_config->name)
+ sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name);
+ else
+ sprintf(mpu_synth_info[m].name,
+ "MPU-401 %d.%d%c Midi interface #%d",
+ (int) (devc->version & 0xf0) >> 4,
+ devc->version & 0x0f,
+ revision_char,
+ n_mpu_devs);
+ }
+
+ strcpy(mpu401_midi_operations[m].info.name,
+ mpu_synth_info[m].name);
+
+ conf_printf(mpu_synth_info[m].name, hw_config);
+
+ mpu401_synth_operations[m]->midi_dev = devc->devno = m;
+ mpu401_synth_operations[devc->devno]->info = &mpu_synth_info[devc->devno];
+
+ if (devc->capabilities & MPU_CAP_INTLG) /* Intelligent mode */
+ hw_config->slots[2] = mpu_timer_init(m);
+
+ midi_devs[m] = &mpu401_midi_operations[devc->devno];
+
+ if (owner)
+ midi_devs[m]->owner = owner;
+
+ hw_config->slots[1] = m;
+ sequencer_init();
+
+ return 0;
+
+out_irq:
+ free_irq(devc->irq, hw_config);
+out_mididev:
+ sound_unload_mididev(m);
+out_err:
+ release_region(hw_config->io_base, 2);
+ return ret;
+}
+
+static int reset_mpu401(struct mpu_config *devc)
+{
+ unsigned long flags;
+ int ok, timeout, n;
+ int timeout_limit;
+
+ /*
+ * Send the RESET command. Try again if no success at the first time.
+ * (If the device is in the UART mode, it will not ack the reset cmd).
+ */
+
+ ok = 0;
+
+ timeout_limit = devc->initialized ? 30000 : 100000;
+ devc->initialized = 1;
+
+ for (n = 0; n < 2 && !ok; n++)
+ {
+ for (timeout = timeout_limit; timeout > 0 && !ok; timeout--)
+ ok = output_ready(devc);
+
+ write_command(devc, MPU_RESET); /*
+ * Send MPU-401 RESET Command
+ */
+
+ /*
+ * Wait at least 25 msec. This method is not accurate so let's make the
+ * loop bit longer. Cannot sleep since this is called during boot.
+ */
+
+ for (timeout = timeout_limit * 2; timeout > 0 && !ok; timeout--)
+ {
+ spin_lock_irqsave(&devc->lock,flags);
+ if (input_avail(devc))
+ if (read_data(devc) == MPU_ACK)
+ ok = 1;
+ spin_unlock_irqrestore(&devc->lock,flags);
+ }
+
+ }
+
+ devc->m_state = ST_INIT;
+ devc->m_ptr = 0;
+ devc->m_left = 0;
+ devc->last_status = 0;
+ devc->uart_mode = 0;
+
+ return ok;
+}
+
+static void set_uart_mode(int dev, struct mpu_config *devc, int arg)
+{
+ if (!arg && (devc->capabilities & MPU_CAP_INTLG))
+ return;
+ if ((devc->uart_mode == 0) == (arg == 0))
+ return; /* Already set */
+ reset_mpu401(devc); /* This exits the uart mode */
+
+ if (arg)
+ {
+ if (mpu_cmd(dev, UART_MODE_ON, 0) < 0)
+ {
+ printk(KERN_ERR "mpu401: Can't enter UART mode\n");
+ devc->uart_mode = 0;
+ return;
+ }
+ }
+ devc->uart_mode = arg;
+
+}
+
+int probe_mpu401(struct address_info *hw_config, struct resource *ports)
+{
+ int ok = 0;
+ struct mpu_config tmp_devc;
+
+ tmp_devc.base = hw_config->io_base;
+ tmp_devc.irq = hw_config->irq;
+ tmp_devc.initialized = 0;
+ tmp_devc.opened = 0;
+ tmp_devc.osp = hw_config->osp;
+
+ if (hw_config->always_detect)
+ return 1;
+
+ if (inb(hw_config->io_base + 1) == 0xff)
+ {
+ DDB(printk("MPU401: Port %x looks dead.\n", hw_config->io_base));
+ return 0; /* Just bus float? */
+ }
+ ok = reset_mpu401(&tmp_devc);
+
+ if (!ok)
+ {
+ DDB(printk("MPU401: Reset failed on port %x\n", hw_config->io_base));
+ }
+ return ok;
+}
+
+void unload_mpu401(struct address_info *hw_config)
+{
+ void *p;
+ int n=hw_config->slots[1];
+
+ if (n != -1) {
+ release_region(hw_config->io_base, 2);
+ if (hw_config->always_detect == 0 && hw_config->irq > 0)
+ free_irq(hw_config->irq, hw_config);
+ p=mpu401_synth_operations[n];
+ sound_unload_mididev(n);
+ sound_unload_timerdev(hw_config->slots[2]);
+ kfree(p);
+ }
+}
+
+/*****************************************************
+ * Timer stuff
+ ****************************************************/
+
+static volatile int timer_initialized = 0, timer_open = 0, tmr_running = 0;
+static volatile int curr_tempo, curr_timebase, hw_timebase;
+static int max_timebase = 8; /* 8*24=192 ppqn */
+static volatile unsigned long next_event_time;
+static volatile unsigned long curr_ticks, curr_clocks;
+static unsigned long prev_event_time;
+static int metronome_mode;
+
+static unsigned long clocks2ticks(unsigned long clocks)
+{
+ /*
+ * The MPU-401 supports just a limited set of possible timebase values.
+ * Since the applications require more choices, the driver has to
+ * program the HW to do its best and to convert between the HW and
+ * actual timebases.
+ */
+ return ((clocks * curr_timebase) + (hw_timebase / 2)) / hw_timebase;
+}
+
+static void set_timebase(int midi_dev, int val)
+{
+ int hw_val;
+
+ if (val < 48)
+ val = 48;
+ if (val > 1000)
+ val = 1000;
+
+ hw_val = val;
+ hw_val = (hw_val + 12) / 24;
+ if (hw_val > max_timebase)
+ hw_val = max_timebase;
+
+ if (mpu_cmd(midi_dev, 0xC0 | (hw_val & 0x0f), 0) < 0)
+ {
+ printk(KERN_WARNING "mpu401: Can't set HW timebase to %d\n", hw_val * 24);
+ return;
+ }
+ hw_timebase = hw_val * 24;
+ curr_timebase = val;
+
+}
+
+static void tmr_reset(struct mpu_config *devc)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ next_event_time = (unsigned long) -1;
+ prev_event_time = 0;
+ curr_ticks = curr_clocks = 0;
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void set_timer_mode(int midi_dev)
+{
+ if (timer_mode & TMR_MODE_CLS)
+ mpu_cmd(midi_dev, 0x3c, 0); /* Use CLS sync */
+ else if (timer_mode & TMR_MODE_SMPTE)
+ mpu_cmd(midi_dev, 0x3d, 0); /* Use SMPTE sync */
+
+ if (timer_mode & TMR_INTERNAL)
+ {
+ mpu_cmd(midi_dev, 0x80, 0); /* Use MIDI sync */
+ }
+ else
+ {
+ if (timer_mode & (TMR_MODE_MIDI | TMR_MODE_CLS))
+ {
+ mpu_cmd(midi_dev, 0x82, 0); /* Use MIDI sync */
+ mpu_cmd(midi_dev, 0x91, 0); /* Enable ext MIDI ctrl */
+ }
+ else if (timer_mode & TMR_MODE_FSK)
+ mpu_cmd(midi_dev, 0x81, 0); /* Use FSK sync */
+ }
+}
+
+static void stop_metronome(int midi_dev)
+{
+ mpu_cmd(midi_dev, 0x84, 0); /* Disable metronome */
+}
+
+static void setup_metronome(int midi_dev)
+{
+ int numerator, denominator;
+ int clks_per_click, num_32nds_per_beat;
+ int beats_per_measure;
+
+ numerator = ((unsigned) metronome_mode >> 24) & 0xff;
+ denominator = ((unsigned) metronome_mode >> 16) & 0xff;
+ clks_per_click = ((unsigned) metronome_mode >> 8) & 0xff;
+ num_32nds_per_beat = (unsigned) metronome_mode & 0xff;
+ beats_per_measure = (numerator * 4) >> denominator;
+
+ if (!metronome_mode)
+ mpu_cmd(midi_dev, 0x84, 0); /* Disable metronome */
+ else
+ {
+ mpu_cmd(midi_dev, 0xE4, clks_per_click);
+ mpu_cmd(midi_dev, 0xE6, beats_per_measure);
+ mpu_cmd(midi_dev, 0x83, 0); /* Enable metronome without accents */
+ }
+}
+
+static int mpu_start_timer(int midi_dev)
+{
+ struct mpu_config *devc= &dev_conf[midi_dev];
+
+ tmr_reset(devc);
+ set_timer_mode(midi_dev);
+
+ if (tmr_running)
+ return TIMER_NOT_ARMED; /* Already running */
+
+ if (timer_mode & TMR_INTERNAL)
+ {
+ mpu_cmd(midi_dev, 0x02, 0); /* Send MIDI start */
+ tmr_running = 1;
+ return TIMER_NOT_ARMED;
+ }
+ else
+ {
+ mpu_cmd(midi_dev, 0x35, 0); /* Enable mode messages to PC */
+ mpu_cmd(midi_dev, 0x38, 0); /* Enable sys common messages to PC */
+ mpu_cmd(midi_dev, 0x39, 0); /* Enable real time messages to PC */
+ mpu_cmd(midi_dev, 0x97, 0); /* Enable system exclusive messages to PC */
+ }
+ return TIMER_ARMED;
+}
+
+static int mpu_timer_open(int dev, int mode)
+{
+ int midi_dev = sound_timer_devs[dev]->devlink;
+ struct mpu_config *devc= &dev_conf[midi_dev];
+
+ if (timer_open)
+ return -EBUSY;
+
+ tmr_reset(devc);
+ curr_tempo = 50;
+ mpu_cmd(midi_dev, 0xE0, 50);
+ curr_timebase = hw_timebase = 120;
+ set_timebase(midi_dev, 120);
+ timer_open = 1;
+ metronome_mode = 0;
+ set_timer_mode(midi_dev);
+
+ mpu_cmd(midi_dev, 0xe7, 0x04); /* Send all clocks to host */
+ mpu_cmd(midi_dev, 0x95, 0); /* Enable clock to host */
+
+ return 0;
+}
+
+static void mpu_timer_close(int dev)
+{
+ int midi_dev = sound_timer_devs[dev]->devlink;
+
+ timer_open = tmr_running = 0;
+ mpu_cmd(midi_dev, 0x15, 0); /* Stop all */
+ mpu_cmd(midi_dev, 0x94, 0); /* Disable clock to host */
+ mpu_cmd(midi_dev, 0x8c, 0); /* Disable measure end messages to host */
+ stop_metronome(midi_dev);
+}
+
+static int mpu_timer_event(int dev, unsigned char *event)
+{
+ unsigned char command = event[1];
+ unsigned long parm = *(unsigned int *) &event[4];
+ int midi_dev = sound_timer_devs[dev]->devlink;
+
+ switch (command)
+ {
+ case TMR_WAIT_REL:
+ parm += prev_event_time;
+ case TMR_WAIT_ABS:
+ if (parm > 0)
+ {
+ long time;
+
+ if (parm <= curr_ticks) /* It's the time */
+ return TIMER_NOT_ARMED;
+ time = parm;
+ next_event_time = prev_event_time = time;
+
+ return TIMER_ARMED;
+ }
+ break;
+
+ case TMR_START:
+ if (tmr_running)
+ break;
+ return mpu_start_timer(midi_dev);
+
+ case TMR_STOP:
+ mpu_cmd(midi_dev, 0x01, 0); /* Send MIDI stop */
+ stop_metronome(midi_dev);
+ tmr_running = 0;
+ break;
+
+ case TMR_CONTINUE:
+ if (tmr_running)
+ break;
+ mpu_cmd(midi_dev, 0x03, 0); /* Send MIDI continue */
+ setup_metronome(midi_dev);
+ tmr_running = 1;
+ break;
+
+ case TMR_TEMPO:
+ if (parm)
+ {
+ if (parm < 8)
+ parm = 8;
+ if (parm > 250)
+ parm = 250;
+ if (mpu_cmd(midi_dev, 0xE0, parm) < 0)
+ printk(KERN_WARNING "mpu401: Can't set tempo to %d\n", (int) parm);
+ curr_tempo = parm;
+ }
+ break;
+
+ case TMR_ECHO:
+ seq_copy_to_input(event, 8);
+ break;
+
+ case TMR_TIMESIG:
+ if (metronome_mode) /* Metronome enabled */
+ {
+ metronome_mode = parm;
+ setup_metronome(midi_dev);
+ }
+ break;
+
+ default:;
+ }
+ return TIMER_NOT_ARMED;
+}
+
+static unsigned long mpu_timer_get_time(int dev)
+{
+ if (!timer_open)
+ return 0;
+
+ return curr_ticks;
+}
+
+static int mpu_timer_ioctl(int dev, unsigned int command, void __user *arg)
+{
+ int midi_dev = sound_timer_devs[dev]->devlink;
+ int __user *p = (int __user *)arg;
+
+ switch (command)
+ {
+ case SNDCTL_TMR_SOURCE:
+ {
+ int parm;
+
+ if (get_user(parm, p))
+ return -EFAULT;
+ parm &= timer_caps;
+
+ if (parm != 0)
+ {
+ timer_mode = parm;
+
+ if (timer_mode & TMR_MODE_CLS)
+ mpu_cmd(midi_dev, 0x3c, 0); /* Use CLS sync */
+ else if (timer_mode & TMR_MODE_SMPTE)
+ mpu_cmd(midi_dev, 0x3d, 0); /* Use SMPTE sync */
+ }
+ if (put_user(timer_mode, p))
+ return -EFAULT;
+ return timer_mode;
+ }
+ break;
+
+ case SNDCTL_TMR_START:
+ mpu_start_timer(midi_dev);
+ return 0;
+
+ case SNDCTL_TMR_STOP:
+ tmr_running = 0;
+ mpu_cmd(midi_dev, 0x01, 0); /* Send MIDI stop */
+ stop_metronome(midi_dev);
+ return 0;
+
+ case SNDCTL_TMR_CONTINUE:
+ if (tmr_running)
+ return 0;
+ tmr_running = 1;
+ mpu_cmd(midi_dev, 0x03, 0); /* Send MIDI continue */
+ return 0;
+
+ case SNDCTL_TMR_TIMEBASE:
+ {
+ int val;
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val)
+ set_timebase(midi_dev, val);
+ if (put_user(curr_timebase, p))
+ return -EFAULT;
+ return curr_timebase;
+ }
+ break;
+
+ case SNDCTL_TMR_TEMPO:
+ {
+ int val;
+ int ret;
+
+ if (get_user(val, p))
+ return -EFAULT;
+
+ if (val)
+ {
+ if (val < 8)
+ val = 8;
+ if (val > 250)
+ val = 250;
+ if ((ret = mpu_cmd(midi_dev, 0xE0, val)) < 0)
+ {
+ printk(KERN_WARNING "mpu401: Can't set tempo to %d\n", (int) val);
+ return ret;
+ }
+ curr_tempo = val;
+ }
+ if (put_user(curr_tempo, p))
+ return -EFAULT;
+ return curr_tempo;
+ }
+ break;
+
+ case SNDCTL_SEQ_CTRLRATE:
+ {
+ int val;
+ if (get_user(val, p))
+ return -EFAULT;
+
+ if (val != 0) /* Can't change */
+ return -EINVAL;
+ val = ((curr_tempo * curr_timebase) + 30)/60;
+ if (put_user(val, p))
+ return -EFAULT;
+ return val;
+ }
+ break;
+
+ case SNDCTL_SEQ_GETTIME:
+ if (put_user(curr_ticks, p))
+ return -EFAULT;
+ return curr_ticks;
+
+ case SNDCTL_TMR_METRONOME:
+ if (get_user(metronome_mode, p))
+ return -EFAULT;
+ setup_metronome(midi_dev);
+ return 0;
+
+ default:;
+ }
+ return -EINVAL;
+}
+
+static void mpu_timer_arm(int dev, long time)
+{
+ if (time < 0)
+ time = curr_ticks + 1;
+ else if (time <= curr_ticks) /* It's the time */
+ return;
+ next_event_time = prev_event_time = time;
+ return;
+}
+
+static struct sound_timer_operations mpu_timer =
+{
+ .owner = THIS_MODULE,
+ .info = {"MPU-401 Timer", 0},
+ .priority = 10, /* Priority */
+ .devlink = 0, /* Local device link */
+ .open = mpu_timer_open,
+ .close = mpu_timer_close,
+ .event = mpu_timer_event,
+ .get_time = mpu_timer_get_time,
+ .ioctl = mpu_timer_ioctl,
+ .arm_timer = mpu_timer_arm
+};
+
+static void mpu_timer_interrupt(void)
+{
+ if (!timer_open)
+ return;
+
+ if (!tmr_running)
+ return;
+
+ curr_clocks++;
+ curr_ticks = clocks2ticks(curr_clocks);
+
+ if (curr_ticks >= next_event_time)
+ {
+ next_event_time = (unsigned long) -1;
+ sequencer_timer(0);
+ }
+}
+
+static void timer_ext_event(struct mpu_config *devc, int event, int parm)
+{
+ int midi_dev = devc->devno;
+
+ if (!devc->timer_flag)
+ return;
+
+ switch (event)
+ {
+ case TMR_CLOCK:
+ printk("<MIDI clk>");
+ break;
+
+ case TMR_START:
+ printk("Ext MIDI start\n");
+ if (!tmr_running)
+ {
+ if (timer_mode & TMR_EXTERNAL)
+ {
+ tmr_running = 1;
+ setup_metronome(midi_dev);
+ next_event_time = 0;
+ STORE(SEQ_START_TIMER());
+ }
+ }
+ break;
+
+ case TMR_STOP:
+ printk("Ext MIDI stop\n");
+ if (timer_mode & TMR_EXTERNAL)
+ {
+ tmr_running = 0;
+ stop_metronome(midi_dev);
+ STORE(SEQ_STOP_TIMER());
+ }
+ break;
+
+ case TMR_CONTINUE:
+ printk("Ext MIDI continue\n");
+ if (timer_mode & TMR_EXTERNAL)
+ {
+ tmr_running = 1;
+ setup_metronome(midi_dev);
+ STORE(SEQ_CONTINUE_TIMER());
+ }
+ break;
+
+ case TMR_SPP:
+ printk("Songpos: %d\n", parm);
+ if (timer_mode & TMR_EXTERNAL)
+ {
+ STORE(SEQ_SONGPOS(parm));
+ }
+ break;
+ }
+}
+
+static int mpu_timer_init(int midi_dev)
+{
+ struct mpu_config *devc;
+ int n;
+
+ devc = &dev_conf[midi_dev];
+
+ if (timer_initialized)
+ return -1; /* There is already a similar timer */
+
+ timer_initialized = 1;
+
+ mpu_timer.devlink = midi_dev;
+ dev_conf[midi_dev].timer_flag = 1;
+
+ n = sound_alloc_timerdev();
+ if (n == -1)
+ n = 0;
+ sound_timer_devs[n] = &mpu_timer;
+
+ if (devc->version < 0x20) /* Original MPU-401 */
+ timer_caps = TMR_INTERNAL | TMR_EXTERNAL | TMR_MODE_FSK | TMR_MODE_MIDI;
+ else
+ {
+ /*
+ * The version number 2.0 is used (at least) by the
+ * MusicQuest cards and the Roland Super-MPU.
+ *
+ * MusicQuest has given a special meaning to the bits of the
+ * revision number. The Super-MPU returns 0.
+ */
+
+ if (devc->revision)
+ timer_caps |= TMR_EXTERNAL | TMR_MODE_MIDI;
+
+ if (devc->revision & 0x02)
+ timer_caps |= TMR_MODE_CLS;
+
+
+ if (devc->revision & 0x40)
+ max_timebase = 10; /* Has the 216 and 240 ppqn modes */
+ }
+
+ timer_mode = (TMR_INTERNAL | TMR_MODE_MIDI) & timer_caps;
+ return n;
+
+}
+
+EXPORT_SYMBOL(probe_mpu401);
+EXPORT_SYMBOL(attach_mpu401);
+EXPORT_SYMBOL(unload_mpu401);
+
+static struct address_info cfg;
+
+static int io = -1;
+static int irq = -1;
+
+module_param(irq, int, 0);
+module_param(io, int, 0);
+
+static int __init init_mpu401(void)
+{
+ int ret;
+ /* Can be loaded either for module use or to provide functions
+ to others */
+ if (io != -1 && irq != -1) {
+ struct resource *ports;
+ cfg.irq = irq;
+ cfg.io_base = io;
+ ports = request_region(io, 2, "mpu401");
+ if (!ports)
+ return -EBUSY;
+ if (probe_mpu401(&cfg, ports) == 0) {
+ release_region(io, 2);
+ return -ENODEV;
+ }
+ if ((ret = attach_mpu401(&cfg, THIS_MODULE)))
+ return ret;
+ }
+
+ return 0;
+}
+
+static void __exit cleanup_mpu401(void)
+{
+ if (io != -1 && irq != -1) {
+ /* Check for use by, for example, sscape driver */
+ unload_mpu401(&cfg);
+ }
+}
+
+module_init(init_mpu401);
+module_exit(cleanup_mpu401);
+
+#ifndef MODULE
+static int __init setup_mpu401(char *str)
+{
+ /* io, irq */
+ int ints[3];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+
+ return 1;
+}
+
+__setup("mpu401=", setup_mpu401);
+#endif
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/mpu401.h b/sound/oss/mpu401.h
new file mode 100644
index 0000000..0ad1e9e
--- /dev/null
+++ b/sound/oss/mpu401.h
@@ -0,0 +1,11 @@
+
+/* From uart401.c */
+int probe_uart401 (struct address_info *hw_config, struct module *owner);
+void unload_uart401 (struct address_info *hw_config);
+
+irqreturn_t uart401intr (int irq, void *dev_id);
+
+/* From mpu401.c */
+int probe_mpu401(struct address_info *hw_config, struct resource *ports);
+int attach_mpu401(struct address_info * hw_config, struct module *owner);
+void unload_mpu401(struct address_info *hw_info);
diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c
new file mode 100644
index 0000000..e4282d9
--- /dev/null
+++ b/sound/oss/msnd.c
@@ -0,0 +1,414 @@
+/*********************************************************************
+ *
+ * msnd.c - Driver Base
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/types.h>
+#include <linux/delay.h>
+#include <linux/mm.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+
+#include <asm/io.h>
+#include <asm/uaccess.h>
+#include <linux/spinlock.h>
+#include <asm/irq.h>
+#include "msnd.h"
+
+#define LOGNAME "msnd"
+
+#define MSND_MAX_DEVS 4
+
+static multisound_dev_t *devs[MSND_MAX_DEVS];
+static int num_devs;
+
+int msnd_register(multisound_dev_t *dev)
+{
+ int i;
+
+ for (i = 0; i < MSND_MAX_DEVS; ++i)
+ if (devs[i] == NULL)
+ break;
+
+ if (i == MSND_MAX_DEVS)
+ return -ENOMEM;
+
+ devs[i] = dev;
+ ++num_devs;
+ return 0;
+}
+
+void msnd_unregister(multisound_dev_t *dev)
+{
+ int i;
+
+ for (i = 0; i < MSND_MAX_DEVS; ++i)
+ if (devs[i] == dev)
+ break;
+
+ if (i == MSND_MAX_DEVS) {
+ printk(KERN_WARNING LOGNAME ": Unregistering unknown device\n");
+ return;
+ }
+
+ devs[i] = NULL;
+ --num_devs;
+}
+
+void msnd_init_queue(void __iomem *base, int start, int size)
+{
+ writew(PCTODSP_BASED(start), base + JQS_wStart);
+ writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize);
+ writew(0, base + JQS_wHead);
+ writew(0, base + JQS_wTail);
+}
+
+void msnd_fifo_init(msnd_fifo *f)
+{
+ f->data = NULL;
+}
+
+void msnd_fifo_free(msnd_fifo *f)
+{
+ vfree(f->data);
+ f->data = NULL;
+}
+
+int msnd_fifo_alloc(msnd_fifo *f, size_t n)
+{
+ msnd_fifo_free(f);
+ f->data = (char *)vmalloc(n);
+ f->n = n;
+ f->tail = 0;
+ f->head = 0;
+ f->len = 0;
+
+ if (!f->data)
+ return -ENOMEM;
+
+ return 0;
+}
+
+void msnd_fifo_make_empty(msnd_fifo *f)
+{
+ f->len = f->tail = f->head = 0;
+}
+
+int msnd_fifo_write_io(msnd_fifo *f, char __iomem *buf, size_t len)
+{
+ int count = 0;
+
+ while ((count < len) && (f->len != f->n)) {
+
+ int nwritten;
+
+ if (f->head <= f->tail) {
+ nwritten = len - count;
+ if (nwritten > f->n - f->tail)
+ nwritten = f->n - f->tail;
+ }
+ else {
+ nwritten = f->head - f->tail;
+ if (nwritten > len - count)
+ nwritten = len - count;
+ }
+
+ memcpy_fromio(f->data + f->tail, buf, nwritten);
+
+ count += nwritten;
+ buf += nwritten;
+ f->len += nwritten;
+ f->tail += nwritten;
+ f->tail %= f->n;
+ }
+
+ return count;
+}
+
+int msnd_fifo_write(msnd_fifo *f, const char *buf, size_t len)
+{
+ int count = 0;
+
+ while ((count < len) && (f->len != f->n)) {
+
+ int nwritten;
+
+ if (f->head <= f->tail) {
+ nwritten = len - count;
+ if (nwritten > f->n - f->tail)
+ nwritten = f->n - f->tail;
+ }
+ else {
+ nwritten = f->head - f->tail;
+ if (nwritten > len - count)
+ nwritten = len - count;
+ }
+
+ memcpy(f->data + f->tail, buf, nwritten);
+
+ count += nwritten;
+ buf += nwritten;
+ f->len += nwritten;
+ f->tail += nwritten;
+ f->tail %= f->n;
+ }
+
+ return count;
+}
+
+int msnd_fifo_read_io(msnd_fifo *f, char __iomem *buf, size_t len)
+{
+ int count = 0;
+
+ while ((count < len) && (f->len > 0)) {
+
+ int nread;
+
+ if (f->tail <= f->head) {
+ nread = len - count;
+ if (nread > f->n - f->head)
+ nread = f->n - f->head;
+ }
+ else {
+ nread = f->tail - f->head;
+ if (nread > len - count)
+ nread = len - count;
+ }
+
+ memcpy_toio(buf, f->data + f->head, nread);
+
+ count += nread;
+ buf += nread;
+ f->len -= nread;
+ f->head += nread;
+ f->head %= f->n;
+ }
+
+ return count;
+}
+
+int msnd_fifo_read(msnd_fifo *f, char *buf, size_t len)
+{
+ int count = 0;
+
+ while ((count < len) && (f->len > 0)) {
+
+ int nread;
+
+ if (f->tail <= f->head) {
+ nread = len - count;
+ if (nread > f->n - f->head)
+ nread = f->n - f->head;
+ }
+ else {
+ nread = f->tail - f->head;
+ if (nread > len - count)
+ nread = len - count;
+ }
+
+ memcpy(buf, f->data + f->head, nread);
+
+ count += nread;
+ buf += nread;
+ f->len -= nread;
+ f->head += nread;
+ f->head %= f->n;
+ }
+
+ return count;
+}
+
+static int msnd_wait_TXDE(multisound_dev_t *dev)
+{
+ register unsigned int io = dev->io;
+ register int timeout = 1000;
+
+ while(timeout-- > 0)
+ if (msnd_inb(io + HP_ISR) & HPISR_TXDE)
+ return 0;
+
+ return -EIO;
+}
+
+static int msnd_wait_HC0(multisound_dev_t *dev)
+{
+ register unsigned int io = dev->io;
+ register int timeout = 1000;
+
+ while(timeout-- > 0)
+ if (!(msnd_inb(io + HP_CVR) & HPCVR_HC))
+ return 0;
+
+ return -EIO;
+}
+
+int msnd_send_dsp_cmd(multisound_dev_t *dev, BYTE cmd)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&dev->lock, flags);
+ if (msnd_wait_HC0(dev) == 0) {
+ msnd_outb(cmd, dev->io + HP_CVR);
+ spin_unlock_irqrestore(&dev->lock, flags);
+ return 0;
+ }
+ spin_unlock_irqrestore(&dev->lock, flags);
+
+ printk(KERN_DEBUG LOGNAME ": Send DSP command timeout\n");
+
+ return -EIO;
+}
+
+int msnd_send_word(multisound_dev_t *dev, unsigned char high,
+ unsigned char mid, unsigned char low)
+{
+ register unsigned int io = dev->io;
+
+ if (msnd_wait_TXDE(dev) == 0) {
+ msnd_outb(high, io + HP_TXH);
+ msnd_outb(mid, io + HP_TXM);
+ msnd_outb(low, io + HP_TXL);
+ return 0;
+ }
+
+ printk(KERN_DEBUG LOGNAME ": Send host word timeout\n");
+
+ return -EIO;
+}
+
+int msnd_upload_host(multisound_dev_t *dev, char *bin, int len)
+{
+ int i;
+
+ if (len % 3 != 0) {
+ printk(KERN_WARNING LOGNAME ": Upload host data not multiple of 3!\n");
+ return -EINVAL;
+ }
+
+ for (i = 0; i < len; i += 3)
+ if (msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2]) != 0)
+ return -EIO;
+
+ msnd_inb(dev->io + HP_RXL);
+ msnd_inb(dev->io + HP_CVR);
+
+ return 0;
+}
+
+int msnd_enable_irq(multisound_dev_t *dev)
+{
+ unsigned long flags;
+
+ if (dev->irq_ref++)
+ return 0;
+
+ printk(KERN_DEBUG LOGNAME ": Enabling IRQ\n");
+
+ spin_lock_irqsave(&dev->lock, flags);
+ if (msnd_wait_TXDE(dev) == 0) {
+ msnd_outb(msnd_inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR);
+ if (dev->type == msndClassic)
+ msnd_outb(dev->irqid, dev->io + HP_IRQM);
+ msnd_outb(msnd_inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR);
+ msnd_outb(msnd_inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR);
+ enable_irq(dev->irq);
+ msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, dev->dspq_buff_size);
+ spin_unlock_irqrestore(&dev->lock, flags);
+ return 0;
+ }
+ spin_unlock_irqrestore(&dev->lock, flags);
+
+ printk(KERN_DEBUG LOGNAME ": Enable IRQ failed\n");
+
+ return -EIO;
+}
+
+int msnd_disable_irq(multisound_dev_t *dev)
+{
+ unsigned long flags;
+
+ if (--dev->irq_ref > 0)
+ return 0;
+
+ if (dev->irq_ref < 0)
+ printk(KERN_DEBUG LOGNAME ": IRQ ref count is %d\n", dev->irq_ref);
+
+ printk(KERN_DEBUG LOGNAME ": Disabling IRQ\n");
+
+ spin_lock_irqsave(&dev->lock, flags);
+ if (msnd_wait_TXDE(dev) == 0) {
+ msnd_outb(msnd_inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR);
+ if (dev->type == msndClassic)
+ msnd_outb(HPIRQ_NONE, dev->io + HP_IRQM);
+ disable_irq(dev->irq);
+ spin_unlock_irqrestore(&dev->lock, flags);
+ return 0;
+ }
+ spin_unlock_irqrestore(&dev->lock, flags);
+
+ printk(KERN_DEBUG LOGNAME ": Disable IRQ failed\n");
+
+ return -EIO;
+}
+
+#ifndef LINUX20
+EXPORT_SYMBOL(msnd_register);
+EXPORT_SYMBOL(msnd_unregister);
+
+EXPORT_SYMBOL(msnd_init_queue);
+
+EXPORT_SYMBOL(msnd_fifo_init);
+EXPORT_SYMBOL(msnd_fifo_free);
+EXPORT_SYMBOL(msnd_fifo_alloc);
+EXPORT_SYMBOL(msnd_fifo_make_empty);
+EXPORT_SYMBOL(msnd_fifo_write_io);
+EXPORT_SYMBOL(msnd_fifo_read_io);
+EXPORT_SYMBOL(msnd_fifo_write);
+EXPORT_SYMBOL(msnd_fifo_read);
+
+EXPORT_SYMBOL(msnd_send_dsp_cmd);
+EXPORT_SYMBOL(msnd_send_word);
+EXPORT_SYMBOL(msnd_upload_host);
+
+EXPORT_SYMBOL(msnd_enable_irq);
+EXPORT_SYMBOL(msnd_disable_irq);
+#endif
+
+#ifdef MODULE
+MODULE_AUTHOR ("Andrew Veliath <andrewtv@usa.net>");
+MODULE_DESCRIPTION ("Turtle Beach MultiSound Driver Base");
+MODULE_LICENSE("GPL");
+
+
+int init_module(void)
+{
+ return 0;
+}
+
+void cleanup_module(void)
+{
+}
+#endif
diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h
new file mode 100644
index 0000000..c8be47e
--- /dev/null
+++ b/sound/oss/msnd.h
@@ -0,0 +1,278 @@
+/*********************************************************************
+ *
+ * msnd.h
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Some parts of this header file were derived from the Turtle Beach
+ * MultiSound Driver Development Kit.
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ * Copyright (C) 1993 Turtle Beach Systems, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+#ifndef __MSND_H
+#define __MSND_H
+
+#define VERSION "0.8.3.1"
+
+#define DEFSAMPLERATE DSP_DEFAULT_SPEED
+#define DEFSAMPLESIZE AFMT_U8
+#define DEFCHANNELS 1
+
+#define DEFFIFOSIZE 128
+
+#define SNDCARD_MSND 38
+
+#define SRAM_BANK_SIZE 0x8000
+#define SRAM_CNTL_START 0x7F00
+
+#define DSP_BASE_ADDR 0x4000
+#define DSP_BANK_BASE 0x4000
+
+#define HP_ICR 0x00
+#define HP_CVR 0x01
+#define HP_ISR 0x02
+#define HP_IVR 0x03
+#define HP_NU 0x04
+#define HP_INFO 0x04
+#define HP_TXH 0x05
+#define HP_RXH 0x05
+#define HP_TXM 0x06
+#define HP_RXM 0x06
+#define HP_TXL 0x07
+#define HP_RXL 0x07
+
+#define HP_ICR_DEF 0x00
+#define HP_CVR_DEF 0x12
+#define HP_ISR_DEF 0x06
+#define HP_IVR_DEF 0x0f
+#define HP_NU_DEF 0x00
+
+#define HP_IRQM 0x09
+
+#define HPR_BLRC 0x08
+#define HPR_SPR1 0x09
+#define HPR_SPR2 0x0A
+#define HPR_TCL0 0x0B
+#define HPR_TCL1 0x0C
+#define HPR_TCL2 0x0D
+#define HPR_TCL3 0x0E
+#define HPR_TCL4 0x0F
+
+#define HPICR_INIT 0x80
+#define HPICR_HM1 0x40
+#define HPICR_HM0 0x20
+#define HPICR_HF1 0x10
+#define HPICR_HF0 0x08
+#define HPICR_TREQ 0x02
+#define HPICR_RREQ 0x01
+
+#define HPCVR_HC 0x80
+
+#define HPISR_HREQ 0x80
+#define HPISR_DMA 0x40
+#define HPISR_HF3 0x10
+#define HPISR_HF2 0x08
+#define HPISR_TRDY 0x04
+#define HPISR_TXDE 0x02
+#define HPISR_RXDF 0x01
+
+#define HPIO_290 0
+#define HPIO_260 1
+#define HPIO_250 2
+#define HPIO_240 3
+#define HPIO_230 4
+#define HPIO_220 5
+#define HPIO_210 6
+#define HPIO_3E0 7
+
+#define HPMEM_NONE 0
+#define HPMEM_B000 1
+#define HPMEM_C800 2
+#define HPMEM_D000 3
+#define HPMEM_D400 4
+#define HPMEM_D800 5
+#define HPMEM_E000 6
+#define HPMEM_E800 7
+
+#define HPIRQ_NONE 0
+#define HPIRQ_5 1
+#define HPIRQ_7 2
+#define HPIRQ_9 3
+#define HPIRQ_10 4
+#define HPIRQ_11 5
+#define HPIRQ_12 6
+#define HPIRQ_15 7
+
+#define HIMT_PLAY_DONE 0x00
+#define HIMT_RECORD_DONE 0x01
+#define HIMT_MIDI_EOS 0x02
+#define HIMT_MIDI_OUT 0x03
+
+#define HIMT_MIDI_IN_UCHAR 0x0E
+#define HIMT_DSP 0x0F
+
+#define HDEX_BASE 0x92
+#define HDEX_PLAY_START (0 + HDEX_BASE)
+#define HDEX_PLAY_STOP (1 + HDEX_BASE)
+#define HDEX_PLAY_PAUSE (2 + HDEX_BASE)
+#define HDEX_PLAY_RESUME (3 + HDEX_BASE)
+#define HDEX_RECORD_START (4 + HDEX_BASE)
+#define HDEX_RECORD_STOP (5 + HDEX_BASE)
+#define HDEX_MIDI_IN_START (6 + HDEX_BASE)
+#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE)
+#define HDEX_MIDI_OUT_START (8 + HDEX_BASE)
+#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE)
+#define HDEX_AUX_REQ (10 + HDEX_BASE)
+
+#define HIWORD(l) ((WORD)((((DWORD)(l)) >> 16) & 0xFFFF))
+#define LOWORD(l) ((WORD)(DWORD)(l))
+#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF))
+#define LOBYTE(w) ((BYTE)(w))
+#define MAKELONG(low,hi) ((long)(((WORD)(low))|(((DWORD)((WORD)(hi)))<<16)))
+#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8)))
+
+#define PCTODSP_OFFSET(w) (USHORT)((w)/2)
+#define PCTODSP_BASED(w) (USHORT)(((w)/2) + DSP_BASE_ADDR)
+#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2)
+
+#ifdef SLOWIO
+#define msnd_outb outb_p
+#define msnd_inb inb_p
+#else
+#define msnd_outb outb
+#define msnd_inb inb
+#endif
+
+/* JobQueueStruct */
+#define JQS_wStart 0x00
+#define JQS_wSize 0x02
+#define JQS_wHead 0x04
+#define JQS_wTail 0x06
+#define JQS__size 0x08
+
+/* DAQueueDataStruct */
+#define DAQDS_wStart 0x00
+#define DAQDS_wSize 0x02
+#define DAQDS_wFormat 0x04
+#define DAQDS_wSampleSize 0x06
+#define DAQDS_wChannels 0x08
+#define DAQDS_wSampleRate 0x0A
+#define DAQDS_wIntMsg 0x0C
+#define DAQDS_wFlags 0x0E
+#define DAQDS__size 0x10
+
+typedef u8 BYTE;
+typedef u16 USHORT;
+typedef u16 WORD;
+typedef u32 DWORD;
+typedef void __iomem * LPDAQD;
+
+/* Generic FIFO */
+typedef struct {
+ size_t n, len;
+ char *data;
+ int head, tail;
+} msnd_fifo;
+
+typedef struct multisound_dev {
+ /* Linux device info */
+ char *name;
+ int dsp_minor, mixer_minor;
+ int ext_midi_dev, hdr_midi_dev;
+
+ /* Hardware resources */
+ int io, numio;
+ int memid, irqid;
+ int irq, irq_ref;
+ unsigned char info;
+ void __iomem *base;
+
+ /* Motorola 56k DSP SMA */
+ void __iomem *SMA;
+ void __iomem *DAPQ, *DARQ, *MODQ, *MIDQ, *DSPQ;
+ void __iomem *pwDSPQData, *pwMIDQData, *pwMODQData;
+ int dspq_data_buff, dspq_buff_size;
+
+ /* State variables */
+ enum { msndClassic, msndPinnacle } type;
+ fmode_t mode;
+ unsigned long flags;
+#define F_RESETTING 0
+#define F_HAVEDIGITAL 1
+#define F_AUDIO_WRITE_INUSE 2
+#define F_WRITING 3
+#define F_WRITEBLOCK 4
+#define F_WRITEFLUSH 5
+#define F_AUDIO_READ_INUSE 6
+#define F_READING 7
+#define F_READBLOCK 8
+#define F_EXT_MIDI_INUSE 9
+#define F_HDR_MIDI_INUSE 10
+#define F_DISABLE_WRITE_NDELAY 11
+ wait_queue_head_t writeblock;
+ wait_queue_head_t readblock;
+ wait_queue_head_t writeflush;
+ spinlock_t lock;
+ int nresets;
+ unsigned long recsrc;
+ int left_levels[32];
+ int right_levels[32];
+ int mixer_mod_count;
+ int calibrate_signal;
+ int play_sample_size, play_sample_rate, play_channels;
+ int play_ndelay;
+ int rec_sample_size, rec_sample_rate, rec_channels;
+ int rec_ndelay;
+ BYTE bCurrentMidiPatch;
+
+ /* Digital audio FIFOs */
+ msnd_fifo DAPF, DARF;
+ int fifosize;
+ int last_playbank, last_recbank;
+
+ /* MIDI in callback */
+ void (*midi_in_interrupt)(struct multisound_dev *);
+} multisound_dev_t;
+
+#ifndef mdelay
+# define mdelay(a) udelay((a) * 1000)
+#endif
+
+int msnd_register(multisound_dev_t *dev);
+void msnd_unregister(multisound_dev_t *dev);
+
+void msnd_init_queue(void __iomem *, int start, int size);
+
+void msnd_fifo_init(msnd_fifo *f);
+void msnd_fifo_free(msnd_fifo *f);
+int msnd_fifo_alloc(msnd_fifo *f, size_t n);
+void msnd_fifo_make_empty(msnd_fifo *f);
+int msnd_fifo_write_io(msnd_fifo *f, char __iomem *buf, size_t len);
+int msnd_fifo_read_io(msnd_fifo *f, char __iomem *buf, size_t len);
+int msnd_fifo_write(msnd_fifo *f, const char *buf, size_t len);
+int msnd_fifo_read(msnd_fifo *f, char *buf, size_t len);
+
+int msnd_send_dsp_cmd(multisound_dev_t *dev, BYTE cmd);
+int msnd_send_word(multisound_dev_t *dev, unsigned char high,
+ unsigned char mid, unsigned char low);
+int msnd_upload_host(multisound_dev_t *dev, char *bin, int len);
+int msnd_enable_irq(multisound_dev_t *dev);
+int msnd_disable_irq(multisound_dev_t *dev);
+
+#endif /* __MSND_H */
diff --git a/sound/oss/msnd_classic.c b/sound/oss/msnd_classic.c
new file mode 100644
index 0000000..3b23a09
--- /dev/null
+++ b/sound/oss/msnd_classic.c
@@ -0,0 +1,3 @@
+/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */
+#define MSND_CLASSIC
+#include "msnd_pinnacle.c"
diff --git a/sound/oss/msnd_classic.h b/sound/oss/msnd_classic.h
new file mode 100644
index 0000000..1a17dde
--- /dev/null
+++ b/sound/oss/msnd_classic.h
@@ -0,0 +1,185 @@
+/*********************************************************************
+ *
+ * msnd_classic.h
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Some parts of this header file were derived from the Turtle Beach
+ * MultiSound Driver Development Kit.
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ * Copyright (C) 1993 Turtle Beach Systems, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+#ifndef __MSND_CLASSIC_H
+#define __MSND_CLASSIC_H
+
+
+#define DSP_NUMIO 0x10
+
+#define HP_MEMM 0x08
+
+#define HP_BITM 0x0E
+#define HP_WAIT 0x0D
+#define HP_DSPR 0x0A
+#define HP_PROR 0x0B
+#define HP_BLKS 0x0C
+
+#define HPPRORESET_OFF 0
+#define HPPRORESET_ON 1
+
+#define HPDSPRESET_OFF 0
+#define HPDSPRESET_ON 1
+
+#define HPBLKSEL_0 0
+#define HPBLKSEL_1 1
+
+#define HPWAITSTATE_0 0
+#define HPWAITSTATE_1 1
+
+#define HPBITMODE_16 0
+#define HPBITMODE_8 1
+
+#define HIDSP_INT_PLAY_UNDER 0x00
+#define HIDSP_INT_RECORD_OVER 0x01
+#define HIDSP_INPUT_CLIPPING 0x02
+#define HIDSP_MIDI_IN_OVER 0x10
+#define HIDSP_MIDI_OVERRUN_ERR 0x13
+
+#define HDEXAR_CLEAR_PEAKS 1
+#define HDEXAR_IN_SET_POTS 2
+#define HDEXAR_AUX_SET_POTS 3
+#define HDEXAR_CAL_A_TO_D 4
+#define HDEXAR_RD_EXT_DSP_BITS 5
+
+#define TIME_PRO_RESET_DONE 0x028A
+#define TIME_PRO_SYSEX 0x0040
+#define TIME_PRO_RESET 0x0032
+
+#define AGND 0x01
+#define SIGNAL 0x02
+
+#define EXT_DSP_BIT_DCAL 0x0001
+#define EXT_DSP_BIT_MIDI_CON 0x0002
+
+#define BUFFSIZE 0x8000
+#define HOSTQ_SIZE 0x40
+
+#define SRAM_CNTL_START 0x7F00
+#define SMA_STRUCT_START 0x7F40
+
+#define DAP_BUFF_SIZE 0x2400
+#define DAR_BUFF_SIZE 0x2000
+
+#define DAPQ_STRUCT_SIZE 0x10
+#define DARQ_STRUCT_SIZE 0x10
+#define DAPQ_BUFF_SIZE (3 * 0x10)
+#define DARQ_BUFF_SIZE (3 * 0x10)
+#define MODQ_BUFF_SIZE 0x400
+#define MIDQ_BUFF_SIZE 0x200
+#define DSPQ_BUFF_SIZE 0x40
+
+#define DAPQ_DATA_BUFF 0x6C00
+#define DARQ_DATA_BUFF 0x6C30
+#define MODQ_DATA_BUFF 0x6C60
+#define MIDQ_DATA_BUFF 0x7060
+#define DSPQ_DATA_BUFF 0x7260
+
+#define DAPQ_OFFSET SRAM_CNTL_START
+#define DARQ_OFFSET (SRAM_CNTL_START + 0x08)
+#define MODQ_OFFSET (SRAM_CNTL_START + 0x10)
+#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18)
+#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20)
+
+#define MOP_SYNTH 0x10
+#define MOP_EXTOUT 0x32
+#define MOP_EXTTHRU 0x02
+#define MOP_OUTMASK 0x01
+
+#define MIP_EXTIN 0x01
+#define MIP_SYNTH 0x00
+#define MIP_INMASK 0x32
+
+/* Classic SMA Common Data */
+#define SMA_wCurrPlayBytes 0x0000
+#define SMA_wCurrRecordBytes 0x0002
+#define SMA_wCurrPlayVolLeft 0x0004
+#define SMA_wCurrPlayVolRight 0x0006
+#define SMA_wCurrInVolLeft 0x0008
+#define SMA_wCurrInVolRight 0x000a
+#define SMA_wUser_3 0x000c
+#define SMA_wUser_4 0x000e
+#define SMA_dwUser_5 0x0010
+#define SMA_dwUser_6 0x0014
+#define SMA_wUser_7 0x0018
+#define SMA_wReserved_A 0x001a
+#define SMA_wReserved_B 0x001c
+#define SMA_wReserved_C 0x001e
+#define SMA_wReserved_D 0x0020
+#define SMA_wReserved_E 0x0022
+#define SMA_wReserved_F 0x0024
+#define SMA_wReserved_G 0x0026
+#define SMA_wReserved_H 0x0028
+#define SMA_wCurrDSPStatusFlags 0x002a
+#define SMA_wCurrHostStatusFlags 0x002c
+#define SMA_wCurrInputTagBits 0x002e
+#define SMA_wCurrLeftPeak 0x0030
+#define SMA_wCurrRightPeak 0x0032
+#define SMA_wExtDSPbits 0x0034
+#define SMA_bExtHostbits 0x0036
+#define SMA_bBoardLevel 0x0037
+#define SMA_bInPotPosRight 0x0038
+#define SMA_bInPotPosLeft 0x0039
+#define SMA_bAuxPotPosRight 0x003a
+#define SMA_bAuxPotPosLeft 0x003b
+#define SMA_wCurrMastVolLeft 0x003c
+#define SMA_wCurrMastVolRight 0x003e
+#define SMA_bUser_12 0x0040
+#define SMA_bUser_13 0x0041
+#define SMA_wUser_14 0x0042
+#define SMA_wUser_15 0x0044
+#define SMA_wCalFreqAtoD 0x0046
+#define SMA_wUser_16 0x0048
+#define SMA_wUser_17 0x004a
+#define SMA__size 0x004c
+
+#ifdef HAVE_DSPCODEH
+# include "msndperm.c"
+# include "msndinit.c"
+# define PERMCODE msndperm
+# define INITCODE msndinit
+# define PERMCODESIZE sizeof(msndperm)
+# define INITCODESIZE sizeof(msndinit)
+#else
+# ifndef CONFIG_MSNDCLAS_INIT_FILE
+# define CONFIG_MSNDCLAS_INIT_FILE \
+ "/etc/sound/msndinit.bin"
+# endif
+# ifndef CONFIG_MSNDCLAS_PERM_FILE
+# define CONFIG_MSNDCLAS_PERM_FILE \
+ "/etc/sound/msndperm.bin"
+# endif
+# define PERMCODEFILE CONFIG_MSNDCLAS_PERM_FILE
+# define INITCODEFILE CONFIG_MSNDCLAS_INIT_FILE
+# define PERMCODE dspini
+# define INITCODE permini
+# define PERMCODESIZE sizeof_dspini
+# define INITCODESIZE sizeof_permini
+#endif
+#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)"
+
+#endif /* __MSND_CLASSIC_H */
diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c
new file mode 100644
index 0000000..bf27e00
--- /dev/null
+++ b/sound/oss/msnd_pinnacle.c
@@ -0,0 +1,1916 @@
+/*********************************************************************
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ * Linux 2.0/2.2 Version
+ *
+ * msnd_pinnacle.c / msnd_classic.c
+ *
+ * -- If MSND_CLASSIC is defined:
+ *
+ * -> driver for Turtle Beach Classic/Monterey/Tahiti
+ *
+ * -- Else
+ *
+ * -> driver for Turtle Beach Pinnacle/Fiji
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * 12-3-2000 Modified IO port validation Steve Sycamore
+ *
+ ********************************************************************/
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/types.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/smp_lock.h>
+#include <asm/irq.h>
+#include <asm/io.h>
+#include "sound_config.h"
+#include "sound_firmware.h"
+#ifdef MSND_CLASSIC
+# ifndef __alpha__
+# define SLOWIO
+# endif
+#endif
+#include "msnd.h"
+#ifdef MSND_CLASSIC
+# ifdef CONFIG_MSNDCLAS_HAVE_BOOT
+# define HAVE_DSPCODEH
+# endif
+# include "msnd_classic.h"
+# define LOGNAME "msnd_classic"
+#else
+# ifdef CONFIG_MSNDPIN_HAVE_BOOT
+# define HAVE_DSPCODEH
+# endif
+# include "msnd_pinnacle.h"
+# define LOGNAME "msnd_pinnacle"
+#endif
+
+#ifndef CONFIG_MSND_WRITE_NDELAY
+# define CONFIG_MSND_WRITE_NDELAY 1
+#endif
+
+#define get_play_delay_jiffies(size) ((size) * HZ * \
+ dev.play_sample_size / 8 / \
+ dev.play_sample_rate / \
+ dev.play_channels)
+
+#define get_rec_delay_jiffies(size) ((size) * HZ * \
+ dev.rec_sample_size / 8 / \
+ dev.rec_sample_rate / \
+ dev.rec_channels)
+
+static multisound_dev_t dev;
+
+#ifndef HAVE_DSPCODEH
+static char *dspini, *permini;
+static int sizeof_dspini, sizeof_permini;
+#endif
+
+static int dsp_full_reset(void);
+static void dsp_write_flush(void);
+
+static __inline__ int chk_send_dsp_cmd(multisound_dev_t *dev, register BYTE cmd)
+{
+ if (msnd_send_dsp_cmd(dev, cmd) == 0)
+ return 0;
+ dsp_full_reset();
+ return msnd_send_dsp_cmd(dev, cmd);
+}
+
+static void reset_play_queue(void)
+{
+ int n;
+ LPDAQD lpDAQ;
+
+ dev.last_playbank = -1;
+ writew(PCTODSP_OFFSET(0 * DAQDS__size), dev.DAPQ + JQS_wHead);
+ writew(PCTODSP_OFFSET(0 * DAQDS__size), dev.DAPQ + JQS_wTail);
+
+ for (n = 0, lpDAQ = dev.base + DAPQ_DATA_BUFF; n < 3; ++n, lpDAQ += DAQDS__size) {
+ writew(PCTODSP_BASED((DWORD)(DAP_BUFF_SIZE * n)), lpDAQ + DAQDS_wStart);
+ writew(0, lpDAQ + DAQDS_wSize);
+ writew(1, lpDAQ + DAQDS_wFormat);
+ writew(dev.play_sample_size, lpDAQ + DAQDS_wSampleSize);
+ writew(dev.play_channels, lpDAQ + DAQDS_wChannels);
+ writew(dev.play_sample_rate, lpDAQ + DAQDS_wSampleRate);
+ writew(HIMT_PLAY_DONE * 0x100 + n, lpDAQ + DAQDS_wIntMsg);
+ writew(n, lpDAQ + DAQDS_wFlags);
+ }
+}
+
+static void reset_record_queue(void)
+{
+ int n;
+ LPDAQD lpDAQ;
+ unsigned long flags;
+
+ dev.last_recbank = 2;
+ writew(PCTODSP_OFFSET(0 * DAQDS__size), dev.DARQ + JQS_wHead);
+ writew(PCTODSP_OFFSET(dev.last_recbank * DAQDS__size), dev.DARQ + JQS_wTail);
+
+ /* Critical section: bank 1 access */
+ spin_lock_irqsave(&dev.lock, flags);
+ msnd_outb(HPBLKSEL_1, dev.io + HP_BLKS);
+ memset_io(dev.base, 0, DAR_BUFF_SIZE * 3);
+ msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
+ spin_unlock_irqrestore(&dev.lock, flags);
+
+ for (n = 0, lpDAQ = dev.base + DARQ_DATA_BUFF; n < 3; ++n, lpDAQ += DAQDS__size) {
+ writew(PCTODSP_BASED((DWORD)(DAR_BUFF_SIZE * n)) + 0x4000, lpDAQ + DAQDS_wStart);
+ writew(DAR_BUFF_SIZE, lpDAQ + DAQDS_wSize);
+ writew(1, lpDAQ + DAQDS_wFormat);
+ writew(dev.rec_sample_size, lpDAQ + DAQDS_wSampleSize);
+ writew(dev.rec_channels, lpDAQ + DAQDS_wChannels);
+ writew(dev.rec_sample_rate, lpDAQ + DAQDS_wSampleRate);
+ writew(HIMT_RECORD_DONE * 0x100 + n, lpDAQ + DAQDS_wIntMsg);
+ writew(n, lpDAQ + DAQDS_wFlags);
+ }
+}
+
+static void reset_queues(void)
+{
+ if (dev.mode & FMODE_WRITE) {
+ msnd_fifo_make_empty(&dev.DAPF);
+ reset_play_queue();
+ }
+ if (dev.mode & FMODE_READ) {
+ msnd_fifo_make_empty(&dev.DARF);
+ reset_record_queue();
+ }
+}
+
+static int dsp_set_format(struct file *file, int val)
+{
+ int data, i;
+ LPDAQD lpDAQ, lpDARQ;
+
+ lpDAQ = dev.base + DAPQ_DATA_BUFF;
+ lpDARQ = dev.base + DARQ_DATA_BUFF;
+
+ switch (val) {
+ case AFMT_U8:
+ case AFMT_S16_LE:
+ data = val;
+ break;
+ default:
+ data = DEFSAMPLESIZE;
+ break;
+ }
+
+ for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size, lpDARQ += DAQDS__size) {
+ if (file->f_mode & FMODE_WRITE)
+ writew(data, lpDAQ + DAQDS_wSampleSize);
+ if (file->f_mode & FMODE_READ)
+ writew(data, lpDARQ + DAQDS_wSampleSize);
+ }
+ if (file->f_mode & FMODE_WRITE)
+ dev.play_sample_size = data;
+ if (file->f_mode & FMODE_READ)
+ dev.rec_sample_size = data;
+
+ return data;
+}
+
+static int dsp_get_frag_size(void)
+{
+ int size;
+ size = dev.fifosize / 4;
+ if (size > 32 * 1024)
+ size = 32 * 1024;
+ return size;
+}
+
+static int dsp_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ int val, i, data, tmp;
+ LPDAQD lpDAQ, lpDARQ;
+ audio_buf_info abinfo;
+ unsigned long flags;
+ int __user *p = (int __user *)arg;
+
+ lpDAQ = dev.base + DAPQ_DATA_BUFF;
+ lpDARQ = dev.base + DARQ_DATA_BUFF;
+
+ switch (cmd) {
+ case SNDCTL_DSP_SUBDIVIDE:
+ case SNDCTL_DSP_SETFRAGMENT:
+ case SNDCTL_DSP_SETDUPLEX:
+ case SNDCTL_DSP_POST:
+ return 0;
+
+ case SNDCTL_DSP_GETIPTR:
+ case SNDCTL_DSP_GETOPTR:
+ case SNDCTL_DSP_MAPINBUF:
+ case SNDCTL_DSP_MAPOUTBUF:
+ return -EINVAL;
+
+ case SNDCTL_DSP_GETOSPACE:
+ if (!(file->f_mode & FMODE_WRITE))
+ return -EINVAL;
+ spin_lock_irqsave(&dev.lock, flags);
+ abinfo.fragsize = dsp_get_frag_size();
+ abinfo.bytes = dev.DAPF.n - dev.DAPF.len;
+ abinfo.fragstotal = dev.DAPF.n / abinfo.fragsize;
+ abinfo.fragments = abinfo.bytes / abinfo.fragsize;
+ spin_unlock_irqrestore(&dev.lock, flags);
+ return copy_to_user((void __user *)arg, &abinfo, sizeof(abinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_GETISPACE:
+ if (!(file->f_mode & FMODE_READ))
+ return -EINVAL;
+ spin_lock_irqsave(&dev.lock, flags);
+ abinfo.fragsize = dsp_get_frag_size();
+ abinfo.bytes = dev.DARF.n - dev.DARF.len;
+ abinfo.fragstotal = dev.DARF.n / abinfo.fragsize;
+ abinfo.fragments = abinfo.bytes / abinfo.fragsize;
+ spin_unlock_irqrestore(&dev.lock, flags);
+ return copy_to_user((void __user *)arg, &abinfo, sizeof(abinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_RESET:
+ dev.nresets = 0;
+ reset_queues();
+ return 0;
+
+ case SNDCTL_DSP_SYNC:
+ dsp_write_flush();
+ return 0;
+
+ case SNDCTL_DSP_GETBLKSIZE:
+ tmp = dsp_get_frag_size();
+ if (put_user(tmp, p))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_GETFMTS:
+ val = AFMT_S16_LE | AFMT_U8;
+ if (put_user(val, p))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_SETFMT:
+ if (get_user(val, p))
+ return -EFAULT;
+
+ if (file->f_mode & FMODE_WRITE)
+ data = val == AFMT_QUERY
+ ? dev.play_sample_size
+ : dsp_set_format(file, val);
+ else
+ data = val == AFMT_QUERY
+ ? dev.rec_sample_size
+ : dsp_set_format(file, val);
+
+ if (put_user(data, p))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_NONBLOCK:
+ if (!test_bit(F_DISABLE_WRITE_NDELAY, &dev.flags) &&
+ file->f_mode & FMODE_WRITE)
+ dev.play_ndelay = 1;
+ if (file->f_mode & FMODE_READ)
+ dev.rec_ndelay = 1;
+ return 0;
+
+ case SNDCTL_DSP_GETCAPS:
+ val = DSP_CAP_DUPLEX | DSP_CAP_BATCH;
+ if (put_user(val, p))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_SPEED:
+ if (get_user(val, p))
+ return -EFAULT;
+
+ if (val < 8000)
+ val = 8000;
+
+ if (val > 48000)
+ val = 48000;
+
+ data = val;
+
+ for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size, lpDARQ += DAQDS__size) {
+ if (file->f_mode & FMODE_WRITE)
+ writew(data, lpDAQ + DAQDS_wSampleRate);
+ if (file->f_mode & FMODE_READ)
+ writew(data, lpDARQ + DAQDS_wSampleRate);
+ }
+ if (file->f_mode & FMODE_WRITE)
+ dev.play_sample_rate = data;
+ if (file->f_mode & FMODE_READ)
+ dev.rec_sample_rate = data;
+
+ if (put_user(data, p))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_CHANNELS:
+ case SNDCTL_DSP_STEREO:
+ if (get_user(val, p))
+ return -EFAULT;
+
+ if (cmd == SNDCTL_DSP_CHANNELS) {
+ switch (val) {
+ case 1:
+ case 2:
+ data = val;
+ break;
+ default:
+ val = data = 2;
+ break;
+ }
+ } else {
+ switch (val) {
+ case 0:
+ data = 1;
+ break;
+ default:
+ val = 1;
+ case 1:
+ data = 2;
+ break;
+ }
+ }
+
+ for (i = 0; i < 3; ++i, lpDAQ += DAQDS__size, lpDARQ += DAQDS__size) {
+ if (file->f_mode & FMODE_WRITE)
+ writew(data, lpDAQ + DAQDS_wChannels);
+ if (file->f_mode & FMODE_READ)
+ writew(data, lpDARQ + DAQDS_wChannels);
+ }
+ if (file->f_mode & FMODE_WRITE)
+ dev.play_channels = data;
+ if (file->f_mode & FMODE_READ)
+ dev.rec_channels = data;
+
+ if (put_user(val, p))
+ return -EFAULT;
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
+static int mixer_get(int d)
+{
+ if (d > 31)
+ return -EINVAL;
+
+ switch (d) {
+ case SOUND_MIXER_VOLUME:
+ case SOUND_MIXER_PCM:
+ case SOUND_MIXER_LINE:
+ case SOUND_MIXER_IMIX:
+ case SOUND_MIXER_LINE1:
+#ifndef MSND_CLASSIC
+ case SOUND_MIXER_MIC:
+ case SOUND_MIXER_SYNTH:
+#endif
+ return (dev.left_levels[d] >> 8) * 100 / 0xff |
+ (((dev.right_levels[d] >> 8) * 100 / 0xff) << 8);
+ default:
+ return 0;
+ }
+}
+
+#define update_volm(a,b) \
+ writew((dev.left_levels[a] >> 1) * \
+ readw(dev.SMA + SMA_wCurrMastVolLeft) / 0xffff, \
+ dev.SMA + SMA_##b##Left); \
+ writew((dev.right_levels[a] >> 1) * \
+ readw(dev.SMA + SMA_wCurrMastVolRight) / 0xffff, \
+ dev.SMA + SMA_##b##Right);
+
+#define update_potm(d,s,ar) \
+ writeb((dev.left_levels[d] >> 8) * \
+ readw(dev.SMA + SMA_wCurrMastVolLeft) / 0xffff, \
+ dev.SMA + SMA_##s##Left); \
+ writeb((dev.right_levels[d] >> 8) * \
+ readw(dev.SMA + SMA_wCurrMastVolRight) / 0xffff, \
+ dev.SMA + SMA_##s##Right); \
+ if (msnd_send_word(&dev, 0, 0, ar) == 0) \
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+
+#define update_pot(d,s,ar) \
+ writeb(dev.left_levels[d] >> 8, \
+ dev.SMA + SMA_##s##Left); \
+ writeb(dev.right_levels[d] >> 8, \
+ dev.SMA + SMA_##s##Right); \
+ if (msnd_send_word(&dev, 0, 0, ar) == 0) \
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+
+static int mixer_set(int d, int value)
+{
+ int left = value & 0x000000ff;
+ int right = (value & 0x0000ff00) >> 8;
+ int bLeft, bRight;
+ int wLeft, wRight;
+ int updatemaster = 0;
+
+ if (d > 31)
+ return -EINVAL;
+
+ bLeft = left * 0xff / 100;
+ wLeft = left * 0xffff / 100;
+
+ bRight = right * 0xff / 100;
+ wRight = right * 0xffff / 100;
+
+ dev.left_levels[d] = wLeft;
+ dev.right_levels[d] = wRight;
+
+ switch (d) {
+ /* master volume unscaled controls */
+ case SOUND_MIXER_LINE: /* line pot control */
+ /* scaled by IMIX in digital mix */
+ writeb(bLeft, dev.SMA + SMA_bInPotPosLeft);
+ writeb(bRight, dev.SMA + SMA_bInPotPosRight);
+ if (msnd_send_word(&dev, 0, 0, HDEXAR_IN_SET_POTS) == 0)
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+ break;
+#ifndef MSND_CLASSIC
+ case SOUND_MIXER_MIC: /* mic pot control */
+ /* scaled by IMIX in digital mix */
+ writeb(bLeft, dev.SMA + SMA_bMicPotPosLeft);
+ writeb(bRight, dev.SMA + SMA_bMicPotPosRight);
+ if (msnd_send_word(&dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0)
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+ break;
+#endif
+ case SOUND_MIXER_VOLUME: /* master volume */
+ writew(wLeft, dev.SMA + SMA_wCurrMastVolLeft);
+ writew(wRight, dev.SMA + SMA_wCurrMastVolRight);
+ /* fall through */
+
+ case SOUND_MIXER_LINE1: /* aux pot control */
+ /* scaled by master volume */
+ /* fall through */
+
+ /* digital controls */
+ case SOUND_MIXER_SYNTH: /* synth vol (dsp mix) */
+ case SOUND_MIXER_PCM: /* pcm vol (dsp mix) */
+ case SOUND_MIXER_IMIX: /* input monitor (dsp mix) */
+ /* scaled by master volume */
+ updatemaster = 1;
+ break;
+
+ default:
+ return 0;
+ }
+
+ if (updatemaster) {
+ /* update master volume scaled controls */
+ update_volm(SOUND_MIXER_PCM, wCurrPlayVol);
+ update_volm(SOUND_MIXER_IMIX, wCurrInVol);
+#ifndef MSND_CLASSIC
+ update_volm(SOUND_MIXER_SYNTH, wCurrMHdrVol);
+#endif
+ update_potm(SOUND_MIXER_LINE1, bAuxPotPos, HDEXAR_AUX_SET_POTS);
+ }
+
+ return mixer_get(d);
+}
+
+static void mixer_setup(void)
+{
+ update_pot(SOUND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS);
+ update_potm(SOUND_MIXER_LINE1, bAuxPotPos, HDEXAR_AUX_SET_POTS);
+ update_volm(SOUND_MIXER_PCM, wCurrPlayVol);
+ update_volm(SOUND_MIXER_IMIX, wCurrInVol);
+#ifndef MSND_CLASSIC
+ update_pot(SOUND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS);
+ update_volm(SOUND_MIXER_SYNTH, wCurrMHdrVol);
+#endif
+}
+
+static unsigned long set_recsrc(unsigned long recsrc)
+{
+ if (dev.recsrc == recsrc)
+ return dev.recsrc;
+#ifdef HAVE_NORECSRC
+ else if (recsrc == 0)
+ dev.recsrc = 0;
+#endif
+ else
+ dev.recsrc ^= recsrc;
+
+#ifndef MSND_CLASSIC
+ if (dev.recsrc & SOUND_MASK_IMIX) {
+ if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_ANA_IN) == 0)
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+ }
+ else if (dev.recsrc & SOUND_MASK_SYNTH) {
+ if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_SYNTH_IN) == 0)
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+ }
+ else if ((dev.recsrc & SOUND_MASK_DIGITAL1) && test_bit(F_HAVEDIGITAL, &dev.flags)) {
+ if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_DAT_IN) == 0)
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+ }
+ else {
+#ifdef HAVE_NORECSRC
+ /* Select no input (?) */
+ dev.recsrc = 0;
+#else
+ dev.recsrc = SOUND_MASK_IMIX;
+ if (msnd_send_word(&dev, 0, 0, HDEXAR_SET_ANA_IN) == 0)
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ);
+#endif
+ }
+#endif /* MSND_CLASSIC */
+
+ return dev.recsrc;
+}
+
+static unsigned long force_recsrc(unsigned long recsrc)
+{
+ dev.recsrc = 0;
+ return set_recsrc(recsrc);
+}
+
+#define set_mixer_info() \
+ memset(&info, 0, sizeof(info)); \
+ strlcpy(info.id, "MSNDMIXER", sizeof(info.id)); \
+ strlcpy(info.name, "MultiSound Mixer", sizeof(info.name));
+
+static int mixer_ioctl(unsigned int cmd, unsigned long arg)
+{
+ if (cmd == SOUND_MIXER_INFO) {
+ mixer_info info;
+ set_mixer_info();
+ info.modify_counter = dev.mixer_mod_count;
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ } else if (cmd == SOUND_OLD_MIXER_INFO) {
+ _old_mixer_info info;
+ set_mixer_info();
+ if (copy_to_user((void __user *)arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ } else if (cmd == SOUND_MIXER_PRIVATE1) {
+ dev.nresets = 0;
+ dsp_full_reset();
+ return 0;
+ } else if (((cmd >> 8) & 0xff) == 'M') {
+ int val = 0;
+
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE) {
+ switch (cmd & 0xff) {
+ case SOUND_MIXER_RECSRC:
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ val = set_recsrc(val);
+ break;
+
+ default:
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ val = mixer_set(cmd & 0xff, val);
+ break;
+ }
+ ++dev.mixer_mod_count;
+ return put_user(val, (int __user *)arg);
+ } else {
+ switch (cmd & 0xff) {
+ case SOUND_MIXER_RECSRC:
+ val = dev.recsrc;
+ break;
+
+ case SOUND_MIXER_DEVMASK:
+ case SOUND_MIXER_STEREODEVS:
+ val = SOUND_MASK_PCM |
+ SOUND_MASK_LINE |
+ SOUND_MASK_IMIX |
+ SOUND_MASK_LINE1 |
+#ifndef MSND_CLASSIC
+ SOUND_MASK_MIC |
+ SOUND_MASK_SYNTH |
+#endif
+ SOUND_MASK_VOLUME;
+ break;
+
+ case SOUND_MIXER_RECMASK:
+#ifdef MSND_CLASSIC
+ val = 0;
+#else
+ val = SOUND_MASK_IMIX |
+ SOUND_MASK_SYNTH;
+ if (test_bit(F_HAVEDIGITAL, &dev.flags))
+ val |= SOUND_MASK_DIGITAL1;
+#endif
+ break;
+
+ case SOUND_MIXER_CAPS:
+ val = SOUND_CAP_EXCL_INPUT;
+ break;
+
+ default:
+ if ((val = mixer_get(cmd & 0xff)) < 0)
+ return -EINVAL;
+ break;
+ }
+ }
+
+ return put_user(val, (int __user *)arg);
+ }
+
+ return -EINVAL;
+}
+
+static int dev_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long arg)
+{
+ int minor = iminor(inode);
+
+ if (cmd == OSS_GETVERSION) {
+ int sound_version = SOUND_VERSION;
+ return put_user(sound_version, (int __user *)arg);
+ }
+
+ if (minor == dev.dsp_minor)
+ return dsp_ioctl(file, cmd, arg);
+ else if (minor == dev.mixer_minor)
+ return mixer_ioctl(cmd, arg);
+
+ return -EINVAL;
+}
+
+static void dsp_write_flush(void)
+{
+ if (!(dev.mode & FMODE_WRITE) || !test_bit(F_WRITING, &dev.flags))
+ return;
+ set_bit(F_WRITEFLUSH, &dev.flags);
+ interruptible_sleep_on_timeout(
+ &dev.writeflush,
+ get_play_delay_jiffies(dev.DAPF.len));
+ clear_bit(F_WRITEFLUSH, &dev.flags);
+ if (!signal_pending(current)) {
+ current->state = TASK_INTERRUPTIBLE;
+ schedule_timeout(get_play_delay_jiffies(DAP_BUFF_SIZE));
+ }
+ clear_bit(F_WRITING, &dev.flags);
+}
+
+static void dsp_halt(struct file *file)
+{
+ if ((file ? file->f_mode : dev.mode) & FMODE_READ) {
+ clear_bit(F_READING, &dev.flags);
+ chk_send_dsp_cmd(&dev, HDEX_RECORD_STOP);
+ msnd_disable_irq(&dev);
+ if (file) {
+ printk(KERN_DEBUG LOGNAME ": Stopping read for %p\n", file);
+ dev.mode &= ~FMODE_READ;
+ }
+ clear_bit(F_AUDIO_READ_INUSE, &dev.flags);
+ }
+ if ((file ? file->f_mode : dev.mode) & FMODE_WRITE) {
+ if (test_bit(F_WRITING, &dev.flags)) {
+ dsp_write_flush();
+ chk_send_dsp_cmd(&dev, HDEX_PLAY_STOP);
+ }
+ msnd_disable_irq(&dev);
+ if (file) {
+ printk(KERN_DEBUG LOGNAME ": Stopping write for %p\n", file);
+ dev.mode &= ~FMODE_WRITE;
+ }
+ clear_bit(F_AUDIO_WRITE_INUSE, &dev.flags);
+ }
+}
+
+static int dsp_release(struct file *file)
+{
+ dsp_halt(file);
+ return 0;
+}
+
+static int dsp_open(struct file *file)
+{
+ if ((file ? file->f_mode : dev.mode) & FMODE_WRITE) {
+ set_bit(F_AUDIO_WRITE_INUSE, &dev.flags);
+ clear_bit(F_WRITING, &dev.flags);
+ msnd_fifo_make_empty(&dev.DAPF);
+ reset_play_queue();
+ if (file) {
+ printk(KERN_DEBUG LOGNAME ": Starting write for %p\n", file);
+ dev.mode |= FMODE_WRITE;
+ }
+ msnd_enable_irq(&dev);
+ }
+ if ((file ? file->f_mode : dev.mode) & FMODE_READ) {
+ set_bit(F_AUDIO_READ_INUSE, &dev.flags);
+ clear_bit(F_READING, &dev.flags);
+ msnd_fifo_make_empty(&dev.DARF);
+ reset_record_queue();
+ if (file) {
+ printk(KERN_DEBUG LOGNAME ": Starting read for %p\n", file);
+ dev.mode |= FMODE_READ;
+ }
+ msnd_enable_irq(&dev);
+ }
+ return 0;
+}
+
+static void set_default_play_audio_parameters(void)
+{
+ dev.play_sample_size = DEFSAMPLESIZE;
+ dev.play_sample_rate = DEFSAMPLERATE;
+ dev.play_channels = DEFCHANNELS;
+}
+
+static void set_default_rec_audio_parameters(void)
+{
+ dev.rec_sample_size = DEFSAMPLESIZE;
+ dev.rec_sample_rate = DEFSAMPLERATE;
+ dev.rec_channels = DEFCHANNELS;
+}
+
+static void set_default_audio_parameters(void)
+{
+ set_default_play_audio_parameters();
+ set_default_rec_audio_parameters();
+}
+
+static int dev_open(struct inode *inode, struct file *file)
+{
+ int minor = iminor(inode);
+ int err = 0;
+
+ if (minor == dev.dsp_minor) {
+ if ((file->f_mode & FMODE_WRITE &&
+ test_bit(F_AUDIO_WRITE_INUSE, &dev.flags)) ||
+ (file->f_mode & FMODE_READ &&
+ test_bit(F_AUDIO_READ_INUSE, &dev.flags)))
+ return -EBUSY;
+
+ if ((err = dsp_open(file)) >= 0) {
+ dev.nresets = 0;
+ if (file->f_mode & FMODE_WRITE) {
+ set_default_play_audio_parameters();
+ if (!test_bit(F_DISABLE_WRITE_NDELAY, &dev.flags))
+ dev.play_ndelay = (file->f_flags & O_NDELAY) ? 1 : 0;
+ else
+ dev.play_ndelay = 0;
+ }
+ if (file->f_mode & FMODE_READ) {
+ set_default_rec_audio_parameters();
+ dev.rec_ndelay = (file->f_flags & O_NDELAY) ? 1 : 0;
+ }
+ }
+ }
+ else if (minor == dev.mixer_minor) {
+ /* nothing */
+ } else
+ err = -EINVAL;
+
+ return err;
+}
+
+static int dev_release(struct inode *inode, struct file *file)
+{
+ int minor = iminor(inode);
+ int err = 0;
+
+ lock_kernel();
+ if (minor == dev.dsp_minor)
+ err = dsp_release(file);
+ else if (minor == dev.mixer_minor) {
+ /* nothing */
+ } else
+ err = -EINVAL;
+ unlock_kernel();
+ return err;
+}
+
+static __inline__ int pack_DARQ_to_DARF(register int bank)
+{
+ register int size, timeout = 3;
+ register WORD wTmp;
+ LPDAQD DAQD;
+
+ /* Increment the tail and check for queue wrap */
+ wTmp = readw(dev.DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size);
+ if (wTmp > readw(dev.DARQ + JQS_wSize))
+ wTmp = 0;
+ while (wTmp == readw(dev.DARQ + JQS_wHead) && timeout--)
+ udelay(1);
+ writew(wTmp, dev.DARQ + JQS_wTail);
+
+ /* Get our digital audio queue struct */
+ DAQD = bank * DAQDS__size + dev.base + DARQ_DATA_BUFF;
+
+ /* Get length of data */
+ size = readw(DAQD + DAQDS_wSize);
+
+ /* Read data from the head (unprotected bank 1 access okay
+ since this is only called inside an interrupt) */
+ msnd_outb(HPBLKSEL_1, dev.io + HP_BLKS);
+ msnd_fifo_write_io(
+ &dev.DARF,
+ dev.base + bank * DAR_BUFF_SIZE,
+ size);
+ msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
+
+ return 1;
+}
+
+static __inline__ int pack_DAPF_to_DAPQ(register int start)
+{
+ register WORD DAPQ_tail;
+ register int protect = start, nbanks = 0;
+ LPDAQD DAQD;
+
+ DAPQ_tail = readw(dev.DAPQ + JQS_wTail);
+ while (DAPQ_tail != readw(dev.DAPQ + JQS_wHead) || start) {
+ register int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size);
+ register int n;
+ unsigned long flags;
+
+ /* Write the data to the new tail */
+ if (protect) {
+ /* Critical section: protect fifo in non-interrupt */
+ spin_lock_irqsave(&dev.lock, flags);
+ n = msnd_fifo_read_io(
+ &dev.DAPF,
+ dev.base + bank_num * DAP_BUFF_SIZE,
+ DAP_BUFF_SIZE);
+ spin_unlock_irqrestore(&dev.lock, flags);
+ } else {
+ n = msnd_fifo_read_io(
+ &dev.DAPF,
+ dev.base + bank_num * DAP_BUFF_SIZE,
+ DAP_BUFF_SIZE);
+ }
+ if (!n)
+ break;
+
+ if (start)
+ start = 0;
+
+ /* Get our digital audio queue struct */
+ DAQD = bank_num * DAQDS__size + dev.base + DAPQ_DATA_BUFF;
+
+ /* Write size of this bank */
+ writew(n, DAQD + DAQDS_wSize);
+ ++nbanks;
+
+ /* Then advance the tail */
+ DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size);
+ writew(DAPQ_tail, dev.DAPQ + JQS_wTail);
+ /* Tell the DSP to play the bank */
+ msnd_send_dsp_cmd(&dev, HDEX_PLAY_START);
+ }
+ return nbanks;
+}
+
+static int dsp_read(char __user *buf, size_t len)
+{
+ int count = len;
+ char *page = (char *)__get_free_page(GFP_KERNEL);
+
+ if (!page)
+ return -ENOMEM;
+
+ while (count > 0) {
+ int n, k;
+ unsigned long flags;
+
+ k = PAGE_SIZE;
+ if (k > count)
+ k = count;
+
+ /* Critical section: protect fifo in non-interrupt */
+ spin_lock_irqsave(&dev.lock, flags);
+ n = msnd_fifo_read(&dev.DARF, page, k);
+ spin_unlock_irqrestore(&dev.lock, flags);
+ if (copy_to_user(buf, page, n)) {
+ free_page((unsigned long)page);
+ return -EFAULT;
+ }
+ buf += n;
+ count -= n;
+
+ if (n == k && count)
+ continue;
+
+ if (!test_bit(F_READING, &dev.flags) && dev.mode & FMODE_READ) {
+ dev.last_recbank = -1;
+ if (chk_send_dsp_cmd(&dev, HDEX_RECORD_START) == 0)
+ set_bit(F_READING, &dev.flags);
+ }
+
+ if (dev.rec_ndelay) {
+ free_page((unsigned long)page);
+ return count == len ? -EAGAIN : len - count;
+ }
+
+ if (count > 0) {
+ set_bit(F_READBLOCK, &dev.flags);
+ if (!interruptible_sleep_on_timeout(
+ &dev.readblock,
+ get_rec_delay_jiffies(DAR_BUFF_SIZE)))
+ clear_bit(F_READING, &dev.flags);
+ clear_bit(F_READBLOCK, &dev.flags);
+ if (signal_pending(current)) {
+ free_page((unsigned long)page);
+ return -EINTR;
+ }
+ }
+ }
+ free_page((unsigned long)page);
+ return len - count;
+}
+
+static int dsp_write(const char __user *buf, size_t len)
+{
+ int count = len;
+ char *page = (char *)__get_free_page(GFP_KERNEL);
+
+ if (!page)
+ return -ENOMEM;
+
+ while (count > 0) {
+ int n, k;
+ unsigned long flags;
+
+ k = PAGE_SIZE;
+ if (k > count)
+ k = count;
+
+ if (copy_from_user(page, buf, k)) {
+ free_page((unsigned long)page);
+ return -EFAULT;
+ }
+
+ /* Critical section: protect fifo in non-interrupt */
+ spin_lock_irqsave(&dev.lock, flags);
+ n = msnd_fifo_write(&dev.DAPF, page, k);
+ spin_unlock_irqrestore(&dev.lock, flags);
+ buf += n;
+ count -= n;
+
+ if (count && n == k)
+ continue;
+
+ if (!test_bit(F_WRITING, &dev.flags) && (dev.mode & FMODE_WRITE)) {
+ dev.last_playbank = -1;
+ if (pack_DAPF_to_DAPQ(1) > 0)
+ set_bit(F_WRITING, &dev.flags);
+ }
+
+ if (dev.play_ndelay) {
+ free_page((unsigned long)page);
+ return count == len ? -EAGAIN : len - count;
+ }
+
+ if (count > 0) {
+ set_bit(F_WRITEBLOCK, &dev.flags);
+ interruptible_sleep_on_timeout(
+ &dev.writeblock,
+ get_play_delay_jiffies(DAP_BUFF_SIZE));
+ clear_bit(F_WRITEBLOCK, &dev.flags);
+ if (signal_pending(current)) {
+ free_page((unsigned long)page);
+ return -EINTR;
+ }
+ }
+ }
+
+ free_page((unsigned long)page);
+ return len - count;
+}
+
+static ssize_t dev_read(struct file *file, char __user *buf, size_t count, loff_t *off)
+{
+ int minor = iminor(file->f_path.dentry->d_inode);
+ if (minor == dev.dsp_minor)
+ return dsp_read(buf, count);
+ else
+ return -EINVAL;
+}
+
+static ssize_t dev_write(struct file *file, const char __user *buf, size_t count, loff_t *off)
+{
+ int minor = iminor(file->f_path.dentry->d_inode);
+ if (minor == dev.dsp_minor)
+ return dsp_write(buf, count);
+ else
+ return -EINVAL;
+}
+
+static __inline__ void eval_dsp_msg(register WORD wMessage)
+{
+ switch (HIBYTE(wMessage)) {
+ case HIMT_PLAY_DONE:
+ if (dev.last_playbank == LOBYTE(wMessage) || !test_bit(F_WRITING, &dev.flags))
+ break;
+ dev.last_playbank = LOBYTE(wMessage);
+
+ if (pack_DAPF_to_DAPQ(0) <= 0) {
+ if (!test_bit(F_WRITEBLOCK, &dev.flags)) {
+ if (test_and_clear_bit(F_WRITEFLUSH, &dev.flags))
+ wake_up_interruptible(&dev.writeflush);
+ }
+ clear_bit(F_WRITING, &dev.flags);
+ }
+
+ if (test_bit(F_WRITEBLOCK, &dev.flags))
+ wake_up_interruptible(&dev.writeblock);
+ break;
+
+ case HIMT_RECORD_DONE:
+ if (dev.last_recbank == LOBYTE(wMessage))
+ break;
+ dev.last_recbank = LOBYTE(wMessage);
+
+ pack_DARQ_to_DARF(dev.last_recbank);
+
+ if (test_bit(F_READBLOCK, &dev.flags))
+ wake_up_interruptible(&dev.readblock);
+ break;
+
+ case HIMT_DSP:
+ switch (LOBYTE(wMessage)) {
+#ifndef MSND_CLASSIC
+ case HIDSP_PLAY_UNDER:
+#endif
+ case HIDSP_INT_PLAY_UNDER:
+/* printk(KERN_DEBUG LOGNAME ": Play underflow\n"); */
+ clear_bit(F_WRITING, &dev.flags);
+ break;
+
+ case HIDSP_INT_RECORD_OVER:
+/* printk(KERN_DEBUG LOGNAME ": Record overflow\n"); */
+ clear_bit(F_READING, &dev.flags);
+ break;
+
+ default:
+/* printk(KERN_DEBUG LOGNAME ": DSP message %d 0x%02x\n",
+ LOBYTE(wMessage), LOBYTE(wMessage)); */
+ break;
+ }
+ break;
+
+ case HIMT_MIDI_IN_UCHAR:
+ if (dev.midi_in_interrupt)
+ (*dev.midi_in_interrupt)(&dev);
+ break;
+
+ default:
+/* printk(KERN_DEBUG LOGNAME ": HIMT message %d 0x%02x\n", HIBYTE(wMessage), HIBYTE(wMessage)); */
+ break;
+ }
+}
+
+static irqreturn_t intr(int irq, void *dev_id)
+{
+ /* Send ack to DSP */
+ msnd_inb(dev.io + HP_RXL);
+
+ /* Evaluate queued DSP messages */
+ while (readw(dev.DSPQ + JQS_wTail) != readw(dev.DSPQ + JQS_wHead)) {
+ register WORD wTmp;
+
+ eval_dsp_msg(readw(dev.pwDSPQData + 2*readw(dev.DSPQ + JQS_wHead)));
+
+ if ((wTmp = readw(dev.DSPQ + JQS_wHead) + 1) > readw(dev.DSPQ + JQS_wSize))
+ writew(0, dev.DSPQ + JQS_wHead);
+ else
+ writew(wTmp, dev.DSPQ + JQS_wHead);
+ }
+ return IRQ_HANDLED;
+}
+
+static const struct file_operations dev_fileops = {
+ .owner = THIS_MODULE,
+ .read = dev_read,
+ .write = dev_write,
+ .ioctl = dev_ioctl,
+ .open = dev_open,
+ .release = dev_release,
+};
+
+static int reset_dsp(void)
+{
+ int timeout = 100;
+
+ msnd_outb(HPDSPRESET_ON, dev.io + HP_DSPR);
+ mdelay(1);
+#ifndef MSND_CLASSIC
+ dev.info = msnd_inb(dev.io + HP_INFO);
+#endif
+ msnd_outb(HPDSPRESET_OFF, dev.io + HP_DSPR);
+ mdelay(1);
+ while (timeout-- > 0) {
+ if (msnd_inb(dev.io + HP_CVR) == HP_CVR_DEF)
+ return 0;
+ mdelay(1);
+ }
+ printk(KERN_ERR LOGNAME ": Cannot reset DSP\n");
+
+ return -EIO;
+}
+
+static int __init probe_multisound(void)
+{
+#ifndef MSND_CLASSIC
+ char *xv, *rev = NULL;
+ char *pin = "Pinnacle", *fiji = "Fiji";
+ char *pinfiji = "Pinnacle/Fiji";
+#endif
+
+ if (!request_region(dev.io, dev.numio, "probing")) {
+ printk(KERN_ERR LOGNAME ": I/O port conflict\n");
+ return -ENODEV;
+ }
+
+ if (reset_dsp() < 0) {
+ release_region(dev.io, dev.numio);
+ return -ENODEV;
+ }
+
+#ifdef MSND_CLASSIC
+ dev.name = "Classic/Tahiti/Monterey";
+ printk(KERN_INFO LOGNAME ": %s, "
+#else
+ switch (dev.info >> 4) {
+ case 0xf: xv = "<= 1.15"; break;
+ case 0x1: xv = "1.18/1.2"; break;
+ case 0x2: xv = "1.3"; break;
+ case 0x3: xv = "1.4"; break;
+ default: xv = "unknown"; break;
+ }
+
+ switch (dev.info & 0x7) {
+ case 0x0: rev = "I"; dev.name = pin; break;
+ case 0x1: rev = "F"; dev.name = pin; break;
+ case 0x2: rev = "G"; dev.name = pin; break;
+ case 0x3: rev = "H"; dev.name = pin; break;
+ case 0x4: rev = "E"; dev.name = fiji; break;
+ case 0x5: rev = "C"; dev.name = fiji; break;
+ case 0x6: rev = "D"; dev.name = fiji; break;
+ case 0x7:
+ rev = "A-B (Fiji) or A-E (Pinnacle)";
+ dev.name = pinfiji;
+ break;
+ }
+ printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, "
+#endif /* MSND_CLASSIC */
+ "I/O 0x%x-0x%x, IRQ %d, memory mapped to %p-%p\n",
+ dev.name,
+#ifndef MSND_CLASSIC
+ rev, xv,
+#endif
+ dev.io, dev.io + dev.numio - 1,
+ dev.irq,
+ dev.base, dev.base + 0x7fff);
+
+ release_region(dev.io, dev.numio);
+ return 0;
+}
+
+static int init_sma(void)
+{
+ static int initted;
+ WORD mastVolLeft, mastVolRight;
+ unsigned long flags;
+
+#ifdef MSND_CLASSIC
+ msnd_outb(dev.memid, dev.io + HP_MEMM);
+#endif
+ msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
+ if (initted) {
+ mastVolLeft = readw(dev.SMA + SMA_wCurrMastVolLeft);
+ mastVolRight = readw(dev.SMA + SMA_wCurrMastVolRight);
+ } else
+ mastVolLeft = mastVolRight = 0;
+ memset_io(dev.base, 0, 0x8000);
+
+ /* Critical section: bank 1 access */
+ spin_lock_irqsave(&dev.lock, flags);
+ msnd_outb(HPBLKSEL_1, dev.io + HP_BLKS);
+ memset_io(dev.base, 0, 0x8000);
+ msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
+ spin_unlock_irqrestore(&dev.lock, flags);
+
+ dev.pwDSPQData = (dev.base + DSPQ_DATA_BUFF);
+ dev.pwMODQData = (dev.base + MODQ_DATA_BUFF);
+ dev.pwMIDQData = (dev.base + MIDQ_DATA_BUFF);
+
+ /* Motorola 56k shared memory base */
+ dev.SMA = dev.base + SMA_STRUCT_START;
+
+ /* Digital audio play queue */
+ dev.DAPQ = dev.base + DAPQ_OFFSET;
+ msnd_init_queue(dev.DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE);
+
+ /* Digital audio record queue */
+ dev.DARQ = dev.base + DARQ_OFFSET;
+ msnd_init_queue(dev.DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE);
+
+ /* MIDI out queue */
+ dev.MODQ = dev.base + MODQ_OFFSET;
+ msnd_init_queue(dev.MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE);
+
+ /* MIDI in queue */
+ dev.MIDQ = dev.base + MIDQ_OFFSET;
+ msnd_init_queue(dev.MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE);
+
+ /* DSP -> host message queue */
+ dev.DSPQ = dev.base + DSPQ_OFFSET;
+ msnd_init_queue(dev.DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE);
+
+ /* Setup some DSP values */
+#ifndef MSND_CLASSIC
+ writew(1, dev.SMA + SMA_wCurrPlayFormat);
+ writew(dev.play_sample_size, dev.SMA + SMA_wCurrPlaySampleSize);
+ writew(dev.play_channels, dev.SMA + SMA_wCurrPlayChannels);
+ writew(dev.play_sample_rate, dev.SMA + SMA_wCurrPlaySampleRate);
+#endif
+ writew(dev.play_sample_rate, dev.SMA + SMA_wCalFreqAtoD);
+ writew(mastVolLeft, dev.SMA + SMA_wCurrMastVolLeft);
+ writew(mastVolRight, dev.SMA + SMA_wCurrMastVolRight);
+#ifndef MSND_CLASSIC
+ writel(0x00010000, dev.SMA + SMA_dwCurrPlayPitch);
+ writel(0x00000001, dev.SMA + SMA_dwCurrPlayRate);
+#endif
+ writew(0x303, dev.SMA + SMA_wCurrInputTagBits);
+
+ initted = 1;
+
+ return 0;
+}
+
+static int __init calibrate_adc(WORD srate)
+{
+ writew(srate, dev.SMA + SMA_wCalFreqAtoD);
+ if (dev.calibrate_signal == 0)
+ writew(readw(dev.SMA + SMA_wCurrHostStatusFlags)
+ | 0x0001, dev.SMA + SMA_wCurrHostStatusFlags);
+ else
+ writew(readw(dev.SMA + SMA_wCurrHostStatusFlags)
+ & ~0x0001, dev.SMA + SMA_wCurrHostStatusFlags);
+ if (msnd_send_word(&dev, 0, 0, HDEXAR_CAL_A_TO_D) == 0 &&
+ chk_send_dsp_cmd(&dev, HDEX_AUX_REQ) == 0) {
+ current->state = TASK_INTERRUPTIBLE;
+ schedule_timeout(HZ / 3);
+ return 0;
+ }
+ printk(KERN_WARNING LOGNAME ": ADC calibration failed\n");
+
+ return -EIO;
+}
+
+static int upload_dsp_code(void)
+{
+ msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
+#ifndef HAVE_DSPCODEH
+ INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE);
+ if (!INITCODE) {
+ printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE);
+ return -EBUSY;
+ }
+
+ PERMCODESIZE = mod_firmware_load(PERMCODEFILE, &PERMCODE);
+ if (!PERMCODE) {
+ printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE);
+ vfree(INITCODE);
+ return -EBUSY;
+ }
+#endif
+ memcpy_toio(dev.base, PERMCODE, PERMCODESIZE);
+ if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) {
+ printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n");
+ return -ENODEV;
+ }
+#ifdef HAVE_DSPCODEH
+ printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n");
+#else
+ printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n");
+#endif
+
+#ifndef HAVE_DSPCODEH
+ vfree(INITCODE);
+ vfree(PERMCODE);
+#endif
+
+ return 0;
+}
+
+#ifdef MSND_CLASSIC
+static void reset_proteus(void)
+{
+ msnd_outb(HPPRORESET_ON, dev.io + HP_PROR);
+ mdelay(TIME_PRO_RESET);
+ msnd_outb(HPPRORESET_OFF, dev.io + HP_PROR);
+ mdelay(TIME_PRO_RESET_DONE);
+}
+#endif
+
+static int initialize(void)
+{
+ int err, timeout;
+
+#ifdef MSND_CLASSIC
+ msnd_outb(HPWAITSTATE_0, dev.io + HP_WAIT);
+ msnd_outb(HPBITMODE_16, dev.io + HP_BITM);
+
+ reset_proteus();
+#endif
+ if ((err = init_sma()) < 0) {
+ printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n");
+ return err;
+ }
+
+ if ((err = reset_dsp()) < 0)
+ return err;
+
+ if ((err = upload_dsp_code()) < 0) {
+ printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n");
+ return err;
+ }
+
+ timeout = 200;
+ while (readw(dev.base)) {
+ mdelay(1);
+ if (!timeout--) {
+ printk(KERN_DEBUG LOGNAME ": DSP reset timeout\n");
+ return -EIO;
+ }
+ }
+
+ mixer_setup();
+
+ return 0;
+}
+
+static int dsp_full_reset(void)
+{
+ int rv;
+
+ if (test_bit(F_RESETTING, &dev.flags) || ++dev.nresets > 10)
+ return 0;
+
+ set_bit(F_RESETTING, &dev.flags);
+ printk(KERN_INFO LOGNAME ": DSP reset\n");
+ dsp_halt(NULL); /* Unconditionally halt */
+ if ((rv = initialize()))
+ printk(KERN_WARNING LOGNAME ": DSP reset failed\n");
+ force_recsrc(dev.recsrc);
+ dsp_open(NULL);
+ clear_bit(F_RESETTING, &dev.flags);
+
+ return rv;
+}
+
+static int __init attach_multisound(void)
+{
+ int err;
+
+ if ((err = request_irq(dev.irq, intr, 0, dev.name, &dev)) < 0) {
+ printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", dev.irq);
+ return err;
+ }
+ request_region(dev.io, dev.numio, dev.name);
+
+ if ((err = dsp_full_reset()) < 0) {
+ release_region(dev.io, dev.numio);
+ free_irq(dev.irq, &dev);
+ return err;
+ }
+
+ if ((err = msnd_register(&dev)) < 0) {
+ printk(KERN_ERR LOGNAME ": Unable to register MultiSound\n");
+ release_region(dev.io, dev.numio);
+ free_irq(dev.irq, &dev);
+ return err;
+ }
+
+ if ((dev.dsp_minor = register_sound_dsp(&dev_fileops, -1)) < 0) {
+ printk(KERN_ERR LOGNAME ": Unable to register DSP operations\n");
+ msnd_unregister(&dev);
+ release_region(dev.io, dev.numio);
+ free_irq(dev.irq, &dev);
+ return dev.dsp_minor;
+ }
+
+ if ((dev.mixer_minor = register_sound_mixer(&dev_fileops, -1)) < 0) {
+ printk(KERN_ERR LOGNAME ": Unable to register mixer operations\n");
+ unregister_sound_mixer(dev.mixer_minor);
+ msnd_unregister(&dev);
+ release_region(dev.io, dev.numio);
+ free_irq(dev.irq, &dev);
+ return dev.mixer_minor;
+ }
+
+ dev.ext_midi_dev = dev.hdr_midi_dev = -1;
+
+ disable_irq(dev.irq);
+ calibrate_adc(dev.play_sample_rate);
+#ifndef MSND_CLASSIC
+ force_recsrc(SOUND_MASK_IMIX);
+#endif
+
+ return 0;
+}
+
+static void __exit unload_multisound(void)
+{
+ release_region(dev.io, dev.numio);
+ free_irq(dev.irq, &dev);
+ unregister_sound_mixer(dev.mixer_minor);
+ unregister_sound_dsp(dev.dsp_minor);
+ msnd_unregister(&dev);
+}
+
+#ifndef MSND_CLASSIC
+
+/* Pinnacle/Fiji Logical Device Configuration */
+
+static int __init msnd_write_cfg(int cfg, int reg, int value)
+{
+ msnd_outb(reg, cfg);
+ msnd_outb(value, cfg + 1);
+ if (value != msnd_inb(cfg + 1)) {
+ printk(KERN_ERR LOGNAME ": msnd_write_cfg: I/O error\n");
+ return -EIO;
+ }
+ return 0;
+}
+
+static int __init msnd_write_cfg_io0(int cfg, int num, WORD io)
+{
+ if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io)))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io)))
+ return -EIO;
+ return 0;
+}
+
+static int __init msnd_write_cfg_io1(int cfg, int num, WORD io)
+{
+ if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io)))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io)))
+ return -EIO;
+ return 0;
+}
+
+static int __init msnd_write_cfg_irq(int cfg, int num, WORD irq)
+{
+ if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq)))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE))
+ return -EIO;
+ return 0;
+}
+
+static int __init msnd_write_cfg_mem(int cfg, int num, int mem)
+{
+ WORD wmem;
+
+ mem >>= 8;
+ mem &= 0xfff;
+ wmem = (WORD)mem;
+ if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem)))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem)))
+ return -EIO;
+ if (wmem && msnd_write_cfg(cfg, IREG_MEMCONTROL, (MEMTYPE_HIADDR | MEMTYPE_16BIT)))
+ return -EIO;
+ return 0;
+}
+
+static int __init msnd_activate_logical(int cfg, int num)
+{
+ if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+ return -EIO;
+ if (msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE))
+ return -EIO;
+ return 0;
+}
+
+static int __init msnd_write_cfg_logical(int cfg, int num, WORD io0, WORD io1, WORD irq, int mem)
+{
+ if (msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+ return -EIO;
+ if (msnd_write_cfg_io0(cfg, num, io0))
+ return -EIO;
+ if (msnd_write_cfg_io1(cfg, num, io1))
+ return -EIO;
+ if (msnd_write_cfg_irq(cfg, num, irq))
+ return -EIO;
+ if (msnd_write_cfg_mem(cfg, num, mem))
+ return -EIO;
+ if (msnd_activate_logical(cfg, num))
+ return -EIO;
+ return 0;
+}
+
+typedef struct msnd_pinnacle_cfg_device {
+ WORD io0, io1, irq;
+ int mem;
+} msnd_pinnacle_cfg_t[4];
+
+static int __init msnd_pinnacle_cfg_devices(int cfg, int reset, msnd_pinnacle_cfg_t device)
+{
+ int i;
+
+ /* Reset devices if told to */
+ if (reset) {
+ printk(KERN_INFO LOGNAME ": Resetting all devices\n");
+ for (i = 0; i < 4; ++i)
+ if (msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0))
+ return -EIO;
+ }
+
+ /* Configure specified devices */
+ for (i = 0; i < 4; ++i) {
+
+ switch (i) {
+ case 0: /* DSP */
+ if (!(device[i].io0 && device[i].irq && device[i].mem))
+ continue;
+ break;
+ case 1: /* MPU */
+ if (!(device[i].io0 && device[i].irq))
+ continue;
+ printk(KERN_INFO LOGNAME
+ ": Configuring MPU to I/O 0x%x IRQ %d\n",
+ device[i].io0, device[i].irq);
+ break;
+ case 2: /* IDE */
+ if (!(device[i].io0 && device[i].io1 && device[i].irq))
+ continue;
+ printk(KERN_INFO LOGNAME
+ ": Configuring IDE to I/O 0x%x, 0x%x IRQ %d\n",
+ device[i].io0, device[i].io1, device[i].irq);
+ break;
+ case 3: /* Joystick */
+ if (!(device[i].io0))
+ continue;
+ printk(KERN_INFO LOGNAME
+ ": Configuring joystick to I/O 0x%x\n",
+ device[i].io0);
+ break;
+ }
+
+ /* Configure the device */
+ if (msnd_write_cfg_logical(cfg, i, device[i].io0, device[i].io1, device[i].irq, device[i].mem))
+ return -EIO;
+ }
+
+ return 0;
+}
+#endif
+
+#ifdef MODULE
+MODULE_AUTHOR ("Andrew Veliath <andrewtv@usa.net>");
+MODULE_DESCRIPTION ("Turtle Beach " LONGNAME " Linux Driver");
+MODULE_LICENSE("GPL");
+
+static int io __initdata = -1;
+static int irq __initdata = -1;
+static int mem __initdata = -1;
+static int write_ndelay __initdata = -1;
+
+#ifndef MSND_CLASSIC
+/* Pinnacle/Fiji non-PnP Config Port */
+static int cfg __initdata = -1;
+
+/* Extra Peripheral Configuration */
+static int reset __initdata = 0;
+static int mpu_io __initdata = 0;
+static int mpu_irq __initdata = 0;
+static int ide_io0 __initdata = 0;
+static int ide_io1 __initdata = 0;
+static int ide_irq __initdata = 0;
+static int joystick_io __initdata = 0;
+
+/* If we have the digital daugherboard... */
+static int digital __initdata = 0;
+#endif
+
+static int fifosize __initdata = DEFFIFOSIZE;
+static int calibrate_signal __initdata = 0;
+
+#else /* not a module */
+
+static int write_ndelay __initdata = -1;
+
+#ifdef MSND_CLASSIC
+static int io __initdata = CONFIG_MSNDCLAS_IO;
+static int irq __initdata = CONFIG_MSNDCLAS_IRQ;
+static int mem __initdata = CONFIG_MSNDCLAS_MEM;
+#else /* Pinnacle/Fiji */
+
+static int io __initdata = CONFIG_MSNDPIN_IO;
+static int irq __initdata = CONFIG_MSNDPIN_IRQ;
+static int mem __initdata = CONFIG_MSNDPIN_MEM;
+
+/* Pinnacle/Fiji non-PnP Config Port */
+#ifdef CONFIG_MSNDPIN_NONPNP
+# ifndef CONFIG_MSNDPIN_CFG
+# define CONFIG_MSNDPIN_CFG 0x250
+# endif
+#else
+# ifdef CONFIG_MSNDPIN_CFG
+# undef CONFIG_MSNDPIN_CFG
+# endif
+# define CONFIG_MSNDPIN_CFG -1
+#endif
+static int cfg __initdata = CONFIG_MSNDPIN_CFG;
+/* If not a module, we don't need to bother with reset=1 */
+static int reset;
+
+/* Extra Peripheral Configuration (Default: Disable) */
+#ifndef CONFIG_MSNDPIN_MPU_IO
+# define CONFIG_MSNDPIN_MPU_IO 0
+#endif
+static int mpu_io __initdata = CONFIG_MSNDPIN_MPU_IO;
+
+#ifndef CONFIG_MSNDPIN_MPU_IRQ
+# define CONFIG_MSNDPIN_MPU_IRQ 0
+#endif
+static int mpu_irq __initdata = CONFIG_MSNDPIN_MPU_IRQ;
+
+#ifndef CONFIG_MSNDPIN_IDE_IO0
+# define CONFIG_MSNDPIN_IDE_IO0 0
+#endif
+static int ide_io0 __initdata = CONFIG_MSNDPIN_IDE_IO0;
+
+#ifndef CONFIG_MSNDPIN_IDE_IO1
+# define CONFIG_MSNDPIN_IDE_IO1 0
+#endif
+static int ide_io1 __initdata = CONFIG_MSNDPIN_IDE_IO1;
+
+#ifndef CONFIG_MSNDPIN_IDE_IRQ
+# define CONFIG_MSNDPIN_IDE_IRQ 0
+#endif
+static int ide_irq __initdata = CONFIG_MSNDPIN_IDE_IRQ;
+
+#ifndef CONFIG_MSNDPIN_JOYSTICK_IO
+# define CONFIG_MSNDPIN_JOYSTICK_IO 0
+#endif
+static int joystick_io __initdata = CONFIG_MSNDPIN_JOYSTICK_IO;
+
+/* Have SPDIF (Digital) Daughterboard */
+#ifndef CONFIG_MSNDPIN_DIGITAL
+# define CONFIG_MSNDPIN_DIGITAL 0
+#endif
+static int digital __initdata = CONFIG_MSNDPIN_DIGITAL;
+
+#endif /* MSND_CLASSIC */
+
+#ifndef CONFIG_MSND_FIFOSIZE
+# define CONFIG_MSND_FIFOSIZE DEFFIFOSIZE
+#endif
+static int fifosize __initdata = CONFIG_MSND_FIFOSIZE;
+
+#ifndef CONFIG_MSND_CALSIGNAL
+# define CONFIG_MSND_CALSIGNAL 0
+#endif
+static int
+calibrate_signal __initdata = CONFIG_MSND_CALSIGNAL;
+#endif /* MODULE */
+
+module_param (io, int, 0);
+module_param (irq, int, 0);
+module_param (mem, int, 0);
+module_param (write_ndelay, int, 0);
+module_param (fifosize, int, 0);
+module_param (calibrate_signal, int, 0);
+#ifndef MSND_CLASSIC
+module_param (digital, bool, 0);
+module_param (cfg, int, 0);
+module_param (reset, int, 0);
+module_param (mpu_io, int, 0);
+module_param (mpu_irq, int, 0);
+module_param (ide_io0, int, 0);
+module_param (ide_io1, int, 0);
+module_param (ide_irq, int, 0);
+module_param (joystick_io, int, 0);
+#endif
+
+static int __init msnd_init(void)
+{
+ int err;
+#ifndef MSND_CLASSIC
+ static msnd_pinnacle_cfg_t pinnacle_devs;
+#endif /* MSND_CLASSIC */
+
+ printk(KERN_INFO LOGNAME ": Turtle Beach " LONGNAME " Linux Driver Version "
+ VERSION ", Copyright (C) 1998 Andrew Veliath\n");
+
+ if (io == -1 || irq == -1 || mem == -1)
+ printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n");
+
+#ifdef MSND_CLASSIC
+ if (io == -1 ||
+ !(io == 0x290 ||
+ io == 0x260 ||
+ io == 0x250 ||
+ io == 0x240 ||
+ io == 0x230 ||
+ io == 0x220 ||
+ io == 0x210 ||
+ io == 0x3e0)) {
+ printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, or 0x3E0\n");
+ return -EINVAL;
+ }
+#else
+ if (io == -1 ||
+ io < 0x100 ||
+ io > 0x3e0 ||
+ (io % 0x10) != 0) {
+ printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must within the range 0x100 to 0x3E0 and must be evenly divisible by 0x10\n");
+ return -EINVAL;
+ }
+#endif /* MSND_CLASSIC */
+
+ if (irq == -1 ||
+ !(irq == 5 ||
+ irq == 7 ||
+ irq == 9 ||
+ irq == 10 ||
+ irq == 11 ||
+ irq == 12)) {
+ printk(KERN_ERR LOGNAME ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n");
+ return -EINVAL;
+ }
+
+ if (mem == -1 ||
+ !(mem == 0xb0000 ||
+ mem == 0xc8000 ||
+ mem == 0xd0000 ||
+ mem == 0xd8000 ||
+ mem == 0xe0000 ||
+ mem == 0xe8000)) {
+ printk(KERN_ERR LOGNAME ": \"mem\" - must be set to "
+ "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000\n");
+ return -EINVAL;
+ }
+
+#ifdef MSND_CLASSIC
+ switch (irq) {
+ case 5: dev.irqid = HPIRQ_5; break;
+ case 7: dev.irqid = HPIRQ_7; break;
+ case 9: dev.irqid = HPIRQ_9; break;
+ case 10: dev.irqid = HPIRQ_10; break;
+ case 11: dev.irqid = HPIRQ_11; break;
+ case 12: dev.irqid = HPIRQ_12; break;
+ }
+
+ switch (mem) {
+ case 0xb0000: dev.memid = HPMEM_B000; break;
+ case 0xc8000: dev.memid = HPMEM_C800; break;
+ case 0xd0000: dev.memid = HPMEM_D000; break;
+ case 0xd8000: dev.memid = HPMEM_D800; break;
+ case 0xe0000: dev.memid = HPMEM_E000; break;
+ case 0xe8000: dev.memid = HPMEM_E800; break;
+ }
+#else
+ if (cfg == -1) {
+ printk(KERN_INFO LOGNAME ": Assuming PnP mode\n");
+ } else if (cfg != 0x250 && cfg != 0x260 && cfg != 0x270) {
+ printk(KERN_INFO LOGNAME ": Config port must be 0x250, 0x260 or 0x270 (or unspecified for PnP mode)\n");
+ return -EINVAL;
+ } else {
+ printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%x\n", cfg);
+
+ /* DSP */
+ pinnacle_devs[0].io0 = io;
+ pinnacle_devs[0].irq = irq;
+ pinnacle_devs[0].mem = mem;
+
+ /* The following are Pinnacle specific */
+
+ /* MPU */
+ pinnacle_devs[1].io0 = mpu_io;
+ pinnacle_devs[1].irq = mpu_irq;
+
+ /* IDE */
+ pinnacle_devs[2].io0 = ide_io0;
+ pinnacle_devs[2].io1 = ide_io1;
+ pinnacle_devs[2].irq = ide_irq;
+
+ /* Joystick */
+ pinnacle_devs[3].io0 = joystick_io;
+
+ if (!request_region(cfg, 2, "Pinnacle/Fiji Config")) {
+ printk(KERN_ERR LOGNAME ": Config port 0x%x conflict\n", cfg);
+ return -EIO;
+ }
+
+ if (msnd_pinnacle_cfg_devices(cfg, reset, pinnacle_devs)) {
+ printk(KERN_ERR LOGNAME ": Device configuration error\n");
+ release_region(cfg, 2);
+ return -EIO;
+ }
+ release_region(cfg, 2);
+ }
+#endif /* MSND_CLASSIC */
+
+ if (fifosize < 16)
+ fifosize = 16;
+
+ if (fifosize > 1024)
+ fifosize = 1024;
+
+ set_default_audio_parameters();
+#ifdef MSND_CLASSIC
+ dev.type = msndClassic;
+#else
+ dev.type = msndPinnacle;
+#endif
+ dev.io = io;
+ dev.numio = DSP_NUMIO;
+ dev.irq = irq;
+ dev.base = ioremap(mem, 0x8000);
+ dev.fifosize = fifosize * 1024;
+ dev.calibrate_signal = calibrate_signal ? 1 : 0;
+ dev.recsrc = 0;
+ dev.dspq_data_buff = DSPQ_DATA_BUFF;
+ dev.dspq_buff_size = DSPQ_BUFF_SIZE;
+ if (write_ndelay == -1)
+ write_ndelay = CONFIG_MSND_WRITE_NDELAY;
+ if (write_ndelay)
+ clear_bit(F_DISABLE_WRITE_NDELAY, &dev.flags);
+ else
+ set_bit(F_DISABLE_WRITE_NDELAY, &dev.flags);
+#ifndef MSND_CLASSIC
+ if (digital)
+ set_bit(F_HAVEDIGITAL, &dev.flags);
+#endif
+ init_waitqueue_head(&dev.writeblock);
+ init_waitqueue_head(&dev.readblock);
+ init_waitqueue_head(&dev.writeflush);
+ msnd_fifo_init(&dev.DAPF);
+ msnd_fifo_init(&dev.DARF);
+ spin_lock_init(&dev.lock);
+ printk(KERN_INFO LOGNAME ": %u byte audio FIFOs (x2)\n", dev.fifosize);
+ if ((err = msnd_fifo_alloc(&dev.DAPF, dev.fifosize)) < 0) {
+ printk(KERN_ERR LOGNAME ": Couldn't allocate write FIFO\n");
+ return err;
+ }
+
+ if ((err = msnd_fifo_alloc(&dev.DARF, dev.fifosize)) < 0) {
+ printk(KERN_ERR LOGNAME ": Couldn't allocate read FIFO\n");
+ msnd_fifo_free(&dev.DAPF);
+ return err;
+ }
+
+ if ((err = probe_multisound()) < 0) {
+ printk(KERN_ERR LOGNAME ": Probe failed\n");
+ msnd_fifo_free(&dev.DAPF);
+ msnd_fifo_free(&dev.DARF);
+ return err;
+ }
+
+ if ((err = attach_multisound()) < 0) {
+ printk(KERN_ERR LOGNAME ": Attach failed\n");
+ msnd_fifo_free(&dev.DAPF);
+ msnd_fifo_free(&dev.DARF);
+ return err;
+ }
+
+ return 0;
+}
+
+static void __exit msdn_cleanup(void)
+{
+ unload_multisound();
+ msnd_fifo_free(&dev.DAPF);
+ msnd_fifo_free(&dev.DARF);
+}
+
+module_init(msnd_init);
+module_exit(msdn_cleanup);
diff --git a/sound/oss/msnd_pinnacle.h b/sound/oss/msnd_pinnacle.h
new file mode 100644
index 0000000..c18d66c
--- /dev/null
+++ b/sound/oss/msnd_pinnacle.h
@@ -0,0 +1,246 @@
+/*********************************************************************
+ *
+ * msnd_pinnacle.h
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Some parts of this header file were derived from the Turtle Beach
+ * MultiSound Driver Development Kit.
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ * Copyright (C) 1993 Turtle Beach Systems, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+#ifndef __MSND_PINNACLE_H
+#define __MSND_PINNACLE_H
+
+
+#define DSP_NUMIO 0x08
+
+#define IREG_LOGDEVICE 0x07
+#define IREG_ACTIVATE 0x30
+#define LD_ACTIVATE 0x01
+#define LD_DISACTIVATE 0x00
+#define IREG_EECONTROL 0x3F
+#define IREG_MEMBASEHI 0x40
+#define IREG_MEMBASELO 0x41
+#define IREG_MEMCONTROL 0x42
+#define IREG_MEMRANGEHI 0x43
+#define IREG_MEMRANGELO 0x44
+#define MEMTYPE_8BIT 0x00
+#define MEMTYPE_16BIT 0x02
+#define MEMTYPE_RANGE 0x00
+#define MEMTYPE_HIADDR 0x01
+#define IREG_IO0_BASEHI 0x60
+#define IREG_IO0_BASELO 0x61
+#define IREG_IO1_BASEHI 0x62
+#define IREG_IO1_BASELO 0x63
+#define IREG_IRQ_NUMBER 0x70
+#define IREG_IRQ_TYPE 0x71
+#define IRQTYPE_HIGH 0x02
+#define IRQTYPE_LOW 0x00
+#define IRQTYPE_LEVEL 0x01
+#define IRQTYPE_EDGE 0x00
+
+#define HP_DSPR 0x04
+#define HP_BLKS 0x04
+
+#define HPDSPRESET_OFF 2
+#define HPDSPRESET_ON 0
+
+#define HPBLKSEL_0 2
+#define HPBLKSEL_1 3
+
+#define HIMT_DAT_OFF 0x03
+
+#define HIDSP_PLAY_UNDER 0x00
+#define HIDSP_INT_PLAY_UNDER 0x01
+#define HIDSP_SSI_TX_UNDER 0x02
+#define HIDSP_RECQ_OVERFLOW 0x08
+#define HIDSP_INT_RECORD_OVER 0x09
+#define HIDSP_SSI_RX_OVERFLOW 0x0a
+
+#define HIDSP_MIDI_IN_OVER 0x10
+
+#define HIDSP_MIDI_FRAME_ERR 0x11
+#define HIDSP_MIDI_PARITY_ERR 0x12
+#define HIDSP_MIDI_OVERRUN_ERR 0x13
+
+#define HIDSP_INPUT_CLIPPING 0x20
+#define HIDSP_MIX_CLIPPING 0x30
+#define HIDSP_DAT_IN_OFF 0x21
+
+#define HDEXAR_SET_ANA_IN 0
+#define HDEXAR_CLEAR_PEAKS 1
+#define HDEXAR_IN_SET_POTS 2
+#define HDEXAR_AUX_SET_POTS 3
+#define HDEXAR_CAL_A_TO_D 4
+#define HDEXAR_RD_EXT_DSP_BITS 5
+
+#define HDEXAR_SET_SYNTH_IN 4
+#define HDEXAR_READ_DAT_IN 5
+#define HDEXAR_MIC_SET_POTS 6
+#define HDEXAR_SET_DAT_IN 7
+
+#define HDEXAR_SET_SYNTH_48 8
+#define HDEXAR_SET_SYNTH_44 9
+
+#define TIME_PRO_RESET_DONE 0x028A
+#define TIME_PRO_SYSEX 0x001E
+#define TIME_PRO_RESET 0x0032
+
+#define AGND 0x01
+#define SIGNAL 0x02
+
+#define EXT_DSP_BIT_DCAL 0x0001
+#define EXT_DSP_BIT_MIDI_CON 0x0002
+
+#define BUFFSIZE 0x8000
+#define HOSTQ_SIZE 0x40
+
+#define SRAM_CNTL_START 0x7F00
+#define SMA_STRUCT_START 0x7F40
+
+#define DAP_BUFF_SIZE 0x2400
+#define DAR_BUFF_SIZE 0x2000
+
+#define DAPQ_STRUCT_SIZE 0x10
+#define DARQ_STRUCT_SIZE 0x10
+#define DAPQ_BUFF_SIZE (3 * 0x10)
+#define DARQ_BUFF_SIZE (3 * 0x10)
+#define MODQ_BUFF_SIZE 0x400
+#define MIDQ_BUFF_SIZE 0x800
+#define DSPQ_BUFF_SIZE 0x5A0
+
+#define DAPQ_DATA_BUFF 0x6C00
+#define DARQ_DATA_BUFF 0x6C30
+#define MODQ_DATA_BUFF 0x6C60
+#define MIDQ_DATA_BUFF 0x7060
+#define DSPQ_DATA_BUFF 0x7860
+
+#define DAPQ_OFFSET SRAM_CNTL_START
+#define DARQ_OFFSET (SRAM_CNTL_START + 0x08)
+#define MODQ_OFFSET (SRAM_CNTL_START + 0x10)
+#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18)
+#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20)
+
+#define MOP_WAVEHDR 0
+#define MOP_EXTOUT 1
+#define MOP_HWINIT 0xfe
+#define MOP_NONE 0xff
+#define MOP_MAX 1
+
+#define MIP_EXTIN 0
+#define MIP_WAVEHDR 1
+#define MIP_HWINIT 0xfe
+#define MIP_MAX 1
+
+/* Pinnacle/Fiji SMA Common Data */
+#define SMA_wCurrPlayBytes 0x0000
+#define SMA_wCurrRecordBytes 0x0002
+#define SMA_wCurrPlayVolLeft 0x0004
+#define SMA_wCurrPlayVolRight 0x0006
+#define SMA_wCurrInVolLeft 0x0008
+#define SMA_wCurrInVolRight 0x000a
+#define SMA_wCurrMHdrVolLeft 0x000c
+#define SMA_wCurrMHdrVolRight 0x000e
+#define SMA_dwCurrPlayPitch 0x0010
+#define SMA_dwCurrPlayRate 0x0014
+#define SMA_wCurrMIDIIOPatch 0x0018
+#define SMA_wCurrPlayFormat 0x001a
+#define SMA_wCurrPlaySampleSize 0x001c
+#define SMA_wCurrPlayChannels 0x001e
+#define SMA_wCurrPlaySampleRate 0x0020
+#define SMA_wCurrRecordFormat 0x0022
+#define SMA_wCurrRecordSampleSize 0x0024
+#define SMA_wCurrRecordChannels 0x0026
+#define SMA_wCurrRecordSampleRate 0x0028
+#define SMA_wCurrDSPStatusFlags 0x002a
+#define SMA_wCurrHostStatusFlags 0x002c
+#define SMA_wCurrInputTagBits 0x002e
+#define SMA_wCurrLeftPeak 0x0030
+#define SMA_wCurrRightPeak 0x0032
+#define SMA_bMicPotPosLeft 0x0034
+#define SMA_bMicPotPosRight 0x0035
+#define SMA_bMicPotMaxLeft 0x0036
+#define SMA_bMicPotMaxRight 0x0037
+#define SMA_bInPotPosLeft 0x0038
+#define SMA_bInPotPosRight 0x0039
+#define SMA_bAuxPotPosLeft 0x003a
+#define SMA_bAuxPotPosRight 0x003b
+#define SMA_bInPotMaxLeft 0x003c
+#define SMA_bInPotMaxRight 0x003d
+#define SMA_bAuxPotMaxLeft 0x003e
+#define SMA_bAuxPotMaxRight 0x003f
+#define SMA_bInPotMaxMethod 0x0040
+#define SMA_bAuxPotMaxMethod 0x0041
+#define SMA_wCurrMastVolLeft 0x0042
+#define SMA_wCurrMastVolRight 0x0044
+#define SMA_wCalFreqAtoD 0x0046
+#define SMA_wCurrAuxVolLeft 0x0048
+#define SMA_wCurrAuxVolRight 0x004a
+#define SMA_wCurrPlay1VolLeft 0x004c
+#define SMA_wCurrPlay1VolRight 0x004e
+#define SMA_wCurrPlay2VolLeft 0x0050
+#define SMA_wCurrPlay2VolRight 0x0052
+#define SMA_wCurrPlay3VolLeft 0x0054
+#define SMA_wCurrPlay3VolRight 0x0056
+#define SMA_wCurrPlay4VolLeft 0x0058
+#define SMA_wCurrPlay4VolRight 0x005a
+#define SMA_wCurrPlay1PeakLeft 0x005c
+#define SMA_wCurrPlay1PeakRight 0x005e
+#define SMA_wCurrPlay2PeakLeft 0x0060
+#define SMA_wCurrPlay2PeakRight 0x0062
+#define SMA_wCurrPlay3PeakLeft 0x0064
+#define SMA_wCurrPlay3PeakRight 0x0066
+#define SMA_wCurrPlay4PeakLeft 0x0068
+#define SMA_wCurrPlay4PeakRight 0x006a
+#define SMA_wCurrPlayPeakLeft 0x006c
+#define SMA_wCurrPlayPeakRight 0x006e
+#define SMA_wCurrDATSR 0x0070
+#define SMA_wCurrDATRXCHNL 0x0072
+#define SMA_wCurrDATTXCHNL 0x0074
+#define SMA_wCurrDATRXRate 0x0076
+#define SMA_dwDSPPlayCount 0x0078
+#define SMA__size 0x007c
+
+#ifdef HAVE_DSPCODEH
+# include "pndsperm.c"
+# include "pndspini.c"
+# define PERMCODE pndsperm
+# define INITCODE pndspini
+# define PERMCODESIZE sizeof(pndsperm)
+# define INITCODESIZE sizeof(pndspini)
+#else
+# ifndef CONFIG_MSNDPIN_INIT_FILE
+# define CONFIG_MSNDPIN_INIT_FILE \
+ "/etc/sound/pndspini.bin"
+# endif
+# ifndef CONFIG_MSNDPIN_PERM_FILE
+# define CONFIG_MSNDPIN_PERM_FILE \
+ "/etc/sound/pndsperm.bin"
+# endif
+# define PERMCODEFILE CONFIG_MSNDPIN_PERM_FILE
+# define INITCODEFILE CONFIG_MSNDPIN_INIT_FILE
+# define PERMCODE dspini
+# define INITCODE permini
+# define PERMCODESIZE sizeof_dspini
+# define INITCODESIZE sizeof_permini
+#endif
+#define LONGNAME "MultiSound (Pinnacle/Fiji)"
+
+#endif /* __MSND_PINNACLE_H */
diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c
new file mode 100644
index 0000000..7781c13
--- /dev/null
+++ b/sound/oss/opl3.c
@@ -0,0 +1,1250 @@
+/*
+ * sound/oss/opl3.c
+ *
+ * A low level driver for Yamaha YM3812 and OPL-3 -chips
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Changes
+ * Thomas Sailer ioctl code reworked (vmalloc/vfree removed)
+ * Alan Cox modularisation, fixed sound_mem allocs.
+ * Christoph Hellwig Adapted to module_init/module_exit
+ * Arnaldo C. de Melo get rid of check_region, use request_region for
+ * OPL4, release it on exit, some cleanups.
+ *
+ * Status
+ * Believed to work. Badly needs rewriting a bit to support multiple
+ * OPL3 devices.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+
+/*
+ * Major improvements to the FM handling 30AUG92 by Rob Hooft,
+ * hooft@chem.ruu.nl
+ */
+
+#include "sound_config.h"
+
+#include "opl3_hw.h"
+
+#define MAX_VOICE 18
+#define OFFS_4OP 11
+
+struct voice_info
+{
+ unsigned char keyon_byte;
+ long bender;
+ long bender_range;
+ unsigned long orig_freq;
+ unsigned long current_freq;
+ int volume;
+ int mode;
+ int panning; /* 0xffff means not set */
+};
+
+typedef struct opl_devinfo
+{
+ int base;
+ int left_io, right_io;
+ int nr_voice;
+ int lv_map[MAX_VOICE];
+
+ struct voice_info voc[MAX_VOICE];
+ struct voice_alloc_info *v_alloc;
+ struct channel_info *chn_info;
+
+ struct sbi_instrument i_map[SBFM_MAXINSTR];
+ struct sbi_instrument *act_i[MAX_VOICE];
+
+ struct synth_info fm_info;
+
+ int busy;
+ int model;
+ unsigned char cmask;
+
+ int is_opl4;
+} opl_devinfo;
+
+static struct opl_devinfo *devc = NULL;
+
+static int detected_model;
+
+static int store_instr(int instr_no, struct sbi_instrument *instr);
+static void freq_to_fnum(int freq, int *block, int *fnum);
+static void opl3_command(int io_addr, unsigned int addr, unsigned int val);
+static int opl3_kill_note(int dev, int voice, int note, int velocity);
+
+static void enter_4op_mode(void)
+{
+ int i;
+ static int v4op[MAX_VOICE] = {
+ 0, 1, 2, 9, 10, 11, 6, 7, 8, 15, 16, 17
+ };
+
+ devc->cmask = 0x3f; /* Connect all possible 4 OP voice operators */
+ opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, 0x3f);
+
+ for (i = 0; i < 3; i++)
+ pv_map[i].voice_mode = 4;
+ for (i = 3; i < 6; i++)
+ pv_map[i].voice_mode = 0;
+
+ for (i = 9; i < 12; i++)
+ pv_map[i].voice_mode = 4;
+ for (i = 12; i < 15; i++)
+ pv_map[i].voice_mode = 0;
+
+ for (i = 0; i < 12; i++)
+ devc->lv_map[i] = v4op[i];
+ devc->v_alloc->max_voice = devc->nr_voice = 12;
+}
+
+static int opl3_ioctl(int dev, unsigned int cmd, void __user * arg)
+{
+ struct sbi_instrument ins;
+
+ switch (cmd) {
+ case SNDCTL_FM_LOAD_INSTR:
+ printk(KERN_WARNING "Warning: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n");
+ if (copy_from_user(&ins, arg, sizeof(ins)))
+ return -EFAULT;
+ if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR) {
+ printk(KERN_WARNING "FM Error: Invalid instrument number %d\n", ins.channel);
+ return -EINVAL;
+ }
+ return store_instr(ins.channel, &ins);
+
+ case SNDCTL_SYNTH_INFO:
+ devc->fm_info.nr_voices = (devc->nr_voice == 12) ? 6 : devc->nr_voice;
+ if (copy_to_user(arg, &devc->fm_info, sizeof(devc->fm_info)))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_SYNTH_MEMAVL:
+ return 0x7fffffff;
+
+ case SNDCTL_FM_4OP_ENABLE:
+ if (devc->model == 2)
+ enter_4op_mode();
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static int opl3_detect(int ioaddr)
+{
+ /*
+ * This function returns 1 if the FM chip is present at the given I/O port
+ * The detection algorithm plays with the timer built in the FM chip and
+ * looks for a change in the status register.
+ *
+ * Note! The timers of the FM chip are not connected to AdLib (and compatible)
+ * boards.
+ *
+ * Note2! The chip is initialized if detected.
+ */
+
+ unsigned char stat1, signature;
+ int i;
+
+ if (devc != NULL)
+ {
+ printk(KERN_ERR "opl3: Only one OPL3 supported.\n");
+ return 0;
+ }
+
+ devc = kzalloc(sizeof(*devc), GFP_KERNEL);
+
+ if (devc == NULL)
+ {
+ printk(KERN_ERR "opl3: Can't allocate memory for the device control "
+ "structure \n ");
+ return 0;
+ }
+
+ strcpy(devc->fm_info.name, "OPL2");
+
+ if (!request_region(ioaddr, 4, devc->fm_info.name)) {
+ printk(KERN_WARNING "opl3: I/O port 0x%x already in use\n", ioaddr);
+ goto cleanup_devc;
+ }
+
+ devc->base = ioaddr;
+
+ /* Reset timers 1 and 2 */
+ opl3_command(ioaddr, TIMER_CONTROL_REGISTER, TIMER1_MASK | TIMER2_MASK);
+
+ /* Reset the IRQ of the FM chip */
+ opl3_command(ioaddr, TIMER_CONTROL_REGISTER, IRQ_RESET);
+
+ signature = stat1 = inb(ioaddr); /* Status register */
+
+ if (signature != 0x00 && signature != 0x06 && signature != 0x02 &&
+ signature != 0x0f)
+ {
+ MDB(printk(KERN_INFO "OPL3 not detected %x\n", signature));
+ goto cleanup_region;
+ }
+
+ if (signature == 0x06) /* OPL2 */
+ {
+ detected_model = 2;
+ }
+ else if (signature == 0x00 || signature == 0x0f) /* OPL3 or OPL4 */
+ {
+ unsigned char tmp;
+
+ detected_model = 3;
+
+ /*
+ * Detect availability of OPL4 (_experimental_). Works probably
+ * only after a cold boot. In addition the OPL4 port
+ * of the chip may not be connected to the PC bus at all.
+ */
+
+ opl3_command(ioaddr + 2, OPL3_MODE_REGISTER, 0x00);
+ opl3_command(ioaddr + 2, OPL3_MODE_REGISTER, OPL3_ENABLE | OPL4_ENABLE);
+
+ if ((tmp = inb(ioaddr)) == 0x02) /* Have a OPL4 */
+ {
+ detected_model = 4;
+ }
+
+ if (request_region(ioaddr - 8, 2, "OPL4")) /* OPL4 port was free */
+ {
+ int tmp;
+
+ outb((0x02), ioaddr - 8); /* Select OPL4 ID register */
+ udelay(10);
+ tmp = inb(ioaddr - 7); /* Read it */
+ udelay(10);
+
+ if (tmp == 0x20) /* OPL4 should return 0x20 here */
+ {
+ detected_model = 4;
+ outb((0xF8), ioaddr - 8); /* Select OPL4 FM mixer control */
+ udelay(10);
+ outb((0x1B), ioaddr - 7); /* Write value */
+ udelay(10);
+ }
+ else
+ { /* release OPL4 port */
+ release_region(ioaddr - 8, 2);
+ detected_model = 3;
+ }
+ }
+ opl3_command(ioaddr + 2, OPL3_MODE_REGISTER, 0);
+ }
+ for (i = 0; i < 9; i++)
+ opl3_command(ioaddr, KEYON_BLOCK + i, 0); /*
+ * Note off
+ */
+
+ opl3_command(ioaddr, TEST_REGISTER, ENABLE_WAVE_SELECT);
+ opl3_command(ioaddr, PERCOSSION_REGISTER, 0x00); /*
+ * Melodic mode.
+ */
+ return 1;
+cleanup_region:
+ release_region(ioaddr, 4);
+cleanup_devc:
+ kfree(devc);
+ devc = NULL;
+ return 0;
+}
+
+static int opl3_kill_note (int devno, int voice, int note, int velocity)
+{
+ struct physical_voice_info *map;
+
+ if (voice < 0 || voice >= devc->nr_voice)
+ return 0;
+
+ devc->v_alloc->map[voice] = 0;
+
+ map = &pv_map[devc->lv_map[voice]];
+ DEB(printk("Kill note %d\n", voice));
+
+ if (map->voice_mode == 0)
+ return 0;
+
+ opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, devc->voc[voice].keyon_byte & ~0x20);
+ devc->voc[voice].keyon_byte = 0;
+ devc->voc[voice].bender = 0;
+ devc->voc[voice].volume = 64;
+ devc->voc[voice].panning = 0xffff; /* Not set */
+ devc->voc[voice].bender_range = 200;
+ devc->voc[voice].orig_freq = 0;
+ devc->voc[voice].current_freq = 0;
+ devc->voc[voice].mode = 0;
+ return 0;
+}
+
+#define HIHAT 0
+#define CYMBAL 1
+#define TOMTOM 2
+#define SNARE 3
+#define BDRUM 4
+#define UNDEFINED TOMTOM
+#define DEFAULT TOMTOM
+
+static int store_instr(int instr_no, struct sbi_instrument *instr)
+{
+ if (instr->key != FM_PATCH && (instr->key != OPL3_PATCH || devc->model != 2))
+ printk(KERN_WARNING "FM warning: Invalid patch format field (key) 0x%x\n", instr->key);
+ memcpy((char *) &(devc->i_map[instr_no]), (char *) instr, sizeof(*instr));
+ return 0;
+}
+
+static int opl3_set_instr (int dev, int voice, int instr_no)
+{
+ if (voice < 0 || voice >= devc->nr_voice)
+ return 0;
+ if (instr_no < 0 || instr_no >= SBFM_MAXINSTR)
+ instr_no = 0; /* Acoustic piano (usually) */
+
+ devc->act_i[voice] = &devc->i_map[instr_no];
+ return 0;
+}
+
+/*
+ * The next table looks magical, but it certainly is not. Its values have
+ * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception
+ * for i=0. This log-table converts a linear volume-scaling (0..127) to a
+ * logarithmic scaling as present in the FM-synthesizer chips. so : Volume
+ * 64 = 0 db = relative volume 0 and: Volume 32 = -6 db = relative
+ * volume -8 it was implemented as a table because it is only 128 bytes and
+ * it saves a lot of log() calculations. (RH)
+ */
+
+static char fm_volume_table[128] =
+{
+ -64, -48, -40, -35, -32, -29, -27, -26,
+ -24, -23, -21, -20, -19, -18, -18, -17,
+ -16, -15, -15, -14, -13, -13, -12, -12,
+ -11, -11, -10, -10, -10, -9, -9, -8,
+ -8, -8, -7, -7, -7, -6, -6, -6,
+ -5, -5, -5, -5, -4, -4, -4, -4,
+ -3, -3, -3, -3, -2, -2, -2, -2,
+ -2, -1, -1, -1, -1, 0, 0, 0,
+ 0, 0, 0, 1, 1, 1, 1, 1,
+ 1, 2, 2, 2, 2, 2, 2, 2,
+ 3, 3, 3, 3, 3, 3, 3, 4,
+ 4, 4, 4, 4, 4, 4, 4, 5,
+ 5, 5, 5, 5, 5, 5, 5, 5,
+ 6, 6, 6, 6, 6, 6, 6, 6,
+ 6, 7, 7, 7, 7, 7, 7, 7,
+ 7, 7, 7, 8, 8, 8, 8, 8
+};
+
+static void calc_vol(unsigned char *regbyte, int volume, int main_vol)
+{
+ int level = (~*regbyte & 0x3f);
+
+ if (main_vol > 127)
+ main_vol = 127;
+ volume = (volume * main_vol) / 127;
+
+ if (level)
+ level += fm_volume_table[volume];
+
+ if (level > 0x3f)
+ level = 0x3f;
+ if (level < 0)
+ level = 0;
+
+ *regbyte = (*regbyte & 0xc0) | (~level & 0x3f);
+}
+
+static void set_voice_volume(int voice, int volume, int main_vol)
+{
+ unsigned char vol1, vol2, vol3, vol4;
+ struct sbi_instrument *instr;
+ struct physical_voice_info *map;
+
+ if (voice < 0 || voice >= devc->nr_voice)
+ return;
+
+ map = &pv_map[devc->lv_map[voice]];
+ instr = devc->act_i[voice];
+
+ if (!instr)
+ instr = &devc->i_map[0];
+
+ if (instr->channel < 0)
+ return;
+
+ if (devc->voc[voice].mode == 0)
+ return;
+
+ if (devc->voc[voice].mode == 2)
+ {
+ vol1 = instr->operators[2];
+ vol2 = instr->operators[3];
+ if ((instr->operators[10] & 0x01))
+ {
+ calc_vol(&vol1, volume, main_vol);
+ calc_vol(&vol2, volume, main_vol);
+ }
+ else
+ {
+ calc_vol(&vol2, volume, main_vol);
+ }
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[0], vol1);
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[1], vol2);
+ }
+ else
+ { /*
+ * 4 OP voice
+ */
+ int connection;
+
+ vol1 = instr->operators[2];
+ vol2 = instr->operators[3];
+ vol3 = instr->operators[OFFS_4OP + 2];
+ vol4 = instr->operators[OFFS_4OP + 3];
+
+ /*
+ * The connection method for 4 OP devc->voc is defined by the rightmost
+ * bits at the offsets 10 and 10+OFFS_4OP
+ */
+
+ connection = ((instr->operators[10] & 0x01) << 1) | (instr->operators[10 + OFFS_4OP] & 0x01);
+
+ switch (connection)
+ {
+ case 0:
+ calc_vol(&vol4, volume, main_vol);
+ break;
+
+ case 1:
+ calc_vol(&vol2, volume, main_vol);
+ calc_vol(&vol4, volume, main_vol);
+ break;
+
+ case 2:
+ calc_vol(&vol1, volume, main_vol);
+ calc_vol(&vol4, volume, main_vol);
+ break;
+
+ case 3:
+ calc_vol(&vol1, volume, main_vol);
+ calc_vol(&vol3, volume, main_vol);
+ calc_vol(&vol4, volume, main_vol);
+ break;
+
+ default:
+ ;
+ }
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[0], vol1);
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[1], vol2);
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[2], vol3);
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[3], vol4);
+ }
+}
+
+static int opl3_start_note (int dev, int voice, int note, int volume)
+{
+ unsigned char data, fpc;
+ int block, fnum, freq, voice_mode, pan;
+ struct sbi_instrument *instr;
+ struct physical_voice_info *map;
+
+ if (voice < 0 || voice >= devc->nr_voice)
+ return 0;
+
+ map = &pv_map[devc->lv_map[voice]];
+ pan = devc->voc[voice].panning;
+
+ if (map->voice_mode == 0)
+ return 0;
+
+ if (note == 255) /*
+ * Just change the volume
+ */
+ {
+ set_voice_volume(voice, volume, devc->voc[voice].volume);
+ return 0;
+ }
+
+ /*
+ * Kill previous note before playing
+ */
+
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[1], 0xff); /*
+ * Carrier
+ * volume to
+ * min
+ */
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[0], 0xff); /*
+ * Modulator
+ * volume to
+ */
+
+ if (map->voice_mode == 4)
+ {
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[2], 0xff);
+ opl3_command(map->ioaddr, KSL_LEVEL + map->op[3], 0xff);
+ }
+
+ opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, 0x00); /*
+ * Note
+ * off
+ */
+
+ instr = devc->act_i[voice];
+
+ if (!instr)
+ instr = &devc->i_map[0];
+
+ if (instr->channel < 0)
+ {
+ printk(KERN_WARNING "opl3: Initializing voice %d with undefined instrument\n", voice);
+ return 0;
+ }
+
+ if (map->voice_mode == 2 && instr->key == OPL3_PATCH)
+ return 0; /*
+ * Cannot play
+ */
+
+ voice_mode = map->voice_mode;
+
+ if (voice_mode == 4)
+ {
+ int voice_shift;
+
+ voice_shift = (map->ioaddr == devc->left_io) ? 0 : 3;
+ voice_shift += map->voice_num;
+
+ if (instr->key != OPL3_PATCH) /*
+ * Just 2 OP patch
+ */
+ {
+ voice_mode = 2;
+ devc->cmask &= ~(1 << voice_shift);
+ }
+ else
+ {
+ devc->cmask |= (1 << voice_shift);
+ }
+
+ opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, devc->cmask);
+ }
+
+ /*
+ * Set Sound Characteristics
+ */
+
+ opl3_command(map->ioaddr, AM_VIB + map->op[0], instr->operators[0]);
+ opl3_command(map->ioaddr, AM_VIB + map->op[1], instr->operators[1]);
+
+ /*
+ * Set Attack/Decay
+ */
+
+ opl3_command(map->ioaddr, ATTACK_DECAY + map->op[0], instr->operators[4]);
+ opl3_command(map->ioaddr, ATTACK_DECAY + map->op[1], instr->operators[5]);
+
+ /*
+ * Set Sustain/Release
+ */
+
+ opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[0], instr->operators[6]);
+ opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[1], instr->operators[7]);
+
+ /*
+ * Set Wave Select
+ */
+
+ opl3_command(map->ioaddr, WAVE_SELECT + map->op[0], instr->operators[8]);
+ opl3_command(map->ioaddr, WAVE_SELECT + map->op[1], instr->operators[9]);
+
+ /*
+ * Set Feedback/Connection
+ */
+
+ fpc = instr->operators[10];
+
+ if (pan != 0xffff)
+ {
+ fpc &= ~STEREO_BITS;
+ if (pan < -64)
+ fpc |= VOICE_TO_LEFT;
+ else
+ if (pan > 64)
+ fpc |= VOICE_TO_RIGHT;
+ else
+ fpc |= (VOICE_TO_LEFT | VOICE_TO_RIGHT);
+ }
+
+ if (!(fpc & 0x30))
+ fpc |= 0x30; /*
+ * Ensure that at least one chn is enabled
+ */
+ opl3_command(map->ioaddr, FEEDBACK_CONNECTION + map->voice_num, fpc);
+
+ /*
+ * If the voice is a 4 OP one, initialize the operators 3 and 4 also
+ */
+
+ if (voice_mode == 4)
+ {
+ /*
+ * Set Sound Characteristics
+ */
+
+ opl3_command(map->ioaddr, AM_VIB + map->op[2], instr->operators[OFFS_4OP + 0]);
+ opl3_command(map->ioaddr, AM_VIB + map->op[3], instr->operators[OFFS_4OP + 1]);
+
+ /*
+ * Set Attack/Decay
+ */
+
+ opl3_command(map->ioaddr, ATTACK_DECAY + map->op[2], instr->operators[OFFS_4OP + 4]);
+ opl3_command(map->ioaddr, ATTACK_DECAY + map->op[3], instr->operators[OFFS_4OP + 5]);
+
+ /*
+ * Set Sustain/Release
+ */
+
+ opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[2], instr->operators[OFFS_4OP + 6]);
+ opl3_command(map->ioaddr, SUSTAIN_RELEASE + map->op[3], instr->operators[OFFS_4OP + 7]);
+
+ /*
+ * Set Wave Select
+ */
+
+ opl3_command(map->ioaddr, WAVE_SELECT + map->op[2], instr->operators[OFFS_4OP + 8]);
+ opl3_command(map->ioaddr, WAVE_SELECT + map->op[3], instr->operators[OFFS_4OP + 9]);
+
+ /*
+ * Set Feedback/Connection
+ */
+
+ fpc = instr->operators[OFFS_4OP + 10];
+ if (!(fpc & 0x30))
+ fpc |= 0x30; /*
+ * Ensure that at least one chn is enabled
+ */
+ opl3_command(map->ioaddr, FEEDBACK_CONNECTION + map->voice_num + 3, fpc);
+ }
+
+ devc->voc[voice].mode = voice_mode;
+ set_voice_volume(voice, volume, devc->voc[voice].volume);
+
+ freq = devc->voc[voice].orig_freq = note_to_freq(note) / 1000;
+
+ /*
+ * Since the pitch bender may have been set before playing the note, we
+ * have to calculate the bending now.
+ */
+
+ freq = compute_finetune(devc->voc[voice].orig_freq, devc->voc[voice].bender, devc->voc[voice].bender_range, 0);
+ devc->voc[voice].current_freq = freq;
+
+ freq_to_fnum(freq, &block, &fnum);
+
+ /*
+ * Play note
+ */
+
+ data = fnum & 0xff; /*
+ * Least significant bits of fnumber
+ */
+ opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data);
+
+ data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3);
+ devc->voc[voice].keyon_byte = data;
+ opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data);
+ if (voice_mode == 4)
+ opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data);
+
+ return 0;
+}
+
+static void freq_to_fnum (int freq, int *block, int *fnum)
+{
+ int f, octave;
+
+ /*
+ * Converts the note frequency to block and fnum values for the FM chip
+ */
+ /*
+ * First try to compute the block -value (octave) where the note belongs
+ */
+
+ f = freq;
+
+ octave = 5;
+
+ if (f == 0)
+ octave = 0;
+ else if (f < 261)
+ {
+ while (f < 261)
+ {
+ octave--;
+ f <<= 1;
+ }
+ }
+ else if (f > 493)
+ {
+ while (f > 493)
+ {
+ octave++;
+ f >>= 1;
+ }
+ }
+
+ if (octave > 7)
+ octave = 7;
+
+ *fnum = freq * (1 << (20 - octave)) / 49716;
+ *block = octave;
+}
+
+static void opl3_command (int io_addr, unsigned int addr, unsigned int val)
+{
+ int i;
+
+ /*
+ * The original 2-OP synth requires a quite long delay after writing to a
+ * register. The OPL-3 survives with just two INBs
+ */
+
+ outb(((unsigned char) (addr & 0xff)), io_addr);
+
+ if (devc->model != 2)
+ udelay(10);
+ else
+ for (i = 0; i < 2; i++)
+ inb(io_addr);
+
+ outb(((unsigned char) (val & 0xff)), io_addr + 1);
+
+ if (devc->model != 2)
+ udelay(30);
+ else
+ for (i = 0; i < 2; i++)
+ inb(io_addr);
+}
+
+static void opl3_reset(int devno)
+{
+ int i;
+
+ for (i = 0; i < 18; i++)
+ devc->lv_map[i] = i;
+
+ for (i = 0; i < devc->nr_voice; i++)
+ {
+ opl3_command(pv_map[devc->lv_map[i]].ioaddr,
+ KSL_LEVEL + pv_map[devc->lv_map[i]].op[0], 0xff);
+
+ opl3_command(pv_map[devc->lv_map[i]].ioaddr,
+ KSL_LEVEL + pv_map[devc->lv_map[i]].op[1], 0xff);
+
+ if (pv_map[devc->lv_map[i]].voice_mode == 4)
+ {
+ opl3_command(pv_map[devc->lv_map[i]].ioaddr,
+ KSL_LEVEL + pv_map[devc->lv_map[i]].op[2], 0xff);
+
+ opl3_command(pv_map[devc->lv_map[i]].ioaddr,
+ KSL_LEVEL + pv_map[devc->lv_map[i]].op[3], 0xff);
+ }
+
+ opl3_kill_note(devno, i, 0, 64);
+ }
+
+ if (devc->model == 2)
+ {
+ devc->v_alloc->max_voice = devc->nr_voice = 18;
+
+ for (i = 0; i < 18; i++)
+ pv_map[i].voice_mode = 2;
+
+ }
+}
+
+static int opl3_open(int dev, int mode)
+{
+ int i;
+
+ if (devc->busy)
+ return -EBUSY;
+ devc->busy = 1;
+
+ devc->v_alloc->max_voice = devc->nr_voice = (devc->model == 2) ? 18 : 9;
+ devc->v_alloc->timestamp = 0;
+
+ for (i = 0; i < 18; i++)
+ {
+ devc->v_alloc->map[i] = 0;
+ devc->v_alloc->alloc_times[i] = 0;
+ }
+
+ devc->cmask = 0x00; /*
+ * Just 2 OP mode
+ */
+ if (devc->model == 2)
+ opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, devc->cmask);
+ return 0;
+}
+
+static void opl3_close(int dev)
+{
+ devc->busy = 0;
+ devc->v_alloc->max_voice = devc->nr_voice = (devc->model == 2) ? 18 : 9;
+
+ devc->fm_info.nr_drums = 0;
+ devc->fm_info.perc_mode = 0;
+
+ opl3_reset(dev);
+}
+
+static void opl3_hw_control(int dev, unsigned char *event)
+{
+}
+
+static int opl3_load_patch(int dev, int format, const char __user *addr,
+ int offs, int count, int pmgr_flag)
+{
+ struct sbi_instrument ins;
+
+ if (count <sizeof(ins))
+ {
+ printk(KERN_WARNING "FM Error: Patch record too short\n");
+ return -EINVAL;
+ }
+
+ /*
+ * What the fuck is going on here? We leave junk in the beginning
+ * of ins and then check the field pretty close to that beginning?
+ */
+ if(copy_from_user(&((char *) &ins)[offs], addr + offs, sizeof(ins) - offs))
+ return -EFAULT;
+
+ if (ins.channel < 0 || ins.channel >= SBFM_MAXINSTR)
+ {
+ printk(KERN_WARNING "FM Error: Invalid instrument number %d\n", ins.channel);
+ return -EINVAL;
+ }
+ ins.key = format;
+
+ return store_instr(ins.channel, &ins);
+}
+
+static void opl3_panning(int dev, int voice, int value)
+{
+ devc->voc[voice].panning = value;
+}
+
+static void opl3_volume_method(int dev, int mode)
+{
+}
+
+#define SET_VIBRATO(cell) { \
+ tmp = instr->operators[(cell-1)+(((cell-1)/2)*OFFS_4OP)]; \
+ if (pressure > 110) \
+ tmp |= 0x40; /* Vibrato on */ \
+ opl3_command (map->ioaddr, AM_VIB + map->op[cell-1], tmp);}
+
+static void opl3_aftertouch(int dev, int voice, int pressure)
+{
+ int tmp;
+ struct sbi_instrument *instr;
+ struct physical_voice_info *map;
+
+ if (voice < 0 || voice >= devc->nr_voice)
+ return;
+
+ map = &pv_map[devc->lv_map[voice]];
+
+ DEB(printk("Aftertouch %d\n", voice));
+
+ if (map->voice_mode == 0)
+ return;
+
+ /*
+ * Adjust the amount of vibrato depending the pressure
+ */
+
+ instr = devc->act_i[voice];
+
+ if (!instr)
+ instr = &devc->i_map[0];
+
+ if (devc->voc[voice].mode == 4)
+ {
+ int connection = ((instr->operators[10] & 0x01) << 1) | (instr->operators[10 + OFFS_4OP] & 0x01);
+
+ switch (connection)
+ {
+ case 0:
+ SET_VIBRATO(4);
+ break;
+
+ case 1:
+ SET_VIBRATO(2);
+ SET_VIBRATO(4);
+ break;
+
+ case 2:
+ SET_VIBRATO(1);
+ SET_VIBRATO(4);
+ break;
+
+ case 3:
+ SET_VIBRATO(1);
+ SET_VIBRATO(3);
+ SET_VIBRATO(4);
+ break;
+
+ }
+ /*
+ * Not implemented yet
+ */
+ }
+ else
+ {
+ SET_VIBRATO(1);
+
+ if ((instr->operators[10] & 0x01)) /*
+ * Additive synthesis
+ */
+ SET_VIBRATO(2);
+ }
+}
+
+#undef SET_VIBRATO
+
+static void bend_pitch(int dev, int voice, int value)
+{
+ unsigned char data;
+ int block, fnum, freq;
+ struct physical_voice_info *map;
+
+ map = &pv_map[devc->lv_map[voice]];
+
+ if (map->voice_mode == 0)
+ return;
+
+ devc->voc[voice].bender = value;
+ if (!value)
+ return;
+ if (!(devc->voc[voice].keyon_byte & 0x20))
+ return; /*
+ * Not keyed on
+ */
+
+ freq = compute_finetune(devc->voc[voice].orig_freq, devc->voc[voice].bender, devc->voc[voice].bender_range, 0);
+ devc->voc[voice].current_freq = freq;
+
+ freq_to_fnum(freq, &block, &fnum);
+
+ data = fnum & 0xff; /*
+ * Least significant bits of fnumber
+ */
+ opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data);
+
+ data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3);
+ devc->voc[voice].keyon_byte = data;
+ opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data);
+}
+
+static void opl3_controller (int dev, int voice, int ctrl_num, int value)
+{
+ if (voice < 0 || voice >= devc->nr_voice)
+ return;
+
+ switch (ctrl_num)
+ {
+ case CTRL_PITCH_BENDER:
+ bend_pitch(dev, voice, value);
+ break;
+
+ case CTRL_PITCH_BENDER_RANGE:
+ devc->voc[voice].bender_range = value;
+ break;
+
+ case CTL_MAIN_VOLUME:
+ devc->voc[voice].volume = value / 128;
+ break;
+
+ case CTL_PAN:
+ devc->voc[voice].panning = (value * 2) - 128;
+ break;
+ }
+}
+
+static void opl3_bender(int dev, int voice, int value)
+{
+ if (voice < 0 || voice >= devc->nr_voice)
+ return;
+
+ bend_pitch(dev, voice, value - 8192);
+}
+
+static int opl3_alloc_voice(int dev, int chn, int note, struct voice_alloc_info *alloc)
+{
+ int i, p, best, first, avail, best_time = 0x7fffffff;
+ struct sbi_instrument *instr;
+ int is4op;
+ int instr_no;
+
+ if (chn < 0 || chn > 15)
+ instr_no = 0;
+ else
+ instr_no = devc->chn_info[chn].pgm_num;
+
+ instr = &devc->i_map[instr_no];
+ if (instr->channel < 0 || /* Instrument not loaded */
+ devc->nr_voice != 12) /* Not in 4 OP mode */
+ is4op = 0;
+ else if (devc->nr_voice == 12) /* 4 OP mode */
+ is4op = (instr->key == OPL3_PATCH);
+ else
+ is4op = 0;
+
+ if (is4op)
+ {
+ first = p = 0;
+ avail = 6;
+ }
+ else
+ {
+ if (devc->nr_voice == 12) /* 4 OP mode. Use the '2 OP only' operators first */
+ first = p = 6;
+ else
+ first = p = 0;
+ avail = devc->nr_voice;
+ }
+
+ /*
+ * Now try to find a free voice
+ */
+ best = first;
+
+ for (i = 0; i < avail; i++)
+ {
+ if (alloc->map[p] == 0)
+ {
+ return p;
+ }
+ if (alloc->alloc_times[p] < best_time) /* Find oldest playing note */
+ {
+ best_time = alloc->alloc_times[p];
+ best = p;
+ }
+ p = (p + 1) % avail;
+ }
+
+ /*
+ * Insert some kind of priority mechanism here.
+ */
+
+ if (best < 0)
+ best = 0;
+ if (best > devc->nr_voice)
+ best -= devc->nr_voice;
+
+ return best; /* All devc->voc in use. Select the first one. */
+}
+
+static void opl3_setup_voice(int dev, int voice, int chn)
+{
+ struct channel_info *info =
+ &synth_devs[dev]->chn_info[chn];
+
+ opl3_set_instr(dev, voice, info->pgm_num);
+
+ devc->voc[voice].bender = 0;
+ devc->voc[voice].bender_range = info->bender_range;
+ devc->voc[voice].volume = info->controllers[CTL_MAIN_VOLUME];
+ devc->voc[voice].panning = (info->controllers[CTL_PAN] * 2) - 128;
+}
+
+static struct synth_operations opl3_operations =
+{
+ .owner = THIS_MODULE,
+ .id = "OPL",
+ .info = NULL,
+ .midi_dev = 0,
+ .synth_type = SYNTH_TYPE_FM,
+ .synth_subtype = FM_TYPE_ADLIB,
+ .open = opl3_open,
+ .close = opl3_close,
+ .ioctl = opl3_ioctl,
+ .kill_note = opl3_kill_note,
+ .start_note = opl3_start_note,
+ .set_instr = opl3_set_instr,
+ .reset = opl3_reset,
+ .hw_control = opl3_hw_control,
+ .load_patch = opl3_load_patch,
+ .aftertouch = opl3_aftertouch,
+ .controller = opl3_controller,
+ .panning = opl3_panning,
+ .volume_method = opl3_volume_method,
+ .bender = opl3_bender,
+ .alloc_voice = opl3_alloc_voice,
+ .setup_voice = opl3_setup_voice
+};
+
+static int opl3_init(int ioaddr, struct module *owner)
+{
+ int i;
+ int me;
+
+ if (devc == NULL)
+ {
+ printk(KERN_ERR "opl3: Device control structure not initialized.\n");
+ return -1;
+ }
+
+ if ((me = sound_alloc_synthdev()) == -1)
+ {
+ printk(KERN_WARNING "opl3: Too many synthesizers\n");
+ return -1;
+ }
+
+ devc->nr_voice = 9;
+
+ devc->fm_info.device = 0;
+ devc->fm_info.synth_type = SYNTH_TYPE_FM;
+ devc->fm_info.synth_subtype = FM_TYPE_ADLIB;
+ devc->fm_info.perc_mode = 0;
+ devc->fm_info.nr_voices = 9;
+ devc->fm_info.nr_drums = 0;
+ devc->fm_info.instr_bank_size = SBFM_MAXINSTR;
+ devc->fm_info.capabilities = 0;
+ devc->left_io = ioaddr;
+ devc->right_io = ioaddr + 2;
+
+ if (detected_model <= 2)
+ devc->model = 1;
+ else
+ {
+ devc->model = 2;
+ if (detected_model == 4)
+ devc->is_opl4 = 1;
+ }
+
+ opl3_operations.info = &devc->fm_info;
+
+ synth_devs[me] = &opl3_operations;
+
+ if (owner)
+ synth_devs[me]->owner = owner;
+
+ sequencer_init();
+ devc->v_alloc = &opl3_operations.alloc;
+ devc->chn_info = &opl3_operations.chn_info[0];
+
+ if (devc->model == 2)
+ {
+ if (devc->is_opl4)
+ strcpy(devc->fm_info.name, "Yamaha OPL4/OPL3 FM");
+ else
+ strcpy(devc->fm_info.name, "Yamaha OPL3");
+
+ devc->v_alloc->max_voice = devc->nr_voice = 18;
+ devc->fm_info.nr_drums = 0;
+ devc->fm_info.synth_subtype = FM_TYPE_OPL3;
+ devc->fm_info.capabilities |= SYNTH_CAP_OPL3;
+
+ for (i = 0; i < 18; i++)
+ {
+ if (pv_map[i].ioaddr == USE_LEFT)
+ pv_map[i].ioaddr = devc->left_io;
+ else
+ pv_map[i].ioaddr = devc->right_io;
+ }
+ opl3_command(devc->right_io, OPL3_MODE_REGISTER, OPL3_ENABLE);
+ opl3_command(devc->right_io, CONNECTION_SELECT_REGISTER, 0x00);
+ }
+ else
+ {
+ strcpy(devc->fm_info.name, "Yamaha OPL2");
+ devc->v_alloc->max_voice = devc->nr_voice = 9;
+ devc->fm_info.nr_drums = 0;
+
+ for (i = 0; i < 18; i++)
+ pv_map[i].ioaddr = devc->left_io;
+ };
+ conf_printf2(devc->fm_info.name, ioaddr, 0, -1, -1);
+
+ for (i = 0; i < SBFM_MAXINSTR; i++)
+ devc->i_map[i].channel = -1;
+
+ return me;
+}
+
+static int me;
+
+static int io = -1;
+
+module_param(io, int, 0);
+
+static int __init init_opl3 (void)
+{
+ printk(KERN_INFO "YM3812 and OPL-3 driver Copyright (C) by Hannu Savolainen, Rob Hooft 1993-1996\n");
+
+ if (io != -1) /* User loading pure OPL3 module */
+ {
+ if (!opl3_detect(io))
+ {
+ return -ENODEV;
+ }
+
+ me = opl3_init(io, THIS_MODULE);
+ }
+
+ return 0;
+}
+
+static void __exit cleanup_opl3(void)
+{
+ if (devc && io != -1)
+ {
+ if (devc->base) {
+ release_region(devc->base,4);
+ if (devc->is_opl4)
+ release_region(devc->base - 8, 2);
+ }
+ kfree(devc);
+ devc = NULL;
+ sound_unload_synthdev(me);
+ }
+}
+
+module_init(init_opl3);
+module_exit(cleanup_opl3);
+
+#ifndef MODULE
+static int __init setup_opl3(char *str)
+{
+ /* io */
+ int ints[2];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+
+ return 1;
+}
+
+__setup("opl3=", setup_opl3);
+#endif
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/opl3_hw.h b/sound/oss/opl3_hw.h
new file mode 100644
index 0000000..8b11c89
--- /dev/null
+++ b/sound/oss/opl3_hw.h
@@ -0,0 +1,246 @@
+/*
+ * opl3_hw.h - Definitions of the OPL-3 registers
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * The OPL-3 mode is switched on by writing 0x01, to the offset 5
+ * of the right side.
+ *
+ * Another special register at the right side is at offset 4. It contains
+ * a bit mask defining which voices are used as 4 OP voices.
+ *
+ * The percussive mode is implemented in the left side only.
+ *
+ * With the above exceptions the both sides can be operated independently.
+ *
+ * A 4 OP voice can be created by setting the corresponding
+ * bit at offset 4 of the right side.
+ *
+ * For example setting the rightmost bit (0x01) changes the
+ * first voice on the right side to the 4 OP mode. The fourth
+ * voice is made inaccessible.
+ *
+ * If a voice is set to the 2 OP mode, it works like 2 OP modes
+ * of the original YM3812 (AdLib). In addition the voice can
+ * be connected the left, right or both stereo channels. It can
+ * even be left unconnected. This works with 4 OP voices also.
+ *
+ * The stereo connection bits are located in the FEEDBACK_CONNECTION
+ * register of the voice (0xC0-0xC8). In 4 OP voices these bits are
+ * in the second half of the voice.
+ */
+
+/*
+ * Register numbers for the global registers
+ */
+
+#define TEST_REGISTER 0x01
+#define ENABLE_WAVE_SELECT 0x20
+
+#define TIMER1_REGISTER 0x02
+#define TIMER2_REGISTER 0x03
+#define TIMER_CONTROL_REGISTER 0x04 /* Left side */
+#define IRQ_RESET 0x80
+#define TIMER1_MASK 0x40
+#define TIMER2_MASK 0x20
+#define TIMER1_START 0x01
+#define TIMER2_START 0x02
+
+#define CONNECTION_SELECT_REGISTER 0x04 /* Right side */
+#define RIGHT_4OP_0 0x01
+#define RIGHT_4OP_1 0x02
+#define RIGHT_4OP_2 0x04
+#define LEFT_4OP_0 0x08
+#define LEFT_4OP_1 0x10
+#define LEFT_4OP_2 0x20
+
+#define OPL3_MODE_REGISTER 0x05 /* Right side */
+#define OPL3_ENABLE 0x01
+#define OPL4_ENABLE 0x02
+
+#define KBD_SPLIT_REGISTER 0x08 /* Left side */
+#define COMPOSITE_SINE_WAVE_MODE 0x80 /* Don't use with OPL-3? */
+#define KEYBOARD_SPLIT 0x40
+
+#define PERCOSSION_REGISTER 0xbd /* Left side only */
+#define TREMOLO_DEPTH 0x80
+#define VIBRATO_DEPTH 0x40
+#define PERCOSSION_ENABLE 0x20
+#define BASSDRUM_ON 0x10
+#define SNAREDRUM_ON 0x08
+#define TOMTOM_ON 0x04
+#define CYMBAL_ON 0x02
+#define HIHAT_ON 0x01
+
+/*
+ * Offsets to the register banks for operators. To get the
+ * register number just add the operator offset to the bank offset
+ *
+ * AM/VIB/EG/KSR/Multiple (0x20 to 0x35)
+ */
+#define AM_VIB 0x20
+#define TREMOLO_ON 0x80
+#define VIBRATO_ON 0x40
+#define SUSTAIN_ON 0x20
+#define KSR 0x10 /* Key scaling rate */
+#define MULTIPLE_MASK 0x0f /* Frequency multiplier */
+
+ /*
+ * KSL/Total level (0x40 to 0x55)
+ */
+#define KSL_LEVEL 0x40
+#define KSL_MASK 0xc0 /* Envelope scaling bits */
+#define TOTAL_LEVEL_MASK 0x3f /* Strength (volume) of OP */
+
+/*
+ * Attack / Decay rate (0x60 to 0x75)
+ */
+#define ATTACK_DECAY 0x60
+#define ATTACK_MASK 0xf0
+#define DECAY_MASK 0x0f
+
+/*
+ * Sustain level / Release rate (0x80 to 0x95)
+ */
+#define SUSTAIN_RELEASE 0x80
+#define SUSTAIN_MASK 0xf0
+#define RELEASE_MASK 0x0f
+
+/*
+ * Wave select (0xE0 to 0xF5)
+ */
+#define WAVE_SELECT 0xe0
+
+/*
+ * Offsets to the register banks for voices. Just add to the
+ * voice number to get the register number.
+ *
+ * F-Number low bits (0xA0 to 0xA8).
+ */
+#define FNUM_LOW 0xa0
+
+/*
+ * F-number high bits / Key on / Block (octave) (0xB0 to 0xB8)
+ */
+#define KEYON_BLOCK 0xb0
+#define KEYON_BIT 0x20
+#define BLOCKNUM_MASK 0x1c
+#define FNUM_HIGH_MASK 0x03
+
+/*
+ * Feedback / Connection (0xc0 to 0xc8)
+ *
+ * These registers have two new bits when the OPL-3 mode
+ * is selected. These bits controls connecting the voice
+ * to the stereo channels. For 4 OP voices this bit is
+ * defined in the second half of the voice (add 3 to the
+ * register offset).
+ *
+ * For 4 OP voices the connection bit is used in the
+ * both halves (gives 4 ways to connect the operators).
+ */
+#define FEEDBACK_CONNECTION 0xc0
+#define FEEDBACK_MASK 0x0e /* Valid just for 1st OP of a voice */
+#define CONNECTION_BIT 0x01
+/*
+ * In the 4 OP mode there is four possible configurations how the
+ * operators can be connected together (in 2 OP modes there is just
+ * AM or FM). The 4 OP connection mode is defined by the rightmost
+ * bit of the FEEDBACK_CONNECTION (0xC0-0xC8) on the both halves.
+ *
+ * First half Second half Mode
+ *
+ * +---+
+ * v |
+ * 0 0 >+-1-+--2--3--4-->
+ *
+ *
+ *
+ * +---+
+ * | |
+ * 0 1 >+-1-+--2-+
+ * |->
+ * >--3----4-+
+ *
+ * +---+
+ * | |
+ * 1 0 >+-1-+-----+
+ * |->
+ * >--2--3--4-+
+ *
+ * +---+
+ * | |
+ * 1 1 >+-1-+--+
+ * |
+ * >--2--3-+->
+ * |
+ * >--4----+
+ */
+#define STEREO_BITS 0x30 /* OPL-3 only */
+#define VOICE_TO_LEFT 0x10
+#define VOICE_TO_RIGHT 0x20
+
+/*
+ * Definition table for the physical voices
+ */
+
+struct physical_voice_info {
+ unsigned char voice_num;
+ unsigned char voice_mode; /* 0=unavailable, 2=2 OP, 4=4 OP */
+ unsigned short ioaddr; /* I/O port (left or right side) */
+ unsigned char op[4]; /* Operator offsets */
+ };
+
+/*
+ * There is 18 possible 2 OP voices
+ * (9 in the left and 9 in the right).
+ * The first OP is the modulator and 2nd is the carrier.
+ *
+ * The first three voices in the both sides may be connected
+ * with another voice to a 4 OP voice. For example voice 0
+ * can be connected with voice 3. The operators of voice 3 are
+ * used as operators 3 and 4 of the new 4 OP voice.
+ * In this case the 2 OP voice number 0 is the 'first half' and
+ * voice 3 is the second.
+ */
+
+#define USE_LEFT 0
+#define USE_RIGHT 1
+
+static struct physical_voice_info pv_map[18] =
+{
+/* No Mode Side OP1 OP2 OP3 OP4 */
+/* --------------------------------------------------- */
+ { 0, 2, USE_LEFT, {0x00, 0x03, 0x08, 0x0b}},
+ { 1, 2, USE_LEFT, {0x01, 0x04, 0x09, 0x0c}},
+ { 2, 2, USE_LEFT, {0x02, 0x05, 0x0a, 0x0d}},
+
+ { 3, 2, USE_LEFT, {0x08, 0x0b, 0x00, 0x00}},
+ { 4, 2, USE_LEFT, {0x09, 0x0c, 0x00, 0x00}},
+ { 5, 2, USE_LEFT, {0x0a, 0x0d, 0x00, 0x00}},
+
+ { 6, 2, USE_LEFT, {0x10, 0x13, 0x00, 0x00}}, /* Used by percussive voices */
+ { 7, 2, USE_LEFT, {0x11, 0x14, 0x00, 0x00}}, /* if the percussive mode */
+ { 8, 2, USE_LEFT, {0x12, 0x15, 0x00, 0x00}}, /* is selected */
+
+ { 0, 2, USE_RIGHT, {0x00, 0x03, 0x08, 0x0b}},
+ { 1, 2, USE_RIGHT, {0x01, 0x04, 0x09, 0x0c}},
+ { 2, 2, USE_RIGHT, {0x02, 0x05, 0x0a, 0x0d}},
+
+ { 3, 2, USE_RIGHT, {0x08, 0x0b, 0x00, 0x00}},
+ { 4, 2, USE_RIGHT, {0x09, 0x0c, 0x00, 0x00}},
+ { 5, 2, USE_RIGHT, {0x0a, 0x0d, 0x00, 0x00}},
+
+ { 6, 2, USE_RIGHT, {0x10, 0x13, 0x00, 0x00}},
+ { 7, 2, USE_RIGHT, {0x11, 0x14, 0x00, 0x00}},
+ { 8, 2, USE_RIGHT, {0x12, 0x15, 0x00, 0x00}}
+};
+/*
+ * DMA buffer calls
+ */
diff --git a/sound/oss/os.h b/sound/oss/os.h
new file mode 100644
index 0000000..a1a962d
--- /dev/null
+++ b/sound/oss/os.h
@@ -0,0 +1,46 @@
+#define ALLOW_SELECT
+#undef NO_INLINE_ASM
+#define SHORT_BANNERS
+#define MANUAL_PNP
+#undef DO_TIMINGS
+
+#include <linux/module.h>
+
+#ifdef __KERNEL__
+#include <linux/string.h>
+#include <linux/fs.h>
+#include <asm/dma.h>
+#include <asm/io.h>
+#include <asm/param.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <asm/page.h>
+#include <asm/system.h>
+#include <linux/vmalloc.h>
+#include <asm/uaccess.h>
+#include <linux/poll.h>
+#include <linux/pci.h>
+#endif
+
+#include <linux/soundcard.h>
+
+#define FALSE 0
+#define TRUE 1
+
+extern int sound_alloc_dma(int chn, char *deviceID);
+extern int sound_open_dma(int chn, char *deviceID);
+extern void sound_free_dma(int chn);
+extern void sound_close_dma(int chn);
+
+extern void reprogram_timer(void);
+
+#define USE_AUTOINIT_DMA
+
+extern void *sound_mem_blocks[1024];
+extern int sound_nblocks;
+
+#undef PSEUDO_DMA_AUTOINIT
+#define ALLOW_BUFFER_MAPPING
+
+extern const struct file_operations oss_sound_fops;
diff --git a/sound/oss/pas2.h b/sound/oss/pas2.h
new file mode 100644
index 0000000..fa12c55
--- /dev/null
+++ b/sound/oss/pas2.h
@@ -0,0 +1,17 @@
+
+/* From pas_card.c */
+int pas_set_intr(int mask);
+int pas_remove_intr(int mask);
+unsigned char pas_read(int ioaddr);
+void pas_write(unsigned char data, int ioaddr);
+
+/* From pas_audio.c */
+void pas_pcm_interrupt(unsigned char status, int cause);
+void pas_pcm_init(struct address_info *hw_config);
+
+/* From pas_mixer.c */
+int pas_init_mixer(void);
+
+/* From pas_midi.c */
+void pas_midi_init(void);
+void pas_midi_interrupt(void);
diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c
new file mode 100644
index 0000000..25f3a22
--- /dev/null
+++ b/sound/oss/pas2_card.c
@@ -0,0 +1,457 @@
+/*
+ * sound/oss/pas2_card.c
+ *
+ * Detection routine for the Pro Audio Spectrum cards.
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/spinlock.h>
+#include "sound_config.h"
+
+#include "pas2.h"
+#include "sb.h"
+
+static unsigned char dma_bits[] = {
+ 4, 1, 2, 3, 0, 5, 6, 7
+};
+
+static unsigned char irq_bits[] = {
+ 0, 0, 1, 2, 3, 4, 5, 6, 0, 1, 7, 8, 9, 0, 10, 11
+};
+
+static unsigned char sb_irq_bits[] = {
+ 0x00, 0x00, 0x08, 0x10, 0x00, 0x18, 0x00, 0x20,
+ 0x00, 0x08, 0x28, 0x30, 0x38, 0, 0
+};
+
+static unsigned char sb_dma_bits[] = {
+ 0x00, 0x40, 0x80, 0xC0, 0, 0, 0, 0
+};
+
+/*
+ * The Address Translation code is used to convert I/O register addresses to
+ * be relative to the given base -register
+ */
+
+int pas_translate_code = 0;
+static int pas_intr_mask;
+static int pas_irq;
+static int pas_sb_base;
+DEFINE_SPINLOCK(pas_lock);
+#ifndef CONFIG_PAS_JOYSTICK
+static int joystick;
+#else
+static int joystick = 1;
+#endif
+#ifdef SYMPHONY_PAS
+static int symphony = 1;
+#else
+static int symphony;
+#endif
+#ifdef BROKEN_BUS_CLOCK
+static int broken_bus_clock = 1;
+#else
+static int broken_bus_clock;
+#endif
+
+static struct address_info cfg;
+static struct address_info cfg2;
+
+char pas_model = 0;
+static char *pas_model_names[] = {
+ "",
+ "Pro AudioSpectrum+",
+ "CDPC",
+ "Pro AudioSpectrum 16",
+ "Pro AudioSpectrum 16D"
+};
+
+/*
+ * pas_read() and pas_write() are equivalents of inb and outb
+ * These routines perform the I/O address translation required
+ * to support other than the default base address
+ */
+
+extern void mix_write(unsigned char data, int ioaddr);
+
+unsigned char pas_read(int ioaddr)
+{
+ return inb(ioaddr + pas_translate_code);
+}
+
+void pas_write(unsigned char data, int ioaddr)
+{
+ outb((data), ioaddr + pas_translate_code);
+}
+
+/******************* Begin of the Interrupt Handler ********************/
+
+static irqreturn_t pasintr(int irq, void *dev_id)
+{
+ int status;
+
+ status = pas_read(0x0B89);
+ pas_write(status, 0x0B89); /* Clear interrupt */
+
+ if (status & 0x08)
+ {
+ pas_pcm_interrupt(status, 1);
+ status &= ~0x08;
+ }
+ if (status & 0x10)
+ {
+ pas_midi_interrupt();
+ status &= ~0x10;
+ }
+ return IRQ_HANDLED;
+}
+
+int pas_set_intr(int mask)
+{
+ if (!mask)
+ return 0;
+
+ pas_intr_mask |= mask;
+
+ pas_write(pas_intr_mask, 0x0B8B);
+ return 0;
+}
+
+int pas_remove_intr(int mask)
+{
+ if (!mask)
+ return 0;
+
+ pas_intr_mask &= ~mask;
+ pas_write(pas_intr_mask, 0x0B8B);
+
+ return 0;
+}
+
+/******************* End of the Interrupt handler **********************/
+
+/******************* Begin of the Initialization Code ******************/
+
+static int __init config_pas_hw(struct address_info *hw_config)
+{
+ char ok = 1;
+ unsigned int_ptrs; /* scsi/sound interrupt pointers */
+
+ pas_irq = hw_config->irq;
+
+ pas_write(0x00, 0x0B8B);
+ pas_write(0x36, 0x138B);
+ pas_write(0x36, 0x1388);
+ pas_write(0, 0x1388);
+ pas_write(0x74, 0x138B);
+ pas_write(0x74, 0x1389);
+ pas_write(0, 0x1389);
+
+ pas_write(0x80 | 0x40 | 0x20 | 1, 0x0B8A);
+ pas_write(0x80 | 0x20 | 0x10 | 0x08 | 0x01, 0xF8A);
+ pas_write(0x01 | 0x02 | 0x04 | 0x10 /*
+ * |
+ * 0x80
+ */ , 0xB88);
+
+ pas_write(0x80
+ | joystick?0x40:0
+ ,0xF388);
+
+ if (pas_irq < 0 || pas_irq > 15)
+ {
+ printk(KERN_ERR "PAS16: Invalid IRQ %d", pas_irq);
+ hw_config->irq=-1;
+ ok = 0;
+ }
+ else
+ {
+ int_ptrs = pas_read(0xF38A);
+ int_ptrs = (int_ptrs & 0xf0) | irq_bits[pas_irq];
+ pas_write(int_ptrs, 0xF38A);
+ if (!irq_bits[pas_irq])
+ {
+ printk(KERN_ERR "PAS16: Invalid IRQ %d", pas_irq);
+ hw_config->irq=-1;
+ ok = 0;
+ }
+ else
+ {
+ if (request_irq(pas_irq, pasintr, 0, "PAS16",hw_config) < 0) {
+ printk(KERN_ERR "PAS16: Cannot allocate IRQ %d\n",pas_irq);
+ hw_config->irq=-1;
+ ok = 0;
+ }
+ }
+ }
+
+ if (hw_config->dma < 0 || hw_config->dma > 7)
+ {
+ printk(KERN_ERR "PAS16: Invalid DMA selection %d", hw_config->dma);
+ hw_config->dma=-1;
+ ok = 0;
+ }
+ else
+ {
+ pas_write(dma_bits[hw_config->dma], 0xF389);
+ if (!dma_bits[hw_config->dma])
+ {
+ printk(KERN_ERR "PAS16: Invalid DMA selection %d", hw_config->dma);
+ hw_config->dma=-1;
+ ok = 0;
+ }
+ else
+ {
+ if (sound_alloc_dma(hw_config->dma, "PAS16"))
+ {
+ printk(KERN_ERR "pas2_card.c: Can't allocate DMA channel\n");
+ hw_config->dma=-1;
+ ok = 0;
+ }
+ }
+ }
+
+ /*
+ * This fixes the timing problems of the PAS due to the Symphony chipset
+ * as per Media Vision. Only define this if your PAS doesn't work correctly.
+ */
+
+ if(symphony)
+ {
+ outb((0x05), 0xa8);
+ outb((0x60), 0xa9);
+ }
+
+ if(broken_bus_clock)
+ pas_write(0x01 | 0x10 | 0x20 | 0x04, 0x8388);
+ else
+ /*
+ * pas_write(0x01, 0x8388);
+ */
+ pas_write(0x01 | 0x10 | 0x20, 0x8388);
+
+ pas_write(0x18, 0x838A); /* ??? */
+ pas_write(0x20 | 0x01, 0x0B8A); /* Mute off, filter = 17.897 kHz */
+ pas_write(8, 0xBF8A);
+
+ mix_write(0x80 | 5, 0x078B);
+ mix_write(5, 0x078B);
+
+ {
+ struct address_info *sb_config;
+
+ sb_config = &cfg2;
+ if (sb_config->io_base)
+ {
+ unsigned char irq_dma;
+
+ /*
+ * Turn on Sound Blaster compatibility
+ * bit 1 = SB emulation
+ * bit 0 = MPU401 emulation (CDPC only :-( )
+ */
+
+ pas_write(0x02, 0xF788);
+
+ /*
+ * "Emulation address"
+ */
+
+ pas_write((sb_config->io_base >> 4) & 0x0f, 0xF789);
+ pas_sb_base = sb_config->io_base;
+
+ if (!sb_dma_bits[sb_config->dma])
+ printk(KERN_ERR "PAS16 Warning: Invalid SB DMA %d\n\n", sb_config->dma);
+
+ if (!sb_irq_bits[sb_config->irq])
+ printk(KERN_ERR "PAS16 Warning: Invalid SB IRQ %d\n\n", sb_config->irq);
+
+ irq_dma = sb_dma_bits[sb_config->dma] |
+ sb_irq_bits[sb_config->irq];
+
+ pas_write(irq_dma, 0xFB8A);
+ }
+ else
+ pas_write(0x00, 0xF788);
+ }
+
+ if (!ok)
+ printk(KERN_WARNING "PAS16: Driver not enabled\n");
+
+ return ok;
+}
+
+static int __init detect_pas_hw(struct address_info *hw_config)
+{
+ unsigned char board_id, foo;
+
+ /*
+ * WARNING: Setting an option like W:1 or so that disables warm boot reset
+ * of the card will screw up this detect code something fierce. Adding code
+ * to handle this means possibly interfering with other cards on the bus if
+ * you have something on base port 0x388. SO be forewarned.
+ */
+
+ outb((0xBC), 0x9A01); /* Activate first board */
+ outb((hw_config->io_base >> 2), 0x9A01); /* Set base address */
+ pas_translate_code = hw_config->io_base - 0x388;
+ pas_write(1, 0xBF88); /* Select one wait states */
+
+ board_id = pas_read(0x0B8B);
+
+ if (board_id == 0xff)
+ return 0;
+
+ /*
+ * We probably have a PAS-series board, now check for a PAS16-series board
+ * by trying to change the board revision bits. PAS16-series hardware won't
+ * let you do this - the bits are read-only.
+ */
+
+ foo = board_id ^ 0xe0;
+
+ pas_write(foo, 0x0B8B);
+ foo = pas_read(0x0B8B);
+ pas_write(board_id, 0x0B8B);
+
+ if (board_id != foo)
+ return 0;
+
+ pas_model = pas_read(0xFF88);
+
+ return pas_model;
+}
+
+static void __init attach_pas_card(struct address_info *hw_config)
+{
+ pas_irq = hw_config->irq;
+
+ if (detect_pas_hw(hw_config))
+ {
+
+ if ((pas_model = pas_read(0xFF88)))
+ {
+ char temp[100];
+
+ sprintf(temp,
+ "%s rev %d", pas_model_names[(int) pas_model],
+ pas_read(0x2789));
+ conf_printf(temp, hw_config);
+ }
+ if (config_pas_hw(hw_config))
+ {
+ pas_pcm_init(hw_config);
+ pas_midi_init();
+ pas_init_mixer();
+ }
+ }
+}
+
+static inline int __init probe_pas(struct address_info *hw_config)
+{
+ return detect_pas_hw(hw_config);
+}
+
+static void __exit unload_pas(struct address_info *hw_config)
+{
+ extern int pas_audiodev;
+ extern int pas2_mididev;
+
+ if (hw_config->dma>0)
+ sound_free_dma(hw_config->dma);
+ if (hw_config->irq>0)
+ free_irq(hw_config->irq, hw_config);
+
+ if(pas_audiodev!=-1)
+ sound_unload_mixerdev(audio_devs[pas_audiodev]->mixer_dev);
+ if(pas2_mididev!=-1)
+ sound_unload_mididev(pas2_mididev);
+ if(pas_audiodev!=-1)
+ sound_unload_audiodev(pas_audiodev);
+}
+
+static int __initdata io = -1;
+static int __initdata irq = -1;
+static int __initdata dma = -1;
+static int __initdata dma16 = -1; /* Set this for modules that need it */
+
+static int __initdata sb_io = 0;
+static int __initdata sb_irq = -1;
+static int __initdata sb_dma = -1;
+static int __initdata sb_dma16 = -1;
+
+module_param(io, int, 0);
+module_param(irq, int, 0);
+module_param(dma, int, 0);
+module_param(dma16, int, 0);
+
+module_param(sb_io, int, 0);
+module_param(sb_irq, int, 0);
+module_param(sb_dma, int, 0);
+module_param(sb_dma16, int, 0);
+
+module_param(joystick, bool, 0);
+module_param(symphony, bool, 0);
+module_param(broken_bus_clock, bool, 0);
+
+MODULE_LICENSE("GPL");
+
+static int __init init_pas2(void)
+{
+ printk(KERN_INFO "Pro Audio Spectrum driver Copyright (C) by Hannu Savolainen 1993-1996\n");
+
+ cfg.io_base = io;
+ cfg.irq = irq;
+ cfg.dma = dma;
+ cfg.dma2 = dma16;
+
+ cfg2.io_base = sb_io;
+ cfg2.irq = sb_irq;
+ cfg2.dma = sb_dma;
+ cfg2.dma2 = sb_dma16;
+
+ if (cfg.io_base == -1 || cfg.dma == -1 || cfg.irq == -1) {
+ printk(KERN_INFO "I/O, IRQ, DMA and type are mandatory\n");
+ return -EINVAL;
+ }
+
+ if (!probe_pas(&cfg))
+ return -ENODEV;
+ attach_pas_card(&cfg);
+
+ return 0;
+}
+
+static void __exit cleanup_pas2(void)
+{
+ unload_pas(&cfg);
+}
+
+module_init(init_pas2);
+module_exit(cleanup_pas2);
+
+#ifndef MODULE
+static int __init setup_pas2(char *str)
+{
+ /* io, irq, dma, dma2, sb_io, sb_irq, sb_dma, sb_dma2 */
+ int ints[9];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+ dma = ints[3];
+ dma16 = ints[4];
+
+ sb_io = ints[5];
+ sb_irq = ints[6];
+ sb_dma = ints[7];
+ sb_dma16 = ints[8];
+
+ return 1;
+}
+
+__setup("pas2=", setup_pas2);
+#endif
diff --git a/sound/oss/pas2_midi.c b/sound/oss/pas2_midi.c
new file mode 100644
index 0000000..1122d10
--- /dev/null
+++ b/sound/oss/pas2_midi.c
@@ -0,0 +1,262 @@
+/*
+ * sound/oss/pas2_midi.c
+ *
+ * The low level driver for the PAS Midi Interface.
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ * Bartlomiej Zolnierkiewicz : Added __init to pas_init_mixer()
+ */
+
+#include <linux/init.h>
+#include <linux/spinlock.h>
+#include "sound_config.h"
+
+#include "pas2.h"
+
+extern spinlock_t pas_lock;
+
+static int midi_busy, input_opened;
+static int my_dev;
+
+int pas2_mididev=-1;
+
+static unsigned char tmp_queue[256];
+static volatile int qlen;
+static volatile unsigned char qhead, qtail;
+
+static void (*midi_input_intr) (int dev, unsigned char data);
+
+static int pas_midi_open(int dev, int mode,
+ void (*input) (int dev, unsigned char data),
+ void (*output) (int dev)
+)
+{
+ int err;
+ unsigned long flags;
+ unsigned char ctrl;
+
+
+ if (midi_busy)
+ return -EBUSY;
+
+ /*
+ * Reset input and output FIFO pointers
+ */
+ pas_write(0x20 | 0x40,
+ 0x178b);
+
+ spin_lock_irqsave(&pas_lock, flags);
+
+ if ((err = pas_set_intr(0x10)) < 0)
+ {
+ spin_unlock_irqrestore(&pas_lock, flags);
+ return err;
+ }
+ /*
+ * Enable input available and output FIFO empty interrupts
+ */
+
+ ctrl = 0;
+ input_opened = 0;
+ midi_input_intr = input;
+
+ if (mode == OPEN_READ || mode == OPEN_READWRITE)
+ {
+ ctrl |= 0x04; /* Enable input */
+ input_opened = 1;
+ }
+ if (mode == OPEN_WRITE || mode == OPEN_READWRITE)
+ {
+ ctrl |= 0x08 | 0x10; /* Enable output */
+ }
+ pas_write(ctrl, 0x178b);
+
+ /*
+ * Acknowledge any pending interrupts
+ */
+
+ pas_write(0xff, 0x1B88);
+
+ spin_unlock_irqrestore(&pas_lock, flags);
+
+ midi_busy = 1;
+ qlen = qhead = qtail = 0;
+ return 0;
+}
+
+static void pas_midi_close(int dev)
+{
+
+ /*
+ * Reset FIFO pointers, disable intrs
+ */
+ pas_write(0x20 | 0x40, 0x178b);
+
+ pas_remove_intr(0x10);
+ midi_busy = 0;
+}
+
+static int dump_to_midi(unsigned char midi_byte)
+{
+ int fifo_space, x;
+
+ fifo_space = ((x = pas_read(0x1B89)) >> 4) & 0x0f;
+
+ /*
+ * The MIDI FIFO space register and it's documentation is nonunderstandable.
+ * There seem to be no way to differentiate between buffer full and buffer
+ * empty situations. For this reason we don't never write the buffer
+ * completely full. In this way we can assume that 0 (or is it 15)
+ * means that the buffer is empty.
+ */
+
+ if (fifo_space < 2 && fifo_space != 0) /* Full (almost) */
+ return 0; /* Ask upper layers to retry after some time */
+
+ pas_write(midi_byte, 0x178A);
+
+ return 1;
+}
+
+static int pas_midi_out(int dev, unsigned char midi_byte)
+{
+
+ unsigned long flags;
+
+ /*
+ * Drain the local queue first
+ */
+
+ spin_lock_irqsave(&pas_lock, flags);
+
+ while (qlen && dump_to_midi(tmp_queue[qhead]))
+ {
+ qlen--;
+ qhead++;
+ }
+
+ spin_unlock_irqrestore(&pas_lock, flags);
+
+ /*
+ * Output the byte if the local queue is empty.
+ */
+
+ if (!qlen)
+ if (dump_to_midi(midi_byte))
+ return 1;
+
+ /*
+ * Put to the local queue
+ */
+
+ if (qlen >= 256)
+ return 0; /* Local queue full */
+
+ spin_lock_irqsave(&pas_lock, flags);
+
+ tmp_queue[qtail] = midi_byte;
+ qlen++;
+ qtail++;
+
+ spin_unlock_irqrestore(&pas_lock, flags);
+
+ return 1;
+}
+
+static int pas_midi_start_read(int dev)
+{
+ return 0;
+}
+
+static int pas_midi_end_read(int dev)
+{
+ return 0;
+}
+
+static void pas_midi_kick(int dev)
+{
+}
+
+static int pas_buffer_status(int dev)
+{
+ return qlen;
+}
+
+#define MIDI_SYNTH_NAME "Pro Audio Spectrum Midi"
+#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT
+#include "midi_synth.h"
+
+static struct midi_operations pas_midi_operations =
+{
+ .owner = THIS_MODULE,
+ .info = {"Pro Audio Spectrum", 0, 0, SNDCARD_PAS},
+ .converter = &std_midi_synth,
+ .in_info = {0},
+ .open = pas_midi_open,
+ .close = pas_midi_close,
+ .outputc = pas_midi_out,
+ .start_read = pas_midi_start_read,
+ .end_read = pas_midi_end_read,
+ .kick = pas_midi_kick,
+ .buffer_status = pas_buffer_status,
+};
+
+void __init pas_midi_init(void)
+{
+ int dev = sound_alloc_mididev();
+
+ if (dev == -1)
+ {
+ printk(KERN_WARNING "pas_midi_init: Too many midi devices detected\n");
+ return;
+ }
+ std_midi_synth.midi_dev = my_dev = dev;
+ midi_devs[dev] = &pas_midi_operations;
+ pas2_mididev = dev;
+ sequencer_init();
+}
+
+void pas_midi_interrupt(void)
+{
+ unsigned char stat;
+ int i, incount;
+
+ stat = pas_read(0x1B88);
+
+ if (stat & 0x04) /* Input data available */
+ {
+ incount = pas_read(0x1B89) & 0x0f; /* Input FIFO size */
+ if (!incount)
+ incount = 16;
+
+ for (i = 0; i < incount; i++)
+ if (input_opened)
+ {
+ midi_input_intr(my_dev, pas_read(0x178A));
+ } else
+ pas_read(0x178A); /* Flush */
+ }
+ if (stat & (0x08 | 0x10))
+ {
+ spin_lock(&pas_lock);/* called in irq context */
+
+ while (qlen && dump_to_midi(tmp_queue[qhead]))
+ {
+ qlen--;
+ qhead++;
+ }
+
+ spin_unlock(&pas_lock);
+ }
+ if (stat & 0x40)
+ {
+ printk(KERN_WARNING "MIDI output overrun %x,%x\n", pas_read(0x1B89), stat);
+ }
+ pas_write(stat, 0x1B88); /* Acknowledge interrupts */
+}
diff --git a/sound/oss/pas2_mixer.c b/sound/oss/pas2_mixer.c
new file mode 100644
index 0000000..a0bcb85
--- /dev/null
+++ b/sound/oss/pas2_mixer.c
@@ -0,0 +1,336 @@
+
+/*
+ * sound/oss/pas2_mixer.c
+ *
+ * Mixer routines for the Pro Audio Spectrum cards.
+ */
+
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+/*
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Bartlomiej Zolnierkiewicz : added __init to pas_init_mixer()
+ */
+#include <linux/init.h>
+#include "sound_config.h"
+
+#include "pas2.h"
+
+#ifndef DEB
+#define DEB(what) /* (what) */
+#endif
+
+extern int pas_translate_code;
+extern char pas_model;
+extern int *pas_osp;
+extern int pas_audiodev;
+
+static int rec_devices = (SOUND_MASK_MIC); /* Default recording source */
+static int mode_control;
+
+#define POSSIBLE_RECORDING_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \
+ SOUND_MASK_CD | SOUND_MASK_ALTPCM)
+
+#define SUPPORTED_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \
+ SOUND_MASK_CD | SOUND_MASK_ALTPCM | SOUND_MASK_IMIX | \
+ SOUND_MASK_VOLUME | SOUND_MASK_BASS | SOUND_MASK_TREBLE | SOUND_MASK_RECLEV)
+
+static int *levels;
+
+static int default_levels[32] =
+{
+ 0x3232, /* Master Volume */
+ 0x3232, /* Bass */
+ 0x3232, /* Treble */
+ 0x5050, /* FM */
+ 0x4b4b, /* PCM */
+ 0x3232, /* PC Speaker */
+ 0x4b4b, /* Ext Line */
+ 0x4b4b, /* Mic */
+ 0x4b4b, /* CD */
+ 0x6464, /* Recording monitor */
+ 0x4b4b, /* SB PCM */
+ 0x6464 /* Recording level */
+};
+
+void
+mix_write(unsigned char data, int ioaddr)
+{
+ /*
+ * The Revision D cards have a problem with their MVA508 interface. The
+ * kludge-o-rama fix is to make a 16-bit quantity with identical LSB and
+ * MSBs out of the output byte and to do a 16-bit out to the mixer port -
+ * 1. We need to do this because it isn't timing problem but chip access
+ * sequence problem.
+ */
+
+ if (pas_model == 4)
+ {
+ outw(data | (data << 8), (ioaddr + pas_translate_code) - 1);
+ outb((0x80), 0);
+ } else
+ pas_write(data, ioaddr);
+}
+
+static int
+mixer_output(int right_vol, int left_vol, int div, int bits,
+ int mixer) /* Input or output mixer */
+{
+ int left = left_vol * div / 100;
+ int right = right_vol * div / 100;
+
+
+ if (bits & 0x10)
+ {
+ left |= mixer;
+ right |= mixer;
+ }
+ if (bits == 0x03 || bits == 0x04)
+ {
+ mix_write(0x80 | bits, 0x078B);
+ mix_write(left, 0x078B);
+ right_vol = left_vol;
+ } else
+ {
+ mix_write(0x80 | 0x20 | bits, 0x078B);
+ mix_write(left, 0x078B);
+ mix_write(0x80 | 0x40 | bits, 0x078B);
+ mix_write(right, 0x078B);
+ }
+
+ return (left_vol | (right_vol << 8));
+}
+
+static void
+set_mode(int new_mode)
+{
+ mix_write(0x80 | 0x05, 0x078B);
+ mix_write(new_mode, 0x078B);
+
+ mode_control = new_mode;
+}
+
+static int
+pas_mixer_set(int whichDev, unsigned int level)
+{
+ int left, right, devmask, changed, i, mixer = 0;
+
+ DEB(printk("static int pas_mixer_set(int whichDev = %d, unsigned int level = %X)\n", whichDev, level));
+
+ left = level & 0x7f;
+ right = (level & 0x7f00) >> 8;
+
+ if (whichDev < SOUND_MIXER_NRDEVICES) {
+ if ((1 << whichDev) & rec_devices)
+ mixer = 0x20;
+ else
+ mixer = 0x00;
+ }
+
+ switch (whichDev)
+ {
+ case SOUND_MIXER_VOLUME: /* Master volume (0-63) */
+ levels[whichDev] = mixer_output(right, left, 63, 0x01, 0);
+ break;
+
+ /*
+ * Note! Bass and Treble are mono devices. Will use just the left
+ * channel.
+ */
+ case SOUND_MIXER_BASS: /* Bass (0-12) */
+ levels[whichDev] = mixer_output(right, left, 12, 0x03, 0);
+ break;
+ case SOUND_MIXER_TREBLE: /* Treble (0-12) */
+ levels[whichDev] = mixer_output(right, left, 12, 0x04, 0);
+ break;
+
+ case SOUND_MIXER_SYNTH: /* Internal synthesizer (0-31) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x00, mixer);
+ break;
+ case SOUND_MIXER_PCM: /* PAS PCM (0-31) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x05, mixer);
+ break;
+ case SOUND_MIXER_ALTPCM: /* SB PCM (0-31) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x07, mixer);
+ break;
+ case SOUND_MIXER_SPEAKER: /* PC speaker (0-31) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x06, mixer);
+ break;
+ case SOUND_MIXER_LINE: /* External line (0-31) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x02, mixer);
+ break;
+ case SOUND_MIXER_CD: /* CD (0-31) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x03, mixer);
+ break;
+ case SOUND_MIXER_MIC: /* External microphone (0-31) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x04, mixer);
+ break;
+ case SOUND_MIXER_IMIX: /* Recording monitor (0-31) (Output mixer only) */
+ levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x01,
+ 0x00);
+ break;
+ case SOUND_MIXER_RECLEV: /* Recording level (0-15) */
+ levels[whichDev] = mixer_output(right, left, 15, 0x02, 0);
+ break;
+
+
+ case SOUND_MIXER_RECSRC:
+ devmask = level & POSSIBLE_RECORDING_DEVICES;
+
+ changed = devmask ^ rec_devices;
+ rec_devices = devmask;
+
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++)
+ if (changed & (1 << i))
+ {
+ pas_mixer_set(i, levels[i]);
+ }
+ return rec_devices;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return (levels[whichDev]);
+}
+
+/*****/
+
+static void
+pas_mixer_reset(void)
+{
+ int foo;
+
+ DEB(printk("pas2_mixer.c: void pas_mixer_reset(void)\n"));
+
+ for (foo = 0; foo < SOUND_MIXER_NRDEVICES; foo++)
+ pas_mixer_set(foo, levels[foo]);
+
+ set_mode(0x04 | 0x01);
+}
+
+static int pas_mixer_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ int level,v ;
+ int __user *p = (int __user *)arg;
+
+ DEB(printk("pas2_mixer.c: int pas_mixer_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg));
+ if (cmd == SOUND_MIXER_PRIVATE1) { /* Set loudness bit */
+ if (get_user(level, p))
+ return -EFAULT;
+ if (level == -1) /* Return current settings */
+ level = (mode_control & 0x04);
+ else {
+ mode_control &= ~0x04;
+ if (level)
+ mode_control |= 0x04;
+ set_mode(mode_control);
+ }
+ level = !!level;
+ return put_user(level, p);
+ }
+ if (cmd == SOUND_MIXER_PRIVATE2) { /* Set enhance bit */
+ if (get_user(level, p))
+ return -EFAULT;
+ if (level == -1) { /* Return current settings */
+ if (!(mode_control & 0x03))
+ level = 0;
+ else
+ level = ((mode_control & 0x03) + 1) * 20;
+ } else {
+ int i = 0;
+
+ level &= 0x7f;
+ if (level)
+ i = (level / 20) - 1;
+ mode_control &= ~0x03;
+ mode_control |= i & 0x03;
+ set_mode(mode_control);
+ if (i)
+ i = (i + 1) * 20;
+ level = i;
+ }
+ return put_user(level, p);
+ }
+ if (cmd == SOUND_MIXER_PRIVATE3) { /* Set mute bit */
+ if (get_user(level, p))
+ return -EFAULT;
+ if (level == -1) /* Return current settings */
+ level = !(pas_read(0x0B8A) & 0x20);
+ else {
+ if (level)
+ pas_write(pas_read(0x0B8A) & (~0x20), 0x0B8A);
+ else
+ pas_write(pas_read(0x0B8A) | 0x20, 0x0B8A);
+
+ level = !(pas_read(0x0B8A) & 0x20);
+ }
+ return put_user(level, p);
+ }
+ if (((cmd >> 8) & 0xff) == 'M') {
+ if (get_user(v, p))
+ return -EFAULT;
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE) {
+ v = pas_mixer_set(cmd & 0xff, v);
+ } else {
+ switch (cmd & 0xff) {
+ case SOUND_MIXER_RECSRC:
+ v = rec_devices;
+ break;
+
+ case SOUND_MIXER_STEREODEVS:
+ v = SUPPORTED_MIXER_DEVICES & ~(SOUND_MASK_BASS | SOUND_MASK_TREBLE);
+ break;
+
+ case SOUND_MIXER_DEVMASK:
+ v = SUPPORTED_MIXER_DEVICES;
+ break;
+
+ case SOUND_MIXER_RECMASK:
+ v = POSSIBLE_RECORDING_DEVICES & SUPPORTED_MIXER_DEVICES;
+ break;
+
+ case SOUND_MIXER_CAPS:
+ v = 0; /* No special capabilities */
+ break;
+
+ default:
+ v = levels[cmd & 0xff];
+ break;
+ }
+ }
+ return put_user(v, p);
+ }
+ return -EINVAL;
+}
+
+static struct mixer_operations pas_mixer_operations =
+{
+ .owner = THIS_MODULE,
+ .id = "PAS16",
+ .name = "Pro Audio Spectrum 16",
+ .ioctl = pas_mixer_ioctl
+};
+
+int __init
+pas_init_mixer(void)
+{
+ int d;
+
+ levels = load_mixer_volumes("PAS16_1", default_levels, 1);
+
+ pas_mixer_reset();
+
+ if ((d = sound_alloc_mixerdev()) != -1)
+ {
+ audio_devs[pas_audiodev]->mixer_dev = d;
+ mixer_devs[d] = &pas_mixer_operations;
+ }
+ return 1;
+}
diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c
new file mode 100644
index 0000000..36c3ea6
--- /dev/null
+++ b/sound/oss/pas2_pcm.c
@@ -0,0 +1,437 @@
+/*
+ * pas2_pcm.c Audio routines for PAS16
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Alan Cox : Swatted a double allocation of device bug. Made a few
+ * more things module options.
+ * Bartlomiej Zolnierkiewicz : Added __init to pas_pcm_init()
+ */
+
+#include <linux/init.h>
+#include <linux/spinlock.h>
+#include <asm/timex.h>
+#include "sound_config.h"
+
+#include "pas2.h"
+
+#ifndef DEB
+#define DEB(WHAT)
+#endif
+
+#define PAS_PCM_INTRBITS (0x08)
+/*
+ * Sample buffer timer interrupt enable
+ */
+
+#define PCM_NON 0
+#define PCM_DAC 1
+#define PCM_ADC 2
+
+static unsigned long pcm_speed; /* sampling rate */
+static unsigned char pcm_channels = 1; /* channels (1 or 2) */
+static unsigned char pcm_bits = 8; /* bits/sample (8 or 16) */
+static unsigned char pcm_filter; /* filter FLAG */
+static unsigned char pcm_mode = PCM_NON;
+static unsigned long pcm_count;
+static unsigned short pcm_bitsok = 8; /* mask of OK bits */
+static int pcm_busy;
+int pas_audiodev = -1;
+static int open_mode;
+
+extern spinlock_t pas_lock;
+
+static int pcm_set_speed(int arg)
+{
+ int foo, tmp;
+ unsigned long flags;
+
+ if (arg == 0)
+ return pcm_speed;
+
+ if (arg > 44100)
+ arg = 44100;
+ if (arg < 5000)
+ arg = 5000;
+
+ if (pcm_channels & 2)
+ {
+ foo = ((CLOCK_TICK_RATE / 2) + (arg / 2)) / arg;
+ arg = ((CLOCK_TICK_RATE / 2) + (foo / 2)) / foo;
+ }
+ else
+ {
+ foo = (CLOCK_TICK_RATE + (arg / 2)) / arg;
+ arg = (CLOCK_TICK_RATE + (foo / 2)) / foo;
+ }
+
+ pcm_speed = arg;
+
+ tmp = pas_read(0x0B8A);
+
+ /*
+ * Set anti-aliasing filters according to sample rate. You really *NEED*
+ * to enable this feature for all normal recording unless you want to
+ * experiment with aliasing effects.
+ * These filters apply to the selected "recording" source.
+ * I (pfw) don't know the encoding of these 5 bits. The values shown
+ * come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/.
+ *
+ * I cleared bit 5 of these values, since that bit controls the master
+ * mute flag. (Olav Wölfelschneider)
+ *
+ */
+#if !defined NO_AUTO_FILTER_SET
+ tmp &= 0xe0;
+ if (pcm_speed >= 2 * 17897)
+ tmp |= 0x01;
+ else if (pcm_speed >= 2 * 15909)
+ tmp |= 0x02;
+ else if (pcm_speed >= 2 * 11931)
+ tmp |= 0x09;
+ else if (pcm_speed >= 2 * 8948)
+ tmp |= 0x11;
+ else if (pcm_speed >= 2 * 5965)
+ tmp |= 0x19;
+ else if (pcm_speed >= 2 * 2982)
+ tmp |= 0x04;
+ pcm_filter = tmp;
+#endif
+
+ spin_lock_irqsave(&pas_lock, flags);
+
+ pas_write(tmp & ~(0x40 | 0x80), 0x0B8A);
+ pas_write(0x00 | 0x30 | 0x04, 0x138B);
+ pas_write(foo & 0xff, 0x1388);
+ pas_write((foo >> 8) & 0xff, 0x1388);
+ pas_write(tmp, 0x0B8A);
+
+ spin_unlock_irqrestore(&pas_lock, flags);
+
+ return pcm_speed;
+}
+
+static int pcm_set_channels(int arg)
+{
+
+ if ((arg != 1) && (arg != 2))
+ return pcm_channels;
+
+ if (arg != pcm_channels)
+ {
+ pas_write(pas_read(0xF8A) ^ 0x20, 0xF8A);
+
+ pcm_channels = arg;
+ pcm_set_speed(pcm_speed); /* The speed must be reinitialized */
+ }
+ return pcm_channels;
+}
+
+static int pcm_set_bits(int arg)
+{
+ if (arg == 0)
+ return pcm_bits;
+
+ if ((arg & pcm_bitsok) != arg)
+ return pcm_bits;
+
+ if (arg != pcm_bits)
+ {
+ pas_write(pas_read(0x8389) ^ 0x04, 0x8389);
+
+ pcm_bits = arg;
+ }
+ return pcm_bits;
+}
+
+static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ int val, ret;
+ int __user *p = arg;
+
+ DEB(printk("pas2_pcm.c: static int pas_audio_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg));
+
+ switch (cmd)
+ {
+ case SOUND_PCM_WRITE_RATE:
+ if (get_user(val, p))
+ return -EFAULT;
+ ret = pcm_set_speed(val);
+ break;
+
+ case SOUND_PCM_READ_RATE:
+ ret = pcm_speed;
+ break;
+
+ case SNDCTL_DSP_STEREO:
+ if (get_user(val, p))
+ return -EFAULT;
+ ret = pcm_set_channels(val + 1) - 1;
+ break;
+
+ case SOUND_PCM_WRITE_CHANNELS:
+ if (get_user(val, p))
+ return -EFAULT;
+ ret = pcm_set_channels(val);
+ break;
+
+ case SOUND_PCM_READ_CHANNELS:
+ ret = pcm_channels;
+ break;
+
+ case SNDCTL_DSP_SETFMT:
+ if (get_user(val, p))
+ return -EFAULT;
+ ret = pcm_set_bits(val);
+ break;
+
+ case SOUND_PCM_READ_BITS:
+ ret = pcm_bits;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return put_user(ret, p);
+}
+
+static void pas_audio_reset(int dev)
+{
+ DEB(printk("pas2_pcm.c: static void pas_audio_reset(void)\n"));
+
+ pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); /* Disable PCM */
+}
+
+static int pas_audio_open(int dev, int mode)
+{
+ int err;
+ unsigned long flags;
+
+ DEB(printk("pas2_pcm.c: static int pas_audio_open(int mode = %X)\n", mode));
+
+ spin_lock_irqsave(&pas_lock, flags);
+ if (pcm_busy)
+ {
+ spin_unlock_irqrestore(&pas_lock, flags);
+ return -EBUSY;
+ }
+ pcm_busy = 1;
+ spin_unlock_irqrestore(&pas_lock, flags);
+
+ if ((err = pas_set_intr(PAS_PCM_INTRBITS)) < 0)
+ return err;
+
+
+ pcm_count = 0;
+ open_mode = mode;
+
+ return 0;
+}
+
+static void pas_audio_close(int dev)
+{
+ unsigned long flags;
+
+ DEB(printk("pas2_pcm.c: static void pas_audio_close(void)\n"));
+
+ spin_lock_irqsave(&pas_lock, flags);
+
+ pas_audio_reset(dev);
+ pas_remove_intr(PAS_PCM_INTRBITS);
+ pcm_mode = PCM_NON;
+
+ pcm_busy = 0;
+ spin_unlock_irqrestore(&pas_lock, flags);
+}
+
+static void pas_audio_output_block(int dev, unsigned long buf, int count,
+ int intrflag)
+{
+ unsigned long flags, cnt;
+
+ DEB(printk("pas2_pcm.c: static void pas_audio_output_block(char *buf = %P, int count = %X)\n", buf, count));
+
+ cnt = count;
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ cnt >>= 1;
+
+ if (audio_devs[dev]->flags & DMA_AUTOMODE &&
+ intrflag &&
+ cnt == pcm_count)
+ return;
+
+ spin_lock_irqsave(&pas_lock, flags);
+
+ pas_write(pas_read(0xF8A) & ~0x40,
+ 0xF8A);
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */
+
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ count >>= 1;
+
+ if (count != pcm_count)
+ {
+ pas_write(pas_read(0x0B8A) & ~0x80, 0x0B8A);
+ pas_write(0x40 | 0x30 | 0x04, 0x138B);
+ pas_write(count & 0xff, 0x1389);
+ pas_write((count >> 8) & 0xff, 0x1389);
+ pas_write(pas_read(0x0B8A) | 0x80, 0x0B8A);
+
+ pcm_count = count;
+ }
+ pas_write(pas_read(0x0B8A) | 0x80 | 0x40, 0x0B8A);
+#ifdef NO_TRIGGER
+ pas_write(pas_read(0xF8A) | 0x40 | 0x10, 0xF8A);
+#endif
+
+ pcm_mode = PCM_DAC;
+
+ spin_unlock_irqrestore(&pas_lock, flags);
+}
+
+static void pas_audio_start_input(int dev, unsigned long buf, int count,
+ int intrflag)
+{
+ unsigned long flags;
+ int cnt;
+
+ DEB(printk("pas2_pcm.c: static void pas_audio_start_input(char *buf = %P, int count = %X)\n", buf, count));
+
+ cnt = count;
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ cnt >>= 1;
+
+ if (audio_devs[pas_audiodev]->flags & DMA_AUTOMODE &&
+ intrflag &&
+ cnt == pcm_count)
+ return;
+
+ spin_lock_irqsave(&pas_lock, flags);
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */
+
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ count >>= 1;
+
+ if (count != pcm_count)
+ {
+ pas_write(pas_read(0x0B8A) & ~0x80, 0x0B8A);
+ pas_write(0x40 | 0x30 | 0x04, 0x138B);
+ pas_write(count & 0xff, 0x1389);
+ pas_write((count >> 8) & 0xff, 0x1389);
+ pas_write(pas_read(0x0B8A) | 0x80, 0x0B8A);
+
+ pcm_count = count;
+ }
+ pas_write(pas_read(0x0B8A) | 0x80 | 0x40, 0x0B8A);
+#ifdef NO_TRIGGER
+ pas_write((pas_read(0xF8A) | 0x40) & ~0x10, 0xF8A);
+#endif
+
+ pcm_mode = PCM_ADC;
+
+ spin_unlock_irqrestore(&pas_lock, flags);
+}
+
+#ifndef NO_TRIGGER
+static void pas_audio_trigger(int dev, int state)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&pas_lock, flags);
+ state &= open_mode;
+
+ if (state & PCM_ENABLE_OUTPUT)
+ pas_write(pas_read(0xF8A) | 0x40 | 0x10, 0xF8A);
+ else if (state & PCM_ENABLE_INPUT)
+ pas_write((pas_read(0xF8A) | 0x40) & ~0x10, 0xF8A);
+ else
+ pas_write(pas_read(0xF8A) & ~0x40, 0xF8A);
+
+ spin_unlock_irqrestore(&pas_lock, flags);
+}
+#endif
+
+static int pas_audio_prepare_for_input(int dev, int bsize, int bcount)
+{
+ pas_audio_reset(dev);
+ return 0;
+}
+
+static int pas_audio_prepare_for_output(int dev, int bsize, int bcount)
+{
+ pas_audio_reset(dev);
+ return 0;
+}
+
+static struct audio_driver pas_audio_driver =
+{
+ .owner = THIS_MODULE,
+ .open = pas_audio_open,
+ .close = pas_audio_close,
+ .output_block = pas_audio_output_block,
+ .start_input = pas_audio_start_input,
+ .ioctl = pas_audio_ioctl,
+ .prepare_for_input = pas_audio_prepare_for_input,
+ .prepare_for_output = pas_audio_prepare_for_output,
+ .halt_io = pas_audio_reset,
+ .trigger = pas_audio_trigger
+};
+
+void __init pas_pcm_init(struct address_info *hw_config)
+{
+ DEB(printk("pas2_pcm.c: long pas_pcm_init()\n"));
+
+ pcm_bitsok = 8;
+ if (pas_read(0xEF8B) & 0x08)
+ pcm_bitsok |= 16;
+
+ pcm_set_speed(DSP_DEFAULT_SPEED);
+
+ if ((pas_audiodev = sound_install_audiodrv(AUDIO_DRIVER_VERSION,
+ "Pro Audio Spectrum",
+ &pas_audio_driver,
+ sizeof(struct audio_driver),
+ DMA_AUTOMODE,
+ AFMT_U8 | AFMT_S16_LE,
+ NULL,
+ hw_config->dma,
+ hw_config->dma)) < 0)
+ printk(KERN_WARNING "PAS16: Too many PCM devices available\n");
+}
+
+void pas_pcm_interrupt(unsigned char status, int cause)
+{
+ if (cause == 1)
+ {
+ /*
+ * Halt the PCM first. Otherwise we don't have time to start a new
+ * block before the PCM chip proceeds to the next sample
+ */
+
+ if (!(audio_devs[pas_audiodev]->flags & DMA_AUTOMODE))
+ pas_write(pas_read(0xF8A) & ~0x40, 0xF8A);
+
+ switch (pcm_mode)
+ {
+ case PCM_DAC:
+ DMAbuf_outputintr(pas_audiodev, 1);
+ break;
+
+ case PCM_ADC:
+ DMAbuf_inputintr(pas_audiodev);
+ break;
+
+ default:
+ printk(KERN_WARNING "PAS: Unexpected PCM interrupt\n");
+ }
+ }
+}
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
new file mode 100644
index 0000000..16ed069
--- /dev/null
+++ b/sound/oss/pss.c
@@ -0,0 +1,1266 @@
+/*
+ * sound/oss/pss.c
+ *
+ * The low level driver for the Personal Sound System (ECHO ESC614).
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Thomas Sailer ioctl code reworked (vmalloc/vfree removed)
+ * Alan Cox modularisation, clean up.
+ *
+ * 98-02-21: Vladimir Michl <vladimir.michl@upol.cz>
+ * Added mixer device for Beethoven ADSP-16 (master volume,
+ * bass, treble, synth), only for speakers.
+ * Fixed bug in pss_write (exchange parameters)
+ * Fixed config port of SB
+ * Requested two regions for PSS (PSS mixer, PSS config)
+ * Modified pss_download_boot
+ * To probe_pss_mss added test for initialize AD1848
+ * 98-05-28: Vladimir Michl <vladimir.michl@upol.cz>
+ * Fixed computation of mixer volumes
+ * 04-05-1999: Anthony Barbachan <barbcode@xmen.cis.fordham.edu>
+ * Added code that allows the user to enable his cdrom and/or
+ * joystick through the module parameters pss_cdrom_port and
+ * pss_enable_joystick. pss_cdrom_port takes a port address as its
+ * argument. pss_enable_joystick takes either a 0 or a non-0 as its
+ * argument.
+ * 04-06-1999: Anthony Barbachan <barbcode@xmen.cis.fordham.edu>
+ * Separated some code into new functions for easier reuse.
+ * Cleaned up and streamlined new code. Added code to allow a user
+ * to only use this driver for enabling non-sound components
+ * through the new module parameter pss_no_sound (flag). Added
+ * code that would allow a user to decide whether the driver should
+ * reset the configured hardware settings for the PSS board through
+ * the module parameter pss_keep_settings (flag). This flag will
+ * allow a user to free up resources in use by this card if needbe,
+ * furthermore it allows him to use this driver to just enable the
+ * emulations and then be unloaded as it is no longer needed. Both
+ * new settings are only available to this driver if compiled as a
+ * module. The default settings of all new parameters are set to
+ * load the driver as it did in previous versions.
+ * 04-07-1999: Anthony Barbachan <barbcode@xmen.cis.fordham.edu>
+ * Added module parameter pss_firmware to allow the user to tell
+ * the driver where the fireware file is located. The default
+ * setting is the previous hardcoded setting "/etc/sound/pss_synth".
+ * 00-03-03: Christoph Hellwig <chhellwig@infradead.org>
+ * Adapted to module_init/module_exit
+ * 11-10-2000: Bartlomiej Zolnierkiewicz <bkz@linux-ide.org>
+ * Added __init to probe_pss(), attach_pss() and probe_pss_mpu()
+ * 02-Jan-2001: Chris Rankin
+ * Specify that this module owns the coprocessor
+ */
+
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/spinlock.h>
+
+#include "sound_config.h"
+#include "sound_firmware.h"
+
+#include "ad1848.h"
+#include "mpu401.h"
+
+/*
+ * PSS registers.
+ */
+#define REG(x) (devc->base+x)
+#define PSS_DATA 0
+#define PSS_STATUS 2
+#define PSS_CONTROL 2
+#define PSS_ID 4
+#define PSS_IRQACK 4
+#define PSS_PIO 0x1a
+
+/*
+ * Config registers
+ */
+#define CONF_PSS 0x10
+#define CONF_WSS 0x12
+#define CONF_SB 0x14
+#define CONF_CDROM 0x16
+#define CONF_MIDI 0x18
+
+/*
+ * Status bits.
+ */
+#define PSS_FLAG3 0x0800
+#define PSS_FLAG2 0x0400
+#define PSS_FLAG1 0x1000
+#define PSS_FLAG0 0x0800
+#define PSS_WRITE_EMPTY 0x8000
+#define PSS_READ_FULL 0x4000
+
+/*
+ * WSS registers
+ */
+#define WSS_INDEX 4
+#define WSS_DATA 5
+
+/*
+ * WSS status bits
+ */
+#define WSS_INITIALIZING 0x80
+#define WSS_AUTOCALIBRATION 0x20
+
+#define NO_WSS_MIXER -1
+
+#include "coproc.h"
+
+#include "pss_boot.h"
+
+/* If compiled into kernel, it enable or disable pss mixer */
+#ifdef CONFIG_PSS_MIXER
+static int pss_mixer = 1;
+#else
+static int pss_mixer;
+#endif
+
+
+typedef struct pss_mixerdata {
+ unsigned int volume_l;
+ unsigned int volume_r;
+ unsigned int bass;
+ unsigned int treble;
+ unsigned int synth;
+} pss_mixerdata;
+
+typedef struct pss_confdata {
+ int base;
+ int irq;
+ int dma;
+ int *osp;
+ pss_mixerdata mixer;
+ int ad_mixer_dev;
+} pss_confdata;
+
+static pss_confdata pss_data;
+static pss_confdata *devc = &pss_data;
+static DEFINE_SPINLOCK(lock);
+
+static int pss_initialized;
+static int nonstandard_microcode;
+static int pss_cdrom_port = -1; /* Parameter for the PSS cdrom port */
+static int pss_enable_joystick; /* Parameter for enabling the joystick */
+static coproc_operations pss_coproc_operations;
+
+static void pss_write(pss_confdata *devc, int data)
+{
+ unsigned long i, limit;
+
+ limit = jiffies + HZ/10; /* The timeout is 0.1 seconds */
+ /*
+ * Note! the i<5000000 is an emergency exit. The dsp_command() is sometimes
+ * called while interrupts are disabled. This means that the timer is
+ * disabled also. However the timeout situation is a abnormal condition.
+ * Normally the DSP should be ready to accept commands after just couple of
+ * loops.
+ */
+
+ for (i = 0; i < 5000000 && time_before(jiffies, limit); i++)
+ {
+ if (inw(REG(PSS_STATUS)) & PSS_WRITE_EMPTY)
+ {
+ outw(data, REG(PSS_DATA));
+ return;
+ }
+ }
+ printk(KERN_WARNING "PSS: DSP Command (%04x) Timeout.\n", data);
+}
+
+static int __init probe_pss(struct address_info *hw_config)
+{
+ unsigned short id;
+ int irq, dma;
+
+ devc->base = hw_config->io_base;
+ irq = devc->irq = hw_config->irq;
+ dma = devc->dma = hw_config->dma;
+ devc->osp = hw_config->osp;
+
+ if (devc->base != 0x220 && devc->base != 0x240)
+ if (devc->base != 0x230 && devc->base != 0x250) /* Some cards use these */
+ return 0;
+
+ if (!request_region(devc->base, 0x10, "PSS mixer, SB emulation")) {
+ printk(KERN_ERR "PSS: I/O port conflict\n");
+ return 0;
+ }
+ id = inw(REG(PSS_ID));
+ if ((id >> 8) != 'E') {
+ printk(KERN_ERR "No PSS signature detected at 0x%x (0x%x)\n", devc->base, id);
+ release_region(devc->base, 0x10);
+ return 0;
+ }
+ if (!request_region(devc->base + 0x10, 0x9, "PSS config")) {
+ printk(KERN_ERR "PSS: I/O port conflict\n");
+ release_region(devc->base, 0x10);
+ return 0;
+ }
+ return 1;
+}
+
+static int set_irq(pss_confdata * devc, int dev, int irq)
+{
+ static unsigned short irq_bits[16] =
+ {
+ 0x0000, 0x0000, 0x0000, 0x0008,
+ 0x0000, 0x0010, 0x0000, 0x0018,
+ 0x0000, 0x0020, 0x0028, 0x0030,
+ 0x0038, 0x0000, 0x0000, 0x0000
+ };
+
+ unsigned short tmp, bits;
+
+ if (irq < 0 || irq > 15)
+ return 0;
+
+ tmp = inw(REG(dev)) & ~0x38; /* Load confreg, mask IRQ bits out */
+
+ if ((bits = irq_bits[irq]) == 0 && irq != 0)
+ {
+ printk(KERN_ERR "PSS: Invalid IRQ %d\n", irq);
+ return 0;
+ }
+ outw(tmp | bits, REG(dev));
+ return 1;
+}
+
+static void set_io_base(pss_confdata * devc, int dev, int base)
+{
+ unsigned short tmp = inw(REG(dev)) & 0x003f;
+ unsigned short bits = (base & 0x0ffc) << 4;
+
+ outw(bits | tmp, REG(dev));
+}
+
+static int set_dma(pss_confdata * devc, int dev, int dma)
+{
+ static unsigned short dma_bits[8] =
+ {
+ 0x0001, 0x0002, 0x0000, 0x0003,
+ 0x0000, 0x0005, 0x0006, 0x0007
+ };
+
+ unsigned short tmp, bits;
+
+ if (dma < 0 || dma > 7)
+ return 0;
+
+ tmp = inw(REG(dev)) & ~0x07; /* Load confreg, mask DMA bits out */
+
+ if ((bits = dma_bits[dma]) == 0 && dma != 4)
+ {
+ printk(KERN_ERR "PSS: Invalid DMA %d\n", dma);
+ return 0;
+ }
+ outw(tmp | bits, REG(dev));
+ return 1;
+}
+
+static int pss_reset_dsp(pss_confdata * devc)
+{
+ unsigned long i, limit = jiffies + HZ/10;
+
+ outw(0x2000, REG(PSS_CONTROL));
+ for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+ inw(REG(PSS_CONTROL));
+ outw(0x0000, REG(PSS_CONTROL));
+ return 1;
+}
+
+static int pss_put_dspword(pss_confdata * devc, unsigned short word)
+{
+ int i, val;
+
+ for (i = 0; i < 327680; i++)
+ {
+ val = inw(REG(PSS_STATUS));
+ if (val & PSS_WRITE_EMPTY)
+ {
+ outw(word, REG(PSS_DATA));
+ return 1;
+ }
+ }
+ return 0;
+}
+
+static int pss_get_dspword(pss_confdata * devc, unsigned short *word)
+{
+ int i, val;
+
+ for (i = 0; i < 327680; i++)
+ {
+ val = inw(REG(PSS_STATUS));
+ if (val & PSS_READ_FULL)
+ {
+ *word = inw(REG(PSS_DATA));
+ return 1;
+ }
+ }
+ return 0;
+}
+
+static int pss_download_boot(pss_confdata * devc, unsigned char *block, int size, int flags)
+{
+ int i, val, count;
+ unsigned long limit;
+
+ if (flags & CPF_FIRST)
+ {
+/*_____ Warn DSP software that a boot is coming */
+ outw(0x00fe, REG(PSS_DATA));
+
+ limit = jiffies + HZ/10;
+ for (i = 0; i < 32768 && time_before(jiffies, limit); i++)
+ if (inw(REG(PSS_DATA)) == 0x5500)
+ break;
+
+ outw(*block++, REG(PSS_DATA));
+ pss_reset_dsp(devc);
+ }
+ count = 1;
+ while ((flags&CPF_LAST) || count<size )
+ {
+ int j;
+
+ for (j = 0; j < 327670; j++)
+ {
+/*_____ Wait for BG to appear */
+ if (inw(REG(PSS_STATUS)) & PSS_FLAG3)
+ break;
+ }
+
+ if (j == 327670)
+ {
+ /* It's ok we timed out when the file was empty */
+ if (count >= size && flags & CPF_LAST)
+ break;
+ else
+ {
+ printk("\n");
+ printk(KERN_ERR "PSS: Download timeout problems, byte %d=%d\n", count, size);
+ return 0;
+ }
+ }
+/*_____ Send the next byte */
+ if (count >= size)
+ {
+ /* If not data in block send 0xffff */
+ outw (0xffff, REG (PSS_DATA));
+ }
+ else
+ {
+ /*_____ Send the next byte */
+ outw (*block++, REG (PSS_DATA));
+ };
+ count++;
+ }
+
+ if (flags & CPF_LAST)
+ {
+/*_____ Why */
+ outw(0, REG(PSS_DATA));
+
+ limit = jiffies + HZ/10;
+ for (i = 0; i < 32768 && (limit - jiffies >= 0); i++)
+ val = inw(REG(PSS_STATUS));
+
+ limit = jiffies + HZ/10;
+ for (i = 0; i < 32768 && (limit-jiffies >= 0); i++)
+ {
+ val = inw(REG(PSS_STATUS));
+ if (val & 0x4000)
+ break;
+ }
+
+ /* now read the version */
+ for (i = 0; i < 32000; i++)
+ {
+ val = inw(REG(PSS_STATUS));
+ if (val & PSS_READ_FULL)
+ break;
+ }
+ if (i == 32000)
+ return 0;
+
+ val = inw(REG(PSS_DATA));
+ /* printk( "<PSS: microcode version %d.%d loaded>", val/16, val % 16); */
+ }
+ return 1;
+}
+
+/* Mixer */
+static void set_master_volume(pss_confdata *devc, int left, int right)
+{
+ static unsigned char log_scale[101] = {
+ 0xdb, 0xe0, 0xe3, 0xe5, 0xe7, 0xe9, 0xea, 0xeb, 0xec, 0xed, 0xed, 0xee,
+ 0xef, 0xef, 0xf0, 0xf0, 0xf1, 0xf1, 0xf2, 0xf2, 0xf2, 0xf3, 0xf3, 0xf3,
+ 0xf4, 0xf4, 0xf4, 0xf5, 0xf5, 0xf5, 0xf5, 0xf6, 0xf6, 0xf6, 0xf6, 0xf7,
+ 0xf7, 0xf7, 0xf7, 0xf7, 0xf8, 0xf8, 0xf8, 0xf8, 0xf8, 0xf9, 0xf9, 0xf9,
+ 0xf9, 0xf9, 0xf9, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfa, 0xfb, 0xfb,
+ 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfb, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc, 0xfc,
+ 0xfc, 0xfc, 0xfc, 0xfc, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd, 0xfd,
+ 0xfd, 0xfd, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe, 0xfe,
+ 0xfe, 0xfe, 0xff, 0xff, 0xff
+ };
+ pss_write(devc, 0x0010);
+ pss_write(devc, log_scale[left] | 0x0000);
+ pss_write(devc, 0x0010);
+ pss_write(devc, log_scale[right] | 0x0100);
+}
+
+static void set_synth_volume(pss_confdata *devc, int volume)
+{
+ int vol = ((0x8000*volume)/100L);
+ pss_write(devc, 0x0080);
+ pss_write(devc, vol);
+ pss_write(devc, 0x0081);
+ pss_write(devc, vol);
+}
+
+static void set_bass(pss_confdata *devc, int level)
+{
+ int vol = (int)(((0xfd - 0xf0) * level)/100L) + 0xf0;
+ pss_write(devc, 0x0010);
+ pss_write(devc, vol | 0x0200);
+};
+
+static void set_treble(pss_confdata *devc, int level)
+{
+ int vol = (((0xfd - 0xf0) * level)/100L) + 0xf0;
+ pss_write(devc, 0x0010);
+ pss_write(devc, vol | 0x0300);
+};
+
+static void pss_mixer_reset(pss_confdata *devc)
+{
+ set_master_volume(devc, 33, 33);
+ set_bass(devc, 50);
+ set_treble(devc, 50);
+ set_synth_volume(devc, 30);
+ pss_write (devc, 0x0010);
+ pss_write (devc, 0x0800 | 0xce); /* Stereo */
+
+ if(pss_mixer)
+ {
+ devc->mixer.volume_l = devc->mixer.volume_r = 33;
+ devc->mixer.bass = 50;
+ devc->mixer.treble = 50;
+ devc->mixer.synth = 30;
+ }
+}
+
+static int set_volume_mono(unsigned __user *p, int *aleft)
+{
+ int left;
+ unsigned volume;
+ if (get_user(volume, p))
+ return -EFAULT;
+
+ left = volume & 0xff;
+ if (left > 100)
+ left = 100;
+ *aleft = left;
+ return 0;
+}
+
+static int set_volume_stereo(unsigned __user *p, int *aleft, int *aright)
+{
+ int left, right;
+ unsigned volume;
+ if (get_user(volume, p))
+ return -EFAULT;
+
+ left = volume & 0xff;
+ if (left > 100)
+ left = 100;
+ right = (volume >> 8) & 0xff;
+ if (right > 100)
+ right = 100;
+ *aleft = left;
+ *aright = right;
+ return 0;
+}
+
+static int ret_vol_mono(int left)
+{
+ return ((left << 8) | left);
+}
+
+static int ret_vol_stereo(int left, int right)
+{
+ return ((right << 8) | left);
+}
+
+static int call_ad_mixer(pss_confdata *devc,unsigned int cmd, void __user *arg)
+{
+ if (devc->ad_mixer_dev != NO_WSS_MIXER)
+ return mixer_devs[devc->ad_mixer_dev]->ioctl(devc->ad_mixer_dev, cmd, arg);
+ else
+ return -EINVAL;
+}
+
+static int pss_mixer_ioctl (int dev, unsigned int cmd, void __user *arg)
+{
+ pss_confdata *devc = mixer_devs[dev]->devc;
+ int cmdf = cmd & 0xff;
+
+ if ((cmdf != SOUND_MIXER_VOLUME) && (cmdf != SOUND_MIXER_BASS) &&
+ (cmdf != SOUND_MIXER_TREBLE) && (cmdf != SOUND_MIXER_SYNTH) &&
+ (cmdf != SOUND_MIXER_DEVMASK) && (cmdf != SOUND_MIXER_STEREODEVS) &&
+ (cmdf != SOUND_MIXER_RECMASK) && (cmdf != SOUND_MIXER_CAPS) &&
+ (cmdf != SOUND_MIXER_RECSRC))
+ {
+ return call_ad_mixer(devc, cmd, arg);
+ }
+
+ if (((cmd >> 8) & 0xff) != 'M')
+ return -EINVAL;
+
+ if (_SIOC_DIR (cmd) & _SIOC_WRITE)
+ {
+ switch (cmdf)
+ {
+ case SOUND_MIXER_RECSRC:
+ if (devc->ad_mixer_dev != NO_WSS_MIXER)
+ return call_ad_mixer(devc, cmd, arg);
+ else
+ {
+ int v;
+ if (get_user(v, (int __user *)arg))
+ return -EFAULT;
+ if (v != 0)
+ return -EINVAL;
+ return 0;
+ }
+ case SOUND_MIXER_VOLUME:
+ if (set_volume_stereo(arg,
+ &devc->mixer.volume_l,
+ &devc->mixer.volume_r))
+ return -EFAULT;
+ set_master_volume(devc, devc->mixer.volume_l,
+ devc->mixer.volume_r);
+ return ret_vol_stereo(devc->mixer.volume_l,
+ devc->mixer.volume_r);
+
+ case SOUND_MIXER_BASS:
+ if (set_volume_mono(arg, &devc->mixer.bass))
+ return -EFAULT;
+ set_bass(devc, devc->mixer.bass);
+ return ret_vol_mono(devc->mixer.bass);
+
+ case SOUND_MIXER_TREBLE:
+ if (set_volume_mono(arg, &devc->mixer.treble))
+ return -EFAULT;
+ set_treble(devc, devc->mixer.treble);
+ return ret_vol_mono(devc->mixer.treble);
+
+ case SOUND_MIXER_SYNTH:
+ if (set_volume_mono(arg, &devc->mixer.synth))
+ return -EFAULT;
+ set_synth_volume(devc, devc->mixer.synth);
+ return ret_vol_mono(devc->mixer.synth);
+
+ default:
+ return -EINVAL;
+ }
+ }
+ else
+ {
+ int val, and_mask = 0, or_mask = 0;
+ /*
+ * Return parameters
+ */
+ switch (cmdf)
+ {
+ case SOUND_MIXER_DEVMASK:
+ if (call_ad_mixer(devc, cmd, arg) == -EINVAL)
+ break;
+ and_mask = ~0;
+ or_mask = SOUND_MASK_VOLUME | SOUND_MASK_BASS | SOUND_MASK_TREBLE | SOUND_MASK_SYNTH;
+ break;
+
+ case SOUND_MIXER_STEREODEVS:
+ if (call_ad_mixer(devc, cmd, arg) == -EINVAL)
+ break;
+ and_mask = ~0;
+ or_mask = SOUND_MASK_VOLUME;
+ break;
+
+ case SOUND_MIXER_RECMASK:
+ if (devc->ad_mixer_dev != NO_WSS_MIXER)
+ return call_ad_mixer(devc, cmd, arg);
+ break;
+
+ case SOUND_MIXER_CAPS:
+ if (devc->ad_mixer_dev != NO_WSS_MIXER)
+ return call_ad_mixer(devc, cmd, arg);
+ or_mask = SOUND_CAP_EXCL_INPUT;
+ break;
+
+ case SOUND_MIXER_RECSRC:
+ if (devc->ad_mixer_dev != NO_WSS_MIXER)
+ return call_ad_mixer(devc, cmd, arg);
+ break;
+
+ case SOUND_MIXER_VOLUME:
+ or_mask = ret_vol_stereo(devc->mixer.volume_l, devc->mixer.volume_r);
+ break;
+
+ case SOUND_MIXER_BASS:
+ or_mask = ret_vol_mono(devc->mixer.bass);
+ break;
+
+ case SOUND_MIXER_TREBLE:
+ or_mask = ret_vol_mono(devc->mixer.treble);
+ break;
+
+ case SOUND_MIXER_SYNTH:
+ or_mask = ret_vol_mono(devc->mixer.synth);
+ break;
+ default:
+ return -EINVAL;
+ }
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+ val &= and_mask;
+ val |= or_mask;
+ if (put_user(val, (int __user *)arg))
+ return -EFAULT;
+ return val;
+ }
+}
+
+static struct mixer_operations pss_mixer_operations =
+{
+ .owner = THIS_MODULE,
+ .id = "SOUNDPORT",
+ .name = "PSS-AD1848",
+ .ioctl = pss_mixer_ioctl
+};
+
+static void disable_all_emulations(void)
+{
+ outw(0x0000, REG(CONF_PSS)); /* 0x0400 enables joystick */
+ outw(0x0000, REG(CONF_WSS));
+ outw(0x0000, REG(CONF_SB));
+ outw(0x0000, REG(CONF_MIDI));
+ outw(0x0000, REG(CONF_CDROM));
+}
+
+static void configure_nonsound_components(void)
+{
+ /* Configure Joystick port */
+
+ if(pss_enable_joystick)
+ {
+ outw(0x0400, REG(CONF_PSS)); /* 0x0400 enables joystick */
+ printk(KERN_INFO "PSS: joystick enabled.\n");
+ }
+ else
+ {
+ printk(KERN_INFO "PSS: joystick port not enabled.\n");
+ }
+
+ /* Configure CDROM port */
+
+ if (pss_cdrom_port == -1) { /* If cdrom port enablation wasn't requested */
+ printk(KERN_INFO "PSS: CDROM port not enabled.\n");
+ } else if (check_region(pss_cdrom_port, 2)) {
+ printk(KERN_ERR "PSS: CDROM I/O port conflict.\n");
+ } else {
+ set_io_base(devc, CONF_CDROM, pss_cdrom_port);
+ printk(KERN_INFO "PSS: CDROM I/O port set to 0x%x.\n", pss_cdrom_port);
+ }
+}
+
+static int __init attach_pss(struct address_info *hw_config)
+{
+ unsigned short id;
+ char tmp[100];
+
+ devc->base = hw_config->io_base;
+ devc->irq = hw_config->irq;
+ devc->dma = hw_config->dma;
+ devc->osp = hw_config->osp;
+ devc->ad_mixer_dev = NO_WSS_MIXER;
+
+ if (!probe_pss(hw_config))
+ return 0;
+
+ id = inw(REG(PSS_ID)) & 0x00ff;
+
+ /*
+ * Disable all emulations. Will be enabled later (if required).
+ */
+
+ disable_all_emulations();
+
+#ifdef YOU_REALLY_WANT_TO_ALLOCATE_THESE_RESOURCES
+ if (sound_alloc_dma(hw_config->dma, "PSS"))
+ {
+ printk("pss.c: Can't allocate DMA channel.\n");
+ release_region(hw_config->io_base, 0x10);
+ release_region(hw_config->io_base+0x10, 0x9);
+ return 0;
+ }
+ if (!set_irq(devc, CONF_PSS, devc->irq))
+ {
+ printk("PSS: IRQ allocation error.\n");
+ release_region(hw_config->io_base, 0x10);
+ release_region(hw_config->io_base+0x10, 0x9);
+ return 0;
+ }
+ if (!set_dma(devc, CONF_PSS, devc->dma))
+ {
+ printk(KERN_ERR "PSS: DMA allocation error\n");
+ release_region(hw_config->io_base, 0x10);
+ release_region(hw_config->io_base+0x10, 0x9);
+ return 0;
+ }
+#endif
+
+ configure_nonsound_components();
+ pss_initialized = 1;
+ sprintf(tmp, "ECHO-PSS Rev. %d", id);
+ conf_printf(tmp, hw_config);
+ return 1;
+}
+
+static int __init probe_pss_mpu(struct address_info *hw_config)
+{
+ struct resource *ports;
+ int timeout;
+
+ if (!pss_initialized)
+ return 0;
+
+ ports = request_region(hw_config->io_base, 2, "mpu401");
+
+ if (!ports) {
+ printk(KERN_ERR "PSS: MPU I/O port conflict\n");
+ return 0;
+ }
+ set_io_base(devc, CONF_MIDI, hw_config->io_base);
+ if (!set_irq(devc, CONF_MIDI, hw_config->irq)) {
+ printk(KERN_ERR "PSS: MIDI IRQ allocation error.\n");
+ goto fail;
+ }
+ if (!pss_synthLen) {
+ printk(KERN_ERR "PSS: Can't enable MPU. MIDI synth microcode not available.\n");
+ goto fail;
+ }
+ if (!pss_download_boot(devc, pss_synth, pss_synthLen, CPF_FIRST | CPF_LAST)) {
+ printk(KERN_ERR "PSS: Unable to load MIDI synth microcode to DSP.\n");
+ goto fail;
+ }
+
+ /*
+ * Finally wait until the DSP algorithm has initialized itself and
+ * deactivates receive interrupt.
+ */
+
+ for (timeout = 900000; timeout > 0; timeout--)
+ {
+ if ((inb(hw_config->io_base + 1) & 0x80) == 0) /* Input data avail */
+ inb(hw_config->io_base); /* Discard it */
+ else
+ break; /* No more input */
+ }
+
+ if (!probe_mpu401(hw_config, ports))
+ goto fail;
+
+ attach_mpu401(hw_config, THIS_MODULE); /* Slot 1 */
+ if (hw_config->slots[1] != -1) /* The MPU driver installed itself */
+ midi_devs[hw_config->slots[1]]->coproc = &pss_coproc_operations;
+ return 1;
+fail:
+ release_region(hw_config->io_base, 2);
+ return 0;
+}
+
+static int pss_coproc_open(void *dev_info, int sub_device)
+{
+ switch (sub_device)
+ {
+ case COPR_MIDI:
+ if (pss_synthLen == 0)
+ {
+ printk(KERN_ERR "PSS: MIDI synth microcode not available.\n");
+ return -EIO;
+ }
+ if (nonstandard_microcode)
+ if (!pss_download_boot(devc, pss_synth, pss_synthLen, CPF_FIRST | CPF_LAST))
+ {
+ printk(KERN_ERR "PSS: Unable to load MIDI synth microcode to DSP.\n");
+ return -EIO;
+ }
+ nonstandard_microcode = 0;
+ break;
+
+ default:
+ break;
+ }
+ return 0;
+}
+
+static void pss_coproc_close(void *dev_info, int sub_device)
+{
+ return;
+}
+
+static void pss_coproc_reset(void *dev_info)
+{
+ if (pss_synthLen)
+ if (!pss_download_boot(devc, pss_synth, pss_synthLen, CPF_FIRST | CPF_LAST))
+ {
+ printk(KERN_ERR "PSS: Unable to load MIDI synth microcode to DSP.\n");
+ }
+ nonstandard_microcode = 0;
+}
+
+static int download_boot_block(void *dev_info, copr_buffer * buf)
+{
+ if (buf->len <= 0 || buf->len > sizeof(buf->data))
+ return -EINVAL;
+
+ if (!pss_download_boot(devc, buf->data, buf->len, buf->flags))
+ {
+ printk(KERN_ERR "PSS: Unable to load microcode block to DSP.\n");
+ return -EIO;
+ }
+ nonstandard_microcode = 1; /* The MIDI microcode has been overwritten */
+ return 0;
+}
+
+static int pss_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, int local)
+{
+ copr_buffer *buf;
+ copr_msg *mbuf;
+ copr_debug_buf dbuf;
+ unsigned short tmp;
+ unsigned long flags;
+ unsigned short *data;
+ int i, err;
+ /* printk( "PSS coproc ioctl %x %x %d\n", cmd, arg, local); */
+
+ switch (cmd)
+ {
+ case SNDCTL_COPR_RESET:
+ pss_coproc_reset(dev_info);
+ return 0;
+
+ case SNDCTL_COPR_LOAD:
+ buf = (copr_buffer *) vmalloc(sizeof(copr_buffer));
+ if (buf == NULL)
+ return -ENOSPC;
+ if (copy_from_user(buf, arg, sizeof(copr_buffer))) {
+ vfree(buf);
+ return -EFAULT;
+ }
+ err = download_boot_block(dev_info, buf);
+ vfree(buf);
+ return err;
+
+ case SNDCTL_COPR_SENDMSG:
+ mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+ if (mbuf == NULL)
+ return -ENOSPC;
+ if (copy_from_user(mbuf, arg, sizeof(copr_msg))) {
+ vfree(mbuf);
+ return -EFAULT;
+ }
+ data = (unsigned short *)(mbuf->data);
+ spin_lock_irqsave(&lock, flags);
+ for (i = 0; i < mbuf->len; i++) {
+ if (!pss_put_dspword(devc, *data++)) {
+ spin_unlock_irqrestore(&lock,flags);
+ mbuf->len = i; /* feed back number of WORDs sent */
+ err = copy_to_user(arg, mbuf, sizeof(copr_msg));
+ vfree(mbuf);
+ return err ? -EFAULT : -EIO;
+ }
+ }
+ spin_unlock_irqrestore(&lock,flags);
+ vfree(mbuf);
+ return 0;
+
+ case SNDCTL_COPR_RCVMSG:
+ err = 0;
+ mbuf = (copr_msg *)vmalloc(sizeof(copr_msg));
+ if (mbuf == NULL)
+ return -ENOSPC;
+ data = (unsigned short *)mbuf->data;
+ spin_lock_irqsave(&lock, flags);
+ for (i = 0; i < sizeof(mbuf->data)/sizeof(unsigned short); i++) {
+ mbuf->len = i; /* feed back number of WORDs read */
+ if (!pss_get_dspword(devc, data++)) {
+ if (i == 0)
+ err = -EIO;
+ break;
+ }
+ }
+ spin_unlock_irqrestore(&lock,flags);
+ if (copy_to_user(arg, mbuf, sizeof(copr_msg)))
+ err = -EFAULT;
+ vfree(mbuf);
+ return err;
+
+ case SNDCTL_COPR_RDATA:
+ if (copy_from_user(&dbuf, arg, sizeof(dbuf)))
+ return -EFAULT;
+ spin_lock_irqsave(&lock, flags);
+ if (!pss_put_dspword(devc, 0x00d0)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ if (!pss_put_dspword(devc, (unsigned short)(dbuf.parm1 & 0xffff))) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ if (!pss_get_dspword(devc, &tmp)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ dbuf.parm1 = tmp;
+ spin_unlock_irqrestore(&lock,flags);
+ if (copy_to_user(arg, &dbuf, sizeof(dbuf)))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_COPR_WDATA:
+ if (copy_from_user(&dbuf, arg, sizeof(dbuf)))
+ return -EFAULT;
+ spin_lock_irqsave(&lock, flags);
+ if (!pss_put_dspword(devc, 0x00d1)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ if (!pss_put_dspword(devc, (unsigned short) (dbuf.parm1 & 0xffff))) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ tmp = (unsigned int)dbuf.parm2 & 0xffff;
+ if (!pss_put_dspword(devc, tmp)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ spin_unlock_irqrestore(&lock,flags);
+ return 0;
+
+ case SNDCTL_COPR_WCODE:
+ if (copy_from_user(&dbuf, arg, sizeof(dbuf)))
+ return -EFAULT;
+ spin_lock_irqsave(&lock, flags);
+ if (!pss_put_dspword(devc, 0x00d3)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ if (!pss_put_dspword(devc, (unsigned short)(dbuf.parm1 & 0xffff))) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ tmp = (unsigned int)dbuf.parm2 & 0x00ff;
+ if (!pss_put_dspword(devc, tmp)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ tmp = ((unsigned int)dbuf.parm2 >> 8) & 0xffff;
+ if (!pss_put_dspword(devc, tmp)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ spin_unlock_irqrestore(&lock,flags);
+ return 0;
+
+ case SNDCTL_COPR_RCODE:
+ if (copy_from_user(&dbuf, arg, sizeof(dbuf)))
+ return -EFAULT;
+ spin_lock_irqsave(&lock, flags);
+ if (!pss_put_dspword(devc, 0x00d2)) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ if (!pss_put_dspword(devc, (unsigned short)(dbuf.parm1 & 0xffff))) {
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ if (!pss_get_dspword(devc, &tmp)) { /* Read MSB */
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ dbuf.parm1 = tmp << 8;
+ if (!pss_get_dspword(devc, &tmp)) { /* Read LSB */
+ spin_unlock_irqrestore(&lock,flags);
+ return -EIO;
+ }
+ dbuf.parm1 |= tmp & 0x00ff;
+ spin_unlock_irqrestore(&lock,flags);
+ if (copy_to_user(arg, &dbuf, sizeof(dbuf)))
+ return -EFAULT;
+ return 0;
+
+ default:
+ return -EINVAL;
+ }
+ return -EINVAL;
+}
+
+static coproc_operations pss_coproc_operations =
+{
+ "ADSP-2115",
+ THIS_MODULE,
+ pss_coproc_open,
+ pss_coproc_close,
+ pss_coproc_ioctl,
+ pss_coproc_reset,
+ &pss_data
+};
+
+static int __init probe_pss_mss(struct address_info *hw_config)
+{
+ volatile int timeout;
+ struct resource *ports;
+ int my_mix = -999; /* gcc shut up */
+
+ if (!pss_initialized)
+ return 0;
+
+ if (!request_region(hw_config->io_base, 4, "WSS config")) {
+ printk(KERN_ERR "PSS: WSS I/O port conflicts.\n");
+ return 0;
+ }
+ ports = request_region(hw_config->io_base + 4, 4, "ad1848");
+ if (!ports) {
+ printk(KERN_ERR "PSS: WSS I/O port conflicts.\n");
+ release_region(hw_config->io_base, 4);
+ return 0;
+ }
+ set_io_base(devc, CONF_WSS, hw_config->io_base);
+ if (!set_irq(devc, CONF_WSS, hw_config->irq)) {
+ printk("PSS: WSS IRQ allocation error.\n");
+ goto fail;
+ }
+ if (!set_dma(devc, CONF_WSS, hw_config->dma)) {
+ printk(KERN_ERR "PSS: WSS DMA allocation error\n");
+ goto fail;
+ }
+ /*
+ * For some reason the card returns 0xff in the WSS status register
+ * immediately after boot. Probably MIDI+SB emulation algorithm
+ * downloaded to the ADSP2115 spends some time initializing the card.
+ * Let's try to wait until it finishes this task.
+ */
+ for (timeout = 0; timeout < 100000 && (inb(hw_config->io_base + WSS_INDEX) &
+ WSS_INITIALIZING); timeout++)
+ ;
+
+ outb((0x0b), hw_config->io_base + WSS_INDEX); /* Required by some cards */
+
+ for (timeout = 0; (inb(hw_config->io_base + WSS_DATA) & WSS_AUTOCALIBRATION) &&
+ (timeout < 100000); timeout++)
+ ;
+
+ if (!probe_ms_sound(hw_config, ports))
+ goto fail;
+
+ devc->ad_mixer_dev = NO_WSS_MIXER;
+ if (pss_mixer)
+ {
+ if ((my_mix = sound_install_mixer (MIXER_DRIVER_VERSION,
+ "PSS-SPEAKERS and AD1848 (through MSS audio codec)",
+ &pss_mixer_operations,
+ sizeof (struct mixer_operations),
+ devc)) < 0)
+ {
+ printk(KERN_ERR "Could not install PSS mixer\n");
+ goto fail;
+ }
+ }
+ pss_mixer_reset(devc);
+ attach_ms_sound(hw_config, ports, THIS_MODULE); /* Slot 0 */
+
+ if (hw_config->slots[0] != -1)
+ {
+ /* The MSS driver installed itself */
+ audio_devs[hw_config->slots[0]]->coproc = &pss_coproc_operations;
+ if (pss_mixer && (num_mixers == (my_mix + 2)))
+ {
+ /* The MSS mixer installed */
+ devc->ad_mixer_dev = audio_devs[hw_config->slots[0]]->mixer_dev;
+ }
+ }
+ return 1;
+fail:
+ release_region(hw_config->io_base + 4, 4);
+ release_region(hw_config->io_base, 4);
+ return 0;
+}
+
+static inline void __exit unload_pss(struct address_info *hw_config)
+{
+ release_region(hw_config->io_base, 0x10);
+ release_region(hw_config->io_base+0x10, 0x9);
+}
+
+static inline void __exit unload_pss_mpu(struct address_info *hw_config)
+{
+ unload_mpu401(hw_config);
+}
+
+static inline void __exit unload_pss_mss(struct address_info *hw_config)
+{
+ unload_ms_sound(hw_config);
+}
+
+
+static struct address_info cfg;
+static struct address_info cfg2;
+static struct address_info cfg_mpu;
+
+static int pss_io __initdata = -1;
+static int mss_io __initdata = -1;
+static int mss_irq __initdata = -1;
+static int mss_dma __initdata = -1;
+static int mpu_io __initdata = -1;
+static int mpu_irq __initdata = -1;
+static int pss_no_sound = 0; /* Just configure non-sound components */
+static int pss_keep_settings = 1; /* Keep hardware settings at module exit */
+static char *pss_firmware = "/etc/sound/pss_synth";
+
+module_param(pss_io, int, 0);
+MODULE_PARM_DESC(pss_io, "Set i/o base of PSS card (probably 0x220 or 0x240)");
+module_param(mss_io, int, 0);
+MODULE_PARM_DESC(mss_io, "Set WSS (audio) i/o base (0x530, 0x604, 0xE80, 0xF40, or other. Address must end in 0 or 4 and must be from 0x100 to 0xFF4)");
+module_param(mss_irq, int, 0);
+MODULE_PARM_DESC(mss_irq, "Set WSS (audio) IRQ (3, 5, 7, 9, 10, 11, 12)");
+module_param(mss_dma, int, 0);
+MODULE_PARM_DESC(mss_dma, "Set WSS (audio) DMA (0, 1, 3)");
+module_param(mpu_io, int, 0);
+MODULE_PARM_DESC(mpu_io, "Set MIDI i/o base (0x330 or other. Address must be on 4 location boundaries and must be from 0x100 to 0xFFC)");
+module_param(mpu_irq, int, 0);
+MODULE_PARM_DESC(mpu_irq, "Set MIDI IRQ (3, 5, 7, 9, 10, 11, 12)");
+module_param(pss_cdrom_port, int, 0);
+MODULE_PARM_DESC(pss_cdrom_port, "Set the PSS CDROM port i/o base (0x340 or other)");
+module_param(pss_enable_joystick, bool, 0);
+MODULE_PARM_DESC(pss_enable_joystick, "Enables the PSS joystick port (1 to enable, 0 to disable)");
+module_param(pss_no_sound, bool, 0);
+MODULE_PARM_DESC(pss_no_sound, "Configure sound compoents (0 - no, 1 - yes)");
+module_param(pss_keep_settings, bool, 0);
+MODULE_PARM_DESC(pss_keep_settings, "Keep hardware setting at driver unloading (0 - no, 1 - yes)");
+module_param(pss_firmware, charp, 0);
+MODULE_PARM_DESC(pss_firmware, "Location of the firmware file (default - /etc/sound/pss_synth)");
+module_param(pss_mixer, bool, 0);
+MODULE_PARM_DESC(pss_mixer, "Enable (1) or disable (0) PSS mixer (controlling of output volume, bass, treble, synth volume). The mixer is not available on all PSS cards.");
+MODULE_AUTHOR("Hannu Savolainen, Vladimir Michl");
+MODULE_DESCRIPTION("Module for PSS sound cards (based on AD1848, ADSP-2115 and ESC614). This module includes control of output amplifier and synth volume of the Beethoven ADSP-16 card (this may work with other PSS cards).");
+MODULE_LICENSE("GPL");
+
+
+static int fw_load = 0;
+static int pssmpu = 0, pssmss = 0;
+
+/*
+ * Load a PSS sound card module
+ */
+
+static int __init init_pss(void)
+{
+
+ if(pss_no_sound) /* If configuring only nonsound components */
+ {
+ cfg.io_base = pss_io;
+ if(!probe_pss(&cfg))
+ return -ENODEV;
+ printk(KERN_INFO "ECHO-PSS Rev. %d\n", inw(REG(PSS_ID)) & 0x00ff);
+ printk(KERN_INFO "PSS: loading in no sound mode.\n");
+ disable_all_emulations();
+ configure_nonsound_components();
+ release_region(pss_io, 0x10);
+ release_region(pss_io + 0x10, 0x9);
+ return 0;
+ }
+
+ cfg.io_base = pss_io;
+
+ cfg2.io_base = mss_io;
+ cfg2.irq = mss_irq;
+ cfg2.dma = mss_dma;
+
+ cfg_mpu.io_base = mpu_io;
+ cfg_mpu.irq = mpu_irq;
+
+ if (cfg.io_base == -1 || cfg2.io_base == -1 || cfg2.irq == -1 || cfg.dma == -1) {
+ printk(KERN_INFO "pss: mss_io, mss_dma, mss_irq and pss_io must be set.\n");
+ return -EINVAL;
+ }
+
+ if (!pss_synth) {
+ fw_load = 1;
+ pss_synthLen = mod_firmware_load(pss_firmware, (void *) &pss_synth);
+ }
+ if (!attach_pss(&cfg))
+ return -ENODEV;
+ /*
+ * Attach stuff
+ */
+ if (probe_pss_mpu(&cfg_mpu))
+ pssmpu = 1;
+
+ if (probe_pss_mss(&cfg2))
+ pssmss = 1;
+
+ return 0;
+}
+
+static void __exit cleanup_pss(void)
+{
+ if(!pss_no_sound)
+ {
+ if(fw_load && pss_synth)
+ vfree(pss_synth);
+ if(pssmss)
+ unload_pss_mss(&cfg2);
+ if(pssmpu)
+ unload_pss_mpu(&cfg_mpu);
+ unload_pss(&cfg);
+ }
+
+ if(!pss_keep_settings) /* Keep hardware settings if asked */
+ {
+ disable_all_emulations();
+ printk(KERN_INFO "Resetting PSS sound card configurations.\n");
+ }
+}
+
+module_init(init_pss);
+module_exit(cleanup_pss);
+
+#ifndef MODULE
+static int __init setup_pss(char *str)
+{
+ /* io, mss_io, mss_irq, mss_dma, mpu_io, mpu_irq */
+ int ints[7];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ pss_io = ints[1];
+ mss_io = ints[2];
+ mss_irq = ints[3];
+ mss_dma = ints[4];
+ mpu_io = ints[5];
+ mpu_irq = ints[6];
+
+ return 1;
+}
+
+__setup("pss=", setup_pss);
+#endif
diff --git a/sound/oss/sb.h b/sound/oss/sb.h
new file mode 100644
index 0000000..77e8891
--- /dev/null
+++ b/sound/oss/sb.h
@@ -0,0 +1,185 @@
+#define DSP_RESET (devc->base + 0x6)
+#define DSP_READ (devc->base + 0xA)
+#define DSP_WRITE (devc->base + 0xC)
+#define DSP_COMMAND (devc->base + 0xC)
+#define DSP_STATUS (devc->base + 0xC)
+#define DSP_DATA_AVAIL (devc->base + 0xE)
+#define DSP_DATA_AVL16 (devc->base + 0xF)
+#define MIXER_ADDR (devc->base + 0x4)
+#define MIXER_DATA (devc->base + 0x5)
+#define OPL3_LEFT (devc->base + 0x0)
+#define OPL3_RIGHT (devc->base + 0x2)
+#define OPL3_BOTH (devc->base + 0x8)
+/* DSP Commands */
+
+#define DSP_CMD_SPKON 0xD1
+#define DSP_CMD_SPKOFF 0xD3
+#define DSP_CMD_DMAON 0xD0
+#define DSP_CMD_DMAOFF 0xD4
+
+#define IMODE_NONE 0
+#define IMODE_OUTPUT PCM_ENABLE_OUTPUT
+#define IMODE_INPUT PCM_ENABLE_INPUT
+#define IMODE_INIT 3
+#define IMODE_MIDI 4
+
+#define NORMAL_MIDI 0
+#define UART_MIDI 1
+
+
+/*
+ * Device models
+ */
+#define MDL_NONE 0
+#define MDL_SB1 1 /* SB1.0 or 1.5 */
+#define MDL_SB2 2 /* SB2.0 */
+#define MDL_SB201 3 /* SB2.01 */
+#define MDL_SBPRO 4 /* SB Pro */
+#define MDL_SB16 5 /* SB16/32/AWE */
+#define MDL_SBPNP 6 /* SB16/32/AWE PnP */
+#define MDL_JAZZ 10 /* Media Vision Jazz16 */
+#define MDL_SMW 11 /* Logitech SoundMan Wave (Jazz16) */
+#define MDL_ESS 12 /* ESS ES688 and ES1688 */
+#define MDL_AZTECH 13 /* Aztech Sound Galaxy family */
+#define MDL_ES1868MIDI 14 /* MIDI port of ESS1868 */
+#define MDL_AEDSP 15 /* Audio Excel DSP 16 */
+#define MDL_ESSPCI 16 /* ESS PCI card */
+#define MDL_YMPCI 17 /* Yamaha PCI sb in emulation */
+
+#define SUBMDL_ALS007 42 /* ALS-007 differs from SB16 only in mixer */
+ /* register assignment */
+#define SUBMDL_ALS100 43 /* ALS-100 allows sampling rates of up */
+ /* to 48kHz */
+
+/*
+ * Config flags
+ */
+#define SB_NO_MIDI 0x00000001
+#define SB_NO_MIXER 0x00000002
+#define SB_NO_AUDIO 0x00000004
+#define SB_NO_RECORDING 0x00000008 /* No audio recording */
+#define SB_MIDI_ONLY (SB_NO_AUDIO|SB_NO_MIXER)
+#define SB_PCI_IRQ 0x00000010 /* PCI shared IRQ */
+
+struct mixer_def {
+ unsigned int regno: 8;
+ unsigned int bitoffs:4;
+ unsigned int nbits:4;
+};
+
+typedef struct mixer_def mixer_tab[32][2];
+typedef struct mixer_def mixer_ent;
+
+struct sb_module_options
+{
+ int esstype; /* ESS chip type */
+ int acer; /* Do acer notebook init? */
+ int sm_games; /* Logitech soundman games? */
+};
+
+typedef struct sb_devc {
+ int dev;
+
+ /* Hardware parameters */
+ int *osp;
+ int minor, major;
+ int type;
+ int model, submodel;
+ int caps;
+# define SBCAP_STEREO 0x00000001
+# define SBCAP_16BITS 0x00000002
+
+ /* Hardware resources */
+ int base;
+ int irq;
+ int dma8, dma16;
+
+ int pcibase; /* For ESS Maestro etc */
+
+ /* State variables */
+ int opened;
+ /* new audio fields for full duplex support */
+ int fullduplex;
+ int duplex;
+ int speed, bits, channels;
+ volatile int irq_ok;
+ volatile int intr_active, irq_mode;
+ /* duplicate audio fields for full duplex support */
+ volatile int intr_active_16, irq_mode_16;
+
+ /* Mixer fields */
+ int *levels;
+ mixer_tab *iomap;
+ size_t iomap_sz; /* number or records in the iomap table */
+ int mixer_caps, recmask, outmask, supported_devices;
+ int supported_rec_devices, supported_out_devices;
+ int my_mixerdev;
+ int sbmixnum;
+
+ /* Audio fields */
+ unsigned long trg_buf;
+ int trigger_bits;
+ int trg_bytes;
+ int trg_intrflag;
+ int trg_restart;
+ /* duplicate audio fields for full duplex support */
+ unsigned long trg_buf_16;
+ int trigger_bits_16;
+ int trg_bytes_16;
+ int trg_intrflag_16;
+ int trg_restart_16;
+
+ unsigned char tconst;
+
+ /* MIDI fields */
+ int my_mididev;
+ int input_opened;
+ int midi_broken;
+ void (*midi_input_intr) (int dev, unsigned char data);
+ void *midi_irq_cookie; /* IRQ cookie for the midi */
+
+ spinlock_t lock;
+
+ struct sb_module_options sbmo; /* Module options */
+
+ } sb_devc;
+
+/*
+ * PCI card types
+ */
+
+#define SB_PCI_ESSMAESTRO 1 /* ESS Maestro Legacy */
+#define SB_PCI_YAMAHA 2 /* Yamaha Legacy */
+
+/*
+ * Functions
+ */
+
+int sb_dsp_command (sb_devc *devc, unsigned char val);
+int sb_dsp_get_byte(sb_devc * devc);
+int sb_dsp_reset (sb_devc *devc);
+void sb_setmixer (sb_devc *devc, unsigned int port, unsigned int value);
+unsigned int sb_getmixer (sb_devc *devc, unsigned int port);
+int sb_dsp_detect (struct address_info *hw_config, int pci, int pciio, struct sb_module_options *sbmo);
+int sb_dsp_init (struct address_info *hw_config, struct module *owner);
+void sb_dsp_unload(struct address_info *hw_config, int sbmpu);
+int sb_mixer_init(sb_devc *devc, struct module *owner);
+void sb_mixer_unload(sb_devc *devc);
+void sb_mixer_set_stereo (sb_devc *devc, int mode);
+void smw_mixer_init(sb_devc *devc);
+void sb_dsp_midi_init (sb_devc *devc, struct module *owner);
+void sb_audio_init (sb_devc *devc, char *name, struct module *owner);
+void sb_midi_interrupt (sb_devc *devc);
+void sb_chgmixer (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val);
+int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right);
+
+int sb_audio_open(int dev, int mode);
+void sb_audio_close(int dev);
+
+/* From sb_common.c */
+void sb_dsp_disable_midi(int port);
+int probe_sbmpu (struct address_info *hw_config, struct module *owner);
+void unload_sbmpu (struct address_info *hw_config);
+
+void unload_sb16(struct address_info *hw_info);
+void unload_sb16midi(struct address_info *hw_info);
diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c
new file mode 100644
index 0000000..733b014
--- /dev/null
+++ b/sound/oss/sb_audio.c
@@ -0,0 +1,1098 @@
+/*
+ * sound/oss/sb_audio.c
+ *
+ * Audio routines for Sound Blaster compatible cards.
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ * Changes
+ * Alan Cox : Formatting and clean ups
+ *
+ * Status
+ * Mostly working. Weird uart bug causing irq storms
+ *
+ * Daniel J. Rodriksson: Changes to make sb16 work full duplex.
+ * Maybe other 16 bit cards in this code could behave
+ * the same.
+ * Chris Rankin: Use spinlocks instead of CLI/STI
+ */
+
+#include <linux/spinlock.h>
+
+#include "sound_config.h"
+
+#include "sb_mixer.h"
+#include "sb.h"
+
+#include "sb_ess.h"
+
+int sb_audio_open(int dev, int mode)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned long flags;
+
+ if (devc == NULL)
+ {
+ printk(KERN_ERR "Sound Blaster: incomplete initialization.\n");
+ return -ENXIO;
+ }
+ if (devc->caps & SB_NO_RECORDING && mode & OPEN_READ)
+ {
+ if (mode == OPEN_READ)
+ return -EPERM;
+ }
+ spin_lock_irqsave(&devc->lock, flags);
+ if (devc->opened)
+ {
+ spin_unlock_irqrestore(&devc->lock, flags);
+ return -EBUSY;
+ }
+ if (devc->dma16 != -1 && devc->dma16 != devc->dma8 && !devc->duplex)
+ {
+ if (sound_open_dma(devc->dma16, "Sound Blaster 16 bit"))
+ {
+ spin_unlock_irqrestore(&devc->lock, flags);
+ return -EBUSY;
+ }
+ }
+ devc->opened = mode;
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ devc->irq_mode = IMODE_NONE;
+ devc->irq_mode_16 = IMODE_NONE;
+ devc->fullduplex = devc->duplex &&
+ ((mode & OPEN_READ) && (mode & OPEN_WRITE));
+ sb_dsp_reset(devc);
+
+ /* At first glance this check isn't enough, some ESS chips might not
+ * have a RECLEV. However if they don't common_mixer_set will refuse
+ * cause devc->iomap has no register mapping for RECLEV
+ */
+ if (devc->model == MDL_ESS) ess_mixer_reload (devc, SOUND_MIXER_RECLEV);
+
+ /* The ALS007 seems to require that the DSP be removed from the output */
+ /* in order for recording to be activated properly. This is done by */
+ /* setting the appropriate bits of the output control register 4ch to */
+ /* zero. This code assumes that the output control registers are not */
+ /* used anywhere else and therefore the DSP bits are *always* ON for */
+ /* output and OFF for sampling. */
+
+ if (devc->submodel == SUBMDL_ALS007)
+ {
+ if (mode & OPEN_READ)
+ sb_setmixer(devc,ALS007_OUTPUT_CTRL2,
+ sb_getmixer(devc,ALS007_OUTPUT_CTRL2) & 0xf9);
+ else
+ sb_setmixer(devc,ALS007_OUTPUT_CTRL2,
+ sb_getmixer(devc,ALS007_OUTPUT_CTRL2) | 0x06);
+ }
+ return 0;
+}
+
+void sb_audio_close(int dev)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ /* fix things if mmap turned off fullduplex */
+ if(devc->duplex
+ && !devc->fullduplex
+ && (devc->opened & OPEN_READ) && (devc->opened & OPEN_WRITE))
+ {
+ struct dma_buffparms *dmap_temp;
+ dmap_temp = audio_devs[dev]->dmap_out;
+ audio_devs[dev]->dmap_out = audio_devs[dev]->dmap_in;
+ audio_devs[dev]->dmap_in = dmap_temp;
+ }
+ audio_devs[dev]->dmap_out->dma = devc->dma8;
+ audio_devs[dev]->dmap_in->dma = ( devc->duplex ) ?
+ devc->dma16 : devc->dma8;
+
+ if (devc->dma16 != -1 && devc->dma16 != devc->dma8 && !devc->duplex)
+ sound_close_dma(devc->dma16);
+
+ /* For ALS007, turn DSP output back on if closing the device for read */
+
+ if ((devc->submodel == SUBMDL_ALS007) && (devc->opened & OPEN_READ))
+ {
+ sb_setmixer(devc,ALS007_OUTPUT_CTRL2,
+ sb_getmixer(devc,ALS007_OUTPUT_CTRL2) | 0x06);
+ }
+ devc->opened = 0;
+}
+
+static void sb_set_output_parms(int dev, unsigned long buf, int nr_bytes,
+ int intrflag)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (!devc->fullduplex || devc->bits == AFMT_S16_LE)
+ {
+ devc->trg_buf = buf;
+ devc->trg_bytes = nr_bytes;
+ devc->trg_intrflag = intrflag;
+ devc->irq_mode = IMODE_OUTPUT;
+ }
+ else
+ {
+ devc->trg_buf_16 = buf;
+ devc->trg_bytes_16 = nr_bytes;
+ devc->trg_intrflag_16 = intrflag;
+ devc->irq_mode_16 = IMODE_OUTPUT;
+ }
+}
+
+static void sb_set_input_parms(int dev, unsigned long buf, int count, int intrflag)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (!devc->fullduplex || devc->bits != AFMT_S16_LE)
+ {
+ devc->trg_buf = buf;
+ devc->trg_bytes = count;
+ devc->trg_intrflag = intrflag;
+ devc->irq_mode = IMODE_INPUT;
+ }
+ else
+ {
+ devc->trg_buf_16 = buf;
+ devc->trg_bytes_16 = count;
+ devc->trg_intrflag_16 = intrflag;
+ devc->irq_mode_16 = IMODE_INPUT;
+ }
+}
+
+/*
+ * SB1.x compatible routines
+ */
+
+static void sb1_audio_output_block(int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ unsigned long flags;
+ int count = nr_bytes;
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */
+
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ count >>= 1;
+ count--;
+
+ devc->irq_mode = IMODE_OUTPUT;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x14)) /* 8 bit DAC using DMA */
+ {
+ sb_dsp_command(devc, (unsigned char) (count & 0xff));
+ sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff));
+ }
+ else
+ printk(KERN_WARNING "Sound Blaster: unable to start DAC.\n");
+ spin_unlock_irqrestore(&devc->lock, flags);
+ devc->intr_active = 1;
+}
+
+static void sb1_audio_start_input(int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ unsigned long flags;
+ int count = nr_bytes;
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ /*
+ * Start a DMA input to the buffer pointed by dmaqtail
+ */
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */
+
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ count >>= 1;
+ count--;
+
+ devc->irq_mode = IMODE_INPUT;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x24)) /* 8 bit ADC using DMA */
+ {
+ sb_dsp_command(devc, (unsigned char) (count & 0xff));
+ sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff));
+ }
+ else
+ printk(KERN_ERR "Sound Blaster: unable to start ADC.\n");
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ devc->intr_active = 1;
+}
+
+static void sb1_audio_trigger(int dev, int bits)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ bits &= devc->irq_mode;
+
+ if (!bits)
+ sb_dsp_command(devc, 0xd0); /* Halt DMA */
+ else
+ {
+ switch (devc->irq_mode)
+ {
+ case IMODE_INPUT:
+ sb1_audio_start_input(dev, devc->trg_buf, devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+
+ case IMODE_OUTPUT:
+ sb1_audio_output_block(dev, devc->trg_buf, devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+ }
+ }
+ devc->trigger_bits = bits;
+}
+
+static int sb1_audio_prepare_for_input(int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x40))
+ sb_dsp_command(devc, devc->tconst);
+ sb_dsp_command(devc, DSP_CMD_SPKOFF);
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int sb1_audio_prepare_for_output(int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x40))
+ sb_dsp_command(devc, devc->tconst);
+ sb_dsp_command(devc, DSP_CMD_SPKON);
+ spin_unlock_irqrestore(&devc->lock, flags);
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int sb1_audio_set_speed(int dev, int speed)
+{
+ int max_speed = 23000;
+ sb_devc *devc = audio_devs[dev]->devc;
+ int tmp;
+
+ if (devc->opened & OPEN_READ)
+ max_speed = 13000;
+
+ if (speed > 0)
+ {
+ if (speed < 4000)
+ speed = 4000;
+
+ if (speed > max_speed)
+ speed = max_speed;
+
+ devc->tconst = (256 - ((1000000 + speed / 2) / speed)) & 0xff;
+ tmp = 256 - devc->tconst;
+ speed = (1000000 + tmp / 2) / tmp;
+
+ devc->speed = speed;
+ }
+ return devc->speed;
+}
+
+static short sb1_audio_set_channels(int dev, short channels)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ return devc->channels = 1;
+}
+
+static unsigned int sb1_audio_set_bits(int dev, unsigned int bits)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ return devc->bits = 8;
+}
+
+static void sb1_audio_halt_xfer(int dev)
+{
+ unsigned long flags;
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ sb_dsp_reset(devc);
+ spin_unlock_irqrestore(&devc->lock, flags);
+}
+
+/*
+ * SB 2.0 and SB 2.01 compatible routines
+ */
+
+static void sb20_audio_output_block(int dev, unsigned long buf, int nr_bytes,
+ int intrflag)
+{
+ unsigned long flags;
+ int count = nr_bytes;
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned char cmd;
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */
+
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ count >>= 1;
+ count--;
+
+ devc->irq_mode = IMODE_OUTPUT;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x48)) /* DSP Block size */
+ {
+ sb_dsp_command(devc, (unsigned char) (count & 0xff));
+ sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff));
+
+ if (devc->speed * devc->channels <= 23000)
+ cmd = 0x1c; /* 8 bit PCM output */
+ else
+ cmd = 0x90; /* 8 bit high speed PCM output (SB2.01/Pro) */
+
+ if (!sb_dsp_command(devc, cmd))
+ printk(KERN_ERR "Sound Blaster: unable to start DAC.\n");
+ }
+ else
+ printk(KERN_ERR "Sound Blaster: unable to start DAC.\n");
+ spin_unlock_irqrestore(&devc->lock, flags);
+ devc->intr_active = 1;
+}
+
+static void sb20_audio_start_input(int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ unsigned long flags;
+ int count = nr_bytes;
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned char cmd;
+
+ /*
+ * Start a DMA input to the buffer pointed by dmaqtail
+ */
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */
+
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ count >>= 1;
+ count--;
+
+ devc->irq_mode = IMODE_INPUT;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x48)) /* DSP Block size */
+ {
+ sb_dsp_command(devc, (unsigned char) (count & 0xff));
+ sb_dsp_command(devc, (unsigned char) ((count >> 8) & 0xff));
+
+ if (devc->speed * devc->channels <= (devc->major == 3 ? 23000 : 13000))
+ cmd = 0x2c; /* 8 bit PCM input */
+ else
+ cmd = 0x98; /* 8 bit high speed PCM input (SB2.01/Pro) */
+
+ if (!sb_dsp_command(devc, cmd))
+ printk(KERN_ERR "Sound Blaster: unable to start ADC.\n");
+ }
+ else
+ printk(KERN_ERR "Sound Blaster: unable to start ADC.\n");
+ spin_unlock_irqrestore(&devc->lock, flags);
+ devc->intr_active = 1;
+}
+
+static void sb20_audio_trigger(int dev, int bits)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ bits &= devc->irq_mode;
+
+ if (!bits)
+ sb_dsp_command(devc, 0xd0); /* Halt DMA */
+ else
+ {
+ switch (devc->irq_mode)
+ {
+ case IMODE_INPUT:
+ sb20_audio_start_input(dev, devc->trg_buf, devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+
+ case IMODE_OUTPUT:
+ sb20_audio_output_block(dev, devc->trg_buf, devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+ }
+ }
+ devc->trigger_bits = bits;
+}
+
+/*
+ * SB2.01 specific speed setup
+ */
+
+static int sb201_audio_set_speed(int dev, int speed)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ int tmp;
+ int s = speed * devc->channels;
+
+ if (speed > 0)
+ {
+ if (speed < 4000)
+ speed = 4000;
+ if (speed > 44100)
+ speed = 44100;
+ if (devc->opened & OPEN_READ && speed > 15000)
+ speed = 15000;
+ devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff;
+ tmp = 256 - devc->tconst;
+ speed = ((1000000 + tmp / 2) / tmp) / devc->channels;
+
+ devc->speed = speed;
+ }
+ return devc->speed;
+}
+
+/*
+ * SB Pro specific routines
+ */
+
+static int sbpro_audio_prepare_for_input(int dev, int bsize, int bcount)
+{ /* For SB Pro and Jazz16 */
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned long flags;
+ unsigned char bits = 0;
+
+ if (devc->dma16 >= 0 && devc->dma16 != devc->dma8)
+ audio_devs[dev]->dmap_out->dma = audio_devs[dev]->dmap_in->dma =
+ devc->bits == 16 ? devc->dma16 : devc->dma8;
+
+ if (devc->model == MDL_JAZZ || devc->model == MDL_SMW)
+ if (devc->bits == AFMT_S16_LE)
+ bits = 0x04; /* 16 bit mode */
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x40))
+ sb_dsp_command(devc, devc->tconst);
+ sb_dsp_command(devc, DSP_CMD_SPKOFF);
+ if (devc->channels == 1)
+ sb_dsp_command(devc, 0xa0 | bits); /* Mono input */
+ else
+ sb_dsp_command(devc, 0xa8 | bits); /* Stereo input */
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int sbpro_audio_prepare_for_output(int dev, int bsize, int bcount)
+{ /* For SB Pro and Jazz16 */
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned long flags;
+ unsigned char tmp;
+ unsigned char bits = 0;
+
+ if (devc->dma16 >= 0 && devc->dma16 != devc->dma8)
+ audio_devs[dev]->dmap_out->dma = audio_devs[dev]->dmap_in->dma = devc->bits == 16 ? devc->dma16 : devc->dma8;
+ if (devc->model == MDL_SBPRO)
+ sb_mixer_set_stereo(devc, devc->channels == 2);
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (sb_dsp_command(devc, 0x40))
+ sb_dsp_command(devc, devc->tconst);
+ sb_dsp_command(devc, DSP_CMD_SPKON);
+
+ if (devc->model == MDL_JAZZ || devc->model == MDL_SMW)
+ {
+ if (devc->bits == AFMT_S16_LE)
+ bits = 0x04; /* 16 bit mode */
+
+ if (devc->channels == 1)
+ sb_dsp_command(devc, 0xa0 | bits); /* Mono output */
+ else
+ sb_dsp_command(devc, 0xa8 | bits); /* Stereo output */
+ spin_unlock_irqrestore(&devc->lock, flags);
+ }
+ else
+ {
+ spin_unlock_irqrestore(&devc->lock, flags);
+ tmp = sb_getmixer(devc, 0x0e);
+ if (devc->channels == 1)
+ tmp &= ~0x02;
+ else
+ tmp |= 0x02;
+ sb_setmixer(devc, 0x0e, tmp);
+ }
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int sbpro_audio_set_speed(int dev, int speed)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (speed > 0)
+ {
+ if (speed < 4000)
+ speed = 4000;
+ if (speed > 44100)
+ speed = 44100;
+ if (devc->channels > 1 && speed > 22050)
+ speed = 22050;
+ sb201_audio_set_speed(dev, speed);
+ }
+ return devc->speed;
+}
+
+static short sbpro_audio_set_channels(int dev, short channels)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (channels == 1 || channels == 2)
+ {
+ if (channels != devc->channels)
+ {
+ devc->channels = channels;
+ if (devc->model == MDL_SBPRO && devc->channels == 2)
+ sbpro_audio_set_speed(dev, devc->speed);
+ }
+ }
+ return devc->channels;
+}
+
+static int jazz16_audio_set_speed(int dev, int speed)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (speed > 0)
+ {
+ int tmp;
+ int s = speed * devc->channels;
+
+ if (speed < 5000)
+ speed = 5000;
+ if (speed > 44100)
+ speed = 44100;
+
+ devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff;
+
+ tmp = 256 - devc->tconst;
+ speed = ((1000000 + tmp / 2) / tmp) / devc->channels;
+
+ devc->speed = speed;
+ }
+ return devc->speed;
+}
+
+/*
+ * SB16 specific routines
+ */
+
+static int sb16_audio_set_speed(int dev, int speed)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ int max_speed = devc->submodel == SUBMDL_ALS100 ? 48000 : 44100;
+
+ if (speed > 0)
+ {
+ if (speed < 5000)
+ speed = 5000;
+
+ if (speed > max_speed)
+ speed = max_speed;
+
+ devc->speed = speed;
+ }
+ return devc->speed;
+}
+
+static unsigned int sb16_audio_set_bits(int dev, unsigned int bits)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (bits != 0)
+ {
+ if (bits == AFMT_U8 || bits == AFMT_S16_LE)
+ devc->bits = bits;
+ else
+ devc->bits = AFMT_U8;
+ }
+
+ return devc->bits;
+}
+
+static int sb16_audio_prepare_for_input(int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (!devc->fullduplex)
+ {
+ audio_devs[dev]->dmap_out->dma =
+ audio_devs[dev]->dmap_in->dma =
+ devc->bits == AFMT_S16_LE ?
+ devc->dma16 : devc->dma8;
+ }
+ else if (devc->bits == AFMT_S16_LE)
+ {
+ audio_devs[dev]->dmap_out->dma = devc->dma8;
+ audio_devs[dev]->dmap_in->dma = devc->dma16;
+ }
+ else
+ {
+ audio_devs[dev]->dmap_out->dma = devc->dma16;
+ audio_devs[dev]->dmap_in->dma = devc->dma8;
+ }
+
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int sb16_audio_prepare_for_output(int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (!devc->fullduplex)
+ {
+ audio_devs[dev]->dmap_out->dma =
+ audio_devs[dev]->dmap_in->dma =
+ devc->bits == AFMT_S16_LE ?
+ devc->dma16 : devc->dma8;
+ }
+ else if (devc->bits == AFMT_S16_LE)
+ {
+ audio_devs[dev]->dmap_out->dma = devc->dma8;
+ audio_devs[dev]->dmap_in->dma = devc->dma16;
+ }
+ else
+ {
+ audio_devs[dev]->dmap_out->dma = devc->dma16;
+ audio_devs[dev]->dmap_in->dma = devc->dma8;
+ }
+
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static void sb16_audio_output_block(int dev, unsigned long buf, int count,
+ int intrflag)
+{
+ unsigned long flags, cnt;
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned long bits;
+
+ if (!devc->fullduplex || devc->bits == AFMT_S16_LE)
+ {
+ devc->irq_mode = IMODE_OUTPUT;
+ devc->intr_active = 1;
+ }
+ else
+ {
+ devc->irq_mode_16 = IMODE_OUTPUT;
+ devc->intr_active_16 = 1;
+ }
+
+ /* save value */
+ spin_lock_irqsave(&devc->lock, flags);
+ bits = devc->bits;
+ if (devc->fullduplex)
+ devc->bits = (devc->bits == AFMT_S16_LE) ?
+ AFMT_U8 : AFMT_S16_LE;
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ cnt = count;
+ if (devc->bits == AFMT_S16_LE)
+ cnt >>= 1;
+ cnt--;
+
+ spin_lock_irqsave(&devc->lock, flags);
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */
+
+ sb_dsp_command(devc, 0x41);
+ sb_dsp_command(devc, (unsigned char) ((devc->speed >> 8) & 0xff));
+ sb_dsp_command(devc, (unsigned char) (devc->speed & 0xff));
+
+ sb_dsp_command(devc, (devc->bits == AFMT_S16_LE ? 0xb6 : 0xc6));
+ sb_dsp_command(devc, ((devc->channels == 2 ? 0x20 : 0) +
+ (devc->bits == AFMT_S16_LE ? 0x10 : 0)));
+ sb_dsp_command(devc, (unsigned char) (cnt & 0xff));
+ sb_dsp_command(devc, (unsigned char) (cnt >> 8));
+
+ /* restore real value after all programming */
+ devc->bits = bits;
+ spin_unlock_irqrestore(&devc->lock, flags);
+}
+
+
+/*
+ * This fails on the Cyrix MediaGX. If you don't have the DMA enabled
+ * before the first sample arrives it locks up. However even if you
+ * do enable the DMA in time you just get DMA timeouts and missing
+ * interrupts and stuff, so for now I've not bothered fixing this either.
+ */
+
+static void sb16_audio_start_input(int dev, unsigned long buf, int count, int intrflag)
+{
+ unsigned long flags, cnt;
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (!devc->fullduplex || devc->bits != AFMT_S16_LE)
+ {
+ devc->irq_mode = IMODE_INPUT;
+ devc->intr_active = 1;
+ }
+ else
+ {
+ devc->irq_mode_16 = IMODE_INPUT;
+ devc->intr_active_16 = 1;
+ }
+
+ cnt = count;
+ if (devc->bits == AFMT_S16_LE)
+ cnt >>= 1;
+ cnt--;
+
+ spin_lock_irqsave(&devc->lock, flags);
+
+ /* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */
+
+ sb_dsp_command(devc, 0x42);
+ sb_dsp_command(devc, (unsigned char) ((devc->speed >> 8) & 0xff));
+ sb_dsp_command(devc, (unsigned char) (devc->speed & 0xff));
+
+ sb_dsp_command(devc, (devc->bits == AFMT_S16_LE ? 0xbe : 0xce));
+ sb_dsp_command(devc, ((devc->channels == 2 ? 0x20 : 0) +
+ (devc->bits == AFMT_S16_LE ? 0x10 : 0)));
+ sb_dsp_command(devc, (unsigned char) (cnt & 0xff));
+ sb_dsp_command(devc, (unsigned char) (cnt >> 8));
+
+ spin_unlock_irqrestore(&devc->lock, flags);
+}
+
+static void sb16_audio_trigger(int dev, int bits)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ int bits_16 = bits & devc->irq_mode_16;
+ bits &= devc->irq_mode;
+
+ if (!bits && !bits_16)
+ sb_dsp_command(devc, 0xd0); /* Halt DMA */
+ else
+ {
+ if (bits)
+ {
+ switch (devc->irq_mode)
+ {
+ case IMODE_INPUT:
+ sb16_audio_start_input(dev,
+ devc->trg_buf,
+ devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+
+ case IMODE_OUTPUT:
+ sb16_audio_output_block(dev,
+ devc->trg_buf,
+ devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+ }
+ }
+ if (bits_16)
+ {
+ switch (devc->irq_mode_16)
+ {
+ case IMODE_INPUT:
+ sb16_audio_start_input(dev,
+ devc->trg_buf_16,
+ devc->trg_bytes_16,
+ devc->trg_intrflag_16);
+ break;
+
+ case IMODE_OUTPUT:
+ sb16_audio_output_block(dev,
+ devc->trg_buf_16,
+ devc->trg_bytes_16,
+ devc->trg_intrflag_16);
+ break;
+ }
+ }
+ }
+
+ devc->trigger_bits = bits | bits_16;
+}
+
+static unsigned char lbuf8[2048];
+static signed short *lbuf16 = (signed short *)lbuf8;
+#define LBUFCOPYSIZE 1024
+static void
+sb16_copy_from_user(int dev,
+ char *localbuf, int localoffs,
+ const char __user *userbuf, int useroffs,
+ int max_in, int max_out,
+ int *used, int *returned,
+ int len)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ int i, c, p, locallen;
+ unsigned char *buf8;
+ signed short *buf16;
+
+ /* if not duplex no conversion */
+ if (!devc->fullduplex)
+ {
+ if (copy_from_user(localbuf + localoffs,
+ userbuf + useroffs, len))
+ return;
+ *used = len;
+ *returned = len;
+ }
+ else if (devc->bits == AFMT_S16_LE)
+ {
+ /* 16 -> 8 */
+ /* max_in >> 1, max number of samples in ( 16 bits ) */
+ /* max_out, max number of samples out ( 8 bits ) */
+ /* len, number of samples that will be taken ( 16 bits )*/
+ /* c, count of samples remaining in buffer ( 16 bits )*/
+ /* p, count of samples already processed ( 16 bits )*/
+ len = ( (max_in >> 1) > max_out) ? max_out : (max_in >> 1);
+ c = len;
+ p = 0;
+ buf8 = (unsigned char *)(localbuf + localoffs);
+ while (c)
+ {
+ locallen = (c >= LBUFCOPYSIZE ? LBUFCOPYSIZE : c);
+ /* << 1 in order to get 16 bit samples */
+ if (copy_from_user(lbuf16,
+ userbuf + useroffs + (p << 1),
+ locallen << 1))
+ return;
+ for (i = 0; i < locallen; i++)
+ {
+ buf8[p+i] = ~((lbuf16[i] >> 8) & 0xff) ^ 0x80;
+ }
+ c -= locallen; p += locallen;
+ }
+ /* used = ( samples * 16 bits size ) */
+ *used = max_in > ( max_out << 1) ? (max_out << 1) : max_in;
+ /* returned = ( samples * 8 bits size ) */
+ *returned = len;
+ }
+ else
+ {
+ /* 8 -> 16 */
+ /* max_in, max number of samples in ( 8 bits ) */
+ /* max_out >> 1, max number of samples out ( 16 bits ) */
+ /* len, number of samples that will be taken ( 8 bits )*/
+ /* c, count of samples remaining in buffer ( 8 bits )*/
+ /* p, count of samples already processed ( 8 bits )*/
+ len = max_in > (max_out >> 1) ? (max_out >> 1) : max_in;
+ c = len;
+ p = 0;
+ buf16 = (signed short *)(localbuf + localoffs);
+ while (c)
+ {
+ locallen = (c >= LBUFCOPYSIZE ? LBUFCOPYSIZE : c);
+ if (copy_from_user(lbuf8,
+ userbuf+useroffs + p,
+ locallen))
+ return;
+ for (i = 0; i < locallen; i++)
+ {
+ buf16[p+i] = (~lbuf8[i] ^ 0x80) << 8;
+ }
+ c -= locallen; p += locallen;
+ }
+ /* used = ( samples * 8 bits size ) */
+ *used = len;
+ /* returned = ( samples * 16 bits size ) */
+ *returned = len << 1;
+ }
+}
+
+static void
+sb16_audio_mmap(int dev)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ devc->fullduplex = 0;
+}
+
+static struct audio_driver sb1_audio_driver = /* SB1.x */
+{
+ .owner = THIS_MODULE,
+ .open = sb_audio_open,
+ .close = sb_audio_close,
+ .output_block = sb_set_output_parms,
+ .start_input = sb_set_input_parms,
+ .prepare_for_input = sb1_audio_prepare_for_input,
+ .prepare_for_output = sb1_audio_prepare_for_output,
+ .halt_io = sb1_audio_halt_xfer,
+ .trigger = sb1_audio_trigger,
+ .set_speed = sb1_audio_set_speed,
+ .set_bits = sb1_audio_set_bits,
+ .set_channels = sb1_audio_set_channels
+};
+
+static struct audio_driver sb20_audio_driver = /* SB2.0 */
+{
+ .owner = THIS_MODULE,
+ .open = sb_audio_open,
+ .close = sb_audio_close,
+ .output_block = sb_set_output_parms,
+ .start_input = sb_set_input_parms,
+ .prepare_for_input = sb1_audio_prepare_for_input,
+ .prepare_for_output = sb1_audio_prepare_for_output,
+ .halt_io = sb1_audio_halt_xfer,
+ .trigger = sb20_audio_trigger,
+ .set_speed = sb1_audio_set_speed,
+ .set_bits = sb1_audio_set_bits,
+ .set_channels = sb1_audio_set_channels
+};
+
+static struct audio_driver sb201_audio_driver = /* SB2.01 */
+{
+ .owner = THIS_MODULE,
+ .open = sb_audio_open,
+ .close = sb_audio_close,
+ .output_block = sb_set_output_parms,
+ .start_input = sb_set_input_parms,
+ .prepare_for_input = sb1_audio_prepare_for_input,
+ .prepare_for_output = sb1_audio_prepare_for_output,
+ .halt_io = sb1_audio_halt_xfer,
+ .trigger = sb20_audio_trigger,
+ .set_speed = sb201_audio_set_speed,
+ .set_bits = sb1_audio_set_bits,
+ .set_channels = sb1_audio_set_channels
+};
+
+static struct audio_driver sbpro_audio_driver = /* SB Pro */
+{
+ .owner = THIS_MODULE,
+ .open = sb_audio_open,
+ .close = sb_audio_close,
+ .output_block = sb_set_output_parms,
+ .start_input = sb_set_input_parms,
+ .prepare_for_input = sbpro_audio_prepare_for_input,
+ .prepare_for_output = sbpro_audio_prepare_for_output,
+ .halt_io = sb1_audio_halt_xfer,
+ .trigger = sb20_audio_trigger,
+ .set_speed = sbpro_audio_set_speed,
+ .set_bits = sb1_audio_set_bits,
+ .set_channels = sbpro_audio_set_channels
+};
+
+static struct audio_driver jazz16_audio_driver = /* Jazz16 and SM Wave */
+{
+ .owner = THIS_MODULE,
+ .open = sb_audio_open,
+ .close = sb_audio_close,
+ .output_block = sb_set_output_parms,
+ .start_input = sb_set_input_parms,
+ .prepare_for_input = sbpro_audio_prepare_for_input,
+ .prepare_for_output = sbpro_audio_prepare_for_output,
+ .halt_io = sb1_audio_halt_xfer,
+ .trigger = sb20_audio_trigger,
+ .set_speed = jazz16_audio_set_speed,
+ .set_bits = sb16_audio_set_bits,
+ .set_channels = sbpro_audio_set_channels
+};
+
+static struct audio_driver sb16_audio_driver = /* SB16 */
+{
+ .owner = THIS_MODULE,
+ .open = sb_audio_open,
+ .close = sb_audio_close,
+ .output_block = sb_set_output_parms,
+ .start_input = sb_set_input_parms,
+ .prepare_for_input = sb16_audio_prepare_for_input,
+ .prepare_for_output = sb16_audio_prepare_for_output,
+ .halt_io = sb1_audio_halt_xfer,
+ .copy_user = sb16_copy_from_user,
+ .trigger = sb16_audio_trigger,
+ .set_speed = sb16_audio_set_speed,
+ .set_bits = sb16_audio_set_bits,
+ .set_channels = sbpro_audio_set_channels,
+ .mmap = sb16_audio_mmap
+};
+
+void sb_audio_init(sb_devc * devc, char *name, struct module *owner)
+{
+ int audio_flags = 0;
+ int format_mask = AFMT_U8;
+
+ struct audio_driver *driver = &sb1_audio_driver;
+
+ switch (devc->model)
+ {
+ case MDL_SB1: /* SB1.0 or SB 1.5 */
+ DDB(printk("Will use standard SB1.x driver\n"));
+ audio_flags = DMA_HARDSTOP;
+ break;
+
+ case MDL_SB2:
+ DDB(printk("Will use SB2.0 driver\n"));
+ audio_flags = DMA_AUTOMODE;
+ driver = &sb20_audio_driver;
+ break;
+
+ case MDL_SB201:
+ DDB(printk("Will use SB2.01 (high speed) driver\n"));
+ audio_flags = DMA_AUTOMODE;
+ driver = &sb201_audio_driver;
+ break;
+
+ case MDL_JAZZ:
+ case MDL_SMW:
+ DDB(printk("Will use Jazz16 driver\n"));
+ audio_flags = DMA_AUTOMODE;
+ format_mask |= AFMT_S16_LE;
+ driver = &jazz16_audio_driver;
+ break;
+
+ case MDL_ESS:
+ DDB(printk("Will use ESS ES688/1688 driver\n"));
+ driver = ess_audio_init (devc, &audio_flags, &format_mask);
+ break;
+
+ case MDL_SB16:
+ DDB(printk("Will use SB16 driver\n"));
+ audio_flags = DMA_AUTOMODE;
+ format_mask |= AFMT_S16_LE;
+ if (devc->dma8 != devc->dma16 && devc->dma16 != -1)
+ {
+ audio_flags |= DMA_DUPLEX;
+ devc->duplex = 1;
+ }
+ driver = &sb16_audio_driver;
+ break;
+
+ default:
+ DDB(printk("Will use SB Pro driver\n"));
+ audio_flags = DMA_AUTOMODE;
+ driver = &sbpro_audio_driver;
+ }
+
+ if (owner)
+ driver->owner = owner;
+
+ if ((devc->dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION,
+ name,driver, sizeof(struct audio_driver),
+ audio_flags, format_mask, devc,
+ devc->dma8,
+ devc->duplex ? devc->dma16 : devc->dma8)) < 0)
+ {
+ printk(KERN_ERR "Sound Blaster: unable to install audio.\n");
+ return;
+ }
+ audio_devs[devc->dev]->mixer_dev = devc->my_mixerdev;
+ audio_devs[devc->dev]->min_fragment = 5;
+}
diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c
new file mode 100644
index 0000000..7de18b5
--- /dev/null
+++ b/sound/oss/sb_card.c
@@ -0,0 +1,353 @@
+/*
+ * sound/oss/sb_card.c
+ *
+ * Detection routine for the ISA Sound Blaster and compatable sound
+ * cards.
+ *
+ * This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this
+ * software for more info.
+ *
+ * This is a complete rewrite of the detection routines. This was
+ * prompted by the PnP API change during v2.5 and the ugly state the
+ * code was in.
+ *
+ * Copyright (C) by Paul Laufer 2002. Based on code originally by
+ * Hannu Savolainen which was modified by many others over the
+ * years. Authors specifically mentioned in the previous version were:
+ * Daniel Stone, Alessandro Zummo, Jeff Garzik, Arnaldo Carvalho de
+ * Melo, Daniel Church, and myself.
+ *
+ * 02-05-2003 Original Release, Paul Laufer <paul@laufernet.com>
+ * 02-07-2003 Bug made it into first release. Take two.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include "sound_config.h"
+#include "sb_mixer.h"
+#include "sb.h"
+#ifdef CONFIG_PNP
+#include <linux/pnp.h>
+#endif /* CONFIG_PNP */
+#include "sb_card.h"
+
+MODULE_DESCRIPTION("OSS Soundblaster ISA PnP and legacy sound driver");
+MODULE_LICENSE("GPL");
+
+extern void *smw_free;
+
+static int __initdata mpu_io = 0;
+static int __initdata io = -1;
+static int __initdata irq = -1;
+static int __initdata dma = -1;
+static int __initdata dma16 = -1;
+static int __initdata type = 0; /* Can set this to a specific card type */
+static int __initdata esstype = 0; /* ESS chip type */
+static int __initdata acer = 0; /* Do acer notebook init? */
+static int __initdata sm_games = 0; /* Logitech soundman games? */
+
+static struct sb_card_config *legacy = NULL;
+
+#ifdef CONFIG_PNP
+static int pnp_registered;
+static int __initdata pnp = 1;
+/*
+static int __initdata uart401 = 0;
+*/
+#else
+static int __initdata pnp = 0;
+#endif
+
+module_param(io, int, 000);
+MODULE_PARM_DESC(io, "Soundblaster i/o base address (0x220,0x240,0x260,0x280)");
+module_param(irq, int, 000);
+MODULE_PARM_DESC(irq, "IRQ (5,7,9,10)");
+module_param(dma, int, 000);
+MODULE_PARM_DESC(dma, "8-bit DMA channel (0,1,3)");
+module_param(dma16, int, 000);
+MODULE_PARM_DESC(dma16, "16-bit DMA channel (5,6,7)");
+module_param(mpu_io, int, 000);
+MODULE_PARM_DESC(mpu_io, "MPU base address");
+module_param(type, int, 000);
+MODULE_PARM_DESC(type, "You can set this to specific card type (doesn't " \
+ "work with pnp)");
+module_param(sm_games, int, 000);
+MODULE_PARM_DESC(sm_games, "Enable support for Logitech soundman games " \
+ "(doesn't work with pnp)");
+module_param(esstype, int, 000);
+MODULE_PARM_DESC(esstype, "ESS chip type (doesn't work with pnp)");
+module_param(acer, int, 000);
+MODULE_PARM_DESC(acer, "Set this to detect cards in some ACER notebooks "\
+ "(doesn't work with pnp)");
+
+#ifdef CONFIG_PNP
+module_param(pnp, int, 000);
+MODULE_PARM_DESC(pnp, "Went set to 0 will disable detection using PnP. "\
+ "Default is 1.\n");
+/* Not done yet.... */
+/*
+module_param(uart401, int, 000);
+MODULE_PARM_DESC(uart401, "When set to 1, will attempt to detect and enable"\
+ "the mpu on some clones");
+*/
+#endif /* CONFIG_PNP */
+
+/* OSS subsystem card registration shared by PnP and legacy routines */
+static int sb_register_oss(struct sb_card_config *scc, struct sb_module_options *sbmo)
+{
+ if (!request_region(scc->conf.io_base, 16, "soundblaster")) {
+ printk(KERN_ERR "sb: ports busy.\n");
+ kfree(scc);
+ return -EBUSY;
+ }
+
+ if (!sb_dsp_detect(&scc->conf, 0, 0, sbmo)) {
+ release_region(scc->conf.io_base, 16);
+ printk(KERN_ERR "sb: Failed DSP Detect.\n");
+ kfree(scc);
+ return -ENODEV;
+ }
+ if(!sb_dsp_init(&scc->conf, THIS_MODULE)) {
+ printk(KERN_ERR "sb: Failed DSP init.\n");
+ kfree(scc);
+ return -ENODEV;
+ }
+ if(scc->mpucnf.io_base > 0) {
+ scc->mpu = 1;
+ printk(KERN_INFO "sb: Turning on MPU\n");
+ if(!probe_sbmpu(&scc->mpucnf, THIS_MODULE))
+ scc->mpu = 0;
+ }
+
+ return 1;
+}
+
+static void sb_unload(struct sb_card_config *scc)
+{
+ sb_dsp_unload(&scc->conf, 0);
+ if(scc->mpu)
+ unload_sbmpu(&scc->mpucnf);
+ kfree(scc);
+}
+
+/* Register legacy card with OSS subsystem */
+static int __init sb_init_legacy(void)
+{
+ struct sb_module_options sbmo = {0};
+
+ if((legacy = kzalloc(sizeof(struct sb_card_config), GFP_KERNEL)) == NULL) {
+ printk(KERN_ERR "sb: Error: Could not allocate memory\n");
+ return -ENOMEM;
+ }
+
+ legacy->conf.io_base = io;
+ legacy->conf.irq = irq;
+ legacy->conf.dma = dma;
+ legacy->conf.dma2 = dma16;
+ legacy->conf.card_subtype = type;
+
+ legacy->mpucnf.io_base = mpu_io;
+ legacy->mpucnf.irq = -1;
+ legacy->mpucnf.dma = -1;
+ legacy->mpucnf.dma2 = -1;
+
+ sbmo.esstype = esstype;
+ sbmo.sm_games = sm_games;
+ sbmo.acer = acer;
+
+ return sb_register_oss(legacy, &sbmo);
+}
+
+#ifdef CONFIG_PNP
+
+/* Populate the OSS subsystem structures with information from PnP */
+static void sb_dev2cfg(struct pnp_dev *dev, struct sb_card_config *scc)
+{
+ scc->conf.io_base = -1;
+ scc->conf.irq = -1;
+ scc->conf.dma = -1;
+ scc->conf.dma2 = -1;
+ scc->mpucnf.io_base = -1;
+ scc->mpucnf.irq = -1;
+ scc->mpucnf.dma = -1;
+ scc->mpucnf.dma2 = -1;
+
+ /* All clones layout their PnP tables differently and some use
+ different logical devices for the MPU */
+ if(!strncmp("CTL",scc->card_id,3)) {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,0);
+ scc->conf.dma2 = pnp_dma(dev,1);
+ scc->mpucnf.io_base = pnp_port_start(dev,1);
+ return;
+ }
+ if(!strncmp("tBA",scc->card_id,3)) {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,0);
+ scc->conf.dma2 = pnp_dma(dev,1);
+ return;
+ }
+ if(!strncmp("ESS",scc->card_id,3)) {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,0);
+ scc->conf.dma2 = pnp_dma(dev,1);
+ scc->mpucnf.io_base = pnp_port_start(dev,2);
+ return;
+ }
+ if(!strncmp("CMI",scc->card_id,3)) {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,0);
+ scc->conf.dma2 = pnp_dma(dev,1);
+ return;
+ }
+ if(!strncmp("RWB",scc->card_id,3)) {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,0);
+ return;
+ }
+ if(!strncmp("ALS",scc->card_id,3)) {
+ if(!strncmp("ALS0007",scc->card_id,7)) {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,0);
+ } else {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,1);
+ scc->conf.dma2 = pnp_dma(dev,0);
+ }
+ return;
+ }
+ if(!strncmp("RTL",scc->card_id,3)) {
+ scc->conf.io_base = pnp_port_start(dev,0);
+ scc->conf.irq = pnp_irq(dev,0);
+ scc->conf.dma = pnp_dma(dev,1);
+ scc->conf.dma2 = pnp_dma(dev,0);
+ }
+}
+
+static unsigned int sb_pnp_devices;
+
+/* Probe callback function for the PnP API */
+static int sb_pnp_probe(struct pnp_card_link *card, const struct pnp_card_device_id *card_id)
+{
+ struct sb_card_config *scc;
+ struct sb_module_options sbmo = {0}; /* Default to 0 for PnP */
+ struct pnp_dev *dev = pnp_request_card_device(card, card_id->devs[0].id, NULL);
+
+ if(!dev){
+ return -EBUSY;
+ }
+
+ if((scc = kzalloc(sizeof(struct sb_card_config), GFP_KERNEL)) == NULL) {
+ printk(KERN_ERR "sb: Error: Could not allocate memory\n");
+ return -ENOMEM;
+ }
+
+ printk(KERN_INFO "sb: PnP: Found Card Named = \"%s\", Card PnP id = " \
+ "%s, Device PnP id = %s\n", card->card->name, card_id->id,
+ dev->id->id);
+
+ scc->card_id = card_id->id;
+ scc->dev_id = dev->id->id;
+ sb_dev2cfg(dev, scc);
+
+ printk(KERN_INFO "sb: PnP: Detected at: io=0x%x, irq=%d, " \
+ "dma=%d, dma16=%d\n", scc->conf.io_base, scc->conf.irq,
+ scc->conf.dma, scc->conf.dma2);
+
+ pnp_set_card_drvdata(card, scc);
+ sb_pnp_devices++;
+
+ return sb_register_oss(scc, &sbmo);
+}
+
+static void sb_pnp_remove(struct pnp_card_link *card)
+{
+ struct sb_card_config *scc = pnp_get_card_drvdata(card);
+
+ if(!scc)
+ return;
+
+ printk(KERN_INFO "sb: PnP: Removing %s\n", scc->card_id);
+
+ sb_unload(scc);
+}
+
+static struct pnp_card_driver sb_pnp_driver = {
+ .name = "OSS SndBlstr", /* 16 character limit */
+ .id_table = sb_pnp_card_table,
+ .probe = sb_pnp_probe,
+ .remove = sb_pnp_remove,
+};
+MODULE_DEVICE_TABLE(pnp_card, sb_pnp_card_table);
+#endif /* CONFIG_PNP */
+
+static void sb_unregister_all(void)
+{
+#ifdef CONFIG_PNP
+ if (pnp_registered)
+ pnp_unregister_card_driver(&sb_pnp_driver);
+#endif
+}
+
+static int __init sb_init(void)
+{
+ int lres = 0;
+ int pres = 0;
+
+ printk(KERN_INFO "sb: Init: Starting Probe...\n");
+
+ if(io != -1 && irq != -1 && dma != -1) {
+ printk(KERN_INFO "sb: Probing legacy card with io=%x, "\
+ "irq=%d, dma=%d, dma16=%d\n",io, irq, dma, dma16);
+ lres = sb_init_legacy();
+ } else if((io != -1 || irq != -1 || dma != -1) ||
+ (!pnp && (io == -1 && irq == -1 && dma == -1)))
+ printk(KERN_ERR "sb: Error: At least io, irq, and dma "\
+ "must be set for legacy cards.\n");
+
+#ifdef CONFIG_PNP
+ if(pnp) {
+ int err = pnp_register_card_driver(&sb_pnp_driver);
+ if (!err)
+ pnp_registered = 1;
+ pres = sb_pnp_devices;
+ }
+#endif
+ printk(KERN_INFO "sb: Init: Done\n");
+
+ /* If either PnP or Legacy registered a card then return
+ * success */
+ if (pres == 0 && lres <= 0) {
+ sb_unregister_all();
+ return -ENODEV;
+ }
+ return 0;
+}
+
+static void __exit sb_exit(void)
+{
+ printk(KERN_INFO "sb: Unloading...\n");
+
+ /* Unload legacy card */
+ if (legacy) {
+ printk (KERN_INFO "sb: Unloading legacy card\n");
+ sb_unload(legacy);
+ }
+
+ sb_unregister_all();
+
+ vfree(smw_free);
+ smw_free = NULL;
+}
+
+module_init(sb_init);
+module_exit(sb_exit);
diff --git a/sound/oss/sb_card.h b/sound/oss/sb_card.h
new file mode 100644
index 0000000..5535cff
--- /dev/null
+++ b/sound/oss/sb_card.h
@@ -0,0 +1,149 @@
+/*
+ * sound/oss/sb_card.h
+ *
+ * This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this
+ * software for more info.
+ *
+ * 02-05-2002 Original Release, Paul Laufer <paul@laufernet.com>
+ */
+
+struct sb_card_config {
+ struct address_info conf;
+ struct address_info mpucnf;
+ const char *card_id;
+ const char *dev_id;
+ int mpu;
+};
+
+#ifdef CONFIG_PNP
+
+/*
+ * SoundBlaster PnP tables and structures.
+ */
+
+/* Card PnP ID Table */
+static struct pnp_card_device_id sb_pnp_card_table[] = {
+ /* Sound Blaster 16 */
+ {.id = "CTL0024", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL0025", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL0026", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL0027", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL0028", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL0029", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL002a", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL002b", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL002c", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL00ed", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
+ /* Sound Blaster 16 */
+ {.id = "CTL0086", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
+ /* Sound Blaster Vibra16S */
+ {.id = "CTL0051", .driver_data = 0, .devs = { {.id="CTL0001"}, } },
+ /* Sound Blaster Vibra16C */
+ {.id = "CTL0070", .driver_data = 0, .devs = { {.id="CTL0001"}, } },
+ /* Sound Blaster Vibra16CL */
+ {.id = "CTL0080", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
+ /* Sound Blaster Vibra16CL */
+ {.id = "CTL00F0", .driver_data = 0, .devs = { {.id="CTL0043"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0039", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0042", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0043", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0044", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0045", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0046", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0047", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0048", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL0054", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
+ /* Sound Blaster AWE 32 */
+ {.id = "CTL009C", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
+ /* Createive SB32 PnP */
+ {.id = "CTL009F", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
+ /* Sound Blaster AWE 64 */
+ {.id = "CTL009D", .driver_data = 0, .devs = { {.id="CTL0042"}, } },
+ /* Sound Blaster AWE 64 Gold */
+ {.id = "CTL009E", .driver_data = 0, .devs = { {.id="CTL0044"}, } },
+ /* Sound Blaster AWE 64 Gold */
+ {.id = "CTL00B2", .driver_data = 0, .devs = { {.id="CTL0044"}, } },
+ /* Sound Blaster AWE 64 */
+ {.id = "CTL00C1", .driver_data = 0, .devs = { {.id="CTL0042"}, } },
+ /* Sound Blaster AWE 64 */
+ {.id = "CTL00C3", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
+ /* Sound Blaster AWE 64 */
+ {.id = "CTL00C5", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
+ /* Sound Blaster AWE 64 */
+ {.id = "CTL00C7", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
+ /* Sound Blaster AWE 64 */
+ {.id = "CTL00E4", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
+ /* Sound Blaster AWE 64 */
+ {.id = "CTL00E9", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
+ /* ESS 1868 */
+ {.id = "ESS0968", .driver_data = 0, .devs = { {.id="ESS0968"}, } },
+ /* ESS 1868 */
+ {.id = "ESS1868", .driver_data = 0, .devs = { {.id="ESS1868"}, } },
+ /* ESS 1868 */
+ {.id = "ESS1868", .driver_data = 0, .devs = { {.id="ESS8611"}, } },
+ /* ESS 1869 PnP AudioDrive */
+ {.id = "ESS0003", .driver_data = 0, .devs = { {.id="ESS1869"}, } },
+ /* ESS 1869 */
+ {.id = "ESS1869", .driver_data = 0, .devs = { {.id="ESS1869"}, } },
+ /* ESS 1878 */
+ {.id = "ESS1878", .driver_data = 0, .devs = { {.id="ESS1878"}, } },
+ /* ESS 1879 */
+ {.id = "ESS1879", .driver_data = 0, .devs = { {.id="ESS1879"}, } },
+ /* CMI 8330 SoundPRO */
+ {.id = "CMI0001", .driver_data = 0, .devs = { {.id="@X@0001"},
+ {.id="@H@0001"},
+ {.id="@@@0001"}, } },
+ /* Diamond DT0197H */
+ {.id = "RWR1688", .driver_data = 0, .devs = { {.id="@@@0001"},
+ {.id="@X@0001"},
+ {.id="@H@0001"}, } },
+ /* ALS007 */
+ {.id = "ALS0007", .driver_data = 0, .devs = { {.id="@@@0001"},
+ {.id="@X@0001"},
+ {.id="@H@0001"}, } },
+ /* ALS100 */
+ {.id = "ALS0001", .driver_data = 0, .devs = { {.id="@@@0001"},
+ {.id="@X@0001"},
+ {.id="@H@0001"}, } },
+ /* ALS110 */
+ {.id = "ALS0110", .driver_data = 0, .devs = { {.id="@@@1001"},
+ {.id="@X@1001"},
+ {.id="@H@0001"}, } },
+ /* ALS120 */
+ {.id = "ALS0120", .driver_data = 0, .devs = { {.id="@@@2001"},
+ {.id="@X@2001"},
+ {.id="@H@0001"}, } },
+ /* ALS200 */
+ {.id = "ALS0200", .driver_data = 0, .devs = { {.id="@@@0020"},
+ {.id="@X@0030"},
+ {.id="@H@0001"}, } },
+ /* ALS200 */
+ {.id = "RTL3000", .driver_data = 0, .devs = { {.id="@@@2001"},
+ {.id="@X@2001"},
+ {.id="@H@0001"}, } },
+ /* Sound Blaster 16 (Virtual PC 2004) */
+ {.id = "tBA03b0", .driver_data = 0, .devs = { {.id="PNPb003"}, } },
+ /* -end- */
+ {.id = "", }
+};
+
+#endif
diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c
new file mode 100644
index 0000000..77d0e5e
--- /dev/null
+++ b/sound/oss/sb_common.c
@@ -0,0 +1,1291 @@
+/*
+ * sound/oss/sb_common.c
+ *
+ * Common routines for Sound Blaster compatible cards.
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Daniel J. Rodriksson: Modified sbintr to handle 8 and 16 bit interrupts
+ * for full duplex support ( only sb16 by now )
+ * Rolf Fokkens: Added (BETA?) support for ES1887 chips.
+ * (fokkensr@vertis.nl) Which means: You can adjust the recording levels.
+ *
+ * 2000/01/18 - separated sb_card and sb_common -
+ * Jeff Garzik <jgarzik@pobox.com>
+ *
+ * 2000/09/18 - got rid of attach_uart401
+ * Arnaldo Carvalho de Melo <acme@conectiva.com.br>
+ *
+ * 2001/01/26 - replaced CLI/STI with spinlocks
+ * Chris Rankin <rankinc@zipworld.com.au>
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/spinlock.h>
+
+#include "sound_config.h"
+#include "sound_firmware.h"
+
+#include "mpu401.h"
+
+#include "sb_mixer.h"
+#include "sb.h"
+#include "sb_ess.h"
+
+/*
+ * global module flag
+ */
+
+int sb_be_quiet;
+
+static sb_devc *detected_devc; /* For communication from probe to init */
+static sb_devc *last_devc; /* For MPU401 initialization */
+
+static unsigned char jazz_irq_bits[] = {
+ 0, 0, 2, 3, 0, 1, 0, 4, 0, 2, 5, 0, 0, 0, 0, 6
+};
+
+static unsigned char jazz_dma_bits[] = {
+ 0, 1, 0, 2, 0, 3, 0, 4
+};
+
+void *smw_free;
+
+/*
+ * Jazz16 chipset specific control variables
+ */
+
+static int jazz16_base; /* Not detected */
+static unsigned char jazz16_bits; /* I/O relocation bits */
+static DEFINE_SPINLOCK(jazz16_lock);
+
+/*
+ * Logitech Soundman Wave specific initialization code
+ */
+
+#ifdef SMW_MIDI0001_INCLUDED
+#include "smw-midi0001.h"
+#else
+static unsigned char *smw_ucode;
+static int smw_ucodeLen;
+
+#endif
+
+static sb_devc *last_sb; /* Last sb loaded */
+
+int sb_dsp_command(sb_devc * devc, unsigned char val)
+{
+ int i;
+ unsigned long limit;
+
+ limit = jiffies + HZ / 10; /* Timeout */
+
+ /*
+ * Note! the i<500000 is an emergency exit. The sb_dsp_command() is sometimes
+ * called while interrupts are disabled. This means that the timer is
+ * disabled also. However the timeout situation is a abnormal condition.
+ * Normally the DSP should be ready to accept commands after just couple of
+ * loops.
+ */
+
+ for (i = 0; i < 500000 && (limit-jiffies)>0; i++)
+ {
+ if ((inb(DSP_STATUS) & 0x80) == 0)
+ {
+ outb((val), DSP_COMMAND);
+ return 1;
+ }
+ }
+ printk(KERN_WARNING "Sound Blaster: DSP command(%x) timeout.\n", val);
+ return 0;
+}
+
+int sb_dsp_get_byte(sb_devc * devc)
+{
+ int i;
+
+ for (i = 1000; i; i--)
+ {
+ if (inb(DSP_DATA_AVAIL) & 0x80)
+ return inb(DSP_READ);
+ }
+ return 0xffff;
+}
+
+static void sb_intr (sb_devc *devc)
+{
+ int status;
+ unsigned char src = 0xff;
+
+ if (devc->model == MDL_SB16)
+ {
+ src = sb_getmixer(devc, IRQ_STAT); /* Interrupt source register */
+
+ if (src & 4) /* MPU401 interrupt */
+ if(devc->midi_irq_cookie)
+ uart401intr(devc->irq, devc->midi_irq_cookie);
+
+ if (!(src & 3))
+ return; /* Not a DSP interrupt */
+ }
+ if (devc->intr_active && (!devc->fullduplex || (src & 0x01)))
+ {
+ switch (devc->irq_mode)
+ {
+ case IMODE_OUTPUT:
+ DMAbuf_outputintr(devc->dev, 1);
+ break;
+
+ case IMODE_INPUT:
+ DMAbuf_inputintr(devc->dev);
+ break;
+
+ case IMODE_INIT:
+ break;
+
+ case IMODE_MIDI:
+ sb_midi_interrupt(devc);
+ break;
+
+ default:
+ /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */
+ ;
+ }
+ }
+ else if (devc->intr_active_16 && (src & 0x02))
+ {
+ switch (devc->irq_mode_16)
+ {
+ case IMODE_OUTPUT:
+ DMAbuf_outputintr(devc->dev, 1);
+ break;
+
+ case IMODE_INPUT:
+ DMAbuf_inputintr(devc->dev);
+ break;
+
+ case IMODE_INIT:
+ break;
+
+ default:
+ /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */
+ ;
+ }
+ }
+ /*
+ * Acknowledge interrupts
+ */
+
+ if (src & 0x01)
+ status = inb(DSP_DATA_AVAIL);
+
+ if (devc->model == MDL_SB16 && src & 0x02)
+ status = inb(DSP_DATA_AVL16);
+}
+
+static void pci_intr(sb_devc *devc)
+{
+ int src = inb(devc->pcibase+0x1A);
+ src&=3;
+ if(src)
+ sb_intr(devc);
+}
+
+static irqreturn_t sbintr(int irq, void *dev_id)
+{
+ sb_devc *devc = dev_id;
+
+ devc->irq_ok = 1;
+
+ switch (devc->model) {
+ case MDL_ESSPCI:
+ pci_intr (devc);
+ break;
+
+ case MDL_ESS:
+ ess_intr (devc);
+ break;
+ default:
+ sb_intr (devc);
+ break;
+ }
+ return IRQ_HANDLED;
+}
+
+int sb_dsp_reset(sb_devc * devc)
+{
+ int loopc;
+
+ DEB(printk("Entered sb_dsp_reset()\n"));
+
+ if (devc->model == MDL_ESS) return ess_dsp_reset (devc);
+
+ /* This is only for non-ESS chips */
+
+ outb(1, DSP_RESET);
+
+ udelay(10);
+ outb(0, DSP_RESET);
+ udelay(30);
+
+ for (loopc = 0; loopc < 1000 && !(inb(DSP_DATA_AVAIL) & 0x80); loopc++);
+
+ if (inb(DSP_READ) != 0xAA)
+ {
+ DDB(printk("sb: No response to RESET\n"));
+ return 0; /* Sorry */
+ }
+
+ DEB(printk("sb_dsp_reset() OK\n"));
+
+ return 1;
+}
+
+static void dsp_get_vers(sb_devc * devc)
+{
+ int i;
+
+ unsigned long flags;
+
+ DDB(printk("Entered dsp_get_vers()\n"));
+ spin_lock_irqsave(&devc->lock, flags);
+ devc->major = devc->minor = 0;
+ sb_dsp_command(devc, 0xe1); /* Get version */
+
+ for (i = 100000; i; i--)
+ {
+ if (inb(DSP_DATA_AVAIL) & 0x80)
+ {
+ if (devc->major == 0)
+ devc->major = inb(DSP_READ);
+ else
+ {
+ devc->minor = inb(DSP_READ);
+ break;
+ }
+ }
+ }
+ spin_unlock_irqrestore(&devc->lock, flags);
+ DDB(printk("DSP version %d.%02d\n", devc->major, devc->minor));
+}
+
+static int sb16_set_dma_hw(sb_devc * devc)
+{
+ int bits;
+
+ if (devc->dma8 != 0 && devc->dma8 != 1 && devc->dma8 != 3)
+ {
+ printk(KERN_ERR "SB16: Invalid 8 bit DMA (%d)\n", devc->dma8);
+ return 0;
+ }
+ bits = (1 << devc->dma8);
+
+ if (devc->dma16 >= 5 && devc->dma16 <= 7)
+ bits |= (1 << devc->dma16);
+
+ sb_setmixer(devc, DMA_NR, bits);
+ return 1;
+}
+
+static void sb16_set_mpu_port(sb_devc * devc, struct address_info *hw_config)
+{
+ /*
+ * This routine initializes new MIDI port setup register of SB Vibra (CT2502).
+ */
+ unsigned char bits = sb_getmixer(devc, 0x84) & ~0x06;
+
+ switch (hw_config->io_base)
+ {
+ case 0x300:
+ sb_setmixer(devc, 0x84, bits | 0x04);
+ break;
+
+ case 0x330:
+ sb_setmixer(devc, 0x84, bits | 0x00);
+ break;
+
+ default:
+ sb_setmixer(devc, 0x84, bits | 0x02); /* Disable MPU */
+ printk(KERN_ERR "SB16: Invalid MIDI I/O port %x\n", hw_config->io_base);
+ }
+}
+
+static int sb16_set_irq_hw(sb_devc * devc, int level)
+{
+ int ival;
+
+ switch (level)
+ {
+ case 5:
+ ival = 2;
+ break;
+ case 7:
+ ival = 4;
+ break;
+ case 9:
+ ival = 1;
+ break;
+ case 10:
+ ival = 8;
+ break;
+ default:
+ printk(KERN_ERR "SB16: Invalid IRQ%d\n", level);
+ return 0;
+ }
+ sb_setmixer(devc, IRQ_NR, ival);
+ return 1;
+}
+
+static void relocate_Jazz16(sb_devc * devc, struct address_info *hw_config)
+{
+ unsigned char bits = 0;
+ unsigned long flags;
+
+ if (jazz16_base != 0 && jazz16_base != hw_config->io_base)
+ return;
+
+ switch (hw_config->io_base)
+ {
+ case 0x220:
+ bits = 1;
+ break;
+ case 0x240:
+ bits = 2;
+ break;
+ case 0x260:
+ bits = 3;
+ break;
+ default:
+ return;
+ }
+ bits = jazz16_bits = bits << 5;
+ jazz16_base = hw_config->io_base;
+
+ /*
+ * Magic wake up sequence by writing to 0x201 (aka Joystick port)
+ */
+ spin_lock_irqsave(&jazz16_lock, flags);
+ outb((0xAF), 0x201);
+ outb((0x50), 0x201);
+ outb((bits), 0x201);
+ spin_unlock_irqrestore(&jazz16_lock, flags);
+}
+
+static int init_Jazz16(sb_devc * devc, struct address_info *hw_config)
+{
+ char name[100];
+ /*
+ * First try to check that the card has Jazz16 chip. It identifies itself
+ * by returning 0x12 as response to DSP command 0xfa.
+ */
+
+ if (!sb_dsp_command(devc, 0xfa))
+ return 0;
+
+ if (sb_dsp_get_byte(devc) != 0x12)
+ return 0;
+
+ /*
+ * OK so far. Now configure the IRQ and DMA channel used by the card.
+ */
+ if (hw_config->irq < 1 || hw_config->irq > 15 || jazz_irq_bits[hw_config->irq] == 0)
+ {
+ printk(KERN_ERR "Jazz16: Invalid interrupt (IRQ%d)\n", hw_config->irq);
+ return 0;
+ }
+ if (hw_config->dma < 0 || hw_config->dma > 3 || jazz_dma_bits[hw_config->dma] == 0)
+ {
+ printk(KERN_ERR "Jazz16: Invalid 8 bit DMA (DMA%d)\n", hw_config->dma);
+ return 0;
+ }
+ if (hw_config->dma2 < 0)
+ {
+ printk(KERN_ERR "Jazz16: No 16 bit DMA channel defined\n");
+ return 0;
+ }
+ if (hw_config->dma2 < 5 || hw_config->dma2 > 7 || jazz_dma_bits[hw_config->dma2] == 0)
+ {
+ printk(KERN_ERR "Jazz16: Invalid 16 bit DMA (DMA%d)\n", hw_config->dma2);
+ return 0;
+ }
+ devc->dma16 = hw_config->dma2;
+
+ if (!sb_dsp_command(devc, 0xfb))
+ return 0;
+
+ if (!sb_dsp_command(devc, jazz_dma_bits[hw_config->dma] |
+ (jazz_dma_bits[hw_config->dma2] << 4)))
+ return 0;
+
+ if (!sb_dsp_command(devc, jazz_irq_bits[hw_config->irq]))
+ return 0;
+
+ /*
+ * Now we have configured a standard Jazz16 device.
+ */
+ devc->model = MDL_JAZZ;
+ strcpy(name, "Jazz16");
+
+ hw_config->name = "Jazz16";
+ devc->caps |= SB_NO_MIDI;
+ return 1;
+}
+
+static void relocate_ess1688(sb_devc * devc)
+{
+ unsigned char bits;
+
+ switch (devc->base)
+ {
+ case 0x220:
+ bits = 0x04;
+ break;
+ case 0x230:
+ bits = 0x05;
+ break;
+ case 0x240:
+ bits = 0x06;
+ break;
+ case 0x250:
+ bits = 0x07;
+ break;
+ default:
+ return; /* Wrong port */
+ }
+
+ DDB(printk("Doing ESS1688 address selection\n"));
+
+ /*
+ * ES1688 supports two alternative ways for software address config.
+ * First try the so called Read-Sequence-Key method.
+ */
+
+ /* Reset the sequence logic */
+ inb(0x229);
+ inb(0x229);
+ inb(0x229);
+
+ /* Perform the read sequence */
+ inb(0x22b);
+ inb(0x229);
+ inb(0x22b);
+ inb(0x229);
+ inb(0x229);
+ inb(0x22b);
+ inb(0x229);
+
+ /* Select the base address by reading from it. Then probe using the port. */
+ inb(devc->base);
+ if (sb_dsp_reset(devc)) /* Bingo */
+ return;
+
+#if 0 /* This causes system lockups (Nokia 386/25 at least) */
+ /*
+ * The last resort is the system control register method.
+ */
+
+ outb((0x00), 0xfb); /* 0xFB is the unlock register */
+ outb((0x00), 0xe0); /* Select index 0 */
+ outb((bits), 0xe1); /* Write the config bits */
+ outb((0x00), 0xf9); /* 0xFB is the lock register */
+#endif
+}
+
+int sb_dsp_detect(struct address_info *hw_config, int pci, int pciio, struct sb_module_options *sbmo)
+{
+ sb_devc sb_info;
+ sb_devc *devc = &sb_info;
+
+ memset((char *) &sb_info, 0, sizeof(sb_info)); /* Zero everything */
+
+ /* Copy module options in place */
+ if(sbmo) memcpy(&devc->sbmo, sbmo, sizeof(struct sb_module_options));
+
+ sb_info.my_mididev = -1;
+ sb_info.my_mixerdev = -1;
+ sb_info.dev = -1;
+
+ /*
+ * Initialize variables
+ */
+
+ DDB(printk("sb_dsp_detect(%x) entered\n", hw_config->io_base));
+
+ spin_lock_init(&devc->lock);
+ devc->type = hw_config->card_subtype;
+
+ devc->base = hw_config->io_base;
+ devc->irq = hw_config->irq;
+ devc->dma8 = hw_config->dma;
+
+ devc->dma16 = -1;
+ devc->pcibase = pciio;
+
+ if(pci == SB_PCI_ESSMAESTRO)
+ {
+ devc->model = MDL_ESSPCI;
+ devc->caps |= SB_PCI_IRQ;
+ hw_config->driver_use_1 |= SB_PCI_IRQ;
+ hw_config->card_subtype = MDL_ESSPCI;
+ }
+
+ if(pci == SB_PCI_YAMAHA)
+ {
+ devc->model = MDL_YMPCI;
+ devc->caps |= SB_PCI_IRQ;
+ hw_config->driver_use_1 |= SB_PCI_IRQ;
+ hw_config->card_subtype = MDL_YMPCI;
+
+ printk("Yamaha PCI mode.\n");
+ }
+
+ if (devc->sbmo.acer)
+ {
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ inb(devc->base + 0x09);
+ inb(devc->base + 0x09);
+ inb(devc->base + 0x09);
+ inb(devc->base + 0x0b);
+ inb(devc->base + 0x09);
+ inb(devc->base + 0x0b);
+ inb(devc->base + 0x09);
+ inb(devc->base + 0x09);
+ inb(devc->base + 0x0b);
+ inb(devc->base + 0x09);
+ inb(devc->base + 0x00);
+ spin_unlock_irqrestore(&devc->lock, flags);
+ }
+ /*
+ * Detect the device
+ */
+
+ if (sb_dsp_reset(devc))
+ dsp_get_vers(devc);
+ else
+ devc->major = 0;
+
+ if (devc->type == 0 || devc->type == MDL_JAZZ || devc->type == MDL_SMW)
+ if (devc->major == 0 || (devc->major == 3 && devc->minor == 1))
+ relocate_Jazz16(devc, hw_config);
+
+ if (devc->major == 0 && (devc->type == MDL_ESS || devc->type == 0))
+ relocate_ess1688(devc);
+
+ if (!sb_dsp_reset(devc))
+ {
+ DDB(printk("SB reset failed\n"));
+#ifdef MODULE
+ printk(KERN_INFO "sb: dsp reset failed.\n");
+#endif
+ return 0;
+ }
+ if (devc->major == 0)
+ dsp_get_vers(devc);
+
+ if (devc->major == 3 && devc->minor == 1)
+ {
+ if (devc->type == MDL_AZTECH) /* SG Washington? */
+ {
+ if (sb_dsp_command(devc, 0x09))
+ if (sb_dsp_command(devc, 0x00)) /* Enter WSS mode */
+ {
+ int i;
+
+ /* Have some delay */
+ for (i = 0; i < 10000; i++)
+ inb(DSP_DATA_AVAIL);
+ devc->caps = SB_NO_AUDIO | SB_NO_MIDI; /* Mixer only */
+ devc->model = MDL_AZTECH;
+ }
+ }
+ }
+
+ if(devc->type == MDL_ESSPCI)
+ devc->model = MDL_ESSPCI;
+
+ if(devc->type == MDL_YMPCI)
+ {
+ printk("YMPCI selected\n");
+ devc->model = MDL_YMPCI;
+ }
+
+ /*
+ * Save device information for sb_dsp_init()
+ */
+
+
+ detected_devc = kmalloc(sizeof(sb_devc), GFP_KERNEL);
+ if (detected_devc == NULL)
+ {
+ printk(KERN_ERR "sb: Can't allocate memory for device information\n");
+ return 0;
+ }
+ memcpy(detected_devc, devc, sizeof(sb_devc));
+ MDB(printk(KERN_INFO "SB %d.%02d detected OK (%x)\n", devc->major, devc->minor, hw_config->io_base));
+ return 1;
+}
+
+int sb_dsp_init(struct address_info *hw_config, struct module *owner)
+{
+ sb_devc *devc;
+ char name[100];
+ extern int sb_be_quiet;
+ int mixer22, mixer30;
+
+/*
+ * Check if we had detected a SB device earlier
+ */
+ DDB(printk("sb_dsp_init(%x) entered\n", hw_config->io_base));
+ name[0] = 0;
+
+ if (detected_devc == NULL)
+ {
+ MDB(printk("No detected device\n"));
+ return 0;
+ }
+ devc = detected_devc;
+ detected_devc = NULL;
+
+ if (devc->base != hw_config->io_base)
+ {
+ DDB(printk("I/O port mismatch\n"));
+ release_region(devc->base, 16);
+ return 0;
+ }
+ /*
+ * Now continue initialization of the device
+ */
+
+ devc->caps = hw_config->driver_use_1;
+
+ if (!((devc->caps & SB_NO_AUDIO) && (devc->caps & SB_NO_MIDI)) && hw_config->irq > 0)
+ { /* IRQ setup */
+
+ /*
+ * ESS PCI cards do shared PCI IRQ stuff. Since they
+ * will get shared PCI irq lines we must cope.
+ */
+
+ int i=(devc->caps&SB_PCI_IRQ)?IRQF_SHARED:0;
+
+ if (request_irq(hw_config->irq, sbintr, i, "soundblaster", devc) < 0)
+ {
+ printk(KERN_ERR "SB: Can't allocate IRQ%d\n", hw_config->irq);
+ release_region(devc->base, 16);
+ return 0;
+ }
+ devc->irq_ok = 0;
+
+ if (devc->major == 4)
+ if (!sb16_set_irq_hw(devc, devc->irq)) /* Unsupported IRQ */
+ {
+ free_irq(devc->irq, devc);
+ release_region(devc->base, 16);
+ return 0;
+ }
+ if ((devc->type == 0 || devc->type == MDL_ESS) &&
+ devc->major == 3 && devc->minor == 1)
+ { /* Handle various chipsets which claim they are SB Pro compatible */
+ if ((devc->type != 0 && devc->type != MDL_ESS) ||
+ !ess_init(devc, hw_config))
+ {
+ if ((devc->type != 0 && devc->type != MDL_JAZZ &&
+ devc->type != MDL_SMW) || !init_Jazz16(devc, hw_config))
+ {
+ DDB(printk("This is a genuine SB Pro\n"));
+ }
+ }
+ }
+ if (devc->major == 4 && devc->minor <= 11 ) /* Won't work */
+ devc->irq_ok = 1;
+ else
+ {
+ int n;
+
+ for (n = 0; n < 3 && devc->irq_ok == 0; n++)
+ {
+ if (sb_dsp_command(devc, 0xf2)) /* Cause interrupt immediately */
+ {
+ int i;
+
+ for (i = 0; !devc->irq_ok && i < 10000; i++);
+ }
+ }
+ if (!devc->irq_ok)
+ printk(KERN_WARNING "sb: Interrupt test on IRQ%d failed - Probable IRQ conflict\n", devc->irq);
+ else
+ {
+ DDB(printk("IRQ test OK (IRQ%d)\n", devc->irq));
+ }
+ }
+ } /* IRQ setup */
+
+ last_sb = devc;
+
+ switch (devc->major)
+ {
+ case 1: /* SB 1.0 or 1.5 */
+ devc->model = hw_config->card_subtype = MDL_SB1;
+ break;
+
+ case 2: /* SB 2.x */
+ if (devc->minor == 0)
+ devc->model = hw_config->card_subtype = MDL_SB2;
+ else
+ devc->model = hw_config->card_subtype = MDL_SB201;
+ break;
+
+ case 3: /* SB Pro and most clones */
+ switch (devc->model) {
+ case 0:
+ devc->model = hw_config->card_subtype = MDL_SBPRO;
+ if (hw_config->name == NULL)
+ hw_config->name = "Sound Blaster Pro (8 BIT ONLY)";
+ break;
+ case MDL_ESS:
+ ess_dsp_init(devc, hw_config);
+ break;
+ }
+ break;
+
+ case 4:
+ devc->model = hw_config->card_subtype = MDL_SB16;
+ /*
+ * ALS007 and ALS100 return DSP version 4.2 and have 2 post-reset !=0
+ * registers at 0x3c and 0x4c (output ctrl registers on ALS007) whereas
+ * a "standard" SB16 doesn't have a register at 0x4c. ALS100 actively
+ * updates register 0x22 whenever 0x30 changes, as per the SB16 spec.
+ * Since ALS007 doesn't, this can be used to differentiate the 2 cards.
+ */
+ if ((devc->minor == 2) && sb_getmixer(devc,0x3c) && sb_getmixer(devc,0x4c))
+ {
+ mixer30 = sb_getmixer(devc,0x30);
+ sb_setmixer(devc,0x22,(mixer22=sb_getmixer(devc,0x22)) & 0x0f);
+ sb_setmixer(devc,0x30,0xff);
+ /* ALS100 will force 0x30 to 0xf8 like SB16; ALS007 will allow 0xff. */
+ /* Register 0x22 & 0xf0 on ALS100 == 0xf0; on ALS007 it == 0x10. */
+ if ((sb_getmixer(devc,0x30) != 0xff) || ((sb_getmixer(devc,0x22) & 0xf0) != 0x10))
+ {
+ devc->submodel = SUBMDL_ALS100;
+ if (hw_config->name == NULL)
+ hw_config->name = "Sound Blaster 16 (ALS-100)";
+ }
+ else
+ {
+ sb_setmixer(devc,0x3c,0x1f); /* Enable all inputs */
+ sb_setmixer(devc,0x4c,0x1f);
+ sb_setmixer(devc,0x22,mixer22); /* Restore 0x22 to original value */
+ devc->submodel = SUBMDL_ALS007;
+ if (hw_config->name == NULL)
+ hw_config->name = "Sound Blaster 16 (ALS-007)";
+ }
+ sb_setmixer(devc,0x30,mixer30);
+ }
+ else if (hw_config->name == NULL)
+ hw_config->name = "Sound Blaster 16";
+
+ if (hw_config->dma2 == -1)
+ devc->dma16 = devc->dma8;
+ else if (hw_config->dma2 < 5 || hw_config->dma2 > 7)
+ {
+ printk(KERN_WARNING "SB16: Bad or missing 16 bit DMA channel\n");
+ devc->dma16 = devc->dma8;
+ }
+ else
+ devc->dma16 = hw_config->dma2;
+
+ if(!sb16_set_dma_hw(devc)) {
+ free_irq(devc->irq, devc);
+ release_region(hw_config->io_base, 16);
+ return 0;
+ }
+
+ devc->caps |= SB_NO_MIDI;
+ }
+
+ if (!(devc->caps & SB_NO_MIXER))
+ if (devc->major == 3 || devc->major == 4)
+ sb_mixer_init(devc, owner);
+
+ if (!(devc->caps & SB_NO_MIDI))
+ sb_dsp_midi_init(devc, owner);
+
+ if (hw_config->name == NULL)
+ hw_config->name = "Sound Blaster (8 BIT/MONO ONLY)";
+
+ sprintf(name, "%s (%d.%02d)", hw_config->name, devc->major, devc->minor);
+ conf_printf(name, hw_config);
+
+ /*
+ * Assuming that a sound card is Sound Blaster (compatible) is the most common
+ * configuration error and the mother of all problems. Usually sound cards
+ * emulate SB Pro but in addition they have a 16 bit native mode which should be
+ * used in Unix. See Readme.cards for more information about configuring OSS/Free
+ * properly.
+ */
+ if (devc->model <= MDL_SBPRO)
+ {
+ if (devc->major == 3 && devc->minor != 1) /* "True" SB Pro should have v3.1 (rare ones may have 3.2). */
+ {
+ printk(KERN_INFO "This sound card may not be fully Sound Blaster Pro compatible.\n");
+ printk(KERN_INFO "In many cases there is another way to configure OSS so that\n");
+ printk(KERN_INFO "it works properly with OSS (for example in 16 bit mode).\n");
+ printk(KERN_INFO "Please ignore this message if you _really_ have a SB Pro.\n");
+ }
+ else if (!sb_be_quiet && devc->model == MDL_SBPRO)
+ {
+ printk(KERN_INFO "SB DSP version is just %d.%02d which means that your card is\n", devc->major, devc->minor);
+ printk(KERN_INFO "several years old (8 bit only device) or alternatively the sound driver\n");
+ printk(KERN_INFO "is incorrectly configured.\n");
+ }
+ }
+ hw_config->card_subtype = devc->model;
+ hw_config->slots[0]=devc->dev;
+ last_devc = devc; /* For SB MPU detection */
+
+ if (!(devc->caps & SB_NO_AUDIO) && devc->dma8 >= 0)
+ {
+ if (sound_alloc_dma(devc->dma8, "SoundBlaster8"))
+ {
+ printk(KERN_WARNING "Sound Blaster: Can't allocate 8 bit DMA channel %d\n", devc->dma8);
+ }
+ if (devc->dma16 >= 0 && devc->dma16 != devc->dma8)
+ {
+ if (sound_alloc_dma(devc->dma16, "SoundBlaster16"))
+ printk(KERN_WARNING "Sound Blaster: can't allocate 16 bit DMA channel %d.\n", devc->dma16);
+ }
+ sb_audio_init(devc, name, owner);
+ hw_config->slots[0]=devc->dev;
+ }
+ else
+ {
+ MDB(printk("Sound Blaster: no audio devices found.\n"));
+ }
+ return 1;
+}
+
+/* if (sbmpu) below we allow mpu401 to manage the midi devs
+ otherwise we have to unload them. (Andrzej Krzysztofowicz) */
+
+void sb_dsp_unload(struct address_info *hw_config, int sbmpu)
+{
+ sb_devc *devc;
+
+ devc = audio_devs[hw_config->slots[0]]->devc;
+
+ if (devc && devc->base == hw_config->io_base)
+ {
+ if ((devc->model & MDL_ESS) && devc->pcibase)
+ release_region(devc->pcibase, 8);
+
+ release_region(devc->base, 16);
+
+ if (!(devc->caps & SB_NO_AUDIO))
+ {
+ sound_free_dma(devc->dma8);
+ if (devc->dma16 >= 0)
+ sound_free_dma(devc->dma16);
+ }
+ if (!(devc->caps & SB_NO_AUDIO && devc->caps & SB_NO_MIDI))
+ {
+ if (devc->irq > 0)
+ free_irq(devc->irq, devc);
+
+ sb_mixer_unload(devc);
+ /* We don't have to do this bit any more the UART401 is its own
+ master -- Krzysztof Halasa */
+ /* But we have to do it, if UART401 is not detected */
+ if (!sbmpu)
+ sound_unload_mididev(devc->my_mididev);
+ sound_unload_audiodev(devc->dev);
+ }
+ kfree(devc);
+ }
+ else
+ release_region(hw_config->io_base, 16);
+
+ kfree(detected_devc);
+}
+
+/*
+ * Mixer access routines
+ *
+ * ES1887 modifications: some mixer registers reside in the
+ * range above 0xa0. These must be accessed in another way.
+ */
+
+void sb_setmixer(sb_devc * devc, unsigned int port, unsigned int value)
+{
+ unsigned long flags;
+
+ if (devc->model == MDL_ESS) {
+ ess_setmixer (devc, port, value);
+ return;
+ }
+
+ spin_lock_irqsave(&devc->lock, flags);
+
+ outb(((unsigned char) (port & 0xff)), MIXER_ADDR);
+ udelay(20);
+ outb(((unsigned char) (value & 0xff)), MIXER_DATA);
+ udelay(20);
+
+ spin_unlock_irqrestore(&devc->lock, flags);
+}
+
+unsigned int sb_getmixer(sb_devc * devc, unsigned int port)
+{
+ unsigned int val;
+ unsigned long flags;
+
+ if (devc->model == MDL_ESS) return ess_getmixer (devc, port);
+
+ spin_lock_irqsave(&devc->lock, flags);
+
+ outb(((unsigned char) (port & 0xff)), MIXER_ADDR);
+ udelay(20);
+ val = inb(MIXER_DATA);
+ udelay(20);
+
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ return val;
+}
+
+void sb_chgmixer
+ (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val)
+{
+ int value;
+
+ value = sb_getmixer(devc, reg);
+ value = (value & ~mask) | (val & mask);
+ sb_setmixer(devc, reg, value);
+}
+
+/*
+ * MPU401 MIDI initialization.
+ */
+
+static void smw_putmem(sb_devc * devc, int base, int addr, unsigned char val)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&jazz16_lock, flags); /* NOT the SB card? */
+
+ outb((addr & 0xff), base + 1); /* Low address bits */
+ outb((addr >> 8), base + 2); /* High address bits */
+ outb((val), base); /* Data */
+
+ spin_unlock_irqrestore(&jazz16_lock, flags);
+}
+
+static unsigned char smw_getmem(sb_devc * devc, int base, int addr)
+{
+ unsigned long flags;
+ unsigned char val;
+
+ spin_lock_irqsave(&jazz16_lock, flags); /* NOT the SB card? */
+
+ outb((addr & 0xff), base + 1); /* Low address bits */
+ outb((addr >> 8), base + 2); /* High address bits */
+ val = inb(base); /* Data */
+
+ spin_unlock_irqrestore(&jazz16_lock, flags);
+ return val;
+}
+
+static int smw_midi_init(sb_devc * devc, struct address_info *hw_config)
+{
+ int mpu_base = hw_config->io_base;
+ int mp_base = mpu_base + 4; /* Microcontroller base */
+ int i;
+ unsigned char control;
+
+
+ /*
+ * Reset the microcontroller so that the RAM can be accessed
+ */
+
+ control = inb(mpu_base + 7);
+ outb((control | 3), mpu_base + 7); /* Set last two bits to 1 (?) */
+ outb(((control & 0xfe) | 2), mpu_base + 7); /* xxxxxxx0 resets the mc */
+
+ mdelay(3); /* Wait at least 1ms */
+
+ outb((control & 0xfc), mpu_base + 7); /* xxxxxx00 enables RAM */
+
+ /*
+ * Detect microcontroller by probing the 8k RAM area
+ */
+ smw_putmem(devc, mp_base, 0, 0x00);
+ smw_putmem(devc, mp_base, 1, 0xff);
+ udelay(10);
+
+ if (smw_getmem(devc, mp_base, 0) != 0x00 || smw_getmem(devc, mp_base, 1) != 0xff)
+ {
+ DDB(printk("SM Wave: No microcontroller RAM detected (%02x, %02x)\n", smw_getmem(devc, mp_base, 0), smw_getmem(devc, mp_base, 1)));
+ return 0; /* No RAM */
+ }
+ /*
+ * There is RAM so assume it's really a SM Wave
+ */
+
+ devc->model = MDL_SMW;
+ smw_mixer_init(devc);
+
+#ifdef MODULE
+ if (!smw_ucode)
+ {
+ smw_ucodeLen = mod_firmware_load("/etc/sound/midi0001.bin", (void *) &smw_ucode);
+ smw_free = smw_ucode;
+ }
+#endif
+ if (smw_ucodeLen > 0)
+ {
+ if (smw_ucodeLen != 8192)
+ {
+ printk(KERN_ERR "SM Wave: Invalid microcode (MIDI0001.BIN) length\n");
+ return 1;
+ }
+ /*
+ * Download microcode
+ */
+
+ for (i = 0; i < 8192; i++)
+ smw_putmem(devc, mp_base, i, smw_ucode[i]);
+
+ /*
+ * Verify microcode
+ */
+
+ for (i = 0; i < 8192; i++)
+ if (smw_getmem(devc, mp_base, i) != smw_ucode[i])
+ {
+ printk(KERN_ERR "SM Wave: Microcode verification failed\n");
+ return 0;
+ }
+ }
+ control = 0;
+#ifdef SMW_SCSI_IRQ
+ /*
+ * Set the SCSI interrupt (IRQ2/9, IRQ3 or IRQ10). The SCSI interrupt
+ * is disabled by default.
+ *
+ * FIXME - make this a module option
+ *
+ * BTW the Zilog 5380 SCSI controller is located at MPU base + 0x10.
+ */
+ {
+ static unsigned char scsi_irq_bits[] = {
+ 0, 0, 3, 1, 0, 0, 0, 0, 0, 3, 2, 0, 0, 0, 0, 0
+ };
+ control |= scsi_irq_bits[SMW_SCSI_IRQ] << 6;
+ }
+#endif
+
+#ifdef SMW_OPL4_ENABLE
+ /*
+ * Make the OPL4 chip visible on the PC bus at 0x380.
+ *
+ * There is no need to enable this feature since this driver
+ * doesn't support OPL4 yet. Also there is no RAM in SM Wave so
+ * enabling OPL4 is pretty useless.
+ */
+ control |= 0x10; /* Uses IRQ12 if bit 0x20 == 0 */
+ /* control |= 0x20; Uncomment this if you want to use IRQ7 */
+#endif
+ outb((control | 0x03), mpu_base + 7); /* xxxxxx11 restarts */
+ hw_config->name = "SoundMan Wave";
+ return 1;
+}
+
+static int init_Jazz16_midi(sb_devc * devc, struct address_info *hw_config)
+{
+ int mpu_base = hw_config->io_base;
+ int sb_base = devc->base;
+ int irq = hw_config->irq;
+
+ unsigned char bits = 0;
+ unsigned long flags;
+
+ if (irq < 0)
+ irq *= -1;
+
+ if (irq < 1 || irq > 15 ||
+ jazz_irq_bits[irq] == 0)
+ {
+ printk(KERN_ERR "Jazz16: Invalid MIDI interrupt (IRQ%d)\n", irq);
+ return 0;
+ }
+ switch (sb_base)
+ {
+ case 0x220:
+ bits = 1;
+ break;
+ case 0x240:
+ bits = 2;
+ break;
+ case 0x260:
+ bits = 3;
+ break;
+ default:
+ return 0;
+ }
+ bits = jazz16_bits = bits << 5;
+ switch (mpu_base)
+ {
+ case 0x310:
+ bits |= 1;
+ break;
+ case 0x320:
+ bits |= 2;
+ break;
+ case 0x330:
+ bits |= 3;
+ break;
+ default:
+ printk(KERN_ERR "Jazz16: Invalid MIDI I/O port %x\n", mpu_base);
+ return 0;
+ }
+ /*
+ * Magic wake up sequence by writing to 0x201 (aka Joystick port)
+ */
+ spin_lock_irqsave(&jazz16_lock, flags);
+ outb(0xAF, 0x201);
+ outb(0x50, 0x201);
+ outb(bits, 0x201);
+ spin_unlock_irqrestore(&jazz16_lock, flags);
+
+ hw_config->name = "Jazz16";
+ smw_midi_init(devc, hw_config);
+
+ if (!sb_dsp_command(devc, 0xfb))
+ return 0;
+
+ if (!sb_dsp_command(devc, jazz_dma_bits[devc->dma8] |
+ (jazz_dma_bits[devc->dma16] << 4)))
+ return 0;
+
+ if (!sb_dsp_command(devc, jazz_irq_bits[devc->irq] |
+ (jazz_irq_bits[irq] << 4)))
+ return 0;
+
+ return 1;
+}
+
+int probe_sbmpu(struct address_info *hw_config, struct module *owner)
+{
+ sb_devc *devc = last_devc;
+ int ret;
+
+ if (last_devc == NULL)
+ return 0;
+
+ last_devc = NULL;
+
+ if (hw_config->io_base <= 0)
+ {
+ /* The real vibra16 is fine about this, but we have to go
+ wipe up after Cyrix again */
+
+ if(devc->model == MDL_SB16 && devc->minor >= 12)
+ {
+ unsigned char bits = sb_getmixer(devc, 0x84) & ~0x06;
+ sb_setmixer(devc, 0x84, bits | 0x02); /* Disable MPU */
+ }
+ return 0;
+ }
+
+#if defined(CONFIG_SOUND_MPU401)
+ if (devc->model == MDL_ESS)
+ {
+ struct resource *ports;
+ ports = request_region(hw_config->io_base, 2, "mpu401");
+ if (!ports) {
+ printk(KERN_ERR "sbmpu: I/O port conflict (%x)\n", hw_config->io_base);
+ return 0;
+ }
+ if (!ess_midi_init(devc, hw_config)) {
+ release_region(hw_config->io_base, 2);
+ return 0;
+ }
+ hw_config->name = "ESS1xxx MPU";
+ devc->midi_irq_cookie = NULL;
+ if (!probe_mpu401(hw_config, ports)) {
+ release_region(hw_config->io_base, 2);
+ return 0;
+ }
+ attach_mpu401(hw_config, owner);
+ if (last_sb->irq == -hw_config->irq)
+ last_sb->midi_irq_cookie =
+ (void *)(long) hw_config->slots[1];
+ return 1;
+ }
+#endif
+
+ switch (devc->model)
+ {
+ case MDL_SB16:
+ if (hw_config->io_base != 0x300 && hw_config->io_base != 0x330)
+ {
+ printk(KERN_ERR "SB16: Invalid MIDI port %x\n", hw_config->io_base);
+ return 0;
+ }
+ hw_config->name = "Sound Blaster 16";
+ if (hw_config->irq < 3 || hw_config->irq == devc->irq)
+ hw_config->irq = -devc->irq;
+ if (devc->minor > 12) /* What is Vibra's version??? */
+ sb16_set_mpu_port(devc, hw_config);
+ break;
+
+ case MDL_JAZZ:
+ if (hw_config->irq < 3 || hw_config->irq == devc->irq)
+ hw_config->irq = -devc->irq;
+ if (!init_Jazz16_midi(devc, hw_config))
+ return 0;
+ break;
+
+ case MDL_YMPCI:
+ hw_config->name = "Yamaha PCI Legacy";
+ printk("Yamaha PCI legacy UART401 check.\n");
+ break;
+ default:
+ return 0;
+ }
+
+ ret = probe_uart401(hw_config, owner);
+ if (ret)
+ last_sb->midi_irq_cookie=midi_devs[hw_config->slots[4]]->devc;
+ return ret;
+}
+
+void unload_sbmpu(struct address_info *hw_config)
+{
+#if defined(CONFIG_SOUND_MPU401)
+ if (!strcmp (hw_config->name, "ESS1xxx MPU")) {
+ unload_mpu401(hw_config);
+ return;
+ }
+#endif
+ unload_uart401(hw_config);
+}
+
+EXPORT_SYMBOL(sb_dsp_init);
+EXPORT_SYMBOL(sb_dsp_detect);
+EXPORT_SYMBOL(sb_dsp_unload);
+EXPORT_SYMBOL(sb_be_quiet);
+EXPORT_SYMBOL(probe_sbmpu);
+EXPORT_SYMBOL(unload_sbmpu);
+EXPORT_SYMBOL(smw_free);
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c
new file mode 100644
index 0000000..180e95c
--- /dev/null
+++ b/sound/oss/sb_ess.c
@@ -0,0 +1,1832 @@
+#undef FKS_LOGGING
+#undef FKS_TEST
+
+/*
+ * tabs should be 4 spaces, in vi(m): set tabstop=4
+ *
+ * TODO: consistency speed calculations!!
+ * cleanup!
+ * ????: Did I break MIDI support?
+ *
+ * History:
+ *
+ * Rolf Fokkens (Dec 20 1998): ES188x recording level support on a per
+ * fokkensr@vertis.nl input basis.
+ * (Dec 24 1998): Recognition of ES1788, ES1887, ES1888,
+ * ES1868, ES1869 and ES1878. Could be used for
+ * specific handling in the future. All except
+ * ES1887 and ES1888 and ES688 are handled like
+ * ES1688.
+ * (Dec 27 1998): RECLEV for all (?) ES1688+ chips. ES188x now
+ * have the "Dec 20" support + RECLEV
+ * (Jan 2 1999): Preparation for Full Duplex. This means
+ * Audio 2 is now used for playback when dma16
+ * is specified. The next step would be to use
+ * Audio 1 and Audio 2 at the same time.
+ * (Jan 9 1999): Put all ESS stuff into sb_ess.[ch], this
+ * includes both the ESS stuff that has been in
+ * sb_*[ch] before I touched it and the ESS support
+ * I added later
+ * (Jan 23 1999): Full Duplex seems to work. I wrote a small
+ * test proggy which works OK. Haven't found
+ * any applications to test it though. So why did
+ * I bother to create it anyway?? :) Just for
+ * fun.
+ * (May 2 1999): I tried to be too smart by "introducing"
+ * ess_calc_best_speed (). The idea was that two
+ * dividers could be used to setup a samplerate,
+ * ess_calc_best_speed () would choose the best.
+ * This works for playback, but results in
+ * recording problems for high samplerates. I
+ * fixed this by removing ess_calc_best_speed ()
+ * and just doing what the documentation says.
+ * Andy Sloane (Jun 4 1999): Stole some code from ALSA to fix the playback
+ * andy@guildsoftware.com speed on ES1869, ES1879, ES1887, and ES1888.
+ * 1879's were previously ignored by this driver;
+ * added (untested) support for those.
+ * Cvetan Ivanov (Oct 27 1999): Fixed ess_dsp_init to call ess_set_dma_hw for
+ * zezo@inet.bg _ALL_ ESS models, not only ES1887
+ *
+ * This files contains ESS chip specifics. It's based on the existing ESS
+ * handling as it resided in sb_common.c, sb_mixer.c and sb_audio.c. This
+ * file adds features like:
+ * - Chip Identification (as shown in /proc/sound)
+ * - RECLEV support for ES1688 and later
+ * - 6 bits playback level support chips later than ES1688
+ * - Recording level support on a per-device basis for ES1887
+ * - Full-Duplex for ES1887
+ *
+ * Full duplex is enabled by specifying dma16. While the normal dma must
+ * be one of 0, 1 or 3, dma16 can be one of 0, 1, 3 or 5. DMA 5 is a 16 bit
+ * DMA channel, while the others are 8 bit..
+ *
+ * ESS detection isn't full proof (yet). If it fails an additional module
+ * parameter esstype can be specified to be one of the following:
+ * -1, 0, 688, 1688, 1868, 1869, 1788, 1887, 1888
+ * -1 means: mimic 2.0 behaviour,
+ * 0 means: auto detect.
+ * others: explicitly specify chip
+ * -1 is default, cause auto detect still doesn't work.
+ */
+
+/*
+ * About the documentation
+ *
+ * I don't know if the chips all are OK, but the documentation is buggy. 'cause
+ * I don't have all the cips myself, there's a lot I cannot verify. I'll try to
+ * keep track of my latest insights about his here. If you have additional info,
+ * please enlighten me (fokkensr@vertis.nl)!
+ *
+ * I had the impression that ES1688 also has 6 bit master volume control. The
+ * documentation about ES1888 (rev C, october '95) claims that ES1888 has
+ * the following features ES1688 doesn't have:
+ * - 6 bit master volume
+ * - Full Duplex
+ * So ES1688 apparently doesn't have 6 bit master volume control, but the
+ * ES1688 does have RECLEV control. Makes me wonder: does ES688 have it too?
+ * Without RECLEV ES688 won't be much fun I guess.
+ *
+ * From the ES1888 (rev C, october '95) documentation I got the impression
+ * that registers 0x68 to 0x6e don't exist which means: no recording volume
+ * controls. To my surprise the ES888 documentation (1/14/96) claims that
+ * ES888 does have these record mixer registers, but that ES1888 doesn't have
+ * 0x69 and 0x6b. So the rest should be there.
+ *
+ * I'm trying to get ES1887 Full Duplex. Audio 2 is playback only, while Audio 2
+ * is both record and playback. I think I should use Audio 2 for all playback.
+ *
+ * The documentation is an adventure: it's close but not fully accurate. I
+ * found out that after a reset some registers are *NOT* reset, though the
+ * docs say the would be. Interesting ones are 0x7f, 0x7d and 0x7a. They are
+ * related to the Audio 2 channel. I also was surprised about the consequences
+ * of writing 0x00 to 0x7f (which should be done by reset): The ES1887 moves
+ * into ES1888 mode. This means that it claims IRQ 11, which happens to be my
+ * ISDN adapter. Needless to say it no longer worked. I now understand why
+ * after rebooting 0x7f already was 0x05, the value of my choice: the BIOS
+ * did it.
+ *
+ * Oh, and this is another trap: in ES1887 docs mixer register 0x70 is
+ * described as if it's exactly the same as register 0xa1. This is *NOT* true.
+ * The description of 0x70 in ES1869 docs is accurate however.
+ * Well, the assumption about ES1869 was wrong: register 0x70 is very much
+ * like register 0xa1, except that bit 7 is always 1, whatever you want
+ * it to be.
+ *
+ * When using audio 2 mixer register 0x72 seems te be meaningless. Only 0xa2
+ * has effect.
+ *
+ * Software reset not being able to reset all registers is great! Especially
+ * the fact that register 0x78 isn't reset is great when you wanna change back
+ * to single dma operation (simplex): audio 2 is still operational, and uses
+ * the same dma as audio 1: your ess changes into a funny echo machine.
+ *
+ * Received the news that ES1688 is detected as a ES1788. Did some thinking:
+ * the ES1887 detection scheme suggests in step 2 to try if bit 3 of register
+ * 0x64 can be changed. This is inaccurate, first I inverted the * check: "If
+ * can be modified, it's a 1688", which lead to a correct detection
+ * of my ES1887. It resulted however in bad detection of 1688 (reported by mail)
+ * and 1868 (if no PnP detection first): they result in a 1788 being detected.
+ * I don't have docs on 1688, but I do have docs on 1868: The documentation is
+ * probably inaccurate in the fact that I should check bit 2, not bit 3. This
+ * is what I do now.
+ */
+
+/*
+ * About recognition of ESS chips
+ *
+ * The distinction of ES688, ES1688, ES1788, ES1887 and ES1888 is described in
+ * a (preliminary ??) datasheet on ES1887. Its aim is to identify ES1887, but
+ * during detection the text claims that "this chip may be ..." when a step
+ * fails. This scheme is used to distinct between the above chips.
+ * It appears however that some PnP chips like ES1868 are recognized as ES1788
+ * by the ES1887 detection scheme. These PnP chips can be detected in another
+ * way however: ES1868, ES1869 and ES1878 can be recognized (full proof I think)
+ * by repeatedly reading mixer register 0x40. This is done by ess_identify in
+ * sb_common.c.
+ * This results in the following detection steps:
+ * - distinct between ES688 and ES1688+ (as always done in this driver)
+ * if ES688 we're ready
+ * - try to detect ES1868, ES1869 or ES1878
+ * if successful we're ready
+ * - try to detect ES1888, ES1887 or ES1788
+ * if successful we're ready
+ * - Dunno. Must be 1688. Will do in general
+ *
+ * About RECLEV support:
+ *
+ * The existing ES1688 support didn't take care of the ES1688+ recording
+ * levels very well. Whenever a device was selected (recmask) for recording
+ * its recording level was loud, and it couldn't be changed. The fact that
+ * internal register 0xb4 could take care of RECLEV, didn't work meaning until
+ * its value was restored every time the chip was reset; this reset the
+ * value of 0xb4 too. I guess that's what 4front also had (have?) trouble with.
+ *
+ * About ES1887 support:
+ *
+ * The ES1887 has separate registers to control the recording levels, for all
+ * inputs. The ES1887 specific software makes these levels the same as their
+ * corresponding playback levels, unless recmask says they aren't recorded. In
+ * the latter case the recording volumes are 0.
+ * Now recording levels of inputs can be controlled, by changing the playback
+ * levels. Futhermore several devices can be recorded together (which is not
+ * possible with the ES1688).
+ * Besides the separate recording level control for each input, the common
+ * recording level can also be controlled by RECLEV as described above.
+ *
+ * Not only ES1887 have this recording mixer. I know the following from the
+ * documentation:
+ * ES688 no
+ * ES1688 no
+ * ES1868 no
+ * ES1869 yes
+ * ES1878 no
+ * ES1879 yes
+ * ES1888 no/yes Contradicting documentation; most recent: yes
+ * ES1946 yes This is a PCI chip; not handled by this driver
+ */
+
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+
+#include "sound_config.h"
+#include "sb_mixer.h"
+#include "sb.h"
+
+#include "sb_ess.h"
+
+#define ESSTYPE_LIKE20 -1 /* Mimic 2.0 behaviour */
+#define ESSTYPE_DETECT 0 /* Mimic 2.0 behaviour */
+
+#define SUBMDL_ES1788 0x10 /* Subtype ES1788 for specific handling */
+#define SUBMDL_ES1868 0x11 /* Subtype ES1868 for specific handling */
+#define SUBMDL_ES1869 0x12 /* Subtype ES1869 for specific handling */
+#define SUBMDL_ES1878 0x13 /* Subtype ES1878 for specific handling */
+#define SUBMDL_ES1879 0x16 /* ES1879 was initially forgotten */
+#define SUBMDL_ES1887 0x14 /* Subtype ES1887 for specific handling */
+#define SUBMDL_ES1888 0x15 /* Subtype ES1888 for specific handling */
+
+#define SB_CAP_ES18XX_RATE 0x100
+
+#define ES1688_CLOCK1 795444 /* 128 - div */
+#define ES1688_CLOCK2 397722 /* 256 - div */
+#define ES18XX_CLOCK1 793800 /* 128 - div */
+#define ES18XX_CLOCK2 768000 /* 256 - div */
+
+#ifdef FKS_LOGGING
+static void ess_show_mixerregs (sb_devc *devc);
+#endif
+static int ess_read (sb_devc * devc, unsigned char reg);
+static int ess_write (sb_devc * devc, unsigned char reg, unsigned char data);
+static void ess_chgmixer
+ (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val);
+
+/****************************************************************************
+ * *
+ * ESS audio *
+ * *
+ ****************************************************************************/
+
+struct ess_command {short cmd; short data;};
+
+/*
+ * Commands for initializing Audio 1 for input (record)
+ */
+static struct ess_command ess_i08m[] = /* input 8 bit mono */
+ { {0xb7, 0x51}, {0xb7, 0xd0}, {-1, 0} };
+static struct ess_command ess_i16m[] = /* input 16 bit mono */
+ { {0xb7, 0x71}, {0xb7, 0xf4}, {-1, 0} };
+static struct ess_command ess_i08s[] = /* input 8 bit stereo */
+ { {0xb7, 0x51}, {0xb7, 0x98}, {-1, 0} };
+static struct ess_command ess_i16s[] = /* input 16 bit stereo */
+ { {0xb7, 0x71}, {0xb7, 0xbc}, {-1, 0} };
+
+static struct ess_command *ess_inp_cmds[] =
+ { ess_i08m, ess_i16m, ess_i08s, ess_i16s };
+
+
+/*
+ * Commands for initializing Audio 1 for output (playback)
+ */
+static struct ess_command ess_o08m[] = /* output 8 bit mono */
+ { {0xb6, 0x80}, {0xb7, 0x51}, {0xb7, 0xd0}, {-1, 0} };
+static struct ess_command ess_o16m[] = /* output 16 bit mono */
+ { {0xb6, 0x00}, {0xb7, 0x71}, {0xb7, 0xf4}, {-1, 0} };
+static struct ess_command ess_o08s[] = /* output 8 bit stereo */
+ { {0xb6, 0x80}, {0xb7, 0x51}, {0xb7, 0x98}, {-1, 0} };
+static struct ess_command ess_o16s[] = /* output 16 bit stereo */
+ { {0xb6, 0x00}, {0xb7, 0x71}, {0xb7, 0xbc}, {-1, 0} };
+
+static struct ess_command *ess_out_cmds[] =
+ { ess_o08m, ess_o16m, ess_o08s, ess_o16s };
+
+static void ess_exec_commands
+ (sb_devc *devc, struct ess_command *cmdtab[])
+{
+ struct ess_command *cmd;
+
+ cmd = cmdtab [ ((devc->channels != 1) << 1) + (devc->bits != AFMT_U8) ];
+
+ while (cmd->cmd != -1) {
+ ess_write (devc, cmd->cmd, cmd->data);
+ cmd++;
+ }
+}
+
+static void ess_change
+ (sb_devc *devc, unsigned int reg, unsigned int mask, unsigned int val)
+{
+ int value;
+
+ value = ess_read (devc, reg);
+ value = (value & ~mask) | (val & mask);
+ ess_write (devc, reg, value);
+}
+
+static void ess_set_output_parms
+ (int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (devc->duplex) {
+ devc->trg_buf_16 = buf;
+ devc->trg_bytes_16 = nr_bytes;
+ devc->trg_intrflag_16 = intrflag;
+ devc->irq_mode_16 = IMODE_OUTPUT;
+ } else {
+ devc->trg_buf = buf;
+ devc->trg_bytes = nr_bytes;
+ devc->trg_intrflag = intrflag;
+ devc->irq_mode = IMODE_OUTPUT;
+ }
+}
+
+static void ess_set_input_parms
+ (int dev, unsigned long buf, int count, int intrflag)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ devc->trg_buf = buf;
+ devc->trg_bytes = count;
+ devc->trg_intrflag = intrflag;
+ devc->irq_mode = IMODE_INPUT;
+}
+
+static int ess_calc_div (int clock, int revert, int *speedp, int *diffp)
+{
+ int divider;
+ int speed, diff;
+ int retval;
+
+ speed = *speedp;
+ divider = (clock + speed / 2) / speed;
+ retval = revert - divider;
+ if (retval > revert - 1) {
+ retval = revert - 1;
+ divider = revert - retval;
+ }
+ /* This line is suggested. Must be wrong I think
+ *speedp = (clock + divider / 2) / divider;
+ So I chose the next one */
+
+ *speedp = clock / divider;
+ diff = speed - *speedp;
+ if (diff < 0) diff =-diff;
+ *diffp = diff;
+
+ return retval;
+}
+
+static int ess_calc_best_speed
+ (int clock1, int rev1, int clock2, int rev2, int *divp, int *speedp)
+{
+ int speed1 = *speedp, speed2 = *speedp;
+ int div1, div2;
+ int diff1, diff2;
+ int retval;
+
+ div1 = ess_calc_div (clock1, rev1, &speed1, &diff1);
+ div2 = ess_calc_div (clock2, rev2, &speed2, &diff2);
+
+ if (diff1 < diff2) {
+ *divp = div1;
+ *speedp = speed1;
+ retval = 1;
+ } else {
+ /* *divp = div2; */
+ *divp = 0x80 | div2;
+ *speedp = speed2;
+ retval = 2;
+ }
+
+ return retval;
+}
+
+/*
+ * Depending on the audiochannel ESS devices can
+ * have different clock settings. These are made consistent for duplex
+ * however.
+ * callers of ess_speed only do an audionum suggestion, which means
+ * input suggests 1, output suggests 2. This suggestion is only true
+ * however when doing duplex.
+ */
+static void ess_common_speed (sb_devc *devc, int *speedp, int *divp)
+{
+ int diff = 0, div;
+
+ if (devc->duplex) {
+ /*
+ * The 0x80 is important for the first audio channel
+ */
+ if (devc->submodel == SUBMDL_ES1888) {
+ div = 0x80 | ess_calc_div (795500, 256, speedp, &diff);
+ } else {
+ div = 0x80 | ess_calc_div (795500, 128, speedp, &diff);
+ }
+ } else if(devc->caps & SB_CAP_ES18XX_RATE) {
+ if (devc->submodel == SUBMDL_ES1888) {
+ ess_calc_best_speed(397700, 128, 795500, 256,
+ &div, speedp);
+ } else {
+ ess_calc_best_speed(ES18XX_CLOCK1, 128, ES18XX_CLOCK2, 256,
+ &div, speedp);
+ }
+ } else {
+ if (*speedp > 22000) {
+ div = 0x80 | ess_calc_div (ES1688_CLOCK1, 256, speedp, &diff);
+ } else {
+ div = 0x00 | ess_calc_div (ES1688_CLOCK2, 128, speedp, &diff);
+ }
+ }
+ *divp = div;
+}
+
+static void ess_speed (sb_devc *devc, int audionum)
+{
+ int speed;
+ int div, div2;
+
+ ess_common_speed (devc, &(devc->speed), &div);
+
+#ifdef FKS_REG_LOGGING
+printk (KERN_INFO "FKS: ess_speed (%d) b speed = %d, div=%x\n", audionum, devc->speed, div);
+#endif
+
+ /* Set filter roll-off to 90% of speed/2 */
+ speed = (devc->speed * 9) / 20;
+
+ div2 = 256 - 7160000 / (speed * 82);
+
+ if (!devc->duplex) audionum = 1;
+
+ if (audionum == 1) {
+ /* Change behaviour of register A1 *
+ sb_chg_mixer(devc, 0x71, 0x20, 0x20)
+ * For ES1869 only??? */
+ ess_write (devc, 0xa1, div);
+ ess_write (devc, 0xa2, div2);
+ } else {
+ ess_setmixer (devc, 0x70, div);
+ /*
+ * FKS: fascinating: 0x72 doesn't seem to work.
+ */
+ ess_write (devc, 0xa2, div2);
+ ess_setmixer (devc, 0x72, div2);
+ }
+}
+
+static int ess_audio_prepare_for_input(int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ ess_speed(devc, 1);
+
+ sb_dsp_command(devc, DSP_CMD_SPKOFF);
+
+ ess_write (devc, 0xb8, 0x0e); /* Auto init DMA mode */
+ ess_change (devc, 0xa8, 0x03, 3 - devc->channels); /* Mono/stereo */
+ ess_write (devc, 0xb9, 2); /* Demand mode (4 bytes/DMA request) */
+
+ ess_exec_commands (devc, ess_inp_cmds);
+
+ ess_change (devc, 0xb1, 0xf0, 0x50);
+ ess_change (devc, 0xb2, 0xf0, 0x50);
+
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int ess_audio_prepare_for_output_audio1 (int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ sb_dsp_reset(devc);
+ ess_speed(devc, 1);
+ ess_write (devc, 0xb8, 4); /* Auto init DMA mode */
+ ess_change (devc, 0xa8, 0x03, 3 - devc->channels); /* Mono/stereo */
+ ess_write (devc, 0xb9, 2); /* Demand mode (4 bytes/request) */
+
+ ess_exec_commands (devc, ess_out_cmds);
+
+ ess_change (devc, 0xb1, 0xf0, 0x50); /* Enable DMA */
+ ess_change (devc, 0xb2, 0xf0, 0x50); /* Enable IRQ */
+
+ sb_dsp_command(devc, DSP_CMD_SPKON); /* There be sound! */
+
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int ess_audio_prepare_for_output_audio2 (int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ unsigned char bits;
+
+/* FKS: qqq
+ sb_dsp_reset(devc);
+*/
+
+ /*
+ * Auto-Initialize:
+ * DMA mode + demand mode (8 bytes/request, yes I want it all!)
+ * But leave 16-bit DMA bit untouched!
+ */
+ ess_chgmixer (devc, 0x78, 0xd0, 0xd0);
+
+ ess_speed(devc, 2);
+
+ /* bits 4:3 on ES1887 represent recording source. Keep them! */
+ bits = ess_getmixer (devc, 0x7a) & 0x18;
+
+ /* Set stereo/mono */
+ if (devc->channels != 1) bits |= 0x02;
+
+ /* Init DACs; UNSIGNED mode for 8 bit; SIGNED mode for 16 bit */
+ if (devc->bits != AFMT_U8) bits |= 0x05; /* 16 bit */
+
+ /* Enable DMA, IRQ will be shared (hopefully)*/
+ bits |= 0x60;
+
+ ess_setmixer (devc, 0x7a, bits);
+
+ ess_mixer_reload (devc, SOUND_MIXER_PCM); /* There be sound! */
+
+ devc->trigger_bits = 0;
+ return 0;
+}
+
+static int ess_audio_prepare_for_output(int dev, int bsize, int bcount)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+#ifdef FKS_REG_LOGGING
+printk(KERN_INFO "ess_audio_prepare_for_output: dma_out=%d,dma_in=%d\n"
+, audio_devs[dev]->dmap_out->dma, audio_devs[dev]->dmap_in->dma);
+#endif
+
+ if (devc->duplex) {
+ return ess_audio_prepare_for_output_audio2 (dev, bsize, bcount);
+ } else {
+ return ess_audio_prepare_for_output_audio1 (dev, bsize, bcount);
+ }
+}
+
+static void ess_audio_halt_xfer(int dev)
+{
+ unsigned long flags;
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ sb_dsp_reset(devc);
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ /*
+ * Audio 2 may still be operational! Creates awful sounds!
+ */
+ if (devc->duplex) ess_chgmixer(devc, 0x78, 0x03, 0x00);
+}
+
+static void ess_audio_start_input
+ (int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ int count = nr_bytes;
+ sb_devc *devc = audio_devs[dev]->devc;
+ short c = -nr_bytes;
+
+ /*
+ * Start a DMA input to the buffer pointed by dmaqtail
+ */
+
+ if (audio_devs[dev]->dmap_in->dma > 3) count >>= 1;
+ count--;
+
+ devc->irq_mode = IMODE_INPUT;
+
+ ess_write (devc, 0xa4, (unsigned char) ((unsigned short) c & 0xff));
+ ess_write (devc, 0xa5, (unsigned char) (((unsigned short) c >> 8) & 0xff));
+
+ ess_change (devc, 0xb8, 0x0f, 0x0f); /* Go */
+ devc->intr_active = 1;
+}
+
+static void ess_audio_output_block_audio1
+ (int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ int count = nr_bytes;
+ sb_devc *devc = audio_devs[dev]->devc;
+ short c = -nr_bytes;
+
+ if (audio_devs[dev]->dmap_out->dma > 3)
+ count >>= 1;
+ count--;
+
+ devc->irq_mode = IMODE_OUTPUT;
+
+ ess_write (devc, 0xa4, (unsigned char) ((unsigned short) c & 0xff));
+ ess_write (devc, 0xa5, (unsigned char) (((unsigned short) c >> 8) & 0xff));
+
+ ess_change (devc, 0xb8, 0x05, 0x05); /* Go */
+ devc->intr_active = 1;
+}
+
+static void ess_audio_output_block_audio2
+ (int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ int count = nr_bytes;
+ sb_devc *devc = audio_devs[dev]->devc;
+ short c = -nr_bytes;
+
+ if (audio_devs[dev]->dmap_out->dma > 3) count >>= 1;
+ count--;
+
+ ess_setmixer (devc, 0x74, (unsigned char) ((unsigned short) c & 0xff));
+ ess_setmixer (devc, 0x76, (unsigned char) (((unsigned short) c >> 8) & 0xff));
+ ess_chgmixer (devc, 0x78, 0x03, 0x03); /* Go */
+
+ devc->irq_mode_16 = IMODE_OUTPUT;
+ devc->intr_active_16 = 1;
+}
+
+static void ess_audio_output_block
+ (int dev, unsigned long buf, int nr_bytes, int intrflag)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (devc->duplex) {
+ ess_audio_output_block_audio2 (dev, buf, nr_bytes, intrflag);
+ } else {
+ ess_audio_output_block_audio1 (dev, buf, nr_bytes, intrflag);
+ }
+}
+
+/*
+ * FKS: the if-statements for both bits and bits_16 are quite alike.
+ * Combine this...
+ */
+static void ess_audio_trigger(int dev, int bits)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ int bits_16 = bits & devc->irq_mode_16;
+ bits &= devc->irq_mode;
+
+ if (!bits && !bits_16) {
+ /* FKS oh oh.... wrong?? for dma 16? */
+ sb_dsp_command(devc, 0xd0); /* Halt DMA */
+ }
+
+ if (bits) {
+ switch (devc->irq_mode)
+ {
+ case IMODE_INPUT:
+ ess_audio_start_input(dev, devc->trg_buf, devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+
+ case IMODE_OUTPUT:
+ ess_audio_output_block(dev, devc->trg_buf, devc->trg_bytes,
+ devc->trg_intrflag);
+ break;
+ }
+ }
+
+ if (bits_16) {
+ switch (devc->irq_mode_16) {
+ case IMODE_INPUT:
+ ess_audio_start_input(dev, devc->trg_buf_16, devc->trg_bytes_16,
+ devc->trg_intrflag_16);
+ break;
+
+ case IMODE_OUTPUT:
+ ess_audio_output_block(dev, devc->trg_buf_16, devc->trg_bytes_16,
+ devc->trg_intrflag_16);
+ break;
+ }
+ }
+
+ devc->trigger_bits = bits | bits_16;
+}
+
+static int ess_audio_set_speed(int dev, int speed)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+ int minspeed, maxspeed, dummydiv;
+
+ if (speed > 0) {
+ minspeed = (devc->duplex ? 6215 : 5000 );
+ maxspeed = (devc->duplex ? 44100 : 48000);
+ if (speed < minspeed) speed = minspeed;
+ if (speed > maxspeed) speed = maxspeed;
+
+ ess_common_speed (devc, &speed, &dummydiv);
+
+ devc->speed = speed;
+ }
+ return devc->speed;
+}
+
+/*
+ * FKS: This is a one-on-one copy of sb1_audio_set_bits
+ */
+static unsigned int ess_audio_set_bits(int dev, unsigned int bits)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (bits != 0) {
+ if (bits == AFMT_U8 || bits == AFMT_S16_LE) {
+ devc->bits = bits;
+ } else {
+ devc->bits = AFMT_U8;
+ }
+ }
+
+ return devc->bits;
+}
+
+/*
+ * FKS: This is a one-on-one copy of sbpro_audio_set_channels
+ * (*) Modified it!!
+ */
+static short ess_audio_set_channels(int dev, short channels)
+{
+ sb_devc *devc = audio_devs[dev]->devc;
+
+ if (channels == 1 || channels == 2) devc->channels = channels;
+
+ return devc->channels;
+}
+
+static struct audio_driver ess_audio_driver = /* ESS ES688/1688 */
+{
+ .owner = THIS_MODULE,
+ .open = sb_audio_open,
+ .close = sb_audio_close,
+ .output_block = ess_set_output_parms,
+ .start_input = ess_set_input_parms,
+ .prepare_for_input = ess_audio_prepare_for_input,
+ .prepare_for_output = ess_audio_prepare_for_output,
+ .halt_io = ess_audio_halt_xfer,
+ .trigger = ess_audio_trigger,
+ .set_speed = ess_audio_set_speed,
+ .set_bits = ess_audio_set_bits,
+ .set_channels = ess_audio_set_channels
+};
+
+/*
+ * ess_audio_init must be called from sb_audio_init
+ */
+struct audio_driver *ess_audio_init
+ (sb_devc *devc, int *audio_flags, int *format_mask)
+{
+ *audio_flags = DMA_AUTOMODE;
+ *format_mask |= AFMT_S16_LE;
+
+ if (devc->duplex) {
+ int tmp_dma;
+ /*
+ * sb_audio_init thinks dma8 is for playback and
+ * dma16 is for record. Not now! So swap them.
+ */
+ tmp_dma = devc->dma16;
+ devc->dma16 = devc->dma8;
+ devc->dma8 = tmp_dma;
+
+ *audio_flags |= DMA_DUPLEX;
+ }
+
+ return &ess_audio_driver;
+}
+
+/****************************************************************************
+ * *
+ * ESS common *
+ * *
+ ****************************************************************************/
+static void ess_handle_channel
+ (char *channel, int dev, int intr_active, unsigned char flag, int irq_mode)
+{
+ if (!intr_active || !flag) return;
+#ifdef FKS_REG_LOGGING
+printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode);
+#endif
+ switch (irq_mode) {
+ case IMODE_OUTPUT:
+ DMAbuf_outputintr (dev, 1);
+ break;
+
+ case IMODE_INPUT:
+ DMAbuf_inputintr (dev);
+ break;
+
+ case IMODE_INIT:
+ break;
+
+ default:;
+ /* printk(KERN_WARN "ESS: Unexpected interrupt\n"); */
+ }
+}
+
+/*
+ * FKS: TODO!!! Finish this!
+ *
+ * I think midi stuff uses uart401, without interrupts.
+ * So IMODE_MIDI isn't a value for devc->irq_mode.
+ */
+void ess_intr (sb_devc *devc)
+{
+ int status;
+ unsigned char src;
+
+ if (devc->submodel == SUBMDL_ES1887) {
+ src = ess_getmixer (devc, 0x7f) >> 4;
+ } else {
+ src = 0xff;
+ }
+
+#ifdef FKS_REG_LOGGING
+printk(KERN_INFO "FKS: sbintr src=%x\n",(int)src);
+#endif
+ ess_handle_channel
+ ( "Audio 1"
+ , devc->dev, devc->intr_active , src & 0x01, devc->irq_mode );
+ ess_handle_channel
+ ( "Audio 2"
+ , devc->dev, devc->intr_active_16, src & 0x02, devc->irq_mode_16);
+ /*
+ * Acknowledge interrupts
+ */
+ if (devc->submodel == SUBMDL_ES1887 && (src & 0x02)) {
+ ess_chgmixer (devc, 0x7a, 0x80, 0x00);
+ }
+
+ if (src & 0x01) {
+ status = inb(DSP_DATA_AVAIL);
+ }
+}
+
+static void ess_extended (sb_devc * devc)
+{
+ /* Enable extended mode */
+
+ sb_dsp_command(devc, 0xc6);
+}
+
+static int ess_write (sb_devc * devc, unsigned char reg, unsigned char data)
+{
+#ifdef FKS_REG_LOGGING
+printk(KERN_INFO "FKS: write reg %x: %x\n", reg, data);
+#endif
+ /* Write a byte to an extended mode register of ES1688 */
+
+ if (!sb_dsp_command(devc, reg))
+ return 0;
+
+ return sb_dsp_command(devc, data);
+}
+
+static int ess_read (sb_devc * devc, unsigned char reg)
+{
+ /* Read a byte from an extended mode register of ES1688 */
+
+ /* Read register command */
+ if (!sb_dsp_command(devc, 0xc0)) return -1;
+
+ if (!sb_dsp_command(devc, reg )) return -1;
+
+ return sb_dsp_get_byte(devc);
+}
+
+int ess_dsp_reset(sb_devc * devc)
+{
+ int loopc;
+
+#ifdef FKS_REG_LOGGING
+printk(KERN_INFO "FKS: ess_dsp_reset 1\n");
+ess_show_mixerregs (devc);
+#endif
+
+ DEB(printk("Entered ess_dsp_reset()\n"));
+
+ outb(3, DSP_RESET); /* Reset FIFO too */
+
+ udelay(10);
+ outb(0, DSP_RESET);
+ udelay(30);
+
+ for (loopc = 0; loopc < 1000 && !(inb(DSP_DATA_AVAIL) & 0x80); loopc++);
+
+ if (inb(DSP_READ) != 0xAA) {
+ DDB(printk("sb: No response to RESET\n"));
+ return 0; /* Sorry */
+ }
+ ess_extended (devc);
+
+ DEB(printk("sb_dsp_reset() OK\n"));
+
+#ifdef FKS_LOGGING
+printk(KERN_INFO "FKS: dsp_reset 2\n");
+ess_show_mixerregs (devc);
+#endif
+
+ return 1;
+}
+
+static int ess_irq_bits (int irq)
+{
+ switch (irq) {
+ case 2:
+ case 9:
+ return 0;
+
+ case 5:
+ return 1;
+
+ case 7:
+ return 2;
+
+ case 10:
+ return 3;
+
+ default:
+ printk(KERN_ERR "ESS1688: Invalid IRQ %d\n", irq);
+ return -1;
+ }
+}
+
+/*
+ * Set IRQ configuration register for all ESS models
+ */
+static int ess_common_set_irq_hw (sb_devc * devc)
+{
+ int irq_bits;
+
+ if ((irq_bits = ess_irq_bits (devc->irq)) == -1) return 0;
+
+ if (!ess_write (devc, 0xb1, 0x50 | (irq_bits << 2))) {
+ printk(KERN_ERR "ES1688: Failed to write to IRQ config register\n");
+ return 0;
+ }
+ return 1;
+}
+
+/*
+ * I wanna use modern ES1887 mixer irq handling. Funny is the
+ * fact that my BIOS wants the same. But suppose someone's BIOS
+ * doesn't do this!
+ * This is independent of duplex. If there's a 1887 this will
+ * prevent it from going into 1888 mode.
+ */
+static void ess_es1887_set_irq_hw (sb_devc * devc)
+{
+ int irq_bits;
+
+ if ((irq_bits = ess_irq_bits (devc->irq)) == -1) return;
+
+ ess_chgmixer (devc, 0x7f, 0x0f, 0x01 | ((irq_bits + 1) << 1));
+}
+
+static int ess_set_irq_hw (sb_devc * devc)
+{
+ if (devc->submodel == SUBMDL_ES1887) ess_es1887_set_irq_hw (devc);
+
+ return ess_common_set_irq_hw (devc);
+}
+
+#ifdef FKS_TEST
+
+/*
+ * FKS_test:
+ * for ES1887: 00, 18, non wr bits: 0001 1000
+ * for ES1868: 00, b8, non wr bits: 1011 1000
+ * for ES1888: 00, f8, non wr bits: 1111 1000
+ * for ES1688: 00, f8, non wr bits: 1111 1000
+ * + ES968
+ */
+
+static void FKS_test (sb_devc * devc)
+{
+ int val1, val2;
+ val1 = ess_getmixer (devc, 0x64);
+ ess_setmixer (devc, 0x64, ~val1);
+ val2 = ess_getmixer (devc, 0x64) ^ ~val1;
+ ess_setmixer (devc, 0x64, val1);
+ val1 ^= ess_getmixer (devc, 0x64);
+printk (KERN_INFO "FKS: FKS_test %02x, %02x\n", (val1 & 0x0ff), (val2 & 0x0ff));
+};
+#endif
+
+static unsigned int ess_identify (sb_devc * devc)
+{
+ unsigned int val;
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ outb(((unsigned char) (0x40 & 0xff)), MIXER_ADDR);
+
+ udelay(20);
+ val = inb(MIXER_DATA) << 8;
+ udelay(20);
+ val |= inb(MIXER_DATA);
+ udelay(20);
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ return val;
+}
+
+/*
+ * ESS technology describes a detection scheme in their docs. It involves
+ * fiddling with the bits in certain mixer registers. ess_probe is supposed
+ * to help.
+ *
+ * FKS: tracing shows ess_probe writes wrong value to 0x64. Bit 3 reads 1, but
+ * should be written 0 only. Check this.
+ */
+static int ess_probe (sb_devc * devc, int reg, int xorval)
+{
+ int val1, val2, val3;
+
+ val1 = ess_getmixer (devc, reg);
+ val2 = val1 ^ xorval;
+ ess_setmixer (devc, reg, val2);
+ val3 = ess_getmixer (devc, reg);
+ ess_setmixer (devc, reg, val1);
+
+ return (val2 == val3);
+}
+
+int ess_init(sb_devc * devc, struct address_info *hw_config)
+{
+ unsigned char cfg;
+ int ess_major = 0, ess_minor = 0;
+ int i;
+ static char name[100], modelname[10];
+
+ /*
+ * Try to detect ESS chips.
+ */
+
+ sb_dsp_command(devc, 0xe7); /* Return identification */
+
+ for (i = 1000; i; i--) {
+ if (inb(DSP_DATA_AVAIL) & 0x80) {
+ if (ess_major == 0) {
+ ess_major = inb(DSP_READ);
+ } else {
+ ess_minor = inb(DSP_READ);
+ break;
+ }
+ }
+ }
+
+ if (ess_major == 0) return 0;
+
+ if (ess_major == 0x48 && (ess_minor & 0xf0) == 0x80) {
+ sprintf(name, "ESS ES488 AudioDrive (rev %d)",
+ ess_minor & 0x0f);
+ hw_config->name = name;
+ devc->model = MDL_SBPRO;
+ return 1;
+ }
+
+ /*
+ * This the detection heuristic of ESS technology, though somewhat
+ * changed to actually make it work.
+ * This results in the following detection steps:
+ * - distinct between ES688 and ES1688+ (as always done in this driver)
+ * if ES688 we're ready
+ * - try to detect ES1868, ES1869 or ES1878 (ess_identify)
+ * if successful we're ready
+ * - try to detect ES1888, ES1887 or ES1788 (aim: detect ES1887)
+ * if successful we're ready
+ * - Dunno. Must be 1688. Will do in general
+ *
+ * This is the most BETA part of the software: Will the detection
+ * always work?
+ */
+ devc->model = MDL_ESS;
+ devc->submodel = ess_minor & 0x0f;
+
+ if (ess_major == 0x68 && (ess_minor & 0xf0) == 0x80) {
+ char *chip = NULL;
+ int submodel = -1;
+
+ switch (devc->sbmo.esstype) {
+ case ESSTYPE_DETECT:
+ case ESSTYPE_LIKE20:
+ break;
+ case 688:
+ submodel = 0x00;
+ break;
+ case 1688:
+ submodel = 0x08;
+ break;
+ case 1868:
+ submodel = SUBMDL_ES1868;
+ break;
+ case 1869:
+ submodel = SUBMDL_ES1869;
+ break;
+ case 1788:
+ submodel = SUBMDL_ES1788;
+ break;
+ case 1878:
+ submodel = SUBMDL_ES1878;
+ break;
+ case 1879:
+ submodel = SUBMDL_ES1879;
+ break;
+ case 1887:
+ submodel = SUBMDL_ES1887;
+ break;
+ case 1888:
+ submodel = SUBMDL_ES1888;
+ break;
+ default:
+ printk (KERN_ERR "Invalid esstype=%d specified\n", devc->sbmo.esstype);
+ return 0;
+ };
+ if (submodel != -1) {
+ devc->submodel = submodel;
+ sprintf (modelname, "ES%d", devc->sbmo.esstype);
+ chip = modelname;
+ };
+ if (chip == NULL && (ess_minor & 0x0f) < 8) {
+ chip = "ES688";
+ };
+#ifdef FKS_TEST
+FKS_test (devc);
+#endif
+ /*
+ * If Nothing detected yet, and we want 2.0 behaviour...
+ * Then let's assume it's ES1688.
+ */
+ if (chip == NULL && devc->sbmo.esstype == ESSTYPE_LIKE20) {
+ chip = "ES1688";
+ };
+
+ if (chip == NULL) {
+ int type;
+
+ type = ess_identify (devc);
+
+ switch (type) {
+ case 0x1868:
+ chip = "ES1868";
+ devc->submodel = SUBMDL_ES1868;
+ break;
+ case 0x1869:
+ chip = "ES1869";
+ devc->submodel = SUBMDL_ES1869;
+ break;
+ case 0x1878:
+ chip = "ES1878";
+ devc->submodel = SUBMDL_ES1878;
+ break;
+ case 0x1879:
+ chip = "ES1879";
+ devc->submodel = SUBMDL_ES1879;
+ break;
+ default:
+ if ((type & 0x00ff) != ((type >> 8) & 0x00ff)) {
+ printk ("ess_init: Unrecognized %04x\n", type);
+ }
+ };
+ };
+#if 0
+ /*
+ * this one failed:
+ * the probing of bit 4 is another thought: from ES1788 and up, all
+ * chips seem to have hardware volume control. Bit 4 is readonly to
+ * check if a hardware volume interrupt has fired.
+ * Cause ES688/ES1688 don't have this feature, bit 4 might be writeable
+ * for these chips.
+ */
+ if (chip == NULL && !ess_probe(devc, 0x64, (1 << 4))) {
+#endif
+ /*
+ * the probing of bit 2 is my idea. The ES1887 docs want me to probe
+ * bit 3. This results in ES1688 being detected as ES1788.
+ * Bit 2 is for "Enable HWV IRQE", but as ES(1)688 chips don't have
+ * HardWare Volume, I think they don't have this IRQE.
+ */
+ if (chip == NULL && ess_probe(devc, 0x64, (1 << 2))) {
+ if (ess_probe (devc, 0x70, 0x7f)) {
+ if (ess_probe (devc, 0x64, (1 << 5))) {
+ chip = "ES1887";
+ devc->submodel = SUBMDL_ES1887;
+ } else {
+ chip = "ES1888";
+ devc->submodel = SUBMDL_ES1888;
+ }
+ } else {
+ chip = "ES1788";
+ devc->submodel = SUBMDL_ES1788;
+ }
+ };
+ if (chip == NULL) {
+ chip = "ES1688";
+ };
+
+ printk ( KERN_INFO "ESS chip %s %s%s\n"
+ , chip
+ , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20
+ ? "detected"
+ : "specified"
+ )
+ , ( devc->sbmo.esstype == ESSTYPE_LIKE20
+ ? " (kernel 2.0 compatible)"
+ : ""
+ )
+ );
+
+ sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f);
+ } else {
+ strcpy(name, "Jazz16");
+ }
+
+ /* AAS: info stolen from ALSA: these boards have different clocks */
+ switch(devc->submodel) {
+/* APPARENTLY NOT 1869 AND 1887
+ case SUBMDL_ES1869:
+ case SUBMDL_ES1887:
+*/
+ case SUBMDL_ES1888:
+ devc->caps |= SB_CAP_ES18XX_RATE;
+ break;
+ }
+
+ hw_config->name = name;
+ /* FKS: sb_dsp_reset to enable extended mode???? */
+ sb_dsp_reset(devc); /* Turn on extended mode */
+
+ /*
+ * Enable joystick and OPL3
+ */
+ cfg = ess_getmixer (devc, 0x40);
+ ess_setmixer (devc, 0x40, cfg | 0x03);
+ if (devc->submodel >= 8) { /* ES1688 */
+ devc->caps |= SB_NO_MIDI; /* ES1688 uses MPU401 MIDI mode */
+ }
+ sb_dsp_reset (devc);
+
+ /*
+ * This is important! If it's not done, the IRQ probe in sb_dsp_init
+ * may fail.
+ */
+ return ess_set_irq_hw (devc);
+}
+
+static int ess_set_dma_hw(sb_devc * devc)
+{
+ unsigned char cfg, dma_bits = 0, dma16_bits;
+ int dma;
+
+#ifdef FKS_LOGGING
+printk(KERN_INFO "ess_set_dma_hw: dma8=%d,dma16=%d,dup=%d\n"
+, devc->dma8, devc->dma16, devc->duplex);
+#endif
+
+ /*
+ * FKS: It seems as if this duplex flag isn't set yet. Check it.
+ */
+ dma = devc->dma8;
+
+ if (dma > 3 || dma < 0 || dma == 2) {
+ dma_bits = 0;
+ printk(KERN_ERR "ESS1688: Invalid DMA8 %d\n", dma);
+ return 0;
+ } else {
+ /* Extended mode DMA enable */
+ cfg = 0x50;
+
+ if (dma == 3) {
+ dma_bits = 3;
+ } else {
+ dma_bits = dma + 1;
+ }
+ }
+
+ if (!ess_write (devc, 0xb2, cfg | (dma_bits << 2))) {
+ printk(KERN_ERR "ESS1688: Failed to write to DMA config register\n");
+ return 0;
+ }
+
+ if (devc->duplex) {
+ dma = devc->dma16;
+ dma16_bits = 0;
+
+ if (dma >= 0) {
+ switch (dma) {
+ case 0:
+ dma_bits = 0x04;
+ break;
+ case 1:
+ dma_bits = 0x05;
+ break;
+ case 3:
+ dma_bits = 0x06;
+ break;
+ case 5:
+ dma_bits = 0x07;
+ dma16_bits = 0x20;
+ break;
+ default:
+ printk(KERN_ERR "ESS1887: Invalid DMA16 %d\n", dma);
+ return 0;
+ };
+ ess_chgmixer (devc, 0x78, 0x20, dma16_bits);
+ ess_chgmixer (devc, 0x7d, 0x07, dma_bits);
+ }
+ }
+ return 1;
+}
+
+/*
+ * This one is called from sb_dsp_init.
+ *
+ * Return values:
+ * 0: Failed
+ * 1: Succeeded or doesn't apply (not SUBMDL_ES1887)
+ */
+int ess_dsp_init (sb_devc *devc, struct address_info *hw_config)
+{
+ /*
+ * Caller also checks this, but anyway
+ */
+ if (devc->model != MDL_ESS) {
+ printk (KERN_INFO "ess_dsp_init for non ESS chip\n");
+ return 1;
+ }
+ /*
+ * This for ES1887 to run Full Duplex. Actually ES1888
+ * is allowed to do so too. I have no idea yet if this
+ * will work for ES1888 however.
+ *
+ * For SB16 having both dma8 and dma16 means enable
+ * Full Duplex. Let's try this for ES1887 too
+ *
+ */
+ if (devc->submodel == SUBMDL_ES1887) {
+ if (hw_config->dma2 != -1) {
+ devc->dma16 = hw_config->dma2;
+ }
+ /*
+ * devc->duplex initialization is put here, cause
+ * ess_set_dma_hw needs it.
+ */
+ if (devc->dma8 != devc->dma16 && devc->dma16 != -1) {
+ devc->duplex = 1;
+ }
+ }
+ if (!ess_set_dma_hw (devc)) {
+ free_irq(devc->irq, devc);
+ return 0;
+ }
+ return 1;
+}
+
+/****************************************************************************
+ * *
+ * ESS mixer *
+ * *
+ ****************************************************************************/
+
+#define ES688_RECORDING_DEVICES \
+ ( SOUND_MASK_LINE | SOUND_MASK_MIC | SOUND_MASK_CD )
+#define ES688_MIXER_DEVICES \
+ ( SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_LINE \
+ | SOUND_MASK_MIC | SOUND_MASK_CD | SOUND_MASK_VOLUME \
+ | SOUND_MASK_LINE2 | SOUND_MASK_SPEAKER )
+
+#define ES1688_RECORDING_DEVICES \
+ ( ES688_RECORDING_DEVICES )
+#define ES1688_MIXER_DEVICES \
+ ( ES688_MIXER_DEVICES | SOUND_MASK_RECLEV )
+
+#define ES1887_RECORDING_DEVICES \
+ ( ES1688_RECORDING_DEVICES | SOUND_MASK_LINE2 | SOUND_MASK_SYNTH)
+#define ES1887_MIXER_DEVICES \
+ ( ES1688_MIXER_DEVICES )
+
+/*
+ * Mixer registers of ES1887
+ *
+ * These registers specifically take care of recording levels. To make the
+ * mapping from playback devices to recording devices every recording
+ * devices = playback device + ES_REC_MIXER_RECDIFF
+ */
+#define ES_REC_MIXER_RECBASE (SOUND_MIXER_LINE3 + 1)
+#define ES_REC_MIXER_RECDIFF (ES_REC_MIXER_RECBASE - SOUND_MIXER_SYNTH)
+
+#define ES_REC_MIXER_RECSYNTH (SOUND_MIXER_SYNTH + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECPCM (SOUND_MIXER_PCM + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECSPEAKER (SOUND_MIXER_SPEAKER + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECLINE (SOUND_MIXER_LINE + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECMIC (SOUND_MIXER_MIC + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECCD (SOUND_MIXER_CD + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECIMIX (SOUND_MIXER_IMIX + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECALTPCM (SOUND_MIXER_ALTPCM + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECRECLEV (SOUND_MIXER_RECLEV + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECIGAIN (SOUND_MIXER_IGAIN + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECOGAIN (SOUND_MIXER_OGAIN + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECLINE1 (SOUND_MIXER_LINE1 + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECLINE2 (SOUND_MIXER_LINE2 + ES_REC_MIXER_RECDIFF)
+#define ES_REC_MIXER_RECLINE3 (SOUND_MIXER_LINE3 + ES_REC_MIXER_RECDIFF)
+
+static mixer_tab es688_mix = {
+MIX_ENT(SOUND_MIXER_VOLUME, 0x32, 7, 4, 0x32, 3, 4),
+MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4),
+MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4),
+MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4),
+MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4),
+MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4),
+MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0)
+};
+
+/*
+ * The ES1688 specifics... hopefully correct...
+ * - 6 bit master volume
+ * I was wrong, ES1888 docs say ES1688 didn't have it.
+ * - RECLEV control
+ * These may apply to ES688 too. I have no idea.
+ */
+static mixer_tab es1688_mix = {
+MIX_ENT(SOUND_MIXER_VOLUME, 0x32, 7, 4, 0x32, 3, 4),
+MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4),
+MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4),
+MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4),
+MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4),
+MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4),
+MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4),
+MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0)
+};
+
+static mixer_tab es1688later_mix = {
+MIX_ENT(SOUND_MIXER_VOLUME, 0x60, 5, 6, 0x62, 5, 6),
+MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4),
+MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4),
+MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4),
+MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4),
+MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4),
+MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4),
+MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0)
+};
+
+/*
+ * This one is for all ESS chips with a record mixer.
+ * It's not used (yet) however
+ */
+static mixer_tab es_rec_mix = {
+MIX_ENT(SOUND_MIXER_VOLUME, 0x60, 5, 6, 0x62, 5, 6),
+MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4),
+MIX_ENT(SOUND_MIXER_PCM, 0x14, 7, 4, 0x14, 3, 4),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4),
+MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4),
+MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4),
+MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4),
+MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4),
+MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECSYNTH, 0x6b, 7, 4, 0x6b, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECSPEAKER, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECLINE, 0x6e, 7, 4, 0x6e, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECMIC, 0x68, 7, 4, 0x68, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECCD, 0x6a, 7, 4, 0x6a, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECIMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECRECLEV, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECIGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECOGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECLINE1, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECLINE2, 0x6c, 7, 4, 0x6c, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECLINE3, 0x00, 0, 0, 0x00, 0, 0)
+};
+
+/*
+ * This one is for ES1887. It's little different from es_rec_mix: it
+ * has 0x7c for PCM playback level. This is because ES1887 uses
+ * Audio 2 for playback.
+ */
+static mixer_tab es1887_mix = {
+MIX_ENT(SOUND_MIXER_VOLUME, 0x60, 5, 6, 0x62, 5, 6),
+MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x36, 7, 4, 0x36, 3, 4),
+MIX_ENT(SOUND_MIXER_PCM, 0x7c, 7, 4, 0x7c, 3, 4),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x3c, 2, 3, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x3e, 7, 4, 0x3e, 3, 4),
+MIX_ENT(SOUND_MIXER_MIC, 0x1a, 7, 4, 0x1a, 3, 4),
+MIX_ENT(SOUND_MIXER_CD, 0x38, 7, 4, 0x38, 3, 4),
+MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0xb4, 7, 4, 0xb4, 3, 4),
+MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE1, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE2, 0x3a, 7, 4, 0x3a, 3, 4),
+MIX_ENT(SOUND_MIXER_LINE3, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECSYNTH, 0x6b, 7, 4, 0x6b, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECSPEAKER, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECLINE, 0x6e, 7, 4, 0x6e, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECMIC, 0x68, 7, 4, 0x68, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECCD, 0x6a, 7, 4, 0x6a, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECIMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECRECLEV, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECIGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECOGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECLINE1, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(ES_REC_MIXER_RECLINE2, 0x6c, 7, 4, 0x6c, 3, 4),
+MIX_ENT(ES_REC_MIXER_RECLINE3, 0x00, 0, 0, 0x00, 0, 0)
+};
+
+static int ess_has_rec_mixer (int submodel)
+{
+ switch (submodel) {
+ case SUBMDL_ES1887:
+ return 1;
+ default:
+ return 0;
+ };
+};
+
+#ifdef FKS_LOGGING
+static int ess_mixer_mon_regs[]
+ = { 0x70, 0x71, 0x72, 0x74, 0x76, 0x78, 0x7a, 0x7c, 0x7d, 0x7f
+ , 0xa1, 0xa2, 0xa4, 0xa5, 0xa8, 0xa9
+ , 0xb1, 0xb2, 0xb4, 0xb5, 0xb6, 0xb7, 0xb9
+ , 0x00};
+
+static void ess_show_mixerregs (sb_devc *devc)
+{
+ int *mp = ess_mixer_mon_regs;
+
+return;
+
+ while (*mp != 0) {
+ printk (KERN_INFO "res (%x)=%x\n", *mp, (int)(ess_getmixer (devc, *mp)));
+ mp++;
+ }
+}
+#endif
+
+void ess_setmixer (sb_devc * devc, unsigned int port, unsigned int value)
+{
+ unsigned long flags;
+
+#ifdef FKS_LOGGING
+printk(KERN_INFO "FKS: write mixer %x: %x\n", port, value);
+#endif
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (port >= 0xa0) {
+ ess_write (devc, port, value);
+ } else {
+ outb(((unsigned char) (port & 0xff)), MIXER_ADDR);
+
+ udelay(20);
+ outb(((unsigned char) (value & 0xff)), MIXER_DATA);
+ udelay(20);
+ };
+ spin_unlock_irqrestore(&devc->lock, flags);
+}
+
+unsigned int ess_getmixer (sb_devc * devc, unsigned int port)
+{
+ unsigned int val;
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock, flags);
+
+ if (port >= 0xa0) {
+ val = ess_read (devc, port);
+ } else {
+ outb(((unsigned char) (port & 0xff)), MIXER_ADDR);
+
+ udelay(20);
+ val = inb(MIXER_DATA);
+ udelay(20);
+ }
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ return val;
+}
+
+static void ess_chgmixer
+ (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val)
+{
+ int value;
+
+ value = ess_getmixer (devc, reg);
+ value = (value & ~mask) | (val & mask);
+ ess_setmixer (devc, reg, value);
+}
+
+/*
+ * ess_mixer_init must be called from sb_mixer_init
+ */
+void ess_mixer_init (sb_devc * devc)
+{
+ devc->mixer_caps = SOUND_CAP_EXCL_INPUT;
+
+ /*
+ * Take care of ES1887 specifics...
+ */
+ switch (devc->submodel) {
+ case SUBMDL_ES1887:
+ devc->supported_devices = ES1887_MIXER_DEVICES;
+ devc->supported_rec_devices = ES1887_RECORDING_DEVICES;
+#ifdef FKS_LOGGING
+printk (KERN_INFO "FKS: ess_mixer_init dup = %d\n", devc->duplex);
+#endif
+ if (devc->duplex) {
+ devc->iomap = &es1887_mix;
+ devc->iomap_sz = ARRAY_SIZE(es1887_mix);
+ } else {
+ devc->iomap = &es_rec_mix;
+ devc->iomap_sz = ARRAY_SIZE(es_rec_mix);
+ }
+ break;
+ default:
+ if (devc->submodel < 8) {
+ devc->supported_devices = ES688_MIXER_DEVICES;
+ devc->supported_rec_devices = ES688_RECORDING_DEVICES;
+ devc->iomap = &es688_mix;
+ devc->iomap_sz = ARRAY_SIZE(es688_mix);
+ } else {
+ /*
+ * es1688 has 4 bits master vol.
+ * later chips have 6 bits (?)
+ */
+ devc->supported_devices = ES1688_MIXER_DEVICES;
+ devc->supported_rec_devices = ES1688_RECORDING_DEVICES;
+ if (devc->submodel < 0x10) {
+ devc->iomap = &es1688_mix;
+ devc->iomap_sz = ARRAY_SIZE(es688_mix);
+ } else {
+ devc->iomap = &es1688later_mix;
+ devc->iomap_sz = ARRAY_SIZE(es1688later_mix);
+ }
+ }
+ }
+}
+
+/*
+ * Changing playback levels at an ESS chip with record mixer means having to
+ * take care of recording levels of recorded inputs (devc->recmask) too!
+ */
+int ess_mixer_set(sb_devc *devc, int dev, int left, int right)
+{
+ if (ess_has_rec_mixer (devc->submodel) && (devc->recmask & (1 << dev))) {
+ sb_common_mixer_set (devc, dev + ES_REC_MIXER_RECDIFF, left, right);
+ }
+ return sb_common_mixer_set (devc, dev, left, right);
+}
+
+/*
+ * After a sb_dsp_reset extended register 0xb4 (RECLEV) is reset too. After
+ * sb_dsp_reset RECLEV has to be restored. This is where ess_mixer_reload
+ * helps.
+ */
+void ess_mixer_reload (sb_devc *devc, int dev)
+{
+ int left, right, value;
+
+ value = devc->levels[dev];
+ left = value & 0x000000ff;
+ right = (value & 0x0000ff00) >> 8;
+
+ sb_common_mixer_set(devc, dev, left, right);
+}
+
+static int es_rec_set_recmask(sb_devc * devc, int mask)
+{
+ int i, i_mask, cur_mask, diff_mask;
+ int value, left, right;
+
+#ifdef FKS_LOGGING
+printk (KERN_INFO "FKS: es_rec_set_recmask mask = %x\n", mask);
+#endif
+ /*
+ * Changing the recmask on an ESS chip with recording mixer means:
+ * (1) Find the differences
+ * (2) For "turned-on" inputs: make the recording level the playback level
+ * (3) For "turned-off" inputs: make the recording level zero
+ */
+ cur_mask = devc->recmask;
+ diff_mask = (cur_mask ^ mask);
+
+ for (i = 0; i < 32; i++) {
+ i_mask = (1 << i);
+ if (diff_mask & i_mask) { /* Difference? (1) */
+ if (mask & i_mask) { /* Turn it on (2) */
+ value = devc->levels[i];
+ left = value & 0x000000ff;
+ right = (value & 0x0000ff00) >> 8;
+ } else { /* Turn it off (3) */
+ left = 0;
+ left = 0;
+ right = 0;
+ }
+ sb_common_mixer_set(devc, i + ES_REC_MIXER_RECDIFF, left, right);
+ }
+ }
+ return mask;
+}
+
+int ess_set_recmask(sb_devc * devc, int *mask)
+{
+ /* This applies to ESS chips with record mixers only! */
+
+ if (ess_has_rec_mixer (devc->submodel)) {
+ *mask = es_rec_set_recmask (devc, *mask);
+ return 1; /* Applied */
+ } else {
+ return 0; /* Not applied */
+ }
+}
+
+/*
+ * ess_mixer_reset must be called from sb_mixer_reset
+ */
+int ess_mixer_reset (sb_devc * devc)
+{
+ /*
+ * Separate actions for ESS chips with a record mixer:
+ */
+ if (ess_has_rec_mixer (devc->submodel)) {
+ switch (devc->submodel) {
+ case SUBMDL_ES1887:
+ /*
+ * Separate actions for ES1887:
+ * Change registers 7a and 1c to make the record mixer the
+ * actual recording source.
+ */
+ ess_chgmixer(devc, 0x7a, 0x18, 0x08);
+ ess_chgmixer(devc, 0x1c, 0x07, 0x07);
+ break;
+ };
+ /*
+ * Call set_recmask for proper initialization
+ */
+ devc->recmask = devc->supported_rec_devices;
+ es_rec_set_recmask(devc, 0);
+ devc->recmask = 0;
+
+ return 1; /* We took care of recmask. */
+ } else {
+ return 0; /* We didn't take care; caller do it */
+ }
+}
+
+/****************************************************************************
+ * *
+ * ESS midi *
+ * *
+ ****************************************************************************/
+
+/*
+ * FKS: IRQ may be shared. Hm. And if so? Then What?
+ */
+int ess_midi_init(sb_devc * devc, struct address_info *hw_config)
+{
+ unsigned char cfg, tmp;
+
+ cfg = ess_getmixer (devc, 0x40) & 0x03;
+
+ if (devc->submodel < 8) {
+ ess_setmixer (devc, 0x40, cfg | 0x03); /* Enable OPL3 & joystick */
+ return 0; /* ES688 doesn't support MPU401 mode */
+ }
+ tmp = (hw_config->io_base & 0x0f0) >> 4;
+
+ if (tmp > 3) {
+ ess_setmixer (devc, 0x40, cfg);
+ return 0;
+ }
+ cfg |= tmp << 3;
+
+ tmp = 1; /* MPU enabled without interrupts */
+
+ /* May be shared: if so the value is -ve */
+
+ switch (abs(hw_config->irq)) {
+ case 9:
+ tmp = 0x4;
+ break;
+ case 5:
+ tmp = 0x5;
+ break;
+ case 7:
+ tmp = 0x6;
+ break;
+ case 10:
+ tmp = 0x7;
+ break;
+ default:
+ return 0;
+ }
+
+ cfg |= tmp << 5;
+ ess_setmixer (devc, 0x40, cfg | 0x03);
+
+ return 1;
+}
+
diff --git a/sound/oss/sb_ess.h b/sound/oss/sb_ess.h
new file mode 100644
index 0000000..38aa072
--- /dev/null
+++ b/sound/oss/sb_ess.h
@@ -0,0 +1,34 @@
+/*
+ * Created: 9-Jan-1999 Rolf Fokkens
+ */
+
+extern void ess_intr
+ (sb_devc *devc);
+extern int ess_dsp_init
+ (sb_devc *devc, struct address_info *hw_config);
+
+extern struct audio_driver *ess_audio_init
+ (sb_devc *devc, int *audio_flags, int *format_mask);
+extern int ess_midi_init
+ (sb_devc *devc, struct address_info *hw_config);
+extern void ess_mixer_init
+ (sb_devc *devc);
+
+extern int ess_init
+ (sb_devc *devc, struct address_info *hw_config);
+extern int ess_dsp_reset
+ (sb_devc *devc);
+
+extern void ess_setmixer
+ (sb_devc *devc, unsigned int port, unsigned int value);
+extern unsigned int ess_getmixer
+ (sb_devc *devc, unsigned int port);
+extern int ess_mixer_set
+ (sb_devc *devc, int dev, int left, int right);
+extern int ess_mixer_reset
+ (sb_devc *devc);
+extern void ess_mixer_reload
+ (sb_devc * devc, int dev);
+extern int ess_set_recmask
+ (sb_devc *devc, int *mask);
+
diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c
new file mode 100644
index 0000000..8b79670
--- /dev/null
+++ b/sound/oss/sb_midi.c
@@ -0,0 +1,205 @@
+/*
+ * sound/oss/sb_midi.c
+ *
+ * The low level driver for the Sound Blaster DS chips.
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+
+#include <linux/spinlock.h>
+
+#include "sound_config.h"
+
+#include "sb.h"
+#undef SB_TEST_IRQ
+
+/*
+ * The DSP channel can be used either for input or output. Variable
+ * 'sb_irq_mode' will be set when the program calls read or write first time
+ * after open. Current version doesn't support mode changes without closing
+ * and reopening the device. Support for this feature may be implemented in a
+ * future version of this driver.
+ */
+
+
+static int sb_midi_open(int dev, int mode,
+ void (*input) (int dev, unsigned char data),
+ void (*output) (int dev)
+)
+{
+ sb_devc *devc = midi_devs[dev]->devc;
+ unsigned long flags;
+
+ if (devc == NULL)
+ return -ENXIO;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ if (devc->opened)
+ {
+ spin_unlock_irqrestore(&devc->lock, flags);
+ return -EBUSY;
+ }
+ devc->opened = 1;
+ spin_unlock_irqrestore(&devc->lock, flags);
+
+ devc->irq_mode = IMODE_MIDI;
+ devc->midi_broken = 0;
+
+ sb_dsp_reset(devc);
+
+ if (!sb_dsp_command(devc, 0x35)) /* Start MIDI UART mode */
+ {
+ devc->opened = 0;
+ return -EIO;
+ }
+ devc->intr_active = 1;
+
+ if (mode & OPEN_READ)
+ {
+ devc->input_opened = 1;
+ devc->midi_input_intr = input;
+ }
+ return 0;
+}
+
+static void sb_midi_close(int dev)
+{
+ sb_devc *devc = midi_devs[dev]->devc;
+ unsigned long flags;
+
+ if (devc == NULL)
+ return;
+
+ spin_lock_irqsave(&devc->lock, flags);
+ sb_dsp_reset(devc);
+ devc->intr_active = 0;
+ devc->input_opened = 0;
+ devc->opened = 0;
+ spin_unlock_irqrestore(&devc->lock, flags);
+}
+
+static int sb_midi_out(int dev, unsigned char midi_byte)
+{
+ sb_devc *devc = midi_devs[dev]->devc;
+
+ if (devc == NULL)
+ return 1;
+
+ if (devc->midi_broken)
+ return 1;
+
+ if (!sb_dsp_command(devc, midi_byte))
+ {
+ devc->midi_broken = 1;
+ return 1;
+ }
+ return 1;
+}
+
+static int sb_midi_start_read(int dev)
+{
+ return 0;
+}
+
+static int sb_midi_end_read(int dev)
+{
+ sb_devc *devc = midi_devs[dev]->devc;
+
+ if (devc == NULL)
+ return -ENXIO;
+
+ sb_dsp_reset(devc);
+ devc->intr_active = 0;
+ return 0;
+}
+
+static int sb_midi_ioctl(int dev, unsigned cmd, void __user *arg)
+{
+ return -EINVAL;
+}
+
+void sb_midi_interrupt(sb_devc * devc)
+{
+ unsigned long flags;
+ unsigned char data;
+
+ if (devc == NULL)
+ return;
+
+ spin_lock_irqsave(&devc->lock, flags);
+
+ data = inb(DSP_READ);
+ if (devc->input_opened)
+ devc->midi_input_intr(devc->my_mididev, data);
+
+ spin_unlock_irqrestore(&devc->lock, flags);
+}
+
+#define MIDI_SYNTH_NAME "Sound Blaster Midi"
+#define MIDI_SYNTH_CAPS 0
+#include "midi_synth.h"
+
+static struct midi_operations sb_midi_operations =
+{
+ .owner = THIS_MODULE,
+ .info = {"Sound Blaster", 0, 0, SNDCARD_SB},
+ .converter = &std_midi_synth,
+ .in_info = {0},
+ .open = sb_midi_open,
+ .close = sb_midi_close,
+ .ioctl = sb_midi_ioctl,
+ .outputc = sb_midi_out,
+ .start_read = sb_midi_start_read,
+ .end_read = sb_midi_end_read,
+};
+
+void sb_dsp_midi_init(sb_devc * devc, struct module *owner)
+{
+ int dev;
+
+ if (devc->model < 2) /* No MIDI support for SB 1.x */
+ return;
+
+ dev = sound_alloc_mididev();
+
+ if (dev == -1)
+ {
+ printk(KERN_ERR "sb_midi: too many MIDI devices detected\n");
+ return;
+ }
+ std_midi_synth.midi_dev = devc->my_mididev = dev;
+ midi_devs[dev] = kmalloc(sizeof(struct midi_operations), GFP_KERNEL);
+ if (midi_devs[dev] == NULL)
+ {
+ printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n");
+ sound_unload_mididev(dev);
+ return;
+ }
+ memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations,
+ sizeof(struct midi_operations));
+
+ if (owner)
+ midi_devs[dev]->owner = owner;
+
+ midi_devs[dev]->devc = devc;
+
+
+ midi_devs[dev]->converter = kmalloc(sizeof(struct synth_operations), GFP_KERNEL);
+ if (midi_devs[dev]->converter == NULL)
+ {
+ printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n");
+ kfree(midi_devs[dev]);
+ sound_unload_mididev(dev);
+ return;
+ }
+ memcpy((char *) midi_devs[dev]->converter, (char *) &std_midi_synth,
+ sizeof(struct synth_operations));
+
+ midi_devs[dev]->converter->id = "SBMIDI";
+ sequencer_init();
+}
diff --git a/sound/oss/sb_mixer.c b/sound/oss/sb_mixer.c
new file mode 100644
index 0000000..fad1a4f
--- /dev/null
+++ b/sound/oss/sb_mixer.c
@@ -0,0 +1,768 @@
+/*
+ * sound/oss/sb_mixer.c
+ *
+ * The low level mixer driver for the Sound Blaster compatible cards.
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Rolf Fokkens (Dec 20 1998) : Moved ESS stuff into sb_ess.[ch]
+ * Stanislav Voronyi <stas@esc.kharkov.com> : Support for AWE 3DSE device (Jun 7 1999)
+ */
+
+#include "sound_config.h"
+
+#define __SB_MIXER_C__
+
+#include "sb.h"
+#include "sb_mixer.h"
+
+#include "sb_ess.h"
+
+#define SBPRO_RECORDING_DEVICES (SOUND_MASK_LINE | SOUND_MASK_MIC | SOUND_MASK_CD)
+
+/* Same as SB Pro, unless I find otherwise */
+#define SGNXPRO_RECORDING_DEVICES SBPRO_RECORDING_DEVICES
+
+#define SBPRO_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_LINE | SOUND_MASK_MIC | \
+ SOUND_MASK_CD | SOUND_MASK_VOLUME)
+
+/* SG NX Pro has treble and bass settings on the mixer. The 'speaker'
+ * channel is the COVOX/DisneySoundSource emulation volume control
+ * on the mixer. It does NOT control speaker volume. Should have own
+ * mask eventually?
+ */
+#define SGNXPRO_MIXER_DEVICES (SBPRO_MIXER_DEVICES|SOUND_MASK_BASS| \
+ SOUND_MASK_TREBLE|SOUND_MASK_SPEAKER )
+
+#define SB16_RECORDING_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_LINE | SOUND_MASK_MIC | \
+ SOUND_MASK_CD)
+
+#define SB16_OUTFILTER_DEVICES (SOUND_MASK_LINE | SOUND_MASK_MIC | \
+ SOUND_MASK_CD)
+
+#define SB16_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \
+ SOUND_MASK_CD | \
+ SOUND_MASK_IGAIN | SOUND_MASK_OGAIN | \
+ SOUND_MASK_VOLUME | SOUND_MASK_BASS | SOUND_MASK_TREBLE | \
+ SOUND_MASK_IMIX)
+
+/* These are the only devices that are working at the moment. Others could
+ * be added once they are identified and a method is found to control them.
+ */
+#define ALS007_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_LINE | \
+ SOUND_MASK_PCM | SOUND_MASK_MIC | \
+ SOUND_MASK_CD | \
+ SOUND_MASK_VOLUME)
+
+static mixer_tab sbpro_mix = {
+MIX_ENT(SOUND_MIXER_VOLUME, 0x22, 7, 4, 0x22, 3, 4),
+MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x26, 7, 4, 0x26, 3, 4),
+MIX_ENT(SOUND_MIXER_PCM, 0x04, 7, 4, 0x04, 3, 4),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x2e, 7, 4, 0x2e, 3, 4),
+MIX_ENT(SOUND_MIXER_MIC, 0x0a, 2, 3, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_CD, 0x28, 7, 4, 0x28, 3, 4),
+MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0x00, 0, 0, 0x00, 0, 0)
+};
+
+static mixer_tab sb16_mix = {
+MIX_ENT(SOUND_MIXER_VOLUME, 0x30, 7, 5, 0x31, 7, 5),
+MIX_ENT(SOUND_MIXER_BASS, 0x46, 7, 4, 0x47, 7, 4),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x44, 7, 4, 0x45, 7, 4),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x34, 7, 5, 0x35, 7, 5),
+MIX_ENT(SOUND_MIXER_PCM, 0x32, 7, 5, 0x33, 7, 5),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x3b, 7, 2, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x38, 7, 5, 0x39, 7, 5),
+MIX_ENT(SOUND_MIXER_MIC, 0x3a, 7, 5, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_CD, 0x36, 7, 5, 0x37, 7, 5),
+MIX_ENT(SOUND_MIXER_IMIX, 0x3c, 0, 1, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0x3f, 7, 2, 0x40, 7, 2), /* Obsolete. Use IGAIN */
+MIX_ENT(SOUND_MIXER_IGAIN, 0x3f, 7, 2, 0x40, 7, 2),
+MIX_ENT(SOUND_MIXER_OGAIN, 0x41, 7, 2, 0x42, 7, 2)
+};
+
+static mixer_tab als007_mix =
+{
+MIX_ENT(SOUND_MIXER_VOLUME, 0x62, 7, 4, 0x62, 3, 4),
+MIX_ENT(SOUND_MIXER_BASS, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_TREBLE, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_SYNTH, 0x66, 7, 4, 0x66, 3, 4),
+MIX_ENT(SOUND_MIXER_PCM, 0x64, 7, 4, 0x64, 3, 4),
+MIX_ENT(SOUND_MIXER_SPEAKER, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_LINE, 0x6e, 7, 4, 0x6e, 3, 4),
+MIX_ENT(SOUND_MIXER_MIC, 0x6a, 2, 3, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_CD, 0x68, 7, 4, 0x68, 3, 4),
+MIX_ENT(SOUND_MIXER_IMIX, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_ALTPCM, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_RECLEV, 0x00, 0, 0, 0x00, 0, 0), /* Obsolete. Use IGAIN */
+MIX_ENT(SOUND_MIXER_IGAIN, 0x00, 0, 0, 0x00, 0, 0),
+MIX_ENT(SOUND_MIXER_OGAIN, 0x00, 0, 0, 0x00, 0, 0)
+};
+
+
+/* SM_GAMES Master volume is lower and PCM & FM volumes
+ higher than with SB Pro. This improves the
+ sound quality */
+
+static int smg_default_levels[32] =
+{
+ 0x2020, /* Master Volume */
+ 0x4b4b, /* Bass */
+ 0x4b4b, /* Treble */
+ 0x6464, /* FM */
+ 0x6464, /* PCM */
+ 0x4b4b, /* PC Speaker */
+ 0x4b4b, /* Ext Line */
+ 0x0000, /* Mic */
+ 0x4b4b, /* CD */
+ 0x4b4b, /* Recording monitor */
+ 0x4b4b, /* SB PCM */
+ 0x4b4b, /* Recording level */
+ 0x4b4b, /* Input gain */
+ 0x4b4b, /* Output gain */
+ 0x4040, /* Line1 */
+ 0x4040, /* Line2 */
+ 0x1515 /* Line3 */
+};
+
+static int sb_default_levels[32] =
+{
+ 0x5a5a, /* Master Volume */
+ 0x4b4b, /* Bass */
+ 0x4b4b, /* Treble */
+ 0x4b4b, /* FM */
+ 0x4b4b, /* PCM */
+ 0x4b4b, /* PC Speaker */
+ 0x4b4b, /* Ext Line */
+ 0x1010, /* Mic */
+ 0x4b4b, /* CD */
+ 0x0000, /* Recording monitor */
+ 0x4b4b, /* SB PCM */
+ 0x4b4b, /* Recording level */
+ 0x4b4b, /* Input gain */
+ 0x4b4b, /* Output gain */
+ 0x4040, /* Line1 */
+ 0x4040, /* Line2 */
+ 0x1515 /* Line3 */
+};
+
+static unsigned char sb16_recmasks_L[SOUND_MIXER_NRDEVICES] =
+{
+ 0x00, /* SOUND_MIXER_VOLUME */
+ 0x00, /* SOUND_MIXER_BASS */
+ 0x00, /* SOUND_MIXER_TREBLE */
+ 0x40, /* SOUND_MIXER_SYNTH */
+ 0x00, /* SOUND_MIXER_PCM */
+ 0x00, /* SOUND_MIXER_SPEAKER */
+ 0x10, /* SOUND_MIXER_LINE */
+ 0x01, /* SOUND_MIXER_MIC */
+ 0x04, /* SOUND_MIXER_CD */
+ 0x00, /* SOUND_MIXER_IMIX */
+ 0x00, /* SOUND_MIXER_ALTPCM */
+ 0x00, /* SOUND_MIXER_RECLEV */
+ 0x00, /* SOUND_MIXER_IGAIN */
+ 0x00 /* SOUND_MIXER_OGAIN */
+};
+
+static unsigned char sb16_recmasks_R[SOUND_MIXER_NRDEVICES] =
+{
+ 0x00, /* SOUND_MIXER_VOLUME */
+ 0x00, /* SOUND_MIXER_BASS */
+ 0x00, /* SOUND_MIXER_TREBLE */
+ 0x20, /* SOUND_MIXER_SYNTH */
+ 0x00, /* SOUND_MIXER_PCM */
+ 0x00, /* SOUND_MIXER_SPEAKER */
+ 0x08, /* SOUND_MIXER_LINE */
+ 0x01, /* SOUND_MIXER_MIC */
+ 0x02, /* SOUND_MIXER_CD */
+ 0x00, /* SOUND_MIXER_IMIX */
+ 0x00, /* SOUND_MIXER_ALTPCM */
+ 0x00, /* SOUND_MIXER_RECLEV */
+ 0x00, /* SOUND_MIXER_IGAIN */
+ 0x00 /* SOUND_MIXER_OGAIN */
+};
+
+static char smw_mix_regs[] = /* Left mixer registers */
+{
+ 0x0b, /* SOUND_MIXER_VOLUME */
+ 0x0d, /* SOUND_MIXER_BASS */
+ 0x0d, /* SOUND_MIXER_TREBLE */
+ 0x05, /* SOUND_MIXER_SYNTH */
+ 0x09, /* SOUND_MIXER_PCM */
+ 0x00, /* SOUND_MIXER_SPEAKER */
+ 0x03, /* SOUND_MIXER_LINE */
+ 0x01, /* SOUND_MIXER_MIC */
+ 0x07, /* SOUND_MIXER_CD */
+ 0x00, /* SOUND_MIXER_IMIX */
+ 0x00, /* SOUND_MIXER_ALTPCM */
+ 0x00, /* SOUND_MIXER_RECLEV */
+ 0x00, /* SOUND_MIXER_IGAIN */
+ 0x00, /* SOUND_MIXER_OGAIN */
+ 0x00, /* SOUND_MIXER_LINE1 */
+ 0x00, /* SOUND_MIXER_LINE2 */
+ 0x00 /* SOUND_MIXER_LINE3 */
+};
+
+static int sbmixnum = 1;
+
+static void sb_mixer_reset(sb_devc * devc);
+
+void sb_mixer_set_stereo(sb_devc * devc, int mode)
+{
+ sb_chgmixer(devc, OUT_FILTER, STEREO_DAC, (mode ? STEREO_DAC : MONO_DAC));
+}
+
+static int detect_mixer(sb_devc * devc)
+{
+ /* Just trust the mixer is there */
+ return 1;
+}
+
+static void change_bits(sb_devc * devc, unsigned char *regval, int dev, int chn, int newval)
+{
+ unsigned char mask;
+ int shift;
+
+ mask = (1 << (*devc->iomap)[dev][chn].nbits) - 1;
+ newval = (int) ((newval * mask) + 50) / 100; /* Scale */
+
+ shift = (*devc->iomap)[dev][chn].bitoffs - (*devc->iomap)[dev][LEFT_CHN].nbits + 1;
+
+ *regval &= ~(mask << shift); /* Mask out previous value */
+ *regval |= (newval & mask) << shift; /* Set the new value */
+}
+
+static int sb_mixer_get(sb_devc * devc, int dev)
+{
+ if (!((1 << dev) & devc->supported_devices))
+ return -EINVAL;
+ return devc->levels[dev];
+}
+
+void smw_mixer_init(sb_devc * devc)
+{
+ int i;
+
+ sb_setmixer(devc, 0x00, 0x18); /* Mute unused (Telephone) line */
+ sb_setmixer(devc, 0x10, 0x38); /* Config register 2 */
+
+ devc->supported_devices = 0;
+ for (i = 0; i < sizeof(smw_mix_regs); i++)
+ if (smw_mix_regs[i] != 0)
+ devc->supported_devices |= (1 << i);
+
+ devc->supported_rec_devices = devc->supported_devices &
+ ~(SOUND_MASK_BASS | SOUND_MASK_TREBLE | SOUND_MASK_PCM | SOUND_MASK_VOLUME);
+ sb_mixer_reset(devc);
+}
+
+int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right)
+{
+ int regoffs;
+ unsigned char val;
+
+ if ((dev < 0) || (dev >= devc->iomap_sz))
+ return -EINVAL;
+
+ regoffs = (*devc->iomap)[dev][LEFT_CHN].regno;
+
+ if (regoffs == 0)
+ return -EINVAL;
+
+ val = sb_getmixer(devc, regoffs);
+ change_bits(devc, &val, dev, LEFT_CHN, left);
+
+ if ((*devc->iomap)[dev][RIGHT_CHN].regno != regoffs) /*
+ * Change register
+ */
+ {
+ sb_setmixer(devc, regoffs, val); /*
+ * Save the old one
+ */
+ regoffs = (*devc->iomap)[dev][RIGHT_CHN].regno;
+
+ if (regoffs == 0)
+ return left | (left << 8); /*
+ * Just left channel present
+ */
+
+ val = sb_getmixer(devc, regoffs); /*
+ * Read the new one
+ */
+ }
+ change_bits(devc, &val, dev, RIGHT_CHN, right);
+
+ sb_setmixer(devc, regoffs, val);
+
+ return left | (right << 8);
+}
+
+static int smw_mixer_set(sb_devc * devc, int dev, int left, int right)
+{
+ int reg, val;
+
+ switch (dev)
+ {
+ case SOUND_MIXER_VOLUME:
+ sb_setmixer(devc, 0x0b, 96 - (96 * left / 100)); /* 96=mute, 0=max */
+ sb_setmixer(devc, 0x0c, 96 - (96 * right / 100));
+ break;
+
+ case SOUND_MIXER_BASS:
+ case SOUND_MIXER_TREBLE:
+ devc->levels[dev] = left | (right << 8);
+ /* Set left bass and treble values */
+ val = ((devc->levels[SOUND_MIXER_TREBLE] & 0xff) * 16 / (unsigned) 100) << 4;
+ val |= ((devc->levels[SOUND_MIXER_BASS] & 0xff) * 16 / (unsigned) 100) & 0x0f;
+ sb_setmixer(devc, 0x0d, val);
+
+ /* Set right bass and treble values */
+ val = (((devc->levels[SOUND_MIXER_TREBLE] >> 8) & 0xff) * 16 / (unsigned) 100) << 4;
+ val |= (((devc->levels[SOUND_MIXER_BASS] >> 8) & 0xff) * 16 / (unsigned) 100) & 0x0f;
+ sb_setmixer(devc, 0x0e, val);
+
+ break;
+
+ default:
+ /* bounds check */
+ if (dev < 0 || dev >= ARRAY_SIZE(smw_mix_regs))
+ return -EINVAL;
+ reg = smw_mix_regs[dev];
+ if (reg == 0)
+ return -EINVAL;
+ sb_setmixer(devc, reg, (24 - (24 * left / 100)) | 0x20); /* 24=mute, 0=max */
+ sb_setmixer(devc, reg + 1, (24 - (24 * right / 100)) | 0x40);
+ }
+
+ devc->levels[dev] = left | (right << 8);
+ return left | (right << 8);
+}
+
+static int sb_mixer_set(sb_devc * devc, int dev, int value)
+{
+ int left = value & 0x000000ff;
+ int right = (value & 0x0000ff00) >> 8;
+ int retval;
+
+ if (left > 100)
+ left = 100;
+ if (right > 100)
+ right = 100;
+
+ if ((dev < 0) || (dev > 31))
+ return -EINVAL;
+
+ if (!(devc->supported_devices & (1 << dev))) /*
+ * Not supported
+ */
+ return -EINVAL;
+
+ /* Differentiate depending on the chipsets */
+ switch (devc->model) {
+ case MDL_SMW:
+ retval = smw_mixer_set(devc, dev, left, right);
+ break;
+ case MDL_ESS:
+ retval = ess_mixer_set(devc, dev, left, right);
+ break;
+ default:
+ retval = sb_common_mixer_set(devc, dev, left, right);
+ }
+ if (retval >= 0) devc->levels[dev] = retval;
+
+ return retval;
+}
+
+/*
+ * set_recsrc doesn't apply to ES188x
+ */
+static void set_recsrc(sb_devc * devc, int src)
+{
+ sb_setmixer(devc, RECORD_SRC, (sb_getmixer(devc, RECORD_SRC) & ~7) | (src & 0x7));
+}
+
+static int set_recmask(sb_devc * devc, int mask)
+{
+ int devmask, i;
+ unsigned char regimageL, regimageR;
+
+ devmask = mask & devc->supported_rec_devices;
+
+ switch (devc->model)
+ {
+ case MDL_SBPRO:
+ case MDL_ESS:
+ case MDL_JAZZ:
+ case MDL_SMW:
+ if (devc->model == MDL_ESS && ess_set_recmask (devc, &devmask)) {
+ break;
+ };
+ if (devmask != SOUND_MASK_MIC &&
+ devmask != SOUND_MASK_LINE &&
+ devmask != SOUND_MASK_CD)
+ {
+ /*
+ * More than one device selected. Drop the
+ * previous selection
+ */
+ devmask &= ~devc->recmask;
+ }
+ if (devmask != SOUND_MASK_MIC &&
+ devmask != SOUND_MASK_LINE &&
+ devmask != SOUND_MASK_CD)
+ {
+ /*
+ * More than one device selected. Default to
+ * mic
+ */
+ devmask = SOUND_MASK_MIC;
+ }
+ if (devmask ^ devc->recmask) /*
+ * Input source changed
+ */
+ {
+ switch (devmask)
+ {
+ case SOUND_MASK_MIC:
+ set_recsrc(devc, SRC__MIC);
+ break;
+
+ case SOUND_MASK_LINE:
+ set_recsrc(devc, SRC__LINE);
+ break;
+
+ case SOUND_MASK_CD:
+ set_recsrc(devc, SRC__CD);
+ break;
+
+ default:
+ set_recsrc(devc, SRC__MIC);
+ }
+ }
+ break;
+
+ case MDL_SB16:
+ if (!devmask)
+ devmask = SOUND_MASK_MIC;
+
+ if (devc->submodel == SUBMDL_ALS007)
+ {
+ switch (devmask)
+ {
+ case SOUND_MASK_LINE:
+ sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_LINE);
+ break;
+ case SOUND_MASK_CD:
+ sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_CD);
+ break;
+ case SOUND_MASK_SYNTH:
+ sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_SYNTH);
+ break;
+ default: /* Also takes care of SOUND_MASK_MIC case */
+ sb_setmixer(devc, ALS007_RECORD_SRC, ALS007_MIC);
+ break;
+ }
+ }
+ else
+ {
+ regimageL = regimageR = 0;
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++)
+ {
+ if ((1 << i) & devmask)
+ {
+ regimageL |= sb16_recmasks_L[i];
+ regimageR |= sb16_recmasks_R[i];
+ }
+ sb_setmixer (devc, SB16_IMASK_L, regimageL);
+ sb_setmixer (devc, SB16_IMASK_R, regimageR);
+ }
+ }
+ break;
+ }
+ devc->recmask = devmask;
+ return devc->recmask;
+}
+
+static int set_outmask(sb_devc * devc, int mask)
+{
+ int devmask, i;
+ unsigned char regimage;
+
+ devmask = mask & devc->supported_out_devices;
+
+ switch (devc->model)
+ {
+ case MDL_SB16:
+ if (devc->submodel == SUBMDL_ALS007)
+ break;
+ else
+ {
+ regimage = 0;
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++)
+ {
+ if ((1 << i) & devmask)
+ {
+ regimage |= (sb16_recmasks_L[i] | sb16_recmasks_R[i]);
+ }
+ sb_setmixer (devc, SB16_OMASK, regimage);
+ }
+ }
+ break;
+ default:
+ break;
+ }
+
+ devc->outmask = devmask;
+ return devc->outmask;
+}
+
+static int sb_mixer_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ sb_devc *devc = mixer_devs[dev]->devc;
+ int val, ret;
+ int __user *p = arg;
+
+ /*
+ * Use ioctl(fd, SOUND_MIXER_AGC, &mode) to turn AGC off (0) or on (1).
+ * Use ioctl(fd, SOUND_MIXER_3DSE, &mode) to turn 3DSE off (0) or on (1)
+ * or mode==2 put 3DSE state to mode.
+ */
+ if (devc->model == MDL_SB16) {
+ if (cmd == SOUND_MIXER_AGC)
+ {
+ if (get_user(val, p))
+ return -EFAULT;
+ sb_setmixer(devc, 0x43, (~val) & 0x01);
+ return 0;
+ }
+ if (cmd == SOUND_MIXER_3DSE)
+ {
+ /* I put here 15, but I don't know the exact version.
+ At least my 4.13 havn't 3DSE, 4.16 has it. */
+ if (devc->minor < 15)
+ return -EINVAL;
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val == 0 || val == 1)
+ sb_chgmixer(devc, AWE_3DSE, 0x01, val);
+ else if (val == 2)
+ {
+ ret = sb_getmixer(devc, AWE_3DSE)&0x01;
+ return put_user(ret, p);
+ }
+ else
+ return -EINVAL;
+ return 0;
+ }
+ }
+ if (((cmd >> 8) & 0xff) == 'M')
+ {
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE)
+ {
+ if (get_user(val, p))
+ return -EFAULT;
+ switch (cmd & 0xff)
+ {
+ case SOUND_MIXER_RECSRC:
+ ret = set_recmask(devc, val);
+ break;
+
+ case SOUND_MIXER_OUTSRC:
+ ret = set_outmask(devc, val);
+ break;
+
+ default:
+ ret = sb_mixer_set(devc, cmd & 0xff, val);
+ }
+ }
+ else switch (cmd & 0xff)
+ {
+ case SOUND_MIXER_RECSRC:
+ ret = devc->recmask;
+ break;
+
+ case SOUND_MIXER_OUTSRC:
+ ret = devc->outmask;
+ break;
+
+ case SOUND_MIXER_DEVMASK:
+ ret = devc->supported_devices;
+ break;
+
+ case SOUND_MIXER_STEREODEVS:
+ ret = devc->supported_devices;
+ /* The ESS seems to have stereo mic controls */
+ if (devc->model == MDL_ESS)
+ ret &= ~(SOUND_MASK_SPEAKER|SOUND_MASK_IMIX);
+ else if (devc->model != MDL_JAZZ && devc->model != MDL_SMW)
+ ret &= ~(SOUND_MASK_MIC | SOUND_MASK_SPEAKER | SOUND_MASK_IMIX);
+ break;
+
+ case SOUND_MIXER_RECMASK:
+ ret = devc->supported_rec_devices;
+ break;
+
+ case SOUND_MIXER_OUTMASK:
+ ret = devc->supported_out_devices;
+ break;
+
+ case SOUND_MIXER_CAPS:
+ ret = devc->mixer_caps;
+ break;
+
+ default:
+ ret = sb_mixer_get(devc, cmd & 0xff);
+ break;
+ }
+ return put_user(ret, p);
+ } else
+ return -EINVAL;
+}
+
+static struct mixer_operations sb_mixer_operations =
+{
+ .owner = THIS_MODULE,
+ .id = "SB",
+ .name = "Sound Blaster",
+ .ioctl = sb_mixer_ioctl
+};
+
+static struct mixer_operations als007_mixer_operations =
+{
+ .owner = THIS_MODULE,
+ .id = "ALS007",
+ .name = "Avance ALS-007",
+ .ioctl = sb_mixer_ioctl
+};
+
+static void sb_mixer_reset(sb_devc * devc)
+{
+ char name[32];
+ int i;
+
+ sprintf(name, "SB_%d", devc->sbmixnum);
+
+ if (devc->sbmo.sm_games)
+ devc->levels = load_mixer_volumes(name, smg_default_levels, 1);
+ else
+ devc->levels = load_mixer_volumes(name, sb_default_levels, 1);
+
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++)
+ sb_mixer_set(devc, i, devc->levels[i]);
+
+ if (devc->model != MDL_ESS || !ess_mixer_reset (devc)) {
+ set_recmask(devc, SOUND_MASK_MIC);
+ };
+}
+
+int sb_mixer_init(sb_devc * devc, struct module *owner)
+{
+ int mixer_type = 0;
+ int m;
+
+ devc->sbmixnum = sbmixnum++;
+ devc->levels = NULL;
+
+ sb_setmixer(devc, 0x00, 0); /* Reset mixer */
+
+ if (!(mixer_type = detect_mixer(devc)))
+ return 0; /* No mixer. Why? */
+
+ switch (devc->model)
+ {
+ case MDL_ESSPCI:
+ case MDL_YMPCI:
+ case MDL_SBPRO:
+ case MDL_AZTECH:
+ case MDL_JAZZ:
+ devc->mixer_caps = SOUND_CAP_EXCL_INPUT;
+ devc->supported_devices = SBPRO_MIXER_DEVICES;
+ devc->supported_rec_devices = SBPRO_RECORDING_DEVICES;
+ devc->iomap = &sbpro_mix;
+ devc->iomap_sz = ARRAY_SIZE(sbpro_mix);
+ break;
+
+ case MDL_ESS:
+ ess_mixer_init (devc);
+ break;
+
+ case MDL_SMW:
+ devc->mixer_caps = SOUND_CAP_EXCL_INPUT;
+ devc->supported_devices = 0;
+ devc->supported_rec_devices = 0;
+ devc->iomap = &sbpro_mix;
+ devc->iomap_sz = ARRAY_SIZE(sbpro_mix);
+ smw_mixer_init(devc);
+ break;
+
+ case MDL_SB16:
+ devc->mixer_caps = 0;
+ devc->supported_rec_devices = SB16_RECORDING_DEVICES;
+ devc->supported_out_devices = SB16_OUTFILTER_DEVICES;
+ if (devc->submodel != SUBMDL_ALS007)
+ {
+ devc->supported_devices = SB16_MIXER_DEVICES;
+ devc->iomap = &sb16_mix;
+ devc->iomap_sz = ARRAY_SIZE(sb16_mix);
+ }
+ else
+ {
+ devc->supported_devices = ALS007_MIXER_DEVICES;
+ devc->iomap = &als007_mix;
+ devc->iomap_sz = ARRAY_SIZE(als007_mix);
+ }
+ break;
+
+ default:
+ printk(KERN_WARNING "sb_mixer: Unsupported mixer type %d\n", devc->model);
+ return 0;
+ }
+
+ m = sound_alloc_mixerdev();
+ if (m == -1)
+ return 0;
+
+ mixer_devs[m] = kmalloc(sizeof(struct mixer_operations), GFP_KERNEL);
+ if (mixer_devs[m] == NULL)
+ {
+ printk(KERN_ERR "sb_mixer: Can't allocate memory\n");
+ sound_unload_mixerdev(m);
+ return 0;
+ }
+
+ if (devc->submodel != SUBMDL_ALS007)
+ memcpy ((char *) mixer_devs[m], (char *) &sb_mixer_operations, sizeof (struct mixer_operations));
+ else
+ memcpy ((char *) mixer_devs[m], (char *) &als007_mixer_operations, sizeof (struct mixer_operations));
+
+ mixer_devs[m]->devc = devc;
+
+ if (owner)
+ mixer_devs[m]->owner = owner;
+
+ devc->my_mixerdev = m;
+ sb_mixer_reset(devc);
+ return 1;
+}
+
+void sb_mixer_unload(sb_devc *devc)
+{
+ if (devc->my_mixerdev == -1)
+ return;
+
+ kfree(mixer_devs[devc->my_mixerdev]);
+ sound_unload_mixerdev(devc->my_mixerdev);
+ sbmixnum--;
+}
diff --git a/sound/oss/sb_mixer.h b/sound/oss/sb_mixer.h
new file mode 100644
index 0000000..4b9425f
--- /dev/null
+++ b/sound/oss/sb_mixer.h
@@ -0,0 +1,105 @@
+/*
+ * sound/oss/sb_mixer.h
+ *
+ * Definitions for the SB Pro and SB16 mixers
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+
+/*
+ * Modified:
+ * Hunyue Yau Jan 6 1994
+ * Added defines for the Sound Galaxy NX Pro mixer.
+ *
+ * Rolf Fokkens Dec 20 1998
+ * Added defines for some ES188x chips.
+ *
+ * Rolf Fokkens Dec 27 1998
+ * Moved static stuff to sb_mixer.c
+ *
+ */
+/*
+ * Mixer registers
+ *
+ * NOTE! RECORD_SRC == IN_FILTER
+ */
+
+/*
+ * Mixer registers of SB Pro
+ */
+#define VOC_VOL 0x04
+#define MIC_VOL 0x0A
+#define MIC_MIX 0x0A
+#define RECORD_SRC 0x0C
+#define IN_FILTER 0x0C
+#define OUT_FILTER 0x0E
+#define MASTER_VOL 0x22
+#define FM_VOL 0x26
+#define CD_VOL 0x28
+#define LINE_VOL 0x2E
+#define IRQ_NR 0x80
+#define DMA_NR 0x81
+#define IRQ_STAT 0x82
+#define OPSW 0x3c
+
+/*
+ * Additional registers on the SG NX Pro
+ */
+#define COVOX_VOL 0x42
+#define TREBLE_LVL 0x44
+#define BASS_LVL 0x46
+
+#define FREQ_HI (1 << 3)/* Use High-frequency ANFI filters */
+#define FREQ_LOW 0 /* Use Low-frequency ANFI filters */
+#define FILT_ON 0 /* Yes, 0 to turn it on, 1 for off */
+#define FILT_OFF (1 << 5)
+
+#define MONO_DAC 0x00
+#define STEREO_DAC 0x02
+
+/*
+ * Mixer registers of SB16
+ */
+#define SB16_OMASK 0x3c
+#define SB16_IMASK_L 0x3d
+#define SB16_IMASK_R 0x3e
+
+#define LEFT_CHN 0
+#define RIGHT_CHN 1
+
+/*
+ * 3DSE register of AWE32/64
+ */
+#define AWE_3DSE 0x90
+
+/*
+ * Mixer registers of ALS007
+ */
+#define ALS007_RECORD_SRC 0x6c
+#define ALS007_OUTPUT_CTRL1 0x3c
+#define ALS007_OUTPUT_CTRL2 0x4c
+
+#define MIX_ENT(name, reg_l, bit_l, len_l, reg_r, bit_r, len_r) \
+ {{reg_l, bit_l, len_l}, {reg_r, bit_r, len_r}}
+
+/*
+ * Recording sources (SB Pro)
+ */
+
+#define SRC__MIC 1 /* Select Microphone recording source */
+#define SRC__CD 3 /* Select CD recording source */
+#define SRC__LINE 7 /* Use Line-in for recording source */
+
+/*
+ * Recording sources for ALS-007
+ */
+
+#define ALS007_MIC 4
+#define ALS007_LINE 6
+#define ALS007_CD 2
+#define ALS007_SYNTH 7
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
new file mode 100644
index 0000000..5c215f7
--- /dev/null
+++ b/sound/oss/sequencer.c
@@ -0,0 +1,1674 @@
+/*
+ * sound/oss/sequencer.c
+ *
+ * The sequencer personality manager.
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+/*
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Alan Cox : reformatted and fixed a pair of null pointer bugs
+ */
+#include <linux/kmod.h>
+#include <linux/spinlock.h>
+#include "sound_config.h"
+
+#include "midi_ctrl.h"
+
+static int sequencer_ok;
+static struct sound_timer_operations *tmr;
+static int tmr_no = -1; /* Currently selected timer */
+static int pending_timer = -1; /* For timer change operation */
+extern unsigned long seq_time;
+
+static int obsolete_api_used;
+static DEFINE_SPINLOCK(lock);
+
+/*
+ * Local counts for number of synth and MIDI devices. These are initialized
+ * by the sequencer_open.
+ */
+static int max_mididev;
+static int max_synthdev;
+
+/*
+ * The seq_mode gives the operating mode of the sequencer:
+ * 1 = level1 (the default)
+ * 2 = level2 (extended capabilities)
+ */
+
+#define SEQ_1 1
+#define SEQ_2 2
+static int seq_mode = SEQ_1;
+
+static DECLARE_WAIT_QUEUE_HEAD(seq_sleeper);
+static DECLARE_WAIT_QUEUE_HEAD(midi_sleeper);
+
+static int midi_opened[MAX_MIDI_DEV];
+
+static int midi_written[MAX_MIDI_DEV];
+
+static unsigned long prev_input_time;
+static int prev_event_time;
+
+#include "tuning.h"
+
+#define EV_SZ 8
+#define IEV_SZ 8
+
+static unsigned char *queue;
+static unsigned char *iqueue;
+
+static volatile int qhead, qtail, qlen;
+static volatile int iqhead, iqtail, iqlen;
+static volatile int seq_playing;
+static volatile int sequencer_busy;
+static int output_threshold;
+static long pre_event_timeout;
+static unsigned synth_open_mask;
+
+static int seq_queue(unsigned char *note, char nonblock);
+static void seq_startplay(void);
+static int seq_sync(void);
+static void seq_reset(void);
+
+#if MAX_SYNTH_DEV > 15
+#error Too many synthesizer devices enabled.
+#endif
+
+int sequencer_read(int dev, struct file *file, char __user *buf, int count)
+{
+ int c = count, p = 0;
+ int ev_len;
+ unsigned long flags;
+
+ dev = dev >> 4;
+
+ ev_len = seq_mode == SEQ_1 ? 4 : 8;
+
+ spin_lock_irqsave(&lock,flags);
+
+ if (!iqlen)
+ {
+ spin_unlock_irqrestore(&lock,flags);
+ if (file->f_flags & O_NONBLOCK) {
+ return -EAGAIN;
+ }
+
+ interruptible_sleep_on_timeout(&midi_sleeper,
+ pre_event_timeout);
+ spin_lock_irqsave(&lock,flags);
+ if (!iqlen)
+ {
+ spin_unlock_irqrestore(&lock,flags);
+ return 0;
+ }
+ }
+ while (iqlen && c >= ev_len)
+ {
+ char *fixit = (char *) &iqueue[iqhead * IEV_SZ];
+ spin_unlock_irqrestore(&lock,flags);
+ if (copy_to_user(&(buf)[p], fixit, ev_len))
+ return count - c;
+ p += ev_len;
+ c -= ev_len;
+
+ spin_lock_irqsave(&lock,flags);
+ iqhead = (iqhead + 1) % SEQ_MAX_QUEUE;
+ iqlen--;
+ }
+ spin_unlock_irqrestore(&lock,flags);
+ return count - c;
+}
+
+static void sequencer_midi_output(int dev)
+{
+ /*
+ * Currently NOP
+ */
+}
+
+void seq_copy_to_input(unsigned char *event_rec, int len)
+{
+ unsigned long flags;
+
+ /*
+ * Verify that the len is valid for the current mode.
+ */
+
+ if (len != 4 && len != 8)
+ return;
+ if ((seq_mode == SEQ_1) != (len == 4))
+ return;
+
+ if (iqlen >= (SEQ_MAX_QUEUE - 1))
+ return; /* Overflow */
+
+ spin_lock_irqsave(&lock,flags);
+ memcpy(&iqueue[iqtail * IEV_SZ], event_rec, len);
+ iqlen++;
+ iqtail = (iqtail + 1) % SEQ_MAX_QUEUE;
+ wake_up(&midi_sleeper);
+ spin_unlock_irqrestore(&lock,flags);
+}
+EXPORT_SYMBOL(seq_copy_to_input);
+
+static void sequencer_midi_input(int dev, unsigned char data)
+{
+ unsigned int tstamp;
+ unsigned char event_rec[4];
+
+ if (data == 0xfe) /* Ignore active sensing */
+ return;
+
+ tstamp = jiffies - seq_time;
+
+ if (tstamp != prev_input_time)
+ {
+ tstamp = (tstamp << 8) | SEQ_WAIT;
+ seq_copy_to_input((unsigned char *) &tstamp, 4);
+ prev_input_time = tstamp;
+ }
+ event_rec[0] = SEQ_MIDIPUTC;
+ event_rec[1] = data;
+ event_rec[2] = dev;
+ event_rec[3] = 0;
+
+ seq_copy_to_input(event_rec, 4);
+}
+
+void seq_input_event(unsigned char *event_rec, int len)
+{
+ unsigned long this_time;
+
+ if (seq_mode == SEQ_2)
+ this_time = tmr->get_time(tmr_no);
+ else
+ this_time = jiffies - seq_time;
+
+ if (this_time != prev_input_time)
+ {
+ unsigned char tmp_event[8];
+
+ tmp_event[0] = EV_TIMING;
+ tmp_event[1] = TMR_WAIT_ABS;
+ tmp_event[2] = 0;
+ tmp_event[3] = 0;
+ *(unsigned int *) &tmp_event[4] = this_time;
+
+ seq_copy_to_input(tmp_event, 8);
+ prev_input_time = this_time;
+ }
+ seq_copy_to_input(event_rec, len);
+}
+EXPORT_SYMBOL(seq_input_event);
+
+int sequencer_write(int dev, struct file *file, const char __user *buf, int count)
+{
+ unsigned char event_rec[EV_SZ], ev_code;
+ int p = 0, c, ev_size;
+ int err;
+ int mode = translate_mode(file);
+
+ dev = dev >> 4;
+
+ DEB(printk("sequencer_write(dev=%d, count=%d)\n", dev, count));
+
+ if (mode == OPEN_READ)
+ return -EIO;
+
+ c = count;
+
+ while (c >= 4)
+ {
+ if (copy_from_user((char *) event_rec, &(buf)[p], 4))
+ goto out;
+ ev_code = event_rec[0];
+
+ if (ev_code == SEQ_FULLSIZE)
+ {
+ int err, fmt;
+
+ dev = *(unsigned short *) &event_rec[2];
+ if (dev < 0 || dev >= max_synthdev || synth_devs[dev] == NULL)
+ return -ENXIO;
+
+ if (!(synth_open_mask & (1 << dev)))
+ return -ENXIO;
+
+ fmt = (*(short *) &event_rec[0]) & 0xffff;
+ err = synth_devs[dev]->load_patch(dev, fmt, buf, p + 4, c, 0);
+ if (err < 0)
+ return err;
+
+ return err;
+ }
+ if (ev_code >= 128)
+ {
+ if (seq_mode == SEQ_2 && ev_code == SEQ_EXTENDED)
+ {
+ printk(KERN_WARNING "Sequencer: Invalid level 2 event %x\n", ev_code);
+ return -EINVAL;
+ }
+ ev_size = 8;
+
+ if (c < ev_size)
+ {
+ if (!seq_playing)
+ seq_startplay();
+ return count - c;
+ }
+ if (copy_from_user((char *)&event_rec[4],
+ &(buf)[p + 4], 4))
+ goto out;
+
+ }
+ else
+ {
+ if (seq_mode == SEQ_2)
+ {
+ printk(KERN_WARNING "Sequencer: 4 byte event in level 2 mode\n");
+ return -EINVAL;
+ }
+ ev_size = 4;
+
+ if (event_rec[0] != SEQ_MIDIPUTC)
+ obsolete_api_used = 1;
+ }
+
+ if (event_rec[0] == SEQ_MIDIPUTC)
+ {
+ if (!midi_opened[event_rec[2]])
+ {
+ int mode;
+ int dev = event_rec[2];
+
+ if (dev >= max_mididev || midi_devs[dev]==NULL)
+ {
+ /*printk("Sequencer Error: Nonexistent MIDI device %d\n", dev);*/
+ return -ENXIO;
+ }
+ mode = translate_mode(file);
+
+ if ((err = midi_devs[dev]->open(dev, mode,
+ sequencer_midi_input, sequencer_midi_output)) < 0)
+ {
+ seq_reset();
+ printk(KERN_WARNING "Sequencer Error: Unable to open Midi #%d\n", dev);
+ return err;
+ }
+ midi_opened[dev] = 1;
+ }
+ }
+ if (!seq_queue(event_rec, (file->f_flags & (O_NONBLOCK) ? 1 : 0)))
+ {
+ int processed = count - c;
+
+ if (!seq_playing)
+ seq_startplay();
+
+ if (!processed && (file->f_flags & O_NONBLOCK))
+ return -EAGAIN;
+ else
+ return processed;
+ }
+ p += ev_size;
+ c -= ev_size;
+ }
+
+ if (!seq_playing)
+ seq_startplay();
+out:
+ return count;
+}
+
+static int seq_queue(unsigned char *note, char nonblock)
+{
+
+ /*
+ * Test if there is space in the queue
+ */
+
+ if (qlen >= SEQ_MAX_QUEUE)
+ if (!seq_playing)
+ seq_startplay(); /*
+ * Give chance to drain the queue
+ */
+
+ if (!nonblock && qlen >= SEQ_MAX_QUEUE && !waitqueue_active(&seq_sleeper)) {
+ /*
+ * Sleep until there is enough space on the queue
+ */
+ interruptible_sleep_on(&seq_sleeper);
+ }
+ if (qlen >= SEQ_MAX_QUEUE)
+ {
+ return 0; /*
+ * To be sure
+ */
+ }
+ memcpy(&queue[qtail * EV_SZ], note, EV_SZ);
+
+ qtail = (qtail + 1) % SEQ_MAX_QUEUE;
+ qlen++;
+
+ return 1;
+}
+
+static int extended_event(unsigned char *q)
+{
+ int dev = q[2];
+
+ if (dev < 0 || dev >= max_synthdev)
+ return -ENXIO;
+
+ if (!(synth_open_mask & (1 << dev)))
+ return -ENXIO;
+
+ switch (q[1])
+ {
+ case SEQ_NOTEOFF:
+ synth_devs[dev]->kill_note(dev, q[3], q[4], q[5]);
+ break;
+
+ case SEQ_NOTEON:
+ if (q[4] > 127 && q[4] != 255)
+ return 0;
+
+ if (q[5] == 0)
+ {
+ synth_devs[dev]->kill_note(dev, q[3], q[4], q[5]);
+ break;
+ }
+ synth_devs[dev]->start_note(dev, q[3], q[4], q[5]);
+ break;
+
+ case SEQ_PGMCHANGE:
+ synth_devs[dev]->set_instr(dev, q[3], q[4]);
+ break;
+
+ case SEQ_AFTERTOUCH:
+ synth_devs[dev]->aftertouch(dev, q[3], q[4]);
+ break;
+
+ case SEQ_BALANCE:
+ synth_devs[dev]->panning(dev, q[3], (char) q[4]);
+ break;
+
+ case SEQ_CONTROLLER:
+ synth_devs[dev]->controller(dev, q[3], q[4], (short) (q[5] | (q[6] << 8)));
+ break;
+
+ case SEQ_VOLMODE:
+ if (synth_devs[dev]->volume_method != NULL)
+ synth_devs[dev]->volume_method(dev, q[3]);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int find_voice(int dev, int chn, int note)
+{
+ unsigned short key;
+ int i;
+
+ key = (chn << 8) | (note + 1);
+ for (i = 0; i < synth_devs[dev]->alloc.max_voice; i++)
+ if (synth_devs[dev]->alloc.map[i] == key)
+ return i;
+ return -1;
+}
+
+static int alloc_voice(int dev, int chn, int note)
+{
+ unsigned short key;
+ int voice;
+
+ key = (chn << 8) | (note + 1);
+
+ voice = synth_devs[dev]->alloc_voice(dev, chn, note,
+ &synth_devs[dev]->alloc);
+ synth_devs[dev]->alloc.map[voice] = key;
+ synth_devs[dev]->alloc.alloc_times[voice] =
+ synth_devs[dev]->alloc.timestamp++;
+ return voice;
+}
+
+static void seq_chn_voice_event(unsigned char *event_rec)
+{
+#define dev event_rec[1]
+#define cmd event_rec[2]
+#define chn event_rec[3]
+#define note event_rec[4]
+#define parm event_rec[5]
+
+ int voice = -1;
+
+ if ((int) dev > max_synthdev || synth_devs[dev] == NULL)
+ return;
+ if (!(synth_open_mask & (1 << dev)))
+ return;
+ if (!synth_devs[dev])
+ return;
+
+ if (seq_mode == SEQ_2)
+ {
+ if (synth_devs[dev]->alloc_voice)
+ voice = find_voice(dev, chn, note);
+
+ if (cmd == MIDI_NOTEON && parm == 0)
+ {
+ cmd = MIDI_NOTEOFF;
+ parm = 64;
+ }
+ }
+
+ switch (cmd)
+ {
+ case MIDI_NOTEON:
+ if (note > 127 && note != 255) /* Not a seq2 feature */
+ return;
+
+ if (voice == -1 && seq_mode == SEQ_2 && synth_devs[dev]->alloc_voice)
+ {
+ /* Internal synthesizer (FM, GUS, etc) */
+ voice = alloc_voice(dev, chn, note);
+ }
+ if (voice == -1)
+ voice = chn;
+
+ if (seq_mode == SEQ_2 && (int) dev < num_synths)
+ {
+ /*
+ * The MIDI channel 10 is a percussive channel. Use the note
+ * number to select the proper patch (128 to 255) to play.
+ */
+
+ if (chn == 9)
+ {
+ synth_devs[dev]->set_instr(dev, voice, 128 + note);
+ synth_devs[dev]->chn_info[chn].pgm_num = 128 + note;
+ }
+ synth_devs[dev]->setup_voice(dev, voice, chn);
+ }
+ synth_devs[dev]->start_note(dev, voice, note, parm);
+ break;
+
+ case MIDI_NOTEOFF:
+ if (voice == -1)
+ voice = chn;
+ synth_devs[dev]->kill_note(dev, voice, note, parm);
+ break;
+
+ case MIDI_KEY_PRESSURE:
+ if (voice == -1)
+ voice = chn;
+ synth_devs[dev]->aftertouch(dev, voice, parm);
+ break;
+
+ default:;
+ }
+#undef dev
+#undef cmd
+#undef chn
+#undef note
+#undef parm
+}
+
+
+static void seq_chn_common_event(unsigned char *event_rec)
+{
+ unsigned char dev = event_rec[1];
+ unsigned char cmd = event_rec[2];
+ unsigned char chn = event_rec[3];
+ unsigned char p1 = event_rec[4];
+
+ /* unsigned char p2 = event_rec[5]; */
+ unsigned short w14 = *(short *) &event_rec[6];
+
+ if ((int) dev > max_synthdev || synth_devs[dev] == NULL)
+ return;
+ if (!(synth_open_mask & (1 << dev)))
+ return;
+ if (!synth_devs[dev])
+ return;
+
+ switch (cmd)
+ {
+ case MIDI_PGM_CHANGE:
+ if (seq_mode == SEQ_2)
+ {
+ synth_devs[dev]->chn_info[chn].pgm_num = p1;
+ if ((int) dev >= num_synths)
+ synth_devs[dev]->set_instr(dev, chn, p1);
+ }
+ else
+ synth_devs[dev]->set_instr(dev, chn, p1);
+
+ break;
+
+ case MIDI_CTL_CHANGE:
+ if (seq_mode == SEQ_2)
+ {
+ if (chn > 15 || p1 > 127)
+ break;
+
+ synth_devs[dev]->chn_info[chn].controllers[p1] = w14 & 0x7f;
+
+ if (p1 < 32) /* Setting MSB should clear LSB to 0 */
+ synth_devs[dev]->chn_info[chn].controllers[p1 + 32] = 0;
+
+ if ((int) dev < num_synths)
+ {
+ int val = w14 & 0x7f;
+ int i, key;
+
+ if (p1 < 64) /* Combine MSB and LSB */
+ {
+ val = ((synth_devs[dev]->
+ chn_info[chn].controllers[p1 & ~32] & 0x7f) << 7)
+ | (synth_devs[dev]->
+ chn_info[chn].controllers[p1 | 32] & 0x7f);
+ p1 &= ~32;
+ }
+ /* Handle all playing notes on this channel */
+
+ key = ((int) chn << 8);
+
+ for (i = 0; i < synth_devs[dev]->alloc.max_voice; i++)
+ if ((synth_devs[dev]->alloc.map[i] & 0xff00) == key)
+ synth_devs[dev]->controller(dev, i, p1, val);
+ }
+ else
+ synth_devs[dev]->controller(dev, chn, p1, w14);
+ }
+ else /* Mode 1 */
+ synth_devs[dev]->controller(dev, chn, p1, w14);
+ break;
+
+ case MIDI_PITCH_BEND:
+ if (seq_mode == SEQ_2)
+ {
+ synth_devs[dev]->chn_info[chn].bender_value = w14;
+
+ if ((int) dev < num_synths)
+ {
+ /* Handle all playing notes on this channel */
+ int i, key;
+
+ key = (chn << 8);
+
+ for (i = 0; i < synth_devs[dev]->alloc.max_voice; i++)
+ if ((synth_devs[dev]->alloc.map[i] & 0xff00) == key)
+ synth_devs[dev]->bender(dev, i, w14);
+ }
+ else
+ synth_devs[dev]->bender(dev, chn, w14);
+ }
+ else /* MODE 1 */
+ synth_devs[dev]->bender(dev, chn, w14);
+ break;
+
+ default:;
+ }
+}
+
+static int seq_timing_event(unsigned char *event_rec)
+{
+ unsigned char cmd = event_rec[1];
+ unsigned int parm = *(int *) &event_rec[4];
+
+ if (seq_mode == SEQ_2)
+ {
+ int ret;
+
+ if ((ret = tmr->event(tmr_no, event_rec)) == TIMER_ARMED)
+ if ((SEQ_MAX_QUEUE - qlen) >= output_threshold)
+ wake_up(&seq_sleeper);
+ return ret;
+ }
+ switch (cmd)
+ {
+ case TMR_WAIT_REL:
+ parm += prev_event_time;
+
+ /*
+ * NOTE! No break here. Execution of TMR_WAIT_REL continues in the
+ * next case (TMR_WAIT_ABS)
+ */
+
+ case TMR_WAIT_ABS:
+ if (parm > 0)
+ {
+ long time;
+
+ time = parm;
+ prev_event_time = time;
+
+ seq_playing = 1;
+ request_sound_timer(time);
+
+ if ((SEQ_MAX_QUEUE - qlen) >= output_threshold)
+ wake_up(&seq_sleeper);
+ return TIMER_ARMED;
+ }
+ break;
+
+ case TMR_START:
+ seq_time = jiffies;
+ prev_input_time = 0;
+ prev_event_time = 0;
+ break;
+
+ case TMR_STOP:
+ break;
+
+ case TMR_CONTINUE:
+ break;
+
+ case TMR_TEMPO:
+ break;
+
+ case TMR_ECHO:
+ if (seq_mode == SEQ_2)
+ seq_copy_to_input(event_rec, 8);
+ else
+ {
+ parm = (parm << 8 | SEQ_ECHO);
+ seq_copy_to_input((unsigned char *) &parm, 4);
+ }
+ break;
+
+ default:;
+ }
+
+ return TIMER_NOT_ARMED;
+}
+
+static void seq_local_event(unsigned char *event_rec)
+{
+ unsigned char cmd = event_rec[1];
+ unsigned int parm = *((unsigned int *) &event_rec[4]);
+
+ switch (cmd)
+ {
+ case LOCL_STARTAUDIO:
+ DMAbuf_start_devices(parm);
+ break;
+
+ default:;
+ }
+}
+
+static void seq_sysex_message(unsigned char *event_rec)
+{
+ unsigned int dev = event_rec[1];
+ int i, l = 0;
+ unsigned char *buf = &event_rec[2];
+
+ if (dev > max_synthdev)
+ return;
+ if (!(synth_open_mask & (1 << dev)))
+ return;
+ if (!synth_devs[dev])
+ return;
+
+ l = 0;
+ for (i = 0; i < 6 && buf[i] != 0xff; i++)
+ l = i + 1;
+
+ if (!synth_devs[dev]->send_sysex)
+ return;
+ if (l > 0)
+ synth_devs[dev]->send_sysex(dev, buf, l);
+}
+
+static int play_event(unsigned char *q)
+{
+ /*
+ * NOTE! This routine returns
+ * 0 = normal event played.
+ * 1 = Timer armed. Suspend playback until timer callback.
+ * 2 = MIDI output buffer full. Restore queue and suspend until timer
+ */
+ unsigned int *delay;
+
+ switch (q[0])
+ {
+ case SEQ_NOTEOFF:
+ if (synth_open_mask & (1 << 0))
+ if (synth_devs[0])
+ synth_devs[0]->kill_note(0, q[1], 255, q[3]);
+ break;
+
+ case SEQ_NOTEON:
+ if (q[4] < 128 || q[4] == 255)
+ if (synth_open_mask & (1 << 0))
+ if (synth_devs[0])
+ synth_devs[0]->start_note(0, q[1], q[2], q[3]);
+ break;
+
+ case SEQ_WAIT:
+ delay = (unsigned int *) q; /*
+ * Bytes 1 to 3 are containing the *
+ * delay in 'ticks'
+ */
+ *delay = (*delay >> 8) & 0xffffff;
+
+ if (*delay > 0)
+ {
+ long time;
+
+ seq_playing = 1;
+ time = *delay;
+ prev_event_time = time;
+
+ request_sound_timer(time);
+
+ if ((SEQ_MAX_QUEUE - qlen) >= output_threshold)
+ wake_up(&seq_sleeper);
+ /*
+ * The timer is now active and will reinvoke this function
+ * after the timer expires. Return to the caller now.
+ */
+ return 1;
+ }
+ break;
+
+ case SEQ_PGMCHANGE:
+ if (synth_open_mask & (1 << 0))
+ if (synth_devs[0])
+ synth_devs[0]->set_instr(0, q[1], q[2]);
+ break;
+
+ case SEQ_SYNCTIMER: /*
+ * Reset timer
+ */
+ seq_time = jiffies;
+ prev_input_time = 0;
+ prev_event_time = 0;
+ break;
+
+ case SEQ_MIDIPUTC: /*
+ * Put a midi character
+ */
+ if (midi_opened[q[2]])
+ {
+ int dev;
+
+ dev = q[2];
+
+ if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL)
+ break;
+
+ if (!midi_devs[dev]->outputc(dev, q[1]))
+ {
+ /*
+ * Output FIFO is full. Wait one timer cycle and try again.
+ */
+
+ seq_playing = 1;
+ request_sound_timer(-1);
+ return 2;
+ }
+ else
+ midi_written[dev] = 1;
+ }
+ break;
+
+ case SEQ_ECHO:
+ seq_copy_to_input(q, 4); /*
+ * Echo back to the process
+ */
+ break;
+
+ case SEQ_PRIVATE:
+ if ((int) q[1] < max_synthdev)
+ synth_devs[q[1]]->hw_control(q[1], q);
+ break;
+
+ case SEQ_EXTENDED:
+ extended_event(q);
+ break;
+
+ case EV_CHN_VOICE:
+ seq_chn_voice_event(q);
+ break;
+
+ case EV_CHN_COMMON:
+ seq_chn_common_event(q);
+ break;
+
+ case EV_TIMING:
+ if (seq_timing_event(q) == TIMER_ARMED)
+ {
+ return 1;
+ }
+ break;
+
+ case EV_SEQ_LOCAL:
+ seq_local_event(q);
+ break;
+
+ case EV_SYSEX:
+ seq_sysex_message(q);
+ break;
+
+ default:;
+ }
+ return 0;
+}
+
+/* called also as timer in irq context */
+static void seq_startplay(void)
+{
+ int this_one, action;
+ unsigned long flags;
+
+ while (qlen > 0)
+ {
+
+ spin_lock_irqsave(&lock,flags);
+ qhead = ((this_one = qhead) + 1) % SEQ_MAX_QUEUE;
+ qlen--;
+ spin_unlock_irqrestore(&lock,flags);
+
+ seq_playing = 1;
+
+ if ((action = play_event(&queue[this_one * EV_SZ])))
+ { /* Suspend playback. Next timer routine invokes this routine again */
+ if (action == 2)
+ {
+ qlen++;
+ qhead = this_one;
+ }
+ return;
+ }
+ }
+
+ seq_playing = 0;
+
+ if ((SEQ_MAX_QUEUE - qlen) >= output_threshold)
+ wake_up(&seq_sleeper);
+}
+
+static void reset_controllers(int dev, unsigned char *controller, int update_dev)
+{
+ int i;
+ for (i = 0; i < 128; i++)
+ controller[i] = ctrl_def_values[i];
+}
+
+static void setup_mode2(void)
+{
+ int dev;
+
+ max_synthdev = num_synths;
+
+ for (dev = 0; dev < num_midis; dev++)
+ {
+ if (midi_devs[dev] && midi_devs[dev]->converter != NULL)
+ {
+ synth_devs[max_synthdev++] = midi_devs[dev]->converter;
+ }
+ }
+
+ for (dev = 0; dev < max_synthdev; dev++)
+ {
+ int chn;
+
+ synth_devs[dev]->sysex_ptr = 0;
+ synth_devs[dev]->emulation = 0;
+
+ for (chn = 0; chn < 16; chn++)
+ {
+ synth_devs[dev]->chn_info[chn].pgm_num = 0;
+ reset_controllers(dev,
+ synth_devs[dev]->chn_info[chn].controllers,0);
+ synth_devs[dev]->chn_info[chn].bender_value = (1 << 7); /* Neutral */
+ synth_devs[dev]->chn_info[chn].bender_range = 200;
+ }
+ }
+ max_mididev = 0;
+ seq_mode = SEQ_2;
+}
+
+int sequencer_open(int dev, struct file *file)
+{
+ int retval, mode, i;
+ int level, tmp;
+
+ if (!sequencer_ok)
+ sequencer_init();
+
+ level = ((dev & 0x0f) == SND_DEV_SEQ2) ? 2 : 1;
+
+ dev = dev >> 4;
+ mode = translate_mode(file);
+
+ DEB(printk("sequencer_open(dev=%d)\n", dev));
+
+ if (!sequencer_ok)
+ {
+/* printk("Sound card: sequencer not initialized\n");*/
+ return -ENXIO;
+ }
+ if (dev) /* Patch manager device (obsolete) */
+ return -ENXIO;
+
+ if(synth_devs[dev] == NULL)
+ request_module("synth0");
+
+ if (mode == OPEN_READ)
+ {
+ if (!num_midis)
+ {
+ /*printk("Sequencer: No MIDI devices. Input not possible\n");*/
+ sequencer_busy = 0;
+ return -ENXIO;
+ }
+ }
+ if (sequencer_busy)
+ {
+ return -EBUSY;
+ }
+ sequencer_busy = 1;
+ obsolete_api_used = 0;
+
+ max_mididev = num_midis;
+ max_synthdev = num_synths;
+ pre_event_timeout = MAX_SCHEDULE_TIMEOUT;
+ seq_mode = SEQ_1;
+
+ if (pending_timer != -1)
+ {
+ tmr_no = pending_timer;
+ pending_timer = -1;
+ }
+ if (tmr_no == -1) /* Not selected yet */
+ {
+ int i, best;
+
+ best = -1;
+ for (i = 0; i < num_sound_timers; i++)
+ if (sound_timer_devs[i] && sound_timer_devs[i]->priority > best)
+ {
+ tmr_no = i;
+ best = sound_timer_devs[i]->priority;
+ }
+ if (tmr_no == -1) /* Should not be */
+ tmr_no = 0;
+ }
+ tmr = sound_timer_devs[tmr_no];
+
+ if (level == 2)
+ {
+ if (tmr == NULL)
+ {
+ /*printk("sequencer: No timer for level 2\n");*/
+ sequencer_busy = 0;
+ return -ENXIO;
+ }
+ setup_mode2();
+ }
+ if (!max_synthdev && !max_mididev)
+ {
+ sequencer_busy=0;
+ return -ENXIO;
+ }
+
+ synth_open_mask = 0;
+
+ for (i = 0; i < max_mididev; i++)
+ {
+ midi_opened[i] = 0;
+ midi_written[i] = 0;
+ }
+
+ for (i = 0; i < max_synthdev; i++)
+ {
+ if (synth_devs[i]==NULL)
+ continue;
+
+ if (!try_module_get(synth_devs[i]->owner))
+ continue;
+
+ if ((tmp = synth_devs[i]->open(i, mode)) < 0)
+ {
+ printk(KERN_WARNING "Sequencer: Warning! Cannot open synth device #%d (%d)\n", i, tmp);
+ if (synth_devs[i]->midi_dev)
+ printk(KERN_WARNING "(Maps to MIDI dev #%d)\n", synth_devs[i]->midi_dev);
+ }
+ else
+ {
+ synth_open_mask |= (1 << i);
+ if (synth_devs[i]->midi_dev)
+ midi_opened[synth_devs[i]->midi_dev] = 1;
+ }
+ }
+
+ seq_time = jiffies;
+
+ prev_input_time = 0;
+ prev_event_time = 0;
+
+ if (seq_mode == SEQ_1 && (mode == OPEN_READ || mode == OPEN_READWRITE))
+ {
+ /*
+ * Initialize midi input devices
+ */
+
+ for (i = 0; i < max_mididev; i++)
+ if (!midi_opened[i] && midi_devs[i])
+ {
+ if (!try_module_get(midi_devs[i]->owner))
+ continue;
+
+ if ((retval = midi_devs[i]->open(i, mode,
+ sequencer_midi_input, sequencer_midi_output)) >= 0)
+ {
+ midi_opened[i] = 1;
+ }
+ }
+ }
+
+ if (seq_mode == SEQ_2) {
+ if (try_module_get(tmr->owner))
+ tmr->open(tmr_no, seq_mode);
+ }
+
+ init_waitqueue_head(&seq_sleeper);
+ init_waitqueue_head(&midi_sleeper);
+ output_threshold = SEQ_MAX_QUEUE / 2;
+
+ return 0;
+}
+
+static void seq_drain_midi_queues(void)
+{
+ int i, n;
+
+ /*
+ * Give the Midi drivers time to drain their output queues
+ */
+
+ n = 1;
+
+ while (!signal_pending(current) && n)
+ {
+ n = 0;
+
+ for (i = 0; i < max_mididev; i++)
+ if (midi_opened[i] && midi_written[i])
+ if (midi_devs[i]->buffer_status != NULL)
+ if (midi_devs[i]->buffer_status(i))
+ n++;
+
+ /*
+ * Let's have a delay
+ */
+
+ if (n)
+ interruptible_sleep_on_timeout(&seq_sleeper,
+ HZ/10);
+ }
+}
+
+void sequencer_release(int dev, struct file *file)
+{
+ int i;
+ int mode = translate_mode(file);
+
+ dev = dev >> 4;
+
+ DEB(printk("sequencer_release(dev=%d)\n", dev));
+
+ /*
+ * Wait until the queue is empty (if we don't have nonblock)
+ */
+
+ if (mode != OPEN_READ && !(file->f_flags & O_NONBLOCK))
+ {
+ while (!signal_pending(current) && qlen > 0)
+ {
+ seq_sync();
+ interruptible_sleep_on_timeout(&seq_sleeper,
+ 3*HZ);
+ /* Extra delay */
+ }
+ }
+
+ if (mode != OPEN_READ)
+ seq_drain_midi_queues(); /*
+ * Ensure the output queues are empty
+ */
+ seq_reset();
+ if (mode != OPEN_READ)
+ seq_drain_midi_queues(); /*
+ * Flush the all notes off messages
+ */
+
+ for (i = 0; i < max_synthdev; i++)
+ {
+ if (synth_open_mask & (1 << i)) /*
+ * Actually opened
+ */
+ if (synth_devs[i])
+ {
+ synth_devs[i]->close(i);
+
+ module_put(synth_devs[i]->owner);
+
+ if (synth_devs[i]->midi_dev)
+ midi_opened[synth_devs[i]->midi_dev] = 0;
+ }
+ }
+
+ for (i = 0; i < max_mididev; i++)
+ {
+ if (midi_opened[i]) {
+ midi_devs[i]->close(i);
+ module_put(midi_devs[i]->owner);
+ }
+ }
+
+ if (seq_mode == SEQ_2) {
+ tmr->close(tmr_no);
+ module_put(tmr->owner);
+ }
+
+ if (obsolete_api_used)
+ printk(KERN_WARNING "/dev/music: Obsolete (4 byte) API was used by %s\n", current->comm);
+ sequencer_busy = 0;
+}
+
+static int seq_sync(void)
+{
+ if (qlen && !seq_playing && !signal_pending(current))
+ seq_startplay();
+
+ if (qlen > 0)
+ interruptible_sleep_on_timeout(&seq_sleeper, HZ);
+ return qlen;
+}
+
+static void midi_outc(int dev, unsigned char data)
+{
+ /*
+ * NOTE! Calls sleep(). Don't call this from interrupt.
+ */
+
+ int n;
+ unsigned long flags;
+
+ /*
+ * This routine sends one byte to the Midi channel.
+ * If the output FIFO is full, it waits until there
+ * is space in the queue
+ */
+
+ n = 3 * HZ; /* Timeout */
+
+ spin_lock_irqsave(&lock,flags);
+ while (n && !midi_devs[dev]->outputc(dev, data)) {
+ interruptible_sleep_on_timeout(&seq_sleeper, HZ/25);
+ n--;
+ }
+ spin_unlock_irqrestore(&lock,flags);
+}
+
+static void seq_reset(void)
+{
+ /*
+ * NOTE! Calls sleep(). Don't call this from interrupt.
+ */
+
+ int i;
+ int chn;
+ unsigned long flags;
+
+ sound_stop_timer();
+
+ seq_time = jiffies;
+ prev_input_time = 0;
+ prev_event_time = 0;
+
+ qlen = qhead = qtail = 0;
+ iqlen = iqhead = iqtail = 0;
+
+ for (i = 0; i < max_synthdev; i++)
+ if (synth_open_mask & (1 << i))
+ if (synth_devs[i])
+ synth_devs[i]->reset(i);
+
+ if (seq_mode == SEQ_2)
+ {
+ for (chn = 0; chn < 16; chn++)
+ for (i = 0; i < max_synthdev; i++)
+ if (synth_open_mask & (1 << i))
+ if (synth_devs[i])
+ {
+ synth_devs[i]->controller(i, chn, 123, 0); /* All notes off */
+ synth_devs[i]->controller(i, chn, 121, 0); /* Reset all ctl */
+ synth_devs[i]->bender(i, chn, 1 << 13); /* Bender off */
+ }
+ }
+ else /* seq_mode == SEQ_1 */
+ {
+ for (i = 0; i < max_mididev; i++)
+ if (midi_written[i]) /*
+ * Midi used. Some notes may still be playing
+ */
+ {
+ /*
+ * Sending just a ACTIVE SENSING message should be enough to stop all
+ * playing notes. Since there are devices not recognizing the
+ * active sensing, we have to send some all notes off messages also.
+ */
+ midi_outc(i, 0xfe);
+
+ for (chn = 0; chn < 16; chn++)
+ {
+ midi_outc(i, (unsigned char) (0xb0 + (chn & 0x0f))); /* control change */
+ midi_outc(i, 0x7b); /* All notes off */
+ midi_outc(i, 0); /* Dummy parameter */
+ }
+
+ midi_devs[i]->close(i);
+
+ midi_written[i] = 0;
+ midi_opened[i] = 0;
+ }
+ }
+
+ seq_playing = 0;
+
+ spin_lock_irqsave(&lock,flags);
+
+ if (waitqueue_active(&seq_sleeper)) {
+ /* printk( "Sequencer Warning: Unexpected sleeping process - Waking up\n"); */
+ wake_up(&seq_sleeper);
+ }
+ spin_unlock_irqrestore(&lock,flags);
+}
+
+static void seq_panic(void)
+{
+ /*
+ * This routine is called by the application in case the user
+ * wants to reset the system to the default state.
+ */
+
+ seq_reset();
+
+ /*
+ * Since some of the devices don't recognize the active sensing and
+ * all notes off messages, we have to shut all notes manually.
+ *
+ * TO BE IMPLEMENTED LATER
+ */
+
+ /*
+ * Also return the controllers to their default states
+ */
+}
+
+int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg)
+{
+ int midi_dev, orig_dev, val, err;
+ int mode = translate_mode(file);
+ struct synth_info inf;
+ struct seq_event_rec event_rec;
+ unsigned long flags;
+ int __user *p = arg;
+
+ orig_dev = dev = dev >> 4;
+
+ switch (cmd)
+ {
+ case SNDCTL_TMR_TIMEBASE:
+ case SNDCTL_TMR_TEMPO:
+ case SNDCTL_TMR_START:
+ case SNDCTL_TMR_STOP:
+ case SNDCTL_TMR_CONTINUE:
+ case SNDCTL_TMR_METRONOME:
+ case SNDCTL_TMR_SOURCE:
+ if (seq_mode != SEQ_2)
+ return -EINVAL;
+ return tmr->ioctl(tmr_no, cmd, arg);
+
+ case SNDCTL_TMR_SELECT:
+ if (seq_mode != SEQ_2)
+ return -EINVAL;
+ if (get_user(pending_timer, p))
+ return -EFAULT;
+ if (pending_timer < 0 || pending_timer >= num_sound_timers || sound_timer_devs[pending_timer] == NULL)
+ {
+ pending_timer = -1;
+ return -EINVAL;
+ }
+ val = pending_timer;
+ break;
+
+ case SNDCTL_SEQ_PANIC:
+ seq_panic();
+ return -EINVAL;
+
+ case SNDCTL_SEQ_SYNC:
+ if (mode == OPEN_READ)
+ return 0;
+ while (qlen > 0 && !signal_pending(current))
+ seq_sync();
+ return qlen ? -EINTR : 0;
+
+ case SNDCTL_SEQ_RESET:
+ seq_reset();
+ return 0;
+
+ case SNDCTL_SEQ_TESTMIDI:
+ if (__get_user(midi_dev, p))
+ return -EFAULT;
+ if (midi_dev < 0 || midi_dev >= max_mididev || !midi_devs[midi_dev])
+ return -ENXIO;
+
+ if (!midi_opened[midi_dev] &&
+ (err = midi_devs[midi_dev]->open(midi_dev, mode, sequencer_midi_input,
+ sequencer_midi_output)) < 0)
+ return err;
+ midi_opened[midi_dev] = 1;
+ return 0;
+
+ case SNDCTL_SEQ_GETINCOUNT:
+ if (mode == OPEN_WRITE)
+ return 0;
+ val = iqlen;
+ break;
+
+ case SNDCTL_SEQ_GETOUTCOUNT:
+ if (mode == OPEN_READ)
+ return 0;
+ val = SEQ_MAX_QUEUE - qlen;
+ break;
+
+ case SNDCTL_SEQ_GETTIME:
+ if (seq_mode == SEQ_2)
+ return tmr->ioctl(tmr_no, cmd, arg);
+ val = jiffies - seq_time;
+ break;
+
+ case SNDCTL_SEQ_CTRLRATE:
+ /*
+ * If *arg == 0, just return the current rate
+ */
+ if (seq_mode == SEQ_2)
+ return tmr->ioctl(tmr_no, cmd, arg);
+
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val != 0)
+ return -EINVAL;
+ val = HZ;
+ break;
+
+ case SNDCTL_SEQ_RESETSAMPLES:
+ case SNDCTL_SYNTH_REMOVESAMPLE:
+ case SNDCTL_SYNTH_CONTROL:
+ if (get_user(dev, p))
+ return -EFAULT;
+ if (dev < 0 || dev >= num_synths || synth_devs[dev] == NULL)
+ return -ENXIO;
+ if (!(synth_open_mask & (1 << dev)) && !orig_dev)
+ return -EBUSY;
+ return synth_devs[dev]->ioctl(dev, cmd, arg);
+
+ case SNDCTL_SEQ_NRSYNTHS:
+ val = max_synthdev;
+ break;
+
+ case SNDCTL_SEQ_NRMIDIS:
+ val = max_mididev;
+ break;
+
+ case SNDCTL_SYNTH_MEMAVL:
+ if (get_user(dev, p))
+ return -EFAULT;
+ if (dev < 0 || dev >= num_synths || synth_devs[dev] == NULL)
+ return -ENXIO;
+ if (!(synth_open_mask & (1 << dev)) && !orig_dev)
+ return -EBUSY;
+ val = synth_devs[dev]->ioctl(dev, cmd, arg);
+ break;
+
+ case SNDCTL_FM_4OP_ENABLE:
+ if (get_user(dev, p))
+ return -EFAULT;
+ if (dev < 0 || dev >= num_synths || synth_devs[dev] == NULL)
+ return -ENXIO;
+ if (!(synth_open_mask & (1 << dev)))
+ return -ENXIO;
+ synth_devs[dev]->ioctl(dev, cmd, arg);
+ return 0;
+
+ case SNDCTL_SYNTH_INFO:
+ if (get_user(dev, &((struct synth_info __user *)arg)->device))
+ return -EFAULT;
+ if (dev < 0 || dev >= max_synthdev)
+ return -ENXIO;
+ if (!(synth_open_mask & (1 << dev)) && !orig_dev)
+ return -EBUSY;
+ return synth_devs[dev]->ioctl(dev, cmd, arg);
+
+ /* Like SYNTH_INFO but returns ID in the name field */
+ case SNDCTL_SYNTH_ID:
+ if (get_user(dev, &((struct synth_info __user *)arg)->device))
+ return -EFAULT;
+ if (dev < 0 || dev >= max_synthdev)
+ return -ENXIO;
+ if (!(synth_open_mask & (1 << dev)) && !orig_dev)
+ return -EBUSY;
+ memcpy(&inf, synth_devs[dev]->info, sizeof(inf));
+ strlcpy(inf.name, synth_devs[dev]->id, sizeof(inf.name));
+ inf.device = dev;
+ return copy_to_user(arg, &inf, sizeof(inf))?-EFAULT:0;
+
+ case SNDCTL_SEQ_OUTOFBAND:
+ if (copy_from_user(&event_rec, arg, sizeof(event_rec)))
+ return -EFAULT;
+ spin_lock_irqsave(&lock,flags);
+ play_event(event_rec.arr);
+ spin_unlock_irqrestore(&lock,flags);
+ return 0;
+
+ case SNDCTL_MIDI_INFO:
+ if (get_user(dev, &((struct midi_info __user *)arg)->device))
+ return -EFAULT;
+ if (dev < 0 || dev >= max_mididev || !midi_devs[dev])
+ return -ENXIO;
+ midi_devs[dev]->info.device = dev;
+ return copy_to_user(arg, &midi_devs[dev]->info, sizeof(struct midi_info))?-EFAULT:0;
+
+ case SNDCTL_SEQ_THRESHOLD:
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val < 1)
+ val = 1;
+ if (val >= SEQ_MAX_QUEUE)
+ val = SEQ_MAX_QUEUE - 1;
+ output_threshold = val;
+ return 0;
+
+ case SNDCTL_MIDI_PRETIME:
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val < 0)
+ val = 0;
+ val = (HZ * val) / 10;
+ pre_event_timeout = val;
+ break;
+
+ default:
+ if (mode == OPEN_READ)
+ return -EIO;
+ if (!synth_devs[0])
+ return -ENXIO;
+ if (!(synth_open_mask & (1 << 0)))
+ return -ENXIO;
+ if (!synth_devs[0]->ioctl)
+ return -EINVAL;
+ return synth_devs[0]->ioctl(0, cmd, arg);
+ }
+ return put_user(val, p);
+}
+
+/* No kernel lock - we're using the global irq lock here */
+unsigned int sequencer_poll(int dev, struct file *file, poll_table * wait)
+{
+ unsigned long flags;
+ unsigned int mask = 0;
+
+ dev = dev >> 4;
+
+ spin_lock_irqsave(&lock,flags);
+ /* input */
+ poll_wait(file, &midi_sleeper, wait);
+ if (iqlen)
+ mask |= POLLIN | POLLRDNORM;
+
+ /* output */
+ poll_wait(file, &seq_sleeper, wait);
+ if ((SEQ_MAX_QUEUE - qlen) >= output_threshold)
+ mask |= POLLOUT | POLLWRNORM;
+ spin_unlock_irqrestore(&lock,flags);
+ return mask;
+}
+
+
+void sequencer_timer(unsigned long dummy)
+{
+ seq_startplay();
+}
+EXPORT_SYMBOL(sequencer_timer);
+
+int note_to_freq(int note_num)
+{
+
+ /*
+ * This routine converts a midi note to a frequency (multiplied by 1000)
+ */
+
+ int note, octave, note_freq;
+ static int notes[] =
+ {
+ 261632, 277189, 293671, 311132, 329632, 349232,
+ 369998, 391998, 415306, 440000, 466162, 493880
+ };
+
+#define BASE_OCTAVE 5
+
+ octave = note_num / 12;
+ note = note_num % 12;
+
+ note_freq = notes[note];
+
+ if (octave < BASE_OCTAVE)
+ note_freq >>= (BASE_OCTAVE - octave);
+ else if (octave > BASE_OCTAVE)
+ note_freq <<= (octave - BASE_OCTAVE);
+
+ /*
+ * note_freq >>= 1;
+ */
+
+ return note_freq;
+}
+EXPORT_SYMBOL(note_to_freq);
+
+unsigned long compute_finetune(unsigned long base_freq, int bend, int range,
+ int vibrato_cents)
+{
+ unsigned long amount;
+ int negative, semitones, cents, multiplier = 1;
+
+ if (!bend)
+ return base_freq;
+ if (!range)
+ return base_freq;
+
+ if (!base_freq)
+ return base_freq;
+
+ if (range >= 8192)
+ range = 8192;
+
+ bend = bend * range / 8192; /* Convert to cents */
+ bend += vibrato_cents;
+
+ if (!bend)
+ return base_freq;
+
+ negative = bend < 0 ? 1 : 0;
+
+ if (bend < 0)
+ bend *= -1;
+ if (bend > range)
+ bend = range;
+
+ /*
+ if (bend > 2399)
+ bend = 2399;
+ */
+ while (bend > 2399)
+ {
+ multiplier *= 4;
+ bend -= 2400;
+ }
+
+ semitones = bend / 100;
+ if (semitones > 99)
+ semitones = 99;
+ cents = bend % 100;
+
+ amount = (int) (semitone_tuning[semitones] * multiplier * cent_tuning[cents]) / 10000;
+
+ if (negative)
+ return (base_freq * 10000) / amount; /* Bend down */
+ else
+ return (base_freq * amount) / 10000; /* Bend up */
+}
+EXPORT_SYMBOL(compute_finetune);
+
+void sequencer_init(void)
+{
+ if (sequencer_ok)
+ return;
+ queue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * EV_SZ);
+ if (queue == NULL)
+ {
+ printk(KERN_ERR "sequencer: Can't allocate memory for sequencer output queue\n");
+ return;
+ }
+ iqueue = (unsigned char *)vmalloc(SEQ_MAX_QUEUE * IEV_SZ);
+ if (iqueue == NULL)
+ {
+ printk(KERN_ERR "sequencer: Can't allocate memory for sequencer input queue\n");
+ vfree(queue);
+ return;
+ }
+ sequencer_ok = 1;
+}
+EXPORT_SYMBOL(sequencer_init);
+
+void sequencer_unload(void)
+{
+ vfree(queue);
+ vfree(iqueue);
+ queue = iqueue = NULL;
+}
diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c
new file mode 100644
index 0000000..e5d4239
--- /dev/null
+++ b/sound/oss/sh_dac_audio.c
@@ -0,0 +1,331 @@
+/*
+ * sound/oss/sh_dac_audio.c
+ *
+ * SH DAC based sound :(
+ *
+ * Copyright (C) 2004,2005 Andriy Skulysh
+ *
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License. See the file "COPYING" in the main directory of this archive
+ * for more details.
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/sched.h>
+#include <linux/linkage.h>
+#include <linux/slab.h>
+#include <linux/fs.h>
+#include <linux/sound.h>
+#include <linux/soundcard.h>
+#include <linux/interrupt.h>
+#include <asm/io.h>
+#include <asm/uaccess.h>
+#include <asm/irq.h>
+#include <asm/delay.h>
+#include <asm/clock.h>
+#include <asm/cpu/dac.h>
+#include <asm/cpu/timer.h>
+#include <asm/machvec.h>
+#include <mach/hp6xx.h>
+#include <asm/hd64461.h>
+
+#define MODNAME "sh_dac_audio"
+
+#define TMU_TOCR_INIT 0x00
+
+#define TMU1_TCR_INIT 0x0020 /* Clock/4, rising edge; interrupt on */
+#define TMU1_TSTR_INIT 0x02 /* Bit to turn on TMU1 */
+
+#define BUFFER_SIZE 48000
+
+static int rate;
+static int empty;
+static char *data_buffer, *buffer_begin, *buffer_end;
+static int in_use, device_major;
+
+static void dac_audio_start_timer(void)
+{
+ u8 tstr;
+
+ tstr = ctrl_inb(TMU_TSTR);
+ tstr |= TMU1_TSTR_INIT;
+ ctrl_outb(tstr, TMU_TSTR);
+}
+
+static void dac_audio_stop_timer(void)
+{
+ u8 tstr;
+
+ tstr = ctrl_inb(TMU_TSTR);
+ tstr &= ~TMU1_TSTR_INIT;
+ ctrl_outb(tstr, TMU_TSTR);
+}
+
+static void dac_audio_reset(void)
+{
+ dac_audio_stop_timer();
+ buffer_begin = buffer_end = data_buffer;
+ empty = 1;
+}
+
+static void dac_audio_sync(void)
+{
+ while (!empty)
+ schedule();
+}
+
+static void dac_audio_start(void)
+{
+ if (mach_is_hp6xx()) {
+ u16 v = inw(HD64461_GPADR);
+ v &= ~HD64461_GPADR_SPEAKER;
+ outw(v, HD64461_GPADR);
+ }
+
+ sh_dac_enable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
+ ctrl_outw(TMU1_TCR_INIT, TMU1_TCR);
+}
+static void dac_audio_stop(void)
+{
+ dac_audio_stop_timer();
+
+ if (mach_is_hp6xx()) {
+ u16 v = inw(HD64461_GPADR);
+ v |= HD64461_GPADR_SPEAKER;
+ outw(v, HD64461_GPADR);
+ }
+
+ sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
+ sh_dac_disable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
+}
+
+static void dac_audio_set_rate(void)
+{
+ unsigned long interval;
+ struct clk *clk;
+
+ clk = clk_get(NULL, "module_clk");
+ interval = (clk_get_rate(clk) / 4) / rate;
+ clk_put(clk);
+ ctrl_outl(interval, TMU1_TCOR);
+ ctrl_outl(interval, TMU1_TCNT);
+}
+
+static int dac_audio_ioctl(struct inode *inode, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ int val;
+
+ switch (cmd) {
+ case OSS_GETVERSION:
+ return put_user(SOUND_VERSION, (int *)arg);
+
+ case SNDCTL_DSP_SYNC:
+ dac_audio_sync();
+ return 0;
+
+ case SNDCTL_DSP_RESET:
+ dac_audio_reset();
+ return 0;
+
+ case SNDCTL_DSP_GETFMTS:
+ return put_user(AFMT_U8, (int *)arg);
+
+ case SNDCTL_DSP_SETFMT:
+ return put_user(AFMT_U8, (int *)arg);
+
+ case SNDCTL_DSP_NONBLOCK:
+ file->f_flags |= O_NONBLOCK;
+ return 0;
+
+ case SNDCTL_DSP_GETCAPS:
+ return 0;
+
+ case SOUND_PCM_WRITE_RATE:
+ val = *(int *)arg;
+ if (val > 0) {
+ rate = val;
+ dac_audio_set_rate();
+ }
+ return put_user(rate, (int *)arg);
+
+ case SNDCTL_DSP_STEREO:
+ return put_user(0, (int *)arg);
+
+ case SOUND_PCM_WRITE_CHANNELS:
+ return put_user(1, (int *)arg);
+
+ case SNDCTL_DSP_SETDUPLEX:
+ return -EINVAL;
+
+ case SNDCTL_DSP_PROFILE:
+ return -EINVAL;
+
+ case SNDCTL_DSP_GETBLKSIZE:
+ return put_user(BUFFER_SIZE, (int *)arg);
+
+ case SNDCTL_DSP_SETFRAGMENT:
+ return 0;
+
+ default:
+ printk(KERN_ERR "sh_dac_audio: unimplemented ioctl=0x%x\n",
+ cmd);
+ return -EINVAL;
+ }
+ return -EINVAL;
+}
+
+static ssize_t dac_audio_write(struct file *file, const char *buf, size_t count,
+ loff_t * ppos)
+{
+ int free;
+ int nbytes;
+
+ if (count < 0)
+ return -EINVAL;
+
+ if (!count) {
+ dac_audio_sync();
+ return 0;
+ }
+
+ free = buffer_begin - buffer_end;
+
+ if (free < 0)
+ free += BUFFER_SIZE;
+ if ((free == 0) && (empty))
+ free = BUFFER_SIZE;
+ if (count > free)
+ count = free;
+ if (buffer_begin > buffer_end) {
+ if (copy_from_user((void *)buffer_end, buf, count))
+ return -EFAULT;
+
+ buffer_end += count;
+ } else {
+ nbytes = data_buffer + BUFFER_SIZE - buffer_end;
+ if (nbytes > count) {
+ if (copy_from_user((void *)buffer_end, buf, count))
+ return -EFAULT;
+ buffer_end += count;
+ } else {
+ if (copy_from_user((void *)buffer_end, buf, nbytes))
+ return -EFAULT;
+ if (copy_from_user
+ ((void *)data_buffer, buf + nbytes, count - nbytes))
+ return -EFAULT;
+ buffer_end = data_buffer + count - nbytes;
+ }
+ }
+
+ if (empty) {
+ empty = 0;
+ dac_audio_start_timer();
+ }
+
+ return count;
+}
+
+static ssize_t dac_audio_read(struct file *file, char *buf, size_t count,
+ loff_t * ppos)
+{
+ return -EINVAL;
+}
+
+static int dac_audio_open(struct inode *inode, struct file *file)
+{
+ if (file->f_mode & FMODE_READ)
+ return -ENODEV;
+ if (in_use)
+ return -EBUSY;
+
+ in_use = 1;
+
+ dac_audio_start();
+
+ return 0;
+}
+
+static int dac_audio_release(struct inode *inode, struct file *file)
+{
+ dac_audio_sync();
+ dac_audio_stop();
+ in_use = 0;
+
+ return 0;
+}
+
+const struct file_operations dac_audio_fops = {
+ .read = dac_audio_read,
+ .write = dac_audio_write,
+ .ioctl = dac_audio_ioctl,
+ .open = dac_audio_open,
+ .release = dac_audio_release,
+};
+
+static irqreturn_t timer1_interrupt(int irq, void *dev)
+{
+ unsigned long timer_status;
+
+ timer_status = ctrl_inw(TMU1_TCR);
+ timer_status &= ~0x100;
+ ctrl_outw(timer_status, TMU1_TCR);
+
+ if (!empty) {
+ sh_dac_output(*buffer_begin, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
+ buffer_begin++;
+
+ if (buffer_begin == data_buffer + BUFFER_SIZE)
+ buffer_begin = data_buffer;
+ if (buffer_begin == buffer_end) {
+ empty = 1;
+ dac_audio_stop_timer();
+ }
+ }
+ return IRQ_HANDLED;
+}
+
+static int __init dac_audio_init(void)
+{
+ int retval;
+
+ if ((device_major = register_sound_dsp(&dac_audio_fops, -1)) < 0) {
+ printk(KERN_ERR "Cannot register dsp device");
+ return device_major;
+ }
+
+ in_use = 0;
+
+ data_buffer = kmalloc(BUFFER_SIZE, GFP_KERNEL);
+ if (data_buffer == NULL)
+ return -ENOMEM;
+
+ dac_audio_reset();
+ rate = 8000;
+ dac_audio_set_rate();
+
+ retval =
+ request_irq(TIMER1_IRQ, timer1_interrupt, IRQF_DISABLED, MODNAME, 0);
+ if (retval < 0) {
+ printk(KERN_ERR "sh_dac_audio: IRQ %d request failed\n",
+ TIMER1_IRQ);
+ return retval;
+ }
+
+ return 0;
+}
+
+static void __exit dac_audio_exit(void)
+{
+ free_irq(TIMER1_IRQ, 0);
+
+ unregister_sound_dsp(device_major);
+ kfree((void *)data_buffer);
+}
+
+module_init(dac_audio_init);
+module_exit(dac_audio_exit);
+
+MODULE_AUTHOR("Andriy Skulysh, askulysh@image.kiev.ua");
+MODULE_DESCRIPTION("SH DAC sound driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/sound_calls.h b/sound/oss/sound_calls.h
new file mode 100644
index 0000000..87d8ad4
--- /dev/null
+++ b/sound/oss/sound_calls.h
@@ -0,0 +1,87 @@
+/*
+ * DMA buffer calls
+ */
+
+int DMAbuf_open(int dev, int mode);
+int DMAbuf_release(int dev, int mode);
+int DMAbuf_getwrbuffer(int dev, char **buf, int *size, int dontblock);
+int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock);
+int DMAbuf_rmchars(int dev, int buff_no, int c);
+int DMAbuf_start_output(int dev, int buff_no, int l);
+int DMAbuf_move_wrpointer(int dev, int l);
+/* int DMAbuf_ioctl(int dev, unsigned int cmd, void __user *arg, int local); */
+void DMAbuf_init(int dev, int dma1, int dma2);
+void DMAbuf_deinit(int dev);
+int DMAbuf_start_dma (int dev, unsigned long physaddr, int count, int dma_mode);
+void DMAbuf_inputintr(int dev);
+void DMAbuf_outputintr(int dev, int underflow_flag);
+struct dma_buffparms;
+int DMAbuf_space_in_queue (int dev);
+int DMAbuf_activate_recording (int dev, struct dma_buffparms *dmap);
+int DMAbuf_get_buffer_pointer (int dev, struct dma_buffparms *dmap, int direction);
+void DMAbuf_launch_output(int dev, struct dma_buffparms *dmap);
+unsigned int DMAbuf_poll(struct file *file, int dev, poll_table *wait);
+void DMAbuf_start_devices(unsigned int devmask);
+void DMAbuf_reset (int dev);
+int DMAbuf_sync (int dev);
+
+/*
+ * System calls for /dev/dsp and /dev/audio (audio.c)
+ */
+
+int audio_read (int dev, struct file *file, char __user *buf, int count);
+int audio_write (int dev, struct file *file, const char __user *buf, int count);
+int audio_open (int dev, struct file *file);
+void audio_release (int dev, struct file *file);
+int audio_ioctl (int dev, struct file *file,
+ unsigned int cmd, void __user *arg);
+void audio_init_devices (void);
+void reorganize_buffers (int dev, struct dma_buffparms *dmap, int recording);
+
+/*
+ * System calls for the /dev/sequencer
+ */
+
+int sequencer_read (int dev, struct file *file, char __user *buf, int count);
+int sequencer_write (int dev, struct file *file, const char __user *buf, int count);
+int sequencer_open (int dev, struct file *file);
+void sequencer_release (int dev, struct file *file);
+int sequencer_ioctl (int dev, struct file *file, unsigned int cmd, void __user *arg);
+unsigned int sequencer_poll(int dev, struct file *file, poll_table * wait);
+
+void sequencer_init (void);
+void sequencer_unload (void);
+void sequencer_timer(unsigned long dummy);
+int note_to_freq(int note_num);
+unsigned long compute_finetune(unsigned long base_freq, int bend, int range,
+ int vibrato_bend);
+void seq_input_event(unsigned char *event, int len);
+void seq_copy_to_input (unsigned char *event, int len);
+
+/*
+ * System calls for the /dev/midi
+ */
+
+int MIDIbuf_read (int dev, struct file *file, char __user *buf, int count);
+int MIDIbuf_write (int dev, struct file *file, const char __user *buf, int count);
+int MIDIbuf_open (int dev, struct file *file);
+void MIDIbuf_release (int dev, struct file *file);
+int MIDIbuf_ioctl (int dev, struct file *file, unsigned int cmd, void __user *arg);
+unsigned int MIDIbuf_poll(int dev, struct file *file, poll_table * wait);
+int MIDIbuf_avail(int dev);
+
+void MIDIbuf_bytes_received(int dev, unsigned char *buf, int count);
+
+
+/* From soundcard.c */
+void request_sound_timer (int count);
+void sound_stop_timer(void);
+void conf_printf(char *name, struct address_info *hw_config);
+void conf_printf2(char *name, int base, int irq, int dma, int dma2);
+
+/* From sound_timer.c */
+void sound_timer_interrupt(void);
+void sound_timer_syncinterval(unsigned int new_usecs);
+
+/* From midi_synth.c */
+void do_midi_msg (int synthno, unsigned char *msg, int mlen);
diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h
new file mode 100644
index 0000000..55271fb
--- /dev/null
+++ b/sound/oss/sound_config.h
@@ -0,0 +1,145 @@
+/* sound_config.h
+ *
+ * A driver for sound cards, misc. configuration parameters.
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+
+
+#ifndef _SOUND_CONFIG_H_
+#define _SOUND_CONFIG_H_
+
+#include <linux/fs.h>
+#include <linux/sound.h>
+
+#include "os.h"
+#include "soundvers.h"
+
+
+#ifndef SND_DEFAULT_ENABLE
+#define SND_DEFAULT_ENABLE 1
+#endif
+
+#ifndef MAX_REALTIME_FACTOR
+#define MAX_REALTIME_FACTOR 4
+#endif
+
+/*
+ * Use always 64k buffer size. There is no reason to use shorter.
+ */
+#undef DSP_BUFFSIZE
+#define DSP_BUFFSIZE (64*1024)
+
+#ifndef DSP_BUFFCOUNT
+#define DSP_BUFFCOUNT 1 /* 1 is recommended. */
+#endif
+
+#define FM_MONO 0x388 /* This is the I/O address used by AdLib */
+
+#ifndef CONFIG_PAS_BASE
+#define CONFIG_PAS_BASE 0x388
+#endif
+
+/* SEQ_MAX_QUEUE is the maximum number of sequencer events buffered by the
+ driver. (There is no need to alter this) */
+#define SEQ_MAX_QUEUE 1024
+
+#define SBFM_MAXINSTR (256) /* Size of the FM Instrument bank */
+/* 128 instruments for general MIDI setup and 16 unassigned */
+
+#define SND_NDEVS 256 /* Number of supported devices */
+
+#define DSP_DEFAULT_SPEED 8000
+
+#define MAX_AUDIO_DEV 5
+#define MAX_MIXER_DEV 5
+#define MAX_SYNTH_DEV 5
+#define MAX_MIDI_DEV 6
+#define MAX_TIMER_DEV 4
+
+struct address_info {
+ int io_base;
+ int irq;
+ int dma;
+ int dma2;
+ int always_detect; /* 1=Trust me, it's there */
+ char *name;
+ int driver_use_1; /* Driver defined field 1 */
+ int driver_use_2; /* Driver defined field 2 */
+ int *osp; /* OS specific info */
+ int card_subtype; /* Driver specific. Usually 0 */
+ void *memptr; /* Module memory chainer */
+ int slots[6]; /* To remember driver slot ids */
+};
+
+#define SYNTH_MAX_VOICES 32
+
+struct voice_alloc_info {
+ int max_voice;
+ int used_voices;
+ int ptr; /* For device specific use */
+ unsigned short map[SYNTH_MAX_VOICES]; /* (ch << 8) | (note+1) */
+ int timestamp;
+ int alloc_times[SYNTH_MAX_VOICES];
+ };
+
+struct channel_info {
+ int pgm_num;
+ int bender_value;
+ int bender_range;
+ unsigned char controllers[128];
+ };
+
+/*
+ * Process wakeup reasons
+ */
+#define WK_NONE 0x00
+#define WK_WAKEUP 0x01
+#define WK_TIMEOUT 0x02
+#define WK_SIGNAL 0x04
+#define WK_SLEEP 0x08
+#define WK_SELECT 0x10
+#define WK_ABORT 0x20
+
+#define OPEN_READ PCM_ENABLE_INPUT
+#define OPEN_WRITE PCM_ENABLE_OUTPUT
+#define OPEN_READWRITE (OPEN_READ|OPEN_WRITE)
+
+static inline int translate_mode(struct file *file)
+{
+ if (OPEN_READ == (__force int)FMODE_READ &&
+ OPEN_WRITE == (__force int)FMODE_WRITE)
+ return (__force int)(file->f_mode & (FMODE_READ | FMODE_WRITE));
+ else
+ return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) |
+ ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0);
+}
+
+#include "sound_calls.h"
+#include "dev_table.h"
+
+#ifndef DEB
+#define DEB(x)
+#endif
+
+#ifndef DDB
+#define DDB(x) do {} while (0)
+#endif
+
+#ifndef MDB
+#ifdef MODULE
+#define MDB(x) x
+#else
+#define MDB(x)
+#endif
+#endif
+
+#define TIMER_ARMED 121234
+#define TIMER_NOT_ARMED 1
+
+#endif
diff --git a/sound/oss/sound_firmware.h b/sound/oss/sound_firmware.h
new file mode 100644
index 0000000..0a0cbfd
--- /dev/null
+++ b/sound/oss/sound_firmware.h
@@ -0,0 +1,2 @@
+extern int mod_firmware_load(const char *fn, char **fp);
+
diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c
new file mode 100644
index 0000000..f0f0c19
--- /dev/null
+++ b/sound/oss/sound_timer.c
@@ -0,0 +1,327 @@
+/*
+ * sound/oss/sound_timer.c
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+/*
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ */
+#include <linux/string.h>
+#include <linux/spinlock.h>
+
+#include "sound_config.h"
+
+static volatile int initialized, opened, tmr_running;
+static volatile time_t tmr_offs, tmr_ctr;
+static volatile unsigned long ticks_offs;
+static volatile int curr_tempo, curr_timebase;
+static volatile unsigned long curr_ticks;
+static volatile unsigned long next_event_time;
+static unsigned long prev_event_time;
+static volatile unsigned long usecs_per_tmr; /* Length of the current interval */
+
+static struct sound_lowlev_timer *tmr;
+static spinlock_t lock;
+
+static unsigned long tmr2ticks(int tmr_value)
+{
+ /*
+ * Convert timer ticks to MIDI ticks
+ */
+
+ unsigned long tmp;
+ unsigned long scale;
+
+ tmp = tmr_value * usecs_per_tmr; /* Convert to usecs */
+ scale = (60 * 1000000) / (curr_tempo * curr_timebase); /* usecs per MIDI tick */
+ return (tmp + (scale / 2)) / scale;
+}
+
+void reprogram_timer(void)
+{
+ unsigned long usecs_per_tick;
+
+ /*
+ * The user is changing the timer rate before setting a timer
+ * slap, bad bad not allowed.
+ */
+
+ if(!tmr)
+ return;
+
+ usecs_per_tick = (60 * 1000000) / (curr_tempo * curr_timebase);
+
+ /*
+ * Don't kill the system by setting too high timer rate
+ */
+ if (usecs_per_tick < 2000)
+ usecs_per_tick = 2000;
+
+ usecs_per_tmr = tmr->tmr_start(tmr->dev, usecs_per_tick);
+}
+
+void sound_timer_syncinterval(unsigned int new_usecs)
+{
+ /*
+ * This routine is called by the hardware level if
+ * the clock frequency has changed for some reason.
+ */
+ tmr_offs = tmr_ctr;
+ ticks_offs += tmr2ticks(tmr_ctr);
+ tmr_ctr = 0;
+ usecs_per_tmr = new_usecs;
+}
+EXPORT_SYMBOL(sound_timer_syncinterval);
+
+static void tmr_reset(void)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&lock,flags);
+ tmr_offs = 0;
+ ticks_offs = 0;
+ tmr_ctr = 0;
+ next_event_time = (unsigned long) -1;
+ prev_event_time = 0;
+ curr_ticks = 0;
+ spin_unlock_irqrestore(&lock,flags);
+}
+
+static int timer_open(int dev, int mode)
+{
+ if (opened)
+ return -EBUSY;
+ tmr_reset();
+ curr_tempo = 60;
+ curr_timebase = 100;
+ opened = 1;
+ reprogram_timer();
+ return 0;
+}
+
+static void timer_close(int dev)
+{
+ opened = tmr_running = 0;
+ tmr->tmr_disable(tmr->dev);
+}
+
+static int timer_event(int dev, unsigned char *event)
+{
+ unsigned char cmd = event[1];
+ unsigned long parm = *(int *) &event[4];
+
+ switch (cmd)
+ {
+ case TMR_WAIT_REL:
+ parm += prev_event_time;
+ case TMR_WAIT_ABS:
+ if (parm > 0)
+ {
+ long time;
+
+ if (parm <= curr_ticks) /* It's the time */
+ return TIMER_NOT_ARMED;
+ time = parm;
+ next_event_time = prev_event_time = time;
+ return TIMER_ARMED;
+ }
+ break;
+
+ case TMR_START:
+ tmr_reset();
+ tmr_running = 1;
+ reprogram_timer();
+ break;
+
+ case TMR_STOP:
+ tmr_running = 0;
+ break;
+
+ case TMR_CONTINUE:
+ tmr_running = 1;
+ reprogram_timer();
+ break;
+
+ case TMR_TEMPO:
+ if (parm)
+ {
+ if (parm < 8)
+ parm = 8;
+ if (parm > 250)
+ parm = 250;
+ tmr_offs = tmr_ctr;
+ ticks_offs += tmr2ticks(tmr_ctr);
+ tmr_ctr = 0;
+ curr_tempo = parm;
+ reprogram_timer();
+ }
+ break;
+
+ case TMR_ECHO:
+ seq_copy_to_input(event, 8);
+ break;
+
+ default:;
+ }
+ return TIMER_NOT_ARMED;
+}
+
+static unsigned long timer_get_time(int dev)
+{
+ if (!opened)
+ return 0;
+ return curr_ticks;
+}
+
+static int timer_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ int __user *p = arg;
+ int val;
+
+ switch (cmd)
+ {
+ case SNDCTL_TMR_SOURCE:
+ val = TMR_INTERNAL;
+ break;
+
+ case SNDCTL_TMR_START:
+ tmr_reset();
+ tmr_running = 1;
+ return 0;
+
+ case SNDCTL_TMR_STOP:
+ tmr_running = 0;
+ return 0;
+
+ case SNDCTL_TMR_CONTINUE:
+ tmr_running = 1;
+ return 0;
+
+ case SNDCTL_TMR_TIMEBASE:
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val)
+ {
+ if (val < 1)
+ val = 1;
+ if (val > 1000)
+ val = 1000;
+ curr_timebase = val;
+ }
+ val = curr_timebase;
+ break;
+
+ case SNDCTL_TMR_TEMPO:
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val)
+ {
+ if (val < 8)
+ val = 8;
+ if (val > 250)
+ val = 250;
+ tmr_offs = tmr_ctr;
+ ticks_offs += tmr2ticks(tmr_ctr);
+ tmr_ctr = 0;
+ curr_tempo = val;
+ reprogram_timer();
+ }
+ val = curr_tempo;
+ break;
+
+ case SNDCTL_SEQ_CTRLRATE:
+ if (get_user(val, p))
+ return -EFAULT;
+ if (val != 0) /* Can't change */
+ return -EINVAL;
+ val = ((curr_tempo * curr_timebase) + 30) / 60;
+ break;
+
+ case SNDCTL_SEQ_GETTIME:
+ val = curr_ticks;
+ break;
+
+ case SNDCTL_TMR_METRONOME:
+ default:
+ return -EINVAL;
+ }
+ return put_user(val, p);
+}
+
+static void timer_arm(int dev, long time)
+{
+ if (time < 0)
+ time = curr_ticks + 1;
+ else if (time <= curr_ticks) /* It's the time */
+ return;
+
+ next_event_time = prev_event_time = time;
+ return;
+}
+
+static struct sound_timer_operations sound_timer =
+{
+ .owner = THIS_MODULE,
+ .info = {"Sound Timer", 0},
+ .priority = 1, /* Priority */
+ .devlink = 0, /* Local device link */
+ .open = timer_open,
+ .close = timer_close,
+ .event = timer_event,
+ .get_time = timer_get_time,
+ .ioctl = timer_ioctl,
+ .arm_timer = timer_arm
+};
+
+void sound_timer_interrupt(void)
+{
+ unsigned long flags;
+
+ if (!opened)
+ return;
+
+ tmr->tmr_restart(tmr->dev);
+
+ if (!tmr_running)
+ return;
+
+ spin_lock_irqsave(&lock,flags);
+ tmr_ctr++;
+ curr_ticks = ticks_offs + tmr2ticks(tmr_ctr);
+
+ if (curr_ticks >= next_event_time)
+ {
+ next_event_time = (unsigned long) -1;
+ sequencer_timer(0);
+ }
+ spin_unlock_irqrestore(&lock,flags);
+}
+EXPORT_SYMBOL(sound_timer_interrupt);
+
+void sound_timer_init(struct sound_lowlev_timer *t, char *name)
+{
+ int n;
+
+ if (initialized)
+ {
+ if (t->priority <= tmr->priority)
+ return; /* There is already a similar or better timer */
+ tmr = t;
+ return;
+ }
+ initialized = 1;
+ tmr = t;
+
+ n = sound_alloc_timerdev();
+ if (n == -1)
+ n = 0; /* Overwrite the system timer */
+ strcpy(sound_timer.info.name, name);
+ sound_timer_devs[n] = &sound_timer;
+}
+EXPORT_SYMBOL(sound_timer_init);
+
diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c
new file mode 100644
index 0000000..61aaeda
--- /dev/null
+++ b/sound/oss/soundcard.c
@@ -0,0 +1,744 @@
+/*
+ * linux/sound/oss/soundcard.c
+ *
+ * Sound card driver for Linux
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * integrated sound_switch.c
+ * Stefan Reinauer : integrated /proc/sound (equals to /dev/sndstat,
+ * which should disappear in the near future)
+ * Eric Dumas : devfs support (22-Jan-98) <dumas@linux.eu.org> with
+ * fixups by C. Scott Ananian <cananian@alumni.princeton.edu>
+ * Richard Gooch : moved common (non OSS-specific) devices to sound_core.c
+ * Rob Riggs : Added persistent DMA buffers support (1998/10/17)
+ * Christoph Hellwig : Some cleanup work (2000/03/01)
+ */
+
+
+#include "sound_config.h"
+#include <linux/init.h>
+#include <linux/types.h>
+#include <linux/errno.h>
+#include <linux/signal.h>
+#include <linux/fcntl.h>
+#include <linux/ctype.h>
+#include <linux/stddef.h>
+#include <linux/kmod.h>
+#include <linux/kernel.h>
+#include <asm/dma.h>
+#include <asm/io.h>
+#include <linux/wait.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/major.h>
+#include <linux/delay.h>
+#include <linux/proc_fs.h>
+#include <linux/smp_lock.h>
+#include <linux/module.h>
+#include <linux/mm.h>
+#include <linux/device.h>
+
+/*
+ * This ought to be moved into include/asm/dma.h
+ */
+#ifndef valid_dma
+#define valid_dma(n) ((n) >= 0 && (n) < MAX_DMA_CHANNELS && (n) != 4)
+#endif
+
+/*
+ * Table for permanently allocated memory (used when unloading the module)
+ */
+void * sound_mem_blocks[1024];
+int sound_nblocks = 0;
+
+/* Persistent DMA buffers */
+#ifdef CONFIG_SOUND_DMAP
+int sound_dmap_flag = 1;
+#else
+int sound_dmap_flag = 0;
+#endif
+
+static char dma_alloc_map[MAX_DMA_CHANNELS];
+
+#define DMA_MAP_UNAVAIL 0
+#define DMA_MAP_FREE 1
+#define DMA_MAP_BUSY 2
+
+
+unsigned long seq_time = 0; /* Time for /dev/sequencer */
+extern struct class *sound_class;
+
+/*
+ * Table for configurable mixer volume handling
+ */
+static mixer_vol_table mixer_vols[MAX_MIXER_DEV];
+static int num_mixer_volumes;
+
+int *load_mixer_volumes(char *name, int *levels, int present)
+{
+ int i, n;
+
+ for (i = 0; i < num_mixer_volumes; i++) {
+ if (strcmp(name, mixer_vols[i].name) == 0) {
+ if (present)
+ mixer_vols[i].num = i;
+ return mixer_vols[i].levels;
+ }
+ }
+ if (num_mixer_volumes >= MAX_MIXER_DEV) {
+ printk(KERN_ERR "Sound: Too many mixers (%s)\n", name);
+ return levels;
+ }
+ n = num_mixer_volumes++;
+
+ strcpy(mixer_vols[n].name, name);
+
+ if (present)
+ mixer_vols[n].num = n;
+ else
+ mixer_vols[n].num = -1;
+
+ for (i = 0; i < 32; i++)
+ mixer_vols[n].levels[i] = levels[i];
+ return mixer_vols[n].levels;
+}
+EXPORT_SYMBOL(load_mixer_volumes);
+
+static int set_mixer_levels(void __user * arg)
+{
+ /* mixer_vol_table is 174 bytes, so IMHO no reason to not allocate it on the stack */
+ mixer_vol_table buf;
+
+ if (__copy_from_user(&buf, arg, sizeof(buf)))
+ return -EFAULT;
+ load_mixer_volumes(buf.name, buf.levels, 0);
+ if (__copy_to_user(arg, &buf, sizeof(buf)))
+ return -EFAULT;
+ return 0;
+}
+
+static int get_mixer_levels(void __user * arg)
+{
+ int n;
+
+ if (__get_user(n, (int __user *)(&(((mixer_vol_table __user *)arg)->num))))
+ return -EFAULT;
+ if (n < 0 || n >= num_mixer_volumes)
+ return -EINVAL;
+ if (__copy_to_user(arg, &mixer_vols[n], sizeof(mixer_vol_table)))
+ return -EFAULT;
+ return 0;
+}
+
+/* 4K page size but our output routines use some slack for overruns */
+#define PROC_BLOCK_SIZE (3*1024)
+
+static ssize_t sound_read(struct file *file, char __user *buf, size_t count, loff_t *ppos)
+{
+ int dev = iminor(file->f_path.dentry->d_inode);
+ int ret = -EINVAL;
+
+ /*
+ * The OSS drivers aren't remotely happy without this locking,
+ * and unless someone fixes them when they are about to bite the
+ * big one anyway, we might as well bandage here..
+ */
+
+ lock_kernel();
+
+ DEB(printk("sound_read(dev=%d, count=%d)\n", dev, count));
+ switch (dev & 0x0f) {
+ case SND_DEV_DSP:
+ case SND_DEV_DSP16:
+ case SND_DEV_AUDIO:
+ ret = audio_read(dev, file, buf, count);
+ break;
+
+ case SND_DEV_SEQ:
+ case SND_DEV_SEQ2:
+ ret = sequencer_read(dev, file, buf, count);
+ break;
+
+ case SND_DEV_MIDIN:
+ ret = MIDIbuf_read(dev, file, buf, count);
+ }
+ unlock_kernel();
+ return ret;
+}
+
+static ssize_t sound_write(struct file *file, const char __user *buf, size_t count, loff_t *ppos)
+{
+ int dev = iminor(file->f_path.dentry->d_inode);
+ int ret = -EINVAL;
+
+ lock_kernel();
+ DEB(printk("sound_write(dev=%d, count=%d)\n", dev, count));
+ switch (dev & 0x0f) {
+ case SND_DEV_SEQ:
+ case SND_DEV_SEQ2:
+ ret = sequencer_write(dev, file, buf, count);
+ break;
+
+ case SND_DEV_DSP:
+ case SND_DEV_DSP16:
+ case SND_DEV_AUDIO:
+ ret = audio_write(dev, file, buf, count);
+ break;
+
+ case SND_DEV_MIDIN:
+ ret = MIDIbuf_write(dev, file, buf, count);
+ break;
+ }
+ unlock_kernel();
+ return ret;
+}
+
+static int sound_open(struct inode *inode, struct file *file)
+{
+ int dev = iminor(inode);
+ int retval;
+
+ DEB(printk("sound_open(dev=%d)\n", dev));
+ if ((dev >= SND_NDEVS) || (dev < 0)) {
+ printk(KERN_ERR "Invalid minor device %d\n", dev);
+ return -ENXIO;
+ }
+ switch (dev & 0x0f) {
+ case SND_DEV_CTL:
+ dev >>= 4;
+ if (dev >= 0 && dev < MAX_MIXER_DEV && mixer_devs[dev] == NULL) {
+ request_module("mixer%d", dev);
+ }
+ if (dev && (dev >= num_mixers || mixer_devs[dev] == NULL))
+ return -ENXIO;
+
+ if (!try_module_get(mixer_devs[dev]->owner))
+ return -ENXIO;
+ break;
+
+ case SND_DEV_SEQ:
+ case SND_DEV_SEQ2:
+ if ((retval = sequencer_open(dev, file)) < 0)
+ return retval;
+ break;
+
+ case SND_DEV_MIDIN:
+ if ((retval = MIDIbuf_open(dev, file)) < 0)
+ return retval;
+ break;
+
+ case SND_DEV_DSP:
+ case SND_DEV_DSP16:
+ case SND_DEV_AUDIO:
+ if ((retval = audio_open(dev, file)) < 0)
+ return retval;
+ break;
+
+ default:
+ printk(KERN_ERR "Invalid minor device %d\n", dev);
+ return -ENXIO;
+ }
+
+ return 0;
+}
+
+static int sound_release(struct inode *inode, struct file *file)
+{
+ int dev = iminor(inode);
+
+ lock_kernel();
+ DEB(printk("sound_release(dev=%d)\n", dev));
+ switch (dev & 0x0f) {
+ case SND_DEV_CTL:
+ module_put(mixer_devs[dev >> 4]->owner);
+ break;
+
+ case SND_DEV_SEQ:
+ case SND_DEV_SEQ2:
+ sequencer_release(dev, file);
+ break;
+
+ case SND_DEV_MIDIN:
+ MIDIbuf_release(dev, file);
+ break;
+
+ case SND_DEV_DSP:
+ case SND_DEV_DSP16:
+ case SND_DEV_AUDIO:
+ audio_release(dev, file);
+ break;
+
+ default:
+ printk(KERN_ERR "Sound error: Releasing unknown device 0x%02x\n", dev);
+ }
+ unlock_kernel();
+
+ return 0;
+}
+
+static int get_mixer_info(int dev, void __user *arg)
+{
+ mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, mixer_devs[dev]->id, sizeof(info.id));
+ strlcpy(info.name, mixer_devs[dev]->name, sizeof(info.name));
+ info.modify_counter = mixer_devs[dev]->modify_counter;
+ if (__copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int get_old_mixer_info(int dev, void __user *arg)
+{
+ _old_mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, mixer_devs[dev]->id, sizeof(info.id));
+ strlcpy(info.name, mixer_devs[dev]->name, sizeof(info.name));
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+}
+
+static int sound_mixer_ioctl(int mixdev, unsigned int cmd, void __user *arg)
+{
+ if (mixdev < 0 || mixdev >= MAX_MIXER_DEV)
+ return -ENXIO;
+ /* Try to load the mixer... */
+ if (mixer_devs[mixdev] == NULL) {
+ request_module("mixer%d", mixdev);
+ }
+ if (mixdev >= num_mixers || !mixer_devs[mixdev])
+ return -ENXIO;
+ if (cmd == SOUND_MIXER_INFO)
+ return get_mixer_info(mixdev, arg);
+ if (cmd == SOUND_OLD_MIXER_INFO)
+ return get_old_mixer_info(mixdev, arg);
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE)
+ mixer_devs[mixdev]->modify_counter++;
+ if (!mixer_devs[mixdev]->ioctl)
+ return -EINVAL;
+ return mixer_devs[mixdev]->ioctl(mixdev, cmd, arg);
+}
+
+static int sound_ioctl(struct inode *inode, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ int len = 0, dtype;
+ int dev = iminor(inode);
+ void __user *p = (void __user *)arg;
+
+ if (_SIOC_DIR(cmd) != _SIOC_NONE && _SIOC_DIR(cmd) != 0) {
+ /*
+ * Have to validate the address given by the process.
+ */
+ len = _SIOC_SIZE(cmd);
+ if (len < 1 || len > 65536 || !p)
+ return -EFAULT;
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE)
+ if (!access_ok(VERIFY_READ, p, len))
+ return -EFAULT;
+ if (_SIOC_DIR(cmd) & _SIOC_READ)
+ if (!access_ok(VERIFY_WRITE, p, len))
+ return -EFAULT;
+ }
+ DEB(printk("sound_ioctl(dev=%d, cmd=0x%x, arg=0x%x)\n", dev, cmd, arg));
+ if (cmd == OSS_GETVERSION)
+ return __put_user(SOUND_VERSION, (int __user *)p);
+
+ if (_IOC_TYPE(cmd) == 'M' && num_mixers > 0 && /* Mixer ioctl */
+ (dev & 0x0f) != SND_DEV_CTL) {
+ dtype = dev & 0x0f;
+ switch (dtype) {
+ case SND_DEV_DSP:
+ case SND_DEV_DSP16:
+ case SND_DEV_AUDIO:
+ return sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev,
+ cmd, p);
+
+ default:
+ return sound_mixer_ioctl(dev >> 4, cmd, p);
+ }
+ }
+ switch (dev & 0x0f) {
+ case SND_DEV_CTL:
+ if (cmd == SOUND_MIXER_GETLEVELS)
+ return get_mixer_levels(p);
+ if (cmd == SOUND_MIXER_SETLEVELS)
+ return set_mixer_levels(p);
+ return sound_mixer_ioctl(dev >> 4, cmd, p);
+
+ case SND_DEV_SEQ:
+ case SND_DEV_SEQ2:
+ return sequencer_ioctl(dev, file, cmd, p);
+
+ case SND_DEV_DSP:
+ case SND_DEV_DSP16:
+ case SND_DEV_AUDIO:
+ return audio_ioctl(dev, file, cmd, p);
+ break;
+
+ case SND_DEV_MIDIN:
+ return MIDIbuf_ioctl(dev, file, cmd, p);
+ break;
+
+ }
+ return -EINVAL;
+}
+
+static unsigned int sound_poll(struct file *file, poll_table * wait)
+{
+ struct inode *inode = file->f_path.dentry->d_inode;
+ int dev = iminor(inode);
+
+ DEB(printk("sound_poll(dev=%d)\n", dev));
+ switch (dev & 0x0f) {
+ case SND_DEV_SEQ:
+ case SND_DEV_SEQ2:
+ return sequencer_poll(dev, file, wait);
+
+ case SND_DEV_MIDIN:
+ return MIDIbuf_poll(dev, file, wait);
+
+ case SND_DEV_DSP:
+ case SND_DEV_DSP16:
+ case SND_DEV_AUDIO:
+ return DMAbuf_poll(file, dev >> 4, wait);
+ }
+ return 0;
+}
+
+static int sound_mmap(struct file *file, struct vm_area_struct *vma)
+{
+ int dev_class;
+ unsigned long size;
+ struct dma_buffparms *dmap = NULL;
+ int dev = iminor(file->f_path.dentry->d_inode);
+
+ dev_class = dev & 0x0f;
+ dev >>= 4;
+
+ if (dev_class != SND_DEV_DSP && dev_class != SND_DEV_DSP16 && dev_class != SND_DEV_AUDIO) {
+ printk(KERN_ERR "Sound: mmap() not supported for other than audio devices\n");
+ return -EINVAL;
+ }
+ lock_kernel();
+ if (vma->vm_flags & VM_WRITE) /* Map write and read/write to the output buf */
+ dmap = audio_devs[dev]->dmap_out;
+ else if (vma->vm_flags & VM_READ)
+ dmap = audio_devs[dev]->dmap_in;
+ else {
+ printk(KERN_ERR "Sound: Undefined mmap() access\n");
+ unlock_kernel();
+ return -EINVAL;
+ }
+
+ if (dmap == NULL) {
+ printk(KERN_ERR "Sound: mmap() error. dmap == NULL\n");
+ unlock_kernel();
+ return -EIO;
+ }
+ if (dmap->raw_buf == NULL) {
+ printk(KERN_ERR "Sound: mmap() called when raw_buf == NULL\n");
+ unlock_kernel();
+ return -EIO;
+ }
+ if (dmap->mapping_flags) {
+ printk(KERN_ERR "Sound: mmap() called twice for the same DMA buffer\n");
+ unlock_kernel();
+ return -EIO;
+ }
+ if (vma->vm_pgoff != 0) {
+ printk(KERN_ERR "Sound: mmap() offset must be 0.\n");
+ unlock_kernel();
+ return -EINVAL;
+ }
+ size = vma->vm_end - vma->vm_start;
+
+ if (size != dmap->bytes_in_use) {
+ printk(KERN_WARNING "Sound: mmap() size = %ld. Should be %d\n", size, dmap->bytes_in_use);
+ }
+ if (remap_pfn_range(vma, vma->vm_start,
+ virt_to_phys(dmap->raw_buf) >> PAGE_SHIFT,
+ vma->vm_end - vma->vm_start, vma->vm_page_prot)) {
+ unlock_kernel();
+ return -EAGAIN;
+ }
+
+ dmap->mapping_flags |= DMA_MAP_MAPPED;
+
+ if( audio_devs[dev]->d->mmap)
+ audio_devs[dev]->d->mmap(dev);
+
+ memset(dmap->raw_buf,
+ dmap->neutral_byte,
+ dmap->bytes_in_use);
+ unlock_kernel();
+ return 0;
+}
+
+const struct file_operations oss_sound_fops = {
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .read = sound_read,
+ .write = sound_write,
+ .poll = sound_poll,
+ .ioctl = sound_ioctl,
+ .mmap = sound_mmap,
+ .open = sound_open,
+ .release = sound_release,
+};
+
+/*
+ * Create the required special subdevices
+ */
+
+static int create_special_devices(void)
+{
+ int seq1,seq2;
+ seq1=register_sound_special(&oss_sound_fops, 1);
+ if(seq1==-1)
+ goto bad;
+ seq2=register_sound_special(&oss_sound_fops, 8);
+ if(seq2!=-1)
+ return 0;
+ unregister_sound_special(1);
+bad:
+ return -1;
+}
+
+
+/* These device names follow the official Linux device list,
+ * Documentation/devices.txt. Let us know if there are other
+ * common names we should support for compatibility.
+ * Only those devices not created by the generic code in sound_core.c are
+ * registered here.
+ */
+static const struct {
+ unsigned short minor;
+ char *name;
+ umode_t mode;
+ int *num;
+} dev_list[] = { /* list of minor devices */
+/* seems to be some confusion here -- this device is not in the device list */
+ {SND_DEV_DSP16, "dspW", S_IWUGO | S_IRUSR | S_IRGRP,
+ &num_audiodevs},
+ {SND_DEV_AUDIO, "audio", S_IWUGO | S_IRUSR | S_IRGRP,
+ &num_audiodevs},
+};
+
+static int dmabuf;
+static int dmabug;
+
+module_param(dmabuf, int, 0444);
+module_param(dmabug, int, 0444);
+
+static int __init oss_init(void)
+{
+ int err;
+ int i, j;
+
+#ifdef CONFIG_PCI
+ if(dmabug)
+ isa_dma_bridge_buggy = dmabug;
+#endif
+
+ err = create_special_devices();
+ if (err) {
+ printk(KERN_ERR "sound: driver already loaded/included in kernel\n");
+ return err;
+ }
+
+ /* Protecting the innocent */
+ sound_dmap_flag = (dmabuf > 0 ? 1 : 0);
+
+ for (i = 0; i < ARRAY_SIZE(dev_list); i++) {
+ device_create(sound_class, NULL,
+ MKDEV(SOUND_MAJOR, dev_list[i].minor), NULL,
+ "%s", dev_list[i].name);
+
+ if (!dev_list[i].num)
+ continue;
+
+ for (j = 1; j < *dev_list[i].num; j++)
+ device_create(sound_class, NULL,
+ MKDEV(SOUND_MAJOR,
+ dev_list[i].minor + (j*0x10)),
+ NULL, "%s%d", dev_list[i].name, j);
+ }
+
+ if (sound_nblocks >= 1024)
+ printk(KERN_ERR "Sound warning: Deallocation table was too small.\n");
+
+ return 0;
+}
+
+static void __exit oss_cleanup(void)
+{
+ int i, j;
+
+ for (i = 0; i < ARRAY_SIZE(dev_list); i++) {
+ device_destroy(sound_class, MKDEV(SOUND_MAJOR, dev_list[i].minor));
+ if (!dev_list[i].num)
+ continue;
+ for (j = 1; j < *dev_list[i].num; j++)
+ device_destroy(sound_class, MKDEV(SOUND_MAJOR, dev_list[i].minor + (j*0x10)));
+ }
+
+ unregister_sound_special(1);
+ unregister_sound_special(8);
+
+ sound_stop_timer();
+
+ sequencer_unload();
+
+ for (i = 0; i < MAX_DMA_CHANNELS; i++)
+ if (dma_alloc_map[i] != DMA_MAP_UNAVAIL) {
+ printk(KERN_ERR "Sound: Hmm, DMA%d was left allocated - fixed\n", i);
+ sound_free_dma(i);
+ }
+
+ for (i = 0; i < sound_nblocks; i++)
+ vfree(sound_mem_blocks[i]);
+
+}
+
+module_init(oss_init);
+module_exit(oss_cleanup);
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("OSS Sound subsystem");
+MODULE_AUTHOR("Hannu Savolainen, et al.");
+
+
+int sound_alloc_dma(int chn, char *deviceID)
+{
+ int err;
+
+ if ((err = request_dma(chn, deviceID)) != 0)
+ return err;
+
+ dma_alloc_map[chn] = DMA_MAP_FREE;
+
+ return 0;
+}
+EXPORT_SYMBOL(sound_alloc_dma);
+
+int sound_open_dma(int chn, char *deviceID)
+{
+ if (!valid_dma(chn)) {
+ printk(KERN_ERR "sound_open_dma: Invalid DMA channel %d\n", chn);
+ return 1;
+ }
+
+ if (dma_alloc_map[chn] != DMA_MAP_FREE) {
+ printk("sound_open_dma: DMA channel %d busy or not allocated (%d)\n", chn, dma_alloc_map[chn]);
+ return 1;
+ }
+ dma_alloc_map[chn] = DMA_MAP_BUSY;
+ return 0;
+}
+EXPORT_SYMBOL(sound_open_dma);
+
+void sound_free_dma(int chn)
+{
+ if (dma_alloc_map[chn] == DMA_MAP_UNAVAIL) {
+ /* printk( "sound_free_dma: Bad access to DMA channel %d\n", chn); */
+ return;
+ }
+ free_dma(chn);
+ dma_alloc_map[chn] = DMA_MAP_UNAVAIL;
+}
+EXPORT_SYMBOL(sound_free_dma);
+
+void sound_close_dma(int chn)
+{
+ if (dma_alloc_map[chn] != DMA_MAP_BUSY) {
+ printk(KERN_ERR "sound_close_dma: Bad access to DMA channel %d\n", chn);
+ return;
+ }
+ dma_alloc_map[chn] = DMA_MAP_FREE;
+}
+EXPORT_SYMBOL(sound_close_dma);
+
+static void do_sequencer_timer(unsigned long dummy)
+{
+ sequencer_timer(0);
+}
+
+
+static DEFINE_TIMER(seq_timer, do_sequencer_timer, 0, 0);
+
+void request_sound_timer(int count)
+{
+ extern unsigned long seq_time;
+
+ if (count < 0) {
+ seq_timer.expires = (-count) + jiffies;
+ add_timer(&seq_timer);
+ return;
+ }
+ count += seq_time;
+
+ count -= jiffies;
+
+ if (count < 1)
+ count = 1;
+
+ seq_timer.expires = (count) + jiffies;
+ add_timer(&seq_timer);
+}
+
+void sound_stop_timer(void)
+{
+ del_timer(&seq_timer);
+}
+
+void conf_printf(char *name, struct address_info *hw_config)
+{
+#ifndef CONFIG_SOUND_TRACEINIT
+ return;
+#else
+ printk("<%s> at 0x%03x", name, hw_config->io_base);
+
+ if (hw_config->irq)
+ printk(" irq %d", (hw_config->irq > 0) ? hw_config->irq : -hw_config->irq);
+
+ if (hw_config->dma != -1 || hw_config->dma2 != -1)
+ {
+ printk(" dma %d", hw_config->dma);
+ if (hw_config->dma2 != -1)
+ printk(",%d", hw_config->dma2);
+ }
+ printk("\n");
+#endif
+}
+EXPORT_SYMBOL(conf_printf);
+
+void conf_printf2(char *name, int base, int irq, int dma, int dma2)
+{
+#ifndef CONFIG_SOUND_TRACEINIT
+ return;
+#else
+ printk("<%s> at 0x%03x", name, base);
+
+ if (irq)
+ printk(" irq %d", (irq > 0) ? irq : -irq);
+
+ if (dma != -1 || dma2 != -1)
+ {
+ printk(" dma %d", dma);
+ if (dma2 != -1)
+ printk(",%d", dma2);
+ }
+ printk("\n");
+#endif
+}
+EXPORT_SYMBOL(conf_printf2);
+
diff --git a/sound/oss/soundvers.h b/sound/oss/soundvers.h
new file mode 100644
index 0000000..e9084d2
--- /dev/null
+++ b/sound/oss/soundvers.h
@@ -0,0 +1,2 @@
+#define SOUND_VERSION_STRING "3.8s2++-971130"
+#define SOUND_INTERNAL_VERSION 0x030804
diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c
new file mode 100644
index 0000000..30c36d1
--- /dev/null
+++ b/sound/oss/sscape.c
@@ -0,0 +1,1480 @@
+/*
+ * sound/oss/sscape.c
+ *
+ * Low level driver for Ensoniq SoundScape
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ *
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Sergey Smitienko : ensoniq p'n'p support
+ * Christoph Hellwig : adapted to module_init/module_exit
+ * Bartlomiej Zolnierkiewicz : added __init to attach_sscape()
+ * Chris Rankin : Specify that this module owns the coprocessor
+ * Arnaldo C. de Melo : added missing restore_flags in sscape_pnp_upload_file
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+
+#include "sound_config.h"
+#include "sound_firmware.h"
+
+#include <linux/types.h>
+#include <linux/errno.h>
+#include <linux/signal.h>
+#include <linux/fcntl.h>
+#include <linux/ctype.h>
+#include <linux/stddef.h>
+#include <linux/kmod.h>
+#include <asm/dma.h>
+#include <asm/io.h>
+#include <linux/wait.h>
+#include <linux/slab.h>
+#include <linux/ioport.h>
+#include <linux/delay.h>
+#include <linux/proc_fs.h>
+#include <linux/mm.h>
+#include <linux/spinlock.h>
+
+#include "coproc.h"
+
+#include "ad1848.h"
+#include "mpu401.h"
+
+/*
+ * I/O ports
+ */
+#define MIDI_DATA 0
+#define MIDI_CTRL 1
+#define HOST_CTRL 2
+#define TX_READY 0x02
+#define RX_READY 0x01
+#define HOST_DATA 3
+#define ODIE_ADDR 4
+#define ODIE_DATA 5
+
+/*
+ * Indirect registers
+ */
+
+#define GA_INTSTAT_REG 0
+#define GA_INTENA_REG 1
+#define GA_DMAA_REG 2
+#define GA_DMAB_REG 3
+#define GA_INTCFG_REG 4
+#define GA_DMACFG_REG 5
+#define GA_CDCFG_REG 6
+#define GA_SMCFGA_REG 7
+#define GA_SMCFGB_REG 8
+#define GA_HMCTL_REG 9
+
+/*
+ * DMA channel identifiers (A and B)
+ */
+
+#define SSCAPE_DMA_A 0
+#define SSCAPE_DMA_B 1
+
+#define PORT(name) (devc->base+name)
+
+/*
+ * Host commands recognized by the OBP microcode
+ */
+
+#define CMD_GEN_HOST_ACK 0x80
+#define CMD_GEN_MPU_ACK 0x81
+#define CMD_GET_BOARD_TYPE 0x82
+#define CMD_SET_CONTROL 0x88 /* Old firmware only */
+#define CMD_GET_CONTROL 0x89 /* Old firmware only */
+#define CTL_MASTER_VOL 0
+#define CTL_MIC_MODE 2
+#define CTL_SYNTH_VOL 4
+#define CTL_WAVE_VOL 7
+#define CMD_SET_EXTMIDI 0x8a
+#define CMD_GET_EXTMIDI 0x8b
+#define CMD_SET_MT32 0x8c
+#define CMD_GET_MT32 0x8d
+
+#define CMD_ACK 0x80
+
+#define IC_ODIE 1
+#define IC_OPUS 2
+
+typedef struct sscape_info
+{
+ int base, irq, dma;
+
+ int codec, codec_irq; /* required to setup pnp cards*/
+ int codec_type;
+ int ic_type;
+ char* raw_buf;
+ unsigned long raw_buf_phys;
+ int buffsize; /* -------------------------- */
+ spinlock_t lock;
+ int ok; /* Properly detected */
+ int failed;
+ int dma_allocated;
+ int codec_audiodev;
+ int opened;
+ int *osp;
+ int my_audiodev;
+} sscape_info;
+
+static struct sscape_info adev_info = {
+ 0
+};
+
+static struct sscape_info *devc = &adev_info;
+static int sscape_mididev = -1;
+
+/* Some older cards have assigned interrupt bits differently than new ones */
+static char valid_interrupts_old[] = {
+ 9, 7, 5, 15
+};
+
+static char valid_interrupts_new[] = {
+ 9, 5, 7, 10
+};
+
+static char *valid_interrupts = valid_interrupts_new;
+
+/*
+ * See the bottom of the driver. This can be set by spea =0/1.
+ */
+
+#ifdef REVEAL_SPEA
+static char old_hardware = 1;
+#else
+static char old_hardware;
+#endif
+
+static void sleep(unsigned howlong)
+{
+ current->state = TASK_INTERRUPTIBLE;
+ schedule_timeout(howlong);
+}
+
+static unsigned char sscape_read(struct sscape_info *devc, int reg)
+{
+ unsigned long flags;
+ unsigned char val;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ outb(reg, PORT(ODIE_ADDR));
+ val = inb(PORT(ODIE_DATA));
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return val;
+}
+
+static void __sscape_write(int reg, int data)
+{
+ outb(reg, PORT(ODIE_ADDR));
+ outb(data, PORT(ODIE_DATA));
+}
+
+static void sscape_write(struct sscape_info *devc, int reg, int data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ __sscape_write(reg, data);
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static unsigned char sscape_pnp_read_codec(sscape_info* devc, unsigned char reg)
+{
+ unsigned char res;
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ outb( reg, devc -> codec);
+ res = inb (devc -> codec + 1);
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return res;
+
+}
+
+static void sscape_pnp_write_codec(sscape_info* devc, unsigned char reg, unsigned char data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ outb( reg, devc -> codec);
+ outb( data, devc -> codec + 1);
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static void host_open(struct sscape_info *devc)
+{
+ outb((0x00), PORT(HOST_CTRL)); /* Put the board to the host mode */
+}
+
+static void host_close(struct sscape_info *devc)
+{
+ outb((0x03), PORT(HOST_CTRL)); /* Put the board to the MIDI mode */
+}
+
+static int host_write(struct sscape_info *devc, unsigned char *data, int count)
+{
+ unsigned long flags;
+ int i, timeout_val;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ /*
+ * Send the command and data bytes
+ */
+
+ for (i = 0; i < count; i++)
+ {
+ for (timeout_val = 10000; timeout_val > 0; timeout_val--)
+ if (inb(PORT(HOST_CTRL)) & TX_READY)
+ break;
+
+ if (timeout_val <= 0)
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return 0;
+ }
+ outb(data[i], PORT(HOST_DATA));
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return 1;
+}
+
+static int host_read(struct sscape_info *devc)
+{
+ unsigned long flags;
+ int timeout_val;
+ unsigned char data;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ /*
+ * Read a byte
+ */
+
+ for (timeout_val = 10000; timeout_val > 0; timeout_val--)
+ if (inb(PORT(HOST_CTRL)) & RX_READY)
+ break;
+
+ if (timeout_val <= 0)
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return -1;
+ }
+ data = inb(PORT(HOST_DATA));
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return data;
+}
+
+#if 0 /* unused */
+static int host_command1(struct sscape_info *devc, int cmd)
+{
+ unsigned char buf[10];
+ buf[0] = (unsigned char) (cmd & 0xff);
+ return host_write(devc, buf, 1);
+}
+#endif /* unused */
+
+
+static int host_command2(struct sscape_info *devc, int cmd, int parm1)
+{
+ unsigned char buf[10];
+
+ buf[0] = (unsigned char) (cmd & 0xff);
+ buf[1] = (unsigned char) (parm1 & 0xff);
+
+ return host_write(devc, buf, 2);
+}
+
+static int host_command3(struct sscape_info *devc, int cmd, int parm1, int parm2)
+{
+ unsigned char buf[10];
+
+ buf[0] = (unsigned char) (cmd & 0xff);
+ buf[1] = (unsigned char) (parm1 & 0xff);
+ buf[2] = (unsigned char) (parm2 & 0xff);
+ return host_write(devc, buf, 3);
+}
+
+static void set_mt32(struct sscape_info *devc, int value)
+{
+ host_open(devc);
+ host_command2(devc, CMD_SET_MT32, value ? 1 : 0);
+ if (host_read(devc) != CMD_ACK)
+ {
+ /* printk( "SNDSCAPE: Setting MT32 mode failed\n"); */
+ }
+ host_close(devc);
+}
+
+static void set_control(struct sscape_info *devc, int ctrl, int value)
+{
+ host_open(devc);
+ host_command3(devc, CMD_SET_CONTROL, ctrl, value);
+ if (host_read(devc) != CMD_ACK)
+ {
+ /* printk( "SNDSCAPE: Setting control (%d) failed\n", ctrl); */
+ }
+ host_close(devc);
+}
+
+static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, int blk_size, int mode)
+{
+ unsigned char temp;
+
+ if (dma_chan != SSCAPE_DMA_A)
+ {
+ printk(KERN_WARNING "soundscape: Tried to use DMA channel != A. Why?\n");
+ return;
+ }
+ audio_devs[devc->codec_audiodev]->flags &= ~DMA_AUTOMODE;
+ DMAbuf_start_dma(devc->codec_audiodev, buf, blk_size, mode);
+ audio_devs[devc->codec_audiodev]->flags |= DMA_AUTOMODE;
+
+ temp = devc->dma << 4; /* Setup DMA channel select bits */
+ if (devc->dma <= 3)
+ temp |= 0x80; /* 8 bit DMA channel */
+
+ temp |= 1; /* Trigger DMA */
+ sscape_write(devc, GA_DMAA_REG, temp);
+ temp &= 0xfe; /* Clear DMA trigger */
+ sscape_write(devc, GA_DMAA_REG, temp);
+}
+
+static int verify_mpu(struct sscape_info *devc)
+{
+ /*
+ * The SoundScape board could be in three modes (MPU, 8250 and host).
+ * If the card is not in the MPU mode, enabling the MPU driver will
+ * cause infinite loop (the driver believes that there is always some
+ * received data in the buffer.
+ *
+ * Detect this by looking if there are more than 10 received MIDI bytes
+ * (0x00) in the buffer.
+ */
+
+ int i;
+
+ for (i = 0; i < 10; i++)
+ {
+ if (inb(devc->base + HOST_CTRL) & 0x80)
+ return 1;
+
+ if (inb(devc->base) != 0x00)
+ return 1;
+ }
+ printk(KERN_WARNING "SoundScape: The device is not in the MPU-401 mode\n");
+ return 0;
+}
+
+static int sscape_coproc_open(void *dev_info, int sub_device)
+{
+ if (sub_device == COPR_MIDI)
+ {
+ set_mt32(devc, 0);
+ if (!verify_mpu(devc))
+ return -EIO;
+ }
+ return 0;
+}
+
+static void sscape_coproc_close(void *dev_info, int sub_device)
+{
+ struct sscape_info *devc = dev_info;
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ if (devc->dma_allocated)
+ {
+ __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */
+ devc->dma_allocated = 0;
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return;
+}
+
+static void sscape_coproc_reset(void *dev_info)
+{
+}
+
+static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, int size, int flag)
+{
+ unsigned long flags;
+ unsigned char temp;
+ volatile int done, timeout_val;
+ static unsigned char codec_dma_bits;
+
+ if (flag & CPF_FIRST)
+ {
+ /*
+ * First block. Have to allocate DMA and to reset the board
+ * before continuing.
+ */
+
+ spin_lock_irqsave(&devc->lock,flags);
+ codec_dma_bits = sscape_read(devc, GA_CDCFG_REG);
+
+ if (devc->dma_allocated == 0)
+ devc->dma_allocated = 1;
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ sscape_write(devc, GA_HMCTL_REG,
+ (temp = sscape_read(devc, GA_HMCTL_REG)) & 0x3f); /*Reset */
+
+ for (timeout_val = 10000; timeout_val > 0; timeout_val--)
+ sscape_read(devc, GA_HMCTL_REG); /* Delay */
+
+ /* Take board out of reset */
+ sscape_write(devc, GA_HMCTL_REG,
+ (temp = sscape_read(devc, GA_HMCTL_REG)) | 0x80);
+ }
+ /*
+ * Transfer one code block using DMA
+ */
+ if (audio_devs[devc->codec_audiodev]->dmap_out->raw_buf == NULL)
+ {
+ printk(KERN_WARNING "soundscape: DMA buffer not available\n");
+ return 0;
+ }
+ memcpy(audio_devs[devc->codec_audiodev]->dmap_out->raw_buf, block, size);
+
+ spin_lock_irqsave(&devc->lock,flags);
+
+ /******** INTERRUPTS DISABLED NOW ********/
+
+ do_dma(devc, SSCAPE_DMA_A,
+ audio_devs[devc->codec_audiodev]->dmap_out->raw_buf_phys,
+ size, DMA_MODE_WRITE);
+
+ /*
+ * Wait until transfer completes.
+ */
+
+ done = 0;
+ timeout_val = 30;
+ while (!done && timeout_val-- > 0)
+ {
+ int resid;
+
+ if (HZ / 50)
+ sleep(HZ / 50);
+ clear_dma_ff(devc->dma);
+ if ((resid = get_dma_residue(devc->dma)) == 0)
+ done = 1;
+ }
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+ if (!done)
+ return 0;
+
+ if (flag & CPF_LAST)
+ {
+ /*
+ * Take the board out of reset
+ */
+ outb((0x00), PORT(HOST_CTRL));
+ outb((0x00), PORT(MIDI_CTRL));
+
+ temp = sscape_read(devc, GA_HMCTL_REG);
+ temp |= 0x40;
+ sscape_write(devc, GA_HMCTL_REG, temp); /* Kickstart the board */
+
+ /*
+ * Wait until the ODB wakes up
+ */
+ spin_lock_irqsave(&devc->lock,flags);
+ done = 0;
+ timeout_val = 5 * HZ;
+ while (!done && timeout_val-- > 0)
+ {
+ unsigned char x;
+
+ sleep(1);
+ x = inb(PORT(HOST_DATA));
+ if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */
+ {
+ DDB(printk("Soundscape: Acknowledge = %x\n", x));
+ done = 1;
+ }
+ }
+ sscape_write(devc, GA_CDCFG_REG, codec_dma_bits);
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+ if (!done)
+ {
+ printk(KERN_ERR "soundscape: The OBP didn't respond after code download\n");
+ return 0;
+ }
+ spin_lock_irqsave(&devc->lock,flags);
+ done = 0;
+ timeout_val = 5 * HZ;
+ while (!done && timeout_val-- > 0)
+ {
+ sleep(1);
+ if (inb(PORT(HOST_DATA)) == 0xfe) /* Host startup acknowledge */
+ done = 1;
+ }
+ spin_unlock_irqrestore(&devc->lock,flags);
+ if (!done)
+ {
+ printk(KERN_ERR "soundscape: OBP Initialization failed.\n");
+ return 0;
+ }
+ printk(KERN_INFO "SoundScape board initialized OK\n");
+ set_control(devc, CTL_MASTER_VOL, 100);
+ set_control(devc, CTL_SYNTH_VOL, 100);
+
+#ifdef SSCAPE_DEBUG3
+ /*
+ * Temporary debugging aid. Print contents of the registers after
+ * downloading the code.
+ */
+ {
+ int i;
+
+ for (i = 0; i < 13; i++)
+ printk("I%d = %02x (new value)\n", i, sscape_read(devc, i));
+ }
+#endif
+
+ }
+ return 1;
+}
+
+static int download_boot_block(void *dev_info, copr_buffer * buf)
+{
+ if (buf->len <= 0 || buf->len > sizeof(buf->data))
+ return -EINVAL;
+
+ if (!sscape_download_boot(devc, buf->data, buf->len, buf->flags))
+ {
+ printk(KERN_ERR "soundscape: Unable to load microcode block to the OBP.\n");
+ return -EIO;
+ }
+ return 0;
+}
+
+static int sscape_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, int local)
+{
+ copr_buffer *buf;
+ int err;
+
+ switch (cmd)
+ {
+ case SNDCTL_COPR_RESET:
+ sscape_coproc_reset(dev_info);
+ return 0;
+
+ case SNDCTL_COPR_LOAD:
+ buf = (copr_buffer *) vmalloc(sizeof(copr_buffer));
+ if (buf == NULL)
+ return -ENOSPC;
+ if (copy_from_user(buf, arg, sizeof(copr_buffer)))
+ {
+ vfree(buf);
+ return -EFAULT;
+ }
+ err = download_boot_block(dev_info, buf);
+ vfree(buf);
+ return err;
+
+ default:
+ return -EINVAL;
+ }
+}
+
+static coproc_operations sscape_coproc_operations =
+{
+ "SoundScape M68K",
+ THIS_MODULE,
+ sscape_coproc_open,
+ sscape_coproc_close,
+ sscape_coproc_ioctl,
+ sscape_coproc_reset,
+ &adev_info
+};
+
+static struct resource *sscape_ports;
+static int sscape_is_pnp;
+
+static void __init attach_sscape(struct address_info *hw_config)
+{
+#ifndef SSCAPE_REGS
+ /*
+ * Config register values for Spea/V7 Media FX and Ensoniq S-2000.
+ * These values are card
+ * dependent. If you have another SoundScape based card, you have to
+ * find the correct values. Do the following:
+ * - Compile this driver with SSCAPE_DEBUG1 defined.
+ * - Shut down and power off your machine.
+ * - Boot with DOS so that the SSINIT.EXE program is run.
+ * - Warm boot to {Linux|SYSV|BSD} and write down the lines displayed
+ * when detecting the SoundScape.
+ * - Modify the following list to use the values printed during boot.
+ * Undefine the SSCAPE_DEBUG1
+ */
+#define SSCAPE_REGS { \
+/* I0 */ 0x00, \
+/* I1 */ 0xf0, /* Note! Ignored. Set always to 0xf0 */ \
+/* I2 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \
+/* I3 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \
+/* I4 */ 0xf5, /* Ignored */ \
+/* I5 */ 0x10, \
+/* I6 */ 0x00, \
+/* I7 */ 0x2e, /* I7 MEM config A. Likely to vary between models */ \
+/* I8 */ 0x00, /* I8 MEM config B. Likely to vary between models */ \
+/* I9 */ 0x40 /* Ignored */ \
+ }
+#endif
+
+ unsigned long flags;
+ static unsigned char regs[10] = SSCAPE_REGS;
+
+ int i, irq_bits = 0xff;
+
+ if (old_hardware)
+ {
+ valid_interrupts = valid_interrupts_old;
+ conf_printf("Ensoniq SoundScape (old)", hw_config);
+ }
+ else
+ conf_printf("Ensoniq SoundScape", hw_config);
+
+ for (i = 0; i < 4; i++)
+ {
+ if (hw_config->irq == valid_interrupts[i])
+ {
+ irq_bits = i;
+ break;
+ }
+ }
+ if (hw_config->irq > 15 || (regs[4] = irq_bits == 0xff))
+ {
+ printk(KERN_ERR "Invalid IRQ%d\n", hw_config->irq);
+ release_region(devc->base, 2);
+ release_region(devc->base + 2, 6);
+ if (sscape_is_pnp)
+ release_region(devc->codec, 2);
+ return;
+ }
+
+ if (!sscape_is_pnp) {
+
+ spin_lock_irqsave(&devc->lock,flags);
+ /* Host interrupt enable */
+ sscape_write(devc, 1, 0xf0); /* All interrupts enabled */
+ /* DMA A status/trigger register */
+ sscape_write(devc, 2, 0x20); /* DMA channel disabled */
+ /* DMA B status/trigger register */
+ sscape_write(devc, 3, 0x20); /* DMA channel disabled */
+ /* Host interrupt config reg */
+ sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits);
+ /* Don't destroy CD-ROM DMA config bits (0xc0) */
+ sscape_write(devc, 5, (regs[5] & 0x3f) | (sscape_read(devc, 5) & 0xc0));
+ /* CD-ROM config (WSS codec actually) */
+ sscape_write(devc, 6, regs[6]);
+ sscape_write(devc, 7, regs[7]);
+ sscape_write(devc, 8, regs[8]);
+ /* Master control reg. Don't modify CR-ROM bits. Disable SB emul */
+ sscape_write(devc, 9, (sscape_read(devc, 9) & 0xf0) | 0x08);
+ spin_unlock_irqrestore(&devc->lock,flags);
+ }
+#ifdef SSCAPE_DEBUG2
+ /*
+ * Temporary debugging aid. Print contents of the registers after
+ * changing them.
+ */
+ {
+ int i;
+
+ for (i = 0; i < 13; i++)
+ printk("I%d = %02x (new value)\n", i, sscape_read(devc, i));
+ }
+#endif
+
+ if (probe_mpu401(hw_config, sscape_ports))
+ hw_config->always_detect = 1;
+ hw_config->name = "SoundScape";
+
+ hw_config->irq *= -1; /* Negative value signals IRQ sharing */
+ attach_mpu401(hw_config, THIS_MODULE);
+ hw_config->irq *= -1; /* Restore it */
+
+ if (hw_config->slots[1] != -1) /* The MPU driver installed itself */
+ {
+ sscape_mididev = hw_config->slots[1];
+ midi_devs[hw_config->slots[1]]->coproc = &sscape_coproc_operations;
+ }
+ sscape_write(devc, GA_INTENA_REG, 0x80); /* Master IRQ enable */
+ devc->ok = 1;
+ devc->failed = 0;
+}
+
+static int detect_ga(sscape_info * devc)
+{
+ unsigned char save;
+
+ DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base));
+
+ /*
+ * First check that the address register of "ODIE" is
+ * there and that it has exactly 4 writable bits.
+ * First 4 bits
+ */
+
+ if ((save = inb(PORT(ODIE_ADDR))) & 0xf0)
+ {
+ DDB(printk("soundscape: Detect error A\n"));
+ return 0;
+ }
+ outb((0x00), PORT(ODIE_ADDR));
+ if (inb(PORT(ODIE_ADDR)) != 0x00)
+ {
+ DDB(printk("soundscape: Detect error B\n"));
+ return 0;
+ }
+ outb((0xff), PORT(ODIE_ADDR));
+ if (inb(PORT(ODIE_ADDR)) != 0x0f)
+ {
+ DDB(printk("soundscape: Detect error C\n"));
+ return 0;
+ }
+ outb((save), PORT(ODIE_ADDR));
+
+ /*
+ * Now verify that some indirect registers return zero on some bits.
+ * This may break the driver with some future revisions of "ODIE" but...
+ */
+
+ if (sscape_read(devc, 0) & 0x0c)
+ {
+ DDB(printk("soundscape: Detect error D (%x)\n", sscape_read(devc, 0)));
+ return 0;
+ }
+ if (sscape_read(devc, 1) & 0x0f)
+ {
+ DDB(printk("soundscape: Detect error E\n"));
+ return 0;
+ }
+ if (sscape_read(devc, 5) & 0x0f)
+ {
+ DDB(printk("soundscape: Detect error F\n"));
+ return 0;
+ }
+ return 1;
+}
+
+static int sscape_read_host_ctrl(sscape_info* devc)
+{
+ return host_read(devc);
+}
+
+static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b)
+{
+ host_command2(devc, a, b);
+}
+
+static int sscape_alloc_dma(sscape_info *devc)
+{
+ char *start_addr, *end_addr;
+ int dma_pagesize;
+ int sz, size;
+ struct page *page;
+
+ if (devc->raw_buf != NULL) return 0; /* Already done */
+ dma_pagesize = (devc->dma < 4) ? (64 * 1024) : (128 * 1024);
+ devc->raw_buf = NULL;
+ devc->buffsize = 8192*4;
+ if (devc->buffsize > dma_pagesize) devc->buffsize = dma_pagesize;
+ start_addr = NULL;
+ /*
+ * Now loop until we get a free buffer. Try to get smaller buffer if
+ * it fails. Don't accept smaller than 8k buffer for performance
+ * reasons.
+ */
+ while (start_addr == NULL && devc->buffsize > PAGE_SIZE) {
+ for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1);
+ devc->buffsize = PAGE_SIZE * (1 << sz);
+ start_addr = (char *) __get_free_pages(GFP_ATOMIC|GFP_DMA, sz);
+ if (start_addr == NULL) devc->buffsize /= 2;
+ }
+
+ if (start_addr == NULL) {
+ printk(KERN_ERR "sscape pnp init error: Couldn't allocate DMA buffer\n");
+ return 0;
+ } else {
+ /* make some checks */
+ end_addr = start_addr + devc->buffsize - 1;
+ /* now check if it fits into the same dma-pagesize */
+
+ if (((long) start_addr & ~(dma_pagesize - 1)) != ((long) end_addr & ~(dma_pagesize - 1))
+ || end_addr >= (char *) (MAX_DMA_ADDRESS)) {
+ printk(KERN_ERR "sscape pnp: Got invalid address 0x%lx for %db DMA-buffer\n", (long) start_addr, devc->buffsize);
+ return 0;
+ }
+ }
+ devc->raw_buf = start_addr;
+ devc->raw_buf_phys = virt_to_bus(start_addr);
+
+ for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
+ SetPageReserved(page);
+ return 1;
+}
+
+static void sscape_free_dma(sscape_info *devc)
+{
+ int sz, size;
+ unsigned long start_addr, end_addr;
+ struct page *page;
+
+ if (devc->raw_buf == NULL) return;
+ for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1);
+ start_addr = (unsigned long) devc->raw_buf;
+ end_addr = start_addr + devc->buffsize;
+
+ for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
+ ClearPageReserved(page);
+
+ free_pages((unsigned long) devc->raw_buf, sz);
+ devc->raw_buf = NULL;
+}
+
+/* Intel version !!!!!!!!! */
+
+static int sscape_start_dma(int chan, unsigned long physaddr, int count, int dma_mode)
+{
+ unsigned long flags;
+
+ flags = claim_dma_lock();
+ disable_dma(chan);
+ clear_dma_ff(chan);
+ set_dma_mode(chan, dma_mode);
+ set_dma_addr(chan, physaddr);
+ set_dma_count(chan, count);
+ enable_dma(chan);
+ release_dma_lock(flags);
+ return 0;
+}
+
+static void sscape_pnp_start_dma(sscape_info* devc, int arg )
+{
+ int reg;
+ if (arg == 0) reg = 2;
+ else reg = 3;
+
+ sscape_write(devc, reg, sscape_read( devc, reg) | 0x01);
+ sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE);
+}
+
+static int sscape_pnp_wait_dma (sscape_info* devc, int arg )
+{
+ int reg;
+ unsigned long i;
+ unsigned char d;
+
+ if (arg == 0) reg = 2;
+ else reg = 3;
+
+ sleep ( 1 );
+ i = 0;
+ do {
+ d = sscape_read(devc, reg) & 1;
+ if ( d == 1) break;
+ i++;
+ } while (i < 500000);
+ d = sscape_read(devc, reg) & 1;
+ return d;
+}
+
+static int sscape_pnp_alloc_dma(sscape_info* devc)
+{
+ /* printk(KERN_INFO "sscape: requesting dma\n"); */
+ if (request_dma(devc -> dma, "sscape")) return 0;
+ /* printk(KERN_INFO "sscape: dma channel allocated\n"); */
+ if (!sscape_alloc_dma(devc)) {
+ free_dma(devc -> dma);
+ return 0;
+ };
+ return 1;
+}
+
+static void sscape_pnp_free_dma(sscape_info* devc)
+{
+ sscape_free_dma( devc);
+ free_dma(devc -> dma );
+ /* printk(KERN_INFO "sscape: dma released\n"); */
+}
+
+static int sscape_pnp_upload_file(sscape_info* devc, char* fn)
+{
+ int done = 0;
+ int timeout_val;
+ char* data,*dt;
+ int len,l;
+ unsigned long flags;
+
+ sscape_write( devc, 9, sscape_read(devc, 9 ) & 0x3F );
+ sscape_write( devc, 2, (devc -> dma << 4) | 0x80 );
+ sscape_write( devc, 3, 0x20 );
+ sscape_write( devc, 9, sscape_read( devc, 9 ) | 0x80 );
+
+ len = mod_firmware_load(fn, &data);
+ if (len == 0) {
+ printk(KERN_ERR "sscape: file not found: %s\n", fn);
+ return 0;
+ }
+ dt = data;
+ spin_lock_irqsave(&devc->lock,flags);
+ while ( len > 0 ) {
+ if (len > devc -> buffsize) l = devc->buffsize;
+ else l = len;
+ len -= l;
+ memcpy(devc->raw_buf, dt, l); dt += l;
+ sscape_start_dma(devc->dma, devc->raw_buf_phys, l, 0x48);
+ sscape_pnp_start_dma ( devc, 0 );
+ if (sscape_pnp_wait_dma ( devc, 0 ) == 0) {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return 0;
+ }
+ }
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+ vfree(data);
+
+ outb(0, devc -> base + 2);
+ outb(0, devc -> base);
+
+ sscape_write ( devc, 9, sscape_read( devc, 9 ) | 0x40);
+
+ timeout_val = 5 * HZ;
+ while (!done && timeout_val-- > 0)
+ {
+ unsigned char x;
+ sleep(1);
+ x = inb( devc -> base + 3);
+ if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */
+ {
+ //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x);
+ done = 1;
+ }
+ }
+ timeout_val = 5 * HZ;
+ done = 0;
+ while (!done && timeout_val-- > 0)
+ {
+ unsigned char x;
+ sleep(1);
+ x = inb( devc -> base + 3);
+ if (x == 0xfe) /* OBP startup acknowledge */
+ {
+ //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x);
+ done = 1;
+ }
+ }
+
+ if ( !done ) printk(KERN_ERR "soundscape: OBP Initialization failed.\n");
+
+ sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40);
+ sscape_write( devc, 3, (devc -> dma << 4) + 0x80);
+ return 1;
+}
+
+static void __init sscape_pnp_init_hw(sscape_info* devc)
+{
+ unsigned char midi_irq = 0, sb_irq = 0;
+ unsigned i;
+ static char code_file_name[23] = "/sndscape/sndscape.cox";
+
+ int sscape_joystic_enable = 0x7f;
+ int sscape_mic_enable = 0;
+ int sscape_ext_midi = 0;
+
+ if ( !sscape_pnp_alloc_dma(devc) ) {
+ printk(KERN_ERR "sscape: faild to allocate dma\n");
+ return;
+ }
+
+ for (i = 0; i < 4; i++) {
+ if ( devc -> irq == valid_interrupts[i] )
+ midi_irq = i;
+ if ( devc -> codec_irq == valid_interrupts[i] )
+ sb_irq = i;
+ }
+
+ sscape_write( devc, 5, 0x50);
+ sscape_write( devc, 7, 0x2e);
+ sscape_write( devc, 8, 0x00);
+
+ sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40);
+ sscape_write( devc, 3, ( devc -> dma << 4) | 0x80);
+
+ sscape_write (devc, 4, 0xF0 | (midi_irq<<2) | midi_irq);
+
+ i = 0x10; //sscape_read(devc, 9) & (devc->ic_type == IC_ODIE ? 0xf0 : 0xc0);
+ if (sscape_joystic_enable) i |= 8;
+
+ sscape_write (devc, 9, i);
+ sscape_write (devc, 6, 0x80);
+ sscape_write (devc, 1, 0x80);
+
+ if (devc -> codec_type == 2) {
+ sscape_pnp_write_codec( devc, 0x0C, 0x50);
+ sscape_pnp_write_codec( devc, 0x10, sscape_pnp_read_codec( devc, 0x10) & 0x3F);
+ sscape_pnp_write_codec( devc, 0x11, sscape_pnp_read_codec( devc, 0x11) | 0xC0);
+ sscape_pnp_write_codec( devc, 29, 0x20);
+ }
+
+ if (sscape_pnp_upload_file(devc, "/sndscape/scope.cod") == 0 ) {
+ printk(KERN_ERR "sscape: faild to upload file /sndscape/scope.cod\n");
+ sscape_pnp_free_dma(devc);
+ return;
+ }
+
+ i = sscape_read_host_ctrl( devc );
+
+ if ( (i & 0x0F) > 7 ) {
+ printk(KERN_ERR "sscape: scope.cod faild\n");
+ sscape_pnp_free_dma(devc);
+ return;
+ }
+ if ( i & 0x10 ) sscape_write( devc, 7, 0x2F);
+ code_file_name[21] = (char) ( i & 0x0F) + 0x30;
+ if (sscape_pnp_upload_file( devc, code_file_name) == 0) {
+ printk(KERN_ERR "sscape: faild to upload file %s\n", code_file_name);
+ sscape_pnp_free_dma(devc);
+ return;
+ }
+
+ if (devc->ic_type != IC_ODIE) {
+ sscape_pnp_write_codec( devc, 10, (sscape_pnp_read_codec(devc, 10) & 0x7f) |
+ ( sscape_mic_enable == 0 ? 0x00 : 0x80) );
+ }
+ sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */
+ sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */
+ sscape_write_host_ctrl2( devc, 0x8A, sscape_ext_midi);
+
+ sscape_pnp_write_codec ( devc, 6, 0x3f ); //WAV_VOL
+ sscape_pnp_write_codec ( devc, 7, 0x3f ); //WAV_VOL
+ sscape_pnp_write_codec ( devc, 2, 0x1F ); //WD_CDXVOLL
+ sscape_pnp_write_codec ( devc, 3, 0x1F ); //WD_CDXVOLR
+
+ if (devc -> codec_type == 1) {
+ sscape_pnp_write_codec ( devc, 4, 0x1F );
+ sscape_pnp_write_codec ( devc, 5, 0x1F );
+ sscape_write_host_ctrl2( devc, 0x88, sscape_mic_enable);
+ } else {
+ int t;
+ sscape_pnp_write_codec ( devc, 0x10, 0x1F << 1);
+ sscape_pnp_write_codec ( devc, 0x11, 0xC0 | (0x1F << 1));
+
+ t = sscape_pnp_read_codec( devc, 0x00) & 0xDF;
+ if ( (sscape_mic_enable == 0)) t |= 0;
+ else t |= 0x20;
+ sscape_pnp_write_codec ( devc, 0x00, t);
+ t = sscape_pnp_read_codec( devc, 0x01) & 0xDF;
+ if ( (sscape_mic_enable == 0) ) t |= 0;
+ else t |= 0x20;
+ sscape_pnp_write_codec ( devc, 0x01, t);
+ sscape_pnp_write_codec ( devc, 0x40 | 29 , 0x20);
+ outb(0, devc -> codec);
+ }
+ if (devc -> ic_type == IC_OPUS ) {
+ int i = sscape_read( devc, 9 );
+ sscape_write( devc, 9, i | 3 );
+ sscape_write( devc, 3, 0x40);
+
+ if (request_region(0x228, 1, "sscape setup junk")) {
+ outb(0, 0x228);
+ release_region(0x228,1);
+ }
+ sscape_write( devc, 3, (devc -> dma << 4) | 0x80);
+ sscape_write( devc, 9, i );
+ }
+
+ host_close ( devc );
+ sscape_pnp_free_dma(devc);
+}
+
+static int __init detect_sscape_pnp(sscape_info* devc)
+{
+ long i, irq_bits = 0xff;
+ unsigned int d;
+
+ DDB(printk("Entered detect_sscape_pnp(%x)\n", devc->base));
+
+ if (!request_region(devc->codec, 2, "sscape codec")) {
+ printk(KERN_ERR "detect_sscape_pnp: port %x is not free\n", devc->codec);
+ return 0;
+ }
+
+ if ((inb(devc->base + 2) & 0x78) != 0)
+ goto fail;
+
+ d = inb ( devc -> base + 4) & 0xF0;
+ if (d & 0x80)
+ goto fail;
+
+ if (d == 0) {
+ devc->codec_type = 1;
+ devc->ic_type = IC_ODIE;
+ } else if ( (d & 0x60) != 0) {
+ devc->codec_type = 2;
+ devc->ic_type = IC_OPUS;
+ } else if ( (d & 0x40) != 0) { /* WTF? */
+ devc->codec_type = 2;
+ devc->ic_type = IC_ODIE;
+ } else
+ goto fail;
+
+ sscape_is_pnp = 1;
+
+ outb(0xFA, devc -> base+4);
+ if ((inb( devc -> base+4) & 0x9F) != 0x0A)
+ goto fail;
+ outb(0xFE, devc -> base+4);
+ if ( (inb(devc -> base+4) & 0x9F) != 0x0E)
+ goto fail;
+ if ( (inb(devc -> base+5) & 0x9F) != 0x0E)
+ goto fail;
+
+ if (devc->codec_type == 2) {
+ if (devc->codec != devc->base + 8) {
+ printk("soundscape warning: incorrect codec port specified\n");
+ goto fail;
+ }
+ d = 0x10 | (sscape_read(devc, 9) & 0xCF);
+ sscape_write(devc, 9, d);
+ sscape_write(devc, 6, 0x80);
+ } else {
+ //todo: check codec is not base + 8
+ }
+
+ d = (sscape_read(devc, 9) & 0x3F) | 0xC0;
+ sscape_write(devc, 9, d);
+
+ for (i = 0; i < 550000; i++)
+ if ( !(inb(devc -> codec) & 0x80) ) break;
+
+ d = inb(devc -> codec);
+ if (d & 0x80)
+ goto fail;
+ if ( inb(devc -> codec + 2) == 0xFF)
+ goto fail;
+
+ sscape_write(devc, 9, sscape_read(devc, 9) & 0x3F );
+
+ d = inb(devc -> codec) & 0x80;
+ if ( d == 0) {
+ printk(KERN_INFO "soundscape: hardware detected\n");
+ valid_interrupts = valid_interrupts_new;
+ } else {
+ printk(KERN_INFO "soundscape: board looks like media fx\n");
+ valid_interrupts = valid_interrupts_old;
+ old_hardware = 1;
+ }
+
+ sscape_write( devc, 9, 0xC0 | (sscape_read(devc, 9) & 0x3F) );
+
+ for (i = 0; i < 550000; i++)
+ if ( !(inb(devc -> codec) & 0x80))
+ break;
+
+ sscape_pnp_init_hw(devc);
+
+ for (i = 0; i < 4; i++)
+ {
+ if (devc->codec_irq == valid_interrupts[i]) {
+ irq_bits = i;
+ break;
+ }
+ }
+ sscape_write(devc, GA_INTENA_REG, 0x00);
+ sscape_write(devc, GA_DMACFG_REG, 0x50);
+ sscape_write(devc, GA_DMAA_REG, 0x70);
+ sscape_write(devc, GA_DMAB_REG, 0x20);
+ sscape_write(devc, GA_INTCFG_REG, 0xf0);
+ sscape_write(devc, GA_CDCFG_REG, 0x89 | (devc->dma << 4) | (irq_bits << 1));
+
+ sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 0) | 0x20);
+ sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 1) | 0x20);
+
+ return 1;
+fail:
+ release_region(devc->codec, 2);
+ return 0;
+}
+
+static int __init probe_sscape(struct address_info *hw_config)
+{
+ devc->base = hw_config->io_base;
+ devc->irq = hw_config->irq;
+ devc->dma = hw_config->dma;
+ devc->osp = hw_config->osp;
+
+#ifdef SSCAPE_DEBUG1
+ /*
+ * Temporary debugging aid. Print contents of the registers before
+ * changing them.
+ */
+ {
+ int i;
+
+ for (i = 0; i < 13; i++)
+ printk("I%d = %02x (old value)\n", i, sscape_read(devc, i));
+ }
+#endif
+ devc->failed = 1;
+
+ sscape_ports = request_region(devc->base, 2, "mpu401");
+ if (!sscape_ports)
+ return 0;
+
+ if (!request_region(devc->base + 2, 6, "SoundScape")) {
+ release_region(devc->base, 2);
+ return 0;
+ }
+
+ if (!detect_ga(devc)) {
+ if (detect_sscape_pnp(devc))
+ return 1;
+ release_region(devc->base, 2);
+ release_region(devc->base + 2, 6);
+ return 0;
+ }
+
+ if (old_hardware) /* Check that it's really an old Spea/Reveal card. */
+ {
+ unsigned char tmp;
+ int cc;
+
+ if (!((tmp = sscape_read(devc, GA_HMCTL_REG)) & 0xc0))
+ {
+ sscape_write(devc, GA_HMCTL_REG, tmp | 0x80);
+ for (cc = 0; cc < 200000; ++cc)
+ inb(devc->base + ODIE_ADDR);
+ }
+ }
+ return 1;
+}
+
+static int __init init_ss_ms_sound(struct address_info *hw_config)
+{
+ int i, irq_bits = 0xff;
+ int ad_flags = 0;
+ struct resource *ports;
+
+ if (devc->failed)
+ {
+ printk(KERN_ERR "soundscape: Card not detected\n");
+ return 0;
+ }
+ if (devc->ok == 0)
+ {
+ printk(KERN_ERR "soundscape: Invalid initialization order.\n");
+ return 0;
+ }
+ for (i = 0; i < 4; i++)
+ {
+ if (hw_config->irq == valid_interrupts[i])
+ {
+ irq_bits = i;
+ break;
+ }
+ }
+ if (irq_bits == 0xff) {
+ printk(KERN_ERR "soundscape: Invalid MSS IRQ%d\n", hw_config->irq);
+ return 0;
+ }
+
+ if (old_hardware)
+ ad_flags = 0x12345677; /* Tell that we may have a CS4248 chip (Spea-V7 Media FX) */
+ else if (sscape_is_pnp)
+ ad_flags = 0x87654321; /* Tell that we have a soundscape pnp with 1845 chip */
+
+ ports = request_region(hw_config->io_base, 4, "ad1848");
+ if (!ports) {
+ printk(KERN_ERR "soundscape: ports busy\n");
+ return 0;
+ }
+
+ if (!ad1848_detect(ports, &ad_flags, hw_config->osp)) {
+ release_region(hw_config->io_base, 4);
+ return 0;
+ }
+
+ if (!sscape_is_pnp) /*pnp is already setup*/
+ {
+ /*
+ * Setup the DMA polarity.
+ */
+ sscape_write(devc, GA_DMACFG_REG, 0x50);
+
+ /*
+ * Take the gate-array off of the DMA channel.
+ */
+ sscape_write(devc, GA_DMAB_REG, 0x20);
+
+ /*
+ * Init the AD1848 (CD-ROM) config reg.
+ */
+ sscape_write(devc, GA_CDCFG_REG, 0x89 | (hw_config->dma << 4) | (irq_bits << 1));
+ }
+
+ if (hw_config->irq == devc->irq)
+ printk(KERN_WARNING "soundscape: Warning! The WSS mode can't share IRQ with MIDI\n");
+
+ hw_config->slots[0] = ad1848_init(
+ sscape_is_pnp ? "SoundScape" : "SoundScape PNP",
+ ports,
+ hw_config->irq,
+ hw_config->dma,
+ hw_config->dma,
+ 0,
+ devc->osp,
+ THIS_MODULE);
+
+
+ if (hw_config->slots[0] != -1) /* The AD1848 driver installed itself */
+ {
+ audio_devs[hw_config->slots[0]]->coproc = &sscape_coproc_operations;
+ devc->codec_audiodev = hw_config->slots[0];
+ devc->my_audiodev = hw_config->slots[0];
+
+ /* Set proper routings here (what are they) */
+ AD1848_REROUTE(SOUND_MIXER_LINE1, SOUND_MIXER_LINE);
+ }
+
+#ifdef SSCAPE_DEBUG5
+ /*
+ * Temporary debugging aid. Print contents of the registers
+ * after the AD1848 device has been initialized.
+ */
+ {
+ int i;
+
+ for (i = 0; i < 13; i++)
+ printk("I%d = %02x\n", i, sscape_read(devc, i));
+ }
+#endif
+ return 1;
+}
+
+static void __exit unload_sscape(struct address_info *hw_config)
+{
+ release_region(devc->base + 2, 6);
+ unload_mpu401(hw_config);
+ if (sscape_is_pnp)
+ release_region(devc->codec, 2);
+}
+
+static void __exit unload_ss_ms_sound(struct address_info *hw_config)
+{
+ ad1848_unload(hw_config->io_base,
+ hw_config->irq,
+ devc->dma,
+ devc->dma,
+ 0);
+ sound_unload_audiodev(hw_config->slots[0]);
+}
+
+static struct address_info cfg;
+static struct address_info cfg_mpu;
+
+static int __initdata spea = -1;
+static int mss = 0;
+static int __initdata dma = -1;
+static int __initdata irq = -1;
+static int __initdata io = -1;
+static int __initdata mpu_irq = -1;
+static int __initdata mpu_io = -1;
+
+module_param(dma, int, 0);
+module_param(irq, int, 0);
+module_param(io, int, 0);
+module_param(spea, int, 0); /* spea=0/1 set the old_hardware */
+module_param(mpu_irq, int, 0);
+module_param(mpu_io, int, 0);
+module_param(mss, int, 0);
+
+static int __init init_sscape(void)
+{
+ printk(KERN_INFO "Soundscape driver Copyright (C) by Hannu Savolainen 1993-1996\n");
+
+ cfg.irq = irq;
+ cfg.dma = dma;
+ cfg.io_base = io;
+
+ cfg_mpu.irq = mpu_irq;
+ cfg_mpu.io_base = mpu_io;
+ /* WEH - Try to get right dma channel */
+ cfg_mpu.dma = dma;
+
+ devc->codec = cfg.io_base;
+ devc->codec_irq = cfg.irq;
+ devc->codec_type = 0;
+ devc->ic_type = 0;
+ devc->raw_buf = NULL;
+ spin_lock_init(&devc->lock);
+
+ if (cfg.dma == -1 || cfg.irq == -1 || cfg.io_base == -1) {
+ printk(KERN_ERR "DMA, IRQ, and IO port must be specified.\n");
+ return -EINVAL;
+ }
+
+ if (cfg_mpu.irq == -1 && cfg_mpu.io_base != -1) {
+ printk(KERN_ERR "MPU_IRQ must be specified if MPU_IO is set.\n");
+ return -EINVAL;
+ }
+
+ if(spea != -1) {
+ old_hardware = spea;
+ printk(KERN_INFO "Forcing %s hardware support.\n",
+ spea?"new":"old");
+ }
+ if (probe_sscape(&cfg_mpu) == 0)
+ return -ENODEV;
+
+ attach_sscape(&cfg_mpu);
+
+ mss = init_ss_ms_sound(&cfg);
+
+ return 0;
+}
+
+static void __exit cleanup_sscape(void)
+{
+ if (mss)
+ unload_ss_ms_sound(&cfg);
+ unload_sscape(&cfg_mpu);
+}
+
+module_init(init_sscape);
+module_exit(cleanup_sscape);
+
+#ifndef MODULE
+static int __init setup_sscape(char *str)
+{
+ /* io, irq, dma, mpu_io, mpu_irq */
+ int ints[6];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+ dma = ints[3];
+ mpu_io = ints[4];
+ mpu_irq = ints[5];
+
+ return 1;
+}
+
+__setup("sscape=", setup_sscape);
+#endif
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
new file mode 100644
index 0000000..41562ec
--- /dev/null
+++ b/sound/oss/swarm_cs4297a.c
@@ -0,0 +1,2740 @@
+/*******************************************************************************
+*
+* "swarm_cs4297a.c" -- Cirrus Logic-Crystal CS4297a linux audio driver.
+*
+* Copyright (C) 2001 Broadcom Corporation.
+* Copyright (C) 2000,2001 Cirrus Logic Corp.
+* -- adapted from drivers by Thomas Sailer,
+* -- but don't bug him; Problems should go to:
+* -- tom woller (twoller@crystal.cirrus.com) or
+* (audio@crystal.cirrus.com).
+* -- adapted from cs4281 PCI driver for cs4297a on
+* BCM1250 Synchronous Serial interface
+* (Kip Walker, Broadcom Corp.)
+* Copyright (C) 2004 Maciej W. Rozycki
+* Copyright (C) 2005 Ralf Baechle (ralf@linux-mips.org)
+*
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of the GNU General Public License as published by
+* the Free Software Foundation; either version 2 of the License, or
+* (at your option) any later version.
+*
+* This program is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+* GNU General Public License for more details.
+*
+* You should have received a copy of the GNU General Public License
+* along with this program; if not, write to the Free Software
+* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*
+* Module command line parameters:
+* none
+*
+* Supported devices:
+* /dev/dsp standard /dev/dsp device, (mostly) OSS compatible
+* /dev/mixer standard /dev/mixer device, (mostly) OSS compatible
+* /dev/midi simple MIDI UART interface, no ioctl
+*
+* Modification History
+* 08/20/00 trw - silence and no stopping DAC until release
+* 08/23/00 trw - added CS_DBG statements, fix interrupt hang issue on DAC stop.
+* 09/18/00 trw - added 16bit only record with conversion
+* 09/24/00 trw - added Enhanced Full duplex (separate simultaneous
+* capture/playback rates)
+* 10/03/00 trw - fixed mmap (fixed GRECORD and the XMMS mmap test plugin
+* libOSSm.so)
+* 10/11/00 trw - modified for 2.4.0-test9 kernel enhancements (NR_MAP removal)
+* 11/03/00 trw - fixed interrupt loss/stutter, added debug.
+* 11/10/00 bkz - added __devinit to cs4297a_hw_init()
+* 11/10/00 trw - fixed SMP and capture spinlock hang.
+* 12/04/00 trw - cleaned up CSDEBUG flags and added "defaultorder" moduleparm.
+* 12/05/00 trw - fixed polling (myth2), and added underrun swptr fix.
+* 12/08/00 trw - added PM support.
+* 12/14/00 trw - added wrapper code, builds under 2.4.0, 2.2.17-20, 2.2.17-8
+* (RH/Dell base), 2.2.18, 2.2.12. cleaned up code mods by ident.
+* 12/19/00 trw - added PM support for 2.2 base (apm_callback). other PM cleanup.
+* 12/21/00 trw - added fractional "defaultorder" inputs. if >100 then use
+* defaultorder-100 as power of 2 for the buffer size. example:
+* 106 = 2^(106-100) = 2^6 = 64 bytes for the buffer size.
+*
+*******************************************************************************/
+
+#include <linux/list.h>
+#include <linux/module.h>
+#include <linux/string.h>
+#include <linux/ioport.h>
+#include <linux/sched.h>
+#include <linux/delay.h>
+#include <linux/sound.h>
+#include <linux/slab.h>
+#include <linux/soundcard.h>
+#include <linux/ac97_codec.h>
+#include <linux/pci.h>
+#include <linux/bitops.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/poll.h>
+#include <linux/mutex.h>
+#include <linux/kernel.h>
+
+#include <asm/byteorder.h>
+#include <asm/dma.h>
+#include <asm/io.h>
+#include <asm/uaccess.h>
+
+#include <asm/sibyte/sb1250_regs.h>
+#include <asm/sibyte/sb1250_int.h>
+#include <asm/sibyte/sb1250_dma.h>
+#include <asm/sibyte/sb1250_scd.h>
+#include <asm/sibyte/sb1250_syncser.h>
+#include <asm/sibyte/sb1250_mac.h>
+#include <asm/sibyte/sb1250.h>
+
+struct cs4297a_state;
+
+static void stop_dac(struct cs4297a_state *s);
+static void stop_adc(struct cs4297a_state *s);
+static void start_dac(struct cs4297a_state *s);
+static void start_adc(struct cs4297a_state *s);
+#undef OSS_DOCUMENTED_MIXER_SEMANTICS
+
+// ---------------------------------------------------------------------
+
+#define CS4297a_MAGIC 0xf00beef1
+
+// buffer order determines the size of the dma buffer for the driver.
+// under Linux, a smaller buffer allows more responsiveness from many of the
+// applications (e.g. games). A larger buffer allows some of the apps (esound)
+// to not underrun the dma buffer as easily. As default, use 32k (order=3)
+// rather than 64k as some of the games work more responsively.
+// log base 2( buff sz = 32k).
+
+//static unsigned long defaultorder = 3;
+//MODULE_PARM(defaultorder, "i");
+
+//
+// Turn on/off debugging compilation by commenting out "#define CSDEBUG"
+//
+#define CSDEBUG 0
+#if CSDEBUG
+#define CSDEBUG_INTERFACE 1
+#else
+#undef CSDEBUG_INTERFACE
+#endif
+//
+// cs_debugmask areas
+//
+#define CS_INIT 0x00000001 // initialization and probe functions
+#define CS_ERROR 0x00000002 // tmp debugging bit placeholder
+#define CS_INTERRUPT 0x00000004 // interrupt handler (separate from all other)
+#define CS_FUNCTION 0x00000008 // enter/leave functions
+#define CS_WAVE_WRITE 0x00000010 // write information for wave
+#define CS_WAVE_READ 0x00000020 // read information for wave
+#define CS_AC97 0x00000040 // AC97 register access
+#define CS_DESCR 0x00000080 // descriptor management
+#define CS_OPEN 0x00000400 // all open functions in the driver
+#define CS_RELEASE 0x00000800 // all release functions in the driver
+#define CS_PARMS 0x00001000 // functional and operational parameters
+#define CS_IOCTL 0x00002000 // ioctl (non-mixer)
+#define CS_TMP 0x10000000 // tmp debug mask bit
+
+//
+// CSDEBUG is usual mode is set to 1, then use the
+// cs_debuglevel and cs_debugmask to turn on or off debugging.
+// Debug level of 1 has been defined to be kernel errors and info
+// that should be printed on any released driver.
+//
+#if CSDEBUG
+#define CS_DBGOUT(mask,level,x) if((cs_debuglevel >= (level)) && ((mask) & cs_debugmask) ) {x;}
+#else
+#define CS_DBGOUT(mask,level,x)
+#endif
+
+#if CSDEBUG
+static unsigned long cs_debuglevel = 4; // levels range from 1-9
+static unsigned long cs_debugmask = CS_INIT /*| CS_IOCTL*/;
+module_param(cs_debuglevel, int, 0);
+module_param(cs_debugmask, int, 0);
+#endif
+#define CS_TRUE 1
+#define CS_FALSE 0
+
+#define CS_TYPE_ADC 0
+#define CS_TYPE_DAC 1
+
+#define SER_BASE (A_SER_BASE_1 + KSEG1)
+#define SS_CSR(t) (SER_BASE+t)
+#define SS_TXTBL(t) (SER_BASE+R_SER_TX_TABLE_BASE+(t*8))
+#define SS_RXTBL(t) (SER_BASE+R_SER_RX_TABLE_BASE+(t*8))
+
+#define FRAME_BYTES 32
+#define FRAME_SAMPLE_BYTES 4
+
+/* Should this be variable? */
+#define SAMPLE_BUF_SIZE (16*1024)
+#define SAMPLE_FRAME_COUNT (SAMPLE_BUF_SIZE / FRAME_SAMPLE_BYTES)
+/* The driver can explode/shrink the frames to/from a smaller sample
+ buffer */
+#define DMA_BLOAT_FACTOR 1
+#define DMA_DESCR (SAMPLE_FRAME_COUNT / DMA_BLOAT_FACTOR)
+#define DMA_BUF_SIZE (DMA_DESCR * FRAME_BYTES)
+
+/* Use the maxmium count (255 == 5.1 ms between interrupts) */
+#define DMA_INT_CNT ((1 << S_DMA_INT_PKTCNT) - 1)
+
+/* Figure this out: how many TX DMAs ahead to schedule a reg access */
+#define REG_LATENCY 150
+
+#define FRAME_TX_US 20
+
+#define SERDMA_NEXTBUF(d,f) (((d)->f+1) % (d)->ringsz)
+
+static const char invalid_magic[] =
+ KERN_CRIT "cs4297a: invalid magic value\n";
+
+#define VALIDATE_STATE(s) \
+({ \
+ if (!(s) || (s)->magic != CS4297a_MAGIC) { \
+ printk(invalid_magic); \
+ return -ENXIO; \
+ } \
+})
+
+struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs };
+
+typedef struct serdma_descr_s {
+ u64 descr_a;
+ u64 descr_b;
+} serdma_descr_t;
+
+typedef unsigned long paddr_t;
+
+typedef struct serdma_s {
+ unsigned ringsz;
+ serdma_descr_t *descrtab;
+ serdma_descr_t *descrtab_end;
+ paddr_t descrtab_phys;
+
+ serdma_descr_t *descr_add;
+ serdma_descr_t *descr_rem;
+
+ u64 *dma_buf; // buffer for DMA contents (frames)
+ paddr_t dma_buf_phys;
+ u16 *sample_buf; // tmp buffer for sample conversions
+ u16 *sb_swptr;
+ u16 *sb_hwptr;
+ u16 *sb_end;
+
+ dma_addr_t dmaaddr;
+// unsigned buforder; // Log base 2 of 'dma_buf' size in bytes..
+ unsigned numfrag; // # of 'fragments' in the buffer.
+ unsigned fragshift; // Log base 2 of fragment size.
+ unsigned hwptr, swptr;
+ unsigned total_bytes; // # bytes process since open.
+ unsigned blocks; // last returned blocks value GETOPTR
+ unsigned wakeup; // interrupt occurred on block
+ int count;
+ unsigned underrun; // underrun flag
+ unsigned error; // over/underrun
+ wait_queue_head_t wait;
+ wait_queue_head_t reg_wait;
+ // redundant, but makes calculations easier
+ unsigned fragsize; // 2**fragshift..
+ unsigned sbufsz; // 2**buforder.
+ unsigned fragsamples;
+ // OSS stuff
+ unsigned mapped:1; // Buffer mapped in cs4297a_mmap()?
+ unsigned ready:1; // prog_dmabuf_dac()/adc() successful?
+ unsigned endcleared:1;
+ unsigned type:1; // adc or dac buffer (CS_TYPE_XXX)
+ unsigned ossfragshift;
+ int ossmaxfrags;
+ unsigned subdivision;
+} serdma_t;
+
+struct cs4297a_state {
+ // magic
+ unsigned int magic;
+
+ struct list_head list;
+
+ // soundcore stuff
+ int dev_audio;
+ int dev_mixer;
+
+ // hardware resources
+ unsigned int irq;
+
+ struct {
+ unsigned int rx_ovrrn; /* FIFO */
+ unsigned int rx_overflow; /* staging buffer */
+ unsigned int tx_underrun;
+ unsigned int rx_bad;
+ unsigned int rx_good;
+ } stats;
+
+ // mixer registers
+ struct {
+ unsigned short vol[10];
+ unsigned int recsrc;
+ unsigned int modcnt;
+ unsigned short micpreamp;
+ } mix;
+
+ // wave stuff
+ struct properties {
+ unsigned fmt;
+ unsigned fmt_original; // original requested format
+ unsigned channels;
+ unsigned rate;
+ } prop_dac, prop_adc;
+ unsigned conversion:1; // conversion from 16 to 8 bit in progress
+ unsigned ena;
+ spinlock_t lock;
+ struct mutex open_mutex;
+ struct mutex open_sem_adc;
+ struct mutex open_sem_dac;
+ fmode_t open_mode;
+ wait_queue_head_t open_wait;
+ wait_queue_head_t open_wait_adc;
+ wait_queue_head_t open_wait_dac;
+
+ dma_addr_t dmaaddr_sample_buf;
+ unsigned buforder_sample_buf; // Log base 2 of 'dma_buf' size in bytes..
+
+ serdma_t dma_dac, dma_adc;
+
+ volatile u16 read_value;
+ volatile u16 read_reg;
+ volatile u64 reg_request;
+};
+
+#if 1
+#define prog_codec(a,b)
+#define dealloc_dmabuf(a,b);
+#endif
+
+static int prog_dmabuf_adc(struct cs4297a_state *s)
+{
+ s->dma_adc.ready = 1;
+ return 0;
+}
+
+
+static int prog_dmabuf_dac(struct cs4297a_state *s)
+{
+ s->dma_dac.ready = 1;
+ return 0;
+}
+
+static void clear_advance(void *buf, unsigned bsize, unsigned bptr,
+ unsigned len, unsigned char c)
+{
+ if (bptr + len > bsize) {
+ unsigned x = bsize - bptr;
+ memset(((char *) buf) + bptr, c, x);
+ bptr = 0;
+ len -= x;
+ }
+ CS_DBGOUT(CS_WAVE_WRITE, 4, printk(KERN_INFO
+ "cs4297a: clear_advance(): memset %d at 0x%.8x for %d size \n",
+ (unsigned)c, (unsigned)((char *) buf) + bptr, len));
+ memset(((char *) buf) + bptr, c, len);
+}
+
+#if CSDEBUG
+
+// DEBUG ROUTINES
+
+#define SOUND_MIXER_CS_GETDBGLEVEL _SIOWR('M',120, int)
+#define SOUND_MIXER_CS_SETDBGLEVEL _SIOWR('M',121, int)
+#define SOUND_MIXER_CS_GETDBGMASK _SIOWR('M',122, int)
+#define SOUND_MIXER_CS_SETDBGMASK _SIOWR('M',123, int)
+
+static void cs_printioctl(unsigned int x)
+{
+ unsigned int i;
+ unsigned char vidx;
+ // Index of mixtable1[] member is Device ID
+ // and must be <= SOUND_MIXER_NRDEVICES.
+ // Value of array member is index into s->mix.vol[]
+ static const unsigned char mixtable1[SOUND_MIXER_NRDEVICES] = {
+ [SOUND_MIXER_PCM] = 1, // voice
+ [SOUND_MIXER_LINE1] = 2, // AUX
+ [SOUND_MIXER_CD] = 3, // CD
+ [SOUND_MIXER_LINE] = 4, // Line
+ [SOUND_MIXER_SYNTH] = 5, // FM
+ [SOUND_MIXER_MIC] = 6, // Mic
+ [SOUND_MIXER_SPEAKER] = 7, // Speaker
+ [SOUND_MIXER_RECLEV] = 8, // Recording level
+ [SOUND_MIXER_VOLUME] = 9 // Master Volume
+ };
+
+ switch (x) {
+ case SOUND_MIXER_CS_GETDBGMASK:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_CS_GETDBGMASK:\n"));
+ break;
+ case SOUND_MIXER_CS_GETDBGLEVEL:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_CS_GETDBGLEVEL:\n"));
+ break;
+ case SOUND_MIXER_CS_SETDBGMASK:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_CS_SETDBGMASK:\n"));
+ break;
+ case SOUND_MIXER_CS_SETDBGLEVEL:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_CS_SETDBGLEVEL:\n"));
+ break;
+ case OSS_GETVERSION:
+ CS_DBGOUT(CS_IOCTL, 4, printk("OSS_GETVERSION:\n"));
+ break;
+ case SNDCTL_DSP_SYNC:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SYNC:\n"));
+ break;
+ case SNDCTL_DSP_SETDUPLEX:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETDUPLEX:\n"));
+ break;
+ case SNDCTL_DSP_GETCAPS:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETCAPS:\n"));
+ break;
+ case SNDCTL_DSP_RESET:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_RESET:\n"));
+ break;
+ case SNDCTL_DSP_SPEED:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SPEED:\n"));
+ break;
+ case SNDCTL_DSP_STEREO:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_STEREO:\n"));
+ break;
+ case SNDCTL_DSP_CHANNELS:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_CHANNELS:\n"));
+ break;
+ case SNDCTL_DSP_GETFMTS:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETFMTS:\n"));
+ break;
+ case SNDCTL_DSP_SETFMT:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETFMT:\n"));
+ break;
+ case SNDCTL_DSP_POST:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_POST:\n"));
+ break;
+ case SNDCTL_DSP_GETTRIGGER:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETTRIGGER:\n"));
+ break;
+ case SNDCTL_DSP_SETTRIGGER:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETTRIGGER:\n"));
+ break;
+ case SNDCTL_DSP_GETOSPACE:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETOSPACE:\n"));
+ break;
+ case SNDCTL_DSP_GETISPACE:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETISPACE:\n"));
+ break;
+ case SNDCTL_DSP_NONBLOCK:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_NONBLOCK:\n"));
+ break;
+ case SNDCTL_DSP_GETODELAY:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETODELAY:\n"));
+ break;
+ case SNDCTL_DSP_GETIPTR:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETIPTR:\n"));
+ break;
+ case SNDCTL_DSP_GETOPTR:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETOPTR:\n"));
+ break;
+ case SNDCTL_DSP_GETBLKSIZE:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_GETBLKSIZE:\n"));
+ break;
+ case SNDCTL_DSP_SETFRAGMENT:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SNDCTL_DSP_SETFRAGMENT:\n"));
+ break;
+ case SNDCTL_DSP_SUBDIVIDE:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SUBDIVIDE:\n"));
+ break;
+ case SOUND_PCM_READ_RATE:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_RATE:\n"));
+ break;
+ case SOUND_PCM_READ_CHANNELS:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_PCM_READ_CHANNELS:\n"));
+ break;
+ case SOUND_PCM_READ_BITS:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_BITS:\n"));
+ break;
+ case SOUND_PCM_WRITE_FILTER:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_PCM_WRITE_FILTER:\n"));
+ break;
+ case SNDCTL_DSP_SETSYNCRO:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SNDCTL_DSP_SETSYNCRO:\n"));
+ break;
+ case SOUND_PCM_READ_FILTER:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_PCM_READ_FILTER:\n"));
+ break;
+ case SOUND_MIXER_PRIVATE1:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE1:\n"));
+ break;
+ case SOUND_MIXER_PRIVATE2:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE2:\n"));
+ break;
+ case SOUND_MIXER_PRIVATE3:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE3:\n"));
+ break;
+ case SOUND_MIXER_PRIVATE4:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE4:\n"));
+ break;
+ case SOUND_MIXER_PRIVATE5:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_PRIVATE5:\n"));
+ break;
+ case SOUND_MIXER_INFO:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_INFO:\n"));
+ break;
+ case SOUND_OLD_MIXER_INFO:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_OLD_MIXER_INFO:\n"));
+ break;
+
+ default:
+ switch (_IOC_NR(x)) {
+ case SOUND_MIXER_VOLUME:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_VOLUME:\n"));
+ break;
+ case SOUND_MIXER_SPEAKER:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_SPEAKER:\n"));
+ break;
+ case SOUND_MIXER_RECLEV:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_RECLEV:\n"));
+ break;
+ case SOUND_MIXER_MIC:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_MIC:\n"));
+ break;
+ case SOUND_MIXER_SYNTH:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_SYNTH:\n"));
+ break;
+ case SOUND_MIXER_RECSRC:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_RECSRC:\n"));
+ break;
+ case SOUND_MIXER_DEVMASK:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_DEVMASK:\n"));
+ break;
+ case SOUND_MIXER_RECMASK:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_RECMASK:\n"));
+ break;
+ case SOUND_MIXER_STEREODEVS:
+ CS_DBGOUT(CS_IOCTL, 4,
+ printk("SOUND_MIXER_STEREODEVS:\n"));
+ break;
+ case SOUND_MIXER_CAPS:
+ CS_DBGOUT(CS_IOCTL, 4, printk("SOUND_MIXER_CAPS:\n"));
+ break;
+ default:
+ i = _IOC_NR(x);
+ if (i >= SOUND_MIXER_NRDEVICES
+ || !(vidx = mixtable1[i])) {
+ CS_DBGOUT(CS_IOCTL, 4, printk
+ ("UNKNOWN IOCTL: 0x%.8x NR=%d\n",
+ x, i));
+ } else {
+ CS_DBGOUT(CS_IOCTL, 4, printk
+ ("SOUND_MIXER_IOCTL AC9x: 0x%.8x NR=%d\n",
+ x, i));
+ }
+ break;
+ }
+ }
+}
+#endif
+
+
+static int ser_init(struct cs4297a_state *s)
+{
+ int i;
+
+ CS_DBGOUT(CS_INIT, 2,
+ printk(KERN_INFO "cs4297a: Setting up serial parameters\n"));
+
+ __raw_writeq(M_SYNCSER_CMD_RX_RESET | M_SYNCSER_CMD_TX_RESET, SS_CSR(R_SER_CMD));
+
+ __raw_writeq(M_SYNCSER_MSB_FIRST, SS_CSR(R_SER_MODE));
+ __raw_writeq(32, SS_CSR(R_SER_MINFRM_SZ));
+ __raw_writeq(32, SS_CSR(R_SER_MAXFRM_SZ));
+
+ __raw_writeq(1, SS_CSR(R_SER_TX_RD_THRSH));
+ __raw_writeq(4, SS_CSR(R_SER_TX_WR_THRSH));
+ __raw_writeq(8, SS_CSR(R_SER_RX_RD_THRSH));
+
+ /* This looks good from experimentation */
+ __raw_writeq((M_SYNCSER_TXSYNC_INT | V_SYNCSER_TXSYNC_DLY(0) | M_SYNCSER_TXCLK_EXT |
+ M_SYNCSER_RXSYNC_INT | V_SYNCSER_RXSYNC_DLY(1) | M_SYNCSER_RXCLK_EXT | M_SYNCSER_RXSYNC_EDGE),
+ SS_CSR(R_SER_LINE_MODE));
+
+ /* This looks good from experimentation */
+ __raw_writeq(V_SYNCSER_SEQ_COUNT(14) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_STROBE,
+ SS_TXTBL(0));
+ __raw_writeq(V_SYNCSER_SEQ_COUNT(15) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE,
+ SS_TXTBL(1));
+ __raw_writeq(V_SYNCSER_SEQ_COUNT(13) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE,
+ SS_TXTBL(2));
+ __raw_writeq(V_SYNCSER_SEQ_COUNT( 0) | M_SYNCSER_SEQ_ENABLE |
+ M_SYNCSER_SEQ_STROBE | M_SYNCSER_SEQ_LAST, SS_TXTBL(3));
+
+ __raw_writeq(V_SYNCSER_SEQ_COUNT(14) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_STROBE,
+ SS_RXTBL(0));
+ __raw_writeq(V_SYNCSER_SEQ_COUNT(15) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE,
+ SS_RXTBL(1));
+ __raw_writeq(V_SYNCSER_SEQ_COUNT(13) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_BYTE,
+ SS_RXTBL(2));
+ __raw_writeq(V_SYNCSER_SEQ_COUNT( 0) | M_SYNCSER_SEQ_ENABLE | M_SYNCSER_SEQ_STROBE |
+ M_SYNCSER_SEQ_LAST, SS_RXTBL(3));
+
+ for (i=4; i<16; i++) {
+ /* Just in case... */
+ __raw_writeq(M_SYNCSER_SEQ_LAST, SS_TXTBL(i));
+ __raw_writeq(M_SYNCSER_SEQ_LAST, SS_RXTBL(i));
+ }
+
+ return 0;
+}
+
+static int init_serdma(serdma_t *dma)
+{
+ CS_DBGOUT(CS_INIT, 2,
+ printk(KERN_ERR "cs4297a: desc - %d sbufsize - %d dbufsize - %d\n",
+ DMA_DESCR, SAMPLE_BUF_SIZE, DMA_BUF_SIZE));
+
+ /* Descriptors */
+ dma->ringsz = DMA_DESCR;
+ dma->descrtab = kzalloc(dma->ringsz * sizeof(serdma_descr_t), GFP_KERNEL);
+ if (!dma->descrtab) {
+ printk(KERN_ERR "cs4297a: kzalloc descrtab failed\n");
+ return -1;
+ }
+ dma->descrtab_end = dma->descrtab + dma->ringsz;
+ /* XXX bloddy mess, use proper DMA API here ... */
+ dma->descrtab_phys = CPHYSADDR((long)dma->descrtab);
+ dma->descr_add = dma->descr_rem = dma->descrtab;
+
+ /* Frame buffer area */
+ dma->dma_buf = kzalloc(DMA_BUF_SIZE, GFP_KERNEL);
+ if (!dma->dma_buf) {
+ printk(KERN_ERR "cs4297a: kzalloc dma_buf failed\n");
+ kfree(dma->descrtab);
+ return -1;
+ }
+ dma->dma_buf_phys = CPHYSADDR((long)dma->dma_buf);
+
+ /* Samples buffer area */
+ dma->sbufsz = SAMPLE_BUF_SIZE;
+ dma->sample_buf = kmalloc(dma->sbufsz, GFP_KERNEL);
+ if (!dma->sample_buf) {
+ printk(KERN_ERR "cs4297a: kmalloc sample_buf failed\n");
+ kfree(dma->descrtab);
+ kfree(dma->dma_buf);
+ return -1;
+ }
+ dma->sb_swptr = dma->sb_hwptr = dma->sample_buf;
+ dma->sb_end = (u16 *)((void *)dma->sample_buf + dma->sbufsz);
+ dma->fragsize = dma->sbufsz >> 1;
+
+ CS_DBGOUT(CS_INIT, 4,
+ printk(KERN_ERR "cs4297a: descrtab - %08x dma_buf - %x sample_buf - %x\n",
+ (int)dma->descrtab, (int)dma->dma_buf,
+ (int)dma->sample_buf));
+
+ return 0;
+}
+
+static int dma_init(struct cs4297a_state *s)
+{
+ int i;
+
+ CS_DBGOUT(CS_INIT, 2,
+ printk(KERN_INFO "cs4297a: Setting up DMA\n"));
+
+ if (init_serdma(&s->dma_adc) ||
+ init_serdma(&s->dma_dac))
+ return -1;
+
+ if (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_RX))||
+ __raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX))) {
+ panic("DMA state corrupted?!");
+ }
+
+ /* Initialize now - the descr/buffer pairings will never
+ change... */
+ for (i=0; i<DMA_DESCR; i++) {
+ s->dma_dac.descrtab[i].descr_a = M_DMA_SERRX_SOP | V_DMA_DSCRA_A_SIZE(1) |
+ (s->dma_dac.dma_buf_phys + i*FRAME_BYTES);
+ s->dma_dac.descrtab[i].descr_b = V_DMA_DSCRB_PKT_SIZE(FRAME_BYTES);
+ s->dma_adc.descrtab[i].descr_a = V_DMA_DSCRA_A_SIZE(1) |
+ (s->dma_adc.dma_buf_phys + i*FRAME_BYTES);
+ s->dma_adc.descrtab[i].descr_b = 0;
+ }
+
+ __raw_writeq((M_DMA_EOP_INT_EN | V_DMA_INT_PKTCNT(DMA_INT_CNT) |
+ V_DMA_RINGSZ(DMA_DESCR) | M_DMA_TDX_EN),
+ SS_CSR(R_SER_DMA_CONFIG0_RX));
+ __raw_writeq(M_DMA_L2CA, SS_CSR(R_SER_DMA_CONFIG1_RX));
+ __raw_writeq(s->dma_adc.descrtab_phys, SS_CSR(R_SER_DMA_DSCR_BASE_RX));
+
+ __raw_writeq(V_DMA_RINGSZ(DMA_DESCR), SS_CSR(R_SER_DMA_CONFIG0_TX));
+ __raw_writeq(M_DMA_L2CA | M_DMA_NO_DSCR_UPDT, SS_CSR(R_SER_DMA_CONFIG1_TX));
+ __raw_writeq(s->dma_dac.descrtab_phys, SS_CSR(R_SER_DMA_DSCR_BASE_TX));
+
+ /* Prep the receive DMA descriptor ring */
+ __raw_writeq(DMA_DESCR, SS_CSR(R_SER_DMA_DSCR_COUNT_RX));
+
+ __raw_writeq(M_SYNCSER_DMA_RX_EN | M_SYNCSER_DMA_TX_EN, SS_CSR(R_SER_DMA_ENABLE));
+
+ __raw_writeq((M_SYNCSER_RX_SYNC_ERR | M_SYNCSER_RX_OVERRUN | M_SYNCSER_RX_EOP_COUNT),
+ SS_CSR(R_SER_INT_MASK));
+
+ /* Enable the rx/tx; let the codec warm up to the sync and
+ start sending good frames before the receive FIFO is
+ enabled */
+ __raw_writeq(M_SYNCSER_CMD_TX_EN, SS_CSR(R_SER_CMD));
+ udelay(1000);
+ __raw_writeq(M_SYNCSER_CMD_RX_EN | M_SYNCSER_CMD_TX_EN, SS_CSR(R_SER_CMD));
+
+ /* XXXKW is this magic? (the "1" part) */
+ while ((__raw_readq(SS_CSR(R_SER_STATUS)) & 0xf1) != 1)
+ ;
+
+ CS_DBGOUT(CS_INIT, 4,
+ printk(KERN_INFO "cs4297a: status: %08x\n",
+ (unsigned int)(__raw_readq(SS_CSR(R_SER_STATUS)) & 0xffffffff)));
+
+ return 0;
+}
+
+static int serdma_reg_access(struct cs4297a_state *s, u64 data)
+{
+ serdma_t *d = &s->dma_dac;
+ u64 *data_p;
+ unsigned swptr;
+ unsigned long flags;
+ serdma_descr_t *descr;
+
+ if (s->reg_request) {
+ printk(KERN_ERR "cs4297a: attempt to issue multiple reg_access\n");
+ return -1;
+ }
+
+ if (s->ena & FMODE_WRITE) {
+ /* Since a writer has the DSP open, we have to mux the
+ request in */
+ s->reg_request = data;
+ interruptible_sleep_on(&s->dma_dac.reg_wait);
+ /* XXXKW how can I deal with the starvation case where
+ the opener isn't writing? */
+ } else {
+ /* Be safe when changing ring pointers */
+ spin_lock_irqsave(&s->lock, flags);
+ if (d->hwptr != d->swptr) {
+ printk(KERN_ERR "cs4297a: reg access found bookkeeping error (hw/sw = %d/%d\n",
+ d->hwptr, d->swptr);
+ spin_unlock_irqrestore(&s->lock, flags);
+ return -1;
+ }
+ swptr = d->swptr;
+ d->hwptr = d->swptr = (d->swptr + 1) % d->ringsz;
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ descr = &d->descrtab[swptr];
+ data_p = &d->dma_buf[swptr * 4];
+ *data_p = cpu_to_be64(data);
+ __raw_writeq(1, SS_CSR(R_SER_DMA_DSCR_COUNT_TX));
+ CS_DBGOUT(CS_DESCR, 4,
+ printk(KERN_INFO "cs4297a: add_tx %p (%x -> %x)\n",
+ data_p, swptr, d->hwptr));
+ }
+
+ CS_DBGOUT(CS_FUNCTION, 6,
+ printk(KERN_INFO "cs4297a: serdma_reg_access()-\n"));
+
+ return 0;
+}
+
+//****************************************************************************
+// "cs4297a_read_ac97" -- Reads an AC97 register
+//****************************************************************************
+static int cs4297a_read_ac97(struct cs4297a_state *s, u32 offset,
+ u32 * value)
+{
+ CS_DBGOUT(CS_AC97, 1,
+ printk(KERN_INFO "cs4297a: read reg %2x\n", offset));
+ if (serdma_reg_access(s, (0xCLL << 60) | (1LL << 47) | ((u64)(offset & 0x7F) << 40)))
+ return -1;
+
+ interruptible_sleep_on(&s->dma_adc.reg_wait);
+ *value = s->read_value;
+ CS_DBGOUT(CS_AC97, 2,
+ printk(KERN_INFO "cs4297a: rdr reg %x -> %x\n", s->read_reg, s->read_value));
+
+ return 0;
+}
+
+
+//****************************************************************************
+// "cs4297a_write_ac97()"-- writes an AC97 register
+//****************************************************************************
+static int cs4297a_write_ac97(struct cs4297a_state *s, u32 offset,
+ u32 value)
+{
+ CS_DBGOUT(CS_AC97, 1,
+ printk(KERN_INFO "cs4297a: write reg %2x -> %04x\n", offset, value));
+ return (serdma_reg_access(s, (0xELL << 60) | ((u64)(offset & 0x7F) << 40) | ((value & 0xffff) << 12)));
+}
+
+static void stop_dac(struct cs4297a_state *s)
+{
+ unsigned long flags;
+
+ CS_DBGOUT(CS_WAVE_WRITE, 3, printk(KERN_INFO "cs4297a: stop_dac():\n"));
+ spin_lock_irqsave(&s->lock, flags);
+ s->ena &= ~FMODE_WRITE;
+#if 0
+ /* XXXKW what do I really want here? My theory for now is
+ that I just flip the "ena" bit, and the interrupt handler
+ will stop processing the xmit channel */
+ __raw_writeq((s->ena & FMODE_READ) ? M_SYNCSER_DMA_RX_EN : 0,
+ SS_CSR(R_SER_DMA_ENABLE));
+#endif
+
+ spin_unlock_irqrestore(&s->lock, flags);
+}
+
+
+static void start_dac(struct cs4297a_state *s)
+{
+ unsigned long flags;
+
+ CS_DBGOUT(CS_FUNCTION, 3, printk(KERN_INFO "cs4297a: start_dac()+\n"));
+ spin_lock_irqsave(&s->lock, flags);
+ if (!(s->ena & FMODE_WRITE) && (s->dma_dac.mapped ||
+ (s->dma_dac.count > 0
+ && s->dma_dac.ready))) {
+ s->ena |= FMODE_WRITE;
+ /* XXXKW what do I really want here? My theory for
+ now is that I just flip the "ena" bit, and the
+ interrupt handler will start processing the xmit
+ channel */
+
+ CS_DBGOUT(CS_WAVE_WRITE | CS_PARMS, 8, printk(KERN_INFO
+ "cs4297a: start_dac(): start dma\n"));
+
+ }
+ spin_unlock_irqrestore(&s->lock, flags);
+ CS_DBGOUT(CS_FUNCTION, 3,
+ printk(KERN_INFO "cs4297a: start_dac()-\n"));
+}
+
+
+static void stop_adc(struct cs4297a_state *s)
+{
+ unsigned long flags;
+
+ CS_DBGOUT(CS_FUNCTION, 3,
+ printk(KERN_INFO "cs4297a: stop_adc()+\n"));
+
+ spin_lock_irqsave(&s->lock, flags);
+ s->ena &= ~FMODE_READ;
+
+ if (s->conversion == 1) {
+ s->conversion = 0;
+ s->prop_adc.fmt = s->prop_adc.fmt_original;
+ }
+ /* Nothing to do really, I need to keep the DMA going
+ XXXKW when do I get here, and is there more I should do? */
+ spin_unlock_irqrestore(&s->lock, flags);
+ CS_DBGOUT(CS_FUNCTION, 3,
+ printk(KERN_INFO "cs4297a: stop_adc()-\n"));
+}
+
+
+static void start_adc(struct cs4297a_state *s)
+{
+ unsigned long flags;
+
+ CS_DBGOUT(CS_FUNCTION, 2,
+ printk(KERN_INFO "cs4297a: start_adc()+\n"));
+
+ if (!(s->ena & FMODE_READ) &&
+ (s->dma_adc.mapped || s->dma_adc.count <=
+ (signed) (s->dma_adc.sbufsz - 2 * s->dma_adc.fragsize))
+ && s->dma_adc.ready) {
+ if (s->prop_adc.fmt & AFMT_S8 || s->prop_adc.fmt & AFMT_U8) {
+ //
+ // now only use 16 bit capture, due to truncation issue
+ // in the chip, noticable distortion occurs.
+ // allocate buffer and then convert from 16 bit to
+ // 8 bit for the user buffer.
+ //
+ s->prop_adc.fmt_original = s->prop_adc.fmt;
+ if (s->prop_adc.fmt & AFMT_S8) {
+ s->prop_adc.fmt &= ~AFMT_S8;
+ s->prop_adc.fmt |= AFMT_S16_LE;
+ }
+ if (s->prop_adc.fmt & AFMT_U8) {
+ s->prop_adc.fmt &= ~AFMT_U8;
+ s->prop_adc.fmt |= AFMT_U16_LE;
+ }
+ //
+ // prog_dmabuf_adc performs a stop_adc() but that is
+ // ok since we really haven't started the DMA yet.
+ //
+ prog_codec(s, CS_TYPE_ADC);
+
+ prog_dmabuf_adc(s);
+ s->conversion = 1;
+ }
+ spin_lock_irqsave(&s->lock, flags);
+ s->ena |= FMODE_READ;
+ /* Nothing to do really, I am probably already
+ DMAing... XXXKW when do I get here, and is there
+ more I should do? */
+ spin_unlock_irqrestore(&s->lock, flags);
+
+ CS_DBGOUT(CS_PARMS, 6, printk(KERN_INFO
+ "cs4297a: start_adc(): start adc\n"));
+ }
+ CS_DBGOUT(CS_FUNCTION, 2,
+ printk(KERN_INFO "cs4297a: start_adc()-\n"));
+
+}
+
+
+// call with spinlock held!
+static void cs4297a_update_ptr(struct cs4297a_state *s, int intflag)
+{
+ int good_diff, diff, diff2;
+ u64 *data_p, data;
+ u32 *s_ptr;
+ unsigned hwptr;
+ u32 status;
+ serdma_t *d;
+ serdma_descr_t *descr;
+
+ // update ADC pointer
+ status = intflag ? __raw_readq(SS_CSR(R_SER_STATUS)) : 0;
+
+ if ((s->ena & FMODE_READ) || (status & (M_SYNCSER_RX_EOP_COUNT))) {
+ d = &s->dma_adc;
+ hwptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_RX)) & M_DMA_CURDSCR_ADDR) -
+ d->descrtab_phys) / sizeof(serdma_descr_t));
+
+ if (s->ena & FMODE_READ) {
+ CS_DBGOUT(CS_FUNCTION, 2,
+ printk(KERN_INFO "cs4297a: upd_rcv sw->hw->hw %x/%x/%x (int-%d)n",
+ d->swptr, d->hwptr, hwptr, intflag));
+ /* Number of DMA buffers available for software: */
+ diff2 = diff = (d->ringsz + hwptr - d->hwptr) % d->ringsz;
+ d->hwptr = hwptr;
+ good_diff = 0;
+ s_ptr = (u32 *)&(d->dma_buf[d->swptr*4]);
+ descr = &d->descrtab[d->swptr];
+ while (diff2--) {
+ u64 data = be64_to_cpu(*(u64 *)s_ptr);
+ u64 descr_a;
+ u16 left, right;
+ descr_a = descr->descr_a;
+ descr->descr_a &= ~M_DMA_SERRX_SOP;
+ if ((descr_a & M_DMA_DSCRA_A_ADDR) != CPHYSADDR((long)s_ptr)) {
+ printk(KERN_ERR "cs4297a: RX Bad address (read)\n");
+ }
+ if (((data & 0x9800000000000000) != 0x9800000000000000) ||
+ (!(descr_a & M_DMA_SERRX_SOP)) ||
+ (G_DMA_DSCRB_PKT_SIZE(descr->descr_b) != FRAME_BYTES)) {
+ s->stats.rx_bad++;
+ printk(KERN_DEBUG "cs4297a: RX Bad attributes (read)\n");
+ continue;
+ }
+ s->stats.rx_good++;
+ if ((data >> 61) == 7) {
+ s->read_value = (data >> 12) & 0xffff;
+ s->read_reg = (data >> 40) & 0x7f;
+ wake_up(&d->reg_wait);
+ }
+ if (d->count && (d->sb_hwptr == d->sb_swptr)) {
+ s->stats.rx_overflow++;
+ printk(KERN_DEBUG "cs4297a: RX overflow\n");
+ continue;
+ }
+ good_diff++;
+ left = ((be32_to_cpu(s_ptr[1]) & 0xff) << 8) |
+ ((be32_to_cpu(s_ptr[2]) >> 24) & 0xff);
+ right = (be32_to_cpu(s_ptr[2]) >> 4) & 0xffff;
+ *d->sb_hwptr++ = cpu_to_be16(left);
+ *d->sb_hwptr++ = cpu_to_be16(right);
+ if (d->sb_hwptr == d->sb_end)
+ d->sb_hwptr = d->sample_buf;
+ descr++;
+ if (descr == d->descrtab_end) {
+ descr = d->descrtab;
+ s_ptr = (u32 *)s->dma_adc.dma_buf;
+ } else {
+ s_ptr += 8;
+ }
+ }
+ d->total_bytes += good_diff * FRAME_SAMPLE_BYTES;
+ d->count += good_diff * FRAME_SAMPLE_BYTES;
+ if (d->count > d->sbufsz) {
+ printk(KERN_ERR "cs4297a: bogus receive overflow!!\n");
+ }
+ d->swptr = (d->swptr + diff) % d->ringsz;
+ __raw_writeq(diff, SS_CSR(R_SER_DMA_DSCR_COUNT_RX));
+ if (d->mapped) {
+ if (d->count >= (signed) d->fragsize)
+ wake_up(&d->wait);
+ } else {
+ if (d->count > 0) {
+ CS_DBGOUT(CS_WAVE_READ, 4,
+ printk(KERN_INFO
+ "cs4297a: update count -> %d\n", d->count));
+ wake_up(&d->wait);
+ }
+ }
+ } else {
+ /* Receive is going even if no one is
+ listening (for register accesses and to
+ avoid FIFO overrun) */
+ diff2 = diff = (hwptr + d->ringsz - d->hwptr) % d->ringsz;
+ if (!diff) {
+ printk(KERN_ERR "cs4297a: RX full or empty?\n");
+ }
+
+ descr = &d->descrtab[d->swptr];
+ data_p = &d->dma_buf[d->swptr*4];
+
+ /* Force this to happen at least once; I got
+ here because of an interrupt, so there must
+ be a buffer to process. */
+ do {
+ data = be64_to_cpu(*data_p);
+ if ((descr->descr_a & M_DMA_DSCRA_A_ADDR) != CPHYSADDR((long)data_p)) {
+ printk(KERN_ERR "cs4297a: RX Bad address %d (%llx %lx)\n", d->swptr,
+ (long long)(descr->descr_a & M_DMA_DSCRA_A_ADDR),
+ (long)CPHYSADDR((long)data_p));
+ }
+ if (!(data & (1LL << 63)) ||
+ !(descr->descr_a & M_DMA_SERRX_SOP) ||
+ (G_DMA_DSCRB_PKT_SIZE(descr->descr_b) != FRAME_BYTES)) {
+ s->stats.rx_bad++;
+ printk(KERN_DEBUG "cs4297a: RX Bad attributes\n");
+ } else {
+ s->stats.rx_good++;
+ if ((data >> 61) == 7) {
+ s->read_value = (data >> 12) & 0xffff;
+ s->read_reg = (data >> 40) & 0x7f;
+ wake_up(&d->reg_wait);
+ }
+ }
+ descr->descr_a &= ~M_DMA_SERRX_SOP;
+ descr++;
+ d->swptr++;
+ data_p += 4;
+ if (descr == d->descrtab_end) {
+ descr = d->descrtab;
+ d->swptr = 0;
+ data_p = d->dma_buf;
+ }
+ __raw_writeq(1, SS_CSR(R_SER_DMA_DSCR_COUNT_RX));
+ } while (--diff);
+ d->hwptr = hwptr;
+
+ CS_DBGOUT(CS_DESCR, 6,
+ printk(KERN_INFO "cs4297a: hw/sw %x/%x\n", d->hwptr, d->swptr));
+ }
+
+ CS_DBGOUT(CS_PARMS, 8, printk(KERN_INFO
+ "cs4297a: cs4297a_update_ptr(): s=0x%.8x hwptr=%d total_bytes=%d count=%d \n",
+ (unsigned)s, d->hwptr,
+ d->total_bytes, d->count));
+ }
+
+ /* XXXKW worry about s->reg_request -- there is a starvation
+ case if s->ena has FMODE_WRITE on, but the client isn't
+ doing writes */
+
+ // update DAC pointer
+ //
+ // check for end of buffer, means that we are going to wait for another interrupt
+ // to allow silence to fill the fifos on the part, to keep pops down to a minimum.
+ //
+ if (s->ena & FMODE_WRITE) {
+ serdma_t *d = &s->dma_dac;
+ hwptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) -
+ d->descrtab_phys) / sizeof(serdma_descr_t));
+ diff = (d->ringsz + hwptr - d->hwptr) % d->ringsz;
+ CS_DBGOUT(CS_WAVE_WRITE, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_update_ptr(): hw/hw/sw %x/%x/%x diff %d count %d\n",
+ d->hwptr, hwptr, d->swptr, diff, d->count));
+ d->hwptr = hwptr;
+ /* XXXKW stereo? conversion? Just assume 2 16-bit samples for now */
+ d->total_bytes += diff * FRAME_SAMPLE_BYTES;
+ if (d->mapped) {
+ d->count += diff * FRAME_SAMPLE_BYTES;
+ if (d->count >= d->fragsize) {
+ d->wakeup = 1;
+ wake_up(&d->wait);
+ if (d->count > d->sbufsz)
+ d->count &= d->sbufsz - 1;
+ }
+ } else {
+ d->count -= diff * FRAME_SAMPLE_BYTES;
+ if (d->count <= 0) {
+ //
+ // fill with silence, and do not shut down the DAC.
+ // Continue to play silence until the _release.
+ //
+ CS_DBGOUT(CS_WAVE_WRITE, 6, printk(KERN_INFO
+ "cs4297a: cs4297a_update_ptr(): memset %d at 0x%.8x for %d size \n",
+ (unsigned)(s->prop_dac.fmt &
+ (AFMT_U8 | AFMT_U16_LE)) ? 0x80 : 0,
+ (unsigned)d->dma_buf,
+ d->ringsz));
+ memset(d->dma_buf, 0, d->ringsz * FRAME_BYTES);
+ if (d->count < 0) {
+ d->underrun = 1;
+ s->stats.tx_underrun++;
+ d->count = 0;
+ CS_DBGOUT(CS_ERROR, 9, printk(KERN_INFO
+ "cs4297a: cs4297a_update_ptr(): underrun\n"));
+ }
+ } else if (d->count <=
+ (signed) d->fragsize
+ && !d->endcleared) {
+ /* XXXKW what is this for? */
+ clear_advance(d->dma_buf,
+ d->sbufsz,
+ d->swptr,
+ d->fragsize,
+ 0);
+ d->endcleared = 1;
+ }
+ if ( (d->count <= (signed) d->sbufsz/2) || intflag)
+ {
+ CS_DBGOUT(CS_WAVE_WRITE, 4,
+ printk(KERN_INFO
+ "cs4297a: update count -> %d\n", d->count));
+ wake_up(&d->wait);
+ }
+ }
+ CS_DBGOUT(CS_PARMS, 8, printk(KERN_INFO
+ "cs4297a: cs4297a_update_ptr(): s=0x%.8x hwptr=%d total_bytes=%d count=%d \n",
+ (unsigned) s, d->hwptr,
+ d->total_bytes, d->count));
+ }
+}
+
+static int mixer_ioctl(struct cs4297a_state *s, unsigned int cmd,
+ unsigned long arg)
+{
+ // Index to mixer_src[] is value of AC97 Input Mux Select Reg.
+ // Value of array member is recording source Device ID Mask.
+ static const unsigned int mixer_src[8] = {
+ SOUND_MASK_MIC, SOUND_MASK_CD, 0, SOUND_MASK_LINE1,
+ SOUND_MASK_LINE, SOUND_MASK_VOLUME, 0, 0
+ };
+
+ // Index of mixtable1[] member is Device ID
+ // and must be <= SOUND_MIXER_NRDEVICES.
+ // Value of array member is index into s->mix.vol[]
+ static const unsigned char mixtable1[SOUND_MIXER_NRDEVICES] = {
+ [SOUND_MIXER_PCM] = 1, // voice
+ [SOUND_MIXER_LINE1] = 2, // AUX
+ [SOUND_MIXER_CD] = 3, // CD
+ [SOUND_MIXER_LINE] = 4, // Line
+ [SOUND_MIXER_SYNTH] = 5, // FM
+ [SOUND_MIXER_MIC] = 6, // Mic
+ [SOUND_MIXER_SPEAKER] = 7, // Speaker
+ [SOUND_MIXER_RECLEV] = 8, // Recording level
+ [SOUND_MIXER_VOLUME] = 9 // Master Volume
+ };
+
+ static const unsigned mixreg[] = {
+ AC97_PCMOUT_VOL,
+ AC97_AUX_VOL,
+ AC97_CD_VOL,
+ AC97_LINEIN_VOL
+ };
+ unsigned char l, r, rl, rr, vidx;
+ unsigned char attentbl[11] =
+ { 63, 42, 26, 17, 14, 11, 8, 6, 4, 2, 0 };
+ unsigned temp1;
+ int i, val;
+
+ VALIDATE_STATE(s);
+ CS_DBGOUT(CS_FUNCTION, 4, printk(KERN_INFO
+ "cs4297a: mixer_ioctl(): s=0x%.8x cmd=0x%.8x\n",
+ (unsigned) s, cmd));
+#if CSDEBUG
+ cs_printioctl(cmd);
+#endif
+#if CSDEBUG_INTERFACE
+
+ if ((cmd == SOUND_MIXER_CS_GETDBGMASK) ||
+ (cmd == SOUND_MIXER_CS_SETDBGMASK) ||
+ (cmd == SOUND_MIXER_CS_GETDBGLEVEL) ||
+ (cmd == SOUND_MIXER_CS_SETDBGLEVEL))
+ {
+ switch (cmd) {
+
+ case SOUND_MIXER_CS_GETDBGMASK:
+ return put_user(cs_debugmask,
+ (unsigned long *) arg);
+
+ case SOUND_MIXER_CS_GETDBGLEVEL:
+ return put_user(cs_debuglevel,
+ (unsigned long *) arg);
+
+ case SOUND_MIXER_CS_SETDBGMASK:
+ if (get_user(val, (unsigned long *) arg))
+ return -EFAULT;
+ cs_debugmask = val;
+ return 0;
+
+ case SOUND_MIXER_CS_SETDBGLEVEL:
+ if (get_user(val, (unsigned long *) arg))
+ return -EFAULT;
+ cs_debuglevel = val;
+ return 0;
+ default:
+ CS_DBGOUT(CS_ERROR, 1, printk(KERN_INFO
+ "cs4297a: mixer_ioctl(): ERROR unknown debug cmd\n"));
+ return 0;
+ }
+ }
+#endif
+
+ if (cmd == SOUND_MIXER_PRIVATE1) {
+ return -EINVAL;
+ }
+ if (cmd == SOUND_MIXER_PRIVATE2) {
+ // enable/disable/query spatializer
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (val != -1) {
+ temp1 = (val & 0x3f) >> 2;
+ cs4297a_write_ac97(s, AC97_3D_CONTROL, temp1);
+ cs4297a_read_ac97(s, AC97_GENERAL_PURPOSE,
+ &temp1);
+ cs4297a_write_ac97(s, AC97_GENERAL_PURPOSE,
+ temp1 | 0x2000);
+ }
+ cs4297a_read_ac97(s, AC97_3D_CONTROL, &temp1);
+ return put_user((temp1 << 2) | 3, (int *) arg);
+ }
+ if (cmd == SOUND_MIXER_INFO) {
+ mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, "CS4297a", sizeof(info.id));
+ strlcpy(info.name, "Crystal CS4297a", sizeof(info.name));
+ info.modify_counter = s->mix.modcnt;
+ if (copy_to_user((void *) arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ }
+ if (cmd == SOUND_OLD_MIXER_INFO) {
+ _old_mixer_info info;
+ memset(&info, 0, sizeof(info));
+ strlcpy(info.id, "CS4297a", sizeof(info.id));
+ strlcpy(info.name, "Crystal CS4297a", sizeof(info.name));
+ if (copy_to_user((void *) arg, &info, sizeof(info)))
+ return -EFAULT;
+ return 0;
+ }
+ if (cmd == OSS_GETVERSION)
+ return put_user(SOUND_VERSION, (int *) arg);
+
+ if (_IOC_TYPE(cmd) != 'M' || _SIOC_SIZE(cmd) != sizeof(int))
+ return -EINVAL;
+
+ // If ioctl has only the SIOC_READ bit(bit 31)
+ // on, process the only-read commands.
+ if (_SIOC_DIR(cmd) == _SIOC_READ) {
+ switch (_IOC_NR(cmd)) {
+ case SOUND_MIXER_RECSRC: // Arg contains a bit for each recording source
+ cs4297a_read_ac97(s, AC97_RECORD_SELECT,
+ &temp1);
+ return put_user(mixer_src[temp1 & 7], (int *) arg);
+
+ case SOUND_MIXER_DEVMASK: // Arg contains a bit for each supported device
+ return put_user(SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_VOLUME | SOUND_MASK_RECLEV,
+ (int *) arg);
+
+ case SOUND_MIXER_RECMASK: // Arg contains a bit for each supported recording source
+ return put_user(SOUND_MASK_LINE | SOUND_MASK_VOLUME,
+ (int *) arg);
+
+ case SOUND_MIXER_STEREODEVS: // Mixer channels supporting stereo
+ return put_user(SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_VOLUME | SOUND_MASK_RECLEV,
+ (int *) arg);
+
+ case SOUND_MIXER_CAPS:
+ return put_user(SOUND_CAP_EXCL_INPUT, (int *) arg);
+
+ default:
+ i = _IOC_NR(cmd);
+ if (i >= SOUND_MIXER_NRDEVICES
+ || !(vidx = mixtable1[i]))
+ return -EINVAL;
+ return put_user(s->mix.vol[vidx - 1], (int *) arg);
+ }
+ }
+ // If ioctl doesn't have both the SIOC_READ and
+ // the SIOC_WRITE bit set, return invalid.
+ if (_SIOC_DIR(cmd) != (_SIOC_READ | _SIOC_WRITE))
+ return -EINVAL;
+
+ // Increment the count of volume writes.
+ s->mix.modcnt++;
+
+ // Isolate the command; it must be a write.
+ switch (_IOC_NR(cmd)) {
+
+ case SOUND_MIXER_RECSRC: // Arg contains a bit for each recording source
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ i = hweight32(val); // i = # bits on in val.
+ if (i != 1) // One & only 1 bit must be on.
+ return 0;
+ for (i = 0; i < sizeof(mixer_src) / sizeof(int); i++) {
+ if (val == mixer_src[i]) {
+ temp1 = (i << 8) | i;
+ cs4297a_write_ac97(s,
+ AC97_RECORD_SELECT,
+ temp1);
+ return 0;
+ }
+ }
+ return 0;
+
+ case SOUND_MIXER_VOLUME:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ l = val & 0xff;
+ if (l > 100)
+ l = 100; // Max soundcard.h vol is 100.
+ if (l < 6) {
+ rl = 63;
+ l = 0;
+ } else
+ rl = attentbl[(10 * l) / 100]; // Convert 0-100 vol to 63-0 atten.
+
+ r = (val >> 8) & 0xff;
+ if (r > 100)
+ r = 100; // Max right volume is 100, too
+ if (r < 6) {
+ rr = 63;
+ r = 0;
+ } else
+ rr = attentbl[(10 * r) / 100]; // Convert volume to attenuation.
+
+ if ((rl > 60) && (rr > 60)) // If both l & r are 'low',
+ temp1 = 0x8000; // turn on the mute bit.
+ else
+ temp1 = 0;
+
+ temp1 |= (rl << 8) | rr;
+
+ cs4297a_write_ac97(s, AC97_MASTER_VOL_STEREO, temp1);
+ cs4297a_write_ac97(s, AC97_PHONE_VOL, temp1);
+
+#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS
+ s->mix.vol[8] = ((unsigned int) r << 8) | l;
+#else
+ s->mix.vol[8] = val;
+#endif
+ return put_user(s->mix.vol[8], (int *) arg);
+
+ case SOUND_MIXER_SPEAKER:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ l = val & 0xff;
+ if (l > 100)
+ l = 100;
+ if (l < 3) {
+ rl = 0;
+ l = 0;
+ } else {
+ rl = (l * 2 - 5) / 13; // Convert 0-100 range to 0-15.
+ l = (rl * 13 + 5) / 2;
+ }
+
+ if (rl < 3) {
+ temp1 = 0x8000;
+ rl = 0;
+ } else
+ temp1 = 0;
+ rl = 15 - rl; // Convert volume to attenuation.
+ temp1 |= rl << 1;
+ cs4297a_write_ac97(s, AC97_PCBEEP_VOL, temp1);
+
+#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS
+ s->mix.vol[6] = l << 8;
+#else
+ s->mix.vol[6] = val;
+#endif
+ return put_user(s->mix.vol[6], (int *) arg);
+
+ case SOUND_MIXER_RECLEV:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ l = val & 0xff;
+ if (l > 100)
+ l = 100;
+ r = (val >> 8) & 0xff;
+ if (r > 100)
+ r = 100;
+ rl = (l * 2 - 5) / 13; // Convert 0-100 scale to 0-15.
+ rr = (r * 2 - 5) / 13;
+ if (rl < 3 && rr < 3)
+ temp1 = 0x8000;
+ else
+ temp1 = 0;
+
+ temp1 = temp1 | (rl << 8) | rr;
+ cs4297a_write_ac97(s, AC97_RECORD_GAIN, temp1);
+
+#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS
+ s->mix.vol[7] = ((unsigned int) r << 8) | l;
+#else
+ s->mix.vol[7] = val;
+#endif
+ return put_user(s->mix.vol[7], (int *) arg);
+
+ case SOUND_MIXER_MIC:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ l = val & 0xff;
+ if (l > 100)
+ l = 100;
+ if (l < 1) {
+ l = 0;
+ rl = 0;
+ } else {
+ rl = ((unsigned) l * 5 - 4) / 16; // Convert 0-100 range to 0-31.
+ l = (rl * 16 + 4) / 5;
+ }
+ cs4297a_read_ac97(s, AC97_MIC_VOL, &temp1);
+ temp1 &= 0x40; // Isolate 20db gain bit.
+ if (rl < 3) {
+ temp1 |= 0x8000;
+ rl = 0;
+ }
+ rl = 31 - rl; // Convert volume to attenuation.
+ temp1 |= rl;
+ cs4297a_write_ac97(s, AC97_MIC_VOL, temp1);
+
+#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS
+ s->mix.vol[5] = val << 8;
+#else
+ s->mix.vol[5] = val;
+#endif
+ return put_user(s->mix.vol[5], (int *) arg);
+
+
+ case SOUND_MIXER_SYNTH:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ l = val & 0xff;
+ if (l > 100)
+ l = 100;
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ r = (val >> 8) & 0xff;
+ if (r > 100)
+ r = 100;
+ rl = (l * 2 - 11) / 3; // Convert 0-100 range to 0-63.
+ rr = (r * 2 - 11) / 3;
+ if (rl < 3) // If l is low, turn on
+ temp1 = 0x0080; // the mute bit.
+ else
+ temp1 = 0;
+
+ rl = 63 - rl; // Convert vol to attenuation.
+// writel(temp1 | rl, s->pBA0 + FMLVC);
+ if (rr < 3) // If rr is low, turn on
+ temp1 = 0x0080; // the mute bit.
+ else
+ temp1 = 0;
+ rr = 63 - rr; // Convert vol to attenuation.
+// writel(temp1 | rr, s->pBA0 + FMRVC);
+
+#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS
+ s->mix.vol[4] = (r << 8) | l;
+#else
+ s->mix.vol[4] = val;
+#endif
+ return put_user(s->mix.vol[4], (int *) arg);
+
+
+ default:
+ CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO
+ "cs4297a: mixer_ioctl(): default\n"));
+
+ i = _IOC_NR(cmd);
+ if (i >= SOUND_MIXER_NRDEVICES || !(vidx = mixtable1[i]))
+ return -EINVAL;
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ l = val & 0xff;
+ if (l > 100)
+ l = 100;
+ if (l < 1) {
+ l = 0;
+ rl = 31;
+ } else
+ rl = (attentbl[(l * 10) / 100]) >> 1;
+
+ r = (val >> 8) & 0xff;
+ if (r > 100)
+ r = 100;
+ if (r < 1) {
+ r = 0;
+ rr = 31;
+ } else
+ rr = (attentbl[(r * 10) / 100]) >> 1;
+ if ((rl > 30) && (rr > 30))
+ temp1 = 0x8000;
+ else
+ temp1 = 0;
+ temp1 = temp1 | (rl << 8) | rr;
+ cs4297a_write_ac97(s, mixreg[vidx - 1], temp1);
+
+#ifdef OSS_DOCUMENTED_MIXER_SEMANTICS
+ s->mix.vol[vidx - 1] = ((unsigned int) r << 8) | l;
+#else
+ s->mix.vol[vidx - 1] = val;
+#endif
+ return put_user(s->mix.vol[vidx - 1], (int *) arg);
+ }
+}
+
+
+// ---------------------------------------------------------------------
+
+static int cs4297a_open_mixdev(struct inode *inode, struct file *file)
+{
+ int minor = iminor(inode);
+ struct cs4297a_state *s=NULL;
+ struct list_head *entry;
+
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4,
+ printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()+\n"));
+
+ list_for_each(entry, &cs4297a_devs)
+ {
+ s = list_entry(entry, struct cs4297a_state, list);
+ if(s->dev_mixer == minor)
+ break;
+ }
+ if (!s)
+ {
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2,
+ printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- -ENODEV\n"));
+ return -ENODEV;
+ }
+ VALIDATE_STATE(s);
+ file->private_data = s;
+
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4,
+ printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- 0\n"));
+
+ return nonseekable_open(inode, file);
+}
+
+
+static int cs4297a_release_mixdev(struct inode *inode, struct file *file)
+{
+ struct cs4297a_state *s =
+ (struct cs4297a_state *) file->private_data;
+
+ VALIDATE_STATE(s);
+ return 0;
+}
+
+
+static int cs4297a_ioctl_mixdev(struct inode *inode, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return mixer_ioctl((struct cs4297a_state *) file->private_data, cmd,
+ arg);
+}
+
+
+// ******************************************************************************************
+// Mixer file operations struct.
+// ******************************************************************************************
+static const struct file_operations cs4297a_mixer_fops = {
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .ioctl = cs4297a_ioctl_mixdev,
+ .open = cs4297a_open_mixdev,
+ .release = cs4297a_release_mixdev,
+};
+
+// ---------------------------------------------------------------------
+
+
+static int drain_adc(struct cs4297a_state *s, int nonblock)
+{
+ /* This routine serves no purpose currently - any samples
+ sitting in the receive queue will just be processed by the
+ background consumer. This would be different if DMA
+ actually stopped when there were no clients. */
+ return 0;
+}
+
+static int drain_dac(struct cs4297a_state *s, int nonblock)
+{
+ DECLARE_WAITQUEUE(wait, current);
+ unsigned long flags;
+ unsigned hwptr;
+ unsigned tmo;
+ int count;
+
+ if (s->dma_dac.mapped)
+ return 0;
+ if (nonblock)
+ return -EBUSY;
+ add_wait_queue(&s->dma_dac.wait, &wait);
+ while ((count = __raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX))) ||
+ (s->dma_dac.count > 0)) {
+ if (!signal_pending(current)) {
+ set_current_state(TASK_INTERRUPTIBLE);
+ /* XXXKW is this calculation working? */
+ tmo = ((count * FRAME_TX_US) * HZ) / 1000000;
+ schedule_timeout(tmo + 1);
+ } else {
+ /* XXXKW do I care if there is a signal pending? */
+ }
+ }
+ spin_lock_irqsave(&s->lock, flags);
+ /* Reset the bookkeeping */
+ hwptr = (int)(((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) -
+ s->dma_dac.descrtab_phys) / sizeof(serdma_descr_t));
+ s->dma_dac.hwptr = s->dma_dac.swptr = hwptr;
+ spin_unlock_irqrestore(&s->lock, flags);
+ remove_wait_queue(&s->dma_dac.wait, &wait);
+ current->state = TASK_RUNNING;
+ return 0;
+}
+
+
+// ---------------------------------------------------------------------
+
+static ssize_t cs4297a_read(struct file *file, char *buffer, size_t count,
+ loff_t * ppos)
+{
+ struct cs4297a_state *s =
+ (struct cs4297a_state *) file->private_data;
+ ssize_t ret;
+ unsigned long flags;
+ int cnt, count_fr, cnt_by;
+ unsigned copied = 0;
+
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_READ, 2,
+ printk(KERN_INFO "cs4297a: cs4297a_read()+ %d \n", count));
+
+ VALIDATE_STATE(s);
+ if (s->dma_adc.mapped)
+ return -ENXIO;
+ if (!s->dma_adc.ready && (ret = prog_dmabuf_adc(s)))
+ return ret;
+ if (!access_ok(VERIFY_WRITE, buffer, count))
+ return -EFAULT;
+ ret = 0;
+//
+// "count" is the amount of bytes to read (from app), is decremented each loop
+// by the amount of bytes that have been returned to the user buffer.
+// "cnt" is the running total of each read from the buffer (changes each loop)
+// "buffer" points to the app's buffer
+// "ret" keeps a running total of the amount of bytes that have been copied
+// to the user buffer.
+// "copied" is the total bytes copied into the user buffer for each loop.
+//
+ while (count > 0) {
+ CS_DBGOUT(CS_WAVE_READ, 8, printk(KERN_INFO
+ "_read() count>0 count=%d .count=%d .swptr=%d .hwptr=%d \n",
+ count, s->dma_adc.count,
+ s->dma_adc.swptr, s->dma_adc.hwptr));
+ spin_lock_irqsave(&s->lock, flags);
+
+ /* cnt will be the number of available samples (16-bit
+ stereo); it starts out as the maxmimum consequetive
+ samples */
+ cnt = (s->dma_adc.sb_end - s->dma_adc.sb_swptr) / 2;
+ count_fr = s->dma_adc.count / FRAME_SAMPLE_BYTES;
+
+ // dma_adc.count is the current total bytes that have not been read.
+ // if the amount of unread bytes from the current sw pointer to the
+ // end of the buffer is greater than the current total bytes that
+ // have not been read, then set the "cnt" (unread bytes) to the
+ // amount of unread bytes.
+
+ if (count_fr < cnt)
+ cnt = count_fr;
+ cnt_by = cnt * FRAME_SAMPLE_BYTES;
+ spin_unlock_irqrestore(&s->lock, flags);
+ //
+ // if we are converting from 8/16 then we need to copy
+ // twice the number of 16 bit bytes then 8 bit bytes.
+ //
+ if (s->conversion) {
+ if (cnt_by > (count * 2)) {
+ cnt = (count * 2) / FRAME_SAMPLE_BYTES;
+ cnt_by = count * 2;
+ }
+ } else {
+ if (cnt_by > count) {
+ cnt = count / FRAME_SAMPLE_BYTES;
+ cnt_by = count;
+ }
+ }
+ //
+ // "cnt" NOW is the smaller of the amount that will be read,
+ // and the amount that is requested in this read (or partial).
+ // if there are no bytes in the buffer to read, then start the
+ // ADC and wait for the interrupt handler to wake us up.
+ //
+ if (cnt <= 0) {
+
+ // start up the dma engine and then continue back to the top of
+ // the loop when wake up occurs.
+ start_adc(s);
+ if (file->f_flags & O_NONBLOCK)
+ return ret ? ret : -EAGAIN;
+ interruptible_sleep_on(&s->dma_adc.wait);
+ if (signal_pending(current))
+ return ret ? ret : -ERESTARTSYS;
+ continue;
+ }
+ // there are bytes in the buffer to read.
+ // copy from the hw buffer over to the user buffer.
+ // user buffer is designated by "buffer"
+ // virtual address to copy from is dma_buf+swptr
+ // the "cnt" is the number of bytes to read.
+
+ CS_DBGOUT(CS_WAVE_READ, 2, printk(KERN_INFO
+ "_read() copy_to cnt=%d count=%d ", cnt_by, count));
+ CS_DBGOUT(CS_WAVE_READ, 8, printk(KERN_INFO
+ " .sbufsz=%d .count=%d buffer=0x%.8x ret=%d\n",
+ s->dma_adc.sbufsz, s->dma_adc.count,
+ (unsigned) buffer, ret));
+
+ if (copy_to_user (buffer, ((void *)s->dma_adc.sb_swptr), cnt_by))
+ return ret ? ret : -EFAULT;
+ copied = cnt_by;
+
+ /* Return the descriptors */
+ spin_lock_irqsave(&s->lock, flags);
+ CS_DBGOUT(CS_FUNCTION, 2,
+ printk(KERN_INFO "cs4297a: upd_rcv sw->hw %x/%x\n", s->dma_adc.swptr, s->dma_adc.hwptr));
+ s->dma_adc.count -= cnt_by;
+ s->dma_adc.sb_swptr += cnt * 2;
+ if (s->dma_adc.sb_swptr == s->dma_adc.sb_end)
+ s->dma_adc.sb_swptr = s->dma_adc.sample_buf;
+ spin_unlock_irqrestore(&s->lock, flags);
+ count -= copied;
+ buffer += copied;
+ ret += copied;
+ start_adc(s);
+ }
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_READ, 2,
+ printk(KERN_INFO "cs4297a: cs4297a_read()- %d\n", ret));
+ return ret;
+}
+
+
+static ssize_t cs4297a_write(struct file *file, const char *buffer,
+ size_t count, loff_t * ppos)
+{
+ struct cs4297a_state *s =
+ (struct cs4297a_state *) file->private_data;
+ ssize_t ret;
+ unsigned long flags;
+ unsigned swptr, hwptr;
+ int cnt;
+
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE, 2,
+ printk(KERN_INFO "cs4297a: cs4297a_write()+ count=%d\n",
+ count));
+ VALIDATE_STATE(s);
+
+ if (s->dma_dac.mapped)
+ return -ENXIO;
+ if (!s->dma_dac.ready && (ret = prog_dmabuf_dac(s)))
+ return ret;
+ if (!access_ok(VERIFY_READ, buffer, count))
+ return -EFAULT;
+ ret = 0;
+ while (count > 0) {
+ serdma_t *d = &s->dma_dac;
+ int copy_cnt;
+ u32 *s_tmpl;
+ u32 *t_tmpl;
+ u32 left, right;
+ int swap = (s->prop_dac.fmt == AFMT_S16_LE) || (s->prop_dac.fmt == AFMT_U16_LE);
+
+ /* XXXXXX this is broken for BLOAT_FACTOR */
+ spin_lock_irqsave(&s->lock, flags);
+ if (d->count < 0) {
+ d->count = 0;
+ d->swptr = d->hwptr;
+ }
+ if (d->underrun) {
+ d->underrun = 0;
+ hwptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) -
+ d->descrtab_phys) / sizeof(serdma_descr_t));
+ d->swptr = d->hwptr = hwptr;
+ }
+ swptr = d->swptr;
+ cnt = d->sbufsz - (swptr * FRAME_SAMPLE_BYTES);
+ /* Will this write fill up the buffer? */
+ if (d->count + cnt > d->sbufsz)
+ cnt = d->sbufsz - d->count;
+ spin_unlock_irqrestore(&s->lock, flags);
+ if (cnt > count)
+ cnt = count;
+ if (cnt <= 0) {
+ start_dac(s);
+ if (file->f_flags & O_NONBLOCK)
+ return ret ? ret : -EAGAIN;
+ interruptible_sleep_on(&d->wait);
+ if (signal_pending(current))
+ return ret ? ret : -ERESTARTSYS;
+ continue;
+ }
+ if (copy_from_user(d->sample_buf, buffer, cnt))
+ return ret ? ret : -EFAULT;
+
+ copy_cnt = cnt;
+ s_tmpl = (u32 *)d->sample_buf;
+ t_tmpl = (u32 *)(d->dma_buf + (swptr * 4));
+
+ /* XXXKW assuming 16-bit stereo! */
+ do {
+ u32 tmp;
+
+ t_tmpl[0] = cpu_to_be32(0x98000000);
+
+ tmp = be32_to_cpu(s_tmpl[0]);
+ left = tmp & 0xffff;
+ right = tmp >> 16;
+ if (swap) {
+ left = swab16(left);
+ right = swab16(right);
+ }
+ t_tmpl[1] = cpu_to_be32(left >> 8);
+ t_tmpl[2] = cpu_to_be32(((left & 0xff) << 24) |
+ (right << 4));
+
+ s_tmpl++;
+ t_tmpl += 8;
+ copy_cnt -= 4;
+ } while (copy_cnt);
+
+ /* Mux in any pending read/write accesses */
+ if (s->reg_request) {
+ *(u64 *)(d->dma_buf + (swptr * 4)) |=
+ cpu_to_be64(s->reg_request);
+ s->reg_request = 0;
+ wake_up(&s->dma_dac.reg_wait);
+ }
+
+ CS_DBGOUT(CS_WAVE_WRITE, 4,
+ printk(KERN_INFO
+ "cs4297a: copy in %d to swptr %x\n", cnt, swptr));
+
+ swptr = (swptr + (cnt/FRAME_SAMPLE_BYTES)) % d->ringsz;
+ __raw_writeq(cnt/FRAME_SAMPLE_BYTES, SS_CSR(R_SER_DMA_DSCR_COUNT_TX));
+ spin_lock_irqsave(&s->lock, flags);
+ d->swptr = swptr;
+ d->count += cnt;
+ d->endcleared = 0;
+ spin_unlock_irqrestore(&s->lock, flags);
+ count -= cnt;
+ buffer += cnt;
+ ret += cnt;
+ start_dac(s);
+ }
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE, 2,
+ printk(KERN_INFO "cs4297a: cs4297a_write()- %d\n", ret));
+ return ret;
+}
+
+
+static unsigned int cs4297a_poll(struct file *file,
+ struct poll_table_struct *wait)
+{
+ struct cs4297a_state *s =
+ (struct cs4297a_state *) file->private_data;
+ unsigned long flags;
+ unsigned int mask = 0;
+
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4,
+ printk(KERN_INFO "cs4297a: cs4297a_poll()+\n"));
+ VALIDATE_STATE(s);
+ if (file->f_mode & FMODE_WRITE) {
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4,
+ printk(KERN_INFO
+ "cs4297a: cs4297a_poll() wait on FMODE_WRITE\n"));
+ if(!s->dma_dac.ready && prog_dmabuf_dac(s))
+ return 0;
+ poll_wait(file, &s->dma_dac.wait, wait);
+ }
+ if (file->f_mode & FMODE_READ) {
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4,
+ printk(KERN_INFO
+ "cs4297a: cs4297a_poll() wait on FMODE_READ\n"));
+ if(!s->dma_dac.ready && prog_dmabuf_adc(s))
+ return 0;
+ poll_wait(file, &s->dma_adc.wait, wait);
+ }
+ spin_lock_irqsave(&s->lock, flags);
+ cs4297a_update_ptr(s,CS_FALSE);
+ if (file->f_mode & FMODE_WRITE) {
+ if (s->dma_dac.mapped) {
+ if (s->dma_dac.count >=
+ (signed) s->dma_dac.fragsize) {
+ if (s->dma_dac.wakeup)
+ mask |= POLLOUT | POLLWRNORM;
+ else
+ mask = 0;
+ s->dma_dac.wakeup = 0;
+ }
+ } else {
+ if ((signed) (s->dma_dac.sbufsz/2) >= s->dma_dac.count)
+ mask |= POLLOUT | POLLWRNORM;
+ }
+ } else if (file->f_mode & FMODE_READ) {
+ if (s->dma_adc.mapped) {
+ if (s->dma_adc.count >= (signed) s->dma_adc.fragsize)
+ mask |= POLLIN | POLLRDNORM;
+ } else {
+ if (s->dma_adc.count > 0)
+ mask |= POLLIN | POLLRDNORM;
+ }
+ }
+ spin_unlock_irqrestore(&s->lock, flags);
+ CS_DBGOUT(CS_FUNCTION | CS_WAVE_WRITE | CS_WAVE_READ, 4,
+ printk(KERN_INFO "cs4297a: cs4297a_poll()- 0x%.8x\n",
+ mask));
+ return mask;
+}
+
+
+static int cs4297a_mmap(struct file *file, struct vm_area_struct *vma)
+{
+ /* XXXKW currently no mmap support */
+ return -EINVAL;
+ return 0;
+}
+
+
+static int cs4297a_ioctl(struct inode *inode, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct cs4297a_state *s =
+ (struct cs4297a_state *) file->private_data;
+ unsigned long flags;
+ audio_buf_info abinfo;
+ count_info cinfo;
+ int val, mapped, ret;
+
+ CS_DBGOUT(CS_FUNCTION|CS_IOCTL, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): file=0x%.8x cmd=0x%.8x\n",
+ (unsigned) file, cmd));
+#if CSDEBUG
+ cs_printioctl(cmd);
+#endif
+ VALIDATE_STATE(s);
+ mapped = ((file->f_mode & FMODE_WRITE) && s->dma_dac.mapped) ||
+ ((file->f_mode & FMODE_READ) && s->dma_adc.mapped);
+ switch (cmd) {
+ case OSS_GETVERSION:
+ CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): SOUND_VERSION=0x%.8x\n",
+ SOUND_VERSION));
+ return put_user(SOUND_VERSION, (int *) arg);
+
+ case SNDCTL_DSP_SYNC:
+ CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_SYNC\n"));
+ if (file->f_mode & FMODE_WRITE)
+ return drain_dac(s,
+ 0 /*file->f_flags & O_NONBLOCK */
+ );
+ return 0;
+
+ case SNDCTL_DSP_SETDUPLEX:
+ return 0;
+
+ case SNDCTL_DSP_GETCAPS:
+ return put_user(DSP_CAP_DUPLEX | DSP_CAP_REALTIME |
+ DSP_CAP_TRIGGER | DSP_CAP_MMAP,
+ (int *) arg);
+
+ case SNDCTL_DSP_RESET:
+ CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_RESET\n"));
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ synchronize_irq(s->irq);
+ s->dma_dac.count = s->dma_dac.total_bytes =
+ s->dma_dac.blocks = s->dma_dac.wakeup = 0;
+ s->dma_dac.swptr = s->dma_dac.hwptr =
+ (int)(((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_TX)) & M_DMA_CURDSCR_ADDR) -
+ s->dma_dac.descrtab_phys) / sizeof(serdma_descr_t));
+ }
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ synchronize_irq(s->irq);
+ s->dma_adc.count = s->dma_adc.total_bytes =
+ s->dma_adc.blocks = s->dma_dac.wakeup = 0;
+ s->dma_adc.swptr = s->dma_adc.hwptr =
+ (int)(((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_RX)) & M_DMA_CURDSCR_ADDR) -
+ s->dma_adc.descrtab_phys) / sizeof(serdma_descr_t));
+ }
+ return 0;
+
+ case SNDCTL_DSP_SPEED:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_SPEED val=%d -> 48000\n", val));
+ val = 48000;
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_STEREO:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_STEREO val=%d\n", val));
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ s->dma_adc.ready = 0;
+ s->prop_adc.channels = val ? 2 : 1;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ s->dma_dac.ready = 0;
+ s->prop_dac.channels = val ? 2 : 1;
+ }
+ return 0;
+
+ case SNDCTL_DSP_CHANNELS:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_CHANNELS val=%d\n",
+ val));
+ if (val != 0) {
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ s->dma_adc.ready = 0;
+ if (val >= 2)
+ s->prop_adc.channels = 2;
+ else
+ s->prop_adc.channels = 1;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ s->dma_dac.ready = 0;
+ if (val >= 2)
+ s->prop_dac.channels = 2;
+ else
+ s->prop_dac.channels = 1;
+ }
+ }
+
+ if (file->f_mode & FMODE_WRITE)
+ val = s->prop_dac.channels;
+ else if (file->f_mode & FMODE_READ)
+ val = s->prop_adc.channels;
+
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_GETFMTS: // Returns a mask
+ CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_GETFMT val=0x%.8x\n",
+ AFMT_S16_LE | AFMT_U16_LE | AFMT_S8 |
+ AFMT_U8));
+ return put_user(AFMT_S16_LE | AFMT_U16_LE | AFMT_S8 |
+ AFMT_U8, (int *) arg);
+
+ case SNDCTL_DSP_SETFMT:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_SETFMT val=0x%.8x\n",
+ val));
+ if (val != AFMT_QUERY) {
+ if (file->f_mode & FMODE_READ) {
+ stop_adc(s);
+ s->dma_adc.ready = 0;
+ if (val != AFMT_S16_LE
+ && val != AFMT_U16_LE && val != AFMT_S8
+ && val != AFMT_U8)
+ val = AFMT_U8;
+ s->prop_adc.fmt = val;
+ s->prop_adc.fmt_original = s->prop_adc.fmt;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ stop_dac(s);
+ s->dma_dac.ready = 0;
+ if (val != AFMT_S16_LE
+ && val != AFMT_U16_LE && val != AFMT_S8
+ && val != AFMT_U8)
+ val = AFMT_U8;
+ s->prop_dac.fmt = val;
+ s->prop_dac.fmt_original = s->prop_dac.fmt;
+ }
+ } else {
+ if (file->f_mode & FMODE_WRITE)
+ val = s->prop_dac.fmt_original;
+ else if (file->f_mode & FMODE_READ)
+ val = s->prop_adc.fmt_original;
+ }
+ CS_DBGOUT(CS_IOCTL | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_SETFMT return val=0x%.8x\n",
+ val));
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_POST:
+ CS_DBGOUT(CS_IOCTL, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): DSP_POST\n"));
+ return 0;
+
+ case SNDCTL_DSP_GETTRIGGER:
+ val = 0;
+ if (file->f_mode & s->ena & FMODE_READ)
+ val |= PCM_ENABLE_INPUT;
+ if (file->f_mode & s->ena & FMODE_WRITE)
+ val |= PCM_ENABLE_OUTPUT;
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_SETTRIGGER:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (file->f_mode & FMODE_READ) {
+ if (val & PCM_ENABLE_INPUT) {
+ if (!s->dma_adc.ready
+ && (ret = prog_dmabuf_adc(s)))
+ return ret;
+ start_adc(s);
+ } else
+ stop_adc(s);
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ if (val & PCM_ENABLE_OUTPUT) {
+ if (!s->dma_dac.ready
+ && (ret = prog_dmabuf_dac(s)))
+ return ret;
+ start_dac(s);
+ } else
+ stop_dac(s);
+ }
+ return 0;
+
+ case SNDCTL_DSP_GETOSPACE:
+ if (!(file->f_mode & FMODE_WRITE))
+ return -EINVAL;
+ if (!s->dma_dac.ready && (val = prog_dmabuf_dac(s)))
+ return val;
+ spin_lock_irqsave(&s->lock, flags);
+ cs4297a_update_ptr(s,CS_FALSE);
+ abinfo.fragsize = s->dma_dac.fragsize;
+ if (s->dma_dac.mapped)
+ abinfo.bytes = s->dma_dac.sbufsz;
+ else
+ abinfo.bytes =
+ s->dma_dac.sbufsz - s->dma_dac.count;
+ abinfo.fragstotal = s->dma_dac.numfrag;
+ abinfo.fragments = abinfo.bytes >> s->dma_dac.fragshift;
+ CS_DBGOUT(CS_FUNCTION | CS_PARMS, 4, printk(KERN_INFO
+ "cs4297a: cs4297a_ioctl(): GETOSPACE .fragsize=%d .bytes=%d .fragstotal=%d .fragments=%d\n",
+ abinfo.fragsize,abinfo.bytes,abinfo.fragstotal,
+ abinfo.fragments));
+ spin_unlock_irqrestore(&s->lock, flags);
+ return copy_to_user((void *) arg, &abinfo,
+ sizeof(abinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_GETISPACE:
+ if (!(file->f_mode & FMODE_READ))
+ return -EINVAL;
+ if (!s->dma_adc.ready && (val = prog_dmabuf_adc(s)))
+ return val;
+ spin_lock_irqsave(&s->lock, flags);
+ cs4297a_update_ptr(s,CS_FALSE);
+ if (s->conversion) {
+ abinfo.fragsize = s->dma_adc.fragsize / 2;
+ abinfo.bytes = s->dma_adc.count / 2;
+ abinfo.fragstotal = s->dma_adc.numfrag;
+ abinfo.fragments =
+ abinfo.bytes >> (s->dma_adc.fragshift - 1);
+ } else {
+ abinfo.fragsize = s->dma_adc.fragsize;
+ abinfo.bytes = s->dma_adc.count;
+ abinfo.fragstotal = s->dma_adc.numfrag;
+ abinfo.fragments =
+ abinfo.bytes >> s->dma_adc.fragshift;
+ }
+ spin_unlock_irqrestore(&s->lock, flags);
+ return copy_to_user((void *) arg, &abinfo,
+ sizeof(abinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_NONBLOCK:
+ file->f_flags |= O_NONBLOCK;
+ return 0;
+
+ case SNDCTL_DSP_GETODELAY:
+ if (!(file->f_mode & FMODE_WRITE))
+ return -EINVAL;
+ if(!s->dma_dac.ready && prog_dmabuf_dac(s))
+ return 0;
+ spin_lock_irqsave(&s->lock, flags);
+ cs4297a_update_ptr(s,CS_FALSE);
+ val = s->dma_dac.count;
+ spin_unlock_irqrestore(&s->lock, flags);
+ return put_user(val, (int *) arg);
+
+ case SNDCTL_DSP_GETIPTR:
+ if (!(file->f_mode & FMODE_READ))
+ return -EINVAL;
+ if(!s->dma_adc.ready && prog_dmabuf_adc(s))
+ return 0;
+ spin_lock_irqsave(&s->lock, flags);
+ cs4297a_update_ptr(s,CS_FALSE);
+ cinfo.bytes = s->dma_adc.total_bytes;
+ if (s->dma_adc.mapped) {
+ cinfo.blocks =
+ (cinfo.bytes >> s->dma_adc.fragshift) -
+ s->dma_adc.blocks;
+ s->dma_adc.blocks =
+ cinfo.bytes >> s->dma_adc.fragshift;
+ } else {
+ if (s->conversion) {
+ cinfo.blocks =
+ s->dma_adc.count /
+ 2 >> (s->dma_adc.fragshift - 1);
+ } else
+ cinfo.blocks =
+ s->dma_adc.count >> s->dma_adc.
+ fragshift;
+ }
+ if (s->conversion)
+ cinfo.ptr = s->dma_adc.hwptr / 2;
+ else
+ cinfo.ptr = s->dma_adc.hwptr;
+ if (s->dma_adc.mapped)
+ s->dma_adc.count &= s->dma_adc.fragsize - 1;
+ spin_unlock_irqrestore(&s->lock, flags);
+ return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_GETOPTR:
+ if (!(file->f_mode & FMODE_WRITE))
+ return -EINVAL;
+ if(!s->dma_dac.ready && prog_dmabuf_dac(s))
+ return 0;
+ spin_lock_irqsave(&s->lock, flags);
+ cs4297a_update_ptr(s,CS_FALSE);
+ cinfo.bytes = s->dma_dac.total_bytes;
+ if (s->dma_dac.mapped) {
+ cinfo.blocks =
+ (cinfo.bytes >> s->dma_dac.fragshift) -
+ s->dma_dac.blocks;
+ s->dma_dac.blocks =
+ cinfo.bytes >> s->dma_dac.fragshift;
+ } else {
+ cinfo.blocks =
+ s->dma_dac.count >> s->dma_dac.fragshift;
+ }
+ cinfo.ptr = s->dma_dac.hwptr;
+ if (s->dma_dac.mapped)
+ s->dma_dac.count &= s->dma_dac.fragsize - 1;
+ spin_unlock_irqrestore(&s->lock, flags);
+ return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)) ? -EFAULT : 0;
+
+ case SNDCTL_DSP_GETBLKSIZE:
+ if (file->f_mode & FMODE_WRITE) {
+ if ((val = prog_dmabuf_dac(s)))
+ return val;
+ return put_user(s->dma_dac.fragsize, (int *) arg);
+ }
+ if ((val = prog_dmabuf_adc(s)))
+ return val;
+ if (s->conversion)
+ return put_user(s->dma_adc.fragsize / 2,
+ (int *) arg);
+ else
+ return put_user(s->dma_adc.fragsize, (int *) arg);
+
+ case SNDCTL_DSP_SETFRAGMENT:
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ return 0; // Say OK, but do nothing.
+
+ case SNDCTL_DSP_SUBDIVIDE:
+ if ((file->f_mode & FMODE_READ && s->dma_adc.subdivision)
+ || (file->f_mode & FMODE_WRITE
+ && s->dma_dac.subdivision)) return -EINVAL;
+ if (get_user(val, (int *) arg))
+ return -EFAULT;
+ if (val != 1 && val != 2 && val != 4)
+ return -EINVAL;
+ if (file->f_mode & FMODE_READ)
+ s->dma_adc.subdivision = val;
+ else if (file->f_mode & FMODE_WRITE)
+ s->dma_dac.subdivision = val;
+ return 0;
+
+ case SOUND_PCM_READ_RATE:
+ if (file->f_mode & FMODE_READ)
+ return put_user(s->prop_adc.rate, (int *) arg);
+ else if (file->f_mode & FMODE_WRITE)
+ return put_user(s->prop_dac.rate, (int *) arg);
+
+ case SOUND_PCM_READ_CHANNELS:
+ if (file->f_mode & FMODE_READ)
+ return put_user(s->prop_adc.channels, (int *) arg);
+ else if (file->f_mode & FMODE_WRITE)
+ return put_user(s->prop_dac.channels, (int *) arg);
+
+ case SOUND_PCM_READ_BITS:
+ if (file->f_mode & FMODE_READ)
+ return
+ put_user(
+ (s->prop_adc.
+ fmt & (AFMT_S8 | AFMT_U8)) ? 8 : 16,
+ (int *) arg);
+ else if (file->f_mode & FMODE_WRITE)
+ return
+ put_user(
+ (s->prop_dac.
+ fmt & (AFMT_S8 | AFMT_U8)) ? 8 : 16,
+ (int *) arg);
+
+ case SOUND_PCM_WRITE_FILTER:
+ case SNDCTL_DSP_SETSYNCRO:
+ case SOUND_PCM_READ_FILTER:
+ return -EINVAL;
+ }
+ return mixer_ioctl(s, cmd, arg);
+}
+
+
+static int cs4297a_release(struct inode *inode, struct file *file)
+{
+ struct cs4297a_state *s =
+ (struct cs4297a_state *) file->private_data;
+
+ CS_DBGOUT(CS_FUNCTION | CS_RELEASE, 2, printk(KERN_INFO
+ "cs4297a: cs4297a_release(): inode=0x%.8x file=0x%.8x f_mode=0x%x\n",
+ (unsigned) inode, (unsigned) file, file->f_mode));
+ VALIDATE_STATE(s);
+
+ if (file->f_mode & FMODE_WRITE) {
+ drain_dac(s, file->f_flags & O_NONBLOCK);
+ mutex_lock(&s->open_sem_dac);
+ stop_dac(s);
+ dealloc_dmabuf(s, &s->dma_dac);
+ s->open_mode &= ~FMODE_WRITE;
+ mutex_unlock(&s->open_sem_dac);
+ wake_up(&s->open_wait_dac);
+ }
+ if (file->f_mode & FMODE_READ) {
+ drain_adc(s, file->f_flags & O_NONBLOCK);
+ mutex_lock(&s->open_sem_adc);
+ stop_adc(s);
+ dealloc_dmabuf(s, &s->dma_adc);
+ s->open_mode &= ~FMODE_READ;
+ mutex_unlock(&s->open_sem_adc);
+ wake_up(&s->open_wait_adc);
+ }
+ return 0;
+}
+
+static int cs4297a_open(struct inode *inode, struct file *file)
+{
+ int minor = iminor(inode);
+ struct cs4297a_state *s=NULL;
+ struct list_head *entry;
+
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, printk(KERN_INFO
+ "cs4297a: cs4297a_open(): inode=0x%.8x file=0x%.8x f_mode=0x%x\n",
+ (unsigned) inode, (unsigned) file, file->f_mode));
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, printk(KERN_INFO
+ "cs4297a: status = %08x\n", (int)__raw_readq(SS_CSR(R_SER_STATUS_DEBUG))));
+
+ list_for_each(entry, &cs4297a_devs)
+ {
+ s = list_entry(entry, struct cs4297a_state, list);
+
+ if (!((s->dev_audio ^ minor) & ~0xf))
+ break;
+ }
+ if (entry == &cs4297a_devs)
+ return -ENODEV;
+ if (!s) {
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2, printk(KERN_INFO
+ "cs4297a: cs4297a_open(): Error - unable to find audio state struct\n"));
+ return -ENODEV;
+ }
+ VALIDATE_STATE(s);
+ file->private_data = s;
+
+ // wait for device to become free
+ if (!(file->f_mode & (FMODE_WRITE | FMODE_READ))) {
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2, printk(KERN_INFO
+ "cs4297a: cs4297a_open(): Error - must open READ and/or WRITE\n"));
+ return -ENODEV;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ if (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX)) != 0) {
+ printk(KERN_ERR "cs4297a: TX pipe needs to drain\n");
+ while (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_TX)))
+ ;
+ }
+
+ mutex_lock(&s->open_sem_dac);
+ while (s->open_mode & FMODE_WRITE) {
+ if (file->f_flags & O_NONBLOCK) {
+ mutex_unlock(&s->open_sem_dac);
+ return -EBUSY;
+ }
+ mutex_unlock(&s->open_sem_dac);
+ interruptible_sleep_on(&s->open_wait_dac);
+
+ if (signal_pending(current)) {
+ printk("open - sig pending\n");
+ return -ERESTARTSYS;
+ }
+ mutex_lock(&s->open_sem_dac);
+ }
+ }
+ if (file->f_mode & FMODE_READ) {
+ mutex_lock(&s->open_sem_adc);
+ while (s->open_mode & FMODE_READ) {
+ if (file->f_flags & O_NONBLOCK) {
+ mutex_unlock(&s->open_sem_adc);
+ return -EBUSY;
+ }
+ mutex_unlock(&s->open_sem_adc);
+ interruptible_sleep_on(&s->open_wait_adc);
+
+ if (signal_pending(current)) {
+ printk("open - sig pending\n");
+ return -ERESTARTSYS;
+ }
+ mutex_lock(&s->open_sem_adc);
+ }
+ }
+ s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE);
+ if (file->f_mode & FMODE_READ) {
+ s->prop_adc.fmt = AFMT_S16_BE;
+ s->prop_adc.fmt_original = s->prop_adc.fmt;
+ s->prop_adc.channels = 2;
+ s->prop_adc.rate = 48000;
+ s->conversion = 0;
+ s->ena &= ~FMODE_READ;
+ s->dma_adc.ossfragshift = s->dma_adc.ossmaxfrags =
+ s->dma_adc.subdivision = 0;
+ mutex_unlock(&s->open_sem_adc);
+
+ if (prog_dmabuf_adc(s)) {
+ CS_DBGOUT(CS_OPEN | CS_ERROR, 2, printk(KERN_ERR
+ "cs4297a: adc Program dmabufs failed.\n"));
+ cs4297a_release(inode, file);
+ return -ENOMEM;
+ }
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ s->prop_dac.fmt = AFMT_S16_BE;
+ s->prop_dac.fmt_original = s->prop_dac.fmt;
+ s->prop_dac.channels = 2;
+ s->prop_dac.rate = 48000;
+ s->conversion = 0;
+ s->ena &= ~FMODE_WRITE;
+ s->dma_dac.ossfragshift = s->dma_dac.ossmaxfrags =
+ s->dma_dac.subdivision = 0;
+ mutex_unlock(&s->open_sem_dac);
+
+ if (prog_dmabuf_dac(s)) {
+ CS_DBGOUT(CS_OPEN | CS_ERROR, 2, printk(KERN_ERR
+ "cs4297a: dac Program dmabufs failed.\n"));
+ cs4297a_release(inode, file);
+ return -ENOMEM;
+ }
+ }
+ CS_DBGOUT(CS_FUNCTION | CS_OPEN, 2,
+ printk(KERN_INFO "cs4297a: cs4297a_open()- 0\n"));
+ return nonseekable_open(inode, file);
+}
+
+
+// ******************************************************************************************
+// Wave (audio) file operations struct.
+// ******************************************************************************************
+static const struct file_operations cs4297a_audio_fops = {
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .read = cs4297a_read,
+ .write = cs4297a_write,
+ .poll = cs4297a_poll,
+ .ioctl = cs4297a_ioctl,
+ .mmap = cs4297a_mmap,
+ .open = cs4297a_open,
+ .release = cs4297a_release,
+};
+
+static void cs4297a_interrupt(int irq, void *dev_id)
+{
+ struct cs4297a_state *s = (struct cs4297a_state *) dev_id;
+ u32 status;
+
+ status = __raw_readq(SS_CSR(R_SER_STATUS_DEBUG));
+
+ CS_DBGOUT(CS_INTERRUPT, 6, printk(KERN_INFO
+ "cs4297a: cs4297a_interrupt() HISR=0x%.8x\n", status));
+
+#if 0
+ /* XXXKW what check *should* be done here? */
+ if (!(status & (M_SYNCSER_RX_EOP_COUNT | M_SYNCSER_RX_OVERRUN | M_SYNCSER_RX_SYNC_ERR))) {
+ status = __raw_readq(SS_CSR(R_SER_STATUS));
+ printk(KERN_ERR "cs4297a: unexpected interrupt (status %08x)\n", status);
+ return;
+ }
+#endif
+
+ if (status & M_SYNCSER_RX_SYNC_ERR) {
+ status = __raw_readq(SS_CSR(R_SER_STATUS));
+ printk(KERN_ERR "cs4297a: rx sync error (status %08x)\n", status);
+ return;
+ }
+
+ if (status & M_SYNCSER_RX_OVERRUN) {
+ int newptr, i;
+ s->stats.rx_ovrrn++;
+ printk(KERN_ERR "cs4297a: receive FIFO overrun\n");
+
+ /* Fix things up: get the receive descriptor pool
+ clean and give them back to the hardware */
+ while (__raw_readq(SS_CSR(R_SER_DMA_DSCR_COUNT_RX)))
+ ;
+ newptr = (unsigned) (((__raw_readq(SS_CSR(R_SER_DMA_CUR_DSCR_ADDR_RX)) & M_DMA_CURDSCR_ADDR) -
+ s->dma_adc.descrtab_phys) / sizeof(serdma_descr_t));
+ for (i=0; i<DMA_DESCR; i++) {
+ s->dma_adc.descrtab[i].descr_a &= ~M_DMA_SERRX_SOP;
+ }
+ s->dma_adc.swptr = s->dma_adc.hwptr = newptr;
+ s->dma_adc.count = 0;
+ s->dma_adc.sb_swptr = s->dma_adc.sb_hwptr = s->dma_adc.sample_buf;
+ __raw_writeq(DMA_DESCR, SS_CSR(R_SER_DMA_DSCR_COUNT_RX));
+ }
+
+ spin_lock(&s->lock);
+ cs4297a_update_ptr(s,CS_TRUE);
+ spin_unlock(&s->lock);
+
+ CS_DBGOUT(CS_INTERRUPT, 6, printk(KERN_INFO
+ "cs4297a: cs4297a_interrupt()-\n"));
+}
+
+#if 0
+static struct initvol {
+ int mixch;
+ int vol;
+} initvol[] __initdata = {
+
+ {SOUND_MIXER_WRITE_VOLUME, 0x4040},
+ {SOUND_MIXER_WRITE_PCM, 0x4040},
+ {SOUND_MIXER_WRITE_SYNTH, 0x4040},
+ {SOUND_MIXER_WRITE_CD, 0x4040},
+ {SOUND_MIXER_WRITE_LINE, 0x4040},
+ {SOUND_MIXER_WRITE_LINE1, 0x4040},
+ {SOUND_MIXER_WRITE_RECLEV, 0x0000},
+ {SOUND_MIXER_WRITE_SPEAKER, 0x4040},
+ {SOUND_MIXER_WRITE_MIC, 0x0000}
+};
+#endif
+
+static int __init cs4297a_init(void)
+{
+ struct cs4297a_state *s;
+ u32 pwr, id;
+ mm_segment_t fs;
+ int rval;
+#ifndef CONFIG_BCM_CS4297A_CSWARM
+ u64 cfg;
+ int mdio_val;
+#endif
+
+ CS_DBGOUT(CS_INIT | CS_FUNCTION, 2, printk(KERN_INFO
+ "cs4297a: cs4297a_init_module()+ \n"));
+
+#ifndef CONFIG_BCM_CS4297A_CSWARM
+ mdio_val = __raw_readq(KSEG1 + A_MAC_REGISTER(2, R_MAC_MDIO)) &
+ (M_MAC_MDIO_DIR|M_MAC_MDIO_OUT);
+
+ /* Check syscfg for synchronous serial on port 1 */
+ cfg = __raw_readq(KSEG1 + A_SCD_SYSTEM_CFG);
+ if (!(cfg & M_SYS_SER1_ENABLE)) {
+ __raw_writeq(cfg | M_SYS_SER1_ENABLE, KSEG1+A_SCD_SYSTEM_CFG);
+ cfg = __raw_readq(KSEG1 + A_SCD_SYSTEM_CFG);
+ if (!(cfg & M_SYS_SER1_ENABLE)) {
+ printk(KERN_INFO "cs4297a: serial port 1 not configured for synchronous operation\n");
+ return -1;
+ }
+
+ printk(KERN_INFO "cs4297a: serial port 1 switching to synchronous operation\n");
+
+ /* Force the codec (on SWARM) to reset by clearing
+ GENO, preserving MDIO (no effect on CSWARM) */
+ __raw_writeq(mdio_val, KSEG1+A_MAC_REGISTER(2, R_MAC_MDIO));
+ udelay(10);
+ }
+
+ /* Now set GENO */
+ __raw_writeq(mdio_val | M_MAC_GENC, KSEG1+A_MAC_REGISTER(2, R_MAC_MDIO));
+ /* Give the codec some time to finish resetting (start the bit clock) */
+ udelay(100);
+#endif
+
+ if (!(s = kzalloc(sizeof(struct cs4297a_state), GFP_KERNEL))) {
+ CS_DBGOUT(CS_ERROR, 1, printk(KERN_ERR
+ "cs4297a: probe() no memory for state struct.\n"));
+ return -1;
+ }
+ s->magic = CS4297a_MAGIC;
+ init_waitqueue_head(&s->dma_adc.wait);
+ init_waitqueue_head(&s->dma_dac.wait);
+ init_waitqueue_head(&s->dma_adc.reg_wait);
+ init_waitqueue_head(&s->dma_dac.reg_wait);
+ init_waitqueue_head(&s->open_wait);
+ init_waitqueue_head(&s->open_wait_adc);
+ init_waitqueue_head(&s->open_wait_dac);
+ mutex_init(&s->open_sem_adc);
+ mutex_init(&s->open_sem_dac);
+ spin_lock_init(&s->lock);
+
+ s->irq = K_INT_SER_1;
+
+ if (request_irq
+ (s->irq, cs4297a_interrupt, 0, "Crystal CS4297a", s)) {
+ CS_DBGOUT(CS_INIT | CS_ERROR, 1,
+ printk(KERN_ERR "cs4297a: irq %u in use\n", s->irq));
+ goto err_irq;
+ }
+ if ((s->dev_audio = register_sound_dsp(&cs4297a_audio_fops, -1)) <
+ 0) {
+ CS_DBGOUT(CS_INIT | CS_ERROR, 1, printk(KERN_ERR
+ "cs4297a: probe() register_sound_dsp() failed.\n"));
+ goto err_dev1;
+ }
+ if ((s->dev_mixer = register_sound_mixer(&cs4297a_mixer_fops, -1)) <
+ 0) {
+ CS_DBGOUT(CS_INIT | CS_ERROR, 1, printk(KERN_ERR
+ "cs4297a: probe() register_sound_mixer() failed.\n"));
+ goto err_dev2;
+ }
+
+ if (ser_init(s) || dma_init(s)) {
+ CS_DBGOUT(CS_INIT | CS_ERROR, 1, printk(KERN_ERR
+ "cs4297a: ser_init failed.\n"));
+ goto err_dev3;
+ }
+
+ do {
+ udelay(4000);
+ rval = cs4297a_read_ac97(s, AC97_POWER_CONTROL, &pwr);
+ } while (!rval && (pwr != 0xf));
+
+ if (!rval) {
+ char *sb1250_duart_present;
+
+ fs = get_fs();
+ set_fs(KERNEL_DS);
+#if 0
+ val = SOUND_MASK_LINE;
+ mixer_ioctl(s, SOUND_MIXER_WRITE_RECSRC, (unsigned long) &val);
+ for (i = 0; i < ARRAY_SIZE(initvol); i++) {
+ val = initvol[i].vol;
+ mixer_ioctl(s, initvol[i].mixch, (unsigned long) &val);
+ }
+// cs4297a_write_ac97(s, 0x18, 0x0808);
+#else
+ // cs4297a_write_ac97(s, 0x5e, 0x180);
+ cs4297a_write_ac97(s, 0x02, 0x0808);
+ cs4297a_write_ac97(s, 0x18, 0x0808);
+#endif
+ set_fs(fs);
+
+ list_add(&s->list, &cs4297a_devs);
+
+ cs4297a_read_ac97(s, AC97_VENDOR_ID1, &id);
+
+ sb1250_duart_present = symbol_get(sb1250_duart_present);
+ if (sb1250_duart_present)
+ sb1250_duart_present[1] = 0;
+
+ printk(KERN_INFO "cs4297a: initialized (vendor id = %x)\n", id);
+
+ CS_DBGOUT(CS_INIT | CS_FUNCTION, 2,
+ printk(KERN_INFO "cs4297a: cs4297a_init_module()-\n"));
+
+ return 0;
+ }
+
+ err_dev3:
+ unregister_sound_mixer(s->dev_mixer);
+ err_dev2:
+ unregister_sound_dsp(s->dev_audio);
+ err_dev1:
+ free_irq(s->irq, s);
+ err_irq:
+ kfree(s);
+
+ printk(KERN_INFO "cs4297a: initialization failed\n");
+
+ return -1;
+}
+
+static void __exit cs4297a_cleanup(void)
+{
+ /*
+ XXXKW
+ disable_irq, free_irq
+ drain DMA queue
+ disable DMA
+ disable TX/RX
+ free memory
+ */
+ CS_DBGOUT(CS_INIT | CS_FUNCTION, 2,
+ printk(KERN_INFO "cs4297a: cleanup_cs4297a() finished\n"));
+}
+
+// ---------------------------------------------------------------------
+
+MODULE_AUTHOR("Kip Walker, Broadcom Corp.");
+MODULE_DESCRIPTION("Cirrus Logic CS4297a Driver for Broadcom SWARM board");
+
+// ---------------------------------------------------------------------
+
+module_init(cs4297a_init);
+module_exit(cs4297a_cleanup);
diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c
new file mode 100644
index 0000000..1075344
--- /dev/null
+++ b/sound/oss/sys_timer.c
@@ -0,0 +1,288 @@
+/*
+ * sound/oss/sys_timer.c
+ *
+ * The default timer for the Level 2 sequencer interface
+ * Uses the (1/HZ sec) timer of kernel.
+ */
+/*
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ */
+/*
+ * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
+ * Andrew Veliath : adapted tmr2ticks from level 1 sequencer (avoid overflow)
+ */
+#include <linux/spinlock.h>
+#include "sound_config.h"
+
+static volatile int opened, tmr_running;
+static volatile time_t tmr_offs, tmr_ctr;
+static volatile unsigned long ticks_offs;
+static volatile int curr_tempo, curr_timebase;
+static volatile unsigned long curr_ticks;
+static volatile unsigned long next_event_time;
+static unsigned long prev_event_time;
+
+static void poll_def_tmr(unsigned long dummy);
+static DEFINE_SPINLOCK(lock);
+static DEFINE_TIMER(def_tmr, poll_def_tmr, 0, 0);
+
+static unsigned long
+tmr2ticks(int tmr_value)
+{
+ /*
+ * Convert timer ticks to MIDI ticks
+ */
+
+ unsigned long tmp;
+ unsigned long scale;
+
+ /* tmr_value (ticks per sec) *
+ 1000000 (usecs per sec) / HZ (ticks per sec) -=> usecs */
+ tmp = tmr_value * (1000000 / HZ);
+ scale = (60 * 1000000) / (curr_tempo * curr_timebase); /* usecs per MIDI tick */
+ return (tmp + scale / 2) / scale;
+}
+
+static void
+poll_def_tmr(unsigned long dummy)
+{
+
+ if (opened)
+ {
+
+ {
+ def_tmr.expires = (1) + jiffies;
+ add_timer(&def_tmr);
+ };
+
+ if (tmr_running)
+ {
+ spin_lock(&lock);
+ tmr_ctr++;
+ curr_ticks = ticks_offs + tmr2ticks(tmr_ctr);
+
+ if (curr_ticks >= next_event_time)
+ {
+ next_event_time = (unsigned long) -1;
+ sequencer_timer(0);
+ }
+ spin_unlock(&lock);
+ }
+ }
+}
+
+static void
+tmr_reset(void)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&lock,flags);
+ tmr_offs = 0;
+ ticks_offs = 0;
+ tmr_ctr = 0;
+ next_event_time = (unsigned long) -1;
+ prev_event_time = 0;
+ curr_ticks = 0;
+ spin_unlock_irqrestore(&lock,flags);
+}
+
+static int
+def_tmr_open(int dev, int mode)
+{
+ if (opened)
+ return -EBUSY;
+
+ tmr_reset();
+ curr_tempo = 60;
+ curr_timebase = 100;
+ opened = 1;
+
+ ;
+
+ {
+ def_tmr.expires = (1) + jiffies;
+ add_timer(&def_tmr);
+ };
+
+ return 0;
+}
+
+static void
+def_tmr_close(int dev)
+{
+ opened = tmr_running = 0;
+ del_timer(&def_tmr);
+}
+
+static int
+def_tmr_event(int dev, unsigned char *event)
+{
+ unsigned char cmd = event[1];
+ unsigned long parm = *(int *) &event[4];
+
+ switch (cmd)
+ {
+ case TMR_WAIT_REL:
+ parm += prev_event_time;
+ case TMR_WAIT_ABS:
+ if (parm > 0)
+ {
+ long time;
+
+ if (parm <= curr_ticks) /* It's the time */
+ return TIMER_NOT_ARMED;
+
+ time = parm;
+ next_event_time = prev_event_time = time;
+
+ return TIMER_ARMED;
+ }
+ break;
+
+ case TMR_START:
+ tmr_reset();
+ tmr_running = 1;
+ break;
+
+ case TMR_STOP:
+ tmr_running = 0;
+ break;
+
+ case TMR_CONTINUE:
+ tmr_running = 1;
+ break;
+
+ case TMR_TEMPO:
+ if (parm)
+ {
+ if (parm < 8)
+ parm = 8;
+ if (parm > 360)
+ parm = 360;
+ tmr_offs = tmr_ctr;
+ ticks_offs += tmr2ticks(tmr_ctr);
+ tmr_ctr = 0;
+ curr_tempo = parm;
+ }
+ break;
+
+ case TMR_ECHO:
+ seq_copy_to_input(event, 8);
+ break;
+
+ default:;
+ }
+
+ return TIMER_NOT_ARMED;
+}
+
+static unsigned long
+def_tmr_get_time(int dev)
+{
+ if (!opened)
+ return 0;
+
+ return curr_ticks;
+}
+
+/* same as sound_timer.c:timer_ioctl!? */
+static int def_tmr_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ int __user *p = arg;
+ int val;
+
+ switch (cmd) {
+ case SNDCTL_TMR_SOURCE:
+ return __put_user(TMR_INTERNAL, p);
+
+ case SNDCTL_TMR_START:
+ tmr_reset();
+ tmr_running = 1;
+ return 0;
+
+ case SNDCTL_TMR_STOP:
+ tmr_running = 0;
+ return 0;
+
+ case SNDCTL_TMR_CONTINUE:
+ tmr_running = 1;
+ return 0;
+
+ case SNDCTL_TMR_TIMEBASE:
+ if (__get_user(val, p))
+ return -EFAULT;
+ if (val) {
+ if (val < 1)
+ val = 1;
+ if (val > 1000)
+ val = 1000;
+ curr_timebase = val;
+ }
+ return __put_user(curr_timebase, p);
+
+ case SNDCTL_TMR_TEMPO:
+ if (__get_user(val, p))
+ return -EFAULT;
+ if (val) {
+ if (val < 8)
+ val = 8;
+ if (val > 250)
+ val = 250;
+ tmr_offs = tmr_ctr;
+ ticks_offs += tmr2ticks(tmr_ctr);
+ tmr_ctr = 0;
+ curr_tempo = val;
+ reprogram_timer();
+ }
+ return __put_user(curr_tempo, p);
+
+ case SNDCTL_SEQ_CTRLRATE:
+ if (__get_user(val, p))
+ return -EFAULT;
+ if (val != 0) /* Can't change */
+ return -EINVAL;
+ val = ((curr_tempo * curr_timebase) + 30) / 60;
+ return __put_user(val, p);
+
+ case SNDCTL_SEQ_GETTIME:
+ return __put_user(curr_ticks, p);
+
+ case SNDCTL_TMR_METRONOME:
+ /* NOP */
+ break;
+
+ default:;
+ }
+ return -EINVAL;
+}
+
+static void
+def_tmr_arm(int dev, long time)
+{
+ if (time < 0)
+ time = curr_ticks + 1;
+ else if (time <= curr_ticks) /* It's the time */
+ return;
+
+ next_event_time = prev_event_time = time;
+
+ return;
+}
+
+struct sound_timer_operations default_sound_timer =
+{
+ .owner = THIS_MODULE,
+ .info = {"System clock", 0},
+ .priority = 0, /* Priority */
+ .devlink = 0, /* Local device link */
+ .open = def_tmr_open,
+ .close = def_tmr_close,
+ .event = def_tmr_event,
+ .get_time = def_tmr_get_time,
+ .ioctl = def_tmr_ioctl,
+ .arm_timer = def_tmr_arm
+};
diff --git a/sound/oss/trix.c b/sound/oss/trix.c
new file mode 100644
index 0000000..e04169e
--- /dev/null
+++ b/sound/oss/trix.c
@@ -0,0 +1,525 @@
+/*
+ * sound/oss/trix.c
+ *
+ * Low level driver for the MediaTrix AudioTrix Pro
+ * (MT-0002-PC Control Chip)
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ * Changes
+ * Alan Cox Modularisation, cleanup.
+ * Christoph Hellwig Adapted to module_init/module_exit
+ * Arnaldo C. de Melo Got rid of attach_uart401
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+
+#include "sound_config.h"
+#include "sb.h"
+#include "sound_firmware.h"
+
+#include "ad1848.h"
+#include "mpu401.h"
+
+#include "trix_boot.h"
+
+static int mpu;
+
+static int joystick;
+
+static unsigned char trix_read(int addr)
+{
+ outb(((unsigned char) addr), 0x390); /* MT-0002-PC ASIC address */
+ return inb(0x391); /* MT-0002-PC ASIC data */
+}
+
+static void trix_write(int addr, int data)
+{
+ outb(((unsigned char) addr), 0x390); /* MT-0002-PC ASIC address */
+ outb(((unsigned char) data), 0x391); /* MT-0002-PC ASIC data */
+}
+
+static void download_boot(int base)
+{
+ int i = 0, n = trix_boot_len;
+
+ if (trix_boot_len == 0)
+ return;
+
+ trix_write(0xf8, 0x00); /* ??????? */
+ outb((0x01), base + 6); /* Clear the internal data pointer */
+ outb((0x00), base + 6); /* Restart */
+
+ /*
+ * Write the boot code to the RAM upload/download register.
+ * Each write increments the internal data pointer.
+ */
+ outb((0x01), base + 6); /* Clear the internal data pointer */
+ outb((0x1A), 0x390); /* Select RAM download/upload port */
+
+ for (i = 0; i < n; i++)
+ outb((trix_boot[i]), 0x391);
+ for (i = n; i < 10016; i++) /* Clear up to first 16 bytes of data RAM */
+ outb((0x00), 0x391);
+ outb((0x00), base + 6); /* Reset */
+ outb((0x50), 0x390); /* ?????? */
+
+}
+
+static int trix_set_wss_port(struct address_info *hw_config)
+{
+ unsigned char addr_bits;
+
+ if (trix_read(0x15) != 0x71) /* No ASIC signature */
+ {
+ MDB(printk(KERN_ERR "No AudioTrix ASIC signature found\n"));
+ return 0;
+ }
+
+ /*
+ * Reset some registers.
+ */
+
+ trix_write(0x13, 0);
+ trix_write(0x14, 0);
+
+ /*
+ * Configure the ASIC to place the codec to the proper I/O location
+ */
+
+ switch (hw_config->io_base)
+ {
+ case 0x530:
+ addr_bits = 0;
+ break;
+ case 0x604:
+ addr_bits = 1;
+ break;
+ case 0xE80:
+ addr_bits = 2;
+ break;
+ case 0xF40:
+ addr_bits = 3;
+ break;
+ default:
+ return 0;
+ }
+
+ trix_write(0x19, (trix_read(0x19) & 0x03) | addr_bits);
+ return 1;
+}
+
+/*
+ * Probe and attach routines for the Windows Sound System mode of
+ * AudioTrix Pro
+ */
+
+static int __init init_trix_wss(struct address_info *hw_config)
+{
+ static unsigned char dma_bits[4] = {
+ 1, 2, 0, 3
+ };
+ struct resource *ports;
+ int config_port = hw_config->io_base + 0;
+ int dma1 = hw_config->dma, dma2 = hw_config->dma2;
+ int old_num_mixers = num_mixers;
+ u8 config, bits;
+ int ret;
+
+ switch(hw_config->irq) {
+ case 7:
+ bits = 8;
+ break;
+ case 9:
+ bits = 0x10;
+ break;
+ case 10:
+ bits = 0x18;
+ break;
+ case 11:
+ bits = 0x20;
+ break;
+ default:
+ printk(KERN_ERR "AudioTrix: Bad WSS IRQ %d\n", hw_config->irq);
+ return 0;
+ }
+
+ switch (dma1) {
+ case 0:
+ case 1:
+ case 3:
+ break;
+ default:
+ printk(KERN_ERR "AudioTrix: Bad WSS DMA %d\n", dma1);
+ return 0;
+ }
+
+ switch (dma2) {
+ case -1:
+ case 0:
+ case 1:
+ case 3:
+ break;
+ default:
+ printk(KERN_ERR "AudioTrix: Bad capture DMA %d\n", dma2);
+ return 0;
+ }
+
+ /*
+ * Check if the IO port returns valid signature. The original MS Sound
+ * system returns 0x04 while some cards (AudioTrix Pro for example)
+ * return 0x00.
+ */
+ ports = request_region(hw_config->io_base + 4, 4, "ad1848");
+ if (!ports) {
+ printk(KERN_ERR "AudioTrix: MSS I/O port conflict (%x)\n", hw_config->io_base);
+ return 0;
+ }
+
+ if (!request_region(hw_config->io_base, 4, "MSS config")) {
+ printk(KERN_ERR "AudioTrix: MSS I/O port conflict (%x)\n", hw_config->io_base);
+ release_region(hw_config->io_base + 4, 4);
+ return 0;
+ }
+
+ if (!trix_set_wss_port(hw_config))
+ goto fail;
+
+ config = inb(hw_config->io_base + 3);
+
+ if ((config & 0x3f) != 0x00)
+ {
+ MDB(printk(KERN_ERR "No MSS signature detected on port 0x%x\n", hw_config->io_base));
+ goto fail;
+ }
+
+ /*
+ * Check that DMA0 is not in use with a 8 bit board.
+ */
+
+ if (dma1 == 0 && config & 0x80)
+ {
+ printk(KERN_ERR "AudioTrix: Can't use DMA0 with a 8 bit card slot\n");
+ goto fail;
+ }
+ if (hw_config->irq > 9 && config & 0x80)
+ {
+ printk(KERN_ERR "AudioTrix: Can't use IRQ%d with a 8 bit card slot\n", hw_config->irq);
+ goto fail;
+ }
+
+ ret = ad1848_detect(ports, NULL, hw_config->osp);
+ if (!ret)
+ goto fail;
+
+ if (joystick==1)
+ trix_write(0x15, 0x80);
+
+ /*
+ * Set the IRQ and DMA addresses.
+ */
+
+ outb((bits | 0x40), config_port);
+
+ if (dma2 == -1 || dma2 == dma1)
+ {
+ bits |= dma_bits[dma1];
+ dma2 = dma1;
+ }
+ else
+ {
+ unsigned char tmp;
+
+ tmp = trix_read(0x13) & ~30;
+ trix_write(0x13, tmp | 0x80 | (dma1 << 4));
+
+ tmp = trix_read(0x14) & ~30;
+ trix_write(0x14, tmp | 0x80 | (dma2 << 4));
+ }
+
+ outb((bits), config_port); /* Write IRQ+DMA setup */
+
+ hw_config->slots[0] = ad1848_init("AudioTrix Pro", ports,
+ hw_config->irq,
+ dma1,
+ dma2,
+ 0,
+ hw_config->osp,
+ THIS_MODULE);
+
+ if (num_mixers > old_num_mixers) /* Mixer got installed */
+ {
+ AD1848_REROUTE(SOUND_MIXER_LINE1, SOUND_MIXER_LINE); /* Line in */
+ AD1848_REROUTE(SOUND_MIXER_LINE2, SOUND_MIXER_CD);
+ AD1848_REROUTE(SOUND_MIXER_LINE3, SOUND_MIXER_SYNTH); /* OPL4 */
+ AD1848_REROUTE(SOUND_MIXER_SPEAKER, SOUND_MIXER_ALTPCM); /* SB */
+ }
+ return 1;
+
+fail:
+ release_region(hw_config->io_base, 4);
+ release_region(hw_config->io_base + 4, 4);
+ return 0;
+}
+
+static int __init probe_trix_sb(struct address_info *hw_config)
+{
+
+ int tmp;
+ unsigned char conf;
+ extern int sb_be_quiet;
+ int old_quiet;
+ static signed char irq_translate[] = {
+ -1, -1, -1, 0, 1, 2, -1, 3
+ };
+
+ if (trix_boot_len == 0)
+ return 0; /* No boot code -> no fun */
+
+ if ((hw_config->io_base & 0xffffff8f) != 0x200)
+ return 0;
+
+ tmp = hw_config->irq;
+ if (tmp > 7)
+ return 0;
+ if (irq_translate[tmp] == -1)
+ return 0;
+
+ tmp = hw_config->dma;
+ if (tmp != 1 && tmp != 3)
+ return 0;
+
+ if (!request_region(hw_config->io_base, 16, "soundblaster")) {
+ printk(KERN_ERR "AudioTrix: SB I/O port conflict (%x)\n", hw_config->io_base);
+ return 0;
+ }
+
+ conf = 0x84; /* DMA and IRQ enable */
+ conf |= hw_config->io_base & 0x70; /* I/O address bits */
+ conf |= irq_translate[hw_config->irq];
+ if (hw_config->dma == 3)
+ conf |= 0x08;
+ trix_write(0x1b, conf);
+
+ download_boot(hw_config->io_base);
+
+ hw_config->name = "AudioTrix SB";
+ if (!sb_dsp_detect(hw_config, 0, 0, NULL)) {
+ release_region(hw_config->io_base, 16);
+ return 0;
+ }
+
+ hw_config->driver_use_1 = SB_NO_MIDI | SB_NO_MIXER | SB_NO_RECORDING;
+
+ /* Prevent false alarms */
+ old_quiet = sb_be_quiet;
+ sb_be_quiet = 1;
+
+ sb_dsp_init(hw_config, THIS_MODULE);
+
+ sb_be_quiet = old_quiet;
+ return 1;
+}
+
+static int __init probe_trix_mpu(struct address_info *hw_config)
+{
+ unsigned char conf;
+ static int irq_bits[] = {
+ -1, -1, -1, 1, 2, 3, -1, 4, -1, 5
+ };
+
+ if (hw_config->irq > 9)
+ {
+ printk(KERN_ERR "AudioTrix: Bad MPU IRQ %d\n", hw_config->irq);
+ return 0;
+ }
+ if (irq_bits[hw_config->irq] == -1)
+ {
+ printk(KERN_ERR "AudioTrix: Bad MPU IRQ %d\n", hw_config->irq);
+ return 0;
+ }
+ switch (hw_config->io_base)
+ {
+ case 0x330:
+ conf = 0x00;
+ break;
+ case 0x370:
+ conf = 0x04;
+ break;
+ case 0x3b0:
+ conf = 0x08;
+ break;
+ case 0x3f0:
+ conf = 0x0c;
+ break;
+ default:
+ return 0; /* Invalid port */
+ }
+
+ conf |= irq_bits[hw_config->irq] << 4;
+ trix_write(0x19, (trix_read(0x19) & 0x83) | conf);
+ hw_config->name = "AudioTrix Pro";
+ return probe_uart401(hw_config, THIS_MODULE);
+}
+
+static void __exit unload_trix_wss(struct address_info *hw_config)
+{
+ int dma2 = hw_config->dma2;
+
+ if (dma2 == -1)
+ dma2 = hw_config->dma;
+
+ release_region(0x390, 2);
+ release_region(hw_config->io_base, 4);
+
+ ad1848_unload(hw_config->io_base + 4,
+ hw_config->irq,
+ hw_config->dma,
+ dma2,
+ 0);
+ sound_unload_audiodev(hw_config->slots[0]);
+}
+
+static inline void __exit unload_trix_mpu(struct address_info *hw_config)
+{
+ unload_uart401(hw_config);
+}
+
+static inline void __exit unload_trix_sb(struct address_info *hw_config)
+{
+ sb_dsp_unload(hw_config, mpu);
+}
+
+static struct address_info cfg;
+static struct address_info cfg2;
+static struct address_info cfg_mpu;
+
+static int sb;
+static int fw_load;
+
+static int __initdata io = -1;
+static int __initdata irq = -1;
+static int __initdata dma = -1;
+static int __initdata dma2 = -1; /* Set this for modules that need it */
+static int __initdata sb_io = -1;
+static int __initdata sb_dma = -1;
+static int __initdata sb_irq = -1;
+static int __initdata mpu_io = -1;
+static int __initdata mpu_irq = -1;
+
+module_param(io, int, 0);
+module_param(irq, int, 0);
+module_param(dma, int, 0);
+module_param(dma2, int, 0);
+module_param(sb_io, int, 0);
+module_param(sb_dma, int, 0);
+module_param(sb_irq, int, 0);
+module_param(mpu_io, int, 0);
+module_param(mpu_irq, int, 0);
+module_param(joystick, bool, 0);
+
+static int __init init_trix(void)
+{
+ printk(KERN_INFO "MediaTrix audio driver Copyright (C) by Hannu Savolainen 1993-1996\n");
+
+ cfg.io_base = io;
+ cfg.irq = irq;
+ cfg.dma = dma;
+ cfg.dma2 = dma2;
+
+ cfg2.io_base = sb_io;
+ cfg2.irq = sb_irq;
+ cfg2.dma = sb_dma;
+
+ cfg_mpu.io_base = mpu_io;
+ cfg_mpu.irq = mpu_irq;
+
+ if (cfg.io_base == -1 || cfg.dma == -1 || cfg.irq == -1) {
+ printk(KERN_INFO "I/O, IRQ, DMA and type are mandatory\n");
+ return -EINVAL;
+ }
+
+ if (cfg2.io_base != -1 && (cfg2.irq == -1 || cfg2.dma == -1)) {
+ printk(KERN_INFO "CONFIG_SB_IRQ and CONFIG_SB_DMA must be specified if SB_IO is set.\n");
+ return -EINVAL;
+ }
+ if (cfg_mpu.io_base != -1 && cfg_mpu.irq == -1) {
+ printk(KERN_INFO "CONFIG_MPU_IRQ must be specified if MPU_IO is set.\n");
+ return -EINVAL;
+ }
+ if (!trix_boot)
+ {
+ fw_load = 1;
+ trix_boot_len = mod_firmware_load("/etc/sound/trxpro.bin",
+ (char **) &trix_boot);
+ }
+
+ if (!request_region(0x390, 2, "AudioTrix")) {
+ printk(KERN_ERR "AudioTrix: Config port I/O conflict\n");
+ return -ENODEV;
+ }
+
+ if (!init_trix_wss(&cfg)) {
+ release_region(0x390, 2);
+ return -ENODEV;
+ }
+
+ /*
+ * We must attach in the right order to get the firmware
+ * loaded up in time.
+ */
+
+ if (cfg2.io_base != -1) {
+ sb = probe_trix_sb(&cfg2);
+ }
+
+ if (cfg_mpu.io_base != -1)
+ mpu = probe_trix_mpu(&cfg_mpu);
+
+ return 0;
+}
+
+static void __exit cleanup_trix(void)
+{
+ if (fw_load && trix_boot)
+ vfree(trix_boot);
+ if (sb)
+ unload_trix_sb(&cfg2);
+ if (mpu)
+ unload_trix_mpu(&cfg_mpu);
+ unload_trix_wss(&cfg);
+}
+
+module_init(init_trix);
+module_exit(cleanup_trix);
+
+#ifndef MODULE
+static int __init setup_trix (char *str)
+{
+ /* io, irq, dma, dma2, sb_io, sb_irq, sb_dma, mpu_io, mpu_irq */
+ int ints[9];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+ dma = ints[3];
+ dma2 = ints[4];
+ sb_io = ints[5];
+ sb_irq = ints[6];
+ sb_dma = ints[6];
+ mpu_io = ints[7];
+ mpu_irq = ints[8];
+
+ return 1;
+}
+
+__setup("trix=", setup_trix);
+#endif
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/tuning.h b/sound/oss/tuning.h
new file mode 100644
index 0000000..a73e3dd
--- /dev/null
+++ b/sound/oss/tuning.h
@@ -0,0 +1,23 @@
+static unsigned short semitone_tuning[24] =
+{
+/* 0 */ 10000, 10595, 11225, 11892, 12599, 13348, 14142, 14983,
+/* 8 */ 15874, 16818, 17818, 18877, 20000, 21189, 22449, 23784,
+/* 16 */ 25198, 26697, 28284, 29966, 31748, 33636, 35636, 37755
+};
+
+static unsigned short cent_tuning[100] =
+{
+/* 0 */ 10000, 10006, 10012, 10017, 10023, 10029, 10035, 10041,
+/* 8 */ 10046, 10052, 10058, 10064, 10070, 10075, 10081, 10087,
+/* 16 */ 10093, 10099, 10105, 10110, 10116, 10122, 10128, 10134,
+/* 24 */ 10140, 10145, 10151, 10157, 10163, 10169, 10175, 10181,
+/* 32 */ 10187, 10192, 10198, 10204, 10210, 10216, 10222, 10228,
+/* 40 */ 10234, 10240, 10246, 10251, 10257, 10263, 10269, 10275,
+/* 48 */ 10281, 10287, 10293, 10299, 10305, 10311, 10317, 10323,
+/* 56 */ 10329, 10335, 10341, 10347, 10353, 10359, 10365, 10371,
+/* 64 */ 10377, 10383, 10389, 10395, 10401, 10407, 10413, 10419,
+/* 72 */ 10425, 10431, 10437, 10443, 10449, 10455, 10461, 10467,
+/* 80 */ 10473, 10479, 10485, 10491, 10497, 10503, 10509, 10515,
+/* 88 */ 10521, 10528, 10534, 10540, 10546, 10552, 10558, 10564,
+/* 96 */ 10570, 10576, 10582, 10589
+};
diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c
new file mode 100644
index 0000000..a446b82
--- /dev/null
+++ b/sound/oss/uart401.c
@@ -0,0 +1,481 @@
+/*
+ * sound/oss/uart401.c
+ *
+ * MPU-401 UART driver (formerly uart401_midi.c)
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ * Changes:
+ * Alan Cox Reformatted, removed sound_mem usage, use normal Linux
+ * interrupt allocation. Protect against bogus unload
+ * Fixed to allow IRQ > 15
+ * Christoph Hellwig Adapted to module_init/module_exit
+ * Arnaldo C. de Melo got rid of check_region
+ *
+ * Status:
+ * Untested
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/spinlock.h>
+#include "sound_config.h"
+
+#include "mpu401.h"
+
+typedef struct uart401_devc
+{
+ int base;
+ int irq;
+ int *osp;
+ void (*midi_input_intr) (int dev, unsigned char data);
+ int opened, disabled;
+ volatile unsigned char input_byte;
+ int my_dev;
+ int share_irq;
+ spinlock_t lock;
+}
+uart401_devc;
+
+#define DATAPORT (devc->base)
+#define COMDPORT (devc->base+1)
+#define STATPORT (devc->base+1)
+
+static int uart401_status(uart401_devc * devc)
+{
+ return inb(STATPORT);
+}
+
+#define input_avail(devc) (!(uart401_status(devc)&INPUT_AVAIL))
+#define output_ready(devc) (!(uart401_status(devc)&OUTPUT_READY))
+
+static void uart401_cmd(uart401_devc * devc, unsigned char cmd)
+{
+ outb((cmd), COMDPORT);
+}
+
+static int uart401_read(uart401_devc * devc)
+{
+ return inb(DATAPORT);
+}
+
+static void uart401_write(uart401_devc * devc, unsigned char byte)
+{
+ outb((byte), DATAPORT);
+}
+
+#define OUTPUT_READY 0x40
+#define INPUT_AVAIL 0x80
+#define MPU_ACK 0xFE
+#define MPU_RESET 0xFF
+#define UART_MODE_ON 0x3F
+
+static int reset_uart401(uart401_devc * devc);
+static void enter_uart_mode(uart401_devc * devc);
+
+static void uart401_input_loop(uart401_devc * devc)
+{
+ int work_limit=30000;
+
+ while (input_avail(devc) && --work_limit)
+ {
+ unsigned char c = uart401_read(devc);
+
+ if (c == MPU_ACK)
+ devc->input_byte = c;
+ else if (devc->opened & OPEN_READ && devc->midi_input_intr)
+ devc->midi_input_intr(devc->my_dev, c);
+ }
+ if(work_limit==0)
+ printk(KERN_WARNING "Too much work in interrupt on uart401 (0x%X). UART jabbering ??\n", devc->base);
+}
+
+irqreturn_t uart401intr(int irq, void *dev_id)
+{
+ uart401_devc *devc = dev_id;
+
+ if (devc == NULL)
+ {
+ printk(KERN_ERR "uart401: bad devc\n");
+ return IRQ_NONE;
+ }
+
+ if (input_avail(devc))
+ uart401_input_loop(devc);
+ return IRQ_HANDLED;
+}
+
+static int
+uart401_open(int dev, int mode,
+ void (*input) (int dev, unsigned char data),
+ void (*output) (int dev)
+)
+{
+ uart401_devc *devc = (uart401_devc *) midi_devs[dev]->devc;
+
+ if (devc->opened)
+ return -EBUSY;
+
+ /* Flush the UART */
+
+ while (input_avail(devc))
+ uart401_read(devc);
+
+ devc->midi_input_intr = input;
+ devc->opened = mode;
+ enter_uart_mode(devc);
+ devc->disabled = 0;
+
+ return 0;
+}
+
+static void uart401_close(int dev)
+{
+ uart401_devc *devc = (uart401_devc *) midi_devs[dev]->devc;
+
+ reset_uart401(devc);
+ devc->opened = 0;
+}
+
+static int uart401_out(int dev, unsigned char midi_byte)
+{
+ int timeout;
+ unsigned long flags;
+ uart401_devc *devc = (uart401_devc *) midi_devs[dev]->devc;
+
+ if (devc->disabled)
+ return 1;
+ /*
+ * Test for input since pending input seems to block the output.
+ */
+
+ spin_lock_irqsave(&devc->lock,flags);
+ if (input_avail(devc))
+ uart401_input_loop(devc);
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ /*
+ * Sometimes it takes about 13000 loops before the output becomes ready
+ * (After reset). Normally it takes just about 10 loops.
+ */
+
+ for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--);
+
+ if (!output_ready(devc))
+ {
+ printk(KERN_WARNING "uart401: Timeout - Device not responding\n");
+ devc->disabled = 1;
+ reset_uart401(devc);
+ enter_uart_mode(devc);
+ return 1;
+ }
+ uart401_write(devc, midi_byte);
+ return 1;
+}
+
+static inline int uart401_start_read(int dev)
+{
+ return 0;
+}
+
+static inline int uart401_end_read(int dev)
+{
+ return 0;
+}
+
+static inline void uart401_kick(int dev)
+{
+}
+
+static inline int uart401_buffer_status(int dev)
+{
+ return 0;
+}
+
+#define MIDI_SYNTH_NAME "MPU-401 UART"
+#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT
+#include "midi_synth.h"
+
+static const struct midi_operations uart401_operations =
+{
+ .owner = THIS_MODULE,
+ .info = {"MPU-401 (UART) MIDI", 0, 0, SNDCARD_MPU401},
+ .converter = &std_midi_synth,
+ .in_info = {0},
+ .open = uart401_open,
+ .close = uart401_close,
+ .outputc = uart401_out,
+ .start_read = uart401_start_read,
+ .end_read = uart401_end_read,
+ .kick = uart401_kick,
+ .buffer_status = uart401_buffer_status,
+};
+
+static void enter_uart_mode(uart401_devc * devc)
+{
+ int ok, timeout;
+ unsigned long flags;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--);
+
+ devc->input_byte = 0;
+ uart401_cmd(devc, UART_MODE_ON);
+
+ ok = 0;
+ for (timeout = 50000; timeout > 0 && !ok; timeout--)
+ if (devc->input_byte == MPU_ACK)
+ ok = 1;
+ else if (input_avail(devc))
+ if (uart401_read(devc) == MPU_ACK)
+ ok = 1;
+
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static int reset_uart401(uart401_devc * devc)
+{
+ int ok, timeout, n;
+
+ /*
+ * Send the RESET command. Try again if no success at the first time.
+ */
+
+ ok = 0;
+
+ for (n = 0; n < 2 && !ok; n++)
+ {
+ for (timeout = 30000; timeout > 0 && !output_ready(devc); timeout--);
+ devc->input_byte = 0;
+ uart401_cmd(devc, MPU_RESET);
+
+ /*
+ * Wait at least 25 msec. This method is not accurate so let's make the
+ * loop bit longer. Cannot sleep since this is called during boot.
+ */
+
+ for (timeout = 50000; timeout > 0 && !ok; timeout--)
+ {
+ if (devc->input_byte == MPU_ACK) /* Interrupt */
+ ok = 1;
+ else if (input_avail(devc))
+ {
+ if (uart401_read(devc) == MPU_ACK)
+ ok = 1;
+ }
+ }
+ }
+
+
+ if (ok)
+ {
+ DEB(printk("Reset UART401 OK\n"));
+ }
+ else
+ DDB(printk("Reset UART401 failed - No hardware detected.\n"));
+
+ if (ok)
+ uart401_input_loop(devc); /*
+ * Flush input before enabling interrupts
+ */
+
+ return ok;
+}
+
+int probe_uart401(struct address_info *hw_config, struct module *owner)
+{
+ uart401_devc *devc;
+ char *name = "MPU-401 (UART) MIDI";
+ int ok = 0;
+ unsigned long flags;
+
+ DDB(printk("Entered probe_uart401()\n"));
+
+ /* Default to "not found" */
+ hw_config->slots[4] = -1;
+
+ if (!request_region(hw_config->io_base, 4, "MPU-401 UART")) {
+ printk(KERN_INFO "uart401: could not request_region(%d, 4)\n", hw_config->io_base);
+ return 0;
+ }
+
+ devc = kmalloc(sizeof(uart401_devc), GFP_KERNEL);
+ if (!devc) {
+ printk(KERN_WARNING "uart401: Can't allocate memory\n");
+ goto cleanup_region;
+ }
+
+ devc->base = hw_config->io_base;
+ devc->irq = hw_config->irq;
+ devc->osp = hw_config->osp;
+ devc->midi_input_intr = NULL;
+ devc->opened = 0;
+ devc->input_byte = 0;
+ devc->my_dev = 0;
+ devc->share_irq = 0;
+ spin_lock_init(&devc->lock);
+
+ spin_lock_irqsave(&devc->lock,flags);
+ ok = reset_uart401(devc);
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ if (!ok)
+ goto cleanup_devc;
+
+ if (hw_config->name)
+ name = hw_config->name;
+
+ if (devc->irq < 0) {
+ devc->share_irq = 1;
+ devc->irq *= -1;
+ } else
+ devc->share_irq = 0;
+
+ if (!devc->share_irq)
+ if (request_irq(devc->irq, uart401intr, 0, "MPU-401 UART", devc) < 0) {
+ printk(KERN_WARNING "uart401: Failed to allocate IRQ%d\n", devc->irq);
+ devc->share_irq = 1;
+ }
+ devc->my_dev = sound_alloc_mididev();
+ enter_uart_mode(devc);
+
+ if (devc->my_dev == -1) {
+ printk(KERN_INFO "uart401: Too many midi devices detected\n");
+ goto cleanup_irq;
+ }
+ conf_printf(name, hw_config);
+ midi_devs[devc->my_dev] = kmalloc(sizeof(struct midi_operations), GFP_KERNEL);
+ if (!midi_devs[devc->my_dev]) {
+ printk(KERN_ERR "uart401: Failed to allocate memory\n");
+ goto cleanup_unload_mididev;
+ }
+ memcpy(midi_devs[devc->my_dev], &uart401_operations, sizeof(struct midi_operations));
+
+ if (owner)
+ midi_devs[devc->my_dev]->owner = owner;
+
+ midi_devs[devc->my_dev]->devc = devc;
+ midi_devs[devc->my_dev]->converter = kmalloc(sizeof(struct synth_operations), GFP_KERNEL);
+ if (!midi_devs[devc->my_dev]->converter) {
+ printk(KERN_WARNING "uart401: Failed to allocate memory\n");
+ goto cleanup_midi_devs;
+ }
+ memcpy(midi_devs[devc->my_dev]->converter, &std_midi_synth, sizeof(struct synth_operations));
+ strcpy(midi_devs[devc->my_dev]->info.name, name);
+ midi_devs[devc->my_dev]->converter->id = "UART401";
+ midi_devs[devc->my_dev]->converter->midi_dev = devc->my_dev;
+
+ if (owner)
+ midi_devs[devc->my_dev]->converter->owner = owner;
+
+ hw_config->slots[4] = devc->my_dev;
+ sequencer_init();
+ devc->opened = 0;
+ return 1;
+cleanup_midi_devs:
+ kfree(midi_devs[devc->my_dev]);
+cleanup_unload_mididev:
+ sound_unload_mididev(devc->my_dev);
+cleanup_irq:
+ if (!devc->share_irq)
+ free_irq(devc->irq, devc);
+cleanup_devc:
+ kfree(devc);
+cleanup_region:
+ release_region(hw_config->io_base, 4);
+ return 0;
+}
+
+void unload_uart401(struct address_info *hw_config)
+{
+ uart401_devc *devc;
+ int n=hw_config->slots[4];
+
+ /* Not set up */
+ if(n==-1 || midi_devs[n]==NULL)
+ return;
+
+ /* Not allocated (erm ??) */
+
+ devc = midi_devs[hw_config->slots[4]]->devc;
+ if (devc == NULL)
+ return;
+
+ reset_uart401(devc);
+ release_region(hw_config->io_base, 4);
+
+ if (!devc->share_irq)
+ free_irq(devc->irq, devc);
+ if (devc)
+ {
+ kfree(midi_devs[devc->my_dev]->converter);
+ kfree(midi_devs[devc->my_dev]);
+ kfree(devc);
+ devc = NULL;
+ }
+ /* This kills midi_devs[x] */
+ sound_unload_mididev(hw_config->slots[4]);
+}
+
+EXPORT_SYMBOL(probe_uart401);
+EXPORT_SYMBOL(unload_uart401);
+EXPORT_SYMBOL(uart401intr);
+
+static struct address_info cfg_mpu;
+
+static int io = -1;
+static int irq = -1;
+
+module_param(io, int, 0444);
+module_param(irq, int, 0444);
+
+
+static int __init init_uart401(void)
+{
+ cfg_mpu.irq = irq;
+ cfg_mpu.io_base = io;
+
+ /* Can be loaded either for module use or to provide functions
+ to others */
+ if (cfg_mpu.io_base != -1 && cfg_mpu.irq != -1) {
+ printk(KERN_INFO "MPU-401 UART driver Copyright (C) Hannu Savolainen 1993-1997");
+ if (!probe_uart401(&cfg_mpu, THIS_MODULE))
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static void __exit cleanup_uart401(void)
+{
+ if (cfg_mpu.io_base != -1 && cfg_mpu.irq != -1)
+ unload_uart401(&cfg_mpu);
+}
+
+module_init(init_uart401);
+module_exit(cleanup_uart401);
+
+#ifndef MODULE
+static int __init setup_uart401(char *str)
+{
+ /* io, irq */
+ int ints[3];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+
+ return 1;
+}
+
+__setup("uart401=", setup_uart401);
+#endif
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/uart6850.c b/sound/oss/uart6850.c
new file mode 100644
index 0000000..f3f914a
--- /dev/null
+++ b/sound/oss/uart6850.c
@@ -0,0 +1,361 @@
+/*
+ * sound/oss/uart6850.c
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ * Extended by Alan Cox for Red Hat Software. Now a loadable MIDI driver.
+ * 28/4/97 - (C) Copyright Alan Cox. Released under the GPL version 2.
+ *
+ * Alan Cox: Updated for new modular code. Removed snd_* irq handling. Now
+ * uses native linux resources
+ * Christoph Hellwig: Adapted to module_init/module_exit
+ * Jeff Garzik: Made it work again, in theory
+ * FIXME: If the request_irq() succeeds, the probe succeeds. Ug.
+ *
+ * Status: Testing required (no shit -jgarzik)
+ *
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/spinlock.h>
+/* Mon Nov 22 22:38:35 MET 1993 marco@driq.home.usn.nl:
+ * added 6850 support, used with COVOX SoundMaster II and custom cards.
+ */
+
+#include "sound_config.h"
+
+static int uart6850_base = 0x330;
+
+static int *uart6850_osp;
+
+#define DATAPORT (uart6850_base)
+#define COMDPORT (uart6850_base+1)
+#define STATPORT (uart6850_base+1)
+
+static int uart6850_status(void)
+{
+ return inb(STATPORT);
+}
+
+#define input_avail() (uart6850_status()&INPUT_AVAIL)
+#define output_ready() (uart6850_status()&OUTPUT_READY)
+
+static void uart6850_cmd(unsigned char cmd)
+{
+ outb(cmd, COMDPORT);
+}
+
+static int uart6850_read(void)
+{
+ return inb(DATAPORT);
+}
+
+static void uart6850_write(unsigned char byte)
+{
+ outb(byte, DATAPORT);
+}
+
+#define OUTPUT_READY 0x02 /* Mask for data ready Bit */
+#define INPUT_AVAIL 0x01 /* Mask for Data Send Ready Bit */
+
+#define UART_RESET 0x95
+#define UART_MODE_ON 0x03
+
+static int uart6850_opened;
+static int uart6850_irq;
+static int uart6850_detected;
+static int my_dev;
+static DEFINE_SPINLOCK(lock);
+
+static void (*midi_input_intr) (int dev, unsigned char data);
+static void poll_uart6850(unsigned long dummy);
+
+
+static DEFINE_TIMER(uart6850_timer, poll_uart6850, 0, 0);
+
+static void uart6850_input_loop(void)
+{
+ int count = 10;
+
+ while (count)
+ {
+ /*
+ * Not timed out
+ */
+ if (input_avail())
+ {
+ unsigned char c = uart6850_read();
+ count = 100;
+ if (uart6850_opened & OPEN_READ)
+ midi_input_intr(my_dev, c);
+ }
+ else
+ {
+ while (!input_avail() && count)
+ count--;
+ }
+ }
+}
+
+static irqreturn_t m6850intr(int irq, void *dev_id)
+{
+ if (input_avail())
+ uart6850_input_loop();
+ return IRQ_HANDLED;
+}
+
+/*
+ * It looks like there is no input interrupts in the UART mode. Let's try
+ * polling.
+ */
+
+static void poll_uart6850(unsigned long dummy)
+{
+ unsigned long flags;
+
+ if (!(uart6850_opened & OPEN_READ))
+ return; /* Device has been closed */
+
+ spin_lock_irqsave(&lock,flags);
+ if (input_avail())
+ uart6850_input_loop();
+
+ uart6850_timer.expires = 1 + jiffies;
+ add_timer(&uart6850_timer);
+
+ /*
+ * Come back later
+ */
+
+ spin_unlock_irqrestore(&lock,flags);
+}
+
+static int uart6850_open(int dev, int mode,
+ void (*input) (int dev, unsigned char data),
+ void (*output) (int dev)
+)
+{
+ if (uart6850_opened)
+ {
+/* printk("Midi6850: Midi busy\n");*/
+ return -EBUSY;
+ };
+
+ uart6850_cmd(UART_RESET);
+ uart6850_input_loop();
+ midi_input_intr = input;
+ uart6850_opened = mode;
+ poll_uart6850(0); /*
+ * Enable input polling
+ */
+
+ return 0;
+}
+
+static void uart6850_close(int dev)
+{
+ uart6850_cmd(UART_MODE_ON);
+ del_timer(&uart6850_timer);
+ uart6850_opened = 0;
+}
+
+static int uart6850_out(int dev, unsigned char midi_byte)
+{
+ int timeout;
+ unsigned long flags;
+
+ /*
+ * Test for input since pending input seems to block the output.
+ */
+
+ spin_lock_irqsave(&lock,flags);
+
+ if (input_avail())
+ uart6850_input_loop();
+
+ spin_unlock_irqrestore(&lock,flags);
+
+ /*
+ * Sometimes it takes about 13000 loops before the output becomes ready
+ * (After reset). Normally it takes just about 10 loops.
+ */
+
+ for (timeout = 30000; timeout > 0 && !output_ready(); timeout--); /*
+ * Wait
+ */
+ if (!output_ready())
+ {
+ printk(KERN_WARNING "Midi6850: Timeout\n");
+ return 0;
+ }
+ uart6850_write(midi_byte);
+ return 1;
+}
+
+static inline int uart6850_command(int dev, unsigned char *midi_byte)
+{
+ return 1;
+}
+
+static inline int uart6850_start_read(int dev)
+{
+ return 0;
+}
+
+static inline int uart6850_end_read(int dev)
+{
+ return 0;
+}
+
+static inline void uart6850_kick(int dev)
+{
+}
+
+static inline int uart6850_buffer_status(int dev)
+{
+ return 0; /*
+ * No data in buffers
+ */
+}
+
+#define MIDI_SYNTH_NAME "6850 UART Midi"
+#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT
+#include "midi_synth.h"
+
+static struct midi_operations uart6850_operations =
+{
+ .owner = THIS_MODULE,
+ .info = {"6850 UART", 0, 0, SNDCARD_UART6850},
+ .converter = &std_midi_synth,
+ .in_info = {0},
+ .open = uart6850_open,
+ .close = uart6850_close,
+ .outputc = uart6850_out,
+ .start_read = uart6850_start_read,
+ .end_read = uart6850_end_read,
+ .kick = uart6850_kick,
+ .command = uart6850_command,
+ .buffer_status = uart6850_buffer_status
+};
+
+
+static void __init attach_uart6850(struct address_info *hw_config)
+{
+ int ok, timeout;
+ unsigned long flags;
+
+ if (!uart6850_detected)
+ return;
+
+ if ((my_dev = sound_alloc_mididev()) == -1)
+ {
+ printk(KERN_INFO "uart6850: Too many midi devices detected\n");
+ return;
+ }
+ uart6850_base = hw_config->io_base;
+ uart6850_osp = hw_config->osp;
+ uart6850_irq = hw_config->irq;
+
+ spin_lock_irqsave(&lock,flags);
+
+ for (timeout = 30000; timeout > 0 && !output_ready(); timeout--); /*
+ * Wait
+ */
+ uart6850_cmd(UART_MODE_ON);
+ ok = 1;
+ spin_unlock_irqrestore(&lock,flags);
+
+ conf_printf("6850 Midi Interface", hw_config);
+
+ std_midi_synth.midi_dev = my_dev;
+ hw_config->slots[4] = my_dev;
+ midi_devs[my_dev] = &uart6850_operations;
+ sequencer_init();
+}
+
+static inline int reset_uart6850(void)
+{
+ uart6850_read();
+ return 1; /*
+ * OK
+ */
+}
+
+static int __init probe_uart6850(struct address_info *hw_config)
+{
+ int ok;
+
+ uart6850_osp = hw_config->osp;
+ uart6850_base = hw_config->io_base;
+ uart6850_irq = hw_config->irq;
+
+ if (request_irq(uart6850_irq, m6850intr, 0, "MIDI6850", NULL) < 0)
+ return 0;
+
+ ok = reset_uart6850();
+ uart6850_detected = ok;
+ return ok;
+}
+
+static void __exit unload_uart6850(struct address_info *hw_config)
+{
+ free_irq(hw_config->irq, NULL);
+ sound_unload_mididev(hw_config->slots[4]);
+}
+
+static struct address_info cfg_mpu;
+
+static int __initdata io = -1;
+static int __initdata irq = -1;
+
+module_param(io, int, 0);
+module_param(irq, int, 0);
+
+static int __init init_uart6850(void)
+{
+ cfg_mpu.io_base = io;
+ cfg_mpu.irq = irq;
+
+ if (cfg_mpu.io_base == -1 || cfg_mpu.irq == -1) {
+ printk(KERN_INFO "uart6850: irq and io must be set.\n");
+ return -EINVAL;
+ }
+
+ if (probe_uart6850(&cfg_mpu))
+ return -ENODEV;
+ attach_uart6850(&cfg_mpu);
+
+ return 0;
+}
+
+static void __exit cleanup_uart6850(void)
+{
+ unload_uart6850(&cfg_mpu);
+}
+
+module_init(init_uart6850);
+module_exit(cleanup_uart6850);
+
+#ifndef MODULE
+static int __init setup_uart6850(char *str)
+{
+ /* io, irq */
+ int ints[3];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+
+ return 1;
+}
+__setup("uart6850=", setup_uart6850);
+#endif
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/ulaw.h b/sound/oss/ulaw.h
new file mode 100644
index 0000000..0ff8c0a
--- /dev/null
+++ b/sound/oss/ulaw.h
@@ -0,0 +1,69 @@
+static unsigned char ulaw_dsp[] = {
+ 3, 7, 11, 15, 19, 23, 27, 31,
+ 35, 39, 43, 47, 51, 55, 59, 63,
+ 66, 68, 70, 72, 74, 76, 78, 80,
+ 82, 84, 86, 88, 90, 92, 94, 96,
+ 98, 99, 100, 101, 102, 103, 104, 105,
+ 106, 107, 108, 109, 110, 111, 112, 113,
+ 113, 114, 114, 115, 115, 116, 116, 117,
+ 117, 118, 118, 119, 119, 120, 120, 121,
+ 121, 121, 122, 122, 122, 122, 123, 123,
+ 123, 123, 124, 124, 124, 124, 125, 125,
+ 125, 125, 125, 125, 126, 126, 126, 126,
+ 126, 126, 126, 126, 127, 127, 127, 127,
+ 127, 127, 127, 127, 127, 127, 127, 127,
+ 128, 128, 128, 128, 128, 128, 128, 128,
+ 128, 128, 128, 128, 128, 128, 128, 128,
+ 128, 128, 128, 128, 128, 128, 128, 128,
+ 253, 249, 245, 241, 237, 233, 229, 225,
+ 221, 217, 213, 209, 205, 201, 197, 193,
+ 190, 188, 186, 184, 182, 180, 178, 176,
+ 174, 172, 170, 168, 166, 164, 162, 160,
+ 158, 157, 156, 155, 154, 153, 152, 151,
+ 150, 149, 148, 147, 146, 145, 144, 143,
+ 143, 142, 142, 141, 141, 140, 140, 139,
+ 139, 138, 138, 137, 137, 136, 136, 135,
+ 135, 135, 134, 134, 134, 134, 133, 133,
+ 133, 133, 132, 132, 132, 132, 131, 131,
+ 131, 131, 131, 131, 130, 130, 130, 130,
+ 130, 130, 130, 130, 129, 129, 129, 129,
+ 129, 129, 129, 129, 129, 129, 129, 129,
+ 128, 128, 128, 128, 128, 128, 128, 128,
+ 128, 128, 128, 128, 128, 128, 128, 128,
+ 128, 128, 128, 128, 128, 128, 128, 128,
+};
+
+static unsigned char dsp_ulaw[] = {
+ 0, 0, 0, 0, 0, 1, 1, 1,
+ 1, 2, 2, 2, 2, 3, 3, 3,
+ 3, 4, 4, 4, 4, 5, 5, 5,
+ 5, 6, 6, 6, 6, 7, 7, 7,
+ 7, 8, 8, 8, 8, 9, 9, 9,
+ 9, 10, 10, 10, 10, 11, 11, 11,
+ 11, 12, 12, 12, 12, 13, 13, 13,
+ 13, 14, 14, 14, 14, 15, 15, 15,
+ 15, 16, 16, 17, 17, 18, 18, 19,
+ 19, 20, 20, 21, 21, 22, 22, 23,
+ 23, 24, 24, 25, 25, 26, 26, 27,
+ 27, 28, 28, 29, 29, 30, 30, 31,
+ 31, 32, 33, 34, 35, 36, 37, 38,
+ 39, 40, 41, 42, 43, 44, 45, 46,
+ 47, 49, 51, 53, 55, 57, 59, 61,
+ 63, 66, 70, 74, 78, 84, 92, 104,
+ 254, 231, 219, 211, 205, 201, 197, 193,
+ 190, 188, 186, 184, 182, 180, 178, 176,
+ 175, 174, 173, 172, 171, 170, 169, 168,
+ 167, 166, 165, 164, 163, 162, 161, 160,
+ 159, 159, 158, 158, 157, 157, 156, 156,
+ 155, 155, 154, 154, 153, 153, 152, 152,
+ 151, 151, 150, 150, 149, 149, 148, 148,
+ 147, 147, 146, 146, 145, 145, 144, 144,
+ 143, 143, 143, 143, 142, 142, 142, 142,
+ 141, 141, 141, 141, 140, 140, 140, 140,
+ 139, 139, 139, 139, 138, 138, 138, 138,
+ 137, 137, 137, 137, 136, 136, 136, 136,
+ 135, 135, 135, 135, 134, 134, 134, 134,
+ 133, 133, 133, 133, 132, 132, 132, 132,
+ 131, 131, 131, 131, 130, 130, 130, 130,
+ 129, 129, 129, 129, 128, 128, 128, 128,
+};
diff --git a/sound/oss/v_midi.c b/sound/oss/v_midi.c
new file mode 100644
index 0000000..103940f
--- /dev/null
+++ b/sound/oss/v_midi.c
@@ -0,0 +1,289 @@
+/*
+ * sound/oss/v_midi.c
+ *
+ * The low level driver for the Sound Blaster DS chips.
+ *
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1996
+ *
+ * USS/Lite for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ * ??
+ *
+ * Changes
+ * Alan Cox Modularisation, changed memory allocations
+ * Christoph Hellwig Adapted to module_init/module_exit
+ *
+ * Status
+ * Untested
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/spinlock.h>
+#include "sound_config.h"
+
+#include "v_midi.h"
+
+static vmidi_devc *v_devc[2] = { NULL, NULL};
+static int midi1,midi2;
+static void *midi_mem = NULL;
+
+/*
+ * The DSP channel can be used either for input or output. Variable
+ * 'sb_irq_mode' will be set when the program calls read or write first time
+ * after open. Current version doesn't support mode changes without closing
+ * and reopening the device. Support for this feature may be implemented in a
+ * future version of this driver.
+ */
+
+
+static int v_midi_open (int dev, int mode,
+ void (*input) (int dev, unsigned char data),
+ void (*output) (int dev)
+)
+{
+ vmidi_devc *devc = midi_devs[dev]->devc;
+ unsigned long flags;
+
+ if (devc == NULL)
+ return -(ENXIO);
+
+ spin_lock_irqsave(&devc->lock,flags);
+ if (devc->opened)
+ {
+ spin_unlock_irqrestore(&devc->lock,flags);
+ return -(EBUSY);
+ }
+ devc->opened = 1;
+ spin_unlock_irqrestore(&devc->lock,flags);
+
+ devc->intr_active = 1;
+
+ if (mode & OPEN_READ)
+ {
+ devc->input_opened = 1;
+ devc->midi_input_intr = input;
+ }
+
+ return 0;
+}
+
+static void v_midi_close (int dev)
+{
+ vmidi_devc *devc = midi_devs[dev]->devc;
+ unsigned long flags;
+
+ if (devc == NULL)
+ return;
+
+ spin_lock_irqsave(&devc->lock,flags);
+ devc->intr_active = 0;
+ devc->input_opened = 0;
+ devc->opened = 0;
+ spin_unlock_irqrestore(&devc->lock,flags);
+}
+
+static int v_midi_out (int dev, unsigned char midi_byte)
+{
+ vmidi_devc *devc = midi_devs[dev]->devc;
+ vmidi_devc *pdevc;
+
+ if (devc == NULL)
+ return -ENXIO;
+
+ pdevc = midi_devs[devc->pair_mididev]->devc;
+ if (pdevc->input_opened > 0){
+ if (MIDIbuf_avail(pdevc->my_mididev) > 500)
+ return 0;
+ pdevc->midi_input_intr (pdevc->my_mididev, midi_byte);
+ }
+ return 1;
+}
+
+static inline int v_midi_start_read (int dev)
+{
+ return 0;
+}
+
+static int v_midi_end_read (int dev)
+{
+ vmidi_devc *devc = midi_devs[dev]->devc;
+ if (devc == NULL)
+ return -ENXIO;
+
+ devc->intr_active = 0;
+ return 0;
+}
+
+/* why -EPERM and not -EINVAL?? */
+
+static inline int v_midi_ioctl (int dev, unsigned cmd, void __user *arg)
+{
+ return -EPERM;
+}
+
+
+#define MIDI_SYNTH_NAME "Loopback MIDI"
+#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT
+
+#include "midi_synth.h"
+
+static struct midi_operations v_midi_operations =
+{
+ .owner = THIS_MODULE,
+ .info = {"Loopback MIDI Port 1", 0, 0, SNDCARD_VMIDI},
+ .converter = &std_midi_synth,
+ .in_info = {0},
+ .open = v_midi_open,
+ .close = v_midi_close,
+ .ioctl = v_midi_ioctl,
+ .outputc = v_midi_out,
+ .start_read = v_midi_start_read,
+ .end_read = v_midi_end_read,
+};
+
+static struct midi_operations v_midi_operations2 =
+{
+ .owner = THIS_MODULE,
+ .info = {"Loopback MIDI Port 2", 0, 0, SNDCARD_VMIDI},
+ .converter = &std_midi_synth,
+ .in_info = {0},
+ .open = v_midi_open,
+ .close = v_midi_close,
+ .ioctl = v_midi_ioctl,
+ .outputc = v_midi_out,
+ .start_read = v_midi_start_read,
+ .end_read = v_midi_end_read,
+};
+
+/*
+ * We kmalloc just one of these - it makes life simpler and the code
+ * cleaner and the memory handling far more efficient
+ */
+
+struct vmidi_memory
+{
+ /* Must be first */
+ struct midi_operations m_ops[2];
+ struct synth_operations s_ops[2];
+ struct vmidi_devc v_ops[2];
+};
+
+static void __init attach_v_midi (struct address_info *hw_config)
+{
+ struct vmidi_memory *m;
+ /* printk("Attaching v_midi device.....\n"); */
+
+ midi1 = sound_alloc_mididev();
+ if (midi1 == -1)
+ {
+ printk(KERN_ERR "v_midi: Too many midi devices detected\n");
+ return;
+ }
+
+ m = kmalloc(sizeof(struct vmidi_memory), GFP_KERNEL);
+ if (m == NULL)
+ {
+ printk(KERN_WARNING "Loopback MIDI: Failed to allocate memory\n");
+ sound_unload_mididev(midi1);
+ return;
+ }
+
+ midi_mem = m;
+
+ midi_devs[midi1] = &m->m_ops[0];
+
+
+ midi2 = sound_alloc_mididev();
+ if (midi2 == -1)
+ {
+ printk (KERN_ERR "v_midi: Too many midi devices detected\n");
+ kfree(m);
+ sound_unload_mididev(midi1);
+ return;
+ }
+
+ midi_devs[midi2] = &m->m_ops[1];
+
+ /* printk("VMIDI1: %d VMIDI2: %d\n",midi1,midi2); */
+
+ /* for MIDI-1 */
+ v_devc[0] = &m->v_ops[0];
+ memcpy ((char *) midi_devs[midi1], (char *) &v_midi_operations,
+ sizeof (struct midi_operations));
+
+ v_devc[0]->my_mididev = midi1;
+ v_devc[0]->pair_mididev = midi2;
+ v_devc[0]->opened = v_devc[0]->input_opened = 0;
+ v_devc[0]->intr_active = 0;
+ v_devc[0]->midi_input_intr = NULL;
+ spin_lock_init(&v_devc[0]->lock);
+
+ midi_devs[midi1]->devc = v_devc[0];
+
+ midi_devs[midi1]->converter = &m->s_ops[0];
+ std_midi_synth.midi_dev = midi1;
+ memcpy ((char *) midi_devs[midi1]->converter, (char *) &std_midi_synth,
+ sizeof (struct synth_operations));
+ midi_devs[midi1]->converter->id = "V_MIDI 1";
+
+ /* for MIDI-2 */
+ v_devc[1] = &m->v_ops[1];
+
+ memcpy ((char *) midi_devs[midi2], (char *) &v_midi_operations2,
+ sizeof (struct midi_operations));
+
+ v_devc[1]->my_mididev = midi2;
+ v_devc[1]->pair_mididev = midi1;
+ v_devc[1]->opened = v_devc[1]->input_opened = 0;
+ v_devc[1]->intr_active = 0;
+ v_devc[1]->midi_input_intr = NULL;
+ spin_lock_init(&v_devc[1]->lock);
+
+ midi_devs[midi2]->devc = v_devc[1];
+ midi_devs[midi2]->converter = &m->s_ops[1];
+
+ std_midi_synth.midi_dev = midi2;
+ memcpy ((char *) midi_devs[midi2]->converter, (char *) &std_midi_synth,
+ sizeof (struct synth_operations));
+ midi_devs[midi2]->converter->id = "V_MIDI 2";
+
+ sequencer_init();
+ /* printk("Attached v_midi device\n"); */
+}
+
+static inline int __init probe_v_midi(struct address_info *hw_config)
+{
+ return(1); /* always OK */
+}
+
+
+static void __exit unload_v_midi(struct address_info *hw_config)
+{
+ sound_unload_mididev(midi1);
+ sound_unload_mididev(midi2);
+ kfree(midi_mem);
+}
+
+static struct address_info cfg; /* dummy */
+
+static int __init init_vmidi(void)
+{
+ printk("MIDI Loopback device driver\n");
+ if (!probe_v_midi(&cfg))
+ return -ENODEV;
+ attach_v_midi(&cfg);
+
+ return 0;
+}
+
+static void __exit cleanup_vmidi(void)
+{
+ unload_v_midi(&cfg);
+}
+
+module_init(init_vmidi);
+module_exit(cleanup_vmidi);
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/v_midi.h b/sound/oss/v_midi.h
new file mode 100644
index 0000000..1b86cb4
--- /dev/null
+++ b/sound/oss/v_midi.h
@@ -0,0 +1,15 @@
+typedef struct vmidi_devc {
+ int dev;
+
+ /* State variables */
+ int opened;
+ spinlock_t lock;
+
+ /* MIDI fields */
+ int my_mididev;
+ int pair_mididev;
+ int input_opened;
+ int intr_active;
+ void (*midi_input_intr) (int dev, unsigned char data);
+ } vmidi_devc;
+
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
new file mode 100644
index 0000000..725fef0
--- /dev/null
+++ b/sound/oss/vidc.c
@@ -0,0 +1,560 @@
+/*
+ * linux/drivers/sound/vidc.c
+ *
+ * Copyright (C) 1997-2000 by Russell King <rmk@arm.linux.org.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * VIDC20 audio driver.
+ *
+ * The VIDC20 sound hardware consists of the VIDC20 itself, a DAC and a DMA
+ * engine. The DMA transfers fixed-format (16-bit little-endian linear)
+ * samples to the VIDC20, which then transfers this data serially to the
+ * DACs. The samplerate is controlled by the VIDC.
+ *
+ * We currently support a mixer device, but it is currently non-functional.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/interrupt.h>
+
+#include <mach/hardware.h>
+#include <asm/dma.h>
+#include <asm/io.h>
+#include <asm/hardware/iomd.h>
+#include <asm/irq.h>
+#include <asm/system.h>
+
+#include "sound_config.h"
+#include "vidc.h"
+
+#ifndef _SIOC_TYPE
+#define _SIOC_TYPE(x) _IOC_TYPE(x)
+#endif
+#ifndef _SIOC_NR
+#define _SIOC_NR(x) _IOC_NR(x)
+#endif
+
+#define VIDC_SOUND_CLOCK (250000)
+#define VIDC_SOUND_CLOCK_EXT (176400)
+
+/*
+ * When using SERIAL SOUND mode (external DAC), the number of physical
+ * channels is fixed at 2.
+ */
+static int vidc_busy;
+static int vidc_adev;
+static int vidc_audio_rate;
+static char vidc_audio_format;
+static char vidc_audio_channels;
+
+static unsigned char vidc_level_l[SOUND_MIXER_NRDEVICES] = {
+ 85, /* master */
+ 50, /* bass */
+ 50, /* treble */
+ 0, /* synth */
+ 75, /* pcm */
+ 0, /* speaker */
+ 100, /* ext line */
+ 0, /* mic */
+ 100, /* CD */
+ 0,
+};
+
+static unsigned char vidc_level_r[SOUND_MIXER_NRDEVICES] = {
+ 85, /* master */
+ 50, /* bass */
+ 50, /* treble */
+ 0, /* synth */
+ 75, /* pcm */
+ 0, /* speaker */
+ 100, /* ext line */
+ 0, /* mic */
+ 100, /* CD */
+ 0,
+};
+
+static unsigned int vidc_audio_volume_l; /* left PCM vol, 0 - 65536 */
+static unsigned int vidc_audio_volume_r; /* right PCM vol, 0 - 65536 */
+
+extern void vidc_update_filler(int bits, int channels);
+extern int softoss_dev;
+
+static void
+vidc_mixer_set(int mdev, unsigned int level)
+{
+ unsigned int lev_l = level & 0x007f;
+ unsigned int lev_r = (level & 0x7f00) >> 8;
+ unsigned int mlev_l, mlev_r;
+
+ if (lev_l > 100)
+ lev_l = 100;
+ if (lev_r > 100)
+ lev_r = 100;
+
+#define SCALE(lev,master) ((lev) * (master) * 65536 / 10000)
+
+ mlev_l = vidc_level_l[SOUND_MIXER_VOLUME];
+ mlev_r = vidc_level_r[SOUND_MIXER_VOLUME];
+
+ switch (mdev) {
+ case SOUND_MIXER_VOLUME:
+ case SOUND_MIXER_PCM:
+ vidc_level_l[mdev] = lev_l;
+ vidc_level_r[mdev] = lev_r;
+
+ vidc_audio_volume_l = SCALE(lev_l, mlev_l);
+ vidc_audio_volume_r = SCALE(lev_r, mlev_r);
+/*printk("VIDC: PCM vol %05X %05X\n", vidc_audio_volume_l, vidc_audio_volume_r);*/
+ break;
+ }
+#undef SCALE
+}
+
+static int vidc_mixer_ioctl(int dev, unsigned int cmd, void __user *arg)
+{
+ unsigned int val;
+ unsigned int mdev;
+
+ if (_SIOC_TYPE(cmd) != 'M')
+ return -EINVAL;
+
+ mdev = _SIOC_NR(cmd);
+
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE) {
+ if (get_user(val, (unsigned int __user *)arg))
+ return -EFAULT;
+
+ if (mdev < SOUND_MIXER_NRDEVICES)
+ vidc_mixer_set(mdev, val);
+ else
+ return -EINVAL;
+ }
+
+ /*
+ * Return parameters
+ */
+ switch (mdev) {
+ case SOUND_MIXER_RECSRC:
+ val = 0;
+ break;
+
+ case SOUND_MIXER_DEVMASK:
+ val = SOUND_MASK_VOLUME | SOUND_MASK_PCM | SOUND_MASK_SYNTH;
+ break;
+
+ case SOUND_MIXER_STEREODEVS:
+ val = SOUND_MASK_VOLUME | SOUND_MASK_PCM | SOUND_MASK_SYNTH;
+ break;
+
+ case SOUND_MIXER_RECMASK:
+ val = 0;
+ break;
+
+ case SOUND_MIXER_CAPS:
+ val = 0;
+ break;
+
+ default:
+ if (mdev < SOUND_MIXER_NRDEVICES)
+ val = vidc_level_l[mdev] | vidc_level_r[mdev] << 8;
+ else
+ return -EINVAL;
+ }
+
+ return put_user(val, (unsigned int __user *)arg) ? -EFAULT : 0;
+}
+
+static unsigned int vidc_audio_set_format(int dev, unsigned int fmt)
+{
+ switch (fmt) {
+ default:
+ fmt = AFMT_S16_LE;
+ case AFMT_U8:
+ case AFMT_S8:
+ case AFMT_S16_LE:
+ vidc_audio_format = fmt;
+ vidc_update_filler(vidc_audio_format, vidc_audio_channels);
+ case AFMT_QUERY:
+ break;
+ }
+ return vidc_audio_format;
+}
+
+#define my_abs(i) ((i)<0 ? -(i) : (i))
+
+static int vidc_audio_set_speed(int dev, int rate)
+{
+ if (rate) {
+ unsigned int hwctrl, hwrate, hwrate_ext, rate_int, rate_ext;
+ unsigned int diff_int, diff_ext;
+ unsigned int newsize, new2size;
+
+ hwctrl = 0x00000003;
+
+ /* Using internal clock */
+ hwrate = (((VIDC_SOUND_CLOCK * 2) / rate) + 1) >> 1;
+ if (hwrate < 3)
+ hwrate = 3;
+ if (hwrate > 255)
+ hwrate = 255;
+
+ /* Using exernal clock */
+ hwrate_ext = (((VIDC_SOUND_CLOCK_EXT * 2) / rate) + 1) >> 1;
+ if (hwrate_ext < 3)
+ hwrate_ext = 3;
+ if (hwrate_ext > 255)
+ hwrate_ext = 255;
+
+ rate_int = VIDC_SOUND_CLOCK / hwrate;
+ rate_ext = VIDC_SOUND_CLOCK_EXT / hwrate_ext;
+
+ /* Chose between external and internal clock */
+ diff_int = my_abs(rate_ext-rate);
+ diff_ext = my_abs(rate_int-rate);
+ if (diff_ext < diff_int) {
+ /*printk("VIDC: external %d %d %d\n", rate, rate_ext, hwrate_ext);*/
+ hwrate=hwrate_ext;
+ hwctrl=0x00000002;
+ /* Allow roughly 0.4% tolerance */
+ if (diff_ext > (rate/256))
+ rate=rate_ext;
+ } else {
+ /*printk("VIDC: internal %d %d %d\n", rate, rate_int, hwrate);*/
+ hwctrl=0x00000003;
+ /* Allow rougly 0.4% tolerance */
+ if (diff_int > (rate/256))
+ rate=rate_int;
+ }
+
+ vidc_writel(0xb0000000 | (hwrate - 2));
+ vidc_writel(0xb1000000 | hwctrl);
+
+ newsize = (10000 / hwrate) & ~3;
+ if (newsize < 208)
+ newsize = 208;
+ if (newsize > 4096)
+ newsize = 4096;
+ for (new2size = 128; new2size < newsize; new2size <<= 1);
+ if (new2size - newsize > newsize - (new2size >> 1))
+ new2size >>= 1;
+ if (new2size > 4096) {
+ printk(KERN_ERR "VIDC: error: dma buffer (%d) %d > 4K\n",
+ newsize, new2size);
+ new2size = 4096;
+ }
+ /*printk("VIDC: dma size %d\n", new2size);*/
+ dma_bufsize = new2size;
+ vidc_audio_rate = rate;
+ }
+ return vidc_audio_rate;
+}
+
+static short vidc_audio_set_channels(int dev, short channels)
+{
+ switch (channels) {
+ default:
+ channels = 2;
+ case 1:
+ case 2:
+ vidc_audio_channels = channels;
+ vidc_update_filler(vidc_audio_format, vidc_audio_channels);
+ case 0:
+ break;
+ }
+ return vidc_audio_channels;
+}
+
+/*
+ * Open the device
+ */
+static int vidc_audio_open(int dev, int mode)
+{
+ /* This audio device does not have recording capability */
+ if (mode == OPEN_READ)
+ return -EPERM;
+
+ if (vidc_busy)
+ return -EBUSY;
+
+ vidc_busy = 1;
+ return 0;
+}
+
+/*
+ * Close the device
+ */
+static void vidc_audio_close(int dev)
+{
+ vidc_busy = 0;
+}
+
+/*
+ * Output a block via DMA to sound device.
+ *
+ * We just set the DMA start and count; the DMA interrupt routine
+ * will take care of formatting the samples (via the appropriate
+ * vidc_filler routine), and flag via vidc_audio_dma_interrupt when
+ * more data is required.
+ */
+static void
+vidc_audio_output_block(int dev, unsigned long buf, int total_count, int one)
+{
+ struct dma_buffparms *dmap = audio_devs[dev]->dmap_out;
+ unsigned long flags;
+
+ local_irq_save(flags);
+ dma_start = buf - (unsigned long)dmap->raw_buf_phys + (unsigned long)dmap->raw_buf;
+ dma_count = total_count;
+ local_irq_restore(flags);
+}
+
+static void
+vidc_audio_start_input(int dev, unsigned long buf, int count, int intrflag)
+{
+}
+
+static int vidc_audio_prepare_for_input(int dev, int bsize, int bcount)
+{
+ return -EINVAL;
+}
+
+static irqreturn_t vidc_audio_dma_interrupt(void)
+{
+ DMAbuf_outputintr(vidc_adev, 1);
+ return IRQ_HANDLED;
+}
+
+/*
+ * Prepare for outputting samples.
+ *
+ * Each buffer that will be passed will be `bsize' bytes long,
+ * with a total of `bcount' buffers.
+ */
+static int vidc_audio_prepare_for_output(int dev, int bsize, int bcount)
+{
+ struct audio_operations *adev = audio_devs[dev];
+
+ dma_interrupt = NULL;
+ adev->dmap_out->flags |= DMA_NODMA;
+
+ return 0;
+}
+
+/*
+ * Stop our current operation.
+ */
+static void vidc_audio_reset(int dev)
+{
+ dma_interrupt = NULL;
+}
+
+static int vidc_audio_local_qlen(int dev)
+{
+ return /*dma_count !=*/ 0;
+}
+
+static void vidc_audio_trigger(int dev, int enable_bits)
+{
+ struct audio_operations *adev = audio_devs[dev];
+
+ if (enable_bits & PCM_ENABLE_OUTPUT) {
+ if (!(adev->flags & DMA_ACTIVE)) {
+ unsigned long flags;
+
+ local_irq_save(flags);
+
+ /* prevent recusion */
+ adev->flags |= DMA_ACTIVE;
+
+ dma_interrupt = vidc_audio_dma_interrupt;
+ vidc_sound_dma_irq(0, NULL);
+ iomd_writeb(DMA_CR_E | 0x10, IOMD_SD0CR);
+
+ local_irq_restore(flags);
+ }
+ }
+}
+
+static struct audio_driver vidc_audio_driver =
+{
+ .owner = THIS_MODULE,
+ .open = vidc_audio_open,
+ .close = vidc_audio_close,
+ .output_block = vidc_audio_output_block,
+ .start_input = vidc_audio_start_input,
+ .prepare_for_input = vidc_audio_prepare_for_input,
+ .prepare_for_output = vidc_audio_prepare_for_output,
+ .halt_io = vidc_audio_reset,
+ .local_qlen = vidc_audio_local_qlen,
+ .trigger = vidc_audio_trigger,
+ .set_speed = vidc_audio_set_speed,
+ .set_bits = vidc_audio_set_format,
+ .set_channels = vidc_audio_set_channels
+};
+
+static struct mixer_operations vidc_mixer_operations = {
+ .owner = THIS_MODULE,
+ .id = "VIDC",
+ .name = "VIDCsound",
+ .ioctl = vidc_mixer_ioctl
+};
+
+void vidc_update_filler(int format, int channels)
+{
+#define TYPE(fmt,ch) (((fmt)<<2) | ((ch)&3))
+
+ switch (TYPE(format, channels)) {
+ default:
+ case TYPE(AFMT_U8, 1):
+ vidc_filler = vidc_fill_1x8_u;
+ break;
+
+ case TYPE(AFMT_U8, 2):
+ vidc_filler = vidc_fill_2x8_u;
+ break;
+
+ case TYPE(AFMT_S8, 1):
+ vidc_filler = vidc_fill_1x8_s;
+ break;
+
+ case TYPE(AFMT_S8, 2):
+ vidc_filler = vidc_fill_2x8_s;
+ break;
+
+ case TYPE(AFMT_S16_LE, 1):
+ vidc_filler = vidc_fill_1x16_s;
+ break;
+
+ case TYPE(AFMT_S16_LE, 2):
+ vidc_filler = vidc_fill_2x16_s;
+ break;
+ }
+}
+
+static void __init attach_vidc(struct address_info *hw_config)
+{
+ char name[32];
+ int i, adev;
+
+ sprintf(name, "VIDC %d-bit sound", hw_config->card_subtype);
+ conf_printf(name, hw_config);
+ memset(dma_buf, 0, sizeof(dma_buf));
+
+ adev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, name,
+ &vidc_audio_driver, sizeof(vidc_audio_driver),
+ DMA_AUTOMODE, AFMT_U8 | AFMT_S8 | AFMT_S16_LE,
+ NULL, hw_config->dma, hw_config->dma2);
+
+ if (adev < 0)
+ goto audio_failed;
+
+ /*
+ * 1024 bytes => 64 buffers
+ */
+ audio_devs[adev]->min_fragment = 10;
+ audio_devs[adev]->mixer_dev = num_mixers;
+
+ audio_devs[adev]->mixer_dev =
+ sound_install_mixer(MIXER_DRIVER_VERSION,
+ name, &vidc_mixer_operations,
+ sizeof(vidc_mixer_operations), NULL);
+
+ if (audio_devs[adev]->mixer_dev < 0)
+ goto mixer_failed;
+
+ for (i = 0; i < 2; i++) {
+ dma_buf[i] = get_zeroed_page(GFP_KERNEL);
+ if (!dma_buf[i]) {
+ printk(KERN_ERR "%s: can't allocate required buffers\n",
+ name);
+ goto mem_failed;
+ }
+ dma_pbuf[i] = virt_to_phys((void *)dma_buf[i]);
+ }
+
+ if (sound_alloc_dma(hw_config->dma, hw_config->name)) {
+ printk(KERN_ERR "%s: DMA %d is in use\n", name, hw_config->dma);
+ goto dma_failed;
+ }
+
+ if (request_irq(hw_config->irq, vidc_sound_dma_irq, 0,
+ hw_config->name, &dma_start)) {
+ printk(KERN_ERR "%s: IRQ %d is in use\n", name, hw_config->irq);
+ goto irq_failed;
+ }
+ vidc_adev = adev;
+ vidc_mixer_set(SOUND_MIXER_VOLUME, (85 | 85 << 8));
+
+#if defined(CONFIG_SOUND_SOFTOSS) || defined(CONFIG_SOUND_SOFTOSS_MODULE)
+ softoss_dev = adev;
+#endif
+ return;
+
+irq_failed:
+ sound_free_dma(hw_config->dma);
+dma_failed:
+mem_failed:
+ for (i = 0; i < 2; i++)
+ free_page(dma_buf[i]);
+ sound_unload_mixerdev(audio_devs[adev]->mixer_dev);
+mixer_failed:
+ sound_unload_audiodev(adev);
+audio_failed:
+ return;
+}
+
+static int __init probe_vidc(struct address_info *hw_config)
+{
+ hw_config->irq = IRQ_DMAS0;
+ hw_config->dma = DMA_VIRTUAL_SOUND;
+ hw_config->dma2 = -1;
+ hw_config->card_subtype = 16;
+ hw_config->name = "VIDC20";
+ return 1;
+}
+
+static void __exit unload_vidc(struct address_info *hw_config)
+{
+ int i, adev = vidc_adev;
+
+ vidc_adev = -1;
+
+ free_irq(hw_config->irq, &dma_start);
+ sound_free_dma(hw_config->dma);
+
+ if (adev >= 0) {
+ sound_unload_mixerdev(audio_devs[adev]->mixer_dev);
+ sound_unload_audiodev(adev);
+ for (i = 0; i < 2; i++)
+ free_page(dma_buf[i]);
+ }
+}
+
+static struct address_info cfg;
+
+static int __init init_vidc(void)
+{
+ if (probe_vidc(&cfg) == 0)
+ return -ENODEV;
+
+ attach_vidc(&cfg);
+
+ return 0;
+}
+
+static void __exit cleanup_vidc(void)
+{
+ unload_vidc(&cfg);
+}
+
+module_init(init_vidc);
+module_exit(cleanup_vidc);
+
+MODULE_AUTHOR("Russell King");
+MODULE_DESCRIPTION("VIDC20 audio driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/vidc.h b/sound/oss/vidc.h
new file mode 100644
index 0000000..0d14247
--- /dev/null
+++ b/sound/oss/vidc.h
@@ -0,0 +1,63 @@
+/*
+ * linux/drivers/sound/vidc.h
+ *
+ * Copyright (C) 1997 Russell King <rmk@arm.linux.org.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * VIDC sound function prototypes
+ */
+
+/* vidc_fill.S */
+
+/*
+ * Filler routines for different channels and sample sizes
+ */
+
+extern unsigned long vidc_fill_1x8_u(unsigned long ibuf, unsigned long iend,
+ unsigned long obuf, int mask);
+extern unsigned long vidc_fill_2x8_u(unsigned long ibuf, unsigned long iend,
+ unsigned long obuf, int mask);
+extern unsigned long vidc_fill_1x8_s(unsigned long ibuf, unsigned long iend,
+ unsigned long obuf, int mask);
+extern unsigned long vidc_fill_2x8_s(unsigned long ibuf, unsigned long iend,
+ unsigned long obuf, int mask);
+extern unsigned long vidc_fill_1x16_s(unsigned long ibuf, unsigned long iend,
+ unsigned long obuf, int mask);
+extern unsigned long vidc_fill_2x16_s(unsigned long ibuf, unsigned long iend,
+ unsigned long obuf, int mask);
+
+/*
+ * DMA Interrupt handler
+ */
+
+extern irqreturn_t vidc_sound_dma_irq(int irqnr, void *ref);
+
+/*
+ * Filler routine pointer
+ */
+
+extern unsigned long (*vidc_filler) (unsigned long ibuf, unsigned long iend,
+ unsigned long obuf, int mask);
+
+/*
+ * Virtual DMA buffer exhausted
+ */
+
+extern irqreturn_t (*dma_interrupt) (void);
+
+/*
+ * Virtual DMA buffer addresses
+ */
+
+extern unsigned long dma_start, dma_count, dma_bufsize;
+extern unsigned long dma_buf[2], dma_pbuf[2];
+
+/* vidc_synth.c */
+
+extern void vidc_synth_init(struct address_info *hw_config);
+extern void vidc_synth_exit(struct address_info *hw_config);
+extern int vidc_synth_get_volume(void);
+extern int vidc_synth_set_volume(int vol);
diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S
new file mode 100644
index 0000000..bed3492
--- /dev/null
+++ b/sound/oss/vidc_fill.S
@@ -0,0 +1,218 @@
+/*
+ * linux/drivers/sound/vidc_fill.S
+ *
+ * Copyright (C) 1997 Russell King
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Filler routines for DMA buffers
+ */
+#include <linux/linkage.h>
+#include <asm/assembler.h>
+#include <mach/hardware.h>
+#include <asm/hardware/iomd.h>
+
+ .text
+
+ENTRY(vidc_fill_1x8_u)
+ mov ip, #0xff00
+1: cmp r0, r1
+ bge vidc_clear
+ ldrb r4, [r0], #1
+ eor r4, r4, #0x80
+ and r4, ip, r4, lsl #8
+ orr r4, r4, r4, lsl #16
+ str r4, [r2], #4
+ cmp r2, r3
+ blt 1b
+ mov pc, lr
+
+ENTRY(vidc_fill_2x8_u)
+ mov ip, #0xff00
+1: cmp r0, r1
+ bge vidc_clear
+ ldr r4, [r0], #2
+ and r5, r4, ip
+ and r4, ip, r4, lsl #8
+ orr r4, r4, r5, lsl #16
+ orr r4, r4, r4, lsr #8
+ str r4, [r2], #4
+ cmp r2, r3
+ blt 1b
+ mov pc, lr
+
+ENTRY(vidc_fill_1x8_s)
+ mov ip, #0xff00
+1: cmp r0, r1
+ bge vidc_clear
+ ldrb r4, [r0], #1
+ and r4, ip, r4, lsl #8
+ orr r4, r4, r4, lsl #16
+ str r4, [r2], #4
+ cmp r2, r3
+ blt 1b
+ mov pc, lr
+
+ENTRY(vidc_fill_2x8_s)
+ mov ip, #0xff00
+1: cmp r0, r1
+ bge vidc_clear
+ ldr r4, [r0], #2
+ and r5, r4, ip
+ and r4, ip, r4, lsl #8
+ orr r4, r4, r5, lsl #16
+ orr r4, r4, r4, lsr #8
+ str r4, [r2], #4
+ cmp r2, r3
+ blt 1b
+ mov pc, lr
+
+ENTRY(vidc_fill_1x16_s)
+ mov ip, #0xff00
+ orr ip, ip, ip, lsr #8
+1: cmp r0, r1
+ bge vidc_clear
+ ldr r5, [r0], #2
+ and r4, r5, ip
+ orr r4, r4, r4, lsl #16
+ str r4, [r2], #4
+ cmp r0, r1
+ addlt r0, r0, #2
+ andlt r4, r5, ip, lsl #16
+ orrlt r4, r4, r4, lsr #16
+ strlt r4, [r2], #4
+ cmp r2, r3
+ blt 1b
+ mov pc, lr
+
+ENTRY(vidc_fill_2x16_s)
+ mov ip, #0xff00
+ orr ip, ip, ip, lsr #8
+1: cmp r0, r1
+ bge vidc_clear
+ ldr r4, [r0], #4
+ str r4, [r2], #4
+ cmp r0, r1
+ ldrlt r4, [r0], #4
+ strlt r4, [r2], #4
+ cmp r2, r3
+ blt 1b
+ mov pc, lr
+
+ENTRY(vidc_fill_noaudio)
+ mov r0, #0
+ mov r1, #0
+2: mov r4, #0
+ mov r5, #0
+1: cmp r2, r3
+ stmltia r2!, {r0, r1, r4, r5}
+ blt 1b
+ mov pc, lr
+
+ENTRY(vidc_clear)
+ mov r0, #0
+ mov r1, #0
+ tst r2, #4
+ str r0, [r2], #4
+ tst r2, #8
+ stmia r2!, {r0, r1}
+ b 2b
+
+/*
+ * Call filler routines with:
+ * r0 = phys address
+ * r1 = phys end
+ * r2 = buffer
+ * Returns:
+ * r0 = new buffer address
+ * r2 = new buffer finish
+ * r4 = corrupted
+ * r5 = corrupted
+ * ip = corrupted
+ */
+
+ENTRY(vidc_sound_dma_irq)
+ stmfd sp!, {r4 - r8, lr}
+ ldr r8, =dma_start
+ ldmia r8, {r0, r1, r2, r3, r4, r5}
+ teq r1, #0
+ adreq r4, vidc_fill_noaudio
+ moveq r7, #1 << 31
+ movne r7, #0
+ mov ip, #IOMD_BASE & 0xff000000
+ orr ip, ip, #IOMD_BASE & 0x00ff0000
+ ldrb r6, [ip, #IOMD_SD0ST]
+ tst r6, #DMA_ST_OFL @ Check for overrun
+ eorne r6, r6, #DMA_ST_AB
+ tst r6, #DMA_ST_AB
+ moveq r2, r3 @ DMAing A, update B
+ add r3, r2, r5 @ End of DMA buffer
+ add r1, r1, r0 @ End of virtual DMA buffer
+ mov lr, pc
+ mov pc, r4 @ Call fill routine (uses r4, ip)
+ sub r1, r1, r0 @ Remaining length
+ stmia r8, {r0, r1}
+ mov r0, #0
+ tst r2, #4 @ Round buffer up to 4 words
+ strne r0, [r2], #4
+ tst r2, #8
+ strne r0, [r2], #4
+ strne r0, [r2], #4
+ sub r2, r2, #16
+ mov r2, r2, lsl #20
+ movs r2, r2, lsr #20
+ orreq r2, r2, #1 << 30 @ Set L bit
+ orr r2, r2, r7
+ ldmdb r8, {r3, r4, r5}
+ tst r6, #DMA_ST_AB
+ mov ip, #IOMD_BASE & 0xff000000
+ orr ip, ip, #IOMD_BASE & 0x00ff0000
+ streq r4, [ip, #IOMD_SD0CURB]
+ strne r5, [ip, #IOMD_SD0CURA]
+ streq r2, [ip, #IOMD_SD0ENDB]
+ strne r2, [ip, #IOMD_SD0ENDA]
+ ldr lr, [ip, #IOMD_SD0ST]
+ tst lr, #DMA_ST_OFL
+ bne 1f
+ tst r6, #DMA_ST_AB
+ strne r4, [ip, #IOMD_SD0CURB]
+ streq r5, [ip, #IOMD_SD0CURA]
+ strne r2, [ip, #IOMD_SD0ENDB]
+ streq r2, [ip, #IOMD_SD0ENDA]
+1: teq r7, #0
+ mov r0, #0x10
+ strneb r0, [ip, #IOMD_SD0CR]
+ ldmfd sp!, {r4 - r8, lr}
+ mov r0, #1 @ IRQ_HANDLED
+ teq r1, #0 @ If we have no more
+ movne pc, lr
+ teq r3, #0
+ movne pc, r3 @ Call interrupt routine
+ mov pc, lr
+
+ .data
+ .globl dma_interrupt
+dma_interrupt:
+ .long 0 @ r3
+ .globl dma_pbuf
+dma_pbuf:
+ .long 0 @ r4
+ .long 0 @ r5
+ .globl dma_start
+dma_start:
+ .long 0 @ r0
+ .globl dma_count
+dma_count:
+ .long 0 @ r1
+ .globl dma_buf
+dma_buf:
+ .long 0 @ r2
+ .long 0 @ r3
+ .globl vidc_filler
+vidc_filler:
+ .long vidc_fill_noaudio @ r4
+ .globl dma_bufsize
+dma_bufsize:
+ .long 0x1000 @ r5
diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c
new file mode 100644
index 0000000..78b8acc
--- /dev/null
+++ b/sound/oss/vwsnd.c
@@ -0,0 +1,3485 @@
+/*
+ * Sound driver for Silicon Graphics 320 and 540 Visual Workstations'
+ * onboard audio. See notes in Documentation/sound/oss/vwsnd .
+ *
+ * Copyright 1999 Silicon Graphics, Inc. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#undef VWSND_DEBUG /* define for debugging */
+
+/*
+ * XXX to do -
+ *
+ * External sync.
+ * Rename swbuf, hwbuf, u&i, hwptr&swptr to something rational.
+ * Bug - if select() called before read(), pcm_setup() not called.
+ * Bug - output doesn't stop soon enough if process killed.
+ */
+
+/*
+ * Things to test -
+ *
+ * Will readv/writev work? Write a test.
+ *
+ * insmod/rmmod 100 million times.
+ *
+ * Run I/O until int ptrs wrap around (roughly 6.2 hours @ DAT
+ * rate).
+ *
+ * Concurrent threads banging on mixer simultaneously, both UP
+ * and SMP kernels. Especially, watch for thread A changing
+ * OUTSRC while thread B changes gain -- both write to the same
+ * ad1843 register.
+ *
+ * What happens if a client opens /dev/audio then forks?
+ * Do two procs have /dev/audio open? Test.
+ *
+ * Pump audio through the CD, MIC and line inputs and verify that
+ * they mix/mute into the output.
+ *
+ * Apps:
+ * amp
+ * mpg123
+ * x11amp
+ * mxv
+ * kmedia
+ * esound
+ * need more input apps
+ *
+ * Run tests while bombarding with signals. setitimer(2) will do it... */
+
+/*
+ * This driver is organized in nine sections.
+ * The nine sections are:
+ *
+ * debug stuff
+ * low level lithium access
+ * high level lithium access
+ * AD1843 access
+ * PCM I/O
+ * audio driver
+ * mixer driver
+ * probe/attach/unload
+ * initialization and loadable kernel module interface
+ *
+ * That is roughly the order of increasing abstraction, so forward
+ * dependencies are minimal.
+ */
+
+/*
+ * Locking Notes
+ *
+ * INC_USE_COUNT and DEC_USE_COUNT keep track of the number of
+ * open descriptors to this driver. They store it in vwsnd_use_count.
+ * The global device list, vwsnd_dev_list, is immutable when the IN_USE
+ * is true.
+ *
+ * devc->open_lock is a semaphore that is used to enforce the
+ * single reader/single writer rule for /dev/audio. The rule is
+ * that each device may have at most one reader and one writer.
+ * Open will block until the previous client has closed the
+ * device, unless O_NONBLOCK is specified.
+ *
+ * The semaphore devc->io_mutex serializes PCM I/O syscalls. This
+ * is unnecessary in Linux 2.2, because the kernel lock
+ * serializes read, write, and ioctl globally, but it's there,
+ * ready for the brave, new post-kernel-lock world.
+ *
+ * Locking between interrupt and baselevel is handled by the
+ * "lock" spinlock in vwsnd_port (one lock each for read and
+ * write). Each half holds the lock just long enough to see what
+ * area it owns and update its pointers. See pcm_output() and
+ * pcm_input() for most of the gory stuff.
+ *
+ * devc->mix_mutex serializes all mixer ioctls. This is also
+ * redundant because of the kernel lock.
+ *
+ * The lowest level lock is lith->lithium_lock. It is a
+ * spinlock which is held during the two-register tango of
+ * reading/writing an AD1843 register. See
+ * li_{read,write}_ad1843_reg().
+ */
+
+/*
+ * Sample Format Notes
+ *
+ * Lithium's DMA engine has two formats: 16-bit 2's complement
+ * and 8-bit unsigned . 16-bit transfers the data unmodified, 2
+ * bytes per sample. 8-bit unsigned transfers 1 byte per sample
+ * and XORs each byte with 0x80. Lithium can input or output
+ * either mono or stereo in either format.
+ *
+ * The AD1843 has four formats: 16-bit 2's complement, 8-bit
+ * unsigned, 8-bit mu-Law and 8-bit A-Law.
+ *
+ * This driver supports five formats: AFMT_S8, AFMT_U8,
+ * AFMT_MU_LAW, AFMT_A_LAW, and AFMT_S16_LE.
+ *
+ * For AFMT_U8 output, we keep the AD1843 in 16-bit mode, and
+ * rely on Lithium's XOR to translate between U8 and S8.
+ *
+ * For AFMT_S8, AFMT_MU_LAW and AFMT_A_LAW output, we have to XOR
+ * the 0x80 bit in software to compensate for Lithium's XOR.
+ * This happens in pcm_copy_{in,out}().
+ *
+ * Changes:
+ * 11-10-2000 Bartlomiej Zolnierkiewicz <bkz@linux-ide.org>
+ * Added some __init/__exit
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+
+#include <linux/spinlock.h>
+#include <linux/smp_lock.h>
+#include <linux/wait.h>
+#include <linux/interrupt.h>
+#include <linux/mutex.h>
+
+#include <asm/visws/cobalt.h>
+
+#include "sound_config.h"
+
+/*****************************************************************************/
+/* debug stuff */
+
+#ifdef VWSND_DEBUG
+
+static int shut_up = 1;
+
+/*
+ * dbgassert - called when an assertion fails.
+ */
+
+static void dbgassert(const char *fcn, int line, const char *expr)
+{
+ if (in_interrupt())
+ panic("ASSERTION FAILED IN INTERRUPT, %s:%s:%d %s\n",
+ __FILE__, fcn, line, expr);
+ else {
+ int x;
+ printk(KERN_ERR "ASSERTION FAILED, %s:%s:%d %s\n",
+ __FILE__, fcn, line, expr);
+ x = * (volatile int *) 0; /* force proc to exit */
+ }
+}
+
+/*
+ * Bunch of useful debug macros:
+ *
+ * ASSERT - print unless e nonzero (panic if in interrupt)
+ * DBGDO - include arbitrary code if debugging
+ * DBGX - debug print raw (w/o function name)
+ * DBGP - debug print w/ function name
+ * DBGE - debug print function entry
+ * DBGC - debug print function call
+ * DBGR - debug print function return
+ * DBGXV - debug print raw when verbose
+ * DBGPV - debug print when verbose
+ * DBGEV - debug print function entry when verbose
+ * DBGRV - debug print function return when verbose
+ */
+
+#define ASSERT(e) ((e) ? (void) 0 : dbgassert(__func__, __LINE__, #e))
+#define DBGDO(x) x
+#define DBGX(fmt, args...) (in_interrupt() ? 0 : printk(KERN_ERR fmt, ##args))
+#define DBGP(fmt, args...) (DBGX("%s: " fmt, __func__ , ##args))
+#define DBGE(fmt, args...) (DBGX("%s" fmt, __func__ , ##args))
+#define DBGC(rtn) (DBGP("calling %s\n", rtn))
+#define DBGR() (DBGP("returning\n"))
+#define DBGXV(fmt, args...) (shut_up ? 0 : DBGX(fmt, ##args))
+#define DBGPV(fmt, args...) (shut_up ? 0 : DBGP(fmt, ##args))
+#define DBGEV(fmt, args...) (shut_up ? 0 : DBGE(fmt, ##args))
+#define DBGCV(rtn) (shut_up ? 0 : DBGC(rtn))
+#define DBGRV() (shut_up ? 0 : DBGR())
+
+#else /* !VWSND_DEBUG */
+
+#define ASSERT(e) ((void) 0)
+#define DBGDO(x) /* don't */
+#define DBGX(fmt, args...) ((void) 0)
+#define DBGP(fmt, args...) ((void) 0)
+#define DBGE(fmt, args...) ((void) 0)
+#define DBGC(rtn) ((void) 0)
+#define DBGR() ((void) 0)
+#define DBGPV(fmt, args...) ((void) 0)
+#define DBGXV(fmt, args...) ((void) 0)
+#define DBGEV(fmt, args...) ((void) 0)
+#define DBGCV(rtn) ((void) 0)
+#define DBGRV() ((void) 0)
+
+#endif /* !VWSND_DEBUG */
+
+/*****************************************************************************/
+/* low level lithium access */
+
+/*
+ * We need to talk to Lithium registers on three pages. Here are
+ * the pages' offsets from the base address (0xFF001000).
+ */
+
+enum {
+ LI_PAGE0_OFFSET = 0x01000 - 0x1000, /* FF001000 */
+ LI_PAGE1_OFFSET = 0x0F000 - 0x1000, /* FF00F000 */
+ LI_PAGE2_OFFSET = 0x10000 - 0x1000, /* FF010000 */
+};
+
+/* low-level lithium data */
+
+typedef struct lithium {
+ void * page0; /* virtual addresses */
+ void * page1;
+ void * page2;
+ spinlock_t lock; /* protects codec and UST/MSC access */
+} lithium_t;
+
+/*
+ * li_destroy destroys the lithium_t structure and vm mappings.
+ */
+
+static void li_destroy(lithium_t *lith)
+{
+ if (lith->page0) {
+ iounmap(lith->page0);
+ lith->page0 = NULL;
+ }
+ if (lith->page1) {
+ iounmap(lith->page1);
+ lith->page1 = NULL;
+ }
+ if (lith->page2) {
+ iounmap(lith->page2);
+ lith->page2 = NULL;
+ }
+}
+
+/*
+ * li_create initializes the lithium_t structure and sets up vm mappings
+ * to access the registers.
+ * Returns 0 on success, -errno on failure.
+ */
+
+static int __init li_create(lithium_t *lith, unsigned long baseaddr)
+{
+ spin_lock_init(&lith->lock);
+ lith->page0 = ioremap_nocache(baseaddr + LI_PAGE0_OFFSET, PAGE_SIZE);
+ lith->page1 = ioremap_nocache(baseaddr + LI_PAGE1_OFFSET, PAGE_SIZE);
+ lith->page2 = ioremap_nocache(baseaddr + LI_PAGE2_OFFSET, PAGE_SIZE);
+ if (!lith->page0 || !lith->page1 || !lith->page2) {
+ li_destroy(lith);
+ return -ENOMEM;
+ }
+ return 0;
+}
+
+/*
+ * basic register accessors - read/write long/byte
+ */
+
+static __inline__ unsigned long li_readl(lithium_t *lith, int off)
+{
+ return * (volatile unsigned long *) (lith->page0 + off);
+}
+
+static __inline__ unsigned char li_readb(lithium_t *lith, int off)
+{
+ return * (volatile unsigned char *) (lith->page0 + off);
+}
+
+static __inline__ void li_writel(lithium_t *lith, int off, unsigned long val)
+{
+ * (volatile unsigned long *) (lith->page0 + off) = val;
+}
+
+static __inline__ void li_writeb(lithium_t *lith, int off, unsigned char val)
+{
+ * (volatile unsigned char *) (lith->page0 + off) = val;
+}
+
+/*****************************************************************************/
+/* High Level Lithium Access */
+
+/*
+ * Lithium DMA Notes
+ *
+ * Lithium has two dedicated DMA channels for audio. They are known
+ * as comm1 and comm2 (communication areas 1 and 2). Comm1 is for
+ * input, and comm2 is for output. Each is controlled by three
+ * registers: BASE (base address), CFG (config) and CCTL
+ * (config/control).
+ *
+ * Each DMA channel points to a physically contiguous ring buffer in
+ * main memory of up to 8 Kbytes. (This driver always uses 8 Kb.)
+ * There are three pointers into the ring buffer: read, write, and
+ * trigger. The pointers are 8 bits each. Each pointer points to
+ * 32-byte "chunks" of data. The DMA engine moves 32 bytes at a time,
+ * so there is no finer-granularity control.
+ *
+ * In comm1, the hardware updates the write ptr, and software updates
+ * the read ptr. In comm2, it's the opposite: hardware updates the
+ * read ptr, and software updates the write ptr. I designate the
+ * hardware-updated ptr as the hwptr, and the software-updated ptr as
+ * the swptr.
+ *
+ * The trigger ptr and trigger mask are used to trigger interrupts.
+ * From the Lithium spec, section 5.6.8, revision of 12/15/1998:
+ *
+ * Trigger Mask Value
+ *
+ * A three bit wide field that represents a power of two mask
+ * that is used whenever the trigger pointer is compared to its
+ * respective read or write pointer. A value of zero here
+ * implies a mask of 0xFF and a value of seven implies a mask
+ * 0x01. This value can be used to sub-divide the ring buffer
+ * into pie sections so that interrupts monitor the progress of
+ * hardware from section to section.
+ *
+ * My interpretation of that is, whenever the hw ptr is updated, it is
+ * compared with the trigger ptr, and the result is masked by the
+ * trigger mask. (Actually, by the complement of the trigger mask.)
+ * If the result is zero, an interrupt is triggered. I.e., interrupt
+ * if ((hwptr & ~mask) == (trptr & ~mask)). The mask is formed from
+ * the trigger register value as mask = (1 << (8 - tmreg)) - 1.
+ *
+ * In yet different words, setting tmreg to 0 causes an interrupt after
+ * every 256 DMA chunks (8192 bytes) or once per traversal of the
+ * ring buffer. Setting it to 7 caues an interrupt every 2 DMA chunks
+ * (64 bytes) or 128 times per traversal of the ring buffer.
+ */
+
+/* Lithium register offsets and bit definitions */
+
+#define LI_HOST_CONTROLLER 0x000
+# define LI_HC_RESET 0x00008000
+# define LI_HC_LINK_ENABLE 0x00004000
+# define LI_HC_LINK_FAILURE 0x00000004
+# define LI_HC_LINK_CODEC 0x00000002
+# define LI_HC_LINK_READY 0x00000001
+
+#define LI_INTR_STATUS 0x010
+#define LI_INTR_MASK 0x014
+# define LI_INTR_LINK_ERR 0x00008000
+# define LI_INTR_COMM2_TRIG 0x00000008
+# define LI_INTR_COMM2_UNDERFLOW 0x00000004
+# define LI_INTR_COMM1_TRIG 0x00000002
+# define LI_INTR_COMM1_OVERFLOW 0x00000001
+
+#define LI_CODEC_COMMAND 0x018
+# define LI_CC_BUSY 0x00008000
+# define LI_CC_DIR 0x00000080
+# define LI_CC_DIR_RD LI_CC_DIR
+# define LI_CC_DIR_WR (!LI_CC_DIR)
+# define LI_CC_ADDR_MASK 0x0000007F
+
+#define LI_CODEC_DATA 0x01C
+
+#define LI_COMM1_BASE 0x100
+#define LI_COMM1_CTL 0x104
+# define LI_CCTL_RESET 0x80000000
+# define LI_CCTL_SIZE 0x70000000
+# define LI_CCTL_DMA_ENABLE 0x08000000
+# define LI_CCTL_TMASK 0x07000000 /* trigger mask */
+# define LI_CCTL_TPTR 0x00FF0000 /* trigger pointer */
+# define LI_CCTL_RPTR 0x0000FF00
+# define LI_CCTL_WPTR 0x000000FF
+#define LI_COMM1_CFG 0x108
+# define LI_CCFG_LOCK 0x00008000
+# define LI_CCFG_SLOT 0x00000070
+# define LI_CCFG_DIRECTION 0x00000008
+# define LI_CCFG_DIR_IN (!LI_CCFG_DIRECTION)
+# define LI_CCFG_DIR_OUT LI_CCFG_DIRECTION
+# define LI_CCFG_MODE 0x00000004
+# define LI_CCFG_MODE_MONO (!LI_CCFG_MODE)
+# define LI_CCFG_MODE_STEREO LI_CCFG_MODE
+# define LI_CCFG_FORMAT 0x00000003
+# define LI_CCFG_FMT_8BIT 0x00000000
+# define LI_CCFG_FMT_16BIT 0x00000001
+#define LI_COMM2_BASE 0x10C
+#define LI_COMM2_CTL 0x110
+ /* bit definitions are the same as LI_COMM1_CTL */
+#define LI_COMM2_CFG 0x114
+ /* bit definitions are the same as LI_COMM1_CFG */
+
+#define LI_UST_LOW 0x200 /* 64-bit Unadjusted System Time is */
+#define LI_UST_HIGH 0x204 /* microseconds since boot */
+
+#define LI_AUDIO1_UST 0x300 /* UST-MSC pairs */
+#define LI_AUDIO1_MSC 0x304 /* MSC (Media Stream Counter) */
+#define LI_AUDIO2_UST 0x308 /* counts samples actually */
+#define LI_AUDIO2_MSC 0x30C /* processed as of time UST */
+
+/*
+ * Lithium's DMA engine operates on chunks of 32 bytes. We call that
+ * a DMACHUNK.
+ */
+
+#define DMACHUNK_SHIFT 5
+#define DMACHUNK_SIZE (1 << DMACHUNK_SHIFT)
+#define BYTES_TO_CHUNKS(bytes) ((bytes) >> DMACHUNK_SHIFT)
+#define CHUNKS_TO_BYTES(chunks) ((chunks) << DMACHUNK_SHIFT)
+
+/*
+ * Two convenient macros to shift bitfields into/out of position.
+ *
+ * Observe that (mask & -mask) is (1 << low_set_bit_of(mask)).
+ * As long as mask is constant, we trust the compiler will change the
+ * multipy and divide into shifts.
+ */
+
+#define SHIFT_FIELD(val, mask) (((val) * ((mask) & -(mask))) & (mask))
+#define UNSHIFT_FIELD(val, mask) (((val) & (mask)) / ((mask) & -(mask)))
+
+/*
+ * dma_chan_desc is invariant information about a Lithium
+ * DMA channel. There are two instances, li_comm1 and li_comm2.
+ *
+ * Note that the CCTL register fields are write ptr and read ptr, but what
+ * we care about are which pointer is updated by software and which by
+ * hardware.
+ */
+
+typedef struct dma_chan_desc {
+ int basereg;
+ int cfgreg;
+ int ctlreg;
+ int hwptrreg;
+ int swptrreg;
+ int ustreg;
+ int mscreg;
+ unsigned long swptrmask;
+ int ad1843_slot;
+ int direction; /* LI_CCTL_DIR_IN/OUT */
+} dma_chan_desc_t;
+
+static const dma_chan_desc_t li_comm1 = {
+ LI_COMM1_BASE, /* base register offset */
+ LI_COMM1_CFG, /* config register offset */
+ LI_COMM1_CTL, /* control register offset */
+ LI_COMM1_CTL + 0, /* hw ptr reg offset (write ptr) */
+ LI_COMM1_CTL + 1, /* sw ptr reg offset (read ptr) */
+ LI_AUDIO1_UST, /* ust reg offset */
+ LI_AUDIO1_MSC, /* msc reg offset */
+ LI_CCTL_RPTR, /* sw ptr bitmask in ctlval */
+ 2, /* ad1843 serial slot */
+ LI_CCFG_DIR_IN /* direction */
+};
+
+static const dma_chan_desc_t li_comm2 = {
+ LI_COMM2_BASE, /* base register offset */
+ LI_COMM2_CFG, /* config register offset */
+ LI_COMM2_CTL, /* control register offset */
+ LI_COMM2_CTL + 1, /* hw ptr reg offset (read ptr) */
+ LI_COMM2_CTL + 0, /* sw ptr reg offset (writr ptr) */
+ LI_AUDIO2_UST, /* ust reg offset */
+ LI_AUDIO2_MSC, /* msc reg offset */
+ LI_CCTL_WPTR, /* sw ptr bitmask in ctlval */
+ 2, /* ad1843 serial slot */
+ LI_CCFG_DIR_OUT /* direction */
+};
+
+/*
+ * dma_chan is variable information about a Lithium DMA channel.
+ *
+ * The desc field points to invariant information.
+ * The lith field points to a lithium_t which is passed
+ * to li_read* and li_write* to access the registers.
+ * The *val fields shadow the lithium registers' contents.
+ */
+
+typedef struct dma_chan {
+ const dma_chan_desc_t *desc;
+ lithium_t *lith;
+ unsigned long baseval;
+ unsigned long cfgval;
+ unsigned long ctlval;
+} dma_chan_t;
+
+/*
+ * ustmsc is a UST/MSC pair (Unadjusted System Time/Media Stream Counter).
+ * UST is time in microseconds since the system booted, and MSC is a
+ * counter that increments with every audio sample.
+ */
+
+typedef struct ustmsc {
+ unsigned long long ust;
+ unsigned long msc;
+} ustmsc_t;
+
+/*
+ * li_ad1843_wait waits until lithium says the AD1843 register
+ * exchange is not busy. Returns 0 on success, -EBUSY on timeout.
+ *
+ * Locking: must be called with lithium_lock held.
+ */
+
+static int li_ad1843_wait(lithium_t *lith)
+{
+ unsigned long later = jiffies + 2;
+ while (li_readl(lith, LI_CODEC_COMMAND) & LI_CC_BUSY)
+ if (time_after_eq(jiffies, later))
+ return -EBUSY;
+ return 0;
+}
+
+/*
+ * li_read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
+ *
+ * Returns unsigned register value on success, -errno on failure.
+ */
+
+static int li_read_ad1843_reg(lithium_t *lith, int reg)
+{
+ int val;
+
+ ASSERT(!in_interrupt());
+ spin_lock(&lith->lock);
+ {
+ val = li_ad1843_wait(lith);
+ if (val == 0) {
+ li_writel(lith, LI_CODEC_COMMAND, LI_CC_DIR_RD | reg);
+ val = li_ad1843_wait(lith);
+ }
+ if (val == 0)
+ val = li_readl(lith, LI_CODEC_DATA);
+ }
+ spin_unlock(&lith->lock);
+
+ DBGXV("li_read_ad1843_reg(lith=0x%p, reg=%d) returns 0x%04x\n",
+ lith, reg, val);
+
+ return val;
+}
+
+/*
+ * li_write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
+ */
+
+static void li_write_ad1843_reg(lithium_t *lith, int reg, int newval)
+{
+ spin_lock(&lith->lock);
+ {
+ if (li_ad1843_wait(lith) == 0) {
+ li_writel(lith, LI_CODEC_DATA, newval);
+ li_writel(lith, LI_CODEC_COMMAND, LI_CC_DIR_WR | reg);
+ }
+ }
+ spin_unlock(&lith->lock);
+}
+
+/*
+ * li_setup_dma calculates all the register settings for DMA in a particular
+ * mode. It takes too many arguments.
+ */
+
+static void li_setup_dma(dma_chan_t *chan,
+ const dma_chan_desc_t *desc,
+ lithium_t *lith,
+ unsigned long buffer_paddr,
+ int bufshift,
+ int fragshift,
+ int channels,
+ int sampsize)
+{
+ unsigned long mode, format;
+ unsigned long size, tmask;
+
+ DBGEV("(chan=0x%p, desc=0x%p, lith=0x%p, buffer_paddr=0x%lx, "
+ "bufshift=%d, fragshift=%d, channels=%d, sampsize=%d)\n",
+ chan, desc, lith, buffer_paddr,
+ bufshift, fragshift, channels, sampsize);
+
+ /* Reset the channel first. */
+
+ li_writel(lith, desc->ctlreg, LI_CCTL_RESET);
+
+ ASSERT(channels == 1 || channels == 2);
+ if (channels == 2)
+ mode = LI_CCFG_MODE_STEREO;
+ else
+ mode = LI_CCFG_MODE_MONO;
+ ASSERT(sampsize == 1 || sampsize == 2);
+ if (sampsize == 2)
+ format = LI_CCFG_FMT_16BIT;
+ else
+ format = LI_CCFG_FMT_8BIT;
+ chan->desc = desc;
+ chan->lith = lith;
+
+ /*
+ * Lithium DMA address register takes a 40-bit physical
+ * address, right-shifted by 8 so it fits in 32 bits. Bit 37
+ * must be set -- it enables cache coherence.
+ */
+
+ ASSERT(!(buffer_paddr & 0xFF));
+ chan->baseval = (buffer_paddr >> 8) | 1 << (37 - 8);
+
+ chan->cfgval = (!LI_CCFG_LOCK |
+ SHIFT_FIELD(desc->ad1843_slot, LI_CCFG_SLOT) |
+ desc->direction |
+ mode |
+ format);
+
+ size = bufshift - 6;
+ tmask = 13 - fragshift; /* See Lithium DMA Notes above. */
+ ASSERT(size >= 2 && size <= 7);
+ ASSERT(tmask >= 1 && tmask <= 7);
+ chan->ctlval = (!LI_CCTL_RESET |
+ SHIFT_FIELD(size, LI_CCTL_SIZE) |
+ !LI_CCTL_DMA_ENABLE |
+ SHIFT_FIELD(tmask, LI_CCTL_TMASK) |
+ SHIFT_FIELD(0, LI_CCTL_TPTR));
+
+ DBGPV("basereg 0x%x = 0x%lx\n", desc->basereg, chan->baseval);
+ DBGPV("cfgreg 0x%x = 0x%lx\n", desc->cfgreg, chan->cfgval);
+ DBGPV("ctlreg 0x%x = 0x%lx\n", desc->ctlreg, chan->ctlval);
+
+ li_writel(lith, desc->basereg, chan->baseval);
+ li_writel(lith, desc->cfgreg, chan->cfgval);
+ li_writel(lith, desc->ctlreg, chan->ctlval);
+
+ DBGRV();
+}
+
+static void li_shutdown_dma(dma_chan_t *chan)
+{
+ lithium_t *lith = chan->lith;
+ void * lith1 = lith->page1;
+
+ DBGEV("(chan=0x%p)\n", chan);
+
+ chan->ctlval &= ~LI_CCTL_DMA_ENABLE;
+ DBGPV("ctlreg 0x%x = 0x%lx\n", chan->desc->ctlreg, chan->ctlval);
+ li_writel(lith, chan->desc->ctlreg, chan->ctlval);
+
+ /*
+ * Offset 0x500 on Lithium page 1 is an undocumented,
+ * unsupported register that holds the zero sample value.
+ * Lithium is supposed to output zero samples when DMA is
+ * inactive, and repeat the last sample when DMA underflows.
+ * But it has a bug, where, after underflow occurs, the zero
+ * sample is not reset.
+ *
+ * I expect this to break in a future rev of Lithium.
+ */
+
+ if (lith1 && chan->desc->direction == LI_CCFG_DIR_OUT)
+ * (volatile unsigned long *) (lith1 + 0x500) = 0;
+}
+
+/*
+ * li_activate_dma always starts dma at the beginning of the buffer.
+ *
+ * N.B., these may be called from interrupt.
+ */
+
+static __inline__ void li_activate_dma(dma_chan_t *chan)
+{
+ chan->ctlval |= LI_CCTL_DMA_ENABLE;
+ DBGPV("ctlval = 0x%lx\n", chan->ctlval);
+ li_writel(chan->lith, chan->desc->ctlreg, chan->ctlval);
+}
+
+static void li_deactivate_dma(dma_chan_t *chan)
+{
+ lithium_t *lith = chan->lith;
+ void * lith2 = lith->page2;
+
+ chan->ctlval &= ~(LI_CCTL_DMA_ENABLE | LI_CCTL_RPTR | LI_CCTL_WPTR);
+ DBGPV("ctlval = 0x%lx\n", chan->ctlval);
+ DBGPV("ctlreg 0x%x = 0x%lx\n", chan->desc->ctlreg, chan->ctlval);
+ li_writel(lith, chan->desc->ctlreg, chan->ctlval);
+
+ /*
+ * Offsets 0x98 and 0x9C on Lithium page 2 are undocumented,
+ * unsupported registers that are internal copies of the DMA
+ * read and write pointers. Because of a Lithium bug, these
+ * registers aren't zeroed correctly when DMA is shut off. So
+ * we whack them directly.
+ *
+ * I expect this to break in a future rev of Lithium.
+ */
+
+ if (lith2 && chan->desc->direction == LI_CCFG_DIR_OUT) {
+ * (volatile unsigned long *) (lith2 + 0x98) = 0;
+ * (volatile unsigned long *) (lith2 + 0x9C) = 0;
+ }
+}
+
+/*
+ * read/write the ring buffer pointers. These routines' arguments and results
+ * are byte offsets from the beginning of the ring buffer.
+ */
+
+static __inline__ int li_read_swptr(dma_chan_t *chan)
+{
+ const unsigned long mask = chan->desc->swptrmask;
+
+ return CHUNKS_TO_BYTES(UNSHIFT_FIELD(chan->ctlval, mask));
+}
+
+static __inline__ int li_read_hwptr(dma_chan_t *chan)
+{
+ return CHUNKS_TO_BYTES(li_readb(chan->lith, chan->desc->hwptrreg));
+}
+
+static __inline__ void li_write_swptr(dma_chan_t *chan, int val)
+{
+ const unsigned long mask = chan->desc->swptrmask;
+
+ ASSERT(!(val & ~CHUNKS_TO_BYTES(0xFF)));
+ val = BYTES_TO_CHUNKS(val);
+ chan->ctlval = (chan->ctlval & ~mask) | SHIFT_FIELD(val, mask);
+ li_writeb(chan->lith, chan->desc->swptrreg, val);
+}
+
+/* li_read_USTMSC() returns a UST/MSC pair for the given channel. */
+
+static void li_read_USTMSC(dma_chan_t *chan, ustmsc_t *ustmsc)
+{
+ lithium_t *lith = chan->lith;
+ const dma_chan_desc_t *desc = chan->desc;
+ unsigned long now_low, now_high0, now_high1, chan_ust;
+
+ spin_lock(&lith->lock);
+ {
+ /*
+ * retry until we do all five reads without the
+ * high word changing. (High word increments
+ * every 2^32 microseconds, i.e., not often)
+ */
+ do {
+ now_high0 = li_readl(lith, LI_UST_HIGH);
+ now_low = li_readl(lith, LI_UST_LOW);
+
+ /*
+ * Lithium guarantees these two reads will be
+ * atomic -- ust will not increment after msc
+ * is read.
+ */
+
+ ustmsc->msc = li_readl(lith, desc->mscreg);
+ chan_ust = li_readl(lith, desc->ustreg);
+
+ now_high1 = li_readl(lith, LI_UST_HIGH);
+ } while (now_high0 != now_high1);
+ }
+ spin_unlock(&lith->lock);
+ ustmsc->ust = ((unsigned long long) now_high0 << 32 | chan_ust);
+}
+
+static void li_enable_interrupts(lithium_t *lith, unsigned int mask)
+{
+ DBGEV("(lith=0x%p, mask=0x%x)\n", lith, mask);
+
+ /* clear any already-pending interrupts. */
+
+ li_writel(lith, LI_INTR_STATUS, mask);
+
+ /* enable the interrupts. */
+
+ mask |= li_readl(lith, LI_INTR_MASK);
+ li_writel(lith, LI_INTR_MASK, mask);
+}
+
+static void li_disable_interrupts(lithium_t *lith, unsigned int mask)
+{
+ unsigned int keepmask;
+
+ DBGEV("(lith=0x%p, mask=0x%x)\n", lith, mask);
+
+ /* disable the interrupts */
+
+ keepmask = li_readl(lith, LI_INTR_MASK) & ~mask;
+ li_writel(lith, LI_INTR_MASK, keepmask);
+
+ /* clear any pending interrupts. */
+
+ li_writel(lith, LI_INTR_STATUS, mask);
+}
+
+/* Get the interrupt status and clear all pending interrupts. */
+
+static unsigned int li_get_clear_intr_status(lithium_t *lith)
+{
+ unsigned int status;
+
+ status = li_readl(lith, LI_INTR_STATUS);
+ li_writel(lith, LI_INTR_STATUS, ~0);
+ return status & li_readl(lith, LI_INTR_MASK);
+}
+
+static int li_init(lithium_t *lith)
+{
+ /* 1. System power supplies stabilize. */
+
+ /* 2. Assert the ~RESET signal. */
+
+ li_writel(lith, LI_HOST_CONTROLLER, LI_HC_RESET);
+ udelay(1);
+
+ /* 3. Deassert the ~RESET signal and enter a wait period to allow
+ the AD1843 internal clocks and the external crystal oscillator
+ to stabilize. */
+
+ li_writel(lith, LI_HOST_CONTROLLER, LI_HC_LINK_ENABLE);
+ udelay(1);
+
+ return 0;
+}
+
+/*****************************************************************************/
+/* AD1843 access */
+
+/*
+ * AD1843 bitfield definitions. All are named as in the AD1843 data
+ * sheet, with ad1843_ prepended and individual bit numbers removed.
+ *
+ * E.g., bits LSS0 through LSS2 become ad1843_LSS.
+ *
+ * Only the bitfields we need are defined.
+ */
+
+typedef struct ad1843_bitfield {
+ char reg;
+ char lo_bit;
+ char nbits;
+} ad1843_bitfield_t;
+
+static const ad1843_bitfield_t
+ ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */
+ ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */
+ ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */
+ ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */
+ ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */
+ ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */
+ ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */
+ ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */
+ ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */
+ ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */
+ ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */
+ ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */
+ ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */
+ ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */
+ ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */
+ ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */
+ ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */
+ ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */
+ ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */
+ ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */
+ ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */
+ ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */
+ ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */
+ ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */
+ ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */
+ ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */
+ ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */
+ ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */
+ ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */
+ ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */
+ ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */
+ ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */
+ ad1843_C2C = { 20, 0, 16 }, /* Clock 1 Sample Rate Select */
+ ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */
+ ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */
+ ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */
+ ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */
+ ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */
+ ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */
+ ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */
+ ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */
+ ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */
+ ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */
+ ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */
+ ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */
+ ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */
+ ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */
+ ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */
+ ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */
+ ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */
+ ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */
+
+/*
+ * The various registers of the AD1843 use three different formats for
+ * specifying gain. The ad1843_gain structure parameterizes the
+ * formats.
+ */
+
+typedef struct ad1843_gain {
+
+ int negative; /* nonzero if gain is negative. */
+ const ad1843_bitfield_t *lfield;
+ const ad1843_bitfield_t *rfield;
+
+} ad1843_gain_t;
+
+static const ad1843_gain_t ad1843_gain_RECLEV
+ = { 0, &ad1843_LIG, &ad1843_RIG };
+static const ad1843_gain_t ad1843_gain_LINE
+ = { 1, &ad1843_LX1M, &ad1843_RX1M };
+static const ad1843_gain_t ad1843_gain_CD
+ = { 1, &ad1843_LX2M, &ad1843_RX2M };
+static const ad1843_gain_t ad1843_gain_MIC
+ = { 1, &ad1843_LMCM, &ad1843_RMCM };
+static const ad1843_gain_t ad1843_gain_PCM
+ = { 1, &ad1843_LDA1G, &ad1843_RDA1G };
+
+/* read the current value of an AD1843 bitfield. */
+
+static int ad1843_read_bits(lithium_t *lith, const ad1843_bitfield_t *field)
+{
+ int w = li_read_ad1843_reg(lith, field->reg);
+ int val = w >> field->lo_bit & ((1 << field->nbits) - 1);
+
+ DBGXV("ad1843_read_bits(lith=0x%p, field->{%d %d %d}) returns 0x%x\n",
+ lith, field->reg, field->lo_bit, field->nbits, val);
+
+ return val;
+}
+
+/*
+ * write a new value to an AD1843 bitfield and return the old value.
+ */
+
+static int ad1843_write_bits(lithium_t *lith,
+ const ad1843_bitfield_t *field,
+ int newval)
+{
+ int w = li_read_ad1843_reg(lith, field->reg);
+ int mask = ((1 << field->nbits) - 1) << field->lo_bit;
+ int oldval = (w & mask) >> field->lo_bit;
+ int newbits = (newval << field->lo_bit) & mask;
+ w = (w & ~mask) | newbits;
+ (void) li_write_ad1843_reg(lith, field->reg, w);
+
+ DBGXV("ad1843_write_bits(lith=0x%p, field->{%d %d %d}, val=0x%x) "
+ "returns 0x%x\n",
+ lith, field->reg, field->lo_bit, field->nbits, newval,
+ oldval);
+
+ return oldval;
+}
+
+/*
+ * ad1843_read_multi reads multiple bitfields from the same AD1843
+ * register. It uses a single read cycle to do it. (Reading the
+ * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20
+ * microseconds.)
+ *
+ * Called ike this.
+ *
+ * ad1843_read_multi(lith, nfields,
+ * &ad1843_FIELD1, &val1,
+ * &ad1843_FIELD2, &val2, ...);
+ */
+
+static void ad1843_read_multi(lithium_t *lith, int argcount, ...)
+{
+ va_list ap;
+ const ad1843_bitfield_t *fp;
+ int w = 0, mask, *value, reg = -1;
+
+ va_start(ap, argcount);
+ while (--argcount >= 0) {
+ fp = va_arg(ap, const ad1843_bitfield_t *);
+ value = va_arg(ap, int *);
+ if (reg == -1) {
+ reg = fp->reg;
+ w = li_read_ad1843_reg(lith, reg);
+ }
+ ASSERT(reg == fp->reg);
+ mask = (1 << fp->nbits) - 1;
+ *value = w >> fp->lo_bit & mask;
+ }
+ va_end(ap);
+}
+
+/*
+ * ad1843_write_multi stores multiple bitfields into the same AD1843
+ * register. It uses one read and one write cycle to do it.
+ *
+ * Called like this.
+ *
+ * ad1843_write_multi(lith, nfields,
+ * &ad1843_FIELD1, val1,
+ * &ad1843_FIELF2, val2, ...);
+ */
+
+static void ad1843_write_multi(lithium_t *lith, int argcount, ...)
+{
+ va_list ap;
+ int reg;
+ const ad1843_bitfield_t *fp;
+ int value;
+ int w, m, mask, bits;
+
+ mask = 0;
+ bits = 0;
+ reg = -1;
+
+ va_start(ap, argcount);
+ while (--argcount >= 0) {
+ fp = va_arg(ap, const ad1843_bitfield_t *);
+ value = va_arg(ap, int);
+ if (reg == -1)
+ reg = fp->reg;
+ ASSERT(fp->reg == reg);
+ m = ((1 << fp->nbits) - 1) << fp->lo_bit;
+ mask |= m;
+ bits |= (value << fp->lo_bit) & m;
+ }
+ va_end(ap);
+ ASSERT(!(bits & ~mask));
+ if (~mask & 0xFFFF)
+ w = li_read_ad1843_reg(lith, reg);
+ else
+ w = 0;
+ w = (w & ~mask) | bits;
+ (void) li_write_ad1843_reg(lith, reg, w);
+}
+
+/*
+ * ad1843_get_gain reads the specified register and extracts the gain value
+ * using the supplied gain type. It returns the gain in OSS format.
+ */
+
+static int ad1843_get_gain(lithium_t *lith, const ad1843_gain_t *gp)
+{
+ int lg, rg;
+ unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+ ad1843_read_multi(lith, 2, gp->lfield, &lg, gp->rfield, &rg);
+ if (gp->negative) {
+ lg = mask - lg;
+ rg = mask - rg;
+ }
+ lg = (lg * 100 + (mask >> 1)) / mask;
+ rg = (rg * 100 + (mask >> 1)) / mask;
+ return lg << 0 | rg << 8;
+}
+
+/*
+ * Set an audio channel's gain. Converts from OSS format to AD1843's
+ * format.
+ *
+ * Returns the new gain, which may be lower than the old gain.
+ */
+
+static int ad1843_set_gain(lithium_t *lith,
+ const ad1843_gain_t *gp,
+ int newval)
+{
+ unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+ int lg = newval >> 0 & 0xFF;
+ int rg = newval >> 8;
+ if (lg < 0 || lg > 100 || rg < 0 || rg > 100)
+ return -EINVAL;
+ lg = (lg * mask + (mask >> 1)) / 100;
+ rg = (rg * mask + (mask >> 1)) / 100;
+ if (gp->negative) {
+ lg = mask - lg;
+ rg = mask - rg;
+ }
+ ad1843_write_multi(lith, 2, gp->lfield, lg, gp->rfield, rg);
+ return ad1843_get_gain(lith, gp);
+}
+
+/* Returns the current recording source, in OSS format. */
+
+static int ad1843_get_recsrc(lithium_t *lith)
+{
+ int ls = ad1843_read_bits(lith, &ad1843_LSS);
+
+ switch (ls) {
+ case 1:
+ return SOUND_MASK_MIC;
+ case 2:
+ return SOUND_MASK_LINE;
+ case 3:
+ return SOUND_MASK_CD;
+ case 6:
+ return SOUND_MASK_PCM;
+ default:
+ ASSERT(0);
+ return -1;
+ }
+}
+
+/*
+ * Enable/disable digital resample mode in the AD1843.
+ *
+ * The AD1843 requires that ADL, ADR, DA1 and DA2 be powered down
+ * while switching modes. So we save DA1's state (DA2's state is not
+ * interesting), power them down, switch into/out of resample mode,
+ * power them up, and restore state.
+ *
+ * This will cause audible glitches if D/A or A/D is going on, so the
+ * driver disallows that (in mixer_write_ioctl()).
+ *
+ * The open question is, is this worth doing? I'm leaving it in,
+ * because it's written, but...
+ */
+
+static void ad1843_set_resample_mode(lithium_t *lith, int onoff)
+{
+ /* Save DA1 mute and gain (addr 9 is DA1 analog gain/attenuation) */
+ int save_da1 = li_read_ad1843_reg(lith, 9);
+
+ /* Power down A/D and D/A. */
+ ad1843_write_multi(lith, 4,
+ &ad1843_DA1EN, 0,
+ &ad1843_DA2EN, 0,
+ &ad1843_ADLEN, 0,
+ &ad1843_ADREN, 0);
+
+ /* Switch mode */
+ ASSERT(onoff == 0 || onoff == 1);
+ ad1843_write_bits(lith, &ad1843_DRSFLT, onoff);
+
+ /* Power up A/D and D/A. */
+ ad1843_write_multi(lith, 3,
+ &ad1843_DA1EN, 1,
+ &ad1843_ADLEN, 1,
+ &ad1843_ADREN, 1);
+
+ /* Restore DA1 mute and gain. */
+ li_write_ad1843_reg(lith, 9, save_da1);
+}
+
+/*
+ * Set recording source. Arg newsrc specifies an OSS channel mask.
+ *
+ * The complication is that when we switch into/out of loopback mode
+ * (i.e., src = SOUND_MASK_PCM), we change the AD1843 into/out of
+ * digital resampling mode.
+ *
+ * Returns newsrc on success, -errno on failure.
+ */
+
+static int ad1843_set_recsrc(lithium_t *lith, int newsrc)
+{
+ int bits;
+ int oldbits;
+
+ switch (newsrc) {
+ case SOUND_MASK_PCM:
+ bits = 6;
+ break;
+
+ case SOUND_MASK_MIC:
+ bits = 1;
+ break;
+
+ case SOUND_MASK_LINE:
+ bits = 2;
+ break;
+
+ case SOUND_MASK_CD:
+ bits = 3;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ oldbits = ad1843_read_bits(lith, &ad1843_LSS);
+ if (newsrc == SOUND_MASK_PCM && oldbits != 6) {
+ DBGP("enabling digital resample mode\n");
+ ad1843_set_resample_mode(lith, 1);
+ ad1843_write_multi(lith, 2,
+ &ad1843_DAADL, 2,
+ &ad1843_DAADR, 2);
+ } else if (newsrc != SOUND_MASK_PCM && oldbits == 6) {
+ DBGP("disabling digital resample mode\n");
+ ad1843_set_resample_mode(lith, 0);
+ ad1843_write_multi(lith, 2,
+ &ad1843_DAADL, 0,
+ &ad1843_DAADR, 0);
+ }
+ ad1843_write_multi(lith, 2, &ad1843_LSS, bits, &ad1843_RSS, bits);
+ return newsrc;
+}
+
+/*
+ * Return current output sources, in OSS format.
+ */
+
+static int ad1843_get_outsrc(lithium_t *lith)
+{
+ int pcm, line, mic, cd;
+
+ pcm = ad1843_read_bits(lith, &ad1843_LDA1GM) ? 0 : SOUND_MASK_PCM;
+ line = ad1843_read_bits(lith, &ad1843_LX1MM) ? 0 : SOUND_MASK_LINE;
+ cd = ad1843_read_bits(lith, &ad1843_LX2MM) ? 0 : SOUND_MASK_CD;
+ mic = ad1843_read_bits(lith, &ad1843_LMCMM) ? 0 : SOUND_MASK_MIC;
+
+ return pcm | line | cd | mic;
+}
+
+/*
+ * Set output sources. Arg is a mask of active sources in OSS format.
+ *
+ * Returns source mask on success, -errno on failure.
+ */
+
+static int ad1843_set_outsrc(lithium_t *lith, int mask)
+{
+ int pcm, line, mic, cd;
+
+ if (mask & ~(SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_CD | SOUND_MASK_MIC))
+ return -EINVAL;
+ pcm = (mask & SOUND_MASK_PCM) ? 0 : 1;
+ line = (mask & SOUND_MASK_LINE) ? 0 : 1;
+ mic = (mask & SOUND_MASK_MIC) ? 0 : 1;
+ cd = (mask & SOUND_MASK_CD) ? 0 : 1;
+
+ ad1843_write_multi(lith, 2, &ad1843_LDA1GM, pcm, &ad1843_RDA1GM, pcm);
+ ad1843_write_multi(lith, 2, &ad1843_LX1MM, line, &ad1843_RX1MM, line);
+ ad1843_write_multi(lith, 2, &ad1843_LX2MM, cd, &ad1843_RX2MM, cd);
+ ad1843_write_multi(lith, 2, &ad1843_LMCMM, mic, &ad1843_RMCMM, mic);
+
+ return mask;
+}
+
+/* Setup ad1843 for D/A conversion. */
+
+static void ad1843_setup_dac(lithium_t *lith,
+ int framerate,
+ int fmt,
+ int channels)
+{
+ int ad_fmt = 0, ad_mode = 0;
+
+ DBGEV("(lith=0x%p, framerate=%d, fmt=%d, channels=%d)\n",
+ lith, framerate, fmt, channels);
+
+ switch (fmt) {
+ case AFMT_S8: ad_fmt = 1; break;
+ case AFMT_U8: ad_fmt = 1; break;
+ case AFMT_S16_LE: ad_fmt = 1; break;
+ case AFMT_MU_LAW: ad_fmt = 2; break;
+ case AFMT_A_LAW: ad_fmt = 3; break;
+ default: ASSERT(0);
+ }
+
+ switch (channels) {
+ case 2: ad_mode = 0; break;
+ case 1: ad_mode = 1; break;
+ default: ASSERT(0);
+ }
+
+ DBGPV("ad_mode = %d, ad_fmt = %d\n", ad_mode, ad_fmt);
+ ASSERT(framerate >= 4000 && framerate <= 49000);
+ ad1843_write_bits(lith, &ad1843_C1C, framerate);
+ ad1843_write_multi(lith, 2,
+ &ad1843_DA1SM, ad_mode, &ad1843_DA1F, ad_fmt);
+}
+
+static void ad1843_shutdown_dac(lithium_t *lith)
+{
+ ad1843_write_bits(lith, &ad1843_DA1F, 1);
+}
+
+static void ad1843_setup_adc(lithium_t *lith, int framerate, int fmt, int channels)
+{
+ int da_fmt = 0;
+
+ DBGEV("(lith=0x%p, framerate=%d, fmt=%d, channels=%d)\n",
+ lith, framerate, fmt, channels);
+
+ switch (fmt) {
+ case AFMT_S8: da_fmt = 1; break;
+ case AFMT_U8: da_fmt = 1; break;
+ case AFMT_S16_LE: da_fmt = 1; break;
+ case AFMT_MU_LAW: da_fmt = 2; break;
+ case AFMT_A_LAW: da_fmt = 3; break;
+ default: ASSERT(0);
+ }
+
+ DBGPV("da_fmt = %d\n", da_fmt);
+ ASSERT(framerate >= 4000 && framerate <= 49000);
+ ad1843_write_bits(lith, &ad1843_C2C, framerate);
+ ad1843_write_multi(lith, 2,
+ &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt);
+}
+
+static void ad1843_shutdown_adc(lithium_t *lith)
+{
+ /* nothing to do */
+}
+
+/*
+ * Fully initialize the ad1843. As described in the AD1843 data
+ * sheet, section "START-UP SEQUENCE". The numbered comments are
+ * subsection headings from the data sheet. See the data sheet, pages
+ * 52-54, for more info.
+ *
+ * return 0 on success, -errno on failure. */
+
+static int __init ad1843_init(lithium_t *lith)
+{
+ unsigned long later;
+ int err;
+
+ err = li_init(lith);
+ if (err)
+ return err;
+
+ if (ad1843_read_bits(lith, &ad1843_INIT) != 0) {
+ printk(KERN_ERR "vwsnd sound: AD1843 won't initialize\n");
+ return -EIO;
+ }
+
+ ad1843_write_bits(lith, &ad1843_SCF, 1);
+
+ /* 4. Put the conversion resources into standby. */
+
+ ad1843_write_bits(lith, &ad1843_PDNI, 0);
+ later = jiffies + HZ / 2; /* roughly half a second */
+ DBGDO(shut_up++);
+ while (ad1843_read_bits(lith, &ad1843_PDNO)) {
+ if (time_after(jiffies, later)) {
+ printk(KERN_ERR
+ "vwsnd audio: AD1843 won't power up\n");
+ return -EIO;
+ }
+ schedule();
+ }
+ DBGDO(shut_up--);
+
+ /* 5. Power up the clock generators and enable clock output pins. */
+
+ ad1843_write_multi(lith, 2, &ad1843_C1EN, 1, &ad1843_C2EN, 1);
+
+ /* 6. Configure conversion resources while they are in standby. */
+
+ /* DAC1 uses clock 1 as source, ADC uses clock 2. Always. */
+
+ ad1843_write_multi(lith, 3,
+ &ad1843_DA1C, 1,
+ &ad1843_ADLC, 2,
+ &ad1843_ADRC, 2);
+
+ /* 7. Enable conversion resources. */
+
+ ad1843_write_bits(lith, &ad1843_ADTLK, 1);
+ ad1843_write_multi(lith, 5,
+ &ad1843_ANAEN, 1,
+ &ad1843_AAMEN, 1,
+ &ad1843_DA1EN, 1,
+ &ad1843_ADLEN, 1,
+ &ad1843_ADREN, 1);
+
+ /* 8. Configure conversion resources while they are enabled. */
+
+ ad1843_write_bits(lith, &ad1843_DA1C, 1);
+
+ /* Unmute all channels. */
+
+ ad1843_set_outsrc(lith,
+ (SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_MIC | SOUND_MASK_CD));
+ ad1843_write_multi(lith, 2, &ad1843_LDA1AM, 0, &ad1843_RDA1AM, 0);
+
+ /* Set default recording source to Line In and set
+ * mic gain to +20 dB.
+ */
+
+ ad1843_set_recsrc(lith, SOUND_MASK_LINE);
+ ad1843_write_multi(lith, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1);
+
+ /* Set Speaker Out level to +/- 4V and unmute it. */
+
+ ad1843_write_multi(lith, 2, &ad1843_HPOS, 1, &ad1843_HPOM, 0);
+
+ return 0;
+}
+
+/*****************************************************************************/
+/* PCM I/O */
+
+#define READ_INTR_MASK (LI_INTR_COMM1_TRIG | LI_INTR_COMM1_OVERFLOW)
+#define WRITE_INTR_MASK (LI_INTR_COMM2_TRIG | LI_INTR_COMM2_UNDERFLOW)
+
+typedef enum vwsnd_port_swstate { /* software state */
+ SW_OFF,
+ SW_INITIAL,
+ SW_RUN,
+ SW_DRAIN,
+} vwsnd_port_swstate_t;
+
+typedef enum vwsnd_port_hwstate { /* hardware state */
+ HW_STOPPED,
+ HW_RUNNING,
+} vwsnd_port_hwstate_t;
+
+/*
+ * These flags are read by ISR, but only written at baseline.
+ */
+
+typedef enum vwsnd_port_flags {
+ DISABLED = 1 << 0,
+ ERFLOWN = 1 << 1, /* overflown or underflown */
+ HW_BUSY = 1 << 2,
+} vwsnd_port_flags_t;
+
+/*
+ * vwsnd_port is the per-port data structure. Each device has two
+ * ports, one for input and one for output.
+ *
+ * Locking:
+ *
+ * port->lock protects: hwstate, flags, swb_[iu]_avail.
+ *
+ * devc->io_mutex protects: swstate, sw_*, swb_[iu]_idx.
+ *
+ * everything else is only written by open/release or
+ * pcm_{setup,shutdown}(), which are serialized by a
+ * combination of devc->open_mutex and devc->io_mutex.
+ */
+
+typedef struct vwsnd_port {
+
+ spinlock_t lock;
+ wait_queue_head_t queue;
+ vwsnd_port_swstate_t swstate;
+ vwsnd_port_hwstate_t hwstate;
+ vwsnd_port_flags_t flags;
+
+ int sw_channels;
+ int sw_samplefmt;
+ int sw_framerate;
+ int sample_size;
+ int frame_size;
+ unsigned int zero_word; /* zero for the sample format */
+
+ int sw_fragshift;
+ int sw_fragcount;
+ int sw_subdivshift;
+
+ unsigned int hw_fragshift;
+ unsigned int hw_fragsize;
+ unsigned int hw_fragcount;
+
+ int hwbuf_size;
+ unsigned long hwbuf_paddr;
+ unsigned long hwbuf_vaddr;
+ void * hwbuf; /* hwbuf == hwbuf_vaddr */
+ int hwbuf_max; /* max bytes to preload */
+
+ void * swbuf;
+ unsigned int swbuf_size; /* size in bytes */
+ unsigned int swb_u_idx; /* index of next user byte */
+ unsigned int swb_i_idx; /* index of next intr byte */
+ unsigned int swb_u_avail; /* # bytes avail to user */
+ unsigned int swb_i_avail; /* # bytes avail to intr */
+
+ dma_chan_t chan;
+
+ /* Accounting */
+
+ int byte_count;
+ int frag_count;
+ int MSC_offset;
+
+} vwsnd_port_t;
+
+/* vwsnd_dev is the per-device data structure. */
+
+typedef struct vwsnd_dev {
+ struct vwsnd_dev *next_dev;
+ int audio_minor; /* minor number of audio device */
+ int mixer_minor; /* minor number of mixer device */
+
+ struct mutex open_mutex;
+ struct mutex io_mutex;
+ struct mutex mix_mutex;
+ fmode_t open_mode;
+ wait_queue_head_t open_wait;
+
+ lithium_t lith;
+
+ vwsnd_port_t rport;
+ vwsnd_port_t wport;
+} vwsnd_dev_t;
+
+static vwsnd_dev_t *vwsnd_dev_list; /* linked list of all devices */
+
+static atomic_t vwsnd_use_count = ATOMIC_INIT(0);
+
+# define INC_USE_COUNT (atomic_inc(&vwsnd_use_count))
+# define DEC_USE_COUNT (atomic_dec(&vwsnd_use_count))
+# define IN_USE (atomic_read(&vwsnd_use_count) != 0)
+
+/*
+ * Lithium can only DMA multiples of 32 bytes. Its DMA buffer may
+ * be up to 8 Kb. This driver always uses 8 Kb.
+ *
+ * Memory bug workaround -- I'm not sure what's going on here, but
+ * somehow pcm_copy_out() was triggering segv's going on to the next
+ * page of the hw buffer. So, I make the hw buffer one size bigger
+ * than we actually use. That way, the following page is allocated
+ * and mapped, and no error. I suspect that something is broken
+ * in Cobalt, but haven't really investigated. HBO is the actual
+ * size of the buffer, and HWBUF_ORDER is what we allocate.
+ */
+
+#define HWBUF_SHIFT 13
+#define HWBUF_SIZE (1 << HWBUF_SHIFT)
+# define HBO (HWBUF_SHIFT > PAGE_SHIFT ? HWBUF_SHIFT - PAGE_SHIFT : 0)
+# define HWBUF_ORDER (HBO + 1) /* next size bigger */
+#define MIN_SPEED 4000
+#define MAX_SPEED 49000
+
+#define MIN_FRAGSHIFT (DMACHUNK_SHIFT + 1)
+#define MAX_FRAGSHIFT (PAGE_SHIFT)
+#define MIN_FRAGSIZE (1 << MIN_FRAGSHIFT)
+#define MAX_FRAGSIZE (1 << MAX_FRAGSHIFT)
+#define MIN_FRAGCOUNT(fragsize) 3
+#define MAX_FRAGCOUNT(fragsize) (32 * PAGE_SIZE / (fragsize))
+#define DEFAULT_FRAGSHIFT 12
+#define DEFAULT_FRAGCOUNT 16
+#define DEFAULT_SUBDIVSHIFT 0
+
+/*
+ * The software buffer (swbuf) is a ring buffer shared between user
+ * level and interrupt level. Each level owns some of the bytes in
+ * the buffer, and may give bytes away by calling swb_inc_{u,i}().
+ * User level calls _u for user, and interrupt level calls _i for
+ * interrupt.
+ *
+ * port->swb_{u,i}_avail is the number of bytes available to that level.
+ *
+ * port->swb_{u,i}_idx is the index of the first available byte in the
+ * buffer.
+ *
+ * Each level calls swb_inc_{u,i}() to atomically increment its index,
+ * recalculate the number of bytes available for both sides, and
+ * return the number of bytes available. Since each side can only
+ * give away bytes, the other side can only increase the number of
+ * bytes available to this side. Each side updates its own index
+ * variable, swb_{u,i}_idx, so no lock is needed to read it.
+ *
+ * To query the number of bytes available, call swb_inc_{u,i} with an
+ * increment of zero.
+ */
+
+static __inline__ unsigned int __swb_inc_u(vwsnd_port_t *port, int inc)
+{
+ if (inc) {
+ port->swb_u_idx += inc;
+ port->swb_u_idx %= port->swbuf_size;
+ port->swb_u_avail -= inc;
+ port->swb_i_avail += inc;
+ }
+ return port->swb_u_avail;
+}
+
+static __inline__ unsigned int swb_inc_u(vwsnd_port_t *port, int inc)
+{
+ unsigned long flags;
+ unsigned int ret;
+
+ spin_lock_irqsave(&port->lock, flags);
+ {
+ ret = __swb_inc_u(port, inc);
+ }
+ spin_unlock_irqrestore(&port->lock, flags);
+ return ret;
+}
+
+static __inline__ unsigned int __swb_inc_i(vwsnd_port_t *port, int inc)
+{
+ if (inc) {
+ port->swb_i_idx += inc;
+ port->swb_i_idx %= port->swbuf_size;
+ port->swb_i_avail -= inc;
+ port->swb_u_avail += inc;
+ }
+ return port->swb_i_avail;
+}
+
+static __inline__ unsigned int swb_inc_i(vwsnd_port_t *port, int inc)
+{
+ unsigned long flags;
+ unsigned int ret;
+
+ spin_lock_irqsave(&port->lock, flags);
+ {
+ ret = __swb_inc_i(port, inc);
+ }
+ spin_unlock_irqrestore(&port->lock, flags);
+ return ret;
+}
+
+/*
+ * pcm_setup - this routine initializes all port state after
+ * mode-setting ioctls have been done, but before the first I/O is
+ * done.
+ *
+ * Locking: called with devc->io_mutex held.
+ *
+ * Returns 0 on success, -errno on failure.
+ */
+
+static int pcm_setup(vwsnd_dev_t *devc,
+ vwsnd_port_t *rport,
+ vwsnd_port_t *wport)
+{
+ vwsnd_port_t *aport = rport ? rport : wport;
+ int sample_size;
+ unsigned int zero_word;
+
+ DBGEV("(devc=0x%p, rport=0x%p, wport=0x%p)\n", devc, rport, wport);
+
+ ASSERT(aport != NULL);
+ if (aport->swbuf != NULL)
+ return 0;
+ switch (aport->sw_samplefmt) {
+ case AFMT_MU_LAW:
+ sample_size = 1;
+ zero_word = 0xFFFFFFFF ^ 0x80808080;
+ break;
+
+ case AFMT_A_LAW:
+ sample_size = 1;
+ zero_word = 0xD5D5D5D5 ^ 0x80808080;
+ break;
+
+ case AFMT_U8:
+ sample_size = 1;
+ zero_word = 0x80808080;
+ break;
+
+ case AFMT_S8:
+ sample_size = 1;
+ zero_word = 0x00000000;
+ break;
+
+ case AFMT_S16_LE:
+ sample_size = 2;
+ zero_word = 0x00000000;
+ break;
+
+ default:
+ sample_size = 0; /* prevent compiler warning */
+ zero_word = 0;
+ ASSERT(0);
+ }
+ aport->sample_size = sample_size;
+ aport->zero_word = zero_word;
+ aport->frame_size = aport->sw_channels * aport->sample_size;
+ aport->hw_fragshift = aport->sw_fragshift - aport->sw_subdivshift;
+ aport->hw_fragsize = 1 << aport->hw_fragshift;
+ aport->hw_fragcount = aport->sw_fragcount << aport->sw_subdivshift;
+ ASSERT(aport->hw_fragsize >= MIN_FRAGSIZE);
+ ASSERT(aport->hw_fragsize <= MAX_FRAGSIZE);
+ ASSERT(aport->hw_fragcount >= MIN_FRAGCOUNT(aport->hw_fragsize));
+ ASSERT(aport->hw_fragcount <= MAX_FRAGCOUNT(aport->hw_fragsize));
+ if (rport) {
+ int hwfrags, swfrags;
+ rport->hwbuf_max = aport->hwbuf_size - DMACHUNK_SIZE;
+ hwfrags = rport->hwbuf_max >> aport->hw_fragshift;
+ swfrags = aport->hw_fragcount - hwfrags;
+ if (swfrags < 2)
+ swfrags = 2;
+ rport->swbuf_size = swfrags * aport->hw_fragsize;
+ DBGPV("hwfrags = %d, swfrags = %d\n", hwfrags, swfrags);
+ DBGPV("read hwbuf_max = %d, swbuf_size = %d\n",
+ rport->hwbuf_max, rport->swbuf_size);
+ }
+ if (wport) {
+ int hwfrags, swfrags;
+ int total_bytes = aport->hw_fragcount * aport->hw_fragsize;
+ wport->hwbuf_max = aport->hwbuf_size - DMACHUNK_SIZE;
+ if (wport->hwbuf_max > total_bytes)
+ wport->hwbuf_max = total_bytes;
+ hwfrags = wport->hwbuf_max >> aport->hw_fragshift;
+ DBGPV("hwfrags = %d\n", hwfrags);
+ swfrags = aport->hw_fragcount - hwfrags;
+ if (swfrags < 2)
+ swfrags = 2;
+ wport->swbuf_size = swfrags * aport->hw_fragsize;
+ DBGPV("hwfrags = %d, swfrags = %d\n", hwfrags, swfrags);
+ DBGPV("write hwbuf_max = %d, swbuf_size = %d\n",
+ wport->hwbuf_max, wport->swbuf_size);
+ }
+
+ aport->swb_u_idx = 0;
+ aport->swb_i_idx = 0;
+ aport->byte_count = 0;
+
+ /*
+ * Is this a Cobalt bug? We need to make this buffer extend
+ * one page further than we actually use -- somehow memcpy
+ * causes an exceptoin otherwise. I suspect there's a bug in
+ * Cobalt (or somewhere) where it's generating a fault on a
+ * speculative load or something. Obviously, I haven't taken
+ * the time to track it down.
+ */
+
+ aport->swbuf = vmalloc(aport->swbuf_size + PAGE_SIZE);
+ if (!aport->swbuf)
+ return -ENOMEM;
+ if (rport && wport) {
+ ASSERT(aport == rport);
+ ASSERT(wport->swbuf == NULL);
+ /* One extra page - see comment above. */
+ wport->swbuf = vmalloc(aport->swbuf_size + PAGE_SIZE);
+ if (!wport->swbuf) {
+ vfree(aport->swbuf);
+ aport->swbuf = NULL;
+ return -ENOMEM;
+ }
+ wport->sample_size = rport->sample_size;
+ wport->zero_word = rport->zero_word;
+ wport->frame_size = rport->frame_size;
+ wport->hw_fragshift = rport->hw_fragshift;
+ wport->hw_fragsize = rport->hw_fragsize;
+ wport->hw_fragcount = rport->hw_fragcount;
+ wport->swbuf_size = rport->swbuf_size;
+ wport->hwbuf_max = rport->hwbuf_max;
+ wport->swb_u_idx = rport->swb_u_idx;
+ wport->swb_i_idx = rport->swb_i_idx;
+ wport->byte_count = rport->byte_count;
+ }
+ if (rport) {
+ rport->swb_u_avail = 0;
+ rport->swb_i_avail = rport->swbuf_size;
+ rport->swstate = SW_RUN;
+ li_setup_dma(&rport->chan,
+ &li_comm1,
+ &devc->lith,
+ rport->hwbuf_paddr,
+ HWBUF_SHIFT,
+ rport->hw_fragshift,
+ rport->sw_channels,
+ rport->sample_size);
+ ad1843_setup_adc(&devc->lith,
+ rport->sw_framerate,
+ rport->sw_samplefmt,
+ rport->sw_channels);
+ li_enable_interrupts(&devc->lith, READ_INTR_MASK);
+ if (!(rport->flags & DISABLED)) {
+ ustmsc_t ustmsc;
+ rport->hwstate = HW_RUNNING;
+ li_activate_dma(&rport->chan);
+ li_read_USTMSC(&rport->chan, &ustmsc);
+ rport->MSC_offset = ustmsc.msc;
+ }
+ }
+ if (wport) {
+ if (wport->hwbuf_max > wport->swbuf_size)
+ wport->hwbuf_max = wport->swbuf_size;
+ wport->flags &= ~ERFLOWN;
+ wport->swb_u_avail = wport->swbuf_size;
+ wport->swb_i_avail = 0;
+ wport->swstate = SW_RUN;
+ li_setup_dma(&wport->chan,
+ &li_comm2,
+ &devc->lith,
+ wport->hwbuf_paddr,
+ HWBUF_SHIFT,
+ wport->hw_fragshift,
+ wport->sw_channels,
+ wport->sample_size);
+ ad1843_setup_dac(&devc->lith,
+ wport->sw_framerate,
+ wport->sw_samplefmt,
+ wport->sw_channels);
+ li_enable_interrupts(&devc->lith, WRITE_INTR_MASK);
+ }
+ DBGRV();
+ return 0;
+}
+
+/*
+ * pcm_shutdown_port - shut down one port (direction) for PCM I/O.
+ * Only called from pcm_shutdown.
+ */
+
+static void pcm_shutdown_port(vwsnd_dev_t *devc,
+ vwsnd_port_t *aport,
+ unsigned int mask)
+{
+ unsigned long flags;
+ vwsnd_port_hwstate_t hwstate;
+ DECLARE_WAITQUEUE(wait, current);
+
+ aport->swstate = SW_INITIAL;
+ add_wait_queue(&aport->queue, &wait);
+ while (1) {
+ set_current_state(TASK_UNINTERRUPTIBLE);
+ spin_lock_irqsave(&aport->lock, flags);
+ {
+ hwstate = aport->hwstate;
+ }
+ spin_unlock_irqrestore(&aport->lock, flags);
+ if (hwstate == HW_STOPPED)
+ break;
+ schedule();
+ }
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&aport->queue, &wait);
+ li_disable_interrupts(&devc->lith, mask);
+ if (aport == &devc->rport)
+ ad1843_shutdown_adc(&devc->lith);
+ else /* aport == &devc->wport) */
+ ad1843_shutdown_dac(&devc->lith);
+ li_shutdown_dma(&aport->chan);
+ vfree(aport->swbuf);
+ aport->swbuf = NULL;
+ aport->byte_count = 0;
+}
+
+/*
+ * pcm_shutdown undoes what pcm_setup did.
+ * Also sets the ports' swstate to newstate.
+ */
+
+static void pcm_shutdown(vwsnd_dev_t *devc,
+ vwsnd_port_t *rport,
+ vwsnd_port_t *wport)
+{
+ DBGEV("(devc=0x%p, rport=0x%p, wport=0x%p)\n", devc, rport, wport);
+
+ if (rport && rport->swbuf) {
+ DBGPV("shutting down rport\n");
+ pcm_shutdown_port(devc, rport, READ_INTR_MASK);
+ }
+ if (wport && wport->swbuf) {
+ DBGPV("shutting down wport\n");
+ pcm_shutdown_port(devc, wport, WRITE_INTR_MASK);
+ }
+ DBGRV();
+}
+
+static void pcm_copy_in(vwsnd_port_t *rport, int swidx, int hwidx, int nb)
+{
+ char *src = rport->hwbuf + hwidx;
+ char *dst = rport->swbuf + swidx;
+ int fmt = rport->sw_samplefmt;
+
+ DBGPV("swidx = %d, hwidx = %d\n", swidx, hwidx);
+ ASSERT(rport->hwbuf != NULL);
+ ASSERT(rport->swbuf != NULL);
+ ASSERT(nb > 0 && (nb % 32) == 0);
+ ASSERT(swidx % 32 == 0 && hwidx % 32 == 0);
+ ASSERT(swidx >= 0 && swidx + nb <= rport->swbuf_size);
+ ASSERT(hwidx >= 0 && hwidx + nb <= rport->hwbuf_size);
+
+ if (fmt == AFMT_MU_LAW || fmt == AFMT_A_LAW || fmt == AFMT_S8) {
+
+ /* See Sample Format Notes above. */
+
+ char *end = src + nb;
+ while (src < end)
+ *dst++ = *src++ ^ 0x80;
+ } else
+ memcpy(dst, src, nb);
+}
+
+static void pcm_copy_out(vwsnd_port_t *wport, int swidx, int hwidx, int nb)
+{
+ char *src = wport->swbuf + swidx;
+ char *dst = wport->hwbuf + hwidx;
+ int fmt = wport->sw_samplefmt;
+
+ ASSERT(nb > 0 && (nb % 32) == 0);
+ ASSERT(wport->hwbuf != NULL);
+ ASSERT(wport->swbuf != NULL);
+ ASSERT(swidx % 32 == 0 && hwidx % 32 == 0);
+ ASSERT(swidx >= 0 && swidx + nb <= wport->swbuf_size);
+ ASSERT(hwidx >= 0 && hwidx + nb <= wport->hwbuf_size);
+ if (fmt == AFMT_MU_LAW || fmt == AFMT_A_LAW || fmt == AFMT_S8) {
+
+ /* See Sample Format Notes above. */
+
+ char *end = src + nb;
+ while (src < end)
+ *dst++ = *src++ ^ 0x80;
+ } else
+ memcpy(dst, src, nb);
+}
+
+/*
+ * pcm_output() is called both from baselevel and from interrupt level.
+ * This is where audio frames are copied into the hardware-accessible
+ * ring buffer.
+ *
+ * Locking note: The part of this routine that figures out what to do
+ * holds wport->lock. The longer part releases wport->lock, but sets
+ * wport->flags & HW_BUSY. Afterward, it reacquires wport->lock, and
+ * checks for more work to do.
+ *
+ * If another thread calls pcm_output() while HW_BUSY is set, it
+ * returns immediately, knowing that the thread that set HW_BUSY will
+ * look for more work to do before returning.
+ *
+ * This has the advantage that port->lock is held for several short
+ * periods instead of one long period. Also, when pcm_output is
+ * called from base level, it reenables interrupts.
+ */
+
+static void pcm_output(vwsnd_dev_t *devc, int erflown, int nb)
+{
+ vwsnd_port_t *wport = &devc->wport;
+ const int hwmax = wport->hwbuf_max;
+ const int hwsize = wport->hwbuf_size;
+ const int swsize = wport->swbuf_size;
+ const int fragsize = wport->hw_fragsize;
+ unsigned long iflags;
+
+ DBGEV("(devc=0x%p, erflown=%d, nb=%d)\n", devc, erflown, nb);
+ spin_lock_irqsave(&wport->lock, iflags);
+ if (erflown)
+ wport->flags |= ERFLOWN;
+ (void) __swb_inc_u(wport, nb);
+ if (wport->flags & HW_BUSY) {
+ spin_unlock_irqrestore(&wport->lock, iflags);
+ DBGPV("returning: HW BUSY\n");
+ return;
+ }
+ if (wport->flags & DISABLED) {
+ spin_unlock_irqrestore(&wport->lock, iflags);
+ DBGPV("returning: DISABLED\n");
+ return;
+ }
+ wport->flags |= HW_BUSY;
+ while (1) {
+ int swptr, hwptr, hw_avail, sw_avail, swidx;
+ vwsnd_port_hwstate_t hwstate = wport->hwstate;
+ vwsnd_port_swstate_t swstate = wport->swstate;
+ int hw_unavail;
+ ustmsc_t ustmsc;
+
+ hwptr = li_read_hwptr(&wport->chan);
+ swptr = li_read_swptr(&wport->chan);
+ hw_unavail = (swptr - hwptr + hwsize) % hwsize;
+ hw_avail = (hwmax - hw_unavail) & -fragsize;
+ sw_avail = wport->swb_i_avail & -fragsize;
+ if (sw_avail && swstate == SW_RUN) {
+ if (wport->flags & ERFLOWN) {
+ wport->flags &= ~ERFLOWN;
+ }
+ } else if (swstate == SW_INITIAL ||
+ swstate == SW_OFF ||
+ (swstate == SW_DRAIN &&
+ !sw_avail &&
+ (wport->flags & ERFLOWN))) {
+ DBGP("stopping. hwstate = %d\n", hwstate);
+ if (hwstate != HW_STOPPED) {
+ li_deactivate_dma(&wport->chan);
+ wport->hwstate = HW_STOPPED;
+ }
+ wake_up(&wport->queue);
+ break;
+ }
+ if (!sw_avail || !hw_avail)
+ break;
+ spin_unlock_irqrestore(&wport->lock, iflags);
+
+ /*
+ * We gave up the port lock, but we have the HW_BUSY flag.
+ * Proceed without accessing any nonlocal state.
+ * Do not exit the loop -- must check for more work.
+ */
+
+ swidx = wport->swb_i_idx;
+ nb = hw_avail;
+ if (nb > sw_avail)
+ nb = sw_avail;
+ if (nb > hwsize - swptr)
+ nb = hwsize - swptr; /* don't overflow hwbuf */
+ if (nb > swsize - swidx)
+ nb = swsize - swidx; /* don't overflow swbuf */
+ ASSERT(nb > 0);
+ if (nb % fragsize) {
+ DBGP("nb = %d, fragsize = %d\n", nb, fragsize);
+ DBGP("hw_avail = %d\n", hw_avail);
+ DBGP("sw_avail = %d\n", sw_avail);
+ DBGP("hwsize = %d, swptr = %d\n", hwsize, swptr);
+ DBGP("swsize = %d, swidx = %d\n", swsize, swidx);
+ }
+ ASSERT(!(nb % fragsize));
+ DBGPV("copying swb[%d..%d] to hwb[%d..%d]\n",
+ swidx, swidx + nb, swptr, swptr + nb);
+ pcm_copy_out(wport, swidx, swptr, nb);
+ li_write_swptr(&wport->chan, (swptr + nb) % hwsize);
+ spin_lock_irqsave(&wport->lock, iflags);
+ if (hwstate == HW_STOPPED) {
+ DBGPV("starting\n");
+ li_activate_dma(&wport->chan);
+ wport->hwstate = HW_RUNNING;
+ li_read_USTMSC(&wport->chan, &ustmsc);
+ ASSERT(wport->byte_count % wport->frame_size == 0);
+ wport->MSC_offset = ustmsc.msc - wport->byte_count / wport->frame_size;
+ }
+ __swb_inc_i(wport, nb);
+ wport->byte_count += nb;
+ wport->frag_count += nb / fragsize;
+ ASSERT(nb % fragsize == 0);
+ wake_up(&wport->queue);
+ }
+ wport->flags &= ~HW_BUSY;
+ spin_unlock_irqrestore(&wport->lock, iflags);
+ DBGRV();
+}
+
+/*
+ * pcm_input() is called both from baselevel and from interrupt level.
+ * This is where audio frames are copied out of the hardware-accessible
+ * ring buffer.
+ *
+ * Locking note: The part of this routine that figures out what to do
+ * holds rport->lock. The longer part releases rport->lock, but sets
+ * rport->flags & HW_BUSY. Afterward, it reacquires rport->lock, and
+ * checks for more work to do.
+ *
+ * If another thread calls pcm_input() while HW_BUSY is set, it
+ * returns immediately, knowing that the thread that set HW_BUSY will
+ * look for more work to do before returning.
+ *
+ * This has the advantage that port->lock is held for several short
+ * periods instead of one long period. Also, when pcm_input is
+ * called from base level, it reenables interrupts.
+ */
+
+static void pcm_input(vwsnd_dev_t *devc, int erflown, int nb)
+{
+ vwsnd_port_t *rport = &devc->rport;
+ const int hwmax = rport->hwbuf_max;
+ const int hwsize = rport->hwbuf_size;
+ const int swsize = rport->swbuf_size;
+ const int fragsize = rport->hw_fragsize;
+ unsigned long iflags;
+
+ DBGEV("(devc=0x%p, erflown=%d, nb=%d)\n", devc, erflown, nb);
+
+ spin_lock_irqsave(&rport->lock, iflags);
+ if (erflown)
+ rport->flags |= ERFLOWN;
+ (void) __swb_inc_u(rport, nb);
+ if (rport->flags & HW_BUSY || !rport->swbuf) {
+ spin_unlock_irqrestore(&rport->lock, iflags);
+ DBGPV("returning: HW BUSY or !swbuf\n");
+ return;
+ }
+ if (rport->flags & DISABLED) {
+ spin_unlock_irqrestore(&rport->lock, iflags);
+ DBGPV("returning: DISABLED\n");
+ return;
+ }
+ rport->flags |= HW_BUSY;
+ while (1) {
+ int swptr, hwptr, hw_avail, sw_avail, swidx;
+ vwsnd_port_hwstate_t hwstate = rport->hwstate;
+ vwsnd_port_swstate_t swstate = rport->swstate;
+
+ hwptr = li_read_hwptr(&rport->chan);
+ swptr = li_read_swptr(&rport->chan);
+ hw_avail = (hwptr - swptr + hwsize) % hwsize & -fragsize;
+ if (hw_avail > hwmax)
+ hw_avail = hwmax;
+ sw_avail = rport->swb_i_avail & -fragsize;
+ if (swstate != SW_RUN) {
+ DBGP("stopping. hwstate = %d\n", hwstate);
+ if (hwstate != HW_STOPPED) {
+ li_deactivate_dma(&rport->chan);
+ rport->hwstate = HW_STOPPED;
+ }
+ wake_up(&rport->queue);
+ break;
+ }
+ if (!sw_avail || !hw_avail)
+ break;
+ spin_unlock_irqrestore(&rport->lock, iflags);
+
+ /*
+ * We gave up the port lock, but we have the HW_BUSY flag.
+ * Proceed without accessing any nonlocal state.
+ * Do not exit the loop -- must check for more work.
+ */
+
+ swidx = rport->swb_i_idx;
+ nb = hw_avail;
+ if (nb > sw_avail)
+ nb = sw_avail;
+ if (nb > hwsize - swptr)
+ nb = hwsize - swptr; /* don't overflow hwbuf */
+ if (nb > swsize - swidx)
+ nb = swsize - swidx; /* don't overflow swbuf */
+ ASSERT(nb > 0);
+ if (nb % fragsize) {
+ DBGP("nb = %d, fragsize = %d\n", nb, fragsize);
+ DBGP("hw_avail = %d\n", hw_avail);
+ DBGP("sw_avail = %d\n", sw_avail);
+ DBGP("hwsize = %d, swptr = %d\n", hwsize, swptr);
+ DBGP("swsize = %d, swidx = %d\n", swsize, swidx);
+ }
+ ASSERT(!(nb % fragsize));
+ DBGPV("copying hwb[%d..%d] to swb[%d..%d]\n",
+ swptr, swptr + nb, swidx, swidx + nb);
+ pcm_copy_in(rport, swidx, swptr, nb);
+ li_write_swptr(&rport->chan, (swptr + nb) % hwsize);
+ spin_lock_irqsave(&rport->lock, iflags);
+ __swb_inc_i(rport, nb);
+ rport->byte_count += nb;
+ rport->frag_count += nb / fragsize;
+ ASSERT(nb % fragsize == 0);
+ wake_up(&rport->queue);
+ }
+ rport->flags &= ~HW_BUSY;
+ spin_unlock_irqrestore(&rport->lock, iflags);
+ DBGRV();
+}
+
+/*
+ * pcm_flush_frag() writes zero samples to fill the current fragment,
+ * then flushes it to the hardware.
+ *
+ * It is only meaningful to flush output, not input.
+ */
+
+static void pcm_flush_frag(vwsnd_dev_t *devc)
+{
+ vwsnd_port_t *wport = &devc->wport;
+
+ DBGPV("swstate = %d\n", wport->swstate);
+ if (wport->swstate == SW_RUN) {
+ int idx = wport->swb_u_idx;
+ int end = (idx + wport->hw_fragsize - 1)
+ >> wport->hw_fragshift
+ << wport->hw_fragshift;
+ int nb = end - idx;
+ DBGPV("clearing %d bytes\n", nb);
+ if (nb)
+ memset(wport->swbuf + idx,
+ (char) wport->zero_word,
+ nb);
+ wport->swstate = SW_DRAIN;
+ pcm_output(devc, 0, nb);
+ }
+ DBGRV();
+}
+
+/*
+ * Wait for output to drain. This sleeps uninterruptibly because
+ * there is nothing intelligent we can do if interrupted. This
+ * means the process will be delayed in responding to the signal.
+ */
+
+static void pcm_write_sync(vwsnd_dev_t *devc)
+{
+ vwsnd_port_t *wport = &devc->wport;
+ DECLARE_WAITQUEUE(wait, current);
+ unsigned long flags;
+ vwsnd_port_hwstate_t hwstate;
+
+ DBGEV("(devc=0x%p)\n", devc);
+ add_wait_queue(&wport->queue, &wait);
+ while (1) {
+ set_current_state(TASK_UNINTERRUPTIBLE);
+ spin_lock_irqsave(&wport->lock, flags);
+ {
+ hwstate = wport->hwstate;
+ }
+ spin_unlock_irqrestore(&wport->lock, flags);
+ if (hwstate == HW_STOPPED)
+ break;
+ schedule();
+ }
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&wport->queue, &wait);
+ DBGPV("swstate = %d, hwstate = %d\n", wport->swstate, wport->hwstate);
+ DBGRV();
+}
+
+/*****************************************************************************/
+/* audio driver */
+
+/*
+ * seek on an audio device always fails.
+ */
+
+static void vwsnd_audio_read_intr(vwsnd_dev_t *devc, unsigned int status)
+{
+ int overflown = status & LI_INTR_COMM1_OVERFLOW;
+
+ if (status & READ_INTR_MASK)
+ pcm_input(devc, overflown, 0);
+}
+
+static void vwsnd_audio_write_intr(vwsnd_dev_t *devc, unsigned int status)
+{
+ int underflown = status & LI_INTR_COMM2_UNDERFLOW;
+
+ if (status & WRITE_INTR_MASK)
+ pcm_output(devc, underflown, 0);
+}
+
+static irqreturn_t vwsnd_audio_intr(int irq, void *dev_id)
+{
+ vwsnd_dev_t *devc = dev_id;
+ unsigned int status;
+
+ DBGEV("(irq=%d, dev_id=0x%p)\n", irq, dev_id);
+
+ status = li_get_clear_intr_status(&devc->lith);
+ vwsnd_audio_read_intr(devc, status);
+ vwsnd_audio_write_intr(devc, status);
+ return IRQ_HANDLED;
+}
+
+static ssize_t vwsnd_audio_do_read(struct file *file,
+ char *buffer,
+ size_t count,
+ loff_t *ppos)
+{
+ vwsnd_dev_t *devc = file->private_data;
+ vwsnd_port_t *rport = ((file->f_mode & FMODE_READ) ?
+ &devc->rport : NULL);
+ int ret, nb;
+
+ DBGEV("(file=0x%p, buffer=0x%p, count=%d, ppos=0x%p)\n",
+ file, buffer, count, ppos);
+
+ if (!rport)
+ return -EINVAL;
+
+ if (rport->swbuf == NULL) {
+ vwsnd_port_t *wport = (file->f_mode & FMODE_WRITE) ?
+ &devc->wport : NULL;
+ ret = pcm_setup(devc, rport, wport);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (!access_ok(VERIFY_READ, buffer, count))
+ return -EFAULT;
+ ret = 0;
+ while (count) {
+ DECLARE_WAITQUEUE(wait, current);
+ add_wait_queue(&rport->queue, &wait);
+ while ((nb = swb_inc_u(rport, 0)) == 0) {
+ DBGPV("blocking\n");
+ set_current_state(TASK_INTERRUPTIBLE);
+ if (rport->flags & DISABLED ||
+ file->f_flags & O_NONBLOCK) {
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&rport->queue, &wait);
+ return ret ? ret : -EAGAIN;
+ }
+ schedule();
+ if (signal_pending(current)) {
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&rport->queue, &wait);
+ return ret ? ret : -ERESTARTSYS;
+ }
+ }
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&rport->queue, &wait);
+ pcm_input(devc, 0, 0);
+ /* nb bytes are available in userbuf. */
+ if (nb > count)
+ nb = count;
+ DBGPV("nb = %d\n", nb);
+ if (copy_to_user(buffer, rport->swbuf + rport->swb_u_idx, nb))
+ return -EFAULT;
+ (void) swb_inc_u(rport, nb);
+ buffer += nb;
+ count -= nb;
+ ret += nb;
+ }
+ DBGPV("returning %d\n", ret);
+ return ret;
+}
+
+static ssize_t vwsnd_audio_read(struct file *file,
+ char *buffer,
+ size_t count,
+ loff_t *ppos)
+{
+ vwsnd_dev_t *devc = file->private_data;
+ ssize_t ret;
+
+ mutex_lock(&devc->io_mutex);
+ ret = vwsnd_audio_do_read(file, buffer, count, ppos);
+ mutex_unlock(&devc->io_mutex);
+ return ret;
+}
+
+static ssize_t vwsnd_audio_do_write(struct file *file,
+ const char *buffer,
+ size_t count,
+ loff_t *ppos)
+{
+ vwsnd_dev_t *devc = file->private_data;
+ vwsnd_port_t *wport = ((file->f_mode & FMODE_WRITE) ?
+ &devc->wport : NULL);
+ int ret, nb;
+
+ DBGEV("(file=0x%p, buffer=0x%p, count=%d, ppos=0x%p)\n",
+ file, buffer, count, ppos);
+
+ if (!wport)
+ return -EINVAL;
+
+ if (wport->swbuf == NULL) {
+ vwsnd_port_t *rport = (file->f_mode & FMODE_READ) ?
+ &devc->rport : NULL;
+ ret = pcm_setup(devc, rport, wport);
+ if (ret < 0)
+ return ret;
+ }
+ if (!access_ok(VERIFY_WRITE, buffer, count))
+ return -EFAULT;
+ ret = 0;
+ while (count) {
+ DECLARE_WAITQUEUE(wait, current);
+ add_wait_queue(&wport->queue, &wait);
+ while ((nb = swb_inc_u(wport, 0)) == 0) {
+ set_current_state(TASK_INTERRUPTIBLE);
+ if (wport->flags & DISABLED ||
+ file->f_flags & O_NONBLOCK) {
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&wport->queue, &wait);
+ return ret ? ret : -EAGAIN;
+ }
+ schedule();
+ if (signal_pending(current)) {
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&wport->queue, &wait);
+ return ret ? ret : -ERESTARTSYS;
+ }
+ }
+ current->state = TASK_RUNNING;
+ remove_wait_queue(&wport->queue, &wait);
+ /* nb bytes are available in userbuf. */
+ if (nb > count)
+ nb = count;
+ DBGPV("nb = %d\n", nb);
+ if (copy_from_user(wport->swbuf + wport->swb_u_idx, buffer, nb))
+ return -EFAULT;
+ pcm_output(devc, 0, nb);
+ buffer += nb;
+ count -= nb;
+ ret += nb;
+ }
+ DBGPV("returning %d\n", ret);
+ return ret;
+}
+
+static ssize_t vwsnd_audio_write(struct file *file,
+ const char *buffer,
+ size_t count,
+ loff_t *ppos)
+{
+ vwsnd_dev_t *devc = file->private_data;
+ ssize_t ret;
+
+ mutex_lock(&devc->io_mutex);
+ ret = vwsnd_audio_do_write(file, buffer, count, ppos);
+ mutex_unlock(&devc->io_mutex);
+ return ret;
+}
+
+/* No kernel lock - fine */
+static unsigned int vwsnd_audio_poll(struct file *file,
+ struct poll_table_struct *wait)
+{
+ vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data;
+ vwsnd_port_t *rport = (file->f_mode & FMODE_READ) ?
+ &devc->rport : NULL;
+ vwsnd_port_t *wport = (file->f_mode & FMODE_WRITE) ?
+ &devc->wport : NULL;
+ unsigned int mask = 0;
+
+ DBGEV("(file=0x%p, wait=0x%p)\n", file, wait);
+
+ ASSERT(rport || wport);
+ if (rport) {
+ poll_wait(file, &rport->queue, wait);
+ if (swb_inc_u(rport, 0))
+ mask |= (POLLIN | POLLRDNORM);
+ }
+ if (wport) {
+ poll_wait(file, &wport->queue, wait);
+ if (wport->swbuf == NULL || swb_inc_u(wport, 0))
+ mask |= (POLLOUT | POLLWRNORM);
+ }
+
+ DBGPV("returning 0x%x\n", mask);
+ return mask;
+}
+
+static int vwsnd_audio_do_ioctl(struct inode *inode,
+ struct file *file,
+ unsigned int cmd,
+ unsigned long arg)
+{
+ vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data;
+ vwsnd_port_t *rport = (file->f_mode & FMODE_READ) ?
+ &devc->rport : NULL;
+ vwsnd_port_t *wport = (file->f_mode & FMODE_WRITE) ?
+ &devc->wport : NULL;
+ vwsnd_port_t *aport = rport ? rport : wport;
+ struct audio_buf_info buf_info;
+ struct count_info info;
+ unsigned long flags;
+ int ival;
+
+
+ DBGEV("(inode=0x%p, file=0x%p, cmd=0x%x, arg=0x%lx)\n",
+ inode, file, cmd, arg);
+ switch (cmd) {
+ case OSS_GETVERSION: /* _SIOR ('M', 118, int) */
+ DBGX("OSS_GETVERSION\n");
+ ival = SOUND_VERSION;
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_GETCAPS: /* _SIOR ('P',15, int) */
+ DBGX("SNDCTL_DSP_GETCAPS\n");
+ ival = DSP_CAP_DUPLEX | DSP_CAP_REALTIME | DSP_CAP_TRIGGER;
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_GETFMTS: /* _SIOR ('P',11, int) */
+ DBGX("SNDCTL_DSP_GETFMTS\n");
+ ival = (AFMT_S16_LE | AFMT_MU_LAW | AFMT_A_LAW |
+ AFMT_U8 | AFMT_S8);
+ return put_user(ival, (int *) arg);
+ break;
+
+ case SOUND_PCM_READ_RATE: /* _SIOR ('P', 2, int) */
+ DBGX("SOUND_PCM_READ_RATE\n");
+ ival = aport->sw_framerate;
+ return put_user(ival, (int *) arg);
+
+ case SOUND_PCM_READ_CHANNELS: /* _SIOR ('P', 6, int) */
+ DBGX("SOUND_PCM_READ_CHANNELS\n");
+ ival = aport->sw_channels;
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_SPEED: /* _SIOWR('P', 2, int) */
+ if (get_user(ival, (int *) arg))
+ return -EFAULT;
+ DBGX("SNDCTL_DSP_SPEED %d\n", ival);
+ if (ival) {
+ if (aport->swstate != SW_INITIAL) {
+ DBGX("SNDCTL_DSP_SPEED failed: swstate = %d\n",
+ aport->swstate);
+ return -EINVAL;
+ }
+ if (ival < MIN_SPEED)
+ ival = MIN_SPEED;
+ if (ival > MAX_SPEED)
+ ival = MAX_SPEED;
+ if (rport)
+ rport->sw_framerate = ival;
+ if (wport)
+ wport->sw_framerate = ival;
+ } else
+ ival = aport->sw_framerate;
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_STEREO: /* _SIOWR('P', 3, int) */
+ if (get_user(ival, (int *) arg))
+ return -EFAULT;
+ DBGX("SNDCTL_DSP_STEREO %d\n", ival);
+ if (ival != 0 && ival != 1)
+ return -EINVAL;
+ if (aport->swstate != SW_INITIAL)
+ return -EINVAL;
+ if (rport)
+ rport->sw_channels = ival + 1;
+ if (wport)
+ wport->sw_channels = ival + 1;
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_CHANNELS: /* _SIOWR('P', 6, int) */
+ if (get_user(ival, (int *) arg))
+ return -EFAULT;
+ DBGX("SNDCTL_DSP_CHANNELS %d\n", ival);
+ if (ival != 1 && ival != 2)
+ return -EINVAL;
+ if (aport->swstate != SW_INITIAL)
+ return -EINVAL;
+ if (rport)
+ rport->sw_channels = ival;
+ if (wport)
+ wport->sw_channels = ival;
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_GETBLKSIZE: /* _SIOWR('P', 4, int) */
+ ival = pcm_setup(devc, rport, wport);
+ if (ival < 0) {
+ DBGX("SNDCTL_DSP_GETBLKSIZE failed, errno %d\n", ival);
+ return ival;
+ }
+ ival = 1 << aport->sw_fragshift;
+ DBGX("SNDCTL_DSP_GETBLKSIZE returning %d\n", ival);
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_SETFRAGMENT: /* _SIOWR('P',10, int) */
+ if (get_user(ival, (int *) arg))
+ return -EFAULT;
+ DBGX("SNDCTL_DSP_SETFRAGMENT %d:%d\n",
+ ival >> 16, ival & 0xFFFF);
+ if (aport->swstate != SW_INITIAL)
+ return -EINVAL;
+ {
+ int sw_fragshift = ival & 0xFFFF;
+ int sw_subdivshift = aport->sw_subdivshift;
+ int hw_fragshift = sw_fragshift - sw_subdivshift;
+ int sw_fragcount = (ival >> 16) & 0xFFFF;
+ int hw_fragsize;
+ if (hw_fragshift < MIN_FRAGSHIFT)
+ hw_fragshift = MIN_FRAGSHIFT;
+ if (hw_fragshift > MAX_FRAGSHIFT)
+ hw_fragshift = MAX_FRAGSHIFT;
+ sw_fragshift = hw_fragshift + aport->sw_subdivshift;
+ hw_fragsize = 1 << hw_fragshift;
+ if (sw_fragcount < MIN_FRAGCOUNT(hw_fragsize))
+ sw_fragcount = MIN_FRAGCOUNT(hw_fragsize);
+ if (sw_fragcount > MAX_FRAGCOUNT(hw_fragsize))
+ sw_fragcount = MAX_FRAGCOUNT(hw_fragsize);
+ DBGPV("sw_fragshift = %d\n", sw_fragshift);
+ DBGPV("rport = 0x%p, wport = 0x%p\n", rport, wport);
+ if (rport) {
+ rport->sw_fragshift = sw_fragshift;
+ rport->sw_fragcount = sw_fragcount;
+ }
+ if (wport) {
+ wport->sw_fragshift = sw_fragshift;
+ wport->sw_fragcount = sw_fragcount;
+ }
+ ival = sw_fragcount << 16 | sw_fragshift;
+ }
+ DBGX("SNDCTL_DSP_SETFRAGMENT returns %d:%d\n",
+ ival >> 16, ival & 0xFFFF);
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_SUBDIVIDE: /* _SIOWR('P', 9, int) */
+ if (get_user(ival, (int *) arg))
+ return -EFAULT;
+ DBGX("SNDCTL_DSP_SUBDIVIDE %d\n", ival);
+ if (aport->swstate != SW_INITIAL)
+ return -EINVAL;
+ {
+ int subdivshift;
+ int hw_fragshift, hw_fragsize, hw_fragcount;
+ switch (ival) {
+ case 1: subdivshift = 0; break;
+ case 2: subdivshift = 1; break;
+ case 4: subdivshift = 2; break;
+ default: return -EINVAL;
+ }
+ hw_fragshift = aport->sw_fragshift - subdivshift;
+ if (hw_fragshift < MIN_FRAGSHIFT ||
+ hw_fragshift > MAX_FRAGSHIFT)
+ return -EINVAL;
+ hw_fragsize = 1 << hw_fragshift;
+ hw_fragcount = aport->sw_fragcount >> subdivshift;
+ if (hw_fragcount < MIN_FRAGCOUNT(hw_fragsize) ||
+ hw_fragcount > MAX_FRAGCOUNT(hw_fragsize))
+ return -EINVAL;
+ if (rport)
+ rport->sw_subdivshift = subdivshift;
+ if (wport)
+ wport->sw_subdivshift = subdivshift;
+ }
+ return 0;
+
+ case SNDCTL_DSP_SETFMT: /* _SIOWR('P',5, int) */
+ if (get_user(ival, (int *) arg))
+ return -EFAULT;
+ DBGX("SNDCTL_DSP_SETFMT %d\n", ival);
+ if (ival != AFMT_QUERY) {
+ if (aport->swstate != SW_INITIAL) {
+ DBGP("SETFMT failed, swstate = %d\n",
+ aport->swstate);
+ return -EINVAL;
+ }
+ switch (ival) {
+ case AFMT_MU_LAW:
+ case AFMT_A_LAW:
+ case AFMT_U8:
+ case AFMT_S8:
+ case AFMT_S16_LE:
+ if (rport)
+ rport->sw_samplefmt = ival;
+ if (wport)
+ wport->sw_samplefmt = ival;
+ break;
+ default:
+ return -EINVAL;
+ }
+ }
+ ival = aport->sw_samplefmt;
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_GETOSPACE: /* _SIOR ('P',12, audio_buf_info) */
+ DBGXV("SNDCTL_DSP_GETOSPACE\n");
+ if (!wport)
+ return -EINVAL;
+ ival = pcm_setup(devc, rport, wport);
+ if (ival < 0)
+ return ival;
+ ival = swb_inc_u(wport, 0);
+ buf_info.fragments = ival >> wport->sw_fragshift;
+ buf_info.fragstotal = wport->sw_fragcount;
+ buf_info.fragsize = 1 << wport->sw_fragshift;
+ buf_info.bytes = ival;
+ DBGXV("SNDCTL_DSP_GETOSPACE returns { %d %d %d %d }\n",
+ buf_info.fragments, buf_info.fragstotal,
+ buf_info.fragsize, buf_info.bytes);
+ if (copy_to_user((void *) arg, &buf_info, sizeof buf_info))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_GETISPACE: /* _SIOR ('P',13, audio_buf_info) */
+ DBGX("SNDCTL_DSP_GETISPACE\n");
+ if (!rport)
+ return -EINVAL;
+ ival = pcm_setup(devc, rport, wport);
+ if (ival < 0)
+ return ival;
+ ival = swb_inc_u(rport, 0);
+ buf_info.fragments = ival >> rport->sw_fragshift;
+ buf_info.fragstotal = rport->sw_fragcount;
+ buf_info.fragsize = 1 << rport->sw_fragshift;
+ buf_info.bytes = ival;
+ DBGX("SNDCTL_DSP_GETISPACE returns { %d %d %d %d }\n",
+ buf_info.fragments, buf_info.fragstotal,
+ buf_info.fragsize, buf_info.bytes);
+ if (copy_to_user((void *) arg, &buf_info, sizeof buf_info))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_NONBLOCK: /* _SIO ('P',14) */
+ DBGX("SNDCTL_DSP_NONBLOCK\n");
+ file->f_flags |= O_NONBLOCK;
+ return 0;
+
+ case SNDCTL_DSP_RESET: /* _SIO ('P', 0) */
+ DBGX("SNDCTL_DSP_RESET\n");
+ /*
+ * Nothing special needs to be done for input. Input
+ * samples sit in swbuf, but it will be reinitialized
+ * to empty when pcm_setup() is called.
+ */
+ if (wport && wport->swbuf) {
+ wport->swstate = SW_INITIAL;
+ pcm_output(devc, 0, 0);
+ pcm_write_sync(devc);
+ }
+ pcm_shutdown(devc, rport, wport);
+ return 0;
+
+ case SNDCTL_DSP_SYNC: /* _SIO ('P', 1) */
+ DBGX("SNDCTL_DSP_SYNC\n");
+ if (wport) {
+ pcm_flush_frag(devc);
+ pcm_write_sync(devc);
+ }
+ pcm_shutdown(devc, rport, wport);
+ return 0;
+
+ case SNDCTL_DSP_POST: /* _SIO ('P', 8) */
+ DBGX("SNDCTL_DSP_POST\n");
+ if (!wport)
+ return -EINVAL;
+ pcm_flush_frag(devc);
+ return 0;
+
+ case SNDCTL_DSP_GETIPTR: /* _SIOR ('P', 17, count_info) */
+ DBGX("SNDCTL_DSP_GETIPTR\n");
+ if (!rport)
+ return -EINVAL;
+ spin_lock_irqsave(&rport->lock, flags);
+ {
+ ustmsc_t ustmsc;
+ if (rport->hwstate == HW_RUNNING) {
+ ASSERT(rport->swstate == SW_RUN);
+ li_read_USTMSC(&rport->chan, &ustmsc);
+ info.bytes = ustmsc.msc - rport->MSC_offset;
+ info.bytes *= rport->frame_size;
+ } else {
+ info.bytes = rport->byte_count;
+ }
+ info.blocks = rport->frag_count;
+ info.ptr = 0; /* not implemented */
+ rport->frag_count = 0;
+ }
+ spin_unlock_irqrestore(&rport->lock, flags);
+ if (copy_to_user((void *) arg, &info, sizeof info))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_GETOPTR: /* _SIOR ('P',18, count_info) */
+ DBGX("SNDCTL_DSP_GETOPTR\n");
+ if (!wport)
+ return -EINVAL;
+ spin_lock_irqsave(&wport->lock, flags);
+ {
+ ustmsc_t ustmsc;
+ if (wport->hwstate == HW_RUNNING) {
+ ASSERT(wport->swstate == SW_RUN);
+ li_read_USTMSC(&wport->chan, &ustmsc);
+ info.bytes = ustmsc.msc - wport->MSC_offset;
+ info.bytes *= wport->frame_size;
+ } else {
+ info.bytes = wport->byte_count;
+ }
+ info.blocks = wport->frag_count;
+ info.ptr = 0; /* not implemented */
+ wport->frag_count = 0;
+ }
+ spin_unlock_irqrestore(&wport->lock, flags);
+ if (copy_to_user((void *) arg, &info, sizeof info))
+ return -EFAULT;
+ return 0;
+
+ case SNDCTL_DSP_GETODELAY: /* _SIOR ('P', 23, int) */
+ DBGX("SNDCTL_DSP_GETODELAY\n");
+ if (!wport)
+ return -EINVAL;
+ spin_lock_irqsave(&wport->lock, flags);
+ {
+ int fsize = wport->frame_size;
+ ival = wport->swb_i_avail / fsize;
+ if (wport->hwstate == HW_RUNNING) {
+ int swptr, hwptr, hwframes, hwbytes, hwsize;
+ int totalhwbytes;
+ ustmsc_t ustmsc;
+
+ hwsize = wport->hwbuf_size;
+ swptr = li_read_swptr(&wport->chan);
+ li_read_USTMSC(&wport->chan, &ustmsc);
+ hwframes = ustmsc.msc - wport->MSC_offset;
+ totalhwbytes = hwframes * fsize;
+ hwptr = totalhwbytes % hwsize;
+ hwbytes = (swptr - hwptr + hwsize) % hwsize;
+ ival += hwbytes / fsize;
+ }
+ }
+ spin_unlock_irqrestore(&wport->lock, flags);
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_PROFILE: /* _SIOW ('P', 23, int) */
+ DBGX("SNDCTL_DSP_PROFILE\n");
+
+ /*
+ * Thomas Sailer explains SNDCTL_DSP_PROFILE
+ * (private email, March 24, 1999):
+ *
+ * This gives the sound driver a hint on what it
+ * should do with partial fragments
+ * (i.e. fragments partially filled with write).
+ * This can direct the driver to zero them or
+ * leave them alone. But don't ask me what this
+ * is good for, my driver just zeroes the last
+ * fragment before the receiver stops, no idea
+ * what good for any other behaviour could
+ * be. Implementing it as NOP seems safe.
+ */
+
+ break;
+
+ case SNDCTL_DSP_GETTRIGGER: /* _SIOR ('P',16, int) */
+ DBGX("SNDCTL_DSP_GETTRIGGER\n");
+ ival = 0;
+ if (rport) {
+ spin_lock_irqsave(&rport->lock, flags);
+ {
+ if (!(rport->flags & DISABLED))
+ ival |= PCM_ENABLE_INPUT;
+ }
+ spin_unlock_irqrestore(&rport->lock, flags);
+ }
+ if (wport) {
+ spin_lock_irqsave(&wport->lock, flags);
+ {
+ if (!(wport->flags & DISABLED))
+ ival |= PCM_ENABLE_OUTPUT;
+ }
+ spin_unlock_irqrestore(&wport->lock, flags);
+ }
+ return put_user(ival, (int *) arg);
+
+ case SNDCTL_DSP_SETTRIGGER: /* _SIOW ('P',16, int) */
+ if (get_user(ival, (int *) arg))
+ return -EFAULT;
+ DBGX("SNDCTL_DSP_SETTRIGGER %d\n", ival);
+
+ /*
+ * If user is disabling I/O and port is not in initial
+ * state, fail with EINVAL.
+ */
+
+ if (((rport && !(ival & PCM_ENABLE_INPUT)) ||
+ (wport && !(ival & PCM_ENABLE_OUTPUT))) &&
+ aport->swstate != SW_INITIAL)
+ return -EINVAL;
+
+ if (rport) {
+ vwsnd_port_hwstate_t hwstate;
+ spin_lock_irqsave(&rport->lock, flags);
+ {
+ hwstate = rport->hwstate;
+ if (ival & PCM_ENABLE_INPUT)
+ rport->flags &= ~DISABLED;
+ else
+ rport->flags |= DISABLED;
+ }
+ spin_unlock_irqrestore(&rport->lock, flags);
+ if (hwstate != HW_RUNNING && ival & PCM_ENABLE_INPUT) {
+
+ if (rport->swstate == SW_INITIAL)
+ pcm_setup(devc, rport, wport);
+ else
+ li_activate_dma(&rport->chan);
+ }
+ }
+ if (wport) {
+ vwsnd_port_flags_t pflags;
+ spin_lock_irqsave(&wport->lock, flags);
+ {
+ pflags = wport->flags;
+ if (ival & PCM_ENABLE_OUTPUT)
+ wport->flags &= ~DISABLED;
+ else
+ wport->flags |= DISABLED;
+ }
+ spin_unlock_irqrestore(&wport->lock, flags);
+ if (pflags & DISABLED && ival & PCM_ENABLE_OUTPUT) {
+ if (wport->swstate == SW_RUN)
+ pcm_output(devc, 0, 0);
+ }
+ }
+ return 0;
+
+ default:
+ DBGP("unknown ioctl 0x%x\n", cmd);
+ return -EINVAL;
+ }
+ DBGP("unimplemented ioctl 0x%x\n", cmd);
+ return -EINVAL;
+}
+
+static int vwsnd_audio_ioctl(struct inode *inode,
+ struct file *file,
+ unsigned int cmd,
+ unsigned long arg)
+{
+ vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data;
+ int ret;
+
+ mutex_lock(&devc->io_mutex);
+ ret = vwsnd_audio_do_ioctl(inode, file, cmd, arg);
+ mutex_unlock(&devc->io_mutex);
+ return ret;
+}
+
+/* No mmap. */
+
+static int vwsnd_audio_mmap(struct file *file, struct vm_area_struct *vma)
+{
+ DBGE("(file=0x%p, vma=0x%p)\n", file, vma);
+ return -ENODEV;
+}
+
+/*
+ * Open the audio device for read and/or write.
+ *
+ * Returns 0 on success, -errno on failure.
+ */
+
+static int vwsnd_audio_open(struct inode *inode, struct file *file)
+{
+ vwsnd_dev_t *devc;
+ int minor = iminor(inode);
+ int sw_samplefmt;
+
+ DBGE("(inode=0x%p, file=0x%p)\n", inode, file);
+
+ INC_USE_COUNT;
+ for (devc = vwsnd_dev_list; devc; devc = devc->next_dev)
+ if ((devc->audio_minor & ~0x0F) == (minor & ~0x0F))
+ break;
+
+ if (devc == NULL) {
+ DEC_USE_COUNT;
+ return -ENODEV;
+ }
+
+ mutex_lock(&devc->open_mutex);
+ while (devc->open_mode & file->f_mode) {
+ mutex_unlock(&devc->open_mutex);
+ if (file->f_flags & O_NONBLOCK) {
+ DEC_USE_COUNT;
+ return -EBUSY;
+ }
+ interruptible_sleep_on(&devc->open_wait);
+ if (signal_pending(current)) {
+ DEC_USE_COUNT;
+ return -ERESTARTSYS;
+ }
+ mutex_lock(&devc->open_mutex);
+ }
+ devc->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE);
+ mutex_unlock(&devc->open_mutex);
+
+ /* get default sample format from minor number. */
+
+ sw_samplefmt = 0;
+ if ((minor & 0xF) == SND_DEV_DSP)
+ sw_samplefmt = AFMT_U8;
+ else if ((minor & 0xF) == SND_DEV_AUDIO)
+ sw_samplefmt = AFMT_MU_LAW;
+ else if ((minor & 0xF) == SND_DEV_DSP16)
+ sw_samplefmt = AFMT_S16_LE;
+ else
+ ASSERT(0);
+
+ /* Initialize vwsnd_ports. */
+
+ mutex_lock(&devc->io_mutex);
+ {
+ if (file->f_mode & FMODE_READ) {
+ devc->rport.swstate = SW_INITIAL;
+ devc->rport.flags = 0;
+ devc->rport.sw_channels = 1;
+ devc->rport.sw_samplefmt = sw_samplefmt;
+ devc->rport.sw_framerate = 8000;
+ devc->rport.sw_fragshift = DEFAULT_FRAGSHIFT;
+ devc->rport.sw_fragcount = DEFAULT_FRAGCOUNT;
+ devc->rport.sw_subdivshift = DEFAULT_SUBDIVSHIFT;
+ devc->rport.byte_count = 0;
+ devc->rport.frag_count = 0;
+ }
+ if (file->f_mode & FMODE_WRITE) {
+ devc->wport.swstate = SW_INITIAL;
+ devc->wport.flags = 0;
+ devc->wport.sw_channels = 1;
+ devc->wport.sw_samplefmt = sw_samplefmt;
+ devc->wport.sw_framerate = 8000;
+ devc->wport.sw_fragshift = DEFAULT_FRAGSHIFT;
+ devc->wport.sw_fragcount = DEFAULT_FRAGCOUNT;
+ devc->wport.sw_subdivshift = DEFAULT_SUBDIVSHIFT;
+ devc->wport.byte_count = 0;
+ devc->wport.frag_count = 0;
+ }
+ }
+ mutex_unlock(&devc->io_mutex);
+
+ file->private_data = devc;
+ DBGRV();
+ return 0;
+}
+
+/*
+ * Release (close) the audio device.
+ */
+
+static int vwsnd_audio_release(struct inode *inode, struct file *file)
+{
+ vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data;
+ vwsnd_port_t *wport = NULL, *rport = NULL;
+ int err = 0;
+
+ lock_kernel();
+ mutex_lock(&devc->io_mutex);
+ {
+ DBGEV("(inode=0x%p, file=0x%p)\n", inode, file);
+
+ if (file->f_mode & FMODE_READ)
+ rport = &devc->rport;
+ if (file->f_mode & FMODE_WRITE) {
+ wport = &devc->wport;
+ pcm_flush_frag(devc);
+ pcm_write_sync(devc);
+ }
+ pcm_shutdown(devc, rport, wport);
+ if (rport)
+ rport->swstate = SW_OFF;
+ if (wport)
+ wport->swstate = SW_OFF;
+ }
+ mutex_unlock(&devc->io_mutex);
+
+ mutex_lock(&devc->open_mutex);
+ {
+ devc->open_mode &= ~file->f_mode;
+ }
+ mutex_unlock(&devc->open_mutex);
+ wake_up(&devc->open_wait);
+ DEC_USE_COUNT;
+ DBGR();
+ unlock_kernel();
+ return err;
+}
+
+static const struct file_operations vwsnd_audio_fops = {
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .read = vwsnd_audio_read,
+ .write = vwsnd_audio_write,
+ .poll = vwsnd_audio_poll,
+ .ioctl = vwsnd_audio_ioctl,
+ .mmap = vwsnd_audio_mmap,
+ .open = vwsnd_audio_open,
+ .release = vwsnd_audio_release,
+};
+
+/*****************************************************************************/
+/* mixer driver */
+
+/* open the mixer device. */
+
+static int vwsnd_mixer_open(struct inode *inode, struct file *file)
+{
+ vwsnd_dev_t *devc;
+
+ DBGEV("(inode=0x%p, file=0x%p)\n", inode, file);
+
+ INC_USE_COUNT;
+ for (devc = vwsnd_dev_list; devc; devc = devc->next_dev)
+ if (devc->mixer_minor == iminor(inode))
+ break;
+
+ if (devc == NULL) {
+ DEC_USE_COUNT;
+ return -ENODEV;
+ }
+ file->private_data = devc;
+ return 0;
+}
+
+/* release (close) the mixer device. */
+
+static int vwsnd_mixer_release(struct inode *inode, struct file *file)
+{
+ DBGEV("(inode=0x%p, file=0x%p)\n", inode, file);
+ DEC_USE_COUNT;
+ return 0;
+}
+
+/* mixer_read_ioctl handles all read ioctls on the mixer device. */
+
+static int mixer_read_ioctl(vwsnd_dev_t *devc, unsigned int nr, void __user *arg)
+{
+ int val = -1;
+
+ DBGEV("(devc=0x%p, nr=0x%x, arg=0x%p)\n", devc, nr, arg);
+
+ switch (nr) {
+ case SOUND_MIXER_CAPS:
+ val = SOUND_CAP_EXCL_INPUT;
+ break;
+
+ case SOUND_MIXER_DEVMASK:
+ val = (SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_MIC | SOUND_MASK_CD | SOUND_MASK_RECLEV);
+ break;
+
+ case SOUND_MIXER_STEREODEVS:
+ val = (SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_MIC | SOUND_MASK_CD | SOUND_MASK_RECLEV);
+ break;
+
+ case SOUND_MIXER_OUTMASK:
+ val = (SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_MIC | SOUND_MASK_CD);
+ break;
+
+ case SOUND_MIXER_RECMASK:
+ val = (SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_MIC | SOUND_MASK_CD);
+ break;
+
+ case SOUND_MIXER_PCM:
+ val = ad1843_get_gain(&devc->lith, &ad1843_gain_PCM);
+ break;
+
+ case SOUND_MIXER_LINE:
+ val = ad1843_get_gain(&devc->lith, &ad1843_gain_LINE);
+ break;
+
+ case SOUND_MIXER_MIC:
+ val = ad1843_get_gain(&devc->lith, &ad1843_gain_MIC);
+ break;
+
+ case SOUND_MIXER_CD:
+ val = ad1843_get_gain(&devc->lith, &ad1843_gain_CD);
+ break;
+
+ case SOUND_MIXER_RECLEV:
+ val = ad1843_get_gain(&devc->lith, &ad1843_gain_RECLEV);
+ break;
+
+ case SOUND_MIXER_RECSRC:
+ val = ad1843_get_recsrc(&devc->lith);
+ break;
+
+ case SOUND_MIXER_OUTSRC:
+ val = ad1843_get_outsrc(&devc->lith);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return put_user(val, (int __user *) arg);
+}
+
+/* mixer_write_ioctl handles all write ioctls on the mixer device. */
+
+static int mixer_write_ioctl(vwsnd_dev_t *devc, unsigned int nr, void __user *arg)
+{
+ int val;
+ int err;
+
+ DBGEV("(devc=0x%p, nr=0x%x, arg=0x%p)\n", devc, nr, arg);
+
+ err = get_user(val, (int __user *) arg);
+ if (err)
+ return -EFAULT;
+ switch (nr) {
+ case SOUND_MIXER_PCM:
+ val = ad1843_set_gain(&devc->lith, &ad1843_gain_PCM, val);
+ break;
+
+ case SOUND_MIXER_LINE:
+ val = ad1843_set_gain(&devc->lith, &ad1843_gain_LINE, val);
+ break;
+
+ case SOUND_MIXER_MIC:
+ val = ad1843_set_gain(&devc->lith, &ad1843_gain_MIC, val);
+ break;
+
+ case SOUND_MIXER_CD:
+ val = ad1843_set_gain(&devc->lith, &ad1843_gain_CD, val);
+ break;
+
+ case SOUND_MIXER_RECLEV:
+ val = ad1843_set_gain(&devc->lith, &ad1843_gain_RECLEV, val);
+ break;
+
+ case SOUND_MIXER_RECSRC:
+ if (devc->rport.swbuf || devc->wport.swbuf)
+ return -EBUSY; /* can't change recsrc while running */
+ val = ad1843_set_recsrc(&devc->lith, val);
+ break;
+
+ case SOUND_MIXER_OUTSRC:
+ val = ad1843_set_outsrc(&devc->lith, val);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ if (val < 0)
+ return val;
+ return put_user(val, (int __user *) arg);
+}
+
+/* This is the ioctl entry to the mixer driver. */
+
+static int vwsnd_mixer_ioctl(struct inode *ioctl,
+ struct file *file,
+ unsigned int cmd,
+ unsigned long arg)
+{
+ vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data;
+ const unsigned int nrmask = _IOC_NRMASK << _IOC_NRSHIFT;
+ const unsigned int nr = (cmd & nrmask) >> _IOC_NRSHIFT;
+ int retval;
+
+ DBGEV("(devc=0x%p, cmd=0x%x, arg=0x%lx)\n", devc, cmd, arg);
+
+ mutex_lock(&devc->mix_mutex);
+ {
+ if ((cmd & ~nrmask) == MIXER_READ(0))
+ retval = mixer_read_ioctl(devc, nr, (void __user *) arg);
+ else if ((cmd & ~nrmask) == MIXER_WRITE(0))
+ retval = mixer_write_ioctl(devc, nr, (void __user *) arg);
+ else
+ retval = -EINVAL;
+ }
+ mutex_unlock(&devc->mix_mutex);
+ return retval;
+}
+
+static const struct file_operations vwsnd_mixer_fops = {
+ .owner = THIS_MODULE,
+ .llseek = no_llseek,
+ .ioctl = vwsnd_mixer_ioctl,
+ .open = vwsnd_mixer_open,
+ .release = vwsnd_mixer_release,
+};
+
+/*****************************************************************************/
+/* probe/attach/unload */
+
+/* driver probe routine. Return nonzero if hardware is found. */
+
+static int __init probe_vwsnd(struct address_info *hw_config)
+{
+ lithium_t lith;
+ int w;
+ unsigned long later;
+
+ DBGEV("(hw_config=0x%p)\n", hw_config);
+
+ /* XXX verify lithium present (to prevent crash on non-vw) */
+
+ if (li_create(&lith, hw_config->io_base) != 0) {
+ printk(KERN_WARNING "probe_vwsnd: can't map lithium\n");
+ return 0;
+ }
+ later = jiffies + 2;
+ li_writel(&lith, LI_HOST_CONTROLLER, LI_HC_LINK_ENABLE);
+ do {
+ w = li_readl(&lith, LI_HOST_CONTROLLER);
+ } while (w == LI_HC_LINK_ENABLE && time_before(jiffies, later));
+
+ li_destroy(&lith);
+
+ DBGPV("HC = 0x%04x\n", w);
+
+ if ((w == LI_HC_LINK_ENABLE) || (w & LI_HC_LINK_CODEC)) {
+
+ /* This may indicate a beta machine with no audio,
+ * or a future machine with different audio.
+ * On beta-release 320 w/ no audio, HC == 0x4000 */
+
+ printk(KERN_WARNING "probe_vwsnd: audio codec not found\n");
+ return 0;
+ }
+
+ if (w & LI_HC_LINK_FAILURE) {
+ printk(KERN_WARNING "probe_vwsnd: can't init audio codec\n");
+ return 0;
+ }
+
+ printk(KERN_INFO "vwsnd: lithium audio at mmio %#x irq %d\n",
+ hw_config->io_base, hw_config->irq);
+
+ return 1;
+}
+
+/*
+ * driver attach routine. Initialize driver data structures and
+ * initialize hardware. A new vwsnd_dev_t is allocated and put
+ * onto the global list, vwsnd_dev_list.
+ *
+ * Return +minor_dev on success, -errno on failure.
+ */
+
+static int __init attach_vwsnd(struct address_info *hw_config)
+{
+ vwsnd_dev_t *devc = NULL;
+ int err = -ENOMEM;
+
+ DBGEV("(hw_config=0x%p)\n", hw_config);
+
+ devc = kmalloc(sizeof (vwsnd_dev_t), GFP_KERNEL);
+ if (devc == NULL)
+ goto fail0;
+
+ err = li_create(&devc->lith, hw_config->io_base);
+ if (err)
+ goto fail1;
+
+ init_waitqueue_head(&devc->open_wait);
+
+ devc->rport.hwbuf_size = HWBUF_SIZE;
+ devc->rport.hwbuf_vaddr = __get_free_pages(GFP_KERNEL, HWBUF_ORDER);
+ if (!devc->rport.hwbuf_vaddr)
+ goto fail2;
+ devc->rport.hwbuf = (void *) devc->rport.hwbuf_vaddr;
+ devc->rport.hwbuf_paddr = virt_to_phys(devc->rport.hwbuf);
+
+ /*
+ * Quote from the NT driver:
+ *
+ * // WARNING!!! HACK to setup output dma!!!
+ * // This is required because even on output there is some data
+ * // trickling into the input DMA channel. This is a bug in the
+ * // Lithium microcode.
+ * // --sde
+ *
+ * We set the input side's DMA base address here. It will remain
+ * valid until the driver is unloaded.
+ */
+
+ li_writel(&devc->lith, LI_COMM1_BASE,
+ devc->rport.hwbuf_paddr >> 8 | 1 << (37 - 8));
+
+ devc->wport.hwbuf_size = HWBUF_SIZE;
+ devc->wport.hwbuf_vaddr = __get_free_pages(GFP_KERNEL, HWBUF_ORDER);
+ if (!devc->wport.hwbuf_vaddr)
+ goto fail3;
+ devc->wport.hwbuf = (void *) devc->wport.hwbuf_vaddr;
+ devc->wport.hwbuf_paddr = virt_to_phys(devc->wport.hwbuf);
+ DBGP("wport hwbuf = 0x%p\n", devc->wport.hwbuf);
+
+ DBGDO(shut_up++);
+ err = ad1843_init(&devc->lith);
+ DBGDO(shut_up--);
+ if (err)
+ goto fail4;
+
+ /* install interrupt handler */
+
+ err = request_irq(hw_config->irq, vwsnd_audio_intr, 0, "vwsnd", devc);
+ if (err)
+ goto fail5;
+
+ /* register this device's drivers. */
+
+ devc->audio_minor = register_sound_dsp(&vwsnd_audio_fops, -1);
+ if ((err = devc->audio_minor) < 0) {
+ DBGDO(printk(KERN_WARNING
+ "attach_vwsnd: register_sound_dsp error %d\n",
+ err));
+ goto fail6;
+ }
+ devc->mixer_minor = register_sound_mixer(&vwsnd_mixer_fops,
+ devc->audio_minor >> 4);
+ if ((err = devc->mixer_minor) < 0) {
+ DBGDO(printk(KERN_WARNING
+ "attach_vwsnd: register_sound_mixer error %d\n",
+ err));
+ goto fail7;
+ }
+
+ /* Squirrel away device indices for unload routine. */
+
+ hw_config->slots[0] = devc->audio_minor;
+
+ /* Initialize as much of *devc as possible */
+
+ mutex_init(&devc->open_mutex);
+ mutex_init(&devc->io_mutex);
+ mutex_init(&devc->mix_mutex);
+ devc->open_mode = 0;
+ spin_lock_init(&devc->rport.lock);
+ init_waitqueue_head(&devc->rport.queue);
+ devc->rport.swstate = SW_OFF;
+ devc->rport.hwstate = HW_STOPPED;
+ devc->rport.flags = 0;
+ devc->rport.swbuf = NULL;
+ spin_lock_init(&devc->wport.lock);
+ init_waitqueue_head(&devc->wport.queue);
+ devc->wport.swstate = SW_OFF;
+ devc->wport.hwstate = HW_STOPPED;
+ devc->wport.flags = 0;
+ devc->wport.swbuf = NULL;
+
+ /* Success. Link us onto the local device list. */
+
+ devc->next_dev = vwsnd_dev_list;
+ vwsnd_dev_list = devc;
+ return devc->audio_minor;
+
+ /* So many ways to fail. Undo what we did. */
+
+ fail7:
+ unregister_sound_dsp(devc->audio_minor);
+ fail6:
+ free_irq(hw_config->irq, devc);
+ fail5:
+ fail4:
+ free_pages(devc->wport.hwbuf_vaddr, HWBUF_ORDER);
+ fail3:
+ free_pages(devc->rport.hwbuf_vaddr, HWBUF_ORDER);
+ fail2:
+ li_destroy(&devc->lith);
+ fail1:
+ kfree(devc);
+ fail0:
+ return err;
+}
+
+static int __exit unload_vwsnd(struct address_info *hw_config)
+{
+ vwsnd_dev_t *devc, **devcp;
+
+ DBGE("()\n");
+
+ devcp = &vwsnd_dev_list;
+ while ((devc = *devcp)) {
+ if (devc->audio_minor == hw_config->slots[0]) {
+ *devcp = devc->next_dev;
+ break;
+ }
+ devcp = &devc->next_dev;
+ }
+
+ if (!devc)
+ return -ENODEV;
+
+ unregister_sound_mixer(devc->mixer_minor);
+ unregister_sound_dsp(devc->audio_minor);
+ free_irq(hw_config->irq, devc);
+ free_pages(devc->wport.hwbuf_vaddr, HWBUF_ORDER);
+ free_pages(devc->rport.hwbuf_vaddr, HWBUF_ORDER);
+ li_destroy(&devc->lith);
+ kfree(devc);
+
+ return 0;
+}
+
+/*****************************************************************************/
+/* initialization and loadable kernel module interface */
+
+static struct address_info the_hw_config = {
+ 0xFF001000, /* lithium phys addr */
+ CO_IRQ(CO_APIC_LI_AUDIO) /* irq */
+};
+
+MODULE_DESCRIPTION("SGI Visual Workstation sound module");
+MODULE_AUTHOR("Bob Miller <kbob@sgi.com>");
+MODULE_LICENSE("GPL");
+
+static int __init init_vwsnd(void)
+{
+ int err;
+
+ DBGXV("\n");
+ DBGXV("sound::vwsnd::init_module()\n");
+
+ if (!probe_vwsnd(&the_hw_config))
+ return -ENODEV;
+
+ err = attach_vwsnd(&the_hw_config);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static void __exit cleanup_vwsnd(void)
+{
+ DBGX("sound::vwsnd::cleanup_module()\n");
+
+ unload_vwsnd(&the_hw_config);
+}
+
+module_init(init_vwsnd);
+module_exit(cleanup_vwsnd);
diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c
new file mode 100644
index 0000000..c47842f
--- /dev/null
+++ b/sound/oss/waveartist.c
@@ -0,0 +1,2032 @@
+/*
+ * linux/sound/oss/waveartist.c
+ *
+ * The low level driver for the RWA010 Rockwell Wave Artist
+ * codec chip used in the Rebel.com NetWinder.
+ *
+ * Cleaned up and integrated into 2.1 by Russell King (rmk@arm.linux.org.uk)
+ * and Pat Beirne (patb@corel.ca)
+ *
+ *
+ * Copyright (C) by Rebel.com 1998-1999
+ *
+ * RWA010 specs received under NDA from Rockwell
+ *
+ * Copyright (C) by Hannu Savolainen 1993-1997
+ *
+ * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
+ * Version 2 (June 1991). See the "COPYING" file distributed with this software
+ * for more info.
+ *
+ * Changes:
+ * 11-10-2000 Bartlomiej Zolnierkiewicz <bkz@linux-ide.org>
+ * Added __init to waveartist_init()
+ */
+
+/* Debugging */
+#define DEBUG_CMD 1
+#define DEBUG_OUT 2
+#define DEBUG_IN 4
+#define DEBUG_INTR 8
+#define DEBUG_MIXER 16
+#define DEBUG_TRIGGER 32
+
+#define debug_flg (0)
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/sched.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <linux/spinlock.h>
+#include <linux/bitops.h>
+
+#include <asm/system.h>
+
+#include "sound_config.h"
+#include "waveartist.h"
+
+#ifdef CONFIG_ARM
+#include <mach/hardware.h>
+#include <asm/mach-types.h>
+#endif
+
+#ifndef NO_DMA
+#define NO_DMA 255
+#endif
+
+#define SUPPORTED_MIXER_DEVICES (SOUND_MASK_SYNTH |\
+ SOUND_MASK_PCM |\
+ SOUND_MASK_LINE |\
+ SOUND_MASK_MIC |\
+ SOUND_MASK_LINE1 |\
+ SOUND_MASK_RECLEV |\
+ SOUND_MASK_VOLUME |\
+ SOUND_MASK_IMIX)
+
+static unsigned short levels[SOUND_MIXER_NRDEVICES] = {
+ 0x5555, /* Master Volume */
+ 0x0000, /* Bass */
+ 0x0000, /* Treble */
+ 0x2323, /* Synth (FM) */
+ 0x4b4b, /* PCM */
+ 0x6464, /* PC Speaker */
+ 0x0000, /* Ext Line */
+ 0x0000, /* Mic */
+ 0x0000, /* CD */
+ 0x6464, /* Recording monitor */
+ 0x0000, /* SB PCM (ALT PCM) */
+ 0x0000, /* Recording level */
+ 0x6464, /* Input gain */
+ 0x6464, /* Output gain */
+ 0x0000, /* Line1 (Aux1) */
+ 0x0000, /* Line2 (Aux2) */
+ 0x0000, /* Line3 (Aux3) */
+ 0x0000, /* Digital1 */
+ 0x0000, /* Digital2 */
+ 0x0000, /* Digital3 */
+ 0x0000, /* Phone In */
+ 0x6464, /* Phone Out */
+ 0x0000, /* Video */
+ 0x0000, /* Radio */
+ 0x0000 /* Monitor */
+};
+
+typedef struct {
+ struct address_info hw; /* hardware */
+ char *chip_name;
+
+ int xfer_count;
+ int audio_mode;
+ int open_mode;
+ int audio_flags;
+ int record_dev;
+ int playback_dev;
+ int dev_no;
+
+ /* Mixer parameters */
+ const struct waveartist_mixer_info *mix;
+
+ unsigned short *levels; /* cache of volume settings */
+ int recmask; /* currently enabled recording device! */
+
+#ifdef CONFIG_ARCH_NETWINDER
+ signed int slider_vol; /* hardware slider volume */
+ unsigned int handset_detect :1;
+ unsigned int telephone_detect:1;
+ unsigned int no_autoselect :1;/* handset/telephone autoselects a path */
+ unsigned int spkr_mute_state :1;/* set by ioctl or autoselect */
+ unsigned int line_mute_state :1;/* set by ioctl or autoselect */
+ unsigned int use_slider :1;/* use slider setting for o/p vol */
+#endif
+} wavnc_info;
+
+/*
+ * This is the implementation specific mixer information.
+ */
+struct waveartist_mixer_info {
+ unsigned int supported_devs; /* Supported devices */
+ unsigned int recording_devs; /* Recordable devies */
+ unsigned int stereo_devs; /* Stereo devices */
+
+ unsigned int (*select_input)(wavnc_info *, unsigned int,
+ unsigned char *, unsigned char *);
+ int (*decode_mixer)(wavnc_info *, int,
+ unsigned char, unsigned char);
+ int (*get_mixer)(wavnc_info *, int);
+};
+
+typedef struct wavnc_port_info {
+ int open_mode;
+ int speed;
+ int channels;
+ int audio_format;
+} wavnc_port_info;
+
+static int nr_waveartist_devs;
+static wavnc_info adev_info[MAX_AUDIO_DEV];
+static DEFINE_SPINLOCK(waveartist_lock);
+
+#ifndef CONFIG_ARCH_NETWINDER
+#define machine_is_netwinder() 0
+#else
+static struct timer_list vnc_timer;
+static void vnc_configure_mixer(wavnc_info *devc, unsigned int input_mask);
+static int vnc_private_ioctl(int dev, unsigned int cmd, int __user *arg);
+static void vnc_slider_tick(unsigned long data);
+#endif
+
+static inline void
+waveartist_set_ctlr(struct address_info *hw, unsigned char clear, unsigned char set)
+{
+ unsigned int ctlr_port = hw->io_base + CTLR;
+
+ clear = ~clear & inb(ctlr_port);
+
+ outb(clear | set, ctlr_port);
+}
+
+/* Toggle IRQ acknowledge line
+ */
+static inline void
+waveartist_iack(wavnc_info *devc)
+{
+ unsigned int ctlr_port = devc->hw.io_base + CTLR;
+ int old_ctlr;
+
+ old_ctlr = inb(ctlr_port) & ~IRQ_ACK;
+
+ outb(old_ctlr | IRQ_ACK, ctlr_port);
+ outb(old_ctlr, ctlr_port);
+}
+
+static inline int
+waveartist_sleep(int timeout_ms)
+{
+ unsigned int timeout = timeout_ms * 10 * HZ / 100;
+
+ do {
+ set_current_state(TASK_INTERRUPTIBLE);
+ timeout = schedule_timeout(timeout);
+ } while (timeout);
+
+ return 0;
+}
+
+static int
+waveartist_reset(wavnc_info *devc)
+{
+ struct address_info *hw = &devc->hw;
+ unsigned int timeout, res = -1;
+
+ waveartist_set_ctlr(hw, -1, RESET);
+ waveartist_sleep(2);
+ waveartist_set_ctlr(hw, RESET, 0);
+
+ timeout = 500;
+ do {
+ mdelay(2);
+
+ if (inb(hw->io_base + STATR) & CMD_RF) {
+ res = inw(hw->io_base + CMDR);
+ if (res == 0x55aa)
+ break;
+ }
+ } while (--timeout);
+
+ if (timeout == 0) {
+ printk(KERN_WARNING "WaveArtist: reset timeout ");
+ if (res != (unsigned int)-1)
+ printk("(res=%04X)", res);
+ printk("\n");
+ return 1;
+ }
+ return 0;
+}
+
+/* Helper function to send and receive words
+ * from WaveArtist. It handles all the handshaking
+ * and can send or receive multiple words.
+ */
+static int
+waveartist_cmd(wavnc_info *devc,
+ int nr_cmd, unsigned int *cmd,
+ int nr_resp, unsigned int *resp)
+{
+ unsigned int io_base = devc->hw.io_base;
+ unsigned int timed_out = 0;
+ unsigned int i;
+
+ if (debug_flg & DEBUG_CMD) {
+ printk("waveartist_cmd: cmd=");
+
+ for (i = 0; i < nr_cmd; i++)
+ printk("%04X ", cmd[i]);
+
+ printk("\n");
+ }
+
+ if (inb(io_base + STATR) & CMD_RF) {
+ int old_data;
+
+ /* flush the port
+ */
+
+ old_data = inw(io_base + CMDR);
+
+ if (debug_flg & DEBUG_CMD)
+ printk("flushed %04X...", old_data);
+
+ udelay(10);
+ }
+
+ for (i = 0; !timed_out && i < nr_cmd; i++) {
+ int count;
+
+ for (count = 5000; count; count--)
+ if (inb(io_base + STATR) & CMD_WE)
+ break;
+
+ if (!count)
+ timed_out = 1;
+ else
+ outw(cmd[i], io_base + CMDR);
+ }
+
+ for (i = 0; !timed_out && i < nr_resp; i++) {
+ int count;
+
+ for (count = 5000; count; count--)
+ if (inb(io_base + STATR) & CMD_RF)
+ break;
+
+ if (!count)
+ timed_out = 1;
+ else
+ resp[i] = inw(io_base + CMDR);
+ }
+
+ if (debug_flg & DEBUG_CMD) {
+ if (!timed_out) {
+ printk("waveartist_cmd: resp=");
+
+ for (i = 0; i < nr_resp; i++)
+ printk("%04X ", resp[i]);
+
+ printk("\n");
+ } else
+ printk("waveartist_cmd: timed out\n");
+ }
+
+ return timed_out ? 1 : 0;
+}
+
+/*
+ * Send one command word
+ */
+static inline int
+waveartist_cmd1(wavnc_info *devc, unsigned int cmd)
+{
+ return waveartist_cmd(devc, 1, &cmd, 0, NULL);
+}
+
+/*
+ * Send one command, receive one word
+ */
+static inline unsigned int
+waveartist_cmd1_r(wavnc_info *devc, unsigned int cmd)
+{
+ unsigned int ret;
+
+ waveartist_cmd(devc, 1, &cmd, 1, &ret);
+
+ return ret;
+}
+
+/*
+ * Send a double command, receive one
+ * word (and throw it away)
+ */
+static inline int
+waveartist_cmd2(wavnc_info *devc, unsigned int cmd, unsigned int arg)
+{
+ unsigned int vals[2];
+
+ vals[0] = cmd;
+ vals[1] = arg;
+
+ return waveartist_cmd(devc, 2, vals, 1, vals);
+}
+
+/*
+ * Send a triple command
+ */
+static inline int
+waveartist_cmd3(wavnc_info *devc, unsigned int cmd,
+ unsigned int arg1, unsigned int arg2)
+{
+ unsigned int vals[3];
+
+ vals[0] = cmd;
+ vals[1] = arg1;
+ vals[2] = arg2;
+
+ return waveartist_cmd(devc, 3, vals, 0, NULL);
+}
+
+static int
+waveartist_getrev(wavnc_info *devc, char *rev)
+{
+ unsigned int temp[2];
+ unsigned int cmd = WACMD_GETREV;
+
+ waveartist_cmd(devc, 1, &cmd, 2, temp);
+
+ rev[0] = temp[0] >> 8;
+ rev[1] = temp[0] & 255;
+ rev[2] = '\0';
+
+ return temp[0];
+}
+
+static void waveartist_halt_output(int dev);
+static void waveartist_halt_input(int dev);
+static void waveartist_halt(int dev);
+static void waveartist_trigger(int dev, int state);
+
+static int
+waveartist_open(int dev, int mode)
+{
+ wavnc_info *devc;
+ wavnc_port_info *portc;
+ unsigned long flags;
+
+ if (dev < 0 || dev >= num_audiodevs)
+ return -ENXIO;
+
+ devc = (wavnc_info *) audio_devs[dev]->devc;
+ portc = (wavnc_port_info *) audio_devs[dev]->portc;
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+ if (portc->open_mode || (devc->open_mode & mode)) {
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+ return -EBUSY;
+ }
+
+ devc->audio_mode = 0;
+ devc->open_mode |= mode;
+ portc->open_mode = mode;
+ waveartist_trigger(dev, 0);
+
+ if (mode & OPEN_READ)
+ devc->record_dev = dev;
+ if (mode & OPEN_WRITE)
+ devc->playback_dev = dev;
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+
+ return 0;
+}
+
+static void
+waveartist_close(int dev)
+{
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+ unsigned long flags;
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ waveartist_halt(dev);
+
+ devc->audio_mode = 0;
+ devc->open_mode &= ~portc->open_mode;
+ portc->open_mode = 0;
+
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+}
+
+static void
+waveartist_output_block(int dev, unsigned long buf, int __count, int intrflag)
+{
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ unsigned long flags;
+ unsigned int count = __count;
+
+ if (debug_flg & DEBUG_OUT)
+ printk("waveartist: output block, buf=0x%lx, count=0x%x...\n",
+ buf, count);
+ /*
+ * 16 bit data
+ */
+ if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE))
+ count >>= 1;
+
+ if (portc->channels > 1)
+ count >>= 1;
+
+ count -= 1;
+
+ if (devc->audio_mode & PCM_ENABLE_OUTPUT &&
+ audio_devs[dev]->flags & DMA_AUTOMODE &&
+ intrflag &&
+ count == devc->xfer_count) {
+ devc->audio_mode |= PCM_ENABLE_OUTPUT;
+ return; /*
+ * Auto DMA mode on. No need to react
+ */
+ }
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ /*
+ * set sample count
+ */
+ waveartist_cmd2(devc, WACMD_OUTPUTSIZE, count);
+
+ devc->xfer_count = count;
+ devc->audio_mode |= PCM_ENABLE_OUTPUT;
+
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+}
+
+static void
+waveartist_start_input(int dev, unsigned long buf, int __count, int intrflag)
+{
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ unsigned long flags;
+ unsigned int count = __count;
+
+ if (debug_flg & DEBUG_IN)
+ printk("waveartist: start input, buf=0x%lx, count=0x%x...\n",
+ buf, count);
+
+ if (portc->audio_format & (AFMT_S16_LE | AFMT_S16_BE)) /* 16 bit data */
+ count >>= 1;
+
+ if (portc->channels > 1)
+ count >>= 1;
+
+ count -= 1;
+
+ if (devc->audio_mode & PCM_ENABLE_INPUT &&
+ audio_devs[dev]->flags & DMA_AUTOMODE &&
+ intrflag &&
+ count == devc->xfer_count) {
+ devc->audio_mode |= PCM_ENABLE_INPUT;
+ return; /*
+ * Auto DMA mode on. No need to react
+ */
+ }
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ /*
+ * set sample count
+ */
+ waveartist_cmd2(devc, WACMD_INPUTSIZE, count);
+
+ devc->xfer_count = count;
+ devc->audio_mode |= PCM_ENABLE_INPUT;
+
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+}
+
+static int
+waveartist_ioctl(int dev, unsigned int cmd, void __user * arg)
+{
+ return -EINVAL;
+}
+
+static unsigned int
+waveartist_get_speed(wavnc_port_info *portc)
+{
+ unsigned int speed;
+
+ /*
+ * program the speed, channels, bits
+ */
+ if (portc->speed == 8000)
+ speed = 0x2E71;
+ else if (portc->speed == 11025)
+ speed = 0x4000;
+ else if (portc->speed == 22050)
+ speed = 0x8000;
+ else if (portc->speed == 44100)
+ speed = 0x0;
+ else {
+ /*
+ * non-standard - just calculate
+ */
+ speed = portc->speed << 16;
+
+ speed = (speed / 44100) & 65535;
+ }
+
+ return speed;
+}
+
+static unsigned int
+waveartist_get_bits(wavnc_port_info *portc)
+{
+ unsigned int bits;
+
+ if (portc->audio_format == AFMT_S16_LE)
+ bits = 1;
+ else if (portc->audio_format == AFMT_S8)
+ bits = 0;
+ else
+ bits = 2; //default AFMT_U8
+
+ return bits;
+}
+
+static int
+waveartist_prepare_for_input(int dev, int bsize, int bcount)
+{
+ unsigned long flags;
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+ unsigned int speed, bits;
+
+ if (devc->audio_mode)
+ return 0;
+
+ speed = waveartist_get_speed(portc);
+ bits = waveartist_get_bits(portc);
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ if (waveartist_cmd2(devc, WACMD_INPUTFORMAT, bits))
+ printk(KERN_WARNING "waveartist: error setting the "
+ "record format to %d\n", portc->audio_format);
+
+ if (waveartist_cmd2(devc, WACMD_INPUTCHANNELS, portc->channels))
+ printk(KERN_WARNING "waveartist: error setting record "
+ "to %d channels\n", portc->channels);
+
+ /*
+ * write cmd SetSampleSpeedTimeConstant
+ */
+ if (waveartist_cmd2(devc, WACMD_INPUTSPEED, speed))
+ printk(KERN_WARNING "waveartist: error setting the record "
+ "speed to %dHz.\n", portc->speed);
+
+ if (waveartist_cmd2(devc, WACMD_INPUTDMA, 1))
+ printk(KERN_WARNING "waveartist: error setting the record "
+ "data path to 0x%X\n", 1);
+
+ if (waveartist_cmd2(devc, WACMD_INPUTFORMAT, bits))
+ printk(KERN_WARNING "waveartist: error setting the record "
+ "format to %d\n", portc->audio_format);
+
+ devc->xfer_count = 0;
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+ waveartist_halt_input(dev);
+
+ if (debug_flg & DEBUG_INTR) {
+ printk("WA CTLR reg: 0x%02X.\n",
+ inb(devc->hw.io_base + CTLR));
+ printk("WA STAT reg: 0x%02X.\n",
+ inb(devc->hw.io_base + STATR));
+ printk("WA IRQS reg: 0x%02X.\n",
+ inb(devc->hw.io_base + IRQSTAT));
+ }
+
+ return 0;
+}
+
+static int
+waveartist_prepare_for_output(int dev, int bsize, int bcount)
+{
+ unsigned long flags;
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+ unsigned int speed, bits;
+
+ /*
+ * program the speed, channels, bits
+ */
+ speed = waveartist_get_speed(portc);
+ bits = waveartist_get_bits(portc);
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ if (waveartist_cmd2(devc, WACMD_OUTPUTSPEED, speed) &&
+ waveartist_cmd2(devc, WACMD_OUTPUTSPEED, speed))
+ printk(KERN_WARNING "waveartist: error setting the playback "
+ "speed to %dHz.\n", portc->speed);
+
+ if (waveartist_cmd2(devc, WACMD_OUTPUTCHANNELS, portc->channels))
+ printk(KERN_WARNING "waveartist: error setting the playback "
+ "to %d channels\n", portc->channels);
+
+ if (waveartist_cmd2(devc, WACMD_OUTPUTDMA, 0))
+ printk(KERN_WARNING "waveartist: error setting the playback "
+ "data path to 0x%X\n", 0);
+
+ if (waveartist_cmd2(devc, WACMD_OUTPUTFORMAT, bits))
+ printk(KERN_WARNING "waveartist: error setting the playback "
+ "format to %d\n", portc->audio_format);
+
+ devc->xfer_count = 0;
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+ waveartist_halt_output(dev);
+
+ if (debug_flg & DEBUG_INTR) {
+ printk("WA CTLR reg: 0x%02X.\n",inb(devc->hw.io_base + CTLR));
+ printk("WA STAT reg: 0x%02X.\n",inb(devc->hw.io_base + STATR));
+ printk("WA IRQS reg: 0x%02X.\n",inb(devc->hw.io_base + IRQSTAT));
+ }
+
+ return 0;
+}
+
+static void
+waveartist_halt(int dev)
+{
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+ wavnc_info *devc;
+
+ if (portc->open_mode & OPEN_WRITE)
+ waveartist_halt_output(dev);
+
+ if (portc->open_mode & OPEN_READ)
+ waveartist_halt_input(dev);
+
+ devc = (wavnc_info *) audio_devs[dev]->devc;
+ devc->audio_mode = 0;
+}
+
+static void
+waveartist_halt_input(int dev)
+{
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ unsigned long flags;
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ /*
+ * Stop capture
+ */
+ waveartist_cmd1(devc, WACMD_INPUTSTOP);
+
+ devc->audio_mode &= ~PCM_ENABLE_INPUT;
+
+ /*
+ * Clear interrupt by toggling
+ * the IRQ_ACK bit in CTRL
+ */
+ if (inb(devc->hw.io_base + STATR) & IRQ_REQ)
+ waveartist_iack(devc);
+
+// devc->audio_mode &= ~PCM_ENABLE_INPUT;
+
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+}
+
+static void
+waveartist_halt_output(int dev)
+{
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ unsigned long flags;
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ waveartist_cmd1(devc, WACMD_OUTPUTSTOP);
+
+ devc->audio_mode &= ~PCM_ENABLE_OUTPUT;
+
+ /*
+ * Clear interrupt by toggling
+ * the IRQ_ACK bit in CTRL
+ */
+ if (inb(devc->hw.io_base + STATR) & IRQ_REQ)
+ waveartist_iack(devc);
+
+// devc->audio_mode &= ~PCM_ENABLE_OUTPUT;
+
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+}
+
+static void
+waveartist_trigger(int dev, int state)
+{
+ wavnc_info *devc = (wavnc_info *) audio_devs[dev]->devc;
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+ unsigned long flags;
+
+ if (debug_flg & DEBUG_TRIGGER) {
+ printk("wavnc: audio trigger ");
+ if (state & PCM_ENABLE_INPUT)
+ printk("in ");
+ if (state & PCM_ENABLE_OUTPUT)
+ printk("out");
+ printk("\n");
+ }
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ state &= devc->audio_mode;
+
+ if (portc->open_mode & OPEN_READ &&
+ state & PCM_ENABLE_INPUT)
+ /*
+ * enable ADC Data Transfer to PC
+ */
+ waveartist_cmd1(devc, WACMD_INPUTSTART);
+
+ if (portc->open_mode & OPEN_WRITE &&
+ state & PCM_ENABLE_OUTPUT)
+ /*
+ * enable DAC data transfer from PC
+ */
+ waveartist_cmd1(devc, WACMD_OUTPUTSTART);
+
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+}
+
+static int
+waveartist_set_speed(int dev, int arg)
+{
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+
+ if (arg <= 0)
+ return portc->speed;
+
+ if (arg < 5000)
+ arg = 5000;
+ if (arg > 44100)
+ arg = 44100;
+
+ portc->speed = arg;
+ return portc->speed;
+
+}
+
+static short
+waveartist_set_channels(int dev, short arg)
+{
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+
+ if (arg != 1 && arg != 2)
+ return portc->channels;
+
+ portc->channels = arg;
+ return arg;
+}
+
+static unsigned int
+waveartist_set_bits(int dev, unsigned int arg)
+{
+ wavnc_port_info *portc = (wavnc_port_info *) audio_devs[dev]->portc;
+
+ if (arg == 0)
+ return portc->audio_format;
+
+ if ((arg != AFMT_U8) && (arg != AFMT_S16_LE) && (arg != AFMT_S8))
+ arg = AFMT_U8;
+
+ portc->audio_format = arg;
+
+ return arg;
+}
+
+static struct audio_driver waveartist_audio_driver = {
+ .owner = THIS_MODULE,
+ .open = waveartist_open,
+ .close = waveartist_close,
+ .output_block = waveartist_output_block,
+ .start_input = waveartist_start_input,
+ .ioctl = waveartist_ioctl,
+ .prepare_for_input = waveartist_prepare_for_input,
+ .prepare_for_output = waveartist_prepare_for_output,
+ .halt_io = waveartist_halt,
+ .halt_input = waveartist_halt_input,
+ .halt_output = waveartist_halt_output,
+ .trigger = waveartist_trigger,
+ .set_speed = waveartist_set_speed,
+ .set_bits = waveartist_set_bits,
+ .set_channels = waveartist_set_channels
+};
+
+
+static irqreturn_t
+waveartist_intr(int irq, void *dev_id)
+{
+ wavnc_info *devc = dev_id;
+ int irqstatus, status;
+
+ spin_lock(&waveartist_lock);
+ irqstatus = inb(devc->hw.io_base + IRQSTAT);
+ status = inb(devc->hw.io_base + STATR);
+
+ if (debug_flg & DEBUG_INTR)
+ printk("waveartist_intr: stat=%02x, irqstat=%02x\n",
+ status, irqstatus);
+
+ if (status & IRQ_REQ) /* Clear interrupt */
+ waveartist_iack(devc);
+ else
+ printk(KERN_WARNING "waveartist: unexpected interrupt\n");
+
+ if (irqstatus & 0x01) {
+ int temp = 1;
+
+ /* PCM buffer done
+ */
+ if ((status & DMA0) && (devc->audio_mode & PCM_ENABLE_OUTPUT)) {
+ DMAbuf_outputintr(devc->playback_dev, 1);
+ temp = 0;
+ }
+ if ((status & DMA1) && (devc->audio_mode & PCM_ENABLE_INPUT)) {
+ DMAbuf_inputintr(devc->record_dev);
+ temp = 0;
+ }
+ if (temp) //default:
+ printk(KERN_WARNING "waveartist: Unknown interrupt\n");
+ }
+ if (irqstatus & 0x2)
+ // We do not use SB mode natively...
+ printk(KERN_WARNING "waveartist: Unexpected SB interrupt...\n");
+ spin_unlock(&waveartist_lock);
+ return IRQ_HANDLED;
+}
+
+/* -------------------------------------------------------------------------
+ * Mixer stuff
+ */
+struct mix_ent {
+ unsigned char reg_l;
+ unsigned char reg_r;
+ unsigned char shift;
+ unsigned char max;
+};
+
+static const struct mix_ent mix_devs[SOUND_MIXER_NRDEVICES] = {
+ { 2, 6, 1, 7 }, /* SOUND_MIXER_VOLUME */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_BASS */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_TREBLE */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_SYNTH */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_PCM */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_SPEAKER */
+ { 0, 4, 6, 31 }, /* SOUND_MIXER_LINE */
+ { 2, 6, 4, 3 }, /* SOUND_MIXER_MIC */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_CD */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_IMIX */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_ALTPCM */
+#if 0
+ { 3, 7, 0, 10 }, /* SOUND_MIXER_RECLEV */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_IGAIN */
+#else
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_RECLEV */
+ { 3, 7, 0, 7 }, /* SOUND_MIXER_IGAIN */
+#endif
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_OGAIN */
+ { 0, 4, 1, 31 }, /* SOUND_MIXER_LINE1 */
+ { 1, 5, 6, 31 }, /* SOUND_MIXER_LINE2 */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_LINE3 */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_DIGITAL1 */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_DIGITAL2 */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_DIGITAL3 */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_PHONEIN */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_PHONEOUT */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_VIDEO */
+ { 0, 0, 0, 0 }, /* SOUND_MIXER_RADIO */
+ { 0, 0, 0, 0 } /* SOUND_MIXER_MONITOR */
+};
+
+static void
+waveartist_mixer_update(wavnc_info *devc, int whichDev)
+{
+ unsigned int lev_left, lev_right;
+
+ lev_left = devc->levels[whichDev] & 0xff;
+ lev_right = devc->levels[whichDev] >> 8;
+
+ if (lev_left > 100)
+ lev_left = 100;
+ if (lev_right > 100)
+ lev_right = 100;
+
+#define SCALE(lev,max) ((lev) * (max) / 100)
+
+ if (machine_is_netwinder() && whichDev == SOUND_MIXER_PHONEOUT)
+ whichDev = SOUND_MIXER_VOLUME;
+
+ if (mix_devs[whichDev].reg_l || mix_devs[whichDev].reg_r) {
+ const struct mix_ent *mix = mix_devs + whichDev;
+ unsigned int mask, left, right;
+
+ mask = mix->max << mix->shift;
+ lev_left = SCALE(lev_left, mix->max) << mix->shift;
+ lev_right = SCALE(lev_right, mix->max) << mix->shift;
+
+ /* read left setting */
+ left = waveartist_cmd1_r(devc, WACMD_GET_LEVEL |
+ mix->reg_l << 8);
+
+ /* read right setting */
+ right = waveartist_cmd1_r(devc, WACMD_GET_LEVEL |
+ mix->reg_r << 8);
+
+ left = (left & ~mask) | (lev_left & mask);
+ right = (right & ~mask) | (lev_right & mask);
+
+ /* write left,right back */
+ waveartist_cmd3(devc, WACMD_SET_MIXER, left, right);
+ } else {
+ switch(whichDev) {
+ case SOUND_MIXER_PCM:
+ waveartist_cmd3(devc, WACMD_SET_LEVEL,
+ SCALE(lev_left, 32767),
+ SCALE(lev_right, 32767));
+ break;
+
+ case SOUND_MIXER_SYNTH:
+ waveartist_cmd3(devc, 0x0100 | WACMD_SET_LEVEL,
+ SCALE(lev_left, 32767),
+ SCALE(lev_right, 32767));
+ break;
+ }
+ }
+}
+
+/*
+ * Set the ADC MUX to the specified values. We do NOT do any
+ * checking of the values passed, since we assume that the
+ * relevant *_select_input function has done that for us.
+ */
+static void
+waveartist_set_adc_mux(wavnc_info *devc, char left_dev, char right_dev)
+{
+ unsigned int reg_08, reg_09;
+
+ reg_08 = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | 0x0800);
+ reg_09 = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | 0x0900);
+
+ reg_08 = (reg_08 & ~0x3f) | right_dev << 3 | left_dev;
+
+ waveartist_cmd3(devc, WACMD_SET_MIXER, reg_08, reg_09);
+}
+
+/*
+ * Decode a recording mask into a mixer selection as follows:
+ *
+ * OSS Source WA Source Actual source
+ * SOUND_MASK_IMIX Mixer Mixer output (same as AD1848)
+ * SOUND_MASK_LINE Line Line in
+ * SOUND_MASK_LINE1 Aux 1 Aux 1 in
+ * SOUND_MASK_LINE2 Aux 2 Aux 2 in
+ * SOUND_MASK_MIC Mic Microphone
+ */
+static unsigned int
+waveartist_select_input(wavnc_info *devc, unsigned int recmask,
+ unsigned char *dev_l, unsigned char *dev_r)
+{
+ unsigned int recdev = ADC_MUX_NONE;
+
+ if (recmask & SOUND_MASK_IMIX) {
+ recmask = SOUND_MASK_IMIX;
+ recdev = ADC_MUX_MIXER;
+ } else if (recmask & SOUND_MASK_LINE2) {
+ recmask = SOUND_MASK_LINE2;
+ recdev = ADC_MUX_AUX2;
+ } else if (recmask & SOUND_MASK_LINE1) {
+ recmask = SOUND_MASK_LINE1;
+ recdev = ADC_MUX_AUX1;
+ } else if (recmask & SOUND_MASK_LINE) {
+ recmask = SOUND_MASK_LINE;
+ recdev = ADC_MUX_LINE;
+ } else if (recmask & SOUND_MASK_MIC) {
+ recmask = SOUND_MASK_MIC;
+ recdev = ADC_MUX_MIC;
+ }
+
+ *dev_l = *dev_r = recdev;
+
+ return recmask;
+}
+
+static int
+waveartist_decode_mixer(wavnc_info *devc, int dev, unsigned char lev_l,
+ unsigned char lev_r)
+{
+ switch (dev) {
+ case SOUND_MIXER_VOLUME:
+ case SOUND_MIXER_SYNTH:
+ case SOUND_MIXER_PCM:
+ case SOUND_MIXER_LINE:
+ case SOUND_MIXER_MIC:
+ case SOUND_MIXER_IGAIN:
+ case SOUND_MIXER_LINE1:
+ case SOUND_MIXER_LINE2:
+ devc->levels[dev] = lev_l | lev_r << 8;
+ break;
+
+ case SOUND_MIXER_IMIX:
+ break;
+
+ default:
+ dev = -EINVAL;
+ break;
+ }
+
+ return dev;
+}
+
+static int waveartist_get_mixer(wavnc_info *devc, int dev)
+{
+ return devc->levels[dev];
+}
+
+static const struct waveartist_mixer_info waveartist_mixer = {
+ .supported_devs = SUPPORTED_MIXER_DEVICES | SOUND_MASK_IGAIN,
+ .recording_devs = SOUND_MASK_LINE | SOUND_MASK_MIC |
+ SOUND_MASK_LINE1 | SOUND_MASK_LINE2 |
+ SOUND_MASK_IMIX,
+ .stereo_devs = (SUPPORTED_MIXER_DEVICES | SOUND_MASK_IGAIN) & ~
+ (SOUND_MASK_SPEAKER | SOUND_MASK_IMIX),
+ .select_input = waveartist_select_input,
+ .decode_mixer = waveartist_decode_mixer,
+ .get_mixer = waveartist_get_mixer,
+};
+
+static void
+waveartist_set_recmask(wavnc_info *devc, unsigned int recmask)
+{
+ unsigned char dev_l, dev_r;
+
+ recmask &= devc->mix->recording_devs;
+
+ /*
+ * If more than one recording device selected,
+ * disable the device that is currently in use.
+ */
+ if (hweight32(recmask) > 1)
+ recmask &= ~devc->recmask;
+
+ /*
+ * Translate the recording device mask into
+ * the ADC multiplexer settings.
+ */
+ devc->recmask = devc->mix->select_input(devc, recmask,
+ &dev_l, &dev_r);
+
+ waveartist_set_adc_mux(devc, dev_l, dev_r);
+}
+
+static int
+waveartist_set_mixer(wavnc_info *devc, int dev, unsigned int level)
+{
+ unsigned int lev_left = level & 0x00ff;
+ unsigned int lev_right = (level & 0xff00) >> 8;
+
+ if (lev_left > 100)
+ lev_left = 100;
+ if (lev_right > 100)
+ lev_right = 100;
+
+ /*
+ * Mono devices have their right volume forced to their
+ * left volume. (from ALSA driver OSS emulation).
+ */
+ if (!(devc->mix->stereo_devs & (1 << dev)))
+ lev_right = lev_left;
+
+ dev = devc->mix->decode_mixer(devc, dev, lev_left, lev_right);
+
+ if (dev >= 0)
+ waveartist_mixer_update(devc, dev);
+
+ return dev < 0 ? dev : 0;
+}
+
+static int
+waveartist_mixer_ioctl(int dev, unsigned int cmd, void __user * arg)
+{
+ wavnc_info *devc = (wavnc_info *)audio_devs[dev]->devc;
+ int ret = 0, val, nr;
+
+ /*
+ * All SOUND_MIXER_* ioctls use type 'M'
+ */
+ if (((cmd >> 8) & 255) != 'M')
+ return -ENOIOCTLCMD;
+
+#ifdef CONFIG_ARCH_NETWINDER
+ if (machine_is_netwinder()) {
+ ret = vnc_private_ioctl(dev, cmd, arg);
+ if (ret != -ENOIOCTLCMD)
+ return ret;
+ else
+ ret = 0;
+ }
+#endif
+
+ nr = cmd & 0xff;
+
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE) {
+ if (get_user(val, (int __user *)arg))
+ return -EFAULT;
+
+ switch (nr) {
+ case SOUND_MIXER_RECSRC:
+ waveartist_set_recmask(devc, val);
+ break;
+
+ default:
+ ret = -EINVAL;
+ if (nr < SOUND_MIXER_NRDEVICES &&
+ devc->mix->supported_devs & (1 << nr))
+ ret = waveartist_set_mixer(devc, nr, val);
+ }
+ }
+
+ if (ret == 0 && _SIOC_DIR(cmd) & _SIOC_READ) {
+ ret = -EINVAL;
+
+ switch (nr) {
+ case SOUND_MIXER_RECSRC:
+ ret = devc->recmask;
+ break;
+
+ case SOUND_MIXER_DEVMASK:
+ ret = devc->mix->supported_devs;
+ break;
+
+ case SOUND_MIXER_STEREODEVS:
+ ret = devc->mix->stereo_devs;
+ break;
+
+ case SOUND_MIXER_RECMASK:
+ ret = devc->mix->recording_devs;
+ break;
+
+ case SOUND_MIXER_CAPS:
+ ret = SOUND_CAP_EXCL_INPUT;
+ break;
+
+ default:
+ if (nr < SOUND_MIXER_NRDEVICES)
+ ret = devc->mix->get_mixer(devc, nr);
+ break;
+ }
+
+ if (ret >= 0)
+ ret = put_user(ret, (int __user *)arg) ? -EFAULT : 0;
+ }
+
+ return ret;
+}
+
+static struct mixer_operations waveartist_mixer_operations =
+{
+ .owner = THIS_MODULE,
+ .id = "WaveArtist",
+ .name = "WaveArtist",
+ .ioctl = waveartist_mixer_ioctl
+};
+
+static void
+waveartist_mixer_reset(wavnc_info *devc)
+{
+ int i;
+
+ if (debug_flg & DEBUG_MIXER)
+ printk("%s: mixer_reset\n", devc->hw.name);
+
+ /*
+ * reset mixer cmd
+ */
+ waveartist_cmd1(devc, WACMD_RST_MIXER);
+
+ /*
+ * set input for ADC to come from 'quiet'
+ * turn on default modes
+ */
+ waveartist_cmd3(devc, WACMD_SET_MIXER, 0x9800, 0xa836);
+
+ /*
+ * set mixer input select to none, RX filter gains 0 dB
+ */
+ waveartist_cmd3(devc, WACMD_SET_MIXER, 0x4c00, 0x8c00);
+
+ /*
+ * set bit 0 reg 2 to 1 - unmute MonoOut
+ */
+ waveartist_cmd3(devc, WACMD_SET_MIXER, 0x2801, 0x6800);
+
+ /* set default input device = internal mic
+ * current recording device = none
+ */
+ waveartist_set_recmask(devc, 0);
+
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++)
+ waveartist_mixer_update(devc, i);
+}
+
+static int __init waveartist_init(wavnc_info *devc)
+{
+ wavnc_port_info *portc;
+ char rev[3], dev_name[64];
+ int my_dev;
+
+ if (waveartist_reset(devc))
+ return -ENODEV;
+
+ sprintf(dev_name, "%s (%s", devc->hw.name, devc->chip_name);
+
+ if (waveartist_getrev(devc, rev)) {
+ strcat(dev_name, " rev. ");
+ strcat(dev_name, rev);
+ }
+ strcat(dev_name, ")");
+
+ conf_printf2(dev_name, devc->hw.io_base, devc->hw.irq,
+ devc->hw.dma, devc->hw.dma2);
+
+ portc = kzalloc(sizeof(wavnc_port_info), GFP_KERNEL);
+ if (portc == NULL)
+ goto nomem;
+
+ my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, dev_name,
+ &waveartist_audio_driver, sizeof(struct audio_driver),
+ devc->audio_flags, AFMT_U8 | AFMT_S16_LE | AFMT_S8,
+ devc, devc->hw.dma, devc->hw.dma2);
+
+ if (my_dev < 0)
+ goto free;
+
+ audio_devs[my_dev]->portc = portc;
+
+ waveartist_mixer_reset(devc);
+
+ /*
+ * clear any pending interrupt
+ */
+ waveartist_iack(devc);
+
+ if (request_irq(devc->hw.irq, waveartist_intr, 0, devc->hw.name, devc) < 0) {
+ printk(KERN_ERR "%s: IRQ %d in use\n",
+ devc->hw.name, devc->hw.irq);
+ goto uninstall;
+ }
+
+ if (sound_alloc_dma(devc->hw.dma, devc->hw.name)) {
+ printk(KERN_ERR "%s: Can't allocate DMA%d\n",
+ devc->hw.name, devc->hw.dma);
+ goto uninstall_irq;
+ }
+
+ if (devc->hw.dma != devc->hw.dma2 && devc->hw.dma2 != NO_DMA)
+ if (sound_alloc_dma(devc->hw.dma2, devc->hw.name)) {
+ printk(KERN_ERR "%s: can't allocate DMA%d\n",
+ devc->hw.name, devc->hw.dma2);
+ goto uninstall_dma;
+ }
+
+ waveartist_set_ctlr(&devc->hw, 0, DMA1_IE | DMA0_IE);
+
+ audio_devs[my_dev]->mixer_dev =
+ sound_install_mixer(MIXER_DRIVER_VERSION,
+ dev_name,
+ &waveartist_mixer_operations,
+ sizeof(struct mixer_operations),
+ devc);
+
+ return my_dev;
+
+uninstall_dma:
+ sound_free_dma(devc->hw.dma);
+
+uninstall_irq:
+ free_irq(devc->hw.irq, devc);
+
+uninstall:
+ sound_unload_audiodev(my_dev);
+
+free:
+ kfree(portc);
+
+nomem:
+ return -1;
+}
+
+static int __init probe_waveartist(struct address_info *hw_config)
+{
+ wavnc_info *devc = &adev_info[nr_waveartist_devs];
+
+ if (nr_waveartist_devs >= MAX_AUDIO_DEV) {
+ printk(KERN_WARNING "waveartist: too many audio devices\n");
+ return 0;
+ }
+
+ if (!request_region(hw_config->io_base, 15, hw_config->name)) {
+ printk(KERN_WARNING "WaveArtist: I/O port conflict\n");
+ return 0;
+ }
+
+ if (hw_config->irq > 15 || hw_config->irq < 0) {
+ release_region(hw_config->io_base, 15);
+ printk(KERN_WARNING "WaveArtist: Bad IRQ %d\n",
+ hw_config->irq);
+ return 0;
+ }
+
+ if (hw_config->dma != 3) {
+ release_region(hw_config->io_base, 15);
+ printk(KERN_WARNING "WaveArtist: Bad DMA %d\n",
+ hw_config->dma);
+ return 0;
+ }
+
+ hw_config->name = "WaveArtist";
+ devc->hw = *hw_config;
+ devc->open_mode = 0;
+ devc->chip_name = "RWA-010";
+
+ return 1;
+}
+
+static void __init
+attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *mix)
+{
+ wavnc_info *devc = &adev_info[nr_waveartist_devs];
+
+ /*
+ * NOTE! If irq < 0, there is another driver which has allocated the
+ * IRQ so that this driver doesn't need to allocate/deallocate it.
+ * The actually used IRQ is ABS(irq).
+ */
+ devc->hw = *hw;
+ devc->hw.irq = (hw->irq > 0) ? hw->irq : 0;
+ devc->open_mode = 0;
+ devc->playback_dev = 0;
+ devc->record_dev = 0;
+ devc->audio_flags = DMA_AUTOMODE;
+ devc->levels = levels;
+
+ if (hw->dma != hw->dma2 && hw->dma2 != NO_DMA)
+ devc->audio_flags |= DMA_DUPLEX;
+
+ devc->mix = mix;
+ devc->dev_no = waveartist_init(devc);
+
+ if (devc->dev_no < 0)
+ release_region(hw->io_base, 15);
+ else {
+#ifdef CONFIG_ARCH_NETWINDER
+ if (machine_is_netwinder()) {
+ init_timer(&vnc_timer);
+ vnc_timer.function = vnc_slider_tick;
+ vnc_timer.expires = jiffies;
+ vnc_timer.data = nr_waveartist_devs;
+ add_timer(&vnc_timer);
+
+ vnc_configure_mixer(devc, 0);
+
+ devc->no_autoselect = 1;
+ }
+#endif
+ nr_waveartist_devs += 1;
+ }
+}
+
+static void __exit unload_waveartist(struct address_info *hw)
+{
+ wavnc_info *devc = NULL;
+ int i;
+
+ for (i = 0; i < nr_waveartist_devs; i++)
+ if (hw->io_base == adev_info[i].hw.io_base) {
+ devc = adev_info + i;
+ break;
+ }
+
+ if (devc != NULL) {
+ int mixer;
+
+#ifdef CONFIG_ARCH_NETWINDER
+ if (machine_is_netwinder())
+ del_timer(&vnc_timer);
+#endif
+
+ release_region(devc->hw.io_base, 15);
+
+ waveartist_set_ctlr(&devc->hw, DMA1_IE|DMA0_IE, 0);
+
+ if (devc->hw.irq >= 0)
+ free_irq(devc->hw.irq, devc);
+
+ sound_free_dma(devc->hw.dma);
+
+ if (devc->hw.dma != devc->hw.dma2 &&
+ devc->hw.dma2 != NO_DMA)
+ sound_free_dma(devc->hw.dma2);
+
+ mixer = audio_devs[devc->dev_no]->mixer_dev;
+
+ if (mixer >= 0)
+ sound_unload_mixerdev(mixer);
+
+ if (devc->dev_no >= 0)
+ sound_unload_audiodev(devc->dev_no);
+
+ nr_waveartist_devs -= 1;
+
+ for (; i < nr_waveartist_devs; i++)
+ adev_info[i] = adev_info[i + 1];
+ } else
+ printk(KERN_WARNING "waveartist: can't find device "
+ "to unload\n");
+}
+
+#ifdef CONFIG_ARCH_NETWINDER
+
+/*
+ * Rebel.com Netwinder specifics...
+ */
+
+#include <asm/hardware/dec21285.h>
+
+#define VNC_TIMER_PERIOD (HZ/4) //check slider 4 times/sec
+
+#define MIXER_PRIVATE3_RESET 0x53570000
+#define MIXER_PRIVATE3_READ 0x53570001
+#define MIXER_PRIVATE3_WRITE 0x53570002
+
+#define VNC_MUTE_INTERNAL_SPKR 0x01 //the sw mute on/off control bit
+#define VNC_MUTE_LINE_OUT 0x10
+#define VNC_PHONE_DETECT 0x20
+#define VNC_HANDSET_DETECT 0x40
+#define VNC_DISABLE_AUTOSWITCH 0x80
+
+extern spinlock_t gpio_lock;
+
+static inline void
+vnc_mute_spkr(wavnc_info *devc)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&gpio_lock, flags);
+ cpld_modify(CPLD_UNMUTE, devc->spkr_mute_state ? 0 : CPLD_UNMUTE);
+ spin_unlock_irqrestore(&gpio_lock, flags);
+}
+
+static void
+vnc_mute_lout(wavnc_info *devc)
+{
+ unsigned int left, right;
+
+ left = waveartist_cmd1_r(devc, WACMD_GET_LEVEL);
+ right = waveartist_cmd1_r(devc, WACMD_GET_LEVEL | 0x400);
+
+ if (devc->line_mute_state) {
+ left &= ~1;
+ right &= ~1;
+ } else {
+ left |= 1;
+ right |= 1;
+ }
+ waveartist_cmd3(devc, WACMD_SET_MIXER, left, right);
+
+}
+
+static int
+vnc_volume_slider(wavnc_info *devc)
+{
+ static signed int old_slider_volume;
+ unsigned long flags;
+ signed int volume = 255;
+
+ *CSR_TIMER1_LOAD = 0x00ffffff;
+
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ outb(0xFF, 0x201);
+ *CSR_TIMER1_CNTL = TIMER_CNTL_ENABLE | TIMER_CNTL_DIV1;
+
+ while (volume && (inb(0x201) & 0x01))
+ volume--;
+
+ *CSR_TIMER1_CNTL = 0;
+
+ spin_unlock_irqrestore(&waveartist_lock,flags);
+
+ volume = 0x00ffffff - *CSR_TIMER1_VALUE;
+
+
+#ifndef REVERSE
+ volume = 150 - (volume >> 5);
+#else
+ volume = (volume >> 6) - 25;
+#endif
+
+ if (volume < 0)
+ volume = 0;
+
+ if (volume > 100)
+ volume = 100;
+
+ /*
+ * slider quite often reads +-8, so debounce this random noise
+ */
+ if (abs(volume - old_slider_volume) > 7) {
+ old_slider_volume = volume;
+
+ if (debug_flg & DEBUG_MIXER)
+ printk(KERN_DEBUG "Slider volume: %d.\n", volume);
+ }
+
+ return old_slider_volume;
+}
+
+/*
+ * Decode a recording mask into a mixer selection on the NetWinder
+ * as follows:
+ *
+ * OSS Source WA Source Actual source
+ * SOUND_MASK_IMIX Mixer Mixer output (same as AD1848)
+ * SOUND_MASK_LINE Line Line in
+ * SOUND_MASK_LINE1 Left Mic Handset
+ * SOUND_MASK_PHONEIN Left Aux Telephone microphone
+ * SOUND_MASK_MIC Right Mic Builtin microphone
+ */
+static unsigned int
+netwinder_select_input(wavnc_info *devc, unsigned int recmask,
+ unsigned char *dev_l, unsigned char *dev_r)
+{
+ unsigned int recdev_l = ADC_MUX_NONE, recdev_r = ADC_MUX_NONE;
+
+ if (recmask & SOUND_MASK_IMIX) {
+ recmask = SOUND_MASK_IMIX;
+ recdev_l = ADC_MUX_MIXER;
+ recdev_r = ADC_MUX_MIXER;
+ } else if (recmask & SOUND_MASK_LINE) {
+ recmask = SOUND_MASK_LINE;
+ recdev_l = ADC_MUX_LINE;
+ recdev_r = ADC_MUX_LINE;
+ } else if (recmask & SOUND_MASK_LINE1) {
+ recmask = SOUND_MASK_LINE1;
+ waveartist_cmd1(devc, WACMD_SET_MONO); /* left */
+ recdev_l = ADC_MUX_MIC;
+ recdev_r = ADC_MUX_NONE;
+ } else if (recmask & SOUND_MASK_PHONEIN) {
+ recmask = SOUND_MASK_PHONEIN;
+ waveartist_cmd1(devc, WACMD_SET_MONO); /* left */
+ recdev_l = ADC_MUX_AUX1;
+ recdev_r = ADC_MUX_NONE;
+ } else if (recmask & SOUND_MASK_MIC) {
+ recmask = SOUND_MASK_MIC;
+ waveartist_cmd1(devc, WACMD_SET_MONO | 0x100); /* right */
+ recdev_l = ADC_MUX_NONE;
+ recdev_r = ADC_MUX_MIC;
+ }
+
+ *dev_l = recdev_l;
+ *dev_r = recdev_r;
+
+ return recmask;
+}
+
+static int
+netwinder_decode_mixer(wavnc_info *devc, int dev, unsigned char lev_l,
+ unsigned char lev_r)
+{
+ switch (dev) {
+ case SOUND_MIXER_VOLUME:
+ case SOUND_MIXER_SYNTH:
+ case SOUND_MIXER_PCM:
+ case SOUND_MIXER_LINE:
+ case SOUND_MIXER_IGAIN:
+ devc->levels[dev] = lev_l | lev_r << 8;
+ break;
+
+ case SOUND_MIXER_MIC: /* right mic only */
+ devc->levels[SOUND_MIXER_MIC] &= 0xff;
+ devc->levels[SOUND_MIXER_MIC] |= lev_l << 8;
+ break;
+
+ case SOUND_MIXER_LINE1: /* left mic only */
+ devc->levels[SOUND_MIXER_MIC] &= 0xff00;
+ devc->levels[SOUND_MIXER_MIC] |= lev_l;
+ dev = SOUND_MIXER_MIC;
+ break;
+
+ case SOUND_MIXER_PHONEIN: /* left aux only */
+ devc->levels[SOUND_MIXER_LINE1] = lev_l;
+ dev = SOUND_MIXER_LINE1;
+ break;
+
+ case SOUND_MIXER_IMIX:
+ case SOUND_MIXER_PHONEOUT:
+ break;
+
+ default:
+ dev = -EINVAL;
+ break;
+ }
+ return dev;
+}
+
+static int netwinder_get_mixer(wavnc_info *devc, int dev)
+{
+ int levels;
+
+ switch (dev) {
+ case SOUND_MIXER_VOLUME:
+ case SOUND_MIXER_SYNTH:
+ case SOUND_MIXER_PCM:
+ case SOUND_MIXER_LINE:
+ case SOUND_MIXER_IGAIN:
+ levels = devc->levels[dev];
+ break;
+
+ case SOUND_MIXER_MIC: /* builtin mic: right mic only */
+ levels = devc->levels[SOUND_MIXER_MIC] >> 8;
+ levels |= levels << 8;
+ break;
+
+ case SOUND_MIXER_LINE1: /* handset mic: left mic only */
+ levels = devc->levels[SOUND_MIXER_MIC] & 0xff;
+ levels |= levels << 8;
+ break;
+
+ case SOUND_MIXER_PHONEIN: /* phone mic: left aux1 only */
+ levels = devc->levels[SOUND_MIXER_LINE1] & 0xff;
+ levels |= levels << 8;
+ break;
+
+ default:
+ levels = 0;
+ }
+
+ return levels;
+}
+
+/*
+ * Waveartist specific mixer information.
+ */
+static const struct waveartist_mixer_info netwinder_mixer = {
+ .supported_devs = SOUND_MASK_VOLUME | SOUND_MASK_SYNTH |
+ SOUND_MASK_PCM | SOUND_MASK_SPEAKER |
+ SOUND_MASK_LINE | SOUND_MASK_MIC |
+ SOUND_MASK_IMIX | SOUND_MASK_LINE1 |
+ SOUND_MASK_PHONEIN | SOUND_MASK_PHONEOUT|
+ SOUND_MASK_IGAIN,
+
+ .recording_devs = SOUND_MASK_LINE | SOUND_MASK_MIC |
+ SOUND_MASK_IMIX | SOUND_MASK_LINE1 |
+ SOUND_MASK_PHONEIN,
+
+ .stereo_devs = SOUND_MASK_VOLUME | SOUND_MASK_SYNTH |
+ SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_IMIX | SOUND_MASK_IGAIN,
+
+ .select_input = netwinder_select_input,
+ .decode_mixer = netwinder_decode_mixer,
+ .get_mixer = netwinder_get_mixer,
+};
+
+static void
+vnc_configure_mixer(wavnc_info *devc, unsigned int recmask)
+{
+ if (!devc->no_autoselect) {
+ if (devc->handset_detect) {
+ recmask = SOUND_MASK_LINE1;
+ devc->spkr_mute_state = devc->line_mute_state = 1;
+ } else if (devc->telephone_detect) {
+ recmask = SOUND_MASK_PHONEIN;
+ devc->spkr_mute_state = devc->line_mute_state = 1;
+ } else {
+ /* unless someone has asked for LINE-IN,
+ * we default to MIC
+ */
+ if ((devc->recmask & SOUND_MASK_LINE) == 0)
+ devc->recmask = SOUND_MASK_MIC;
+ devc->spkr_mute_state = devc->line_mute_state = 0;
+ }
+ vnc_mute_spkr(devc);
+ vnc_mute_lout(devc);
+
+ if (recmask != devc->recmask)
+ waveartist_set_recmask(devc, recmask);
+ }
+}
+
+static int
+vnc_slider(wavnc_info *devc)
+{
+ signed int slider_volume;
+ unsigned int temp, old_hs, old_td;
+
+ /*
+ * read the "buttons" state.
+ * Bit 4 = 0 means handset present
+ * Bit 5 = 1 means phone offhook
+ */
+ temp = inb(0x201);
+
+ old_hs = devc->handset_detect;
+ old_td = devc->telephone_detect;
+
+ devc->handset_detect = !(temp & 0x10);
+ devc->telephone_detect = !!(temp & 0x20);
+
+ if (!devc->no_autoselect &&
+ (old_hs != devc->handset_detect ||
+ old_td != devc->telephone_detect))
+ vnc_configure_mixer(devc, devc->recmask);
+
+ slider_volume = vnc_volume_slider(devc);
+
+ /*
+ * If we're using software controlled volume, and
+ * the slider moves by more than 20%, then we
+ * switch back to slider controlled volume.
+ */
+ if (abs(devc->slider_vol - slider_volume) > 20)
+ devc->use_slider = 1;
+
+ /*
+ * use only left channel
+ */
+ temp = levels[SOUND_MIXER_VOLUME] & 0xFF;
+
+ if (slider_volume != temp && devc->use_slider) {
+ devc->slider_vol = slider_volume;
+
+ waveartist_set_mixer(devc, SOUND_MIXER_VOLUME,
+ slider_volume | slider_volume << 8);
+
+ return 1;
+ }
+
+ return 0;
+}
+
+static void
+vnc_slider_tick(unsigned long data)
+{
+ int next_timeout;
+
+ if (vnc_slider(adev_info + data))
+ next_timeout = 5; // mixer reported change
+ else
+ next_timeout = VNC_TIMER_PERIOD;
+
+ mod_timer(&vnc_timer, jiffies + next_timeout);
+}
+
+static int
+vnc_private_ioctl(int dev, unsigned int cmd, int __user * arg)
+{
+ wavnc_info *devc = (wavnc_info *)audio_devs[dev]->devc;
+ int val;
+
+ switch (cmd) {
+ case SOUND_MIXER_PRIVATE1:
+ {
+ u_int prev_spkr_mute, prev_line_mute, prev_auto_state;
+ int val;
+
+ if (get_user(val, arg))
+ return -EFAULT;
+
+ /* check if parameter is logical */
+ if (val & ~(VNC_MUTE_INTERNAL_SPKR |
+ VNC_MUTE_LINE_OUT |
+ VNC_DISABLE_AUTOSWITCH))
+ return -EINVAL;
+
+ prev_auto_state = devc->no_autoselect;
+ prev_spkr_mute = devc->spkr_mute_state;
+ prev_line_mute = devc->line_mute_state;
+
+ devc->no_autoselect = (val & VNC_DISABLE_AUTOSWITCH) ? 1 : 0;
+ devc->spkr_mute_state = (val & VNC_MUTE_INTERNAL_SPKR) ? 1 : 0;
+ devc->line_mute_state = (val & VNC_MUTE_LINE_OUT) ? 1 : 0;
+
+ if (prev_spkr_mute != devc->spkr_mute_state)
+ vnc_mute_spkr(devc);
+
+ if (prev_line_mute != devc->line_mute_state)
+ vnc_mute_lout(devc);
+
+ if (prev_auto_state != devc->no_autoselect)
+ vnc_configure_mixer(devc, devc->recmask);
+
+ return 0;
+ }
+
+ case SOUND_MIXER_PRIVATE2:
+ if (get_user(val, arg))
+ return -EFAULT;
+
+ switch (val) {
+#define VNC_SOUND_PAUSE 0x53 //to pause the DSP
+#define VNC_SOUND_RESUME 0x57 //to unpause the DSP
+ case VNC_SOUND_PAUSE:
+ waveartist_cmd1(devc, 0x16);
+ break;
+
+ case VNC_SOUND_RESUME:
+ waveartist_cmd1(devc, 0x18);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+
+ /* private ioctl to allow bulk access to waveartist */
+ case SOUND_MIXER_PRIVATE3:
+ {
+ unsigned long flags;
+ int mixer_reg[15], i, val;
+
+ if (get_user(val, arg))
+ return -EFAULT;
+ if (copy_from_user(mixer_reg, (void *)val, sizeof(mixer_reg)))
+ return -EFAULT;
+
+ switch (mixer_reg[14]) {
+ case MIXER_PRIVATE3_RESET:
+ waveartist_mixer_reset(devc);
+ break;
+
+ case MIXER_PRIVATE3_WRITE:
+ waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[0], mixer_reg[4]);
+ waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[1], mixer_reg[5]);
+ waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[2], mixer_reg[6]);
+ waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[3], mixer_reg[7]);
+ waveartist_cmd3(devc, WACMD_SET_MIXER, mixer_reg[8], mixer_reg[9]);
+
+ waveartist_cmd3(devc, WACMD_SET_LEVEL, mixer_reg[10], mixer_reg[11]);
+ waveartist_cmd3(devc, WACMD_SET_LEVEL, mixer_reg[12], mixer_reg[13]);
+ break;
+
+ case MIXER_PRIVATE3_READ:
+ spin_lock_irqsave(&waveartist_lock, flags);
+
+ for (i = 0x30; i < 14 << 8; i += 1 << 8)
+ waveartist_cmd(devc, 1, &i, 1, mixer_reg + (i >> 8));
+
+ spin_unlock_irqrestore(&waveartist_lock, flags);
+
+ if (copy_to_user((void *)val, mixer_reg, sizeof(mixer_reg)))
+ return -EFAULT;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+ }
+
+ /* read back the state from PRIVATE1 */
+ case SOUND_MIXER_PRIVATE4:
+ val = (devc->spkr_mute_state ? VNC_MUTE_INTERNAL_SPKR : 0) |
+ (devc->line_mute_state ? VNC_MUTE_LINE_OUT : 0) |
+ (devc->handset_detect ? VNC_HANDSET_DETECT : 0) |
+ (devc->telephone_detect ? VNC_PHONE_DETECT : 0) |
+ (devc->no_autoselect ? VNC_DISABLE_AUTOSWITCH : 0);
+
+ return put_user(val, arg) ? -EFAULT : 0;
+ }
+
+ if (_SIOC_DIR(cmd) & _SIOC_WRITE) {
+ /*
+ * special case for master volume: if we
+ * received this call - switch from hw
+ * volume control to a software volume
+ * control, till the hw volume is modified
+ * to signal that user wants to be back in
+ * hardware...
+ */
+ if ((cmd & 0xff) == SOUND_MIXER_VOLUME)
+ devc->use_slider = 0;
+
+ /* speaker output */
+ if ((cmd & 0xff) == SOUND_MIXER_SPEAKER) {
+ unsigned int val, l, r;
+
+ if (get_user(val, arg))
+ return -EFAULT;
+
+ l = val & 0x7f;
+ r = (val & 0x7f00) >> 8;
+ val = (l + r) / 2;
+ devc->levels[SOUND_MIXER_SPEAKER] = val | (val << 8);
+ devc->spkr_mute_state = (val <= 50);
+ vnc_mute_spkr(devc);
+ return 0;
+ }
+ }
+
+ return -ENOIOCTLCMD;
+}
+
+#endif
+
+static struct address_info cfg;
+
+static int attached;
+
+static int __initdata io = 0;
+static int __initdata irq = 0;
+static int __initdata dma = 0;
+static int __initdata dma2 = 0;
+
+
+static int __init init_waveartist(void)
+{
+ const struct waveartist_mixer_info *mix;
+
+ if (!io && machine_is_netwinder()) {
+ /*
+ * The NetWinder WaveArtist is at a fixed address.
+ * If the user does not supply an address, use the
+ * well-known parameters.
+ */
+ io = 0x250;
+ irq = 12;
+ dma = 3;
+ dma2 = 7;
+ }
+
+ mix = &waveartist_mixer;
+#ifdef CONFIG_ARCH_NETWINDER
+ if (machine_is_netwinder())
+ mix = &netwinder_mixer;
+#endif
+
+ cfg.io_base = io;
+ cfg.irq = irq;
+ cfg.dma = dma;
+ cfg.dma2 = dma2;
+
+ if (!probe_waveartist(&cfg))
+ return -ENODEV;
+
+ attach_waveartist(&cfg, mix);
+ attached = 1;
+
+ return 0;
+}
+
+static void __exit cleanup_waveartist(void)
+{
+ if (attached)
+ unload_waveartist(&cfg);
+}
+
+module_init(init_waveartist);
+module_exit(cleanup_waveartist);
+
+#ifndef MODULE
+static int __init setup_waveartist(char *str)
+{
+ /* io, irq, dma, dma2 */
+ int ints[5];
+
+ str = get_options(str, ARRAY_SIZE(ints), ints);
+
+ io = ints[1];
+ irq = ints[2];
+ dma = ints[3];
+ dma2 = ints[4];
+
+ return 1;
+}
+__setup("waveartist=", setup_waveartist);
+#endif
+
+MODULE_DESCRIPTION("Rockwell WaveArtist RWA-010 sound driver");
+module_param(io, int, 0); /* IO base */
+module_param(irq, int, 0); /* IRQ */
+module_param(dma, int, 0); /* DMA */
+module_param(dma2, int, 0); /* DMA2 */
+MODULE_LICENSE("GPL");
diff --git a/sound/oss/waveartist.h b/sound/oss/waveartist.h
new file mode 100644
index 0000000..dac4ca9
--- /dev/null
+++ b/sound/oss/waveartist.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/oss/waveartist.h
+ *
+ * def file for Rockwell RWA010 chip set, as installed in Rebel.com NetWinder
+ */
+
+//registers
+#define CMDR 0
+#define DATR 2
+#define CTLR 4
+#define STATR 5
+#define IRQSTAT 12
+
+//bit defs
+//reg STATR
+#define CMD_WE 0x80
+#define CMD_RF 0x40
+#define DAT_WE 0x20
+#define DAT_RF 0x10
+
+#define IRQ_REQ 0x08
+#define DMA1 0x04
+#define DMA0 0x02
+
+//bit defs
+//reg CTLR
+#define CMD_WEIE 0x80
+#define CMD_RFIE 0x40
+#define DAT_WEIE 0x20
+#define DAT_RFIE 0x10
+
+#define RESET 0x08
+#define DMA1_IE 0x04
+#define DMA0_IE 0x02
+#define IRQ_ACK 0x01
+
+//commands
+
+#define WACMD_SYSTEMID 0x00
+#define WACMD_GETREV 0x00
+#define WACMD_INPUTFORMAT 0x10 //0-8S, 1-16S, 2-8U
+#define WACMD_INPUTCHANNELS 0x11 //1-Mono, 2-Stereo
+#define WACMD_INPUTSPEED 0x12 //sampling rate
+#define WACMD_INPUTDMA 0x13 //0-8bit, 1-16bit, 2-PIO
+#define WACMD_INPUTSIZE 0x14 //samples to interrupt
+#define WACMD_INPUTSTART 0x15 //start ADC
+#define WACMD_INPUTPAUSE 0x16 //pause ADC
+#define WACMD_INPUTSTOP 0x17 //stop ADC
+#define WACMD_INPUTRESUME 0x18 //resume ADC
+#define WACMD_INPUTPIO 0x19 //PIO ADC
+
+#define WACMD_OUTPUTFORMAT 0x20 //0-8S, 1-16S, 2-8U
+#define WACMD_OUTPUTCHANNELS 0x21 //1-Mono, 2-Stereo
+#define WACMD_OUTPUTSPEED 0x22 //sampling rate
+#define WACMD_OUTPUTDMA 0x23 //0-8bit, 1-16bit, 2-PIO
+#define WACMD_OUTPUTSIZE 0x24 //samples to interrupt
+#define WACMD_OUTPUTSTART 0x25 //start ADC
+#define WACMD_OUTPUTPAUSE 0x26 //pause ADC
+#define WACMD_OUTPUTSTOP 0x27 //stop ADC
+#define WACMD_OUTPUTRESUME 0x28 //resume ADC
+#define WACMD_OUTPUTPIO 0x29 //PIO ADC
+
+#define WACMD_GET_LEVEL 0x30
+#define WACMD_SET_LEVEL 0x31
+#define WACMD_SET_MIXER 0x32
+#define WACMD_RST_MIXER 0x33
+#define WACMD_SET_MONO 0x34
+
+/*
+ * Definitions for left/right recording input mux
+ */
+#define ADC_MUX_NONE 0
+#define ADC_MUX_MIXER 1
+#define ADC_MUX_LINE 2
+#define ADC_MUX_AUX2 3
+#define ADC_MUX_AUX1 4
+#define ADC_MUX_MIC 5
+
+/*
+ * Definitions for mixer gain settings
+ */
+#define MIX_GAIN_LINE 0 /* line in */
+#define MIX_GAIN_AUX1 1 /* aux1 */
+#define MIX_GAIN_AUX2 2 /* aux2 */
+#define MIX_GAIN_XMIC 3 /* crossover mic */
+#define MIX_GAIN_MIC 4 /* normal mic */
+#define MIX_GAIN_PREMIC 5 /* preamp mic */
+#define MIX_GAIN_OUT 6 /* output */
+#define MIX_GAIN_MONO 7 /* mono in */
+
+int wa_sendcmd(unsigned int cmd);
+int wa_writecmd(unsigned int cmd, unsigned int arg);
diff --git a/sound/parisc/Kconfig b/sound/parisc/Kconfig
new file mode 100644
index 0000000..9b61d95
--- /dev/null
+++ b/sound/parisc/Kconfig
@@ -0,0 +1,20 @@
+# ALSA PA-RISC drivers
+
+menuconfig SND_GSC
+ bool "GSC sound devices"
+ depends on GSC
+ default y
+ help
+ Support for GSC sound devices on PA-RISC architectures.
+
+if SND_GSC
+
+config SND_HARMONY
+ tristate "Harmony/Vivace sound chip"
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for the Harmony/Vivace sound
+ chip found in most GSC-based PA-RISC workstations. It's frequently
+ provided as part of the Lasi multi-function IC.
+
+endif # SND_GSC
diff --git a/sound/parisc/Makefile b/sound/parisc/Makefile
new file mode 100644
index 0000000..b91e750
--- /dev/null
+++ b/sound/parisc/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+#
+
+snd-harmony-objs := harmony.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_HARMONY) += snd-harmony.o
diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c
new file mode 100644
index 0000000..41f870f
--- /dev/null
+++ b/sound/parisc/harmony.c
@@ -0,0 +1,1043 @@
+/* Hewlett-Packard Harmony audio driver
+ *
+ * This is a driver for the Harmony audio chipset found
+ * on the LASI ASIC of various early HP PA-RISC workstations.
+ *
+ * Copyright (C) 2004, Kyle McMartin <kyle@{debian.org,parisc-linux.org}>
+ *
+ * Based on the previous Harmony incarnations by,
+ * Copyright 2000 (c) Linuxcare Canada, Alex deVries
+ * Copyright 2000-2003 (c) Helge Deller
+ * Copyright 2001 (c) Matthieu Delahaye
+ * Copyright 2001 (c) Jean-Christophe Vaugeois
+ * Copyright 2003 (c) Laurent Canet
+ * Copyright 2004 (c) Stuart Brady
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Notes:
+ * - graveyard and silence buffers last for lifetime of
+ * the driver. playback and capture buffers are allocated
+ * per _open()/_close().
+ *
+ * TODO:
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/rawmidi.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+
+#include <asm/io.h>
+#include <asm/hardware.h>
+#include <asm/parisc-device.h>
+
+#include "harmony.h"
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for Harmony driver.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for Harmony driver.");
+
+
+static struct parisc_device_id snd_harmony_devtable[] = {
+ /* bushmaster / flounder */
+ { HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007A },
+ /* 712 / 715 */
+ { HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007B },
+ /* pace */
+ { HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007E },
+ /* outfield / coral II */
+ { HPHW_FIO, HVERSION_REV_ANY_ID, HVERSION_ANY_ID, 0x0007F },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(parisc, snd_harmony_devtable);
+
+#define NAME "harmony"
+#define PFX NAME ": "
+
+static unsigned int snd_harmony_rates[] = {
+ 5512, 6615, 8000, 9600,
+ 11025, 16000, 18900, 22050,
+ 27428, 32000, 33075, 37800,
+ 44100, 48000
+};
+
+static unsigned int rate_bits[14] = {
+ HARMONY_SR_5KHZ, HARMONY_SR_6KHZ, HARMONY_SR_8KHZ,
+ HARMONY_SR_9KHZ, HARMONY_SR_11KHZ, HARMONY_SR_16KHZ,
+ HARMONY_SR_18KHZ, HARMONY_SR_22KHZ, HARMONY_SR_27KHZ,
+ HARMONY_SR_32KHZ, HARMONY_SR_33KHZ, HARMONY_SR_37KHZ,
+ HARMONY_SR_44KHZ, HARMONY_SR_48KHZ
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraint_rates = {
+ .count = ARRAY_SIZE(snd_harmony_rates),
+ .list = snd_harmony_rates,
+ .mask = 0,
+};
+
+static inline unsigned long
+harmony_read(struct snd_harmony *h, unsigned r)
+{
+ return __raw_readl(h->iobase + r);
+}
+
+static inline void
+harmony_write(struct snd_harmony *h, unsigned r, unsigned long v)
+{
+ __raw_writel(v, h->iobase + r);
+}
+
+static inline void
+harmony_wait_for_control(struct snd_harmony *h)
+{
+ while (harmony_read(h, HARMONY_CNTL) & HARMONY_CNTL_C) ;
+}
+
+static inline void
+harmony_reset(struct snd_harmony *h)
+{
+ harmony_write(h, HARMONY_RESET, 1);
+ mdelay(50);
+ harmony_write(h, HARMONY_RESET, 0);
+}
+
+static void
+harmony_disable_interrupts(struct snd_harmony *h)
+{
+ u32 dstatus;
+ harmony_wait_for_control(h);
+ dstatus = harmony_read(h, HARMONY_DSTATUS);
+ dstatus &= ~HARMONY_DSTATUS_IE;
+ harmony_write(h, HARMONY_DSTATUS, dstatus);
+}
+
+static void
+harmony_enable_interrupts(struct snd_harmony *h)
+{
+ u32 dstatus;
+ harmony_wait_for_control(h);
+ dstatus = harmony_read(h, HARMONY_DSTATUS);
+ dstatus |= HARMONY_DSTATUS_IE;
+ harmony_write(h, HARMONY_DSTATUS, dstatus);
+}
+
+static void
+harmony_mute(struct snd_harmony *h)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&h->mixer_lock, flags);
+ harmony_wait_for_control(h);
+ harmony_write(h, HARMONY_GAINCTL, HARMONY_GAIN_SILENCE);
+ spin_unlock_irqrestore(&h->mixer_lock, flags);
+}
+
+static void
+harmony_unmute(struct snd_harmony *h)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&h->mixer_lock, flags);
+ harmony_wait_for_control(h);
+ harmony_write(h, HARMONY_GAINCTL, h->st.gain);
+ spin_unlock_irqrestore(&h->mixer_lock, flags);
+}
+
+static void
+harmony_set_control(struct snd_harmony *h)
+{
+ u32 ctrl;
+ unsigned long flags;
+
+ spin_lock_irqsave(&h->lock, flags);
+
+ ctrl = (HARMONY_CNTL_C |
+ (h->st.format << 6) |
+ (h->st.stereo << 5) |
+ (h->st.rate));
+
+ harmony_wait_for_control(h);
+ harmony_write(h, HARMONY_CNTL, ctrl);
+
+ spin_unlock_irqrestore(&h->lock, flags);
+}
+
+static irqreturn_t
+snd_harmony_interrupt(int irq, void *dev)
+{
+ u32 dstatus;
+ struct snd_harmony *h = dev;
+
+ spin_lock(&h->lock);
+ harmony_disable_interrupts(h);
+ harmony_wait_for_control(h);
+ dstatus = harmony_read(h, HARMONY_DSTATUS);
+ spin_unlock(&h->lock);
+
+ if (dstatus & HARMONY_DSTATUS_PN) {
+ if (h->psubs && h->st.playing) {
+ spin_lock(&h->lock);
+ h->pbuf.buf += h->pbuf.count; /* PAGE_SIZE */
+ h->pbuf.buf %= h->pbuf.size; /* MAX_BUFS*PAGE_SIZE */
+
+ harmony_write(h, HARMONY_PNXTADD,
+ h->pbuf.addr + h->pbuf.buf);
+ h->stats.play_intr++;
+ spin_unlock(&h->lock);
+ snd_pcm_period_elapsed(h->psubs);
+ } else {
+ spin_lock(&h->lock);
+ harmony_write(h, HARMONY_PNXTADD, h->sdma.addr);
+ h->stats.silence_intr++;
+ spin_unlock(&h->lock);
+ }
+ }
+
+ if (dstatus & HARMONY_DSTATUS_RN) {
+ if (h->csubs && h->st.capturing) {
+ spin_lock(&h->lock);
+ h->cbuf.buf += h->cbuf.count;
+ h->cbuf.buf %= h->cbuf.size;
+
+ harmony_write(h, HARMONY_RNXTADD,
+ h->cbuf.addr + h->cbuf.buf);
+ h->stats.rec_intr++;
+ spin_unlock(&h->lock);
+ snd_pcm_period_elapsed(h->csubs);
+ } else {
+ spin_lock(&h->lock);
+ harmony_write(h, HARMONY_RNXTADD, h->gdma.addr);
+ h->stats.graveyard_intr++;
+ spin_unlock(&h->lock);
+ }
+ }
+
+ spin_lock(&h->lock);
+ harmony_enable_interrupts(h);
+ spin_unlock(&h->lock);
+
+ return IRQ_HANDLED;
+}
+
+static unsigned int
+snd_harmony_rate_bits(int rate)
+{
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_harmony_rates); i++)
+ if (snd_harmony_rates[i] == rate)
+ return rate_bits[i];
+
+ return HARMONY_SR_44KHZ;
+}
+
+static struct snd_pcm_hardware snd_harmony_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER),
+ .formats = (SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_MU_LAW |
+ SNDRV_PCM_FMTBIT_A_LAW),
+ .rates = (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 5512,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MAX_BUF_SIZE,
+ .period_bytes_min = BUF_SIZE,
+ .period_bytes_max = BUF_SIZE,
+ .periods_min = 1,
+ .periods_max = MAX_BUFS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_harmony_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER),
+ .formats = (SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_MU_LAW |
+ SNDRV_PCM_FMTBIT_A_LAW),
+ .rates = (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 5512,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MAX_BUF_SIZE,
+ .period_bytes_min = BUF_SIZE,
+ .period_bytes_max = BUF_SIZE,
+ .periods_min = 1,
+ .periods_max = MAX_BUFS,
+ .fifo_size = 0,
+};
+
+static int
+snd_harmony_playback_trigger(struct snd_pcm_substream *ss, int cmd)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+
+ if (h->st.capturing)
+ return -EBUSY;
+
+ spin_lock(&h->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ h->st.playing = 1;
+ harmony_write(h, HARMONY_PNXTADD, h->pbuf.addr);
+ harmony_write(h, HARMONY_RNXTADD, h->gdma.addr);
+ harmony_unmute(h);
+ harmony_enable_interrupts(h);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ h->st.playing = 0;
+ harmony_mute(h);
+ harmony_write(h, HARMONY_PNXTADD, h->sdma.addr);
+ harmony_disable_interrupts(h);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ default:
+ spin_unlock(&h->lock);
+ snd_BUG();
+ return -EINVAL;
+ }
+ spin_unlock(&h->lock);
+
+ return 0;
+}
+
+static int
+snd_harmony_capture_trigger(struct snd_pcm_substream *ss, int cmd)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+
+ if (h->st.playing)
+ return -EBUSY;
+
+ spin_lock(&h->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ h->st.capturing = 1;
+ harmony_write(h, HARMONY_PNXTADD, h->sdma.addr);
+ harmony_write(h, HARMONY_RNXTADD, h->cbuf.addr);
+ harmony_unmute(h);
+ harmony_enable_interrupts(h);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ h->st.capturing = 0;
+ harmony_mute(h);
+ harmony_write(h, HARMONY_RNXTADD, h->gdma.addr);
+ harmony_disable_interrupts(h);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ default:
+ spin_unlock(&h->lock);
+ snd_BUG();
+ return -EINVAL;
+ }
+ spin_unlock(&h->lock);
+
+ return 0;
+}
+
+static int
+snd_harmony_set_data_format(struct snd_harmony *h, int fmt, int force)
+{
+ int o = h->st.format;
+ int n;
+
+ switch(fmt) {
+ case SNDRV_PCM_FORMAT_S16_BE:
+ n = HARMONY_DF_16BIT_LINEAR;
+ break;
+ case SNDRV_PCM_FORMAT_A_LAW:
+ n = HARMONY_DF_8BIT_ALAW;
+ break;
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ n = HARMONY_DF_8BIT_ULAW;
+ break;
+ default:
+ n = HARMONY_DF_16BIT_LINEAR;
+ break;
+ }
+
+ if (force || o != n) {
+ snd_pcm_format_set_silence(fmt, h->sdma.area, SILENCE_BUFSZ /
+ (snd_pcm_format_physical_width(fmt)
+ / 8));
+ }
+
+ return n;
+}
+
+static int
+snd_harmony_playback_prepare(struct snd_pcm_substream *ss)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+
+ if (h->st.capturing)
+ return -EBUSY;
+
+ h->pbuf.size = snd_pcm_lib_buffer_bytes(ss);
+ h->pbuf.count = snd_pcm_lib_period_bytes(ss);
+ if (h->pbuf.buf >= h->pbuf.size)
+ h->pbuf.buf = 0;
+ h->st.playing = 0;
+
+ h->st.rate = snd_harmony_rate_bits(rt->rate);
+ h->st.format = snd_harmony_set_data_format(h, rt->format, 0);
+
+ if (rt->channels == 2)
+ h->st.stereo = HARMONY_SS_STEREO;
+ else
+ h->st.stereo = HARMONY_SS_MONO;
+
+ harmony_set_control(h);
+
+ h->pbuf.addr = rt->dma_addr;
+
+ return 0;
+}
+
+static int
+snd_harmony_capture_prepare(struct snd_pcm_substream *ss)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+
+ if (h->st.playing)
+ return -EBUSY;
+
+ h->cbuf.size = snd_pcm_lib_buffer_bytes(ss);
+ h->cbuf.count = snd_pcm_lib_period_bytes(ss);
+ if (h->cbuf.buf >= h->cbuf.size)
+ h->cbuf.buf = 0;
+ h->st.capturing = 0;
+
+ h->st.rate = snd_harmony_rate_bits(rt->rate);
+ h->st.format = snd_harmony_set_data_format(h, rt->format, 0);
+
+ if (rt->channels == 2)
+ h->st.stereo = HARMONY_SS_STEREO;
+ else
+ h->st.stereo = HARMONY_SS_MONO;
+
+ harmony_set_control(h);
+
+ h->cbuf.addr = rt->dma_addr;
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+snd_harmony_playback_pointer(struct snd_pcm_substream *ss)
+{
+ struct snd_pcm_runtime *rt = ss->runtime;
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ unsigned long pcuradd;
+ unsigned long played;
+
+ if (!(h->st.playing) || (h->psubs == NULL))
+ return 0;
+
+ if ((h->pbuf.addr == 0) || (h->pbuf.size == 0))
+ return 0;
+
+ pcuradd = harmony_read(h, HARMONY_PCURADD);
+ played = pcuradd - h->pbuf.addr;
+
+#ifdef HARMONY_DEBUG
+ printk(KERN_DEBUG PFX "playback_pointer is 0x%lx-0x%lx = %d bytes\n",
+ pcuradd, h->pbuf.addr, played);
+#endif
+
+ if (pcuradd > h->pbuf.addr + h->pbuf.size) {
+ return 0;
+ }
+
+ return bytes_to_frames(rt, played);
+}
+
+static snd_pcm_uframes_t
+snd_harmony_capture_pointer(struct snd_pcm_substream *ss)
+{
+ struct snd_pcm_runtime *rt = ss->runtime;
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ unsigned long rcuradd;
+ unsigned long caught;
+
+ if (!(h->st.capturing) || (h->csubs == NULL))
+ return 0;
+
+ if ((h->cbuf.addr == 0) || (h->cbuf.size == 0))
+ return 0;
+
+ rcuradd = harmony_read(h, HARMONY_RCURADD);
+ caught = rcuradd - h->cbuf.addr;
+
+#ifdef HARMONY_DEBUG
+ printk(KERN_DEBUG PFX "capture_pointer is 0x%lx-0x%lx = %d bytes\n",
+ rcuradd, h->cbuf.addr, caught);
+#endif
+
+ if (rcuradd > h->cbuf.addr + h->cbuf.size) {
+ return 0;
+ }
+
+ return bytes_to_frames(rt, caught);
+}
+
+static int
+snd_harmony_playback_open(struct snd_pcm_substream *ss)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+ int err;
+
+ h->psubs = ss;
+ rt->hw = snd_harmony_playback;
+ snd_pcm_hw_constraint_list(rt, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraint_rates);
+
+ err = snd_pcm_hw_constraint_integer(rt, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int
+snd_harmony_capture_open(struct snd_pcm_substream *ss)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+ int err;
+
+ h->csubs = ss;
+ rt->hw = snd_harmony_capture;
+ snd_pcm_hw_constraint_list(rt, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraint_rates);
+
+ err = snd_pcm_hw_constraint_integer(rt, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int
+snd_harmony_playback_close(struct snd_pcm_substream *ss)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ h->psubs = NULL;
+ return 0;
+}
+
+static int
+snd_harmony_capture_close(struct snd_pcm_substream *ss)
+{
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+ h->csubs = NULL;
+ return 0;
+}
+
+static int
+snd_harmony_hw_params(struct snd_pcm_substream *ss,
+ struct snd_pcm_hw_params *hw)
+{
+ int err;
+ struct snd_harmony *h = snd_pcm_substream_chip(ss);
+
+ err = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(hw));
+ if (err > 0 && h->dma.type == SNDRV_DMA_TYPE_CONTINUOUS)
+ ss->runtime->dma_addr = __pa(ss->runtime->dma_area);
+
+ return err;
+}
+
+static int
+snd_harmony_hw_free(struct snd_pcm_substream *ss)
+{
+ return snd_pcm_lib_free_pages(ss);
+}
+
+static struct snd_pcm_ops snd_harmony_playback_ops = {
+ .open = snd_harmony_playback_open,
+ .close = snd_harmony_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_harmony_hw_params,
+ .hw_free = snd_harmony_hw_free,
+ .prepare = snd_harmony_playback_prepare,
+ .trigger = snd_harmony_playback_trigger,
+ .pointer = snd_harmony_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_harmony_capture_ops = {
+ .open = snd_harmony_capture_open,
+ .close = snd_harmony_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_harmony_hw_params,
+ .hw_free = snd_harmony_hw_free,
+ .prepare = snd_harmony_capture_prepare,
+ .trigger = snd_harmony_capture_trigger,
+ .pointer = snd_harmony_capture_pointer,
+};
+
+static int
+snd_harmony_pcm_init(struct snd_harmony *h)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ harmony_disable_interrupts(h);
+
+ err = snd_pcm_new(h->card, "harmony", 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_harmony_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_harmony_capture_ops);
+
+ pcm->private_data = h;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "harmony");
+ h->pcm = pcm;
+
+ h->psubs = NULL;
+ h->csubs = NULL;
+
+ /* initialize graveyard buffer */
+ h->dma.type = SNDRV_DMA_TYPE_DEV;
+ h->dma.dev = &h->dev->dev;
+ err = snd_dma_alloc_pages(h->dma.type,
+ h->dma.dev,
+ BUF_SIZE*GRAVEYARD_BUFS,
+ &h->gdma);
+ if (err < 0) {
+ printk(KERN_ERR PFX "cannot allocate graveyard buffer!\n");
+ return err;
+ }
+
+ /* initialize silence buffers */
+ err = snd_dma_alloc_pages(h->dma.type,
+ h->dma.dev,
+ BUF_SIZE*SILENCE_BUFS,
+ &h->sdma);
+ if (err < 0) {
+ printk(KERN_ERR PFX "cannot allocate silence buffer!\n");
+ return err;
+ }
+
+ /* pre-allocate space for DMA */
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm, h->dma.type,
+ h->dma.dev,
+ MAX_BUF_SIZE,
+ MAX_BUF_SIZE);
+ if (err < 0) {
+ printk(KERN_ERR PFX "buffer allocation error: %d\n", err);
+ return err;
+ }
+
+ h->st.format = snd_harmony_set_data_format(h,
+ SNDRV_PCM_FORMAT_S16_BE, 1);
+
+ return 0;
+}
+
+static void
+snd_harmony_set_new_gain(struct snd_harmony *h)
+{
+ harmony_wait_for_control(h);
+ harmony_write(h, HARMONY_GAINCTL, h->st.gain);
+}
+
+static int
+snd_harmony_mixercontrol_info(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kc->private_value >> 16) & 0xff;
+ int left_shift = (kc->private_value) & 0xff;
+ int right_shift = (kc->private_value >> 8) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN :
+ SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = left_shift == right_shift ? 1 : 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+
+ return 0;
+}
+
+static int
+snd_harmony_volume_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_harmony *h = snd_kcontrol_chip(kc);
+ int shift_left = (kc->private_value) & 0xff;
+ int shift_right = (kc->private_value >> 8) & 0xff;
+ int mask = (kc->private_value >> 16) & 0xff;
+ int invert = (kc->private_value >> 24) & 0xff;
+ int left, right;
+
+ spin_lock_irq(&h->mixer_lock);
+
+ left = (h->st.gain >> shift_left) & mask;
+ right = (h->st.gain >> shift_right) & mask;
+ if (invert) {
+ left = mask - left;
+ right = mask - right;
+ }
+
+ ucontrol->value.integer.value[0] = left;
+ if (shift_left != shift_right)
+ ucontrol->value.integer.value[1] = right;
+
+ spin_unlock_irq(&h->mixer_lock);
+
+ return 0;
+}
+
+static int
+snd_harmony_volume_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_harmony *h = snd_kcontrol_chip(kc);
+ int shift_left = (kc->private_value) & 0xff;
+ int shift_right = (kc->private_value >> 8) & 0xff;
+ int mask = (kc->private_value >> 16) & 0xff;
+ int invert = (kc->private_value >> 24) & 0xff;
+ int left, right;
+ int old_gain = h->st.gain;
+
+ spin_lock_irq(&h->mixer_lock);
+
+ left = ucontrol->value.integer.value[0] & mask;
+ if (invert)
+ left = mask - left;
+ h->st.gain &= ~( (mask << shift_left ) );
+ h->st.gain |= (left << shift_left);
+
+ if (shift_left != shift_right) {
+ right = ucontrol->value.integer.value[1] & mask;
+ if (invert)
+ right = mask - right;
+ h->st.gain &= ~( (mask << shift_right) );
+ h->st.gain |= (right << shift_right);
+ }
+
+ snd_harmony_set_new_gain(h);
+
+ spin_unlock_irq(&h->mixer_lock);
+
+ return h->st.gain != old_gain;
+}
+
+static int
+snd_harmony_captureroute_info(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "Line", "Mic" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int
+snd_harmony_captureroute_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_harmony *h = snd_kcontrol_chip(kc);
+ int value;
+
+ spin_lock_irq(&h->mixer_lock);
+
+ value = (h->st.gain >> HARMONY_GAIN_IS_SHIFT) & 1;
+ ucontrol->value.enumerated.item[0] = value;
+
+ spin_unlock_irq(&h->mixer_lock);
+
+ return 0;
+}
+
+static int
+snd_harmony_captureroute_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_harmony *h = snd_kcontrol_chip(kc);
+ int value;
+ int old_gain = h->st.gain;
+
+ spin_lock_irq(&h->mixer_lock);
+
+ value = ucontrol->value.enumerated.item[0] & 1;
+ h->st.gain &= ~HARMONY_GAIN_IS_MASK;
+ h->st.gain |= value << HARMONY_GAIN_IS_SHIFT;
+
+ snd_harmony_set_new_gain(h);
+
+ spin_unlock_irq(&h->mixer_lock);
+
+ return h->st.gain != old_gain;
+}
+
+#define HARMONY_CONTROLS ARRAY_SIZE(snd_harmony_controls)
+
+#define HARMONY_VOLUME(xname, left_shift, right_shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_harmony_mixercontrol_info, \
+ .get = snd_harmony_volume_get, .put = snd_harmony_volume_put, \
+ .private_value = ((left_shift) | ((right_shift) << 8) | \
+ ((mask) << 16) | ((invert) << 24)) }
+
+static struct snd_kcontrol_new snd_harmony_controls[] = {
+ HARMONY_VOLUME("Master Playback Volume", HARMONY_GAIN_LO_SHIFT,
+ HARMONY_GAIN_RO_SHIFT, HARMONY_GAIN_OUT, 1),
+ HARMONY_VOLUME("Capture Volume", HARMONY_GAIN_LI_SHIFT,
+ HARMONY_GAIN_RI_SHIFT, HARMONY_GAIN_IN, 0),
+ HARMONY_VOLUME("Monitor Volume", HARMONY_GAIN_MA_SHIFT,
+ HARMONY_GAIN_MA_SHIFT, HARMONY_GAIN_MA, 1),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Route",
+ .info = snd_harmony_captureroute_info,
+ .get = snd_harmony_captureroute_get,
+ .put = snd_harmony_captureroute_put
+ },
+ HARMONY_VOLUME("Internal Speaker Switch", HARMONY_GAIN_SE_SHIFT,
+ HARMONY_GAIN_SE_SHIFT, 1, 0),
+ HARMONY_VOLUME("Line-Out Switch", HARMONY_GAIN_LE_SHIFT,
+ HARMONY_GAIN_LE_SHIFT, 1, 0),
+ HARMONY_VOLUME("Headphones Switch", HARMONY_GAIN_HE_SHIFT,
+ HARMONY_GAIN_HE_SHIFT, 1, 0),
+};
+
+static void __devinit
+snd_harmony_mixer_reset(struct snd_harmony *h)
+{
+ harmony_mute(h);
+ harmony_reset(h);
+ h->st.gain = HARMONY_GAIN_DEFAULT;
+ harmony_unmute(h);
+}
+
+static int __devinit
+snd_harmony_mixer_init(struct snd_harmony *h)
+{
+ struct snd_card *card = h->card;
+ int idx, err;
+
+ if (snd_BUG_ON(!h))
+ return -EINVAL;
+ strcpy(card->mixername, "Harmony Gain control interface");
+
+ for (idx = 0; idx < HARMONY_CONTROLS; idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_harmony_controls[idx], h));
+ if (err < 0)
+ return err;
+ }
+
+ snd_harmony_mixer_reset(h);
+
+ return 0;
+}
+
+static int
+snd_harmony_free(struct snd_harmony *h)
+{
+ if (h->gdma.addr)
+ snd_dma_free_pages(&h->gdma);
+ if (h->sdma.addr)
+ snd_dma_free_pages(&h->sdma);
+
+ if (h->irq >= 0)
+ free_irq(h->irq, h);
+
+ if (h->iobase)
+ iounmap(h->iobase);
+
+ parisc_set_drvdata(h->dev, NULL);
+
+ kfree(h);
+ return 0;
+}
+
+static int
+snd_harmony_dev_free(struct snd_device *dev)
+{
+ struct snd_harmony *h = dev->device_data;
+ return snd_harmony_free(h);
+}
+
+static int __devinit
+snd_harmony_create(struct snd_card *card,
+ struct parisc_device *padev,
+ struct snd_harmony **rchip)
+{
+ int err;
+ struct snd_harmony *h;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_harmony_dev_free,
+ };
+
+ *rchip = NULL;
+
+ h = kzalloc(sizeof(*h), GFP_KERNEL);
+ if (h == NULL)
+ return -ENOMEM;
+
+ h->hpa = padev->hpa.start;
+ h->card = card;
+ h->dev = padev;
+ h->irq = -1;
+ h->iobase = ioremap_nocache(padev->hpa.start, HARMONY_SIZE);
+ if (h->iobase == NULL) {
+ printk(KERN_ERR PFX "unable to remap hpa 0x%lx\n",
+ padev->hpa.start);
+ err = -EBUSY;
+ goto free_and_ret;
+ }
+
+ err = request_irq(padev->irq, snd_harmony_interrupt, 0,
+ "harmony", h);
+ if (err) {
+ printk(KERN_ERR PFX "could not obtain interrupt %d",
+ padev->irq);
+ goto free_and_ret;
+ }
+ h->irq = padev->irq;
+
+ spin_lock_init(&h->mixer_lock);
+ spin_lock_init(&h->lock);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ h, &ops)) < 0) {
+ goto free_and_ret;
+ }
+
+ snd_card_set_dev(card, &padev->dev);
+
+ *rchip = h;
+
+ return 0;
+
+free_and_ret:
+ snd_harmony_free(h);
+ return err;
+}
+
+static int __devinit
+snd_harmony_probe(struct parisc_device *padev)
+{
+ int err;
+ struct snd_card *card;
+ struct snd_harmony *h;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ err = snd_harmony_create(card, padev, &h);
+ if (err < 0)
+ goto free_and_ret;
+
+ err = snd_harmony_pcm_init(h);
+ if (err < 0)
+ goto free_and_ret;
+
+ err = snd_harmony_mixer_init(h);
+ if (err < 0)
+ goto free_and_ret;
+
+ strcpy(card->driver, "harmony");
+ strcpy(card->shortname, "Harmony");
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, h->hpa, h->irq);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto free_and_ret;
+
+ parisc_set_drvdata(padev, card);
+ return 0;
+
+free_and_ret:
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit
+snd_harmony_remove(struct parisc_device *padev)
+{
+ snd_card_free(parisc_get_drvdata(padev));
+ parisc_set_drvdata(padev, NULL);
+ return 0;
+}
+
+static struct parisc_driver snd_harmony_driver = {
+ .name = "harmony",
+ .id_table = snd_harmony_devtable,
+ .probe = snd_harmony_probe,
+ .remove = snd_harmony_remove,
+};
+
+static int __init
+alsa_harmony_init(void)
+{
+ return register_parisc_driver(&snd_harmony_driver);
+}
+
+static void __exit
+alsa_harmony_fini(void)
+{
+ unregister_parisc_driver(&snd_harmony_driver);
+}
+
+MODULE_LICENSE("GPL");
+MODULE_AUTHOR("Kyle McMartin <kyle@parisc-linux.org>");
+MODULE_DESCRIPTION("Harmony sound driver");
+
+module_init(alsa_harmony_init);
+module_exit(alsa_harmony_fini);
diff --git a/sound/parisc/harmony.h b/sound/parisc/harmony.h
new file mode 100644
index 0000000..2e43452
--- /dev/null
+++ b/sound/parisc/harmony.h
@@ -0,0 +1,154 @@
+/* Hewlett-Packard Harmony audio driver
+ * Copyright (C) 2004, Kyle McMartin <kyle@parisc-linux.org>
+ */
+
+#ifndef __HARMONY_H__
+#define __HARMONY_H__
+
+struct harmony_buffer {
+ unsigned long addr;
+ int buf;
+ int count;
+ int size;
+ int coherent;
+};
+
+struct snd_harmony {
+ int irq;
+
+ unsigned long hpa; /* hard physical address */
+ void __iomem *iobase; /* remapped io address */
+
+ struct parisc_device *dev;
+
+ struct {
+ u32 gain;
+ u32 rate;
+ u32 format;
+ u32 stereo;
+ int playing;
+ int capturing;
+ } st;
+
+ struct snd_dma_device dma; /* playback/capture */
+ struct harmony_buffer pbuf;
+ struct harmony_buffer cbuf;
+
+ struct snd_dma_buffer gdma; /* graveyard */
+ struct snd_dma_buffer sdma; /* silence */
+
+ struct {
+ unsigned long play_intr;
+ unsigned long rec_intr;
+ unsigned long graveyard_intr;
+ unsigned long silence_intr;
+ } stats;
+
+ struct snd_pcm *pcm;
+ struct snd_card *card;
+ struct snd_pcm_substream *psubs;
+ struct snd_pcm_substream *csubs;
+ struct snd_info_entry *proc;
+
+ spinlock_t lock;
+ spinlock_t mixer_lock;
+};
+
+#define MAX_PCM_DEVICES 1
+#define MAX_PCM_SUBSTREAMS 4
+#define MAX_MIDI_DEVICES 0
+
+#define HARMONY_SIZE 64
+
+#define BUF_SIZE PAGE_SIZE
+#define MAX_BUFS 16
+#define MAX_BUF_SIZE (MAX_BUFS * BUF_SIZE)
+
+#define PLAYBACK_BUFS MAX_BUFS
+#define RECORD_BUFS MAX_BUFS
+#define GRAVEYARD_BUFS 1
+#define GRAVEYARD_BUFSZ (GRAVEYARD_BUFS*BUF_SIZE)
+#define SILENCE_BUFS 1
+#define SILENCE_BUFSZ (SILENCE_BUFS*BUF_SIZE)
+
+#define HARMONY_ID 0x000
+#define HARMONY_RESET 0x004
+#define HARMONY_CNTL 0x008
+#define HARMONY_GAINCTL 0x00c
+#define HARMONY_PNXTADD 0x010
+#define HARMONY_PCURADD 0x014
+#define HARMONY_RNXTADD 0x018
+#define HARMONY_RCURADD 0x01c
+#define HARMONY_DSTATUS 0x020
+#define HARMONY_OV 0x024
+#define HARMONY_PIO 0x028
+#define HARMONY_DIAG 0x03c
+
+#define HARMONY_CNTL_C 0x80000000
+#define HARMONY_CNTL_ST 0x00000020
+#define HARMONY_CNTL_44100 0x00000015 /* HARMONY_SR_44KHZ */
+#define HARMONY_CNTL_8000 0x00000008 /* HARMONY_SR_8KHZ */
+
+#define HARMONY_DSTATUS_ID 0x00000000 /* interrupts off */
+#define HARMONY_DSTATUS_PN 0x00000200 /* playback fill */
+#define HARMONY_DSTATUS_RN 0x00000002 /* record fill */
+#define HARMONY_DSTATUS_IE 0x80000000 /* interrupts on */
+
+#define HARMONY_DF_16BIT_LINEAR 0x00000000
+#define HARMONY_DF_8BIT_ULAW 0x00000001
+#define HARMONY_DF_8BIT_ALAW 0x00000002
+
+#define HARMONY_SS_MONO 0x00000000
+#define HARMONY_SS_STEREO 0x00000001
+
+#define HARMONY_GAIN_SILENCE 0x01F00FFF
+#define HARMONY_GAIN_DEFAULT 0x01F00FFF
+
+#define HARMONY_GAIN_HE_SHIFT 27 /* headphones enabled */
+#define HARMONY_GAIN_HE_MASK (1 << HARMONY_GAIN_HE_SHIFT)
+#define HARMONY_GAIN_LE_SHIFT 26 /* line-out enabled */
+#define HARMONY_GAIN_LE_MASK (1 << HARMONY_GAIN_LE_SHIFT)
+#define HARMONY_GAIN_SE_SHIFT 25 /* internal-speaker enabled */
+#define HARMONY_GAIN_SE_MASK (1 << HARMONY_GAIN_SE_SHIFT)
+#define HARMONY_GAIN_IS_SHIFT 24 /* input select - 0 for line, 1 for mic */
+#define HARMONY_GAIN_IS_MASK (1 << HARMONY_GAIN_IS_SHIFT)
+
+/* monitor attenuation */
+#define HARMONY_GAIN_MA 0x0f
+#define HARMONY_GAIN_MA_SHIFT 20
+#define HARMONY_GAIN_MA_MASK (HARMONY_GAIN_MA << HARMONY_GAIN_MA_SHIFT)
+
+/* input gain */
+#define HARMONY_GAIN_IN 0x0f
+#define HARMONY_GAIN_LI_SHIFT 16
+#define HARMONY_GAIN_LI_MASK (HARMONY_GAIN_IN << HARMONY_GAIN_LI_SHIFT)
+#define HARMONY_GAIN_RI_SHIFT 12
+#define HARMONY_GAIN_RI_MASK (HARMONY_GAIN_IN << HARMONY_GAIN_RI_SHIFT)
+
+/* output gain (master volume) */
+#define HARMONY_GAIN_OUT 0x3f
+#define HARMONY_GAIN_LO_SHIFT 6
+#define HARMONY_GAIN_LO_MASK (HARMONY_GAIN_OUT << HARMONY_GAIN_LO_SHIFT)
+#define HARMONY_GAIN_RO_SHIFT 0
+#define HARMONY_GAIN_RO_MASK (HARMONY_GAIN_OUT << HARMONY_GAIN_RO_SHIFT)
+
+#define HARMONY_MAX_OUT (HARMONY_GAIN_RO_MASK >> HARMONY_GAIN_RO_SHIFT)
+#define HARMONY_MAX_IN (HARMONY_GAIN_RI_MASK >> HARMONY_GAIN_RI_SHIFT)
+#define HARMONY_MAX_MON (HARMONY_GAIN_MA_MASK >> HARMONY_GAIN_MA_SHIFT)
+
+#define HARMONY_SR_8KHZ 0x08
+#define HARMONY_SR_16KHZ 0x09
+#define HARMONY_SR_27KHZ 0x0A
+#define HARMONY_SR_32KHZ 0x0B
+#define HARMONY_SR_48KHZ 0x0E
+#define HARMONY_SR_9KHZ 0x0F
+#define HARMONY_SR_5KHZ 0x10
+#define HARMONY_SR_11KHZ 0x11
+#define HARMONY_SR_18KHZ 0x12
+#define HARMONY_SR_22KHZ 0x13
+#define HARMONY_SR_37KHZ 0x14
+#define HARMONY_SR_44KHZ 0x15
+#define HARMONY_SR_33KHZ 0x16
+#define HARMONY_SR_6KHZ 0x17
+
+#endif /* __HARMONY_H__ */
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
new file mode 100644
index 0000000..7003711
--- /dev/null
+++ b/sound/pci/Kconfig
@@ -0,0 +1,895 @@
+# ALSA PCI drivers
+
+menuconfig SND_PCI
+ bool "PCI sound devices"
+ depends on PCI
+ default y
+ help
+ Support for sound devices connected via the PCI bus.
+
+if SND_PCI
+
+config SND_AD1889
+ tristate "Analog Devices AD1889"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated AC97 sound
+ device found in particular on the Hewlett-Packard [BCJ]-xxx0
+ class PA-RISC workstations, using the AD1819 codec.
+
+ To compile this as a module, choose M here: the module
+ will be called snd-ad1889.
+
+config SND_ALS300
+ tristate "Avance Logic ALS300/ALS300+"
+ select SND_PCM
+ select SND_AC97_CODEC
+ select SND_OPL3_LIB
+ help
+ Say 'Y' or 'M' to include support for Avance Logic ALS300/ALS300+
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-als300
+
+config SND_ALS4000
+ tristate "Avance Logic ALS4000"
+ depends on ISA_DMA_API
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ select SND_SB_COMMON
+ help
+ Say Y here to include support for soundcards based on Avance Logic
+ ALS4000 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-als4000.
+
+config SND_ALI5451
+ tristate "ALi M5451 PCI Audio Controller"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated AC97 sound
+ device on motherboards using the ALi M5451 Audio Controller
+ (M1535/M1535D/M1535+/M1535D+ south bridges). Newer chipsets
+ use the "Intel/SiS/nVidia/AMD/ALi AC97 Controller" driver.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ali5451.
+
+config SND_ATIIXP
+ tristate "ATI IXP AC97 Controller"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated AC97 sound
+ device on motherboards with ATI chipsets (ATI IXP 150/200/250/
+ 300/400).
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-atiixp.
+
+config SND_ATIIXP_MODEM
+ tristate "ATI IXP Modem"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated MC97 modem on
+ motherboards with ATI chipsets (ATI IXP 150/200/250).
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-atiixp-modem.
+
+config SND_AU8810
+ tristate "Aureal Advantage"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for Aureal Advantage soundcards.
+
+ Supported features: Hardware Mixer, SRC, EQ and SPDIF output.
+ 3D support code is in place, but not yet useable. For more info,
+ email the ALSA developer list, or <mjander@users.sourceforge.net>.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-au8810.
+
+config SND_AU8820
+ tristate "Aureal Vortex"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for Aureal Vortex soundcards.
+
+ Supported features: Hardware Mixer and SRC. For more info, email
+ the ALSA developer list, or <mjander@users.sourceforge.net>.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-au8820.
+
+config SND_AU8830
+ tristate "Aureal Vortex 2"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for Aureal Vortex 2 soundcards.
+
+ Supported features: Hardware Mixer, SRC, EQ and SPDIF output.
+ 3D support code is in place, but not yet useable. For more info,
+ email the ALSA developer list, or <mjander@users.sourceforge.net>.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-au8830.
+
+config SND_AW2
+ tristate "Emagic Audiowerk 2"
+ help
+ Say Y here to include support for Emagic Audiowerk 2 soundcards.
+
+ Supported features: Analog and SPDIF output. Analog or SPDIF input.
+ Note: Switch between analog and digital input does not always work.
+ It can produce continuous noise. The workaround is to switch again
+ (and again) between digital and analog input until it works.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-aw2.
+
+
+config SND_AZT3328
+ tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)"
+ depends on EXPERIMENTAL
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ help
+ Say Y here to include support for Aztech AZF3328 (PCI168)
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-azt3328.
+
+config SND_BT87X
+ tristate "Bt87x Audio Capture"
+ select SND_PCM
+ help
+ If you want to record audio from TV cards based on
+ Brooktree Bt878/Bt879 chips, say Y here and read
+ <file:Documentation/sound/alsa/Bt87x.txt>.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-bt87x.
+
+config SND_BT87X_OVERCLOCK
+ bool "Bt87x Audio overclocking"
+ depends on SND_BT87X
+ help
+ Say Y here if 448000 Hz isn't enough for you and you want to
+ record from the analog input with up to 1792000 Hz.
+
+ Higher sample rates won't hurt your hardware, but audio
+ quality may suffer.
+
+config SND_CA0106
+ tristate "SB Audigy LS / Live 24bit"
+ select SND_AC97_CODEC
+ select SND_RAWMIDI
+ select SND_VMASTER
+ help
+ Say Y here to include support for the Sound Blaster Audigy LS
+ and Live 24bit.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ca0106.
+
+config SND_CMIPCI
+ tristate "C-Media 8338, 8738, 8768, 8770"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_PCM
+ help
+ If you want to use soundcards based on C-Media CMI8338, CMI8738,
+ CMI8768 or CMI8770 chips, say Y here and read
+ <file:Documentation/sound/alsa/CMIPCI.txt>.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cmipci.
+
+config SND_OXYGEN_LIB
+ tristate
+ select SND_PCM
+ select SND_MPU401_UART
+
+config SND_OXYGEN
+ tristate "C-Media 8788 (Oxygen)"
+ select SND_OXYGEN_LIB
+ help
+ Say Y here to include support for sound cards based on the
+ C-Media CMI8788 (Oxygen HD Audio) chip:
+ * Asound A-8788
+ * AuzenTech X-Meridian
+ * Bgears b-Enspirer
+ * Club3D Theatron DTS
+ * HT-Omega Claro
+ * Razer Barracuda AC-1
+ * Sondigo Inferno
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-oxygen.
+
+config SND_CS4281
+ tristate "Cirrus Logic (Sound Fusion) CS4281"
+ select SND_OPL3_LIB
+ select SND_RAWMIDI
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for Cirrus Logic CS4281 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cs4281.
+
+config SND_CS46XX
+ tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x"
+ select SND_RAWMIDI
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for Cirrus Logic CS4610/CS4612/
+ CS4614/CS4615/CS4622/CS4624/CS4630/CS4280 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cs46xx.
+
+config SND_CS46XX_NEW_DSP
+ bool "Cirrus Logic (Sound Fusion) New DSP support"
+ depends on SND_CS46XX
+ default y
+ help
+ Say Y here to use a new DSP image for SPDIF and dual codecs.
+
+ This works better than the old code, so say Y.
+
+config SND_CS5530
+ tristate "CS5530 Audio"
+ depends on ISA_DMA_API
+ select SND_SB16_DSP
+ help
+ Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cs5530.
+
+config SND_CS5535AUDIO
+ tristate "CS5535/CS5536 Audio"
+ depends on X86 && !X86_64
+ select SND_PCM
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for audio on CS5535 chips. It is
+ referred to as NS CS5535 IO or AMD CS5535 IO companion in
+ various literature. This driver also supports the CS5536 audio
+ device. However, for both chips, on certain boards, you may
+ need to use ac97_quirk=hp_only if your board has physically
+ mapped headphone out to master output. If that works for you,
+ send lspci -vvv output to the mailing list so that your board
+ can be identified in the quirks list.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-cs5535audio.
+
+config SND_DARLA20
+ tristate "(Echoaudio) Darla20"
+ select FW_LOADER
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Darla.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-darla20
+
+config SND_GINA20
+ tristate "(Echoaudio) Gina20"
+ select FW_LOADER
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Gina.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-gina20
+
+config SND_LAYLA20
+ tristate "(Echoaudio) Layla20"
+ select FW_LOADER
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Layla.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-layla20
+
+config SND_DARLA24
+ tristate "(Echoaudio) Darla24"
+ select FW_LOADER
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Darla24.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-darla24
+
+config SND_GINA24
+ tristate "(Echoaudio) Gina24"
+ select FW_LOADER
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Gina24.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-gina24
+
+config SND_LAYLA24
+ tristate "(Echoaudio) Layla24"
+ select FW_LOADER
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Layla24.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-layla24
+
+config SND_MONA
+ tristate "(Echoaudio) Mona"
+ select FW_LOADER
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Mona.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-mona
+
+config SND_MIA
+ tristate "(Echoaudio) Mia"
+ select FW_LOADER
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Mia and Mia-midi.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-mia
+
+config SND_ECHO3G
+ tristate "(Echoaudio) 3G cards"
+ select FW_LOADER
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Gina3G and Layla3G.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-echo3g
+
+config SND_INDIGO
+ tristate "(Echoaudio) Indigo"
+ select FW_LOADER
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Indigo.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-indigo
+
+config SND_INDIGOIO
+ tristate "(Echoaudio) Indigo IO"
+ select FW_LOADER
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Indigo IO.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-indigoio
+
+config SND_INDIGODJ
+ tristate "(Echoaudio) Indigo DJ"
+ select FW_LOADER
+ select SND_PCM
+ help
+ Say 'Y' or 'M' to include support for Echoaudio Indigo DJ.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-indigodj
+
+config SND_EMU10K1
+ tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)"
+ select FW_LOADER
+ select SND_HWDEP
+ select SND_RAWMIDI
+ select SND_AC97_CODEC
+ help
+ Say Y to include support for Sound Blaster PCI 512, Live!,
+ Audigy and E-mu APS (partially supported) soundcards.
+
+ The confusing multitude of mixer controls is documented in
+ <file:Documentation/sound/alsa/SB-Live-mixer.txt> and
+ <file:Documentation/sound/alsa/Audigy-mixer.txt>.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-emu10k1.
+
+config SND_EMU10K1X
+ tristate "Emu10k1X (Dell OEM Version)"
+ select SND_AC97_CODEC
+ select SND_RAWMIDI
+ help
+ Say Y here to include support for the Dell OEM version of the
+ Sound Blaster Live!.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-emu10k1x.
+
+config SND_ENS1370
+ tristate "(Creative) Ensoniq AudioPCI 1370"
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for Ensoniq AudioPCI ES1370 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ens1370.
+
+config SND_ENS1371
+ tristate "(Creative) Ensoniq AudioPCI 1371/1373"
+ select SND_RAWMIDI
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for Ensoniq AudioPCI ES1371 chips and
+ Sound Blaster PCI 64 or 128 soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ens1371.
+
+config SND_ES1938
+ tristate "ESS ES1938/1946/1969 (Solo-1)"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for soundcards based on ESS Solo-1
+ (ES1938, ES1946, ES1969) chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-es1938.
+
+config SND_ES1968
+ tristate "ESS ES1968/1978 (Maestro-1/2/2E)"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for soundcards based on ESS Maestro
+ 1/2/2E chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-es1968.
+
+config SND_FM801
+ tristate "ForteMedia FM801"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for soundcards based on the ForteMedia
+ FM801 chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-fm801.
+
+config SND_FM801_TEA575X_BOOL
+ bool "ForteMedia FM801 + TEA5757 tuner"
+ depends on SND_FM801
+ depends on VIDEO_V4L1=y || VIDEO_V4L1=SND_FM801
+ help
+ Say Y here to include support for soundcards based on the ForteMedia
+ FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media
+ Forte SF256-PCS-02) into the snd-fm801 driver.
+
+config SND_FM801_TEA575X
+ tristate
+ depends on SND_FM801_TEA575X_BOOL
+ default SND_FM801
+
+config SND_HDA_INTEL
+ tristate "Intel HD Audio"
+ select SND_PCM
+ select SND_VMASTER
+ help
+ Say Y here to include support for Intel "High Definition
+ Audio" (Azalia) motherboard devices.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-hda-intel.
+
+config SND_HDA_HWDEP
+ bool "Build hwdep interface for HD-audio driver"
+ depends on SND_HDA_INTEL
+ select SND_HWDEP
+ help
+ Say Y here to build a hwdep interface for HD-audio driver.
+ This interface can be used for out-of-band communication
+ with codecs for debugging purposes.
+
+config SND_HDA_INPUT_BEEP
+ bool "Support digital beep via input layer"
+ depends on SND_HDA_INTEL
+ depends on INPUT=y || INPUT=SND_HDA_INTEL
+ help
+ Say Y here to build a digital beep interface for HD-audio
+ driver. This interface is used to generate digital beeps.
+
+config SND_HDA_CODEC_REALTEK
+ bool "Build Realtek HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Realtek HD-audio codec support in
+ snd-hda-intel driver, such as ALC880.
+
+config SND_HDA_CODEC_ANALOG
+ bool "Build Analog Device HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Analog Device HD-audio codec support in
+ snd-hda-intel driver, such as AD1986A.
+
+config SND_HDA_CODEC_SIGMATEL
+ bool "Build IDT/Sigmatel HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include IDT (Sigmatel) HD-audio codec support in
+ snd-hda-intel driver, such as STAC9200.
+
+config SND_HDA_CODEC_VIA
+ bool "Build VIA HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include VIA HD-audio codec support in
+ snd-hda-intel driver, such as VT1708.
+
+config SND_HDA_CODEC_ATIHDMI
+ bool "Build ATI HDMI HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include ATI HDMI HD-audio codec support in
+ snd-hda-intel driver, such as ATI RS600 HDMI.
+
+config SND_HDA_CODEC_NVHDMI
+ bool "Build NVIDIA HDMI HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include NVIDIA HDMI HD-audio codec support in
+ snd-hda-intel driver, such as NVIDIA MCP78 HDMI.
+
+config SND_HDA_CODEC_CONEXANT
+ bool "Build Conexant HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Conexant HD-audio codec support in
+ snd-hda-intel driver, such as CX20549.
+
+config SND_HDA_CODEC_CMEDIA
+ bool "Build C-Media HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include C-Media HD-audio codec support in
+ snd-hda-intel driver, such as CMI9880.
+
+config SND_HDA_CODEC_SI3054
+ bool "Build Silicon Labs 3054 HD-modem codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Silicon Labs 3054 HD-modem codec
+ (and compatibles) support in snd-hda-intel driver.
+
+config SND_HDA_GENERIC
+ bool "Enable generic HD-audio codec parser"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to enable the generic HD-audio codec parser
+ in snd-hda-intel driver.
+
+config SND_HDA_POWER_SAVE
+ bool "Aggressive power-saving on HD-audio"
+ depends on SND_HDA_INTEL && EXPERIMENTAL
+ help
+ Say Y here to enable more aggressive power-saving mode on
+ HD-audio driver. The power-saving timeout can be configured
+ via power_save option or over sysfs on-the-fly.
+
+config SND_HDA_POWER_SAVE_DEFAULT
+ int "Default time-out for HD-audio power-save mode"
+ depends on SND_HDA_POWER_SAVE
+ default 0
+ help
+ The default time-out value in seconds for HD-audio automatic
+ power-save mode. 0 means to disable the power-save mode.
+
+config SND_HDSP
+ tristate "RME Hammerfall DSP Audio"
+ select SND_HWDEP
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for RME Hammerfall DSP Audio
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-hdsp.
+
+config SND_HDSPM
+ tristate "RME Hammerfall DSP MADI"
+ select SND_HWDEP
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for RME Hammerfall DSP MADI
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-hdspm.
+
+config SND_HIFIER
+ tristate "TempoTec HiFier Fantasia"
+ select SND_OXYGEN_LIB
+ help
+ Say Y here to include support for the MediaTek/TempoTec HiFier
+ Fantasia sound card.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-hifier.
+
+config SND_ICE1712
+ tristate "ICEnsemble ICE1712 (Envy24)"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for soundcards based on the
+ ICE1712 (Envy24) chip.
+
+ Currently supported hardware is: M-Audio Delta 1010(LT),
+ DiO 2496, 66, 44, 410, Audiophile 24/96; Digigram VX442;
+ TerraTec EWX 24/96, EWS 88MT/D, DMX 6Fire, Phase 88;
+ Hoontech SoundTrack DSP 24/Value/Media7.1; Event EZ8;
+ Lionstracs Mediastation, Terrasoniq TS 88.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ice1712.
+
+config SND_ICE1724
+ tristate "ICE/VT1724/1720 (Envy24HT/PT)"
+ select SND_RAWMIDI
+ select SND_AC97_CODEC
+ select SND_VMASTER
+ help
+ Say Y here to include support for soundcards based on
+ ICE/VT1724/1720 (Envy24HT/PT) chips.
+
+ Currently supported hardware is: AMP AUDIO2000; M-Audio
+ Revolution 5.1, 7.1, Audiophile 192; TerraTec Aureon 5.1 Sky,
+ 7.1 Space/Universe, Phase 22/28; Onkyo SE-90PCI, SE-200PCI;
+ AudioTrak Prodigy 192, 7.1 (HIFI/LT/XT), HD2; Hercules
+ Fortissimo IV; ESI Juli@; Pontis MS300; EGO-SYS WaveTerminal
+ 192M; Albatron K8X800 Pro II; Chaintech ZNF3-150/250, 9CJS,
+ AV-710; Shuttle SN25P.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ice1724.
+
+config SND_INTEL8X0
+ tristate "Intel/SiS/nVidia/AMD/ALi AC97 Controller"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated AC97 sound
+ device on motherboards with Intel/SiS/nVidia/AMD chipsets, or
+ ALi chipsets using the M5455 Audio Controller. (There is a
+ separate driver for ALi M5451 Audio Controllers.)
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-intel8x0.
+
+config SND_INTEL8X0M
+ tristate "Intel/SiS/nVidia/AMD MC97 Modem"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated MC97 modem on
+ motherboards with Intel/SiS/nVidia/AMD chipsets.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-intel8x0m.
+
+config SND_KORG1212
+ tristate "Korg 1212 IO"
+ select SND_PCM
+ help
+ Say Y here to include support for Korg 1212IO soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-korg1212.
+
+config SND_MAESTRO3
+ tristate "ESS Allegro/Maestro3"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for soundcards based on ESS Maestro 3
+ (Allegro) chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-maestro3.
+
+config SND_MIXART
+ tristate "Digigram miXart"
+ select SND_HWDEP
+ select SND_PCM
+ help
+ If you want to use Digigram miXart soundcards, say Y here and
+ read <file:Documentation/sound/alsa/MIXART.txt>.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-mixart.
+
+config SND_NM256
+ tristate "NeoMagic NM256AV/ZX"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for NeoMagic NM256AV/ZX chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-nm256.
+
+config SND_PCXHR
+ tristate "Digigram PCXHR"
+ select SND_PCM
+ select SND_HWDEP
+ help
+ Say Y here to include support for Digigram PCXHR boards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-pcxhr.
+
+config SND_RIPTIDE
+ tristate "Conexant Riptide"
+ select FW_LOADER
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say 'Y' or 'M' to include support for Conexant Riptide chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-riptide
+
+config SND_RME32
+ tristate "RME Digi32, 32/8, 32 PRO"
+ select SND_PCM
+ help
+ Say Y to include support for RME Digi32, Digi32 PRO and
+ Digi32/8 (Sek'd Prodif32, Prodif96 and Prodif Gold) audio
+ devices.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-rme32.
+
+config SND_RME96
+ tristate "RME Digi96, 96/8, 96/8 PRO"
+ select SND_PCM
+ help
+ Say Y here to include support for RME Digi96, Digi96/8 and
+ Digi96/8 PRO/PAD/PST soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-rme96.
+
+config SND_RME9652
+ tristate "RME Digi9652 (Hammerfall)"
+ select SND_PCM
+ help
+ Say Y here to include support for RME Hammerfall (RME
+ Digi9652/Digi9636) soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-rme9652.
+
+config SND_SIS7019
+ tristate "SiS 7019 Audio Accelerator"
+ depends on X86 && !X86_64
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the SiS 7019 Audio Accelerator.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sis7019.
+
+config SND_SONICVIBES
+ tristate "S3 SonicVibes"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for soundcards based on the S3
+ SonicVibes chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sonicvibes.
+
+config SND_TRIDENT
+ tristate "Trident 4D-Wave DX/NX; SiS 7018"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for soundcards based on Trident
+ 4D-Wave DX/NX or SiS 7018 chips.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-trident.
+
+config SND_VIA82XX
+ tristate "VIA 82C686A/B, 8233/8235 AC97 Controller"
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated AC97 sound
+ device on motherboards with VIA chipsets.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-via82xx.
+
+config SND_VIA82XX_MODEM
+ tristate "VIA 82C686A/B, 8233 based Modems"
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for the integrated MC97 modem on
+ motherboards with VIA chipsets.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-via82xx-modem.
+
+config SND_VIRTUOSO
+ tristate "Asus Virtuoso 100/200 (Xonar)"
+ select SND_OXYGEN_LIB
+ help
+ Say Y here to include support for sound cards based on the
+ Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and
+ HDAV1.3 (Deluxe).
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-virtuoso.
+
+config SND_VX222
+ tristate "Digigram VX222"
+ select SND_VX_LIB
+ help
+ Say Y here to include support for Digigram VX222 soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-vx222.
+
+config SND_YMFPCI
+ tristate "Yamaha YMF724/740/744/754"
+ select SND_OPL3_LIB
+ select SND_MPU401_UART
+ select SND_AC97_CODEC
+ help
+ Say Y here to include support for Yamaha PCI audio chips -
+ YMF724, YMF724F, YMF740, YMF740C, YMF744, YMF754.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ymfpci.
+
+endif # SND_PCI
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
new file mode 100644
index 0000000..65b25d2
--- /dev/null
+++ b/sound/pci/Makefile
@@ -0,0 +1,78 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ad1889-objs := ad1889.o
+snd-als300-objs := als300.o
+snd-als4000-objs := als4000.o
+snd-atiixp-objs := atiixp.o
+snd-atiixp-modem-objs := atiixp_modem.o
+snd-azt3328-objs := azt3328.o
+snd-bt87x-objs := bt87x.o
+snd-cmipci-objs := cmipci.o
+snd-cs4281-objs := cs4281.o
+snd-cs5530-objs := cs5530.o
+snd-ens1370-objs := ens1370.o ak4531_codec.o
+snd-ens1371-objs := ens1371.o
+snd-es1938-objs := es1938.o
+snd-es1968-objs := es1968.o
+snd-fm801-objs := fm801.o
+snd-intel8x0-objs := intel8x0.o
+snd-intel8x0m-objs := intel8x0m.o
+snd-maestro3-objs := maestro3.o
+snd-rme32-objs := rme32.o
+snd-rme96-objs := rme96.o
+snd-sis7019-objs := sis7019.o
+snd-sonicvibes-objs := sonicvibes.o
+snd-via82xx-objs := via82xx.o
+snd-via82xx-modem-objs := via82xx_modem.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AD1889) += snd-ad1889.o
+obj-$(CONFIG_SND_ALS300) += snd-als300.o
+obj-$(CONFIG_SND_ALS4000) += snd-als4000.o
+obj-$(CONFIG_SND_ATIIXP) += snd-atiixp.o
+obj-$(CONFIG_SND_ATIIXP_MODEM) += snd-atiixp-modem.o
+obj-$(CONFIG_SND_AZT3328) += snd-azt3328.o
+obj-$(CONFIG_SND_BT87X) += snd-bt87x.o
+obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o
+obj-$(CONFIG_SND_CS4281) += snd-cs4281.o
+obj-$(CONFIG_SND_CS5530) += snd-cs5530.o
+obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o
+obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o
+obj-$(CONFIG_SND_ES1938) += snd-es1938.o
+obj-$(CONFIG_SND_ES1968) += snd-es1968.o
+obj-$(CONFIG_SND_FM801) += snd-fm801.o
+obj-$(CONFIG_SND_INTEL8X0) += snd-intel8x0.o
+obj-$(CONFIG_SND_INTEL8X0M) += snd-intel8x0m.o
+obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o
+obj-$(CONFIG_SND_RME32) += snd-rme32.o
+obj-$(CONFIG_SND_RME96) += snd-rme96.o
+obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o
+obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o
+obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o
+obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o
+
+obj-$(CONFIG_SND) += \
+ ac97/ \
+ ali5451/ \
+ au88x0/ \
+ aw2/ \
+ ca0106/ \
+ cs46xx/ \
+ cs5535audio/ \
+ echoaudio/ \
+ emu10k1/ \
+ hda/ \
+ ice1712/ \
+ korg1212/ \
+ mixart/ \
+ nm256/ \
+ oxygen/ \
+ pcxhr/ \
+ riptide/ \
+ rme9652/ \
+ trident/ \
+ ymfpci/ \
+ vx222/
diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile
new file mode 100644
index 0000000..41fa322
--- /dev/null
+++ b/sound/pci/ac97/Makefile
@@ -0,0 +1,10 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ac97-codec-y := ac97_codec.o ac97_pcm.o
+snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
new file mode 100644
index 0000000..bd510ec
--- /dev/null
+++ b/sound/pci/ac97/ac97_codec.c
@@ -0,0 +1,2903 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.2
+ * by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include <sound/ac97_codec.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include "ac97_id.h"
+
+#include "ac97_patch.c"
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Universal interface for Audio Codec '97");
+MODULE_LICENSE("GPL");
+
+static int enable_loopback;
+
+module_param(enable_loopback, bool, 0444);
+MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control");
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT;
+module_param(power_save, int, 0644);
+MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
+ "(in second, 0 = disable).");
+#endif
+/*
+
+ */
+
+struct ac97_codec_id {
+ unsigned int id;
+ unsigned int mask;
+ const char *name;
+ int (*patch)(struct snd_ac97 *ac97);
+ int (*mpatch)(struct snd_ac97 *ac97);
+ unsigned int flags;
+};
+
+static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = {
+{ 0x41445300, 0xffffff00, "Analog Devices", NULL, NULL },
+{ 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL },
+{ 0x414c4300, 0xffffff00, "Realtek", NULL, NULL },
+{ 0x414c4700, 0xffffff00, "Realtek", NULL, NULL },
+{ 0x434d4900, 0xffffff00, "C-Media Electronics", NULL, NULL },
+{ 0x43525900, 0xffffff00, "Cirrus Logic", NULL, NULL },
+{ 0x43585400, 0xffffff00, "Conexant", NULL, NULL },
+{ 0x44543000, 0xffffff00, "Diamond Technology", NULL, NULL },
+{ 0x454d4300, 0xffffff00, "eMicro", NULL, NULL },
+{ 0x45838300, 0xffffff00, "ESS Technology", NULL, NULL },
+{ 0x48525300, 0xffffff00, "Intersil", NULL, NULL },
+{ 0x49434500, 0xffffff00, "ICEnsemble", NULL, NULL },
+{ 0x49544500, 0xffffff00, "ITE Tech.Inc", NULL, NULL },
+{ 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL },
+{ 0x50534300, 0xffffff00, "Philips", NULL, NULL },
+{ 0x53494c00, 0xffffff00, "Silicon Laboratory", NULL, NULL },
+{ 0x54524100, 0xffffff00, "TriTech", NULL, NULL },
+{ 0x54584e00, 0xffffff00, "Texas Instruments", NULL, NULL },
+{ 0x56494100, 0xffffff00, "VIA Technologies", NULL, NULL },
+{ 0x57454300, 0xffffff00, "Winbond", NULL, NULL },
+{ 0x574d4c00, 0xffffff00, "Wolfson", NULL, NULL },
+{ 0x594d4800, 0xffffff00, "Yamaha", NULL, NULL },
+{ 0x83847600, 0xffffff00, "SigmaTel", NULL, NULL },
+{ 0, 0, NULL, NULL, NULL }
+};
+
+static const struct ac97_codec_id snd_ac97_codec_ids[] = {
+{ 0x41445303, 0xffffffff, "AD1819", patch_ad1819, NULL },
+{ 0x41445340, 0xffffffff, "AD1881", patch_ad1881, NULL },
+{ 0x41445348, 0xffffffff, "AD1881A", patch_ad1881, NULL },
+{ 0x41445360, 0xffffffff, "AD1885", patch_ad1885, NULL },
+{ 0x41445361, 0xffffffff, "AD1886", patch_ad1886, NULL },
+{ 0x41445362, 0xffffffff, "AD1887", patch_ad1881, NULL },
+{ 0x41445363, 0xffffffff, "AD1886A", patch_ad1881, NULL },
+{ 0x41445368, 0xffffffff, "AD1888", patch_ad1888, NULL },
+{ 0x41445370, 0xffffffff, "AD1980", patch_ad1980, NULL },
+{ 0x41445372, 0xffffffff, "AD1981A", patch_ad1981a, NULL },
+{ 0x41445374, 0xffffffff, "AD1981B", patch_ad1981b, NULL },
+{ 0x41445375, 0xffffffff, "AD1985", patch_ad1985, NULL },
+{ 0x41445378, 0xffffffff, "AD1986", patch_ad1986, NULL },
+{ 0x414b4d00, 0xffffffff, "AK4540", NULL, NULL },
+{ 0x414b4d01, 0xffffffff, "AK4542", NULL, NULL },
+{ 0x414b4d02, 0xffffffff, "AK4543", NULL, NULL },
+{ 0x414b4d06, 0xffffffff, "AK4544A", NULL, NULL },
+{ 0x414b4d07, 0xffffffff, "AK4545", NULL, NULL },
+{ 0x414c4300, 0xffffff00, "ALC100,100P", NULL, NULL },
+{ 0x414c4710, 0xfffffff0, "ALC200,200P", NULL, NULL },
+{ 0x414c4721, 0xffffffff, "ALC650D", NULL, NULL }, /* already patched */
+{ 0x414c4722, 0xffffffff, "ALC650E", NULL, NULL }, /* already patched */
+{ 0x414c4723, 0xffffffff, "ALC650F", NULL, NULL }, /* already patched */
+{ 0x414c4720, 0xfffffff0, "ALC650", patch_alc650, NULL },
+{ 0x414c4730, 0xffffffff, "ALC101", NULL, NULL },
+{ 0x414c4740, 0xfffffff0, "ALC202", NULL, NULL },
+{ 0x414c4750, 0xfffffff0, "ALC250", NULL, NULL },
+{ 0x414c4760, 0xfffffff0, "ALC655", patch_alc655, NULL },
+{ 0x414c4770, 0xfffffff0, "ALC203", patch_alc203, NULL },
+{ 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */
+{ 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL },
+{ 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL },
+{ 0x434d4941, 0xffffffff, "CMI9738", patch_cm9738, NULL },
+{ 0x434d4961, 0xffffffff, "CMI9739", patch_cm9739, NULL },
+{ 0x434d4969, 0xffffffff, "CMI9780", patch_cm9780, NULL },
+{ 0x434d4978, 0xffffffff, "CMI9761A", patch_cm9761, NULL },
+{ 0x434d4982, 0xffffffff, "CMI9761B", patch_cm9761, NULL },
+{ 0x434d4983, 0xffffffff, "CMI9761A+", patch_cm9761, NULL },
+{ 0x43525900, 0xfffffff8, "CS4297", NULL, NULL },
+{ 0x43525910, 0xfffffff8, "CS4297A", patch_cirrus_spdif, NULL },
+{ 0x43525920, 0xfffffff8, "CS4298", patch_cirrus_spdif, NULL },
+{ 0x43525928, 0xfffffff8, "CS4294", NULL, NULL },
+{ 0x43525930, 0xfffffff8, "CS4299", patch_cirrus_cs4299, NULL },
+{ 0x43525948, 0xfffffff8, "CS4201", NULL, NULL },
+{ 0x43525958, 0xfffffff8, "CS4205", patch_cirrus_spdif, NULL },
+{ 0x43525960, 0xfffffff8, "CS4291", NULL, NULL },
+{ 0x43525970, 0xfffffff8, "CS4202", NULL, NULL },
+{ 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, // SmartMC II
+{ 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, // SmartAMC fixme: the mask might be different
+{ 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL },
+{ 0x44543031, 0xfffffff0, "DT0398", NULL, NULL },
+{ 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, // same as TR28028?
+{ 0x45838308, 0xffffffff, "ESS1988", NULL, NULL },
+{ 0x48525300, 0xffffff00, "HMP9701", NULL, NULL },
+{ 0x49434501, 0xffffffff, "ICE1230", NULL, NULL },
+{ 0x49434511, 0xffffffff, "ICE1232", NULL, NULL }, // alias VIA VT1611A?
+{ 0x49434514, 0xffffffff, "ICE1232A", NULL, NULL },
+{ 0x49434551, 0xffffffff, "VT1616", patch_vt1616, NULL },
+{ 0x49434552, 0xffffffff, "VT1616i", patch_vt1616, NULL }, // VT1616 compatible (chipset integrated)
+{ 0x49544520, 0xffffffff, "IT2226E", NULL, NULL },
+{ 0x49544561, 0xffffffff, "IT2646E", patch_it2646, NULL },
+{ 0x4e534300, 0xffffffff, "LM4540,43,45,46,48", NULL, NULL }, // only guess --jk
+{ 0x4e534331, 0xffffffff, "LM4549", NULL, NULL },
+{ 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix
+{ 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL },
+{ 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH },
+{ 0x54524102, 0xffffffff, "TR28022", NULL, NULL },
+{ 0x54524103, 0xffffffff, "TR28023", NULL, NULL },
+{ 0x54524106, 0xffffffff, "TR28026", NULL, NULL },
+{ 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99]
+{ 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)]
+{ 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL },
+{ 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF
+{ 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF
+{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL },
+{ 0x57454301, 0xffffffff, "W83971D", NULL, NULL },
+{ 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL },
+{ 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL},
+{ 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL},
+{ 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL},
+{ 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL},
+{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL},
+{ 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF},
+{ 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL },
+{ 0x594d4802, 0xffffffff, "YMF752", NULL, NULL },
+{ 0x594d4803, 0xffffffff, "YMF753", patch_yamaha_ymf753, NULL },
+{ 0x83847600, 0xffffffff, "STAC9700,83,84", patch_sigmatel_stac9700, NULL },
+{ 0x83847604, 0xffffffff, "STAC9701,3,4,5", NULL, NULL },
+{ 0x83847605, 0xffffffff, "STAC9704", NULL, NULL },
+{ 0x83847608, 0xffffffff, "STAC9708,11", patch_sigmatel_stac9708, NULL },
+{ 0x83847609, 0xffffffff, "STAC9721,23", patch_sigmatel_stac9721, NULL },
+{ 0x83847644, 0xffffffff, "STAC9744", patch_sigmatel_stac9744, NULL },
+{ 0x83847650, 0xffffffff, "STAC9750,51", NULL, NULL }, // patch?
+{ 0x83847652, 0xffffffff, "STAC9752,53", NULL, NULL }, // patch?
+{ 0x83847656, 0xffffffff, "STAC9756,57", patch_sigmatel_stac9756, NULL },
+{ 0x83847658, 0xffffffff, "STAC9758,59", patch_sigmatel_stac9758, NULL },
+{ 0x83847666, 0xffffffff, "STAC9766,67", NULL, NULL }, // patch?
+{ 0, 0, NULL, NULL, NULL }
+};
+
+
+static void update_power_regs(struct snd_ac97 *ac97);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+#define ac97_is_power_save_mode(ac97) \
+ ((ac97->scaps & AC97_SCAP_POWER_SAVE) && power_save)
+#else
+#define ac97_is_power_save_mode(ac97) 0
+#endif
+
+
+/*
+ * I/O routines
+ */
+
+static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg)
+{
+ /* filter some registers for buggy codecs */
+ switch (ac97->id) {
+ case AC97_ID_AK4540:
+ case AC97_ID_AK4542:
+ if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c)
+ return 1;
+ return 0;
+ case AC97_ID_AD1819: /* AD1819 */
+ case AC97_ID_AD1881: /* AD1881 */
+ case AC97_ID_AD1881A: /* AD1881A */
+ if (reg >= 0x3a && reg <= 0x6e) /* 0x59 */
+ return 0;
+ return 1;
+ case AC97_ID_AD1885: /* AD1885 */
+ case AC97_ID_AD1886: /* AD1886 */
+ case AC97_ID_AD1886A: /* AD1886A - !!verify!! --jk */
+ case AC97_ID_AD1887: /* AD1887 - !!verify!! --jk */
+ if (reg == 0x5a)
+ return 1;
+ if (reg >= 0x3c && reg <= 0x6e) /* 0x59 */
+ return 0;
+ return 1;
+ case AC97_ID_STAC9700:
+ case AC97_ID_STAC9704:
+ case AC97_ID_STAC9705:
+ case AC97_ID_STAC9708:
+ case AC97_ID_STAC9721:
+ case AC97_ID_STAC9744:
+ case AC97_ID_STAC9756:
+ if (reg <= 0x3a || reg >= 0x5a)
+ return 1;
+ return 0;
+ }
+ return 1;
+}
+
+/**
+ * snd_ac97_write - write a value on the given register
+ * @ac97: the ac97 instance
+ * @reg: the register to change
+ * @value: the value to set
+ *
+ * Writes a value on the given register. This will invoke the write
+ * callback directly after the register check.
+ * This function doesn't change the register cache unlike
+ * #snd_ca97_write_cache(), so use this only when you don't want to
+ * reflect the change to the suspend/resume state.
+ */
+void snd_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short value)
+{
+ if (!snd_ac97_valid_reg(ac97, reg))
+ return;
+ if ((ac97->id & 0xffffff00) == AC97_ID_ALC100) {
+ /* Fix H/W bug of ALC100/100P */
+ if (reg == AC97_MASTER || reg == AC97_HEADPHONE)
+ ac97->bus->ops->write(ac97, AC97_RESET, 0); /* reset audio codec */
+ }
+ ac97->bus->ops->write(ac97, reg, value);
+}
+
+EXPORT_SYMBOL(snd_ac97_write);
+
+/**
+ * snd_ac97_read - read a value from the given register
+ *
+ * @ac97: the ac97 instance
+ * @reg: the register to read
+ *
+ * Reads a value from the given register. This will invoke the read
+ * callback directly after the register check.
+ *
+ * Returns the read value.
+ */
+unsigned short snd_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ if (!snd_ac97_valid_reg(ac97, reg))
+ return 0;
+ return ac97->bus->ops->read(ac97, reg);
+}
+
+/* read a register - return the cached value if already read */
+static inline unsigned short snd_ac97_read_cache(struct snd_ac97 *ac97, unsigned short reg)
+{
+ if (! test_bit(reg, ac97->reg_accessed)) {
+ ac97->regs[reg] = ac97->bus->ops->read(ac97, reg);
+ // set_bit(reg, ac97->reg_accessed);
+ }
+ return ac97->regs[reg];
+}
+
+EXPORT_SYMBOL(snd_ac97_read);
+
+/**
+ * snd_ac97_write_cache - write a value on the given register and update the cache
+ * @ac97: the ac97 instance
+ * @reg: the register to change
+ * @value: the value to set
+ *
+ * Writes a value on the given register and updates the register
+ * cache. The cached values are used for the cached-read and the
+ * suspend/resume.
+ */
+void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned short value)
+{
+ if (!snd_ac97_valid_reg(ac97, reg))
+ return;
+ mutex_lock(&ac97->reg_mutex);
+ ac97->regs[reg] = value;
+ ac97->bus->ops->write(ac97, reg, value);
+ set_bit(reg, ac97->reg_accessed);
+ mutex_unlock(&ac97->reg_mutex);
+}
+
+EXPORT_SYMBOL(snd_ac97_write_cache);
+
+/**
+ * snd_ac97_update - update the value on the given register
+ * @ac97: the ac97 instance
+ * @reg: the register to change
+ * @value: the value to set
+ *
+ * Compares the value with the register cache and updates the value
+ * only when the value is changed.
+ *
+ * Returns 1 if the value is changed, 0 if no change, or a negative
+ * code on failure.
+ */
+int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short value)
+{
+ int change;
+
+ if (!snd_ac97_valid_reg(ac97, reg))
+ return -EINVAL;
+ mutex_lock(&ac97->reg_mutex);
+ change = ac97->regs[reg] != value;
+ if (change) {
+ ac97->regs[reg] = value;
+ ac97->bus->ops->write(ac97, reg, value);
+ }
+ set_bit(reg, ac97->reg_accessed);
+ mutex_unlock(&ac97->reg_mutex);
+ return change;
+}
+
+EXPORT_SYMBOL(snd_ac97_update);
+
+/**
+ * snd_ac97_update_bits - update the bits on the given register
+ * @ac97: the ac97 instance
+ * @reg: the register to change
+ * @mask: the bit-mask to change
+ * @value: the value to set
+ *
+ * Updates the masked-bits on the given register only when the value
+ * is changed.
+ *
+ * Returns 1 if the bits are changed, 0 if no change, or a negative
+ * code on failure.
+ */
+int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value)
+{
+ int change;
+
+ if (!snd_ac97_valid_reg(ac97, reg))
+ return -EINVAL;
+ mutex_lock(&ac97->reg_mutex);
+ change = snd_ac97_update_bits_nolock(ac97, reg, mask, value);
+ mutex_unlock(&ac97->reg_mutex);
+ return change;
+}
+
+EXPORT_SYMBOL(snd_ac97_update_bits);
+
+/* no lock version - see snd_ac97_updat_bits() */
+int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short mask, unsigned short value)
+{
+ int change;
+ unsigned short old, new;
+
+ old = snd_ac97_read_cache(ac97, reg);
+ new = (old & ~mask) | (value & mask);
+ change = old != new;
+ if (change) {
+ ac97->regs[reg] = new;
+ ac97->bus->ops->write(ac97, reg, new);
+ }
+ set_bit(reg, ac97->reg_accessed);
+ return change;
+}
+
+static int snd_ac97_ad18xx_update_pcm_bits(struct snd_ac97 *ac97, int codec, unsigned short mask, unsigned short value)
+{
+ int change;
+ unsigned short old, new, cfg;
+
+ mutex_lock(&ac97->page_mutex);
+ old = ac97->spec.ad18xx.pcmreg[codec];
+ new = (old & ~mask) | (value & mask);
+ change = old != new;
+ if (change) {
+ mutex_lock(&ac97->reg_mutex);
+ cfg = snd_ac97_read_cache(ac97, AC97_AD_SERIAL_CFG);
+ ac97->spec.ad18xx.pcmreg[codec] = new;
+ /* select single codec */
+ ac97->bus->ops->write(ac97, AC97_AD_SERIAL_CFG,
+ (cfg & ~0x7000) |
+ ac97->spec.ad18xx.unchained[codec] | ac97->spec.ad18xx.chained[codec]);
+ /* update PCM bits */
+ ac97->bus->ops->write(ac97, AC97_PCM, new);
+ /* select all codecs */
+ ac97->bus->ops->write(ac97, AC97_AD_SERIAL_CFG,
+ cfg | 0x7000);
+ mutex_unlock(&ac97->reg_mutex);
+ }
+ mutex_unlock(&ac97->page_mutex);
+ return change;
+}
+
+/*
+ * Controls
+ */
+
+static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
+ uinfo->value.enumerated.items = e->mask;
+
+ if (uinfo->value.enumerated.item > e->mask - 1)
+ uinfo->value.enumerated.item = e->mask - 1;
+ strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value;
+ unsigned short val, bitmask;
+
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
+ val = snd_ac97_read_cache(ac97, e->reg);
+ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+ if (e->shift_l != e->shift_r)
+ ucontrol->value.enumerated.item[1] = (val >> e->shift_r) & (bitmask - 1);
+
+ return 0;
+}
+
+static int snd_ac97_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value;
+ unsigned short val;
+ unsigned short mask, bitmask;
+
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
+ if (ucontrol->value.enumerated.item[0] > e->mask - 1)
+ return -EINVAL;
+ val = ucontrol->value.enumerated.item[0] << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
+ if (e->shift_l != e->shift_r) {
+ if (ucontrol->value.enumerated.item[1] > e->mask - 1)
+ return -EINVAL;
+ val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
+ }
+ return snd_ac97_update_bits(ac97, e->reg, mask, val);
+}
+
+/* save/restore ac97 v2.3 paging */
+static int snd_ac97_page_save(struct snd_ac97 *ac97, int reg, struct snd_kcontrol *kcontrol)
+{
+ int page_save = -1;
+ if ((kcontrol->private_value & (1<<25)) &&
+ (ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23 &&
+ (reg >= 0x60 && reg < 0x70)) {
+ unsigned short page = (kcontrol->private_value >> 26) & 0x0f;
+ mutex_lock(&ac97->page_mutex); /* lock paging */
+ page_save = snd_ac97_read(ac97, AC97_INT_PAGING) & AC97_PAGE_MASK;
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page);
+ }
+ return page_save;
+}
+
+static void snd_ac97_page_restore(struct snd_ac97 *ac97, int page_save)
+{
+ if (page_save >= 0) {
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page_save);
+ mutex_unlock(&ac97->page_mutex); /* unlock paging */
+ }
+}
+
+/* volume and switch controls */
+static int snd_ac97_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = shift == rshift ? 1 : 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_ac97_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0x01;
+ int page_save;
+
+ page_save = snd_ac97_page_save(ac97, reg, kcontrol);
+ ucontrol->value.integer.value[0] = (snd_ac97_read_cache(ac97, reg) >> shift) & mask;
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] = (snd_ac97_read_cache(ac97, reg) >> rshift) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ snd_ac97_page_restore(ac97, page_save);
+ return 0;
+}
+
+static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0x01;
+ int err, page_save;
+ unsigned short val, val2, val_mask;
+
+ page_save = snd_ac97_page_save(ac97, reg, kcontrol);
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val_mask = mask << shift;
+ val = val << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1] & mask);
+ if (invert)
+ val2 = mask - val2;
+ val_mask |= mask << rshift;
+ val |= val2 << rshift;
+ }
+ err = snd_ac97_update_bits(ac97, reg, val_mask, val);
+ snd_ac97_page_restore(ac97, page_save);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ /* check analog mixer power-down */
+ if ((val_mask & 0x8000) &&
+ (kcontrol->private_value & (1<<30))) {
+ if (val & 0x8000)
+ ac97->power_up &= ~(1 << (reg>>1));
+ else
+ ac97->power_up |= 1 << (reg>>1);
+ update_power_regs(ac97);
+ }
+#endif
+ return err;
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_master_mono[2] = {
+AC97_SINGLE("Master Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+AC97_SINGLE("Master Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_tone[2] = {
+AC97_SINGLE("Tone Control - Bass", AC97_MASTER_TONE, 8, 15, 1),
+AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_pc_beep[2] = {
+AC97_SINGLE("PC Speaker Playback Switch", AC97_PC_BEEP, 15, 1, 1),
+AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_mic_boost =
+ AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0);
+
+
+static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"};
+static const char* std_3d_path[] = {"pre 3D", "post 3D"};
+static const char* std_mix[] = {"Mix", "Mic"};
+static const char* std_mic[] = {"Mic1", "Mic2"};
+
+static const struct ac97_enum std_enum[] = {
+AC97_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, std_rec_sel),
+AC97_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, std_3d_path),
+AC97_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, std_mix),
+AC97_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, std_mic),
+};
+
+static const struct snd_kcontrol_new snd_ac97_control_capture_src =
+AC97_ENUM("Capture Source", std_enum[0]);
+
+static const struct snd_kcontrol_new snd_ac97_control_capture_vol =
+AC97_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0);
+
+static const struct snd_kcontrol_new snd_ac97_controls_mic_capture[2] = {
+AC97_SINGLE("Mic Capture Switch", AC97_REC_GAIN_MIC, 15, 1, 1),
+AC97_SINGLE("Mic Capture Volume", AC97_REC_GAIN_MIC, 0, 15, 0)
+};
+
+enum {
+ AC97_GENERAL_PCM_OUT = 0,
+ AC97_GENERAL_STEREO_ENHANCEMENT,
+ AC97_GENERAL_3D,
+ AC97_GENERAL_LOUDNESS,
+ AC97_GENERAL_MONO,
+ AC97_GENERAL_MIC,
+ AC97_GENERAL_LOOPBACK
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_general[7] = {
+AC97_ENUM("PCM Out Path & Mute", std_enum[1]),
+AC97_SINGLE("Simulated Stereo Enhancement", AC97_GENERAL_PURPOSE, 14, 1, 0),
+AC97_SINGLE("3D Control - Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+AC97_SINGLE("Loudness (bass boost)", AC97_GENERAL_PURPOSE, 12, 1, 0),
+AC97_ENUM("Mono Output Select", std_enum[2]),
+AC97_ENUM("Mic Select", std_enum[3]),
+AC97_SINGLE("ADC/DAC Loopback", AC97_GENERAL_PURPOSE, 7, 1, 0)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_3d[2] = {
+AC97_SINGLE("3D Control - Center", AC97_3D_CONTROL, 8, 15, 0),
+AC97_SINGLE("3D Control - Depth", AC97_3D_CONTROL, 0, 15, 0)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_center[2] = {
+AC97_SINGLE("Center Playback Switch", AC97_CENTER_LFE_MASTER, 7, 1, 1),
+AC97_SINGLE("Center Playback Volume", AC97_CENTER_LFE_MASTER, 0, 31, 1)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_lfe[2] = {
+AC97_SINGLE("LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 1, 1),
+AC97_SINGLE("LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 31, 1)
+};
+
+static const struct snd_kcontrol_new snd_ac97_control_eapd =
+AC97_SINGLE("External Amplifier", AC97_POWERDOWN, 15, 1, 1);
+
+static const struct snd_kcontrol_new snd_ac97_controls_modem_switches[2] = {
+AC97_SINGLE("Off-hook Switch", AC97_GPIO_STATUS, 0, 1, 0),
+AC97_SINGLE("Caller ID Switch", AC97_GPIO_STATUS, 2, 1, 0)
+};
+
+/* change the existing EAPD control as inverted */
+static void set_inv_eapd(struct snd_ac97 *ac97, struct snd_kcontrol *kctl)
+{
+ kctl->private_value = AC97_SINGLE_VALUE(AC97_POWERDOWN, 15, 1, 0);
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN, (1<<15), (1<<15)); /* EAPD up */
+ ac97->scaps |= AC97_SCAP_INV_EAPD;
+}
+
+static int snd_ac97_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ac97_spdif_cmask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_NONAUDIO |
+ IEC958_AES0_CON_EMPHASIS_5015 |
+ IEC958_AES0_CON_NOT_COPYRIGHT;
+ ucontrol->value.iec958.status[1] = IEC958_AES1_CON_CATEGORY |
+ IEC958_AES1_CON_ORIGINAL;
+ ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS;
+ return 0;
+}
+
+static int snd_ac97_spdif_pmask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ /* FIXME: AC'97 spec doesn't say which bits are used for what */
+ ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PRO_FS |
+ IEC958_AES0_PRO_EMPHASIS_5015;
+ return 0;
+}
+
+static int snd_ac97_spdif_default_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ac97->reg_mutex);
+ ucontrol->value.iec958.status[0] = ac97->spdif_status & 0xff;
+ ucontrol->value.iec958.status[1] = (ac97->spdif_status >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (ac97->spdif_status >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (ac97->spdif_status >> 24) & 0xff;
+ mutex_unlock(&ac97->reg_mutex);
+ return 0;
+}
+
+static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned int new = 0;
+ unsigned short val = 0;
+ int change;
+
+ new = val = ucontrol->value.iec958.status[0] & (IEC958_AES0_PROFESSIONAL|IEC958_AES0_NONAUDIO);
+ if (ucontrol->value.iec958.status[0] & IEC958_AES0_PROFESSIONAL) {
+ new |= ucontrol->value.iec958.status[0] & (IEC958_AES0_PRO_FS|IEC958_AES0_PRO_EMPHASIS_5015);
+ switch (new & IEC958_AES0_PRO_FS) {
+ case IEC958_AES0_PRO_FS_44100: val |= 0<<12; break;
+ case IEC958_AES0_PRO_FS_48000: val |= 2<<12; break;
+ case IEC958_AES0_PRO_FS_32000: val |= 3<<12; break;
+ default: val |= 1<<12; break;
+ }
+ if ((new & IEC958_AES0_PRO_EMPHASIS) == IEC958_AES0_PRO_EMPHASIS_5015)
+ val |= 1<<3;
+ } else {
+ new |= ucontrol->value.iec958.status[0] & (IEC958_AES0_CON_EMPHASIS_5015|IEC958_AES0_CON_NOT_COPYRIGHT);
+ new |= ((ucontrol->value.iec958.status[1] & (IEC958_AES1_CON_CATEGORY|IEC958_AES1_CON_ORIGINAL)) << 8);
+ new |= ((ucontrol->value.iec958.status[3] & IEC958_AES3_CON_FS) << 24);
+ if ((new & IEC958_AES0_CON_EMPHASIS) == IEC958_AES0_CON_EMPHASIS_5015)
+ val |= 1<<3;
+ if (!(new & IEC958_AES0_CON_NOT_COPYRIGHT))
+ val |= 1<<2;
+ val |= ((new >> 8) & 0xff) << 4; // category + original
+ switch ((new >> 24) & 0xff) {
+ case IEC958_AES3_CON_FS_44100: val |= 0<<12; break;
+ case IEC958_AES3_CON_FS_48000: val |= 2<<12; break;
+ case IEC958_AES3_CON_FS_32000: val |= 3<<12; break;
+ default: val |= 1<<12; break;
+ }
+ }
+
+ mutex_lock(&ac97->reg_mutex);
+ change = ac97->spdif_status != new;
+ ac97->spdif_status = new;
+
+ if (ac97->flags & AC97_CS_SPDIF) {
+ int x = (val >> 12) & 0x03;
+ switch (x) {
+ case 0: x = 1; break; // 44.1
+ case 2: x = 0; break; // 48.0
+ default: x = 0; break; // illegal.
+ }
+ change |= snd_ac97_update_bits_nolock(ac97, AC97_CSR_SPDIF, 0x3fff, ((val & 0xcfff) | (x << 12)));
+ } else if (ac97->flags & AC97_CX_SPDIF) {
+ int v;
+ v = new & (IEC958_AES0_CON_EMPHASIS_5015|IEC958_AES0_CON_NOT_COPYRIGHT) ? 0 : AC97_CXR_COPYRGT;
+ v |= new & IEC958_AES0_NONAUDIO ? AC97_CXR_SPDIF_AC3 : AC97_CXR_SPDIF_PCM;
+ change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC,
+ AC97_CXR_SPDIF_MASK | AC97_CXR_COPYRGT,
+ v);
+ } else if (ac97->id == AC97_ID_YMF743) {
+ change |= snd_ac97_update_bits_nolock(ac97,
+ AC97_YMF7X3_DIT_CTRL,
+ 0xff38,
+ ((val << 4) & 0xff00) |
+ ((val << 2) & 0x0038));
+ } else {
+ unsigned short extst = snd_ac97_read_cache(ac97, AC97_EXTENDED_STATUS);
+ snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); /* turn off */
+
+ change |= snd_ac97_update_bits_nolock(ac97, AC97_SPDIF, 0x3fff, val);
+ if (extst & AC97_EA_SPDIF) {
+ snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, AC97_EA_SPDIF); /* turn on again */
+ }
+ }
+ mutex_unlock(&ac97->reg_mutex);
+
+ return change;
+}
+
+static int snd_ac97_put_spsa(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ // int invert = (kcontrol->private_value >> 24) & 0xff;
+ unsigned short value, old, new;
+ int change;
+
+ value = (ucontrol->value.integer.value[0] & mask);
+
+ mutex_lock(&ac97->reg_mutex);
+ mask <<= shift;
+ value <<= shift;
+ old = snd_ac97_read_cache(ac97, reg);
+ new = (old & ~mask) | value;
+ change = old != new;
+
+ if (change) {
+ unsigned short extst = snd_ac97_read_cache(ac97, AC97_EXTENDED_STATUS);
+ snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); /* turn off */
+ change = snd_ac97_update_bits_nolock(ac97, reg, mask, value);
+ if (extst & AC97_EA_SPDIF)
+ snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, AC97_EA_SPDIF); /* turn on again */
+ }
+ mutex_unlock(&ac97->reg_mutex);
+ return change;
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_spdif[5] = {
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_ac97_spdif_mask_info,
+ .get = snd_ac97_spdif_cmask_get,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+ .info = snd_ac97_spdif_mask_info,
+ .get = snd_ac97_spdif_pmask_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_ac97_spdif_mask_info,
+ .get = snd_ac97_spdif_default_get,
+ .put = snd_ac97_spdif_default_put,
+ },
+
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH),AC97_EXTENDED_STATUS, 2, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "AC97-SPSA",
+ .info = snd_ac97_info_volsw,
+ .get = snd_ac97_get_volsw,
+ .put = snd_ac97_put_spsa,
+ .private_value = AC97_SINGLE_VALUE(AC97_EXTENDED_STATUS, 4, 3, 0)
+ },
+};
+
+#define AD18XX_PCM_BITS(xname, codec, lshift, rshift, mask) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_ac97_ad18xx_pcm_info_bits, \
+ .get = snd_ac97_ad18xx_pcm_get_bits, .put = snd_ac97_ad18xx_pcm_put_bits, \
+ .private_value = (codec) | ((lshift) << 8) | ((rshift) << 12) | ((mask) << 16) }
+
+static int snd_ac97_ad18xx_pcm_info_bits(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int mask = (kcontrol->private_value >> 16) & 0x0f;
+ int lshift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ if (lshift != rshift && (ac97->flags & AC97_STEREO_MUTES))
+ uinfo->count = 2;
+ else
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_ac97_ad18xx_pcm_get_bits(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int codec = kcontrol->private_value & 3;
+ int lshift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ ucontrol->value.integer.value[0] = mask - ((ac97->spec.ad18xx.pcmreg[codec] >> lshift) & mask);
+ if (lshift != rshift && (ac97->flags & AC97_STEREO_MUTES))
+ ucontrol->value.integer.value[1] = mask - ((ac97->spec.ad18xx.pcmreg[codec] >> rshift) & mask);
+ return 0;
+}
+
+static int snd_ac97_ad18xx_pcm_put_bits(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int codec = kcontrol->private_value & 3;
+ int lshift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ unsigned short val, valmask;
+
+ val = (mask - (ucontrol->value.integer.value[0] & mask)) << lshift;
+ valmask = mask << lshift;
+ if (lshift != rshift && (ac97->flags & AC97_STEREO_MUTES)) {
+ val |= (mask - (ucontrol->value.integer.value[1] & mask)) << rshift;
+ valmask |= mask << rshift;
+ }
+ return snd_ac97_ad18xx_update_pcm_bits(ac97, codec, valmask, val);
+}
+
+#define AD18XX_PCM_VOLUME(xname, codec) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_ac97_ad18xx_pcm_info_volume, \
+ .get = snd_ac97_ad18xx_pcm_get_volume, .put = snd_ac97_ad18xx_pcm_put_volume, \
+ .private_value = codec }
+
+static int snd_ac97_ad18xx_pcm_info_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 31;
+ return 0;
+}
+
+static int snd_ac97_ad18xx_pcm_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int codec = kcontrol->private_value & 3;
+
+ mutex_lock(&ac97->page_mutex);
+ ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31);
+ ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31);
+ mutex_unlock(&ac97->page_mutex);
+ return 0;
+}
+
+static int snd_ac97_ad18xx_pcm_put_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int codec = kcontrol->private_value & 3;
+ unsigned short val1, val2;
+
+ val1 = 31 - (ucontrol->value.integer.value[0] & 31);
+ val2 = 31 - (ucontrol->value.integer.value[1] & 31);
+ return snd_ac97_ad18xx_update_pcm_bits(ac97, codec, 0x1f1f, (val1 << 8) | val2);
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_ad18xx_pcm[2] = {
+AD18XX_PCM_BITS("PCM Playback Switch", 0, 15, 7, 1),
+AD18XX_PCM_VOLUME("PCM Playback Volume", 0)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_ad18xx_surround[2] = {
+AD18XX_PCM_BITS("Surround Playback Switch", 1, 15, 7, 1),
+AD18XX_PCM_VOLUME("Surround Playback Volume", 1)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_ad18xx_center[2] = {
+AD18XX_PCM_BITS("Center Playback Switch", 2, 15, 15, 1),
+AD18XX_PCM_BITS("Center Playback Volume", 2, 8, 8, 31)
+};
+
+static const struct snd_kcontrol_new snd_ac97_controls_ad18xx_lfe[2] = {
+AD18XX_PCM_BITS("LFE Playback Switch", 2, 7, 7, 1),
+AD18XX_PCM_BITS("LFE Playback Volume", 2, 0, 0, 31)
+};
+
+/*
+ *
+ */
+
+static void snd_ac97_powerdown(struct snd_ac97 *ac97);
+
+static int snd_ac97_bus_free(struct snd_ac97_bus *bus)
+{
+ if (bus) {
+ snd_ac97_bus_proc_done(bus);
+ kfree(bus->pcms);
+ if (bus->private_free)
+ bus->private_free(bus);
+ kfree(bus);
+ }
+ return 0;
+}
+
+static int snd_ac97_bus_dev_free(struct snd_device *device)
+{
+ struct snd_ac97_bus *bus = device->device_data;
+ return snd_ac97_bus_free(bus);
+}
+
+static int snd_ac97_free(struct snd_ac97 *ac97)
+{
+ if (ac97) {
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ cancel_delayed_work(&ac97->power_work);
+ flush_scheduled_work();
+#endif
+ snd_ac97_proc_done(ac97);
+ if (ac97->bus)
+ ac97->bus->codec[ac97->num] = NULL;
+ if (ac97->private_free)
+ ac97->private_free(ac97);
+ kfree(ac97);
+ }
+ return 0;
+}
+
+static int snd_ac97_dev_free(struct snd_device *device)
+{
+ struct snd_ac97 *ac97 = device->device_data;
+ snd_ac97_powerdown(ac97); /* for avoiding click noises during shut down */
+ return snd_ac97_free(ac97);
+}
+
+static int snd_ac97_try_volume_mix(struct snd_ac97 * ac97, int reg)
+{
+ unsigned short val, mask = 0x8000;
+
+ if (! snd_ac97_valid_reg(ac97, reg))
+ return 0;
+
+ switch (reg) {
+ case AC97_MASTER_TONE:
+ return ac97->caps & 0x04 ? 1 : 0;
+ case AC97_HEADPHONE:
+ return ac97->caps & 0x10 ? 1 : 0;
+ case AC97_REC_GAIN_MIC:
+ return ac97->caps & 0x01 ? 1 : 0;
+ case AC97_3D_CONTROL:
+ if (ac97->caps & 0x7c00) {
+ val = snd_ac97_read(ac97, reg);
+ /* if nonzero - fixed and we can't set it */
+ return val == 0;
+ }
+ return 0;
+ case AC97_CENTER_LFE_MASTER: /* center */
+ if ((ac97->ext_id & AC97_EI_CDAC) == 0)
+ return 0;
+ break;
+ case AC97_CENTER_LFE_MASTER+1: /* lfe */
+ if ((ac97->ext_id & AC97_EI_LDAC) == 0)
+ return 0;
+ reg = AC97_CENTER_LFE_MASTER;
+ mask = 0x0080;
+ break;
+ case AC97_SURROUND_MASTER:
+ if ((ac97->ext_id & AC97_EI_SDAC) == 0)
+ return 0;
+ break;
+ }
+
+ val = snd_ac97_read(ac97, reg);
+ if (!(val & mask)) {
+ /* nothing seems to be here - mute flag is not set */
+ /* try another test */
+ snd_ac97_write_cache(ac97, reg, val | mask);
+ val = snd_ac97_read(ac97, reg);
+ val = snd_ac97_read(ac97, reg);
+ if (!(val & mask))
+ return 0; /* nothing here */
+ }
+ return 1; /* success, useable */
+}
+
+static void check_volume_resolution(struct snd_ac97 *ac97, int reg, unsigned char *lo_max, unsigned char *hi_max)
+{
+ unsigned short cbit[3] = { 0x20, 0x10, 0x01 };
+ unsigned char max[3] = { 63, 31, 15 };
+ int i;
+
+ /* first look up the static resolution table */
+ if (ac97->res_table) {
+ const struct snd_ac97_res_table *tbl;
+ for (tbl = ac97->res_table; tbl->reg; tbl++) {
+ if (tbl->reg == reg) {
+ *lo_max = tbl->bits & 0xff;
+ *hi_max = (tbl->bits >> 8) & 0xff;
+ return;
+ }
+ }
+ }
+
+ *lo_max = *hi_max = 0;
+ for (i = 0 ; i < ARRAY_SIZE(cbit); i++) {
+ unsigned short val;
+ snd_ac97_write(ac97, reg, 0x8080 | cbit[i] | (cbit[i] << 8));
+ /* Do the read twice due to buffers on some ac97 codecs.
+ * e.g. The STAC9704 returns exactly what you wrote to the register
+ * if you read it immediately. This causes the detect routine to fail.
+ */
+ val = snd_ac97_read(ac97, reg);
+ val = snd_ac97_read(ac97, reg);
+ if (! *lo_max && (val & 0x7f) == cbit[i])
+ *lo_max = max[i];
+ if (! *hi_max && ((val >> 8) & 0x7f) == cbit[i])
+ *hi_max = max[i];
+ if (*lo_max && *hi_max)
+ break;
+ }
+}
+
+static int snd_ac97_try_bit(struct snd_ac97 * ac97, int reg, int bit)
+{
+ unsigned short mask, val, orig, res;
+
+ mask = 1 << bit;
+ orig = snd_ac97_read(ac97, reg);
+ val = orig ^ mask;
+ snd_ac97_write(ac97, reg, val);
+ res = snd_ac97_read(ac97, reg);
+ snd_ac97_write_cache(ac97, reg, orig);
+ return res == val;
+}
+
+/* check the volume resolution of center/lfe */
+static void snd_ac97_change_volume_params2(struct snd_ac97 * ac97, int reg, int shift, unsigned char *max)
+{
+ unsigned short val, val1;
+
+ *max = 63;
+ val = 0x8080 | (0x20 << shift);
+ snd_ac97_write(ac97, reg, val);
+ val1 = snd_ac97_read(ac97, reg);
+ if (val != val1) {
+ *max = 31;
+ }
+ /* reset volume to zero */
+ snd_ac97_write_cache(ac97, reg, 0x8080);
+}
+
+static inline int printable(unsigned int x)
+{
+ x &= 0xff;
+ if (x < ' ' || x >= 0x71) {
+ if (x <= 0x89)
+ return x - 0x71 + 'A';
+ return '?';
+ }
+ return x;
+}
+
+static struct snd_kcontrol *snd_ac97_cnew(const struct snd_kcontrol_new *_template,
+ struct snd_ac97 * ac97)
+{
+ struct snd_kcontrol_new template;
+ memcpy(&template, _template, sizeof(template));
+ template.index = ac97->num;
+ return snd_ctl_new1(&template, ac97);
+}
+
+/*
+ * create mute switch(es) for normal stereo controls
+ */
+static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg,
+ int check_stereo, int check_amix,
+ struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+ unsigned short val, val1, mute_mask;
+
+ if (! snd_ac97_valid_reg(ac97, reg))
+ return 0;
+
+ mute_mask = 0x8000;
+ val = snd_ac97_read(ac97, reg);
+ if (check_stereo || (ac97->flags & AC97_STEREO_MUTES)) {
+ /* check whether both mute bits work */
+ val1 = val | 0x8080;
+ snd_ac97_write(ac97, reg, val1);
+ if (val1 == snd_ac97_read(ac97, reg))
+ mute_mask = 0x8080;
+ }
+ if (mute_mask == 0x8080) {
+ struct snd_kcontrol_new tmp = AC97_DOUBLE(name, reg, 15, 7, 1, 1);
+ if (check_amix)
+ tmp.private_value |= (1 << 30);
+ tmp.index = ac97->num;
+ kctl = snd_ctl_new1(&tmp, ac97);
+ } else {
+ struct snd_kcontrol_new tmp = AC97_SINGLE(name, reg, 15, 1, 1);
+ if (check_amix)
+ tmp.private_value |= (1 << 30);
+ tmp.index = ac97->num;
+ kctl = snd_ctl_new1(&tmp, ac97);
+ }
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
+ return err;
+ /* mute as default */
+ snd_ac97_write_cache(ac97, reg, val | mute_mask);
+ return 0;
+}
+
+/*
+ * set dB information
+ */
+static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0);
+
+static const unsigned int *find_db_scale(unsigned int maxval)
+{
+ switch (maxval) {
+ case 0x0f: return db_scale_4bit;
+ case 0x1f: return db_scale_5bit;
+ case 0x3f: return db_scale_6bit;
+ }
+ return NULL;
+}
+
+static void set_tlv_db_scale(struct snd_kcontrol *kctl, const unsigned int *tlv)
+{
+ kctl->tlv.p = tlv;
+ if (tlv)
+ kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+}
+
+/*
+ * create a volume for normal stereo/mono controls
+ */
+static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigned int lo_max,
+ unsigned int hi_max, struct snd_ac97 *ac97)
+{
+ int err;
+ struct snd_kcontrol *kctl;
+
+ if (! snd_ac97_valid_reg(ac97, reg))
+ return 0;
+ if (hi_max) {
+ /* invert */
+ struct snd_kcontrol_new tmp = AC97_DOUBLE(name, reg, 8, 0, lo_max, 1);
+ tmp.index = ac97->num;
+ kctl = snd_ctl_new1(&tmp, ac97);
+ } else {
+ /* invert */
+ struct snd_kcontrol_new tmp = AC97_SINGLE(name, reg, 0, lo_max, 1);
+ tmp.index = ac97->num;
+ kctl = snd_ctl_new1(&tmp, ac97);
+ }
+ if (reg >= AC97_PHONE && reg <= AC97_PCM)
+ set_tlv_db_scale(kctl, db_scale_5bit_12db_max);
+ else
+ set_tlv_db_scale(kctl, find_db_scale(lo_max));
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
+ return err;
+ snd_ac97_write_cache(ac97, reg,
+ (snd_ac97_read(ac97, reg) & 0x8080) |
+ lo_max | (hi_max << 8));
+ return 0;
+}
+
+/*
+ * create a mute-switch and a volume for normal stereo/mono controls
+ */
+static int snd_ac97_cmix_new_stereo(struct snd_card *card, const char *pfx,
+ int reg, int check_stereo, int check_amix,
+ struct snd_ac97 *ac97)
+{
+ int err;
+ char name[44];
+ unsigned char lo_max, hi_max;
+
+ if (! snd_ac97_valid_reg(ac97, reg))
+ return 0;
+
+ if (snd_ac97_try_bit(ac97, reg, 15)) {
+ sprintf(name, "%s Switch", pfx);
+ if ((err = snd_ac97_cmute_new_stereo(card, name, reg,
+ check_stereo, check_amix,
+ ac97)) < 0)
+ return err;
+ }
+ check_volume_resolution(ac97, reg, &lo_max, &hi_max);
+ if (lo_max) {
+ sprintf(name, "%s Volume", pfx);
+ if ((err = snd_ac97_cvol_new(card, name, reg, lo_max, hi_max, ac97)) < 0)
+ return err;
+ }
+ return 0;
+}
+
+#define snd_ac97_cmix_new(card, pfx, reg, acheck, ac97) \
+ snd_ac97_cmix_new_stereo(card, pfx, reg, 0, acheck, ac97)
+#define snd_ac97_cmute_new(card, name, reg, acheck, ac97) \
+ snd_ac97_cmute_new_stereo(card, name, reg, 0, acheck, ac97)
+
+static unsigned int snd_ac97_determine_spdif_rates(struct snd_ac97 *ac97);
+
+static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
+{
+ struct snd_card *card = ac97->bus->card;
+ struct snd_kcontrol *kctl;
+ int err;
+ unsigned int idx;
+ unsigned char max;
+
+ /* build master controls */
+ /* AD claims to remove this control from AD1887, although spec v2.2 does not allow this */
+ if (snd_ac97_try_volume_mix(ac97, AC97_MASTER)) {
+ if (ac97->flags & AC97_HAS_NO_MASTER_VOL)
+ err = snd_ac97_cmute_new(card, "Master Playback Switch",
+ AC97_MASTER, 0, ac97);
+ else
+ err = snd_ac97_cmix_new(card, "Master Playback",
+ AC97_MASTER, 0, ac97);
+ if (err < 0)
+ return err;
+ }
+
+ ac97->regs[AC97_CENTER_LFE_MASTER] = 0x8080;
+
+ /* build center controls */
+ if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER))
+ && !(ac97->flags & AC97_AD_MULTI)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_center[0], ac97))) < 0)
+ return err;
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_center[1], ac97))) < 0)
+ return err;
+ snd_ac97_change_volume_params2(ac97, AC97_CENTER_LFE_MASTER, 0, &max);
+ kctl->private_value &= ~(0xff << 16);
+ kctl->private_value |= (int)max << 16;
+ set_tlv_db_scale(kctl, find_db_scale(max));
+ snd_ac97_write_cache(ac97, AC97_CENTER_LFE_MASTER, ac97->regs[AC97_CENTER_LFE_MASTER] | max);
+ }
+
+ /* build LFE controls */
+ if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER+1))
+ && !(ac97->flags & AC97_AD_MULTI)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_lfe[0], ac97))) < 0)
+ return err;
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_lfe[1], ac97))) < 0)
+ return err;
+ snd_ac97_change_volume_params2(ac97, AC97_CENTER_LFE_MASTER, 8, &max);
+ kctl->private_value &= ~(0xff << 16);
+ kctl->private_value |= (int)max << 16;
+ set_tlv_db_scale(kctl, find_db_scale(max));
+ snd_ac97_write_cache(ac97, AC97_CENTER_LFE_MASTER, ac97->regs[AC97_CENTER_LFE_MASTER] | max << 8);
+ }
+
+ /* build surround controls */
+ if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER))
+ && !(ac97->flags & AC97_AD_MULTI)) {
+ /* Surround Master (0x38) is with stereo mutes */
+ if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback",
+ AC97_SURROUND_MASTER, 1, 0,
+ ac97)) < 0)
+ return err;
+ }
+
+ /* build headphone controls */
+ if (snd_ac97_try_volume_mix(ac97, AC97_HEADPHONE)) {
+ if ((err = snd_ac97_cmix_new(card, "Headphone Playback",
+ AC97_HEADPHONE, 0, ac97)) < 0)
+ return err;
+ }
+
+ /* build master mono controls */
+ if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_MONO)) {
+ if ((err = snd_ac97_cmix_new(card, "Master Mono Playback",
+ AC97_MASTER_MONO, 0, ac97)) < 0)
+ return err;
+ }
+
+ /* build master tone controls */
+ if (!(ac97->flags & AC97_HAS_NO_TONE)) {
+ if (snd_ac97_try_volume_mix(ac97, AC97_MASTER_TONE)) {
+ for (idx = 0; idx < 2; idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0)
+ return err;
+ if (ac97->id == AC97_ID_YMF743 ||
+ ac97->id == AC97_ID_YMF753) {
+ kctl->private_value &= ~(0xff << 16);
+ kctl->private_value |= 7 << 16;
+ }
+ }
+ snd_ac97_write_cache(ac97, AC97_MASTER_TONE, 0x0f0f);
+ }
+ }
+
+ /* build PC Speaker controls */
+ if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) &&
+ ((ac97->flags & AC97_HAS_PC_BEEP) ||
+ snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) {
+ for (idx = 0; idx < 2; idx++)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_pc_beep[idx], ac97))) < 0)
+ return err;
+ set_tlv_db_scale(kctl, db_scale_4bit);
+ snd_ac97_write_cache(ac97, AC97_PC_BEEP,
+ snd_ac97_read(ac97, AC97_PC_BEEP) | 0x801e);
+ }
+
+ /* build Phone controls */
+ if (!(ac97->flags & AC97_HAS_NO_PHONE)) {
+ if (snd_ac97_try_volume_mix(ac97, AC97_PHONE)) {
+ if ((err = snd_ac97_cmix_new(card, "Phone Playback",
+ AC97_PHONE, 1, ac97)) < 0)
+ return err;
+ }
+ }
+
+ /* build MIC controls */
+ if (!(ac97->flags & AC97_HAS_NO_MIC)) {
+ if (snd_ac97_try_volume_mix(ac97, AC97_MIC)) {
+ if ((err = snd_ac97_cmix_new(card, "Mic Playback",
+ AC97_MIC, 1, ac97)) < 0)
+ return err;
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_mic_boost, ac97))) < 0)
+ return err;
+ }
+ }
+
+ /* build Line controls */
+ if (snd_ac97_try_volume_mix(ac97, AC97_LINE)) {
+ if ((err = snd_ac97_cmix_new(card, "Line Playback",
+ AC97_LINE, 1, ac97)) < 0)
+ return err;
+ }
+
+ /* build CD controls */
+ if (!(ac97->flags & AC97_HAS_NO_CD)) {
+ if (snd_ac97_try_volume_mix(ac97, AC97_CD)) {
+ if ((err = snd_ac97_cmix_new(card, "CD Playback",
+ AC97_CD, 1, ac97)) < 0)
+ return err;
+ }
+ }
+
+ /* build Video controls */
+ if (!(ac97->flags & AC97_HAS_NO_VIDEO)) {
+ if (snd_ac97_try_volume_mix(ac97, AC97_VIDEO)) {
+ if ((err = snd_ac97_cmix_new(card, "Video Playback",
+ AC97_VIDEO, 1, ac97)) < 0)
+ return err;
+ }
+ }
+
+ /* build Aux controls */
+ if (!(ac97->flags & AC97_HAS_NO_AUX)) {
+ if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) {
+ if ((err = snd_ac97_cmix_new(card, "Aux Playback",
+ AC97_AUX, 1, ac97)) < 0)
+ return err;
+ }
+ }
+
+ /* build PCM controls */
+ if (ac97->flags & AC97_AD_MULTI) {
+ unsigned short init_val;
+ if (ac97->flags & AC97_STEREO_MUTES)
+ init_val = 0x9f9f;
+ else
+ init_val = 0x9f1f;
+ for (idx = 0; idx < 2; idx++)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_pcm[idx], ac97))) < 0)
+ return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
+ ac97->spec.ad18xx.pcmreg[0] = init_val;
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC) {
+ for (idx = 0; idx < 2; idx++)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_surround[idx], ac97))) < 0)
+ return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
+ ac97->spec.ad18xx.pcmreg[1] = init_val;
+ }
+ if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC) {
+ for (idx = 0; idx < 2; idx++)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_center[idx], ac97))) < 0)
+ return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
+ for (idx = 0; idx < 2; idx++)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_ad18xx_lfe[idx], ac97))) < 0)
+ return err;
+ set_tlv_db_scale(kctl, db_scale_5bit);
+ ac97->spec.ad18xx.pcmreg[2] = init_val;
+ }
+ snd_ac97_write_cache(ac97, AC97_PCM, init_val);
+ } else {
+ if (!(ac97->flags & AC97_HAS_NO_STD_PCM)) {
+ if (ac97->flags & AC97_HAS_NO_PCM_VOL)
+ err = snd_ac97_cmute_new(card,
+ "PCM Playback Switch",
+ AC97_PCM, 0, ac97);
+ else
+ err = snd_ac97_cmix_new(card, "PCM Playback",
+ AC97_PCM, 0, ac97);
+ if (err < 0)
+ return err;
+ }
+ }
+
+ /* build Capture controls */
+ if (!(ac97->flags & AC97_HAS_NO_REC_GAIN)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_control_capture_src, ac97))) < 0)
+ return err;
+ if (snd_ac97_try_bit(ac97, AC97_REC_GAIN, 15)) {
+ err = snd_ac97_cmute_new(card, "Capture Switch",
+ AC97_REC_GAIN, 0, ac97);
+ if (err < 0)
+ return err;
+ }
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_control_capture_vol, ac97))) < 0)
+ return err;
+ set_tlv_db_scale(kctl, db_scale_rec_gain);
+ snd_ac97_write_cache(ac97, AC97_REC_SEL, 0x0000);
+ snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x0000);
+ }
+ /* build MIC Capture controls */
+ if (snd_ac97_try_volume_mix(ac97, AC97_REC_GAIN_MIC)) {
+ for (idx = 0; idx < 2; idx++)
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_mic_capture[idx], ac97))) < 0)
+ return err;
+ set_tlv_db_scale(kctl, db_scale_rec_gain);
+ snd_ac97_write_cache(ac97, AC97_REC_GAIN_MIC, 0x0000);
+ }
+
+ /* build PCM out path & mute control */
+ if (snd_ac97_try_bit(ac97, AC97_GENERAL_PURPOSE, 15)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_PCM_OUT], ac97))) < 0)
+ return err;
+ }
+
+ /* build Simulated Stereo Enhancement control */
+ if (ac97->caps & 0x0008) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_STEREO_ENHANCEMENT], ac97))) < 0)
+ return err;
+ }
+
+ /* build 3D Stereo Enhancement control */
+ if (snd_ac97_try_bit(ac97, AC97_GENERAL_PURPOSE, 13)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_3D], ac97))) < 0)
+ return err;
+ }
+
+ /* build Loudness control */
+ if (ac97->caps & 0x0020) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_LOUDNESS], ac97))) < 0)
+ return err;
+ }
+
+ /* build Mono output select control */
+ if (snd_ac97_try_bit(ac97, AC97_GENERAL_PURPOSE, 9)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_MONO], ac97))) < 0)
+ return err;
+ }
+
+ /* build Mic select control */
+ if (snd_ac97_try_bit(ac97, AC97_GENERAL_PURPOSE, 8)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_MIC], ac97))) < 0)
+ return err;
+ }
+
+ /* build ADC/DAC loopback control */
+ if (enable_loopback && snd_ac97_try_bit(ac97, AC97_GENERAL_PURPOSE, 7)) {
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_LOOPBACK], ac97))) < 0)
+ return err;
+ }
+
+ snd_ac97_update_bits(ac97, AC97_GENERAL_PURPOSE, ~AC97_GP_DRSS_MASK, 0x0000);
+
+ /* build 3D controls */
+ if (ac97->build_ops->build_3d) {
+ ac97->build_ops->build_3d(ac97);
+ } else {
+ if (snd_ac97_try_volume_mix(ac97, AC97_3D_CONTROL)) {
+ unsigned short val;
+ val = 0x0707;
+ snd_ac97_write(ac97, AC97_3D_CONTROL, val);
+ val = snd_ac97_read(ac97, AC97_3D_CONTROL);
+ val = val == 0x0606;
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0)
+ return err;
+ if (val)
+ kctl->private_value = AC97_3D_CONTROL | (9 << 8) | (7 << 16);
+ if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[1], ac97))) < 0)
+ return err;
+ if (val)
+ kctl->private_value = AC97_3D_CONTROL | (1 << 8) | (7 << 16);
+ snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
+ }
+ }
+
+ /* build S/PDIF controls */
+
+ /* Hack for ASUS P5P800-VM, which does not indicate S/PDIF capability */
+ if (ac97->subsystem_vendor == 0x1043 &&
+ ac97->subsystem_device == 0x810f)
+ ac97->ext_id |= AC97_EI_SPDIF;
+
+ if ((ac97->ext_id & AC97_EI_SPDIF) && !(ac97->scaps & AC97_SCAP_NO_SPDIF)) {
+ if (ac97->build_ops->build_spdif) {
+ if ((err = ac97->build_ops->build_spdif(ac97)) < 0)
+ return err;
+ } else {
+ for (idx = 0; idx < 5; idx++)
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_spdif[idx], ac97))) < 0)
+ return err;
+ if (ac97->build_ops->build_post_spdif) {
+ if ((err = ac97->build_ops->build_post_spdif(ac97)) < 0)
+ return err;
+ }
+ /* set default PCM S/PDIF params */
+ /* consumer,PCM audio,no copyright,no preemphasis,PCM coder,original,48000Hz */
+ snd_ac97_write_cache(ac97, AC97_SPDIF, 0x2a20);
+ ac97->rates[AC97_RATES_SPDIF] = snd_ac97_determine_spdif_rates(ac97);
+ }
+ ac97->spdif_status = SNDRV_PCM_DEFAULT_CON_SPDIF;
+ }
+
+ /* build chip specific controls */
+ if (ac97->build_ops->build_specific)
+ if ((err = ac97->build_ops->build_specific(ac97)) < 0)
+ return err;
+
+ if (snd_ac97_try_bit(ac97, AC97_POWERDOWN, 15)) {
+ kctl = snd_ac97_cnew(&snd_ac97_control_eapd, ac97);
+ if (! kctl)
+ return -ENOMEM;
+ if (ac97->scaps & AC97_SCAP_INV_EAPD)
+ set_inv_eapd(ac97, kctl);
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 * ac97)
+{
+ int err, idx;
+
+ //printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG));
+ snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH));
+ snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH));
+ snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff);
+ snd_ac97_write(ac97, AC97_GPIO_WAKEUP, 0x0);
+ snd_ac97_write(ac97, AC97_MISC_AFE, 0x0);
+
+ /* build modem switches */
+ for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_modem_switches); idx++)
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_ac97_controls_modem_switches[idx], ac97))) < 0)
+ return err;
+
+ /* build chip specific controls */
+ if (ac97->build_ops->build_specific)
+ if ((err = ac97->build_ops->build_specific(ac97)) < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_ac97_test_rate(struct snd_ac97 *ac97, int reg, int shadow_reg, int rate)
+{
+ unsigned short val;
+ unsigned int tmp;
+
+ tmp = ((unsigned int)rate * ac97->bus->clock) / 48000;
+ snd_ac97_write_cache(ac97, reg, tmp & 0xffff);
+ if (shadow_reg)
+ snd_ac97_write_cache(ac97, shadow_reg, tmp & 0xffff);
+ val = snd_ac97_read(ac97, reg);
+ return val == (tmp & 0xffff);
+}
+
+static void snd_ac97_determine_rates(struct snd_ac97 *ac97, int reg, int shadow_reg, unsigned int *r_result)
+{
+ unsigned int result = 0;
+ unsigned short saved;
+
+ if (ac97->bus->no_vra) {
+ *r_result = SNDRV_PCM_RATE_48000;
+ if ((ac97->flags & AC97_DOUBLE_RATE) &&
+ reg == AC97_PCM_FRONT_DAC_RATE)
+ *r_result |= SNDRV_PCM_RATE_96000;
+ return;
+ }
+
+ saved = snd_ac97_read(ac97, reg);
+ if ((ac97->ext_id & AC97_EI_DRA) && reg == AC97_PCM_FRONT_DAC_RATE)
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
+ AC97_EA_DRA, 0);
+ /* test a non-standard rate */
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 11000))
+ result |= SNDRV_PCM_RATE_CONTINUOUS;
+ /* let's try to obtain standard rates */
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 8000))
+ result |= SNDRV_PCM_RATE_8000;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 11025))
+ result |= SNDRV_PCM_RATE_11025;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 16000))
+ result |= SNDRV_PCM_RATE_16000;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 22050))
+ result |= SNDRV_PCM_RATE_22050;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 32000))
+ result |= SNDRV_PCM_RATE_32000;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 44100))
+ result |= SNDRV_PCM_RATE_44100;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 48000))
+ result |= SNDRV_PCM_RATE_48000;
+ if ((ac97->flags & AC97_DOUBLE_RATE) &&
+ reg == AC97_PCM_FRONT_DAC_RATE) {
+ /* test standard double rates */
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
+ AC97_EA_DRA, AC97_EA_DRA);
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 64000 / 2))
+ result |= SNDRV_PCM_RATE_64000;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 88200 / 2))
+ result |= SNDRV_PCM_RATE_88200;
+ if (snd_ac97_test_rate(ac97, reg, shadow_reg, 96000 / 2))
+ result |= SNDRV_PCM_RATE_96000;
+ /* some codecs don't support variable double rates */
+ if (!snd_ac97_test_rate(ac97, reg, shadow_reg, 76100 / 2))
+ result &= ~SNDRV_PCM_RATE_CONTINUOUS;
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
+ AC97_EA_DRA, 0);
+ }
+ /* restore the default value */
+ snd_ac97_write_cache(ac97, reg, saved);
+ if (shadow_reg)
+ snd_ac97_write_cache(ac97, shadow_reg, saved);
+ *r_result = result;
+}
+
+/* check AC97_SPDIF register to accept which sample rates */
+static unsigned int snd_ac97_determine_spdif_rates(struct snd_ac97 *ac97)
+{
+ unsigned int result = 0;
+ int i;
+ static unsigned short ctl_bits[] = {
+ AC97_SC_SPSR_44K, AC97_SC_SPSR_32K, AC97_SC_SPSR_48K
+ };
+ static unsigned int rate_bits[] = {
+ SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_32000, SNDRV_PCM_RATE_48000
+ };
+
+ for (i = 0; i < (int)ARRAY_SIZE(ctl_bits); i++) {
+ snd_ac97_update_bits(ac97, AC97_SPDIF, AC97_SC_SPSR_MASK, ctl_bits[i]);
+ if ((snd_ac97_read(ac97, AC97_SPDIF) & AC97_SC_SPSR_MASK) == ctl_bits[i])
+ result |= rate_bits[i];
+ }
+ return result;
+}
+
+/* look for the codec id table matching with the given id */
+static const struct ac97_codec_id *look_for_codec_id(const struct ac97_codec_id *table,
+ unsigned int id)
+{
+ const struct ac97_codec_id *pid;
+
+ for (pid = table; pid->id; pid++)
+ if (pid->id == (id & pid->mask))
+ return pid;
+ return NULL;
+}
+
+void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int modem)
+{
+ const struct ac97_codec_id *pid;
+
+ sprintf(name, "0x%x %c%c%c", id,
+ printable(id >> 24),
+ printable(id >> 16),
+ printable(id >> 8));
+ pid = look_for_codec_id(snd_ac97_codec_id_vendors, id);
+ if (! pid)
+ return;
+
+ strcpy(name, pid->name);
+ if (ac97 && pid->patch) {
+ if ((modem && (pid->flags & AC97_MODEM_PATCH)) ||
+ (! modem && ! (pid->flags & AC97_MODEM_PATCH)))
+ pid->patch(ac97);
+ }
+
+ pid = look_for_codec_id(snd_ac97_codec_ids, id);
+ if (pid) {
+ strcat(name, " ");
+ strcat(name, pid->name);
+ if (pid->mask != 0xffffffff)
+ sprintf(name + strlen(name), " rev %d", id & ~pid->mask);
+ if (ac97 && pid->patch) {
+ if ((modem && (pid->flags & AC97_MODEM_PATCH)) ||
+ (! modem && ! (pid->flags & AC97_MODEM_PATCH)))
+ pid->patch(ac97);
+ }
+ } else
+ sprintf(name + strlen(name), " id %x", id & 0xff);
+}
+
+/**
+ * snd_ac97_get_short_name - retrieve codec name
+ * @ac97: the codec instance
+ *
+ * Returns the short identifying name of the codec.
+ */
+const char *snd_ac97_get_short_name(struct snd_ac97 *ac97)
+{
+ const struct ac97_codec_id *pid;
+
+ for (pid = snd_ac97_codec_ids; pid->id; pid++)
+ if (pid->id == (ac97->id & pid->mask))
+ return pid->name;
+ return "unknown codec";
+}
+
+EXPORT_SYMBOL(snd_ac97_get_short_name);
+
+/* wait for a while until registers are accessible after RESET
+ * return 0 if ok, negative not ready
+ */
+static int ac97_reset_wait(struct snd_ac97 *ac97, int timeout, int with_modem)
+{
+ unsigned long end_time;
+ unsigned short val;
+
+ end_time = jiffies + timeout;
+ do {
+
+ /* use preliminary reads to settle the communication */
+ snd_ac97_read(ac97, AC97_RESET);
+ snd_ac97_read(ac97, AC97_VENDOR_ID1);
+ snd_ac97_read(ac97, AC97_VENDOR_ID2);
+ /* modem? */
+ if (with_modem) {
+ val = snd_ac97_read(ac97, AC97_EXTENDED_MID);
+ if (val != 0xffff && (val & 1) != 0)
+ return 0;
+ }
+ if (ac97->scaps & AC97_SCAP_DETECT_BY_VENDOR) {
+ /* probably only Xbox issue - all registers are read as zero */
+ val = snd_ac97_read(ac97, AC97_VENDOR_ID1);
+ if (val != 0 && val != 0xffff)
+ return 0;
+ } else {
+ /* because the PCM or MASTER volume registers can be modified,
+ * the REC_GAIN register is used for tests
+ */
+ /* test if we can write to the record gain volume register */
+ snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x8a05);
+ if ((snd_ac97_read(ac97, AC97_REC_GAIN) & 0x7fff) == 0x0a05)
+ return 0;
+ }
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ return -ENODEV;
+}
+
+/**
+ * snd_ac97_bus - create an AC97 bus component
+ * @card: the card instance
+ * @num: the bus number
+ * @ops: the bus callbacks table
+ * @private_data: private data pointer for the new instance
+ * @rbus: the pointer to store the new AC97 bus instance.
+ *
+ * Creates an AC97 bus component. An struct snd_ac97_bus instance is newly
+ * allocated and initialized.
+ *
+ * The ops table must include valid callbacks (at least read and
+ * write). The other callbacks, wait and reset, are not mandatory.
+ *
+ * The clock is set to 48000. If another clock is needed, set
+ * (*rbus)->clock manually.
+ *
+ * The AC97 bus instance is registered as a low-level device, so you don't
+ * have to release it manually.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops,
+ void *private_data, struct snd_ac97_bus **rbus)
+{
+ int err;
+ struct snd_ac97_bus *bus;
+ static struct snd_device_ops dev_ops = {
+ .dev_free = snd_ac97_bus_dev_free,
+ };
+
+ if (snd_BUG_ON(!card))
+ return -EINVAL;
+ bus = kzalloc(sizeof(*bus), GFP_KERNEL);
+ if (bus == NULL)
+ return -ENOMEM;
+ bus->card = card;
+ bus->num = num;
+ bus->ops = ops;
+ bus->private_data = private_data;
+ bus->clock = 48000;
+ spin_lock_init(&bus->bus_lock);
+ snd_ac97_bus_proc_init(bus);
+ if ((err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops)) < 0) {
+ snd_ac97_bus_free(bus);
+ return err;
+ }
+ if (rbus)
+ *rbus = bus;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_bus);
+
+/* stop no dev release warning */
+static void ac97_device_release(struct device * dev)
+{
+}
+
+/* register ac97 codec to bus */
+static int snd_ac97_dev_register(struct snd_device *device)
+{
+ struct snd_ac97 *ac97 = device->device_data;
+ int err;
+
+ ac97->dev.bus = &ac97_bus_type;
+ ac97->dev.parent = ac97->bus->card->dev;
+ ac97->dev.release = ac97_device_release;
+ dev_set_name(&ac97->dev, "%d-%d:%s",
+ ac97->bus->card->number, ac97->num,
+ snd_ac97_get_short_name(ac97));
+ if ((err = device_register(&ac97->dev)) < 0) {
+ snd_printk(KERN_ERR "Can't register ac97 bus\n");
+ ac97->dev.bus = NULL;
+ return err;
+ }
+ return 0;
+}
+
+/* disconnect ac97 codec */
+static int snd_ac97_dev_disconnect(struct snd_device *device)
+{
+ struct snd_ac97 *ac97 = device->device_data;
+ if (ac97->dev.bus)
+ device_unregister(&ac97->dev);
+ return 0;
+}
+
+/* build_ops to do nothing */
+static struct snd_ac97_build_ops null_build_ops;
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+static void do_update_power(struct work_struct *work)
+{
+ update_power_regs(
+ container_of(work, struct snd_ac97, power_work.work));
+}
+#endif
+
+/**
+ * snd_ac97_mixer - create an Codec97 component
+ * @bus: the AC97 bus which codec is attached to
+ * @template: the template of ac97, including index, callbacks and
+ * the private data.
+ * @rac97: the pointer to store the new ac97 instance.
+ *
+ * Creates an Codec97 component. An struct snd_ac97 instance is newly
+ * allocated and initialized from the template. The codec
+ * is then initialized by the standard procedure.
+ *
+ * The template must include the codec number (num) and address (addr),
+ * and the private data (private_data).
+ *
+ * The ac97 instance is registered as a low-level device, so you don't
+ * have to release it manually.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, struct snd_ac97 **rac97)
+{
+ int err;
+ struct snd_ac97 *ac97;
+ struct snd_card *card;
+ char name[64];
+ unsigned long end_time;
+ unsigned int reg;
+ const struct ac97_codec_id *pid;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ac97_dev_free,
+ .dev_register = snd_ac97_dev_register,
+ .dev_disconnect = snd_ac97_dev_disconnect,
+ };
+
+ if (rac97)
+ *rac97 = NULL;
+ if (snd_BUG_ON(!bus || !template))
+ return -EINVAL;
+ if (snd_BUG_ON(template->num >= 4))
+ return -EINVAL;
+ if (bus->codec[template->num])
+ return -EBUSY;
+
+ card = bus->card;
+ ac97 = kzalloc(sizeof(*ac97), GFP_KERNEL);
+ if (ac97 == NULL)
+ return -ENOMEM;
+ ac97->private_data = template->private_data;
+ ac97->private_free = template->private_free;
+ ac97->bus = bus;
+ ac97->pci = template->pci;
+ ac97->num = template->num;
+ ac97->addr = template->addr;
+ ac97->scaps = template->scaps;
+ ac97->res_table = template->res_table;
+ bus->codec[ac97->num] = ac97;
+ mutex_init(&ac97->reg_mutex);
+ mutex_init(&ac97->page_mutex);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ INIT_DELAYED_WORK(&ac97->power_work, do_update_power);
+#endif
+
+#ifdef CONFIG_PCI
+ if (ac97->pci) {
+ pci_read_config_word(ac97->pci, PCI_SUBSYSTEM_VENDOR_ID, &ac97->subsystem_vendor);
+ pci_read_config_word(ac97->pci, PCI_SUBSYSTEM_ID, &ac97->subsystem_device);
+ }
+#endif
+ if (bus->ops->reset) {
+ bus->ops->reset(ac97);
+ goto __access_ok;
+ }
+
+ ac97->id = snd_ac97_read(ac97, AC97_VENDOR_ID1) << 16;
+ ac97->id |= snd_ac97_read(ac97, AC97_VENDOR_ID2);
+ if (ac97->id && ac97->id != (unsigned int)-1) {
+ pid = look_for_codec_id(snd_ac97_codec_ids, ac97->id);
+ if (pid && (pid->flags & AC97_DEFAULT_POWER_OFF))
+ goto __access_ok;
+ }
+
+ /* reset to defaults */
+ if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO))
+ snd_ac97_write(ac97, AC97_RESET, 0);
+ if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM))
+ snd_ac97_write(ac97, AC97_EXTENDED_MID, 0);
+ if (bus->ops->wait)
+ bus->ops->wait(ac97);
+ else {
+ udelay(50);
+ if (ac97->scaps & AC97_SCAP_SKIP_AUDIO)
+ err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 1);
+ else {
+ err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 0);
+ if (err < 0)
+ err = ac97_reset_wait(ac97,
+ msecs_to_jiffies(500), 1);
+ }
+ if (err < 0) {
+ snd_printk(KERN_WARNING "AC'97 %d does not respond - RESET\n", ac97->num);
+ /* proceed anyway - it's often non-critical */
+ }
+ }
+ __access_ok:
+ ac97->id = snd_ac97_read(ac97, AC97_VENDOR_ID1) << 16;
+ ac97->id |= snd_ac97_read(ac97, AC97_VENDOR_ID2);
+ if (! (ac97->scaps & AC97_SCAP_DETECT_BY_VENDOR) &&
+ (ac97->id == 0x00000000 || ac97->id == 0xffffffff)) {
+ snd_printk(KERN_ERR "AC'97 %d access is not valid [0x%x], removing mixer.\n", ac97->num, ac97->id);
+ snd_ac97_free(ac97);
+ return -EIO;
+ }
+ pid = look_for_codec_id(snd_ac97_codec_ids, ac97->id);
+ if (pid)
+ ac97->flags |= pid->flags;
+
+ /* test for AC'97 */
+ if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO) && !(ac97->scaps & AC97_SCAP_AUDIO)) {
+ /* test if we can write to the record gain volume register */
+ snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x8a06);
+ if (((err = snd_ac97_read(ac97, AC97_REC_GAIN)) & 0x7fff) == 0x0a06)
+ ac97->scaps |= AC97_SCAP_AUDIO;
+ }
+ if (ac97->scaps & AC97_SCAP_AUDIO) {
+ ac97->caps = snd_ac97_read(ac97, AC97_RESET);
+ ac97->ext_id = snd_ac97_read(ac97, AC97_EXTENDED_ID);
+ if (ac97->ext_id == 0xffff) /* invalid combination */
+ ac97->ext_id = 0;
+ }
+
+ /* test for MC'97 */
+ if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM) && !(ac97->scaps & AC97_SCAP_MODEM)) {
+ ac97->ext_mid = snd_ac97_read(ac97, AC97_EXTENDED_MID);
+ if (ac97->ext_mid == 0xffff) /* invalid combination */
+ ac97->ext_mid = 0;
+ if (ac97->ext_mid & 1)
+ ac97->scaps |= AC97_SCAP_MODEM;
+ }
+
+ if (!ac97_is_audio(ac97) && !ac97_is_modem(ac97)) {
+ if (!(ac97->scaps & (AC97_SCAP_SKIP_AUDIO|AC97_SCAP_SKIP_MODEM)))
+ snd_printk(KERN_ERR "AC'97 %d access error (not audio or modem codec)\n", ac97->num);
+ snd_ac97_free(ac97);
+ return -EACCES;
+ }
+
+ if (bus->ops->reset) // FIXME: always skipping?
+ goto __ready_ok;
+
+ /* FIXME: add powerdown control */
+ if (ac97_is_audio(ac97)) {
+ /* nothing should be in powerdown mode */
+ snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0);
+ if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) {
+ snd_ac97_write_cache(ac97, AC97_RESET, 0); /* reset to defaults */
+ udelay(100);
+ snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0);
+ }
+ /* nothing should be in powerdown mode */
+ snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0);
+ end_time = jiffies + msecs_to_jiffies(100);
+ do {
+ if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f)
+ goto __ready_ok;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_WARNING "AC'97 %d analog subsections not ready\n", ac97->num);
+ }
+
+ /* FIXME: add powerdown control */
+ if (ac97_is_modem(ac97)) {
+ unsigned char tmp;
+
+ /* nothing should be in powerdown mode */
+ /* note: it's important to set the rate at first */
+ tmp = AC97_MEA_GPIO;
+ if (ac97->ext_mid & AC97_MEI_LINE1) {
+ snd_ac97_write_cache(ac97, AC97_LINE1_RATE, 8000);
+ tmp |= AC97_MEA_ADC1 | AC97_MEA_DAC1;
+ }
+ if (ac97->ext_mid & AC97_MEI_LINE2) {
+ snd_ac97_write_cache(ac97, AC97_LINE2_RATE, 8000);
+ tmp |= AC97_MEA_ADC2 | AC97_MEA_DAC2;
+ }
+ if (ac97->ext_mid & AC97_MEI_HANDSET) {
+ snd_ac97_write_cache(ac97, AC97_HANDSET_RATE, 8000);
+ tmp |= AC97_MEA_HADC | AC97_MEA_HDAC;
+ }
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0);
+ udelay(100);
+ /* nothing should be in powerdown mode */
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0);
+ end_time = jiffies + msecs_to_jiffies(100);
+ do {
+ if ((snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS) & tmp) == tmp)
+ goto __ready_ok;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_WARNING "MC'97 %d converters and GPIO not ready (0x%x)\n", ac97->num, snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS));
+ }
+
+ __ready_ok:
+ if (ac97_is_audio(ac97))
+ ac97->addr = (ac97->ext_id & AC97_EI_ADDR_MASK) >> AC97_EI_ADDR_SHIFT;
+ else
+ ac97->addr = (ac97->ext_mid & AC97_MEI_ADDR_MASK) >> AC97_MEI_ADDR_SHIFT;
+ if (ac97->ext_id & 0x01c9) { /* L/R, MIC, SDAC, LDAC VRA support */
+ reg = snd_ac97_read(ac97, AC97_EXTENDED_STATUS);
+ reg |= ac97->ext_id & 0x01c0; /* LDAC/SDAC/CDAC */
+ if (! bus->no_vra)
+ reg |= ac97->ext_id & 0x0009; /* VRA/VRM */
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_STATUS, reg);
+ }
+ if ((ac97->ext_id & AC97_EI_DRA) && bus->dra) {
+ /* Intel controllers require double rate data to be put in
+ * slots 7+8, so let's hope the codec supports it. */
+ snd_ac97_update_bits(ac97, AC97_GENERAL_PURPOSE, AC97_GP_DRSS_MASK, AC97_GP_DRSS_78);
+ if ((snd_ac97_read(ac97, AC97_GENERAL_PURPOSE) & AC97_GP_DRSS_MASK) == AC97_GP_DRSS_78)
+ ac97->flags |= AC97_DOUBLE_RATE;
+ /* restore to slots 10/11 to avoid the confliction with surrounds */
+ snd_ac97_update_bits(ac97, AC97_GENERAL_PURPOSE, AC97_GP_DRSS_MASK, 0);
+ }
+ if (ac97->ext_id & AC97_EI_VRA) { /* VRA support */
+ snd_ac97_determine_rates(ac97, AC97_PCM_FRONT_DAC_RATE, 0, &ac97->rates[AC97_RATES_FRONT_DAC]);
+ snd_ac97_determine_rates(ac97, AC97_PCM_LR_ADC_RATE, 0, &ac97->rates[AC97_RATES_ADC]);
+ } else {
+ ac97->rates[AC97_RATES_FRONT_DAC] = SNDRV_PCM_RATE_48000;
+ if (ac97->flags & AC97_DOUBLE_RATE)
+ ac97->rates[AC97_RATES_FRONT_DAC] |= SNDRV_PCM_RATE_96000;
+ ac97->rates[AC97_RATES_ADC] = SNDRV_PCM_RATE_48000;
+ }
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ /* codec specific code (patch) should override these values */
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_32000;
+ }
+ if (ac97->ext_id & AC97_EI_VRM) { /* MIC VRA support */
+ snd_ac97_determine_rates(ac97, AC97_PCM_MIC_ADC_RATE, 0, &ac97->rates[AC97_RATES_MIC_ADC]);
+ } else {
+ ac97->rates[AC97_RATES_MIC_ADC] = SNDRV_PCM_RATE_48000;
+ }
+ if (ac97->ext_id & AC97_EI_SDAC) { /* SDAC support */
+ snd_ac97_determine_rates(ac97, AC97_PCM_SURR_DAC_RATE, AC97_PCM_FRONT_DAC_RATE, &ac97->rates[AC97_RATES_SURR_DAC]);
+ ac97->scaps |= AC97_SCAP_SURROUND_DAC;
+ }
+ if (ac97->ext_id & AC97_EI_LDAC) { /* LDAC support */
+ snd_ac97_determine_rates(ac97, AC97_PCM_LFE_DAC_RATE, AC97_PCM_FRONT_DAC_RATE, &ac97->rates[AC97_RATES_LFE_DAC]);
+ ac97->scaps |= AC97_SCAP_CENTER_LFE_DAC;
+ }
+ /* additional initializations */
+ if (bus->ops->init)
+ bus->ops->init(ac97);
+ snd_ac97_get_name(ac97, ac97->id, name, !ac97_is_audio(ac97));
+ snd_ac97_get_name(NULL, ac97->id, name, !ac97_is_audio(ac97)); // ac97->id might be changed in the special setup code
+ if (! ac97->build_ops)
+ ac97->build_ops = &null_build_ops;
+
+ if (ac97_is_audio(ac97)) {
+ char comp[16];
+ if (card->mixername[0] == '\0') {
+ strcpy(card->mixername, name);
+ } else {
+ if (strlen(card->mixername) + 1 + strlen(name) + 1 <= sizeof(card->mixername)) {
+ strcat(card->mixername, ",");
+ strcat(card->mixername, name);
+ }
+ }
+ sprintf(comp, "AC97a:%08x", ac97->id);
+ if ((err = snd_component_add(card, comp)) < 0) {
+ snd_ac97_free(ac97);
+ return err;
+ }
+ if (snd_ac97_mixer_build(ac97) < 0) {
+ snd_ac97_free(ac97);
+ return -ENOMEM;
+ }
+ }
+ if (ac97_is_modem(ac97)) {
+ char comp[16];
+ if (card->mixername[0] == '\0') {
+ strcpy(card->mixername, name);
+ } else {
+ if (strlen(card->mixername) + 1 + strlen(name) + 1 <= sizeof(card->mixername)) {
+ strcat(card->mixername, ",");
+ strcat(card->mixername, name);
+ }
+ }
+ sprintf(comp, "AC97m:%08x", ac97->id);
+ if ((err = snd_component_add(card, comp)) < 0) {
+ snd_ac97_free(ac97);
+ return err;
+ }
+ if (snd_ac97_modem_build(card, ac97) < 0) {
+ snd_ac97_free(ac97);
+ return -ENOMEM;
+ }
+ }
+ if (ac97_is_audio(ac97))
+ update_power_regs(ac97);
+ snd_ac97_proc_init(ac97);
+ if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ac97, &ops)) < 0) {
+ snd_ac97_free(ac97);
+ return err;
+ }
+ *rac97 = ac97;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_mixer);
+
+/*
+ * Power down the chip.
+ *
+ * MASTER and HEADPHONE registers are muted but the register cache values
+ * are not changed, so that the values can be restored in snd_ac97_resume().
+ */
+static void snd_ac97_powerdown(struct snd_ac97 *ac97)
+{
+ unsigned short power;
+
+ if (ac97_is_audio(ac97)) {
+ /* some codecs have stereo mute bits */
+ snd_ac97_write(ac97, AC97_MASTER, 0x9f9f);
+ snd_ac97_write(ac97, AC97_HEADPHONE, 0x9f9f);
+ }
+
+ /* surround, CLFE, mic powerdown */
+ power = ac97->regs[AC97_EXTENDED_STATUS];
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
+ power |= AC97_EA_PRJ;
+ if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)
+ power |= AC97_EA_PRI | AC97_EA_PRK;
+ power |= AC97_EA_PRL;
+ snd_ac97_write(ac97, AC97_EXTENDED_STATUS, power);
+
+ /* powerdown external amplifier */
+ if (ac97->scaps & AC97_SCAP_INV_EAPD)
+ power = ac97->regs[AC97_POWERDOWN] & ~AC97_PD_EAPD;
+ else if (! (ac97->scaps & AC97_SCAP_EAPD_LED))
+ power = ac97->regs[AC97_POWERDOWN] | AC97_PD_EAPD;
+ power |= AC97_PD_PR6; /* Headphone amplifier powerdown */
+ power |= AC97_PD_PR0 | AC97_PD_PR1; /* ADC & DAC powerdown */
+ snd_ac97_write(ac97, AC97_POWERDOWN, power);
+ udelay(100);
+ power |= AC97_PD_PR2; /* Analog Mixer powerdown (Vref on) */
+ snd_ac97_write(ac97, AC97_POWERDOWN, power);
+ if (ac97_is_power_save_mode(ac97)) {
+ power |= AC97_PD_PR3; /* Analog Mixer powerdown */
+ snd_ac97_write(ac97, AC97_POWERDOWN, power);
+ udelay(100);
+ /* AC-link powerdown, internal Clk disable */
+ /* FIXME: this may cause click noises on some boards */
+ power |= AC97_PD_PR4 | AC97_PD_PR5;
+ snd_ac97_write(ac97, AC97_POWERDOWN, power);
+ }
+}
+
+
+struct ac97_power_reg {
+ unsigned short reg;
+ unsigned short power_reg;
+ unsigned short mask;
+};
+
+enum { PWIDX_ADC, PWIDX_FRONT, PWIDX_CLFE, PWIDX_SURR, PWIDX_MIC, PWIDX_SIZE };
+
+static struct ac97_power_reg power_regs[PWIDX_SIZE] = {
+ [PWIDX_ADC] = { AC97_PCM_LR_ADC_RATE, AC97_POWERDOWN, AC97_PD_PR0},
+ [PWIDX_FRONT] = { AC97_PCM_FRONT_DAC_RATE, AC97_POWERDOWN, AC97_PD_PR1},
+ [PWIDX_CLFE] = { AC97_PCM_LFE_DAC_RATE, AC97_EXTENDED_STATUS,
+ AC97_EA_PRI | AC97_EA_PRK},
+ [PWIDX_SURR] = { AC97_PCM_SURR_DAC_RATE, AC97_EXTENDED_STATUS,
+ AC97_EA_PRJ},
+ [PWIDX_MIC] = { AC97_PCM_MIC_ADC_RATE, AC97_EXTENDED_STATUS,
+ AC97_EA_PRL},
+};
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+/**
+ * snd_ac97_update_power - update the powerdown register
+ * @ac97: the codec instance
+ * @reg: the rate register, e.g. AC97_PCM_FRONT_DAC_RATE
+ * @powerup: non-zero when power up the part
+ *
+ * Update the AC97 powerdown register bits of the given part.
+ */
+int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup)
+{
+ int i;
+
+ if (! ac97)
+ return 0;
+
+ if (reg) {
+ /* SPDIF requires DAC power, too */
+ if (reg == AC97_SPDIF)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ for (i = 0; i < PWIDX_SIZE; i++) {
+ if (power_regs[i].reg == reg) {
+ if (powerup)
+ ac97->power_up |= (1 << i);
+ else
+ ac97->power_up &= ~(1 << i);
+ break;
+ }
+ }
+ }
+
+ if (ac97_is_power_save_mode(ac97) && !powerup)
+ /* adjust power-down bits after two seconds delay
+ * (for avoiding loud click noises for many (OSS) apps
+ * that open/close frequently)
+ */
+ schedule_delayed_work(&ac97->power_work,
+ msecs_to_jiffies(power_save * 1000));
+ else {
+ cancel_delayed_work(&ac97->power_work);
+ update_power_regs(ac97);
+ }
+
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_update_power);
+#endif /* CONFIG_SND_AC97_POWER_SAVE */
+
+static void update_power_regs(struct snd_ac97 *ac97)
+{
+ unsigned int power_up, bits;
+ int i;
+
+ power_up = (1 << PWIDX_FRONT) | (1 << PWIDX_ADC);
+ power_up |= (1 << PWIDX_MIC);
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
+ power_up |= (1 << PWIDX_SURR);
+ if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)
+ power_up |= (1 << PWIDX_CLFE);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ if (ac97_is_power_save_mode(ac97))
+ power_up = ac97->power_up;
+#endif
+ if (power_up) {
+ if (ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2) {
+ /* needs power-up analog mix and vref */
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR3, 0);
+ msleep(1);
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR2, 0);
+ }
+ }
+ for (i = 0; i < PWIDX_SIZE; i++) {
+ if (power_up & (1 << i))
+ bits = 0;
+ else
+ bits = power_regs[i].mask;
+ snd_ac97_update_bits(ac97, power_regs[i].power_reg,
+ power_regs[i].mask, bits);
+ }
+ if (! power_up) {
+ if (! (ac97->regs[AC97_POWERDOWN] & AC97_PD_PR2)) {
+ /* power down analog mix and vref */
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR2, AC97_PD_PR2);
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN,
+ AC97_PD_PR3, AC97_PD_PR3);
+ }
+ }
+}
+
+
+#ifdef CONFIG_PM
+/**
+ * snd_ac97_suspend - General suspend function for AC97 codec
+ * @ac97: the ac97 instance
+ *
+ * Suspends the codec, power down the chip.
+ */
+void snd_ac97_suspend(struct snd_ac97 *ac97)
+{
+ if (! ac97)
+ return;
+ if (ac97->build_ops->suspend)
+ ac97->build_ops->suspend(ac97);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ cancel_delayed_work(&ac97->power_work);
+ flush_scheduled_work();
+#endif
+ snd_ac97_powerdown(ac97);
+}
+
+EXPORT_SYMBOL(snd_ac97_suspend);
+
+/*
+ * restore ac97 status
+ */
+static void snd_ac97_restore_status(struct snd_ac97 *ac97)
+{
+ int i;
+
+ for (i = 2; i < 0x7c ; i += 2) {
+ if (i == AC97_POWERDOWN || i == AC97_EXTENDED_ID)
+ continue;
+ /* restore only accessible registers
+ * some chip (e.g. nm256) may hang up when unsupported registers
+ * are accessed..!
+ */
+ if (test_bit(i, ac97->reg_accessed)) {
+ snd_ac97_write(ac97, i, ac97->regs[i]);
+ snd_ac97_read(ac97, i);
+ }
+ }
+}
+
+/*
+ * restore IEC958 status
+ */
+static void snd_ac97_restore_iec958(struct snd_ac97 *ac97)
+{
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ if (ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_SPDIF) {
+ /* reset spdif status */
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0);
+ snd_ac97_write(ac97, AC97_EXTENDED_STATUS, ac97->regs[AC97_EXTENDED_STATUS]);
+ if (ac97->flags & AC97_CS_SPDIF)
+ snd_ac97_write(ac97, AC97_CSR_SPDIF, ac97->regs[AC97_CSR_SPDIF]);
+ else
+ snd_ac97_write(ac97, AC97_SPDIF, ac97->regs[AC97_SPDIF]);
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, AC97_EA_SPDIF); /* turn on again */
+ }
+ }
+}
+
+/**
+ * snd_ac97_resume - General resume function for AC97 codec
+ * @ac97: the ac97 instance
+ *
+ * Do the standard resume procedure, power up and restoring the
+ * old register values.
+ */
+void snd_ac97_resume(struct snd_ac97 *ac97)
+{
+ unsigned long end_time;
+
+ if (! ac97)
+ return;
+
+ if (ac97->bus->ops->reset) {
+ ac97->bus->ops->reset(ac97);
+ goto __reset_ready;
+ }
+
+ snd_ac97_write(ac97, AC97_POWERDOWN, 0);
+ if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) {
+ if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO))
+ snd_ac97_write(ac97, AC97_RESET, 0);
+ else if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM))
+ snd_ac97_write(ac97, AC97_EXTENDED_MID, 0);
+ udelay(100);
+ snd_ac97_write(ac97, AC97_POWERDOWN, 0);
+ }
+ snd_ac97_write(ac97, AC97_GENERAL_PURPOSE, 0);
+
+ snd_ac97_write(ac97, AC97_POWERDOWN, ac97->regs[AC97_POWERDOWN]);
+ if (ac97_is_audio(ac97)) {
+ ac97->bus->ops->write(ac97, AC97_MASTER, 0x8101);
+ end_time = jiffies + msecs_to_jiffies(100);
+ do {
+ if (snd_ac97_read(ac97, AC97_MASTER) == 0x8101)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ /* FIXME: extra delay */
+ ac97->bus->ops->write(ac97, AC97_MASTER, 0x8000);
+ if (snd_ac97_read(ac97, AC97_MASTER) != 0x8000)
+ msleep(250);
+ } else {
+ end_time = jiffies + msecs_to_jiffies(100);
+ do {
+ unsigned short val = snd_ac97_read(ac97, AC97_EXTENDED_MID);
+ if (val != 0xffff && (val & 1) != 0)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ }
+__reset_ready:
+
+ if (ac97->bus->ops->init)
+ ac97->bus->ops->init(ac97);
+
+ if (ac97->build_ops->resume)
+ ac97->build_ops->resume(ac97);
+ else {
+ snd_ac97_restore_status(ac97);
+ snd_ac97_restore_iec958(ac97);
+ }
+}
+
+EXPORT_SYMBOL(snd_ac97_resume);
+#endif
+
+
+/*
+ * Hardware tuning
+ */
+static void set_ctl_name(char *dst, const char *src, const char *suffix)
+{
+ if (suffix)
+ sprintf(dst, "%s %s", src, suffix);
+ else
+ strcpy(dst, src);
+}
+
+/* remove the control with the given name and optional suffix */
+static int snd_ac97_remove_ctl(struct snd_ac97 *ac97, const char *name,
+ const char *suffix)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ set_ctl_name(id.name, name, suffix);
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_remove_id(ac97->bus->card, &id);
+}
+
+static struct snd_kcontrol *ctl_find(struct snd_ac97 *ac97, const char *name, const char *suffix)
+{
+ struct snd_ctl_elem_id sid;
+ memset(&sid, 0, sizeof(sid));
+ set_ctl_name(sid.name, name, suffix);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_find_id(ac97->bus->card, &sid);
+}
+
+/* rename the control with the given name and optional suffix */
+static int snd_ac97_rename_ctl(struct snd_ac97 *ac97, const char *src,
+ const char *dst, const char *suffix)
+{
+ struct snd_kcontrol *kctl = ctl_find(ac97, src, suffix);
+ if (kctl) {
+ set_ctl_name(kctl->id.name, dst, suffix);
+ return 0;
+ }
+ return -ENOENT;
+}
+
+/* rename both Volume and Switch controls - don't check the return value */
+static void snd_ac97_rename_vol_ctl(struct snd_ac97 *ac97, const char *src,
+ const char *dst)
+{
+ snd_ac97_rename_ctl(ac97, src, dst, "Switch");
+ snd_ac97_rename_ctl(ac97, src, dst, "Volume");
+}
+
+/* swap controls */
+static int snd_ac97_swap_ctl(struct snd_ac97 *ac97, const char *s1,
+ const char *s2, const char *suffix)
+{
+ struct snd_kcontrol *kctl1, *kctl2;
+ kctl1 = ctl_find(ac97, s1, suffix);
+ kctl2 = ctl_find(ac97, s2, suffix);
+ if (kctl1 && kctl2) {
+ set_ctl_name(kctl1->id.name, s2, suffix);
+ set_ctl_name(kctl2->id.name, s1, suffix);
+ return 0;
+ }
+ return -ENOENT;
+}
+
+#if 1
+/* bind hp and master controls instead of using only hp control */
+static int bind_hp_volsw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ int err = snd_ac97_put_volsw(kcontrol, ucontrol);
+ if (err > 0) {
+ unsigned long priv_saved = kcontrol->private_value;
+ kcontrol->private_value = (kcontrol->private_value & ~0xff) | AC97_HEADPHONE;
+ snd_ac97_put_volsw(kcontrol, ucontrol);
+ kcontrol->private_value = priv_saved;
+ }
+ return err;
+}
+
+/* ac97 tune: bind Master and Headphone controls */
+static int tune_hp_only(struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *msw = ctl_find(ac97, "Master Playback Switch", NULL);
+ struct snd_kcontrol *mvol = ctl_find(ac97, "Master Playback Volume", NULL);
+ if (! msw || ! mvol)
+ return -ENOENT;
+ msw->put = bind_hp_volsw_put;
+ mvol->put = bind_hp_volsw_put;
+ snd_ac97_remove_ctl(ac97, "Headphone Playback", "Switch");
+ snd_ac97_remove_ctl(ac97, "Headphone Playback", "Volume");
+ return 0;
+}
+
+#else
+/* ac97 tune: use Headphone control as master */
+static int tune_hp_only(struct snd_ac97 *ac97)
+{
+ if (ctl_find(ac97, "Headphone Playback Switch", NULL) == NULL)
+ return -ENOENT;
+ snd_ac97_remove_ctl(ac97, "Master Playback", "Switch");
+ snd_ac97_remove_ctl(ac97, "Master Playback", "Volume");
+ snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback");
+ return 0;
+}
+#endif
+
+/* ac97 tune: swap Headphone and Master controls */
+static int tune_swap_hp(struct snd_ac97 *ac97)
+{
+ if (ctl_find(ac97, "Headphone Playback Switch", NULL) == NULL)
+ return -ENOENT;
+ snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Line-Out Playback");
+ snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback");
+ return 0;
+}
+
+/* ac97 tune: swap Surround and Master controls */
+static int tune_swap_surround(struct snd_ac97 *ac97)
+{
+ if (snd_ac97_swap_ctl(ac97, "Master Playback", "Surround Playback", "Switch") ||
+ snd_ac97_swap_ctl(ac97, "Master Playback", "Surround Playback", "Volume"))
+ return -ENOENT;
+ return 0;
+}
+
+/* ac97 tune: set up mic sharing for AD codecs */
+static int tune_ad_sharing(struct snd_ac97 *ac97)
+{
+ unsigned short scfg;
+ if ((ac97->id & 0xffffff00) != 0x41445300) {
+ snd_printk(KERN_ERR "ac97_quirk AD_SHARING is only for AD codecs\n");
+ return -EINVAL;
+ }
+ /* Turn on OMS bit to route microphone to back panel */
+ scfg = snd_ac97_read(ac97, AC97_AD_SERIAL_CFG);
+ snd_ac97_write_cache(ac97, AC97_AD_SERIAL_CFG, scfg | 0x0200);
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_alc_jack_detect =
+AC97_SINGLE("Jack Detect", AC97_ALC650_CLOCK, 5, 1, 0);
+
+/* ac97 tune: set up ALC jack-select */
+static int tune_alc_jack(struct snd_ac97 *ac97)
+{
+ if ((ac97->id & 0xffffff00) != 0x414c4700) {
+ snd_printk(KERN_ERR "ac97_quirk ALC_JACK is only for Realtek codecs\n");
+ return -EINVAL;
+ }
+ snd_ac97_update_bits(ac97, 0x7a, 0x20, 0x20); /* select jack detect function */
+ snd_ac97_update_bits(ac97, 0x7a, 0x01, 0x01); /* Line-out auto mute */
+ if (ac97->id == AC97_ID_ALC658D)
+ snd_ac97_update_bits(ac97, 0x74, 0x0800, 0x0800);
+ return snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_alc_jack_detect, ac97));
+}
+
+/* ac97 tune: inversed EAPD bit */
+static int tune_inv_eapd(struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *kctl = ctl_find(ac97, "External Amplifier", NULL);
+ if (! kctl)
+ return -ENOENT;
+ set_inv_eapd(ac97, kctl);
+ return 0;
+}
+
+static int master_mute_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ int err = snd_ac97_put_volsw(kcontrol, ucontrol);
+ if (err > 0) {
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ unsigned short mask;
+ if (shift != rshift)
+ mask = 0x8080;
+ else
+ mask = 0x8000;
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000,
+ (ac97->regs[AC97_MASTER] & mask) == mask ?
+ 0x8000 : 0);
+ }
+ return err;
+}
+
+/* ac97 tune: EAPD controls mute LED bound with the master mute */
+static int tune_mute_led(struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *msw = ctl_find(ac97, "Master Playback Switch", NULL);
+ if (! msw)
+ return -ENOENT;
+ msw->put = master_mute_sw_put;
+ snd_ac97_remove_ctl(ac97, "External Amplifier", NULL);
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */
+ ac97->scaps |= AC97_SCAP_EAPD_LED;
+ return 0;
+}
+
+static int hp_master_mute_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int err = bind_hp_volsw_put(kcontrol, ucontrol);
+ if (err > 0) {
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int rshift = (kcontrol->private_value >> 12) & 0x0f;
+ unsigned short mask;
+ if (shift != rshift)
+ mask = 0x8080;
+ else
+ mask = 0x8000;
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000,
+ (ac97->regs[AC97_MASTER] & mask) == mask ?
+ 0x8000 : 0);
+ }
+ return err;
+}
+
+static int tune_hp_mute_led(struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *msw = ctl_find(ac97, "Master Playback Switch", NULL);
+ struct snd_kcontrol *mvol = ctl_find(ac97, "Master Playback Volume", NULL);
+ if (! msw || ! mvol)
+ return -ENOENT;
+ msw->put = hp_master_mute_sw_put;
+ mvol->put = bind_hp_volsw_put;
+ snd_ac97_remove_ctl(ac97, "External Amplifier", NULL);
+ snd_ac97_remove_ctl(ac97, "Headphone Playback", "Switch");
+ snd_ac97_remove_ctl(ac97, "Headphone Playback", "Volume");
+ snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */
+ return 0;
+}
+
+struct quirk_table {
+ const char *name;
+ int (*func)(struct snd_ac97 *);
+};
+
+static struct quirk_table applicable_quirks[] = {
+ { "none", NULL },
+ { "hp_only", tune_hp_only },
+ { "swap_hp", tune_swap_hp },
+ { "swap_surround", tune_swap_surround },
+ { "ad_sharing", tune_ad_sharing },
+ { "alc_jack", tune_alc_jack },
+ { "inv_eapd", tune_inv_eapd },
+ { "mute_led", tune_mute_led },
+ { "hp_mute_led", tune_hp_mute_led },
+};
+
+/* apply the quirk with the given type */
+static int apply_quirk(struct snd_ac97 *ac97, int type)
+{
+ if (type <= 0)
+ return 0;
+ else if (type >= ARRAY_SIZE(applicable_quirks))
+ return -EINVAL;
+ if (applicable_quirks[type].func)
+ return applicable_quirks[type].func(ac97);
+ return 0;
+}
+
+/* apply the quirk with the given name */
+static int apply_quirk_str(struct snd_ac97 *ac97, const char *typestr)
+{
+ int i;
+ struct quirk_table *q;
+
+ for (i = 0; i < ARRAY_SIZE(applicable_quirks); i++) {
+ q = &applicable_quirks[i];
+ if (q->name && ! strcmp(typestr, q->name))
+ return apply_quirk(ac97, i);
+ }
+ /* for compatibility, accept the numbers, too */
+ if (*typestr >= '0' && *typestr <= '9')
+ return apply_quirk(ac97, (int)simple_strtoul(typestr, NULL, 10));
+ return -EINVAL;
+}
+
+/**
+ * snd_ac97_tune_hardware - tune up the hardware
+ * @ac97: the ac97 instance
+ * @quirk: quirk list
+ * @override: explicit quirk value (overrides the list if non-NULL)
+ *
+ * Do some workaround for each pci device, such as renaming of the
+ * headphone (true line-out) control as "Master".
+ * The quirk-list must be terminated with a zero-filled entry.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+
+int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override)
+{
+ int result;
+
+ /* quirk overriden? */
+ if (override && strcmp(override, "-1") && strcmp(override, "default")) {
+ result = apply_quirk_str(ac97, override);
+ if (result < 0)
+ snd_printk(KERN_ERR "applying quirk type %s failed (%d)\n", override, result);
+ return result;
+ }
+
+ if (! quirk)
+ return -EINVAL;
+
+ for (; quirk->subvendor; quirk++) {
+ if (quirk->subvendor != ac97->subsystem_vendor)
+ continue;
+ if ((! quirk->mask && quirk->subdevice == ac97->subsystem_device) ||
+ quirk->subdevice == (quirk->mask & ac97->subsystem_device)) {
+ if (quirk->codec_id && quirk->codec_id != ac97->id)
+ continue;
+ snd_printdd("ac97 quirk for %s (%04x:%04x)\n", quirk->name, ac97->subsystem_vendor, ac97->subsystem_device);
+ result = apply_quirk(ac97, quirk->type);
+ if (result < 0)
+ snd_printk(KERN_ERR "applying quirk type %d for %s failed (%d)\n", quirk->type, quirk->name, result);
+ return result;
+ }
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_tune_hardware);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_ac97_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_ac97_exit(void)
+{
+}
+
+module_init(alsa_ac97_init)
+module_exit(alsa_ac97_exit)
diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h
new file mode 100644
index 0000000..c129492
--- /dev/null
+++ b/sound/pci/ac97/ac97_id.h
@@ -0,0 +1,64 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.2
+ * by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define AC97_ID_AK4540 0x414b4d00
+#define AC97_ID_AK4542 0x414b4d01
+#define AC97_ID_AD1819 0x41445303
+#define AC97_ID_AD1881 0x41445340
+#define AC97_ID_AD1881A 0x41445348
+#define AC97_ID_AD1885 0x41445360
+#define AC97_ID_AD1886 0x41445361
+#define AC97_ID_AD1887 0x41445362
+#define AC97_ID_AD1886A 0x41445363
+#define AC97_ID_AD1980 0x41445370
+#define AC97_ID_TR28028 0x54524108
+#define AC97_ID_STAC9700 0x83847600
+#define AC97_ID_STAC9704 0x83847604
+#define AC97_ID_STAC9705 0x83847605
+#define AC97_ID_STAC9708 0x83847608
+#define AC97_ID_STAC9721 0x83847609
+#define AC97_ID_STAC9744 0x83847644
+#define AC97_ID_STAC9756 0x83847656
+#define AC97_ID_CS4297A 0x43525910
+#define AC97_ID_CS4299 0x43525930
+#define AC97_ID_CS4201 0x43525948
+#define AC97_ID_CS4205 0x43525958
+#define AC97_ID_CS_MASK 0xfffffff8 /* bit 0-2: rev */
+#define AC97_ID_ALC100 0x414c4300
+#define AC97_ID_ALC650 0x414c4720
+#define AC97_ID_ALC650D 0x414c4721
+#define AC97_ID_ALC650E 0x414c4722
+#define AC97_ID_ALC650F 0x414c4723
+#define AC97_ID_ALC655 0x414c4760
+#define AC97_ID_ALC658 0x414c4780
+#define AC97_ID_ALC658D 0x414c4781
+#define AC97_ID_ALC850 0x414c4790
+#define AC97_ID_YMF743 0x594d4800
+#define AC97_ID_YMF753 0x594d4803
+#define AC97_ID_VT1616 0x49434551
+#define AC97_ID_CM9738 0x434d4941
+#define AC97_ID_CM9739 0x434d4961
+#define AC97_ID_CM9761_78 0x434d4978
+#define AC97_ID_CM9761_82 0x434d4982
+#define AC97_ID_CM9761_83 0x434d4983
diff --git a/sound/pci/ac97/ac97_local.h b/sound/pci/ac97/ac97_local.h
new file mode 100644
index 0000000..c276a5e
--- /dev/null
+++ b/sound/pci/ac97/ac97_local.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.2
+ * by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name,
+ int modem);
+int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short mask, unsigned short value);
+
+/* ac97_proc.c */
+#ifdef CONFIG_PROC_FS
+void snd_ac97_bus_proc_init(struct snd_ac97_bus * ac97);
+void snd_ac97_bus_proc_done(struct snd_ac97_bus * ac97);
+void snd_ac97_proc_init(struct snd_ac97 * ac97);
+void snd_ac97_proc_done(struct snd_ac97 * ac97);
+#else
+#define snd_ac97_bus_proc_init(ac97_bus_t) do { } while (0)
+#define snd_ac97_bus_proc_done(ac97_bus_t) do { } while (0)
+#define snd_ac97_proc_init(ac97_t) do { } while (0)
+#define snd_ac97_proc_done(ac97_t) do { } while (0)
+#endif
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
new file mode 100644
index 0000000..6e831af
--- /dev/null
+++ b/sound/pci/ac97/ac97_patch.c
@@ -0,0 +1,3918 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.2
+ * by Intel Corporation (http://developer.intel.com) and to datasheets
+ * for specific codecs.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include "ac97_local.h"
+#include "ac97_patch.h"
+
+/*
+ * Chip specific initialization
+ */
+
+static int patch_build_controls(struct snd_ac97 * ac97, const struct snd_kcontrol_new *controls, int count)
+{
+ int idx, err;
+
+ for (idx = 0; idx < count; idx++)
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&controls[idx], ac97))) < 0)
+ return err;
+ return 0;
+}
+
+/* replace with a new TLV */
+static void reset_tlv(struct snd_ac97 *ac97, const char *name,
+ const unsigned int *tlv)
+{
+ struct snd_ctl_elem_id sid;
+ struct snd_kcontrol *kctl;
+ memset(&sid, 0, sizeof(sid));
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ kctl = snd_ctl_find_id(ac97->bus->card, &sid);
+ if (kctl && kctl->tlv.p)
+ kctl->tlv.p = tlv;
+}
+
+/* set to the page, update bits and restore the page */
+static int ac97_update_bits_page(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value, unsigned short page)
+{
+ unsigned short page_save;
+ int ret;
+
+ mutex_lock(&ac97->page_mutex);
+ page_save = snd_ac97_read(ac97, AC97_INT_PAGING) & AC97_PAGE_MASK;
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page);
+ ret = snd_ac97_update_bits(ac97, reg, mask, value);
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page_save);
+ mutex_unlock(&ac97->page_mutex); /* unlock paging */
+ return ret;
+}
+
+/*
+ * shared line-in/mic controls
+ */
+static int ac97_enum_text_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo,
+ const char **texts, unsigned int nums)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = nums;
+ if (uinfo->value.enumerated.item > nums - 1)
+ uinfo->value.enumerated.item = nums - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int ac97_surround_jack_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static const char *texts[] = { "Shared", "Independent" };
+ return ac97_enum_text_info(kcontrol, uinfo, texts, 2);
+}
+
+static int ac97_surround_jack_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = ac97->indep_surround;
+ return 0;
+}
+
+static int ac97_surround_jack_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned char indep = !!ucontrol->value.enumerated.item[0];
+
+ if (indep != ac97->indep_surround) {
+ ac97->indep_surround = indep;
+ if (ac97->build_ops->update_jacks)
+ ac97->build_ops->update_jacks(ac97);
+ return 1;
+ }
+ return 0;
+}
+
+static int ac97_channel_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static const char *texts[] = { "2ch", "4ch", "6ch", "8ch" };
+ return ac97_enum_text_info(kcontrol, uinfo, texts,
+ kcontrol->private_value);
+}
+
+static int ac97_channel_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = ac97->channel_mode;
+ return 0;
+}
+
+static int ac97_channel_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned char mode = ucontrol->value.enumerated.item[0];
+
+ if (mode >= kcontrol->private_value)
+ return -EINVAL;
+
+ if (mode != ac97->channel_mode) {
+ ac97->channel_mode = mode;
+ if (ac97->build_ops->update_jacks)
+ ac97->build_ops->update_jacks(ac97);
+ return 1;
+ }
+ return 0;
+}
+
+#define AC97_SURROUND_JACK_MODE_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Surround Jack Mode", \
+ .info = ac97_surround_jack_mode_info, \
+ .get = ac97_surround_jack_mode_get, \
+ .put = ac97_surround_jack_mode_put, \
+ }
+/* 6ch */
+#define AC97_CHANNEL_MODE_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Channel Mode", \
+ .info = ac97_channel_mode_info, \
+ .get = ac97_channel_mode_get, \
+ .put = ac97_channel_mode_put, \
+ .private_value = 3, \
+ }
+/* 4ch */
+#define AC97_CHANNEL_MODE_4CH_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Channel Mode", \
+ .info = ac97_channel_mode_info, \
+ .get = ac97_channel_mode_get, \
+ .put = ac97_channel_mode_put, \
+ .private_value = 2, \
+ }
+/* 8ch */
+#define AC97_CHANNEL_MODE_8CH_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Channel Mode", \
+ .info = ac97_channel_mode_info, \
+ .get = ac97_channel_mode_get, \
+ .put = ac97_channel_mode_put, \
+ .private_value = 4, \
+ }
+
+static inline int is_surround_on(struct snd_ac97 *ac97)
+{
+ return ac97->channel_mode >= 1;
+}
+
+static inline int is_clfe_on(struct snd_ac97 *ac97)
+{
+ return ac97->channel_mode >= 2;
+}
+
+/* system has shared jacks with surround out enabled */
+static inline int is_shared_surrout(struct snd_ac97 *ac97)
+{
+ return !ac97->indep_surround && is_surround_on(ac97);
+}
+
+/* system has shared jacks with center/lfe out enabled */
+static inline int is_shared_clfeout(struct snd_ac97 *ac97)
+{
+ return !ac97->indep_surround && is_clfe_on(ac97);
+}
+
+/* system has shared jacks with line in enabled */
+static inline int is_shared_linein(struct snd_ac97 *ac97)
+{
+ return !ac97->indep_surround && !is_surround_on(ac97);
+}
+
+/* system has shared jacks with mic in enabled */
+static inline int is_shared_micin(struct snd_ac97 *ac97)
+{
+ return !ac97->indep_surround && !is_clfe_on(ac97);
+}
+
+static inline int alc850_is_aux_back_surround(struct snd_ac97 *ac97)
+{
+ return is_surround_on(ac97);
+}
+
+/* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */
+/* Modified for YMF743 by Keita Maehara <maehara@debian.org> */
+
+/* It is possible to indicate to the Yamaha YMF7x3 the type of
+ speakers being used. */
+
+static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {
+ "Standard", "Small", "Smaller"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_YMF7X3_3D_MODE_SEL];
+ val = (val >> 10) & 3;
+ if (val > 0) /* 0 = invalid */
+ val--;
+ ucontrol->value.enumerated.item[0] = val;
+ return 0;
+}
+
+static int snd_ac97_ymf7x3_put_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ val = (ucontrol->value.enumerated.item[0] + 1) << 10;
+ return snd_ac97_update(ac97, AC97_YMF7X3_3D_MODE_SEL, val);
+}
+
+static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "3D Control - Speaker",
+ .info = snd_ac97_ymf7x3_info_speaker,
+ .get = snd_ac97_ymf7x3_get_speaker,
+ .put = snd_ac97_ymf7x3_put_speaker,
+};
+
+/* It is possible to indicate to the Yamaha YMF7x3 the source to
+ direct to the S/PDIF output. */
+static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "AC-Link", "A/D Converter" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_YMF7X3_DIT_CTRL];
+ ucontrol->value.enumerated.item[0] = (val >> 1) & 1;
+ return 0;
+}
+
+static int snd_ac97_ymf7x3_spdif_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ val = ucontrol->value.enumerated.item[0] << 1;
+ return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val);
+}
+
+static int patch_yamaha_ymf7x3_3d(struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+
+ kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97);
+ err = snd_ctl_add(ac97->bus->card, kctl);
+ if (err < 0)
+ return err;
+ strcpy(kctl->id.name, "3D Control - Wide");
+ kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0);
+ snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
+ err = snd_ctl_add(ac97->bus->card,
+ snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker,
+ ac97));
+ if (err < 0)
+ return err;
+ snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00);
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_yamaha_ymf743_controls_spdif[3] =
+{
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
+ AC97_YMF7X3_DIT_CTRL, 0, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Source",
+ .info = snd_ac97_ymf7x3_spdif_source_info,
+ .get = snd_ac97_ymf7x3_spdif_source_get,
+ .put = snd_ac97_ymf7x3_spdif_source_put,
+ },
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute",
+ AC97_YMF7X3_DIT_CTRL, 2, 1, 1)
+};
+
+static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 *ac97)
+{
+ int err;
+
+ err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3);
+ if (err < 0)
+ return err;
+ err = patch_build_controls(ac97,
+ snd_ac97_yamaha_ymf743_controls_spdif, 3);
+ if (err < 0)
+ return err;
+ /* set default PCM S/PDIF params */
+ /* PCM audio,no copyright,no preemphasis,PCM coder,original */
+ snd_ac97_write_cache(ac97, AC97_YMF7X3_DIT_CTRL, 0xa201);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = {
+ .build_spdif = patch_yamaha_ymf743_build_spdif,
+ .build_3d = patch_yamaha_ymf7x3_3d,
+};
+
+static int patch_yamaha_ymf743(struct snd_ac97 *ac97)
+{
+ ac97->build_ops = &patch_yamaha_ymf743_ops;
+ ac97->caps |= AC97_BC_BASS_TREBLE;
+ ac97->caps |= 0x04 << 10; /* Yamaha 3D enhancement */
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+ ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
+ return 0;
+}
+
+/* The AC'97 spec states that the S/PDIF signal is to be output at pin 48.
+ The YMF753 will output the S/PDIF signal to pin 43, 47 (EAPD), or 48.
+ By default, no output pin is selected, and the S/PDIF signal is not output.
+ There is also a bit to mute S/PDIF output in a vendor-specific register. */
+static int snd_ac97_ymf753_spdif_output_pin_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = { "Disabled", "Pin 43", "Pin 48" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_ymf753_spdif_output_pin_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_YMF7X3_DIT_CTRL];
+ ucontrol->value.enumerated.item[0] = (val & 0x0008) ? 2 : (val & 0x0020) ? 1 : 0;
+ return 0;
+}
+
+static int snd_ac97_ymf753_spdif_output_pin_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ val = (ucontrol->value.enumerated.item[0] == 2) ? 0x0008 :
+ (ucontrol->value.enumerated.item[0] == 1) ? 0x0020 : 0;
+ return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0028, val);
+ /* The following can be used to direct S/PDIF output to pin 47 (EAPD).
+ snd_ac97_write_cache(ac97, 0x62, snd_ac97_read(ac97, 0x62) | 0x0008); */
+}
+
+static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = snd_ac97_ymf7x3_spdif_source_info,
+ .get = snd_ac97_ymf7x3_spdif_source_get,
+ .put = snd_ac97_ymf7x3_spdif_source_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Output Pin",
+ .info = snd_ac97_ymf753_spdif_output_pin_info,
+ .get = snd_ac97_ymf753_spdif_output_pin_get,
+ .put = snd_ac97_ymf753_spdif_output_pin_put,
+ },
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute",
+ AC97_YMF7X3_DIT_CTRL, 2, 1, 1)
+};
+
+static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97)
+{
+ int err;
+
+ if ((err = patch_build_controls(ac97, snd_ac97_ymf753_controls_spdif, ARRAY_SIZE(snd_ac97_ymf753_controls_spdif))) < 0)
+ return err;
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = {
+ .build_3d = patch_yamaha_ymf7x3_3d,
+ .build_post_spdif = patch_yamaha_ymf753_post_spdif
+};
+
+static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
+{
+ /* Patch for Yamaha YMF753, Copyright (c) by David Shust, dshust@shustring.com.
+ This chip has nonstandard and extended behaviour with regard to its S/PDIF output.
+ The AC'97 spec states that the S/PDIF signal is to be output at pin 48.
+ The YMF753 will ouput the S/PDIF signal to pin 43, 47 (EAPD), or 48.
+ By default, no output pin is selected, and the S/PDIF signal is not output.
+ There is also a bit to mute S/PDIF output in a vendor-specific register.
+ */
+ ac97->build_ops = &patch_yamaha_ymf753_ops;
+ ac97->caps |= AC97_BC_BASS_TREBLE;
+ ac97->caps |= 0x04 << 10; /* Yamaha 3D enhancement */
+ return 0;
+}
+
+/*
+ * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
+ * removed broken wolfson00 patch.
+ * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
+ */
+
+static const struct snd_kcontrol_new wm97xx_snd_ac97_controls[] = {
+AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1),
+AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1),
+};
+
+static int patch_wolfson_wm9703_specific(struct snd_ac97 * ac97)
+{
+ /* This is known to work for the ViewSonic ViewPad 1000
+ * Randolph Bentson <bentson@holmsjoen.com>
+ * WM9703/9707/9708/9717
+ */
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) {
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0)
+ return err;
+ }
+ snd_ac97_write_cache(ac97, AC97_WM97XX_FMIXER_VOL, 0x0808);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_wolfson_wm9703_ops = {
+ .build_specific = patch_wolfson_wm9703_specific,
+};
+
+static int patch_wolfson03(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_wolfson_wm9703_ops;
+ return 0;
+}
+
+static const struct snd_kcontrol_new wm9704_snd_ac97_controls[] = {
+AC97_DOUBLE("Front Playback Volume", AC97_WM97XX_FMIXER_VOL, 8, 0, 31, 1),
+AC97_SINGLE("Front Playback Switch", AC97_WM97XX_FMIXER_VOL, 15, 1, 1),
+AC97_DOUBLE("Rear Playback Volume", AC97_WM9704_RMIXER_VOL, 8, 0, 31, 1),
+AC97_SINGLE("Rear Playback Switch", AC97_WM9704_RMIXER_VOL, 15, 1, 1),
+AC97_DOUBLE("Rear DAC Volume", AC97_WM9704_RPCM_VOL, 8, 0, 31, 1),
+AC97_DOUBLE("Surround Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
+};
+
+static int patch_wolfson_wm9704_specific(struct snd_ac97 * ac97)
+{
+ int err, i;
+ for (i = 0; i < ARRAY_SIZE(wm9704_snd_ac97_controls); i++) {
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm9704_snd_ac97_controls[i], ac97))) < 0)
+ return err;
+ }
+ /* patch for DVD noise */
+ snd_ac97_write_cache(ac97, AC97_WM9704_TEST, 0x0200);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_wolfson_wm9704_ops = {
+ .build_specific = patch_wolfson_wm9704_specific,
+};
+
+static int patch_wolfson04(struct snd_ac97 * ac97)
+{
+ /* WM9704M/9704Q */
+ ac97->build_ops = &patch_wolfson_wm9704_ops;
+ return 0;
+}
+
+static int patch_wolfson_wm9705_specific(struct snd_ac97 * ac97)
+{
+ int err, i;
+ for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) {
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0)
+ return err;
+ }
+ snd_ac97_write_cache(ac97, 0x72, 0x0808);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_wolfson_wm9705_ops = {
+ .build_specific = patch_wolfson_wm9705_specific,
+};
+
+static int patch_wolfson05(struct snd_ac97 * ac97)
+{
+ /* WM9705, WM9710 */
+ ac97->build_ops = &patch_wolfson_wm9705_ops;
+#ifdef CONFIG_TOUCHSCREEN_WM9705
+ /* WM9705 touchscreen uses AUX and VIDEO for touch */
+ ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX;
+#endif
+ return 0;
+}
+
+static const char* wm9711_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char* wm9711_alc_mix[] = {"Stereo", "Right", "Left", "None"};
+static const char* wm9711_out3_src[] = {"Left", "VREF", "Left + Right", "Mono"};
+static const char* wm9711_out3_lrsrc[] = {"Master Mix", "Headphone Mix"};
+static const char* wm9711_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
+static const char* wm9711_base[] = {"Linear Control", "Adaptive Boost"};
+static const char* wm9711_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char* wm9711_mic[] = {"Mic 1", "Differential", "Mic 2", "Stereo"};
+static const char* wm9711_rec_sel[] =
+ {"Mic 1", "NC", "NC", "Master Mix", "Line", "Headphone Mix", "Phone Mix", "Phone"};
+static const char* wm9711_ng_type[] = {"Constant Gain", "Mute"};
+
+static const struct ac97_enum wm9711_enum[] = {
+AC97_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9711_alc_select),
+AC97_ENUM_SINGLE(AC97_VIDEO, 10, 4, wm9711_alc_mix),
+AC97_ENUM_SINGLE(AC97_AUX, 9, 4, wm9711_out3_src),
+AC97_ENUM_SINGLE(AC97_AUX, 8, 2, wm9711_out3_lrsrc),
+AC97_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9711_rec_adc),
+AC97_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9711_base),
+AC97_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9711_rec_gain),
+AC97_ENUM_SINGLE(AC97_MIC, 5, 4, wm9711_mic),
+AC97_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, wm9711_rec_sel),
+AC97_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9711_ng_type),
+};
+
+static const struct snd_kcontrol_new wm9711_snd_ac97_controls[] = {
+AC97_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+AC97_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+AC97_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
+AC97_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+AC97_ENUM("ALC Function", wm9711_enum[0]),
+AC97_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 1),
+AC97_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
+AC97_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+AC97_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+AC97_ENUM("ALC NG Type", wm9711_enum[9]),
+AC97_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
+
+AC97_SINGLE("Side Tone Switch", AC97_VIDEO, 15, 1, 1),
+AC97_SINGLE("Side Tone Volume", AC97_VIDEO, 12, 7, 1),
+AC97_ENUM("ALC Headphone Mux", wm9711_enum[1]),
+AC97_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
+
+AC97_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
+AC97_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 0),
+AC97_ENUM("Out3 Mux", wm9711_enum[2]),
+AC97_ENUM("Out3 LR Mux", wm9711_enum[3]),
+AC97_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
+
+AC97_SINGLE("Beep to Headphone Switch", AC97_PC_BEEP, 15, 1, 1),
+AC97_SINGLE("Beep to Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
+AC97_SINGLE("Beep to Side Tone Switch", AC97_PC_BEEP, 11, 1, 1),
+AC97_SINGLE("Beep to Side Tone Volume", AC97_PC_BEEP, 8, 7, 1),
+AC97_SINGLE("Beep to Phone Switch", AC97_PC_BEEP, 7, 1, 1),
+AC97_SINGLE("Beep to Phone Volume", AC97_PC_BEEP, 4, 7, 1),
+
+AC97_SINGLE("Aux to Headphone Switch", AC97_CD, 15, 1, 1),
+AC97_SINGLE("Aux to Headphone Volume", AC97_CD, 12, 7, 1),
+AC97_SINGLE("Aux to Side Tone Switch", AC97_CD, 11, 1, 1),
+AC97_SINGLE("Aux to Side Tone Volume", AC97_CD, 8, 7, 1),
+AC97_SINGLE("Aux to Phone Switch", AC97_CD, 7, 1, 1),
+AC97_SINGLE("Aux to Phone Volume", AC97_CD, 4, 7, 1),
+
+AC97_SINGLE("Phone to Headphone Switch", AC97_PHONE, 15, 1, 1),
+AC97_SINGLE("Phone to Master Switch", AC97_PHONE, 14, 1, 1),
+
+AC97_SINGLE("Line to Headphone Switch", AC97_LINE, 15, 1, 1),
+AC97_SINGLE("Line to Master Switch", AC97_LINE, 14, 1, 1),
+AC97_SINGLE("Line to Phone Switch", AC97_LINE, 13, 1, 1),
+
+AC97_SINGLE("PCM Playback to Headphone Switch", AC97_PCM, 15, 1, 1),
+AC97_SINGLE("PCM Playback to Master Switch", AC97_PCM, 14, 1, 1),
+AC97_SINGLE("PCM Playback to Phone Switch", AC97_PCM, 13, 1, 1),
+
+AC97_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
+AC97_ENUM("Capture to Phone Mux", wm9711_enum[4]),
+AC97_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
+AC97_ENUM("Capture Select", wm9711_enum[8]),
+
+AC97_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
+AC97_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
+
+AC97_ENUM("Bass Control", wm9711_enum[5]),
+AC97_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
+AC97_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
+AC97_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
+
+AC97_SINGLE("ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+AC97_ENUM("Capture Volume Steps", wm9711_enum[6]),
+AC97_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+AC97_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
+
+AC97_SINGLE("Mic 1 to Phone Switch", AC97_MIC, 14, 1, 1),
+AC97_SINGLE("Mic 2 to Phone Switch", AC97_MIC, 13, 1, 1),
+AC97_ENUM("Mic Select Source", wm9711_enum[7]),
+AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
+
+AC97_SINGLE("Master Left Inv Switch", AC97_MASTER, 6, 1, 0),
+AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0),
+AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0),
+AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
+};
+
+static int patch_wolfson_wm9711_specific(struct snd_ac97 * ac97)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm9711_snd_ac97_controls); i++) {
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm9711_snd_ac97_controls[i], ac97))) < 0)
+ return err;
+ }
+ snd_ac97_write_cache(ac97, AC97_CODEC_CLASS_REV, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_PCI_SVID, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_VIDEO, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_AUX, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_PC_BEEP, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_CD, 0x0000);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_wolfson_wm9711_ops = {
+ .build_specific = patch_wolfson_wm9711_specific,
+};
+
+static int patch_wolfson11(struct snd_ac97 * ac97)
+{
+ /* WM9711, WM9712 */
+ ac97->build_ops = &patch_wolfson_wm9711_ops;
+
+ ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_MIC |
+ AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD;
+
+ return 0;
+}
+
+static const char* wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
+static const char* wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
+static const char* wm9713_rec_src[] =
+ {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone Mix", "Master Mix",
+ "Mono Mix", "Zh"};
+static const char* wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char* wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char* wm9713_mono_pga[] = {"Vmid", "Zh", "Mono Mix", "Inv 1"};
+static const char* wm9713_spk_pga[] =
+ {"Vmid", "Zh", "Headphone Mix", "Master Mix", "Inv", "NC", "NC", "NC"};
+static const char* wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone Mix", "NC"};
+static const char* wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "NC"};
+static const char* wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "NC"};
+static const char* wm9713_dac_inv[] =
+ {"Off", "Mono Mix", "Master Mix", "Headphone Mix L", "Headphone Mix R",
+ "Headphone Mix Mono", "NC", "Vmid"};
+static const char* wm9713_base[] = {"Linear Control", "Adaptive Boost"};
+static const char* wm9713_ng_type[] = {"Constant Gain", "Mute"};
+
+static const struct ac97_enum wm9713_enum[] = {
+AC97_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer),
+AC97_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux),
+AC97_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux),
+AC97_ENUM_DOUBLE(AC97_VIDEO, 3, 0, 8, wm9713_rec_src),
+AC97_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain),
+AC97_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select),
+AC97_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga),
+AC97_ENUM_DOUBLE(AC97_REC_GAIN, 11, 8, 8, wm9713_spk_pga),
+AC97_ENUM_DOUBLE(AC97_REC_GAIN, 6, 4, 4, wm9713_hp_pga),
+AC97_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga),
+AC97_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga),
+AC97_ENUM_DOUBLE(AC97_REC_GAIN_MIC, 13, 10, 8, wm9713_dac_inv),
+AC97_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_base),
+AC97_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type),
+};
+
+static const struct snd_kcontrol_new wm13_snd_ac97_controls[] = {
+AC97_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1),
+AC97_SINGLE("Line In to Headphone Switch", AC97_PC_BEEP, 15, 1, 1),
+AC97_SINGLE("Line In to Master Switch", AC97_PC_BEEP, 14, 1, 1),
+AC97_SINGLE("Line In to Mono Switch", AC97_PC_BEEP, 13, 1, 1),
+
+AC97_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1),
+AC97_SINGLE("PCM Playback to Headphone Switch", AC97_PHONE, 15, 1, 1),
+AC97_SINGLE("PCM Playback to Master Switch", AC97_PHONE, 14, 1, 1),
+AC97_SINGLE("PCM Playback to Mono Switch", AC97_PHONE, 13, 1, 1),
+
+AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+AC97_SINGLE("Mic 1 to Mono Switch", AC97_LINE, 7, 1, 1),
+AC97_SINGLE("Mic 2 to Mono Switch", AC97_LINE, 6, 1, 1),
+AC97_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
+AC97_ENUM("Mic to Headphone Mux", wm9713_enum[0]),
+AC97_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
+
+AC97_SINGLE("Capture Switch", AC97_CD, 15, 1, 1),
+AC97_ENUM("Capture Volume Steps", wm9713_enum[4]),
+AC97_DOUBLE("Capture Volume", AC97_CD, 8, 0, 15, 0),
+AC97_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
+
+AC97_ENUM("Capture to Headphone Mux", wm9713_enum[1]),
+AC97_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1),
+AC97_ENUM("Capture to Mono Mux", wm9713_enum[2]),
+AC97_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
+AC97_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
+AC97_ENUM("Capture Select", wm9713_enum[3]),
+
+AC97_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+AC97_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+AC97_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0),
+AC97_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+AC97_ENUM("ALC Function", wm9713_enum[5]),
+AC97_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
+AC97_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0),
+AC97_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+AC97_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+AC97_ENUM("ALC NG Type", wm9713_enum[13]),
+AC97_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0),
+
+AC97_DOUBLE("Master ZC Switch", AC97_MASTER, 14, 6, 1, 0),
+AC97_DOUBLE("Headphone ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
+AC97_DOUBLE("Out3/4 ZC Switch", AC97_MASTER_MONO, 14, 6, 1, 0),
+AC97_SINGLE("Master Right Switch", AC97_MASTER, 7, 1, 1),
+AC97_SINGLE("Headphone Right Switch", AC97_HEADPHONE, 7, 1, 1),
+AC97_SINGLE("Out3/4 Right Switch", AC97_MASTER_MONO, 7, 1, 1),
+
+AC97_SINGLE("Mono In to Headphone Switch", AC97_MASTER_TONE, 15, 1, 1),
+AC97_SINGLE("Mono In to Master Switch", AC97_MASTER_TONE, 14, 1, 1),
+AC97_SINGLE("Mono In Volume", AC97_MASTER_TONE, 8, 31, 1),
+AC97_SINGLE("Mono Switch", AC97_MASTER_TONE, 7, 1, 1),
+AC97_SINGLE("Mono ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
+AC97_SINGLE("Mono Volume", AC97_MASTER_TONE, 0, 31, 1),
+
+AC97_SINGLE("PC Beep to Headphone Switch", AC97_AUX, 15, 1, 1),
+AC97_SINGLE("PC Beep to Headphone Volume", AC97_AUX, 12, 7, 1),
+AC97_SINGLE("PC Beep to Master Switch", AC97_AUX, 11, 1, 1),
+AC97_SINGLE("PC Beep to Master Volume", AC97_AUX, 8, 7, 1),
+AC97_SINGLE("PC Beep to Mono Switch", AC97_AUX, 7, 1, 1),
+AC97_SINGLE("PC Beep to Mono Volume", AC97_AUX, 4, 7, 1),
+
+AC97_SINGLE("Voice to Headphone Switch", AC97_PCM, 15, 1, 1),
+AC97_SINGLE("Voice to Headphone Volume", AC97_PCM, 12, 7, 1),
+AC97_SINGLE("Voice to Master Switch", AC97_PCM, 11, 1, 1),
+AC97_SINGLE("Voice to Master Volume", AC97_PCM, 8, 7, 1),
+AC97_SINGLE("Voice to Mono Switch", AC97_PCM, 7, 1, 1),
+AC97_SINGLE("Voice to Mono Volume", AC97_PCM, 4, 7, 1),
+
+AC97_SINGLE("Aux to Headphone Switch", AC97_REC_SEL, 15, 1, 1),
+AC97_SINGLE("Aux to Headphone Volume", AC97_REC_SEL, 12, 7, 1),
+AC97_SINGLE("Aux to Master Switch", AC97_REC_SEL, 11, 1, 1),
+AC97_SINGLE("Aux to Master Volume", AC97_REC_SEL, 8, 7, 1),
+AC97_SINGLE("Aux to Mono Switch", AC97_REC_SEL, 7, 1, 1),
+AC97_SINGLE("Aux to Mono Volume", AC97_REC_SEL, 4, 7, 1),
+
+AC97_ENUM("Mono Input Mux", wm9713_enum[6]),
+AC97_ENUM("Master Input Mux", wm9713_enum[7]),
+AC97_ENUM("Headphone Input Mux", wm9713_enum[8]),
+AC97_ENUM("Out 3 Input Mux", wm9713_enum[9]),
+AC97_ENUM("Out 4 Input Mux", wm9713_enum[10]),
+
+AC97_ENUM("Bass Control", wm9713_enum[12]),
+AC97_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
+AC97_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1),
+AC97_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0),
+AC97_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1),
+AC97_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1),
+};
+
+static const struct snd_kcontrol_new wm13_snd_ac97_controls_3d[] = {
+AC97_ENUM("Inv Input Mux", wm9713_enum[11]),
+AC97_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0),
+AC97_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
+AC97_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
+};
+
+static int patch_wolfson_wm9713_3d (struct snd_ac97 * ac97)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls_3d); i++) {
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls_3d[i], ac97))) < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int patch_wolfson_wm9713_specific(struct snd_ac97 * ac97)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm13_snd_ac97_controls); i++) {
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm13_snd_ac97_controls[i], ac97))) < 0)
+ return err;
+ }
+ snd_ac97_write_cache(ac97, AC97_PC_BEEP, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_PHONE, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_MIC, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_LINE, 0x00da);
+ snd_ac97_write_cache(ac97, AC97_CD, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_VIDEO, 0xd612);
+ snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x1ba0);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static void patch_wolfson_wm9713_suspend (struct snd_ac97 * ac97)
+{
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MID, 0xfeff);
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0xffff);
+}
+
+static void patch_wolfson_wm9713_resume (struct snd_ac97 * ac97)
+{
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MID, 0xda00);
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0x3810);
+ snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0x0);
+}
+#endif
+
+static struct snd_ac97_build_ops patch_wolfson_wm9713_ops = {
+ .build_specific = patch_wolfson_wm9713_specific,
+ .build_3d = patch_wolfson_wm9713_3d,
+#ifdef CONFIG_PM
+ .suspend = patch_wolfson_wm9713_suspend,
+ .resume = patch_wolfson_wm9713_resume
+#endif
+};
+
+static int patch_wolfson13(struct snd_ac97 * ac97)
+{
+ /* WM9713, WM9714 */
+ ac97->build_ops = &patch_wolfson_wm9713_ops;
+
+ ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_PHONE |
+ AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD | AC97_HAS_NO_TONE |
+ AC97_HAS_NO_STD_PCM;
+ ac97->scaps &= ~AC97_SCAP_MODEM;
+
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MID, 0xda00);
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0x3810);
+ snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0x0);
+
+ return 0;
+}
+
+/*
+ * Tritech codec
+ */
+static int patch_tritech_tr28028(struct snd_ac97 * ac97)
+{
+ snd_ac97_write_cache(ac97, 0x26, 0x0300);
+ snd_ac97_write_cache(ac97, 0x26, 0x0000);
+ snd_ac97_write_cache(ac97, AC97_SURROUND_MASTER, 0x0000);
+ snd_ac97_write_cache(ac97, AC97_SPDIF, 0x0000);
+ return 0;
+}
+
+/*
+ * Sigmatel STAC97xx codecs
+ */
+static int patch_sigmatel_stac9700_3d(struct snd_ac97 * ac97)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+
+ if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0)
+ return err;
+ strcpy(kctl->id.name, "3D Control Sigmatel - Depth");
+ kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 2, 3, 0);
+ snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
+ return 0;
+}
+
+static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+
+ if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0)
+ return err;
+ strcpy(kctl->id.name, "3D Control Sigmatel - Depth");
+ kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 0, 3, 0);
+ if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0)
+ return err;
+ strcpy(kctl->id.name, "3D Control Sigmatel - Rear Depth");
+ kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 2, 3, 0);
+ snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker =
+AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+
+static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert =
+AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
+
+static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = {
+AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0),
+AC97_SINGLE("Sigmatel ADC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 0, 1, 0)
+};
+
+static int patch_sigmatel_stac97xx_specific(struct snd_ac97 * ac97)
+{
+ int err;
+
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_ANALOG, snd_ac97_read(ac97, AC97_SIGMATEL_ANALOG) & ~0x0003);
+ if (snd_ac97_try_bit(ac97, AC97_SIGMATEL_ANALOG, 1))
+ if ((err = patch_build_controls(ac97, &snd_ac97_sigmatel_controls[0], 1)) < 0)
+ return err;
+ if (snd_ac97_try_bit(ac97, AC97_SIGMATEL_ANALOG, 0))
+ if ((err = patch_build_controls(ac97, &snd_ac97_sigmatel_controls[1], 1)) < 0)
+ return err;
+ if (snd_ac97_try_bit(ac97, AC97_SIGMATEL_DAC2INVERT, 2))
+ if ((err = patch_build_controls(ac97, &snd_ac97_sigmatel_4speaker, 1)) < 0)
+ return err;
+ if (snd_ac97_try_bit(ac97, AC97_SIGMATEL_DAC2INVERT, 3))
+ if ((err = patch_build_controls(ac97, &snd_ac97_sigmatel_phaseinvert, 1)) < 0)
+ return err;
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = {
+ .build_3d = patch_sigmatel_stac9700_3d,
+ .build_specific = patch_sigmatel_stac97xx_specific
+};
+
+static int patch_sigmatel_stac9700(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_sigmatel_stac9700_ops;
+ return 0;
+}
+
+static int snd_ac97_stac9708_put_bias(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int err;
+
+ mutex_lock(&ac97->page_mutex);
+ snd_ac97_write(ac97, AC97_SIGMATEL_BIAS1, 0xabba);
+ err = snd_ac97_update_bits(ac97, AC97_SIGMATEL_BIAS2, 0x0010,
+ (ucontrol->value.integer.value[0] & 1) << 4);
+ snd_ac97_write(ac97, AC97_SIGMATEL_BIAS1, 0);
+ mutex_unlock(&ac97->page_mutex);
+ return err;
+}
+
+static const struct snd_kcontrol_new snd_ac97_stac9708_bias_control = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Sigmatel Output Bias Switch",
+ .info = snd_ac97_info_volsw,
+ .get = snd_ac97_get_volsw,
+ .put = snd_ac97_stac9708_put_bias,
+ .private_value = AC97_SINGLE_VALUE(AC97_SIGMATEL_BIAS2, 4, 1, 0),
+};
+
+static int patch_sigmatel_stac9708_specific(struct snd_ac97 *ac97)
+{
+ int err;
+
+ /* the register bit is writable, but the function is not implemented: */
+ snd_ac97_remove_ctl(ac97, "PCM Out Path & Mute", NULL);
+
+ snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Sigmatel Surround Playback");
+ if ((err = patch_build_controls(ac97, &snd_ac97_stac9708_bias_control, 1)) < 0)
+ return err;
+ return patch_sigmatel_stac97xx_specific(ac97);
+}
+
+static struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = {
+ .build_3d = patch_sigmatel_stac9708_3d,
+ .build_specific = patch_sigmatel_stac9708_specific
+};
+
+static int patch_sigmatel_stac9708(struct snd_ac97 * ac97)
+{
+ unsigned int codec72, codec6c;
+
+ ac97->build_ops = &patch_sigmatel_stac9708_ops;
+ ac97->caps |= 0x10; /* HP (sigmatel surround) support */
+
+ codec72 = snd_ac97_read(ac97, AC97_SIGMATEL_BIAS2) & 0x8000;
+ codec6c = snd_ac97_read(ac97, AC97_SIGMATEL_ANALOG);
+
+ if ((codec72==0) && (codec6c==0)) {
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x1000);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS2, 0x0007);
+ } else if ((codec72==0x8000) && (codec6c==0)) {
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x1001);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_DAC2INVERT, 0x0008);
+ } else if ((codec72==0x8000) && (codec6c==0x0080)) {
+ /* nothing */
+ }
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_MULTICHN, 0x0000);
+ return 0;
+}
+
+static int patch_sigmatel_stac9721(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_sigmatel_stac9700_ops;
+ if (snd_ac97_read(ac97, AC97_SIGMATEL_ANALOG) == 0) {
+ // patch for SigmaTel
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x4000);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS2, 0x0002);
+ }
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_MULTICHN, 0x0000);
+ return 0;
+}
+
+static int patch_sigmatel_stac9744(struct snd_ac97 * ac97)
+{
+ // patch for SigmaTel
+ ac97->build_ops = &patch_sigmatel_stac9700_ops;
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x0000); /* is this correct? --jk */
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS2, 0x0002);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_MULTICHN, 0x0000);
+ return 0;
+}
+
+static int patch_sigmatel_stac9756(struct snd_ac97 * ac97)
+{
+ // patch for SigmaTel
+ ac97->build_ops = &patch_sigmatel_stac9700_ops;
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_CIC2, 0x0000); /* is this correct? --jk */
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS1, 0xabba);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_BIAS2, 0x0002);
+ snd_ac97_write_cache(ac97, AC97_SIGMATEL_MULTICHN, 0x0000);
+ return 0;
+}
+
+static int snd_ac97_stac9758_output_jack_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[5] = { "Input/Disabled", "Front Output",
+ "Rear Output", "Center/LFE Output", "Mixer Output" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 5;
+ if (uinfo->value.enumerated.item > 4)
+ uinfo->value.enumerated.item = 4;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_stac9758_output_jack_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value;
+ unsigned short val;
+
+ val = ac97->regs[AC97_SIGMATEL_OUTSEL] >> shift;
+ if (!(val & 4))
+ ucontrol->value.enumerated.item[0] = 0;
+ else
+ ucontrol->value.enumerated.item[0] = 1 + (val & 3);
+ return 0;
+}
+
+static int snd_ac97_stac9758_output_jack_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value;
+ unsigned short val;
+
+ if (ucontrol->value.enumerated.item[0] > 4)
+ return -EINVAL;
+ if (ucontrol->value.enumerated.item[0] == 0)
+ val = 0;
+ else
+ val = 4 | (ucontrol->value.enumerated.item[0] - 1);
+ return ac97_update_bits_page(ac97, AC97_SIGMATEL_OUTSEL,
+ 7 << shift, val << shift, 0);
+}
+
+static int snd_ac97_stac9758_input_jack_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[7] = { "Mic2 Jack", "Mic1 Jack", "Line In Jack",
+ "Front Jack", "Rear Jack", "Center/LFE Jack", "Mute" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 7;
+ if (uinfo->value.enumerated.item > 6)
+ uinfo->value.enumerated.item = 6;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_stac9758_input_jack_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value;
+ unsigned short val;
+
+ val = ac97->regs[AC97_SIGMATEL_INSEL];
+ ucontrol->value.enumerated.item[0] = (val >> shift) & 7;
+ return 0;
+}
+
+static int snd_ac97_stac9758_input_jack_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value;
+
+ return ac97_update_bits_page(ac97, AC97_SIGMATEL_INSEL, 7 << shift,
+ ucontrol->value.enumerated.item[0] << shift, 0);
+}
+
+static int snd_ac97_stac9758_phonesel_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = { "None", "Front Jack", "Rear Jack" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_stac9758_phonesel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = ac97->regs[AC97_SIGMATEL_IOMISC] & 3;
+ return 0;
+}
+
+static int snd_ac97_stac9758_phonesel_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ return ac97_update_bits_page(ac97, AC97_SIGMATEL_IOMISC, 3,
+ ucontrol->value.enumerated.item[0], 0);
+}
+
+#define STAC9758_OUTPUT_JACK(xname, shift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_ac97_stac9758_output_jack_info, \
+ .get = snd_ac97_stac9758_output_jack_get, \
+ .put = snd_ac97_stac9758_output_jack_put, \
+ .private_value = shift }
+#define STAC9758_INPUT_JACK(xname, shift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_ac97_stac9758_input_jack_info, \
+ .get = snd_ac97_stac9758_input_jack_get, \
+ .put = snd_ac97_stac9758_input_jack_put, \
+ .private_value = shift }
+static const struct snd_kcontrol_new snd_ac97_sigmatel_stac9758_controls[] = {
+ STAC9758_OUTPUT_JACK("Mic1 Jack", 1),
+ STAC9758_OUTPUT_JACK("LineIn Jack", 4),
+ STAC9758_OUTPUT_JACK("Front Jack", 7),
+ STAC9758_OUTPUT_JACK("Rear Jack", 10),
+ STAC9758_OUTPUT_JACK("Center/LFE Jack", 13),
+ STAC9758_INPUT_JACK("Mic Input Source", 0),
+ STAC9758_INPUT_JACK("Line Input Source", 8),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Amp",
+ .info = snd_ac97_stac9758_phonesel_info,
+ .get = snd_ac97_stac9758_phonesel_get,
+ .put = snd_ac97_stac9758_phonesel_put
+ },
+ AC97_SINGLE("Exchange Center/LFE", AC97_SIGMATEL_IOMISC, 4, 1, 0),
+ AC97_SINGLE("Headphone +3dB Boost", AC97_SIGMATEL_IOMISC, 8, 1, 0)
+};
+
+static int patch_sigmatel_stac9758_specific(struct snd_ac97 *ac97)
+{
+ int err;
+
+ err = patch_sigmatel_stac97xx_specific(ac97);
+ if (err < 0)
+ return err;
+ err = patch_build_controls(ac97, snd_ac97_sigmatel_stac9758_controls,
+ ARRAY_SIZE(snd_ac97_sigmatel_stac9758_controls));
+ if (err < 0)
+ return err;
+ /* DAC-A direct */
+ snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Front Playback");
+ /* DAC-A to Mix = PCM */
+ /* DAC-B direct = Surround */
+ /* DAC-B to Mix */
+ snd_ac97_rename_vol_ctl(ac97, "Video Playback", "Surround Mix Playback");
+ /* DAC-C direct = Center/LFE */
+
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = {
+ .build_3d = patch_sigmatel_stac9700_3d,
+ .build_specific = patch_sigmatel_stac9758_specific
+};
+
+static int patch_sigmatel_stac9758(struct snd_ac97 * ac97)
+{
+ static unsigned short regs[4] = {
+ AC97_SIGMATEL_OUTSEL,
+ AC97_SIGMATEL_IOMISC,
+ AC97_SIGMATEL_INSEL,
+ AC97_SIGMATEL_VARIOUS
+ };
+ static unsigned short def_regs[4] = {
+ /* OUTSEL */ 0xd794, /* CL:CL, SR:SR, LO:MX, LI:DS, MI:DS */
+ /* IOMISC */ 0x2001,
+ /* INSEL */ 0x0201, /* LI:LI, MI:M1 */
+ /* VARIOUS */ 0x0040
+ };
+ static unsigned short m675_regs[4] = {
+ /* OUTSEL */ 0xfc70, /* CL:MX, SR:MX, LO:DS, LI:MX, MI:DS */
+ /* IOMISC */ 0x2102, /* HP amp on */
+ /* INSEL */ 0x0203, /* LI:LI, MI:FR */
+ /* VARIOUS */ 0x0041 /* stereo mic */
+ };
+ unsigned short *pregs = def_regs;
+ int i;
+
+ /* Gateway M675 notebook */
+ if (ac97->pci &&
+ ac97->subsystem_vendor == 0x107b &&
+ ac97->subsystem_device == 0x0601)
+ pregs = m675_regs;
+
+ // patch for SigmaTel
+ ac97->build_ops = &patch_sigmatel_stac9758_ops;
+ /* FIXME: assume only page 0 for writing cache */
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, AC97_PAGE_VENDOR);
+ for (i = 0; i < 4; i++)
+ snd_ac97_write_cache(ac97, regs[i], pregs[i]);
+
+ ac97->flags |= AC97_STEREO_MUTES;
+ return 0;
+}
+
+/*
+ * Cirrus Logic CS42xx codecs
+ */
+static const struct snd_kcontrol_new snd_ac97_cirrus_controls_spdif[2] = {
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), AC97_CSR_SPDIF, 15, 1, 0),
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "AC97-SPSA", AC97_CSR_ACMODE, 0, 3, 0)
+};
+
+static int patch_cirrus_build_spdif(struct snd_ac97 * ac97)
+{
+ int err;
+
+ /* con mask, pro mask, default */
+ if ((err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3)) < 0)
+ return err;
+ /* switch, spsa */
+ if ((err = patch_build_controls(ac97, &snd_ac97_cirrus_controls_spdif[0], 1)) < 0)
+ return err;
+ switch (ac97->id & AC97_ID_CS_MASK) {
+ case AC97_ID_CS4205:
+ if ((err = patch_build_controls(ac97, &snd_ac97_cirrus_controls_spdif[1], 1)) < 0)
+ return err;
+ break;
+ }
+ /* set default PCM S/PDIF params */
+ /* consumer,PCM audio,no copyright,no preemphasis,PCM coder,original,48000Hz */
+ snd_ac97_write_cache(ac97, AC97_CSR_SPDIF, 0x0a20);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_cirrus_ops = {
+ .build_spdif = patch_cirrus_build_spdif
+};
+
+static int patch_cirrus_spdif(struct snd_ac97 * ac97)
+{
+ /* Basically, the cs4201/cs4205/cs4297a has non-standard sp/dif registers.
+ WHY CAN'T ANYONE FOLLOW THE BLOODY SPEC? *sigh*
+ - sp/dif EA ID is not set, but sp/dif is always present.
+ - enable/disable is spdif register bit 15.
+ - sp/dif control register is 0x68. differs from AC97:
+ - valid is bit 14 (vs 15)
+ - no DRS
+ - only 44.1/48k [00 = 48, 01=44,1] (AC97 is 00=44.1, 10=48)
+ - sp/dif ssource select is in 0x5e bits 0,1.
+ */
+
+ ac97->build_ops = &patch_cirrus_ops;
+ ac97->flags |= AC97_CS_SPDIF;
+ ac97->rates[AC97_RATES_SPDIF] &= ~SNDRV_PCM_RATE_32000;
+ ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
+ snd_ac97_write_cache(ac97, AC97_CSR_ACMODE, 0x0080);
+ return 0;
+}
+
+static int patch_cirrus_cs4299(struct snd_ac97 * ac97)
+{
+ /* force the detection of PC Beep */
+ ac97->flags |= AC97_HAS_PC_BEEP;
+
+ return patch_cirrus_spdif(ac97);
+}
+
+/*
+ * Conexant codecs
+ */
+static const struct snd_kcontrol_new snd_ac97_conexant_controls_spdif[1] = {
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), AC97_CXR_AUDIO_MISC, 3, 1, 0),
+};
+
+static int patch_conexant_build_spdif(struct snd_ac97 * ac97)
+{
+ int err;
+
+ /* con mask, pro mask, default */
+ if ((err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3)) < 0)
+ return err;
+ /* switch */
+ if ((err = patch_build_controls(ac97, &snd_ac97_conexant_controls_spdif[0], 1)) < 0)
+ return err;
+ /* set default PCM S/PDIF params */
+ /* consumer,PCM audio,no copyright,no preemphasis,PCM coder,original,48000Hz */
+ snd_ac97_write_cache(ac97, AC97_CXR_AUDIO_MISC,
+ snd_ac97_read(ac97, AC97_CXR_AUDIO_MISC) & ~(AC97_CXR_SPDIFEN|AC97_CXR_COPYRGT|AC97_CXR_SPDIF_MASK));
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_conexant_ops = {
+ .build_spdif = patch_conexant_build_spdif
+};
+
+static int patch_conexant(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_conexant_ops;
+ ac97->flags |= AC97_CX_SPDIF;
+ ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+ return 0;
+}
+
+static int patch_cx20551(struct snd_ac97 *ac97)
+{
+ snd_ac97_update_bits(ac97, 0x5c, 0x01, 0x01);
+ return 0;
+}
+
+/*
+ * Analog Device AD18xx, AD19xx codecs
+ */
+#ifdef CONFIG_PM
+static void ad18xx_resume(struct snd_ac97 *ac97)
+{
+ static unsigned short setup_regs[] = {
+ AC97_AD_MISC, AC97_AD_SERIAL_CFG, AC97_AD_JACK_SPDIF,
+ };
+ int i, codec;
+
+ for (i = 0; i < (int)ARRAY_SIZE(setup_regs); i++) {
+ unsigned short reg = setup_regs[i];
+ if (test_bit(reg, ac97->reg_accessed)) {
+ snd_ac97_write(ac97, reg, ac97->regs[reg]);
+ snd_ac97_read(ac97, reg);
+ }
+ }
+
+ if (! (ac97->flags & AC97_AD_MULTI))
+ /* normal restore */
+ snd_ac97_restore_status(ac97);
+ else {
+ /* restore the AD18xx codec configurations */
+ for (codec = 0; codec < 3; codec++) {
+ if (! ac97->spec.ad18xx.id[codec])
+ continue;
+ /* select single codec */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000,
+ ac97->spec.ad18xx.unchained[codec] | ac97->spec.ad18xx.chained[codec]);
+ ac97->bus->ops->write(ac97, AC97_AD_CODEC_CFG, ac97->spec.ad18xx.codec_cfg[codec]);
+ }
+ /* select all codecs */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000);
+
+ /* restore status */
+ for (i = 2; i < 0x7c ; i += 2) {
+ if (i == AC97_POWERDOWN || i == AC97_EXTENDED_ID)
+ continue;
+ if (test_bit(i, ac97->reg_accessed)) {
+ /* handle multi codecs for AD18xx */
+ if (i == AC97_PCM) {
+ for (codec = 0; codec < 3; codec++) {
+ if (! ac97->spec.ad18xx.id[codec])
+ continue;
+ /* select single codec */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000,
+ ac97->spec.ad18xx.unchained[codec] | ac97->spec.ad18xx.chained[codec]);
+ /* update PCM bits */
+ ac97->bus->ops->write(ac97, AC97_PCM, ac97->spec.ad18xx.pcmreg[codec]);
+ }
+ /* select all codecs */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000);
+ continue;
+ } else if (i == AC97_AD_TEST ||
+ i == AC97_AD_CODEC_CFG ||
+ i == AC97_AD_SERIAL_CFG)
+ continue; /* ignore */
+ }
+ snd_ac97_write(ac97, i, ac97->regs[i]);
+ snd_ac97_read(ac97, i);
+ }
+ }
+
+ snd_ac97_restore_iec958(ac97);
+}
+
+static void ad1888_resume(struct snd_ac97 *ac97)
+{
+ ad18xx_resume(ac97);
+ snd_ac97_write_cache(ac97, AC97_CODEC_CLASS_REV, 0x8080);
+}
+
+#endif
+
+static const struct snd_ac97_res_table ad1819_restbl[] = {
+ { AC97_PHONE, 0x9f1f },
+ { AC97_MIC, 0x9f1f },
+ { AC97_LINE, 0x9f1f },
+ { AC97_CD, 0x9f1f },
+ { AC97_VIDEO, 0x9f1f },
+ { AC97_AUX, 0x9f1f },
+ { AC97_PCM, 0x9f1f },
+ { } /* terminator */
+};
+
+static int patch_ad1819(struct snd_ac97 * ac97)
+{
+ unsigned short scfg;
+
+ // patch for Analog Devices
+ scfg = snd_ac97_read(ac97, AC97_AD_SERIAL_CFG);
+ snd_ac97_write_cache(ac97, AC97_AD_SERIAL_CFG, scfg | 0x7000); /* select all codecs */
+ ac97->res_table = ad1819_restbl;
+ return 0;
+}
+
+static unsigned short patch_ad1881_unchained(struct snd_ac97 * ac97, int idx, unsigned short mask)
+{
+ unsigned short val;
+
+ // test for unchained codec
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, mask);
+ snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0000); /* ID0C, ID1C, SDIE = off */
+ val = snd_ac97_read(ac97, AC97_VENDOR_ID2);
+ if ((val & 0xff40) != 0x5340)
+ return 0;
+ ac97->spec.ad18xx.unchained[idx] = mask;
+ ac97->spec.ad18xx.id[idx] = val;
+ ac97->spec.ad18xx.codec_cfg[idx] = 0x0000;
+ return mask;
+}
+
+static int patch_ad1881_chained1(struct snd_ac97 * ac97, int idx, unsigned short codec_bits)
+{
+ static int cfg_bits[3] = { 1<<12, 1<<14, 1<<13 };
+ unsigned short val;
+
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, cfg_bits[idx]);
+ snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0004); // SDIE
+ val = snd_ac97_read(ac97, AC97_VENDOR_ID2);
+ if ((val & 0xff40) != 0x5340)
+ return 0;
+ if (codec_bits)
+ snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, codec_bits);
+ ac97->spec.ad18xx.chained[idx] = cfg_bits[idx];
+ ac97->spec.ad18xx.id[idx] = val;
+ ac97->spec.ad18xx.codec_cfg[idx] = codec_bits ? codec_bits : 0x0004;
+ return 1;
+}
+
+static void patch_ad1881_chained(struct snd_ac97 * ac97, int unchained_idx, int cidx1, int cidx2)
+{
+ // already detected?
+ if (ac97->spec.ad18xx.unchained[cidx1] || ac97->spec.ad18xx.chained[cidx1])
+ cidx1 = -1;
+ if (ac97->spec.ad18xx.unchained[cidx2] || ac97->spec.ad18xx.chained[cidx2])
+ cidx2 = -1;
+ if (cidx1 < 0 && cidx2 < 0)
+ return;
+ // test for chained codecs
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000,
+ ac97->spec.ad18xx.unchained[unchained_idx]);
+ snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0002); // ID1C
+ ac97->spec.ad18xx.codec_cfg[unchained_idx] = 0x0002;
+ if (cidx1 >= 0) {
+ if (cidx2 < 0)
+ patch_ad1881_chained1(ac97, cidx1, 0);
+ else if (patch_ad1881_chained1(ac97, cidx1, 0x0006)) // SDIE | ID1C
+ patch_ad1881_chained1(ac97, cidx2, 0);
+ else if (patch_ad1881_chained1(ac97, cidx2, 0x0006)) // SDIE | ID1C
+ patch_ad1881_chained1(ac97, cidx1, 0);
+ } else if (cidx2 >= 0) {
+ patch_ad1881_chained1(ac97, cidx2, 0);
+ }
+}
+
+static struct snd_ac97_build_ops patch_ad1881_build_ops = {
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume
+#endif
+};
+
+static int patch_ad1881(struct snd_ac97 * ac97)
+{
+ static const char cfg_idxs[3][2] = {
+ {2, 1},
+ {0, 2},
+ {0, 1}
+ };
+
+ // patch for Analog Devices
+ unsigned short codecs[3];
+ unsigned short val;
+ int idx, num;
+
+ val = snd_ac97_read(ac97, AC97_AD_SERIAL_CFG);
+ snd_ac97_write_cache(ac97, AC97_AD_SERIAL_CFG, val);
+ codecs[0] = patch_ad1881_unchained(ac97, 0, (1<<12));
+ codecs[1] = patch_ad1881_unchained(ac97, 1, (1<<14));
+ codecs[2] = patch_ad1881_unchained(ac97, 2, (1<<13));
+
+ if (! (codecs[0] || codecs[1] || codecs[2]))
+ goto __end;
+
+ for (idx = 0; idx < 3; idx++)
+ if (ac97->spec.ad18xx.unchained[idx])
+ patch_ad1881_chained(ac97, idx, cfg_idxs[idx][0], cfg_idxs[idx][1]);
+
+ if (ac97->spec.ad18xx.id[1]) {
+ ac97->flags |= AC97_AD_MULTI;
+ ac97->scaps |= AC97_SCAP_SURROUND_DAC;
+ }
+ if (ac97->spec.ad18xx.id[2]) {
+ ac97->flags |= AC97_AD_MULTI;
+ ac97->scaps |= AC97_SCAP_CENTER_LFE_DAC;
+ }
+
+ __end:
+ /* select all codecs */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000);
+ /* check if only one codec is present */
+ for (idx = num = 0; idx < 3; idx++)
+ if (ac97->spec.ad18xx.id[idx])
+ num++;
+ if (num == 1) {
+ /* ok, deselect all ID bits */
+ snd_ac97_write_cache(ac97, AC97_AD_CODEC_CFG, 0x0000);
+ ac97->spec.ad18xx.codec_cfg[0] =
+ ac97->spec.ad18xx.codec_cfg[1] =
+ ac97->spec.ad18xx.codec_cfg[2] = 0x0000;
+ }
+ /* required for AD1886/AD1885 combination */
+ ac97->ext_id = snd_ac97_read(ac97, AC97_EXTENDED_ID);
+ if (ac97->spec.ad18xx.id[0]) {
+ ac97->id &= 0xffff0000;
+ ac97->id |= ac97->spec.ad18xx.id[0];
+ }
+ ac97->build_ops = &patch_ad1881_build_ops;
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_ad1885[] = {
+ AC97_SINGLE("Digital Mono Direct", AC97_AD_MISC, 11, 1, 0),
+ /* AC97_SINGLE("Digital Audio Mode", AC97_AD_MISC, 12, 1, 0), */ /* seems problematic */
+ AC97_SINGLE("Low Power Mixer", AC97_AD_MISC, 14, 1, 0),
+ AC97_SINGLE("Zero Fill DAC", AC97_AD_MISC, 15, 1, 0),
+ AC97_SINGLE("Headphone Jack Sense", AC97_AD_JACK_SPDIF, 9, 1, 1), /* inverted */
+ AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 8, 1, 1), /* inverted */
+};
+
+static const DECLARE_TLV_DB_SCALE(db_scale_6bit_6db_max, -8850, 150, 0);
+
+static int patch_ad1885_specific(struct snd_ac97 * ac97)
+{
+ int err;
+
+ if ((err = patch_build_controls(ac97, snd_ac97_controls_ad1885, ARRAY_SIZE(snd_ac97_controls_ad1885))) < 0)
+ return err;
+ reset_tlv(ac97, "Headphone Playback Volume",
+ db_scale_6bit_6db_max);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_ad1885_build_ops = {
+ .build_specific = &patch_ad1885_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume
+#endif
+};
+
+static int patch_ad1885(struct snd_ac97 * ac97)
+{
+ patch_ad1881(ac97);
+ /* This is required to deal with the Intel D815EEAL2 */
+ /* i.e. Line out is actually headphone out from codec */
+
+ /* set default */
+ snd_ac97_write_cache(ac97, AC97_AD_MISC, 0x0404);
+
+ ac97->build_ops = &patch_ad1885_build_ops;
+ return 0;
+}
+
+static int patch_ad1886_specific(struct snd_ac97 * ac97)
+{
+ reset_tlv(ac97, "Headphone Playback Volume",
+ db_scale_6bit_6db_max);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_ad1886_build_ops = {
+ .build_specific = &patch_ad1886_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume
+#endif
+};
+
+static int patch_ad1886(struct snd_ac97 * ac97)
+{
+ patch_ad1881(ac97);
+ /* Presario700 workaround */
+ /* for Jack Sense/SPDIF Register misetting causing */
+ snd_ac97_write_cache(ac97, AC97_AD_JACK_SPDIF, 0x0010);
+ ac97->build_ops = &patch_ad1886_build_ops;
+ return 0;
+}
+
+/* MISC bits (AD1888/AD1980/AD1985 register 0x76) */
+#define AC97_AD198X_MBC 0x0003 /* mic boost */
+#define AC97_AD198X_MBC_20 0x0000 /* +20dB */
+#define AC97_AD198X_MBC_10 0x0001 /* +10dB */
+#define AC97_AD198X_MBC_30 0x0002 /* +30dB */
+#define AC97_AD198X_VREFD 0x0004 /* VREF high-Z */
+#define AC97_AD198X_VREFH 0x0008 /* 0=2.25V, 1=3.7V */
+#define AC97_AD198X_VREF_0 0x000c /* 0V (AD1985 only) */
+#define AC97_AD198X_VREF_MASK (AC97_AD198X_VREFH | AC97_AD198X_VREFD)
+#define AC97_AD198X_VREF_SHIFT 2
+#define AC97_AD198X_SRU 0x0010 /* sample rate unlock */
+#define AC97_AD198X_LOSEL 0x0020 /* LINE_OUT amplifiers input select */
+#define AC97_AD198X_2MIC 0x0040 /* 2-channel mic select */
+#define AC97_AD198X_SPRD 0x0080 /* SPREAD enable */
+#define AC97_AD198X_DMIX0 0x0100 /* downmix mode: */
+ /* 0 = 6-to-4, 1 = 6-to-2 downmix */
+#define AC97_AD198X_DMIX1 0x0200 /* downmix mode: 1 = enabled */
+#define AC97_AD198X_HPSEL 0x0400 /* headphone amplifier input select */
+#define AC97_AD198X_CLDIS 0x0800 /* center/lfe disable */
+#define AC97_AD198X_LODIS 0x1000 /* LINE_OUT disable */
+#define AC97_AD198X_MSPLT 0x2000 /* mute split */
+#define AC97_AD198X_AC97NC 0x4000 /* AC97 no compatible mode */
+#define AC97_AD198X_DACZ 0x8000 /* DAC zero-fill mode */
+
+/* MISC 1 bits (AD1986 register 0x76) */
+#define AC97_AD1986_MBC 0x0003 /* mic boost */
+#define AC97_AD1986_MBC_20 0x0000 /* +20dB */
+#define AC97_AD1986_MBC_10 0x0001 /* +10dB */
+#define AC97_AD1986_MBC_30 0x0002 /* +30dB */
+#define AC97_AD1986_LISEL0 0x0004 /* LINE_IN select bit 0 */
+#define AC97_AD1986_LISEL1 0x0008 /* LINE_IN select bit 1 */
+#define AC97_AD1986_LISEL_MASK (AC97_AD1986_LISEL1 | AC97_AD1986_LISEL0)
+#define AC97_AD1986_LISEL_LI 0x0000 /* LINE_IN pins as LINE_IN source */
+#define AC97_AD1986_LISEL_SURR 0x0004 /* SURROUND pins as LINE_IN source */
+#define AC97_AD1986_LISEL_MIC 0x0008 /* MIC_1/2 pins as LINE_IN source */
+#define AC97_AD1986_SRU 0x0010 /* sample rate unlock */
+#define AC97_AD1986_SOSEL 0x0020 /* SURROUND_OUT amplifiers input sel */
+#define AC97_AD1986_2MIC 0x0040 /* 2-channel mic select */
+#define AC97_AD1986_SPRD 0x0080 /* SPREAD enable */
+#define AC97_AD1986_DMIX0 0x0100 /* downmix mode: */
+ /* 0 = 6-to-4, 1 = 6-to-2 downmix */
+#define AC97_AD1986_DMIX1 0x0200 /* downmix mode: 1 = enabled */
+#define AC97_AD1986_CLDIS 0x0800 /* center/lfe disable */
+#define AC97_AD1986_SODIS 0x1000 /* SURROUND_OUT disable */
+#define AC97_AD1986_MSPLT 0x2000 /* mute split (read only 1) */
+#define AC97_AD1986_AC97NC 0x4000 /* AC97 no compatible mode (r/o 1) */
+#define AC97_AD1986_DACZ 0x8000 /* DAC zero-fill mode */
+
+/* MISC 2 bits (AD1986 register 0x70) */
+#define AC97_AD_MISC2 0x70 /* Misc Control Bits 2 (AD1986) */
+
+#define AC97_AD1986_CVREF0 0x0004 /* C/LFE VREF_OUT 2.25V */
+#define AC97_AD1986_CVREF1 0x0008 /* C/LFE VREF_OUT 0V */
+#define AC97_AD1986_CVREF2 0x0010 /* C/LFE VREF_OUT 3.7V */
+#define AC97_AD1986_CVREF_MASK \
+ (AC97_AD1986_CVREF2 | AC97_AD1986_CVREF1 | AC97_AD1986_CVREF0)
+#define AC97_AD1986_JSMAP 0x0020 /* Jack Sense Mapping 1 = alternate */
+#define AC97_AD1986_MMDIS 0x0080 /* Mono Mute Disable */
+#define AC97_AD1986_MVREF0 0x0400 /* MIC VREF_OUT 2.25V */
+#define AC97_AD1986_MVREF1 0x0800 /* MIC VREF_OUT 0V */
+#define AC97_AD1986_MVREF2 0x1000 /* MIC VREF_OUT 3.7V */
+#define AC97_AD1986_MVREF_MASK \
+ (AC97_AD1986_MVREF2 | AC97_AD1986_MVREF1 | AC97_AD1986_MVREF0)
+
+/* MISC 3 bits (AD1986 register 0x7a) */
+#define AC97_AD_MISC3 0x7a /* Misc Control Bits 3 (AD1986) */
+
+#define AC97_AD1986_MMIX 0x0004 /* Mic Mix, left/right */
+#define AC97_AD1986_GPO 0x0008 /* General Purpose Out */
+#define AC97_AD1986_LOHPEN 0x0010 /* LINE_OUT headphone drive */
+#define AC97_AD1986_LVREF0 0x0100 /* LINE_OUT VREF_OUT 2.25V */
+#define AC97_AD1986_LVREF1 0x0200 /* LINE_OUT VREF_OUT 0V */
+#define AC97_AD1986_LVREF2 0x0400 /* LINE_OUT VREF_OUT 3.7V */
+#define AC97_AD1986_LVREF_MASK \
+ (AC97_AD1986_LVREF2 | AC97_AD1986_LVREF1 | AC97_AD1986_LVREF0)
+#define AC97_AD1986_JSINVA 0x0800 /* Jack Sense Invert SENSE_A */
+#define AC97_AD1986_LOSEL 0x1000 /* LINE_OUT amplifiers input select */
+#define AC97_AD1986_HPSEL0 0x2000 /* Headphone amplifiers */
+ /* input select Surround DACs */
+#define AC97_AD1986_HPSEL1 0x4000 /* Headphone amplifiers input */
+ /* select C/LFE DACs */
+#define AC97_AD1986_JSINVB 0x8000 /* Jack Sense Invert SENSE_B */
+
+/* Serial Config bits (AD1986 register 0x74) (incomplete) */
+#define AC97_AD1986_OMS0 0x0100 /* Optional Mic Selector bit 0 */
+#define AC97_AD1986_OMS1 0x0200 /* Optional Mic Selector bit 1 */
+#define AC97_AD1986_OMS2 0x0400 /* Optional Mic Selector bit 2 */
+#define AC97_AD1986_OMS_MASK \
+ (AC97_AD1986_OMS2 | AC97_AD1986_OMS1 | AC97_AD1986_OMS0)
+#define AC97_AD1986_OMS_M 0x0000 /* MIC_1/2 pins are MIC sources */
+#define AC97_AD1986_OMS_L 0x0100 /* LINE_IN pins are MIC sources */
+#define AC97_AD1986_OMS_C 0x0200 /* Center/LFE pins are MCI sources */
+#define AC97_AD1986_OMS_MC 0x0400 /* Mix of MIC and C/LFE pins */
+ /* are MIC sources */
+#define AC97_AD1986_OMS_ML 0x0500 /* MIX of MIC and LINE_IN pins */
+ /* are MIC sources */
+#define AC97_AD1986_OMS_LC 0x0600 /* MIX of LINE_IN and C/LFE pins */
+ /* are MIC sources */
+#define AC97_AD1986_OMS_MLC 0x0700 /* MIX of MIC, LINE_IN, C/LFE pins */
+ /* are MIC sources */
+
+
+static int snd_ac97_ad198x_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "AC-Link", "A/D Converter" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_ad198x_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_AD_SERIAL_CFG];
+ ucontrol->value.enumerated.item[0] = (val >> 2) & 1;
+ return 0;
+}
+
+static int snd_ac97_ad198x_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ val = ucontrol->value.enumerated.item[0] << 2;
+ return snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x0004, val);
+}
+
+static const struct snd_kcontrol_new snd_ac97_ad198x_spdif_source = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = snd_ac97_ad198x_spdif_source_info,
+ .get = snd_ac97_ad198x_spdif_source_get,
+ .put = snd_ac97_ad198x_spdif_source_put,
+};
+
+static int patch_ad198x_post_spdif(struct snd_ac97 * ac97)
+{
+ return patch_build_controls(ac97, &snd_ac97_ad198x_spdif_source, 1);
+}
+
+static const struct snd_kcontrol_new snd_ac97_ad1981x_jack_sense[] = {
+ AC97_SINGLE("Headphone Jack Sense", AC97_AD_JACK_SPDIF, 11, 1, 0),
+ AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0),
+};
+
+/* black list to avoid HP/Line jack-sense controls
+ * (SS vendor << 16 | device)
+ */
+static unsigned int ad1981_jacks_blacklist[] = {
+ 0x10140523, /* Thinkpad R40 */
+ 0x10140534, /* Thinkpad X31 */
+ 0x10140537, /* Thinkpad T41p */
+ 0x10140554, /* Thinkpad T42p/R50p */
+ 0x10140567, /* Thinkpad T43p 2668-G7U */
+ 0x10140581, /* Thinkpad X41-2527 */
+ 0x104380b0, /* Asus A7V8X-MX */
+ 0x11790241, /* Toshiba Satellite A-15 S127 */
+ 0x144dc01a, /* Samsung NP-X20C004/SEG */
+ 0 /* end */
+};
+
+static int check_list(struct snd_ac97 *ac97, const unsigned int *list)
+{
+ u32 subid = ((u32)ac97->subsystem_vendor << 16) | ac97->subsystem_device;
+ for (; *list; list++)
+ if (*list == subid)
+ return 1;
+ return 0;
+}
+
+static int patch_ad1981a_specific(struct snd_ac97 * ac97)
+{
+ if (check_list(ac97, ad1981_jacks_blacklist))
+ return 0;
+ return patch_build_controls(ac97, snd_ac97_ad1981x_jack_sense,
+ ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
+}
+
+static struct snd_ac97_build_ops patch_ad1981a_build_ops = {
+ .build_post_spdif = patch_ad198x_post_spdif,
+ .build_specific = patch_ad1981a_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume
+#endif
+};
+
+/* white list to enable HP jack-sense bits
+ * (SS vendor << 16 | device)
+ */
+static unsigned int ad1981_jacks_whitelist[] = {
+ 0x0e11005a, /* HP nc4000/4010 */
+ 0x103c0890, /* HP nc6000 */
+ 0x103c0938, /* HP nc4220 */
+ 0x103c099c, /* HP nx6110 */
+ 0x103c0944, /* HP nc6220 */
+ 0x103c0934, /* HP nc8220 */
+ 0x103c006d, /* HP nx9105 */
+ 0x17340088, /* FSC Scenic-W */
+ 0 /* end */
+};
+
+static void check_ad1981_hp_jack_sense(struct snd_ac97 *ac97)
+{
+ if (check_list(ac97, ad1981_jacks_whitelist))
+ /* enable headphone jack sense */
+ snd_ac97_update_bits(ac97, AC97_AD_JACK_SPDIF, 1<<11, 1<<11);
+}
+
+static int patch_ad1981a(struct snd_ac97 *ac97)
+{
+ patch_ad1881(ac97);
+ ac97->build_ops = &patch_ad1981a_build_ops;
+ snd_ac97_update_bits(ac97, AC97_AD_MISC, AC97_AD198X_MSPLT, AC97_AD198X_MSPLT);
+ ac97->flags |= AC97_STEREO_MUTES;
+ check_ad1981_hp_jack_sense(ac97);
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_ad198x_2cmic =
+AC97_SINGLE("Stereo Mic", AC97_AD_MISC, 6, 1, 0);
+
+static int patch_ad1981b_specific(struct snd_ac97 *ac97)
+{
+ int err;
+
+ if ((err = patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1)) < 0)
+ return err;
+ if (check_list(ac97, ad1981_jacks_blacklist))
+ return 0;
+ return patch_build_controls(ac97, snd_ac97_ad1981x_jack_sense,
+ ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
+}
+
+static struct snd_ac97_build_ops patch_ad1981b_build_ops = {
+ .build_post_spdif = patch_ad198x_post_spdif,
+ .build_specific = patch_ad1981b_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume
+#endif
+};
+
+static int patch_ad1981b(struct snd_ac97 *ac97)
+{
+ patch_ad1881(ac97);
+ ac97->build_ops = &patch_ad1981b_build_ops;
+ snd_ac97_update_bits(ac97, AC97_AD_MISC, AC97_AD198X_MSPLT, AC97_AD198X_MSPLT);
+ ac97->flags |= AC97_STEREO_MUTES;
+ check_ad1981_hp_jack_sense(ac97);
+ return 0;
+}
+
+#define snd_ac97_ad1888_lohpsel_info snd_ctl_boolean_mono_info
+
+static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_AD_MISC];
+ ucontrol->value.integer.value[0] = !(val & AC97_AD198X_LOSEL);
+ if (ac97->spec.ad18xx.lo_as_master)
+ ucontrol->value.integer.value[0] =
+ !ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = !ucontrol->value.integer.value[0];
+ if (ac97->spec.ad18xx.lo_as_master)
+ val = !val;
+ val = val ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
+ return snd_ac97_update_bits(ac97, AC97_AD_MISC,
+ AC97_AD198X_LOSEL | AC97_AD198X_HPSEL, val);
+}
+
+static int snd_ac97_ad1888_downmix_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {"Off", "6 -> 4", "6 -> 2"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_ad1888_downmix_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_AD_MISC];
+ if (!(val & AC97_AD198X_DMIX1))
+ ucontrol->value.enumerated.item[0] = 0;
+ else
+ ucontrol->value.enumerated.item[0] = 1 + ((val >> 8) & 1);
+ return 0;
+}
+
+static int snd_ac97_ad1888_downmix_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ if (ucontrol->value.enumerated.item[0] > 2)
+ return -EINVAL;
+ if (ucontrol->value.enumerated.item[0] == 0)
+ val = 0;
+ else
+ val = AC97_AD198X_DMIX1 |
+ ((ucontrol->value.enumerated.item[0] - 1) << 8);
+ return snd_ac97_update_bits(ac97, AC97_AD_MISC,
+ AC97_AD198X_DMIX0 | AC97_AD198X_DMIX1, val);
+}
+
+static void ad1888_update_jacks(struct snd_ac97 *ac97)
+{
+ unsigned short val = 0;
+ /* clear LODIS if shared jack is to be used for Surround out */
+ if (!ac97->spec.ad18xx.lo_as_master && is_shared_linein(ac97))
+ val |= (1 << 12);
+ /* clear CLDIS if shared jack is to be used for C/LFE out */
+ if (is_shared_micin(ac97))
+ val |= (1 << 11);
+ /* shared Line-In */
+ snd_ac97_update_bits(ac97, AC97_AD_MISC, (1 << 11) | (1 << 12), val);
+}
+
+static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Exchange Front/Surround",
+ .info = snd_ac97_ad1888_lohpsel_info,
+ .get = snd_ac97_ad1888_lohpsel_get,
+ .put = snd_ac97_ad1888_lohpsel_put
+ },
+ AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, 2, 1, 1),
+ AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, 12, 1, 1),
+ AC97_SINGLE("Spread Front to Surround and Center/LFE", AC97_AD_MISC, 7, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Downmix",
+ .info = snd_ac97_ad1888_downmix_info,
+ .get = snd_ac97_ad1888_downmix_get,
+ .put = snd_ac97_ad1888_downmix_put
+ },
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+
+ AC97_SINGLE("Headphone Jack Sense", AC97_AD_JACK_SPDIF, 10, 1, 0),
+ AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0),
+};
+
+static int patch_ad1888_specific(struct snd_ac97 *ac97)
+{
+ if (!ac97->spec.ad18xx.lo_as_master) {
+ /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
+ snd_ac97_rename_vol_ctl(ac97, "Master Playback",
+ "Master Surround Playback");
+ snd_ac97_rename_vol_ctl(ac97, "Headphone Playback",
+ "Master Playback");
+ }
+ return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls));
+}
+
+static struct snd_ac97_build_ops patch_ad1888_build_ops = {
+ .build_post_spdif = patch_ad198x_post_spdif,
+ .build_specific = patch_ad1888_specific,
+#ifdef CONFIG_PM
+ .resume = ad1888_resume,
+#endif
+ .update_jacks = ad1888_update_jacks,
+};
+
+static int patch_ad1888(struct snd_ac97 * ac97)
+{
+ unsigned short misc;
+
+ patch_ad1881(ac97);
+ ac97->build_ops = &patch_ad1888_build_ops;
+
+ /*
+ * LO can be used as a real line-out on some devices,
+ * and we need to revert the front/surround mixer switches
+ */
+ if (ac97->subsystem_vendor == 0x1043 &&
+ ac97->subsystem_device == 0x1193) /* ASUS A9T laptop */
+ ac97->spec.ad18xx.lo_as_master = 1;
+
+ misc = snd_ac97_read(ac97, AC97_AD_MISC);
+ /* AD-compatible mode */
+ /* Stereo mutes enabled */
+ misc |= AC97_AD198X_MSPLT | AC97_AD198X_AC97NC;
+ if (!ac97->spec.ad18xx.lo_as_master)
+ /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
+ /* it seems that most vendors connect line-out connector to
+ * headphone out of AC'97
+ */
+ misc |= AC97_AD198X_LOSEL | AC97_AD198X_HPSEL;
+
+ snd_ac97_write_cache(ac97, AC97_AD_MISC, misc);
+ ac97->flags |= AC97_STEREO_MUTES;
+ return 0;
+}
+
+static int patch_ad1980_specific(struct snd_ac97 *ac97)
+{
+ int err;
+
+ if ((err = patch_ad1888_specific(ac97)) < 0)
+ return err;
+ return patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1);
+}
+
+static struct snd_ac97_build_ops patch_ad1980_build_ops = {
+ .build_post_spdif = patch_ad198x_post_spdif,
+ .build_specific = patch_ad1980_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume,
+#endif
+ .update_jacks = ad1888_update_jacks,
+};
+
+static int patch_ad1980(struct snd_ac97 * ac97)
+{
+ patch_ad1888(ac97);
+ ac97->build_ops = &patch_ad1980_build_ops;
+ return 0;
+}
+
+static int snd_ac97_ad1985_vrefout_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {"High-Z", "3.7 V", "2.25 V", "0 V"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_ad1985_vrefout_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ static const int reg2ctrl[4] = {2, 0, 1, 3};
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+ val = (ac97->regs[AC97_AD_MISC] & AC97_AD198X_VREF_MASK)
+ >> AC97_AD198X_VREF_SHIFT;
+ ucontrol->value.enumerated.item[0] = reg2ctrl[val];
+ return 0;
+}
+
+static int snd_ac97_ad1985_vrefout_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ static const int ctrl2reg[4] = {1, 2, 0, 3};
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ if (ucontrol->value.enumerated.item[0] > 3)
+ return -EINVAL;
+ val = ctrl2reg[ucontrol->value.enumerated.item[0]]
+ << AC97_AD198X_VREF_SHIFT;
+ return snd_ac97_update_bits(ac97, AC97_AD_MISC,
+ AC97_AD198X_VREF_MASK, val);
+}
+
+static const struct snd_kcontrol_new snd_ac97_ad1985_controls[] = {
+ AC97_SINGLE("Exchange Center/LFE", AC97_AD_SERIAL_CFG, 3, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Exchange Front/Surround",
+ .info = snd_ac97_ad1888_lohpsel_info,
+ .get = snd_ac97_ad1888_lohpsel_get,
+ .put = snd_ac97_ad1888_lohpsel_put
+ },
+ AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, 12, 1, 1),
+ AC97_SINGLE("Spread Front to Surround and Center/LFE",
+ AC97_AD_MISC, 7, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Downmix",
+ .info = snd_ac97_ad1888_downmix_info,
+ .get = snd_ac97_ad1888_downmix_get,
+ .put = snd_ac97_ad1888_downmix_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "V_REFOUT",
+ .info = snd_ac97_ad1985_vrefout_info,
+ .get = snd_ac97_ad1985_vrefout_get,
+ .put = snd_ac97_ad1985_vrefout_put
+ },
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+
+ AC97_SINGLE("Headphone Jack Sense", AC97_AD_JACK_SPDIF, 10, 1, 0),
+ AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0),
+};
+
+static void ad1985_update_jacks(struct snd_ac97 *ac97)
+{
+ ad1888_update_jacks(ac97);
+ /* clear OMS if shared jack is to be used for C/LFE out */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 1 << 9,
+ is_shared_micin(ac97) ? 1 << 9 : 0);
+}
+
+static int patch_ad1985_specific(struct snd_ac97 *ac97)
+{
+ int err;
+
+ /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
+ snd_ac97_rename_vol_ctl(ac97, "Master Playback",
+ "Master Surround Playback");
+ snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback");
+
+ if ((err = patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1)) < 0)
+ return err;
+
+ return patch_build_controls(ac97, snd_ac97_ad1985_controls,
+ ARRAY_SIZE(snd_ac97_ad1985_controls));
+}
+
+static struct snd_ac97_build_ops patch_ad1985_build_ops = {
+ .build_post_spdif = patch_ad198x_post_spdif,
+ .build_specific = patch_ad1985_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume,
+#endif
+ .update_jacks = ad1985_update_jacks,
+};
+
+static int patch_ad1985(struct snd_ac97 * ac97)
+{
+ unsigned short misc;
+
+ patch_ad1881(ac97);
+ ac97->build_ops = &patch_ad1985_build_ops;
+ misc = snd_ac97_read(ac97, AC97_AD_MISC);
+ /* switch front/surround line-out/hp-out */
+ /* AD-compatible mode */
+ /* Stereo mutes enabled */
+ snd_ac97_write_cache(ac97, AC97_AD_MISC, misc |
+ AC97_AD198X_LOSEL |
+ AC97_AD198X_HPSEL |
+ AC97_AD198X_MSPLT |
+ AC97_AD198X_AC97NC);
+ ac97->flags |= AC97_STEREO_MUTES;
+
+ /* update current jack configuration */
+ ad1985_update_jacks(ac97);
+
+ /* on AD1985 rev. 3, AC'97 revision bits are zero */
+ ac97->ext_id = (ac97->ext_id & ~AC97_EI_REV_MASK) | AC97_EI_REV_23;
+ return 0;
+}
+
+#define snd_ac97_ad1986_bool_info snd_ctl_boolean_mono_info
+
+static int snd_ac97_ad1986_lososel_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_AD_MISC3];
+ ucontrol->value.integer.value[0] = (val & AC97_AD1986_LOSEL) != 0;
+ return 0;
+}
+
+static int snd_ac97_ad1986_lososel_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int ret0;
+ int ret1;
+ int sprd = (ac97->regs[AC97_AD_MISC] & AC97_AD1986_SPRD) != 0;
+
+ ret0 = snd_ac97_update_bits(ac97, AC97_AD_MISC3, AC97_AD1986_LOSEL,
+ ucontrol->value.integer.value[0] != 0
+ ? AC97_AD1986_LOSEL : 0);
+ if (ret0 < 0)
+ return ret0;
+
+ /* SOSEL is set to values of "Spread" or "Exchange F/S" controls */
+ ret1 = snd_ac97_update_bits(ac97, AC97_AD_MISC, AC97_AD1986_SOSEL,
+ (ucontrol->value.integer.value[0] != 0
+ || sprd)
+ ? AC97_AD1986_SOSEL : 0);
+ if (ret1 < 0)
+ return ret1;
+
+ return (ret0 > 0 || ret1 > 0) ? 1 : 0;
+}
+
+static int snd_ac97_ad1986_spread_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_AD_MISC];
+ ucontrol->value.integer.value[0] = (val & AC97_AD1986_SPRD) != 0;
+ return 0;
+}
+
+static int snd_ac97_ad1986_spread_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ int ret0;
+ int ret1;
+ int sprd = (ac97->regs[AC97_AD_MISC3] & AC97_AD1986_LOSEL) != 0;
+
+ ret0 = snd_ac97_update_bits(ac97, AC97_AD_MISC, AC97_AD1986_SPRD,
+ ucontrol->value.integer.value[0] != 0
+ ? AC97_AD1986_SPRD : 0);
+ if (ret0 < 0)
+ return ret0;
+
+ /* SOSEL is set to values of "Spread" or "Exchange F/S" controls */
+ ret1 = snd_ac97_update_bits(ac97, AC97_AD_MISC, AC97_AD1986_SOSEL,
+ (ucontrol->value.integer.value[0] != 0
+ || sprd)
+ ? AC97_AD1986_SOSEL : 0);
+ if (ret1 < 0)
+ return ret1;
+
+ return (ret0 > 0 || ret1 > 0) ? 1 : 0;
+}
+
+static int snd_ac97_ad1986_miclisel_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = ac97->spec.ad18xx.swap_mic_linein;
+ return 0;
+}
+
+static int snd_ac97_ad1986_miclisel_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned char swap = ucontrol->value.integer.value[0] != 0;
+
+ if (swap != ac97->spec.ad18xx.swap_mic_linein) {
+ ac97->spec.ad18xx.swap_mic_linein = swap;
+ if (ac97->build_ops->update_jacks)
+ ac97->build_ops->update_jacks(ac97);
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_ac97_ad1986_vrefout_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ /* Use MIC_1/2 V_REFOUT as the "get" value */
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+ unsigned short reg = ac97->regs[AC97_AD_MISC2];
+ if ((reg & AC97_AD1986_MVREF0) != 0)
+ val = 2;
+ else if ((reg & AC97_AD1986_MVREF1) != 0)
+ val = 3;
+ else if ((reg & AC97_AD1986_MVREF2) != 0)
+ val = 1;
+ else
+ val = 0;
+ ucontrol->value.enumerated.item[0] = val;
+ return 0;
+}
+
+static int snd_ac97_ad1986_vrefout_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short cval;
+ unsigned short lval;
+ unsigned short mval;
+ int cret;
+ int lret;
+ int mret;
+
+ switch (ucontrol->value.enumerated.item[0])
+ {
+ case 0: /* High-Z */
+ cval = 0;
+ lval = 0;
+ mval = 0;
+ break;
+ case 1: /* 3.7 V */
+ cval = AC97_AD1986_CVREF2;
+ lval = AC97_AD1986_LVREF2;
+ mval = AC97_AD1986_MVREF2;
+ break;
+ case 2: /* 2.25 V */
+ cval = AC97_AD1986_CVREF0;
+ lval = AC97_AD1986_LVREF0;
+ mval = AC97_AD1986_MVREF0;
+ break;
+ case 3: /* 0 V */
+ cval = AC97_AD1986_CVREF1;
+ lval = AC97_AD1986_LVREF1;
+ mval = AC97_AD1986_MVREF1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ cret = snd_ac97_update_bits(ac97, AC97_AD_MISC2,
+ AC97_AD1986_CVREF_MASK, cval);
+ if (cret < 0)
+ return cret;
+ lret = snd_ac97_update_bits(ac97, AC97_AD_MISC3,
+ AC97_AD1986_LVREF_MASK, lval);
+ if (lret < 0)
+ return lret;
+ mret = snd_ac97_update_bits(ac97, AC97_AD_MISC2,
+ AC97_AD1986_MVREF_MASK, mval);
+ if (mret < 0)
+ return mret;
+
+ return (cret > 0 || lret > 0 || mret > 0) ? 1 : 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_ad1986_controls[] = {
+ AC97_SINGLE("Exchange Center/LFE", AC97_AD_SERIAL_CFG, 3, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Exchange Front/Surround",
+ .info = snd_ac97_ad1986_bool_info,
+ .get = snd_ac97_ad1986_lososel_get,
+ .put = snd_ac97_ad1986_lososel_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Exchange Mic/Line In",
+ .info = snd_ac97_ad1986_bool_info,
+ .get = snd_ac97_ad1986_miclisel_get,
+ .put = snd_ac97_ad1986_miclisel_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Spread Front to Surround and Center/LFE",
+ .info = snd_ac97_ad1986_bool_info,
+ .get = snd_ac97_ad1986_spread_get,
+ .put = snd_ac97_ad1986_spread_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Downmix",
+ .info = snd_ac97_ad1888_downmix_info,
+ .get = snd_ac97_ad1888_downmix_get,
+ .put = snd_ac97_ad1888_downmix_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "V_REFOUT",
+ .info = snd_ac97_ad1985_vrefout_info,
+ .get = snd_ac97_ad1986_vrefout_get,
+ .put = snd_ac97_ad1986_vrefout_put
+ },
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+
+ AC97_SINGLE("Headphone Jack Sense", AC97_AD_JACK_SPDIF, 10, 1, 0),
+ AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0)
+};
+
+static void ad1986_update_jacks(struct snd_ac97 *ac97)
+{
+ unsigned short misc_val = 0;
+ unsigned short ser_val;
+
+ /* disable SURROUND and CENTER/LFE if not surround mode */
+ if (!is_surround_on(ac97))
+ misc_val |= AC97_AD1986_SODIS;
+ if (!is_clfe_on(ac97))
+ misc_val |= AC97_AD1986_CLDIS;
+
+ /* select line input (default=LINE_IN, SURROUND or MIC_1/2) */
+ if (is_shared_linein(ac97))
+ misc_val |= AC97_AD1986_LISEL_SURR;
+ else if (ac97->spec.ad18xx.swap_mic_linein != 0)
+ misc_val |= AC97_AD1986_LISEL_MIC;
+ snd_ac97_update_bits(ac97, AC97_AD_MISC,
+ AC97_AD1986_SODIS | AC97_AD1986_CLDIS |
+ AC97_AD1986_LISEL_MASK,
+ misc_val);
+
+ /* select microphone input (MIC_1/2, Center/LFE or LINE_IN) */
+ if (is_shared_micin(ac97))
+ ser_val = AC97_AD1986_OMS_C;
+ else if (ac97->spec.ad18xx.swap_mic_linein != 0)
+ ser_val = AC97_AD1986_OMS_L;
+ else
+ ser_val = AC97_AD1986_OMS_M;
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG,
+ AC97_AD1986_OMS_MASK,
+ ser_val);
+}
+
+static int patch_ad1986_specific(struct snd_ac97 *ac97)
+{
+ int err;
+
+ if ((err = patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1)) < 0)
+ return err;
+
+ return patch_build_controls(ac97, snd_ac97_ad1986_controls,
+ ARRAY_SIZE(snd_ac97_ad1985_controls));
+}
+
+static struct snd_ac97_build_ops patch_ad1986_build_ops = {
+ .build_post_spdif = patch_ad198x_post_spdif,
+ .build_specific = patch_ad1986_specific,
+#ifdef CONFIG_PM
+ .resume = ad18xx_resume,
+#endif
+ .update_jacks = ad1986_update_jacks,
+};
+
+static int patch_ad1986(struct snd_ac97 * ac97)
+{
+ patch_ad1881(ac97);
+ ac97->build_ops = &patch_ad1986_build_ops;
+ ac97->flags |= AC97_STEREO_MUTES;
+
+ /* update current jack configuration */
+ ad1986_update_jacks(ac97);
+
+ return 0;
+}
+
+/*
+ * realtek ALC203: use mono-out for pin 37
+ */
+static int patch_alc203(struct snd_ac97 *ac97)
+{
+ snd_ac97_update_bits(ac97, 0x7a, 0x400, 0x400);
+ return 0;
+}
+
+/*
+ * realtek ALC65x/850 codecs
+ */
+static void alc650_update_jacks(struct snd_ac97 *ac97)
+{
+ int shared;
+
+ /* shared Line-In / Surround Out */
+ shared = is_shared_surrout(ac97);
+ snd_ac97_update_bits(ac97, AC97_ALC650_MULTICH, 1 << 9,
+ shared ? (1 << 9) : 0);
+ /* update shared Mic In / Center/LFE Out */
+ shared = is_shared_clfeout(ac97);
+ /* disable/enable vref */
+ snd_ac97_update_bits(ac97, AC97_ALC650_CLOCK, 1 << 12,
+ shared ? (1 << 12) : 0);
+ /* turn on/off center-on-mic */
+ snd_ac97_update_bits(ac97, AC97_ALC650_MULTICH, 1 << 10,
+ shared ? (1 << 10) : 0);
+ /* GPIO0 high for mic */
+ snd_ac97_update_bits(ac97, AC97_ALC650_GPIO_STATUS, 0x100,
+ shared ? 0 : 0x100);
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_alc650[] = {
+ AC97_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0),
+ AC97_SINGLE("Surround Down Mix", AC97_ALC650_MULTICH, 1, 1, 0),
+ AC97_SINGLE("Center/LFE Down Mix", AC97_ALC650_MULTICH, 2, 1, 0),
+ AC97_SINGLE("Exchange Center/LFE", AC97_ALC650_MULTICH, 3, 1, 0),
+ /* 4: Analog Input To Surround */
+ /* 5: Analog Input To Center/LFE */
+ /* 6: Independent Master Volume Right */
+ /* 7: Independent Master Volume Left */
+ /* 8: reserved */
+ /* 9: Line-In/Surround share */
+ /* 10: Mic/CLFE share */
+ /* 11-13: in IEC958 controls */
+ AC97_SINGLE("Swap Surround Slot", AC97_ALC650_MULTICH, 14, 1, 0),
+#if 0 /* always set in patch_alc650 */
+ AC97_SINGLE("IEC958 Input Clock Enable", AC97_ALC650_CLOCK, 0, 1, 0),
+ AC97_SINGLE("IEC958 Input Pin Enable", AC97_ALC650_CLOCK, 1, 1, 0),
+ AC97_SINGLE("Surround DAC Switch", AC97_ALC650_SURR_DAC_VOL, 15, 1, 1),
+ AC97_DOUBLE("Surround DAC Volume", AC97_ALC650_SURR_DAC_VOL, 8, 0, 31, 1),
+ AC97_SINGLE("Center/LFE DAC Switch", AC97_ALC650_LFE_DAC_VOL, 15, 1, 1),
+ AC97_DOUBLE("Center/LFE DAC Volume", AC97_ALC650_LFE_DAC_VOL, 8, 0, 31, 1),
+#endif
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+};
+
+static const struct snd_kcontrol_new snd_ac97_spdif_controls_alc650[] = {
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_ALC650_MULTICH, 11, 1, 0),
+ AC97_SINGLE("Analog to IEC958 Output", AC97_ALC650_MULTICH, 12, 1, 0),
+ /* disable this controls since it doesn't work as expected */
+ /* AC97_SINGLE("IEC958 Input Monitor", AC97_ALC650_MULTICH, 13, 1, 0), */
+};
+
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_max, -4350, 150, 0);
+
+static int patch_alc650_specific(struct snd_ac97 * ac97)
+{
+ int err;
+
+ if ((err = patch_build_controls(ac97, snd_ac97_controls_alc650, ARRAY_SIZE(snd_ac97_controls_alc650))) < 0)
+ return err;
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ if ((err = patch_build_controls(ac97, snd_ac97_spdif_controls_alc650, ARRAY_SIZE(snd_ac97_spdif_controls_alc650))) < 0)
+ return err;
+ }
+ if (ac97->id != AC97_ID_ALC650F)
+ reset_tlv(ac97, "Master Playback Volume",
+ db_scale_5bit_3db_max);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_alc650_ops = {
+ .build_specific = patch_alc650_specific,
+ .update_jacks = alc650_update_jacks
+};
+
+static int patch_alc650(struct snd_ac97 * ac97)
+{
+ unsigned short val;
+
+ ac97->build_ops = &patch_alc650_ops;
+
+ /* determine the revision */
+ val = snd_ac97_read(ac97, AC97_ALC650_REVISION) & 0x3f;
+ if (val < 3)
+ ac97->id = 0x414c4720; /* Old version */
+ else if (val < 0x10)
+ ac97->id = 0x414c4721; /* D version */
+ else if (val < 0x20)
+ ac97->id = 0x414c4722; /* E version */
+ else if (val < 0x30)
+ ac97->id = 0x414c4723; /* F version */
+
+ /* revision E or F */
+ /* FIXME: what about revision D ? */
+ ac97->spec.dev_flags = (ac97->id == 0x414c4722 ||
+ ac97->id == 0x414c4723);
+
+ /* enable AC97_ALC650_GPIO_SETUP, AC97_ALC650_CLOCK for R/W */
+ snd_ac97_write_cache(ac97, AC97_ALC650_GPIO_STATUS,
+ snd_ac97_read(ac97, AC97_ALC650_GPIO_STATUS) | 0x8000);
+
+ /* Enable SPDIF-IN only on Rev.E and above */
+ val = snd_ac97_read(ac97, AC97_ALC650_CLOCK);
+ /* SPDIF IN with pin 47 */
+ if (ac97->spec.dev_flags &&
+ /* ASUS A6KM requires EAPD */
+ ! (ac97->subsystem_vendor == 0x1043 &&
+ ac97->subsystem_device == 0x1103))
+ val |= 0x03; /* enable */
+ else
+ val &= ~0x03; /* disable */
+ snd_ac97_write_cache(ac97, AC97_ALC650_CLOCK, val);
+
+ /* set default: slot 3,4,7,8,6,9
+ spdif-in monitor off, analog-spdif off, spdif-in off
+ center on mic off, surround on line-in off
+ downmix off, duplicate front off
+ */
+ snd_ac97_write_cache(ac97, AC97_ALC650_MULTICH, 0);
+
+ /* set GPIO0 for mic bias */
+ /* GPIO0 pin output, no interrupt, high */
+ snd_ac97_write_cache(ac97, AC97_ALC650_GPIO_SETUP,
+ snd_ac97_read(ac97, AC97_ALC650_GPIO_SETUP) | 0x01);
+ snd_ac97_write_cache(ac97, AC97_ALC650_GPIO_STATUS,
+ (snd_ac97_read(ac97, AC97_ALC650_GPIO_STATUS) | 0x100) & ~0x10);
+
+ /* full DAC volume */
+ snd_ac97_write_cache(ac97, AC97_ALC650_SURR_DAC_VOL, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_ALC650_LFE_DAC_VOL, 0x0808);
+ return 0;
+}
+
+static void alc655_update_jacks(struct snd_ac97 *ac97)
+{
+ int shared;
+
+ /* shared Line-In / Surround Out */
+ shared = is_shared_surrout(ac97);
+ ac97_update_bits_page(ac97, AC97_ALC650_MULTICH, 1 << 9,
+ shared ? (1 << 9) : 0, 0);
+ /* update shared Mic In / Center/LFE Out */
+ shared = is_shared_clfeout(ac97);
+ /* misc control; vrefout disable */
+ snd_ac97_update_bits(ac97, AC97_ALC650_CLOCK, 1 << 12,
+ shared ? (1 << 12) : 0);
+ ac97_update_bits_page(ac97, AC97_ALC650_MULTICH, 1 << 10,
+ shared ? (1 << 10) : 0, 0);
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_alc655[] = {
+ AC97_PAGE_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0, 0),
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+};
+
+static int alc655_iec958_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts_655[3] = { "PCM", "Analog In", "IEC958 In" };
+ static char *texts_658[4] = { "PCM", "Analog1 In", "Analog2 In", "IEC958 In" };
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = ac97->spec.dev_flags ? 4 : 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ ac97->spec.dev_flags ?
+ texts_658[uinfo->value.enumerated.item] :
+ texts_655[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int alc655_iec958_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_ALC650_MULTICH];
+ val = (val >> 12) & 3;
+ if (ac97->spec.dev_flags && val == 3)
+ val = 0;
+ ucontrol->value.enumerated.item[0] = val;
+ return 0;
+}
+
+static int alc655_iec958_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ return ac97_update_bits_page(ac97, AC97_ALC650_MULTICH, 3 << 12,
+ (unsigned short)ucontrol->value.enumerated.item[0] << 12,
+ 0);
+}
+
+static const struct snd_kcontrol_new snd_ac97_spdif_controls_alc655[] = {
+ AC97_PAGE_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_ALC650_MULTICH, 11, 1, 0, 0),
+ /* disable this controls since it doesn't work as expected */
+ /* AC97_PAGE_SINGLE("IEC958 Input Monitor", AC97_ALC650_MULTICH, 14, 1, 0, 0), */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = alc655_iec958_route_info,
+ .get = alc655_iec958_route_get,
+ .put = alc655_iec958_route_put,
+ },
+};
+
+static int patch_alc655_specific(struct snd_ac97 * ac97)
+{
+ int err;
+
+ if ((err = patch_build_controls(ac97, snd_ac97_controls_alc655, ARRAY_SIZE(snd_ac97_controls_alc655))) < 0)
+ return err;
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ if ((err = patch_build_controls(ac97, snd_ac97_spdif_controls_alc655, ARRAY_SIZE(snd_ac97_spdif_controls_alc655))) < 0)
+ return err;
+ }
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_alc655_ops = {
+ .build_specific = patch_alc655_specific,
+ .update_jacks = alc655_update_jacks
+};
+
+static int patch_alc655(struct snd_ac97 * ac97)
+{
+ unsigned int val;
+
+ if (ac97->id == AC97_ID_ALC658) {
+ ac97->spec.dev_flags = 1; /* ALC658 */
+ if ((snd_ac97_read(ac97, AC97_ALC650_REVISION) & 0x3f) == 2) {
+ ac97->id = AC97_ID_ALC658D;
+ ac97->spec.dev_flags = 2;
+ }
+ }
+
+ ac97->build_ops = &patch_alc655_ops;
+
+ /* assume only page 0 for writing cache */
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, AC97_PAGE_VENDOR);
+
+ /* adjust default values */
+ val = snd_ac97_read(ac97, 0x7a); /* misc control */
+ if (ac97->spec.dev_flags) /* ALC658 */
+ val &= ~(1 << 1); /* Pin 47 is spdif input pin */
+ else { /* ALC655 */
+ if (ac97->subsystem_vendor == 0x1462 &&
+ (ac97->subsystem_device == 0x0131 || /* MSI S270 laptop */
+ ac97->subsystem_device == 0x0161 || /* LG K1 Express */
+ ac97->subsystem_device == 0x0351 || /* MSI L725 laptop */
+ ac97->subsystem_device == 0x0471 || /* MSI L720 laptop */
+ ac97->subsystem_device == 0x0061)) /* MSI S250 laptop */
+ val &= ~(1 << 1); /* Pin 47 is EAPD (for internal speaker) */
+ else
+ val |= (1 << 1); /* Pin 47 is spdif input pin */
+ }
+ val &= ~(1 << 12); /* vref enable */
+ snd_ac97_write_cache(ac97, 0x7a, val);
+ /* set default: spdif-in enabled,
+ spdif-in monitor off, spdif-in PCM off
+ center on mic off, surround on line-in off
+ duplicate front off
+ */
+ snd_ac97_write_cache(ac97, AC97_ALC650_MULTICH, 1<<15);
+
+ /* full DAC volume */
+ snd_ac97_write_cache(ac97, AC97_ALC650_SURR_DAC_VOL, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_ALC650_LFE_DAC_VOL, 0x0808);
+
+ /* update undocumented bit... */
+ if (ac97->id == AC97_ID_ALC658D)
+ snd_ac97_update_bits(ac97, 0x74, 0x0800, 0x0800);
+
+ return 0;
+}
+
+
+#define AC97_ALC850_JACK_SELECT 0x76
+#define AC97_ALC850_MISC1 0x7a
+#define AC97_ALC850_MULTICH 0x6a
+
+static void alc850_update_jacks(struct snd_ac97 *ac97)
+{
+ int shared;
+ int aux_is_back_surround;
+
+ /* shared Line-In / Surround Out */
+ shared = is_shared_surrout(ac97);
+ /* SURR 1kOhm (bit4), Amp (bit5) */
+ snd_ac97_update_bits(ac97, AC97_ALC850_MISC1, (1<<4)|(1<<5),
+ shared ? (1<<5) : (1<<4));
+ /* LINE-IN = 0, SURROUND = 2 */
+ snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 12,
+ shared ? (2<<12) : (0<<12));
+ /* update shared Mic In / Center/LFE Out */
+ shared = is_shared_clfeout(ac97);
+ /* Vref disable (bit12), 1kOhm (bit13) */
+ snd_ac97_update_bits(ac97, AC97_ALC850_MISC1, (1<<12)|(1<<13),
+ shared ? (1<<12) : (1<<13));
+ /* MIC-IN = 1, CENTER-LFE = 5 */
+ snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 4,
+ shared ? (5<<4) : (1<<4));
+
+ aux_is_back_surround = alc850_is_aux_back_surround(ac97);
+ /* Aux is Back Surround */
+ snd_ac97_update_bits(ac97, AC97_ALC850_MULTICH, 1 << 10,
+ aux_is_back_surround ? (1<<10) : (0<<10));
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_alc850[] = {
+ AC97_PAGE_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0, 0),
+ AC97_SINGLE("Mic Front Input Switch", AC97_ALC850_JACK_SELECT, 15, 1, 1),
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_8CH_CTL,
+};
+
+static int patch_alc850_specific(struct snd_ac97 *ac97)
+{
+ int err;
+
+ if ((err = patch_build_controls(ac97, snd_ac97_controls_alc850, ARRAY_SIZE(snd_ac97_controls_alc850))) < 0)
+ return err;
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ if ((err = patch_build_controls(ac97, snd_ac97_spdif_controls_alc655, ARRAY_SIZE(snd_ac97_spdif_controls_alc655))) < 0)
+ return err;
+ }
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_alc850_ops = {
+ .build_specific = patch_alc850_specific,
+ .update_jacks = alc850_update_jacks
+};
+
+static int patch_alc850(struct snd_ac97 *ac97)
+{
+ ac97->build_ops = &patch_alc850_ops;
+
+ ac97->spec.dev_flags = 0; /* for IEC958 playback route - ALC655 compatible */
+ ac97->flags |= AC97_HAS_8CH;
+
+ /* assume only page 0 for writing cache */
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, AC97_PAGE_VENDOR);
+
+ /* adjust default values */
+ /* set default: spdif-in enabled,
+ spdif-in monitor off, spdif-in PCM off
+ center on mic off, surround on line-in off
+ duplicate front off
+ NB default bit 10=0 = Aux is Capture, not Back Surround
+ */
+ snd_ac97_write_cache(ac97, AC97_ALC650_MULTICH, 1<<15);
+ /* SURR_OUT: on, Surr 1kOhm: on, Surr Amp: off, Front 1kOhm: off
+ * Front Amp: on, Vref: enable, Center 1kOhm: on, Mix: on
+ */
+ snd_ac97_write_cache(ac97, 0x7a, (1<<1)|(1<<4)|(0<<5)|(1<<6)|
+ (1<<7)|(0<<12)|(1<<13)|(0<<14));
+ /* detection UIO2,3: all path floating, UIO3: MIC, Vref2: disable,
+ * UIO1: FRONT, Vref3: disable, UIO3: LINE, Front-Mic: mute
+ */
+ snd_ac97_write_cache(ac97, 0x76, (0<<0)|(0<<2)|(1<<4)|(1<<7)|(2<<8)|
+ (1<<11)|(0<<12)|(1<<15));
+
+ /* full DAC volume */
+ snd_ac97_write_cache(ac97, AC97_ALC650_SURR_DAC_VOL, 0x0808);
+ snd_ac97_write_cache(ac97, AC97_ALC650_LFE_DAC_VOL, 0x0808);
+ return 0;
+}
+
+
+/*
+ * C-Media CM97xx codecs
+ */
+static void cm9738_update_jacks(struct snd_ac97 *ac97)
+{
+ /* shared Line-In / Surround Out */
+ snd_ac97_update_bits(ac97, AC97_CM9738_VENDOR_CTRL, 1 << 10,
+ is_shared_surrout(ac97) ? (1 << 10) : 0);
+}
+
+static const struct snd_kcontrol_new snd_ac97_cm9738_controls[] = {
+ AC97_SINGLE("Duplicate Front", AC97_CM9738_VENDOR_CTRL, 13, 1, 0),
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_4CH_CTL,
+};
+
+static int patch_cm9738_specific(struct snd_ac97 * ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_cm9738_controls, ARRAY_SIZE(snd_ac97_cm9738_controls));
+}
+
+static struct snd_ac97_build_ops patch_cm9738_ops = {
+ .build_specific = patch_cm9738_specific,
+ .update_jacks = cm9738_update_jacks
+};
+
+static int patch_cm9738(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_cm9738_ops;
+ /* FIXME: can anyone confirm below? */
+ /* CM9738 has no PCM volume although the register reacts */
+ ac97->flags |= AC97_HAS_NO_PCM_VOL;
+ snd_ac97_write_cache(ac97, AC97_PCM, 0x8000);
+
+ return 0;
+}
+
+static int snd_ac97_cmedia_spdif_playback_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "Analog", "Digital" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ac97_cmedia_spdif_playback_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = ac97->regs[AC97_CM9739_SPDIF_CTRL];
+ ucontrol->value.enumerated.item[0] = (val >> 1) & 0x01;
+ return 0;
+}
+
+static int snd_ac97_cmedia_spdif_playback_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ return snd_ac97_update_bits(ac97, AC97_CM9739_SPDIF_CTRL,
+ 0x01 << 1,
+ (ucontrol->value.enumerated.item[0] & 0x01) << 1);
+}
+
+static const struct snd_kcontrol_new snd_ac97_cm9739_controls_spdif[] = {
+ /* BIT 0: SPDI_EN - always true */
+ { /* BIT 1: SPDIFS */
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = snd_ac97_cmedia_spdif_playback_source_info,
+ .get = snd_ac97_cmedia_spdif_playback_source_get,
+ .put = snd_ac97_cmedia_spdif_playback_source_put,
+ },
+ /* BIT 2: IG_SPIV */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Valid Switch", AC97_CM9739_SPDIF_CTRL, 2, 1, 0),
+ /* BIT 3: SPI2F */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Monitor", AC97_CM9739_SPDIF_CTRL, 3, 1, 0),
+ /* BIT 4: SPI2SDI */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_CM9739_SPDIF_CTRL, 4, 1, 0),
+ /* BIT 8: SPD32 - 32bit SPDIF - not supported yet */
+};
+
+static void cm9739_update_jacks(struct snd_ac97 *ac97)
+{
+ /* shared Line-In / Surround Out */
+ snd_ac97_update_bits(ac97, AC97_CM9739_MULTI_CHAN, 1 << 10,
+ is_shared_surrout(ac97) ? (1 << 10) : 0);
+ /* shared Mic In / Center/LFE Out **/
+ snd_ac97_update_bits(ac97, AC97_CM9739_MULTI_CHAN, 0x3000,
+ is_shared_clfeout(ac97) ? 0x1000 : 0x2000);
+}
+
+static const struct snd_kcontrol_new snd_ac97_cm9739_controls[] = {
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+};
+
+static int patch_cm9739_specific(struct snd_ac97 * ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_cm9739_controls, ARRAY_SIZE(snd_ac97_cm9739_controls));
+}
+
+static int patch_cm9739_post_spdif(struct snd_ac97 * ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_cm9739_controls_spdif, ARRAY_SIZE(snd_ac97_cm9739_controls_spdif));
+}
+
+static struct snd_ac97_build_ops patch_cm9739_ops = {
+ .build_specific = patch_cm9739_specific,
+ .build_post_spdif = patch_cm9739_post_spdif,
+ .update_jacks = cm9739_update_jacks
+};
+
+static int patch_cm9739(struct snd_ac97 * ac97)
+{
+ unsigned short val;
+
+ ac97->build_ops = &patch_cm9739_ops;
+
+ /* CM9739/A has no Master and PCM volume although the register reacts */
+ ac97->flags |= AC97_HAS_NO_MASTER_VOL | AC97_HAS_NO_PCM_VOL;
+ snd_ac97_write_cache(ac97, AC97_MASTER, 0x8000);
+ snd_ac97_write_cache(ac97, AC97_PCM, 0x8000);
+
+ /* check spdif */
+ val = snd_ac97_read(ac97, AC97_EXTENDED_STATUS);
+ if (val & AC97_EA_SPCV) {
+ /* enable spdif in */
+ snd_ac97_write_cache(ac97, AC97_CM9739_SPDIF_CTRL,
+ snd_ac97_read(ac97, AC97_CM9739_SPDIF_CTRL) | 0x01);
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+ } else {
+ ac97->ext_id &= ~AC97_EI_SPDIF; /* disable extended-id */
+ ac97->rates[AC97_RATES_SPDIF] = 0;
+ }
+
+ /* set-up multi channel */
+ /* bit 14: 0 = SPDIF, 1 = EAPD */
+ /* bit 13: enable internal vref output for mic */
+ /* bit 12: disable center/lfe (swithable) */
+ /* bit 10: disable surround/line (switchable) */
+ /* bit 9: mix 2 surround off */
+ /* bit 4: undocumented; 0 mutes the CM9739A, which defaults to 1 */
+ /* bit 3: undocumented; surround? */
+ /* bit 0: dB */
+ val = snd_ac97_read(ac97, AC97_CM9739_MULTI_CHAN) & (1 << 4);
+ val |= (1 << 3);
+ val |= (1 << 13);
+ if (! (ac97->ext_id & AC97_EI_SPDIF))
+ val |= (1 << 14);
+ snd_ac97_write_cache(ac97, AC97_CM9739_MULTI_CHAN, val);
+
+ /* FIXME: set up GPIO */
+ snd_ac97_write_cache(ac97, 0x70, 0x0100);
+ snd_ac97_write_cache(ac97, 0x72, 0x0020);
+ /* Special exception for ASUS W1000/CMI9739. It does not have an SPDIF in. */
+ if (ac97->pci &&
+ ac97->subsystem_vendor == 0x1043 &&
+ ac97->subsystem_device == 0x1843) {
+ snd_ac97_write_cache(ac97, AC97_CM9739_SPDIF_CTRL,
+ snd_ac97_read(ac97, AC97_CM9739_SPDIF_CTRL) & ~0x01);
+ snd_ac97_write_cache(ac97, AC97_CM9739_MULTI_CHAN,
+ snd_ac97_read(ac97, AC97_CM9739_MULTI_CHAN) | (1 << 14));
+ }
+
+ return 0;
+}
+
+#define AC97_CM9761_MULTI_CHAN 0x64
+#define AC97_CM9761_FUNC 0x66
+#define AC97_CM9761_SPDIF_CTRL 0x6c
+
+static void cm9761_update_jacks(struct snd_ac97 *ac97)
+{
+ /* FIXME: check the bits for each model
+ * model 83 is confirmed to work
+ */
+ static unsigned short surr_on[3][2] = {
+ { 0x0008, 0x0000 }, /* 9761-78 & 82 */
+ { 0x0000, 0x0008 }, /* 9761-82 rev.B */
+ { 0x0000, 0x0008 }, /* 9761-83 */
+ };
+ static unsigned short clfe_on[3][2] = {
+ { 0x0000, 0x1000 }, /* 9761-78 & 82 */
+ { 0x1000, 0x0000 }, /* 9761-82 rev.B */
+ { 0x0000, 0x1000 }, /* 9761-83 */
+ };
+ static unsigned short surr_shared[3][2] = {
+ { 0x0000, 0x0400 }, /* 9761-78 & 82 */
+ { 0x0000, 0x0400 }, /* 9761-82 rev.B */
+ { 0x0000, 0x0400 }, /* 9761-83 */
+ };
+ static unsigned short clfe_shared[3][2] = {
+ { 0x2000, 0x0880 }, /* 9761-78 & 82 */
+ { 0x0000, 0x2880 }, /* 9761-82 rev.B */
+ { 0x2000, 0x0800 }, /* 9761-83 */
+ };
+ unsigned short val = 0;
+
+ val |= surr_on[ac97->spec.dev_flags][is_surround_on(ac97)];
+ val |= clfe_on[ac97->spec.dev_flags][is_clfe_on(ac97)];
+ val |= surr_shared[ac97->spec.dev_flags][is_shared_surrout(ac97)];
+ val |= clfe_shared[ac97->spec.dev_flags][is_shared_clfeout(ac97)];
+
+ snd_ac97_update_bits(ac97, AC97_CM9761_MULTI_CHAN, 0x3c88, val);
+}
+
+static const struct snd_kcontrol_new snd_ac97_cm9761_controls[] = {
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+};
+
+static int cm9761_spdif_out_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "AC-Link", "ADC", "SPDIF-In" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int cm9761_spdif_out_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ if (ac97->regs[AC97_CM9761_FUNC] & 0x1)
+ ucontrol->value.enumerated.item[0] = 2; /* SPDIF-loopback */
+ else if (ac97->regs[AC97_CM9761_SPDIF_CTRL] & 0x2)
+ ucontrol->value.enumerated.item[0] = 1; /* ADC loopback */
+ else
+ ucontrol->value.enumerated.item[0] = 0; /* AC-link */
+ return 0;
+}
+
+static int cm9761_spdif_out_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.enumerated.item[0] == 2)
+ return snd_ac97_update_bits(ac97, AC97_CM9761_FUNC, 0x1, 0x1);
+ snd_ac97_update_bits(ac97, AC97_CM9761_FUNC, 0x1, 0);
+ return snd_ac97_update_bits(ac97, AC97_CM9761_SPDIF_CTRL, 0x2,
+ ucontrol->value.enumerated.item[0] == 1 ? 0x2 : 0);
+}
+
+static const char *cm9761_dac_clock[] = { "AC-Link", "SPDIF-In", "Both" };
+static const struct ac97_enum cm9761_dac_clock_enum =
+ AC97_ENUM_SINGLE(AC97_CM9761_SPDIF_CTRL, 9, 3, cm9761_dac_clock);
+
+static const struct snd_kcontrol_new snd_ac97_cm9761_controls_spdif[] = {
+ { /* BIT 1: SPDIFS */
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = cm9761_spdif_out_source_info,
+ .get = cm9761_spdif_out_source_get,
+ .put = cm9761_spdif_out_source_put,
+ },
+ /* BIT 2: IG_SPIV */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Valid Switch", AC97_CM9761_SPDIF_CTRL, 2, 1, 0),
+ /* BIT 3: SPI2F */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Monitor", AC97_CM9761_SPDIF_CTRL, 3, 1, 0),
+ /* BIT 4: SPI2SDI */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_CM9761_SPDIF_CTRL, 4, 1, 0),
+ /* BIT 9-10: DAC_CTL */
+ AC97_ENUM("DAC Clock Source", cm9761_dac_clock_enum),
+};
+
+static int patch_cm9761_post_spdif(struct snd_ac97 * ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_cm9761_controls_spdif, ARRAY_SIZE(snd_ac97_cm9761_controls_spdif));
+}
+
+static int patch_cm9761_specific(struct snd_ac97 * ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_cm9761_controls, ARRAY_SIZE(snd_ac97_cm9761_controls));
+}
+
+static struct snd_ac97_build_ops patch_cm9761_ops = {
+ .build_specific = patch_cm9761_specific,
+ .build_post_spdif = patch_cm9761_post_spdif,
+ .update_jacks = cm9761_update_jacks
+};
+
+static int patch_cm9761(struct snd_ac97 *ac97)
+{
+ unsigned short val;
+
+ /* CM9761 has no PCM volume although the register reacts */
+ /* Master volume seems to have _some_ influence on the analog
+ * input sounds
+ */
+ ac97->flags |= /*AC97_HAS_NO_MASTER_VOL |*/ AC97_HAS_NO_PCM_VOL;
+ snd_ac97_write_cache(ac97, AC97_MASTER, 0x8808);
+ snd_ac97_write_cache(ac97, AC97_PCM, 0x8808);
+
+ ac97->spec.dev_flags = 0; /* 1 = model 82 revision B, 2 = model 83 */
+ if (ac97->id == AC97_ID_CM9761_82) {
+ unsigned short tmp;
+ /* check page 1, reg 0x60 */
+ val = snd_ac97_read(ac97, AC97_INT_PAGING);
+ snd_ac97_write_cache(ac97, AC97_INT_PAGING, (val & ~0x0f) | 0x01);
+ tmp = snd_ac97_read(ac97, 0x60);
+ ac97->spec.dev_flags = tmp & 1; /* revision B? */
+ snd_ac97_write_cache(ac97, AC97_INT_PAGING, val);
+ } else if (ac97->id == AC97_ID_CM9761_83)
+ ac97->spec.dev_flags = 2;
+
+ ac97->build_ops = &patch_cm9761_ops;
+
+ /* enable spdif */
+ /* force the SPDIF bit in ext_id - codec doesn't set this bit! */
+ ac97->ext_id |= AC97_EI_SPDIF;
+ /* to be sure: we overwrite the ext status bits */
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_STATUS, 0x05c0);
+ /* Don't set 0x0200 here. This results in the silent analog output */
+ snd_ac97_write_cache(ac97, AC97_CM9761_SPDIF_CTRL, 0x0001); /* enable spdif-in */
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+
+ /* set-up multi channel */
+ /* bit 15: pc master beep off
+ * bit 14: pin47 = EAPD/SPDIF
+ * bit 13: vref ctl [= cm9739]
+ * bit 12: CLFE control (reverted on rev B)
+ * bit 11: Mic/center share (reverted on rev B)
+ * bit 10: suddound/line share
+ * bit 9: Analog-in mix -> surround
+ * bit 8: Analog-in mix -> CLFE
+ * bit 7: Mic/LFE share (mic/center/lfe)
+ * bit 5: vref select (9761A)
+ * bit 4: front control
+ * bit 3: surround control (revereted with rev B)
+ * bit 2: front mic
+ * bit 1: stereo mic
+ * bit 0: mic boost level (0=20dB, 1=30dB)
+ */
+
+#if 0
+ if (ac97->spec.dev_flags)
+ val = 0x0214;
+ else
+ val = 0x321c;
+#endif
+ val = snd_ac97_read(ac97, AC97_CM9761_MULTI_CHAN);
+ val |= (1 << 4); /* front on */
+ snd_ac97_write_cache(ac97, AC97_CM9761_MULTI_CHAN, val);
+
+ /* FIXME: set up GPIO */
+ snd_ac97_write_cache(ac97, 0x70, 0x0100);
+ snd_ac97_write_cache(ac97, 0x72, 0x0020);
+
+ return 0;
+}
+
+#define AC97_CM9780_SIDE 0x60
+#define AC97_CM9780_JACK 0x62
+#define AC97_CM9780_MIXER 0x64
+#define AC97_CM9780_MULTI_CHAN 0x66
+#define AC97_CM9780_SPDIF 0x6c
+
+static const char *cm9780_ch_select[] = { "Front", "Side", "Center/LFE", "Rear" };
+static const struct ac97_enum cm9780_ch_select_enum =
+ AC97_ENUM_SINGLE(AC97_CM9780_MULTI_CHAN, 6, 4, cm9780_ch_select);
+static const struct snd_kcontrol_new cm9780_controls[] = {
+ AC97_DOUBLE("Side Playback Switch", AC97_CM9780_SIDE, 15, 7, 1, 1),
+ AC97_DOUBLE("Side Playback Volume", AC97_CM9780_SIDE, 8, 0, 31, 0),
+ AC97_ENUM("Side Playback Route", cm9780_ch_select_enum),
+};
+
+static int patch_cm9780_specific(struct snd_ac97 *ac97)
+{
+ return patch_build_controls(ac97, cm9780_controls, ARRAY_SIZE(cm9780_controls));
+}
+
+static struct snd_ac97_build_ops patch_cm9780_ops = {
+ .build_specific = patch_cm9780_specific,
+ .build_post_spdif = patch_cm9761_post_spdif /* identical with CM9761 */
+};
+
+static int patch_cm9780(struct snd_ac97 *ac97)
+{
+ unsigned short val;
+
+ ac97->build_ops = &patch_cm9780_ops;
+
+ /* enable spdif */
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+ val = snd_ac97_read(ac97, AC97_CM9780_SPDIF);
+ val |= 0x1; /* SPDI_EN */
+ snd_ac97_write_cache(ac97, AC97_CM9780_SPDIF, val);
+ }
+
+ return 0;
+}
+
+/*
+ * VIA VT1616 codec
+ */
+static const struct snd_kcontrol_new snd_ac97_controls_vt1616[] = {
+AC97_SINGLE("DC Offset removal", 0x5a, 10, 1, 0),
+AC97_SINGLE("Alternate Level to Surround Out", 0x5a, 15, 1, 0),
+AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0),
+AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0),
+};
+
+static const char *slave_vols_vt1616[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ NULL
+};
+
+static const char *slave_sws_vt1616[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ NULL
+};
+
+/* find a mixer control element with the given name */
+static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
+ const char *name)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, name);
+ return snd_ctl_find_id(ac97->bus->card, &id);
+}
+
+/* create a virtual master control and add slaves */
+static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
+ const unsigned int *tlv, const char **slaves)
+{
+ struct snd_kcontrol *kctl;
+ const char **s;
+ int err;
+
+ kctl = snd_ctl_make_virtual_master(name, tlv);
+ if (!kctl)
+ return -ENOMEM;
+ err = snd_ctl_add(ac97->bus->card, kctl);
+ if (err < 0)
+ return err;
+
+ for (s = slaves; *s; s++) {
+ struct snd_kcontrol *sctl;
+
+ sctl = snd_ac97_find_mixer_ctl(ac97, *s);
+ if (!sctl) {
+ snd_printdd("Cannot find slave %s, skipped\n", *s);
+ continue;
+ }
+ err = snd_ctl_add_slave(kctl, sctl);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int patch_vt1616_specific(struct snd_ac97 * ac97)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+
+ if (snd_ac97_try_bit(ac97, 0x5a, 9))
+ if ((err = patch_build_controls(ac97, &snd_ac97_controls_vt1616[0], 1)) < 0)
+ return err;
+ if ((err = patch_build_controls(ac97, &snd_ac97_controls_vt1616[1], ARRAY_SIZE(snd_ac97_controls_vt1616) - 1)) < 0)
+ return err;
+
+ /* There is already a misnamed master switch. Rename it. */
+ kctl = snd_ac97_find_mixer_ctl(ac97, "Master Playback Volume");
+ if (!kctl)
+ return -EINVAL;
+
+ snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback");
+
+ err = snd_ac97_add_vmaster(ac97, "Master Playback Volume",
+ kctl->tlv.p, slave_vols_vt1616);
+ if (err < 0)
+ return err;
+
+ err = snd_ac97_add_vmaster(ac97, "Master Playback Switch",
+ NULL, slave_sws_vt1616);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_vt1616_ops = {
+ .build_specific = patch_vt1616_specific
+};
+
+static int patch_vt1616(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_vt1616_ops;
+ return 0;
+}
+
+/*
+ * VT1617A codec
+ */
+
+/*
+ * unfortunately, the vt1617a stashes the twiddlers required for
+ * noodling the i/o jacks on 2 different regs. that means that we can't
+ * use the easy way provided by AC97_ENUM_DOUBLE() we have to write
+ * are own funcs.
+ *
+ * NB: this is absolutely and utterly different from the vt1618. dunno
+ * about the 1616.
+ */
+
+/* copied from ac97_surround_jack_mode_info() */
+static int snd_ac97_vt1617a_smart51_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ /* ordering in this list reflects vt1617a docs for Reg 20 and
+ * 7a and Table 6 that lays out the matrix NB WRT Table6: SM51
+ * is SM51EN *AND* it's Bit14, not Bit15 so the table is very
+ * counter-intuitive */
+
+ static const char* texts[] = { "LineIn Mic1", "LineIn Mic1 Mic3",
+ "Surr LFE/C Mic3", "LineIn LFE/C Mic3",
+ "LineIn Mic2", "LineIn Mic2 Mic1",
+ "Surr LFE Mic1", "Surr LFE Mic1 Mic2"};
+ return ac97_enum_text_info(kcontrol, uinfo, texts, 8);
+}
+
+static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ushort usSM51, usMS;
+
+ struct snd_ac97 *pac97;
+
+ pac97 = snd_kcontrol_chip(kcontrol); /* grab codec handle */
+
+ /* grab our desired bits, then mash them together in a manner
+ * consistent with Table 6 on page 17 in the 1617a docs */
+
+ usSM51 = snd_ac97_read(pac97, 0x7a) >> 14;
+ usMS = snd_ac97_read(pac97, 0x20) >> 8;
+
+ ucontrol->value.enumerated.item[0] = (usSM51 << 1) + usMS;
+
+ return 0;
+}
+
+static int snd_ac97_vt1617a_smart51_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ushort usSM51, usMS, usReg;
+
+ struct snd_ac97 *pac97;
+
+ pac97 = snd_kcontrol_chip(kcontrol); /* grab codec handle */
+
+ usSM51 = ucontrol->value.enumerated.item[0] >> 1;
+ usMS = ucontrol->value.enumerated.item[0] & 1;
+
+ /* push our values into the register - consider that things will be left
+ * in a funky state if the write fails */
+
+ usReg = snd_ac97_read(pac97, 0x7a);
+ snd_ac97_write_cache(pac97, 0x7a, (usReg & 0x3FFF) + (usSM51 << 14));
+ usReg = snd_ac97_read(pac97, 0x20);
+ snd_ac97_write_cache(pac97, 0x20, (usReg & 0xFEFF) + (usMS << 8));
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_vt1617a[] = {
+
+ AC97_SINGLE("Center/LFE Exchange", 0x5a, 8, 1, 0),
+ /*
+ * These are used to enable/disable surround sound on motherboards
+ * that have 3 bidirectional analog jacks
+ */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Smart 5.1 Select",
+ .info = snd_ac97_vt1617a_smart51_info,
+ .get = snd_ac97_vt1617a_smart51_get,
+ .put = snd_ac97_vt1617a_smart51_put,
+ },
+};
+
+static int patch_vt1617a(struct snd_ac97 * ac97)
+{
+ int err = 0;
+ int val;
+
+ /* we choose to not fail out at this point, but we tell the
+ caller when we return */
+
+ err = patch_build_controls(ac97, &snd_ac97_controls_vt1617a[0],
+ ARRAY_SIZE(snd_ac97_controls_vt1617a));
+
+ /* bring analog power consumption to normal by turning off the
+ * headphone amplifier, like WinXP driver for EPIA SP
+ */
+ /* We need to check the bit before writing it.
+ * On some (many?) hardwares, setting bit actually clears it!
+ */
+ val = snd_ac97_read(ac97, 0x5c);
+ if (!(val & 0x20))
+ snd_ac97_write_cache(ac97, 0x5c, 0x20);
+
+ ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
+ ac97->build_ops = &patch_vt1616_ops;
+
+ return err;
+}
+
+/* VIA VT1618 8 CHANNEL AC97 CODEC
+ *
+ * VIA implements 'Smart 5.1' completely differently on the 1618 than
+ * it does on the 1617a. awesome! They seem to have sourced this
+ * particular revision of the technology from somebody else, it's
+ * called Universal Audio Jack and it shows up on some other folk's chips
+ * as well.
+ *
+ * ordering in this list reflects vt1618 docs for Reg 60h and
+ * the block diagram, DACs are as follows:
+ *
+ * OUT_O -> Front,
+ * OUT_1 -> Surround,
+ * OUT_2 -> C/LFE
+ *
+ * Unlike the 1617a, each OUT has a consistent set of mappings
+ * for all bitpatterns other than 00:
+ *
+ * 01 Unmixed Output
+ * 10 Line In
+ * 11 Mic In
+ *
+ * Special Case of 00:
+ *
+ * OUT_0 Mixed Output
+ * OUT_1 Reserved
+ * OUT_2 Reserved
+ *
+ * I have no idea what the hell Reserved does, but on an MSI
+ * CN700T, i have to set it to get 5.1 output - YMMV, bad
+ * shit may happen.
+ *
+ * If other chips use Universal Audio Jack, then this code might be applicable
+ * to them.
+ */
+
+struct vt1618_uaj_item {
+ unsigned short mask;
+ unsigned short shift;
+ const char *items[4];
+};
+
+/* This list reflects the vt1618 docs for Vendor Defined Register 0x60. */
+
+static struct vt1618_uaj_item vt1618_uaj[3] = {
+ {
+ /* speaker jack */
+ .mask = 0x03,
+ .shift = 0,
+ .items = {
+ "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In"
+ }
+ },
+ {
+ /* line jack */
+ .mask = 0x0c,
+ .shift = 2,
+ .items = {
+ "Surround Out", "DAC Unmixed Out", "Line In", "Mic In"
+ }
+ },
+ {
+ /* mic jack */
+ .mask = 0x30,
+ .shift = 4,
+ .items = {
+ "Center LFE Out", "DAC Unmixed Out", "Line In", "Mic In"
+ },
+ },
+};
+
+static int snd_ac97_vt1618_UAJ_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ return ac97_enum_text_info(kcontrol, uinfo,
+ vt1618_uaj[kcontrol->private_value].items,
+ 4);
+}
+
+/* All of the vt1618 Universal Audio Jack twiddlers are on
+ * Vendor Defined Register 0x60, page 0. The bits, and thus
+ * the mask, are the only thing that changes
+ */
+static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned short datpag, uaj;
+ struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&pac97->page_mutex);
+
+ datpag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK;
+ snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0);
+
+ uaj = snd_ac97_read(pac97, 0x60) &
+ vt1618_uaj[kcontrol->private_value].mask;
+
+ snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, datpag);
+ mutex_unlock(&pac97->page_mutex);
+
+ ucontrol->value.enumerated.item[0] = uaj >>
+ vt1618_uaj[kcontrol->private_value].shift;
+
+ return 0;
+}
+
+static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return ac97_update_bits_page(snd_kcontrol_chip(kcontrol), 0x60,
+ vt1618_uaj[kcontrol->private_value].mask,
+ ucontrol->value.enumerated.item[0]<<
+ vt1618_uaj[kcontrol->private_value].shift,
+ 0);
+}
+
+/* config aux in jack - not found on 3 jack motherboards or soundcards */
+
+static int snd_ac97_vt1618_aux_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char *txt_aux[] = {"Aux In", "Back Surr Out"};
+
+ return ac97_enum_text_info(kcontrol, uinfo, txt_aux, 2);
+}
+
+static int snd_ac97_vt1618_aux_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.enumerated.item[0] =
+ (snd_ac97_read(snd_kcontrol_chip(kcontrol), 0x5c) & 0x0008)>>3;
+ return 0;
+}
+
+static int snd_ac97_vt1618_aux_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ /* toggle surround rear dac power */
+
+ snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x5c, 0x0008,
+ ucontrol->value.enumerated.item[0] << 3);
+
+ /* toggle aux in surround rear out jack */
+
+ return snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x76, 0x0008,
+ ucontrol->value.enumerated.item[0] << 3);
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = {
+ AC97_SINGLE("Exchange Center/LFE", 0x5a, 8, 1, 0),
+ AC97_SINGLE("DC Offset", 0x5a, 10, 1, 0),
+ AC97_SINGLE("Soft Mute", 0x5c, 0, 1, 1),
+ AC97_SINGLE("Headphone Amp", 0x5c, 5, 1, 1),
+ AC97_DOUBLE("Back Surr Volume", 0x5e, 8, 0, 31, 1),
+ AC97_SINGLE("Back Surr Switch", 0x5e, 15, 1, 1),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Speaker Jack Mode",
+ .info = snd_ac97_vt1618_UAJ_info,
+ .get = snd_ac97_vt1618_UAJ_get,
+ .put = snd_ac97_vt1618_UAJ_put,
+ .private_value = 0
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Jack Mode",
+ .info = snd_ac97_vt1618_UAJ_info,
+ .get = snd_ac97_vt1618_UAJ_get,
+ .put = snd_ac97_vt1618_UAJ_put,
+ .private_value = 1
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Jack Mode",
+ .info = snd_ac97_vt1618_UAJ_info,
+ .get = snd_ac97_vt1618_UAJ_get,
+ .put = snd_ac97_vt1618_UAJ_put,
+ .private_value = 2
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Aux Jack Mode",
+ .info = snd_ac97_vt1618_aux_info,
+ .get = snd_ac97_vt1618_aux_get,
+ .put = snd_ac97_vt1618_aux_put,
+ }
+};
+
+static int patch_vt1618(struct snd_ac97 *ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_controls_vt1618,
+ ARRAY_SIZE(snd_ac97_controls_vt1618));
+}
+
+/*
+ */
+static void it2646_update_jacks(struct snd_ac97 *ac97)
+{
+ /* shared Line-In / Surround Out */
+ snd_ac97_update_bits(ac97, 0x76, 1 << 9,
+ is_shared_surrout(ac97) ? (1<<9) : 0);
+ /* shared Mic / Center/LFE Out */
+ snd_ac97_update_bits(ac97, 0x76, 1 << 10,
+ is_shared_clfeout(ac97) ? (1<<10) : 0);
+}
+
+static const struct snd_kcontrol_new snd_ac97_controls_it2646[] = {
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
+};
+
+static const struct snd_kcontrol_new snd_ac97_spdif_controls_it2646[] = {
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0x76, 11, 1, 0),
+ AC97_SINGLE("Analog to IEC958 Output", 0x76, 12, 1, 0),
+ AC97_SINGLE("IEC958 Input Monitor", 0x76, 13, 1, 0),
+};
+
+static int patch_it2646_specific(struct snd_ac97 * ac97)
+{
+ int err;
+ if ((err = patch_build_controls(ac97, snd_ac97_controls_it2646, ARRAY_SIZE(snd_ac97_controls_it2646))) < 0)
+ return err;
+ if ((err = patch_build_controls(ac97, snd_ac97_spdif_controls_it2646, ARRAY_SIZE(snd_ac97_spdif_controls_it2646))) < 0)
+ return err;
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_it2646_ops = {
+ .build_specific = patch_it2646_specific,
+ .update_jacks = it2646_update_jacks
+};
+
+static int patch_it2646(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_it2646_ops;
+ /* full DAC volume */
+ snd_ac97_write_cache(ac97, 0x5E, 0x0808);
+ snd_ac97_write_cache(ac97, 0x7A, 0x0808);
+ return 0;
+}
+
+/*
+ * Si3036 codec
+ */
+
+#define AC97_SI3036_CHIP_ID 0x5a
+#define AC97_SI3036_LINE_CFG 0x5c
+
+static const struct snd_kcontrol_new snd_ac97_controls_si3036[] = {
+AC97_DOUBLE("Modem Speaker Volume", 0x5c, 14, 12, 3, 1)
+};
+
+static int patch_si3036_specific(struct snd_ac97 * ac97)
+{
+ int idx, err;
+ for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_si3036); idx++)
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ctl_new1(&snd_ac97_controls_si3036[idx], ac97))) < 0)
+ return err;
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_si3036_ops = {
+ .build_specific = patch_si3036_specific,
+};
+
+static int mpatch_si3036(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_si3036_ops;
+ snd_ac97_write_cache(ac97, 0x5c, 0xf210 );
+ snd_ac97_write_cache(ac97, 0x68, 0);
+ return 0;
+}
+
+/*
+ * LM 4550 Codec
+ *
+ * We use a static resolution table since LM4550 codec cannot be
+ * properly autoprobed to determine the resolution via
+ * check_volume_resolution().
+ */
+
+static struct snd_ac97_res_table lm4550_restbl[] = {
+ { AC97_MASTER, 0x1f1f },
+ { AC97_HEADPHONE, 0x1f1f },
+ { AC97_MASTER_MONO, 0x001f },
+ { AC97_PC_BEEP, 0x001f }, /* LSB is ignored */
+ { AC97_PHONE, 0x001f },
+ { AC97_MIC, 0x001f },
+ { AC97_LINE, 0x1f1f },
+ { AC97_CD, 0x1f1f },
+ { AC97_VIDEO, 0x1f1f },
+ { AC97_AUX, 0x1f1f },
+ { AC97_PCM, 0x1f1f },
+ { AC97_REC_GAIN, 0x0f0f },
+ { } /* terminator */
+};
+
+static int patch_lm4550(struct snd_ac97 *ac97)
+{
+ ac97->res_table = lm4550_restbl;
+ return 0;
+}
+
+/*
+ * UCB1400 codec (http://www.semiconductors.philips.com/acrobat_download/datasheets/UCB1400-02.pdf)
+ */
+static const struct snd_kcontrol_new snd_ac97_controls_ucb1400[] = {
+/* enable/disable headphone driver which allows direct connection to
+ stereo headphone without the use of external DC blocking
+ capacitors */
+AC97_SINGLE("Headphone Driver", 0x6a, 6, 1, 0),
+/* Filter used to compensate the DC offset is added in the ADC to remove idle
+ tones from the audio band. */
+AC97_SINGLE("DC Filter", 0x6a, 4, 1, 0),
+/* Control smart-low-power mode feature. Allows automatic power down
+ of unused blocks in the ADC analog front end and the PLL. */
+AC97_SINGLE("Smart Low Power Mode", 0x6c, 4, 3, 0),
+};
+
+static int patch_ucb1400_specific(struct snd_ac97 * ac97)
+{
+ int idx, err;
+ for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_ucb1400); idx++)
+ if ((err = snd_ctl_add(ac97->bus->card, snd_ctl_new1(&snd_ac97_controls_ucb1400[idx], ac97))) < 0)
+ return err;
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_ucb1400_ops = {
+ .build_specific = patch_ucb1400_specific,
+};
+
+static int patch_ucb1400(struct snd_ac97 * ac97)
+{
+ ac97->build_ops = &patch_ucb1400_ops;
+ /* enable headphone driver and smart low power mode by default */
+ snd_ac97_write_cache(ac97, 0x6a, 0x0050);
+ snd_ac97_write_cache(ac97, 0x6c, 0x0030);
+ return 0;
+}
diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h
new file mode 100644
index 0000000..47bf8df
--- /dev/null
+++ b/sound/pci/ac97/ac97_patch.h
@@ -0,0 +1,95 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.2
+ * by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define AC97_SINGLE_VALUE(reg,shift,mask,invert) \
+ ((reg) | ((shift) << 8) | ((shift) << 12) | ((mask) << 16) | \
+ ((invert) << 24))
+#define AC97_PAGE_SINGLE_VALUE(reg,shift,mask,invert,page) \
+ (AC97_SINGLE_VALUE(reg,shift,mask,invert) | (1<<25) | ((page) << 26))
+#define AC97_SINGLE(xname, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_ac97_info_volsw, \
+ .get = snd_ac97_get_volsw, .put = snd_ac97_put_volsw, \
+ .private_value = AC97_SINGLE_VALUE(reg, shift, mask, invert) }
+#define AC97_PAGE_SINGLE(xname, reg, shift, mask, invert, page) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_ac97_info_volsw, \
+ .get = snd_ac97_get_volsw, .put = snd_ac97_put_volsw, \
+ .private_value = AC97_PAGE_SINGLE_VALUE(reg, shift, mask, invert, page) }
+#define AC97_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .info = snd_ac97_info_volsw, \
+ .get = snd_ac97_get_volsw, .put = snd_ac97_put_volsw, \
+ .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) | ((mask) << 16) | ((invert) << 24) }
+
+/* enum control */
+struct ac97_enum {
+ unsigned char reg;
+ unsigned char shift_l;
+ unsigned char shift_r;
+ unsigned short mask;
+ const char **texts;
+};
+
+#define AC97_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \
+{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
+ .mask = xmask, .texts = xtexts }
+#define AC97_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \
+ AC97_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts)
+#define AC97_ENUM(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_ac97_info_enum_double, \
+ .get = snd_ac97_get_enum_double, .put = snd_ac97_put_enum_double, \
+ .private_value = (unsigned long)&xenum }
+
+/* ac97_codec.c */
+static const struct snd_kcontrol_new snd_ac97_controls_3d[];
+static const struct snd_kcontrol_new snd_ac97_controls_spdif[];
+static struct snd_kcontrol *snd_ac97_cnew(const struct snd_kcontrol_new *_template,
+ struct snd_ac97 * ac97);
+static int snd_ac97_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+static int snd_ac97_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+static int snd_ac97_try_bit(struct snd_ac97 * ac97, int reg, int bit);
+static int snd_ac97_remove_ctl(struct snd_ac97 *ac97, const char *name,
+ const char *suffix);
+static int snd_ac97_rename_ctl(struct snd_ac97 *ac97, const char *src,
+ const char *dst, const char *suffix);
+static int snd_ac97_swap_ctl(struct snd_ac97 *ac97, const char *s1,
+ const char *s2, const char *suffix);
+static void snd_ac97_rename_vol_ctl(struct snd_ac97 *ac97, const char *src,
+ const char *dst);
+#ifdef CONFIG_PM
+static void snd_ac97_restore_status(struct snd_ac97 *ac97);
+static void snd_ac97_restore_iec958(struct snd_ac97 *ac97);
+#endif
+static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+static int snd_ac97_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c
new file mode 100644
index 0000000..48cbda9
--- /dev/null
+++ b/sound/pci/ac97/ac97_pcm.c
@@ -0,0 +1,736 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.2
+ * by Intel Corporation (http://developer.intel.com) and to datasheets
+ * for specific codecs.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/ac97_codec.h>
+#include <sound/asoundef.h>
+#include "ac97_id.h"
+#include "ac97_local.h"
+
+/*
+ * PCM support
+ */
+
+static unsigned char rate_reg_tables[2][4][9] = {
+{
+ /* standard rates */
+ {
+ /* 3&4 front, 7&8 rear, 6&9 center/lfe */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 3 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 4 */
+ 0xff, /* slot 5 */
+ AC97_PCM_LFE_DAC_RATE, /* slot 6 */
+ AC97_PCM_SURR_DAC_RATE, /* slot 7 */
+ AC97_PCM_SURR_DAC_RATE, /* slot 8 */
+ AC97_PCM_LFE_DAC_RATE, /* slot 9 */
+ 0xff, /* slot 10 */
+ 0xff, /* slot 11 */
+ },
+ {
+ /* 7&8 front, 6&9 rear, 10&11 center/lfe */
+ 0xff, /* slot 3 */
+ 0xff, /* slot 4 */
+ 0xff, /* slot 5 */
+ AC97_PCM_SURR_DAC_RATE, /* slot 6 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 7 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 8 */
+ AC97_PCM_SURR_DAC_RATE, /* slot 9 */
+ AC97_PCM_LFE_DAC_RATE, /* slot 10 */
+ AC97_PCM_LFE_DAC_RATE, /* slot 11 */
+ },
+ {
+ /* 6&9 front, 10&11 rear, 3&4 center/lfe */
+ AC97_PCM_LFE_DAC_RATE, /* slot 3 */
+ AC97_PCM_LFE_DAC_RATE, /* slot 4 */
+ 0xff, /* slot 5 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 6 */
+ 0xff, /* slot 7 */
+ 0xff, /* slot 8 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 9 */
+ AC97_PCM_SURR_DAC_RATE, /* slot 10 */
+ AC97_PCM_SURR_DAC_RATE, /* slot 11 */
+ },
+ {
+ /* 10&11 front, 3&4 rear, 7&8 center/lfe */
+ AC97_PCM_SURR_DAC_RATE, /* slot 3 */
+ AC97_PCM_SURR_DAC_RATE, /* slot 4 */
+ 0xff, /* slot 5 */
+ 0xff, /* slot 6 */
+ AC97_PCM_LFE_DAC_RATE, /* slot 7 */
+ AC97_PCM_LFE_DAC_RATE, /* slot 8 */
+ 0xff, /* slot 9 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 10 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 11 */
+ },
+},
+{
+ /* double rates */
+ {
+ /* 3&4 front, 7&8 front (t+1) */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 3 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 4 */
+ 0xff, /* slot 5 */
+ 0xff, /* slot 6 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 7 */
+ AC97_PCM_FRONT_DAC_RATE, /* slot 8 */
+ 0xff, /* slot 9 */
+ 0xff, /* slot 10 */
+ 0xff, /* slot 11 */
+ },
+ {
+ /* not specified in the specification */
+ 0xff, /* slot 3 */
+ 0xff, /* slot 4 */
+ 0xff, /* slot 5 */
+ 0xff, /* slot 6 */
+ 0xff, /* slot 7 */
+ 0xff, /* slot 8 */
+ 0xff, /* slot 9 */
+ 0xff, /* slot 10 */
+ 0xff, /* slot 11 */
+ },
+ {
+ 0xff, /* slot 3 */
+ 0xff, /* slot 4 */
+ 0xff, /* slot 5 */
+ 0xff, /* slot 6 */
+ 0xff, /* slot 7 */
+ 0xff, /* slot 8 */
+ 0xff, /* slot 9 */
+ 0xff, /* slot 10 */
+ 0xff, /* slot 11 */
+ },
+ {
+ 0xff, /* slot 3 */
+ 0xff, /* slot 4 */
+ 0xff, /* slot 5 */
+ 0xff, /* slot 6 */
+ 0xff, /* slot 7 */
+ 0xff, /* slot 8 */
+ 0xff, /* slot 9 */
+ 0xff, /* slot 10 */
+ 0xff, /* slot 11 */
+ }
+}};
+
+/* FIXME: more various mappings for ADC? */
+static unsigned char rate_cregs[9] = {
+ AC97_PCM_LR_ADC_RATE, /* 3 */
+ AC97_PCM_LR_ADC_RATE, /* 4 */
+ 0xff, /* 5 */
+ AC97_PCM_MIC_ADC_RATE, /* 6 */
+ 0xff, /* 7 */
+ 0xff, /* 8 */
+ 0xff, /* 9 */
+ 0xff, /* 10 */
+ 0xff, /* 11 */
+};
+
+static unsigned char get_slot_reg(struct ac97_pcm *pcm, unsigned short cidx,
+ unsigned short slot, int dbl)
+{
+ if (slot < 3)
+ return 0xff;
+ if (slot > 11)
+ return 0xff;
+ if (pcm->spdif)
+ return AC97_SPDIF; /* pseudo register */
+ if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return rate_reg_tables[dbl][pcm->r[dbl].rate_table[cidx]][slot - 3];
+ else
+ return rate_cregs[slot - 3];
+}
+
+static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate)
+{
+ unsigned short old, bits, reg, mask;
+ unsigned int sbits;
+
+ if (! (ac97->ext_id & AC97_EI_SPDIF))
+ return -ENODEV;
+
+ /* TODO: double rate support */
+ if (ac97->flags & AC97_CS_SPDIF) {
+ switch (rate) {
+ case 48000: bits = 0; break;
+ case 44100: bits = 1 << AC97_SC_SPSR_SHIFT; break;
+ default: /* invalid - disable output */
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0);
+ return -EINVAL;
+ }
+ reg = AC97_CSR_SPDIF;
+ mask = 1 << AC97_SC_SPSR_SHIFT;
+ } else {
+ if (ac97->id == AC97_ID_CM9739 && rate != 48000) {
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0);
+ return -EINVAL;
+ }
+ switch (rate) {
+ case 44100: bits = AC97_SC_SPSR_44K; break;
+ case 48000: bits = AC97_SC_SPSR_48K; break;
+ case 32000: bits = AC97_SC_SPSR_32K; break;
+ default: /* invalid - disable output */
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0);
+ return -EINVAL;
+ }
+ reg = AC97_SPDIF;
+ mask = AC97_SC_SPSR_MASK;
+ }
+
+ mutex_lock(&ac97->reg_mutex);
+ old = snd_ac97_read(ac97, reg) & mask;
+ if (old != bits) {
+ snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0);
+ snd_ac97_update_bits_nolock(ac97, reg, mask, bits);
+ /* update the internal spdif bits */
+ sbits = ac97->spdif_status;
+ if (sbits & IEC958_AES0_PROFESSIONAL) {
+ sbits &= ~IEC958_AES0_PRO_FS;
+ switch (rate) {
+ case 44100: sbits |= IEC958_AES0_PRO_FS_44100; break;
+ case 48000: sbits |= IEC958_AES0_PRO_FS_48000; break;
+ case 32000: sbits |= IEC958_AES0_PRO_FS_32000; break;
+ }
+ } else {
+ sbits &= ~(IEC958_AES3_CON_FS << 24);
+ switch (rate) {
+ case 44100: sbits |= IEC958_AES3_CON_FS_44100<<24; break;
+ case 48000: sbits |= IEC958_AES3_CON_FS_48000<<24; break;
+ case 32000: sbits |= IEC958_AES3_CON_FS_32000<<24; break;
+ }
+ }
+ ac97->spdif_status = sbits;
+ }
+ snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, AC97_EA_SPDIF);
+ mutex_unlock(&ac97->reg_mutex);
+ return 0;
+}
+
+/**
+ * snd_ac97_set_rate - change the rate of the given input/output.
+ * @ac97: the ac97 instance
+ * @reg: the register to change
+ * @rate: the sample rate to set
+ *
+ * Changes the rate of the given input/output on the codec.
+ * If the codec doesn't support VAR, the rate must be 48000 (except
+ * for SPDIF).
+ *
+ * The valid registers are AC97_PMC_MIC_ADC_RATE,
+ * AC97_PCM_FRONT_DAC_RATE, AC97_PCM_LR_ADC_RATE.
+ * AC97_PCM_SURR_DAC_RATE and AC97_PCM_LFE_DAC_RATE are accepted
+ * if the codec supports them.
+ * AC97_SPDIF is accepted as a pseudo register to modify the SPDIF
+ * status bits.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate)
+{
+ int dbl;
+ unsigned int tmp;
+
+ dbl = rate > 48000;
+ if (dbl) {
+ if (!(ac97->flags & AC97_DOUBLE_RATE))
+ return -EINVAL;
+ if (reg != AC97_PCM_FRONT_DAC_RATE)
+ return -EINVAL;
+ }
+
+ snd_ac97_update_power(ac97, reg, 1);
+ switch (reg) {
+ case AC97_PCM_MIC_ADC_RATE:
+ if ((ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_VRM) == 0) /* MIC VRA */
+ if (rate != 48000)
+ return -EINVAL;
+ break;
+ case AC97_PCM_FRONT_DAC_RATE:
+ case AC97_PCM_LR_ADC_RATE:
+ if ((ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_VRA) == 0) /* VRA */
+ if (rate != 48000 && rate != 96000)
+ return -EINVAL;
+ break;
+ case AC97_PCM_SURR_DAC_RATE:
+ if (! (ac97->scaps & AC97_SCAP_SURROUND_DAC))
+ return -EINVAL;
+ break;
+ case AC97_PCM_LFE_DAC_RATE:
+ if (! (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC))
+ return -EINVAL;
+ break;
+ case AC97_SPDIF:
+ /* special case */
+ return set_spdif_rate(ac97, rate);
+ default:
+ return -EINVAL;
+ }
+ if (dbl)
+ rate /= 2;
+ tmp = (rate * ac97->bus->clock) / 48000;
+ if (tmp > 65535)
+ return -EINVAL;
+ if ((ac97->ext_id & AC97_EI_DRA) && reg == AC97_PCM_FRONT_DAC_RATE)
+ snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
+ AC97_EA_DRA, dbl ? AC97_EA_DRA : 0);
+ snd_ac97_update(ac97, reg, tmp & 0xffff);
+ snd_ac97_read(ac97, reg);
+ if ((ac97->ext_id & AC97_EI_DRA) && reg == AC97_PCM_FRONT_DAC_RATE) {
+ /* Intel controllers require double rate data to be put in
+ * slots 7+8
+ */
+ snd_ac97_update_bits(ac97, AC97_GENERAL_PURPOSE,
+ AC97_GP_DRSS_MASK,
+ dbl ? AC97_GP_DRSS_78 : 0);
+ snd_ac97_read(ac97, AC97_GENERAL_PURPOSE);
+ }
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_set_rate);
+
+static unsigned short get_pslots(struct snd_ac97 *ac97, unsigned char *rate_table, unsigned short *spdif_slots)
+{
+ if (!ac97_is_audio(ac97))
+ return 0;
+ if (ac97_is_rev22(ac97) || ac97_can_amap(ac97)) {
+ unsigned short slots = 0;
+ if (ac97_is_rev22(ac97)) {
+ /* Note: it's simply emulation of AMAP behaviour */
+ u16 es;
+ es = ac97->regs[AC97_EXTENDED_ID] &= ~AC97_EI_DACS_SLOT_MASK;
+ switch (ac97->addr) {
+ case 1:
+ case 2: es |= (1<<AC97_EI_DACS_SLOT_SHIFT); break;
+ case 3: es |= (2<<AC97_EI_DACS_SLOT_SHIFT); break;
+ }
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_ID, es);
+ }
+ switch (ac97->addr) {
+ case 0:
+ slots |= (1<<AC97_SLOT_PCM_LEFT)|(1<<AC97_SLOT_PCM_RIGHT);
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
+ slots |= (1<<AC97_SLOT_PCM_SLEFT)|(1<<AC97_SLOT_PCM_SRIGHT);
+ if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)
+ slots |= (1<<AC97_SLOT_PCM_CENTER)|(1<<AC97_SLOT_LFE);
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ if (!(ac97->scaps & AC97_SCAP_SURROUND_DAC))
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT)|(1<<AC97_SLOT_SPDIF_RIGHT);
+ else if (!(ac97->scaps & AC97_SCAP_CENTER_LFE_DAC))
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT1)|(1<<AC97_SLOT_SPDIF_RIGHT1);
+ else
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT2)|(1<<AC97_SLOT_SPDIF_RIGHT2);
+ }
+ *rate_table = 0;
+ break;
+ case 1:
+ case 2:
+ slots |= (1<<AC97_SLOT_PCM_SLEFT)|(1<<AC97_SLOT_PCM_SRIGHT);
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
+ slots |= (1<<AC97_SLOT_PCM_CENTER)|(1<<AC97_SLOT_LFE);
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ if (!(ac97->scaps & AC97_SCAP_SURROUND_DAC))
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT1)|(1<<AC97_SLOT_SPDIF_RIGHT1);
+ else
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT2)|(1<<AC97_SLOT_SPDIF_RIGHT2);
+ }
+ *rate_table = 1;
+ break;
+ case 3:
+ slots |= (1<<AC97_SLOT_PCM_CENTER)|(1<<AC97_SLOT_LFE);
+ if (ac97->ext_id & AC97_EI_SPDIF)
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT2)|(1<<AC97_SLOT_SPDIF_RIGHT2);
+ *rate_table = 2;
+ break;
+ }
+ return slots;
+ } else {
+ unsigned short slots;
+ slots = (1<<AC97_SLOT_PCM_LEFT)|(1<<AC97_SLOT_PCM_RIGHT);
+ if (ac97->scaps & AC97_SCAP_SURROUND_DAC)
+ slots |= (1<<AC97_SLOT_PCM_SLEFT)|(1<<AC97_SLOT_PCM_SRIGHT);
+ if (ac97->scaps & AC97_SCAP_CENTER_LFE_DAC)
+ slots |= (1<<AC97_SLOT_PCM_CENTER)|(1<<AC97_SLOT_LFE);
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ if (!(ac97->scaps & AC97_SCAP_SURROUND_DAC))
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT)|(1<<AC97_SLOT_SPDIF_RIGHT);
+ else if (!(ac97->scaps & AC97_SCAP_CENTER_LFE_DAC))
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT1)|(1<<AC97_SLOT_SPDIF_RIGHT1);
+ else
+ *spdif_slots = (1<<AC97_SLOT_SPDIF_LEFT2)|(1<<AC97_SLOT_SPDIF_RIGHT2);
+ }
+ *rate_table = 0;
+ return slots;
+ }
+}
+
+static unsigned short get_cslots(struct snd_ac97 *ac97)
+{
+ unsigned short slots;
+
+ if (!ac97_is_audio(ac97))
+ return 0;
+ slots = (1<<AC97_SLOT_PCM_LEFT)|(1<<AC97_SLOT_PCM_RIGHT);
+ slots |= (1<<AC97_SLOT_MIC);
+ return slots;
+}
+
+static unsigned int get_rates(struct ac97_pcm *pcm, unsigned int cidx, unsigned short slots, int dbl)
+{
+ int i, idx;
+ unsigned int rates = ~0;
+ unsigned char reg;
+
+ for (i = 3; i < 12; i++) {
+ if (!(slots & (1 << i)))
+ continue;
+ reg = get_slot_reg(pcm, cidx, i, dbl);
+ switch (reg) {
+ case AC97_PCM_FRONT_DAC_RATE: idx = AC97_RATES_FRONT_DAC; break;
+ case AC97_PCM_SURR_DAC_RATE: idx = AC97_RATES_SURR_DAC; break;
+ case AC97_PCM_LFE_DAC_RATE: idx = AC97_RATES_LFE_DAC; break;
+ case AC97_PCM_LR_ADC_RATE: idx = AC97_RATES_ADC; break;
+ case AC97_PCM_MIC_ADC_RATE: idx = AC97_RATES_MIC_ADC; break;
+ default: idx = AC97_RATES_SPDIF; break;
+ }
+ rates &= pcm->r[dbl].codec[cidx]->rates[idx];
+ }
+ if (!dbl)
+ rates &= ~(SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000);
+ return rates;
+}
+
+/**
+ * snd_ac97_pcm_assign - assign AC97 slots to given PCM streams
+ * @bus: the ac97 bus instance
+ * @pcms_count: count of PCMs to be assigned
+ * @pcms: PCMs to be assigned
+ *
+ * It assigns available AC97 slots for given PCMs. If none or only
+ * some slots are available, pcm->xxx.slots and pcm->xxx.rslots[] members
+ * are reduced and might be zero.
+ */
+int snd_ac97_pcm_assign(struct snd_ac97_bus *bus,
+ unsigned short pcms_count,
+ const struct ac97_pcm *pcms)
+{
+ int i, j, k;
+ const struct ac97_pcm *pcm;
+ struct ac97_pcm *rpcms, *rpcm;
+ unsigned short avail_slots[2][4];
+ unsigned char rate_table[2][4];
+ unsigned short tmp, slots;
+ unsigned short spdif_slots[4];
+ unsigned int rates;
+ struct snd_ac97 *codec;
+
+ rpcms = kcalloc(pcms_count, sizeof(struct ac97_pcm), GFP_KERNEL);
+ if (rpcms == NULL)
+ return -ENOMEM;
+ memset(avail_slots, 0, sizeof(avail_slots));
+ memset(rate_table, 0, sizeof(rate_table));
+ memset(spdif_slots, 0, sizeof(spdif_slots));
+ for (i = 0; i < 4; i++) {
+ codec = bus->codec[i];
+ if (!codec)
+ continue;
+ avail_slots[0][i] = get_pslots(codec, &rate_table[0][i], &spdif_slots[i]);
+ avail_slots[1][i] = get_cslots(codec);
+ if (!(codec->scaps & AC97_SCAP_INDEP_SDIN)) {
+ for (j = 0; j < i; j++) {
+ if (bus->codec[j])
+ avail_slots[1][i] &= ~avail_slots[1][j];
+ }
+ }
+ }
+ /* first step - exclusive devices */
+ for (i = 0; i < pcms_count; i++) {
+ pcm = &pcms[i];
+ rpcm = &rpcms[i];
+ /* low-level driver thinks that it's more clever */
+ if (pcm->copy_flag) {
+ *rpcm = *pcm;
+ continue;
+ }
+ rpcm->stream = pcm->stream;
+ rpcm->exclusive = pcm->exclusive;
+ rpcm->spdif = pcm->spdif;
+ rpcm->private_value = pcm->private_value;
+ rpcm->bus = bus;
+ rpcm->rates = ~0;
+ slots = pcm->r[0].slots;
+ for (j = 0; j < 4 && slots; j++) {
+ if (!bus->codec[j])
+ continue;
+ rates = ~0;
+ if (pcm->spdif && pcm->stream == 0)
+ tmp = spdif_slots[j];
+ else
+ tmp = avail_slots[pcm->stream][j];
+ if (pcm->exclusive) {
+ /* exclusive access */
+ tmp &= slots;
+ for (k = 0; k < i; k++) {
+ if (rpcm->stream == rpcms[k].stream)
+ tmp &= ~rpcms[k].r[0].rslots[j];
+ }
+ } else {
+ /* non-exclusive access */
+ tmp &= pcm->r[0].slots;
+ }
+ if (tmp) {
+ rpcm->r[0].rslots[j] = tmp;
+ rpcm->r[0].codec[j] = bus->codec[j];
+ rpcm->r[0].rate_table[j] = rate_table[pcm->stream][j];
+ if (bus->no_vra)
+ rates = SNDRV_PCM_RATE_48000;
+ else
+ rates = get_rates(rpcm, j, tmp, 0);
+ if (pcm->exclusive)
+ avail_slots[pcm->stream][j] &= ~tmp;
+ }
+ slots &= ~tmp;
+ rpcm->r[0].slots |= tmp;
+ rpcm->rates &= rates;
+ }
+ /* for double rate, we check the first codec only */
+ if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ bus->codec[0] && (bus->codec[0]->flags & AC97_DOUBLE_RATE) &&
+ rate_table[pcm->stream][0] == 0) {
+ tmp = (1<<AC97_SLOT_PCM_LEFT) | (1<<AC97_SLOT_PCM_RIGHT) |
+ (1<<AC97_SLOT_PCM_LEFT_0) | (1<<AC97_SLOT_PCM_RIGHT_0);
+ if ((tmp & pcm->r[1].slots) == tmp) {
+ rpcm->r[1].slots = tmp;
+ rpcm->r[1].rslots[0] = tmp;
+ rpcm->r[1].rate_table[0] = 0;
+ rpcm->r[1].codec[0] = bus->codec[0];
+ if (pcm->exclusive)
+ avail_slots[pcm->stream][0] &= ~tmp;
+ if (bus->no_vra)
+ rates = SNDRV_PCM_RATE_96000;
+ else
+ rates = get_rates(rpcm, 0, tmp, 1);
+ rpcm->rates |= rates;
+ }
+ }
+ if (rpcm->rates == ~0)
+ rpcm->rates = 0; /* not used */
+ }
+ bus->pcms_count = pcms_count;
+ bus->pcms = rpcms;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_pcm_assign);
+
+/**
+ * snd_ac97_pcm_open - opens the given AC97 pcm
+ * @pcm: the ac97 pcm instance
+ * @rate: rate in Hz, if codec does not support VRA, this value must be 48000Hz
+ * @cfg: output stream characteristics
+ * @slots: a subset of allocated slots (snd_ac97_pcm_assign) for this pcm
+ *
+ * It locks the specified slots and sets the given rate to AC97 registers.
+ */
+int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
+ enum ac97_pcm_cfg cfg, unsigned short slots)
+{
+ struct snd_ac97_bus *bus;
+ int i, cidx, r, ok_flag;
+ unsigned int reg_ok[4] = {0,0,0,0};
+ unsigned char reg;
+ int err = 0;
+
+ r = rate > 48000;
+ bus = pcm->bus;
+ if (cfg == AC97_PCM_CFG_SPDIF) {
+ for (cidx = 0; cidx < 4; cidx++)
+ if (bus->codec[cidx] && (bus->codec[cidx]->ext_id & AC97_EI_SPDIF)) {
+ err = set_spdif_rate(bus->codec[cidx], rate);
+ if (err < 0)
+ return err;
+ }
+ }
+ spin_lock_irq(&pcm->bus->bus_lock);
+ for (i = 3; i < 12; i++) {
+ if (!(slots & (1 << i)))
+ continue;
+ ok_flag = 0;
+ for (cidx = 0; cidx < 4; cidx++) {
+ if (bus->used_slots[pcm->stream][cidx] & (1 << i)) {
+ spin_unlock_irq(&pcm->bus->bus_lock);
+ err = -EBUSY;
+ goto error;
+ }
+ if (pcm->r[r].rslots[cidx] & (1 << i)) {
+ bus->used_slots[pcm->stream][cidx] |= (1 << i);
+ ok_flag++;
+ }
+ }
+ if (!ok_flag) {
+ spin_unlock_irq(&pcm->bus->bus_lock);
+ snd_printk(KERN_ERR "cannot find configuration for AC97 slot %i\n", i);
+ err = -EAGAIN;
+ goto error;
+ }
+ }
+ pcm->cur_dbl = r;
+ spin_unlock_irq(&pcm->bus->bus_lock);
+ for (i = 3; i < 12; i++) {
+ if (!(slots & (1 << i)))
+ continue;
+ for (cidx = 0; cidx < 4; cidx++) {
+ if (pcm->r[r].rslots[cidx] & (1 << i)) {
+ reg = get_slot_reg(pcm, cidx, i, r);
+ if (reg == 0xff) {
+ snd_printk(KERN_ERR "invalid AC97 slot %i?\n", i);
+ continue;
+ }
+ if (reg_ok[cidx] & (1 << (reg - AC97_PCM_FRONT_DAC_RATE)))
+ continue;
+ //printk(KERN_DEBUG "setting ac97 reg 0x%x to rate %d\n", reg, rate);
+ err = snd_ac97_set_rate(pcm->r[r].codec[cidx], reg, rate);
+ if (err < 0)
+ snd_printk(KERN_ERR "error in snd_ac97_set_rate: cidx=%d, reg=0x%x, rate=%d, err=%d\n", cidx, reg, rate, err);
+ else
+ reg_ok[cidx] |= (1 << (reg - AC97_PCM_FRONT_DAC_RATE));
+ }
+ }
+ }
+ pcm->aslots = slots;
+ return 0;
+
+ error:
+ pcm->aslots = slots;
+ snd_ac97_pcm_close(pcm);
+ return err;
+}
+
+EXPORT_SYMBOL(snd_ac97_pcm_open);
+
+/**
+ * snd_ac97_pcm_close - closes the given AC97 pcm
+ * @pcm: the ac97 pcm instance
+ *
+ * It frees the locked AC97 slots.
+ */
+int snd_ac97_pcm_close(struct ac97_pcm *pcm)
+{
+ struct snd_ac97_bus *bus;
+ unsigned short slots = pcm->aslots;
+ int i, cidx;
+
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ int r = pcm->cur_dbl;
+ for (i = 3; i < 12; i++) {
+ if (!(slots & (1 << i)))
+ continue;
+ for (cidx = 0; cidx < 4; cidx++) {
+ if (pcm->r[r].rslots[cidx] & (1 << i)) {
+ int reg = get_slot_reg(pcm, cidx, i, r);
+ snd_ac97_update_power(pcm->r[r].codec[cidx],
+ reg, 0);
+ }
+ }
+ }
+#endif
+
+ bus = pcm->bus;
+ spin_lock_irq(&pcm->bus->bus_lock);
+ for (i = 3; i < 12; i++) {
+ if (!(slots & (1 << i)))
+ continue;
+ for (cidx = 0; cidx < 4; cidx++)
+ bus->used_slots[pcm->stream][cidx] &= ~(1 << i);
+ }
+ pcm->aslots = 0;
+ pcm->cur_dbl = 0;
+ spin_unlock_irq(&pcm->bus->bus_lock);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_ac97_pcm_close);
+
+static int double_rate_hw_constraint_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ if (channels->min > 2) {
+ static const struct snd_interval single_rates = {
+ .min = 1,
+ .max = 48000,
+ };
+ struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ return snd_interval_refine(rate, &single_rates);
+ }
+ return 0;
+}
+
+static int double_rate_hw_constraint_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (rate->min > 48000) {
+ static const struct snd_interval double_rate_channels = {
+ .min = 2,
+ .max = 2,
+ };
+ struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ return snd_interval_refine(channels, &double_rate_channels);
+ }
+ return 0;
+}
+
+/**
+ * snd_ac97_pcm_double_rate_rules - set double rate constraints
+ * @runtime: the runtime of the ac97 front playback pcm
+ *
+ * Installs the hardware constraint rules to prevent using double rates and
+ * more than two channels at the same time.
+ */
+int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ double_rate_hw_constraint_rate, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ double_rate_hw_constraint_channels, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ return err;
+}
+
+EXPORT_SYMBOL(snd_ac97_pcm_double_rate_rules);
diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c
new file mode 100644
index 0000000..060ea59
--- /dev/null
+++ b/sound/pci/ac97/ac97_proc.c
@@ -0,0 +1,489 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal interface for Audio Codec '97
+ *
+ * For more details look to AC '97 component specification revision 2.2
+ * by Intel Corporation (http://developer.intel.com).
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/ac97_codec.h>
+#include <sound/asoundef.h>
+#include "ac97_local.h"
+#include "ac97_id.h"
+
+/*
+ * proc interface
+ */
+
+static void snd_ac97_proc_read_functions(struct snd_ac97 *ac97, struct snd_info_buffer *buffer)
+{
+ int header = 0, function;
+ unsigned short info, sense_info;
+ static const char *function_names[12] = {
+ "Master Out", "AUX Out", "Center/LFE Out", "SPDIF Out",
+ "Phone In", "Mic 1", "Mic 2", "Line In", "CD In", "Video In",
+ "Aux In", "Mono Out"
+ };
+ static const char *locations[8] = {
+ "Rear I/O Panel", "Front Panel", "Motherboard", "Dock/External",
+ "reserved", "reserved", "reserved", "NC/unused"
+ };
+
+ for (function = 0; function < 12; ++function) {
+ snd_ac97_write(ac97, AC97_FUNC_SELECT, function << 1);
+ info = snd_ac97_read(ac97, AC97_FUNC_INFO);
+ if (!(info & 0x0001))
+ continue;
+ if (!header) {
+ snd_iprintf(buffer, "\n Gain Inverted Buffer delay Location\n");
+ header = 1;
+ }
+ sense_info = snd_ac97_read(ac97, AC97_SENSE_INFO);
+ snd_iprintf(buffer, "%-17s: %3d.%d dBV %c %2d/fs %s\n",
+ function_names[function],
+ (info & 0x8000 ? -1 : 1) * ((info & 0x7000) >> 12) * 3 / 2,
+ ((info & 0x0800) >> 11) * 5,
+ info & 0x0400 ? 'X' : '-',
+ (info & 0x03e0) >> 5,
+ locations[sense_info >> 13]);
+ }
+}
+
+static const char *snd_ac97_stereo_enhancements[] =
+{
+ /* 0 */ "No 3D Stereo Enhancement",
+ /* 1 */ "Analog Devices Phat Stereo",
+ /* 2 */ "Creative Stereo Enhancement",
+ /* 3 */ "National Semi 3D Stereo Enhancement",
+ /* 4 */ "YAMAHA Ymersion",
+ /* 5 */ "BBE 3D Stereo Enhancement",
+ /* 6 */ "Crystal Semi 3D Stereo Enhancement",
+ /* 7 */ "Qsound QXpander",
+ /* 8 */ "Spatializer 3D Stereo Enhancement",
+ /* 9 */ "SRS 3D Stereo Enhancement",
+ /* 10 */ "Platform Tech 3D Stereo Enhancement",
+ /* 11 */ "AKM 3D Audio",
+ /* 12 */ "Aureal Stereo Enhancement",
+ /* 13 */ "Aztech 3D Enhancement",
+ /* 14 */ "Binaura 3D Audio Enhancement",
+ /* 15 */ "ESS Technology Stereo Enhancement",
+ /* 16 */ "Harman International VMAx",
+ /* 17 */ "Nvidea/IC Ensemble/KS Waves 3D Stereo Enhancement",
+ /* 18 */ "Philips Incredible Sound",
+ /* 19 */ "Texas Instruments 3D Stereo Enhancement",
+ /* 20 */ "VLSI Technology 3D Stereo Enhancement",
+ /* 21 */ "TriTech 3D Stereo Enhancement",
+ /* 22 */ "Realtek 3D Stereo Enhancement",
+ /* 23 */ "Samsung 3D Stereo Enhancement",
+ /* 24 */ "Wolfson Microelectronics 3D Enhancement",
+ /* 25 */ "Delta Integration 3D Enhancement",
+ /* 26 */ "SigmaTel 3D Enhancement",
+ /* 27 */ "IC Ensemble/KS Waves",
+ /* 28 */ "Rockwell 3D Stereo Enhancement",
+ /* 29 */ "Reserved 29",
+ /* 30 */ "Reserved 30",
+ /* 31 */ "Reserved 31"
+};
+
+static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffer *buffer, int subidx)
+{
+ char name[64];
+ unsigned short val, tmp, ext, mext;
+ static const char *spdif_slots[4] = { " SPDIF=3/4", " SPDIF=7/8", " SPDIF=6/9", " SPDIF=10/11" };
+ static const char *spdif_rates[4] = { " Rate=44.1kHz", " Rate=res", " Rate=48kHz", " Rate=32kHz" };
+ static const char *spdif_rates_cs4205[4] = { " Rate=48kHz", " Rate=44.1kHz", " Rate=res", " Rate=res" };
+ static const char *double_rate_slots[4] = { "10/11", "7/8", "reserved", "reserved" };
+
+ snd_ac97_get_name(NULL, ac97->id, name, 0);
+ snd_iprintf(buffer, "%d-%d/%d: %s\n\n", ac97->addr, ac97->num, subidx, name);
+
+ if ((ac97->scaps & AC97_SCAP_AUDIO) == 0)
+ goto __modem;
+
+ snd_iprintf(buffer, "PCI Subsys Vendor: 0x%04x\n",
+ ac97->subsystem_vendor);
+ snd_iprintf(buffer, "PCI Subsys Device: 0x%04x\n\n",
+ ac97->subsystem_device);
+
+ if ((ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23) {
+ val = snd_ac97_read(ac97, AC97_INT_PAGING);
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING,
+ AC97_PAGE_MASK, AC97_PAGE_1);
+ tmp = snd_ac97_read(ac97, AC97_CODEC_CLASS_REV);
+ snd_iprintf(buffer, "Revision : 0x%02x\n", tmp & 0xff);
+ snd_iprintf(buffer, "Compat. Class : 0x%02x\n", (tmp >> 8) & 0x1f);
+ snd_iprintf(buffer, "Subsys. Vendor ID: 0x%04x\n",
+ snd_ac97_read(ac97, AC97_PCI_SVID));
+ snd_iprintf(buffer, "Subsys. ID : 0x%04x\n\n",
+ snd_ac97_read(ac97, AC97_PCI_SID));
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING,
+ AC97_PAGE_MASK, val & AC97_PAGE_MASK);
+ }
+
+ // val = snd_ac97_read(ac97, AC97_RESET);
+ val = ac97->caps;
+ snd_iprintf(buffer, "Capabilities :%s%s%s%s%s%s\n",
+ val & AC97_BC_DEDICATED_MIC ? " -dedicated MIC PCM IN channel-" : "",
+ val & AC97_BC_RESERVED1 ? " -reserved1-" : "",
+ val & AC97_BC_BASS_TREBLE ? " -bass & treble-" : "",
+ val & AC97_BC_SIM_STEREO ? " -simulated stereo-" : "",
+ val & AC97_BC_HEADPHONE ? " -headphone out-" : "",
+ val & AC97_BC_LOUDNESS ? " -loudness-" : "");
+ tmp = ac97->caps & AC97_BC_DAC_MASK;
+ snd_iprintf(buffer, "DAC resolution : %s%s%s%s\n",
+ tmp == AC97_BC_16BIT_DAC ? "16-bit" : "",
+ tmp == AC97_BC_18BIT_DAC ? "18-bit" : "",
+ tmp == AC97_BC_20BIT_DAC ? "20-bit" : "",
+ tmp == AC97_BC_DAC_MASK ? "???" : "");
+ tmp = ac97->caps & AC97_BC_ADC_MASK;
+ snd_iprintf(buffer, "ADC resolution : %s%s%s%s\n",
+ tmp == AC97_BC_16BIT_ADC ? "16-bit" : "",
+ tmp == AC97_BC_18BIT_ADC ? "18-bit" : "",
+ tmp == AC97_BC_20BIT_ADC ? "20-bit" : "",
+ tmp == AC97_BC_ADC_MASK ? "???" : "");
+ snd_iprintf(buffer, "3D enhancement : %s\n",
+ snd_ac97_stereo_enhancements[(val >> 10) & 0x1f]);
+ snd_iprintf(buffer, "\nCurrent setup\n");
+ val = snd_ac97_read(ac97, AC97_MIC);
+ snd_iprintf(buffer, "Mic gain : %s [%s]\n", val & 0x0040 ? "+20dB" : "+0dB", ac97->regs[AC97_MIC] & 0x0040 ? "+20dB" : "+0dB");
+ val = snd_ac97_read(ac97, AC97_GENERAL_PURPOSE);
+ snd_iprintf(buffer, "POP path : %s 3D\n"
+ "Sim. stereo : %s\n"
+ "3D enhancement : %s\n"
+ "Loudness : %s\n"
+ "Mono output : %s\n"
+ "Mic select : %s\n"
+ "ADC/DAC loopback : %s\n",
+ val & 0x8000 ? "post" : "pre",
+ val & 0x4000 ? "on" : "off",
+ val & 0x2000 ? "on" : "off",
+ val & 0x1000 ? "on" : "off",
+ val & 0x0200 ? "Mic" : "MIX",
+ val & 0x0100 ? "Mic2" : "Mic1",
+ val & 0x0080 ? "on" : "off");
+ if (ac97->ext_id & AC97_EI_DRA)
+ snd_iprintf(buffer, "Double rate slots: %s\n",
+ double_rate_slots[(val >> 10) & 3]);
+
+ ext = snd_ac97_read(ac97, AC97_EXTENDED_ID);
+ if (ext == 0)
+ goto __modem;
+
+ snd_iprintf(buffer, "Extended ID : codec=%i rev=%i%s%s%s%s DSA=%i%s%s%s%s\n",
+ (ext & AC97_EI_ADDR_MASK) >> AC97_EI_ADDR_SHIFT,
+ (ext & AC97_EI_REV_MASK) >> AC97_EI_REV_SHIFT,
+ ext & AC97_EI_AMAP ? " AMAP" : "",
+ ext & AC97_EI_LDAC ? " LDAC" : "",
+ ext & AC97_EI_SDAC ? " SDAC" : "",
+ ext & AC97_EI_CDAC ? " CDAC" : "",
+ (ext & AC97_EI_DACS_SLOT_MASK) >> AC97_EI_DACS_SLOT_SHIFT,
+ ext & AC97_EI_VRM ? " VRM" : "",
+ ext & AC97_EI_SPDIF ? " SPDIF" : "",
+ ext & AC97_EI_DRA ? " DRA" : "",
+ ext & AC97_EI_VRA ? " VRA" : "");
+ val = snd_ac97_read(ac97, AC97_EXTENDED_STATUS);
+ snd_iprintf(buffer, "Extended status :%s%s%s%s%s%s%s%s%s%s%s%s%s%s\n",
+ val & AC97_EA_PRL ? " PRL" : "",
+ val & AC97_EA_PRK ? " PRK" : "",
+ val & AC97_EA_PRJ ? " PRJ" : "",
+ val & AC97_EA_PRI ? " PRI" : "",
+ val & AC97_EA_SPCV ? " SPCV" : "",
+ val & AC97_EA_MDAC ? " MADC" : "",
+ val & AC97_EA_LDAC ? " LDAC" : "",
+ val & AC97_EA_SDAC ? " SDAC" : "",
+ val & AC97_EA_CDAC ? " CDAC" : "",
+ ext & AC97_EI_SPDIF ? spdif_slots[(val & AC97_EA_SPSA_SLOT_MASK) >> AC97_EA_SPSA_SLOT_SHIFT] : "",
+ val & AC97_EA_VRM ? " VRM" : "",
+ val & AC97_EA_SPDIF ? " SPDIF" : "",
+ val & AC97_EA_DRA ? " DRA" : "",
+ val & AC97_EA_VRA ? " VRA" : "");
+ if (ext & AC97_EI_VRA) { /* VRA */
+ val = snd_ac97_read(ac97, AC97_PCM_FRONT_DAC_RATE);
+ snd_iprintf(buffer, "PCM front DAC : %iHz\n", val);
+ if (ext & AC97_EI_SDAC) {
+ val = snd_ac97_read(ac97, AC97_PCM_SURR_DAC_RATE);
+ snd_iprintf(buffer, "PCM Surr DAC : %iHz\n", val);
+ }
+ if (ext & AC97_EI_LDAC) {
+ val = snd_ac97_read(ac97, AC97_PCM_LFE_DAC_RATE);
+ snd_iprintf(buffer, "PCM LFE DAC : %iHz\n", val);
+ }
+ val = snd_ac97_read(ac97, AC97_PCM_LR_ADC_RATE);
+ snd_iprintf(buffer, "PCM ADC : %iHz\n", val);
+ }
+ if (ext & AC97_EI_VRM) {
+ val = snd_ac97_read(ac97, AC97_PCM_MIC_ADC_RATE);
+ snd_iprintf(buffer, "PCM MIC ADC : %iHz\n", val);
+ }
+ if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF) ||
+ (ac97->id == AC97_ID_YMF743)) {
+ if (ac97->flags & AC97_CS_SPDIF)
+ val = snd_ac97_read(ac97, AC97_CSR_SPDIF);
+ else if (ac97->id == AC97_ID_YMF743) {
+ val = snd_ac97_read(ac97, AC97_YMF7X3_DIT_CTRL);
+ val = 0x2000 | (val & 0xff00) >> 4 | (val & 0x38) >> 2;
+ } else
+ val = snd_ac97_read(ac97, AC97_SPDIF);
+
+ snd_iprintf(buffer, "SPDIF Control :%s%s%s%s Category=0x%x Generation=%i%s%s%s\n",
+ val & AC97_SC_PRO ? " PRO" : " Consumer",
+ val & AC97_SC_NAUDIO ? " Non-audio" : " PCM",
+ val & AC97_SC_COPY ? "" : " Copyright",
+ val & AC97_SC_PRE ? " Preemph50/15" : "",
+ (val & AC97_SC_CC_MASK) >> AC97_SC_CC_SHIFT,
+ (val & AC97_SC_L) >> 11,
+ (ac97->flags & AC97_CS_SPDIF) ?
+ spdif_rates_cs4205[(val & AC97_SC_SPSR_MASK) >> AC97_SC_SPSR_SHIFT] :
+ spdif_rates[(val & AC97_SC_SPSR_MASK) >> AC97_SC_SPSR_SHIFT],
+ (ac97->flags & AC97_CS_SPDIF) ?
+ (val & AC97_SC_DRS ? " Validity" : "") :
+ (val & AC97_SC_DRS ? " DRS" : ""),
+ (ac97->flags & AC97_CS_SPDIF) ?
+ (val & AC97_SC_V ? " Enabled" : "") :
+ (val & AC97_SC_V ? " Validity" : ""));
+ /* ALC650 specific*/
+ if ((ac97->id & 0xfffffff0) == 0x414c4720 &&
+ (snd_ac97_read(ac97, AC97_ALC650_CLOCK) & 0x01)) {
+ val = snd_ac97_read(ac97, AC97_ALC650_SPDIF_INPUT_STATUS2);
+ if (val & AC97_ALC650_CLOCK_LOCK) {
+ val = snd_ac97_read(ac97, AC97_ALC650_SPDIF_INPUT_STATUS1);
+ snd_iprintf(buffer, "SPDIF In Status :%s%s%s%s Category=0x%x Generation=%i",
+ val & AC97_ALC650_PRO ? " PRO" : " Consumer",
+ val & AC97_ALC650_NAUDIO ? " Non-audio" : " PCM",
+ val & AC97_ALC650_COPY ? "" : " Copyright",
+ val & AC97_ALC650_PRE ? " Preemph50/15" : "",
+ (val & AC97_ALC650_CC_MASK) >> AC97_ALC650_CC_SHIFT,
+ (val & AC97_ALC650_L) >> 15);
+ val = snd_ac97_read(ac97, AC97_ALC650_SPDIF_INPUT_STATUS2);
+ snd_iprintf(buffer, "%s Accuracy=%i%s%s\n",
+ spdif_rates[(val & AC97_ALC650_SPSR_MASK) >> AC97_ALC650_SPSR_SHIFT],
+ (val & AC97_ALC650_CLOCK_ACCURACY) >> AC97_ALC650_CLOCK_SHIFT,
+ (val & AC97_ALC650_CLOCK_LOCK ? " Locked" : " Unlocked"),
+ (val & AC97_ALC650_V ? " Validity?" : ""));
+ } else {
+ snd_iprintf(buffer, "SPDIF In Status : Not Locked\n");
+ }
+ }
+ }
+ if ((ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23) {
+ val = snd_ac97_read(ac97, AC97_INT_PAGING);
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING,
+ AC97_PAGE_MASK, AC97_PAGE_1);
+ snd_ac97_proc_read_functions(ac97, buffer);
+ snd_ac97_update_bits(ac97, AC97_INT_PAGING,
+ AC97_PAGE_MASK, val & AC97_PAGE_MASK);
+ }
+
+
+ __modem:
+ mext = snd_ac97_read(ac97, AC97_EXTENDED_MID);
+ if (mext == 0)
+ return;
+
+ snd_iprintf(buffer, "Extended modem ID: codec=%i%s%s%s%s%s\n",
+ (mext & AC97_MEI_ADDR_MASK) >> AC97_MEI_ADDR_SHIFT,
+ mext & AC97_MEI_CID2 ? " CID2" : "",
+ mext & AC97_MEI_CID1 ? " CID1" : "",
+ mext & AC97_MEI_HANDSET ? " HSET" : "",
+ mext & AC97_MEI_LINE2 ? " LIN2" : "",
+ mext & AC97_MEI_LINE1 ? " LIN1" : "");
+ val = snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS);
+ snd_iprintf(buffer, "Modem status :%s%s%s%s%s%s%s%s%s%s%s%s%s%s%s%s\n",
+ val & AC97_MEA_GPIO ? " GPIO" : "",
+ val & AC97_MEA_MREF ? " MREF" : "",
+ val & AC97_MEA_ADC1 ? " ADC1" : "",
+ val & AC97_MEA_DAC1 ? " DAC1" : "",
+ val & AC97_MEA_ADC2 ? " ADC2" : "",
+ val & AC97_MEA_DAC2 ? " DAC2" : "",
+ val & AC97_MEA_HADC ? " HADC" : "",
+ val & AC97_MEA_HDAC ? " HDAC" : "",
+ val & AC97_MEA_PRA ? " PRA(GPIO)" : "",
+ val & AC97_MEA_PRB ? " PRB(res)" : "",
+ val & AC97_MEA_PRC ? " PRC(ADC1)" : "",
+ val & AC97_MEA_PRD ? " PRD(DAC1)" : "",
+ val & AC97_MEA_PRE ? " PRE(ADC2)" : "",
+ val & AC97_MEA_PRF ? " PRF(DAC2)" : "",
+ val & AC97_MEA_PRG ? " PRG(HADC)" : "",
+ val & AC97_MEA_PRH ? " PRH(HDAC)" : "");
+ if (mext & AC97_MEI_LINE1) {
+ val = snd_ac97_read(ac97, AC97_LINE1_RATE);
+ snd_iprintf(buffer, "Line1 rate : %iHz\n", val);
+ }
+ if (mext & AC97_MEI_LINE2) {
+ val = snd_ac97_read(ac97, AC97_LINE2_RATE);
+ snd_iprintf(buffer, "Line2 rate : %iHz\n", val);
+ }
+ if (mext & AC97_MEI_HANDSET) {
+ val = snd_ac97_read(ac97, AC97_HANDSET_RATE);
+ snd_iprintf(buffer, "Headset rate : %iHz\n", val);
+ }
+}
+
+static void snd_ac97_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_ac97 *ac97 = entry->private_data;
+
+ mutex_lock(&ac97->page_mutex);
+ if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { // Analog Devices AD1881/85/86
+ int idx;
+ for (idx = 0; idx < 3; idx++)
+ if (ac97->spec.ad18xx.id[idx]) {
+ /* select single codec */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000,
+ ac97->spec.ad18xx.unchained[idx] | ac97->spec.ad18xx.chained[idx]);
+ snd_ac97_proc_read_main(ac97, buffer, idx);
+ snd_iprintf(buffer, "\n\n");
+ }
+ /* select all codecs */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000);
+
+ snd_iprintf(buffer, "\nAD18XX configuration\n");
+ snd_iprintf(buffer, "Unchained : 0x%04x,0x%04x,0x%04x\n",
+ ac97->spec.ad18xx.unchained[0],
+ ac97->spec.ad18xx.unchained[1],
+ ac97->spec.ad18xx.unchained[2]);
+ snd_iprintf(buffer, "Chained : 0x%04x,0x%04x,0x%04x\n",
+ ac97->spec.ad18xx.chained[0],
+ ac97->spec.ad18xx.chained[1],
+ ac97->spec.ad18xx.chained[2]);
+ } else {
+ snd_ac97_proc_read_main(ac97, buffer, 0);
+ }
+ mutex_unlock(&ac97->page_mutex);
+}
+
+#ifdef CONFIG_SND_DEBUG
+/* direct register write for debugging */
+static void snd_ac97_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_ac97 *ac97 = entry->private_data;
+ char line[64];
+ unsigned int reg, val;
+ mutex_lock(&ac97->page_mutex);
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ /* register must be even */
+ if (reg < 0x80 && (reg & 1) == 0 && val <= 0xffff)
+ snd_ac97_write_cache(ac97, reg, val);
+ }
+ mutex_unlock(&ac97->page_mutex);
+}
+#endif
+
+static void snd_ac97_proc_regs_read_main(struct snd_ac97 *ac97, struct snd_info_buffer *buffer, int subidx)
+{
+ int reg, val;
+
+ for (reg = 0; reg < 0x80; reg += 2) {
+ val = snd_ac97_read(ac97, reg);
+ snd_iprintf(buffer, "%i:%02x = %04x\n", subidx, reg, val);
+ }
+}
+
+static void snd_ac97_proc_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ac97 *ac97 = entry->private_data;
+
+ mutex_lock(&ac97->page_mutex);
+ if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { // Analog Devices AD1881/85/86
+
+ int idx;
+ for (idx = 0; idx < 3; idx++)
+ if (ac97->spec.ad18xx.id[idx]) {
+ /* select single codec */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000,
+ ac97->spec.ad18xx.unchained[idx] | ac97->spec.ad18xx.chained[idx]);
+ snd_ac97_proc_regs_read_main(ac97, buffer, idx);
+ }
+ /* select all codecs */
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 0x7000, 0x7000);
+ } else {
+ snd_ac97_proc_regs_read_main(ac97, buffer, 0);
+ }
+ mutex_unlock(&ac97->page_mutex);
+}
+
+void snd_ac97_proc_init(struct snd_ac97 * ac97)
+{
+ struct snd_info_entry *entry;
+ char name[32];
+ const char *prefix;
+
+ if (ac97->bus->proc == NULL)
+ return;
+ prefix = ac97_is_audio(ac97) ? "ac97" : "mc97";
+ sprintf(name, "%s#%d-%d", prefix, ac97->addr, ac97->num);
+ if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) {
+ snd_info_set_text_ops(entry, ac97, snd_ac97_proc_read);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ac97->proc = entry;
+ sprintf(name, "%s#%d-%d+regs", prefix, ac97->addr, ac97->num);
+ if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) {
+ snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read);
+#ifdef CONFIG_SND_DEBUG
+ entry->mode |= S_IWUSR;
+ entry->c.text.write = snd_ac97_proc_regs_write;
+#endif
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ac97->proc_regs = entry;
+}
+
+void snd_ac97_proc_done(struct snd_ac97 * ac97)
+{
+ snd_info_free_entry(ac97->proc_regs);
+ ac97->proc_regs = NULL;
+ snd_info_free_entry(ac97->proc);
+ ac97->proc = NULL;
+}
+
+void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus)
+{
+ struct snd_info_entry *entry;
+ char name[32];
+
+ sprintf(name, "codec97#%d", bus->num);
+ if ((entry = snd_info_create_card_entry(bus->card, name, bus->card->proc_root)) != NULL) {
+ entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ bus->proc = entry;
+}
+
+void snd_ac97_bus_proc_done(struct snd_ac97_bus * bus)
+{
+ snd_info_free_entry(bus->proc);
+ bus->proc = NULL;
+}
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
new file mode 100644
index 0000000..a7f38e6
--- /dev/null
+++ b/sound/pci/ad1889.c
@@ -0,0 +1,1077 @@
+/* Analog Devices 1889 audio driver
+ *
+ * This is a driver for the AD1889 PCI audio chipset found
+ * on the HP PA-RISC [BCJ]-xxx0 workstations.
+ *
+ * Copyright (C) 2004-2005, Kyle McMartin <kyle@parisc-linux.org>
+ * Copyright (C) 2005, Thibaut Varene <varenet@parisc-linux.org>
+ * Based on the OSS AD1889 driver by Randolph Chung <tausq@debian.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * TODO:
+ * Do we need to take care of CCS register?
+ * Maybe we could use finer grained locking (separate locks for pb/cap)?
+ * Wishlist:
+ * Control Interface (mixer) support
+ * Better AC97 support (VSR...)?
+ * PM support
+ * MIDI support
+ * Game Port support
+ * SG DMA support (this will need *alot* of work)
+ */
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/compiler.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include <asm/io.h>
+
+#include "ad1889.h"
+#include "ac97/ac97_id.h"
+
+#define AD1889_DRVVER "Version: 1.7"
+
+MODULE_AUTHOR("Kyle McMartin <kyle@parisc-linux.org>, Thibaut Varene <t-bone@parisc-linux.org>");
+MODULE_DESCRIPTION("Analog Devices AD1889 ALSA sound driver");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Analog Devices,AD1889}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the AD1889 soundcard.");
+
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the AD1889 soundcard.");
+
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable AD1889 soundcard.");
+
+static char *ac97_quirk[SNDRV_CARDS];
+module_param_array(ac97_quirk, charp, NULL, 0444);
+MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
+
+#define DEVNAME "ad1889"
+#define PFX DEVNAME ": "
+
+/* let's use the global sound debug interfaces */
+#define ad1889_debug(fmt, arg...) snd_printd(KERN_DEBUG fmt, ## arg)
+
+/* keep track of some hw registers */
+struct ad1889_register_state {
+ u16 reg; /* reg setup */
+ u32 addr; /* dma base address */
+ unsigned long size; /* DMA buffer size */
+};
+
+struct snd_ad1889 {
+ struct snd_card *card;
+ struct pci_dev *pci;
+
+ int irq;
+ unsigned long bar;
+ void __iomem *iobase;
+
+ struct snd_ac97 *ac97;
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_pcm *pcm;
+ struct snd_info_entry *proc;
+
+ struct snd_pcm_substream *psubs;
+ struct snd_pcm_substream *csubs;
+
+ /* playback register state */
+ struct ad1889_register_state wave;
+ struct ad1889_register_state ramc;
+
+ spinlock_t lock;
+};
+
+static inline u16
+ad1889_readw(struct snd_ad1889 *chip, unsigned reg)
+{
+ return readw(chip->iobase + reg);
+}
+
+static inline void
+ad1889_writew(struct snd_ad1889 *chip, unsigned reg, u16 val)
+{
+ writew(val, chip->iobase + reg);
+}
+
+static inline u32
+ad1889_readl(struct snd_ad1889 *chip, unsigned reg)
+{
+ return readl(chip->iobase + reg);
+}
+
+static inline void
+ad1889_writel(struct snd_ad1889 *chip, unsigned reg, u32 val)
+{
+ writel(val, chip->iobase + reg);
+}
+
+static inline void
+ad1889_unmute(struct snd_ad1889 *chip)
+{
+ u16 st;
+ st = ad1889_readw(chip, AD_DS_WADA) &
+ ~(AD_DS_WADA_RWAM | AD_DS_WADA_LWAM);
+ ad1889_writew(chip, AD_DS_WADA, st);
+ ad1889_readw(chip, AD_DS_WADA);
+}
+
+static inline void
+ad1889_mute(struct snd_ad1889 *chip)
+{
+ u16 st;
+ st = ad1889_readw(chip, AD_DS_WADA) | AD_DS_WADA_RWAM | AD_DS_WADA_LWAM;
+ ad1889_writew(chip, AD_DS_WADA, st);
+ ad1889_readw(chip, AD_DS_WADA);
+}
+
+static inline void
+ad1889_load_adc_buffer_address(struct snd_ad1889 *chip, u32 address)
+{
+ ad1889_writel(chip, AD_DMA_ADCBA, address);
+ ad1889_writel(chip, AD_DMA_ADCCA, address);
+}
+
+static inline void
+ad1889_load_adc_buffer_count(struct snd_ad1889 *chip, u32 count)
+{
+ ad1889_writel(chip, AD_DMA_ADCBC, count);
+ ad1889_writel(chip, AD_DMA_ADCCC, count);
+}
+
+static inline void
+ad1889_load_adc_interrupt_count(struct snd_ad1889 *chip, u32 count)
+{
+ ad1889_writel(chip, AD_DMA_ADCIB, count);
+ ad1889_writel(chip, AD_DMA_ADCIC, count);
+}
+
+static inline void
+ad1889_load_wave_buffer_address(struct snd_ad1889 *chip, u32 address)
+{
+ ad1889_writel(chip, AD_DMA_WAVBA, address);
+ ad1889_writel(chip, AD_DMA_WAVCA, address);
+}
+
+static inline void
+ad1889_load_wave_buffer_count(struct snd_ad1889 *chip, u32 count)
+{
+ ad1889_writel(chip, AD_DMA_WAVBC, count);
+ ad1889_writel(chip, AD_DMA_WAVCC, count);
+}
+
+static inline void
+ad1889_load_wave_interrupt_count(struct snd_ad1889 *chip, u32 count)
+{
+ ad1889_writel(chip, AD_DMA_WAVIB, count);
+ ad1889_writel(chip, AD_DMA_WAVIC, count);
+}
+
+static void
+ad1889_channel_reset(struct snd_ad1889 *chip, unsigned int channel)
+{
+ u16 reg;
+
+ if (channel & AD_CHAN_WAV) {
+ /* Disable wave channel */
+ reg = ad1889_readw(chip, AD_DS_WSMC) & ~AD_DS_WSMC_WAEN;
+ ad1889_writew(chip, AD_DS_WSMC, reg);
+ chip->wave.reg = reg;
+
+ /* disable IRQs */
+ reg = ad1889_readw(chip, AD_DMA_WAV);
+ reg &= AD_DMA_IM_DIS;
+ reg &= ~AD_DMA_LOOP;
+ ad1889_writew(chip, AD_DMA_WAV, reg);
+
+ /* clear IRQ and address counters and pointers */
+ ad1889_load_wave_buffer_address(chip, 0x0);
+ ad1889_load_wave_buffer_count(chip, 0x0);
+ ad1889_load_wave_interrupt_count(chip, 0x0);
+
+ /* flush */
+ ad1889_readw(chip, AD_DMA_WAV);
+ }
+
+ if (channel & AD_CHAN_ADC) {
+ /* Disable ADC channel */
+ reg = ad1889_readw(chip, AD_DS_RAMC) & ~AD_DS_RAMC_ADEN;
+ ad1889_writew(chip, AD_DS_RAMC, reg);
+ chip->ramc.reg = reg;
+
+ reg = ad1889_readw(chip, AD_DMA_ADC);
+ reg &= AD_DMA_IM_DIS;
+ reg &= ~AD_DMA_LOOP;
+ ad1889_writew(chip, AD_DMA_ADC, reg);
+
+ ad1889_load_adc_buffer_address(chip, 0x0);
+ ad1889_load_adc_buffer_count(chip, 0x0);
+ ad1889_load_adc_interrupt_count(chip, 0x0);
+
+ /* flush */
+ ad1889_readw(chip, AD_DMA_ADC);
+ }
+}
+
+static u16
+snd_ad1889_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct snd_ad1889 *chip = ac97->private_data;
+ return ad1889_readw(chip, AD_AC97_BASE + reg);
+}
+
+static void
+snd_ad1889_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
+{
+ struct snd_ad1889 *chip = ac97->private_data;
+ ad1889_writew(chip, AD_AC97_BASE + reg, val);
+}
+
+static int
+snd_ad1889_ac97_ready(struct snd_ad1889 *chip)
+{
+ int retry = 400; /* average needs 352 msec */
+
+ while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY)
+ && --retry)
+ mdelay(1);
+ if (!retry) {
+ snd_printk(KERN_ERR PFX "[%s] Link is not ready.\n",
+ __func__);
+ return -EIO;
+ }
+ ad1889_debug("[%s] ready after %d ms\n", __func__, 400 - retry);
+
+ return 0;
+}
+
+static int
+snd_ad1889_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int
+snd_ad1889_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_pcm_hardware snd_ad1889_playback_hw = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000, /* docs say 7000, but we're lazy */
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = BUFFER_BYTES_MAX,
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = PERIOD_BYTES_MAX,
+ .periods_min = PERIODS_MIN,
+ .periods_max = PERIODS_MAX,
+ /*.fifo_size = 0,*/
+};
+
+static struct snd_pcm_hardware snd_ad1889_capture_hw = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000, /* docs say we could to VSR, but we're lazy */
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = BUFFER_BYTES_MAX,
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = PERIOD_BYTES_MAX,
+ .periods_min = PERIODS_MIN,
+ .periods_max = PERIODS_MAX,
+ /*.fifo_size = 0,*/
+};
+
+static int
+snd_ad1889_playback_open(struct snd_pcm_substream *ss)
+{
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+
+ chip->psubs = ss;
+ rt->hw = snd_ad1889_playback_hw;
+
+ return 0;
+}
+
+static int
+snd_ad1889_capture_open(struct snd_pcm_substream *ss)
+{
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+
+ chip->csubs = ss;
+ rt->hw = snd_ad1889_capture_hw;
+
+ return 0;
+}
+
+static int
+snd_ad1889_playback_close(struct snd_pcm_substream *ss)
+{
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+ chip->psubs = NULL;
+ return 0;
+}
+
+static int
+snd_ad1889_capture_close(struct snd_pcm_substream *ss)
+{
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+ chip->csubs = NULL;
+ return 0;
+}
+
+static int
+snd_ad1889_playback_prepare(struct snd_pcm_substream *ss)
+{
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(ss);
+ unsigned int count = snd_pcm_lib_period_bytes(ss);
+ u16 reg;
+
+ ad1889_channel_reset(chip, AD_CHAN_WAV);
+
+ reg = ad1889_readw(chip, AD_DS_WSMC);
+
+ /* Mask out 16-bit / Stereo */
+ reg &= ~(AD_DS_WSMC_WA16 | AD_DS_WSMC_WAST);
+
+ if (snd_pcm_format_width(rt->format) == 16)
+ reg |= AD_DS_WSMC_WA16;
+
+ if (rt->channels > 1)
+ reg |= AD_DS_WSMC_WAST;
+
+ /* let's make sure we don't clobber ourselves */
+ spin_lock_irq(&chip->lock);
+
+ chip->wave.size = size;
+ chip->wave.reg = reg;
+ chip->wave.addr = rt->dma_addr;
+
+ ad1889_writew(chip, AD_DS_WSMC, chip->wave.reg);
+
+ /* Set sample rates on the codec */
+ ad1889_writew(chip, AD_DS_WAS, rt->rate);
+
+ /* Set up DMA */
+ ad1889_load_wave_buffer_address(chip, chip->wave.addr);
+ ad1889_load_wave_buffer_count(chip, size);
+ ad1889_load_wave_interrupt_count(chip, count);
+
+ /* writes flush */
+ ad1889_readw(chip, AD_DS_WSMC);
+
+ spin_unlock_irq(&chip->lock);
+
+ ad1889_debug("prepare playback: addr = 0x%x, count = %u, "
+ "size = %u, reg = 0x%x, rate = %u\n", chip->wave.addr,
+ count, size, reg, rt->rate);
+ return 0;
+}
+
+static int
+snd_ad1889_capture_prepare(struct snd_pcm_substream *ss)
+{
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+ struct snd_pcm_runtime *rt = ss->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(ss);
+ unsigned int count = snd_pcm_lib_period_bytes(ss);
+ u16 reg;
+
+ ad1889_channel_reset(chip, AD_CHAN_ADC);
+
+ reg = ad1889_readw(chip, AD_DS_RAMC);
+
+ /* Mask out 16-bit / Stereo */
+ reg &= ~(AD_DS_RAMC_AD16 | AD_DS_RAMC_ADST);
+
+ if (snd_pcm_format_width(rt->format) == 16)
+ reg |= AD_DS_RAMC_AD16;
+
+ if (rt->channels > 1)
+ reg |= AD_DS_RAMC_ADST;
+
+ /* let's make sure we don't clobber ourselves */
+ spin_lock_irq(&chip->lock);
+
+ chip->ramc.size = size;
+ chip->ramc.reg = reg;
+ chip->ramc.addr = rt->dma_addr;
+
+ ad1889_writew(chip, AD_DS_RAMC, chip->ramc.reg);
+
+ /* Set up DMA */
+ ad1889_load_adc_buffer_address(chip, chip->ramc.addr);
+ ad1889_load_adc_buffer_count(chip, size);
+ ad1889_load_adc_interrupt_count(chip, count);
+
+ /* writes flush */
+ ad1889_readw(chip, AD_DS_RAMC);
+
+ spin_unlock_irq(&chip->lock);
+
+ ad1889_debug("prepare capture: addr = 0x%x, count = %u, "
+ "size = %u, reg = 0x%x, rate = %u\n", chip->ramc.addr,
+ count, size, reg, rt->rate);
+ return 0;
+}
+
+/* this is called in atomic context with IRQ disabled.
+ Must be as fast as possible and not sleep.
+ DMA should be *triggered* by this call.
+ The WSMC "WAEN" bit triggers DMA Wave On/Off */
+static int
+snd_ad1889_playback_trigger(struct snd_pcm_substream *ss, int cmd)
+{
+ u16 wsmc;
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+
+ wsmc = ad1889_readw(chip, AD_DS_WSMC);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* enable DMA loop & interrupts */
+ ad1889_writew(chip, AD_DMA_WAV, AD_DMA_LOOP | AD_DMA_IM_CNT);
+ wsmc |= AD_DS_WSMC_WAEN;
+ /* 1 to clear CHSS bit */
+ ad1889_writel(chip, AD_DMA_CHSS, AD_DMA_CHSS_WAVS);
+ ad1889_unmute(chip);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ ad1889_mute(chip);
+ wsmc &= ~AD_DS_WSMC_WAEN;
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+
+ chip->wave.reg = wsmc;
+ ad1889_writew(chip, AD_DS_WSMC, wsmc);
+ ad1889_readw(chip, AD_DS_WSMC); /* flush */
+
+ /* reset the chip when STOP - will disable IRQs */
+ if (cmd == SNDRV_PCM_TRIGGER_STOP)
+ ad1889_channel_reset(chip, AD_CHAN_WAV);
+
+ return 0;
+}
+
+/* this is called in atomic context with IRQ disabled.
+ Must be as fast as possible and not sleep.
+ DMA should be *triggered* by this call.
+ The RAMC "ADEN" bit triggers DMA ADC On/Off */
+static int
+snd_ad1889_capture_trigger(struct snd_pcm_substream *ss, int cmd)
+{
+ u16 ramc;
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+
+ ramc = ad1889_readw(chip, AD_DS_RAMC);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* enable DMA loop & interrupts */
+ ad1889_writew(chip, AD_DMA_ADC, AD_DMA_LOOP | AD_DMA_IM_CNT);
+ ramc |= AD_DS_RAMC_ADEN;
+ /* 1 to clear CHSS bit */
+ ad1889_writel(chip, AD_DMA_CHSS, AD_DMA_CHSS_ADCS);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ ramc &= ~AD_DS_RAMC_ADEN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ chip->ramc.reg = ramc;
+ ad1889_writew(chip, AD_DS_RAMC, ramc);
+ ad1889_readw(chip, AD_DS_RAMC); /* flush */
+
+ /* reset the chip when STOP - will disable IRQs */
+ if (cmd == SNDRV_PCM_TRIGGER_STOP)
+ ad1889_channel_reset(chip, AD_CHAN_ADC);
+
+ return 0;
+}
+
+/* Called in atomic context with IRQ disabled */
+static snd_pcm_uframes_t
+snd_ad1889_playback_pointer(struct snd_pcm_substream *ss)
+{
+ size_t ptr = 0;
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+
+ if (unlikely(!(chip->wave.reg & AD_DS_WSMC_WAEN)))
+ return 0;
+
+ ptr = ad1889_readl(chip, AD_DMA_WAVCA);
+ ptr -= chip->wave.addr;
+
+ if (snd_BUG_ON(ptr >= chip->wave.size))
+ return 0;
+
+ return bytes_to_frames(ss->runtime, ptr);
+}
+
+/* Called in atomic context with IRQ disabled */
+static snd_pcm_uframes_t
+snd_ad1889_capture_pointer(struct snd_pcm_substream *ss)
+{
+ size_t ptr = 0;
+ struct snd_ad1889 *chip = snd_pcm_substream_chip(ss);
+
+ if (unlikely(!(chip->ramc.reg & AD_DS_RAMC_ADEN)))
+ return 0;
+
+ ptr = ad1889_readl(chip, AD_DMA_ADCCA);
+ ptr -= chip->ramc.addr;
+
+ if (snd_BUG_ON(ptr >= chip->ramc.size))
+ return 0;
+
+ return bytes_to_frames(ss->runtime, ptr);
+}
+
+static struct snd_pcm_ops snd_ad1889_playback_ops = {
+ .open = snd_ad1889_playback_open,
+ .close = snd_ad1889_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ad1889_hw_params,
+ .hw_free = snd_ad1889_hw_free,
+ .prepare = snd_ad1889_playback_prepare,
+ .trigger = snd_ad1889_playback_trigger,
+ .pointer = snd_ad1889_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_ad1889_capture_ops = {
+ .open = snd_ad1889_capture_open,
+ .close = snd_ad1889_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ad1889_hw_params,
+ .hw_free = snd_ad1889_hw_free,
+ .prepare = snd_ad1889_capture_prepare,
+ .trigger = snd_ad1889_capture_trigger,
+ .pointer = snd_ad1889_capture_pointer,
+};
+
+static irqreturn_t
+snd_ad1889_interrupt(int irq, void *dev_id)
+{
+ unsigned long st;
+ struct snd_ad1889 *chip = dev_id;
+
+ st = ad1889_readl(chip, AD_DMA_DISR);
+
+ /* clear ISR */
+ ad1889_writel(chip, AD_DMA_DISR, st);
+
+ st &= AD_INTR_MASK;
+
+ if (unlikely(!st))
+ return IRQ_NONE;
+
+ if (st & (AD_DMA_DISR_PMAI|AD_DMA_DISR_PTAI))
+ ad1889_debug("Unexpected master or target abort interrupt!\n");
+
+ if ((st & AD_DMA_DISR_WAVI) && chip->psubs)
+ snd_pcm_period_elapsed(chip->psubs);
+ if ((st & AD_DMA_DISR_ADCI) && chip->csubs)
+ snd_pcm_period_elapsed(chip->csubs);
+
+ return IRQ_HANDLED;
+}
+
+static int __devinit
+snd_ad1889_pcm_init(struct snd_ad1889 *chip, int device, struct snd_pcm **rpcm)
+{
+ int err;
+ struct snd_pcm *pcm;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ err = snd_pcm_new(chip->card, chip->card->driver, device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_ad1889_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_ad1889_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, chip->card->shortname);
+
+ chip->pcm = pcm;
+ chip->psubs = NULL;
+ chip->csubs = NULL;
+
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ BUFFER_BYTES_MAX / 2,
+ BUFFER_BYTES_MAX);
+
+ if (err < 0) {
+ snd_printk(KERN_ERR PFX "buffer allocation error: %d\n", err);
+ return err;
+ }
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+static void
+snd_ad1889_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_ad1889 *chip = entry->private_data;
+ u16 reg;
+ int tmp;
+
+ reg = ad1889_readw(chip, AD_DS_WSMC);
+ snd_iprintf(buffer, "Wave output: %s\n",
+ (reg & AD_DS_WSMC_WAEN) ? "enabled" : "disabled");
+ snd_iprintf(buffer, "Wave Channels: %s\n",
+ (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono");
+ snd_iprintf(buffer, "Wave Quality: %d-bit linear\n",
+ (reg & AD_DS_WSMC_WA16) ? 16 : 8);
+
+ /* WARQ is at offset 12 */
+ tmp = (reg & AD_DS_WSMC_WARQ) ?
+ (((reg & AD_DS_WSMC_WARQ >> 12) & 0x01) ? 12 : 18) : 4;
+ tmp /= (reg & AD_DS_WSMC_WAST) ? 2 : 1;
+
+ snd_iprintf(buffer, "Wave FIFO: %d %s words\n\n", tmp,
+ (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono");
+
+
+ snd_iprintf(buffer, "Synthesis output: %s\n",
+ reg & AD_DS_WSMC_SYEN ? "enabled" : "disabled");
+
+ /* SYRQ is at offset 4 */
+ tmp = (reg & AD_DS_WSMC_SYRQ) ?
+ (((reg & AD_DS_WSMC_SYRQ >> 4) & 0x01) ? 12 : 18) : 4;
+ tmp /= (reg & AD_DS_WSMC_WAST) ? 2 : 1;
+
+ snd_iprintf(buffer, "Synthesis FIFO: %d %s words\n\n", tmp,
+ (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono");
+
+ reg = ad1889_readw(chip, AD_DS_RAMC);
+ snd_iprintf(buffer, "ADC input: %s\n",
+ (reg & AD_DS_RAMC_ADEN) ? "enabled" : "disabled");
+ snd_iprintf(buffer, "ADC Channels: %s\n",
+ (reg & AD_DS_RAMC_ADST) ? "stereo" : "mono");
+ snd_iprintf(buffer, "ADC Quality: %d-bit linear\n",
+ (reg & AD_DS_RAMC_AD16) ? 16 : 8);
+
+ /* ACRQ is at offset 4 */
+ tmp = (reg & AD_DS_RAMC_ACRQ) ?
+ (((reg & AD_DS_RAMC_ACRQ >> 4) & 0x01) ? 12 : 18) : 4;
+ tmp /= (reg & AD_DS_RAMC_ADST) ? 2 : 1;
+
+ snd_iprintf(buffer, "ADC FIFO: %d %s words\n\n", tmp,
+ (reg & AD_DS_RAMC_ADST) ? "stereo" : "mono");
+
+ snd_iprintf(buffer, "Resampler input: %s\n",
+ reg & AD_DS_RAMC_REEN ? "enabled" : "disabled");
+
+ /* RERQ is at offset 12 */
+ tmp = (reg & AD_DS_RAMC_RERQ) ?
+ (((reg & AD_DS_RAMC_RERQ >> 12) & 0x01) ? 12 : 18) : 4;
+ tmp /= (reg & AD_DS_RAMC_ADST) ? 2 : 1;
+
+ snd_iprintf(buffer, "Resampler FIFO: %d %s words\n\n", tmp,
+ (reg & AD_DS_WSMC_WAST) ? "stereo" : "mono");
+
+
+ /* doc says LSB represents -1.5dB, but the max value (-94.5dB)
+ suggests that LSB is -3dB, which is more coherent with the logarithmic
+ nature of the dB scale */
+ reg = ad1889_readw(chip, AD_DS_WADA);
+ snd_iprintf(buffer, "Left: %s, -%d dB\n",
+ (reg & AD_DS_WADA_LWAM) ? "mute" : "unmute",
+ ((reg & AD_DS_WADA_LWAA) >> 8) * 3);
+ reg = ad1889_readw(chip, AD_DS_WADA);
+ snd_iprintf(buffer, "Right: %s, -%d dB\n",
+ (reg & AD_DS_WADA_RWAM) ? "mute" : "unmute",
+ ((reg & AD_DS_WADA_RWAA) >> 8) * 3);
+
+ reg = ad1889_readw(chip, AD_DS_WAS);
+ snd_iprintf(buffer, "Wave samplerate: %u Hz\n", reg);
+ reg = ad1889_readw(chip, AD_DS_RES);
+ snd_iprintf(buffer, "Resampler samplerate: %u Hz\n", reg);
+}
+
+static void __devinit
+snd_ad1889_proc_init(struct snd_ad1889 *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(chip->card, chip->card->driver, &entry))
+ snd_info_set_text_ops(entry, chip, snd_ad1889_proc_read);
+}
+
+static struct ac97_quirk ac97_quirks[] = {
+ {
+ .subvendor = 0x11d4, /* AD */
+ .subdevice = 0x1889, /* AD1889 */
+ .codec_id = AC97_ID_AD1819,
+ .name = "AD1889",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ { } /* terminator */
+};
+
+static void __devinit
+snd_ad1889_ac97_xinit(struct snd_ad1889 *chip)
+{
+ u16 reg;
+
+ reg = ad1889_readw(chip, AD_AC97_ACIC);
+ reg |= AD_AC97_ACIC_ACRD; /* Reset Disable */
+ ad1889_writew(chip, AD_AC97_ACIC, reg);
+ ad1889_readw(chip, AD_AC97_ACIC); /* flush posted write */
+ udelay(10);
+ /* Interface Enable */
+ reg |= AD_AC97_ACIC_ACIE;
+ ad1889_writew(chip, AD_AC97_ACIC, reg);
+
+ snd_ad1889_ac97_ready(chip);
+
+ /* Audio Stream Output | Variable Sample Rate Mode */
+ reg = ad1889_readw(chip, AD_AC97_ACIC);
+ reg |= AD_AC97_ACIC_ASOE | AD_AC97_ACIC_VSRM;
+ ad1889_writew(chip, AD_AC97_ACIC, reg);
+ ad1889_readw(chip, AD_AC97_ACIC); /* flush posted write */
+
+}
+
+static void
+snd_ad1889_ac97_bus_free(struct snd_ac97_bus *bus)
+{
+ struct snd_ad1889 *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void
+snd_ad1889_ac97_free(struct snd_ac97 *ac97)
+{
+ struct snd_ad1889 *chip = ac97->private_data;
+ chip->ac97 = NULL;
+}
+
+static int __devinit
+snd_ad1889_ac97_init(struct snd_ad1889 *chip, const char *quirk_override)
+{
+ int err;
+ struct snd_ac97_template ac97;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_ad1889_ac97_write,
+ .read = snd_ad1889_ac97_read,
+ };
+
+ /* doing that here, it works. */
+ snd_ad1889_ac97_xinit(chip);
+
+ err = snd_ac97_bus(chip->card, 0, &ops, chip, &chip->ac97_bus);
+ if (err < 0)
+ return err;
+
+ chip->ac97_bus->private_free = snd_ad1889_ac97_bus_free;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_ad1889_ac97_free;
+ ac97.pci = chip->pci;
+
+ err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97);
+ if (err < 0)
+ return err;
+
+ snd_ac97_tune_hardware(chip->ac97, ac97_quirks, quirk_override);
+
+ return 0;
+}
+
+static int
+snd_ad1889_free(struct snd_ad1889 *chip)
+{
+ if (chip->irq < 0)
+ goto skip_hw;
+
+ spin_lock_irq(&chip->lock);
+
+ ad1889_mute(chip);
+
+ /* Turn off interrupt on count and zero DMA registers */
+ ad1889_channel_reset(chip, AD_CHAN_WAV | AD_CHAN_ADC);
+
+ /* clear DISR. If we don't, we'd better jump off the Eiffel Tower */
+ ad1889_writel(chip, AD_DMA_DISR, AD_DMA_DISR_PTAI | AD_DMA_DISR_PMAI);
+ ad1889_readl(chip, AD_DMA_DISR); /* flush, dammit! */
+
+ spin_unlock_irq(&chip->lock);
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+
+skip_hw:
+ if (chip->iobase)
+ iounmap(chip->iobase);
+
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+
+ kfree(chip);
+ return 0;
+}
+
+static int
+snd_ad1889_dev_free(struct snd_device *device)
+{
+ struct snd_ad1889 *chip = device->device_data;
+ return snd_ad1889_free(chip);
+}
+
+static int __devinit
+snd_ad1889_init(struct snd_ad1889 *chip)
+{
+ ad1889_writew(chip, AD_DS_CCS, AD_DS_CCS_CLKEN); /* turn on clock */
+ ad1889_readw(chip, AD_DS_CCS); /* flush posted write */
+
+ mdelay(10);
+
+ /* enable Master and Target abort interrupts */
+ ad1889_writel(chip, AD_DMA_DISR, AD_DMA_DISR_PMAE | AD_DMA_DISR_PTAE);
+
+ return 0;
+}
+
+static int __devinit
+snd_ad1889_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct snd_ad1889 **rchip)
+{
+ int err;
+
+ struct snd_ad1889 *chip;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ad1889_dev_free,
+ };
+
+ *rchip = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ /* check PCI availability (32bit DMA) */
+ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
+ printk(KERN_ERR PFX "error setting 32-bit DMA mask.\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ /* allocate chip specific data with zero-filled memory */
+ if ((chip = kzalloc(sizeof(*chip), GFP_KERNEL)) == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ card->private_data = chip;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* (1) PCI resource allocation */
+ if ((err = pci_request_regions(pci, card->driver)) < 0)
+ goto free_and_ret;
+
+ chip->bar = pci_resource_start(pci, 0);
+ chip->iobase = pci_ioremap_bar(pci, 0);
+ if (chip->iobase == NULL) {
+ printk(KERN_ERR PFX "unable to reserve region.\n");
+ err = -EBUSY;
+ goto free_and_ret;
+ }
+
+ pci_set_master(pci);
+
+ spin_lock_init(&chip->lock); /* only now can we call ad1889_free */
+
+ if (request_irq(pci->irq, snd_ad1889_interrupt,
+ IRQF_SHARED, card->driver, chip)) {
+ printk(KERN_ERR PFX "cannot obtain IRQ %d\n", pci->irq);
+ snd_ad1889_free(chip);
+ return -EBUSY;
+ }
+
+ chip->irq = pci->irq;
+ synchronize_irq(chip->irq);
+
+ /* (2) initialization of the chip hardware */
+ if ((err = snd_ad1889_init(chip)) < 0) {
+ snd_ad1889_free(chip);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_ad1889_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+
+ return 0;
+
+free_and_ret:
+ kfree(chip);
+ pci_disable_device(pci);
+
+ return err;
+}
+
+static int __devinit
+snd_ad1889_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ int err;
+ static int devno;
+ struct snd_card *card;
+ struct snd_ad1889 *chip;
+
+ /* (1) */
+ if (devno >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[devno]) {
+ devno++;
+ return -ENOENT;
+ }
+
+ /* (2) */
+ card = snd_card_new(index[devno], id[devno], THIS_MODULE, 0);
+ /* XXX REVISIT: we can probably allocate chip in this call */
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "AD1889");
+ strcpy(card->shortname, "Analog Devices AD1889");
+
+ /* (3) */
+ err = snd_ad1889_create(card, pci, &chip);
+ if (err < 0)
+ goto free_and_ret;
+
+ /* (4) */
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->bar, chip->irq);
+
+ /* (5) */
+ /* register AC97 mixer */
+ err = snd_ad1889_ac97_init(chip, ac97_quirk[devno]);
+ if (err < 0)
+ goto free_and_ret;
+
+ err = snd_ad1889_pcm_init(chip, 0, NULL);
+ if (err < 0)
+ goto free_and_ret;
+
+ /* register proc interface */
+ snd_ad1889_proc_init(chip);
+
+ /* (6) */
+ err = snd_card_register(card);
+ if (err < 0)
+ goto free_and_ret;
+
+ /* (7) */
+ pci_set_drvdata(pci, card);
+
+ devno++;
+ return 0;
+
+free_and_ret:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit
+snd_ad1889_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_device_id snd_ad1889_ids[] = {
+ { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) },
+ { 0, },
+};
+MODULE_DEVICE_TABLE(pci, snd_ad1889_ids);
+
+static struct pci_driver ad1889_pci_driver = {
+ .name = "AD1889 Audio",
+ .id_table = snd_ad1889_ids,
+ .probe = snd_ad1889_probe,
+ .remove = __devexit_p(snd_ad1889_remove),
+};
+
+static int __init
+alsa_ad1889_init(void)
+{
+ return pci_register_driver(&ad1889_pci_driver);
+}
+
+static void __exit
+alsa_ad1889_fini(void)
+{
+ pci_unregister_driver(&ad1889_pci_driver);
+}
+
+module_init(alsa_ad1889_init);
+module_exit(alsa_ad1889_fini);
diff --git a/sound/pci/ad1889.h b/sound/pci/ad1889.h
new file mode 100644
index 0000000..5e6dad5
--- /dev/null
+++ b/sound/pci/ad1889.h
@@ -0,0 +1,189 @@
+/* Analog Devices 1889 audio driver
+ * Copyright (C) 2004, Kyle McMartin <kyle@parisc-linux.org>
+ */
+
+#ifndef __AD1889_H__
+#define __AD1889_H__
+
+#define AD_DS_WSMC 0x00 /* wave/synthesis channel mixer control */
+#define AD_DS_WSMC_SYEN 0x0004 /* synthesis channel enable */
+#define AD_DS_WSMC_SYRQ 0x0030 /* synth. fifo request point */
+#define AD_DS_WSMC_WA16 0x0100 /* wave channel 16bit select */
+#define AD_DS_WSMC_WAST 0x0200 /* wave channel stereo select */
+#define AD_DS_WSMC_WAEN 0x0400 /* wave channel enable */
+#define AD_DS_WSMC_WARQ 0x3000 /* wave fifo request point */
+
+#define AD_DS_RAMC 0x02 /* resampler/ADC channel mixer control */
+#define AD_DS_RAMC_AD16 0x0001 /* ADC channel 16bit select */
+#define AD_DS_RAMC_ADST 0x0002 /* ADC channel stereo select */
+#define AD_DS_RAMC_ADEN 0x0004 /* ADC channel enable */
+#define AD_DS_RAMC_ACRQ 0x0030 /* ADC fifo request point */
+#define AD_DS_RAMC_REEN 0x0400 /* resampler channel enable */
+#define AD_DS_RAMC_RERQ 0x3000 /* res. fifo request point */
+
+#define AD_DS_WADA 0x04 /* wave channel mix attenuation */
+#define AD_DS_WADA_RWAM 0x0080 /* right wave mute */
+#define AD_DS_WADA_RWAA 0x001f /* right wave attenuation */
+#define AD_DS_WADA_LWAM 0x8000 /* left wave mute */
+#define AD_DS_WADA_LWAA 0x3e00 /* left wave attenuation */
+
+#define AD_DS_SYDA 0x06 /* synthesis channel mix attenuation */
+#define AD_DS_SYDA_RSYM 0x0080 /* right synthesis mute */
+#define AD_DS_SYDA_RSYA 0x001f /* right synthesis attenuation */
+#define AD_DS_SYDA_LSYM 0x8000 /* left synthesis mute */
+#define AD_DS_SYDA_LSYA 0x3e00 /* left synthesis attenuation */
+
+#define AD_DS_WAS 0x08 /* wave channel sample rate */
+#define AD_DS_WAS_WAS 0xffff /* sample rate mask */
+
+#define AD_DS_RES 0x0a /* resampler channel sample rate */
+#define AD_DS_RES_RES 0xffff /* sample rate mask */
+
+#define AD_DS_CCS 0x0c /* chip control/status */
+#define AD_DS_CCS_ADO 0x0001 /* ADC channel overflow */
+#define AD_DS_CCS_REO 0x0002 /* resampler channel overflow */
+#define AD_DS_CCS_SYU 0x0004 /* synthesis channel underflow */
+#define AD_DS_CCS_WAU 0x0008 /* wave channel underflow */
+/* bits 4 -> 7, 9, 11 -> 14 reserved */
+#define AD_DS_CCS_XTD 0x0100 /* xtd delay control (4096 clock cycles) */
+#define AD_DS_CCS_PDALL 0x0400 /* power */
+#define AD_DS_CCS_CLKEN 0x8000 /* clock */
+
+#define AD_DMA_RESBA 0x40 /* RES base address */
+#define AD_DMA_RESCA 0x44 /* RES current address */
+#define AD_DMA_RESBC 0x48 /* RES base count */
+#define AD_DMA_RESCC 0x4c /* RES current count */
+
+#define AD_DMA_ADCBA 0x50 /* ADC base address */
+#define AD_DMA_ADCCA 0x54 /* ADC current address */
+#define AD_DMA_ADCBC 0x58 /* ADC base count */
+#define AD_DMA_ADCCC 0x5c /* ADC current count */
+
+#define AD_DMA_SYNBA 0x60 /* synth base address */
+#define AD_DMA_SYNCA 0x64 /* synth current address */
+#define AD_DMA_SYNBC 0x68 /* synth base count */
+#define AD_DMA_SYNCC 0x6c /* synth current count */
+
+#define AD_DMA_WAVBA 0x70 /* wave base address */
+#define AD_DMA_WAVCA 0x74 /* wave current address */
+#define AD_DMA_WAVBC 0x78 /* wave base count */
+#define AD_DMA_WAVCC 0x7c /* wave current count */
+
+#define AD_DMA_RESIC 0x80 /* RES dma interrupt current byte count */
+#define AD_DMA_RESIB 0x84 /* RES dma interrupt base byte count */
+
+#define AD_DMA_ADCIC 0x88 /* ADC dma interrupt current byte count */
+#define AD_DMA_ADCIB 0x8c /* ADC dma interrupt base byte count */
+
+#define AD_DMA_SYNIC 0x90 /* synth dma interrupt current byte count */
+#define AD_DMA_SYNIB 0x94 /* synth dma interrupt base byte count */
+
+#define AD_DMA_WAVIC 0x98 /* wave dma interrupt current byte count */
+#define AD_DMA_WAVIB 0x9c /* wave dma interrupt base byte count */
+
+#define AD_DMA_ICC 0xffffff /* current byte count mask */
+#define AD_DMA_IBC 0xffffff /* base byte count mask */
+/* bits 24 -> 31 reserved */
+
+/* 4 bytes pad */
+#define AD_DMA_ADC 0xa8 /* ADC dma control and status */
+#define AD_DMA_SYNTH 0xb0 /* Synth dma control and status */
+#define AD_DMA_WAV 0xb8 /* wave dma control and status */
+#define AD_DMA_RES 0xa0 /* Resample dma control and status */
+
+#define AD_DMA_SGDE 0x0001 /* SGD mode enable */
+#define AD_DMA_LOOP 0x0002 /* loop enable */
+#define AD_DMA_IM 0x000c /* interrupt mode mask */
+#define AD_DMA_IM_DIS (~AD_DMA_IM) /* disable */
+#define AD_DMA_IM_CNT 0x0004 /* interrupt on count */
+#define AD_DMA_IM_SGD 0x0008 /* interrupt on SGD flag */
+#define AD_DMA_IM_EOL 0x000c /* interrupt on End of Linked List */
+#define AD_DMA_SGDS 0x0030 /* SGD status */
+#define AD_DMA_SFLG 0x0040 /* SGD flag */
+#define AD_DMA_EOL 0x0080 /* SGD end of list */
+/* bits 8 -> 15 reserved */
+
+#define AD_DMA_DISR 0xc0 /* dma interrupt status */
+#define AD_DMA_DISR_RESI 0x000001 /* resampler channel interrupt */
+#define AD_DMA_DISR_ADCI 0x000002 /* ADC channel interrupt */
+#define AD_DMA_DISR_SYNI 0x000004 /* synthesis channel interrupt */
+#define AD_DMA_DISR_WAVI 0x000008 /* wave channel interrupt */
+/* bits 4, 5 reserved */
+#define AD_DMA_DISR_SEPS 0x000040 /* serial eeprom status */
+/* bits 7 -> 13 reserved */
+#define AD_DMA_DISR_PMAI 0x004000 /* pci master abort interrupt */
+#define AD_DMA_DISR_PTAI 0x008000 /* pci target abort interrupt */
+#define AD_DMA_DISR_PTAE 0x010000 /* pci target abort interrupt enable */
+#define AD_DMA_DISR_PMAE 0x020000 /* pci master abort interrupt enable */
+/* bits 19 -> 31 reserved */
+
+/* interrupt mask */
+#define AD_INTR_MASK (AD_DMA_DISR_RESI|AD_DMA_DISR_ADCI| \
+ AD_DMA_DISR_WAVI|AD_DMA_DISR_SYNI| \
+ AD_DMA_DISR_PMAI|AD_DMA_DISR_PTAI)
+
+#define AD_DMA_CHSS 0xc4 /* dma channel stop status */
+#define AD_DMA_CHSS_RESS 0x000001 /* resampler channel stopped */
+#define AD_DMA_CHSS_ADCS 0x000002 /* ADC channel stopped */
+#define AD_DMA_CHSS_SYNS 0x000004 /* synthesis channel stopped */
+#define AD_DMA_CHSS_WAVS 0x000008 /* wave channel stopped */
+
+#define AD_GPIO_IPC 0xc8 /* gpio port control */
+#define AD_GPIO_OP 0xca /* gpio output port status */
+#define AD_GPIO_IP 0xcc /* gpio input port status */
+
+#define AD_AC97_BASE 0x100 /* ac97 base register */
+
+#define AD_AC97_RESET 0x100 /* reset */
+
+#define AD_AC97_PWR_CTL 0x126 /* == AC97_POWERDOWN */
+#define AD_AC97_PWR_ADC 0x0001 /* ADC ready status */
+#define AD_AC97_PWR_DAC 0x0002 /* DAC ready status */
+#define AD_AC97_PWR_PR0 0x0100 /* PR0 (ADC) powerdown */
+#define AD_AC97_PWR_PR1 0x0200 /* PR1 (DAC) powerdown */
+
+#define AD_MISC_CTL 0x176 /* misc control */
+#define AD_MISC_CTL_DACZ 0x8000 /* set for zero fill, unset for repeat */
+#define AD_MISC_CTL_ARSR 0x0001 /* set for SR1, unset for SR0 */
+#define AD_MISC_CTL_ALSR 0x0100
+#define AD_MISC_CTL_DLSR 0x0400
+#define AD_MISC_CTL_DRSR 0x0004
+
+#define AD_AC97_SR0 0x178 /* sample rate 0, 0xbb80 == 48K */
+#define AD_AC97_SR0_48K 0xbb80 /* 48KHz */
+#define AD_AC97_SR1 0x17a /* sample rate 1 */
+
+#define AD_AC97_ACIC 0x180 /* ac97 codec interface control */
+#define AD_AC97_ACIC_ACIE 0x0001 /* analog codec interface enable */
+#define AD_AC97_ACIC_ACRD 0x0002 /* analog codec reset disable */
+#define AD_AC97_ACIC_ASOE 0x0004 /* audio stream output enable */
+#define AD_AC97_ACIC_VSRM 0x0008 /* variable sample rate mode */
+#define AD_AC97_ACIC_FSDH 0x0100 /* force SDATA_OUT high */
+#define AD_AC97_ACIC_FSYH 0x0200 /* force sync high */
+#define AD_AC97_ACIC_ACRDY 0x8000 /* analog codec ready status */
+/* bits 10 -> 14 reserved */
+
+
+#define AD_DS_MEMSIZE 512
+#define AD_OPL_MEMSIZE 16
+#define AD_MIDI_MEMSIZE 16
+
+#define AD_WAV_STATE 0
+#define AD_ADC_STATE 1
+#define AD_MAX_STATES 2
+
+#define AD_CHAN_WAV 0x0001
+#define AD_CHAN_ADC 0x0002
+#define AD_CHAN_RES 0x0004
+#define AD_CHAN_SYN 0x0008
+
+
+/* The chip would support 4 GB buffers and 16 MB periods,
+ * but let's not overdo it ... */
+#define BUFFER_BYTES_MAX (256 * 1024)
+#define PERIOD_BYTES_MIN 32
+#define PERIOD_BYTES_MAX (BUFFER_BYTES_MAX / 2)
+#define PERIODS_MIN 2
+#define PERIODS_MAX (BUFFER_BYTES_MAX / PERIOD_BYTES_MIN)
+
+#endif /* __AD1889_H__ */
diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c
new file mode 100644
index 0000000..0f819dd
--- /dev/null
+++ b/sound/pci/ak4531_codec.c
@@ -0,0 +1,492 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Universal routines for AK4531 codec
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/ak4531_codec.h>
+#include <sound/tlv.h>
+
+/*
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Universal routines for AK4531 codec");
+MODULE_LICENSE("GPL");
+*/
+
+#ifdef CONFIG_PROC_FS
+static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531);
+#else
+#define snd_ak4531_proc_init(card,ak)
+#endif
+
+/*
+ *
+ */
+
+#if 0
+
+static void snd_ak4531_dump(struct snd_ak4531 *ak4531)
+{
+ int idx;
+
+ for (idx = 0; idx < 0x19; idx++)
+ printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]);
+}
+
+#endif
+
+/*
+ *
+ */
+
+#define AK4531_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_ak4531_info_single, \
+ .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \
+ .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22) }
+#define AK4531_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .index = xindex, \
+ .info = snd_ak4531_info_single, \
+ .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \
+ .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = (xtlv) } }
+
+static int snd_ak4531_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_ak4531_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 16) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int val;
+
+ mutex_lock(&ak4531->reg_mutex);
+ val = (ak4531->regs[reg] >> shift) & mask;
+ mutex_unlock(&ak4531->reg_mutex);
+ if (invert) {
+ val = mask - val;
+ }
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+}
+
+static int snd_ak4531_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 16) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ int val;
+
+ val = ucontrol->value.integer.value[0] & mask;
+ if (invert) {
+ val = mask - val;
+ }
+ val <<= shift;
+ mutex_lock(&ak4531->reg_mutex);
+ val = (ak4531->regs[reg] & ~(mask << shift)) | val;
+ change = val != ak4531->regs[reg];
+ ak4531->write(ak4531, reg, ak4531->regs[reg] = val);
+ mutex_unlock(&ak4531->reg_mutex);
+ return change;
+}
+
+#define AK4531_DOUBLE(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_ak4531_info_double, \
+ .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22) }
+#define AK4531_DOUBLE_TLV(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .index = xindex, \
+ .info = snd_ak4531_info_double, \
+ .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = (xtlv) } }
+
+static int snd_ak4531_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_ak4531_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x07;
+ int right_shift = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int left, right;
+
+ mutex_lock(&ak4531->reg_mutex);
+ left = (ak4531->regs[left_reg] >> left_shift) & mask;
+ right = (ak4531->regs[right_reg] >> right_shift) & mask;
+ mutex_unlock(&ak4531->reg_mutex);
+ if (invert) {
+ left = mask - left;
+ right = mask - right;
+ }
+ ucontrol->value.integer.value[0] = left;
+ ucontrol->value.integer.value[1] = right;
+ return 0;
+}
+
+static int snd_ak4531_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x07;
+ int right_shift = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ int left, right;
+
+ left = ucontrol->value.integer.value[0] & mask;
+ right = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ left = mask - left;
+ right = mask - right;
+ }
+ left <<= left_shift;
+ right <<= right_shift;
+ mutex_lock(&ak4531->reg_mutex);
+ if (left_reg == right_reg) {
+ left = (ak4531->regs[left_reg] & ~((mask << left_shift) | (mask << right_shift))) | left | right;
+ change = left != ak4531->regs[left_reg];
+ ak4531->write(ak4531, left_reg, ak4531->regs[left_reg] = left);
+ } else {
+ left = (ak4531->regs[left_reg] & ~(mask << left_shift)) | left;
+ right = (ak4531->regs[right_reg] & ~(mask << right_shift)) | right;
+ change = left != ak4531->regs[left_reg] || right != ak4531->regs[right_reg];
+ ak4531->write(ak4531, left_reg, ak4531->regs[left_reg] = left);
+ ak4531->write(ak4531, right_reg, ak4531->regs[right_reg] = right);
+ }
+ mutex_unlock(&ak4531->reg_mutex);
+ return change;
+}
+
+#define AK4531_INPUT_SW(xname, xindex, reg1, reg2, left_shift, right_shift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_ak4531_info_input_sw, \
+ .get = snd_ak4531_get_input_sw, .put = snd_ak4531_put_input_sw, \
+ .private_value = reg1 | (reg2 << 8) | (left_shift << 16) | (right_shift << 24) }
+
+static int snd_ak4531_info_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 4;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_ak4531_get_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol);
+ int reg1 = kcontrol->private_value & 0xff;
+ int reg2 = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x0f;
+ int right_shift = (kcontrol->private_value >> 24) & 0x0f;
+
+ mutex_lock(&ak4531->reg_mutex);
+ ucontrol->value.integer.value[0] = (ak4531->regs[reg1] >> left_shift) & 1;
+ ucontrol->value.integer.value[1] = (ak4531->regs[reg2] >> left_shift) & 1;
+ ucontrol->value.integer.value[2] = (ak4531->regs[reg1] >> right_shift) & 1;
+ ucontrol->value.integer.value[3] = (ak4531->regs[reg2] >> right_shift) & 1;
+ mutex_unlock(&ak4531->reg_mutex);
+ return 0;
+}
+
+static int snd_ak4531_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol);
+ int reg1 = kcontrol->private_value & 0xff;
+ int reg2 = (kcontrol->private_value >> 8) & 0xff;
+ int left_shift = (kcontrol->private_value >> 16) & 0x0f;
+ int right_shift = (kcontrol->private_value >> 24) & 0x0f;
+ int change;
+ int val1, val2;
+
+ mutex_lock(&ak4531->reg_mutex);
+ val1 = ak4531->regs[reg1] & ~((1 << left_shift) | (1 << right_shift));
+ val2 = ak4531->regs[reg2] & ~((1 << left_shift) | (1 << right_shift));
+ val1 |= (ucontrol->value.integer.value[0] & 1) << left_shift;
+ val2 |= (ucontrol->value.integer.value[1] & 1) << left_shift;
+ val1 |= (ucontrol->value.integer.value[2] & 1) << right_shift;
+ val2 |= (ucontrol->value.integer.value[3] & 1) << right_shift;
+ change = val1 != ak4531->regs[reg1] || val2 != ak4531->regs[reg2];
+ ak4531->write(ak4531, reg1, ak4531->regs[reg1] = val1);
+ ak4531->write(ak4531, reg2, ak4531->regs[reg2] = val2);
+ mutex_unlock(&ak4531->reg_mutex);
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0);
+
+static struct snd_kcontrol_new snd_ak4531_controls[] __devinitdata = {
+
+AK4531_DOUBLE_TLV("Master Playback Switch", 0,
+ AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1,
+ db_scale_master),
+AK4531_DOUBLE("Master Playback Volume", 0, AK4531_LMASTER, AK4531_RMASTER, 0, 0, 0x1f, 1),
+
+AK4531_SINGLE_TLV("Master Mono Playback Switch", 0, AK4531_MONO_OUT, 7, 1, 1,
+ db_scale_mono),
+AK4531_SINGLE("Master Mono Playback Volume", 0, AK4531_MONO_OUT, 0, 0x07, 1),
+
+AK4531_DOUBLE("PCM Switch", 0, AK4531_LVOICE, AK4531_RVOICE, 7, 7, 1, 1),
+AK4531_DOUBLE_TLV("PCM Volume", 0, AK4531_LVOICE, AK4531_RVOICE, 0, 0, 0x1f, 1,
+ db_scale_input),
+AK4531_DOUBLE("PCM Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 3, 2, 1, 0),
+AK4531_DOUBLE("PCM Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 2, 2, 1, 0),
+
+AK4531_DOUBLE("PCM Switch", 1, AK4531_LFM, AK4531_RFM, 7, 7, 1, 1),
+AK4531_DOUBLE_TLV("PCM Volume", 1, AK4531_LFM, AK4531_RFM, 0, 0, 0x1f, 1,
+ db_scale_input),
+AK4531_DOUBLE("PCM Playback Switch", 1, AK4531_OUT_SW1, AK4531_OUT_SW1, 6, 5, 1, 0),
+AK4531_INPUT_SW("PCM Capture Route", 1, AK4531_LIN_SW1, AK4531_RIN_SW1, 6, 5),
+
+AK4531_DOUBLE("CD Switch", 0, AK4531_LCD, AK4531_RCD, 7, 7, 1, 1),
+AK4531_DOUBLE_TLV("CD Volume", 0, AK4531_LCD, AK4531_RCD, 0, 0, 0x1f, 1,
+ db_scale_input),
+AK4531_DOUBLE("CD Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 2, 1, 1, 0),
+AK4531_INPUT_SW("CD Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 2, 1),
+
+AK4531_DOUBLE("Line Switch", 0, AK4531_LLINE, AK4531_RLINE, 7, 7, 1, 1),
+AK4531_DOUBLE_TLV("Line Volume", 0, AK4531_LLINE, AK4531_RLINE, 0, 0, 0x1f, 1,
+ db_scale_input),
+AK4531_DOUBLE("Line Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 4, 3, 1, 0),
+AK4531_INPUT_SW("Line Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 4, 3),
+
+AK4531_DOUBLE("Aux Switch", 0, AK4531_LAUXA, AK4531_RAUXA, 7, 7, 1, 1),
+AK4531_DOUBLE_TLV("Aux Volume", 0, AK4531_LAUXA, AK4531_RAUXA, 0, 0, 0x1f, 1,
+ db_scale_input),
+AK4531_DOUBLE("Aux Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 5, 4, 1, 0),
+AK4531_INPUT_SW("Aux Capture Route", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 4, 3),
+
+AK4531_SINGLE("Mono Switch", 0, AK4531_MONO1, 7, 1, 1),
+AK4531_SINGLE_TLV("Mono Volume", 0, AK4531_MONO1, 0, 0x1f, 1, db_scale_input),
+AK4531_SINGLE("Mono Playback Switch", 0, AK4531_OUT_SW2, 0, 1, 0),
+AK4531_DOUBLE("Mono Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 0, 0, 1, 0),
+
+AK4531_SINGLE("Mono Switch", 1, AK4531_MONO2, 7, 1, 1),
+AK4531_SINGLE_TLV("Mono Volume", 1, AK4531_MONO2, 0, 0x1f, 1, db_scale_input),
+AK4531_SINGLE("Mono Playback Switch", 1, AK4531_OUT_SW2, 1, 1, 0),
+AK4531_DOUBLE("Mono Capture Switch", 1, AK4531_LIN_SW2, AK4531_RIN_SW2, 1, 1, 1, 0),
+
+AK4531_SINGLE_TLV("Mic Volume", 0, AK4531_MIC, 0, 0x1f, 1, db_scale_input),
+AK4531_SINGLE("Mic Switch", 0, AK4531_MIC, 7, 1, 1),
+AK4531_SINGLE("Mic Playback Switch", 0, AK4531_OUT_SW1, 0, 1, 0),
+AK4531_DOUBLE("Mic Capture Switch", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 0, 0, 1, 0),
+
+AK4531_DOUBLE("Mic Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 7, 7, 1, 0),
+AK4531_DOUBLE("Mono1 Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 6, 6, 1, 0),
+AK4531_DOUBLE("Mono2 Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 5, 5, 1, 0),
+
+AK4531_SINGLE("AD Input Select", 0, AK4531_AD_IN, 0, 1, 0),
+AK4531_SINGLE("Mic Boost (+30dB)", 0, AK4531_MIC_GAIN, 0, 1, 0)
+};
+
+static int snd_ak4531_free(struct snd_ak4531 *ak4531)
+{
+ if (ak4531) {
+ if (ak4531->private_free)
+ ak4531->private_free(ak4531);
+ kfree(ak4531);
+ }
+ return 0;
+}
+
+static int snd_ak4531_dev_free(struct snd_device *device)
+{
+ struct snd_ak4531 *ak4531 = device->device_data;
+ return snd_ak4531_free(ak4531);
+}
+
+static u8 snd_ak4531_initial_map[0x19 + 1] = {
+ 0x9f, /* 00: Master Volume Lch */
+ 0x9f, /* 01: Master Volume Rch */
+ 0x9f, /* 02: Voice Volume Lch */
+ 0x9f, /* 03: Voice Volume Rch */
+ 0x9f, /* 04: FM Volume Lch */
+ 0x9f, /* 05: FM Volume Rch */
+ 0x9f, /* 06: CD Audio Volume Lch */
+ 0x9f, /* 07: CD Audio Volume Rch */
+ 0x9f, /* 08: Line Volume Lch */
+ 0x9f, /* 09: Line Volume Rch */
+ 0x9f, /* 0a: Aux Volume Lch */
+ 0x9f, /* 0b: Aux Volume Rch */
+ 0x9f, /* 0c: Mono1 Volume */
+ 0x9f, /* 0d: Mono2 Volume */
+ 0x9f, /* 0e: Mic Volume */
+ 0x87, /* 0f: Mono-out Volume */
+ 0x00, /* 10: Output Mixer SW1 */
+ 0x00, /* 11: Output Mixer SW2 */
+ 0x00, /* 12: Lch Input Mixer SW1 */
+ 0x00, /* 13: Rch Input Mixer SW1 */
+ 0x00, /* 14: Lch Input Mixer SW2 */
+ 0x00, /* 15: Rch Input Mixer SW2 */
+ 0x00, /* 16: Reset & Power Down */
+ 0x00, /* 17: Clock Select */
+ 0x00, /* 18: AD Input Select */
+ 0x01 /* 19: Mic Amp Setup */
+};
+
+int __devinit snd_ak4531_mixer(struct snd_card *card,
+ struct snd_ak4531 *_ak4531,
+ struct snd_ak4531 **rak4531)
+{
+ unsigned int idx;
+ int err;
+ struct snd_ak4531 *ak4531;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ak4531_dev_free,
+ };
+
+ if (snd_BUG_ON(!card || !_ak4531))
+ return -EINVAL;
+ if (rak4531)
+ *rak4531 = NULL;
+ ak4531 = kzalloc(sizeof(*ak4531), GFP_KERNEL);
+ if (ak4531 == NULL)
+ return -ENOMEM;
+ *ak4531 = *_ak4531;
+ mutex_init(&ak4531->reg_mutex);
+ if ((err = snd_component_add(card, "AK4531")) < 0) {
+ snd_ak4531_free(ak4531);
+ return err;
+ }
+ strcpy(card->mixername, "Asahi Kasei AK4531");
+ ak4531->write(ak4531, AK4531_RESET, 0x03); /* no RST, PD */
+ udelay(100);
+ ak4531->write(ak4531, AK4531_CLOCK, 0x00); /* CODEC ADC and CODEC DAC use {LR,B}CLK2 and run off LRCLK2 PLL */
+ for (idx = 0; idx <= 0x19; idx++) {
+ if (idx == AK4531_RESET || idx == AK4531_CLOCK)
+ continue;
+ ak4531->write(ak4531, idx, ak4531->regs[idx] = snd_ak4531_initial_map[idx]); /* recording source is mixer */
+ }
+ for (idx = 0; idx < ARRAY_SIZE(snd_ak4531_controls); idx++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_ak4531_controls[idx], ak4531))) < 0) {
+ snd_ak4531_free(ak4531);
+ return err;
+ }
+ }
+ snd_ak4531_proc_init(card, ak4531);
+ if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ak4531, &ops)) < 0) {
+ snd_ak4531_free(ak4531);
+ return err;
+ }
+
+#if 0
+ snd_ak4531_dump(ak4531);
+#endif
+ if (rak4531)
+ *rak4531 = ak4531;
+ return 0;
+}
+
+/*
+ * power management
+ */
+#ifdef CONFIG_PM
+void snd_ak4531_suspend(struct snd_ak4531 *ak4531)
+{
+ /* mute */
+ ak4531->write(ak4531, AK4531_LMASTER, 0x9f);
+ ak4531->write(ak4531, AK4531_RMASTER, 0x9f);
+ /* powerdown */
+ ak4531->write(ak4531, AK4531_RESET, 0x01);
+}
+
+void snd_ak4531_resume(struct snd_ak4531 *ak4531)
+{
+ int idx;
+
+ /* initialize */
+ ak4531->write(ak4531, AK4531_RESET, 0x03);
+ udelay(100);
+ ak4531->write(ak4531, AK4531_CLOCK, 0x00);
+ /* restore mixer registers */
+ for (idx = 0; idx <= 0x19; idx++) {
+ if (idx == AK4531_RESET || idx == AK4531_CLOCK)
+ continue;
+ ak4531->write(ak4531, idx, ak4531->regs[idx]);
+ }
+}
+#endif
+
+#ifdef CONFIG_PROC_FS
+/*
+ * /proc interface
+ */
+
+static void snd_ak4531_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ak4531 *ak4531 = entry->private_data;
+
+ snd_iprintf(buffer, "Asahi Kasei AK4531\n\n");
+ snd_iprintf(buffer, "Recording source : %s\n"
+ "MIC gain : %s\n",
+ ak4531->regs[AK4531_AD_IN] & 1 ? "external" : "mixer",
+ ak4531->regs[AK4531_MIC_GAIN] & 1 ? "+30dB" : "+0dB");
+}
+
+static void __devinit
+snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(card, "ak4531", &entry))
+ snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read);
+}
+#endif
diff --git a/sound/pci/ali5451/Makefile b/sound/pci/ali5451/Makefile
new file mode 100644
index 0000000..713459c
--- /dev/null
+++ b/sound/pci/ali5451/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ali5451-objs := ali5451.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_ALI5451) += snd-ali5451.o
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
new file mode 100644
index 0000000..1a0fd65
--- /dev/null
+++ b/sound/pci/ali5451/ali5451.c
@@ -0,0 +1,2378 @@
+/*
+ * Matt Wu <Matt_Wu@acersoftech.com.cn>
+ * Apr 26, 2001
+ * Routines for control of ALi pci audio M5451
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public Lcodecnse as published by
+ * the Free Software Foundation; either version 2 of the Lcodecnse, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public Lcodecnse for more details.
+ *
+ * You should have received a copy of the GNU General Public Lcodecnse
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/ac97_codec.h>
+#include <sound/mpu401.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Matt Wu <Matt_Wu@acersoftech.com.cn>");
+MODULE_DESCRIPTION("ALI M5451");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int pcm_channels = 32;
+static int spdif;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for ALI M5451 PCI Audio.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for ALI M5451 PCI Audio.");
+module_param(pcm_channels, int, 0444);
+MODULE_PARM_DESC(pcm_channels, "PCM Channels");
+module_param(spdif, bool, 0444);
+MODULE_PARM_DESC(spdif, "Support SPDIF I/O");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
+
+/*
+ * Debug part definitions
+ */
+
+/* #define ALI_DEBUG */
+
+#ifdef ALI_DEBUG
+#define snd_ali_printk(format, args...) printk(KERN_DEBUG format, ##args);
+#else
+#define snd_ali_printk(format, args...)
+#endif
+
+/*
+ * Constants definition
+ */
+
+#define DEVICE_ID_ALI5451 ((PCI_VENDOR_ID_AL<<16)|PCI_DEVICE_ID_AL_M5451)
+
+
+#define ALI_CHANNELS 32
+
+#define ALI_PCM_IN_CHANNEL 31
+#define ALI_SPDIF_IN_CHANNEL 19
+#define ALI_SPDIF_OUT_CHANNEL 15
+#define ALI_CENTER_CHANNEL 24
+#define ALI_LEF_CHANNEL 23
+#define ALI_SURR_LEFT_CHANNEL 26
+#define ALI_SURR_RIGHT_CHANNEL 25
+#define ALI_MODEM_IN_CHANNEL 21
+#define ALI_MODEM_OUT_CHANNEL 20
+
+#define SNDRV_ALI_VOICE_TYPE_PCM 01
+#define SNDRV_ALI_VOICE_TYPE_OTH 02
+
+#define ALI_5451_V02 0x02
+
+/*
+ * Direct Registers
+ */
+
+#define ALI_LEGACY_DMAR0 0x00 /* ADR0 */
+#define ALI_LEGACY_DMAR4 0x04 /* CNT0 */
+#define ALI_LEGACY_DMAR11 0x0b /* MOD */
+#define ALI_LEGACY_DMAR15 0x0f /* MMR */
+#define ALI_MPUR0 0x20
+#define ALI_MPUR1 0x21
+#define ALI_MPUR2 0x22
+#define ALI_MPUR3 0x23
+
+#define ALI_AC97_WRITE 0x40
+#define ALI_AC97_READ 0x44
+
+#define ALI_SCTRL 0x48
+#define ALI_SPDIF_OUT_ENABLE 0x20
+#define ALI_SCTRL_LINE_IN2 (1 << 9)
+#define ALI_SCTRL_GPIO_IN2 (1 << 13)
+#define ALI_SCTRL_LINE_OUT_EN (1 << 20)
+#define ALI_SCTRL_GPIO_OUT_EN (1 << 23)
+#define ALI_SCTRL_CODEC1_READY (1 << 24)
+#define ALI_SCTRL_CODEC2_READY (1 << 25)
+#define ALI_AC97_GPIO 0x4c
+#define ALI_AC97_GPIO_ENABLE 0x8000
+#define ALI_AC97_GPIO_DATA_SHIFT 16
+#define ALI_SPDIF_CS 0x70
+#define ALI_SPDIF_CTRL 0x74
+#define ALI_SPDIF_IN_FUNC_ENABLE 0x02
+#define ALI_SPDIF_IN_CH_STATUS 0x40
+#define ALI_SPDIF_OUT_CH_STATUS 0xbf
+#define ALI_START 0x80
+#define ALI_STOP 0x84
+#define ALI_CSPF 0x90
+#define ALI_AINT 0x98
+#define ALI_GC_CIR 0xa0
+ #define ENDLP_IE 0x00001000
+ #define MIDLP_IE 0x00002000
+#define ALI_AINTEN 0xa4
+#define ALI_VOLUME 0xa8
+#define ALI_SBDELTA_DELTA_R 0xac
+#define ALI_MISCINT 0xb0
+ #define ADDRESS_IRQ 0x00000020
+ #define TARGET_REACHED 0x00008000
+ #define MIXER_OVERFLOW 0x00000800
+ #define MIXER_UNDERFLOW 0x00000400
+ #define GPIO_IRQ 0x01000000
+#define ALI_SBBL_SBCL 0xc0
+#define ALI_SBCTRL_SBE2R_SBDD 0xc4
+#define ALI_STIMER 0xc8
+#define ALI_GLOBAL_CONTROL 0xd4
+#define ALI_SPDIF_OUT_SEL_PCM 0x00000400 /* bit 10 */
+#define ALI_SPDIF_IN_SUPPORT 0x00000800 /* bit 11 */
+#define ALI_SPDIF_OUT_CH_ENABLE 0x00008000 /* bit 15 */
+#define ALI_SPDIF_IN_CH_ENABLE 0x00080000 /* bit 19 */
+#define ALI_PCM_IN_ENABLE 0x80000000 /* bit 31 */
+
+#define ALI_CSO_ALPHA_FMS 0xe0
+#define ALI_LBA 0xe4
+#define ALI_ESO_DELTA 0xe8
+#define ALI_GVSEL_PAN_VOC_CTRL_EC 0xf0
+#define ALI_EBUF1 0xf4
+#define ALI_EBUF2 0xf8
+
+#define ALI_REG(codec, x) ((codec)->port + x)
+
+#define MAX_CODECS 2
+
+
+struct snd_ali;
+struct snd_ali_voice;
+
+struct snd_ali_channel_control {
+ /* register data */
+ struct REGDATA {
+ unsigned int start;
+ unsigned int stop;
+ unsigned int aint;
+ unsigned int ainten;
+ } data;
+
+ /* register addresses */
+ struct REGS {
+ unsigned int start;
+ unsigned int stop;
+ unsigned int aint;
+ unsigned int ainten;
+ unsigned int ac97read;
+ unsigned int ac97write;
+ } regs;
+
+};
+
+struct snd_ali_voice {
+ unsigned int number;
+ unsigned int use :1,
+ pcm :1,
+ midi :1,
+ mode :1,
+ synth :1,
+ running :1;
+
+ /* PCM data */
+ struct snd_ali *codec;
+ struct snd_pcm_substream *substream;
+ struct snd_ali_voice *extra;
+
+ int eso; /* final ESO value for channel */
+ int count; /* runtime->period_size */
+
+ /* --- */
+
+ void *private_data;
+ void (*private_free)(void *private_data);
+};
+
+
+struct snd_alidev {
+
+ struct snd_ali_voice voices[ALI_CHANNELS];
+
+ unsigned int chcnt; /* num of opened channels */
+ unsigned int chmap; /* bitmap for opened channels */
+ unsigned int synthcount;
+
+};
+
+
+#define ALI_GLOBAL_REGS 56
+#define ALI_CHANNEL_REGS 8
+struct snd_ali_image {
+ u32 regs[ALI_GLOBAL_REGS];
+ u32 channel_regs[ALI_CHANNELS][ALI_CHANNEL_REGS];
+};
+
+
+struct snd_ali {
+ int irq;
+ unsigned long port;
+ unsigned char revision;
+
+ unsigned int hw_initialized :1;
+ unsigned int spdif_support :1;
+
+ struct pci_dev *pci;
+ struct pci_dev *pci_m1533;
+ struct pci_dev *pci_m7101;
+
+ struct snd_card *card;
+ struct snd_pcm *pcm[MAX_CODECS];
+ struct snd_alidev synth;
+ struct snd_ali_channel_control chregs;
+
+ /* S/PDIF Mask */
+ unsigned int spdif_mask;
+
+ unsigned int spurious_irq_count;
+ unsigned int spurious_irq_max_delta;
+
+ unsigned int num_of_codecs;
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97[MAX_CODECS];
+ unsigned short ac97_ext_id;
+ unsigned short ac97_ext_status;
+
+ spinlock_t reg_lock;
+ spinlock_t voice_alloc;
+
+#ifdef CONFIG_PM
+ struct snd_ali_image *image;
+#endif
+};
+
+static struct pci_device_id snd_ali_ids[] = {
+ {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0},
+ {0, }
+};
+MODULE_DEVICE_TABLE(pci, snd_ali_ids);
+
+static void snd_ali_clear_voices(struct snd_ali *, unsigned int, unsigned int);
+static unsigned short snd_ali_codec_peek(struct snd_ali *, int, unsigned short);
+static void snd_ali_codec_poke(struct snd_ali *, int, unsigned short,
+ unsigned short);
+
+/*
+ * AC97 ACCESS
+ */
+
+static inline unsigned int snd_ali_5451_peek(struct snd_ali *codec,
+ unsigned int port)
+{
+ return (unsigned int)inl(ALI_REG(codec, port));
+}
+
+static inline void snd_ali_5451_poke(struct snd_ali *codec,
+ unsigned int port,
+ unsigned int val)
+{
+ outl((unsigned int)val, ALI_REG(codec, port));
+}
+
+static int snd_ali_codec_ready(struct snd_ali *codec,
+ unsigned int port)
+{
+ unsigned long end_time;
+ unsigned int res;
+
+ end_time = jiffies + msecs_to_jiffies(250);
+ do {
+ res = snd_ali_5451_peek(codec,port);
+ if (!(res & 0x8000))
+ return 0;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ snd_ali_5451_poke(codec, port, res & ~0x8000);
+ snd_printdd("ali_codec_ready: codec is not ready.\n ");
+ return -EIO;
+}
+
+static int snd_ali_stimer_ready(struct snd_ali *codec)
+{
+ unsigned long end_time;
+ unsigned long dwChk1,dwChk2;
+
+ dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER);
+ dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER);
+
+ end_time = jiffies + msecs_to_jiffies(250);
+ do {
+ dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER);
+ if (dwChk2 != dwChk1)
+ return 0;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "ali_stimer_read: stimer is not ready.\n");
+ return -EIO;
+}
+
+static void snd_ali_codec_poke(struct snd_ali *codec,int secondary,
+ unsigned short reg,
+ unsigned short val)
+{
+ unsigned int dwVal;
+ unsigned int port;
+
+ if (reg >= 0x80) {
+ snd_printk(KERN_ERR "ali_codec_poke: reg(%xh) invalid.\n", reg);
+ return;
+ }
+
+ port = codec->chregs.regs.ac97write;
+
+ if (snd_ali_codec_ready(codec, port) < 0)
+ return;
+ if (snd_ali_stimer_ready(codec) < 0)
+ return;
+
+ dwVal = (unsigned int) (reg & 0xff);
+ dwVal |= 0x8000 | (val << 16);
+ if (secondary)
+ dwVal |= 0x0080;
+ if (codec->revision == ALI_5451_V02)
+ dwVal |= 0x0100;
+
+ snd_ali_5451_poke(codec, port, dwVal);
+
+ return ;
+}
+
+static unsigned short snd_ali_codec_peek(struct snd_ali *codec,
+ int secondary,
+ unsigned short reg)
+{
+ unsigned int dwVal;
+ unsigned int port;
+
+ if (reg >= 0x80) {
+ snd_printk(KERN_ERR "ali_codec_peek: reg(%xh) invalid.\n", reg);
+ return ~0;
+ }
+
+ port = codec->chregs.regs.ac97read;
+
+ if (snd_ali_codec_ready(codec, port) < 0)
+ return ~0;
+ if (snd_ali_stimer_ready(codec) < 0)
+ return ~0;
+
+ dwVal = (unsigned int) (reg & 0xff);
+ dwVal |= 0x8000; /* bit 15*/
+ if (secondary)
+ dwVal |= 0x0080;
+
+ snd_ali_5451_poke(codec, port, dwVal);
+
+ if (snd_ali_stimer_ready(codec) < 0)
+ return ~0;
+ if (snd_ali_codec_ready(codec, port) < 0)
+ return ~0;
+
+ return (snd_ali_5451_peek(codec, port) & 0xffff0000) >> 16;
+}
+
+static void snd_ali_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val )
+{
+ struct snd_ali *codec = ac97->private_data;
+
+ snd_ali_printk("codec_write: reg=%xh data=%xh.\n", reg, val);
+ if (reg == AC97_GPIO_STATUS) {
+ outl((val << ALI_AC97_GPIO_DATA_SHIFT) | ALI_AC97_GPIO_ENABLE,
+ ALI_REG(codec, ALI_AC97_GPIO));
+ return;
+ }
+ snd_ali_codec_poke(codec, ac97->num, reg, val);
+ return ;
+}
+
+
+static unsigned short snd_ali_codec_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct snd_ali *codec = ac97->private_data;
+
+ snd_ali_printk("codec_read reg=%xh.\n", reg);
+ return snd_ali_codec_peek(codec, ac97->num, reg);
+}
+
+/*
+ * AC97 Reset
+ */
+
+static int snd_ali_reset_5451(struct snd_ali *codec)
+{
+ struct pci_dev *pci_dev;
+ unsigned short wCount, wReg;
+ unsigned int dwVal;
+
+ pci_dev = codec->pci_m1533;
+ if (pci_dev) {
+ pci_read_config_dword(pci_dev, 0x7c, &dwVal);
+ pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000);
+ udelay(5000);
+ pci_read_config_dword(pci_dev, 0x7c, &dwVal);
+ pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff);
+ udelay(5000);
+ }
+
+ pci_dev = codec->pci;
+ pci_read_config_dword(pci_dev, 0x44, &dwVal);
+ pci_write_config_dword(pci_dev, 0x44, dwVal | 0x000c0000);
+ udelay(500);
+ pci_read_config_dword(pci_dev, 0x44, &dwVal);
+ pci_write_config_dword(pci_dev, 0x44, dwVal & 0xfffbffff);
+ udelay(5000);
+
+ wCount = 200;
+ while(wCount--) {
+ wReg = snd_ali_codec_peek(codec, 0, AC97_POWERDOWN);
+ if ((wReg & 0x000f) == 0x000f)
+ return 0;
+ udelay(5000);
+ }
+
+ /* non-fatal if you have a non PM capable codec */
+ /* snd_printk(KERN_WARNING "ali5451: reset time out\n"); */
+ return 0;
+}
+
+#ifdef CODEC_RESET
+
+static int snd_ali_reset_codec(struct snd_ali *codec)
+{
+ struct pci_dev *pci_dev;
+ unsigned char bVal;
+ unsigned int dwVal;
+ unsigned short wCount, wReg;
+
+ pci_dev = codec->pci_m1533;
+
+ pci_read_config_dword(pci_dev, 0x7c, &dwVal);
+ pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000);
+ udelay(5000);
+ pci_read_config_dword(pci_dev, 0x7c, &dwVal);
+ pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff);
+ udelay(5000);
+
+ bVal = inb(ALI_REG(codec,ALI_SCTRL));
+ bVal |= 0x02;
+ outb(ALI_REG(codec,ALI_SCTRL),bVal);
+ udelay(5000);
+ bVal = inb(ALI_REG(codec,ALI_SCTRL));
+ bVal &= 0xfd;
+ outb(ALI_REG(codec,ALI_SCTRL),bVal);
+ udelay(15000);
+
+ wCount = 200;
+ while (wCount--) {
+ wReg = snd_ali_codec_read(codec->ac97, AC97_POWERDOWN);
+ if ((wReg & 0x000f) == 0x000f)
+ return 0;
+ udelay(5000);
+ }
+ return -1;
+}
+
+#endif
+
+/*
+ * ALI 5451 Controller
+ */
+
+static void snd_ali_enable_special_channel(struct snd_ali *codec,
+ unsigned int channel)
+{
+ unsigned long dwVal;
+
+ dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ dwVal |= 1 << (channel & 0x0000001f);
+ outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+}
+
+static void snd_ali_disable_special_channel(struct snd_ali *codec,
+ unsigned int channel)
+{
+ unsigned long dwVal;
+
+ dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ dwVal &= ~(1 << (channel & 0x0000001f));
+ outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+}
+
+static void snd_ali_enable_address_interrupt(struct snd_ali *codec)
+{
+ unsigned int gc;
+
+ gc = inl(ALI_REG(codec, ALI_GC_CIR));
+ gc |= ENDLP_IE;
+ gc |= MIDLP_IE;
+ outl( gc, ALI_REG(codec, ALI_GC_CIR));
+}
+
+static void snd_ali_disable_address_interrupt(struct snd_ali *codec)
+{
+ unsigned int gc;
+
+ gc = inl(ALI_REG(codec, ALI_GC_CIR));
+ gc &= ~ENDLP_IE;
+ gc &= ~MIDLP_IE;
+ outl(gc, ALI_REG(codec, ALI_GC_CIR));
+}
+
+#if 0 /* not used */
+static void snd_ali_enable_voice_irq(struct snd_ali *codec,
+ unsigned int channel)
+{
+ unsigned int mask;
+ struct snd_ali_channel_control *pchregs = &(codec->chregs);
+
+ snd_ali_printk("enable_voice_irq channel=%d\n",channel);
+
+ mask = 1 << (channel & 0x1f);
+ pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten));
+ pchregs->data.ainten |= mask;
+ outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten));
+}
+#endif
+
+static void snd_ali_disable_voice_irq(struct snd_ali *codec,
+ unsigned int channel)
+{
+ unsigned int mask;
+ struct snd_ali_channel_control *pchregs = &(codec->chregs);
+
+ snd_ali_printk("disable_voice_irq channel=%d\n",channel);
+
+ mask = 1 << (channel & 0x1f);
+ pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten));
+ pchregs->data.ainten &= ~mask;
+ outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten));
+}
+
+static int snd_ali_alloc_pcm_channel(struct snd_ali *codec, int channel)
+{
+ unsigned int idx = channel & 0x1f;
+
+ if (codec->synth.chcnt >= ALI_CHANNELS){
+ snd_printk(KERN_ERR
+ "ali_alloc_pcm_channel: no free channels.\n");
+ return -1;
+ }
+
+ if (!(codec->synth.chmap & (1 << idx))) {
+ codec->synth.chmap |= 1 << idx;
+ codec->synth.chcnt++;
+ snd_ali_printk("alloc_pcm_channel no. %d.\n",idx);
+ return idx;
+ }
+ return -1;
+}
+
+static int snd_ali_find_free_channel(struct snd_ali * codec, int rec)
+{
+ int idx;
+ int result = -1;
+
+ snd_ali_printk("find_free_channel: for %s\n",rec ? "rec" : "pcm");
+
+ /* recording */
+ if (rec) {
+ if (codec->spdif_support &&
+ (inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)) &
+ ALI_SPDIF_IN_SUPPORT))
+ idx = ALI_SPDIF_IN_CHANNEL;
+ else
+ idx = ALI_PCM_IN_CHANNEL;
+
+ result = snd_ali_alloc_pcm_channel(codec, idx);
+ if (result >= 0)
+ return result;
+ else {
+ snd_printk(KERN_ERR "ali_find_free_channel: "
+ "record channel is busy now.\n");
+ return -1;
+ }
+ }
+
+ /* playback... */
+ if (codec->spdif_support &&
+ (inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)) &
+ ALI_SPDIF_OUT_CH_ENABLE)) {
+ idx = ALI_SPDIF_OUT_CHANNEL;
+ result = snd_ali_alloc_pcm_channel(codec, idx);
+ if (result >= 0)
+ return result;
+ else
+ snd_printk(KERN_ERR "ali_find_free_channel: "
+ "S/PDIF out channel is in busy now.\n");
+ }
+
+ for (idx = 0; idx < ALI_CHANNELS; idx++) {
+ result = snd_ali_alloc_pcm_channel(codec, idx);
+ if (result >= 0)
+ return result;
+ }
+ snd_printk(KERN_ERR "ali_find_free_channel: no free channels.\n");
+ return -1;
+}
+
+static void snd_ali_free_channel_pcm(struct snd_ali *codec, int channel)
+{
+ unsigned int idx = channel & 0x0000001f;
+
+ snd_ali_printk("free_channel_pcm channel=%d\n",channel);
+
+ if (channel < 0 || channel >= ALI_CHANNELS)
+ return;
+
+ if (!(codec->synth.chmap & (1 << idx))) {
+ snd_printk(KERN_ERR "ali_free_channel_pcm: "
+ "channel %d is not in use.\n", channel);
+ return;
+ } else {
+ codec->synth.chmap &= ~(1 << idx);
+ codec->synth.chcnt--;
+ }
+}
+
+#if 0 /* not used */
+static void snd_ali_start_voice(struct snd_ali *codec, unsigned int channel)
+{
+ unsigned int mask = 1 << (channel & 0x1f);
+
+ snd_ali_printk("start_voice: channel=%d\n",channel);
+ outl(mask, ALI_REG(codec,codec->chregs.regs.start));
+}
+#endif
+
+static void snd_ali_stop_voice(struct snd_ali *codec, unsigned int channel)
+{
+ unsigned int mask = 1 << (channel & 0x1f);
+
+ snd_ali_printk("stop_voice: channel=%d\n",channel);
+ outl(mask, ALI_REG(codec, codec->chregs.regs.stop));
+}
+
+/*
+ * S/PDIF Part
+ */
+
+static void snd_ali_delay(struct snd_ali *codec,int interval)
+{
+ unsigned long begintimer,currenttimer;
+
+ begintimer = inl(ALI_REG(codec, ALI_STIMER));
+ currenttimer = inl(ALI_REG(codec, ALI_STIMER));
+
+ while (currenttimer < begintimer + interval) {
+ if (snd_ali_stimer_ready(codec) < 0)
+ break;
+ currenttimer = inl(ALI_REG(codec, ALI_STIMER));
+ cpu_relax();
+ }
+}
+
+static void snd_ali_detect_spdif_rate(struct snd_ali *codec)
+{
+ u16 wval;
+ u16 count = 0;
+ u8 bval, R1 = 0, R2;
+
+ bval = inb(ALI_REG(codec, ALI_SPDIF_CTRL + 1));
+ bval |= 0x1F;
+ outb(bval, ALI_REG(codec, ALI_SPDIF_CTRL + 1));
+
+ while ((R1 < 0x0b || R1 > 0x0e) && R1 != 0x12 && count <= 50000) {
+ count ++;
+ snd_ali_delay(codec, 6);
+ bval = inb(ALI_REG(codec, ALI_SPDIF_CTRL + 1));
+ R1 = bval & 0x1F;
+ }
+
+ if (count > 50000) {
+ snd_printk(KERN_ERR "ali_detect_spdif_rate: timeout!\n");
+ return;
+ }
+
+ for (count = 0; count <= 50000; count++) {
+ snd_ali_delay(codec, 6);
+ bval = inb(ALI_REG(codec,ALI_SPDIF_CTRL + 1));
+ R2 = bval & 0x1F;
+ if (R2 != R1)
+ R1 = R2;
+ else
+ break;
+ }
+
+ if (count > 50000) {
+ snd_printk(KERN_ERR "ali_detect_spdif_rate: timeout!\n");
+ return;
+ }
+
+ if (R2 >= 0x0b && R2 <= 0x0e) {
+ wval = inw(ALI_REG(codec, ALI_SPDIF_CTRL + 2));
+ wval &= 0xe0f0;
+ wval |= (0x09 << 8) | 0x05;
+ outw(wval, ALI_REG(codec, ALI_SPDIF_CTRL + 2));
+
+ bval = inb(ALI_REG(codec, ALI_SPDIF_CS + 3)) & 0xf0;
+ outb(bval | 0x02, ALI_REG(codec, ALI_SPDIF_CS + 3));
+ } else if (R2 == 0x12) {
+ wval = inw(ALI_REG(codec, ALI_SPDIF_CTRL + 2));
+ wval &= 0xe0f0;
+ wval |= (0x0e << 8) | 0x08;
+ outw(wval, ALI_REG(codec, ALI_SPDIF_CTRL + 2));
+
+ bval = inb(ALI_REG(codec,ALI_SPDIF_CS + 3)) & 0xf0;
+ outb(bval | 0x03, ALI_REG(codec, ALI_SPDIF_CS + 3));
+ }
+}
+
+static unsigned int snd_ali_get_spdif_in_rate(struct snd_ali *codec)
+{
+ u32 dwRate;
+ u8 bval;
+
+ bval = inb(ALI_REG(codec, ALI_SPDIF_CTRL));
+ bval &= 0x7f;
+ bval |= 0x40;
+ outb(bval, ALI_REG(codec, ALI_SPDIF_CTRL));
+
+ snd_ali_detect_spdif_rate(codec);
+
+ bval = inb(ALI_REG(codec, ALI_SPDIF_CS + 3));
+ bval &= 0x0f;
+
+ switch (bval) {
+ case 0: dwRate = 44100; break;
+ case 1: dwRate = 48000; break;
+ case 2: dwRate = 32000; break;
+ default: dwRate = 0; break;
+ }
+
+ return dwRate;
+}
+
+static void snd_ali_enable_spdif_in(struct snd_ali *codec)
+{
+ unsigned int dwVal;
+
+ dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ dwVal |= ALI_SPDIF_IN_SUPPORT;
+ outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+
+ dwVal = inb(ALI_REG(codec, ALI_SPDIF_CTRL));
+ dwVal |= 0x02;
+ outb(dwVal, ALI_REG(codec, ALI_SPDIF_CTRL));
+
+ snd_ali_enable_special_channel(codec, ALI_SPDIF_IN_CHANNEL);
+}
+
+static void snd_ali_disable_spdif_in(struct snd_ali *codec)
+{
+ unsigned int dwVal;
+
+ dwVal = inl(ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ dwVal &= ~ALI_SPDIF_IN_SUPPORT;
+ outl(dwVal, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+
+ snd_ali_disable_special_channel(codec, ALI_SPDIF_IN_CHANNEL);
+}
+
+
+static void snd_ali_set_spdif_out_rate(struct snd_ali *codec, unsigned int rate)
+{
+ unsigned char bVal;
+ unsigned int dwRate;
+
+ switch (rate) {
+ case 32000: dwRate = 0x300; break;
+ case 48000: dwRate = 0x200; break;
+ default: dwRate = 0; break;
+ }
+
+ bVal = inb(ALI_REG(codec, ALI_SPDIF_CTRL));
+ bVal &= (unsigned char)(~(1<<6));
+
+ bVal |= 0x80; /* select right */
+ outb(bVal, ALI_REG(codec, ALI_SPDIF_CTRL));
+ outb(dwRate | 0x20, ALI_REG(codec, ALI_SPDIF_CS + 2));
+
+ bVal &= ~0x80; /* select left */
+ outb(bVal, ALI_REG(codec, ALI_SPDIF_CTRL));
+ outw(rate | 0x10, ALI_REG(codec, ALI_SPDIF_CS + 2));
+}
+
+static void snd_ali_enable_spdif_out(struct snd_ali *codec)
+{
+ unsigned short wVal;
+ unsigned char bVal;
+ struct pci_dev *pci_dev;
+
+ pci_dev = codec->pci_m1533;
+ if (pci_dev == NULL)
+ return;
+ pci_read_config_byte(pci_dev, 0x61, &bVal);
+ bVal |= 0x40;
+ pci_write_config_byte(pci_dev, 0x61, bVal);
+ pci_read_config_byte(pci_dev, 0x7d, &bVal);
+ bVal |= 0x01;
+ pci_write_config_byte(pci_dev, 0x7d, bVal);
+
+ pci_read_config_byte(pci_dev, 0x7e, &bVal);
+ bVal &= (~0x20);
+ bVal |= 0x10;
+ pci_write_config_byte(pci_dev, 0x7e, bVal);
+
+ bVal = inb(ALI_REG(codec, ALI_SCTRL));
+ outb(bVal | ALI_SPDIF_OUT_ENABLE, ALI_REG(codec, ALI_SCTRL));
+
+ bVal = inb(ALI_REG(codec, ALI_SPDIF_CTRL));
+ outb(bVal & ALI_SPDIF_OUT_CH_STATUS, ALI_REG(codec, ALI_SPDIF_CTRL));
+
+ wVal = inw(ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ wVal |= ALI_SPDIF_OUT_SEL_PCM;
+ outw(wVal, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ snd_ali_disable_special_channel(codec, ALI_SPDIF_OUT_CHANNEL);
+}
+
+static void snd_ali_enable_spdif_chnout(struct snd_ali *codec)
+{
+ unsigned short wVal;
+
+ wVal = inw(ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ wVal &= ~ALI_SPDIF_OUT_SEL_PCM;
+ outw(wVal, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+/*
+ wVal = inw(ALI_REG(codec, ALI_SPDIF_CS));
+ if (flag & ALI_SPDIF_OUT_NON_PCM)
+ wVal |= 0x0002;
+ else
+ wVal &= (~0x0002);
+ outw(wVal, ALI_REG(codec, ALI_SPDIF_CS));
+*/
+ snd_ali_enable_special_channel(codec, ALI_SPDIF_OUT_CHANNEL);
+}
+
+static void snd_ali_disable_spdif_chnout(struct snd_ali *codec)
+{
+ unsigned short wVal;
+
+ wVal = inw(ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ wVal |= ALI_SPDIF_OUT_SEL_PCM;
+ outw(wVal, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+
+ snd_ali_enable_special_channel(codec, ALI_SPDIF_OUT_CHANNEL);
+}
+
+static void snd_ali_disable_spdif_out(struct snd_ali *codec)
+{
+ unsigned char bVal;
+
+ bVal = inb(ALI_REG(codec, ALI_SCTRL));
+ outb(bVal & ~ALI_SPDIF_OUT_ENABLE, ALI_REG(codec, ALI_SCTRL));
+
+ snd_ali_disable_spdif_chnout(codec);
+}
+
+static void snd_ali_update_ptr(struct snd_ali *codec, int channel)
+{
+ struct snd_ali_voice *pvoice;
+ struct snd_pcm_runtime *runtime;
+ struct snd_ali_channel_control *pchregs;
+ unsigned int old, mask;
+#ifdef ALI_DEBUG
+ unsigned int temp, cspf;
+#endif
+
+ pchregs = &(codec->chregs);
+
+ /* check if interrupt occurred for channel */
+ old = pchregs->data.aint;
+ mask = 1U << (channel & 0x1f);
+
+ if (!(old & mask))
+ return;
+
+ pvoice = &codec->synth.voices[channel];
+ runtime = pvoice->substream->runtime;
+
+ udelay(100);
+ spin_lock(&codec->reg_lock);
+
+ if (pvoice->pcm && pvoice->substream) {
+ /* pcm interrupt */
+#ifdef ALI_DEBUG
+ outb((u8)(pvoice->number), ALI_REG(codec, ALI_GC_CIR));
+ temp = inw(ALI_REG(codec, ALI_CSO_ALPHA_FMS + 2));
+ cspf = (inl(ALI_REG(codec, ALI_CSPF)) & mask) == mask;
+#endif
+ if (pvoice->running) {
+ snd_ali_printk("update_ptr: cso=%4.4x cspf=%d.\n",
+ (u16)temp, cspf);
+ spin_unlock(&codec->reg_lock);
+ snd_pcm_period_elapsed(pvoice->substream);
+ spin_lock(&codec->reg_lock);
+ } else {
+ snd_ali_stop_voice(codec, channel);
+ snd_ali_disable_voice_irq(codec, channel);
+ }
+ } else if (codec->synth.voices[channel].synth) {
+ /* synth interrupt */
+ } else if (codec->synth.voices[channel].midi) {
+ /* midi interrupt */
+ } else {
+ /* unknown interrupt */
+ snd_ali_stop_voice(codec, channel);
+ snd_ali_disable_voice_irq(codec, channel);
+ }
+ spin_unlock(&codec->reg_lock);
+ outl(mask,ALI_REG(codec,pchregs->regs.aint));
+ pchregs->data.aint = old & (~mask);
+}
+
+static irqreturn_t snd_ali_card_interrupt(int irq, void *dev_id)
+{
+ struct snd_ali *codec = dev_id;
+ int channel;
+ unsigned int audio_int;
+ struct snd_ali_channel_control *pchregs;
+
+ if (codec == NULL || !codec->hw_initialized)
+ return IRQ_NONE;
+
+ audio_int = inl(ALI_REG(codec, ALI_MISCINT));
+ if (!audio_int)
+ return IRQ_NONE;
+
+ pchregs = &(codec->chregs);
+ if (audio_int & ADDRESS_IRQ) {
+ /* get interrupt status for all channels */
+ pchregs->data.aint = inl(ALI_REG(codec, pchregs->regs.aint));
+ for (channel = 0; channel < ALI_CHANNELS; channel++)
+ snd_ali_update_ptr(codec, channel);
+ }
+ outl((TARGET_REACHED | MIXER_OVERFLOW | MIXER_UNDERFLOW),
+ ALI_REG(codec, ALI_MISCINT));
+
+ return IRQ_HANDLED;
+}
+
+
+static struct snd_ali_voice *snd_ali_alloc_voice(struct snd_ali * codec,
+ int type, int rec, int channel)
+{
+ struct snd_ali_voice *pvoice;
+ int idx;
+
+ snd_ali_printk("alloc_voice: type=%d rec=%d\n", type, rec);
+
+ spin_lock_irq(&codec->voice_alloc);
+ if (type == SNDRV_ALI_VOICE_TYPE_PCM) {
+ idx = channel > 0 ? snd_ali_alloc_pcm_channel(codec, channel) :
+ snd_ali_find_free_channel(codec,rec);
+ if (idx < 0) {
+ snd_printk(KERN_ERR "ali_alloc_voice: err.\n");
+ spin_unlock_irq(&codec->voice_alloc);
+ return NULL;
+ }
+ pvoice = &(codec->synth.voices[idx]);
+ pvoice->codec = codec;
+ pvoice->use = 1;
+ pvoice->pcm = 1;
+ pvoice->mode = rec;
+ spin_unlock_irq(&codec->voice_alloc);
+ return pvoice;
+ }
+ spin_unlock_irq(&codec->voice_alloc);
+ return NULL;
+}
+
+
+static void snd_ali_free_voice(struct snd_ali * codec,
+ struct snd_ali_voice *pvoice)
+{
+ void (*private_free)(void *);
+ void *private_data;
+
+ snd_ali_printk("free_voice: channel=%d\n",pvoice->number);
+ if (pvoice == NULL || !pvoice->use)
+ return;
+ snd_ali_clear_voices(codec, pvoice->number, pvoice->number);
+ spin_lock_irq(&codec->voice_alloc);
+ private_free = pvoice->private_free;
+ private_data = pvoice->private_data;
+ pvoice->private_free = NULL;
+ pvoice->private_data = NULL;
+ if (pvoice->pcm)
+ snd_ali_free_channel_pcm(codec, pvoice->number);
+ pvoice->use = pvoice->pcm = pvoice->synth = 0;
+ pvoice->substream = NULL;
+ spin_unlock_irq(&codec->voice_alloc);
+ if (private_free)
+ private_free(private_data);
+}
+
+
+static void snd_ali_clear_voices(struct snd_ali *codec,
+ unsigned int v_min,
+ unsigned int v_max)
+{
+ unsigned int i;
+
+ for (i = v_min; i <= v_max; i++) {
+ snd_ali_stop_voice(codec, i);
+ snd_ali_disable_voice_irq(codec, i);
+ }
+}
+
+static void snd_ali_write_voice_regs(struct snd_ali *codec,
+ unsigned int Channel,
+ unsigned int LBA,
+ unsigned int CSO,
+ unsigned int ESO,
+ unsigned int DELTA,
+ unsigned int ALPHA_FMS,
+ unsigned int GVSEL,
+ unsigned int PAN,
+ unsigned int VOL,
+ unsigned int CTRL,
+ unsigned int EC)
+{
+ unsigned int ctlcmds[4];
+
+ outb((unsigned char)(Channel & 0x001f), ALI_REG(codec, ALI_GC_CIR));
+
+ ctlcmds[0] = (CSO << 16) | (ALPHA_FMS & 0x0000ffff);
+ ctlcmds[1] = LBA;
+ ctlcmds[2] = (ESO << 16) | (DELTA & 0x0ffff);
+ ctlcmds[3] = (GVSEL << 31) |
+ ((PAN & 0x0000007f) << 24) |
+ ((VOL & 0x000000ff) << 16) |
+ ((CTRL & 0x0000000f) << 12) |
+ (EC & 0x00000fff);
+
+ outb(Channel, ALI_REG(codec, ALI_GC_CIR));
+
+ outl(ctlcmds[0], ALI_REG(codec, ALI_CSO_ALPHA_FMS));
+ outl(ctlcmds[1], ALI_REG(codec, ALI_LBA));
+ outl(ctlcmds[2], ALI_REG(codec, ALI_ESO_DELTA));
+ outl(ctlcmds[3], ALI_REG(codec, ALI_GVSEL_PAN_VOC_CTRL_EC));
+
+ outl(0x30000000, ALI_REG(codec, ALI_EBUF1)); /* Still Mode */
+ outl(0x30000000, ALI_REG(codec, ALI_EBUF2)); /* Still Mode */
+}
+
+static unsigned int snd_ali_convert_rate(unsigned int rate, int rec)
+{
+ unsigned int delta;
+
+ if (rate < 4000)
+ rate = 4000;
+ if (rate > 48000)
+ rate = 48000;
+
+ if (rec) {
+ if (rate == 44100)
+ delta = 0x116a;
+ else if (rate == 8000)
+ delta = 0x6000;
+ else if (rate == 48000)
+ delta = 0x1000;
+ else
+ delta = ((48000 << 12) / rate) & 0x0000ffff;
+ } else {
+ if (rate == 44100)
+ delta = 0xeb3;
+ else if (rate == 8000)
+ delta = 0x2ab;
+ else if (rate == 48000)
+ delta = 0x1000;
+ else
+ delta = (((rate << 12) + rate) / 48000) & 0x0000ffff;
+ }
+
+ return delta;
+}
+
+static unsigned int snd_ali_control_mode(struct snd_pcm_substream *substream)
+{
+ unsigned int CTRL;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ /* set ctrl mode
+ CTRL default: 8-bit (unsigned) mono, loop mode enabled
+ */
+ CTRL = 0x00000001;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ CTRL |= 0x00000008; /* 16-bit data */
+ if (!snd_pcm_format_unsigned(runtime->format))
+ CTRL |= 0x00000002; /* signed data */
+ if (runtime->channels > 1)
+ CTRL |= 0x00000004; /* stereo data */
+ return CTRL;
+}
+
+/*
+ * PCM part
+ */
+
+static int snd_ali_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+ unsigned int what, whati, capture_flag;
+ struct snd_ali_voice *pvoice, *evoice;
+ unsigned int val;
+ int do_start;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ do_start = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ do_start = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ what = whati = capture_flag = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) {
+ pvoice = s->runtime->private_data;
+ evoice = pvoice->extra;
+ what |= 1 << (pvoice->number & 0x1f);
+ if (evoice == NULL)
+ whati |= 1 << (pvoice->number & 0x1f);
+ else {
+ whati |= 1 << (evoice->number & 0x1f);
+ what |= 1 << (evoice->number & 0x1f);
+ }
+ if (do_start) {
+ pvoice->running = 1;
+ if (evoice != NULL)
+ evoice->running = 1;
+ } else {
+ pvoice->running = 0;
+ if (evoice != NULL)
+ evoice->running = 0;
+ }
+ snd_pcm_trigger_done(s, substream);
+ if (pvoice->mode)
+ capture_flag = 1;
+ }
+ }
+ spin_lock(&codec->reg_lock);
+ if (!do_start)
+ outl(what, ALI_REG(codec, ALI_STOP));
+ val = inl(ALI_REG(codec, ALI_AINTEN));
+ if (do_start)
+ val |= whati;
+ else
+ val &= ~whati;
+ outl(val, ALI_REG(codec, ALI_AINTEN));
+ if (do_start)
+ outl(what, ALI_REG(codec, ALI_START));
+ snd_ali_printk("trigger: what=%xh whati=%xh\n", what, whati);
+ spin_unlock(&codec->reg_lock);
+
+ return 0;
+}
+
+static int snd_ali_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ali_voice *pvoice = runtime->private_data;
+ struct snd_ali_voice *evoice = pvoice->extra;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ /* voice management */
+
+ if (params_buffer_size(hw_params) / 2 !=
+ params_period_size(hw_params)) {
+ if (!evoice) {
+ evoice = snd_ali_alloc_voice(codec,
+ SNDRV_ALI_VOICE_TYPE_PCM,
+ 0, -1);
+ if (!evoice)
+ return -ENOMEM;
+ pvoice->extra = evoice;
+ evoice->substream = substream;
+ }
+ } else {
+ if (evoice) {
+ snd_ali_free_voice(codec, evoice);
+ pvoice->extra = evoice = NULL;
+ }
+ }
+
+ return 0;
+}
+
+static int snd_ali_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ali_voice *pvoice = runtime->private_data;
+ struct snd_ali_voice *evoice = pvoice ? pvoice->extra : NULL;
+
+ snd_pcm_lib_free_pages(substream);
+ if (evoice) {
+ snd_ali_free_voice(codec, evoice);
+ pvoice->extra = NULL;
+ }
+ return 0;
+}
+
+static int snd_ali_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int snd_ali_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_ali_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ali_voice *pvoice = runtime->private_data;
+ struct snd_ali_voice *evoice = pvoice->extra;
+
+ unsigned int LBA;
+ unsigned int Delta;
+ unsigned int ESO;
+ unsigned int CTRL;
+ unsigned int GVSEL;
+ unsigned int PAN;
+ unsigned int VOL;
+ unsigned int EC;
+
+ snd_ali_printk("playback_prepare ...\n");
+
+ spin_lock_irq(&codec->reg_lock);
+
+ /* set Delta (rate) value */
+ Delta = snd_ali_convert_rate(runtime->rate, 0);
+
+ if (pvoice->number == ALI_SPDIF_IN_CHANNEL ||
+ pvoice->number == ALI_PCM_IN_CHANNEL)
+ snd_ali_disable_special_channel(codec, pvoice->number);
+ else if (codec->spdif_support &&
+ (inl(ALI_REG(codec, ALI_GLOBAL_CONTROL)) &
+ ALI_SPDIF_OUT_CH_ENABLE)
+ && pvoice->number == ALI_SPDIF_OUT_CHANNEL) {
+ snd_ali_set_spdif_out_rate(codec, runtime->rate);
+ Delta = 0x1000;
+ }
+
+ /* set Loop Back Address */
+ LBA = runtime->dma_addr;
+
+ /* set interrupt count size */
+ pvoice->count = runtime->period_size;
+
+ /* set target ESO for channel */
+ pvoice->eso = runtime->buffer_size;
+
+ snd_ali_printk("playback_prepare: eso=%xh count=%xh\n",
+ pvoice->eso, pvoice->count);
+
+ /* set ESO to capture first MIDLP interrupt */
+ ESO = pvoice->eso -1;
+ /* set ctrl mode */
+ CTRL = snd_ali_control_mode(substream);
+
+ GVSEL = 1;
+ PAN = 0;
+ VOL = 0;
+ EC = 0;
+ snd_ali_printk("playback_prepare:\n");
+ snd_ali_printk("ch=%d, Rate=%d Delta=%xh,GVSEL=%xh,PAN=%xh,CTRL=%xh\n",
+ pvoice->number,runtime->rate,Delta,GVSEL,PAN,CTRL);
+ snd_ali_write_voice_regs(codec,
+ pvoice->number,
+ LBA,
+ 0, /* cso */
+ ESO,
+ Delta,
+ 0, /* alpha */
+ GVSEL,
+ PAN,
+ VOL,
+ CTRL,
+ EC);
+ if (evoice) {
+ evoice->count = pvoice->count;
+ evoice->eso = pvoice->count << 1;
+ ESO = evoice->eso - 1;
+ snd_ali_write_voice_regs(codec,
+ evoice->number,
+ LBA,
+ 0, /* cso */
+ ESO,
+ Delta,
+ 0, /* alpha */
+ GVSEL,
+ 0x7f,
+ 0x3ff,
+ CTRL,
+ EC);
+ }
+ spin_unlock_irq(&codec->reg_lock);
+ return 0;
+}
+
+
+static int snd_ali_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ali_voice *pvoice = runtime->private_data;
+ unsigned int LBA;
+ unsigned int Delta;
+ unsigned int ESO;
+ unsigned int CTRL;
+ unsigned int GVSEL;
+ unsigned int PAN;
+ unsigned int VOL;
+ unsigned int EC;
+ u8 bValue;
+
+ spin_lock_irq(&codec->reg_lock);
+
+ snd_ali_printk("ali_prepare...\n");
+
+ snd_ali_enable_special_channel(codec,pvoice->number);
+
+ Delta = (pvoice->number == ALI_MODEM_IN_CHANNEL ||
+ pvoice->number == ALI_MODEM_OUT_CHANNEL) ?
+ 0x1000 : snd_ali_convert_rate(runtime->rate, pvoice->mode);
+
+ /* Prepare capture intr channel */
+ if (pvoice->number == ALI_SPDIF_IN_CHANNEL) {
+
+ unsigned int rate;
+
+ spin_unlock_irq(&codec->reg_lock);
+ if (codec->revision != ALI_5451_V02)
+ return -1;
+
+ rate = snd_ali_get_spdif_in_rate(codec);
+ if (rate == 0) {
+ snd_printk(KERN_WARNING "ali_capture_preapre: "
+ "spdif rate detect err!\n");
+ rate = 48000;
+ }
+ spin_lock_irq(&codec->reg_lock);
+ bValue = inb(ALI_REG(codec,ALI_SPDIF_CTRL));
+ if (bValue & 0x10) {
+ outb(bValue,ALI_REG(codec,ALI_SPDIF_CTRL));
+ printk(KERN_WARNING "clear SPDIF parity error flag.\n");
+ }
+
+ if (rate != 48000)
+ Delta = ((rate << 12) / runtime->rate) & 0x00ffff;
+ }
+
+ /* set target ESO for channel */
+ pvoice->eso = runtime->buffer_size;
+
+ /* set interrupt count size */
+ pvoice->count = runtime->period_size;
+
+ /* set Loop Back Address */
+ LBA = runtime->dma_addr;
+
+ /* set ESO to capture first MIDLP interrupt */
+ ESO = pvoice->eso - 1;
+ CTRL = snd_ali_control_mode(substream);
+ GVSEL = 0;
+ PAN = 0x00;
+ VOL = 0x00;
+ EC = 0;
+
+ snd_ali_write_voice_regs( codec,
+ pvoice->number,
+ LBA,
+ 0, /* cso */
+ ESO,
+ Delta,
+ 0, /* alpha */
+ GVSEL,
+ PAN,
+ VOL,
+ CTRL,
+ EC);
+
+ spin_unlock_irq(&codec->reg_lock);
+
+ return 0;
+}
+
+
+static snd_pcm_uframes_t
+snd_ali_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ali_voice *pvoice = runtime->private_data;
+ unsigned int cso;
+
+ spin_lock(&codec->reg_lock);
+ if (!pvoice->running) {
+ spin_unlock(&codec->reg_lock);
+ return 0;
+ }
+ outb(pvoice->number, ALI_REG(codec, ALI_GC_CIR));
+ cso = inw(ALI_REG(codec, ALI_CSO_ALPHA_FMS + 2));
+ spin_unlock(&codec->reg_lock);
+ snd_ali_printk("playback pointer returned cso=%xh.\n", cso);
+
+ return cso;
+}
+
+
+static snd_pcm_uframes_t snd_ali_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ali_voice *pvoice = runtime->private_data;
+ unsigned int cso;
+
+ spin_lock(&codec->reg_lock);
+ if (!pvoice->running) {
+ spin_unlock_irq(&codec->reg_lock);
+ return 0;
+ }
+ outb(pvoice->number, ALI_REG(codec, ALI_GC_CIR));
+ cso = inw(ALI_REG(codec, ALI_CSO_ALPHA_FMS + 2));
+ spin_unlock(&codec->reg_lock);
+
+ return cso;
+}
+
+static struct snd_pcm_hardware snd_ali_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (256*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (256*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+ * Capture support device description
+ */
+
+static struct snd_pcm_hardware snd_ali_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static void snd_ali_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ struct snd_ali_voice *pvoice = runtime->private_data;
+ struct snd_ali *codec;
+
+ if (pvoice) {
+ codec = pvoice->codec;
+ snd_ali_free_voice(pvoice->codec, pvoice);
+ }
+}
+
+static int snd_ali_open(struct snd_pcm_substream *substream, int rec,
+ int channel, struct snd_pcm_hardware *phw)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ali_voice *pvoice;
+
+ pvoice = snd_ali_alloc_voice(codec, SNDRV_ALI_VOICE_TYPE_PCM, rec,
+ channel);
+ if (!pvoice)
+ return -EAGAIN;
+
+ pvoice->substream = substream;
+ runtime->private_data = pvoice;
+ runtime->private_free = snd_ali_pcm_free_substream;
+
+ runtime->hw = *phw;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ 0, 64*1024);
+ return 0;
+}
+
+static int snd_ali_playback_open(struct snd_pcm_substream *substream)
+{
+ return snd_ali_open(substream, 0, -1, &snd_ali_playback);
+}
+
+static int snd_ali_capture_open(struct snd_pcm_substream *substream)
+{
+ return snd_ali_open(substream, 1, -1, &snd_ali_capture);
+}
+
+static int snd_ali_playback_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static int snd_ali_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ali *codec = snd_pcm_substream_chip(substream);
+ struct snd_ali_voice *pvoice = substream->runtime->private_data;
+
+ snd_ali_disable_special_channel(codec,pvoice->number);
+
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ali_playback_ops = {
+ .open = snd_ali_playback_open,
+ .close = snd_ali_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ali_playback_hw_params,
+ .hw_free = snd_ali_playback_hw_free,
+ .prepare = snd_ali_playback_prepare,
+ .trigger = snd_ali_trigger,
+ .pointer = snd_ali_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_ali_capture_ops = {
+ .open = snd_ali_capture_open,
+ .close = snd_ali_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ali_hw_params,
+ .hw_free = snd_ali_hw_free,
+ .prepare = snd_ali_prepare,
+ .trigger = snd_ali_trigger,
+ .pointer = snd_ali_pointer,
+};
+
+/*
+ * Modem PCM
+ */
+
+static int snd_ali_modem_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_ali *chip = snd_pcm_substream_chip(substream);
+ unsigned int modem_num = chip->num_of_codecs - 1;
+ snd_ac97_write(chip->ac97[modem_num], AC97_LINE1_RATE,
+ params_rate(hw_params));
+ snd_ac97_write(chip->ac97[modem_num], AC97_LINE1_LEVEL, 0);
+ return snd_ali_hw_params(substream, hw_params);
+}
+
+static struct snd_pcm_hardware snd_ali_modem =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000),
+ .rate_min = 8000,
+ .rate_max = 16000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = (256*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (256*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_ali_modem_open(struct snd_pcm_substream *substream, int rec,
+ int channel)
+{
+ static unsigned int rates[] = {8000, 9600, 12000, 16000};
+ static struct snd_pcm_hw_constraint_list hw_constraint_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+ int err = snd_ali_open(substream, rec, channel, &snd_ali_modem);
+
+ if (err)
+ return err;
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraint_rates);
+}
+
+static int snd_ali_modem_playback_open(struct snd_pcm_substream *substream)
+{
+ return snd_ali_modem_open(substream, 0, ALI_MODEM_OUT_CHANNEL);
+}
+
+static int snd_ali_modem_capture_open(struct snd_pcm_substream *substream)
+{
+ return snd_ali_modem_open(substream, 1, ALI_MODEM_IN_CHANNEL);
+}
+
+static struct snd_pcm_ops snd_ali_modem_playback_ops = {
+ .open = snd_ali_modem_playback_open,
+ .close = snd_ali_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ali_modem_hw_params,
+ .hw_free = snd_ali_hw_free,
+ .prepare = snd_ali_prepare,
+ .trigger = snd_ali_trigger,
+ .pointer = snd_ali_pointer,
+};
+
+static struct snd_pcm_ops snd_ali_modem_capture_ops = {
+ .open = snd_ali_modem_capture_open,
+ .close = snd_ali_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ali_modem_hw_params,
+ .hw_free = snd_ali_hw_free,
+ .prepare = snd_ali_prepare,
+ .trigger = snd_ali_trigger,
+ .pointer = snd_ali_pointer,
+};
+
+
+struct ali_pcm_description {
+ char *name;
+ unsigned int playback_num;
+ unsigned int capture_num;
+ struct snd_pcm_ops *playback_ops;
+ struct snd_pcm_ops *capture_ops;
+ unsigned short class;
+};
+
+
+static void snd_ali_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_ali *codec = pcm->private_data;
+ codec->pcm[pcm->device] = NULL;
+}
+
+
+static int __devinit snd_ali_pcm(struct snd_ali * codec, int device,
+ struct ali_pcm_description *desc)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(codec->card, desc->name, device,
+ desc->playback_num, desc->capture_num, &pcm);
+ if (err < 0) {
+ snd_printk(KERN_ERR "snd_ali_pcm: err called snd_pcm_new.\n");
+ return err;
+ }
+ pcm->private_data = codec;
+ pcm->private_free = snd_ali_pcm_free;
+ if (desc->playback_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ desc->playback_ops);
+ if (desc->capture_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ desc->capture_ops);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(codec->pci),
+ 64*1024, 128*1024);
+
+ pcm->info_flags = 0;
+ pcm->dev_class = desc->class;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strcpy(pcm->name, desc->name);
+ codec->pcm[0] = pcm;
+ return 0;
+}
+
+static struct ali_pcm_description ali_pcms[] = {
+ { .name = "ALI 5451",
+ .playback_num = ALI_CHANNELS,
+ .capture_num = 1,
+ .playback_ops = &snd_ali_playback_ops,
+ .capture_ops = &snd_ali_capture_ops
+ },
+ { .name = "ALI 5451 modem",
+ .playback_num = 1,
+ .capture_num = 1,
+ .playback_ops = &snd_ali_modem_playback_ops,
+ .capture_ops = &snd_ali_modem_capture_ops,
+ .class = SNDRV_PCM_CLASS_MODEM
+ }
+};
+
+static int __devinit snd_ali_build_pcms(struct snd_ali *codec)
+{
+ int i, err;
+ for (i = 0; i < codec->num_of_codecs && i < ARRAY_SIZE(ali_pcms); i++) {
+ err = snd_ali_pcm(codec, i, &ali_pcms[i]);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+
+#define ALI5451_SPDIF(xname, xindex, value) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\
+.info = snd_ali5451_spdif_info, .get = snd_ali5451_spdif_get, \
+.put = snd_ali5451_spdif_put, .private_value = value}
+
+#define snd_ali5451_spdif_info snd_ctl_boolean_mono_info
+
+static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ali *codec = kcontrol->private_data;
+ unsigned int spdif_enable;
+
+ spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0;
+
+ spin_lock_irq(&codec->reg_lock);
+ switch (kcontrol->private_value) {
+ case 0:
+ spdif_enable = (codec->spdif_mask & 0x02) ? 1 : 0;
+ break;
+ case 1:
+ spdif_enable = ((codec->spdif_mask & 0x02) &&
+ (codec->spdif_mask & 0x04)) ? 1 : 0;
+ break;
+ case 2:
+ spdif_enable = (codec->spdif_mask & 0x01) ? 1 : 0;
+ break;
+ default:
+ break;
+ }
+ ucontrol->value.integer.value[0] = spdif_enable;
+ spin_unlock_irq(&codec->reg_lock);
+ return 0;
+}
+
+static int snd_ali5451_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ali *codec = kcontrol->private_data;
+ unsigned int change = 0, spdif_enable = 0;
+
+ spdif_enable = ucontrol->value.integer.value[0] ? 1 : 0;
+
+ spin_lock_irq(&codec->reg_lock);
+ switch (kcontrol->private_value) {
+ case 0:
+ change = (codec->spdif_mask & 0x02) ? 1 : 0;
+ change = change ^ spdif_enable;
+ if (change) {
+ if (spdif_enable) {
+ codec->spdif_mask |= 0x02;
+ snd_ali_enable_spdif_out(codec);
+ } else {
+ codec->spdif_mask &= ~(0x02);
+ codec->spdif_mask &= ~(0x04);
+ snd_ali_disable_spdif_out(codec);
+ }
+ }
+ break;
+ case 1:
+ change = (codec->spdif_mask & 0x04) ? 1 : 0;
+ change = change ^ spdif_enable;
+ if (change && (codec->spdif_mask & 0x02)) {
+ if (spdif_enable) {
+ codec->spdif_mask |= 0x04;
+ snd_ali_enable_spdif_chnout(codec);
+ } else {
+ codec->spdif_mask &= ~(0x04);
+ snd_ali_disable_spdif_chnout(codec);
+ }
+ }
+ break;
+ case 2:
+ change = (codec->spdif_mask & 0x01) ? 1 : 0;
+ change = change ^ spdif_enable;
+ if (change) {
+ if (spdif_enable) {
+ codec->spdif_mask |= 0x01;
+ snd_ali_enable_spdif_in(codec);
+ } else {
+ codec->spdif_mask &= ~(0x01);
+ snd_ali_disable_spdif_in(codec);
+ }
+ }
+ break;
+ default:
+ break;
+ }
+ spin_unlock_irq(&codec->reg_lock);
+
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ali5451_mixer_spdif[] __devinitdata = {
+ /* spdif aplayback switch */
+ /* FIXME: "IEC958 Playback Switch" may conflict with one on ac97_codec */
+ ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH), 0, 0),
+ /* spdif out to spdif channel */
+ ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("Channel Output ",NONE,SWITCH), 0, 1),
+ /* spdif in from spdif channel */
+ ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, 2)
+};
+
+static int __devinit snd_ali_mixer(struct snd_ali * codec)
+{
+ struct snd_ac97_template ac97;
+ unsigned int idx;
+ int i, err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_ali_codec_write,
+ .read = snd_ali_codec_read,
+ };
+
+ err = snd_ac97_bus(codec->card, 0, &ops, codec, &codec->ac97_bus);
+ if (err < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = codec;
+
+ for (i = 0; i < codec->num_of_codecs; i++) {
+ ac97.num = i;
+ err = snd_ac97_mixer(codec->ac97_bus, &ac97, &codec->ac97[i]);
+ if (err < 0) {
+ snd_printk(KERN_ERR
+ "ali mixer %d creating error.\n", i);
+ if (i == 0)
+ return err;
+ codec->num_of_codecs = 1;
+ break;
+ }
+ }
+
+ if (codec->spdif_support) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_ali5451_mixer_spdif); idx++) {
+ err = snd_ctl_add(codec->card,
+ snd_ctl_new1(&snd_ali5451_mixer_spdif[idx], codec));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int ali_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_ali *chip = card->private_data;
+ struct snd_ali_image *im;
+ int i, j;
+
+ im = chip->image;
+ if (!im)
+ return 0;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < chip->num_of_codecs; i++) {
+ snd_pcm_suspend_all(chip->pcm[i]);
+ snd_ac97_suspend(chip->ac97[i]);
+ }
+
+ spin_lock_irq(&chip->reg_lock);
+
+ im->regs[ALI_MISCINT >> 2] = inl(ALI_REG(chip, ALI_MISCINT));
+ /* im->regs[ALI_START >> 2] = inl(ALI_REG(chip, ALI_START)); */
+ im->regs[ALI_STOP >> 2] = inl(ALI_REG(chip, ALI_STOP));
+
+ /* disable all IRQ bits */
+ outl(0, ALI_REG(chip, ALI_MISCINT));
+
+ for (i = 0; i < ALI_GLOBAL_REGS; i++) {
+ if ((i*4 == ALI_MISCINT) || (i*4 == ALI_STOP))
+ continue;
+ im->regs[i] = inl(ALI_REG(chip, i*4));
+ }
+
+ for (i = 0; i < ALI_CHANNELS; i++) {
+ outb(i, ALI_REG(chip, ALI_GC_CIR));
+ for (j = 0; j < ALI_CHANNEL_REGS; j++)
+ im->channel_regs[i][j] = inl(ALI_REG(chip, j*4 + 0xe0));
+ }
+
+ /* stop all HW channel */
+ outl(0xffffffff, ALI_REG(chip, ALI_STOP));
+
+ spin_unlock_irq(&chip->reg_lock);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int ali_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_ali *chip = card->private_data;
+ struct snd_ali_image *im;
+ int i, j;
+
+ im = chip->image;
+ if (!im)
+ return 0;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "ali5451: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ spin_lock_irq(&chip->reg_lock);
+
+ for (i = 0; i < ALI_CHANNELS; i++) {
+ outb(i, ALI_REG(chip, ALI_GC_CIR));
+ for (j = 0; j < ALI_CHANNEL_REGS; j++)
+ outl(im->channel_regs[i][j], ALI_REG(chip, j*4 + 0xe0));
+ }
+
+ for (i = 0; i < ALI_GLOBAL_REGS; i++) {
+ if ((i*4 == ALI_MISCINT) || (i*4 == ALI_STOP) ||
+ (i*4 == ALI_START))
+ continue;
+ outl(im->regs[i], ALI_REG(chip, i*4));
+ }
+
+ /* start HW channel */
+ outl(im->regs[ALI_START >> 2], ALI_REG(chip, ALI_START));
+ /* restore IRQ enable bits */
+ outl(im->regs[ALI_MISCINT >> 2], ALI_REG(chip, ALI_MISCINT));
+
+ spin_unlock_irq(&chip->reg_lock);
+
+ for (i = 0 ; i < chip->num_of_codecs; i++)
+ snd_ac97_resume(chip->ac97[i]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static int snd_ali_free(struct snd_ali * codec)
+{
+ if (codec->hw_initialized)
+ snd_ali_disable_address_interrupt(codec);
+ if (codec->irq >= 0)
+ free_irq(codec->irq, codec);
+ if (codec->port)
+ pci_release_regions(codec->pci);
+ pci_disable_device(codec->pci);
+#ifdef CONFIG_PM
+ kfree(codec->image);
+#endif
+ pci_dev_put(codec->pci_m1533);
+ pci_dev_put(codec->pci_m7101);
+ kfree(codec);
+ return 0;
+}
+
+static int snd_ali_chip_init(struct snd_ali *codec)
+{
+ unsigned int legacy;
+ unsigned char temp;
+ struct pci_dev *pci_dev;
+
+ snd_ali_printk("chip initializing ... \n");
+
+ if (snd_ali_reset_5451(codec)) {
+ snd_printk(KERN_ERR "ali_chip_init: reset 5451 error.\n");
+ return -1;
+ }
+
+ if (codec->revision == ALI_5451_V02) {
+ pci_dev = codec->pci_m1533;
+ pci_read_config_byte(pci_dev, 0x59, &temp);
+ temp |= 0x80;
+ pci_write_config_byte(pci_dev, 0x59, temp);
+
+ pci_dev = codec->pci_m7101;
+ pci_read_config_byte(pci_dev, 0xb8, &temp);
+ temp |= 0x20;
+ pci_write_config_byte(pci_dev, 0xB8, temp);
+ }
+
+ pci_read_config_dword(codec->pci, 0x44, &legacy);
+ legacy &= 0xff00ff00;
+ legacy |= 0x000800aa;
+ pci_write_config_dword(codec->pci, 0x44, legacy);
+
+ outl(0x80000001, ALI_REG(codec, ALI_GLOBAL_CONTROL));
+ outl(0x00000000, ALI_REG(codec, ALI_AINTEN));
+ outl(0xffffffff, ALI_REG(codec, ALI_AINT));
+ outl(0x00000000, ALI_REG(codec, ALI_VOLUME));
+ outb(0x10, ALI_REG(codec, ALI_MPUR2));
+
+ codec->ac97_ext_id = snd_ali_codec_peek(codec, 0, AC97_EXTENDED_ID);
+ codec->ac97_ext_status = snd_ali_codec_peek(codec, 0,
+ AC97_EXTENDED_STATUS);
+ if (codec->spdif_support) {
+ snd_ali_enable_spdif_out(codec);
+ codec->spdif_mask = 0x00000002;
+ }
+
+ codec->num_of_codecs = 1;
+
+ /* secondary codec - modem */
+ if (inl(ALI_REG(codec, ALI_SCTRL)) & ALI_SCTRL_CODEC2_READY) {
+ codec->num_of_codecs++;
+ outl(inl(ALI_REG(codec, ALI_SCTRL)) |
+ (ALI_SCTRL_LINE_IN2 | ALI_SCTRL_GPIO_IN2 |
+ ALI_SCTRL_LINE_OUT_EN),
+ ALI_REG(codec, ALI_SCTRL));
+ }
+
+ snd_ali_printk("chip initialize succeed.\n");
+ return 0;
+
+}
+
+/* proc for register dump */
+static void snd_ali_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buf)
+{
+ struct snd_ali *codec = entry->private_data;
+ int i;
+ for (i = 0; i < 256 ; i+= 4)
+ snd_iprintf(buf, "%02x: %08x\n", i, inl(ALI_REG(codec, i)));
+}
+
+static void __devinit snd_ali_proc_init(struct snd_ali *codec)
+{
+ struct snd_info_entry *entry;
+ if (!snd_card_proc_new(codec->card, "ali5451", &entry))
+ snd_info_set_text_ops(entry, codec, snd_ali_proc_read);
+}
+
+static int __devinit snd_ali_resources(struct snd_ali *codec)
+{
+ int err;
+
+ snd_ali_printk("resouces allocation ...\n");
+ err = pci_request_regions(codec->pci, "ALI 5451");
+ if (err < 0)
+ return err;
+ codec->port = pci_resource_start(codec->pci, 0);
+
+ if (request_irq(codec->pci->irq, snd_ali_card_interrupt,
+ IRQF_SHARED, "ALI 5451", codec)) {
+ snd_printk(KERN_ERR "Unable to request irq.\n");
+ return -EBUSY;
+ }
+ codec->irq = codec->pci->irq;
+ snd_ali_printk("resouces allocated.\n");
+ return 0;
+}
+static int snd_ali_dev_free(struct snd_device *device)
+{
+ struct snd_ali *codec = device->device_data;
+ snd_ali_free(codec);
+ return 0;
+}
+
+static int __devinit snd_ali_create(struct snd_card *card,
+ struct pci_dev *pci,
+ int pcm_streams,
+ int spdif_support,
+ struct snd_ali ** r_ali)
+{
+ struct snd_ali *codec;
+ int i, err;
+ unsigned short cmdw;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ali_dev_free,
+ };
+
+ *r_ali = NULL;
+
+ snd_ali_printk("creating ...\n");
+
+ /* enable PCI device */
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+ /* check, if we can restrict PCI DMA transfers to 31 bits */
+ if (pci_set_dma_mask(pci, DMA_31BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_31BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support "
+ "31bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ codec = kzalloc(sizeof(*codec), GFP_KERNEL);
+ if (!codec) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&codec->reg_lock);
+ spin_lock_init(&codec->voice_alloc);
+
+ codec->card = card;
+ codec->pci = pci;
+ codec->irq = -1;
+ codec->revision = pci->revision;
+ codec->spdif_support = spdif_support;
+
+ if (pcm_streams < 1)
+ pcm_streams = 1;
+ if (pcm_streams > 32)
+ pcm_streams = 32;
+
+ pci_set_master(pci);
+ pci_read_config_word(pci, PCI_COMMAND, &cmdw);
+ if ((cmdw & PCI_COMMAND_IO) != PCI_COMMAND_IO) {
+ cmdw |= PCI_COMMAND_IO;
+ pci_write_config_word(pci, PCI_COMMAND, cmdw);
+ }
+ pci_set_master(pci);
+
+ if (snd_ali_resources(codec)) {
+ snd_ali_free(codec);
+ return -EBUSY;
+ }
+
+ synchronize_irq(pci->irq);
+
+ codec->synth.chmap = 0;
+ codec->synth.chcnt = 0;
+ codec->spdif_mask = 0;
+ codec->synth.synthcount = 0;
+
+ if (codec->revision == ALI_5451_V02)
+ codec->chregs.regs.ac97read = ALI_AC97_WRITE;
+ else
+ codec->chregs.regs.ac97read = ALI_AC97_READ;
+ codec->chregs.regs.ac97write = ALI_AC97_WRITE;
+
+ codec->chregs.regs.start = ALI_START;
+ codec->chregs.regs.stop = ALI_STOP;
+ codec->chregs.regs.aint = ALI_AINT;
+ codec->chregs.regs.ainten = ALI_AINTEN;
+
+ codec->chregs.data.start = 0x00;
+ codec->chregs.data.stop = 0x00;
+ codec->chregs.data.aint = 0x00;
+ codec->chregs.data.ainten = 0x00;
+
+ /* M1533: southbridge */
+ codec->pci_m1533 = pci_get_device(0x10b9, 0x1533, NULL);
+ if (!codec->pci_m1533) {
+ snd_printk(KERN_ERR "ali5451: cannot find ALi 1533 chip.\n");
+ snd_ali_free(codec);
+ return -ENODEV;
+ }
+ /* M7101: power management */
+ codec->pci_m7101 = pci_get_device(0x10b9, 0x7101, NULL);
+ if (!codec->pci_m7101 && codec->revision == ALI_5451_V02) {
+ snd_printk(KERN_ERR "ali5451: cannot find ALi 7101 chip.\n");
+ snd_ali_free(codec);
+ return -ENODEV;
+ }
+
+ snd_ali_printk("snd_device_new is called.\n");
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, codec, &ops);
+ if (err < 0) {
+ snd_ali_free(codec);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ /* initialise synth voices*/
+ for (i = 0; i < ALI_CHANNELS; i++)
+ codec->synth.voices[i].number = i;
+
+ err = snd_ali_chip_init(codec);
+ if (err < 0) {
+ snd_printk(KERN_ERR "ali create: chip init error.\n");
+ return err;
+ }
+
+#ifdef CONFIG_PM
+ codec->image = kmalloc(sizeof(*codec->image), GFP_KERNEL);
+ if (!codec->image)
+ snd_printk(KERN_WARNING "can't allocate apm buffer\n");
+#endif
+
+ snd_ali_enable_address_interrupt(codec);
+ codec->hw_initialized = 1;
+
+ *r_ali = codec;
+ snd_ali_printk("created.\n");
+ return 0;
+}
+
+static int __devinit snd_ali_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct snd_ali *codec;
+ int err;
+
+ snd_ali_printk("probe ...\n");
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (!card)
+ return -ENOMEM;
+
+ err = snd_ali_create(card, pci, pcm_channels, spdif, &codec);
+ if (err < 0)
+ goto error;
+ card->private_data = codec;
+
+ snd_ali_printk("mixer building ...\n");
+ err = snd_ali_mixer(codec);
+ if (err < 0)
+ goto error;
+
+ snd_ali_printk("pcm building ...\n");
+ err = snd_ali_build_pcms(codec);
+ if (err < 0)
+ goto error;
+
+ snd_ali_proc_init(codec);
+
+ strcpy(card->driver, "ALI5451");
+ strcpy(card->shortname, "ALI 5451");
+
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, codec->port, codec->irq);
+
+ snd_ali_printk("register card.\n");
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ pci_set_drvdata(pci, card);
+ return 0;
+
+ error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_ali_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "ALI 5451",
+ .id_table = snd_ali_ids,
+ .probe = snd_ali_probe,
+ .remove = __devexit_p(snd_ali_remove),
+#ifdef CONFIG_PM
+ .suspend = ali_suspend,
+ .resume = ali_resume,
+#endif
+};
+
+static int __init alsa_card_ali_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_ali_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_ali_init)
+module_exit(alsa_card_ali_exit)
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
new file mode 100644
index 0000000..8df6824
--- /dev/null
+++ b/sound/pci/als300.c
@@ -0,0 +1,870 @@
+/*
+ * als300.c - driver for Avance Logic ALS300/ALS300+ soundcards.
+ * Copyright (C) 2005 by Ash Willis <ashwillis@programmer.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * TODO
+ * 4 channel playback for ALS300+
+ * gameport
+ * mpu401
+ * opl3
+ *
+ * NOTES
+ * The BLOCK_COUNTER registers for the ALS300(+) return a figure related to
+ * the position in the current period, NOT the whole buffer. It is important
+ * to know which period we are in so we can calculate the correct pointer.
+ * This is why we always use 2 periods. We can then use a flip-flop variable
+ * to keep track of what period we are in.
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+
+#include <asm/io.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/ac97_codec.h>
+#include <sound/opl3.h>
+
+/* snd_als300_set_irq_flag */
+#define IRQ_DISABLE 0
+#define IRQ_ENABLE 1
+
+/* I/O port layout */
+#define AC97_ACCESS 0x00
+#define AC97_READ 0x04
+#define AC97_STATUS 0x06
+#define AC97_DATA_AVAIL (1<<6)
+#define AC97_BUSY (1<<7)
+#define ALS300_IRQ_STATUS 0x07 /* ALS300 Only */
+#define IRQ_PLAYBACK (1<<3)
+#define IRQ_CAPTURE (1<<2)
+#define GCR_DATA 0x08
+#define GCR_INDEX 0x0C
+#define ALS300P_DRAM_IRQ_STATUS 0x0D /* ALS300+ Only */
+#define MPU_IRQ_STATUS 0x0E /* ALS300 Rev. E+, ALS300+ */
+#define ALS300P_IRQ_STATUS 0x0F /* ALS300+ Only */
+
+/* General Control Registers */
+#define PLAYBACK_START 0x80
+#define PLAYBACK_END 0x81
+#define PLAYBACK_CONTROL 0x82
+#define TRANSFER_START (1<<16)
+#define FIFO_PAUSE (1<<17)
+#define RECORD_START 0x83
+#define RECORD_END 0x84
+#define RECORD_CONTROL 0x85
+#define DRAM_WRITE_CONTROL 0x8B
+#define WRITE_TRANS_START (1<<16)
+#define DRAM_MODE_2 (1<<17)
+#define MISC_CONTROL 0x8C
+#define IRQ_SET_BIT (1<<15)
+#define VMUTE_NORMAL (1<<20)
+#define MMUTE_NORMAL (1<<21)
+#define MUS_VOC_VOL 0x8E
+#define PLAYBACK_BLOCK_COUNTER 0x9A
+#define RECORD_BLOCK_COUNTER 0x9B
+
+#define DEBUG_CALLS 0
+#define DEBUG_PLAY_REC 0
+
+#if DEBUG_CALLS
+#define snd_als300_dbgcalls(format, args...) printk(format, ##args)
+#define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__)
+#define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__)
+#else
+#define snd_als300_dbgcalls(format, args...)
+#define snd_als300_dbgcallenter()
+#define snd_als300_dbgcallleave()
+#endif
+
+#if DEBUG_PLAY_REC
+#define snd_als300_dbgplay(format, args...) printk(KERN_ERR format, ##args)
+#else
+#define snd_als300_dbgplay(format, args...)
+#endif
+
+enum {DEVICE_ALS300, DEVICE_ALS300_PLUS};
+
+MODULE_AUTHOR("Ash Willis <ashwillis@programmer.net>");
+MODULE_DESCRIPTION("Avance Logic ALS300");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS300},{Avance Logic,ALS300+}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+struct snd_als300 {
+ unsigned long port;
+ spinlock_t reg_lock;
+ struct snd_card *card;
+ struct pci_dev *pci;
+
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ struct snd_ac97 *ac97;
+ struct snd_opl3 *opl3;
+
+ struct resource *res_port;
+
+ int irq;
+
+ int chip_type; /* ALS300 or ALS300+ */
+
+ char revision;
+};
+
+struct snd_als300_substream_data {
+ int period_flipflop;
+ int control_register;
+ int block_counter_register;
+};
+
+static struct pci_device_id snd_als300_ids[] = {
+ { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 },
+ { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_als300_ids);
+
+static inline u32 snd_als300_gcr_read(unsigned long port, unsigned short reg)
+{
+ outb(reg, port+GCR_INDEX);
+ return inl(port+GCR_DATA);
+}
+
+static inline void snd_als300_gcr_write(unsigned long port,
+ unsigned short reg, u32 val)
+{
+ outb(reg, port+GCR_INDEX);
+ outl(val, port+GCR_DATA);
+}
+
+/* Enable/Disable Interrupts */
+static void snd_als300_set_irq_flag(struct snd_als300 *chip, int cmd)
+{
+ u32 tmp = snd_als300_gcr_read(chip->port, MISC_CONTROL);
+ snd_als300_dbgcallenter();
+
+ /* boolean XOR check, since old vs. new hardware have
+ directly reversed bit setting for ENABLE and DISABLE.
+ ALS300+ acts like newer versions of ALS300 */
+ if (((chip->revision > 5 || chip->chip_type == DEVICE_ALS300_PLUS) ^
+ (cmd == IRQ_ENABLE)) == 0)
+ tmp |= IRQ_SET_BIT;
+ else
+ tmp &= ~IRQ_SET_BIT;
+ snd_als300_gcr_write(chip->port, MISC_CONTROL, tmp);
+ snd_als300_dbgcallleave();
+}
+
+static int snd_als300_free(struct snd_als300 *chip)
+{
+ snd_als300_dbgcallenter();
+ snd_als300_set_irq_flag(chip, IRQ_DISABLE);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static int snd_als300_dev_free(struct snd_device *device)
+{
+ struct snd_als300 *chip = device->device_data;
+ return snd_als300_free(chip);
+}
+
+static irqreturn_t snd_als300_interrupt(int irq, void *dev_id)
+{
+ u8 status;
+ struct snd_als300 *chip = dev_id;
+ struct snd_als300_substream_data *data;
+
+ status = inb(chip->port+ALS300_IRQ_STATUS);
+ if (!status) /* shared IRQ, for different device?? Exit ASAP! */
+ return IRQ_NONE;
+
+ /* ACK everything ASAP */
+ outb(status, chip->port+ALS300_IRQ_STATUS);
+ if (status & IRQ_PLAYBACK) {
+ if (chip->pcm && chip->playback_substream) {
+ data = chip->playback_substream->runtime->private_data;
+ data->period_flipflop ^= 1;
+ snd_pcm_period_elapsed(chip->playback_substream);
+ snd_als300_dbgplay("IRQ_PLAYBACK\n");
+ }
+ }
+ if (status & IRQ_CAPTURE) {
+ if (chip->pcm && chip->capture_substream) {
+ data = chip->capture_substream->runtime->private_data;
+ data->period_flipflop ^= 1;
+ snd_pcm_period_elapsed(chip->capture_substream);
+ snd_als300_dbgplay("IRQ_CAPTURE\n");
+ }
+ }
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t snd_als300plus_interrupt(int irq, void *dev_id)
+{
+ u8 general, mpu, dram;
+ struct snd_als300 *chip = dev_id;
+ struct snd_als300_substream_data *data;
+
+ general = inb(chip->port+ALS300P_IRQ_STATUS);
+ mpu = inb(chip->port+MPU_IRQ_STATUS);
+ dram = inb(chip->port+ALS300P_DRAM_IRQ_STATUS);
+
+ /* shared IRQ, for different device?? Exit ASAP! */
+ if ((general == 0) && ((mpu & 0x80) == 0) && ((dram & 0x01) == 0))
+ return IRQ_NONE;
+
+ if (general & IRQ_PLAYBACK) {
+ if (chip->pcm && chip->playback_substream) {
+ outb(IRQ_PLAYBACK, chip->port+ALS300P_IRQ_STATUS);
+ data = chip->playback_substream->runtime->private_data;
+ data->period_flipflop ^= 1;
+ snd_pcm_period_elapsed(chip->playback_substream);
+ snd_als300_dbgplay("IRQ_PLAYBACK\n");
+ }
+ }
+ if (general & IRQ_CAPTURE) {
+ if (chip->pcm && chip->capture_substream) {
+ outb(IRQ_CAPTURE, chip->port+ALS300P_IRQ_STATUS);
+ data = chip->capture_substream->runtime->private_data;
+ data->period_flipflop ^= 1;
+ snd_pcm_period_elapsed(chip->capture_substream);
+ snd_als300_dbgplay("IRQ_CAPTURE\n");
+ }
+ }
+ /* FIXME: Ack other interrupt types. Not important right now as
+ * those other devices aren't enabled. */
+ return IRQ_HANDLED;
+}
+
+static void __devexit snd_als300_remove(struct pci_dev *pci)
+{
+ snd_als300_dbgcallenter();
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+ snd_als300_dbgcallleave();
+}
+
+static unsigned short snd_als300_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ int i;
+ struct snd_als300 *chip = ac97->private_data;
+
+ for (i = 0; i < 1000; i++) {
+ if ((inb(chip->port+AC97_STATUS) & (AC97_BUSY)) == 0)
+ break;
+ udelay(10);
+ }
+ outl((reg << 24) | (1 << 31), chip->port+AC97_ACCESS);
+
+ for (i = 0; i < 1000; i++) {
+ if ((inb(chip->port+AC97_STATUS) & (AC97_DATA_AVAIL)) != 0)
+ break;
+ udelay(10);
+ }
+ return inw(chip->port+AC97_READ);
+}
+
+static void snd_als300_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ int i;
+ struct snd_als300 *chip = ac97->private_data;
+
+ for (i = 0; i < 1000; i++) {
+ if ((inb(chip->port+AC97_STATUS) & (AC97_BUSY)) == 0)
+ break;
+ udelay(10);
+ }
+ outl((reg << 24) | val, chip->port+AC97_ACCESS);
+}
+
+static int snd_als300_ac97(struct snd_als300 *chip)
+{
+ struct snd_ac97_bus *bus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_als300_ac97_write,
+ .read = snd_als300_ac97_read,
+ };
+
+ snd_als300_dbgcallenter();
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus)) < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+
+ snd_als300_dbgcallleave();
+ return snd_ac97_mixer(bus, &ac97, &chip->ac97);
+}
+
+/* hardware definition
+ *
+ * In AC97 mode, we always use 48k/16bit/stereo.
+ * Any request to change data type is ignored by
+ * the card when it is running outside of legacy
+ * mode.
+ */
+static struct snd_pcm_hardware snd_als300_playback_hw =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = 32 * 1024,
+ .periods_min = 2,
+ .periods_max = 2,
+};
+
+static struct snd_pcm_hardware snd_als300_capture_hw =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = 32 * 1024,
+ .periods_min = 2,
+ .periods_max = 2,
+};
+
+static int snd_als300_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_als300_substream_data *data = kzalloc(sizeof(*data),
+ GFP_KERNEL);
+
+ snd_als300_dbgcallenter();
+ chip->playback_substream = substream;
+ runtime->hw = snd_als300_playback_hw;
+ runtime->private_data = data;
+ data->control_register = PLAYBACK_CONTROL;
+ data->block_counter_register = PLAYBACK_BLOCK_COUNTER;
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static int snd_als300_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ struct snd_als300_substream_data *data;
+
+ data = substream->runtime->private_data;
+ snd_als300_dbgcallenter();
+ kfree(data);
+ chip->playback_substream = NULL;
+ snd_pcm_lib_free_pages(substream);
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static int snd_als300_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_als300_substream_data *data = kzalloc(sizeof(*data),
+ GFP_KERNEL);
+
+ snd_als300_dbgcallenter();
+ chip->capture_substream = substream;
+ runtime->hw = snd_als300_capture_hw;
+ runtime->private_data = data;
+ data->control_register = RECORD_CONTROL;
+ data->block_counter_register = RECORD_BLOCK_COUNTER;
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static int snd_als300_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ struct snd_als300_substream_data *data;
+
+ data = substream->runtime->private_data;
+ snd_als300_dbgcallenter();
+ kfree(data);
+ chip->capture_substream = NULL;
+ snd_pcm_lib_free_pages(substream);
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static int snd_als300_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int snd_als300_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_als300_playback_prepare(struct snd_pcm_substream *substream)
+{
+ u32 tmp;
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short period_bytes = snd_pcm_lib_period_bytes(substream);
+ unsigned short buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+
+ snd_als300_dbgcallenter();
+ spin_lock_irq(&chip->reg_lock);
+ tmp = snd_als300_gcr_read(chip->port, PLAYBACK_CONTROL);
+ tmp &= ~TRANSFER_START;
+
+ snd_als300_dbgplay("Period bytes: %d Buffer bytes %d\n",
+ period_bytes, buffer_bytes);
+
+ /* set block size */
+ tmp &= 0xffff0000;
+ tmp |= period_bytes - 1;
+ snd_als300_gcr_write(chip->port, PLAYBACK_CONTROL, tmp);
+
+ /* set dma area */
+ snd_als300_gcr_write(chip->port, PLAYBACK_START,
+ runtime->dma_addr);
+ snd_als300_gcr_write(chip->port, PLAYBACK_END,
+ runtime->dma_addr + buffer_bytes - 1);
+ spin_unlock_irq(&chip->reg_lock);
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static int snd_als300_capture_prepare(struct snd_pcm_substream *substream)
+{
+ u32 tmp;
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short period_bytes = snd_pcm_lib_period_bytes(substream);
+ unsigned short buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+
+ snd_als300_dbgcallenter();
+ spin_lock_irq(&chip->reg_lock);
+ tmp = snd_als300_gcr_read(chip->port, RECORD_CONTROL);
+ tmp &= ~TRANSFER_START;
+
+ snd_als300_dbgplay("Period bytes: %d Buffer bytes %d\n", period_bytes,
+ buffer_bytes);
+
+ /* set block size */
+ tmp &= 0xffff0000;
+ tmp |= period_bytes - 1;
+
+ /* set dma area */
+ snd_als300_gcr_write(chip->port, RECORD_CONTROL, tmp);
+ snd_als300_gcr_write(chip->port, RECORD_START,
+ runtime->dma_addr);
+ snd_als300_gcr_write(chip->port, RECORD_END,
+ runtime->dma_addr + buffer_bytes - 1);
+ spin_unlock_irq(&chip->reg_lock);
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static int snd_als300_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ u32 tmp;
+ struct snd_als300_substream_data *data;
+ unsigned short reg;
+ int ret = 0;
+
+ data = substream->runtime->private_data;
+ reg = data->control_register;
+
+ snd_als300_dbgcallenter();
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ tmp = snd_als300_gcr_read(chip->port, reg);
+ data->period_flipflop = 1;
+ snd_als300_gcr_write(chip->port, reg, tmp | TRANSFER_START);
+ snd_als300_dbgplay("TRIGGER START\n");
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ tmp = snd_als300_gcr_read(chip->port, reg);
+ snd_als300_gcr_write(chip->port, reg, tmp & ~TRANSFER_START);
+ snd_als300_dbgplay("TRIGGER STOP\n");
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ tmp = snd_als300_gcr_read(chip->port, reg);
+ snd_als300_gcr_write(chip->port, reg, tmp | FIFO_PAUSE);
+ snd_als300_dbgplay("TRIGGER PAUSE\n");
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ tmp = snd_als300_gcr_read(chip->port, reg);
+ snd_als300_gcr_write(chip->port, reg, tmp & ~FIFO_PAUSE);
+ snd_als300_dbgplay("TRIGGER RELEASE\n");
+ break;
+ default:
+ snd_als300_dbgplay("TRIGGER INVALID\n");
+ ret = -EINVAL;
+ }
+ spin_unlock(&chip->reg_lock);
+ snd_als300_dbgcallleave();
+ return ret;
+}
+
+static snd_pcm_uframes_t snd_als300_pointer(struct snd_pcm_substream *substream)
+{
+ u16 current_ptr;
+ struct snd_als300 *chip = snd_pcm_substream_chip(substream);
+ struct snd_als300_substream_data *data;
+ unsigned short period_bytes;
+
+ data = substream->runtime->private_data;
+ period_bytes = snd_pcm_lib_period_bytes(substream);
+
+ snd_als300_dbgcallenter();
+ spin_lock(&chip->reg_lock);
+ current_ptr = (u16) snd_als300_gcr_read(chip->port,
+ data->block_counter_register) + 4;
+ spin_unlock(&chip->reg_lock);
+ if (current_ptr > period_bytes)
+ current_ptr = 0;
+ else
+ current_ptr = period_bytes - current_ptr;
+
+ if (data->period_flipflop == 0)
+ current_ptr += period_bytes;
+ snd_als300_dbgplay("Pointer (bytes): %d\n", current_ptr);
+ snd_als300_dbgcallleave();
+ return bytes_to_frames(substream->runtime, current_ptr);
+}
+
+static struct snd_pcm_ops snd_als300_playback_ops = {
+ .open = snd_als300_playback_open,
+ .close = snd_als300_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_als300_pcm_hw_params,
+ .hw_free = snd_als300_pcm_hw_free,
+ .prepare = snd_als300_playback_prepare,
+ .trigger = snd_als300_trigger,
+ .pointer = snd_als300_pointer,
+};
+
+static struct snd_pcm_ops snd_als300_capture_ops = {
+ .open = snd_als300_capture_open,
+ .close = snd_als300_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_als300_pcm_hw_params,
+ .hw_free = snd_als300_pcm_hw_free,
+ .prepare = snd_als300_capture_prepare,
+ .trigger = snd_als300_trigger,
+ .pointer = snd_als300_pointer,
+};
+
+static int __devinit snd_als300_new_pcm(struct snd_als300 *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ snd_als300_dbgcallenter();
+ err = snd_pcm_new(chip->card, "ALS300", 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ pcm->private_data = chip;
+ strcpy(pcm->name, "ALS300");
+ chip->pcm = pcm;
+
+ /* set operators */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_als300_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_als300_capture_ops);
+
+ /* pre-allocation of buffers */
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 64*1024);
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+static void snd_als300_init(struct snd_als300 *chip)
+{
+ unsigned long flags;
+ u32 tmp;
+
+ snd_als300_dbgcallenter();
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ chip->revision = (snd_als300_gcr_read(chip->port, MISC_CONTROL) >> 16)
+ & 0x0000000F;
+ /* Setup DRAM */
+ tmp = snd_als300_gcr_read(chip->port, DRAM_WRITE_CONTROL);
+ snd_als300_gcr_write(chip->port, DRAM_WRITE_CONTROL,
+ (tmp | DRAM_MODE_2)
+ & ~WRITE_TRANS_START);
+
+ /* Enable IRQ output */
+ snd_als300_set_irq_flag(chip, IRQ_ENABLE);
+
+ /* Unmute hardware devices so their outputs get routed to
+ * the onboard mixer */
+ tmp = snd_als300_gcr_read(chip->port, MISC_CONTROL);
+ snd_als300_gcr_write(chip->port, MISC_CONTROL,
+ tmp | VMUTE_NORMAL | MMUTE_NORMAL);
+
+ /* Reset volumes */
+ snd_als300_gcr_write(chip->port, MUS_VOC_VOL, 0);
+
+ /* Make sure playback transfer is stopped */
+ tmp = snd_als300_gcr_read(chip->port, PLAYBACK_CONTROL);
+ snd_als300_gcr_write(chip->port, PLAYBACK_CONTROL,
+ tmp & ~TRANSFER_START);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_als300_dbgcallleave();
+}
+
+static int __devinit snd_als300_create(struct snd_card *card,
+ struct pci_dev *pci, int chip_type,
+ struct snd_als300 **rchip)
+{
+ struct snd_als300 *chip;
+ void *irq_handler;
+ int err;
+
+ static struct snd_device_ops ops = {
+ .dev_free = snd_als300_dev_free,
+ };
+ *rchip = NULL;
+
+ snd_als300_dbgcallenter();
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) {
+ printk(KERN_ERR "error setting 28bit DMA mask\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+ pci_set_master(pci);
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->chip_type = chip_type;
+ spin_lock_init(&chip->reg_lock);
+
+ if ((err = pci_request_regions(pci, "ALS300")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->port = pci_resource_start(pci, 0);
+
+ if (chip->chip_type == DEVICE_ALS300_PLUS)
+ irq_handler = snd_als300plus_interrupt;
+ else
+ irq_handler = snd_als300_interrupt;
+
+ if (request_irq(pci->irq, irq_handler, IRQF_SHARED,
+ card->shortname, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_als300_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+
+ snd_als300_init(chip);
+
+ err = snd_als300_ac97(chip);
+ if (err < 0) {
+ snd_printk(KERN_WARNING "Could not create ac97\n");
+ snd_als300_free(chip);
+ return err;
+ }
+
+ if ((err = snd_als300_new_pcm(chip)) < 0) {
+ snd_printk(KERN_WARNING "Could not create PCM\n");
+ snd_als300_free(chip);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &ops)) < 0) {
+ snd_als300_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ snd_als300_dbgcallleave();
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_als300 *chip = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_ac97_suspend(chip->ac97);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_als300_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_als300 *chip = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "als300: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_als300_init(chip);
+ snd_ac97_resume(chip->ac97);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static int __devinit snd_als300_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_als300 *chip;
+ int err, chip_type;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+
+ if (card == NULL)
+ return -ENOMEM;
+
+ chip_type = pci_id->driver_data;
+
+ if ((err = snd_als300_create(card, pci, chip_type, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ strcpy(card->driver, "ALS300");
+ if (chip->chip_type == DEVICE_ALS300_PLUS)
+ /* don't know much about ALS300+ yet
+ * print revision number for now */
+ sprintf(card->shortname, "ALS300+ (Rev. %d)", chip->revision);
+ else
+ sprintf(card->shortname, "ALS300 (Rev. %c)", 'A' +
+ chip->revision - 1);
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->port, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static struct pci_driver driver = {
+ .name = "ALS300",
+ .id_table = snd_als300_ids,
+ .probe = snd_als300_probe,
+ .remove = __devexit_p(snd_als300_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_als300_suspend,
+ .resume = snd_als300_resume,
+#endif
+};
+
+static int __init alsa_card_als300_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_als300_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_als300_init)
+module_exit(alsa_card_als300_exit)
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
new file mode 100644
index 0000000..ba57005
--- /dev/null
+++ b/sound/pci/als4000.c
@@ -0,0 +1,1060 @@
+/*
+ * card-als4000.c - driver for Avance Logic ALS4000 based soundcards.
+ * Copyright (C) 2000 by Bart Hartgers <bart@etpmod.phys.tue.nl>,
+ * Jaroslav Kysela <perex@perex.cz>
+ * Copyright (C) 2002, 2008 by Andreas Mohr <hw7oshyuv3001@sneakemail.com>
+ *
+ * Framework borrowed from Massimo Piccioni's card-als100.c.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * NOTES
+ *
+ * Since Avance does not provide any meaningful documentation, and I
+ * bought an ALS4000 based soundcard, I was forced to base this driver
+ * on reverse engineering.
+ *
+ * Note: this is no longer true (thank you!):
+ * pretty verbose chip docu (ALS4000a.PDF) can be found on the ALSA web site.
+ * Page numbers stated anywhere below with the "SPECS_PAGE:" tag
+ * refer to: ALS4000a.PDF specs Ver 1.0, May 28th, 1998.
+ *
+ * The ALS4000 seems to be the PCI-cousin of the ALS100. It contains an
+ * ALS100-like SB DSP/mixer, an OPL3 synth, a MPU401 and a gameport
+ * interface. These subsystems can be mapped into ISA io-port space,
+ * using the PCI-interface. In addition, the PCI-bit provides DMA and IRQ
+ * services to the subsystems.
+ *
+ * While ALS4000 is very similar to a SoundBlaster, the differences in
+ * DMA and capturing require more changes to the SoundBlaster than
+ * desirable, so I made this separate driver.
+ *
+ * The ALS4000 can do real full duplex playback/capture.
+ *
+ * FMDAC:
+ * - 0x4f -> port 0x14
+ * - port 0x15 |= 1
+ *
+ * Enable/disable 3D sound:
+ * - 0x50 -> port 0x14
+ * - change bit 6 (0x40) of port 0x15
+ *
+ * Set QSound:
+ * - 0xdb -> port 0x14
+ * - set port 0x15:
+ * 0x3e (mode 3), 0x3c (mode 2), 0x3a (mode 1), 0x38 (mode 0)
+ *
+ * Set KSound:
+ * - value -> some port 0x0c0d
+ *
+ * ToDo:
+ * - by default, don't enable legacy game and use PCI game I/O
+ * - power management? (card can do voice wakeup according to datasheet!!)
+ */
+
+#include <asm/io.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/sb.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Bart Hartgers <bart@etpmod.phys.tue.nl>, Andreas Mohr");
+MODULE_DESCRIPTION("Avance Logic ALS4000");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS4000}}");
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK 1
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+#ifdef SUPPORT_JOYSTICK
+static int joystick_port[SNDRV_CARDS];
+#endif
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for ALS4000 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for ALS4000 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable ALS4000 soundcard.");
+#ifdef SUPPORT_JOYSTICK
+module_param_array(joystick_port, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_port, "Joystick port address for ALS4000 soundcard. (0 = disabled)");
+#endif
+
+struct snd_card_als4000 {
+ /* most frequent access first */
+ unsigned long iobase;
+ struct pci_dev *pci;
+ struct snd_sb *chip;
+#ifdef SUPPORT_JOYSTICK
+ struct gameport *gameport;
+#endif
+};
+
+static struct pci_device_id snd_als4000_ids[] = {
+ { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_als4000_ids);
+
+enum als4k_iobase_t {
+ /* IOx: B == Byte, W = Word, D = DWord; SPECS_PAGE: 37 */
+ ALS4K_IOD_00_AC97_ACCESS = 0x00,
+ ALS4K_IOW_04_AC97_READ = 0x04,
+ ALS4K_IOB_06_AC97_STATUS = 0x06,
+ ALS4K_IOB_07_IRQSTATUS = 0x07,
+ ALS4K_IOD_08_GCR_DATA = 0x08,
+ ALS4K_IOB_0C_GCR_INDEX = 0x0c,
+ ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU = 0x0e,
+ ALS4K_IOB_10_ADLIB_ADDR0 = 0x10,
+ ALS4K_IOB_11_ADLIB_ADDR1 = 0x11,
+ ALS4K_IOB_12_ADLIB_ADDR2 = 0x12,
+ ALS4K_IOB_13_ADLIB_ADDR3 = 0x13,
+ ALS4K_IOB_14_MIXER_INDEX = 0x14,
+ ALS4K_IOB_15_MIXER_DATA = 0x15,
+ ALS4K_IOB_16_ESP_RESET = 0x16,
+ ALS4K_IOB_16_ACK_FOR_CR1E = 0x16, /* 2nd function */
+ ALS4K_IOB_18_OPL_ADDR0 = 0x18,
+ ALS4K_IOB_19_OPL_ADDR1 = 0x19,
+ ALS4K_IOB_1A_ESP_RD_DATA = 0x1a,
+ ALS4K_IOB_1C_ESP_CMD_DATA = 0x1c,
+ ALS4K_IOB_1C_ESP_WR_STATUS = 0x1c, /* 2nd function */
+ ALS4K_IOB_1E_ESP_RD_STATUS8 = 0x1e,
+ ALS4K_IOB_1F_ESP_RD_STATUS16 = 0x1f,
+ ALS4K_IOB_20_ESP_GAMEPORT_200 = 0x20,
+ ALS4K_IOB_21_ESP_GAMEPORT_201 = 0x21,
+ ALS4K_IOB_30_MIDI_DATA = 0x30,
+ ALS4K_IOB_31_MIDI_STATUS = 0x31,
+ ALS4K_IOB_31_MIDI_COMMAND = 0x31, /* 2nd function */
+};
+
+enum als4k_iobase_0e_t {
+ ALS4K_IOB_0E_MPU_IRQ = 0x10,
+ ALS4K_IOB_0E_CR1E_IRQ = 0x40,
+ ALS4K_IOB_0E_SB_DMA_IRQ = 0x80,
+};
+
+enum als4k_gcr_t { /* all registers 32bit wide; SPECS_PAGE: 38 to 42 */
+ ALS4K_GCR8C_MISC_CTRL = 0x8c,
+ ALS4K_GCR90_TEST_MODE_REG = 0x90,
+ ALS4K_GCR91_DMA0_ADDR = 0x91,
+ ALS4K_GCR92_DMA0_MODE_COUNT = 0x92,
+ ALS4K_GCR93_DMA1_ADDR = 0x93,
+ ALS4K_GCR94_DMA1_MODE_COUNT = 0x94,
+ ALS4K_GCR95_DMA3_ADDR = 0x95,
+ ALS4K_GCR96_DMA3_MODE_COUNT = 0x96,
+ ALS4K_GCR99_DMA_EMULATION_CTRL = 0x99,
+ ALS4K_GCRA0_FIFO1_CURRENT_ADDR = 0xa0,
+ ALS4K_GCRA1_FIFO1_STATUS_BYTECOUNT = 0xa1,
+ ALS4K_GCRA2_FIFO2_PCIADDR = 0xa2,
+ ALS4K_GCRA3_FIFO2_COUNT = 0xa3,
+ ALS4K_GCRA4_FIFO2_CURRENT_ADDR = 0xa4,
+ ALS4K_GCRA5_FIFO1_STATUS_BYTECOUNT = 0xa5,
+ ALS4K_GCRA6_PM_CTRL = 0xa6,
+ ALS4K_GCRA7_PCI_ACCESS_STORAGE = 0xa7,
+ ALS4K_GCRA8_LEGACY_CFG1 = 0xa8,
+ ALS4K_GCRA9_LEGACY_CFG2 = 0xa9,
+ ALS4K_GCRFF_DUMMY_SCRATCH = 0xff,
+};
+
+enum als4k_gcr8c_t {
+ ALS4K_GCR8C_IRQ_MASK_CTRL_ENABLE = 0x8000,
+ ALS4K_GCR8C_CHIP_REV_MASK = 0xf0000
+};
+
+static inline void snd_als4k_iobase_writeb(unsigned long iobase,
+ enum als4k_iobase_t reg,
+ u8 val)
+{
+ outb(val, iobase + reg);
+}
+
+static inline void snd_als4k_iobase_writel(unsigned long iobase,
+ enum als4k_iobase_t reg,
+ u32 val)
+{
+ outl(val, iobase + reg);
+}
+
+static inline u8 snd_als4k_iobase_readb(unsigned long iobase,
+ enum als4k_iobase_t reg)
+{
+ return inb(iobase + reg);
+}
+
+static inline u32 snd_als4k_iobase_readl(unsigned long iobase,
+ enum als4k_iobase_t reg)
+{
+ return inl(iobase + reg);
+}
+
+static inline void snd_als4k_gcr_write_addr(unsigned long iobase,
+ enum als4k_gcr_t reg,
+ u32 val)
+{
+ snd_als4k_iobase_writeb(iobase, ALS4K_IOB_0C_GCR_INDEX, reg);
+ snd_als4k_iobase_writel(iobase, ALS4K_IOD_08_GCR_DATA, val);
+}
+
+static inline void snd_als4k_gcr_write(struct snd_sb *sb,
+ enum als4k_gcr_t reg,
+ u32 val)
+{
+ snd_als4k_gcr_write_addr(sb->alt_port, reg, val);
+}
+
+static inline u32 snd_als4k_gcr_read_addr(unsigned long iobase,
+ enum als4k_gcr_t reg)
+{
+ /* SPECS_PAGE: 37/38 */
+ snd_als4k_iobase_writeb(iobase, ALS4K_IOB_0C_GCR_INDEX, reg);
+ return snd_als4k_iobase_readl(iobase, ALS4K_IOD_08_GCR_DATA);
+}
+
+static inline u32 snd_als4k_gcr_read(struct snd_sb *sb, enum als4k_gcr_t reg)
+{
+ return snd_als4k_gcr_read_addr(sb->alt_port, reg);
+}
+
+enum als4k_cr_t { /* all registers 8bit wide; SPECS_PAGE: 20 to 23 */
+ ALS4K_CR0_SB_CONFIG = 0x00,
+ ALS4K_CR2_MISC_CONTROL = 0x02,
+ ALS4K_CR3_CONFIGURATION = 0x03,
+ ALS4K_CR17_FIFO_STATUS = 0x17,
+ ALS4K_CR18_ESP_MAJOR_VERSION = 0x18,
+ ALS4K_CR19_ESP_MINOR_VERSION = 0x19,
+ ALS4K_CR1A_MPU401_UART_MODE_CONTROL = 0x1a,
+ ALS4K_CR1C_FIFO2_BLOCK_LENGTH_LO = 0x1c,
+ ALS4K_CR1D_FIFO2_BLOCK_LENGTH_HI = 0x1d,
+ ALS4K_CR1E_FIFO2_CONTROL = 0x1e, /* secondary PCM FIFO (recording) */
+ ALS4K_CR3A_MISC_CONTROL = 0x3a,
+ ALS4K_CR3B_CRC32_BYTE0 = 0x3b, /* for testing, activate via CR3A */
+ ALS4K_CR3C_CRC32_BYTE1 = 0x3c,
+ ALS4K_CR3D_CRC32_BYTE2 = 0x3d,
+ ALS4K_CR3E_CRC32_BYTE3 = 0x3e,
+};
+
+enum als4k_cr0_t {
+ ALS4K_CR0_DMA_CONTIN_MODE_CTRL = 0x02, /* IRQ/FIFO controlled for 0/1 */
+ ALS4K_CR0_DMA_90H_MODE_CTRL = 0x04, /* IRQ/FIFO controlled for 0/1 */
+ ALS4K_CR0_MX80_81_REG_WRITE_ENABLE = 0x80,
+};
+
+static inline void snd_als4_cr_write(struct snd_sb *chip,
+ enum als4k_cr_t reg,
+ u8 data)
+{
+ /* Control Register is reg | 0xc0 (bit 7, 6 set) on sbmixer_index
+ * NOTE: assumes chip->mixer_lock to be locked externally already!
+ * SPECS_PAGE: 6 */
+ snd_sbmixer_write(chip, reg | 0xc0, data);
+}
+
+static inline u8 snd_als4_cr_read(struct snd_sb *chip,
+ enum als4k_cr_t reg)
+{
+ /* NOTE: assumes chip->mixer_lock to be locked externally already! */
+ return snd_sbmixer_read(chip, reg | 0xc0);
+}
+
+
+
+static void snd_als4000_set_rate(struct snd_sb *chip, unsigned int rate)
+{
+ if (!(chip->mode & SB_RATE_LOCK)) {
+ snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE_OUT);
+ snd_sbdsp_command(chip, rate>>8);
+ snd_sbdsp_command(chip, rate);
+ }
+}
+
+static inline void snd_als4000_set_capture_dma(struct snd_sb *chip,
+ dma_addr_t addr, unsigned size)
+{
+ /* SPECS_PAGE: 40 */
+ snd_als4k_gcr_write(chip, ALS4K_GCRA2_FIFO2_PCIADDR, addr);
+ snd_als4k_gcr_write(chip, ALS4K_GCRA3_FIFO2_COUNT, (size-1));
+}
+
+static inline void snd_als4000_set_playback_dma(struct snd_sb *chip,
+ dma_addr_t addr,
+ unsigned size)
+{
+ /* SPECS_PAGE: 38 */
+ snd_als4k_gcr_write(chip, ALS4K_GCR91_DMA0_ADDR, addr);
+ snd_als4k_gcr_write(chip, ALS4K_GCR92_DMA0_MODE_COUNT,
+ (size-1)|0x180000);
+}
+
+#define ALS4000_FORMAT_SIGNED (1<<0)
+#define ALS4000_FORMAT_16BIT (1<<1)
+#define ALS4000_FORMAT_STEREO (1<<2)
+
+static int snd_als4000_get_format(struct snd_pcm_runtime *runtime)
+{
+ int result;
+
+ result = 0;
+ if (snd_pcm_format_signed(runtime->format))
+ result |= ALS4000_FORMAT_SIGNED;
+ if (snd_pcm_format_physical_width(runtime->format) == 16)
+ result |= ALS4000_FORMAT_16BIT;
+ if (runtime->channels > 1)
+ result |= ALS4000_FORMAT_STEREO;
+ return result;
+}
+
+/* structure for setting up playback */
+static const struct {
+ unsigned char dsp_cmd, dma_on, dma_off, format;
+} playback_cmd_vals[]={
+/* ALS4000_FORMAT_U8_MONO */
+{ SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_UNS_MONO },
+/* ALS4000_FORMAT_S8_MONO */
+{ SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_SIGN_MONO },
+/* ALS4000_FORMAT_U16L_MONO */
+{ SB_DSP4_OUT16_AI, SB_DSP_DMA16_ON, SB_DSP_DMA16_OFF, SB_DSP4_MODE_UNS_MONO },
+/* ALS4000_FORMAT_S16L_MONO */
+{ SB_DSP4_OUT16_AI, SB_DSP_DMA16_ON, SB_DSP_DMA16_OFF, SB_DSP4_MODE_SIGN_MONO },
+/* ALS4000_FORMAT_U8_STEREO */
+{ SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_UNS_STEREO },
+/* ALS4000_FORMAT_S8_STEREO */
+{ SB_DSP4_OUT8_AI, SB_DSP_DMA8_ON, SB_DSP_DMA8_OFF, SB_DSP4_MODE_SIGN_STEREO },
+/* ALS4000_FORMAT_U16L_STEREO */
+{ SB_DSP4_OUT16_AI, SB_DSP_DMA16_ON, SB_DSP_DMA16_OFF, SB_DSP4_MODE_UNS_STEREO },
+/* ALS4000_FORMAT_S16L_STEREO */
+{ SB_DSP4_OUT16_AI, SB_DSP_DMA16_ON, SB_DSP_DMA16_OFF, SB_DSP4_MODE_SIGN_STEREO },
+};
+#define playback_cmd(chip) (playback_cmd_vals[(chip)->playback_format])
+
+/* structure for setting up capture */
+enum { CMD_WIDTH8=0x04, CMD_SIGNED=0x10, CMD_MONO=0x80, CMD_STEREO=0xA0 };
+static const unsigned char capture_cmd_vals[]=
+{
+CMD_WIDTH8|CMD_MONO, /* ALS4000_FORMAT_U8_MONO */
+CMD_WIDTH8|CMD_SIGNED|CMD_MONO, /* ALS4000_FORMAT_S8_MONO */
+CMD_MONO, /* ALS4000_FORMAT_U16L_MONO */
+CMD_SIGNED|CMD_MONO, /* ALS4000_FORMAT_S16L_MONO */
+CMD_WIDTH8|CMD_STEREO, /* ALS4000_FORMAT_U8_STEREO */
+CMD_WIDTH8|CMD_SIGNED|CMD_STEREO, /* ALS4000_FORMAT_S8_STEREO */
+CMD_STEREO, /* ALS4000_FORMAT_U16L_STEREO */
+CMD_SIGNED|CMD_STEREO, /* ALS4000_FORMAT_S16L_STEREO */
+};
+#define capture_cmd(chip) (capture_cmd_vals[(chip)->capture_format])
+
+static int snd_als4000_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_als4000_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_als4000_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long size;
+ unsigned count;
+
+ chip->capture_format = snd_als4000_get_format(runtime);
+
+ size = snd_pcm_lib_buffer_bytes(substream);
+ count = snd_pcm_lib_period_bytes(substream);
+
+ if (chip->capture_format & ALS4000_FORMAT_16BIT)
+ count >>= 1;
+ count--;
+
+ spin_lock_irq(&chip->reg_lock);
+ snd_als4000_set_rate(chip, runtime->rate);
+ snd_als4000_set_capture_dma(chip, runtime->dma_addr, size);
+ spin_unlock_irq(&chip->reg_lock);
+ spin_lock_irq(&chip->mixer_lock);
+ snd_als4_cr_write(chip, ALS4K_CR1C_FIFO2_BLOCK_LENGTH_LO, count & 0xff);
+ snd_als4_cr_write(chip, ALS4K_CR1D_FIFO2_BLOCK_LENGTH_HI, count >> 8);
+ spin_unlock_irq(&chip->mixer_lock);
+ return 0;
+}
+
+static int snd_als4000_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long size;
+ unsigned count;
+
+ chip->playback_format = snd_als4000_get_format(runtime);
+
+ size = snd_pcm_lib_buffer_bytes(substream);
+ count = snd_pcm_lib_period_bytes(substream);
+
+ if (chip->playback_format & ALS4000_FORMAT_16BIT)
+ count >>= 1;
+ count--;
+
+ /* FIXME: from second playback on, there's a lot more clicks and pops
+ * involved here than on first playback. Fiddling with
+ * tons of different settings didn't help (DMA, speaker on/off,
+ * reordering, ...). Something seems to get enabled on playback
+ * that I haven't found out how to disable again, which then causes
+ * the switching pops to reach the speakers the next time here. */
+ spin_lock_irq(&chip->reg_lock);
+ snd_als4000_set_rate(chip, runtime->rate);
+ snd_als4000_set_playback_dma(chip, runtime->dma_addr, size);
+
+ /* SPEAKER_ON not needed, since dma_on seems to also enable speaker */
+ /* snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); */
+ snd_sbdsp_command(chip, playback_cmd(chip).dsp_cmd);
+ snd_sbdsp_command(chip, playback_cmd(chip).format);
+ snd_sbdsp_command(chip, count & 0xff);
+ snd_sbdsp_command(chip, count >> 8);
+ snd_sbdsp_command(chip, playback_cmd(chip).dma_off);
+ spin_unlock_irq(&chip->reg_lock);
+
+ return 0;
+}
+
+static int snd_als4000_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ int result = 0;
+
+ /* FIXME race condition in here!!!
+ chip->mode non-atomic update gets consistently protected
+ by reg_lock always, _except_ for this place!!
+ Probably need to take reg_lock as outer (or inner??) lock, too.
+ (or serialize both lock operations? probably not, though... - racy?)
+ */
+ spin_lock(&chip->mixer_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ chip->mode |= SB_RATE_LOCK_CAPTURE;
+ snd_als4_cr_write(chip, ALS4K_CR1E_FIFO2_CONTROL,
+ capture_cmd(chip));
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ chip->mode &= ~SB_RATE_LOCK_CAPTURE;
+ snd_als4_cr_write(chip, ALS4K_CR1E_FIFO2_CONTROL,
+ capture_cmd(chip));
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ spin_unlock(&chip->mixer_lock);
+ return result;
+}
+
+static int snd_als4000_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ int result = 0;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ chip->mode |= SB_RATE_LOCK_PLAYBACK;
+ snd_sbdsp_command(chip, playback_cmd(chip).dma_on);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ snd_sbdsp_command(chip, playback_cmd(chip).dma_off);
+ chip->mode &= ~SB_RATE_LOCK_PLAYBACK;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ spin_unlock(&chip->reg_lock);
+ return result;
+}
+
+static snd_pcm_uframes_t snd_als4000_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ unsigned int result;
+
+ spin_lock(&chip->reg_lock);
+ result = snd_als4k_gcr_read(chip, ALS4K_GCRA4_FIFO2_CURRENT_ADDR);
+ spin_unlock(&chip->reg_lock);
+ result &= 0xffff;
+ return bytes_to_frames( substream->runtime, result );
+}
+
+static snd_pcm_uframes_t snd_als4000_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ unsigned result;
+
+ spin_lock(&chip->reg_lock);
+ result = snd_als4k_gcr_read(chip, ALS4K_GCRA0_FIFO1_CURRENT_ADDR);
+ spin_unlock(&chip->reg_lock);
+ result &= 0xffff;
+ return bytes_to_frames( substream->runtime, result );
+}
+
+/* FIXME: this IRQ routine doesn't really support IRQ sharing (we always
+ * return IRQ_HANDLED no matter whether we actually had an IRQ flag or not).
+ * ALS4000a.PDF writes that while ACKing IRQ in PCI block will *not* ACK
+ * the IRQ in the SB core, ACKing IRQ in SB block *will* ACK the PCI IRQ
+ * register (alt_port + ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU). Probably something
+ * could be optimized here to query/write one register only...
+ * And even if both registers need to be queried, then there's still the
+ * question of whether it's actually correct to ACK PCI IRQ before reading
+ * SB IRQ like we do now, since ALS4000a.PDF mentions that PCI IRQ will *clear*
+ * SB IRQ status.
+ * (hmm, SPECS_PAGE: 38 mentions it the other way around!)
+ * And do we *really* need the lock here for *reading* SB_DSP4_IRQSTATUS??
+ * */
+static irqreturn_t snd_als4000_interrupt(int irq, void *dev_id)
+{
+ struct snd_sb *chip = dev_id;
+ unsigned pci_irqstatus;
+ unsigned sb_irqstatus;
+
+ /* find out which bit of the ALS4000 PCI block produced the interrupt,
+ SPECS_PAGE: 38, 5 */
+ pci_irqstatus = snd_als4k_iobase_readb(chip->alt_port,
+ ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU);
+ if ((pci_irqstatus & ALS4K_IOB_0E_SB_DMA_IRQ)
+ && (chip->playback_substream)) /* playback */
+ snd_pcm_period_elapsed(chip->playback_substream);
+ if ((pci_irqstatus & ALS4K_IOB_0E_CR1E_IRQ)
+ && (chip->capture_substream)) /* capturing */
+ snd_pcm_period_elapsed(chip->capture_substream);
+ if ((pci_irqstatus & ALS4K_IOB_0E_MPU_IRQ)
+ && (chip->rmidi)) /* MPU401 interrupt */
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+ /* ACK the PCI block IRQ */
+ snd_als4k_iobase_writeb(chip->alt_port,
+ ALS4K_IOB_0E_IRQTYPE_SB_CR1E_MPU, pci_irqstatus);
+
+ spin_lock(&chip->mixer_lock);
+ /* SPECS_PAGE: 20 */
+ sb_irqstatus = snd_sbmixer_read(chip, SB_DSP4_IRQSTATUS);
+ spin_unlock(&chip->mixer_lock);
+
+ if (sb_irqstatus & SB_IRQTYPE_8BIT)
+ snd_sb_ack_8bit(chip);
+ if (sb_irqstatus & SB_IRQTYPE_16BIT)
+ snd_sb_ack_16bit(chip);
+ if (sb_irqstatus & SB_IRQTYPE_MPUIN)
+ inb(chip->mpu_port);
+ if (sb_irqstatus & ALS4K_IRQTYPE_CR1E_DMA)
+ snd_als4k_iobase_readb(chip->alt_port,
+ ALS4K_IOB_16_ACK_FOR_CR1E);
+
+ /* printk(KERN_INFO "als4000: irq 0x%04x 0x%04x\n",
+ pci_irqstatus, sb_irqstatus); */
+
+ /* only ack the things we actually handled above */
+ return IRQ_RETVAL(
+ (pci_irqstatus & (ALS4K_IOB_0E_SB_DMA_IRQ|ALS4K_IOB_0E_CR1E_IRQ|
+ ALS4K_IOB_0E_MPU_IRQ))
+ || (sb_irqstatus & (SB_IRQTYPE_8BIT|SB_IRQTYPE_16BIT|
+ SB_IRQTYPE_MPUIN|ALS4K_IRQTYPE_CR1E_DMA))
+ );
+}
+
+/*****************************************************************/
+
+static struct snd_pcm_hardware snd_als4000_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE, /* formats */
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0
+};
+
+static struct snd_pcm_hardware snd_als4000_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE, /* formats */
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0
+};
+
+/*****************************************************************/
+
+static int snd_als4000_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ chip->playback_substream = substream;
+ runtime->hw = snd_als4000_playback;
+ return 0;
+}
+
+static int snd_als4000_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+
+ chip->playback_substream = NULL;
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_als4000_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ chip->capture_substream = substream;
+ runtime->hw = snd_als4000_capture;
+ return 0;
+}
+
+static int snd_als4000_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_sb *chip = snd_pcm_substream_chip(substream);
+
+ chip->capture_substream = NULL;
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+/******************************************************************/
+
+static struct snd_pcm_ops snd_als4000_playback_ops = {
+ .open = snd_als4000_playback_open,
+ .close = snd_als4000_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_als4000_hw_params,
+ .hw_free = snd_als4000_hw_free,
+ .prepare = snd_als4000_playback_prepare,
+ .trigger = snd_als4000_playback_trigger,
+ .pointer = snd_als4000_playback_pointer
+};
+
+static struct snd_pcm_ops snd_als4000_capture_ops = {
+ .open = snd_als4000_capture_open,
+ .close = snd_als4000_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_als4000_hw_params,
+ .hw_free = snd_als4000_hw_free,
+ .prepare = snd_als4000_capture_prepare,
+ .trigger = snd_als4000_capture_trigger,
+ .pointer = snd_als4000_capture_pointer
+};
+
+static int __devinit snd_als4000_pcm(struct snd_sb *chip, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(chip->card, "ALS4000 DSP", device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ pcm->private_data = chip;
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_als4000_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_als4000_capture_ops);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ 64*1024, 64*1024);
+
+ chip->pcm = pcm;
+
+ return 0;
+}
+
+/******************************************************************/
+
+static void snd_als4000_set_addr(unsigned long iobase,
+ unsigned int sb_io,
+ unsigned int mpu_io,
+ unsigned int opl_io,
+ unsigned int game_io)
+{
+ u32 cfg1 = 0;
+ u32 cfg2 = 0;
+
+ if (mpu_io > 0)
+ cfg2 |= (mpu_io | 1) << 16;
+ if (sb_io > 0)
+ cfg2 |= (sb_io | 1);
+ if (game_io > 0)
+ cfg1 |= (game_io | 1) << 16;
+ if (opl_io > 0)
+ cfg1 |= (opl_io | 1);
+ snd_als4k_gcr_write_addr(iobase, ALS4K_GCRA8_LEGACY_CFG1, cfg1);
+ snd_als4k_gcr_write_addr(iobase, ALS4K_GCRA9_LEGACY_CFG2, cfg2);
+}
+
+static void snd_als4000_configure(struct snd_sb *chip)
+{
+ u8 tmp;
+ int i;
+
+ /* do some more configuration */
+ spin_lock_irq(&chip->mixer_lock);
+ tmp = snd_als4_cr_read(chip, ALS4K_CR0_SB_CONFIG);
+ snd_als4_cr_write(chip, ALS4K_CR0_SB_CONFIG,
+ tmp|ALS4K_CR0_MX80_81_REG_WRITE_ENABLE);
+ /* always select DMA channel 0, since we do not actually use DMA
+ * SPECS_PAGE: 19/20 */
+ snd_sbmixer_write(chip, SB_DSP4_DMASETUP, SB_DMASETUP_DMA0);
+ snd_als4_cr_write(chip, ALS4K_CR0_SB_CONFIG,
+ tmp & ~ALS4K_CR0_MX80_81_REG_WRITE_ENABLE);
+ spin_unlock_irq(&chip->mixer_lock);
+
+ spin_lock_irq(&chip->reg_lock);
+ /* enable interrupts */
+ snd_als4k_gcr_write(chip, ALS4K_GCR8C_MISC_CTRL,
+ ALS4K_GCR8C_IRQ_MASK_CTRL_ENABLE);
+
+ /* SPECS_PAGE: 39 */
+ for (i = ALS4K_GCR91_DMA0_ADDR; i <= ALS4K_GCR96_DMA3_MODE_COUNT; ++i)
+ snd_als4k_gcr_write(chip, i, 0);
+
+ snd_als4k_gcr_write(chip, ALS4K_GCR99_DMA_EMULATION_CTRL,
+ snd_als4k_gcr_read(chip, ALS4K_GCR99_DMA_EMULATION_CTRL));
+ spin_unlock_irq(&chip->reg_lock);
+}
+
+#ifdef SUPPORT_JOYSTICK
+static int __devinit snd_als4000_create_gameport(struct snd_card_als4000 *acard, int dev)
+{
+ struct gameport *gp;
+ struct resource *r;
+ int io_port;
+
+ if (joystick_port[dev] == 0)
+ return -ENODEV;
+
+ if (joystick_port[dev] == 1) { /* auto-detect */
+ for (io_port = 0x200; io_port <= 0x218; io_port += 8) {
+ r = request_region(io_port, 8, "ALS4000 gameport");
+ if (r)
+ break;
+ }
+ } else {
+ io_port = joystick_port[dev];
+ r = request_region(io_port, 8, "ALS4000 gameport");
+ }
+
+ if (!r) {
+ printk(KERN_WARNING "als4000: cannot reserve joystick ports\n");
+ return -EBUSY;
+ }
+
+ acard->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "als4000: cannot allocate memory for gameport\n");
+ release_and_free_resource(r);
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "ALS4000 Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(acard->pci));
+ gameport_set_dev_parent(gp, &acard->pci->dev);
+ gp->io = io_port;
+ gameport_set_port_data(gp, r);
+
+ /* Enable legacy joystick port */
+ snd_als4000_set_addr(acard->iobase, 0, 0, 0, 1);
+
+ gameport_register_port(acard->gameport);
+
+ return 0;
+}
+
+static void snd_als4000_free_gameport(struct snd_card_als4000 *acard)
+{
+ if (acard->gameport) {
+ struct resource *r = gameport_get_port_data(acard->gameport);
+
+ gameport_unregister_port(acard->gameport);
+ acard->gameport = NULL;
+
+ /* disable joystick */
+ snd_als4000_set_addr(acard->iobase, 0, 0, 0, 0);
+
+ release_and_free_resource(r);
+ }
+}
+#else
+static inline int snd_als4000_create_gameport(struct snd_card_als4000 *acard, int dev) { return -ENOSYS; }
+static inline void snd_als4000_free_gameport(struct snd_card_als4000 *acard) { }
+#endif
+
+static void snd_card_als4000_free( struct snd_card *card )
+{
+ struct snd_card_als4000 *acard = card->private_data;
+
+ /* make sure that interrupts are disabled */
+ snd_als4k_gcr_write_addr(acard->iobase, ALS4K_GCR8C_MISC_CTRL, 0);
+ /* free resources */
+ snd_als4000_free_gameport(acard);
+ pci_release_regions(acard->pci);
+ pci_disable_device(acard->pci);
+}
+
+static int __devinit snd_card_als4000_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_card_als4000 *acard;
+ unsigned long iobase;
+ struct snd_sb *chip;
+ struct snd_opl3 *opl3;
+ unsigned short word;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0) {
+ return err;
+ }
+ /* check, if we can restrict PCI DMA transfers to 24 bits */
+ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ if ((err = pci_request_regions(pci, "ALS4000")) < 0) {
+ pci_disable_device(pci);
+ return err;
+ }
+ iobase = pci_resource_start(pci, 0);
+
+ pci_read_config_word(pci, PCI_COMMAND, &word);
+ pci_write_config_word(pci, PCI_COMMAND, word | PCI_COMMAND_IO);
+ pci_set_master(pci);
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(*acard) /* private_data: acard */);
+ if (card == NULL) {
+ pci_release_regions(pci);
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ acard = card->private_data;
+ acard->pci = pci;
+ acard->iobase = iobase;
+ card->private_free = snd_card_als4000_free;
+
+ /* disable all legacy ISA stuff */
+ snd_als4000_set_addr(acard->iobase, 0, 0, 0, 0);
+
+ if ((err = snd_sbdsp_create(card,
+ iobase + ALS4K_IOB_10_ADLIB_ADDR0,
+ pci->irq,
+ /* internally registered as IRQF_SHARED in case of ALS4000 SB */
+ snd_als4000_interrupt,
+ -1,
+ -1,
+ SB_HW_ALS4000,
+ &chip)) < 0) {
+ goto out_err;
+ }
+ acard->chip = chip;
+
+ chip->pci = pci;
+ chip->alt_port = iobase;
+ snd_card_set_dev(card, &pci->dev);
+
+ snd_als4000_configure(chip);
+
+ strcpy(card->driver, "ALS4000");
+ strcpy(card->shortname, "Avance Logic ALS4000");
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, chip->alt_port, chip->irq);
+
+ if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000,
+ iobase + ALS4K_IOB_30_MIDI_DATA,
+ MPU401_INFO_INTEGRATED,
+ pci->irq, 0, &chip->rmidi)) < 0) {
+ printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n",
+ iobase + ALS4K_IOB_30_MIDI_DATA);
+ goto out_err;
+ }
+ /* FIXME: ALS4000 has interesting MPU401 configuration features
+ * at ALS4K_CR1A_MPU401_UART_MODE_CONTROL
+ * (pass-thru / UART switching, fast MIDI clock, etc.),
+ * however there doesn't seem to be an ALSA API for this...
+ * SPECS_PAGE: 21 */
+
+ if ((err = snd_als4000_pcm(chip, 0)) < 0) {
+ goto out_err;
+ }
+ if ((err = snd_sbmixer_new(chip)) < 0) {
+ goto out_err;
+ }
+
+ if (snd_opl3_create(card,
+ iobase + ALS4K_IOB_10_ADLIB_ADDR0,
+ iobase + ALS4K_IOB_12_ADLIB_ADDR2,
+ OPL3_HW_AUTO, 1, &opl3) < 0) {
+ printk(KERN_ERR "als4000: no OPL device at 0x%lx-0x%lx?\n",
+ iobase + ALS4K_IOB_10_ADLIB_ADDR0,
+ iobase + ALS4K_IOB_12_ADLIB_ADDR2);
+ } else {
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ goto out_err;
+ }
+ }
+
+ snd_als4000_create_gameport(acard, dev);
+
+ if ((err = snd_card_register(card)) < 0) {
+ goto out_err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ err = 0;
+ goto out;
+
+out_err:
+ snd_card_free(card);
+
+out:
+ return err;
+}
+
+static void __devexit snd_card_als4000_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_card_als4000 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ snd_pcm_suspend_all(chip->pcm);
+ snd_sbmixer_suspend(chip);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_als4000_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_card_als4000 *acard = card->private_data;
+ struct snd_sb *chip = acard->chip;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "als4000: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_als4000_configure(chip);
+ snd_sbdsp_reset(chip);
+ snd_sbmixer_resume(chip);
+
+#ifdef SUPPORT_JOYSTICK
+ if (acard->gameport)
+ snd_als4000_set_addr(acard->iobase, 0, 0, 0, 1);
+#endif
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+static struct pci_driver driver = {
+ .name = "ALS4000",
+ .id_table = snd_als4000_ids,
+ .probe = snd_card_als4000_probe,
+ .remove = __devexit_p(snd_card_als4000_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_als4000_suspend,
+ .resume = snd_als4000_resume,
+#endif
+};
+
+static int __init alsa_card_als4000_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_als4000_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_als4000_init)
+module_exit(alsa_card_als4000_exit)
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
new file mode 100644
index 0000000..226fe82
--- /dev/null
+++ b/sound/pci/atiixp.c
@@ -0,0 +1,1719 @@
+/*
+ * ALSA driver for ATI IXP 150/200/250/300 AC97 controllers
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/info.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("ATI IXP AC97 controller");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250/300/400/600}}");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int ac97_clock = 48000;
+static char *ac97_quirk;
+static int spdif_aclink = 1;
+static int ac97_codec = -1;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for ATI IXP controller.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for ATI IXP controller.");
+module_param(ac97_clock, int, 0444);
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz).");
+module_param(ac97_quirk, charp, 0444);
+MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
+module_param(ac97_codec, int, 0444);
+MODULE_PARM_DESC(ac97_codec, "Specify codec instead of probing.");
+module_param(spdif_aclink, bool, 0444);
+MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link.");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
+
+/*
+ */
+
+#define ATI_REG_ISR 0x00 /* interrupt source */
+#define ATI_REG_ISR_IN_XRUN (1U<<0)
+#define ATI_REG_ISR_IN_STATUS (1U<<1)
+#define ATI_REG_ISR_OUT_XRUN (1U<<2)
+#define ATI_REG_ISR_OUT_STATUS (1U<<3)
+#define ATI_REG_ISR_SPDF_XRUN (1U<<4)
+#define ATI_REG_ISR_SPDF_STATUS (1U<<5)
+#define ATI_REG_ISR_PHYS_INTR (1U<<8)
+#define ATI_REG_ISR_PHYS_MISMATCH (1U<<9)
+#define ATI_REG_ISR_CODEC0_NOT_READY (1U<<10)
+#define ATI_REG_ISR_CODEC1_NOT_READY (1U<<11)
+#define ATI_REG_ISR_CODEC2_NOT_READY (1U<<12)
+#define ATI_REG_ISR_NEW_FRAME (1U<<13)
+
+#define ATI_REG_IER 0x04 /* interrupt enable */
+#define ATI_REG_IER_IN_XRUN_EN (1U<<0)
+#define ATI_REG_IER_IO_STATUS_EN (1U<<1)
+#define ATI_REG_IER_OUT_XRUN_EN (1U<<2)
+#define ATI_REG_IER_OUT_XRUN_COND (1U<<3)
+#define ATI_REG_IER_SPDF_XRUN_EN (1U<<4)
+#define ATI_REG_IER_SPDF_STATUS_EN (1U<<5)
+#define ATI_REG_IER_PHYS_INTR_EN (1U<<8)
+#define ATI_REG_IER_PHYS_MISMATCH_EN (1U<<9)
+#define ATI_REG_IER_CODEC0_INTR_EN (1U<<10)
+#define ATI_REG_IER_CODEC1_INTR_EN (1U<<11)
+#define ATI_REG_IER_CODEC2_INTR_EN (1U<<12)
+#define ATI_REG_IER_NEW_FRAME_EN (1U<<13) /* (RO */
+#define ATI_REG_IER_SET_BUS_BUSY (1U<<14) /* (WO) audio is running */
+
+#define ATI_REG_CMD 0x08 /* command */
+#define ATI_REG_CMD_POWERDOWN (1U<<0)
+#define ATI_REG_CMD_RECEIVE_EN (1U<<1)
+#define ATI_REG_CMD_SEND_EN (1U<<2)
+#define ATI_REG_CMD_STATUS_MEM (1U<<3)
+#define ATI_REG_CMD_SPDF_OUT_EN (1U<<4)
+#define ATI_REG_CMD_SPDF_STATUS_MEM (1U<<5)
+#define ATI_REG_CMD_SPDF_THRESHOLD (3U<<6)
+#define ATI_REG_CMD_SPDF_THRESHOLD_SHIFT 6
+#define ATI_REG_CMD_IN_DMA_EN (1U<<8)
+#define ATI_REG_CMD_OUT_DMA_EN (1U<<9)
+#define ATI_REG_CMD_SPDF_DMA_EN (1U<<10)
+#define ATI_REG_CMD_SPDF_OUT_STOPPED (1U<<11)
+#define ATI_REG_CMD_SPDF_CONFIG_MASK (7U<<12)
+#define ATI_REG_CMD_SPDF_CONFIG_34 (1U<<12)
+#define ATI_REG_CMD_SPDF_CONFIG_78 (2U<<12)
+#define ATI_REG_CMD_SPDF_CONFIG_69 (3U<<12)
+#define ATI_REG_CMD_SPDF_CONFIG_01 (4U<<12)
+#define ATI_REG_CMD_INTERLEAVE_SPDF (1U<<16)
+#define ATI_REG_CMD_AUDIO_PRESENT (1U<<20)
+#define ATI_REG_CMD_INTERLEAVE_IN (1U<<21)
+#define ATI_REG_CMD_INTERLEAVE_OUT (1U<<22)
+#define ATI_REG_CMD_LOOPBACK_EN (1U<<23)
+#define ATI_REG_CMD_PACKED_DIS (1U<<24)
+#define ATI_REG_CMD_BURST_EN (1U<<25)
+#define ATI_REG_CMD_PANIC_EN (1U<<26)
+#define ATI_REG_CMD_MODEM_PRESENT (1U<<27)
+#define ATI_REG_CMD_ACLINK_ACTIVE (1U<<28)
+#define ATI_REG_CMD_AC_SOFT_RESET (1U<<29)
+#define ATI_REG_CMD_AC_SYNC (1U<<30)
+#define ATI_REG_CMD_AC_RESET (1U<<31)
+
+#define ATI_REG_PHYS_OUT_ADDR 0x0c
+#define ATI_REG_PHYS_OUT_CODEC_MASK (3U<<0)
+#define ATI_REG_PHYS_OUT_RW (1U<<2)
+#define ATI_REG_PHYS_OUT_ADDR_EN (1U<<8)
+#define ATI_REG_PHYS_OUT_ADDR_SHIFT 9
+#define ATI_REG_PHYS_OUT_DATA_SHIFT 16
+
+#define ATI_REG_PHYS_IN_ADDR 0x10
+#define ATI_REG_PHYS_IN_READ_FLAG (1U<<8)
+#define ATI_REG_PHYS_IN_ADDR_SHIFT 9
+#define ATI_REG_PHYS_IN_DATA_SHIFT 16
+
+#define ATI_REG_SLOTREQ 0x14
+
+#define ATI_REG_COUNTER 0x18
+#define ATI_REG_COUNTER_SLOT (3U<<0) /* slot # */
+#define ATI_REG_COUNTER_BITCLOCK (31U<<8)
+
+#define ATI_REG_IN_FIFO_THRESHOLD 0x1c
+
+#define ATI_REG_IN_DMA_LINKPTR 0x20
+#define ATI_REG_IN_DMA_DT_START 0x24 /* RO */
+#define ATI_REG_IN_DMA_DT_NEXT 0x28 /* RO */
+#define ATI_REG_IN_DMA_DT_CUR 0x2c /* RO */
+#define ATI_REG_IN_DMA_DT_SIZE 0x30
+
+#define ATI_REG_OUT_DMA_SLOT 0x34
+#define ATI_REG_OUT_DMA_SLOT_BIT(x) (1U << ((x) - 3))
+#define ATI_REG_OUT_DMA_SLOT_MASK 0x1ff
+#define ATI_REG_OUT_DMA_THRESHOLD_MASK 0xf800
+#define ATI_REG_OUT_DMA_THRESHOLD_SHIFT 11
+
+#define ATI_REG_OUT_DMA_LINKPTR 0x38
+#define ATI_REG_OUT_DMA_DT_START 0x3c /* RO */
+#define ATI_REG_OUT_DMA_DT_NEXT 0x40 /* RO */
+#define ATI_REG_OUT_DMA_DT_CUR 0x44 /* RO */
+#define ATI_REG_OUT_DMA_DT_SIZE 0x48
+
+#define ATI_REG_SPDF_CMD 0x4c
+#define ATI_REG_SPDF_CMD_LFSR (1U<<4)
+#define ATI_REG_SPDF_CMD_SINGLE_CH (1U<<5)
+#define ATI_REG_SPDF_CMD_LFSR_ACC (0xff<<8) /* RO */
+
+#define ATI_REG_SPDF_DMA_LINKPTR 0x50
+#define ATI_REG_SPDF_DMA_DT_START 0x54 /* RO */
+#define ATI_REG_SPDF_DMA_DT_NEXT 0x58 /* RO */
+#define ATI_REG_SPDF_DMA_DT_CUR 0x5c /* RO */
+#define ATI_REG_SPDF_DMA_DT_SIZE 0x60
+
+#define ATI_REG_MODEM_MIRROR 0x7c
+#define ATI_REG_AUDIO_MIRROR 0x80
+
+#define ATI_REG_6CH_REORDER 0x84 /* reorder slots for 6ch */
+#define ATI_REG_6CH_REORDER_EN (1U<<0) /* 3,4,7,8,6,9 -> 3,4,6,9,7,8 */
+
+#define ATI_REG_FIFO_FLUSH 0x88
+#define ATI_REG_FIFO_OUT_FLUSH (1U<<0)
+#define ATI_REG_FIFO_IN_FLUSH (1U<<1)
+
+/* LINKPTR */
+#define ATI_REG_LINKPTR_EN (1U<<0)
+
+/* [INT|OUT|SPDIF]_DMA_DT_SIZE */
+#define ATI_REG_DMA_DT_SIZE (0xffffU<<0)
+#define ATI_REG_DMA_FIFO_USED (0x1fU<<16)
+#define ATI_REG_DMA_FIFO_FREE (0x1fU<<21)
+#define ATI_REG_DMA_STATE (7U<<26)
+
+
+#define ATI_MAX_DESCRIPTORS 256 /* max number of descriptor packets */
+
+
+struct atiixp;
+
+/*
+ * DMA packate descriptor
+ */
+
+struct atiixp_dma_desc {
+ u32 addr; /* DMA buffer address */
+ u16 status; /* status bits */
+ u16 size; /* size of the packet in dwords */
+ u32 next; /* address of the next packet descriptor */
+};
+
+/*
+ * stream enum
+ */
+enum { ATI_DMA_PLAYBACK, ATI_DMA_CAPTURE, ATI_DMA_SPDIF, NUM_ATI_DMAS }; /* DMAs */
+enum { ATI_PCM_OUT, ATI_PCM_IN, ATI_PCM_SPDIF, NUM_ATI_PCMS }; /* AC97 pcm slots */
+enum { ATI_PCMDEV_ANALOG, ATI_PCMDEV_DIGITAL, NUM_ATI_PCMDEVS }; /* pcm devices */
+
+#define NUM_ATI_CODECS 3
+
+
+/*
+ * constants and callbacks for each DMA type
+ */
+struct atiixp_dma_ops {
+ int type; /* ATI_DMA_XXX */
+ unsigned int llp_offset; /* LINKPTR offset */
+ unsigned int dt_cur; /* DT_CUR offset */
+ /* called from open callback */
+ void (*enable_dma)(struct atiixp *chip, int on);
+ /* called from trigger (START/STOP) */
+ void (*enable_transfer)(struct atiixp *chip, int on);
+ /* called from trigger (STOP only) */
+ void (*flush_dma)(struct atiixp *chip);
+};
+
+/*
+ * DMA stream
+ */
+struct atiixp_dma {
+ const struct atiixp_dma_ops *ops;
+ struct snd_dma_buffer desc_buf;
+ struct snd_pcm_substream *substream; /* assigned PCM substream */
+ unsigned int buf_addr, buf_bytes; /* DMA buffer address, bytes */
+ unsigned int period_bytes, periods;
+ int opened;
+ int running;
+ int suspended;
+ int pcm_open_flag;
+ int ac97_pcm_type; /* index # of ac97_pcm to access, -1 = not used */
+ unsigned int saved_curptr;
+};
+
+/*
+ * ATI IXP chip
+ */
+struct atiixp {
+ struct snd_card *card;
+ struct pci_dev *pci;
+
+ unsigned long addr;
+ void __iomem *remap_addr;
+ int irq;
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97[NUM_ATI_CODECS];
+
+ spinlock_t reg_lock;
+
+ struct atiixp_dma dmas[NUM_ATI_DMAS];
+ struct ac97_pcm *pcms[NUM_ATI_PCMS];
+ struct snd_pcm *pcmdevs[NUM_ATI_PCMDEVS];
+
+ int max_channels; /* max. channels for PCM out */
+
+ unsigned int codec_not_ready_bits; /* for codec detection */
+
+ int spdif_over_aclink; /* passed from the module option */
+ struct mutex open_mutex; /* playback open mutex */
+};
+
+
+/*
+ */
+static struct pci_device_id snd_atiixp_ids[] = {
+ { 0x1002, 0x4341, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */
+ { 0x1002, 0x4361, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB300 */
+ { 0x1002, 0x4370, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */
+ { 0x1002, 0x4382, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB600 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
+
+static struct snd_pci_quirk atiixp_quirks[] __devinitdata = {
+ SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0),
+ { } /* terminator */
+};
+
+/*
+ * lowlevel functions
+ */
+
+/*
+ * update the bits of the given register.
+ * return 1 if the bits changed.
+ */
+static int snd_atiixp_update_bits(struct atiixp *chip, unsigned int reg,
+ unsigned int mask, unsigned int value)
+{
+ void __iomem *addr = chip->remap_addr + reg;
+ unsigned int data, old_data;
+ old_data = data = readl(addr);
+ data &= ~mask;
+ data |= value;
+ if (old_data == data)
+ return 0;
+ writel(data, addr);
+ return 1;
+}
+
+/*
+ * macros for easy use
+ */
+#define atiixp_write(chip,reg,value) \
+ writel(value, chip->remap_addr + ATI_REG_##reg)
+#define atiixp_read(chip,reg) \
+ readl(chip->remap_addr + ATI_REG_##reg)
+#define atiixp_update(chip,reg,mask,val) \
+ snd_atiixp_update_bits(chip, ATI_REG_##reg, mask, val)
+
+/*
+ * handling DMA packets
+ *
+ * we allocate a linear buffer for the DMA, and split it to each packet.
+ * in a future version, a scatter-gather buffer should be implemented.
+ */
+
+#define ATI_DESC_LIST_SIZE \
+ PAGE_ALIGN(ATI_MAX_DESCRIPTORS * sizeof(struct atiixp_dma_desc))
+
+/*
+ * build packets ring for the given buffer size.
+ *
+ * IXP handles the buffer descriptors, which are connected as a linked
+ * list. although we can change the list dynamically, in this version,
+ * a static RING of buffer descriptors is used.
+ *
+ * the ring is built in this function, and is set up to the hardware.
+ */
+static int atiixp_build_dma_packets(struct atiixp *chip, struct atiixp_dma *dma,
+ struct snd_pcm_substream *substream,
+ unsigned int periods,
+ unsigned int period_bytes)
+{
+ unsigned int i;
+ u32 addr, desc_addr;
+ unsigned long flags;
+
+ if (periods > ATI_MAX_DESCRIPTORS)
+ return -ENOMEM;
+
+ if (dma->desc_buf.area == NULL) {
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ ATI_DESC_LIST_SIZE,
+ &dma->desc_buf) < 0)
+ return -ENOMEM;
+ dma->period_bytes = dma->periods = 0; /* clear */
+ }
+
+ if (dma->periods == periods && dma->period_bytes == period_bytes)
+ return 0;
+
+ /* reset DMA before changing the descriptor table */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ writel(0, chip->remap_addr + dma->ops->llp_offset);
+ dma->ops->enable_dma(chip, 0);
+ dma->ops->enable_dma(chip, 1);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ /* fill the entries */
+ addr = (u32)substream->runtime->dma_addr;
+ desc_addr = (u32)dma->desc_buf.addr;
+ for (i = 0; i < periods; i++) {
+ struct atiixp_dma_desc *desc;
+ desc = &((struct atiixp_dma_desc *)dma->desc_buf.area)[i];
+ desc->addr = cpu_to_le32(addr);
+ desc->status = 0;
+ desc->size = period_bytes >> 2; /* in dwords */
+ desc_addr += sizeof(struct atiixp_dma_desc);
+ if (i == periods - 1)
+ desc->next = cpu_to_le32((u32)dma->desc_buf.addr);
+ else
+ desc->next = cpu_to_le32(desc_addr);
+ addr += period_bytes;
+ }
+
+ writel((u32)dma->desc_buf.addr | ATI_REG_LINKPTR_EN,
+ chip->remap_addr + dma->ops->llp_offset);
+
+ dma->period_bytes = period_bytes;
+ dma->periods = periods;
+
+ return 0;
+}
+
+/*
+ * remove the ring buffer and release it if assigned
+ */
+static void atiixp_clear_dma_packets(struct atiixp *chip, struct atiixp_dma *dma,
+ struct snd_pcm_substream *substream)
+{
+ if (dma->desc_buf.area) {
+ writel(0, chip->remap_addr + dma->ops->llp_offset);
+ snd_dma_free_pages(&dma->desc_buf);
+ dma->desc_buf.area = NULL;
+ }
+}
+
+/*
+ * AC97 interface
+ */
+static int snd_atiixp_acquire_codec(struct atiixp *chip)
+{
+ int timeout = 1000;
+
+ while (atiixp_read(chip, PHYS_OUT_ADDR) & ATI_REG_PHYS_OUT_ADDR_EN) {
+ if (! timeout--) {
+ snd_printk(KERN_WARNING "atiixp: codec acquire timeout\n");
+ return -EBUSY;
+ }
+ udelay(1);
+ }
+ return 0;
+}
+
+static unsigned short snd_atiixp_codec_read(struct atiixp *chip, unsigned short codec, unsigned short reg)
+{
+ unsigned int data;
+ int timeout;
+
+ if (snd_atiixp_acquire_codec(chip) < 0)
+ return 0xffff;
+ data = (reg << ATI_REG_PHYS_OUT_ADDR_SHIFT) |
+ ATI_REG_PHYS_OUT_ADDR_EN |
+ ATI_REG_PHYS_OUT_RW |
+ codec;
+ atiixp_write(chip, PHYS_OUT_ADDR, data);
+ if (snd_atiixp_acquire_codec(chip) < 0)
+ return 0xffff;
+ timeout = 1000;
+ do {
+ data = atiixp_read(chip, PHYS_IN_ADDR);
+ if (data & ATI_REG_PHYS_IN_READ_FLAG)
+ return data >> ATI_REG_PHYS_IN_DATA_SHIFT;
+ udelay(1);
+ } while (--timeout);
+ /* time out may happen during reset */
+ if (reg < 0x7c)
+ snd_printk(KERN_WARNING "atiixp: codec read timeout (reg %x)\n", reg);
+ return 0xffff;
+}
+
+
+static void snd_atiixp_codec_write(struct atiixp *chip, unsigned short codec,
+ unsigned short reg, unsigned short val)
+{
+ unsigned int data;
+
+ if (snd_atiixp_acquire_codec(chip) < 0)
+ return;
+ data = ((unsigned int)val << ATI_REG_PHYS_OUT_DATA_SHIFT) |
+ ((unsigned int)reg << ATI_REG_PHYS_OUT_ADDR_SHIFT) |
+ ATI_REG_PHYS_OUT_ADDR_EN | codec;
+ atiixp_write(chip, PHYS_OUT_ADDR, data);
+}
+
+
+static unsigned short snd_atiixp_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct atiixp *chip = ac97->private_data;
+ return snd_atiixp_codec_read(chip, ac97->num, reg);
+
+}
+
+static void snd_atiixp_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct atiixp *chip = ac97->private_data;
+ snd_atiixp_codec_write(chip, ac97->num, reg, val);
+}
+
+/*
+ * reset AC link
+ */
+static int snd_atiixp_aclink_reset(struct atiixp *chip)
+{
+ int timeout;
+
+ /* reset powerdoewn */
+ if (atiixp_update(chip, CMD, ATI_REG_CMD_POWERDOWN, 0))
+ udelay(10);
+
+ /* perform a software reset */
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SOFT_RESET, ATI_REG_CMD_AC_SOFT_RESET);
+ atiixp_read(chip, CMD);
+ udelay(10);
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SOFT_RESET, 0);
+
+ timeout = 10;
+ while (! (atiixp_read(chip, CMD) & ATI_REG_CMD_ACLINK_ACTIVE)) {
+ /* do a hard reset */
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET,
+ ATI_REG_CMD_AC_SYNC);
+ atiixp_read(chip, CMD);
+ mdelay(1);
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_RESET, ATI_REG_CMD_AC_RESET);
+ if (--timeout) {
+ snd_printk(KERN_ERR "atiixp: codec reset timeout\n");
+ break;
+ }
+ }
+
+ /* deassert RESET and assert SYNC to make sure */
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET,
+ ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_atiixp_aclink_down(struct atiixp *chip)
+{
+ // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */
+ // return -EBUSY;
+ atiixp_update(chip, CMD,
+ ATI_REG_CMD_POWERDOWN | ATI_REG_CMD_AC_RESET,
+ ATI_REG_CMD_POWERDOWN);
+ return 0;
+}
+#endif
+
+/*
+ * auto-detection of codecs
+ *
+ * the IXP chip can generate interrupts for the non-existing codecs.
+ * NEW_FRAME interrupt is used to make sure that the interrupt is generated
+ * even if all three codecs are connected.
+ */
+
+#define ALL_CODEC_NOT_READY \
+ (ATI_REG_ISR_CODEC0_NOT_READY |\
+ ATI_REG_ISR_CODEC1_NOT_READY |\
+ ATI_REG_ISR_CODEC2_NOT_READY)
+#define CODEC_CHECK_BITS (ALL_CODEC_NOT_READY|ATI_REG_ISR_NEW_FRAME)
+
+static int __devinit ac97_probing_bugs(struct pci_dev *pci)
+{
+ const struct snd_pci_quirk *q;
+
+ q = snd_pci_quirk_lookup(pci, atiixp_quirks);
+ if (q) {
+ snd_printdd(KERN_INFO "Atiixp quirk for %s. "
+ "Forcing codec %d\n", q->name, q->value);
+ return q->value;
+ }
+ /* this hardware doesn't need workarounds. Probe for codec */
+ return -1;
+}
+
+static int __devinit snd_atiixp_codec_detect(struct atiixp *chip)
+{
+ int timeout;
+
+ chip->codec_not_ready_bits = 0;
+ if (ac97_codec == -1)
+ ac97_codec = ac97_probing_bugs(chip->pci);
+ if (ac97_codec >= 0) {
+ chip->codec_not_ready_bits |=
+ CODEC_CHECK_BITS ^ (1 << (ac97_codec + 10));
+ return 0;
+ }
+
+ atiixp_write(chip, IER, CODEC_CHECK_BITS);
+ /* wait for the interrupts */
+ timeout = 50;
+ while (timeout-- > 0) {
+ mdelay(1);
+ if (chip->codec_not_ready_bits)
+ break;
+ }
+ atiixp_write(chip, IER, 0); /* disable irqs */
+
+ if ((chip->codec_not_ready_bits & ALL_CODEC_NOT_READY) == ALL_CODEC_NOT_READY) {
+ snd_printk(KERN_ERR "atiixp: no codec detected!\n");
+ return -ENXIO;
+ }
+ return 0;
+}
+
+
+/*
+ * enable DMA and irqs
+ */
+static int snd_atiixp_chip_start(struct atiixp *chip)
+{
+ unsigned int reg;
+
+ /* set up spdif, enable burst mode */
+ reg = atiixp_read(chip, CMD);
+ reg |= 0x02 << ATI_REG_CMD_SPDF_THRESHOLD_SHIFT;
+ reg |= ATI_REG_CMD_BURST_EN;
+ atiixp_write(chip, CMD, reg);
+
+ reg = atiixp_read(chip, SPDF_CMD);
+ reg &= ~(ATI_REG_SPDF_CMD_LFSR|ATI_REG_SPDF_CMD_SINGLE_CH);
+ atiixp_write(chip, SPDF_CMD, reg);
+
+ /* clear all interrupt source */
+ atiixp_write(chip, ISR, 0xffffffff);
+ /* enable irqs */
+ atiixp_write(chip, IER,
+ ATI_REG_IER_IO_STATUS_EN |
+ ATI_REG_IER_IN_XRUN_EN |
+ ATI_REG_IER_OUT_XRUN_EN |
+ ATI_REG_IER_SPDF_XRUN_EN |
+ ATI_REG_IER_SPDF_STATUS_EN);
+ return 0;
+}
+
+
+/*
+ * disable DMA and IRQs
+ */
+static int snd_atiixp_chip_stop(struct atiixp *chip)
+{
+ /* clear interrupt source */
+ atiixp_write(chip, ISR, atiixp_read(chip, ISR));
+ /* disable irqs */
+ atiixp_write(chip, IER, 0);
+ return 0;
+}
+
+
+/*
+ * PCM section
+ */
+
+/*
+ * pointer callback simplly reads XXX_DMA_DT_CUR register as the current
+ * position. when SG-buffer is implemented, the offset must be calculated
+ * correctly...
+ */
+static snd_pcm_uframes_t snd_atiixp_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct atiixp_dma *dma = runtime->private_data;
+ unsigned int curptr;
+ int timeout = 1000;
+
+ while (timeout--) {
+ curptr = readl(chip->remap_addr + dma->ops->dt_cur);
+ if (curptr < dma->buf_addr)
+ continue;
+ curptr -= dma->buf_addr;
+ if (curptr >= dma->buf_bytes)
+ continue;
+ return bytes_to_frames(runtime, curptr);
+ }
+ snd_printd("atiixp: invalid DMA pointer read 0x%x (buf=%x)\n",
+ readl(chip->remap_addr + dma->ops->dt_cur), dma->buf_addr);
+ return 0;
+}
+
+/*
+ * XRUN detected, and stop the PCM substream
+ */
+static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma)
+{
+ if (! dma->substream || ! dma->running)
+ return;
+ snd_printdd("atiixp: XRUN detected (DMA %d)\n", dma->ops->type);
+ snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN);
+}
+
+/*
+ * the period ack. update the substream.
+ */
+static void snd_atiixp_update_dma(struct atiixp *chip, struct atiixp_dma *dma)
+{
+ if (! dma->substream || ! dma->running)
+ return;
+ snd_pcm_period_elapsed(dma->substream);
+}
+
+/* set BUS_BUSY interrupt bit if any DMA is running */
+/* call with spinlock held */
+static void snd_atiixp_check_bus_busy(struct atiixp *chip)
+{
+ unsigned int bus_busy;
+ if (atiixp_read(chip, CMD) & (ATI_REG_CMD_SEND_EN |
+ ATI_REG_CMD_RECEIVE_EN |
+ ATI_REG_CMD_SPDF_OUT_EN))
+ bus_busy = ATI_REG_IER_SET_BUS_BUSY;
+ else
+ bus_busy = 0;
+ atiixp_update(chip, IER, ATI_REG_IER_SET_BUS_BUSY, bus_busy);
+}
+
+/* common trigger callback
+ * calling the lowlevel callbacks in it
+ */
+static int snd_atiixp_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ struct atiixp_dma *dma = substream->runtime->private_data;
+ int err = 0;
+
+ if (snd_BUG_ON(!dma->ops->enable_transfer ||
+ !dma->ops->flush_dma))
+ return -EINVAL;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ dma->ops->enable_transfer(chip, 1);
+ dma->running = 1;
+ dma->suspended = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ dma->ops->enable_transfer(chip, 0);
+ dma->running = 0;
+ dma->suspended = cmd == SNDRV_PCM_TRIGGER_SUSPEND;
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ if (! err) {
+ snd_atiixp_check_bus_busy(chip);
+ if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ dma->ops->flush_dma(chip);
+ snd_atiixp_check_bus_busy(chip);
+ }
+ }
+ spin_unlock(&chip->reg_lock);
+ return err;
+}
+
+
+/*
+ * lowlevel callbacks for each DMA type
+ *
+ * every callback is supposed to be called in chip->reg_lock spinlock
+ */
+
+/* flush FIFO of analog OUT DMA */
+static void atiixp_out_flush_dma(struct atiixp *chip)
+{
+ atiixp_write(chip, FIFO_FLUSH, ATI_REG_FIFO_OUT_FLUSH);
+}
+
+/* enable/disable analog OUT DMA */
+static void atiixp_out_enable_dma(struct atiixp *chip, int on)
+{
+ unsigned int data;
+ data = atiixp_read(chip, CMD);
+ if (on) {
+ if (data & ATI_REG_CMD_OUT_DMA_EN)
+ return;
+ atiixp_out_flush_dma(chip);
+ data |= ATI_REG_CMD_OUT_DMA_EN;
+ } else
+ data &= ~ATI_REG_CMD_OUT_DMA_EN;
+ atiixp_write(chip, CMD, data);
+}
+
+/* start/stop transfer over OUT DMA */
+static void atiixp_out_enable_transfer(struct atiixp *chip, int on)
+{
+ atiixp_update(chip, CMD, ATI_REG_CMD_SEND_EN,
+ on ? ATI_REG_CMD_SEND_EN : 0);
+}
+
+/* enable/disable analog IN DMA */
+static void atiixp_in_enable_dma(struct atiixp *chip, int on)
+{
+ atiixp_update(chip, CMD, ATI_REG_CMD_IN_DMA_EN,
+ on ? ATI_REG_CMD_IN_DMA_EN : 0);
+}
+
+/* start/stop analog IN DMA */
+static void atiixp_in_enable_transfer(struct atiixp *chip, int on)
+{
+ if (on) {
+ unsigned int data = atiixp_read(chip, CMD);
+ if (! (data & ATI_REG_CMD_RECEIVE_EN)) {
+ data |= ATI_REG_CMD_RECEIVE_EN;
+#if 0 /* FIXME: this causes the endless loop */
+ /* wait until slot 3/4 are finished */
+ while ((atiixp_read(chip, COUNTER) &
+ ATI_REG_COUNTER_SLOT) != 5)
+ ;
+#endif
+ atiixp_write(chip, CMD, data);
+ }
+ } else
+ atiixp_update(chip, CMD, ATI_REG_CMD_RECEIVE_EN, 0);
+}
+
+/* flush FIFO of analog IN DMA */
+static void atiixp_in_flush_dma(struct atiixp *chip)
+{
+ atiixp_write(chip, FIFO_FLUSH, ATI_REG_FIFO_IN_FLUSH);
+}
+
+/* enable/disable SPDIF OUT DMA */
+static void atiixp_spdif_enable_dma(struct atiixp *chip, int on)
+{
+ atiixp_update(chip, CMD, ATI_REG_CMD_SPDF_DMA_EN,
+ on ? ATI_REG_CMD_SPDF_DMA_EN : 0);
+}
+
+/* start/stop SPDIF OUT DMA */
+static void atiixp_spdif_enable_transfer(struct atiixp *chip, int on)
+{
+ unsigned int data;
+ data = atiixp_read(chip, CMD);
+ if (on)
+ data |= ATI_REG_CMD_SPDF_OUT_EN;
+ else
+ data &= ~ATI_REG_CMD_SPDF_OUT_EN;
+ atiixp_write(chip, CMD, data);
+}
+
+/* flush FIFO of SPDIF OUT DMA */
+static void atiixp_spdif_flush_dma(struct atiixp *chip)
+{
+ int timeout;
+
+ /* DMA off, transfer on */
+ atiixp_spdif_enable_dma(chip, 0);
+ atiixp_spdif_enable_transfer(chip, 1);
+
+ timeout = 100;
+ do {
+ if (! (atiixp_read(chip, SPDF_DMA_DT_SIZE) & ATI_REG_DMA_FIFO_USED))
+ break;
+ udelay(1);
+ } while (timeout-- > 0);
+
+ atiixp_spdif_enable_transfer(chip, 0);
+}
+
+/* set up slots and formats for SPDIF OUT */
+static int snd_atiixp_spdif_prepare(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ if (chip->spdif_over_aclink) {
+ unsigned int data;
+ /* enable slots 10/11 */
+ atiixp_update(chip, CMD, ATI_REG_CMD_SPDF_CONFIG_MASK,
+ ATI_REG_CMD_SPDF_CONFIG_01);
+ data = atiixp_read(chip, OUT_DMA_SLOT) & ~ATI_REG_OUT_DMA_SLOT_MASK;
+ data |= ATI_REG_OUT_DMA_SLOT_BIT(10) |
+ ATI_REG_OUT_DMA_SLOT_BIT(11);
+ data |= 0x04 << ATI_REG_OUT_DMA_THRESHOLD_SHIFT;
+ atiixp_write(chip, OUT_DMA_SLOT, data);
+ atiixp_update(chip, CMD, ATI_REG_CMD_INTERLEAVE_OUT,
+ substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE ?
+ ATI_REG_CMD_INTERLEAVE_OUT : 0);
+ } else {
+ atiixp_update(chip, CMD, ATI_REG_CMD_SPDF_CONFIG_MASK, 0);
+ atiixp_update(chip, CMD, ATI_REG_CMD_INTERLEAVE_SPDF, 0);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+/* set up slots and formats for analog OUT */
+static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ unsigned int data;
+
+ spin_lock_irq(&chip->reg_lock);
+ data = atiixp_read(chip, OUT_DMA_SLOT) & ~ATI_REG_OUT_DMA_SLOT_MASK;
+ switch (substream->runtime->channels) {
+ case 8:
+ data |= ATI_REG_OUT_DMA_SLOT_BIT(10) |
+ ATI_REG_OUT_DMA_SLOT_BIT(11);
+ /* fallthru */
+ case 6:
+ data |= ATI_REG_OUT_DMA_SLOT_BIT(7) |
+ ATI_REG_OUT_DMA_SLOT_BIT(8);
+ /* fallthru */
+ case 4:
+ data |= ATI_REG_OUT_DMA_SLOT_BIT(6) |
+ ATI_REG_OUT_DMA_SLOT_BIT(9);
+ /* fallthru */
+ default:
+ data |= ATI_REG_OUT_DMA_SLOT_BIT(3) |
+ ATI_REG_OUT_DMA_SLOT_BIT(4);
+ break;
+ }
+
+ /* set output threshold */
+ data |= 0x04 << ATI_REG_OUT_DMA_THRESHOLD_SHIFT;
+ atiixp_write(chip, OUT_DMA_SLOT, data);
+
+ atiixp_update(chip, CMD, ATI_REG_CMD_INTERLEAVE_OUT,
+ substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE ?
+ ATI_REG_CMD_INTERLEAVE_OUT : 0);
+
+ /*
+ * enable 6 channel re-ordering bit if needed
+ */
+ atiixp_update(chip, 6CH_REORDER, ATI_REG_6CH_REORDER_EN,
+ substream->runtime->channels >= 6 ? ATI_REG_6CH_REORDER_EN: 0);
+
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+/* set up slots and formats for analog IN */
+static int snd_atiixp_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ atiixp_update(chip, CMD, ATI_REG_CMD_INTERLEAVE_IN,
+ substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE ?
+ ATI_REG_CMD_INTERLEAVE_IN : 0);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+/*
+ * hw_params - allocate the buffer and set up buffer descriptors
+ */
+static int snd_atiixp_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ struct atiixp_dma *dma = substream->runtime->private_data;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ dma->buf_addr = substream->runtime->dma_addr;
+ dma->buf_bytes = params_buffer_bytes(hw_params);
+
+ err = atiixp_build_dma_packets(chip, dma, substream,
+ params_periods(hw_params),
+ params_period_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (dma->ac97_pcm_type >= 0) {
+ struct ac97_pcm *pcm = chip->pcms[dma->ac97_pcm_type];
+ /* PCM is bound to AC97 codec(s)
+ * set up the AC97 codecs
+ */
+ if (dma->pcm_open_flag) {
+ snd_ac97_pcm_close(pcm);
+ dma->pcm_open_flag = 0;
+ }
+ err = snd_ac97_pcm_open(pcm, params_rate(hw_params),
+ params_channels(hw_params),
+ pcm->r[0].slots);
+ if (err >= 0)
+ dma->pcm_open_flag = 1;
+ }
+
+ return err;
+}
+
+static int snd_atiixp_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ struct atiixp_dma *dma = substream->runtime->private_data;
+
+ if (dma->pcm_open_flag) {
+ struct ac97_pcm *pcm = chip->pcms[dma->ac97_pcm_type];
+ snd_ac97_pcm_close(pcm);
+ dma->pcm_open_flag = 0;
+ }
+ atiixp_clear_dma_packets(chip, dma, substream);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+
+/*
+ * pcm hardware definition, identical for all DMA types
+ */
+static struct snd_pcm_hardware snd_atiixp_pcm_hw =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 256 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 128 * 1024,
+ .periods_min = 2,
+ .periods_max = ATI_MAX_DESCRIPTORS,
+};
+
+static int snd_atiixp_pcm_open(struct snd_pcm_substream *substream,
+ struct atiixp_dma *dma, int pcm_type)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if (snd_BUG_ON(!dma->ops || !dma->ops->enable_dma))
+ return -EINVAL;
+
+ if (dma->opened)
+ return -EBUSY;
+ dma->substream = substream;
+ runtime->hw = snd_atiixp_pcm_hw;
+ dma->ac97_pcm_type = pcm_type;
+ if (pcm_type >= 0) {
+ runtime->hw.rates = chip->pcms[pcm_type]->rates;
+ snd_pcm_limit_hw_rates(runtime);
+ } else {
+ /* direct SPDIF */
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE;
+ }
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ runtime->private_data = dma;
+
+ /* enable DMA bits */
+ spin_lock_irq(&chip->reg_lock);
+ dma->ops->enable_dma(chip, 1);
+ spin_unlock_irq(&chip->reg_lock);
+ dma->opened = 1;
+
+ return 0;
+}
+
+static int snd_atiixp_pcm_close(struct snd_pcm_substream *substream,
+ struct atiixp_dma *dma)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ /* disable DMA bits */
+ if (snd_BUG_ON(!dma->ops || !dma->ops->enable_dma))
+ return -EINVAL;
+ spin_lock_irq(&chip->reg_lock);
+ dma->ops->enable_dma(chip, 0);
+ spin_unlock_irq(&chip->reg_lock);
+ dma->substream = NULL;
+ dma->opened = 0;
+ return 0;
+}
+
+/*
+ */
+static int snd_atiixp_playback_open(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ mutex_lock(&chip->open_mutex);
+ err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_PLAYBACK], 0);
+ mutex_unlock(&chip->open_mutex);
+ if (err < 0)
+ return err;
+ substream->runtime->hw.channels_max = chip->max_channels;
+ if (chip->max_channels > 2)
+ /* channels must be even */
+ snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 2);
+ return 0;
+}
+
+static int snd_atiixp_playback_close(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ int err;
+ mutex_lock(&chip->open_mutex);
+ err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_PLAYBACK]);
+ mutex_unlock(&chip->open_mutex);
+ return err;
+}
+
+static int snd_atiixp_capture_open(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ return snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_CAPTURE], 1);
+}
+
+static int snd_atiixp_capture_close(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ return snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_CAPTURE]);
+}
+
+static int snd_atiixp_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ int err;
+ mutex_lock(&chip->open_mutex);
+ if (chip->spdif_over_aclink) /* share DMA_PLAYBACK */
+ err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_PLAYBACK], 2);
+ else
+ err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_SPDIF], -1);
+ mutex_unlock(&chip->open_mutex);
+ return err;
+}
+
+static int snd_atiixp_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct atiixp *chip = snd_pcm_substream_chip(substream);
+ int err;
+ mutex_lock(&chip->open_mutex);
+ if (chip->spdif_over_aclink)
+ err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_PLAYBACK]);
+ else
+ err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_SPDIF]);
+ mutex_unlock(&chip->open_mutex);
+ return err;
+}
+
+/* AC97 playback */
+static struct snd_pcm_ops snd_atiixp_playback_ops = {
+ .open = snd_atiixp_playback_open,
+ .close = snd_atiixp_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_atiixp_pcm_hw_params,
+ .hw_free = snd_atiixp_pcm_hw_free,
+ .prepare = snd_atiixp_playback_prepare,
+ .trigger = snd_atiixp_pcm_trigger,
+ .pointer = snd_atiixp_pcm_pointer,
+};
+
+/* AC97 capture */
+static struct snd_pcm_ops snd_atiixp_capture_ops = {
+ .open = snd_atiixp_capture_open,
+ .close = snd_atiixp_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_atiixp_pcm_hw_params,
+ .hw_free = snd_atiixp_pcm_hw_free,
+ .prepare = snd_atiixp_capture_prepare,
+ .trigger = snd_atiixp_pcm_trigger,
+ .pointer = snd_atiixp_pcm_pointer,
+};
+
+/* SPDIF playback */
+static struct snd_pcm_ops snd_atiixp_spdif_ops = {
+ .open = snd_atiixp_spdif_open,
+ .close = snd_atiixp_spdif_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_atiixp_pcm_hw_params,
+ .hw_free = snd_atiixp_pcm_hw_free,
+ .prepare = snd_atiixp_spdif_prepare,
+ .trigger = snd_atiixp_pcm_trigger,
+ .pointer = snd_atiixp_pcm_pointer,
+};
+
+static struct ac97_pcm atiixp_pcm_defs[] __devinitdata = {
+ /* front PCM */
+ {
+ .exclusive = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT) |
+ (1 << AC97_SLOT_PCM_CENTER) |
+ (1 << AC97_SLOT_PCM_SLEFT) |
+ (1 << AC97_SLOT_PCM_SRIGHT) |
+ (1 << AC97_SLOT_LFE)
+ }
+ }
+ },
+ /* PCM IN #1 */
+ {
+ .stream = 1,
+ .exclusive = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT)
+ }
+ }
+ },
+ /* S/PDIF OUT (optional) */
+ {
+ .exclusive = 1,
+ .spdif = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_SPDIF_LEFT2) |
+ (1 << AC97_SLOT_SPDIF_RIGHT2)
+ }
+ }
+ },
+};
+
+static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = {
+ .type = ATI_DMA_PLAYBACK,
+ .llp_offset = ATI_REG_OUT_DMA_LINKPTR,
+ .dt_cur = ATI_REG_OUT_DMA_DT_CUR,
+ .enable_dma = atiixp_out_enable_dma,
+ .enable_transfer = atiixp_out_enable_transfer,
+ .flush_dma = atiixp_out_flush_dma,
+};
+
+static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = {
+ .type = ATI_DMA_CAPTURE,
+ .llp_offset = ATI_REG_IN_DMA_LINKPTR,
+ .dt_cur = ATI_REG_IN_DMA_DT_CUR,
+ .enable_dma = atiixp_in_enable_dma,
+ .enable_transfer = atiixp_in_enable_transfer,
+ .flush_dma = atiixp_in_flush_dma,
+};
+
+static struct atiixp_dma_ops snd_atiixp_spdif_dma_ops = {
+ .type = ATI_DMA_SPDIF,
+ .llp_offset = ATI_REG_SPDF_DMA_LINKPTR,
+ .dt_cur = ATI_REG_SPDF_DMA_DT_CUR,
+ .enable_dma = atiixp_spdif_enable_dma,
+ .enable_transfer = atiixp_spdif_enable_transfer,
+ .flush_dma = atiixp_spdif_flush_dma,
+};
+
+
+static int __devinit snd_atiixp_pcm_new(struct atiixp *chip)
+{
+ struct snd_pcm *pcm;
+ struct snd_ac97_bus *pbus = chip->ac97_bus;
+ int err, i, num_pcms;
+
+ /* initialize constants */
+ chip->dmas[ATI_DMA_PLAYBACK].ops = &snd_atiixp_playback_dma_ops;
+ chip->dmas[ATI_DMA_CAPTURE].ops = &snd_atiixp_capture_dma_ops;
+ if (! chip->spdif_over_aclink)
+ chip->dmas[ATI_DMA_SPDIF].ops = &snd_atiixp_spdif_dma_ops;
+
+ /* assign AC97 pcm */
+ if (chip->spdif_over_aclink)
+ num_pcms = 3;
+ else
+ num_pcms = 2;
+ err = snd_ac97_pcm_assign(pbus, num_pcms, atiixp_pcm_defs);
+ if (err < 0)
+ return err;
+ for (i = 0; i < num_pcms; i++)
+ chip->pcms[i] = &pbus->pcms[i];
+
+ chip->max_channels = 2;
+ if (pbus->pcms[ATI_PCM_OUT].r[0].slots & (1 << AC97_SLOT_PCM_SLEFT)) {
+ if (pbus->pcms[ATI_PCM_OUT].r[0].slots & (1 << AC97_SLOT_LFE))
+ chip->max_channels = 6;
+ else
+ chip->max_channels = 4;
+ }
+
+ /* PCM #0: analog I/O */
+ err = snd_pcm_new(chip->card, "ATI IXP AC97",
+ ATI_PCMDEV_ANALOG, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_atiixp_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_atiixp_capture_ops);
+ pcm->private_data = chip;
+ strcpy(pcm->name, "ATI IXP AC97");
+ chip->pcmdevs[ATI_PCMDEV_ANALOG] = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 128*1024);
+
+ /* no SPDIF support on codec? */
+ if (chip->pcms[ATI_PCM_SPDIF] && ! chip->pcms[ATI_PCM_SPDIF]->rates)
+ return 0;
+
+ /* FIXME: non-48k sample rate doesn't work on my test machine with AD1888 */
+ if (chip->pcms[ATI_PCM_SPDIF])
+ chip->pcms[ATI_PCM_SPDIF]->rates = SNDRV_PCM_RATE_48000;
+
+ /* PCM #1: spdif playback */
+ err = snd_pcm_new(chip->card, "ATI IXP IEC958",
+ ATI_PCMDEV_DIGITAL, 1, 0, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_atiixp_spdif_ops);
+ pcm->private_data = chip;
+ if (chip->spdif_over_aclink)
+ strcpy(pcm->name, "ATI IXP IEC958 (AC97)");
+ else
+ strcpy(pcm->name, "ATI IXP IEC958 (Direct)");
+ chip->pcmdevs[ATI_PCMDEV_DIGITAL] = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 128*1024);
+
+ /* pre-select AC97 SPDIF slots 10/11 */
+ for (i = 0; i < NUM_ATI_CODECS; i++) {
+ if (chip->ac97[i])
+ snd_ac97_update_bits(chip->ac97[i],
+ AC97_EXTENDED_STATUS,
+ 0x03 << 4, 0x03 << 4);
+ }
+
+ return 0;
+}
+
+
+
+/*
+ * interrupt handler
+ */
+static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id)
+{
+ struct atiixp *chip = dev_id;
+ unsigned int status;
+
+ status = atiixp_read(chip, ISR);
+
+ if (! status)
+ return IRQ_NONE;
+
+ /* process audio DMA */
+ if (status & ATI_REG_ISR_OUT_XRUN)
+ snd_atiixp_xrun_dma(chip, &chip->dmas[ATI_DMA_PLAYBACK]);
+ else if (status & ATI_REG_ISR_OUT_STATUS)
+ snd_atiixp_update_dma(chip, &chip->dmas[ATI_DMA_PLAYBACK]);
+ if (status & ATI_REG_ISR_IN_XRUN)
+ snd_atiixp_xrun_dma(chip, &chip->dmas[ATI_DMA_CAPTURE]);
+ else if (status & ATI_REG_ISR_IN_STATUS)
+ snd_atiixp_update_dma(chip, &chip->dmas[ATI_DMA_CAPTURE]);
+ if (! chip->spdif_over_aclink) {
+ if (status & ATI_REG_ISR_SPDF_XRUN)
+ snd_atiixp_xrun_dma(chip, &chip->dmas[ATI_DMA_SPDIF]);
+ else if (status & ATI_REG_ISR_SPDF_STATUS)
+ snd_atiixp_update_dma(chip, &chip->dmas[ATI_DMA_SPDIF]);
+ }
+
+ /* for codec detection */
+ if (status & CODEC_CHECK_BITS) {
+ unsigned int detected;
+ detected = status & CODEC_CHECK_BITS;
+ spin_lock(&chip->reg_lock);
+ chip->codec_not_ready_bits |= detected;
+ atiixp_update(chip, IER, detected, 0); /* disable the detected irqs */
+ spin_unlock(&chip->reg_lock);
+ }
+
+ /* ack */
+ atiixp_write(chip, ISR, status);
+
+ return IRQ_HANDLED;
+}
+
+
+/*
+ * ac97 mixer section
+ */
+
+static struct ac97_quirk ac97_quirks[] __devinitdata = {
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x006b,
+ .name = "HP Pavilion ZV5030US",
+ .type = AC97_TUNE_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x308b,
+ .name = "HP nx6125",
+ .type = AC97_TUNE_MUTE_LED
+ },
+ { } /* terminator */
+};
+
+static int __devinit snd_atiixp_mixer_new(struct atiixp *chip, int clock,
+ const char *quirk_override)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int i, err;
+ int codec_count;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_atiixp_ac97_write,
+ .read = snd_atiixp_ac97_read,
+ };
+ static unsigned int codec_skip[NUM_ATI_CODECS] = {
+ ATI_REG_ISR_CODEC0_NOT_READY,
+ ATI_REG_ISR_CODEC1_NOT_READY,
+ ATI_REG_ISR_CODEC2_NOT_READY,
+ };
+
+ if (snd_atiixp_codec_detect(chip) < 0)
+ return -ENXIO;
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &pbus)) < 0)
+ return err;
+ pbus->clock = clock;
+ chip->ac97_bus = pbus;
+
+ codec_count = 0;
+ for (i = 0; i < NUM_ATI_CODECS; i++) {
+ if (chip->codec_not_ready_bits & codec_skip[i])
+ continue;
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.pci = chip->pci;
+ ac97.num = i;
+ ac97.scaps = AC97_SCAP_SKIP_MODEM | AC97_SCAP_POWER_SAVE;
+ if (! chip->spdif_over_aclink)
+ ac97.scaps |= AC97_SCAP_NO_SPDIF;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97[i])) < 0) {
+ chip->ac97[i] = NULL; /* to be sure */
+ snd_printdd("atiixp: codec %d not available for audio\n", i);
+ continue;
+ }
+ codec_count++;
+ }
+
+ if (! codec_count) {
+ snd_printk(KERN_ERR "atiixp: no codec available\n");
+ return -ENODEV;
+ }
+
+ snd_ac97_tune_hardware(chip->ac97[0], ac97_quirks, quirk_override);
+
+ return 0;
+}
+
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct atiixp *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < NUM_ATI_PCMDEVS; i++)
+ if (chip->pcmdevs[i]) {
+ struct atiixp_dma *dma = &chip->dmas[i];
+ if (dma->substream && dma->running)
+ dma->saved_curptr = readl(chip->remap_addr +
+ dma->ops->dt_cur);
+ snd_pcm_suspend_all(chip->pcmdevs[i]);
+ }
+ for (i = 0; i < NUM_ATI_CODECS; i++)
+ snd_ac97_suspend(chip->ac97[i]);
+ snd_atiixp_aclink_down(chip);
+ snd_atiixp_chip_stop(chip);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_atiixp_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct atiixp *chip = card->private_data;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "atiixp: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_atiixp_aclink_reset(chip);
+ snd_atiixp_chip_start(chip);
+
+ for (i = 0; i < NUM_ATI_CODECS; i++)
+ snd_ac97_resume(chip->ac97[i]);
+
+ for (i = 0; i < NUM_ATI_PCMDEVS; i++)
+ if (chip->pcmdevs[i]) {
+ struct atiixp_dma *dma = &chip->dmas[i];
+ if (dma->substream && dma->suspended) {
+ dma->ops->enable_dma(chip, 1);
+ dma->substream->ops->prepare(dma->substream);
+ writel((u32)dma->desc_buf.addr | ATI_REG_LINKPTR_EN,
+ chip->remap_addr + dma->ops->llp_offset);
+ writel(dma->saved_curptr, chip->remap_addr +
+ dma->ops->dt_cur);
+ }
+ }
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+#ifdef CONFIG_PROC_FS
+/*
+ * proc interface for register dump
+ */
+
+static void snd_atiixp_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct atiixp *chip = entry->private_data;
+ int i;
+
+ for (i = 0; i < 256; i += 4)
+ snd_iprintf(buffer, "%02x: %08x\n", i, readl(chip->remap_addr + i));
+}
+
+static void __devinit snd_atiixp_proc_init(struct atiixp *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "atiixp", &entry))
+ snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read);
+}
+#else /* !CONFIG_PROC_FS */
+#define snd_atiixp_proc_init(chip)
+#endif
+
+
+/*
+ * destructor
+ */
+
+static int snd_atiixp_free(struct atiixp *chip)
+{
+ if (chip->irq < 0)
+ goto __hw_end;
+ snd_atiixp_chip_stop(chip);
+
+ __hw_end:
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (chip->remap_addr)
+ iounmap(chip->remap_addr);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_atiixp_dev_free(struct snd_device *device)
+{
+ struct atiixp *chip = device->device_data;
+ return snd_atiixp_free(chip);
+}
+
+/*
+ * constructor for chip instance
+ */
+static int __devinit snd_atiixp_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct atiixp **r_chip)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_atiixp_dev_free,
+ };
+ struct atiixp *chip;
+ int err;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&chip->reg_lock);
+ mutex_init(&chip->open_mutex);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ if ((err = pci_request_regions(pci, "ATI IXP AC97")) < 0) {
+ pci_disable_device(pci);
+ kfree(chip);
+ return err;
+ }
+ chip->addr = pci_resource_start(pci, 0);
+ chip->remap_addr = pci_ioremap_bar(pci, 0);
+ if (chip->remap_addr == NULL) {
+ snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
+ snd_atiixp_free(chip);
+ return -EIO;
+ }
+
+ if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
+ card->shortname, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_atiixp_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ pci_set_master(pci);
+ synchronize_irq(chip->irq);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_atiixp_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_chip = chip;
+ return 0;
+}
+
+
+static int __devinit snd_atiixp_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct atiixp *chip;
+ int err;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, spdif_aclink ? "ATIIXP" : "ATIIXP-SPDMA");
+ strcpy(card->shortname, "ATI IXP");
+ if ((err = snd_atiixp_create(card, pci, &chip)) < 0)
+ goto __error;
+ card->private_data = chip;
+
+ if ((err = snd_atiixp_aclink_reset(chip)) < 0)
+ goto __error;
+
+ chip->spdif_over_aclink = spdif_aclink;
+
+ if ((err = snd_atiixp_mixer_new(chip, ac97_clock, ac97_quirk)) < 0)
+ goto __error;
+
+ if ((err = snd_atiixp_pcm_new(chip)) < 0)
+ goto __error;
+
+ snd_atiixp_proc_init(chip);
+
+ snd_atiixp_chip_start(chip);
+
+ snprintf(card->longname, sizeof(card->longname),
+ "%s rev %x with %s at %#lx, irq %i", card->shortname,
+ pci->revision,
+ chip->ac97[0] ? snd_ac97_get_short_name(chip->ac97[0]) : "?",
+ chip->addr, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto __error;
+
+ pci_set_drvdata(pci, card);
+ return 0;
+
+ __error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_atiixp_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "ATI IXP AC97 controller",
+ .id_table = snd_atiixp_ids,
+ .probe = snd_atiixp_probe,
+ .remove = __devexit_p(snd_atiixp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_atiixp_suspend,
+ .resume = snd_atiixp_resume,
+#endif
+};
+
+
+static int __init alsa_card_atiixp_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_atiixp_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_atiixp_init)
+module_exit(alsa_card_atiixp_exit)
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
new file mode 100644
index 0000000..0e6e5cc
--- /dev/null
+++ b/sound/pci/atiixp_modem.c
@@ -0,0 +1,1357 @@
+/*
+ * ALSA driver for ATI IXP 150/200/250 AC97 modem controllers
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/info.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("ATI IXP MC97 controller");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ATI,IXP150/200/250}}");
+
+static int index = -2; /* Exclude the first card */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int ac97_clock = 48000;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for ATI IXP controller.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for ATI IXP controller.");
+module_param(ac97_clock, int, 0444);
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz).");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
+
+/*
+ */
+
+#define ATI_REG_ISR 0x00 /* interrupt source */
+#define ATI_REG_ISR_MODEM_IN_XRUN (1U<<0)
+#define ATI_REG_ISR_MODEM_IN_STATUS (1U<<1)
+#define ATI_REG_ISR_MODEM_OUT1_XRUN (1U<<2)
+#define ATI_REG_ISR_MODEM_OUT1_STATUS (1U<<3)
+#define ATI_REG_ISR_MODEM_OUT2_XRUN (1U<<4)
+#define ATI_REG_ISR_MODEM_OUT2_STATUS (1U<<5)
+#define ATI_REG_ISR_MODEM_OUT3_XRUN (1U<<6)
+#define ATI_REG_ISR_MODEM_OUT3_STATUS (1U<<7)
+#define ATI_REG_ISR_PHYS_INTR (1U<<8)
+#define ATI_REG_ISR_PHYS_MISMATCH (1U<<9)
+#define ATI_REG_ISR_CODEC0_NOT_READY (1U<<10)
+#define ATI_REG_ISR_CODEC1_NOT_READY (1U<<11)
+#define ATI_REG_ISR_CODEC2_NOT_READY (1U<<12)
+#define ATI_REG_ISR_NEW_FRAME (1U<<13)
+#define ATI_REG_ISR_MODEM_GPIO_DATA (1U<<14)
+
+#define ATI_REG_IER 0x04 /* interrupt enable */
+#define ATI_REG_IER_MODEM_IN_XRUN_EN (1U<<0)
+#define ATI_REG_IER_MODEM_STATUS_EN (1U<<1)
+#define ATI_REG_IER_MODEM_OUT1_XRUN_EN (1U<<2)
+#define ATI_REG_IER_MODEM_OUT2_XRUN_EN (1U<<4)
+#define ATI_REG_IER_MODEM_OUT3_XRUN_EN (1U<<6)
+#define ATI_REG_IER_PHYS_INTR_EN (1U<<8)
+#define ATI_REG_IER_PHYS_MISMATCH_EN (1U<<9)
+#define ATI_REG_IER_CODEC0_INTR_EN (1U<<10)
+#define ATI_REG_IER_CODEC1_INTR_EN (1U<<11)
+#define ATI_REG_IER_CODEC2_INTR_EN (1U<<12)
+#define ATI_REG_IER_NEW_FRAME_EN (1U<<13) /* (RO */
+#define ATI_REG_IER_MODEM_GPIO_DATA_EN (1U<<14) /* (WO) modem is running */
+#define ATI_REG_IER_MODEM_SET_BUS_BUSY (1U<<15)
+
+#define ATI_REG_CMD 0x08 /* command */
+#define ATI_REG_CMD_POWERDOWN (1U<<0)
+#define ATI_REG_CMD_MODEM_RECEIVE_EN (1U<<1) /* modem only */
+#define ATI_REG_CMD_MODEM_SEND1_EN (1U<<2) /* modem only */
+#define ATI_REG_CMD_MODEM_SEND2_EN (1U<<3) /* modem only */
+#define ATI_REG_CMD_MODEM_SEND3_EN (1U<<4) /* modem only */
+#define ATI_REG_CMD_MODEM_STATUS_MEM (1U<<5) /* modem only */
+#define ATI_REG_CMD_MODEM_IN_DMA_EN (1U<<8) /* modem only */
+#define ATI_REG_CMD_MODEM_OUT_DMA1_EN (1U<<9) /* modem only */
+#define ATI_REG_CMD_MODEM_OUT_DMA2_EN (1U<<10) /* modem only */
+#define ATI_REG_CMD_MODEM_OUT_DMA3_EN (1U<<11) /* modem only */
+#define ATI_REG_CMD_AUDIO_PRESENT (1U<<20)
+#define ATI_REG_CMD_MODEM_GPIO_THRU_DMA (1U<<22) /* modem only */
+#define ATI_REG_CMD_LOOPBACK_EN (1U<<23)
+#define ATI_REG_CMD_PACKED_DIS (1U<<24)
+#define ATI_REG_CMD_BURST_EN (1U<<25)
+#define ATI_REG_CMD_PANIC_EN (1U<<26)
+#define ATI_REG_CMD_MODEM_PRESENT (1U<<27)
+#define ATI_REG_CMD_ACLINK_ACTIVE (1U<<28)
+#define ATI_REG_CMD_AC_SOFT_RESET (1U<<29)
+#define ATI_REG_CMD_AC_SYNC (1U<<30)
+#define ATI_REG_CMD_AC_RESET (1U<<31)
+
+#define ATI_REG_PHYS_OUT_ADDR 0x0c
+#define ATI_REG_PHYS_OUT_CODEC_MASK (3U<<0)
+#define ATI_REG_PHYS_OUT_RW (1U<<2)
+#define ATI_REG_PHYS_OUT_ADDR_EN (1U<<8)
+#define ATI_REG_PHYS_OUT_ADDR_SHIFT 9
+#define ATI_REG_PHYS_OUT_DATA_SHIFT 16
+
+#define ATI_REG_PHYS_IN_ADDR 0x10
+#define ATI_REG_PHYS_IN_READ_FLAG (1U<<8)
+#define ATI_REG_PHYS_IN_ADDR_SHIFT 9
+#define ATI_REG_PHYS_IN_DATA_SHIFT 16
+
+#define ATI_REG_SLOTREQ 0x14
+
+#define ATI_REG_COUNTER 0x18
+#define ATI_REG_COUNTER_SLOT (3U<<0) /* slot # */
+#define ATI_REG_COUNTER_BITCLOCK (31U<<8)
+
+#define ATI_REG_IN_FIFO_THRESHOLD 0x1c
+
+#define ATI_REG_MODEM_IN_DMA_LINKPTR 0x20
+#define ATI_REG_MODEM_IN_DMA_DT_START 0x24 /* RO */
+#define ATI_REG_MODEM_IN_DMA_DT_NEXT 0x28 /* RO */
+#define ATI_REG_MODEM_IN_DMA_DT_CUR 0x2c /* RO */
+#define ATI_REG_MODEM_IN_DMA_DT_SIZE 0x30
+#define ATI_REG_MODEM_OUT_FIFO 0x34 /* output threshold */
+#define ATI_REG_MODEM_OUT1_DMA_THRESHOLD_MASK (0xf<<16)
+#define ATI_REG_MODEM_OUT1_DMA_THRESHOLD_SHIFT 16
+#define ATI_REG_MODEM_OUT_DMA1_LINKPTR 0x38
+#define ATI_REG_MODEM_OUT_DMA2_LINKPTR 0x3c
+#define ATI_REG_MODEM_OUT_DMA3_LINKPTR 0x40
+#define ATI_REG_MODEM_OUT_DMA1_DT_START 0x44
+#define ATI_REG_MODEM_OUT_DMA1_DT_NEXT 0x48
+#define ATI_REG_MODEM_OUT_DMA1_DT_CUR 0x4c
+#define ATI_REG_MODEM_OUT_DMA2_DT_START 0x50
+#define ATI_REG_MODEM_OUT_DMA2_DT_NEXT 0x54
+#define ATI_REG_MODEM_OUT_DMA2_DT_CUR 0x58
+#define ATI_REG_MODEM_OUT_DMA3_DT_START 0x5c
+#define ATI_REG_MODEM_OUT_DMA3_DT_NEXT 0x60
+#define ATI_REG_MODEM_OUT_DMA3_DT_CUR 0x64
+#define ATI_REG_MODEM_OUT_DMA12_DT_SIZE 0x68
+#define ATI_REG_MODEM_OUT_DMA3_DT_SIZE 0x6c
+#define ATI_REG_MODEM_OUT_FIFO_USED 0x70
+#define ATI_REG_MODEM_OUT_GPIO 0x74
+#define ATI_REG_MODEM_OUT_GPIO_EN 1
+#define ATI_REG_MODEM_OUT_GPIO_DATA_SHIFT 5
+#define ATI_REG_MODEM_IN_GPIO 0x78
+
+#define ATI_REG_MODEM_MIRROR 0x7c
+#define ATI_REG_AUDIO_MIRROR 0x80
+
+#define ATI_REG_MODEM_FIFO_FLUSH 0x88
+#define ATI_REG_MODEM_FIFO_OUT1_FLUSH (1U<<0)
+#define ATI_REG_MODEM_FIFO_OUT2_FLUSH (1U<<1)
+#define ATI_REG_MODEM_FIFO_OUT3_FLUSH (1U<<2)
+#define ATI_REG_MODEM_FIFO_IN_FLUSH (1U<<3)
+
+/* LINKPTR */
+#define ATI_REG_LINKPTR_EN (1U<<0)
+
+#define ATI_MAX_DESCRIPTORS 256 /* max number of descriptor packets */
+
+
+struct atiixp_modem;
+
+/*
+ * DMA packate descriptor
+ */
+
+struct atiixp_dma_desc {
+ u32 addr; /* DMA buffer address */
+ u16 status; /* status bits */
+ u16 size; /* size of the packet in dwords */
+ u32 next; /* address of the next packet descriptor */
+};
+
+/*
+ * stream enum
+ */
+enum { ATI_DMA_PLAYBACK, ATI_DMA_CAPTURE, NUM_ATI_DMAS }; /* DMAs */
+enum { ATI_PCM_OUT, ATI_PCM_IN, NUM_ATI_PCMS }; /* AC97 pcm slots */
+enum { ATI_PCMDEV_ANALOG, NUM_ATI_PCMDEVS }; /* pcm devices */
+
+#define NUM_ATI_CODECS 3
+
+
+/*
+ * constants and callbacks for each DMA type
+ */
+struct atiixp_dma_ops {
+ int type; /* ATI_DMA_XXX */
+ unsigned int llp_offset; /* LINKPTR offset */
+ unsigned int dt_cur; /* DT_CUR offset */
+ /* called from open callback */
+ void (*enable_dma)(struct atiixp_modem *chip, int on);
+ /* called from trigger (START/STOP) */
+ void (*enable_transfer)(struct atiixp_modem *chip, int on);
+ /* called from trigger (STOP only) */
+ void (*flush_dma)(struct atiixp_modem *chip);
+};
+
+/*
+ * DMA stream
+ */
+struct atiixp_dma {
+ const struct atiixp_dma_ops *ops;
+ struct snd_dma_buffer desc_buf;
+ struct snd_pcm_substream *substream; /* assigned PCM substream */
+ unsigned int buf_addr, buf_bytes; /* DMA buffer address, bytes */
+ unsigned int period_bytes, periods;
+ int opened;
+ int running;
+ int pcm_open_flag;
+ int ac97_pcm_type; /* index # of ac97_pcm to access, -1 = not used */
+};
+
+/*
+ * ATI IXP chip
+ */
+struct atiixp_modem {
+ struct snd_card *card;
+ struct pci_dev *pci;
+
+ struct resource *res; /* memory i/o */
+ unsigned long addr;
+ void __iomem *remap_addr;
+ int irq;
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97[NUM_ATI_CODECS];
+
+ spinlock_t reg_lock;
+
+ struct atiixp_dma dmas[NUM_ATI_DMAS];
+ struct ac97_pcm *pcms[NUM_ATI_PCMS];
+ struct snd_pcm *pcmdevs[NUM_ATI_PCMDEVS];
+
+ int max_channels; /* max. channels for PCM out */
+
+ unsigned int codec_not_ready_bits; /* for codec detection */
+
+ int spdif_over_aclink; /* passed from the module option */
+ struct mutex open_mutex; /* playback open mutex */
+};
+
+
+/*
+ */
+static struct pci_device_id snd_atiixp_ids[] = {
+ { 0x1002, 0x434d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */
+ { 0x1002, 0x4378, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
+
+
+/*
+ * lowlevel functions
+ */
+
+/*
+ * update the bits of the given register.
+ * return 1 if the bits changed.
+ */
+static int snd_atiixp_update_bits(struct atiixp_modem *chip, unsigned int reg,
+ unsigned int mask, unsigned int value)
+{
+ void __iomem *addr = chip->remap_addr + reg;
+ unsigned int data, old_data;
+ old_data = data = readl(addr);
+ data &= ~mask;
+ data |= value;
+ if (old_data == data)
+ return 0;
+ writel(data, addr);
+ return 1;
+}
+
+/*
+ * macros for easy use
+ */
+#define atiixp_write(chip,reg,value) \
+ writel(value, chip->remap_addr + ATI_REG_##reg)
+#define atiixp_read(chip,reg) \
+ readl(chip->remap_addr + ATI_REG_##reg)
+#define atiixp_update(chip,reg,mask,val) \
+ snd_atiixp_update_bits(chip, ATI_REG_##reg, mask, val)
+
+/*
+ * handling DMA packets
+ *
+ * we allocate a linear buffer for the DMA, and split it to each packet.
+ * in a future version, a scatter-gather buffer should be implemented.
+ */
+
+#define ATI_DESC_LIST_SIZE \
+ PAGE_ALIGN(ATI_MAX_DESCRIPTORS * sizeof(struct atiixp_dma_desc))
+
+/*
+ * build packets ring for the given buffer size.
+ *
+ * IXP handles the buffer descriptors, which are connected as a linked
+ * list. although we can change the list dynamically, in this version,
+ * a static RING of buffer descriptors is used.
+ *
+ * the ring is built in this function, and is set up to the hardware.
+ */
+static int atiixp_build_dma_packets(struct atiixp_modem *chip,
+ struct atiixp_dma *dma,
+ struct snd_pcm_substream *substream,
+ unsigned int periods,
+ unsigned int period_bytes)
+{
+ unsigned int i;
+ u32 addr, desc_addr;
+ unsigned long flags;
+
+ if (periods > ATI_MAX_DESCRIPTORS)
+ return -ENOMEM;
+
+ if (dma->desc_buf.area == NULL) {
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ ATI_DESC_LIST_SIZE, &dma->desc_buf) < 0)
+ return -ENOMEM;
+ dma->period_bytes = dma->periods = 0; /* clear */
+ }
+
+ if (dma->periods == periods && dma->period_bytes == period_bytes)
+ return 0;
+
+ /* reset DMA before changing the descriptor table */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ writel(0, chip->remap_addr + dma->ops->llp_offset);
+ dma->ops->enable_dma(chip, 0);
+ dma->ops->enable_dma(chip, 1);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ /* fill the entries */
+ addr = (u32)substream->runtime->dma_addr;
+ desc_addr = (u32)dma->desc_buf.addr;
+ for (i = 0; i < periods; i++) {
+ struct atiixp_dma_desc *desc;
+ desc = &((struct atiixp_dma_desc *)dma->desc_buf.area)[i];
+ desc->addr = cpu_to_le32(addr);
+ desc->status = 0;
+ desc->size = period_bytes >> 2; /* in dwords */
+ desc_addr += sizeof(struct atiixp_dma_desc);
+ if (i == periods - 1)
+ desc->next = cpu_to_le32((u32)dma->desc_buf.addr);
+ else
+ desc->next = cpu_to_le32(desc_addr);
+ addr += period_bytes;
+ }
+
+ writel((u32)dma->desc_buf.addr | ATI_REG_LINKPTR_EN,
+ chip->remap_addr + dma->ops->llp_offset);
+
+ dma->period_bytes = period_bytes;
+ dma->periods = periods;
+
+ return 0;
+}
+
+/*
+ * remove the ring buffer and release it if assigned
+ */
+static void atiixp_clear_dma_packets(struct atiixp_modem *chip,
+ struct atiixp_dma *dma,
+ struct snd_pcm_substream *substream)
+{
+ if (dma->desc_buf.area) {
+ writel(0, chip->remap_addr + dma->ops->llp_offset);
+ snd_dma_free_pages(&dma->desc_buf);
+ dma->desc_buf.area = NULL;
+ }
+}
+
+/*
+ * AC97 interface
+ */
+static int snd_atiixp_acquire_codec(struct atiixp_modem *chip)
+{
+ int timeout = 1000;
+
+ while (atiixp_read(chip, PHYS_OUT_ADDR) & ATI_REG_PHYS_OUT_ADDR_EN) {
+ if (! timeout--) {
+ snd_printk(KERN_WARNING "atiixp-modem: codec acquire timeout\n");
+ return -EBUSY;
+ }
+ udelay(1);
+ }
+ return 0;
+}
+
+static unsigned short snd_atiixp_codec_read(struct atiixp_modem *chip,
+ unsigned short codec,
+ unsigned short reg)
+{
+ unsigned int data;
+ int timeout;
+
+ if (snd_atiixp_acquire_codec(chip) < 0)
+ return 0xffff;
+ data = (reg << ATI_REG_PHYS_OUT_ADDR_SHIFT) |
+ ATI_REG_PHYS_OUT_ADDR_EN |
+ ATI_REG_PHYS_OUT_RW |
+ codec;
+ atiixp_write(chip, PHYS_OUT_ADDR, data);
+ if (snd_atiixp_acquire_codec(chip) < 0)
+ return 0xffff;
+ timeout = 1000;
+ do {
+ data = atiixp_read(chip, PHYS_IN_ADDR);
+ if (data & ATI_REG_PHYS_IN_READ_FLAG)
+ return data >> ATI_REG_PHYS_IN_DATA_SHIFT;
+ udelay(1);
+ } while (--timeout);
+ /* time out may happen during reset */
+ if (reg < 0x7c)
+ snd_printk(KERN_WARNING "atiixp-modem: codec read timeout (reg %x)\n", reg);
+ return 0xffff;
+}
+
+
+static void snd_atiixp_codec_write(struct atiixp_modem *chip,
+ unsigned short codec,
+ unsigned short reg, unsigned short val)
+{
+ unsigned int data;
+
+ if (snd_atiixp_acquire_codec(chip) < 0)
+ return;
+ data = ((unsigned int)val << ATI_REG_PHYS_OUT_DATA_SHIFT) |
+ ((unsigned int)reg << ATI_REG_PHYS_OUT_ADDR_SHIFT) |
+ ATI_REG_PHYS_OUT_ADDR_EN | codec;
+ atiixp_write(chip, PHYS_OUT_ADDR, data);
+}
+
+
+static unsigned short snd_atiixp_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct atiixp_modem *chip = ac97->private_data;
+ return snd_atiixp_codec_read(chip, ac97->num, reg);
+
+}
+
+static void snd_atiixp_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct atiixp_modem *chip = ac97->private_data;
+ if (reg == AC97_GPIO_STATUS) {
+ atiixp_write(chip, MODEM_OUT_GPIO,
+ (val << ATI_REG_MODEM_OUT_GPIO_DATA_SHIFT) | ATI_REG_MODEM_OUT_GPIO_EN);
+ return;
+ }
+ snd_atiixp_codec_write(chip, ac97->num, reg, val);
+}
+
+/*
+ * reset AC link
+ */
+static int snd_atiixp_aclink_reset(struct atiixp_modem *chip)
+{
+ int timeout;
+
+ /* reset powerdoewn */
+ if (atiixp_update(chip, CMD, ATI_REG_CMD_POWERDOWN, 0))
+ udelay(10);
+
+ /* perform a software reset */
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SOFT_RESET, ATI_REG_CMD_AC_SOFT_RESET);
+ atiixp_read(chip, CMD);
+ udelay(10);
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SOFT_RESET, 0);
+
+ timeout = 10;
+ while (! (atiixp_read(chip, CMD) & ATI_REG_CMD_ACLINK_ACTIVE)) {
+ /* do a hard reset */
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET,
+ ATI_REG_CMD_AC_SYNC);
+ atiixp_read(chip, CMD);
+ msleep(1);
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_RESET, ATI_REG_CMD_AC_RESET);
+ if (--timeout) {
+ snd_printk(KERN_ERR "atiixp-modem: codec reset timeout\n");
+ break;
+ }
+ }
+
+ /* deassert RESET and assert SYNC to make sure */
+ atiixp_update(chip, CMD, ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET,
+ ATI_REG_CMD_AC_SYNC|ATI_REG_CMD_AC_RESET);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_atiixp_aclink_down(struct atiixp_modem *chip)
+{
+ // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */
+ // return -EBUSY;
+ atiixp_update(chip, CMD,
+ ATI_REG_CMD_POWERDOWN | ATI_REG_CMD_AC_RESET,
+ ATI_REG_CMD_POWERDOWN);
+ return 0;
+}
+#endif
+
+/*
+ * auto-detection of codecs
+ *
+ * the IXP chip can generate interrupts for the non-existing codecs.
+ * NEW_FRAME interrupt is used to make sure that the interrupt is generated
+ * even if all three codecs are connected.
+ */
+
+#define ALL_CODEC_NOT_READY \
+ (ATI_REG_ISR_CODEC0_NOT_READY |\
+ ATI_REG_ISR_CODEC1_NOT_READY |\
+ ATI_REG_ISR_CODEC2_NOT_READY)
+#define CODEC_CHECK_BITS (ALL_CODEC_NOT_READY|ATI_REG_ISR_NEW_FRAME)
+
+static int snd_atiixp_codec_detect(struct atiixp_modem *chip)
+{
+ int timeout;
+
+ chip->codec_not_ready_bits = 0;
+ atiixp_write(chip, IER, CODEC_CHECK_BITS);
+ /* wait for the interrupts */
+ timeout = 50;
+ while (timeout-- > 0) {
+ msleep(1);
+ if (chip->codec_not_ready_bits)
+ break;
+ }
+ atiixp_write(chip, IER, 0); /* disable irqs */
+
+ if ((chip->codec_not_ready_bits & ALL_CODEC_NOT_READY) == ALL_CODEC_NOT_READY) {
+ snd_printk(KERN_ERR "atiixp-modem: no codec detected!\n");
+ return -ENXIO;
+ }
+ return 0;
+}
+
+
+/*
+ * enable DMA and irqs
+ */
+static int snd_atiixp_chip_start(struct atiixp_modem *chip)
+{
+ unsigned int reg;
+
+ /* set up spdif, enable burst mode */
+ reg = atiixp_read(chip, CMD);
+ reg |= ATI_REG_CMD_BURST_EN;
+ if(!(reg & ATI_REG_CMD_MODEM_PRESENT))
+ reg |= ATI_REG_CMD_MODEM_PRESENT;
+ atiixp_write(chip, CMD, reg);
+
+ /* clear all interrupt source */
+ atiixp_write(chip, ISR, 0xffffffff);
+ /* enable irqs */
+ atiixp_write(chip, IER,
+ ATI_REG_IER_MODEM_STATUS_EN |
+ ATI_REG_IER_MODEM_IN_XRUN_EN |
+ ATI_REG_IER_MODEM_OUT1_XRUN_EN);
+ return 0;
+}
+
+
+/*
+ * disable DMA and IRQs
+ */
+static int snd_atiixp_chip_stop(struct atiixp_modem *chip)
+{
+ /* clear interrupt source */
+ atiixp_write(chip, ISR, atiixp_read(chip, ISR));
+ /* disable irqs */
+ atiixp_write(chip, IER, 0);
+ return 0;
+}
+
+
+/*
+ * PCM section
+ */
+
+/*
+ * pointer callback simplly reads XXX_DMA_DT_CUR register as the current
+ * position. when SG-buffer is implemented, the offset must be calculated
+ * correctly...
+ */
+static snd_pcm_uframes_t snd_atiixp_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct atiixp_dma *dma = runtime->private_data;
+ unsigned int curptr;
+ int timeout = 1000;
+
+ while (timeout--) {
+ curptr = readl(chip->remap_addr + dma->ops->dt_cur);
+ if (curptr < dma->buf_addr)
+ continue;
+ curptr -= dma->buf_addr;
+ if (curptr >= dma->buf_bytes)
+ continue;
+ return bytes_to_frames(runtime, curptr);
+ }
+ snd_printd("atiixp-modem: invalid DMA pointer read 0x%x (buf=%x)\n",
+ readl(chip->remap_addr + dma->ops->dt_cur), dma->buf_addr);
+ return 0;
+}
+
+/*
+ * XRUN detected, and stop the PCM substream
+ */
+static void snd_atiixp_xrun_dma(struct atiixp_modem *chip,
+ struct atiixp_dma *dma)
+{
+ if (! dma->substream || ! dma->running)
+ return;
+ snd_printdd("atiixp-modem: XRUN detected (DMA %d)\n", dma->ops->type);
+ snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN);
+}
+
+/*
+ * the period ack. update the substream.
+ */
+static void snd_atiixp_update_dma(struct atiixp_modem *chip,
+ struct atiixp_dma *dma)
+{
+ if (! dma->substream || ! dma->running)
+ return;
+ snd_pcm_period_elapsed(dma->substream);
+}
+
+/* set BUS_BUSY interrupt bit if any DMA is running */
+/* call with spinlock held */
+static void snd_atiixp_check_bus_busy(struct atiixp_modem *chip)
+{
+ unsigned int bus_busy;
+ if (atiixp_read(chip, CMD) & (ATI_REG_CMD_MODEM_SEND1_EN |
+ ATI_REG_CMD_MODEM_RECEIVE_EN))
+ bus_busy = ATI_REG_IER_MODEM_SET_BUS_BUSY;
+ else
+ bus_busy = 0;
+ atiixp_update(chip, IER, ATI_REG_IER_MODEM_SET_BUS_BUSY, bus_busy);
+}
+
+/* common trigger callback
+ * calling the lowlevel callbacks in it
+ */
+static int snd_atiixp_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ struct atiixp_dma *dma = substream->runtime->private_data;
+ int err = 0;
+
+ if (snd_BUG_ON(!dma->ops->enable_transfer ||
+ !dma->ops->flush_dma))
+ return -EINVAL;
+
+ spin_lock(&chip->reg_lock);
+ switch(cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ dma->ops->enable_transfer(chip, 1);
+ dma->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ dma->ops->enable_transfer(chip, 0);
+ dma->running = 0;
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ if (! err) {
+ snd_atiixp_check_bus_busy(chip);
+ if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ dma->ops->flush_dma(chip);
+ snd_atiixp_check_bus_busy(chip);
+ }
+ }
+ spin_unlock(&chip->reg_lock);
+ return err;
+}
+
+
+/*
+ * lowlevel callbacks for each DMA type
+ *
+ * every callback is supposed to be called in chip->reg_lock spinlock
+ */
+
+/* flush FIFO of analog OUT DMA */
+static void atiixp_out_flush_dma(struct atiixp_modem *chip)
+{
+ atiixp_write(chip, MODEM_FIFO_FLUSH, ATI_REG_MODEM_FIFO_OUT1_FLUSH);
+}
+
+/* enable/disable analog OUT DMA */
+static void atiixp_out_enable_dma(struct atiixp_modem *chip, int on)
+{
+ unsigned int data;
+ data = atiixp_read(chip, CMD);
+ if (on) {
+ if (data & ATI_REG_CMD_MODEM_OUT_DMA1_EN)
+ return;
+ atiixp_out_flush_dma(chip);
+ data |= ATI_REG_CMD_MODEM_OUT_DMA1_EN;
+ } else
+ data &= ~ATI_REG_CMD_MODEM_OUT_DMA1_EN;
+ atiixp_write(chip, CMD, data);
+}
+
+/* start/stop transfer over OUT DMA */
+static void atiixp_out_enable_transfer(struct atiixp_modem *chip, int on)
+{
+ atiixp_update(chip, CMD, ATI_REG_CMD_MODEM_SEND1_EN,
+ on ? ATI_REG_CMD_MODEM_SEND1_EN : 0);
+}
+
+/* enable/disable analog IN DMA */
+static void atiixp_in_enable_dma(struct atiixp_modem *chip, int on)
+{
+ atiixp_update(chip, CMD, ATI_REG_CMD_MODEM_IN_DMA_EN,
+ on ? ATI_REG_CMD_MODEM_IN_DMA_EN : 0);
+}
+
+/* start/stop analog IN DMA */
+static void atiixp_in_enable_transfer(struct atiixp_modem *chip, int on)
+{
+ if (on) {
+ unsigned int data = atiixp_read(chip, CMD);
+ if (! (data & ATI_REG_CMD_MODEM_RECEIVE_EN)) {
+ data |= ATI_REG_CMD_MODEM_RECEIVE_EN;
+ atiixp_write(chip, CMD, data);
+ }
+ } else
+ atiixp_update(chip, CMD, ATI_REG_CMD_MODEM_RECEIVE_EN, 0);
+}
+
+/* flush FIFO of analog IN DMA */
+static void atiixp_in_flush_dma(struct atiixp_modem *chip)
+{
+ atiixp_write(chip, MODEM_FIFO_FLUSH, ATI_REG_MODEM_FIFO_IN_FLUSH);
+}
+
+/* set up slots and formats for analog OUT */
+static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ unsigned int data;
+
+ spin_lock_irq(&chip->reg_lock);
+ /* set output threshold */
+ data = atiixp_read(chip, MODEM_OUT_FIFO);
+ data &= ~ATI_REG_MODEM_OUT1_DMA_THRESHOLD_MASK;
+ data |= 0x04 << ATI_REG_MODEM_OUT1_DMA_THRESHOLD_SHIFT;
+ atiixp_write(chip, MODEM_OUT_FIFO, data);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+/* set up slots and formats for analog IN */
+static int snd_atiixp_capture_prepare(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+/*
+ * hw_params - allocate the buffer and set up buffer descriptors
+ */
+static int snd_atiixp_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ struct atiixp_dma *dma = substream->runtime->private_data;
+ int err;
+ int i;
+
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ dma->buf_addr = substream->runtime->dma_addr;
+ dma->buf_bytes = params_buffer_bytes(hw_params);
+
+ err = atiixp_build_dma_packets(chip, dma, substream,
+ params_periods(hw_params),
+ params_period_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ /* set up modem rate */
+ for (i = 0; i < NUM_ATI_CODECS; i++) {
+ if (! chip->ac97[i])
+ continue;
+ snd_ac97_write(chip->ac97[i], AC97_LINE1_RATE, params_rate(hw_params));
+ snd_ac97_write(chip->ac97[i], AC97_LINE1_LEVEL, 0);
+ }
+
+ return err;
+}
+
+static int snd_atiixp_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ struct atiixp_dma *dma = substream->runtime->private_data;
+
+ atiixp_clear_dma_packets(chip, dma, substream);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+
+/*
+ * pcm hardware definition, identical for all DMA types
+ */
+static struct snd_pcm_hardware snd_atiixp_pcm_hw =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 16000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 256 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 128 * 1024,
+ .periods_min = 2,
+ .periods_max = ATI_MAX_DESCRIPTORS,
+};
+
+static int snd_atiixp_pcm_open(struct snd_pcm_substream *substream,
+ struct atiixp_dma *dma, int pcm_type)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ static unsigned int rates[] = { 8000, 9600, 12000, 16000 };
+ static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+
+ if (snd_BUG_ON(!dma->ops || !dma->ops->enable_dma))
+ return -EINVAL;
+
+ if (dma->opened)
+ return -EBUSY;
+ dma->substream = substream;
+ runtime->hw = snd_atiixp_pcm_hw;
+ dma->ac97_pcm_type = pcm_type;
+ if ((err = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ runtime->private_data = dma;
+
+ /* enable DMA bits */
+ spin_lock_irq(&chip->reg_lock);
+ dma->ops->enable_dma(chip, 1);
+ spin_unlock_irq(&chip->reg_lock);
+ dma->opened = 1;
+
+ return 0;
+}
+
+static int snd_atiixp_pcm_close(struct snd_pcm_substream *substream,
+ struct atiixp_dma *dma)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ /* disable DMA bits */
+ if (snd_BUG_ON(!dma->ops || !dma->ops->enable_dma))
+ return -EINVAL;
+ spin_lock_irq(&chip->reg_lock);
+ dma->ops->enable_dma(chip, 0);
+ spin_unlock_irq(&chip->reg_lock);
+ dma->substream = NULL;
+ dma->opened = 0;
+ return 0;
+}
+
+/*
+ */
+static int snd_atiixp_playback_open(struct snd_pcm_substream *substream)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ mutex_lock(&chip->open_mutex);
+ err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_PLAYBACK], 0);
+ mutex_unlock(&chip->open_mutex);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int snd_atiixp_playback_close(struct snd_pcm_substream *substream)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ int err;
+ mutex_lock(&chip->open_mutex);
+ err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_PLAYBACK]);
+ mutex_unlock(&chip->open_mutex);
+ return err;
+}
+
+static int snd_atiixp_capture_open(struct snd_pcm_substream *substream)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ return snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_CAPTURE], 1);
+}
+
+static int snd_atiixp_capture_close(struct snd_pcm_substream *substream)
+{
+ struct atiixp_modem *chip = snd_pcm_substream_chip(substream);
+ return snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_CAPTURE]);
+}
+
+
+/* AC97 playback */
+static struct snd_pcm_ops snd_atiixp_playback_ops = {
+ .open = snd_atiixp_playback_open,
+ .close = snd_atiixp_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_atiixp_pcm_hw_params,
+ .hw_free = snd_atiixp_pcm_hw_free,
+ .prepare = snd_atiixp_playback_prepare,
+ .trigger = snd_atiixp_pcm_trigger,
+ .pointer = snd_atiixp_pcm_pointer,
+};
+
+/* AC97 capture */
+static struct snd_pcm_ops snd_atiixp_capture_ops = {
+ .open = snd_atiixp_capture_open,
+ .close = snd_atiixp_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_atiixp_pcm_hw_params,
+ .hw_free = snd_atiixp_pcm_hw_free,
+ .prepare = snd_atiixp_capture_prepare,
+ .trigger = snd_atiixp_pcm_trigger,
+ .pointer = snd_atiixp_pcm_pointer,
+};
+
+static struct atiixp_dma_ops snd_atiixp_playback_dma_ops = {
+ .type = ATI_DMA_PLAYBACK,
+ .llp_offset = ATI_REG_MODEM_OUT_DMA1_LINKPTR,
+ .dt_cur = ATI_REG_MODEM_OUT_DMA1_DT_CUR,
+ .enable_dma = atiixp_out_enable_dma,
+ .enable_transfer = atiixp_out_enable_transfer,
+ .flush_dma = atiixp_out_flush_dma,
+};
+
+static struct atiixp_dma_ops snd_atiixp_capture_dma_ops = {
+ .type = ATI_DMA_CAPTURE,
+ .llp_offset = ATI_REG_MODEM_IN_DMA_LINKPTR,
+ .dt_cur = ATI_REG_MODEM_IN_DMA_DT_CUR,
+ .enable_dma = atiixp_in_enable_dma,
+ .enable_transfer = atiixp_in_enable_transfer,
+ .flush_dma = atiixp_in_flush_dma,
+};
+
+static int __devinit snd_atiixp_pcm_new(struct atiixp_modem *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ /* initialize constants */
+ chip->dmas[ATI_DMA_PLAYBACK].ops = &snd_atiixp_playback_dma_ops;
+ chip->dmas[ATI_DMA_CAPTURE].ops = &snd_atiixp_capture_dma_ops;
+
+ /* PCM #0: analog I/O */
+ err = snd_pcm_new(chip->card, "ATI IXP MC97", ATI_PCMDEV_ANALOG, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_atiixp_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_atiixp_capture_ops);
+ pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
+ pcm->private_data = chip;
+ strcpy(pcm->name, "ATI IXP MC97");
+ chip->pcmdevs[ATI_PCMDEV_ANALOG] = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 128*1024);
+
+ return 0;
+}
+
+
+
+/*
+ * interrupt handler
+ */
+static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id)
+{
+ struct atiixp_modem *chip = dev_id;
+ unsigned int status;
+
+ status = atiixp_read(chip, ISR);
+
+ if (! status)
+ return IRQ_NONE;
+
+ /* process audio DMA */
+ if (status & ATI_REG_ISR_MODEM_OUT1_XRUN)
+ snd_atiixp_xrun_dma(chip, &chip->dmas[ATI_DMA_PLAYBACK]);
+ else if (status & ATI_REG_ISR_MODEM_OUT1_STATUS)
+ snd_atiixp_update_dma(chip, &chip->dmas[ATI_DMA_PLAYBACK]);
+ if (status & ATI_REG_ISR_MODEM_IN_XRUN)
+ snd_atiixp_xrun_dma(chip, &chip->dmas[ATI_DMA_CAPTURE]);
+ else if (status & ATI_REG_ISR_MODEM_IN_STATUS)
+ snd_atiixp_update_dma(chip, &chip->dmas[ATI_DMA_CAPTURE]);
+
+ /* for codec detection */
+ if (status & CODEC_CHECK_BITS) {
+ unsigned int detected;
+ detected = status & CODEC_CHECK_BITS;
+ spin_lock(&chip->reg_lock);
+ chip->codec_not_ready_bits |= detected;
+ atiixp_update(chip, IER, detected, 0); /* disable the detected irqs */
+ spin_unlock(&chip->reg_lock);
+ }
+
+ /* ack */
+ atiixp_write(chip, ISR, status);
+
+ return IRQ_HANDLED;
+}
+
+
+/*
+ * ac97 mixer section
+ */
+
+static int __devinit snd_atiixp_mixer_new(struct atiixp_modem *chip, int clock)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int i, err;
+ int codec_count;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_atiixp_ac97_write,
+ .read = snd_atiixp_ac97_read,
+ };
+ static unsigned int codec_skip[NUM_ATI_CODECS] = {
+ ATI_REG_ISR_CODEC0_NOT_READY,
+ ATI_REG_ISR_CODEC1_NOT_READY,
+ ATI_REG_ISR_CODEC2_NOT_READY,
+ };
+
+ if (snd_atiixp_codec_detect(chip) < 0)
+ return -ENXIO;
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &pbus)) < 0)
+ return err;
+ pbus->clock = clock;
+ chip->ac97_bus = pbus;
+
+ codec_count = 0;
+ for (i = 0; i < NUM_ATI_CODECS; i++) {
+ if (chip->codec_not_ready_bits & codec_skip[i])
+ continue;
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.pci = chip->pci;
+ ac97.num = i;
+ ac97.scaps = AC97_SCAP_SKIP_AUDIO | AC97_SCAP_POWER_SAVE;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97[i])) < 0) {
+ chip->ac97[i] = NULL; /* to be sure */
+ snd_printdd("atiixp-modem: codec %d not available for modem\n", i);
+ continue;
+ }
+ codec_count++;
+ }
+
+ if (! codec_count) {
+ snd_printk(KERN_ERR "atiixp-modem: no codec available\n");
+ return -ENODEV;
+ }
+
+ /* snd_ac97_tune_hardware(chip->ac97, ac97_quirks); */
+
+ return 0;
+}
+
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct atiixp_modem *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < NUM_ATI_PCMDEVS; i++)
+ snd_pcm_suspend_all(chip->pcmdevs[i]);
+ for (i = 0; i < NUM_ATI_CODECS; i++)
+ snd_ac97_suspend(chip->ac97[i]);
+ snd_atiixp_aclink_down(chip);
+ snd_atiixp_chip_stop(chip);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_atiixp_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct atiixp_modem *chip = card->private_data;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "atiixp-modem: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_atiixp_aclink_reset(chip);
+ snd_atiixp_chip_start(chip);
+
+ for (i = 0; i < NUM_ATI_CODECS; i++)
+ snd_ac97_resume(chip->ac97[i]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+#ifdef CONFIG_PROC_FS
+/*
+ * proc interface for register dump
+ */
+
+static void snd_atiixp_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct atiixp_modem *chip = entry->private_data;
+ int i;
+
+ for (i = 0; i < 256; i += 4)
+ snd_iprintf(buffer, "%02x: %08x\n", i, readl(chip->remap_addr + i));
+}
+
+static void __devinit snd_atiixp_proc_init(struct atiixp_modem *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "atiixp-modem", &entry))
+ snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read);
+}
+#else
+#define snd_atiixp_proc_init(chip)
+#endif
+
+
+/*
+ * destructor
+ */
+
+static int snd_atiixp_free(struct atiixp_modem *chip)
+{
+ if (chip->irq < 0)
+ goto __hw_end;
+ snd_atiixp_chip_stop(chip);
+
+ __hw_end:
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (chip->remap_addr)
+ iounmap(chip->remap_addr);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_atiixp_dev_free(struct snd_device *device)
+{
+ struct atiixp_modem *chip = device->device_data;
+ return snd_atiixp_free(chip);
+}
+
+/*
+ * constructor for chip instance
+ */
+static int __devinit snd_atiixp_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct atiixp_modem **r_chip)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_atiixp_dev_free,
+ };
+ struct atiixp_modem *chip;
+ int err;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&chip->reg_lock);
+ mutex_init(&chip->open_mutex);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ if ((err = pci_request_regions(pci, "ATI IXP MC97")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->addr = pci_resource_start(pci, 0);
+ chip->remap_addr = pci_ioremap_bar(pci, 0);
+ if (chip->remap_addr == NULL) {
+ snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
+ snd_atiixp_free(chip);
+ return -EIO;
+ }
+
+ if (request_irq(pci->irq, snd_atiixp_interrupt, IRQF_SHARED,
+ card->shortname, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_atiixp_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ pci_set_master(pci);
+ synchronize_irq(chip->irq);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_atiixp_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_chip = chip;
+ return 0;
+}
+
+
+static int __devinit snd_atiixp_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct atiixp_modem *chip;
+ int err;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "ATIIXP-MODEM");
+ strcpy(card->shortname, "ATI IXP Modem");
+ if ((err = snd_atiixp_create(card, pci, &chip)) < 0)
+ goto __error;
+ card->private_data = chip;
+
+ if ((err = snd_atiixp_aclink_reset(chip)) < 0)
+ goto __error;
+
+ if ((err = snd_atiixp_mixer_new(chip, ac97_clock)) < 0)
+ goto __error;
+
+ if ((err = snd_atiixp_pcm_new(chip)) < 0)
+ goto __error;
+
+ snd_atiixp_proc_init(chip);
+
+ snd_atiixp_chip_start(chip);
+
+ sprintf(card->longname, "%s rev %x at 0x%lx, irq %i",
+ card->shortname, pci->revision, chip->addr, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto __error;
+
+ pci_set_drvdata(pci, card);
+ return 0;
+
+ __error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_atiixp_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "ATI IXP MC97 controller",
+ .id_table = snd_atiixp_ids,
+ .probe = snd_atiixp_probe,
+ .remove = __devexit_p(snd_atiixp_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_atiixp_suspend,
+ .resume = snd_atiixp_resume,
+#endif
+};
+
+
+static int __init alsa_card_atiixp_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_atiixp_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_atiixp_init)
+module_exit(alsa_card_atiixp_exit)
diff --git a/sound/pci/au88x0/Makefile b/sound/pci/au88x0/Makefile
new file mode 100644
index 0000000..d0a66bc
--- /dev/null
+++ b/sound/pci/au88x0/Makefile
@@ -0,0 +1,7 @@
+snd-au8810-objs := au8810.o
+snd-au8820-objs := au8820.o
+snd-au8830-objs := au8830.o
+
+obj-$(CONFIG_SND_AU8810) += snd-au8810.o
+obj-$(CONFIG_SND_AU8820) += snd-au8820.o
+obj-$(CONFIG_SND_AU8830) += snd-au8830.o
diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c
new file mode 100644
index 0000000..fce22c7
--- /dev/null
+++ b/sound/pci/au88x0/au8810.c
@@ -0,0 +1,17 @@
+#include "au8810.h"
+#include "au88x0.h"
+static struct pci_device_id snd_vortex_ids[] = {
+ {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1,},
+ {0,}
+};
+
+#include "au88x0_core.c"
+#include "au88x0_pcm.c"
+#include "au88x0_mixer.c"
+#include "au88x0_mpu401.c"
+#include "au88x0_game.c"
+#include "au88x0_eq.c"
+#include "au88x0_a3d.c"
+#include "au88x0_xtalk.c"
+#include "au88x0.c"
diff --git a/sound/pci/au88x0/au8810.h b/sound/pci/au88x0/au8810.h
new file mode 100644
index 0000000..5d69c31
--- /dev/null
+++ b/sound/pci/au88x0/au8810.h
@@ -0,0 +1,224 @@
+/*
+ Aureal Advantage Soundcard driver.
+ */
+
+#define CHIP_AU8810
+
+#define CARD_NAME "Aureal Advantage 3D Sound Processor"
+#define CARD_NAME_SHORT "au8810"
+
+#define NR_ADB 0x10
+#define NR_WT 0x00
+#define NR_SRC 0x10
+#define NR_A3D 0x10
+#define NR_MIXIN 0x20
+#define NR_MIXOUT 0x10
+
+
+/* ADBDMA */
+#define VORTEX_ADBDMA_STAT 0x27e00 /* read only, subbuffer, DMA pos */
+#define POS_MASK 0x00000fff
+#define POS_SHIFT 0x0
+#define ADB_SUBBUF_MASK 0x00003000 /* ADB only. */
+#define ADB_SUBBUF_SHIFT 0xc /* ADB only. */
+#define VORTEX_ADBDMA_CTRL 0x27180 /* write only; format, flags, DMA pos */
+#define OFFSET_MASK 0x00000fff
+#define OFFSET_SHIFT 0x0
+#define IE_MASK 0x00001000 /* interrupt enable. */
+#define IE_SHIFT 0xc
+#define DIR_MASK 0x00002000 /* Direction */
+#define DIR_SHIFT 0xd
+#define FMT_MASK 0x0003c000
+#define FMT_SHIFT 0xe
+// The ADB masks and shift also are valid for the wtdma, except if specified otherwise.
+#define VORTEX_ADBDMA_BUFCFG0 0x27100
+#define VORTEX_ADBDMA_BUFCFG1 0x27104
+#define VORTEX_ADBDMA_BUFBASE 0x27000
+#define VORTEX_ADBDMA_START 0x27c00 /* Which subbuffer starts */
+
+#define VORTEX_ADBDMA_STATUS 0x27A90 /* stored at AdbDma->this_10 / 2 DWORD in size. */
+
+/* WTDMA */
+#define VORTEX_WTDMA_CTRL 0x27fd8 /* format, DMA pos */
+#define VORTEX_WTDMA_STAT 0x27fe8 /* DMA subbuf, DMA pos */
+#define WT_SUBBUF_MASK 0x3
+#define WT_SUBBUF_SHIFT 0xc
+#define VORTEX_WTDMA_BUFBASE 0x27fc0
+#define VORTEX_WTDMA_BUFCFG0 0x27fd0
+#define VORTEX_WTDMA_BUFCFG1 0x27fd4
+#define VORTEX_WTDMA_START 0x27fe4 /* which subbuffer is first */
+
+/* ADB */
+#define VORTEX_ADB_SR 0x28400 /* Samplerates enable/disable */
+#define VORTEX_ADB_RTBASE 0x28000
+#define VORTEX_ADB_RTBASE_COUNT 173
+#define VORTEX_ADB_CHNBASE 0x282b4
+#define VORTEX_ADB_CHNBASE_COUNT 24
+#define ROUTE_MASK 0xffff
+#define SOURCE_MASK 0xff00
+#define ADB_MASK 0xff
+#define ADB_SHIFT 0x8
+
+/* ADB address */
+#define OFFSET_ADBDMA 0x00
+#define OFFSET_SRCIN 0x40
+#define OFFSET_SRCOUT 0x20
+#define OFFSET_MIXIN 0x50
+#define OFFSET_MIXOUT 0x30
+#define OFFSET_CODECIN 0x70
+#define OFFSET_CODECOUT 0x88
+#define OFFSET_SPORTIN 0x78 /* ch 0x13 */
+#define OFFSET_SPORTOUT 0x90
+#define OFFSET_SPDIFOUT 0x92 /* ch 0x14 check this! */
+#define OFFSET_EQIN 0xa0
+#define OFFSET_EQOUT 0x7e /* 2 routes on ch 0x11 */
+#define OFFSET_XTALKOUT 0x66 /* crosstalk canceller (source) */
+#define OFFSET_XTALKIN 0x96 /* crosstalk canceller (sink) */
+#define OFFSET_A3DIN 0x70 /* ADB sink. */
+#define OFFSET_A3DOUT 0xA6 /* ADB source. 2 routes per slice = 8 */
+#define OFFSET_EFXIN 0x80 /* ADB sink. */
+#define OFFSET_EFXOUT 0x68 /* ADB source. */
+
+/* ADB route translate helper */
+#define ADB_DMA(x) (x)
+#define ADB_SRCOUT(x) (x + OFFSET_SRCOUT)
+#define ADB_SRCIN(x) (x + OFFSET_SRCIN)
+#define ADB_MIXOUT(x) (x + OFFSET_MIXOUT)
+#define ADB_MIXIN(x) (x + OFFSET_MIXIN)
+#define ADB_CODECIN(x) (x + OFFSET_CODECIN)
+#define ADB_CODECOUT(x) (x + OFFSET_CODECOUT)
+#define ADB_SPORTIN(x) (x + OFFSET_SPORTIN)
+#define ADB_SPORTOUT(x) (x + OFFSET_SPORTOUT)
+#define ADB_SPDIFOUT(x) (x + OFFSET_SPDIFOUT)
+#define ADB_EQIN(x) (x + OFFSET_EQIN)
+#define ADB_EQOUT(x) (x + OFFSET_EQOUT)
+#define ADB_A3DOUT(x) (x + OFFSET_A3DOUT) /* 0x10 A3D blocks */
+#define ADB_A3DIN(x) (x + OFFSET_A3DIN)
+#define ADB_XTALKIN(x) (x + OFFSET_XTALKIN)
+#define ADB_XTALKOUT(x) (x + OFFSET_XTALKOUT)
+
+#define MIX_OUTL 0xe
+#define MIX_OUTR 0xf
+#define MIX_INL 0x1e
+#define MIX_INR 0x1f
+#define MIX_DEFIGAIN 0x08 /* 0x8 => 6dB */
+#define MIX_DEFOGAIN 0x08
+
+/* MIXER */
+#define VORTEX_MIXER_SR 0x21f00
+#define VORTEX_MIXER_CLIP 0x21f80
+#define VORTEX_MIXER_CHNBASE 0x21e40
+#define VORTEX_MIXER_RTBASE 0x21e00
+#define MIXER_RTBASE_SIZE 0x38
+#define VORTEX_MIX_ENIN 0x21a00 /* Input enable bits. 4 bits wide. */
+#define VORTEX_MIX_SMP 0x21c00 /* AU8820: 0x9c00 */
+
+/* MIX */
+#define VORTEX_MIX_INVOL_A 0x21000 /* in? */
+#define VORTEX_MIX_INVOL_B 0x20000 /* out? */
+#define VORTEX_MIX_VOL_A 0x21800
+#define VORTEX_MIX_VOL_B 0x20800
+
+#define VOL_MIN 0x80 /* Input volume when muted. */
+#define VOL_MAX 0x7f /* FIXME: Not confirmed! Just guessed. */
+
+/* SRC */
+#define VORTEX_SRC_CHNBASE 0x26c40
+#define VORTEX_SRC_RTBASE 0x26c00
+#define VORTEX_SRCBLOCK_SR 0x26cc0
+#define VORTEX_SRC_SOURCE 0x26cc4
+#define VORTEX_SRC_SOURCESIZE 0x26cc8
+/* Params
+ 0x26e00 : 1 U0
+ 0x26e40 : 2 CR
+ 0x26e80 : 3 U3
+ 0x26ec0 : 4 DRIFT1
+ 0x26f00 : 5 U1
+ 0x26f40 : 6 DRIFT2
+ 0x26f80 : 7 U2 : Target rate, direction
+*/
+
+#define VORTEX_SRC_CONVRATIO 0x26e40
+#define VORTEX_SRC_DRIFT0 0x26e80
+#define VORTEX_SRC_DRIFT1 0x26ec0
+#define VORTEX_SRC_DRIFT2 0x26f40
+#define VORTEX_SRC_U0 0x26e00
+#define U0_SLOWLOCK 0x200
+#define VORTEX_SRC_U1 0x26f00
+#define VORTEX_SRC_U2 0x26f80
+#define VORTEX_SRC_DATA 0x26800 /* 0xc800 */
+#define VORTEX_SRC_DATA0 0x26000
+
+/* FIFO */
+#define VORTEX_FIFO_ADBCTRL 0x16100 /* Control bits. */
+#define VORTEX_FIFO_WTCTRL 0x16000
+#define FIFO_RDONLY 0x00000001
+#define FIFO_CTRL 0x00000002 /* Allow ctrl. ? */
+#define FIFO_VALID 0x00000010
+#define FIFO_EMPTY 0x00000020
+#define FIFO_U0 0x00001000 /* Unknown. */
+#define FIFO_U1 0x00010000
+#define FIFO_SIZE_BITS 5
+#define FIFO_SIZE (1<<FIFO_SIZE_BITS) // 0x20
+#define FIFO_MASK (FIFO_SIZE-1) //0x1f /* at shift left 0xc */
+//#define FIFO_MASK 0x1f /* at shift left 0xb */
+//#define FIFO_SIZE 0x20
+#define FIFO_BITS 0x03880000
+#define VORTEX_FIFO_ADBDATA 0x14000
+#define VORTEX_FIFO_WTDATA 0x10000
+
+/* CODEC */
+#define VORTEX_CODEC_CTRL 0x29184
+#define VORTEX_CODEC_EN 0x29190
+#define EN_CODEC0 0x00000300
+#define EN_AC98 0x00000c00 /* Modem AC98 slots. */
+#define EN_CODEC1 0x00003000
+#define EN_CODEC (EN_CODEC0 | EN_CODEC1)
+#define EN_SPORT 0x00030000
+#define EN_SPDIF 0x000c0000
+
+#define VORTEX_CODEC_CHN 0x29080
+#define VORTEX_CODEC_IO 0x29188
+
+/* SPDIF */
+#define VORTEX_SPDIF_FLAGS 0x2205c
+#define VORTEX_SPDIF_CFG0 0x291D0
+#define VORTEX_SPDIF_CFG1 0x291D4
+#define VORTEX_SPDIF_SMPRATE 0x29194
+
+/* Sample timer */
+#define VORTEX_SMP_TIME 0x29198
+
+#define VORTEX_MODEM_CTRL 0x291ac
+
+/* IRQ */
+#define VORTEX_IRQ_SOURCE 0x2a000 /* Interrupt source flags. */
+#define VORTEX_IRQ_CTRL 0x2a004 /* Interrupt source mask. */
+
+#define VORTEX_STAT 0x2a008 /* Status */
+
+#define VORTEX_CTRL 0x2a00c
+#define CTRL_MIDI_EN 0x00000001
+#define CTRL_MIDI_PORT 0x00000060
+#define CTRL_GAME_EN 0x00000008
+#define CTRL_GAME_PORT 0x00000e00
+//#define CTRL_IRQ_ENABLE 0x01004000
+#define CTRL_IRQ_ENABLE 0x00004000
+
+/* write: Timer period config / read: TIMER IRQ ack. */
+#define VORTEX_IRQ_STAT 0x2919c
+
+/* DMA */
+#define VORTEX_ENGINE_CTRL 0x27ae8
+#define ENGINE_INIT 0x1380000
+
+/* MIDI *//* GAME. */
+#define VORTEX_MIDI_DATA 0x28800
+#define VORTEX_MIDI_CMD 0x28804 /* Write command / Read status */
+
+#define VORTEX_CTRL2 0x2880c
+#define CTRL2_GAME_ADCMODE 0x40
+#define VORTEX_GAME_LEGACY 0x28808
+#define VORTEX_GAME_AXIS 0x28810
+#define AXIS_SIZE 4
+#define AXIS_RANGE 0x1fff
diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c
new file mode 100644
index 0000000..d1fbcce
--- /dev/null
+++ b/sound/pci/au88x0/au8820.c
@@ -0,0 +1,15 @@
+#include "au8820.h"
+#include "au88x0.h"
+static struct pci_device_id snd_vortex_ids[] = {
+ {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,},
+ {0,}
+};
+
+#include "au88x0_synth.c"
+#include "au88x0_core.c"
+#include "au88x0_pcm.c"
+#include "au88x0_mpu401.c"
+#include "au88x0_game.c"
+#include "au88x0_mixer.c"
+#include "au88x0.c"
diff --git a/sound/pci/au88x0/au8820.h b/sound/pci/au88x0/au8820.h
new file mode 100644
index 0000000..abbe85e
--- /dev/null
+++ b/sound/pci/au88x0/au8820.h
@@ -0,0 +1,204 @@
+/*
+ Aureal Vortex Soundcard driver.
+
+ IO addr collected from asp4core.vxd:
+ function address
+ 0005D5A0 13004
+ 00080674 14004
+ 00080AFF 12818
+
+ */
+
+#define CHIP_AU8820
+
+#define CARD_NAME "Aureal Vortex 3D Sound Processor"
+#define CARD_NAME_SHORT "au8820"
+
+/* Number of ADB and WT channels */
+#define NR_ADB 0x10
+#define NR_WT 0x20
+#define NR_SRC 0x10
+#define NR_A3D 0x00
+#define NR_MIXIN 0x10
+#define NR_MIXOUT 0x10
+
+
+/* ADBDMA */
+#define VORTEX_ADBDMA_STAT 0x105c0 /* read only, subbuffer, DMA pos */
+#define POS_MASK 0x00000fff
+#define POS_SHIFT 0x0
+#define ADB_SUBBUF_MASK 0x00003000 /* ADB only. */
+#define ADB_SUBBUF_SHIFT 0xc /* ADB only. */
+#define VORTEX_ADBDMA_CTRL 0x10580 /* write only, format, flags, DMA pos */
+#define OFFSET_MASK 0x00000fff
+#define OFFSET_SHIFT 0x0
+#define IE_MASK 0x00001000 /* interrupt enable. */
+#define IE_SHIFT 0xc
+#define DIR_MASK 0x00002000 /* Direction. */
+#define DIR_SHIFT 0xd
+#define FMT_MASK 0x0003c000
+#define FMT_SHIFT 0xe
+// The masks and shift also work for the wtdma, if not specified otherwise.
+#define VORTEX_ADBDMA_BUFCFG0 0x10400
+#define VORTEX_ADBDMA_BUFCFG1 0x10404
+#define VORTEX_ADBDMA_BUFBASE 0x10200
+#define VORTEX_ADBDMA_START 0x106c0 /* Which subbuffer starts */
+#define VORTEX_ADBDMA_STATUS 0x10600 /* stored at AdbDma->this_10 / 2 DWORD in size. */
+
+/* ADB */
+#define VORTEX_ADB_SR 0x10a00 /* Samplerates enable/disable */
+#define VORTEX_ADB_RTBASE 0x10800
+#define VORTEX_ADB_RTBASE_COUNT 103
+#define VORTEX_ADB_CHNBASE 0x1099c
+#define VORTEX_ADB_CHNBASE_COUNT 22
+#define ROUTE_MASK 0x3fff
+#define ADB_MASK 0x7f
+#define ADB_SHIFT 0x7
+//#define ADB_MIX_MASK 0xf
+/* ADB address */
+#define OFFSET_ADBDMA 0x00
+#define OFFSET_SRCOUT 0x10 /* on channel 0x11 */
+#define OFFSET_SRCIN 0x10 /* on channel < 0x11 */
+#define OFFSET_MIXOUT 0x20 /* source */
+#define OFFSET_MIXIN 0x30 /* sink */
+#define OFFSET_CODECIN 0x48 /* ADB source */
+#define OFFSET_CODECOUT 0x58 /* ADB sink/target */
+#define OFFSET_SPORTOUT 0x60 /* sink */
+#define OFFSET_SPORTIN 0x50 /* source */
+#define OFFSET_EFXOUT 0x50 /* sink */
+#define OFFSET_EFXIN 0x40 /* source */
+#define OFFSET_A3DOUT 0x00 /* This card has no HRTF :( */
+#define OFFSET_A3DIN 0x00
+#define OFFSET_WTOUT 0x58 /* */
+
+/* ADB route translate helper */
+#define ADB_DMA(x) (x + OFFSET_ADBDMA)
+#define ADB_SRCOUT(x) (x + OFFSET_SRCOUT)
+#define ADB_SRCIN(x) (x + OFFSET_SRCIN)
+#define ADB_MIXOUT(x) (x + OFFSET_MIXOUT)
+#define ADB_MIXIN(x) (x + OFFSET_MIXIN)
+#define ADB_CODECIN(x) (x + OFFSET_CODECIN)
+#define ADB_CODECOUT(x) (x + OFFSET_CODECOUT)
+#define ADB_SPORTOUT(x) (x + OFFSET_SPORTOUT)
+#define ADB_SPORTIN(x) (x + OFFSET_SPORTIN) /* */
+#define ADB_A3DOUT(x) (x + OFFSET_A3DOUT) /* 8 A3D blocks */
+#define ADB_A3DIN(x) (x + OFFSET_A3DIN)
+#define ADB_WTOUT(x,y) (y + OFFSET_WTOUT)
+
+/* WTDMA */
+#define VORTEX_WTDMA_CTRL 0x10500 /* format, DMA pos */
+#define VORTEX_WTDMA_STAT 0x10500 /* DMA subbuf, DMA pos */
+#define WT_SUBBUF_MASK (0x3 << WT_SUBBUF_SHIFT)
+#define WT_SUBBUF_SHIFT 0x15
+#define VORTEX_WTDMA_BUFBASE 0x10000
+#define VORTEX_WTDMA_BUFCFG0 0x10300
+#define VORTEX_WTDMA_BUFCFG1 0x10304
+#define VORTEX_WTDMA_START 0x10640 /* which subbuffer is first */
+
+#define VORTEX_WT_BASE 0x9000
+
+/* MIXER */
+#define VORTEX_MIXER_SR 0x9f00
+#define VORTEX_MIXER_CLIP 0x9f80
+#define VORTEX_MIXER_CHNBASE 0x9e40
+#define VORTEX_MIXER_RTBASE 0x9e00
+#define MIXER_RTBASE_SIZE 0x26
+#define VORTEX_MIX_ENIN 0x9a00 /* Input enable bits. 4 bits wide. */
+#define VORTEX_MIX_SMP 0x9c00
+
+/* MIX */
+#define VORTEX_MIX_INVOL_A 0x9000 /* in? */
+#define VORTEX_MIX_INVOL_B 0x8000 /* out? */
+#define VORTEX_MIX_VOL_A 0x9800
+#define VORTEX_MIX_VOL_B 0x8800
+
+#define VOL_MIN 0x80 /* Input volume when muted. */
+#define VOL_MAX 0x7f /* FIXME: Not confirmed! Just guessed. */
+
+//#define MIX_OUTL 0xe
+//#define MIX_OUTR 0xf
+//#define MIX_INL 0xe
+//#define MIX_INR 0xf
+#define MIX_DEFIGAIN 0x08 /* 0x8 => 6dB */
+#define MIX_DEFOGAIN 0x08
+
+/* SRC */
+#define VORTEX_SRCBLOCK_SR 0xccc0
+#define VORTEX_SRC_CHNBASE 0xcc40
+#define VORTEX_SRC_RTBASE 0xcc00
+#define VORTEX_SRC_SOURCE 0xccc4
+#define VORTEX_SRC_SOURCESIZE 0xccc8
+#define VORTEX_SRC_U0 0xce00
+#define VORTEX_SRC_DRIFT0 0xce80
+#define VORTEX_SRC_DRIFT1 0xcec0
+#define VORTEX_SRC_U1 0xcf00
+#define VORTEX_SRC_DRIFT2 0xcf40
+#define VORTEX_SRC_U2 0xcf80
+#define VORTEX_SRC_DATA 0xc800
+#define VORTEX_SRC_DATA0 0xc000
+#define VORTEX_SRC_CONVRATIO 0xce40
+//#define SRC_RATIO(x) ((((x<<15)/48000) + 1)/2) /* Playback */
+//#define SRC_RATIO2(x) ((((48000<<15)/x) + 1)/2) /* Recording */
+
+/* FIFO */
+#define VORTEX_FIFO_ADBCTRL 0xf800 /* Control bits. */
+#define VORTEX_FIFO_WTCTRL 0xf840
+#define FIFO_RDONLY 0x00000001
+#define FIFO_CTRL 0x00000002 /* Allow ctrl. ? */
+#define FIFO_VALID 0x00000010
+#define FIFO_EMPTY 0x00000020
+#define FIFO_U0 0x00001000 /* Unknown. */
+#define FIFO_U1 0x00010000
+#define FIFO_SIZE_BITS 5
+#define FIFO_SIZE (1<<FIFO_SIZE_BITS) // 0x20
+#define FIFO_MASK (FIFO_SIZE-1) //0x1f /* at shift left 0xc */
+#define VORTEX_FIFO_ADBDATA 0xe000
+#define VORTEX_FIFO_WTDATA 0xe800
+
+/* CODEC */
+#define VORTEX_CODEC_CTRL 0x11984
+#define VORTEX_CODEC_EN 0x11990
+#define EN_CODEC 0x00000300
+#define EN_SPORT 0x00030000
+#define EN_SPDIF 0x000c0000
+#define VORTEX_CODEC_CHN 0x11880
+#define VORTEX_CODEC_IO 0x11988
+
+#define VORTEX_SPDIF_FLAGS 0x1005c /* FIXME */
+#define VORTEX_SPDIF_CFG0 0x119D0
+#define VORTEX_SPDIF_CFG1 0x119D4
+#define VORTEX_SPDIF_SMPRATE 0x11994
+
+/* Sample timer */
+#define VORTEX_SMP_TIME 0x11998
+
+/* IRQ */
+#define VORTEX_IRQ_SOURCE 0x12800 /* Interrupt source flags. */
+#define VORTEX_IRQ_CTRL 0x12804 /* Interrupt source mask. */
+
+#define VORTEX_STAT 0x12808 /* ?? */
+
+#define VORTEX_CTRL 0x1280c
+#define CTRL_MIDI_EN 0x00000001
+#define CTRL_MIDI_PORT 0x00000060
+#define CTRL_GAME_EN 0x00000008
+#define CTRL_GAME_PORT 0x00000e00
+#define CTRL_IRQ_ENABLE 0x4000
+
+/* write: Timer period config / read: TIMER IRQ ack. */
+#define VORTEX_IRQ_STAT 0x1199c
+
+/* DMA */
+#define VORTEX_DMA_BUFFER 0x10200
+#define VORTEX_ENGINE_CTRL 0x1060c
+#define ENGINE_INIT 0x0L
+
+ /* MIDI *//* GAME. */
+#define VORTEX_MIDI_DATA 0x11000
+#define VORTEX_MIDI_CMD 0x11004 /* Write command / Read status */
+#define VORTEX_GAME_LEGACY 0x11008
+#define VORTEX_CTRL2 0x1100c
+#define CTRL2_GAME_ADCMODE 0x40
+#define VORTEX_GAME_AXIS 0x11010
+#define AXIS_SIZE 4
+#define AXIS_RANGE 0x1fff
diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c
new file mode 100644
index 0000000..d4f2717
--- /dev/null
+++ b/sound/pci/au88x0/au8830.c
@@ -0,0 +1,18 @@
+#include "au8830.h"
+#include "au88x0.h"
+static struct pci_device_id snd_vortex_ids[] = {
+ {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,},
+ {0,}
+};
+
+#include "au88x0_synth.c"
+#include "au88x0_core.c"
+#include "au88x0_pcm.c"
+#include "au88x0_mixer.c"
+#include "au88x0_mpu401.c"
+#include "au88x0_game.c"
+#include "au88x0_eq.c"
+#include "au88x0_a3d.c"
+#include "au88x0_xtalk.c"
+#include "au88x0.c"
diff --git a/sound/pci/au88x0/au8830.h b/sound/pci/au88x0/au8830.h
new file mode 100644
index 0000000..04ece1b
--- /dev/null
+++ b/sound/pci/au88x0/au8830.h
@@ -0,0 +1,251 @@
+/*
+ Aureal Vortex Soundcard driver.
+
+ IO addr collected from asp4core.vxd:
+ function address
+ 0005D5A0 13004
+ 00080674 14004
+ 00080AFF 12818
+
+ */
+
+#define CHIP_AU8830
+
+#define CARD_NAME "Aureal Vortex 2 3D Sound Processor"
+#define CARD_NAME_SHORT "au8830"
+
+#define NR_ADB 0x20
+#define NR_SRC 0x10
+#define NR_A3D 0x10
+#define NR_MIXIN 0x20
+#define NR_MIXOUT 0x10
+#define NR_WT 0x40
+
+/* ADBDMA */
+#define VORTEX_ADBDMA_STAT 0x27e00 /* read only, subbuffer, DMA pos */
+#define POS_MASK 0x00000fff
+#define POS_SHIFT 0x0
+#define ADB_SUBBUF_MASK 0x00003000 /* ADB only. */
+#define ADB_SUBBUF_SHIFT 0xc /* ADB only. */
+#define VORTEX_ADBDMA_CTRL 0x27a00 /* write only; format, flags, DMA pos */
+#define OFFSET_MASK 0x00000fff
+#define OFFSET_SHIFT 0x0
+#define IE_MASK 0x00001000 /* interrupt enable. */
+#define IE_SHIFT 0xc
+#define DIR_MASK 0x00002000 /* Direction. */
+#define DIR_SHIFT 0xd
+#define FMT_MASK 0x0003c000
+#define FMT_SHIFT 0xe
+#define ADB_FIFO_EN_SHIFT 0x15
+#define ADB_FIFO_EN (1 << 0x15)
+// The ADB masks and shift also are valid for the wtdma, except if specified otherwise.
+#define VORTEX_ADBDMA_BUFCFG0 0x27800
+#define VORTEX_ADBDMA_BUFCFG1 0x27804
+#define VORTEX_ADBDMA_BUFBASE 0x27400
+#define VORTEX_ADBDMA_START 0x27c00 /* Which subbuffer starts */
+
+#define VORTEX_ADBDMA_STATUS 0x27A90 /* stored at AdbDma->this_10 / 2 DWORD in size. */
+/* Starting at the MSB, each pair of bits seem to be the current DMA page. */
+/* This current page bits are consistent (same value) with VORTEX_ADBDMA_STAT) */
+
+/* DMA */
+#define VORTEX_ENGINE_CTRL 0x27ae8
+#define ENGINE_INIT 0x1380000
+
+/* WTDMA */
+#define VORTEX_WTDMA_CTRL 0x27900 /* format, DMA pos */
+#define VORTEX_WTDMA_STAT 0x27d00 /* DMA subbuf, DMA pos */
+#define WT_SUBBUF_MASK 0x3
+#define WT_SUBBUF_SHIFT 0xc
+#define VORTEX_WTDMA_BUFBASE 0x27000
+#define VORTEX_WTDMA_BUFCFG0 0x27600
+#define VORTEX_WTDMA_BUFCFG1 0x27604
+#define VORTEX_WTDMA_START 0x27b00 /* which subbuffer is first */
+
+/* ADB */
+#define VORTEX_ADB_SR 0x28400 /* Samplerates enable/disable */
+#define VORTEX_ADB_RTBASE 0x28000
+#define VORTEX_ADB_RTBASE_COUNT 173
+#define VORTEX_ADB_CHNBASE 0x282b4
+#define VORTEX_ADB_CHNBASE_COUNT 24
+#define ROUTE_MASK 0xffff
+#define SOURCE_MASK 0xff00
+#define ADB_MASK 0xff
+#define ADB_SHIFT 0x8
+/* ADB address */
+#define OFFSET_ADBDMA 0x00
+#define OFFSET_ADBDMAB 0x20
+#define OFFSET_SRCIN 0x40
+#define OFFSET_SRCOUT 0x20 /* ch 0x11 */
+#define OFFSET_MIXIN 0x50 /* ch 0x11 */
+#define OFFSET_MIXOUT 0x30 /* ch 0x11 */
+#define OFFSET_CODECIN 0x70 /* ch 0x11 */ /* adb source */
+#define OFFSET_CODECOUT 0x88 /* ch 0x11 */ /* adb target */
+#define OFFSET_SPORTIN 0x78 /* ch 0x13 ADB source. 2 routes. */
+#define OFFSET_SPORTOUT 0x90 /* ch 0x13 ADB sink. 2 routes. */
+#define OFFSET_SPDIFIN 0x7A /* ch 0x14 ADB source. */
+#define OFFSET_SPDIFOUT 0x92 /* ch 0x14 ADB sink. */
+#define OFFSET_AC98IN 0x7c /* ch 0x14 ADB source. */
+#define OFFSET_AC98OUT 0x94 /* ch 0x14 ADB sink. */
+#define OFFSET_EQIN 0xa0 /* ch 0x11 */
+#define OFFSET_EQOUT 0x7e /* ch 0x11 */ /* 2 routes on ch 0x11 */
+#define OFFSET_A3DIN 0x70 /* ADB sink. */
+#define OFFSET_A3DOUT 0xA6 /* ADB source. 2 routes per slice = 8 */
+#define OFFSET_WT0 0x40 /* WT bank 0 output. 0x40 - 0x65 */
+#define OFFSET_WT1 0x80 /* WT bank 1 output. 0x80 - 0xA5 */
+/* WT sources offset : 0x00-0x1f Direct stream. */
+/* WT sources offset : 0x20-0x25 Mixed Output. */
+#define OFFSET_XTALKOUT 0x66 /* crosstalk canceller (source) 2 routes */
+#define OFFSET_XTALKIN 0x96 /* crosstalk canceller (sink). 10 routes */
+#define OFFSET_EFXOUT 0x68 /* ADB source. 8 routes. */
+#define OFFSET_EFXIN 0x80 /* ADB sink. 8 routes. */
+
+/* ADB route translate helper */
+#define ADB_DMA(x) (x)
+#define ADB_SRCOUT(x) (x + OFFSET_SRCOUT)
+#define ADB_SRCIN(x) (x + OFFSET_SRCIN)
+#define ADB_MIXOUT(x) (x + OFFSET_MIXOUT)
+#define ADB_MIXIN(x) (x + OFFSET_MIXIN)
+#define ADB_CODECIN(x) (x + OFFSET_CODECIN)
+#define ADB_CODECOUT(x) (x + OFFSET_CODECOUT)
+#define ADB_SPORTIN(x) (x + OFFSET_SPORTIN)
+#define ADB_SPORTOUT(x) (x + OFFSET_SPORTOUT)
+#define ADB_SPDIFIN(x) (x + OFFSET_SPDIFIN)
+#define ADB_SPDIFOUT(x) (x + OFFSET_SPDIFOUT)
+#define ADB_EQIN(x) (x + OFFSET_EQIN)
+#define ADB_EQOUT(x) (x + OFFSET_EQOUT)
+#define ADB_A3DOUT(x) (x + OFFSET_A3DOUT) /* 0x10 A3D blocks */
+#define ADB_A3DIN(x) (x + OFFSET_A3DIN)
+//#define ADB_WTOUT(x) ((x<x20)?(x + OFFSET_WT0):(x + OFFSET_WT1))
+#define ADB_WTOUT(x,y) (((x)==0)?((y) + OFFSET_WT0):((y) + OFFSET_WT1))
+#define ADB_XTALKIN(x) ((x) + OFFSET_XTALKIN)
+#define ADB_XTALKOUT(x) ((x) + OFFSET_XTALKOUT)
+
+#define MIX_DEFIGAIN 0x08
+#define MIX_DEFOGAIN 0x08 /* 0x8->6dB (6dB = x4) 16 to 18 bit conversion? */
+
+/* MIXER */
+#define VORTEX_MIXER_SR 0x21f00
+#define VORTEX_MIXER_CLIP 0x21f80
+#define VORTEX_MIXER_CHNBASE 0x21e40
+#define VORTEX_MIXER_RTBASE 0x21e00
+#define MIXER_RTBASE_SIZE 0x38
+#define VORTEX_MIX_ENIN 0x21a00 /* Input enable bits. 4 bits wide. */
+#define VORTEX_MIX_SMP 0x21c00 /* wave data buffers. AU8820: 0x9c00 */
+
+/* MIX */
+#define VORTEX_MIX_INVOL_B 0x20000 /* Input volume current */
+#define VORTEX_MIX_VOL_B 0x20800 /* Output Volume current */
+#define VORTEX_MIX_INVOL_A 0x21000 /* Input Volume target */
+#define VORTEX_MIX_VOL_A 0x21800 /* Output Volume target */
+
+#define VOL_MIN 0x80 /* Input volume when muted. */
+#define VOL_MAX 0x7f /* FIXME: Not confirmed! Just guessed. */
+
+/* SRC */
+#define VORTEX_SRC_CHNBASE 0x26c40
+#define VORTEX_SRC_RTBASE 0x26c00
+#define VORTEX_SRCBLOCK_SR 0x26cc0
+#define VORTEX_SRC_SOURCE 0x26cc4
+#define VORTEX_SRC_SOURCESIZE 0x26cc8
+/* Params
+ 0x26e00 : 1 U0
+ 0x26e40 : 2 CR
+ 0x26e80 : 3 U3
+ 0x26ec0 : 4 DRIFT1
+ 0x26f00 : 5 U1
+ 0x26f40 : 6 DRIFT2
+ 0x26f80 : 7 U2 : Target rate, direction
+*/
+
+#define VORTEX_SRC_CONVRATIO 0x26e40
+#define VORTEX_SRC_DRIFT0 0x26e80
+#define VORTEX_SRC_DRIFT1 0x26ec0
+#define VORTEX_SRC_DRIFT2 0x26f40
+#define VORTEX_SRC_U0 0x26e00
+#define U0_SLOWLOCK 0x200
+#define VORTEX_SRC_U1 0x26f00
+#define VORTEX_SRC_U2 0x26f80
+#define VORTEX_SRC_DATA 0x26800 /* 0xc800 */
+#define VORTEX_SRC_DATA0 0x26000
+
+/* FIFO */
+#define VORTEX_FIFO_ADBCTRL 0x16100 /* Control bits. */
+#define VORTEX_FIFO_WTCTRL 0x16000
+#define FIFO_RDONLY 0x00000001
+#define FIFO_CTRL 0x00000002 /* Allow ctrl. ? */
+#define FIFO_VALID 0x00000010
+#define FIFO_EMPTY 0x00000020
+#define FIFO_U0 0x00002000 /* Unknown. */
+#define FIFO_U1 0x00040000
+#define FIFO_SIZE_BITS 6
+#define FIFO_SIZE (1<<(FIFO_SIZE_BITS)) // 0x40
+#define FIFO_MASK (FIFO_SIZE-1) //0x3f /* at shift left 0xc */
+#define FIFO_BITS 0x1c400000
+#define VORTEX_FIFO_ADBDATA 0x14000
+#define VORTEX_FIFO_WTDATA 0x10000
+
+#define VORTEX_FIFO_GIRT 0x17000 /* wt0, wt1, adb */
+#define GIRT_COUNT 3
+
+/* CODEC */
+
+#define VORTEX_CODEC_CHN 0x29080 /* The name "CHN" is wrong. */
+
+#define VORTEX_CODEC_CTRL 0x29184
+#define VORTEX_CODEC_IO 0x29188
+
+#define VORTEX_CODEC_SPORTCTRL 0x2918c
+
+#define VORTEX_CODEC_EN 0x29190
+#define EN_AUDIO0 0x00000300
+#define EN_MODEM 0x00000c00
+#define EN_AUDIO1 0x00003000
+#define EN_SPORT 0x00030000
+#define EN_SPDIF 0x000c0000
+#define EN_CODEC (EN_AUDIO1 | EN_AUDIO0)
+
+#define VORTEX_SPDIF_SMPRATE 0x29194
+
+#define VORTEX_SPDIF_FLAGS 0x2205c
+#define VORTEX_SPDIF_CFG0 0x291D0 /* status data */
+#define VORTEX_SPDIF_CFG1 0x291D4
+
+#define VORTEX_SMP_TIME 0x29198 /* Sample counter/timer */
+#define VORTEX_SMP_TIMER 0x2919c
+#define VORTEX_CODEC2_CTRL 0x291a0
+
+#define VORTEX_MODEM_CTRL 0x291ac
+
+/* IRQ */
+#define VORTEX_IRQ_SOURCE 0x2a000 /* Interrupt source flags. */
+#define VORTEX_IRQ_CTRL 0x2a004 /* Interrupt source mask. */
+
+//#define VORTEX_IRQ_U0 0x2a008 /* ?? */
+#define VORTEX_STAT 0x2a008 /* Some sort of status */
+#define STAT_IRQ 0x00000001 /* This bitis set if the IRQ is valid. */
+
+#define VORTEX_CTRL 0x2a00c
+#define CTRL_MIDI_EN 0x00000001
+#define CTRL_MIDI_PORT 0x00000060
+#define CTRL_GAME_EN 0x00000008
+#define CTRL_GAME_PORT 0x00000e00
+#define CTRL_IRQ_ENABLE 0x00004000
+#define CTRL_SPDIF 0x00000000 /* unknown. Please find this value */
+#define CTRL_SPORT 0x00200000
+#define CTRL_RST 0x00800000
+#define CTRL_UNKNOWN 0x01000000
+
+/* write: Timer period config / read: TIMER IRQ ack. */
+#define VORTEX_IRQ_STAT 0x2919c
+
+ /* MIDI *//* GAME. */
+#define VORTEX_MIDI_DATA 0x28800
+#define VORTEX_MIDI_CMD 0x28804 /* Write command / Read status */
+
+#define VORTEX_GAME_LEGACY 0x28808
+#define VORTEX_CTRL2 0x2880c
+#define CTRL2_GAME_ADCMODE 0x40
+#define VORTEX_GAME_AXIS 0x28810 /* Axis base register. 4 axis's */
+#define AXIS_SIZE 4
+#define AXIS_RANGE 0x1fff
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
new file mode 100644
index 0000000..a36d4d1
--- /dev/null
+++ b/sound/pci/au88x0/au88x0.c
@@ -0,0 +1,397 @@
+/*
+ * ALSA driver for the Aureal Vortex family of soundprocessors.
+ * Author: Manuel Jander (mjander@embedded.cl)
+ *
+ * This driver is the result of the OpenVortex Project from Savannah
+ * (savannah.nongnu.org/projects/openvortex). I would like to thank
+ * the developers of OpenVortex, Jeff Muizelaar and Kester Maddock, from
+ * whom i got plenty of help, and their codebase was invaluable.
+ * Thanks to the ALSA developers, they helped a lot working out
+ * the ALSA part.
+ * Thanks also to Sourceforge for maintaining the old binary drivers,
+ * and the forum, where developers could comunicate.
+ *
+ * Now at least i can play Legacy DOOM with MIDI music :-)
+ */
+
+#include "au88x0.h"
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
+#include <sound/initval.h>
+
+// module parameters (see "Module Parameters")
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+static int pcifix[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 255 };
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
+module_param_array(pcifix, int, NULL, 0444);
+MODULE_PARM_DESC(pcifix, "Enable VIA-workaround for " CARD_NAME " soundcard.");
+
+MODULE_DESCRIPTION("Aureal vortex");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Aureal Semiconductor Inc., Aureal Vortex Sound Processor}}");
+
+MODULE_DEVICE_TABLE(pci, snd_vortex_ids);
+
+static void vortex_fix_latency(struct pci_dev *vortex)
+{
+ int rc;
+ if (!(rc = pci_write_config_byte(vortex, 0x40, 0xff))) {
+ printk(KERN_INFO CARD_NAME
+ ": vortex latency is 0xff\n");
+ } else {
+ printk(KERN_WARNING CARD_NAME
+ ": could not set vortex latency: pci error 0x%x\n", rc);
+ }
+}
+
+static void vortex_fix_agp_bridge(struct pci_dev *via)
+{
+ int rc;
+ u8 value;
+
+ /*
+ * only set the bit (Extend PCI#2 Internal Master for
+ * Efficient Handling of Dummy Requests) if the can
+ * read the config and it is not already set
+ */
+
+ if (!(rc = pci_read_config_byte(via, 0x42, &value))
+ && ((value & 0x10)
+ || !(rc = pci_write_config_byte(via, 0x42, value | 0x10)))) {
+ printk(KERN_INFO CARD_NAME
+ ": bridge config is 0x%x\n", value | 0x10);
+ } else {
+ printk(KERN_WARNING CARD_NAME
+ ": could not set vortex latency: pci error 0x%x\n", rc);
+ }
+}
+
+static void __devinit snd_vortex_workaround(struct pci_dev *vortex, int fix)
+{
+ struct pci_dev *via = NULL;
+
+ /* autodetect if workarounds are required */
+ if (fix == 255) {
+ /* VIA KT133 */
+ via = pci_get_device(PCI_VENDOR_ID_VIA,
+ PCI_DEVICE_ID_VIA_8365_1, NULL);
+ /* VIA Apollo */
+ if (via == NULL) {
+ via = pci_get_device(PCI_VENDOR_ID_VIA,
+ PCI_DEVICE_ID_VIA_82C598_1, NULL);
+ /* AMD Irongate */
+ if (via == NULL)
+ via = pci_get_device(PCI_VENDOR_ID_AMD,
+ PCI_DEVICE_ID_AMD_FE_GATE_7007, NULL);
+ }
+ if (via) {
+ printk(KERN_INFO CARD_NAME ": Activating latency workaround...\n");
+ vortex_fix_latency(vortex);
+ vortex_fix_agp_bridge(via);
+ }
+ } else {
+ if (fix & 0x1)
+ vortex_fix_latency(vortex);
+ if ((fix & 0x2) && (via = pci_get_device(PCI_VENDOR_ID_VIA,
+ PCI_DEVICE_ID_VIA_8365_1, NULL)))
+ vortex_fix_agp_bridge(via);
+ if ((fix & 0x4) && (via = pci_get_device(PCI_VENDOR_ID_VIA,
+ PCI_DEVICE_ID_VIA_82C598_1, NULL)))
+ vortex_fix_agp_bridge(via);
+ if ((fix & 0x8) && (via = pci_get_device(PCI_VENDOR_ID_AMD,
+ PCI_DEVICE_ID_AMD_FE_GATE_7007, NULL)))
+ vortex_fix_agp_bridge(via);
+ }
+ pci_dev_put(via);
+}
+
+// component-destructor
+// (see "Management of Cards and Components")
+static int snd_vortex_dev_free(struct snd_device *device)
+{
+ vortex_t *vortex = device->device_data;
+
+ vortex_gameport_unregister(vortex);
+ vortex_core_shutdown(vortex);
+ // Take down PCI interface.
+ free_irq(vortex->irq, vortex);
+ iounmap(vortex->mmio);
+ pci_release_regions(vortex->pci_dev);
+ pci_disable_device(vortex->pci_dev);
+ kfree(vortex);
+
+ return 0;
+}
+
+// chip-specific constructor
+// (see "Management of Cards and Components")
+static int __devinit
+snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
+{
+ vortex_t *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_vortex_dev_free,
+ };
+
+ *rchip = NULL;
+
+ // check PCI availability (DMA).
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
+ printk(KERN_ERR "error to set DMA mask\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+
+ // initialize the stuff
+ chip->pci_dev = pci;
+ chip->io = pci_resource_start(pci, 0);
+ chip->vendor = pci->vendor;
+ chip->device = pci->device;
+ chip->card = card;
+ chip->irq = -1;
+
+ // (1) PCI resource allocation
+ // Get MMIO area
+ //
+ if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0)
+ goto regions_out;
+
+ chip->mmio = pci_ioremap_bar(pci, 0);
+ if (!chip->mmio) {
+ printk(KERN_ERR "MMIO area remap failed.\n");
+ err = -ENOMEM;
+ goto ioremap_out;
+ }
+
+ /* Init audio core.
+ * This must be done before we do request_irq otherwise we can get spurious
+ * interrupts that we do not handle properly and make a mess of things */
+ if ((err = vortex_core_init(chip)) != 0) {
+ printk(KERN_ERR "hw core init failed\n");
+ goto core_out;
+ }
+
+ if ((err = request_irq(pci->irq, vortex_interrupt,
+ IRQF_SHARED, CARD_NAME_SHORT,
+ chip)) != 0) {
+ printk(KERN_ERR "cannot grab irq\n");
+ goto irq_out;
+ }
+ chip->irq = pci->irq;
+
+ pci_set_master(pci);
+ // End of PCI setup.
+
+ // Register alsa root device.
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ goto alloc_out;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+
+ return 0;
+
+ alloc_out:
+ free_irq(chip->irq, chip);
+ irq_out:
+ vortex_core_shutdown(chip);
+ core_out:
+ iounmap(chip->mmio);
+ ioremap_out:
+ pci_release_regions(chip->pci_dev);
+ regions_out:
+ pci_disable_device(chip->pci_dev);
+ //FIXME: this not the right place to unregister the gameport
+ vortex_gameport_unregister(chip);
+ kfree(chip);
+ return err;
+}
+
+// constructor -- see "Constructor" sub-section
+static int __devinit
+snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ vortex_t *chip;
+ int err;
+
+ // (1)
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+ // (2)
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ // (3)
+ if ((err = snd_vortex_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_vortex_workaround(pci, pcifix[dev]);
+
+ // Card details needed in snd_vortex_midi
+ strcpy(card->driver, CARD_NAME_SHORT);
+ sprintf(card->shortname, "Aureal Vortex %s", CARD_NAME_SHORT);
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->io, chip->irq);
+
+ // (4) Alloc components.
+ // ADB pcm.
+ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#ifndef CHIP_AU8820
+ // ADB SPDIF
+ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_SPDIF, 1)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ // A3D
+ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_A3D, NR_A3D)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#endif
+ /*
+ // ADB I2S
+ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_I2S, 1)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ */
+#ifndef CHIP_AU8810
+ // WT pcm.
+ if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_WT, NR_WT)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#endif
+ // snd_ac97_mixer and Vortex mixer.
+ if ((err = snd_vortex_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_vortex_midi(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ vortex_gameport_register(chip);
+
+#if 0
+ if (snd_seq_device_new(card, 1, SNDRV_SEQ_DEV_ID_VORTEX_SYNTH,
+ sizeof(snd_vortex_synth_arg_t), &wave) < 0
+ || wave == NULL) {
+ snd_printk(KERN_ERR "Can't initialize Aureal wavetable synth\n");
+ } else {
+ snd_vortex_synth_arg_t *arg;
+
+ arg = SNDRV_SEQ_DEVICE_ARGPTR(wave);
+ strcpy(wave->name, "Aureal Synth");
+ arg->hwptr = vortex;
+ arg->index = 1;
+ arg->seq_ports = seq_ports[dev];
+ arg->max_voices = max_synth_voices[dev];
+ }
+#endif
+
+ // (5)
+ if ((err = pci_read_config_word(pci, PCI_DEVICE_ID,
+ &(chip->device))) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = pci_read_config_word(pci, PCI_VENDOR_ID,
+ &(chip->vendor))) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ chip->rev = pci->revision;
+#ifdef CHIP_AU8830
+ if ((chip->rev) != 0xfe && (chip->rev) != 0xfa) {
+ printk(KERN_ALERT
+ "vortex: The revision (%x) of your card has not been seen before.\n",
+ chip->rev);
+ printk(KERN_ALERT
+ "vortex: Please email the results of 'lspci -vv' to openvortex-dev@nongnu.org.\n");
+ snd_card_free(card);
+ err = -ENODEV;
+ return err;
+ }
+#endif
+
+ // (6)
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ // (7)
+ pci_set_drvdata(pci, card);
+ dev++;
+ vortex_connect_default(chip, 1);
+ vortex_enable_int(chip);
+ return 0;
+}
+
+// destructor -- see "Destructor" sub-section
+static void __devexit snd_vortex_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+// pci_driver definition
+static struct pci_driver driver = {
+ .name = CARD_NAME_SHORT,
+ .id_table = snd_vortex_ids,
+ .probe = snd_vortex_probe,
+ .remove = __devexit_p(snd_vortex_remove),
+};
+
+// initialization of the module
+static int __init alsa_card_vortex_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+// clean up the module
+static void __exit alsa_card_vortex_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_vortex_init)
+module_exit(alsa_card_vortex_exit)
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
new file mode 100644
index 0000000..cf46bba
--- /dev/null
+++ b/sound/pci/au88x0/au88x0.h
@@ -0,0 +1,285 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __SOUND_AU88X0_H
+#define __SOUND_AU88X0_H
+
+#ifdef __KERNEL__
+#include <linux/pci.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
+#include <sound/hwdep.h>
+#include <sound/ac97_codec.h>
+
+#endif
+
+#ifndef CHIP_AU8820
+#include "au88x0_eq.h"
+#include "au88x0_a3d.h"
+#endif
+#ifndef CHIP_AU8810
+#include "au88x0_wt.h"
+#endif
+
+#define hwread(x,y) readl((x)+(y))
+#define hwwrite(x,y,z) writel((z),(x)+(y))
+
+/* Vortex MPU401 defines. */
+#define MIDI_CLOCK_DIV 0x61
+/* Standart MPU401 defines. */
+#define MPU401_RESET 0xff
+#define MPU401_ENTER_UART 0x3f
+#define MPU401_ACK 0xfe
+
+// Get src register value to convert from x to y.
+#define SRC_RATIO(x,y) ((((x<<15)/y) + 1)/2)
+
+/* FIFO software state constants. */
+#define FIFO_STOP 0
+#define FIFO_START 1
+#define FIFO_PAUSE 2
+
+/* IRQ flags */
+#define IRQ_ERR_MASK 0x00ff
+#define IRQ_FATAL 0x0001
+#define IRQ_PARITY 0x0002
+#define IRQ_REG 0x0004
+#define IRQ_FIFO 0x0008
+#define IRQ_DMA 0x0010
+#define IRQ_PCMOUT 0x0020 /* PCM OUT page crossing */
+#define IRQ_TIMER 0x1000
+#define IRQ_MIDI 0x2000
+#define IRQ_MODEM 0x4000
+
+/* ADB Resource */
+#define VORTEX_RESOURCE_DMA 0x00000000
+#define VORTEX_RESOURCE_SRC 0x00000001
+#define VORTEX_RESOURCE_MIXIN 0x00000002
+#define VORTEX_RESOURCE_MIXOUT 0x00000003
+#define VORTEX_RESOURCE_A3D 0x00000004
+#define VORTEX_RESOURCE_LAST 0x00000005
+
+/* codec io: VORTEX_CODEC_IO bits */
+#define VORTEX_CODEC_ID_SHIFT 24
+#define VORTEX_CODEC_WRITE 0x00800000
+#define VORTEX_CODEC_ADDSHIFT 16
+#define VORTEX_CODEC_ADDMASK 0x7f0000
+#define VORTEX_CODEC_DATSHIFT 0
+#define VORTEX_CODEC_DATMASK 0xffff
+
+/* Check for SDAC bit in "Extended audio ID" AC97 register */
+//#define VORTEX_IS_QUAD(x) (((x)->codec == NULL) ? 0 : ((x)->codec->ext_id&0x80))
+#define VORTEX_IS_QUAD(x) ((x)->isquad)
+/* Check if chip has bug. */
+#define IS_BAD_CHIP(x) (\
+ (x->rev == 0xfe && x->device == PCI_DEVICE_ID_AUREAL_VORTEX_2) || \
+ (x->rev == 0xfe && x->device == PCI_DEVICE_ID_AUREAL_ADVANTAGE))
+
+
+/* PCM devices */
+#define VORTEX_PCM_ADB 0
+#define VORTEX_PCM_SPDIF 1
+#define VORTEX_PCM_A3D 2
+#define VORTEX_PCM_WT 3
+#define VORTEX_PCM_I2S 4
+#define VORTEX_PCM_LAST 5
+
+#define MIX_CAPT(x) (vortex->mixcapt[x])
+#define MIX_PLAYB(x) (vortex->mixplayb[x])
+#define MIX_SPDIF(x) (vortex->mixspdif[x])
+
+#define NR_WTPB 0x20 /* WT channels per eahc bank. */
+
+/* Structs */
+typedef struct {
+ //int this_08; /* Still unknown */
+ int fifo_enabled; /* this_24 */
+ int fifo_status; /* this_1c */
+ u32 dma_ctrl; /* this_78 (ADB), this_7c (WT) */
+ int dma_unknown; /* this_74 (ADB), this_78 (WT). WDM: +8 */
+ int cfg0;
+ int cfg1;
+
+ int nr_ch; /* Nr of PCM channels in use */
+ int type; /* Output type (ac97, a3d, spdif, i2s, dsp) */
+ int dma; /* Hardware DMA index. */
+ int dir; /* Stream Direction. */
+ u32 resources[5];
+
+ /* Virtual page extender stuff */
+ int nr_periods;
+ int period_bytes;
+ int period_real;
+ int period_virt;
+
+ struct snd_pcm_substream *substream;
+} stream_t;
+
+typedef struct snd_vortex vortex_t;
+struct snd_vortex {
+ /* ALSA structs. */
+ struct snd_card *card;
+ struct snd_pcm *pcm[VORTEX_PCM_LAST];
+
+ struct snd_rawmidi *rmidi; /* Legacy Midi interface. */
+ struct snd_ac97 *codec;
+
+ /* Stream structs. */
+ stream_t dma_adb[NR_ADB];
+ int spdif_sr;
+#ifndef CHIP_AU8810
+ stream_t dma_wt[NR_WT];
+ wt_voice_t wt_voice[NR_WT]; /* WT register cache. */
+ char mixwt[(NR_WT / NR_WTPB) * 6]; /* WT mixin objects */
+#endif
+
+ /* Global resources */
+ s8 mixcapt[2];
+ s8 mixplayb[4];
+#ifndef CHIP_AU8820
+ s8 mixspdif[2];
+ s8 mixa3d[2]; /* mixers which collect all a3d streams. */
+ s8 mixxtlk[2]; /* crosstalk canceler mixer inputs. */
+#endif
+ u32 fixed_res[5];
+
+#ifndef CHIP_AU8820
+ /* Hardware equalizer structs */
+ eqlzr_t eq;
+ /* A3D structs */
+ a3dsrc_t a3d[NR_A3D];
+ /* Xtalk canceler */
+ int xt_mode; /* 1: speakers, 0:headphones. */
+#endif
+
+ int isquad; /* cache of extended ID codec flag. */
+
+ /* Gameport stuff. */
+ struct gameport *gameport;
+
+ /* PCI hardware resources */
+ unsigned long io;
+ void __iomem *mmio;
+ unsigned int irq;
+ spinlock_t lock;
+
+ /* PCI device */
+ struct pci_dev *pci_dev;
+ u16 vendor;
+ u16 device;
+ u8 rev;
+};
+
+/* Functions. */
+
+/* SRC */
+static void vortex_adb_setsrc(vortex_t * vortex, int adbdma,
+ unsigned int cvrt, int dir);
+
+/* DMA Engines. */
+static void vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
+ int size, int count);
+static void vortex_adbdma_setmode(vortex_t * vortex, int adbdma, int ie,
+ int dir, int fmt, int d,
+ u32 offset);
+static void vortex_adbdma_setstartbuffer(vortex_t * vortex, int adbdma, int sb);
+#ifndef CHIP_AU8810
+static void vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma,
+ int size, int count);
+static void vortex_wtdma_setmode(vortex_t * vortex, int wtdma, int ie, int fmt, int d, /*int e, */
+ u32 offset);
+static void vortex_wtdma_setstartbuffer(vortex_t * vortex, int wtdma, int sb);
+#endif
+
+static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma);
+//static void vortex_adbdma_stopfifo(vortex_t *vortex, int adbdma);
+static void vortex_adbdma_pausefifo(vortex_t * vortex, int adbdma);
+static void vortex_adbdma_resumefifo(vortex_t * vortex, int adbdma);
+static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma);
+static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma);
+
+#ifndef CHIP_AU8810
+static void vortex_wtdma_startfifo(vortex_t * vortex, int wtdma);
+static void vortex_wtdma_stopfifo(vortex_t * vortex, int wtdma);
+static void vortex_wtdma_pausefifo(vortex_t * vortex, int wtdma);
+static void vortex_wtdma_resumefifo(vortex_t * vortex, int wtdma);
+static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma);
+#endif
+
+/* global stuff. */
+static void vortex_codec_init(vortex_t * vortex);
+static void vortex_codec_write(struct snd_ac97 * codec, unsigned short addr,
+ unsigned short data);
+static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short addr);
+static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode);
+
+static int vortex_core_init(vortex_t * card);
+static int vortex_core_shutdown(vortex_t * card);
+static void vortex_enable_int(vortex_t * card);
+static irqreturn_t vortex_interrupt(int irq, void *dev_id);
+static int vortex_alsafmt_aspfmt(int alsafmt);
+
+/* Connection stuff. */
+static void vortex_connect_default(vortex_t * vortex, int en);
+static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch,
+ int dir, int type);
+static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out,
+ int restype);
+#ifndef CHIP_AU8810
+static int vortex_wt_allocroute(vortex_t * vortex, int dma, int nr_ch);
+static void vortex_wt_connect(vortex_t * vortex, int en);
+static void vortex_wt_init(vortex_t * vortex);
+#endif
+
+static void vortex_route(vortex_t * vortex, int en, unsigned char channel,
+ unsigned char source, unsigned char dest);
+#if 0
+static void vortex_routes(vortex_t * vortex, int en, unsigned char channel,
+ unsigned char source, unsigned char dest0,
+ unsigned char dest1);
+#endif
+static void vortex_connection_mixin_mix(vortex_t * vortex, int en,
+ unsigned char mixin,
+ unsigned char mix, int a);
+static void vortex_mix_setinputvolumebyte(vortex_t * vortex,
+ unsigned char mix, int mixin,
+ unsigned char vol);
+static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix,
+ unsigned char vol);
+
+/* A3D functions. */
+#ifndef CHIP_AU8820
+static void vortex_Vort3D_enable(vortex_t * v);
+static void vortex_Vort3D_disable(vortex_t * v);
+static void vortex_Vort3D_connect(vortex_t * vortex, int en);
+static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en);
+#endif
+
+/* Driver stuff. */
+static int vortex_gameport_register(vortex_t * card);
+static void vortex_gameport_unregister(vortex_t * card);
+#ifndef CHIP_AU8820
+static int vortex_eq_init(vortex_t * vortex);
+static int vortex_eq_free(vortex_t * vortex);
+#endif
+/* ALSA stuff. */
+static int snd_vortex_new_pcm(vortex_t * vortex, int idx, int nr);
+static int snd_vortex_mixer(vortex_t * vortex);
+static int snd_vortex_midi(vortex_t * vortex);
+#endif
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
new file mode 100644
index 0000000..649849e
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -0,0 +1,913 @@
+/***************************************************************************
+ * au88x0_a3d.c
+ *
+ * Fri Jul 18 14:16:22 2003
+ * Copyright 2003 mjander
+ * mjander@users.sourceforge.net
+ *
+ * A3D. You may think i'm crazy, but this may work someday. Who knows...
+ ****************************************************************************/
+
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#include "au88x0_a3d.h"
+#include "au88x0_a3ddata.c"
+#include "au88x0_xtalk.h"
+#include "au88x0.h"
+
+static void
+a3dsrc_SetTimeConsts(a3dsrc_t * a, short HrtfTrack, short ItdTrack,
+ short GTrack, short CTrack)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_HrtfTrackTC), HrtfTrack);
+ hwwrite(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_ITDTrackTC), ItdTrack);
+ hwwrite(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_GainTrackTC), GTrack);
+ hwwrite(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_CoeffTrackTC), CTrack);
+}
+
+#if 0
+static void
+a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack,
+ short *GTrack, short *CTrack)
+{
+ // stub!
+}
+
+#endif
+/* Atmospheric absorbtion. */
+
+static void
+a3dsrc_SetAtmosTarget(a3dsrc_t * a, short aa, short b, short c, short d,
+ short e)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_A21Target),
+ (e << 0x10) | d);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_B10Target),
+ (b << 0x10) | aa);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_B2Target), c);
+}
+
+static void
+a3dsrc_SetAtmosCurrent(a3dsrc_t * a, short aa, short b, short c, short d,
+ short e)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_A12Current),
+ (e << 0x10) | d);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_B01Current),
+ (b << 0x10) | aa);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_B2Current), c);
+}
+
+static void
+a3dsrc_SetAtmosState(a3dsrc_t * a, short x1, short x2, short y1, short y2)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio, a3d_addrA(a->slice, a->source, A3D_A_x1), x1);
+ hwwrite(vortex->mmio, a3d_addrA(a->slice, a->source, A3D_A_x2), x2);
+ hwwrite(vortex->mmio, a3d_addrA(a->slice, a->source, A3D_A_y1), y1);
+ hwwrite(vortex->mmio, a3d_addrA(a->slice, a->source, A3D_A_y2), y2);
+}
+
+#if 0
+static void
+a3dsrc_GetAtmosTarget(a3dsrc_t * a, short *aa, short *b, short *c,
+ short *d, short *e)
+{
+}
+static void
+a3dsrc_GetAtmosCurrent(a3dsrc_t * a, short *bb01, short *ab01, short *b2,
+ short *aa12, short *ba12)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *aa12 =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_A12Current));
+ *ba12 =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_A12Current));
+ *ab01 =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_B01Current));
+ *bb01 =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_B01Current));
+ *b2 =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_B2Current));
+}
+
+static void
+a3dsrc_GetAtmosState(a3dsrc_t * a, short *x1, short *x2, short *y1, short *y2)
+{
+
+}
+
+#endif
+/* HRTF */
+
+static void
+a3dsrc_SetHrtfTarget(a3dsrc_t * a, a3d_Hrtf_t const aa, a3d_Hrtf_t const b)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+
+ for (i = 0; i < HRTF_SZ; i++)
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source,
+ A3D_B_HrtfTarget) + (i << 2),
+ (b[i] << 0x10) | aa[i]);
+}
+
+static void
+a3dsrc_SetHrtfCurrent(a3dsrc_t * a, a3d_Hrtf_t const aa, a3d_Hrtf_t const b)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+
+ for (i = 0; i < HRTF_SZ; i++)
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source,
+ A3D_B_HrtfCurrent) + (i << 2),
+ (b[i] << 0x10) | aa[i]);
+}
+
+static void
+a3dsrc_SetHrtfState(a3dsrc_t * a, a3d_Hrtf_t const aa, a3d_Hrtf_t const b)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+
+ for (i = 0; i < HRTF_SZ; i++)
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source,
+ A3D_B_HrtfDelayLine) + (i << 2),
+ (b[i] << 0x10) | aa[i]);
+}
+
+static void a3dsrc_SetHrtfOutput(a3dsrc_t * a, short left, short right)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_HrtfOutL), left);
+ hwwrite(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_HrtfOutR), right);
+}
+
+#if 0
+static void a3dsrc_GetHrtfTarget(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+
+ for (i = 0; i < HRTF_SZ; i++)
+ aa[i] =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source,
+ A3D_A_HrtfTarget + (i << 2)));
+ for (i = 0; i < HRTF_SZ; i++)
+ b[i] =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source,
+ A3D_B_HrtfTarget + (i << 2)));
+}
+
+static void a3dsrc_GetHrtfCurrent(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+
+ for (i = 0; i < HRTF_SZ; i++)
+ aa[i] =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source,
+ A3D_A_HrtfCurrent + (i << 2)));
+ for (i = 0; i < HRTF_SZ; i++)
+ b[i] =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source,
+ A3D_B_HrtfCurrent + (i << 2)));
+}
+
+static void a3dsrc_GetHrtfState(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+ // FIXME: verify this!
+ for (i = 0; i < HRTF_SZ; i++)
+ aa[i] =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source,
+ A3D_A_HrtfDelayLine + (i << 2)));
+ for (i = 0; i < HRTF_SZ; i++)
+ b[i] =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source,
+ A3D_B_HrtfDelayLine + (i << 2)));
+}
+
+static void a3dsrc_GetHrtfOutput(a3dsrc_t * a, short *left, short *right)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *left =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_HrtfOutL));
+ *right =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_HrtfOutR));
+}
+
+#endif
+
+/* Interaural Time Difference.
+ * "The other main clue that humans use to locate sounds, is called
+ * Interaural Time Difference (ITD). The differences in distance from
+ * the sound source to a listeners ears means that the sound will
+ * reach one ear slightly before the other....", found somewhere with google.*/
+static void a3dsrc_SetItdTarget(a3dsrc_t * a, short litd, short ritd)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+
+ if (litd < 0)
+ litd = 0;
+ if (litd > 0x57FF)
+ litd = 0x57FF;
+ if (ritd < 0)
+ ritd = 0;
+ if (ritd > 0x57FF)
+ ritd = 0x57FF;
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_ITDTarget),
+ (ritd << 0x10) | litd);
+ //hwwrite(vortex->mmio, addr(0x191DF+5, this04, this08), (ritd<<0x10)|litd);
+}
+
+static void a3dsrc_SetItdCurrent(a3dsrc_t * a, short litd, short ritd)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+
+ if (litd < 0)
+ litd = 0;
+ if (litd > 0x57FF)
+ litd = 0x57FF;
+ if (ritd < 0)
+ ritd = 0;
+ if (ritd > 0x57FF)
+ ritd = 0x57FF;
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_ITDCurrent),
+ (ritd << 0x10) | litd);
+ //hwwrite(vortex->mmio, addr(0x191DF+1, this04, this08), (ritd<<0x10)|litd);
+}
+
+static void a3dsrc_SetItdDline(a3dsrc_t * a, a3d_ItdDline_t const dline)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+ /* 45 != 40 -> Check this ! */
+ for (i = 0; i < DLINE_SZ; i++)
+ hwwrite(vortex->mmio,
+ a3d_addrA(a->slice, a->source,
+ A3D_A_ITDDelayLine) + (i << 2), dline[i]);
+}
+
+#if 0
+static void a3dsrc_GetItdTarget(a3dsrc_t * a, short *litd, short *ritd)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *ritd =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_ITDTarget));
+ *litd =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_ITDTarget));
+}
+
+static void a3dsrc_GetItdCurrent(a3dsrc_t * a, short *litd, short *ritd)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+
+ *ritd =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_ITDCurrent));
+ *litd =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_ITDCurrent));
+}
+
+static void a3dsrc_GetItdDline(a3dsrc_t * a, a3d_ItdDline_t dline)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+
+ for (i = 0; i < DLINE_SZ; i++)
+ dline[i] =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source,
+ A3D_A_ITDDelayLine + (i << 2)));
+}
+
+#endif
+/* This is may be used for ILD Interaural Level Difference. */
+
+static void a3dsrc_SetGainTarget(a3dsrc_t * a, short left, short right)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_GainTarget),
+ (right << 0x10) | left);
+}
+
+static void a3dsrc_SetGainCurrent(a3dsrc_t * a, short left, short right)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_GainCurrent),
+ (right << 0x10) | left);
+}
+
+#if 0
+static void a3dsrc_GetGainTarget(a3dsrc_t * a, short *left, short *right)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *right =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_GainTarget));
+ *left =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_GainTarget));
+}
+
+static void a3dsrc_GetGainCurrent(a3dsrc_t * a, short *left, short *right)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *right =
+ hwread(vortex->mmio,
+ a3d_addrA(a->slice, a->source, A3D_A_GainCurrent));
+ *left =
+ hwread(vortex->mmio,
+ a3d_addrB(a->slice, a->source, A3D_B_GainCurrent));
+}
+
+/* CA3dIO this func seems to be inlined all over this place. */
+static void CA3dIO_WriteReg(a3dsrc_t * a, unsigned long addr, short aa, short b)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio, addr, (aa << 0x10) | b);
+}
+
+#endif
+/* Generic A3D stuff */
+
+static void a3dsrc_SetA3DSampleRate(a3dsrc_t * a, int sr)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int esp0 = 0;
+
+ esp0 = (((esp0 & 0x7fffffff) | 0xB8000000) & 0x7) | ((sr & 0x1f) << 3);
+ hwwrite(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd), esp0);
+ //hwwrite(vortex->mmio, 0x19C38 + (this08<<0xd), esp0);
+}
+
+static void a3dsrc_EnableA3D(a3dsrc_t * a)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd),
+ 0xF0000001);
+ //hwwrite(vortex->mmio, 0x19C38 + (this08<<0xd), 0xF0000001);
+}
+
+static void a3dsrc_DisableA3D(a3dsrc_t * a)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd),
+ 0xF0000000);
+}
+
+static void a3dsrc_SetA3DControlReg(a3dsrc_t * a, unsigned long ctrl)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd), ctrl);
+}
+
+static void a3dsrc_SetA3DPointerReg(a3dsrc_t * a, unsigned long ptr)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ hwwrite(vortex->mmio, A3D_SLICE_Pointers + ((a->slice) << 0xd), ptr);
+}
+
+#if 0
+static void a3dsrc_GetA3DSampleRate(a3dsrc_t * a, int *sr)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *sr = ((hwread(vortex->mmio, A3D_SLICE_Control + (a->slice << 0xd))
+ >> 3) & 0x1f);
+ //*sr = ((hwread(vortex->mmio, 0x19C38 + (this08<<0xd))>>3)&0x1f);
+}
+
+static void a3dsrc_GetA3DControlReg(a3dsrc_t * a, unsigned long *ctrl)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *ctrl = hwread(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd));
+}
+
+static void a3dsrc_GetA3DPointerReg(a3dsrc_t * a, unsigned long *ptr)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ *ptr = hwread(vortex->mmio, A3D_SLICE_Pointers + ((a->slice) << 0xd));
+}
+
+#endif
+static void a3dsrc_ZeroSliceIO(a3dsrc_t * a)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+ int i;
+
+ for (i = 0; i < 8; i++)
+ hwwrite(vortex->mmio,
+ A3D_SLICE_VDBDest +
+ ((((a->slice) << 0xb) + i) << 2), 0);
+ for (i = 0; i < 4; i++)
+ hwwrite(vortex->mmio,
+ A3D_SLICE_VDBSource +
+ ((((a->slice) << 0xb) + i) << 2), 0);
+}
+
+/* Reset Single A3D source. */
+static void a3dsrc_ZeroState(a3dsrc_t * a)
+{
+
+ //printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source);
+
+ a3dsrc_SetAtmosState(a, 0, 0, 0, 0);
+ a3dsrc_SetHrtfState(a, A3dHrirZeros, A3dHrirZeros);
+ a3dsrc_SetItdDline(a, A3dItdDlineZeros);
+ a3dsrc_SetHrtfOutput(a, 0, 0);
+ a3dsrc_SetTimeConsts(a, 0, 0, 0, 0);
+
+ a3dsrc_SetAtmosCurrent(a, 0, 0, 0, 0, 0);
+ a3dsrc_SetAtmosTarget(a, 0, 0, 0, 0, 0);
+ a3dsrc_SetItdCurrent(a, 0, 0);
+ a3dsrc_SetItdTarget(a, 0, 0);
+ a3dsrc_SetGainCurrent(a, 0, 0);
+ a3dsrc_SetGainTarget(a, 0, 0);
+
+ a3dsrc_SetHrtfCurrent(a, A3dHrirZeros, A3dHrirZeros);
+ a3dsrc_SetHrtfTarget(a, A3dHrirZeros, A3dHrirZeros);
+}
+
+/* Reset entire A3D engine */
+static void a3dsrc_ZeroStateA3D(a3dsrc_t * a)
+{
+ int i, var, var2;
+
+ if ((a->vortex) == NULL) {
+ printk(KERN_ERR "vortex: ZeroStateA3D: ERROR: a->vortex is NULL\n");
+ return;
+ }
+
+ a3dsrc_SetA3DControlReg(a, 0);
+ a3dsrc_SetA3DPointerReg(a, 0);
+
+ var = a->slice;
+ var2 = a->source;
+ for (i = 0; i < 4; i++) {
+ a->slice = i;
+ a3dsrc_ZeroSliceIO(a);
+ //a3dsrc_ZeroState(a);
+ }
+ a->source = var2;
+ a->slice = var;
+}
+
+/* Program A3D block as pass through */
+static void a3dsrc_ProgramPipe(a3dsrc_t * a)
+{
+ a3dsrc_SetTimeConsts(a, 0, 0, 0, 0);
+ a3dsrc_SetAtmosCurrent(a, 0, 0x4000, 0, 0, 0);
+ a3dsrc_SetAtmosTarget(a, 0x4000, 0, 0, 0, 0);
+ a3dsrc_SetItdCurrent(a, 0, 0);
+ a3dsrc_SetItdTarget(a, 0, 0);
+ a3dsrc_SetGainCurrent(a, 0x7fff, 0x7fff);
+ a3dsrc_SetGainTarget(a, 0x7fff, 0x7fff);
+
+ /* SET HRTF HERE */
+
+ /* Single spike leads to identity transfer function. */
+ a3dsrc_SetHrtfCurrent(a, A3dHrirImpulse, A3dHrirImpulse);
+ a3dsrc_SetHrtfTarget(a, A3dHrirImpulse, A3dHrirImpulse);
+
+ /* Test: Sounds saturated. */
+ //a3dsrc_SetHrtfCurrent(a, A3dHrirSatTest, A3dHrirSatTest);
+ //a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest);
+}
+
+/* VDB = Vortex audio Dataflow Bus */
+#if 0
+static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa)
+{
+ vortex_t *vortex = (vortex_t *) (a->vortex);
+
+ // ((aa >> 2) << 8) - (aa >> 2)
+ hwwrite(vortex->mmio,
+ a3d_addrS(a->slice, A3D_SLICE_VDBDest) + (a->source << 2), 0);
+ hwwrite(vortex->mmio,
+ a3d_addrS(a->slice,
+ A3D_SLICE_VDBDest + 4) + (a->source << 2), 0);
+ /*
+ hwwrite(vortex->mmio, 0x19c00 + (((aa>>2)*255*4)+aa)*8, 0);
+ hwwrite(vortex->mmio, 0x19c04 + (((aa>>2)*255*4)+aa)*8, 0);
+ */
+}
+#endif
+
+/* A3D HwSource stuff. */
+
+static void vortex_A3dSourceHw_Initialize(vortex_t * v, int source, int slice)
+{
+ a3dsrc_t *a3dsrc = &(v->a3d[source + (slice * 4)]);
+ //a3dsrc_t *a3dsrc = &(v->a3d[source + (slice*4)]);
+
+ a3dsrc->vortex = (void *)v;
+ a3dsrc->source = source; /* source */
+ a3dsrc->slice = slice; /* slice */
+ a3dsrc_ZeroState(a3dsrc);
+ /* Added by me. */
+ a3dsrc_SetA3DSampleRate(a3dsrc, 0x11);
+}
+
+static int Vort3DRend_Initialize(vortex_t * v, unsigned short mode)
+{
+ v->xt_mode = mode; /* this_14 */
+
+ vortex_XtalkHw_init(v);
+ vortex_XtalkHw_SetGainsAllChan(v);
+ switch (v->xt_mode) {
+ case XT_SPEAKER0:
+ vortex_XtalkHw_ProgramXtalkNarrow(v);
+ break;
+ case XT_SPEAKER1:
+ vortex_XtalkHw_ProgramXtalkWide(v);
+ break;
+ default:
+ case XT_HEADPHONE:
+ vortex_XtalkHw_ProgramPipe(v);
+ break;
+ case XT_DIAMOND:
+ vortex_XtalkHw_ProgramDiamondXtalk(v);
+ break;
+ }
+ vortex_XtalkHw_SetSampleRate(v, 0x11);
+ vortex_XtalkHw_Enable(v);
+ return 0;
+}
+
+/* 3D Sound entry points. */
+
+static int vortex_a3d_register_controls(vortex_t * vortex);
+static void vortex_a3d_unregister_controls(vortex_t * vortex);
+/* A3D base support init/shudown */
+static void __devinit vortex_Vort3D_enable(vortex_t * v)
+{
+ int i;
+
+ Vort3DRend_Initialize(v, XT_HEADPHONE);
+ for (i = 0; i < NR_A3D; i++) {
+ vortex_A3dSourceHw_Initialize(v, i % 4, i >> 2);
+ a3dsrc_ZeroStateA3D(&(v->a3d[0]));
+ }
+ /* Register ALSA controls */
+ vortex_a3d_register_controls(v);
+}
+
+static void vortex_Vort3D_disable(vortex_t * v)
+{
+ vortex_XtalkHw_Disable(v);
+ vortex_a3d_unregister_controls(v);
+}
+
+/* Make A3D subsystem connections. */
+static void vortex_Vort3D_connect(vortex_t * v, int en)
+{
+ int i;
+
+// Disable AU8810 routes, since they seem to be wrong (in au8810.h).
+#ifdef CHIP_AU8810
+ return;
+#endif
+
+#if 1
+ /* Alloc Xtalk mixin resources */
+ v->mixxtlk[0] =
+ vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN);
+ if (v->mixxtlk[0] < 0) {
+ printk
+ ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n");
+ return;
+ }
+ v->mixxtlk[1] =
+ vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN);
+ if (v->mixxtlk[1] < 0) {
+ printk
+ ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n");
+ return;
+ }
+#endif
+
+ /* Connect A3D -> XTALK */
+ for (i = 0; i < 4; i++) {
+ // 2 outputs per each A3D slice.
+ vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2), ADB_XTALKIN(i));
+ vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2) + 1, ADB_XTALKIN(5 + i));
+ }
+#if 0
+ vortex_route(v, en, 0x11, ADB_XTALKOUT(0), ADB_EQIN(2));
+ vortex_route(v, en, 0x11, ADB_XTALKOUT(1), ADB_EQIN(3));
+#else
+ /* Connect XTalk -> mixer */
+ vortex_route(v, en, 0x11, ADB_XTALKOUT(0), ADB_MIXIN(v->mixxtlk[0]));
+ vortex_route(v, en, 0x11, ADB_XTALKOUT(1), ADB_MIXIN(v->mixxtlk[1]));
+ vortex_connection_mixin_mix(v, en, v->mixxtlk[0], v->mixplayb[0], 0);
+ vortex_connection_mixin_mix(v, en, v->mixxtlk[1], v->mixplayb[1], 0);
+ vortex_mix_setinputvolumebyte(v, v->mixplayb[0], v->mixxtlk[0],
+ en ? MIX_DEFIGAIN : VOL_MIN);
+ vortex_mix_setinputvolumebyte(v, v->mixplayb[1], v->mixxtlk[1],
+ en ? MIX_DEFIGAIN : VOL_MIN);
+ if (VORTEX_IS_QUAD(v)) {
+ vortex_connection_mixin_mix(v, en, v->mixxtlk[0],
+ v->mixplayb[2], 0);
+ vortex_connection_mixin_mix(v, en, v->mixxtlk[1],
+ v->mixplayb[3], 0);
+ vortex_mix_setinputvolumebyte(v, v->mixplayb[2],
+ v->mixxtlk[0],
+ en ? MIX_DEFIGAIN : VOL_MIN);
+ vortex_mix_setinputvolumebyte(v, v->mixplayb[3],
+ v->mixxtlk[1],
+ en ? MIX_DEFIGAIN : VOL_MIN);
+ }
+#endif
+}
+
+/* Initialize one single A3D source. */
+static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en)
+{
+ if (a->vortex == NULL) {
+ printk
+ ("vortex: Vort3D_InitializeSource: A3D source not initialized\n");
+ return;
+ }
+ if (en) {
+ a3dsrc_ProgramPipe(a);
+ a3dsrc_SetA3DSampleRate(a, 0x11);
+ a3dsrc_SetTimeConsts(a, HrtfTCDefault,
+ ItdTCDefault, GainTCDefault,
+ CoefTCDefault);
+ /* Remark: zero gain is muted. */
+ //a3dsrc_SetGainTarget(a,0,0);
+ //a3dsrc_SetGainCurrent(a,0,0);
+ a3dsrc_EnableA3D(a);
+ } else {
+ a3dsrc_DisableA3D(a);
+ a3dsrc_ZeroState(a);
+ }
+}
+
+/* Conversion of coordinates into 3D parameters. */
+
+static void vortex_a3d_coord2hrtf(a3d_Hrtf_t hrtf, int *coord)
+{
+ /* FIXME: implement this. */
+
+}
+static void vortex_a3d_coord2itd(a3d_Itd_t itd, int *coord)
+{
+ /* FIXME: implement this. */
+
+}
+static void vortex_a3d_coord2ild(a3d_LRGains_t ild, int left, int right)
+{
+ /* FIXME: implement this. */
+
+}
+static void vortex_a3d_translate_filter(a3d_atmos_t filter, int *params)
+{
+ /* FIXME: implement this. */
+
+}
+
+/* ALSA control interface. */
+
+static int
+snd_vortex_a3d_hrtf_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 6;
+ uinfo->value.integer.min = 0x00000000;
+ uinfo->value.integer.max = 0xffffffff;
+ return 0;
+}
+static int
+snd_vortex_a3d_itd_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0x00000000;
+ uinfo->value.integer.max = 0xffffffff;
+ return 0;
+}
+static int
+snd_vortex_a3d_ild_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0x00000000;
+ uinfo->value.integer.max = 0xffffffff;
+ return 0;
+}
+static int
+snd_vortex_a3d_filter_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 4;
+ uinfo->value.integer.min = 0x00000000;
+ uinfo->value.integer.max = 0xffffffff;
+ return 0;
+}
+
+static int
+snd_vortex_a3d_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ //a3dsrc_t *a = kcontrol->private_data;
+ /* No read yet. Would this be really useable/needed ? */
+
+ return 0;
+}
+
+static int
+snd_vortex_a3d_hrtf_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ a3dsrc_t *a = kcontrol->private_data;
+ int changed = 1, i;
+ int coord[6];
+ for (i = 0; i < 6; i++)
+ coord[i] = ucontrol->value.integer.value[i];
+ /* Translate orientation coordinates to a3d params. */
+ vortex_a3d_coord2hrtf(a->hrtf[0], coord);
+ vortex_a3d_coord2hrtf(a->hrtf[1], coord);
+ a3dsrc_SetHrtfTarget(a, a->hrtf[0], a->hrtf[1]);
+ a3dsrc_SetHrtfCurrent(a, a->hrtf[0], a->hrtf[1]);
+ return changed;
+}
+
+static int
+snd_vortex_a3d_itd_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ a3dsrc_t *a = kcontrol->private_data;
+ int coord[6];
+ int i, changed = 1;
+ for (i = 0; i < 6; i++)
+ coord[i] = ucontrol->value.integer.value[i];
+ /* Translate orientation coordinates to a3d params. */
+ vortex_a3d_coord2itd(a->hrtf[0], coord);
+ vortex_a3d_coord2itd(a->hrtf[1], coord);
+ /* Inter aural time difference. */
+ a3dsrc_SetItdTarget(a, a->itd[0], a->itd[1]);
+ a3dsrc_SetItdCurrent(a, a->itd[0], a->itd[1]);
+ a3dsrc_SetItdDline(a, a->dline);
+ return changed;
+}
+
+static int
+snd_vortex_a3d_ild_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ a3dsrc_t *a = kcontrol->private_data;
+ int changed = 1;
+ int l, r;
+ /* There may be some scale tranlation needed here. */
+ l = ucontrol->value.integer.value[0];
+ r = ucontrol->value.integer.value[1];
+ vortex_a3d_coord2ild(a->ild, l, r);
+ /* Left Right panning. */
+ a3dsrc_SetGainTarget(a, l, r);
+ a3dsrc_SetGainCurrent(a, l, r);
+ return changed;
+}
+
+static int
+snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ a3dsrc_t *a = kcontrol->private_data;
+ int i, changed = 1;
+ int params[6];
+ for (i = 0; i < 6; i++)
+ params[i] = ucontrol->value.integer.value[i];
+ /* Translate generic filter params to a3d filter params. */
+ vortex_a3d_translate_filter(a->filter, params);
+ /* Atmospheric absorbtion and filtering. */
+ a3dsrc_SetAtmosTarget(a, a->filter[0],
+ a->filter[1], a->filter[2],
+ a->filter[3], a->filter[4]);
+ a3dsrc_SetAtmosCurrent(a, a->filter[0],
+ a->filter[1], a->filter[2],
+ a->filter[3], a->filter[4]);
+ return changed;
+}
+
+static struct snd_kcontrol_new vortex_a3d_kcontrol __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "Playback PCM advanced processing",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_vortex_a3d_hrtf_info,
+ .get = snd_vortex_a3d_get,
+ .put = snd_vortex_a3d_hrtf_put,
+};
+
+/* Control (un)registration. */
+static int __devinit vortex_a3d_register_controls(vortex_t * vortex)
+{
+ struct snd_kcontrol *kcontrol;
+ int err, i;
+ /* HRTF controls. */
+ for (i = 0; i < NR_A3D; i++) {
+ if ((kcontrol =
+ snd_ctl_new1(&vortex_a3d_kcontrol, &vortex->a3d[i])) == NULL)
+ return -ENOMEM;
+ kcontrol->id.numid = CTRLID_HRTF;
+ kcontrol->info = snd_vortex_a3d_hrtf_info;
+ kcontrol->put = snd_vortex_a3d_hrtf_put;
+ if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0)
+ return err;
+ }
+ /* ITD controls. */
+ for (i = 0; i < NR_A3D; i++) {
+ if ((kcontrol =
+ snd_ctl_new1(&vortex_a3d_kcontrol, &vortex->a3d[i])) == NULL)
+ return -ENOMEM;
+ kcontrol->id.numid = CTRLID_ITD;
+ kcontrol->info = snd_vortex_a3d_itd_info;
+ kcontrol->put = snd_vortex_a3d_itd_put;
+ if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0)
+ return err;
+ }
+ /* ILD (gains) controls. */
+ for (i = 0; i < NR_A3D; i++) {
+ if ((kcontrol =
+ snd_ctl_new1(&vortex_a3d_kcontrol, &vortex->a3d[i])) == NULL)
+ return -ENOMEM;
+ kcontrol->id.numid = CTRLID_GAINS;
+ kcontrol->info = snd_vortex_a3d_ild_info;
+ kcontrol->put = snd_vortex_a3d_ild_put;
+ if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0)
+ return err;
+ }
+ /* Filter controls. */
+ for (i = 0; i < NR_A3D; i++) {
+ if ((kcontrol =
+ snd_ctl_new1(&vortex_a3d_kcontrol, &vortex->a3d[i])) == NULL)
+ return -ENOMEM;
+ kcontrol->id.numid = CTRLID_FILTER;
+ kcontrol->info = snd_vortex_a3d_filter_info;
+ kcontrol->put = snd_vortex_a3d_filter_put;
+ if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0)
+ return err;
+ }
+ return 0;
+}
+
+static void vortex_a3d_unregister_controls(vortex_t * vortex)
+{
+
+}
+
+/* End of File*/
diff --git a/sound/pci/au88x0/au88x0_a3d.h b/sound/pci/au88x0/au88x0_a3d.h
new file mode 100644
index 0000000..0584c65
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_a3d.h
@@ -0,0 +1,123 @@
+/***************************************************************************
+ * au88x0_a3d.h
+ *
+ * Fri Jul 18 14:16:03 2003
+ * Copyright 2003 mjander
+ * mjander@users.sourceforge.net
+ ****************************************************************************/
+
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#ifndef _AU88X0_A3D_H
+#define _AU88X0_A3D_H
+
+//#include <openal.h>
+
+#define HRTF_SZ 0x38
+#define DLINE_SZ 0x28
+
+#define CTRLID_HRTF 1
+#define CTRLID_ITD 2
+#define CTRLID_ILD 4
+#define CTRLID_FILTER 8
+#define CTRLID_GAINS 16
+
+/* 3D parameter structs */
+typedef unsigned short int a3d_Hrtf_t[HRTF_SZ];
+typedef unsigned short int a3d_ItdDline_t[DLINE_SZ];
+typedef unsigned short int a3d_atmos_t[5];
+typedef unsigned short int a3d_LRGains_t[2];
+typedef unsigned short int a3d_Itd_t[2];
+typedef unsigned short int a3d_Ild_t[2];
+
+typedef struct {
+ void *vortex; // Formerly CAsp4HwIO*, now vortex_t*.
+ unsigned int source; /* this_04 */
+ unsigned int slice; /* this_08 */
+ a3d_Hrtf_t hrtf[2];
+ a3d_Itd_t itd;
+ a3d_Ild_t ild;
+ a3d_ItdDline_t dline;
+ a3d_atmos_t filter;
+} a3dsrc_t;
+
+/* First Register bank */
+
+#define A3D_A_HrtfCurrent 0x18000 /* 56 ULONG */
+#define A3D_A_GainCurrent 0x180E0
+#define A3D_A_GainTarget 0x180E4
+#define A3D_A_A12Current 0x180E8 /* Atmospheric current. */
+#define A3D_A_A21Target 0x180EC /* Atmospheric target */
+#define A3D_A_B01Current 0x180F0 /* Atmospheric current */
+#define A3D_A_B10Target 0x180F4 /* Atmospheric target */
+#define A3D_A_B2Current 0x180F8 /* Atmospheric current */
+#define A3D_A_B2Target 0x180FC /* Atmospheric target */
+#define A3D_A_HrtfTarget 0x18100 /* 56 ULONG */
+#define A3D_A_ITDCurrent 0x181E0
+#define A3D_A_ITDTarget 0x181E4
+#define A3D_A_HrtfDelayLine 0x181E8 /* 56 ULONG */
+#define A3D_A_ITDDelayLine 0x182C8 /* 40/45 ULONG */
+#define A3D_A_HrtfTrackTC 0x1837C /* Time Constants */
+#define A3D_A_GainTrackTC 0x18380
+#define A3D_A_CoeffTrackTC 0x18384
+#define A3D_A_ITDTrackTC 0x18388
+#define A3D_A_x1 0x1838C
+#define A3D_A_x2 0x18390
+#define A3D_A_y1 0x18394
+#define A3D_A_y2 0x18398
+#define A3D_A_HrtfOutL 0x1839C
+#define A3D_A_HrtfOutR 0x183A0
+#define A3D_A_TAIL 0x183A4
+
+/* Second register bank */
+#define A3D_B_HrtfCurrent 0x19000 /* 56 ULONG */
+#define A3D_B_GainCurrent 0x190E0
+#define A3D_B_GainTarget 0x190E4
+#define A3D_B_A12Current 0x190E8
+#define A3D_B_A21Target 0x190EC
+#define A3D_B_B01Current 0x190F0
+#define A3D_B_B10Target 0x190F4
+#define A3D_B_B2Current 0x190F8
+#define A3D_B_B2Target 0x190FC
+#define A3D_B_HrtfTarget 0x19100 /* 56 ULONG */
+#define A3D_B_ITDCurrent 0x191E0
+#define A3D_B_ITDTarget 0x191E4
+#define A3D_B_HrtfDelayLine 0x191E8 /* 56 ULONG */
+#define A3D_B_TAIL 0x192C8
+
+/* There are 4 slices, 4 a3d each = 16 a3d sources. */
+#define A3D_SLICE_BANK_A 0x18000 /* 4 sources */
+#define A3D_SLICE_BANK_B 0x19000 /* 4 sources */
+#define A3D_SLICE_VDBDest 0x19C00 /* 8 ULONG */
+#define A3D_SLICE_VDBSource 0x19C20 /* 4 ULONG */
+#define A3D_SLICE_ABReg 0x19C30
+#define A3D_SLICE_CReg 0x19C34
+#define A3D_SLICE_Control 0x19C38
+#define A3D_SLICE_DebugReserved 0x19C3c /* Dangerous! */
+#define A3D_SLICE_Pointers 0x19C40
+#define A3D_SLICE_TAIL 0x1A000
+
+// Slice size: 0x2000
+// Source size: 0x3A4, 0x2C8
+
+/* Address generator macro. */
+#define a3d_addrA(slice,source,reg) (((slice)<<0xd)+((source)*0x3A4)+(reg))
+#define a3d_addrB(slice,source,reg) (((slice)<<0xd)+((source)*0x2C8)+(reg))
+#define a3d_addrS(slice,reg) (((slice)<<0xd)+(reg))
+//#define a3d_addr(slice,source,reg) (((reg)>=0x19000) ? a3d_addr2((slice),(source),(reg)) : a3d_addr1((slice),(source),(reg)))
+
+#endif /* _AU88X0_A3D_H */
diff --git a/sound/pci/au88x0/au88x0_a3ddata.c b/sound/pci/au88x0/au88x0_a3ddata.c
new file mode 100644
index 0000000..6fab4bb
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_a3ddata.c
@@ -0,0 +1,91 @@
+/***************************************************************************
+ * au88x0_a3ddata.c
+ *
+ * Wed Nov 19 21:11:32 2003
+ * Copyright 2003 mjander
+ * mjander@users.sourceforge.org
+ ****************************************************************************/
+
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+/* Constant initializer values. */
+
+static const a3d_Hrtf_t A3dHrirZeros = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0
+};
+
+static const a3d_Hrtf_t A3dHrirImpulse = {
+ 0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0
+};
+
+static const a3d_Hrtf_t A3dHrirOnes = {
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
+ 0x7fff,
+ 0x7fff,
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
+ 0x7fff,
+ 0x7fff,
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
+ 0x7fff,
+ 0x7fff,
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
+ 0x7fff,
+ 0x7fff,
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff
+};
+
+static const a3d_Hrtf_t A3dHrirSatTest = {
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
+ 0x7fff,
+ 0x7fff,
+ 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001, 0x8001,
+ 0x8001,
+ 0x8001,
+ 0x7fff, 0x0000, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
+};
+
+static const a3d_Hrtf_t A3dHrirDImpulse = {
+ 0, 0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0,
+ 0, 0, 0
+};
+
+static const a3d_ItdDline_t A3dItdDlineZeros = {
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
+};
+
+static short const GainTCDefault = 0x300;
+static short const ItdTCDefault = 0x0C8;
+static short const HrtfTCDefault = 0x147;
+static short const CoefTCDefault = 0x300;
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
new file mode 100644
index 0000000..b070e57
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -0,0 +1,2835 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+/*
+ Vortex core low level functions.
+
+ Author: Manuel Jander (mjander@users.sourceforge.cl)
+ These functions are mainly the result of translations made
+ from the original disassembly of the au88x0 binary drivers,
+ written by Aureal before they went down.
+ Many thanks to the Jeff Muizelaar, Kester Maddock, and whoever
+ contributed to the OpenVortex project.
+ The author of this file, put the few available pieces together
+ and translated the rest of the riddle (Mix, Src and connection stuff).
+ Some things are still to be discovered, and their meanings are unclear.
+
+ Some of these functions aren't intended to be really used, rather
+ to help to understand how does the AU88X0 chips work. Keep them in, because
+ they could be used somewhere in the future.
+
+ This code hasn't been tested or proof read thoroughly. If you wanna help,
+ take a look at the AU88X0 assembly and check if this matches.
+ Functions tested ok so far are (they show the desired effect
+ at least):
+ vortex_routes(); (1 bug fixed).
+ vortex_adb_addroute();
+ vortex_adb_addroutes();
+ vortex_connect_codecplay();
+ vortex_src_flushbuffers();
+ vortex_adbdma_setmode(); note: still some unknown arguments!
+ vortex_adbdma_startfifo();
+ vortex_adbdma_stopfifo();
+ vortex_fifo_setadbctrl(); note: still some unknown arguments!
+ vortex_mix_setinputvolumebyte();
+ vortex_mix_enableinput();
+ vortex_mixer_addWTD(); (fixed)
+ vortex_connection_adbdma_src_src();
+ vortex_connection_adbdma_src();
+ vortex_src_change_convratio();
+ vortex_src_addWTD(); (fixed)
+
+ History:
+
+ 01-03-2003 First revision.
+ 01-21-2003 Some bug fixes.
+ 17-02-2003 many bugfixes after a big versioning mess.
+ 18-02-2003 JAAAAAHHHUUUUUU!!!! The mixer works !! I'm just so happy !
+ (2 hours later...) I cant believe it! Im really lucky today.
+ Now the SRC is working too! Yeah! XMMS works !
+ 20-02-2003 First steps into the ALSA world.
+ 28-02-2003 As my birthday present, i discovered how the DMA buffer pages really
+ work :-). It was all wrong.
+ 12-03-2003 ALSA driver starts working (2 channels).
+ 16-03-2003 More srcblock_setupchannel discoveries.
+ 12-04-2003 AU8830 playback support. Recording in the works.
+ 17-04-2003 vortex_route() and vortex_routes() bug fixes. AU8830 recording
+ works now, but chipn' dale effect is still there.
+ 16-05-2003 SrcSetupChannel cleanup. Moved the Src setup stuff entirely
+ into au88x0_pcm.c .
+ 06-06-2003 Buffer shifter bugfix. Mixer volume fix.
+ 07-12-2003 A3D routing finally fixed. Believed to be OK.
+ 25-03-2004 Many thanks to Claudia, for such valuable bug reports.
+
+*/
+
+#include "au88x0.h"
+#include "au88x0_a3d.h"
+#include <linux/delay.h>
+
+/* MIXER (CAsp4Mix.s and CAsp4Mixer.s) */
+
+// FIXME: get rid of this.
+static int mchannels[NR_MIXIN];
+static int rampchs[NR_MIXIN];
+
+static void vortex_mixer_en_sr(vortex_t * vortex, int channel)
+{
+ hwwrite(vortex->mmio, VORTEX_MIXER_SR,
+ hwread(vortex->mmio, VORTEX_MIXER_SR) | (0x1 << channel));
+}
+static void vortex_mixer_dis_sr(vortex_t * vortex, int channel)
+{
+ hwwrite(vortex->mmio, VORTEX_MIXER_SR,
+ hwread(vortex->mmio, VORTEX_MIXER_SR) & ~(0x1 << channel));
+}
+
+#if 0
+static void
+vortex_mix_muteinputgain(vortex_t * vortex, unsigned char mix,
+ unsigned char channel)
+{
+ hwwrite(vortex->mmio, VORTEX_MIX_INVOL_A + ((mix << 5) + channel),
+ 0x80);
+ hwwrite(vortex->mmio, VORTEX_MIX_INVOL_B + ((mix << 5) + channel),
+ 0x80);
+}
+
+static int vortex_mix_getvolume(vortex_t * vortex, unsigned char mix)
+{
+ int a;
+ a = hwread(vortex->mmio, VORTEX_MIX_VOL_A + (mix << 2)) & 0xff;
+ //FP2LinearFrac(a);
+ return (a);
+}
+
+static int
+vortex_mix_getinputvolume(vortex_t * vortex, unsigned char mix,
+ int channel, int *vol)
+{
+ int a;
+ if (!(mchannels[mix] & (1 << channel)))
+ return 0;
+ a = hwread(vortex->mmio,
+ VORTEX_MIX_INVOL_A + (((mix << 5) + channel) << 2));
+ /*
+ if (rampchs[mix] == 0)
+ a = FP2LinearFrac(a);
+ else
+ a = FP2LinearFracWT(a);
+ */
+ *vol = a;
+ return (0);
+}
+
+static unsigned int vortex_mix_boost6db(unsigned char vol)
+{
+ return (vol + 8); /* WOW! what a complex function! */
+}
+
+static void vortex_mix_rampvolume(vortex_t * vortex, int mix)
+{
+ int ch;
+ char a;
+ // This function is intended for ramping down only (see vortex_disableinput()).
+ for (ch = 0; ch < 0x20; ch++) {
+ if (((1 << ch) & rampchs[mix]) == 0)
+ continue;
+ a = hwread(vortex->mmio,
+ VORTEX_MIX_INVOL_B + (((mix << 5) + ch) << 2));
+ if (a > -126) {
+ a -= 2;
+ hwwrite(vortex->mmio,
+ VORTEX_MIX_INVOL_A +
+ (((mix << 5) + ch) << 2), a);
+ hwwrite(vortex->mmio,
+ VORTEX_MIX_INVOL_B +
+ (((mix << 5) + ch) << 2), a);
+ } else
+ vortex_mix_killinput(vortex, mix, ch);
+ }
+}
+
+static int
+vortex_mix_getenablebit(vortex_t * vortex, unsigned char mix, int mixin)
+{
+ int addr, temp;
+ if (mixin >= 0)
+ addr = mixin;
+ else
+ addr = mixin + 3;
+ addr = ((mix << 3) + (addr >> 2)) << 2;
+ temp = hwread(vortex->mmio, VORTEX_MIX_ENIN + addr);
+ return ((temp >> (mixin & 3)) & 1);
+}
+#endif
+static void
+vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix,
+ unsigned char vol)
+{
+ int temp;
+ hwwrite(vortex->mmio, VORTEX_MIX_VOL_A + (mix << 2), vol);
+ if (1) { /*if (this_10) */
+ temp = hwread(vortex->mmio, VORTEX_MIX_VOL_B + (mix << 2));
+ if ((temp != 0x80) || (vol == 0x80))
+ return;
+ }
+ hwwrite(vortex->mmio, VORTEX_MIX_VOL_B + (mix << 2), vol);
+}
+
+static void
+vortex_mix_setinputvolumebyte(vortex_t * vortex, unsigned char mix,
+ int mixin, unsigned char vol)
+{
+ int temp;
+
+ hwwrite(vortex->mmio,
+ VORTEX_MIX_INVOL_A + (((mix << 5) + mixin) << 2), vol);
+ if (1) { /* this_10, initialized to 1. */
+ temp =
+ hwread(vortex->mmio,
+ VORTEX_MIX_INVOL_B + (((mix << 5) + mixin) << 2));
+ if ((temp != 0x80) || (vol == 0x80))
+ return;
+ }
+ hwwrite(vortex->mmio,
+ VORTEX_MIX_INVOL_B + (((mix << 5) + mixin) << 2), vol);
+}
+
+static void
+vortex_mix_setenablebit(vortex_t * vortex, unsigned char mix, int mixin, int en)
+{
+ int temp, addr;
+
+ if (mixin < 0)
+ addr = (mixin + 3);
+ else
+ addr = mixin;
+ addr = ((mix << 3) + (addr >> 2)) << 2;
+ temp = hwread(vortex->mmio, VORTEX_MIX_ENIN + addr);
+ if (en)
+ temp |= (1 << (mixin & 3));
+ else
+ temp &= ~(1 << (mixin & 3));
+ /* Mute input. Astatic void crackling? */
+ hwwrite(vortex->mmio,
+ VORTEX_MIX_INVOL_B + (((mix << 5) + mixin) << 2), 0x80);
+ /* Looks like clear buffer. */
+ hwwrite(vortex->mmio, VORTEX_MIX_SMP + (mixin << 2), 0x0);
+ hwwrite(vortex->mmio, VORTEX_MIX_SMP + 4 + (mixin << 2), 0x0);
+ /* Write enable bit. */
+ hwwrite(vortex->mmio, VORTEX_MIX_ENIN + addr, temp);
+}
+
+static void
+vortex_mix_killinput(vortex_t * vortex, unsigned char mix, int mixin)
+{
+ rampchs[mix] &= ~(1 << mixin);
+ vortex_mix_setinputvolumebyte(vortex, mix, mixin, 0x80);
+ mchannels[mix] &= ~(1 << mixin);
+ vortex_mix_setenablebit(vortex, mix, mixin, 0);
+}
+
+static void
+vortex_mix_enableinput(vortex_t * vortex, unsigned char mix, int mixin)
+{
+ vortex_mix_killinput(vortex, mix, mixin);
+ if ((mchannels[mix] & (1 << mixin)) == 0) {
+ vortex_mix_setinputvolumebyte(vortex, mix, mixin, 0x80); /*0x80 : mute */
+ mchannels[mix] |= (1 << mixin);
+ }
+ vortex_mix_setenablebit(vortex, mix, mixin, 1);
+}
+
+static void
+vortex_mix_disableinput(vortex_t * vortex, unsigned char mix, int channel,
+ int ramp)
+{
+ if (ramp) {
+ rampchs[mix] |= (1 << channel);
+ // Register callback.
+ //vortex_mix_startrampvolume(vortex);
+ vortex_mix_killinput(vortex, mix, channel);
+ } else
+ vortex_mix_killinput(vortex, mix, channel);
+}
+
+static int
+vortex_mixer_addWTD(vortex_t * vortex, unsigned char mix, unsigned char ch)
+{
+ int temp, lifeboat = 0, prev;
+
+ temp = hwread(vortex->mmio, VORTEX_MIXER_SR);
+ if ((temp & (1 << ch)) == 0) {
+ hwwrite(vortex->mmio, VORTEX_MIXER_CHNBASE + (ch << 2), mix);
+ vortex_mixer_en_sr(vortex, ch);
+ return 1;
+ }
+ prev = VORTEX_MIXER_CHNBASE + (ch << 2);
+ temp = hwread(vortex->mmio, prev);
+ while (temp & 0x10) {
+ prev = VORTEX_MIXER_RTBASE + ((temp & 0xf) << 2);
+ temp = hwread(vortex->mmio, prev);
+ //printk(KERN_INFO "vortex: mixAddWTD: while addr=%x, val=%x\n", prev, temp);
+ if ((++lifeboat) > 0xf) {
+ printk(KERN_ERR
+ "vortex_mixer_addWTD: lifeboat overflow\n");
+ return 0;
+ }
+ }
+ hwwrite(vortex->mmio, VORTEX_MIXER_RTBASE + ((temp & 0xf) << 2), mix);
+ hwwrite(vortex->mmio, prev, (temp & 0xf) | 0x10);
+ return 1;
+}
+
+static int
+vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch)
+{
+ int esp14 = -1, esp18, eax, ebx, edx, ebp, esi = 0;
+ //int esp1f=edi(while)=src, esp10=ch;
+
+ eax = hwread(vortex->mmio, VORTEX_MIXER_SR);
+ if (((1 << ch) & eax) == 0) {
+ printk(KERN_ERR "mix ALARM %x\n", eax);
+ return 0;
+ }
+ ebp = VORTEX_MIXER_CHNBASE + (ch << 2);
+ esp18 = hwread(vortex->mmio, ebp);
+ if (esp18 & 0x10) {
+ ebx = (esp18 & 0xf);
+ if (mix == ebx) {
+ ebx = VORTEX_MIXER_RTBASE + (mix << 2);
+ edx = hwread(vortex->mmio, ebx);
+ //7b60
+ hwwrite(vortex->mmio, ebp, edx);
+ hwwrite(vortex->mmio, ebx, 0);
+ } else {
+ //7ad3
+ edx =
+ hwread(vortex->mmio,
+ VORTEX_MIXER_RTBASE + (ebx << 2));
+ //printk(KERN_INFO "vortex: mixdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src);
+ while ((edx & 0xf) != mix) {
+ if ((esi) > 0xf) {
+ printk(KERN_ERR
+ "vortex: mixdelWTD: error lifeboat overflow\n");
+ return 0;
+ }
+ esp14 = ebx;
+ ebx = edx & 0xf;
+ ebp = ebx << 2;
+ edx =
+ hwread(vortex->mmio,
+ VORTEX_MIXER_RTBASE + ebp);
+ //printk(KERN_INFO "vortex: mixdelWTD: while addr=%x, val=%x\n", ebp, edx);
+ esi++;
+ }
+ //7b30
+ ebp = ebx << 2;
+ if (edx & 0x10) { /* Delete entry in between others */
+ ebx = VORTEX_MIXER_RTBASE + ((edx & 0xf) << 2);
+ edx = hwread(vortex->mmio, ebx);
+ //7b60
+ hwwrite(vortex->mmio,
+ VORTEX_MIXER_RTBASE + ebp, edx);
+ hwwrite(vortex->mmio, ebx, 0);
+ //printk(KERN_INFO "vortex mixdelWTD between addr= 0x%x, val= 0x%x\n", ebp, edx);
+ } else { /* Delete last entry */
+ //7b83
+ if (esp14 == -1)
+ hwwrite(vortex->mmio,
+ VORTEX_MIXER_CHNBASE +
+ (ch << 2), esp18 & 0xef);
+ else {
+ ebx = (0xffffffe0 & edx) | (0xf & ebx);
+ hwwrite(vortex->mmio,
+ VORTEX_MIXER_RTBASE +
+ (esp14 << 2), ebx);
+ //printk(KERN_INFO "vortex mixdelWTD last addr= 0x%x, val= 0x%x\n", esp14, ebx);
+ }
+ hwwrite(vortex->mmio,
+ VORTEX_MIXER_RTBASE + ebp, 0);
+ return 1;
+ }
+ }
+ } else {
+ //printk(KERN_INFO "removed last mix\n");
+ //7be0
+ vortex_mixer_dis_sr(vortex, ch);
+ hwwrite(vortex->mmio, ebp, 0);
+ }
+ return 1;
+}
+
+static void vortex_mixer_init(vortex_t * vortex)
+{
+ u32 addr;
+ int x;
+
+ // FIXME: get rid of this crap.
+ memset(mchannels, 0, NR_MIXOUT * sizeof(int));
+ memset(rampchs, 0, NR_MIXOUT * sizeof(int));
+
+ addr = VORTEX_MIX_SMP + 0x17c;
+ for (x = 0x5f; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0);
+ addr -= 4;
+ }
+ addr = VORTEX_MIX_ENIN + 0x1fc;
+ for (x = 0x7f; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0);
+ addr -= 4;
+ }
+ addr = VORTEX_MIX_SMP + 0x17c;
+ for (x = 0x5f; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0);
+ addr -= 4;
+ }
+ addr = VORTEX_MIX_INVOL_A + 0x7fc;
+ for (x = 0x1ff; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0x80);
+ addr -= 4;
+ }
+ addr = VORTEX_MIX_VOL_A + 0x3c;
+ for (x = 0xf; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0x80);
+ addr -= 4;
+ }
+ addr = VORTEX_MIX_INVOL_B + 0x7fc;
+ for (x = 0x1ff; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0x80);
+ addr -= 4;
+ }
+ addr = VORTEX_MIX_VOL_B + 0x3c;
+ for (x = 0xf; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0x80);
+ addr -= 4;
+ }
+ addr = VORTEX_MIXER_RTBASE + (MIXER_RTBASE_SIZE - 1) * 4;
+ for (x = (MIXER_RTBASE_SIZE - 1); x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0x0);
+ addr -= 4;
+ }
+ hwwrite(vortex->mmio, VORTEX_MIXER_SR, 0);
+
+ /* Set clipping ceiling (this may be all wrong). */
+ /*
+ for (x = 0; x < 0x80; x++) {
+ hwwrite(vortex->mmio, VORTEX_MIXER_CLIP + (x << 2), 0x3ffff);
+ }
+ */
+ /*
+ call CAsp4Mix__Initialize_CAsp4HwIO____CAsp4Mixer____
+ Register ISR callback for volume smooth fade out.
+ Maybe this avoids clicks when press "stop" ?
+ */
+}
+
+/* SRC (CAsp4Src.s and CAsp4SrcBlock) */
+
+static void vortex_src_en_sr(vortex_t * vortex, int channel)
+{
+ hwwrite(vortex->mmio, VORTEX_SRCBLOCK_SR,
+ hwread(vortex->mmio, VORTEX_SRCBLOCK_SR) | (0x1 << channel));
+}
+
+static void vortex_src_dis_sr(vortex_t * vortex, int channel)
+{
+ hwwrite(vortex->mmio, VORTEX_SRCBLOCK_SR,
+ hwread(vortex->mmio, VORTEX_SRCBLOCK_SR) & ~(0x1 << channel));
+}
+
+static void vortex_src_flushbuffers(vortex_t * vortex, unsigned char src)
+{
+ int i;
+
+ for (i = 0x1f; i >= 0; i--)
+ hwwrite(vortex->mmio,
+ VORTEX_SRC_DATA0 + (src << 7) + (i << 2), 0);
+ hwwrite(vortex->mmio, VORTEX_SRC_DATA + (src << 3), 0);
+ hwwrite(vortex->mmio, VORTEX_SRC_DATA + (src << 3) + 4, 0);
+}
+
+static void vortex_src_cleardrift(vortex_t * vortex, unsigned char src)
+{
+ hwwrite(vortex->mmio, VORTEX_SRC_DRIFT0 + (src << 2), 0);
+ hwwrite(vortex->mmio, VORTEX_SRC_DRIFT1 + (src << 2), 0);
+ hwwrite(vortex->mmio, VORTEX_SRC_DRIFT2 + (src << 2), 1);
+}
+
+static void
+vortex_src_set_throttlesource(vortex_t * vortex, unsigned char src, int en)
+{
+ int temp;
+
+ temp = hwread(vortex->mmio, VORTEX_SRC_SOURCE);
+ if (en)
+ temp |= 1 << src;
+ else
+ temp &= ~(1 << src);
+ hwwrite(vortex->mmio, VORTEX_SRC_SOURCE, temp);
+}
+
+static int
+vortex_src_persist_convratio(vortex_t * vortex, unsigned char src, int ratio)
+{
+ int temp, lifeboat = 0;
+
+ do {
+ hwwrite(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2), ratio);
+ temp = hwread(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2));
+ if ((++lifeboat) > 0x9) {
+ printk(KERN_ERR "Vortex: Src cvr fail\n");
+ break;
+ }
+ }
+ while (temp != ratio);
+ return temp;
+}
+
+#if 0
+static void vortex_src_slowlock(vortex_t * vortex, unsigned char src)
+{
+ int temp;
+
+ hwwrite(vortex->mmio, VORTEX_SRC_DRIFT2 + (src << 2), 1);
+ hwwrite(vortex->mmio, VORTEX_SRC_DRIFT0 + (src << 2), 0);
+ temp = hwread(vortex->mmio, VORTEX_SRC_U0 + (src << 2));
+ if (temp & 0x200)
+ hwwrite(vortex->mmio, VORTEX_SRC_U0 + (src << 2),
+ temp & ~0x200L);
+}
+
+static void
+vortex_src_change_convratio(vortex_t * vortex, unsigned char src, int ratio)
+{
+ int temp, a;
+
+ if ((ratio & 0x10000) && (ratio != 0x10000)) {
+ if (ratio & 0x3fff)
+ a = (0x11 - ((ratio >> 0xe) & 0x3)) - 1;
+ else
+ a = (0x11 - ((ratio >> 0xe) & 0x3)) - 2;
+ } else
+ a = 0xc;
+ temp = hwread(vortex->mmio, VORTEX_SRC_U0 + (src << 2));
+ if (((temp >> 4) & 0xf) != a)
+ hwwrite(vortex->mmio, VORTEX_SRC_U0 + (src << 2),
+ (temp & 0xf) | ((a & 0xf) << 4));
+
+ vortex_src_persist_convratio(vortex, src, ratio);
+}
+
+static int
+vortex_src_checkratio(vortex_t * vortex, unsigned char src,
+ unsigned int desired_ratio)
+{
+ int hw_ratio, lifeboat = 0;
+
+ hw_ratio = hwread(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2));
+
+ while (hw_ratio != desired_ratio) {
+ hwwrite(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2), desired_ratio);
+
+ if ((lifeboat++) > 15) {
+ printk(KERN_ERR "Vortex: could not set src-%d from %d to %d\n",
+ src, hw_ratio, desired_ratio);
+ break;
+ }
+ }
+
+ return hw_ratio;
+}
+
+#endif
+/*
+ Objective: Set samplerate for given SRC module.
+ Arguments:
+ card: pointer to vortex_t strcut.
+ src: Integer index of the SRC module.
+ cr: Current sample rate conversion factor.
+ b: unknown 16 bit value.
+ sweep: Enable Samplerate fade from cr toward tr flag.
+ dirplay: 1: playback, 0: recording.
+ sl: Slow Lock flag.
+ tr: Target samplerate conversion.
+ thsource: Throttle source flag (no idea what that means).
+*/
+static void vortex_src_setupchannel(vortex_t * card, unsigned char src,
+ unsigned int cr, unsigned int b, int sweep, int d,
+ int dirplay, int sl, unsigned int tr, int thsource)
+{
+ // noplayback: d=2,4,7,0xa,0xb when using first 2 src's.
+ // c: enables pitch sweep.
+ // looks like g is c related. Maybe g is a sweep parameter ?
+ // g = cvr
+ // dirplay: 0 = recording, 1 = playback
+ // d = src hw index.
+
+ int esi, ebp = 0, esp10;
+
+ vortex_src_flushbuffers(card, src);
+
+ if (sweep) {
+ if ((tr & 0x10000) && (tr != 0x10000)) {
+ tr = 0;
+ esi = 0x7;
+ } else {
+ if ((((short)tr) < 0) && (tr != 0x8000)) {
+ tr = 0;
+ esi = 0x8;
+ } else {
+ tr = 1;
+ esi = 0xc;
+ }
+ }
+ } else {
+ if ((cr & 0x10000) && (cr != 0x10000)) {
+ tr = 0; /*ebx = 0 */
+ esi = 0x11 - ((cr >> 0xe) & 7);
+ if (cr & 0x3fff)
+ esi -= 1;
+ else
+ esi -= 2;
+ } else {
+ tr = 1;
+ esi = 0xc;
+ }
+ }
+ vortex_src_cleardrift(card, src);
+ vortex_src_set_throttlesource(card, src, thsource);
+
+ if ((dirplay == 0) && (sweep == 0)) {
+ if (tr)
+ esp10 = 0xf;
+ else
+ esp10 = 0xc;
+ ebp = 0;
+ } else {
+ if (tr)
+ ebp = 0xf;
+ else
+ ebp = 0xc;
+ esp10 = 0;
+ }
+ hwwrite(card->mmio, VORTEX_SRC_U0 + (src << 2),
+ (sl << 0x9) | (sweep << 0x8) | ((esi & 0xf) << 4) | d);
+ /* 0xc0 esi=0xc c=f=0 d=0 */
+ vortex_src_persist_convratio(card, src, cr);
+ hwwrite(card->mmio, VORTEX_SRC_U1 + (src << 2), b & 0xffff);
+ /* 0 b=0 */
+ hwwrite(card->mmio, VORTEX_SRC_U2 + (src << 2),
+ (tr << 0x11) | (dirplay << 0x10) | (ebp << 0x8) | esp10);
+ /* 0x30f00 e=g=1 esp10=0 ebp=f */
+ //printk(KERN_INFO "vortex: SRC %d, d=0x%x, esi=0x%x, esp10=0x%x, ebp=0x%x\n", src, d, esi, esp10, ebp);
+}
+
+static void vortex_srcblock_init(vortex_t * vortex)
+{
+ u32 addr;
+ int x;
+ hwwrite(vortex->mmio, VORTEX_SRC_SOURCESIZE, 0x1ff);
+ /*
+ for (x=0; x<0x10; x++) {
+ vortex_src_init(&vortex_src[x], x);
+ }
+ */
+ //addr = 0xcc3c;
+ //addr = 0x26c3c;
+ addr = VORTEX_SRC_RTBASE + 0x3c;
+ for (x = 0xf; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0);
+ addr -= 4;
+ }
+ //addr = 0xcc94;
+ //addr = 0x26c94;
+ addr = VORTEX_SRC_CHNBASE + 0x54;
+ for (x = 0x15; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, 0);
+ addr -= 4;
+ }
+}
+
+static int
+vortex_src_addWTD(vortex_t * vortex, unsigned char src, unsigned char ch)
+{
+ int temp, lifeboat = 0, prev;
+ // esp13 = src
+
+ temp = hwread(vortex->mmio, VORTEX_SRCBLOCK_SR);
+ if ((temp & (1 << ch)) == 0) {
+ hwwrite(vortex->mmio, VORTEX_SRC_CHNBASE + (ch << 2), src);
+ vortex_src_en_sr(vortex, ch);
+ return 1;
+ }
+ prev = VORTEX_SRC_CHNBASE + (ch << 2); /*ebp */
+ temp = hwread(vortex->mmio, prev);
+ //while (temp & NR_SRC) {
+ while (temp & 0x10) {
+ prev = VORTEX_SRC_RTBASE + ((temp & 0xf) << 2); /*esp12 */
+ //prev = VORTEX_SRC_RTBASE + ((temp & (NR_SRC-1)) << 2); /*esp12*/
+ temp = hwread(vortex->mmio, prev);
+ //printk(KERN_INFO "vortex: srcAddWTD: while addr=%x, val=%x\n", prev, temp);
+ if ((++lifeboat) > 0xf) {
+ printk(KERN_ERR
+ "vortex_src_addWTD: lifeboat overflow\n");
+ return 0;
+ }
+ }
+ hwwrite(vortex->mmio, VORTEX_SRC_RTBASE + ((temp & 0xf) << 2), src);
+ //hwwrite(vortex->mmio, prev, (temp & (NR_SRC-1)) | NR_SRC);
+ hwwrite(vortex->mmio, prev, (temp & 0xf) | 0x10);
+ return 1;
+}
+
+static int
+vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch)
+{
+ int esp14 = -1, esp18, eax, ebx, edx, ebp, esi = 0;
+ //int esp1f=edi(while)=src, esp10=ch;
+
+ eax = hwread(vortex->mmio, VORTEX_SRCBLOCK_SR);
+ if (((1 << ch) & eax) == 0) {
+ printk(KERN_ERR "src alarm\n");
+ return 0;
+ }
+ ebp = VORTEX_SRC_CHNBASE + (ch << 2);
+ esp18 = hwread(vortex->mmio, ebp);
+ if (esp18 & 0x10) {
+ ebx = (esp18 & 0xf);
+ if (src == ebx) {
+ ebx = VORTEX_SRC_RTBASE + (src << 2);
+ edx = hwread(vortex->mmio, ebx);
+ //7b60
+ hwwrite(vortex->mmio, ebp, edx);
+ hwwrite(vortex->mmio, ebx, 0);
+ } else {
+ //7ad3
+ edx =
+ hwread(vortex->mmio,
+ VORTEX_SRC_RTBASE + (ebx << 2));
+ //printk(KERN_INFO "vortex: srcdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src);
+ while ((edx & 0xf) != src) {
+ if ((esi) > 0xf) {
+ printk
+ ("vortex: srcdelWTD: error, lifeboat overflow\n");
+ return 0;
+ }
+ esp14 = ebx;
+ ebx = edx & 0xf;
+ ebp = ebx << 2;
+ edx =
+ hwread(vortex->mmio,
+ VORTEX_SRC_RTBASE + ebp);
+ //printk(KERN_INFO "vortex: srcdelWTD: while addr=%x, val=%x\n", ebp, edx);
+ esi++;
+ }
+ //7b30
+ ebp = ebx << 2;
+ if (edx & 0x10) { /* Delete entry in between others */
+ ebx = VORTEX_SRC_RTBASE + ((edx & 0xf) << 2);
+ edx = hwread(vortex->mmio, ebx);
+ //7b60
+ hwwrite(vortex->mmio,
+ VORTEX_SRC_RTBASE + ebp, edx);
+ hwwrite(vortex->mmio, ebx, 0);
+ //printk(KERN_INFO "vortex srcdelWTD between addr= 0x%x, val= 0x%x\n", ebp, edx);
+ } else { /* Delete last entry */
+ //7b83
+ if (esp14 == -1)
+ hwwrite(vortex->mmio,
+ VORTEX_SRC_CHNBASE +
+ (ch << 2), esp18 & 0xef);
+ else {
+ ebx = (0xffffffe0 & edx) | (0xf & ebx);
+ hwwrite(vortex->mmio,
+ VORTEX_SRC_RTBASE +
+ (esp14 << 2), ebx);
+ //printk(KERN_INFO"vortex srcdelWTD last addr= 0x%x, val= 0x%x\n", esp14, ebx);
+ }
+ hwwrite(vortex->mmio,
+ VORTEX_SRC_RTBASE + ebp, 0);
+ return 1;
+ }
+ }
+ } else {
+ //7be0
+ vortex_src_dis_sr(vortex, ch);
+ hwwrite(vortex->mmio, ebp, 0);
+ }
+ return 1;
+}
+
+ /*FIFO*/
+
+static void
+vortex_fifo_clearadbdata(vortex_t * vortex, int fifo, int x)
+{
+ for (x--; x >= 0; x--)
+ hwwrite(vortex->mmio,
+ VORTEX_FIFO_ADBDATA +
+ (((fifo << FIFO_SIZE_BITS) + x) << 2), 0);
+}
+
+#if 0
+static void vortex_fifo_adbinitialize(vortex_t * vortex, int fifo, int j)
+{
+ vortex_fifo_clearadbdata(vortex, fifo, FIFO_SIZE);
+#ifdef CHIP_AU8820
+ hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2),
+ (FIFO_U1 | ((j & FIFO_MASK) << 0xb)));
+#else
+ hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2),
+ (FIFO_U1 | ((j & FIFO_MASK) << 0xc)));
+#endif
+}
+#endif
+static void vortex_fifo_setadbvalid(vortex_t * vortex, int fifo, int en)
+{
+ hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2),
+ (hwread(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2)) &
+ 0xffffffef) | ((1 & en) << 4) | FIFO_U1);
+}
+
+static void
+vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int b, int priority,
+ int empty, int valid, int f)
+{
+ int temp, lifeboat = 0;
+ //int this_8[NR_ADB] = {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0}; /* position */
+ int this_4 = 0x2;
+ /* f seems priority related.
+ * CAsp4AdbDma::SetPriority is the only place that calls SetAdbCtrl with f set to 1
+ * every where else it is set to 0. It seems, however, that CAsp4AdbDma::SetPriority
+ * is never called, thus the f related bits remain a mystery for now.
+ */
+ do {
+ temp = hwread(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2));
+ if (lifeboat++ > 0xbb8) {
+ printk(KERN_ERR
+ "Vortex: vortex_fifo_setadbctrl fail\n");
+ break;
+ }
+ }
+ while (temp & FIFO_RDONLY);
+
+ // AU8830 semes to take some special care about fifo content (data).
+ // But i'm just to lazy to translate that :)
+ if (valid) {
+ if ((temp & FIFO_VALID) == 0) {
+ //this_8[fifo] = 0;
+ vortex_fifo_clearadbdata(vortex, fifo, FIFO_SIZE); // this_4
+#ifdef CHIP_AU8820
+ temp = (this_4 & 0x1f) << 0xb;
+#else
+ temp = (this_4 & 0x3f) << 0xc;
+#endif
+ temp = (temp & 0xfffffffd) | ((b & 1) << 1);
+ temp = (temp & 0xfffffff3) | ((priority & 3) << 2);
+ temp = (temp & 0xffffffef) | ((valid & 1) << 4);
+ temp |= FIFO_U1;
+ temp = (temp & 0xffffffdf) | ((empty & 1) << 5);
+#ifdef CHIP_AU8820
+ temp = (temp & 0xfffbffff) | ((f & 1) << 0x12);
+#endif
+#ifdef CHIP_AU8830
+ temp = (temp & 0xf7ffffff) | ((f & 1) << 0x1b);
+ temp = (temp & 0xefffffff) | ((f & 1) << 0x1c);
+#endif
+#ifdef CHIP_AU8810
+ temp = (temp & 0xfeffffff) | ((f & 1) << 0x18);
+ temp = (temp & 0xfdffffff) | ((f & 1) << 0x19);
+#endif
+ }
+ } else {
+ if (temp & FIFO_VALID) {
+#ifdef CHIP_AU8820
+ temp = ((f & 1) << 0x12) | (temp & 0xfffbffef);
+#endif
+#ifdef CHIP_AU8830
+ temp =
+ ((f & 1) << 0x1b) | (temp & 0xe7ffffef) | FIFO_BITS;
+#endif
+#ifdef CHIP_AU8810
+ temp =
+ ((f & 1) << 0x18) | (temp & 0xfcffffef) | FIFO_BITS;
+#endif
+ } else
+ /*if (this_8[fifo]) */
+ vortex_fifo_clearadbdata(vortex, fifo, FIFO_SIZE);
+ }
+ hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2), temp);
+ hwread(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2));
+}
+
+#ifndef CHIP_AU8810
+static void vortex_fifo_clearwtdata(vortex_t * vortex, int fifo, int x)
+{
+ if (x < 1)
+ return;
+ for (x--; x >= 0; x--)
+ hwwrite(vortex->mmio,
+ VORTEX_FIFO_WTDATA +
+ (((fifo << FIFO_SIZE_BITS) + x) << 2), 0);
+}
+
+static void vortex_fifo_wtinitialize(vortex_t * vortex, int fifo, int j)
+{
+ vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE);
+#ifdef CHIP_AU8820
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2),
+ (FIFO_U1 | ((j & FIFO_MASK) << 0xb)));
+#else
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2),
+ (FIFO_U1 | ((j & FIFO_MASK) << 0xc)));
+#endif
+}
+
+static void vortex_fifo_setwtvalid(vortex_t * vortex, int fifo, int en)
+{
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2),
+ (hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2)) &
+ 0xffffffef) | ((en & 1) << 4) | FIFO_U1);
+}
+
+static void
+vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority,
+ int empty, int valid, int f)
+{
+ int temp = 0, lifeboat = 0;
+ int this_4 = 2;
+
+ do {
+ temp = hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2));
+ if (lifeboat++ > 0xbb8) {
+ printk(KERN_ERR "Vortex: vortex_fifo_setwtctrl fail\n");
+ break;
+ }
+ }
+ while (temp & FIFO_RDONLY);
+
+ if (valid) {
+ if ((temp & FIFO_VALID) == 0) {
+ vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE); // this_4
+#ifdef CHIP_AU8820
+ temp = (this_4 & 0x1f) << 0xb;
+#else
+ temp = (this_4 & 0x3f) << 0xc;
+#endif
+ temp = (temp & 0xfffffffd) | ((ctrl & 1) << 1);
+ temp = (temp & 0xfffffff3) | ((priority & 3) << 2);
+ temp = (temp & 0xffffffef) | ((valid & 1) << 4);
+ temp |= FIFO_U1;
+ temp = (temp & 0xffffffdf) | ((empty & 1) << 5);
+#ifdef CHIP_AU8820
+ temp = (temp & 0xfffbffff) | ((f & 1) << 0x12);
+#endif
+#ifdef CHIP_AU8830
+ temp = (temp & 0xf7ffffff) | ((f & 1) << 0x1b);
+ temp = (temp & 0xefffffff) | ((f & 1) << 0x1c);
+#endif
+#ifdef CHIP_AU8810
+ temp = (temp & 0xfeffffff) | ((f & 1) << 0x18);
+ temp = (temp & 0xfdffffff) | ((f & 1) << 0x19);
+#endif
+ }
+ } else {
+ if (temp & FIFO_VALID) {
+#ifdef CHIP_AU8820
+ temp = ((f & 1) << 0x12) | (temp & 0xfffbffef);
+#endif
+#ifdef CHIP_AU8830
+ temp =
+ ((f & 1) << 0x1b) | (temp & 0xe7ffffef) | FIFO_BITS;
+#endif
+#ifdef CHIP_AU8810
+ temp =
+ ((f & 1) << 0x18) | (temp & 0xfcffffef) | FIFO_BITS;
+#endif
+ } else
+ /*if (this_8[fifo]) */
+ vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE);
+ }
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp);
+ hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2));
+
+/*
+ do {
+ temp = hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2));
+ if (lifeboat++ > 0xbb8) {
+ printk(KERN_ERR "Vortex: vortex_fifo_setwtctrl fail (hanging)\n");
+ break;
+ }
+ } while ((temp & FIFO_RDONLY)&&(temp & FIFO_VALID)&&(temp != 0xFFFFFFFF));
+
+
+ if (valid) {
+ if (temp & FIFO_VALID) {
+ temp = 0x40000;
+ //temp |= 0x08000000;
+ //temp |= 0x10000000;
+ //temp |= 0x04000000;
+ //temp |= 0x00400000;
+ temp |= 0x1c400000;
+ temp &= 0xFFFFFFF3;
+ temp &= 0xFFFFFFEF;
+ temp |= (valid & 1) << 4;
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp);
+ return;
+ } else {
+ vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE);
+ return;
+ }
+ } else {
+ temp &= 0xffffffef;
+ temp |= 0x08000000;
+ temp |= 0x10000000;
+ temp |= 0x04000000;
+ temp |= 0x00400000;
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp);
+ temp = hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2));
+ //((temp >> 6) & 0x3f)
+
+ priority = 0;
+ if (((temp & 0x0fc0) ^ ((temp >> 6) & 0x0fc0)) & 0FFFFFFC0)
+ vortex_fifo_clearwtdata(vortex, fifo, FIFO_SIZE);
+ valid = 0xfb;
+ temp = (temp & 0xfffffffd) | ((ctrl & 1) << 1);
+ temp = (temp & 0xfffdffff) | ((f & 1) << 0x11);
+ temp = (temp & 0xfffffff3) | ((priority & 3) << 2);
+ temp = (temp & 0xffffffef) | ((valid & 1) << 4);
+ temp = (temp & 0xffffffdf) | ((empty & 1) << 5);
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp);
+ }
+
+ */
+
+ /*
+ temp = (temp & 0xfffffffd) | ((ctrl & 1) << 1);
+ temp = (temp & 0xfffdffff) | ((f & 1) << 0x11);
+ temp = (temp & 0xfffffff3) | ((priority & 3) << 2);
+ temp = (temp & 0xffffffef) | ((valid & 1) << 4);
+ temp = (temp & 0xffffffdf) | ((empty & 1) << 5);
+ #ifdef FIFO_BITS
+ temp = temp | FIFO_BITS | 40000;
+ #endif
+ // 0x1c440010, 0x1c400000
+ hwwrite(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2), temp);
+ */
+}
+
+#endif
+static void vortex_fifo_init(vortex_t * vortex)
+{
+ int x;
+ u32 addr;
+
+ /* ADB DMA channels fifos. */
+ addr = VORTEX_FIFO_ADBCTRL + ((NR_ADB - 1) * 4);
+ for (x = NR_ADB - 1; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, (FIFO_U0 | FIFO_U1));
+ if (hwread(vortex->mmio, addr) != (FIFO_U0 | FIFO_U1))
+ printk(KERN_ERR "bad adb fifo reset!");
+ vortex_fifo_clearadbdata(vortex, x, FIFO_SIZE);
+ addr -= 4;
+ }
+
+#ifndef CHIP_AU8810
+ /* WT DMA channels fifos. */
+ addr = VORTEX_FIFO_WTCTRL + ((NR_WT - 1) * 4);
+ for (x = NR_WT - 1; x >= 0; x--) {
+ hwwrite(vortex->mmio, addr, FIFO_U0);
+ if (hwread(vortex->mmio, addr) != FIFO_U0)
+ printk(KERN_ERR
+ "bad wt fifo reset (0x%08x, 0x%08x)!\n",
+ addr, hwread(vortex->mmio, addr));
+ vortex_fifo_clearwtdata(vortex, x, FIFO_SIZE);
+ addr -= 4;
+ }
+#endif
+ /* trigger... */
+#ifdef CHIP_AU8820
+ hwwrite(vortex->mmio, 0xf8c0, 0xd03); //0x0843 0xd6b
+#else
+#ifdef CHIP_AU8830
+ hwwrite(vortex->mmio, 0x17000, 0x61); /* wt a */
+ hwwrite(vortex->mmio, 0x17004, 0x61); /* wt b */
+#endif
+ hwwrite(vortex->mmio, 0x17008, 0x61); /* adb */
+#endif
+}
+
+/* ADBDMA */
+
+static void vortex_adbdma_init(vortex_t * vortex)
+{
+}
+
+static void vortex_adbdma_setfirstbuffer(vortex_t * vortex, int adbdma)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_CTRL + (adbdma << 2),
+ dma->dma_ctrl);
+}
+
+static void vortex_adbdma_setstartbuffer(vortex_t * vortex, int adbdma, int sb)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+ //hwwrite(vortex->mmio, VORTEX_ADBDMA_START + (adbdma << 2), sb << (((NR_ADB-1)-((adbdma&0xf)*2))));
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_START + (adbdma << 2),
+ sb << ((0xf - (adbdma & 0xf)) * 2));
+ dma->period_real = dma->period_virt = sb;
+}
+
+static void
+vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
+ int psize, int count)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+
+ dma->period_bytes = psize;
+ dma->nr_periods = count;
+
+ dma->cfg0 = 0;
+ dma->cfg1 = 0;
+ switch (count) {
+ /* Four or more pages */
+ default:
+ case 4:
+ dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize - 1);
+ hwwrite(vortex->mmio,
+ VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0xc,
+ snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+ /* 3 pages */
+ case 3:
+ dma->cfg0 |= 0x12000000;
+ dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc);
+ hwwrite(vortex->mmio,
+ VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x8,
+ snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+ /* 2 pages */
+ case 2:
+ dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize - 1);
+ hwwrite(vortex->mmio,
+ VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x4,
+ snd_pcm_sgbuf_get_addr(dma->substream, psize));
+ /* 1 page */
+ case 1:
+ dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc);
+ hwwrite(vortex->mmio,
+ VORTEX_ADBDMA_BUFBASE + (adbdma << 4),
+ snd_pcm_sgbuf_get_addr(dma->substream, 0));
+ break;
+ }
+ //printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1);
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG0 + (adbdma << 3), dma->cfg0);
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG1 + (adbdma << 3), dma->cfg1);
+
+ vortex_adbdma_setfirstbuffer(vortex, adbdma);
+ vortex_adbdma_setstartbuffer(vortex, adbdma, 0);
+}
+
+static void
+vortex_adbdma_setmode(vortex_t * vortex, int adbdma, int ie, int dir,
+ int fmt, int d, u32 offset)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+
+ dma->dma_unknown = d;
+ dma->dma_ctrl =
+ ((offset & OFFSET_MASK) | (dma->dma_ctrl & ~OFFSET_MASK));
+ /* Enable PCMOUT interrupts. */
+ dma->dma_ctrl =
+ (dma->dma_ctrl & ~IE_MASK) | ((ie << IE_SHIFT) & IE_MASK);
+
+ dma->dma_ctrl =
+ (dma->dma_ctrl & ~DIR_MASK) | ((dir << DIR_SHIFT) & DIR_MASK);
+ dma->dma_ctrl =
+ (dma->dma_ctrl & ~FMT_MASK) | ((fmt << FMT_SHIFT) & FMT_MASK);
+
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_CTRL + (adbdma << 2),
+ dma->dma_ctrl);
+ hwread(vortex->mmio, VORTEX_ADBDMA_CTRL + (adbdma << 2));
+}
+
+static int vortex_adbdma_bufshift(vortex_t * vortex, int adbdma)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+ int page, p, pp, delta, i;
+
+ page =
+ (hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2)) &
+ ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+ if (dma->nr_periods >= 4)
+ delta = (page - dma->period_real) & 3;
+ else {
+ delta = (page - dma->period_real);
+ if (delta < 0)
+ delta += dma->nr_periods;
+ }
+ if (delta == 0)
+ return 0;
+
+ /* refresh hw page table */
+ if (dma->nr_periods > 4) {
+ for (i = 0; i < delta; i++) {
+ /* p: audio buffer page index */
+ p = dma->period_virt + i + 4;
+ if (p >= dma->nr_periods)
+ p -= dma->nr_periods;
+ /* pp: hardware DMA page index. */
+ pp = dma->period_real + i;
+ if (pp >= 4)
+ pp -= 4;
+ //hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFBASE+(((adbdma << 2)+pp) << 2), dma->table[p].addr);
+ hwwrite(vortex->mmio,
+ VORTEX_ADBDMA_BUFBASE + (((adbdma << 2) + pp) << 2),
+ snd_pcm_sgbuf_get_addr(dma->substream,
+ dma->period_bytes * p));
+ /* Force write thru cache. */
+ hwread(vortex->mmio, VORTEX_ADBDMA_BUFBASE +
+ (((adbdma << 2) + pp) << 2));
+ }
+ }
+ dma->period_virt += delta;
+ dma->period_real = page;
+ if (dma->period_virt >= dma->nr_periods)
+ dma->period_virt -= dma->nr_periods;
+ if (delta != 1)
+ printk(KERN_INFO "vortex: %d virt=%d, real=%d, delta=%d\n",
+ adbdma, dma->period_virt, dma->period_real, delta);
+
+ return delta;
+}
+
+
+static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
+ stream_t *dma = &vortex->dma_adb[adbdma];
+ int p, pp, i;
+
+ /* refresh hw page table */
+ for (i=0 ; i < 4 && i < dma->nr_periods; i++) {
+ /* p: audio buffer page index */
+ p = dma->period_virt + i;
+ if (p >= dma->nr_periods)
+ p -= dma->nr_periods;
+ /* pp: hardware DMA page index. */
+ pp = dma->period_real + i;
+ if (dma->nr_periods < 4) {
+ if (pp >= dma->nr_periods)
+ pp -= dma->nr_periods;
+ }
+ else {
+ if (pp >= 4)
+ pp -= 4;
+ }
+ hwwrite(vortex->mmio,
+ VORTEX_ADBDMA_BUFBASE + (((adbdma << 2) + pp) << 2),
+ snd_pcm_sgbuf_get_addr(dma->substream,
+ dma->period_bytes * p));
+ /* Force write thru cache. */
+ hwread(vortex->mmio, VORTEX_ADBDMA_BUFBASE + (((adbdma << 2)+pp) << 2));
+ }
+}
+
+static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+ int temp;
+
+ temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
+ temp = (dma->period_virt * dma->period_bytes) + (temp & POS_MASK);
+ return (temp);
+}
+
+static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
+{
+ int this_8 = 0 /*empty */ , this_4 = 0 /*priority */ ;
+ stream_t *dma = &vortex->dma_adb[adbdma];
+
+ switch (dma->fifo_status) {
+ case FIFO_START:
+ vortex_fifo_setadbvalid(vortex, adbdma,
+ dma->fifo_enabled ? 1 : 0);
+ break;
+ case FIFO_STOP:
+ this_8 = 1;
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_CTRL + (adbdma << 2),
+ dma->dma_ctrl);
+ vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ case FIFO_PAUSE:
+ vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ }
+ dma->fifo_status = FIFO_START;
+}
+
+static void vortex_adbdma_resumefifo(vortex_t * vortex, int adbdma)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+
+ int this_8 = 1, this_4 = 0;
+ switch (dma->fifo_status) {
+ case FIFO_STOP:
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_CTRL + (adbdma << 2),
+ dma->dma_ctrl);
+ vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ case FIFO_PAUSE:
+ vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ }
+ dma->fifo_status = FIFO_START;
+}
+
+static void vortex_adbdma_pausefifo(vortex_t * vortex, int adbdma)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+
+ int this_8 = 0, this_4 = 0;
+ switch (dma->fifo_status) {
+ case FIFO_START:
+ vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown,
+ this_4, this_8, 0, 0);
+ break;
+ case FIFO_STOP:
+ hwwrite(vortex->mmio, VORTEX_ADBDMA_CTRL + (adbdma << 2),
+ dma->dma_ctrl);
+ vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown,
+ this_4, this_8, 0, 0);
+ break;
+ }
+ dma->fifo_status = FIFO_PAUSE;
+}
+
+#if 0 // Using pause instead
+static void vortex_adbdma_stopfifo(vortex_t * vortex, int adbdma)
+{
+ stream_t *dma = &vortex->dma_adb[adbdma];
+
+ int this_4 = 0, this_8 = 0;
+ if (dma->fifo_status == FIFO_START)
+ vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown,
+ this_4, this_8, 0, 0);
+ else if (dma->fifo_status == FIFO_STOP)
+ return;
+ dma->fifo_status = FIFO_STOP;
+ dma->fifo_enabled = 0;
+}
+
+#endif
+/* WTDMA */
+
+#ifndef CHIP_AU8810
+static void vortex_wtdma_setfirstbuffer(vortex_t * vortex, int wtdma)
+{
+ //int this_7c=dma_ctrl;
+ stream_t *dma = &vortex->dma_wt[wtdma];
+
+ hwwrite(vortex->mmio, VORTEX_WTDMA_CTRL + (wtdma << 2), dma->dma_ctrl);
+}
+
+static void vortex_wtdma_setstartbuffer(vortex_t * vortex, int wtdma, int sb)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+ //hwwrite(vortex->mmio, VORTEX_WTDMA_START + (wtdma << 2), sb << ((0x1f-(wtdma&0xf)*2)));
+ hwwrite(vortex->mmio, VORTEX_WTDMA_START + (wtdma << 2),
+ sb << ((0xf - (wtdma & 0xf)) * 2));
+ dma->period_real = dma->period_virt = sb;
+}
+
+static void
+vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma,
+ int psize, int count)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+
+ dma->period_bytes = psize;
+ dma->nr_periods = count;
+
+ dma->cfg0 = 0;
+ dma->cfg1 = 0;
+ switch (count) {
+ /* Four or more pages */
+ default:
+ case 4:
+ dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize-1);
+ hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0xc,
+ snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
+ /* 3 pages */
+ case 3:
+ dma->cfg0 |= 0x12000000;
+ dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
+ hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x8,
+ snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
+ /* 2 pages */
+ case 2:
+ dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize-1);
+ hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x4,
+ snd_pcm_sgbuf_get_addr(dma->substream, psize));
+ /* 1 page */
+ case 1:
+ dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
+ hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4),
+ snd_pcm_sgbuf_get_addr(dma->substream, 0));
+ break;
+ }
+ hwwrite(vortex->mmio, VORTEX_WTDMA_BUFCFG0 + (wtdma << 3), dma->cfg0);
+ hwwrite(vortex->mmio, VORTEX_WTDMA_BUFCFG1 + (wtdma << 3), dma->cfg1);
+
+ vortex_wtdma_setfirstbuffer(vortex, wtdma);
+ vortex_wtdma_setstartbuffer(vortex, wtdma, 0);
+}
+
+static void
+vortex_wtdma_setmode(vortex_t * vortex, int wtdma, int ie, int fmt, int d,
+ /*int e, */ u32 offset)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+
+ //dma->this_08 = e;
+ dma->dma_unknown = d;
+ dma->dma_ctrl = 0;
+ dma->dma_ctrl =
+ ((offset & OFFSET_MASK) | (dma->dma_ctrl & ~OFFSET_MASK));
+ /* PCMOUT interrupt */
+ dma->dma_ctrl =
+ (dma->dma_ctrl & ~IE_MASK) | ((ie << IE_SHIFT) & IE_MASK);
+ /* Always playback. */
+ dma->dma_ctrl |= (1 << DIR_SHIFT);
+ /* Audio Format */
+ dma->dma_ctrl =
+ (dma->dma_ctrl & FMT_MASK) | ((fmt << FMT_SHIFT) & FMT_MASK);
+ /* Write into hardware */
+ hwwrite(vortex->mmio, VORTEX_WTDMA_CTRL + (wtdma << 2), dma->dma_ctrl);
+}
+
+static int vortex_wtdma_bufshift(vortex_t * vortex, int wtdma)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+ int page, p, pp, delta, i;
+
+ page =
+ (hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) &
+ WT_SUBBUF_MASK)
+ >> WT_SUBBUF_SHIFT;
+ if (dma->nr_periods >= 4)
+ delta = (page - dma->period_real) & 3;
+ else {
+ delta = (page - dma->period_real);
+ if (delta < 0)
+ delta += dma->nr_periods;
+ }
+ if (delta == 0)
+ return 0;
+
+ /* refresh hw page table */
+ if (dma->nr_periods > 4) {
+ for (i = 0; i < delta; i++) {
+ /* p: audio buffer page index */
+ p = dma->period_virt + i + 4;
+ if (p >= dma->nr_periods)
+ p -= dma->nr_periods;
+ /* pp: hardware DMA page index. */
+ pp = dma->period_real + i;
+ if (pp >= 4)
+ pp -= 4;
+ hwwrite(vortex->mmio,
+ VORTEX_WTDMA_BUFBASE +
+ (((wtdma << 2) + pp) << 2),
+ snd_pcm_sgbuf_get_addr(dma->substream,
+ dma->period_bytes * p));
+ /* Force write thru cache. */
+ hwread(vortex->mmio, VORTEX_WTDMA_BUFBASE +
+ (((wtdma << 2) + pp) << 2));
+ }
+ }
+ dma->period_virt += delta;
+ if (dma->period_virt >= dma->nr_periods)
+ dma->period_virt -= dma->nr_periods;
+ dma->period_real = page;
+
+ if (delta != 1)
+ printk(KERN_WARNING "vortex: wt virt = %d, delta = %d\n",
+ dma->period_virt, delta);
+
+ return delta;
+}
+
+#if 0
+static void
+vortex_wtdma_getposition(vortex_t * vortex, int wtdma, int *subbuf, int *pos)
+{
+ int temp;
+ temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2));
+ *subbuf = (temp >> WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK;
+ *pos = temp & POS_MASK;
+}
+
+static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma)
+{
+ return ((hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) >>
+ POS_SHIFT) & POS_MASK);
+}
+#endif
+static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+ int temp;
+
+ temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2));
+ //temp = (temp & POS_MASK) + (((temp>>WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK)*(dma->cfg0&POS_MASK));
+ temp = (temp & POS_MASK) + ((dma->period_virt) * (dma->period_bytes));
+ return temp;
+}
+
+static void vortex_wtdma_startfifo(vortex_t * vortex, int wtdma)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+ int this_8 = 0, this_4 = 0;
+
+ switch (dma->fifo_status) {
+ case FIFO_START:
+ vortex_fifo_setwtvalid(vortex, wtdma,
+ dma->fifo_enabled ? 1 : 0);
+ break;
+ case FIFO_STOP:
+ this_8 = 1;
+ hwwrite(vortex->mmio, VORTEX_WTDMA_CTRL + (wtdma << 2),
+ dma->dma_ctrl);
+ vortex_fifo_setwtctrl(vortex, wtdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ case FIFO_PAUSE:
+ vortex_fifo_setwtctrl(vortex, wtdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ }
+ dma->fifo_status = FIFO_START;
+}
+
+static void vortex_wtdma_resumefifo(vortex_t * vortex, int wtdma)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+
+ int this_8 = 0, this_4 = 0;
+ switch (dma->fifo_status) {
+ case FIFO_STOP:
+ hwwrite(vortex->mmio, VORTEX_WTDMA_CTRL + (wtdma << 2),
+ dma->dma_ctrl);
+ vortex_fifo_setwtctrl(vortex, wtdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ case FIFO_PAUSE:
+ vortex_fifo_setwtctrl(vortex, wtdma, dma->dma_unknown,
+ this_4, this_8,
+ dma->fifo_enabled ? 1 : 0, 0);
+ break;
+ }
+ dma->fifo_status = FIFO_START;
+}
+
+static void vortex_wtdma_pausefifo(vortex_t * vortex, int wtdma)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+
+ int this_8 = 0, this_4 = 0;
+ switch (dma->fifo_status) {
+ case FIFO_START:
+ vortex_fifo_setwtctrl(vortex, wtdma, dma->dma_unknown,
+ this_4, this_8, 0, 0);
+ break;
+ case FIFO_STOP:
+ hwwrite(vortex->mmio, VORTEX_WTDMA_CTRL + (wtdma << 2),
+ dma->dma_ctrl);
+ vortex_fifo_setwtctrl(vortex, wtdma, dma->dma_unknown,
+ this_4, this_8, 0, 0);
+ break;
+ }
+ dma->fifo_status = FIFO_PAUSE;
+}
+
+static void vortex_wtdma_stopfifo(vortex_t * vortex, int wtdma)
+{
+ stream_t *dma = &vortex->dma_wt[wtdma];
+
+ int this_4 = 0, this_8 = 0;
+ if (dma->fifo_status == FIFO_START)
+ vortex_fifo_setwtctrl(vortex, wtdma, dma->dma_unknown,
+ this_4, this_8, 0, 0);
+ else if (dma->fifo_status == FIFO_STOP)
+ return;
+ dma->fifo_status = FIFO_STOP;
+ dma->fifo_enabled = 0;
+}
+
+#endif
+/* ADB Routes */
+
+typedef int ADBRamLink;
+static void vortex_adb_init(vortex_t * vortex)
+{
+ int i;
+ /* it looks like we are writing more than we need to...
+ * if we write what we are supposed to it breaks things... */
+ hwwrite(vortex->mmio, VORTEX_ADB_SR, 0);
+ for (i = 0; i < VORTEX_ADB_RTBASE_COUNT; i++)
+ hwwrite(vortex->mmio, VORTEX_ADB_RTBASE + (i << 2),
+ hwread(vortex->mmio,
+ VORTEX_ADB_RTBASE + (i << 2)) | ROUTE_MASK);
+ for (i = 0; i < VORTEX_ADB_CHNBASE_COUNT; i++) {
+ hwwrite(vortex->mmio, VORTEX_ADB_CHNBASE + (i << 2),
+ hwread(vortex->mmio,
+ VORTEX_ADB_CHNBASE + (i << 2)) | ROUTE_MASK);
+ }
+}
+
+static void vortex_adb_en_sr(vortex_t * vortex, int channel)
+{
+ hwwrite(vortex->mmio, VORTEX_ADB_SR,
+ hwread(vortex->mmio, VORTEX_ADB_SR) | (0x1 << channel));
+}
+
+static void vortex_adb_dis_sr(vortex_t * vortex, int channel)
+{
+ hwwrite(vortex->mmio, VORTEX_ADB_SR,
+ hwread(vortex->mmio, VORTEX_ADB_SR) & ~(0x1 << channel));
+}
+
+static void
+vortex_adb_addroutes(vortex_t * vortex, unsigned char channel,
+ ADBRamLink * route, int rnum)
+{
+ int temp, prev, lifeboat = 0;
+
+ if ((rnum <= 0) || (route == NULL))
+ return;
+ /* Write last routes. */
+ rnum--;
+ hwwrite(vortex->mmio,
+ VORTEX_ADB_RTBASE + ((route[rnum] & ADB_MASK) << 2),
+ ROUTE_MASK);
+ while (rnum > 0) {
+ hwwrite(vortex->mmio,
+ VORTEX_ADB_RTBASE +
+ ((route[rnum - 1] & ADB_MASK) << 2), route[rnum]);
+ rnum--;
+ }
+ /* Write first route. */
+ temp =
+ hwread(vortex->mmio,
+ VORTEX_ADB_CHNBASE + (channel << 2)) & ADB_MASK;
+ if (temp == ADB_MASK) {
+ /* First entry on this channel. */
+ hwwrite(vortex->mmio, VORTEX_ADB_CHNBASE + (channel << 2),
+ route[0]);
+ vortex_adb_en_sr(vortex, channel);
+ return;
+ }
+ /* Not first entry on this channel. Need to link. */
+ do {
+ prev = temp;
+ temp =
+ hwread(vortex->mmio,
+ VORTEX_ADB_RTBASE + (temp << 2)) & ADB_MASK;
+ if ((lifeboat++) > ADB_MASK) {
+ printk(KERN_ERR
+ "vortex_adb_addroutes: unending route! 0x%x\n",
+ *route);
+ return;
+ }
+ }
+ while (temp != ADB_MASK);
+ hwwrite(vortex->mmio, VORTEX_ADB_RTBASE + (prev << 2), route[0]);
+}
+
+static void
+vortex_adb_delroutes(vortex_t * vortex, unsigned char channel,
+ ADBRamLink route0, ADBRamLink route1)
+{
+ int temp, lifeboat = 0, prev;
+
+ /* Find route. */
+ temp =
+ hwread(vortex->mmio,
+ VORTEX_ADB_CHNBASE + (channel << 2)) & ADB_MASK;
+ if (temp == (route0 & ADB_MASK)) {
+ temp =
+ hwread(vortex->mmio,
+ VORTEX_ADB_RTBASE + ((route1 & ADB_MASK) << 2));
+ if ((temp & ADB_MASK) == ADB_MASK)
+ vortex_adb_dis_sr(vortex, channel);
+ hwwrite(vortex->mmio, VORTEX_ADB_CHNBASE + (channel << 2),
+ temp);
+ return;
+ }
+ do {
+ prev = temp;
+ temp =
+ hwread(vortex->mmio,
+ VORTEX_ADB_RTBASE + (prev << 2)) & ADB_MASK;
+ if (((lifeboat++) > ADB_MASK) || (temp == ADB_MASK)) {
+ printk(KERN_ERR
+ "vortex_adb_delroutes: route not found! 0x%x\n",
+ route0);
+ return;
+ }
+ }
+ while (temp != (route0 & ADB_MASK));
+ temp = hwread(vortex->mmio, VORTEX_ADB_RTBASE + (temp << 2));
+ if ((temp & ADB_MASK) == route1)
+ temp = hwread(vortex->mmio, VORTEX_ADB_RTBASE + (temp << 2));
+ /* Make bridge over deleted route. */
+ hwwrite(vortex->mmio, VORTEX_ADB_RTBASE + (prev << 2), temp);
+}
+
+static void
+vortex_route(vortex_t * vortex, int en, unsigned char channel,
+ unsigned char source, unsigned char dest)
+{
+ ADBRamLink route;
+
+ route = ((source & ADB_MASK) << ADB_SHIFT) | (dest & ADB_MASK);
+ if (en) {
+ vortex_adb_addroutes(vortex, channel, &route, 1);
+ if ((source < (OFFSET_SRCOUT + NR_SRC))
+ && (source >= OFFSET_SRCOUT))
+ vortex_src_addWTD(vortex, (source - OFFSET_SRCOUT),
+ channel);
+ else if ((source < (OFFSET_MIXOUT + NR_MIXOUT))
+ && (source >= OFFSET_MIXOUT))
+ vortex_mixer_addWTD(vortex,
+ (source - OFFSET_MIXOUT), channel);
+ } else {
+ vortex_adb_delroutes(vortex, channel, route, route);
+ if ((source < (OFFSET_SRCOUT + NR_SRC))
+ && (source >= OFFSET_SRCOUT))
+ vortex_src_delWTD(vortex, (source - OFFSET_SRCOUT),
+ channel);
+ else if ((source < (OFFSET_MIXOUT + NR_MIXOUT))
+ && (source >= OFFSET_MIXOUT))
+ vortex_mixer_delWTD(vortex,
+ (source - OFFSET_MIXOUT), channel);
+ }
+}
+
+#if 0
+static void
+vortex_routes(vortex_t * vortex, int en, unsigned char channel,
+ unsigned char source, unsigned char dest0, unsigned char dest1)
+{
+ ADBRamLink route[2];
+
+ route[0] = ((source & ADB_MASK) << ADB_SHIFT) | (dest0 & ADB_MASK);
+ route[1] = ((source & ADB_MASK) << ADB_SHIFT) | (dest1 & ADB_MASK);
+
+ if (en) {
+ vortex_adb_addroutes(vortex, channel, route, 2);
+ if ((source < (OFFSET_SRCOUT + NR_SRC))
+ && (source >= (OFFSET_SRCOUT)))
+ vortex_src_addWTD(vortex, (source - OFFSET_SRCOUT),
+ channel);
+ else if ((source < (OFFSET_MIXOUT + NR_MIXOUT))
+ && (source >= (OFFSET_MIXOUT)))
+ vortex_mixer_addWTD(vortex,
+ (source - OFFSET_MIXOUT), channel);
+ } else {
+ vortex_adb_delroutes(vortex, channel, route[0], route[1]);
+ if ((source < (OFFSET_SRCOUT + NR_SRC))
+ && (source >= (OFFSET_SRCOUT)))
+ vortex_src_delWTD(vortex, (source - OFFSET_SRCOUT),
+ channel);
+ else if ((source < (OFFSET_MIXOUT + NR_MIXOUT))
+ && (source >= (OFFSET_MIXOUT)))
+ vortex_mixer_delWTD(vortex,
+ (source - OFFSET_MIXOUT), channel);
+ }
+}
+
+#endif
+/* Route two sources to same target. Sources must be of same class !!! */
+static void
+vortex_routeLRT(vortex_t * vortex, int en, unsigned char ch,
+ unsigned char source0, unsigned char source1,
+ unsigned char dest)
+{
+ ADBRamLink route[2];
+
+ route[0] = ((source0 & ADB_MASK) << ADB_SHIFT) | (dest & ADB_MASK);
+ route[1] = ((source1 & ADB_MASK) << ADB_SHIFT) | (dest & ADB_MASK);
+
+ if (dest < 0x10)
+ route[1] = (route[1] & ~ADB_MASK) | (dest + 0x20); /* fifo A */
+
+ if (en) {
+ vortex_adb_addroutes(vortex, ch, route, 2);
+ if ((source0 < (OFFSET_SRCOUT + NR_SRC))
+ && (source0 >= OFFSET_SRCOUT)) {
+ vortex_src_addWTD(vortex,
+ (source0 - OFFSET_SRCOUT), ch);
+ vortex_src_addWTD(vortex,
+ (source1 - OFFSET_SRCOUT), ch);
+ } else if ((source0 < (OFFSET_MIXOUT + NR_MIXOUT))
+ && (source0 >= OFFSET_MIXOUT)) {
+ vortex_mixer_addWTD(vortex,
+ (source0 - OFFSET_MIXOUT), ch);
+ vortex_mixer_addWTD(vortex,
+ (source1 - OFFSET_MIXOUT), ch);
+ }
+ } else {
+ vortex_adb_delroutes(vortex, ch, route[0], route[1]);
+ if ((source0 < (OFFSET_SRCOUT + NR_SRC))
+ && (source0 >= OFFSET_SRCOUT)) {
+ vortex_src_delWTD(vortex,
+ (source0 - OFFSET_SRCOUT), ch);
+ vortex_src_delWTD(vortex,
+ (source1 - OFFSET_SRCOUT), ch);
+ } else if ((source0 < (OFFSET_MIXOUT + NR_MIXOUT))
+ && (source0 >= OFFSET_MIXOUT)) {
+ vortex_mixer_delWTD(vortex,
+ (source0 - OFFSET_MIXOUT), ch);
+ vortex_mixer_delWTD(vortex,
+ (source1 - OFFSET_MIXOUT), ch);
+ }
+ }
+}
+
+/* Connection stuff */
+
+// Connect adbdma to src('s).
+static void
+vortex_connection_adbdma_src(vortex_t * vortex, int en, unsigned char ch,
+ unsigned char adbdma, unsigned char src)
+{
+ vortex_route(vortex, en, ch, ADB_DMA(adbdma), ADB_SRCIN(src));
+}
+
+// Connect SRC to mixin.
+static void
+vortex_connection_src_mixin(vortex_t * vortex, int en,
+ unsigned char channel, unsigned char src,
+ unsigned char mixin)
+{
+ vortex_route(vortex, en, channel, ADB_SRCOUT(src), ADB_MIXIN(mixin));
+}
+
+// Connect mixin with mix output.
+static void
+vortex_connection_mixin_mix(vortex_t * vortex, int en, unsigned char mixin,
+ unsigned char mix, int a)
+{
+ if (en) {
+ vortex_mix_enableinput(vortex, mix, mixin);
+ vortex_mix_setinputvolumebyte(vortex, mix, mixin, MIX_DEFIGAIN); // added to original code.
+ } else
+ vortex_mix_disableinput(vortex, mix, mixin, a);
+}
+
+// Connect absolut address to mixin.
+static void
+vortex_connection_adb_mixin(vortex_t * vortex, int en,
+ unsigned char channel, unsigned char source,
+ unsigned char mixin)
+{
+ vortex_route(vortex, en, channel, source, ADB_MIXIN(mixin));
+}
+
+static void
+vortex_connection_src_adbdma(vortex_t * vortex, int en, unsigned char ch,
+ unsigned char src, unsigned char adbdma)
+{
+ vortex_route(vortex, en, ch, ADB_SRCOUT(src), ADB_DMA(adbdma));
+}
+
+static void
+vortex_connection_src_src_adbdma(vortex_t * vortex, int en,
+ unsigned char ch, unsigned char src0,
+ unsigned char src1, unsigned char adbdma)
+{
+
+ vortex_routeLRT(vortex, en, ch, ADB_SRCOUT(src0), ADB_SRCOUT(src1),
+ ADB_DMA(adbdma));
+}
+
+// mix to absolut address.
+static void
+vortex_connection_mix_adb(vortex_t * vortex, int en, unsigned char ch,
+ unsigned char mix, unsigned char dest)
+{
+ vortex_route(vortex, en, ch, ADB_MIXOUT(mix), dest);
+ vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); // added to original code.
+}
+
+// mixer to src.
+static void
+vortex_connection_mix_src(vortex_t * vortex, int en, unsigned char ch,
+ unsigned char mix, unsigned char src)
+{
+ vortex_route(vortex, en, ch, ADB_MIXOUT(mix), ADB_SRCIN(src));
+ vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); // added to original code.
+}
+
+#if 0
+static void
+vortex_connection_adbdma_src_src(vortex_t * vortex, int en,
+ unsigned char channel,
+ unsigned char adbdma, unsigned char src0,
+ unsigned char src1)
+{
+ vortex_routes(vortex, en, channel, ADB_DMA(adbdma),
+ ADB_SRCIN(src0), ADB_SRCIN(src1));
+}
+
+// Connect two mix to AdbDma.
+static void
+vortex_connection_mix_mix_adbdma(vortex_t * vortex, int en,
+ unsigned char ch, unsigned char mix0,
+ unsigned char mix1, unsigned char adbdma)
+{
+
+ ADBRamLink routes[2];
+ routes[0] =
+ (((mix0 +
+ OFFSET_MIXOUT) & ADB_MASK) << ADB_SHIFT) | (adbdma & ADB_MASK);
+ routes[1] =
+ (((mix1 + OFFSET_MIXOUT) & ADB_MASK) << ADB_SHIFT) | ((adbdma +
+ 0x20) &
+ ADB_MASK);
+ if (en) {
+ vortex_adb_addroutes(vortex, ch, routes, 0x2);
+ vortex_mixer_addWTD(vortex, mix0, ch);
+ vortex_mixer_addWTD(vortex, mix1, ch);
+ } else {
+ vortex_adb_delroutes(vortex, ch, routes[0], routes[1]);
+ vortex_mixer_delWTD(vortex, mix0, ch);
+ vortex_mixer_delWTD(vortex, mix1, ch);
+ }
+}
+#endif
+
+/* CODEC connect. */
+
+static void
+vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[])
+{
+#ifdef CHIP_AU8820
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_CODECOUT(0));
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1));
+#else
+#if 1
+ // Connect front channels through EQ.
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_EQIN(0));
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_EQIN(1));
+ /* Lower volume, since EQ has some gain. */
+ vortex_mix_setvolumebyte(vortex, mixers[0], 0);
+ vortex_mix_setvolumebyte(vortex, mixers[1], 0);
+ vortex_route(vortex, en, 0x11, ADB_EQOUT(0), ADB_CODECOUT(0));
+ vortex_route(vortex, en, 0x11, ADB_EQOUT(1), ADB_CODECOUT(1));
+
+ /* Check if reg 0x28 has SDAC bit set. */
+ if (VORTEX_IS_QUAD(vortex)) {
+ /* Rear channel. Note: ADB_CODECOUT(0+2) and (1+2) is for AC97 modem */
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[2],
+ ADB_CODECOUT(0 + 4));
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[3],
+ ADB_CODECOUT(1 + 4));
+ //printk("SDAC detected ");
+ }
+#else
+ // Use plain direct output to codec.
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_CODECOUT(0));
+ vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1));
+#endif
+#endif
+}
+
+static void
+vortex_connect_codecrec(vortex_t * vortex, int en, unsigned char mixin0,
+ unsigned char mixin1)
+{
+ /*
+ Enable: 0x1, 0x1
+ Channel: 0x11, 0x11
+ ADB Source address: 0x48, 0x49
+ Destination Asp4Topology_0x9c,0x98
+ */
+ vortex_connection_adb_mixin(vortex, en, 0x11, ADB_CODECIN(0), mixin0);
+ vortex_connection_adb_mixin(vortex, en, 0x11, ADB_CODECIN(1), mixin1);
+}
+
+// Higher level ADB audio path (de)allocator.
+
+/* Resource manager */
+static int resnum[VORTEX_RESOURCE_LAST] =
+ { NR_ADB, NR_SRC, NR_MIXIN, NR_MIXOUT, NR_A3D };
+/*
+ Checkout/Checkin resource of given type.
+ resmap: resource map to be used. If NULL means that we want to allocate
+ a DMA resource (root of all other resources of a dma channel).
+ out: Mean checkout if != 0. Else mean Checkin resource.
+ restype: Indicates type of resource to be checked in or out.
+*/
+static char
+vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
+{
+ int i, qty = resnum[restype], resinuse = 0;
+
+ if (out) {
+ /* Gather used resources by all streams. */
+ for (i = 0; i < NR_ADB; i++) {
+ resinuse |= vortex->dma_adb[i].resources[restype];
+ }
+ resinuse |= vortex->fixed_res[restype];
+ /* Find and take free resource. */
+ for (i = 0; i < qty; i++) {
+ if ((resinuse & (1 << i)) == 0) {
+ if (resmap != NULL)
+ resmap[restype] |= (1 << i);
+ else
+ vortex->dma_adb[i].resources[restype] |= (1 << i);
+ //printk("vortex: ResManager: type %d out %d\n", restype, i);
+ return i;
+ }
+ }
+ } else {
+ if (resmap == NULL)
+ return -EINVAL;
+ /* Checkin first resource of type restype. */
+ for (i = 0; i < qty; i++) {
+ if (resmap[restype] & (1 << i)) {
+ resmap[restype] &= ~(1 << i);
+ //printk("vortex: ResManager: type %d in %d\n",restype, i);
+ return i;
+ }
+ }
+ }
+ printk(KERN_ERR "vortex: FATAL: ResManager: resource type %d exhausted.\n", restype);
+ return -ENOMEM;
+}
+
+/* Default Connections */
+static int
+vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type);
+
+static void vortex_connect_default(vortex_t * vortex, int en)
+{
+ // Connect AC97 codec.
+ vortex->mixplayb[0] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXOUT);
+ vortex->mixplayb[1] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXOUT);
+ if (VORTEX_IS_QUAD(vortex)) {
+ vortex->mixplayb[2] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXOUT);
+ vortex->mixplayb[3] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXOUT);
+ }
+ vortex_connect_codecplay(vortex, en, vortex->mixplayb);
+
+ vortex->mixcapt[0] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXIN);
+ vortex->mixcapt[1] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXIN);
+ vortex_connect_codecrec(vortex, en, MIX_CAPT(0), MIX_CAPT(1));
+
+ // Connect SPDIF
+#ifndef CHIP_AU8820
+ vortex->mixspdif[0] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXOUT);
+ vortex->mixspdif[1] = vortex_adb_checkinout(vortex, vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXOUT);
+ vortex_connection_mix_adb(vortex, en, 0x14, vortex->mixspdif[0],
+ ADB_SPDIFOUT(0));
+ vortex_connection_mix_adb(vortex, en, 0x14, vortex->mixspdif[1],
+ ADB_SPDIFOUT(1));
+#endif
+ // Connect WT
+#ifndef CHIP_AU8810
+ vortex_wt_connect(vortex, en);
+#endif
+ // A3D (crosstalk canceler and A3D slices). AU8810 disabled for now.
+#ifndef CHIP_AU8820
+ vortex_Vort3D_connect(vortex, en);
+#endif
+ // Connect I2S
+
+ // Connect DSP interface for SQ3500 turbo (not here i think...)
+
+ // Connect AC98 modem codec
+
+}
+
+/*
+ Allocate nr_ch pcm audio routes if dma < 0. If dma >= 0, existing routes
+ are deallocated.
+ dma: DMA engine routes to be deallocated when dma >= 0.
+ nr_ch: Number of channels to be de/allocated.
+ dir: direction of stream. Uses same values as substream->stream.
+ type: Type of audio output/source (codec, spdif, i2s, dsp, etc)
+ Return: Return allocated DMA or same DMA passed as "dma" when dma >= 0.
+*/
+static int
+vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type)
+{
+ stream_t *stream;
+ int i, en;
+
+ if ((nr_ch == 3)
+ || ((dir == SNDRV_PCM_STREAM_CAPTURE) && (nr_ch > 2)))
+ return -EBUSY;
+
+ if (dma >= 0) {
+ en = 0;
+ vortex_adb_checkinout(vortex,
+ vortex->dma_adb[dma].resources, en,
+ VORTEX_RESOURCE_DMA);
+ } else {
+ en = 1;
+ if ((dma =
+ vortex_adb_checkinout(vortex, NULL, en,
+ VORTEX_RESOURCE_DMA)) < 0)
+ return -EBUSY;
+ }
+
+ stream = &vortex->dma_adb[dma];
+ stream->dma = dma;
+ stream->dir = dir;
+ stream->type = type;
+
+ /* PLAYBACK ROUTES. */
+ if (dir == SNDRV_PCM_STREAM_PLAYBACK) {
+ int src[4], mix[4], ch_top;
+#ifndef CHIP_AU8820
+ int a3d = 0;
+#endif
+ /* Get SRC and MIXER hardware resources. */
+ if (stream->type != VORTEX_PCM_SPDIF) {
+ for (i = 0; i < nr_ch; i++) {
+ if ((src[i] = vortex_adb_checkinout(vortex,
+ stream->resources, en,
+ VORTEX_RESOURCE_SRC)) < 0) {
+ memset(stream->resources, 0,
+ sizeof(unsigned char) *
+ VORTEX_RESOURCE_LAST);
+ return -EBUSY;
+ }
+ if (stream->type != VORTEX_PCM_A3D) {
+ if ((mix[i] = vortex_adb_checkinout(vortex,
+ stream->resources,
+ en,
+ VORTEX_RESOURCE_MIXIN)) < 0) {
+ memset(stream->resources,
+ 0,
+ sizeof(unsigned char) * VORTEX_RESOURCE_LAST);
+ return -EBUSY;
+ }
+ }
+ }
+ }
+#ifndef CHIP_AU8820
+ if (stream->type == VORTEX_PCM_A3D) {
+ if ((a3d =
+ vortex_adb_checkinout(vortex,
+ stream->resources, en,
+ VORTEX_RESOURCE_A3D)) < 0) {
+ memset(stream->resources, 0,
+ sizeof(unsigned char) *
+ VORTEX_RESOURCE_LAST);
+ printk(KERN_ERR "vortex: out of A3D sources. Sorry\n");
+ return -EBUSY;
+ }
+ /* (De)Initialize A3D hardware source. */
+ vortex_Vort3D_InitializeSource(&(vortex->a3d[a3d]), en);
+ }
+ /* Make SPDIF out exclusive to "spdif" device when in use. */
+ if ((stream->type == VORTEX_PCM_SPDIF) && (en)) {
+ vortex_route(vortex, 0, 0x14,
+ ADB_MIXOUT(vortex->mixspdif[0]),
+ ADB_SPDIFOUT(0));
+ vortex_route(vortex, 0, 0x14,
+ ADB_MIXOUT(vortex->mixspdif[1]),
+ ADB_SPDIFOUT(1));
+ }
+#endif
+ /* Make playback routes. */
+ for (i = 0; i < nr_ch; i++) {
+ if (stream->type == VORTEX_PCM_ADB) {
+ vortex_connection_adbdma_src(vortex, en,
+ src[nr_ch - 1],
+ dma,
+ src[i]);
+ vortex_connection_src_mixin(vortex, en,
+ 0x11, src[i],
+ mix[i]);
+ vortex_connection_mixin_mix(vortex, en,
+ mix[i],
+ MIX_PLAYB(i), 0);
+#ifndef CHIP_AU8820
+ vortex_connection_mixin_mix(vortex, en,
+ mix[i],
+ MIX_SPDIF(i % 2), 0);
+ vortex_mix_setinputvolumebyte(vortex,
+ MIX_SPDIF(i % 2),
+ mix[i],
+ MIX_DEFIGAIN);
+#endif
+ }
+#ifndef CHIP_AU8820
+ if (stream->type == VORTEX_PCM_A3D) {
+ vortex_connection_adbdma_src(vortex, en,
+ src[nr_ch - 1],
+ dma,
+ src[i]);
+ vortex_route(vortex, en, 0x11, ADB_SRCOUT(src[i]), ADB_A3DIN(a3d));
+ /* XTalk test. */
+ //vortex_route(vortex, en, 0x11, dma, ADB_XTALKIN(i?9:4));
+ //vortex_route(vortex, en, 0x11, ADB_SRCOUT(src[i]), ADB_XTALKIN(i?4:9));
+ }
+ if (stream->type == VORTEX_PCM_SPDIF)
+ vortex_route(vortex, en, 0x14,
+ ADB_DMA(stream->dma),
+ ADB_SPDIFOUT(i));
+#endif
+ }
+ if (stream->type != VORTEX_PCM_SPDIF && stream->type != VORTEX_PCM_A3D) {
+ ch_top = (VORTEX_IS_QUAD(vortex) ? 4 : 2);
+ for (i = nr_ch; i < ch_top; i++) {
+ vortex_connection_mixin_mix(vortex, en,
+ mix[i % nr_ch],
+ MIX_PLAYB(i), 0);
+#ifndef CHIP_AU8820
+ vortex_connection_mixin_mix(vortex, en,
+ mix[i % nr_ch],
+ MIX_SPDIF(i % 2),
+ 0);
+ vortex_mix_setinputvolumebyte(vortex,
+ MIX_SPDIF(i % 2),
+ mix[i % nr_ch],
+ MIX_DEFIGAIN);
+#endif
+ }
+ }
+#ifndef CHIP_AU8820
+ else {
+ if (nr_ch == 1 && stream->type == VORTEX_PCM_SPDIF)
+ vortex_route(vortex, en, 0x14,
+ ADB_DMA(stream->dma),
+ ADB_SPDIFOUT(1));
+ }
+ /* Reconnect SPDIF out when "spdif" device is down. */
+ if ((stream->type == VORTEX_PCM_SPDIF) && (!en)) {
+ vortex_route(vortex, 1, 0x14,
+ ADB_MIXOUT(vortex->mixspdif[0]),
+ ADB_SPDIFOUT(0));
+ vortex_route(vortex, 1, 0x14,
+ ADB_MIXOUT(vortex->mixspdif[1]),
+ ADB_SPDIFOUT(1));
+ }
+#endif
+ /* CAPTURE ROUTES. */
+ } else {
+ int src[2], mix[2];
+
+ /* Get SRC and MIXER hardware resources. */
+ for (i = 0; i < nr_ch; i++) {
+ if ((mix[i] =
+ vortex_adb_checkinout(vortex,
+ stream->resources, en,
+ VORTEX_RESOURCE_MIXOUT))
+ < 0) {
+ memset(stream->resources, 0,
+ sizeof(unsigned char) *
+ VORTEX_RESOURCE_LAST);
+ return -EBUSY;
+ }
+ if ((src[i] =
+ vortex_adb_checkinout(vortex,
+ stream->resources, en,
+ VORTEX_RESOURCE_SRC)) < 0) {
+ memset(stream->resources, 0,
+ sizeof(unsigned char) *
+ VORTEX_RESOURCE_LAST);
+ return -EBUSY;
+ }
+ }
+
+ /* Make capture routes. */
+ vortex_connection_mixin_mix(vortex, en, MIX_CAPT(0), mix[0], 0);
+ vortex_connection_mix_src(vortex, en, 0x11, mix[0], src[0]);
+ if (nr_ch == 1) {
+ vortex_connection_mixin_mix(vortex, en,
+ MIX_CAPT(1), mix[0], 0);
+ vortex_connection_src_adbdma(vortex, en,
+ src[0],
+ src[0], dma);
+ } else {
+ vortex_connection_mixin_mix(vortex, en,
+ MIX_CAPT(1), mix[1], 0);
+ vortex_connection_mix_src(vortex, en, 0x11, mix[1],
+ src[1]);
+ vortex_connection_src_src_adbdma(vortex, en,
+ src[1], src[0],
+ src[1], dma);
+ }
+ }
+ vortex->dma_adb[dma].nr_ch = nr_ch;
+
+#if 0
+ /* AC97 Codec channel setup. FIXME: this has no effect on some cards !! */
+ if (nr_ch < 4) {
+ /* Copy stereo to rear channel (surround) */
+ snd_ac97_write_cache(vortex->codec,
+ AC97_SIGMATEL_DAC2INVERT,
+ snd_ac97_read(vortex->codec,
+ AC97_SIGMATEL_DAC2INVERT)
+ | 4);
+ } else {
+ /* Allow separate front and rear channels. */
+ snd_ac97_write_cache(vortex->codec,
+ AC97_SIGMATEL_DAC2INVERT,
+ snd_ac97_read(vortex->codec,
+ AC97_SIGMATEL_DAC2INVERT)
+ & ~((u32)
+ 4));
+ }
+#endif
+ return dma;
+}
+
+/*
+ Set the SampleRate of the SRC's attached to the given DMA engine.
+ */
+static void
+vortex_adb_setsrc(vortex_t * vortex, int adbdma, unsigned int rate, int dir)
+{
+ stream_t *stream = &(vortex->dma_adb[adbdma]);
+ int i, cvrt;
+
+ /* dir=1:play ; dir=0:rec */
+ if (dir)
+ cvrt = SRC_RATIO(rate, 48000);
+ else
+ cvrt = SRC_RATIO(48000, rate);
+
+ /* Setup SRC's */
+ for (i = 0; i < NR_SRC; i++) {
+ if (stream->resources[VORTEX_RESOURCE_SRC] & (1 << i))
+ vortex_src_setupchannel(vortex, i, cvrt, 0, 0, i, dir, 1, cvrt, dir);
+ }
+}
+
+// Timer and ISR functions.
+
+static void vortex_settimer(vortex_t * vortex, int period)
+{
+ //set the timer period to <period> 48000ths of a second.
+ hwwrite(vortex->mmio, VORTEX_IRQ_STAT, period);
+}
+
+#if 0
+static void vortex_enable_timer_int(vortex_t * card)
+{
+ hwwrite(card->mmio, VORTEX_IRQ_CTRL,
+ hwread(card->mmio, VORTEX_IRQ_CTRL) | IRQ_TIMER | 0x60);
+}
+
+static void vortex_disable_timer_int(vortex_t * card)
+{
+ hwwrite(card->mmio, VORTEX_IRQ_CTRL,
+ hwread(card->mmio, VORTEX_IRQ_CTRL) & ~IRQ_TIMER);
+}
+
+#endif
+static void vortex_enable_int(vortex_t * card)
+{
+ // CAsp4ISR__EnableVortexInt_void_
+ hwwrite(card->mmio, VORTEX_CTRL,
+ hwread(card->mmio, VORTEX_CTRL) | CTRL_IRQ_ENABLE);
+ hwwrite(card->mmio, VORTEX_IRQ_CTRL,
+ (hwread(card->mmio, VORTEX_IRQ_CTRL) & 0xffffefc0) | 0x24);
+}
+
+static void vortex_disable_int(vortex_t * card)
+{
+ hwwrite(card->mmio, VORTEX_CTRL,
+ hwread(card->mmio, VORTEX_CTRL) & ~CTRL_IRQ_ENABLE);
+}
+
+static irqreturn_t vortex_interrupt(int irq, void *dev_id)
+{
+ vortex_t *vortex = dev_id;
+ int i, handled;
+ u32 source;
+
+ //check if the interrupt is ours.
+ if (!(hwread(vortex->mmio, VORTEX_STAT) & 0x1))
+ return IRQ_NONE;
+
+ // This is the Interrupt Enable flag we set before (consistency check).
+ if (!(hwread(vortex->mmio, VORTEX_CTRL) & CTRL_IRQ_ENABLE))
+ return IRQ_NONE;
+
+ source = hwread(vortex->mmio, VORTEX_IRQ_SOURCE);
+ // Reset IRQ flags.
+ hwwrite(vortex->mmio, VORTEX_IRQ_SOURCE, source);
+ hwread(vortex->mmio, VORTEX_IRQ_SOURCE);
+ // Is at least one IRQ flag set?
+ if (source == 0) {
+ printk(KERN_ERR "vortex: missing irq source\n");
+ return IRQ_NONE;
+ }
+
+ handled = 0;
+ // Attend every interrupt source.
+ if (unlikely(source & IRQ_ERR_MASK)) {
+ if (source & IRQ_FATAL) {
+ printk(KERN_ERR "vortex: IRQ fatal error\n");
+ }
+ if (source & IRQ_PARITY) {
+ printk(KERN_ERR "vortex: IRQ parity error\n");
+ }
+ if (source & IRQ_REG) {
+ printk(KERN_ERR "vortex: IRQ reg error\n");
+ }
+ if (source & IRQ_FIFO) {
+ printk(KERN_ERR "vortex: IRQ fifo error\n");
+ }
+ if (source & IRQ_DMA) {
+ printk(KERN_ERR "vortex: IRQ dma error\n");
+ }
+ handled = 1;
+ }
+ if (source & IRQ_PCMOUT) {
+ /* ALSA period acknowledge. */
+ spin_lock(&vortex->lock);
+ for (i = 0; i < NR_ADB; i++) {
+ if (vortex->dma_adb[i].fifo_status == FIFO_START) {
+ if (vortex_adbdma_bufshift(vortex, i)) ;
+ spin_unlock(&vortex->lock);
+ snd_pcm_period_elapsed(vortex->dma_adb[i].
+ substream);
+ spin_lock(&vortex->lock);
+ }
+ }
+#ifndef CHIP_AU8810
+ for (i = 0; i < NR_WT; i++) {
+ if (vortex->dma_wt[i].fifo_status == FIFO_START) {
+ if (vortex_wtdma_bufshift(vortex, i)) ;
+ spin_unlock(&vortex->lock);
+ snd_pcm_period_elapsed(vortex->dma_wt[i].
+ substream);
+ spin_lock(&vortex->lock);
+ }
+ }
+#endif
+ spin_unlock(&vortex->lock);
+ handled = 1;
+ }
+ //Acknowledge the Timer interrupt
+ if (source & IRQ_TIMER) {
+ hwread(vortex->mmio, VORTEX_IRQ_STAT);
+ handled = 1;
+ }
+ if (source & IRQ_MIDI) {
+ snd_mpu401_uart_interrupt(vortex->irq,
+ vortex->rmidi->private_data);
+ handled = 1;
+ }
+
+ if (!handled) {
+ printk(KERN_ERR "vortex: unknown irq source %x\n", source);
+ }
+ return IRQ_RETVAL(handled);
+}
+
+/* Codec */
+
+#define POLL_COUNT 1000
+static void vortex_codec_init(vortex_t * vortex)
+{
+ int i;
+
+ for (i = 0; i < 32; i++) {
+ /* the windows driver writes -i, so we write -i */
+ hwwrite(vortex->mmio, (VORTEX_CODEC_CHN + (i << 2)), -i);
+ msleep(2);
+ }
+ if (0) {
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x8068);
+ msleep(1);
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x00e8);
+ msleep(1);
+ } else {
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x00a8);
+ msleep(2);
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x80a8);
+ msleep(2);
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x80e8);
+ msleep(2);
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x80a8);
+ msleep(2);
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x00a8);
+ msleep(2);
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0x00e8);
+ }
+ for (i = 0; i < 32; i++) {
+ hwwrite(vortex->mmio, (VORTEX_CODEC_CHN + (i << 2)), -i);
+ msleep(5);
+ }
+ hwwrite(vortex->mmio, VORTEX_CODEC_CTRL, 0xe8);
+ msleep(1);
+ /* Enable codec channels 0 and 1. */
+ hwwrite(vortex->mmio, VORTEX_CODEC_EN,
+ hwread(vortex->mmio, VORTEX_CODEC_EN) | EN_CODEC);
+}
+
+static void
+vortex_codec_write(struct snd_ac97 * codec, unsigned short addr, unsigned short data)
+{
+
+ vortex_t *card = (vortex_t *) codec->private_data;
+ unsigned int lifeboat = 0;
+
+ /* wait for transactions to clear */
+ while (!(hwread(card->mmio, VORTEX_CODEC_CTRL) & 0x100)) {
+ udelay(100);
+ if (lifeboat++ > POLL_COUNT) {
+ printk(KERN_ERR "vortex: ac97 codec stuck busy\n");
+ return;
+ }
+ }
+ /* write register */
+ hwwrite(card->mmio, VORTEX_CODEC_IO,
+ ((addr << VORTEX_CODEC_ADDSHIFT) & VORTEX_CODEC_ADDMASK) |
+ ((data << VORTEX_CODEC_DATSHIFT) & VORTEX_CODEC_DATMASK) |
+ VORTEX_CODEC_WRITE |
+ (codec->num << VORTEX_CODEC_ID_SHIFT) );
+
+ /* Flush Caches. */
+ hwread(card->mmio, VORTEX_CODEC_IO);
+}
+
+static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short addr)
+{
+
+ vortex_t *card = (vortex_t *) codec->private_data;
+ u32 read_addr, data;
+ unsigned lifeboat = 0;
+
+ /* wait for transactions to clear */
+ while (!(hwread(card->mmio, VORTEX_CODEC_CTRL) & 0x100)) {
+ udelay(100);
+ if (lifeboat++ > POLL_COUNT) {
+ printk(KERN_ERR "vortex: ac97 codec stuck busy\n");
+ return 0xffff;
+ }
+ }
+ /* set up read address */
+ read_addr = ((addr << VORTEX_CODEC_ADDSHIFT) & VORTEX_CODEC_ADDMASK) |
+ (codec->num << VORTEX_CODEC_ID_SHIFT) ;
+ hwwrite(card->mmio, VORTEX_CODEC_IO, read_addr);
+
+ /* wait for address */
+ do {
+ udelay(100);
+ data = hwread(card->mmio, VORTEX_CODEC_IO);
+ if (lifeboat++ > POLL_COUNT) {
+ printk(KERN_ERR "vortex: ac97 address never arrived\n");
+ return 0xffff;
+ }
+ } while ((data & VORTEX_CODEC_ADDMASK) !=
+ (addr << VORTEX_CODEC_ADDSHIFT));
+
+ /* return data. */
+ return (u16) (data & VORTEX_CODEC_DATMASK);
+}
+
+/* SPDIF support */
+
+static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode)
+{
+ int i, this_38 = 0, this_04 = 0, this_08 = 0, this_0c = 0;
+
+ /* CAsp4Spdif::InitializeSpdifHardware(void) */
+ hwwrite(vortex->mmio, VORTEX_SPDIF_FLAGS,
+ hwread(vortex->mmio, VORTEX_SPDIF_FLAGS) & 0xfff3fffd);
+ //for (i=0x291D4; i<0x29200; i+=4)
+ for (i = 0; i < 11; i++)
+ hwwrite(vortex->mmio, VORTEX_SPDIF_CFG1 + (i << 2), 0);
+ //hwwrite(vortex->mmio, 0x29190, hwread(vortex->mmio, 0x29190) | 0xc0000);
+ hwwrite(vortex->mmio, VORTEX_CODEC_EN,
+ hwread(vortex->mmio, VORTEX_CODEC_EN) | EN_SPDIF);
+
+ /* CAsp4Spdif::ProgramSRCInHardware(enum SPDIF_SR,enum SPDIFMODE) */
+ if (this_04 && this_08) {
+ int edi;
+
+ i = (((0x5DC00000 / spdif_sr) + 1) >> 1);
+ if (i > 0x800) {
+ if (i < 0x1ffff)
+ edi = (i >> 1);
+ else
+ edi = 0x1ffff;
+ } else {
+ i = edi = 0x800;
+ }
+ /* this_04 and this_08 are the CASp4Src's (samplerate converters) */
+ vortex_src_setupchannel(vortex, this_04, edi, 0, 1,
+ this_0c, 1, 0, edi, 1);
+ vortex_src_setupchannel(vortex, this_08, edi, 0, 1,
+ this_0c, 1, 0, edi, 1);
+ }
+
+ i = spdif_sr;
+ spdif_sr |= 0x8c;
+ switch (i) {
+ case 32000:
+ this_38 &= 0xFFFFFFFE;
+ this_38 &= 0xFFFFFFFD;
+ this_38 &= 0xF3FFFFFF;
+ this_38 |= 0x03000000; /* set 32khz samplerate */
+ this_38 &= 0xFFFFFF3F;
+ spdif_sr &= 0xFFFFFFFD;
+ spdif_sr |= 1;
+ break;
+ case 44100:
+ this_38 &= 0xFFFFFFFE;
+ this_38 &= 0xFFFFFFFD;
+ this_38 &= 0xF0FFFFFF;
+ this_38 |= 0x03000000;
+ this_38 &= 0xFFFFFF3F;
+ spdif_sr &= 0xFFFFFFFC;
+ break;
+ case 48000:
+ if (spdif_mode == 1) {
+ this_38 &= 0xFFFFFFFE;
+ this_38 &= 0xFFFFFFFD;
+ this_38 &= 0xF2FFFFFF;
+ this_38 |= 0x02000000; /* set 48khz samplerate */
+ this_38 &= 0xFFFFFF3F;
+ } else {
+ /* J. Gordon Wolfe: I think this stuff is for AC3 */
+ this_38 |= 0x00000003;
+ this_38 &= 0xFFFFFFBF;
+ this_38 |= 0x80;
+ }
+ spdif_sr |= 2;
+ spdif_sr &= 0xFFFFFFFE;
+ break;
+
+ }
+ /* looks like the next 2 lines transfer a 16-bit value into 2 8-bit
+ registers. seems to be for the standard IEC/SPDIF initialization
+ stuff */
+ hwwrite(vortex->mmio, VORTEX_SPDIF_CFG0, this_38 & 0xffff);
+ hwwrite(vortex->mmio, VORTEX_SPDIF_CFG1, this_38 >> 0x10);
+ hwwrite(vortex->mmio, VORTEX_SPDIF_SMPRATE, spdif_sr);
+}
+
+/* Initialization */
+
+static int __devinit vortex_core_init(vortex_t * vortex)
+{
+
+ printk(KERN_INFO "Vortex: init.... ");
+ /* Hardware Init. */
+ hwwrite(vortex->mmio, VORTEX_CTRL, 0xffffffff);
+ msleep(5);
+ hwwrite(vortex->mmio, VORTEX_CTRL,
+ hwread(vortex->mmio, VORTEX_CTRL) & 0xffdfffff);
+ msleep(5);
+ /* Reset IRQ flags */
+ hwwrite(vortex->mmio, VORTEX_IRQ_SOURCE, 0xffffffff);
+ hwread(vortex->mmio, VORTEX_IRQ_STAT);
+
+ vortex_codec_init(vortex);
+
+#ifdef CHIP_AU8830
+ hwwrite(vortex->mmio, VORTEX_CTRL,
+ hwread(vortex->mmio, VORTEX_CTRL) | 0x1000000);
+#endif
+
+ /* Init audio engine. */
+ vortex_adbdma_init(vortex);
+ hwwrite(vortex->mmio, VORTEX_ENGINE_CTRL, 0x0); //, 0xc83c7e58, 0xc5f93e58
+ vortex_adb_init(vortex);
+ /* Init processing blocks. */
+ vortex_fifo_init(vortex);
+ vortex_mixer_init(vortex);
+ vortex_srcblock_init(vortex);
+#ifndef CHIP_AU8820
+ vortex_eq_init(vortex);
+ vortex_spdif_init(vortex, 48000, 1);
+ vortex_Vort3D_enable(vortex);
+#endif
+#ifndef CHIP_AU8810
+ vortex_wt_init(vortex);
+#endif
+ // Moved to au88x0.c
+ //vortex_connect_default(vortex, 1);
+
+ vortex_settimer(vortex, 0x90);
+ // Enable Interrupts.
+ // vortex_enable_int() must be first !!
+ // hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, 0);
+ // vortex_enable_int(vortex);
+ //vortex_enable_timer_int(vortex);
+ //vortex_disable_timer_int(vortex);
+
+ printk(KERN_INFO "done.\n");
+ spin_lock_init(&vortex->lock);
+
+ return 0;
+}
+
+static int vortex_core_shutdown(vortex_t * vortex)
+{
+
+ printk(KERN_INFO "Vortex: shutdown...");
+#ifndef CHIP_AU8820
+ vortex_eq_free(vortex);
+ vortex_Vort3D_disable(vortex);
+#endif
+ //vortex_disable_timer_int(vortex);
+ vortex_disable_int(vortex);
+ vortex_connect_default(vortex, 0);
+ /* Reset all DMA fifos. */
+ vortex_fifo_init(vortex);
+ /* Erase all audio routes. */
+ vortex_adb_init(vortex);
+
+ /* Disable MPU401 */
+ //hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, hwread(vortex->mmio, VORTEX_IRQ_CTRL) & ~IRQ_MIDI);
+ //hwwrite(vortex->mmio, VORTEX_CTRL, hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_EN);
+
+ hwwrite(vortex->mmio, VORTEX_IRQ_CTRL, 0);
+ hwwrite(vortex->mmio, VORTEX_CTRL, 0);
+ msleep(5);
+ hwwrite(vortex->mmio, VORTEX_IRQ_SOURCE, 0xffff);
+
+ printk(KERN_INFO "done.\n");
+ return 0;
+}
+
+/* Alsa support. */
+
+static int vortex_alsafmt_aspfmt(int alsafmt)
+{
+ int fmt;
+
+ switch (alsafmt) {
+ case SNDRV_PCM_FORMAT_U8:
+ fmt = 0x1;
+ break;
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ fmt = 0x2;
+ break;
+ case SNDRV_PCM_FORMAT_A_LAW:
+ fmt = 0x3;
+ break;
+ case SNDRV_PCM_FORMAT_SPECIAL:
+ fmt = 0x4; /* guess. */
+ break;
+ case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE:
+ fmt = 0x5; /* guess. */
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ fmt = 0x8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ fmt = 0x9; /* check this... */
+ break;
+ default:
+ fmt = 0x8;
+ printk(KERN_ERR "vortex: format unsupported %d\n", alsafmt);
+ break;
+ }
+ return fmt;
+}
+
+/* Some not yet useful translations. */
+#if 0
+typedef enum {
+ ASPFMTLINEAR16 = 0, /* 0x8 */
+ ASPFMTLINEAR8, /* 0x1 */
+ ASPFMTULAW, /* 0x2 */
+ ASPFMTALAW, /* 0x3 */
+ ASPFMTSPORT, /* ? */
+ ASPFMTSPDIF, /* ? */
+} ASPENCODING;
+
+static int
+vortex_translateformat(vortex_t * vortex, char bits, char nch, int encod)
+{
+ int a, this_194;
+
+ if ((bits != 8) || (bits != 16))
+ return -1;
+
+ switch (encod) {
+ case 0:
+ if (bits == 0x10)
+ a = 8; // 16 bit
+ break;
+ case 1:
+ if (bits == 8)
+ a = 1; // 8 bit
+ break;
+ case 2:
+ a = 2; // U_LAW
+ break;
+ case 3:
+ a = 3; // A_LAW
+ break;
+ }
+ switch (nch) {
+ case 1:
+ this_194 = 0;
+ break;
+ case 2:
+ this_194 = 1;
+ break;
+ case 4:
+ this_194 = 1;
+ break;
+ case 6:
+ this_194 = 1;
+ break;
+ }
+ return (a);
+}
+
+static void vortex_cdmacore_setformat(vortex_t * vortex, int bits, int nch)
+{
+ short int d, this_148;
+
+ d = ((bits >> 3) * nch);
+ this_148 = 0xbb80 / d;
+}
+#endif
diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c
new file mode 100644
index 0000000..38602b8
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_eq.c
@@ -0,0 +1,928 @@
+/***************************************************************************
+ * au88x0_eq.c
+ * Aureal Vortex Hardware EQ control/access.
+ *
+ * Sun Jun 8 18:19:19 2003
+ * 2003 Manuel Jander (mjander@users.sourceforge.net)
+ *
+ * 02 July 2003: First time something works :)
+ * November 2003: A3D Bypass code completed but untested.
+ *
+ * TODO:
+ * - Debug (testing)
+ * - Test peak visualization support.
+ *
+ ****************************************************************************/
+
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+/*
+ The Aureal Hardware EQ is found on AU8810 and AU8830 chips only.
+ it has 4 inputs (2 for general mix, 2 for A3D) and 2 outputs (supposed
+ to be routed to the codec).
+*/
+
+#include "au88x0.h"
+#include "au88x0_eq.h"
+#include "au88x0_eqdata.c"
+
+#define VORTEX_EQ_BASE 0x2b000
+#define VORTEX_EQ_DEST (VORTEX_EQ_BASE + 0x410)
+#define VORTEX_EQ_SOURCE (VORTEX_EQ_BASE + 0x430)
+#define VORTEX_EQ_CTRL (VORTEX_EQ_BASE + 0x440)
+
+#define VORTEX_BAND_COEFF_SIZE 0x30
+
+/* CEqHw.s */
+static void vortex_EqHw_SetTimeConsts(vortex_t * vortex, u16 gain, u16 level)
+{
+ hwwrite(vortex->mmio, 0x2b3c4, gain);
+ hwwrite(vortex->mmio, 0x2b3c8, level);
+}
+
+static inline u16 sign_invert(u16 a)
+{
+ /* -(-32768) -> -32768 so we do -(-32768) -> 32767 to make the result positive */
+ if (a == (u16)-32768)
+ return 32767;
+ else
+ return -a;
+}
+
+static void vortex_EqHw_SetLeftCoefs(vortex_t * vortex, u16 coefs[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int i = 0, n /*esp2c */;
+
+ for (n = 0; n < eqhw->this04; n++) {
+ hwwrite(vortex->mmio, 0x2b000 + n * 0x30, coefs[i + 0]);
+ hwwrite(vortex->mmio, 0x2b004 + n * 0x30, coefs[i + 1]);
+
+ if (eqhw->this08 == 0) {
+ hwwrite(vortex->mmio, 0x2b008 + n * 0x30, coefs[i + 2]);
+ hwwrite(vortex->mmio, 0x2b00c + n * 0x30, coefs[i + 3]);
+ hwwrite(vortex->mmio, 0x2b010 + n * 0x30, coefs[i + 4]);
+ } else {
+ hwwrite(vortex->mmio, 0x2b008 + n * 0x30, sign_invert(coefs[2 + i]));
+ hwwrite(vortex->mmio, 0x2b00c + n * 0x30, sign_invert(coefs[3 + i]));
+ hwwrite(vortex->mmio, 0x2b010 + n * 0x30, sign_invert(coefs[4 + i]));
+ }
+ i += 5;
+ }
+}
+
+static void vortex_EqHw_SetRightCoefs(vortex_t * vortex, u16 coefs[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int i = 0, n /*esp2c */;
+
+ for (n = 0; n < eqhw->this04; n++) {
+ hwwrite(vortex->mmio, 0x2b1e0 + n * 0x30, coefs[0 + i]);
+ hwwrite(vortex->mmio, 0x2b1e4 + n * 0x30, coefs[1 + i]);
+
+ if (eqhw->this08 == 0) {
+ hwwrite(vortex->mmio, 0x2b1e8 + n * 0x30, coefs[2 + i]);
+ hwwrite(vortex->mmio, 0x2b1ec + n * 0x30, coefs[3 + i]);
+ hwwrite(vortex->mmio, 0x2b1f0 + n * 0x30, coefs[4 + i]);
+ } else {
+ hwwrite(vortex->mmio, 0x2b1e8 + n * 0x30, sign_invert(coefs[2 + i]));
+ hwwrite(vortex->mmio, 0x2b1ec + n * 0x30, sign_invert(coefs[3 + i]));
+ hwwrite(vortex->mmio, 0x2b1f0 + n * 0x30, sign_invert(coefs[4 + i]));
+ }
+ i += 5;
+ }
+
+}
+
+static void vortex_EqHw_SetLeftStates(vortex_t * vortex, u16 a[], u16 b[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int i = 0, ebx;
+
+ hwwrite(vortex->mmio, 0x2b3fc, a[0]);
+ hwwrite(vortex->mmio, 0x2b400, a[1]);
+
+ for (ebx = 0; ebx < eqhw->this04; ebx++) {
+ hwwrite(vortex->mmio, 0x2b014 + (i * 0xc), b[i]);
+ hwwrite(vortex->mmio, 0x2b018 + (i * 0xc), b[1 + i]);
+ hwwrite(vortex->mmio, 0x2b01c + (i * 0xc), b[2 + i]);
+ hwwrite(vortex->mmio, 0x2b020 + (i * 0xc), b[3 + i]);
+ i += 4;
+ }
+}
+
+static void vortex_EqHw_SetRightStates(vortex_t * vortex, u16 a[], u16 b[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int i = 0, ebx;
+
+ hwwrite(vortex->mmio, 0x2b404, a[0]);
+ hwwrite(vortex->mmio, 0x2b408, a[1]);
+
+ for (ebx = 0; ebx < eqhw->this04; ebx++) {
+ hwwrite(vortex->mmio, 0x2b1f4 + (i * 0xc), b[i]);
+ hwwrite(vortex->mmio, 0x2b1f8 + (i * 0xc), b[1 + i]);
+ hwwrite(vortex->mmio, 0x2b1fc + (i * 0xc), b[2 + i]);
+ hwwrite(vortex->mmio, 0x2b200 + (i * 0xc), b[3 + i]);
+ i += 4;
+ }
+}
+
+#if 0
+static void vortex_EqHw_GetTimeConsts(vortex_t * vortex, u16 * a, u16 * b)
+{
+ *a = hwread(vortex->mmio, 0x2b3c4);
+ *b = hwread(vortex->mmio, 0x2b3c8);
+}
+
+static void vortex_EqHw_GetLeftCoefs(vortex_t * vortex, u16 a[])
+{
+
+}
+
+static void vortex_EqHw_GetRightCoefs(vortex_t * vortex, u16 a[])
+{
+
+}
+
+static void vortex_EqHw_GetLeftStates(vortex_t * vortex, u16 * a, u16 b[])
+{
+
+}
+
+static void vortex_EqHw_GetRightStates(vortex_t * vortex, u16 * a, u16 b[])
+{
+
+}
+
+#endif
+/* Mix Gains */
+static void vortex_EqHw_SetBypassGain(vortex_t * vortex, u16 a, u16 b)
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ if (eqhw->this08 == 0) {
+ hwwrite(vortex->mmio, 0x2b3d4, a);
+ hwwrite(vortex->mmio, 0x2b3ec, b);
+ } else {
+ hwwrite(vortex->mmio, 0x2b3d4, sign_invert(a));
+ hwwrite(vortex->mmio, 0x2b3ec, sign_invert(b));
+ }
+}
+
+static void vortex_EqHw_SetA3DBypassGain(vortex_t * vortex, u16 a, u16 b)
+{
+
+ hwwrite(vortex->mmio, 0x2b3e0, a);
+ hwwrite(vortex->mmio, 0x2b3f8, b);
+}
+
+#if 0
+static void vortex_EqHw_SetCurrBypassGain(vortex_t * vortex, u16 a, u16 b)
+{
+
+ hwwrite(vortex->mmio, 0x2b3d0, a);
+ hwwrite(vortex->mmio, 0x2b3e8, b);
+}
+
+static void vortex_EqHw_SetCurrA3DBypassGain(vortex_t * vortex, u16 a, u16 b)
+{
+
+ hwwrite(vortex->mmio, 0x2b3dc, a);
+ hwwrite(vortex->mmio, 0x2b3f4, b);
+}
+
+#endif
+static void
+vortex_EqHw_SetLeftGainsSingleTarget(vortex_t * vortex, u16 index, u16 b)
+{
+ hwwrite(vortex->mmio, 0x2b02c + (index * 0x30), b);
+}
+
+static void
+vortex_EqHw_SetRightGainsSingleTarget(vortex_t * vortex, u16 index, u16 b)
+{
+ hwwrite(vortex->mmio, 0x2b20c + (index * 0x30), b);
+}
+
+static void vortex_EqHw_SetLeftGainsTarget(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx;
+
+ for (ebx = 0; ebx < eqhw->this04; ebx++) {
+ hwwrite(vortex->mmio, 0x2b02c + ebx * 0x30, a[ebx]);
+ }
+}
+
+static void vortex_EqHw_SetRightGainsTarget(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx;
+
+ for (ebx = 0; ebx < eqhw->this04; ebx++) {
+ hwwrite(vortex->mmio, 0x2b20c + ebx * 0x30, a[ebx]);
+ }
+}
+
+static void vortex_EqHw_SetLeftGainsCurrent(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx;
+
+ for (ebx = 0; ebx < eqhw->this04; ebx++) {
+ hwwrite(vortex->mmio, 0x2b028 + ebx * 0x30, a[ebx]);
+ }
+}
+
+static void vortex_EqHw_SetRightGainsCurrent(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx;
+
+ for (ebx = 0; ebx < eqhw->this04; ebx++) {
+ hwwrite(vortex->mmio, 0x2b208 + ebx * 0x30, a[ebx]);
+ }
+}
+
+#if 0
+static void vortex_EqHw_GetLeftGainsTarget(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx = 0;
+
+ if (eqhw->this04 < 0)
+ return;
+
+ do {
+ a[ebx] = hwread(vortex->mmio, 0x2b02c + ebx * 0x30);
+ ebx++;
+ }
+ while (ebx < eqhw->this04);
+}
+
+static void vortex_EqHw_GetRightGainsTarget(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx = 0;
+
+ if (eqhw->this04 < 0)
+ return;
+
+ do {
+ a[ebx] = hwread(vortex->mmio, 0x2b20c + ebx * 0x30);
+ ebx++;
+ }
+ while (ebx < eqhw->this04);
+}
+
+static void vortex_EqHw_GetLeftGainsCurrent(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx = 0;
+
+ if (eqhw->this04 < 0)
+ return;
+
+ do {
+ a[ebx] = hwread(vortex->mmio, 0x2b028 + ebx * 0x30);
+ ebx++;
+ }
+ while (ebx < eqhw->this04);
+}
+
+static void vortex_EqHw_GetRightGainsCurrent(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx = 0;
+
+ if (eqhw->this04 < 0)
+ return;
+
+ do {
+ a[ebx] = hwread(vortex->mmio, 0x2b208 + ebx * 0x30);
+ ebx++;
+ }
+ while (ebx < eqhw->this04);
+}
+
+#endif
+/* EQ band levels settings */
+static void vortex_EqHw_SetLevels(vortex_t * vortex, u16 peaks[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int i;
+
+ /* set left peaks */
+ for (i = 0; i < eqhw->this04; i++) {
+ hwwrite(vortex->mmio, 0x2b024 + i * VORTEX_BAND_COEFF_SIZE, peaks[i]);
+ }
+
+ hwwrite(vortex->mmio, 0x2b3cc, peaks[eqhw->this04]);
+ hwwrite(vortex->mmio, 0x2b3d8, peaks[eqhw->this04 + 1]);
+
+ /* set right peaks */
+ for (i = 0; i < eqhw->this04; i++) {
+ hwwrite(vortex->mmio, 0x2b204 + i * VORTEX_BAND_COEFF_SIZE,
+ peaks[i + (eqhw->this04 + 2)]);
+ }
+
+ hwwrite(vortex->mmio, 0x2b3e4, peaks[2 + (eqhw->this04 * 2)]);
+ hwwrite(vortex->mmio, 0x2b3f0, peaks[3 + (eqhw->this04 * 2)]);
+}
+
+#if 0
+static void vortex_EqHw_GetLevels(vortex_t * vortex, u16 a[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int ebx;
+
+ if (eqhw->this04 < 0)
+ return;
+
+ ebx = 0;
+ do {
+ a[ebx] = hwread(vortex->mmio, 0x2b024 + ebx * 0x30);
+ ebx++;
+ }
+ while (ebx < eqhw->this04);
+
+ a[eqhw->this04] = hwread(vortex->mmio, 0x2b3cc);
+ a[eqhw->this04 + 1] = hwread(vortex->mmio, 0x2b3d8);
+
+ ebx = 0;
+ do {
+ a[ebx + (eqhw->this04 + 2)] =
+ hwread(vortex->mmio, 0x2b204 + ebx * 0x30);
+ ebx++;
+ }
+ while (ebx < eqhw->this04);
+
+ a[2 + (eqhw->this04 * 2)] = hwread(vortex->mmio, 0x2b3e4);
+ a[3 + (eqhw->this04 * 2)] = hwread(vortex->mmio, 0x2b3f0);
+}
+
+#endif
+/* Global Control */
+static void vortex_EqHw_SetControlReg(vortex_t * vortex, u32 reg)
+{
+ hwwrite(vortex->mmio, 0x2b440, reg);
+}
+
+static void vortex_EqHw_SetSampleRate(vortex_t * vortex, u32 sr)
+{
+ hwwrite(vortex->mmio, 0x2b440, ((sr & 0x1f) << 3) | 0xb800);
+}
+
+#if 0
+static void vortex_EqHw_GetControlReg(vortex_t * vortex, u32 *reg)
+{
+ *reg = hwread(vortex->mmio, 0x2b440);
+}
+
+static void vortex_EqHw_GetSampleRate(vortex_t * vortex, u32 *sr)
+{
+ *sr = (hwread(vortex->mmio, 0x2b440) >> 3) & 0x1f;
+}
+
+#endif
+static void vortex_EqHw_Enable(vortex_t * vortex)
+{
+ hwwrite(vortex->mmio, VORTEX_EQ_CTRL, 0xf001);
+}
+
+static void vortex_EqHw_Disable(vortex_t * vortex)
+{
+ hwwrite(vortex->mmio, VORTEX_EQ_CTRL, 0xf000);
+}
+
+/* Reset (zero) buffers */
+static void vortex_EqHw_ZeroIO(vortex_t * vortex)
+{
+ int i;
+ for (i = 0; i < 0x8; i++)
+ hwwrite(vortex->mmio, VORTEX_EQ_DEST + (i << 2), 0x0);
+ for (i = 0; i < 0x4; i++)
+ hwwrite(vortex->mmio, VORTEX_EQ_SOURCE + (i << 2), 0x0);
+}
+
+static void vortex_EqHw_ZeroA3DIO(vortex_t * vortex)
+{
+ int i;
+ for (i = 0; i < 0x4; i++)
+ hwwrite(vortex->mmio, VORTEX_EQ_DEST + (i << 2), 0x0);
+}
+
+static void vortex_EqHw_ZeroState(vortex_t * vortex)
+{
+
+ vortex_EqHw_SetControlReg(vortex, 0);
+ vortex_EqHw_ZeroIO(vortex);
+ hwwrite(vortex->mmio, 0x2b3c0, 0);
+
+ vortex_EqHw_SetTimeConsts(vortex, 0, 0);
+
+ vortex_EqHw_SetLeftCoefs(vortex, asEqCoefsZeros);
+ vortex_EqHw_SetRightCoefs(vortex, asEqCoefsZeros);
+
+ vortex_EqHw_SetLeftGainsCurrent(vortex, eq_gains_zero);
+ vortex_EqHw_SetRightGainsCurrent(vortex, eq_gains_zero);
+ vortex_EqHw_SetLeftGainsTarget(vortex, eq_gains_zero);
+ vortex_EqHw_SetRightGainsTarget(vortex, eq_gains_zero);
+
+ vortex_EqHw_SetBypassGain(vortex, 0, 0);
+ //vortex_EqHw_SetCurrBypassGain(vortex, 0, 0);
+ vortex_EqHw_SetA3DBypassGain(vortex, 0, 0);
+ //vortex_EqHw_SetCurrA3DBypassGain(vortex, 0, 0);
+ vortex_EqHw_SetLeftStates(vortex, eq_states_zero, asEqOutStateZeros);
+ vortex_EqHw_SetRightStates(vortex, eq_states_zero, asEqOutStateZeros);
+ vortex_EqHw_SetLevels(vortex, (u16 *) eq_levels);
+}
+
+/* Program coeficients as pass through */
+static void vortex_EqHw_ProgramPipe(vortex_t * vortex)
+{
+ vortex_EqHw_SetTimeConsts(vortex, 0, 0);
+
+ vortex_EqHw_SetLeftCoefs(vortex, asEqCoefsPipes);
+ vortex_EqHw_SetRightCoefs(vortex, asEqCoefsPipes);
+
+ vortex_EqHw_SetLeftGainsCurrent(vortex, eq_gains_current);
+ vortex_EqHw_SetRightGainsCurrent(vortex, eq_gains_current);
+ vortex_EqHw_SetLeftGainsTarget(vortex, eq_gains_current);
+ vortex_EqHw_SetRightGainsTarget(vortex, eq_gains_current);
+}
+
+/* Program EQ block as 10 band Equalizer */
+static void
+vortex_EqHw_Program10Band(vortex_t * vortex, auxxEqCoeffSet_t * coefset)
+{
+
+ vortex_EqHw_SetTimeConsts(vortex, 0xc, 0x7fe0);
+
+ vortex_EqHw_SetLeftCoefs(vortex, coefset->LeftCoefs);
+ vortex_EqHw_SetRightCoefs(vortex, coefset->RightCoefs);
+
+ vortex_EqHw_SetLeftGainsCurrent(vortex, coefset->LeftGains);
+
+ vortex_EqHw_SetRightGainsTarget(vortex, coefset->RightGains);
+ vortex_EqHw_SetLeftGainsTarget(vortex, coefset->LeftGains);
+
+ vortex_EqHw_SetRightGainsCurrent(vortex, coefset->RightGains);
+}
+
+/* Read all EQ peaks. (think VU meter) */
+static void vortex_EqHw_GetTenBandLevels(vortex_t * vortex, u16 peaks[])
+{
+ eqhw_t *eqhw = &(vortex->eq.this04);
+ int i;
+
+ if (eqhw->this04 <= 0)
+ return;
+
+ for (i = 0; i < eqhw->this04; i++)
+ peaks[i] = hwread(vortex->mmio, 0x2B024 + i * 0x30);
+ for (i = 0; i < eqhw->this04; i++)
+ peaks[i + eqhw->this04] =
+ hwread(vortex->mmio, 0x2B204 + i * 0x30);
+}
+
+/* CEqlzr.s */
+
+static int vortex_Eqlzr_GetLeftGain(vortex_t * vortex, u16 index, u16 * gain)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ if (eq->this28) {
+ *gain = eq->this130[index];
+ return 0;
+ }
+ return 1;
+}
+
+static void vortex_Eqlzr_SetLeftGain(vortex_t * vortex, u16 index, u16 gain)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ if (eq->this28 == 0)
+ return;
+
+ eq->this130[index] = gain;
+ if (eq->this54)
+ return;
+
+ vortex_EqHw_SetLeftGainsSingleTarget(vortex, index, gain);
+}
+
+static int vortex_Eqlzr_GetRightGain(vortex_t * vortex, u16 index, u16 * gain)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ if (eq->this28) {
+ *gain = eq->this130[index + eq->this10];
+ return 0;
+ }
+ return 1;
+}
+
+static void vortex_Eqlzr_SetRightGain(vortex_t * vortex, u16 index, u16 gain)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ if (eq->this28 == 0)
+ return;
+
+ eq->this130[index + eq->this10] = gain;
+ if (eq->this54)
+ return;
+
+ vortex_EqHw_SetRightGainsSingleTarget(vortex, index, gain);
+}
+
+#if 0
+static int
+vortex_Eqlzr_GetAllBands(vortex_t * vortex, u16 * gains, s32 *cnt)
+{
+ eqlzr_t *eq = &(vortex->eq);
+ int si = 0;
+
+ if (eq->this10 == 0)
+ return 1;
+
+ {
+ if (vortex_Eqlzr_GetLeftGain(vortex, si, &gains[si]))
+ return 1;
+ if (vortex_Eqlzr_GetRightGain
+ (vortex, si, &gains[si + eq->this10]))
+ return 1;
+ si++;
+ }
+ while (eq->this10 > si) ;
+ *cnt = si * 2;
+ return 0;
+}
+#endif
+static int vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex_t * vortex)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ vortex_EqHw_SetLeftGainsTarget(vortex, eq->this130);
+ vortex_EqHw_SetRightGainsTarget(vortex, &(eq->this130[eq->this10]));
+
+ return 0;
+}
+
+static int
+vortex_Eqlzr_SetAllBands(vortex_t * vortex, u16 gains[], s32 count)
+{
+ eqlzr_t *eq = &(vortex->eq);
+ int i;
+
+ if (((eq->this10) * 2 != count) || (eq->this28 == 0))
+ return 1;
+
+ for (i = 0; i < count; i++) {
+ eq->this130[i] = gains[i];
+ }
+
+ if (eq->this54)
+ return 0;
+ return vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex);
+}
+
+static void
+vortex_Eqlzr_SetA3dBypassGain(vortex_t * vortex, u32 a, u32 b)
+{
+ eqlzr_t *eq = &(vortex->eq);
+ u32 eax, ebx;
+
+ eq->this58 = a;
+ eq->this5c = b;
+ if (eq->this54)
+ eax = eq->this0e;
+ else
+ eax = eq->this0a;
+ ebx = (eax * eq->this58) >> 0x10;
+ eax = (eax * eq->this5c) >> 0x10;
+ vortex_EqHw_SetA3DBypassGain(vortex, ebx, eax);
+}
+
+static void vortex_Eqlzr_ProgramA3dBypassGain(vortex_t * vortex)
+{
+ eqlzr_t *eq = &(vortex->eq);
+ u32 eax, ebx;
+
+ if (eq->this54)
+ eax = eq->this0e;
+ else
+ eax = eq->this0a;
+ ebx = (eax * eq->this58) >> 0x10;
+ eax = (eax * eq->this5c) >> 0x10;
+ vortex_EqHw_SetA3DBypassGain(vortex, ebx, eax);
+}
+
+static void vortex_Eqlzr_ShutDownA3d(vortex_t * vortex)
+{
+ if (vortex != NULL)
+ vortex_EqHw_ZeroA3DIO(vortex);
+}
+
+static void vortex_Eqlzr_SetBypass(vortex_t * vortex, u32 bp)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ if ((eq->this28) && (bp == 0)) {
+ /* EQ enabled */
+ vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex);
+ vortex_EqHw_SetBypassGain(vortex, eq->this08, eq->this08);
+ } else {
+ /* EQ disabled. */
+ vortex_EqHw_SetLeftGainsTarget(vortex, eq->this14_array);
+ vortex_EqHw_SetRightGainsTarget(vortex, eq->this14_array);
+ vortex_EqHw_SetBypassGain(vortex, eq->this0c, eq->this0c);
+ }
+ vortex_Eqlzr_ProgramA3dBypassGain(vortex);
+}
+
+static void vortex_Eqlzr_ReadAndSetActiveCoefSet(vortex_t * vortex)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ /* Set EQ BiQuad filter coeficients */
+ memcpy(&(eq->coefset), &asEqCoefsNormal, sizeof(auxxEqCoeffSet_t));
+ /* Set EQ Band gain levels and dump into hardware registers. */
+ vortex_Eqlzr_SetAllBands(vortex, eq_gains_normal, eq->this10 * 2);
+}
+
+static int vortex_Eqlzr_GetAllPeaks(vortex_t * vortex, u16 * peaks, int *count)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ if (eq->this10 == 0)
+ return 1;
+ *count = eq->this10 * 2;
+ vortex_EqHw_GetTenBandLevels(vortex, peaks);
+ return 0;
+}
+
+#if 0
+static auxxEqCoeffSet_t *vortex_Eqlzr_GetActiveCoefSet(vortex_t * vortex)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ return (&(eq->coefset));
+}
+#endif
+static void vortex_Eqlzr_init(vortex_t * vortex)
+{
+ eqlzr_t *eq = &(vortex->eq);
+
+ /* Object constructor */
+ //eq->this04 = 0;
+ eq->this08 = 0; /* Bypass gain with EQ in use. */
+ eq->this0a = 0x5999;
+ eq->this0c = 0x5999; /* Bypass gain with EQ disabled. */
+ eq->this0e = 0x5999;
+
+ eq->this10 = 0xa; /* 10 eq frequency bands. */
+ eq->this04.this04 = eq->this10;
+ eq->this28 = 0x1; /* if 1 => Allow read access to this130 (gains) */
+ eq->this54 = 0x0; /* if 1 => Dont Allow access to hardware (gains) */
+ eq->this58 = 0xffff;
+ eq->this5c = 0xffff;
+
+ /* Set gains. */
+ memset(eq->this14_array, 0, sizeof(eq->this14_array));
+
+ /* Actual init. */
+ vortex_EqHw_ZeroState(vortex);
+ vortex_EqHw_SetSampleRate(vortex, 0x11);
+ vortex_Eqlzr_ReadAndSetActiveCoefSet(vortex);
+
+ vortex_EqHw_Program10Band(vortex, &(eq->coefset));
+ vortex_Eqlzr_SetBypass(vortex, eq->this54);
+ vortex_Eqlzr_SetA3dBypassGain(vortex, 0, 0);
+ vortex_EqHw_Enable(vortex);
+}
+
+static void vortex_Eqlzr_shutdown(vortex_t * vortex)
+{
+ vortex_Eqlzr_ShutDownA3d(vortex);
+ vortex_EqHw_ProgramPipe(vortex);
+ vortex_EqHw_Disable(vortex);
+}
+
+/* ALSA interface */
+
+/* Control interface */
+#define snd_vortex_eqtoggle_info snd_ctl_boolean_mono_info
+
+static int
+snd_vortex_eqtoggle_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ eqlzr_t *eq = &(vortex->eq);
+ //int i = kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = eq->this54 ? 0 : 1;
+
+ return 0;
+}
+
+static int
+snd_vortex_eqtoggle_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ eqlzr_t *eq = &(vortex->eq);
+ //int i = kcontrol->private_value;
+
+ eq->this54 = ucontrol->value.integer.value[0] ? 0 : 1;
+ vortex_Eqlzr_SetBypass(vortex, eq->this54);
+
+ return 1; /* Allways changes */
+}
+
+static struct snd_kcontrol_new vortex_eqtoggle_kcontrol __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "EQ Enable",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0,
+ .info = snd_vortex_eqtoggle_info,
+ .get = snd_vortex_eqtoggle_get,
+ .put = snd_vortex_eqtoggle_put
+};
+
+static int
+snd_vortex_eq_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0x0000;
+ uinfo->value.integer.max = 0x7fff;
+ return 0;
+}
+
+static int
+snd_vortex_eq_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ int i = kcontrol->private_value;
+ u16 gainL = 0, gainR = 0;
+
+ vortex_Eqlzr_GetLeftGain(vortex, i, &gainL);
+ vortex_Eqlzr_GetRightGain(vortex, i, &gainR);
+ ucontrol->value.integer.value[0] = gainL;
+ ucontrol->value.integer.value[1] = gainR;
+ return 0;
+}
+
+static int
+snd_vortex_eq_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ int changed = 0, i = kcontrol->private_value;
+ u16 gainL = 0, gainR = 0;
+
+ vortex_Eqlzr_GetLeftGain(vortex, i, &gainL);
+ vortex_Eqlzr_GetRightGain(vortex, i, &gainR);
+
+ if (gainL != ucontrol->value.integer.value[0]) {
+ vortex_Eqlzr_SetLeftGain(vortex, i,
+ ucontrol->value.integer.value[0]);
+ changed = 1;
+ }
+ if (gainR != ucontrol->value.integer.value[1]) {
+ vortex_Eqlzr_SetRightGain(vortex, i,
+ ucontrol->value.integer.value[1]);
+ changed = 1;
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new vortex_eq_kcontrol __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = " .",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0,
+ .info = snd_vortex_eq_info,
+ .get = snd_vortex_eq_get,
+ .put = snd_vortex_eq_put
+};
+
+static int
+snd_vortex_peaks_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 20;
+ uinfo->value.integer.min = 0x0000;
+ uinfo->value.integer.max = 0x7fff;
+ return 0;
+}
+
+static int
+snd_vortex_peaks_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ int i, count = 0;
+ u16 peaks[20];
+
+ vortex_Eqlzr_GetAllPeaks(vortex, peaks, &count);
+ if (count != 20) {
+ printk(KERN_ERR "vortex: peak count error 20 != %d \n", count);
+ return -1;
+ }
+ for (i = 0; i < 20; i++)
+ ucontrol->value.integer.value[i] = peaks[i];
+
+ return 0;
+}
+
+static struct snd_kcontrol_new vortex_levels_kcontrol __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "EQ Peaks",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_vortex_peaks_info,
+ .get = snd_vortex_peaks_get,
+};
+
+/* EQ band gain labels. */
+static char *EqBandLabels[10] __devinitdata = {
+ "EQ0 31Hz\0",
+ "EQ1 63Hz\0",
+ "EQ2 125Hz\0",
+ "EQ3 250Hz\0",
+ "EQ4 500Hz\0",
+ "EQ5 1KHz\0",
+ "EQ6 2KHz\0",
+ "EQ7 4KHz\0",
+ "EQ8 8KHz\0",
+ "EQ9 16KHz\0",
+};
+
+/* ALSA driver entry points. Init and exit. */
+static int __devinit vortex_eq_init(vortex_t * vortex)
+{
+ struct snd_kcontrol *kcontrol;
+ int err, i;
+
+ vortex_Eqlzr_init(vortex);
+
+ if ((kcontrol =
+ snd_ctl_new1(&vortex_eqtoggle_kcontrol, vortex)) == NULL)
+ return -ENOMEM;
+ kcontrol->private_value = 0;
+ if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0)
+ return err;
+
+ /* EQ gain controls */
+ for (i = 0; i < 10; i++) {
+ if ((kcontrol =
+ snd_ctl_new1(&vortex_eq_kcontrol, vortex)) == NULL)
+ return -ENOMEM;
+ strcpy(kcontrol->id.name, EqBandLabels[i]);
+ kcontrol->private_value = i;
+ if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0)
+ return err;
+ //vortex->eqctrl[i] = kcontrol;
+ }
+ /* EQ band levels */
+ if ((kcontrol = snd_ctl_new1(&vortex_levels_kcontrol, vortex)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(vortex->card, kcontrol)) < 0)
+ return err;
+
+ return 0;
+}
+
+static int vortex_eq_free(vortex_t * vortex)
+{
+ /*
+ //FIXME: segfault because vortex->eqctrl[i] == 4
+ int i;
+ for (i=0; i<10; i++) {
+ if (vortex->eqctrl[i])
+ snd_ctl_remove(vortex->card, vortex->eqctrl[i]);
+ }
+ */
+ vortex_Eqlzr_shutdown(vortex);
+ return 0;
+}
+
+/* End */
diff --git a/sound/pci/au88x0/au88x0_eq.h b/sound/pci/au88x0/au88x0_eq.h
new file mode 100644
index 0000000..474cd00
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_eq.h
@@ -0,0 +1,43 @@
+#ifndef AU88X0_EQ_H
+#define AU88X0_EQ_H
+
+/***************************************************************************
+ * au88x0_eq.h
+ *
+ * Definitions and constant data for the Aureal Hardware EQ.
+ *
+ * Sun Jun 8 18:23:38 2003
+ * Author: Manuel Jander (mjander@users.sourceforge.net)
+ ****************************************************************************/
+
+typedef struct {
+ u16 LeftCoefs[50]; //0x4
+ u16 RightCoefs[50]; // 0x68
+ u16 LeftGains[10]; //0xd0
+ u16 RightGains[10]; //0xe4
+} auxxEqCoeffSet_t;
+
+typedef struct {
+ s32 this04; /* How many filters for each side (default = 10) */
+ s32 this08; /* inited to cero. Stereo flag? */
+} eqhw_t;
+
+typedef struct {
+ eqhw_t this04; /* CHwEq */
+ u16 this08; /* Bad codec flag ? SetBypassGain: bypass gain */
+ u16 this0a;
+ u16 this0c; /* SetBypassGain: bypass gain when this28 is not set. */
+ u16 this0e;
+
+ s32 this10; /* How many gains are used for each side (right or left). */
+ u16 this14_array[10]; /* SetLeftGainsTarget: Left (and right?) EQ gains */
+ s32 this28; /* flag related to EQ enabled or not. Gang flag ? */
+ s32 this54; /* SetBypass */
+ s32 this58;
+ s32 this5c;
+ /*0x60 */ auxxEqCoeffSet_t coefset;
+ /* 50 u16 word each channel. */
+ u16 this130[20]; /* Left and Right gains */
+} eqlzr_t;
+
+#endif
diff --git a/sound/pci/au88x0/au88x0_eqdata.c b/sound/pci/au88x0/au88x0_eqdata.c
new file mode 100644
index 0000000..ce8dca8
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_eqdata.c
@@ -0,0 +1,116 @@
+/* Data structs */
+
+static u16 asEqCoefsZeros[50] = {
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+};
+
+static u16 asEqCoefsPipes[64] = {
+ 0x0000, 0x0000,
+ 0x0000, 0x0666, 0x0000, 0x0000, 0x0666,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0666, 0x0000, 0x0000, 0x0666,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0666, 0x0000, 0x0000, 0x0666,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0666, 0x0000, 0x0000, 0x0666,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0666, 0x0000, 0x0000, 0x066a,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000
+};
+
+/* More coef sets can be found in the win2k "inf" file. */
+static auxxEqCoeffSet_t asEqCoefsNormal = {
+ .LeftCoefs = {
+ 0x7e60, 0xc19e, 0x0001, 0x0002, 0x0001,
+ 0x7fa0, 0xc05f, 0x004f, 0x0000, 0xffb1,
+ 0x7f3f, 0xc0bc, 0x00c2, 0x0000, 0xff3e,
+ 0x7e78, 0xc177, 0x011f, 0x0000, 0xfee1,
+ 0x7cd6, 0xc2e5, 0x025c, 0x0000, 0xfda4,
+ 0x7949, 0xc5aa, 0x0467, 0x0000, 0xfb99,
+ 0x7120, 0xcadf, 0x0864, 0x0000, 0xf79c,
+ 0x5d33, 0xd430, 0x0f7e, 0x0000, 0xf082,
+ 0x2beb, 0xe3ca, 0x1bd3, 0x0000, 0xe42d,
+ 0xd740, 0xf01d, 0x2ac5, 0x0000, 0xd53b},
+
+ .RightCoefs = {
+ 0x7e60, 0xc19e, 0x0001, 0x0002, 0x0001,
+ 0x7fa0, 0xc05f, 0x004f, 0x0000, 0xffb1,
+ 0x7f3f, 0xc0bc, 0x00c2, 0x0000, 0xff3e,
+ 0x7e78, 0xc177, 0x011f, 0x0000, 0xfee1,
+ 0x7cd6, 0xc2e5, 0x025c, 0x0000, 0xfda4,
+ 0x7949, 0xc5aa, 0x0467, 0x0000, 0xfb99,
+ 0x7120, 0xcadf, 0x0864, 0x0000, 0xf79c,
+ 0x5d33, 0xd430, 0x0f7e, 0x0000, 0xf082,
+ 0x2beb, 0xe3ca, 0x1bd3, 0x0000, 0xe42d,
+ 0xd740, 0xf01d, 0x2ac5, 0x0000, 0xd53b},
+
+ .LeftGains = {
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96,
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96},
+ .RightGains = {
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96,
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96}
+};
+
+static u16 eq_gains_normal[20] = {
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96,
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96,
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96,
+ 0x3e96, 0x3e96, 0x3e96, 0x3e96, 0x3e96
+};
+
+/* _rodatab60 */
+static u16 eq_gains_zero[10] = {
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000
+};
+
+/* _rodatab7c: ProgramPipe */
+static u16 eq_gains_current[12] = {
+ 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
+ 0x7fff,
+ 0x7fff, 0x7fff, 0x7fff
+};
+
+/* _rodatab78 */
+static u16 eq_states_zero[2] = { 0x0000, 0x0000 };
+
+static u16 asEqOutStateZeros[48] = {
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000
+};
+
+/*_rodataba0:*/
+static u16 eq_levels[64] = {
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000
+};
diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c
new file mode 100644
index 0000000..e291aa5
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_game.c
@@ -0,0 +1,132 @@
+/*
+ * Manuel Jander.
+ *
+ * Based on the work of:
+ * Vojtech Pavlik
+ * Raymond Ingles
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * Should you need to contact me, the author, you can do so either by
+ * e-mail - mail your message to <vojtech@suse.cz>, or by paper mail:
+ * Vojtech Pavlik, Ucitelska 1576, Prague 8, 182 00 Czech Republic
+ *
+ * Based 90% on Vojtech Pavlik pcigame driver.
+ * Merged and modified by Manuel Jander, for the OpenVortex
+ * driver. (email: mjander@embedded.cl).
+ */
+
+#include <linux/time.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include "au88x0.h"
+#include <linux/gameport.h>
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+
+#define VORTEX_GAME_DWAIT 20 /* 20 ms */
+
+static unsigned char vortex_game_read(struct gameport *gameport)
+{
+ vortex_t *vortex = gameport_get_port_data(gameport);
+ return hwread(vortex->mmio, VORTEX_GAME_LEGACY);
+}
+
+static void vortex_game_trigger(struct gameport *gameport)
+{
+ vortex_t *vortex = gameport_get_port_data(gameport);
+ hwwrite(vortex->mmio, VORTEX_GAME_LEGACY, 0xff);
+}
+
+static int
+vortex_game_cooked_read(struct gameport *gameport, int *axes, int *buttons)
+{
+ vortex_t *vortex = gameport_get_port_data(gameport);
+ int i;
+
+ *buttons = (~hwread(vortex->mmio, VORTEX_GAME_LEGACY) >> 4) & 0xf;
+
+ for (i = 0; i < 4; i++) {
+ axes[i] =
+ hwread(vortex->mmio, VORTEX_GAME_AXIS + (i * AXIS_SIZE));
+ if (axes[i] == AXIS_RANGE)
+ axes[i] = -1;
+ }
+ return 0;
+}
+
+static int vortex_game_open(struct gameport *gameport, int mode)
+{
+ vortex_t *vortex = gameport_get_port_data(gameport);
+
+ switch (mode) {
+ case GAMEPORT_MODE_COOKED:
+ hwwrite(vortex->mmio, VORTEX_CTRL2,
+ hwread(vortex->mmio,
+ VORTEX_CTRL2) | CTRL2_GAME_ADCMODE);
+ msleep(VORTEX_GAME_DWAIT);
+ return 0;
+ case GAMEPORT_MODE_RAW:
+ hwwrite(vortex->mmio, VORTEX_CTRL2,
+ hwread(vortex->mmio,
+ VORTEX_CTRL2) & ~CTRL2_GAME_ADCMODE);
+ return 0;
+ default:
+ return -1;
+ }
+
+ return 0;
+}
+
+static int __devinit vortex_gameport_register(vortex_t * vortex)
+{
+ struct gameport *gp;
+
+ vortex->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "vortex: cannot allocate memory for gameport\n");
+ return -ENOMEM;
+ };
+
+ gameport_set_name(gp, "AU88x0 Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(vortex->pci_dev));
+ gameport_set_dev_parent(gp, &vortex->pci_dev->dev);
+
+ gp->read = vortex_game_read;
+ gp->trigger = vortex_game_trigger;
+ gp->cooked_read = vortex_game_cooked_read;
+ gp->open = vortex_game_open;
+
+ gameport_set_port_data(gp, vortex);
+ gp->fuzz = 64;
+
+ gameport_register_port(gp);
+
+ return 0;
+}
+
+static void vortex_gameport_unregister(vortex_t * vortex)
+{
+ if (vortex->gameport) {
+ gameport_unregister_port(vortex->gameport);
+ vortex->gameport = NULL;
+ }
+}
+
+#else
+static inline int vortex_gameport_register(vortex_t * vortex) { return -ENOSYS; }
+static inline void vortex_gameport_unregister(vortex_t * vortex) { }
+#endif
diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c
new file mode 100644
index 0000000..c92f493
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_mixer.c
@@ -0,0 +1,32 @@
+/*
+ * Vortex Mixer support.
+ *
+ * There is much more than just the AC97 mixer...
+ *
+ */
+
+#include <linux/time.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include "au88x0.h"
+
+static int __devinit snd_vortex_mixer(vortex_t * vortex)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = vortex_codec_write,
+ .read = vortex_codec_read,
+ };
+
+ if ((err = snd_ac97_bus(vortex->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+ memset(&ac97, 0, sizeof(ac97));
+ // Intialize AC97 codec stuff.
+ ac97.private_data = vortex;
+ ac97.scaps = AC97_SCAP_NO_SPDIF;
+ err = snd_ac97_mixer(pbus, &ac97, &vortex->codec);
+ vortex->isquad = ((vortex->codec == NULL) ? 0 : (vortex->codec->ext_id&0x80));
+ return err;
+}
diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c
new file mode 100644
index 0000000..0dc8d25
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -0,0 +1,112 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of MPU-401 in UART mode
+ *
+ * Modified for the Aureal Vortex based Soundcards
+ * by Manuel Jander (mjande@embedded.cl).
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/mpu401.h>
+#include "au88x0.h"
+
+/* Check for mpu401 mmio support. */
+/* MPU401 legacy support is only provided as a emergency fallback *
+ * for older versions of ALSA. Its usage is strongly discouraged. */
+#ifndef MPU401_HW_AUREAL
+#define VORTEX_MPU401_LEGACY
+#endif
+
+/* Vortex MPU401 defines. */
+#define MIDI_CLOCK_DIV 0x61
+/* Standart MPU401 defines. */
+#define MPU401_RESET 0xff
+#define MPU401_ENTER_UART 0x3f
+#define MPU401_ACK 0xfe
+
+static int __devinit snd_vortex_midi(vortex_t * vortex)
+{
+ struct snd_rawmidi *rmidi;
+ int temp, mode;
+ struct snd_mpu401 *mpu;
+ unsigned long port;
+
+#ifdef VORTEX_MPU401_LEGACY
+ /* EnableHardCodedMPU401Port() */
+ /* Enable Legacy MIDI Interface port. */
+ port = (0x03 << 5); /* FIXME: static address. 0x330 */
+ temp =
+ (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) |
+ CTRL_MIDI_EN | port;
+ hwwrite(vortex->mmio, VORTEX_CTRL, temp);
+#else
+ /* Disable Legacy MIDI Interface port. */
+ temp =
+ (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) &
+ ~CTRL_MIDI_EN;
+ hwwrite(vortex->mmio, VORTEX_CTRL, temp);
+#endif
+ /* Mpu401UartInit() */
+ mode = 1;
+ temp = hwread(vortex->mmio, VORTEX_CTRL2) & 0xffff00cf;
+ temp |= (MIDI_CLOCK_DIV << 8) | ((mode >> 24) & 0xff) << 4;
+ hwwrite(vortex->mmio, VORTEX_CTRL2, temp);
+ hwwrite(vortex->mmio, VORTEX_MIDI_CMD, MPU401_RESET);
+
+ /* Check if anything is OK. */
+ temp = hwread(vortex->mmio, VORTEX_MIDI_DATA);
+ if (temp != MPU401_ACK /*0xfe */ ) {
+ printk(KERN_ERR "midi port doesn't acknowledge!\n");
+ return -ENODEV;
+ }
+ /* Enable MPU401 interrupts. */
+ hwwrite(vortex->mmio, VORTEX_IRQ_CTRL,
+ hwread(vortex->mmio, VORTEX_IRQ_CTRL) | IRQ_MIDI);
+
+ /* Create MPU401 instance. */
+#ifdef VORTEX_MPU401_LEGACY
+ if ((temp =
+ snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_MPU401, 0x330,
+ 0, 0, 0, &rmidi)) != 0) {
+ hwwrite(vortex->mmio, VORTEX_CTRL,
+ (hwread(vortex->mmio, VORTEX_CTRL) &
+ ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
+ return temp;
+ }
+#else
+ port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA);
+ if ((temp =
+ snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port,
+ MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO,
+ 0, 0, &rmidi)) != 0) {
+ hwwrite(vortex->mmio, VORTEX_CTRL,
+ (hwread(vortex->mmio, VORTEX_CTRL) &
+ ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
+ return temp;
+ }
+ mpu = rmidi->private_data;
+ mpu->cport = (unsigned long)(vortex->mmio + VORTEX_MIDI_CMD);
+#endif
+ /* Overwrite MIDI name */
+ snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI %d", CARD_NAME_SHORT , vortex->card->number);
+
+ vortex->rmidi = rmidi;
+ return 0;
+}
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
new file mode 100644
index 0000000..b9d2f20
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -0,0 +1,539 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+/*
+ * Vortex PCM ALSA driver.
+ *
+ * Supports ADB and WT DMA. Unfortunately, WT channels do not run yet.
+ * It remains stuck,and DMA transfers do not happen.
+ */
+#include <sound/asoundef.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "au88x0.h"
+
+#define VORTEX_PCM_TYPE(x) (x->name[40])
+
+/* hardware definition */
+static struct snd_pcm_hardware snd_vortex_playback_hw_adb = {
+ .info =
+ (SNDRV_PCM_INFO_MMAP | /* SNDRV_PCM_INFO_RESUME | */
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats =
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 5000,
+ .rate_max = 48000,
+ .channels_min = 1,
+#ifdef CHIP_AU8830
+ .channels_max = 4,
+#else
+ .channels_max = 2,
+#endif
+ .buffer_bytes_max = 0x10000,
+ .period_bytes_min = 0x1,
+ .period_bytes_max = 0x1000,
+ .periods_min = 2,
+ .periods_max = 32,
+};
+
+#ifndef CHIP_AU8820
+static struct snd_pcm_hardware snd_vortex_playback_hw_a3d = {
+ .info =
+ (SNDRV_PCM_INFO_MMAP | /* SNDRV_PCM_INFO_RESUME | */
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats =
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 5000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = 0x10000,
+ .period_bytes_min = 0x100,
+ .period_bytes_max = 0x1000,
+ .periods_min = 2,
+ .periods_max = 64,
+};
+#endif
+static struct snd_pcm_hardware snd_vortex_playback_hw_spdif = {
+ .info =
+ (SNDRV_PCM_INFO_MMAP | /* SNDRV_PCM_INFO_RESUME | */
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats =
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | SNDRV_PCM_FMTBIT_MU_LAW |
+ SNDRV_PCM_FMTBIT_A_LAW,
+ .rates =
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 32000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 0x10000,
+ .period_bytes_min = 0x100,
+ .period_bytes_max = 0x1000,
+ .periods_min = 2,
+ .periods_max = 64,
+};
+
+#ifndef CHIP_AU8810
+static struct snd_pcm_hardware snd_vortex_playback_hw_wt = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, // SNDRV_PCM_RATE_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 0x10000,
+ .period_bytes_min = 0x0400,
+ .period_bytes_max = 0x1000,
+ .periods_min = 2,
+ .periods_max = 64,
+};
+#endif
+/* open callback */
+static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
+{
+ vortex_t *vortex = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ /* Force equal size periods */
+ if ((err =
+ snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ /* Avoid PAGE_SIZE boundary to fall inside of a period. */
+ if ((err =
+ snd_pcm_hw_constraint_pow2(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0)
+ return err;
+
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
+#ifndef CHIP_AU8820
+ if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) {
+ runtime->hw = snd_vortex_playback_hw_a3d;
+ }
+#endif
+ if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_SPDIF) {
+ runtime->hw = snd_vortex_playback_hw_spdif;
+ switch (vortex->spdif_sr) {
+ case 32000:
+ runtime->hw.rates = SNDRV_PCM_RATE_32000;
+ break;
+ case 44100:
+ runtime->hw.rates = SNDRV_PCM_RATE_44100;
+ break;
+ case 48000:
+ runtime->hw.rates = SNDRV_PCM_RATE_48000;
+ break;
+ }
+ }
+ if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB
+ || VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_I2S)
+ runtime->hw = snd_vortex_playback_hw_adb;
+ substream->runtime->private_data = NULL;
+ }
+#ifndef CHIP_AU8810
+ else {
+ runtime->hw = snd_vortex_playback_hw_wt;
+ substream->runtime->private_data = NULL;
+ }
+#endif
+ return 0;
+}
+
+/* close callback */
+static int snd_vortex_pcm_close(struct snd_pcm_substream *substream)
+{
+ //vortex_t *chip = snd_pcm_substream_chip(substream);
+ stream_t *stream = (stream_t *) substream->runtime->private_data;
+
+ // the hardware-specific codes will be here
+ if (stream != NULL) {
+ stream->substream = NULL;
+ stream->nr_ch = 0;
+ }
+ substream->runtime->private_data = NULL;
+ return 0;
+}
+
+/* hw_params callback */
+static int
+snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ vortex_t *chip = snd_pcm_substream_chip(substream);
+ stream_t *stream = (stream_t *) (substream->runtime->private_data);
+ int err;
+
+ // Alloc buffer memory.
+ err =
+ snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ if (err < 0) {
+ printk(KERN_ERR "Vortex: pcm page alloc failed!\n");
+ return err;
+ }
+ /*
+ printk(KERN_INFO "Vortex: periods %d, period_bytes %d, channels = %d\n", params_periods(hw_params),
+ params_period_bytes(hw_params), params_channels(hw_params));
+ */
+ spin_lock_irq(&chip->lock);
+ // Make audio routes and config buffer DMA.
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
+ int dma, type = VORTEX_PCM_TYPE(substream->pcm);
+ /* Dealloc any routes. */
+ if (stream != NULL)
+ vortex_adb_allocroute(chip, stream->dma,
+ stream->nr_ch, stream->dir,
+ stream->type);
+ /* Alloc routes. */
+ dma =
+ vortex_adb_allocroute(chip, -1,
+ params_channels(hw_params),
+ substream->stream, type);
+ if (dma < 0) {
+ spin_unlock_irq(&chip->lock);
+ return dma;
+ }
+ stream = substream->runtime->private_data = &chip->dma_adb[dma];
+ stream->substream = substream;
+ /* Setup Buffers. */
+ vortex_adbdma_setbuffers(chip, dma,
+ params_period_bytes(hw_params),
+ params_periods(hw_params));
+ }
+#ifndef CHIP_AU8810
+ else {
+ /* if (stream != NULL)
+ vortex_wt_allocroute(chip, substream->number, 0); */
+ vortex_wt_allocroute(chip, substream->number,
+ params_channels(hw_params));
+ stream = substream->runtime->private_data =
+ &chip->dma_wt[substream->number];
+ stream->dma = substream->number;
+ stream->substream = substream;
+ vortex_wtdma_setbuffers(chip, substream->number,
+ params_period_bytes(hw_params),
+ params_periods(hw_params));
+ }
+#endif
+ spin_unlock_irq(&chip->lock);
+ return 0;
+}
+
+/* hw_free callback */
+static int snd_vortex_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ vortex_t *chip = snd_pcm_substream_chip(substream);
+ stream_t *stream = (stream_t *) (substream->runtime->private_data);
+
+ spin_lock_irq(&chip->lock);
+ // Delete audio routes.
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
+ if (stream != NULL)
+ vortex_adb_allocroute(chip, stream->dma,
+ stream->nr_ch, stream->dir,
+ stream->type);
+ }
+#ifndef CHIP_AU8810
+ else {
+ if (stream != NULL)
+ vortex_wt_allocroute(chip, stream->dma, 0);
+ }
+#endif
+ substream->runtime->private_data = NULL;
+ spin_unlock_irq(&chip->lock);
+
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare callback */
+static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ vortex_t *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ stream_t *stream = (stream_t *) substream->runtime->private_data;
+ int dma = stream->dma, fmt, dir;
+
+ // set up the hardware with the current configuration.
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = 1;
+ else
+ dir = 0;
+ fmt = vortex_alsafmt_aspfmt(runtime->format);
+ spin_lock_irq(&chip->lock);
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
+ vortex_adbdma_setmode(chip, dma, 1, dir, fmt, 0 /*? */ ,
+ 0);
+ vortex_adbdma_setstartbuffer(chip, dma, 0);
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_SPDIF)
+ vortex_adb_setsrc(chip, dma, runtime->rate, dir);
+ }
+#ifndef CHIP_AU8810
+ else {
+ vortex_wtdma_setmode(chip, dma, 1, fmt, 0, 0);
+ // FIXME: Set rate (i guess using vortex_wt_writereg() somehow).
+ vortex_wtdma_setstartbuffer(chip, dma, 0);
+ }
+#endif
+ spin_unlock_irq(&chip->lock);
+ return 0;
+}
+
+/* trigger callback */
+static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ vortex_t *chip = snd_pcm_substream_chip(substream);
+ stream_t *stream = (stream_t *) substream->runtime->private_data;
+ int dma = stream->dma;
+
+ spin_lock(&chip->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ // do something to start the PCM engine
+ //printk(KERN_INFO "vortex: start %d\n", dma);
+ stream->fifo_enabled = 1;
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
+ vortex_adbdma_resetup(chip, dma);
+ vortex_adbdma_startfifo(chip, dma);
+ }
+#ifndef CHIP_AU8810
+ else {
+ printk(KERN_INFO "vortex: wt start %d\n", dma);
+ vortex_wtdma_startfifo(chip, dma);
+ }
+#endif
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ // do something to stop the PCM engine
+ //printk(KERN_INFO "vortex: stop %d\n", dma);
+ stream->fifo_enabled = 0;
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT)
+ vortex_adbdma_pausefifo(chip, dma);
+ //vortex_adbdma_stopfifo(chip, dma);
+#ifndef CHIP_AU8810
+ else {
+ printk(KERN_INFO "vortex: wt stop %d\n", dma);
+ vortex_wtdma_stopfifo(chip, dma);
+ }
+#endif
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ //printk(KERN_INFO "vortex: pause %d\n", dma);
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT)
+ vortex_adbdma_pausefifo(chip, dma);
+#ifndef CHIP_AU8810
+ else
+ vortex_wtdma_pausefifo(chip, dma);
+#endif
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ //printk(KERN_INFO "vortex: resume %d\n", dma);
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT)
+ vortex_adbdma_resumefifo(chip, dma);
+#ifndef CHIP_AU8810
+ else
+ vortex_wtdma_resumefifo(chip, dma);
+#endif
+ break;
+ default:
+ spin_unlock(&chip->lock);
+ return -EINVAL;
+ }
+ spin_unlock(&chip->lock);
+ return 0;
+}
+
+/* pointer callback */
+static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ vortex_t *chip = snd_pcm_substream_chip(substream);
+ stream_t *stream = (stream_t *) substream->runtime->private_data;
+ int dma = stream->dma;
+ snd_pcm_uframes_t current_ptr = 0;
+
+ spin_lock(&chip->lock);
+ if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT)
+ current_ptr = vortex_adbdma_getlinearpos(chip, dma);
+#ifndef CHIP_AU8810
+ else
+ current_ptr = vortex_wtdma_getlinearpos(chip, dma);
+#endif
+ //printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr);
+ spin_unlock(&chip->lock);
+ return (bytes_to_frames(substream->runtime, current_ptr));
+}
+
+/* operators */
+static struct snd_pcm_ops snd_vortex_playback_ops = {
+ .open = snd_vortex_pcm_open,
+ .close = snd_vortex_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_vortex_pcm_hw_params,
+ .hw_free = snd_vortex_pcm_hw_free,
+ .prepare = snd_vortex_pcm_prepare,
+ .trigger = snd_vortex_pcm_trigger,
+ .pointer = snd_vortex_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/*
+* definitions of capture are omitted here...
+*/
+
+static char *vortex_pcm_prettyname[VORTEX_PCM_LAST] = {
+ "AU88x0 ADB",
+ "AU88x0 SPDIF",
+ "AU88x0 A3D",
+ "AU88x0 WT",
+ "AU88x0 I2S",
+};
+static char *vortex_pcm_name[VORTEX_PCM_LAST] = {
+ "adb",
+ "spdif",
+ "a3d",
+ "wt",
+ "i2s",
+};
+
+/* SPDIF kcontrol */
+
+static int snd_vortex_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_vortex_spdif_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS;
+ return 0;
+}
+
+static int snd_vortex_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.iec958.status[0] = 0x00;
+ ucontrol->value.iec958.status[1] = IEC958_AES1_CON_ORIGINAL|IEC958_AES1_CON_DIGDIGCONV_ID;
+ ucontrol->value.iec958.status[2] = 0x00;
+ switch (vortex->spdif_sr) {
+ case 32000: ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS_32000; break;
+ case 44100: ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS_44100; break;
+ case 48000: ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS_48000; break;
+ }
+ return 0;
+}
+
+static int snd_vortex_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ int spdif_sr = 48000;
+ switch (ucontrol->value.iec958.status[3] & IEC958_AES3_CON_FS) {
+ case IEC958_AES3_CON_FS_32000: spdif_sr = 32000; break;
+ case IEC958_AES3_CON_FS_44100: spdif_sr = 44100; break;
+ case IEC958_AES3_CON_FS_48000: spdif_sr = 48000; break;
+ }
+ if (spdif_sr == vortex->spdif_sr)
+ return 0;
+ vortex->spdif_sr = spdif_sr;
+ vortex_spdif_init(vortex, vortex->spdif_sr, 1);
+ return 1;
+}
+
+/* spdif controls */
+static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_vortex_spdif_info,
+ .get = snd_vortex_spdif_get,
+ .put = snd_vortex_spdif_put,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_vortex_spdif_info,
+ .get = snd_vortex_spdif_mask_get
+ },
+};
+
+/* create a pcm device */
+static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
+{
+ struct snd_pcm *pcm;
+ struct snd_kcontrol *kctl;
+ int i;
+ int err, nr_capt;
+
+ if (!chip || idx < 0 || idx >= VORTEX_PCM_LAST)
+ return -ENODEV;
+
+ /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the
+ * same dma engine. WT uses it own separate dma engine whcih cant capture. */
+ if (idx == VORTEX_PCM_ADB)
+ nr_capt = nr;
+ else
+ nr_capt = 0;
+ err = snd_pcm_new(chip->card, vortex_pcm_prettyname[idx], idx, nr,
+ nr_capt, &pcm);
+ if (err < 0)
+ return err;
+ strcpy(pcm->name, vortex_pcm_name[idx]);
+ chip->pcm[idx] = pcm;
+ // This is an evil hack, but it saves a lot of duplicated code.
+ VORTEX_PCM_TYPE(pcm) = idx;
+ pcm->private_data = chip;
+ /* set operators */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_vortex_playback_ops);
+ if (idx == VORTEX_PCM_ADB)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_vortex_playback_ops);
+
+ /* pre-allocation of Scatter-Gather buffers */
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci_dev),
+ 0x10000, 0x10000);
+
+ if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_SPDIF) {
+ for (i = 0; i < ARRAY_SIZE(snd_vortex_mixer_spdif); i++) {
+ kctl = snd_ctl_new1(&snd_vortex_mixer_spdif[i], chip);
+ if (!kctl)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(chip->card, kctl)) < 0)
+ return err;
+ }
+ }
+ return 0;
+}
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
new file mode 100644
index 0000000..978b856
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -0,0 +1,395 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+/*
+ * Someday its supposed to make use of the WT DMA engine
+ * for a Wavetable synthesizer.
+ */
+
+#include "au88x0.h"
+#include "au88x0_wt.h"
+
+static void vortex_fifo_setwtvalid(vortex_t * vortex, int fifo, int en);
+static void vortex_connection_adb_mixin(vortex_t * vortex, int en,
+ unsigned char channel,
+ unsigned char source,
+ unsigned char mixin);
+static void vortex_connection_mixin_mix(vortex_t * vortex, int en,
+ unsigned char mixin,
+ unsigned char mix, int a);
+static void vortex_fifo_wtinitialize(vortex_t * vortex, int fifo, int j);
+static int vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt,
+ u32 val);
+
+/* WT */
+
+/* Put 2 WT channels together for one stereo interlaced channel. */
+static void vortex_wt_setstereo(vortex_t * vortex, u32 wt, u32 stereo)
+{
+ int temp;
+
+ //temp = hwread(vortex->mmio, 0x80 + ((wt >> 0x5)<< 0xf) + (((wt & 0x1f) >> 1) << 2));
+ temp = hwread(vortex->mmio, WT_STEREO(wt));
+ temp = (temp & 0xfe) | (stereo & 1);
+ //hwwrite(vortex->mmio, 0x80 + ((wt >> 0x5)<< 0xf) + (((wt & 0x1f) >> 1) << 2), temp);
+ hwwrite(vortex->mmio, WT_STEREO(wt), temp);
+}
+
+/* Join to mixdown route. */
+static void vortex_wt_setdsout(vortex_t * vortex, u32 wt, int en)
+{
+ int temp;
+
+ /* There is one DSREG register for each bank (32 voices each). */
+ temp = hwread(vortex->mmio, WT_DSREG((wt >= 0x20) ? 1 : 0));
+ if (en)
+ temp |= (1 << (wt & 0x1f));
+ else
+ temp &= (1 << ~(wt & 0x1f));
+ hwwrite(vortex->mmio, WT_DSREG((wt >= 0x20) ? 1 : 0), temp);
+}
+
+/* Setup WT route. */
+static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch)
+{
+ wt_voice_t *voice = &(vortex->wt_voice[wt]);
+ int temp;
+
+ //FIXME: WT audio routing.
+ if (nr_ch) {
+ vortex_fifo_wtinitialize(vortex, wt, 1);
+ vortex_fifo_setwtvalid(vortex, wt, 1);
+ vortex_wt_setstereo(vortex, wt, nr_ch - 1);
+ } else
+ vortex_fifo_setwtvalid(vortex, wt, 0);
+
+ /* Set mixdown mode. */
+ vortex_wt_setdsout(vortex, wt, 1);
+ /* Set other parameter registers. */
+ hwwrite(vortex->mmio, WT_SRAMP(0), 0x880000);
+ //hwwrite(vortex->mmio, WT_GMODE(0), 0xffffffff);
+#ifdef CHIP_AU8830
+ hwwrite(vortex->mmio, WT_SRAMP(1), 0x880000);
+ //hwwrite(vortex->mmio, WT_GMODE(1), 0xffffffff);
+#endif
+ hwwrite(vortex->mmio, WT_PARM(wt, 0), 0);
+ hwwrite(vortex->mmio, WT_PARM(wt, 1), 0);
+ hwwrite(vortex->mmio, WT_PARM(wt, 2), 0);
+
+ temp = hwread(vortex->mmio, WT_PARM(wt, 3));
+ printk(KERN_DEBUG "vortex: WT PARM3: %x\n", temp);
+ //hwwrite(vortex->mmio, WT_PARM(wt, 3), temp);
+
+ hwwrite(vortex->mmio, WT_DELAY(wt, 0), 0);
+ hwwrite(vortex->mmio, WT_DELAY(wt, 1), 0);
+ hwwrite(vortex->mmio, WT_DELAY(wt, 2), 0);
+ hwwrite(vortex->mmio, WT_DELAY(wt, 3), 0);
+
+ printk(KERN_DEBUG "vortex: WT GMODE: %x\n", hwread(vortex->mmio, WT_GMODE(wt)));
+
+ hwwrite(vortex->mmio, WT_PARM(wt, 2), 0xffffffff);
+ hwwrite(vortex->mmio, WT_PARM(wt, 3), 0xcff1c810);
+
+ voice->parm0 = voice->parm1 = 0xcfb23e2f;
+ hwwrite(vortex->mmio, WT_PARM(wt, 0), voice->parm0);
+ hwwrite(vortex->mmio, WT_PARM(wt, 1), voice->parm1);
+ printk(KERN_DEBUG "vortex: WT GMODE 2 : %x\n", hwread(vortex->mmio, WT_GMODE(wt)));
+ return 0;
+}
+
+
+static void vortex_wt_connect(vortex_t * vortex, int en)
+{
+ int i, ii, mix;
+
+#define NR_WTROUTES 6
+#ifdef CHIP_AU8830
+#define NR_WTBLOCKS 2
+#else
+#define NR_WTBLOCKS 1
+#endif
+
+ for (i = 0; i < NR_WTBLOCKS; i++) {
+ for (ii = 0; ii < NR_WTROUTES; ii++) {
+ mix =
+ vortex_adb_checkinout(vortex,
+ vortex->fixed_res, en,
+ VORTEX_RESOURCE_MIXIN);
+ vortex->mixwt[(i * NR_WTROUTES) + ii] = mix;
+
+ vortex_route(vortex, en, 0x11,
+ ADB_WTOUT(i, ii + 0x20), ADB_MIXIN(mix));
+
+ vortex_connection_mixin_mix(vortex, en, mix,
+ vortex->mixplayb[ii % 2], 0);
+ if (VORTEX_IS_QUAD(vortex))
+ vortex_connection_mixin_mix(vortex, en,
+ mix,
+ vortex->mixplayb[2 +
+ (ii % 2)], 0);
+ }
+ }
+ for (i = 0; i < NR_WT; i++) {
+ hwwrite(vortex->mmio, WT_RUN(i), 1);
+ }
+}
+
+/* Read WT Register */
+#if 0
+static int vortex_wt_GetReg(vortex_t * vortex, char reg, int wt)
+{
+ //int eax, esi;
+
+ if (reg == 4) {
+ return hwread(vortex->mmio, WT_PARM(wt, 3));
+ }
+ if (reg == 7) {
+ return hwread(vortex->mmio, WT_GMODE(wt));
+ }
+
+ return 0;
+}
+
+/* WT hardware abstraction layer generic register interface. */
+static int
+vortex_wt_SetReg2(vortex_t * vortex, unsigned char reg, int wt,
+ u16 val)
+{
+ /*
+ int eax, edx;
+
+ if (wt >= NR_WT) // 0x40 -> NR_WT
+ return 0;
+
+ if ((reg - 0x20) > 0) {
+ if ((reg - 0x21) != 0)
+ return 0;
+ eax = ((((b & 0xff) << 0xb) + (edx & 0xff)) << 4) + 0x208; // param 2
+ } else {
+ eax = ((((b & 0xff) << 0xb) + (edx & 0xff)) << 4) + 0x20a; // param 3
+ }
+ hwwrite(vortex->mmio, eax, c);
+ */
+ return 1;
+}
+
+/*public: static void __thiscall CWTHal::SetReg(unsigned char,int,unsigned long) */
+#endif
+static int
+vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt,
+ u32 val)
+{
+ int ecx;
+
+ if ((reg == 5) || ((reg >= 7) && (reg <= 10)) || (reg == 0xc)) {
+ if (wt >= (NR_WT / NR_WT_PB)) {
+ printk
+ ("vortex: WT SetReg: bank out of range. reg=0x%x, wt=%d\n",
+ reg, wt);
+ return 0;
+ }
+ } else {
+ if (wt >= NR_WT) {
+ printk(KERN_ERR "vortex: WT SetReg: voice out of range\n");
+ return 0;
+ }
+ }
+ if (reg > 0xc)
+ return 0;
+
+ switch (reg) {
+ /* Voice specific parameters */
+ case 0: /* running */
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val);
+ hwwrite(vortex->mmio, WT_RUN(wt), val);
+ return 0xc;
+ break;
+ case 1: /* param 0 */
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val);
+ hwwrite(vortex->mmio, WT_PARM(wt, 0), val);
+ return 0xc;
+ break;
+ case 2: /* param 1 */
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val);
+ hwwrite(vortex->mmio, WT_PARM(wt, 1), val);
+ return 0xc;
+ break;
+ case 3: /* param 2 */
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val);
+ hwwrite(vortex->mmio, WT_PARM(wt, 2), val);
+ return 0xc;
+ break;
+ case 4: /* param 3 */
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val);
+ hwwrite(vortex->mmio, WT_PARM(wt, 3), val);
+ return 0xc;
+ break;
+ case 6: /* mute */
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val);
+ hwwrite(vortex->mmio, WT_MUTE(wt), val);
+ return 0xc;
+ break;
+ case 0xb:
+ { /* delay */
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val);
+ hwwrite(vortex->mmio, WT_DELAY(wt, 3), val);
+ hwwrite(vortex->mmio, WT_DELAY(wt, 2), val);
+ hwwrite(vortex->mmio, WT_DELAY(wt, 1), val);
+ hwwrite(vortex->mmio, WT_DELAY(wt, 0), val);
+ return 0xc;
+ }
+ break;
+ /* Global WT block parameters */
+ case 5: /* sramp */
+ ecx = WT_SRAMP(wt);
+ break;
+ case 8: /* aramp */
+ ecx = WT_ARAMP(wt);
+ break;
+ case 9: /* mramp */
+ ecx = WT_MRAMP(wt);
+ break;
+ case 0xa: /* ctrl */
+ ecx = WT_CTRL(wt);
+ break;
+ case 0xc: /* ds_reg */
+ ecx = WT_DSREG(wt);
+ break;
+ default:
+ return 0;
+ break;
+ }
+ //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val);
+ hwwrite(vortex->mmio, ecx, val);
+ return 1;
+}
+
+static void vortex_wt_init(vortex_t * vortex)
+{
+ u32 var4, var8, varc, var10 = 0, edi;
+
+ var10 &= 0xFFFFFFE3;
+ var10 |= 0x22;
+ var10 &= 0xFFFFFEBF;
+ var10 |= 0x80;
+ var10 |= 0x200;
+ var10 &= 0xfffffffe;
+ var10 &= 0xfffffbff;
+ var10 |= 0x1800;
+ // var10 = 0x1AA2
+ var4 = 0x10000000;
+ varc = 0x00830000;
+ var8 = 0x00830000;
+
+ /* Init Bank registers. */
+ for (edi = 0; edi < (NR_WT / NR_WT_PB); edi++) {
+ vortex_wt_SetReg(vortex, 0xc, edi, 0); /* ds_reg */
+ vortex_wt_SetReg(vortex, 0xa, edi, var10); /* ctrl */
+ vortex_wt_SetReg(vortex, 0x9, edi, var4); /* mramp */
+ vortex_wt_SetReg(vortex, 0x8, edi, varc); /* aramp */
+ vortex_wt_SetReg(vortex, 0x5, edi, var8); /* sramp */
+ }
+ /* Init Voice registers. */
+ for (edi = 0; edi < NR_WT; edi++) {
+ vortex_wt_SetReg(vortex, 0x4, edi, 0); /* param 3 0x20c */
+ vortex_wt_SetReg(vortex, 0x3, edi, 0); /* param 2 0x208 */
+ vortex_wt_SetReg(vortex, 0x2, edi, 0); /* param 1 0x204 */
+ vortex_wt_SetReg(vortex, 0x1, edi, 0); /* param 0 0x200 */
+ vortex_wt_SetReg(vortex, 0xb, edi, 0); /* delay 0x400 - 0x40c */
+ }
+ var10 |= 1;
+ for (edi = 0; edi < (NR_WT / NR_WT_PB); edi++)
+ vortex_wt_SetReg(vortex, 0xa, edi, var10); /* ctrl */
+}
+
+/* Extract of CAdbTopology::SetVolume(struct _ASPVOLUME *) */
+#if 0
+static void vortex_wt_SetVolume(vortex_t * vortex, int wt, int vol[])
+{
+ wt_voice_t *voice = &(vortex->wt_voice[wt]);
+ int ecx = vol[1], eax = vol[0];
+
+ /* This is pure guess */
+ voice->parm0 &= 0xff00ffff;
+ voice->parm0 |= (vol[0] & 0xff) << 0x10;
+ voice->parm1 &= 0xff00ffff;
+ voice->parm1 |= (vol[1] & 0xff) << 0x10;
+
+ /* This is real */
+ hwwrite(vortex, WT_PARM(wt, 0), voice->parm0);
+ hwwrite(vortex, WT_PARM(wt, 1), voice->parm0);
+
+ if (voice->this_1D0 & 4) {
+ eax >>= 8;
+ ecx = eax;
+ if (ecx < 0x80)
+ ecx = 0x7f;
+ voice->parm3 &= 0xFFFFC07F;
+ voice->parm3 |= (ecx & 0x7f) << 7;
+ voice->parm3 &= 0xFFFFFF80;
+ voice->parm3 |= (eax & 0x7f);
+ } else {
+ voice->parm3 &= 0xFFE03FFF;
+ voice->parm3 |= (eax & 0xFE00) << 5;
+ }
+
+ hwwrite(vortex, WT_PARM(wt, 3), voice->parm3);
+}
+
+/* Extract of CAdbTopology::SetFrequency(unsigned long arg_0) */
+static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr)
+{
+ wt_voice_t *voice = &(vortex->wt_voice[wt]);
+ u32 eax, edx;
+
+ //FIXME: 64 bit operation.
+ eax = ((sr << 0xf) * 0x57619F1) & 0xffffffff;
+ edx = (((sr << 0xf) * 0x57619F1)) >> 0x20;
+
+ edx >>= 0xa;
+ edx <<= 1;
+ if (edx) {
+ if (edx & 0x0FFF80000)
+ eax = 0x7fff;
+ else {
+ edx <<= 0xd;
+ eax = 7;
+ while ((edx & 0x80000000) == 0) {
+ edx <<= 1;
+ eax--;
+ if (eax == 0)
+ break;
+ }
+ if (eax)
+ edx <<= 1;
+ eax <<= 0xc;
+ edx >>= 0x14;
+ eax |= edx;
+ }
+ } else
+ eax = 0;
+ voice->parm0 &= 0xffff0001;
+ voice->parm0 |= (eax & 0x7fff) << 1;
+ voice->parm1 = voice->parm0 | 1;
+ // Wt: this_1D4
+ //AuWt::WriteReg((ulong)(this_1DC<<4)+0x200, (ulong)this_1E4);
+ //AuWt::WriteReg((ulong)(this_1DC<<4)+0x204, (ulong)this_1E8);
+ hwwrite(vortex->mmio, WT_PARM(wt, 0), voice->parm0);
+ hwwrite(vortex->mmio, WT_PARM(wt, 1), voice->parm1);
+}
+#endif
+
+/* End of File */
diff --git a/sound/pci/au88x0/au88x0_wt.h b/sound/pci/au88x0/au88x0_wt.h
new file mode 100644
index 0000000..38d98f8
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_wt.h
@@ -0,0 +1,65 @@
+/***************************************************************************
+ * WT register offsets.
+ *
+ * Wed Oct 22 13:50:20 2003
+ * Copyright 2003 mjander
+ * mjander@users.sourceforge.org
+ ****************************************************************************/
+#ifndef _AU88X0_WT_H
+#define _AU88X0_WT_H
+
+/* WT channels are grouped in banks. Each bank has 0x20 channels. */
+/* Bank register address boundary is 0x8000 */
+
+#define NR_WT_PB 0x20
+
+/* WT bank base register (as dword address). */
+#define WT_BAR(x) (((x)&0xffe0)<<0x8)
+#define WT_BANK(x) (x>>5)
+/* WT Bank registers */
+#define WT_CTRL(bank) (((((bank)&1)<<0xd) + 0x00)<<2) /* 0x0000 */
+#define WT_SRAMP(bank) (((((bank)&1)<<0xd) + 0x01)<<2) /* 0x0004 */
+#define WT_DSREG(bank) (((((bank)&1)<<0xd) + 0x02)<<2) /* 0x0008 */
+#define WT_MRAMP(bank) (((((bank)&1)<<0xd) + 0x03)<<2) /* 0x000c */
+#define WT_GMODE(bank) (((((bank)&1)<<0xd) + 0x04)<<2) /* 0x0010 */
+#define WT_ARAMP(bank) (((((bank)&1)<<0xd) + 0x05)<<2) /* 0x0014 */
+/* WT Voice registers */
+#define WT_STEREO(voice) ((WT_BAR(voice)+ 0x20 +(((voice)&0x1f)>>1))<<2) /* 0x0080 */
+#define WT_MUTE(voice) ((WT_BAR(voice)+ 0x40 +((voice)&0x1f))<<2) /* 0x0100 */
+#define WT_RUN(voice) ((WT_BAR(voice)+ 0x60 +((voice)&0x1f))<<2) /* 0x0180 */
+/* Some kind of parameters. */
+/* PARM0, PARM1 : Filter (0xFF000000), SampleRate (0x0000FFFF) */
+/* PARM2, PARM3 : Still unknown */
+#define WT_PARM(x,y) (((WT_BAR(x))+ 0x80 +(((x)&0x1f)<<2)+(y))<<2) /* 0x0200 */
+#define WT_DELAY(x,y) (((WT_BAR(x))+ 0x100 +(((x)&0x1f)<<2)+(y))<<2) /* 0x0400 */
+
+/* Numeric indexes used by SetReg() and GetReg() */
+#if 0
+enum {
+ run = 0, /* 0 W 1:run 0:stop */
+ parm0, /* 1 W filter, samplerate */
+ parm1, /* 2 W filter, samplerate */
+ parm2, /* 3 W */
+ parm3, /* 4 RW volume. This value is calculated using floating point ops. */
+ sramp, /* 5 W */
+ mute, /* 6 W 1:mute, 0:unmute */
+ gmode, /* 7 RO Looks like only bit0 is used. */
+ aramp, /* 8 W */
+ mramp, /* 9 W */
+ ctrl, /* a W */
+ delay, /* b W All 4 values are written at once with same value. */
+ dsreg, /* c (R)W */
+} wt_reg;
+#endif
+
+typedef struct {
+ u32 parm0; /* this_1E4 */
+ u32 parm1; /* this_1E8 */
+ u32 parm2; /* this_1EC */
+ u32 parm3; /* this_1F0 */
+ u32 this_1D0;
+} wt_voice_t;
+
+#endif /* _AU88X0_WT_H */
+
+/* End of file */
diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c
new file mode 100644
index 0000000..b4151e2
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_xtalk.c
@@ -0,0 +1,770 @@
+/***************************************************************************
+ * au88x0_cxtalk.c
+ *
+ * Wed Nov 19 16:29:47 2003
+ * Copyright 2003 mjander
+ * mjander@users.sourceforge.org
+ ****************************************************************************/
+
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#include "au88x0_xtalk.h"
+
+/* Data (a whole lot of data.... ) */
+
+static short const sXtalkWideKLeftEq = 0x269C;
+static short const sXtalkWideKRightEq = 0x269C;
+static short const sXtalkWideKLeftXt = 0xF25E;
+static short const sXtalkWideKRightXt = 0xF25E;
+static short const sXtalkWideShiftLeftEq = 1;
+static short const sXtalkWideShiftRightEq = 1;
+static short const sXtalkWideShiftLeftXt = 0;
+static short const sXtalkWideShiftRightXt = 0;
+static unsigned short const wXtalkWideLeftDelay = 0xd;
+static unsigned short const wXtalkWideRightDelay = 0xd;
+static short const sXtalkNarrowKLeftEq = 0x468D;
+static short const sXtalkNarrowKRightEq = 0x468D;
+static short const sXtalkNarrowKLeftXt = 0xF82E;
+static short const sXtalkNarrowKRightXt = 0xF82E;
+static short const sXtalkNarrowShiftLeftEq = 0x3;
+static short const sXtalkNarrowShiftRightEq = 0x3;
+static short const sXtalkNarrowShiftLeftXt = 0;
+static short const sXtalkNarrowShiftRightXt = 0;
+static unsigned short const wXtalkNarrowLeftDelay = 0x7;
+static unsigned short const wXtalkNarrowRightDelay = 0x7;
+
+static xtalk_gains_t const asXtalkGainsDefault = {
+ 0x4000, 0x4000, 4000, 0x4000, 4000, 0x4000, 4000, 0x4000, 4000,
+ 0x4000
+};
+
+static xtalk_gains_t const asXtalkGainsTest = {
+ 0x8000, 0x7FFF, 0, 0xFFFF, 0x0001, 0xC000, 0x4000, 0xFFFE, 0x0002,
+ 0
+};
+static xtalk_gains_t const asXtalkGains1Chan = {
+ 0x7FFF, 0, 0, 0, 0x7FFF, 0, 0, 0, 0, 0
+};
+
+// Input gain for 4 A3D slices. One possible input pair is left zero.
+static xtalk_gains_t const asXtalkGainsAllChan = {
+ 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF, 0, 0x7FFF, 0x7FFF, 0x7FFF, 0x7FFF,
+ 0
+ //0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff,0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff
+};
+static xtalk_gains_t const asXtalkGainsZeros;
+
+static xtalk_dline_t const alXtalkDlineZeros;
+static xtalk_dline_t const alXtalkDlineTest = {
+ 0xFC18, 0x03E8FFFF, 0x186A0, 0x7960FFFE, 1, 0xFFFFFFFF,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0
+};
+
+static xtalk_instate_t const asXtalkInStateZeros;
+static xtalk_instate_t const asXtalkInStateTest =
+ { 0xFF80, 0x0080, 0xFFFF, 0x0001 };
+static xtalk_state_t const asXtalkOutStateZeros;
+
+static short const sDiamondKLeftEq = 0x401d;
+static short const sDiamondKRightEq = 0x401d;
+static short const sDiamondKLeftXt = 0xF90E;
+static short const sDiamondKRightXt = 0xF90E;
+static short const sDiamondShiftLeftEq = 1; /* 0xF90E Is this a bug ??? */
+static short const sDiamondShiftRightEq = 1;
+static short const sDiamondShiftLeftXt = 0;
+static short const sDiamondShiftRightXt = 0;
+static unsigned short const wDiamondLeftDelay = 0xb;
+static unsigned short const wDiamondRightDelay = 0xb;
+
+static xtalk_coefs_t const asXtalkWideCoefsLeftEq = {
+ {0xEC4C, 0xDCE9, 0xFDC2, 0xFEEC, 0},
+ {0x5F60, 0xCBCB, 0xFC26, 0x0305, 0},
+ {0x340B, 0xf504, 0x6CE8, 0x0D23, 0x00E4},
+ {0xD500, 0x8D76, 0xACC7, 0x5B05, 0x00FA},
+ {0x7F04, 0xC0FA, 0x0263, 0xFDA2, 0}
+};
+static xtalk_coefs_t const asXtalkWideCoefsRightEq = {
+ {0xEC4C, 0xDCE9, 0xFDC2, 0xFEEC, 0},
+ {0x5F60, 0xCBCB, 0xFC26, 0x0305, 0},
+ {0x340B, 0xF504, 0x6CE8, 0x0D23, 0x00E4},
+ {0xD500, 0x8D76, 0xACC7, 0x5B05, 0x00FA},
+ {0x7F04, 0xC0FA, 0x0263, 0xFDA2, 0}
+};
+static xtalk_coefs_t const asXtalkWideCoefsLeftXt = {
+ {0x86C3, 0x7B55, 0x89C3, 0x005B, 0x0047},
+ {0x6000, 0x206A, 0xC6CA, 0x40FF, 0},
+ {0x1100, 0x1164, 0xA1D7, 0x90FC, 0x0001},
+ {0xDC00, 0x9E77, 0xB8C7, 0x0AFF, 0},
+ {0, 0, 0, 0, 0}
+};
+static xtalk_coefs_t const asXtalkWideCoefsRightXt = {
+ {0x86C3, 0x7B55, 0x89C3, 0x005B, 0x0047},
+ {0x6000, 0x206A, 0xC6CA, 0x40FF, 0},
+ {0x1100, 0x1164, 0xA1D7, 0x90FC, 0x0001},
+ {0xDC00, 0x9E77, 0xB8C7, 0x0AFF, 0},
+ {0, 0, 0, 0, 0}
+};
+static xtalk_coefs_t const asXtalkNarrowCoefsLeftEq = {
+ {0x50B5, 0xD07C, 0x026D, 0xFD21, 0},
+ {0x460F, 0xE44F, 0xF75E, 0xEFA6, 0},
+ {0x556D, 0xDCAB, 0x2098, 0xF0F2, 0},
+ {0x7E03, 0xC1F0, 0x007D, 0xFF89, 0},
+ {0x383E, 0xFD9D, 0xB278, 0x4547, 0}
+};
+
+static xtalk_coefs_t const asXtalkNarrowCoefsRightEq = {
+ {0x50B5, 0xD07C, 0x026D, 0xFD21, 0},
+ {0x460F, 0xE44F, 0xF75E, 0xEFA6, 0},
+ {0x556D, 0xDCAB, 0x2098, 0xF0F2, 0},
+ {0x7E03, 0xC1F0, 0x007D, 0xFF89, 0},
+ {0x383E, 0xFD9D, 0xB278, 0x4547, 0}
+};
+
+static xtalk_coefs_t const asXtalkNarrowCoefsLeftXt = {
+ {0x3CB2, 0xDF49, 0xF6EA, 0x095B, 0},
+ {0x6777, 0xC915, 0xFEAF, 0x00B1, 0},
+ {0x7762, 0xC7D9, 0x025B, 0xFDA6, 0},
+ {0x6B7A, 0xD2AA, 0xF2FB, 0x0B64, 0},
+ {0, 0, 0, 0, 0}
+};
+
+static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = {
+ {0x3CB2, 0xDF49, 0xF6EA, 0x095B, 0},
+ {0x6777, 0xC915, 0xFEAF, 0x00B1, 0},
+ {0x7762, 0xC7D9, 0x025B, 0xFDA6, 0},
+ {0x6B7A, 0xD2AA, 0xF2FB, 0x0B64, 0},
+ {0, 0, 0, 0, 0}
+};
+
+static xtalk_coefs_t const asXtalkCoefsZeros;
+static xtalk_coefs_t const asXtalkCoefsPipe = {
+ {0, 0, 0x0FA0, 0, 0},
+ {0, 0, 0x0FA0, 0, 0},
+ {0, 0, 0x0FA0, 0, 0},
+ {0, 0, 0x0FA0, 0, 0},
+ {0, 0, 0x1180, 0, 0},
+};
+static xtalk_coefs_t const asXtalkCoefsNegPipe = {
+ {0, 0, 0xF380, 0, 0},
+ {0, 0, 0xF380, 0, 0},
+ {0, 0, 0xF380, 0, 0},
+ {0, 0, 0xF380, 0, 0},
+ {0, 0, 0xF200, 0, 0}
+};
+
+static xtalk_coefs_t const asXtalkCoefsNumTest = {
+ {0, 0, 0xF380, 0x8000, 0x6D60},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0}
+};
+
+static xtalk_coefs_t const asXtalkCoefsDenTest = {
+ {0xC000, 0x2000, 0x4000, 0, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0}
+};
+
+static xtalk_state_t const asXtalkOutStateTest = {
+ {0x7FFF, 0x0004, 0xFFFC, 0},
+ {0xFE00, 0x0008, 0xFFF8, 0x4000},
+ {0x200, 0x0010, 0xFFF0, 0xC000},
+ {0x8000, 0x0020, 0xFFE0, 0},
+ {0, 0, 0, 0}
+};
+
+static xtalk_coefs_t const asDiamondCoefsLeftEq = {
+ {0x0F1E, 0x2D05, 0xF8E3, 0x07C8, 0},
+ {0x45E2, 0xCA51, 0x0448, 0xFCE7, 0},
+ {0xA93E, 0xDBD5, 0x022C, 0x028A, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0}
+};
+
+static xtalk_coefs_t const asDiamondCoefsRightEq = {
+ {0x0F1E, 0x2D05, 0xF8E3, 0x07C8, 0},
+ {0x45E2, 0xCA51, 0x0448, 0xFCE7, 0},
+ {0xA93E, 0xDBD5, 0x022C, 0x028A, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0}
+};
+
+static xtalk_coefs_t const asDiamondCoefsLeftXt = {
+ {0x3B50, 0xFE08, 0xF959, 0x0060, 0},
+ {0x9FCB, 0xD8F1, 0x00A2, 0x003A, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0}
+};
+
+static xtalk_coefs_t const asDiamondCoefsRightXt = {
+ {0x3B50, 0xFE08, 0xF959, 0x0060, 0},
+ {0x9FCB, 0xD8F1, 0x00A2, 0x003A, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0},
+ {0, 0, 0, 0, 0}
+};
+
+ /**/
+/* XTalk EQ and XT */
+static void
+vortex_XtalkHw_SetLeftEQ(vortex_t * vortex, short arg_0, short arg_4,
+ xtalk_coefs_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x24200 + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x24204 + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x24208 + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x2420c + i * 0x24, coefs[i][3]);
+ hwwrite(vortex->mmio, 0x24210 + i * 0x24, coefs[i][4]);
+ }
+ hwwrite(vortex->mmio, 0x24538, arg_0 & 0xffff);
+ hwwrite(vortex->mmio, 0x2453C, arg_4 & 0xffff);
+}
+
+static void
+vortex_XtalkHw_SetRightEQ(vortex_t * vortex, short arg_0, short arg_4,
+ xtalk_coefs_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x242b4 + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x242b8 + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x242bc + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x242c0 + i * 0x24, coefs[i][3]);
+ hwwrite(vortex->mmio, 0x242c4 + i * 0x24, coefs[i][4]);
+ }
+ hwwrite(vortex->mmio, 0x24540, arg_0 & 0xffff);
+ hwwrite(vortex->mmio, 0x24544, arg_4 & 0xffff);
+}
+
+static void
+vortex_XtalkHw_SetLeftXT(vortex_t * vortex, short arg_0, short arg_4,
+ xtalk_coefs_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x24368 + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x2436c + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x24370 + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x24374 + i * 0x24, coefs[i][3]);
+ hwwrite(vortex->mmio, 0x24378 + i * 0x24, coefs[i][4]);
+ }
+ hwwrite(vortex->mmio, 0x24548, arg_0 & 0xffff);
+ hwwrite(vortex->mmio, 0x2454C, arg_4 & 0xffff);
+}
+
+static void
+vortex_XtalkHw_SetRightXT(vortex_t * vortex, short arg_0, short arg_4,
+ xtalk_coefs_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x2441C + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x24420 + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x24424 + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x24428 + i * 0x24, coefs[i][3]);
+ hwwrite(vortex->mmio, 0x2442C + i * 0x24, coefs[i][4]);
+ }
+ hwwrite(vortex->mmio, 0x24550, arg_0 & 0xffff);
+ hwwrite(vortex->mmio, 0x24554, arg_4 & 0xffff);
+}
+
+static void
+vortex_XtalkHw_SetLeftEQStates(vortex_t * vortex,
+ xtalk_instate_t const arg_0,
+ xtalk_state_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x24214 + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x24218 + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x2421C + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x24220 + i * 0x24, coefs[i][3]);
+ }
+ hwwrite(vortex->mmio, 0x244F8 + i * 0x24, arg_0[0]);
+ hwwrite(vortex->mmio, 0x244FC + i * 0x24, arg_0[1]);
+ hwwrite(vortex->mmio, 0x24500 + i * 0x24, arg_0[2]);
+ hwwrite(vortex->mmio, 0x24504 + i * 0x24, arg_0[3]);
+}
+
+static void
+vortex_XtalkHw_SetRightEQStates(vortex_t * vortex,
+ xtalk_instate_t const arg_0,
+ xtalk_state_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x242C8 + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x242CC + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x242D0 + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x244D4 + i * 0x24, coefs[i][3]);
+ }
+ hwwrite(vortex->mmio, 0x24508 + i * 0x24, arg_0[0]);
+ hwwrite(vortex->mmio, 0x2450C + i * 0x24, arg_0[1]);
+ hwwrite(vortex->mmio, 0x24510 + i * 0x24, arg_0[2]);
+ hwwrite(vortex->mmio, 0x24514 + i * 0x24, arg_0[3]);
+}
+
+static void
+vortex_XtalkHw_SetLeftXTStates(vortex_t * vortex,
+ xtalk_instate_t const arg_0,
+ xtalk_state_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x2437C + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x24380 + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x24384 + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x24388 + i * 0x24, coefs[i][3]);
+ }
+ hwwrite(vortex->mmio, 0x24518 + i * 0x24, arg_0[0]);
+ hwwrite(vortex->mmio, 0x2451C + i * 0x24, arg_0[1]);
+ hwwrite(vortex->mmio, 0x24520 + i * 0x24, arg_0[2]);
+ hwwrite(vortex->mmio, 0x24524 + i * 0x24, arg_0[3]);
+}
+
+static void
+vortex_XtalkHw_SetRightXTStates(vortex_t * vortex,
+ xtalk_instate_t const arg_0,
+ xtalk_state_t const coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ hwwrite(vortex->mmio, 0x24430 + i * 0x24, coefs[i][0]);
+ hwwrite(vortex->mmio, 0x24434 + i * 0x24, coefs[i][1]);
+ hwwrite(vortex->mmio, 0x24438 + i * 0x24, coefs[i][2]);
+ hwwrite(vortex->mmio, 0x2443C + i * 0x24, coefs[i][3]);
+ }
+ hwwrite(vortex->mmio, 0x24528 + i * 0x24, arg_0[0]);
+ hwwrite(vortex->mmio, 0x2452C + i * 0x24, arg_0[1]);
+ hwwrite(vortex->mmio, 0x24530 + i * 0x24, arg_0[2]);
+ hwwrite(vortex->mmio, 0x24534 + i * 0x24, arg_0[3]);
+}
+
+#if 0
+static void
+vortex_XtalkHw_GetLeftEQ(vortex_t * vortex, short *arg_0, short *arg_4,
+ xtalk_coefs_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x24200 + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x24204 + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x24208 + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x2420c + i * 0x24);
+ coefs[i][4] = hwread(vortex->mmio, 0x24210 + i * 0x24);
+ }
+ *arg_0 = hwread(vortex->mmio, 0x24538) & 0xffff;
+ *arg_4 = hwread(vortex->mmio, 0x2453c) & 0xffff;
+}
+
+static void
+vortex_XtalkHw_GetRightEQ(vortex_t * vortex, short *arg_0, short *arg_4,
+ xtalk_coefs_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x242b4 + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x242b8 + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x242bc + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x242c0 + i * 0x24);
+ coefs[i][4] = hwread(vortex->mmio, 0x242c4 + i * 0x24);
+ }
+ *arg_0 = hwread(vortex->mmio, 0x24540) & 0xffff;
+ *arg_4 = hwread(vortex->mmio, 0x24544) & 0xffff;
+}
+
+static void
+vortex_XtalkHw_GetLeftXT(vortex_t * vortex, short *arg_0, short *arg_4,
+ xtalk_coefs_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x24368 + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x2436C + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x24370 + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x24374 + i * 0x24);
+ coefs[i][4] = hwread(vortex->mmio, 0x24378 + i * 0x24);
+ }
+ *arg_0 = hwread(vortex->mmio, 0x24548) & 0xffff;
+ *arg_4 = hwread(vortex->mmio, 0x2454C) & 0xffff;
+}
+
+static void
+vortex_XtalkHw_GetRightXT(vortex_t * vortex, short *arg_0, short *arg_4,
+ xtalk_coefs_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x2441C + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x24420 + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x24424 + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x24428 + i * 0x24);
+ coefs[i][4] = hwread(vortex->mmio, 0x2442C + i * 0x24);
+ }
+ *arg_0 = hwread(vortex->mmio, 0x24550) & 0xffff;
+ *arg_4 = hwread(vortex->mmio, 0x24554) & 0xffff;
+}
+
+static void
+vortex_XtalkHw_GetLeftEQStates(vortex_t * vortex, xtalk_instate_t arg_0,
+ xtalk_state_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x24214 + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x24218 + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x2421C + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x24220 + i * 0x24);
+ }
+ arg_0[0] = hwread(vortex->mmio, 0x244F8 + i * 0x24);
+ arg_0[1] = hwread(vortex->mmio, 0x244FC + i * 0x24);
+ arg_0[2] = hwread(vortex->mmio, 0x24500 + i * 0x24);
+ arg_0[3] = hwread(vortex->mmio, 0x24504 + i * 0x24);
+}
+
+static void
+vortex_XtalkHw_GetRightEQStates(vortex_t * vortex, xtalk_instate_t arg_0,
+ xtalk_state_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x242C8 + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x242CC + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x242D0 + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x242D4 + i * 0x24);
+ }
+ arg_0[0] = hwread(vortex->mmio, 0x24508 + i * 0x24);
+ arg_0[1] = hwread(vortex->mmio, 0x2450C + i * 0x24);
+ arg_0[2] = hwread(vortex->mmio, 0x24510 + i * 0x24);
+ arg_0[3] = hwread(vortex->mmio, 0x24514 + i * 0x24);
+}
+
+static void
+vortex_XtalkHw_GetLeftXTStates(vortex_t * vortex, xtalk_instate_t arg_0,
+ xtalk_state_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x2437C + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x24380 + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x24384 + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x24388 + i * 0x24);
+ }
+ arg_0[0] = hwread(vortex->mmio, 0x24518 + i * 0x24);
+ arg_0[1] = hwread(vortex->mmio, 0x2451C + i * 0x24);
+ arg_0[2] = hwread(vortex->mmio, 0x24520 + i * 0x24);
+ arg_0[3] = hwread(vortex->mmio, 0x24524 + i * 0x24);
+}
+
+static void
+vortex_XtalkHw_GetRightXTStates(vortex_t * vortex, xtalk_instate_t arg_0,
+ xtalk_state_t coefs)
+{
+ int i;
+
+ for (i = 0; i < 5; i++) {
+ coefs[i][0] = hwread(vortex->mmio, 0x24430 + i * 0x24);
+ coefs[i][1] = hwread(vortex->mmio, 0x24434 + i * 0x24);
+ coefs[i][2] = hwread(vortex->mmio, 0x24438 + i * 0x24);
+ coefs[i][3] = hwread(vortex->mmio, 0x2443C + i * 0x24);
+ }
+ arg_0[0] = hwread(vortex->mmio, 0x24528 + i * 0x24);
+ arg_0[1] = hwread(vortex->mmio, 0x2452C + i * 0x24);
+ arg_0[2] = hwread(vortex->mmio, 0x24530 + i * 0x24);
+ arg_0[3] = hwread(vortex->mmio, 0x24534 + i * 0x24);
+}
+
+#endif
+/* Gains */
+
+static void
+vortex_XtalkHw_SetGains(vortex_t * vortex, xtalk_gains_t const gains)
+{
+ int i;
+
+ for (i = 0; i < XTGAINS_SZ; i++) {
+ hwwrite(vortex->mmio, 0x244D0 + (i * 4), gains[i]);
+ }
+}
+
+static void
+vortex_XtalkHw_SetGainsAllChan(vortex_t * vortex)
+{
+ vortex_XtalkHw_SetGains(vortex, asXtalkGainsAllChan);
+}
+
+#if 0
+static void vortex_XtalkHw_GetGains(vortex_t * vortex, xtalk_gains_t gains)
+{
+ int i;
+
+ for (i = 0; i < XTGAINS_SZ; i++)
+ gains[i] = hwread(vortex->mmio, 0x244D0 + i * 4);
+}
+
+#endif
+/* Delay parameters */
+
+static void
+vortex_XtalkHw_SetDelay(vortex_t * vortex, unsigned short right,
+ unsigned short left)
+{
+ u32 esp0 = 0;
+
+ esp0 &= 0x1FFFFFFF;
+ esp0 |= 0xA0000000;
+ esp0 = (esp0 & 0xffffE0ff) | ((right & 0x1F) << 8);
+ esp0 = (esp0 & 0xfffc1fff) | ((left & 0x1F) << 0xd);
+
+ hwwrite(vortex->mmio, 0x24660, esp0);
+}
+
+static void
+vortex_XtalkHw_SetLeftDline(vortex_t * vortex, xtalk_dline_t const dline)
+{
+ int i;
+
+ for (i = 0; i < 0x20; i++) {
+ hwwrite(vortex->mmio, 0x24000 + (i << 2), dline[i] & 0xffff);
+ hwwrite(vortex->mmio, 0x24080 + (i << 2), dline[i] >> 0x10);
+ }
+}
+
+static void
+vortex_XtalkHw_SetRightDline(vortex_t * vortex, xtalk_dline_t const dline)
+{
+ int i;
+
+ for (i = 0; i < 0x20; i++) {
+ hwwrite(vortex->mmio, 0x24100 + (i << 2), dline[i] & 0xffff);
+ hwwrite(vortex->mmio, 0x24180 + (i << 2), dline[i] >> 0x10);
+ }
+}
+
+#if 0
+static void
+vortex_XtalkHw_GetDelay(vortex_t * vortex, unsigned short *right,
+ unsigned short *left)
+{
+ int esp0;
+
+ esp0 = hwread(vortex->mmio, 0x24660);
+ *right = (esp0 >> 8) & 0x1f;
+ *left = (esp0 >> 0xd) & 0x1f;
+}
+
+static void vortex_XtalkHw_GetLeftDline(vortex_t * vortex, xtalk_dline_t dline)
+{
+ int i;
+
+ for (i = 0; i < 0x20; i++) {
+ dline[i] =
+ (hwread(vortex->mmio, 0x24000 + (i << 2)) & 0xffff) |
+ (hwread(vortex->mmio, 0x24080 + (i << 2)) << 0x10);
+ }
+}
+
+static void vortex_XtalkHw_GetRightDline(vortex_t * vortex, xtalk_dline_t dline)
+{
+ int i;
+
+ for (i = 0; i < 0x20; i++) {
+ dline[i] =
+ (hwread(vortex->mmio, 0x24100 + (i << 2)) & 0xffff) |
+ (hwread(vortex->mmio, 0x24180 + (i << 2)) << 0x10);
+ }
+}
+
+#endif
+/* Control/Global stuff */
+
+#if 0
+static void vortex_XtalkHw_SetControlReg(vortex_t * vortex, u32 ctrl)
+{
+ hwwrite(vortex->mmio, 0x24660, ctrl);
+}
+static void vortex_XtalkHw_GetControlReg(vortex_t * vortex, u32 *ctrl)
+{
+ *ctrl = hwread(vortex->mmio, 0x24660);
+}
+#endif
+static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, u32 sr)
+{
+ u32 temp;
+
+ temp = (hwread(vortex->mmio, 0x24660) & 0x1FFFFFFF) | 0xC0000000;
+ temp = (temp & 0xffffff07) | ((sr & 0x1f) << 3);
+ hwwrite(vortex->mmio, 0x24660, temp);
+}
+
+#if 0
+static void vortex_XtalkHw_GetSampleRate(vortex_t * vortex, u32 *sr)
+{
+ *sr = (hwread(vortex->mmio, 0x24660) >> 3) & 0x1f;
+}
+
+#endif
+static void vortex_XtalkHw_Enable(vortex_t * vortex)
+{
+ u32 temp;
+
+ temp = (hwread(vortex->mmio, 0x24660) & 0x1FFFFFFF) | 0xC0000000;
+ temp |= 1;
+ hwwrite(vortex->mmio, 0x24660, temp);
+
+}
+
+static void vortex_XtalkHw_Disable(vortex_t * vortex)
+{
+ u32 temp;
+
+ temp = (hwread(vortex->mmio, 0x24660) & 0x1FFFFFFF) | 0xC0000000;
+ temp &= 0xfffffffe;
+ hwwrite(vortex->mmio, 0x24660, temp);
+
+}
+
+static void vortex_XtalkHw_ZeroIO(vortex_t * vortex)
+{
+ int i;
+
+ for (i = 0; i < 20; i++)
+ hwwrite(vortex->mmio, 0x24600 + (i << 2), 0);
+ for (i = 0; i < 4; i++)
+ hwwrite(vortex->mmio, 0x24650 + (i << 2), 0);
+}
+
+static void vortex_XtalkHw_ZeroState(vortex_t * vortex)
+{
+ vortex_XtalkHw_ZeroIO(vortex); // inlined
+
+ vortex_XtalkHw_SetLeftEQ(vortex, 0, 0, asXtalkCoefsZeros);
+ vortex_XtalkHw_SetRightEQ(vortex, 0, 0, asXtalkCoefsZeros);
+
+ vortex_XtalkHw_SetLeftXT(vortex, 0, 0, asXtalkCoefsZeros);
+ vortex_XtalkHw_SetRightXT(vortex, 0, 0, asXtalkCoefsZeros);
+
+ vortex_XtalkHw_SetGains(vortex, asXtalkGainsZeros); // inlined
+
+ vortex_XtalkHw_SetDelay(vortex, 0, 0); // inlined
+
+ vortex_XtalkHw_SetLeftDline(vortex, alXtalkDlineZeros); // inlined
+ vortex_XtalkHw_SetRightDline(vortex, alXtalkDlineZeros); // inlined
+ vortex_XtalkHw_SetLeftDline(vortex, alXtalkDlineZeros); // inlined
+ vortex_XtalkHw_SetRightDline(vortex, alXtalkDlineZeros); // inlined
+
+ vortex_XtalkHw_SetLeftEQStates(vortex, asXtalkInStateZeros,
+ asXtalkOutStateZeros);
+ vortex_XtalkHw_SetRightEQStates(vortex, asXtalkInStateZeros,
+ asXtalkOutStateZeros);
+ vortex_XtalkHw_SetLeftXTStates(vortex, asXtalkInStateZeros,
+ asXtalkOutStateZeros);
+ vortex_XtalkHw_SetRightXTStates(vortex, asXtalkInStateZeros,
+ asXtalkOutStateZeros);
+}
+
+static void vortex_XtalkHw_ProgramPipe(vortex_t * vortex)
+{
+
+ vortex_XtalkHw_SetLeftEQ(vortex, 0, 1, asXtalkCoefsPipe);
+ vortex_XtalkHw_SetRightEQ(vortex, 0, 1, asXtalkCoefsPipe);
+ vortex_XtalkHw_SetLeftXT(vortex, 0, 0, asXtalkCoefsZeros);
+ vortex_XtalkHw_SetRightXT(vortex, 0, 0, asXtalkCoefsZeros);
+
+ vortex_XtalkHw_SetDelay(vortex, 0, 0); // inlined
+}
+
+static void vortex_XtalkHw_ProgramXtalkWide(vortex_t * vortex)
+{
+
+ vortex_XtalkHw_SetLeftEQ(vortex, sXtalkWideKLeftEq,
+ sXtalkWideShiftLeftEq, asXtalkWideCoefsLeftEq);
+ vortex_XtalkHw_SetRightEQ(vortex, sXtalkWideKRightEq,
+ sXtalkWideShiftRightEq,
+ asXtalkWideCoefsRightEq);
+ vortex_XtalkHw_SetLeftXT(vortex, sXtalkWideKLeftXt,
+ sXtalkWideShiftLeftXt, asXtalkWideCoefsLeftXt);
+ vortex_XtalkHw_SetRightXT(vortex, sXtalkWideKLeftXt,
+ sXtalkWideShiftLeftXt,
+ asXtalkWideCoefsLeftXt);
+
+ vortex_XtalkHw_SetDelay(vortex, wXtalkWideRightDelay, wXtalkWideLeftDelay); // inlined
+}
+
+static void vortex_XtalkHw_ProgramXtalkNarrow(vortex_t * vortex)
+{
+
+ vortex_XtalkHw_SetLeftEQ(vortex, sXtalkNarrowKLeftEq,
+ sXtalkNarrowShiftLeftEq,
+ asXtalkNarrowCoefsLeftEq);
+ vortex_XtalkHw_SetRightEQ(vortex, sXtalkNarrowKRightEq,
+ sXtalkNarrowShiftRightEq,
+ asXtalkNarrowCoefsRightEq);
+ vortex_XtalkHw_SetLeftXT(vortex, sXtalkNarrowKLeftXt,
+ sXtalkNarrowShiftLeftXt,
+ asXtalkNarrowCoefsLeftXt);
+ vortex_XtalkHw_SetRightXT(vortex, sXtalkNarrowKLeftXt,
+ sXtalkNarrowShiftLeftXt,
+ asXtalkNarrowCoefsLeftXt);
+
+ vortex_XtalkHw_SetDelay(vortex, wXtalkNarrowRightDelay, wXtalkNarrowLeftDelay); // inlined
+}
+
+static void vortex_XtalkHw_ProgramDiamondXtalk(vortex_t * vortex)
+{
+
+ //sDiamondKLeftEq,sDiamondKRightXt,asDiamondCoefsLeftEq
+ vortex_XtalkHw_SetLeftEQ(vortex, sDiamondKLeftEq,
+ sDiamondShiftLeftEq, asDiamondCoefsLeftEq);
+ vortex_XtalkHw_SetRightEQ(vortex, sDiamondKRightEq,
+ sDiamondShiftRightEq, asDiamondCoefsRightEq);
+ vortex_XtalkHw_SetLeftXT(vortex, sDiamondKLeftXt,
+ sDiamondShiftLeftXt, asDiamondCoefsLeftXt);
+ vortex_XtalkHw_SetRightXT(vortex, sDiamondKLeftXt,
+ sDiamondShiftLeftXt, asDiamondCoefsLeftXt);
+
+ vortex_XtalkHw_SetDelay(vortex, wDiamondRightDelay, wDiamondLeftDelay); // inlined
+}
+
+static void vortex_XtalkHw_init(vortex_t * vortex)
+{
+ vortex_XtalkHw_ZeroState(vortex);
+}
+
+/* End of file */
diff --git a/sound/pci/au88x0/au88x0_xtalk.h b/sound/pci/au88x0/au88x0_xtalk.h
new file mode 100644
index 0000000..7f4534b
--- /dev/null
+++ b/sound/pci/au88x0/au88x0_xtalk.h
@@ -0,0 +1,61 @@
+/***************************************************************************
+ * au88x0_cxtalk.h
+ *
+ * Wed Nov 19 19:07:17 2003
+ * Copyright 2003 mjander
+ * mjander@users.sourceforge.org
+ ****************************************************************************/
+
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+/* The crosstalk canceler supports 5 stereo input channels. The result is
+ available at one single output route pair (stereo). */
+
+#ifndef _AU88X0_CXTALK_H
+#define _AU88X0_CXTALK_H
+
+#include "au88x0.h"
+
+#define XTDLINE_SZ 32
+#define XTGAINS_SZ 10
+#define XTINST_SZ 4
+
+#define XT_HEADPHONE 1
+#define XT_SPEAKER0 2
+#define XT_SPEAKER1 3
+#define XT_DIAMOND 4
+
+typedef u32 xtalk_dline_t[XTDLINE_SZ];
+typedef u16 xtalk_gains_t[XTGAINS_SZ];
+typedef u16 xtalk_instate_t[XTINST_SZ];
+typedef u16 xtalk_coefs_t[5][5];
+typedef u16 xtalk_state_t[5][4];
+
+static void vortex_XtalkHw_SetGains(vortex_t * vortex,
+ xtalk_gains_t const gains);
+static void vortex_XtalkHw_SetGainsAllChan(vortex_t * vortex);
+static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, u32 sr);
+static void vortex_XtalkHw_ProgramPipe(vortex_t * vortex);
+static void vortex_XtalkHw_ProgramPipe(vortex_t * vortex);
+static void vortex_XtalkHw_ProgramXtalkWide(vortex_t * vortex);
+static void vortex_XtalkHw_ProgramXtalkNarrow(vortex_t * vortex);
+static void vortex_XtalkHw_ProgramDiamondXtalk(vortex_t * vortex);
+static void vortex_XtalkHw_Enable(vortex_t * vortex);
+static void vortex_XtalkHw_Disable(vortex_t * vortex);
+static void vortex_XtalkHw_init(vortex_t * vortex);
+
+#endif /* _AU88X0_CXTALK_H */
diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile
new file mode 100644
index 0000000..842335d
--- /dev/null
+++ b/sound/pci/aw2/Makefile
@@ -0,0 +1,3 @@
+snd-aw2-objs := aw2-alsa.o aw2-saa7146.o
+
+obj-$(CONFIG_SND_AW2) += snd-aw2.o
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
new file mode 100644
index 0000000..c7c54e7
--- /dev/null
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -0,0 +1,794 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/control.h>
+
+#include "saa7146.h"
+#include "aw2-saa7146.h"
+
+MODULE_AUTHOR("Cedric Bregardis <cedric.bregardis@free.fr>, "
+ "Jean-Christian Hassler <jhassler@free.fr>");
+MODULE_DESCRIPTION("Emagic Audiowerk 2 sound driver");
+MODULE_LICENSE("GPL");
+
+/*********************************
+ * DEFINES
+ ********************************/
+#define PCI_VENDOR_ID_SAA7146 0x1131
+#define PCI_DEVICE_ID_SAA7146 0x7146
+
+#define CTL_ROUTE_ANALOG 0
+#define CTL_ROUTE_DIGITAL 1
+
+/*********************************
+ * TYPEDEFS
+ ********************************/
+ /* hardware definition */
+static struct snd_pcm_hardware snd_aw2_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .channels_min = 2,
+ .channels_max = 4,
+ .buffer_bytes_max = 32768,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 32768,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static struct snd_pcm_hardware snd_aw2_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 32768,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 32768,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+struct aw2_pcm_device {
+ struct snd_pcm *pcm;
+ unsigned int stream_number;
+ struct aw2 *chip;
+};
+
+struct aw2 {
+ struct snd_aw2_saa7146 saa7146;
+
+ struct pci_dev *pci;
+ int irq;
+ spinlock_t reg_lock;
+ struct mutex mtx;
+
+ unsigned long iobase_phys;
+ void __iomem *iobase_virt;
+
+ struct snd_card *card;
+
+ struct aw2_pcm_device device_playback[NB_STREAM_PLAYBACK];
+ struct aw2_pcm_device device_capture[NB_STREAM_CAPTURE];
+};
+
+/*********************************
+ * FUNCTION DECLARATIONS
+ ********************************/
+static int __init alsa_card_aw2_init(void);
+static void __exit alsa_card_aw2_exit(void);
+static int snd_aw2_dev_free(struct snd_device *device);
+static int __devinit snd_aw2_create(struct snd_card *card,
+ struct pci_dev *pci, struct aw2 **rchip);
+static int __devinit snd_aw2_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id);
+static void __devexit snd_aw2_remove(struct pci_dev *pci);
+static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params);
+static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream);
+static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream,
+ int cmd);
+static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd);
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream
+ *substream);
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream
+ *substream);
+static int __devinit snd_aw2_new_pcm(struct aw2 *chip);
+
+static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol);
+static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol);
+
+/*********************************
+ * VARIABLES
+ ********************************/
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Audiowerk2 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
+
+static struct pci_device_id snd_aw2_ids[] = {
+ {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
+ 0, 0, 0},
+ {0}
+};
+
+MODULE_DEVICE_TABLE(pci, snd_aw2_ids);
+
+/* pci_driver definition */
+static struct pci_driver driver = {
+ .name = "Emagic Audiowerk 2",
+ .id_table = snd_aw2_ids,
+ .probe = snd_aw2_probe,
+ .remove = __devexit_p(snd_aw2_remove),
+};
+
+/* operators for playback PCM alsa interface */
+static struct snd_pcm_ops snd_aw2_playback_ops = {
+ .open = snd_aw2_pcm_playback_open,
+ .close = snd_aw2_pcm_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_aw2_pcm_hw_params,
+ .hw_free = snd_aw2_pcm_hw_free,
+ .prepare = snd_aw2_pcm_prepare_playback,
+ .trigger = snd_aw2_pcm_trigger_playback,
+ .pointer = snd_aw2_pcm_pointer_playback,
+};
+
+/* operators for capture PCM alsa interface */
+static struct snd_pcm_ops snd_aw2_capture_ops = {
+ .open = snd_aw2_pcm_capture_open,
+ .close = snd_aw2_pcm_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_aw2_pcm_hw_params,
+ .hw_free = snd_aw2_pcm_hw_free,
+ .prepare = snd_aw2_pcm_prepare_capture,
+ .trigger = snd_aw2_pcm_trigger_capture,
+ .pointer = snd_aw2_pcm_pointer_capture,
+};
+
+static struct snd_kcontrol_new aw2_control __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Capture Route",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0xffff,
+ .info = snd_aw2_control_switch_capture_info,
+ .get = snd_aw2_control_switch_capture_get,
+ .put = snd_aw2_control_switch_capture_put
+};
+
+/*********************************
+ * FUNCTION IMPLEMENTATIONS
+ ********************************/
+
+/* initialization of the module */
+static int __init alsa_card_aw2_init(void)
+{
+ snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n");
+ return pci_register_driver(&driver);
+}
+
+/* clean up the module */
+static void __exit alsa_card_aw2_exit(void)
+{
+ snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n");
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_aw2_init);
+module_exit(alsa_card_aw2_exit);
+
+/* component-destructor */
+static int snd_aw2_dev_free(struct snd_device *device)
+{
+ struct aw2 *chip = device->device_data;
+
+ /* Free hardware */
+ snd_aw2_saa7146_free(&chip->saa7146);
+
+ /* release the irq */
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *)chip);
+ /* release the i/o ports & memory */
+ if (chip->iobase_virt)
+ iounmap(chip->iobase_virt);
+
+ pci_release_regions(chip->pci);
+ /* disable the PCI entry */
+ pci_disable_device(chip->pci);
+ /* release the data */
+ kfree(chip);
+
+ return 0;
+}
+
+/* chip-specific constructor */
+static int __devinit snd_aw2_create(struct snd_card *card,
+ struct pci_dev *pci, struct aw2 **rchip)
+{
+ struct aw2 *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_aw2_dev_free,
+ };
+
+ *rchip = NULL;
+
+ /* initialize the PCI entry */
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+ pci_set_master(pci);
+
+ /* check PCI availability (32bit DMA) */
+ if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) ||
+ (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) {
+ printk(KERN_ERR "aw2: Impossible to set 32bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ /* initialize the stuff */
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* (1) PCI resource allocation */
+ err = pci_request_regions(pci, "Audiowerk2");
+ if (err < 0) {
+ pci_disable_device(pci);
+ kfree(chip);
+ return err;
+ }
+ chip->iobase_phys = pci_resource_start(pci, 0);
+ chip->iobase_virt =
+ ioremap_nocache(chip->iobase_phys,
+ pci_resource_len(pci, 0));
+
+ if (chip->iobase_virt == NULL) {
+ printk(KERN_ERR "aw2: unable to remap memory region");
+ pci_release_regions(pci);
+ pci_disable_device(pci);
+ kfree(chip);
+ return -ENOMEM;
+ }
+
+ /* (2) initialization of the chip hardware */
+ snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
+
+ if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
+ IRQF_SHARED, "Audiowerk2", chip)) {
+ printk(KERN_ERR "aw2: Cannot grab irq %d\n", pci->irq);
+
+ iounmap(chip->iobase_virt);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ free_irq(chip->irq, (void *)chip);
+ iounmap(chip->iobase_virt);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+ *rchip = chip;
+
+ printk(KERN_INFO
+ "Audiowerk 2 sound card (saa7146 chipset) detected and "
+ "managed\n");
+ return 0;
+}
+
+/* constructor */
+static int __devinit snd_aw2_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct aw2 *chip;
+ int err;
+
+ /* (1) Continue if device is not enabled, else inc dev */
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ /* (2) Create card instance */
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ /* (3) Create main component */
+ err = snd_aw2_create(card, pci, &chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* initialize mutex */
+ mutex_init(&chip->mtx);
+ /* init spinlock */
+ spin_lock_init(&chip->reg_lock);
+ /* (4) Define driver ID and name string */
+ strcpy(card->driver, "aw2");
+ strcpy(card->shortname, "Audiowerk2");
+
+ sprintf(card->longname, "%s with SAA7146 irq %i",
+ card->shortname, chip->irq);
+
+ /* (5) Create other components */
+ snd_aw2_new_pcm(chip);
+
+ /* (6) Register card instance */
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* (7) Set PCI driver data */
+ pci_set_drvdata(pci, card);
+
+ dev++;
+ return 0;
+}
+
+/* destructor */
+static void __devexit snd_aw2_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+/* open callback */
+static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_printdd(KERN_DEBUG "aw2: Playback_open \n");
+ runtime->hw = snd_aw2_playback_hw;
+ return 0;
+}
+
+/* close callback */
+static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+
+}
+
+static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_printdd(KERN_DEBUG "aw2: Capture_open \n");
+ runtime->hw = snd_aw2_capture_hw;
+ return 0;
+}
+
+/* close callback */
+static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ /* TODO: something to do ? */
+ return 0;
+}
+
+ /* hw_params callback */
+static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare callback for playback */
+static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long period_size, buffer_size;
+
+ mutex_lock(&chip->mtx);
+
+ period_size = snd_pcm_lib_period_bytes(substream);
+ buffer_size = snd_pcm_lib_buffer_bytes(substream);
+
+ snd_aw2_saa7146_pcm_init_playback(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_addr, period_size,
+ buffer_size);
+
+ /* Define Interrupt callback */
+ snd_aw2_saa7146_define_it_playback_callback(pcm_device->stream_number,
+ (snd_aw2_saa7146_it_cb)
+ snd_pcm_period_elapsed,
+ (void *)substream);
+
+ mutex_unlock(&chip->mtx);
+
+ return 0;
+}
+
+/* prepare callback for capture */
+static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long period_size, buffer_size;
+
+ mutex_lock(&chip->mtx);
+
+ period_size = snd_pcm_lib_period_bytes(substream);
+ buffer_size = snd_pcm_lib_buffer_bytes(substream);
+
+ snd_aw2_saa7146_pcm_init_capture(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_addr, period_size,
+ buffer_size);
+
+ /* Define Interrupt callback */
+ snd_aw2_saa7146_define_it_capture_callback(pcm_device->stream_number,
+ (snd_aw2_saa7146_it_cb)
+ snd_pcm_period_elapsed,
+ (void *)substream);
+
+ mutex_unlock(&chip->mtx);
+
+ return 0;
+}
+
+/* playback trigger callback */
+static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ int status = 0;
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_aw2_saa7146_pcm_trigger_start_playback(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_aw2_saa7146_pcm_trigger_stop_playback(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ default:
+ status = -EINVAL;
+ }
+ spin_unlock(&chip->reg_lock);
+ return status;
+}
+
+/* capture trigger callback */
+static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ int status = 0;
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_aw2_saa7146_pcm_trigger_start_capture(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_aw2_saa7146_pcm_trigger_stop_capture(&chip->saa7146,
+ pcm_device->
+ stream_number);
+ break;
+ default:
+ status = -EINVAL;
+ }
+ spin_unlock(&chip->reg_lock);
+ return status;
+}
+
+/* playback pointer callback */
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream
+ *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ unsigned int current_ptr;
+
+ /* get the current hardware pointer */
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ current_ptr =
+ snd_aw2_saa7146_get_hw_ptr_playback(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_area,
+ runtime->buffer_size);
+
+ return bytes_to_frames(substream->runtime, current_ptr);
+}
+
+/* capture pointer callback */
+static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream
+ *substream)
+{
+ struct aw2_pcm_device *pcm_device = snd_pcm_substream_chip(substream);
+ struct aw2 *chip = pcm_device->chip;
+ unsigned int current_ptr;
+
+ /* get the current hardware pointer */
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ current_ptr =
+ snd_aw2_saa7146_get_hw_ptr_capture(&chip->saa7146,
+ pcm_device->stream_number,
+ runtime->dma_area,
+ runtime->buffer_size);
+
+ return bytes_to_frames(substream->runtime, current_ptr);
+}
+
+/* create a pcm device */
+static int __devinit snd_aw2_new_pcm(struct aw2 *chip)
+{
+ struct snd_pcm *pcm_playback_ana;
+ struct snd_pcm *pcm_playback_num;
+ struct snd_pcm *pcm_capture;
+ struct aw2_pcm_device *pcm_device;
+ int err = 0;
+
+ /* Create new Alsa PCM device */
+
+ err = snd_pcm_new(chip->card, "Audiowerk2 analog playback", 0, 1, 0,
+ &pcm_playback_ana);
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err);
+ return err;
+ }
+
+ /* Creation ok */
+ pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_ANA];
+
+ /* Set PCM device name */
+ strcpy(pcm_playback_ana->name, "Analog playback");
+ /* Associate private data to PCM device */
+ pcm_playback_ana->private_data = pcm_device;
+ /* set operators of PCM device */
+ snd_pcm_set_ops(pcm_playback_ana, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_aw2_playback_ops);
+ /* store PCM device */
+ pcm_device->pcm = pcm_playback_ana;
+ /* give base chip pointer to our internal pcm device
+ structure */
+ pcm_device->chip = chip;
+ /* Give stream number to PCM device */
+ pcm_device->stream_number = NUM_STREAM_PLAYBACK_ANA;
+
+ /* pre-allocation of buffers */
+ /* Preallocate continuous pages. */
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_ana,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data
+ (chip->pci),
+ 64 * 1024, 64 * 1024);
+ if (err)
+ printk(KERN_ERR "aw2: snd_pcm_lib_preallocate_pages_for_all "
+ "error (0x%X)\n", err);
+
+ err = snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0,
+ &pcm_playback_num);
+
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err);
+ return err;
+ }
+ /* Creation ok */
+ pcm_device = &chip->device_playback[NUM_STREAM_PLAYBACK_DIG];
+
+ /* Set PCM device name */
+ strcpy(pcm_playback_num->name, "Digital playback");
+ /* Associate private data to PCM device */
+ pcm_playback_num->private_data = pcm_device;
+ /* set operators of PCM device */
+ snd_pcm_set_ops(pcm_playback_num, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_aw2_playback_ops);
+ /* store PCM device */
+ pcm_device->pcm = pcm_playback_num;
+ /* give base chip pointer to our internal pcm device
+ structure */
+ pcm_device->chip = chip;
+ /* Give stream number to PCM device */
+ pcm_device->stream_number = NUM_STREAM_PLAYBACK_DIG;
+
+ /* pre-allocation of buffers */
+ /* Preallocate continuous pages. */
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm_playback_num,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data
+ (chip->pci),
+ 64 * 1024, 64 * 1024);
+ if (err)
+ printk(KERN_ERR
+ "aw2: snd_pcm_lib_preallocate_pages_for_all error "
+ "(0x%X)\n", err);
+
+
+
+ err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1,
+ &pcm_capture);
+
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_pcm_new error (0x%X)\n", err);
+ return err;
+ }
+
+ /* Creation ok */
+ pcm_device = &chip->device_capture[NUM_STREAM_CAPTURE_ANA];
+
+ /* Set PCM device name */
+ strcpy(pcm_capture->name, "Capture");
+ /* Associate private data to PCM device */
+ pcm_capture->private_data = pcm_device;
+ /* set operators of PCM device */
+ snd_pcm_set_ops(pcm_capture, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_aw2_capture_ops);
+ /* store PCM device */
+ pcm_device->pcm = pcm_capture;
+ /* give base chip pointer to our internal pcm device
+ structure */
+ pcm_device->chip = chip;
+ /* Give stream number to PCM device */
+ pcm_device->stream_number = NUM_STREAM_CAPTURE_ANA;
+
+ /* pre-allocation of buffers */
+ /* Preallocate continuous pages. */
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm_capture,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data
+ (chip->pci),
+ 64 * 1024, 64 * 1024);
+ if (err)
+ printk(KERN_ERR
+ "aw2: snd_pcm_lib_preallocate_pages_for_all error "
+ "(0x%X)\n", err);
+
+
+ /* Create control */
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&aw2_control, chip));
+ if (err < 0) {
+ printk(KERN_ERR "aw2: snd_ctl_add error (0x%X)\n", err);
+ return err;
+ }
+
+ return 0;
+}
+
+static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = {
+ "Analog", "Digital"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) {
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ }
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol)
+{
+ struct aw2 *chip = snd_kcontrol_chip(kcontrol);
+ if (snd_aw2_saa7146_is_using_digital_input(&chip->saa7146))
+ ucontrol->value.enumerated.item[0] = CTL_ROUTE_DIGITAL;
+ else
+ ucontrol->value.enumerated.item[0] = CTL_ROUTE_ANALOG;
+ return 0;
+}
+
+static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value
+ *ucontrol)
+{
+ struct aw2 *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int is_disgital =
+ snd_aw2_saa7146_is_using_digital_input(&chip->saa7146);
+
+ if (((ucontrol->value.integer.value[0] == CTL_ROUTE_DIGITAL)
+ && !is_disgital)
+ || ((ucontrol->value.integer.value[0] == CTL_ROUTE_ANALOG)
+ && is_disgital)) {
+ snd_aw2_saa7146_use_digital_input(&chip->saa7146, !is_disgital);
+ changed = 1;
+ }
+ return changed;
+}
diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c
new file mode 100644
index 0000000..6a3891a
--- /dev/null
+++ b/sound/pci/aw2/aw2-saa7146.c
@@ -0,0 +1,465 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+#define AW2_SAA7146_M
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <asm/system.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "saa7146.h"
+#include "aw2-saa7146.h"
+
+#include "aw2-tsl.c"
+
+#define WRITEREG(value, addr) writel((value), chip->base_addr + (addr))
+#define READREG(addr) readl(chip->base_addr + (addr))
+
+static struct snd_aw2_saa7146_cb_param
+ arr_substream_it_playback_cb[NB_STREAM_PLAYBACK];
+static struct snd_aw2_saa7146_cb_param
+ arr_substream_it_capture_cb[NB_STREAM_CAPTURE];
+
+static int snd_aw2_saa7146_get_limit(int size);
+
+/* chip-specific destructor */
+int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip)
+{
+ /* disable all irqs */
+ WRITEREG(0, IER);
+
+ /* reset saa7146 */
+ WRITEREG((MRST_N << 16), MC1);
+
+ /* Unset base addr */
+ chip->base_addr = NULL;
+
+ return 0;
+}
+
+void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip,
+ void __iomem *pci_base_addr)
+{
+ /* set PCI burst/threshold
+
+ Burst length definition
+ VALUE BURST LENGTH
+ 000 1 Dword
+ 001 2 Dwords
+ 010 4 Dwords
+ 011 8 Dwords
+ 100 16 Dwords
+ 101 32 Dwords
+ 110 64 Dwords
+ 111 128 Dwords
+
+ Threshold definition
+ VALUE WRITE MODE READ MODE
+ 00 1 Dword of valid data 1 empty Dword
+ 01 4 Dwords of valid data 4 empty Dwords
+ 10 8 Dwords of valid data 8 empty Dwords
+ 11 16 Dwords of valid data 16 empty Dwords */
+
+ unsigned int acon2;
+ unsigned int acon1 = 0;
+ int i;
+
+ /* Set base addr */
+ chip->base_addr = pci_base_addr;
+
+ /* disable all irqs */
+ WRITEREG(0, IER);
+
+ /* reset saa7146 */
+ WRITEREG((MRST_N << 16), MC1);
+
+ /* enable audio interface */
+#ifdef __BIG_ENDIAN
+ acon1 |= A1_SWAP;
+ acon1 |= A2_SWAP;
+#endif
+ /* WS0_CTRL, WS0_SYNC: input TSL1, I2S */
+
+ /* At initialization WS1 and WS2 are disbaled (configured as input */
+ acon1 |= 0 * WS1_CTRL;
+ acon1 |= 0 * WS2_CTRL;
+
+ /* WS4 is not used. So it must not restart A2.
+ This is why it is configured as output (force to low) */
+ acon1 |= 3 * WS4_CTRL;
+
+ /* WS3_CTRL, WS3_SYNC: output TSL2, I2S */
+ acon1 |= 2 * WS3_CTRL;
+
+ /* A1 and A2 are active and asynchronous */
+ acon1 |= 3 * AUDIO_MODE;
+ WRITEREG(acon1, ACON1);
+
+ /* The following comes from original windows driver.
+ It is needed to have a correct behavior of input and output
+ simultenously, but I don't know why ! */
+ WRITEREG(3 * (BurstA1_in) + 3 * (ThreshA1_in) +
+ 3 * (BurstA1_out) + 3 * (ThreshA1_out) +
+ 3 * (BurstA2_out) + 3 * (ThreshA2_out), PCI_BT_A);
+
+ /* enable audio port pins */
+ WRITEREG((EAP << 16) | EAP, MC1);
+
+ /* enable I2C */
+ WRITEREG((EI2C << 16) | EI2C, MC1);
+ /* enable interrupts */
+ WRITEREG(A1_out | A2_out | A1_in | IIC_S | IIC_E, IER);
+
+ /* audio configuration */
+ acon2 = A2_CLKSRC | BCLK1_OEN;
+ WRITEREG(acon2, ACON2);
+
+ /* By default use analog input */
+ snd_aw2_saa7146_use_digital_input(chip, 0);
+
+ /* TSL setup */
+ for (i = 0; i < 8; ++i) {
+ WRITEREG(tsl1[i], TSL1 + (i * 4));
+ WRITEREG(tsl2[i], TSL2 + (i * 4));
+ }
+
+}
+
+void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size)
+{
+ unsigned long dw_page, dw_limit;
+
+ /* Configure DMA for substream
+ Configuration informations: ALSA has allocated continuous memory
+ pages. So we don't need to use MMU of saa7146.
+ */
+
+ /* No MMU -> nothing to do with PageA1, we only configure the limit of
+ PageAx_out register */
+ /* Disable MMU */
+ dw_page = (0L << 11);
+
+ /* Configure Limit for DMA access.
+ The limit register defines an address limit, which generates
+ an interrupt if passed by the actual PCI address pointer.
+ '0001' means an interrupt will be generated if the lower
+ 6 bits (64 bytes) of the PCI address are zero. '0010'
+ defines a limit of 128 bytes, '0011' one of 256 bytes, and
+ so on up to 1 Mbyte defined by '1111'. This interrupt range
+ can be calculated as follows:
+ Range = 2^(5 + Limit) bytes.
+ */
+ dw_limit = snd_aw2_saa7146_get_limit(period_size);
+ dw_page |= (dw_limit << 4);
+
+ if (stream_number == 0) {
+ WRITEREG(dw_page, PageA2_out);
+
+ /* Base address for DMA transfert. */
+ /* This address has been reserved by ALSA. */
+ /* This is a physical address */
+ WRITEREG(dma_addr, BaseA2_out);
+
+ /* Define upper limit for DMA access */
+ WRITEREG(dma_addr + buffer_size, ProtA2_out);
+
+ } else if (stream_number == 1) {
+ WRITEREG(dw_page, PageA1_out);
+
+ /* Base address for DMA transfert. */
+ /* This address has been reserved by ALSA. */
+ /* This is a physical address */
+ WRITEREG(dma_addr, BaseA1_out);
+
+ /* Define upper limit for DMA access */
+ WRITEREG(dma_addr + buffer_size, ProtA1_out);
+ } else {
+ printk(KERN_ERR
+ "aw2: snd_aw2_saa7146_pcm_init_playback: "
+ "Substream number is not 0 or 1 -> not managed\n");
+ }
+}
+
+void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number, unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size)
+{
+ unsigned long dw_page, dw_limit;
+
+ /* Configure DMA for substream
+ Configuration informations: ALSA has allocated continuous memory
+ pages. So we don't need to use MMU of saa7146.
+ */
+
+ /* No MMU -> nothing to do with PageA1, we only configure the limit of
+ PageAx_out register */
+ /* Disable MMU */
+ dw_page = (0L << 11);
+
+ /* Configure Limit for DMA access.
+ The limit register defines an address limit, which generates
+ an interrupt if passed by the actual PCI address pointer.
+ '0001' means an interrupt will be generated if the lower
+ 6 bits (64 bytes) of the PCI address are zero. '0010'
+ defines a limit of 128 bytes, '0011' one of 256 bytes, and
+ so on up to 1 Mbyte defined by '1111'. This interrupt range
+ can be calculated as follows:
+ Range = 2^(5 + Limit) bytes.
+ */
+ dw_limit = snd_aw2_saa7146_get_limit(period_size);
+ dw_page |= (dw_limit << 4);
+
+ if (stream_number == 0) {
+ WRITEREG(dw_page, PageA1_in);
+
+ /* Base address for DMA transfert. */
+ /* This address has been reserved by ALSA. */
+ /* This is a physical address */
+ WRITEREG(dma_addr, BaseA1_in);
+
+ /* Define upper limit for DMA access */
+ WRITEREG(dma_addr + buffer_size, ProtA1_in);
+ } else {
+ printk(KERN_ERR
+ "aw2: snd_aw2_saa7146_pcm_init_capture: "
+ "Substream number is not 0 -> not managed\n");
+ }
+}
+
+void snd_aw2_saa7146_define_it_playback_callback(unsigned int stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param)
+{
+ if (stream_number < NB_STREAM_PLAYBACK) {
+ arr_substream_it_playback_cb[stream_number].p_it_callback =
+ (snd_aw2_saa7146_it_cb) p_it_callback;
+ arr_substream_it_playback_cb[stream_number].p_callback_param =
+ (void *)p_callback_param;
+ }
+}
+
+void snd_aw2_saa7146_define_it_capture_callback(unsigned int stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param)
+{
+ if (stream_number < NB_STREAM_CAPTURE) {
+ arr_substream_it_capture_cb[stream_number].p_it_callback =
+ (snd_aw2_saa7146_it_cb) p_it_callback;
+ arr_substream_it_capture_cb[stream_number].p_callback_param =
+ (void *)p_callback_param;
+ }
+}
+
+void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ unsigned int acon1 = 0;
+ /* In aw8 driver, dma transfert is always active. It is
+ started and stopped in a larger "space" */
+ acon1 = READREG(ACON1);
+ if (stream_number == 0) {
+ WRITEREG((TR_E_A2_OUT << 16) | TR_E_A2_OUT, MC1);
+
+ /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */
+ acon1 |= 2 * WS2_CTRL;
+ WRITEREG(acon1, ACON1);
+
+ } else if (stream_number == 1) {
+ WRITEREG((TR_E_A1_OUT << 16) | TR_E_A1_OUT, MC1);
+
+ /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */
+ acon1 |= 1 * WS1_CTRL;
+ WRITEREG(acon1, ACON1);
+ }
+}
+
+void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ unsigned int acon1 = 0;
+ acon1 = READREG(ACON1);
+ if (stream_number == 0) {
+ /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */
+ acon1 &= ~(3 * WS2_CTRL);
+ WRITEREG(acon1, ACON1);
+
+ WRITEREG((TR_E_A2_OUT << 16), MC1);
+ } else if (stream_number == 1) {
+ /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */
+ acon1 &= ~(3 * WS1_CTRL);
+ WRITEREG(acon1, ACON1);
+
+ WRITEREG((TR_E_A1_OUT << 16), MC1);
+ }
+}
+
+void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ /* In aw8 driver, dma transfert is always active. It is
+ started and stopped in a larger "space" */
+ if (stream_number == 0)
+ WRITEREG((TR_E_A1_IN << 16) | TR_E_A1_IN, MC1);
+}
+
+void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number)
+{
+ if (stream_number == 0)
+ WRITEREG((TR_E_A1_IN << 16), MC1);
+}
+
+irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id)
+{
+ unsigned int isr;
+ unsigned int iicsta;
+ struct snd_aw2_saa7146 *chip = dev_id;
+
+ isr = READREG(ISR);
+ if (!isr)
+ return IRQ_NONE;
+
+ WRITEREG(isr, ISR);
+
+ if (isr & (IIC_S | IIC_E)) {
+ iicsta = READREG(IICSTA);
+ WRITEREG(0x100, IICSTA);
+ }
+
+ if (isr & A1_out) {
+ if (arr_substream_it_playback_cb[1].p_it_callback != NULL) {
+ arr_substream_it_playback_cb[1].
+ p_it_callback(arr_substream_it_playback_cb[1].
+ p_callback_param);
+ }
+ }
+ if (isr & A2_out) {
+ if (arr_substream_it_playback_cb[0].p_it_callback != NULL) {
+ arr_substream_it_playback_cb[0].
+ p_it_callback(arr_substream_it_playback_cb[0].
+ p_callback_param);
+ }
+
+ }
+ if (isr & A1_in) {
+ if (arr_substream_it_capture_cb[0].p_it_callback != NULL) {
+ arr_substream_it_capture_cb[0].
+ p_it_callback(arr_substream_it_capture_cb[0].
+ p_callback_param);
+ }
+ }
+ return IRQ_HANDLED;
+}
+
+unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned char *start_addr,
+ unsigned int buffer_size)
+{
+ long pci_adp = 0;
+ size_t ptr = 0;
+
+ if (stream_number == 0) {
+ pci_adp = READREG(PCI_ADP3);
+ ptr = pci_adp - (long)start_addr;
+
+ if (ptr == buffer_size)
+ ptr = 0;
+ }
+ if (stream_number == 1) {
+ pci_adp = READREG(PCI_ADP1);
+ ptr = pci_adp - (size_t) start_addr;
+
+ if (ptr == buffer_size)
+ ptr = 0;
+ }
+ return ptr;
+}
+
+unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned char *start_addr,
+ unsigned int buffer_size)
+{
+ size_t pci_adp = 0;
+ size_t ptr = 0;
+ if (stream_number == 0) {
+ pci_adp = READREG(PCI_ADP2);
+ ptr = pci_adp - (size_t) start_addr;
+
+ if (ptr == buffer_size)
+ ptr = 0;
+ }
+ return ptr;
+}
+
+void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip,
+ int use_digital)
+{
+ /* FIXME: switch between analog and digital input does not always work.
+ It can produce a kind of white noise. It seams that received data
+ are inverted sometime (endian inversion). Why ? I don't know, maybe
+ a problem of synchronization... However for the time being I have
+ not found the problem. Workaround: switch again (and again) between
+ digital and analog input until it works. */
+ if (use_digital)
+ WRITEREG(0x40, GPIO_CTRL);
+ else
+ WRITEREG(0x50, GPIO_CTRL);
+}
+
+int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146 *chip)
+{
+ unsigned int reg_val = READREG(GPIO_CTRL);
+ if ((reg_val & 0xFF) == 0x40)
+ return 1;
+ else
+ return 0;
+}
+
+
+static int snd_aw2_saa7146_get_limit(int size)
+{
+ int limitsize = 32;
+ int limit = 0;
+ while (limitsize < size) {
+ limitsize *= 2;
+ limit++;
+ }
+ return limit;
+}
diff --git a/sound/pci/aw2/aw2-saa7146.h b/sound/pci/aw2/aw2-saa7146.h
new file mode 100644
index 0000000..5b35e35
--- /dev/null
+++ b/sound/pci/aw2/aw2-saa7146.h
@@ -0,0 +1,105 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+#ifndef AW2_SAA7146_H
+#define AW2_SAA7146_H
+
+#define NB_STREAM_PLAYBACK 2
+#define NB_STREAM_CAPTURE 1
+
+#define NUM_STREAM_PLAYBACK_ANA 0
+#define NUM_STREAM_PLAYBACK_DIG 1
+
+#define NUM_STREAM_CAPTURE_ANA 0
+
+typedef void (*snd_aw2_saa7146_it_cb) (void *);
+
+struct snd_aw2_saa7146_cb_param {
+ snd_aw2_saa7146_it_cb p_it_callback;
+ void *p_callback_param;
+};
+
+/* definition of the chip-specific record */
+
+struct snd_aw2_saa7146 {
+ void __iomem *base_addr;
+};
+
+extern void snd_aw2_saa7146_setup(struct snd_aw2_saa7146 *chip,
+ void __iomem *pci_base_addr);
+extern int snd_aw2_saa7146_free(struct snd_aw2_saa7146 *chip);
+
+extern void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size);
+extern void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146 *chip,
+ int stream_number,
+ unsigned long dma_addr,
+ unsigned long period_size,
+ unsigned long buffer_size);
+extern void snd_aw2_saa7146_define_it_playback_callback(unsigned int
+ stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param);
+extern void snd_aw2_saa7146_define_it_capture_callback(unsigned int
+ stream_number,
+ snd_aw2_saa7146_it_cb
+ p_it_callback,
+ void *p_callback_param);
+extern void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146
+ *chip, int stream_number);
+extern void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146
+ *chip, int stream_number);
+
+extern void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146
+ *chip,
+ int stream_number);
+extern void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146
+ *chip, int stream_number);
+
+extern irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id);
+extern unsigned int snd_aw2_saa7146_get_hw_ptr_playback(struct snd_aw2_saa7146
+ *chip,
+ int stream_number,
+ unsigned char
+ *start_addr,
+ unsigned int
+ buffer_size);
+extern unsigned int snd_aw2_saa7146_get_hw_ptr_capture(struct snd_aw2_saa7146
+ *chip,
+ int stream_number,
+ unsigned char
+ *start_addr,
+ unsigned int
+ buffer_size);
+
+extern void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146 *chip,
+ int use_digital);
+
+extern int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146
+ *chip);
+
+#endif
diff --git a/sound/pci/aw2/aw2-tsl.c b/sound/pci/aw2/aw2-tsl.c
new file mode 100644
index 0000000..459b031
--- /dev/null
+++ b/sound/pci/aw2/aw2-tsl.c
@@ -0,0 +1,110 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ * Copyright 1998 Emagic Soft- und Hardware GmbH
+ * Copyright 2002 Martijn Sipkema
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+#define TSL_WS0 (1UL << 31)
+#define TSL_WS1 (1UL << 30)
+#define TSL_WS2 (1UL << 29)
+#define TSL_WS3 (1UL << 28)
+#define TSL_WS4 (1UL << 27)
+#define TSL_DIS_A1 (1UL << 24)
+#define TSL_SDW_A1 (1UL << 23)
+#define TSL_SIB_A1 (1UL << 22)
+#define TSL_SF_A1 (1UL << 21)
+#define TSL_LF_A1 (1UL << 20)
+#define TSL_BSEL_A1 (1UL << 17)
+#define TSL_DOD_A1 (1UL << 15)
+#define TSL_LOW_A1 (1UL << 14)
+#define TSL_DIS_A2 (1UL << 11)
+#define TSL_SDW_A2 (1UL << 10)
+#define TSL_SIB_A2 (1UL << 9)
+#define TSL_SF_A2 (1UL << 8)
+#define TSL_LF_A2 (1UL << 7)
+#define TSL_BSEL_A2 (1UL << 4)
+#define TSL_DOD_A2 (1UL << 2)
+#define TSL_LOW_A2 (1UL << 1)
+#define TSL_EOS (1UL << 0)
+
+ /* Audiowerk8 hardware setup: */
+ /* WS0, SD4, TSL1 - Analog/ digital in */
+ /* WS1, SD0, TSL1 - Analog out #1, digital out */
+ /* WS2, SD2, TSL1 - Analog out #2 */
+ /* WS3, SD1, TSL2 - Analog out #3 */
+ /* WS4, SD3, TSL2 - Analog out #4 */
+
+ /* Audiowerk8 timing: */
+ /* Timeslot: | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | ... */
+
+ /* A1_INPUT: */
+ /* SD4: <_ADC-L_>-------<_ADC-R_>-------< */
+ /* WS0: _______________/---------------\_ */
+
+ /* A1_OUTPUT: */
+ /* SD0: <_1-L___>-------<_1-R___>-------< */
+ /* WS1: _______________/---------------\_ */
+ /* SD2: >-------<_2-L___>-------<_2-R___> */
+ /* WS2: -------\_______________/--------- */
+
+ /* A2_OUTPUT: */
+ /* SD1: <_3-L___>-------<_3-R___>-------< */
+ /* WS3: _______________/---------------\_ */
+ /* SD3: >-------<_4-L___>-------<_4-R___> */
+ /* WS4: -------\_______________/--------- */
+
+static int tsl1[8] = {
+ 1 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_LF_A1,
+
+ 1 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1,
+
+ 0 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1,
+
+ 0 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1,
+
+ 1 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0,
+
+ 1 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0,
+
+ 0 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 |
+ 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0,
+
+ 0 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | 0 * TSL_DIS_A1 |
+ 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0 | TSL_SF_A1 | TSL_EOS,
+};
+
+static int tsl2[8] = {
+ 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_LF_A2,
+ 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2,
+ 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2,
+ 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2,
+ 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2,
+ 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2,
+ 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2,
+ 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2 | TSL_EOS
+};
diff --git a/sound/pci/aw2/saa7146.h b/sound/pci/aw2/saa7146.h
new file mode 100644
index 0000000..ce0ab5f
--- /dev/null
+++ b/sound/pci/aw2/saa7146.h
@@ -0,0 +1,168 @@
+/*****************************************************************************
+ *
+ * Copyright (C) 2008 Cedric Bregardis <cedric.bregardis@free.fr> and
+ * Jean-Christian Hassler <jhassler@free.fr>
+ *
+ * This file is part of the Audiowerk2 ALSA driver
+ *
+ * The Audiowerk2 ALSA driver is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2.
+ *
+ * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with the Audiowerk2 ALSA driver; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301,
+ * USA.
+ *
+ *****************************************************************************/
+
+/* SAA7146 registers */
+#define PCI_BT_A 0x4C
+#define IICTFR 0x8C
+#define IICSTA 0x90
+#define BaseA1_in 0x94
+#define ProtA1_in 0x98
+#define PageA1_in 0x9C
+#define BaseA1_out 0xA0
+#define ProtA1_out 0xA4
+#define PageA1_out 0xA8
+#define BaseA2_in 0xAC
+#define ProtA2_in 0xB0
+#define PageA2_in 0xB4
+#define BaseA2_out 0xB8
+#define ProtA2_out 0xBC
+#define PageA2_out 0xC0
+#define IER 0xDC
+#define GPIO_CTRL 0xE0
+#define ACON1 0xF4
+#define ACON2 0xF8
+#define MC1 0xFC
+#define MC2 0x100
+#define ISR 0x10C
+#define PSR 0x110
+#define SSR 0x114
+#define PCI_ADP1 0x12C
+#define PCI_ADP2 0x130
+#define PCI_ADP3 0x134
+#define PCI_ADP4 0x138
+#define LEVEL_REP 0x140
+#define FB_BUFFER1 0x144
+#define FB_BUFFER2 0x148
+#define TSL1 0x180
+#define TSL2 0x1C0
+
+#define ME (1UL << 11)
+#define LIMIT (1UL << 4)
+#define PV (1UL << 3)
+
+/* PSR/ISR/IER */
+#define PPEF (1UL << 31)
+#define PABO (1UL << 30)
+#define IIC_S (1UL << 17)
+#define IIC_E (1UL << 16)
+#define A2_in (1UL << 15)
+#define A2_out (1UL << 14)
+#define A1_in (1UL << 13)
+#define A1_out (1UL << 12)
+#define AFOU (1UL << 11)
+#define PIN3 (1UL << 6)
+#define PIN2 (1UL << 5)
+#define PIN1 (1UL << 4)
+#define PIN0 (1UL << 3)
+#define ECS (1UL << 2)
+#define EC3S (1UL << 1)
+#define EC0S (1UL << 0)
+
+/* SSR */
+#define PRQ (1UL << 31)
+#define PMA (1UL << 30)
+#define IIC_EA (1UL << 21)
+#define IIC_EW (1UL << 20)
+#define IIC_ER (1UL << 19)
+#define IIC_EL (1UL << 18)
+#define IIC_EF (1UL << 17)
+#define AF2_in (1UL << 10)
+#define AF2_out (1UL << 9)
+#define AF1_in (1UL << 8)
+#define AF1_out (1UL << 7)
+#define EC5S (1UL << 3)
+#define EC4S (1UL << 2)
+#define EC2S (1UL << 1)
+#define EC1S (1UL << 0)
+
+/* PCI_BT_A */
+#define BurstA1_in (1UL << 26)
+#define ThreshA1_in (1UL << 24)
+#define BurstA1_out (1UL << 18)
+#define ThreshA1_out (1UL << 16)
+#define BurstA2_in (1UL << 10)
+#define ThreshA2_in (1UL << 8)
+#define BurstA2_out (1UL << 2)
+#define ThreshA2_out (1UL << 0)
+
+/* MC1 */
+#define MRST_N (1UL << 15)
+#define EAP (1UL << 9)
+#define EI2C (1UL << 8)
+#define TR_E_A2_OUT (1UL << 3)
+#define TR_E_A2_IN (1UL << 2)
+#define TR_E_A1_OUT (1UL << 1)
+#define TR_E_A1_IN (1UL << 0)
+
+/* MC2 */
+#define UPLD_IIC (1UL << 0)
+
+/* ACON1 */
+#define AUDIO_MODE (1UL << 29)
+#define MAXLEVEL (1UL << 22)
+#define A1_SWAP (1UL << 21)
+#define A2_SWAP (1UL << 20)
+#define WS0_CTRL (1UL << 18)
+#define WS0_SYNC (1UL << 16)
+#define WS1_CTRL (1UL << 14)
+#define WS1_SYNC (1UL << 12)
+#define WS2_CTRL (1UL << 10)
+#define WS2_SYNC (1UL << 8)
+#define WS3_CTRL (1UL << 6)
+#define WS3_SYNC (1UL << 4)
+#define WS4_CTRL (1UL << 2)
+#define WS4_SYNC (1UL << 0)
+
+/* ACON2 */
+#define A1_CLKSRC (1UL << 27)
+#define A2_CLKSRC (1UL << 22)
+#define INVERT_BCLK1 (1UL << 21)
+#define INVERT_BCLK2 (1UL << 20)
+#define BCLK1_OEN (1UL << 19)
+#define BCLK2_OEN (1UL << 18)
+
+/* IICSTA */
+#define IICCC (1UL << 8)
+#define ABORT (1UL << 7)
+#define SPERR (1UL << 6)
+#define APERR (1UL << 5)
+#define DTERR (1UL << 4)
+#define DRERR (1UL << 3)
+#define AL (1UL << 2)
+#define ERR (1UL << 1)
+#define BUSY (1UL << 0)
+
+/* IICTFR */
+#define BYTE2 (1UL << 24)
+#define BYTE1 (1UL << 16)
+#define BYTE0 (1UL << 8)
+#define ATRR2 (1UL << 6)
+#define ATRR1 (1UL << 4)
+#define ATRR0 (1UL << 2)
+#define ERR (1UL << 1)
+#define BUSY (1UL << 0)
+
+#define START 3
+#define CONT 2
+#define STOP 1
+#define NOP 0
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
new file mode 100644
index 0000000..333007c
--- /dev/null
+++ b/sound/pci/azt3328.c
@@ -0,0 +1,2412 @@
+/*
+ * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168).
+ * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr <andi AT lisas.de>
+ *
+ * Framework borrowed from Bart Hartgers's als4000.c.
+ * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801),
+ * found in a Fujitsu-Siemens PC ("Cordant", aluminum case).
+ * Other versions are:
+ * PCI168 A(W), sub ID 1800
+ * PCI168 A/AP, sub ID 8000
+ * Please give me feedback in case you try my driver with one of these!!
+ *
+ * GPL LICENSE
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * NOTES
+ * Since Aztech does not provide any chipset documentation,
+ * even on repeated request to various addresses,
+ * and the answer that was finally given was negative
+ * (and I was stupid enough to manage to get hold of a PCI168 soundcard
+ * in the first place >:-P}),
+ * I was forced to base this driver on reverse engineering
+ * (3 weeks' worth of evenings filled with driver work).
+ * (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros)
+ *
+ * It is quite likely that the AZF3328 chip is the PCI cousin of the
+ * AZF3318 ("azt1020 pnp", "MM Pro 16") ISA chip, given very similar specs.
+ *
+ * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name
+ * for compatibility reasons) from Azfin (joint-venture of Aztech and Fincitec,
+ * Fincitec acquired by National Semiconductor in 2002, together with the
+ * Fincitec-related company ARSmikro) has the following features:
+ *
+ * - compatibility & compliance:
+ * - Microsoft PC 97 ("PC 97 Hardware Design Guide",
+ * http://www.microsoft.com/whdc/archive/pcguides.mspx)
+ * - Microsoft PC 98 Baseline Audio
+ * - MPU401 UART
+ * - Sound Blaster Emulation (DOS Box)
+ * - builtin AC97 conformant codec (SNR over 80dB)
+ * Note that "conformant" != "compliant"!! this chip's mixer register layout
+ * *differs* from the standard AC97 layout:
+ * they chose to not implement the headphone register (which is not a
+ * problem since it's merely optional), yet when doing this, they committed
+ * the grave sin of letting other registers follow immediately instead of
+ * keeping a headphone dummy register, thereby shifting the mixer register
+ * addresses illegally. So far unfortunately it looks like the very flexible
+ * ALSA AC97 support is still not enough to easily compensate for such a
+ * grave layout violation despite all tweaks and quirks mechanisms it offers.
+ * - builtin genuine OPL3 - verified to work fine, 20080506
+ * - full duplex 16bit playback/record at independent sampling rate
+ * - MPU401 (+ legacy address support, claimed by one official spec sheet)
+ * FIXME: how to enable legacy addr??
+ * - game port (legacy address support)
+ * - builtin DirectInput support, helps reduce CPU overhead (interrupt-driven
+ * features supported). - See common term "Digital Enhanced Game Port"...
+ * (probably DirectInput 3.0 spec - confirm)
+ * - builtin 3D enhancement (said to be YAMAHA Ymersion)
+ * - built-in General DirectX timer having a 20 bits counter
+ * with 1us resolution (see below!)
+ * - I2S serial output port for external DAC
+ * - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI
+ * - supports hardware volume control
+ * - single chip low cost solution (128 pin QFP)
+ * - supports programmable Sub-vendor and Sub-system ID
+ * required for Microsoft's logo compliance (FIXME: where?)
+ * At least the Trident 4D Wave DX has one bit somewhere
+ * to enable writes to PCI subsystem VID registers, that should be it.
+ * This might easily be in extended PCI reg space, since PCI168 also has
+ * some custom data starting at 0x80. What kind of config settings
+ * are located in our extended PCI space anyway??
+ * - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms
+ *
+ * Note that this driver now is actually *better* than the Windows driver,
+ * since it additionally supports the card's 1MHz DirectX timer - just try
+ * the following snd-seq module parameters etc.:
+ * - options snd-seq seq_default_timer_class=2 seq_default_timer_sclass=0
+ * seq_default_timer_card=0 seq_client_load=1 seq_default_timer_device=0
+ * seq_default_timer_subdevice=0 seq_default_timer_resolution=1000000
+ * - "timidity -iAv -B2,8 -Os -EFreverb=0"
+ * - "pmidi -p 128:0 jazz.mid"
+ *
+ * OPL3 hardware playback testing, try something like:
+ * cat /proc/asound/hwdep
+ * and
+ * aconnect -o
+ * Then use
+ * sbiload -Dhw:x,y --opl3 /usr/share/sounds/opl3/std.o3 ......./drums.o3
+ * where x,y is the xx-yy number as given in hwdep.
+ * Then try
+ * pmidi -p a:b jazz.mid
+ * where a:b is the client number plus 0 usually, as given by aconnect above.
+ * Oh, and make sure to unmute the FM mixer control (doh!)
+ * NOTE: power use during OPL3 playback is _VERY_ high (70W --> 90W!)
+ * despite no CPU activity, possibly due to hindering ACPI idling somehow.
+ * Shouldn't be a problem of the AZF3328 chip itself, I'd hope.
+ * Higher PCM / FM mixer levels seem to conflict (causes crackling),
+ * at least sometimes. Maybe even use with hardware sequencer timer above :)
+ * adplay/adplug-utils might soon offer hardware-based OPL3 playback, too.
+ *
+ * Certain PCI versions of this card are susceptible to DMA traffic underruns
+ * in some systems (resulting in sound crackling/clicking/popping),
+ * probably because they don't have a DMA FIFO buffer or so.
+ * Overview (PCI ID/PCI subID/PCI rev.):
+ * - no DMA crackling on SiS735: 0x50DC/0x1801/16
+ * - unknown performance: 0x50DC/0x1801/10
+ * (well, it's not bad on an Athlon 1800 with now very optimized IRQ handler)
+ *
+ * Crackling happens with VIA chipsets or, in my case, an SiS735, which is
+ * supposed to be very fast and supposed to get rid of crackling much
+ * better than a VIA, yet ironically I still get crackling, like many other
+ * people with the same chipset.
+ * Possible remedies:
+ * - use speaker (amplifier) output instead of headphone output
+ * (in case crackling is due to overloaded output clipping)
+ * - plug card into a different PCI slot, preferrably one that isn't shared
+ * too much (this helps a lot, but not completely!)
+ * - get rid of PCI VGA card, use AGP instead
+ * - upgrade or downgrade BIOS
+ * - fiddle with PCI latency settings (setpci -v -s BUSID latency_timer=XX)
+ * Not too helpful.
+ * - Disable ACPI/power management/"Auto Detect RAM/PCI Clk" in BIOS
+ *
+ * BUGS
+ * - full-duplex might *still* be problematic, however a recent test was fine
+ * - (non-bug) "Bass/Treble or 3D settings don't work" - they do get evaluated
+ * if you set PCM output switch to "pre 3D" instead of "post 3D".
+ * If this can't be set, then get a mixer application that Isn't Stupid (tm)
+ * (e.g. kmix, gamix) - unfortunately several are!!
+ * - locking is not entirely clean, especially the audio stream activity
+ * ints --> may be racy
+ * - an _unconnected_ secondary joystick at the gameport will be reported
+ * to be "active" (floating values, not precisely -1) due to the way we need
+ * to read the Digital Enhanced Game Port. Not sure whether it is fixable.
+ *
+ * TODO
+ * - test MPU401 MIDI playback etc.
+ * - add more power micro-management (disable various units of the card
+ * as long as they're unused). However this requires more I/O ports which I
+ * haven't figured out yet and which thus might not even exist...
+ * The standard suspend/resume functionality could probably make use of
+ * some improvement, too...
+ * - figure out what all unknown port bits are responsible for
+ * - figure out some cleverly evil scheme to possibly make ALSA AC97 code
+ * fully accept our quite incompatible ""AC97"" mixer and thus save some
+ * code (but I'm not too optimistic that doing this is possible at all)
+ * - use MMIO (memory-mapped I/O)? Slightly faster access, e.g. for gameport.
+ */
+
+#include <asm/io.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+#include "azt3328.h"
+
+MODULE_AUTHOR("Andreas Mohr <andi AT lisas.de>");
+MODULE_DESCRIPTION("Aztech AZF3328 (PCI168)");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_GAMEPORT 1
+#endif
+
+#define DEBUG_MISC 0
+#define DEBUG_CALLS 0
+#define DEBUG_MIXER 0
+#define DEBUG_PLAY_REC 0
+#define DEBUG_IO 0
+#define DEBUG_TIMER 0
+#define DEBUG_GAME 0
+#define MIXER_TESTING 0
+
+#if DEBUG_MISC
+#define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args)
+#else
+#define snd_azf3328_dbgmisc(format, args...)
+#endif
+
+#if DEBUG_CALLS
+#define snd_azf3328_dbgcalls(format, args...) printk(format, ##args)
+#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__)
+#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__)
+#else
+#define snd_azf3328_dbgcalls(format, args...)
+#define snd_azf3328_dbgcallenter()
+#define snd_azf3328_dbgcallleave()
+#endif
+
+#if DEBUG_MIXER
+#define snd_azf3328_dbgmixer(format, args...) printk(format, ##args)
+#else
+#define snd_azf3328_dbgmixer(format, args...)
+#endif
+
+#if DEBUG_PLAY_REC
+#define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args)
+#else
+#define snd_azf3328_dbgplay(format, args...)
+#endif
+
+#if DEBUG_MISC
+#define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args)
+#else
+#define snd_azf3328_dbgtimer(format, args...)
+#endif
+
+#if DEBUG_GAME
+#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args)
+#else
+#define snd_azf3328_dbggame(format, args...)
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for AZF3328 soundcard.");
+
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for AZF3328 soundcard.");
+
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard.");
+
+static int seqtimer_scaling = 128;
+module_param(seqtimer_scaling, int, 0444);
+MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128.");
+
+struct snd_azf3328_audio_stream {
+ struct snd_pcm_substream *substream;
+ int enabled;
+ int running;
+ unsigned long portbase;
+};
+
+enum snd_azf3328_stream_index {
+ AZF_PLAYBACK = 0,
+ AZF_CAPTURE = 1,
+};
+
+struct snd_azf3328 {
+ /* often-used fields towards beginning, then grouped */
+
+ unsigned long codec_io; /* usually 0xb000, size 128 */
+ unsigned long game_io; /* usually 0xb400, size 8 */
+ unsigned long mpu_io; /* usually 0xb800, size 4 */
+ unsigned long opl3_io; /* usually 0xbc00, size 8 */
+ unsigned long mixer_io; /* usually 0xc000, size 64 */
+
+ spinlock_t reg_lock;
+
+ struct snd_timer *timer;
+
+ struct snd_pcm *pcm;
+ struct snd_azf3328_audio_stream audio_stream[2];
+
+ struct snd_card *card;
+ struct snd_rawmidi *rmidi;
+
+#ifdef SUPPORT_GAMEPORT
+ struct gameport *gameport;
+ int axes[4];
+#endif
+
+ struct pci_dev *pci;
+ int irq;
+
+ /* register 0x6a is write-only, thus need to remember setting.
+ * If we need to add more registers here, then we might try to fold this
+ * into some transparent combined shadow register handling with
+ * CONFIG_PM register storage below, but that's slightly difficult. */
+ u16 shadow_reg_codec_6AH;
+
+#ifdef CONFIG_PM
+ /* register value containers for power management
+ * Note: not always full I/O range preserved (just like Win driver!) */
+ u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2];
+ u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2];
+ u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2];
+ u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2];
+ u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2];
+#endif
+};
+
+static const struct pci_device_id snd_azf3328_ids[] = {
+ { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */
+ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_azf3328_ids);
+
+
+static int
+snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set)
+{
+ u8 prev = inb(reg), new;
+
+ new = (do_set) ? (prev|mask) : (prev & ~mask);
+ /* we need to always write the new value no matter whether it differs
+ * or not, since some register bits don't indicate their setting */
+ outb(new, reg);
+ if (new != prev)
+ return 1;
+
+ return 0;
+}
+
+static inline void
+snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value)
+{
+ outb(value, chip->codec_io + reg);
+}
+
+static inline u8
+snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg)
+{
+ return inb(chip->codec_io + reg);
+}
+
+static inline void
+snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
+{
+ outw(value, chip->codec_io + reg);
+}
+
+static inline u16
+snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg)
+{
+ return inw(chip->codec_io + reg);
+}
+
+static inline void
+snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value)
+{
+ outl(value, chip->codec_io + reg);
+}
+
+static inline u32
+snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg)
+{
+ return inl(chip->codec_io + reg);
+}
+
+static inline void
+snd_azf3328_game_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value)
+{
+ outb(value, chip->game_io + reg);
+}
+
+static inline void
+snd_azf3328_game_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
+{
+ outw(value, chip->game_io + reg);
+}
+
+static inline u8
+snd_azf3328_game_inb(const struct snd_azf3328 *chip, unsigned reg)
+{
+ return inb(chip->game_io + reg);
+}
+
+static inline u16
+snd_azf3328_game_inw(const struct snd_azf3328 *chip, unsigned reg)
+{
+ return inw(chip->game_io + reg);
+}
+
+static inline void
+snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value)
+{
+ outw(value, chip->mixer_io + reg);
+}
+
+static inline u16
+snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg)
+{
+ return inw(chip->mixer_io + reg);
+}
+
+#define AZF_MUTE_BIT 0x80
+
+static int
+snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip,
+ unsigned reg, int do_mute
+)
+{
+ unsigned long portbase = chip->mixer_io + reg + 1;
+ int updated;
+
+ /* the mute bit is on the *second* (i.e. right) register of a
+ * left/right channel setting */
+ updated = snd_azf3328_io_reg_setb(portbase, AZF_MUTE_BIT, do_mute);
+
+ /* indicate whether it was muted before */
+ return (do_mute) ? !updated : updated;
+}
+
+static void
+snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip,
+ unsigned reg,
+ unsigned char dst_vol_left,
+ unsigned char dst_vol_right,
+ int chan_sel, int delay
+)
+{
+ unsigned long portbase = chip->mixer_io + reg;
+ unsigned char curr_vol_left = 0, curr_vol_right = 0;
+ int left_change = 0, right_change = 0;
+
+ snd_azf3328_dbgcallenter();
+
+ if (chan_sel & SET_CHAN_LEFT) {
+ curr_vol_left = inb(portbase + 1);
+
+ /* take care of muting flag contained in left channel */
+ if (curr_vol_left & AZF_MUTE_BIT)
+ dst_vol_left |= AZF_MUTE_BIT;
+ else
+ dst_vol_left &= ~AZF_MUTE_BIT;
+
+ left_change = (curr_vol_left > dst_vol_left) ? -1 : 1;
+ }
+
+ if (chan_sel & SET_CHAN_RIGHT) {
+ curr_vol_right = inb(portbase + 0);
+
+ right_change = (curr_vol_right > dst_vol_right) ? -1 : 1;
+ }
+
+ do {
+ if (left_change) {
+ if (curr_vol_left != dst_vol_left) {
+ curr_vol_left += left_change;
+ outb(curr_vol_left, portbase + 1);
+ } else
+ left_change = 0;
+ }
+ if (right_change) {
+ if (curr_vol_right != dst_vol_right) {
+ curr_vol_right += right_change;
+
+ /* during volume change, the right channel is crackling
+ * somewhat more than the left channel, unfortunately.
+ * This seems to be a hardware issue. */
+ outb(curr_vol_right, portbase + 0);
+ } else
+ right_change = 0;
+ }
+ if (delay)
+ mdelay(delay);
+ } while ((left_change) || (right_change));
+ snd_azf3328_dbgcallleave();
+}
+
+/*
+ * general mixer element
+ */
+struct azf3328_mixer_reg {
+ unsigned reg;
+ unsigned int lchan_shift, rchan_shift;
+ unsigned int mask;
+ unsigned int invert: 1;
+ unsigned int stereo: 1;
+ unsigned int enum_c: 4;
+};
+
+#define COMPOSE_MIXER_REG(reg,lchan_shift,rchan_shift,mask,invert,stereo,enum_c) \
+ ((reg) | (lchan_shift << 8) | (rchan_shift << 12) | \
+ (mask << 16) | \
+ (invert << 24) | \
+ (stereo << 25) | \
+ (enum_c << 26))
+
+static void snd_azf3328_mixer_reg_decode(struct azf3328_mixer_reg *r, unsigned long val)
+{
+ r->reg = val & 0xff;
+ r->lchan_shift = (val >> 8) & 0x0f;
+ r->rchan_shift = (val >> 12) & 0x0f;
+ r->mask = (val >> 16) & 0xff;
+ r->invert = (val >> 24) & 1;
+ r->stereo = (val >> 25) & 1;
+ r->enum_c = (val >> 26) & 0x0f;
+}
+
+/*
+ * mixer switches/volumes
+ */
+
+#define AZF3328_MIXER_SWITCH(xname, reg, shift, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_azf3328_info_mixer, \
+ .get = snd_azf3328_get_mixer, .put = snd_azf3328_put_mixer, \
+ .private_value = COMPOSE_MIXER_REG(reg, shift, 0, 0x1, invert, 0, 0), \
+}
+
+#define AZF3328_MIXER_VOL_STEREO(xname, reg, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_azf3328_info_mixer, \
+ .get = snd_azf3328_get_mixer, .put = snd_azf3328_put_mixer, \
+ .private_value = COMPOSE_MIXER_REG(reg, 8, 0, mask, invert, 1, 0), \
+}
+
+#define AZF3328_MIXER_VOL_MONO(xname, reg, mask, is_right_chan) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_azf3328_info_mixer, \
+ .get = snd_azf3328_get_mixer, .put = snd_azf3328_put_mixer, \
+ .private_value = COMPOSE_MIXER_REG(reg, is_right_chan ? 0 : 8, 0, mask, 1, 0, 0), \
+}
+
+#define AZF3328_MIXER_VOL_SPECIAL(xname, reg, mask, shift, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_azf3328_info_mixer, \
+ .get = snd_azf3328_get_mixer, .put = snd_azf3328_put_mixer, \
+ .private_value = COMPOSE_MIXER_REG(reg, shift, 0, mask, invert, 0, 0), \
+}
+
+#define AZF3328_MIXER_ENUM(xname, reg, enum_c, shift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_azf3328_info_mixer_enum, \
+ .get = snd_azf3328_get_mixer_enum, .put = snd_azf3328_put_mixer_enum, \
+ .private_value = COMPOSE_MIXER_REG(reg, shift, 0, 0, 0, 0, enum_c), \
+}
+
+static int
+snd_azf3328_info_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct azf3328_mixer_reg reg;
+
+ snd_azf3328_dbgcallenter();
+ snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
+ uinfo->type = reg.mask == 1 ?
+ SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = reg.stereo + 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = reg.mask;
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_get_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
+ struct azf3328_mixer_reg reg;
+ unsigned int oreg, val;
+
+ snd_azf3328_dbgcallenter();
+ snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
+
+ oreg = snd_azf3328_mixer_inw(chip, reg.reg);
+ val = (oreg >> reg.lchan_shift) & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ ucontrol->value.integer.value[0] = val;
+ if (reg.stereo) {
+ val = (oreg >> reg.rchan_shift) & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ ucontrol->value.integer.value[1] = val;
+ }
+ snd_azf3328_dbgmixer("get: %02x is %04x -> vol %02lx|%02lx "
+ "(shift %02d|%02d, mask %02x, inv. %d, stereo %d)\n",
+ reg.reg, oreg,
+ ucontrol->value.integer.value[0], ucontrol->value.integer.value[1],
+ reg.lchan_shift, reg.rchan_shift, reg.mask, reg.invert, reg.stereo);
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_put_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
+ struct azf3328_mixer_reg reg;
+ unsigned int oreg, nreg, val;
+
+ snd_azf3328_dbgcallenter();
+ snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
+ oreg = snd_azf3328_mixer_inw(chip, reg.reg);
+ val = ucontrol->value.integer.value[0] & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ nreg = oreg & ~(reg.mask << reg.lchan_shift);
+ nreg |= (val << reg.lchan_shift);
+ if (reg.stereo) {
+ val = ucontrol->value.integer.value[1] & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ nreg &= ~(reg.mask << reg.rchan_shift);
+ nreg |= (val << reg.rchan_shift);
+ }
+ if (reg.mask >= 0x07) /* it's a volume control, so better take care */
+ snd_azf3328_mixer_write_volume_gradually(
+ chip, reg.reg, nreg >> 8, nreg & 0xff,
+ /* just set both channels, doesn't matter */
+ SET_CHAN_LEFT|SET_CHAN_RIGHT,
+ 0);
+ else
+ snd_azf3328_mixer_outw(chip, reg.reg, nreg);
+
+ snd_azf3328_dbgmixer("put: %02x to %02lx|%02lx, "
+ "oreg %04x; shift %02d|%02d -> nreg %04x; after: %04x\n",
+ reg.reg, ucontrol->value.integer.value[0], ucontrol->value.integer.value[1],
+ oreg, reg.lchan_shift, reg.rchan_shift,
+ nreg, snd_azf3328_mixer_inw(chip, reg.reg));
+ snd_azf3328_dbgcallleave();
+ return (nreg != oreg);
+}
+
+static int
+snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts1[] = {
+ "Mic1", "Mic2"
+ };
+ static const char * const texts2[] = {
+ "Mix", "Mic"
+ };
+ static const char * const texts3[] = {
+ "Mic", "CD", "Video", "Aux",
+ "Line", "Mix", "Mix Mono", "Phone"
+ };
+ static const char * const texts4[] = {
+ "pre 3D", "post 3D"
+ };
+ struct azf3328_mixer_reg reg;
+ const char * const *p = NULL;
+
+ snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = (reg.reg == IDX_MIXER_REC_SELECT) ? 2 : 1;
+ uinfo->value.enumerated.items = reg.enum_c;
+ if (uinfo->value.enumerated.item > reg.enum_c - 1U)
+ uinfo->value.enumerated.item = reg.enum_c - 1U;
+ if (reg.reg == IDX_MIXER_ADVCTL2) {
+ switch(reg.lchan_shift) {
+ case 8: /* modem out sel */
+ p = texts1;
+ break;
+ case 9: /* mono sel source */
+ p = texts2;
+ break;
+ case 15: /* PCM Out Path */
+ p = texts4;
+ break;
+ }
+ } else
+ if (reg.reg == IDX_MIXER_REC_SELECT)
+ p = texts3;
+
+ strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int
+snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
+ struct azf3328_mixer_reg reg;
+ unsigned short val;
+
+ snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
+ val = snd_azf3328_mixer_inw(chip, reg.reg);
+ if (reg.reg == IDX_MIXER_REC_SELECT) {
+ ucontrol->value.enumerated.item[0] = (val >> 8) & (reg.enum_c - 1);
+ ucontrol->value.enumerated.item[1] = (val >> 0) & (reg.enum_c - 1);
+ } else
+ ucontrol->value.enumerated.item[0] = (val >> reg.lchan_shift) & (reg.enum_c - 1);
+
+ snd_azf3328_dbgmixer("get_enum: %02x is %04x -> %d|%d (shift %02d, enum_c %d)\n",
+ reg.reg, val, ucontrol->value.enumerated.item[0], ucontrol->value.enumerated.item[1],
+ reg.lchan_shift, reg.enum_c);
+ return 0;
+}
+
+static int
+snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol);
+ struct azf3328_mixer_reg reg;
+ unsigned int oreg, nreg, val;
+
+ snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
+ oreg = snd_azf3328_mixer_inw(chip, reg.reg);
+ val = oreg;
+ if (reg.reg == IDX_MIXER_REC_SELECT) {
+ if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U ||
+ ucontrol->value.enumerated.item[1] > reg.enum_c - 1U)
+ return -EINVAL;
+ val = (ucontrol->value.enumerated.item[0] << 8) |
+ (ucontrol->value.enumerated.item[1] << 0);
+ } else {
+ if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U)
+ return -EINVAL;
+ val &= ~((reg.enum_c - 1) << reg.lchan_shift);
+ val |= (ucontrol->value.enumerated.item[0] << reg.lchan_shift);
+ }
+ snd_azf3328_mixer_outw(chip, reg.reg, val);
+ nreg = val;
+
+ snd_azf3328_dbgmixer("put_enum: %02x to %04x, oreg %04x\n", reg.reg, val, oreg);
+ return (nreg != oreg);
+}
+
+static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = {
+ AZF3328_MIXER_SWITCH("Master Playback Switch", IDX_MIXER_PLAY_MASTER, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("Master Playback Volume", IDX_MIXER_PLAY_MASTER, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("PCM Playback Switch", IDX_MIXER_WAVEOUT, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("PCM Playback Volume",
+ IDX_MIXER_WAVEOUT, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("PCM 3D Bypass Playback Switch",
+ IDX_MIXER_ADVCTL2, 7, 1),
+ AZF3328_MIXER_SWITCH("FM Playback Switch", IDX_MIXER_FMSYNTH, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("FM Playback Volume", IDX_MIXER_FMSYNTH, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("CD Playback Switch", IDX_MIXER_CDAUDIO, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("CD Playback Volume", IDX_MIXER_CDAUDIO, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("Capture Switch", IDX_MIXER_REC_VOLUME, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("Capture Volume", IDX_MIXER_REC_VOLUME, 0x0f, 0),
+ AZF3328_MIXER_ENUM("Capture Source", IDX_MIXER_REC_SELECT, 8, 0),
+ AZF3328_MIXER_SWITCH("Mic Playback Switch", IDX_MIXER_MIC, 15, 1),
+ AZF3328_MIXER_VOL_MONO("Mic Playback Volume", IDX_MIXER_MIC, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("Mic Boost (+20dB)", IDX_MIXER_MIC, 6, 0),
+ AZF3328_MIXER_SWITCH("Line Playback Switch", IDX_MIXER_LINEIN, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("Line Playback Volume", IDX_MIXER_LINEIN, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("PC Speaker Playback Switch", IDX_MIXER_PCBEEP, 15, 1),
+ AZF3328_MIXER_VOL_SPECIAL("PC Speaker Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1),
+ AZF3328_MIXER_SWITCH("Video Playback Switch", IDX_MIXER_VIDEO, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("Video Playback Volume", IDX_MIXER_VIDEO, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("Aux Playback Switch", IDX_MIXER_AUX, 15, 1),
+ AZF3328_MIXER_VOL_STEREO("Aux Playback Volume", IDX_MIXER_AUX, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("Modem Playback Switch", IDX_MIXER_MODEMOUT, 15, 1),
+ AZF3328_MIXER_VOL_MONO("Modem Playback Volume", IDX_MIXER_MODEMOUT, 0x1f, 1),
+ AZF3328_MIXER_SWITCH("Modem Capture Switch", IDX_MIXER_MODEMIN, 15, 1),
+ AZF3328_MIXER_VOL_MONO("Modem Capture Volume", IDX_MIXER_MODEMIN, 0x1f, 1),
+ AZF3328_MIXER_ENUM("Mic Select", IDX_MIXER_ADVCTL2, 2, 8),
+ AZF3328_MIXER_ENUM("Mono Output Select", IDX_MIXER_ADVCTL2, 2, 9),
+ AZF3328_MIXER_ENUM("PCM Output Route", IDX_MIXER_ADVCTL2, 2, 15), /* PCM Out Path, place in front since it controls *both* 3D and Bass/Treble! */
+ AZF3328_MIXER_VOL_SPECIAL("Tone Control - Treble", IDX_MIXER_BASSTREBLE, 0x07, 1, 0),
+ AZF3328_MIXER_VOL_SPECIAL("Tone Control - Bass", IDX_MIXER_BASSTREBLE, 0x07, 9, 0),
+ AZF3328_MIXER_SWITCH("3D Control - Switch", IDX_MIXER_ADVCTL2, 13, 0),
+ AZF3328_MIXER_VOL_SPECIAL("3D Control - Width", IDX_MIXER_ADVCTL1, 0x07, 1, 0), /* "3D Width" */
+ AZF3328_MIXER_VOL_SPECIAL("3D Control - Depth", IDX_MIXER_ADVCTL1, 0x03, 8, 0), /* "Hifi 3D" */
+#if MIXER_TESTING
+ AZF3328_MIXER_SWITCH("0", IDX_MIXER_ADVCTL2, 0, 0),
+ AZF3328_MIXER_SWITCH("1", IDX_MIXER_ADVCTL2, 1, 0),
+ AZF3328_MIXER_SWITCH("2", IDX_MIXER_ADVCTL2, 2, 0),
+ AZF3328_MIXER_SWITCH("3", IDX_MIXER_ADVCTL2, 3, 0),
+ AZF3328_MIXER_SWITCH("4", IDX_MIXER_ADVCTL2, 4, 0),
+ AZF3328_MIXER_SWITCH("5", IDX_MIXER_ADVCTL2, 5, 0),
+ AZF3328_MIXER_SWITCH("6", IDX_MIXER_ADVCTL2, 6, 0),
+ AZF3328_MIXER_SWITCH("7", IDX_MIXER_ADVCTL2, 7, 0),
+ AZF3328_MIXER_SWITCH("8", IDX_MIXER_ADVCTL2, 8, 0),
+ AZF3328_MIXER_SWITCH("9", IDX_MIXER_ADVCTL2, 9, 0),
+ AZF3328_MIXER_SWITCH("10", IDX_MIXER_ADVCTL2, 10, 0),
+ AZF3328_MIXER_SWITCH("11", IDX_MIXER_ADVCTL2, 11, 0),
+ AZF3328_MIXER_SWITCH("12", IDX_MIXER_ADVCTL2, 12, 0),
+ AZF3328_MIXER_SWITCH("13", IDX_MIXER_ADVCTL2, 13, 0),
+ AZF3328_MIXER_SWITCH("14", IDX_MIXER_ADVCTL2, 14, 0),
+ AZF3328_MIXER_SWITCH("15", IDX_MIXER_ADVCTL2, 15, 0),
+#endif
+};
+
+static u16 __devinitdata snd_azf3328_init_values[][2] = {
+ { IDX_MIXER_PLAY_MASTER, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_MODEMOUT, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_BASSTREBLE, 0x0000 },
+ { IDX_MIXER_PCBEEP, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_MODEMIN, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_MIC, MIXER_MUTE_MASK|0x001f },
+ { IDX_MIXER_LINEIN, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_CDAUDIO, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_VIDEO, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_AUX, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_WAVEOUT, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_FMSYNTH, MIXER_MUTE_MASK|0x1f1f },
+ { IDX_MIXER_REC_VOLUME, MIXER_MUTE_MASK|0x0707 },
+};
+
+static int __devinit
+snd_azf3328_mixer_new(struct snd_azf3328 *chip)
+{
+ struct snd_card *card;
+ const struct snd_kcontrol_new *sw;
+ unsigned int idx;
+ int err;
+
+ snd_azf3328_dbgcallenter();
+ if (snd_BUG_ON(!chip || !chip->card))
+ return -EINVAL;
+
+ card = chip->card;
+
+ /* mixer reset */
+ snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000);
+
+ /* mute and zero volume channels */
+ for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); ++idx) {
+ snd_azf3328_mixer_outw(chip,
+ snd_azf3328_init_values[idx][0],
+ snd_azf3328_init_values[idx][1]);
+ }
+
+ /* add mixer controls */
+ sw = snd_azf3328_mixer_controls;
+ for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls);
+ ++idx, ++sw) {
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(sw, chip))) < 0)
+ return err;
+ }
+ snd_component_add(card, "AZF3328 mixer");
+ strcpy(card->mixername, "AZF3328 mixer");
+
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ int res;
+ snd_azf3328_dbgcallenter();
+ res = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ snd_azf3328_dbgcallleave();
+ return res;
+}
+
+static int
+snd_azf3328_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_azf3328_dbgcallenter();
+ snd_pcm_lib_free_pages(substream);
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static void
+snd_azf3328_codec_setfmt(struct snd_azf3328 *chip,
+ unsigned reg,
+ enum azf_freq_t bitrate,
+ unsigned int format_width,
+ unsigned int channels
+)
+{
+ u16 val = 0xff00;
+ unsigned long flags;
+
+ snd_azf3328_dbgcallenter();
+ switch (bitrate) {
+ case AZF_FREQ_4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+ case AZF_FREQ_4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+ case AZF_FREQ_5512:
+ /* the AZF3328 names it "5510" for some strange reason */
+ val |= SOUNDFORMAT_FREQ_5510; break;
+ case AZF_FREQ_6620: val |= SOUNDFORMAT_FREQ_6620; break;
+ case AZF_FREQ_8000: val |= SOUNDFORMAT_FREQ_8000; break;
+ case AZF_FREQ_9600: val |= SOUNDFORMAT_FREQ_9600; break;
+ case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break;
+ case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+ case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break;
+ case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break;
+ case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break;
+ default:
+ snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
+ /* fall-through */
+ case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break;
+ case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break;
+ case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
+ }
+ /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
+ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
+ /* val = 0xff0a; 47m30.599s (4764,891Hz; -> 4800Hz???) yup, 4803Hz */
+ /* val = 0xff0c; 57m0.510s (4010,263Hz; -> 4000Hz???) yup, 4003Hz */
+ /* val = 0xff05; 5m11.556s (... -> 44100Hz) */
+ /* val = 0xff03; 10m21.529s (21872,463Hz; -> 22050Hz???) */
+ /* val = 0xff0f; 20m41.883s (10937,993Hz; -> 11025Hz???) */
+ /* val = 0xff0d; 41m23.135s (5523,600Hz; -> 5512Hz???) */
+ /* val = 0xff0e; 28m30.777s (8017Hz; -> 8000Hz???) */
+
+ if (channels == 2)
+ val |= SOUNDFORMAT_FLAG_2CHANNELS;
+
+ if (format_width == 16)
+ val |= SOUNDFORMAT_FLAG_16BIT;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ /* set bitrate/format */
+ snd_azf3328_codec_outw(chip, reg, val);
+
+ /* changing the bitrate/format settings switches off the
+ * audio output with an annoying click in case of 8/16bit format change
+ * (maybe shutting down DAC/ADC?), thus immediately
+ * do some tweaking to reenable it and get rid of the clicking
+ * (FIXME: yes, it works, but what exactly am I doing here?? :)
+ * FIXME: does this have some side effects for full-duplex
+ * or other dramatic side effects? */
+ if (reg == IDX_IO_PLAY_SOUNDFORMAT) /* only do it for playback */
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) |
+ DMA_PLAY_SOMETHING1 |
+ DMA_PLAY_SOMETHING2 |
+ SOMETHING_ALMOST_ALWAYS_SET |
+ DMA_EPILOGUE_SOMETHING |
+ DMA_SOMETHING_ELSE
+ );
+
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_azf3328_dbgcallleave();
+}
+
+static inline void
+snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip,
+ unsigned reg
+)
+{
+ /* choose lowest frequency for low power consumption.
+ * While this will cause louder noise due to rather coarse frequency,
+ * it should never matter since output should always
+ * get disabled properly when idle anyway. */
+ snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1);
+}
+
+static void
+snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip,
+ unsigned bitmask,
+ int enable
+)
+{
+ if (enable)
+ chip->shadow_reg_codec_6AH &= ~bitmask;
+ else
+ chip->shadow_reg_codec_6AH |= bitmask;
+ snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n",
+ bitmask, enable, chip->shadow_reg_codec_6AH);
+ snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH);
+}
+
+static inline void
+snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable)
+{
+ snd_azf3328_dbgplay("codec_enable %d\n", enable);
+ /* no idea what exactly is being done here, but I strongly assume it's
+ * PM related */
+ snd_azf3328_codec_reg_6AH_update(
+ chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable
+ );
+}
+
+static void
+snd_azf3328_codec_activity(struct snd_azf3328 *chip,
+ enum snd_azf3328_stream_index stream_type,
+ int enable
+)
+{
+ int need_change = (chip->audio_stream[stream_type].running != enable);
+
+ snd_azf3328_dbgplay(
+ "codec_activity: type %d, enable %d, need_change %d\n",
+ stream_type, enable, need_change
+ );
+ if (need_change) {
+ enum snd_azf3328_stream_index other =
+ (stream_type == AZF_PLAYBACK) ?
+ AZF_CAPTURE : AZF_PLAYBACK;
+ /* small check to prevent shutting down the other party
+ * in case it's active */
+ if ((enable) || !(chip->audio_stream[other].running))
+ snd_azf3328_codec_enable(chip, enable);
+
+ /* ...and adjust clock, too
+ * (reduce noise and power consumption) */
+ if (!enable)
+ snd_azf3328_codec_setfmt_lowpower(
+ chip,
+ chip->audio_stream[stream_type].portbase
+ + IDX_IO_PLAY_SOUNDFORMAT
+ );
+ }
+ chip->audio_stream[stream_type].running = enable;
+}
+
+static void
+snd_azf3328_setdmaa(struct snd_azf3328 *chip,
+ long unsigned int addr,
+ unsigned int count,
+ unsigned int size,
+ enum snd_azf3328_stream_index stream_type
+)
+{
+ snd_azf3328_dbgcallenter();
+ if (!chip->audio_stream[stream_type].running) {
+ /* AZF3328 uses a two buffer pointer DMA playback approach */
+
+ unsigned long flags, portbase, addr_area2;
+
+ /* width 32bit (prevent overflow): */
+ unsigned long count_areas, count_tmp;
+
+ portbase = chip->audio_stream[stream_type].portbase;
+ count_areas = size/2;
+ addr_area2 = addr+count_areas;
+ count_areas--; /* max. index */
+ snd_azf3328_dbgplay("set DMA: buf1 %08lx[%lu], buf2 %08lx[%lu]\n", addr, count_areas, addr_area2, count_areas);
+
+ /* build combined I/O buffer length word */
+ count_tmp = count_areas;
+ count_areas |= (count_tmp << 16);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ outl(addr, portbase + IDX_IO_PLAY_DMA_START_1);
+ outl(addr_area2, portbase + IDX_IO_PLAY_DMA_START_2);
+ outl(count_areas, portbase + IDX_IO_PLAY_DMA_LEN_1);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
+ snd_azf3328_dbgcallleave();
+}
+
+static int
+snd_azf3328_playback_prepare(struct snd_pcm_substream *substream)
+{
+#if 0
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+#endif
+
+ snd_azf3328_dbgcallenter();
+#if 0
+ snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
+ runtime->rate,
+ snd_pcm_format_width(runtime->format),
+ runtime->channels);
+ snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK);
+#endif
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_capture_prepare(struct snd_pcm_substream *substream)
+{
+#if 0
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+#endif
+
+ snd_azf3328_dbgcallenter();
+#if 0
+ snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
+ runtime->rate,
+ snd_pcm_format_width(runtime->format),
+ runtime->channels);
+ snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE);
+#endif
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int result = 0;
+ unsigned int status1;
+ int previously_muted;
+
+ snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_azf3328_dbgplay("START PLAYBACK\n");
+
+ /* mute WaveOut (avoid clicking during setup) */
+ previously_muted =
+ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+
+ snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT,
+ runtime->rate,
+ snd_pcm_format_width(runtime->format),
+ runtime->channels);
+
+ spin_lock(&chip->reg_lock);
+ /* first, remember current value: */
+ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS);
+
+ /* stop playback */
+ status1 &= ~DMA_RESUME;
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+
+ /* FIXME: clear interrupts or what??? */
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff);
+ spin_unlock(&chip->reg_lock);
+
+ snd_azf3328_setdmaa(chip, runtime->dma_addr,
+ snd_pcm_lib_period_bytes(substream),
+ snd_pcm_lib_buffer_bytes(substream),
+ AZF_PLAYBACK);
+
+ spin_lock(&chip->reg_lock);
+#ifdef WIN9X
+ /* FIXME: enable playback/recording??? */
+ status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2;
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+
+ /* start playback again */
+ /* FIXME: what is this value (0x0010)??? */
+ status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING;
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+#else /* NT4 */
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ 0x0000);
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ DMA_PLAY_SOMETHING1);
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ DMA_PLAY_SOMETHING1 |
+ DMA_PLAY_SOMETHING2);
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ DMA_RESUME |
+ SOMETHING_ALMOST_ALWAYS_SET |
+ DMA_EPILOGUE_SOMETHING |
+ DMA_SOMETHING_ELSE);
+#endif
+ spin_unlock(&chip->reg_lock);
+ snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1);
+
+ /* now unmute WaveOut */
+ if (!previously_muted)
+ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
+
+ snd_azf3328_dbgplay("STARTED PLAYBACK\n");
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ snd_azf3328_dbgplay("RESUME PLAYBACK\n");
+ /* resume playback if we were active */
+ spin_lock(&chip->reg_lock);
+ if (chip->audio_stream[AZF_PLAYBACK].running)
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME);
+ spin_unlock(&chip->reg_lock);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_azf3328_dbgplay("STOP PLAYBACK\n");
+
+ /* mute WaveOut (avoid clicking during setup) */
+ previously_muted =
+ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+
+ spin_lock(&chip->reg_lock);
+ /* first, remember current value: */
+ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS);
+
+ /* stop playback */
+ status1 &= ~DMA_RESUME;
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+
+ /* hmm, is this really required? we're resetting the same bit
+ * immediately thereafter... */
+ status1 |= DMA_PLAY_SOMETHING1;
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+
+ status1 &= ~DMA_PLAY_SOMETHING1;
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1);
+ spin_unlock(&chip->reg_lock);
+ snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0);
+
+ /* now unmute WaveOut */
+ if (!previously_muted)
+ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0);
+
+ snd_azf3328_dbgplay("STOPPED PLAYBACK\n");
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ snd_azf3328_dbgplay("SUSPEND PLAYBACK\n");
+ /* make sure playback is stopped */
+ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+ snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) & ~DMA_RESUME);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n");
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n");
+ break;
+ default:
+ printk(KERN_ERR "FIXME: unknown trigger mode!\n");
+ return -EINVAL;
+ }
+
+ snd_azf3328_dbgcallleave();
+ return result;
+}
+
+/* this is just analogous to playback; I'm not quite sure whether recording
+ * should actually be triggered like that */
+static int
+snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int result = 0;
+ unsigned int status1;
+
+ snd_azf3328_dbgcalls("snd_azf3328_capture_trigger cmd %d\n", cmd);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+
+ snd_azf3328_dbgplay("START CAPTURE\n");
+
+ snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT,
+ runtime->rate,
+ snd_pcm_format_width(runtime->format),
+ runtime->channels);
+
+ spin_lock(&chip->reg_lock);
+ /* first, remember current value: */
+ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS);
+
+ /* stop recording */
+ status1 &= ~DMA_RESUME;
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
+
+ /* FIXME: clear interrupts or what??? */
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff);
+ spin_unlock(&chip->reg_lock);
+
+ snd_azf3328_setdmaa(chip, runtime->dma_addr,
+ snd_pcm_lib_period_bytes(substream),
+ snd_pcm_lib_buffer_bytes(substream),
+ AZF_CAPTURE);
+
+ spin_lock(&chip->reg_lock);
+#ifdef WIN9X
+ /* FIXME: enable playback/recording??? */
+ status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2;
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
+
+ /* start capture again */
+ /* FIXME: what is this value (0x0010)??? */
+ status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING;
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
+#else
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+ 0x0000);
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+ DMA_PLAY_SOMETHING1);
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+ DMA_PLAY_SOMETHING1 |
+ DMA_PLAY_SOMETHING2);
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+ DMA_RESUME |
+ SOMETHING_ALMOST_ALWAYS_SET |
+ DMA_EPILOGUE_SOMETHING |
+ DMA_SOMETHING_ELSE);
+#endif
+ spin_unlock(&chip->reg_lock);
+ snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1);
+
+ snd_azf3328_dbgplay("STARTED CAPTURE\n");
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ snd_azf3328_dbgplay("RESUME CAPTURE\n");
+ /* resume recording if we were active */
+ spin_lock(&chip->reg_lock);
+ if (chip->audio_stream[AZF_CAPTURE].running)
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+ snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME);
+ spin_unlock(&chip->reg_lock);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_azf3328_dbgplay("STOP CAPTURE\n");
+
+ spin_lock(&chip->reg_lock);
+ /* first, remember current value: */
+ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS);
+
+ /* stop recording */
+ status1 &= ~DMA_RESUME;
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
+
+ status1 |= DMA_PLAY_SOMETHING1;
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
+
+ status1 &= ~DMA_PLAY_SOMETHING1;
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1);
+ spin_unlock(&chip->reg_lock);
+ snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0);
+
+ snd_azf3328_dbgplay("STOPPED CAPTURE\n");
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ snd_azf3328_dbgplay("SUSPEND CAPTURE\n");
+ /* make sure recording is stopped */
+ snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+ snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) & ~DMA_RESUME);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n");
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n");
+ break;
+ default:
+ printk(KERN_ERR "FIXME: unknown trigger mode!\n");
+ return -EINVAL;
+ }
+
+ snd_azf3328_dbgcallleave();
+ return result;
+}
+
+static snd_pcm_uframes_t
+snd_azf3328_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ unsigned long bufptr, result;
+ snd_pcm_uframes_t frmres;
+
+#ifdef QUERY_HARDWARE
+ bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1);
+#else
+ bufptr = substream->runtime->dma_addr;
+#endif
+ result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS);
+
+ /* calculate offset */
+ result -= bufptr;
+ frmres = bytes_to_frames( substream->runtime, result);
+ snd_azf3328_dbgplay("PLAY @ 0x%8lx, frames %8ld\n", result, frmres);
+ return frmres;
+}
+
+static snd_pcm_uframes_t
+snd_azf3328_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ unsigned long bufptr, result;
+ snd_pcm_uframes_t frmres;
+
+#ifdef QUERY_HARDWARE
+ bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1);
+#else
+ bufptr = substream->runtime->dma_addr;
+#endif
+ result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS);
+
+ /* calculate offset */
+ result -= bufptr;
+ frmres = bytes_to_frames( substream->runtime, result);
+ snd_azf3328_dbgplay("REC @ 0x%8lx, frames %8ld\n", result, frmres);
+ return frmres;
+}
+
+/******************************************************************/
+
+#ifdef SUPPORT_GAMEPORT
+static inline void
+snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable)
+{
+ snd_azf3328_io_reg_setb(
+ chip->game_io+IDX_GAME_HWCONFIG,
+ GAME_HWCFG_IRQ_ENABLE,
+ enable
+ );
+}
+
+static inline void
+snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable)
+{
+ snd_azf3328_io_reg_setb(
+ chip->game_io+IDX_GAME_HWCONFIG,
+ GAME_HWCFG_LEGACY_ADDRESS_ENABLE,
+ enable
+ );
+}
+
+static inline void
+snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable)
+{
+ snd_azf3328_codec_reg_6AH_update(
+ chip, IO_6A_SOMETHING2_GAMEPORT, enable
+ );
+}
+
+static inline void
+snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip)
+{
+ /*
+ * skeleton handler only
+ * (we do not want axis reading in interrupt handler - too much load!)
+ */
+ snd_azf3328_dbggame("gameport irq\n");
+
+ /* this should ACK the gameport IRQ properly, hopefully. */
+ snd_azf3328_game_inw(chip, IDX_GAME_AXIS_VALUE);
+}
+
+static int
+snd_azf3328_gameport_open(struct gameport *gameport, int mode)
+{
+ struct snd_azf3328 *chip = gameport_get_port_data(gameport);
+ int res;
+
+ snd_azf3328_dbggame("gameport_open, mode %d\n", mode);
+ switch (mode) {
+ case GAMEPORT_MODE_COOKED:
+ case GAMEPORT_MODE_RAW:
+ res = 0;
+ break;
+ default:
+ res = -1;
+ break;
+ }
+
+ snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0));
+
+ return res;
+}
+
+static void
+snd_azf3328_gameport_close(struct gameport *gameport)
+{
+ struct snd_azf3328 *chip = gameport_get_port_data(gameport);
+
+ snd_azf3328_dbggame("gameport_close\n");
+ snd_azf3328_gameport_axis_circuit_enable(chip, 0);
+}
+
+static int
+snd_azf3328_gameport_cooked_read(struct gameport *gameport,
+ int *axes,
+ int *buttons
+)
+{
+ struct snd_azf3328 *chip = gameport_get_port_data(gameport);
+ int i;
+ u8 val;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = snd_azf3328_game_inb(chip, IDX_GAME_LEGACY_COMPATIBLE);
+ *buttons = (~(val) >> 4) & 0xf;
+
+ /* ok, this one is a bit dirty: cooked_read is being polled by a timer,
+ * thus we're atomic and cannot actively wait in here
+ * (which would be useful for us since it probably would be better
+ * to trigger a measurement in here, then wait a short amount of
+ * time until it's finished, then read values of _this_ measurement).
+ *
+ * Thus we simply resort to reading values if they're available already
+ * and trigger the next measurement.
+ */
+
+ val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG);
+ if (val & GAME_AXES_SAMPLING_READY) {
+ for (i = 0; i < 4; ++i) {
+ /* configure the axis to read */
+ val = (i << 4) | 0x0f;
+ snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val);
+
+ chip->axes[i] = snd_azf3328_game_inw(
+ chip, IDX_GAME_AXIS_VALUE
+ );
+ }
+ }
+
+ /* trigger next axes sampling, to be evaluated the next time we
+ * enter this function */
+
+ /* for some very, very strange reason we cannot enable
+ * Measurement Ready monitoring for all axes here,
+ * at least not when only one joystick connected */
+ val = 0x03; /* we're able to monitor axes 1 and 2 only */
+ snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val);
+
+ snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ for (i = 0; i < 4; i++) {
+ axes[i] = chip->axes[i];
+ if (axes[i] == 0xffff)
+ axes[i] = -1;
+ }
+
+ snd_azf3328_dbggame("cooked_read: axes %d %d %d %d buttons %d\n",
+ axes[0], axes[1], axes[2], axes[3], *buttons
+ );
+
+ return 0;
+}
+
+static int __devinit
+snd_azf3328_gameport(struct snd_azf3328 *chip, int dev)
+{
+ struct gameport *gp;
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n");
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "AZF3328 Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+ gp->io = chip->game_io;
+ gameport_set_port_data(gp, chip);
+
+ gp->open = snd_azf3328_gameport_open;
+ gp->close = snd_azf3328_gameport_close;
+ gp->fuzz = 16; /* seems ok */
+ gp->cooked_read = snd_azf3328_gameport_cooked_read;
+
+ /* DISABLE legacy address: we don't need it! */
+ snd_azf3328_gameport_legacy_address_enable(chip, 0);
+
+ snd_azf3328_gameport_axis_circuit_enable(chip, 0);
+
+ gameport_register_port(chip->gameport);
+
+ return 0;
+}
+
+static void
+snd_azf3328_gameport_free(struct snd_azf3328 *chip)
+{
+ if (chip->gameport) {
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+ }
+ snd_azf3328_gameport_irq_enable(chip, 0);
+}
+#else
+static inline int
+snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { return -ENOSYS; }
+static inline void
+snd_azf3328_gameport_free(struct snd_azf3328 *chip) { }
+static inline void
+snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip)
+{
+ printk(KERN_WARNING "huh, game port IRQ occurred!?\n");
+}
+#endif /* SUPPORT_GAMEPORT */
+
+/******************************************************************/
+
+static inline void
+snd_azf3328_irq_log_unknown_type(u8 which)
+{
+ snd_azf3328_dbgplay(
+ "azt3328: unknown IRQ type (%x) occurred, please report!\n",
+ which
+ );
+}
+
+static irqreturn_t
+snd_azf3328_interrupt(int irq, void *dev_id)
+{
+ struct snd_azf3328 *chip = dev_id;
+ u8 status, which;
+#if DEBUG_PLAY_REC
+ static unsigned long irq_count;
+#endif
+
+ status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS);
+
+ /* fast path out, to ease interrupt sharing */
+ if (!(status &
+ (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER)
+ ))
+ return IRQ_NONE; /* must be interrupt for another device */
+
+ snd_azf3328_dbgplay(
+ "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, "
+ "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n",
+ irq_count++ /* debug-only */,
+ snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS),
+ snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE),
+ status
+ );
+
+ if (status & IRQ_TIMER) {
+ /* snd_azf3328_dbgplay("timer %ld\n",
+ snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE)
+ & TIMER_VALUE_MASK
+ ); */
+ if (chip->timer)
+ snd_timer_interrupt(chip->timer, chip->timer->sticks);
+ /* ACK timer */
+ spin_lock(&chip->reg_lock);
+ snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x07);
+ spin_unlock(&chip->reg_lock);
+ snd_azf3328_dbgplay("azt3328: timer IRQ\n");
+ }
+ if (status & IRQ_PLAYBACK) {
+ spin_lock(&chip->reg_lock);
+ which = snd_azf3328_codec_inb(chip, IDX_IO_PLAY_IRQTYPE);
+ /* ack all IRQ types immediately */
+ snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which);
+ spin_unlock(&chip->reg_lock);
+
+ if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) {
+ snd_pcm_period_elapsed(
+ chip->audio_stream[AZF_PLAYBACK].substream
+ );
+ snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n",
+ which,
+ snd_azf3328_codec_inl(
+ chip, IDX_IO_PLAY_DMA_CURRPOS
+ )
+ );
+ } else
+ printk(KERN_WARNING "azt3328: irq handler problem!\n");
+ if (which & IRQ_PLAY_SOMETHING)
+ snd_azf3328_irq_log_unknown_type(which);
+ }
+ if (status & IRQ_RECORDING) {
+ spin_lock(&chip->reg_lock);
+ which = snd_azf3328_codec_inb(chip, IDX_IO_REC_IRQTYPE);
+ /* ack all IRQ types immediately */
+ snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which);
+ spin_unlock(&chip->reg_lock);
+
+ if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) {
+ snd_pcm_period_elapsed(
+ chip->audio_stream[AZF_CAPTURE].substream
+ );
+ snd_azf3328_dbgplay("REC period done (#%x), @ %x\n",
+ which,
+ snd_azf3328_codec_inl(
+ chip, IDX_IO_REC_DMA_CURRPOS
+ )
+ );
+ } else
+ printk(KERN_WARNING "azt3328: irq handler problem!\n");
+ if (which & IRQ_REC_SOMETHING)
+ snd_azf3328_irq_log_unknown_type(which);
+ }
+ if (status & IRQ_GAMEPORT)
+ snd_azf3328_gameport_interrupt(chip);
+ /* MPU401 has less critical IRQ requirements
+ * than timer and playback/recording, right? */
+ if (status & IRQ_MPU401) {
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+
+ /* hmm, do we have to ack the IRQ here somehow?
+ * If so, then I don't know how... */
+ snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n");
+ }
+ return IRQ_HANDLED;
+}
+
+/*****************************************************************/
+
+static const struct snd_pcm_hardware snd_azf3328_playback =
+{
+ /* FIXME!! Correct? */
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_LE,
+ .rates = SNDRV_PCM_RATE_5512 |
+ SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_KNOT,
+ .rate_min = AZF_FREQ_4000,
+ .rate_max = AZF_FREQ_66200,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ /* FIXME: maybe that card actually has a FIFO?
+ * Hmm, it seems newer revisions do have one, but we still don't know
+ * its size... */
+ .fifo_size = 0,
+};
+
+static const struct snd_pcm_hardware snd_azf3328_capture =
+{
+ /* FIXME */
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_LE,
+ .rates = SNDRV_PCM_RATE_5512 |
+ SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_KNOT,
+ .rate_min = AZF_FREQ_4000,
+ .rate_max = AZF_FREQ_66200,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 64,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+
+static unsigned int snd_azf3328_fixed_rates[] = {
+ AZF_FREQ_4000,
+ AZF_FREQ_4800,
+ AZF_FREQ_5512,
+ AZF_FREQ_6620,
+ AZF_FREQ_8000,
+ AZF_FREQ_9600,
+ AZF_FREQ_11025,
+ AZF_FREQ_13240,
+ AZF_FREQ_16000,
+ AZF_FREQ_22050,
+ AZF_FREQ_32000,
+ AZF_FREQ_44100,
+ AZF_FREQ_48000,
+ AZF_FREQ_66200
+};
+
+static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = {
+ .count = ARRAY_SIZE(snd_azf3328_fixed_rates),
+ .list = snd_azf3328_fixed_rates,
+ .mask = 0,
+};
+
+/*****************************************************************/
+
+static int
+snd_azf3328_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_azf3328_dbgcallenter();
+ chip->audio_stream[AZF_PLAYBACK].substream = substream;
+ runtime->hw = snd_azf3328_playback;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_azf3328_hw_constraints_rates);
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_azf3328_dbgcallenter();
+ chip->audio_stream[AZF_CAPTURE].substream = substream;
+ runtime->hw = snd_azf3328_capture;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_azf3328_hw_constraints_rates);
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+
+ snd_azf3328_dbgcallenter();
+ chip->audio_stream[AZF_PLAYBACK].substream = NULL;
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream);
+
+ snd_azf3328_dbgcallenter();
+ chip->audio_stream[AZF_CAPTURE].substream = NULL;
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+/******************************************************************/
+
+static struct snd_pcm_ops snd_azf3328_playback_ops = {
+ .open = snd_azf3328_playback_open,
+ .close = snd_azf3328_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_azf3328_hw_params,
+ .hw_free = snd_azf3328_hw_free,
+ .prepare = snd_azf3328_playback_prepare,
+ .trigger = snd_azf3328_playback_trigger,
+ .pointer = snd_azf3328_playback_pointer
+};
+
+static struct snd_pcm_ops snd_azf3328_capture_ops = {
+ .open = snd_azf3328_capture_open,
+ .close = snd_azf3328_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_azf3328_hw_params,
+ .hw_free = snd_azf3328_hw_free,
+ .prepare = snd_azf3328_capture_prepare,
+ .trigger = snd_azf3328_capture_trigger,
+ .pointer = snd_azf3328_capture_pointer
+};
+
+static int __devinit
+snd_azf3328_pcm(struct snd_azf3328 *chip, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ snd_azf3328_dbgcallenter();
+ if ((err = snd_pcm_new(chip->card, "AZF3328 DSP", device, 1, 1, &pcm)) < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_azf3328_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_azf3328_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 64*1024);
+
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+/******************************************************************/
+
+/*** NOTE: the physical timer resolution actually is 1024000 ticks per second
+ *** (probably derived from main crystal via a divider of 24),
+ *** but announcing those attributes to user-space would make programs
+ *** configure the timer to a 1 tick value, resulting in an absolutely fatal
+ *** timer IRQ storm.
+ *** Thus I chose to announce a down-scaled virtual timer to the outside and
+ *** calculate real timer countdown values internally.
+ *** (the scale factor can be set via module parameter "seqtimer_scaling").
+ ***/
+
+static int
+snd_azf3328_timer_start(struct snd_timer *timer)
+{
+ struct snd_azf3328 *chip;
+ unsigned long flags;
+ unsigned int delay;
+
+ snd_azf3328_dbgcallenter();
+ chip = snd_timer_chip(timer);
+ delay = ((timer->sticks * seqtimer_scaling) - 1) & TIMER_VALUE_MASK;
+ if (delay < 49) {
+ /* uhoh, that's not good, since user-space won't know about
+ * this timing tweak
+ * (we need to do it to avoid a lockup, though) */
+
+ snd_azf3328_dbgtimer("delay was too low (%d)!\n", delay);
+ delay = 49; /* minimum time is 49 ticks */
+ }
+ snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay);
+ delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static int
+snd_azf3328_timer_stop(struct snd_timer *timer)
+{
+ struct snd_azf3328 *chip;
+ unsigned long flags;
+
+ snd_azf3328_dbgcallenter();
+ chip = snd_timer_chip(timer);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ /* disable timer countdown and interrupt */
+ /* FIXME: should we write TIMER_IRQ_ACK here? */
+ snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+
+static int
+snd_azf3328_timer_precise_resolution(struct snd_timer *timer,
+ unsigned long *num, unsigned long *den)
+{
+ snd_azf3328_dbgcallenter();
+ *num = 1;
+ *den = 1024000 / seqtimer_scaling;
+ snd_azf3328_dbgcallleave();
+ return 0;
+}
+
+static struct snd_timer_hardware snd_azf3328_timer_hw = {
+ .flags = SNDRV_TIMER_HW_AUTO,
+ .resolution = 977, /* 1000000/1024000 = 0.9765625us */
+ .ticks = 1024000, /* max tick count, defined by the value register; actually it's not 1024000, but 1048576, but we don't care */
+ .start = snd_azf3328_timer_start,
+ .stop = snd_azf3328_timer_stop,
+ .precise_resolution = snd_azf3328_timer_precise_resolution,
+};
+
+static int __devinit
+snd_azf3328_timer(struct snd_azf3328 *chip, int device)
+{
+ struct snd_timer *timer = NULL;
+ struct snd_timer_id tid;
+ int err;
+
+ snd_azf3328_dbgcallenter();
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = chip->card->number;
+ tid.device = device;
+ tid.subdevice = 0;
+
+ snd_azf3328_timer_hw.resolution *= seqtimer_scaling;
+ snd_azf3328_timer_hw.ticks /= seqtimer_scaling;
+
+ err = snd_timer_new(chip->card, "AZF3328", &tid, &timer);
+ if (err < 0)
+ goto out;
+
+ strcpy(timer->name, "AZF3328 timer");
+ timer->private_data = chip;
+ timer->hw = snd_azf3328_timer_hw;
+
+ chip->timer = timer;
+
+ snd_azf3328_timer_stop(timer);
+
+ err = 0;
+
+out:
+ snd_azf3328_dbgcallleave();
+ return err;
+}
+
+/******************************************************************/
+
+static int
+snd_azf3328_free(struct snd_azf3328 *chip)
+{
+ if (chip->irq < 0)
+ goto __end_hw;
+
+ /* reset (close) mixer:
+ * first mute master volume, then reset
+ */
+ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1);
+ snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000);
+
+ snd_azf3328_timer_stop(chip->timer);
+ snd_azf3328_gameport_free(chip);
+
+ if (chip->irq >= 0)
+ synchronize_irq(chip->irq);
+__end_hw:
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+
+ kfree(chip);
+ return 0;
+}
+
+static int
+snd_azf3328_dev_free(struct snd_device *device)
+{
+ struct snd_azf3328 *chip = device->device_data;
+ return snd_azf3328_free(chip);
+}
+
+#if 0
+/* check whether a bit can be modified */
+static void
+snd_azf3328_test_bit(unsigned unsigned reg, int bit)
+{
+ unsigned char val, valoff, valon;
+
+ val = inb(reg);
+
+ outb(val & ~(1 << bit), reg);
+ valoff = inb(reg);
+
+ outb(val|(1 << bit), reg);
+ valon = inb(reg);
+
+ outb(val, reg);
+
+ printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n",
+ reg, bit, val, valoff, valon
+ );
+}
+#endif
+
+static inline void
+snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip)
+{
+#if DEBUG_MISC
+ u16 tmp;
+
+ snd_azf3328_dbgmisc(
+ "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, "
+ "opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n",
+ chip->codec_io, chip->game_io, chip->mpu_io,
+ chip->opl3_io, chip->mixer_io, chip->irq
+ );
+
+ snd_azf3328_dbgmisc("game %02x %02x %02x %02x %02x %02x\n",
+ snd_azf3328_game_inb(chip, 0),
+ snd_azf3328_game_inb(chip, 1),
+ snd_azf3328_game_inb(chip, 2),
+ snd_azf3328_game_inb(chip, 3),
+ snd_azf3328_game_inb(chip, 4),
+ snd_azf3328_game_inb(chip, 5)
+ );
+
+ for (tmp = 0; tmp < 0x07; tmp += 1)
+ snd_azf3328_dbgmisc("mpu_io 0x%04x\n", inb(chip->mpu_io + tmp));
+
+ for (tmp = 0; tmp <= 0x07; tmp += 1)
+ snd_azf3328_dbgmisc("0x%02x: game200 0x%04x, game208 0x%04x\n",
+ tmp, inb(0x200 + tmp), inb(0x208 + tmp));
+
+ for (tmp = 0; tmp <= 0x01; tmp += 1)
+ snd_azf3328_dbgmisc(
+ "0x%02x: mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, "
+ "mpu330 0x%04x opl388 0x%04x opl38c 0x%04x\n",
+ tmp,
+ inb(0x300 + tmp),
+ inb(0x310 + tmp),
+ inb(0x320 + tmp),
+ inb(0x330 + tmp),
+ inb(0x388 + tmp),
+ inb(0x38c + tmp)
+ );
+
+ for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2)
+ snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n",
+ tmp, snd_azf3328_codec_inw(chip, tmp)
+ );
+
+ for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2)
+ snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n",
+ tmp, snd_azf3328_mixer_inw(chip, tmp)
+ );
+#endif /* DEBUG_MISC */
+}
+
+static int __devinit
+snd_azf3328_create(struct snd_card *card,
+ struct pci_dev *pci,
+ unsigned long device_type,
+ struct snd_azf3328 **rchip)
+{
+ struct snd_azf3328 *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_azf3328_dev_free,
+ };
+ u16 tmp;
+
+ *rchip = NULL;
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ err = -ENOMEM;
+ goto out_err;
+ }
+ spin_lock_init(&chip->reg_lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* check if we can restrict PCI DMA transfers to 24 bits */
+ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support "
+ "24bit PCI busmaster DMA\n"
+ );
+ err = -ENXIO;
+ goto out_err;
+ }
+
+ err = pci_request_regions(pci, "Aztech AZF3328");
+ if (err < 0)
+ goto out_err;
+
+ chip->codec_io = pci_resource_start(pci, 0);
+ chip->game_io = pci_resource_start(pci, 1);
+ chip->mpu_io = pci_resource_start(pci, 2);
+ chip->opl3_io = pci_resource_start(pci, 3);
+ chip->mixer_io = pci_resource_start(pci, 4);
+
+ chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00;
+ chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20;
+
+ if (request_irq(pci->irq, snd_azf3328_interrupt,
+ IRQF_SHARED, card->shortname, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ err = -EBUSY;
+ goto out_err;
+ }
+ chip->irq = pci->irq;
+ pci_set_master(pci);
+ synchronize_irq(chip->irq);
+
+ snd_azf3328_debug_show_ports(chip);
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0)
+ goto out_err;
+
+ /* create mixer interface & switches */
+ err = snd_azf3328_mixer_new(chip);
+ if (err < 0)
+ goto out_err;
+
+ /* shutdown codecs to save power */
+ /* have snd_azf3328_codec_activity() act properly */
+ chip->audio_stream[AZF_PLAYBACK].running = 1;
+ snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0);
+
+ /* standard chip init stuff */
+ /* default IRQ init value */
+ tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE;
+
+ spin_lock_irq(&chip->reg_lock);
+ snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp);
+ snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp);
+ snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp);
+ spin_unlock_irq(&chip->reg_lock);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+
+ err = 0;
+ goto out;
+
+out_err:
+ if (chip)
+ snd_azf3328_free(chip);
+ pci_disable_device(pci);
+
+out:
+ return err;
+}
+
+static int __devinit
+snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_azf3328 *chip;
+ struct snd_opl3 *opl3;
+ int err;
+
+ snd_azf3328_dbgcallenter();
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "AZF3328");
+ strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
+
+ err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip);
+ if (err < 0)
+ goto out_err;
+
+ card->private_data = chip;
+
+ err = snd_mpu401_uart_new(
+ card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED,
+ pci->irq, 0, &chip->rmidi
+ );
+ if (err < 0) {
+ snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n",
+ chip->mpu_io
+ );
+ goto out_err;
+ }
+
+ err = snd_azf3328_timer(chip, 0);
+ if (err < 0)
+ goto out_err;
+
+ err = snd_azf3328_pcm(chip, 0);
+ if (err < 0)
+ goto out_err;
+
+ if (snd_opl3_create(card, chip->opl3_io, chip->opl3_io+2,
+ OPL3_HW_AUTO, 1, &opl3) < 0) {
+ snd_printk(KERN_ERR "azf3328: no OPL3 device at 0x%lx-0x%lx?\n",
+ chip->opl3_io, chip->opl3_io+2
+ );
+ } else {
+ /* need to use IDs 1, 2 since ID 0 is snd_azf3328_timer above */
+ err = snd_opl3_timer_new(opl3, 1, 2);
+ if (err < 0)
+ goto out_err;
+ err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+ if (err < 0)
+ goto out_err;
+ }
+
+ opl3->private_data = chip;
+
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, chip->codec_io, chip->irq);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto out_err;
+
+#ifdef MODULE
+ printk(
+"azt3328: Sound driver for Aztech AZF3328-based soundcards such as PCI168.\n"
+"azt3328: Hardware was completely undocumented, unfortunately.\n"
+"azt3328: Feel free to contact andi AT lisas.de for bug reports etc.!\n"
+"azt3328: User-scalable sequencer timer set to %dHz (1024000Hz / %d).\n",
+ 1024000 / seqtimer_scaling, seqtimer_scaling);
+#endif
+
+ snd_azf3328_gameport(chip, dev);
+
+ pci_set_drvdata(pci, card);
+ dev++;
+
+ err = 0;
+ goto out;
+
+out_err:
+ snd_printk(KERN_ERR "azf3328: something failed, exiting\n");
+ snd_card_free(card);
+
+out:
+ snd_azf3328_dbgcallleave();
+ return err;
+}
+
+static void __devexit
+snd_azf3328_remove(struct pci_dev *pci)
+{
+ snd_azf3328_dbgcallenter();
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+ snd_azf3328_dbgcallleave();
+}
+
+#ifdef CONFIG_PM
+static int
+snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_azf3328 *chip = card->private_data;
+ unsigned reg;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ snd_pcm_suspend_all(chip->pcm);
+
+ for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg)
+ chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2);
+
+ /* make sure to disable master volume etc. to prevent looping sound */
+ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1);
+ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+
+ for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg)
+ chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2);
+
+ /* manually store the one currently relevant write-only reg, too */
+ chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH;
+
+ for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg)
+ chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2);
+ for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg)
+ chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2);
+ for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg)
+ chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int
+snd_azf3328_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_azf3328 *chip = card->private_data;
+ unsigned reg;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "azt3328: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg)
+ outw(chip->saved_regs_game[reg], chip->game_io + reg * 2);
+ for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg)
+ outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2);
+ for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg)
+ outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2);
+ for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg)
+ outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2);
+ for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg)
+ outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+static struct pci_driver driver = {
+ .name = "AZF3328",
+ .id_table = snd_azf3328_ids,
+ .probe = snd_azf3328_probe,
+ .remove = __devexit_p(snd_azf3328_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_azf3328_suspend,
+ .resume = snd_azf3328_resume,
+#endif
+};
+
+static int __init
+alsa_card_azf3328_init(void)
+{
+ int err;
+ snd_azf3328_dbgcallenter();
+ err = pci_register_driver(&driver);
+ snd_azf3328_dbgcallleave();
+ return err;
+}
+
+static void __exit
+alsa_card_azf3328_exit(void)
+{
+ snd_azf3328_dbgcallenter();
+ pci_unregister_driver(&driver);
+ snd_azf3328_dbgcallleave();
+}
+
+module_init(alsa_card_azf3328_init)
+module_exit(alsa_card_azf3328_exit)
diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h
new file mode 100644
index 0000000..974e051
--- /dev/null
+++ b/sound/pci/azt3328.h
@@ -0,0 +1,343 @@
+#ifndef __SOUND_AZT3328_H
+#define __SOUND_AZT3328_H
+
+/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10
+ * "WRITE_ONLY" == register does not indicate actual bit values */
+
+/*** main I/O area port indices ***/
+/* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */
+#define AZF_IO_SIZE_CODEC 0x80
+#define AZF_IO_SIZE_CODEC_PM 0x70
+
+/* the driver initialisation suggests a layout of 4 main areas:
+ * from 0x00 (playback), from 0x20 (recording) and from 0x40 (maybe MPU401??).
+ * And another area from 0x60 to 0x6f (DirectX timer, IRQ management,
+ * power management etc.???). */
+
+/** playback area **/
+#define IDX_IO_PLAY_FLAGS 0x00 /* PU:0x0000 */
+ /* able to reactivate output after output muting due to 8/16bit
+ * output change, just like 0x0002.
+ * 0x0001 is the only bit that's able to start the DMA counter */
+ #define DMA_RESUME 0x0001 /* paused if cleared ? */
+ /* 0x0002 *temporarily* set during DMA stopping. hmm
+ * both 0x0002 and 0x0004 set in playback setup. */
+ /* able to reactivate output after output muting due to 8/16bit
+ * output change, just like 0x0001. */
+ #define DMA_PLAY_SOMETHING1 0x0002 /* \ alternated (toggled) */
+ /* 0x0004: NOT able to reactivate output */
+ #define DMA_PLAY_SOMETHING2 0x0004 /* / bits */
+ #define SOMETHING_ALMOST_ALWAYS_SET 0x0008 /* ???; can be modified */
+ #define DMA_EPILOGUE_SOMETHING 0x0010
+ #define DMA_SOMETHING_ELSE 0x0020 /* ??? */
+ #define SOMETHING_UNMODIFIABLE 0xffc0 /* unused ? not modifiable */
+#define IDX_IO_PLAY_IRQTYPE 0x02 /* PU:0x0001 */
+ /* write back to flags in case flags are set, in order to ACK IRQ in handler
+ * (bit 1 of port 0x64 indicates interrupt for one of these three types)
+ * sometimes in this case it just writes 0xffff to globally ACK all IRQs
+ * settings written are not reflected when reading back, though.
+ * seems to be IRQ, too (frequently used: port |= 0x07 !), but who knows ? */
+ #define IRQ_PLAY_SOMETHING 0x0001 /* something & ACK */
+ #define IRQ_FINISHED_PLAYBUF_1 0x0002 /* 1st dmabuf finished & ACK */
+ #define IRQ_FINISHED_PLAYBUF_2 0x0004 /* 2nd dmabuf finished & ACK */
+ #define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */
+ #define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */
+ #define IRQMASK_UNMODIFIABLE 0xffe0 /* unused ? not modifiable */
+#define IDX_IO_PLAY_DMA_START_1 0x04 /* start address of 1st DMA play area, PU:0x00000000 */
+#define IDX_IO_PLAY_DMA_START_2 0x08 /* start address of 2nd DMA play area, PU:0x00000000 */
+#define IDX_IO_PLAY_DMA_LEN_1 0x0c /* length of 1st DMA play area, PU:0x0000 */
+#define IDX_IO_PLAY_DMA_LEN_2 0x0e /* length of 2nd DMA play area, PU:0x0000 */
+#define IDX_IO_PLAY_DMA_CURRPOS 0x10 /* current DMA position, PU:0x00000000 */
+#define IDX_IO_PLAY_DMA_CURROFS 0x14 /* offset within current DMA play area, PU:0x0000 */
+#define IDX_IO_PLAY_SOUNDFORMAT 0x16 /* PU:0x0010 */
+ /* all unspecified bits can't be modified */
+ #define SOUNDFORMAT_FREQUENCY_MASK 0x000f
+ #define SOUNDFORMAT_XTAL1 0x00
+ #define SOUNDFORMAT_XTAL2 0x01
+ /* all _SUSPECTED_ values are not used by Windows drivers, so we don't
+ * have any hard facts, only rough measurements.
+ * All we know is that the crystal used on the board has 24.576MHz,
+ * like many soundcards (which results in the frequencies below when
+ * using certain divider values selected by the values below) */
+ #define SOUNDFORMAT_FREQ_SUSPECTED_4000 0x0c | SOUNDFORMAT_XTAL1
+ #define SOUNDFORMAT_FREQ_SUSPECTED_4800 0x0a | SOUNDFORMAT_XTAL1
+ #define SOUNDFORMAT_FREQ_5510 0x0c | SOUNDFORMAT_XTAL2
+ #define SOUNDFORMAT_FREQ_6620 0x0a | SOUNDFORMAT_XTAL2
+ #define SOUNDFORMAT_FREQ_8000 0x00 | SOUNDFORMAT_XTAL1 /* also 0x0e | SOUNDFORMAT_XTAL1? */
+ #define SOUNDFORMAT_FREQ_9600 0x08 | SOUNDFORMAT_XTAL1
+ #define SOUNDFORMAT_FREQ_11025 0x00 | SOUNDFORMAT_XTAL2 /* also 0x0e | SOUNDFORMAT_XTAL2? */
+ #define SOUNDFORMAT_FREQ_SUSPECTED_13240 0x08 | SOUNDFORMAT_XTAL2 /* seems to be 6620 *2 */
+ #define SOUNDFORMAT_FREQ_16000 0x02 | SOUNDFORMAT_XTAL1
+ #define SOUNDFORMAT_FREQ_22050 0x02 | SOUNDFORMAT_XTAL2
+ #define SOUNDFORMAT_FREQ_32000 0x04 | SOUNDFORMAT_XTAL1
+ #define SOUNDFORMAT_FREQ_44100 0x04 | SOUNDFORMAT_XTAL2
+ #define SOUNDFORMAT_FREQ_48000 0x06 | SOUNDFORMAT_XTAL1
+ #define SOUNDFORMAT_FREQ_SUSPECTED_66200 0x06 | SOUNDFORMAT_XTAL2 /* 66200 (13240 * 5); 64000 may have been nicer :-\ */
+ #define SOUNDFORMAT_FLAG_16BIT 0x0010
+ #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020
+
+/* define frequency helpers, for maximum value safety */
+enum azf_freq_t {
+#define AZF_FREQ(rate) AZF_FREQ_##rate = rate
+ AZF_FREQ(4000),
+ AZF_FREQ(4800),
+ AZF_FREQ(5512),
+ AZF_FREQ(6620),
+ AZF_FREQ(8000),
+ AZF_FREQ(9600),
+ AZF_FREQ(11025),
+ AZF_FREQ(13240),
+ AZF_FREQ(16000),
+ AZF_FREQ(22050),
+ AZF_FREQ(32000),
+ AZF_FREQ(44100),
+ AZF_FREQ(48000),
+ AZF_FREQ(66200),
+#undef AZF_FREQ
+};
+
+/** recording area (see also: playback bit flag definitions) **/
+#define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */
+#define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */
+ #define IRQ_REC_SOMETHING 0x0001 /* something & ACK */
+ #define IRQ_FINISHED_RECBUF_1 0x0002 /* 1st dmabuf finished & ACK */
+ #define IRQ_FINISHED_RECBUF_2 0x0004 /* 2nd dmabuf finished & ACK */
+ /* hmm, maybe these are just the corresponding *recording* flags ?
+ * but OTOH they are most likely at port 0x22 instead */
+ #define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */
+ #define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */
+#define IDX_IO_REC_DMA_START_1 0x24 /* PU:0x00000000 */
+#define IDX_IO_REC_DMA_START_2 0x28 /* PU:0x00000000 */
+#define IDX_IO_REC_DMA_LEN_1 0x2c /* PU:0x0000 */
+#define IDX_IO_REC_DMA_LEN_2 0x2e /* PU:0x0000 */
+#define IDX_IO_REC_DMA_CURRPOS 0x30 /* PU:0x00000000 */
+#define IDX_IO_REC_DMA_CURROFS 0x34 /* PU:0x00000000 */
+#define IDX_IO_REC_SOUNDFORMAT 0x36 /* PU:0x0000 */
+
+/** hmm, what is this I/O area for? MPU401?? or external DAC via I2S?? (after playback, recording, ???, timer) **/
+#define IDX_IO_SOMETHING_FLAGS 0x40 /* gets set to 0x34 just like port 0x0 and 0x20 on card init, PU:0x0000 */
+/* general */
+#define IDX_IO_42H 0x42 /* PU:0x0001 */
+
+/** DirectX timer, main interrupt area (FIXME: and something else?) **/
+#define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */
+ /* timer countdown value; triggers IRQ when timer is finished */
+ #define TIMER_VALUE_MASK 0x000fffffUL
+ /* activate timer countdown */
+ #define TIMER_COUNTDOWN_ENABLE 0x01000000UL
+ /* trigger timer IRQ on zero transition */
+ #define TIMER_IRQ_ENABLE 0x02000000UL
+ /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?)
+ * had 0x0020 set upon IRQ handler */
+ #define TIMER_IRQ_ACK 0x04000000UL
+#define IDX_IO_IRQSTATUS 0x64
+ /* some IRQ bit in here might also be used to signal a power-management timer
+ * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing).
+ * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which
+ * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */
+
+ #define IRQ_PLAYBACK 0x0001
+ #define IRQ_RECORDING 0x0002
+ #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */
+ #define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */
+ #define IRQ_MPU401 0x0010
+ #define IRQ_TIMER 0x0020 /* DirectX timer */
+ #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly I2S port? */
+ #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly I2S port? */
+#define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */
+ /* this is set to e.g. 0x3ff or 0x300, and writable;
+ * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */
+#define IDX_IO_SOME_VALUE 0x68
+ #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */
+ #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */
+ /* umm, nope, behaviour of these bits changes depending on what we wrote
+ * to 0x6b!!
+ * And they change upon playback/stop, too:
+ * Writing a value to 0x68 will display this exact value during playback,
+ * too but when stopped it can fall back to a rather different
+ * seemingly random value). Hmm, possibly this is a register which
+ * has a remote shadow which needs proper device supply which only exists
+ * in case playback is active? Or is this driver-induced?
+ */
+
+/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??);
+ * actually inhibits PCM playback!!! maybe power management??: */
+#define IDX_IO_6AH 0x6A /* WRITE_ONLY! */
+ /* bit 5: enabling this will activate permanent counting of bytes 2/3
+ * at gameport I/O (0xb402/3) (equal values each) and cause
+ * gameport legacy I/O at 0x0200 to be _DISABLED_!
+ * Is this Digital Enhanced Game Port Enable??? Or maybe it's Testmode
+ * for Enhanced Digital Gameport (see 4D Wave DX card): */
+ #define IO_6A_SOMETHING1_GAMEPORT 0x0020
+ /* bit 8; sure, this _pauses_ playback (later resumes at same spot!),
+ * but what the heck is this really about??: */
+ #define IO_6A_PAUSE_PLAYBACK_BIT8 0x0100
+ /* bit 9; sure, this _pauses_ playback (later resumes at same spot!),
+ * but what the heck is this really about??: */
+ #define IO_6A_PAUSE_PLAYBACK_BIT9 0x0200
+ /* BIT8 and BIT9 are _NOT_ able to affect OPL3 MIDI playback,
+ * thus it suggests influence on PCM only!!
+ * However OTOH there seems to be no bit anywhere around here
+ * which is able to disable OPL3... */
+ /* bit 10: enabling this actually changes values at legacy gameport
+ * I/O address (0x200); is this enabling of the Digital Enhanced Game Port???
+ * Or maybe this simply switches off the NE558 circuit, since enabling this
+ * still lets us evaluate button states, but not axis states */
+ #define IO_6A_SOMETHING2_GAMEPORT 0x0400
+ /* writing 0x0300: causes quite some crackling during
+ * PC activity such as switching windows (PCI traffic??
+ * --> FIFO/timing settings???) */
+ /* writing 0x0100 plus/or 0x0200 inhibits playback */
+ /* since the Windows .INF file has Flag_Enable_JoyStick and
+ * Flag_Enable_SB_DOS_Emulation directly together, it stands to reason
+ * that some other bit in this same register might be responsible
+ * for SB DOS Emulation activation (note that the file did NOT define
+ * a switch for OPL3!) */
+#define IDX_IO_6CH 0x6C /* unknown; fully read-writable */
+#define IDX_IO_6EH 0x6E
+ /* writing 0xffff returns 0x83fe (or 0x03fe only).
+ * writing 0x83 (and only 0x83!!) to 0x6f will cause 0x6c to switch
+ * from 0000 to ffff. */
+
+/* further I/O indices not saved/restored and not readable after writing,
+ * so probably not used */
+
+
+/*** Gameport area port indices ***/
+/* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */
+#define AZF_IO_SIZE_GAME 0x08
+#define AZF_IO_SIZE_GAME_PM 0x06
+
+enum {
+ AZF_GAME_LEGACY_IO_PORT = 0x200
+};
+
+#define IDX_GAME_LEGACY_COMPATIBLE 0x00
+ /* in some operation mode, writing anything to this port
+ * triggers an interrupt:
+ * yup, that's in case IDX_GAME_01H has one of the
+ * axis measurement bits enabled
+ * (and of course one needs to have GAME_HWCFG_IRQ_ENABLE, too) */
+
+#define IDX_GAME_AXES_CONFIG 0x01
+ /* NOTE: layout of this register awfully similar (read: "identical??")
+ * to AD1815JS.pdf (p.29) */
+
+ /* enables axis 1 (X axis) measurement: */
+ #define GAME_AXES_ENABLE_1 0x01
+ /* enables axis 2 (Y axis) measurement: */
+ #define GAME_AXES_ENABLE_2 0x02
+ /* enables axis 3 (X axis) measurement: */
+ #define GAME_AXES_ENABLE_3 0x04
+ /* enables axis 4 (Y axis) measurement: */
+ #define GAME_AXES_ENABLE_4 0x08
+ /* selects the current axis to read the measured value of
+ * (at IDX_GAME_AXIS_VALUE):
+ * 00 = axis 1, 01 = axis 2, 10 = axis 3, 11 = axis 4: */
+ #define GAME_AXES_READ_MASK 0x30
+ /* enable to have the latch continuously accept ADC values
+ * (and continuously cause interrupts in case interrupts are enabled);
+ * AD1815JS.pdf says it's ~16ms interval there: */
+ #define GAME_AXES_LATCH_ENABLE 0x40
+ /* joystick data (measured axes) ready for reading: */
+ #define GAME_AXES_SAMPLING_READY 0x80
+
+ /* NOTE: other card specs (SiS960 and others!) state that the
+ * game position latches should be frozen when reading and be freed
+ * (== reset?) after reading!!!
+ * Freezing most likely means disabling 0x40 (GAME_AXES_LATCH_ENABLE),
+ * but how to free the value? */
+ /* An internet search for "gameport latch ADC" should provide some insight
+ * into how to program such a gameport system. */
+
+ /* writing 0xf0 to 01H once reset both counters to 0, in some special mode!?
+ * yup, in case 6AH 0x20 is not enabled
+ * (and 0x40 is sufficient, 0xf0 is not needed) */
+
+#define IDX_GAME_AXIS_VALUE 0x02
+ /* R: value of currently configured axis (word value!);
+ * W: trigger axis measurement */
+
+#define IDX_GAME_HWCONFIG 0x04
+ /* note: bits 4 to 7 are never set (== 0) when reading!
+ * --> reserved bits? */
+ /* enables IRQ notification upon axes measurement ready: */
+ #define GAME_HWCFG_IRQ_ENABLE 0x01
+ /* these bits choose a different frequency for the
+ * internal ADC counter increment.
+ * hmm, seems to be a combo of bits:
+ * 00 --> standard frequency
+ * 10 --> 1/2
+ * 01 --> 1/20
+ * 11 --> 1/200: */
+ #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06
+
+ /* enable gameport legacy I/O address (0x200)
+ * I was unable to locate any configurability for a different address: */
+ #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08
+
+/*** MPU401 ***/
+#define AZF_IO_SIZE_MPU 0x04
+#define AZF_IO_SIZE_MPU_PM 0x04
+
+/*** OPL3 synth ***/
+#define AZF_IO_SIZE_OPL3 0x08
+#define AZF_IO_SIZE_OPL3_PM 0x06
+/* hmm, given that a standard OPL3 has 4 registers only,
+ * there might be some enhanced functionality lurking at the end
+ * (especially since register 0x04 has a "non-empty" value 0xfe) */
+
+/*** mixer I/O area port indices ***/
+/* (only 0x22 of 0x40 bytes saved/restored by Windows driver)
+ * UNFORTUNATELY azf3328 is NOT truly AC97 compliant: see main file intro */
+#define AZF_IO_SIZE_MIXER 0x40
+#define AZF_IO_SIZE_MIXER_PM 0x22
+
+ #define MIXER_VOLUME_RIGHT_MASK 0x001f
+ #define MIXER_VOLUME_LEFT_MASK 0x1f00
+ #define MIXER_MUTE_MASK 0x8000
+#define IDX_MIXER_RESET 0x00 /* does NOT seem to have AC97 ID bits */
+#define IDX_MIXER_PLAY_MASTER 0x02
+#define IDX_MIXER_MODEMOUT 0x04
+#define IDX_MIXER_BASSTREBLE 0x06
+ #define MIXER_BASSTREBLE_TREBLE_VOLUME_MASK 0x000e
+ #define MIXER_BASSTREBLE_BASS_VOLUME_MASK 0x0e00
+#define IDX_MIXER_PCBEEP 0x08
+#define IDX_MIXER_MODEMIN 0x0a
+#define IDX_MIXER_MIC 0x0c
+ #define MIXER_MIC_MICGAIN_20DB_ENHANCEMENT_MASK 0x0040
+#define IDX_MIXER_LINEIN 0x0e
+#define IDX_MIXER_CDAUDIO 0x10
+#define IDX_MIXER_VIDEO 0x12
+#define IDX_MIXER_AUX 0x14
+#define IDX_MIXER_WAVEOUT 0x16
+#define IDX_MIXER_FMSYNTH 0x18
+#define IDX_MIXER_REC_SELECT 0x1a
+ #define MIXER_REC_SELECT_MIC 0x00
+ #define MIXER_REC_SELECT_CD 0x01
+ #define MIXER_REC_SELECT_VIDEO 0x02
+ #define MIXER_REC_SELECT_AUX 0x03
+ #define MIXER_REC_SELECT_LINEIN 0x04
+ #define MIXER_REC_SELECT_MIXSTEREO 0x05
+ #define MIXER_REC_SELECT_MIXMONO 0x06
+ #define MIXER_REC_SELECT_MONOIN 0x07
+#define IDX_MIXER_REC_VOLUME 0x1c
+#define IDX_MIXER_ADVCTL1 0x1e
+ /* unlisted bits are unmodifiable */
+ #define MIXER_ADVCTL1_3DWIDTH_MASK 0x000e
+ #define MIXER_ADVCTL1_HIFI3D_MASK 0x0300 /* yup, this is missing the high bit that official AC97 contains, plus it doesn't have linear bit value range behaviour but instead acts weirdly (possibly we're dealing with two *different* 3D settings here??) */
+#define IDX_MIXER_ADVCTL2 0x20 /* subset of AC97_GENERAL_PURPOSE reg! */
+ /* unlisted bits are unmodifiable */
+ #define MIXER_ADVCTL2_LPBK 0x0080 /* Loopback mode -- Win driver: "WaveOut3DBypass"? mutes WaveOut at LineOut */
+ #define MIXER_ADVCTL2_MS 0x0100 /* Mic Select 0=Mic1, 1=Mic2 -- Win driver: "ModemOutSelect"?? */
+ #define MIXER_ADVCTL2_MIX 0x0200 /* Mono output select 0=Mix, 1=Mic; Win driver: "MonoSelectSource"?? */
+ #define MIXER_ADVCTL2_3D 0x2000 /* 3D Enhancement 1=on */
+ #define MIXER_ADVCTL2_POP 0x8000 /* Pcm Out Path, 0=pre 3D, 1=post 3D */
+
+#define IDX_MIXER_SOMETHING30H 0x30 /* used, but unknown??? */
+
+/* driver internal flags */
+#define SET_CHAN_LEFT 1
+#define SET_CHAN_RIGHT 2
+
+#endif /* __SOUND_AZT3328_H */
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
new file mode 100644
index 0000000..1aa1c04
--- /dev/null
+++ b/sound/pci/bt87x.c
@@ -0,0 +1,987 @@
+/*
+ * bt87x.c - Brooktree Bt878/Bt879 driver for ALSA
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ * based on btaudio.c by Gerd Knorr <kraxel@bytesex.org>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/bitops.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("Brooktree Bt87x audio driver");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878},"
+ "{Brooktree,Bt879}}");
+
+static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the first card */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int digital_rate[SNDRV_CARDS]; /* digital input rate */
+static int load_all; /* allow to load the non-whitelisted cards */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Bt87x soundcard");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Bt87x soundcard");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Bt87x soundcard");
+module_param_array(digital_rate, int, NULL, 0444);
+MODULE_PARM_DESC(digital_rate, "Digital input rate for Bt87x soundcard");
+module_param(load_all, bool, 0444);
+MODULE_PARM_DESC(load_all, "Allow to load the non-whitelisted cards");
+
+
+/* register offsets */
+#define REG_INT_STAT 0x100 /* interrupt status */
+#define REG_INT_MASK 0x104 /* interrupt mask */
+#define REG_GPIO_DMA_CTL 0x10c /* audio control */
+#define REG_PACKET_LEN 0x110 /* audio packet lengths */
+#define REG_RISC_STRT_ADD 0x114 /* RISC program start address */
+#define REG_RISC_COUNT 0x120 /* RISC program counter */
+
+/* interrupt bits */
+#define INT_OFLOW (1 << 3) /* audio A/D overflow */
+#define INT_RISCI (1 << 11) /* RISC instruction IRQ bit set */
+#define INT_FBUS (1 << 12) /* FIFO overrun due to bus access latency */
+#define INT_FTRGT (1 << 13) /* FIFO overrun due to target latency */
+#define INT_FDSR (1 << 14) /* FIFO data stream resynchronization */
+#define INT_PPERR (1 << 15) /* PCI parity error */
+#define INT_RIPERR (1 << 16) /* RISC instruction parity error */
+#define INT_PABORT (1 << 17) /* PCI master or target abort */
+#define INT_OCERR (1 << 18) /* invalid opcode */
+#define INT_SCERR (1 << 19) /* sync counter overflow */
+#define INT_RISC_EN (1 << 27) /* DMA controller running */
+#define INT_RISCS_SHIFT 28 /* RISC status bits */
+
+/* audio control bits */
+#define CTL_FIFO_ENABLE (1 << 0) /* enable audio data FIFO */
+#define CTL_RISC_ENABLE (1 << 1) /* enable audio DMA controller */
+#define CTL_PKTP_4 (0 << 2) /* packet mode FIFO trigger point - 4 DWORDs */
+#define CTL_PKTP_8 (1 << 2) /* 8 DWORDs */
+#define CTL_PKTP_16 (2 << 2) /* 16 DWORDs */
+#define CTL_ACAP_EN (1 << 4) /* enable audio capture */
+#define CTL_DA_APP (1 << 5) /* GPIO input */
+#define CTL_DA_IOM_AFE (0 << 6) /* audio A/D input */
+#define CTL_DA_IOM_DA (1 << 6) /* digital audio input */
+#define CTL_DA_SDR_SHIFT 8 /* DDF first stage decimation rate */
+#define CTL_DA_SDR_MASK (0xf<< 8)
+#define CTL_DA_LMT (1 << 12) /* limit audio data values */
+#define CTL_DA_ES2 (1 << 13) /* enable DDF stage 2 */
+#define CTL_DA_SBR (1 << 14) /* samples rounded to 8 bits */
+#define CTL_DA_DPM (1 << 15) /* data packet mode */
+#define CTL_DA_LRD_SHIFT 16 /* ALRCK delay */
+#define CTL_DA_MLB (1 << 21) /* MSB/LSB format */
+#define CTL_DA_LRI (1 << 22) /* left/right indication */
+#define CTL_DA_SCE (1 << 23) /* sample clock edge */
+#define CTL_A_SEL_STV (0 << 24) /* TV tuner audio input */
+#define CTL_A_SEL_SFM (1 << 24) /* FM audio input */
+#define CTL_A_SEL_SML (2 << 24) /* mic/line audio input */
+#define CTL_A_SEL_SMXC (3 << 24) /* MUX bypass */
+#define CTL_A_SEL_SHIFT 24
+#define CTL_A_SEL_MASK (3 << 24)
+#define CTL_A_PWRDN (1 << 26) /* analog audio power-down */
+#define CTL_A_G2X (1 << 27) /* audio gain boost */
+#define CTL_A_GAIN_SHIFT 28 /* audio input gain */
+#define CTL_A_GAIN_MASK (0xf<<28)
+
+/* RISC instruction opcodes */
+#define RISC_WRITE (0x1 << 28) /* write FIFO data to memory at address */
+#define RISC_WRITEC (0x5 << 28) /* write FIFO data to memory at current address */
+#define RISC_SKIP (0x2 << 28) /* skip FIFO data */
+#define RISC_JUMP (0x7 << 28) /* jump to address */
+#define RISC_SYNC (0x8 << 28) /* synchronize with FIFO */
+
+/* RISC instruction bits */
+#define RISC_BYTES_ENABLE (0xf << 12) /* byte enable bits */
+#define RISC_RESYNC ( 1 << 15) /* disable FDSR errors */
+#define RISC_SET_STATUS_SHIFT 16 /* set status bits */
+#define RISC_RESET_STATUS_SHIFT 20 /* clear status bits */
+#define RISC_IRQ ( 1 << 24) /* interrupt */
+#define RISC_EOL ( 1 << 26) /* end of line */
+#define RISC_SOL ( 1 << 27) /* start of line */
+
+/* SYNC status bits values */
+#define RISC_SYNC_FM1 0x6
+#define RISC_SYNC_VRO 0xc
+
+#define ANALOG_CLOCK 1792000
+#ifdef CONFIG_SND_BT87X_OVERCLOCK
+#define CLOCK_DIV_MIN 1
+#else
+#define CLOCK_DIV_MIN 4
+#endif
+#define CLOCK_DIV_MAX 15
+
+#define ERROR_INTERRUPTS (INT_FBUS | INT_FTRGT | INT_PPERR | \
+ INT_RIPERR | INT_PABORT | INT_OCERR)
+#define MY_INTERRUPTS (INT_RISCI | ERROR_INTERRUPTS)
+
+/* SYNC, one WRITE per line, one extra WRITE per page boundary, SYNC, JUMP */
+#define MAX_RISC_SIZE ((1 + 255 + (PAGE_ALIGN(255 * 4092) / PAGE_SIZE - 1) + 1 + 1) * 8)
+
+/* Cards with configuration information */
+enum snd_bt87x_boardid {
+ SND_BT87X_BOARD_UNKNOWN,
+ SND_BT87X_BOARD_GENERIC, /* both an & dig interfaces, 32kHz */
+ SND_BT87X_BOARD_ANALOG, /* board with no external A/D */
+ SND_BT87X_BOARD_OSPREY2x0,
+ SND_BT87X_BOARD_OSPREY440,
+ SND_BT87X_BOARD_AVPHONE98,
+};
+
+/* Card configuration */
+struct snd_bt87x_board {
+ int dig_rate; /* Digital input sampling rate */
+ u32 digital_fmt; /* Register settings for digital input */
+ unsigned no_analog:1; /* No analog input */
+ unsigned no_digital:1; /* No digital input */
+};
+
+static __devinitdata struct snd_bt87x_board snd_bt87x_boards[] = {
+ [SND_BT87X_BOARD_UNKNOWN] = {
+ .dig_rate = 32000, /* just a guess */
+ },
+ [SND_BT87X_BOARD_GENERIC] = {
+ .dig_rate = 32000,
+ },
+ [SND_BT87X_BOARD_ANALOG] = {
+ .no_digital = 1,
+ },
+ [SND_BT87X_BOARD_OSPREY2x0] = {
+ .dig_rate = 44100,
+ .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT),
+ },
+ [SND_BT87X_BOARD_OSPREY440] = {
+ .dig_rate = 32000,
+ .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT),
+ .no_analog = 1,
+ },
+ [SND_BT87X_BOARD_AVPHONE98] = {
+ .dig_rate = 48000,
+ },
+};
+
+struct snd_bt87x {
+ struct snd_card *card;
+ struct pci_dev *pci;
+ struct snd_bt87x_board board;
+
+ void __iomem *mmio;
+ int irq;
+
+ spinlock_t reg_lock;
+ unsigned long opened;
+ struct snd_pcm_substream *substream;
+
+ struct snd_dma_buffer dma_risc;
+ unsigned int line_bytes;
+ unsigned int lines;
+
+ u32 reg_control;
+ u32 interrupt_mask;
+
+ int current_line;
+
+ int pci_parity_errors;
+};
+
+enum { DEVICE_DIGITAL, DEVICE_ANALOG };
+
+static inline u32 snd_bt87x_readl(struct snd_bt87x *chip, u32 reg)
+{
+ return readl(chip->mmio + reg);
+}
+
+static inline void snd_bt87x_writel(struct snd_bt87x *chip, u32 reg, u32 value)
+{
+ writel(value, chip->mmio + reg);
+}
+
+static int snd_bt87x_create_risc(struct snd_bt87x *chip, struct snd_pcm_substream *substream,
+ unsigned int periods, unsigned int period_bytes)
+{
+ unsigned int i, offset;
+ u32 *risc;
+
+ if (chip->dma_risc.area == NULL) {
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ PAGE_ALIGN(MAX_RISC_SIZE), &chip->dma_risc) < 0)
+ return -ENOMEM;
+ }
+ risc = (u32 *)chip->dma_risc.area;
+ offset = 0;
+ *risc++ = cpu_to_le32(RISC_SYNC | RISC_SYNC_FM1);
+ *risc++ = cpu_to_le32(0);
+ for (i = 0; i < periods; ++i) {
+ u32 rest;
+
+ rest = period_bytes;
+ do {
+ u32 cmd, len;
+ unsigned int addr;
+
+ len = PAGE_SIZE - (offset % PAGE_SIZE);
+ if (len > rest)
+ len = rest;
+ cmd = RISC_WRITE | len;
+ if (rest == period_bytes) {
+ u32 block = i * 16 / periods;
+ cmd |= RISC_SOL;
+ cmd |= block << RISC_SET_STATUS_SHIFT;
+ cmd |= (~block & 0xf) << RISC_RESET_STATUS_SHIFT;
+ }
+ if (len == rest)
+ cmd |= RISC_EOL | RISC_IRQ;
+ *risc++ = cpu_to_le32(cmd);
+ addr = snd_pcm_sgbuf_get_addr(substream, offset);
+ *risc++ = cpu_to_le32(addr);
+ offset += len;
+ rest -= len;
+ } while (rest > 0);
+ }
+ *risc++ = cpu_to_le32(RISC_SYNC | RISC_SYNC_VRO);
+ *risc++ = cpu_to_le32(0);
+ *risc++ = cpu_to_le32(RISC_JUMP);
+ *risc++ = cpu_to_le32(chip->dma_risc.addr);
+ chip->line_bytes = period_bytes;
+ chip->lines = periods;
+ return 0;
+}
+
+static void snd_bt87x_free_risc(struct snd_bt87x *chip)
+{
+ if (chip->dma_risc.area) {
+ snd_dma_free_pages(&chip->dma_risc);
+ chip->dma_risc.area = NULL;
+ }
+}
+
+static void snd_bt87x_pci_error(struct snd_bt87x *chip, unsigned int status)
+{
+ u16 pci_status;
+
+ pci_read_config_word(chip->pci, PCI_STATUS, &pci_status);
+ pci_status &= PCI_STATUS_PARITY | PCI_STATUS_SIG_TARGET_ABORT |
+ PCI_STATUS_REC_TARGET_ABORT | PCI_STATUS_REC_MASTER_ABORT |
+ PCI_STATUS_SIG_SYSTEM_ERROR | PCI_STATUS_DETECTED_PARITY;
+ pci_write_config_word(chip->pci, PCI_STATUS, pci_status);
+ if (pci_status != PCI_STATUS_DETECTED_PARITY)
+ snd_printk(KERN_ERR "Aieee - PCI error! status %#08x, PCI status %#04x\n",
+ status & ERROR_INTERRUPTS, pci_status);
+ else {
+ snd_printk(KERN_ERR "Aieee - PCI parity error detected!\n");
+ /* error 'handling' similar to aic7xxx_pci.c: */
+ chip->pci_parity_errors++;
+ if (chip->pci_parity_errors > 20) {
+ snd_printk(KERN_ERR "Too many PCI parity errors observed.\n");
+ snd_printk(KERN_ERR "Some device on this bus is generating bad parity.\n");
+ snd_printk(KERN_ERR "This is an error *observed by*, not *generated by*, this card.\n");
+ snd_printk(KERN_ERR "PCI parity error checking has been disabled.\n");
+ chip->interrupt_mask &= ~(INT_PPERR | INT_RIPERR);
+ snd_bt87x_writel(chip, REG_INT_MASK, chip->interrupt_mask);
+ }
+ }
+}
+
+static irqreturn_t snd_bt87x_interrupt(int irq, void *dev_id)
+{
+ struct snd_bt87x *chip = dev_id;
+ unsigned int status, irq_status;
+
+ status = snd_bt87x_readl(chip, REG_INT_STAT);
+ irq_status = status & chip->interrupt_mask;
+ if (!irq_status)
+ return IRQ_NONE;
+ snd_bt87x_writel(chip, REG_INT_STAT, irq_status);
+
+ if (irq_status & ERROR_INTERRUPTS) {
+ if (irq_status & (INT_FBUS | INT_FTRGT))
+ snd_printk(KERN_WARNING "FIFO overrun, status %#08x\n", status);
+ if (irq_status & INT_OCERR)
+ snd_printk(KERN_ERR "internal RISC error, status %#08x\n", status);
+ if (irq_status & (INT_PPERR | INT_RIPERR | INT_PABORT))
+ snd_bt87x_pci_error(chip, irq_status);
+ }
+ if ((irq_status & INT_RISCI) && (chip->reg_control & CTL_ACAP_EN)) {
+ int current_block, irq_block;
+
+ /* assume that exactly one line has been recorded */
+ chip->current_line = (chip->current_line + 1) % chip->lines;
+ /* but check if some interrupts have been skipped */
+ current_block = chip->current_line * 16 / chip->lines;
+ irq_block = status >> INT_RISCS_SHIFT;
+ if (current_block != irq_block)
+ chip->current_line = (irq_block * chip->lines + 15) / 16;
+
+ snd_pcm_period_elapsed(chip->substream);
+ }
+ return IRQ_HANDLED;
+}
+
+static struct snd_pcm_hardware snd_bt87x_digital_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = 0, /* set at runtime */
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 255 * 4092,
+ .period_bytes_min = 32,
+ .period_bytes_max = 4092,
+ .periods_min = 2,
+ .periods_max = 255,
+};
+
+static struct snd_pcm_hardware snd_bt87x_analog_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX,
+ .rate_max = ANALOG_CLOCK / CLOCK_DIV_MIN,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = 255 * 4092,
+ .period_bytes_min = 32,
+ .period_bytes_max = 4092,
+ .periods_min = 2,
+ .periods_max = 255,
+};
+
+static int snd_bt87x_set_digital_hw(struct snd_bt87x *chip, struct snd_pcm_runtime *runtime)
+{
+ chip->reg_control |= CTL_DA_IOM_DA | CTL_A_PWRDN;
+ runtime->hw = snd_bt87x_digital_hw;
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(chip->board.dig_rate);
+ runtime->hw.rate_min = chip->board.dig_rate;
+ runtime->hw.rate_max = chip->board.dig_rate;
+ return 0;
+}
+
+static int snd_bt87x_set_analog_hw(struct snd_bt87x *chip, struct snd_pcm_runtime *runtime)
+{
+ static struct snd_ratnum analog_clock = {
+ .num = ANALOG_CLOCK,
+ .den_min = CLOCK_DIV_MIN,
+ .den_max = CLOCK_DIV_MAX,
+ .den_step = 1
+ };
+ static struct snd_pcm_hw_constraint_ratnums constraint_rates = {
+ .nrats = 1,
+ .rats = &analog_clock
+ };
+
+ chip->reg_control &= ~(CTL_DA_IOM_DA | CTL_A_PWRDN);
+ runtime->hw = snd_bt87x_analog_hw;
+ return snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraint_rates);
+}
+
+static int snd_bt87x_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if (test_and_set_bit(0, &chip->opened))
+ return -EBUSY;
+
+ if (substream->pcm->device == DEVICE_DIGITAL)
+ err = snd_bt87x_set_digital_hw(chip, runtime);
+ else
+ err = snd_bt87x_set_analog_hw(chip, runtime);
+ if (err < 0)
+ goto _error;
+
+ err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0)
+ goto _error;
+
+ chip->substream = substream;
+ return 0;
+
+_error:
+ clear_bit(0, &chip->opened);
+ smp_mb__after_clear_bit();
+ return err;
+}
+
+static int snd_bt87x_close(struct snd_pcm_substream *substream)
+{
+ struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->reg_control |= CTL_A_PWRDN;
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ spin_unlock_irq(&chip->reg_lock);
+
+ chip->substream = NULL;
+ clear_bit(0, &chip->opened);
+ smp_mb__after_clear_bit();
+ return 0;
+}
+
+static int snd_bt87x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ return snd_bt87x_create_risc(chip, substream,
+ params_periods(hw_params),
+ params_period_bytes(hw_params));
+}
+
+static int snd_bt87x_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+
+ snd_bt87x_free_risc(chip);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_bt87x_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int decimation;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->reg_control &= ~(CTL_DA_SDR_MASK | CTL_DA_SBR);
+ decimation = (ANALOG_CLOCK + runtime->rate / 4) / runtime->rate;
+ chip->reg_control |= decimation << CTL_DA_SDR_SHIFT;
+ if (runtime->format == SNDRV_PCM_FORMAT_S8)
+ chip->reg_control |= CTL_DA_SBR;
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_bt87x_start(struct snd_bt87x *chip)
+{
+ spin_lock(&chip->reg_lock);
+ chip->current_line = 0;
+ chip->reg_control |= CTL_FIFO_ENABLE | CTL_RISC_ENABLE | CTL_ACAP_EN;
+ snd_bt87x_writel(chip, REG_RISC_STRT_ADD, chip->dma_risc.addr);
+ snd_bt87x_writel(chip, REG_PACKET_LEN,
+ chip->line_bytes | (chip->lines << 16));
+ snd_bt87x_writel(chip, REG_INT_MASK, chip->interrupt_mask);
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ spin_unlock(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_bt87x_stop(struct snd_bt87x *chip)
+{
+ spin_lock(&chip->reg_lock);
+ chip->reg_control &= ~(CTL_FIFO_ENABLE | CTL_RISC_ENABLE | CTL_ACAP_EN);
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ snd_bt87x_writel(chip, REG_INT_MASK, 0);
+ snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS);
+ spin_unlock(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_bt87x_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ return snd_bt87x_start(chip);
+ case SNDRV_PCM_TRIGGER_STOP:
+ return snd_bt87x_stop(chip);
+ default:
+ return -EINVAL;
+ }
+}
+
+static snd_pcm_uframes_t snd_bt87x_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return (snd_pcm_uframes_t)bytes_to_frames(runtime, chip->current_line * chip->line_bytes);
+}
+
+static struct snd_pcm_ops snd_bt87x_pcm_ops = {
+ .open = snd_bt87x_pcm_open,
+ .close = snd_bt87x_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_bt87x_hw_params,
+ .hw_free = snd_bt87x_hw_free,
+ .prepare = snd_bt87x_prepare,
+ .trigger = snd_bt87x_trigger,
+ .pointer = snd_bt87x_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static int snd_bt87x_capture_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *info)
+{
+ info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ info->count = 1;
+ info->value.integer.min = 0;
+ info->value.integer.max = 15;
+ return 0;
+}
+
+static int snd_bt87x_capture_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *value)
+{
+ struct snd_bt87x *chip = snd_kcontrol_chip(kcontrol);
+
+ value->value.integer.value[0] = (chip->reg_control & CTL_A_GAIN_MASK) >> CTL_A_GAIN_SHIFT;
+ return 0;
+}
+
+static int snd_bt87x_capture_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *value)
+{
+ struct snd_bt87x *chip = snd_kcontrol_chip(kcontrol);
+ u32 old_control;
+ int changed;
+
+ spin_lock_irq(&chip->reg_lock);
+ old_control = chip->reg_control;
+ chip->reg_control = (chip->reg_control & ~CTL_A_GAIN_MASK)
+ | (value->value.integer.value[0] << CTL_A_GAIN_SHIFT);
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ changed = old_control != chip->reg_control;
+ spin_unlock_irq(&chip->reg_lock);
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_bt87x_capture_volume = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Volume",
+ .info = snd_bt87x_capture_volume_info,
+ .get = snd_bt87x_capture_volume_get,
+ .put = snd_bt87x_capture_volume_put,
+};
+
+#define snd_bt87x_capture_boost_info snd_ctl_boolean_mono_info
+
+static int snd_bt87x_capture_boost_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *value)
+{
+ struct snd_bt87x *chip = snd_kcontrol_chip(kcontrol);
+
+ value->value.integer.value[0] = !! (chip->reg_control & CTL_A_G2X);
+ return 0;
+}
+
+static int snd_bt87x_capture_boost_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *value)
+{
+ struct snd_bt87x *chip = snd_kcontrol_chip(kcontrol);
+ u32 old_control;
+ int changed;
+
+ spin_lock_irq(&chip->reg_lock);
+ old_control = chip->reg_control;
+ chip->reg_control = (chip->reg_control & ~CTL_A_G2X)
+ | (value->value.integer.value[0] ? CTL_A_G2X : 0);
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ changed = chip->reg_control != old_control;
+ spin_unlock_irq(&chip->reg_lock);
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_bt87x_capture_boost = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Boost",
+ .info = snd_bt87x_capture_boost_info,
+ .get = snd_bt87x_capture_boost_get,
+ .put = snd_bt87x_capture_boost_put,
+};
+
+static int snd_bt87x_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *info)
+{
+ static char *texts[3] = {"TV Tuner", "FM", "Mic/Line"};
+
+ info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ info->count = 1;
+ info->value.enumerated.items = 3;
+ if (info->value.enumerated.item > 2)
+ info->value.enumerated.item = 2;
+ strcpy(info->value.enumerated.name, texts[info->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_bt87x_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *value)
+{
+ struct snd_bt87x *chip = snd_kcontrol_chip(kcontrol);
+
+ value->value.enumerated.item[0] = (chip->reg_control & CTL_A_SEL_MASK) >> CTL_A_SEL_SHIFT;
+ return 0;
+}
+
+static int snd_bt87x_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *value)
+{
+ struct snd_bt87x *chip = snd_kcontrol_chip(kcontrol);
+ u32 old_control;
+ int changed;
+
+ spin_lock_irq(&chip->reg_lock);
+ old_control = chip->reg_control;
+ chip->reg_control = (chip->reg_control & ~CTL_A_SEL_MASK)
+ | (value->value.enumerated.item[0] << CTL_A_SEL_SHIFT);
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ changed = chip->reg_control != old_control;
+ spin_unlock_irq(&chip->reg_lock);
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_bt87x_capture_source = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_bt87x_capture_source_info,
+ .get = snd_bt87x_capture_source_get,
+ .put = snd_bt87x_capture_source_put,
+};
+
+static int snd_bt87x_free(struct snd_bt87x *chip)
+{
+ if (chip->mmio)
+ snd_bt87x_stop(chip);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (chip->mmio)
+ iounmap(chip->mmio);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_bt87x_dev_free(struct snd_device *device)
+{
+ struct snd_bt87x *chip = device->device_data;
+ return snd_bt87x_free(chip);
+}
+
+static int __devinit snd_bt87x_pcm(struct snd_bt87x *chip, int device, char *name)
+{
+ int err;
+ struct snd_pcm *pcm;
+
+ err = snd_pcm_new(chip->card, name, device, 0, 1, &pcm);
+ if (err < 0)
+ return err;
+ pcm->private_data = chip;
+ strcpy(pcm->name, name);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_bt87x_pcm_ops);
+ return snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 128 * 1024,
+ ALIGN(255 * 4092, 1024));
+}
+
+static int __devinit snd_bt87x_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct snd_bt87x **rchip)
+{
+ struct snd_bt87x *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_bt87x_dev_free
+ };
+
+ *rchip = NULL;
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (!chip) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ spin_lock_init(&chip->reg_lock);
+
+ if ((err = pci_request_regions(pci, "Bt87x audio")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->mmio = pci_ioremap_bar(pci, 0);
+ if (!chip->mmio) {
+ snd_printk(KERN_ERR "cannot remap io memory\n");
+ err = -ENOMEM;
+ goto fail;
+ }
+
+ chip->reg_control = CTL_A_PWRDN | CTL_DA_ES2 |
+ CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT);
+ chip->interrupt_mask = MY_INTERRUPTS;
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ snd_bt87x_writel(chip, REG_INT_MASK, 0);
+ snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS);
+
+ err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
+ "Bt87x audio", chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
+ goto fail;
+ }
+ chip->irq = pci->irq;
+ pci_set_master(pci);
+ synchronize_irq(chip->irq);
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0)
+ goto fail;
+
+ snd_card_set_dev(card, &pci->dev);
+ *rchip = chip;
+ return 0;
+
+fail:
+ snd_bt87x_free(chip);
+ return err;
+}
+
+#define BT_DEVICE(chip, subvend, subdev, id) \
+ { .vendor = PCI_VENDOR_ID_BROOKTREE, \
+ .device = chip, \
+ .subvendor = subvend, .subdevice = subdev, \
+ .driver_data = SND_BT87X_BOARD_ ## id }
+/* driver_data is the card id for that device */
+
+static struct pci_device_id snd_bt87x_ids[] = {
+ /* Hauppauge WinTV series */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC),
+ /* Hauppauge WinTV series */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, GENERIC),
+ /* Viewcast Osprey 200 */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, OSPREY2x0),
+ /* Viewcast Osprey 440 (rate is configurable via gpio) */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, OSPREY440),
+ /* ATI TV-Wonder */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC),
+ /* Leadtek Winfast tv 2000xp delux */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
+ /* Voodoo TV 200 */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
+ /* AVerMedia Studio No. 103, 203, ...? */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, AVPHONE98),
+ /* Prolink PixelView PV-M4900 */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1554, 0x4011, GENERIC),
+ /* Pinnacle Studio PCTV rave */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0xbd11, 0x1200, GENERIC),
+ { }
+};
+MODULE_DEVICE_TABLE(pci, snd_bt87x_ids);
+
+/* cards known not to have audio
+ * (DVB cards use the audio function to transfer MPEG data) */
+static struct {
+ unsigned short subvendor, subdevice;
+} blacklist[] __devinitdata = {
+ {0x0071, 0x0101}, /* Nebula Electronics DigiTV */
+ {0x11bd, 0x001c}, /* Pinnacle PCTV Sat */
+ {0x11bd, 0x0026}, /* Pinnacle PCTV SAT CI */
+ {0x1461, 0x0761}, /* AVermedia AverTV DVB-T */
+ {0x1461, 0x0771}, /* AVermedia DVB-T 771 */
+ {0x1822, 0x0001}, /* Twinhan VisionPlus DVB-T */
+ {0x18ac, 0xd500}, /* DVICO FusionHDTV 5 Lite */
+ {0x18ac, 0xdb10}, /* DVICO FusionHDTV DVB-T Lite */
+ {0x18ac, 0xdb11}, /* Ultraview DVB-T Lite */
+ {0x270f, 0xfc00}, /* Chaintech Digitop DST-1000 DVB-S */
+ {0x7063, 0x2000}, /* pcHDTV HD-2000 TV */
+};
+
+static struct pci_driver driver;
+
+/* return the id of the card, or a negative value if it's blacklisted */
+static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
+{
+ int i;
+ const struct pci_device_id *supported;
+
+ supported = pci_match_id(snd_bt87x_ids, pci);
+ if (supported && supported->driver_data > 0)
+ return supported->driver_data;
+
+ for (i = 0; i < ARRAY_SIZE(blacklist); ++i)
+ if (blacklist[i].subvendor == pci->subsystem_vendor &&
+ blacklist[i].subdevice == pci->subsystem_device) {
+ snd_printdd(KERN_INFO "card %#04x-%#04x:%#04x has no audio\n",
+ pci->device, pci->subsystem_vendor, pci->subsystem_device);
+ return -EBUSY;
+ }
+
+ snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x\n",
+ pci->device, pci->subsystem_vendor, pci->subsystem_device);
+ snd_printk(KERN_DEBUG "please mail id, board name, and, "
+ "if it works, the correct digital_rate option to "
+ "<alsa-devel@alsa-project.org>\n");
+ return SND_BT87X_BOARD_UNKNOWN;
+}
+
+static int __devinit snd_bt87x_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_bt87x *chip;
+ int err;
+ enum snd_bt87x_boardid boardid;
+
+ if (!pci_id->driver_data) {
+ err = snd_bt87x_detect_card(pci);
+ if (err < 0)
+ return -ENODEV;
+ boardid = err;
+ } else
+ boardid = pci_id->driver_data;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ ++dev;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (!card)
+ return -ENOMEM;
+
+ err = snd_bt87x_create(card, pci, &chip);
+ if (err < 0)
+ goto _error;
+
+ memcpy(&chip->board, &snd_bt87x_boards[boardid], sizeof(chip->board));
+
+ if (!chip->board.no_digital) {
+ if (digital_rate[dev] > 0)
+ chip->board.dig_rate = digital_rate[dev];
+
+ chip->reg_control |= chip->board.digital_fmt;
+
+ err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital");
+ if (err < 0)
+ goto _error;
+ }
+ if (!chip->board.no_analog) {
+ err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog");
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_volume, chip));
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_boost, chip));
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_source, chip));
+ if (err < 0)
+ goto _error;
+ }
+ snd_printk(KERN_INFO "bt87x%d: Using board %d, %sanalog, %sdigital "
+ "(rate %d Hz)\n", dev, boardid,
+ chip->board.no_analog ? "no " : "",
+ chip->board.no_digital ? "no " : "", chip->board.dig_rate);
+
+ strcpy(card->driver, "Bt87x");
+ sprintf(card->shortname, "Brooktree Bt%x", pci->device);
+ sprintf(card->longname, "%s at %#llx, irq %i",
+ card->shortname, (unsigned long long)pci_resource_start(pci, 0),
+ chip->irq);
+ strcpy(card->mixername, "Bt87x");
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto _error;
+
+ pci_set_drvdata(pci, card);
+ ++dev;
+ return 0;
+
+_error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_bt87x_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+/* default entries for all Bt87x cards - it's not exported */
+/* driver_data is set to 0 to call detection */
+static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = {
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN),
+ { }
+};
+
+static struct pci_driver driver = {
+ .name = "Bt87x",
+ .id_table = snd_bt87x_ids,
+ .probe = snd_bt87x_probe,
+ .remove = __devexit_p(snd_bt87x_remove),
+};
+
+static int __init alsa_card_bt87x_init(void)
+{
+ if (load_all)
+ driver.id_table = snd_bt87x_default_ids;
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_bt87x_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_bt87x_init)
+module_exit(alsa_card_bt87x_exit)
diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile
new file mode 100644
index 0000000..dcbae7b
--- /dev/null
+++ b/sound/pci/ca0106/Makefile
@@ -0,0 +1,3 @@
+snd-ca0106-objs := ca0106_main.o ca0106_proc.o ca0106_mixer.o ca_midi.o
+
+obj-$(CONFIG_SND_CA0106) += snd-ca0106.o
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
new file mode 100644
index 0000000..74175fc
--- /dev/null
+++ b/sound/pci/ca0106/ca0106.h
@@ -0,0 +1,723 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.22
+ *
+ * FEATURES currently supported:
+ * See ca0106_main.c for features.
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Separated ca0106.c into separate functional .c files.
+ * 0.0.16
+ * Implement 192000 sample rate.
+ * 0.0.17
+ * Add support for SB0410 and SB0413.
+ * 0.0.18
+ * Modified Copyright message.
+ * 0.0.19
+ * Added I2C and SPI registers. Filled in interrupt enable.
+ * 0.0.20
+ * Added GPIO info for SB Live 24bit.
+ * 0.0.21
+ * Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.22
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ *
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/************************************************************************************************/
+/* PCI function 0 registers, address = <val> + PCIBASE0 */
+/************************************************************************************************/
+
+#define PTR 0x00 /* Indexed register set pointer register */
+ /* NOTE: The CHANNELNUM and ADDRESS words can */
+ /* be modified independently of each other. */
+ /* CNL[1:0], ADDR[27:16] */
+
+#define DATA 0x04 /* Indexed register set data register */
+ /* DATA[31:0] */
+
+#define IPR 0x08 /* Global interrupt pending register */
+ /* Clear pending interrupts by writing a 1 to */
+ /* the relevant bits and zero to the other bits */
+#define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */
+#define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */
+#define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */
+#define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */
+#define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */
+#define IPR_SPI 0x00000800 /* SPI transaction completed */
+#define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */
+#define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */
+#define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */
+#define IPR_GPI 0x00000080 /* General Purpose input changed */
+#define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */
+#define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */
+#define IPR_TIMER2 0x00000010 /* 192000Hz Timer */
+#define IPR_TIMER1 0x00000008 /* 44100Hz Timer */
+#define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */
+#define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */
+#define IPR_PCI 0x00000001 /* PCI Bus error */
+
+#define INTE 0x0c /* Interrupt enable register */
+
+#define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */
+#define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */
+#define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */
+#define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */
+#define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */
+#define INTE_SPI 0x00000800 /* SPI transaction completed */
+#define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */
+#define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */
+#define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */
+#define INTE_GPI 0x00000080 /* General Purpose input changed */
+#define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */
+#define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */
+#define INTE_TIMER2 0x00000010 /* 192000Hz Timer */
+#define INTE_TIMER1 0x00000008 /* 44100Hz Timer */
+#define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */
+#define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */
+#define INTE_PCI 0x00000001 /* PCI Bus error */
+
+#define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */
+#define HCFG 0x14 /* Hardware config register */
+ /* 0x1000 causes AC3 to fails. It adds a dither bit. */
+
+#define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */
+#define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */
+#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */
+#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */
+#define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */
+#define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */
+#define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */
+#define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */
+#define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */
+#define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/
+#define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/
+#define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */
+#define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */
+#define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */
+#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */
+ /* NOTE: This should generally never be used. */
+#define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */
+ /* NOTE: This should generally never be used. */
+#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */
+ /* Should be set to 1 when the EMU10K1 is */
+ /* completely initialized. */
+#define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */
+ /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */
+ /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */
+ /* SB Live 24bit:
+ * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in.
+ * bit 9 0 = Mute / 1 = Analog out.
+ * bit 10 0 = Line-in / 1 = Mic-in.
+ * bit 11 0 = ? / 1 = ?
+ * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit.
+ * bit 13 0 = ? / 1 = ?
+ * bit 14 0 = Mute / 1 = Analog out
+ * bit 15 0 = ? / 1 = ?
+ * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit.
+ */
+ /* 8 general purpose programmable In/Out pins.
+ * GPI [8:0] Read only. Default 0.
+ * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF)
+ * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin.
+ */
+#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
+
+#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
+
+/********************************************************************************************************/
+/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */
+/********************************************************************************************************/
+
+/* Initally all registers from 0x00 to 0x3f have zero contents. */
+#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
+ /* One list entry: 4 bytes for DMA address,
+ * 4 bytes for period_size << 16.
+ * One list entry is 8 bytes long.
+ * One list entry for each period in the buffer.
+ */
+ /* ADDR[31:0], Default: 0x0 */
+#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
+ /* SIZE[21:16], Default: 0x8 */
+#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
+ /* PTR[5:0], Default: 0x0 */
+#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */
+#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
+ /* DMA[31:0], Default: 0x0 */
+#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
+ /* SIZE[31:16], Default: 0x0 */
+#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
+ /* POINTER[15:0], Default: 0x0 */
+#define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */
+ /* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */
+#define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */
+ /* Cache size valid [5:0] */
+#define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */
+#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
+ /* DMA[31:0], Default: 0x0 */
+#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
+ /* SIZE[31:16], Default: 0x0 */
+#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
+ /* POINTER[15:0], Default: 0x0 */
+#define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */
+ /* Cache size valid [5:0] */
+#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */
+/* 0x21 - 0x3f unused */
+#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
+ /* Playback (0x1<<channel_id) */
+ /* Capture (0x100<<channel_id) */
+ /* Playback sample rate 96000 = 0x20000 */
+ /* Start Playback [3:0] (one bit per channel)
+ * Start Capture [11:8] (one bit per channel)
+ * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * Playback mixer in enable [27:24] (one bit per channel)
+ * Playback mixer out enable [31:28] (one bit per channel)
+ */
+/* The Digital out jack is shared with the Center/LFE Analogue output.
+ * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
+ * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
+ * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
+ * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red.
+ * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
+ */
+/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
+ * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS
+ * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM.
+ * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output
+ */
+/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
+ * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs.
+ */
+#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */
+#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */
+#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */
+#define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */
+ /* When Channel set to 0: */
+#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */
+#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */
+#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */
+#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */
+#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */
+#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */
+#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */
+#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */
+#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */
+#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */
+#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */
+#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */
+#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */
+#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */
+#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */
+#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */
+#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */
+#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */
+#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */
+#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */
+#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */
+#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */
+#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */
+
+ /* When Channel set to 1: */
+#define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */
+#define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */
+#define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */
+#define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */
+#define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */
+#define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */
+#define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */
+#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */
+#define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */
+#define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */
+#define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */
+#define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */
+#define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */
+#define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */
+
+#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
+ /* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
+ * But as the jack is shared, use 0xf00.
+ * The Windows2000 driver uses 0x0000000f for both digital and analog.
+ * 0xf00 introduces interesting noises onto the Center/LFE.
+ * If you turn the volume up, you hear computer noise,
+ * e.g. mouse moving, changing between app windows etc.
+ * So, I am going to set this to 0x0000000f all the time now,
+ * same as the windows driver does.
+ * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
+ */
+ /* When Channel = 0:
+ * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit)
+ * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate)
+ * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass)
+ */
+ /* When Channel = 1:
+ * SPDIF 0 User data [7:0]
+ * SPDIF 1 User data [15:8]
+ * SPDIF 0 User data [23:16]
+ * SPDIF 0 User data [31:24]
+ * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts.
+ */
+#define WATERMARK 0x46 /* Test bit to indicate cache usage level */
+#define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS.
+ * When Channel = 0: Bits the same as SPCS channel 0.
+ * When Channel = 1: Bits the same as SPCS channel 1.
+ * When Channel = 2:
+ * SPDIF Input User data [16:0]
+ * SPDIF Input Frame count [21:16]
+ */
+#define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */
+#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */
+#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */
+#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */
+#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */
+#define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */
+#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */
+ /* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3
+ * Record source select for channel 0 [18:16]
+ * Record source select for channel 1 [22:20]
+ * Record source select for channel 2 [26:24]
+ * Record source select for channel 3 [30:28]
+ * 0 - SPDIF mixer output.
+ * 1 - i2s mixer output.
+ * 2 - SPDIF input.
+ * 3 - i2s input.
+ * 4 - AC97 capture.
+ * 5 - SRC output.
+ */
+#define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */
+#define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */
+
+#define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */
+#define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
+#define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
+#define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
+#define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
+ /* Channel_id's handle stereo channels. Channel X is a single mono channel */
+ /* Host is input from the PCI bus. */
+ /* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
+ * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
+ */
+
+#define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */
+ /* SRC is input from the capture inputs. */
+ /* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
+ * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
+ */
+
+#define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */
+ /* SPDIF Mixer input control:
+ * Invert SRC to SPDIF Mixer [7-0] (One bit per channel)
+ * Invert Host to SPDIF Mixer [15:8] (One bit per channel)
+ * SRC to SPDIF Mixer disable [23:16] (One bit per channel)
+ * Host to SPDIF Mixer disable [31:24] (One bit per channel)
+ */
+#define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */
+ /* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
+ /* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */
+ /* One register for each of the 4 stereo streams. */
+ /* SRC Right volume [7:0]
+ * SRC Left volume [15:8]
+ * Host Right volume [23:16]
+ * Host Left volume [31:24]
+ */
+#define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */
+ /* Similar to register 0x63, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */
+ /* Similar to register 0x64, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
+ /* Similar to register 0x65, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */
+ /* Similar to register 0x66, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
+#define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
+#define MIDI_UART_A_DATA 0x6c /* Midi Uart A Data */
+#define MIDI_UART_A_CMD 0x6d /* Midi Uart A Command/Status */
+#define MIDI_UART_B_DATA 0x6e /* Midi Uart B Data (currently unused) */
+#define MIDI_UART_B_CMD 0x6f /* Midi Uart B Command/Status (currently unused) */
+
+/* unique channel identifier for midi->channel */
+
+#define CA0106_MIDI_CHAN_A 0x1
+#define CA0106_MIDI_CHAN_B 0x2
+
+/* from mpu401 */
+
+#define CA0106_MIDI_INPUT_AVAIL 0x80
+#define CA0106_MIDI_OUTPUT_READY 0x40
+#define CA0106_MPU401_RESET 0xff
+#define CA0106_MPU401_ENTER_UART 0x3f
+#define CA0106_MPU401_ACK 0xfe
+
+#define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */
+ /* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0
+ * Rate Locked [20]
+ * SPDIF Locked [21] For SPDIF channel only.
+ * Valid Audio [22] For SPDIF channel only.
+ */
+#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
+ /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
+ /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
+ /* Sample rate output control register Channel=0
+ * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
+ * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source.
+ * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
+ * Record mixer output enable [12:10]
+ * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * I2S output source select [18] (0=Audio from host, 1=Audio from SRC)
+ * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0)
+ * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.)
+ * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.)
+ * I2S input mode [23] (0=Slave, 1=Master)
+ * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+ * SPDIF output source select [26] (0=host, 1=SRC)
+ * Not used [27]
+ * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
+ * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
+ */
+ /* Sample rate output control register Channel=1
+ * I2S Input 0 volume Right [7:0]
+ * I2S Input 0 volume Left [15:8]
+ * I2S Input 1 volume Right [23:16]
+ * I2S Input 1 volume Left [31:24]
+ */
+ /* Sample rate output control register Channel=2
+ * SPDIF Input volume Right [23:16]
+ * SPDIF Input volume Left [31:24]
+ */
+ /* Sample rate output control register Channel=3
+ * No used
+ */
+#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
+#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */
+#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */
+#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */
+ /* Audio output control
+ * AC97 output enable [5:0]
+ * I2S output enable [19:16]
+ * SPDIF output enable [27:24]
+ */
+#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */
+#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */
+#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */
+ /* Sets which Interrupts are enabled. */
+ /* 0x00000001 = Half period. Playback.
+ * 0x00000010 = Full period. Playback.
+ * 0x00000100 = Half buffer. Playback.
+ * 0x00001000 = Full buffer. Playback.
+ * 0x00010000 = Half buffer. Capture.
+ * 0x00100000 = Full buffer. Capture.
+ * Capture can only do 2 periods.
+ * 0x01000000 = End audio. Playback.
+ * 0x40000000 = Half buffer Playback,Caputre xrun.
+ * 0x80000000 = Full buffer Playback,Caputre xrun.
+ */
+#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */
+ /* Shows which interrupts are active at the moment. */
+ /* Same bit layout as EXTENDED_INT_MASK */
+#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */
+#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */
+#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */
+ /* Causes interrupts based on timer intervals. */
+#define SPI 0x7a /* SPI: Serial Interface Register */
+#define I2C_A 0x7b /* I2C Address. 32 bit */
+#define I2C_D0 0x7c /* I2C Data Port 0. 32 bit */
+#define I2C_D1 0x7d /* I2C Data Port 1. 32 bit */
+//I2C values
+#define I2C_A_ADC_ADD_MASK 0x000000fe //The address is a 7 bit address
+#define I2C_A_ADC_RW_MASK 0x00000001 //bit mask for R/W
+#define I2C_A_ADC_TRANS_MASK 0x00000010 //Bit mask for I2c address DAC value
+#define I2C_A_ADC_ABORT_MASK 0x00000020 //Bit mask for I2C transaction abort flag
+#define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction
+#define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode
+
+#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC
+#define I2C_A_ADC_READ 0x00000001 //To perform a read operation
+#define I2C_A_ADC_START 0x00000100 //Start I2C transaction
+#define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort
+#define I2C_A_ADC_LAST 0x00000400 //I2C last transaction
+#define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode
+
+#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register
+#define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register
+
+#define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable
+#define ADC_IFC_CTRL 0x0000000b //ADC Interface Control
+#define ADC_MASTER 0x0000000c //ADC Master Mode Control
+#define ADC_POWER 0x0000000d //ADC PowerDown Control
+#define ADC_ATTEN_ADCL 0x0000000e //ADC Attenuation ADCL
+#define ADC_ATTEN_ADCR 0x0000000f //ADC Attenuation ADCR
+#define ADC_ALC_CTRL1 0x00000010 //ADC ALC Control 1
+#define ADC_ALC_CTRL2 0x00000011 //ADC ALC Control 2
+#define ADC_ALC_CTRL3 0x00000012 //ADC ALC Control 3
+#define ADC_NOISE_CTRL 0x00000013 //ADC Noise Gate Control
+#define ADC_LIMIT_CTRL 0x00000014 //ADC Limiter Control
+#define ADC_MUX 0x00000015 //ADC Mux offset
+
+#if 0
+/* FIXME: Not tested yet. */
+#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain
+#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB
+#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute
+#define ADC_MUTE 0x000000c0 //Value to mute ADC
+#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select
+#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock
+#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter
+#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window
+#endif
+
+#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux
+#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used)
+#define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux
+#define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux
+#define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux
+
+#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
+#define PCM_FRONT_CHANNEL 0
+#define PCM_REAR_CHANNEL 1
+#define PCM_CENTER_LFE_CHANNEL 2
+#define PCM_UNKNOWN_CHANNEL 3
+#define CONTROL_FRONT_CHANNEL 0
+#define CONTROL_REAR_CHANNEL 3
+#define CONTROL_CENTER_LFE_CHANNEL 1
+#define CONTROL_UNKNOWN_CHANNEL 2
+
+
+/* Based on WM8768 Datasheet Rev 4.2 page 32 */
+#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */
+#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */
+
+#define SPI_LDA1_REG 0 /* digital attenuation */
+#define SPI_RDA1_REG 1
+#define SPI_LDA2_REG 4
+#define SPI_RDA2_REG 5
+#define SPI_LDA3_REG 6
+#define SPI_RDA3_REG 7
+#define SPI_LDA4_REG 13
+#define SPI_RDA4_REG 14
+#define SPI_MASTDA_REG 8
+
+#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */
+#define SPI_DA_BIT_0dB 0xff /* 0 dB */
+#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */
+
+#define SPI_PL_REG 2
+#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */
+#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */
+#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */
+#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */
+#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */
+#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */
+#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */
+#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */
+#define SPI_IZD_REG 2
+#define SPI_IZD_BIT (1<<4) /* infinite zero detect */
+
+#define SPI_FMT_REG 3
+#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */
+#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */
+#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */
+#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */
+#define SPI_LRP_REG 3
+#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */
+#define SPI_BCP_REG 3
+#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */
+#define SPI_IWL_REG 3
+#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */
+#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */
+#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */
+#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */
+
+#define SPI_MS_REG 10
+#define SPI_MS_BIT (1<<5) /* master mode */
+#define SPI_RATE_REG 10 /* only applies in master mode */
+#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */
+#define SPI_RATE_BIT_192 (1<<6)
+#define SPI_RATE_BIT_256 (2<<6)
+#define SPI_RATE_BIT_384 (3<<6)
+#define SPI_RATE_BIT_512 (4<<6)
+#define SPI_RATE_BIT_768 (5<<6)
+
+/* They really do label the bit for the 4th channel "4" and not "3" */
+#define SPI_DMUTE0_REG 9
+#define SPI_DMUTE1_REG 9
+#define SPI_DMUTE2_REG 9
+#define SPI_DMUTE4_REG 15
+#define SPI_DMUTE0_BIT (1<<3)
+#define SPI_DMUTE1_BIT (1<<4)
+#define SPI_DMUTE2_BIT (1<<5)
+#define SPI_DMUTE4_BIT (1<<2)
+
+#define SPI_PHASE0_REG 3
+#define SPI_PHASE1_REG 3
+#define SPI_PHASE2_REG 3
+#define SPI_PHASE4_REG 15
+#define SPI_PHASE0_BIT (1<<6)
+#define SPI_PHASE1_BIT (1<<7)
+#define SPI_PHASE2_BIT (1<<8)
+#define SPI_PHASE4_BIT (1<<3)
+
+#define SPI_PDWN_REG 2 /* power down all DACs */
+#define SPI_PDWN_BIT (1<<2)
+#define SPI_DACD0_REG 10 /* power down individual DACs */
+#define SPI_DACD1_REG 10
+#define SPI_DACD2_REG 10
+#define SPI_DACD4_REG 15
+#define SPI_DACD0_BIT (1<<1)
+#define SPI_DACD1_BIT (1<<2)
+#define SPI_DACD2_BIT (1<<3)
+#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */
+
+#define SPI_PWRDNALL_REG 10 /* power down everything */
+#define SPI_PWRDNALL_BIT (1<<4)
+
+#include "ca_midi.h"
+
+struct snd_ca0106;
+
+struct snd_ca0106_channel {
+ struct snd_ca0106 *emu;
+ int number;
+ int use;
+ void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel);
+ struct snd_ca0106_pcm *epcm;
+};
+
+struct snd_ca0106_pcm {
+ struct snd_ca0106 *emu;
+ struct snd_pcm_substream *substream;
+ int channel_id;
+ unsigned short running;
+};
+
+struct snd_ca0106_details {
+ u32 serial;
+ char * name;
+ int ac97;
+ int gpio_type;
+ int i2c_adc;
+ int spi_dac;
+};
+
+// definition of the chip-specific record
+struct snd_ca0106 {
+ struct snd_card *card;
+ struct snd_ca0106_details *details;
+ struct pci_dev *pci;
+
+ unsigned long port;
+ struct resource *res_port;
+ int irq;
+
+ unsigned int serial; /* serial number */
+ unsigned short model; /* subsystem id */
+
+ spinlock_t emu_lock;
+
+ struct snd_ac97 *ac97;
+ struct snd_pcm *pcm;
+
+ struct snd_ca0106_channel playback_channels[4];
+ struct snd_ca0106_channel capture_channels[4];
+ u32 spdif_bits[4]; /* s/pdif out setup */
+ int spdif_enable;
+ int capture_source;
+ int i2c_capture_source;
+ u8 i2c_capture_volume[4][2];
+ int capture_mic_line_in;
+
+ struct snd_dma_buffer buffer;
+
+ struct snd_ca_midi midi;
+ struct snd_ca_midi midi2;
+
+ u16 spi_dac_reg[16];
+};
+
+int snd_ca0106_mixer(struct snd_ca0106 *emu);
+int snd_ca0106_proc_init(struct snd_ca0106 * emu);
+
+unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu,
+ unsigned int reg,
+ unsigned int chn);
+
+void snd_ca0106_ptr_write(struct snd_ca0106 *emu,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data);
+
+int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
+
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+ unsigned int data);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
new file mode 100644
index 0000000..88fbf28
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -0,0 +1,1735 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.25
+ *
+ * FEATURES currently supported:
+ * Front, Rear and Center/LFE.
+ * Surround40 and Surround51.
+ * Capture from MIC an LINE IN input.
+ * SPDIF digital playback of PCM stereo and AC3/DTS works.
+ * (One can use a standard mono mini-jack to one RCA plugs cable.
+ * or one can use a standard stereo mini-jack to two RCA plugs cable.
+ * Plug one of the RCA plugs into the Coax input of the external decoder/receiver.)
+ * ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. )
+ * Notes on how to capture sound:
+ * The AC97 is used in the PLAYBACK direction.
+ * The output from the AC97 chip, instead of reaching the speakers, is fed into the Philips 1361T ADC.
+ * So, to record from the MIC, set the MIC Playback volume to max,
+ * unmute the MIC and turn up the MASTER Playback volume.
+ * So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume.
+ *
+ * The only playback controls that currently do anything are: -
+ * Analog Front
+ * Analog Rear
+ * Analog Center/LFE
+ * SPDIF Front
+ * SPDIF Rear
+ * SPDIF Center/LFE
+ *
+ * For capture from Mic in or Line in.
+ * Digital/Analog ( switch must be in Analog mode for CAPTURE. )
+ *
+ * CAPTURE feedback into PLAYBACK
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Minor updates.
+ * 0.0.16
+ * Implement 192000 sample rate.
+ * 0.0.17
+ * Add support for SB0410 and SB0413.
+ * 0.0.18
+ * Modified Copyright message.
+ * 0.0.19
+ * Finally fix support for SB Live 24 bit. SB0410 and SB0413.
+ * The output codec needs resetting, otherwise all output is muted.
+ * 0.0.20
+ * Merge "pci_disable_device(pci);" fixes.
+ * 0.0.21
+ * Add 4 capture channels. (SPDIF only comes in on channel 0. )
+ * Add SPDIF capture using optional digital I/O module for SB Live 24bit. (Analog capture does not yet work.)
+ * 0.0.22
+ * Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901
+ * 0.0.23
+ * Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.24
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ * 0.0.25
+ * Powerdown SPI DAC channels when not in use
+ *
+ * BUGS:
+ * Some stability problems when unloading the snd-ca0106 kernel module.
+ * --
+ *
+ * TODO:
+ * 4 Capture channels, only one implemented so far.
+ * Other capture rates apart from 48khz not implemented.
+ * MIDI
+ * --
+ * GENERAL INFO:
+ * Model: SB0310
+ * P17 Chip: CA0106-DAT
+ * AC97 Codec: STAC 9721
+ * ADC: Philips 1361T (Stereo 24bit)
+ * DAC: WM8746EDS (6-channel, 24bit, 192Khz)
+ *
+ * GENERAL INFO:
+ * Model: SB0410
+ * P17 Chip: CA0106-DAT
+ * AC97 Codec: None
+ * ADC: WM8775EDS (4 Channel)
+ * DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support)
+ * SPDIF Out control switches between Mic in and SPDIF out.
+ * No sound out or mic input working yet.
+ *
+ * GENERAL INFO:
+ * Model: SB0413
+ * P17 Chip: CA0106-DAT
+ * AC97 Codec: None.
+ * ADC: Unknown
+ * DAC: Unknown
+ * Trying to handle it like the SB0410.
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+
+MODULE_AUTHOR("James Courtier-Dutton <James@superbug.demon.co.uk>");
+MODULE_DESCRIPTION("CA0106");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}");
+
+// module parameters (see "Module Parameters")
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the CA0106 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the CA0106 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable the CA0106 soundcard.");
+module_param_array(subsystem, uint, NULL, 0444);
+MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
+
+#include "ca0106.h"
+
+static struct snd_ca0106_details ca0106_chip_details[] = {
+ /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+ /* It is really just a normal SB Live 24bit. */
+ /* Tested:
+ * See ALSA bug#3251
+ */
+ { .serial = 0x10131102,
+ .name = "X-Fi Extreme Audio [SBxxxx]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+ /* It is really just a normal SB Live 24bit. */
+ /*
+ * CTRL:CA0111-WTLF
+ * ADC: WM8775SEDS
+ * DAC: CS4382-KQZ
+ */
+ /* Tested:
+ * Playback on front, rear, center/lfe speakers
+ * Capture from Mic in.
+ * Not-Tested:
+ * Capture from Line in.
+ * Playback to digital out.
+ */
+ { .serial = 0x10121102,
+ .name = "X-Fi Extreme Audio [SB0790]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
+ /* AudigyLS[SB0310] */
+ { .serial = 0x10021102,
+ .name = "AudigyLS [SB0310]",
+ .ac97 = 1 } ,
+ /* Unknown AudigyLS that also says SB0310 on it */
+ { .serial = 0x10051102,
+ .name = "AudigyLS [SB0310b]",
+ .ac97 = 1 } ,
+ /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
+ { .serial = 0x10061102,
+ .name = "Live! 7.1 24bit [SB0410]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
+ { .serial = 0x10071102,
+ .name = "Live! 7.1 24bit [SB0413]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* New Audigy SE. Has a different DAC. */
+ /* SB0570:
+ * CTRL:CA0106-DAT
+ * ADC: WM8775EDS
+ * DAC: WM8768GEDS
+ */
+ { .serial = 0x100a1102,
+ .name = "Audigy SE [SB0570]",
+ .gpio_type = 1,
+ .i2c_adc = 1,
+ .spi_dac = 1 } ,
+ /* New Audigy LS. Has a different DAC. */
+ /* SB0570:
+ * CTRL:CA0106-DAT
+ * ADC: WM8775EDS
+ * DAC: WM8768GEDS
+ */
+ { .serial = 0x10111102,
+ .name = "Audigy SE OEM [SB0570a]",
+ .gpio_type = 1,
+ .i2c_adc = 1,
+ .spi_dac = 1 } ,
+ /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */
+ /* SB0438
+ * CTRL:CA0106-DAT
+ * ADC: WM8775SEDS
+ * DAC: CS4382-KQZ
+ */
+ { .serial = 0x10091462,
+ .name = "MSI K8N Diamond MB [SB0438]",
+ .gpio_type = 2,
+ .i2c_adc = 1 } ,
+ /* MSI K8N Diamond PLUS MB */
+ { .serial = 0x10091102,
+ .name = "MSI K8N Diamond MB",
+ .gpio_type = 2,
+ .i2c_adc = 1,
+ .spi_dac = 2 } ,
+ /* Shuttle XPC SD31P which has an onboard Creative Labs
+ * Sound Blaster Live! 24-bit EAX
+ * high-definition 7.1 audio processor".
+ * Added using info from andrewvegan in alsa bug #1298
+ */
+ { .serial = 0x30381297,
+ .name = "Shuttle XPC SD31P [SD31P]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* Shuttle XPC SD11G5 which has an onboard Creative Labs
+ * Sound Blaster Live! 24-bit EAX
+ * high-definition 7.1 audio processor".
+ * Fixes ALSA bug#1600
+ */
+ { .serial = 0x30411297,
+ .name = "Shuttle XPC SD11G5 [SD11G5]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ { .serial = 0,
+ .name = "AudigyLS [Unknown]" }
+};
+
+/* hardware definition */
+static struct snd_pcm_hardware snd_ca0106_playback_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000),
+ .rate_min = 48000,
+ .rate_max = 192000,
+ .channels_min = 2, //1,
+ .channels_max = 2, //6,
+ .buffer_bytes_max = ((65536 - 64) * 8),
+ .period_bytes_min = 64,
+ .period_bytes_max = (65536 - 64),
+ .periods_min = 2,
+ .periods_max = 8,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_ca0106_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
+ .rate_min = 44100,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = ((65536 - 64) * 8),
+ .period_bytes_min = 64,
+ .period_bytes_max = (65536 - 64),
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu,
+ unsigned int reg,
+ unsigned int chn)
+{
+ unsigned long flags;
+ unsigned int regptr, val;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+void snd_ca0106_ptr_write(struct snd_ca0106 *emu,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data)
+{
+ unsigned int regptr;
+ unsigned long flags;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ outl(data, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+ unsigned int data)
+{
+ unsigned int reset, set;
+ unsigned int reg, tmp;
+ int n, result;
+ reg = SPI;
+ if (data > 0xffff) /* Only 16bit values allowed */
+ return 1;
+ tmp = snd_ca0106_ptr_read(emu, reg, 0);
+ reset = (tmp & ~0x3ffff) | 0x20000; /* Set xxx20000 */
+ set = reset | 0x10000; /* Set xxx1xxxx */
+ snd_ca0106_ptr_write(emu, reg, 0, reset | data);
+ tmp = snd_ca0106_ptr_read(emu, reg, 0); /* write post */
+ snd_ca0106_ptr_write(emu, reg, 0, set | data);
+ result = 1;
+ /* Wait for status bit to return to 0 */
+ for (n = 0; n < 100; n++) {
+ udelay(10);
+ tmp = snd_ca0106_ptr_read(emu, reg, 0);
+ if (!(tmp & 0x10000)) {
+ result = 0;
+ break;
+ }
+ }
+ if (result) /* Timed out */
+ return 1;
+ snd_ca0106_ptr_write(emu, reg, 0, reset | data);
+ tmp = snd_ca0106_ptr_read(emu, reg, 0); /* Write post */
+ return 0;
+}
+
+/* The ADC does not support i2c read, so only write is implemented */
+int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
+ u32 reg,
+ u32 value)
+{
+ u32 tmp;
+ int timeout = 0;
+ int status;
+ int retry;
+ if ((reg > 0x7f) || (value > 0x1ff)) {
+ snd_printk(KERN_ERR "i2c_write: invalid values.\n");
+ return -EINVAL;
+ }
+
+ tmp = reg << 25 | value << 16;
+ // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+ /* Not sure what this I2C channel controls. */
+ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
+
+ /* This controls the I2C connected to the WM8775 ADC Codec */
+ snd_ca0106_ptr_write(emu, I2C_D1, 0, tmp);
+
+ for (retry = 0; retry < 10; retry++) {
+ /* Send the data to i2c */
+ //tmp = snd_ca0106_ptr_read(emu, I2C_A, 0);
+ //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK);
+ tmp = 0;
+ tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD);
+ snd_ca0106_ptr_write(emu, I2C_A, 0, tmp);
+
+ /* Wait till the transaction ends */
+ while (1) {
+ status = snd_ca0106_ptr_read(emu, I2C_A, 0);
+ //snd_printk("I2C:status=0x%x\n", status);
+ timeout++;
+ if ((status & I2C_A_ADC_START) == 0)
+ break;
+
+ if (timeout > 1000)
+ break;
+ }
+ //Read back and see if the transaction is successful
+ if ((status & I2C_A_ADC_ABORT) == 0)
+ break;
+ }
+
+ if (retry == 10) {
+ snd_printk(KERN_ERR "Writing to ADC failed!\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int intr_enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ intr_enable = inl(emu->port + INTE) | intrenb;
+ outl(intr_enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int intr_enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ intr_enable = inl(emu->port + INTE) & ~intrenb;
+ outl(intr_enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+
+static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static const int spi_dacd_reg[] = {
+ [PCM_FRONT_CHANNEL] = SPI_DACD4_REG,
+ [PCM_REAR_CHANNEL] = SPI_DACD0_REG,
+ [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_REG,
+ [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_REG,
+};
+static const int spi_dacd_bit[] = {
+ [PCM_FRONT_CHANNEL] = SPI_DACD4_BIT,
+ [PCM_REAR_CHANNEL] = SPI_DACD0_BIT,
+ [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_BIT,
+ [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT,
+};
+
+/* open_playback callback */
+static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream,
+ int channel_id)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]);
+ struct snd_ca0106_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = chip;
+ epcm->substream = substream;
+ epcm->channel_id=channel_id;
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_ca0106_pcm_free_substream;
+
+ runtime->hw = snd_ca0106_playback_hw;
+
+ channel->emu = chip;
+ channel->number = channel_id;
+
+ channel->use = 1;
+ //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+ //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+ channel->epcm = epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+ snd_pcm_set_sync(substream);
+
+ if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) {
+ const int reg = spi_dacd_reg[channel_id];
+
+ /* Power up dac */
+ chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id];
+ err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ chip->playback_channels[epcm->channel_id].use = 0;
+
+ if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) {
+ const int reg = spi_dacd_reg[epcm->channel_id];
+
+ /* Power down DAC */
+ chip->spi_dac_reg[reg] |= spi_dacd_bit[epcm->channel_id];
+ snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ }
+ /* FIXME: maybe zero others */
+ return 0;
+}
+
+static int snd_ca0106_pcm_open_playback_front(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_center_lfe(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_unknown(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_rear(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL);
+}
+
+/* open_capture callback */
+static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substream,
+ int channel_id)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]);
+ struct snd_ca0106_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL) {
+ snd_printk(KERN_ERR "open_capture_channel: failed epcm alloc\n");
+ return -ENOMEM;
+ }
+ epcm->emu = chip;
+ epcm->substream = substream;
+ epcm->channel_id=channel_id;
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_ca0106_pcm_free_substream;
+
+ runtime->hw = snd_ca0106_capture_hw;
+
+ channel->emu = chip;
+ channel->number = channel_id;
+
+ channel->use = 1;
+ //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+ //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+ channel->epcm = epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes);
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+ return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ chip->capture_channels[epcm->channel_id].use = 0;
+ /* FIXME: maybe zero others */
+ return 0;
+}
+
+static int snd_ca0106_pcm_open_0_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 0);
+}
+
+static int snd_ca0106_pcm_open_1_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 1);
+}
+
+static int snd_ca0106_pcm_open_2_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 2);
+}
+
+static int snd_ca0106_pcm_open_3_capture(struct snd_pcm_substream *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 3);
+}
+
+/* hw_params callback */
+static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_ca0106_pcm_hw_free_playback(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* hw_params callback */
+static int snd_ca0106_pcm_hw_params_capture(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_ca0106_pcm_hw_free_capture(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare playback callback */
+static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ int channel = epcm->channel_id;
+ u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel));
+ u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
+ u32 hcfg_mask = HCFG_PLAYBACK_S32_LE;
+ u32 hcfg_set = 0x00000000;
+ u32 hcfg;
+ u32 reg40_mask = 0x30000 << (channel<<1);
+ u32 reg40_set = 0;
+ u32 reg40;
+ /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */
+ u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */
+ u32 reg71_set = 0;
+ u32 reg71;
+ int i;
+
+ //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
+ //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
+ //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+ /* Rate can be set per channel. */
+ /* reg40 control host to fifo */
+ /* reg71 controls DAC rate. */
+ switch (runtime->rate) {
+ case 44100:
+ reg40_set = 0x10000 << (channel<<1);
+ reg71_set = 0x01010000;
+ break;
+ case 48000:
+ reg40_set = 0;
+ reg71_set = 0;
+ break;
+ case 96000:
+ reg40_set = 0x20000 << (channel<<1);
+ reg71_set = 0x02020000;
+ break;
+ case 192000:
+ reg40_set = 0x30000 << (channel<<1);
+ reg71_set = 0x03030000;
+ break;
+ default:
+ reg40_set = 0;
+ reg71_set = 0;
+ break;
+ }
+ /* Format is a global setting */
+ /* FIXME: Only let the first channel accessed set this. */
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hcfg_set = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ hcfg_set = HCFG_PLAYBACK_S32_LE;
+ break;
+ default:
+ hcfg_set = 0;
+ break;
+ }
+ hcfg = inl(emu->port + HCFG) ;
+ hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+ outl(hcfg, emu->port + HCFG);
+ reg40 = snd_ca0106_ptr_read(emu, 0x40, 0);
+ reg40 = (reg40 & ~reg40_mask) | reg40_set;
+ snd_ca0106_ptr_write(emu, 0x40, 0, reg40);
+ reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
+ reg71 = (reg71 & ~reg71_mask) | reg71_set;
+ snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
+
+ /* FIXME: Check emu->buffer.size before actually writing to it. */
+ for(i=0; i < runtime->periods; i++) {
+ table_base[i*2] = runtime->dma_addr + (i * period_size_bytes);
+ table_base[i*2+1] = period_size_bytes << 16;
+ }
+
+ snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel));
+ snd_ca0106_ptr_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19);
+ snd_ca0106_ptr_write(emu, PLAYBACK_LIST_PTR, channel, 0);
+ snd_ca0106_ptr_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr);
+ snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
+ /* FIXME test what 0 bytes does. */
+ snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes
+ snd_ca0106_ptr_write(emu, PLAYBACK_POINTER, channel, 0);
+ snd_ca0106_ptr_write(emu, 0x07, channel, 0x0);
+ snd_ca0106_ptr_write(emu, 0x08, channel, 0);
+ snd_ca0106_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */
+#if 0
+ snd_ca0106_ptr_write(emu, SPCS0, 0,
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT );
+#endif
+
+ return 0;
+}
+
+/* prepare capture callback */
+static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ int channel = epcm->channel_id;
+ u32 hcfg_mask = HCFG_CAPTURE_S32_LE;
+ u32 hcfg_set = 0x00000000;
+ u32 hcfg;
+ u32 over_sampling=0x2;
+ u32 reg71_mask = 0x0000c000 ; /* Global. Set ADC rate. */
+ u32 reg71_set = 0;
+ u32 reg71;
+
+ //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
+ //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
+ //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+ /* reg71 controls ADC rate. */
+ switch (runtime->rate) {
+ case 44100:
+ reg71_set = 0x00004000;
+ break;
+ case 48000:
+ reg71_set = 0;
+ break;
+ case 96000:
+ reg71_set = 0x00008000;
+ over_sampling=0xa;
+ break;
+ case 192000:
+ reg71_set = 0x0000c000;
+ over_sampling=0xa;
+ break;
+ default:
+ reg71_set = 0;
+ break;
+ }
+ /* Format is a global setting */
+ /* FIXME: Only let the first channel accessed set this. */
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hcfg_set = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ hcfg_set = HCFG_CAPTURE_S32_LE;
+ break;
+ default:
+ hcfg_set = 0;
+ break;
+ }
+ hcfg = inl(emu->port + HCFG) ;
+ hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+ outl(hcfg, emu->port + HCFG);
+ reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
+ reg71 = (reg71 & ~reg71_mask) | reg71_set;
+ snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
+ if (emu->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
+ snd_ca0106_i2c_write(emu, ADC_MASTER, over_sampling); /* Adjust the over sampler to better suit the capture rate. */
+ }
+
+
+ //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1));
+ snd_ca0106_ptr_write(emu, 0x13, channel, 0);
+ snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
+ snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
+ snd_ca0106_ptr_write(emu, CAPTURE_POINTER, channel, 0);
+
+ return 0;
+}
+
+/* trigger_playback callback */
+static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime;
+ struct snd_ca0106_pcm *epcm;
+ int channel;
+ int result = 0;
+ struct snd_pcm_substream *s;
+ u32 basic = 0;
+ u32 extended = 0;
+ int running=0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ running=1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ default:
+ running=0;
+ break;
+ }
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) != emu ||
+ s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ continue;
+ runtime = s->runtime;
+ epcm = runtime->private_data;
+ channel = epcm->channel_id;
+ //snd_printk("channel=%d\n",channel);
+ epcm->running = running;
+ basic |= (0x1<<channel);
+ extended |= (0x10<<channel);
+ snd_pcm_trigger_done(s, substream);
+ }
+ //snd_printk("basic=0x%x, extended=0x%x\n",basic, extended);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (extended));
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(basic));
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(basic));
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(extended));
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* trigger_capture callback */
+static int snd_ca0106_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ int channel = epcm->channel_id;
+ int result = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<channel));
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel));
+ epcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel));
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel));
+ epcm->running = 0;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* pointer_playback callback */
+static snd_pcm_uframes_t
+snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0;
+ int channel = epcm->channel_id;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr3 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+ ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel);
+ ptr4 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+ if (ptr3 != ptr4) ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr2+= (ptr4 >> 3) * runtime->period_size;
+ ptr=ptr2;
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+ //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
+
+ return ptr;
+}
+
+/* pointer_capture callback */
+static snd_pcm_uframes_t
+snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ca0106_pcm *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
+ int channel = channel=epcm->channel_id;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr1 = snd_ca0106_ptr_read(emu, CAPTURE_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr=ptr2;
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+ //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
+
+ return ptr;
+}
+
+/* operators */
+static struct snd_pcm_ops snd_ca0106_playback_front_ops = {
+ .open = snd_ca0106_pcm_open_playback_front,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_0_ops = {
+ .open = snd_ca0106_pcm_open_0_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_1_ops = {
+ .open = snd_ca0106_pcm_open_1_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_2_ops = {
+ .open = snd_ca0106_pcm_open_2_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_capture_3_ops = {
+ .open = snd_ca0106_pcm_open_3_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = {
+ .open = snd_ca0106_pcm_open_playback_center_lfe,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static struct snd_pcm_ops snd_ca0106_playback_unknown_ops = {
+ .open = snd_ca0106_pcm_open_playback_unknown,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static struct snd_pcm_ops snd_ca0106_playback_rear_ops = {
+ .open = snd_ca0106_pcm_open_playback_rear,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+
+static unsigned short snd_ca0106_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct snd_ca0106 *emu = ac97->private_data;
+ unsigned long flags;
+ unsigned short val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ val = inw(emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+static void snd_ca0106_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ struct snd_ca0106 *emu = ac97->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ outw(val, emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static int snd_ca0106_ac97(struct snd_ca0106 *chip)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_ca0106_ac97_write,
+ .read = snd_ca0106_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+ pbus->no_vra = 1; /* we don't need VRA */
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.scaps = AC97_SCAP_NO_SPDIF;
+ return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
+}
+
+static int snd_ca0106_free(struct snd_ca0106 *chip)
+{
+ if (chip->res_port != NULL) { /* avoid access to already used hardware */
+ // disable interrupts
+ snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
+ outl(0, chip->port + INTE);
+ snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
+ udelay(1000);
+ // disable audio
+ //outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
+ outl(0, chip->port + HCFG);
+ /* FIXME: We need to stop and DMA transfers here.
+ * But as I am not sure how yet, we cannot from the dma pages.
+ * So we can fix: snd-malloc: Memory leak? pages not freed = 8
+ */
+ }
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ // release the data
+#if 1
+ if (chip->buffer.area)
+ snd_dma_free_pages(&chip->buffer);
+#endif
+
+ // release the i/o port
+ release_and_free_resource(chip->res_port);
+
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_ca0106_dev_free(struct snd_device *device)
+{
+ struct snd_ca0106 *chip = device->device_data;
+ return snd_ca0106_free(chip);
+}
+
+static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
+{
+ unsigned int status;
+
+ struct snd_ca0106 *chip = dev_id;
+ int i;
+ int mask;
+ unsigned int stat76;
+ struct snd_ca0106_channel *pchannel;
+
+ status = inl(chip->port + IPR);
+ if (! status)
+ return IRQ_NONE;
+
+ stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
+ //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76);
+ //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+ mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
+ for(i = 0; i < 4; i++) {
+ pchannel = &(chip->playback_channels[i]);
+ if (stat76 & mask) {
+/* FIXME: Select the correct substream for period elapsed */
+ if(pchannel->use) {
+ snd_pcm_period_elapsed(pchannel->epcm->substream);
+ //printk(KERN_INFO "interrupt [%d] used\n", i);
+ }
+ }
+ //printk(KERN_INFO "channel=%p\n",pchannel);
+ //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
+ mask <<= 1;
+ }
+ mask = 0x110000; /* 0x1 for one half, 0x10 for the other half period. */
+ for(i = 0; i < 4; i++) {
+ pchannel = &(chip->capture_channels[i]);
+ if (stat76 & mask) {
+/* FIXME: Select the correct substream for period elapsed */
+ if(pchannel->use) {
+ snd_pcm_period_elapsed(pchannel->epcm->substream);
+ //printk(KERN_INFO "interrupt [%d] used\n", i);
+ }
+ }
+ //printk(KERN_INFO "channel=%p\n",pchannel);
+ //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
+ mask <<= 1;
+ }
+
+ snd_ca0106_ptr_write(chip, EXTENDED_INT, 0, stat76);
+
+ if (chip->midi.dev_id &&
+ (status & (chip->midi.ipr_tx|chip->midi.ipr_rx))) {
+ if (chip->midi.interrupt)
+ chip->midi.interrupt(&chip->midi, status);
+ else
+ chip->midi.interrupt_disable(&chip->midi, chip->midi.tx_enable | chip->midi.rx_enable);
+ }
+
+ // acknowledge the interrupt if necessary
+ outl(status, chip->port+IPR);
+
+ return IRQ_HANDLED;
+}
+
+static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+
+ switch (device) {
+ case 0:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_front_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_0_ops);
+ break;
+ case 1:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_rear_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_1_ops);
+ break;
+ case 2:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_center_lfe_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_2_ops);
+ break;
+ case 3:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_unknown_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_3_ops);
+ break;
+ }
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strcpy(pcm->name, "CA0106");
+ emu->pcm = pcm;
+
+ for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */
+ return err;
+ }
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 64*1024, 64*1024)) < 0)
+ return err;
+ }
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value))
+static unsigned int spi_dac_init[] = {
+ SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */
+ SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT),
+ SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24),
+ SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB),
+ SPI_REG(9, 0x00),
+ SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT),
+ SPI_REG(12, 0x00),
+ SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE),
+ SPI_REG(SPI_DACD4_REG, 0x00),
+};
+
+static unsigned int i2c_adc_init[][2] = {
+ { 0x17, 0x00 }, /* Reset */
+ { 0x07, 0x00 }, /* Timeout */
+ { 0x0b, 0x22 }, /* Interface control */
+ { 0x0c, 0x22 }, /* Master mode control */
+ { 0x0d, 0x08 }, /* Powerdown control */
+ { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */
+ { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */
+ { 0x10, 0x7b }, /* ALC Control 1 */
+ { 0x11, 0x00 }, /* ALC Control 2 */
+ { 0x12, 0x32 }, /* ALC Control 3 */
+ { 0x13, 0x00 }, /* Noise gate control */
+ { 0x14, 0xa6 }, /* Limiter control */
+ { 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */
+};
+
+static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
+ struct pci_dev *pci,
+ struct snd_ca0106 **rchip)
+{
+ struct snd_ca0106 *chip;
+ struct snd_ca0106_details *c;
+ int err;
+ int ch;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ca0106_dev_free,
+ };
+
+ *rchip = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
+ printk(KERN_ERR "error to set 32bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ spin_lock_init(&chip->emu_lock);
+
+ chip->port = pci_resource_start(pci, 0);
+ if ((chip->res_port = request_region(chip->port, 0x20,
+ "snd_ca0106")) == NULL) {
+ snd_ca0106_free(chip);
+ printk(KERN_ERR "cannot allocate the port\n");
+ return -EBUSY;
+ }
+
+ if (request_irq(pci->irq, snd_ca0106_interrupt,
+ IRQF_SHARED, "snd_ca0106", chip)) {
+ snd_ca0106_free(chip);
+ printk(KERN_ERR "cannot grab irq\n");
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ /* This stores the periods table. */
+ if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) {
+ snd_ca0106_free(chip);
+ return -ENOMEM;
+ }
+
+ pci_set_master(pci);
+ /* read serial */
+ pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
+ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
+#if 1
+ printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model,
+ pci->revision, chip->serial);
+#endif
+ strcpy(card->driver, "CA0106");
+ strcpy(card->shortname, "CA0106");
+
+ for (c = ca0106_chip_details; c->serial; c++) {
+ if (subsystem[dev]) {
+ if (c->serial == subsystem[dev])
+ break;
+ } else if (c->serial == chip->serial)
+ break;
+ }
+ chip->details = c;
+ if (subsystem[dev]) {
+ printk(KERN_INFO "snd-ca0106: Sound card name=%s, subsystem=0x%x. Forced to subsystem=0x%x\n",
+ c->name, chip->serial, subsystem[dev]);
+ }
+
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ c->name, chip->port, chip->irq);
+
+ outl(0, chip->port + INTE);
+
+ /*
+ * Init to 0x02109204 :
+ * Clock accuracy = 0 (1000ppm)
+ * Sample Rate = 2 (48kHz)
+ * Audio Channel = 1 (Left of 2)
+ * Source Number = 0 (Unspecified)
+ * Generation Status = 1 (Original for Cat Code 12)
+ * Cat Code = 12 (Digital Signal Mixer)
+ * Mode = 0 (Mode 0)
+ * Emphasis = 0 (None)
+ * CP = 1 (Copyright unasserted)
+ * AN = 0 (Audio data)
+ * P = 0 (Consumer)
+ */
+ snd_ca0106_ptr_write(chip, SPCS0, 0,
+ chip->spdif_bits[0] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ /* Only SPCS1 has been tested */
+ snd_ca0106_ptr_write(chip, SPCS1, 0,
+ chip->spdif_bits[1] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ snd_ca0106_ptr_write(chip, SPCS2, 0,
+ chip->spdif_bits[2] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ snd_ca0106_ptr_write(chip, SPCS3, 0,
+ chip->spdif_bits[3] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+
+ snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000);
+ snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000);
+
+ /* Write 0x8000 to AC97_REC_GAIN to mute it. */
+ outb(AC97_REC_GAIN, chip->port + AC97ADDRESS);
+ outw(0x8000, chip->port + AC97DATA);
+#if 0
+ snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006);
+ snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006);
+ snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006);
+ snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006);
+#endif
+
+ //snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */
+ /* Analog or Digital output */
+ snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. Use 0x000f0000 for surround71 */
+ chip->spdif_enable = 0; /* Set digital SPDIF output off */
+ //snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
+ //snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */
+
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
+ snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410);
+ snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676);
+ snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410);
+ snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676);
+ for(ch = 0; ch < 4; ch++) {
+ snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */
+ snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
+ //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */
+ //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */
+ }
+ if (chip->details->i2c_adc == 1) {
+ /* Select MIC, Line in, TAD in, AUX in */
+ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
+ /* Default to CAPTURE_SOURCE to i2s in */
+ chip->capture_source = 3;
+ } else if (chip->details->ac97 == 1) {
+ /* Default to AC97 in */
+ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4);
+ /* Default to CAPTURE_SOURCE to AC97 in */
+ chip->capture_source = 4;
+ } else {
+ /* Select MIC, Line in, TAD in, AUX in */
+ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
+ /* Default to Set CAPTURE_SOURCE to i2s in */
+ chip->capture_source = 3;
+ }
+
+ if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */
+ /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
+ outl(0x0, chip->port+GPIO);
+ //outl(0x00f0e000, chip->port+GPIO); /* Analog */
+ outl(0x005f5301, chip->port+GPIO); /* Analog */
+ } else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */
+ /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
+ outl(0x0, chip->port+GPIO);
+ //outl(0x00f0e000, chip->port+GPIO); /* Analog */
+ outl(0x005f5301, chip->port+GPIO); /* Analog */
+ } else {
+ outl(0x0, chip->port+GPIO);
+ outl(0x005f03a3, chip->port+GPIO); /* Analog */
+ //outl(0x005f02a2, chip->port+GPIO); /* SPDIF */
+ }
+ snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
+
+ //outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
+ //outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
+ //outl(0x00000009, chip->port+HCFG);
+ outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */
+
+ if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
+ int size, n;
+
+ size = ARRAY_SIZE(i2c_adc_init);
+ //snd_printk("I2C:array size=0x%x\n", size);
+ for (n=0; n < size; n++) {
+ snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]);
+ }
+ for (n=0; n < 4; n++) {
+ chip->i2c_capture_volume[n][0]= 0xcf;
+ chip->i2c_capture_volume[n][1]= 0xcf;
+ }
+ chip->i2c_capture_source=2; /* Line in */
+ //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */
+ }
+ if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */
+ int size, n;
+
+ size = ARRAY_SIZE(spi_dac_init);
+ for (n = 0; n < size; n++) {
+ int reg = spi_dac_init[n] >> SPI_REG_SHIFT;
+
+ snd_ca0106_spi_write(chip, spi_dac_init[n]);
+ if (reg < ARRAY_SIZE(chip->spi_dac_reg))
+ chip->spi_dac_reg[reg] = spi_dac_init[n];
+ }
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &ops)) < 0) {
+ snd_ca0106_free(chip);
+ return err;
+ }
+ *rchip = chip;
+ return 0;
+}
+
+
+static void ca0106_midi_interrupt_enable(struct snd_ca_midi *midi, int intr)
+{
+ snd_ca0106_intr_enable((struct snd_ca0106 *)(midi->dev_id), intr);
+}
+
+static void ca0106_midi_interrupt_disable(struct snd_ca_midi *midi, int intr)
+{
+ snd_ca0106_intr_disable((struct snd_ca0106 *)(midi->dev_id), intr);
+}
+
+static unsigned char ca0106_midi_read(struct snd_ca_midi *midi, int idx)
+{
+ return (unsigned char)snd_ca0106_ptr_read((struct snd_ca0106 *)(midi->dev_id),
+ midi->port + idx, 0);
+}
+
+static void ca0106_midi_write(struct snd_ca_midi *midi, int data, int idx)
+{
+ snd_ca0106_ptr_write((struct snd_ca0106 *)(midi->dev_id), midi->port + idx, 0, data);
+}
+
+static struct snd_card *ca0106_dev_id_card(void *dev_id)
+{
+ return ((struct snd_ca0106 *)dev_id)->card;
+}
+
+static int ca0106_dev_id_port(void *dev_id)
+{
+ return ((struct snd_ca0106 *)dev_id)->port;
+}
+
+static int __devinit snd_ca0106_midi(struct snd_ca0106 *chip, unsigned int channel)
+{
+ struct snd_ca_midi *midi;
+ char *name;
+ int err;
+
+ if (channel == CA0106_MIDI_CHAN_B) {
+ name = "CA0106 MPU-401 (UART) B";
+ midi = &chip->midi2;
+ midi->tx_enable = INTE_MIDI_TX_B;
+ midi->rx_enable = INTE_MIDI_RX_B;
+ midi->ipr_tx = IPR_MIDI_TX_B;
+ midi->ipr_rx = IPR_MIDI_RX_B;
+ midi->port = MIDI_UART_B_DATA;
+ } else {
+ name = "CA0106 MPU-401 (UART)";
+ midi = &chip->midi;
+ midi->tx_enable = INTE_MIDI_TX_A;
+ midi->rx_enable = INTE_MIDI_TX_B;
+ midi->ipr_tx = IPR_MIDI_TX_A;
+ midi->ipr_rx = IPR_MIDI_RX_A;
+ midi->port = MIDI_UART_A_DATA;
+ }
+
+ midi->reset = CA0106_MPU401_RESET;
+ midi->enter_uart = CA0106_MPU401_ENTER_UART;
+ midi->ack = CA0106_MPU401_ACK;
+
+ midi->input_avail = CA0106_MIDI_INPUT_AVAIL;
+ midi->output_ready = CA0106_MIDI_OUTPUT_READY;
+
+ midi->channel = channel;
+
+ midi->interrupt_enable = ca0106_midi_interrupt_enable;
+ midi->interrupt_disable = ca0106_midi_interrupt_disable;
+
+ midi->read = ca0106_midi_read;
+ midi->write = ca0106_midi_write;
+
+ midi->get_dev_id_card = ca0106_dev_id_card;
+ midi->get_dev_id_port = ca0106_dev_id_port;
+
+ midi->dev_id = chip;
+
+ if ((err = ca_midi_init(chip, midi, 0, name)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+static int __devinit snd_ca0106_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_ca0106 *chip;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_ca0106_create(dev, card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if (chip->details->ac97 == 1) { /* The SB0410 and SB0413 do not have an AC97 chip. */
+ if ((err = snd_ca0106_ac97(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+ if ((err = snd_ca0106_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_printdd("ca0106: probe for MIDI channel A ...");
+ if ((err = snd_ca0106_midi(chip,CA0106_MIDI_CHAN_A)) < 0) {
+ snd_card_free(card);
+ snd_printdd(" failed, err=0x%x\n",err);
+ return err;
+ }
+ snd_printdd(" done.\n");
+
+#ifdef CONFIG_PROC_FS
+ snd_ca0106_proc_init(chip);
+#endif
+
+ snd_card_set_dev(card, &pci->dev);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_ca0106_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+// PCI IDs
+static struct pci_device_id snd_ca0106_ids[] = {
+ { 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */
+ { 0, }
+};
+MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
+
+// pci_driver definition
+static struct pci_driver driver = {
+ .name = "CA0106",
+ .id_table = snd_ca0106_ids,
+ .probe = snd_ca0106_probe,
+ .remove = __devexit_p(snd_ca0106_remove),
+};
+
+// initialization of the module
+static int __init alsa_card_ca0106_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+// clean up the module
+static void __exit alsa_card_ca0106_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_ca0106_init)
+module_exit(alsa_card_ca0106_exit)
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
new file mode 100644
index 0000000..3025ed1
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -0,0 +1,775 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.18
+ *
+ * FEATURES currently supported:
+ * See ca0106_main.c for features.
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Separated ca0106.c into separate functional .c files.
+ * 0.0.16
+ * Modified Copyright message.
+ * 0.0.17
+ * Implement Mic and Line in Capture.
+ * 0.0.18
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+#include <asm/io.h>
+
+#include "ca0106.h"
+
+static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
+static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
+
+#define snd_ca0106_shared_spdif_info snd_ctl_boolean_mono_info
+
+static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = emu->spdif_enable;
+ return 0;
+}
+
+static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 mask;
+
+ val = !!ucontrol->value.integer.value[0];
+ change = (emu->spdif_enable != val);
+ if (change) {
+ emu->spdif_enable = val;
+ if (val) {
+ /* Digital */
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
+ snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
+ snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000);
+ mask = inl(emu->port + GPIO) & ~0x101;
+ outl(mask, emu->port + GPIO);
+
+ } else {
+ /* Analog */
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
+ snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
+ snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000);
+ mask = inl(emu->port + GPIO) | 0x101;
+ outl(mask, emu->port + GPIO);
+ }
+ }
+ return change;
+}
+
+static int snd_ca0106_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[6] = {
+ "IEC958 out", "i2s mixer out", "IEC958 in", "i2s in", "AC97 in", "SRC out"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 6;
+ if (uinfo->value.enumerated.item > 5)
+ uinfo->value.enumerated.item = 5;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ca0106_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->capture_source;
+ return 0;
+}
+
+static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 mask;
+ u32 source;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ if (val >= 6)
+ return -EINVAL;
+ change = (emu->capture_source != val);
+ if (change) {
+ emu->capture_source = val;
+ source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
+ mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
+ snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
+ }
+ return change;
+}
+
+static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[6] = {
+ "Phone", "Mic", "Line in", "Aux"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->i2c_capture_source;
+ return 0;
+}
+
+static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int source_id;
+ unsigned int ngain, ogain;
+ int change = 0;
+ u32 source;
+ /* If the capture source has changed,
+ * update the capture volume from the cached value
+ * for the particular source.
+ */
+ source_id = ucontrol->value.enumerated.item[0] ;
+ if (source_id >= 4)
+ return -EINVAL;
+ change = (emu->i2c_capture_source != source_id);
+ if (change) {
+ snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+ ngain = emu->i2c_capture_volume[source_id][0]; /* Left */
+ ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
+ if (ngain != ogain)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff));
+ ngain = emu->i2c_capture_volume[source_id][1]; /* Left */
+ ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Left */
+ if (ngain != ogain)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+ source = 1 << source_id;
+ snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
+ emu->i2c_capture_source = source_id;
+ }
+ return change;
+}
+
+static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "Side out", "Line in" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "Line in", "Mic in" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ca0106_capture_mic_line_in_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->capture_mic_line_in;
+ return 0;
+}
+
+static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 tmp;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ if (val > 1)
+ return -EINVAL;
+ change = (emu->capture_mic_line_in != val);
+ if (change) {
+ emu->capture_mic_line_in = val;
+ if (val) {
+ //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+ tmp = inl(emu->port+GPIO) & ~0x400;
+ tmp = tmp | 0x400;
+ outl(tmp, emu->port+GPIO);
+ //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC);
+ } else {
+ //snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+ tmp = inl(emu->port+GPIO) & ~0x400;
+ outl(tmp, emu->port+GPIO);
+ //snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN);
+ }
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Shared Mic/Line in Capture Switch",
+ .info = snd_ca0106_capture_mic_line_in_info,
+ .get = snd_ca0106_capture_mic_line_in_get,
+ .put = snd_ca0106_capture_mic_line_in_put
+};
+
+static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Shared Line in/Side out Capture Switch",
+ .info = snd_ca0106_capture_line_in_side_out_info,
+ .get = snd_ca0106_capture_mic_line_in_get,
+ .put = snd_ca0106_capture_mic_line_in_put
+};
+
+
+static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ca0106_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff;
+ return 0;
+}
+
+static int snd_ca0106_spdif_get_mask(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static int snd_ca0106_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ int change;
+ unsigned int val;
+
+ val = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+ change = val != emu->spdif_bits[idx];
+ if (change) {
+ snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, val);
+ emu->spdif_bits[idx] = val;
+ }
+ return change;
+}
+
+static int snd_ca0106_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_ca0106_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int value;
+ int channel_id, reg;
+
+ channel_id = (kcontrol->private_value >> 8) & 0xff;
+ reg = kcontrol->private_value & 0xff;
+
+ value = snd_ca0106_ptr_read(emu, reg, channel_id);
+ ucontrol->value.integer.value[0] = 0xff - ((value >> 24) & 0xff); /* Left */
+ ucontrol->value.integer.value[1] = 0xff - ((value >> 16) & 0xff); /* Right */
+ return 0;
+}
+
+static int snd_ca0106_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int oval, nval;
+ int channel_id, reg;
+
+ channel_id = (kcontrol->private_value >> 8) & 0xff;
+ reg = kcontrol->private_value & 0xff;
+
+ oval = snd_ca0106_ptr_read(emu, reg, channel_id);
+ nval = ((0xff - ucontrol->value.integer.value[0]) << 24) |
+ ((0xff - ucontrol->value.integer.value[1]) << 16);
+ nval |= ((0xff - ucontrol->value.integer.value[0]) << 8) |
+ ((0xff - ucontrol->value.integer.value[1]) );
+ if (oval == nval)
+ return 0;
+ snd_ca0106_ptr_write(emu, reg, channel_id, nval);
+ return 1;
+}
+
+static int snd_ca0106_i2c_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_ca0106_i2c_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ int source_id;
+
+ source_id = kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0];
+ ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1];
+ return 0;
+}
+
+static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int ogain;
+ unsigned int ngain;
+ int source_id;
+ int change = 0;
+
+ source_id = kcontrol->private_value;
+ ogain = emu->i2c_capture_volume[source_id][0]; /* Left */
+ ngain = ucontrol->value.integer.value[0];
+ if (ngain > 0xff)
+ return -EINVAL;
+ if (ogain != ngain) {
+ if (emu->i2c_capture_source == source_id)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) );
+ emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0];
+ change = 1;
+ }
+ ogain = emu->i2c_capture_volume[source_id][1]; /* Right */
+ ngain = ucontrol->value.integer.value[1];
+ if (ngain > 0xff)
+ return -EINVAL;
+ if (ogain != ngain) {
+ if (emu->i2c_capture_source == source_id)
+ snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+ emu->i2c_capture_volume[source_id][1] = ucontrol->value.integer.value[1];
+ change = 1;
+ }
+
+ return change;
+}
+
+#define spi_mute_info snd_ctl_boolean_mono_info
+
+static int spi_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+
+ ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit);
+ return 0;
+}
+
+static int spi_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+ int ret;
+
+ ret = emu->spi_dac_reg[reg] & bit;
+ if (ucontrol->value.integer.value[0]) {
+ if (!ret) /* bit already cleared, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] &= ~bit;
+ } else {
+ if (ret) /* bit already set, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] |= bit;
+ }
+
+ ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]);
+ return ret ? -EINVAL : 1;
+}
+
+#define CA_VOLUME(xname,chid,reg) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = snd_ca0106_volume_info, \
+ .get = snd_ca0106_volume_get, \
+ .put = snd_ca0106_volume_put, \
+ .tlv = { .p = snd_ca0106_db_scale1 }, \
+ .private_value = ((chid) << 8) | (reg) \
+}
+
+static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
+ CA_VOLUME("Analog Front Playback Volume",
+ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2),
+ CA_VOLUME("Analog Rear Playback Volume",
+ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2),
+ CA_VOLUME("Analog Center/LFE Playback Volume",
+ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2),
+ CA_VOLUME("Analog Side Playback Volume",
+ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2),
+
+ CA_VOLUME("IEC958 Front Playback Volume",
+ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1),
+ CA_VOLUME("IEC958 Rear Playback Volume",
+ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1),
+ CA_VOLUME("IEC958 Center/LFE Playback Volume",
+ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1),
+ CA_VOLUME("IEC958 Unknown Playback Volume",
+ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1),
+
+ CA_VOLUME("CAPTURE feedback Playback Volume",
+ 1, CAPTURE_CONTROL),
+
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .count = 4,
+ .info = snd_ca0106_spdif_info,
+ .get = snd_ca0106_spdif_get_mask
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Playback Switch",
+ .info = snd_ca0106_shared_spdif_info,
+ .get = snd_ca0106_shared_spdif_get,
+ .put = snd_ca0106_shared_spdif_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Source Capture Enum",
+ .info = snd_ca0106_capture_source_info,
+ .get = snd_ca0106_capture_source_get,
+ .put = snd_ca0106_capture_source_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Source Capture Enum",
+ .info = snd_ca0106_i2c_capture_source_info,
+ .get = snd_ca0106_i2c_capture_source_get,
+ .put = snd_ca0106_i2c_capture_source_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .count = 4,
+ .info = snd_ca0106_spdif_info,
+ .get = snd_ca0106_spdif_get,
+ .put = snd_ca0106_spdif_put
+ },
+};
+
+#define I2C_VOLUME(xname,chid) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = snd_ca0106_i2c_volume_info, \
+ .get = snd_ca0106_i2c_volume_get, \
+ .put = snd_ca0106_i2c_volume_put, \
+ .tlv = { .p = snd_ca0106_db_scale2 }, \
+ .private_value = chid \
+}
+
+static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] __devinitdata = {
+ I2C_VOLUME("Phone Capture Volume", 0),
+ I2C_VOLUME("Mic Capture Volume", 1),
+ I2C_VOLUME("Line in Capture Volume", 2),
+ I2C_VOLUME("Aux Capture Volume", 3),
+};
+
+#define SPI_SWITCH(xname,reg,bit) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = spi_mute_info, \
+ .get = spi_mute_get, \
+ .put = spi_mute_put, \
+ .private_value = (reg<<SPI_REG_SHIFT) | (bit) \
+}
+
+static struct snd_kcontrol_new snd_ca0106_volume_spi_dac_ctls[]
+__devinitdata = {
+ SPI_SWITCH("Analog Front Playback Switch",
+ SPI_DMUTE4_REG, SPI_DMUTE4_BIT),
+ SPI_SWITCH("Analog Rear Playback Switch",
+ SPI_DMUTE0_REG, SPI_DMUTE0_BIT),
+ SPI_SWITCH("Analog Center/LFE Playback Switch",
+ SPI_DMUTE2_REG, SPI_DMUTE2_BIT),
+ SPI_SWITCH("Analog Side Playback Switch",
+ SPI_DMUTE1_REG, SPI_DMUTE1_BIT),
+};
+
+static int __devinit remove_ctl(struct snd_card *card, const char *name)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ strcpy(id.name, name);
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_remove_id(card, &id);
+}
+
+static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card, const char *name)
+{
+ struct snd_ctl_elem_id sid;
+ memset(&sid, 0, sizeof(sid));
+ /* FIXME: strcpy is bad. */
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_find_id(card, &sid);
+}
+
+static int __devinit rename_ctl(struct snd_card *card, const char *src, const char *dst)
+{
+ struct snd_kcontrol *kctl = ctl_find(card, src);
+ if (kctl) {
+ strcpy(kctl->id.name, dst);
+ return 0;
+ }
+ return -ENOENT;
+}
+
+#define ADD_CTLS(emu, ctls) \
+ do { \
+ int i, _err; \
+ for (i = 0; i < ARRAY_SIZE(ctls); i++) { \
+ _err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \
+ if (_err < 0) \
+ return _err; \
+ } \
+ } while (0)
+
+static __devinitdata
+DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 50, 1);
+
+static char *slave_vols[] __devinitdata = {
+ "Analog Front Playback Volume",
+ "Analog Rear Playback Volume",
+ "Analog Center/LFE Playback Volume",
+ "Analog Side Playback Volume",
+ "IEC958 Front Playback Volume",
+ "IEC958 Rear Playback Volume",
+ "IEC958 Center/LFE Playback Volume",
+ "IEC958 Unknown Playback Volume",
+ "CAPTURE feedback Playback Volume",
+ NULL
+};
+
+static char *slave_sws[] __devinitdata = {
+ "Analog Front Playback Switch",
+ "Analog Rear Playback Switch",
+ "Analog Center/LFE Playback Switch",
+ "Analog Side Playback Switch",
+ "IEC958 Playback Switch",
+ NULL
+};
+
+static void __devinit add_slaves(struct snd_card *card,
+ struct snd_kcontrol *master, char **list)
+{
+ for (; *list; list++) {
+ struct snd_kcontrol *slave = ctl_find(card, *list);
+ if (slave)
+ snd_ctl_add_slave(master, slave);
+ }
+}
+
+int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
+{
+ int err;
+ struct snd_card *card = emu->card;
+ char **c;
+ struct snd_kcontrol *vmaster;
+ static char *ca0106_remove_ctls[] = {
+ "Master Mono Playback Switch",
+ "Master Mono Playback Volume",
+ "3D Control - Switch",
+ "3D Control Sigmatel - Depth",
+ "PCM Playback Switch",
+ "PCM Playback Volume",
+ "CD Playback Switch",
+ "CD Playback Volume",
+ "Phone Playback Switch",
+ "Phone Playback Volume",
+ "Video Playback Switch",
+ "Video Playback Volume",
+ "PC Speaker Playback Switch",
+ "PC Speaker Playback Volume",
+ "Mono Output Select",
+ "Capture Source",
+ "Capture Switch",
+ "Capture Volume",
+ "External Amplifier",
+ "Sigmatel 4-Speaker Stereo Playback Switch",
+ "Sigmatel Surround Phase Inversion Playback ",
+ NULL
+ };
+ static char *ca0106_rename_ctls[] = {
+ "Master Playback Switch", "Capture Switch",
+ "Master Playback Volume", "Capture Volume",
+ "Line Playback Switch", "AC97 Line Capture Switch",
+ "Line Playback Volume", "AC97 Line Capture Volume",
+ "Aux Playback Switch", "AC97 Aux Capture Switch",
+ "Aux Playback Volume", "AC97 Aux Capture Volume",
+ "Mic Playback Switch", "AC97 Mic Capture Switch",
+ "Mic Playback Volume", "AC97 Mic Capture Volume",
+ "Mic Select", "AC97 Mic Select",
+ "Mic Boost (+20dB)", "AC97 Mic Boost (+20dB)",
+ NULL
+ };
+#if 1
+ for (c = ca0106_remove_ctls; *c; c++)
+ remove_ctl(card, *c);
+ for (c = ca0106_rename_ctls; *c; c += 2)
+ rename_ctl(card, c[0], c[1]);
+#endif
+
+ ADD_CTLS(emu, snd_ca0106_volume_ctls);
+ if (emu->details->i2c_adc == 1) {
+ ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls);
+ if (emu->details->gpio_type == 1)
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu));
+ else /* gpio_type == 2 */
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu));
+ if (err < 0)
+ return err;
+ }
+ if (emu->details->spi_dac == 1)
+ ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls);
+
+ /* Create virtual master controls */
+ vmaster = snd_ctl_make_virtual_master("Master Playback Volume",
+ snd_ca0106_master_db_scale);
+ if (!vmaster)
+ return -ENOMEM;
+ add_slaves(card, vmaster, slave_vols);
+
+ if (emu->details->spi_dac == 1) {
+ vmaster = snd_ctl_make_virtual_master("Master Playback Switch",
+ NULL);
+ if (!vmaster)
+ return -ENOMEM;
+ add_slaves(card, vmaster, slave_sws);
+ }
+ return 0;
+}
+
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
new file mode 100644
index 0000000..c62b7d1
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -0,0 +1,458 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.18
+ *
+ * FEATURES currently supported:
+ * See ca0106_main.c for features.
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
+ * 0.0.9
+ * Fix AC3 output.
+ * Enable S32_LE format support.
+ * 0.0.10
+ * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ * 0.0.11
+ * Add Model name recognition.
+ * 0.0.12
+ * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ * Remove redundent "voice" handling.
+ * 0.0.13
+ * Single trigger call for multi channels.
+ * 0.0.14
+ * Set limits based on what the sound card hardware can do.
+ * playback periods_min=2, periods_max=8
+ * capture hw constraints require period_size = n * 64 bytes.
+ * playback hw constraints require period_size = n * 64 bytes.
+ * 0.0.15
+ * Separate ca0106.c into separate functional .c files.
+ * 0.0.16
+ * Modified Copyright message.
+ * 0.0.17
+ * Add iec958 file in proc file system to show status of SPDIF in.
+ * 0.0.18
+ * Implement support for Line-in capture on SB Live 24bit.
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+#include <asm/io.h>
+
+#include "ca0106.h"
+
+
+#ifdef CONFIG_PROC_FS
+
+struct snd_ca0106_category_str {
+ int val;
+ const char *name;
+};
+
+static struct snd_ca0106_category_str snd_ca0106_con_category[] = {
+ { IEC958_AES1_CON_DAT, "DAT" },
+ { IEC958_AES1_CON_VCR, "VCR" },
+ { IEC958_AES1_CON_MICROPHONE, "microphone" },
+ { IEC958_AES1_CON_SYNTHESIZER, "synthesizer" },
+ { IEC958_AES1_CON_RATE_CONVERTER, "rate converter" },
+ { IEC958_AES1_CON_MIXER, "mixer" },
+ { IEC958_AES1_CON_SAMPLER, "sampler" },
+ { IEC958_AES1_CON_PCM_CODER, "PCM coder" },
+ { IEC958_AES1_CON_IEC908_CD, "CD" },
+ { IEC958_AES1_CON_NON_IEC908_CD, "non-IEC908 CD" },
+ { IEC958_AES1_CON_GENERAL, "general" },
+};
+
+
+static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 value)
+{
+ int i;
+ u32 status[4];
+ status[0] = value & 0xff;
+ status[1] = (value >> 8) & 0xff;
+ status[2] = (value >> 16) & 0xff;
+ status[3] = (value >> 24) & 0xff;
+
+ if (! (status[0] & IEC958_AES0_PROFESSIONAL)) {
+ /* consumer */
+ snd_iprintf(buffer, "Mode: consumer\n");
+ snd_iprintf(buffer, "Data: ");
+ if (!(status[0] & IEC958_AES0_NONAUDIO)) {
+ snd_iprintf(buffer, "audio\n");
+ } else {
+ snd_iprintf(buffer, "non-audio\n");
+ }
+ snd_iprintf(buffer, "Rate: ");
+ switch (status[3] & IEC958_AES3_CON_FS) {
+ case IEC958_AES3_CON_FS_44100:
+ snd_iprintf(buffer, "44100 Hz\n");
+ break;
+ case IEC958_AES3_CON_FS_48000:
+ snd_iprintf(buffer, "48000 Hz\n");
+ break;
+ case IEC958_AES3_CON_FS_32000:
+ snd_iprintf(buffer, "32000 Hz\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Copyright: ");
+ if (status[0] & IEC958_AES0_CON_NOT_COPYRIGHT) {
+ snd_iprintf(buffer, "permitted\n");
+ } else {
+ snd_iprintf(buffer, "protected\n");
+ }
+ snd_iprintf(buffer, "Emphasis: ");
+ if ((status[0] & IEC958_AES0_CON_EMPHASIS) != IEC958_AES0_CON_EMPHASIS_5015) {
+ snd_iprintf(buffer, "none\n");
+ } else {
+ snd_iprintf(buffer, "50/15us\n");
+ }
+ snd_iprintf(buffer, "Category: ");
+ for (i = 0; i < ARRAY_SIZE(snd_ca0106_con_category); i++) {
+ if ((status[1] & IEC958_AES1_CON_CATEGORY) == snd_ca0106_con_category[i].val) {
+ snd_iprintf(buffer, "%s\n", snd_ca0106_con_category[i].name);
+ break;
+ }
+ }
+ if (i >= ARRAY_SIZE(snd_ca0106_con_category)) {
+ snd_iprintf(buffer, "unknown 0x%x\n", status[1] & IEC958_AES1_CON_CATEGORY);
+ }
+ snd_iprintf(buffer, "Original: ");
+ if (status[1] & IEC958_AES1_CON_ORIGINAL) {
+ snd_iprintf(buffer, "original\n");
+ } else {
+ snd_iprintf(buffer, "1st generation\n");
+ }
+ snd_iprintf(buffer, "Clock: ");
+ switch (status[3] & IEC958_AES3_CON_CLOCK) {
+ case IEC958_AES3_CON_CLOCK_1000PPM:
+ snd_iprintf(buffer, "1000 ppm\n");
+ break;
+ case IEC958_AES3_CON_CLOCK_50PPM:
+ snd_iprintf(buffer, "50 ppm\n");
+ break;
+ case IEC958_AES3_CON_CLOCK_VARIABLE:
+ snd_iprintf(buffer, "variable pitch\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ } else {
+ snd_iprintf(buffer, "Mode: professional\n");
+ snd_iprintf(buffer, "Data: ");
+ if (!(status[0] & IEC958_AES0_NONAUDIO)) {
+ snd_iprintf(buffer, "audio\n");
+ } else {
+ snd_iprintf(buffer, "non-audio\n");
+ }
+ snd_iprintf(buffer, "Rate: ");
+ switch (status[0] & IEC958_AES0_PRO_FS) {
+ case IEC958_AES0_PRO_FS_44100:
+ snd_iprintf(buffer, "44100 Hz\n");
+ break;
+ case IEC958_AES0_PRO_FS_48000:
+ snd_iprintf(buffer, "48000 Hz\n");
+ break;
+ case IEC958_AES0_PRO_FS_32000:
+ snd_iprintf(buffer, "32000 Hz\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Rate Locked: ");
+ if (status[0] & IEC958_AES0_PRO_FREQ_UNLOCKED)
+ snd_iprintf(buffer, "no\n");
+ else
+ snd_iprintf(buffer, "yes\n");
+ snd_iprintf(buffer, "Emphasis: ");
+ switch (status[0] & IEC958_AES0_PRO_EMPHASIS) {
+ case IEC958_AES0_PRO_EMPHASIS_CCITT:
+ snd_iprintf(buffer, "CCITT J.17\n");
+ break;
+ case IEC958_AES0_PRO_EMPHASIS_NONE:
+ snd_iprintf(buffer, "none\n");
+ break;
+ case IEC958_AES0_PRO_EMPHASIS_5015:
+ snd_iprintf(buffer, "50/15us\n");
+ break;
+ case IEC958_AES0_PRO_EMPHASIS_NOTID:
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Stereophonic: ");
+ if ((status[1] & IEC958_AES1_PRO_MODE) == IEC958_AES1_PRO_MODE_STEREOPHONIC) {
+ snd_iprintf(buffer, "stereo\n");
+ } else {
+ snd_iprintf(buffer, "not indicated\n");
+ }
+ snd_iprintf(buffer, "Userbits: ");
+ switch (status[1] & IEC958_AES1_PRO_USERBITS) {
+ case IEC958_AES1_PRO_USERBITS_192:
+ snd_iprintf(buffer, "192bit\n");
+ break;
+ case IEC958_AES1_PRO_USERBITS_UDEF:
+ snd_iprintf(buffer, "user-defined\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unkown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Sample Bits: ");
+ switch (status[2] & IEC958_AES2_PRO_SBITS) {
+ case IEC958_AES2_PRO_SBITS_20:
+ snd_iprintf(buffer, "20 bit\n");
+ break;
+ case IEC958_AES2_PRO_SBITS_24:
+ snd_iprintf(buffer, "24 bit\n");
+ break;
+ case IEC958_AES2_PRO_SBITS_UDEF:
+ snd_iprintf(buffer, "user defined\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ snd_iprintf(buffer, "Word Length: ");
+ switch (status[2] & IEC958_AES2_PRO_WORDLEN) {
+ case IEC958_AES2_PRO_WORDLEN_22_18:
+ snd_iprintf(buffer, "22 bit or 18 bit\n");
+ break;
+ case IEC958_AES2_PRO_WORDLEN_23_19:
+ snd_iprintf(buffer, "23 bit or 19 bit\n");
+ break;
+ case IEC958_AES2_PRO_WORDLEN_24_20:
+ snd_iprintf(buffer, "24 bit or 20 bit\n");
+ break;
+ case IEC958_AES2_PRO_WORDLEN_20_16:
+ snd_iprintf(buffer, "20 bit or 16 bit\n");
+ break;
+ default:
+ snd_iprintf(buffer, "unknown\n");
+ break;
+ }
+ }
+}
+
+static void snd_ca0106_proc_iec958(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ u32 value;
+
+ value = snd_ca0106_ptr_read(emu, SAMPLE_RATE_TRACKER_STATUS, 0);
+ snd_iprintf(buffer, "Status: %s, %s, %s\n",
+ (value & 0x100000) ? "Rate Locked" : "Not Rate Locked",
+ (value & 0x200000) ? "SPDIF Locked" : "No SPDIF Lock",
+ (value & 0x400000) ? "Audio Valid" : "No valid audio" );
+ snd_iprintf(buffer, "Estimated sample rate: %u\n",
+ ((value & 0xfffff) * 48000) / 0x8000 );
+ if (value & 0x200000) {
+ snd_iprintf(buffer, "IEC958/SPDIF input status:\n");
+ value = snd_ca0106_ptr_read(emu, SPDIF_INPUT_STATUS, 0);
+ snd_ca0106_proc_dump_iec958(buffer, value);
+ }
+
+ snd_iprintf(buffer, "\n");
+}
+
+static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long flags;
+ char line[64];
+ u32 reg, val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(val, emu->port + (reg & 0xfffffffc));
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ }
+ }
+}
+
+static void snd_ca0106_proc_reg_read32(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "Registers:\n\n");
+ for(i = 0; i < 0x20; i+=4) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inl(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "Register %02X: %08lX\n", i, value);
+ }
+}
+
+static void snd_ca0106_proc_reg_read16(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned int value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "Registers:\n\n");
+ for(i = 0; i < 0x20; i+=2) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inw(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "Register %02X: %04X\n", i, value);
+ }
+}
+
+static void snd_ca0106_proc_reg_read8(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned int value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "Registers:\n\n");
+ for(i = 0; i < 0x20; i+=1) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inb(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "Register %02X: %02X\n", i, value);
+ }
+}
+
+static void snd_ca0106_proc_reg_read1(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long value;
+ int i,j;
+
+ snd_iprintf(buffer, "Registers\n");
+ for(i = 0; i < 0x40; i++) {
+ snd_iprintf(buffer, "%02X: ",i);
+ for (j = 0; j < 4; j++) {
+ value = snd_ca0106_ptr_read(emu, i, j);
+ snd_iprintf(buffer, "%08lX ", value);
+ }
+ snd_iprintf(buffer, "\n");
+ }
+}
+
+static void snd_ca0106_proc_reg_read2(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ unsigned long value;
+ int i,j;
+
+ snd_iprintf(buffer, "Registers\n");
+ for(i = 0x40; i < 0x80; i++) {
+ snd_iprintf(buffer, "%02X: ",i);
+ for (j = 0; j < 4; j++) {
+ value = snd_ca0106_ptr_read(emu, i, j);
+ snd_iprintf(buffer, "%08lX ", value);
+ }
+ snd_iprintf(buffer, "\n");
+ }
+}
+
+static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ char line[64];
+ unsigned int reg, channel_id , val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
+ continue;
+ if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) )
+ snd_ca0106_ptr_write(emu, reg, channel_id, val);
+ }
+}
+
+static void snd_ca0106_proc_i2c_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ca0106 *emu = entry->private_data;
+ char line[64];
+ unsigned int reg, val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ if ((reg <= 0x7f) || (val <= 0x1ff)) {
+ snd_ca0106_i2c_write(emu, reg, val);
+ }
+ }
+}
+
+int __devinit snd_ca0106_proc_init(struct snd_ca0106 * emu)
+{
+ struct snd_info_entry *entry;
+
+ if(! snd_card_proc_new(emu->card, "iec958", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_iec958);
+ if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32);
+ entry->c.text.write = snd_ca0106_proc_reg_write32;
+ entry->mode |= S_IWUSR;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16);
+ if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read8);
+ if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1);
+ entry->c.text.write = snd_ca0106_proc_reg_write;
+ entry->mode |= S_IWUSR;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) {
+ entry->c.text.write = snd_ca0106_proc_i2c_write;
+ entry->private_data = emu;
+ entry->mode |= S_IWUSR;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry))
+ snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2);
+ return 0;
+}
+
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c
new file mode 100644
index 0000000..c788511
--- /dev/null
+++ b/sound/pci/ca0106/ca_midi.c
@@ -0,0 +1,316 @@
+/*
+ * Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de>
+ * Creative Audio MIDI, for the CA0106 Driver
+ * Version: 0.0.1
+ *
+ * Changelog:
+ * Implementation is based on mpu401 and emu10k1x and
+ * tested with ca0106.
+ * mpu401: Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * emu10k1x: Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ */
+
+#include <linux/spinlock.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+
+#include "ca_midi.h"
+
+#define ca_midi_write_data(midi, data) midi->write(midi, data, 0)
+#define ca_midi_write_cmd(midi, data) midi->write(midi, data, 1)
+#define ca_midi_read_data(midi) midi->read(midi, 0)
+#define ca_midi_read_stat(midi) midi->read(midi, 1)
+#define ca_midi_input_avail(midi) (!(ca_midi_read_stat(midi) & midi->input_avail))
+#define ca_midi_output_ready(midi) (!(ca_midi_read_stat(midi) & midi->output_ready))
+
+static void ca_midi_clear_rx(struct snd_ca_midi *midi)
+{
+ int timeout = 100000;
+ for (; timeout > 0 && ca_midi_input_avail(midi); timeout--)
+ ca_midi_read_data(midi);
+#ifdef CONFIG_SND_DEBUG
+ if (timeout <= 0)
+ snd_printk(KERN_ERR "ca_midi_clear_rx: timeout (status = 0x%x)\n",
+ ca_midi_read_stat(midi));
+#endif
+}
+
+static void ca_midi_interrupt(struct snd_ca_midi *midi, unsigned int status)
+{
+ unsigned char byte;
+
+ if (midi->rmidi == NULL) {
+ midi->interrupt_disable(midi,midi->tx_enable | midi->rx_enable);
+ return;
+ }
+
+ spin_lock(&midi->input_lock);
+ if ((status & midi->ipr_rx) && ca_midi_input_avail(midi)) {
+ if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+ ca_midi_clear_rx(midi);
+ } else {
+ byte = ca_midi_read_data(midi);
+ if(midi->substream_input)
+ snd_rawmidi_receive(midi->substream_input, &byte, 1);
+
+
+ }
+ }
+ spin_unlock(&midi->input_lock);
+
+ spin_lock(&midi->output_lock);
+ if ((status & midi->ipr_tx) && ca_midi_output_ready(midi)) {
+ if (midi->substream_output &&
+ snd_rawmidi_transmit(midi->substream_output, &byte, 1) == 1) {
+ ca_midi_write_data(midi, byte);
+ } else {
+ midi->interrupt_disable(midi,midi->tx_enable);
+ }
+ }
+ spin_unlock(&midi->output_lock);
+
+}
+
+static void ca_midi_cmd(struct snd_ca_midi *midi, unsigned char cmd, int ack)
+{
+ unsigned long flags;
+ int timeout, ok;
+
+ spin_lock_irqsave(&midi->input_lock, flags);
+ ca_midi_write_data(midi, 0x00);
+ /* ca_midi_clear_rx(midi); */
+
+ ca_midi_write_cmd(midi, cmd);
+ if (ack) {
+ ok = 0;
+ timeout = 10000;
+ while (!ok && timeout-- > 0) {
+ if (ca_midi_input_avail(midi)) {
+ if (ca_midi_read_data(midi) == midi->ack)
+ ok = 1;
+ }
+ }
+ if (!ok && ca_midi_read_data(midi) == midi->ack)
+ ok = 1;
+ } else {
+ ok = 1;
+ }
+ spin_unlock_irqrestore(&midi->input_lock, flags);
+ if (!ok)
+ snd_printk(KERN_ERR "ca_midi_cmd: 0x%x failed at 0x%x (status = 0x%x, data = 0x%x)!!!\n",
+ cmd,
+ midi->get_dev_id_port(midi->dev_id),
+ ca_midi_read_stat(midi),
+ ca_midi_read_data(midi));
+}
+
+static int ca_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= CA_MIDI_MODE_INPUT;
+ midi->substream_input = substream;
+ if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 1);
+ ca_midi_cmd(midi, midi->enter_uart, 1);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static int ca_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= CA_MIDI_MODE_OUTPUT;
+ midi->substream_output = substream;
+ if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 1);
+ ca_midi_cmd(midi, midi->enter_uart, 1);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static int ca_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->interrupt_disable(midi,midi->rx_enable);
+ midi->midi_mode &= ~CA_MIDI_MODE_INPUT;
+ midi->substream_input = NULL;
+ if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static int ca_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return -ENXIO;
+
+ spin_lock_irqsave(&midi->open_lock, flags);
+
+ midi->interrupt_disable(midi,midi->tx_enable);
+ midi->midi_mode &= ~CA_MIDI_MODE_OUTPUT;
+ midi->substream_output = NULL;
+
+ if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ ca_midi_cmd(midi, midi->reset, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+}
+
+static void ca_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return;
+
+ if (up) {
+ midi->interrupt_enable(midi,midi->rx_enable);
+ } else {
+ midi->interrupt_disable(midi, midi->rx_enable);
+ }
+}
+
+static void ca_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_ca_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!midi->dev_id))
+ return;
+
+ if (up) {
+ int max = 4;
+ unsigned char byte;
+
+ spin_lock_irqsave(&midi->output_lock, flags);
+
+ /* try to send some amount of bytes here before interrupts */
+ while (max > 0) {
+ if (ca_midi_output_ready(midi)) {
+ if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT) ||
+ snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ /* no more data */
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ return;
+ }
+ ca_midi_write_data(midi, byte);
+ max--;
+ } else {
+ break;
+ }
+ }
+
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ midi->interrupt_enable(midi,midi->tx_enable);
+
+ } else {
+ midi->interrupt_disable(midi,midi->tx_enable);
+ }
+}
+
+static struct snd_rawmidi_ops ca_midi_output =
+{
+ .open = ca_midi_output_open,
+ .close = ca_midi_output_close,
+ .trigger = ca_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops ca_midi_input =
+{
+ .open = ca_midi_input_open,
+ .close = ca_midi_input_close,
+ .trigger = ca_midi_input_trigger,
+};
+
+static void ca_midi_free(struct snd_ca_midi *midi)
+{
+ midi->interrupt = NULL;
+ midi->interrupt_enable = NULL;
+ midi->interrupt_disable = NULL;
+ midi->read = NULL;
+ midi->write = NULL;
+ midi->get_dev_id_card = NULL;
+ midi->get_dev_id_port = NULL;
+ midi->rmidi = NULL;
+}
+
+static void ca_rmidi_free(struct snd_rawmidi *rmidi)
+{
+ ca_midi_free(rmidi->private_data);
+}
+
+int __devinit ca_midi_init(void *dev_id, struct snd_ca_midi *midi, int device, char *name)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if ((err = snd_rawmidi_new(midi->get_dev_id_card(midi->dev_id), name, device, 1, 1, &rmidi)) < 0)
+ return err;
+
+ midi->dev_id = dev_id;
+ midi->interrupt = ca_midi_interrupt;
+
+ spin_lock_init(&midi->open_lock);
+ spin_lock_init(&midi->input_lock);
+ spin_lock_init(&midi->output_lock);
+
+ strcpy(rmidi->name, name);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &ca_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &ca_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = midi;
+ rmidi->private_free = ca_rmidi_free;
+
+ midi->rmidi = rmidi;
+ return 0;
+}
+
diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h
new file mode 100644
index 0000000..922ed3e
--- /dev/null
+++ b/sound/pci/ca0106/ca_midi.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de>
+ * Creative Audio MIDI, for the CA0106 Driver
+ * Version: 0.0.1
+ *
+ * Changelog:
+ * See ca_midi.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/spinlock.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
+
+#define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT
+#define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT
+
+struct snd_ca_midi {
+
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream_input;
+ struct snd_rawmidi_substream *substream_output;
+
+ void *dev_id;
+
+ spinlock_t input_lock;
+ spinlock_t output_lock;
+ spinlock_t open_lock;
+
+ unsigned int channel;
+
+ unsigned int midi_mode;
+ int port;
+ int tx_enable, rx_enable;
+ int ipr_tx, ipr_rx;
+
+ int input_avail, output_ready;
+ int ack, reset, enter_uart;
+
+ void (*interrupt)(struct snd_ca_midi *midi, unsigned int status);
+ void (*interrupt_enable)(struct snd_ca_midi *midi, int intr);
+ void (*interrupt_disable)(struct snd_ca_midi *midi, int intr);
+
+ unsigned char (*read)(struct snd_ca_midi *midi, int idx);
+ void (*write)(struct snd_ca_midi *midi, int data, int idx);
+
+ /* get info from dev_id */
+ struct snd_card *(*get_dev_id_card)(void *dev_id);
+ int (*get_dev_id_port)(void *dev_id);
+};
+
+int ca_midi_init(void *card, struct snd_ca_midi *midi, int device, char *name);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
new file mode 100644
index 0000000..1a74ca6
--- /dev/null
+++ b/sound/pci/cmipci.c
@@ -0,0 +1,3421 @@
+/*
+ * Driver for C-Media CMI8338 and 8738 PCI soundcards.
+ * Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/* Does not work. Warning may block system in capture mode */
+/* #define USE_VAR48KRATE */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/sb.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("C-Media CMI8x38 PCI");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8738},"
+ "{C-Media,CMI8738B},"
+ "{C-Media,CMI8338A},"
+ "{C-Media,CMI8338B}}");
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK 1
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */
+static long mpu_port[SNDRV_CARDS];
+static long fm_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1};
+static int soft_ac3[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)]=1};
+#ifdef SUPPORT_JOYSTICK
+static int joystick_port[SNDRV_CARDS];
+#endif
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for C-Media PCI soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for C-Media PCI soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable C-Media PCI soundcard.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port.");
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM port.");
+module_param_array(soft_ac3, bool, NULL, 0444);
+MODULE_PARM_DESC(soft_ac3, "Sofware-conversion of raw SPDIF packets (model 033 only).");
+#ifdef SUPPORT_JOYSTICK
+module_param_array(joystick_port, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_port, "Joystick port address.");
+#endif
+
+/*
+ * CM8x38 registers definition
+ */
+
+#define CM_REG_FUNCTRL0 0x00
+#define CM_RST_CH1 0x00080000
+#define CM_RST_CH0 0x00040000
+#define CM_CHEN1 0x00020000 /* ch1: enable */
+#define CM_CHEN0 0x00010000 /* ch0: enable */
+#define CM_PAUSE1 0x00000008 /* ch1: pause */
+#define CM_PAUSE0 0x00000004 /* ch0: pause */
+#define CM_CHADC1 0x00000002 /* ch1, 0:playback, 1:record */
+#define CM_CHADC0 0x00000001 /* ch0, 0:playback, 1:record */
+
+#define CM_REG_FUNCTRL1 0x04
+#define CM_DSFC_MASK 0x0000E000 /* channel 1 (DAC?) sampling frequency */
+#define CM_DSFC_SHIFT 13
+#define CM_ASFC_MASK 0x00001C00 /* channel 0 (ADC?) sampling frequency */
+#define CM_ASFC_SHIFT 10
+#define CM_SPDF_1 0x00000200 /* SPDIF IN/OUT at channel B */
+#define CM_SPDF_0 0x00000100 /* SPDIF OUT only channel A */
+#define CM_SPDFLOOP 0x00000080 /* ext. SPDIIF/IN -> OUT loopback */
+#define CM_SPDO2DAC 0x00000040 /* SPDIF/OUT can be heard from internal DAC */
+#define CM_INTRM 0x00000020 /* master control block (MCB) interrupt enabled */
+#define CM_BREQ 0x00000010 /* bus master enabled */
+#define CM_VOICE_EN 0x00000008 /* legacy voice (SB16,FM) */
+#define CM_UART_EN 0x00000004 /* legacy UART */
+#define CM_JYSTK_EN 0x00000002 /* legacy joystick */
+#define CM_ZVPORT 0x00000001 /* ZVPORT */
+
+#define CM_REG_CHFORMAT 0x08
+
+#define CM_CHB3D5C 0x80000000 /* 5,6 channels */
+#define CM_FMOFFSET2 0x40000000 /* initial FM PCM offset 2 when Fmute=1 */
+#define CM_CHB3D 0x20000000 /* 4 channels */
+
+#define CM_CHIP_MASK1 0x1f000000
+#define CM_CHIP_037 0x01000000
+#define CM_SETLAT48 0x00800000 /* set latency timer 48h */
+#define CM_EDGEIRQ 0x00400000 /* emulated edge trigger legacy IRQ */
+#define CM_SPD24SEL39 0x00200000 /* 24-bit spdif: model 039 */
+#define CM_AC3EN1 0x00100000 /* enable AC3: model 037 */
+#define CM_SPDIF_SELECT1 0x00080000 /* for model <= 037 ? */
+#define CM_SPD24SEL 0x00020000 /* 24bit spdif: model 037 */
+/* #define CM_SPDIF_INVERSE 0x00010000 */ /* ??? */
+
+#define CM_ADCBITLEN_MASK 0x0000C000
+#define CM_ADCBITLEN_16 0x00000000
+#define CM_ADCBITLEN_15 0x00004000
+#define CM_ADCBITLEN_14 0x00008000
+#define CM_ADCBITLEN_13 0x0000C000
+
+#define CM_ADCDACLEN_MASK 0x00003000 /* model 037 */
+#define CM_ADCDACLEN_060 0x00000000
+#define CM_ADCDACLEN_066 0x00001000
+#define CM_ADCDACLEN_130 0x00002000
+#define CM_ADCDACLEN_280 0x00003000
+
+#define CM_ADCDLEN_MASK 0x00003000 /* model 039 */
+#define CM_ADCDLEN_ORIGINAL 0x00000000
+#define CM_ADCDLEN_EXTRA 0x00001000
+#define CM_ADCDLEN_24K 0x00002000
+#define CM_ADCDLEN_WEIGHT 0x00003000
+
+#define CM_CH1_SRATE_176K 0x00000800
+#define CM_CH1_SRATE_96K 0x00000800 /* model 055? */
+#define CM_CH1_SRATE_88K 0x00000400
+#define CM_CH0_SRATE_176K 0x00000200
+#define CM_CH0_SRATE_96K 0x00000200 /* model 055? */
+#define CM_CH0_SRATE_88K 0x00000100
+#define CM_CH0_SRATE_128K 0x00000300
+#define CM_CH0_SRATE_MASK 0x00000300
+
+#define CM_SPDIF_INVERSE2 0x00000080 /* model 055? */
+#define CM_DBLSPDS 0x00000040 /* double SPDIF sample rate 88.2/96 */
+#define CM_POLVALID 0x00000020 /* inverse SPDIF/IN valid bit */
+#define CM_SPDLOCKED 0x00000010
+
+#define CM_CH1FMT_MASK 0x0000000C /* bit 3: 16 bits, bit 2: stereo */
+#define CM_CH1FMT_SHIFT 2
+#define CM_CH0FMT_MASK 0x00000003 /* bit 1: 16 bits, bit 0: stereo */
+#define CM_CH0FMT_SHIFT 0
+
+#define CM_REG_INT_HLDCLR 0x0C
+#define CM_CHIP_MASK2 0xff000000
+#define CM_CHIP_8768 0x20000000
+#define CM_CHIP_055 0x08000000
+#define CM_CHIP_039 0x04000000
+#define CM_CHIP_039_6CH 0x01000000
+#define CM_UNKNOWN_INT_EN 0x00080000 /* ? */
+#define CM_TDMA_INT_EN 0x00040000
+#define CM_CH1_INT_EN 0x00020000
+#define CM_CH0_INT_EN 0x00010000
+
+#define CM_REG_INT_STATUS 0x10
+#define CM_INTR 0x80000000
+#define CM_VCO 0x08000000 /* Voice Control? CMI8738 */
+#define CM_MCBINT 0x04000000 /* Master Control Block abort cond.? */
+#define CM_UARTINT 0x00010000
+#define CM_LTDMAINT 0x00008000
+#define CM_HTDMAINT 0x00004000
+#define CM_XDO46 0x00000080 /* Modell 033? Direct programming EEPROM (read data register) */
+#define CM_LHBTOG 0x00000040 /* High/Low status from DMA ctrl register */
+#define CM_LEG_HDMA 0x00000020 /* Legacy is in High DMA channel */
+#define CM_LEG_STEREO 0x00000010 /* Legacy is in Stereo mode */
+#define CM_CH1BUSY 0x00000008
+#define CM_CH0BUSY 0x00000004
+#define CM_CHINT1 0x00000002
+#define CM_CHINT0 0x00000001
+
+#define CM_REG_LEGACY_CTRL 0x14
+#define CM_NXCHG 0x80000000 /* don't map base reg dword->sample */
+#define CM_VMPU_MASK 0x60000000 /* MPU401 i/o port address */
+#define CM_VMPU_330 0x00000000
+#define CM_VMPU_320 0x20000000
+#define CM_VMPU_310 0x40000000
+#define CM_VMPU_300 0x60000000
+#define CM_ENWR8237 0x10000000 /* enable bus master to write 8237 base reg */
+#define CM_VSBSEL_MASK 0x0C000000 /* SB16 base address */
+#define CM_VSBSEL_220 0x00000000
+#define CM_VSBSEL_240 0x04000000
+#define CM_VSBSEL_260 0x08000000
+#define CM_VSBSEL_280 0x0C000000
+#define CM_FMSEL_MASK 0x03000000 /* FM OPL3 base address */
+#define CM_FMSEL_388 0x00000000
+#define CM_FMSEL_3C8 0x01000000
+#define CM_FMSEL_3E0 0x02000000
+#define CM_FMSEL_3E8 0x03000000
+#define CM_ENSPDOUT 0x00800000 /* enable XSPDIF/OUT to I/O interface */
+#define CM_SPDCOPYRHT 0x00400000 /* spdif in/out copyright bit */
+#define CM_DAC2SPDO 0x00200000 /* enable wave+fm_midi -> SPDIF/OUT */
+#define CM_INVIDWEN 0x00100000 /* internal vendor ID write enable, model 039? */
+#define CM_SETRETRY 0x00100000 /* 0: legacy i/o wait (default), 1: legacy i/o bus retry */
+#define CM_C_EEACCESS 0x00080000 /* direct programming eeprom regs */
+#define CM_C_EECS 0x00040000
+#define CM_C_EEDI46 0x00020000
+#define CM_C_EECK46 0x00010000
+#define CM_CHB3D6C 0x00008000 /* 5.1 channels support */
+#define CM_CENTR2LIN 0x00004000 /* line-in as center out */
+#define CM_BASE2LIN 0x00002000 /* line-in as bass out */
+#define CM_EXBASEN 0x00001000 /* external bass input enable */
+
+#define CM_REG_MISC_CTRL 0x18
+#define CM_PWD 0x80000000 /* power down */
+#define CM_RESET 0x40000000
+#define CM_SFIL_MASK 0x30000000 /* filter control at front end DAC, model 037? */
+#define CM_VMGAIN 0x10000000 /* analog master amp +6dB, model 039? */
+#define CM_TXVX 0x08000000 /* model 037? */
+#define CM_N4SPK3D 0x04000000 /* copy front to rear */
+#define CM_SPDO5V 0x02000000 /* 5V spdif output (1 = 0.5v (coax)) */
+#define CM_SPDIF48K 0x01000000 /* write */
+#define CM_SPATUS48K 0x01000000 /* read */
+#define CM_ENDBDAC 0x00800000 /* enable double dac */
+#define CM_XCHGDAC 0x00400000 /* 0: front=ch0, 1: front=ch1 */
+#define CM_SPD32SEL 0x00200000 /* 0: 16bit SPDIF, 1: 32bit */
+#define CM_SPDFLOOPI 0x00100000 /* int. SPDIF-OUT -> int. IN */
+#define CM_FM_EN 0x00080000 /* enable legacy FM */
+#define CM_AC3EN2 0x00040000 /* enable AC3: model 039 */
+#define CM_ENWRASID 0x00010000 /* choose writable internal SUBID (audio) */
+#define CM_VIDWPDSB 0x00010000 /* model 037? */
+#define CM_SPDF_AC97 0x00008000 /* 0: SPDIF/OUT 44.1K, 1: 48K */
+#define CM_MASK_EN 0x00004000 /* activate channel mask on legacy DMA */
+#define CM_ENWRMSID 0x00002000 /* choose writable internal SUBID (modem) */
+#define CM_VIDWPPRT 0x00002000 /* model 037? */
+#define CM_SFILENB 0x00001000 /* filter stepping at front end DAC, model 037? */
+#define CM_MMODE_MASK 0x00000E00 /* model DAA interface mode */
+#define CM_SPDIF_SELECT2 0x00000100 /* for model > 039 ? */
+#define CM_ENCENTER 0x00000080
+#define CM_FLINKON 0x00000040 /* force modem link detection on, model 037 */
+#define CM_MUTECH1 0x00000040 /* mute PCI ch1 to DAC */
+#define CM_FLINKOFF 0x00000020 /* force modem link detection off, model 037 */
+#define CM_MIDSMP 0x00000010 /* 1/2 interpolation at front end DAC */
+#define CM_UPDDMA_MASK 0x0000000C /* TDMA position update notification */
+#define CM_UPDDMA_2048 0x00000000
+#define CM_UPDDMA_1024 0x00000004
+#define CM_UPDDMA_512 0x00000008
+#define CM_UPDDMA_256 0x0000000C
+#define CM_TWAIT_MASK 0x00000003 /* model 037 */
+#define CM_TWAIT1 0x00000002 /* FM i/o cycle, 0: 48, 1: 64 PCICLKs */
+#define CM_TWAIT0 0x00000001 /* i/o cycle, 0: 4, 1: 6 PCICLKs */
+
+#define CM_REG_TDMA_POSITION 0x1C
+#define CM_TDMA_CNT_MASK 0xFFFF0000 /* current byte/word count */
+#define CM_TDMA_ADR_MASK 0x0000FFFF /* current address */
+
+ /* byte */
+#define CM_REG_MIXER0 0x20
+#define CM_REG_SBVR 0x20 /* write: sb16 version */
+#define CM_REG_DEV 0x20 /* read: hardware device version */
+
+#define CM_REG_MIXER21 0x21
+#define CM_UNKNOWN_21_MASK 0x78 /* ? */
+#define CM_X_ADPCM 0x04 /* SB16 ADPCM enable */
+#define CM_PROINV 0x02 /* SBPro left/right channel switching */
+#define CM_X_SB16 0x01 /* SB16 compatible */
+
+#define CM_REG_SB16_DATA 0x22
+#define CM_REG_SB16_ADDR 0x23
+
+#define CM_REFFREQ_XIN (315*1000*1000)/22 /* 14.31818 Mhz reference clock frequency pin XIN */
+#define CM_ADCMULT_XIN 512 /* Guessed (487 best for 44.1kHz, not for 88/176kHz) */
+#define CM_TOLERANCE_RATE 0.001 /* Tolerance sample rate pitch (1000ppm) */
+#define CM_MAXIMUM_RATE 80000000 /* Note more than 80MHz */
+
+#define CM_REG_MIXER1 0x24
+#define CM_FMMUTE 0x80 /* mute FM */
+#define CM_FMMUTE_SHIFT 7
+#define CM_WSMUTE 0x40 /* mute PCM */
+#define CM_WSMUTE_SHIFT 6
+#define CM_REAR2LIN 0x20 /* lin-in -> rear line out */
+#define CM_REAR2LIN_SHIFT 5
+#define CM_REAR2FRONT 0x10 /* exchange rear/front */
+#define CM_REAR2FRONT_SHIFT 4
+#define CM_WAVEINL 0x08 /* digital wave rec. left chan */
+#define CM_WAVEINL_SHIFT 3
+#define CM_WAVEINR 0x04 /* digical wave rec. right */
+#define CM_WAVEINR_SHIFT 2
+#define CM_X3DEN 0x02 /* 3D surround enable */
+#define CM_X3DEN_SHIFT 1
+#define CM_CDPLAY 0x01 /* enable SPDIF/IN PCM -> DAC */
+#define CM_CDPLAY_SHIFT 0
+
+#define CM_REG_MIXER2 0x25
+#define CM_RAUXREN 0x80 /* AUX right capture */
+#define CM_RAUXREN_SHIFT 7
+#define CM_RAUXLEN 0x40 /* AUX left capture */
+#define CM_RAUXLEN_SHIFT 6
+#define CM_VAUXRM 0x20 /* AUX right mute */
+#define CM_VAUXRM_SHIFT 5
+#define CM_VAUXLM 0x10 /* AUX left mute */
+#define CM_VAUXLM_SHIFT 4
+#define CM_VADMIC_MASK 0x0e /* mic gain level (0-3) << 1 */
+#define CM_VADMIC_SHIFT 1
+#define CM_MICGAINZ 0x01 /* mic boost */
+#define CM_MICGAINZ_SHIFT 0
+
+#define CM_REG_MIXER3 0x24
+#define CM_REG_AUX_VOL 0x26
+#define CM_VAUXL_MASK 0xf0
+#define CM_VAUXR_MASK 0x0f
+
+#define CM_REG_MISC 0x27
+#define CM_UNKNOWN_27_MASK 0xd8 /* ? */
+#define CM_XGPO1 0x20
+// #define CM_XGPBIO 0x04
+#define CM_MIC_CENTER_LFE 0x04 /* mic as center/lfe out? (model 039 or later?) */
+#define CM_SPDIF_INVERSE 0x04 /* spdif input phase inverse (model 037) */
+#define CM_SPDVALID 0x02 /* spdif input valid check */
+#define CM_DMAUTO 0x01 /* SB16 DMA auto detect */
+
+#define CM_REG_AC97 0x28 /* hmmm.. do we have ac97 link? */
+/*
+ * For CMI-8338 (0x28 - 0x2b) .. is this valid for CMI-8738
+ * or identical with AC97 codec?
+ */
+#define CM_REG_EXTERN_CODEC CM_REG_AC97
+
+/*
+ * MPU401 pci port index address 0x40 - 0x4f (CMI-8738 spec ver. 0.6)
+ */
+#define CM_REG_MPU_PCI 0x40
+
+/*
+ * FM pci port index address 0x50 - 0x5f (CMI-8738 spec ver. 0.6)
+ */
+#define CM_REG_FM_PCI 0x50
+
+/*
+ * access from SB-mixer port
+ */
+#define CM_REG_EXTENT_IND 0xf0
+#define CM_VPHONE_MASK 0xe0 /* Phone volume control (0-3) << 5 */
+#define CM_VPHONE_SHIFT 5
+#define CM_VPHOM 0x10 /* Phone mute control */
+#define CM_VSPKM 0x08 /* Speaker mute control, default high */
+#define CM_RLOOPREN 0x04 /* Rec. R-channel enable */
+#define CM_RLOOPLEN 0x02 /* Rec. L-channel enable */
+#define CM_VADMIC3 0x01 /* Mic record boost */
+
+/*
+ * CMI-8338 spec ver 0.5 (this is not valid for CMI-8738):
+ * the 8 registers 0xf8 - 0xff are used for programming m/n counter by the PLL
+ * unit (readonly?).
+ */
+#define CM_REG_PLL 0xf8
+
+/*
+ * extended registers
+ */
+#define CM_REG_CH0_FRAME1 0x80 /* write: base address */
+#define CM_REG_CH0_FRAME2 0x84 /* read: current address */
+#define CM_REG_CH1_FRAME1 0x88 /* 0-15: count of samples at bus master; buffer size */
+#define CM_REG_CH1_FRAME2 0x8C /* 16-31: count of samples at codec; fragment size */
+
+#define CM_REG_EXT_MISC 0x90
+#define CM_ADC48K44K 0x10000000 /* ADC parameters group, 0: 44k, 1: 48k */
+#define CM_CHB3D8C 0x00200000 /* 7.1 channels support */
+#define CM_SPD32FMT 0x00100000 /* SPDIF/IN 32k sample rate */
+#define CM_ADC2SPDIF 0x00080000 /* ADC output to SPDIF/OUT */
+#define CM_SHAREADC 0x00040000 /* DAC in ADC as Center/LFE */
+#define CM_REALTCMP 0x00020000 /* monitor the CMPL/CMPR of ADC */
+#define CM_INVLRCK 0x00010000 /* invert ZVPORT's LRCK */
+#define CM_UNKNOWN_90_MASK 0x0000FFFF /* ? */
+
+/*
+ * size of i/o region
+ */
+#define CM_EXTENT_CODEC 0x100
+#define CM_EXTENT_MIDI 0x2
+#define CM_EXTENT_SYNTH 0x4
+
+
+/*
+ * channels for playback / capture
+ */
+#define CM_CH_PLAY 0
+#define CM_CH_CAPT 1
+
+/*
+ * flags to check device open/close
+ */
+#define CM_OPEN_NONE 0
+#define CM_OPEN_CH_MASK 0x01
+#define CM_OPEN_DAC 0x10
+#define CM_OPEN_ADC 0x20
+#define CM_OPEN_SPDIF 0x40
+#define CM_OPEN_MCHAN 0x80
+#define CM_OPEN_PLAYBACK (CM_CH_PLAY | CM_OPEN_DAC)
+#define CM_OPEN_PLAYBACK2 (CM_CH_CAPT | CM_OPEN_DAC)
+#define CM_OPEN_PLAYBACK_MULTI (CM_CH_PLAY | CM_OPEN_DAC | CM_OPEN_MCHAN)
+#define CM_OPEN_CAPTURE (CM_CH_CAPT | CM_OPEN_ADC)
+#define CM_OPEN_SPDIF_PLAYBACK (CM_CH_PLAY | CM_OPEN_DAC | CM_OPEN_SPDIF)
+#define CM_OPEN_SPDIF_CAPTURE (CM_CH_CAPT | CM_OPEN_ADC | CM_OPEN_SPDIF)
+
+
+#if CM_CH_PLAY == 1
+#define CM_PLAYBACK_SRATE_176K CM_CH1_SRATE_176K
+#define CM_PLAYBACK_SPDF CM_SPDF_1
+#define CM_CAPTURE_SPDF CM_SPDF_0
+#else
+#define CM_PLAYBACK_SRATE_176K CM_CH0_SRATE_176K
+#define CM_PLAYBACK_SPDF CM_SPDF_0
+#define CM_CAPTURE_SPDF CM_SPDF_1
+#endif
+
+
+/*
+ * driver data
+ */
+
+struct cmipci_pcm {
+ struct snd_pcm_substream *substream;
+ u8 running; /* dac/adc running? */
+ u8 fmt; /* format bits */
+ u8 is_dac;
+ u8 needs_silencing;
+ unsigned int dma_size; /* in frames */
+ unsigned int shift;
+ unsigned int ch; /* channel (0/1) */
+ unsigned int offset; /* physical address of the buffer */
+};
+
+/* mixer elements toggled/resumed during ac3 playback */
+struct cmipci_mixer_auto_switches {
+ const char *name; /* switch to toggle */
+ int toggle_on; /* value to change when ac3 mode */
+};
+static const struct cmipci_mixer_auto_switches cm_saved_mixer[] = {
+ {"PCM Playback Switch", 0},
+ {"IEC958 Output Switch", 1},
+ {"IEC958 Mix Analog", 0},
+ // {"IEC958 Out To DAC", 1}, // no longer used
+ {"IEC958 Loop", 0},
+};
+#define CM_SAVED_MIXERS ARRAY_SIZE(cm_saved_mixer)
+
+struct cmipci {
+ struct snd_card *card;
+
+ struct pci_dev *pci;
+ unsigned int device; /* device ID */
+ int irq;
+
+ unsigned long iobase;
+ unsigned int ctrl; /* FUNCTRL0 current value */
+
+ struct snd_pcm *pcm; /* DAC/ADC PCM */
+ struct snd_pcm *pcm2; /* 2nd DAC */
+ struct snd_pcm *pcm_spdif; /* SPDIF */
+
+ int chip_version;
+ int max_channels;
+ unsigned int can_ac3_sw: 1;
+ unsigned int can_ac3_hw: 1;
+ unsigned int can_multi_ch: 1;
+ unsigned int can_96k: 1; /* samplerate above 48k */
+ unsigned int do_soft_ac3: 1;
+
+ unsigned int spdif_playback_avail: 1; /* spdif ready? */
+ unsigned int spdif_playback_enabled: 1; /* spdif switch enabled? */
+ int spdif_counter; /* for software AC3 */
+
+ unsigned int dig_status;
+ unsigned int dig_pcm_status;
+
+ struct snd_pcm_hardware *hw_info[3]; /* for playbacks */
+
+ int opened[2]; /* open mode */
+ struct mutex open_mutex;
+
+ unsigned int mixer_insensitive: 1;
+ struct snd_kcontrol *mixer_res_ctl[CM_SAVED_MIXERS];
+ int mixer_res_status[CM_SAVED_MIXERS];
+
+ struct cmipci_pcm channel[2]; /* ch0 - DAC, ch1 - ADC or 2nd DAC */
+
+ /* external MIDI */
+ struct snd_rawmidi *rmidi;
+
+#ifdef SUPPORT_JOYSTICK
+ struct gameport *gameport;
+#endif
+
+ spinlock_t reg_lock;
+
+#ifdef CONFIG_PM
+ unsigned int saved_regs[0x20];
+ unsigned char saved_mixers[0x20];
+#endif
+};
+
+
+/* read/write operations for dword register */
+static inline void snd_cmipci_write(struct cmipci *cm, unsigned int cmd, unsigned int data)
+{
+ outl(data, cm->iobase + cmd);
+}
+
+static inline unsigned int snd_cmipci_read(struct cmipci *cm, unsigned int cmd)
+{
+ return inl(cm->iobase + cmd);
+}
+
+/* read/write operations for word register */
+static inline void snd_cmipci_write_w(struct cmipci *cm, unsigned int cmd, unsigned short data)
+{
+ outw(data, cm->iobase + cmd);
+}
+
+static inline unsigned short snd_cmipci_read_w(struct cmipci *cm, unsigned int cmd)
+{
+ return inw(cm->iobase + cmd);
+}
+
+/* read/write operations for byte register */
+static inline void snd_cmipci_write_b(struct cmipci *cm, unsigned int cmd, unsigned char data)
+{
+ outb(data, cm->iobase + cmd);
+}
+
+static inline unsigned char snd_cmipci_read_b(struct cmipci *cm, unsigned int cmd)
+{
+ return inb(cm->iobase + cmd);
+}
+
+/* bit operations for dword register */
+static int snd_cmipci_set_bit(struct cmipci *cm, unsigned int cmd, unsigned int flag)
+{
+ unsigned int val, oval;
+ val = oval = inl(cm->iobase + cmd);
+ val |= flag;
+ if (val == oval)
+ return 0;
+ outl(val, cm->iobase + cmd);
+ return 1;
+}
+
+static int snd_cmipci_clear_bit(struct cmipci *cm, unsigned int cmd, unsigned int flag)
+{
+ unsigned int val, oval;
+ val = oval = inl(cm->iobase + cmd);
+ val &= ~flag;
+ if (val == oval)
+ return 0;
+ outl(val, cm->iobase + cmd);
+ return 1;
+}
+
+/* bit operations for byte register */
+static int snd_cmipci_set_bit_b(struct cmipci *cm, unsigned int cmd, unsigned char flag)
+{
+ unsigned char val, oval;
+ val = oval = inb(cm->iobase + cmd);
+ val |= flag;
+ if (val == oval)
+ return 0;
+ outb(val, cm->iobase + cmd);
+ return 1;
+}
+
+static int snd_cmipci_clear_bit_b(struct cmipci *cm, unsigned int cmd, unsigned char flag)
+{
+ unsigned char val, oval;
+ val = oval = inb(cm->iobase + cmd);
+ val &= ~flag;
+ if (val == oval)
+ return 0;
+ outb(val, cm->iobase + cmd);
+ return 1;
+}
+
+
+/*
+ * PCM interface
+ */
+
+/*
+ * calculate frequency
+ */
+
+static unsigned int rates[] = { 5512, 11025, 22050, 44100, 8000, 16000, 32000, 48000 };
+
+static unsigned int snd_cmipci_rate_freq(unsigned int rate)
+{
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_SIZE(rates); i++) {
+ if (rates[i] == rate)
+ return i;
+ }
+ snd_BUG();
+ return 0;
+}
+
+#ifdef USE_VAR48KRATE
+/*
+ * Determine PLL values for frequency setup, maybe the CMI8338 (CMI8738???)
+ * does it this way .. maybe not. Never get any information from C-Media about
+ * that <werner@suse.de>.
+ */
+static int snd_cmipci_pll_rmn(unsigned int rate, unsigned int adcmult, int *r, int *m, int *n)
+{
+ unsigned int delta, tolerance;
+ int xm, xn, xr;
+
+ for (*r = 0; rate < CM_MAXIMUM_RATE/adcmult; *r += (1<<5))
+ rate <<= 1;
+ *n = -1;
+ if (*r > 0xff)
+ goto out;
+ tolerance = rate*CM_TOLERANCE_RATE;
+
+ for (xn = (1+2); xn < (0x1f+2); xn++) {
+ for (xm = (1+2); xm < (0xff+2); xm++) {
+ xr = ((CM_REFFREQ_XIN/adcmult) * xm) / xn;
+
+ if (xr < rate)
+ delta = rate - xr;
+ else
+ delta = xr - rate;
+
+ /*
+ * If we found one, remember this,
+ * and try to find a closer one
+ */
+ if (delta < tolerance) {
+ tolerance = delta;
+ *m = xm - 2;
+ *n = xn - 2;
+ }
+ }
+ }
+out:
+ return (*n > -1);
+}
+
+/*
+ * Program pll register bits, I assume that the 8 registers 0xf8 upto 0xff
+ * are mapped onto the 8 ADC/DAC sampling frequency which can be choosen
+ * at the register CM_REG_FUNCTRL1 (0x04).
+ * Problem: other ways are also possible (any information about that?)
+ */
+static void snd_cmipci_set_pll(struct cmipci *cm, unsigned int rate, unsigned int slot)
+{
+ unsigned int reg = CM_REG_PLL + slot;
+ /*
+ * Guess that this programs at reg. 0x04 the pos 15:13/12:10
+ * for DSFC/ASFC (000 upto 111).
+ */
+
+ /* FIXME: Init (Do we've to set an other register first before programming?) */
+
+ /* FIXME: Is this correct? Or shouldn't the m/n/r values be used for that? */
+ snd_cmipci_write_b(cm, reg, rate>>8);
+ snd_cmipci_write_b(cm, reg, rate&0xff);
+
+ /* FIXME: Setup (Do we've to set an other register first to enable this?) */
+}
+#endif /* USE_VAR48KRATE */
+
+static int snd_cmipci_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_cmipci_playback2_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ if (params_channels(hw_params) > 2) {
+ mutex_lock(&cm->open_mutex);
+ if (cm->opened[CM_CH_PLAY]) {
+ mutex_unlock(&cm->open_mutex);
+ return -EBUSY;
+ }
+ /* reserve the channel A */
+ cm->opened[CM_CH_PLAY] = CM_OPEN_PLAYBACK_MULTI;
+ mutex_unlock(&cm->open_mutex);
+ }
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static void snd_cmipci_ch_reset(struct cmipci *cm, int ch)
+{
+ int reset = CM_RST_CH0 << (cm->channel[ch].ch);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | reset);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~reset);
+ udelay(10);
+}
+
+static int snd_cmipci_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+
+/*
+ */
+
+static unsigned int hw_channels[] = {1, 2, 4, 6, 8};
+static struct snd_pcm_hw_constraint_list hw_constraints_channels_4 = {
+ .count = 3,
+ .list = hw_channels,
+ .mask = 0,
+};
+static struct snd_pcm_hw_constraint_list hw_constraints_channels_6 = {
+ .count = 4,
+ .list = hw_channels,
+ .mask = 0,
+};
+static struct snd_pcm_hw_constraint_list hw_constraints_channels_8 = {
+ .count = 5,
+ .list = hw_channels,
+ .mask = 0,
+};
+
+static int set_dac_channels(struct cmipci *cm, struct cmipci_pcm *rec, int channels)
+{
+ if (channels > 2) {
+ if (!cm->can_multi_ch || !rec->ch)
+ return -EINVAL;
+ if (rec->fmt != 0x03) /* stereo 16bit only */
+ return -EINVAL;
+ }
+
+ if (cm->can_multi_ch) {
+ spin_lock_irq(&cm->reg_lock);
+ if (channels > 2) {
+ snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
+ } else {
+ snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
+ }
+ if (channels == 8)
+ snd_cmipci_set_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C);
+ if (channels == 6) {
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
+ snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
+ } else {
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
+ snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
+ }
+ if (channels == 4)
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ spin_unlock_irq(&cm->reg_lock);
+ }
+ return 0;
+}
+
+
+/*
+ * prepare playback/capture channel
+ * channel to be used must have been set in rec->ch.
+ */
+static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
+ struct snd_pcm_substream *substream)
+{
+ unsigned int reg, freq, freq_ext, val;
+ unsigned int period_size;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ rec->fmt = 0;
+ rec->shift = 0;
+ if (snd_pcm_format_width(runtime->format) >= 16) {
+ rec->fmt |= 0x02;
+ if (snd_pcm_format_width(runtime->format) > 16)
+ rec->shift++; /* 24/32bit */
+ }
+ if (runtime->channels > 1)
+ rec->fmt |= 0x01;
+ if (rec->is_dac && set_dac_channels(cm, rec, runtime->channels) < 0) {
+ snd_printd("cannot set dac channels\n");
+ return -EINVAL;
+ }
+
+ rec->offset = runtime->dma_addr;
+ /* buffer and period sizes in frame */
+ rec->dma_size = runtime->buffer_size << rec->shift;
+ period_size = runtime->period_size << rec->shift;
+ if (runtime->channels > 2) {
+ /* multi-channels */
+ rec->dma_size = (rec->dma_size * runtime->channels) / 2;
+ period_size = (period_size * runtime->channels) / 2;
+ }
+
+ spin_lock_irq(&cm->reg_lock);
+
+ /* set buffer address */
+ reg = rec->ch ? CM_REG_CH1_FRAME1 : CM_REG_CH0_FRAME1;
+ snd_cmipci_write(cm, reg, rec->offset);
+ /* program sample counts */
+ reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
+ snd_cmipci_write_w(cm, reg, rec->dma_size - 1);
+ snd_cmipci_write_w(cm, reg + 2, period_size - 1);
+
+ /* set adc/dac flag */
+ val = rec->ch ? CM_CHADC1 : CM_CHADC0;
+ if (rec->is_dac)
+ cm->ctrl &= ~val;
+ else
+ cm->ctrl |= val;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl);
+ //snd_printd("cmipci: functrl0 = %08x\n", cm->ctrl);
+
+ /* set sample rate */
+ freq = 0;
+ freq_ext = 0;
+ if (runtime->rate > 48000)
+ switch (runtime->rate) {
+ case 88200: freq_ext = CM_CH0_SRATE_88K; break;
+ case 96000: freq_ext = CM_CH0_SRATE_96K; break;
+ case 128000: freq_ext = CM_CH0_SRATE_128K; break;
+ default: snd_BUG(); break;
+ }
+ else
+ freq = snd_cmipci_rate_freq(runtime->rate);
+ val = snd_cmipci_read(cm, CM_REG_FUNCTRL1);
+ if (rec->ch) {
+ val &= ~CM_DSFC_MASK;
+ val |= (freq << CM_DSFC_SHIFT) & CM_DSFC_MASK;
+ } else {
+ val &= ~CM_ASFC_MASK;
+ val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK;
+ }
+ snd_cmipci_write(cm, CM_REG_FUNCTRL1, val);
+ //snd_printd("cmipci: functrl1 = %08x\n", val);
+
+ /* set format */
+ val = snd_cmipci_read(cm, CM_REG_CHFORMAT);
+ if (rec->ch) {
+ val &= ~CM_CH1FMT_MASK;
+ val |= rec->fmt << CM_CH1FMT_SHIFT;
+ } else {
+ val &= ~CM_CH0FMT_MASK;
+ val |= rec->fmt << CM_CH0FMT_SHIFT;
+ }
+ if (cm->can_96k) {
+ val &= ~(CM_CH0_SRATE_MASK << (rec->ch * 2));
+ val |= freq_ext << (rec->ch * 2);
+ }
+ snd_cmipci_write(cm, CM_REG_CHFORMAT, val);
+ //snd_printd("cmipci: chformat = %08x\n", val);
+
+ if (!rec->is_dac && cm->chip_version) {
+ if (runtime->rate > 44100)
+ snd_cmipci_set_bit(cm, CM_REG_EXT_MISC, CM_ADC48K44K);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_EXT_MISC, CM_ADC48K44K);
+ }
+
+ rec->running = 0;
+ spin_unlock_irq(&cm->reg_lock);
+
+ return 0;
+}
+
+/*
+ * PCM trigger/stop
+ */
+static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec,
+ int cmd)
+{
+ unsigned int inthld, chen, reset, pause;
+ int result = 0;
+
+ inthld = CM_CH0_INT_EN << rec->ch;
+ chen = CM_CHEN0 << rec->ch;
+ reset = CM_RST_CH0 << rec->ch;
+ pause = CM_PAUSE0 << rec->ch;
+
+ spin_lock(&cm->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ rec->running = 1;
+ /* set interrupt */
+ snd_cmipci_set_bit(cm, CM_REG_INT_HLDCLR, inthld);
+ cm->ctrl |= chen;
+ /* enable channel */
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl);
+ //snd_printd("cmipci: functrl0 = %08x\n", cm->ctrl);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ rec->running = 0;
+ /* disable interrupt */
+ snd_cmipci_clear_bit(cm, CM_REG_INT_HLDCLR, inthld);
+ /* reset */
+ cm->ctrl &= ~chen;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | reset);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~reset);
+ rec->needs_silencing = rec->is_dac;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ cm->ctrl |= pause;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ cm->ctrl &= ~pause;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl);
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ spin_unlock(&cm->reg_lock);
+ return result;
+}
+
+/*
+ * return the current pointer
+ */
+static snd_pcm_uframes_t snd_cmipci_pcm_pointer(struct cmipci *cm, struct cmipci_pcm *rec,
+ struct snd_pcm_substream *substream)
+{
+ size_t ptr;
+ unsigned int reg;
+ if (!rec->running)
+ return 0;
+#if 1 // this seems better..
+ reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
+ ptr = rec->dma_size - (snd_cmipci_read_w(cm, reg) + 1);
+ ptr >>= rec->shift;
+#else
+ reg = rec->ch ? CM_REG_CH1_FRAME1 : CM_REG_CH0_FRAME1;
+ ptr = snd_cmipci_read(cm, reg) - rec->offset;
+ ptr = bytes_to_frames(substream->runtime, ptr);
+#endif
+ if (substream->runtime->channels > 2)
+ ptr = (ptr * 2) / substream->runtime->channels;
+ return ptr;
+}
+
+/*
+ * playback
+ */
+
+static int snd_cmipci_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_PLAY], cmd);
+}
+
+static snd_pcm_uframes_t snd_cmipci_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ return snd_cmipci_pcm_pointer(cm, &cm->channel[CM_CH_PLAY], substream);
+}
+
+
+
+/*
+ * capture
+ */
+
+static int snd_cmipci_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_CAPT], cmd);
+}
+
+static snd_pcm_uframes_t snd_cmipci_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ return snd_cmipci_pcm_pointer(cm, &cm->channel[CM_CH_CAPT], substream);
+}
+
+
+/*
+ * hw preparation for spdif
+ */
+
+static int snd_cmipci_spdif_default_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cmipci_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *chip = snd_kcontrol_chip(kcontrol);
+ int i;
+
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < 4; i++)
+ ucontrol->value.iec958.status[i] = (chip->dig_status >> (i * 8)) & 0xff;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *chip = snd_kcontrol_chip(kcontrol);
+ int i, change;
+ unsigned int val;
+
+ val = 0;
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < 4; i++)
+ val |= (unsigned int)ucontrol->value.iec958.status[i] << (i * 8);
+ change = val != chip->dig_status;
+ chip->dig_status = val;
+ spin_unlock_irq(&chip->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_cmipci_spdif_default __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_cmipci_spdif_default_info,
+ .get = snd_cmipci_spdif_default_get,
+ .put = snd_cmipci_spdif_default_put
+};
+
+static int snd_cmipci_spdif_mask_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cmipci_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_cmipci_spdif_mask __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_cmipci_spdif_mask_info,
+ .get = snd_cmipci_spdif_mask_get,
+};
+
+static int snd_cmipci_spdif_stream_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cmipci_spdif_stream_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *chip = snd_kcontrol_chip(kcontrol);
+ int i;
+
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < 4; i++)
+ ucontrol->value.iec958.status[i] = (chip->dig_pcm_status >> (i * 8)) & 0xff;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_spdif_stream_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *chip = snd_kcontrol_chip(kcontrol);
+ int i, change;
+ unsigned int val;
+
+ val = 0;
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < 4; i++)
+ val |= (unsigned int)ucontrol->value.iec958.status[i] << (i * 8);
+ change = val != chip->dig_pcm_status;
+ chip->dig_pcm_status = val;
+ spin_unlock_irq(&chip->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_cmipci_spdif_stream __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_cmipci_spdif_stream_info,
+ .get = snd_cmipci_spdif_stream_get,
+ .put = snd_cmipci_spdif_stream_put
+};
+
+/*
+ */
+
+/* save mixer setting and mute for AC3 playback */
+static int save_mixer_state(struct cmipci *cm)
+{
+ if (! cm->mixer_insensitive) {
+ struct snd_ctl_elem_value *val;
+ unsigned int i;
+
+ val = kmalloc(sizeof(*val), GFP_ATOMIC);
+ if (!val)
+ return -ENOMEM;
+ for (i = 0; i < CM_SAVED_MIXERS; i++) {
+ struct snd_kcontrol *ctl = cm->mixer_res_ctl[i];
+ if (ctl) {
+ int event;
+ memset(val, 0, sizeof(*val));
+ ctl->get(ctl, val);
+ cm->mixer_res_status[i] = val->value.integer.value[0];
+ val->value.integer.value[0] = cm_saved_mixer[i].toggle_on;
+ event = SNDRV_CTL_EVENT_MASK_INFO;
+ if (cm->mixer_res_status[i] != val->value.integer.value[0]) {
+ ctl->put(ctl, val); /* toggle */
+ event |= SNDRV_CTL_EVENT_MASK_VALUE;
+ }
+ ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(cm->card, event, &ctl->id);
+ }
+ }
+ kfree(val);
+ cm->mixer_insensitive = 1;
+ }
+ return 0;
+}
+
+
+/* restore the previously saved mixer status */
+static void restore_mixer_state(struct cmipci *cm)
+{
+ if (cm->mixer_insensitive) {
+ struct snd_ctl_elem_value *val;
+ unsigned int i;
+
+ val = kmalloc(sizeof(*val), GFP_KERNEL);
+ if (!val)
+ return;
+ cm->mixer_insensitive = 0; /* at first clear this;
+ otherwise the changes will be ignored */
+ for (i = 0; i < CM_SAVED_MIXERS; i++) {
+ struct snd_kcontrol *ctl = cm->mixer_res_ctl[i];
+ if (ctl) {
+ int event;
+
+ memset(val, 0, sizeof(*val));
+ ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ ctl->get(ctl, val);
+ event = SNDRV_CTL_EVENT_MASK_INFO;
+ if (val->value.integer.value[0] != cm->mixer_res_status[i]) {
+ val->value.integer.value[0] = cm->mixer_res_status[i];
+ ctl->put(ctl, val);
+ event |= SNDRV_CTL_EVENT_MASK_VALUE;
+ }
+ snd_ctl_notify(cm->card, event, &ctl->id);
+ }
+ }
+ kfree(val);
+ }
+}
+
+/* spinlock held! */
+static void setup_ac3(struct cmipci *cm, struct snd_pcm_substream *subs, int do_ac3, int rate)
+{
+ if (do_ac3) {
+ /* AC3EN for 037 */
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_AC3EN1);
+ /* AC3EN for 039 */
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_AC3EN2);
+
+ if (cm->can_ac3_hw) {
+ /* SPD24SEL for 037, 0x02 */
+ /* SPD24SEL for 039, 0x20, but cannot be set */
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_SPD24SEL);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+ } else { /* can_ac3_sw */
+ /* SPD32SEL for 037 & 039, 0x20 */
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+ /* set 176K sample rate to fix 033 HW bug */
+ if (cm->chip_version == 33) {
+ if (rate >= 48000) {
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_PLAYBACK_SRATE_176K);
+ } else {
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_PLAYBACK_SRATE_176K);
+ }
+ }
+ }
+
+ } else {
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_AC3EN1);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_AC3EN2);
+
+ if (cm->can_ac3_hw) {
+ /* chip model >= 37 */
+ if (snd_pcm_format_width(subs->runtime->format) > 16) {
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_SPD24SEL);
+ } else {
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_SPD24SEL);
+ }
+ } else {
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_SPD24SEL);
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_PLAYBACK_SRATE_176K);
+ }
+ }
+}
+
+static int setup_spdif_playback(struct cmipci *cm, struct snd_pcm_substream *subs, int up, int do_ac3)
+{
+ int rate, err;
+
+ rate = subs->runtime->rate;
+
+ if (up && do_ac3)
+ if ((err = save_mixer_state(cm)) < 0)
+ return err;
+
+ spin_lock_irq(&cm->reg_lock);
+ cm->spdif_playback_avail = up;
+ if (up) {
+ /* they are controlled via "IEC958 Output Switch" */
+ /* snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_ENSPDOUT); */
+ /* snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_SPDO2DAC); */
+ if (cm->spdif_playback_enabled)
+ snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
+ setup_ac3(cm, subs, do_ac3, rate);
+
+ if (rate == 48000 || rate == 96000)
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97);
+ if (rate > 48000)
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
+ } else {
+ /* they are controlled via "IEC958 Output Switch" */
+ /* snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_ENSPDOUT); */
+ /* snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_SPDO2DAC); */
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
+ snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
+ setup_ac3(cm, subs, 0, 0);
+ }
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+
+/*
+ * preparation
+ */
+
+/* playback - enable spdif only on the certain condition */
+static int snd_cmipci_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ int rate = substream->runtime->rate;
+ int err, do_spdif, do_ac3 = 0;
+
+ do_spdif = (rate >= 44100 && rate <= 96000 &&
+ substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE &&
+ substream->runtime->channels == 2);
+ if (do_spdif && cm->can_ac3_hw)
+ do_ac3 = cm->dig_pcm_status & IEC958_AES0_NONAUDIO;
+ if ((err = setup_spdif_playback(cm, substream, do_spdif, do_ac3)) < 0)
+ return err;
+ return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_PLAY], substream);
+}
+
+/* playback (via device #2) - enable spdif always */
+static int snd_cmipci_playback_spdif_prepare(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ int err, do_ac3;
+
+ if (cm->can_ac3_hw)
+ do_ac3 = cm->dig_pcm_status & IEC958_AES0_NONAUDIO;
+ else
+ do_ac3 = 1; /* doesn't matter */
+ if ((err = setup_spdif_playback(cm, substream, 1, do_ac3)) < 0)
+ return err;
+ return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_PLAY], substream);
+}
+
+/*
+ * Apparently, the samples last played on channel A stay in some buffer, even
+ * after the channel is reset, and get added to the data for the rear DACs when
+ * playing a multichannel stream on channel B. This is likely to generate
+ * wraparounds and thus distortions.
+ * To avoid this, we play at least one zero sample after the actual stream has
+ * stopped.
+ */
+static void snd_cmipci_silence_hack(struct cmipci *cm, struct cmipci_pcm *rec)
+{
+ struct snd_pcm_runtime *runtime = rec->substream->runtime;
+ unsigned int reg, val;
+
+ if (rec->needs_silencing && runtime && runtime->dma_area) {
+ /* set up a small silence buffer */
+ memset(runtime->dma_area, 0, PAGE_SIZE);
+ reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
+ val = ((PAGE_SIZE / 4) - 1) | (((PAGE_SIZE / 4) / 2 - 1) << 16);
+ snd_cmipci_write(cm, reg, val);
+
+ /* configure for 16 bits, 2 channels, 8 kHz */
+ if (runtime->channels > 2)
+ set_dac_channels(cm, rec, 2);
+ spin_lock_irq(&cm->reg_lock);
+ val = snd_cmipci_read(cm, CM_REG_FUNCTRL1);
+ val &= ~(CM_ASFC_MASK << (rec->ch * 3));
+ val |= (4 << CM_ASFC_SHIFT) << (rec->ch * 3);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL1, val);
+ val = snd_cmipci_read(cm, CM_REG_CHFORMAT);
+ val &= ~(CM_CH0FMT_MASK << (rec->ch * 2));
+ val |= (3 << CM_CH0FMT_SHIFT) << (rec->ch * 2);
+ if (cm->can_96k)
+ val &= ~(CM_CH0_SRATE_MASK << (rec->ch * 2));
+ snd_cmipci_write(cm, CM_REG_CHFORMAT, val);
+
+ /* start stream (we don't need interrupts) */
+ cm->ctrl |= CM_CHEN0 << rec->ch;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl);
+ spin_unlock_irq(&cm->reg_lock);
+
+ msleep(1);
+
+ /* stop and reset stream */
+ spin_lock_irq(&cm->reg_lock);
+ cm->ctrl &= ~(CM_CHEN0 << rec->ch);
+ val = CM_RST_CH0 << rec->ch;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | val);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~val);
+ spin_unlock_irq(&cm->reg_lock);
+
+ rec->needs_silencing = 0;
+ }
+}
+
+static int snd_cmipci_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ setup_spdif_playback(cm, substream, 0, 0);
+ restore_mixer_state(cm);
+ snd_cmipci_silence_hack(cm, &cm->channel[0]);
+ return snd_cmipci_hw_free(substream);
+}
+
+static int snd_cmipci_playback2_hw_free(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ snd_cmipci_silence_hack(cm, &cm->channel[1]);
+ return snd_cmipci_hw_free(substream);
+}
+
+/* capture */
+static int snd_cmipci_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_CAPT], substream);
+}
+
+/* capture with spdif (via device #2) */
+static int snd_cmipci_capture_spdif_prepare(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&cm->reg_lock);
+ snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_CAPTURE_SPDF);
+ if (cm->can_96k) {
+ if (substream->runtime->rate > 48000)
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
+ }
+ if (snd_pcm_format_width(substream->runtime->format) > 16)
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+
+ spin_unlock_irq(&cm->reg_lock);
+
+ return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_CAPT], substream);
+}
+
+static int snd_cmipci_capture_spdif_hw_free(struct snd_pcm_substream *subs)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(subs);
+
+ spin_lock_irq(&cm->reg_lock);
+ snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_CAPTURE_SPDF);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPD32SEL);
+ spin_unlock_irq(&cm->reg_lock);
+
+ return snd_cmipci_hw_free(subs);
+}
+
+
+/*
+ * interrupt handler
+ */
+static irqreturn_t snd_cmipci_interrupt(int irq, void *dev_id)
+{
+ struct cmipci *cm = dev_id;
+ unsigned int status, mask = 0;
+
+ /* fastpath out, to ease interrupt sharing */
+ status = snd_cmipci_read(cm, CM_REG_INT_STATUS);
+ if (!(status & CM_INTR))
+ return IRQ_NONE;
+
+ /* acknowledge interrupt */
+ spin_lock(&cm->reg_lock);
+ if (status & CM_CHINT0)
+ mask |= CM_CH0_INT_EN;
+ if (status & CM_CHINT1)
+ mask |= CM_CH1_INT_EN;
+ snd_cmipci_clear_bit(cm, CM_REG_INT_HLDCLR, mask);
+ snd_cmipci_set_bit(cm, CM_REG_INT_HLDCLR, mask);
+ spin_unlock(&cm->reg_lock);
+
+ if (cm->rmidi && (status & CM_UARTINT))
+ snd_mpu401_uart_interrupt(irq, cm->rmidi->private_data);
+
+ if (cm->pcm) {
+ if ((status & CM_CHINT0) && cm->channel[0].running)
+ snd_pcm_period_elapsed(cm->channel[0].substream);
+ if ((status & CM_CHINT1) && cm->channel[1].running)
+ snd_pcm_period_elapsed(cm->channel[1].substream);
+ }
+ return IRQ_HANDLED;
+}
+
+/*
+ * h/w infos
+ */
+
+/* playback on channel A */
+static struct snd_pcm_hardware snd_cmipci_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5512,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/* capture on channel B */
+static struct snd_pcm_hardware snd_cmipci_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5512,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/* playback on channel B - stereo 16bit only? */
+static struct snd_pcm_hardware snd_cmipci_playback2 =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5512,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/* spdif playback on channel A */
+static struct snd_pcm_hardware snd_cmipci_playback_spdif =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/* spdif playback on channel A (32bit, IEC958 subframes) */
+static struct snd_pcm_hardware snd_cmipci_playback_iec958_subframe =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/* spdif capture on channel B */
+static struct snd_pcm_hardware snd_cmipci_capture_spdif =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static unsigned int rate_constraints[] = { 5512, 8000, 11025, 16000, 22050,
+ 32000, 44100, 48000, 88200, 96000, 128000 };
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rate_constraints),
+ .list = rate_constraints,
+ .mask = 0,
+};
+
+/*
+ * check device open/close
+ */
+static int open_device_check(struct cmipci *cm, int mode, struct snd_pcm_substream *subs)
+{
+ int ch = mode & CM_OPEN_CH_MASK;
+
+ /* FIXME: a file should wait until the device becomes free
+ * when it's opened on blocking mode. however, since the current
+ * pcm framework doesn't pass file pointer before actually opened,
+ * we can't know whether blocking mode or not in open callback..
+ */
+ mutex_lock(&cm->open_mutex);
+ if (cm->opened[ch]) {
+ mutex_unlock(&cm->open_mutex);
+ return -EBUSY;
+ }
+ cm->opened[ch] = mode;
+ cm->channel[ch].substream = subs;
+ if (! (mode & CM_OPEN_DAC)) {
+ /* disable dual DAC mode */
+ cm->channel[ch].is_dac = 0;
+ spin_lock_irq(&cm->reg_lock);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENDBDAC);
+ spin_unlock_irq(&cm->reg_lock);
+ }
+ mutex_unlock(&cm->open_mutex);
+ return 0;
+}
+
+static void close_device_check(struct cmipci *cm, int mode)
+{
+ int ch = mode & CM_OPEN_CH_MASK;
+
+ mutex_lock(&cm->open_mutex);
+ if (cm->opened[ch] == mode) {
+ if (cm->channel[ch].substream) {
+ snd_cmipci_ch_reset(cm, ch);
+ cm->channel[ch].running = 0;
+ cm->channel[ch].substream = NULL;
+ }
+ cm->opened[ch] = 0;
+ if (! cm->channel[ch].is_dac) {
+ /* enable dual DAC mode again */
+ cm->channel[ch].is_dac = 1;
+ spin_lock_irq(&cm->reg_lock);
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_ENDBDAC);
+ spin_unlock_irq(&cm->reg_lock);
+ }
+ }
+ mutex_unlock(&cm->open_mutex);
+}
+
+/*
+ */
+
+static int snd_cmipci_playback_open(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if ((err = open_device_check(cm, CM_OPEN_PLAYBACK, substream)) < 0)
+ return err;
+ runtime->hw = snd_cmipci_playback;
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ } else if (cm->chip_version == 55) {
+ err = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates);
+ if (err < 0)
+ return err;
+ runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
+ runtime->hw.rate_max = 128000;
+ }
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
+ cm->dig_pcm_status = cm->dig_status;
+ return 0;
+}
+
+static int snd_cmipci_capture_open(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if ((err = open_device_check(cm, CM_OPEN_CAPTURE, substream)) < 0)
+ return err;
+ runtime->hw = snd_cmipci_capture;
+ if (cm->chip_version == 68) { // 8768 only supports 44k/48k recording
+ runtime->hw.rate_min = 41000;
+ runtime->hw.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
+ } else if (cm->chip_version == 55) {
+ err = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates);
+ if (err < 0)
+ return err;
+ runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
+ runtime->hw.rate_max = 128000;
+ }
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
+ return 0;
+}
+
+static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if ((err = open_device_check(cm, CM_OPEN_PLAYBACK2, substream)) < 0) /* use channel B */
+ return err;
+ runtime->hw = snd_cmipci_playback2;
+ mutex_lock(&cm->open_mutex);
+ if (! cm->opened[CM_CH_PLAY]) {
+ if (cm->can_multi_ch) {
+ runtime->hw.channels_max = cm->max_channels;
+ if (cm->max_channels == 4)
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_4);
+ else if (cm->max_channels == 6)
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_6);
+ else if (cm->max_channels == 8)
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_8);
+ }
+ }
+ mutex_unlock(&cm->open_mutex);
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ } else if (cm->chip_version == 55) {
+ err = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates);
+ if (err < 0)
+ return err;
+ runtime->hw.rates |= SNDRV_PCM_RATE_KNOT;
+ runtime->hw.rate_max = 128000;
+ }
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
+ return 0;
+}
+
+static int snd_cmipci_playback_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if ((err = open_device_check(cm, CM_OPEN_SPDIF_PLAYBACK, substream)) < 0) /* use channel A */
+ return err;
+ if (cm->can_ac3_hw) {
+ runtime->hw = snd_cmipci_playback_spdif;
+ if (cm->chip_version >= 37) {
+ runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ }
+ if (cm->can_96k) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
+ } else {
+ runtime->hw = snd_cmipci_playback_iec958_subframe;
+ }
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x40000);
+ cm->dig_pcm_status = cm->dig_status;
+ return 0;
+}
+
+static int snd_cmipci_capture_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if ((err = open_device_check(cm, CM_OPEN_SPDIF_CAPTURE, substream)) < 0) /* use channel B */
+ return err;
+ runtime->hw = snd_cmipci_capture_spdif;
+ if (cm->can_96k && !(cm->chip_version == 68)) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x40000);
+ return 0;
+}
+
+
+/*
+ */
+
+static int snd_cmipci_playback_close(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ close_device_check(cm, CM_OPEN_PLAYBACK);
+ return 0;
+}
+
+static int snd_cmipci_capture_close(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ close_device_check(cm, CM_OPEN_CAPTURE);
+ return 0;
+}
+
+static int snd_cmipci_playback2_close(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ close_device_check(cm, CM_OPEN_PLAYBACK2);
+ close_device_check(cm, CM_OPEN_PLAYBACK_MULTI);
+ return 0;
+}
+
+static int snd_cmipci_playback_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ close_device_check(cm, CM_OPEN_SPDIF_PLAYBACK);
+ return 0;
+}
+
+static int snd_cmipci_capture_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ close_device_check(cm, CM_OPEN_SPDIF_CAPTURE);
+ return 0;
+}
+
+
+/*
+ */
+
+static struct snd_pcm_ops snd_cmipci_playback_ops = {
+ .open = snd_cmipci_playback_open,
+ .close = snd_cmipci_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cmipci_hw_params,
+ .hw_free = snd_cmipci_playback_hw_free,
+ .prepare = snd_cmipci_playback_prepare,
+ .trigger = snd_cmipci_playback_trigger,
+ .pointer = snd_cmipci_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_cmipci_capture_ops = {
+ .open = snd_cmipci_capture_open,
+ .close = snd_cmipci_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cmipci_hw_params,
+ .hw_free = snd_cmipci_hw_free,
+ .prepare = snd_cmipci_capture_prepare,
+ .trigger = snd_cmipci_capture_trigger,
+ .pointer = snd_cmipci_capture_pointer,
+};
+
+static struct snd_pcm_ops snd_cmipci_playback2_ops = {
+ .open = snd_cmipci_playback2_open,
+ .close = snd_cmipci_playback2_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cmipci_playback2_hw_params,
+ .hw_free = snd_cmipci_playback2_hw_free,
+ .prepare = snd_cmipci_capture_prepare, /* channel B */
+ .trigger = snd_cmipci_capture_trigger, /* channel B */
+ .pointer = snd_cmipci_capture_pointer, /* channel B */
+};
+
+static struct snd_pcm_ops snd_cmipci_playback_spdif_ops = {
+ .open = snd_cmipci_playback_spdif_open,
+ .close = snd_cmipci_playback_spdif_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cmipci_hw_params,
+ .hw_free = snd_cmipci_playback_hw_free,
+ .prepare = snd_cmipci_playback_spdif_prepare, /* set up rate */
+ .trigger = snd_cmipci_playback_trigger,
+ .pointer = snd_cmipci_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_cmipci_capture_spdif_ops = {
+ .open = snd_cmipci_capture_spdif_open,
+ .close = snd_cmipci_capture_spdif_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cmipci_hw_params,
+ .hw_free = snd_cmipci_capture_spdif_hw_free,
+ .prepare = snd_cmipci_capture_spdif_prepare,
+ .trigger = snd_cmipci_capture_trigger,
+ .pointer = snd_cmipci_capture_pointer,
+};
+
+
+/*
+ */
+
+static int __devinit snd_cmipci_pcm_new(struct cmipci *cm, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(cm->card, cm->card->driver, device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cmipci_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cmipci_capture_ops);
+
+ pcm->private_data = cm;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "C-Media PCI DAC/ADC");
+ cm->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(cm->pci), 64*1024, 128*1024);
+
+ return 0;
+}
+
+static int __devinit snd_cmipci_pcm2_new(struct cmipci *cm, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(cm->card, cm->card->driver, device, 1, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cmipci_playback2_ops);
+
+ pcm->private_data = cm;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "C-Media PCI 2nd DAC");
+ cm->pcm2 = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(cm->pci), 64*1024, 128*1024);
+
+ return 0;
+}
+
+static int __devinit snd_cmipci_pcm_spdif_new(struct cmipci *cm, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(cm->card, cm->card->driver, device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cmipci_playback_spdif_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cmipci_capture_spdif_ops);
+
+ pcm->private_data = cm;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "C-Media PCI IEC958");
+ cm->pcm_spdif = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(cm->pci), 64*1024, 128*1024);
+
+ return 0;
+}
+
+/*
+ * mixer interface:
+ * - CM8338/8738 has a compatible mixer interface with SB16, but
+ * lack of some elements like tone control, i/o gain and AGC.
+ * - Access to native registers:
+ * - A 3D switch
+ * - Output mute switches
+ */
+
+static void snd_cmipci_mixer_write(struct cmipci *s, unsigned char idx, unsigned char data)
+{
+ outb(idx, s->iobase + CM_REG_SB16_ADDR);
+ outb(data, s->iobase + CM_REG_SB16_DATA);
+}
+
+static unsigned char snd_cmipci_mixer_read(struct cmipci *s, unsigned char idx)
+{
+ unsigned char v;
+
+ outb(idx, s->iobase + CM_REG_SB16_ADDR);
+ v = inb(s->iobase + CM_REG_SB16_DATA);
+ return v;
+}
+
+/*
+ * general mixer element
+ */
+struct cmipci_sb_reg {
+ unsigned int left_reg, right_reg;
+ unsigned int left_shift, right_shift;
+ unsigned int mask;
+ unsigned int invert: 1;
+ unsigned int stereo: 1;
+};
+
+#define COMPOSE_SB_REG(lreg,rreg,lshift,rshift,mask,invert,stereo) \
+ ((lreg) | ((rreg) << 8) | (lshift << 16) | (rshift << 19) | (mask << 24) | (invert << 22) | (stereo << 23))
+
+#define CMIPCI_DOUBLE(xname, left_reg, right_reg, left_shift, right_shift, mask, invert, stereo) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_cmipci_info_volume, \
+ .get = snd_cmipci_get_volume, .put = snd_cmipci_put_volume, \
+ .private_value = COMPOSE_SB_REG(left_reg, right_reg, left_shift, right_shift, mask, invert, stereo), \
+}
+
+#define CMIPCI_SB_VOL_STEREO(xname,reg,shift,mask) CMIPCI_DOUBLE(xname, reg, reg+1, shift, shift, mask, 0, 1)
+#define CMIPCI_SB_VOL_MONO(xname,reg,shift,mask) CMIPCI_DOUBLE(xname, reg, reg, shift, shift, mask, 0, 0)
+#define CMIPCI_SB_SW_STEREO(xname,lshift,rshift) CMIPCI_DOUBLE(xname, SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, lshift, rshift, 1, 0, 1)
+#define CMIPCI_SB_SW_MONO(xname,shift) CMIPCI_DOUBLE(xname, SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, shift, shift, 1, 0, 0)
+
+static void cmipci_sb_reg_decode(struct cmipci_sb_reg *r, unsigned long val)
+{
+ r->left_reg = val & 0xff;
+ r->right_reg = (val >> 8) & 0xff;
+ r->left_shift = (val >> 16) & 0x07;
+ r->right_shift = (val >> 19) & 0x07;
+ r->invert = (val >> 22) & 1;
+ r->stereo = (val >> 23) & 1;
+ r->mask = (val >> 24) & 0xff;
+}
+
+static int snd_cmipci_info_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct cmipci_sb_reg reg;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ uinfo->type = reg.mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = reg.stereo + 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = reg.mask;
+ return 0;
+}
+
+static int snd_cmipci_get_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ struct cmipci_sb_reg reg;
+ int val;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ spin_lock_irq(&cm->reg_lock);
+ val = (snd_cmipci_mixer_read(cm, reg.left_reg) >> reg.left_shift) & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ ucontrol->value.integer.value[0] = val;
+ if (reg.stereo) {
+ val = (snd_cmipci_mixer_read(cm, reg.right_reg) >> reg.right_shift) & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ ucontrol->value.integer.value[1] = val;
+ }
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_put_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ struct cmipci_sb_reg reg;
+ int change;
+ int left, right, oleft, oright;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ left = ucontrol->value.integer.value[0] & reg.mask;
+ if (reg.invert)
+ left = reg.mask - left;
+ left <<= reg.left_shift;
+ if (reg.stereo) {
+ right = ucontrol->value.integer.value[1] & reg.mask;
+ if (reg.invert)
+ right = reg.mask - right;
+ right <<= reg.right_shift;
+ } else
+ right = 0;
+ spin_lock_irq(&cm->reg_lock);
+ oleft = snd_cmipci_mixer_read(cm, reg.left_reg);
+ left |= oleft & ~(reg.mask << reg.left_shift);
+ change = left != oleft;
+ if (reg.stereo) {
+ if (reg.left_reg != reg.right_reg) {
+ snd_cmipci_mixer_write(cm, reg.left_reg, left);
+ oright = snd_cmipci_mixer_read(cm, reg.right_reg);
+ } else
+ oright = left;
+ right |= oright & ~(reg.mask << reg.right_shift);
+ change |= right != oright;
+ snd_cmipci_mixer_write(cm, reg.right_reg, right);
+ } else
+ snd_cmipci_mixer_write(cm, reg.left_reg, left);
+ spin_unlock_irq(&cm->reg_lock);
+ return change;
+}
+
+/*
+ * input route (left,right) -> (left,right)
+ */
+#define CMIPCI_SB_INPUT_SW(xname, left_shift, right_shift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_cmipci_info_input_sw, \
+ .get = snd_cmipci_get_input_sw, .put = snd_cmipci_put_input_sw, \
+ .private_value = COMPOSE_SB_REG(SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, left_shift, right_shift, 1, 0, 1), \
+}
+
+static int snd_cmipci_info_input_sw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 4;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_cmipci_get_input_sw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ struct cmipci_sb_reg reg;
+ int val1, val2;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ spin_lock_irq(&cm->reg_lock);
+ val1 = snd_cmipci_mixer_read(cm, reg.left_reg);
+ val2 = snd_cmipci_mixer_read(cm, reg.right_reg);
+ spin_unlock_irq(&cm->reg_lock);
+ ucontrol->value.integer.value[0] = (val1 >> reg.left_shift) & 1;
+ ucontrol->value.integer.value[1] = (val2 >> reg.left_shift) & 1;
+ ucontrol->value.integer.value[2] = (val1 >> reg.right_shift) & 1;
+ ucontrol->value.integer.value[3] = (val2 >> reg.right_shift) & 1;
+ return 0;
+}
+
+static int snd_cmipci_put_input_sw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ struct cmipci_sb_reg reg;
+ int change;
+ int val1, val2, oval1, oval2;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ spin_lock_irq(&cm->reg_lock);
+ oval1 = snd_cmipci_mixer_read(cm, reg.left_reg);
+ oval2 = snd_cmipci_mixer_read(cm, reg.right_reg);
+ val1 = oval1 & ~((1 << reg.left_shift) | (1 << reg.right_shift));
+ val2 = oval2 & ~((1 << reg.left_shift) | (1 << reg.right_shift));
+ val1 |= (ucontrol->value.integer.value[0] & 1) << reg.left_shift;
+ val2 |= (ucontrol->value.integer.value[1] & 1) << reg.left_shift;
+ val1 |= (ucontrol->value.integer.value[2] & 1) << reg.right_shift;
+ val2 |= (ucontrol->value.integer.value[3] & 1) << reg.right_shift;
+ change = val1 != oval1 || val2 != oval2;
+ snd_cmipci_mixer_write(cm, reg.left_reg, val1);
+ snd_cmipci_mixer_write(cm, reg.right_reg, val2);
+ spin_unlock_irq(&cm->reg_lock);
+ return change;
+}
+
+/*
+ * native mixer switches/volumes
+ */
+
+#define CMIPCI_MIXER_SW_STEREO(xname, reg, lshift, rshift, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_cmipci_info_native_mixer, \
+ .get = snd_cmipci_get_native_mixer, .put = snd_cmipci_put_native_mixer, \
+ .private_value = COMPOSE_SB_REG(reg, reg, lshift, rshift, 1, invert, 1), \
+}
+
+#define CMIPCI_MIXER_SW_MONO(xname, reg, shift, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_cmipci_info_native_mixer, \
+ .get = snd_cmipci_get_native_mixer, .put = snd_cmipci_put_native_mixer, \
+ .private_value = COMPOSE_SB_REG(reg, reg, shift, shift, 1, invert, 0), \
+}
+
+#define CMIPCI_MIXER_VOL_STEREO(xname, reg, lshift, rshift, mask) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_cmipci_info_native_mixer, \
+ .get = snd_cmipci_get_native_mixer, .put = snd_cmipci_put_native_mixer, \
+ .private_value = COMPOSE_SB_REG(reg, reg, lshift, rshift, mask, 0, 1), \
+}
+
+#define CMIPCI_MIXER_VOL_MONO(xname, reg, shift, mask) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_cmipci_info_native_mixer, \
+ .get = snd_cmipci_get_native_mixer, .put = snd_cmipci_put_native_mixer, \
+ .private_value = COMPOSE_SB_REG(reg, reg, shift, shift, mask, 0, 0), \
+}
+
+static int snd_cmipci_info_native_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct cmipci_sb_reg reg;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ uinfo->type = reg.mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = reg.stereo + 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = reg.mask;
+ return 0;
+
+}
+
+static int snd_cmipci_get_native_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ struct cmipci_sb_reg reg;
+ unsigned char oreg, val;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ spin_lock_irq(&cm->reg_lock);
+ oreg = inb(cm->iobase + reg.left_reg);
+ val = (oreg >> reg.left_shift) & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ ucontrol->value.integer.value[0] = val;
+ if (reg.stereo) {
+ val = (oreg >> reg.right_shift) & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ ucontrol->value.integer.value[1] = val;
+ }
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_put_native_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ struct cmipci_sb_reg reg;
+ unsigned char oreg, nreg, val;
+
+ cmipci_sb_reg_decode(&reg, kcontrol->private_value);
+ spin_lock_irq(&cm->reg_lock);
+ oreg = inb(cm->iobase + reg.left_reg);
+ val = ucontrol->value.integer.value[0] & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ nreg = oreg & ~(reg.mask << reg.left_shift);
+ nreg |= (val << reg.left_shift);
+ if (reg.stereo) {
+ val = ucontrol->value.integer.value[1] & reg.mask;
+ if (reg.invert)
+ val = reg.mask - val;
+ nreg &= ~(reg.mask << reg.right_shift);
+ nreg |= (val << reg.right_shift);
+ }
+ outb(nreg, cm->iobase + reg.left_reg);
+ spin_unlock_irq(&cm->reg_lock);
+ return (nreg != oreg);
+}
+
+/*
+ * special case - check mixer sensitivity
+ */
+static int snd_cmipci_get_native_mixer_sensitive(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ //struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ return snd_cmipci_get_native_mixer(kcontrol, ucontrol);
+}
+
+static int snd_cmipci_put_native_mixer_sensitive(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ if (cm->mixer_insensitive) {
+ /* ignored */
+ return 0;
+ }
+ return snd_cmipci_put_native_mixer(kcontrol, ucontrol);
+}
+
+
+static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = {
+ CMIPCI_SB_VOL_STEREO("Master Playback Volume", SB_DSP4_MASTER_DEV, 3, 31),
+ CMIPCI_MIXER_SW_MONO("3D Control - Switch", CM_REG_MIXER1, CM_X3DEN_SHIFT, 0),
+ CMIPCI_SB_VOL_STEREO("PCM Playback Volume", SB_DSP4_PCM_DEV, 3, 31),
+ //CMIPCI_MIXER_SW_MONO("PCM Playback Switch", CM_REG_MIXER1, CM_WSMUTE_SHIFT, 1),
+ { /* switch with sensitivity */
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .info = snd_cmipci_info_native_mixer,
+ .get = snd_cmipci_get_native_mixer_sensitive,
+ .put = snd_cmipci_put_native_mixer_sensitive,
+ .private_value = COMPOSE_SB_REG(CM_REG_MIXER1, CM_REG_MIXER1, CM_WSMUTE_SHIFT, CM_WSMUTE_SHIFT, 1, 1, 0),
+ },
+ CMIPCI_MIXER_SW_STEREO("PCM Capture Switch", CM_REG_MIXER1, CM_WAVEINL_SHIFT, CM_WAVEINR_SHIFT, 0),
+ CMIPCI_SB_VOL_STEREO("Synth Playback Volume", SB_DSP4_SYNTH_DEV, 3, 31),
+ CMIPCI_MIXER_SW_MONO("Synth Playback Switch", CM_REG_MIXER1, CM_FMMUTE_SHIFT, 1),
+ CMIPCI_SB_INPUT_SW("Synth Capture Route", 6, 5),
+ CMIPCI_SB_VOL_STEREO("CD Playback Volume", SB_DSP4_CD_DEV, 3, 31),
+ CMIPCI_SB_SW_STEREO("CD Playback Switch", 2, 1),
+ CMIPCI_SB_INPUT_SW("CD Capture Route", 2, 1),
+ CMIPCI_SB_VOL_STEREO("Line Playback Volume", SB_DSP4_LINE_DEV, 3, 31),
+ CMIPCI_SB_SW_STEREO("Line Playback Switch", 4, 3),
+ CMIPCI_SB_INPUT_SW("Line Capture Route", 4, 3),
+ CMIPCI_SB_VOL_MONO("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31),
+ CMIPCI_SB_SW_MONO("Mic Playback Switch", 0),
+ CMIPCI_DOUBLE("Mic Capture Switch", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0, 1, 0, 0),
+ CMIPCI_SB_VOL_MONO("PC Speaker Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
+ CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15),
+ CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0),
+ CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0),
+ CMIPCI_MIXER_SW_MONO("Mic Boost Playback Switch", CM_REG_MIXER2, CM_MICGAINZ_SHIFT, 1),
+ CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7),
+ CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7),
+ CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0),
+ CMIPCI_DOUBLE("PC Speaker Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0),
+ CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0),
+};
+
+/*
+ * other switches
+ */
+
+struct cmipci_switch_args {
+ int reg; /* register index */
+ unsigned int mask; /* mask bits */
+ unsigned int mask_on; /* mask bits to turn on */
+ unsigned int is_byte: 1; /* byte access? */
+ unsigned int ac3_sensitive: 1; /* access forbidden during
+ * non-audio operation?
+ */
+};
+
+#define snd_cmipci_uswitch_info snd_ctl_boolean_mono_info
+
+static int _snd_cmipci_uswitch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol,
+ struct cmipci_switch_args *args)
+{
+ unsigned int val;
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&cm->reg_lock);
+ if (args->ac3_sensitive && cm->mixer_insensitive) {
+ ucontrol->value.integer.value[0] = 0;
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+ }
+ if (args->is_byte)
+ val = inb(cm->iobase + args->reg);
+ else
+ val = snd_cmipci_read(cm, args->reg);
+ ucontrol->value.integer.value[0] = ((val & args->mask) == args->mask_on) ? 1 : 0;
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_uswitch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci_switch_args *args;
+ args = (struct cmipci_switch_args *)kcontrol->private_value;
+ if (snd_BUG_ON(!args))
+ return -EINVAL;
+ return _snd_cmipci_uswitch_get(kcontrol, ucontrol, args);
+}
+
+static int _snd_cmipci_uswitch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol,
+ struct cmipci_switch_args *args)
+{
+ unsigned int val;
+ int change;
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&cm->reg_lock);
+ if (args->ac3_sensitive && cm->mixer_insensitive) {
+ /* ignored */
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+ }
+ if (args->is_byte)
+ val = inb(cm->iobase + args->reg);
+ else
+ val = snd_cmipci_read(cm, args->reg);
+ change = (val & args->mask) != (ucontrol->value.integer.value[0] ?
+ args->mask_on : (args->mask & ~args->mask_on));
+ if (change) {
+ val &= ~args->mask;
+ if (ucontrol->value.integer.value[0])
+ val |= args->mask_on;
+ else
+ val |= (args->mask & ~args->mask_on);
+ if (args->is_byte)
+ outb((unsigned char)val, cm->iobase + args->reg);
+ else
+ snd_cmipci_write(cm, args->reg, val);
+ }
+ spin_unlock_irq(&cm->reg_lock);
+ return change;
+}
+
+static int snd_cmipci_uswitch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci_switch_args *args;
+ args = (struct cmipci_switch_args *)kcontrol->private_value;
+ if (snd_BUG_ON(!args))
+ return -EINVAL;
+ return _snd_cmipci_uswitch_put(kcontrol, ucontrol, args);
+}
+
+#define DEFINE_SWITCH_ARG(sname, xreg, xmask, xmask_on, xis_byte, xac3) \
+static struct cmipci_switch_args cmipci_switch_arg_##sname = { \
+ .reg = xreg, \
+ .mask = xmask, \
+ .mask_on = xmask_on, \
+ .is_byte = xis_byte, \
+ .ac3_sensitive = xac3, \
+}
+
+#define DEFINE_BIT_SWITCH_ARG(sname, xreg, xmask, xis_byte, xac3) \
+ DEFINE_SWITCH_ARG(sname, xreg, xmask, xmask, xis_byte, xac3)
+
+#if 0 /* these will be controlled in pcm device */
+DEFINE_BIT_SWITCH_ARG(spdif_in, CM_REG_FUNCTRL1, CM_SPDF_1, 0, 0);
+DEFINE_BIT_SWITCH_ARG(spdif_out, CM_REG_FUNCTRL1, CM_SPDF_0, 0, 0);
+#endif
+DEFINE_BIT_SWITCH_ARG(spdif_in_sel1, CM_REG_CHFORMAT, CM_SPDIF_SELECT1, 0, 0);
+DEFINE_BIT_SWITCH_ARG(spdif_in_sel2, CM_REG_MISC_CTRL, CM_SPDIF_SELECT2, 0, 0);
+DEFINE_BIT_SWITCH_ARG(spdif_enable, CM_REG_LEGACY_CTRL, CM_ENSPDOUT, 0, 0);
+DEFINE_BIT_SWITCH_ARG(spdo2dac, CM_REG_FUNCTRL1, CM_SPDO2DAC, 0, 1);
+DEFINE_BIT_SWITCH_ARG(spdi_valid, CM_REG_MISC, CM_SPDVALID, 1, 0);
+DEFINE_BIT_SWITCH_ARG(spdif_copyright, CM_REG_LEGACY_CTRL, CM_SPDCOPYRHT, 0, 0);
+DEFINE_BIT_SWITCH_ARG(spdif_dac_out, CM_REG_LEGACY_CTRL, CM_DAC2SPDO, 0, 1);
+DEFINE_SWITCH_ARG(spdo_5v, CM_REG_MISC_CTRL, CM_SPDO5V, 0, 0, 0); /* inverse: 0 = 5V */
+// DEFINE_BIT_SWITCH_ARG(spdo_48k, CM_REG_MISC_CTRL, CM_SPDF_AC97|CM_SPDIF48K, 0, 1);
+DEFINE_BIT_SWITCH_ARG(spdif_loop, CM_REG_FUNCTRL1, CM_SPDFLOOP, 0, 1);
+DEFINE_BIT_SWITCH_ARG(spdi_monitor, CM_REG_MIXER1, CM_CDPLAY, 1, 0);
+/* DEFINE_BIT_SWITCH_ARG(spdi_phase, CM_REG_CHFORMAT, CM_SPDIF_INVERSE, 0, 0); */
+DEFINE_BIT_SWITCH_ARG(spdi_phase, CM_REG_MISC, CM_SPDIF_INVERSE, 1, 0);
+DEFINE_BIT_SWITCH_ARG(spdi_phase2, CM_REG_CHFORMAT, CM_SPDIF_INVERSE2, 0, 0);
+#if CM_CH_PLAY == 1
+DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, 0, 0, 0); /* reversed */
+#else
+DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, CM_XCHGDAC, 0, 0);
+#endif
+DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MISC_CTRL, CM_N4SPK3D, 0, 0);
+// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_REAR2LIN, 1, 0);
+// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_CENTR2LIN|CM_BASE2LIN, 0, 0);
+// DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */
+DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0);
+
+#define DEFINE_SWITCH(sname, stype, sarg) \
+{ .name = sname, \
+ .iface = stype, \
+ .info = snd_cmipci_uswitch_info, \
+ .get = snd_cmipci_uswitch_get, \
+ .put = snd_cmipci_uswitch_put, \
+ .private_value = (unsigned long)&cmipci_switch_arg_##sarg,\
+}
+
+#define DEFINE_CARD_SWITCH(sname, sarg) DEFINE_SWITCH(sname, SNDRV_CTL_ELEM_IFACE_CARD, sarg)
+#define DEFINE_MIXER_SWITCH(sname, sarg) DEFINE_SWITCH(sname, SNDRV_CTL_ELEM_IFACE_MIXER, sarg)
+
+
+/*
+ * callbacks for spdif output switch
+ * needs toggle two registers..
+ */
+static int snd_cmipci_spdout_enable_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int changed;
+ changed = _snd_cmipci_uswitch_get(kcontrol, ucontrol, &cmipci_switch_arg_spdif_enable);
+ changed |= _snd_cmipci_uswitch_get(kcontrol, ucontrol, &cmipci_switch_arg_spdo2dac);
+ return changed;
+}
+
+static int snd_cmipci_spdout_enable_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *chip = snd_kcontrol_chip(kcontrol);
+ int changed;
+ changed = _snd_cmipci_uswitch_put(kcontrol, ucontrol, &cmipci_switch_arg_spdif_enable);
+ changed |= _snd_cmipci_uswitch_put(kcontrol, ucontrol, &cmipci_switch_arg_spdo2dac);
+ if (changed) {
+ if (ucontrol->value.integer.value[0]) {
+ if (chip->spdif_playback_avail)
+ snd_cmipci_set_bit(chip, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
+ } else {
+ if (chip->spdif_playback_avail)
+ snd_cmipci_clear_bit(chip, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
+ }
+ }
+ chip->spdif_playback_enabled = ucontrol->value.integer.value[0];
+ return changed;
+}
+
+
+static int snd_cmipci_line_in_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ static char *texts[3] = { "Line-In", "Rear Output", "Bass Output" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = cm->chip_version >= 39 ? 3 : 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static inline unsigned int get_line_in_mode(struct cmipci *cm)
+{
+ unsigned int val;
+ if (cm->chip_version >= 39) {
+ val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL);
+ if (val & (CM_CENTR2LIN | CM_BASE2LIN))
+ return 2;
+ }
+ val = snd_cmipci_read_b(cm, CM_REG_MIXER1);
+ if (val & CM_REAR2LIN)
+ return 1;
+ return 0;
+}
+
+static int snd_cmipci_line_in_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&cm->reg_lock);
+ ucontrol->value.enumerated.item[0] = get_line_in_mode(cm);
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_line_in_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ int change;
+
+ spin_lock_irq(&cm->reg_lock);
+ if (ucontrol->value.enumerated.item[0] == 2)
+ change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN);
+ else
+ change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN);
+ if (ucontrol->value.enumerated.item[0] == 1)
+ change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN);
+ else
+ change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN);
+ spin_unlock_irq(&cm->reg_lock);
+ return change;
+}
+
+static int snd_cmipci_mic_in_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "Mic-In", "Center/LFE Output" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_cmipci_mic_in_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ /* same bit as spdi_phase */
+ spin_lock_irq(&cm->reg_lock);
+ ucontrol->value.enumerated.item[0] =
+ (snd_cmipci_read_b(cm, CM_REG_MISC) & CM_SPDIF_INVERSE) ? 1 : 0;
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_mic_in_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cmipci *cm = snd_kcontrol_chip(kcontrol);
+ int change;
+
+ spin_lock_irq(&cm->reg_lock);
+ if (ucontrol->value.enumerated.item[0])
+ change = snd_cmipci_set_bit_b(cm, CM_REG_MISC, CM_SPDIF_INVERSE);
+ else
+ change = snd_cmipci_clear_bit_b(cm, CM_REG_MISC, CM_SPDIF_INVERSE);
+ spin_unlock_irq(&cm->reg_lock);
+ return change;
+}
+
+/* both for CM8338/8738 */
+static struct snd_kcontrol_new snd_cmipci_mixer_switches[] __devinitdata = {
+ DEFINE_MIXER_SWITCH("Four Channel Mode", fourch),
+ {
+ .name = "Line-In Mode",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_cmipci_line_in_mode_info,
+ .get = snd_cmipci_line_in_mode_get,
+ .put = snd_cmipci_line_in_mode_put,
+ },
+};
+
+/* for non-multichannel chips */
+static struct snd_kcontrol_new snd_cmipci_nomulti_switch __devinitdata =
+DEFINE_MIXER_SWITCH("Exchange DAC", exchange_dac);
+
+/* only for CM8738 */
+static struct snd_kcontrol_new snd_cmipci_8738_mixer_switches[] __devinitdata = {
+#if 0 /* controlled in pcm device */
+ DEFINE_MIXER_SWITCH("IEC958 In Record", spdif_in),
+ DEFINE_MIXER_SWITCH("IEC958 Out", spdif_out),
+ DEFINE_MIXER_SWITCH("IEC958 Out To DAC", spdo2dac),
+#endif
+ // DEFINE_MIXER_SWITCH("IEC958 Output Switch", spdif_enable),
+ { .name = "IEC958 Output Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_cmipci_uswitch_info,
+ .get = snd_cmipci_spdout_enable_get,
+ .put = snd_cmipci_spdout_enable_put,
+ },
+ DEFINE_MIXER_SWITCH("IEC958 In Valid", spdi_valid),
+ DEFINE_MIXER_SWITCH("IEC958 Copyright", spdif_copyright),
+ DEFINE_MIXER_SWITCH("IEC958 5V", spdo_5v),
+// DEFINE_MIXER_SWITCH("IEC958 In/Out 48KHz", spdo_48k),
+ DEFINE_MIXER_SWITCH("IEC958 Loop", spdif_loop),
+ DEFINE_MIXER_SWITCH("IEC958 In Monitor", spdi_monitor),
+};
+
+/* only for model 033/037 */
+static struct snd_kcontrol_new snd_cmipci_old_mixer_switches[] __devinitdata = {
+ DEFINE_MIXER_SWITCH("IEC958 Mix Analog", spdif_dac_out),
+ DEFINE_MIXER_SWITCH("IEC958 In Phase Inverse", spdi_phase),
+ DEFINE_MIXER_SWITCH("IEC958 In Select", spdif_in_sel1),
+};
+
+/* only for model 039 or later */
+static struct snd_kcontrol_new snd_cmipci_extra_mixer_switches[] __devinitdata = {
+ DEFINE_MIXER_SWITCH("IEC958 In Select", spdif_in_sel2),
+ DEFINE_MIXER_SWITCH("IEC958 In Phase Inverse", spdi_phase2),
+ {
+ .name = "Mic-In Mode",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_cmipci_mic_in_mode_info,
+ .get = snd_cmipci_mic_in_mode_get,
+ .put = snd_cmipci_mic_in_mode_put,
+ }
+};
+
+/* card control switches */
+static struct snd_kcontrol_new snd_cmipci_modem_switch __devinitdata =
+DEFINE_CARD_SWITCH("Modem", modem);
+
+
+static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_device)
+{
+ struct snd_card *card;
+ struct snd_kcontrol_new *sw;
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!cm || !cm->card))
+ return -EINVAL;
+
+ card = cm->card;
+
+ strcpy(card->mixername, "CMedia PCI");
+
+ spin_lock_irq(&cm->reg_lock);
+ snd_cmipci_mixer_write(cm, 0x00, 0x00); /* mixer reset */
+ spin_unlock_irq(&cm->reg_lock);
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_mixers); idx++) {
+ if (cm->chip_version == 68) { // 8768 has no PCM volume
+ if (!strcmp(snd_cmipci_mixers[idx].name,
+ "PCM Playback Volume"))
+ continue;
+ }
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cmipci_mixers[idx], cm))) < 0)
+ return err;
+ }
+
+ /* mixer switches */
+ sw = snd_cmipci_mixer_switches;
+ for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_mixer_switches); idx++, sw++) {
+ err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm));
+ if (err < 0)
+ return err;
+ }
+ if (! cm->can_multi_ch) {
+ err = snd_ctl_add(cm->card, snd_ctl_new1(&snd_cmipci_nomulti_switch, cm));
+ if (err < 0)
+ return err;
+ }
+ if (cm->device == PCI_DEVICE_ID_CMEDIA_CM8738 ||
+ cm->device == PCI_DEVICE_ID_CMEDIA_CM8738B) {
+ sw = snd_cmipci_8738_mixer_switches;
+ for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_8738_mixer_switches); idx++, sw++) {
+ err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm));
+ if (err < 0)
+ return err;
+ }
+ if (cm->can_ac3_hw) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_cmipci_spdif_default, cm))) < 0)
+ return err;
+ kctl->id.device = pcm_spdif_device;
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_cmipci_spdif_mask, cm))) < 0)
+ return err;
+ kctl->id.device = pcm_spdif_device;
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_cmipci_spdif_stream, cm))) < 0)
+ return err;
+ kctl->id.device = pcm_spdif_device;
+ }
+ if (cm->chip_version <= 37) {
+ sw = snd_cmipci_old_mixer_switches;
+ for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_old_mixer_switches); idx++, sw++) {
+ err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm));
+ if (err < 0)
+ return err;
+ }
+ }
+ }
+ if (cm->chip_version >= 39) {
+ sw = snd_cmipci_extra_mixer_switches;
+ for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_extra_mixer_switches); idx++, sw++) {
+ err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ /* card switches */
+ /*
+ * newer chips don't have the register bits to force modem link
+ * detection; the bit that was FLINKON now mutes CH1
+ */
+ if (cm->chip_version < 39) {
+ err = snd_ctl_add(cm->card,
+ snd_ctl_new1(&snd_cmipci_modem_switch, cm));
+ if (err < 0)
+ return err;
+ }
+
+ for (idx = 0; idx < CM_SAVED_MIXERS; idx++) {
+ struct snd_ctl_elem_id elem_id;
+ struct snd_kcontrol *ctl;
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, cm_saved_mixer[idx].name);
+ ctl = snd_ctl_find_id(cm->card, &elem_id);
+ if (ctl)
+ cm->mixer_res_ctl[idx] = ctl;
+ }
+
+ return 0;
+}
+
+
+/*
+ * proc interface
+ */
+
+#ifdef CONFIG_PROC_FS
+static void snd_cmipci_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct cmipci *cm = entry->private_data;
+ int i, v;
+
+ snd_iprintf(buffer, "%s\n", cm->card->longname);
+ for (i = 0; i < 0x94; i++) {
+ if (i == 0x28)
+ i = 0x90;
+ v = inb(cm->iobase + i);
+ if (i % 4 == 0)
+ snd_iprintf(buffer, "\n%02x:", i);
+ snd_iprintf(buffer, " %02x", v);
+ }
+ snd_iprintf(buffer, "\n");
+}
+
+static void __devinit snd_cmipci_proc_init(struct cmipci *cm)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(cm->card, "cmipci", &entry))
+ snd_info_set_text_ops(entry, cm, snd_cmipci_proc_read);
+}
+#else /* !CONFIG_PROC_FS */
+static inline void snd_cmipci_proc_init(struct cmipci *cm) {}
+#endif
+
+
+static struct pci_device_id snd_cmipci_ids[] = {
+ {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {PCI_VENDOR_ID_AL, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {0,},
+};
+
+
+/*
+ * check chip version and capabilities
+ * driver name is modified according to the chip model
+ */
+static void __devinit query_chip(struct cmipci *cm)
+{
+ unsigned int detect;
+
+ /* check reg 0Ch, bit 24-31 */
+ detect = snd_cmipci_read(cm, CM_REG_INT_HLDCLR) & CM_CHIP_MASK2;
+ if (! detect) {
+ /* check reg 08h, bit 24-28 */
+ detect = snd_cmipci_read(cm, CM_REG_CHFORMAT) & CM_CHIP_MASK1;
+ switch (detect) {
+ case 0:
+ cm->chip_version = 33;
+ if (cm->do_soft_ac3)
+ cm->can_ac3_sw = 1;
+ else
+ cm->can_ac3_hw = 1;
+ break;
+ case CM_CHIP_037:
+ cm->chip_version = 37;
+ cm->can_ac3_hw = 1;
+ break;
+ default:
+ cm->chip_version = 39;
+ cm->can_ac3_hw = 1;
+ break;
+ }
+ cm->max_channels = 2;
+ } else {
+ if (detect & CM_CHIP_039) {
+ cm->chip_version = 39;
+ if (detect & CM_CHIP_039_6CH) /* 4 or 6 channels */
+ cm->max_channels = 6;
+ else
+ cm->max_channels = 4;
+ } else if (detect & CM_CHIP_8768) {
+ cm->chip_version = 68;
+ cm->max_channels = 8;
+ cm->can_96k = 1;
+ } else {
+ cm->chip_version = 55;
+ cm->max_channels = 6;
+ cm->can_96k = 1;
+ }
+ cm->can_ac3_hw = 1;
+ cm->can_multi_ch = 1;
+ }
+}
+
+#ifdef SUPPORT_JOYSTICK
+static int __devinit snd_cmipci_create_gameport(struct cmipci *cm, int dev)
+{
+ static int ports[] = { 0x201, 0x200, 0 }; /* FIXME: majority is 0x201? */
+ struct gameport *gp;
+ struct resource *r = NULL;
+ int i, io_port = 0;
+
+ if (joystick_port[dev] == 0)
+ return -ENODEV;
+
+ if (joystick_port[dev] == 1) { /* auto-detect */
+ for (i = 0; ports[i]; i++) {
+ io_port = ports[i];
+ r = request_region(io_port, 1, "CMIPCI gameport");
+ if (r)
+ break;
+ }
+ } else {
+ io_port = joystick_port[dev];
+ r = request_region(io_port, 1, "CMIPCI gameport");
+ }
+
+ if (!r) {
+ printk(KERN_WARNING "cmipci: cannot reserve joystick ports\n");
+ return -EBUSY;
+ }
+
+ cm->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "cmipci: cannot allocate memory for gameport\n");
+ release_and_free_resource(r);
+ return -ENOMEM;
+ }
+ gameport_set_name(gp, "C-Media Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(cm->pci));
+ gameport_set_dev_parent(gp, &cm->pci->dev);
+ gp->io = io_port;
+ gameport_set_port_data(gp, r);
+
+ snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_JYSTK_EN);
+
+ gameport_register_port(cm->gameport);
+
+ return 0;
+}
+
+static void snd_cmipci_free_gameport(struct cmipci *cm)
+{
+ if (cm->gameport) {
+ struct resource *r = gameport_get_port_data(cm->gameport);
+
+ gameport_unregister_port(cm->gameport);
+ cm->gameport = NULL;
+
+ snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_JYSTK_EN);
+ release_and_free_resource(r);
+ }
+}
+#else
+static inline int snd_cmipci_create_gameport(struct cmipci *cm, int dev) { return -ENOSYS; }
+static inline void snd_cmipci_free_gameport(struct cmipci *cm) { }
+#endif
+
+static int snd_cmipci_free(struct cmipci *cm)
+{
+ if (cm->irq >= 0) {
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_FM_EN);
+ snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_ENSPDOUT);
+ snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0); /* disable ints */
+ snd_cmipci_ch_reset(cm, CM_CH_PLAY);
+ snd_cmipci_ch_reset(cm, CM_CH_CAPT);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, 0); /* disable channels */
+ snd_cmipci_write(cm, CM_REG_FUNCTRL1, 0);
+
+ /* reset mixer */
+ snd_cmipci_mixer_write(cm, 0, 0);
+
+ free_irq(cm->irq, cm);
+ }
+
+ snd_cmipci_free_gameport(cm);
+ pci_release_regions(cm->pci);
+ pci_disable_device(cm->pci);
+ kfree(cm);
+ return 0;
+}
+
+static int snd_cmipci_dev_free(struct snd_device *device)
+{
+ struct cmipci *cm = device->device_data;
+ return snd_cmipci_free(cm);
+}
+
+static int __devinit snd_cmipci_create_fm(struct cmipci *cm, long fm_port)
+{
+ long iosynth;
+ unsigned int val;
+ struct snd_opl3 *opl3;
+ int err;
+
+ if (!fm_port)
+ goto disable_fm;
+
+ if (cm->chip_version >= 39) {
+ /* first try FM regs in PCI port range */
+ iosynth = cm->iobase + CM_REG_FM_PCI;
+ err = snd_opl3_create(cm->card, iosynth, iosynth + 2,
+ OPL3_HW_OPL3, 1, &opl3);
+ } else {
+ err = -EIO;
+ }
+ if (err < 0) {
+ /* then try legacy ports */
+ val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL) & ~CM_FMSEL_MASK;
+ iosynth = fm_port;
+ switch (iosynth) {
+ case 0x3E8: val |= CM_FMSEL_3E8; break;
+ case 0x3E0: val |= CM_FMSEL_3E0; break;
+ case 0x3C8: val |= CM_FMSEL_3C8; break;
+ case 0x388: val |= CM_FMSEL_388; break;
+ default:
+ goto disable_fm;
+ }
+ snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
+ /* enable FM */
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_FM_EN);
+
+ if (snd_opl3_create(cm->card, iosynth, iosynth + 2,
+ OPL3_HW_OPL3, 0, &opl3) < 0) {
+ printk(KERN_ERR "cmipci: no OPL device at %#lx, "
+ "skipping...\n", iosynth);
+ goto disable_fm;
+ }
+ }
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ printk(KERN_ERR "cmipci: cannot create OPL3 hwdep\n");
+ return err;
+ }
+ return 0;
+
+ disable_fm:
+ snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_FMSEL_MASK);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_FM_EN);
+ return 0;
+}
+
+static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pci,
+ int dev, struct cmipci **rcmipci)
+{
+ struct cmipci *cm;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_cmipci_dev_free,
+ };
+ unsigned int val;
+ long iomidi;
+ int integrated_midi = 0;
+ char modelstr[16];
+ int pcm_index, pcm_spdif_index;
+ static struct pci_device_id intel_82437vx[] = {
+ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) },
+ { },
+ };
+
+ *rcmipci = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ cm = kzalloc(sizeof(*cm), GFP_KERNEL);
+ if (cm == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&cm->reg_lock);
+ mutex_init(&cm->open_mutex);
+ cm->device = pci->device;
+ cm->card = card;
+ cm->pci = pci;
+ cm->irq = -1;
+ cm->channel[0].ch = 0;
+ cm->channel[1].ch = 1;
+ cm->channel[0].is_dac = cm->channel[1].is_dac = 1; /* dual DAC mode */
+
+ if ((err = pci_request_regions(pci, card->driver)) < 0) {
+ kfree(cm);
+ pci_disable_device(pci);
+ return err;
+ }
+ cm->iobase = pci_resource_start(pci, 0);
+
+ if (request_irq(pci->irq, snd_cmipci_interrupt,
+ IRQF_SHARED, card->driver, cm)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_cmipci_free(cm);
+ return -EBUSY;
+ }
+ cm->irq = pci->irq;
+
+ pci_set_master(cm->pci);
+
+ /*
+ * check chip version, max channels and capabilities
+ */
+
+ cm->chip_version = 0;
+ cm->max_channels = 2;
+ cm->do_soft_ac3 = soft_ac3[dev];
+
+ if (pci->device != PCI_DEVICE_ID_CMEDIA_CM8338A &&
+ pci->device != PCI_DEVICE_ID_CMEDIA_CM8338B)
+ query_chip(cm);
+ /* added -MCx suffix for chip supporting multi-channels */
+ if (cm->can_multi_ch)
+ sprintf(cm->card->driver + strlen(cm->card->driver),
+ "-MC%d", cm->max_channels);
+ else if (cm->can_ac3_sw)
+ strcpy(cm->card->driver + strlen(cm->card->driver), "-SWIEC");
+
+ cm->dig_status = SNDRV_PCM_DEFAULT_CON_SPDIF;
+ cm->dig_pcm_status = SNDRV_PCM_DEFAULT_CON_SPDIF;
+
+#if CM_CH_PLAY == 1
+ cm->ctrl = CM_CHADC0; /* default FUNCNTRL0 */
+#else
+ cm->ctrl = CM_CHADC1; /* default FUNCNTRL0 */
+#endif
+
+ /* initialize codec registers */
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_RESET);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_RESET);
+ snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0); /* disable ints */
+ snd_cmipci_ch_reset(cm, CM_CH_PLAY);
+ snd_cmipci_ch_reset(cm, CM_CH_CAPT);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, 0); /* disable channels */
+ snd_cmipci_write(cm, CM_REG_FUNCTRL1, 0);
+
+ snd_cmipci_write(cm, CM_REG_CHFORMAT, 0);
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_ENDBDAC|CM_N4SPK3D);
+#if CM_CH_PLAY == 1
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
+#else
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
+#endif
+ if (cm->chip_version) {
+ snd_cmipci_write_b(cm, CM_REG_EXT_MISC, 0x20); /* magic */
+ snd_cmipci_write_b(cm, CM_REG_EXT_MISC + 1, 0x09); /* more magic */
+ }
+ /* Set Bus Master Request */
+ snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_BREQ);
+
+ /* Assume TX and compatible chip set (Autodetection required for VX chip sets) */
+ switch (pci->device) {
+ case PCI_DEVICE_ID_CMEDIA_CM8738:
+ case PCI_DEVICE_ID_CMEDIA_CM8738B:
+ if (!pci_dev_present(intel_82437vx))
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_TXVX);
+ break;
+ default:
+ break;
+ }
+
+ if (cm->chip_version < 68) {
+ val = pci->device < 0x110 ? 8338 : 8738;
+ } else {
+ switch (snd_cmipci_read_b(cm, CM_REG_INT_HLDCLR + 3) & 0x03) {
+ case 0:
+ val = 8769;
+ break;
+ case 2:
+ val = 8762;
+ break;
+ default:
+ switch ((pci->subsystem_vendor << 16) |
+ pci->subsystem_device) {
+ case 0x13f69761:
+ case 0x584d3741:
+ case 0x584d3751:
+ case 0x584d3761:
+ case 0x584d3771:
+ case 0x72848384:
+ val = 8770;
+ break;
+ default:
+ val = 8768;
+ break;
+ }
+ }
+ }
+ sprintf(card->shortname, "C-Media CMI%d", val);
+ if (cm->chip_version < 68)
+ sprintf(modelstr, " (model %d)", cm->chip_version);
+ else
+ modelstr[0] = '\0';
+ sprintf(card->longname, "%s%s at %#lx, irq %i",
+ card->shortname, modelstr, cm->iobase, cm->irq);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, cm, &ops)) < 0) {
+ snd_cmipci_free(cm);
+ return err;
+ }
+
+ if (cm->chip_version >= 39) {
+ val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1);
+ if (val != 0x00 && val != 0xff) {
+ iomidi = cm->iobase + CM_REG_MPU_PCI;
+ integrated_midi = 1;
+ }
+ }
+ if (!integrated_midi) {
+ val = 0;
+ iomidi = mpu_port[dev];
+ switch (iomidi) {
+ case 0x320: val = CM_VMPU_320; break;
+ case 0x310: val = CM_VMPU_310; break;
+ case 0x300: val = CM_VMPU_300; break;
+ case 0x330: val = CM_VMPU_330; break;
+ default:
+ iomidi = 0; break;
+ }
+ if (iomidi > 0) {
+ snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
+ /* enable UART */
+ snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN);
+ if (inb(iomidi + 1) == 0xff) {
+ snd_printk(KERN_ERR "cannot enable MPU-401 port"
+ " at %#lx\n", iomidi);
+ snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1,
+ CM_UART_EN);
+ iomidi = 0;
+ }
+ }
+ }
+
+ if (cm->chip_version < 68) {
+ err = snd_cmipci_create_fm(cm, fm_port[dev]);
+ if (err < 0)
+ return err;
+ }
+
+ /* reset mixer */
+ snd_cmipci_mixer_write(cm, 0, 0);
+
+ snd_cmipci_proc_init(cm);
+
+ /* create pcm devices */
+ pcm_index = pcm_spdif_index = 0;
+ if ((err = snd_cmipci_pcm_new(cm, pcm_index)) < 0)
+ return err;
+ pcm_index++;
+ if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0)
+ return err;
+ pcm_index++;
+ if (cm->can_ac3_hw || cm->can_ac3_sw) {
+ pcm_spdif_index = pcm_index;
+ if ((err = snd_cmipci_pcm_spdif_new(cm, pcm_index)) < 0)
+ return err;
+ }
+
+ /* create mixer interface & switches */
+ if ((err = snd_cmipci_mixer_new(cm, pcm_spdif_index)) < 0)
+ return err;
+
+ if (iomidi > 0) {
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
+ iomidi,
+ (integrated_midi ?
+ MPU401_INFO_INTEGRATED : 0),
+ cm->irq, 0, &cm->rmidi)) < 0) {
+ printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi);
+ }
+ }
+
+#ifdef USE_VAR48KRATE
+ for (val = 0; val < ARRAY_SIZE(rates); val++)
+ snd_cmipci_set_pll(cm, rates[val], val);
+
+ /*
+ * (Re-)Enable external switch spdo_48k
+ */
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K|CM_SPDF_AC97);
+#endif /* USE_VAR48KRATE */
+
+ if (snd_cmipci_create_gameport(cm, dev) < 0)
+ snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_JYSTK_EN);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rcmipci = cm;
+ return 0;
+}
+
+/*
+ */
+
+MODULE_DEVICE_TABLE(pci, snd_cmipci_ids);
+
+static int __devinit snd_cmipci_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct cmipci *cm;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (! enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ switch (pci->device) {
+ case PCI_DEVICE_ID_CMEDIA_CM8738:
+ case PCI_DEVICE_ID_CMEDIA_CM8738B:
+ strcpy(card->driver, "CMI8738");
+ break;
+ case PCI_DEVICE_ID_CMEDIA_CM8338A:
+ case PCI_DEVICE_ID_CMEDIA_CM8338B:
+ strcpy(card->driver, "CMI8338");
+ break;
+ default:
+ strcpy(card->driver, "CMIPCI");
+ break;
+ }
+
+ if ((err = snd_cmipci_create(card, pci, dev, &cm)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = cm;
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+
+}
+
+static void __devexit snd_cmipci_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static unsigned char saved_regs[] = {
+ CM_REG_FUNCTRL1, CM_REG_CHFORMAT, CM_REG_LEGACY_CTRL, CM_REG_MISC_CTRL,
+ CM_REG_MIXER0, CM_REG_MIXER1, CM_REG_MIXER2, CM_REG_MIXER3, CM_REG_PLL,
+ CM_REG_CH0_FRAME1, CM_REG_CH0_FRAME2,
+ CM_REG_CH1_FRAME1, CM_REG_CH1_FRAME2, CM_REG_EXT_MISC,
+ CM_REG_INT_STATUS, CM_REG_INT_HLDCLR, CM_REG_FUNCTRL0,
+};
+
+static unsigned char saved_mixers[] = {
+ SB_DSP4_MASTER_DEV, SB_DSP4_MASTER_DEV + 1,
+ SB_DSP4_PCM_DEV, SB_DSP4_PCM_DEV + 1,
+ SB_DSP4_SYNTH_DEV, SB_DSP4_SYNTH_DEV + 1,
+ SB_DSP4_CD_DEV, SB_DSP4_CD_DEV + 1,
+ SB_DSP4_LINE_DEV, SB_DSP4_LINE_DEV + 1,
+ SB_DSP4_MIC_DEV, SB_DSP4_SPEAKER_DEV,
+ CM_REG_EXTENT_IND, SB_DSP4_OUTPUT_SW,
+ SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT,
+};
+
+static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct cmipci *cm = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ snd_pcm_suspend_all(cm->pcm);
+ snd_pcm_suspend_all(cm->pcm2);
+ snd_pcm_suspend_all(cm->pcm_spdif);
+
+ /* save registers */
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ cm->saved_regs[i] = snd_cmipci_read(cm, saved_regs[i]);
+ for (i = 0; i < ARRAY_SIZE(saved_mixers); i++)
+ cm->saved_mixers[i] = snd_cmipci_mixer_read(cm, saved_mixers[i]);
+
+ /* disable ints */
+ snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_cmipci_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct cmipci *cm = card->private_data;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "cmipci: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ /* reset / initialize to a sane state */
+ snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0);
+ snd_cmipci_ch_reset(cm, CM_CH_PLAY);
+ snd_cmipci_ch_reset(cm, CM_CH_CAPT);
+ snd_cmipci_mixer_write(cm, 0, 0);
+
+ /* restore registers */
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ snd_cmipci_write(cm, saved_regs[i], cm->saved_regs[i]);
+ for (i = 0; i < ARRAY_SIZE(saved_mixers); i++)
+ snd_cmipci_mixer_write(cm, saved_mixers[i], cm->saved_mixers[i]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static struct pci_driver driver = {
+ .name = "C-Media PCI",
+ .id_table = snd_cmipci_ids,
+ .probe = snd_cmipci_probe,
+ .remove = __devexit_p(snd_cmipci_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cmipci_suspend,
+ .resume = snd_cmipci_resume,
+#endif
+};
+
+static int __init alsa_card_cmipci_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_cmipci_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_cmipci_init)
+module_exit(alsa_card_cmipci_exit)
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
new file mode 100644
index 0000000..192e784
--- /dev/null
+++ b/sound/pci/cs4281.c
@@ -0,0 +1,2117 @@
+/*
+ * Driver for Cirrus Logic CS4281 based PCI soundcard
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/ac97_codec.h>
+#include <sound/tlv.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Cirrus Logic CS4281");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */
+static int dual_codec[SNDRV_CARDS]; /* dual codec */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for CS4281 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for CS4281 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable CS4281 soundcard.");
+module_param_array(dual_codec, bool, NULL, 0444);
+MODULE_PARM_DESC(dual_codec, "Secondary Codec ID (0 = disabled).");
+
+/*
+ * Direct registers
+ */
+
+#define CS4281_BA0_SIZE 0x1000
+#define CS4281_BA1_SIZE 0x10000
+
+/*
+ * BA0 registers
+ */
+#define BA0_HISR 0x0000 /* Host Interrupt Status Register */
+#define BA0_HISR_INTENA (1<<31) /* Internal Interrupt Enable Bit */
+#define BA0_HISR_MIDI (1<<22) /* MIDI port interrupt */
+#define BA0_HISR_FIFOI (1<<20) /* FIFO polled interrupt */
+#define BA0_HISR_DMAI (1<<18) /* DMA interrupt (half or end) */
+#define BA0_HISR_FIFO(c) (1<<(12+(c))) /* FIFO channel interrupt */
+#define BA0_HISR_DMA(c) (1<<(8+(c))) /* DMA channel interrupt */
+#define BA0_HISR_GPPI (1<<5) /* General Purpose Input (Primary chip) */
+#define BA0_HISR_GPSI (1<<4) /* General Purpose Input (Secondary chip) */
+#define BA0_HISR_GP3I (1<<3) /* GPIO3 pin Interrupt */
+#define BA0_HISR_GP1I (1<<2) /* GPIO1 pin Interrupt */
+#define BA0_HISR_VUPI (1<<1) /* VOLUP pin Interrupt */
+#define BA0_HISR_VDNI (1<<0) /* VOLDN pin Interrupt */
+
+#define BA0_HICR 0x0008 /* Host Interrupt Control Register */
+#define BA0_HICR_CHGM (1<<1) /* INTENA Change Mask */
+#define BA0_HICR_IEV (1<<0) /* INTENA Value */
+#define BA0_HICR_EOI (3<<0) /* End of Interrupt command */
+
+#define BA0_HIMR 0x000c /* Host Interrupt Mask Register */
+ /* Use same contants as for BA0_HISR */
+
+#define BA0_IIER 0x0010 /* ISA Interrupt Enable Register */
+
+#define BA0_HDSR0 0x00f0 /* Host DMA Engine 0 Status Register */
+#define BA0_HDSR1 0x00f4 /* Host DMA Engine 1 Status Register */
+#define BA0_HDSR2 0x00f8 /* Host DMA Engine 2 Status Register */
+#define BA0_HDSR3 0x00fc /* Host DMA Engine 3 Status Register */
+
+#define BA0_HDSR_CH1P (1<<25) /* Channel 1 Pending */
+#define BA0_HDSR_CH2P (1<<24) /* Channel 2 Pending */
+#define BA0_HDSR_DHTC (1<<17) /* DMA Half Terminal Count */
+#define BA0_HDSR_DTC (1<<16) /* DMA Terminal Count */
+#define BA0_HDSR_DRUN (1<<15) /* DMA Running */
+#define BA0_HDSR_RQ (1<<7) /* Pending Request */
+
+#define BA0_DCA0 0x0110 /* Host DMA Engine 0 Current Address */
+#define BA0_DCC0 0x0114 /* Host DMA Engine 0 Current Count */
+#define BA0_DBA0 0x0118 /* Host DMA Engine 0 Base Address */
+#define BA0_DBC0 0x011c /* Host DMA Engine 0 Base Count */
+#define BA0_DCA1 0x0120 /* Host DMA Engine 1 Current Address */
+#define BA0_DCC1 0x0124 /* Host DMA Engine 1 Current Count */
+#define BA0_DBA1 0x0128 /* Host DMA Engine 1 Base Address */
+#define BA0_DBC1 0x012c /* Host DMA Engine 1 Base Count */
+#define BA0_DCA2 0x0130 /* Host DMA Engine 2 Current Address */
+#define BA0_DCC2 0x0134 /* Host DMA Engine 2 Current Count */
+#define BA0_DBA2 0x0138 /* Host DMA Engine 2 Base Address */
+#define BA0_DBC2 0x013c /* Host DMA Engine 2 Base Count */
+#define BA0_DCA3 0x0140 /* Host DMA Engine 3 Current Address */
+#define BA0_DCC3 0x0144 /* Host DMA Engine 3 Current Count */
+#define BA0_DBA3 0x0148 /* Host DMA Engine 3 Base Address */
+#define BA0_DBC3 0x014c /* Host DMA Engine 3 Base Count */
+#define BA0_DMR0 0x0150 /* Host DMA Engine 0 Mode */
+#define BA0_DCR0 0x0154 /* Host DMA Engine 0 Command */
+#define BA0_DMR1 0x0158 /* Host DMA Engine 1 Mode */
+#define BA0_DCR1 0x015c /* Host DMA Engine 1 Command */
+#define BA0_DMR2 0x0160 /* Host DMA Engine 2 Mode */
+#define BA0_DCR2 0x0164 /* Host DMA Engine 2 Command */
+#define BA0_DMR3 0x0168 /* Host DMA Engine 3 Mode */
+#define BA0_DCR3 0x016c /* Host DMA Engine 3 Command */
+
+#define BA0_DMR_DMA (1<<29) /* Enable DMA mode */
+#define BA0_DMR_POLL (1<<28) /* Enable poll mode */
+#define BA0_DMR_TBC (1<<25) /* Transfer By Channel */
+#define BA0_DMR_CBC (1<<24) /* Count By Channel (0 = frame resolution) */
+#define BA0_DMR_SWAPC (1<<22) /* Swap Left/Right Channels */
+#define BA0_DMR_SIZE20 (1<<20) /* Sample is 20-bit */
+#define BA0_DMR_USIGN (1<<19) /* Unsigned */
+#define BA0_DMR_BEND (1<<18) /* Big Endian */
+#define BA0_DMR_MONO (1<<17) /* Mono */
+#define BA0_DMR_SIZE8 (1<<16) /* Sample is 8-bit */
+#define BA0_DMR_TYPE_DEMAND (0<<6)
+#define BA0_DMR_TYPE_SINGLE (1<<6)
+#define BA0_DMR_TYPE_BLOCK (2<<6)
+#define BA0_DMR_TYPE_CASCADE (3<<6) /* Not supported */
+#define BA0_DMR_DEC (1<<5) /* Access Increment (0) or Decrement (1) */
+#define BA0_DMR_AUTO (1<<4) /* Auto-Initialize */
+#define BA0_DMR_TR_VERIFY (0<<2) /* Verify Transfer */
+#define BA0_DMR_TR_WRITE (1<<2) /* Write Transfer */
+#define BA0_DMR_TR_READ (2<<2) /* Read Transfer */
+
+#define BA0_DCR_HTCIE (1<<17) /* Half Terminal Count Interrupt */
+#define BA0_DCR_TCIE (1<<16) /* Terminal Count Interrupt */
+#define BA0_DCR_MSK (1<<0) /* DMA Mask bit */
+
+#define BA0_FCR0 0x0180 /* FIFO Control 0 */
+#define BA0_FCR1 0x0184 /* FIFO Control 1 */
+#define BA0_FCR2 0x0188 /* FIFO Control 2 */
+#define BA0_FCR3 0x018c /* FIFO Control 3 */
+
+#define BA0_FCR_FEN (1<<31) /* FIFO Enable bit */
+#define BA0_FCR_DACZ (1<<30) /* DAC Zero */
+#define BA0_FCR_PSH (1<<29) /* Previous Sample Hold */
+#define BA0_FCR_RS(x) (((x)&0x1f)<<24) /* Right Slot Mapping */
+#define BA0_FCR_LS(x) (((x)&0x1f)<<16) /* Left Slot Mapping */
+#define BA0_FCR_SZ(x) (((x)&0x7f)<<8) /* FIFO buffer size (in samples) */
+#define BA0_FCR_OF(x) (((x)&0x7f)<<0) /* FIFO starting offset (in samples) */
+
+#define BA0_FPDR0 0x0190 /* FIFO Polled Data 0 */
+#define BA0_FPDR1 0x0194 /* FIFO Polled Data 1 */
+#define BA0_FPDR2 0x0198 /* FIFO Polled Data 2 */
+#define BA0_FPDR3 0x019c /* FIFO Polled Data 3 */
+
+#define BA0_FCHS 0x020c /* FIFO Channel Status */
+#define BA0_FCHS_RCO(x) (1<<(7+(((x)&3)<<3))) /* Right Channel Out */
+#define BA0_FCHS_LCO(x) (1<<(6+(((x)&3)<<3))) /* Left Channel Out */
+#define BA0_FCHS_MRP(x) (1<<(5+(((x)&3)<<3))) /* Move Read Pointer */
+#define BA0_FCHS_FE(x) (1<<(4+(((x)&3)<<3))) /* FIFO Empty */
+#define BA0_FCHS_FF(x) (1<<(3+(((x)&3)<<3))) /* FIFO Full */
+#define BA0_FCHS_IOR(x) (1<<(2+(((x)&3)<<3))) /* Internal Overrun Flag */
+#define BA0_FCHS_RCI(x) (1<<(1+(((x)&3)<<3))) /* Right Channel In */
+#define BA0_FCHS_LCI(x) (1<<(0+(((x)&3)<<3))) /* Left Channel In */
+
+#define BA0_FSIC0 0x0210 /* FIFO Status and Interrupt Control 0 */
+#define BA0_FSIC1 0x0214 /* FIFO Status and Interrupt Control 1 */
+#define BA0_FSIC2 0x0218 /* FIFO Status and Interrupt Control 2 */
+#define BA0_FSIC3 0x021c /* FIFO Status and Interrupt Control 3 */
+
+#define BA0_FSIC_FIC(x) (((x)&0x7f)<<24) /* FIFO Interrupt Count */
+#define BA0_FSIC_FORIE (1<<23) /* FIFO OverRun Interrupt Enable */
+#define BA0_FSIC_FURIE (1<<22) /* FIFO UnderRun Interrupt Enable */
+#define BA0_FSIC_FSCIE (1<<16) /* FIFO Sample Count Interrupt Enable */
+#define BA0_FSIC_FSC(x) (((x)&0x7f)<<8) /* FIFO Sample Count */
+#define BA0_FSIC_FOR (1<<7) /* FIFO OverRun */
+#define BA0_FSIC_FUR (1<<6) /* FIFO UnderRun */
+#define BA0_FSIC_FSCR (1<<0) /* FIFO Sample Count Reached */
+
+#define BA0_PMCS 0x0344 /* Power Management Control/Status */
+#define BA0_CWPR 0x03e0 /* Configuration Write Protect */
+
+#define BA0_EPPMC 0x03e4 /* Extended PCI Power Management Control */
+#define BA0_EPPMC_FPDN (1<<14) /* Full Power DowN */
+
+#define BA0_GPIOR 0x03e8 /* GPIO Pin Interface Register */
+
+#define BA0_SPMC 0x03ec /* Serial Port Power Management Control (& ASDIN2 enable) */
+#define BA0_SPMC_GIPPEN (1<<15) /* GP INT Primary PME# Enable */
+#define BA0_SPMC_GISPEN (1<<14) /* GP INT Secondary PME# Enable */
+#define BA0_SPMC_EESPD (1<<9) /* EEPROM Serial Port Disable */
+#define BA0_SPMC_ASDI2E (1<<8) /* ASDIN2 Enable */
+#define BA0_SPMC_ASDO (1<<7) /* Asynchronous ASDOUT Assertion */
+#define BA0_SPMC_WUP2 (1<<3) /* Wakeup for Secondary Input */
+#define BA0_SPMC_WUP1 (1<<2) /* Wakeup for Primary Input */
+#define BA0_SPMC_ASYNC (1<<1) /* Asynchronous ASYNC Assertion */
+#define BA0_SPMC_RSTN (1<<0) /* Reset Not! */
+
+#define BA0_CFLR 0x03f0 /* Configuration Load Register (EEPROM or BIOS) */
+#define BA0_CFLR_DEFAULT 0x00000001 /* CFLR must be in AC97 link mode */
+#define BA0_IISR 0x03f4 /* ISA Interrupt Select */
+#define BA0_TMS 0x03f8 /* Test Register */
+#define BA0_SSVID 0x03fc /* Subsystem ID register */
+
+#define BA0_CLKCR1 0x0400 /* Clock Control Register 1 */
+#define BA0_CLKCR1_CLKON (1<<25) /* Read Only */
+#define BA0_CLKCR1_DLLRDY (1<<24) /* DLL Ready */
+#define BA0_CLKCR1_DLLOS (1<<6) /* DLL Output Select */
+#define BA0_CLKCR1_SWCE (1<<5) /* Clock Enable */
+#define BA0_CLKCR1_DLLP (1<<4) /* DLL PowerUp */
+#define BA0_CLKCR1_DLLSS (((x)&3)<<3) /* DLL Source Select */
+
+#define BA0_FRR 0x0410 /* Feature Reporting Register */
+#define BA0_SLT12O 0x041c /* Slot 12 GPIO Output Register for AC-Link */
+
+#define BA0_SERMC 0x0420 /* Serial Port Master Control */
+#define BA0_SERMC_FCRN (1<<27) /* Force Codec Ready Not */
+#define BA0_SERMC_ODSEN2 (1<<25) /* On-Demand Support Enable ASDIN2 */
+#define BA0_SERMC_ODSEN1 (1<<24) /* On-Demand Support Enable ASDIN1 */
+#define BA0_SERMC_SXLB (1<<21) /* ASDIN2 to ASDOUT Loopback */
+#define BA0_SERMC_SLB (1<<20) /* ASDOUT to ASDIN2 Loopback */
+#define BA0_SERMC_LOVF (1<<19) /* Loopback Output Valid Frame bit */
+#define BA0_SERMC_TCID(x) (((x)&3)<<16) /* Target Secondary Codec ID */
+#define BA0_SERMC_PXLB (5<<1) /* Primary Port External Loopback */
+#define BA0_SERMC_PLB (4<<1) /* Primary Port Internal Loopback */
+#define BA0_SERMC_PTC (7<<1) /* Port Timing Configuration */
+#define BA0_SERMC_PTC_AC97 (1<<1) /* AC97 mode */
+#define BA0_SERMC_MSPE (1<<0) /* Master Serial Port Enable */
+
+#define BA0_SERC1 0x0428 /* Serial Port Configuration 1 */
+#define BA0_SERC1_SO1F(x) (((x)&7)>>1) /* Primary Output Port Format */
+#define BA0_SERC1_AC97 (1<<1)
+#define BA0_SERC1_SO1EN (1<<0) /* Primary Output Port Enable */
+
+#define BA0_SERC2 0x042c /* Serial Port Configuration 2 */
+#define BA0_SERC2_SI1F(x) (((x)&7)>>1) /* Primary Input Port Format */
+#define BA0_SERC2_AC97 (1<<1)
+#define BA0_SERC2_SI1EN (1<<0) /* Primary Input Port Enable */
+
+#define BA0_SLT12M 0x045c /* Slot 12 Monitor Register for Primary AC-Link */
+
+#define BA0_ACCTL 0x0460 /* AC'97 Control */
+#define BA0_ACCTL_TC (1<<6) /* Target Codec */
+#define BA0_ACCTL_CRW (1<<4) /* 0=Write, 1=Read Command */
+#define BA0_ACCTL_DCV (1<<3) /* Dynamic Command Valid */
+#define BA0_ACCTL_VFRM (1<<2) /* Valid Frame */
+#define BA0_ACCTL_ESYN (1<<1) /* Enable Sync */
+
+#define BA0_ACSTS 0x0464 /* AC'97 Status */
+#define BA0_ACSTS_VSTS (1<<1) /* Valid Status */
+#define BA0_ACSTS_CRDY (1<<0) /* Codec Ready */
+
+#define BA0_ACOSV 0x0468 /* AC'97 Output Slot Valid */
+#define BA0_ACOSV_SLV(x) (1<<((x)-3))
+
+#define BA0_ACCAD 0x046c /* AC'97 Command Address */
+#define BA0_ACCDA 0x0470 /* AC'97 Command Data */
+
+#define BA0_ACISV 0x0474 /* AC'97 Input Slot Valid */
+#define BA0_ACISV_SLV(x) (1<<((x)-3))
+
+#define BA0_ACSAD 0x0478 /* AC'97 Status Address */
+#define BA0_ACSDA 0x047c /* AC'97 Status Data */
+#define BA0_JSPT 0x0480 /* Joystick poll/trigger */
+#define BA0_JSCTL 0x0484 /* Joystick control */
+#define BA0_JSC1 0x0488 /* Joystick control */
+#define BA0_JSC2 0x048c /* Joystick control */
+#define BA0_JSIO 0x04a0
+
+#define BA0_MIDCR 0x0490 /* MIDI Control */
+#define BA0_MIDCR_MRST (1<<5) /* Reset MIDI Interface */
+#define BA0_MIDCR_MLB (1<<4) /* MIDI Loop Back Enable */
+#define BA0_MIDCR_TIE (1<<3) /* MIDI Transmuit Interrupt Enable */
+#define BA0_MIDCR_RIE (1<<2) /* MIDI Receive Interrupt Enable */
+#define BA0_MIDCR_RXE (1<<1) /* MIDI Receive Enable */
+#define BA0_MIDCR_TXE (1<<0) /* MIDI Transmit Enable */
+
+#define BA0_MIDCMD 0x0494 /* MIDI Command (wo) */
+
+#define BA0_MIDSR 0x0494 /* MIDI Status (ro) */
+#define BA0_MIDSR_RDA (1<<15) /* Sticky bit (RBE 1->0) */
+#define BA0_MIDSR_TBE (1<<14) /* Sticky bit (TBF 0->1) */
+#define BA0_MIDSR_RBE (1<<7) /* Receive Buffer Empty */
+#define BA0_MIDSR_TBF (1<<6) /* Transmit Buffer Full */
+
+#define BA0_MIDWP 0x0498 /* MIDI Write */
+#define BA0_MIDRP 0x049c /* MIDI Read (ro) */
+
+#define BA0_AODSD1 0x04a8 /* AC'97 On-Demand Slot Disable for primary link (ro) */
+#define BA0_AODSD1_NDS(x) (1<<((x)-3))
+
+#define BA0_AODSD2 0x04ac /* AC'97 On-Demand Slot Disable for secondary link (ro) */
+#define BA0_AODSD2_NDS(x) (1<<((x)-3))
+
+#define BA0_CFGI 0x04b0 /* Configure Interface (EEPROM interface) */
+#define BA0_SLT12M2 0x04dc /* Slot 12 Monitor Register 2 for secondary AC-link */
+#define BA0_ACSTS2 0x04e4 /* AC'97 Status Register 2 */
+#define BA0_ACISV2 0x04f4 /* AC'97 Input Slot Valid Register 2 */
+#define BA0_ACSAD2 0x04f8 /* AC'97 Status Address Register 2 */
+#define BA0_ACSDA2 0x04fc /* AC'97 Status Data Register 2 */
+#define BA0_FMSR 0x0730 /* FM Synthesis Status (ro) */
+#define BA0_B0AP 0x0730 /* FM Bank 0 Address Port (wo) */
+#define BA0_FMDP 0x0734 /* FM Data Port */
+#define BA0_B1AP 0x0738 /* FM Bank 1 Address Port */
+#define BA0_B1DP 0x073c /* FM Bank 1 Data Port */
+
+#define BA0_SSPM 0x0740 /* Sound System Power Management */
+#define BA0_SSPM_MIXEN (1<<6) /* Playback SRC + FM/Wavetable MIX */
+#define BA0_SSPM_CSRCEN (1<<5) /* Capture Sample Rate Converter Enable */
+#define BA0_SSPM_PSRCEN (1<<4) /* Playback Sample Rate Converter Enable */
+#define BA0_SSPM_JSEN (1<<3) /* Joystick Enable */
+#define BA0_SSPM_ACLEN (1<<2) /* Serial Port Engine and AC-Link Enable */
+#define BA0_SSPM_FMEN (1<<1) /* FM Synthesis Block Enable */
+
+#define BA0_DACSR 0x0744 /* DAC Sample Rate - Playback SRC */
+#define BA0_ADCSR 0x0748 /* ADC Sample Rate - Capture SRC */
+
+#define BA0_SSCR 0x074c /* Sound System Control Register */
+#define BA0_SSCR_HVS1 (1<<23) /* Hardwave Volume Step (0=1,1=2) */
+#define BA0_SSCR_MVCS (1<<19) /* Master Volume Codec Select */
+#define BA0_SSCR_MVLD (1<<18) /* Master Volume Line Out Disable */
+#define BA0_SSCR_MVAD (1<<17) /* Master Volume Alternate Out Disable */
+#define BA0_SSCR_MVMD (1<<16) /* Master Volume Mono Out Disable */
+#define BA0_SSCR_XLPSRC (1<<8) /* External SRC Loopback Mode */
+#define BA0_SSCR_LPSRC (1<<7) /* SRC Loopback Mode */
+#define BA0_SSCR_CDTX (1<<5) /* CD Transfer Data */
+#define BA0_SSCR_HVC (1<<3) /* Harware Volume Control Enable */
+
+#define BA0_FMLVC 0x0754 /* FM Synthesis Left Volume Control */
+#define BA0_FMRVC 0x0758 /* FM Synthesis Right Volume Control */
+#define BA0_SRCSA 0x075c /* SRC Slot Assignments */
+#define BA0_PPLVC 0x0760 /* PCM Playback Left Volume Control */
+#define BA0_PPRVC 0x0764 /* PCM Playback Right Volume Control */
+#define BA0_PASR 0x0768 /* playback sample rate */
+#define BA0_CASR 0x076C /* capture sample rate */
+
+/* Source Slot Numbers - Playback */
+#define SRCSLOT_LEFT_PCM_PLAYBACK 0
+#define SRCSLOT_RIGHT_PCM_PLAYBACK 1
+#define SRCSLOT_PHONE_LINE_1_DAC 2
+#define SRCSLOT_CENTER_PCM_PLAYBACK 3
+#define SRCSLOT_LEFT_SURROUND_PCM_PLAYBACK 4
+#define SRCSLOT_RIGHT_SURROUND_PCM_PLAYBACK 5
+#define SRCSLOT_LFE_PCM_PLAYBACK 6
+#define SRCSLOT_PHONE_LINE_2_DAC 7
+#define SRCSLOT_HEADSET_DAC 8
+#define SRCSLOT_LEFT_WT 29 /* invalid for BA0_SRCSA */
+#define SRCSLOT_RIGHT_WT 30 /* invalid for BA0_SRCSA */
+
+/* Source Slot Numbers - Capture */
+#define SRCSLOT_LEFT_PCM_RECORD 10
+#define SRCSLOT_RIGHT_PCM_RECORD 11
+#define SRCSLOT_PHONE_LINE_1_ADC 12
+#define SRCSLOT_MIC_ADC 13
+#define SRCSLOT_PHONE_LINE_2_ADC 17
+#define SRCSLOT_HEADSET_ADC 18
+#define SRCSLOT_SECONDARY_LEFT_PCM_RECORD 20
+#define SRCSLOT_SECONDARY_RIGHT_PCM_RECORD 21
+#define SRCSLOT_SECONDARY_PHONE_LINE_1_ADC 22
+#define SRCSLOT_SECONDARY_MIC_ADC 23
+#define SRCSLOT_SECONDARY_PHONE_LINE_2_ADC 27
+#define SRCSLOT_SECONDARY_HEADSET_ADC 28
+
+/* Source Slot Numbers - Others */
+#define SRCSLOT_POWER_DOWN 31
+
+/* MIDI modes */
+#define CS4281_MODE_OUTPUT (1<<0)
+#define CS4281_MODE_INPUT (1<<1)
+
+/* joystick bits */
+/* Bits for JSPT */
+#define JSPT_CAX 0x00000001
+#define JSPT_CAY 0x00000002
+#define JSPT_CBX 0x00000004
+#define JSPT_CBY 0x00000008
+#define JSPT_BA1 0x00000010
+#define JSPT_BA2 0x00000020
+#define JSPT_BB1 0x00000040
+#define JSPT_BB2 0x00000080
+
+/* Bits for JSCTL */
+#define JSCTL_SP_MASK 0x00000003
+#define JSCTL_SP_SLOW 0x00000000
+#define JSCTL_SP_MEDIUM_SLOW 0x00000001
+#define JSCTL_SP_MEDIUM_FAST 0x00000002
+#define JSCTL_SP_FAST 0x00000003
+#define JSCTL_ARE 0x00000004
+
+/* Data register pairs masks */
+#define JSC1_Y1V_MASK 0x0000FFFF
+#define JSC1_X1V_MASK 0xFFFF0000
+#define JSC1_Y1V_SHIFT 0
+#define JSC1_X1V_SHIFT 16
+#define JSC2_Y2V_MASK 0x0000FFFF
+#define JSC2_X2V_MASK 0xFFFF0000
+#define JSC2_Y2V_SHIFT 0
+#define JSC2_X2V_SHIFT 16
+
+/* JS GPIO */
+#define JSIO_DAX 0x00000001
+#define JSIO_DAY 0x00000002
+#define JSIO_DBX 0x00000004
+#define JSIO_DBY 0x00000008
+#define JSIO_AXOE 0x00000010
+#define JSIO_AYOE 0x00000020
+#define JSIO_BXOE 0x00000040
+#define JSIO_BYOE 0x00000080
+
+/*
+ *
+ */
+
+struct cs4281_dma {
+ struct snd_pcm_substream *substream;
+ unsigned int regDBA; /* offset to DBA register */
+ unsigned int regDCA; /* offset to DCA register */
+ unsigned int regDBC; /* offset to DBC register */
+ unsigned int regDCC; /* offset to DCC register */
+ unsigned int regDMR; /* offset to DMR register */
+ unsigned int regDCR; /* offset to DCR register */
+ unsigned int regHDSR; /* offset to HDSR register */
+ unsigned int regFCR; /* offset to FCR register */
+ unsigned int regFSIC; /* offset to FSIC register */
+ unsigned int valDMR; /* DMA mode */
+ unsigned int valDCR; /* DMA command */
+ unsigned int valFCR; /* FIFO control */
+ unsigned int fifo_offset; /* FIFO offset within BA1 */
+ unsigned char left_slot; /* FIFO left slot */
+ unsigned char right_slot; /* FIFO right slot */
+ int frag; /* period number */
+};
+
+#define SUSPEND_REGISTERS 20
+
+struct cs4281 {
+ int irq;
+
+ void __iomem *ba0; /* virtual (accessible) address */
+ void __iomem *ba1; /* virtual (accessible) address */
+ unsigned long ba0_addr;
+ unsigned long ba1_addr;
+
+ int dual_codec;
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97;
+ struct snd_ac97 *ac97_secondary;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *midi_input;
+ struct snd_rawmidi_substream *midi_output;
+
+ struct cs4281_dma dma[4];
+
+ unsigned char src_left_play_slot;
+ unsigned char src_right_play_slot;
+ unsigned char src_left_rec_slot;
+ unsigned char src_right_rec_slot;
+
+ unsigned int spurious_dhtc_irq;
+ unsigned int spurious_dtc_irq;
+
+ spinlock_t reg_lock;
+ unsigned int midcr;
+ unsigned int uartm;
+
+ struct gameport *gameport;
+
+#ifdef CONFIG_PM
+ u32 suspend_regs[SUSPEND_REGISTERS];
+#endif
+
+};
+
+static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id);
+
+static struct pci_device_id snd_cs4281_ids[] = {
+ { 0x1013, 0x6005, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4281 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_cs4281_ids);
+
+/*
+ * constants
+ */
+
+#define CS4281_FIFO_SIZE 32
+
+/*
+ * common I/O routines
+ */
+
+static inline void snd_cs4281_pokeBA0(struct cs4281 *chip, unsigned long offset,
+ unsigned int val)
+{
+ writel(val, chip->ba0 + offset);
+}
+
+static inline unsigned int snd_cs4281_peekBA0(struct cs4281 *chip, unsigned long offset)
+{
+ return readl(chip->ba0 + offset);
+}
+
+static void snd_cs4281_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ /*
+ * 1. Write ACCAD = Command Address Register = 46Ch for AC97 register address
+ * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97
+ * 3. Write ACCTL = Control Register = 460h for initiating the write
+ * 4. Read ACCTL = 460h, DCV should be reset by now and 460h = 07h
+ * 5. if DCV not cleared, break and return error
+ */
+ struct cs4281 *chip = ac97->private_data;
+ int count;
+
+ /*
+ * Setup the AC97 control registers on the CS461x to send the
+ * appropriate command to the AC97 to perform the read.
+ * ACCAD = Command Address Register = 46Ch
+ * ACCDA = Command Data Register = 470h
+ * ACCTL = Control Register = 460h
+ * set DCV - will clear when process completed
+ * reset CRW - Write command
+ * set VFRM - valid frame enabled
+ * set ESYN - ASYNC generation enabled
+ * set RSTN - ARST# inactive, AC97 codec not reset
+ */
+ snd_cs4281_pokeBA0(chip, BA0_ACCAD, reg);
+ snd_cs4281_pokeBA0(chip, BA0_ACCDA, val);
+ snd_cs4281_pokeBA0(chip, BA0_ACCTL, BA0_ACCTL_DCV | BA0_ACCTL_VFRM |
+ BA0_ACCTL_ESYN | (ac97->num ? BA0_ACCTL_TC : 0));
+ for (count = 0; count < 2000; count++) {
+ /*
+ * First, we want to wait for a short time.
+ */
+ udelay(10);
+ /*
+ * Now, check to see if the write has completed.
+ * ACCTL = 460h, DCV should be reset by now and 460h = 07h
+ */
+ if (!(snd_cs4281_peekBA0(chip, BA0_ACCTL) & BA0_ACCTL_DCV)) {
+ return;
+ }
+ }
+ snd_printk(KERN_ERR "AC'97 write problem, reg = 0x%x, val = 0x%x\n", reg, val);
+}
+
+static unsigned short snd_cs4281_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct cs4281 *chip = ac97->private_data;
+ int count;
+ unsigned short result;
+ // FIXME: volatile is necessary in the following due to a bug of
+ // some gcc versions
+ volatile int ac97_num = ((volatile struct snd_ac97 *)ac97)->num;
+
+ /*
+ * 1. Write ACCAD = Command Address Register = 46Ch for AC97 register address
+ * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97
+ * 3. Write ACCTL = Control Register = 460h for initiating the write
+ * 4. Read ACCTL = 460h, DCV should be reset by now and 460h = 17h
+ * 5. if DCV not cleared, break and return error
+ * 6. Read ACSTS = Status Register = 464h, check VSTS bit
+ */
+
+ snd_cs4281_peekBA0(chip, ac97_num ? BA0_ACSDA2 : BA0_ACSDA);
+
+ /*
+ * Setup the AC97 control registers on the CS461x to send the
+ * appropriate command to the AC97 to perform the read.
+ * ACCAD = Command Address Register = 46Ch
+ * ACCDA = Command Data Register = 470h
+ * ACCTL = Control Register = 460h
+ * set DCV - will clear when process completed
+ * set CRW - Read command
+ * set VFRM - valid frame enabled
+ * set ESYN - ASYNC generation enabled
+ * set RSTN - ARST# inactive, AC97 codec not reset
+ */
+
+ snd_cs4281_pokeBA0(chip, BA0_ACCAD, reg);
+ snd_cs4281_pokeBA0(chip, BA0_ACCDA, 0);
+ snd_cs4281_pokeBA0(chip, BA0_ACCTL, BA0_ACCTL_DCV | BA0_ACCTL_CRW |
+ BA0_ACCTL_VFRM | BA0_ACCTL_ESYN |
+ (ac97_num ? BA0_ACCTL_TC : 0));
+
+
+ /*
+ * Wait for the read to occur.
+ */
+ for (count = 0; count < 500; count++) {
+ /*
+ * First, we want to wait for a short time.
+ */
+ udelay(10);
+ /*
+ * Now, check to see if the read has completed.
+ * ACCTL = 460h, DCV should be reset by now and 460h = 17h
+ */
+ if (!(snd_cs4281_peekBA0(chip, BA0_ACCTL) & BA0_ACCTL_DCV))
+ goto __ok1;
+ }
+
+ snd_printk(KERN_ERR "AC'97 read problem (ACCTL_DCV), reg = 0x%x\n", reg);
+ result = 0xffff;
+ goto __end;
+
+ __ok1:
+ /*
+ * Wait for the valid status bit to go active.
+ */
+ for (count = 0; count < 100; count++) {
+ /*
+ * Read the AC97 status register.
+ * ACSTS = Status Register = 464h
+ * VSTS - Valid Status
+ */
+ if (snd_cs4281_peekBA0(chip, ac97_num ? BA0_ACSTS2 : BA0_ACSTS) & BA0_ACSTS_VSTS)
+ goto __ok2;
+ udelay(10);
+ }
+
+ snd_printk(KERN_ERR "AC'97 read problem (ACSTS_VSTS), reg = 0x%x\n", reg);
+ result = 0xffff;
+ goto __end;
+
+ __ok2:
+ /*
+ * Read the data returned from the AC97 register.
+ * ACSDA = Status Data Register = 474h
+ */
+ result = snd_cs4281_peekBA0(chip, ac97_num ? BA0_ACSDA2 : BA0_ACSDA);
+
+ __end:
+ return result;
+}
+
+/*
+ * PCM part
+ */
+
+static int snd_cs4281_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct cs4281_dma *dma = substream->runtime->private_data;
+ struct cs4281 *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dma->valDCR |= BA0_DCR_MSK;
+ dma->valFCR |= BA0_FCR_FEN;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dma->valDCR &= ~BA0_DCR_MSK;
+ dma->valFCR &= ~BA0_FCR_FEN;
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ snd_cs4281_pokeBA0(chip, dma->regDMR, dma->valDMR & ~BA0_DMR_DMA);
+ dma->valDMR |= BA0_DMR_DMA;
+ dma->valDCR &= ~BA0_DCR_MSK;
+ dma->valFCR |= BA0_FCR_FEN;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ dma->valDMR &= ~(BA0_DMR_DMA|BA0_DMR_POLL);
+ dma->valDCR |= BA0_DCR_MSK;
+ dma->valFCR &= ~BA0_FCR_FEN;
+ /* Leave wave playback FIFO enabled for FM */
+ if (dma->regFCR != BA0_FCR0)
+ dma->valFCR &= ~BA0_FCR_FEN;
+ break;
+ default:
+ spin_unlock(&chip->reg_lock);
+ return -EINVAL;
+ }
+ snd_cs4281_pokeBA0(chip, dma->regDMR, dma->valDMR);
+ snd_cs4281_pokeBA0(chip, dma->regFCR, dma->valFCR);
+ snd_cs4281_pokeBA0(chip, dma->regDCR, dma->valDCR);
+ spin_unlock(&chip->reg_lock);
+ return 0;
+}
+
+static unsigned int snd_cs4281_rate(unsigned int rate, unsigned int *real_rate)
+{
+ unsigned int val = ~0;
+
+ if (real_rate)
+ *real_rate = rate;
+ /* special "hardcoded" rates */
+ switch (rate) {
+ case 8000: return 5;
+ case 11025: return 4;
+ case 16000: return 3;
+ case 22050: return 2;
+ case 44100: return 1;
+ case 48000: return 0;
+ default:
+ goto __variable;
+ }
+ __variable:
+ val = 1536000 / rate;
+ if (real_rate)
+ *real_rate = 1536000 / val;
+ return val;
+}
+
+static void snd_cs4281_mode(struct cs4281 *chip, struct cs4281_dma *dma,
+ struct snd_pcm_runtime *runtime,
+ int capture, int src)
+{
+ int rec_mono;
+
+ dma->valDMR = BA0_DMR_TYPE_SINGLE | BA0_DMR_AUTO |
+ (capture ? BA0_DMR_TR_WRITE : BA0_DMR_TR_READ);
+ if (runtime->channels == 1)
+ dma->valDMR |= BA0_DMR_MONO;
+ if (snd_pcm_format_unsigned(runtime->format) > 0)
+ dma->valDMR |= BA0_DMR_USIGN;
+ if (snd_pcm_format_big_endian(runtime->format) > 0)
+ dma->valDMR |= BA0_DMR_BEND;
+ switch (snd_pcm_format_width(runtime->format)) {
+ case 8: dma->valDMR |= BA0_DMR_SIZE8;
+ if (runtime->channels == 1)
+ dma->valDMR |= BA0_DMR_SWAPC;
+ break;
+ case 32: dma->valDMR |= BA0_DMR_SIZE20; break;
+ }
+ dma->frag = 0; /* for workaround */
+ dma->valDCR = BA0_DCR_TCIE | BA0_DCR_MSK;
+ if (runtime->buffer_size != runtime->period_size)
+ dma->valDCR |= BA0_DCR_HTCIE;
+ /* Initialize DMA */
+ snd_cs4281_pokeBA0(chip, dma->regDBA, runtime->dma_addr);
+ snd_cs4281_pokeBA0(chip, dma->regDBC, runtime->buffer_size - 1);
+ rec_mono = (chip->dma[1].valDMR & BA0_DMR_MONO) == BA0_DMR_MONO;
+ snd_cs4281_pokeBA0(chip, BA0_SRCSA, (chip->src_left_play_slot << 0) |
+ (chip->src_right_play_slot << 8) |
+ (chip->src_left_rec_slot << 16) |
+ ((rec_mono ? 31 : chip->src_right_rec_slot) << 24));
+ if (!src)
+ goto __skip_src;
+ if (!capture) {
+ if (dma->left_slot == chip->src_left_play_slot) {
+ unsigned int val = snd_cs4281_rate(runtime->rate, NULL);
+ snd_BUG_ON(dma->right_slot != chip->src_right_play_slot);
+ snd_cs4281_pokeBA0(chip, BA0_DACSR, val);
+ }
+ } else {
+ if (dma->left_slot == chip->src_left_rec_slot) {
+ unsigned int val = snd_cs4281_rate(runtime->rate, NULL);
+ snd_BUG_ON(dma->right_slot != chip->src_right_rec_slot);
+ snd_cs4281_pokeBA0(chip, BA0_ADCSR, val);
+ }
+ }
+ __skip_src:
+ /* Deactivate wave playback FIFO before changing slot assignments */
+ if (dma->regFCR == BA0_FCR0)
+ snd_cs4281_pokeBA0(chip, dma->regFCR, snd_cs4281_peekBA0(chip, dma->regFCR) & ~BA0_FCR_FEN);
+ /* Initialize FIFO */
+ dma->valFCR = BA0_FCR_LS(dma->left_slot) |
+ BA0_FCR_RS(capture && (dma->valDMR & BA0_DMR_MONO) ? 31 : dma->right_slot) |
+ BA0_FCR_SZ(CS4281_FIFO_SIZE) |
+ BA0_FCR_OF(dma->fifo_offset);
+ snd_cs4281_pokeBA0(chip, dma->regFCR, dma->valFCR | (capture ? BA0_FCR_PSH : 0));
+ /* Activate FIFO again for FM playback */
+ if (dma->regFCR == BA0_FCR0)
+ snd_cs4281_pokeBA0(chip, dma->regFCR, dma->valFCR | BA0_FCR_FEN);
+ /* Clear FIFO Status and Interrupt Control Register */
+ snd_cs4281_pokeBA0(chip, dma->regFSIC, 0);
+}
+
+static int snd_cs4281_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_cs4281_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_cs4281_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cs4281_dma *dma = runtime->private_data;
+ struct cs4281 *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ snd_cs4281_mode(chip, dma, runtime, 0, 1);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cs4281_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cs4281_dma *dma = runtime->private_data;
+ struct cs4281 *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ snd_cs4281_mode(chip, dma, runtime, 1, 1);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cs4281_dma *dma = runtime->private_data;
+ struct cs4281 *chip = snd_pcm_substream_chip(substream);
+
+ // printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies);
+ return runtime->buffer_size -
+ snd_cs4281_peekBA0(chip, dma->regDCC) - 1;
+}
+
+static struct snd_pcm_hardware snd_cs4281_playback =
+{
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (512*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (512*1024),
+ .periods_min = 1,
+ .periods_max = 2,
+ .fifo_size = CS4281_FIFO_SIZE,
+};
+
+static struct snd_pcm_hardware snd_cs4281_capture =
+{
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (512*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (512*1024),
+ .periods_min = 1,
+ .periods_max = 2,
+ .fifo_size = CS4281_FIFO_SIZE,
+};
+
+static int snd_cs4281_playback_open(struct snd_pcm_substream *substream)
+{
+ struct cs4281 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cs4281_dma *dma;
+
+ dma = &chip->dma[0];
+ dma->substream = substream;
+ dma->left_slot = 0;
+ dma->right_slot = 1;
+ runtime->private_data = dma;
+ runtime->hw = snd_cs4281_playback;
+ /* should be detected from the AC'97 layer, but it seems
+ that although CS4297A rev B reports 18-bit ADC resolution,
+ samples are 20-bit */
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 20);
+ return 0;
+}
+
+static int snd_cs4281_capture_open(struct snd_pcm_substream *substream)
+{
+ struct cs4281 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cs4281_dma *dma;
+
+ dma = &chip->dma[1];
+ dma->substream = substream;
+ dma->left_slot = 10;
+ dma->right_slot = 11;
+ runtime->private_data = dma;
+ runtime->hw = snd_cs4281_capture;
+ /* should be detected from the AC'97 layer, but it seems
+ that although CS4297A rev B reports 18-bit ADC resolution,
+ samples are 20-bit */
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 20);
+ return 0;
+}
+
+static int snd_cs4281_playback_close(struct snd_pcm_substream *substream)
+{
+ struct cs4281_dma *dma = substream->runtime->private_data;
+
+ dma->substream = NULL;
+ return 0;
+}
+
+static int snd_cs4281_capture_close(struct snd_pcm_substream *substream)
+{
+ struct cs4281_dma *dma = substream->runtime->private_data;
+
+ dma->substream = NULL;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_cs4281_playback_ops = {
+ .open = snd_cs4281_playback_open,
+ .close = snd_cs4281_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs4281_hw_params,
+ .hw_free = snd_cs4281_hw_free,
+ .prepare = snd_cs4281_playback_prepare,
+ .trigger = snd_cs4281_trigger,
+ .pointer = snd_cs4281_pointer,
+};
+
+static struct snd_pcm_ops snd_cs4281_capture_ops = {
+ .open = snd_cs4281_capture_open,
+ .close = snd_cs4281_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs4281_hw_params,
+ .hw_free = snd_cs4281_hw_free,
+ .prepare = snd_cs4281_capture_prepare,
+ .trigger = snd_cs4281_trigger,
+ .pointer = snd_cs4281_pointer,
+};
+
+static int __devinit snd_cs4281_pcm(struct cs4281 * chip, int device,
+ struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ err = snd_pcm_new(chip->card, "CS4281", device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4281_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cs4281_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "CS4281");
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 512*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/*
+ * Mixer section
+ */
+
+#define CS_VOL_MASK 0x1f
+
+static int snd_cs4281_info_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = CS_VOL_MASK;
+ return 0;
+}
+
+static int snd_cs4281_get_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cs4281 *chip = snd_kcontrol_chip(kcontrol);
+ int regL = (kcontrol->private_value >> 16) & 0xffff;
+ int regR = kcontrol->private_value & 0xffff;
+ int volL, volR;
+
+ volL = CS_VOL_MASK - (snd_cs4281_peekBA0(chip, regL) & CS_VOL_MASK);
+ volR = CS_VOL_MASK - (snd_cs4281_peekBA0(chip, regR) & CS_VOL_MASK);
+
+ ucontrol->value.integer.value[0] = volL;
+ ucontrol->value.integer.value[1] = volR;
+ return 0;
+}
+
+static int snd_cs4281_put_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cs4281 *chip = snd_kcontrol_chip(kcontrol);
+ int change = 0;
+ int regL = (kcontrol->private_value >> 16) & 0xffff;
+ int regR = kcontrol->private_value & 0xffff;
+ int volL, volR;
+
+ volL = CS_VOL_MASK - (snd_cs4281_peekBA0(chip, regL) & CS_VOL_MASK);
+ volR = CS_VOL_MASK - (snd_cs4281_peekBA0(chip, regR) & CS_VOL_MASK);
+
+ if (ucontrol->value.integer.value[0] != volL) {
+ volL = CS_VOL_MASK - (ucontrol->value.integer.value[0] & CS_VOL_MASK);
+ snd_cs4281_pokeBA0(chip, regL, volL);
+ change = 1;
+ }
+ if (ucontrol->value.integer.value[1] != volR) {
+ volR = CS_VOL_MASK - (ucontrol->value.integer.value[1] & CS_VOL_MASK);
+ snd_cs4281_pokeBA0(chip, regR, volR);
+ change = 1;
+ }
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_dsp, -4650, 150, 0);
+
+static struct snd_kcontrol_new snd_cs4281_fm_vol =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Synth Playback Volume",
+ .info = snd_cs4281_info_volume,
+ .get = snd_cs4281_get_volume,
+ .put = snd_cs4281_put_volume,
+ .private_value = ((BA0_FMLVC << 16) | BA0_FMRVC),
+ .tlv = { .p = db_scale_dsp },
+};
+
+static struct snd_kcontrol_new snd_cs4281_pcm_vol =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Stream Playback Volume",
+ .info = snd_cs4281_info_volume,
+ .get = snd_cs4281_get_volume,
+ .put = snd_cs4281_put_volume,
+ .private_value = ((BA0_PPLVC << 16) | BA0_PPRVC),
+ .tlv = { .p = db_scale_dsp },
+};
+
+static void snd_cs4281_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct cs4281 *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void snd_cs4281_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct cs4281 *chip = ac97->private_data;
+ if (ac97->num)
+ chip->ac97_secondary = NULL;
+ else
+ chip->ac97 = NULL;
+}
+
+static int __devinit snd_cs4281_mixer(struct cs4281 * chip)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_cs4281_ac97_write,
+ .read = snd_cs4281_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus)) < 0)
+ return err;
+ chip->ac97_bus->private_free = snd_cs4281_mixer_free_ac97_bus;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_cs4281_mixer_free_ac97;
+ if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97)) < 0)
+ return err;
+ if (chip->dual_codec) {
+ ac97.num = 1;
+ if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97_secondary)) < 0)
+ return err;
+ }
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4281_fm_vol, chip))) < 0)
+ return err;
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4281_pcm_vol, chip))) < 0)
+ return err;
+ return 0;
+}
+
+
+/*
+ * proc interface
+ */
+
+static void snd_cs4281_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct cs4281 *chip = entry->private_data;
+
+ snd_iprintf(buffer, "Cirrus Logic CS4281\n\n");
+ snd_iprintf(buffer, "Spurious half IRQs : %u\n", chip->spurious_dhtc_irq);
+ snd_iprintf(buffer, "Spurious end IRQs : %u\n", chip->spurious_dtc_irq);
+}
+
+static long snd_cs4281_BA0_read(struct snd_info_entry *entry,
+ void *file_private_data,
+ struct file *file, char __user *buf,
+ unsigned long count, unsigned long pos)
+{
+ long size;
+ struct cs4281 *chip = entry->private_data;
+
+ size = count;
+ if (pos + size > CS4281_BA0_SIZE)
+ size = (long)CS4281_BA0_SIZE - pos;
+ if (size > 0) {
+ if (copy_to_user_fromio(buf, chip->ba0 + pos, size))
+ return -EFAULT;
+ }
+ return size;
+}
+
+static long snd_cs4281_BA1_read(struct snd_info_entry *entry,
+ void *file_private_data,
+ struct file *file, char __user *buf,
+ unsigned long count, unsigned long pos)
+{
+ long size;
+ struct cs4281 *chip = entry->private_data;
+
+ size = count;
+ if (pos + size > CS4281_BA1_SIZE)
+ size = (long)CS4281_BA1_SIZE - pos;
+ if (size > 0) {
+ if (copy_to_user_fromio(buf, chip->ba1 + pos, size))
+ return -EFAULT;
+ }
+ return size;
+}
+
+static struct snd_info_entry_ops snd_cs4281_proc_ops_BA0 = {
+ .read = snd_cs4281_BA0_read,
+};
+
+static struct snd_info_entry_ops snd_cs4281_proc_ops_BA1 = {
+ .read = snd_cs4281_BA1_read,
+};
+
+static void __devinit snd_cs4281_proc_init(struct cs4281 * chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "cs4281", &entry))
+ snd_info_set_text_ops(entry, chip, snd_cs4281_proc_read);
+ if (! snd_card_proc_new(chip->card, "cs4281_BA0", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = chip;
+ entry->c.ops = &snd_cs4281_proc_ops_BA0;
+ entry->size = CS4281_BA0_SIZE;
+ }
+ if (! snd_card_proc_new(chip->card, "cs4281_BA1", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = chip;
+ entry->c.ops = &snd_cs4281_proc_ops_BA1;
+ entry->size = CS4281_BA1_SIZE;
+ }
+}
+
+/*
+ * joystick support
+ */
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+
+static void snd_cs4281_gameport_trigger(struct gameport *gameport)
+{
+ struct cs4281 *chip = gameport_get_port_data(gameport);
+
+ if (snd_BUG_ON(!chip))
+ return;
+ snd_cs4281_pokeBA0(chip, BA0_JSPT, 0xff);
+}
+
+static unsigned char snd_cs4281_gameport_read(struct gameport *gameport)
+{
+ struct cs4281 *chip = gameport_get_port_data(gameport);
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+ return snd_cs4281_peekBA0(chip, BA0_JSPT);
+}
+
+#ifdef COOKED_MODE
+static int snd_cs4281_gameport_cooked_read(struct gameport *gameport,
+ int *axes, int *buttons)
+{
+ struct cs4281 *chip = gameport_get_port_data(gameport);
+ unsigned js1, js2, jst;
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+
+ js1 = snd_cs4281_peekBA0(chip, BA0_JSC1);
+ js2 = snd_cs4281_peekBA0(chip, BA0_JSC2);
+ jst = snd_cs4281_peekBA0(chip, BA0_JSPT);
+
+ *buttons = (~jst >> 4) & 0x0F;
+
+ axes[0] = ((js1 & JSC1_Y1V_MASK) >> JSC1_Y1V_SHIFT) & 0xFFFF;
+ axes[1] = ((js1 & JSC1_X1V_MASK) >> JSC1_X1V_SHIFT) & 0xFFFF;
+ axes[2] = ((js2 & JSC2_Y2V_MASK) >> JSC2_Y2V_SHIFT) & 0xFFFF;
+ axes[3] = ((js2 & JSC2_X2V_MASK) >> JSC2_X2V_SHIFT) & 0xFFFF;
+
+ for (jst = 0; jst < 4; ++jst)
+ if (axes[jst] == 0xFFFF) axes[jst] = -1;
+ return 0;
+}
+#else
+#define snd_cs4281_gameport_cooked_read NULL
+#endif
+
+static int snd_cs4281_gameport_open(struct gameport *gameport, int mode)
+{
+ switch (mode) {
+#ifdef COOKED_MODE
+ case GAMEPORT_MODE_COOKED:
+ return 0;
+#endif
+ case GAMEPORT_MODE_RAW:
+ return 0;
+ default:
+ return -1;
+ }
+ return 0;
+}
+
+static int __devinit snd_cs4281_create_gameport(struct cs4281 *chip)
+{
+ struct gameport *gp;
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "cs4281: cannot allocate memory for gameport\n");
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "CS4281 Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+ gp->open = snd_cs4281_gameport_open;
+ gp->read = snd_cs4281_gameport_read;
+ gp->trigger = snd_cs4281_gameport_trigger;
+ gp->cooked_read = snd_cs4281_gameport_cooked_read;
+ gameport_set_port_data(gp, chip);
+
+ snd_cs4281_pokeBA0(chip, BA0_JSIO, 0xFF); // ?
+ snd_cs4281_pokeBA0(chip, BA0_JSCTL, JSCTL_SP_MEDIUM_SLOW);
+
+ gameport_register_port(gp);
+
+ return 0;
+}
+
+static void snd_cs4281_free_gameport(struct cs4281 *chip)
+{
+ if (chip->gameport) {
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+ }
+}
+#else
+static inline int snd_cs4281_create_gameport(struct cs4281 *chip) { return -ENOSYS; }
+static inline void snd_cs4281_free_gameport(struct cs4281 *chip) { }
+#endif /* CONFIG_GAMEPORT || (MODULE && CONFIG_GAMEPORT_MODULE) */
+
+static int snd_cs4281_free(struct cs4281 *chip)
+{
+ snd_cs4281_free_gameport(chip);
+
+ if (chip->irq >= 0)
+ synchronize_irq(chip->irq);
+
+ /* Mask interrupts */
+ snd_cs4281_pokeBA0(chip, BA0_HIMR, 0x7fffffff);
+ /* Stop the DLL Clock logic. */
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, 0);
+ /* Sound System Power Management - Turn Everything OFF */
+ snd_cs4281_pokeBA0(chip, BA0_SSPM, 0);
+ /* PCI interface - D3 state */
+ pci_set_power_state(chip->pci, 3);
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (chip->ba0)
+ iounmap(chip->ba0);
+ if (chip->ba1)
+ iounmap(chip->ba1);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+
+ kfree(chip);
+ return 0;
+}
+
+static int snd_cs4281_dev_free(struct snd_device *device)
+{
+ struct cs4281 *chip = device->device_data;
+ return snd_cs4281_free(chip);
+}
+
+static int snd_cs4281_chip_init(struct cs4281 *chip); /* defined below */
+
+static int __devinit snd_cs4281_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct cs4281 ** rchip,
+ int dual_codec)
+{
+ struct cs4281 *chip;
+ unsigned int tmp;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_cs4281_dev_free,
+ };
+
+ *rchip = NULL;
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&chip->reg_lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ pci_set_master(pci);
+ if (dual_codec < 0 || dual_codec > 3) {
+ snd_printk(KERN_ERR "invalid dual_codec option %d\n", dual_codec);
+ dual_codec = 0;
+ }
+ chip->dual_codec = dual_codec;
+
+ if ((err = pci_request_regions(pci, "CS4281")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->ba0_addr = pci_resource_start(pci, 0);
+ chip->ba1_addr = pci_resource_start(pci, 1);
+
+ chip->ba0 = pci_ioremap_bar(pci, 0);
+ chip->ba1 = pci_ioremap_bar(pci, 1);
+ if (!chip->ba0 || !chip->ba1) {
+ snd_cs4281_free(chip);
+ return -ENOMEM;
+ }
+
+ if (request_irq(pci->irq, snd_cs4281_interrupt, IRQF_SHARED,
+ "CS4281", chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_cs4281_free(chip);
+ return -ENOMEM;
+ }
+ chip->irq = pci->irq;
+
+ tmp = snd_cs4281_chip_init(chip);
+ if (tmp) {
+ snd_cs4281_free(chip);
+ return tmp;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_cs4281_free(chip);
+ return err;
+ }
+
+ snd_cs4281_proc_init(chip);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+}
+
+static int snd_cs4281_chip_init(struct cs4281 *chip)
+{
+ unsigned int tmp;
+ unsigned long end_time;
+ int retry_count = 2;
+
+ /* Having EPPMC.FPDN=1 prevent proper chip initialisation */
+ tmp = snd_cs4281_peekBA0(chip, BA0_EPPMC);
+ if (tmp & BA0_EPPMC_FPDN)
+ snd_cs4281_pokeBA0(chip, BA0_EPPMC, tmp & ~BA0_EPPMC_FPDN);
+
+ __retry:
+ tmp = snd_cs4281_peekBA0(chip, BA0_CFLR);
+ if (tmp != BA0_CFLR_DEFAULT) {
+ snd_cs4281_pokeBA0(chip, BA0_CFLR, BA0_CFLR_DEFAULT);
+ tmp = snd_cs4281_peekBA0(chip, BA0_CFLR);
+ if (tmp != BA0_CFLR_DEFAULT) {
+ snd_printk(KERN_ERR "CFLR setup failed (0x%x)\n", tmp);
+ return -EIO;
+ }
+ }
+
+ /* Set the 'Configuration Write Protect' register
+ * to 4281h. Allows vendor-defined configuration
+ * space between 0e4h and 0ffh to be written. */
+ snd_cs4281_pokeBA0(chip, BA0_CWPR, 0x4281);
+
+ if ((tmp = snd_cs4281_peekBA0(chip, BA0_SERC1)) != (BA0_SERC1_SO1EN | BA0_SERC1_AC97)) {
+ snd_printk(KERN_ERR "SERC1 AC'97 check failed (0x%x)\n", tmp);
+ return -EIO;
+ }
+ if ((tmp = snd_cs4281_peekBA0(chip, BA0_SERC2)) != (BA0_SERC2_SI1EN | BA0_SERC2_AC97)) {
+ snd_printk(KERN_ERR "SERC2 AC'97 check failed (0x%x)\n", tmp);
+ return -EIO;
+ }
+
+ /* Sound System Power Management */
+ snd_cs4281_pokeBA0(chip, BA0_SSPM, BA0_SSPM_MIXEN | BA0_SSPM_CSRCEN |
+ BA0_SSPM_PSRCEN | BA0_SSPM_JSEN |
+ BA0_SSPM_ACLEN | BA0_SSPM_FMEN);
+
+ /* Serial Port Power Management */
+ /* Blast the clock control register to zero so that the
+ * PLL starts out in a known state, and blast the master serial
+ * port control register to zero so that the serial ports also
+ * start out in a known state. */
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, 0);
+ snd_cs4281_pokeBA0(chip, BA0_SERMC, 0);
+
+ /* Make ESYN go to zero to turn off
+ * the Sync pulse on the AC97 link. */
+ snd_cs4281_pokeBA0(chip, BA0_ACCTL, 0);
+ udelay(50);
+
+ /* Drive the ARST# pin low for a minimum of 1uS (as defined in the AC97
+ * spec) and then drive it high. This is done for non AC97 modes since
+ * there might be logic external to the CS4281 that uses the ARST# line
+ * for a reset. */
+ snd_cs4281_pokeBA0(chip, BA0_SPMC, 0);
+ udelay(50);
+ snd_cs4281_pokeBA0(chip, BA0_SPMC, BA0_SPMC_RSTN);
+ msleep(50);
+
+ if (chip->dual_codec)
+ snd_cs4281_pokeBA0(chip, BA0_SPMC, BA0_SPMC_RSTN | BA0_SPMC_ASDI2E);
+
+ /*
+ * Set the serial port timing configuration.
+ */
+ snd_cs4281_pokeBA0(chip, BA0_SERMC,
+ (chip->dual_codec ? BA0_SERMC_TCID(chip->dual_codec) : BA0_SERMC_TCID(1)) |
+ BA0_SERMC_PTC_AC97 | BA0_SERMC_MSPE);
+
+ /*
+ * Start the DLL Clock logic.
+ */
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, BA0_CLKCR1_DLLP);
+ msleep(50);
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, BA0_CLKCR1_SWCE | BA0_CLKCR1_DLLP);
+
+ /*
+ * Wait for the DLL ready signal from the clock logic.
+ */
+ end_time = jiffies + HZ;
+ do {
+ /*
+ * Read the AC97 status register to see if we've seen a CODEC
+ * signal from the AC97 codec.
+ */
+ if (snd_cs4281_peekBA0(chip, BA0_CLKCR1) & BA0_CLKCR1_DLLRDY)
+ goto __ok0;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+
+ snd_printk(KERN_ERR "DLLRDY not seen\n");
+ return -EIO;
+
+ __ok0:
+
+ /*
+ * The first thing we do here is to enable sync generation. As soon
+ * as we start receiving bit clock, we'll start producing the SYNC
+ * signal.
+ */
+ snd_cs4281_pokeBA0(chip, BA0_ACCTL, BA0_ACCTL_ESYN);
+
+ /*
+ * Wait for the codec ready signal from the AC97 codec.
+ */
+ end_time = jiffies + HZ;
+ do {
+ /*
+ * Read the AC97 status register to see if we've seen a CODEC
+ * signal from the AC97 codec.
+ */
+ if (snd_cs4281_peekBA0(chip, BA0_ACSTS) & BA0_ACSTS_CRDY)
+ goto __ok1;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+
+ snd_printk(KERN_ERR "never read codec ready from AC'97 (0x%x)\n", snd_cs4281_peekBA0(chip, BA0_ACSTS));
+ return -EIO;
+
+ __ok1:
+ if (chip->dual_codec) {
+ end_time = jiffies + HZ;
+ do {
+ if (snd_cs4281_peekBA0(chip, BA0_ACSTS2) & BA0_ACSTS_CRDY)
+ goto __codec2_ok;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_INFO "secondary codec doesn't respond. disable it...\n");
+ chip->dual_codec = 0;
+ __codec2_ok: ;
+ }
+
+ /*
+ * Assert the valid frame signal so that we can start sending commands
+ * to the AC97 codec.
+ */
+
+ snd_cs4281_pokeBA0(chip, BA0_ACCTL, BA0_ACCTL_VFRM | BA0_ACCTL_ESYN);
+
+ /*
+ * Wait until we've sampled input slots 3 and 4 as valid, meaning that
+ * the codec is pumping ADC data across the AC-link.
+ */
+
+ end_time = jiffies + HZ;
+ do {
+ /*
+ * Read the input slot valid register and see if input slots 3
+ * 4 are valid yet.
+ */
+ if ((snd_cs4281_peekBA0(chip, BA0_ACISV) & (BA0_ACISV_SLV(3) | BA0_ACISV_SLV(4))) == (BA0_ACISV_SLV(3) | BA0_ACISV_SLV(4)))
+ goto __ok2;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+
+ if (--retry_count > 0)
+ goto __retry;
+ snd_printk(KERN_ERR "never read ISV3 and ISV4 from AC'97\n");
+ return -EIO;
+
+ __ok2:
+
+ /*
+ * Now, assert valid frame and the slot 3 and 4 valid bits. This will
+ * commense the transfer of digital audio data to the AC97 codec.
+ */
+ snd_cs4281_pokeBA0(chip, BA0_ACOSV, BA0_ACOSV_SLV(3) | BA0_ACOSV_SLV(4));
+
+ /*
+ * Initialize DMA structures
+ */
+ for (tmp = 0; tmp < 4; tmp++) {
+ struct cs4281_dma *dma = &chip->dma[tmp];
+ dma->regDBA = BA0_DBA0 + (tmp * 0x10);
+ dma->regDCA = BA0_DCA0 + (tmp * 0x10);
+ dma->regDBC = BA0_DBC0 + (tmp * 0x10);
+ dma->regDCC = BA0_DCC0 + (tmp * 0x10);
+ dma->regDMR = BA0_DMR0 + (tmp * 8);
+ dma->regDCR = BA0_DCR0 + (tmp * 8);
+ dma->regHDSR = BA0_HDSR0 + (tmp * 4);
+ dma->regFCR = BA0_FCR0 + (tmp * 4);
+ dma->regFSIC = BA0_FSIC0 + (tmp * 4);
+ dma->fifo_offset = tmp * CS4281_FIFO_SIZE;
+ snd_cs4281_pokeBA0(chip, dma->regFCR,
+ BA0_FCR_LS(31) |
+ BA0_FCR_RS(31) |
+ BA0_FCR_SZ(CS4281_FIFO_SIZE) |
+ BA0_FCR_OF(dma->fifo_offset));
+ }
+
+ chip->src_left_play_slot = 0; /* AC'97 left PCM playback (3) */
+ chip->src_right_play_slot = 1; /* AC'97 right PCM playback (4) */
+ chip->src_left_rec_slot = 10; /* AC'97 left PCM record (3) */
+ chip->src_right_rec_slot = 11; /* AC'97 right PCM record (4) */
+
+ /* Activate wave playback FIFO for FM playback */
+ chip->dma[0].valFCR = BA0_FCR_FEN | BA0_FCR_LS(0) |
+ BA0_FCR_RS(1) |
+ BA0_FCR_SZ(CS4281_FIFO_SIZE) |
+ BA0_FCR_OF(chip->dma[0].fifo_offset);
+ snd_cs4281_pokeBA0(chip, chip->dma[0].regFCR, chip->dma[0].valFCR);
+ snd_cs4281_pokeBA0(chip, BA0_SRCSA, (chip->src_left_play_slot << 0) |
+ (chip->src_right_play_slot << 8) |
+ (chip->src_left_rec_slot << 16) |
+ (chip->src_right_rec_slot << 24));
+
+ /* Initialize digital volume */
+ snd_cs4281_pokeBA0(chip, BA0_PPLVC, 0);
+ snd_cs4281_pokeBA0(chip, BA0_PPRVC, 0);
+
+ /* Enable IRQs */
+ snd_cs4281_pokeBA0(chip, BA0_HICR, BA0_HICR_EOI);
+ /* Unmask interrupts */
+ snd_cs4281_pokeBA0(chip, BA0_HIMR, 0x7fffffff & ~(
+ BA0_HISR_MIDI |
+ BA0_HISR_DMAI |
+ BA0_HISR_DMA(0) |
+ BA0_HISR_DMA(1) |
+ BA0_HISR_DMA(2) |
+ BA0_HISR_DMA(3)));
+ synchronize_irq(chip->irq);
+
+ return 0;
+}
+
+/*
+ * MIDI section
+ */
+
+static void snd_cs4281_midi_reset(struct cs4281 *chip)
+{
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr | BA0_MIDCR_MRST);
+ udelay(100);
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+}
+
+static int snd_cs4281_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct cs4281 *chip = substream->rmidi->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->midcr |= BA0_MIDCR_RXE;
+ chip->midi_input = substream;
+ if (!(chip->uartm & CS4281_MODE_OUTPUT)) {
+ snd_cs4281_midi_reset(chip);
+ } else {
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cs4281_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct cs4281 *chip = substream->rmidi->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->midcr &= ~(BA0_MIDCR_RXE | BA0_MIDCR_RIE);
+ chip->midi_input = NULL;
+ if (!(chip->uartm & CS4281_MODE_OUTPUT)) {
+ snd_cs4281_midi_reset(chip);
+ } else {
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ chip->uartm &= ~CS4281_MODE_INPUT;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cs4281_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct cs4281 *chip = substream->rmidi->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->uartm |= CS4281_MODE_OUTPUT;
+ chip->midcr |= BA0_MIDCR_TXE;
+ chip->midi_output = substream;
+ if (!(chip->uartm & CS4281_MODE_INPUT)) {
+ snd_cs4281_midi_reset(chip);
+ } else {
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cs4281_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct cs4281 *chip = substream->rmidi->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->midcr &= ~(BA0_MIDCR_TXE | BA0_MIDCR_TIE);
+ chip->midi_output = NULL;
+ if (!(chip->uartm & CS4281_MODE_INPUT)) {
+ snd_cs4281_midi_reset(chip);
+ } else {
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ chip->uartm &= ~CS4281_MODE_OUTPUT;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static void snd_cs4281_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct cs4281 *chip = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (up) {
+ if ((chip->midcr & BA0_MIDCR_RIE) == 0) {
+ chip->midcr |= BA0_MIDCR_RIE;
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ } else {
+ if (chip->midcr & BA0_MIDCR_RIE) {
+ chip->midcr &= ~BA0_MIDCR_RIE;
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static void snd_cs4281_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct cs4281 *chip = substream->rmidi->private_data;
+ unsigned char byte;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (up) {
+ if ((chip->midcr & BA0_MIDCR_TIE) == 0) {
+ chip->midcr |= BA0_MIDCR_TIE;
+ /* fill UART FIFO buffer at first, and turn Tx interrupts only if necessary */
+ while ((chip->midcr & BA0_MIDCR_TIE) &&
+ (snd_cs4281_peekBA0(chip, BA0_MIDSR) & BA0_MIDSR_TBF) == 0) {
+ if (snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ chip->midcr &= ~BA0_MIDCR_TIE;
+ } else {
+ snd_cs4281_pokeBA0(chip, BA0_MIDWP, byte);
+ }
+ }
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ } else {
+ if (chip->midcr & BA0_MIDCR_TIE) {
+ chip->midcr &= ~BA0_MIDCR_TIE;
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static struct snd_rawmidi_ops snd_cs4281_midi_output =
+{
+ .open = snd_cs4281_midi_output_open,
+ .close = snd_cs4281_midi_output_close,
+ .trigger = snd_cs4281_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_cs4281_midi_input =
+{
+ .open = snd_cs4281_midi_input_open,
+ .close = snd_cs4281_midi_input_close,
+ .trigger = snd_cs4281_midi_input_trigger,
+};
+
+static int __devinit snd_cs4281_midi(struct cs4281 * chip, int device,
+ struct snd_rawmidi **rrawmidi)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ if ((err = snd_rawmidi_new(chip->card, "CS4281", device, 1, 1, &rmidi)) < 0)
+ return err;
+ strcpy(rmidi->name, "CS4281");
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_cs4281_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_cs4281_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = chip;
+ chip->rmidi = rmidi;
+ if (rrawmidi)
+ *rrawmidi = rmidi;
+ return 0;
+}
+
+/*
+ * Interrupt handler
+ */
+
+static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id)
+{
+ struct cs4281 *chip = dev_id;
+ unsigned int status, dma, val;
+ struct cs4281_dma *cdma;
+
+ if (chip == NULL)
+ return IRQ_NONE;
+ status = snd_cs4281_peekBA0(chip, BA0_HISR);
+ if ((status & 0x7fffffff) == 0) {
+ snd_cs4281_pokeBA0(chip, BA0_HICR, BA0_HICR_EOI);
+ return IRQ_NONE;
+ }
+
+ if (status & (BA0_HISR_DMA(0)|BA0_HISR_DMA(1)|BA0_HISR_DMA(2)|BA0_HISR_DMA(3))) {
+ for (dma = 0; dma < 4; dma++)
+ if (status & BA0_HISR_DMA(dma)) {
+ cdma = &chip->dma[dma];
+ spin_lock(&chip->reg_lock);
+ /* ack DMA IRQ */
+ val = snd_cs4281_peekBA0(chip, cdma->regHDSR);
+ /* workaround, sometimes CS4281 acknowledges */
+ /* end or middle transfer position twice */
+ cdma->frag++;
+ if ((val & BA0_HDSR_DHTC) && !(cdma->frag & 1)) {
+ cdma->frag--;
+ chip->spurious_dhtc_irq++;
+ spin_unlock(&chip->reg_lock);
+ continue;
+ }
+ if ((val & BA0_HDSR_DTC) && (cdma->frag & 1)) {
+ cdma->frag--;
+ chip->spurious_dtc_irq++;
+ spin_unlock(&chip->reg_lock);
+ continue;
+ }
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(cdma->substream);
+ }
+ }
+
+ if ((status & BA0_HISR_MIDI) && chip->rmidi) {
+ unsigned char c;
+
+ spin_lock(&chip->reg_lock);
+ while ((snd_cs4281_peekBA0(chip, BA0_MIDSR) & BA0_MIDSR_RBE) == 0) {
+ c = snd_cs4281_peekBA0(chip, BA0_MIDRP);
+ if ((chip->midcr & BA0_MIDCR_RIE) == 0)
+ continue;
+ snd_rawmidi_receive(chip->midi_input, &c, 1);
+ }
+ while ((snd_cs4281_peekBA0(chip, BA0_MIDSR) & BA0_MIDSR_TBF) == 0) {
+ if ((chip->midcr & BA0_MIDCR_TIE) == 0)
+ break;
+ if (snd_rawmidi_transmit(chip->midi_output, &c, 1) != 1) {
+ chip->midcr &= ~BA0_MIDCR_TIE;
+ snd_cs4281_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ break;
+ }
+ snd_cs4281_pokeBA0(chip, BA0_MIDWP, c);
+ }
+ spin_unlock(&chip->reg_lock);
+ }
+
+ /* EOI to the PCI part... reenables interrupts */
+ snd_cs4281_pokeBA0(chip, BA0_HICR, BA0_HICR_EOI);
+
+ return IRQ_HANDLED;
+}
+
+
+/*
+ * OPL3 command
+ */
+static void snd_cs4281_opl3_command(struct snd_opl3 *opl3, unsigned short cmd,
+ unsigned char val)
+{
+ unsigned long flags;
+ struct cs4281 *chip = opl3->private_data;
+ void __iomem *port;
+
+ if (cmd & OPL3_RIGHT)
+ port = chip->ba0 + BA0_B1AP; /* right port */
+ else
+ port = chip->ba0 + BA0_B0AP; /* left port */
+
+ spin_lock_irqsave(&opl3->reg_lock, flags);
+
+ writel((unsigned int)cmd, port);
+ udelay(10);
+
+ writel((unsigned int)val, port + 4);
+ udelay(30);
+
+ spin_unlock_irqrestore(&opl3->reg_lock, flags);
+}
+
+static int __devinit snd_cs4281_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct cs4281 *chip;
+ struct snd_opl3 *opl3;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_cs4281_create(card, pci, &chip, dual_codec[dev])) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ if ((err = snd_cs4281_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_cs4281_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_cs4281_midi(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_opl3_new(card, OPL3_HW_OPL3_CS4281, &opl3)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ opl3->private_data = chip;
+ opl3->command = snd_cs4281_opl3_command;
+ snd_opl3_init(opl3);
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_cs4281_create_gameport(chip);
+ strcpy(card->driver, "CS4281");
+ strcpy(card->shortname, "Cirrus Logic CS4281");
+ sprintf(card->longname, "%s at 0x%lx, irq %d",
+ card->shortname,
+ chip->ba0_addr,
+ chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_cs4281_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+/*
+ * Power Management
+ */
+#ifdef CONFIG_PM
+
+static int saved_regs[SUSPEND_REGISTERS] = {
+ BA0_JSCTL,
+ BA0_GPIOR,
+ BA0_SSCR,
+ BA0_MIDCR,
+ BA0_SRCSA,
+ BA0_PASR,
+ BA0_CASR,
+ BA0_DACSR,
+ BA0_ADCSR,
+ BA0_FMLVC,
+ BA0_FMRVC,
+ BA0_PPLVC,
+ BA0_PPRVC,
+};
+
+#define CLKCR1_CKRA 0x00010000L
+
+static int cs4281_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct cs4281 *chip = card->private_data;
+ u32 ulCLK;
+ unsigned int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+
+ snd_ac97_suspend(chip->ac97);
+ snd_ac97_suspend(chip->ac97_secondary);
+
+ ulCLK = snd_cs4281_peekBA0(chip, BA0_CLKCR1);
+ ulCLK |= CLKCR1_CKRA;
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, ulCLK);
+
+ /* Disable interrupts. */
+ snd_cs4281_pokeBA0(chip, BA0_HICR, BA0_HICR_CHGM);
+
+ /* remember the status registers */
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ if (saved_regs[i])
+ chip->suspend_regs[i] = snd_cs4281_peekBA0(chip, saved_regs[i]);
+
+ /* Turn off the serial ports. */
+ snd_cs4281_pokeBA0(chip, BA0_SERMC, 0);
+
+ /* Power off FM, Joystick, AC link, */
+ snd_cs4281_pokeBA0(chip, BA0_SSPM, 0);
+
+ /* DLL off. */
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, 0);
+
+ /* AC link off. */
+ snd_cs4281_pokeBA0(chip, BA0_SPMC, 0);
+
+ ulCLK = snd_cs4281_peekBA0(chip, BA0_CLKCR1);
+ ulCLK &= ~CLKCR1_CKRA;
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, ulCLK);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int cs4281_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct cs4281 *chip = card->private_data;
+ unsigned int i;
+ u32 ulCLK;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "cs4281: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ ulCLK = snd_cs4281_peekBA0(chip, BA0_CLKCR1);
+ ulCLK |= CLKCR1_CKRA;
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, ulCLK);
+
+ snd_cs4281_chip_init(chip);
+
+ /* restore the status registers */
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ if (saved_regs[i])
+ snd_cs4281_pokeBA0(chip, saved_regs[i], chip->suspend_regs[i]);
+
+ snd_ac97_resume(chip->ac97);
+ snd_ac97_resume(chip->ac97_secondary);
+
+ ulCLK = snd_cs4281_peekBA0(chip, BA0_CLKCR1);
+ ulCLK &= ~CLKCR1_CKRA;
+ snd_cs4281_pokeBA0(chip, BA0_CLKCR1, ulCLK);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static struct pci_driver driver = {
+ .name = "CS4281",
+ .id_table = snd_cs4281_ids,
+ .probe = snd_cs4281_probe,
+ .remove = __devexit_p(snd_cs4281_remove),
+#ifdef CONFIG_PM
+ .suspend = cs4281_suspend,
+ .resume = cs4281_resume,
+#endif
+};
+
+static int __init alsa_card_cs4281_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_cs4281_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_cs4281_init)
+module_exit(alsa_card_cs4281_exit)
diff --git a/sound/pci/cs46xx/Makefile b/sound/pci/cs46xx/Makefile
new file mode 100644
index 0000000..67e811e
--- /dev/null
+++ b/sound/pci/cs46xx/Makefile
@@ -0,0 +1,10 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-cs46xx-y := cs46xx.o cs46xx_lib.o
+snd-cs46xx-$(CONFIG_SND_CS46XX_NEW_DSP) += dsp_spos.o dsp_spos_scb_lib.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_CS46XX) += snd-cs46xx.o
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
new file mode 100644
index 0000000..e876b32
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -0,0 +1,186 @@
+/*
+ * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ NOTES:
+ - sometimes the sound is metallic and sibilant, unloading and
+ reloading the module may solve this.
+*/
+
+#include <linux/pci.h>
+#include <linux/time.h>
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/cs46xx.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)},"
+ "{Cirrus Logic,Sound Fusion (CS4610)},"
+ "{Cirrus Logic,Sound Fusion (CS4612)},"
+ "{Cirrus Logic,Sound Fusion (CS4615)},"
+ "{Cirrus Logic,Sound Fusion (CS4622)},"
+ "{Cirrus Logic,Sound Fusion (CS4624)},"
+ "{Cirrus Logic,Sound Fusion (CS4630)}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int external_amp[SNDRV_CARDS];
+static int thinkpad[SNDRV_CARDS];
+static int mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the CS46xx soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the CS46xx soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable CS46xx soundcard.");
+module_param_array(external_amp, bool, NULL, 0444);
+MODULE_PARM_DESC(external_amp, "Force to enable external amplifer.");
+module_param_array(thinkpad, bool, NULL, 0444);
+MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control.");
+module_param_array(mmap_valid, bool, NULL, 0444);
+MODULE_PARM_DESC(mmap_valid, "Support OSS mmap.");
+
+static struct pci_device_id snd_cs46xx_ids[] = {
+ { 0x1013, 0x6001, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4280 */
+ { 0x1013, 0x6003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4612 */
+ { 0x1013, 0x6004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4615 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_cs46xx_ids);
+
+static int __devinit snd_card_cs46xx_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_cs46xx *chip;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ if ((err = snd_cs46xx_create(card, pci,
+ external_amp[dev], thinkpad[dev],
+ &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+ chip->accept_valid = mmap_valid[dev];
+ if ((err = snd_cs46xx_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if ((err = snd_cs46xx_pcm_rear(chip,1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_cs46xx_pcm_iec958(chip,2,NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#endif
+ if ((err = snd_cs46xx_mixer(chip, 2)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (chip->nr_ac97_codecs ==2) {
+ if ((err = snd_cs46xx_pcm_center_lfe(chip,3,NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+#endif
+ if ((err = snd_cs46xx_midi(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_cs46xx_start_dsp(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+
+ snd_cs46xx_gameport(chip);
+
+ strcpy(card->driver, "CS46xx");
+ strcpy(card->shortname, "Sound Fusion CS46xx");
+ sprintf(card->longname, "%s at 0x%lx/0x%lx, irq %i",
+ card->shortname,
+ chip->ba0_addr,
+ chip->ba1_addr,
+ chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_card_cs46xx_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Sound Fusion CS46xx",
+ .id_table = snd_cs46xx_ids,
+ .probe = snd_card_cs46xx_probe,
+ .remove = __devexit_p(snd_card_cs46xx_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cs46xx_suspend,
+ .resume = snd_cs46xx_resume,
+#endif
+};
+
+static int __init alsa_card_cs46xx_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_cs46xx_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_cs46xx_init)
+module_exit(alsa_card_cs46xx_exit)
diff --git a/sound/pci/cs46xx/cs46xx_image.h b/sound/pci/cs46xx/cs46xx_image.h
new file mode 100644
index 0000000..dc93f62
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx_image.h
@@ -0,0 +1,3468 @@
+struct BA1struct {
+ struct {
+ unsigned long offset;
+ unsigned long size;
+ } memory[BA1_MEMORY_COUNT];
+ u32 map[BA1_DWORD_SIZE];
+};
+
+
+static struct BA1struct BA1Struct = {
+{{ 0x00000000, 0x00003000 },{ 0x00010000, 0x00003800 },{ 0x00020000, 0x00007000 }},
+{0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000163,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00200040,0x00008010,0x00000000,
+0x00000000,0x80000001,0x00000001,0x00060000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00900080,0x00000173,0x00000000,
+0x00000000,0x00000010,0x00800000,0x00900000,
+0xf2c0000f,0x00000200,0x00000000,0x00010600,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000163,0x330300c2,
+0x06000000,0x00000000,0x80008000,0x80008000,
+0x3fc0000f,0x00000301,0x00010400,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00b00000,0x00d0806d,0x330480c3,
+0x04800000,0x00000001,0x00800001,0x0000ffff,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x066a0600,0x06350070,0x0000929d,0x929d929d,
+0x00000000,0x0000735a,0x00000600,0x00000000,
+0x929d735a,0x8734abfe,0x00010000,0x735a735a,
+0xa431ac75,0xa431ac75,0xa431ac75,0xa431ac75,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x0000804f,0x000000c3,
+0x05000000,0x00a00010,0x00000000,0x80008000,
+0x00000000,0x00000000,0x00000700,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000080,0x00a00000,0x0000809a,0x000000c2,
+0x07400000,0x00000000,0x80008000,0xffffffff,
+0x00c80028,0x00005555,0x00000000,0x000107a0,
+0x00c80028,0x000000c2,0x06800000,0x00000000,
+0x06e00080,0x00300000,0x000080bb,0x000000c9,
+0x07a00000,0x04000000,0x80008000,0xffffffff,
+0x00c80028,0x00005555,0x00000000,0x00000780,
+0x00c80028,0x000000c5,0xff800000,0x00000000,
+0x00640080,0x00c00000,0x00008197,0x000000c9,
+0x07800000,0x04000000,0x80008000,0xffffffff,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x0000805e,0x000000c1,
+0x00000000,0x00800000,0x80008000,0x80008000,
+0x00020000,0x0000ffff,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
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+0x00000000,0x00000000,0x00000000,0x00000000,
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+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
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+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000,
+0x00000000,0x00000000,0x00000000,0x00000000}
+ };
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
new file mode 100644
index 0000000..fb6dc39
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -0,0 +1,3873 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Abramo Bagnara <abramo@alsa-project.org>
+ * Cirrus Logic, Inc.
+ * Routines for control of Cirrus Logic CS461x chips
+ *
+ * KNOWN BUGS:
+ * - Sometimes the SPDIF input DSP tasks get's unsynchronized
+ * and the SPDIF get somewhat "distorcionated", or/and left right channel
+ * are swapped. To get around this problem when it happens, mute and unmute
+ * the SPDIF input mixer control.
+ * - On the Hercules Game Theater XP the amplifier are sometimes turned
+ * off on inadecuate moments which causes distorcions on sound.
+ *
+ * TODO:
+ * - Secondary CODEC on some soundcards
+ * - SPDIF input support for other sample rates then 48khz
+ * - Posibility to mix the SPDIF output with analog sources.
+ * - PCM channels for Center and LFE on secondary codec
+ *
+ * NOTE: with CONFIG_SND_CS46XX_NEW_DSP unset uses old DSP image (which
+ * is default configuration), no SPDIF, no secondary codec, no
+ * multi channel PCM. But known to work.
+ *
+ * FINALLY: A credit to the developers Tom and Jordan
+ * at Cirrus for have helping me out with the DSP, however we
+ * still don't have sufficient documentation and technical
+ * references to be able to implement all fancy feutures
+ * supported by the cs46xx DSP's.
+ * Benny <benny@hostmobility.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/pci.h>
+#include <linux/pm.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/mutex.h>
+
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/cs46xx.h>
+
+#include <asm/io.h>
+
+#include "cs46xx_lib.h"
+#include "dsp_spos.h"
+
+static void amp_voyetra(struct snd_cs46xx *chip, int change);
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+static struct snd_pcm_ops snd_cs46xx_playback_rear_ops;
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_rear_ops;
+static struct snd_pcm_ops snd_cs46xx_playback_clfe_ops;
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_clfe_ops;
+static struct snd_pcm_ops snd_cs46xx_playback_iec958_ops;
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_iec958_ops;
+#endif
+
+static struct snd_pcm_ops snd_cs46xx_playback_ops;
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_ops;
+static struct snd_pcm_ops snd_cs46xx_capture_ops;
+static struct snd_pcm_ops snd_cs46xx_capture_indirect_ops;
+
+static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
+ unsigned short reg,
+ int codec_index)
+{
+ int count;
+ unsigned short result,tmp;
+ u32 offset = 0;
+
+ if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
+ codec_index != CS46XX_SECONDARY_CODEC_INDEX))
+ return -EINVAL;
+
+ chip->active_ctrl(chip, 1);
+
+ if (codec_index == CS46XX_SECONDARY_CODEC_INDEX)
+ offset = CS46XX_SECONDARY_CODEC_OFFSET;
+
+ /*
+ * 1. Write ACCAD = Command Address Register = 46Ch for AC97 register address
+ * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97
+ * 3. Write ACCTL = Control Register = 460h for initiating the write7---55
+ * 4. Read ACCTL = 460h, DCV should be reset by now and 460h = 17h
+ * 5. if DCV not cleared, break and return error
+ * 6. Read ACSTS = Status Register = 464h, check VSTS bit
+ */
+
+ snd_cs46xx_peekBA0(chip, BA0_ACSDA + offset);
+
+ tmp = snd_cs46xx_peekBA0(chip, BA0_ACCTL);
+ if ((tmp & ACCTL_VFRM) == 0) {
+ snd_printk(KERN_WARNING "cs46xx: ACCTL_VFRM not set 0x%x\n",tmp);
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, (tmp & (~ACCTL_ESYN)) | ACCTL_VFRM );
+ msleep(50);
+ tmp = snd_cs46xx_peekBA0(chip, BA0_ACCTL + offset);
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, tmp | ACCTL_ESYN | ACCTL_VFRM );
+
+ }
+
+ /*
+ * Setup the AC97 control registers on the CS461x to send the
+ * appropriate command to the AC97 to perform the read.
+ * ACCAD = Command Address Register = 46Ch
+ * ACCDA = Command Data Register = 470h
+ * ACCTL = Control Register = 460h
+ * set DCV - will clear when process completed
+ * set CRW - Read command
+ * set VFRM - valid frame enabled
+ * set ESYN - ASYNC generation enabled
+ * set RSTN - ARST# inactive, AC97 codec not reset
+ */
+
+ snd_cs46xx_pokeBA0(chip, BA0_ACCAD, reg);
+ snd_cs46xx_pokeBA0(chip, BA0_ACCDA, 0);
+ if (codec_index == CS46XX_PRIMARY_CODEC_INDEX) {
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL,/* clear ACCTL_DCV */ ACCTL_CRW |
+ ACCTL_VFRM | ACCTL_ESYN |
+ ACCTL_RSTN);
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_DCV | ACCTL_CRW |
+ ACCTL_VFRM | ACCTL_ESYN |
+ ACCTL_RSTN);
+ } else {
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_DCV | ACCTL_TC |
+ ACCTL_CRW | ACCTL_VFRM | ACCTL_ESYN |
+ ACCTL_RSTN);
+ }
+
+ /*
+ * Wait for the read to occur.
+ */
+ for (count = 0; count < 1000; count++) {
+ /*
+ * First, we want to wait for a short time.
+ */
+ udelay(10);
+ /*
+ * Now, check to see if the read has completed.
+ * ACCTL = 460h, DCV should be reset by now and 460h = 17h
+ */
+ if (!(snd_cs46xx_peekBA0(chip, BA0_ACCTL) & ACCTL_DCV))
+ goto ok1;
+ }
+
+ snd_printk(KERN_ERR "AC'97 read problem (ACCTL_DCV), reg = 0x%x\n", reg);
+ result = 0xffff;
+ goto end;
+
+ ok1:
+ /*
+ * Wait for the valid status bit to go active.
+ */
+ for (count = 0; count < 100; count++) {
+ /*
+ * Read the AC97 status register.
+ * ACSTS = Status Register = 464h
+ * VSTS - Valid Status
+ */
+ if (snd_cs46xx_peekBA0(chip, BA0_ACSTS + offset) & ACSTS_VSTS)
+ goto ok2;
+ udelay(10);
+ }
+
+ snd_printk(KERN_ERR "AC'97 read problem (ACSTS_VSTS), codec_index %d, reg = 0x%x\n", codec_index, reg);
+ result = 0xffff;
+ goto end;
+
+ ok2:
+ /*
+ * Read the data returned from the AC97 register.
+ * ACSDA = Status Data Register = 474h
+ */
+#if 0
+ printk("e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg,
+ snd_cs46xx_peekBA0(chip, BA0_ACSDA),
+ snd_cs46xx_peekBA0(chip, BA0_ACCAD));
+#endif
+
+ //snd_cs46xx_peekBA0(chip, BA0_ACCAD);
+ result = snd_cs46xx_peekBA0(chip, BA0_ACSDA + offset);
+ end:
+ chip->active_ctrl(chip, -1);
+ return result;
+}
+
+static unsigned short snd_cs46xx_ac97_read(struct snd_ac97 * ac97,
+ unsigned short reg)
+{
+ struct snd_cs46xx *chip = ac97->private_data;
+ unsigned short val;
+ int codec_index = ac97->num;
+
+ if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
+ codec_index != CS46XX_SECONDARY_CODEC_INDEX))
+ return 0xffff;
+
+ val = snd_cs46xx_codec_read(chip, reg, codec_index);
+
+ return val;
+}
+
+
+static void snd_cs46xx_codec_write(struct snd_cs46xx *chip,
+ unsigned short reg,
+ unsigned short val,
+ int codec_index)
+{
+ int count;
+
+ if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
+ codec_index != CS46XX_SECONDARY_CODEC_INDEX))
+ return;
+
+ chip->active_ctrl(chip, 1);
+
+ /*
+ * 1. Write ACCAD = Command Address Register = 46Ch for AC97 register address
+ * 2. Write ACCDA = Command Data Register = 470h for data to write to AC97
+ * 3. Write ACCTL = Control Register = 460h for initiating the write
+ * 4. Read ACCTL = 460h, DCV should be reset by now and 460h = 07h
+ * 5. if DCV not cleared, break and return error
+ */
+
+ /*
+ * Setup the AC97 control registers on the CS461x to send the
+ * appropriate command to the AC97 to perform the read.
+ * ACCAD = Command Address Register = 46Ch
+ * ACCDA = Command Data Register = 470h
+ * ACCTL = Control Register = 460h
+ * set DCV - will clear when process completed
+ * reset CRW - Write command
+ * set VFRM - valid frame enabled
+ * set ESYN - ASYNC generation enabled
+ * set RSTN - ARST# inactive, AC97 codec not reset
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_ACCAD , reg);
+ snd_cs46xx_pokeBA0(chip, BA0_ACCDA , val);
+ snd_cs46xx_peekBA0(chip, BA0_ACCTL);
+
+ if (codec_index == CS46XX_PRIMARY_CODEC_INDEX) {
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, /* clear ACCTL_DCV */ ACCTL_VFRM |
+ ACCTL_ESYN | ACCTL_RSTN);
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_DCV | ACCTL_VFRM |
+ ACCTL_ESYN | ACCTL_RSTN);
+ } else {
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_DCV | ACCTL_TC |
+ ACCTL_VFRM | ACCTL_ESYN | ACCTL_RSTN);
+ }
+
+ for (count = 0; count < 4000; count++) {
+ /*
+ * First, we want to wait for a short time.
+ */
+ udelay(10);
+ /*
+ * Now, check to see if the write has completed.
+ * ACCTL = 460h, DCV should be reset by now and 460h = 07h
+ */
+ if (!(snd_cs46xx_peekBA0(chip, BA0_ACCTL) & ACCTL_DCV)) {
+ goto end;
+ }
+ }
+ snd_printk(KERN_ERR "AC'97 write problem, codec_index = %d, reg = 0x%x, val = 0x%x\n", codec_index, reg, val);
+ end:
+ chip->active_ctrl(chip, -1);
+}
+
+static void snd_cs46xx_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct snd_cs46xx *chip = ac97->private_data;
+ int codec_index = ac97->num;
+
+ if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
+ codec_index != CS46XX_SECONDARY_CODEC_INDEX))
+ return;
+
+ snd_cs46xx_codec_write(chip, reg, val, codec_index);
+}
+
+
+/*
+ * Chip initialization
+ */
+
+int snd_cs46xx_download(struct snd_cs46xx *chip,
+ u32 *src,
+ unsigned long offset,
+ unsigned long len)
+{
+ void __iomem *dst;
+ unsigned int bank = offset >> 16;
+ offset = offset & 0xffff;
+
+ if (snd_BUG_ON((offset & 3) || (len & 3)))
+ return -EINVAL;
+ dst = chip->region.idx[bank+1].remap_addr + offset;
+ len /= sizeof(u32);
+
+ /* writel already converts 32-bit value to right endianess */
+ while (len-- > 0) {
+ writel(*src++, dst);
+ dst += sizeof(u32);
+ }
+ return 0;
+}
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+
+#include "imgs/cwc4630.h"
+#include "imgs/cwcasync.h"
+#include "imgs/cwcsnoop.h"
+#include "imgs/cwcbinhack.h"
+#include "imgs/cwcdma.h"
+
+int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip,
+ unsigned long offset,
+ unsigned long len)
+{
+ void __iomem *dst;
+ unsigned int bank = offset >> 16;
+ offset = offset & 0xffff;
+
+ if (snd_BUG_ON((offset & 3) || (len & 3)))
+ return -EINVAL;
+ dst = chip->region.idx[bank+1].remap_addr + offset;
+ len /= sizeof(u32);
+
+ /* writel already converts 32-bit value to right endianess */
+ while (len-- > 0) {
+ writel(0, dst);
+ dst += sizeof(u32);
+ }
+ return 0;
+}
+
+#else /* old DSP image */
+
+#include "cs46xx_image.h"
+
+int snd_cs46xx_download_image(struct snd_cs46xx *chip)
+{
+ int idx, err;
+ unsigned long offset = 0;
+
+ for (idx = 0; idx < BA1_MEMORY_COUNT; idx++) {
+ if ((err = snd_cs46xx_download(chip,
+ &BA1Struct.map[offset],
+ BA1Struct.memory[idx].offset,
+ BA1Struct.memory[idx].size)) < 0)
+ return err;
+ offset += BA1Struct.memory[idx].size >> 2;
+ }
+ return 0;
+}
+#endif /* CONFIG_SND_CS46XX_NEW_DSP */
+
+/*
+ * Chip reset
+ */
+
+static void snd_cs46xx_reset(struct snd_cs46xx *chip)
+{
+ int idx;
+
+ /*
+ * Write the reset bit of the SP control register.
+ */
+ snd_cs46xx_poke(chip, BA1_SPCR, SPCR_RSTSP);
+
+ /*
+ * Write the control register.
+ */
+ snd_cs46xx_poke(chip, BA1_SPCR, SPCR_DRQEN);
+
+ /*
+ * Clear the trap registers.
+ */
+ for (idx = 0; idx < 8; idx++) {
+ snd_cs46xx_poke(chip, BA1_DREG, DREG_REGID_TRAP_SELECT + idx);
+ snd_cs46xx_poke(chip, BA1_TWPR, 0xFFFF);
+ }
+ snd_cs46xx_poke(chip, BA1_DREG, 0);
+
+ /*
+ * Set the frame timer to reflect the number of cycles per frame.
+ */
+ snd_cs46xx_poke(chip, BA1_FRMT, 0xadf);
+}
+
+static int cs46xx_wait_for_fifo(struct snd_cs46xx * chip,int retry_timeout)
+{
+ u32 i, status = 0;
+ /*
+ * Make sure the previous FIFO write operation has completed.
+ */
+ for(i = 0; i < 50; i++){
+ status = snd_cs46xx_peekBA0(chip, BA0_SERBST);
+
+ if( !(status & SERBST_WBSY) )
+ break;
+
+ mdelay(retry_timeout);
+ }
+
+ if(status & SERBST_WBSY) {
+ snd_printk( KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n");
+
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static void snd_cs46xx_clear_serial_FIFOs(struct snd_cs46xx *chip)
+{
+ int idx, powerdown = 0;
+ unsigned int tmp;
+
+ /*
+ * See if the devices are powered down. If so, we must power them up first
+ * or they will not respond.
+ */
+ tmp = snd_cs46xx_peekBA0(chip, BA0_CLKCR1);
+ if (!(tmp & CLKCR1_SWCE)) {
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp | CLKCR1_SWCE);
+ powerdown = 1;
+ }
+
+ /*
+ * We want to clear out the serial port FIFOs so we don't end up playing
+ * whatever random garbage happens to be in them. We fill the sample FIFOS
+ * with zero (silence).
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERBWP, 0);
+
+ /*
+ * Fill all 256 sample FIFO locations.
+ */
+ for (idx = 0; idx < 0xFF; idx++) {
+ /*
+ * Make sure the previous FIFO write operation has completed.
+ */
+ if (cs46xx_wait_for_fifo(chip,1)) {
+ snd_printdd ("failed waiting for FIFO at addr (%02X)\n",idx);
+
+ if (powerdown)
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp);
+
+ break;
+ }
+ /*
+ * Write the serial port FIFO index.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERBAD, idx);
+ /*
+ * Tell the serial port to load the new value into the FIFO location.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERBCM, SERBCM_WRC);
+ }
+ /*
+ * Now, if we powered up the devices, then power them back down again.
+ * This is kinda ugly, but should never happen.
+ */
+ if (powerdown)
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp);
+}
+
+static void snd_cs46xx_proc_start(struct snd_cs46xx *chip)
+{
+ int cnt;
+
+ /*
+ * Set the frame timer to reflect the number of cycles per frame.
+ */
+ snd_cs46xx_poke(chip, BA1_FRMT, 0xadf);
+ /*
+ * Turn on the run, run at frame, and DMA enable bits in the local copy of
+ * the SP control register.
+ */
+ snd_cs46xx_poke(chip, BA1_SPCR, SPCR_RUN | SPCR_RUNFR | SPCR_DRQEN);
+ /*
+ * Wait until the run at frame bit resets itself in the SP control
+ * register.
+ */
+ for (cnt = 0; cnt < 25; cnt++) {
+ udelay(50);
+ if (!(snd_cs46xx_peek(chip, BA1_SPCR) & SPCR_RUNFR))
+ break;
+ }
+
+ if (snd_cs46xx_peek(chip, BA1_SPCR) & SPCR_RUNFR)
+ snd_printk(KERN_ERR "SPCR_RUNFR never reset\n");
+}
+
+static void snd_cs46xx_proc_stop(struct snd_cs46xx *chip)
+{
+ /*
+ * Turn off the run, run at frame, and DMA enable bits in the local copy of
+ * the SP control register.
+ */
+ snd_cs46xx_poke(chip, BA1_SPCR, 0);
+}
+
+/*
+ * Sample rate routines
+ */
+
+#define GOF_PER_SEC 200
+
+static void snd_cs46xx_set_play_sample_rate(struct snd_cs46xx *chip, unsigned int rate)
+{
+ unsigned long flags;
+ unsigned int tmp1, tmp2;
+ unsigned int phiIncr;
+ unsigned int correctionPerGOF, correctionPerSec;
+
+ /*
+ * Compute the values used to drive the actual sample rate conversion.
+ * The following formulas are being computed, using inline assembly
+ * since we need to use 64 bit arithmetic to compute the values:
+ *
+ * phiIncr = floor((Fs,in * 2^26) / Fs,out)
+ * correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /
+ * GOF_PER_SEC)
+ * ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -M
+ * GOF_PER_SEC * correctionPerGOF
+ *
+ * i.e.
+ *
+ * phiIncr:other = dividend:remainder((Fs,in * 2^26) / Fs,out)
+ * correctionPerGOF:correctionPerSec =
+ * dividend:remainder(ulOther / GOF_PER_SEC)
+ */
+ tmp1 = rate << 16;
+ phiIncr = tmp1 / 48000;
+ tmp1 -= phiIncr * 48000;
+ tmp1 <<= 10;
+ phiIncr <<= 10;
+ tmp2 = tmp1 / 48000;
+ phiIncr += tmp2;
+ tmp1 -= tmp2 * 48000;
+ correctionPerGOF = tmp1 / GOF_PER_SEC;
+ tmp1 -= correctionPerGOF * GOF_PER_SEC;
+ correctionPerSec = tmp1;
+
+ /*
+ * Fill in the SampleRateConverter control block.
+ */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_cs46xx_poke(chip, BA1_PSRC,
+ ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF));
+ snd_cs46xx_poke(chip, BA1_PPI, phiIncr);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned int rate)
+{
+ unsigned long flags;
+ unsigned int phiIncr, coeffIncr, tmp1, tmp2;
+ unsigned int correctionPerGOF, correctionPerSec, initialDelay;
+ unsigned int frameGroupLength, cnt;
+
+ /*
+ * We can only decimate by up to a factor of 1/9th the hardware rate.
+ * Correct the value if an attempt is made to stray outside that limit.
+ */
+ if ((rate * 9) < 48000)
+ rate = 48000 / 9;
+
+ /*
+ * We can not capture at at rate greater than the Input Rate (48000).
+ * Return an error if an attempt is made to stray outside that limit.
+ */
+ if (rate > 48000)
+ rate = 48000;
+
+ /*
+ * Compute the values used to drive the actual sample rate conversion.
+ * The following formulas are being computed, using inline assembly
+ * since we need to use 64 bit arithmetic to compute the values:
+ *
+ * coeffIncr = -floor((Fs,out * 2^23) / Fs,in)
+ * phiIncr = floor((Fs,in * 2^26) / Fs,out)
+ * correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /
+ * GOF_PER_SEC)
+ * correctionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -
+ * GOF_PER_SEC * correctionPerGOF
+ * initialDelay = ceil((24 * Fs,in) / Fs,out)
+ *
+ * i.e.
+ *
+ * coeffIncr = neg(dividend((Fs,out * 2^23) / Fs,in))
+ * phiIncr:ulOther = dividend:remainder((Fs,in * 2^26) / Fs,out)
+ * correctionPerGOF:correctionPerSec =
+ * dividend:remainder(ulOther / GOF_PER_SEC)
+ * initialDelay = dividend(((24 * Fs,in) + Fs,out - 1) / Fs,out)
+ */
+
+ tmp1 = rate << 16;
+ coeffIncr = tmp1 / 48000;
+ tmp1 -= coeffIncr * 48000;
+ tmp1 <<= 7;
+ coeffIncr <<= 7;
+ coeffIncr += tmp1 / 48000;
+ coeffIncr ^= 0xFFFFFFFF;
+ coeffIncr++;
+ tmp1 = 48000 << 16;
+ phiIncr = tmp1 / rate;
+ tmp1 -= phiIncr * rate;
+ tmp1 <<= 10;
+ phiIncr <<= 10;
+ tmp2 = tmp1 / rate;
+ phiIncr += tmp2;
+ tmp1 -= tmp2 * rate;
+ correctionPerGOF = tmp1 / GOF_PER_SEC;
+ tmp1 -= correctionPerGOF * GOF_PER_SEC;
+ correctionPerSec = tmp1;
+ initialDelay = ((48000 * 24) + rate - 1) / rate;
+
+ /*
+ * Fill in the VariDecimate control block.
+ */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_cs46xx_poke(chip, BA1_CSRC,
+ ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF));
+ snd_cs46xx_poke(chip, BA1_CCI, coeffIncr);
+ snd_cs46xx_poke(chip, BA1_CD,
+ (((BA1_VARIDEC_BUF_1 + (initialDelay << 2)) << 16) & 0xFFFF0000) | 0x80);
+ snd_cs46xx_poke(chip, BA1_CPI, phiIncr);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ /*
+ * Figure out the frame group length for the write back task. Basically,
+ * this is just the factors of 24000 (2^6*3*5^3) that are not present in
+ * the output sample rate.
+ */
+ frameGroupLength = 1;
+ for (cnt = 2; cnt <= 64; cnt *= 2) {
+ if (((rate / cnt) * cnt) != rate)
+ frameGroupLength *= 2;
+ }
+ if (((rate / 3) * 3) != rate) {
+ frameGroupLength *= 3;
+ }
+ for (cnt = 5; cnt <= 125; cnt *= 5) {
+ if (((rate / cnt) * cnt) != rate)
+ frameGroupLength *= 5;
+ }
+
+ /*
+ * Fill in the WriteBack control block.
+ */
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_cs46xx_poke(chip, BA1_CFG1, frameGroupLength);
+ snd_cs46xx_poke(chip, BA1_CFG2, (0x00800000 | frameGroupLength));
+ snd_cs46xx_poke(chip, BA1_CCST, 0x0000FFFF);
+ snd_cs46xx_poke(chip, BA1_CSPB, ((65536 * rate) / 24000));
+ snd_cs46xx_poke(chip, (BA1_CSPB + 4), 0x0000FFFF);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+/*
+ * PCM part
+ */
+
+static void snd_cs46xx_pb_trans_copy(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_cs46xx_pcm * cpcm = runtime->private_data;
+ memcpy(cpcm->hw_buf.area + rec->hw_data, runtime->dma_area + rec->sw_data, bytes);
+}
+
+static int snd_cs46xx_playback_transfer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_cs46xx_pcm * cpcm = runtime->private_data;
+ snd_pcm_indirect_playback_transfer(substream, &cpcm->pcm_rec, snd_cs46xx_pb_trans_copy);
+ return 0;
+}
+
+static void snd_cs46xx_cp_trans_copy(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ memcpy(runtime->dma_area + rec->sw_data,
+ chip->capt.hw_buf.area + rec->hw_data, bytes);
+}
+
+static int snd_cs46xx_capture_transfer(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_indirect_capture_transfer(substream, &chip->capt.pcm_rec, snd_cs46xx_cp_trans_copy);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_cs46xx_playback_direct_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+ struct snd_cs46xx_pcm *cpcm = substream->runtime->private_data;
+
+ if (snd_BUG_ON(!cpcm->pcm_channel))
+ return -ENXIO;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ ptr = snd_cs46xx_peek(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 2) << 2);
+#else
+ ptr = snd_cs46xx_peek(chip, BA1_PBA);
+#endif
+ ptr -= cpcm->hw_buf.addr;
+ return ptr >> cpcm->shift;
+}
+
+static snd_pcm_uframes_t snd_cs46xx_playback_indirect_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+ struct snd_cs46xx_pcm *cpcm = substream->runtime->private_data;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (snd_BUG_ON(!cpcm->pcm_channel))
+ return -ENXIO;
+ ptr = snd_cs46xx_peek(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 2) << 2);
+#else
+ ptr = snd_cs46xx_peek(chip, BA1_PBA);
+#endif
+ ptr -= cpcm->hw_buf.addr;
+ return snd_pcm_indirect_playback_pointer(substream, &cpcm->pcm_rec, ptr);
+}
+
+static snd_pcm_uframes_t snd_cs46xx_capture_direct_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ size_t ptr = snd_cs46xx_peek(chip, BA1_CBA) - chip->capt.hw_buf.addr;
+ return ptr >> chip->capt.shift;
+}
+
+static snd_pcm_uframes_t snd_cs46xx_capture_indirect_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ size_t ptr = snd_cs46xx_peek(chip, BA1_CBA) - chip->capt.hw_buf.addr;
+ return snd_pcm_indirect_capture_pointer(substream, &chip->capt.pcm_rec, ptr);
+}
+
+static int snd_cs46xx_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ /*struct snd_pcm_runtime *runtime = substream->runtime;*/
+ int result = 0;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ struct snd_cs46xx_pcm *cpcm = substream->runtime->private_data;
+ if (! cpcm->pcm_channel) {
+ return -ENXIO;
+ }
+#endif
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ /* magic value to unmute PCM stream playback volume */
+ snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address +
+ SCBVolumeCtrl) << 2, 0x80008000);
+
+ if (cpcm->pcm_channel->unlinked)
+ cs46xx_dsp_pcm_link(chip,cpcm->pcm_channel);
+
+ if (substream->runtime->periods != CS46XX_FRAGS)
+ snd_cs46xx_playback_transfer(substream);
+#else
+ spin_lock(&chip->reg_lock);
+ if (substream->runtime->periods != CS46XX_FRAGS)
+ snd_cs46xx_playback_transfer(substream);
+ { unsigned int tmp;
+ tmp = snd_cs46xx_peek(chip, BA1_PCTL);
+ tmp &= 0x0000ffff;
+ snd_cs46xx_poke(chip, BA1_PCTL, chip->play_ctl | tmp);
+ }
+ spin_unlock(&chip->reg_lock);
+#endif
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ /* magic mute channel */
+ snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address +
+ SCBVolumeCtrl) << 2, 0xffffffff);
+
+ if (!cpcm->pcm_channel->unlinked)
+ cs46xx_dsp_pcm_unlink(chip,cpcm->pcm_channel);
+#else
+ spin_lock(&chip->reg_lock);
+ { unsigned int tmp;
+ tmp = snd_cs46xx_peek(chip, BA1_PCTL);
+ tmp &= 0x0000ffff;
+ snd_cs46xx_poke(chip, BA1_PCTL, tmp);
+ }
+ spin_unlock(&chip->reg_lock);
+#endif
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+
+ return result;
+}
+
+static int snd_cs46xx_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ unsigned int tmp;
+ int result = 0;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ tmp = snd_cs46xx_peek(chip, BA1_CCTL);
+ tmp &= 0xffff0000;
+ snd_cs46xx_poke(chip, BA1_CCTL, chip->capt.ctl | tmp);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ tmp = snd_cs46xx_peek(chip, BA1_CCTL);
+ tmp &= 0xffff0000;
+ snd_cs46xx_poke(chip, BA1_CCTL, tmp);
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ spin_unlock(&chip->reg_lock);
+
+ return result;
+}
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+static int _cs46xx_adjust_sample_rate (struct snd_cs46xx *chip, struct snd_cs46xx_pcm *cpcm,
+ int sample_rate)
+{
+
+ /* If PCMReaderSCB and SrcTaskSCB not created yet ... */
+ if ( cpcm->pcm_channel == NULL) {
+ cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate,
+ cpcm, cpcm->hw_buf.addr,cpcm->pcm_channel_id);
+ if (cpcm->pcm_channel == NULL) {
+ snd_printk(KERN_ERR "cs46xx: failed to create virtual PCM channel\n");
+ return -ENOMEM;
+ }
+ cpcm->pcm_channel->sample_rate = sample_rate;
+ } else
+ /* if sample rate is changed */
+ if ((int)cpcm->pcm_channel->sample_rate != sample_rate) {
+ int unlinked = cpcm->pcm_channel->unlinked;
+ cs46xx_dsp_destroy_pcm_channel (chip,cpcm->pcm_channel);
+
+ if ( (cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate, cpcm,
+ cpcm->hw_buf.addr,
+ cpcm->pcm_channel_id)) == NULL) {
+ snd_printk(KERN_ERR "cs46xx: failed to re-create virtual PCM channel\n");
+ return -ENOMEM;
+ }
+
+ if (!unlinked) cs46xx_dsp_pcm_link (chip,cpcm->pcm_channel);
+ cpcm->pcm_channel->sample_rate = sample_rate;
+ }
+
+ return 0;
+}
+#endif
+
+
+static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_cs46xx_pcm *cpcm;
+ int err;
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ int sample_rate = params_rate(hw_params);
+ int period_size = params_period_bytes(hw_params);
+#endif
+ cpcm = runtime->private_data;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (snd_BUG_ON(!sample_rate))
+ return -ENXIO;
+
+ mutex_lock(&chip->spos_mutex);
+
+ if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) {
+ mutex_unlock(&chip->spos_mutex);
+ return -ENXIO;
+ }
+
+ snd_BUG_ON(!cpcm->pcm_channel);
+ if (!cpcm->pcm_channel) {
+ mutex_unlock(&chip->spos_mutex);
+ return -ENXIO;
+ }
+
+
+ if (cs46xx_dsp_pcm_channel_set_period (chip,cpcm->pcm_channel,period_size)) {
+ mutex_unlock(&chip->spos_mutex);
+ return -EINVAL;
+ }
+
+ snd_printdd ("period_size (%d), periods (%d) buffer_size(%d)\n",
+ period_size, params_periods(hw_params),
+ params_buffer_bytes(hw_params));
+#endif
+
+ if (params_periods(hw_params) == CS46XX_FRAGS) {
+ if (runtime->dma_area != cpcm->hw_buf.area)
+ snd_pcm_lib_free_pages(substream);
+ runtime->dma_area = cpcm->hw_buf.area;
+ runtime->dma_addr = cpcm->hw_buf.addr;
+ runtime->dma_bytes = cpcm->hw_buf.bytes;
+
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (cpcm->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_ops;
+ } else if (cpcm->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_rear_ops;
+ } else if (cpcm->pcm_channel_id == DSP_PCM_CENTER_LFE_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_clfe_ops;
+ } else if (cpcm->pcm_channel_id == DSP_IEC958_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_iec958_ops;
+ } else {
+ snd_BUG();
+ }
+#else
+ substream->ops = &snd_cs46xx_playback_ops;
+#endif
+
+ } else {
+ if (runtime->dma_area == cpcm->hw_buf.area) {
+ runtime->dma_area = NULL;
+ runtime->dma_addr = 0;
+ runtime->dma_bytes = 0;
+ }
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) {
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ mutex_unlock(&chip->spos_mutex);
+#endif
+ return err;
+ }
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (cpcm->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_indirect_ops;
+ } else if (cpcm->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_indirect_rear_ops;
+ } else if (cpcm->pcm_channel_id == DSP_PCM_CENTER_LFE_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_indirect_clfe_ops;
+ } else if (cpcm->pcm_channel_id == DSP_IEC958_CHANNEL) {
+ substream->ops = &snd_cs46xx_playback_indirect_iec958_ops;
+ } else {
+ snd_BUG();
+ }
+#else
+ substream->ops = &snd_cs46xx_playback_indirect_ops;
+#endif
+
+ }
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ mutex_unlock(&chip->spos_mutex);
+#endif
+
+ return 0;
+}
+
+static int snd_cs46xx_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ /*struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);*/
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_cs46xx_pcm *cpcm;
+
+ cpcm = runtime->private_data;
+
+ /* if play_back open fails, then this function
+ is called and cpcm can actually be NULL here */
+ if (!cpcm) return -ENXIO;
+
+ if (runtime->dma_area != cpcm->hw_buf.area)
+ snd_pcm_lib_free_pages(substream);
+
+ runtime->dma_area = NULL;
+ runtime->dma_addr = 0;
+ runtime->dma_bytes = 0;
+
+ return 0;
+}
+
+static int snd_cs46xx_playback_prepare(struct snd_pcm_substream *substream)
+{
+ unsigned int tmp;
+ unsigned int pfie;
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_cs46xx_pcm *cpcm;
+
+ cpcm = runtime->private_data;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (snd_BUG_ON(!cpcm->pcm_channel))
+ return -ENXIO;
+
+ pfie = snd_cs46xx_peek(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 1) << 2 );
+ pfie &= ~0x0000f03f;
+#else
+ /* old dsp */
+ pfie = snd_cs46xx_peek(chip, BA1_PFIE);
+ pfie &= ~0x0000f03f;
+#endif
+
+ cpcm->shift = 2;
+ /* if to convert from stereo to mono */
+ if (runtime->channels == 1) {
+ cpcm->shift--;
+ pfie |= 0x00002000;
+ }
+ /* if to convert from 8 bit to 16 bit */
+ if (snd_pcm_format_width(runtime->format) == 8) {
+ cpcm->shift--;
+ pfie |= 0x00001000;
+ }
+ /* if to convert to unsigned */
+ if (snd_pcm_format_unsigned(runtime->format))
+ pfie |= 0x00008000;
+
+ /* Never convert byte order when sample stream is 8 bit */
+ if (snd_pcm_format_width(runtime->format) != 8) {
+ /* convert from big endian to little endian */
+ if (snd_pcm_format_big_endian(runtime->format))
+ pfie |= 0x00004000;
+ }
+
+ memset(&cpcm->pcm_rec, 0, sizeof(cpcm->pcm_rec));
+ cpcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ cpcm->pcm_rec.hw_buffer_size = runtime->period_size * CS46XX_FRAGS << cpcm->shift;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+
+ tmp = snd_cs46xx_peek(chip, (cpcm->pcm_channel->pcm_reader_scb->address) << 2);
+ tmp &= ~0x000003ff;
+ tmp |= (4 << cpcm->shift) - 1;
+ /* playback transaction count register */
+ snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address) << 2, tmp);
+
+ /* playback format && interrupt enable */
+ snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 1) << 2, pfie | cpcm->pcm_channel->pcm_slot);
+#else
+ snd_cs46xx_poke(chip, BA1_PBA, cpcm->hw_buf.addr);
+ tmp = snd_cs46xx_peek(chip, BA1_PDTC);
+ tmp &= ~0x000003ff;
+ tmp |= (4 << cpcm->shift) - 1;
+ snd_cs46xx_poke(chip, BA1_PDTC, tmp);
+ snd_cs46xx_poke(chip, BA1_PFIE, pfie);
+ snd_cs46xx_set_play_sample_rate(chip, runtime->rate);
+#endif
+
+ return 0;
+}
+
+static int snd_cs46xx_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ cs46xx_dsp_pcm_ostream_set_period (chip, params_period_bytes(hw_params));
+#endif
+ if (runtime->periods == CS46XX_FRAGS) {
+ if (runtime->dma_area != chip->capt.hw_buf.area)
+ snd_pcm_lib_free_pages(substream);
+ runtime->dma_area = chip->capt.hw_buf.area;
+ runtime->dma_addr = chip->capt.hw_buf.addr;
+ runtime->dma_bytes = chip->capt.hw_buf.bytes;
+ substream->ops = &snd_cs46xx_capture_ops;
+ } else {
+ if (runtime->dma_area == chip->capt.hw_buf.area) {
+ runtime->dma_area = NULL;
+ runtime->dma_addr = 0;
+ runtime->dma_bytes = 0;
+ }
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ substream->ops = &snd_cs46xx_capture_indirect_ops;
+ }
+
+ return 0;
+}
+
+static int snd_cs46xx_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (runtime->dma_area != chip->capt.hw_buf.area)
+ snd_pcm_lib_free_pages(substream);
+ runtime->dma_area = NULL;
+ runtime->dma_addr = 0;
+ runtime->dma_bytes = 0;
+
+ return 0;
+}
+
+static int snd_cs46xx_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_cs46xx_poke(chip, BA1_CBA, chip->capt.hw_buf.addr);
+ chip->capt.shift = 2;
+ memset(&chip->capt.pcm_rec, 0, sizeof(chip->capt.pcm_rec));
+ chip->capt.pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ chip->capt.pcm_rec.hw_buffer_size = runtime->period_size * CS46XX_FRAGS << 2;
+ snd_cs46xx_set_capture_sample_rate(chip, runtime->rate);
+
+ return 0;
+}
+
+static irqreturn_t snd_cs46xx_interrupt(int irq, void *dev_id)
+{
+ struct snd_cs46xx *chip = dev_id;
+ u32 status1;
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ u32 status2;
+ int i;
+ struct snd_cs46xx_pcm *cpcm = NULL;
+#endif
+
+ /*
+ * Read the Interrupt Status Register to clear the interrupt
+ */
+ status1 = snd_cs46xx_peekBA0(chip, BA0_HISR);
+ if ((status1 & 0x7fffffff) == 0) {
+ snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_CHGM | HICR_IEV);
+ return IRQ_NONE;
+ }
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ status2 = snd_cs46xx_peekBA0(chip, BA0_HSR0);
+
+ for (i = 0; i < DSP_MAX_PCM_CHANNELS; ++i) {
+ if (i <= 15) {
+ if ( status1 & (1 << i) ) {
+ if (i == CS46XX_DSP_CAPTURE_CHANNEL) {
+ if (chip->capt.substream)
+ snd_pcm_period_elapsed(chip->capt.substream);
+ } else {
+ if (ins->pcm_channels[i].active &&
+ ins->pcm_channels[i].private_data &&
+ !ins->pcm_channels[i].unlinked) {
+ cpcm = ins->pcm_channels[i].private_data;
+ snd_pcm_period_elapsed(cpcm->substream);
+ }
+ }
+ }
+ } else {
+ if ( status2 & (1 << (i - 16))) {
+ if (ins->pcm_channels[i].active &&
+ ins->pcm_channels[i].private_data &&
+ !ins->pcm_channels[i].unlinked) {
+ cpcm = ins->pcm_channels[i].private_data;
+ snd_pcm_period_elapsed(cpcm->substream);
+ }
+ }
+ }
+ }
+
+#else
+ /* old dsp */
+ if ((status1 & HISR_VC0) && chip->playback_pcm) {
+ if (chip->playback_pcm->substream)
+ snd_pcm_period_elapsed(chip->playback_pcm->substream);
+ }
+ if ((status1 & HISR_VC1) && chip->pcm) {
+ if (chip->capt.substream)
+ snd_pcm_period_elapsed(chip->capt.substream);
+ }
+#endif
+
+ if ((status1 & HISR_MIDI) && chip->rmidi) {
+ unsigned char c;
+
+ spin_lock(&chip->reg_lock);
+ while ((snd_cs46xx_peekBA0(chip, BA0_MIDSR) & MIDSR_RBE) == 0) {
+ c = snd_cs46xx_peekBA0(chip, BA0_MIDRP);
+ if ((chip->midcr & MIDCR_RIE) == 0)
+ continue;
+ snd_rawmidi_receive(chip->midi_input, &c, 1);
+ }
+ while ((snd_cs46xx_peekBA0(chip, BA0_MIDSR) & MIDSR_TBF) == 0) {
+ if ((chip->midcr & MIDCR_TIE) == 0)
+ break;
+ if (snd_rawmidi_transmit(chip->midi_output, &c, 1) != 1) {
+ chip->midcr &= ~MIDCR_TIE;
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ break;
+ }
+ snd_cs46xx_pokeBA0(chip, BA0_MIDWP, c);
+ }
+ spin_unlock(&chip->reg_lock);
+ }
+ /*
+ * EOI to the PCI part....reenables interrupts
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_CHGM | HICR_IEV);
+
+ return IRQ_HANDLED;
+}
+
+static struct snd_pcm_hardware snd_cs46xx_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
+ /*SNDRV_PCM_INFO_RESUME*/),
+ .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5500,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (256 * 1024),
+ .period_bytes_min = CS46XX_MIN_PERIOD_SIZE,
+ .period_bytes_max = CS46XX_MAX_PERIOD_SIZE,
+ .periods_min = CS46XX_FRAGS,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_cs46xx_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER /*|*/
+ /*SNDRV_PCM_INFO_RESUME*/),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5500,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (256 * 1024),
+ .period_bytes_min = CS46XX_MIN_PERIOD_SIZE,
+ .period_bytes_max = CS46XX_MAX_PERIOD_SIZE,
+ .periods_min = CS46XX_FRAGS,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+
+static unsigned int period_sizes[] = { 32, 64, 128, 256, 512, 1024, 2048 };
+
+static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes = {
+ .count = ARRAY_SIZE(period_sizes),
+ .list = period_sizes,
+ .mask = 0
+};
+
+#endif
+
+static void snd_cs46xx_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static int _cs46xx_playback_open_channel (struct snd_pcm_substream *substream,int pcm_channel_id)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_cs46xx_pcm * cpcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ cpcm = kzalloc(sizeof(*cpcm), GFP_KERNEL);
+ if (cpcm == NULL)
+ return -ENOMEM;
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ PAGE_SIZE, &cpcm->hw_buf) < 0) {
+ kfree(cpcm);
+ return -ENOMEM;
+ }
+
+ runtime->hw = snd_cs46xx_playback;
+ runtime->private_data = cpcm;
+ runtime->private_free = snd_cs46xx_pcm_free_substream;
+
+ cpcm->substream = substream;
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ mutex_lock(&chip->spos_mutex);
+ cpcm->pcm_channel = NULL;
+ cpcm->pcm_channel_id = pcm_channel_id;
+
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ &hw_constraints_period_sizes);
+
+ mutex_unlock(&chip->spos_mutex);
+#else
+ chip->playback_pcm = cpcm; /* HACK */
+#endif
+
+ if (chip->accept_valid)
+ substream->runtime->hw.info |= SNDRV_PCM_INFO_MMAP_VALID;
+ chip->active_ctrl(chip, 1);
+
+ return 0;
+}
+
+static int snd_cs46xx_playback_open(struct snd_pcm_substream *substream)
+{
+ snd_printdd("open front channel\n");
+ return _cs46xx_playback_open_channel(substream,DSP_PCM_MAIN_CHANNEL);
+}
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+static int snd_cs46xx_playback_open_rear(struct snd_pcm_substream *substream)
+{
+ snd_printdd("open rear channel\n");
+
+ return _cs46xx_playback_open_channel(substream,DSP_PCM_REAR_CHANNEL);
+}
+
+static int snd_cs46xx_playback_open_clfe(struct snd_pcm_substream *substream)
+{
+ snd_printdd("open center - LFE channel\n");
+
+ return _cs46xx_playback_open_channel(substream,DSP_PCM_CENTER_LFE_CHANNEL);
+}
+
+static int snd_cs46xx_playback_open_iec958(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+
+ snd_printdd("open raw iec958 channel\n");
+
+ mutex_lock(&chip->spos_mutex);
+ cs46xx_iec958_pre_open (chip);
+ mutex_unlock(&chip->spos_mutex);
+
+ return _cs46xx_playback_open_channel(substream,DSP_IEC958_CHANNEL);
+}
+
+static int snd_cs46xx_playback_close(struct snd_pcm_substream *substream);
+
+static int snd_cs46xx_playback_close_iec958(struct snd_pcm_substream *substream)
+{
+ int err;
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+
+ snd_printdd("close raw iec958 channel\n");
+
+ err = snd_cs46xx_playback_close(substream);
+
+ mutex_lock(&chip->spos_mutex);
+ cs46xx_iec958_post_close (chip);
+ mutex_unlock(&chip->spos_mutex);
+
+ return err;
+}
+#endif
+
+static int snd_cs46xx_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ PAGE_SIZE, &chip->capt.hw_buf) < 0)
+ return -ENOMEM;
+ chip->capt.substream = substream;
+ substream->runtime->hw = snd_cs46xx_capture;
+
+ if (chip->accept_valid)
+ substream->runtime->hw.info |= SNDRV_PCM_INFO_MMAP_VALID;
+
+ chip->active_ctrl(chip, 1);
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ &hw_constraints_period_sizes);
+#endif
+ return 0;
+}
+
+static int snd_cs46xx_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_cs46xx_pcm * cpcm;
+
+ cpcm = runtime->private_data;
+
+ /* when playback_open fails, then cpcm can be NULL */
+ if (!cpcm) return -ENXIO;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ mutex_lock(&chip->spos_mutex);
+ if (cpcm->pcm_channel) {
+ cs46xx_dsp_destroy_pcm_channel(chip,cpcm->pcm_channel);
+ cpcm->pcm_channel = NULL;
+ }
+ mutex_unlock(&chip->spos_mutex);
+#else
+ chip->playback_pcm = NULL;
+#endif
+
+ cpcm->substream = NULL;
+ snd_dma_free_pages(&cpcm->hw_buf);
+ chip->active_ctrl(chip, -1);
+
+ return 0;
+}
+
+static int snd_cs46xx_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_cs46xx *chip = snd_pcm_substream_chip(substream);
+
+ chip->capt.substream = NULL;
+ snd_dma_free_pages(&chip->capt.hw_buf);
+ chip->active_ctrl(chip, -1);
+
+ return 0;
+}
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+static struct snd_pcm_ops snd_cs46xx_playback_rear_ops = {
+ .open = snd_cs46xx_playback_open_rear,
+ .close = snd_cs46xx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_direct_pointer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_rear_ops = {
+ .open = snd_cs46xx_playback_open_rear,
+ .close = snd_cs46xx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_indirect_pointer,
+ .ack = snd_cs46xx_playback_transfer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_playback_clfe_ops = {
+ .open = snd_cs46xx_playback_open_clfe,
+ .close = snd_cs46xx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_direct_pointer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_clfe_ops = {
+ .open = snd_cs46xx_playback_open_clfe,
+ .close = snd_cs46xx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_indirect_pointer,
+ .ack = snd_cs46xx_playback_transfer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_playback_iec958_ops = {
+ .open = snd_cs46xx_playback_open_iec958,
+ .close = snd_cs46xx_playback_close_iec958,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_direct_pointer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_iec958_ops = {
+ .open = snd_cs46xx_playback_open_iec958,
+ .close = snd_cs46xx_playback_close_iec958,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_indirect_pointer,
+ .ack = snd_cs46xx_playback_transfer,
+};
+
+#endif
+
+static struct snd_pcm_ops snd_cs46xx_playback_ops = {
+ .open = snd_cs46xx_playback_open,
+ .close = snd_cs46xx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_direct_pointer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_playback_indirect_ops = {
+ .open = snd_cs46xx_playback_open,
+ .close = snd_cs46xx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_playback_hw_params,
+ .hw_free = snd_cs46xx_playback_hw_free,
+ .prepare = snd_cs46xx_playback_prepare,
+ .trigger = snd_cs46xx_playback_trigger,
+ .pointer = snd_cs46xx_playback_indirect_pointer,
+ .ack = snd_cs46xx_playback_transfer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_capture_ops = {
+ .open = snd_cs46xx_capture_open,
+ .close = snd_cs46xx_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_capture_hw_params,
+ .hw_free = snd_cs46xx_capture_hw_free,
+ .prepare = snd_cs46xx_capture_prepare,
+ .trigger = snd_cs46xx_capture_trigger,
+ .pointer = snd_cs46xx_capture_direct_pointer,
+};
+
+static struct snd_pcm_ops snd_cs46xx_capture_indirect_ops = {
+ .open = snd_cs46xx_capture_open,
+ .close = snd_cs46xx_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs46xx_capture_hw_params,
+ .hw_free = snd_cs46xx_capture_hw_free,
+ .prepare = snd_cs46xx_capture_prepare,
+ .trigger = snd_cs46xx_capture_trigger,
+ .pointer = snd_cs46xx_capture_indirect_pointer,
+ .ack = snd_cs46xx_capture_transfer,
+};
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+#define MAX_PLAYBACK_CHANNELS (DSP_MAX_PCM_CHANNELS - 1)
+#else
+#define MAX_PLAYBACK_CHANNELS 1
+#endif
+
+int __devinit snd_cs46xx_pcm(struct snd_cs46xx *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(chip->card, "CS46xx", device, MAX_PLAYBACK_CHANNELS, 1, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs46xx_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cs46xx_capture_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "CS46xx");
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+int __devinit snd_cs46xx_pcm_rear(struct snd_cs46xx *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(chip->card, "CS46xx - Rear", device, MAX_PLAYBACK_CHANNELS, 0, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs46xx_playback_rear_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "CS46xx - Rear");
+ chip->pcm_rear = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+int __devinit snd_cs46xx_pcm_center_lfe(struct snd_cs46xx *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(chip->card, "CS46xx - Center LFE", device, MAX_PLAYBACK_CHANNELS, 0, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs46xx_playback_clfe_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "CS46xx - Center LFE");
+ chip->pcm_center_lfe = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+int __devinit snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(chip->card, "CS46xx - IEC958", device, 1, 0, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs46xx_playback_iec958_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "CS46xx - IEC958");
+ chip->pcm_rear = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+#endif
+
+/*
+ * Mixer routines
+ */
+static void snd_cs46xx_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct snd_cs46xx *chip = bus->private_data;
+
+ chip->ac97_bus = NULL;
+}
+
+static void snd_cs46xx_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct snd_cs46xx *chip = ac97->private_data;
+
+ if (snd_BUG_ON(ac97 != chip->ac97[CS46XX_PRIMARY_CODEC_INDEX] &&
+ ac97 != chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]))
+ return;
+
+ if (ac97 == chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]) {
+ chip->ac97[CS46XX_PRIMARY_CODEC_INDEX] = NULL;
+ chip->eapd_switch = NULL;
+ }
+ else
+ chip->ac97[CS46XX_SECONDARY_CODEC_INDEX] = NULL;
+}
+
+static int snd_cs46xx_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0x7fff;
+ return 0;
+}
+
+static int snd_cs46xx_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value;
+ unsigned int val = snd_cs46xx_peek(chip, reg);
+ ucontrol->value.integer.value[0] = 0xffff - (val >> 16);
+ ucontrol->value.integer.value[1] = 0xffff - (val & 0xffff);
+ return 0;
+}
+
+static int snd_cs46xx_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value;
+ unsigned int val = ((0xffff - ucontrol->value.integer.value[0]) << 16 |
+ (0xffff - ucontrol->value.integer.value[1]));
+ unsigned int old = snd_cs46xx_peek(chip, reg);
+ int change = (old != val);
+
+ if (change) {
+ snd_cs46xx_poke(chip, reg, val);
+ }
+
+ return change;
+}
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+
+static int snd_cs46xx_vol_dac_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_left;
+ ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_right;
+
+ return 0;
+}
+
+static int snd_cs46xx_vol_dac_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ int change = 0;
+
+ if (chip->dsp_spos_instance->dac_volume_right != ucontrol->value.integer.value[0] ||
+ chip->dsp_spos_instance->dac_volume_left != ucontrol->value.integer.value[1]) {
+ cs46xx_dsp_set_dac_volume(chip,
+ ucontrol->value.integer.value[0],
+ ucontrol->value.integer.value[1]);
+ change = 1;
+ }
+
+ return change;
+}
+
+#if 0
+static int snd_cs46xx_vol_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = chip->dsp_spos_instance->spdif_input_volume_left;
+ ucontrol->value.integer.value[1] = chip->dsp_spos_instance->spdif_input_volume_right;
+ return 0;
+}
+
+static int snd_cs46xx_vol_iec958_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ int change = 0;
+
+ if (chip->dsp_spos_instance->spdif_input_volume_left != ucontrol->value.integer.value[0] ||
+ chip->dsp_spos_instance->spdif_input_volume_right!= ucontrol->value.integer.value[1]) {
+ cs46xx_dsp_set_iec958_volume (chip,
+ ucontrol->value.integer.value[0],
+ ucontrol->value.integer.value[1]);
+ change = 1;
+ }
+
+ return change;
+}
+#endif
+
+#define snd_mixer_boolean_info snd_ctl_boolean_mono_info
+
+static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value;
+
+ if (reg == CS46XX_MIXER_SPDIF_OUTPUT_ELEMENT)
+ ucontrol->value.integer.value[0] = (chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED);
+ else
+ ucontrol->value.integer.value[0] = chip->dsp_spos_instance->spdif_status_in;
+
+ return 0;
+}
+
+static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ int change, res;
+
+ switch (kcontrol->private_value) {
+ case CS46XX_MIXER_SPDIF_OUTPUT_ELEMENT:
+ mutex_lock(&chip->spos_mutex);
+ change = (chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED);
+ if (ucontrol->value.integer.value[0] && !change)
+ cs46xx_dsp_enable_spdif_out(chip);
+ else if (change && !ucontrol->value.integer.value[0])
+ cs46xx_dsp_disable_spdif_out(chip);
+
+ res = (change != (chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED));
+ mutex_unlock(&chip->spos_mutex);
+ break;
+ case CS46XX_MIXER_SPDIF_INPUT_ELEMENT:
+ change = chip->dsp_spos_instance->spdif_status_in;
+ if (ucontrol->value.integer.value[0] && !change) {
+ cs46xx_dsp_enable_spdif_in(chip);
+ /* restore volume */
+ }
+ else if (change && !ucontrol->value.integer.value[0])
+ cs46xx_dsp_disable_spdif_in(chip);
+
+ res = (change != chip->dsp_spos_instance->spdif_status_in);
+ break;
+ default:
+ res = -EINVAL;
+ snd_BUG(); /* should never happen ... */
+ }
+
+ return res;
+}
+
+static int snd_cs46xx_adc_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (ins->adc_input != NULL)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+static int snd_cs46xx_adc_capture_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int change = 0;
+
+ if (ucontrol->value.integer.value[0] && !ins->adc_input) {
+ cs46xx_dsp_enable_adc_capture(chip);
+ change = 1;
+ } else if (!ucontrol->value.integer.value[0] && ins->adc_input) {
+ cs46xx_dsp_disable_adc_capture(chip);
+ change = 1;
+ }
+ return change;
+}
+
+static int snd_cs46xx_pcm_capture_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (ins->pcm_input != NULL)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+
+static int snd_cs46xx_pcm_capture_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int change = 0;
+
+ if (ucontrol->value.integer.value[0] && !ins->pcm_input) {
+ cs46xx_dsp_enable_pcm_capture(chip);
+ change = 1;
+ } else if (!ucontrol->value.integer.value[0] && ins->pcm_input) {
+ cs46xx_dsp_disable_pcm_capture(chip);
+ change = 1;
+ }
+
+ return change;
+}
+
+static int snd_herc_spdif_select_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+
+ int val1 = snd_cs46xx_peekBA0(chip, BA0_EGPIODR);
+
+ if (val1 & EGPIODR_GPOE0)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+/*
+ * Game Theatre XP card - EGPIO[0] is used to select SPDIF input optical or coaxial.
+ */
+static int snd_herc_spdif_select_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ int val1 = snd_cs46xx_peekBA0(chip, BA0_EGPIODR);
+ int val2 = snd_cs46xx_peekBA0(chip, BA0_EGPIOPTR);
+
+ if (ucontrol->value.integer.value[0]) {
+ /* optical is default */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIODR,
+ EGPIODR_GPOE0 | val1); /* enable EGPIO0 output */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR,
+ EGPIOPTR_GPPT0 | val2); /* open-drain on output */
+ } else {
+ /* coaxial */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, val1 & ~EGPIODR_GPOE0); /* disable */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, val2 & ~EGPIOPTR_GPPT0); /* disable */
+ }
+
+ /* checking diff from the EGPIO direction register
+ should be enough */
+ return (val1 != (int)snd_cs46xx_peekBA0(chip, BA0_EGPIODR));
+}
+
+
+static int snd_cs46xx_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cs46xx_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ mutex_lock(&chip->spos_mutex);
+ ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_default >> 24) & 0xff);
+ ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_default >> 16) & 0xff);
+ ucontrol->value.iec958.status[2] = 0;
+ ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_default) & 0xff);
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+static int snd_cs46xx_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx * chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ unsigned int val;
+ int change;
+
+ mutex_lock(&chip->spos_mutex);
+ val = ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[0]) << 24) |
+ ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[2]) << 16) |
+ ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[3])) |
+ /* left and right validity bit */
+ (1 << 13) | (1 << 12);
+
+
+ change = (unsigned int)ins->spdif_csuv_default != val;
+ ins->spdif_csuv_default = val;
+
+ if ( !(ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) )
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val);
+
+ mutex_unlock(&chip->spos_mutex);
+
+ return change;
+}
+
+static int snd_cs46xx_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0x00;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static int snd_cs46xx_spdif_stream_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ mutex_lock(&chip->spos_mutex);
+ ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_stream >> 24) & 0xff);
+ ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_stream >> 16) & 0xff);
+ ucontrol->value.iec958.status[2] = 0;
+ ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_stream) & 0xff);
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx * chip = snd_kcontrol_chip(kcontrol);
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ unsigned int val;
+ int change;
+
+ mutex_lock(&chip->spos_mutex);
+ val = ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[0]) << 24) |
+ ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[1]) << 16) |
+ ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[3])) |
+ /* left and right validity bit */
+ (1 << 13) | (1 << 12);
+
+
+ change = ins->spdif_csuv_stream != val;
+ ins->spdif_csuv_stream = val;
+
+ if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN )
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val);
+
+ mutex_unlock(&chip->spos_mutex);
+
+ return change;
+}
+
+#endif /* CONFIG_SND_CS46XX_NEW_DSP */
+
+
+static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Volume",
+ .info = snd_cs46xx_vol_info,
+#ifndef CONFIG_SND_CS46XX_NEW_DSP
+ .get = snd_cs46xx_vol_get,
+ .put = snd_cs46xx_vol_put,
+ .private_value = BA1_PVOL,
+#else
+ .get = snd_cs46xx_vol_dac_get,
+ .put = snd_cs46xx_vol_dac_put,
+#endif
+},
+
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Volume",
+ .info = snd_cs46xx_vol_info,
+ .get = snd_cs46xx_vol_get,
+ .put = snd_cs46xx_vol_put,
+#ifndef CONFIG_SND_CS46XX_NEW_DSP
+ .private_value = BA1_CVOL,
+#else
+ .private_value = (VARIDECIMATE_SCB_ADDR + 0xE) << 2,
+#endif
+},
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Capture Switch",
+ .info = snd_mixer_boolean_info,
+ .get = snd_cs46xx_adc_capture_get,
+ .put = snd_cs46xx_adc_capture_put
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Capture Switch",
+ .info = snd_mixer_boolean_info,
+ .get = snd_cs46xx_pcm_capture_get,
+ .put = snd_cs46xx_pcm_capture_put
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH),
+ .info = snd_mixer_boolean_info,
+ .get = snd_cs46xx_iec958_get,
+ .put = snd_cs46xx_iec958_put,
+ .private_value = CS46XX_MIXER_SPDIF_OUTPUT_ELEMENT,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("Input ",NONE,SWITCH),
+ .info = snd_mixer_boolean_info,
+ .get = snd_cs46xx_iec958_get,
+ .put = snd_cs46xx_iec958_put,
+ .private_value = CS46XX_MIXER_SPDIF_INPUT_ELEMENT,
+},
+#if 0
+/* Input IEC958 volume does not work for the moment. (Benny) */
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("Input ",NONE,VOLUME),
+ .info = snd_cs46xx_vol_info,
+ .get = snd_cs46xx_vol_iec958_get,
+ .put = snd_cs46xx_vol_iec958_put,
+ .private_value = (ASYNCRX_SCB_ADDR + 0xE) << 2,
+},
+#endif
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_cs46xx_spdif_info,
+ .get = snd_cs46xx_spdif_default_get,
+ .put = snd_cs46xx_spdif_default_put,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .info = snd_cs46xx_spdif_info,
+ .get = snd_cs46xx_spdif_mask_get,
+ .access = SNDRV_CTL_ELEM_ACCESS_READ
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_cs46xx_spdif_info,
+ .get = snd_cs46xx_spdif_stream_get,
+ .put = snd_cs46xx_spdif_stream_put
+},
+
+#endif
+};
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+/* set primary cs4294 codec into Extended Audio Mode */
+static int snd_cs46xx_front_dup_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+ val = snd_ac97_read(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX], AC97_CSR_ACMODE);
+ ucontrol->value.integer.value[0] = (val & 0x200) ? 0 : 1;
+ return 0;
+}
+
+static int snd_cs46xx_front_dup_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol);
+ return snd_ac97_update_bits(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX],
+ AC97_CSR_ACMODE, 0x200,
+ ucontrol->value.integer.value[0] ? 0 : 0x200);
+}
+
+static struct snd_kcontrol_new snd_cs46xx_front_dup_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Duplicate Front",
+ .info = snd_mixer_boolean_info,
+ .get = snd_cs46xx_front_dup_get,
+ .put = snd_cs46xx_front_dup_put,
+};
+#endif
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+/* Only available on the Hercules Game Theater XP soundcard */
+static struct snd_kcontrol_new snd_hercules_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Optical/Coaxial SPDIF Input Switch",
+ .info = snd_mixer_boolean_info,
+ .get = snd_herc_spdif_select_get,
+ .put = snd_herc_spdif_select_put,
+},
+};
+
+
+static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97)
+{
+ unsigned long end_time;
+ int err;
+
+ /* reset to defaults */
+ snd_ac97_write(ac97, AC97_RESET, 0);
+
+ /* set the desired CODEC mode */
+ if (ac97->num == CS46XX_PRIMARY_CODEC_INDEX) {
+ snd_printdd("cs46xx: CODOEC1 mode %04x\n",0x0);
+ snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x0);
+ } else if (ac97->num == CS46XX_SECONDARY_CODEC_INDEX) {
+ snd_printdd("cs46xx: CODOEC2 mode %04x\n",0x3);
+ snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x3);
+ } else {
+ snd_BUG(); /* should never happen ... */
+ }
+
+ udelay(50);
+
+ /* it's necessary to wait awhile until registers are accessible after RESET */
+ /* because the PCM or MASTER volume registers can be modified, */
+ /* the REC_GAIN register is used for tests */
+ end_time = jiffies + HZ;
+ do {
+ unsigned short ext_mid;
+
+ /* use preliminary reads to settle the communication */
+ snd_ac97_read(ac97, AC97_RESET);
+ snd_ac97_read(ac97, AC97_VENDOR_ID1);
+ snd_ac97_read(ac97, AC97_VENDOR_ID2);
+ /* modem? */
+ ext_mid = snd_ac97_read(ac97, AC97_EXTENDED_MID);
+ if (ext_mid != 0xffff && (ext_mid & 1) != 0)
+ return;
+
+ /* test if we can write to the record gain volume register */
+ snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x8a05);
+ if ((err = snd_ac97_read(ac97, AC97_REC_GAIN)) == 0x8a05)
+ return;
+
+ msleep(10);
+ } while (time_after_eq(end_time, jiffies));
+
+ snd_printk(KERN_ERR "CS46xx secondary codec doesn't respond!\n");
+}
+#endif
+
+static int __devinit cs46xx_detect_codec(struct snd_cs46xx *chip, int codec)
+{
+ int idx, err;
+ struct snd_ac97_template ac97;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_cs46xx_mixer_free_ac97;
+ ac97.num = codec;
+ if (chip->amplifier_ctrl == amp_voyetra)
+ ac97.scaps = AC97_SCAP_INV_EAPD;
+
+ if (codec == CS46XX_SECONDARY_CODEC_INDEX) {
+ snd_cs46xx_codec_write(chip, AC97_RESET, 0, codec);
+ udelay(10);
+ if (snd_cs46xx_codec_read(chip, AC97_RESET, codec) & 0x8000) {
+ snd_printdd("snd_cs46xx: seconadry codec not present\n");
+ return -ENXIO;
+ }
+ }
+
+ snd_cs46xx_codec_write(chip, AC97_MASTER, 0x8000, codec);
+ for (idx = 0; idx < 100; ++idx) {
+ if (snd_cs46xx_codec_read(chip, AC97_MASTER, codec) == 0x8000) {
+ err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97[codec]);
+ return err;
+ }
+ msleep(10);
+ }
+ snd_printdd("snd_cs46xx: codec %d detection timeout\n", codec);
+ return -ENXIO;
+}
+
+int __devinit snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device)
+{
+ struct snd_card *card = chip->card;
+ struct snd_ctl_elem_id id;
+ int err;
+ unsigned int idx;
+ static struct snd_ac97_bus_ops ops = {
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ .reset = snd_cs46xx_codec_reset,
+#endif
+ .write = snd_cs46xx_ac97_write,
+ .read = snd_cs46xx_ac97_read,
+ };
+
+ /* detect primary codec */
+ chip->nr_ac97_codecs = 0;
+ snd_printdd("snd_cs46xx: detecting primary codec\n");
+ if ((err = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus)) < 0)
+ return err;
+ chip->ac97_bus->private_free = snd_cs46xx_mixer_free_ac97_bus;
+
+ if (cs46xx_detect_codec(chip, CS46XX_PRIMARY_CODEC_INDEX) < 0)
+ return -ENXIO;
+ chip->nr_ac97_codecs = 1;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_printdd("snd_cs46xx: detecting seconadry codec\n");
+ /* try detect a secondary codec */
+ if (! cs46xx_detect_codec(chip, CS46XX_SECONDARY_CODEC_INDEX))
+ chip->nr_ac97_codecs = 2;
+#endif /* CONFIG_SND_CS46XX_NEW_DSP */
+
+ /* add cs4630 mixer controls */
+ for (idx = 0; idx < ARRAY_SIZE(snd_cs46xx_controls); idx++) {
+ struct snd_kcontrol *kctl;
+ kctl = snd_ctl_new1(&snd_cs46xx_controls[idx], chip);
+ if (kctl && kctl->id.iface == SNDRV_CTL_ELEM_IFACE_PCM)
+ kctl->id.device = spdif_device;
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ return err;
+ }
+
+ /* get EAPD mixer switch (for voyetra hack) */
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, "External Amplifier");
+ chip->eapd_switch = snd_ctl_find_id(chip->card, &id);
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (chip->nr_ac97_codecs == 1) {
+ unsigned int id2 = chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]->id & 0xffff;
+ if (id2 == 0x592b || id2 == 0x592d) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_cs46xx_front_dup_ctl, chip));
+ if (err < 0)
+ return err;
+ snd_ac97_write_cache(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX],
+ AC97_CSR_ACMODE, 0x200);
+ }
+ }
+ /* do soundcard specific mixer setup */
+ if (chip->mixer_init) {
+ snd_printdd ("calling chip->mixer_init(chip);\n");
+ chip->mixer_init(chip);
+ }
+#endif
+
+ /* turn on amplifier */
+ chip->amplifier_ctrl(chip, 1);
+
+ return 0;
+}
+
+/*
+ * RawMIDI interface
+ */
+
+static void snd_cs46xx_midi_reset(struct snd_cs46xx *chip)
+{
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, MIDCR_MRST);
+ udelay(100);
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+}
+
+static int snd_cs46xx_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_cs46xx *chip = substream->rmidi->private_data;
+
+ chip->active_ctrl(chip, 1);
+ spin_lock_irq(&chip->reg_lock);
+ chip->uartm |= CS46XX_MODE_INPUT;
+ chip->midcr |= MIDCR_RXE;
+ chip->midi_input = substream;
+ if (!(chip->uartm & CS46XX_MODE_OUTPUT)) {
+ snd_cs46xx_midi_reset(chip);
+ } else {
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cs46xx_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_cs46xx *chip = substream->rmidi->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->midcr &= ~(MIDCR_RXE | MIDCR_RIE);
+ chip->midi_input = NULL;
+ if (!(chip->uartm & CS46XX_MODE_OUTPUT)) {
+ snd_cs46xx_midi_reset(chip);
+ } else {
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ chip->uartm &= ~CS46XX_MODE_INPUT;
+ spin_unlock_irq(&chip->reg_lock);
+ chip->active_ctrl(chip, -1);
+ return 0;
+}
+
+static int snd_cs46xx_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_cs46xx *chip = substream->rmidi->private_data;
+
+ chip->active_ctrl(chip, 1);
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->uartm |= CS46XX_MODE_OUTPUT;
+ chip->midcr |= MIDCR_TXE;
+ chip->midi_output = substream;
+ if (!(chip->uartm & CS46XX_MODE_INPUT)) {
+ snd_cs46xx_midi_reset(chip);
+ } else {
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_cs46xx_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_cs46xx *chip = substream->rmidi->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->midcr &= ~(MIDCR_TXE | MIDCR_TIE);
+ chip->midi_output = NULL;
+ if (!(chip->uartm & CS46XX_MODE_INPUT)) {
+ snd_cs46xx_midi_reset(chip);
+ } else {
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ chip->uartm &= ~CS46XX_MODE_OUTPUT;
+ spin_unlock_irq(&chip->reg_lock);
+ chip->active_ctrl(chip, -1);
+ return 0;
+}
+
+static void snd_cs46xx_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct snd_cs46xx *chip = substream->rmidi->private_data;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (up) {
+ if ((chip->midcr & MIDCR_RIE) == 0) {
+ chip->midcr |= MIDCR_RIE;
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ } else {
+ if (chip->midcr & MIDCR_RIE) {
+ chip->midcr &= ~MIDCR_RIE;
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static void snd_cs46xx_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct snd_cs46xx *chip = substream->rmidi->private_data;
+ unsigned char byte;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (up) {
+ if ((chip->midcr & MIDCR_TIE) == 0) {
+ chip->midcr |= MIDCR_TIE;
+ /* fill UART FIFO buffer at first, and turn Tx interrupts only if necessary */
+ while ((chip->midcr & MIDCR_TIE) &&
+ (snd_cs46xx_peekBA0(chip, BA0_MIDSR) & MIDSR_TBF) == 0) {
+ if (snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ chip->midcr &= ~MIDCR_TIE;
+ } else {
+ snd_cs46xx_pokeBA0(chip, BA0_MIDWP, byte);
+ }
+ }
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ } else {
+ if (chip->midcr & MIDCR_TIE) {
+ chip->midcr &= ~MIDCR_TIE;
+ snd_cs46xx_pokeBA0(chip, BA0_MIDCR, chip->midcr);
+ }
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static struct snd_rawmidi_ops snd_cs46xx_midi_output =
+{
+ .open = snd_cs46xx_midi_output_open,
+ .close = snd_cs46xx_midi_output_close,
+ .trigger = snd_cs46xx_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_cs46xx_midi_input =
+{
+ .open = snd_cs46xx_midi_input_open,
+ .close = snd_cs46xx_midi_input_close,
+ .trigger = snd_cs46xx_midi_input_trigger,
+};
+
+int __devinit snd_cs46xx_midi(struct snd_cs46xx *chip, int device, struct snd_rawmidi **rrawmidi)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ if ((err = snd_rawmidi_new(chip->card, "CS46XX", device, 1, 1, &rmidi)) < 0)
+ return err;
+ strcpy(rmidi->name, "CS46XX");
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_cs46xx_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_cs46xx_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = chip;
+ chip->rmidi = rmidi;
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ return 0;
+}
+
+
+/*
+ * gameport interface
+ */
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+
+static void snd_cs46xx_gameport_trigger(struct gameport *gameport)
+{
+ struct snd_cs46xx *chip = gameport_get_port_data(gameport);
+
+ if (snd_BUG_ON(!chip))
+ return;
+ snd_cs46xx_pokeBA0(chip, BA0_JSPT, 0xFF); //outb(gameport->io, 0xFF);
+}
+
+static unsigned char snd_cs46xx_gameport_read(struct gameport *gameport)
+{
+ struct snd_cs46xx *chip = gameport_get_port_data(gameport);
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+ return snd_cs46xx_peekBA0(chip, BA0_JSPT); //inb(gameport->io);
+}
+
+static int snd_cs46xx_gameport_cooked_read(struct gameport *gameport, int *axes, int *buttons)
+{
+ struct snd_cs46xx *chip = gameport_get_port_data(gameport);
+ unsigned js1, js2, jst;
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+
+ js1 = snd_cs46xx_peekBA0(chip, BA0_JSC1);
+ js2 = snd_cs46xx_peekBA0(chip, BA0_JSC2);
+ jst = snd_cs46xx_peekBA0(chip, BA0_JSPT);
+
+ *buttons = (~jst >> 4) & 0x0F;
+
+ axes[0] = ((js1 & JSC1_Y1V_MASK) >> JSC1_Y1V_SHIFT) & 0xFFFF;
+ axes[1] = ((js1 & JSC1_X1V_MASK) >> JSC1_X1V_SHIFT) & 0xFFFF;
+ axes[2] = ((js2 & JSC2_Y2V_MASK) >> JSC2_Y2V_SHIFT) & 0xFFFF;
+ axes[3] = ((js2 & JSC2_X2V_MASK) >> JSC2_X2V_SHIFT) & 0xFFFF;
+
+ for(jst=0;jst<4;++jst)
+ if(axes[jst]==0xFFFF) axes[jst] = -1;
+ return 0;
+}
+
+static int snd_cs46xx_gameport_open(struct gameport *gameport, int mode)
+{
+ switch (mode) {
+ case GAMEPORT_MODE_COOKED:
+ return 0;
+ case GAMEPORT_MODE_RAW:
+ return 0;
+ default:
+ return -1;
+ }
+ return 0;
+}
+
+int __devinit snd_cs46xx_gameport(struct snd_cs46xx *chip)
+{
+ struct gameport *gp;
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "cs46xx: cannot allocate memory for gameport\n");
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "CS46xx Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+ gameport_set_port_data(gp, chip);
+
+ gp->open = snd_cs46xx_gameport_open;
+ gp->read = snd_cs46xx_gameport_read;
+ gp->trigger = snd_cs46xx_gameport_trigger;
+ gp->cooked_read = snd_cs46xx_gameport_cooked_read;
+
+ snd_cs46xx_pokeBA0(chip, BA0_JSIO, 0xFF); // ?
+ snd_cs46xx_pokeBA0(chip, BA0_JSCTL, JSCTL_SP_MEDIUM_SLOW);
+
+ gameport_register_port(gp);
+
+ return 0;
+}
+
+static inline void snd_cs46xx_remove_gameport(struct snd_cs46xx *chip)
+{
+ if (chip->gameport) {
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+ }
+}
+#else
+int __devinit snd_cs46xx_gameport(struct snd_cs46xx *chip) { return -ENOSYS; }
+static inline void snd_cs46xx_remove_gameport(struct snd_cs46xx *chip) { }
+#endif /* CONFIG_GAMEPORT */
+
+#ifdef CONFIG_PROC_FS
+/*
+ * proc interface
+ */
+
+static long snd_cs46xx_io_read(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, char __user *buf,
+ unsigned long count, unsigned long pos)
+{
+ long size;
+ struct snd_cs46xx_region *region = entry->private_data;
+
+ size = count;
+ if (pos + (size_t)size > region->size)
+ size = region->size - pos;
+ if (size > 0) {
+ if (copy_to_user_fromio(buf, region->remap_addr + pos, size))
+ return -EFAULT;
+ }
+ return size;
+}
+
+static struct snd_info_entry_ops snd_cs46xx_proc_io_ops = {
+ .read = snd_cs46xx_io_read,
+};
+
+static int __devinit snd_cs46xx_proc_init(struct snd_card *card, struct snd_cs46xx *chip)
+{
+ struct snd_info_entry *entry;
+ int idx;
+
+ for (idx = 0; idx < 5; idx++) {
+ struct snd_cs46xx_region *region = &chip->region.idx[idx];
+ if (! snd_card_proc_new(card, region->name, &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = chip;
+ entry->c.ops = &snd_cs46xx_proc_io_ops;
+ entry->size = region->size;
+ entry->mode = S_IFREG | S_IRUSR;
+ }
+ }
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ cs46xx_dsp_proc_init(card, chip);
+#endif
+ return 0;
+}
+
+static int snd_cs46xx_proc_done(struct snd_cs46xx *chip)
+{
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ cs46xx_dsp_proc_done(chip);
+#endif
+ return 0;
+}
+#else /* !CONFIG_PROC_FS */
+#define snd_cs46xx_proc_init(card, chip)
+#define snd_cs46xx_proc_done(chip)
+#endif
+
+/*
+ * stop the h/w
+ */
+static void snd_cs46xx_hw_stop(struct snd_cs46xx *chip)
+{
+ unsigned int tmp;
+
+ tmp = snd_cs46xx_peek(chip, BA1_PFIE);
+ tmp &= ~0x0000f03f;
+ tmp |= 0x00000010;
+ snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt disable */
+
+ tmp = snd_cs46xx_peek(chip, BA1_CIE);
+ tmp &= ~0x0000003f;
+ tmp |= 0x00000011;
+ snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt disable */
+
+ /*
+ * Stop playback DMA.
+ */
+ tmp = snd_cs46xx_peek(chip, BA1_PCTL);
+ snd_cs46xx_poke(chip, BA1_PCTL, tmp & 0x0000ffff);
+
+ /*
+ * Stop capture DMA.
+ */
+ tmp = snd_cs46xx_peek(chip, BA1_CCTL);
+ snd_cs46xx_poke(chip, BA1_CCTL, tmp & 0xffff0000);
+
+ /*
+ * Reset the processor.
+ */
+ snd_cs46xx_reset(chip);
+
+ snd_cs46xx_proc_stop(chip);
+
+ /*
+ * Power down the PLL.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, 0);
+
+ /*
+ * Turn off the Processor by turning off the software clock enable flag in
+ * the clock control register.
+ */
+ tmp = snd_cs46xx_peekBA0(chip, BA0_CLKCR1) & ~CLKCR1_SWCE;
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp);
+}
+
+
+static int snd_cs46xx_free(struct snd_cs46xx *chip)
+{
+ int idx;
+
+ if (snd_BUG_ON(!chip))
+ return -EINVAL;
+
+ if (chip->active_ctrl)
+ chip->active_ctrl(chip, 1);
+
+ snd_cs46xx_remove_gameport(chip);
+
+ if (chip->amplifier_ctrl)
+ chip->amplifier_ctrl(chip, -chip->amplifier); /* force to off */
+
+ snd_cs46xx_proc_done(chip);
+
+ if (chip->region.idx[0].resource)
+ snd_cs46xx_hw_stop(chip);
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+
+ if (chip->active_ctrl)
+ chip->active_ctrl(chip, -chip->amplifier);
+
+ for (idx = 0; idx < 5; idx++) {
+ struct snd_cs46xx_region *region = &chip->region.idx[idx];
+ if (region->remap_addr)
+ iounmap(region->remap_addr);
+ release_and_free_resource(region->resource);
+ }
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (chip->dsp_spos_instance) {
+ cs46xx_dsp_spos_destroy(chip);
+ chip->dsp_spos_instance = NULL;
+ }
+#endif
+
+#ifdef CONFIG_PM
+ kfree(chip->saved_regs);
+#endif
+
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_cs46xx_dev_free(struct snd_device *device)
+{
+ struct snd_cs46xx *chip = device->device_data;
+ return snd_cs46xx_free(chip);
+}
+
+/*
+ * initialize chip
+ */
+static int snd_cs46xx_chip_init(struct snd_cs46xx *chip)
+{
+ int timeout;
+
+ /*
+ * First, blast the clock control register to zero so that the PLL starts
+ * out in a known state, and blast the master serial port control register
+ * to zero so that the serial ports also start out in a known state.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, 0);
+ snd_cs46xx_pokeBA0(chip, BA0_SERMC1, 0);
+
+ /*
+ * If we are in AC97 mode, then we must set the part to a host controlled
+ * AC-link. Otherwise, we won't be able to bring up the link.
+ */
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_cs46xx_pokeBA0(chip, BA0_SERACC, SERACC_HSP | SERACC_CHIP_TYPE_2_0 |
+ SERACC_TWO_CODECS); /* 2.00 dual codecs */
+ /* snd_cs46xx_pokeBA0(chip, BA0_SERACC, SERACC_HSP | SERACC_CHIP_TYPE_2_0); */ /* 2.00 codec */
+#else
+ snd_cs46xx_pokeBA0(chip, BA0_SERACC, SERACC_HSP | SERACC_CHIP_TYPE_1_03); /* 1.03 codec */
+#endif
+
+ /*
+ * Drive the ARST# pin low for a minimum of 1uS (as defined in the AC97
+ * spec) and then drive it high. This is done for non AC97 modes since
+ * there might be logic external to the CS461x that uses the ARST# line
+ * for a reset.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, 0);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL2, 0);
+#endif
+ udelay(50);
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_RSTN);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL2, ACCTL_RSTN);
+#endif
+
+ /*
+ * The first thing we do here is to enable sync generation. As soon
+ * as we start receiving bit clock, we'll start producing the SYNC
+ * signal.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_ESYN | ACCTL_RSTN);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL2, ACCTL_ESYN | ACCTL_RSTN);
+#endif
+
+ /*
+ * Now wait for a short while to allow the AC97 part to start
+ * generating bit clock (so we don't try to start the PLL without an
+ * input clock).
+ */
+ mdelay(10);
+
+ /*
+ * Set the serial port timing configuration, so that
+ * the clock control circuit gets its clock from the correct place.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERMC1, SERMC1_PTC_AC97);
+
+ /*
+ * Write the selected clock control setup to the hardware. Do not turn on
+ * SWCE yet (if requested), so that the devices clocked by the output of
+ * PLL are not clocked until the PLL is stable.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_PLLCC, PLLCC_LPF_1050_2780_KHZ | PLLCC_CDR_73_104_MHZ);
+ snd_cs46xx_pokeBA0(chip, BA0_PLLM, 0x3a);
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR2, CLKCR2_PDIVS_8);
+
+ /*
+ * Power up the PLL.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, CLKCR1_PLLP);
+
+ /*
+ * Wait until the PLL has stabilized.
+ */
+ msleep(100);
+
+ /*
+ * Turn on clocking of the core so that we can setup the serial ports.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, CLKCR1_PLLP | CLKCR1_SWCE);
+
+ /*
+ * Enable FIFO Host Bypass
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERBCF, SERBCF_HBP);
+
+ /*
+ * Fill the serial port FIFOs with silence.
+ */
+ snd_cs46xx_clear_serial_FIFOs(chip);
+
+ /*
+ * Set the serial port FIFO pointer to the first sample in the FIFO.
+ */
+ /* snd_cs46xx_pokeBA0(chip, BA0_SERBSP, 0); */
+
+ /*
+ * Write the serial port configuration to the part. The master
+ * enable bit is not set until all other values have been written.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERC1, SERC1_SO1F_AC97 | SERC1_SO1EN);
+ snd_cs46xx_pokeBA0(chip, BA0_SERC2, SERC2_SI1F_AC97 | SERC1_SO1EN);
+ snd_cs46xx_pokeBA0(chip, BA0_SERMC1, SERMC1_PTC_AC97 | SERMC1_MSPE);
+
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_cs46xx_pokeBA0(chip, BA0_SERC7, SERC7_ASDI2EN);
+ snd_cs46xx_pokeBA0(chip, BA0_SERC3, 0);
+ snd_cs46xx_pokeBA0(chip, BA0_SERC4, 0);
+ snd_cs46xx_pokeBA0(chip, BA0_SERC5, 0);
+ snd_cs46xx_pokeBA0(chip, BA0_SERC6, 1);
+#endif
+
+ mdelay(5);
+
+
+ /*
+ * Wait for the codec ready signal from the AC97 codec.
+ */
+ timeout = 150;
+ while (timeout-- > 0) {
+ /*
+ * Read the AC97 status register to see if we've seen a CODEC READY
+ * signal from the AC97 codec.
+ */
+ if (snd_cs46xx_peekBA0(chip, BA0_ACSTS) & ACSTS_CRDY)
+ goto ok1;
+ msleep(10);
+ }
+
+
+ snd_printk(KERN_ERR "create - never read codec ready from AC'97\n");
+ snd_printk(KERN_ERR "it is not probably bug, try to use CS4236 driver\n");
+ return -EIO;
+ ok1:
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ {
+ int count;
+ for (count = 0; count < 150; count++) {
+ /* First, we want to wait for a short time. */
+ udelay(25);
+
+ if (snd_cs46xx_peekBA0(chip, BA0_ACSTS2) & ACSTS_CRDY)
+ break;
+ }
+
+ /*
+ * Make sure CODEC is READY.
+ */
+ if (!(snd_cs46xx_peekBA0(chip, BA0_ACSTS2) & ACSTS_CRDY))
+ snd_printdd("cs46xx: never read card ready from secondary AC'97\n");
+ }
+#endif
+
+ /*
+ * Assert the vaid frame signal so that we can start sending commands
+ * to the AC97 codec.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL, ACCTL_VFRM | ACCTL_ESYN | ACCTL_RSTN);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_cs46xx_pokeBA0(chip, BA0_ACCTL2, ACCTL_VFRM | ACCTL_ESYN | ACCTL_RSTN);
+#endif
+
+
+ /*
+ * Wait until we've sampled input slots 3 and 4 as valid, meaning that
+ * the codec is pumping ADC data across the AC-link.
+ */
+ timeout = 150;
+ while (timeout-- > 0) {
+ /*
+ * Read the input slot valid register and see if input slots 3 and
+ * 4 are valid yet.
+ */
+ if ((snd_cs46xx_peekBA0(chip, BA0_ACISV) & (ACISV_ISV3 | ACISV_ISV4)) == (ACISV_ISV3 | ACISV_ISV4))
+ goto ok2;
+ msleep(10);
+ }
+
+#ifndef CONFIG_SND_CS46XX_NEW_DSP
+ snd_printk(KERN_ERR "create - never read ISV3 & ISV4 from AC'97\n");
+ return -EIO;
+#else
+ /* This may happen on a cold boot with a Terratec SiXPack 5.1.
+ Reloading the driver may help, if there's other soundcards
+ with the same problem I would like to know. (Benny) */
+
+ snd_printk(KERN_ERR "ERROR: snd-cs46xx: never read ISV3 & ISV4 from AC'97\n");
+ snd_printk(KERN_ERR " Try reloading the ALSA driver, if you find something\n");
+ snd_printk(KERN_ERR " broken or not working on your soundcard upon\n");
+ snd_printk(KERN_ERR " this message please report to alsa-devel@alsa-project.org\n");
+
+ return -EIO;
+#endif
+ ok2:
+
+ /*
+ * Now, assert valid frame and the slot 3 and 4 valid bits. This will
+ * commense the transfer of digital audio data to the AC97 codec.
+ */
+
+ snd_cs46xx_pokeBA0(chip, BA0_ACOSV, ACOSV_SLV3 | ACOSV_SLV4);
+
+
+ /*
+ * Power down the DAC and ADC. We will power them up (if) when we need
+ * them.
+ */
+ /* snd_cs46xx_pokeBA0(chip, BA0_AC97_POWERDOWN, 0x300); */
+
+ /*
+ * Turn off the Processor by turning off the software clock enable flag in
+ * the clock control register.
+ */
+ /* tmp = snd_cs46xx_peekBA0(chip, BA0_CLKCR1) & ~CLKCR1_SWCE; */
+ /* snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp); */
+
+ return 0;
+}
+
+/*
+ * start and load DSP
+ */
+
+static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip)
+{
+ unsigned int tmp;
+
+ snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM);
+
+ tmp = snd_cs46xx_peek(chip, BA1_PFIE);
+ tmp &= ~0x0000f03f;
+ snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */
+
+ tmp = snd_cs46xx_peek(chip, BA1_CIE);
+ tmp &= ~0x0000003f;
+ tmp |= 0x00000001;
+ snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */
+}
+
+int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip)
+{
+ unsigned int tmp;
+ /*
+ * Reset the processor.
+ */
+ snd_cs46xx_reset(chip);
+ /*
+ * Download the image to the processor.
+ */
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+#if 0
+ if (cs46xx_dsp_load_module(chip, &cwcemb80_module) < 0) {
+ snd_printk(KERN_ERR "image download error\n");
+ return -EIO;
+ }
+#endif
+
+ if (cs46xx_dsp_load_module(chip, &cwc4630_module) < 0) {
+ snd_printk(KERN_ERR "image download error [cwc4630]\n");
+ return -EIO;
+ }
+
+ if (cs46xx_dsp_load_module(chip, &cwcasync_module) < 0) {
+ snd_printk(KERN_ERR "image download error [cwcasync]\n");
+ return -EIO;
+ }
+
+ if (cs46xx_dsp_load_module(chip, &cwcsnoop_module) < 0) {
+ snd_printk(KERN_ERR "image download error [cwcsnoop]\n");
+ return -EIO;
+ }
+
+ if (cs46xx_dsp_load_module(chip, &cwcbinhack_module) < 0) {
+ snd_printk(KERN_ERR "image download error [cwcbinhack]\n");
+ return -EIO;
+ }
+
+ if (cs46xx_dsp_load_module(chip, &cwcdma_module) < 0) {
+ snd_printk(KERN_ERR "image download error [cwcdma]\n");
+ return -EIO;
+ }
+
+ if (cs46xx_dsp_scb_and_task_init(chip) < 0)
+ return -EIO;
+#else
+ /* old image */
+ if (snd_cs46xx_download_image(chip) < 0) {
+ snd_printk(KERN_ERR "image download error\n");
+ return -EIO;
+ }
+
+ /*
+ * Stop playback DMA.
+ */
+ tmp = snd_cs46xx_peek(chip, BA1_PCTL);
+ chip->play_ctl = tmp & 0xffff0000;
+ snd_cs46xx_poke(chip, BA1_PCTL, tmp & 0x0000ffff);
+#endif
+
+ /*
+ * Stop capture DMA.
+ */
+ tmp = snd_cs46xx_peek(chip, BA1_CCTL);
+ chip->capt.ctl = tmp & 0x0000ffff;
+ snd_cs46xx_poke(chip, BA1_CCTL, tmp & 0xffff0000);
+
+ mdelay(5);
+
+ snd_cs46xx_set_play_sample_rate(chip, 8000);
+ snd_cs46xx_set_capture_sample_rate(chip, 8000);
+
+ snd_cs46xx_proc_start(chip);
+
+ cs46xx_enable_stream_irqs(chip);
+
+#ifndef CONFIG_SND_CS46XX_NEW_DSP
+ /* set the attenuation to 0dB */
+ snd_cs46xx_poke(chip, BA1_PVOL, 0x80008000);
+ snd_cs46xx_poke(chip, BA1_CVOL, 0x80008000);
+#endif
+
+ return 0;
+}
+
+
+/*
+ * AMP control - null AMP
+ */
+
+static void amp_none(struct snd_cs46xx *chip, int change)
+{
+}
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+static int voyetra_setup_eapd_slot(struct snd_cs46xx *chip)
+{
+
+ u32 idx, valid_slots,tmp,powerdown = 0;
+ u16 modem_power,pin_config,logic_type;
+
+ snd_printdd ("cs46xx: cs46xx_setup_eapd_slot()+\n");
+
+ /*
+ * See if the devices are powered down. If so, we must power them up first
+ * or they will not respond.
+ */
+ tmp = snd_cs46xx_peekBA0(chip, BA0_CLKCR1);
+
+ if (!(tmp & CLKCR1_SWCE)) {
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp | CLKCR1_SWCE);
+ powerdown = 1;
+ }
+
+ /*
+ * Clear PRA. The Bonzo chip will be used for GPIO not for modem
+ * stuff.
+ */
+ if(chip->nr_ac97_codecs != 2) {
+ snd_printk (KERN_ERR "cs46xx: cs46xx_setup_eapd_slot() - no secondary codec configured\n");
+ return -EINVAL;
+ }
+
+ modem_power = snd_cs46xx_codec_read (chip,
+ AC97_EXTENDED_MSTATUS,
+ CS46XX_SECONDARY_CODEC_INDEX);
+ modem_power &=0xFEFF;
+
+ snd_cs46xx_codec_write(chip,
+ AC97_EXTENDED_MSTATUS, modem_power,
+ CS46XX_SECONDARY_CODEC_INDEX);
+
+ /*
+ * Set GPIO pin's 7 and 8 so that they are configured for output.
+ */
+ pin_config = snd_cs46xx_codec_read (chip,
+ AC97_GPIO_CFG,
+ CS46XX_SECONDARY_CODEC_INDEX);
+ pin_config &=0x27F;
+
+ snd_cs46xx_codec_write(chip,
+ AC97_GPIO_CFG, pin_config,
+ CS46XX_SECONDARY_CODEC_INDEX);
+
+ /*
+ * Set GPIO pin's 7 and 8 so that they are compatible with CMOS logic.
+ */
+
+ logic_type = snd_cs46xx_codec_read(chip, AC97_GPIO_POLARITY,
+ CS46XX_SECONDARY_CODEC_INDEX);
+ logic_type &=0x27F;
+
+ snd_cs46xx_codec_write (chip, AC97_GPIO_POLARITY, logic_type,
+ CS46XX_SECONDARY_CODEC_INDEX);
+
+ valid_slots = snd_cs46xx_peekBA0(chip, BA0_ACOSV);
+ valid_slots |= 0x200;
+ snd_cs46xx_pokeBA0(chip, BA0_ACOSV, valid_slots);
+
+ if ( cs46xx_wait_for_fifo(chip,1) ) {
+ snd_printdd("FIFO is busy\n");
+
+ return -EINVAL;
+ }
+
+ /*
+ * Fill slots 12 with the correct value for the GPIO pins.
+ */
+ for(idx = 0x90; idx <= 0x9F; idx++) {
+ /*
+ * Initialize the fifo so that bits 7 and 8 are on.
+ *
+ * Remember that the GPIO pins in bonzo are shifted by 4 bits to
+ * the left. 0x1800 corresponds to bits 7 and 8.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERBWP, 0x1800);
+
+ /*
+ * Wait for command to complete
+ */
+ if ( cs46xx_wait_for_fifo(chip,200) ) {
+ snd_printdd("failed waiting for FIFO at addr (%02X)\n",idx);
+
+ return -EINVAL;
+ }
+
+ /*
+ * Write the serial port FIFO index.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERBAD, idx);
+
+ /*
+ * Tell the serial port to load the new value into the FIFO location.
+ */
+ snd_cs46xx_pokeBA0(chip, BA0_SERBCM, SERBCM_WRC);
+ }
+
+ /* wait for last command to complete */
+ cs46xx_wait_for_fifo(chip,200);
+
+ /*
+ * Now, if we powered up the devices, then power them back down again.
+ * This is kinda ugly, but should never happen.
+ */
+ if (powerdown)
+ snd_cs46xx_pokeBA0(chip, BA0_CLKCR1, tmp);
+
+ return 0;
+}
+#endif
+
+/*
+ * Crystal EAPD mode
+ */
+
+static void amp_voyetra(struct snd_cs46xx *chip, int change)
+{
+ /* Manage the EAPD bit on the Crystal 4297
+ and the Analog AD1885 */
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ int old = chip->amplifier;
+#endif
+ int oval, val;
+
+ chip->amplifier += change;
+ oval = snd_cs46xx_codec_read(chip, AC97_POWERDOWN,
+ CS46XX_PRIMARY_CODEC_INDEX);
+ val = oval;
+ if (chip->amplifier) {
+ /* Turn the EAPD amp on */
+ val |= 0x8000;
+ } else {
+ /* Turn the EAPD amp off */
+ val &= ~0x8000;
+ }
+ if (val != oval) {
+ snd_cs46xx_codec_write(chip, AC97_POWERDOWN, val,
+ CS46XX_PRIMARY_CODEC_INDEX);
+ if (chip->eapd_switch)
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->eapd_switch->id);
+ }
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (chip->amplifier && !old) {
+ voyetra_setup_eapd_slot(chip);
+ }
+#endif
+}
+
+static void hercules_init(struct snd_cs46xx *chip)
+{
+ /* default: AMP off, and SPDIF input optical */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, EGPIODR_GPOE0);
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, EGPIODR_GPOE0);
+}
+
+
+/*
+ * Game Theatre XP card - EGPIO[2] is used to enable the external amp.
+ */
+static void amp_hercules(struct snd_cs46xx *chip, int change)
+{
+ int old = chip->amplifier;
+ int val1 = snd_cs46xx_peekBA0(chip, BA0_EGPIODR);
+ int val2 = snd_cs46xx_peekBA0(chip, BA0_EGPIOPTR);
+
+ chip->amplifier += change;
+ if (chip->amplifier && !old) {
+ snd_printdd ("Hercules amplifier ON\n");
+
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIODR,
+ EGPIODR_GPOE2 | val1); /* enable EGPIO2 output */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR,
+ EGPIOPTR_GPPT2 | val2); /* open-drain on output */
+ } else if (old && !chip->amplifier) {
+ snd_printdd ("Hercules amplifier OFF\n");
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, val1 & ~EGPIODR_GPOE2); /* disable */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, val2 & ~EGPIOPTR_GPPT2); /* disable */
+ }
+}
+
+static void voyetra_mixer_init (struct snd_cs46xx *chip)
+{
+ snd_printdd ("initializing Voyetra mixer\n");
+
+ /* Enable SPDIF out */
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIODR, EGPIODR_GPOE0);
+ snd_cs46xx_pokeBA0(chip, BA0_EGPIOPTR, EGPIODR_GPOE0);
+}
+
+static void hercules_mixer_init (struct snd_cs46xx *chip)
+{
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ unsigned int idx;
+ int err;
+ struct snd_card *card = chip->card;
+#endif
+
+ /* set EGPIO to default */
+ hercules_init(chip);
+
+ snd_printdd ("initializing Hercules mixer\n");
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ if (chip->in_suspend)
+ return;
+
+ for (idx = 0 ; idx < ARRAY_SIZE(snd_hercules_controls); idx++) {
+ struct snd_kcontrol *kctl;
+
+ kctl = snd_ctl_new1(&snd_hercules_controls[idx], chip);
+ if ((err = snd_ctl_add(card, kctl)) < 0) {
+ printk (KERN_ERR "cs46xx: failed to initialize Hercules mixer (%d)\n",err);
+ break;
+ }
+ }
+#endif
+}
+
+
+#if 0
+/*
+ * Untested
+ */
+
+static void amp_voyetra_4294(struct snd_cs46xx *chip, int change)
+{
+ chip->amplifier += change;
+
+ if (chip->amplifier) {
+ /* Switch the GPIO pins 7 and 8 to open drain */
+ snd_cs46xx_codec_write(chip, 0x4C,
+ snd_cs46xx_codec_read(chip, 0x4C) & 0xFE7F);
+ snd_cs46xx_codec_write(chip, 0x4E,
+ snd_cs46xx_codec_read(chip, 0x4E) | 0x0180);
+ /* Now wake the AMP (this might be backwards) */
+ snd_cs46xx_codec_write(chip, 0x54,
+ snd_cs46xx_codec_read(chip, 0x54) & ~0x0180);
+ } else {
+ snd_cs46xx_codec_write(chip, 0x54,
+ snd_cs46xx_codec_read(chip, 0x54) | 0x0180);
+ }
+}
+#endif
+
+
+/*
+ * Handle the CLKRUN on a thinkpad. We must disable CLKRUN support
+ * whenever we need to beat on the chip.
+ *
+ * The original idea and code for this hack comes from David Kaiser at
+ * Linuxcare. Perhaps one day Crystal will document their chips well
+ * enough to make them useful.
+ */
+
+static void clkrun_hack(struct snd_cs46xx *chip, int change)
+{
+ u16 control, nval;
+
+ if (!chip->acpi_port)
+ return;
+
+ chip->amplifier += change;
+
+ /* Read ACPI port */
+ nval = control = inw(chip->acpi_port + 0x10);
+
+ /* Flip CLKRUN off while running */
+ if (! chip->amplifier)
+ nval |= 0x2000;
+ else
+ nval &= ~0x2000;
+ if (nval != control)
+ outw(nval, chip->acpi_port + 0x10);
+}
+
+
+/*
+ * detect intel piix4
+ */
+static void clkrun_init(struct snd_cs46xx *chip)
+{
+ struct pci_dev *pdev;
+ u8 pp;
+
+ chip->acpi_port = 0;
+
+ pdev = pci_get_device(PCI_VENDOR_ID_INTEL,
+ PCI_DEVICE_ID_INTEL_82371AB_3, NULL);
+ if (pdev == NULL)
+ return; /* Not a thinkpad thats for sure */
+
+ /* Find the control port */
+ pci_read_config_byte(pdev, 0x41, &pp);
+ chip->acpi_port = pp << 8;
+ pci_dev_put(pdev);
+}
+
+
+/*
+ * Card subid table
+ */
+
+struct cs_card_type
+{
+ u16 vendor;
+ u16 id;
+ char *name;
+ void (*init)(struct snd_cs46xx *);
+ void (*amp)(struct snd_cs46xx *, int);
+ void (*active)(struct snd_cs46xx *, int);
+ void (*mixer_init)(struct snd_cs46xx *);
+};
+
+static struct cs_card_type __devinitdata cards[] = {
+ {
+ .vendor = 0x1489,
+ .id = 0x7001,
+ .name = "Genius Soundmaker 128 value",
+ /* nothing special */
+ },
+ {
+ .vendor = 0x5053,
+ .id = 0x3357,
+ .name = "Voyetra",
+ .amp = amp_voyetra,
+ .mixer_init = voyetra_mixer_init,
+ },
+ {
+ .vendor = 0x1071,
+ .id = 0x6003,
+ .name = "Mitac MI6020/21",
+ .amp = amp_voyetra,
+ },
+ /* Hercules Game Theatre XP */
+ {
+ .vendor = 0x14af, /* Guillemot Corporation */
+ .id = 0x0050,
+ .name = "Hercules Game Theatre XP",
+ .amp = amp_hercules,
+ .mixer_init = hercules_mixer_init,
+ },
+ {
+ .vendor = 0x1681,
+ .id = 0x0050,
+ .name = "Hercules Game Theatre XP",
+ .amp = amp_hercules,
+ .mixer_init = hercules_mixer_init,
+ },
+ {
+ .vendor = 0x1681,
+ .id = 0x0051,
+ .name = "Hercules Game Theatre XP",
+ .amp = amp_hercules,
+ .mixer_init = hercules_mixer_init,
+
+ },
+ {
+ .vendor = 0x1681,
+ .id = 0x0052,
+ .name = "Hercules Game Theatre XP",
+ .amp = amp_hercules,
+ .mixer_init = hercules_mixer_init,
+ },
+ {
+ .vendor = 0x1681,
+ .id = 0x0053,
+ .name = "Hercules Game Theatre XP",
+ .amp = amp_hercules,
+ .mixer_init = hercules_mixer_init,
+ },
+ {
+ .vendor = 0x1681,
+ .id = 0x0054,
+ .name = "Hercules Game Theatre XP",
+ .amp = amp_hercules,
+ .mixer_init = hercules_mixer_init,
+ },
+ /* Herculess Fortissimo */
+ {
+ .vendor = 0x1681,
+ .id = 0xa010,
+ .name = "Hercules Gamesurround Fortissimo II",
+ },
+ {
+ .vendor = 0x1681,
+ .id = 0xa011,
+ .name = "Hercules Gamesurround Fortissimo III 7.1",
+ },
+ /* Teratec */
+ {
+ .vendor = 0x153b,
+ .id = 0x112e,
+ .name = "Terratec DMX XFire 1024",
+ },
+ {
+ .vendor = 0x153b,
+ .id = 0x1136,
+ .name = "Terratec SiXPack 5.1",
+ },
+ /* Not sure if the 570 needs the clkrun hack */
+ {
+ .vendor = PCI_VENDOR_ID_IBM,
+ .id = 0x0132,
+ .name = "Thinkpad 570",
+ .init = clkrun_init,
+ .active = clkrun_hack,
+ },
+ {
+ .vendor = PCI_VENDOR_ID_IBM,
+ .id = 0x0153,
+ .name = "Thinkpad 600X/A20/T20",
+ .init = clkrun_init,
+ .active = clkrun_hack,
+ },
+ {
+ .vendor = PCI_VENDOR_ID_IBM,
+ .id = 0x1010,
+ .name = "Thinkpad 600E (unsupported)",
+ },
+ {} /* terminator */
+};
+
+
+/*
+ * APM support
+ */
+#ifdef CONFIG_PM
+static unsigned int saved_regs[] = {
+ BA0_ACOSV,
+ BA0_ASER_FADDR,
+ BA0_ASER_MASTER,
+ BA1_PVOL,
+ BA1_CVOL,
+};
+
+int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_cs46xx *chip = card->private_data;
+ int i, amp_saved;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ chip->in_suspend = 1;
+ snd_pcm_suspend_all(chip->pcm);
+ // chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL);
+ // chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE);
+
+ snd_ac97_suspend(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]);
+ snd_ac97_suspend(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]);
+
+ /* save some registers */
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ chip->saved_regs[i] = snd_cs46xx_peekBA0(chip, saved_regs[i]);
+
+ amp_saved = chip->amplifier;
+ /* turn off amp */
+ chip->amplifier_ctrl(chip, -chip->amplifier);
+ snd_cs46xx_hw_stop(chip);
+ /* disable CLKRUN */
+ chip->active_ctrl(chip, -chip->amplifier);
+ chip->amplifier = amp_saved; /* restore the status */
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+int snd_cs46xx_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_cs46xx *chip = card->private_data;
+ int i, amp_saved;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "cs46xx: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ amp_saved = chip->amplifier;
+ chip->amplifier = 0;
+ chip->active_ctrl(chip, 1); /* force to on */
+
+ snd_cs46xx_chip_init(chip);
+
+ snd_cs46xx_reset(chip);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ cs46xx_dsp_resume(chip);
+ /* restore some registers */
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ snd_cs46xx_pokeBA0(chip, saved_regs[i], chip->saved_regs[i]);
+#else
+ snd_cs46xx_download_image(chip);
+#endif
+
+#if 0
+ snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE,
+ chip->ac97_general_purpose);
+ snd_cs46xx_codec_write(chip, AC97_POWER_CONTROL,
+ chip->ac97_powerdown);
+ mdelay(10);
+ snd_cs46xx_codec_write(chip, BA0_AC97_POWERDOWN,
+ chip->ac97_powerdown);
+ mdelay(5);
+#endif
+
+ snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]);
+ snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]);
+
+ /* reset playback/capture */
+ snd_cs46xx_set_play_sample_rate(chip, 8000);
+ snd_cs46xx_set_capture_sample_rate(chip, 8000);
+ snd_cs46xx_proc_start(chip);
+
+ cs46xx_enable_stream_irqs(chip);
+
+ if (amp_saved)
+ chip->amplifier_ctrl(chip, 1); /* turn amp on */
+ else
+ chip->active_ctrl(chip, -1); /* disable CLKRUN */
+ chip->amplifier = amp_saved;
+ chip->in_suspend = 0;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+/*
+ */
+
+int __devinit snd_cs46xx_create(struct snd_card *card,
+ struct pci_dev * pci,
+ int external_amp, int thinkpad,
+ struct snd_cs46xx ** rchip)
+{
+ struct snd_cs46xx *chip;
+ int err, idx;
+ struct snd_cs46xx_region *region;
+ struct cs_card_type *cp;
+ u16 ss_card, ss_vendor;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_cs46xx_dev_free,
+ };
+
+ *rchip = NULL;
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&chip->reg_lock);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ mutex_init(&chip->spos_mutex);
+#endif
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->ba0_addr = pci_resource_start(pci, 0);
+ chip->ba1_addr = pci_resource_start(pci, 1);
+ if (chip->ba0_addr == 0 || chip->ba0_addr == (unsigned long)~0 ||
+ chip->ba1_addr == 0 || chip->ba1_addr == (unsigned long)~0) {
+ snd_printk(KERN_ERR "wrong address(es) - ba0 = 0x%lx, ba1 = 0x%lx\n",
+ chip->ba0_addr, chip->ba1_addr);
+ snd_cs46xx_free(chip);
+ return -ENOMEM;
+ }
+
+ region = &chip->region.name.ba0;
+ strcpy(region->name, "CS46xx_BA0");
+ region->base = chip->ba0_addr;
+ region->size = CS46XX_BA0_SIZE;
+
+ region = &chip->region.name.data0;
+ strcpy(region->name, "CS46xx_BA1_data0");
+ region->base = chip->ba1_addr + BA1_SP_DMEM0;
+ region->size = CS46XX_BA1_DATA0_SIZE;
+
+ region = &chip->region.name.data1;
+ strcpy(region->name, "CS46xx_BA1_data1");
+ region->base = chip->ba1_addr + BA1_SP_DMEM1;
+ region->size = CS46XX_BA1_DATA1_SIZE;
+
+ region = &chip->region.name.pmem;
+ strcpy(region->name, "CS46xx_BA1_pmem");
+ region->base = chip->ba1_addr + BA1_SP_PMEM;
+ region->size = CS46XX_BA1_PRG_SIZE;
+
+ region = &chip->region.name.reg;
+ strcpy(region->name, "CS46xx_BA1_reg");
+ region->base = chip->ba1_addr + BA1_SP_REG;
+ region->size = CS46XX_BA1_REG_SIZE;
+
+ /* set up amp and clkrun hack */
+ pci_read_config_word(pci, PCI_SUBSYSTEM_VENDOR_ID, &ss_vendor);
+ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &ss_card);
+
+ for (cp = &cards[0]; cp->name; cp++) {
+ if (cp->vendor == ss_vendor && cp->id == ss_card) {
+ snd_printdd ("hack for %s enabled\n", cp->name);
+
+ chip->amplifier_ctrl = cp->amp;
+ chip->active_ctrl = cp->active;
+ chip->mixer_init = cp->mixer_init;
+
+ if (cp->init)
+ cp->init(chip);
+ break;
+ }
+ }
+
+ if (external_amp) {
+ snd_printk(KERN_INFO "Crystal EAPD support forced on.\n");
+ chip->amplifier_ctrl = amp_voyetra;
+ }
+
+ if (thinkpad) {
+ snd_printk(KERN_INFO "Activating CLKRUN hack for Thinkpad.\n");
+ chip->active_ctrl = clkrun_hack;
+ clkrun_init(chip);
+ }
+
+ if (chip->amplifier_ctrl == NULL)
+ chip->amplifier_ctrl = amp_none;
+ if (chip->active_ctrl == NULL)
+ chip->active_ctrl = amp_none;
+
+ chip->active_ctrl(chip, 1); /* enable CLKRUN */
+
+ pci_set_master(pci);
+
+ for (idx = 0; idx < 5; idx++) {
+ region = &chip->region.idx[idx];
+ if ((region->resource = request_mem_region(region->base, region->size,
+ region->name)) == NULL) {
+ snd_printk(KERN_ERR "unable to request memory region 0x%lx-0x%lx\n",
+ region->base, region->base + region->size - 1);
+ snd_cs46xx_free(chip);
+ return -EBUSY;
+ }
+ region->remap_addr = ioremap_nocache(region->base, region->size);
+ if (region->remap_addr == NULL) {
+ snd_printk(KERN_ERR "%s ioremap problem\n", region->name);
+ snd_cs46xx_free(chip);
+ return -ENOMEM;
+ }
+ }
+
+ if (request_irq(pci->irq, snd_cs46xx_interrupt, IRQF_SHARED,
+ "CS46XX", chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_cs46xx_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ chip->dsp_spos_instance = cs46xx_dsp_spos_create(chip);
+ if (chip->dsp_spos_instance == NULL) {
+ snd_cs46xx_free(chip);
+ return -ENOMEM;
+ }
+#endif
+
+ err = snd_cs46xx_chip_init(chip);
+ if (err < 0) {
+ snd_cs46xx_free(chip);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_cs46xx_free(chip);
+ return err;
+ }
+
+ snd_cs46xx_proc_init(card, chip);
+
+#ifdef CONFIG_PM
+ chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) *
+ ARRAY_SIZE(saved_regs), GFP_KERNEL);
+ if (!chip->saved_regs) {
+ snd_cs46xx_free(chip);
+ return -ENOMEM;
+ }
+#endif
+
+ chip->active_ctrl(chip, -1); /* disable CLKRUN */
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+}
diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h
new file mode 100644
index 0000000..018a7de
--- /dev/null
+++ b/sound/pci/cs46xx/cs46xx_lib.h
@@ -0,0 +1,206 @@
+/*
+ * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifndef __CS46XX_LIB_H__
+#define __CS46XX_LIB_H__
+
+/*
+ * constants
+ */
+
+#define CS46XX_BA0_SIZE 0x1000
+#define CS46XX_BA1_DATA0_SIZE 0x3000
+#define CS46XX_BA1_DATA1_SIZE 0x3800
+#define CS46XX_BA1_PRG_SIZE 0x7000
+#define CS46XX_BA1_REG_SIZE 0x0100
+
+
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+#define CS46XX_MIN_PERIOD_SIZE 1
+#define CS46XX_MAX_PERIOD_SIZE 1024*1024
+#else
+#define CS46XX_MIN_PERIOD_SIZE 2048
+#define CS46XX_MAX_PERIOD_SIZE 2048
+#endif
+
+#define CS46XX_FRAGS 2
+/* #define CS46XX_BUFFER_SIZE CS46XX_MAX_PERIOD_SIZE * CS46XX_FRAGS */
+
+#define SCB_NO_PARENT 0
+#define SCB_ON_PARENT_NEXT_SCB 1
+#define SCB_ON_PARENT_SUBLIST_SCB 2
+
+/* 3*1024 parameter, 3.5*1024 sample, 2*3.5*1024 code */
+#define BA1_DWORD_SIZE (13 * 1024 + 512)
+#define BA1_MEMORY_COUNT 3
+
+/*
+ * common I/O routines
+ */
+
+static inline void snd_cs46xx_poke(struct snd_cs46xx *chip, unsigned long reg, unsigned int val)
+{
+ unsigned int bank = reg >> 16;
+ unsigned int offset = reg & 0xffff;
+
+ /*if (bank == 0) printk("snd_cs46xx_poke: %04X - %08X\n",reg >> 2,val); */
+ writel(val, chip->region.idx[bank+1].remap_addr + offset);
+}
+
+static inline unsigned int snd_cs46xx_peek(struct snd_cs46xx *chip, unsigned long reg)
+{
+ unsigned int bank = reg >> 16;
+ unsigned int offset = reg & 0xffff;
+ return readl(chip->region.idx[bank+1].remap_addr + offset);
+}
+
+static inline void snd_cs46xx_pokeBA0(struct snd_cs46xx *chip, unsigned long offset, unsigned int val)
+{
+ writel(val, chip->region.name.ba0.remap_addr + offset);
+}
+
+static inline unsigned int snd_cs46xx_peekBA0(struct snd_cs46xx *chip, unsigned long offset)
+{
+ return readl(chip->region.name.ba0.remap_addr + offset);
+}
+
+struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip);
+void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip);
+int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module);
+#ifdef CONFIG_PM
+int cs46xx_dsp_resume(struct snd_cs46xx * chip);
+#endif
+struct dsp_symbol_entry *cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name,
+ int symbol_type);
+#ifdef CONFIG_PROC_FS
+int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip);
+int cs46xx_dsp_proc_done (struct snd_cs46xx *chip);
+#else
+#define cs46xx_dsp_proc_init(card, chip)
+#define cs46xx_dsp_proc_done(chip)
+#endif
+int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip);
+int snd_cs46xx_download (struct snd_cs46xx *chip, u32 *src, unsigned long offset,
+ unsigned long len);
+int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip, unsigned long offset, unsigned long len);
+int cs46xx_dsp_enable_spdif_out (struct snd_cs46xx *chip);
+int cs46xx_dsp_enable_spdif_hw (struct snd_cs46xx *chip);
+int cs46xx_dsp_disable_spdif_out (struct snd_cs46xx *chip);
+int cs46xx_dsp_enable_spdif_in (struct snd_cs46xx *chip);
+int cs46xx_dsp_disable_spdif_in (struct snd_cs46xx *chip);
+int cs46xx_dsp_enable_pcm_capture (struct snd_cs46xx *chip);
+int cs46xx_dsp_disable_pcm_capture (struct snd_cs46xx *chip);
+int cs46xx_dsp_enable_adc_capture (struct snd_cs46xx *chip);
+int cs46xx_dsp_disable_adc_capture (struct snd_cs46xx *chip);
+int cs46xx_poke_via_dsp (struct snd_cs46xx *chip, u32 address, u32 data);
+struct dsp_scb_descriptor * cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name,
+ u32 * scb_data, u32 dest);
+#ifdef CONFIG_PROC_FS
+void cs46xx_dsp_proc_free_scb_desc (struct dsp_scb_descriptor * scb);
+void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip,
+ struct dsp_scb_descriptor * scb);
+#else
+#define cs46xx_dsp_proc_free_scb_desc(scb)
+#define cs46xx_dsp_proc_register_scb_desc(chip, scb)
+#endif
+struct dsp_scb_descriptor * cs46xx_dsp_create_timing_master_scb (struct snd_cs46xx *chip);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_codec_out_scb(struct snd_cs46xx * chip,
+ char * codec_name, u16 channel_disp, u16 fifo_addr,
+ u16 child_scb_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx * chip, char * codec_name,
+ u16 channel_disp, u16 fifo_addr,
+ u16 sample_buffer_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip,
+ struct dsp_scb_descriptor * scb);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx * chip, char * codec_name,
+ u16 channel_disp, u16 fifo_addr,
+ u16 sample_buffer_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_src_task_scb(struct snd_cs46xx * chip, char * scb_name,
+ int sample_rate, u16 src_buffer_addr,
+ u16 src_delay_buffer_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type, int pass_through);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_mix_only_scb(struct snd_cs46xx * chip, char * scb_name,
+ u16 mix_buffer_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_vari_decimate_scb(struct snd_cs46xx * chip, char * scb_name,
+ u16 vari_buffer_addr0, u16 vari_buffer_addr1, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_asynch_fg_rx_scb(struct snd_cs46xx * chip, char * scb_name,
+ u32 dest, u16 hfg_scb_address, u16 asynch_buffer_address,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_spio_write_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_mix_to_ostream_scb(struct snd_cs46xx * chip, char * scb_name,
+ u16 mix_buffer_addr, u16 writeback_spb, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_magic_snoop_scb(struct snd_cs46xx * chip, char * scb_name,
+ u32 dest, u16 snoop_buffer_address,
+ struct dsp_scb_descriptor * snoop_scb,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type);
+struct dsp_pcm_channel_descriptor *
+cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip, u32 sample_rate,
+ void * private_data, u32 hw_dma_addr,
+ int pcm_channel_id);
+void cs46xx_dsp_destroy_pcm_channel (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel);
+int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel);
+int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel);
+struct dsp_scb_descriptor *
+cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * source,
+ u16 addr, char * scb_name);
+int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src);
+int cs46xx_src_link(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src);
+int cs46xx_iec958_pre_open (struct snd_cs46xx *chip);
+int cs46xx_iec958_post_close (struct snd_cs46xx *chip);
+int cs46xx_dsp_pcm_channel_set_period (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel,
+ int period_size);
+int cs46xx_dsp_pcm_ostream_set_period (struct snd_cs46xx * chip, int period_size);
+int cs46xx_dsp_set_dac_volume (struct snd_cs46xx * chip, u16 left, u16 right);
+int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right);
+#endif /* __CS46XX_LIB_H__ */
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
new file mode 100644
index 0000000..f4f0c8f
--- /dev/null
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -0,0 +1,1994 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * 2002-07 Benny Sjostrand benny@hostmobility.com
+ */
+
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+#include <sound/cs46xx.h>
+
+#include "cs46xx_lib.h"
+#include "dsp_spos.h"
+
+static int cs46xx_dsp_async_init (struct snd_cs46xx *chip,
+ struct dsp_scb_descriptor * fg_entry);
+
+static enum wide_opcode wide_opcodes[] = {
+ WIDE_FOR_BEGIN_LOOP,
+ WIDE_FOR_BEGIN_LOOP2,
+ WIDE_COND_GOTO_ADDR,
+ WIDE_COND_GOTO_CALL,
+ WIDE_TBEQ_COND_GOTO_ADDR,
+ WIDE_TBEQ_COND_CALL_ADDR,
+ WIDE_TBEQ_NCOND_GOTO_ADDR,
+ WIDE_TBEQ_NCOND_CALL_ADDR,
+ WIDE_TBEQ_COND_GOTO1_ADDR,
+ WIDE_TBEQ_COND_CALL1_ADDR,
+ WIDE_TBEQ_NCOND_GOTOI_ADDR,
+ WIDE_TBEQ_NCOND_CALL1_ADDR
+};
+
+static int shadow_and_reallocate_code (struct snd_cs46xx * chip, u32 * data, u32 size,
+ u32 overlay_begin_address)
+{
+ unsigned int i = 0, j, nreallocated = 0;
+ u32 hival,loval,address;
+ u32 mop_operands,mop_type,wide_op;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(size %2))
+ return -EINVAL;
+
+ while (i < size) {
+ loval = data[i++];
+ hival = data[i++];
+
+ if (ins->code.offset > 0) {
+ mop_operands = (hival >> 6) & 0x03fff;
+ mop_type = mop_operands >> 10;
+
+ /* check for wide type instruction */
+ if (mop_type == 0 &&
+ (mop_operands & WIDE_LADD_INSTR_MASK) == 0 &&
+ (mop_operands & WIDE_INSTR_MASK) != 0) {
+ wide_op = loval & 0x7f;
+ for (j = 0;j < ARRAY_SIZE(wide_opcodes); ++j) {
+ if (wide_opcodes[j] == wide_op) {
+ /* need to reallocate instruction */
+ address = (hival & 0x00FFF) << 5;
+ address |= loval >> 15;
+
+ snd_printdd("handle_wideop[1]: %05x:%05x addr %04x\n",hival,loval,address);
+
+ if ( !(address & 0x8000) ) {
+ address += (ins->code.offset / 2) - overlay_begin_address;
+ } else {
+ snd_printdd("handle_wideop[1]: ROM symbol not reallocated\n");
+ }
+
+ hival &= 0xFF000;
+ loval &= 0x07FFF;
+
+ hival |= ( (address >> 5) & 0x00FFF);
+ loval |= ( (address << 15) & 0xF8000);
+
+ address = (hival & 0x00FFF) << 5;
+ address |= loval >> 15;
+
+ snd_printdd("handle_wideop:[2] %05x:%05x addr %04x\n",hival,loval,address);
+ nreallocated ++;
+ } /* wide_opcodes[j] == wide_op */
+ } /* for */
+ } /* mod_type == 0 ... */
+ } /* ins->code.offset > 0 */
+
+ ins->code.data[ins->code.size++] = loval;
+ ins->code.data[ins->code.size++] = hival;
+ }
+
+ snd_printdd("dsp_spos: %d instructions reallocated\n",nreallocated);
+ return nreallocated;
+}
+
+static struct dsp_segment_desc * get_segment_desc (struct dsp_module_desc * module, int seg_type)
+{
+ int i;
+ for (i = 0;i < module->nsegments; ++i) {
+ if (module->segments[i].segment_type == seg_type) {
+ return (module->segments + i);
+ }
+ }
+
+ return NULL;
+};
+
+static int find_free_symbol_index (struct dsp_spos_instance * ins)
+{
+ int index = ins->symbol_table.nsymbols,i;
+
+ for (i = ins->symbol_table.highest_frag_index; i < ins->symbol_table.nsymbols; ++i) {
+ if (ins->symbol_table.symbols[i].deleted) {
+ index = i;
+ break;
+ }
+ }
+
+ return index;
+}
+
+static int add_symbols (struct snd_cs46xx * chip, struct dsp_module_desc * module)
+{
+ int i;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (module->symbol_table.nsymbols > 0) {
+ if (!strcmp(module->symbol_table.symbols[0].symbol_name, "OVERLAYBEGINADDRESS") &&
+ module->symbol_table.symbols[0].symbol_type == SYMBOL_CONSTANT ) {
+ module->overlay_begin_address = module->symbol_table.symbols[0].address;
+ }
+ }
+
+ for (i = 0;i < module->symbol_table.nsymbols; ++i) {
+ if (ins->symbol_table.nsymbols == (DSP_MAX_SYMBOLS - 1)) {
+ snd_printk(KERN_ERR "dsp_spos: symbol table is full\n");
+ return -ENOMEM;
+ }
+
+
+ if (cs46xx_dsp_lookup_symbol(chip,
+ module->symbol_table.symbols[i].symbol_name,
+ module->symbol_table.symbols[i].symbol_type) == NULL) {
+
+ ins->symbol_table.symbols[ins->symbol_table.nsymbols] = module->symbol_table.symbols[i];
+ ins->symbol_table.symbols[ins->symbol_table.nsymbols].address += ((ins->code.offset / 2) - module->overlay_begin_address);
+ ins->symbol_table.symbols[ins->symbol_table.nsymbols].module = module;
+ ins->symbol_table.symbols[ins->symbol_table.nsymbols].deleted = 0;
+
+ if (ins->symbol_table.nsymbols > ins->symbol_table.highest_frag_index)
+ ins->symbol_table.highest_frag_index = ins->symbol_table.nsymbols;
+
+ ins->symbol_table.nsymbols++;
+ } else {
+ /* if (0) printk ("dsp_spos: symbol <%s> duplicated, probably nothing wrong with that (Cirrus?)\n",
+ module->symbol_table.symbols[i].symbol_name); */
+ }
+ }
+
+ return 0;
+}
+
+static struct dsp_symbol_entry *
+add_symbol (struct snd_cs46xx * chip, char * symbol_name, u32 address, int type)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_symbol_entry * symbol = NULL;
+ int index;
+
+ if (ins->symbol_table.nsymbols == (DSP_MAX_SYMBOLS - 1)) {
+ snd_printk(KERN_ERR "dsp_spos: symbol table is full\n");
+ return NULL;
+ }
+
+ if (cs46xx_dsp_lookup_symbol(chip,
+ symbol_name,
+ type) != NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol <%s> duplicated\n", symbol_name);
+ return NULL;
+ }
+
+ index = find_free_symbol_index (ins);
+
+ strcpy (ins->symbol_table.symbols[index].symbol_name, symbol_name);
+ ins->symbol_table.symbols[index].address = address;
+ ins->symbol_table.symbols[index].symbol_type = type;
+ ins->symbol_table.symbols[index].module = NULL;
+ ins->symbol_table.symbols[index].deleted = 0;
+ symbol = (ins->symbol_table.symbols + index);
+
+ if (index > ins->symbol_table.highest_frag_index)
+ ins->symbol_table.highest_frag_index = index;
+
+ if (index == ins->symbol_table.nsymbols)
+ ins->symbol_table.nsymbols++; /* no frag. in list */
+
+ return symbol;
+}
+
+struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip)
+{
+ struct dsp_spos_instance * ins = kzalloc(sizeof(struct dsp_spos_instance), GFP_KERNEL);
+
+ if (ins == NULL)
+ return NULL;
+
+ /* better to use vmalloc for this big table */
+ ins->symbol_table.nsymbols = 0;
+ ins->symbol_table.symbols = vmalloc(sizeof(struct dsp_symbol_entry) *
+ DSP_MAX_SYMBOLS);
+ ins->symbol_table.highest_frag_index = 0;
+
+ if (ins->symbol_table.symbols == NULL) {
+ cs46xx_dsp_spos_destroy(chip);
+ goto error;
+ }
+
+ ins->code.offset = 0;
+ ins->code.size = 0;
+ ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL);
+
+ if (ins->code.data == NULL) {
+ cs46xx_dsp_spos_destroy(chip);
+ goto error;
+ }
+
+ ins->nscb = 0;
+ ins->ntask = 0;
+
+ ins->nmodules = 0;
+ ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL);
+
+ if (ins->modules == NULL) {
+ cs46xx_dsp_spos_destroy(chip);
+ goto error;
+ }
+
+ /* default SPDIF input sample rate
+ to 48000 khz */
+ ins->spdif_in_sample_rate = 48000;
+
+ /* maximize volume */
+ ins->dac_volume_right = 0x8000;
+ ins->dac_volume_left = 0x8000;
+ ins->spdif_input_volume_right = 0x8000;
+ ins->spdif_input_volume_left = 0x8000;
+
+ /* set left and right validity bits and
+ default channel status */
+ ins->spdif_csuv_default =
+ ins->spdif_csuv_stream =
+ /* byte 0 */ ((unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF & 0xff)) << 24) |
+ /* byte 1 */ ((unsigned int)_wrap_all_bits( ((SNDRV_PCM_DEFAULT_CON_SPDIF >> 8) & 0xff)) << 16) |
+ /* byte 3 */ (unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF >> 24) & 0xff) |
+ /* left and right validity bits */ (1 << 13) | (1 << 12);
+
+ return ins;
+
+error:
+ kfree(ins);
+ return NULL;
+}
+
+void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip)
+{
+ int i;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(!ins))
+ return;
+
+ mutex_lock(&chip->spos_mutex);
+ for (i = 0; i < ins->nscb; ++i) {
+ if (ins->scbs[i].deleted) continue;
+
+ cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) );
+ }
+
+ kfree(ins->code.data);
+ vfree(ins->symbol_table.symbols);
+ kfree(ins->modules);
+ kfree(ins);
+ mutex_unlock(&chip->spos_mutex);
+}
+
+static int dsp_load_parameter(struct snd_cs46xx *chip,
+ struct dsp_segment_desc *parameter)
+{
+ u32 doffset, dsize;
+
+ if (!parameter) {
+ snd_printdd("dsp_spos: module got no parameter segment\n");
+ return 0;
+ }
+
+ doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET);
+ dsize = parameter->size * 4;
+
+ snd_printdd("dsp_spos: "
+ "downloading parameter data to chip (%08x-%08x)\n",
+ doffset,doffset + dsize);
+ if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) {
+ snd_printk(KERN_ERR "dsp_spos: "
+ "failed to download parameter data to DSP\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int dsp_load_sample(struct snd_cs46xx *chip,
+ struct dsp_segment_desc *sample)
+{
+ u32 doffset, dsize;
+
+ if (!sample) {
+ snd_printdd("dsp_spos: module got no sample segment\n");
+ return 0;
+ }
+
+ doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET);
+ dsize = sample->size * 4;
+
+ snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n",
+ doffset,doffset + dsize);
+
+ if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) {
+ snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_segment_desc * code = get_segment_desc (module,SEGTYPE_SP_PROGRAM);
+ u32 doffset, dsize;
+ int err;
+
+ if (ins->nmodules == DSP_MAX_MODULES - 1) {
+ snd_printk(KERN_ERR "dsp_spos: to many modules loaded into DSP\n");
+ return -ENOMEM;
+ }
+
+ snd_printdd("dsp_spos: loading module %s into DSP\n", module->module_name);
+
+ if (ins->nmodules == 0) {
+ snd_printdd("dsp_spos: clearing parameter area\n");
+ snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, DSP_PARAMETER_BYTE_SIZE);
+ }
+
+ err = dsp_load_parameter(chip, get_segment_desc(module,
+ SEGTYPE_SP_PARAMETER));
+ if (err < 0)
+ return err;
+
+ if (ins->nmodules == 0) {
+ snd_printdd("dsp_spos: clearing sample area\n");
+ snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, DSP_SAMPLE_BYTE_SIZE);
+ }
+
+ err = dsp_load_sample(chip, get_segment_desc(module,
+ SEGTYPE_SP_SAMPLE));
+ if (err < 0)
+ return err;
+
+ if (ins->nmodules == 0) {
+ snd_printdd("dsp_spos: clearing code area\n");
+ snd_cs46xx_clear_BA1(chip, DSP_CODE_BYTE_OFFSET, DSP_CODE_BYTE_SIZE);
+ }
+
+ if (code == NULL) {
+ snd_printdd("dsp_spos: module got no code segment\n");
+ } else {
+ if (ins->code.offset + code->size > DSP_CODE_BYTE_SIZE) {
+ snd_printk(KERN_ERR "dsp_spos: no space available in DSP\n");
+ return -ENOMEM;
+ }
+
+ module->load_address = ins->code.offset;
+ module->overlay_begin_address = 0x000;
+
+ /* if module has a code segment it must have
+ symbol table */
+ if (snd_BUG_ON(!module->symbol_table.symbols))
+ return -ENOMEM;
+ if (add_symbols(chip,module)) {
+ snd_printk(KERN_ERR "dsp_spos: failed to load symbol table\n");
+ return -ENOMEM;
+ }
+
+ doffset = (code->offset * 4 + ins->code.offset * 4 + DSP_CODE_BYTE_OFFSET);
+ dsize = code->size * 4;
+ snd_printdd("dsp_spos: downloading code to chip (%08x-%08x)\n",
+ doffset,doffset + dsize);
+
+ module->nfixups = shadow_and_reallocate_code(chip,code->data,code->size,module->overlay_begin_address);
+
+ if (snd_cs46xx_download (chip,(ins->code.data + ins->code.offset),doffset,dsize)) {
+ snd_printk(KERN_ERR "dsp_spos: failed to download code to DSP\n");
+ return -EINVAL;
+ }
+
+ ins->code.offset += code->size;
+ }
+
+ /* NOTE: module segments and symbol table must be
+ statically allocated. Case that module data is
+ not generated by the ospparser */
+ ins->modules[ins->nmodules] = *module;
+ ins->nmodules++;
+
+ return 0;
+}
+
+struct dsp_symbol_entry *
+cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, int symbol_type)
+{
+ int i;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ for ( i = 0; i < ins->symbol_table.nsymbols; ++i ) {
+
+ if (ins->symbol_table.symbols[i].deleted)
+ continue;
+
+ if (!strcmp(ins->symbol_table.symbols[i].symbol_name,symbol_name) &&
+ ins->symbol_table.symbols[i].symbol_type == symbol_type) {
+ return (ins->symbol_table.symbols + i);
+ }
+ }
+
+#if 0
+ printk ("dsp_spos: symbol <%s> type %02x not found\n",
+ symbol_name,symbol_type);
+#endif
+
+ return NULL;
+}
+
+
+#ifdef CONFIG_PROC_FS
+static struct dsp_symbol_entry *
+cs46xx_dsp_lookup_symbol_addr (struct snd_cs46xx * chip, u32 address, int symbol_type)
+{
+ int i;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ for ( i = 0; i < ins->symbol_table.nsymbols; ++i ) {
+
+ if (ins->symbol_table.symbols[i].deleted)
+ continue;
+
+ if (ins->symbol_table.symbols[i].address == address &&
+ ins->symbol_table.symbols[i].symbol_type == symbol_type) {
+ return (ins->symbol_table.symbols + i);
+ }
+ }
+
+
+ return NULL;
+}
+
+
+static void cs46xx_dsp_proc_symbol_table_read (struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_cs46xx *chip = entry->private_data;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int i;
+
+ snd_iprintf(buffer, "SYMBOLS:\n");
+ for ( i = 0; i < ins->symbol_table.nsymbols; ++i ) {
+ char *module_str = "system";
+
+ if (ins->symbol_table.symbols[i].deleted)
+ continue;
+
+ if (ins->symbol_table.symbols[i].module != NULL) {
+ module_str = ins->symbol_table.symbols[i].module->module_name;
+ }
+
+
+ snd_iprintf(buffer, "%04X <%02X> %s [%s]\n",
+ ins->symbol_table.symbols[i].address,
+ ins->symbol_table.symbols[i].symbol_type,
+ ins->symbol_table.symbols[i].symbol_name,
+ module_str);
+ }
+}
+
+
+static void cs46xx_dsp_proc_modules_read (struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_cs46xx *chip = entry->private_data;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int i,j;
+
+ mutex_lock(&chip->spos_mutex);
+ snd_iprintf(buffer, "MODULES:\n");
+ for ( i = 0; i < ins->nmodules; ++i ) {
+ snd_iprintf(buffer, "\n%s:\n", ins->modules[i].module_name);
+ snd_iprintf(buffer, " %d symbols\n", ins->modules[i].symbol_table.nsymbols);
+ snd_iprintf(buffer, " %d fixups\n", ins->modules[i].nfixups);
+
+ for (j = 0; j < ins->modules[i].nsegments; ++ j) {
+ struct dsp_segment_desc * desc = (ins->modules[i].segments + j);
+ snd_iprintf(buffer, " segment %02x offset %08x size %08x\n",
+ desc->segment_type,desc->offset, desc->size);
+ }
+ }
+ mutex_unlock(&chip->spos_mutex);
+}
+
+static void cs46xx_dsp_proc_task_tree_read (struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_cs46xx *chip = entry->private_data;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int i, j, col;
+ void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET;
+
+ mutex_lock(&chip->spos_mutex);
+ snd_iprintf(buffer, "TASK TREES:\n");
+ for ( i = 0; i < ins->ntask; ++i) {
+ snd_iprintf(buffer,"\n%04x %s:\n",ins->tasks[i].address,ins->tasks[i].task_name);
+
+ for (col = 0,j = 0;j < ins->tasks[i].size; j++,col++) {
+ u32 val;
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+ val = readl(dst + (ins->tasks[i].address + j) * sizeof(u32));
+ snd_iprintf(buffer,"%08x ",val);
+ }
+ }
+
+ snd_iprintf(buffer,"\n");
+ mutex_unlock(&chip->spos_mutex);
+}
+
+static void cs46xx_dsp_proc_scb_read (struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_cs46xx *chip = entry->private_data;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int i;
+
+ mutex_lock(&chip->spos_mutex);
+ snd_iprintf(buffer, "SCB's:\n");
+ for ( i = 0; i < ins->nscb; ++i) {
+ if (ins->scbs[i].deleted)
+ continue;
+ snd_iprintf(buffer,"\n%04x %s:\n\n",ins->scbs[i].address,ins->scbs[i].scb_name);
+
+ if (ins->scbs[i].parent_scb_ptr != NULL) {
+ snd_iprintf(buffer,"parent [%s:%04x] ",
+ ins->scbs[i].parent_scb_ptr->scb_name,
+ ins->scbs[i].parent_scb_ptr->address);
+ } else snd_iprintf(buffer,"parent [none] ");
+
+ snd_iprintf(buffer,"sub_list_ptr [%s:%04x]\nnext_scb_ptr [%s:%04x] task_entry [%s:%04x]\n",
+ ins->scbs[i].sub_list_ptr->scb_name,
+ ins->scbs[i].sub_list_ptr->address,
+ ins->scbs[i].next_scb_ptr->scb_name,
+ ins->scbs[i].next_scb_ptr->address,
+ ins->scbs[i].task_entry->symbol_name,
+ ins->scbs[i].task_entry->address);
+ }
+
+ snd_iprintf(buffer,"\n");
+ mutex_unlock(&chip->spos_mutex);
+}
+
+static void cs46xx_dsp_proc_parameter_dump_read (struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_cs46xx *chip = entry->private_data;
+ /*struct dsp_spos_instance * ins = chip->dsp_spos_instance; */
+ unsigned int i, col = 0;
+ void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET;
+ struct dsp_symbol_entry * symbol;
+
+ for (i = 0;i < DSP_PARAMETER_BYTE_SIZE; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if ( (symbol = cs46xx_dsp_lookup_symbol_addr (chip,i / sizeof(u32), SYMBOL_PARAMETER)) != NULL) {
+ col = 0;
+ snd_iprintf (buffer,"\n%s:\n",symbol->symbol_name);
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ", i / (unsigned int)sizeof(u32));
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+}
+
+static void cs46xx_dsp_proc_sample_dump_read (struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_cs46xx *chip = entry->private_data;
+ int i,col = 0;
+ void __iomem *dst = chip->region.idx[2].remap_addr;
+
+ snd_iprintf(buffer,"PCMREADER:\n");
+ for (i = PCM_READER_BUF1;i < PCM_READER_BUF1 + 0x30; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+
+ snd_iprintf(buffer,"\nMIX_SAMPLE_BUF1:\n");
+
+ col = 0;
+ for (i = MIX_SAMPLE_BUF1;i < MIX_SAMPLE_BUF1 + 0x40; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+
+ snd_iprintf(buffer,"\nSRC_TASK_SCB1:\n");
+ col = 0;
+ for (i = 0x2480 ; i < 0x2480 + 0x40 ; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+
+
+ snd_iprintf(buffer,"\nSPDIFO_BUFFER:\n");
+ col = 0;
+ for (i = SPDIFO_IP_OUTPUT_BUFFER1;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x30; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+
+ snd_iprintf(buffer,"\n...\n");
+ col = 0;
+
+ for (i = SPDIFO_IP_OUTPUT_BUFFER1+0xD0;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x110; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+
+
+ snd_iprintf(buffer,"\nOUTPUT_SNOOP:\n");
+ col = 0;
+ for (i = OUTPUT_SNOOP_BUFFER;i < OUTPUT_SNOOP_BUFFER + 0x40; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+
+ snd_iprintf(buffer,"\nCODEC_INPUT_BUF1: \n");
+ col = 0;
+ for (i = CODEC_INPUT_BUF1;i < CODEC_INPUT_BUF1 + 0x40; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+#if 0
+ snd_iprintf(buffer,"\nWRITE_BACK_BUF1: \n");
+ col = 0;
+ for (i = WRITE_BACK_BUF1;i < WRITE_BACK_BUF1 + 0x40; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+#endif
+
+ snd_iprintf(buffer,"\nSPDIFI_IP_OUTPUT_BUFFER1: \n");
+ col = 0;
+ for (i = SPDIFI_IP_OUTPUT_BUFFER1;i < SPDIFI_IP_OUTPUT_BUFFER1 + 0x80; i += sizeof(u32),col ++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+
+ if (col == 0) {
+ snd_iprintf(buffer, "%04X ",i);
+ }
+
+ snd_iprintf(buffer,"%08X ",readl(dst + i));
+ }
+ snd_iprintf(buffer,"\n");
+}
+
+int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
+{
+ struct snd_info_entry *entry;
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int i;
+
+ ins->snd_card = card;
+
+ if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+
+ ins->proc_dsp_dir = entry;
+
+ if (!ins->proc_dsp_dir)
+ return -ENOMEM;
+
+ if ((entry = snd_info_create_card_entry(card, "spos_symbols", ins->proc_dsp_dir)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = chip;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = cs46xx_dsp_proc_symbol_table_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ins->proc_sym_info_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "spos_modules", ins->proc_dsp_dir)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = chip;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = cs46xx_dsp_proc_modules_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ins->proc_modules_info_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "parameter", ins->proc_dsp_dir)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = chip;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ins->proc_parameter_dump_info_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "sample", ins->proc_dsp_dir)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = chip;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = cs46xx_dsp_proc_sample_dump_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ins->proc_sample_dump_info_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "task_tree", ins->proc_dsp_dir)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = chip;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = cs46xx_dsp_proc_task_tree_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ins->proc_task_info_entry = entry;
+
+ if ((entry = snd_info_create_card_entry(card, "scb_info", ins->proc_dsp_dir)) != NULL) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = chip;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+ entry->c.text.read = cs46xx_dsp_proc_scb_read;
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ }
+ }
+ ins->proc_scb_info_entry = entry;
+
+ mutex_lock(&chip->spos_mutex);
+ /* register/update SCB's entries on proc */
+ for (i = 0; i < ins->nscb; ++i) {
+ if (ins->scbs[i].deleted) continue;
+
+ cs46xx_dsp_proc_register_scb_desc (chip, (ins->scbs + i));
+ }
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+int cs46xx_dsp_proc_done (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int i;
+
+ snd_info_free_entry(ins->proc_sym_info_entry);
+ ins->proc_sym_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_modules_info_entry);
+ ins->proc_modules_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_parameter_dump_info_entry);
+ ins->proc_parameter_dump_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_sample_dump_info_entry);
+ ins->proc_sample_dump_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_scb_info_entry);
+ ins->proc_scb_info_entry = NULL;
+
+ snd_info_free_entry(ins->proc_task_info_entry);
+ ins->proc_task_info_entry = NULL;
+
+ mutex_lock(&chip->spos_mutex);
+ for (i = 0; i < ins->nscb; ++i) {
+ if (ins->scbs[i].deleted) continue;
+ cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) );
+ }
+ mutex_unlock(&chip->spos_mutex);
+
+ snd_info_free_entry(ins->proc_dsp_dir);
+ ins->proc_dsp_dir = NULL;
+
+ return 0;
+}
+#endif /* CONFIG_PROC_FS */
+
+static int debug_tree;
+static void _dsp_create_task_tree (struct snd_cs46xx *chip, u32 * task_data,
+ u32 dest, int size)
+{
+ void __iomem *spdst = chip->region.idx[1].remap_addr +
+ DSP_PARAMETER_BYTE_OFFSET + dest * sizeof(u32);
+ int i;
+
+ for (i = 0; i < size; ++i) {
+ if (debug_tree) printk ("addr %p, val %08x\n",spdst,task_data[i]);
+ writel(task_data[i],spdst);
+ spdst += sizeof(u32);
+ }
+}
+
+static int debug_scb;
+static void _dsp_create_scb (struct snd_cs46xx *chip, u32 * scb_data, u32 dest)
+{
+ void __iomem *spdst = chip->region.idx[1].remap_addr +
+ DSP_PARAMETER_BYTE_OFFSET + dest * sizeof(u32);
+ int i;
+
+ for (i = 0; i < 0x10; ++i) {
+ if (debug_scb) printk ("addr %p, val %08x\n",spdst,scb_data[i]);
+ writel(scb_data[i],spdst);
+ spdst += sizeof(u32);
+ }
+}
+
+static int find_free_scb_index (struct dsp_spos_instance * ins)
+{
+ int index = ins->nscb, i;
+
+ for (i = ins->scb_highest_frag_index; i < ins->nscb; ++i) {
+ if (ins->scbs[i].deleted) {
+ index = i;
+ break;
+ }
+ }
+
+ return index;
+}
+
+static struct dsp_scb_descriptor * _map_scb (struct snd_cs46xx *chip, char * name, u32 dest)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * desc = NULL;
+ int index;
+
+ if (ins->nscb == DSP_MAX_SCB_DESC - 1) {
+ snd_printk(KERN_ERR "dsp_spos: got no place for other SCB\n");
+ return NULL;
+ }
+
+ index = find_free_scb_index (ins);
+
+ strcpy(ins->scbs[index].scb_name, name);
+ ins->scbs[index].address = dest;
+ ins->scbs[index].index = index;
+ ins->scbs[index].proc_info = NULL;
+ ins->scbs[index].ref_count = 1;
+ ins->scbs[index].deleted = 0;
+ spin_lock_init(&ins->scbs[index].lock);
+
+ desc = (ins->scbs + index);
+ ins->scbs[index].scb_symbol = add_symbol (chip, name, dest, SYMBOL_PARAMETER);
+
+ if (index > ins->scb_highest_frag_index)
+ ins->scb_highest_frag_index = index;
+
+ if (index == ins->nscb)
+ ins->nscb++;
+
+ return desc;
+}
+
+static struct dsp_task_descriptor *
+_map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_task_descriptor * desc = NULL;
+
+ if (ins->ntask == DSP_MAX_TASK_DESC - 1) {
+ snd_printk(KERN_ERR "dsp_spos: got no place for other TASK\n");
+ return NULL;
+ }
+
+ if (name)
+ strcpy(ins->tasks[ins->ntask].task_name, name);
+ else
+ strcpy(ins->tasks[ins->ntask].task_name, "(NULL)");
+ ins->tasks[ins->ntask].address = dest;
+ ins->tasks[ins->ntask].size = size;
+
+ /* quick find in list */
+ ins->tasks[ins->ntask].index = ins->ntask;
+ desc = (ins->tasks + ins->ntask);
+ ins->ntask++;
+
+ if (name)
+ add_symbol (chip,name,dest,SYMBOL_PARAMETER);
+ return desc;
+}
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest)
+{
+ struct dsp_scb_descriptor * desc;
+
+ desc = _map_scb (chip,name,dest);
+ if (desc) {
+ desc->data = scb_data;
+ _dsp_create_scb(chip,scb_data,dest);
+ } else {
+ snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n");
+ }
+
+ return desc;
+}
+
+
+static struct dsp_task_descriptor *
+cs46xx_dsp_create_task_tree (struct snd_cs46xx *chip, char * name, u32 * task_data,
+ u32 dest, int size)
+{
+ struct dsp_task_descriptor * desc;
+
+ desc = _map_task_tree (chip,name,dest,size);
+ if (desc) {
+ desc->data = task_data;
+ _dsp_create_task_tree(chip,task_data,dest,size);
+ } else {
+ snd_printk(KERN_ERR "dsp_spos: failed to map TASK\n");
+ }
+
+ return desc;
+}
+
+int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_symbol_entry * fg_task_tree_header_code;
+ struct dsp_symbol_entry * task_tree_header_code;
+ struct dsp_symbol_entry * task_tree_thread;
+ struct dsp_symbol_entry * null_algorithm;
+ struct dsp_symbol_entry * magic_snoop_task;
+
+ struct dsp_scb_descriptor * timing_master_scb;
+ struct dsp_scb_descriptor * codec_out_scb;
+ struct dsp_scb_descriptor * codec_in_scb;
+ struct dsp_scb_descriptor * src_task_scb;
+ struct dsp_scb_descriptor * master_mix_scb;
+ struct dsp_scb_descriptor * rear_mix_scb;
+ struct dsp_scb_descriptor * record_mix_scb;
+ struct dsp_scb_descriptor * write_back_scb;
+ struct dsp_scb_descriptor * vari_decimate_scb;
+ struct dsp_scb_descriptor * rear_codec_out_scb;
+ struct dsp_scb_descriptor * clfe_codec_out_scb;
+ struct dsp_scb_descriptor * magic_snoop_scb;
+
+ int fifo_addr, fifo_span, valid_slots;
+
+ static struct dsp_spos_control_block sposcb = {
+ /* 0 */ HFG_TREE_SCB,HFG_STACK,
+ /* 1 */ SPOSCB_ADDR,BG_TREE_SCB_ADDR,
+ /* 2 */ DSP_SPOS_DC,0,
+ /* 3 */ DSP_SPOS_DC,DSP_SPOS_DC,
+ /* 4 */ 0,0,
+ /* 5 */ DSP_SPOS_UU,0,
+ /* 6 */ FG_TASK_HEADER_ADDR,0,
+ /* 7 */ 0,0,
+ /* 8 */ DSP_SPOS_UU,DSP_SPOS_DC,
+ /* 9 */ 0,
+ /* A */ 0,HFG_FIRST_EXECUTE_MODE,
+ /* B */ DSP_SPOS_UU,DSP_SPOS_UU,
+ /* C */ DSP_SPOS_DC_DC,
+ /* D */ DSP_SPOS_DC_DC,
+ /* E */ DSP_SPOS_DC_DC,
+ /* F */ DSP_SPOS_DC_DC
+ };
+
+ cs46xx_dsp_create_task_tree(chip, "sposCB", (u32 *)&sposcb, SPOSCB_ADDR, 0x10);
+
+ null_algorithm = cs46xx_dsp_lookup_symbol(chip, "NULLALGORITHM", SYMBOL_CODE);
+ if (null_algorithm == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol NULLALGORITHM not found\n");
+ return -EIO;
+ }
+
+ fg_task_tree_header_code = cs46xx_dsp_lookup_symbol(chip, "FGTASKTREEHEADERCODE", SYMBOL_CODE);
+ if (fg_task_tree_header_code == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol FGTASKTREEHEADERCODE not found\n");
+ return -EIO;
+ }
+
+ task_tree_header_code = cs46xx_dsp_lookup_symbol(chip, "TASKTREEHEADERCODE", SYMBOL_CODE);
+ if (task_tree_header_code == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol TASKTREEHEADERCODE not found\n");
+ return -EIO;
+ }
+
+ task_tree_thread = cs46xx_dsp_lookup_symbol(chip, "TASKTREETHREAD", SYMBOL_CODE);
+ if (task_tree_thread == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol TASKTREETHREAD not found\n");
+ return -EIO;
+ }
+
+ magic_snoop_task = cs46xx_dsp_lookup_symbol(chip, "MAGICSNOOPTASK", SYMBOL_CODE);
+ if (magic_snoop_task == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol MAGICSNOOPTASK not found\n");
+ return -EIO;
+ }
+
+ {
+ /* create the null SCB */
+ static struct dsp_generic_scb null_scb = {
+ { 0, 0, 0, 0 },
+ { 0, 0, 0, 0, 0 },
+ NULL_SCB_ADDR, NULL_SCB_ADDR,
+ 0, 0, 0, 0, 0,
+ {
+ 0,0,
+ 0,0,
+ }
+ };
+
+ null_scb.entry_point = null_algorithm->address;
+ ins->the_null_scb = cs46xx_dsp_create_scb(chip, "nullSCB", (u32 *)&null_scb, NULL_SCB_ADDR);
+ ins->the_null_scb->task_entry = null_algorithm;
+ ins->the_null_scb->sub_list_ptr = ins->the_null_scb;
+ ins->the_null_scb->next_scb_ptr = ins->the_null_scb;
+ ins->the_null_scb->parent_scb_ptr = NULL;
+ cs46xx_dsp_proc_register_scb_desc (chip,ins->the_null_scb);
+ }
+
+ {
+ /* setup foreground task tree */
+ static struct dsp_task_tree_control_block fg_task_tree_hdr = {
+ { FG_TASK_HEADER_ADDR | (DSP_SPOS_DC << 0x10),
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC_DC,
+ 0x0000,DSP_SPOS_DC,
+ DSP_SPOS_DC, DSP_SPOS_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC },
+
+ {
+ BG_TREE_SCB_ADDR,TIMINGMASTER_SCB_ADDR,
+ 0,
+ FG_TASK_HEADER_ADDR + TCBData,
+ },
+
+ {
+ 4,0,
+ 1,0,
+ 2,SPOSCB_ADDR + HFGFlags,
+ 0,0,
+ FG_TASK_HEADER_ADDR + TCBContextBlk,FG_STACK
+ },
+
+ {
+ DSP_SPOS_DC,0,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_UU,1,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC
+ },
+ {
+ FG_INTERVAL_TIMER_PERIOD,DSP_SPOS_UU,
+ 0,0
+ }
+ };
+
+ fg_task_tree_hdr.links.entry_point = fg_task_tree_header_code->address;
+ fg_task_tree_hdr.context_blk.stack0 = task_tree_thread->address;
+ cs46xx_dsp_create_task_tree(chip,"FGtaskTreeHdr",(u32 *)&fg_task_tree_hdr,FG_TASK_HEADER_ADDR,0x35);
+ }
+
+
+ {
+ /* setup foreground task tree */
+ static struct dsp_task_tree_control_block bg_task_tree_hdr = {
+ { DSP_SPOS_DC_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC, DSP_SPOS_DC,
+ DSP_SPOS_DC, DSP_SPOS_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC },
+
+ {
+ NULL_SCB_ADDR,NULL_SCB_ADDR, /* Set up the background to do nothing */
+ 0,
+ BG_TREE_SCB_ADDR + TCBData,
+ },
+
+ {
+ 9999,0,
+ 0,1,
+ 0,SPOSCB_ADDR + HFGFlags,
+ 0,0,
+ BG_TREE_SCB_ADDR + TCBContextBlk,BG_STACK
+ },
+
+ {
+ DSP_SPOS_DC,0,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DC,DSP_SPOS_DC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_UU,1,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC,
+ DSP_SPOS_DCDC
+ },
+ {
+ BG_INTERVAL_TIMER_PERIOD,DSP_SPOS_UU,
+ 0,0
+ }
+ };
+
+ bg_task_tree_hdr.links.entry_point = task_tree_header_code->address;
+ bg_task_tree_hdr.context_blk.stack0 = task_tree_thread->address;
+ cs46xx_dsp_create_task_tree(chip,"BGtaskTreeHdr",(u32 *)&bg_task_tree_hdr,BG_TREE_SCB_ADDR,0x35);
+ }
+
+ /* create timing master SCB */
+ timing_master_scb = cs46xx_dsp_create_timing_master_scb(chip);
+
+ /* create the CODEC output task */
+ codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,"CodecOutSCB_I",0x0010,0x0000,
+ MASTERMIX_SCB_ADDR,
+ CODECOUT_SCB_ADDR,timing_master_scb,
+ SCB_ON_PARENT_SUBLIST_SCB);
+
+ if (!codec_out_scb) goto _fail_end;
+ /* create the master mix SCB */
+ master_mix_scb = cs46xx_dsp_create_mix_only_scb(chip,"MasterMixSCB",
+ MIX_SAMPLE_BUF1,MASTERMIX_SCB_ADDR,
+ codec_out_scb,
+ SCB_ON_PARENT_SUBLIST_SCB);
+ ins->master_mix_scb = master_mix_scb;
+
+ if (!master_mix_scb) goto _fail_end;
+
+ /* create codec in */
+ codec_in_scb = cs46xx_dsp_create_codec_in_scb(chip,"CodecInSCB",0x0010,0x00A0,
+ CODEC_INPUT_BUF1,
+ CODECIN_SCB_ADDR,codec_out_scb,
+ SCB_ON_PARENT_NEXT_SCB);
+ if (!codec_in_scb) goto _fail_end;
+ ins->codec_in_scb = codec_in_scb;
+
+ /* create write back scb */
+ write_back_scb = cs46xx_dsp_create_mix_to_ostream_scb(chip,"WriteBackSCB",
+ WRITE_BACK_BUF1,WRITE_BACK_SPB,
+ WRITEBACK_SCB_ADDR,
+ timing_master_scb,
+ SCB_ON_PARENT_NEXT_SCB);
+ if (!write_back_scb) goto _fail_end;
+
+ {
+ static struct dsp_mix2_ostream_spb mix2_ostream_spb = {
+ 0x00020000,
+ 0x0000ffff
+ };
+
+ if (!cs46xx_dsp_create_task_tree(chip, NULL,
+ (u32 *)&mix2_ostream_spb,
+ WRITE_BACK_SPB, 2))
+ goto _fail_end;
+ }
+
+ /* input sample converter */
+ vari_decimate_scb = cs46xx_dsp_create_vari_decimate_scb(chip,"VariDecimateSCB",
+ VARI_DECIMATE_BUF0,
+ VARI_DECIMATE_BUF1,
+ VARIDECIMATE_SCB_ADDR,
+ write_back_scb,
+ SCB_ON_PARENT_SUBLIST_SCB);
+ if (!vari_decimate_scb) goto _fail_end;
+
+ /* create the record mixer SCB */
+ record_mix_scb = cs46xx_dsp_create_mix_only_scb(chip,"RecordMixerSCB",
+ MIX_SAMPLE_BUF2,
+ RECORD_MIXER_SCB_ADDR,
+ vari_decimate_scb,
+ SCB_ON_PARENT_SUBLIST_SCB);
+ ins->record_mixer_scb = record_mix_scb;
+
+ if (!record_mix_scb) goto _fail_end;
+
+ valid_slots = snd_cs46xx_peekBA0(chip, BA0_ACOSV);
+
+ if (snd_BUG_ON(chip->nr_ac97_codecs != 1 && chip->nr_ac97_codecs != 2))
+ goto _fail_end;
+
+ if (chip->nr_ac97_codecs == 1) {
+ /* output on slot 5 and 11
+ on primary CODEC */
+ fifo_addr = 0x20;
+ fifo_span = 0x60;
+
+ /* enable slot 5 and 11 */
+ valid_slots |= ACOSV_SLV5 | ACOSV_SLV11;
+ } else {
+ /* output on slot 7 and 8
+ on secondary CODEC */
+ fifo_addr = 0x40;
+ fifo_span = 0x10;
+
+ /* enable slot 7 and 8 */
+ valid_slots |= ACOSV_SLV7 | ACOSV_SLV8;
+ }
+ /* create CODEC tasklet for rear speakers output*/
+ rear_codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,"CodecOutSCB_Rear",fifo_span,fifo_addr,
+ REAR_MIXER_SCB_ADDR,
+ REAR_CODECOUT_SCB_ADDR,codec_in_scb,
+ SCB_ON_PARENT_NEXT_SCB);
+ if (!rear_codec_out_scb) goto _fail_end;
+
+
+ /* create the rear PCM channel mixer SCB */
+ rear_mix_scb = cs46xx_dsp_create_mix_only_scb(chip,"RearMixerSCB",
+ MIX_SAMPLE_BUF3,
+ REAR_MIXER_SCB_ADDR,
+ rear_codec_out_scb,
+ SCB_ON_PARENT_SUBLIST_SCB);
+ ins->rear_mix_scb = rear_mix_scb;
+ if (!rear_mix_scb) goto _fail_end;
+
+ if (chip->nr_ac97_codecs == 2) {
+ /* create CODEC tasklet for rear Center/LFE output
+ slot 6 and 9 on seconadry CODEC */
+ clfe_codec_out_scb = cs46xx_dsp_create_codec_out_scb(chip,"CodecOutSCB_CLFE",0x0030,0x0030,
+ CLFE_MIXER_SCB_ADDR,
+ CLFE_CODEC_SCB_ADDR,
+ rear_codec_out_scb,
+ SCB_ON_PARENT_NEXT_SCB);
+ if (!clfe_codec_out_scb) goto _fail_end;
+
+
+ /* create the rear PCM channel mixer SCB */
+ ins->center_lfe_mix_scb = cs46xx_dsp_create_mix_only_scb(chip,"CLFEMixerSCB",
+ MIX_SAMPLE_BUF4,
+ CLFE_MIXER_SCB_ADDR,
+ clfe_codec_out_scb,
+ SCB_ON_PARENT_SUBLIST_SCB);
+ if (!ins->center_lfe_mix_scb) goto _fail_end;
+
+ /* enable slot 6 and 9 */
+ valid_slots |= ACOSV_SLV6 | ACOSV_SLV9;
+ } else {
+ clfe_codec_out_scb = rear_codec_out_scb;
+ ins->center_lfe_mix_scb = rear_mix_scb;
+ }
+
+ /* enable slots depending on CODEC configuration */
+ snd_cs46xx_pokeBA0(chip, BA0_ACOSV, valid_slots);
+
+ /* the magic snooper */
+ magic_snoop_scb = cs46xx_dsp_create_magic_snoop_scb (chip,"MagicSnoopSCB_I",OUTPUTSNOOP_SCB_ADDR,
+ OUTPUT_SNOOP_BUFFER,
+ codec_out_scb,
+ clfe_codec_out_scb,
+ SCB_ON_PARENT_NEXT_SCB);
+
+
+ if (!magic_snoop_scb) goto _fail_end;
+ ins->ref_snoop_scb = magic_snoop_scb;
+
+ /* SP IO access */
+ if (!cs46xx_dsp_create_spio_write_scb(chip,"SPIOWriteSCB",SPIOWRITE_SCB_ADDR,
+ magic_snoop_scb,
+ SCB_ON_PARENT_NEXT_SCB))
+ goto _fail_end;
+
+ /* SPDIF input sampel rate converter */
+ src_task_scb = cs46xx_dsp_create_src_task_scb(chip,"SrcTaskSCB_SPDIFI",
+ ins->spdif_in_sample_rate,
+ SRC_OUTPUT_BUF1,
+ SRC_DELAY_BUF1,SRCTASK_SCB_ADDR,
+ master_mix_scb,
+ SCB_ON_PARENT_SUBLIST_SCB,1);
+
+ if (!src_task_scb) goto _fail_end;
+ cs46xx_src_unlink(chip,src_task_scb);
+
+ /* NOTE: when we now how to detect the SPDIF input
+ sample rate we will use this SRC to adjust it */
+ ins->spdif_in_src = src_task_scb;
+
+ cs46xx_dsp_async_init(chip,timing_master_scb);
+ return 0;
+
+ _fail_end:
+ snd_printk(KERN_ERR "dsp_spos: failed to setup SCB's in DSP\n");
+ return -EINVAL;
+}
+
+static int cs46xx_dsp_async_init (struct snd_cs46xx *chip,
+ struct dsp_scb_descriptor * fg_entry)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_symbol_entry * s16_async_codec_input_task;
+ struct dsp_symbol_entry * spdifo_task;
+ struct dsp_symbol_entry * spdifi_task;
+ struct dsp_scb_descriptor * spdifi_scb_desc, * spdifo_scb_desc, * async_codec_scb_desc;
+
+ s16_async_codec_input_task = cs46xx_dsp_lookup_symbol(chip, "S16_ASYNCCODECINPUTTASK", SYMBOL_CODE);
+ if (s16_async_codec_input_task == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol S16_ASYNCCODECINPUTTASK not found\n");
+ return -EIO;
+ }
+ spdifo_task = cs46xx_dsp_lookup_symbol(chip, "SPDIFOTASK", SYMBOL_CODE);
+ if (spdifo_task == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol SPDIFOTASK not found\n");
+ return -EIO;
+ }
+
+ spdifi_task = cs46xx_dsp_lookup_symbol(chip, "SPDIFITASK", SYMBOL_CODE);
+ if (spdifi_task == NULL) {
+ snd_printk(KERN_ERR "dsp_spos: symbol SPDIFITASK not found\n");
+ return -EIO;
+ }
+
+ {
+ /* 0xBC0 */
+ struct dsp_spdifoscb spdifo_scb = {
+ /* 0 */ DSP_SPOS_UUUU,
+ {
+ /* 1 */ 0xb0,
+ /* 2 */ 0,
+ /* 3 */ 0,
+ /* 4 */ 0,
+ },
+ /* NOTE: the SPDIF output task read samples in mono
+ format, the AsynchFGTxSCB task writes to buffer
+ in stereo format
+ */
+ /* 5 */ RSCONFIG_SAMPLE_16MONO + RSCONFIG_MODULO_256,
+ /* 6 */ ( SPDIFO_IP_OUTPUT_BUFFER1 << 0x10 ) | 0xFFFC,
+ /* 7 */ 0,0,
+ /* 8 */ 0,
+ /* 9 */ FG_TASK_HEADER_ADDR, NULL_SCB_ADDR,
+ /* A */ spdifo_task->address,
+ SPDIFO_SCB_INST + SPDIFOFIFOPointer,
+ {
+ /* B */ 0x0040, /*DSP_SPOS_UUUU,*/
+ /* C */ 0x20ff, /*DSP_SPOS_UUUU,*/
+ },
+ /* D */ 0x804c,0, /* SPDIFOFIFOPointer:SPDIFOStatRegAddr; */
+ /* E */ 0x0108,0x0001, /* SPDIFOStMoFormat:SPDIFOFIFOBaseAddr; */
+ /* F */ DSP_SPOS_UUUU /* SPDIFOFree; */
+ };
+
+ /* 0xBB0 */
+ struct dsp_spdifiscb spdifi_scb = {
+ /* 0 */ DSP_SPOS_UULO,DSP_SPOS_UUHI,
+ /* 1 */ 0,
+ /* 2 */ 0,
+ /* 3 */ 1,4000, /* SPDIFICountLimit SPDIFICount */
+ /* 4 */ DSP_SPOS_UUUU, /* SPDIFIStatusData */
+ /* 5 */ 0,DSP_SPOS_UUHI, /* StatusData, Free4 */
+ /* 6 */ DSP_SPOS_UUUU, /* Free3 */
+ /* 7 */ DSP_SPOS_UU,DSP_SPOS_DC, /* Free2 BitCount*/
+ /* 8 */ DSP_SPOS_UUUU, /* TempStatus */
+ /* 9 */ SPDIFO_SCB_INST, NULL_SCB_ADDR,
+ /* A */ spdifi_task->address,
+ SPDIFI_SCB_INST + SPDIFIFIFOPointer,
+ /* NOTE: The SPDIF input task write the sample in mono
+ format from the HW FIFO, the AsynchFGRxSCB task reads
+ them in stereo
+ */
+ /* B */ RSCONFIG_SAMPLE_16MONO + RSCONFIG_MODULO_128,
+ /* C */ (SPDIFI_IP_OUTPUT_BUFFER1 << 0x10) | 0xFFFC,
+ /* D */ 0x8048,0,
+ /* E */ 0x01f0,0x0001,
+ /* F */ DSP_SPOS_UUUU /* SPDIN_STATUS monitor */
+ };
+
+ /* 0xBA0 */
+ struct dsp_async_codec_input_scb async_codec_input_scb = {
+ /* 0 */ DSP_SPOS_UUUU,
+ /* 1 */ 0,
+ /* 2 */ 0,
+ /* 3 */ 1,4000,
+ /* 4 */ 0x0118,0x0001,
+ /* 5 */ RSCONFIG_SAMPLE_16MONO + RSCONFIG_MODULO_64,
+ /* 6 */ (ASYNC_IP_OUTPUT_BUFFER1 << 0x10) | 0xFFFC,
+ /* 7 */ DSP_SPOS_UU,0x3,
+ /* 8 */ DSP_SPOS_UUUU,
+ /* 9 */ SPDIFI_SCB_INST,NULL_SCB_ADDR,
+ /* A */ s16_async_codec_input_task->address,
+ HFG_TREE_SCB + AsyncCIOFIFOPointer,
+
+ /* B */ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_64,
+ /* C */ (ASYNC_IP_OUTPUT_BUFFER1 << 0x10), /*(ASYNC_IP_OUTPUT_BUFFER1 << 0x10) | 0xFFFC,*/
+
+#ifdef UseASER1Input
+ /* short AsyncCIFIFOPointer:AsyncCIStatRegAddr;
+ Init. 0000:8042: for ASER1
+ 0000:8044: for ASER2 */
+ /* D */ 0x8042,0,
+
+ /* short AsyncCIStMoFormat:AsyncCIFIFOBaseAddr;
+ Init 1 stero:8050 ASER1
+ Init 0 mono:8070 ASER2
+ Init 1 Stereo : 0100 ASER1 (Set by script) */
+ /* E */ 0x0100,0x0001,
+
+#endif
+
+#ifdef UseASER2Input
+ /* short AsyncCIFIFOPointer:AsyncCIStatRegAddr;
+ Init. 0000:8042: for ASER1
+ 0000:8044: for ASER2 */
+ /* D */ 0x8044,0,
+
+ /* short AsyncCIStMoFormat:AsyncCIFIFOBaseAddr;
+ Init 1 stero:8050 ASER1
+ Init 0 mono:8070 ASER2
+ Init 1 Stereo : 0100 ASER1 (Set by script) */
+ /* E */ 0x0110,0x0001,
+
+#endif
+
+ /* short AsyncCIOutputBufModulo:AsyncCIFree;
+ AsyncCIOutputBufModulo: The modulo size for
+ the output buffer of this task */
+ /* F */ 0, /* DSP_SPOS_UUUU */
+ };
+
+ spdifo_scb_desc = cs46xx_dsp_create_scb(chip,"SPDIFOSCB",(u32 *)&spdifo_scb,SPDIFO_SCB_INST);
+
+ if (snd_BUG_ON(!spdifo_scb_desc))
+ return -EIO;
+ spdifi_scb_desc = cs46xx_dsp_create_scb(chip,"SPDIFISCB",(u32 *)&spdifi_scb,SPDIFI_SCB_INST);
+ if (snd_BUG_ON(!spdifi_scb_desc))
+ return -EIO;
+ async_codec_scb_desc = cs46xx_dsp_create_scb(chip,"AsynCodecInputSCB",(u32 *)&async_codec_input_scb, HFG_TREE_SCB);
+ if (snd_BUG_ON(!async_codec_scb_desc))
+ return -EIO;
+
+ async_codec_scb_desc->parent_scb_ptr = NULL;
+ async_codec_scb_desc->next_scb_ptr = spdifi_scb_desc;
+ async_codec_scb_desc->sub_list_ptr = ins->the_null_scb;
+ async_codec_scb_desc->task_entry = s16_async_codec_input_task;
+
+ spdifi_scb_desc->parent_scb_ptr = async_codec_scb_desc;
+ spdifi_scb_desc->next_scb_ptr = spdifo_scb_desc;
+ spdifi_scb_desc->sub_list_ptr = ins->the_null_scb;
+ spdifi_scb_desc->task_entry = spdifi_task;
+
+ spdifo_scb_desc->parent_scb_ptr = spdifi_scb_desc;
+ spdifo_scb_desc->next_scb_ptr = fg_entry;
+ spdifo_scb_desc->sub_list_ptr = ins->the_null_scb;
+ spdifo_scb_desc->task_entry = spdifo_task;
+
+ /* this one is faked, as the parnet of SPDIFO task
+ is the FG task tree */
+ fg_entry->parent_scb_ptr = spdifo_scb_desc;
+
+ /* for proc fs */
+ cs46xx_dsp_proc_register_scb_desc (chip,spdifo_scb_desc);
+ cs46xx_dsp_proc_register_scb_desc (chip,spdifi_scb_desc);
+ cs46xx_dsp_proc_register_scb_desc (chip,async_codec_scb_desc);
+
+ /* Async MASTER ENABLE, affects both SPDIF input and output */
+ snd_cs46xx_pokeBA0(chip, BA0_ASER_MASTER, 0x1 );
+ }
+
+ return 0;
+}
+
+static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ /* set SPDIF output FIFO slot */
+ snd_cs46xx_pokeBA0(chip, BA0_ASER_FADDR, 0);
+
+ /* SPDIF output MASTER ENABLE */
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CONTROL, 0);
+
+ /* right and left validate bit */
+ /*cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);*/
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, 0x0);
+
+ /* clear fifo pointer */
+ cs46xx_poke_via_dsp (chip,SP_SPDIN_FIFOPTR, 0x0);
+
+ /* monitor state */
+ ins->spdif_status_out &= ~DSP_SPDIF_STATUS_HW_ENABLED;
+}
+
+int cs46xx_dsp_enable_spdif_hw (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ /* if hw-ctrl already enabled, turn off to reset logic ... */
+ cs46xx_dsp_disable_spdif_hw (chip);
+ udelay(50);
+
+ /* set SPDIF output FIFO slot */
+ snd_cs46xx_pokeBA0(chip, BA0_ASER_FADDR, ( 0x8000 | ((SP_SPDOUT_FIFO >> 4) << 4) ));
+
+ /* SPDIF output MASTER ENABLE */
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CONTROL, 0x80000000);
+
+ /* right and left validate bit */
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
+
+ /* monitor state */
+ ins->spdif_status_out |= DSP_SPDIF_STATUS_HW_ENABLED;
+
+ return 0;
+}
+
+int cs46xx_dsp_enable_spdif_in (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ /* turn on amplifier */
+ chip->active_ctrl(chip, 1);
+ chip->amplifier_ctrl(chip, 1);
+
+ if (snd_BUG_ON(ins->asynch_rx_scb))
+ return -EINVAL;
+ if (snd_BUG_ON(!ins->spdif_in_src))
+ return -EINVAL;
+
+ mutex_lock(&chip->spos_mutex);
+
+ if ( ! (ins->spdif_status_out & DSP_SPDIF_STATUS_INPUT_CTRL_ENABLED) ) {
+ /* time countdown enable */
+ cs46xx_poke_via_dsp (chip,SP_ASER_COUNTDOWN, 0x80000005);
+ /* NOTE: 80000005 value is just magic. With all values
+ that I've tested this one seem to give the best result.
+ Got no explication why. (Benny) */
+
+ /* SPDIF input MASTER ENABLE */
+ cs46xx_poke_via_dsp (chip,SP_SPDIN_CONTROL, 0x800003ff);
+
+ ins->spdif_status_out |= DSP_SPDIF_STATUS_INPUT_CTRL_ENABLED;
+ }
+
+ /* create and start the asynchronous receiver SCB */
+ ins->asynch_rx_scb = cs46xx_dsp_create_asynch_fg_rx_scb(chip,"AsynchFGRxSCB",
+ ASYNCRX_SCB_ADDR,
+ SPDIFI_SCB_INST,
+ SPDIFI_IP_OUTPUT_BUFFER1,
+ ins->spdif_in_src,
+ SCB_ON_PARENT_SUBLIST_SCB);
+
+ spin_lock_irq(&chip->reg_lock);
+
+ /* reset SPDIF input sample buffer pointer */
+ /*snd_cs46xx_poke (chip, (SPDIFI_SCB_INST + 0x0c) << 2,
+ (SPDIFI_IP_OUTPUT_BUFFER1 << 0x10) | 0xFFFC);*/
+
+ /* reset FIFO ptr */
+ /*cs46xx_poke_via_dsp (chip,SP_SPDIN_FIFOPTR, 0x0);*/
+ cs46xx_src_link(chip,ins->spdif_in_src);
+
+ /* unmute SRC volume */
+ cs46xx_dsp_scb_set_volume (chip,ins->spdif_in_src,0x7fff,0x7fff);
+
+ spin_unlock_irq(&chip->reg_lock);
+
+ /* set SPDIF input sample rate and unmute
+ NOTE: only 48khz support for SPDIF input this time */
+ /* cs46xx_dsp_set_src_sample_rate(chip,ins->spdif_in_src,48000); */
+
+ /* monitor state */
+ ins->spdif_status_in = 1;
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+int cs46xx_dsp_disable_spdif_in (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(!ins->asynch_rx_scb))
+ return -EINVAL;
+ if (snd_BUG_ON(!ins->spdif_in_src))
+ return -EINVAL;
+
+ mutex_lock(&chip->spos_mutex);
+
+ /* Remove the asynchronous receiver SCB */
+ cs46xx_dsp_remove_scb (chip,ins->asynch_rx_scb);
+ ins->asynch_rx_scb = NULL;
+
+ cs46xx_src_unlink(chip,ins->spdif_in_src);
+
+ /* monitor state */
+ ins->spdif_status_in = 0;
+ mutex_unlock(&chip->spos_mutex);
+
+ /* restore amplifier */
+ chip->active_ctrl(chip, -1);
+ chip->amplifier_ctrl(chip, -1);
+
+ return 0;
+}
+
+int cs46xx_dsp_enable_pcm_capture (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(ins->pcm_input))
+ return -EINVAL;
+ if (snd_BUG_ON(!ins->ref_snoop_scb))
+ return -EINVAL;
+
+ mutex_lock(&chip->spos_mutex);
+ ins->pcm_input = cs46xx_add_record_source(chip,ins->ref_snoop_scb,PCMSERIALIN_PCM_SCB_ADDR,
+ "PCMSerialInput_Wave");
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+int cs46xx_dsp_disable_pcm_capture (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(!ins->pcm_input))
+ return -EINVAL;
+
+ mutex_lock(&chip->spos_mutex);
+ cs46xx_dsp_remove_scb (chip,ins->pcm_input);
+ ins->pcm_input = NULL;
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+int cs46xx_dsp_enable_adc_capture (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(ins->adc_input))
+ return -EINVAL;
+ if (snd_BUG_ON(!ins->codec_in_scb))
+ return -EINVAL;
+
+ mutex_lock(&chip->spos_mutex);
+ ins->adc_input = cs46xx_add_record_source(chip,ins->codec_in_scb,PCMSERIALIN_SCB_ADDR,
+ "PCMSerialInput_ADC");
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+int cs46xx_dsp_disable_adc_capture (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(!ins->adc_input))
+ return -EINVAL;
+
+ mutex_lock(&chip->spos_mutex);
+ cs46xx_dsp_remove_scb (chip,ins->adc_input);
+ ins->adc_input = NULL;
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+int cs46xx_poke_via_dsp (struct snd_cs46xx *chip, u32 address, u32 data)
+{
+ u32 temp;
+ int i;
+
+ /* santiy check the parameters. (These numbers are not 100% correct. They are
+ a rough guess from looking at the controller spec.) */
+ if (address < 0x8000 || address >= 0x9000)
+ return -EINVAL;
+
+ /* initialize the SP_IO_WRITE SCB with the data. */
+ temp = ( address << 16 ) | ( address & 0x0000FFFF); /* offset 0 <-- address2 : address1 */
+
+ snd_cs46xx_poke(chip,( SPIOWRITE_SCB_ADDR << 2), temp);
+ snd_cs46xx_poke(chip,((SPIOWRITE_SCB_ADDR + 1) << 2), data); /* offset 1 <-- data1 */
+ snd_cs46xx_poke(chip,((SPIOWRITE_SCB_ADDR + 2) << 2), data); /* offset 1 <-- data2 */
+
+ /* Poke this location to tell the task to start */
+ snd_cs46xx_poke(chip,((SPIOWRITE_SCB_ADDR + 6) << 2), SPIOWRITE_SCB_ADDR << 0x10);
+
+ /* Verify that the task ran */
+ for (i=0; i<25; i++) {
+ udelay(125);
+
+ temp = snd_cs46xx_peek(chip,((SPIOWRITE_SCB_ADDR + 6) << 2));
+ if (temp == 0x00000000)
+ break;
+ }
+
+ if (i == 25) {
+ snd_printk(KERN_ERR "dsp_spos: SPIOWriteTask not responding\n");
+ return -EBUSY;
+ }
+
+ return 0;
+}
+
+int cs46xx_dsp_set_dac_volume (struct snd_cs46xx * chip, u16 left, u16 right)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * scb;
+
+ mutex_lock(&chip->spos_mutex);
+
+ /* main output */
+ scb = ins->master_mix_scb->sub_list_ptr;
+ while (scb != ins->the_null_scb) {
+ cs46xx_dsp_scb_set_volume (chip,scb,left,right);
+ scb = scb->next_scb_ptr;
+ }
+
+ /* rear output */
+ scb = ins->rear_mix_scb->sub_list_ptr;
+ while (scb != ins->the_null_scb) {
+ cs46xx_dsp_scb_set_volume (chip,scb,left,right);
+ scb = scb->next_scb_ptr;
+ }
+
+ ins->dac_volume_left = left;
+ ins->dac_volume_right = right;
+
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ mutex_lock(&chip->spos_mutex);
+
+ if (ins->asynch_rx_scb != NULL)
+ cs46xx_dsp_scb_set_volume (chip,ins->asynch_rx_scb,
+ left,right);
+
+ ins->spdif_input_volume_left = left;
+ ins->spdif_input_volume_right = right;
+
+ mutex_unlock(&chip->spos_mutex);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+int cs46xx_dsp_resume(struct snd_cs46xx * chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int i, err;
+
+ /* clear parameter, sample and code areas */
+ snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET,
+ DSP_PARAMETER_BYTE_SIZE);
+ snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET,
+ DSP_SAMPLE_BYTE_SIZE);
+ snd_cs46xx_clear_BA1(chip, DSP_CODE_BYTE_OFFSET, DSP_CODE_BYTE_SIZE);
+
+ for (i = 0; i < ins->nmodules; i++) {
+ struct dsp_module_desc *module = &ins->modules[i];
+ struct dsp_segment_desc *seg;
+ u32 doffset, dsize;
+
+ seg = get_segment_desc(module, SEGTYPE_SP_PARAMETER);
+ err = dsp_load_parameter(chip, seg);
+ if (err < 0)
+ return err;
+
+ seg = get_segment_desc(module, SEGTYPE_SP_SAMPLE);
+ err = dsp_load_sample(chip, seg);
+ if (err < 0)
+ return err;
+
+ seg = get_segment_desc(module, SEGTYPE_SP_PROGRAM);
+ if (!seg)
+ continue;
+
+ doffset = seg->offset * 4 + module->load_address * 4
+ + DSP_CODE_BYTE_OFFSET;
+ dsize = seg->size * 4;
+ err = snd_cs46xx_download(chip,
+ ins->code.data + module->load_address,
+ doffset, dsize);
+ if (err < 0)
+ return err;
+ }
+
+ for (i = 0; i < ins->ntask; i++) {
+ struct dsp_task_descriptor *t = &ins->tasks[i];
+ _dsp_create_task_tree(chip, t->data, t->address, t->size);
+ }
+
+ for (i = 0; i < ins->nscb; i++) {
+ struct dsp_scb_descriptor *s = &ins->scbs[i];
+ if (s->deleted)
+ continue;
+ _dsp_create_scb(chip, s->data, s->address);
+ }
+
+ return 0;
+}
+#endif
diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h
new file mode 100644
index 0000000..f9e169d
--- /dev/null
+++ b/sound/pci/cs46xx/dsp_spos.h
@@ -0,0 +1,227 @@
+/*
+ * The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * 2002-07 Benny Sjostrand benny@hostmobility.com
+ */
+
+#ifdef CONFIG_SND_CS46XX_NEW_DSP /* hack ... */
+#ifndef __DSP_SPOS_H__
+#define __DSP_SPOS_H__
+
+#define DSP_MAX_SYMBOLS 1024
+#define DSP_MAX_MODULES 64
+
+#define DSP_CODE_BYTE_SIZE 0x00007000UL
+#define DSP_PARAMETER_BYTE_SIZE 0x00003000UL
+#define DSP_SAMPLE_BYTE_SIZE 0x00003800UL
+#define DSP_PARAMETER_BYTE_OFFSET 0x00000000UL
+#define DSP_SAMPLE_BYTE_OFFSET 0x00010000UL
+#define DSP_CODE_BYTE_OFFSET 0x00020000UL
+
+#define WIDE_INSTR_MASK 0x0040
+#define WIDE_LADD_INSTR_MASK 0x0380
+
+/* this instruction types
+ needs to be reallocated when load
+ code into DSP */
+enum wide_opcode {
+ WIDE_FOR_BEGIN_LOOP = 0x20,
+ WIDE_FOR_BEGIN_LOOP2,
+
+ WIDE_COND_GOTO_ADDR = 0x30,
+ WIDE_COND_GOTO_CALL,
+
+ WIDE_TBEQ_COND_GOTO_ADDR = 0x70,
+ WIDE_TBEQ_COND_CALL_ADDR,
+ WIDE_TBEQ_NCOND_GOTO_ADDR,
+ WIDE_TBEQ_NCOND_CALL_ADDR,
+ WIDE_TBEQ_COND_GOTO1_ADDR,
+ WIDE_TBEQ_COND_CALL1_ADDR,
+ WIDE_TBEQ_NCOND_GOTOI_ADDR,
+ WIDE_TBEQ_NCOND_CALL1_ADDR,
+};
+
+/* SAMPLE segment */
+#define VARI_DECIMATE_BUF1 0x0000
+#define WRITE_BACK_BUF1 0x0400
+#define CODEC_INPUT_BUF1 0x0500
+#define PCM_READER_BUF1 0x0600
+#define SRC_DELAY_BUF1 0x0680
+#define VARI_DECIMATE_BUF0 0x0780
+#define SRC_OUTPUT_BUF1 0x07A0
+#define ASYNC_IP_OUTPUT_BUFFER1 0x0A00
+#define OUTPUT_SNOOP_BUFFER 0x0B00
+#define SPDIFI_IP_OUTPUT_BUFFER1 0x0E00
+#define SPDIFO_IP_OUTPUT_BUFFER1 0x1000
+#define MIX_SAMPLE_BUF1 0x1400
+#define MIX_SAMPLE_BUF2 0x2E80
+#define MIX_SAMPLE_BUF3 0x2F00
+#define MIX_SAMPLE_BUF4 0x2F80
+#define MIX_SAMPLE_BUF5 0x3000
+
+/* Task stack address */
+#define HFG_STACK 0x066A
+#define FG_STACK 0x066E
+#define BG_STACK 0x068E
+
+/* SCB's addresses */
+#define SPOSCB_ADDR 0x070
+#define BG_TREE_SCB_ADDR 0x635
+#define NULL_SCB_ADDR 0x000
+#define TIMINGMASTER_SCB_ADDR 0x010
+#define CODECOUT_SCB_ADDR 0x020
+#define PCMREADER_SCB_ADDR 0x030
+#define WRITEBACK_SCB_ADDR 0x040
+#define CODECIN_SCB_ADDR 0x080
+#define MASTERMIX_SCB_ADDR 0x090
+#define SRCTASK_SCB_ADDR 0x0A0
+#define VARIDECIMATE_SCB_ADDR 0x0B0
+#define PCMSERIALIN_SCB_ADDR 0x0C0
+#define FG_TASK_HEADER_ADDR 0x600
+#define ASYNCTX_SCB_ADDR 0x0E0
+#define ASYNCRX_SCB_ADDR 0x0F0
+#define SRCTASKII_SCB_ADDR 0x100
+#define OUTPUTSNOOP_SCB_ADDR 0x110
+#define PCMSERIALINII_SCB_ADDR 0x120
+#define SPIOWRITE_SCB_ADDR 0x130
+#define REAR_CODECOUT_SCB_ADDR 0x140
+#define OUTPUTSNOOPII_SCB_ADDR 0x150
+#define PCMSERIALIN_PCM_SCB_ADDR 0x160
+#define RECORD_MIXER_SCB_ADDR 0x170
+#define REAR_MIXER_SCB_ADDR 0x180
+#define CLFE_MIXER_SCB_ADDR 0x190
+#define CLFE_CODEC_SCB_ADDR 0x1A0
+
+/* hyperforground SCB's*/
+#define HFG_TREE_SCB 0xBA0
+#define SPDIFI_SCB_INST 0xBB0
+#define SPDIFO_SCB_INST 0xBC0
+#define WRITE_BACK_SPB 0x0D0
+
+/* offsets */
+#define AsyncCIOFIFOPointer 0xd
+#define SPDIFOFIFOPointer 0xd
+#define SPDIFIFIFOPointer 0xd
+#define TCBData 0xb
+#define HFGFlags 0xa
+#define TCBContextBlk 0x10
+#define AFGTxAccumPhi 0x4
+#define SCBsubListPtr 0x9
+#define SCBfuncEntryPtr 0xA
+#define SRCCorPerGof 0x2
+#define SRCPhiIncr6Int26Frac 0xd
+#define SCBVolumeCtrl 0xe
+
+/* conf */
+#define UseASER1Input 1
+
+
+
+/*
+ * The following defines are for the flags in the rsConfig01/23 registers of
+ * the SP.
+ */
+
+#define RSCONFIG_MODULO_SIZE_MASK 0x0000000FL
+#define RSCONFIG_MODULO_16 0x00000001L
+#define RSCONFIG_MODULO_32 0x00000002L
+#define RSCONFIG_MODULO_64 0x00000003L
+#define RSCONFIG_MODULO_128 0x00000004L
+#define RSCONFIG_MODULO_256 0x00000005L
+#define RSCONFIG_MODULO_512 0x00000006L
+#define RSCONFIG_MODULO_1024 0x00000007L
+#define RSCONFIG_MODULO_4 0x00000008L
+#define RSCONFIG_MODULO_8 0x00000009L
+#define RSCONFIG_SAMPLE_SIZE_MASK 0x000000C0L
+#define RSCONFIG_SAMPLE_8MONO 0x00000000L
+#define RSCONFIG_SAMPLE_8STEREO 0x00000040L
+#define RSCONFIG_SAMPLE_16MONO 0x00000080L
+#define RSCONFIG_SAMPLE_16STEREO 0x000000C0L
+#define RSCONFIG_UNDERRUN_ZERO 0x00004000L
+#define RSCONFIG_DMA_TO_HOST 0x00008000L
+#define RSCONFIG_STREAM_NUM_MASK 0x00FF0000L
+#define RSCONFIG_MAX_DMA_SIZE_MASK 0x1F000000L
+#define RSCONFIG_DMA_ENABLE 0x20000000L
+#define RSCONFIG_PRIORITY_MASK 0xC0000000L
+#define RSCONFIG_PRIORITY_HIGH 0x00000000L
+#define RSCONFIG_PRIORITY_MEDIUM_HIGH 0x40000000L
+#define RSCONFIG_PRIORITY_MEDIUM_LOW 0x80000000L
+#define RSCONFIG_PRIORITY_LOW 0xC0000000L
+#define RSCONFIG_STREAM_NUM_SHIFT 16L
+#define RSCONFIG_MAX_DMA_SIZE_SHIFT 24L
+
+/* SP constants */
+#define FG_INTERVAL_TIMER_PERIOD 0x0051
+#define BG_INTERVAL_TIMER_PERIOD 0x0100
+
+
+/* Only SP accessible registers */
+#define SP_ASER_COUNTDOWN 0x8040
+#define SP_SPDOUT_FIFO 0x0108
+#define SP_SPDIN_MI_FIFO 0x01E0
+#define SP_SPDIN_D_FIFO 0x01F0
+#define SP_SPDIN_STATUS 0x8048
+#define SP_SPDIN_CONTROL 0x8049
+#define SP_SPDIN_FIFOPTR 0x804A
+#define SP_SPDOUT_STATUS 0x804C
+#define SP_SPDOUT_CONTROL 0x804D
+#define SP_SPDOUT_CSUV 0x808E
+
+static inline u8 _wrap_all_bits (u8 val)
+{
+ u8 wrapped;
+
+ /* wrap all 8 bits */
+ wrapped =
+ ((val & 0x1 ) << 7) |
+ ((val & 0x2 ) << 5) |
+ ((val & 0x4 ) << 3) |
+ ((val & 0x8 ) << 1) |
+ ((val & 0x10) >> 1) |
+ ((val & 0x20) >> 3) |
+ ((val & 0x40) >> 5) |
+ ((val & 0x80) >> 7);
+
+ return wrapped;
+}
+
+static inline void cs46xx_dsp_spos_update_scb (struct snd_cs46xx * chip,
+ struct dsp_scb_descriptor * scb)
+{
+ /* update nextSCB and subListPtr in SCB */
+ snd_cs46xx_poke(chip,
+ (scb->address + SCBsubListPtr) << 2,
+ (scb->sub_list_ptr->address << 0x10) |
+ (scb->next_scb_ptr->address));
+}
+
+static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip,
+ struct dsp_scb_descriptor * scb,
+ u16 left, u16 right)
+{
+ unsigned int val = ((0xffff - left) << 16 | (0xffff - right));
+
+ snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val);
+ snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val);
+}
+#endif /* __DSP_SPOS_H__ */
+#endif /* CONFIG_SND_CS46XX_NEW_DSP */
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
new file mode 100644
index 0000000..dd7c41b
--- /dev/null
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -0,0 +1,1787 @@
+/*
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * 2002-07 Benny Sjostrand benny@hostmobility.com
+ */
+
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/cs46xx.h>
+
+#include "cs46xx_lib.h"
+#include "dsp_spos.h"
+
+struct proc_scb_info {
+ struct dsp_scb_descriptor * scb_desc;
+ struct snd_cs46xx *chip;
+};
+
+static void remove_symbol (struct snd_cs46xx * chip, struct dsp_symbol_entry * symbol)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ int symbol_index = (int)(symbol - ins->symbol_table.symbols);
+
+ if (snd_BUG_ON(ins->symbol_table.nsymbols <= 0))
+ return;
+ if (snd_BUG_ON(symbol_index < 0 ||
+ symbol_index >= ins->symbol_table.nsymbols))
+ return;
+
+ ins->symbol_table.symbols[symbol_index].deleted = 1;
+
+ if (symbol_index < ins->symbol_table.highest_frag_index) {
+ ins->symbol_table.highest_frag_index = symbol_index;
+ }
+
+ if (symbol_index == ins->symbol_table.nsymbols - 1)
+ ins->symbol_table.nsymbols --;
+
+ if (ins->symbol_table.highest_frag_index > ins->symbol_table.nsymbols) {
+ ins->symbol_table.highest_frag_index = ins->symbol_table.nsymbols;
+ }
+
+}
+
+#ifdef CONFIG_PROC_FS
+static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct proc_scb_info * scb_info = entry->private_data;
+ struct dsp_scb_descriptor * scb = scb_info->scb_desc;
+ struct dsp_spos_instance * ins;
+ struct snd_cs46xx *chip = scb_info->chip;
+ int j,col;
+ void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET;
+
+ ins = chip->dsp_spos_instance;
+
+ mutex_lock(&chip->spos_mutex);
+ snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name);
+
+ for (col = 0,j = 0;j < 0x10; j++,col++) {
+ if (col == 4) {
+ snd_iprintf(buffer,"\n");
+ col = 0;
+ }
+ snd_iprintf(buffer,"%08x ",readl(dst + (scb->address + j) * sizeof(u32)));
+ }
+
+ snd_iprintf(buffer,"\n");
+
+ if (scb->parent_scb_ptr != NULL) {
+ snd_iprintf(buffer,"parent [%s:%04x] ",
+ scb->parent_scb_ptr->scb_name,
+ scb->parent_scb_ptr->address);
+ } else snd_iprintf(buffer,"parent [none] ");
+
+ snd_iprintf(buffer,"sub_list_ptr [%s:%04x]\nnext_scb_ptr [%s:%04x] task_entry [%s:%04x]\n",
+ scb->sub_list_ptr->scb_name,
+ scb->sub_list_ptr->address,
+ scb->next_scb_ptr->scb_name,
+ scb->next_scb_ptr->address,
+ scb->task_entry->symbol_name,
+ scb->task_entry->address);
+
+ snd_iprintf(buffer,"index [%d] ref_count [%d]\n",scb->index,scb->ref_count);
+ mutex_unlock(&chip->spos_mutex);
+}
+#endif
+
+static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ unsigned long flags;
+
+ if ( scb->parent_scb_ptr ) {
+ /* unlink parent SCB */
+ if (snd_BUG_ON(scb->parent_scb_ptr->sub_list_ptr != scb &&
+ scb->parent_scb_ptr->next_scb_ptr != scb))
+ return;
+
+ if (scb->parent_scb_ptr->sub_list_ptr == scb) {
+
+ if (scb->next_scb_ptr == ins->the_null_scb) {
+ /* last and only node in parent sublist */
+ scb->parent_scb_ptr->sub_list_ptr = scb->sub_list_ptr;
+
+ if (scb->sub_list_ptr != ins->the_null_scb) {
+ scb->sub_list_ptr->parent_scb_ptr = scb->parent_scb_ptr;
+ }
+ scb->sub_list_ptr = ins->the_null_scb;
+ } else {
+ /* first node in parent sublist */
+ scb->parent_scb_ptr->sub_list_ptr = scb->next_scb_ptr;
+
+ if (scb->next_scb_ptr != ins->the_null_scb) {
+ /* update next node parent ptr. */
+ scb->next_scb_ptr->parent_scb_ptr = scb->parent_scb_ptr;
+ }
+ scb->next_scb_ptr = ins->the_null_scb;
+ }
+ } else {
+ scb->parent_scb_ptr->next_scb_ptr = scb->next_scb_ptr;
+
+ if (scb->next_scb_ptr != ins->the_null_scb) {
+ /* update next node parent ptr. */
+ scb->next_scb_ptr->parent_scb_ptr = scb->parent_scb_ptr;
+ }
+ scb->next_scb_ptr = ins->the_null_scb;
+ }
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ /* update parent first entry in DSP RAM */
+ cs46xx_dsp_spos_update_scb(chip,scb->parent_scb_ptr);
+
+ /* then update entry in DSP RAM */
+ cs46xx_dsp_spos_update_scb(chip,scb);
+
+ scb->parent_scb_ptr = NULL;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
+}
+
+static void _dsp_clear_sample_buffer (struct snd_cs46xx *chip, u32 sample_buffer_addr,
+ int dword_count)
+{
+ void __iomem *dst = chip->region.idx[2].remap_addr + sample_buffer_addr;
+ int i;
+
+ for (i = 0; i < dword_count ; ++i ) {
+ writel(0, dst);
+ dst += 4;
+ }
+}
+
+void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ unsigned long flags;
+
+ /* check integrety */
+ if (snd_BUG_ON(scb->index < 0 ||
+ scb->index >= ins->nscb ||
+ (ins->scbs + scb->index) != scb))
+ return;
+
+#if 0
+ /* can't remove a SCB with childs before
+ removing childs first */
+ if (snd_BUG_ON(scb->sub_list_ptr != ins->the_null_scb ||
+ scb->next_scb_ptr != ins->the_null_scb))
+ goto _end;
+#endif
+
+ spin_lock_irqsave(&scb->lock, flags);
+ _dsp_unlink_scb (chip,scb);
+ spin_unlock_irqrestore(&scb->lock, flags);
+
+ cs46xx_dsp_proc_free_scb_desc(scb);
+ if (snd_BUG_ON(!scb->scb_symbol))
+ return;
+ remove_symbol (chip,scb->scb_symbol);
+
+ ins->scbs[scb->index].deleted = 1;
+
+ if (scb->index < ins->scb_highest_frag_index)
+ ins->scb_highest_frag_index = scb->index;
+
+ if (scb->index == ins->nscb - 1) {
+ ins->nscb --;
+ }
+
+ if (ins->scb_highest_frag_index > ins->nscb) {
+ ins->scb_highest_frag_index = ins->nscb;
+ }
+
+#if 0
+ /* !!!! THIS IS A PIECE OF SHIT MADE BY ME !!! */
+ for(i = scb->index + 1;i < ins->nscb; ++i) {
+ ins->scbs[i - 1].index = i - 1;
+ }
+#endif
+}
+
+
+#ifdef CONFIG_PROC_FS
+void cs46xx_dsp_proc_free_scb_desc (struct dsp_scb_descriptor * scb)
+{
+ if (scb->proc_info) {
+ struct proc_scb_info * scb_info = scb->proc_info->private_data;
+
+ snd_printdd("cs46xx_dsp_proc_free_scb_desc: freeing %s\n",scb->scb_name);
+
+ snd_info_free_entry(scb->proc_info);
+ scb->proc_info = NULL;
+
+ kfree (scb_info);
+ }
+}
+
+void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip,
+ struct dsp_scb_descriptor * scb)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct snd_info_entry * entry;
+ struct proc_scb_info * scb_info;
+
+ /* register to proc */
+ if (ins->snd_card != NULL && ins->proc_dsp_dir != NULL &&
+ scb->proc_info == NULL) {
+
+ if ((entry = snd_info_create_card_entry(ins->snd_card, scb->scb_name,
+ ins->proc_dsp_dir)) != NULL) {
+ scb_info = kmalloc(sizeof(struct proc_scb_info), GFP_KERNEL);
+ if (!scb_info) {
+ snd_info_free_entry(entry);
+ entry = NULL;
+ goto out;
+ }
+
+ scb_info->chip = chip;
+ scb_info->scb_desc = scb;
+
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = scb_info;
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+
+ entry->c.text.read = cs46xx_dsp_proc_scb_info_read;
+
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ kfree (scb_info);
+ entry = NULL;
+ }
+ }
+out:
+ scb->proc_info = entry;
+ }
+}
+#endif /* CONFIG_PROC_FS */
+
+static struct dsp_scb_descriptor *
+_dsp_create_generic_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest,
+ struct dsp_symbol_entry * task_entry,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * scb;
+
+ unsigned long flags;
+
+ if (snd_BUG_ON(!ins->the_null_scb))
+ return NULL;
+
+ /* fill the data that will be wroten to DSP */
+ scb_data[SCBsubListPtr] =
+ (ins->the_null_scb->address << 0x10) | ins->the_null_scb->address;
+
+ scb_data[SCBfuncEntryPtr] &= 0xFFFF0000;
+ scb_data[SCBfuncEntryPtr] |= task_entry->address;
+
+ snd_printdd("dsp_spos: creating SCB <%s>\n",name);
+
+ scb = cs46xx_dsp_create_scb(chip,name,scb_data,dest);
+
+
+ scb->sub_list_ptr = ins->the_null_scb;
+ scb->next_scb_ptr = ins->the_null_scb;
+
+ scb->parent_scb_ptr = parent_scb;
+ scb->task_entry = task_entry;
+
+
+ /* update parent SCB */
+ if (scb->parent_scb_ptr) {
+#if 0
+ printk ("scb->parent_scb_ptr = %s\n",scb->parent_scb_ptr->scb_name);
+ printk ("scb->parent_scb_ptr->next_scb_ptr = %s\n",scb->parent_scb_ptr->next_scb_ptr->scb_name);
+ printk ("scb->parent_scb_ptr->sub_list_ptr = %s\n",scb->parent_scb_ptr->sub_list_ptr->scb_name);
+#endif
+ /* link to parent SCB */
+ if (scb_child_type == SCB_ON_PARENT_NEXT_SCB) {
+ if (snd_BUG_ON(scb->parent_scb_ptr->next_scb_ptr !=
+ ins->the_null_scb))
+ return NULL;
+
+ scb->parent_scb_ptr->next_scb_ptr = scb;
+
+ } else if (scb_child_type == SCB_ON_PARENT_SUBLIST_SCB) {
+ if (snd_BUG_ON(scb->parent_scb_ptr->sub_list_ptr !=
+ ins->the_null_scb))
+ return NULL;
+
+ scb->parent_scb_ptr->sub_list_ptr = scb;
+ } else {
+ snd_BUG();
+ }
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ /* update entry in DSP RAM */
+ cs46xx_dsp_spos_update_scb(chip,scb->parent_scb_ptr);
+
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ }
+
+
+ cs46xx_dsp_proc_register_scb_desc (chip,scb);
+
+ return scb;
+}
+
+static struct dsp_scb_descriptor *
+cs46xx_dsp_create_generic_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data,
+ u32 dest, char * task_entry_name,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_symbol_entry * task_entry;
+
+ task_entry = cs46xx_dsp_lookup_symbol (chip,task_entry_name,
+ SYMBOL_CODE);
+
+ if (task_entry == NULL) {
+ snd_printk (KERN_ERR "dsp_spos: symbol %s not found\n",task_entry_name);
+ return NULL;
+ }
+
+ return _dsp_create_generic_scb (chip,name,scb_data,dest,task_entry,
+ parent_scb,scb_child_type);
+}
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_timing_master_scb (struct snd_cs46xx *chip)
+{
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_timing_master_scb timing_master_scb = {
+ { 0,
+ 0,
+ 0,
+ 0
+ },
+ { 0,
+ 0,
+ 0,
+ 0,
+ 0
+ },
+ 0,0,
+ 0,NULL_SCB_ADDR,
+ 0,0, /* extraSampleAccum:TMreserved */
+ 0,0, /* codecFIFOptr:codecFIFOsyncd */
+ 0x0001,0x8000, /* fracSampAccumQm1:TMfrmsLeftInGroup */
+ 0x0001,0x0000, /* fracSampCorrectionQm1:TMfrmGroupLength */
+ 0x00060000 /* nSampPerFrmQ15 */
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,"TimingMasterSCBInst",(u32 *)&timing_master_scb,
+ TIMINGMASTER_SCB_ADDR,
+ "TIMINGMASTER",NULL,SCB_NO_PARENT);
+
+ return scb;
+}
+
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_codec_out_scb(struct snd_cs46xx * chip, char * codec_name,
+ u16 channel_disp, u16 fifo_addr, u16 child_scb_addr,
+ u32 dest, struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_codec_output_scb codec_out_scb = {
+ { 0,
+ 0,
+ 0,
+ 0
+ },
+ {
+ 0,
+ 0,
+ 0,
+ 0,
+ 0
+ },
+ 0,0,
+ 0,NULL_SCB_ADDR,
+ 0, /* COstrmRsConfig */
+ 0, /* COstrmBufPtr */
+ channel_disp,fifo_addr, /* leftChanBaseIOaddr:rightChanIOdisp */
+ 0x0000,0x0080, /* (!AC97!) COexpVolChangeRate:COscaleShiftCount */
+ 0,child_scb_addr /* COreserved - need child scb to work with rom code */
+ };
+
+
+ scb = cs46xx_dsp_create_generic_scb(chip,codec_name,(u32 *)&codec_out_scb,
+ dest,"S16_CODECOUTPUTTASK",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_codec_in_scb(struct snd_cs46xx * chip, char * codec_name,
+ u16 channel_disp, u16 fifo_addr, u16 sample_buffer_addr,
+ u32 dest, struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+
+ struct dsp_scb_descriptor * scb;
+ struct dsp_codec_input_scb codec_input_scb = {
+ { 0,
+ 0,
+ 0,
+ 0
+ },
+ {
+ 0,
+ 0,
+ 0,
+ 0,
+ 0
+ },
+
+#if 0 /* cs4620 */
+ SyncIOSCB,NULL_SCB_ADDR
+#else
+ 0 , 0,
+#endif
+ 0,0,
+
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_64, /* strmRsConfig */
+ sample_buffer_addr << 0x10, /* strmBufPtr; defined as a dword ptr, used as a byte ptr */
+ channel_disp,fifo_addr, /* (!AC97!) leftChanBaseINaddr=AC97primary
+ link input slot 3 :rightChanINdisp=""slot 4 */
+ 0x0000,0x0000, /* (!AC97!) ????:scaleShiftCount; no shift needed
+ because AC97 is already 20 bits */
+ 0x80008000 /* ??clw cwcgame.scb has 0 */
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,codec_name,(u32 *)&codec_input_scb,
+ dest,"S16_CODECINPUTTASK",parent_scb,
+ scb_child_type);
+ return scb;
+}
+
+
+static struct dsp_scb_descriptor *
+cs46xx_dsp_create_pcm_reader_scb(struct snd_cs46xx * chip, char * scb_name,
+ u16 sample_buffer_addr, u32 dest,
+ int virtual_channel, u32 playback_hw_addr,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_generic_scb pcm_reader_scb = {
+
+ /*
+ Play DMA Task xfers data from host buffer to SP buffer
+ init/runtime variables:
+ PlayAC: Play Audio Data Conversion - SCB loc: 2nd dword, mask: 0x0000F000L
+ DATA_FMT_16BIT_ST_LTLEND(0x00000000L) from 16-bit stereo, little-endian
+ DATA_FMT_8_BIT_ST_SIGNED(0x00001000L) from 8-bit stereo, signed
+ DATA_FMT_16BIT_MN_LTLEND(0x00002000L) from 16-bit mono, little-endian
+ DATA_FMT_8_BIT_MN_SIGNED(0x00003000L) from 8-bit mono, signed
+ DATA_FMT_16BIT_ST_BIGEND(0x00004000L) from 16-bit stereo, big-endian
+ DATA_FMT_16BIT_MN_BIGEND(0x00006000L) from 16-bit mono, big-endian
+ DATA_FMT_8_BIT_ST_UNSIGNED(0x00009000L) from 8-bit stereo, unsigned
+ DATA_FMT_8_BIT_MN_UNSIGNED(0x0000b000L) from 8-bit mono, unsigned
+ ? Other combinations possible from:
+ DMA_RQ_C2_AUDIO_CONVERT_MASK 0x0000F000L
+ DMA_RQ_C2_AC_NONE 0x00000000L
+ DMA_RQ_C2_AC_8_TO_16_BIT 0x00001000L
+ DMA_RQ_C2_AC_MONO_TO_STEREO 0x00002000L
+ DMA_RQ_C2_AC_ENDIAN_CONVERT 0x00004000L
+ DMA_RQ_C2_AC_SIGNED_CONVERT 0x00008000L
+
+ HostBuffAddr: Host Buffer Physical Byte Address - SCB loc:3rd dword, Mask: 0xFFFFFFFFL
+ aligned to dword boundary
+ */
+ /* Basic (non scatter/gather) DMA requestor (4 ints) */
+ { DMA_RQ_C1_SOURCE_ON_HOST + /* source buffer is on the host */
+ DMA_RQ_C1_SOURCE_MOD1024 + /* source buffer is 1024 dwords (4096 bytes) */
+ DMA_RQ_C1_DEST_MOD32 + /* dest buffer(PCMreaderBuf) is 32 dwords*/
+ DMA_RQ_C1_WRITEBACK_SRC_FLAG + /* ?? */
+ DMA_RQ_C1_WRITEBACK_DEST_FLAG + /* ?? */
+ 15, /* DwordCount-1: picked 16 for DwordCount because Jim */
+ /* Barnette said that is what we should use since */
+ /* we are not running in optimized mode? */
+ DMA_RQ_C2_AC_NONE +
+ DMA_RQ_C2_SIGNAL_SOURCE_PINGPONG + /* set play interrupt (bit0) in HISR when source */
+ /* buffer (on host) crosses half-way point */
+ virtual_channel, /* Play DMA channel arbitrarily set to 0 */
+ playback_hw_addr, /* HostBuffAddr (source) */
+ DMA_RQ_SD_SP_SAMPLE_ADDR + /* destination buffer is in SP Sample Memory */
+ sample_buffer_addr /* SP Buffer Address (destination) */
+ },
+ /* Scatter/gather DMA requestor extension (5 ints) */
+ {
+ 0,
+ 0,
+ 0,
+ 0,
+ 0
+ },
+ /* Sublist pointer & next stream control block (SCB) link. */
+ NULL_SCB_ADDR,NULL_SCB_ADDR,
+ /* Pointer to this tasks parameter block & stream function pointer */
+ 0,NULL_SCB_ADDR,
+ /* rsConfig register for stream buffer (rsDMA reg. is loaded from basicReq.daw */
+ /* for incoming streams, or basicReq.saw, for outgoing streams) */
+ RSCONFIG_DMA_ENABLE + /* enable DMA */
+ (19 << RSCONFIG_MAX_DMA_SIZE_SHIFT) + /* MAX_DMA_SIZE picked to be 19 since SPUD */
+ /* uses it for some reason */
+ ((dest >> 4) << RSCONFIG_STREAM_NUM_SHIFT) + /* stream number = SCBaddr/16 */
+ RSCONFIG_SAMPLE_16STEREO +
+ RSCONFIG_MODULO_32, /* dest buffer(PCMreaderBuf) is 32 dwords (256 bytes) */
+ /* Stream sample pointer & MAC-unit mode for this stream */
+ (sample_buffer_addr << 0x10),
+ /* Fractional increment per output sample in the input sample buffer */
+ 0,
+ {
+ /* Standard stereo volume control
+ default muted */
+ 0xffff,0xffff,
+ 0xffff,0xffff
+ }
+ };
+
+ if (ins->null_algorithm == NULL) {
+ ins->null_algorithm = cs46xx_dsp_lookup_symbol (chip,"NULLALGORITHM",
+ SYMBOL_CODE);
+
+ if (ins->null_algorithm == NULL) {
+ snd_printk (KERN_ERR "dsp_spos: symbol NULLALGORITHM not found\n");
+ return NULL;
+ }
+ }
+
+ scb = _dsp_create_generic_scb(chip,scb_name,(u32 *)&pcm_reader_scb,
+ dest,ins->null_algorithm,parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+#define GOF_PER_SEC 200
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_src_task_scb(struct snd_cs46xx * chip, char * scb_name,
+ int rate,
+ u16 src_buffer_addr,
+ u16 src_delay_buffer_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type,
+ int pass_through)
+{
+
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * scb;
+ unsigned int tmp1, tmp2;
+ unsigned int phiIncr;
+ unsigned int correctionPerGOF, correctionPerSec;
+
+ snd_printdd( "dsp_spos: setting %s rate to %u\n",scb_name,rate);
+
+ /*
+ * Compute the values used to drive the actual sample rate conversion.
+ * The following formulas are being computed, using inline assembly
+ * since we need to use 64 bit arithmetic to compute the values:
+ *
+ * phiIncr = floor((Fs,in * 2^26) / Fs,out)
+ * correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /
+ * GOF_PER_SEC)
+ * ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -M
+ * GOF_PER_SEC * correctionPerGOF
+ *
+ * i.e.
+ *
+ * phiIncr:other = dividend:remainder((Fs,in * 2^26) / Fs,out)
+ * correctionPerGOF:correctionPerSec =
+ * dividend:remainder(ulOther / GOF_PER_SEC)
+ */
+ tmp1 = rate << 16;
+ phiIncr = tmp1 / 48000;
+ tmp1 -= phiIncr * 48000;
+ tmp1 <<= 10;
+ phiIncr <<= 10;
+ tmp2 = tmp1 / 48000;
+ phiIncr += tmp2;
+ tmp1 -= tmp2 * 48000;
+ correctionPerGOF = tmp1 / GOF_PER_SEC;
+ tmp1 -= correctionPerGOF * GOF_PER_SEC;
+ correctionPerSec = tmp1;
+
+ {
+ struct dsp_src_task_scb src_task_scb = {
+ 0x0028,0x00c8,
+ 0x5555,0x0000,
+ 0x0000,0x0000,
+ src_buffer_addr,1,
+ correctionPerGOF,correctionPerSec,
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32,
+ 0x0000,src_delay_buffer_addr,
+ 0x0,
+ 0x080,(src_delay_buffer_addr + (24 * 4)),
+ 0,0, /* next_scb, sub_list_ptr */
+ 0,0, /* entry, this_spb */
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_8,
+ src_buffer_addr << 0x10,
+ phiIncr,
+ {
+ 0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left,
+ 0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left
+ }
+ };
+
+ if (ins->s16_up == NULL) {
+ ins->s16_up = cs46xx_dsp_lookup_symbol (chip,"S16_UPSRC",
+ SYMBOL_CODE);
+
+ if (ins->s16_up == NULL) {
+ snd_printk (KERN_ERR "dsp_spos: symbol S16_UPSRC not found\n");
+ return NULL;
+ }
+ }
+
+ /* clear buffers */
+ _dsp_clear_sample_buffer (chip,src_buffer_addr,8);
+ _dsp_clear_sample_buffer (chip,src_delay_buffer_addr,32);
+
+ if (pass_through) {
+ /* wont work with any other rate than
+ the native DSP rate */
+ snd_BUG_ON(rate != 48000);
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&src_task_scb,
+ dest,"DMAREADER",parent_scb,
+ scb_child_type);
+ } else {
+ scb = _dsp_create_generic_scb(chip,scb_name,(u32 *)&src_task_scb,
+ dest,ins->s16_up,parent_scb,
+ scb_child_type);
+ }
+
+
+ }
+
+ return scb;
+}
+
+#if 0 /* not used */
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_filter_scb(struct snd_cs46xx * chip, char * scb_name,
+ u16 buffer_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type) {
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_filter_scb filter_scb = {
+ .a0_right = 0x41a9,
+ .a0_left = 0x41a9,
+ .a1_right = 0xb8e4,
+ .a1_left = 0xb8e4,
+ .a2_right = 0x3e55,
+ .a2_left = 0x3e55,
+
+ .filter_unused3 = 0x0000,
+ .filter_unused2 = 0x0000,
+
+ .output_buf_ptr = buffer_addr,
+ .init = 0x000,
+
+ .prev_sample_output1 = 0x00000000,
+ .prev_sample_output2 = 0x00000000,
+
+ .prev_sample_input1 = 0x00000000,
+ .prev_sample_input2 = 0x00000000,
+
+ .next_scb_ptr = 0x0000,
+ .sub_list_ptr = 0x0000,
+
+ .entry_point = 0x0000,
+ .spb_ptr = 0x0000,
+
+ .b0_right = 0x0e38,
+ .b0_left = 0x0e38,
+ .b1_right = 0x1c71,
+ .b1_left = 0x1c71,
+ .b2_right = 0x0e38,
+ .b2_left = 0x0e38,
+ };
+
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&filter_scb,
+ dest,"FILTERTASK",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+#endif /* not used */
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_mix_only_scb(struct snd_cs46xx * chip, char * scb_name,
+ u16 mix_buffer_addr, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_mix_only_scb master_mix_scb = {
+ /* 0 */ { 0,
+ /* 1 */ 0,
+ /* 2 */ mix_buffer_addr,
+ /* 3 */ 0
+ /* */ },
+ {
+ /* 4 */ 0,
+ /* 5 */ 0,
+ /* 6 */ 0,
+ /* 7 */ 0,
+ /* 8 */ 0x00000080
+ },
+ /* 9 */ 0,0,
+ /* A */ 0,0,
+ /* B */ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32,
+ /* C */ (mix_buffer_addr + (16 * 4)) << 0x10,
+ /* D */ 0,
+ {
+ /* E */ 0x8000,0x8000,
+ /* F */ 0x8000,0x8000
+ }
+ };
+
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&master_mix_scb,
+ dest,"S16_MIX",parent_scb,
+ scb_child_type);
+ return scb;
+}
+
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_mix_to_ostream_scb(struct snd_cs46xx * chip, char * scb_name,
+ u16 mix_buffer_addr, u16 writeback_spb, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_mix2_ostream_scb mix2_ostream_scb = {
+ /* Basic (non scatter/gather) DMA requestor (4 ints) */
+ {
+ DMA_RQ_C1_SOURCE_MOD64 +
+ DMA_RQ_C1_DEST_ON_HOST +
+ DMA_RQ_C1_DEST_MOD1024 +
+ DMA_RQ_C1_WRITEBACK_SRC_FLAG +
+ DMA_RQ_C1_WRITEBACK_DEST_FLAG +
+ 15,
+
+ DMA_RQ_C2_AC_NONE +
+ DMA_RQ_C2_SIGNAL_DEST_PINGPONG +
+
+ CS46XX_DSP_CAPTURE_CHANNEL,
+ DMA_RQ_SD_SP_SAMPLE_ADDR +
+ mix_buffer_addr,
+ 0x0
+ },
+
+ { 0, 0, 0, 0, 0, },
+ 0,0,
+ 0,writeback_spb,
+
+ RSCONFIG_DMA_ENABLE +
+ (19 << RSCONFIG_MAX_DMA_SIZE_SHIFT) +
+
+ ((dest >> 4) << RSCONFIG_STREAM_NUM_SHIFT) +
+ RSCONFIG_DMA_TO_HOST +
+ RSCONFIG_SAMPLE_16STEREO +
+ RSCONFIG_MODULO_64,
+ (mix_buffer_addr + (32 * 4)) << 0x10,
+ 1,0,
+ 0x0001,0x0080,
+ 0xFFFF,0
+ };
+
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&mix2_ostream_scb,
+
+ dest,"S16_MIX_TO_OSTREAM",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_vari_decimate_scb(struct snd_cs46xx * chip,char * scb_name,
+ u16 vari_buffer_addr0,
+ u16 vari_buffer_addr1,
+ u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_vari_decimate_scb vari_decimate_scb = {
+ 0x0028,0x00c8,
+ 0x5555,0x0000,
+ 0x0000,0x0000,
+ vari_buffer_addr0,vari_buffer_addr1,
+
+ 0x0028,0x00c8,
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_256,
+
+ 0xFF800000,
+ 0,
+ 0x0080,vari_buffer_addr1 + (25 * 4),
+
+ 0,0,
+ 0,0,
+
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_8,
+ vari_buffer_addr0 << 0x10,
+ 0x04000000,
+ {
+ 0x8000,0x8000,
+ 0xFFFF,0xFFFF
+ }
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&vari_decimate_scb,
+ dest,"VARIDECIMATE",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+
+static struct dsp_scb_descriptor *
+cs46xx_dsp_create_pcm_serial_input_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest,
+ struct dsp_scb_descriptor * input_scb,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+
+ struct dsp_scb_descriptor * scb;
+
+
+ struct dsp_pcm_serial_input_scb pcm_serial_input_scb = {
+ { 0,
+ 0,
+ 0,
+ 0
+ },
+ {
+ 0,
+ 0,
+ 0,
+ 0,
+ 0
+ },
+
+ 0,0,
+ 0,0,
+
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_16,
+ 0,
+ /* 0xD */ 0,input_scb->address,
+ {
+ /* 0xE */ 0x8000,0x8000,
+ /* 0xF */ 0x8000,0x8000
+ }
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&pcm_serial_input_scb,
+ dest,"PCMSERIALINPUTTASK",parent_scb,
+ scb_child_type);
+ return scb;
+}
+
+
+static struct dsp_scb_descriptor *
+cs46xx_dsp_create_asynch_fg_tx_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest,
+ u16 hfg_scb_address,
+ u16 asynch_buffer_address,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_asynch_fg_tx_scb asynch_fg_tx_scb = {
+ 0xfc00,0x03ff, /* Prototype sample buffer size of 256 dwords */
+ 0x0058,0x0028, /* Min Delta 7 dwords == 28 bytes */
+ /* : Max delta 25 dwords == 100 bytes */
+ 0,hfg_scb_address, /* Point to HFG task SCB */
+ 0,0, /* Initialize current Delta and Consumer ptr adjustment count */
+ 0, /* Initialize accumulated Phi to 0 */
+ 0,0x2aab, /* Const 1/3 */
+
+ {
+ 0, /* Define the unused elements */
+ 0,
+ 0
+ },
+
+ 0,0,
+ 0,dest + AFGTxAccumPhi,
+
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_256, /* Stereo, 256 dword */
+ (asynch_buffer_address) << 0x10, /* This should be automagically synchronized
+ to the producer pointer */
+
+ /* There is no correct initial value, it will depend upon the detected
+ rate etc */
+ 0x18000000, /* Phi increment for approx 32k operation */
+ 0x8000,0x8000, /* Volume controls are unused at this time */
+ 0x8000,0x8000
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&asynch_fg_tx_scb,
+ dest,"ASYNCHFGTXCODE",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_asynch_fg_rx_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest,
+ u16 hfg_scb_address,
+ u16 asynch_buffer_address,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_asynch_fg_rx_scb asynch_fg_rx_scb = {
+ 0xfe00,0x01ff, /* Prototype sample buffer size of 128 dwords */
+ 0x0064,0x001c, /* Min Delta 7 dwords == 28 bytes */
+ /* : Max delta 25 dwords == 100 bytes */
+ 0,hfg_scb_address, /* Point to HFG task SCB */
+ 0,0, /* Initialize current Delta and Consumer ptr adjustment count */
+ {
+ 0, /* Define the unused elements */
+ 0,
+ 0,
+ 0,
+ 0
+ },
+
+ 0,0,
+ 0,dest,
+
+ RSCONFIG_MODULO_128 |
+ RSCONFIG_SAMPLE_16STEREO, /* Stereo, 128 dword */
+ ( (asynch_buffer_address + (16 * 4)) << 0x10), /* This should be automagically
+ synchrinized to the producer pointer */
+
+ /* There is no correct initial value, it will depend upon the detected
+ rate etc */
+ 0x18000000,
+
+ /* Set IEC958 input volume */
+ 0xffff - ins->spdif_input_volume_right,0xffff - ins->spdif_input_volume_left,
+ 0xffff - ins->spdif_input_volume_right,0xffff - ins->spdif_input_volume_left,
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&asynch_fg_rx_scb,
+ dest,"ASYNCHFGRXCODE",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+
+#if 0 /* not used */
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_output_snoop_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest,
+ u16 snoop_buffer_address,
+ struct dsp_scb_descriptor * snoop_scb,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_output_snoop_scb output_snoop_scb = {
+ { 0, /* not used. Zero */
+ 0,
+ 0,
+ 0,
+ },
+ {
+ 0, /* not used. Zero */
+ 0,
+ 0,
+ 0,
+ 0
+ },
+
+ 0,0,
+ 0,0,
+
+ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_64,
+ snoop_buffer_address << 0x10,
+ 0,0,
+ 0,
+ 0,snoop_scb->address
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&output_snoop_scb,
+ dest,"OUTPUTSNOOP",parent_scb,
+ scb_child_type);
+ return scb;
+}
+#endif /* not used */
+
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_spio_write_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_spio_write_scb spio_write_scb = {
+ 0,0, /* SPIOWAddress2:SPIOWAddress1; */
+ 0, /* SPIOWData1; */
+ 0, /* SPIOWData2; */
+ 0,0, /* SPIOWAddress4:SPIOWAddress3; */
+ 0, /* SPIOWData3; */
+ 0, /* SPIOWData4; */
+ 0,0, /* SPIOWDataPtr:Unused1; */
+ { 0,0 }, /* Unused2[2]; */
+
+ 0,0, /* SPIOWChildPtr:SPIOWSiblingPtr; */
+ 0,0, /* SPIOWThisPtr:SPIOWEntryPoint; */
+
+ {
+ 0,
+ 0,
+ 0,
+ 0,
+ 0 /* Unused3[5]; */
+ }
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&spio_write_scb,
+ dest,"SPIOWRITE",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+struct dsp_scb_descriptor *
+cs46xx_dsp_create_magic_snoop_scb(struct snd_cs46xx * chip, char * scb_name, u32 dest,
+ u16 snoop_buffer_address,
+ struct dsp_scb_descriptor * snoop_scb,
+ struct dsp_scb_descriptor * parent_scb,
+ int scb_child_type)
+{
+ struct dsp_scb_descriptor * scb;
+
+ struct dsp_magic_snoop_task magic_snoop_scb = {
+ /* 0 */ 0, /* i0 */
+ /* 1 */ 0, /* i1 */
+ /* 2 */ snoop_buffer_address << 0x10,
+ /* 3 */ 0,snoop_scb->address,
+ /* 4 */ 0, /* i3 */
+ /* 5 */ 0, /* i4 */
+ /* 6 */ 0, /* i5 */
+ /* 7 */ 0, /* i6 */
+ /* 8 */ 0, /* i7 */
+ /* 9 */ 0,0, /* next_scb, sub_list_ptr */
+ /* A */ 0,0, /* entry_point, this_ptr */
+ /* B */ RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_64,
+ /* C */ snoop_buffer_address << 0x10,
+ /* D */ 0,
+ /* E */ { 0x8000,0x8000,
+ /* F */ 0xffff,0xffff
+ }
+ };
+
+ scb = cs46xx_dsp_create_generic_scb(chip,scb_name,(u32 *)&magic_snoop_scb,
+ dest,"MAGICSNOOPTASK",parent_scb,
+ scb_child_type);
+
+ return scb;
+}
+
+static struct dsp_scb_descriptor *
+find_next_free_scb (struct snd_cs46xx * chip, struct dsp_scb_descriptor * from)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * scb = from;
+
+ while (scb->next_scb_ptr != ins->the_null_scb) {
+ if (snd_BUG_ON(!scb->next_scb_ptr))
+ return NULL;
+
+ scb = scb->next_scb_ptr;
+ }
+
+ return scb;
+}
+
+static u32 pcm_reader_buffer_addr[DSP_MAX_PCM_CHANNELS] = {
+ 0x0600, /* 1 */
+ 0x1500, /* 2 */
+ 0x1580, /* 3 */
+ 0x1600, /* 4 */
+ 0x1680, /* 5 */
+ 0x1700, /* 6 */
+ 0x1780, /* 7 */
+ 0x1800, /* 8 */
+ 0x1880, /* 9 */
+ 0x1900, /* 10 */
+ 0x1980, /* 11 */
+ 0x1A00, /* 12 */
+ 0x1A80, /* 13 */
+ 0x1B00, /* 14 */
+ 0x1B80, /* 15 */
+ 0x1C00, /* 16 */
+ 0x1C80, /* 17 */
+ 0x1D00, /* 18 */
+ 0x1D80, /* 19 */
+ 0x1E00, /* 20 */
+ 0x1E80, /* 21 */
+ 0x1F00, /* 22 */
+ 0x1F80, /* 23 */
+ 0x2000, /* 24 */
+ 0x2080, /* 25 */
+ 0x2100, /* 26 */
+ 0x2180, /* 27 */
+ 0x2200, /* 28 */
+ 0x2280, /* 29 */
+ 0x2300, /* 30 */
+ 0x2380, /* 31 */
+ 0x2400, /* 32 */
+};
+
+static u32 src_output_buffer_addr[DSP_MAX_SRC_NR] = {
+ 0x2B80,
+ 0x2BA0,
+ 0x2BC0,
+ 0x2BE0,
+ 0x2D00,
+ 0x2D20,
+ 0x2D40,
+ 0x2D60,
+ 0x2D80,
+ 0x2DA0,
+ 0x2DC0,
+ 0x2DE0,
+ 0x2E00,
+ 0x2E20
+};
+
+static u32 src_delay_buffer_addr[DSP_MAX_SRC_NR] = {
+ 0x2480,
+ 0x2500,
+ 0x2580,
+ 0x2600,
+ 0x2680,
+ 0x2700,
+ 0x2780,
+ 0x2800,
+ 0x2880,
+ 0x2900,
+ 0x2980,
+ 0x2A00,
+ 0x2A80,
+ 0x2B00
+};
+
+struct dsp_pcm_channel_descriptor *
+cs46xx_dsp_create_pcm_channel (struct snd_cs46xx * chip,
+ u32 sample_rate, void * private_data,
+ u32 hw_dma_addr,
+ int pcm_channel_id)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * src_scb = NULL, * pcm_scb, * mixer_scb = NULL;
+ struct dsp_scb_descriptor * src_parent_scb = NULL;
+
+ /* struct dsp_scb_descriptor * pcm_parent_scb; */
+ char scb_name[DSP_MAX_SCB_NAME];
+ int i, pcm_index = -1, insert_point, src_index = -1, pass_through = 0;
+ unsigned long flags;
+
+ switch (pcm_channel_id) {
+ case DSP_PCM_MAIN_CHANNEL:
+ mixer_scb = ins->master_mix_scb;
+ break;
+ case DSP_PCM_REAR_CHANNEL:
+ mixer_scb = ins->rear_mix_scb;
+ break;
+ case DSP_PCM_CENTER_LFE_CHANNEL:
+ mixer_scb = ins->center_lfe_mix_scb;
+ break;
+ case DSP_PCM_S71_CHANNEL:
+ /* TODO */
+ snd_BUG();
+ break;
+ case DSP_IEC958_CHANNEL:
+ if (snd_BUG_ON(!ins->asynch_tx_scb))
+ return NULL;
+ mixer_scb = ins->asynch_tx_scb;
+
+ /* if sample rate is set to 48khz we pass
+ the Sample Rate Converted (which could
+ alter the raw data stream ...) */
+ if (sample_rate == 48000) {
+ snd_printdd ("IEC958 pass through\n");
+ /* Hack to bypass creating a new SRC */
+ pass_through = 1;
+ }
+ break;
+ default:
+ snd_BUG();
+ return NULL;
+ }
+ /* default sample rate is 44100 */
+ if (!sample_rate) sample_rate = 44100;
+
+ /* search for a already created SRC SCB with the same sample rate */
+ for (i = 0; i < DSP_MAX_PCM_CHANNELS &&
+ (pcm_index == -1 || src_scb == NULL); ++i) {
+
+ /* virtual channel reserved
+ for capture */
+ if (i == CS46XX_DSP_CAPTURE_CHANNEL) continue;
+
+ if (ins->pcm_channels[i].active) {
+ if (!src_scb &&
+ ins->pcm_channels[i].sample_rate == sample_rate &&
+ ins->pcm_channels[i].mixer_scb == mixer_scb) {
+ src_scb = ins->pcm_channels[i].src_scb;
+ ins->pcm_channels[i].src_scb->ref_count ++;
+ src_index = ins->pcm_channels[i].src_slot;
+ }
+ } else if (pcm_index == -1) {
+ pcm_index = i;
+ }
+ }
+
+ if (pcm_index == -1) {
+ snd_printk (KERN_ERR "dsp_spos: no free PCM channel\n");
+ return NULL;
+ }
+
+ if (src_scb == NULL) {
+ if (ins->nsrc_scb >= DSP_MAX_SRC_NR) {
+ snd_printk(KERN_ERR "dsp_spos: to many SRC instances\n!");
+ return NULL;
+ }
+
+ /* find a free slot */
+ for (i = 0; i < DSP_MAX_SRC_NR; ++i) {
+ if (ins->src_scb_slots[i] == 0) {
+ src_index = i;
+ ins->src_scb_slots[i] = 1;
+ break;
+ }
+ }
+ if (snd_BUG_ON(src_index == -1))
+ return NULL;
+
+ /* we need to create a new SRC SCB */
+ if (mixer_scb->sub_list_ptr == ins->the_null_scb) {
+ src_parent_scb = mixer_scb;
+ insert_point = SCB_ON_PARENT_SUBLIST_SCB;
+ } else {
+ src_parent_scb = find_next_free_scb(chip,mixer_scb->sub_list_ptr);
+ insert_point = SCB_ON_PARENT_NEXT_SCB;
+ }
+
+ snprintf (scb_name,DSP_MAX_SCB_NAME,"SrcTask_SCB%d",src_index);
+
+ snd_printdd( "dsp_spos: creating SRC \"%s\"\n",scb_name);
+ src_scb = cs46xx_dsp_create_src_task_scb(chip,scb_name,
+ sample_rate,
+ src_output_buffer_addr[src_index],
+ src_delay_buffer_addr[src_index],
+ /* 0x400 - 0x600 source SCBs */
+ 0x400 + (src_index * 0x10) ,
+ src_parent_scb,
+ insert_point,
+ pass_through);
+
+ if (!src_scb) {
+ snd_printk (KERN_ERR "dsp_spos: failed to create SRCtaskSCB\n");
+ return NULL;
+ }
+
+ /* cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate); */
+
+ ins->nsrc_scb ++;
+ }
+
+
+ snprintf (scb_name,DSP_MAX_SCB_NAME,"PCMReader_SCB%d",pcm_index);
+
+ snd_printdd( "dsp_spos: creating PCM \"%s\" (%d)\n",scb_name,
+ pcm_channel_id);
+
+ pcm_scb = cs46xx_dsp_create_pcm_reader_scb(chip,scb_name,
+ pcm_reader_buffer_addr[pcm_index],
+ /* 0x200 - 400 PCMreader SCBs */
+ (pcm_index * 0x10) + 0x200,
+ pcm_index, /* virtual channel 0-31 */
+ hw_dma_addr, /* pcm hw addr */
+ NULL, /* parent SCB ptr */
+ 0 /* insert point */
+ );
+
+ if (!pcm_scb) {
+ snd_printk (KERN_ERR "dsp_spos: failed to create PCMreaderSCB\n");
+ return NULL;
+ }
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ ins->pcm_channels[pcm_index].sample_rate = sample_rate;
+ ins->pcm_channels[pcm_index].pcm_reader_scb = pcm_scb;
+ ins->pcm_channels[pcm_index].src_scb = src_scb;
+ ins->pcm_channels[pcm_index].unlinked = 1;
+ ins->pcm_channels[pcm_index].private_data = private_data;
+ ins->pcm_channels[pcm_index].src_slot = src_index;
+ ins->pcm_channels[pcm_index].active = 1;
+ ins->pcm_channels[pcm_index].pcm_slot = pcm_index;
+ ins->pcm_channels[pcm_index].mixer_scb = mixer_scb;
+ ins->npcm_channels ++;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ return (ins->pcm_channels + pcm_index);
+}
+
+int cs46xx_dsp_pcm_channel_set_period (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel,
+ int period_size)
+{
+ u32 temp = snd_cs46xx_peek (chip,pcm_channel->pcm_reader_scb->address << 2);
+ temp &= ~DMA_RQ_C1_SOURCE_SIZE_MASK;
+
+ switch (period_size) {
+ case 2048:
+ temp |= DMA_RQ_C1_SOURCE_MOD1024;
+ break;
+ case 1024:
+ temp |= DMA_RQ_C1_SOURCE_MOD512;
+ break;
+ case 512:
+ temp |= DMA_RQ_C1_SOURCE_MOD256;
+ break;
+ case 256:
+ temp |= DMA_RQ_C1_SOURCE_MOD128;
+ break;
+ case 128:
+ temp |= DMA_RQ_C1_SOURCE_MOD64;
+ break;
+ case 64:
+ temp |= DMA_RQ_C1_SOURCE_MOD32;
+ break;
+ case 32:
+ temp |= DMA_RQ_C1_SOURCE_MOD16;
+ break;
+ default:
+ snd_printdd ("period size (%d) not supported by HW\n", period_size);
+ return -EINVAL;
+ }
+
+ snd_cs46xx_poke (chip,pcm_channel->pcm_reader_scb->address << 2,temp);
+
+ return 0;
+}
+
+int cs46xx_dsp_pcm_ostream_set_period (struct snd_cs46xx * chip,
+ int period_size)
+{
+ u32 temp = snd_cs46xx_peek (chip,WRITEBACK_SCB_ADDR << 2);
+ temp &= ~DMA_RQ_C1_DEST_SIZE_MASK;
+
+ switch (period_size) {
+ case 2048:
+ temp |= DMA_RQ_C1_DEST_MOD1024;
+ break;
+ case 1024:
+ temp |= DMA_RQ_C1_DEST_MOD512;
+ break;
+ case 512:
+ temp |= DMA_RQ_C1_DEST_MOD256;
+ break;
+ case 256:
+ temp |= DMA_RQ_C1_DEST_MOD128;
+ break;
+ case 128:
+ temp |= DMA_RQ_C1_DEST_MOD64;
+ break;
+ case 64:
+ temp |= DMA_RQ_C1_DEST_MOD32;
+ break;
+ case 32:
+ temp |= DMA_RQ_C1_DEST_MOD16;
+ break;
+ default:
+ snd_printdd ("period size (%d) not supported by HW\n", period_size);
+ return -EINVAL;
+ }
+
+ snd_cs46xx_poke (chip,WRITEBACK_SCB_ADDR << 2,temp);
+
+ return 0;
+}
+
+void cs46xx_dsp_destroy_pcm_channel (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!pcm_channel->active ||
+ ins->npcm_channels <= 0 ||
+ pcm_channel->src_scb->ref_count <= 0))
+ return;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ pcm_channel->unlinked = 1;
+ pcm_channel->active = 0;
+ pcm_channel->private_data = NULL;
+ pcm_channel->src_scb->ref_count --;
+ ins->npcm_channels --;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ cs46xx_dsp_remove_scb(chip,pcm_channel->pcm_reader_scb);
+
+ if (!pcm_channel->src_scb->ref_count) {
+ cs46xx_dsp_remove_scb(chip,pcm_channel->src_scb);
+
+ if (snd_BUG_ON(pcm_channel->src_slot < 0 ||
+ pcm_channel->src_slot >= DSP_MAX_SRC_NR))
+ return;
+
+ ins->src_scb_slots[pcm_channel->src_slot] = 0;
+ ins->nsrc_scb --;
+ }
+}
+
+int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!pcm_channel->active ||
+ chip->dsp_spos_instance->npcm_channels <= 0))
+ return -EIO;
+
+ spin_lock(&pcm_channel->src_scb->lock);
+
+ if (pcm_channel->unlinked) {
+ spin_unlock(&pcm_channel->src_scb->lock);
+ return -EIO;
+ }
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ pcm_channel->unlinked = 1;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ _dsp_unlink_scb (chip,pcm_channel->pcm_reader_scb);
+
+ spin_unlock(&pcm_channel->src_scb->lock);
+ return 0;
+}
+
+int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip,
+ struct dsp_pcm_channel_descriptor * pcm_channel)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * parent_scb;
+ struct dsp_scb_descriptor * src_scb = pcm_channel->src_scb;
+ unsigned long flags;
+
+ spin_lock(&pcm_channel->src_scb->lock);
+
+ if (pcm_channel->unlinked == 0) {
+ spin_unlock(&pcm_channel->src_scb->lock);
+ return -EIO;
+ }
+
+ parent_scb = src_scb;
+
+ if (src_scb->sub_list_ptr != ins->the_null_scb) {
+ src_scb->sub_list_ptr->parent_scb_ptr = pcm_channel->pcm_reader_scb;
+ pcm_channel->pcm_reader_scb->next_scb_ptr = src_scb->sub_list_ptr;
+ }
+
+ src_scb->sub_list_ptr = pcm_channel->pcm_reader_scb;
+
+ snd_BUG_ON(pcm_channel->pcm_reader_scb->parent_scb_ptr);
+ pcm_channel->pcm_reader_scb->parent_scb_ptr = parent_scb;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ /* update SCB entry in DSP RAM */
+ cs46xx_dsp_spos_update_scb(chip,pcm_channel->pcm_reader_scb);
+
+ /* update parent SCB entry */
+ cs46xx_dsp_spos_update_scb(chip,parent_scb);
+
+ pcm_channel->unlinked = 0;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ spin_unlock(&pcm_channel->src_scb->lock);
+ return 0;
+}
+
+struct dsp_scb_descriptor *
+cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * source,
+ u16 addr, char * scb_name)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * parent;
+ struct dsp_scb_descriptor * pcm_input;
+ int insert_point;
+
+ if (snd_BUG_ON(!ins->record_mixer_scb))
+ return NULL;
+
+ if (ins->record_mixer_scb->sub_list_ptr != ins->the_null_scb) {
+ parent = find_next_free_scb (chip,ins->record_mixer_scb->sub_list_ptr);
+ insert_point = SCB_ON_PARENT_NEXT_SCB;
+ } else {
+ parent = ins->record_mixer_scb;
+ insert_point = SCB_ON_PARENT_SUBLIST_SCB;
+ }
+
+ pcm_input = cs46xx_dsp_create_pcm_serial_input_scb(chip,scb_name,addr,
+ source, parent,
+ insert_point);
+
+ return pcm_input;
+}
+
+int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src)
+{
+ if (snd_BUG_ON(!src->parent_scb_ptr))
+ return -EINVAL;
+
+ /* mute SCB */
+ cs46xx_dsp_scb_set_volume (chip,src,0,0);
+
+ _dsp_unlink_scb (chip,src);
+
+ return 0;
+}
+
+int cs46xx_src_link(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+ struct dsp_scb_descriptor * parent_scb;
+
+ if (snd_BUG_ON(src->parent_scb_ptr))
+ return -EINVAL;
+ if (snd_BUG_ON(!ins->master_mix_scb))
+ return -EINVAL;
+
+ if (ins->master_mix_scb->sub_list_ptr != ins->the_null_scb) {
+ parent_scb = find_next_free_scb (chip,ins->master_mix_scb->sub_list_ptr);
+ parent_scb->next_scb_ptr = src;
+ } else {
+ parent_scb = ins->master_mix_scb;
+ parent_scb->sub_list_ptr = src;
+ }
+
+ src->parent_scb_ptr = parent_scb;
+
+ /* update entry in DSP RAM */
+ cs46xx_dsp_spos_update_scb(chip,parent_scb);
+
+ return 0;
+}
+
+int cs46xx_dsp_enable_spdif_out (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if ( ! (ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) ) {
+ cs46xx_dsp_enable_spdif_hw (chip);
+ }
+
+ /* dont touch anything if SPDIF is open */
+ if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) {
+ /* when cs46xx_iec958_post_close(...) is called it
+ will call this function if necessary depending on
+ this bit */
+ ins->spdif_status_out |= DSP_SPDIF_STATUS_OUTPUT_ENABLED;
+
+ return -EBUSY;
+ }
+
+ if (snd_BUG_ON(ins->asynch_tx_scb))
+ return -EINVAL;
+ if (snd_BUG_ON(ins->master_mix_scb->next_scb_ptr !=
+ ins->the_null_scb))
+ return -EINVAL;
+
+ /* reset output snooper sample buffer pointer */
+ snd_cs46xx_poke (chip, (ins->ref_snoop_scb->address + 2) << 2,
+ (OUTPUT_SNOOP_BUFFER + 0x10) << 0x10 );
+
+ /* The asynch. transfer task */
+ ins->asynch_tx_scb = cs46xx_dsp_create_asynch_fg_tx_scb(chip,"AsynchFGTxSCB",ASYNCTX_SCB_ADDR,
+ SPDIFO_SCB_INST,
+ SPDIFO_IP_OUTPUT_BUFFER1,
+ ins->master_mix_scb,
+ SCB_ON_PARENT_NEXT_SCB);
+ if (!ins->asynch_tx_scb) return -ENOMEM;
+
+ ins->spdif_pcm_input_scb = cs46xx_dsp_create_pcm_serial_input_scb(chip,"PCMSerialInput_II",
+ PCMSERIALINII_SCB_ADDR,
+ ins->ref_snoop_scb,
+ ins->asynch_tx_scb,
+ SCB_ON_PARENT_SUBLIST_SCB);
+
+
+ if (!ins->spdif_pcm_input_scb) return -ENOMEM;
+
+ /* monitor state */
+ ins->spdif_status_out |= DSP_SPDIF_STATUS_OUTPUT_ENABLED;
+
+ return 0;
+}
+
+int cs46xx_dsp_disable_spdif_out (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ /* dont touch anything if SPDIF is open */
+ if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) {
+ ins->spdif_status_out &= ~DSP_SPDIF_STATUS_OUTPUT_ENABLED;
+ return -EBUSY;
+ }
+
+ /* check integrety */
+ if (snd_BUG_ON(!ins->asynch_tx_scb))
+ return -EINVAL;
+ if (snd_BUG_ON(!ins->spdif_pcm_input_scb))
+ return -EINVAL;
+ if (snd_BUG_ON(ins->master_mix_scb->next_scb_ptr != ins->asynch_tx_scb))
+ return -EINVAL;
+ if (snd_BUG_ON(ins->asynch_tx_scb->parent_scb_ptr !=
+ ins->master_mix_scb))
+ return -EINVAL;
+
+ cs46xx_dsp_remove_scb (chip,ins->spdif_pcm_input_scb);
+ cs46xx_dsp_remove_scb (chip,ins->asynch_tx_scb);
+
+ ins->spdif_pcm_input_scb = NULL;
+ ins->asynch_tx_scb = NULL;
+
+ /* clear buffer to prevent any undesired noise */
+ _dsp_clear_sample_buffer(chip,SPDIFO_IP_OUTPUT_BUFFER1,256);
+
+ /* monitor state */
+ ins->spdif_status_out &= ~DSP_SPDIF_STATUS_OUTPUT_ENABLED;
+
+
+ return 0;
+}
+
+int cs46xx_iec958_pre_open (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
+ /* remove AsynchFGTxSCB and and PCMSerialInput_II */
+ cs46xx_dsp_disable_spdif_out (chip);
+
+ /* save state */
+ ins->spdif_status_out |= DSP_SPDIF_STATUS_OUTPUT_ENABLED;
+ }
+
+ /* if not enabled already */
+ if ( !(ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) ) {
+ cs46xx_dsp_enable_spdif_hw (chip);
+ }
+
+ /* Create the asynch. transfer task for playback */
+ ins->asynch_tx_scb = cs46xx_dsp_create_asynch_fg_tx_scb(chip,"AsynchFGTxSCB",ASYNCTX_SCB_ADDR,
+ SPDIFO_SCB_INST,
+ SPDIFO_IP_OUTPUT_BUFFER1,
+ ins->master_mix_scb,
+ SCB_ON_PARENT_NEXT_SCB);
+
+
+ /* set spdif channel status value for streaming */
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_stream);
+
+ ins->spdif_status_out |= DSP_SPDIF_STATUS_PLAYBACK_OPEN;
+
+ return 0;
+}
+
+int cs46xx_iec958_post_close (struct snd_cs46xx *chip)
+{
+ struct dsp_spos_instance * ins = chip->dsp_spos_instance;
+
+ if (snd_BUG_ON(!ins->asynch_tx_scb))
+ return -EINVAL;
+
+ ins->spdif_status_out &= ~DSP_SPDIF_STATUS_PLAYBACK_OPEN;
+
+ /* restore settings */
+ cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
+
+ /* deallocate stuff */
+ if (ins->spdif_pcm_input_scb != NULL) {
+ cs46xx_dsp_remove_scb (chip,ins->spdif_pcm_input_scb);
+ ins->spdif_pcm_input_scb = NULL;
+ }
+
+ cs46xx_dsp_remove_scb (chip,ins->asynch_tx_scb);
+ ins->asynch_tx_scb = NULL;
+
+ /* clear buffer to prevent any undesired noise */
+ _dsp_clear_sample_buffer(chip,SPDIFO_IP_OUTPUT_BUFFER1,256);
+
+ /* restore state */
+ if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
+ cs46xx_dsp_enable_spdif_out (chip);
+ }
+
+ return 0;
+}
diff --git a/sound/pci/cs46xx/imgs/cwc4630.h b/sound/pci/cs46xx/imgs/cwc4630.h
new file mode 100644
index 0000000..37c4f13
--- /dev/null
+++ b/sound/pci/cs46xx/imgs/cwc4630.h
@@ -0,0 +1,320 @@
+/* generated from cwc4630.osp DO NOT MODIFY */
+
+#ifndef __HEADER_cwc4630_H__
+#define __HEADER_cwc4630_H__
+
+static struct dsp_symbol_entry cwc4630_symbols[] = {
+ { 0x0000, "BEGINADDRESS",0x00 },
+ { 0x8000, "EXECCHILD",0x03 },
+ { 0x8001, "EXECCHILD_98",0x03 },
+ { 0x8003, "EXECCHILD_PUSH1IND",0x03 },
+ { 0x8008, "EXECSIBLING",0x03 },
+ { 0x800a, "EXECSIBLING_298",0x03 },
+ { 0x800b, "EXECSIBLING_2IND1",0x03 },
+ { 0x8010, "TIMINGMASTER",0x03 },
+ { 0x804f, "S16_CODECINPUTTASK",0x03 },
+ { 0x805e, "PCMSERIALINPUTTASK",0x03 },
+ { 0x806d, "S16_MIX_TO_OSTREAM",0x03 },
+ { 0x809a, "S16_MIX",0x03 },
+ { 0x80bb, "S16_UPSRC",0x03 },
+ { 0x813b, "MIX3_EXP",0x03 },
+ { 0x8164, "DECIMATEBYPOW2",0x03 },
+ { 0x8197, "VARIDECIMATE",0x03 },
+ { 0x81f2, "_3DINPUTTASK",0x03 },
+ { 0x820a, "_3DPRLGCINPTASK",0x03 },
+ { 0x8227, "_3DSTEREOINPUTTASK",0x03 },
+ { 0x8242, "_3DOUTPUTTASK",0x03 },
+ { 0x82c4, "HRTF_MORPH_TASK",0x03 },
+ { 0x82c6, "WAIT4DATA",0x03 },
+ { 0x82fa, "PROLOGIC",0x03 },
+ { 0x8496, "DECORRELATOR",0x03 },
+ { 0x84a4, "STEREO2MONO",0x03 },
+ { 0x0070, "SPOSCB",0x02 },
+ { 0x0107, "TASKTREETHREAD",0x03 },
+ { 0x013c, "TASKTREEHEADERCODE",0x03 },
+ { 0x0145, "FGTASKTREEHEADERCODE",0x03 },
+ { 0x0169, "NULLALGORITHM",0x03 },
+ { 0x016d, "HFGEXECCHILD",0x03 },
+ { 0x016e, "HFGEXECCHILD_98",0x03 },
+ { 0x0170, "HFGEXECCHILD_PUSH1IND",0x03 },
+ { 0x0173, "HFGEXECSIBLING",0x03 },
+ { 0x0175, "HFGEXECSIBLING_298",0x03 },
+ { 0x0176, "HFGEXECSIBLING_2IND1",0x03 },
+ { 0x0179, "S16_CODECOUTPUTTASK",0x03 },
+ { 0x0194, "#CODE_END",0x00 },
+}; /* cwc4630 symbols */
+
+static u32 cwc4630_code[] = {
+/* BEGINADDRESS */
+/* 0000 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 0002 */ 0x00001705,0x00001400,0x000a411e,0x00001003,
+/* 0004 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 0006 */ 0x00009705,0x00001400,0x000a411e,0x00001003,
+/* 0008 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 000A */ 0x00011705,0x00001400,0x000a411e,0x00001003,
+/* 000C */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 000E */ 0x00019705,0x00001400,0x000a411e,0x00001003,
+/* 0010 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 0012 */ 0x00021705,0x00001400,0x000a411e,0x00001003,
+/* 0014 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 0016 */ 0x00029705,0x00001400,0x000a411e,0x00001003,
+/* 0018 */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 001A */ 0x00031705,0x00001400,0x000a411e,0x00001003,
+/* 001C */ 0x00040730,0x00001002,0x000f619e,0x00001003,
+/* 001E */ 0x00039705,0x00001400,0x000a411e,0x00001003,
+/* 0020 */ 0x000fe19e,0x00001003,0x0009c730,0x00001003,
+/* 0022 */ 0x0008e19c,0x00001003,0x000083c1,0x00093040,
+/* 0024 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
+/* 0026 */ 0x00009705,0x00001400,0x000a211e,0x00001003,
+/* 0028 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
+/* 002A */ 0x00011705,0x00001400,0x000a211e,0x00001003,
+/* 002C */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
+/* 002E */ 0x00019705,0x00001400,0x000a211e,0x00001003,
+/* 0030 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
+/* 0032 */ 0x00021705,0x00001400,0x000a211e,0x00001003,
+/* 0034 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
+/* 0036 */ 0x00029705,0x00001400,0x000a211e,0x00001003,
+/* 0038 */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
+/* 003A */ 0x00031705,0x00001400,0x000a211e,0x00001003,
+/* 003C */ 0x00098730,0x00001002,0x000ee19e,0x00001003,
+/* 003E */ 0x00039705,0x00001400,0x000a211e,0x00001003,
+/* 0040 */ 0x0001a730,0x00001008,0x000e2730,0x00001002,
+/* 0042 */ 0x0000a731,0x00001002,0x0000a731,0x00001002,
+/* 0044 */ 0x0000a731,0x00001002,0x0000a731,0x00001002,
+/* 0046 */ 0x0000a731,0x00001002,0x0000a731,0x00001002,
+/* 0048 */ 0x00000000,0x00000000,0x000f619c,0x00001003,
+/* 004A */ 0x0007f801,0x000c0000,0x00000037,0x00001000,
+/* 004C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 004E */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0050 */ 0x00000000,0x000c0000,0x00000000,0x00000000,
+/* 0052 */ 0x0000373c,0x00001000,0x00000000,0x00000000,
+/* 0054 */ 0x000ee19c,0x00001003,0x0007f801,0x000c0000,
+/* 0056 */ 0x00000037,0x00001000,0x00000000,0x00000000,
+/* 0058 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 005A */ 0x00000000,0x00000000,0x0000273c,0x00001000,
+/* 005C */ 0x00000033,0x00001000,0x000e679e,0x00001003,
+/* 005E */ 0x00007705,0x00001400,0x000ac71e,0x00001003,
+/* 0060 */ 0x00087fc1,0x000c3be0,0x0007f801,0x000c0000,
+/* 0062 */ 0x00000037,0x00001000,0x00000000,0x00000000,
+/* 0064 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0066 */ 0x00000000,0x00000000,0x0000a730,0x00001003,
+/* 0068 */ 0x00000033,0x00001000,0x0007f801,0x000c0000,
+/* 006A */ 0x00000037,0x00001000,0x00000000,0x00000000,
+/* 006C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 006E */ 0x00000000,0x00000000,0x00000000,0x000c0000,
+/* 0070 */ 0x00000032,0x00001000,0x0000273d,0x00001000,
+/* 0072 */ 0x0004a730,0x00001003,0x00000f41,0x00097140,
+/* 0074 */ 0x0000a841,0x0009b240,0x0000a0c1,0x0009f040,
+/* 0076 */ 0x0001c641,0x00093540,0x0001cec1,0x0009b5c0,
+/* 0078 */ 0x00000000,0x00000000,0x0001bf05,0x0003fc40,
+/* 007A */ 0x00002725,0x000aa400,0x00013705,0x00093a00,
+/* 007C */ 0x0000002e,0x0009d6c0,0x0002ef8a,0x00000000,
+/* 007E */ 0x00040630,0x00001004,0x0004ef0a,0x000eb785,
+/* 0080 */ 0x0003fc8a,0x00000000,0x00000000,0x000c70e0,
+/* 0082 */ 0x0007d182,0x0002c640,0x00008630,0x00001004,
+/* 0084 */ 0x000799b8,0x0002c6c0,0x00031705,0x00092240,
+/* 0086 */ 0x00039f05,0x000932c0,0x0003520a,0x00000000,
+/* 0088 */ 0x00070731,0x0000100b,0x00010705,0x000b20c0,
+/* 008A */ 0x00000000,0x000eba44,0x00032108,0x000c60c4,
+/* 008C */ 0x00065208,0x000c2917,0x000486b0,0x00001007,
+/* 008E */ 0x00012f05,0x00036880,0x0002818e,0x000c0000,
+/* 0090 */ 0x0004410a,0x00000000,0x00048630,0x00001007,
+/* 0092 */ 0x00029705,0x000c0000,0x00000000,0x00000000,
+/* 0094 */ 0x00003fc1,0x0003fc40,0x000037c1,0x00091b40,
+/* 0096 */ 0x00003fc1,0x000911c0,0x000037c1,0x000957c0,
+/* 0098 */ 0x00003fc1,0x000951c0,0x000037c1,0x00000000,
+/* 009A */ 0x00003fc1,0x000991c0,0x000037c1,0x00000000,
+/* 009C */ 0x00003fc1,0x0009d1c0,0x000037c1,0x00000000,
+/* 009E */ 0x0001ccc1,0x000915c0,0x0001c441,0x0009d800,
+/* 00A0 */ 0x0009cdc1,0x00091240,0x0001c541,0x00091d00,
+/* 00A2 */ 0x0009cfc1,0x00095240,0x0001c741,0x00095c80,
+/* 00A4 */ 0x000e8ca9,0x00099240,0x000e85ad,0x00095640,
+/* 00A6 */ 0x00069ca9,0x00099d80,0x000e952d,0x00099640,
+/* 00A8 */ 0x000eaca9,0x0009d6c0,0x000ea5ad,0x00091a40,
+/* 00AA */ 0x0006bca9,0x0009de80,0x000eb52d,0x00095a40,
+/* 00AC */ 0x000ecca9,0x00099ac0,0x000ec5ad,0x0009da40,
+/* 00AE */ 0x000edca9,0x0009d300,0x000a6e0a,0x00001000,
+/* 00B0 */ 0x000ed52d,0x00091e40,0x000eeca9,0x00095ec0,
+/* 00B2 */ 0x000ee5ad,0x00099e40,0x0006fca9,0x00002500,
+/* 00B4 */ 0x000fb208,0x000c59a0,0x000ef52d,0x0009de40,
+/* 00B6 */ 0x00068ca9,0x000912c1,0x000683ad,0x00095241,
+/* 00B8 */ 0x00020f05,0x000991c1,0x00000000,0x00000000,
+/* 00BA */ 0x00086f88,0x00001000,0x0009cf81,0x000b5340,
+/* 00BC */ 0x0009c701,0x000b92c0,0x0009de81,0x000bd300,
+/* 00BE */ 0x0009d601,0x000b1700,0x0001fd81,0x000b9d80,
+/* 00C0 */ 0x0009f501,0x000b57c0,0x000a0f81,0x000bd740,
+/* 00C2 */ 0x00020701,0x000b5c80,0x000a1681,0x000b97c0,
+/* 00C4 */ 0x00021601,0x00002500,0x000a0701,0x000b9b40,
+/* 00C6 */ 0x000a0f81,0x000b1bc0,0x00021681,0x00002d00,
+/* 00C8 */ 0x00020f81,0x000bd800,0x000a0701,0x000b5bc0,
+/* 00CA */ 0x00021601,0x00003500,0x000a0f81,0x000b5f40,
+/* 00CC */ 0x000a0701,0x000bdbc0,0x00021681,0x00003d00,
+/* 00CE */ 0x00020f81,0x000b1d00,0x000a0701,0x000b1fc0,
+/* 00D0 */ 0x00021601,0x00020500,0x00020f81,0x000b1341,
+/* 00D2 */ 0x000a0701,0x000b9fc0,0x00021681,0x00020d00,
+/* 00D4 */ 0x00020f81,0x000bde80,0x000a0701,0x000bdfc0,
+/* 00D6 */ 0x00021601,0x00021500,0x00020f81,0x000b9341,
+/* 00D8 */ 0x00020701,0x000b53c1,0x00021681,0x00021d00,
+/* 00DA */ 0x000a0f81,0x000d0380,0x0000b601,0x000b15c0,
+/* 00DC */ 0x00007b01,0x00000000,0x00007b81,0x000bd1c0,
+/* 00DE */ 0x00007b01,0x00000000,0x00007b81,0x000b91c0,
+/* 00E0 */ 0x00007b01,0x000b57c0,0x00007b81,0x000b51c0,
+/* 00E2 */ 0x00007b01,0x000b1b40,0x00007b81,0x000b11c0,
+/* 00E4 */ 0x00087b01,0x000c3dc0,0x0007e488,0x000d7e45,
+/* 00E6 */ 0x00000000,0x000d7a44,0x0007e48a,0x00000000,
+/* 00E8 */ 0x00011f05,0x00084080,0x00000000,0x00000000,
+/* 00EA */ 0x00001705,0x000b3540,0x00008a01,0x000bf040,
+/* 00EC */ 0x00007081,0x000bb5c0,0x00055488,0x00000000,
+/* 00EE */ 0x0000d482,0x0003fc40,0x0003fc88,0x00000000,
+/* 00F0 */ 0x0001e401,0x000b3a00,0x0001ec81,0x000bd6c0,
+/* 00F2 */ 0x0002ef88,0x000e7784,0x00056f08,0x00000000,
+/* 00F4 */ 0x000d86b0,0x00001007,0x00008281,0x000bb240,
+/* 00F6 */ 0x0000b801,0x000b7140,0x00007888,0x00000000,
+/* 00F8 */ 0x0000073c,0x00001000,0x0007f188,0x000c0000,
+/* 00FA */ 0x00000000,0x00000000,0x00055288,0x000c555c,
+/* 00FC */ 0x0005528a,0x000c0000,0x0009fa88,0x000c5d00,
+/* 00FE */ 0x0000fa88,0x00000000,0x00000032,0x00001000,
+/* 0100 */ 0x0000073d,0x00001000,0x0007f188,0x000c0000,
+/* 0102 */ 0x00000000,0x00000000,0x0008c01c,0x00001003,
+/* 0104 */ 0x00002705,0x00001008,0x0008b201,0x000c1392,
+/* 0106 */ 0x0000ba01,0x00000000,
+/* TASKTREETHREAD */
+/* 0107 */ 0x00008731,0x00001400,0x0004c108,0x000fe0c4,
+/* 0109 */ 0x00057488,0x00000000,0x000a6388,0x00001001,
+/* 010B */ 0x0008b334,0x000bc141,0x0003020e,0x00000000,
+/* 010D */ 0x000986b0,0x00001008,0x00003625,0x000c5dfa,
+/* 010F */ 0x000a638a,0x00001001,0x0008020e,0x00001002,
+/* 0111 */ 0x0009a6b0,0x00001008,0x0007f301,0x00000000,
+/* 0113 */ 0x00000000,0x00000000,0x00002725,0x000a8c40,
+/* 0115 */ 0x000000ae,0x00000000,0x000e8630,0x00001008,
+/* 0117 */ 0x00000000,0x000c74e0,0x0007d182,0x0002d640,
+/* 0119 */ 0x000b8630,0x00001008,0x000799b8,0x0002d6c0,
+/* 011B */ 0x0000748a,0x000c3ec5,0x0007420a,0x000c0000,
+/* 011D */ 0x00062208,0x000c4117,0x000a0630,0x00001009,
+/* 011F */ 0x00000000,0x000c0000,0x0001022e,0x00000000,
+/* 0121 */ 0x0006a630,0x00001009,0x00000032,0x00001000,
+/* 0123 */ 0x000ca21c,0x00001003,0x00005a02,0x00000000,
+/* 0125 */ 0x0001a630,0x00001009,0x00000000,0x000c0000,
+/* 0127 */ 0x00000036,0x00001000,0x00000000,0x00000000,
+/* 0129 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 012B */ 0x00000000,0x00000000,0x0003a730,0x00001008,
+/* 012D */ 0x0007f801,0x000c0000,0x00000037,0x00001000,
+/* 012F */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0131 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0133 */ 0x0003a730,0x00001008,0x00000033,0x00001000,
+/* 0135 */ 0x0003a705,0x00001008,0x00007a01,0x000c0000,
+/* 0137 */ 0x000e6288,0x000d550a,0x0006428a,0x00000000,
+/* 0139 */ 0x00090730,0x0000100a,0x00000000,0x000c0000,
+/* 013B */ 0x00000000,0x00000000,
+/* TASKTREEHEADERCODE */
+/* 013C */ 0x0007aab0,0x00034880,0x000a8fb0,0x0000100b,
+/* 013E */ 0x00057488,0x00000000,0x00033b94,0x00081140,
+/* 0140 */ 0x000183ae,0x00000000,0x000a86b0,0x0000100b,
+/* 0142 */ 0x00022f05,0x000c3545,0x0000eb8a,0x00000000,
+/* 0144 */ 0x00042731,0x00001003,
+/* FGTASKTREEHEADERCODE */
+/* 0145 */ 0x0007aab0,0x00034880,0x00078fb0,0x0000100a,
+/* 0147 */ 0x00057488,0x00000000,0x00033b94,0x00081140,
+/* 0149 */ 0x000183ae,0x00000000,0x000b06b0,0x0000100b,
+/* 014B */ 0x00022f05,0x00000000,0x00007401,0x00091140,
+/* 014D */ 0x00048f05,0x000951c0,0x00042731,0x00001003,
+/* 014F */ 0x0000473d,0x00001000,0x000f19b0,0x000bbc47,
+/* 0151 */ 0x00080000,0x000bffc7,0x000fe19e,0x00001003,
+/* 0153 */ 0x00000000,0x00000000,0x0008e19c,0x00001003,
+/* 0155 */ 0x000083c1,0x00093040,0x00000f41,0x00097140,
+/* 0157 */ 0x0000a841,0x0009b240,0x0000a0c1,0x0009f040,
+/* 0159 */ 0x0001c641,0x00093540,0x0001cec1,0x0009b5c0,
+/* 015B */ 0x00000000,0x000fdc44,0x00055208,0x00000000,
+/* 015D */ 0x00010705,0x000a2880,0x0000a23a,0x00093a00,
+/* 015F */ 0x0003fc8a,0x000df6c5,0x0004ef0a,0x000c0000,
+/* 0161 */ 0x00012f05,0x00036880,0x00065308,0x000c2997,
+/* 0163 */ 0x000086b0,0x0000100b,0x0004410a,0x000d40c7,
+/* 0165 */ 0x00000000,0x00000000,0x00088730,0x00001004,
+/* 0167 */ 0x00056f0a,0x000ea105,0x00000000,0x00000000,
+/* NULLALGORITHM */
+/* 0169 */ 0x0000473d,0x00001000,0x000f19b0,0x000bbc47,
+/* 016B */ 0x00080000,0x000bffc7,0x0000273d,0x00001000,
+/* HFGEXECCHILD */
+/* 016D */ 0x00000000,0x000eba44,
+/* HFGEXECCHILD_98 */
+/* 016E */ 0x00048f05,0x0000f440,0x00007401,0x0000f7c0,
+/* HFGEXECCHILD_PUSH1IND */
+/* 0170 */ 0x00000734,0x00001000,0x00010705,0x000a6880,
+/* 0172 */ 0x00006a88,0x000c75c4,
+/* HFGEXECSIBLING */
+/* 0173 */ 0x00000000,0x000e5084,0x00000000,0x000eba44,
+/* HFGEXECSIBLING_298 */
+/* 0175 */ 0x00087401,0x000e4782,
+/* HFGEXECSIBLING_2IND1 */
+/* 0176 */ 0x00000734,0x00001000,0x00010705,0x000a6880,
+/* 0178 */ 0x00006a88,0x000c75c4,
+/* S16_CODECOUTPUTTASK */
+/* 0179 */ 0x0007c108,0x000c0000,0x0007e721,0x000bed40,
+/* 017B */ 0x00005f25,0x000badc0,0x0003ba97,0x000beb80,
+/* 017D */ 0x00065590,0x000b2e00,0x00033217,0x00003ec0,
+/* 017F */ 0x00065590,0x000b8e40,0x0003ed80,0x000491c0,
+/* 0181 */ 0x00073fb0,0x00074c80,0x000583a0,0x0000100c,
+/* 0183 */ 0x000ee388,0x00042970,0x00008301,0x00021ef2,
+/* 0185 */ 0x000b8f14,0x0000000f,0x000c4d8d,0x0000001b,
+/* 0187 */ 0x000d6dc2,0x000e06c6,0x000032ac,0x000c3916,
+/* 0189 */ 0x0004edc2,0x00074c80,0x00078898,0x00001000,
+/* 018B */ 0x00038894,0x00000032,0x000c4d8d,0x00092e1b,
+/* 018D */ 0x000d6dc2,0x000e06c6,0x0004edc2,0x000c1956,
+/* 018F */ 0x0000722c,0x00034a00,0x00041705,0x0009ed40,
+/* 0191 */ 0x00058730,0x00001400,0x000d7488,0x000c3a00,
+/* 0193 */ 0x00048f05,0x00000000
+};
+/* #CODE_END */
+
+static u32 cwc4630_parameter[] = {
+/* 0000 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0004 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0008 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 000C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0010 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0014 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0018 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 001C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0020 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0024 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0028 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 002C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0030 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0034 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0038 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 003C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0040 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0044 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0048 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 004C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0050 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0054 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0058 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 005C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0060 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0064 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0068 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 006C */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0070 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0074 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 0078 */ 0x00000000,0x00000000,0x00000000,0x00000000,
+/* 007C */ 0x00000000,0x00000000,0x00000000,0x00000000
+}; /* #PARAMETER_END */
+
+
+static struct dsp_segment_desc cwc4630_segments[] = {
+ { SEGTYPE_SP_PROGRAM, 0x00000000, 0x00000328, cwc4630_code },
+ { SEGTYPE_SP_PARAMETER, 0x00000000, 0x00000080, cwc4630_parameter },
+};
+
+static struct dsp_module_desc cwc4630_module = {
+ "cwc4630",
+ {
+ 38,
+ cwc4630_symbols
+ },
+ 2,
+ cwc4630_segments,
+};
+
+#endif /* __HEADER_cwc4630_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcasync.h b/sound/pci/cs46xx/imgs/cwcasync.h
new file mode 100644
index 0000000..70e63e1
--- /dev/null
+++ b/sound/pci/cs46xx/imgs/cwcasync.h
@@ -0,0 +1,176 @@
+/* generated from cwcasync.osp DO NOT MODIFY */
+
+#ifndef __HEADER_cwcasync_H__
+#define __HEADER_cwcasync_H__
+
+static struct dsp_symbol_entry cwcasync_symbols[] = {
+ { 0x8000, "EXECCHILD",0x03 },
+ { 0x8001, "EXECCHILD_98",0x03 },
+ { 0x8003, "EXECCHILD_PUSH1IND",0x03 },
+ { 0x8008, "EXECSIBLING",0x03 },
+ { 0x800a, "EXECSIBLING_298",0x03 },
+ { 0x800b, "EXECSIBLING_2IND1",0x03 },
+ { 0x8010, "TIMINGMASTER",0x03 },
+ { 0x804f, "S16_CODECINPUTTASK",0x03 },
+ { 0x805e, "PCMSERIALINPUTTASK",0x03 },
+ { 0x806d, "S16_MIX_TO_OSTREAM",0x03 },
+ { 0x809a, "S16_MIX",0x03 },
+ { 0x80bb, "S16_UPSRC",0x03 },
+ { 0x813b, "MIX3_EXP",0x03 },
+ { 0x8164, "DECIMATEBYPOW2",0x03 },
+ { 0x8197, "VARIDECIMATE",0x03 },
+ { 0x81f2, "_3DINPUTTASK",0x03 },
+ { 0x820a, "_3DPRLGCINPTASK",0x03 },
+ { 0x8227, "_3DSTEREOINPUTTASK",0x03 },
+ { 0x8242, "_3DOUTPUTTASK",0x03 },
+ { 0x82c4, "HRTF_MORPH_TASK",0x03 },
+ { 0x82c6, "WAIT4DATA",0x03 },
+ { 0x82fa, "PROLOGIC",0x03 },
+ { 0x8496, "DECORRELATOR",0x03 },
+ { 0x84a4, "STEREO2MONO",0x03 },
+ { 0x0000, "OVERLAYBEGINADDRESS",0x00 },
+ { 0x0000, "SPIOWRITE",0x03 },
+ { 0x000d, "S16_ASYNCCODECINPUTTASK",0x03 },
+ { 0x0043, "SPDIFITASK",0x03 },
+ { 0x007b, "SPDIFOTASK",0x03 },
+ { 0x0097, "ASYNCHFGTXCODE",0x03 },
+ { 0x00be, "ASYNCHFGRXCODE",0x03 },
+ { 0x00db, "#CODE_END",0x00 },
+}; /* cwcasync symbols */
+
+static u32 cwcasync_code[] = {
+/* OVERLAYBEGINADDRESS */
+/* 0000 */ 0x00002731,0x00001400,0x00003725,0x000a8440,
+/* 0002 */ 0x000000ae,0x00000000,0x00060630,0x00001000,
+/* 0004 */ 0x00000000,0x000c7560,0x00075282,0x0002d640,
+/* 0006 */ 0x00021705,0x00000000,0x00072ab8,0x0002d6c0,
+/* 0008 */ 0x00020630,0x00001000,0x000c74c2,0x000d4b82,
+/* 000A */ 0x000475c2,0x00000000,0x0003430a,0x000c0000,
+/* 000C */ 0x00042730,0x00001400,
+/* S16_ASYNCCODECINPUTTASK */
+/* 000D */ 0x0006a108,0x000cf2c4,0x0004f4c0,0x00000000,
+/* 000F */ 0x000fa418,0x0000101f,0x0005d402,0x0001c500,
+/* 0011 */ 0x000f0630,0x00001000,0x00004418,0x00001380,
+/* 0013 */ 0x000e243d,0x000d394a,0x00049705,0x00000000,
+/* 0015 */ 0x0007d530,0x000b4240,0x000e00f2,0x00001000,
+/* 0017 */ 0x00009134,0x000ca20a,0x00004c90,0x00001000,
+/* 0019 */ 0x0005d705,0x00000000,0x00004f25,0x00098240,
+/* 001B */ 0x00004725,0x00000000,0x0000e48a,0x00000000,
+/* 001D */ 0x00027295,0x0009c2c0,0x0003df25,0x00000000,
+/* 001F */ 0x000e8030,0x00001001,0x0005f718,0x000ac600,
+/* 0021 */ 0x0007cf30,0x000c2a01,0x00082630,0x00001001,
+/* 0023 */ 0x000504a0,0x00001001,0x00029314,0x000bcb80,
+/* 0025 */ 0x0003cf25,0x000b0e00,0x0004f5c0,0x00000000,
+/* 0027 */ 0x00049118,0x000d888a,0x0007dd02,0x000c6efa,
+/* 0029 */ 0x00000000,0x00000000,0x0004f5c0,0x00069c80,
+/* 002B */ 0x0000d402,0x00000000,0x000e8630,0x00001001,
+/* 002D */ 0x00079130,0x00000000,0x00049118,0x00090e00,
+/* 002F */ 0x0006c10a,0x00000000,0x00000000,0x000c0000,
+/* 0031 */ 0x0007cf30,0x00030580,0x00005725,0x00000000,
+/* 0033 */ 0x000d84a0,0x00001001,0x00029314,0x000b4780,
+/* 0035 */ 0x0003cf25,0x000b8600,0x00000000,0x00000000,
+/* 0037 */ 0x00000000,0x000c0000,0x00000000,0x00042c80,
+/* 0039 */ 0x0001dec1,0x000e488c,0x00031114,0x00000000,
+/* 003B */ 0x0004f5c2,0x00000000,0x0003640a,0x00000000,
+/* 003D */ 0x00000000,0x000e5084,0x00000000,0x000eb844,
+/* 003F */ 0x00007001,0x00000000,0x00000734,0x00001000,
+/* 0041 */ 0x00010705,0x000a6880,0x00006a88,0x000c75c4,
+/* SPDIFITASK */
+/* 0043 */ 0x0006a108,0x000cf2c4,0x0004f4c0,0x000d5384,
+/* 0045 */ 0x0007e48a,0x00000000,0x00067718,0x00001000,
+/* 0047 */ 0x0007a418,0x00001000,0x0007221a,0x00000000,
+/* 0049 */ 0x0005d402,0x00014500,0x000b8630,0x00001002,
+/* 004B */ 0x00004418,0x00001780,0x000e243d,0x000d394a,
+/* 004D */ 0x00049705,0x00000000,0x0007d530,0x000b4240,
+/* 004F */ 0x000ac0f2,0x00001002,0x00014414,0x00000000,
+/* 0051 */ 0x00004c90,0x00001000,0x0005d705,0x00000000,
+/* 0053 */ 0x00004f25,0x00098240,0x00004725,0x00000000,
+/* 0055 */ 0x0000e48a,0x00000000,0x00027295,0x0009c2c0,
+/* 0057 */ 0x0007df25,0x00000000,0x000ac030,0x00001003,
+/* 0059 */ 0x0005f718,0x000fe798,0x00029314,0x000bcb80,
+/* 005B */ 0x00000930,0x000b0e00,0x0004f5c0,0x000de204,
+/* 005D */ 0x000884a0,0x00001003,0x0007cf25,0x000e3560,
+/* 005F */ 0x00049118,0x00000000,0x00049118,0x000d888a,
+/* 0061 */ 0x0007dd02,0x000c6efa,0x0000c434,0x00030040,
+/* 0063 */ 0x000fda82,0x000c2312,0x000fdc0e,0x00001001,
+/* 0065 */ 0x00083402,0x000c2b92,0x000706b0,0x00001003,
+/* 0067 */ 0x00075a82,0x00000000,0x0000d625,0x000b0940,
+/* 0069 */ 0x0000840e,0x00001002,0x0000aabc,0x000c511e,
+/* 006B */ 0x00078730,0x00001003,0x0000aaf4,0x000e910a,
+/* 006D */ 0x0004628a,0x00000000,0x00006aca,0x00000000,
+/* 006F */ 0x00000930,0x00000000,0x0004f5c0,0x00069c80,
+/* 0071 */ 0x00046ac0,0x00000000,0x0003c40a,0x000fc898,
+/* 0073 */ 0x00049118,0x00090e00,0x0006c10a,0x00000000,
+/* 0075 */ 0x00000000,0x000e5084,0x00000000,0x000eb844,
+/* 0077 */ 0x00007001,0x00000000,0x00000734,0x00001000,
+/* 0079 */ 0x00010705,0x000a6880,0x00006a88,0x000c75c4,
+/* SPDIFOTASK */
+/* 007B */ 0x0006a108,0x000c0000,0x0004f4c0,0x000c3245,
+/* 007D */ 0x0000a418,0x00001000,0x0003a20a,0x00000000,
+/* 007F */ 0x00004418,0x00001380,0x000e243d,0x000d394a,
+/* 0081 */ 0x000c9705,0x000def92,0x0008c030,0x00001004,
+/* 0083 */ 0x0005f718,0x000fe798,0x00000000,0x000c0000,
+/* 0085 */ 0x00005725,0x00000000,0x000704a0,0x00001004,
+/* 0087 */ 0x00029314,0x000b4780,0x0003cf25,0x000b8600,
+/* 0089 */ 0x00000000,0x00000000,0x00000000,0x000c0000,
+/* 008B */ 0x00000000,0x00042c80,0x0001dec1,0x000e488c,
+/* 008D */ 0x00031114,0x00000000,0x0004f5c2,0x00000000,
+/* 008F */ 0x0004a918,0x00098600,0x0006c28a,0x00000000,
+/* 0091 */ 0x00000000,0x000e5084,0x00000000,0x000eb844,
+/* 0093 */ 0x00007001,0x00000000,0x00000734,0x00001000,
+/* 0095 */ 0x00010705,0x000a6880,0x00006a88,0x000c75c4,
+/* ASYNCHFGTXCODE */
+/* 0097 */ 0x0002a880,0x000b4e40,0x00042214,0x000e5548,
+/* 0099 */ 0x000542bf,0x00000000,0x00000000,0x000481c0,
+/* 009B */ 0x00000000,0x00000000,0x00000000,0x00000030,
+/* 009D */ 0x0000072d,0x000fbf8a,0x00077f94,0x000ea7df,
+/* 009F */ 0x0002ac95,0x000d3145,0x00002731,0x00001400,
+/* 00A1 */ 0x00006288,0x000c71c4,0x00014108,0x000e6044,
+/* 00A3 */ 0x00035408,0x00000000,0x00025418,0x000a0ec0,
+/* 00A5 */ 0x0001443d,0x000ca21e,0x00046595,0x000d730c,
+/* 00A7 */ 0x0006538e,0x00000000,0x00064630,0x00001005,
+/* 00A9 */ 0x000e7b0e,0x000df782,0x000746b0,0x00001005,
+/* 00AB */ 0x00036f05,0x000c0000,0x00043695,0x000d598c,
+/* 00AD */ 0x0005331a,0x000f2185,0x00000000,0x00000000,
+/* 00AF */ 0x000007ae,0x000bdb00,0x00040630,0x00001400,
+/* 00B1 */ 0x0005e708,0x000c0000,0x0007ef30,0x000b1c00,
+/* 00B3 */ 0x000d86a0,0x00001005,0x00066408,0x000c0000,
+/* 00B5 */ 0x00000000,0x00000000,0x00021843,0x00000000,
+/* 00B7 */ 0x00000cac,0x00062c00,0x00001dac,0x00063400,
+/* 00B9 */ 0x00002cac,0x0006cc80,0x000db943,0x000e5ca1,
+/* 00BB */ 0x00000000,0x00000000,0x0006680a,0x000f3205,
+/* 00BD */ 0x00042730,0x00001400,
+/* ASYNCHFGRXCODE */
+/* 00BE */ 0x00014108,0x000f2204,0x00025418,0x000a2ec0,
+/* 00C0 */ 0x00015dbd,0x00038100,0x00015dbc,0x00000000,
+/* 00C2 */ 0x0005e415,0x00034880,0x0001258a,0x000d730c,
+/* 00C4 */ 0x0006538e,0x000baa40,0x00060630,0x00001006,
+/* 00C6 */ 0x00067b0e,0x000ac380,0x0003ef05,0x00000000,
+/* 00C8 */ 0x0000f734,0x0001c300,0x000586b0,0x00001400,
+/* 00CA */ 0x000b6f05,0x000c3a00,0x00048f05,0x00000000,
+/* 00CC */ 0x0005b695,0x0008c380,0x0002058e,0x00000000,
+/* 00CE */ 0x000500b0,0x00001400,0x0002b318,0x000e998d,
+/* 00D0 */ 0x0006430a,0x00000000,0x00000000,0x000ef384,
+/* 00D2 */ 0x00004725,0x000c0000,0x00000000,0x000f3204,
+/* 00D4 */ 0x00004f25,0x000c0000,0x00080000,0x000e5ca1,
+/* 00D6 */ 0x000cb943,0x000e5ca1,0x0004b943,0x00000000,
+/* 00D8 */ 0x00040730,0x00001400,0x000cb943,0x000e5ca1,
+/* 00DA */ 0x0004b943,0x00000000
+};
+/* #CODE_END */
+
+static struct dsp_segment_desc cwcasync_segments[] = {
+ { SEGTYPE_SP_PROGRAM, 0x00000000, 0x000001b6, cwcasync_code },
+};
+
+static struct dsp_module_desc cwcasync_module = {
+ "cwcasync",
+ {
+ 32,
+ cwcasync_symbols
+ },
+ 1,
+ cwcasync_segments,
+};
+
+#endif /* __HEADER_cwcasync_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcbinhack.h b/sound/pci/cs46xx/imgs/cwcbinhack.h
new file mode 100644
index 0000000..f4d9368
--- /dev/null
+++ b/sound/pci/cs46xx/imgs/cwcbinhack.h
@@ -0,0 +1,48 @@
+/* generated by Benny
+ MODIFY ON YOUR OWN RISK */
+
+#ifndef __HEADER_cwcbinhack_H__
+#define __HEADER_cwcbinhack_H__
+
+static struct dsp_symbol_entry cwcbinhack_symbols[] = {
+ { 0x02c8, "OVERLAYBEGINADDRESS",0x00 },
+ { 0x02c8, "MAGICSNOOPTASK",0x03 },
+ { 0x0308, "#CODE_END",0x00 },
+}; /* cwcbinhack symbols */
+
+static u32 cwcbinhack_code[] = {
+ /* 0x02c8 */
+ 0x0007bfb0,0x000bc240,0x00000c2e,0x000c6084, /* 1 */
+ 0x000b8630,0x00001016,0x00006408,0x000efb84, /* 2 */
+ 0x00016008,0x00000000,0x0001c088,0x000c0000, /* 3 */
+ 0x000fc908,0x000e3392,0x0005f488,0x000efb84, /* 4 */
+ 0x0001d402,0x000b2e00,0x0003d418,0x00001000, /* 5 */
+ 0x0008d574,0x000c4293,0x00065625,0x000ea30e, /* 6 */
+ 0x00096c01,0x000c6f92,0x0001a58a,0x000c6085, /* 7 */
+ 0x00002f43,0x00000000,0x000e03a0,0x00001016, /* 8 */
+ 0x0005e608,0x000c0000,0x00000000,0x00000000, /* 9 */
+ 0x000ca108,0x000dcca1,0x00003bac,0x000c3205, /* 10 */
+ 0x00073843,0x00000000,0x00010730,0x00001017, /* 11 */
+ 0x0001600a,0x000c0000,0x00057488,0x00000000, /* 12 */
+ 0x00000000,0x000e5084,0x00000000,0x000eba44, /* 13 */
+ 0x00087401,0x000e4782,0x00000734,0x00001000, /* 14 */
+ 0x00010705,0x000a6880,0x00006a88,0x000c75c4, /* 15 */
+ 0x00000000,0x00000000,0x00000000,0x00000000, /* 16 */
+};
+/* #CODE_END */
+
+static struct dsp_segment_desc cwcbinhack_segments[] = {
+ { SEGTYPE_SP_PROGRAM, 0x00000000, 64, cwcbinhack_code },
+};
+
+static struct dsp_module_desc cwcbinhack_module = {
+ "cwcbinhack",
+ {
+ 3,
+ cwcbinhack_symbols
+ },
+ 1,
+ cwcbinhack_segments,
+};
+
+#endif /* __HEADER_cwcbinhack_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
new file mode 100644
index 0000000..09d24c7
--- /dev/null
+++ b/sound/pci/cs46xx/imgs/cwcdma.asp
@@ -0,0 +1,169 @@
+//
+// Copyright(c) by Benny Sjostrand (benny@hostmobility.com)
+//
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+//
+
+
+//
+// This code runs inside the DSP (cs4610, cs4612, cs4624, or cs4630),
+// to compile it you need a tool named SPASM 3.0 and DSP code owned by
+// Cirrus Logic(R). The SPASM program will generate a object file (cwcdma.osp),
+// the "ospparser" tool will genereate the cwcdma.h file it's included from
+// the cs46xx_lib.c file.
+//
+//
+// The purpose of this code is very simple: make it possible to tranfser
+// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host)
+// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters)
+// task always alters the samples in some how, however it's from 48khz -> 48khz. The
+// alterations are not audible, but AC3 wont work.
+//
+// ...
+// |
+// +---------------+
+// | AsynchFGTxSCB |
+// +---------------+
+// |
+// subListPtr
+// |
+// +--------------+
+// | DMAReader |
+// +--------------+
+// |
+// subListPtr
+// |
+// +-------------+
+// | PCMReader |
+// +-------------+
+// (DMA from host)
+//
+
+struct dmaSCB
+ {
+ long dma_reserved1[3];
+
+ short dma_reserved2:dma_outBufPtr;
+
+ short dma_unused1:dma_unused2;
+
+ long dma_reserved3[4];
+
+ short dma_subListPtr:dma_nextSCB;
+ short dma_SPBptr:dma_entryPoint;
+
+ long dma_strmRsConfig;
+ long dma_strmBufPtr;
+
+ long dma_reserved4;
+
+ VolumeControl s2m_volume;
+ };
+
+#export DMAReader
+void DMAReader()
+{
+ execChild();
+ r2 = r0->dma_subListPtr;
+ r1 = r0->nextSCB;
+
+ rsConfig01 = r2->strmRsConfig;
+ // Load rsConfig for input buffer
+
+ rsDMA01 = r2->basicReq.daw, , tb = Z(0 - rf);
+ // Load rsDMA in case input buffer is a DMA buffer Test to see if there is any data to transfer
+
+ if (tb) goto execSibling_2ind1 after {
+ r5 = rf + (-1);
+ r6 = r1->dma_entryPoint; // r6 = entry point of sibling task
+ r1 = r1->dma_SPBptr, // r1 = pointer to sibling task's SPB
+ , ind = r6; // Load entry point of sibling task
+ }
+
+ rsConfig23 = r0->dma_strmRsConfig;
+ // Load rsConfig for output buffer (never a DMA buffer)
+
+ r4 = r0->dma_outBufPtr;
+
+ rsa0 = r2->strmBufPtr;
+ // rsa0 = input buffer pointer
+
+ for (i = r5; i >= 0; --i)
+ after {
+ rsa2 = r4;
+ // rsa2 = output buffer pointer
+
+ nop;
+ nop;
+ }
+ //*****************************
+ // TODO: cycles to this point *
+ //*****************************
+ {
+ acc0 = (rsd0 = *rsa0++1);
+ // get sample
+
+ nop; // Those "nop"'s are really uggly, but there's
+ nop; // something with DSP's pipelines which I don't
+ nop; // understand, resulting this code to fail without
+ // having those "nop"'s (Benny)
+
+ rsa0?reqDMA = r2;
+ // Trigger DMA transfer on input stream,
+ // if needed to replenish input buffer
+
+ nop;
+ // Yet another magic "nop" to make stuff work
+
+ ,,r98 = acc0 $+>> 0;
+ // store sample in ALU
+
+ nop;
+ // latency on load register.
+ // (this one is understandable)
+
+ *rsa2++1 = r98;
+ // store sample in output buffer
+
+ nop; // The same story
+ nop; // as above again ...
+ nop;
+ }
+ // TODO: cycles per loop iteration
+
+ r2->strmBufPtr = rsa0,, ;
+ // Update the modified buffer pointers
+
+ r4 = rsa2;
+ // Load output pointer position into r4
+
+ r2 = r0->nextSCB;
+ // Sibling task
+
+ goto execSibling_2ind1 // takes 6 cycles
+ after {
+ r98 = r2->thisSPB:entryPoint;
+ // Load child routine entry and data address
+
+ r1 = r9;
+ // r9 is r2->thisSPB
+
+ r0->dma_outBufPtr = r4,,
+ // Store updated output buffer pointer
+
+ ind = r8;
+ // r8 is r2->entryPoint
+ }
+}
diff --git a/sound/pci/cs46xx/imgs/cwcdma.h b/sound/pci/cs46xx/imgs/cwcdma.h
new file mode 100644
index 0000000..7ff0d45
--- /dev/null
+++ b/sound/pci/cs46xx/imgs/cwcdma.h
@@ -0,0 +1,68 @@
+/* generated from cwcdma.osp DO NOT MODIFY */
+
+#ifndef __HEADER_cwcdma_H__
+#define __HEADER_cwcdma_H__
+
+static struct dsp_symbol_entry cwcdma_symbols[] = {
+ { 0x8000, "EXECCHILD",0x03 },
+ { 0x8001, "EXECCHILD_98",0x03 },
+ { 0x8003, "EXECCHILD_PUSH1IND",0x03 },
+ { 0x8008, "EXECSIBLING",0x03 },
+ { 0x800a, "EXECSIBLING_298",0x03 },
+ { 0x800b, "EXECSIBLING_2IND1",0x03 },
+ { 0x8010, "TIMINGMASTER",0x03 },
+ { 0x804f, "S16_CODECINPUTTASK",0x03 },
+ { 0x805e, "PCMSERIALINPUTTASK",0x03 },
+ { 0x806d, "S16_MIX_TO_OSTREAM",0x03 },
+ { 0x809a, "S16_MIX",0x03 },
+ { 0x80bb, "S16_UPSRC",0x03 },
+ { 0x813b, "MIX3_EXP",0x03 },
+ { 0x8164, "DECIMATEBYPOW2",0x03 },
+ { 0x8197, "VARIDECIMATE",0x03 },
+ { 0x81f2, "_3DINPUTTASK",0x03 },
+ { 0x820a, "_3DPRLGCINPTASK",0x03 },
+ { 0x8227, "_3DSTEREOINPUTTASK",0x03 },
+ { 0x8242, "_3DOUTPUTTASK",0x03 },
+ { 0x82c4, "HRTF_MORPH_TASK",0x03 },
+ { 0x82c6, "WAIT4DATA",0x03 },
+ { 0x82fa, "PROLOGIC",0x03 },
+ { 0x8496, "DECORRELATOR",0x03 },
+ { 0x84a4, "STEREO2MONO",0x03 },
+ { 0x0000, "OVERLAYBEGINADDRESS",0x00 },
+ { 0x0000, "DMAREADER",0x03 },
+ { 0x0018, "#CODE_END",0x00 },
+}; /* cwcdma symbols */
+
+static u32 cwcdma_code[] = {
+/* OVERLAYBEGINADDRESS */
+/* 0000 */ 0x00002731,0x00001400,0x0004c108,0x000e5044,
+/* 0002 */ 0x0005f608,0x00000000,0x000007ae,0x000be300,
+/* 0004 */ 0x00058630,0x00001400,0x0007afb0,0x000e9584,
+/* 0006 */ 0x00007301,0x000a9840,0x0005e708,0x000cd104,
+/* 0008 */ 0x00067008,0x00000000,0x000902a0,0x00001000,
+/* 000A */ 0x00012a01,0x000c0000,0x00000000,0x00000000,
+/* 000C */ 0x00021843,0x000c0000,0x00000000,0x000c0000,
+/* 000E */ 0x0000e101,0x000c0000,0x00000cac,0x00000000,
+/* 0010 */ 0x00080000,0x000e5ca1,0x00000000,0x000c0000,
+/* 0012 */ 0x00000000,0x00000000,0x00000000,0x00092c00,
+/* 0014 */ 0x000122c1,0x000e5084,0x00058730,0x00001400,
+/* 0016 */ 0x000d7488,0x000e4782,0x00007401,0x0001c100
+};
+
+/* #CODE_END */
+
+static struct dsp_segment_desc cwcdma_segments[] = {
+ { SEGTYPE_SP_PROGRAM, 0x00000000, 0x00000030, cwcdma_code },
+};
+
+static struct dsp_module_desc cwcdma_module = {
+ "cwcdma",
+ {
+ 27,
+ cwcdma_symbols
+ },
+ 1,
+ cwcdma_segments,
+};
+
+#endif /* __HEADER_cwcdma_H__ */
diff --git a/sound/pci/cs46xx/imgs/cwcsnoop.h b/sound/pci/cs46xx/imgs/cwcsnoop.h
new file mode 100644
index 0000000..6929d0a
--- /dev/null
+++ b/sound/pci/cs46xx/imgs/cwcsnoop.h
@@ -0,0 +1,46 @@
+/* generated from cwcsnoop.osp DO NOT MODIFY */
+
+#ifndef __HEADER_cwcsnoop_H__
+#define __HEADER_cwcsnoop_H__
+
+static struct dsp_symbol_entry cwcsnoop_symbols[] = {
+ { 0x0500, "OVERLAYBEGINADDRESS",0x00 },
+ { 0x0500, "OUTPUTSNOOP",0x03 },
+ { 0x051f, "#CODE_END",0x00 },
+}; /* cwcsnoop symbols */
+
+static u32 cwcsnoop_code[] = {
+/* 0000 */ 0x0007bfb0,0x000b4e40,0x0007c088,0x000c0617,
+/* 0002 */ 0x00049705,0x00000000,0x00080630,0x00001028,
+/* 0004 */ 0x00076408,0x000efb84,0x00066008,0x00000000,
+/* 0006 */ 0x0007c908,0x000c0000,0x00046725,0x000efa44,
+/* 0008 */ 0x0005f708,0x00000000,0x0001d402,0x000b2e00,
+/* 000A */ 0x0003d418,0x00001000,0x0008d574,0x000c4293,
+/* 000C */ 0x00065625,0x000ea30e,0x00096c01,0x000c6f92,
+/* 000E */ 0x0006a58a,0x000f6085,0x00002f43,0x00000000,
+/* 0010 */ 0x000a83a0,0x00001028,0x0005e608,0x000c0000,
+/* 0012 */ 0x00000000,0x00000000,0x000ca108,0x000dcca1,
+/* 0014 */ 0x00003bac,0x000fb205,0x00073843,0x00000000,
+/* 0016 */ 0x000d8730,0x00001028,0x0006600a,0x000c0000,
+/* 0018 */ 0x00057488,0x00000000,0x00000000,0x000e5084,
+/* 001A */ 0x00000000,0x000eba44,0x00087401,0x000e4782,
+/* 001C */ 0x00000734,0x00001000,0x00010705,0x000a6880,
+/* 001E */ 0x00006a88,0x000c75c4
+};
+/* #CODE_END */
+
+static struct dsp_segment_desc cwcsnoop_segments[] = {
+ { SEGTYPE_SP_PROGRAM, 0x00000000, 0x0000003e, cwcsnoop_code },
+};
+
+static struct dsp_module_desc cwcsnoop_module = {
+ "cwcsnoop",
+ {
+ 3,
+ cwcsnoop_symbols
+ },
+ 1,
+ cwcsnoop_segments,
+};
+
+#endif /* __HEADER_cwcsnoop_H__ */
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
new file mode 100644
index 0000000..6dea5b5
--- /dev/null
+++ b/sound/pci/cs5530.c
@@ -0,0 +1,305 @@
+/*
+ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio
+ *
+ * (C) Copyright 2007 Ash Willis <ashwillis@programmer.net>
+ * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
+ *
+ * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did
+ * mess with it a bit. The chip seems to have to have trouble with full duplex
+ * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to
+ * simultaneously play back audio at 16bit 44100kHz, the device actually plays
+ * back in the same format in which it is capturing. By forcing the chip to
+ * always play/capture in 16/44100, we can let alsa-lib convert the samples and
+ * that way we can hack up some full duplex audio.
+ *
+ * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
+ * The older version (VSA1) provides fairly good soundblaster emulation
+ * although there are a couple of bugs: large DMA buffers break record,
+ * and the MPU event handling seems suspect. VSA2 allows the native driver
+ * to control the AC97 audio engine directly and requires a different driver.
+ *
+ * Thanks to National Semiconductor for providing the needed information
+ * on the XpressAudio(tm) internals.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2, or (at your option) any
+ * later version.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * TO DO:
+ * Investigate whether we can portably support Cognac (5520) in the
+ * same manner.
+ */
+
+#include <linux/delay.h>
+#include <linux/moduleparam.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include <sound/sb.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Ash Willis");
+MODULE_DESCRIPTION("CS5530 Audio");
+MODULE_LICENSE("GPL");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+struct snd_cs5530 {
+ struct snd_card *card;
+ struct pci_dev *pci;
+ struct snd_sb *sb;
+ unsigned long pci_base;
+};
+
+static struct pci_device_id snd_cs5530_ids[] = {
+ {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID,
+ PCI_ANY_ID, 0, 0},
+ {0,}
+};
+
+MODULE_DEVICE_TABLE(pci, snd_cs5530_ids);
+
+static int snd_cs5530_free(struct snd_cs5530 *chip)
+{
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_cs5530_dev_free(struct snd_device *device)
+{
+ struct snd_cs5530 *chip = device->device_data;
+ return snd_cs5530_free(chip);
+}
+
+static void __devexit snd_cs5530_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg)
+{
+ outb(reg, io + 4);
+ udelay(20);
+ reg = inb(io + 5);
+ udelay(20);
+ return reg;
+}
+
+static int __devinit snd_cs5530_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct snd_cs5530 **rchip)
+{
+ struct snd_cs5530 *chip;
+ unsigned long sb_base;
+ u8 irq, dma8, dma16 = 0;
+ u16 map;
+ void __iomem *mem;
+ int err;
+
+ static struct snd_device_ops ops = {
+ .dev_free = snd_cs5530_dev_free,
+ };
+ *rchip = NULL;
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+
+ err = pci_request_regions(pci, "CS5530");
+ if (err < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->pci_base = pci_resource_start(pci, 0);
+
+ mem = pci_ioremap_bar(pci, 0);
+ if (mem == NULL) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return -EBUSY;
+ }
+
+ map = readw(mem + 0x18);
+ iounmap(mem);
+
+ /* Map bits
+ 0:1 * 0x20 + 0x200 = sb base
+ 2 sb enable
+ 3 adlib enable
+ 5 MPU enable 0x330
+ 6 MPU enable 0x300
+
+ The other bits may be used internally so must be masked */
+
+ sb_base = 0x220 + 0x20 * (map & 3);
+
+ if (map & (1<<2))
+ printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base);
+ else {
+ printk(KERN_ERR "Could not find XpressAudio!\n");
+ snd_cs5530_free(chip);
+ return -ENODEV;
+ }
+
+ if (map & (1<<5))
+ printk(KERN_INFO "CS5530: MPU at 0x300\n");
+ else if (map & (1<<6))
+ printk(KERN_INFO "CS5530: MPU at 0x330\n");
+
+ irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F;
+ dma8 = snd_cs5530_mixer_read(sb_base, 0x81);
+
+ if (dma8 & 0x20)
+ dma16 = 5;
+ else if (dma8 & 0x40)
+ dma16 = 6;
+ else if (dma8 & 0x80)
+ dma16 = 7;
+ else {
+ printk(KERN_ERR "CS5530: No 16bit DMA enabled\n");
+ snd_cs5530_free(chip);
+ return -ENODEV;
+ }
+
+ if (dma8 & 0x01)
+ dma8 = 0;
+ else if (dma8 & 02)
+ dma8 = 1;
+ else if (dma8 & 0x08)
+ dma8 = 3;
+ else {
+ printk(KERN_ERR "CS5530: No 8bit DMA enabled\n");
+ snd_cs5530_free(chip);
+ return -ENODEV;
+ }
+
+ if (irq & 1)
+ irq = 9;
+ else if (irq & 2)
+ irq = 5;
+ else if (irq & 4)
+ irq = 7;
+ else if (irq & 8)
+ irq = 10;
+ else {
+ printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n");
+ snd_cs5530_free(chip);
+ return -ENODEV;
+ }
+
+ printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8,
+ dma16);
+
+ err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8,
+ dma16, SB_HW_CS5530, &chip->sb);
+ if (err < 0) {
+ printk(KERN_ERR "CS5530: Could not create SoundBlaster\n");
+ snd_cs5530_free(chip);
+ return err;
+ }
+
+ err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm);
+ if (err < 0) {
+ printk(KERN_ERR "CS5530: Could not create PCM\n");
+ snd_cs5530_free(chip);
+ return err;
+ }
+
+ err = snd_sbmixer_new(chip->sb);
+ if (err < 0) {
+ printk(KERN_ERR "CS5530: Could not create Mixer\n");
+ snd_cs5530_free(chip);
+ return err;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ snd_cs5530_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+ *rchip = chip;
+ return 0;
+}
+
+static int __devinit snd_cs5530_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_cs5530 *chip = NULL;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+
+ if (card == NULL)
+ return -ENOMEM;
+
+ err = snd_cs5530_create(card, pci, &chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "CS5530");
+ strcpy(card->shortname, "CS5530 Audio");
+ sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static struct pci_driver driver = {
+ .name = "CS5530_Audio",
+ .id_table = snd_cs5530_ids,
+ .probe = snd_cs5530_probe,
+ .remove = __devexit_p(snd_cs5530_remove),
+};
+
+static int __init alsa_card_cs5530_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_cs5530_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_cs5530_init)
+module_exit(alsa_card_cs5530_exit)
+
diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
new file mode 100644
index 0000000..bb3d57e
--- /dev/null
+++ b/sound/pci/cs5535audio/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for cs5535audio
+#
+
+snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
+snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
new file mode 100644
index 0000000..1d8b160
--- /dev/null
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -0,0 +1,413 @@
+/*
+ * Driver for audio on multifunction CS5535/6 companion device
+ * Copyright (C) Jaya Kumar
+ *
+ * Based on Jaroslav Kysela and Takashi Iwai's examples.
+ * This work was sponsored by CIS(M) Sdn Bhd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/asoundef.h>
+#include "cs5535audio.h"
+
+#define DRIVER_NAME "cs5535audio"
+
+static char *ac97_quirk;
+module_param(ac97_quirk, charp, 0444);
+MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds.");
+
+static struct ac97_quirk ac97_quirks[] __devinitdata = {
+#if 0 /* Not yet confirmed if all 5536 boards are HP only */
+ {
+ .subvendor = PCI_VENDOR_ID_AMD,
+ .subdevice = PCI_DEVICE_ID_AMD_CS5536_AUDIO,
+ .name = "AMD RDK",
+ .type = AC97_TUNE_HP_ONLY
+ },
+#endif
+ {}
+};
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME);
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME);
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME);
+
+static struct pci_device_id snd_cs5535audio_ids[] = {
+ { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) },
+ { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) },
+ {}
+};
+
+MODULE_DEVICE_TABLE(pci, snd_cs5535audio_ids);
+
+static void wait_till_cmd_acked(struct cs5535audio *cs5535au, unsigned long timeout)
+{
+ unsigned int tmp;
+ do {
+ tmp = cs_readl(cs5535au, ACC_CODEC_CNTL);
+ if (!(tmp & CMD_NEW))
+ break;
+ udelay(1);
+ } while (--timeout);
+ if (!timeout)
+ snd_printk(KERN_ERR "Failure writing to cs5535 codec\n");
+}
+
+static unsigned short snd_cs5535audio_codec_read(struct cs5535audio *cs5535au,
+ unsigned short reg)
+{
+ unsigned int regdata;
+ unsigned int timeout;
+ unsigned int val;
+
+ regdata = ((unsigned int) reg) << 24;
+ regdata |= ACC_CODEC_CNTL_RD_CMD;
+ regdata |= CMD_NEW;
+
+ cs_writel(cs5535au, ACC_CODEC_CNTL, regdata);
+ wait_till_cmd_acked(cs5535au, 50);
+
+ timeout = 50;
+ do {
+ val = cs_readl(cs5535au, ACC_CODEC_STATUS);
+ if ((val & STS_NEW) && reg == (val >> 24))
+ break;
+ udelay(1);
+ } while (--timeout);
+ if (!timeout)
+ snd_printk(KERN_ERR "Failure reading codec reg 0x%x,"
+ "Last value=0x%x\n", reg, val);
+
+ return (unsigned short) val;
+}
+
+static void snd_cs5535audio_codec_write(struct cs5535audio *cs5535au,
+ unsigned short reg, unsigned short val)
+{
+ unsigned int regdata;
+
+ regdata = ((unsigned int) reg) << 24;
+ regdata |= val;
+ regdata &= CMD_MASK;
+ regdata |= CMD_NEW;
+ regdata &= ACC_CODEC_CNTL_WR_CMD;
+
+ cs_writel(cs5535au, ACC_CODEC_CNTL, regdata);
+ wait_till_cmd_acked(cs5535au, 50);
+}
+
+static void snd_cs5535audio_ac97_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ struct cs5535audio *cs5535au = ac97->private_data;
+ snd_cs5535audio_codec_write(cs5535au, reg, val);
+}
+
+static unsigned short snd_cs5535audio_ac97_codec_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct cs5535audio *cs5535au = ac97->private_data;
+ return snd_cs5535audio_codec_read(cs5535au, reg);
+}
+
+static int __devinit snd_cs5535audio_mixer(struct cs5535audio *cs5535au)
+{
+ struct snd_card *card = cs5535au->card;
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_cs5535audio_ac97_codec_write,
+ .read = snd_cs5535audio_ac97_codec_read,
+ };
+
+ if ((err = snd_ac97_bus(card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.scaps = AC97_SCAP_AUDIO|AC97_SCAP_SKIP_MODEM;
+ ac97.private_data = cs5535au;
+ ac97.pci = cs5535au->pci;
+
+ if ((err = snd_ac97_mixer(pbus, &ac97, &cs5535au->ac97)) < 0) {
+ snd_printk(KERN_ERR "mixer failed\n");
+ return err;
+ }
+
+ snd_ac97_tune_hardware(cs5535au->ac97, ac97_quirks, ac97_quirk);
+
+ return 0;
+}
+
+static void process_bm0_irq(struct cs5535audio *cs5535au)
+{
+ u8 bm_stat;
+ spin_lock(&cs5535au->reg_lock);
+ bm_stat = cs_readb(cs5535au, ACC_BM0_STATUS);
+ spin_unlock(&cs5535au->reg_lock);
+ if (bm_stat & EOP) {
+ struct cs5535audio_dma *dma;
+ dma = cs5535au->playback_substream->runtime->private_data;
+ snd_pcm_period_elapsed(cs5535au->playback_substream);
+ } else {
+ snd_printk(KERN_ERR "unexpected bm0 irq src, bm_stat=%x\n",
+ bm_stat);
+ }
+}
+
+static void process_bm1_irq(struct cs5535audio *cs5535au)
+{
+ u8 bm_stat;
+ spin_lock(&cs5535au->reg_lock);
+ bm_stat = cs_readb(cs5535au, ACC_BM1_STATUS);
+ spin_unlock(&cs5535au->reg_lock);
+ if (bm_stat & EOP) {
+ struct cs5535audio_dma *dma;
+ dma = cs5535au->capture_substream->runtime->private_data;
+ snd_pcm_period_elapsed(cs5535au->capture_substream);
+ }
+}
+
+static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
+{
+ u16 acc_irq_stat;
+ unsigned char count;
+ struct cs5535audio *cs5535au = dev_id;
+
+ if (cs5535au == NULL)
+ return IRQ_NONE;
+
+ acc_irq_stat = cs_readw(cs5535au, ACC_IRQ_STATUS);
+
+ if (!acc_irq_stat)
+ return IRQ_NONE;
+ for (count = 0; count < 4; count++) {
+ if (acc_irq_stat & (1 << count)) {
+ switch (count) {
+ case IRQ_STS:
+ cs_readl(cs5535au, ACC_GPIO_STATUS);
+ break;
+ case WU_IRQ_STS:
+ cs_readl(cs5535au, ACC_GPIO_STATUS);
+ break;
+ case BM0_IRQ_STS:
+ process_bm0_irq(cs5535au);
+ break;
+ case BM1_IRQ_STS:
+ process_bm1_irq(cs5535au);
+ break;
+ default:
+ snd_printk(KERN_ERR "Unexpected irq src: "
+ "0x%x\n", acc_irq_stat);
+ break;
+ }
+ }
+ }
+ return IRQ_HANDLED;
+}
+
+static int snd_cs5535audio_free(struct cs5535audio *cs5535au)
+{
+ synchronize_irq(cs5535au->irq);
+ pci_set_power_state(cs5535au->pci, 3);
+
+ if (cs5535au->irq >= 0)
+ free_irq(cs5535au->irq, cs5535au);
+
+ pci_release_regions(cs5535au->pci);
+ pci_disable_device(cs5535au->pci);
+ kfree(cs5535au);
+ return 0;
+}
+
+static int snd_cs5535audio_dev_free(struct snd_device *device)
+{
+ struct cs5535audio *cs5535au = device->device_data;
+ return snd_cs5535audio_free(cs5535au);
+}
+
+static int __devinit snd_cs5535audio_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct cs5535audio **rcs5535au)
+{
+ struct cs5535audio *cs5535au;
+
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_cs5535audio_dev_free,
+ };
+
+ *rcs5535au = NULL;
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
+ printk(KERN_WARNING "unable to get 32bit dma\n");
+ err = -ENXIO;
+ goto pcifail;
+ }
+
+ cs5535au = kzalloc(sizeof(*cs5535au), GFP_KERNEL);
+ if (cs5535au == NULL) {
+ err = -ENOMEM;
+ goto pcifail;
+ }
+
+ spin_lock_init(&cs5535au->reg_lock);
+ cs5535au->card = card;
+ cs5535au->pci = pci;
+ cs5535au->irq = -1;
+
+ if ((err = pci_request_regions(pci, "CS5535 Audio")) < 0) {
+ kfree(cs5535au);
+ goto pcifail;
+ }
+
+ cs5535au->port = pci_resource_start(pci, 0);
+
+ if (request_irq(pci->irq, snd_cs5535audio_interrupt,
+ IRQF_SHARED, "CS5535 Audio", cs5535au)) {
+ snd_printk("unable to grab IRQ %d\n", pci->irq);
+ err = -EBUSY;
+ goto sndfail;
+ }
+
+ cs5535au->irq = pci->irq;
+ pci_set_master(pci);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ cs5535au, &ops)) < 0)
+ goto sndfail;
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rcs5535au = cs5535au;
+ return 0;
+
+sndfail: /* leave the device alive, just kill the snd */
+ snd_cs5535audio_free(cs5535au);
+ return err;
+
+pcifail:
+ pci_disable_device(pci);
+ return err;
+}
+
+static int __devinit snd_cs5535audio_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct cs5535audio *cs5535au;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_cs5535audio_create(card, pci, &cs5535au)) < 0)
+ goto probefail_out;
+
+ card->private_data = cs5535au;
+
+ if ((err = snd_cs5535audio_mixer(cs5535au)) < 0)
+ goto probefail_out;
+
+ if ((err = snd_cs5535audio_pcm(cs5535au)) < 0)
+ goto probefail_out;
+
+ strcpy(card->driver, DRIVER_NAME);
+
+ strcpy(card->shortname, "CS5535 Audio");
+ sprintf(card->longname, "%s %s at 0x%lx, irq %i",
+ card->shortname, card->driver,
+ cs5535au->port, cs5535au->irq);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto probefail_out;
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+
+probefail_out:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = DRIVER_NAME,
+ .id_table = snd_cs5535audio_ids,
+ .probe = snd_cs5535audio_probe,
+ .remove = __devexit_p(snd_cs5535audio_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_cs5535audio_suspend,
+ .resume = snd_cs5535audio_resume,
+#endif
+};
+
+static int __init alsa_card_cs5535audio_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_cs5535audio_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_cs5535audio_init)
+module_exit(alsa_card_cs5535audio_exit)
+
+MODULE_AUTHOR("Jaya Kumar");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("CS5535 Audio");
+MODULE_SUPPORTED_DEVICE("CS5535 Audio");
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
new file mode 100644
index 0000000..66bae76
--- /dev/null
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -0,0 +1,101 @@
+#ifndef __SOUND_CS5535AUDIO_H
+#define __SOUND_CS5535AUDIO_H
+
+#define cs_writel(cs5535au, reg, val) outl(val, (cs5535au)->port + reg)
+#define cs_writeb(cs5535au, reg, val) outb(val, (cs5535au)->port + reg)
+#define cs_readl(cs5535au, reg) inl((cs5535au)->port + reg)
+#define cs_readw(cs5535au, reg) inw((cs5535au)->port + reg)
+#define cs_readb(cs5535au, reg) inb((cs5535au)->port + reg)
+
+#define CS5535AUDIO_MAX_DESCRIPTORS 128
+
+/* acc_codec bar0 reg addrs */
+#define ACC_GPIO_STATUS 0x00
+#define ACC_CODEC_STATUS 0x08
+#define ACC_CODEC_CNTL 0x0C
+#define ACC_IRQ_STATUS 0x12
+#define ACC_BM0_CMD 0x20
+#define ACC_BM1_CMD 0x28
+#define ACC_BM0_PRD 0x24
+#define ACC_BM1_PRD 0x2C
+#define ACC_BM0_STATUS 0x21
+#define ACC_BM1_STATUS 0x29
+#define ACC_BM0_PNTR 0x60
+#define ACC_BM1_PNTR 0x64
+
+/* acc_codec bar0 reg bits */
+/* ACC_IRQ_STATUS */
+#define IRQ_STS 0
+#define WU_IRQ_STS 1
+#define BM0_IRQ_STS 2
+#define BM1_IRQ_STS 3
+/* ACC_BMX_STATUS */
+#define EOP (1<<0)
+#define BM_EOP_ERR (1<<1)
+/* ACC_BMX_CTL */
+#define BM_CTL_EN 0x01
+#define BM_CTL_PAUSE 0x03
+#define BM_CTL_DIS 0x00
+#define BM_CTL_BYTE_ORD_LE 0x00
+#define BM_CTL_BYTE_ORD_BE 0x04
+/* cs5535 specific ac97 codec register defines */
+#define CMD_MASK 0xFF00FFFF
+#define CMD_NEW 0x00010000
+#define STS_NEW 0x00020000
+#define PRM_RDY_STS 0x00800000
+#define ACC_CODEC_CNTL_WR_CMD (~0x80000000)
+#define ACC_CODEC_CNTL_RD_CMD 0x80000000
+#define ACC_CODEC_CNTL_LNK_SHUTDOWN 0x00040000
+#define ACC_CODEC_CNTL_LNK_WRM_RST 0x00020000
+#define PRD_JMP 0x2000
+#define PRD_EOP 0x4000
+#define PRD_EOT 0x8000
+
+enum { CS5535AUDIO_DMA_PLAYBACK, CS5535AUDIO_DMA_CAPTURE, NUM_CS5535AUDIO_DMAS };
+
+struct cs5535audio;
+
+struct cs5535audio_dma_ops {
+ int type;
+ void (*enable_dma)(struct cs5535audio *cs5535au);
+ void (*disable_dma)(struct cs5535audio *cs5535au);
+ void (*pause_dma)(struct cs5535audio *cs5535au);
+ void (*setup_prd)(struct cs5535audio *cs5535au, u32 prd_addr);
+ u32 (*read_prd)(struct cs5535audio *cs5535au);
+ u32 (*read_dma_pntr)(struct cs5535audio *cs5535au);
+};
+
+struct cs5535audio_dma_desc {
+ u32 addr;
+ u16 size;
+ u16 ctlreserved;
+};
+
+struct cs5535audio_dma {
+ const struct cs5535audio_dma_ops *ops;
+ struct snd_dma_buffer desc_buf;
+ struct snd_pcm_substream *substream;
+ unsigned int buf_addr, buf_bytes;
+ unsigned int period_bytes, periods;
+ u32 saved_prd;
+};
+
+struct cs5535audio {
+ struct snd_card *card;
+ struct snd_ac97 *ac97;
+ struct snd_pcm *pcm;
+ int irq;
+ struct pci_dev *pci;
+ unsigned long port;
+ spinlock_t reg_lock;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+ struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS];
+};
+
+int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
+int snd_cs5535audio_resume(struct pci_dev *pci);
+int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535audio);
+
+#endif /* __SOUND_CS5535AUDIO_H */
+
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
new file mode 100644
index 0000000..cdcda87
--- /dev/null
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -0,0 +1,440 @@
+/*
+ * Driver for audio on multifunction CS5535 companion device
+ * Copyright (C) Jaya Kumar
+ *
+ * Based on Jaroslav Kysela and Takashi Iwai's examples.
+ * This work was sponsored by CIS(M) Sdn Bhd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * todo: add be fmt support, spdif, pm
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/asoundef.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/ac97_codec.h>
+#include "cs5535audio.h"
+
+static struct snd_pcm_hardware snd_cs5535audio_playback =
+{
+ .info = (
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME
+ ),
+ .formats = (
+ SNDRV_PCM_FMTBIT_S16_LE
+ ),
+ .rates = (
+ SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_48000
+ ),
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (64*1024 - 16),
+ .periods_min = 1,
+ .periods_max = CS5535AUDIO_MAX_DESCRIPTORS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_cs5535audio_capture =
+{
+ .info = (
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID
+ ),
+ .formats = (
+ SNDRV_PCM_FMTBIT_S16_LE
+ ),
+ .rates = (
+ SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_48000
+ ),
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (64*1024 - 16),
+ .periods_min = 1,
+ .periods_max = CS5535AUDIO_MAX_DESCRIPTORS,
+ .fifo_size = 0,
+};
+
+static int snd_cs5535audio_playback_open(struct snd_pcm_substream *substream)
+{
+ int err;
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_cs5535audio_playback;
+ runtime->hw.rates = cs5535au->ac97->rates[AC97_RATES_FRONT_DAC];
+ snd_pcm_limit_hw_rates(runtime);
+ cs5535au->playback_substream = substream;
+ runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_PLAYBACK]);
+ if ((err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_cs5535audio_playback_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+#define CS5535AUDIO_DESC_LIST_SIZE \
+ PAGE_ALIGN(CS5535AUDIO_MAX_DESCRIPTORS * sizeof(struct cs5535audio_dma_desc))
+
+static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
+ struct cs5535audio_dma *dma,
+ struct snd_pcm_substream *substream,
+ unsigned int periods,
+ unsigned int period_bytes)
+{
+ unsigned int i;
+ u32 addr, desc_addr, jmpprd_addr;
+ struct cs5535audio_dma_desc *lastdesc;
+
+ if (periods > CS5535AUDIO_MAX_DESCRIPTORS)
+ return -ENOMEM;
+
+ if (dma->desc_buf.area == NULL) {
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(cs5535au->pci),
+ CS5535AUDIO_DESC_LIST_SIZE+1,
+ &dma->desc_buf) < 0)
+ return -ENOMEM;
+ dma->period_bytes = dma->periods = 0;
+ }
+
+ if (dma->periods == periods && dma->period_bytes == period_bytes)
+ return 0;
+
+ /* the u32 cast is okay because in snd*create we successfully told
+ pci alloc that we're only 32 bit capable so the uppper will be 0 */
+ addr = (u32) substream->runtime->dma_addr;
+ desc_addr = (u32) dma->desc_buf.addr;
+ for (i = 0; i < periods; i++) {
+ struct cs5535audio_dma_desc *desc =
+ &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i];
+ desc->addr = cpu_to_le32(addr);
+ desc->size = cpu_to_le32(period_bytes);
+ desc->ctlreserved = cpu_to_le32(PRD_EOP);
+ desc_addr += sizeof(struct cs5535audio_dma_desc);
+ addr += period_bytes;
+ }
+ /* we reserved one dummy descriptor at the end to do the PRD jump */
+ lastdesc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[periods];
+ lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr);
+ lastdesc->size = 0;
+ lastdesc->ctlreserved = cpu_to_le32(PRD_JMP);
+ jmpprd_addr = cpu_to_le32(lastdesc->addr +
+ (sizeof(struct cs5535audio_dma_desc)*periods));
+
+ dma->substream = substream;
+ dma->period_bytes = period_bytes;
+ dma->periods = periods;
+ spin_lock_irq(&cs5535au->reg_lock);
+ dma->ops->disable_dma(cs5535au);
+ dma->ops->setup_prd(cs5535au, jmpprd_addr);
+ spin_unlock_irq(&cs5535au->reg_lock);
+ return 0;
+}
+
+static void cs5535audio_playback_enable_dma(struct cs5535audio *cs5535au)
+{
+ cs_writeb(cs5535au, ACC_BM0_CMD, BM_CTL_EN);
+}
+
+static void cs5535audio_playback_disable_dma(struct cs5535audio *cs5535au)
+{
+ cs_writeb(cs5535au, ACC_BM0_CMD, 0);
+}
+
+static void cs5535audio_playback_pause_dma(struct cs5535audio *cs5535au)
+{
+ cs_writeb(cs5535au, ACC_BM0_CMD, BM_CTL_PAUSE);
+}
+
+static void cs5535audio_playback_setup_prd(struct cs5535audio *cs5535au,
+ u32 prd_addr)
+{
+ cs_writel(cs5535au, ACC_BM0_PRD, prd_addr);
+}
+
+static u32 cs5535audio_playback_read_prd(struct cs5535audio *cs5535au)
+{
+ return cs_readl(cs5535au, ACC_BM0_PRD);
+}
+
+static u32 cs5535audio_playback_read_dma_pntr(struct cs5535audio *cs5535au)
+{
+ return cs_readl(cs5535au, ACC_BM0_PNTR);
+}
+
+static void cs5535audio_capture_enable_dma(struct cs5535audio *cs5535au)
+{
+ cs_writeb(cs5535au, ACC_BM1_CMD, BM_CTL_EN);
+}
+
+static void cs5535audio_capture_disable_dma(struct cs5535audio *cs5535au)
+{
+ cs_writeb(cs5535au, ACC_BM1_CMD, 0);
+}
+
+static void cs5535audio_capture_pause_dma(struct cs5535audio *cs5535au)
+{
+ cs_writeb(cs5535au, ACC_BM1_CMD, BM_CTL_PAUSE);
+}
+
+static void cs5535audio_capture_setup_prd(struct cs5535audio *cs5535au,
+ u32 prd_addr)
+{
+ cs_writel(cs5535au, ACC_BM1_PRD, prd_addr);
+}
+
+static u32 cs5535audio_capture_read_prd(struct cs5535audio *cs5535au)
+{
+ return cs_readl(cs5535au, ACC_BM1_PRD);
+}
+
+static u32 cs5535audio_capture_read_dma_pntr(struct cs5535audio *cs5535au)
+{
+ return cs_readl(cs5535au, ACC_BM1_PNTR);
+}
+
+static void cs5535audio_clear_dma_packets(struct cs5535audio *cs5535au,
+ struct cs5535audio_dma *dma,
+ struct snd_pcm_substream *substream)
+{
+ snd_dma_free_pages(&dma->desc_buf);
+ dma->desc_buf.area = NULL;
+ dma->substream = NULL;
+}
+
+static int snd_cs5535audio_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ struct cs5535audio_dma *dma = substream->runtime->private_data;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ dma->buf_addr = substream->runtime->dma_addr;
+ dma->buf_bytes = params_buffer_bytes(hw_params);
+
+ err = cs5535audio_build_dma_packets(cs5535au, dma, substream,
+ params_periods(hw_params),
+ params_period_bytes(hw_params));
+ return err;
+}
+
+static int snd_cs5535audio_hw_free(struct snd_pcm_substream *substream)
+{
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ struct cs5535audio_dma *dma = substream->runtime->private_data;
+
+ cs5535audio_clear_dma_packets(cs5535au, dma, substream);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_cs5535audio_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ return snd_ac97_set_rate(cs5535au->ac97, AC97_PCM_FRONT_DAC_RATE,
+ substream->runtime->rate);
+}
+
+static int snd_cs5535audio_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ struct cs5535audio_dma *dma = substream->runtime->private_data;
+ int err = 0;
+
+ spin_lock(&cs5535au->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dma->ops->pause_dma(cs5535au);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dma->ops->enable_dma(cs5535au);
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ dma->ops->enable_dma(cs5535au);
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ dma->ops->enable_dma(cs5535au);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ dma->ops->disable_dma(cs5535au);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ dma->ops->disable_dma(cs5535au);
+ break;
+ default:
+ snd_printk(KERN_ERR "unhandled trigger\n");
+ err = -EINVAL;
+ break;
+ }
+ spin_unlock(&cs5535au->reg_lock);
+ return err;
+}
+
+static snd_pcm_uframes_t snd_cs5535audio_pcm_pointer(struct snd_pcm_substream
+ *substream)
+{
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ u32 curdma;
+ struct cs5535audio_dma *dma;
+
+ dma = substream->runtime->private_data;
+ curdma = dma->ops->read_dma_pntr(cs5535au);
+ if (curdma < dma->buf_addr) {
+ snd_printk(KERN_ERR "curdma=%x < %x bufaddr.\n",
+ curdma, dma->buf_addr);
+ return 0;
+ }
+ curdma -= dma->buf_addr;
+ if (curdma >= dma->buf_bytes) {
+ snd_printk(KERN_ERR "diff=%x >= %x buf_bytes.\n",
+ curdma, dma->buf_bytes);
+ return 0;
+ }
+ return bytes_to_frames(substream->runtime, curdma);
+}
+
+static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream)
+{
+ int err;
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_cs5535audio_capture;
+ runtime->hw.rates = cs5535au->ac97->rates[AC97_RATES_ADC];
+ snd_pcm_limit_hw_rates(runtime);
+ cs5535au->capture_substream = substream;
+ runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_CAPTURE]);
+ if ((err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_cs5535audio_capture_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static int snd_cs5535audio_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct cs5535audio *cs5535au = snd_pcm_substream_chip(substream);
+ return snd_ac97_set_rate(cs5535au->ac97, AC97_PCM_LR_ADC_RATE,
+ substream->runtime->rate);
+}
+
+static struct snd_pcm_ops snd_cs5535audio_playback_ops = {
+ .open = snd_cs5535audio_playback_open,
+ .close = snd_cs5535audio_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs5535audio_hw_params,
+ .hw_free = snd_cs5535audio_hw_free,
+ .prepare = snd_cs5535audio_playback_prepare,
+ .trigger = snd_cs5535audio_trigger,
+ .pointer = snd_cs5535audio_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_cs5535audio_capture_ops = {
+ .open = snd_cs5535audio_capture_open,
+ .close = snd_cs5535audio_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs5535audio_hw_params,
+ .hw_free = snd_cs5535audio_hw_free,
+ .prepare = snd_cs5535audio_capture_prepare,
+ .trigger = snd_cs5535audio_trigger,
+ .pointer = snd_cs5535audio_pcm_pointer,
+};
+
+static struct cs5535audio_dma_ops snd_cs5535audio_playback_dma_ops = {
+ .type = CS5535AUDIO_DMA_PLAYBACK,
+ .enable_dma = cs5535audio_playback_enable_dma,
+ .disable_dma = cs5535audio_playback_disable_dma,
+ .setup_prd = cs5535audio_playback_setup_prd,
+ .read_prd = cs5535audio_playback_read_prd,
+ .pause_dma = cs5535audio_playback_pause_dma,
+ .read_dma_pntr = cs5535audio_playback_read_dma_pntr,
+};
+
+static struct cs5535audio_dma_ops snd_cs5535audio_capture_dma_ops = {
+ .type = CS5535AUDIO_DMA_CAPTURE,
+ .enable_dma = cs5535audio_capture_enable_dma,
+ .disable_dma = cs5535audio_capture_disable_dma,
+ .setup_prd = cs5535audio_capture_setup_prd,
+ .read_prd = cs5535audio_capture_read_prd,
+ .pause_dma = cs5535audio_capture_pause_dma,
+ .read_dma_pntr = cs5535audio_capture_read_dma_pntr,
+};
+
+int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535au)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(cs5535au->card, "CS5535 Audio", 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ cs5535au->dmas[CS5535AUDIO_DMA_PLAYBACK].ops =
+ &snd_cs5535audio_playback_dma_ops;
+ cs5535au->dmas[CS5535AUDIO_DMA_CAPTURE].ops =
+ &snd_cs5535audio_capture_dma_ops;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_cs5535audio_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_cs5535audio_capture_ops);
+
+ pcm->private_data = cs5535au;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "CS5535 Audio");
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(cs5535au->pci),
+ 64*1024, 128*1024);
+ cs5535au->pcm = pcm;
+
+ return 0;
+}
+
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
new file mode 100644
index 0000000..564c33b
--- /dev/null
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -0,0 +1,137 @@
+/*
+ * Power management for audio on multifunction CS5535 companion device
+ * Copyright (C) Jaya Kumar
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/asoundef.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include "cs5535audio.h"
+
+static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au)
+{
+ /*
+ we depend on snd_ac97_suspend to tell the
+ AC97 codec to shutdown. the amd spec suggests
+ that the LNK_SHUTDOWN be done at the same time
+ that the codec power-down is issued. instead,
+ we do it just after rather than at the same
+ time. excluding codec specific build_ops->suspend
+ ac97 powerdown hits:
+ 0x8000 EAPD
+ 0x4000 Headphone amplifier
+ 0x0300 ADC & DAC
+ 0x0400 Analog Mixer powerdown (Vref on)
+ I am not sure if this is the best that we can do.
+ The remainder to be investigated are:
+ - analog mixer (vref off) 0x0800
+ - AC-link powerdown 0x1000
+ - codec internal clock 0x2000
+ */
+
+ /* set LNK_SHUTDOWN to shutdown AC link */
+ cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_SHUTDOWN);
+
+}
+
+int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct cs5535audio *cs5535au = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(cs5535au->pcm);
+ snd_ac97_suspend(cs5535au->ac97);
+ for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
+ struct cs5535audio_dma *dma = &cs5535au->dmas[i];
+ if (dma && dma->substream)
+ dma->saved_prd = dma->ops->read_prd(cs5535au);
+ }
+ /* save important regs, then disable aclink in hw */
+ snd_cs5535audio_stop_hardware(cs5535au);
+
+ if (pci_save_state(pci)) {
+ printk(KERN_ERR "cs5535audio: pci_save_state failed!\n");
+ return -EIO;
+ }
+ pci_disable_device(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+int snd_cs5535audio_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct cs5535audio *cs5535au = card->private_data;
+ u32 tmp;
+ int timeout;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ if (pci_restore_state(pci) < 0) {
+ printk(KERN_ERR "cs5535audio: pci_restore_state failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "cs5535audio: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ /* set LNK_WRM_RST to reset AC link */
+ cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_WRM_RST);
+
+ timeout = 50;
+ do {
+ tmp = cs_readl(cs5535au, ACC_CODEC_STATUS);
+ if (tmp & PRM_RDY_STS)
+ break;
+ udelay(1);
+ } while (--timeout);
+
+ if (!timeout)
+ snd_printk(KERN_ERR "Failure getting AC Link ready\n");
+
+ /* set up rate regs, dma. actual initiation is done in trig */
+ for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
+ struct cs5535audio_dma *dma = &cs5535au->dmas[i];
+ if (dma && dma->substream) {
+ dma->substream->ops->prepare(dma->substream);
+ dma->ops->setup_prd(cs5535au, dma->saved_prd);
+ }
+ }
+
+ /* we depend on ac97 to perform the codec power up */
+ snd_ac97_resume(cs5535au->ac97);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+ return 0;
+}
+
diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile
new file mode 100644
index 0000000..7b576ae
--- /dev/null
+++ b/sound/pci/echoaudio/Makefile
@@ -0,0 +1,30 @@
+#
+# Makefile for ALSA Echoaudio soundcard drivers
+# Copyright (c) 2003 by Giuliano Pochini <pochini@shiny.it>
+#
+
+snd-darla20-objs := darla20.o
+snd-gina20-objs := gina20.o
+snd-layla20-objs := layla20.o
+snd-darla24-objs := darla24.o
+snd-gina24-objs := gina24.o
+snd-layla24-objs := layla24.o
+snd-mona-objs := mona.o
+snd-mia-objs := mia.o
+snd-echo3g-objs := echo3g.o
+snd-indigo-objs := indigo.o
+snd-indigoio-objs := indigoio.o
+snd-indigodj-objs := indigodj.o
+
+obj-$(CONFIG_SND_DARLA20) += snd-darla20.o
+obj-$(CONFIG_SND_GINA20) += snd-gina20.o
+obj-$(CONFIG_SND_LAYLA20) += snd-layla20.o
+obj-$(CONFIG_SND_DARLA24) += snd-darla24.o
+obj-$(CONFIG_SND_GINA24) += snd-gina24.o
+obj-$(CONFIG_SND_LAYLA24) += snd-layla24.o
+obj-$(CONFIG_SND_MONA) += snd-mona.o
+obj-$(CONFIG_SND_MIA) += snd-mia.o
+obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o
+obj-$(CONFIG_SND_INDIGO) += snd-indigo.o
+obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o
+obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o
diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c
new file mode 100644
index 0000000..8c6db3a
--- /dev/null
+++ b/sound/pci/echoaudio/darla20.c
@@ -0,0 +1,101 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_DARLA20
+#define ECHOCARD_NAME "Darla20"
+#define ECHOCARD_HAS_MONITOR
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 0 */
+#define PX_ANALOG_IN 8 /* 2 */
+#define PX_DIGITAL_IN 10 /* 0 */
+#define PX_NUM 10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 8 */
+#define BX_DIGITAL_OUT 8 /* 0 */
+#define BX_ANALOG_IN 8 /* 2 */
+#define BX_DIGITAL_IN 10 /* 0 */
+#define BX_NUM 10
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/darla20_dsp.fw");
+
+#define FW_DARLA20_DSP 0
+
+static const struct firmware card_fw[] = {
+ {0, "darla20_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "darla20_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c
new file mode 100644
index 0000000..2904330
--- /dev/null
+++ b/sound/pci/echoaudio/darla20_dsp.c
@@ -0,0 +1,126 @@
+/***************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Darla20\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP];
+ chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
+ chip->clock_state = GD_CLOCK_UNDEF;
+ /* Since this card has no ASIC, mark it as loaded so everything
+ works OK */
+ chip->asic_loaded = TRUE;
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+/* The Darla20 has no external clock sources */
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The Darla20 has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u8 clock_state, spdif_status;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ switch (rate) {
+ case 44100:
+ clock_state = GD_CLOCK_44;
+ spdif_status = GD_SPDIF_STATUS_44;
+ break;
+ case 48000:
+ clock_state = GD_CLOCK_48;
+ spdif_status = GD_SPDIF_STATUS_48;
+ break;
+ default:
+ clock_state = GD_CLOCK_NOCHANGE;
+ spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+ break;
+ }
+
+ if (chip->clock_state == clock_state)
+ clock_state = GD_CLOCK_NOCHANGE;
+ if (spdif_status == chip->spdif_status)
+ spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ chip->comm_page->gd_clock_state = clock_state;
+ chip->comm_page->gd_spdif_status = spdif_status;
+ chip->comm_page->gd_resampler_state = 3; /* magic number - should always be 3 */
+
+ /* Save the new audio state if it changed */
+ if (clock_state != GD_CLOCK_NOCHANGE)
+ chip->clock_state = clock_state;
+ if (spdif_status != GD_SPDIF_STATUS_NOCHANGE)
+ chip->spdif_status = spdif_status;
+ chip->sample_rate = rate;
+
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+}
diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c
new file mode 100644
index 0000000..04cbf3e
--- /dev/null
+++ b/sound/pci/echoaudio/darla24.c
@@ -0,0 +1,108 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_DARLA24
+#define ECHOCARD_NAME "Darla24"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 0 */
+#define PX_ANALOG_IN 8 /* 2 */
+#define PX_DIGITAL_IN 10 /* 0 */
+#define PX_NUM 10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 8 */
+#define BX_DIGITAL_OUT 8 /* 0 */
+#define BX_ANALOG_IN 8 /* 2 */
+#define BX_DIGITAL_IN 10 /* 0 */
+#define BX_NUM 10
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/darla24_dsp.fw");
+
+#define FW_DARLA24_DSP 0
+
+static const struct firmware card_fw[] = {
+ {0, "darla24_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */
+ {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "darla24_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c
new file mode 100644
index 0000000..6022873
--- /dev/null
+++ b/sound/pci/echoaudio/darla24_dsp.c
@@ -0,0 +1,158 @@
+/***************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Darla24\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP];
+ /* Since this card has no ASIC, mark it as loaded so everything
+ works OK */
+ chip->asic_loaded = TRUE;
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+ ECHO_CLOCK_BIT_ESYNC;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock
+ detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_ESYNC)
+ clock_bits |= ECHO_CLOCK_BIT_ESYNC;
+
+ return clock_bits;
+}
+
+
+
+/* The Darla24 has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u8 clock;
+
+ switch (rate) {
+ case 96000:
+ clock = GD24_96000;
+ break;
+ case 88200:
+ clock = GD24_88200;
+ break;
+ case 48000:
+ clock = GD24_48000;
+ break;
+ case 44100:
+ clock = GD24_44100;
+ break;
+ case 32000:
+ clock = GD24_32000;
+ break;
+ case 22050:
+ clock = GD24_22050;
+ break;
+ case 16000:
+ clock = GD24_16000;
+ break;
+ case 11025:
+ clock = GD24_11025;
+ break;
+ case 8000:
+ clock = GD24_8000;
+ break;
+ default:
+ DE_ACT(("set_sample_rate: Error, invalid sample rate %d\n",
+ rate));
+ return -EINVAL;
+ }
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+ chip->sample_rate = rate;
+
+ /* Override the sample rate if this card is set to Echo sync. */
+ if (chip->input_clock == ECHO_CLOCK_ESYNC)
+ clock = GD24_EXT_SYNC;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP ? */
+ chip->comm_page->gd_clock_state = clock;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+ if (snd_BUG_ON(clock != ECHO_CLOCK_INTERNAL &&
+ clock != ECHO_CLOCK_ESYNC))
+ return -EINVAL;
+ chip->input_clock = clock;
+ return set_sample_rate(chip, chip->sample_rate);
+}
+
diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c
new file mode 100644
index 0000000..4022e43
--- /dev/null
+++ b/sound/pci/echoaudio/echo3g.c
@@ -0,0 +1,122 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO3G_FAMILY
+#define ECHOCARD_ECHO3G
+#define ECHOCARD_NAME "Echo3G"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_ADAT 6
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+#define ECHOCARD_HAS_MIDI
+#define ECHOCARD_HAS_PHANTOM_POWER
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0
+#define PX_DIGITAL_OUT chip->px_digital_out
+#define PX_ANALOG_IN chip->px_analog_in
+#define PX_DIGITAL_IN chip->px_digital_in
+#define PX_NUM chip->px_num
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0
+#define BX_DIGITAL_OUT chip->bx_digital_out
+#define BX_ANALOG_IN chip->bx_analog_in
+#define BX_DIGITAL_IN chip->bx_digital_in
+#define BX_NUM chip->bx_num
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/echo3g_dsp.fw");
+MODULE_FIRMWARE("ea/3g_asic.fw");
+
+#define FW_361_LOADER 0
+#define FW_ECHO3G_DSP 1
+#define FW_3G_ASIC 2
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "echo3g_dsp.fw"},
+ {0, "3g_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 32000,
+ .rate_max = 100000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+};
+
+#include "echo3g_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_3g.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c
new file mode 100644
index 0000000..417e25a
--- /dev/null
+++ b/sound/pci/echoaudio/echo3g_dsp.c
@@ -0,0 +1,134 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+static int load_asic(struct echoaudio *chip);
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int check_asic_status(struct echoaudio *chip);
+static int set_sample_rate(struct echoaudio *chip, u32 rate);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_phantom_power(struct echoaudio *chip, char on);
+static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
+ char force);
+
+#include <linux/interrupt.h>
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ local_irq_enable();
+ DE_INIT(("init_hw() - Echo3G\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->comm_page->e3g_frq_register =
+ __constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2);
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->has_midi = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP];
+
+ /* Load the DSP code and the ASIC on the PCI card and get
+ what type of external box is attached */
+ err = load_firmware(chip);
+
+ if (err < 0) {
+ return err;
+ } else if (err == E3G_GINA3G_BOX_TYPE) {
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+ ECHO_CLOCK_BIT_SPDIF |
+ ECHO_CLOCK_BIT_ADAT;
+ chip->card_name = "Gina3G";
+ chip->px_digital_out = chip->bx_digital_out = 6;
+ chip->px_analog_in = chip->bx_analog_in = 14;
+ chip->px_digital_in = chip->bx_digital_in = 16;
+ chip->px_num = chip->bx_num = 24;
+ chip->has_phantom_power = TRUE;
+ chip->hasnt_input_nominal_level = TRUE;
+ } else if (err == E3G_LAYLA3G_BOX_TYPE) {
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+ ECHO_CLOCK_BIT_SPDIF |
+ ECHO_CLOCK_BIT_ADAT |
+ ECHO_CLOCK_BIT_WORD;
+ chip->card_name = "Layla3G";
+ chip->px_digital_out = chip->bx_digital_out = 8;
+ chip->px_analog_in = chip->bx_analog_in = 16;
+ chip->px_digital_in = chip->bx_digital_in = 24;
+ chip->px_num = chip->bx_num = 32;
+ } else {
+ return -ENODEV;
+ }
+
+ chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+ ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+ chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
+ chip->professional_spdif = FALSE;
+ chip->non_audio_spdif = FALSE;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+ err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+ if (err < 0)
+ return err;
+ err = set_phantom_power(chip, 0);
+ if (err < 0)
+ return err;
+ err = set_professional_spdif(chip, TRUE);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static int set_phantom_power(struct echoaudio *chip, char on)
+{
+ u32 control_reg = le32_to_cpu(chip->comm_page->control_register);
+
+ if (on)
+ control_reg |= E3G_PHANTOM_POWER;
+ else
+ control_reg &= ~E3G_PHANTOM_POWER;
+
+ chip->phantom_power = on;
+ return write_control_reg(chip, control_reg,
+ le32_to_cpu(chip->comm_page->e3g_frq_register),
+ 0);
+}
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
new file mode 100644
index 0000000..8dbc5c4
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -0,0 +1,2181 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+MODULE_AUTHOR("Giuliano Pochini <pochini@shiny.it>");
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Echoaudio " ECHOCARD_NAME " soundcards driver");
+MODULE_SUPPORTED_DEVICE("{{Echoaudio," ECHOCARD_NAME "}}");
+MODULE_DEVICE_TABLE(pci, snd_echo_ids);
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " ECHOCARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " ECHOCARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " ECHOCARD_NAME " soundcard.");
+
+static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999};
+static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1);
+
+static int get_firmware(const struct firmware **fw_entry,
+ const struct firmware *frm, struct echoaudio *chip)
+{
+ int err;
+ char name[30];
+ DE_ACT(("firmware requested: %s\n", frm->data));
+ snprintf(name, sizeof(name), "ea/%s", frm->data);
+ if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0)
+ snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err);
+ return err;
+}
+
+static void free_firmware(const struct firmware *fw_entry)
+{
+ release_firmware(fw_entry);
+ DE_ACT(("firmware released\n"));
+}
+
+
+
+/******************************************************************************
+ PCM interface
+******************************************************************************/
+
+static void audiopipe_free(struct snd_pcm_runtime *runtime)
+{
+ struct audiopipe *pipe = runtime->private_data;
+
+ if (pipe->sgpage.area)
+ snd_dma_free_pages(&pipe->sgpage);
+ kfree(pipe);
+}
+
+
+
+static int hw_rule_capture_format_by_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_mask fmt;
+
+ snd_mask_any(&fmt);
+
+#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+ /* >=2 channels cannot be S32_BE */
+ if (c->min == 2) {
+ fmt.bits[0] &= ~SNDRV_PCM_FMTBIT_S32_BE;
+ return snd_mask_refine(f, &fmt);
+ }
+#endif
+ /* > 2 channels cannot be U8 and S32_BE */
+ if (c->min > 2) {
+ fmt.bits[0] &= ~(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_BE);
+ return snd_mask_refine(f, &fmt);
+ }
+ /* Mono is ok with any format */
+ return 0;
+}
+
+
+
+static int hw_rule_capture_channels_by_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_interval ch;
+
+ snd_interval_any(&ch);
+
+ /* S32_BE is mono (and stereo) only */
+ if (f->bits[0] == SNDRV_PCM_FMTBIT_S32_BE) {
+ ch.min = 1;
+#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+ ch.max = 2;
+#else
+ ch.max = 1;
+#endif
+ ch.integer = 1;
+ return snd_interval_refine(c, &ch);
+ }
+ /* U8 can be only mono or stereo */
+ if (f->bits[0] == SNDRV_PCM_FMTBIT_U8) {
+ ch.min = 1;
+ ch.max = 2;
+ ch.integer = 1;
+ return snd_interval_refine(c, &ch);
+ }
+ /* S16_LE, S24_3LE and S32_LE support any number of channels. */
+ return 0;
+}
+
+
+
+static int hw_rule_playback_format_by_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_mask fmt;
+ u64 fmask;
+ snd_mask_any(&fmt);
+
+ fmask = fmt.bits[0] + ((u64)fmt.bits[1] << 32);
+
+ /* >2 channels must be S16_LE, S24_3LE or S32_LE */
+ if (c->min > 2) {
+ fmask &= SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE;
+ /* 1 channel must be S32_BE or S32_LE */
+ } else if (c->max == 1)
+ fmask &= SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE;
+#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+ /* 2 channels cannot be S32_BE */
+ else if (c->min == 2 && c->max == 2)
+ fmask &= ~SNDRV_PCM_FMTBIT_S32_BE;
+#endif
+ else
+ return 0;
+
+ fmt.bits[0] &= (u32)fmask;
+ fmt.bits[1] &= (u32)(fmask >> 32);
+ return snd_mask_refine(f, &fmt);
+}
+
+
+
+static int hw_rule_playback_channels_by_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_interval ch;
+ u64 fmask;
+
+ snd_interval_any(&ch);
+ ch.integer = 1;
+ fmask = f->bits[0] + ((u64)f->bits[1] << 32);
+
+ /* S32_BE is mono (and stereo) only */
+ if (fmask == SNDRV_PCM_FMTBIT_S32_BE) {
+ ch.min = 1;
+#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+ ch.max = 2;
+#else
+ ch.max = 1;
+#endif
+ /* U8 is stereo only */
+ } else if (fmask == SNDRV_PCM_FMTBIT_U8)
+ ch.min = ch.max = 2;
+ /* S16_LE and S24_3LE must be at least stereo */
+ else if (!(fmask & ~(SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE)))
+ ch.min = 2;
+ else
+ return 0;
+
+ return snd_interval_refine(c, &ch);
+}
+
+
+
+/* Since the sample rate is a global setting, do allow the user to change the
+sample rate only if there is only one pcm device open. */
+static int hw_rule_sample_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct echoaudio *chip = rule->private;
+ struct snd_interval fixed;
+
+ if (!chip->can_set_rate) {
+ snd_interval_any(&fixed);
+ fixed.min = fixed.max = chip->sample_rate;
+ return snd_interval_refine(rate, &fixed);
+ }
+ return 0;
+}
+
+
+static int pcm_open(struct snd_pcm_substream *substream,
+ signed char max_channels)
+{
+ struct echoaudio *chip;
+ struct snd_pcm_runtime *runtime;
+ struct audiopipe *pipe;
+ int err, i;
+
+ if (max_channels <= 0)
+ return -EAGAIN;
+
+ chip = snd_pcm_substream_chip(substream);
+ runtime = substream->runtime;
+
+ pipe = kzalloc(sizeof(struct audiopipe), GFP_KERNEL);
+ if (!pipe)
+ return -ENOMEM;
+ pipe->index = -1; /* Not configured yet */
+
+ /* Set up hw capabilities and contraints */
+ memcpy(&pipe->hw, &pcm_hardware_skel, sizeof(struct snd_pcm_hardware));
+ DE_HWP(("max_channels=%d\n", max_channels));
+ pipe->constr.list = channels_list;
+ pipe->constr.mask = 0;
+ for (i = 0; channels_list[i] <= max_channels; i++);
+ pipe->constr.count = i;
+ if (pipe->hw.channels_max > max_channels)
+ pipe->hw.channels_max = max_channels;
+ if (chip->digital_mode == DIGITAL_MODE_ADAT) {
+ pipe->hw.rate_max = 48000;
+ pipe->hw.rates &= SNDRV_PCM_RATE_8000_48000;
+ }
+
+ runtime->hw = pipe->hw;
+ runtime->private_data = pipe;
+ runtime->private_free = audiopipe_free;
+ snd_pcm_set_sync(substream);
+
+ /* Only mono and any even number of channels are allowed */
+ if ((err = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &pipe->constr)) < 0)
+ return err;
+
+ /* All periods should have the same size */
+ if ((err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ /* The hw accesses memory in chunks 32 frames long and they should be
+ 32-bytes-aligned. It's not a requirement, but it seems that IRQs are
+ generated with a resolution of 32 frames. Thus we need the following */
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ 32)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ 32)) < 0)
+ return err;
+
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_sample_rate, chip,
+ SNDRV_PCM_HW_PARAM_RATE, -1)) < 0)
+ return err;
+
+ /* Finally allocate a page for the scatter-gather list */
+ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ PAGE_SIZE, &pipe->sgpage)) < 0) {
+ DE_HWP(("s-g list allocation failed\n"));
+ return err;
+ }
+
+ return 0;
+}
+
+
+
+static int pcm_analog_in_open(struct snd_pcm_substream *substream)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ DE_ACT(("pcm_analog_in_open\n"));
+ if ((err = pcm_open(substream, num_analog_busses_in(chip) -
+ substream->number)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_capture_channels_by_format, NULL,
+ SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_capture_format_by_channels, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
+ return err;
+ atomic_inc(&chip->opencount);
+ if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+ chip->can_set_rate=0;
+ DE_HWP(("pcm_analog_in_open cs=%d oc=%d r=%d\n",
+ chip->can_set_rate, atomic_read(&chip->opencount),
+ chip->sample_rate));
+ return 0;
+}
+
+
+
+static int pcm_analog_out_open(struct snd_pcm_substream *substream)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+ int max_channels, err;
+
+#ifdef ECHOCARD_HAS_VMIXER
+ max_channels = num_pipes_out(chip);
+#else
+ max_channels = num_analog_busses_out(chip);
+#endif
+ DE_ACT(("pcm_analog_out_open\n"));
+ if ((err = pcm_open(substream, max_channels - substream->number)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_playback_channels_by_format,
+ NULL,
+ SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_playback_format_by_channels,
+ NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
+ return err;
+ atomic_inc(&chip->opencount);
+ if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+ chip->can_set_rate=0;
+ DE_HWP(("pcm_analog_out_open cs=%d oc=%d r=%d\n",
+ chip->can_set_rate, atomic_read(&chip->opencount),
+ chip->sample_rate));
+ return 0;
+}
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+
+static int pcm_digital_in_open(struct snd_pcm_substream *substream)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+ int err, max_channels;
+
+ DE_ACT(("pcm_digital_in_open\n"));
+ max_channels = num_digital_busses_in(chip) - substream->number;
+ mutex_lock(&chip->mode_mutex);
+ if (chip->digital_mode == DIGITAL_MODE_ADAT)
+ err = pcm_open(substream, max_channels);
+ else /* If the card has ADAT, subtract the 6 channels
+ * that S/PDIF doesn't have
+ */
+ err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
+
+ if (err < 0)
+ goto din_exit;
+
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_capture_channels_by_format, NULL,
+ SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
+ goto din_exit;
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_capture_format_by_channels, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
+ goto din_exit;
+
+ atomic_inc(&chip->opencount);
+ if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+ chip->can_set_rate=0;
+
+din_exit:
+ mutex_unlock(&chip->mode_mutex);
+ return err;
+}
+
+
+
+#ifndef ECHOCARD_HAS_VMIXER /* See the note in snd_echo_new_pcm() */
+
+static int pcm_digital_out_open(struct snd_pcm_substream *substream)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+ int err, max_channels;
+
+ DE_ACT(("pcm_digital_out_open\n"));
+ max_channels = num_digital_busses_out(chip) - substream->number;
+ mutex_lock(&chip->mode_mutex);
+ if (chip->digital_mode == DIGITAL_MODE_ADAT)
+ err = pcm_open(substream, max_channels);
+ else /* If the card has ADAT, subtract the 6 channels
+ * that S/PDIF doesn't have
+ */
+ err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
+
+ if (err < 0)
+ goto dout_exit;
+
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_playback_channels_by_format,
+ NULL, SNDRV_PCM_HW_PARAM_FORMAT,
+ -1)) < 0)
+ goto dout_exit;
+ if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_playback_format_by_channels,
+ NULL, SNDRV_PCM_HW_PARAM_CHANNELS,
+ -1)) < 0)
+ goto dout_exit;
+ atomic_inc(&chip->opencount);
+ if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+ chip->can_set_rate=0;
+dout_exit:
+ mutex_unlock(&chip->mode_mutex);
+ return err;
+}
+
+#endif /* !ECHOCARD_HAS_VMIXER */
+
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+ int oc;
+
+ /* Nothing to do here. Audio is already off and pipe will be
+ * freed by its callback
+ */
+ DE_ACT(("pcm_close\n"));
+
+ atomic_dec(&chip->opencount);
+ oc = atomic_read(&chip->opencount);
+ DE_ACT(("pcm_close oc=%d cs=%d rs=%d\n", oc,
+ chip->can_set_rate, chip->rate_set));
+ if (oc < 2)
+ chip->can_set_rate = 1;
+ if (oc == 0)
+ chip->rate_set = 0;
+ DE_ACT(("pcm_close2 oc=%d cs=%d rs=%d\n", oc,
+ chip->can_set_rate,chip->rate_set));
+
+ return 0;
+}
+
+
+
+/* Channel allocation and scatter-gather list setup */
+static int init_engine(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params,
+ int pipe_index, int interleave)
+{
+ struct echoaudio *chip;
+ int err, per, rest, page, edge, offs;
+ struct audiopipe *pipe;
+
+ chip = snd_pcm_substream_chip(substream);
+ pipe = (struct audiopipe *) substream->runtime->private_data;
+
+ /* Sets up che hardware. If it's already initialized, reset and
+ * redo with the new parameters
+ */
+ spin_lock_irq(&chip->lock);
+ if (pipe->index >= 0) {
+ DE_HWP(("hwp_ie free(%d)\n", pipe->index));
+ err = free_pipes(chip, pipe);
+ snd_BUG_ON(err);
+ chip->substream[pipe->index] = NULL;
+ }
+
+ err = allocate_pipes(chip, pipe, pipe_index, interleave);
+ if (err < 0) {
+ spin_unlock_irq(&chip->lock);
+ DE_ACT((KERN_NOTICE "allocate_pipes(%d) err=%d\n",
+ pipe_index, err));
+ return err;
+ }
+ spin_unlock_irq(&chip->lock);
+ DE_ACT((KERN_NOTICE "allocate_pipes()=%d\n", pipe_index));
+
+ DE_HWP(("pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n",
+ params_buffer_bytes(hw_params), params_periods(hw_params),
+ params_period_bytes(hw_params)));
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0) {
+ snd_printk(KERN_ERR "malloc_pages err=%d\n", err);
+ spin_lock_irq(&chip->lock);
+ free_pipes(chip, pipe);
+ spin_unlock_irq(&chip->lock);
+ pipe->index = -1;
+ return err;
+ }
+
+ sglist_init(chip, pipe);
+ edge = PAGE_SIZE;
+ for (offs = page = per = 0; offs < params_buffer_bytes(hw_params);
+ per++) {
+ rest = params_period_bytes(hw_params);
+ if (offs + rest > params_buffer_bytes(hw_params))
+ rest = params_buffer_bytes(hw_params) - offs;
+ while (rest) {
+ dma_addr_t addr;
+ addr = snd_pcm_sgbuf_get_addr(substream, offs);
+ if (rest <= edge - offs) {
+ sglist_add_mapping(chip, pipe, addr, rest);
+ sglist_add_irq(chip, pipe);
+ offs += rest;
+ rest = 0;
+ } else {
+ sglist_add_mapping(chip, pipe, addr,
+ edge - offs);
+ rest -= edge - offs;
+ offs = edge;
+ }
+ if (offs == edge) {
+ edge += PAGE_SIZE;
+ page++;
+ }
+ }
+ }
+
+ /* Close the ring buffer */
+ sglist_wrap(chip, pipe);
+
+ /* This stuff is used by the irq handler, so it must be
+ * initialized before chip->substream
+ */
+ chip->last_period[pipe_index] = 0;
+ pipe->last_counter = 0;
+ pipe->position = 0;
+ smp_wmb();
+ chip->substream[pipe_index] = substream;
+ chip->rate_set = 1;
+ spin_lock_irq(&chip->lock);
+ set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den);
+ spin_unlock_irq(&chip->lock);
+ DE_HWP(("pcm_hw_params ok\n"));
+ return 0;
+}
+
+
+
+static int pcm_analog_in_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+
+ return init_engine(substream, hw_params, px_analog_in(chip) +
+ substream->number, params_channels(hw_params));
+}
+
+
+
+static int pcm_analog_out_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return init_engine(substream, hw_params, substream->number,
+ params_channels(hw_params));
+}
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+
+static int pcm_digital_in_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+
+ return init_engine(substream, hw_params, px_digital_in(chip) +
+ substream->number, params_channels(hw_params));
+}
+
+
+
+#ifndef ECHOCARD_HAS_VMIXER /* See the note in snd_echo_new_pcm() */
+static int pcm_digital_out_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+
+ return init_engine(substream, hw_params, px_digital_out(chip) +
+ substream->number, params_channels(hw_params));
+}
+#endif /* !ECHOCARD_HAS_VMIXER */
+
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+static int pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct echoaudio *chip;
+ struct audiopipe *pipe;
+
+ chip = snd_pcm_substream_chip(substream);
+ pipe = (struct audiopipe *) substream->runtime->private_data;
+
+ spin_lock_irq(&chip->lock);
+ if (pipe->index >= 0) {
+ DE_HWP(("pcm_hw_free(%d)\n", pipe->index));
+ free_pipes(chip, pipe);
+ chip->substream[pipe->index] = NULL;
+ pipe->index = -1;
+ }
+ spin_unlock_irq(&chip->lock);
+
+ DE_HWP(("pcm_hw_freed\n"));
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+
+
+static int pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audioformat format;
+ int pipe_index = ((struct audiopipe *)runtime->private_data)->index;
+
+ DE_HWP(("Prepare rate=%d format=%d channels=%d\n",
+ runtime->rate, runtime->format, runtime->channels));
+ format.interleave = runtime->channels;
+ format.data_are_bigendian = 0;
+ format.mono_to_stereo = 0;
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_U8:
+ format.bits_per_sample = 8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ format.bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ format.bits_per_sample = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_BE:
+ format.data_are_bigendian = 1;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ format.bits_per_sample = 32;
+ break;
+ default:
+ DE_HWP(("Prepare error: unsupported format %d\n",
+ runtime->format));
+ return -EINVAL;
+ }
+
+ if (snd_BUG_ON(pipe_index >= px_num(chip)))
+ return -EINVAL;
+ if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index)))
+ return -EINVAL;
+ set_audio_format(chip, pipe_index, &format);
+ return 0;
+}
+
+
+
+static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct echoaudio *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audiopipe *pipe = runtime->private_data;
+ int i, err;
+ u32 channelmask = 0;
+ struct snd_pcm_substream *s;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ for (i = 0; i < DSP_MAXPIPES; i++) {
+ if (s == chip->substream[i]) {
+ channelmask |= 1 << i;
+ snd_pcm_trigger_done(s, substream);
+ }
+ }
+ }
+
+ spin_lock(&chip->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ DE_ACT(("pcm_trigger start\n"));
+ for (i = 0; i < DSP_MAXPIPES; i++) {
+ if (channelmask & (1 << i)) {
+ pipe = chip->substream[i]->runtime->private_data;
+ switch (pipe->state) {
+ case PIPE_STATE_STOPPED:
+ chip->last_period[i] = 0;
+ pipe->last_counter = 0;
+ pipe->position = 0;
+ *pipe->dma_counter = 0;
+ case PIPE_STATE_PAUSED:
+ pipe->state = PIPE_STATE_STARTED;
+ break;
+ case PIPE_STATE_STARTED:
+ break;
+ }
+ }
+ }
+ err = start_transport(chip, channelmask,
+ chip->pipe_cyclic_mask);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ DE_ACT(("pcm_trigger stop\n"));
+ for (i = 0; i < DSP_MAXPIPES; i++) {
+ if (channelmask & (1 << i)) {
+ pipe = chip->substream[i]->runtime->private_data;
+ pipe->state = PIPE_STATE_STOPPED;
+ }
+ }
+ err = stop_transport(chip, channelmask);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ DE_ACT(("pcm_trigger pause\n"));
+ for (i = 0; i < DSP_MAXPIPES; i++) {
+ if (channelmask & (1 << i)) {
+ pipe = chip->substream[i]->runtime->private_data;
+ pipe->state = PIPE_STATE_PAUSED;
+ }
+ }
+ err = pause_transport(chip, channelmask);
+ break;
+ default:
+ err = -EINVAL;
+ }
+ spin_unlock(&chip->lock);
+ return err;
+}
+
+
+
+static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audiopipe *pipe = runtime->private_data;
+ size_t cnt, bufsize, pos;
+
+ cnt = le32_to_cpu(*pipe->dma_counter);
+ pipe->position += cnt - pipe->last_counter;
+ pipe->last_counter = cnt;
+ bufsize = substream->runtime->buffer_size;
+ pos = bytes_to_frames(substream->runtime, pipe->position);
+
+ while (pos >= bufsize) {
+ pipe->position -= frames_to_bytes(substream->runtime, bufsize);
+ pos -= bufsize;
+ }
+ return pos;
+}
+
+
+
+/* pcm *_ops structures */
+static struct snd_pcm_ops analog_playback_ops = {
+ .open = pcm_analog_out_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_analog_out_hw_params,
+ .hw_free = pcm_hw_free,
+ .prepare = pcm_prepare,
+ .trigger = pcm_trigger,
+ .pointer = pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+static struct snd_pcm_ops analog_capture_ops = {
+ .open = pcm_analog_in_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_analog_in_hw_params,
+ .hw_free = pcm_hw_free,
+ .prepare = pcm_prepare,
+ .trigger = pcm_trigger,
+ .pointer = pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+#ifndef ECHOCARD_HAS_VMIXER
+static struct snd_pcm_ops digital_playback_ops = {
+ .open = pcm_digital_out_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_digital_out_hw_params,
+ .hw_free = pcm_hw_free,
+ .prepare = pcm_prepare,
+ .trigger = pcm_trigger,
+ .pointer = pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+#endif /* !ECHOCARD_HAS_VMIXER */
+static struct snd_pcm_ops digital_capture_ops = {
+ .open = pcm_digital_in_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_digital_in_hw_params,
+ .hw_free = pcm_hw_free,
+ .prepare = pcm_prepare,
+ .trigger = pcm_trigger,
+ .pointer = pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+/* Preallocate memory only for the first substream because it's the most
+ * used one
+ */
+static int snd_echo_preallocate_pages(struct snd_pcm *pcm, struct device *dev)
+{
+ struct snd_pcm_substream *ss;
+ int stream, err;
+
+ for (stream = 0; stream < 2; stream++)
+ for (ss = pcm->streams[stream].substream; ss; ss = ss->next) {
+ err = snd_pcm_lib_preallocate_pages(ss, SNDRV_DMA_TYPE_DEV_SG,
+ dev,
+ ss->number ? 0 : 128<<10,
+ 256<<10);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+
+
+/*<--snd_echo_probe() */
+static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+#ifdef ECHOCARD_HAS_VMIXER
+ /* This card has a Vmixer, that is there is no direct mapping from PCM
+ streams to physical outputs. The user can mix the streams as he wishes
+ via control interface and it's possible to send any stream to any
+ output, thus it makes no sense to keep analog and digital outputs
+ separated */
+
+ /* PCM#0 Virtual outputs and analog inputs */
+ if ((err = snd_pcm_new(chip->card, "PCM", 0, num_pipes_out(chip),
+ num_analog_busses_in(chip), &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+ chip->analog_pcm = pcm;
+ strcpy(pcm->name, chip->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
+ if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+ return err;
+ DE_INIT(("Analog PCM ok\n"));
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+ /* PCM#1 Digital inputs, no outputs */
+ if ((err = snd_pcm_new(chip->card, "Digital PCM", 1, 0,
+ num_digital_busses_in(chip), &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+ chip->digital_pcm = pcm;
+ strcpy(pcm->name, chip->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
+ if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+ return err;
+ DE_INIT(("Digital PCM ok\n"));
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+#else /* ECHOCARD_HAS_VMIXER */
+
+ /* The card can manage substreams formed by analog and digital channels
+ at the same time, but I prefer to keep analog and digital channels
+ separated, because that mixed thing is confusing and useless. So we
+ register two PCM devices: */
+
+ /* PCM#0 Analog i/o */
+ if ((err = snd_pcm_new(chip->card, "Analog PCM", 0,
+ num_analog_busses_out(chip),
+ num_analog_busses_in(chip), &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+ chip->analog_pcm = pcm;
+ strcpy(pcm->name, chip->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
+ if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+ return err;
+ DE_INIT(("Analog PCM ok\n"));
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+ /* PCM#1 Digital i/o */
+ if ((err = snd_pcm_new(chip->card, "Digital PCM", 1,
+ num_digital_busses_out(chip),
+ num_digital_busses_in(chip), &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+ chip->digital_pcm = pcm;
+ strcpy(pcm->name, chip->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &digital_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
+ if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+ return err;
+ DE_INIT(("Digital PCM ok\n"));
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+#endif /* ECHOCARD_HAS_VMIXER */
+
+ return 0;
+}
+
+
+
+
+/******************************************************************************
+ Control interface
+******************************************************************************/
+
+/******************* PCM output volume *******************/
+static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = num_busses_out(chip);
+ uinfo->value.integer.min = ECHOGAIN_MINOUT;
+ uinfo->value.integer.max = ECHOGAIN_MAXOUT;
+ return 0;
+}
+
+static int snd_echo_output_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ for (c = 0; c < num_busses_out(chip); c++)
+ ucontrol->value.integer.value[c] = chip->output_gain[c];
+ return 0;
+}
+
+static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c, changed, gain;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+ spin_lock_irq(&chip->lock);
+ for (c = 0; c < num_busses_out(chip); c++) {
+ gain = ucontrol->value.integer.value[c];
+ /* Ignore out of range values */
+ if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
+ continue;
+ if (chip->output_gain[c] != gain) {
+ set_output_gain(chip, c, gain);
+ changed = 1;
+ }
+ }
+ if (changed)
+ update_output_line_level(chip);
+ spin_unlock_irq(&chip->lock);
+ return changed;
+}
+
+#ifdef ECHOCARD_HAS_VMIXER
+/* On Vmixer cards this one controls the line-out volume */
+static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
+ .name = "Line Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_output_gain_info,
+ .get = snd_echo_output_gain_get,
+ .put = snd_echo_output_gain_put,
+ .tlv = {.p = db_scale_output_gain},
+};
+#else
+static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
+ .name = "PCM Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_output_gain_info,
+ .get = snd_echo_output_gain_get,
+ .put = snd_echo_output_gain_put,
+ .tlv = {.p = db_scale_output_gain},
+};
+#endif
+
+
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+
+/******************* Analog input volume *******************/
+static int snd_echo_input_gain_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = num_analog_busses_in(chip);
+ uinfo->value.integer.min = ECHOGAIN_MININP;
+ uinfo->value.integer.max = ECHOGAIN_MAXINP;
+ return 0;
+}
+
+static int snd_echo_input_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ for (c = 0; c < num_analog_busses_in(chip); c++)
+ ucontrol->value.integer.value[c] = chip->input_gain[c];
+ return 0;
+}
+
+static int snd_echo_input_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c, gain, changed;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+ spin_lock_irq(&chip->lock);
+ for (c = 0; c < num_analog_busses_in(chip); c++) {
+ gain = ucontrol->value.integer.value[c];
+ /* Ignore out of range values */
+ if (gain < ECHOGAIN_MININP || gain > ECHOGAIN_MAXINP)
+ continue;
+ if (chip->input_gain[c] != gain) {
+ set_input_gain(chip, c, gain);
+ changed = 1;
+ }
+ }
+ if (changed)
+ update_input_line_level(chip);
+ spin_unlock_irq(&chip->lock);
+ return changed;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_input_gain, -2500, 50, 0);
+
+static struct snd_kcontrol_new snd_echo_line_input_gain __devinitdata = {
+ .name = "Line Capture Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_input_gain_info,
+ .get = snd_echo_input_gain_get,
+ .put = snd_echo_input_gain_put,
+ .tlv = {.p = db_scale_input_gain},
+};
+
+#endif /* ECHOCARD_HAS_INPUT_GAIN */
+
+
+
+#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+
+/************ Analog output nominal level (+4dBu / -10dBV) ***************/
+static int snd_echo_output_nominal_info (struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = num_analog_busses_out(chip);
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_echo_output_nominal_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ for (c = 0; c < num_analog_busses_out(chip); c++)
+ ucontrol->value.integer.value[c] = chip->nominal_level[c];
+ return 0;
+}
+
+static int snd_echo_output_nominal_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c, changed;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+ spin_lock_irq(&chip->lock);
+ for (c = 0; c < num_analog_busses_out(chip); c++) {
+ if (chip->nominal_level[c] != ucontrol->value.integer.value[c]) {
+ set_nominal_level(chip, c,
+ ucontrol->value.integer.value[c]);
+ changed = 1;
+ }
+ }
+ if (changed)
+ update_output_line_level(chip);
+ spin_unlock_irq(&chip->lock);
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_output_nominal_level __devinitdata = {
+ .name = "Line Playback Switch (-10dBV)",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_echo_output_nominal_info,
+ .get = snd_echo_output_nominal_get,
+ .put = snd_echo_output_nominal_put,
+};
+
+#endif /* ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL */
+
+
+
+#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+
+/*************** Analog input nominal level (+4dBu / -10dBV) ***************/
+static int snd_echo_input_nominal_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = num_analog_busses_in(chip);
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_echo_input_nominal_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ for (c = 0; c < num_analog_busses_in(chip); c++)
+ ucontrol->value.integer.value[c] =
+ chip->nominal_level[bx_analog_in(chip) + c];
+ return 0;
+}
+
+static int snd_echo_input_nominal_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int c, changed;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+ spin_lock_irq(&chip->lock);
+ for (c = 0; c < num_analog_busses_in(chip); c++) {
+ if (chip->nominal_level[bx_analog_in(chip) + c] !=
+ ucontrol->value.integer.value[c]) {
+ set_nominal_level(chip, bx_analog_in(chip) + c,
+ ucontrol->value.integer.value[c]);
+ changed = 1;
+ }
+ }
+ if (changed)
+ update_output_line_level(chip); /* "Output" is not a mistake
+ * here.
+ */
+ spin_unlock_irq(&chip->lock);
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_intput_nominal_level __devinitdata = {
+ .name = "Line Capture Switch (-10dBV)",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_echo_input_nominal_info,
+ .get = snd_echo_input_nominal_get,
+ .put = snd_echo_input_nominal_put,
+};
+
+#endif /* ECHOCARD_HAS_INPUT_NOMINAL_LEVEL */
+
+
+
+#ifdef ECHOCARD_HAS_MONITOR
+
+/******************* Monitor mixer *******************/
+static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = ECHOGAIN_MINOUT;
+ uinfo->value.integer.max = ECHOGAIN_MAXOUT;
+ uinfo->dimen.d[0] = num_busses_out(chip);
+ uinfo->dimen.d[1] = num_busses_in(chip);
+ return 0;
+}
+
+static int snd_echo_mixer_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] =
+ chip->monitor_gain[ucontrol->id.index / num_busses_in(chip)]
+ [ucontrol->id.index % num_busses_in(chip)];
+ return 0;
+}
+
+static int snd_echo_mixer_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int changed, gain;
+ short out, in;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+ out = ucontrol->id.index / num_busses_in(chip);
+ in = ucontrol->id.index % num_busses_in(chip);
+ gain = ucontrol->value.integer.value[0];
+ if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
+ return -EINVAL;
+ if (chip->monitor_gain[out][in] != gain) {
+ spin_lock_irq(&chip->lock);
+ set_monitor_gain(chip, out, in, gain);
+ update_output_line_level(chip);
+ spin_unlock_irq(&chip->lock);
+ changed = 1;
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_monitor_mixer __devinitdata = {
+ .name = "Monitor Mixer Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_mixer_info,
+ .get = snd_echo_mixer_get,
+ .put = snd_echo_mixer_put,
+ .tlv = {.p = db_scale_output_gain},
+};
+
+#endif /* ECHOCARD_HAS_MONITOR */
+
+
+
+#ifdef ECHOCARD_HAS_VMIXER
+
+/******************* Vmixer *******************/
+static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = ECHOGAIN_MINOUT;
+ uinfo->value.integer.max = ECHOGAIN_MAXOUT;
+ uinfo->dimen.d[0] = num_busses_out(chip);
+ uinfo->dimen.d[1] = num_pipes_out(chip);
+ return 0;
+}
+
+static int snd_echo_vmixer_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] =
+ chip->vmixer_gain[ucontrol->id.index / num_pipes_out(chip)]
+ [ucontrol->id.index % num_pipes_out(chip)];
+ return 0;
+}
+
+static int snd_echo_vmixer_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int gain, changed;
+ short vch, out;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+ out = ucontrol->id.index / num_pipes_out(chip);
+ vch = ucontrol->id.index % num_pipes_out(chip);
+ gain = ucontrol->value.integer.value[0];
+ if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
+ return -EINVAL;
+ if (chip->vmixer_gain[out][vch] != ucontrol->value.integer.value[0]) {
+ spin_lock_irq(&chip->lock);
+ set_vmixer_gain(chip, out, vch, ucontrol->value.integer.value[0]);
+ update_vmixer_level(chip);
+ spin_unlock_irq(&chip->lock);
+ changed = 1;
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_vmixer __devinitdata = {
+ .name = "VMixer Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_vmixer_info,
+ .get = snd_echo_vmixer_get,
+ .put = snd_echo_vmixer_put,
+ .tlv = {.p = db_scale_output_gain},
+};
+
+#endif /* ECHOCARD_HAS_VMIXER */
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+
+/******************* Digital mode switch *******************/
+static int snd_echo_digital_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *names[4] = {
+ "S/PDIF Coaxial", "S/PDIF Optical", "ADAT Optical",
+ "S/PDIF Cdrom"
+ };
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->value.enumerated.items = chip->num_digital_modes;
+ uinfo->count = 1;
+ if (uinfo->value.enumerated.item >= chip->num_digital_modes)
+ uinfo->value.enumerated.item = chip->num_digital_modes - 1;
+ strcpy(uinfo->value.enumerated.name, names[
+ chip->digital_mode_list[uinfo->value.enumerated.item]]);
+ return 0;
+}
+
+static int snd_echo_digital_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int i, mode;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ mode = chip->digital_mode;
+ for (i = chip->num_digital_modes - 1; i >= 0; i--)
+ if (mode == chip->digital_mode_list[i]) {
+ ucontrol->value.enumerated.item[0] = i;
+ break;
+ }
+ return 0;
+}
+
+static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int changed;
+ unsigned short emode, dmode;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+
+ emode = ucontrol->value.enumerated.item[0];
+ if (emode >= chip->num_digital_modes)
+ return -EINVAL;
+ dmode = chip->digital_mode_list[emode];
+
+ if (dmode != chip->digital_mode) {
+ /* mode_mutex is required to make this operation atomic wrt
+ pcm_digital_*_open() and set_input_clock() functions. */
+ mutex_lock(&chip->mode_mutex);
+
+ /* Do not allow the user to change the digital mode when a pcm
+ device is open because it also changes the number of channels
+ and the allowed sample rates */
+ if (atomic_read(&chip->opencount)) {
+ changed = -EAGAIN;
+ } else {
+ changed = set_digital_mode(chip, dmode);
+ /* If we had to change the clock source, report it */
+ if (changed > 0 && chip->clock_src_ctl) {
+ snd_ctl_notify(chip->card,
+ SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->clock_src_ctl->id);
+ DE_ACT(("SDM() =%d\n", changed));
+ }
+ if (changed >= 0)
+ changed = 1; /* No errors */
+ }
+ mutex_unlock(&chip->mode_mutex);
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_digital_mode_switch __devinitdata = {
+ .name = "Digital mode Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .info = snd_echo_digital_mode_info,
+ .get = snd_echo_digital_mode_get,
+ .put = snd_echo_digital_mode_put,
+};
+
+#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+
+/******************* S/PDIF mode switch *******************/
+static int snd_echo_spdif_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *names[2] = {"Consumer", "Professional"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->value.enumerated.items = 2;
+ uinfo->count = 1;
+ if (uinfo->value.enumerated.item)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name,
+ names[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_echo_spdif_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = !!chip->professional_spdif;
+ return 0;
+}
+
+static int snd_echo_spdif_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int mode;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ mode = !!ucontrol->value.enumerated.item[0];
+ if (mode != chip->professional_spdif) {
+ spin_lock_irq(&chip->lock);
+ set_professional_spdif(chip, mode);
+ spin_unlock_irq(&chip->lock);
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_echo_spdif_mode_switch __devinitdata = {
+ .name = "S/PDIF mode Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .info = snd_echo_spdif_mode_info,
+ .get = snd_echo_spdif_mode_get,
+ .put = snd_echo_spdif_mode_put,
+};
+
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
+
+/******************* Select input clock source *******************/
+static int snd_echo_clock_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *names[8] = {
+ "Internal", "Word", "Super", "S/PDIF", "ADAT", "ESync",
+ "ESync96", "MTC"
+ };
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->value.enumerated.items = chip->num_clock_sources;
+ uinfo->count = 1;
+ if (uinfo->value.enumerated.item >= chip->num_clock_sources)
+ uinfo->value.enumerated.item = chip->num_clock_sources - 1;
+ strcpy(uinfo->value.enumerated.name, names[
+ chip->clock_source_list[uinfo->value.enumerated.item]]);
+ return 0;
+}
+
+static int snd_echo_clock_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int i, clock;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ clock = chip->input_clock;
+
+ for (i = 0; i < chip->num_clock_sources; i++)
+ if (clock == chip->clock_source_list[i])
+ ucontrol->value.enumerated.item[0] = i;
+
+ return 0;
+}
+
+static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int changed;
+ unsigned int eclock, dclock;
+
+ changed = 0;
+ chip = snd_kcontrol_chip(kcontrol);
+ eclock = ucontrol->value.enumerated.item[0];
+ if (eclock >= chip->input_clock_types)
+ return -EINVAL;
+ dclock = chip->clock_source_list[eclock];
+ if (chip->input_clock != dclock) {
+ mutex_lock(&chip->mode_mutex);
+ spin_lock_irq(&chip->lock);
+ if ((changed = set_input_clock(chip, dclock)) == 0)
+ changed = 1; /* no errors */
+ spin_unlock_irq(&chip->lock);
+ mutex_unlock(&chip->mode_mutex);
+ }
+
+ if (changed < 0)
+ DE_ACT(("seticlk val%d err 0x%x\n", dclock, changed));
+
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_clock_source_switch __devinitdata = {
+ .name = "Sample Clock Source",
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .info = snd_echo_clock_source_info,
+ .get = snd_echo_clock_source_get,
+ .put = snd_echo_clock_source_put,
+};
+
+#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
+
+
+
+#ifdef ECHOCARD_HAS_PHANTOM_POWER
+
+/******************* Phantom power switch *******************/
+#define snd_echo_phantom_power_info snd_ctl_boolean_mono_info
+
+static int snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = chip->phantom_power;
+ return 0;
+}
+
+static int snd_echo_phantom_power_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+ int power, changed = 0;
+
+ power = !!ucontrol->value.integer.value[0];
+ if (chip->phantom_power != power) {
+ spin_lock_irq(&chip->lock);
+ changed = set_phantom_power(chip, power);
+ spin_unlock_irq(&chip->lock);
+ if (changed == 0)
+ changed = 1; /* no errors */
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_phantom_power_switch __devinitdata = {
+ .name = "Phantom power Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .info = snd_echo_phantom_power_info,
+ .get = snd_echo_phantom_power_get,
+ .put = snd_echo_phantom_power_put,
+};
+
+#endif /* ECHOCARD_HAS_PHANTOM_POWER */
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+
+/******************* Digital input automute switch *******************/
+#define snd_echo_automute_info snd_ctl_boolean_mono_info
+
+static int snd_echo_automute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = chip->digital_in_automute;
+ return 0;
+}
+
+static int snd_echo_automute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+ int automute, changed = 0;
+
+ automute = !!ucontrol->value.integer.value[0];
+ if (chip->digital_in_automute != automute) {
+ spin_lock_irq(&chip->lock);
+ changed = set_input_auto_mute(chip, automute);
+ spin_unlock_irq(&chip->lock);
+ if (changed == 0)
+ changed = 1; /* no errors */
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_automute_switch __devinitdata = {
+ .name = "Digital Capture Switch (automute)",
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .info = snd_echo_automute_info,
+ .get = snd_echo_automute_get,
+ .put = snd_echo_automute_put,
+};
+
+#endif /* ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE */
+
+
+
+/******************* VU-meters switch *******************/
+#define snd_echo_vumeters_switch_info snd_ctl_boolean_mono_info
+
+static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ spin_lock_irq(&chip->lock);
+ set_meters_on(chip, ucontrol->value.integer.value[0]);
+ spin_unlock_irq(&chip->lock);
+ return 1;
+}
+
+static struct snd_kcontrol_new snd_echo_vumeters_switch __devinitdata = {
+ .name = "VU-meters Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .access = SNDRV_CTL_ELEM_ACCESS_WRITE,
+ .info = snd_echo_vumeters_switch_info,
+ .put = snd_echo_vumeters_switch_put,
+};
+
+
+
+/***** Read VU-meters (input, output, analog and digital together) *****/
+static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 96;
+ uinfo->value.integer.min = ECHOGAIN_MINOUT;
+ uinfo->value.integer.max = 0;
+#ifdef ECHOCARD_HAS_VMIXER
+ uinfo->dimen.d[0] = 3; /* Out, In, Virt */
+#else
+ uinfo->dimen.d[0] = 2; /* Out, In */
+#endif
+ uinfo->dimen.d[1] = 16; /* 16 channels */
+ uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */
+ return 0;
+}
+
+static int snd_echo_vumeters_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ get_audio_meters(chip, ucontrol->value.integer.value);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_echo_vumeters __devinitdata = {
+ .name = "VU-meters",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_vumeters_info,
+ .get = snd_echo_vumeters_get,
+ .tlv = {.p = db_scale_output_gain},
+};
+
+
+
+/*** Channels info - it exports informations about the number of channels ***/
+static int snd_echo_channels_info_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct echoaudio *chip;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 6;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1 << ECHO_CLOCK_NUMBER;
+ return 0;
+}
+
+static int snd_echo_channels_info_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct echoaudio *chip;
+ int detected, clocks, bit, src;
+
+ chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = num_busses_in(chip);
+ ucontrol->value.integer.value[1] = num_analog_busses_in(chip);
+ ucontrol->value.integer.value[2] = num_busses_out(chip);
+ ucontrol->value.integer.value[3] = num_analog_busses_out(chip);
+ ucontrol->value.integer.value[4] = num_pipes_out(chip);
+
+ /* Compute the bitmask of the currently valid input clocks */
+ detected = detect_input_clocks(chip);
+ clocks = 0;
+ src = chip->num_clock_sources - 1;
+ for (bit = ECHO_CLOCK_NUMBER - 1; bit >= 0; bit--)
+ if (detected & (1 << bit))
+ for (; src >= 0; src--)
+ if (bit == chip->clock_source_list[src]) {
+ clocks |= 1 << src;
+ break;
+ }
+ ucontrol->value.integer.value[5] = clocks;
+
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_echo_channels_info __devinitdata = {
+ .name = "Channels info",
+ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP,
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_echo_channels_info_info,
+ .get = snd_echo_channels_info_get,
+};
+
+
+
+
+/******************************************************************************
+ IRQ Handler
+******************************************************************************/
+
+static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
+{
+ struct echoaudio *chip = dev_id;
+ struct snd_pcm_substream *substream;
+ int period, ss, st;
+
+ spin_lock(&chip->lock);
+ st = service_irq(chip);
+ if (st < 0) {
+ spin_unlock(&chip->lock);
+ return IRQ_NONE;
+ }
+ /* The hardware doesn't tell us which substream caused the irq,
+ thus we have to check all running substreams. */
+ for (ss = 0; ss < DSP_MAXPIPES; ss++) {
+ if ((substream = chip->substream[ss])) {
+ period = pcm_pointer(substream) /
+ substream->runtime->period_size;
+ if (period != chip->last_period[ss]) {
+ chip->last_period[ss] = period;
+ spin_unlock(&chip->lock);
+ snd_pcm_period_elapsed(substream);
+ spin_lock(&chip->lock);
+ }
+ }
+ }
+ spin_unlock(&chip->lock);
+
+#ifdef ECHOCARD_HAS_MIDI
+ if (st > 0 && chip->midi_in) {
+ snd_rawmidi_receive(chip->midi_in, chip->midi_buffer, st);
+ DE_MID(("rawmidi_iread=%d\n", st));
+ }
+#endif
+ return IRQ_HANDLED;
+}
+
+
+
+
+/******************************************************************************
+ Module construction / destruction
+******************************************************************************/
+
+static int snd_echo_free(struct echoaudio *chip)
+{
+ DE_INIT(("Stop DSP...\n"));
+ if (chip->comm_page)
+ rest_in_peace(chip);
+ DE_INIT(("Stopped.\n"));
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+
+ if (chip->comm_page)
+ snd_dma_free_pages(&chip->commpage_dma_buf);
+
+ if (chip->dsp_registers)
+ iounmap(chip->dsp_registers);
+
+ if (chip->iores)
+ release_and_free_resource(chip->iores);
+
+ DE_INIT(("MMIO freed.\n"));
+
+ pci_disable_device(chip->pci);
+
+ /* release chip data */
+ kfree(chip);
+ DE_INIT(("Chip freed.\n"));
+ return 0;
+}
+
+
+
+static int snd_echo_dev_free(struct snd_device *device)
+{
+ struct echoaudio *chip = device->device_data;
+
+ DE_INIT(("snd_echo_dev_free()...\n"));
+ return snd_echo_free(chip);
+}
+
+
+
+/* <--snd_echo_probe() */
+static __devinit int snd_echo_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct echoaudio **rchip)
+{
+ struct echoaudio *chip;
+ int err;
+ size_t sz;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_echo_dev_free,
+ };
+
+ *rchip = NULL;
+
+ pci_write_config_byte(pci, PCI_LATENCY_TIMER, 0xC0);
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ pci_set_master(pci);
+
+ /* allocate a chip-specific data */
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (!chip) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ DE_INIT(("chip=%p\n", chip));
+
+ spin_lock_init(&chip->lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* PCI resource allocation */
+ chip->dsp_registers_phys = pci_resource_start(pci, 0);
+ sz = pci_resource_len(pci, 0);
+ if (sz > PAGE_SIZE)
+ sz = PAGE_SIZE; /* We map only the required part */
+
+ if ((chip->iores = request_mem_region(chip->dsp_registers_phys, sz,
+ ECHOCARD_NAME)) == NULL) {
+ snd_echo_free(chip);
+ snd_printk(KERN_ERR "cannot get memory region\n");
+ return -EBUSY;
+ }
+ chip->dsp_registers = (volatile u32 __iomem *)
+ ioremap_nocache(chip->dsp_registers_phys, sz);
+
+ if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
+ ECHOCARD_NAME, chip)) {
+ snd_echo_free(chip);
+ snd_printk(KERN_ERR "cannot grab irq\n");
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ DE_INIT(("pci=%p irq=%d subdev=%04x Init hardware...\n",
+ chip->pci, chip->irq, chip->pci->subsystem_device));
+
+ /* Create the DSP comm page - this is the area of memory used for most
+ of the communication with the DSP, which accesses it via bus mastering */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ sizeof(struct comm_page),
+ &chip->commpage_dma_buf) < 0) {
+ snd_echo_free(chip);
+ snd_printk(KERN_ERR "cannot allocate the comm page\n");
+ return -ENOMEM;
+ }
+ chip->comm_page_phys = chip->commpage_dma_buf.addr;
+ chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area;
+
+ err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
+ if (err) {
+ DE_INIT(("init_hw err=%d\n", err));
+ snd_echo_free(chip);
+ return err;
+ }
+ DE_INIT(("Card init OK\n"));
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_echo_free(chip);
+ return err;
+ }
+ atomic_set(&chip->opencount, 0);
+ mutex_init(&chip->mode_mutex);
+ chip->can_set_rate = 1;
+ *rchip = chip;
+ /* Init done ! */
+ return 0;
+}
+
+
+
+/* constructor */
+static int __devinit snd_echo_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct echoaudio *chip;
+ char *dsp;
+ int i, err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ DE_INIT(("Echoaudio driver starting...\n"));
+ i = 0;
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ snd_card_set_dev(card, &pci->dev);
+
+ if ((err = snd_echo_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "Echo_" ECHOCARD_NAME);
+ strcpy(card->shortname, chip->card_name);
+
+ dsp = "56301";
+ if (pci_id->device == 0x3410)
+ dsp = "56361";
+
+ sprintf(card->longname, "%s rev.%d (DSP%s) at 0x%lx irq %i",
+ card->shortname, pci_id->subdevice & 0x000f, dsp,
+ chip->dsp_registers_phys, chip->irq);
+
+ if ((err = snd_echo_new_pcm(chip)) < 0) {
+ snd_printk(KERN_ERR "new pcm error %d\n", err);
+ snd_card_free(card);
+ return err;
+ }
+
+#ifdef ECHOCARD_HAS_MIDI
+ if (chip->has_midi) { /* Some Mia's do not have midi */
+ if ((err = snd_echo_midi_create(card, chip)) < 0) {
+ snd_printk(KERN_ERR "new midi error %d\n", err);
+ snd_card_free(card);
+ return err;
+ }
+ }
+#endif
+
+#ifdef ECHOCARD_HAS_VMIXER
+ snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0)
+ goto ctl_error;
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
+ goto ctl_error;
+#else
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
+ goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
+ goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+ if (!chip->hasnt_input_nominal_level)
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_intput_nominal_level, chip))) < 0)
+ goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_output_nominal_level, chip))) < 0)
+ goto ctl_error;
+#endif
+
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters_switch, chip))) < 0)
+ goto ctl_error;
+
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters, chip))) < 0)
+ goto ctl_error;
+
+#ifdef ECHOCARD_HAS_MONITOR
+ snd_echo_monitor_mixer.count = num_busses_in(chip) * num_busses_out(chip);
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_monitor_mixer, chip))) < 0)
+ goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_automute_switch, chip))) < 0)
+ goto ctl_error;
+#endif
+
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_channels_info, chip))) < 0)
+ goto ctl_error;
+
+#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+ /* Creates a list of available digital modes */
+ chip->num_digital_modes = 0;
+ for (i = 0; i < 6; i++)
+ if (chip->digital_modes & (1 << i))
+ chip->digital_mode_list[chip->num_digital_modes++] = i;
+
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_digital_mode_switch, chip))) < 0)
+ goto ctl_error;
+#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
+
+#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
+ /* Creates a list of available clock sources */
+ chip->num_clock_sources = 0;
+ for (i = 0; i < 10; i++)
+ if (chip->input_clock_types & (1 << i))
+ chip->clock_source_list[chip->num_clock_sources++] = i;
+
+ if (chip->num_clock_sources > 1) {
+ chip->clock_src_ctl = snd_ctl_new1(&snd_echo_clock_source_switch, chip);
+ if ((err = snd_ctl_add(chip->card, chip->clock_src_ctl)) < 0)
+ goto ctl_error;
+ }
+#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_spdif_mode_switch, chip))) < 0)
+ goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_PHANTOM_POWER
+ if (chip->has_phantom_power)
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_phantom_power_switch, chip))) < 0)
+ goto ctl_error;
+#endif
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ goto ctl_error;
+ }
+ snd_printk(KERN_INFO "Card registered: %s\n", card->longname);
+
+ pci_set_drvdata(pci, chip);
+ dev++;
+ return 0;
+
+ctl_error:
+ snd_printk(KERN_ERR "new control error %d\n", err);
+ snd_card_free(card);
+ return err;
+}
+
+
+
+static void __devexit snd_echo_remove(struct pci_dev *pci)
+{
+ struct echoaudio *chip;
+
+ chip = pci_get_drvdata(pci);
+ if (chip)
+ snd_card_free(chip->card);
+ pci_set_drvdata(pci, NULL);
+}
+
+
+
+/******************************************************************************
+ Everything starts and ends here
+******************************************************************************/
+
+/* pci_driver definition */
+static struct pci_driver driver = {
+ .name = "Echoaudio " ECHOCARD_NAME,
+ .id_table = snd_echo_ids,
+ .probe = snd_echo_probe,
+ .remove = __devexit_p(snd_echo_remove),
+};
+
+
+
+/* initialization of the module */
+static int __init alsa_card_echo_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+
+
+/* clean up the module */
+static void __exit alsa_card_echo_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+
+module_init(alsa_card_echo_init)
+module_exit(alsa_card_echo_exit)
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
new file mode 100644
index 0000000..1c88e05
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -0,0 +1,590 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ ****************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+ ****************************************************************************
+
+
+ Here's a block diagram of how most of the cards work:
+
+ +-----------+
+ record | |<-------------------- Inputs
+ <-------| | |
+ PCI | Transport | |
+ bus | engine | \|/
+ ------->| | +-------+
+ play | |--->|monitor|-------> Outputs
+ +-----------+ | mixer |
+ +-------+
+
+ The lines going to and from the PCI bus represent "pipes". A pipe performs
+ audio transport - moving audio data to and from buffers on the host via
+ bus mastering.
+
+ The inputs and outputs on the right represent input and output "busses."
+ A bus is a physical, real connection to the outside world. An example
+ of a bus would be the 1/4" analog connectors on the back of Layla or
+ an RCA S/PDIF connector.
+
+ For most cards, there is a one-to-one correspondence between outputs
+ and busses; that is, each individual pipe is hard-wired to a single bus.
+
+ Cards that work this way are Darla20, Gina20, Layla20, Darla24, Gina24,
+ Layla24, Mona, and Indigo.
+
+
+ Mia has a feature called "virtual outputs."
+
+
+ +-----------+
+ record | |<----------------------------- Inputs
+ <-------| | |
+ PCI | Transport | |
+ bus | engine | \|/
+ ------->| | +------+ +-------+
+ play | |-->|vmixer|-->|monitor|-------> Outputs
+ +-----------+ +------+ | mixer |
+ +-------+
+
+
+ Obviously, the difference here is the box labeled "vmixer." Vmixer is
+ short for "virtual output mixer." For Mia, pipes are *not* hard-wired
+ to a single bus; the vmixer lets you mix any pipe to any bus in any
+ combination.
+
+ Note, however, that the left-hand side of the diagram is unchanged.
+ Transport works exactly the same way - the difference is in the mixer stage.
+
+
+ Pipes and busses are numbered starting at zero.
+
+
+
+ Pipe index
+ ==========
+
+ A number of calls in CEchoGals refer to a "pipe index". A pipe index is
+ a unique number for a pipe that unambiguously refers to a playback or record
+ pipe. Pipe indices are numbered starting with analog outputs, followed by
+ digital outputs, then analog inputs, then digital inputs.
+
+ Take Gina24 as an example:
+
+ Pipe index
+
+ 0-7 Analog outputs (0 .. FirstDigitalBusOut-1)
+ 8-15 Digital outputs (FirstDigitalBusOut .. NumBussesOut-1)
+ 16-17 Analog inputs
+ 18-25 Digital inputs
+
+
+ You get the pipe index by calling CEchoGals::OpenAudio; the other transport
+ functions take the pipe index as a parameter. If you need a pipe index for
+ some other reason, use the handy Makepipe_index method.
+
+
+ Some calls take a CChannelMask parameter; CChannelMask is a handy way to
+ group pipe indices.
+
+
+
+ Digital mode switch
+ ===================
+
+ Some cards (right now, Gina24, Layla24, and Mona) have a Digital Mode Switch
+ or DMS. Cards with a DMS can be set to one of three mutually exclusive
+ digital modes: S/PDIF RCA, S/PDIF optical, or ADAT optical.
+
+ This may create some confusion since ADAT optical is 8 channels wide and
+ S/PDIF is only two channels wide. Gina24, Layla24, and Mona handle this
+ by acting as if they always have 8 digital outs and ins. If you are in
+ either S/PDIF mode, the last 6 channels don't do anything - data sent
+ out these channels is thrown away and you will always record zeros.
+
+ Note that with Gina24, Layla24, and Mona, sample rates above 50 kHz are
+ only available if you have the card configured for S/PDIF optical or S/PDIF
+ RCA.
+
+
+
+ Double speed mode
+ =================
+
+ Some of the cards support 88.2 kHz and 96 kHz sampling (Darla24, Gina24,
+ Layla24, Mona, Mia, and Indigo). For these cards, the driver sometimes has
+ to worry about "double speed mode"; double speed mode applies whenever the
+ sampling rate is above 50 kHz.
+
+ For instance, Mona and Layla24 support word clock sync. However, they
+ actually support two different word clock modes - single speed (below
+ 50 kHz) and double speed (above 50 kHz). The hardware detects if a single
+ or double speed word clock signal is present; the generic code uses that
+ information to determine which mode to use.
+
+ The generic code takes care of all this for you.
+*/
+
+
+#ifndef _ECHOAUDIO_H_
+#define _ECHOAUDIO_H_
+
+
+#define TRUE 1
+#define FALSE 0
+
+#include "echoaudio_dsp.h"
+
+
+
+/***********************************************************************
+
+ PCI configuration space
+
+***********************************************************************/
+
+/*
+ * PCI vendor ID and device IDs for the hardware
+ */
+#define VENDOR_ID 0x1057
+#define DEVICE_ID_56301 0x1801
+#define DEVICE_ID_56361 0x3410
+#define SUBVENDOR_ID 0xECC0
+
+
+/*
+ * Valid Echo PCI subsystem card IDs
+ */
+#define DARLA20 0x0010
+#define GINA20 0x0020
+#define LAYLA20 0x0030
+#define DARLA24 0x0040
+#define GINA24 0x0050
+#define LAYLA24 0x0060
+#define MONA 0x0070
+#define MIA 0x0080
+#define INDIGO 0x0090
+#define INDIGO_IO 0x00a0
+#define INDIGO_DJ 0x00b0
+#define ECHO3G 0x0100
+
+
+/************************************************************************
+
+ Array sizes and so forth
+
+***********************************************************************/
+
+/*
+ * Sizes
+ */
+#define ECHO_MAXAUDIOINPUTS 32 /* Max audio input channels */
+#define ECHO_MAXAUDIOOUTPUTS 32 /* Max audio output channels */
+#define ECHO_MAXAUDIOPIPES 32 /* Max number of input and output
+ * pipes */
+#define E3G_MAX_OUTPUTS 16
+#define ECHO_MAXMIDIJACKS 1 /* Max MIDI ports */
+#define ECHO_MIDI_QUEUE_SZ 512 /* Max MIDI input queue entries */
+#define ECHO_MTC_QUEUE_SZ 32 /* Max MIDI time code input queue
+ * entries */
+
+/*
+ * MIDI activity indicator timeout
+ */
+#define MIDI_ACTIVITY_TIMEOUT_USEC 200000
+
+
+/****************************************************************************
+
+ Clocks
+
+*****************************************************************************/
+
+/*
+ * Clock numbers
+ */
+#define ECHO_CLOCK_INTERNAL 0
+#define ECHO_CLOCK_WORD 1
+#define ECHO_CLOCK_SUPER 2
+#define ECHO_CLOCK_SPDIF 3
+#define ECHO_CLOCK_ADAT 4
+#define ECHO_CLOCK_ESYNC 5
+#define ECHO_CLOCK_ESYNC96 6
+#define ECHO_CLOCK_MTC 7
+#define ECHO_CLOCK_NUMBER 8
+#define ECHO_CLOCKS 0xffff
+
+/*
+ * Clock bit numbers - used to report capabilities and whatever clocks
+ * are being detected dynamically.
+ */
+#define ECHO_CLOCK_BIT_INTERNAL (1 << ECHO_CLOCK_INTERNAL)
+#define ECHO_CLOCK_BIT_WORD (1 << ECHO_CLOCK_WORD)
+#define ECHO_CLOCK_BIT_SUPER (1 << ECHO_CLOCK_SUPER)
+#define ECHO_CLOCK_BIT_SPDIF (1 << ECHO_CLOCK_SPDIF)
+#define ECHO_CLOCK_BIT_ADAT (1 << ECHO_CLOCK_ADAT)
+#define ECHO_CLOCK_BIT_ESYNC (1 << ECHO_CLOCK_ESYNC)
+#define ECHO_CLOCK_BIT_ESYNC96 (1 << ECHO_CLOCK_ESYNC96)
+#define ECHO_CLOCK_BIT_MTC (1<<ECHO_CLOCK_MTC)
+
+
+/***************************************************************************
+
+ Digital modes
+
+****************************************************************************/
+
+/*
+ * Digital modes for Mona, Layla24, and Gina24
+ */
+#define DIGITAL_MODE_NONE 0xFF
+#define DIGITAL_MODE_SPDIF_RCA 0
+#define DIGITAL_MODE_SPDIF_OPTICAL 1
+#define DIGITAL_MODE_ADAT 2
+#define DIGITAL_MODE_SPDIF_CDROM 3
+#define DIGITAL_MODES 4
+
+/*
+ * Digital mode capability masks
+ */
+#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA (1 << DIGITAL_MODE_SPDIF_RCA)
+#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL (1 << DIGITAL_MODE_SPDIF_OPTICAL)
+#define ECHOCAPS_HAS_DIGITAL_MODE_ADAT (1 << DIGITAL_MODE_ADAT)
+#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_CDROM (1 << DIGITAL_MODE_SPDIF_CDROM)
+
+
+#define EXT_3GBOX_NC 0x01 /* 3G box not connected */
+#define EXT_3GBOX_NOT_SET 0x02 /* 3G box not detected yet */
+
+
+#define ECHOGAIN_MUTED (-128) /* Minimum possible gain */
+#define ECHOGAIN_MINOUT (-128) /* Min output gain (dB) */
+#define ECHOGAIN_MAXOUT (6) /* Max output gain (dB) */
+#define ECHOGAIN_MININP (-50) /* Min input gain (0.5 dB) */
+#define ECHOGAIN_MAXINP (50) /* Max input gain (0.5 dB) */
+
+#define PIPE_STATE_STOPPED 0 /* Pipe has been reset */
+#define PIPE_STATE_PAUSED 1 /* Pipe has been stopped */
+#define PIPE_STATE_STARTED 2 /* Pipe has been started */
+#define PIPE_STATE_PENDING 3 /* Pipe has pending start */
+
+
+/* Debug initialization */
+#ifdef CONFIG_SND_DEBUG
+#define DE_INIT(x) snd_printk x
+#else
+#define DE_INIT(x)
+#endif
+
+/* Debug hw_params callbacks */
+#ifdef CONFIG_SND_DEBUG
+#define DE_HWP(x) snd_printk x
+#else
+#define DE_HWP(x)
+#endif
+
+/* Debug normal activity (open, start, stop...) */
+#ifdef CONFIG_SND_DEBUG
+#define DE_ACT(x) snd_printk x
+#else
+#define DE_ACT(x)
+#endif
+
+/* Debug midi activity */
+#ifdef CONFIG_SND_DEBUG
+#define DE_MID(x) snd_printk x
+#else
+#define DE_MID(x)
+#endif
+
+
+struct audiopipe {
+ volatile u32 *dma_counter; /* Commpage register that contains
+ * the current dma position
+ * (lower 32 bits only)
+ */
+ u32 last_counter; /* The last position, which is used
+ * to compute...
+ */
+ u32 position; /* ...the number of bytes tranferred
+ * by the DMA engine, modulo the
+ * buffer size
+ */
+ short index; /* Index of the first channel or <0
+ * if hw is not configured yet
+ */
+ short interleave;
+ struct snd_dma_buffer sgpage; /* Room for the scatter-gather list */
+ struct snd_pcm_hardware hw;
+ struct snd_pcm_hw_constraint_list constr;
+ short sglist_head;
+ char state; /* pipe state */
+};
+
+
+struct audioformat {
+ u8 interleave; /* How the data is arranged in memory:
+ * mono = 1, stereo = 2, ...
+ */
+ u8 bits_per_sample; /* 8, 16, 24, 32 (24 bits left aligned) */
+ char mono_to_stereo; /* Only used if interleave is 1 and
+ * if this is an output pipe.
+ */
+ char data_are_bigendian; /* 1 = big endian, 0 = little endian */
+};
+
+
+struct echoaudio {
+ spinlock_t lock;
+ struct snd_pcm_substream *substream[DSP_MAXPIPES];
+ int last_period[DSP_MAXPIPES];
+ struct mutex mode_mutex;
+ u16 num_digital_modes, digital_mode_list[6];
+ u16 num_clock_sources, clock_source_list[10];
+ atomic_t opencount;
+ struct snd_kcontrol *clock_src_ctl;
+ struct snd_pcm *analog_pcm, *digital_pcm;
+ struct snd_card *card;
+ const char *card_name;
+ struct pci_dev *pci;
+ unsigned long dsp_registers_phys;
+ struct resource *iores;
+ struct snd_dma_buffer commpage_dma_buf;
+ int irq;
+#ifdef ECHOCARD_HAS_MIDI
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *midi_in, *midi_out;
+#endif
+ struct timer_list timer;
+ char tinuse; /* Timer in use */
+ char midi_full; /* MIDI output buffer is full */
+ char can_set_rate;
+ char rate_set;
+
+ /* This stuff is used mainly by the lowlevel code */
+ struct comm_page *comm_page; /* Virtual address of the memory
+ * seen by DSP
+ */
+ u32 pipe_alloc_mask; /* Bitmask of allocated pipes */
+ u32 pipe_cyclic_mask; /* Bitmask of pipes with cyclic
+ * buffers
+ */
+ u32 sample_rate; /* Card sample rate in Hz */
+ u8 digital_mode; /* Current digital mode
+ * (see DIGITAL_MODE_*)
+ */
+ u8 spdif_status; /* Gina20, Darla20, Darla24 - only */
+ u8 clock_state; /* Gina20, Darla20, Darla24 - only */
+ u8 input_clock; /* Currently selected sample clock
+ * source
+ */
+ u8 output_clock; /* Layla20 only */
+ char meters_enabled; /* VU-meters status */
+ char asic_loaded; /* Set TRUE when ASIC loaded */
+ char bad_board; /* Set TRUE if DSP won't load */
+ char professional_spdif; /* 0 = consumer; 1 = professional */
+ char non_audio_spdif; /* 3G - only */
+ char digital_in_automute; /* Gina24, Layla24, Mona - only */
+ char has_phantom_power;
+ char hasnt_input_nominal_level; /* Gina3G */
+ char phantom_power; /* Gina3G - only */
+ char has_midi;
+ char midi_input_enabled;
+
+#ifdef ECHOCARD_ECHO3G
+ /* External module -dependent pipe and bus indexes */
+ char px_digital_out, px_analog_in, px_digital_in, px_num;
+ char bx_digital_out, bx_analog_in, bx_digital_in, bx_num;
+#endif
+
+ char nominal_level[ECHO_MAXAUDIOPIPES]; /* True == -10dBV
+ * False == +4dBu */
+ s8 input_gain[ECHO_MAXAUDIOINPUTS]; /* Input level -50..+50
+ * unit is 0.5dB */
+ s8 output_gain[ECHO_MAXAUDIOOUTPUTS]; /* Output level -128..+6 dB
+ * (-128=muted) */
+ s8 monitor_gain[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOINPUTS];
+ /* -128..+6 dB */
+ s8 vmixer_gain[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOOUTPUTS];
+ /* -128..+6 dB */
+
+ u16 digital_modes; /* Bitmask of supported modes
+ * (see ECHOCAPS_HAS_DIGITAL_MODE_*) */
+ u16 input_clock_types; /* Suppoted input clock types */
+ u16 output_clock_types; /* Suppoted output clock types -
+ * Layla20 only */
+ u16 device_id, subdevice_id;
+ u16 *dsp_code; /* Current DSP code loaded,
+ * NULL if nothing loaded */
+ const struct firmware *dsp_code_to_load;/* DSP code to load */
+ const struct firmware *asic_code; /* Current ASIC code */
+ u32 comm_page_phys; /* Physical address of the
+ * memory seen by DSP */
+ volatile u32 __iomem *dsp_registers; /* DSP's register base */
+ u32 active_mask; /* Chs. active mask or
+ * punks out */
+
+#ifdef ECHOCARD_HAS_MIDI
+ u16 mtc_state; /* State for MIDI input parsing state machine */
+ u8 midi_buffer[MIDI_IN_BUFFER_SIZE];
+#endif
+};
+
+
+static int init_dsp_comm_page(struct echoaudio *chip);
+static int init_line_levels(struct echoaudio *chip);
+static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe);
+static int load_firmware(struct echoaudio *chip);
+static int wait_handshake(struct echoaudio *chip);
+static int send_vector(struct echoaudio *chip, u32 command);
+static int get_firmware(const struct firmware **fw_entry,
+ const struct firmware *frm, struct echoaudio *chip);
+static void free_firmware(const struct firmware *fw_entry);
+
+#ifdef ECHOCARD_HAS_MIDI
+static int enable_midi_input(struct echoaudio *chip, char enable);
+static int midi_service_irq(struct echoaudio *chip);
+static int __devinit snd_echo_midi_create(struct snd_card *card,
+ struct echoaudio *chip);
+#endif
+
+
+static inline void clear_handshake(struct echoaudio *chip)
+{
+ chip->comm_page->handshake = 0;
+}
+
+static inline u32 get_dsp_register(struct echoaudio *chip, u32 index)
+{
+ return readl(&chip->dsp_registers[index]);
+}
+
+static inline void set_dsp_register(struct echoaudio *chip, u32 index,
+ u32 value)
+{
+ writel(value, &chip->dsp_registers[index]);
+}
+
+
+/* Pipe and bus indexes. PX_* and BX_* are defined as chip->px_* and chip->bx_*
+for 3G cards because they depend on the external box. They are integer
+constants for all other cards.
+Never use those defines directly, use the following functions instead. */
+
+static inline int px_digital_out(const struct echoaudio *chip)
+{
+ return PX_DIGITAL_OUT;
+}
+
+static inline int px_analog_in(const struct echoaudio *chip)
+{
+ return PX_ANALOG_IN;
+}
+
+static inline int px_digital_in(const struct echoaudio *chip)
+{
+ return PX_DIGITAL_IN;
+}
+
+static inline int px_num(const struct echoaudio *chip)
+{
+ return PX_NUM;
+}
+
+static inline int bx_digital_out(const struct echoaudio *chip)
+{
+ return BX_DIGITAL_OUT;
+}
+
+static inline int bx_analog_in(const struct echoaudio *chip)
+{
+ return BX_ANALOG_IN;
+}
+
+static inline int bx_digital_in(const struct echoaudio *chip)
+{
+ return BX_DIGITAL_IN;
+}
+
+static inline int bx_num(const struct echoaudio *chip)
+{
+ return BX_NUM;
+}
+
+static inline int num_pipes_out(const struct echoaudio *chip)
+{
+ return px_analog_in(chip);
+}
+
+static inline int num_pipes_in(const struct echoaudio *chip)
+{
+ return px_num(chip) - px_analog_in(chip);
+}
+
+static inline int num_busses_out(const struct echoaudio *chip)
+{
+ return bx_analog_in(chip);
+}
+
+static inline int num_busses_in(const struct echoaudio *chip)
+{
+ return bx_num(chip) - bx_analog_in(chip);
+}
+
+static inline int num_analog_busses_out(const struct echoaudio *chip)
+{
+ return bx_digital_out(chip);
+}
+
+static inline int num_analog_busses_in(const struct echoaudio *chip)
+{
+ return bx_digital_in(chip) - bx_analog_in(chip);
+}
+
+static inline int num_digital_busses_out(const struct echoaudio *chip)
+{
+ return num_busses_out(chip) - num_analog_busses_out(chip);
+}
+
+static inline int num_digital_busses_in(const struct echoaudio *chip)
+{
+ return num_busses_in(chip) - num_analog_busses_in(chip);
+}
+
+/* The monitor array is a one-dimensional array; compute the offset
+ * into the array */
+static inline int monitor_index(const struct echoaudio *chip, int out, int in)
+{
+ return out * num_busses_in(chip) + in;
+}
+
+
+#ifndef pci_device
+#define pci_device(chip) (&chip->pci->dev)
+#endif
+
+
+#endif /* _ECHOAUDIO_H_ */
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
new file mode 100644
index 0000000..c3736bb
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -0,0 +1,434 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+
+/* These functions are common for all "3G" cards */
+
+
+static int check_asic_status(struct echoaudio *chip)
+{
+ u32 box_status;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->ext_box_status =
+ __constant_cpu_to_le32(E3G_ASIC_NOT_LOADED);
+ chip->asic_loaded = FALSE;
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_TEST_ASIC);
+
+ if (wait_handshake(chip)) {
+ chip->dsp_code = NULL;
+ return -EIO;
+ }
+
+ box_status = le32_to_cpu(chip->comm_page->ext_box_status);
+ DE_INIT(("box_status=%x\n", box_status));
+ if (box_status == E3G_ASIC_NOT_LOADED)
+ return -ENODEV;
+
+ chip->asic_loaded = TRUE;
+ return box_status & E3G_BOX_TYPE_MASK;
+}
+
+
+
+static inline u32 get_frq_reg(struct echoaudio *chip)
+{
+ return le32_to_cpu(chip->comm_page->e3g_frq_register);
+}
+
+
+
+/* Most configuration of 3G cards is accomplished by writing the control
+register. write_control_reg sends the new control register value to the DSP. */
+static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
+ char force)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+
+ DE_ACT(("WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq));
+
+ ctl = cpu_to_le32(ctl);
+ frq = cpu_to_le32(frq);
+
+ if (ctl != chip->comm_page->control_register ||
+ frq != chip->comm_page->e3g_frq_register || force) {
+ chip->comm_page->e3g_frq_register = frq;
+ chip->comm_page->control_register = ctl;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
+ }
+
+ DE_ACT(("WriteControlReg: not written, no change\n"));
+ return 0;
+}
+
+
+
+/* Set the digital mode - currently for Gina24, Layla24, Mona, 3G */
+static int set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+ u8 previous_mode;
+ int err, i, o;
+
+ /* All audio channels must be closed before changing the digital mode */
+ if (snd_BUG_ON(chip->pipe_alloc_mask))
+ return -EAGAIN;
+
+ if (snd_BUG_ON(!(chip->digital_modes & (1 << mode))))
+ return -EINVAL;
+
+ previous_mode = chip->digital_mode;
+ err = dsp_set_digital_mode(chip, mode);
+
+ /* If we successfully changed the digital mode from or to ADAT,
+ * then make sure all output, input and monitor levels are
+ * updated by the DSP comm object. */
+ if (err >= 0 && previous_mode != mode &&
+ (previous_mode == DIGITAL_MODE_ADAT || mode == DIGITAL_MODE_ADAT)) {
+ spin_lock_irq(&chip->lock);
+ for (o = 0; o < num_busses_out(chip); o++)
+ for (i = 0; i < num_busses_in(chip); i++)
+ set_monitor_gain(chip, o, i,
+ chip->monitor_gain[o][i]);
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+ for (i = 0; i < num_busses_in(chip); i++)
+ set_input_gain(chip, i, chip->input_gain[i]);
+ update_input_line_level(chip);
+#endif
+
+ for (o = 0; o < num_busses_out(chip); o++)
+ set_output_gain(chip, o, chip->output_gain[o]);
+ update_output_line_level(chip);
+ spin_unlock_irq(&chip->lock);
+ }
+
+ return err;
+}
+
+
+
+static u32 set_spdif_bits(struct echoaudio *chip, u32 control_reg, u32 rate)
+{
+ control_reg &= E3G_SPDIF_FORMAT_CLEAR_MASK;
+
+ switch (rate) {
+ case 32000 :
+ control_reg |= E3G_SPDIF_SAMPLE_RATE0 | E3G_SPDIF_SAMPLE_RATE1;
+ break;
+ case 44100 :
+ if (chip->professional_spdif)
+ control_reg |= E3G_SPDIF_SAMPLE_RATE0;
+ break;
+ case 48000 :
+ control_reg |= E3G_SPDIF_SAMPLE_RATE1;
+ break;
+ }
+
+ if (chip->professional_spdif)
+ control_reg |= E3G_SPDIF_PRO_MODE;
+
+ if (chip->non_audio_spdif)
+ control_reg |= E3G_SPDIF_NOT_AUDIO;
+
+ control_reg |= E3G_SPDIF_24_BIT | E3G_SPDIF_TWO_CHANNEL |
+ E3G_SPDIF_COPY_PERMIT;
+
+ return control_reg;
+}
+
+
+
+/* Set the S/PDIF output format */
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+ u32 control_reg;
+
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ chip->professional_spdif = prof;
+ control_reg = set_spdif_bits(chip, control_reg, chip->sample_rate);
+ return write_control_reg(chip, control_reg, get_frq_reg(chip), 0);
+}
+
+
+
+/* detect_input_clocks() returns a bitmask consisting of all the input clocks
+currently connected to the hardware; this changes as the user connects and
+disconnects clock inputs. You should use this information to determine which
+clocks the user is allowed to select. */
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock
+ * detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD)
+ clock_bits |= ECHO_CLOCK_BIT_WORD;
+
+ switch(chip->digital_mode) {
+ case DIGITAL_MODE_SPDIF_RCA:
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF)
+ clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+ break;
+ case DIGITAL_MODE_ADAT:
+ if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_ADAT)
+ clock_bits |= ECHO_CLOCK_BIT_ADAT;
+ break;
+ }
+
+ return clock_bits;
+}
+
+
+
+static int load_asic(struct echoaudio *chip)
+{
+ int box_type, err;
+
+ if (chip->asic_loaded)
+ return 0;
+
+ /* Give the DSP a few milliseconds to settle down */
+ mdelay(2);
+
+ err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC,
+ &card_fw[FW_3G_ASIC]);
+ if (err < 0)
+ return err;
+
+ chip->asic_code = &card_fw[FW_3G_ASIC];
+
+ /* Now give the new ASIC some time to set up */
+ msleep(1000);
+ /* See if it worked */
+ box_type = check_asic_status(chip);
+
+ /* Set up the control register if the load succeeded -
+ * 48 kHz, internal clock, S/PDIF RCA mode */
+ if (box_type >= 0) {
+ err = write_control_reg(chip, E3G_48KHZ,
+ E3G_FREQ_REG_DEFAULT, TRUE);
+ if (err < 0)
+ return err;
+ }
+
+ return box_type;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u32 control_reg, clock, base_rate, frq_reg;
+
+ /* Only set the clock for internal mode. */
+ if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+ DE_ACT(("set_sample_rate: Cannot set sample rate - "
+ "clock not set to CLK_CLOCKININTERNAL\n"));
+ /* Save the rate anyhow */
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ chip->sample_rate = rate;
+ set_input_clock(chip, chip->input_clock);
+ return 0;
+ }
+
+ if (snd_BUG_ON(rate >= 50000 &&
+ chip->digital_mode == DIGITAL_MODE_ADAT))
+ return -EINVAL;
+
+ clock = 0;
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= E3G_CLOCK_CLEAR_MASK;
+
+ switch (rate) {
+ case 96000:
+ clock = E3G_96KHZ;
+ break;
+ case 88200:
+ clock = E3G_88KHZ;
+ break;
+ case 48000:
+ clock = E3G_48KHZ;
+ break;
+ case 44100:
+ clock = E3G_44KHZ;
+ break;
+ case 32000:
+ clock = E3G_32KHZ;
+ break;
+ default:
+ clock = E3G_CONTINUOUS_CLOCK;
+ if (rate > 50000)
+ clock |= E3G_DOUBLE_SPEED_MODE;
+ break;
+ }
+
+ control_reg |= clock;
+ control_reg = set_spdif_bits(chip, control_reg, rate);
+
+ base_rate = rate;
+ if (base_rate > 50000)
+ base_rate /= 2;
+ if (base_rate < 32000)
+ base_rate = 32000;
+
+ frq_reg = E3G_MAGIC_NUMBER / base_rate - 2;
+ if (frq_reg > E3G_FREQ_REG_MAX)
+ frq_reg = E3G_FREQ_REG_MAX;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
+ chip->sample_rate = rate;
+ DE_ACT(("SetSampleRate: %d clock %x\n", rate, control_reg));
+
+ /* Tell the DSP about it - DSP reads both control reg & freq reg */
+ return write_control_reg(chip, control_reg, frq_reg, 0);
+}
+
+
+
+/* Set the sample clock source to internal, S/PDIF, ADAT */
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+ u32 control_reg, clocks_from_dsp;
+
+ DE_ACT(("set_input_clock:\n"));
+
+ /* Mask off the clock select bits */
+ control_reg = le32_to_cpu(chip->comm_page->control_register) &
+ E3G_CLOCK_CLEAR_MASK;
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ switch (clock) {
+ case ECHO_CLOCK_INTERNAL:
+ DE_ACT(("Set Echo3G clock to INTERNAL\n"));
+ chip->input_clock = ECHO_CLOCK_INTERNAL;
+ return set_sample_rate(chip, chip->sample_rate);
+ case ECHO_CLOCK_SPDIF:
+ if (chip->digital_mode == DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ DE_ACT(("Set Echo3G clock to SPDIF\n"));
+ control_reg |= E3G_SPDIF_CLOCK;
+ if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96)
+ control_reg |= E3G_DOUBLE_SPEED_MODE;
+ else
+ control_reg &= ~E3G_DOUBLE_SPEED_MODE;
+ break;
+ case ECHO_CLOCK_ADAT:
+ if (chip->digital_mode != DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ DE_ACT(("Set Echo3G clock to ADAT\n"));
+ control_reg |= E3G_ADAT_CLOCK;
+ control_reg &= ~E3G_DOUBLE_SPEED_MODE;
+ break;
+ case ECHO_CLOCK_WORD:
+ DE_ACT(("Set Echo3G clock to WORD\n"));
+ control_reg |= E3G_WORD_CLOCK;
+ if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96)
+ control_reg |= E3G_DOUBLE_SPEED_MODE;
+ else
+ control_reg &= ~E3G_DOUBLE_SPEED_MODE;
+ break;
+ default:
+ DE_ACT(("Input clock 0x%x not supported for Echo3G\n", clock));
+ return -EINVAL;
+ }
+
+ chip->input_clock = clock;
+ return write_control_reg(chip, control_reg, get_frq_reg(chip), 1);
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+ u32 control_reg;
+ int err, incompatible_clock;
+
+ /* Set clock to "internal" if it's not compatible with the new mode */
+ incompatible_clock = FALSE;
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ case DIGITAL_MODE_SPDIF_RCA:
+ if (chip->input_clock == ECHO_CLOCK_ADAT)
+ incompatible_clock = TRUE;
+ break;
+ case DIGITAL_MODE_ADAT:
+ if (chip->input_clock == ECHO_CLOCK_SPDIF)
+ incompatible_clock = TRUE;
+ break;
+ default:
+ DE_ACT(("Digital mode not supported: %d\n", mode));
+ return -EINVAL;
+ }
+
+ spin_lock_irq(&chip->lock);
+
+ if (incompatible_clock) {
+ chip->sample_rate = 48000;
+ set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+ }
+
+ /* Clear the current digital mode */
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= E3G_DIGITAL_MODE_CLEAR_MASK;
+
+ /* Tweak the control reg */
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ control_reg |= E3G_SPDIF_OPTICAL_MODE;
+ break;
+ case DIGITAL_MODE_SPDIF_RCA:
+ /* E3G_SPDIF_OPTICAL_MODE bit cleared */
+ break;
+ case DIGITAL_MODE_ADAT:
+ control_reg |= E3G_ADAT_MODE;
+ control_reg &= ~E3G_DOUBLE_SPEED_MODE; /* @@ useless */
+ break;
+ }
+
+ err = write_control_reg(chip, control_reg, get_frq_reg(chip), 1);
+ spin_unlock_irq(&chip->lock);
+ if (err < 0)
+ return err;
+ chip->digital_mode = mode;
+
+ DE_ACT(("set_digital_mode(%d)\n", chip->digital_mode));
+ return incompatible_clock;
+}
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
new file mode 100644
index 0000000..be0e181
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -0,0 +1,1130 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+#if PAGE_SIZE < 4096
+#error PAGE_SIZE is < 4k
+#endif
+
+static int restore_dsp_rettings(struct echoaudio *chip);
+
+
+/* Some vector commands involve the DSP reading or writing data to and from the
+comm page; if you send one of these commands to the DSP, it will complete the
+command and then write a non-zero value to the Handshake field in the
+comm page. This function waits for the handshake to show up. */
+static int wait_handshake(struct echoaudio *chip)
+{
+ int i;
+
+ /* Wait up to 20ms for the handshake from the DSP */
+ for (i = 0; i < HANDSHAKE_TIMEOUT; i++) {
+ /* Look for the handshake value */
+ barrier();
+ if (chip->comm_page->handshake) {
+ return 0;
+ }
+ udelay(1);
+ }
+
+ snd_printk(KERN_ERR "wait_handshake(): Timeout waiting for DSP\n");
+ return -EBUSY;
+}
+
+
+
+/* Much of the interaction between the DSP and the driver is done via vector
+commands; send_vector writes a vector command to the DSP. Typically, this
+causes the DSP to read or write fields in the comm page.
+PCI posting is not required thanks to the handshake logic. */
+static int send_vector(struct echoaudio *chip, u32 command)
+{
+ int i;
+
+ wmb(); /* Flush all pending writes before sending the command */
+
+ /* Wait up to 100ms for the "vector busy" bit to be off */
+ for (i = 0; i < VECTOR_BUSY_TIMEOUT; i++) {
+ if (!(get_dsp_register(chip, CHI32_VECTOR_REG) &
+ CHI32_VECTOR_BUSY)) {
+ set_dsp_register(chip, CHI32_VECTOR_REG, command);
+ /*if (i) DE_ACT(("send_vector time: %d\n", i));*/
+ return 0;
+ }
+ udelay(1);
+ }
+
+ DE_ACT((KERN_ERR "timeout on send_vector\n"));
+ return -EBUSY;
+}
+
+
+
+/* write_dsp writes a 32-bit value to the DSP; this is used almost
+exclusively for loading the DSP. */
+static int write_dsp(struct echoaudio *chip, u32 data)
+{
+ u32 status, i;
+
+ for (i = 0; i < 10000000; i++) { /* timeout = 10s */
+ status = get_dsp_register(chip, CHI32_STATUS_REG);
+ if ((status & CHI32_STATUS_HOST_WRITE_EMPTY) != 0) {
+ set_dsp_register(chip, CHI32_DATA_REG, data);
+ wmb(); /* write it immediately */
+ return 0;
+ }
+ udelay(1);
+ cond_resched();
+ }
+
+ chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */
+ DE_ACT((KERN_ERR "write_dsp: Set bad_board to TRUE\n"));
+ return -EIO;
+}
+
+
+
+/* read_dsp reads a 32-bit value from the DSP; this is used almost
+exclusively for loading the DSP and checking the status of the ASIC. */
+static int read_dsp(struct echoaudio *chip, u32 *data)
+{
+ u32 status, i;
+
+ for (i = 0; i < READ_DSP_TIMEOUT; i++) {
+ status = get_dsp_register(chip, CHI32_STATUS_REG);
+ if ((status & CHI32_STATUS_HOST_READ_FULL) != 0) {
+ *data = get_dsp_register(chip, CHI32_DATA_REG);
+ return 0;
+ }
+ udelay(1);
+ cond_resched();
+ }
+
+ chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */
+ DE_INIT((KERN_ERR "read_dsp: Set bad_board to TRUE\n"));
+ return -EIO;
+}
+
+
+
+/****************************************************************************
+ Firmware loading functions
+ ****************************************************************************/
+
+/* This function is used to read back the serial number from the DSP;
+this is triggered by the SET_COMMPAGE_ADDR command.
+Only some early Echogals products have serial numbers in the ROM;
+the serial number is not used, but you still need to do this as
+part of the DSP load process. */
+static int read_sn(struct echoaudio *chip)
+{
+ int i;
+ u32 sn[6];
+
+ for (i = 0; i < 5; i++) {
+ if (read_dsp(chip, &sn[i])) {
+ snd_printk(KERN_ERR "Failed to read serial number\n");
+ return -EIO;
+ }
+ }
+ DE_INIT(("Read serial number %08x %08x %08x %08x %08x\n",
+ sn[0], sn[1], sn[2], sn[3], sn[4]));
+ return 0;
+}
+
+
+
+#ifndef ECHOCARD_HAS_ASIC
+/* This card has no ASIC, just return ok */
+static inline int check_asic_status(struct echoaudio *chip)
+{
+ chip->asic_loaded = TRUE;
+ return 0;
+}
+
+#endif /* !ECHOCARD_HAS_ASIC */
+
+
+
+#ifdef ECHOCARD_HAS_ASIC
+
+/* Load ASIC code - done after the DSP is loaded */
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+ const struct firmware *asic)
+{
+ const struct firmware *fw;
+ int err;
+ u32 i, size;
+ u8 *code;
+
+ if ((err = get_firmware(&fw, asic, chip)) < 0) {
+ snd_printk(KERN_WARNING "Firmware not found !\n");
+ return err;
+ }
+
+ code = (u8 *)fw->data;
+ size = fw->size;
+
+ /* Send the "Here comes the ASIC" command */
+ if (write_dsp(chip, cmd) < 0)
+ goto la_error;
+
+ /* Write length of ASIC file in bytes */
+ if (write_dsp(chip, size) < 0)
+ goto la_error;
+
+ for (i = 0; i < size; i++) {
+ if (write_dsp(chip, code[i]) < 0)
+ goto la_error;
+ }
+
+ DE_INIT(("ASIC loaded\n"));
+ free_firmware(fw);
+ return 0;
+
+la_error:
+ DE_INIT(("failed on write_dsp\n"));
+ free_firmware(fw);
+ return -EIO;
+}
+
+#endif /* ECHOCARD_HAS_ASIC */
+
+
+
+#ifdef DSP_56361
+
+/* Install the resident loader for 56361 DSPs; The resident loader is on
+the EPROM on the board for 56301 DSP. The resident loader is a tiny little
+program that is used to load the real DSP code. */
+static int install_resident_loader(struct echoaudio *chip)
+{
+ u32 address;
+ int index, words, i;
+ u16 *code;
+ u32 status;
+ const struct firmware *fw;
+
+ /* 56361 cards only! This check is required by the old 56301-based
+ Mona and Gina24 */
+ if (chip->device_id != DEVICE_ID_56361)
+ return 0;
+
+ /* Look to see if the resident loader is present. If the resident
+ loader is already installed, host flag 5 will be on. */
+ status = get_dsp_register(chip, CHI32_STATUS_REG);
+ if (status & CHI32_STATUS_REG_HF5) {
+ DE_INIT(("Resident loader already installed; status is 0x%x\n",
+ status));
+ return 0;
+ }
+
+ if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) {
+ snd_printk(KERN_WARNING "Firmware not found !\n");
+ return i;
+ }
+
+ /* The DSP code is an array of 16 bit words. The array is divided up
+ into sections. The first word of each section is the size in words,
+ followed by the section type.
+ Since DSP addresses and data are 24 bits wide, they each take up two
+ 16 bit words in the array.
+ This is a lot like the other loader loop, but it's not a loop, you
+ don't write the memory type, and you don't write a zero at the end. */
+
+ /* Set DSP format bits for 24 bit mode */
+ set_dsp_register(chip, CHI32_CONTROL_REG,
+ get_dsp_register(chip, CHI32_CONTROL_REG) | 0x900);
+
+ code = (u16 *)fw->data;
+
+ /* Skip the header section; the first word in the array is the size
+ of the first section, so the first real section of code is pointed
+ to by Code[0]. */
+ index = code[0];
+
+ /* Skip the section size, LRS block type, and DSP memory type */
+ index += 3;
+
+ /* Get the number of DSP words to write */
+ words = code[index++];
+
+ /* Get the DSP address for this block; 24 bits, so build from two words */
+ address = ((u32)code[index] << 16) + code[index + 1];
+ index += 2;
+
+ /* Write the count to the DSP */
+ if (write_dsp(chip, words)) {
+ DE_INIT(("install_resident_loader: Failed to write word count!\n"));
+ goto irl_error;
+ }
+ /* Write the DSP address */
+ if (write_dsp(chip, address)) {
+ DE_INIT(("install_resident_loader: Failed to write DSP address!\n"));
+ goto irl_error;
+ }
+ /* Write out this block of code to the DSP */
+ for (i = 0; i < words; i++) {
+ u32 data;
+
+ data = ((u32)code[index] << 16) + code[index + 1];
+ if (write_dsp(chip, data)) {
+ DE_INIT(("install_resident_loader: Failed to write DSP code\n"));
+ goto irl_error;
+ }
+ index += 2;
+ }
+
+ /* Wait for flag 5 to come up */
+ for (i = 0; i < 200; i++) { /* Timeout is 50us * 200 = 10ms */
+ udelay(50);
+ status = get_dsp_register(chip, CHI32_STATUS_REG);
+ if (status & CHI32_STATUS_REG_HF5)
+ break;
+ }
+
+ if (i == 200) {
+ DE_INIT(("Resident loader failed to set HF5\n"));
+ goto irl_error;
+ }
+
+ DE_INIT(("Resident loader successfully installed\n"));
+ free_firmware(fw);
+ return 0;
+
+irl_error:
+ free_firmware(fw);
+ return -EIO;
+}
+
+#endif /* DSP_56361 */
+
+
+static int load_dsp(struct echoaudio *chip, u16 *code)
+{
+ u32 address, data;
+ int index, words, i;
+
+ if (chip->dsp_code == code) {
+ DE_INIT(("DSP is already loaded!\n"));
+ return 0;
+ }
+ chip->bad_board = TRUE; /* Set TRUE until DSP loaded */
+ chip->dsp_code = NULL; /* Current DSP code not loaded */
+ chip->asic_loaded = FALSE; /* Loading the DSP code will reset the ASIC */
+
+ DE_INIT(("load_dsp: Set bad_board to TRUE\n"));
+
+ /* If this board requires a resident loader, install it. */
+#ifdef DSP_56361
+ if ((i = install_resident_loader(chip)) < 0)
+ return i;
+#endif
+
+ /* Send software reset command */
+ if (send_vector(chip, DSP_VC_RESET) < 0) {
+ DE_INIT(("LoadDsp: send_vector DSP_VC_RESET failed, Critical Failure\n"));
+ return -EIO;
+ }
+ /* Delay 10us */
+ udelay(10);
+
+ /* Wait 10ms for HF3 to indicate that software reset is complete */
+ for (i = 0; i < 1000; i++) { /* Timeout is 10us * 1000 = 10ms */
+ if (get_dsp_register(chip, CHI32_STATUS_REG) &
+ CHI32_STATUS_REG_HF3)
+ break;
+ udelay(10);
+ }
+
+ if (i == 1000) {
+ DE_INIT(("load_dsp: Timeout waiting for CHI32_STATUS_REG_HF3\n"));
+ return -EIO;
+ }
+
+ /* Set DSP format bits for 24 bit mode now that soft reset is done */
+ set_dsp_register(chip, CHI32_CONTROL_REG,
+ get_dsp_register(chip, CHI32_CONTROL_REG) | 0x900);
+
+ /* Main loader loop */
+
+ index = code[0];
+ for (;;) {
+ int block_type, mem_type;
+
+ /* Total Block Size */
+ index++;
+
+ /* Block Type */
+ block_type = code[index];
+ if (block_type == 4) /* We're finished */
+ break;
+
+ index++;
+
+ /* Memory Type P=0,X=1,Y=2 */
+ mem_type = code[index++];
+
+ /* Block Code Size */
+ words = code[index++];
+ if (words == 0) /* We're finished */
+ break;
+
+ /* Start Address */
+ address = ((u32)code[index] << 16) + code[index + 1];
+ index += 2;
+
+ if (write_dsp(chip, words) < 0) {
+ DE_INIT(("load_dsp: failed to write number of DSP words\n"));
+ return -EIO;
+ }
+ if (write_dsp(chip, address) < 0) {
+ DE_INIT(("load_dsp: failed to write DSP address\n"));
+ return -EIO;
+ }
+ if (write_dsp(chip, mem_type) < 0) {
+ DE_INIT(("load_dsp: failed to write DSP memory type\n"));
+ return -EIO;
+ }
+ /* Code */
+ for (i = 0; i < words; i++, index+=2) {
+ data = ((u32)code[index] << 16) + code[index + 1];
+ if (write_dsp(chip, data) < 0) {
+ DE_INIT(("load_dsp: failed to write DSP data\n"));
+ return -EIO;
+ }
+ }
+ }
+
+ if (write_dsp(chip, 0) < 0) { /* We're done!!! */
+ DE_INIT(("load_dsp: Failed to write final zero\n"));
+ return -EIO;
+ }
+ udelay(10);
+
+ for (i = 0; i < 5000; i++) { /* Timeout is 100us * 5000 = 500ms */
+ /* Wait for flag 4 - indicates that the DSP loaded OK */
+ if (get_dsp_register(chip, CHI32_STATUS_REG) &
+ CHI32_STATUS_REG_HF4) {
+ set_dsp_register(chip, CHI32_CONTROL_REG,
+ get_dsp_register(chip, CHI32_CONTROL_REG) & ~0x1b00);
+
+ if (write_dsp(chip, DSP_FNC_SET_COMMPAGE_ADDR) < 0) {
+ DE_INIT(("load_dsp: Failed to write DSP_FNC_SET_COMMPAGE_ADDR\n"));
+ return -EIO;
+ }
+
+ if (write_dsp(chip, chip->comm_page_phys) < 0) {
+ DE_INIT(("load_dsp: Failed to write comm page address\n"));
+ return -EIO;
+ }
+
+ /* Get the serial number via slave mode.
+ This is triggered by the SET_COMMPAGE_ADDR command.
+ We don't actually use the serial number but we have to
+ get it as part of the DSP init voodoo. */
+ if (read_sn(chip) < 0) {
+ DE_INIT(("load_dsp: Failed to read serial number\n"));
+ return -EIO;
+ }
+
+ chip->dsp_code = code; /* Show which DSP code loaded */
+ chip->bad_board = FALSE; /* DSP OK */
+ DE_INIT(("load_dsp: OK!\n"));
+ return 0;
+ }
+ udelay(100);
+ }
+
+ DE_INIT(("load_dsp: DSP load timed out waiting for HF4\n"));
+ return -EIO;
+}
+
+
+
+/* load_firmware takes care of loading the DSP and any ASIC code. */
+static int load_firmware(struct echoaudio *chip)
+{
+ const struct firmware *fw;
+ int box_type, err;
+
+ if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page))
+ return -EPERM;
+
+ /* See if the ASIC is present and working - only if the DSP is already loaded */
+ if (chip->dsp_code) {
+ if ((box_type = check_asic_status(chip)) >= 0)
+ return box_type;
+ /* ASIC check failed; force the DSP to reload */
+ chip->dsp_code = NULL;
+ }
+
+ if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0)
+ return err;
+ err = load_dsp(chip, (u16 *)fw->data);
+ free_firmware(fw);
+ if (err < 0)
+ return err;
+
+ if ((box_type = load_asic(chip)) < 0)
+ return box_type; /* error */
+
+ if ((err = restore_dsp_rettings(chip)) < 0)
+ return err;
+
+ return box_type;
+}
+
+
+
+/****************************************************************************
+ Mixer functions
+ ****************************************************************************/
+
+#if defined(ECHOCARD_HAS_INPUT_NOMINAL_LEVEL) || \
+ defined(ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL)
+
+/* Set the nominal level for an input or output bus (true = -10dBV, false = +4dBu) */
+static int set_nominal_level(struct echoaudio *chip, u16 index, char consumer)
+{
+ if (snd_BUG_ON(index >= num_busses_out(chip) + num_busses_in(chip)))
+ return -EINVAL;
+
+ /* Wait for the handshake (OK even if ASIC is not loaded) */
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->nominal_level[index] = consumer;
+
+ if (consumer)
+ chip->comm_page->nominal_level_mask |= cpu_to_le32(1 << index);
+ else
+ chip->comm_page->nominal_level_mask &= ~cpu_to_le32(1 << index);
+
+ return 0;
+}
+
+#endif /* ECHOCARD_HAS_*_NOMINAL_LEVEL */
+
+
+
+/* Set the gain for a single physical output channel (dB). */
+static int set_output_gain(struct echoaudio *chip, u16 channel, s8 gain)
+{
+ if (snd_BUG_ON(channel >= num_busses_out(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ /* Save the new value */
+ chip->output_gain[channel] = gain;
+ chip->comm_page->line_out_level[channel] = gain;
+ return 0;
+}
+
+
+
+#ifdef ECHOCARD_HAS_MONITOR
+/* Set the monitor level from an input bus to an output bus. */
+static int set_monitor_gain(struct echoaudio *chip, u16 output, u16 input,
+ s8 gain)
+{
+ if (snd_BUG_ON(output >= num_busses_out(chip) ||
+ input >= num_busses_in(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->monitor_gain[output][input] = gain;
+ chip->comm_page->monitors[monitor_index(chip, output, input)] = gain;
+ return 0;
+}
+#endif /* ECHOCARD_HAS_MONITOR */
+
+
+/* Tell the DSP to read and update output, nominal & monitor levels in comm page. */
+static int update_output_line_level(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_OUTVOL);
+}
+
+
+
+/* Tell the DSP to read and update input levels in comm page */
+static int update_input_line_level(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_INGAIN);
+}
+
+
+
+/* set_meters_on turns the meters on or off. If meters are turned on, the DSP
+will write the meter and clock detect values to the comm page at about 30Hz */
+static void set_meters_on(struct echoaudio *chip, char on)
+{
+ if (on && !chip->meters_enabled) {
+ send_vector(chip, DSP_VC_METERS_ON);
+ chip->meters_enabled = 1;
+ } else if (!on && chip->meters_enabled) {
+ send_vector(chip, DSP_VC_METERS_OFF);
+ chip->meters_enabled = 0;
+ memset((s8 *)chip->comm_page->vu_meter, ECHOGAIN_MUTED,
+ DSP_MAXPIPES);
+ memset((s8 *)chip->comm_page->peak_meter, ECHOGAIN_MUTED,
+ DSP_MAXPIPES);
+ }
+}
+
+
+
+/* Fill out an the given array using the current values in the comm page.
+Meters are written in the comm page by the DSP in this order:
+ Output busses
+ Input busses
+ Output pipes (vmixer cards only)
+
+This function assumes there are no more than 16 in/out busses or pipes
+Meters is an array [3][16][2] of long. */
+static void get_audio_meters(struct echoaudio *chip, long *meters)
+{
+ int i, m, n;
+
+ m = 0;
+ n = 0;
+ for (i = 0; i < num_busses_out(chip); i++, m++) {
+ meters[n++] = chip->comm_page->vu_meter[m];
+ meters[n++] = chip->comm_page->peak_meter[m];
+ }
+ for (; n < 32; n++)
+ meters[n] = 0;
+
+#ifdef ECHOCARD_ECHO3G
+ m = E3G_MAX_OUTPUTS; /* Skip unused meters */
+#endif
+
+ for (i = 0; i < num_busses_in(chip); i++, m++) {
+ meters[n++] = chip->comm_page->vu_meter[m];
+ meters[n++] = chip->comm_page->peak_meter[m];
+ }
+ for (; n < 64; n++)
+ meters[n] = 0;
+
+#ifdef ECHOCARD_HAS_VMIXER
+ for (i = 0; i < num_pipes_out(chip); i++, m++) {
+ meters[n++] = chip->comm_page->vu_meter[m];
+ meters[n++] = chip->comm_page->peak_meter[m];
+ }
+#endif
+ for (; n < 96; n++)
+ meters[n] = 0;
+}
+
+
+
+static int restore_dsp_rettings(struct echoaudio *chip)
+{
+ int err;
+ DE_INIT(("restore_dsp_settings\n"));
+
+ if ((err = check_asic_status(chip)) < 0)
+ return err;
+
+ /* @ Gina20/Darla20 only. Should be harmless for other cards. */
+ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF;
+ chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF;
+ chip->comm_page->handshake = 0xffffffff;
+
+ if ((err = set_sample_rate(chip, chip->sample_rate)) < 0)
+ return err;
+
+ if (chip->meters_enabled)
+ if (send_vector(chip, DSP_VC_METERS_ON) < 0)
+ return -EIO;
+
+#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
+ if (set_input_clock(chip, chip->input_clock) < 0)
+ return -EIO;
+#endif
+
+#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH
+ if (set_output_clock(chip, chip->output_clock) < 0)
+ return -EIO;
+#endif
+
+ if (update_output_line_level(chip) < 0)
+ return -EIO;
+
+ if (update_input_line_level(chip) < 0)
+ return -EIO;
+
+#ifdef ECHOCARD_HAS_VMIXER
+ if (update_vmixer_level(chip) < 0)
+ return -EIO;
+#endif
+
+ if (wait_handshake(chip) < 0)
+ return -EIO;
+ clear_handshake(chip);
+
+ DE_INIT(("restore_dsp_rettings done\n"));
+ return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+/****************************************************************************
+ Transport functions
+ ****************************************************************************/
+
+/* set_audio_format() sets the format of the audio data in host memory for
+this pipe. Note that _MS_ (mono-to-stereo) playback modes are not used by ALSA
+but they are here because they are just mono while capturing */
+static void set_audio_format(struct echoaudio *chip, u16 pipe_index,
+ const struct audioformat *format)
+{
+ u16 dsp_format;
+
+ dsp_format = DSP_AUDIOFORM_SS_16LE;
+
+ /* Look for super-interleave (no big-endian and 8 bits) */
+ if (format->interleave > 2) {
+ switch (format->bits_per_sample) {
+ case 16:
+ dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_16LE;
+ break;
+ case 24:
+ dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_24LE;
+ break;
+ case 32:
+ dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_32LE;
+ break;
+ }
+ dsp_format |= format->interleave;
+ } else if (format->data_are_bigendian) {
+ /* For big-endian data, only 32 bit samples are supported */
+ switch (format->interleave) {
+ case 1:
+ dsp_format = DSP_AUDIOFORM_MM_32BE;
+ break;
+#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+ case 2:
+ dsp_format = DSP_AUDIOFORM_SS_32BE;
+ break;
+#endif
+ }
+ } else if (format->interleave == 1 &&
+ format->bits_per_sample == 32 && !format->mono_to_stereo) {
+ /* 32 bit little-endian mono->mono case */
+ dsp_format = DSP_AUDIOFORM_MM_32LE;
+ } else {
+ /* Handle the other little-endian formats */
+ switch (format->bits_per_sample) {
+ case 8:
+ if (format->interleave == 2)
+ dsp_format = DSP_AUDIOFORM_SS_8;
+ else
+ dsp_format = DSP_AUDIOFORM_MS_8;
+ break;
+ default:
+ case 16:
+ if (format->interleave == 2)
+ dsp_format = DSP_AUDIOFORM_SS_16LE;
+ else
+ dsp_format = DSP_AUDIOFORM_MS_16LE;
+ break;
+ case 24:
+ if (format->interleave == 2)
+ dsp_format = DSP_AUDIOFORM_SS_24LE;
+ else
+ dsp_format = DSP_AUDIOFORM_MS_24LE;
+ break;
+ case 32:
+ if (format->interleave == 2)
+ dsp_format = DSP_AUDIOFORM_SS_32LE;
+ else
+ dsp_format = DSP_AUDIOFORM_MS_32LE;
+ break;
+ }
+ }
+ DE_ACT(("set_audio_format[%d] = %x\n", pipe_index, dsp_format));
+ chip->comm_page->audio_format[pipe_index] = cpu_to_le16(dsp_format);
+}
+
+
+
+/* start_transport starts transport for a set of pipes.
+The bits 1 in channel_mask specify what pipes to start. Only the bit of the
+first channel must be set, regardless its interleave.
+Same thing for pause_ and stop_ -trasport below. */
+static int start_transport(struct echoaudio *chip, u32 channel_mask,
+ u32 cyclic_mask)
+{
+ DE_ACT(("start_transport %x\n", channel_mask));
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->cmd_start |= cpu_to_le32(channel_mask);
+
+ if (chip->comm_page->cmd_start) {
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_START_TRANSFER);
+ if (wait_handshake(chip))
+ return -EIO;
+ /* Keep track of which pipes are transporting */
+ chip->active_mask |= channel_mask;
+ chip->comm_page->cmd_start = 0;
+ return 0;
+ }
+
+ DE_ACT(("start_transport: No pipes to start!\n"));
+ return -EINVAL;
+}
+
+
+
+static int pause_transport(struct echoaudio *chip, u32 channel_mask)
+{
+ DE_ACT(("pause_transport %x\n", channel_mask));
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->cmd_stop |= cpu_to_le32(channel_mask);
+ chip->comm_page->cmd_reset = 0;
+ if (chip->comm_page->cmd_stop) {
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_STOP_TRANSFER);
+ if (wait_handshake(chip))
+ return -EIO;
+ /* Keep track of which pipes are transporting */
+ chip->active_mask &= ~channel_mask;
+ chip->comm_page->cmd_stop = 0;
+ chip->comm_page->cmd_reset = 0;
+ return 0;
+ }
+
+ DE_ACT(("pause_transport: No pipes to stop!\n"));
+ return 0;
+}
+
+
+
+static int stop_transport(struct echoaudio *chip, u32 channel_mask)
+{
+ DE_ACT(("stop_transport %x\n", channel_mask));
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->cmd_stop |= cpu_to_le32(channel_mask);
+ chip->comm_page->cmd_reset |= cpu_to_le32(channel_mask);
+ if (chip->comm_page->cmd_reset) {
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_STOP_TRANSFER);
+ if (wait_handshake(chip))
+ return -EIO;
+ /* Keep track of which pipes are transporting */
+ chip->active_mask &= ~channel_mask;
+ chip->comm_page->cmd_stop = 0;
+ chip->comm_page->cmd_reset = 0;
+ return 0;
+ }
+
+ DE_ACT(("stop_transport: No pipes to stop!\n"));
+ return 0;
+}
+
+
+
+static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index)
+{
+ return (chip->pipe_alloc_mask & (1 << pipe_index));
+}
+
+
+
+/* Stops everything and turns off the DSP. All pipes should be already
+stopped and unallocated. */
+static int rest_in_peace(struct echoaudio *chip)
+{
+ DE_ACT(("rest_in_peace() open=%x\n", chip->pipe_alloc_mask));
+
+ /* Stops all active pipes (just to be sure) */
+ stop_transport(chip, chip->active_mask);
+
+ set_meters_on(chip, FALSE);
+
+#ifdef ECHOCARD_HAS_MIDI
+ enable_midi_input(chip, FALSE);
+#endif
+
+ /* Go to sleep */
+ if (chip->dsp_code) {
+ /* Make load_firmware do a complete reload */
+ chip->dsp_code = NULL;
+ /* Put the DSP to sleep */
+ return send_vector(chip, DSP_VC_GO_COMATOSE);
+ }
+ return 0;
+}
+
+
+
+/* Fills the comm page with default values */
+static int init_dsp_comm_page(struct echoaudio *chip)
+{
+ /* Check if the compiler added extra padding inside the structure */
+ if (offsetof(struct comm_page, midi_output) != 0xbe0) {
+ DE_INIT(("init_dsp_comm_page() - Invalid struct comm_page structure\n"));
+ return -EPERM;
+ }
+
+ /* Init all the basic stuff */
+ chip->card_name = ECHOCARD_NAME;
+ chip->bad_board = TRUE; /* Set TRUE until DSP loaded */
+ chip->dsp_code = NULL; /* Current DSP code not loaded */
+ chip->digital_mode = DIGITAL_MODE_NONE;
+ chip->input_clock = ECHO_CLOCK_INTERNAL;
+ chip->output_clock = ECHO_CLOCK_WORD;
+ chip->asic_loaded = FALSE;
+ memset(chip->comm_page, 0, sizeof(struct comm_page));
+
+ /* Init the comm page */
+ chip->comm_page->comm_size =
+ __constant_cpu_to_le32(sizeof(struct comm_page));
+ chip->comm_page->handshake = 0xffffffff;
+ chip->comm_page->midi_out_free_count =
+ __constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
+ chip->comm_page->sample_rate = __constant_cpu_to_le32(44100);
+ chip->sample_rate = 44100;
+
+ /* Set line levels so we don't blast any inputs on startup */
+ memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE);
+ memset(chip->comm_page->vmixer, ECHOGAIN_MUTED, VMIXER_ARRAY_SIZE);
+
+ return 0;
+}
+
+
+
+/* This function initializes the several volume controls for busses and pipes.
+This MUST be called after the DSP is up and running ! */
+static int init_line_levels(struct echoaudio *chip)
+{
+ int st, i, o;
+
+ DE_INIT(("init_line_levels\n"));
+
+ /* Mute output busses */
+ for (i = 0; i < num_busses_out(chip); i++)
+ if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED)))
+ return st;
+ if ((st = update_output_line_level(chip)))
+ return st;
+
+#ifdef ECHOCARD_HAS_VMIXER
+ /* Mute the Vmixer */
+ for (i = 0; i < num_pipes_out(chip); i++)
+ for (o = 0; o < num_busses_out(chip); o++)
+ if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED)))
+ return st;
+ if ((st = update_vmixer_level(chip)))
+ return st;
+#endif /* ECHOCARD_HAS_VMIXER */
+
+#ifdef ECHOCARD_HAS_MONITOR
+ /* Mute the monitor mixer */
+ for (o = 0; o < num_busses_out(chip); o++)
+ for (i = 0; i < num_busses_in(chip); i++)
+ if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED)))
+ return st;
+ if ((st = update_output_line_level(chip)))
+ return st;
+#endif /* ECHOCARD_HAS_MONITOR */
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+ for (i = 0; i < num_busses_in(chip); i++)
+ if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED)))
+ return st;
+ if ((st = update_input_line_level(chip)))
+ return st;
+#endif /* ECHOCARD_HAS_INPUT_GAIN */
+
+ return 0;
+}
+
+
+
+/* This is low level part of the interrupt handler.
+It returns -1 if the IRQ is not ours, or N>=0 if it is, where N is the number
+of midi data in the input queue. */
+static int service_irq(struct echoaudio *chip)
+{
+ int st;
+
+ /* Read the DSP status register and see if this DSP generated this interrupt */
+ if (get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_IRQ) {
+ st = 0;
+#ifdef ECHOCARD_HAS_MIDI
+ /* Get and parse midi data if present */
+ if (chip->comm_page->midi_input[0]) /* The count is at index 0 */
+ st = midi_service_irq(chip); /* Returns how many midi bytes we received */
+#endif
+ /* Clear the hardware interrupt */
+ chip->comm_page->midi_input[0] = 0;
+ send_vector(chip, DSP_VC_ACK_INT);
+ return st;
+ }
+ return -1;
+}
+
+
+
+
+/******************************************************************************
+ Functions for opening and closing pipes
+ ******************************************************************************/
+
+/* allocate_pipes is used to reserve audio pipes for your exclusive use.
+The call will fail if some pipes are already allocated. */
+static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
+ int pipe_index, int interleave)
+{
+ int i;
+ u32 channel_mask;
+ char is_cyclic;
+
+ DE_ACT(("allocate_pipes: ch=%d int=%d\n", pipe_index, interleave));
+
+ if (chip->bad_board)
+ return -EIO;
+
+ is_cyclic = 1; /* This driver uses cyclic buffers only */
+
+ for (channel_mask = i = 0; i < interleave; i++)
+ channel_mask |= 1 << (pipe_index + i);
+ if (chip->pipe_alloc_mask & channel_mask) {
+ DE_ACT(("allocate_pipes: channel already open\n"));
+ return -EAGAIN;
+ }
+
+ chip->comm_page->position[pipe_index] = 0;
+ chip->pipe_alloc_mask |= channel_mask;
+ if (is_cyclic)
+ chip->pipe_cyclic_mask |= channel_mask;
+ pipe->index = pipe_index;
+ pipe->interleave = interleave;
+ pipe->state = PIPE_STATE_STOPPED;
+
+ /* The counter register is where the DSP writes the 32 bit DMA
+ position for a pipe. The DSP is constantly updating this value as
+ it moves data. The DMA counter is in units of bytes, not samples. */
+ pipe->dma_counter = &chip->comm_page->position[pipe_index];
+ *pipe->dma_counter = 0;
+ DE_ACT(("allocate_pipes: ok\n"));
+ return pipe_index;
+}
+
+
+
+static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe)
+{
+ u32 channel_mask;
+ int i;
+
+ DE_ACT(("free_pipes: Pipe %d\n", pipe->index));
+ if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index)))
+ return -EINVAL;
+ if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED))
+ return -EINVAL;
+
+ for (channel_mask = i = 0; i < pipe->interleave; i++)
+ channel_mask |= 1 << (pipe->index + i);
+
+ chip->pipe_alloc_mask &= ~channel_mask;
+ chip->pipe_cyclic_mask &= ~channel_mask;
+ return 0;
+}
+
+
+
+/******************************************************************************
+ Functions for managing the scatter-gather list
+******************************************************************************/
+
+static int sglist_init(struct echoaudio *chip, struct audiopipe *pipe)
+{
+ pipe->sglist_head = 0;
+ memset(pipe->sgpage.area, 0, PAGE_SIZE);
+ chip->comm_page->sglist_addr[pipe->index].addr =
+ cpu_to_le32(pipe->sgpage.addr);
+ return 0;
+}
+
+
+
+static int sglist_add_mapping(struct echoaudio *chip, struct audiopipe *pipe,
+ dma_addr_t address, size_t length)
+{
+ int head = pipe->sglist_head;
+ struct sg_entry *list = (struct sg_entry *)pipe->sgpage.area;
+
+ if (head < MAX_SGLIST_ENTRIES - 1) {
+ list[head].addr = cpu_to_le32(address);
+ list[head].size = cpu_to_le32(length);
+ pipe->sglist_head++;
+ } else {
+ DE_ACT(("SGlist: too many fragments\n"));
+ return -ENOMEM;
+ }
+ return 0;
+}
+
+
+
+static inline int sglist_add_irq(struct echoaudio *chip, struct audiopipe *pipe)
+{
+ return sglist_add_mapping(chip, pipe, 0, 0);
+}
+
+
+
+static inline int sglist_wrap(struct echoaudio *chip, struct audiopipe *pipe)
+{
+ return sglist_add_mapping(chip, pipe, pipe->sgpage.addr, 0);
+}
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
new file mode 100644
index 0000000..e352f3a
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -0,0 +1,693 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+#ifndef _ECHO_DSP_
+#define _ECHO_DSP_
+
+
+/**** Echogals: Darla20, Gina20, Layla20, and Darla24 ****/
+#if defined(ECHOGALS_FAMILY)
+
+#define NUM_ASIC_TESTS 5
+#define READ_DSP_TIMEOUT 1000000L /* one second */
+
+/**** Echo24: Gina24, Layla24, Mona, Mia, Mia-midi ****/
+#elif defined(ECHO24_FAMILY)
+
+#define DSP_56361 /* Some Echo24 cards use the 56361 DSP */
+#define READ_DSP_TIMEOUT 100000L /* .1 second */
+
+/**** 3G: Gina3G, Layla3G ****/
+#elif defined(ECHO3G_FAMILY)
+
+#define DSP_56361
+#define READ_DSP_TIMEOUT 100000L /* .1 second */
+#define MIN_MTC_1X_RATE 32000
+
+/**** Indigo: Indigo, Indigo IO, Indigo DJ ****/
+#elif defined(INDIGO_FAMILY)
+
+#define DSP_56361
+#define READ_DSP_TIMEOUT 100000L /* .1 second */
+
+#else
+
+#error No family is defined
+
+#endif
+
+
+
+/*
+ *
+ * Max inputs and outputs
+ *
+ */
+
+#define DSP_MAXAUDIOINPUTS 16 /* Max audio input channels */
+#define DSP_MAXAUDIOOUTPUTS 16 /* Max audio output channels */
+#define DSP_MAXPIPES 32 /* Max total pipes (input + output) */
+
+
+/*
+ *
+ * These are the offsets for the memory-mapped DSP registers; the DSP base
+ * address is treated as the start of a u32 array.
+ */
+
+#define CHI32_CONTROL_REG 4
+#define CHI32_STATUS_REG 5
+#define CHI32_VECTOR_REG 6
+#define CHI32_DATA_REG 7
+
+
+/*
+ *
+ * Interesting bits within the DSP registers
+ *
+ */
+
+#define CHI32_VECTOR_BUSY 0x00000001
+#define CHI32_STATUS_REG_HF3 0x00000008
+#define CHI32_STATUS_REG_HF4 0x00000010
+#define CHI32_STATUS_REG_HF5 0x00000020
+#define CHI32_STATUS_HOST_READ_FULL 0x00000004
+#define CHI32_STATUS_HOST_WRITE_EMPTY 0x00000002
+#define CHI32_STATUS_IRQ 0x00000040
+
+
+/*
+ *
+ * DSP commands sent via slave mode; these are sent to the DSP by write_dsp()
+ *
+ */
+
+#define DSP_FNC_SET_COMMPAGE_ADDR 0x02
+#define DSP_FNC_LOAD_LAYLA_ASIC 0xa0
+#define DSP_FNC_LOAD_GINA24_ASIC 0xa0
+#define DSP_FNC_LOAD_MONA_PCI_CARD_ASIC 0xa0
+#define DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC 0xa0
+#define DSP_FNC_LOAD_MONA_EXTERNAL_ASIC 0xa1
+#define DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC 0xa1
+#define DSP_FNC_LOAD_3G_ASIC 0xa0
+
+
+/*
+ *
+ * Defines to handle the MIDI input state engine; these are used to properly
+ * extract MIDI time code bytes and their timestamps from the MIDI input stream.
+ *
+ */
+
+#define MIDI_IN_STATE_NORMAL 0
+#define MIDI_IN_STATE_TS_HIGH 1
+#define MIDI_IN_STATE_TS_LOW 2
+#define MIDI_IN_STATE_F1_DATA 3
+#define MIDI_IN_SKIP_DATA (-1)
+
+
+/*----------------------------------------------------------------------------
+
+Setting the sample rates on Layla24 is somewhat schizophrenic.
+
+For standard rates, it works exactly like Mona and Gina24. That is, for
+8, 11.025, 16, 22.05, 32, 44.1, 48, 88.2, and 96 kHz, you just set the
+appropriate bits in the control register and write the control register.
+
+In order to support MIDI time code sync (and possibly SMPTE LTC sync in
+the future), Layla24 also has "continuous sample rate mode". In this mode,
+Layla24 can generate any sample rate between 25 and 50 kHz inclusive, or
+50 to 100 kHz inclusive for double speed mode.
+
+To use continuous mode:
+
+-Set the clock select bits in the control register to 0xe (see the #define
+ below)
+
+-Set double-speed mode if you want to use sample rates above 50 kHz
+
+-Write the control register as you would normally
+
+-Now, you need to set the frequency register. First, you need to determine the
+ value for the frequency register. This is given by the following formula:
+
+frequency_reg = (LAYLA24_MAGIC_NUMBER / sample_rate) - 2
+
+Note the #define below for the magic number
+
+-Wait for the DSP handshake
+-Write the frequency_reg value to the .SampleRate field of the comm page
+-Send the vector command SET_LAYLA24_FREQUENCY_REG (see vmonkey.h)
+
+Once you have set the control register up for continuous mode, you can just
+write the frequency register to change the sample rate. This could be
+used for MIDI time code sync. For MTC sync, the control register is set for
+continuous mode. The driver then just keeps writing the
+SET_LAYLA24_FREQUENCY_REG command.
+
+-----------------------------------------------------------------------------*/
+
+#define LAYLA24_MAGIC_NUMBER 677376000
+#define LAYLA24_CONTINUOUS_CLOCK 0x000e
+
+
+/*
+ *
+ * DSP vector commands
+ *
+ */
+
+#define DSP_VC_RESET 0x80ff
+
+#ifndef DSP_56361
+
+#define DSP_VC_ACK_INT 0x8073
+#define DSP_VC_SET_VMIXER_GAIN 0x0000 /* Not used, only for compile */
+#define DSP_VC_START_TRANSFER 0x0075 /* Handshke rqd. */
+#define DSP_VC_METERS_ON 0x0079
+#define DSP_VC_METERS_OFF 0x007b
+#define DSP_VC_UPDATE_OUTVOL 0x007d /* Handshke rqd. */
+#define DSP_VC_UPDATE_INGAIN 0x007f /* Handshke rqd. */
+#define DSP_VC_ADD_AUDIO_BUFFER 0x0081 /* Handshke rqd. */
+#define DSP_VC_TEST_ASIC 0x00eb
+#define DSP_VC_UPDATE_CLOCKS 0x00ef /* Handshke rqd. */
+#define DSP_VC_SET_LAYLA_SAMPLE_RATE 0x00f1 /* Handshke rqd. */
+#define DSP_VC_SET_GD_AUDIO_STATE 0x00f1 /* Handshke rqd. */
+#define DSP_VC_WRITE_CONTROL_REG 0x00f1 /* Handshke rqd. */
+#define DSP_VC_MIDI_WRITE 0x00f5 /* Handshke rqd. */
+#define DSP_VC_STOP_TRANSFER 0x00f7 /* Handshke rqd. */
+#define DSP_VC_UPDATE_FLAGS 0x00fd /* Handshke rqd. */
+#define DSP_VC_GO_COMATOSE 0x00f9
+
+#else /* !DSP_56361 */
+
+/* Vector commands for families that use either the 56301 or 56361 */
+#define DSP_VC_ACK_INT 0x80F5
+#define DSP_VC_SET_VMIXER_GAIN 0x00DB /* Handshke rqd. */
+#define DSP_VC_START_TRANSFER 0x00DD /* Handshke rqd. */
+#define DSP_VC_METERS_ON 0x00EF
+#define DSP_VC_METERS_OFF 0x00F1
+#define DSP_VC_UPDATE_OUTVOL 0x00E3 /* Handshke rqd. */
+#define DSP_VC_UPDATE_INGAIN 0x00E5 /* Handshke rqd. */
+#define DSP_VC_ADD_AUDIO_BUFFER 0x00E1 /* Handshke rqd. */
+#define DSP_VC_TEST_ASIC 0x00ED
+#define DSP_VC_UPDATE_CLOCKS 0x00E9 /* Handshke rqd. */
+#define DSP_VC_SET_LAYLA24_FREQUENCY_REG 0x00E9 /* Handshke rqd. */
+#define DSP_VC_SET_LAYLA_SAMPLE_RATE 0x00EB /* Handshke rqd. */
+#define DSP_VC_SET_GD_AUDIO_STATE 0x00EB /* Handshke rqd. */
+#define DSP_VC_WRITE_CONTROL_REG 0x00EB /* Handshke rqd. */
+#define DSP_VC_MIDI_WRITE 0x00E7 /* Handshke rqd. */
+#define DSP_VC_STOP_TRANSFER 0x00DF /* Handshke rqd. */
+#define DSP_VC_UPDATE_FLAGS 0x00FB /* Handshke rqd. */
+#define DSP_VC_GO_COMATOSE 0x00d9
+
+#endif /* !DSP_56361 */
+
+
+/*
+ *
+ * Timeouts
+ *
+ */
+
+#define HANDSHAKE_TIMEOUT 20000 /* send_vector command timeout (20ms) */
+#define VECTOR_BUSY_TIMEOUT 100000 /* 100ms */
+#define MIDI_OUT_DELAY_USEC 2000 /* How long to wait after MIDI fills up */
+
+
+/*
+ *
+ * Flags for .Flags field in the comm page
+ *
+ */
+
+#define DSP_FLAG_MIDI_INPUT 0x0001 /* Enable MIDI input */
+#define DSP_FLAG_SPDIF_NONAUDIO 0x0002 /* Sets the "non-audio" bit
+ * in the S/PDIF out status
+ * bits. Clear this flag for
+ * audio data;
+ * set it for AC3 or WMA or
+ * some such */
+#define DSP_FLAG_PROFESSIONAL_SPDIF 0x0008 /* 1 Professional, 0 Consumer */
+
+
+/*
+ *
+ * Clock detect bits reported by the DSP for Gina20, Layla20, Darla24, and Mia
+ *
+ */
+
+#define GLDM_CLOCK_DETECT_BIT_WORD 0x0002
+#define GLDM_CLOCK_DETECT_BIT_SUPER 0x0004
+#define GLDM_CLOCK_DETECT_BIT_SPDIF 0x0008
+#define GLDM_CLOCK_DETECT_BIT_ESYNC 0x0010
+
+
+/*
+ *
+ * Clock detect bits reported by the DSP for Gina24, Mona, and Layla24
+ *
+ */
+
+#define GML_CLOCK_DETECT_BIT_WORD96 0x0002
+#define GML_CLOCK_DETECT_BIT_WORD48 0x0004
+#define GML_CLOCK_DETECT_BIT_SPDIF48 0x0008
+#define GML_CLOCK_DETECT_BIT_SPDIF96 0x0010
+#define GML_CLOCK_DETECT_BIT_WORD (GML_CLOCK_DETECT_BIT_WORD96 | GML_CLOCK_DETECT_BIT_WORD48)
+#define GML_CLOCK_DETECT_BIT_SPDIF (GML_CLOCK_DETECT_BIT_SPDIF48 | GML_CLOCK_DETECT_BIT_SPDIF96)
+#define GML_CLOCK_DETECT_BIT_ESYNC 0x0020
+#define GML_CLOCK_DETECT_BIT_ADAT 0x0040
+
+
+/*
+ *
+ * Layla clock numbers to send to DSP
+ *
+ */
+
+#define LAYLA20_CLOCK_INTERNAL 0
+#define LAYLA20_CLOCK_SPDIF 1
+#define LAYLA20_CLOCK_WORD 2
+#define LAYLA20_CLOCK_SUPER 3
+
+
+/*
+ *
+ * Gina/Darla clock states
+ *
+ */
+
+#define GD_CLOCK_NOCHANGE 0
+#define GD_CLOCK_44 1
+#define GD_CLOCK_48 2
+#define GD_CLOCK_SPDIFIN 3
+#define GD_CLOCK_UNDEF 0xff
+
+
+/*
+ *
+ * Gina/Darla S/PDIF status bits
+ *
+ */
+
+#define GD_SPDIF_STATUS_NOCHANGE 0
+#define GD_SPDIF_STATUS_44 1
+#define GD_SPDIF_STATUS_48 2
+#define GD_SPDIF_STATUS_UNDEF 0xff
+
+
+/*
+ *
+ * Layla20 output clocks
+ *
+ */
+
+#define LAYLA20_OUTPUT_CLOCK_SUPER 0
+#define LAYLA20_OUTPUT_CLOCK_WORD 1
+
+
+/****************************************************************************
+
+ Magic constants for the Darla24 hardware
+
+ ****************************************************************************/
+
+#define GD24_96000 0x0
+#define GD24_48000 0x1
+#define GD24_44100 0x2
+#define GD24_32000 0x3
+#define GD24_22050 0x4
+#define GD24_16000 0x5
+#define GD24_11025 0x6
+#define GD24_8000 0x7
+#define GD24_88200 0x8
+#define GD24_EXT_SYNC 0x9
+
+
+/*
+ *
+ * Return values from the DSP when ASIC is loaded
+ *
+ */
+
+#define ASIC_ALREADY_LOADED 0x1
+#define ASIC_NOT_LOADED 0x0
+
+
+/*
+ *
+ * DSP Audio formats
+ *
+ * These are the audio formats that the DSP can transfer
+ * via input and output pipes. LE means little-endian,
+ * BE means big-endian.
+ *
+ * DSP_AUDIOFORM_MS_8
+ *
+ * 8-bit mono unsigned samples. For playback,
+ * mono data is duplicated out the left and right channels
+ * of the output bus. The "MS" part of the name
+ * means mono->stereo.
+ *
+ * DSP_AUDIOFORM_MS_16LE
+ *
+ * 16-bit signed little-endian mono samples. Playback works
+ * like the previous code.
+ *
+ * DSP_AUDIOFORM_MS_24LE
+ *
+ * 24-bit signed little-endian mono samples. Data is packed
+ * three bytes per sample; if you had two samples 0x112233 and 0x445566
+ * they would be stored in memory like this: 33 22 11 66 55 44.
+ *
+ * DSP_AUDIOFORM_MS_32LE
+ *
+ * 24-bit signed little-endian mono samples in a 32-bit
+ * container. In other words, each sample is a 32-bit signed
+ * integer, where the actual audio data is left-justified
+ * in the 32 bits and only the 24 most significant bits are valid.
+ *
+ * DSP_AUDIOFORM_SS_8
+ * DSP_AUDIOFORM_SS_16LE
+ * DSP_AUDIOFORM_SS_24LE
+ * DSP_AUDIOFORM_SS_32LE
+ *
+ * Like the previous ones, except now with stereo interleaved
+ * data. "SS" means stereo->stereo.
+ *
+ * DSP_AUDIOFORM_MM_32LE
+ *
+ * Similar to DSP_AUDIOFORM_MS_32LE, except that the mono
+ * data is not duplicated out both the left and right outputs.
+ * This mode is used by the ASIO driver. Here, "MM" means
+ * mono->mono.
+ *
+ * DSP_AUDIOFORM_MM_32BE
+ *
+ * Just like DSP_AUDIOFORM_MM_32LE, but now the data is
+ * in big-endian format.
+ *
+ */
+
+#define DSP_AUDIOFORM_MS_8 0 /* 8 bit mono */
+#define DSP_AUDIOFORM_MS_16LE 1 /* 16 bit mono */
+#define DSP_AUDIOFORM_MS_24LE 2 /* 24 bit mono */
+#define DSP_AUDIOFORM_MS_32LE 3 /* 32 bit mono */
+#define DSP_AUDIOFORM_SS_8 4 /* 8 bit stereo */
+#define DSP_AUDIOFORM_SS_16LE 5 /* 16 bit stereo */
+#define DSP_AUDIOFORM_SS_24LE 6 /* 24 bit stereo */
+#define DSP_AUDIOFORM_SS_32LE 7 /* 32 bit stereo */
+#define DSP_AUDIOFORM_MM_32LE 8 /* 32 bit mono->mono little-endian */
+#define DSP_AUDIOFORM_MM_32BE 9 /* 32 bit mono->mono big-endian */
+#define DSP_AUDIOFORM_SS_32BE 10 /* 32 bit stereo big endian */
+#define DSP_AUDIOFORM_INVALID 0xFF /* Invalid audio format */
+
+
+/*
+ *
+ * Super-interleave is defined as interleaving by 4 or more. Darla20 and Gina20
+ * do not support super interleave.
+ *
+ * 16 bit, 24 bit, and 32 bit little endian samples are supported for super
+ * interleave. The interleave factor must be even. 16 - way interleave is the
+ * current maximum, so you can interleave by 4, 6, 8, 10, 12, 14, and 16.
+ *
+ * The actual format code is derived by taking the define below and or-ing with
+ * the interleave factor. So, 32 bit interleave by 6 is 0x86 and
+ * 16 bit interleave by 16 is (0x40 | 0x10) = 0x50.
+ *
+ */
+
+#define DSP_AUDIOFORM_SUPER_INTERLEAVE_16LE 0x40
+#define DSP_AUDIOFORM_SUPER_INTERLEAVE_24LE 0xc0
+#define DSP_AUDIOFORM_SUPER_INTERLEAVE_32LE 0x80
+
+
+/*
+ *
+ * Gina24, Mona, and Layla24 control register defines
+ *
+ */
+
+#define GML_CONVERTER_ENABLE 0x0010
+#define GML_SPDIF_PRO_MODE 0x0020 /* Professional S/PDIF == 1,
+ consumer == 0 */
+#define GML_SPDIF_SAMPLE_RATE0 0x0040
+#define GML_SPDIF_SAMPLE_RATE1 0x0080
+#define GML_SPDIF_TWO_CHANNEL 0x0100 /* 1 == two channels,
+ 0 == one channel */
+#define GML_SPDIF_NOT_AUDIO 0x0200
+#define GML_SPDIF_COPY_PERMIT 0x0400
+#define GML_SPDIF_24_BIT 0x0800 /* 1 == 24 bit, 0 == 20 bit */
+#define GML_ADAT_MODE 0x1000 /* 1 == ADAT mode, 0 == S/PDIF mode */
+#define GML_SPDIF_OPTICAL_MODE 0x2000 /* 1 == optical mode, 0 == RCA mode */
+#define GML_SPDIF_CDROM_MODE 0x3000 /* 1 == CDROM mode,
+ * 0 == RCA or optical mode */
+#define GML_DOUBLE_SPEED_MODE 0x4000 /* 1 == double speed,
+ 0 == single speed */
+
+#define GML_DIGITAL_IN_AUTO_MUTE 0x800000
+
+#define GML_96KHZ (0x0 | GML_DOUBLE_SPEED_MODE)
+#define GML_88KHZ (0x1 | GML_DOUBLE_SPEED_MODE)
+#define GML_48KHZ 0x2
+#define GML_44KHZ 0x3
+#define GML_32KHZ 0x4
+#define GML_22KHZ 0x5
+#define GML_16KHZ 0x6
+#define GML_11KHZ 0x7
+#define GML_8KHZ 0x8
+#define GML_SPDIF_CLOCK 0x9
+#define GML_ADAT_CLOCK 0xA
+#define GML_WORD_CLOCK 0xB
+#define GML_ESYNC_CLOCK 0xC
+#define GML_ESYNCx2_CLOCK 0xD
+
+#define GML_CLOCK_CLEAR_MASK 0xffffbff0
+#define GML_SPDIF_RATE_CLEAR_MASK (~(GML_SPDIF_SAMPLE_RATE0|GML_SPDIF_SAMPLE_RATE1))
+#define GML_DIGITAL_MODE_CLEAR_MASK 0xffffcfff
+#define GML_SPDIF_FORMAT_CLEAR_MASK 0xfffff01f
+
+
+/*
+ *
+ * Mia sample rate and clock setting constants
+ *
+ */
+
+#define MIA_32000 0x0040
+#define MIA_44100 0x0042
+#define MIA_48000 0x0041
+#define MIA_88200 0x0142
+#define MIA_96000 0x0141
+
+#define MIA_SPDIF 0x00000044
+#define MIA_SPDIF96 0x00000144
+
+#define MIA_MIDI_REV 1 /* Must be Mia rev 1 for MIDI support */
+
+
+/*
+ *
+ * 3G register bits
+ *
+ */
+
+#define E3G_CONVERTER_ENABLE 0x0010
+#define E3G_SPDIF_PRO_MODE 0x0020 /* Professional S/PDIF == 1,
+ consumer == 0 */
+#define E3G_SPDIF_SAMPLE_RATE0 0x0040
+#define E3G_SPDIF_SAMPLE_RATE1 0x0080
+#define E3G_SPDIF_TWO_CHANNEL 0x0100 /* 1 == two channels,
+ 0 == one channel */
+#define E3G_SPDIF_NOT_AUDIO 0x0200
+#define E3G_SPDIF_COPY_PERMIT 0x0400
+#define E3G_SPDIF_24_BIT 0x0800 /* 1 == 24 bit, 0 == 20 bit */
+#define E3G_DOUBLE_SPEED_MODE 0x4000 /* 1 == double speed,
+ 0 == single speed */
+#define E3G_PHANTOM_POWER 0x8000 /* 1 == phantom power on,
+ 0 == phantom power off */
+
+#define E3G_96KHZ (0x0 | E3G_DOUBLE_SPEED_MODE)
+#define E3G_88KHZ (0x1 | E3G_DOUBLE_SPEED_MODE)
+#define E3G_48KHZ 0x2
+#define E3G_44KHZ 0x3
+#define E3G_32KHZ 0x4
+#define E3G_22KHZ 0x5
+#define E3G_16KHZ 0x6
+#define E3G_11KHZ 0x7
+#define E3G_8KHZ 0x8
+#define E3G_SPDIF_CLOCK 0x9
+#define E3G_ADAT_CLOCK 0xA
+#define E3G_WORD_CLOCK 0xB
+#define E3G_CONTINUOUS_CLOCK 0xE
+
+#define E3G_ADAT_MODE 0x1000
+#define E3G_SPDIF_OPTICAL_MODE 0x2000
+
+#define E3G_CLOCK_CLEAR_MASK 0xbfffbff0
+#define E3G_DIGITAL_MODE_CLEAR_MASK 0xffffcfff
+#define E3G_SPDIF_FORMAT_CLEAR_MASK 0xfffff01f
+
+/* Clock detect bits reported by the DSP */
+#define E3G_CLOCK_DETECT_BIT_WORD96 0x0001
+#define E3G_CLOCK_DETECT_BIT_WORD48 0x0002
+#define E3G_CLOCK_DETECT_BIT_SPDIF48 0x0004
+#define E3G_CLOCK_DETECT_BIT_ADAT 0x0004
+#define E3G_CLOCK_DETECT_BIT_SPDIF96 0x0008
+#define E3G_CLOCK_DETECT_BIT_WORD (E3G_CLOCK_DETECT_BIT_WORD96|E3G_CLOCK_DETECT_BIT_WORD48)
+#define E3G_CLOCK_DETECT_BIT_SPDIF (E3G_CLOCK_DETECT_BIT_SPDIF48|E3G_CLOCK_DETECT_BIT_SPDIF96)
+
+/* Frequency control register */
+#define E3G_MAGIC_NUMBER 677376000
+#define E3G_FREQ_REG_DEFAULT (E3G_MAGIC_NUMBER / 48000 - 2)
+#define E3G_FREQ_REG_MAX 0xffff
+
+/* 3G external box types */
+#define E3G_GINA3G_BOX_TYPE 0x00
+#define E3G_LAYLA3G_BOX_TYPE 0x10
+#define E3G_ASIC_NOT_LOADED 0xffff
+#define E3G_BOX_TYPE_MASK 0xf0
+
+#define EXT_3GBOX_NC 0x01
+#define EXT_3GBOX_NOT_SET 0x02
+
+
+/*
+ *
+ * Gina20 & Layla20 have input gain controls for the analog inputs;
+ * this is the magic number for the hardware that gives you 0 dB at -10.
+ *
+ */
+
+#define GL20_INPUT_GAIN_MAGIC_NUMBER 0xC8
+
+
+/*
+ *
+ * Defines how much time must pass between DSP load attempts
+ *
+ */
+
+#define DSP_LOAD_ATTEMPT_PERIOD 1000000L /* One second */
+
+
+/*
+ *
+ * Size of arrays for the comm page. MAX_PLAY_TAPS and MAX_REC_TAPS are
+ * no longer used, but the sizes must still be right for the DSP to see
+ * the comm page correctly.
+ *
+ */
+
+#define MONITOR_ARRAY_SIZE 0x180
+#define VMIXER_ARRAY_SIZE 0x40
+#define MIDI_OUT_BUFFER_SIZE 32
+#define MIDI_IN_BUFFER_SIZE 256
+#define MAX_PLAY_TAPS 168
+#define MAX_REC_TAPS 192
+#define DSP_MIDI_OUT_FIFO_SIZE 64
+
+
+/* sg_entry is a single entry for the scatter-gather list. The array of struct
+sg_entry struct is read by the DSP, so all values must be little-endian. */
+
+#define MAX_SGLIST_ENTRIES 512
+
+struct sg_entry {
+ u32 addr;
+ u32 size;
+};
+
+
+/****************************************************************************
+
+ The comm page. This structure is read and written by the DSP; the
+ DSP code is a firm believer in the byte offsets written in the comments
+ at the end of each line. This structure should not be changed.
+
+ Any reads from or writes to this structure should be in little-endian format.
+
+ ****************************************************************************/
+
+struct comm_page { /* Base Length*/
+ u32 comm_size; /* size of this object 0x000 4 */
+ u32 flags; /* See Appendix A below 0x004 4 */
+ u32 unused; /* Unused entry 0x008 4 */
+ u32 sample_rate; /* Card sample rate in Hz 0x00c 4 */
+ u32 handshake; /* DSP command handshake 0x010 4 */
+ u32 cmd_start; /* Chs. to start mask 0x014 4 */
+ u32 cmd_stop; /* Chs. to stop mask 0x018 4 */
+ u32 cmd_reset; /* Chs. to reset mask 0x01c 4 */
+ u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */
+ struct sg_entry sglist_addr[DSP_MAXPIPES];
+ /* Chs. Physical sglist addrs 0x060 32*8 */
+ u32 position[DSP_MAXPIPES];
+ /* Positions for ea. ch. 0x160 32*4 */
+ s8 vu_meter[DSP_MAXPIPES];
+ /* VU meters 0x1e0 32*1 */
+ s8 peak_meter[DSP_MAXPIPES];
+ /* Peak meters 0x200 32*1 */
+ s8 line_out_level[DSP_MAXAUDIOOUTPUTS];
+ /* Output gain 0x220 16*1 */
+ s8 line_in_level[DSP_MAXAUDIOINPUTS];
+ /* Input gain 0x230 16*1 */
+ s8 monitors[MONITOR_ARRAY_SIZE];
+ /* Monitor map 0x240 0x180 */
+ u32 play_coeff[MAX_PLAY_TAPS];
+ /* Gina/Darla play filters - obsolete 0x3c0 168*4 */
+ u32 rec_coeff[MAX_REC_TAPS];
+ /* Gina/Darla record filters - obsolete 0x660 192*4 */
+ u16 midi_input[MIDI_IN_BUFFER_SIZE];
+ /* MIDI input data transfer buffer 0x960 256*2 */
+ u8 gd_clock_state; /* Chg Gina/Darla clock state 0xb60 1 */
+ u8 gd_spdif_status; /* Chg. Gina/Darla S/PDIF state 0xb61 1 */
+ u8 gd_resampler_state; /* Should always be 3 0xb62 1 */
+ u8 filler2; /* 0xb63 1 */
+ u32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */
+ u16 input_clock; /* Chg. Input clock state 0xb68 2 */
+ u16 output_clock; /* Chg. Output clock state 0xb6a 2 */
+ u32 status_clocks; /* Current Input clock state 0xb6c 4 */
+ u32 ext_box_status; /* External box status 0xb70 4 */
+ u32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */
+ u32 midi_out_free_count;
+ /* # of bytes free in MIDI output FIFO 0xb78 4 */
+ u32 unused2; /* Cyclic pipes 0xb7c 4 */
+ u32 control_register;
+ /* Mona, Gina24, Layla24, 3G ctrl reg 0xb80 4 */
+ u32 e3g_frq_register; /* 3G frequency register 0xb84 4 */
+ u8 filler[24]; /* filler 0xb88 24*1 */
+ s8 vmixer[VMIXER_ARRAY_SIZE];
+ /* Vmixer levels 0xba0 64*1 */
+ u8 midi_output[MIDI_OUT_BUFFER_SIZE];
+ /* MIDI output data 0xbe0 32*1 */
+};
+
+#endif /* _ECHO_DSP_ */
diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c
new file mode 100644
index 0000000..afa2733
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_gml.c
@@ -0,0 +1,200 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+/* These functions are common for Gina24, Layla24 and Mona cards */
+
+
+/* ASIC status check - some cards have one or two ASICs that need to be
+loaded. Once that load is complete, this function is called to see if
+the load was successful.
+If this load fails, it does not necessarily mean that the hardware is
+defective - the external box may be disconnected or turned off. */
+static int check_asic_status(struct echoaudio *chip)
+{
+ u32 asic_status;
+
+ send_vector(chip, DSP_VC_TEST_ASIC);
+
+ /* The DSP will return a value to indicate whether or not the
+ ASIC is currently loaded */
+ if (read_dsp(chip, &asic_status) < 0) {
+ DE_INIT(("check_asic_status: failed on read_dsp\n"));
+ chip->asic_loaded = FALSE;
+ return -EIO;
+ }
+
+ chip->asic_loaded = (asic_status == ASIC_ALREADY_LOADED);
+ return chip->asic_loaded ? 0 : -EIO;
+}
+
+
+
+/* Most configuration of Gina24, Layla24, or Mona is accomplished by writing
+the control register. write_control_reg sends the new control register
+value to the DSP. */
+static int write_control_reg(struct echoaudio *chip, u32 value, char force)
+{
+ /* Handle the digital input auto-mute */
+ if (chip->digital_in_automute)
+ value |= GML_DIGITAL_IN_AUTO_MUTE;
+ else
+ value &= ~GML_DIGITAL_IN_AUTO_MUTE;
+
+ DE_ACT(("write_control_reg: 0x%x\n", value));
+
+ /* Write the control register */
+ value = cpu_to_le32(value);
+ if (value != chip->comm_page->control_register || force) {
+ if (wait_handshake(chip))
+ return -EIO;
+ chip->comm_page->control_register = value;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
+ }
+ return 0;
+}
+
+
+
+/* Gina24, Layla24, and Mona support digital input auto-mute. If the digital
+input auto-mute is enabled, the DSP will only enable the digital inputs if
+the card is syncing to a valid clock on the ADAT or S/PDIF inputs.
+If the auto-mute is disabled, the digital inputs are enabled regardless of
+what the input clock is set or what is connected. */
+static int set_input_auto_mute(struct echoaudio *chip, int automute)
+{
+ DE_ACT(("set_input_auto_mute %d\n", automute));
+
+ chip->digital_in_automute = automute;
+
+ /* Re-set the input clock to the current value - indirectly causes
+ the auto-mute flag to be sent to the DSP */
+ return set_input_clock(chip, chip->input_clock);
+}
+
+
+
+/* S/PDIF coax / S/PDIF optical / ADAT - switch */
+static int set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+ u8 previous_mode;
+ int err, i, o;
+
+ if (chip->bad_board)
+ return -EIO;
+
+ /* All audio channels must be closed before changing the digital mode */
+ if (snd_BUG_ON(chip->pipe_alloc_mask))
+ return -EAGAIN;
+
+ if (snd_BUG_ON(!(chip->digital_modes & (1 << mode))))
+ return -EINVAL;
+
+ previous_mode = chip->digital_mode;
+ err = dsp_set_digital_mode(chip, mode);
+
+ /* If we successfully changed the digital mode from or to ADAT,
+ then make sure all output, input and monitor levels are
+ updated by the DSP comm object. */
+ if (err >= 0 && previous_mode != mode &&
+ (previous_mode == DIGITAL_MODE_ADAT || mode == DIGITAL_MODE_ADAT)) {
+ spin_lock_irq(&chip->lock);
+ for (o = 0; o < num_busses_out(chip); o++)
+ for (i = 0; i < num_busses_in(chip); i++)
+ set_monitor_gain(chip, o, i,
+ chip->monitor_gain[o][i]);
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+ for (i = 0; i < num_busses_in(chip); i++)
+ set_input_gain(chip, i, chip->input_gain[i]);
+ update_input_line_level(chip);
+#endif
+
+ for (o = 0; o < num_busses_out(chip); o++)
+ set_output_gain(chip, o, chip->output_gain[o]);
+ update_output_line_level(chip);
+ spin_unlock_irq(&chip->lock);
+ }
+
+ return err;
+}
+
+
+
+/* Set the S/PDIF output format */
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+ u32 control_reg;
+ int err;
+
+ /* Clear the current S/PDIF flags */
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= GML_SPDIF_FORMAT_CLEAR_MASK;
+
+ /* Set the new S/PDIF flags depending on the mode */
+ control_reg |= GML_SPDIF_TWO_CHANNEL | GML_SPDIF_24_BIT |
+ GML_SPDIF_COPY_PERMIT;
+ if (prof) {
+ /* Professional mode */
+ control_reg |= GML_SPDIF_PRO_MODE;
+
+ switch (chip->sample_rate) {
+ case 32000:
+ control_reg |= GML_SPDIF_SAMPLE_RATE0 |
+ GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 44100:
+ control_reg |= GML_SPDIF_SAMPLE_RATE0;
+ break;
+ case 48000:
+ control_reg |= GML_SPDIF_SAMPLE_RATE1;
+ break;
+ }
+ } else {
+ /* Consumer mode */
+ switch (chip->sample_rate) {
+ case 32000:
+ control_reg |= GML_SPDIF_SAMPLE_RATE0 |
+ GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 48000:
+ control_reg |= GML_SPDIF_SAMPLE_RATE1;
+ break;
+ }
+ }
+
+ if ((err = write_control_reg(chip, control_reg, FALSE)))
+ return err;
+ chip->professional_spdif = prof;
+ DE_ACT(("set_professional_spdif to %s\n",
+ prof ? "Professional" : "Consumer"));
+ return 0;
+}
diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c
new file mode 100644
index 0000000..c0e64b8
--- /dev/null
+++ b/sound/pci/echoaudio/gina20.c
@@ -0,0 +1,105 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_GINA20
+#define ECHOCARD_NAME "Gina20"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_INPUT_GAIN
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT FALSE
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 2 */
+#define PX_ANALOG_IN 10 /* 2 */
+#define PX_DIGITAL_IN 12 /* 2 */
+#define PX_NUM 14
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 8 */
+#define BX_DIGITAL_OUT 8 /* 2 */
+#define BX_ANALOG_IN 10 /* 2 */
+#define BX_DIGITAL_IN 12 /* 2 */
+#define BX_NUM 14
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/gina20_dsp.fw");
+
+#define FW_GINA20_DSP 0
+
+static const struct firmware card_fw[] = {
+ {0, "gina20_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "gina20_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c
new file mode 100644
index 0000000..db6c952
--- /dev/null
+++ b/sound/pci/echoaudio/gina20_dsp.c
@@ -0,0 +1,217 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int update_flags(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Gina20\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP];
+ chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
+ chip->clock_state = GD_CLOCK_UNDEF;
+ /* Since this card has no ASIC, mark it as loaded so everything
+ works OK */
+ chip->asic_loaded = TRUE;
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+ ECHO_CLOCK_BIT_SPDIF;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ err = set_professional_spdif(chip, TRUE);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock
+ detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
+ clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+ return clock_bits;
+}
+
+
+
+/* The Gina20 has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u8 clock_state, spdif_status;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ switch (rate) {
+ case 44100:
+ clock_state = GD_CLOCK_44;
+ spdif_status = GD_SPDIF_STATUS_44;
+ break;
+ case 48000:
+ clock_state = GD_CLOCK_48;
+ spdif_status = GD_SPDIF_STATUS_48;
+ break;
+ default:
+ clock_state = GD_CLOCK_NOCHANGE;
+ spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+ break;
+ }
+
+ if (chip->clock_state == clock_state)
+ clock_state = GD_CLOCK_NOCHANGE;
+ if (spdif_status == chip->spdif_status)
+ spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ chip->comm_page->gd_clock_state = clock_state;
+ chip->comm_page->gd_spdif_status = spdif_status;
+ chip->comm_page->gd_resampler_state = 3; /* magic number - should always be 3 */
+
+ /* Save the new audio state if it changed */
+ if (clock_state != GD_CLOCK_NOCHANGE)
+ chip->clock_state = clock_state;
+ if (spdif_status != GD_SPDIF_STATUS_NOCHANGE)
+ chip->spdif_status = spdif_status;
+ chip->sample_rate = rate;
+
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+ DE_ACT(("set_input_clock:\n"));
+
+ switch (clock) {
+ case ECHO_CLOCK_INTERNAL:
+ /* Reset the audio state to unknown (just in case) */
+ chip->clock_state = GD_CLOCK_UNDEF;
+ chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
+ set_sample_rate(chip, chip->sample_rate);
+ chip->input_clock = clock;
+ DE_ACT(("Set Gina clock to INTERNAL\n"));
+ break;
+ case ECHO_CLOCK_SPDIF:
+ chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN;
+ chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+ chip->clock_state = GD_CLOCK_SPDIFIN;
+ DE_ACT(("Set Gina20 clock to SPDIF\n"));
+ chip->input_clock = clock;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+
+/* Set input bus gain (one unit is 0.5dB !) */
+static int set_input_gain(struct echoaudio *chip, u16 input, int gain)
+{
+ if (snd_BUG_ON(input >= num_busses_in(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->input_gain[input] = gain;
+ gain += GL20_INPUT_GAIN_MAGIC_NUMBER;
+ chip->comm_page->line_in_level[input] = gain;
+ return 0;
+}
+
+
+
+/* Tell the DSP to reread the flags from the comm page */
+static int update_flags(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+ DE_ACT(("set_professional_spdif %d\n", prof));
+ if (prof)
+ chip->comm_page->flags |=
+ __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+ else
+ chip->comm_page->flags &=
+ ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+ chip->professional_spdif = prof;
+ return update_flags(chip);
+}
diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c
new file mode 100644
index 0000000..c36a78d
--- /dev/null
+++ b/sound/pci/echoaudio/gina24.c
@@ -0,0 +1,129 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_GINA24
+#define ECHOCARD_NAME "Gina24"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT 6
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 8 */
+#define PX_ANALOG_IN 16 /* 2 */
+#define PX_DIGITAL_IN 18 /* 8 */
+#define PX_NUM 26
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 8 */
+#define BX_DIGITAL_OUT 8 /* 8 */
+#define BX_ANALOG_IN 16 /* 2 */
+#define BX_DIGITAL_IN 18 /* 8 */
+#define BX_NUM 26
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/gina24_301_dsp.fw");
+MODULE_FIRMWARE("ea/gina24_361_dsp.fw");
+MODULE_FIRMWARE("ea/gina24_301_asic.fw");
+MODULE_FIRMWARE("ea/gina24_361_asic.fw");
+
+#define FW_361_LOADER 0
+#define FW_GINA24_301_DSP 1
+#define FW_GINA24_361_DSP 2
+#define FW_GINA24_301_ASIC 3
+#define FW_GINA24_361_ASIC 4
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "gina24_301_dsp.fw"},
+ {0, "gina24_361_dsp.fw"},
+ {0, "gina24_301_asic.fw"},
+ {0, "gina24_361_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */
+ {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */
+ {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */
+ {0x1057, 0x3410, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56361 Gina24 rev.1 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions.
+ 220 ~= (512 - 1 - (BUFFER_BYTES_MAX / PAGE_SIZE)) / 2 */
+};
+
+#include "gina24_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_gml.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c
new file mode 100644
index 0000000..2fef37a
--- /dev/null
+++ b/sound/pci/echoaudio/gina24_dsp.c
@@ -0,0 +1,349 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int write_control_reg(struct echoaudio *chip, u32 value, char force);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+ const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Gina24\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->input_clock_types =
+ ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+ ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 |
+ ECHO_CLOCK_BIT_ADAT;
+ chip->professional_spdif = FALSE;
+ chip->digital_in_automute = TRUE;
+ chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
+
+ /* Gina24 comes in both '301 and '361 flavors */
+ if (chip->device_id == DEVICE_ID_56361) {
+ chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP];
+ chip->digital_modes =
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+ ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+ } else {
+ chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP];
+ chip->digital_modes =
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+ ECHOCAPS_HAS_DIGITAL_MODE_ADAT |
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_CDROM;
+ }
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+ err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+ if (err < 0)
+ return err;
+ err = set_professional_spdif(chip, TRUE);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock
+ detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
+ clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
+ clock_bits |= ECHO_CLOCK_BIT_ADAT;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ESYNC)
+ clock_bits |= ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96;
+
+ return clock_bits;
+}
+
+
+
+/* Gina24 has an ASIC on the PCI card which must be loaded for anything
+interesting to happen. */
+static int load_asic(struct echoaudio *chip)
+{
+ u32 control_reg;
+ int err;
+ const struct firmware *fw;
+
+ if (chip->asic_loaded)
+ return 1;
+
+ /* Give the DSP a few milliseconds to settle down */
+ mdelay(10);
+
+ /* Pick the correct ASIC for '301 or '361 Gina24 */
+ if (chip->device_id == DEVICE_ID_56361)
+ fw = &card_fw[FW_GINA24_361_ASIC];
+ else
+ fw = &card_fw[FW_GINA24_301_ASIC];
+
+ if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0)
+ return err;
+
+ chip->asic_code = fw;
+
+ /* Now give the new ASIC a little time to set up */
+ mdelay(10);
+ /* See if it worked */
+ err = check_asic_status(chip);
+
+ /* Set up the control register if the load succeeded -
+ 48 kHz, internal clock, S/PDIF RCA mode */
+ if (!err) {
+ control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
+ err = write_control_reg(chip, control_reg, TRUE);
+ }
+ DE_INIT(("load_asic() done\n"));
+ return err;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u32 control_reg, clock;
+
+ if (snd_BUG_ON(rate >= 50000 &&
+ chip->digital_mode == DIGITAL_MODE_ADAT))
+ return -EINVAL;
+
+ /* Only set the clock for internal mode. */
+ if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+ DE_ACT(("set_sample_rate: Cannot set sample rate - "
+ "clock not set to CLK_CLOCKININTERNAL\n"));
+ /* Save the rate anyhow */
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ chip->sample_rate = rate;
+ return 0;
+ }
+
+ clock = 0;
+
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= GML_CLOCK_CLEAR_MASK & GML_SPDIF_RATE_CLEAR_MASK;
+
+ switch (rate) {
+ case 96000:
+ clock = GML_96KHZ;
+ break;
+ case 88200:
+ clock = GML_88KHZ;
+ break;
+ case 48000:
+ clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 44100:
+ clock = GML_44KHZ;
+ /* Professional mode ? */
+ if (control_reg & GML_SPDIF_PRO_MODE)
+ clock |= GML_SPDIF_SAMPLE_RATE0;
+ break;
+ case 32000:
+ clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
+ GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 22050:
+ clock = GML_22KHZ;
+ break;
+ case 16000:
+ clock = GML_16KHZ;
+ break;
+ case 11025:
+ clock = GML_11KHZ;
+ break;
+ case 8000:
+ clock = GML_8KHZ;
+ break;
+ default:
+ DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+ return -EINVAL;
+ }
+
+ control_reg |= clock;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
+ chip->sample_rate = rate;
+ DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+
+ return write_control_reg(chip, control_reg, FALSE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+ u32 control_reg, clocks_from_dsp;
+
+ DE_ACT(("set_input_clock:\n"));
+
+ /* Mask off the clock select bits */
+ control_reg = le32_to_cpu(chip->comm_page->control_register) &
+ GML_CLOCK_CLEAR_MASK;
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ switch (clock) {
+ case ECHO_CLOCK_INTERNAL:
+ DE_ACT(("Set Gina24 clock to INTERNAL\n"));
+ chip->input_clock = ECHO_CLOCK_INTERNAL;
+ return set_sample_rate(chip, chip->sample_rate);
+ case ECHO_CLOCK_SPDIF:
+ if (chip->digital_mode == DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ DE_ACT(("Set Gina24 clock to SPDIF\n"));
+ control_reg |= GML_SPDIF_CLOCK;
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
+ control_reg |= GML_DOUBLE_SPEED_MODE;
+ else
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ case ECHO_CLOCK_ADAT:
+ if (chip->digital_mode != DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ DE_ACT(("Set Gina24 clock to ADAT\n"));
+ control_reg |= GML_ADAT_CLOCK;
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ case ECHO_CLOCK_ESYNC:
+ DE_ACT(("Set Gina24 clock to ESYNC\n"));
+ control_reg |= GML_ESYNC_CLOCK;
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ case ECHO_CLOCK_ESYNC96:
+ DE_ACT(("Set Gina24 clock to ESYNC96\n"));
+ control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE;
+ break;
+ default:
+ DE_ACT(("Input clock 0x%x not supported for Gina24\n", clock));
+ return -EINVAL;
+ }
+
+ chip->input_clock = clock;
+ return write_control_reg(chip, control_reg, TRUE);
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+ u32 control_reg;
+ int err, incompatible_clock;
+
+ /* Set clock to "internal" if it's not compatible with the new mode */
+ incompatible_clock = FALSE;
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ case DIGITAL_MODE_SPDIF_CDROM:
+ case DIGITAL_MODE_SPDIF_RCA:
+ if (chip->input_clock == ECHO_CLOCK_ADAT)
+ incompatible_clock = TRUE;
+ break;
+ case DIGITAL_MODE_ADAT:
+ if (chip->input_clock == ECHO_CLOCK_SPDIF)
+ incompatible_clock = TRUE;
+ break;
+ default:
+ DE_ACT(("Digital mode not supported: %d\n", mode));
+ return -EINVAL;
+ }
+
+ spin_lock_irq(&chip->lock);
+
+ if (incompatible_clock) { /* Switch to 48KHz, internal */
+ chip->sample_rate = 48000;
+ set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+ }
+
+ /* Clear the current digital mode */
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
+
+ /* Tweak the control reg */
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ control_reg |= GML_SPDIF_OPTICAL_MODE;
+ break;
+ case DIGITAL_MODE_SPDIF_CDROM:
+ /* '361 Gina24 cards do not have the S/PDIF CD-ROM mode */
+ if (chip->device_id == DEVICE_ID_56301)
+ control_reg |= GML_SPDIF_CDROM_MODE;
+ break;
+ case DIGITAL_MODE_SPDIF_RCA:
+ /* GML_SPDIF_OPTICAL_MODE bit cleared */
+ break;
+ case DIGITAL_MODE_ADAT:
+ control_reg |= GML_ADAT_MODE;
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ }
+
+ err = write_control_reg(chip, control_reg, TRUE);
+ spin_unlock_irq(&chip->lock);
+ if (err < 0)
+ return err;
+ chip->digital_mode = mode;
+
+ DE_ACT(("set_digital_mode to %d\n", chip->digital_mode));
+ return incompatible_clock;
+}
diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c
new file mode 100644
index 0000000..0a58a7c
--- /dev/null
+++ b/sound/pci/echoaudio/indigo.c
@@ -0,0 +1,107 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO
+#define ECHOCARD_NAME "Indigo"
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 0 */
+#define PX_ANALOG_IN 8 /* 0 */
+#define PX_DIGITAL_IN 8 /* 0 */
+#define PX_NUM 8
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 2 */
+#define BX_DIGITAL_OUT 2 /* 0 */
+#define BX_ANALOG_IN 2 /* 0 */
+#define BX_DIGITAL_IN 2 /* 0 */
+#define BX_NUM 2
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/indigo_dsp.fw");
+
+#define FW_361_LOADER 0
+#define FW_INDIGO_DSP 1
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "indigo_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+};
+
+#include "indigo_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c
new file mode 100644
index 0000000..f05e39f
--- /dev/null
+++ b/sound/pci/echoaudio/indigo_dsp.c
@@ -0,0 +1,172 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Indigo\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP];
+ /* Since this card has no ASIC, mark it as loaded so everything
+ works OK */
+ chip->asic_loaded = TRUE;
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ /* Default routing of the virtual channels: all vchannels are routed
+ to the stereo output */
+ set_vmixer_gain(chip, 0, 0, 0);
+ set_vmixer_gain(chip, 1, 1, 0);
+ set_vmixer_gain(chip, 0, 2, 0);
+ set_vmixer_gain(chip, 1, 3, 0);
+ set_vmixer_gain(chip, 0, 4, 0);
+ set_vmixer_gain(chip, 1, 5, 0);
+ set_vmixer_gain(chip, 0, 6, 0);
+ set_vmixer_gain(chip, 1, 7, 0);
+ err = update_vmixer_level(chip);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The Indigo has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u32 control_reg;
+
+ switch (rate) {
+ case 96000:
+ control_reg = MIA_96000;
+ break;
+ case 88200:
+ control_reg = MIA_88200;
+ break;
+ case 48000:
+ control_reg = MIA_48000;
+ break;
+ case 44100:
+ control_reg = MIA_44100;
+ break;
+ case 32000:
+ control_reg = MIA_32000;
+ break;
+ default:
+ DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+ return -EINVAL;
+ }
+
+ /* Set the control register if it has changed */
+ if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
+ chip->comm_page->control_register = cpu_to_le32(control_reg);
+ chip->sample_rate = rate;
+
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+ }
+ return 0;
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain)
+{
+ int index;
+
+ if (snd_BUG_ON(pipe >= num_pipes_out(chip) ||
+ output >= num_busses_out(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->vmixer_gain[output][pipe] = gain;
+ index = output * num_pipes_out(chip) + pipe;
+ chip->comm_page->vmixer[index] = gain;
+
+ DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+ return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c
new file mode 100644
index 0000000..2db24d2
--- /dev/null
+++ b/sound/pci/echoaudio/indigodj.c
@@ -0,0 +1,107 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_DJ
+#define ECHOCARD_NAME "Indigo DJ"
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 0 */
+#define PX_ANALOG_IN 8 /* 0 */
+#define PX_DIGITAL_IN 8 /* 0 */
+#define PX_NUM 8
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 4 */
+#define BX_DIGITAL_OUT 4 /* 0 */
+#define BX_ANALOG_IN 4 /* 0 */
+#define BX_DIGITAL_IN 4 /* 0 */
+#define BX_NUM 4
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/indigo_dj_dsp.fw");
+
+#define FW_361_LOADER 0
+#define FW_INDIGO_DJ_DSP 1
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "indigo_dj_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 4,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+};
+
+#include "indigodj_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c
new file mode 100644
index 0000000..90730a5
--- /dev/null
+++ b/sound/pci/echoaudio/indigodj_dsp.c
@@ -0,0 +1,172 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Indigo DJ\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP];
+ /* Since this card has no ASIC, mark it as loaded so everything
+ works OK */
+ chip->asic_loaded = TRUE;
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ /* Default routing of the virtual channels: vchannels 0-3 and
+ vchannels 4-7 are routed to real channels 0-4 */
+ set_vmixer_gain(chip, 0, 0, 0);
+ set_vmixer_gain(chip, 1, 1, 0);
+ set_vmixer_gain(chip, 2, 2, 0);
+ set_vmixer_gain(chip, 3, 3, 0);
+ set_vmixer_gain(chip, 0, 4, 0);
+ set_vmixer_gain(chip, 1, 5, 0);
+ set_vmixer_gain(chip, 2, 6, 0);
+ set_vmixer_gain(chip, 3, 7, 0);
+ err = update_vmixer_level(chip);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The IndigoDJ has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u32 control_reg;
+
+ switch (rate) {
+ case 96000:
+ control_reg = MIA_96000;
+ break;
+ case 88200:
+ control_reg = MIA_88200;
+ break;
+ case 48000:
+ control_reg = MIA_48000;
+ break;
+ case 44100:
+ control_reg = MIA_44100;
+ break;
+ case 32000:
+ control_reg = MIA_32000;
+ break;
+ default:
+ DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+ return -EINVAL;
+ }
+
+ /* Set the control register if it has changed */
+ if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
+ chip->comm_page->control_register = cpu_to_le32(control_reg);
+ chip->sample_rate = rate;
+
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+ }
+ return 0;
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain)
+{
+ int index;
+
+ if (snd_BUG_ON(pipe >= num_pipes_out(chip) ||
+ output >= num_busses_out(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->vmixer_gain[output][pipe] = gain;
+ index = output * num_pipes_out(chip) + pipe;
+ chip->comm_page->vmixer[index] = gain;
+
+ DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+ return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c
new file mode 100644
index 0000000..a60c0a0
--- /dev/null
+++ b/sound/pci/echoaudio/indigoio.c
@@ -0,0 +1,108 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_IO
+#define ECHOCARD_NAME "Indigo IO"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 0 */
+#define PX_ANALOG_IN 8 /* 2 */
+#define PX_DIGITAL_IN 10 /* 0 */
+#define PX_NUM 10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 2 */
+#define BX_DIGITAL_OUT 2 /* 0 */
+#define BX_ANALOG_IN 2 /* 2 */
+#define BX_DIGITAL_IN 4 /* 0 */
+#define BX_NUM 4
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/indigo_io_dsp.fw");
+
+#define FW_361_LOADER 0
+#define FW_INDIGO_IO_DSP 1
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "indigo_io_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+};
+
+#include "indigoio_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c
new file mode 100644
index 0000000..a7e09ec
--- /dev/null
+++ b/sound/pci/echoaudio/indigoio_dsp.c
@@ -0,0 +1,143 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Indigo IO\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP];
+ /* Since this card has no ASIC, mark it as loaded so everything
+ works OK */
+ chip->asic_loaded = TRUE;
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ /* Default routing of the virtual channels: all vchannels are routed
+ to the stereo output */
+ set_vmixer_gain(chip, 0, 0, 0);
+ set_vmixer_gain(chip, 1, 1, 0);
+ set_vmixer_gain(chip, 0, 2, 0);
+ set_vmixer_gain(chip, 1, 3, 0);
+ set_vmixer_gain(chip, 0, 4, 0);
+ set_vmixer_gain(chip, 1, 5, 0);
+ set_vmixer_gain(chip, 0, 6, 0);
+ set_vmixer_gain(chip, 1, 7, 0);
+ err = update_vmixer_level(chip);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The IndigoIO has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->sample_rate = rate;
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain)
+{
+ int index;
+
+ if (snd_BUG_ON(pipe >= num_pipes_out(chip) ||
+ output >= num_busses_out(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->vmixer_gain[output][pipe] = gain;
+ index = output * num_pipes_out(chip) + pipe;
+ chip->comm_page->vmixer[index] = gain;
+
+ DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+ return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c
new file mode 100644
index 0000000..5061946
--- /dev/null
+++ b/sound/pci/echoaudio/layla20.c
@@ -0,0 +1,115 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_LAYLA20
+#define ECHOCARD_NAME "Layla20"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_GAIN
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT FALSE
+#define ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH
+#define ECHOCARD_HAS_MIDI
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 10 */
+#define PX_DIGITAL_OUT 10 /* 2 */
+#define PX_ANALOG_IN 12 /* 8 */
+#define PX_DIGITAL_IN 20 /* 2 */
+#define PX_NUM 22
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 10 */
+#define BX_DIGITAL_OUT 10 /* 2 */
+#define BX_ANALOG_IN 12 /* 8 */
+#define BX_DIGITAL_IN 20 /* 2 */
+#define BX_NUM 22
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/layla20_dsp.fw");
+MODULE_FIRMWARE("ea/layla20_asic.fw");
+
+#define FW_LAYLA20_DSP 0
+#define FW_LAYLA20_ASIC 1
+
+static const struct firmware card_fw[] = {
+ {0, "layla20_dsp.fw"},
+ {0, "layla20_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */
+ {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 8000,
+ .rate_max = 50000,
+ .channels_min = 1,
+ .channels_max = 10,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+#include "layla20_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c
new file mode 100644
index 0000000..ede75c6
--- /dev/null
+++ b/sound/pci/echoaudio/layla20_dsp.c
@@ -0,0 +1,293 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int read_dsp(struct echoaudio *chip, u32 *data);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+ const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+static int update_flags(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Layla20\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->has_midi = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP];
+ chip->input_clock_types =
+ ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+ ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER;
+ chip->output_clock_types =
+ ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ err = set_professional_spdif(chip, TRUE);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
+ clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+ if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_WORD) {
+ if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SUPER)
+ clock_bits |= ECHO_CLOCK_BIT_SUPER;
+ else
+ clock_bits |= ECHO_CLOCK_BIT_WORD;
+ }
+
+ return clock_bits;
+}
+
+
+
+/* ASIC status check - some cards have one or two ASICs that need to be
+loaded. Once that load is complete, this function is called to see if
+the load was successful.
+If this load fails, it does not necessarily mean that the hardware is
+defective - the external box may be disconnected or turned off.
+This routine sometimes fails for Layla20; for Layla20, the loop runs
+5 times and succeeds if it wins on three of the loops. */
+static int check_asic_status(struct echoaudio *chip)
+{
+ u32 asic_status;
+ int goodcnt, i;
+
+ chip->asic_loaded = FALSE;
+ for (i = goodcnt = 0; i < 5; i++) {
+ send_vector(chip, DSP_VC_TEST_ASIC);
+
+ /* The DSP will return a value to indicate whether or not
+ the ASIC is currently loaded */
+ if (read_dsp(chip, &asic_status) < 0) {
+ DE_ACT(("check_asic_status: failed on read_dsp\n"));
+ return -EIO;
+ }
+
+ if (asic_status == ASIC_ALREADY_LOADED) {
+ if (++goodcnt == 3) {
+ chip->asic_loaded = TRUE;
+ return 0;
+ }
+ }
+ }
+ return -EIO;
+}
+
+
+
+/* Layla20 has an ASIC in the external box */
+static int load_asic(struct echoaudio *chip)
+{
+ int err;
+
+ if (chip->asic_loaded)
+ return 0;
+
+ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC,
+ &card_fw[FW_LAYLA20_ASIC]);
+ if (err < 0)
+ return err;
+
+ /* Check if ASIC is alive and well. */
+ return check_asic_status(chip);
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ if (snd_BUG_ON(rate < 8000 || rate > 50000))
+ return -EINVAL;
+
+ /* Only set the clock for internal mode. Do not return failure,
+ simply treat it as a non-event. */
+ if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+ DE_ACT(("set_sample_rate: Cannot set sample rate - "
+ "clock not set to CLK_CLOCKININTERNAL\n"));
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ chip->sample_rate = rate;
+ return 0;
+ }
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ DE_ACT(("set_sample_rate(%d)\n", rate));
+ chip->sample_rate = rate;
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_LAYLA_SAMPLE_RATE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock_source)
+{
+ u16 clock;
+ u32 rate;
+
+ DE_ACT(("set_input_clock:\n"));
+ rate = 0;
+ switch (clock_source) {
+ case ECHO_CLOCK_INTERNAL:
+ DE_ACT(("Set Layla20 clock to INTERNAL\n"));
+ rate = chip->sample_rate;
+ clock = LAYLA20_CLOCK_INTERNAL;
+ break;
+ case ECHO_CLOCK_SPDIF:
+ DE_ACT(("Set Layla20 clock to SPDIF\n"));
+ clock = LAYLA20_CLOCK_SPDIF;
+ break;
+ case ECHO_CLOCK_WORD:
+ DE_ACT(("Set Layla20 clock to WORD\n"));
+ clock = LAYLA20_CLOCK_WORD;
+ break;
+ case ECHO_CLOCK_SUPER:
+ DE_ACT(("Set Layla20 clock to SUPER\n"));
+ clock = LAYLA20_CLOCK_SUPER;
+ break;
+ default:
+ DE_ACT(("Input clock 0x%x not supported for Layla24\n",
+ clock_source));
+ return -EINVAL;
+ }
+ chip->input_clock = clock_source;
+
+ chip->comm_page->input_clock = cpu_to_le16(clock);
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+
+ if (rate)
+ set_sample_rate(chip, rate);
+
+ return 0;
+}
+
+
+
+static int set_output_clock(struct echoaudio *chip, u16 clock)
+{
+ DE_ACT(("set_output_clock: %d\n", clock));
+ switch (clock) {
+ case ECHO_CLOCK_SUPER:
+ clock = LAYLA20_OUTPUT_CLOCK_SUPER;
+ break;
+ case ECHO_CLOCK_WORD:
+ clock = LAYLA20_OUTPUT_CLOCK_WORD;
+ break;
+ default:
+ DE_ACT(("set_output_clock wrong clock\n"));
+ return -EINVAL;
+ }
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->output_clock = cpu_to_le16(clock);
+ chip->output_clock = clock;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+}
+
+
+
+/* Set input bus gain (one unit is 0.5dB !) */
+static int set_input_gain(struct echoaudio *chip, u16 input, int gain)
+{
+ if (snd_BUG_ON(input >= num_busses_in(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->input_gain[input] = gain;
+ gain += GL20_INPUT_GAIN_MAGIC_NUMBER;
+ chip->comm_page->line_in_level[input] = gain;
+ return 0;
+}
+
+
+
+/* Tell the DSP to reread the flags from the comm page */
+static int update_flags(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+ DE_ACT(("set_professional_spdif %d\n", prof));
+ if (prof)
+ chip->comm_page->flags |=
+ __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+ else
+ chip->comm_page->flags &=
+ ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+ chip->professional_spdif = prof;
+ return update_flags(chip);
+}
diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c
new file mode 100644
index 0000000..e09e3ea
--- /dev/null
+++ b/sound/pci/echoaudio/layla24.c
@@ -0,0 +1,127 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_LAYLA24
+#define ECHOCARD_NAME "Layla24"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT 6
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+#define ECHOCARD_HAS_MIDI
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 8 */
+#define PX_ANALOG_IN 16 /* 8 */
+#define PX_DIGITAL_IN 24 /* 8 */
+#define PX_NUM 32
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 8 */
+#define BX_DIGITAL_OUT 8 /* 8 */
+#define BX_ANALOG_IN 16 /* 8 */
+#define BX_DIGITAL_IN 24 /* 8 */
+#define BX_NUM 32
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/layla24_dsp.fw");
+MODULE_FIRMWARE("ea/layla24_1_asic.fw");
+MODULE_FIRMWARE("ea/layla24_2A_asic.fw");
+MODULE_FIRMWARE("ea/layla24_2S_asic.fw");
+
+#define FW_361_LOADER 0
+#define FW_LAYLA24_DSP 1
+#define FW_LAYLA24_1_ASIC 2
+#define FW_LAYLA24_2A_ASIC 3
+#define FW_LAYLA24_2S_ASIC 4
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "layla24_dsp.fw"},
+ {0, "layla24_1_asic.fw"},
+ {0, "layla24_2A_asic.fw"},
+ {0, "layla24_2S_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .rate_min = 8000,
+ .rate_max = 100000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "layla24_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_gml.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c
new file mode 100644
index 0000000..d61b5cb
--- /dev/null
+++ b/sound/pci/echoaudio/layla24_dsp.c
@@ -0,0 +1,397 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int write_control_reg(struct echoaudio *chip, u32 value, char force);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+ const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Layla24\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->has_midi = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP];
+ chip->input_clock_types =
+ ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+ ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT;
+ chip->digital_modes =
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+ ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+ chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
+ chip->professional_spdif = FALSE;
+ chip->digital_in_automute = TRUE;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+ if (err < 0)
+ return err;
+ err = set_professional_spdif(chip, TRUE);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
+ clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
+ clock_bits |= ECHO_CLOCK_BIT_ADAT;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD)
+ clock_bits |= ECHO_CLOCK_BIT_WORD;
+
+ return clock_bits;
+}
+
+
+
+/* Layla24 has an ASIC on the PCI card and another ASIC in the external box;
+both need to be loaded. */
+static int load_asic(struct echoaudio *chip)
+{
+ int err;
+
+ if (chip->asic_loaded)
+ return 1;
+
+ DE_INIT(("load_asic\n"));
+
+ /* Give the DSP a few milliseconds to settle down */
+ mdelay(10);
+
+ /* Load the ASIC for the PCI card */
+ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC,
+ &card_fw[FW_LAYLA24_1_ASIC]);
+ if (err < 0)
+ return err;
+
+ chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC];
+
+ /* Now give the new ASIC a little time to set up */
+ mdelay(10);
+
+ /* Do the external one */
+ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC,
+ &card_fw[FW_LAYLA24_2S_ASIC]);
+ if (err < 0)
+ return FALSE;
+
+ /* Now give the external ASIC a little time to set up */
+ mdelay(10);
+
+ /* See if it worked */
+ err = check_asic_status(chip);
+
+ /* Set up the control register if the load succeeded -
+ 48 kHz, internal clock, S/PDIF RCA mode */
+ if (!err)
+ err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ,
+ TRUE);
+
+ DE_INIT(("load_asic() done\n"));
+ return err;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u32 control_reg, clock, base_rate;
+
+ if (snd_BUG_ON(rate >= 50000 &&
+ chip->digital_mode == DIGITAL_MODE_ADAT))
+ return -EINVAL;
+
+ /* Only set the clock for internal mode. */
+ if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+ DE_ACT(("set_sample_rate: Cannot set sample rate - "
+ "clock not set to CLK_CLOCKININTERNAL\n"));
+ /* Save the rate anyhow */
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ chip->sample_rate = rate;
+ return 0;
+ }
+
+ /* Get the control register & clear the appropriate bits */
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= GML_CLOCK_CLEAR_MASK & GML_SPDIF_RATE_CLEAR_MASK;
+
+ clock = 0;
+
+ switch (rate) {
+ case 96000:
+ clock = GML_96KHZ;
+ break;
+ case 88200:
+ clock = GML_88KHZ;
+ break;
+ case 48000:
+ clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 44100:
+ clock = GML_44KHZ;
+ /* Professional mode */
+ if (control_reg & GML_SPDIF_PRO_MODE)
+ clock |= GML_SPDIF_SAMPLE_RATE0;
+ break;
+ case 32000:
+ clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
+ GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 22050:
+ clock = GML_22KHZ;
+ break;
+ case 16000:
+ clock = GML_16KHZ;
+ break;
+ case 11025:
+ clock = GML_11KHZ;
+ break;
+ case 8000:
+ clock = GML_8KHZ;
+ break;
+ default:
+ /* If this is a non-standard rate, then the driver needs to
+ use Layla24's special "continuous frequency" mode */
+ clock = LAYLA24_CONTINUOUS_CLOCK;
+ if (rate > 50000) {
+ base_rate = rate >> 1;
+ control_reg |= GML_DOUBLE_SPEED_MODE;
+ } else {
+ base_rate = rate;
+ }
+
+ if (base_rate < 25000)
+ base_rate = 25000;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->sample_rate =
+ cpu_to_le32(LAYLA24_MAGIC_NUMBER / base_rate - 2);
+
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_SET_LAYLA24_FREQUENCY_REG);
+ }
+
+ control_reg |= clock;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP ? */
+ chip->sample_rate = rate;
+ DE_ACT(("set_sample_rate: %d clock %d\n", rate, control_reg));
+
+ return write_control_reg(chip, control_reg, FALSE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+ u32 control_reg, clocks_from_dsp;
+
+ /* Mask off the clock select bits */
+ control_reg = le32_to_cpu(chip->comm_page->control_register) &
+ GML_CLOCK_CLEAR_MASK;
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ /* Pick the new clock */
+ switch (clock) {
+ case ECHO_CLOCK_INTERNAL:
+ DE_ACT(("Set Layla24 clock to INTERNAL\n"));
+ chip->input_clock = ECHO_CLOCK_INTERNAL;
+ return set_sample_rate(chip, chip->sample_rate);
+ case ECHO_CLOCK_SPDIF:
+ if (chip->digital_mode == DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ control_reg |= GML_SPDIF_CLOCK;
+ /* Layla24 doesn't support 96KHz S/PDIF */
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ DE_ACT(("Set Layla24 clock to SPDIF\n"));
+ break;
+ case ECHO_CLOCK_WORD:
+ control_reg |= GML_WORD_CLOCK;
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96)
+ control_reg |= GML_DOUBLE_SPEED_MODE;
+ else
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ DE_ACT(("Set Layla24 clock to WORD\n"));
+ break;
+ case ECHO_CLOCK_ADAT:
+ if (chip->digital_mode != DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ control_reg |= GML_ADAT_CLOCK;
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ DE_ACT(("Set Layla24 clock to ADAT\n"));
+ break;
+ default:
+ DE_ACT(("Input clock 0x%x not supported for Layla24\n", clock));
+ return -EINVAL;
+ }
+
+ chip->input_clock = clock;
+ return write_control_reg(chip, control_reg, TRUE);
+}
+
+
+
+/* Depending on what digital mode you want, Layla24 needs different ASICs
+loaded. This function checks the ASIC needed for the new mode and sees
+if it matches the one already loaded. */
+static int switch_asic(struct echoaudio *chip, const struct firmware *asic)
+{
+ s8 *monitors;
+
+ /* Check to see if this is already loaded */
+ if (asic != chip->asic_code) {
+ monitors = kmemdup(chip->comm_page->monitors,
+ MONITOR_ARRAY_SIZE, GFP_KERNEL);
+ if (! monitors)
+ return -ENOMEM;
+
+ memset(chip->comm_page->monitors, ECHOGAIN_MUTED,
+ MONITOR_ARRAY_SIZE);
+
+ /* Load the desired ASIC */
+ if (load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC,
+ asic) < 0) {
+ memcpy(chip->comm_page->monitors, monitors,
+ MONITOR_ARRAY_SIZE);
+ kfree(monitors);
+ return -EIO;
+ }
+ chip->asic_code = asic;
+ memcpy(chip->comm_page->monitors, monitors, MONITOR_ARRAY_SIZE);
+ kfree(monitors);
+ }
+
+ return 0;
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+ u32 control_reg;
+ int err, incompatible_clock;
+ const struct firmware *asic;
+
+ /* Set clock to "internal" if it's not compatible with the new mode */
+ incompatible_clock = FALSE;
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ case DIGITAL_MODE_SPDIF_RCA:
+ if (chip->input_clock == ECHO_CLOCK_ADAT)
+ incompatible_clock = TRUE;
+ asic = &card_fw[FW_LAYLA24_2S_ASIC];
+ break;
+ case DIGITAL_MODE_ADAT:
+ if (chip->input_clock == ECHO_CLOCK_SPDIF)
+ incompatible_clock = TRUE;
+ asic = &card_fw[FW_LAYLA24_2A_ASIC];
+ break;
+ default:
+ DE_ACT(("Digital mode not supported: %d\n", mode));
+ return -EINVAL;
+ }
+
+ if (incompatible_clock) { /* Switch to 48KHz, internal */
+ chip->sample_rate = 48000;
+ spin_lock_irq(&chip->lock);
+ set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+ spin_unlock_irq(&chip->lock);
+ }
+
+ /* switch_asic() can sleep */
+ if (switch_asic(chip, asic) < 0)
+ return -EIO;
+
+ spin_lock_irq(&chip->lock);
+
+ /* Tweak the control register */
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
+
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ control_reg |= GML_SPDIF_OPTICAL_MODE;
+ break;
+ case DIGITAL_MODE_SPDIF_RCA:
+ /* GML_SPDIF_OPTICAL_MODE bit cleared */
+ break;
+ case DIGITAL_MODE_ADAT:
+ control_reg |= GML_ADAT_MODE;
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ }
+
+ err = write_control_reg(chip, control_reg, TRUE);
+ spin_unlock_irq(&chip->lock);
+ if (err < 0)
+ return err;
+ chip->digital_mode = mode;
+
+ DE_ACT(("set_digital_mode to %d\n", mode));
+ return incompatible_clock;
+}
diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c
new file mode 100644
index 0000000..f3b9b45
--- /dev/null
+++ b/sound/pci/echoaudio/mia.c
@@ -0,0 +1,120 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_MIA
+#define ECHOCARD_NAME "Mia"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT FALSE
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+#define ECHOCARD_HAS_MIDI
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 8 */
+#define PX_DIGITAL_OUT 8 /* 0 */
+#define PX_ANALOG_IN 8 /* 2 */
+#define PX_DIGITAL_IN 10 /* 2 */
+#define PX_NUM 12
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 2 */
+#define BX_DIGITAL_OUT 2 /* 2 */
+#define BX_ANALOG_IN 4 /* 2 */
+#define BX_DIGITAL_IN 6 /* 2 */
+#define BX_NUM 8
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/mia_dsp.fw");
+
+#define FW_361_LOADER 0
+#define FW_MIA_DSP 1
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "mia_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */
+ {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "mia_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
new file mode 100644
index 0000000..2273866
--- /dev/null
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -0,0 +1,232 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int update_flags(struct echoaudio *chip);
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Mia\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->dsp_code_to_load = &card_fw[FW_MIA_DSP];
+ /* Since this card has no ASIC, mark it as loaded so everything
+ works OK */
+ chip->asic_loaded = TRUE;
+ if ((subdevice_id & 0x0000f) == MIA_MIDI_REV)
+ chip->has_midi = TRUE;
+ chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+ ECHO_CLOCK_BIT_SPDIF;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)))
+ return err;
+
+ /* Default routing of the virtual channels: vchannels 0-3 go to analog
+ outputs and vchannels 4-7 go to S/PDIF outputs */
+ set_vmixer_gain(chip, 0, 0, 0);
+ set_vmixer_gain(chip, 1, 1, 0);
+ set_vmixer_gain(chip, 0, 2, 0);
+ set_vmixer_gain(chip, 1, 3, 0);
+ set_vmixer_gain(chip, 2, 4, 0);
+ set_vmixer_gain(chip, 3, 5, 0);
+ set_vmixer_gain(chip, 2, 6, 0);
+ set_vmixer_gain(chip, 3, 7, 0);
+ err = update_vmixer_level(chip);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock
+ detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
+ clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+ return clock_bits;
+}
+
+
+
+/* The Mia has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u32 control_reg;
+
+ switch (rate) {
+ case 96000:
+ control_reg = MIA_96000;
+ break;
+ case 88200:
+ control_reg = MIA_88200;
+ break;
+ case 48000:
+ control_reg = MIA_48000;
+ break;
+ case 44100:
+ control_reg = MIA_44100;
+ break;
+ case 32000:
+ control_reg = MIA_32000;
+ break;
+ default:
+ DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+ return -EINVAL;
+ }
+
+ /* Override the clock setting if this Mia is set to S/PDIF clock */
+ if (chip->input_clock == ECHO_CLOCK_SPDIF)
+ control_reg |= MIA_SPDIF;
+
+ /* Set the control register if it has changed */
+ if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
+ chip->comm_page->control_register = cpu_to_le32(control_reg);
+ chip->sample_rate = rate;
+
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+ }
+ return 0;
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+ DE_ACT(("set_input_clock(%d)\n", clock));
+ if (snd_BUG_ON(clock != ECHO_CLOCK_INTERNAL &&
+ clock != ECHO_CLOCK_SPDIF))
+ return -EINVAL;
+
+ chip->input_clock = clock;
+ return set_sample_rate(chip, chip->sample_rate);
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+ int gain)
+{
+ int index;
+
+ if (snd_BUG_ON(pipe >= num_pipes_out(chip) ||
+ output >= num_busses_out(chip)))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ chip->vmixer_gain[output][pipe] = gain;
+ index = output * num_pipes_out(chip) + pipe;
+ chip->comm_page->vmixer[index] = gain;
+
+ DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+ return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
+
+
+/* Tell the DSP to reread the flags from the comm page */
+static int update_flags(struct echoaudio *chip)
+{
+ if (wait_handshake(chip))
+ return -EIO;
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+ DE_ACT(("set_professional_spdif %d\n", prof));
+ if (prof)
+ chip->comm_page->flags |=
+ __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+ else
+ chip->comm_page->flags &=
+ ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+ chip->professional_spdif = prof;
+ return update_flags(chip);
+}
+
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
new file mode 100644
index 0000000..77bf2a8
--- /dev/null
+++ b/sound/pci/echoaudio/midi.c
@@ -0,0 +1,331 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+/******************************************************************************
+ MIDI lowlevel code
+******************************************************************************/
+
+/* Start and stop Midi input */
+static int enable_midi_input(struct echoaudio *chip, char enable)
+{
+ DE_MID(("enable_midi_input(%d)\n", enable));
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ if (enable) {
+ chip->mtc_state = MIDI_IN_STATE_NORMAL;
+ chip->comm_page->flags |=
+ __constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+ } else
+ chip->comm_page->flags &=
+ ~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+
+ clear_handshake(chip);
+ return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+/* Send a buffer full of MIDI data to the DSP
+Returns how many actually written or < 0 on error */
+static int write_midi(struct echoaudio *chip, u8 *data, int bytes)
+{
+ if (snd_BUG_ON(bytes <= 0 || bytes >= MIDI_OUT_BUFFER_SIZE))
+ return -EINVAL;
+
+ if (wait_handshake(chip))
+ return -EIO;
+
+ /* HF4 indicates that it is safe to write MIDI output data */
+ if (! (get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_REG_HF4))
+ return 0;
+
+ chip->comm_page->midi_output[0] = bytes;
+ memcpy(&chip->comm_page->midi_output[1], data, bytes);
+ chip->comm_page->midi_out_free_count = 0;
+ clear_handshake(chip);
+ send_vector(chip, DSP_VC_MIDI_WRITE);
+ DE_MID(("write_midi: %d\n", bytes));
+ return bytes;
+}
+
+
+
+/* Run the state machine for MIDI input data
+MIDI time code sync isn't supported by this code right now, but you still need
+this state machine to parse the incoming MIDI data stream. Every time the DSP
+sees a 0xF1 byte come in, it adds the DSP sample position to the MIDI data
+stream. The DSP sample position is represented as a 32 bit unsigned value,
+with the high 16 bits first, followed by the low 16 bits. Since these aren't
+real MIDI bytes, the following logic is needed to skip them. */
+static inline int mtc_process_data(struct echoaudio *chip, short midi_byte)
+{
+ switch (chip->mtc_state) {
+ case MIDI_IN_STATE_NORMAL:
+ if (midi_byte == 0xF1)
+ chip->mtc_state = MIDI_IN_STATE_TS_HIGH;
+ break;
+ case MIDI_IN_STATE_TS_HIGH:
+ chip->mtc_state = MIDI_IN_STATE_TS_LOW;
+ return MIDI_IN_SKIP_DATA;
+ break;
+ case MIDI_IN_STATE_TS_LOW:
+ chip->mtc_state = MIDI_IN_STATE_F1_DATA;
+ return MIDI_IN_SKIP_DATA;
+ break;
+ case MIDI_IN_STATE_F1_DATA:
+ chip->mtc_state = MIDI_IN_STATE_NORMAL;
+ break;
+ }
+ return 0;
+}
+
+
+
+/* This function is called from the IRQ handler and it reads the midi data
+from the DSP's buffer. It returns the number of bytes received. */
+static int midi_service_irq(struct echoaudio *chip)
+{
+ short int count, midi_byte, i, received;
+
+ /* The count is at index 0, followed by actual data */
+ count = le16_to_cpu(chip->comm_page->midi_input[0]);
+
+ if (snd_BUG_ON(count >= MIDI_IN_BUFFER_SIZE))
+ return 0;
+
+ /* Get the MIDI data from the comm page */
+ i = 1;
+ received = 0;
+ for (i = 1; i <= count; i++) {
+ /* Get the MIDI byte */
+ midi_byte = le16_to_cpu(chip->comm_page->midi_input[i]);
+
+ /* Parse the incoming MIDI stream. The incoming MIDI data
+ consists of MIDI bytes and timestamps for the MIDI time code
+ 0xF1 bytes. mtc_process_data() is a little state machine that
+ parses the stream. If you get MIDI_IN_SKIP_DATA back, then
+ this is a timestamp byte, not a MIDI byte, so don't store it
+ in the MIDI input buffer. */
+ if (mtc_process_data(chip, midi_byte) == MIDI_IN_SKIP_DATA)
+ continue;
+
+ chip->midi_buffer[received++] = (u8)midi_byte;
+ }
+
+ return received;
+}
+
+
+
+
+/******************************************************************************
+ MIDI interface
+******************************************************************************/
+
+static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct echoaudio *chip = substream->rmidi->private_data;
+
+ chip->midi_in = substream;
+ DE_MID(("rawmidi_iopen\n"));
+ return 0;
+}
+
+
+
+static void snd_echo_midi_input_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct echoaudio *chip = substream->rmidi->private_data;
+
+ if (up != chip->midi_input_enabled) {
+ spin_lock_irq(&chip->lock);
+ enable_midi_input(chip, up);
+ spin_unlock_irq(&chip->lock);
+ chip->midi_input_enabled = up;
+ }
+}
+
+
+
+static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct echoaudio *chip = substream->rmidi->private_data;
+
+ chip->midi_in = NULL;
+ DE_MID(("rawmidi_iclose\n"));
+ return 0;
+}
+
+
+
+static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct echoaudio *chip = substream->rmidi->private_data;
+
+ chip->tinuse = 0;
+ chip->midi_full = 0;
+ chip->midi_out = substream;
+ DE_MID(("rawmidi_oopen\n"));
+ return 0;
+}
+
+
+
+static void snd_echo_midi_output_write(unsigned long data)
+{
+ struct echoaudio *chip = (struct echoaudio *)data;
+ unsigned long flags;
+ int bytes, sent, time;
+ unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1];
+
+ DE_MID(("snd_echo_midi_output_write\n"));
+ /* No interrupts are involved: we have to check at regular intervals
+ if the card's output buffer has room for new data. */
+ sent = bytes = 0;
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->midi_full = 0;
+ if (!snd_rawmidi_transmit_empty(chip->midi_out)) {
+ bytes = snd_rawmidi_transmit_peek(chip->midi_out, buf,
+ MIDI_OUT_BUFFER_SIZE - 1);
+ DE_MID(("Try to send %d bytes...\n", bytes));
+ sent = write_midi(chip, buf, bytes);
+ if (sent < 0) {
+ snd_printk(KERN_ERR "write_midi() error %d\n", sent);
+ /* retry later */
+ sent = 9000;
+ chip->midi_full = 1;
+ } else if (sent > 0) {
+ DE_MID(("%d bytes sent\n", sent));
+ snd_rawmidi_transmit_ack(chip->midi_out, sent);
+ } else {
+ /* Buffer is full. DSP's internal buffer is 64 (128 ?)
+ bytes long. Let's wait until half of them are sent */
+ DE_MID(("Full\n"));
+ sent = 32;
+ chip->midi_full = 1;
+ }
+ }
+
+ /* We restart the timer only if there is some data left to send */
+ if (!snd_rawmidi_transmit_empty(chip->midi_out) && chip->tinuse) {
+ /* The timer will expire slightly after the data has been
+ sent */
+ time = (sent << 3) / 25 + 1; /* 8/25=0.32ms to send a byte */
+ mod_timer(&chip->timer, jiffies + (time * HZ + 999) / 1000);
+ DE_MID(("Timer armed(%d)\n", ((time * HZ + 999) / 1000)));
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+
+static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct echoaudio *chip = substream->rmidi->private_data;
+
+ DE_MID(("snd_echo_midi_output_trigger(%d)\n", up));
+ spin_lock_irq(&chip->lock);
+ if (up) {
+ if (!chip->tinuse) {
+ init_timer(&chip->timer);
+ chip->timer.function = snd_echo_midi_output_write;
+ chip->timer.data = (unsigned long)chip;
+ chip->tinuse = 1;
+ }
+ } else {
+ if (chip->tinuse) {
+ chip->tinuse = 0;
+ spin_unlock_irq(&chip->lock);
+ del_timer_sync(&chip->timer);
+ DE_MID(("Timer removed\n"));
+ return;
+ }
+ }
+ spin_unlock_irq(&chip->lock);
+
+ if (up && !chip->midi_full)
+ snd_echo_midi_output_write((unsigned long)chip);
+}
+
+
+
+static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct echoaudio *chip = substream->rmidi->private_data;
+
+ chip->midi_out = NULL;
+ DE_MID(("rawmidi_oclose\n"));
+ return 0;
+}
+
+
+
+static struct snd_rawmidi_ops snd_echo_midi_input = {
+ .open = snd_echo_midi_input_open,
+ .close = snd_echo_midi_input_close,
+ .trigger = snd_echo_midi_input_trigger,
+};
+
+static struct snd_rawmidi_ops snd_echo_midi_output = {
+ .open = snd_echo_midi_output_open,
+ .close = snd_echo_midi_output_close,
+ .trigger = snd_echo_midi_output_trigger,
+};
+
+
+
+/* <--snd_echo_probe() */
+static int __devinit snd_echo_midi_create(struct snd_card *card,
+ struct echoaudio *chip)
+{
+ int err;
+
+ if ((err = snd_rawmidi_new(card, card->shortname, 0, 1, 1,
+ &chip->rmidi)) < 0)
+ return err;
+
+ strcpy(chip->rmidi->name, card->shortname);
+ chip->rmidi->private_data = chip;
+
+ snd_rawmidi_set_ops(chip->rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_echo_midi_input);
+ snd_rawmidi_set_ops(chip->rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_echo_midi_output);
+
+ chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
+ DE_INIT(("MIDI ok\n"));
+ return 0;
+}
diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c
new file mode 100644
index 0000000..b05bad9
--- /dev/null
+++ b/sound/pci/echoaudio/mona.c
@@ -0,0 +1,138 @@
+/*
+ * ALSA driver for Echoaudio soundcards.
+ * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_MONA
+#define ECHOCARD_NAME "Mona"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT 6
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT 0 /* 6 */
+#define PX_DIGITAL_OUT 6 /* 8 */
+#define PX_ANALOG_IN 14 /* 4 */
+#define PX_DIGITAL_IN 18 /* 8 */
+#define PX_NUM 26
+
+/* Bus indexes */
+#define BX_ANALOG_OUT 0 /* 6 */
+#define BX_DIGITAL_OUT 6 /* 8 */
+#define BX_ANALOG_IN 14 /* 4 */
+#define BX_DIGITAL_IN 18 /* 8 */
+#define BX_NUM 26
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/mona_301_dsp.fw");
+MODULE_FIRMWARE("ea/mona_361_dsp.fw");
+MODULE_FIRMWARE("ea/mona_301_1_asic_48.fw");
+MODULE_FIRMWARE("ea/mona_301_1_asic_96.fw");
+MODULE_FIRMWARE("ea/mona_361_1_asic_48.fw");
+MODULE_FIRMWARE("ea/mona_361_1_asic_96.fw");
+MODULE_FIRMWARE("ea/mona_2_asic.fw");
+
+#define FW_361_LOADER 0
+#define FW_MONA_301_DSP 1
+#define FW_MONA_361_DSP 2
+#define FW_MONA_301_1_ASIC48 3
+#define FW_MONA_301_1_ASIC96 4
+#define FW_MONA_361_1_ASIC48 5
+#define FW_MONA_361_1_ASIC96 6
+#define FW_MONA_2_ASIC 7
+
+static const struct firmware card_fw[] = {
+ {0, "loader_dsp.fw"},
+ {0, "mona_301_dsp.fw"},
+ {0, "mona_361_dsp.fw"},
+ {0, "mona_301_1_asic_48.fw"},
+ {0, "mona_301_1_asic_96.fw"},
+ {0, "mona_361_1_asic_48.fw"},
+ {0, "mona_361_1_asic_96.fw"},
+ {0, "mona_2_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+ {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */
+ {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */
+ {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */
+ {0x1057, 0x3410, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56361 Mona rev.0 */
+ {0x1057, 0x3410, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56361 Mona rev.1 */
+ {0x1057, 0x3410, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56361 Mona rev.2 */
+ {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 262144,
+ .period_bytes_min = 32,
+ .period_bytes_max = 131072,
+ .periods_min = 2,
+ .periods_max = 220,
+ /* One page (4k) contains 512 instructions. I don't know if the hw
+ supports lists longer than this. In this case periods_max=220 is a
+ safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "mona_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_gml.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c
new file mode 100644
index 0000000..eaa619b
--- /dev/null
+++ b/sound/pci/echoaudio/mona_dsp.c
@@ -0,0 +1,430 @@
+/****************************************************************************
+
+ Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+ All rights reserved
+ www.echoaudio.com
+
+ This file is part of Echo Digital Audio's generic driver library.
+
+ Echo Digital Audio's generic driver library is free software;
+ you can redistribute it and/or modify it under the terms of
+ the GNU General Public License as published by the Free Software
+ Foundation.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+ MA 02111-1307, USA.
+
+ *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int write_control_reg(struct echoaudio *chip, u32 value, char force);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+ const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+ int err;
+
+ DE_INIT(("init_hw() - Mona\n"));
+ if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA))
+ return -ENODEV;
+
+ if ((err = init_dsp_comm_page(chip))) {
+ DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+ return err;
+ }
+
+ chip->device_id = device_id;
+ chip->subdevice_id = subdevice_id;
+ chip->bad_board = TRUE;
+ chip->input_clock_types =
+ ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+ ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT;
+ chip->digital_modes =
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+ ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+ ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+
+ /* Mona comes in both '301 and '361 flavors */
+ if (chip->device_id == DEVICE_ID_56361)
+ chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP];
+ else
+ chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP];
+
+ chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
+ chip->professional_spdif = FALSE;
+ chip->digital_in_automute = TRUE;
+
+ if ((err = load_firmware(chip)) < 0)
+ return err;
+ chip->bad_board = FALSE;
+
+ if ((err = init_line_levels(chip)) < 0)
+ return err;
+
+ err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+ if (err < 0)
+ return err;
+ err = set_professional_spdif(chip, TRUE);
+
+ DE_INIT(("init_hw done\n"));
+ return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+ u32 clocks_from_dsp, clock_bits;
+
+ /* Map the DSP clock detect bits to the generic driver clock
+ detect bits */
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
+ clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
+ clock_bits |= ECHO_CLOCK_BIT_ADAT;
+
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD)
+ clock_bits |= ECHO_CLOCK_BIT_WORD;
+
+ return clock_bits;
+}
+
+
+
+/* Mona has an ASIC on the PCI card and another ASIC in the external box;
+both need to be loaded. */
+static int load_asic(struct echoaudio *chip)
+{
+ u32 control_reg;
+ int err;
+ const struct firmware *asic;
+
+ if (chip->asic_loaded)
+ return 0;
+
+ mdelay(10);
+
+ if (chip->device_id == DEVICE_ID_56361)
+ asic = &card_fw[FW_MONA_361_1_ASIC48];
+ else
+ asic = &card_fw[FW_MONA_301_1_ASIC48];
+
+ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic);
+ if (err < 0)
+ return err;
+
+ chip->asic_code = asic;
+ mdelay(10);
+
+ /* Do the external one */
+ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC,
+ &card_fw[FW_MONA_2_ASIC]);
+ if (err < 0)
+ return err;
+
+ mdelay(10);
+ err = check_asic_status(chip);
+
+ /* Set up the control register if the load succeeded -
+ 48 kHz, internal clock, S/PDIF RCA mode */
+ if (!err) {
+ control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
+ err = write_control_reg(chip, control_reg, TRUE);
+ }
+
+ return err;
+}
+
+
+
+/* Depending on what digital mode you want, Mona needs different ASICs
+loaded. This function checks the ASIC needed for the new mode and sees
+if it matches the one already loaded. */
+static int switch_asic(struct echoaudio *chip, char double_speed)
+{
+ const struct firmware *asic;
+ int err;
+
+ /* Check the clock detect bits to see if this is
+ a single-speed clock or a double-speed clock; load
+ a new ASIC if necessary. */
+ if (chip->device_id == DEVICE_ID_56361) {
+ if (double_speed)
+ asic = &card_fw[FW_MONA_361_1_ASIC96];
+ else
+ asic = &card_fw[FW_MONA_361_1_ASIC48];
+ } else {
+ if (double_speed)
+ asic = &card_fw[FW_MONA_301_1_ASIC96];
+ else
+ asic = &card_fw[FW_MONA_301_1_ASIC48];
+ }
+
+ if (asic != chip->asic_code) {
+ /* Load the desired ASIC */
+ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC,
+ asic);
+ if (err < 0)
+ return err;
+ chip->asic_code = asic;
+ }
+
+ return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+ u32 control_reg, clock;
+ const struct firmware *asic;
+ char force_write;
+
+ /* Only set the clock for internal mode. */
+ if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+ DE_ACT(("set_sample_rate: Cannot set sample rate - "
+ "clock not set to CLK_CLOCKININTERNAL\n"));
+ /* Save the rate anyhow */
+ chip->comm_page->sample_rate = cpu_to_le32(rate);
+ chip->sample_rate = rate;
+ return 0;
+ }
+
+ /* Now, check to see if the required ASIC is loaded */
+ if (rate >= 88200) {
+ if (chip->digital_mode == DIGITAL_MODE_ADAT)
+ return -EINVAL;
+ if (chip->device_id == DEVICE_ID_56361)
+ asic = &card_fw[FW_MONA_361_1_ASIC96];
+ else
+ asic = &card_fw[FW_MONA_301_1_ASIC96];
+ } else {
+ if (chip->device_id == DEVICE_ID_56361)
+ asic = &card_fw[FW_MONA_361_1_ASIC48];
+ else
+ asic = &card_fw[FW_MONA_301_1_ASIC48];
+ }
+
+ force_write = 0;
+ if (asic != chip->asic_code) {
+ int err;
+ /* Load the desired ASIC (load_asic_generic() can sleep) */
+ spin_unlock_irq(&chip->lock);
+ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC,
+ asic);
+ spin_lock_irq(&chip->lock);
+
+ if (err < 0)
+ return err;
+ chip->asic_code = asic;
+ force_write = 1;
+ }
+
+ /* Compute the new control register value */
+ clock = 0;
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= GML_CLOCK_CLEAR_MASK;
+ control_reg &= GML_SPDIF_RATE_CLEAR_MASK;
+
+ switch (rate) {
+ case 96000:
+ clock = GML_96KHZ;
+ break;
+ case 88200:
+ clock = GML_88KHZ;
+ break;
+ case 48000:
+ clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 44100:
+ clock = GML_44KHZ;
+ /* Professional mode */
+ if (control_reg & GML_SPDIF_PRO_MODE)
+ clock |= GML_SPDIF_SAMPLE_RATE0;
+ break;
+ case 32000:
+ clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
+ GML_SPDIF_SAMPLE_RATE1;
+ break;
+ case 22050:
+ clock = GML_22KHZ;
+ break;
+ case 16000:
+ clock = GML_16KHZ;
+ break;
+ case 11025:
+ clock = GML_11KHZ;
+ break;
+ case 8000:
+ clock = GML_8KHZ;
+ break;
+ default:
+ DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+ return -EINVAL;
+ }
+
+ control_reg |= clock;
+
+ chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
+ chip->sample_rate = rate;
+ DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+
+ return write_control_reg(chip, control_reg, force_write);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+ u32 control_reg, clocks_from_dsp;
+ int err;
+
+ DE_ACT(("set_input_clock:\n"));
+
+ /* Prevent two simultaneous calls to switch_asic() */
+ if (atomic_read(&chip->opencount))
+ return -EAGAIN;
+
+ /* Mask off the clock select bits */
+ control_reg = le32_to_cpu(chip->comm_page->control_register) &
+ GML_CLOCK_CLEAR_MASK;
+ clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+ switch (clock) {
+ case ECHO_CLOCK_INTERNAL:
+ DE_ACT(("Set Mona clock to INTERNAL\n"));
+ chip->input_clock = ECHO_CLOCK_INTERNAL;
+ return set_sample_rate(chip, chip->sample_rate);
+ case ECHO_CLOCK_SPDIF:
+ if (chip->digital_mode == DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ spin_unlock_irq(&chip->lock);
+ err = switch_asic(chip, clocks_from_dsp &
+ GML_CLOCK_DETECT_BIT_SPDIF96);
+ spin_lock_irq(&chip->lock);
+ if (err < 0)
+ return err;
+ DE_ACT(("Set Mona clock to SPDIF\n"));
+ control_reg |= GML_SPDIF_CLOCK;
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
+ control_reg |= GML_DOUBLE_SPEED_MODE;
+ else
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ case ECHO_CLOCK_WORD:
+ DE_ACT(("Set Mona clock to WORD\n"));
+ spin_unlock_irq(&chip->lock);
+ err = switch_asic(chip, clocks_from_dsp &
+ GML_CLOCK_DETECT_BIT_WORD96);
+ spin_lock_irq(&chip->lock);
+ if (err < 0)
+ return err;
+ control_reg |= GML_WORD_CLOCK;
+ if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96)
+ control_reg |= GML_DOUBLE_SPEED_MODE;
+ else
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ case ECHO_CLOCK_ADAT:
+ DE_ACT(("Set Mona clock to ADAT\n"));
+ if (chip->digital_mode != DIGITAL_MODE_ADAT)
+ return -EAGAIN;
+ control_reg |= GML_ADAT_CLOCK;
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ default:
+ DE_ACT(("Input clock 0x%x not supported for Mona\n", clock));
+ return -EINVAL;
+ }
+
+ chip->input_clock = clock;
+ return write_control_reg(chip, control_reg, TRUE);
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+ u32 control_reg;
+ int err, incompatible_clock;
+
+ /* Set clock to "internal" if it's not compatible with the new mode */
+ incompatible_clock = FALSE;
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ case DIGITAL_MODE_SPDIF_RCA:
+ if (chip->input_clock == ECHO_CLOCK_ADAT)
+ incompatible_clock = TRUE;
+ break;
+ case DIGITAL_MODE_ADAT:
+ if (chip->input_clock == ECHO_CLOCK_SPDIF)
+ incompatible_clock = TRUE;
+ break;
+ default:
+ DE_ACT(("Digital mode not supported: %d\n", mode));
+ return -EINVAL;
+ }
+
+ spin_lock_irq(&chip->lock);
+
+ if (incompatible_clock) { /* Switch to 48KHz, internal */
+ chip->sample_rate = 48000;
+ set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+ }
+
+ /* Clear the current digital mode */
+ control_reg = le32_to_cpu(chip->comm_page->control_register);
+ control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
+
+ /* Tweak the control reg */
+ switch (mode) {
+ case DIGITAL_MODE_SPDIF_OPTICAL:
+ control_reg |= GML_SPDIF_OPTICAL_MODE;
+ break;
+ case DIGITAL_MODE_SPDIF_RCA:
+ /* GML_SPDIF_OPTICAL_MODE bit cleared */
+ break;
+ case DIGITAL_MODE_ADAT:
+ /* If the current ASIC is the 96KHz ASIC, switch the ASIC
+ and set to 48 KHz */
+ if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] ||
+ chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) {
+ set_sample_rate(chip, 48000);
+ }
+ control_reg |= GML_ADAT_MODE;
+ control_reg &= ~GML_DOUBLE_SPEED_MODE;
+ break;
+ }
+
+ err = write_control_reg(chip, control_reg, FALSE);
+ spin_unlock_irq(&chip->lock);
+ if (err < 0)
+ return err;
+ chip->digital_mode = mode;
+
+ DE_ACT(("set_digital_mode to %d\n", mode));
+ return incompatible_clock;
+}
diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile
new file mode 100644
index 0000000..cf2d563
--- /dev/null
+++ b/sound/pci/emu10k1/Makefile
@@ -0,0 +1,23 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-emu10k1-objs := emu10k1.o emu10k1_main.o \
+ irq.o memory.o voice.o emumpu401.o emupcm.o io.o \
+ emuproc.o emumixer.o emufx.o timer.o p16v.o
+snd-emu10k1-synth-objs := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o
+snd-emu10k1x-objs := emu10k1x.o
+
+#
+# this function returns:
+# "m" - CONFIG_SND_SEQUENCER is m
+# <empty string> - CONFIG_SND_SEQUENCER is undefined
+# otherwise parameter #1 value
+#
+sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_EMU10K1) += snd-emu10k1.o
+obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emu10k1-synth.o
+obj-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
new file mode 100644
index 0000000..8354c1a
--- /dev/null
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -0,0 +1,284 @@
+/*
+ * The driver for the EMU10K1 (SB Live!) based soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Added support for Audigy 2 Value.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/time.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("EMU10K1");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS},"
+ "{Creative Labs,SB Audigy}}");
+
+#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE))
+#define ENABLE_SYNTH
+#include <sound/emu10k1_synth.h>
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int extin[SNDRV_CARDS];
+static int extout[SNDRV_CARDS];
+static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4};
+static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64};
+static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128};
+static int enable_ir[SNDRV_CARDS];
+static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the EMU10K1 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable the EMU10K1 soundcard.");
+module_param_array(extin, int, NULL, 0444);
+MODULE_PARM_DESC(extin, "Available external inputs for FX8010. Zero=default.");
+module_param_array(extout, int, NULL, 0444);
+MODULE_PARM_DESC(extout, "Available external outputs for FX8010. Zero=default.");
+module_param_array(seq_ports, int, NULL, 0444);
+MODULE_PARM_DESC(seq_ports, "Allocated sequencer ports for internal synthesizer.");
+module_param_array(max_synth_voices, int, NULL, 0444);
+MODULE_PARM_DESC(max_synth_voices, "Maximum number of voices for WaveTable.");
+module_param_array(max_buffer_size, int, NULL, 0444);
+MODULE_PARM_DESC(max_buffer_size, "Maximum sample buffer size in MB.");
+module_param_array(enable_ir, bool, NULL, 0444);
+MODULE_PARM_DESC(enable_ir, "Enable IR.");
+module_param_array(subsystem, uint, NULL, 0444);
+MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
+/*
+ * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400
+ */
+static struct pci_device_id snd_emu10k1_ids[] = {
+ { 0x1102, 0x0002, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* EMU10K1 */
+ { 0x1102, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1 }, /* Audigy */
+ { 0x1102, 0x0008, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1 }, /* Audigy 2 Value SB0400 */
+ { 0, }
+};
+
+/*
+ * Audigy 2 Value notes:
+ * A_IOCFG Input (GPIO)
+ * 0x400 = Front analog jack plugged in. (Green socket)
+ * 0x1000 = Read analog jack plugged in. (Black socket)
+ * 0x2000 = Center/LFE analog jack plugged in. (Orange socket)
+ * A_IOCFG Output (GPIO)
+ * 0x60 = Sound out of front Left.
+ * Win sets it to 0xXX61
+ */
+
+MODULE_DEVICE_TABLE(pci, snd_emu10k1_ids);
+
+static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_emu10k1 *emu;
+#ifdef ENABLE_SYNTH
+ struct snd_seq_device *wave = NULL;
+#endif
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ if (max_buffer_size[dev] < 32)
+ max_buffer_size[dev] = 32;
+ else if (max_buffer_size[dev] > 1024)
+ max_buffer_size[dev] = 1024;
+ if ((err = snd_emu10k1_create(card, pci, extin[dev], extout[dev],
+ (long)max_buffer_size[dev] * 1024 * 1024,
+ enable_ir[dev], subsystem[dev],
+ &emu)) < 0)
+ goto error;
+ card->private_data = emu;
+ if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0)
+ goto error;
+ if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0)
+ goto error;
+ if ((err = snd_emu10k1_pcm_efx(emu, 2, NULL)) < 0)
+ goto error;
+ /* This stores the periods table. */
+ if (emu->card_capabilities->ca0151_chip) { /* P16V */
+ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ 1024, &emu->p16v_buffer)) < 0)
+ goto error;
+ }
+
+ if ((err = snd_emu10k1_mixer(emu, 0, 3)) < 0)
+ goto error;
+
+ if ((err = snd_emu10k1_timer(emu, 0)) < 0)
+ goto error;
+
+ if ((err = snd_emu10k1_pcm_multi(emu, 3, NULL)) < 0)
+ goto error;
+ if (emu->card_capabilities->ca0151_chip) { /* P16V */
+ if ((err = snd_p16v_pcm(emu, 4, NULL)) < 0)
+ goto error;
+ }
+ if (emu->audigy) {
+ if ((err = snd_emu10k1_audigy_midi(emu)) < 0)
+ goto error;
+ } else {
+ if ((err = snd_emu10k1_midi(emu)) < 0)
+ goto error;
+ }
+ if ((err = snd_emu10k1_fx8010_new(emu, 0, NULL)) < 0)
+ goto error;
+#ifdef ENABLE_SYNTH
+ if (snd_seq_device_new(card, 1, SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH,
+ sizeof(struct snd_emu10k1_synth_arg), &wave) < 0 ||
+ wave == NULL) {
+ snd_printk(KERN_WARNING "can't initialize Emu10k1 wavetable synth\n");
+ } else {
+ struct snd_emu10k1_synth_arg *arg;
+ arg = SNDRV_SEQ_DEVICE_ARGPTR(wave);
+ strcpy(wave->name, "Emu-10k1 Synth");
+ arg->hwptr = emu;
+ arg->index = 1;
+ arg->seq_ports = seq_ports[dev];
+ arg->max_voices = max_synth_voices[dev];
+ }
+#endif
+
+ strcpy(card->driver, emu->card_capabilities->driver);
+ strcpy(card->shortname, emu->card_capabilities->name);
+ snprintf(card->longname, sizeof(card->longname),
+ "%s (rev.%d, serial:0x%x) at 0x%lx, irq %i",
+ card->shortname, emu->revision, emu->serial, emu->port, emu->irq);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto error;
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+
+ error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_card_emu10k1_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+
+#ifdef CONFIG_PM
+static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_emu10k1 *emu = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ snd_pcm_suspend_all(emu->pcm);
+ snd_pcm_suspend_all(emu->pcm_mic);
+ snd_pcm_suspend_all(emu->pcm_efx);
+ snd_pcm_suspend_all(emu->pcm_multi);
+ snd_pcm_suspend_all(emu->pcm_p16v);
+
+ snd_ac97_suspend(emu->ac97);
+
+ snd_emu10k1_efx_suspend(emu);
+ snd_emu10k1_suspend_regs(emu);
+ if (emu->card_capabilities->ca0151_chip)
+ snd_p16v_suspend(emu);
+
+ snd_emu10k1_done(emu);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_emu10k1_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_emu10k1 *emu = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "emu10k1: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_emu10k1_resume_init(emu);
+ snd_emu10k1_efx_resume(emu);
+ snd_ac97_resume(emu->ac97);
+ snd_emu10k1_resume_regs(emu);
+
+ if (emu->card_capabilities->ca0151_chip)
+ snd_p16v_resume(emu);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct pci_driver driver = {
+ .name = "EMU10K1_Audigy",
+ .id_table = snd_emu10k1_ids,
+ .probe = snd_card_emu10k1_probe,
+ .remove = __devexit_p(snd_card_emu10k1_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_emu10k1_suspend,
+ .resume = snd_emu10k1_resume,
+#endif
+};
+
+static int __init alsa_card_emu10k1_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_emu10k1_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_emu10k1_init)
+module_exit(alsa_card_emu10k1_exit)
diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c
new file mode 100644
index 0000000..0e649dc
--- /dev/null
+++ b/sound/pci/emu10k1/emu10k1_callback.c
@@ -0,0 +1,548 @@
+/*
+ * synth callback routines for Emu10k1
+ *
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "emu10k1_synth_local.h"
+#include <sound/asoundef.h>
+
+/* voice status */
+enum {
+ V_FREE=0, V_OFF, V_RELEASED, V_PLAYING, V_END
+};
+
+/* Keeps track of what we are finding */
+struct best_voice {
+ unsigned int time;
+ int voice;
+};
+
+/*
+ * prototypes
+ */
+static void lookup_voices(struct snd_emux *emux, struct snd_emu10k1 *hw,
+ struct best_voice *best, int active_only);
+static struct snd_emux_voice *get_voice(struct snd_emux *emux,
+ struct snd_emux_port *port);
+static int start_voice(struct snd_emux_voice *vp);
+static void trigger_voice(struct snd_emux_voice *vp);
+static void release_voice(struct snd_emux_voice *vp);
+static void update_voice(struct snd_emux_voice *vp, int update);
+static void terminate_voice(struct snd_emux_voice *vp);
+static void free_voice(struct snd_emux_voice *vp);
+static void set_fmmod(struct snd_emu10k1 *hw, struct snd_emux_voice *vp);
+static void set_fm2frq2(struct snd_emu10k1 *hw, struct snd_emux_voice *vp);
+static void set_filterQ(struct snd_emu10k1 *hw, struct snd_emux_voice *vp);
+
+/*
+ * Ensure a value is between two points
+ * macro evaluates its args more than once, so changed to upper-case.
+ */
+#define LIMITVALUE(x, a, b) do { if ((x) < (a)) (x) = (a); else if ((x) > (b)) (x) = (b); } while (0)
+#define LIMITMAX(x, a) do {if ((x) > (a)) (x) = (a); } while (0)
+
+
+/*
+ * set up operators
+ */
+static struct snd_emux_operators emu10k1_ops = {
+ .owner = THIS_MODULE,
+ .get_voice = get_voice,
+ .prepare = start_voice,
+ .trigger = trigger_voice,
+ .release = release_voice,
+ .update = update_voice,
+ .terminate = terminate_voice,
+ .free_voice = free_voice,
+ .sample_new = snd_emu10k1_sample_new,
+ .sample_free = snd_emu10k1_sample_free,
+};
+
+void
+snd_emu10k1_ops_setup(struct snd_emux *emux)
+{
+ emux->ops = emu10k1_ops;
+}
+
+
+/*
+ * get more voice for pcm
+ *
+ * terminate most inactive voice and give it as a pcm voice.
+ */
+int
+snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw)
+{
+ struct snd_emux *emu;
+ struct snd_emux_voice *vp;
+ struct best_voice best[V_END];
+ unsigned long flags;
+ int i;
+
+ emu = hw->synth;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ lookup_voices(emu, hw, best, 1); /* no OFF voices */
+ for (i = 0; i < V_END; i++) {
+ if (best[i].voice >= 0) {
+ int ch;
+ vp = &emu->voices[best[i].voice];
+ if ((ch = vp->ch) < 0) {
+ //printk("synth_get_voice: ch < 0 (%d) ??", i);
+ continue;
+ }
+ vp->emu->num_voices--;
+ vp->ch = -1;
+ vp->state = SNDRV_EMUX_ST_OFF;
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+ return ch;
+ }
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+
+ /* not found */
+ return -ENOMEM;
+}
+
+
+/*
+ * turn off the voice (not terminated)
+ */
+static void
+release_voice(struct snd_emux_voice *vp)
+{
+ int dcysusv;
+ struct snd_emu10k1 *hw;
+
+ hw = vp->hw;
+ dcysusv = 0x8000 | (unsigned char)vp->reg.parm.modrelease;
+ snd_emu10k1_ptr_write(hw, DCYSUSM, vp->ch, dcysusv);
+ dcysusv = 0x8000 | (unsigned char)vp->reg.parm.volrelease | DCYSUSV_CHANNELENABLE_MASK;
+ snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, dcysusv);
+}
+
+
+/*
+ * terminate the voice
+ */
+static void
+terminate_voice(struct snd_emux_voice *vp)
+{
+ struct snd_emu10k1 *hw;
+
+ if (snd_BUG_ON(!vp))
+ return;
+ hw = vp->hw;
+ snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0x807f | DCYSUSV_CHANNELENABLE_MASK);
+ if (vp->block) {
+ struct snd_emu10k1_memblk *emem;
+ emem = (struct snd_emu10k1_memblk *)vp->block;
+ if (emem->map_locked > 0)
+ emem->map_locked--;
+ }
+}
+
+/*
+ * release the voice to system
+ */
+static void
+free_voice(struct snd_emux_voice *vp)
+{
+ struct snd_emu10k1 *hw;
+
+ hw = vp->hw;
+ /* FIXME: emu10k1_synth is broken. */
+ /* This can get called with hw == 0 */
+ /* Problem apparent on plug, unplug then plug */
+ /* on the Audigy 2 ZS Notebook. */
+ if (hw && (vp->ch >= 0)) {
+ snd_emu10k1_ptr_write(hw, IFATN, vp->ch, 0xff00);
+ snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0x807f | DCYSUSV_CHANNELENABLE_MASK);
+ // snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, 0);
+ snd_emu10k1_ptr_write(hw, VTFT, vp->ch, 0xffff);
+ snd_emu10k1_ptr_write(hw, CVCF, vp->ch, 0xffff);
+ snd_emu10k1_voice_free(hw, &hw->voices[vp->ch]);
+ vp->emu->num_voices--;
+ vp->ch = -1;
+ }
+}
+
+
+/*
+ * update registers
+ */
+static void
+update_voice(struct snd_emux_voice *vp, int update)
+{
+ struct snd_emu10k1 *hw;
+
+ hw = vp->hw;
+ if (update & SNDRV_EMUX_UPDATE_VOLUME)
+ snd_emu10k1_ptr_write(hw, IFATN_ATTENUATION, vp->ch, vp->avol);
+ if (update & SNDRV_EMUX_UPDATE_PITCH)
+ snd_emu10k1_ptr_write(hw, IP, vp->ch, vp->apitch);
+ if (update & SNDRV_EMUX_UPDATE_PAN) {
+ snd_emu10k1_ptr_write(hw, PTRX_FXSENDAMOUNT_A, vp->ch, vp->apan);
+ snd_emu10k1_ptr_write(hw, PTRX_FXSENDAMOUNT_B, vp->ch, vp->aaux);
+ }
+ if (update & SNDRV_EMUX_UPDATE_FMMOD)
+ set_fmmod(hw, vp);
+ if (update & SNDRV_EMUX_UPDATE_TREMFREQ)
+ snd_emu10k1_ptr_write(hw, TREMFRQ, vp->ch, vp->reg.parm.tremfrq);
+ if (update & SNDRV_EMUX_UPDATE_FM2FRQ2)
+ set_fm2frq2(hw, vp);
+ if (update & SNDRV_EMUX_UPDATE_Q)
+ set_filterQ(hw, vp);
+}
+
+
+/*
+ * look up voice table - get the best voice in order of preference
+ */
+/* spinlock held! */
+static void
+lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw,
+ struct best_voice *best, int active_only)
+{
+ struct snd_emux_voice *vp;
+ struct best_voice *bp;
+ int i;
+
+ for (i = 0; i < V_END; i++) {
+ best[i].time = (unsigned int)-1; /* XXX MAX_?INT really */;
+ best[i].voice = -1;
+ }
+
+ /*
+ * Go through them all and get a best one to use.
+ * NOTE: could also look at volume and pick the quietest one.
+ */
+ for (i = 0; i < emu->max_voices; i++) {
+ int state, val;
+
+ vp = &emu->voices[i];
+ state = vp->state;
+ if (state == SNDRV_EMUX_ST_OFF) {
+ if (vp->ch < 0) {
+ if (active_only)
+ continue;
+ bp = best + V_FREE;
+ } else
+ bp = best + V_OFF;
+ }
+ else if (state == SNDRV_EMUX_ST_RELEASED ||
+ state == SNDRV_EMUX_ST_PENDING) {
+ bp = best + V_RELEASED;
+#if 1
+ val = snd_emu10k1_ptr_read(hw, CVCF_CURRENTVOL, vp->ch);
+ if (! val)
+ bp = best + V_OFF;
+#endif
+ }
+ else if (state == SNDRV_EMUX_ST_STANDBY)
+ continue;
+ else if (state & SNDRV_EMUX_ST_ON)
+ bp = best + V_PLAYING;
+ else
+ continue;
+
+ /* check if sample is finished playing (non-looping only) */
+ if (bp != best + V_OFF && bp != best + V_FREE &&
+ (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) {
+ val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch);
+ if (val >= vp->reg.loopstart)
+ bp = best + V_OFF;
+ }
+
+ if (vp->time < bp->time) {
+ bp->time = vp->time;
+ bp->voice = i;
+ }
+ }
+}
+
+/*
+ * get an empty voice
+ *
+ * emu->voice_lock is already held.
+ */
+static struct snd_emux_voice *
+get_voice(struct snd_emux *emu, struct snd_emux_port *port)
+{
+ struct snd_emu10k1 *hw;
+ struct snd_emux_voice *vp;
+ struct best_voice best[V_END];
+ int i;
+
+ hw = emu->hw;
+
+ lookup_voices(emu, hw, best, 0);
+ for (i = 0; i < V_END; i++) {
+ if (best[i].voice >= 0) {
+ vp = &emu->voices[best[i].voice];
+ if (vp->ch < 0) {
+ /* allocate a voice */
+ struct snd_emu10k1_voice *hwvoice;
+ if (snd_emu10k1_voice_alloc(hw, EMU10K1_SYNTH, 1, &hwvoice) < 0 || hwvoice == NULL)
+ continue;
+ vp->ch = hwvoice->number;
+ emu->num_voices++;
+ }
+ return vp;
+ }
+ }
+
+ /* not found */
+ return NULL;
+}
+
+/*
+ * prepare envelopes and LFOs
+ */
+static int
+start_voice(struct snd_emux_voice *vp)
+{
+ unsigned int temp;
+ int ch;
+ unsigned int addr, mapped_offset;
+ struct snd_midi_channel *chan;
+ struct snd_emu10k1 *hw;
+ struct snd_emu10k1_memblk *emem;
+
+ hw = vp->hw;
+ ch = vp->ch;
+ if (snd_BUG_ON(ch < 0))
+ return -EINVAL;
+ chan = vp->chan;
+
+ emem = (struct snd_emu10k1_memblk *)vp->block;
+ if (emem == NULL)
+ return -EINVAL;
+ emem->map_locked++;
+ if (snd_emu10k1_memblk_map(hw, emem) < 0) {
+ // printk("emu: cannot map!\n");
+ return -ENOMEM;
+ }
+ mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1;
+ vp->reg.start += mapped_offset;
+ vp->reg.end += mapped_offset;
+ vp->reg.loopstart += mapped_offset;
+ vp->reg.loopend += mapped_offset;
+
+ /* set channel routing */
+ /* A = left(0), B = right(1), C = reverb(c), D = chorus(d) */
+ if (hw->audigy) {
+ temp = FXBUS_MIDI_LEFT | (FXBUS_MIDI_RIGHT << 8) |
+ (FXBUS_MIDI_REVERB << 16) | (FXBUS_MIDI_CHORUS << 24);
+ snd_emu10k1_ptr_write(hw, A_FXRT1, ch, temp);
+ } else {
+ temp = (FXBUS_MIDI_LEFT << 16) | (FXBUS_MIDI_RIGHT << 20) |
+ (FXBUS_MIDI_REVERB << 24) | (FXBUS_MIDI_CHORUS << 28);
+ snd_emu10k1_ptr_write(hw, FXRT, ch, temp);
+ }
+
+ /* channel to be silent and idle */
+ snd_emu10k1_ptr_write(hw, DCYSUSV, ch, 0x0000);
+ snd_emu10k1_ptr_write(hw, VTFT, ch, 0x0000FFFF);
+ snd_emu10k1_ptr_write(hw, CVCF, ch, 0x0000FFFF);
+ snd_emu10k1_ptr_write(hw, PTRX, ch, 0);
+ snd_emu10k1_ptr_write(hw, CPF, ch, 0);
+
+ /* set pitch offset */
+ snd_emu10k1_ptr_write(hw, IP, vp->ch, vp->apitch);
+
+ /* set envelope parameters */
+ snd_emu10k1_ptr_write(hw, ENVVAL, ch, vp->reg.parm.moddelay);
+ snd_emu10k1_ptr_write(hw, ATKHLDM, ch, vp->reg.parm.modatkhld);
+ snd_emu10k1_ptr_write(hw, DCYSUSM, ch, vp->reg.parm.moddcysus);
+ snd_emu10k1_ptr_write(hw, ENVVOL, ch, vp->reg.parm.voldelay);
+ snd_emu10k1_ptr_write(hw, ATKHLDV, ch, vp->reg.parm.volatkhld);
+ /* decay/sustain parameter for volume envelope is used
+ for triggerg the voice */
+
+ /* cutoff and volume */
+ temp = (unsigned int)vp->acutoff << 8 | (unsigned char)vp->avol;
+ snd_emu10k1_ptr_write(hw, IFATN, vp->ch, temp);
+
+ /* modulation envelope heights */
+ snd_emu10k1_ptr_write(hw, PEFE, ch, vp->reg.parm.pefe);
+
+ /* lfo1/2 delay */
+ snd_emu10k1_ptr_write(hw, LFOVAL1, ch, vp->reg.parm.lfo1delay);
+ snd_emu10k1_ptr_write(hw, LFOVAL2, ch, vp->reg.parm.lfo2delay);
+
+ /* lfo1 pitch & cutoff shift */
+ set_fmmod(hw, vp);
+ /* lfo1 volume & freq */
+ snd_emu10k1_ptr_write(hw, TREMFRQ, vp->ch, vp->reg.parm.tremfrq);
+ /* lfo2 pitch & freq */
+ set_fm2frq2(hw, vp);
+
+ /* reverb and loop start (reverb 8bit, MSB) */
+ temp = vp->reg.parm.reverb;
+ temp += (int)vp->chan->control[MIDI_CTL_E1_REVERB_DEPTH] * 9 / 10;
+ LIMITMAX(temp, 255);
+ addr = vp->reg.loopstart;
+ snd_emu10k1_ptr_write(hw, PSST, vp->ch, (temp << 24) | addr);
+
+ /* chorus & loop end (chorus 8bit, MSB) */
+ addr = vp->reg.loopend;
+ temp = vp->reg.parm.chorus;
+ temp += (int)chan->control[MIDI_CTL_E3_CHORUS_DEPTH] * 9 / 10;
+ LIMITMAX(temp, 255);
+ temp = (temp <<24) | addr;
+ snd_emu10k1_ptr_write(hw, DSL, ch, temp);
+
+ /* clear filter delay memory */
+ snd_emu10k1_ptr_write(hw, Z1, ch, 0);
+ snd_emu10k1_ptr_write(hw, Z2, ch, 0);
+
+ /* invalidate maps */
+ temp = (hw->silent_page.addr << 1) | MAP_PTI_MASK;
+ snd_emu10k1_ptr_write(hw, MAPA, ch, temp);
+ snd_emu10k1_ptr_write(hw, MAPB, ch, temp);
+#if 0
+ /* cache */
+ {
+ unsigned int val, sample;
+ val = 32;
+ if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS)
+ sample = 0x80808080;
+ else {
+ sample = 0;
+ val *= 2;
+ }
+
+ /* cache */
+ snd_emu10k1_ptr_write(hw, CCR, ch, 0x1c << 16);
+ snd_emu10k1_ptr_write(hw, CDE, ch, sample);
+ snd_emu10k1_ptr_write(hw, CDF, ch, sample);
+
+ /* invalidate maps */
+ temp = ((unsigned int)hw->silent_page.addr << 1) | MAP_PTI_MASK;
+ snd_emu10k1_ptr_write(hw, MAPA, ch, temp);
+ snd_emu10k1_ptr_write(hw, MAPB, ch, temp);
+
+ /* fill cache */
+ val -= 4;
+ val <<= 25;
+ val |= 0x1c << 16;
+ snd_emu10k1_ptr_write(hw, CCR, ch, val);
+ }
+#endif
+
+ /* Q & current address (Q 4bit value, MSB) */
+ addr = vp->reg.start;
+ temp = vp->reg.parm.filterQ;
+ temp = (temp<<28) | addr;
+ if (vp->apitch < 0xe400)
+ temp |= CCCA_INTERPROM_0;
+ else {
+ unsigned int shift = (vp->apitch - 0xe000) >> 10;
+ temp |= shift << 25;
+ }
+ if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS)
+ temp |= CCCA_8BITSELECT;
+ snd_emu10k1_ptr_write(hw, CCCA, ch, temp);
+
+ /* reset volume */
+ temp = (unsigned int)vp->vtarget << 16;
+ snd_emu10k1_ptr_write(hw, VTFT, ch, temp | vp->ftarget);
+ snd_emu10k1_ptr_write(hw, CVCF, ch, temp | 0xff00);
+ return 0;
+}
+
+/*
+ * Start envelope
+ */
+static void
+trigger_voice(struct snd_emux_voice *vp)
+{
+ unsigned int temp, ptarget;
+ struct snd_emu10k1 *hw;
+ struct snd_emu10k1_memblk *emem;
+
+ hw = vp->hw;
+
+ emem = (struct snd_emu10k1_memblk *)vp->block;
+ if (! emem || emem->mapped_page < 0)
+ return; /* not mapped */
+
+#if 0
+ ptarget = (unsigned int)vp->ptarget << 16;
+#else
+ ptarget = IP_TO_CP(vp->apitch);
+#endif
+ /* set pitch target and pan (volume) */
+ temp = ptarget | (vp->apan << 8) | vp->aaux;
+ snd_emu10k1_ptr_write(hw, PTRX, vp->ch, temp);
+
+ /* pitch target */
+ snd_emu10k1_ptr_write(hw, CPF, vp->ch, ptarget);
+
+ /* trigger voice */
+ snd_emu10k1_ptr_write(hw, DCYSUSV, vp->ch, vp->reg.parm.voldcysus|DCYSUSV_CHANNELENABLE_MASK);
+}
+
+#define MOD_SENSE 18
+
+/* set lfo1 modulation height and cutoff */
+static void
+set_fmmod(struct snd_emu10k1 *hw, struct snd_emux_voice *vp)
+{
+ unsigned short fmmod;
+ short pitch;
+ unsigned char cutoff;
+ int modulation;
+
+ pitch = (char)(vp->reg.parm.fmmod>>8);
+ cutoff = (vp->reg.parm.fmmod & 0xff);
+ modulation = vp->chan->gm_modulation + vp->chan->midi_pressure;
+ pitch += (MOD_SENSE * modulation) / 1200;
+ LIMITVALUE(pitch, -128, 127);
+ fmmod = ((unsigned char)pitch<<8) | cutoff;
+ snd_emu10k1_ptr_write(hw, FMMOD, vp->ch, fmmod);
+}
+
+/* set lfo2 pitch & frequency */
+static void
+set_fm2frq2(struct snd_emu10k1 *hw, struct snd_emux_voice *vp)
+{
+ unsigned short fm2frq2;
+ short pitch;
+ unsigned char freq;
+ int modulation;
+
+ pitch = (char)(vp->reg.parm.fm2frq2>>8);
+ freq = vp->reg.parm.fm2frq2 & 0xff;
+ modulation = vp->chan->gm_modulation + vp->chan->midi_pressure;
+ pitch += (MOD_SENSE * modulation) / 1200;
+ LIMITVALUE(pitch, -128, 127);
+ fm2frq2 = ((unsigned char)pitch<<8) | freq;
+ snd_emu10k1_ptr_write(hw, FM2FRQ2, vp->ch, fm2frq2);
+}
+
+/* set filterQ */
+static void
+set_filterQ(struct snd_emu10k1 *hw, struct snd_emux_voice *vp)
+{
+ unsigned int val;
+ val = snd_emu10k1_ptr_read(hw, CCCA, vp->ch) & ~CCCA_RESONANCE;
+ val |= (vp->reg.parm.filterQ << 28);
+ snd_emu10k1_ptr_write(hw, CCCA, vp->ch, val);
+}
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
new file mode 100644
index 0000000..de5ee8f
--- /dev/null
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -0,0 +1,2058 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Creative Labs, Inc.
+ * Routines for control of EMU10K1 chips
+ *
+ * Copyright (c) by James Courtier-Dutton <James@superbug.co.uk>
+ * Added support for Audigy 2 Value.
+ * Added EMU 1010 support.
+ * General bug fixes and enhancements.
+ *
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/sched.h>
+#include <linux/kthread.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/mutex.h>
+
+
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+#include <linux/firmware.h>
+#include "p16v.h"
+#include "tina2.h"
+#include "p17v.h"
+
+
+#define HANA_FILENAME "emu/hana.fw"
+#define DOCK_FILENAME "emu/audio_dock.fw"
+#define EMU1010B_FILENAME "emu/emu1010b.fw"
+#define MICRO_DOCK_FILENAME "emu/micro_dock.fw"
+#define EMU0404_FILENAME "emu/emu0404.fw"
+#define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw"
+
+MODULE_FIRMWARE(HANA_FILENAME);
+MODULE_FIRMWARE(DOCK_FILENAME);
+MODULE_FIRMWARE(EMU1010B_FILENAME);
+MODULE_FIRMWARE(MICRO_DOCK_FILENAME);
+MODULE_FIRMWARE(EMU0404_FILENAME);
+MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME);
+
+
+/*************************************************************************
+ * EMU10K1 init / done
+ *************************************************************************/
+
+void snd_emu10k1_voice_init(struct snd_emu10k1 * emu, int ch)
+{
+ snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0);
+ snd_emu10k1_ptr_write(emu, IP, ch, 0);
+ snd_emu10k1_ptr_write(emu, VTFT, ch, 0xffff);
+ snd_emu10k1_ptr_write(emu, CVCF, ch, 0xffff);
+ snd_emu10k1_ptr_write(emu, PTRX, ch, 0);
+ snd_emu10k1_ptr_write(emu, CPF, ch, 0);
+ snd_emu10k1_ptr_write(emu, CCR, ch, 0);
+
+ snd_emu10k1_ptr_write(emu, PSST, ch, 0);
+ snd_emu10k1_ptr_write(emu, DSL, ch, 0x10);
+ snd_emu10k1_ptr_write(emu, CCCA, ch, 0);
+ snd_emu10k1_ptr_write(emu, Z1, ch, 0);
+ snd_emu10k1_ptr_write(emu, Z2, ch, 0);
+ snd_emu10k1_ptr_write(emu, FXRT, ch, 0x32100000);
+
+ snd_emu10k1_ptr_write(emu, ATKHLDM, ch, 0);
+ snd_emu10k1_ptr_write(emu, DCYSUSM, ch, 0);
+ snd_emu10k1_ptr_write(emu, IFATN, ch, 0xffff);
+ snd_emu10k1_ptr_write(emu, PEFE, ch, 0);
+ snd_emu10k1_ptr_write(emu, FMMOD, ch, 0);
+ snd_emu10k1_ptr_write(emu, TREMFRQ, ch, 24); /* 1 Hz */
+ snd_emu10k1_ptr_write(emu, FM2FRQ2, ch, 24); /* 1 Hz */
+ snd_emu10k1_ptr_write(emu, TEMPENV, ch, 0);
+
+ /*** these are last so OFF prevents writing ***/
+ snd_emu10k1_ptr_write(emu, LFOVAL2, ch, 0);
+ snd_emu10k1_ptr_write(emu, LFOVAL1, ch, 0);
+ snd_emu10k1_ptr_write(emu, ATKHLDV, ch, 0);
+ snd_emu10k1_ptr_write(emu, ENVVOL, ch, 0);
+ snd_emu10k1_ptr_write(emu, ENVVAL, ch, 0);
+
+ /* Audigy extra stuffs */
+ if (emu->audigy) {
+ snd_emu10k1_ptr_write(emu, 0x4c, ch, 0); /* ?? */
+ snd_emu10k1_ptr_write(emu, 0x4d, ch, 0); /* ?? */
+ snd_emu10k1_ptr_write(emu, 0x4e, ch, 0); /* ?? */
+ snd_emu10k1_ptr_write(emu, 0x4f, ch, 0); /* ?? */
+ snd_emu10k1_ptr_write(emu, A_FXRT1, ch, 0x03020100);
+ snd_emu10k1_ptr_write(emu, A_FXRT2, ch, 0x3f3f3f3f);
+ snd_emu10k1_ptr_write(emu, A_SENDAMOUNTS, ch, 0);
+ }
+}
+
+static unsigned int spi_dac_init[] = {
+ 0x00ff,
+ 0x02ff,
+ 0x0400,
+ 0x0520,
+ 0x0600,
+ 0x08ff,
+ 0x0aff,
+ 0x0cff,
+ 0x0eff,
+ 0x10ff,
+ 0x1200,
+ 0x1400,
+ 0x1480,
+ 0x1800,
+ 0x1aff,
+ 0x1cff,
+ 0x1e00,
+ 0x0530,
+ 0x0602,
+ 0x0622,
+ 0x1400,
+};
+
+static unsigned int i2c_adc_init[][2] = {
+ { 0x17, 0x00 }, /* Reset */
+ { 0x07, 0x00 }, /* Timeout */
+ { 0x0b, 0x22 }, /* Interface control */
+ { 0x0c, 0x22 }, /* Master mode control */
+ { 0x0d, 0x08 }, /* Powerdown control */
+ { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */
+ { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */
+ { 0x10, 0x7b }, /* ALC Control 1 */
+ { 0x11, 0x00 }, /* ALC Control 2 */
+ { 0x12, 0x32 }, /* ALC Control 3 */
+ { 0x13, 0x00 }, /* Noise gate control */
+ { 0x14, 0xa6 }, /* Limiter control */
+ { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for Audigy 2 ZS Notebook */
+};
+
+static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume)
+{
+ unsigned int silent_page;
+ int ch;
+ u32 tmp;
+
+ /* disable audio and lock cache */
+ outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE,
+ emu->port + HCFG);
+
+ /* reset recording buffers */
+ snd_emu10k1_ptr_write(emu, MICBS, 0, ADCBS_BUFSIZE_NONE);
+ snd_emu10k1_ptr_write(emu, MICBA, 0, 0);
+ snd_emu10k1_ptr_write(emu, FXBS, 0, ADCBS_BUFSIZE_NONE);
+ snd_emu10k1_ptr_write(emu, FXBA, 0, 0);
+ snd_emu10k1_ptr_write(emu, ADCBS, 0, ADCBS_BUFSIZE_NONE);
+ snd_emu10k1_ptr_write(emu, ADCBA, 0, 0);
+
+ /* disable channel interrupt */
+ outl(0, emu->port + INTE);
+ snd_emu10k1_ptr_write(emu, CLIEL, 0, 0);
+ snd_emu10k1_ptr_write(emu, CLIEH, 0, 0);
+ snd_emu10k1_ptr_write(emu, SOLEL, 0, 0);
+ snd_emu10k1_ptr_write(emu, SOLEH, 0, 0);
+
+ if (emu->audigy){
+ /* set SPDIF bypass mode */
+ snd_emu10k1_ptr_write(emu, SPBYPASS, 0, SPBYPASS_FORMAT);
+ /* enable rear left + rear right AC97 slots */
+ snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_REAR_RIGHT |
+ AC97SLOT_REAR_LEFT);
+ }
+
+ /* init envelope engine */
+ for (ch = 0; ch < NUM_G; ch++)
+ snd_emu10k1_voice_init(emu, ch);
+
+ snd_emu10k1_ptr_write(emu, SPCS0, 0, emu->spdif_bits[0]);
+ snd_emu10k1_ptr_write(emu, SPCS1, 0, emu->spdif_bits[1]);
+ snd_emu10k1_ptr_write(emu, SPCS2, 0, emu->spdif_bits[2]);
+
+ if (emu->card_capabilities->ca0151_chip) { /* audigy2 */
+ /* Hacks for Alice3 to work independent of haP16V driver */
+ //Setup SRCMulti_I2S SamplingRate
+ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0);
+ tmp &= 0xfffff1ff;
+ tmp |= (0x2<<9);
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp);
+
+ /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */
+ snd_emu10k1_ptr20_write(emu, SRCSel, 0, 0x14);
+ /* Setup SRCMulti Input Audio Enable */
+ /* Use 0xFFFFFFFF to enable P16V sounds. */
+ snd_emu10k1_ptr20_write(emu, SRCMULTI_ENABLE, 0, 0xFFFFFFFF);
+
+ /* Enabled Phased (8-channel) P16V playback */
+ outl(0x0201, emu->port + HCFG2);
+ /* Set playback routing. */
+ snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, 0x78e4);
+ }
+ if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */
+ /* Hacks for Alice3 to work independent of haP16V driver */
+ snd_printk(KERN_INFO "Audigy2 value: Special config.\n");
+ //Setup SRCMulti_I2S SamplingRate
+ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0);
+ tmp &= 0xfffff1ff;
+ tmp |= (0x2<<9);
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp);
+
+ /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */
+ outl(0x600000, emu->port + 0x20);
+ outl(0x14, emu->port + 0x24);
+
+ /* Setup SRCMulti Input Audio Enable */
+ outl(0x7b0000, emu->port + 0x20);
+ outl(0xFF000000, emu->port + 0x24);
+
+ /* Setup SPDIF Out Audio Enable */
+ /* The Audigy 2 Value has a separate SPDIF out,
+ * so no need for a mixer switch
+ */
+ outl(0x7a0000, emu->port + 0x20);
+ outl(0xFF000000, emu->port + 0x24);
+ tmp = inl(emu->port + A_IOCFG) & ~0x8; /* Clear bit 3 */
+ outl(tmp, emu->port + A_IOCFG);
+ }
+ if (emu->card_capabilities->spi_dac) { /* Audigy 2 ZS Notebook with DAC Wolfson WM8768/WM8568 */
+ int size, n;
+
+ size = ARRAY_SIZE(spi_dac_init);
+ for (n = 0; n < size; n++)
+ snd_emu10k1_spi_write(emu, spi_dac_init[n]);
+
+ snd_emu10k1_ptr20_write(emu, 0x60, 0, 0x10);
+ /* Enable GPIOs
+ * GPIO0: Unknown
+ * GPIO1: Speakers-enabled.
+ * GPIO2: Unknown
+ * GPIO3: Unknown
+ * GPIO4: IEC958 Output on.
+ * GPIO5: Unknown
+ * GPIO6: Unknown
+ * GPIO7: Unknown
+ */
+ outl(0x76, emu->port + A_IOCFG); /* Windows uses 0x3f76 */
+ }
+ if (emu->card_capabilities->i2c_adc) { /* Audigy 2 ZS Notebook with ADC Wolfson WM8775 */
+ int size, n;
+
+ snd_emu10k1_ptr20_write(emu, P17V_I2S_SRC_SEL, 0, 0x2020205f);
+ tmp = inl(emu->port + A_IOCFG);
+ outl(tmp | 0x4, emu->port + A_IOCFG); /* Set bit 2 for mic input */
+ tmp = inl(emu->port + A_IOCFG);
+ size = ARRAY_SIZE(i2c_adc_init);
+ for (n = 0; n < size; n++)
+ snd_emu10k1_i2c_write(emu, i2c_adc_init[n][0], i2c_adc_init[n][1]);
+ for (n=0; n < 4; n++) {
+ emu->i2c_capture_volume[n][0]= 0xcf;
+ emu->i2c_capture_volume[n][1]= 0xcf;
+ }
+ }
+
+
+ snd_emu10k1_ptr_write(emu, PTB, 0, emu->ptb_pages.addr);
+ snd_emu10k1_ptr_write(emu, TCB, 0, 0); /* taken from original driver */
+ snd_emu10k1_ptr_write(emu, TCBS, 0, 4); /* taken from original driver */
+
+ silent_page = (emu->silent_page.addr << 1) | MAP_PTI_MASK;
+ for (ch = 0; ch < NUM_G; ch++) {
+ snd_emu10k1_ptr_write(emu, MAPA, ch, silent_page);
+ snd_emu10k1_ptr_write(emu, MAPB, ch, silent_page);
+ }
+
+ if (emu->card_capabilities->emu_model) {
+ outl(HCFG_AUTOMUTE_ASYNC |
+ HCFG_EMU32_SLAVE |
+ HCFG_AUDIOENABLE, emu->port + HCFG);
+ /*
+ * Hokay, setup HCFG
+ * Mute Disable Audio = 0
+ * Lock Tank Memory = 1
+ * Lock Sound Memory = 0
+ * Auto Mute = 1
+ */
+ } else if (emu->audigy) {
+ if (emu->revision == 4) /* audigy2 */
+ outl(HCFG_AUDIOENABLE |
+ HCFG_AC3ENABLE_CDSPDIF |
+ HCFG_AC3ENABLE_GPSPDIF |
+ HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG);
+ else
+ outl(HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG);
+ /* FIXME: Remove all these emu->model and replace it with a card recognition parameter,
+ * e.g. card_capabilities->joystick */
+ } else if (emu->model == 0x20 ||
+ emu->model == 0xc400 ||
+ (emu->model == 0x21 && emu->revision < 6))
+ outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE, emu->port + HCFG);
+ else
+ // With on-chip joystick
+ outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG);
+
+ if (enable_ir) { /* enable IR for SB Live */
+ if (emu->card_capabilities->emu_model) {
+ ; /* Disable all access to A_IOCFG for the emu1010 */
+ } else if (emu->card_capabilities->i2c_adc) {
+ ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */
+ } else if (emu->audigy) {
+ unsigned int reg = inl(emu->port + A_IOCFG);
+ outl(reg | A_IOCFG_GPOUT2, emu->port + A_IOCFG);
+ udelay(500);
+ outl(reg | A_IOCFG_GPOUT1 | A_IOCFG_GPOUT2, emu->port + A_IOCFG);
+ udelay(100);
+ outl(reg, emu->port + A_IOCFG);
+ } else {
+ unsigned int reg = inl(emu->port + HCFG);
+ outl(reg | HCFG_GPOUT2, emu->port + HCFG);
+ udelay(500);
+ outl(reg | HCFG_GPOUT1 | HCFG_GPOUT2, emu->port + HCFG);
+ udelay(100);
+ outl(reg, emu->port + HCFG);
+ }
+ }
+
+ if (emu->card_capabilities->emu_model) {
+ ; /* Disable all access to A_IOCFG for the emu1010 */
+ } else if (emu->card_capabilities->i2c_adc) {
+ ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */
+ } else if (emu->audigy) { /* enable analog output */
+ unsigned int reg = inl(emu->port + A_IOCFG);
+ outl(reg | A_IOCFG_GPOUT0, emu->port + A_IOCFG);
+ }
+
+ return 0;
+}
+
+static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu)
+{
+ /*
+ * Enable the audio bit
+ */
+ outl(inl(emu->port + HCFG) | HCFG_AUDIOENABLE, emu->port + HCFG);
+
+ /* Enable analog/digital outs on audigy */
+ if (emu->card_capabilities->emu_model) {
+ ; /* Disable all access to A_IOCFG for the emu1010 */
+ } else if (emu->card_capabilities->i2c_adc) {
+ ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */
+ } else if (emu->audigy) {
+ outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG);
+
+ if (emu->card_capabilities->ca0151_chip) { /* audigy2 */
+ /* Unmute Analog now. Set GPO6 to 1 for Apollo.
+ * This has to be done after init ALice3 I2SOut beyond 48KHz.
+ * So, sequence is important. */
+ outl(inl(emu->port + A_IOCFG) | 0x0040, emu->port + A_IOCFG);
+ } else if (emu->card_capabilities->ca0108_chip) { /* audigy2 value */
+ /* Unmute Analog now. */
+ outl(inl(emu->port + A_IOCFG) | 0x0060, emu->port + A_IOCFG);
+ } else {
+ /* Disable routing from AC97 line out to Front speakers */
+ outl(inl(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG);
+ }
+ }
+
+#if 0
+ {
+ unsigned int tmp;
+ /* FIXME: the following routine disables LiveDrive-II !! */
+ // TOSLink detection
+ emu->tos_link = 0;
+ tmp = inl(emu->port + HCFG);
+ if (tmp & (HCFG_GPINPUT0 | HCFG_GPINPUT1)) {
+ outl(tmp|0x800, emu->port + HCFG);
+ udelay(50);
+ if (tmp != (inl(emu->port + HCFG) & ~0x800)) {
+ emu->tos_link = 1;
+ outl(tmp, emu->port + HCFG);
+ }
+ }
+ }
+#endif
+
+ snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE);
+}
+
+int snd_emu10k1_done(struct snd_emu10k1 * emu)
+{
+ int ch;
+
+ outl(0, emu->port + INTE);
+
+ /*
+ * Shutdown the chip
+ */
+ for (ch = 0; ch < NUM_G; ch++)
+ snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0);
+ for (ch = 0; ch < NUM_G; ch++) {
+ snd_emu10k1_ptr_write(emu, VTFT, ch, 0);
+ snd_emu10k1_ptr_write(emu, CVCF, ch, 0);
+ snd_emu10k1_ptr_write(emu, PTRX, ch, 0);
+ snd_emu10k1_ptr_write(emu, CPF, ch, 0);
+ }
+
+ /* reset recording buffers */
+ snd_emu10k1_ptr_write(emu, MICBS, 0, 0);
+ snd_emu10k1_ptr_write(emu, MICBA, 0, 0);
+ snd_emu10k1_ptr_write(emu, FXBS, 0, 0);
+ snd_emu10k1_ptr_write(emu, FXBA, 0, 0);
+ snd_emu10k1_ptr_write(emu, FXWC, 0, 0);
+ snd_emu10k1_ptr_write(emu, ADCBS, 0, ADCBS_BUFSIZE_NONE);
+ snd_emu10k1_ptr_write(emu, ADCBA, 0, 0);
+ snd_emu10k1_ptr_write(emu, TCBS, 0, TCBS_BUFFSIZE_16K);
+ snd_emu10k1_ptr_write(emu, TCB, 0, 0);
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, A_DBG_SINGLE_STEP);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, EMU10K1_DBG_SINGLE_STEP);
+
+ /* disable channel interrupt */
+ snd_emu10k1_ptr_write(emu, CLIEL, 0, 0);
+ snd_emu10k1_ptr_write(emu, CLIEH, 0, 0);
+ snd_emu10k1_ptr_write(emu, SOLEL, 0, 0);
+ snd_emu10k1_ptr_write(emu, SOLEH, 0, 0);
+
+ /* disable audio and lock cache */
+ outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE, emu->port + HCFG);
+ snd_emu10k1_ptr_write(emu, PTB, 0, 0);
+
+ return 0;
+}
+
+/*************************************************************************
+ * ECARD functional implementation
+ *************************************************************************/
+
+/* In A1 Silicon, these bits are in the HC register */
+#define HOOKN_BIT (1L << 12)
+#define HANDN_BIT (1L << 11)
+#define PULSEN_BIT (1L << 10)
+
+#define EC_GDI1 (1 << 13)
+#define EC_GDI0 (1 << 14)
+
+#define EC_NUM_CONTROL_BITS 20
+
+#define EC_AC3_DATA_SELN 0x0001L
+#define EC_EE_DATA_SEL 0x0002L
+#define EC_EE_CNTRL_SELN 0x0004L
+#define EC_EECLK 0x0008L
+#define EC_EECS 0x0010L
+#define EC_EESDO 0x0020L
+#define EC_TRIM_CSN 0x0040L
+#define EC_TRIM_SCLK 0x0080L
+#define EC_TRIM_SDATA 0x0100L
+#define EC_TRIM_MUTEN 0x0200L
+#define EC_ADCCAL 0x0400L
+#define EC_ADCRSTN 0x0800L
+#define EC_DACCAL 0x1000L
+#define EC_DACMUTEN 0x2000L
+#define EC_LEDN 0x4000L
+
+#define EC_SPDIF0_SEL_SHIFT 15
+#define EC_SPDIF1_SEL_SHIFT 17
+#define EC_SPDIF0_SEL_MASK (0x3L << EC_SPDIF0_SEL_SHIFT)
+#define EC_SPDIF1_SEL_MASK (0x7L << EC_SPDIF1_SEL_SHIFT)
+#define EC_SPDIF0_SELECT(_x) (((_x) << EC_SPDIF0_SEL_SHIFT) & EC_SPDIF0_SEL_MASK)
+#define EC_SPDIF1_SELECT(_x) (((_x) << EC_SPDIF1_SEL_SHIFT) & EC_SPDIF1_SEL_MASK)
+#define EC_CURRENT_PROM_VERSION 0x01 /* Self-explanatory. This should
+ * be incremented any time the EEPROM's
+ * format is changed. */
+
+#define EC_EEPROM_SIZE 0x40 /* ECARD EEPROM has 64 16-bit words */
+
+/* Addresses for special values stored in to EEPROM */
+#define EC_PROM_VERSION_ADDR 0x20 /* Address of the current prom version */
+#define EC_BOARDREV0_ADDR 0x21 /* LSW of board rev */
+#define EC_BOARDREV1_ADDR 0x22 /* MSW of board rev */
+
+#define EC_LAST_PROMFILE_ADDR 0x2f
+
+#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The
+ * can be up to 30 characters in length
+ * and is stored as a NULL-terminated
+ * ASCII string. Any unused bytes must be
+ * filled with zeros */
+#define EC_CHECKSUM_ADDR 0x3f /* Location at which checksum is stored */
+
+
+/* Most of this stuff is pretty self-evident. According to the hardware
+ * dudes, we need to leave the ADCCAL bit low in order to avoid a DC
+ * offset problem. Weird.
+ */
+#define EC_RAW_RUN_MODE (EC_DACMUTEN | EC_ADCRSTN | EC_TRIM_MUTEN | \
+ EC_TRIM_CSN)
+
+
+#define EC_DEFAULT_ADC_GAIN 0xC4C4
+#define EC_DEFAULT_SPDIF0_SEL 0x0
+#define EC_DEFAULT_SPDIF1_SEL 0x4
+
+/**************************************************************************
+ * @func Clock bits into the Ecard's control latch. The Ecard uses a
+ * control latch will is loaded bit-serially by toggling the Modem control
+ * lines from function 2 on the E8010. This function hides these details
+ * and presents the illusion that we are actually writing to a distinct
+ * register.
+ */
+
+static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value)
+{
+ unsigned short count;
+ unsigned int data;
+ unsigned long hc_port;
+ unsigned int hc_value;
+
+ hc_port = emu->port + HCFG;
+ hc_value = inl(hc_port) & ~(HOOKN_BIT | HANDN_BIT | PULSEN_BIT);
+ outl(hc_value, hc_port);
+
+ for (count = 0; count < EC_NUM_CONTROL_BITS; count++) {
+
+ /* Set up the value */
+ data = ((value & 0x1) ? PULSEN_BIT : 0);
+ value >>= 1;
+
+ outl(hc_value | data, hc_port);
+
+ /* Clock the shift register */
+ outl(hc_value | data | HANDN_BIT, hc_port);
+ outl(hc_value | data, hc_port);
+ }
+
+ /* Latch the bits */
+ outl(hc_value | HOOKN_BIT, hc_port);
+ outl(hc_value, hc_port);
+}
+
+/**************************************************************************
+ * @func Set the gain of the ECARD's CS3310 Trim/gain controller. The
+ * trim value consists of a 16bit value which is composed of two
+ * 8 bit gain/trim values, one for the left channel and one for the
+ * right channel. The following table maps from the Gain/Attenuation
+ * value in decibels into the corresponding bit pattern for a single
+ * channel.
+ */
+
+static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu,
+ unsigned short gain)
+{
+ unsigned int bit;
+
+ /* Enable writing to the TRIM registers */
+ snd_emu10k1_ecard_write(emu, emu->ecard_ctrl & ~EC_TRIM_CSN);
+
+ /* Do it again to insure that we meet hold time requirements */
+ snd_emu10k1_ecard_write(emu, emu->ecard_ctrl & ~EC_TRIM_CSN);
+
+ for (bit = (1 << 15); bit; bit >>= 1) {
+ unsigned int value;
+
+ value = emu->ecard_ctrl & ~(EC_TRIM_CSN | EC_TRIM_SDATA);
+
+ if (gain & bit)
+ value |= EC_TRIM_SDATA;
+
+ /* Clock the bit */
+ snd_emu10k1_ecard_write(emu, value);
+ snd_emu10k1_ecard_write(emu, value | EC_TRIM_SCLK);
+ snd_emu10k1_ecard_write(emu, value);
+ }
+
+ snd_emu10k1_ecard_write(emu, emu->ecard_ctrl);
+}
+
+static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu)
+{
+ unsigned int hc_value;
+
+ /* Set up the initial settings */
+ emu->ecard_ctrl = EC_RAW_RUN_MODE |
+ EC_SPDIF0_SELECT(EC_DEFAULT_SPDIF0_SEL) |
+ EC_SPDIF1_SELECT(EC_DEFAULT_SPDIF1_SEL);
+
+ /* Step 0: Set the codec type in the hardware control register
+ * and enable audio output */
+ hc_value = inl(emu->port + HCFG);
+ outl(hc_value | HCFG_AUDIOENABLE | HCFG_CODECFORMAT_I2S, emu->port + HCFG);
+ inl(emu->port + HCFG);
+
+ /* Step 1: Turn off the led and deassert TRIM_CS */
+ snd_emu10k1_ecard_write(emu, EC_ADCCAL | EC_LEDN | EC_TRIM_CSN);
+
+ /* Step 2: Calibrate the ADC and DAC */
+ snd_emu10k1_ecard_write(emu, EC_DACCAL | EC_LEDN | EC_TRIM_CSN);
+
+ /* Step 3: Wait for awhile; XXX We can't get away with this
+ * under a real operating system; we'll need to block and wait that
+ * way. */
+ snd_emu10k1_wait(emu, 48000);
+
+ /* Step 4: Switch off the DAC and ADC calibration. Note
+ * That ADC_CAL is actually an inverted signal, so we assert
+ * it here to stop calibration. */
+ snd_emu10k1_ecard_write(emu, EC_ADCCAL | EC_LEDN | EC_TRIM_CSN);
+
+ /* Step 4: Switch into run mode */
+ snd_emu10k1_ecard_write(emu, emu->ecard_ctrl);
+
+ /* Step 5: Set the analog input gain */
+ snd_emu10k1_ecard_setadcgain(emu, EC_DEFAULT_ADC_GAIN);
+
+ return 0;
+}
+
+static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu)
+{
+ unsigned long special_port;
+ unsigned int value;
+
+ /* Special initialisation routine
+ * before the rest of the IO-Ports become active.
+ */
+ special_port = emu->port + 0x38;
+ value = inl(special_port);
+ outl(0x00d00000, special_port);
+ value = inl(special_port);
+ outl(0x00d00001, special_port);
+ value = inl(special_port);
+ outl(0x00d0005f, special_port);
+ value = inl(special_port);
+ outl(0x00d0007f, special_port);
+ value = inl(special_port);
+ outl(0x0090007f, special_port);
+ value = inl(special_port);
+
+ snd_emu10k1_ptr20_write(emu, TINA2_VOLUME, 0, 0xfefefefe); /* Defaults to 0x30303030 */
+ /* Delay to give time for ADC chip to switch on. It needs 113ms */
+ msleep(200);
+ return 0;
+}
+
+static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * filename)
+{
+ int err;
+ int n, i;
+ int reg;
+ int value;
+ unsigned int write_post;
+ unsigned long flags;
+ const struct firmware *fw_entry;
+
+ if ((err = request_firmware(&fw_entry, filename, &emu->pci->dev)) != 0) {
+ snd_printk(KERN_ERR "firmware: %s not found. Err=%d\n",filename, err);
+ return err;
+ }
+ snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size);
+
+ /* The FPGA is a Xilinx Spartan IIE XC2S50E */
+ /* GPIO7 -> FPGA PGMN
+ * GPIO6 -> FPGA CCLK
+ * GPIO5 -> FPGA DIN
+ * FPGA CONFIG OFF -> FPGA PGMN
+ */
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(0x00, emu->port + A_IOCFG); /* Set PGMN low for 1uS. */
+ write_post = inl(emu->port + A_IOCFG);
+ udelay(100);
+ outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */
+ write_post = inl(emu->port + A_IOCFG);
+ udelay(100); /* Allow FPGA memory to clean */
+ for(n = 0; n < fw_entry->size; n++) {
+ value=fw_entry->data[n];
+ for(i = 0; i < 8; i++) {
+ reg = 0x80;
+ if (value & 0x1)
+ reg = reg | 0x20;
+ value = value >> 1;
+ outl(reg, emu->port + A_IOCFG);
+ write_post = inl(emu->port + A_IOCFG);
+ outl(reg | 0x40, emu->port + A_IOCFG);
+ write_post = inl(emu->port + A_IOCFG);
+ }
+ }
+ /* After programming, set GPIO bit 4 high again. */
+ outl(0x10, emu->port + A_IOCFG);
+ write_post = inl(emu->port + A_IOCFG);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+
+ release_firmware(fw_entry);
+ return 0;
+}
+
+static int emu1010_firmware_thread(void *data)
+{
+ struct snd_emu10k1 * emu = data;
+ int tmp,tmp2;
+ int reg;
+ int err;
+
+ for (;;) {
+ /* Delay to allow Audio Dock to settle */
+ msleep_interruptible(1000);
+ if (kthread_should_stop())
+ break;
+ snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); /* OPTIONS: Which cards are attached to the EMU */
+ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) {
+ /* Audio Dock attached */
+ /* Return to Audio Dock programming mode */
+ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
+ if (emu->card_capabilities->emu_model ==
+ EMU_MODEL_EMU1010) {
+ if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
+ continue;
+ }
+ } else if (emu->card_capabilities->emu_model ==
+ EMU_MODEL_EMU1010B) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ continue;
+ }
+ } else if (emu->card_capabilities->emu_model ==
+ EMU_MODEL_EMU1616) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ continue;
+ }
+ }
+
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 );
+ snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
+ snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
+ /* ID, should read & 0x7f = 0x55 when FPGA programmed. */
+ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
+ if ((reg & 0x1f) != 0x15) {
+ /* FPGA failed to be programmed */
+ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
+ continue;
+ }
+ snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
+ snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp );
+ snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 );
+ snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2);
+ /* Sync clocking between 1010 and Dock */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all. Default is muted after a firmware load */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ }
+ }
+ snd_printk(KERN_INFO "emu1010: firmware thread stopping\n");
+ return 0;
+}
+
+/*
+ * EMU-1010 - details found out from this driver, official MS Win drivers,
+ * testing the card:
+ *
+ * Audigy2 (aka Alice2):
+ * ---------------------
+ * * communication over PCI
+ * * conversion of 32-bit data coming over EMU32 links from HANA FPGA
+ * to 2 x 16-bit, using internal DSP instructions
+ * * slave mode, clock supplied by HANA
+ * * linked to HANA using:
+ * 32 x 32-bit serial EMU32 output channels
+ * 16 x EMU32 input channels
+ * (?) x I2S I/O channels (?)
+ *
+ * FPGA (aka HANA):
+ * ---------------
+ * * provides all (?) physical inputs and outputs of the card
+ * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.)
+ * * provides clock signal for the card and Alice2
+ * * two crystals - for 44.1kHz and 48kHz multiples
+ * * provides internal routing of signal sources to signal destinations
+ * * inputs/outputs to Alice2 - see above
+ *
+ * Current status of the driver:
+ * ----------------------------
+ * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz)
+ * * PCM device nb. 2:
+ * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops
+ * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops
+ */
+static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
+{
+ unsigned int i;
+ int tmp,tmp2;
+ int reg;
+ int err;
+ const char *filename = NULL;
+
+ snd_printk(KERN_INFO "emu1010: Special config.\n");
+ /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave,
+ * Lock Sound Memory Cache, Lock Tank Memory Cache,
+ * Mute all codecs.
+ */
+ outl(0x0005a00c, emu->port + HCFG);
+ /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave,
+ * Lock Tank Memory Cache,
+ * Mute all codecs.
+ */
+ outl(0x0005a004, emu->port + HCFG);
+ /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave,
+ * Mute all codecs.
+ */
+ outl(0x0005a000, emu->port + HCFG);
+ /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave,
+ * Mute all codecs.
+ */
+ outl(0x0005a000, emu->port + HCFG);
+
+ /* Disable 48Volt power to Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 );
+
+ /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */
+ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ snd_printdd("reg1=0x%x\n",reg);
+ if ((reg & 0x3f) == 0x15) {
+ /* FPGA netlist already present so clear it */
+ /* Return to programming mode */
+
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02 );
+ }
+ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ snd_printdd("reg2=0x%x\n",reg);
+ if ((reg & 0x3f) == 0x15) {
+ /* FPGA failed to return to programming mode */
+ snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n");
+ return -ENODEV;
+ }
+ snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg);
+ switch (emu->card_capabilities->emu_model) {
+ case EMU_MODEL_EMU1010:
+ filename = HANA_FILENAME;
+ break;
+ case EMU_MODEL_EMU1010B:
+ filename = EMU1010B_FILENAME;
+ break;
+ case EMU_MODEL_EMU1616:
+ filename = EMU1010_NOTEBOOK_FILENAME;
+ break;
+ case EMU_MODEL_EMU0404:
+ filename = EMU0404_FILENAME;
+ break;
+ default:
+ filename = NULL;
+ return -ENODEV;
+ break;
+ }
+ snd_printk(KERN_INFO "emu1010: filename %s testing\n", filename);
+ err = snd_emu1010_load_firmware(emu, filename);
+ if (err != 0) {
+ snd_printk(
+ KERN_INFO "emu1010: Loading Firmware file %s failed\n",
+ filename);
+ return err;
+ }
+
+ /* ID, should read & 0x7f = 0x55 when FPGA programmed. */
+ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ if ((reg & 0x3f) != 0x15) {
+ /* FPGA failed to be programmed */
+ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg);
+ return -ENODEV;
+ }
+
+ snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n");
+ snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp );
+ snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2 );
+ snd_printk("Hana ver:%d.%d\n",tmp ,tmp2);
+ /* Enable 48Volt power to Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON );
+
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
+ snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg);
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
+ snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg);
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp );
+ /* Optical -> ADAT I/O */
+ /* 0 : SPDIF
+ * 1 : ADAT
+ */
+ emu->emu1010.optical_in = 1; /* IN_ADAT */
+ emu->emu1010.optical_out = 1; /* IN_ADAT */
+ tmp = 0;
+ tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) |
+ (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0);
+ snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp );
+ snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp );
+ /* Set no attenuation on Audio Dock pads. */
+ snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 );
+ emu->emu1010.adc_pads = 0x00;
+ snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp );
+ /* Unmute Audio dock DACs, Headphone source DAC-4. */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 );
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 );
+ snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp );
+ /* DAC PADs. */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f );
+ emu->emu1010.dac_pads = 0x0f;
+ snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp );
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 );
+ snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp );
+ /* SPDIF Format. Set Consumer mode, 24bit, copy enable */
+ snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 );
+ /* MIDI routing */
+ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 );
+ /* Unknown. */
+ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c );
+ /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); // IRQ Enable: All on */
+ /* IRQ Enable: All off */
+ snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00 );
+
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
+ snd_printk(KERN_INFO "emu1010: Card options3=0x%x\n",reg);
+ /* Default WCLK set to 48kHz. */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00 );
+ /* Word Clock source, Internal 48kHz x1 */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K );
+ //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X );
+ /* Audio Dock LEDs. */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 );
+
+#if 0
+ /* For 96kHz */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_0, EMU_SRC_HAMOA_ADC_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_1, EMU_SRC_HAMOA_ADC_RIGHT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_4, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_5, EMU_SRC_HAMOA_ADC_RIGHT2);
+#endif
+#if 0
+ /* For 192kHz */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_0, EMU_SRC_HAMOA_ADC_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_1, EMU_SRC_HAMOA_ADC_RIGHT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_2, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_3, EMU_SRC_HAMOA_ADC_RIGHT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_4, EMU_SRC_HAMOA_ADC_LEFT3);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_5, EMU_SRC_HAMOA_ADC_RIGHT3);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_6, EMU_SRC_HAMOA_ADC_LEFT4);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_7, EMU_SRC_HAMOA_ADC_RIGHT4);
+#endif
+#if 1
+ /* For 48kHz */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_0, EMU_SRC_DOCK_MIC_A1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_1, EMU_SRC_DOCK_MIC_B1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_2, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_3, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_4, EMU_SRC_DOCK_ADC1_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_5, EMU_SRC_DOCK_ADC1_RIGHT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1);
+ /* Pavel Hofman - setting defaults for 8 more capture channels
+ * Defaults only, users will set their own values anyways, let's
+ * just copy/paste.
+ */
+
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1);
+#endif
+#if 0
+ /* Original */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_4, EMU_SRC_HANA_ADAT);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_5, EMU_SRC_HANA_ADAT + 1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_6, EMU_SRC_HANA_ADAT + 2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_7, EMU_SRC_HANA_ADAT + 3);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_8, EMU_SRC_HANA_ADAT + 4);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_9, EMU_SRC_HANA_ADAT + 5);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_A, EMU_SRC_HANA_ADAT + 6);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_B, EMU_SRC_HANA_ADAT + 7);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_MIC_A1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_MIC_B1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_E, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_F, EMU_SRC_HAMOA_ADC_LEFT2);
+#endif
+ for (i = 0;i < 0x20; i++ ) {
+ /* AudioDock Elink <- Silence */
+ snd_emu1010_fpga_link_dst_src_write(emu, 0x0100+i, EMU_SRC_SILENCE);
+ }
+ for (i = 0;i < 4; i++) {
+ /* Hana SPDIF Out <- Silence */
+ snd_emu1010_fpga_link_dst_src_write(emu, 0x0200+i, EMU_SRC_SILENCE);
+ }
+ for (i = 0;i < 7; i++) {
+ /* Hamoa DAC <- Silence */
+ snd_emu1010_fpga_link_dst_src_write(emu, 0x0300+i, EMU_SRC_SILENCE);
+ }
+ for (i = 0;i < 7; i++) {
+ /* Hana ADAT Out <- Silence */
+ snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HANA_ADAT + i, EMU_SRC_SILENCE);
+ }
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE_I2S0_LEFT, EMU_SRC_DOCK_ADC1_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE_I2S0_RIGHT, EMU_SRC_DOCK_ADC1_RIGHT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE_I2S1_LEFT, EMU_SRC_DOCK_ADC2_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE_I2S1_RIGHT, EMU_SRC_DOCK_ADC2_RIGHT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1);
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01 ); // Unmute all
+
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp );
+
+ /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave,
+ * Lock Sound Memory Cache, Lock Tank Memory Cache,
+ * Mute all codecs.
+ */
+ outl(0x0000a000, emu->port + HCFG);
+ /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave,
+ * Lock Sound Memory Cache, Lock Tank Memory Cache,
+ * Un-Mute all codecs.
+ */
+ outl(0x0000a001, emu->port + HCFG);
+
+ /* Initial boot complete. Now patches */
+
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp );
+ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */
+ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */
+ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */
+ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */
+ snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp );
+ snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */
+
+ /* Start Micro/Audio Dock firmware loader thread */
+ if (!emu->emu1010.firmware_thread) {
+ emu->emu1010.firmware_thread =
+ kthread_create(emu1010_firmware_thread, emu,
+ "emu1010_firmware");
+ wake_up_process(emu->emu1010.firmware_thread);
+ }
+
+#if 0
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32B + 2); /* ALICE2 bus 0xa2 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HAMOA_DAC_RIGHT1, EMU_SRC_ALICE_EMU32B + 3); /* ALICE2 bus 0xa3 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 2); /* ALICE2 bus 0xb2 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 3); /* ALICE2 bus 0xb3 */
+#endif
+ /* Default outputs */
+ if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) {
+ /* 1616(M) cardbus default outputs */
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC1_LEFT1, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[0] = 17;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC1_RIGHT1, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[1] = 18;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC2_LEFT1, EMU_SRC_ALICE_EMU32A + 2);
+ emu->emu1010.output_source[2] = 19;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC2_RIGHT1, EMU_SRC_ALICE_EMU32A + 3);
+ emu->emu1010.output_source[3] = 20;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC3_LEFT1, EMU_SRC_ALICE_EMU32A + 4);
+ emu->emu1010.output_source[4] = 21;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC3_RIGHT1, EMU_SRC_ALICE_EMU32A + 5);
+ emu->emu1010.output_source[5] = 22;
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_MANA_DAC_LEFT, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[16] = 17;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_MANA_DAC_RIGHT, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[17] = 18;
+ } else {
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC1_LEFT1, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[0] = 21;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC1_RIGHT1, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[1] = 22;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC2_LEFT1, EMU_SRC_ALICE_EMU32A + 2);
+ emu->emu1010.output_source[2] = 23;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC2_RIGHT1, EMU_SRC_ALICE_EMU32A + 3);
+ emu->emu1010.output_source[3] = 24;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC3_LEFT1, EMU_SRC_ALICE_EMU32A + 4);
+ emu->emu1010.output_source[4] = 25;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC3_RIGHT1, EMU_SRC_ALICE_EMU32A + 5);
+ emu->emu1010.output_source[5] = 26;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC4_LEFT1, EMU_SRC_ALICE_EMU32A + 6);
+ emu->emu1010.output_source[6] = 27;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_DAC4_RIGHT1, EMU_SRC_ALICE_EMU32A + 7);
+ emu->emu1010.output_source[7] = 28;
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_PHONES_LEFT1, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[8] = 21;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_PHONES_RIGHT1, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[9] = 22;
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[10] = 21;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_DOCK_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[11] = 22;
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_SPDIF_LEFT1, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[12] = 21;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_SPDIF_RIGHT1, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[13] = 22;
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[14] = 21;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HAMOA_DAC_RIGHT1, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[15] = 22;
+ /* ALICE2 bus 0xa0 */
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT, EMU_SRC_ALICE_EMU32A + 0);
+ emu->emu1010.output_source[16] = 21;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT + 1, EMU_SRC_ALICE_EMU32A + 1);
+ emu->emu1010.output_source[17] = 22;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT + 2, EMU_SRC_ALICE_EMU32A + 2);
+ emu->emu1010.output_source[18] = 23;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT + 3, EMU_SRC_ALICE_EMU32A + 3);
+ emu->emu1010.output_source[19] = 24;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT + 4, EMU_SRC_ALICE_EMU32A + 4);
+ emu->emu1010.output_source[20] = 25;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT + 5, EMU_SRC_ALICE_EMU32A + 5);
+ emu->emu1010.output_source[21] = 26;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT + 6, EMU_SRC_ALICE_EMU32A + 6);
+ emu->emu1010.output_source[22] = 27;
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_HANA_ADAT + 7, EMU_SRC_ALICE_EMU32A + 7);
+ emu->emu1010.output_source[23] = 28;
+ }
+ /* TEMP: Select SPDIF in/out */
+ //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */
+
+ /* TEMP: Select 48kHz SPDIF out */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */
+ /* Word Clock source, Internal 48kHz x1 */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K );
+ //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X );
+ emu->emu1010.internal_clock = 1; /* 48000 */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);/* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */
+ //snd_emu1010_fpga_write(emu, 0x7, 0x0); /* Mute all */
+ //snd_emu1010_fpga_write(emu, 0x7, 0x1); /* Unmute all */
+ //snd_emu1010_fpga_write(emu, 0xe, 0x12); /* Set LEDs on Audio Dock */
+
+ return 0;
+}
+/*
+ * Create the EMU10K1 instance
+ */
+
+#ifdef CONFIG_PM
+static int alloc_pm_buffer(struct snd_emu10k1 *emu);
+static void free_pm_buffer(struct snd_emu10k1 *emu);
+#endif
+
+static int snd_emu10k1_free(struct snd_emu10k1 *emu)
+{
+ if (emu->port) { /* avoid access to already used hardware */
+ snd_emu10k1_fx8010_tram_setup(emu, 0);
+ snd_emu10k1_done(emu);
+ snd_emu10k1_free_efx(emu);
+ }
+ if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) {
+ /* Disable 48Volt power to Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 );
+ }
+ if (emu->emu1010.firmware_thread)
+ kthread_stop(emu->emu1010.firmware_thread);
+ if (emu->irq >= 0)
+ free_irq(emu->irq, emu);
+ /* remove reserved page */
+ if (emu->reserved_page) {
+ snd_emu10k1_synth_free(emu,
+ (struct snd_util_memblk *)emu->reserved_page);
+ emu->reserved_page = NULL;
+ }
+ if (emu->memhdr)
+ snd_util_memhdr_free(emu->memhdr);
+ if (emu->silent_page.area)
+ snd_dma_free_pages(&emu->silent_page);
+ if (emu->ptb_pages.area)
+ snd_dma_free_pages(&emu->ptb_pages);
+ vfree(emu->page_ptr_table);
+ vfree(emu->page_addr_table);
+#ifdef CONFIG_PM
+ free_pm_buffer(emu);
+#endif
+ if (emu->port)
+ pci_release_regions(emu->pci);
+ if (emu->card_capabilities->ca0151_chip) /* P16V */
+ snd_p16v_free(emu);
+ pci_disable_device(emu->pci);
+ kfree(emu);
+ return 0;
+}
+
+static int snd_emu10k1_dev_free(struct snd_device *device)
+{
+ struct snd_emu10k1 *emu = device->device_data;
+ return snd_emu10k1_free(emu);
+}
+
+static struct snd_emu_chip_details emu_chip_details[] = {
+ /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/
+ /* Tested by James@superbug.co.uk 3rd July 2005 */
+ /* DSP: CA0108-IAT
+ * DAC: CS4382-KQ
+ * ADC: Philips 1361T
+ * AC97: STAC9750
+ * CA0151: None
+ */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102,
+ .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .spk71 = 1,
+ .ac97_chip = 1} ,
+ /* Audigy4 (Not PRO) SB0610 */
+ /* Tested by James@superbug.co.uk 4th April 2006 */
+ /* A_IOCFG bits
+ * Output
+ * 0: ?
+ * 1: ?
+ * 2: ?
+ * 3: 0 - Digital Out, 1 - Line in
+ * 4: ?
+ * 5: ?
+ * 6: ?
+ * 7: ?
+ * Input
+ * 8: ?
+ * 9: ?
+ * A: Green jack sense (Front)
+ * B: ?
+ * C: Black jack sense (Rear/Side Right)
+ * D: Yellow jack sense (Center/LFE/Side Left)
+ * E: ?
+ * F: ?
+ *
+ * Digital Out/Line in switch using A_IOCFG bit 3 (0x08)
+ * 0 - Digital Out
+ * 1 - Line in
+ */
+ /* Mic input not tested.
+ * Analog CD input not tested
+ * Digital Out not tested.
+ * Line in working.
+ * Audio output 5.1 working. Side outputs not working.
+ */
+ /* DSP: CA10300-IAT LF
+ * DAC: Cirrus Logic CS4382-KQZ
+ * ADC: Philips 1361T
+ * AC97: Sigmatel STAC9750
+ * CA0151: None
+ */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102,
+ .driver = "Audigy2", .name = "Audigy 4 [SB0610]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .spk71 = 1,
+ .adc_1361t = 1, /* 24 bit capture instead of 16bit */
+ .ac97_chip = 1} ,
+ /* Audigy 2 ZS Notebook Cardbus card.*/
+ /* Tested by James@superbug.co.uk 6th November 2006 */
+ /* Audio output 7.1/Headphones working.
+ * Digital output working. (AC3 not checked, only PCM)
+ * Audio Mic/Line inputs working.
+ * Digital input not tested.
+ */
+ /* DSP: Tina2
+ * DAC: Wolfson WM8768/WM8568
+ * ADC: Wolfson WM8775
+ * AC97: None
+ * CA0151: None
+ */
+ /* Tested by James@superbug.co.uk 4th April 2006 */
+ /* A_IOCFG bits
+ * Output
+ * 0: Not Used
+ * 1: 0 = Mute all the 7.1 channel out. 1 = unmute.
+ * 2: Analog input 0 = line in, 1 = mic in
+ * 3: Not Used
+ * 4: Digital output 0 = off, 1 = on.
+ * 5: Not Used
+ * 6: Not Used
+ * 7: Not Used
+ * Input
+ * All bits 1 (0x3fxx) means nothing plugged in.
+ * 8-9: 0 = Line in/Mic, 2 = Optical in, 3 = Nothing.
+ * A-B: 0 = Headphones, 2 = Optical out, 3 = Nothing.
+ * C-D: 2 = Front/Rear/etc, 3 = nothing.
+ * E-F: Always 0
+ *
+ */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102,
+ .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .ca_cardbus_chip = 1,
+ .spi_dac = 1,
+ .i2c_adc = 1,
+ .spk71 = 1} ,
+ /* Tested by James@superbug.co.uk 4th Nov 2007. */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102,
+ .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]",
+ .id = "EMU1010",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .ca_cardbus_chip = 1,
+ .spk71 = 1 ,
+ .emu_model = EMU_MODEL_EMU1616},
+ /* Tested by James@superbug.co.uk 4th Nov 2007. */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102,
+ .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]",
+ .id = "EMU1010",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .spk71 = 1,
+ .emu_model = EMU_MODEL_EMU1010B},
+ /* Tested by James@superbug.co.uk 8th July 2005. */
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102,
+ .driver = "Audigy2", .name = "E-mu 1010 [4001]",
+ .id = "EMU1010",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .spk71 = 1,
+ .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */
+ /* EMU0404b */
+ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102,
+ .driver = "Audigy2", .name = "E-mu 0404b [4002]",
+ .id = "EMU0404",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .spk71 = 1,
+ .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
+ /* Tested by James@superbug.co.uk 20-3-2007. */
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102,
+ .driver = "Audigy2", .name = "E-mu 0404 [4002]",
+ .id = "EMU0404",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .spk71 = 1,
+ .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
+ /* Audigy4 (Not PRO) SB0610 */
+ {.vendor = 0x1102, .device = 0x0008,
+ .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0108_chip = 1,
+ .ac97_chip = 1} ,
+ /* Tested by James@superbug.co.uk 3rd July 2005 */
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102,
+ .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1,
+ .ac97_chip = 1} ,
+ /* Tested by shane-alsa@cm.nu 5th Nov 2005 */
+ /* The 0x20061102 does have SB0350 written on it
+ * Just like 0x20021102
+ */
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20061102,
+ .driver = "Audigy2", .name = "Audigy 2 [SB0350b]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102,
+ .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102,
+ .driver = "Audigy2", .name = "Audigy 2 ZS [2001]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
+ .ac97_chip = 1} ,
+ /* Audigy 2 */
+ /* Tested by James@superbug.co.uk 3rd July 2005 */
+ /* DSP: CA0102-IAT
+ * DAC: CS4382-KQ
+ * ADC: Philips 1361T
+ * AC97: STAC9721
+ * CA0151: Yes
+ */
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10071102,
+ .driver = "Audigy2", .name = "Audigy 2 [SB0240]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1,
+ .adc_1361t = 1, /* 24 bit capture instead of 16bit */
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102,
+ .driver = "Audigy2", .name = "Audigy 2 EX [1005]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1} ,
+ /* Dell OEM/Creative Labs Audigy 2 ZS */
+ /* See ALSA bug#1365 */
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10031102,
+ .driver = "Audigy2", .name = "Audigy 2 ZS [SB0353]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
+ .driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spk71 = 1,
+ .spdif_bug = 1,
+ .invert_shared_spdif = 1, /* digital/analog switch swapped */
+ .adc_1361t = 1, /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .revision = 0x04,
+ .driver = "Audigy2", .name = "Audigy 2 [Unknown]",
+ .id = "Audigy2",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ca0151_chip = 1,
+ .spdif_bug = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102,
+ .driver = "Audigy", .name = "Audigy 1 [SB0090]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102,
+ .driver = "Audigy", .name = "Audigy 1 ES [SB0160]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .spdif_bug = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102,
+ .driver = "Audigy", .name = "Audigy 1 [SB0090]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004,
+ .driver = "Audigy", .name = "Audigy 1 [Unknown]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102,
+ .driver = "EMU10K1", .name = "SBLive! [SB0105]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102,
+ .driver = "EMU10K1", .name = "SBLive! Value [SB0103]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102,
+ .driver = "EMU10K1", .name = "SBLive! Value [SB0101]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ /* Tested by ALSA bug#1680 26th December 2005 */
+ /* note: It really has SB0220 written on the card. */
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80661102,
+ .driver = "EMU10K1", .name = "SB Live 5.1 Dell OEM [SB0220]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ /* Tested by Thomas Zehetbauer 27th Aug 2005 */
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80651102,
+ .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102,
+ .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102,
+ .driver = "EMU10K1", .name = "SB Live 5.1",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ /* Tested by alsa bugtrack user "hus" bug #1297 12th Aug 2005 */
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102,
+ .driver = "EMU10K1", .name = "SBLive 5.1 [SB0060]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 2, /* ac97 is optional; both SBLive 5.1 and platinum
+ * share the same IDs!
+ */
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4850]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102,
+ .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4871]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80311102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4831]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4870]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ /* Tested by James@superbug.co.uk 3rd July 2005 */
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4832]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4830]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102,
+ .driver = "EMU10K1", .name = "SB PCI512 [CT4790]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4780]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102,
+ .driver = "EMU10K1", .name = "E-mu APS [4001]",
+ .id = "APS",
+ .emu10k1_chip = 1,
+ .ecard = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102,
+ .driver = "EMU10K1", .name = "SBLive! [CT4620]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4670]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002,
+ .driver = "EMU10K1", .name = "SB Live [Unknown]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ { } /* terminator */
+};
+
+int __devinit snd_emu10k1_create(struct snd_card *card,
+ struct pci_dev * pci,
+ unsigned short extin_mask,
+ unsigned short extout_mask,
+ long max_cache_bytes,
+ int enable_ir,
+ uint subsystem,
+ struct snd_emu10k1 ** remu)
+{
+ struct snd_emu10k1 *emu;
+ int idx, err;
+ int is_audigy;
+ unsigned int silent_page;
+ const struct snd_emu_chip_details *c;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_emu10k1_dev_free,
+ };
+
+ *remu = NULL;
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ emu = kzalloc(sizeof(*emu), GFP_KERNEL);
+ if (emu == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ emu->card = card;
+ spin_lock_init(&emu->reg_lock);
+ spin_lock_init(&emu->emu_lock);
+ spin_lock_init(&emu->spi_lock);
+ spin_lock_init(&emu->i2c_lock);
+ spin_lock_init(&emu->voice_lock);
+ spin_lock_init(&emu->synth_lock);
+ spin_lock_init(&emu->memblk_lock);
+ mutex_init(&emu->fx8010.lock);
+ INIT_LIST_HEAD(&emu->mapped_link_head);
+ INIT_LIST_HEAD(&emu->mapped_order_link_head);
+ emu->pci = pci;
+ emu->irq = -1;
+ emu->synth = NULL;
+ emu->get_synth_voice = NULL;
+ /* read revision & serial */
+ emu->revision = pci->revision;
+ pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial);
+ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &emu->model);
+ snd_printdd("vendor=0x%x, device=0x%x, subsystem_vendor_id=0x%x, subsystem_id=0x%x\n",pci->vendor, pci->device, emu->serial, emu->model);
+
+ for (c = emu_chip_details; c->vendor; c++) {
+ if (c->vendor == pci->vendor && c->device == pci->device) {
+ if (subsystem) {
+ if (c->subsystem && (c->subsystem == subsystem) ) {
+ break;
+ } else continue;
+ } else {
+ if (c->subsystem && (c->subsystem != emu->serial) )
+ continue;
+ if (c->revision && c->revision != emu->revision)
+ continue;
+ }
+ break;
+ }
+ }
+ if (c->vendor == 0) {
+ snd_printk(KERN_ERR "emu10k1: Card not recognised\n");
+ kfree(emu);
+ pci_disable_device(pci);
+ return -ENOENT;
+ }
+ emu->card_capabilities = c;
+ if (c->subsystem && !subsystem)
+ snd_printdd("Sound card name=%s\n", c->name);
+ else if (subsystem)
+ snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n",
+ c->name, pci->vendor, pci->device, emu->serial, c->subsystem);
+ else
+ snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n",
+ c->name, pci->vendor, pci->device, emu->serial);
+
+ if (!*card->id && c->id) {
+ int i, n = 0;
+ strlcpy(card->id, c->id, sizeof(card->id));
+ for (;;) {
+ for (i = 0; i < snd_ecards_limit; i++) {
+ if (snd_cards[i] && !strcmp(snd_cards[i]->id, card->id))
+ break;
+ }
+ if (i >= snd_ecards_limit)
+ break;
+ n++;
+ if (n >= SNDRV_CARDS)
+ break;
+ snprintf(card->id, sizeof(card->id), "%s_%d", c->id, n);
+ }
+ }
+
+ is_audigy = emu->audigy = c->emu10k2_chip;
+
+ /* set the DMA transfer mask */
+ emu->dma_mask = is_audigy ? AUDIGY_DMA_MASK : EMU10K1_DMA_MASK;
+ if (pci_set_dma_mask(pci, emu->dma_mask) < 0 ||
+ pci_set_consistent_dma_mask(pci, emu->dma_mask) < 0) {
+ snd_printk(KERN_ERR "architecture does not support PCI busmaster DMA with mask 0x%lx\n", emu->dma_mask);
+ kfree(emu);
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+ if (is_audigy)
+ emu->gpr_base = A_FXGPREGBASE;
+ else
+ emu->gpr_base = FXGPREGBASE;
+
+ if ((err = pci_request_regions(pci, "EMU10K1")) < 0) {
+ kfree(emu);
+ pci_disable_device(pci);
+ return err;
+ }
+ emu->port = pci_resource_start(pci, 0);
+
+ emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT;
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ 32 * 1024, &emu->ptb_pages) < 0) {
+ err = -ENOMEM;
+ goto error;
+ }
+
+ emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *));
+ emu->page_addr_table = vmalloc(emu->max_cache_pages *
+ sizeof(unsigned long));
+ if (emu->page_ptr_table == NULL || emu->page_addr_table == NULL) {
+ err = -ENOMEM;
+ goto error;
+ }
+
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ EMUPAGESIZE, &emu->silent_page) < 0) {
+ err = -ENOMEM;
+ goto error;
+ }
+ emu->memhdr = snd_util_memhdr_new(emu->max_cache_pages * PAGE_SIZE);
+ if (emu->memhdr == NULL) {
+ err = -ENOMEM;
+ goto error;
+ }
+ emu->memhdr->block_extra_size = sizeof(struct snd_emu10k1_memblk) -
+ sizeof(struct snd_util_memblk);
+
+ pci_set_master(pci);
+
+ emu->fx8010.fxbus_mask = 0x303f;
+ if (extin_mask == 0)
+ extin_mask = 0x3fcf;
+ if (extout_mask == 0)
+ extout_mask = 0x7fff;
+ emu->fx8010.extin_mask = extin_mask;
+ emu->fx8010.extout_mask = extout_mask;
+ emu->enable_ir = enable_ir;
+
+ if (emu->card_capabilities->ca_cardbus_chip) {
+ if ((err = snd_emu10k1_cardbus_init(emu)) < 0)
+ goto error;
+ }
+ if (emu->card_capabilities->ecard) {
+ if ((err = snd_emu10k1_ecard_init(emu)) < 0)
+ goto error;
+ } else if (emu->card_capabilities->emu_model) {
+ if ((err = snd_emu10k1_emu1010_init(emu)) < 0) {
+ snd_emu10k1_free(emu);
+ return err;
+ }
+ } else {
+ /* 5.1: Enable the additional AC97 Slots. If the emu10k1 version
+ does not support this, it shouldn't do any harm */
+ snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE);
+ }
+
+ /* initialize TRAM setup */
+ emu->fx8010.itram_size = (16 * 1024)/2;
+ emu->fx8010.etram_pages.area = NULL;
+ emu->fx8010.etram_pages.bytes = 0;
+
+ /* irq handler must be registered after I/O ports are activated */
+ if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
+ "EMU10K1", emu)) {
+ err = -EBUSY;
+ goto error;
+ }
+ emu->irq = pci->irq;
+
+ /*
+ * Init to 0x02109204 :
+ * Clock accuracy = 0 (1000ppm)
+ * Sample Rate = 2 (48kHz)
+ * Audio Channel = 1 (Left of 2)
+ * Source Number = 0 (Unspecified)
+ * Generation Status = 1 (Original for Cat Code 12)
+ * Cat Code = 12 (Digital Signal Mixer)
+ * Mode = 0 (Mode 0)
+ * Emphasis = 0 (None)
+ * CP = 1 (Copyright unasserted)
+ * AN = 0 (Audio data)
+ * P = 0 (Consumer)
+ */
+ emu->spdif_bits[0] = emu->spdif_bits[1] =
+ emu->spdif_bits[2] = SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT;
+
+ emu->reserved_page = (struct snd_emu10k1_memblk *)
+ snd_emu10k1_synth_alloc(emu, 4096);
+ if (emu->reserved_page)
+ emu->reserved_page->map_locked = 1;
+
+ /* Clear silent pages and set up pointers */
+ memset(emu->silent_page.area, 0, PAGE_SIZE);
+ silent_page = emu->silent_page.addr << 1;
+ for (idx = 0; idx < MAXPAGES; idx++)
+ ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx);
+
+ /* set up voice indices */
+ for (idx = 0; idx < NUM_G; idx++) {
+ emu->voices[idx].emu = emu;
+ emu->voices[idx].number = idx;
+ }
+
+ if ((err = snd_emu10k1_init(emu, enable_ir, 0)) < 0)
+ goto error;
+#ifdef CONFIG_PM
+ if ((err = alloc_pm_buffer(emu)) < 0)
+ goto error;
+#endif
+
+ /* Initialize the effect engine */
+ if ((err = snd_emu10k1_init_efx(emu)) < 0)
+ goto error;
+ snd_emu10k1_audio_enable(emu);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops)) < 0)
+ goto error;
+
+#ifdef CONFIG_PROC_FS
+ snd_emu10k1_proc_init(emu);
+#endif
+
+ snd_card_set_dev(card, &pci->dev);
+ *remu = emu;
+ return 0;
+
+ error:
+ snd_emu10k1_free(emu);
+ return err;
+}
+
+#ifdef CONFIG_PM
+static unsigned char saved_regs[] = {
+ CPF, PTRX, CVCF, VTFT, Z1, Z2, PSST, DSL, CCCA, CCR, CLP,
+ FXRT, MAPA, MAPB, ENVVOL, ATKHLDV, DCYSUSV, LFOVAL1, ENVVAL,
+ ATKHLDM, DCYSUSM, LFOVAL2, IP, IFATN, PEFE, FMMOD, TREMFRQ, FM2FRQ2,
+ TEMPENV, ADCCR, FXWC, MICBA, ADCBA, FXBA,
+ MICBS, ADCBS, FXBS, CDCS, GPSCS, SPCS0, SPCS1, SPCS2,
+ SPBYPASS, AC97SLOT, CDSRCS, GPSRCS, ZVSRCS, MICIDX, ADCIDX, FXIDX,
+ 0xff /* end */
+};
+static unsigned char saved_regs_audigy[] = {
+ A_ADCIDX, A_MICIDX, A_FXWC1, A_FXWC2, A_SAMPLE_RATE,
+ A_FXRT2, A_SENDAMOUNTS, A_FXRT1,
+ 0xff /* end */
+};
+
+static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu)
+{
+ int size;
+
+ size = ARRAY_SIZE(saved_regs);
+ if (emu->audigy)
+ size += ARRAY_SIZE(saved_regs_audigy);
+ emu->saved_ptr = vmalloc(4 * NUM_G * size);
+ if (! emu->saved_ptr)
+ return -ENOMEM;
+ if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0)
+ return -ENOMEM;
+ if (emu->card_capabilities->ca0151_chip &&
+ snd_p16v_alloc_pm_buffer(emu) < 0)
+ return -ENOMEM;
+ return 0;
+}
+
+static void free_pm_buffer(struct snd_emu10k1 *emu)
+{
+ vfree(emu->saved_ptr);
+ snd_emu10k1_efx_free_pm_buffer(emu);
+ if (emu->card_capabilities->ca0151_chip)
+ snd_p16v_free_pm_buffer(emu);
+}
+
+void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu)
+{
+ int i;
+ unsigned char *reg;
+ unsigned int *val;
+
+ val = emu->saved_ptr;
+ for (reg = saved_regs; *reg != 0xff; reg++)
+ for (i = 0; i < NUM_G; i++, val++)
+ *val = snd_emu10k1_ptr_read(emu, *reg, i);
+ if (emu->audigy) {
+ for (reg = saved_regs_audigy; *reg != 0xff; reg++)
+ for (i = 0; i < NUM_G; i++, val++)
+ *val = snd_emu10k1_ptr_read(emu, *reg, i);
+ }
+ if (emu->audigy)
+ emu->saved_a_iocfg = inl(emu->port + A_IOCFG);
+ emu->saved_hcfg = inl(emu->port + HCFG);
+}
+
+void snd_emu10k1_resume_init(struct snd_emu10k1 *emu)
+{
+ if (emu->card_capabilities->ca_cardbus_chip)
+ snd_emu10k1_cardbus_init(emu);
+ if (emu->card_capabilities->ecard)
+ snd_emu10k1_ecard_init(emu);
+ else if (emu->card_capabilities->emu_model)
+ snd_emu10k1_emu1010_init(emu);
+ else
+ snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE);
+ snd_emu10k1_init(emu, emu->enable_ir, 1);
+}
+
+void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu)
+{
+ int i;
+ unsigned char *reg;
+ unsigned int *val;
+
+ snd_emu10k1_audio_enable(emu);
+
+ /* resore for spdif */
+ if (emu->audigy)
+ outl(emu->saved_a_iocfg, emu->port + A_IOCFG);
+ outl(emu->saved_hcfg, emu->port + HCFG);
+
+ val = emu->saved_ptr;
+ for (reg = saved_regs; *reg != 0xff; reg++)
+ for (i = 0; i < NUM_G; i++, val++)
+ snd_emu10k1_ptr_write(emu, *reg, i, *val);
+ if (emu->audigy) {
+ for (reg = saved_regs_audigy; *reg != 0xff; reg++)
+ for (i = 0; i < NUM_G; i++, val++)
+ snd_emu10k1_ptr_write(emu, *reg, i, *val);
+ }
+}
+#endif
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
new file mode 100644
index 0000000..e10f027
--- /dev/null
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -0,0 +1,229 @@
+/*
+ * Patch transfer callback for Emu10k1
+ *
+ * Copyright (C) 2000 Takashi iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+/*
+ * All the code for loading in a patch. There is very little that is
+ * chip specific here. Just the actual writing to the board.
+ */
+
+#include "emu10k1_synth_local.h"
+
+/*
+ */
+#define BLANK_LOOP_START 4
+#define BLANK_LOOP_END 8
+#define BLANK_LOOP_SIZE 12
+#define BLANK_HEAD_SIZE 32
+
+/*
+ * allocate a sample block and copy data from userspace
+ */
+int
+snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr,
+ const void __user *data, long count)
+{
+ int offset;
+ int truesize, size, loopsize, blocksize;
+ int loopend, sampleend;
+ unsigned int start_addr;
+ struct snd_emu10k1 *emu;
+
+ emu = rec->hw;
+ if (snd_BUG_ON(!sp || !hdr))
+ return -EINVAL;
+
+ if (sp->v.size == 0) {
+ snd_printd("emu: rom font for sample %d\n", sp->v.sample);
+ return 0;
+ }
+
+ /* recalculate address offset */
+ sp->v.end -= sp->v.start;
+ sp->v.loopstart -= sp->v.start;
+ sp->v.loopend -= sp->v.start;
+ sp->v.start = 0;
+
+ /* some samples have invalid data. the addresses are corrected in voice info */
+ sampleend = sp->v.end;
+ if (sampleend > sp->v.size)
+ sampleend = sp->v.size;
+ loopend = sp->v.loopend;
+ if (loopend > sampleend)
+ loopend = sampleend;
+
+ /* be sure loop points start < end */
+ if (sp->v.loopstart >= sp->v.loopend) {
+ int tmp = sp->v.loopstart;
+ sp->v.loopstart = sp->v.loopend;
+ sp->v.loopend = tmp;
+ }
+
+ /* compute true data size to be loaded */
+ truesize = sp->v.size + BLANK_HEAD_SIZE;
+ loopsize = 0;
+#if 0 /* not supported */
+ if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP))
+ loopsize = sp->v.loopend - sp->v.loopstart;
+ truesize += loopsize;
+#endif
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK)
+ truesize += BLANK_LOOP_SIZE;
+
+ /* try to allocate a memory block */
+ blocksize = truesize;
+ if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
+ blocksize *= 2;
+ sp->block = snd_emu10k1_synth_alloc(emu, blocksize);
+ if (sp->block == NULL) {
+ snd_printd("emu10k1: synth malloc failed (size=%d)\n", blocksize);
+ /* not ENOMEM (for compatibility with OSS) */
+ return -ENOSPC;
+ }
+ /* set the total size */
+ sp->v.truesize = blocksize;
+
+ /* write blank samples at head */
+ offset = 0;
+ size = BLANK_HEAD_SIZE;
+ if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
+ size *= 2;
+ if (offset + size > blocksize)
+ return -EINVAL;
+ snd_emu10k1_synth_bzero(emu, sp->block, offset, size);
+ offset += size;
+
+ /* copy start->loopend */
+ size = loopend;
+ if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
+ size *= 2;
+ if (offset + size > blocksize)
+ return -EINVAL;
+ if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) {
+ snd_emu10k1_synth_free(emu, sp->block);
+ sp->block = NULL;
+ return -EFAULT;
+ }
+ offset += size;
+ data += size;
+
+#if 0 /* not suppported yet */
+ /* handle reverse (or bidirectional) loop */
+ if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) {
+ /* copy loop in reverse */
+ if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) {
+ int woffset;
+ unsigned short *wblock = (unsigned short*)block;
+ woffset = offset / 2;
+ if (offset + loopsize * 2 > blocksize)
+ return -EINVAL;
+ for (i = 0; i < loopsize; i++)
+ wblock[woffset + i] = wblock[woffset - i -1];
+ offset += loopsize * 2;
+ } else {
+ if (offset + loopsize > blocksize)
+ return -EINVAL;
+ for (i = 0; i < loopsize; i++)
+ block[offset + i] = block[offset - i -1];
+ offset += loopsize;
+ }
+
+ /* modify loop pointers */
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_BIDIR_LOOP) {
+ sp->v.loopend += loopsize;
+ } else {
+ sp->v.loopstart += loopsize;
+ sp->v.loopend += loopsize;
+ }
+ /* add sample pointer */
+ sp->v.end += loopsize;
+ }
+#endif
+
+ /* loopend -> sample end */
+ size = sp->v.size - loopend;
+ if (size < 0)
+ return -EINVAL;
+ if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
+ size *= 2;
+ if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) {
+ snd_emu10k1_synth_free(emu, sp->block);
+ sp->block = NULL;
+ return -EFAULT;
+ }
+ offset += size;
+
+ /* clear rest of samples (if any) */
+ if (offset < blocksize)
+ snd_emu10k1_synth_bzero(emu, sp->block, offset, blocksize - offset);
+
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) {
+ /* if no blank loop is attached in the sample, add it */
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) {
+ sp->v.loopstart = sp->v.end + BLANK_LOOP_START;
+ sp->v.loopend = sp->v.end + BLANK_LOOP_END;
+ }
+ }
+
+#if 0 /* not supported yet */
+ if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) {
+ /* unsigned -> signed */
+ if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) {
+ unsigned short *wblock = (unsigned short*)block;
+ for (i = 0; i < truesize; i++)
+ wblock[i] ^= 0x8000;
+ } else {
+ for (i = 0; i < truesize; i++)
+ block[i] ^= 0x80;
+ }
+ }
+#endif
+
+ /* recalculate offset */
+ start_addr = BLANK_HEAD_SIZE * 2;
+ if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS))
+ start_addr >>= 1;
+ sp->v.start += start_addr;
+ sp->v.end += start_addr;
+ sp->v.loopstart += start_addr;
+ sp->v.loopend += start_addr;
+
+ return 0;
+}
+
+/*
+ * free a sample block
+ */
+int
+snd_emu10k1_sample_free(struct snd_emux *rec, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr)
+{
+ struct snd_emu10k1 *emu;
+
+ emu = rec->hw;
+ if (snd_BUG_ON(!sp || !hdr))
+ return -EINVAL;
+
+ if (sp->block) {
+ snd_emu10k1_synth_free(emu, sp->block);
+ sp->block = NULL;
+ }
+ return 0;
+}
+
diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c
new file mode 100644
index 0000000..ad7b714
--- /dev/null
+++ b/sound/pci/emu10k1/emu10k1_synth.c
@@ -0,0 +1,123 @@
+/*
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Routines for control of EMU10K1 WaveTable synth
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "emu10k1_synth_local.h"
+#include <linux/init.h>
+
+MODULE_AUTHOR("Takashi Iwai");
+MODULE_DESCRIPTION("Routines for control of EMU10K1 WaveTable synth");
+MODULE_LICENSE("GPL");
+
+/*
+ * create a new hardware dependent device for Emu10k1
+ */
+static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev)
+{
+ struct snd_emux *emux;
+ struct snd_emu10k1 *hw;
+ struct snd_emu10k1_synth_arg *arg;
+ unsigned long flags;
+
+ arg = SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ if (arg == NULL)
+ return -EINVAL;
+
+ if (arg->seq_ports <= 0)
+ return 0; /* nothing */
+ if (arg->max_voices < 1)
+ arg->max_voices = 1;
+ else if (arg->max_voices > 64)
+ arg->max_voices = 64;
+
+ if (snd_emux_new(&emux) < 0)
+ return -ENOMEM;
+
+ snd_emu10k1_ops_setup(emux);
+ hw = arg->hwptr;
+ emux->hw = hw;
+ emux->max_voices = arg->max_voices;
+ emux->num_ports = arg->seq_ports;
+ emux->pitch_shift = -501;
+ emux->memhdr = hw->memhdr;
+ /* maximum two ports */
+ emux->midi_ports = arg->seq_ports < 2 ? arg->seq_ports : 2;
+ /* audigy has two external midis */
+ emux->midi_devidx = hw->audigy ? 2 : 1;
+ emux->linear_panning = 0;
+ emux->hwdep_idx = 2; /* FIXED */
+
+ if (snd_emux_register(emux, dev->card, arg->index, "Emu10k1") < 0) {
+ snd_emux_free(emux);
+ return -ENOMEM;
+ }
+
+ spin_lock_irqsave(&hw->voice_lock, flags);
+ hw->synth = emux;
+ hw->get_synth_voice = snd_emu10k1_synth_get_voice;
+ spin_unlock_irqrestore(&hw->voice_lock, flags);
+
+ dev->driver_data = emux;
+
+ return 0;
+}
+
+static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev)
+{
+ struct snd_emux *emux;
+ struct snd_emu10k1 *hw;
+ unsigned long flags;
+
+ if (dev->driver_data == NULL)
+ return 0; /* not registered actually */
+
+ emux = dev->driver_data;
+
+ hw = emux->hw;
+ spin_lock_irqsave(&hw->voice_lock, flags);
+ hw->synth = NULL;
+ hw->get_synth_voice = NULL;
+ spin_unlock_irqrestore(&hw->voice_lock, flags);
+
+ snd_emux_free(emux);
+ return 0;
+}
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_emu10k1_synth_init(void)
+{
+
+ static struct snd_seq_dev_ops ops = {
+ snd_emu10k1_synth_new_device,
+ snd_emu10k1_synth_delete_device,
+ };
+ return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, &ops,
+ sizeof(struct snd_emu10k1_synth_arg));
+}
+
+static void __exit alsa_emu10k1_synth_exit(void)
+{
+ snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH);
+}
+
+module_init(alsa_emu10k1_synth_init)
+module_exit(alsa_emu10k1_synth_exit)
diff --git a/sound/pci/emu10k1/emu10k1_synth_local.h b/sound/pci/emu10k1/emu10k1_synth_local.h
new file mode 100644
index 0000000..25f328f
--- /dev/null
+++ b/sound/pci/emu10k1/emu10k1_synth_local.h
@@ -0,0 +1,42 @@
+#ifndef __EMU10K1_SYNTH_LOCAL_H
+#define __EMU10K1_SYNTH_LOCAL_H
+/*
+ * Local defininitons for Emu10k1 wavetable
+ *
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/emu10k1_synth.h>
+
+/* emu10k1_patch.c */
+int snd_emu10k1_sample_new(struct snd_emux *private_data,
+ struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr,
+ const void __user *_data, long count);
+int snd_emu10k1_sample_free(struct snd_emux *private_data,
+ struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr);
+int snd_emu10k1_memhdr_init(struct snd_emux *emu);
+
+/* emu10k1_callback.c */
+void snd_emu10k1_ops_setup(struct snd_emux *emu);
+int snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw);
+
+
+#endif /* __EMU10K1_SYNTH_LOCAL_H */
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
new file mode 100644
index 0000000..5ff4dbb
--- /dev/null
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -0,0 +1,1637 @@
+/*
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ * Driver EMU10K1X chips
+ *
+ * Parts of this code were adapted from audigyls.c driver which is
+ * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ *
+ * Chips (SB0200 model):
+ * - EMU10K1X-DBQ
+ * - STAC 9708T
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/rawmidi.h>
+
+MODULE_AUTHOR("Francisco Moraes <fmoraes@nc.rr.com>");
+MODULE_DESCRIPTION("EMU10K1X");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Dell Creative Labs,SB Live!}");
+
+// module parameters (see "Module Parameters")
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the EMU10K1X soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the EMU10K1X soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard.");
+
+
+// some definitions were borrowed from emu10k1 driver as they seem to be the same
+/************************************************************************************************/
+/* PCI function 0 registers, address = <val> + PCIBASE0 */
+/************************************************************************************************/
+
+#define PTR 0x00 /* Indexed register set pointer register */
+ /* NOTE: The CHANNELNUM and ADDRESS words can */
+ /* be modified independently of each other. */
+
+#define DATA 0x04 /* Indexed register set data register */
+
+#define IPR 0x08 /* Global interrupt pending register */
+ /* Clear pending interrupts by writing a 1 to */
+ /* the relevant bits and zero to the other bits */
+#define IPR_MIDITRANSBUFEMPTY 0x00000001 /* MIDI UART transmit buffer empty */
+#define IPR_MIDIRECVBUFEMPTY 0x00000002 /* MIDI UART receive buffer empty */
+#define IPR_CH_0_LOOP 0x00000800 /* Channel 0 loop */
+#define IPR_CH_0_HALF_LOOP 0x00000100 /* Channel 0 half loop */
+#define IPR_CAP_0_LOOP 0x00080000 /* Channel capture loop */
+#define IPR_CAP_0_HALF_LOOP 0x00010000 /* Channel capture half loop */
+
+#define INTE 0x0c /* Interrupt enable register */
+#define INTE_MIDITXENABLE 0x00000001 /* Enable MIDI transmit-buffer-empty interrupts */
+#define INTE_MIDIRXENABLE 0x00000002 /* Enable MIDI receive-buffer-empty interrupts */
+#define INTE_CH_0_LOOP 0x00000800 /* Channel 0 loop */
+#define INTE_CH_0_HALF_LOOP 0x00000100 /* Channel 0 half loop */
+#define INTE_CAP_0_LOOP 0x00080000 /* Channel capture loop */
+#define INTE_CAP_0_HALF_LOOP 0x00010000 /* Channel capture half loop */
+
+#define HCFG 0x14 /* Hardware config register */
+
+#define HCFG_LOCKSOUNDCACHE 0x00000008 /* 1 = Cancel bustmaster accesses to soundcache */
+ /* NOTE: This should generally never be used. */
+#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */
+ /* Should be set to 1 when the EMU10K1 is */
+ /* completely initialized. */
+#define GPIO 0x18 /* Defaults: 00001080-Analog, 00001000-SPDIF. */
+
+
+#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
+
+#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
+
+/********************************************************************************************************/
+/* Emu10k1x pointer-offset register set, accessed through the PTR and DATA registers */
+/********************************************************************************************************/
+#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
+ /* One list entry: 4 bytes for DMA address,
+ * 4 bytes for period_size << 16.
+ * One list entry is 8 bytes long.
+ * One list entry for each period in the buffer.
+ */
+#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
+#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
+#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
+#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size */
+#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Sample currently in DAC */
+#define PLAYBACK_UNKNOWN1 0x07
+#define PLAYBACK_UNKNOWN2 0x08
+
+/* Only one capture channel supported */
+#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
+#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
+#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
+#define CAPTURE_UNKNOWN 0x13
+
+/* From 0x20 - 0x3f, last samples played on each channel */
+
+#define TRIGGER_CHANNEL 0x40 /* Trigger channel playback */
+#define TRIGGER_CHANNEL_0 0x00000001 /* Trigger channel 0 */
+#define TRIGGER_CHANNEL_1 0x00000002 /* Trigger channel 1 */
+#define TRIGGER_CHANNEL_2 0x00000004 /* Trigger channel 2 */
+#define TRIGGER_CAPTURE 0x00000100 /* Trigger capture channel */
+
+#define ROUTING 0x41 /* Setup sound routing ? */
+#define ROUTING_FRONT_LEFT 0x00000001
+#define ROUTING_FRONT_RIGHT 0x00000002
+#define ROUTING_REAR_LEFT 0x00000004
+#define ROUTING_REAR_RIGHT 0x00000008
+#define ROUTING_CENTER_LFE 0x00010000
+
+#define SPCS0 0x42 /* SPDIF output Channel Status 0 register */
+
+#define SPCS1 0x43 /* SPDIF output Channel Status 1 register */
+
+#define SPCS2 0x44 /* SPDIF output Channel Status 2 register */
+
+#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */
+#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */
+#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */
+#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */
+#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */
+#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */
+#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */
+#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */
+#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */
+#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */
+#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */
+#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */
+#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */
+#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */
+#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */
+#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */
+#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */
+#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */
+#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */
+#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */
+#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */
+#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */
+#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */
+
+#define SPDIF_SELECT 0x45 /* Enables SPDIF or Analogue outputs 0-Analogue, 0x700-SPDIF */
+
+/* This is the MPU port on the card */
+#define MUDATA 0x47
+#define MUCMD 0x48
+#define MUSTAT MUCMD
+
+/* From 0x50 - 0x5f, last samples captured */
+
+/**
+ * The hardware has 3 channels for playback and 1 for capture.
+ * - channel 0 is the front channel
+ * - channel 1 is the rear channel
+ * - channel 2 is the center/lfe chanel
+ * Volume is controlled by the AC97 for the front and rear channels by
+ * the PCM Playback Volume, Sigmatel Surround Playback Volume and
+ * Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects
+ * the front/rear channel mixing in the REAR OUT jack. When using the
+ * 4-Speaker Stereo, both front and rear channels will be mixed in the
+ * REAR OUT.
+ * The center/lfe channel has no volume control and cannot be muted during
+ * playback.
+ */
+
+struct emu10k1x_voice {
+ struct emu10k1x *emu;
+ int number;
+ int use;
+
+ struct emu10k1x_pcm *epcm;
+};
+
+struct emu10k1x_pcm {
+ struct emu10k1x *emu;
+ struct snd_pcm_substream *substream;
+ struct emu10k1x_voice *voice;
+ unsigned short running;
+};
+
+struct emu10k1x_midi {
+ struct emu10k1x *emu;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *substream_input;
+ struct snd_rawmidi_substream *substream_output;
+ unsigned int midi_mode;
+ spinlock_t input_lock;
+ spinlock_t output_lock;
+ spinlock_t open_lock;
+ int tx_enable, rx_enable;
+ int port;
+ int ipr_tx, ipr_rx;
+ void (*interrupt)(struct emu10k1x *emu, unsigned int status);
+};
+
+// definition of the chip-specific record
+struct emu10k1x {
+ struct snd_card *card;
+ struct pci_dev *pci;
+
+ unsigned long port;
+ struct resource *res_port;
+ int irq;
+
+ unsigned char revision; /* chip revision */
+ unsigned int serial; /* serial number */
+ unsigned short model; /* subsystem id */
+
+ spinlock_t emu_lock;
+ spinlock_t voice_lock;
+
+ struct snd_ac97 *ac97;
+ struct snd_pcm *pcm;
+
+ struct emu10k1x_voice voices[3];
+ struct emu10k1x_voice capture_voice;
+ u32 spdif_bits[3]; // SPDIF out setup
+
+ struct snd_dma_buffer dma_buffer;
+
+ struct emu10k1x_midi midi;
+};
+
+/* hardware definition */
+static struct snd_pcm_hardware snd_emu10k1x_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (32*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (16*1024),
+ .periods_min = 2,
+ .periods_max = 8,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_emu10k1x_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (32*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (16*1024),
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static unsigned int snd_emu10k1x_ptr_read(struct emu10k1x * emu,
+ unsigned int reg,
+ unsigned int chn)
+{
+ unsigned long flags;
+ unsigned int regptr, val;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+static void snd_emu10k1x_ptr_write(struct emu10k1x *emu,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data)
+{
+ unsigned int regptr;
+ unsigned long flags;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ outl(data, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_emu10k1x_intr_enable(struct emu10k1x *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int intr_enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ intr_enable = inl(emu->port + INTE) | intrenb;
+ outl(intr_enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_emu10k1x_intr_disable(struct emu10k1x *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int intr_enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ intr_enable = inl(emu->port + INTE) & ~intrenb;
+ outl(intr_enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_emu10k1x_gpio_write(struct emu10k1x *emu, unsigned int value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(value, emu->port + GPIO);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_emu10k1x_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static void snd_emu10k1x_pcm_interrupt(struct emu10k1x *emu, struct emu10k1x_voice *voice)
+{
+ struct emu10k1x_pcm *epcm;
+
+ if ((epcm = voice->epcm) == NULL)
+ return;
+ if (epcm->substream == NULL)
+ return;
+#if 0
+ snd_printk(KERN_INFO "IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n",
+ epcm->substream->ops->pointer(epcm->substream),
+ snd_pcm_lib_period_bytes(epcm->substream),
+ snd_pcm_lib_buffer_bytes(epcm->substream));
+#endif
+ snd_pcm_period_elapsed(epcm->substream);
+}
+
+/* open callback */
+static int snd_emu10k1x_playback_open(struct snd_pcm_substream *substream)
+{
+ struct emu10k1x *chip = snd_pcm_substream_chip(substream);
+ struct emu10k1x_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) {
+ return err;
+ }
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = chip;
+ epcm->substream = substream;
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_emu10k1x_pcm_free_substream;
+
+ runtime->hw = snd_emu10k1x_playback_hw;
+
+ return 0;
+}
+
+/* close callback */
+static int snd_emu10k1x_playback_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+/* hw_params callback */
+static int snd_emu10k1x_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm = runtime->private_data;
+
+ if (! epcm->voice) {
+ epcm->voice = &epcm->emu->voices[substream->pcm->device];
+ epcm->voice->use = 1;
+ epcm->voice->epcm = epcm;
+ }
+
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_emu10k1x_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm;
+
+ if (runtime->private_data == NULL)
+ return 0;
+
+ epcm = runtime->private_data;
+
+ if (epcm->voice) {
+ epcm->voice->use = 0;
+ epcm->voice->epcm = NULL;
+ epcm->voice = NULL;
+ }
+
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare callback */
+static int snd_emu10k1x_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct emu10k1x *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm = runtime->private_data;
+ int voice = epcm->voice->number;
+ u32 *table_base = (u32 *)(emu->dma_buffer.area+1024*voice);
+ u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
+ int i;
+
+ for(i = 0; i < runtime->periods; i++) {
+ *table_base++=runtime->dma_addr+(i*period_size_bytes);
+ *table_base++=period_size_bytes<<16;
+ }
+
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_LIST_ADDR, voice, emu->dma_buffer.addr+1024*voice);
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_LIST_SIZE, voice, (runtime->periods - 1) << 19);
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_LIST_PTR, voice, 0);
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_POINTER, voice, 0);
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_UNKNOWN1, voice, 0);
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_UNKNOWN2, voice, 0);
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_DMA_ADDR, voice, runtime->dma_addr);
+
+ snd_emu10k1x_ptr_write(emu, PLAYBACK_PERIOD_SIZE, voice, frames_to_bytes(runtime, runtime->period_size)<<16);
+
+ return 0;
+}
+
+/* trigger callback */
+static int snd_emu10k1x_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct emu10k1x *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm = runtime->private_data;
+ int channel = epcm->voice->number;
+ int result = 0;
+
+// snd_printk(KERN_INFO "trigger - emu10k1x = 0x%x, cmd = %i, pointer = %d\n", (int)emu, cmd, (int)substream->ops->pointer(substream));
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if(runtime->periods == 2)
+ snd_emu10k1x_intr_enable(emu, (INTE_CH_0_LOOP | INTE_CH_0_HALF_LOOP) << channel);
+ else
+ snd_emu10k1x_intr_enable(emu, INTE_CH_0_LOOP << channel);
+ epcm->running = 1;
+ snd_emu10k1x_ptr_write(emu, TRIGGER_CHANNEL, 0, snd_emu10k1x_ptr_read(emu, TRIGGER_CHANNEL, 0)|(TRIGGER_CHANNEL_0<<channel));
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ epcm->running = 0;
+ snd_emu10k1x_intr_disable(emu, (INTE_CH_0_LOOP | INTE_CH_0_HALF_LOOP) << channel);
+ snd_emu10k1x_ptr_write(emu, TRIGGER_CHANNEL, 0, snd_emu10k1x_ptr_read(emu, TRIGGER_CHANNEL, 0) & ~(TRIGGER_CHANNEL_0<<channel));
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* pointer callback */
+static snd_pcm_uframes_t
+snd_emu10k1x_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct emu10k1x *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm = runtime->private_data;
+ int channel = epcm->voice->number;
+ snd_pcm_uframes_t ptr = 0, ptr1 = 0, ptr2= 0,ptr3 = 0,ptr4 = 0;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr3 = snd_emu10k1x_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+ ptr1 = snd_emu10k1x_ptr_read(emu, PLAYBACK_POINTER, channel);
+ ptr4 = snd_emu10k1x_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+
+ if(ptr4 == 0 && ptr1 == frames_to_bytes(runtime, runtime->buffer_size))
+ return 0;
+
+ if (ptr3 != ptr4)
+ ptr1 = snd_emu10k1x_ptr_read(emu, PLAYBACK_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr2 += (ptr4 >> 3) * runtime->period_size;
+ ptr = ptr2;
+
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+
+ return ptr;
+}
+
+/* operators */
+static struct snd_pcm_ops snd_emu10k1x_playback_ops = {
+ .open = snd_emu10k1x_playback_open,
+ .close = snd_emu10k1x_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1x_pcm_hw_params,
+ .hw_free = snd_emu10k1x_pcm_hw_free,
+ .prepare = snd_emu10k1x_pcm_prepare,
+ .trigger = snd_emu10k1x_pcm_trigger,
+ .pointer = snd_emu10k1x_pcm_pointer,
+};
+
+/* open_capture callback */
+static int snd_emu10k1x_pcm_open_capture(struct snd_pcm_substream *substream)
+{
+ struct emu10k1x *chip = snd_pcm_substream_chip(substream);
+ struct emu10k1x_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL)
+ return -ENOMEM;
+
+ epcm->emu = chip;
+ epcm->substream = substream;
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_emu10k1x_pcm_free_substream;
+
+ runtime->hw = snd_emu10k1x_capture_hw;
+
+ return 0;
+}
+
+/* close callback */
+static int snd_emu10k1x_pcm_close_capture(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+/* hw_params callback */
+static int snd_emu10k1x_pcm_hw_params_capture(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm = runtime->private_data;
+
+ if (! epcm->voice) {
+ if (epcm->emu->capture_voice.use)
+ return -EBUSY;
+ epcm->voice = &epcm->emu->capture_voice;
+ epcm->voice->epcm = epcm;
+ epcm->voice->use = 1;
+ }
+
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_emu10k1x_pcm_hw_free_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ struct emu10k1x_pcm *epcm;
+
+ if (runtime->private_data == NULL)
+ return 0;
+ epcm = runtime->private_data;
+
+ if (epcm->voice) {
+ epcm->voice->use = 0;
+ epcm->voice->epcm = NULL;
+ epcm->voice = NULL;
+ }
+
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare capture callback */
+static int snd_emu10k1x_pcm_prepare_capture(struct snd_pcm_substream *substream)
+{
+ struct emu10k1x *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_emu10k1x_ptr_write(emu, CAPTURE_DMA_ADDR, 0, runtime->dma_addr);
+ snd_emu10k1x_ptr_write(emu, CAPTURE_BUFFER_SIZE, 0, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
+ snd_emu10k1x_ptr_write(emu, CAPTURE_POINTER, 0, 0);
+ snd_emu10k1x_ptr_write(emu, CAPTURE_UNKNOWN, 0, 0);
+
+ return 0;
+}
+
+/* trigger_capture callback */
+static int snd_emu10k1x_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct emu10k1x *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm = runtime->private_data;
+ int result = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_emu10k1x_intr_enable(emu, INTE_CAP_0_LOOP |
+ INTE_CAP_0_HALF_LOOP);
+ snd_emu10k1x_ptr_write(emu, TRIGGER_CHANNEL, 0, snd_emu10k1x_ptr_read(emu, TRIGGER_CHANNEL, 0)|TRIGGER_CAPTURE);
+ epcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ epcm->running = 0;
+ snd_emu10k1x_intr_disable(emu, INTE_CAP_0_LOOP |
+ INTE_CAP_0_HALF_LOOP);
+ snd_emu10k1x_ptr_write(emu, TRIGGER_CHANNEL, 0, snd_emu10k1x_ptr_read(emu, TRIGGER_CHANNEL, 0) & ~(TRIGGER_CAPTURE));
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* pointer_capture callback */
+static snd_pcm_uframes_t
+snd_emu10k1x_pcm_pointer_capture(struct snd_pcm_substream *substream)
+{
+ struct emu10k1x *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct emu10k1x_pcm *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr = bytes_to_frames(runtime, snd_emu10k1x_ptr_read(emu, CAPTURE_POINTER, 0));
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+
+ return ptr;
+}
+
+static struct snd_pcm_ops snd_emu10k1x_capture_ops = {
+ .open = snd_emu10k1x_pcm_open_capture,
+ .close = snd_emu10k1x_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1x_pcm_hw_params_capture,
+ .hw_free = snd_emu10k1x_pcm_hw_free_capture,
+ .prepare = snd_emu10k1x_pcm_prepare_capture,
+ .trigger = snd_emu10k1x_pcm_trigger_capture,
+ .pointer = snd_emu10k1x_pcm_pointer_capture,
+};
+
+static unsigned short snd_emu10k1x_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct emu10k1x *emu = ac97->private_data;
+ unsigned long flags;
+ unsigned short val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ val = inw(emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+static void snd_emu10k1x_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ struct emu10k1x *emu = ac97->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ outw(val, emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static int snd_emu10k1x_ac97(struct emu10k1x *chip)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_emu10k1x_ac97_write,
+ .read = snd_emu10k1x_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+ pbus->no_vra = 1; /* we don't need VRA */
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.scaps = AC97_SCAP_NO_SPDIF;
+ return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
+}
+
+static int snd_emu10k1x_free(struct emu10k1x *chip)
+{
+ snd_emu10k1x_ptr_write(chip, TRIGGER_CHANNEL, 0, 0);
+ // disable interrupts
+ outl(0, chip->port + INTE);
+ // disable audio
+ outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
+
+ /* release the irq */
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+
+ // release the i/o port
+ release_and_free_resource(chip->res_port);
+
+ // release the DMA
+ if (chip->dma_buffer.area) {
+ snd_dma_free_pages(&chip->dma_buffer);
+ }
+
+ pci_disable_device(chip->pci);
+
+ // release the data
+ kfree(chip);
+ return 0;
+}
+
+static int snd_emu10k1x_dev_free(struct snd_device *device)
+{
+ struct emu10k1x *chip = device->device_data;
+ return snd_emu10k1x_free(chip);
+}
+
+static irqreturn_t snd_emu10k1x_interrupt(int irq, void *dev_id)
+{
+ unsigned int status;
+
+ struct emu10k1x *chip = dev_id;
+ struct emu10k1x_voice *pvoice = chip->voices;
+ int i;
+ int mask;
+
+ status = inl(chip->port + IPR);
+
+ if (! status)
+ return IRQ_NONE;
+
+ // capture interrupt
+ if (status & (IPR_CAP_0_LOOP | IPR_CAP_0_HALF_LOOP)) {
+ struct emu10k1x_voice *cap_voice = &chip->capture_voice;
+ if (cap_voice->use)
+ snd_emu10k1x_pcm_interrupt(chip, cap_voice);
+ else
+ snd_emu10k1x_intr_disable(chip,
+ INTE_CAP_0_LOOP |
+ INTE_CAP_0_HALF_LOOP);
+ }
+
+ mask = IPR_CH_0_LOOP|IPR_CH_0_HALF_LOOP;
+ for (i = 0; i < 3; i++) {
+ if (status & mask) {
+ if (pvoice->use)
+ snd_emu10k1x_pcm_interrupt(chip, pvoice);
+ else
+ snd_emu10k1x_intr_disable(chip, mask);
+ }
+ pvoice++;
+ mask <<= 1;
+ }
+
+ if (status & (IPR_MIDITRANSBUFEMPTY|IPR_MIDIRECVBUFEMPTY)) {
+ if (chip->midi.interrupt)
+ chip->midi.interrupt(chip, status);
+ else
+ snd_emu10k1x_intr_disable(chip, INTE_MIDITXENABLE|INTE_MIDIRXENABLE);
+ }
+
+ // acknowledge the interrupt if necessary
+ outl(status, chip->port + IPR);
+
+ // snd_printk(KERN_INFO "interrupt %08x\n", status);
+ return IRQ_HANDLED;
+}
+
+static int __devinit snd_emu10k1x_pcm(struct emu10k1x *emu, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+ int capture = 0;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if (device == 0)
+ capture = 1;
+
+ if ((err = snd_pcm_new(emu->card, "emu10k1x", device, 1, capture, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+
+ switch(device) {
+ case 0:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1x_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_emu10k1x_capture_ops);
+ break;
+ case 1:
+ case 2:
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1x_playback_ops);
+ break;
+ }
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ switch(device) {
+ case 0:
+ strcpy(pcm->name, "EMU10K1X Front");
+ break;
+ case 1:
+ strcpy(pcm->name, "EMU10K1X Rear");
+ break;
+ case 2:
+ strcpy(pcm->name, "EMU10K1X Center/LFE");
+ break;
+ }
+ emu->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 32*1024, 32*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+static int __devinit snd_emu10k1x_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct emu10k1x **rchip)
+{
+ struct emu10k1x *chip;
+ int err;
+ int ch;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_emu10k1x_dev_free,
+ };
+
+ *rchip = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "error to set 28bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ spin_lock_init(&chip->emu_lock);
+ spin_lock_init(&chip->voice_lock);
+
+ chip->port = pci_resource_start(pci, 0);
+ if ((chip->res_port = request_region(chip->port, 8,
+ "EMU10K1X")) == NULL) {
+ snd_printk(KERN_ERR "emu10k1x: cannot allocate the port 0x%lx\n", chip->port);
+ snd_emu10k1x_free(chip);
+ return -EBUSY;
+ }
+
+ if (request_irq(pci->irq, snd_emu10k1x_interrupt,
+ IRQF_SHARED, "EMU10K1X", chip)) {
+ snd_printk(KERN_ERR "emu10k1x: cannot grab irq %d\n", pci->irq);
+ snd_emu10k1x_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ 4 * 1024, &chip->dma_buffer) < 0) {
+ snd_emu10k1x_free(chip);
+ return -ENOMEM;
+ }
+
+ pci_set_master(pci);
+ /* read revision & serial */
+ chip->revision = pci->revision;
+ pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
+ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
+ snd_printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model,
+ chip->revision, chip->serial);
+
+ outl(0, chip->port + INTE);
+
+ for(ch = 0; ch < 3; ch++) {
+ chip->voices[ch].emu = chip;
+ chip->voices[ch].number = ch;
+ }
+
+ /*
+ * Init to 0x02109204 :
+ * Clock accuracy = 0 (1000ppm)
+ * Sample Rate = 2 (48kHz)
+ * Audio Channel = 1 (Left of 2)
+ * Source Number = 0 (Unspecified)
+ * Generation Status = 1 (Original for Cat Code 12)
+ * Cat Code = 12 (Digital Signal Mixer)
+ * Mode = 0 (Mode 0)
+ * Emphasis = 0 (None)
+ * CP = 1 (Copyright unasserted)
+ * AN = 0 (Audio data)
+ * P = 0 (Consumer)
+ */
+ snd_emu10k1x_ptr_write(chip, SPCS0, 0,
+ chip->spdif_bits[0] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ snd_emu10k1x_ptr_write(chip, SPCS1, 0,
+ chip->spdif_bits[1] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ snd_emu10k1x_ptr_write(chip, SPCS2, 0,
+ chip->spdif_bits[2] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+
+ snd_emu10k1x_ptr_write(chip, SPDIF_SELECT, 0, 0x700); // disable SPDIF
+ snd_emu10k1x_ptr_write(chip, ROUTING, 0, 0x1003F); // routing
+ snd_emu10k1x_gpio_write(chip, 0x1080); // analog mode
+
+ outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &ops)) < 0) {
+ snd_emu10k1x_free(chip);
+ return err;
+ }
+ *rchip = chip;
+ return 0;
+}
+
+static void snd_emu10k1x_proc_reg_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct emu10k1x *emu = entry->private_data;
+ unsigned long value,value1,value2;
+ unsigned long flags;
+ int i;
+
+ snd_iprintf(buffer, "Registers:\n\n");
+ for(i = 0; i < 0x20; i+=4) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inl(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "Register %02X: %08lX\n", i, value);
+ }
+ snd_iprintf(buffer, "\nRegisters\n\n");
+ for(i = 0; i <= 0x48; i++) {
+ value = snd_emu10k1x_ptr_read(emu, i, 0);
+ if(i < 0x10 || (i >= 0x20 && i < 0x40)) {
+ value1 = snd_emu10k1x_ptr_read(emu, i, 1);
+ value2 = snd_emu10k1x_ptr_read(emu, i, 2);
+ snd_iprintf(buffer, "%02X: %08lX %08lX %08lX\n", i, value, value1, value2);
+ } else {
+ snd_iprintf(buffer, "%02X: %08lX\n", i, value);
+ }
+ }
+}
+
+static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct emu10k1x *emu = entry->private_data;
+ char line[64];
+ unsigned int reg, channel_id , val;
+
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
+ continue;
+
+ if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff)
+ && (channel_id >= 0) && (channel_id <= 2) )
+ snd_emu10k1x_ptr_write(emu, reg, channel_id, val);
+ }
+}
+
+static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu)
+{
+ struct snd_info_entry *entry;
+
+ if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read);
+ entry->c.text.write = snd_emu10k1x_proc_reg_write;
+ entry->mode |= S_IWUSR;
+ entry->private_data = emu;
+ }
+
+ return 0;
+}
+
+#define snd_emu10k1x_shared_spdif_info snd_ctl_boolean_mono_info
+
+static int snd_emu10k1x_shared_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct emu10k1x *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = (snd_emu10k1x_ptr_read(emu, SPDIF_SELECT, 0) == 0x700) ? 0 : 1;
+
+ return 0;
+}
+
+static int snd_emu10k1x_shared_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct emu10k1x *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+
+ val = ucontrol->value.integer.value[0] ;
+
+ if (val) {
+ // enable spdif output
+ snd_emu10k1x_ptr_write(emu, SPDIF_SELECT, 0, 0x000);
+ snd_emu10k1x_ptr_write(emu, ROUTING, 0, 0x700);
+ snd_emu10k1x_gpio_write(emu, 0x1000);
+ } else {
+ // disable spdif output
+ snd_emu10k1x_ptr_write(emu, SPDIF_SELECT, 0, 0x700);
+ snd_emu10k1x_ptr_write(emu, ROUTING, 0, 0x1003F);
+ snd_emu10k1x_gpio_write(emu, 0x1080);
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1x_shared_spdif __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog/Digital Output Jack",
+ .info = snd_emu10k1x_shared_spdif_info,
+ .get = snd_emu10k1x_shared_spdif_get,
+ .put = snd_emu10k1x_shared_spdif_put
+};
+
+static int snd_emu10k1x_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_emu10k1x_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct emu10k1x *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff;
+ return 0;
+}
+
+static int snd_emu10k1x_spdif_get_mask(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static int snd_emu10k1x_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct emu10k1x *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ int change;
+ unsigned int val;
+
+ val = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+ change = val != emu->spdif_bits[idx];
+ if (change) {
+ snd_emu10k1x_ptr_write(emu, SPCS0 + idx, 0, val);
+ emu->spdif_bits[idx] = val;
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1x_spdif_mask_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .count = 3,
+ .info = snd_emu10k1x_spdif_info,
+ .get = snd_emu10k1x_spdif_get_mask
+};
+
+static struct snd_kcontrol_new snd_emu10k1x_spdif_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .count = 3,
+ .info = snd_emu10k1x_spdif_info,
+ .get = snd_emu10k1x_spdif_get,
+ .put = snd_emu10k1x_spdif_put
+};
+
+static int __devinit snd_emu10k1x_mixer(struct emu10k1x *emu)
+{
+ int err;
+ struct snd_kcontrol *kctl;
+ struct snd_card *card = emu->card;
+
+ if ((kctl = snd_ctl_new1(&snd_emu10k1x_spdif_mask_control, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_emu10k1x_shared_spdif, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_emu10k1x_spdif_control, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+
+ return 0;
+}
+
+#define EMU10K1X_MIDI_MODE_INPUT (1<<0)
+#define EMU10K1X_MIDI_MODE_OUTPUT (1<<1)
+
+static inline unsigned char mpu401_read(struct emu10k1x *emu, struct emu10k1x_midi *mpu, int idx)
+{
+ return (unsigned char)snd_emu10k1x_ptr_read(emu, mpu->port + idx, 0);
+}
+
+static inline void mpu401_write(struct emu10k1x *emu, struct emu10k1x_midi *mpu, int data, int idx)
+{
+ snd_emu10k1x_ptr_write(emu, mpu->port + idx, 0, data);
+}
+
+#define mpu401_write_data(emu, mpu, data) mpu401_write(emu, mpu, data, 0)
+#define mpu401_write_cmd(emu, mpu, data) mpu401_write(emu, mpu, data, 1)
+#define mpu401_read_data(emu, mpu) mpu401_read(emu, mpu, 0)
+#define mpu401_read_stat(emu, mpu) mpu401_read(emu, mpu, 1)
+
+#define mpu401_input_avail(emu,mpu) (!(mpu401_read_stat(emu,mpu) & 0x80))
+#define mpu401_output_ready(emu,mpu) (!(mpu401_read_stat(emu,mpu) & 0x40))
+
+#define MPU401_RESET 0xff
+#define MPU401_ENTER_UART 0x3f
+#define MPU401_ACK 0xfe
+
+static void mpu401_clear_rx(struct emu10k1x *emu, struct emu10k1x_midi *mpu)
+{
+ int timeout = 100000;
+ for (; timeout > 0 && mpu401_input_avail(emu, mpu); timeout--)
+ mpu401_read_data(emu, mpu);
+#ifdef CONFIG_SND_DEBUG
+ if (timeout <= 0)
+ snd_printk(KERN_ERR "cmd: clear rx timeout (status = 0x%x)\n", mpu401_read_stat(emu, mpu));
+#endif
+}
+
+/*
+
+ */
+
+static void do_emu10k1x_midi_interrupt(struct emu10k1x *emu,
+ struct emu10k1x_midi *midi, unsigned int status)
+{
+ unsigned char byte;
+
+ if (midi->rmidi == NULL) {
+ snd_emu10k1x_intr_disable(emu, midi->tx_enable | midi->rx_enable);
+ return;
+ }
+
+ spin_lock(&midi->input_lock);
+ if ((status & midi->ipr_rx) && mpu401_input_avail(emu, midi)) {
+ if (!(midi->midi_mode & EMU10K1X_MIDI_MODE_INPUT)) {
+ mpu401_clear_rx(emu, midi);
+ } else {
+ byte = mpu401_read_data(emu, midi);
+ if (midi->substream_input)
+ snd_rawmidi_receive(midi->substream_input, &byte, 1);
+ }
+ }
+ spin_unlock(&midi->input_lock);
+
+ spin_lock(&midi->output_lock);
+ if ((status & midi->ipr_tx) && mpu401_output_ready(emu, midi)) {
+ if (midi->substream_output &&
+ snd_rawmidi_transmit(midi->substream_output, &byte, 1) == 1) {
+ mpu401_write_data(emu, midi, byte);
+ } else {
+ snd_emu10k1x_intr_disable(emu, midi->tx_enable);
+ }
+ }
+ spin_unlock(&midi->output_lock);
+}
+
+static void snd_emu10k1x_midi_interrupt(struct emu10k1x *emu, unsigned int status)
+{
+ do_emu10k1x_midi_interrupt(emu, &emu->midi, status);
+}
+
+static int snd_emu10k1x_midi_cmd(struct emu10k1x * emu,
+ struct emu10k1x_midi *midi, unsigned char cmd, int ack)
+{
+ unsigned long flags;
+ int timeout, ok;
+
+ spin_lock_irqsave(&midi->input_lock, flags);
+ mpu401_write_data(emu, midi, 0x00);
+ /* mpu401_clear_rx(emu, midi); */
+
+ mpu401_write_cmd(emu, midi, cmd);
+ if (ack) {
+ ok = 0;
+ timeout = 10000;
+ while (!ok && timeout-- > 0) {
+ if (mpu401_input_avail(emu, midi)) {
+ if (mpu401_read_data(emu, midi) == MPU401_ACK)
+ ok = 1;
+ }
+ }
+ if (!ok && mpu401_read_data(emu, midi) == MPU401_ACK)
+ ok = 1;
+ } else {
+ ok = 1;
+ }
+ spin_unlock_irqrestore(&midi->input_lock, flags);
+ if (!ok) {
+ snd_printk(KERN_ERR "midi_cmd: 0x%x failed at 0x%lx (status = 0x%x, data = 0x%x)!!!\n",
+ cmd, emu->port,
+ mpu401_read_stat(emu, midi),
+ mpu401_read_data(emu, midi));
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_emu10k1x_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct emu10k1x *emu;
+ struct emu10k1x_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= EMU10K1X_MIDI_MODE_INPUT;
+ midi->substream_input = substream;
+ if (!(midi->midi_mode & EMU10K1X_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ if (snd_emu10k1x_midi_cmd(emu, midi, MPU401_RESET, 1))
+ goto error_out;
+ if (snd_emu10k1x_midi_cmd(emu, midi, MPU401_ENTER_UART, 1))
+ goto error_out;
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+
+error_out:
+ return -EIO;
+}
+
+static int snd_emu10k1x_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct emu10k1x *emu;
+ struct emu10k1x_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= EMU10K1X_MIDI_MODE_OUTPUT;
+ midi->substream_output = substream;
+ if (!(midi->midi_mode & EMU10K1X_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ if (snd_emu10k1x_midi_cmd(emu, midi, MPU401_RESET, 1))
+ goto error_out;
+ if (snd_emu10k1x_midi_cmd(emu, midi, MPU401_ENTER_UART, 1))
+ goto error_out;
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+
+error_out:
+ return -EIO;
+}
+
+static int snd_emu10k1x_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct emu10k1x *emu;
+ struct emu10k1x_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+ int err = 0;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ snd_emu10k1x_intr_disable(emu, midi->rx_enable);
+ midi->midi_mode &= ~EMU10K1X_MIDI_MODE_INPUT;
+ midi->substream_input = NULL;
+ if (!(midi->midi_mode & EMU10K1X_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ err = snd_emu10k1x_midi_cmd(emu, midi, MPU401_RESET, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return err;
+}
+
+static int snd_emu10k1x_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct emu10k1x *emu;
+ struct emu10k1x_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+ int err = 0;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ snd_emu10k1x_intr_disable(emu, midi->tx_enable);
+ midi->midi_mode &= ~EMU10K1X_MIDI_MODE_OUTPUT;
+ midi->substream_output = NULL;
+ if (!(midi->midi_mode & EMU10K1X_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ err = snd_emu10k1x_midi_cmd(emu, midi, MPU401_RESET, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return err;
+}
+
+static void snd_emu10k1x_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct emu10k1x *emu;
+ struct emu10k1x_midi *midi = substream->rmidi->private_data;
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return;
+
+ if (up)
+ snd_emu10k1x_intr_enable(emu, midi->rx_enable);
+ else
+ snd_emu10k1x_intr_disable(emu, midi->rx_enable);
+}
+
+static void snd_emu10k1x_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct emu10k1x *emu;
+ struct emu10k1x_midi *midi = substream->rmidi->private_data;
+ unsigned long flags;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return;
+
+ if (up) {
+ int max = 4;
+ unsigned char byte;
+
+ /* try to send some amount of bytes here before interrupts */
+ spin_lock_irqsave(&midi->output_lock, flags);
+ while (max > 0) {
+ if (mpu401_output_ready(emu, midi)) {
+ if (!(midi->midi_mode & EMU10K1X_MIDI_MODE_OUTPUT) ||
+ snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ /* no more data */
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ return;
+ }
+ mpu401_write_data(emu, midi, byte);
+ max--;
+ } else {
+ break;
+ }
+ }
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ snd_emu10k1x_intr_enable(emu, midi->tx_enable);
+ } else {
+ snd_emu10k1x_intr_disable(emu, midi->tx_enable);
+ }
+}
+
+/*
+
+ */
+
+static struct snd_rawmidi_ops snd_emu10k1x_midi_output =
+{
+ .open = snd_emu10k1x_midi_output_open,
+ .close = snd_emu10k1x_midi_output_close,
+ .trigger = snd_emu10k1x_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_emu10k1x_midi_input =
+{
+ .open = snd_emu10k1x_midi_input_open,
+ .close = snd_emu10k1x_midi_input_close,
+ .trigger = snd_emu10k1x_midi_input_trigger,
+};
+
+static void snd_emu10k1x_midi_free(struct snd_rawmidi *rmidi)
+{
+ struct emu10k1x_midi *midi = rmidi->private_data;
+ midi->interrupt = NULL;
+ midi->rmidi = NULL;
+}
+
+static int __devinit emu10k1x_midi_init(struct emu10k1x *emu,
+ struct emu10k1x_midi *midi, int device, char *name)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if ((err = snd_rawmidi_new(emu->card, name, device, 1, 1, &rmidi)) < 0)
+ return err;
+ midi->emu = emu;
+ spin_lock_init(&midi->open_lock);
+ spin_lock_init(&midi->input_lock);
+ spin_lock_init(&midi->output_lock);
+ strcpy(rmidi->name, name);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_emu10k1x_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_emu10k1x_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = midi;
+ rmidi->private_free = snd_emu10k1x_midi_free;
+ midi->rmidi = rmidi;
+ return 0;
+}
+
+static int __devinit snd_emu10k1x_midi(struct emu10k1x *emu)
+{
+ struct emu10k1x_midi *midi = &emu->midi;
+ int err;
+
+ if ((err = emu10k1x_midi_init(emu, midi, 0, "EMU10K1X MPU-401 (UART)")) < 0)
+ return err;
+
+ midi->tx_enable = INTE_MIDITXENABLE;
+ midi->rx_enable = INTE_MIDIRXENABLE;
+ midi->port = MUDATA;
+ midi->ipr_tx = IPR_MIDITRANSBUFEMPTY;
+ midi->ipr_rx = IPR_MIDIRECVBUFEMPTY;
+ midi->interrupt = snd_emu10k1x_midi_interrupt;
+ return 0;
+}
+
+static int __devinit snd_emu10k1x_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct emu10k1x *chip;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_emu10k1x_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_emu10k1x_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_emu10k1x_pcm(chip, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_emu10k1x_pcm(chip, 2, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_emu10k1x_ac97(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_emu10k1x_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_emu10k1x_midi(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_emu10k1x_proc_init(chip);
+
+ strcpy(card->driver, "EMU10K1X");
+ strcpy(card->shortname, "Dell Sound Blaster Live!");
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->port, chip->irq);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_emu10k1x_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+// PCI IDs
+static struct pci_device_id snd_emu10k1x_ids[] = {
+ { 0x1102, 0x0006, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Dell OEM version (EMU10K1) */
+ { 0, }
+};
+MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids);
+
+// pci_driver definition
+static struct pci_driver driver = {
+ .name = "EMU10K1X",
+ .id_table = snd_emu10k1x_ids,
+ .probe = snd_emu10k1x_probe,
+ .remove = __devexit_p(snd_emu10k1x_remove),
+};
+
+// initialization of the module
+static int __init alsa_card_emu10k1x_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+// clean up the module
+static void __exit alsa_card_emu10k1x_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_emu10k1x_init)
+module_exit(alsa_card_emu10k1x_exit)
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
new file mode 100644
index 0000000..7dba08f
--- /dev/null
+++ b/sound/pci/emu10k1/emufx.c
@@ -0,0 +1,2745 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Creative Labs, Inc.
+ * Routines for effect processor FX8010
+ *
+ * Copyright (c) by James Courtier-Dutton <James@superbug.co.uk>
+ * Added EMU 1010 support.
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/pci.h>
+#include <linux/capability.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/init.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/tlv.h>
+#include <sound/emu10k1.h>
+
+#if 0 /* for testing purposes - digital out -> capture */
+#define EMU10K1_CAPTURE_DIGITAL_OUT
+#endif
+#if 0 /* for testing purposes - set S/PDIF to AC3 output */
+#define EMU10K1_SET_AC3_IEC958
+#endif
+#if 0 /* for testing purposes - feed the front signal to Center/LFE outputs */
+#define EMU10K1_CENTER_LFE_FROM_FRONT
+#endif
+
+/*
+ * Tables
+ */
+
+static char *fxbuses[16] = {
+ /* 0x00 */ "PCM Left",
+ /* 0x01 */ "PCM Right",
+ /* 0x02 */ "PCM Surround Left",
+ /* 0x03 */ "PCM Surround Right",
+ /* 0x04 */ "MIDI Left",
+ /* 0x05 */ "MIDI Right",
+ /* 0x06 */ "Center",
+ /* 0x07 */ "LFE",
+ /* 0x08 */ NULL,
+ /* 0x09 */ NULL,
+ /* 0x0a */ NULL,
+ /* 0x0b */ NULL,
+ /* 0x0c */ "MIDI Reverb",
+ /* 0x0d */ "MIDI Chorus",
+ /* 0x0e */ NULL,
+ /* 0x0f */ NULL
+};
+
+static char *creative_ins[16] = {
+ /* 0x00 */ "AC97 Left",
+ /* 0x01 */ "AC97 Right",
+ /* 0x02 */ "TTL IEC958 Left",
+ /* 0x03 */ "TTL IEC958 Right",
+ /* 0x04 */ "Zoom Video Left",
+ /* 0x05 */ "Zoom Video Right",
+ /* 0x06 */ "Optical IEC958 Left",
+ /* 0x07 */ "Optical IEC958 Right",
+ /* 0x08 */ "Line/Mic 1 Left",
+ /* 0x09 */ "Line/Mic 1 Right",
+ /* 0x0a */ "Coaxial IEC958 Left",
+ /* 0x0b */ "Coaxial IEC958 Right",
+ /* 0x0c */ "Line/Mic 2 Left",
+ /* 0x0d */ "Line/Mic 2 Right",
+ /* 0x0e */ NULL,
+ /* 0x0f */ NULL
+};
+
+static char *audigy_ins[16] = {
+ /* 0x00 */ "AC97 Left",
+ /* 0x01 */ "AC97 Right",
+ /* 0x02 */ "Audigy CD Left",
+ /* 0x03 */ "Audigy CD Right",
+ /* 0x04 */ "Optical IEC958 Left",
+ /* 0x05 */ "Optical IEC958 Right",
+ /* 0x06 */ NULL,
+ /* 0x07 */ NULL,
+ /* 0x08 */ "Line/Mic 2 Left",
+ /* 0x09 */ "Line/Mic 2 Right",
+ /* 0x0a */ "SPDIF Left",
+ /* 0x0b */ "SPDIF Right",
+ /* 0x0c */ "Aux2 Left",
+ /* 0x0d */ "Aux2 Right",
+ /* 0x0e */ NULL,
+ /* 0x0f */ NULL
+};
+
+static char *creative_outs[32] = {
+ /* 0x00 */ "AC97 Left",
+ /* 0x01 */ "AC97 Right",
+ /* 0x02 */ "Optical IEC958 Left",
+ /* 0x03 */ "Optical IEC958 Right",
+ /* 0x04 */ "Center",
+ /* 0x05 */ "LFE",
+ /* 0x06 */ "Headphone Left",
+ /* 0x07 */ "Headphone Right",
+ /* 0x08 */ "Surround Left",
+ /* 0x09 */ "Surround Right",
+ /* 0x0a */ "PCM Capture Left",
+ /* 0x0b */ "PCM Capture Right",
+ /* 0x0c */ "MIC Capture",
+ /* 0x0d */ "AC97 Surround Left",
+ /* 0x0e */ "AC97 Surround Right",
+ /* 0x0f */ NULL,
+ /* 0x10 */ NULL,
+ /* 0x11 */ "Analog Center",
+ /* 0x12 */ "Analog LFE",
+ /* 0x13 */ NULL,
+ /* 0x14 */ NULL,
+ /* 0x15 */ NULL,
+ /* 0x16 */ NULL,
+ /* 0x17 */ NULL,
+ /* 0x18 */ NULL,
+ /* 0x19 */ NULL,
+ /* 0x1a */ NULL,
+ /* 0x1b */ NULL,
+ /* 0x1c */ NULL,
+ /* 0x1d */ NULL,
+ /* 0x1e */ NULL,
+ /* 0x1f */ NULL,
+};
+
+static char *audigy_outs[32] = {
+ /* 0x00 */ "Digital Front Left",
+ /* 0x01 */ "Digital Front Right",
+ /* 0x02 */ "Digital Center",
+ /* 0x03 */ "Digital LEF",
+ /* 0x04 */ "Headphone Left",
+ /* 0x05 */ "Headphone Right",
+ /* 0x06 */ "Digital Rear Left",
+ /* 0x07 */ "Digital Rear Right",
+ /* 0x08 */ "Front Left",
+ /* 0x09 */ "Front Right",
+ /* 0x0a */ "Center",
+ /* 0x0b */ "LFE",
+ /* 0x0c */ NULL,
+ /* 0x0d */ NULL,
+ /* 0x0e */ "Rear Left",
+ /* 0x0f */ "Rear Right",
+ /* 0x10 */ "AC97 Front Left",
+ /* 0x11 */ "AC97 Front Right",
+ /* 0x12 */ "ADC Caputre Left",
+ /* 0x13 */ "ADC Capture Right",
+ /* 0x14 */ NULL,
+ /* 0x15 */ NULL,
+ /* 0x16 */ NULL,
+ /* 0x17 */ NULL,
+ /* 0x18 */ NULL,
+ /* 0x19 */ NULL,
+ /* 0x1a */ NULL,
+ /* 0x1b */ NULL,
+ /* 0x1c */ NULL,
+ /* 0x1d */ NULL,
+ /* 0x1e */ NULL,
+ /* 0x1f */ NULL,
+};
+
+static const u32 bass_table[41][5] = {
+ { 0x3e4f844f, 0x84ed4cc3, 0x3cc69927, 0x7b03553a, 0xc4da8486 },
+ { 0x3e69a17a, 0x84c280fb, 0x3cd77cd4, 0x7b2f2a6f, 0xc4b08d1d },
+ { 0x3e82ff42, 0x849991d5, 0x3ce7466b, 0x7b5917c6, 0xc48863ee },
+ { 0x3e9bab3c, 0x847267f0, 0x3cf5ffe8, 0x7b813560, 0xc461f22c },
+ { 0x3eb3b275, 0x844ced29, 0x3d03b295, 0x7ba79a1c, 0xc43d223b },
+ { 0x3ecb2174, 0x84290c8b, 0x3d106714, 0x7bcc5ba3, 0xc419dfa5 },
+ { 0x3ee2044b, 0x8406b244, 0x3d1c2561, 0x7bef8e77, 0xc3f8170f },
+ { 0x3ef86698, 0x83e5cb96, 0x3d26f4d8, 0x7c114600, 0xc3d7b625 },
+ { 0x3f0e5390, 0x83c646c9, 0x3d30dc39, 0x7c319498, 0xc3b8ab97 },
+ { 0x3f23d60b, 0x83a81321, 0x3d39e1af, 0x7c508b9c, 0xc39ae704 },
+ { 0x3f38f884, 0x838b20d2, 0x3d420ad2, 0x7c6e3b75, 0xc37e58f1 },
+ { 0x3f4dc52c, 0x836f60ef, 0x3d495cab, 0x7c8ab3a6, 0xc362f2be },
+ { 0x3f6245e8, 0x8354c565, 0x3d4fdbb8, 0x7ca602d6, 0xc348a69b },
+ { 0x3f76845f, 0x833b40ec, 0x3d558bf0, 0x7cc036df, 0xc32f677c },
+ { 0x3f8a8a03, 0x8322c6fb, 0x3d5a70c4, 0x7cd95cd7, 0xc317290b },
+ { 0x3f9e6014, 0x830b4bc3, 0x3d5e8d25, 0x7cf1811a, 0xc2ffdfa5 },
+ { 0x3fb20fae, 0x82f4c420, 0x3d61e37f, 0x7d08af56, 0xc2e9804a },
+ { 0x3fc5a1cc, 0x82df2592, 0x3d6475c3, 0x7d1ef294, 0xc2d40096 },
+ { 0x3fd91f55, 0x82ca6632, 0x3d664564, 0x7d345541, 0xc2bf56b9 },
+ { 0x3fec9120, 0x82b67cac, 0x3d675356, 0x7d48e138, 0xc2ab796e },
+ { 0x40000000, 0x82a36037, 0x3d67a012, 0x7d5c9fc9, 0xc2985fee },
+ { 0x401374c7, 0x8291088a, 0x3d672b93, 0x7d6f99c3, 0xc28601f2 },
+ { 0x4026f857, 0x827f6dd7, 0x3d65f559, 0x7d81d77c, 0xc27457a3 },
+ { 0x403a939f, 0x826e88c5, 0x3d63fc63, 0x7d9360d4, 0xc2635996 },
+ { 0x404e4faf, 0x825e5266, 0x3d613f32, 0x7da43d42, 0xc25300c6 },
+ { 0x406235ba, 0x824ec434, 0x3d5dbbc3, 0x7db473d7, 0xc243468e },
+ { 0x40764f1f, 0x823fd80c, 0x3d596f8f, 0x7dc40b44, 0xc23424a2 },
+ { 0x408aa576, 0x82318824, 0x3d545787, 0x7dd309e2, 0xc2259509 },
+ { 0x409f4296, 0x8223cf0b, 0x3d4e7012, 0x7de175b5, 0xc2179218 },
+ { 0x40b430a0, 0x8216a7a1, 0x3d47b505, 0x7def5475, 0xc20a1670 },
+ { 0x40c97a0a, 0x820a0d12, 0x3d4021a1, 0x7dfcab8d, 0xc1fd1cf5 },
+ { 0x40df29a6, 0x81fdfad6, 0x3d37b08d, 0x7e098028, 0xc1f0a0ca },
+ { 0x40f54ab1, 0x81f26ca9, 0x3d2e5bd1, 0x7e15d72b, 0xc1e49d52 },
+ { 0x410be8da, 0x81e75e89, 0x3d241cce, 0x7e21b544, 0xc1d90e24 },
+ { 0x41231051, 0x81dcccb3, 0x3d18ec37, 0x7e2d1ee6, 0xc1cdef10 },
+ { 0x413acdd0, 0x81d2b39e, 0x3d0cc20a, 0x7e38184e, 0xc1c33c13 },
+ { 0x41532ea7, 0x81c90ffb, 0x3cff9585, 0x7e42a58b, 0xc1b8f15a },
+ { 0x416c40cd, 0x81bfdeb2, 0x3cf15d21, 0x7e4cca7c, 0xc1af0b3f },
+ { 0x418612ea, 0x81b71cdc, 0x3ce20e85, 0x7e568ad3, 0xc1a58640 },
+ { 0x41a0b465, 0x81aec7c5, 0x3cd19e7c, 0x7e5fea1e, 0xc19c5f03 },
+ { 0x41bc3573, 0x81a6dcea, 0x3cc000e9, 0x7e68ebc2, 0xc1939250 }
+};
+
+static const u32 treble_table[41][5] = {
+ { 0x0125cba9, 0xfed5debd, 0x00599b6c, 0x0d2506da, 0xfa85b354 },
+ { 0x0142f67e, 0xfeb03163, 0x0066cd0f, 0x0d14c69d, 0xfa914473 },
+ { 0x016328bd, 0xfe860158, 0x0075b7f2, 0x0d03eb27, 0xfa9d32d2 },
+ { 0x0186b438, 0xfe56c982, 0x00869234, 0x0cf27048, 0xfaa97fca },
+ { 0x01adf358, 0xfe21f5fe, 0x00999842, 0x0ce051c2, 0xfab62ca5 },
+ { 0x01d949fa, 0xfde6e287, 0x00af0d8d, 0x0ccd8b4a, 0xfac33aa7 },
+ { 0x02092669, 0xfda4d8bf, 0x00c73d4c, 0x0cba1884, 0xfad0ab07 },
+ { 0x023e0268, 0xfd5b0e4a, 0x00e27b54, 0x0ca5f509, 0xfade7ef2 },
+ { 0x0278645c, 0xfd08a2b0, 0x01012509, 0x0c911c63, 0xfaecb788 },
+ { 0x02b8e091, 0xfcac9d1a, 0x0123a262, 0x0c7b8a14, 0xfafb55df },
+ { 0x03001a9a, 0xfc45e9ce, 0x014a6709, 0x0c65398f, 0xfb0a5aff },
+ { 0x034ec6d7, 0xfbd3576b, 0x0175f397, 0x0c4e2643, 0xfb19c7e4 },
+ { 0x03a5ac15, 0xfb5393ee, 0x01a6d6ed, 0x0c364b94, 0xfb299d7c },
+ { 0x0405a562, 0xfac52968, 0x01ddafae, 0x0c1da4e2, 0xfb39dca5 },
+ { 0x046fa3fe, 0xfa267a66, 0x021b2ddd, 0x0c042d8d, 0xfb4a8631 },
+ { 0x04e4b17f, 0xf975be0f, 0x0260149f, 0x0be9e0f2, 0xfb5b9ae0 },
+ { 0x0565f220, 0xf8b0fbe5, 0x02ad3c29, 0x0bceba73, 0xfb6d1b60 },
+ { 0x05f4a745, 0xf7d60722, 0x030393d4, 0x0bb2b578, 0xfb7f084d },
+ { 0x06923236, 0xf6e279bd, 0x03642465, 0x0b95cd75, 0xfb916233 },
+ { 0x07401713, 0xf5d3aef9, 0x03d01283, 0x0b77fded, 0xfba42984 },
+ { 0x08000000, 0xf4a6bd88, 0x0448a161, 0x0b594278, 0xfbb75e9f },
+ { 0x08d3c097, 0xf3587131, 0x04cf35a4, 0x0b3996c9, 0xfbcb01cb },
+ { 0x09bd59a2, 0xf1e543f9, 0x05655880, 0x0b18f6b2, 0xfbdf1333 },
+ { 0x0abefd0f, 0xf04956ca, 0x060cbb12, 0x0af75e2c, 0xfbf392e8 },
+ { 0x0bdb123e, 0xee806984, 0x06c739fe, 0x0ad4c962, 0xfc0880dd },
+ { 0x0d143a94, 0xec85d287, 0x0796e150, 0x0ab134b0, 0xfc1ddce5 },
+ { 0x0e6d5664, 0xea547598, 0x087df0a0, 0x0a8c9cb6, 0xfc33a6ad },
+ { 0x0fe98a2a, 0xe7e6ba35, 0x097edf83, 0x0a66fe5b, 0xfc49ddc2 },
+ { 0x118c4421, 0xe536813a, 0x0a9c6248, 0x0a4056d7, 0xfc608185 },
+ { 0x1359422e, 0xe23d19eb, 0x0bd96efb, 0x0a18a3bf, 0xfc77912c },
+ { 0x1554982b, 0xdef33645, 0x0d3942bd, 0x09efe312, 0xfc8f0bc1 },
+ { 0x1782b68a, 0xdb50deb1, 0x0ebf676d, 0x09c6133f, 0xfca6f019 },
+ { 0x19e8715d, 0xd74d64fd, 0x106fb999, 0x099b3337, 0xfcbf3cd6 },
+ { 0x1c8b07b8, 0xd2df56ab, 0x124e6ec8, 0x096f4274, 0xfcd7f060 },
+ { 0x1f702b6d, 0xcdfc6e92, 0x14601c10, 0x0942410b, 0xfcf108e5 },
+ { 0x229e0933, 0xc89985cd, 0x16a9bcfa, 0x09142fb5, 0xfd0a8451 },
+ { 0x261b5118, 0xc2aa8409, 0x1930bab6, 0x08e50fdc, 0xfd24604d },
+ { 0x29ef3f5d, 0xbc224f28, 0x1bfaf396, 0x08b4e3aa, 0xfd3e9a3b },
+ { 0x2e21a59b, 0xb4f2ba46, 0x1f0ec2d6, 0x0883ae15, 0xfd592f33 },
+ { 0x32baf44b, 0xad0c7429, 0x227308a3, 0x085172eb, 0xfd741bfd },
+ { 0x37c4448b, 0xa45ef51d, 0x262f3267, 0x081e36dc, 0xfd8f5d14 }
+};
+
+/* dB gain = (float) 20 * log10( float(db_table_value) / 0x8000000 ) */
+static const u32 db_table[101] = {
+ 0x00000000, 0x01571f82, 0x01674b41, 0x01783a1b, 0x0189f540,
+ 0x019c8651, 0x01aff763, 0x01c45306, 0x01d9a446, 0x01eff6b8,
+ 0x0207567a, 0x021fd03d, 0x0239714c, 0x02544792, 0x027061a1,
+ 0x028dcebb, 0x02ac9edc, 0x02cce2bf, 0x02eeabe8, 0x03120cb0,
+ 0x0337184e, 0x035de2df, 0x03868173, 0x03b10a18, 0x03dd93e9,
+ 0x040c3713, 0x043d0cea, 0x04702ff3, 0x04a5bbf2, 0x04ddcdfb,
+ 0x0518847f, 0x0555ff62, 0x05966005, 0x05d9c95d, 0x06206005,
+ 0x066a4a52, 0x06b7b067, 0x0708bc4c, 0x075d9a01, 0x07b6779d,
+ 0x08138561, 0x0874f5d5, 0x08dafde1, 0x0945d4ed, 0x09b5b4fd,
+ 0x0a2adad1, 0x0aa58605, 0x0b25f936, 0x0bac7a24, 0x0c3951d8,
+ 0x0ccccccc, 0x0d673b17, 0x0e08f093, 0x0eb24510, 0x0f639481,
+ 0x101d3f2d, 0x10dfa9e6, 0x11ab3e3f, 0x12806ac3, 0x135fa333,
+ 0x144960c5, 0x153e2266, 0x163e6cfe, 0x174acbb7, 0x1863d04d,
+ 0x198a1357, 0x1abe349f, 0x1c00db77, 0x1d52b712, 0x1eb47ee6,
+ 0x2026f30f, 0x21aadcb6, 0x23410e7e, 0x24ea64f9, 0x26a7c71d,
+ 0x287a26c4, 0x2a62812c, 0x2c61df84, 0x2e795779, 0x30aa0bcf,
+ 0x32f52cfe, 0x355bf9d8, 0x37dfc033, 0x3a81dda4, 0x3d43c038,
+ 0x4026e73c, 0x432ce40f, 0x46575af8, 0x49a8040f, 0x4d20ac2a,
+ 0x50c335d3, 0x54919a57, 0x588dead1, 0x5cba514a, 0x611911ea,
+ 0x65ac8c2f, 0x6a773c39, 0x6f7bbc23, 0x74bcc56c, 0x7a3d3272,
+ 0x7fffffff,
+};
+
+/* EMU10k1/EMU10k2 DSP control db gain */
+static const DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1);
+
+static const u32 onoff_table[2] = {
+ 0x00000000, 0x00000001
+};
+
+/*
+ */
+
+static inline mm_segment_t snd_enter_user(void)
+{
+ mm_segment_t fs = get_fs();
+ set_fs(get_ds());
+ return fs;
+}
+
+static inline void snd_leave_user(mm_segment_t fs)
+{
+ set_fs(fs);
+}
+
+/*
+ * controls
+ */
+
+static int snd_emu10k1_gpr_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_emu10k1_fx8010_ctl *ctl =
+ (struct snd_emu10k1_fx8010_ctl *) kcontrol->private_value;
+
+ if (ctl->min == 0 && ctl->max == 1)
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = ctl->vcount;
+ uinfo->value.integer.min = ctl->min;
+ uinfo->value.integer.max = ctl->max;
+ return 0;
+}
+
+static int snd_emu10k1_gpr_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_fx8010_ctl *ctl =
+ (struct snd_emu10k1_fx8010_ctl *) kcontrol->private_value;
+ unsigned long flags;
+ unsigned int i;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (i = 0; i < ctl->vcount; i++)
+ ucontrol->value.integer.value[i] = ctl->value[i];
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_gpr_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_fx8010_ctl *ctl =
+ (struct snd_emu10k1_fx8010_ctl *) kcontrol->private_value;
+ unsigned long flags;
+ unsigned int nval, val;
+ unsigned int i, j;
+ int change = 0;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (i = 0; i < ctl->vcount; i++) {
+ nval = ucontrol->value.integer.value[i];
+ if (nval < ctl->min)
+ nval = ctl->min;
+ if (nval > ctl->max)
+ nval = ctl->max;
+ if (nval != ctl->value[i])
+ change = 1;
+ val = ctl->value[i] = nval;
+ switch (ctl->translation) {
+ case EMU10K1_GPR_TRANSLATION_NONE:
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + ctl->gpr[i], 0, val);
+ break;
+ case EMU10K1_GPR_TRANSLATION_TABLE100:
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + ctl->gpr[i], 0, db_table[val]);
+ break;
+ case EMU10K1_GPR_TRANSLATION_BASS:
+ if ((ctl->count % 5) != 0 || (ctl->count / 5) != ctl->vcount) {
+ change = -EIO;
+ goto __error;
+ }
+ for (j = 0; j < 5; j++)
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + ctl->gpr[j * ctl->vcount + i], 0, bass_table[val][j]);
+ break;
+ case EMU10K1_GPR_TRANSLATION_TREBLE:
+ if ((ctl->count % 5) != 0 || (ctl->count / 5) != ctl->vcount) {
+ change = -EIO;
+ goto __error;
+ }
+ for (j = 0; j < 5; j++)
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + ctl->gpr[j * ctl->vcount + i], 0, treble_table[val][j]);
+ break;
+ case EMU10K1_GPR_TRANSLATION_ONOFF:
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + ctl->gpr[i], 0, onoff_table[val]);
+ break;
+ }
+ }
+ __error:
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+/*
+ * Interrupt handler
+ */
+
+static void snd_emu10k1_fx8010_interrupt(struct snd_emu10k1 *emu)
+{
+ struct snd_emu10k1_fx8010_irq *irq, *nirq;
+
+ irq = emu->fx8010.irq_handlers;
+ while (irq) {
+ nirq = irq->next; /* irq ptr can be removed from list */
+ if (snd_emu10k1_ptr_read(emu, emu->gpr_base + irq->gpr_running, 0) & 0xffff0000) {
+ if (irq->handler)
+ irq->handler(emu, irq->private_data);
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + irq->gpr_running, 0, 1);
+ }
+ irq = nirq;
+ }
+}
+
+int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
+ snd_fx8010_irq_handler_t *handler,
+ unsigned char gpr_running,
+ void *private_data,
+ struct snd_emu10k1_fx8010_irq **r_irq)
+{
+ struct snd_emu10k1_fx8010_irq *irq;
+ unsigned long flags;
+
+ irq = kmalloc(sizeof(*irq), GFP_ATOMIC);
+ if (irq == NULL)
+ return -ENOMEM;
+ irq->handler = handler;
+ irq->gpr_running = gpr_running;
+ irq->private_data = private_data;
+ irq->next = NULL;
+ spin_lock_irqsave(&emu->fx8010.irq_lock, flags);
+ if (emu->fx8010.irq_handlers == NULL) {
+ emu->fx8010.irq_handlers = irq;
+ emu->dsp_interrupt = snd_emu10k1_fx8010_interrupt;
+ snd_emu10k1_intr_enable(emu, INTE_FXDSPENABLE);
+ } else {
+ irq->next = emu->fx8010.irq_handlers;
+ emu->fx8010.irq_handlers = irq;
+ }
+ spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags);
+ if (r_irq)
+ *r_irq = irq;
+ return 0;
+}
+
+int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_irq *irq)
+{
+ struct snd_emu10k1_fx8010_irq *tmp;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->fx8010.irq_lock, flags);
+ if ((tmp = emu->fx8010.irq_handlers) == irq) {
+ emu->fx8010.irq_handlers = tmp->next;
+ if (emu->fx8010.irq_handlers == NULL) {
+ snd_emu10k1_intr_disable(emu, INTE_FXDSPENABLE);
+ emu->dsp_interrupt = NULL;
+ }
+ } else {
+ while (tmp && tmp->next != irq)
+ tmp = tmp->next;
+ if (tmp)
+ tmp->next = tmp->next->next;
+ }
+ spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags);
+ kfree(irq);
+ return 0;
+}
+
+/*************************************************************************
+ * EMU10K1 effect manager
+ *************************************************************************/
+
+static void snd_emu10k1_write_op(struct snd_emu10k1_fx8010_code *icode,
+ unsigned int *ptr,
+ u32 op, u32 r, u32 a, u32 x, u32 y)
+{
+ u_int32_t *code;
+ if (snd_BUG_ON(*ptr >= 512))
+ return;
+ code = (u_int32_t __force *)icode->code + (*ptr) * 2;
+ set_bit(*ptr, icode->code_valid);
+ code[0] = ((x & 0x3ff) << 10) | (y & 0x3ff);
+ code[1] = ((op & 0x0f) << 20) | ((r & 0x3ff) << 10) | (a & 0x3ff);
+ (*ptr)++;
+}
+
+#define OP(icode, ptr, op, r, a, x, y) \
+ snd_emu10k1_write_op(icode, ptr, op, r, a, x, y)
+
+static void snd_emu10k1_audigy_write_op(struct snd_emu10k1_fx8010_code *icode,
+ unsigned int *ptr,
+ u32 op, u32 r, u32 a, u32 x, u32 y)
+{
+ u_int32_t *code;
+ if (snd_BUG_ON(*ptr >= 1024))
+ return;
+ code = (u_int32_t __force *)icode->code + (*ptr) * 2;
+ set_bit(*ptr, icode->code_valid);
+ code[0] = ((x & 0x7ff) << 12) | (y & 0x7ff);
+ code[1] = ((op & 0x0f) << 24) | ((r & 0x7ff) << 12) | (a & 0x7ff);
+ (*ptr)++;
+}
+
+#define A_OP(icode, ptr, op, r, a, x, y) \
+ snd_emu10k1_audigy_write_op(icode, ptr, op, r, a, x, y)
+
+static void snd_emu10k1_efx_write(struct snd_emu10k1 *emu, unsigned int pc, unsigned int data)
+{
+ pc += emu->audigy ? A_MICROCODEBASE : MICROCODEBASE;
+ snd_emu10k1_ptr_write(emu, pc, 0, data);
+}
+
+unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc)
+{
+ pc += emu->audigy ? A_MICROCODEBASE : MICROCODEBASE;
+ return snd_emu10k1_ptr_read(emu, pc, 0);
+}
+
+static int snd_emu10k1_gpr_poke(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ int gpr;
+ u32 val;
+
+ for (gpr = 0; gpr < (emu->audigy ? 0x200 : 0x100); gpr++) {
+ if (!test_bit(gpr, icode->gpr_valid))
+ continue;
+ if (get_user(val, &icode->gpr_map[gpr]))
+ return -EFAULT;
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + gpr, 0, val);
+ }
+ return 0;
+}
+
+static int snd_emu10k1_gpr_peek(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ int gpr;
+ u32 val;
+
+ for (gpr = 0; gpr < (emu->audigy ? 0x200 : 0x100); gpr++) {
+ set_bit(gpr, icode->gpr_valid);
+ val = snd_emu10k1_ptr_read(emu, emu->gpr_base + gpr, 0);
+ if (put_user(val, &icode->gpr_map[gpr]))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static int snd_emu10k1_tram_poke(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ int tram;
+ u32 addr, val;
+
+ for (tram = 0; tram < (emu->audigy ? 0x100 : 0xa0); tram++) {
+ if (!test_bit(tram, icode->tram_valid))
+ continue;
+ if (get_user(val, &icode->tram_data_map[tram]) ||
+ get_user(addr, &icode->tram_addr_map[tram]))
+ return -EFAULT;
+ snd_emu10k1_ptr_write(emu, TANKMEMDATAREGBASE + tram, 0, val);
+ if (!emu->audigy) {
+ snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + tram, 0, addr);
+ } else {
+ snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + tram, 0, addr << 12);
+ snd_emu10k1_ptr_write(emu, A_TANKMEMCTLREGBASE + tram, 0, addr >> 20);
+ }
+ }
+ return 0;
+}
+
+static int snd_emu10k1_tram_peek(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ int tram;
+ u32 val, addr;
+
+ memset(icode->tram_valid, 0, sizeof(icode->tram_valid));
+ for (tram = 0; tram < (emu->audigy ? 0x100 : 0xa0); tram++) {
+ set_bit(tram, icode->tram_valid);
+ val = snd_emu10k1_ptr_read(emu, TANKMEMDATAREGBASE + tram, 0);
+ if (!emu->audigy) {
+ addr = snd_emu10k1_ptr_read(emu, TANKMEMADDRREGBASE + tram, 0);
+ } else {
+ addr = snd_emu10k1_ptr_read(emu, TANKMEMADDRREGBASE + tram, 0) >> 12;
+ addr |= snd_emu10k1_ptr_read(emu, A_TANKMEMCTLREGBASE + tram, 0) << 20;
+ }
+ if (put_user(val, &icode->tram_data_map[tram]) ||
+ put_user(addr, &icode->tram_addr_map[tram]))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static int snd_emu10k1_code_poke(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ u32 pc, lo, hi;
+
+ for (pc = 0; pc < (emu->audigy ? 2*1024 : 2*512); pc += 2) {
+ if (!test_bit(pc / 2, icode->code_valid))
+ continue;
+ if (get_user(lo, &icode->code[pc + 0]) ||
+ get_user(hi, &icode->code[pc + 1]))
+ return -EFAULT;
+ snd_emu10k1_efx_write(emu, pc + 0, lo);
+ snd_emu10k1_efx_write(emu, pc + 1, hi);
+ }
+ return 0;
+}
+
+static int snd_emu10k1_code_peek(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ u32 pc;
+
+ memset(icode->code_valid, 0, sizeof(icode->code_valid));
+ for (pc = 0; pc < (emu->audigy ? 2*1024 : 2*512); pc += 2) {
+ set_bit(pc / 2, icode->code_valid);
+ if (put_user(snd_emu10k1_efx_read(emu, pc + 0), &icode->code[pc + 0]))
+ return -EFAULT;
+ if (put_user(snd_emu10k1_efx_read(emu, pc + 1), &icode->code[pc + 1]))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static struct snd_emu10k1_fx8010_ctl *
+snd_emu10k1_look_for_ctl(struct snd_emu10k1 *emu, struct snd_ctl_elem_id *id)
+{
+ struct snd_emu10k1_fx8010_ctl *ctl;
+ struct snd_kcontrol *kcontrol;
+
+ list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) {
+ kcontrol = ctl->kcontrol;
+ if (kcontrol->id.iface == id->iface &&
+ !strcmp(kcontrol->id.name, id->name) &&
+ kcontrol->id.index == id->index)
+ return ctl;
+ }
+ return NULL;
+}
+
+#define MAX_TLV_SIZE 256
+
+static unsigned int *copy_tlv(const unsigned int __user *_tlv)
+{
+ unsigned int data[2];
+ unsigned int *tlv;
+
+ if (!_tlv)
+ return NULL;
+ if (copy_from_user(data, _tlv, sizeof(data)))
+ return NULL;
+ if (data[1] >= MAX_TLV_SIZE)
+ return NULL;
+ tlv = kmalloc(data[1] + sizeof(data), GFP_KERNEL);
+ if (!tlv)
+ return NULL;
+ memcpy(tlv, data, sizeof(data));
+ if (copy_from_user(tlv + 2, _tlv + 2, data[1])) {
+ kfree(tlv);
+ return NULL;
+ }
+ return tlv;
+}
+
+static int copy_gctl(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_control_gpr *gctl,
+ struct snd_emu10k1_fx8010_control_gpr __user *_gctl,
+ int idx)
+{
+ struct snd_emu10k1_fx8010_control_old_gpr __user *octl;
+
+ if (emu->support_tlv)
+ return copy_from_user(gctl, &_gctl[idx], sizeof(*gctl));
+ octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl;
+ if (copy_from_user(gctl, &octl[idx], sizeof(*octl)))
+ return -EFAULT;
+ gctl->tlv = NULL;
+ return 0;
+}
+
+static int copy_gctl_to_user(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_control_gpr __user *_gctl,
+ struct snd_emu10k1_fx8010_control_gpr *gctl,
+ int idx)
+{
+ struct snd_emu10k1_fx8010_control_old_gpr __user *octl;
+
+ if (emu->support_tlv)
+ return copy_to_user(&_gctl[idx], gctl, sizeof(*gctl));
+
+ octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl;
+ return copy_to_user(&octl[idx], gctl, sizeof(*octl));
+}
+
+static int snd_emu10k1_verify_controls(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ unsigned int i;
+ struct snd_ctl_elem_id __user *_id;
+ struct snd_ctl_elem_id id;
+ struct snd_emu10k1_fx8010_control_gpr *gctl;
+ int err;
+
+ for (i = 0, _id = icode->gpr_del_controls;
+ i < icode->gpr_del_control_count; i++, _id++) {
+ if (copy_from_user(&id, _id, sizeof(id)))
+ return -EFAULT;
+ if (snd_emu10k1_look_for_ctl(emu, &id) == NULL)
+ return -ENOENT;
+ }
+ gctl = kmalloc(sizeof(*gctl), GFP_KERNEL);
+ if (! gctl)
+ return -ENOMEM;
+ err = 0;
+ for (i = 0; i < icode->gpr_add_control_count; i++) {
+ if (copy_gctl(emu, gctl, icode->gpr_add_controls, i)) {
+ err = -EFAULT;
+ goto __error;
+ }
+ if (snd_emu10k1_look_for_ctl(emu, &gctl->id))
+ continue;
+ down_read(&emu->card->controls_rwsem);
+ if (snd_ctl_find_id(emu->card, &gctl->id) != NULL) {
+ up_read(&emu->card->controls_rwsem);
+ err = -EEXIST;
+ goto __error;
+ }
+ up_read(&emu->card->controls_rwsem);
+ if (gctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER &&
+ gctl->id.iface != SNDRV_CTL_ELEM_IFACE_PCM) {
+ err = -EINVAL;
+ goto __error;
+ }
+ }
+ for (i = 0; i < icode->gpr_list_control_count; i++) {
+ /* FIXME: we need to check the WRITE access */
+ if (copy_gctl(emu, gctl, icode->gpr_list_controls, i)) {
+ err = -EFAULT;
+ goto __error;
+ }
+ }
+ __error:
+ kfree(gctl);
+ return err;
+}
+
+static void snd_emu10k1_ctl_private_free(struct snd_kcontrol *kctl)
+{
+ struct snd_emu10k1_fx8010_ctl *ctl;
+
+ ctl = (struct snd_emu10k1_fx8010_ctl *) kctl->private_value;
+ kctl->private_value = 0;
+ list_del(&ctl->list);
+ kfree(ctl);
+ if (kctl->tlv.p)
+ kfree(kctl->tlv.p);
+}
+
+static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ unsigned int i, j;
+ struct snd_emu10k1_fx8010_control_gpr *gctl;
+ struct snd_emu10k1_fx8010_ctl *ctl, *nctl;
+ struct snd_kcontrol_new knew;
+ struct snd_kcontrol *kctl;
+ struct snd_ctl_elem_value *val;
+ int err = 0;
+
+ val = kmalloc(sizeof(*val), GFP_KERNEL);
+ gctl = kmalloc(sizeof(*gctl), GFP_KERNEL);
+ nctl = kmalloc(sizeof(*nctl), GFP_KERNEL);
+ if (!val || !gctl || !nctl) {
+ err = -ENOMEM;
+ goto __error;
+ }
+
+ for (i = 0; i < icode->gpr_add_control_count; i++) {
+ if (copy_gctl(emu, gctl, icode->gpr_add_controls, i)) {
+ err = -EFAULT;
+ goto __error;
+ }
+ if (gctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER &&
+ gctl->id.iface != SNDRV_CTL_ELEM_IFACE_PCM) {
+ err = -EINVAL;
+ goto __error;
+ }
+ if (! gctl->id.name[0]) {
+ err = -EINVAL;
+ goto __error;
+ }
+ ctl = snd_emu10k1_look_for_ctl(emu, &gctl->id);
+ memset(&knew, 0, sizeof(knew));
+ knew.iface = gctl->id.iface;
+ knew.name = gctl->id.name;
+ knew.index = gctl->id.index;
+ knew.device = gctl->id.device;
+ knew.subdevice = gctl->id.subdevice;
+ knew.info = snd_emu10k1_gpr_ctl_info;
+ knew.tlv.p = copy_tlv(gctl->tlv);
+ if (knew.tlv.p)
+ knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ knew.get = snd_emu10k1_gpr_ctl_get;
+ knew.put = snd_emu10k1_gpr_ctl_put;
+ memset(nctl, 0, sizeof(*nctl));
+ nctl->vcount = gctl->vcount;
+ nctl->count = gctl->count;
+ for (j = 0; j < 32; j++) {
+ nctl->gpr[j] = gctl->gpr[j];
+ nctl->value[j] = ~gctl->value[j]; /* inverted, we want to write new value in gpr_ctl_put() */
+ val->value.integer.value[j] = gctl->value[j];
+ }
+ nctl->min = gctl->min;
+ nctl->max = gctl->max;
+ nctl->translation = gctl->translation;
+ if (ctl == NULL) {
+ ctl = kmalloc(sizeof(*ctl), GFP_KERNEL);
+ if (ctl == NULL) {
+ err = -ENOMEM;
+ kfree(knew.tlv.p);
+ goto __error;
+ }
+ knew.private_value = (unsigned long)ctl;
+ *ctl = *nctl;
+ if ((err = snd_ctl_add(emu->card, kctl = snd_ctl_new1(&knew, emu))) < 0) {
+ kfree(ctl);
+ kfree(knew.tlv.p);
+ goto __error;
+ }
+ kctl->private_free = snd_emu10k1_ctl_private_free;
+ ctl->kcontrol = kctl;
+ list_add_tail(&ctl->list, &emu->fx8010.gpr_ctl);
+ } else {
+ /* overwrite */
+ nctl->list = ctl->list;
+ nctl->kcontrol = ctl->kcontrol;
+ *ctl = *nctl;
+ snd_ctl_notify(emu->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &ctl->kcontrol->id);
+ }
+ snd_emu10k1_gpr_ctl_put(ctl->kcontrol, val);
+ }
+ __error:
+ kfree(nctl);
+ kfree(gctl);
+ kfree(val);
+ return err;
+}
+
+static int snd_emu10k1_del_controls(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ unsigned int i;
+ struct snd_ctl_elem_id id;
+ struct snd_ctl_elem_id __user *_id;
+ struct snd_emu10k1_fx8010_ctl *ctl;
+ struct snd_card *card = emu->card;
+
+ for (i = 0, _id = icode->gpr_del_controls;
+ i < icode->gpr_del_control_count; i++, _id++) {
+ if (copy_from_user(&id, _id, sizeof(id)))
+ return -EFAULT;
+ down_write(&card->controls_rwsem);
+ ctl = snd_emu10k1_look_for_ctl(emu, &id);
+ if (ctl)
+ snd_ctl_remove(card, ctl->kcontrol);
+ up_write(&card->controls_rwsem);
+ }
+ return 0;
+}
+
+static int snd_emu10k1_list_controls(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ unsigned int i = 0, j;
+ unsigned int total = 0;
+ struct snd_emu10k1_fx8010_control_gpr *gctl;
+ struct snd_emu10k1_fx8010_ctl *ctl;
+ struct snd_ctl_elem_id *id;
+
+ gctl = kmalloc(sizeof(*gctl), GFP_KERNEL);
+ if (! gctl)
+ return -ENOMEM;
+
+ list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) {
+ total++;
+ if (icode->gpr_list_controls &&
+ i < icode->gpr_list_control_count) {
+ memset(gctl, 0, sizeof(*gctl));
+ id = &ctl->kcontrol->id;
+ gctl->id.iface = id->iface;
+ strlcpy(gctl->id.name, id->name, sizeof(gctl->id.name));
+ gctl->id.index = id->index;
+ gctl->id.device = id->device;
+ gctl->id.subdevice = id->subdevice;
+ gctl->vcount = ctl->vcount;
+ gctl->count = ctl->count;
+ for (j = 0; j < 32; j++) {
+ gctl->gpr[j] = ctl->gpr[j];
+ gctl->value[j] = ctl->value[j];
+ }
+ gctl->min = ctl->min;
+ gctl->max = ctl->max;
+ gctl->translation = ctl->translation;
+ if (copy_gctl_to_user(emu, icode->gpr_list_controls,
+ gctl, i)) {
+ kfree(gctl);
+ return -EFAULT;
+ }
+ i++;
+ }
+ }
+ icode->gpr_list_control_total = total;
+ kfree(gctl);
+ return 0;
+}
+
+static int snd_emu10k1_icode_poke(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ int err = 0;
+
+ mutex_lock(&emu->fx8010.lock);
+ if ((err = snd_emu10k1_verify_controls(emu, icode)) < 0)
+ goto __error;
+ strlcpy(emu->fx8010.name, icode->name, sizeof(emu->fx8010.name));
+ /* stop FX processor - this may be dangerous, but it's better to miss
+ some samples than generate wrong ones - [jk] */
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg | A_DBG_SINGLE_STEP);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg | EMU10K1_DBG_SINGLE_STEP);
+ /* ok, do the main job */
+ if ((err = snd_emu10k1_del_controls(emu, icode)) < 0 ||
+ (err = snd_emu10k1_gpr_poke(emu, icode)) < 0 ||
+ (err = snd_emu10k1_tram_poke(emu, icode)) < 0 ||
+ (err = snd_emu10k1_code_poke(emu, icode)) < 0 ||
+ (err = snd_emu10k1_add_controls(emu, icode)) < 0)
+ goto __error;
+ /* start FX processor when the DSP code is updated */
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg);
+ __error:
+ mutex_unlock(&emu->fx8010.lock);
+ return err;
+}
+
+static int snd_emu10k1_icode_peek(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_code *icode)
+{
+ int err;
+
+ mutex_lock(&emu->fx8010.lock);
+ strlcpy(icode->name, emu->fx8010.name, sizeof(icode->name));
+ /* ok, do the main job */
+ err = snd_emu10k1_gpr_peek(emu, icode);
+ if (err >= 0)
+ err = snd_emu10k1_tram_peek(emu, icode);
+ if (err >= 0)
+ err = snd_emu10k1_code_peek(emu, icode);
+ if (err >= 0)
+ err = snd_emu10k1_list_controls(emu, icode);
+ mutex_unlock(&emu->fx8010.lock);
+ return err;
+}
+
+static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_pcm_rec *ipcm)
+{
+ unsigned int i;
+ int err = 0;
+ struct snd_emu10k1_fx8010_pcm *pcm;
+
+ if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
+ return -EINVAL;
+ if (ipcm->channels > 32)
+ return -EINVAL;
+ pcm = &emu->fx8010.pcm[ipcm->substream];
+ mutex_lock(&emu->fx8010.lock);
+ spin_lock_irq(&emu->reg_lock);
+ if (pcm->opened) {
+ err = -EBUSY;
+ goto __error;
+ }
+ if (ipcm->channels == 0) { /* remove */
+ pcm->valid = 0;
+ } else {
+ /* FIXME: we need to add universal code to the PCM transfer routine */
+ if (ipcm->channels != 2) {
+ err = -EINVAL;
+ goto __error;
+ }
+ pcm->valid = 1;
+ pcm->opened = 0;
+ pcm->channels = ipcm->channels;
+ pcm->tram_start = ipcm->tram_start;
+ pcm->buffer_size = ipcm->buffer_size;
+ pcm->gpr_size = ipcm->gpr_size;
+ pcm->gpr_count = ipcm->gpr_count;
+ pcm->gpr_tmpcount = ipcm->gpr_tmpcount;
+ pcm->gpr_ptr = ipcm->gpr_ptr;
+ pcm->gpr_trigger = ipcm->gpr_trigger;
+ pcm->gpr_running = ipcm->gpr_running;
+ for (i = 0; i < pcm->channels; i++)
+ pcm->etram[i] = ipcm->etram[i];
+ }
+ __error:
+ spin_unlock_irq(&emu->reg_lock);
+ mutex_unlock(&emu->fx8010.lock);
+ return err;
+}
+
+static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_pcm_rec *ipcm)
+{
+ unsigned int i;
+ int err = 0;
+ struct snd_emu10k1_fx8010_pcm *pcm;
+
+ if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
+ return -EINVAL;
+ pcm = &emu->fx8010.pcm[ipcm->substream];
+ mutex_lock(&emu->fx8010.lock);
+ spin_lock_irq(&emu->reg_lock);
+ ipcm->channels = pcm->channels;
+ ipcm->tram_start = pcm->tram_start;
+ ipcm->buffer_size = pcm->buffer_size;
+ ipcm->gpr_size = pcm->gpr_size;
+ ipcm->gpr_ptr = pcm->gpr_ptr;
+ ipcm->gpr_count = pcm->gpr_count;
+ ipcm->gpr_tmpcount = pcm->gpr_tmpcount;
+ ipcm->gpr_trigger = pcm->gpr_trigger;
+ ipcm->gpr_running = pcm->gpr_running;
+ for (i = 0; i < pcm->channels; i++)
+ ipcm->etram[i] = pcm->etram[i];
+ ipcm->res1 = ipcm->res2 = 0;
+ ipcm->pad = 0;
+ spin_unlock_irq(&emu->reg_lock);
+ mutex_unlock(&emu->fx8010.lock);
+ return err;
+}
+
+#define SND_EMU10K1_GPR_CONTROLS 44
+#define SND_EMU10K1_INPUTS 12
+#define SND_EMU10K1_PLAYBACK_CHANNELS 8
+#define SND_EMU10K1_CAPTURE_CHANNELS 4
+
+static void __devinit
+snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
+ const char *name, int gpr, int defval)
+{
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, name);
+ ctl->vcount = ctl->count = 1;
+ ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
+ ctl->min = 0;
+ ctl->max = 100;
+ ctl->tlv = snd_emu10k1_db_scale1;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+}
+
+static void __devinit
+snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
+ const char *name, int gpr, int defval)
+{
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, name);
+ ctl->vcount = ctl->count = 2;
+ ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
+ ctl->gpr[1] = gpr + 1; ctl->value[1] = defval;
+ ctl->min = 0;
+ ctl->max = 100;
+ ctl->tlv = snd_emu10k1_db_scale1;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+}
+
+static void __devinit
+snd_emu10k1_init_mono_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
+ const char *name, int gpr, int defval)
+{
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, name);
+ ctl->vcount = ctl->count = 1;
+ ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
+ ctl->min = 0;
+ ctl->max = 1;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF;
+}
+
+static void __devinit
+snd_emu10k1_init_stereo_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
+ const char *name, int gpr, int defval)
+{
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, name);
+ ctl->vcount = ctl->count = 2;
+ ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
+ ctl->gpr[1] = gpr + 1; ctl->value[1] = defval;
+ ctl->min = 0;
+ ctl->max = 1;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF;
+}
+
+/*
+ * Used for emu1010 - conversion from 32-bit capture inputs from HANA
+ * to 2 x 16-bit registers in audigy - their values are read via DMA.
+ * Conversion is performed by Audigy DSP instructions of FX8010.
+ */
+static int snd_emu10k1_audigy_dsp_convert_32_to_2x16(
+ struct snd_emu10k1_fx8010_code *icode,
+ u32 *ptr, int tmp, int bit_shifter16,
+ int reg_in, int reg_out)
+{
+ A_OP(icode, ptr, iACC3, A_GPR(tmp + 1), reg_in, A_C_00000000, A_C_00000000);
+ A_OP(icode, ptr, iANDXOR, A_GPR(tmp), A_GPR(tmp + 1), A_GPR(bit_shifter16 - 1), A_C_00000000);
+ A_OP(icode, ptr, iTSTNEG, A_GPR(tmp + 2), A_GPR(tmp), A_C_80000000, A_GPR(bit_shifter16 - 2));
+ A_OP(icode, ptr, iANDXOR, A_GPR(tmp + 2), A_GPR(tmp + 2), A_C_80000000, A_C_00000000);
+ A_OP(icode, ptr, iANDXOR, A_GPR(tmp), A_GPR(tmp), A_GPR(bit_shifter16 - 3), A_C_00000000);
+ A_OP(icode, ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A_GPR(tmp), A_C_00010000);
+ A_OP(icode, ptr, iANDXOR, reg_out, A_GPR(tmp), A_C_ffffffff, A_GPR(tmp + 2));
+ A_OP(icode, ptr, iACC3, reg_out + 1, A_GPR(tmp + 1), A_C_00000000, A_C_00000000);
+ return 1;
+}
+
+/*
+ * initial DSP configuration for Audigy
+ */
+
+static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu)
+{
+ int err, i, z, gpr, nctl;
+ int bit_shifter16;
+ const int playback = 10;
+ const int capture = playback + (SND_EMU10K1_PLAYBACK_CHANNELS * 2); /* we reserve 10 voices */
+ const int stereo_mix = capture + 2;
+ const int tmp = 0x88;
+ u32 ptr;
+ struct snd_emu10k1_fx8010_code *icode = NULL;
+ struct snd_emu10k1_fx8010_control_gpr *controls = NULL, *ctl;
+ u32 *gpr_map;
+ mm_segment_t seg;
+
+ if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL ||
+ (icode->gpr_map = (u_int32_t __user *)
+ kcalloc(512 + 256 + 256 + 2 * 1024, sizeof(u_int32_t),
+ GFP_KERNEL)) == NULL ||
+ (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS,
+ sizeof(*controls), GFP_KERNEL)) == NULL) {
+ err = -ENOMEM;
+ goto __err;
+ }
+ gpr_map = (u32 __force *)icode->gpr_map;
+
+ icode->tram_data_map = icode->gpr_map + 512;
+ icode->tram_addr_map = icode->tram_data_map + 256;
+ icode->code = icode->tram_addr_map + 256;
+
+ /* clear free GPRs */
+ for (i = 0; i < 512; i++)
+ set_bit(i, icode->gpr_valid);
+
+ /* clear TRAM data & address lines */
+ for (i = 0; i < 256; i++)
+ set_bit(i, icode->tram_valid);
+
+ strcpy(icode->name, "Audigy DSP code for ALSA");
+ ptr = 0;
+ nctl = 0;
+ gpr = stereo_mix + 10;
+ gpr_map[gpr++] = 0x00007fff;
+ gpr_map[gpr++] = 0x00008000;
+ gpr_map[gpr++] = 0x0000ffff;
+ bit_shifter16 = gpr;
+
+ /* stop FX processor */
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP);
+
+#if 1
+ /* PCM front Playback Volume (independent from stereo mix)
+ * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31)
+ * where gpr contains attenuation from corresponding mixer control
+ * (snd_emu10k1_init_stereo_control)
+ */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_FRONT));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* PCM Surround Playback (independent from stereo mix) */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+2), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_REAR));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_REAR));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Surround Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* PCM Side Playback (independent from stereo mix) */
+ if (emu->card_capabilities->spk71) {
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+6), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_SIDE));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+7), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_SIDE));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Side Playback Volume", gpr, 100);
+ gpr += 2;
+ }
+
+ /* PCM Center Playback (independent from stereo mix) */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+4), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_CENTER));
+ snd_emu10k1_init_mono_control(&controls[nctl++], "PCM Center Playback Volume", gpr, 100);
+ gpr++;
+
+ /* PCM LFE Playback (independent from stereo mix) */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+5), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LFE));
+ snd_emu10k1_init_mono_control(&controls[nctl++], "PCM LFE Playback Volume", gpr, 100);
+ gpr++;
+
+ /*
+ * Stereo Mix
+ */
+ /* Wave (PCM) Playback Volume (will be renamed later) */
+ A_OP(icode, &ptr, iMAC0, A_GPR(stereo_mix), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT));
+ A_OP(icode, &ptr, iMAC0, A_GPR(stereo_mix+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Wave Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* Synth Playback */
+ A_OP(icode, &ptr, iMAC0, A_GPR(stereo_mix+0), A_GPR(stereo_mix+0), A_GPR(gpr), A_FXBUS(FXBUS_MIDI_LEFT));
+ A_OP(icode, &ptr, iMAC0, A_GPR(stereo_mix+1), A_GPR(stereo_mix+1), A_GPR(gpr+1), A_FXBUS(FXBUS_MIDI_RIGHT));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Synth Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* Wave (PCM) Capture */
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT));
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Capture Volume", gpr, 0);
+ gpr += 2;
+
+ /* Synth Capture */
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_FXBUS(FXBUS_MIDI_LEFT));
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_FXBUS(FXBUS_MIDI_RIGHT));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Synth Capture Volume", gpr, 0);
+ gpr += 2;
+
+ /*
+ * inputs
+ */
+#define A_ADD_VOLUME_IN(var,vol,input) \
+A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
+
+ /* emu1212 DSP 0 and DSP 1 Capture */
+ if (emu->card_capabilities->emu_model) {
+ if (emu->card_capabilities->ca0108_chip) {
+ /* Note:JCD:No longer bit shift lower 16bits to upper 16bits of 32bit value. */
+ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x0), A_C_00000001);
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_GPR(tmp));
+ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x1), A_C_00000001);
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr), A_GPR(tmp));
+ } else {
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0));
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1));
+ }
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "EMU Capture Volume", gpr, 0);
+ gpr += 2;
+ }
+ /* AC'97 Playback Volume - used only for mic (renamed later) */
+ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_AC97_L);
+ A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_AC97_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "AMic Playback Volume", gpr, 0);
+ gpr += 2;
+ /* AC'97 Capture Volume - used only for mic */
+ A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_AC97_L);
+ A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_AC97_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Mic Capture Volume", gpr, 0);
+ gpr += 2;
+
+ /* mic capture buffer */
+ A_OP(icode, &ptr, iINTERP, A_EXTOUT(A_EXTOUT_MIC_CAP), A_EXTIN(A_EXTIN_AC97_L), 0xcd, A_EXTIN(A_EXTIN_AC97_R));
+
+ /* Audigy CD Playback Volume */
+ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_SPDIF_CD_L);
+ A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_SPDIF_CD_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++],
+ emu->card_capabilities->ac97_chip ? "Audigy CD Playback Volume" : "CD Playback Volume",
+ gpr, 0);
+ gpr += 2;
+ /* Audigy CD Capture Volume */
+ A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_SPDIF_CD_L);
+ A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_SPDIF_CD_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++],
+ emu->card_capabilities->ac97_chip ? "Audigy CD Capture Volume" : "CD Capture Volume",
+ gpr, 0);
+ gpr += 2;
+
+ /* Optical SPDIF Playback Volume */
+ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_OPT_SPDIF_L);
+ A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_OPT_SPDIF_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++], SNDRV_CTL_NAME_IEC958("Optical ",PLAYBACK,VOLUME), gpr, 0);
+ gpr += 2;
+ /* Optical SPDIF Capture Volume */
+ A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_OPT_SPDIF_L);
+ A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_OPT_SPDIF_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++], SNDRV_CTL_NAME_IEC958("Optical ",CAPTURE,VOLUME), gpr, 0);
+ gpr += 2;
+
+ /* Line2 Playback Volume */
+ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_LINE2_L);
+ A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_LINE2_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++],
+ emu->card_capabilities->ac97_chip ? "Line2 Playback Volume" : "Line Playback Volume",
+ gpr, 0);
+ gpr += 2;
+ /* Line2 Capture Volume */
+ A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_LINE2_L);
+ A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_LINE2_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++],
+ emu->card_capabilities->ac97_chip ? "Line2 Capture Volume" : "Line Capture Volume",
+ gpr, 0);
+ gpr += 2;
+
+ /* Philips ADC Playback Volume */
+ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_ADC_L);
+ A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_ADC_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Analog Mix Playback Volume", gpr, 0);
+ gpr += 2;
+ /* Philips ADC Capture Volume */
+ A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_ADC_L);
+ A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_ADC_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Analog Mix Capture Volume", gpr, 0);
+ gpr += 2;
+
+ /* Aux2 Playback Volume */
+ A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_AUX2_L);
+ A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_AUX2_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++],
+ emu->card_capabilities->ac97_chip ? "Aux2 Playback Volume" : "Aux Playback Volume",
+ gpr, 0);
+ gpr += 2;
+ /* Aux2 Capture Volume */
+ A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_AUX2_L);
+ A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_AUX2_R);
+ snd_emu10k1_init_stereo_control(&controls[nctl++],
+ emu->card_capabilities->ac97_chip ? "Aux2 Capture Volume" : "Aux Capture Volume",
+ gpr, 0);
+ gpr += 2;
+
+ /* Stereo Mix Front Playback Volume */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_GPR(playback), A_GPR(gpr), A_GPR(stereo_mix));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_GPR(playback+1), A_GPR(gpr+1), A_GPR(stereo_mix+1));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Front Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* Stereo Mix Surround Playback */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+2), A_GPR(playback+2), A_GPR(gpr), A_GPR(stereo_mix));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_GPR(playback+3), A_GPR(gpr+1), A_GPR(stereo_mix+1));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Surround Playback Volume", gpr, 0);
+ gpr += 2;
+
+ /* Stereo Mix Center Playback */
+ /* Center = sub = Left/2 + Right/2 */
+ A_OP(icode, &ptr, iINTERP, A_GPR(tmp), A_GPR(stereo_mix), 0xcd, A_GPR(stereo_mix+1));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+4), A_GPR(playback+4), A_GPR(gpr), A_GPR(tmp));
+ snd_emu10k1_init_mono_control(&controls[nctl++], "Center Playback Volume", gpr, 0);
+ gpr++;
+
+ /* Stereo Mix LFE Playback */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+5), A_GPR(playback+5), A_GPR(gpr), A_GPR(tmp));
+ snd_emu10k1_init_mono_control(&controls[nctl++], "LFE Playback Volume", gpr, 0);
+ gpr++;
+
+ if (emu->card_capabilities->spk71) {
+ /* Stereo Mix Side Playback */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+6), A_GPR(playback+6), A_GPR(gpr), A_GPR(stereo_mix));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+7), A_GPR(playback+7), A_GPR(gpr+1), A_GPR(stereo_mix+1));
+ snd_emu10k1_init_stereo_control(&controls[nctl++], "Side Playback Volume", gpr, 0);
+ gpr += 2;
+ }
+
+ /*
+ * outputs
+ */
+#define A_PUT_OUTPUT(out,src) A_OP(icode, &ptr, iACC3, A_EXTOUT(out), A_C_00000000, A_C_00000000, A_GPR(src))
+#define A_PUT_STEREO_OUTPUT(out1,out2,src) \
+ {A_PUT_OUTPUT(out1,src); A_PUT_OUTPUT(out2,src+1);}
+
+#define _A_SWITCH(icode, ptr, dst, src, sw) \
+ A_OP((icode), ptr, iMACINT0, dst, A_C_00000000, src, sw);
+#define A_SWITCH(icode, ptr, dst, src, sw) \
+ _A_SWITCH(icode, ptr, A_GPR(dst), A_GPR(src), A_GPR(sw))
+#define _A_SWITCH_NEG(icode, ptr, dst, src) \
+ A_OP((icode), ptr, iANDXOR, dst, src, A_C_00000001, A_C_00000001);
+#define A_SWITCH_NEG(icode, ptr, dst, src) \
+ _A_SWITCH_NEG(icode, ptr, A_GPR(dst), A_GPR(src))
+
+
+ /*
+ * Process tone control
+ */
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 0), A_GPR(playback + 0), A_C_00000000, A_C_00000000); /* left */
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 1), A_GPR(playback + 1), A_C_00000000, A_C_00000000); /* right */
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 2), A_GPR(playback + 2), A_C_00000000, A_C_00000000); /* rear left */
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 3), A_GPR(playback + 3), A_C_00000000, A_C_00000000); /* rear right */
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 4), A_GPR(playback + 4), A_C_00000000, A_C_00000000); /* center */
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 5), A_GPR(playback + 5), A_C_00000000, A_C_00000000); /* LFE */
+ if (emu->card_capabilities->spk71) {
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 6), A_GPR(playback + 6), A_C_00000000, A_C_00000000); /* side left */
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 7), A_GPR(playback + 7), A_C_00000000, A_C_00000000); /* side right */
+ }
+
+
+ ctl = &controls[nctl + 0];
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, "Tone Control - Bass");
+ ctl->vcount = 2;
+ ctl->count = 10;
+ ctl->min = 0;
+ ctl->max = 40;
+ ctl->value[0] = ctl->value[1] = 20;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_BASS;
+ ctl = &controls[nctl + 1];
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, "Tone Control - Treble");
+ ctl->vcount = 2;
+ ctl->count = 10;
+ ctl->min = 0;
+ ctl->max = 40;
+ ctl->value[0] = ctl->value[1] = 20;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_TREBLE;
+
+#define BASS_GPR 0x8c
+#define TREBLE_GPR 0x96
+
+ for (z = 0; z < 5; z++) {
+ int j;
+ for (j = 0; j < 2; j++) {
+ controls[nctl + 0].gpr[z * 2 + j] = BASS_GPR + z * 2 + j;
+ controls[nctl + 1].gpr[z * 2 + j] = TREBLE_GPR + z * 2 + j;
+ }
+ }
+ for (z = 0; z < 4; z++) { /* front/rear/center-lfe/side */
+ int j, k, l, d;
+ for (j = 0; j < 2; j++) { /* left/right */
+ k = 0xb0 + (z * 8) + (j * 4);
+ l = 0xe0 + (z * 8) + (j * 4);
+ d = playback + SND_EMU10K1_PLAYBACK_CHANNELS + z * 2 + j;
+
+ A_OP(icode, &ptr, iMAC0, A_C_00000000, A_C_00000000, A_GPR(d), A_GPR(BASS_GPR + 0 + j));
+ A_OP(icode, &ptr, iMACMV, A_GPR(k+1), A_GPR(k), A_GPR(k+1), A_GPR(BASS_GPR + 4 + j));
+ A_OP(icode, &ptr, iMACMV, A_GPR(k), A_GPR(d), A_GPR(k), A_GPR(BASS_GPR + 2 + j));
+ A_OP(icode, &ptr, iMACMV, A_GPR(k+3), A_GPR(k+2), A_GPR(k+3), A_GPR(BASS_GPR + 8 + j));
+ A_OP(icode, &ptr, iMAC0, A_GPR(k+2), A_GPR_ACCU, A_GPR(k+2), A_GPR(BASS_GPR + 6 + j));
+ A_OP(icode, &ptr, iACC3, A_GPR(k+2), A_GPR(k+2), A_GPR(k+2), A_C_00000000);
+
+ A_OP(icode, &ptr, iMAC0, A_C_00000000, A_C_00000000, A_GPR(k+2), A_GPR(TREBLE_GPR + 0 + j));
+ A_OP(icode, &ptr, iMACMV, A_GPR(l+1), A_GPR(l), A_GPR(l+1), A_GPR(TREBLE_GPR + 4 + j));
+ A_OP(icode, &ptr, iMACMV, A_GPR(l), A_GPR(k+2), A_GPR(l), A_GPR(TREBLE_GPR + 2 + j));
+ A_OP(icode, &ptr, iMACMV, A_GPR(l+3), A_GPR(l+2), A_GPR(l+3), A_GPR(TREBLE_GPR + 8 + j));
+ A_OP(icode, &ptr, iMAC0, A_GPR(l+2), A_GPR_ACCU, A_GPR(l+2), A_GPR(TREBLE_GPR + 6 + j));
+ A_OP(icode, &ptr, iMACINT0, A_GPR(l+2), A_C_00000000, A_GPR(l+2), A_C_00000010);
+
+ A_OP(icode, &ptr, iACC3, A_GPR(d), A_GPR(l+2), A_C_00000000, A_C_00000000);
+
+ if (z == 2) /* center */
+ break;
+ }
+ }
+ nctl += 2;
+
+#undef BASS_GPR
+#undef TREBLE_GPR
+
+ for (z = 0; z < 8; z++) {
+ A_SWITCH(icode, &ptr, tmp + 0, playback + SND_EMU10K1_PLAYBACK_CHANNELS + z, gpr + 0);
+ A_SWITCH_NEG(icode, &ptr, tmp + 1, gpr + 0);
+ A_SWITCH(icode, &ptr, tmp + 1, playback + z, tmp + 1);
+ A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_GPR(tmp + 0), A_GPR(tmp + 1), A_C_00000000);
+ }
+ snd_emu10k1_init_stereo_onoff_control(controls + nctl++, "Tone Control - Switch", gpr, 0);
+ gpr += 2;
+
+ /* Master volume (will be renamed later) */
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS));
+ A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS));
+ snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0);
+ gpr += 2;
+
+ /* analog speakers */
+ A_PUT_STEREO_OUTPUT(A_EXTOUT_AFRONT_L, A_EXTOUT_AFRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS);
+ A_PUT_STEREO_OUTPUT(A_EXTOUT_AREAR_L, A_EXTOUT_AREAR_R, playback+2 + SND_EMU10K1_PLAYBACK_CHANNELS);
+ A_PUT_OUTPUT(A_EXTOUT_ACENTER, playback+4 + SND_EMU10K1_PLAYBACK_CHANNELS);
+ A_PUT_OUTPUT(A_EXTOUT_ALFE, playback+5 + SND_EMU10K1_PLAYBACK_CHANNELS);
+ if (emu->card_capabilities->spk71)
+ A_PUT_STEREO_OUTPUT(A_EXTOUT_ASIDE_L, A_EXTOUT_ASIDE_R, playback+6 + SND_EMU10K1_PLAYBACK_CHANNELS);
+
+ /* headphone */
+ A_PUT_STEREO_OUTPUT(A_EXTOUT_HEADPHONE_L, A_EXTOUT_HEADPHONE_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS);
+
+ /* digital outputs */
+ /* A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS); */
+ if (emu->card_capabilities->emu_model) {
+ /* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */
+ snd_printk("EMU outputs on\n");
+ for (z = 0; z < 8; z++) {
+ if (emu->card_capabilities->ca0108_chip) {
+ A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ } else {
+ A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ }
+ }
+ }
+
+ /* IEC958 Optical Raw Playback Switch */
+ gpr_map[gpr++] = 0;
+ gpr_map[gpr++] = 0x1008;
+ gpr_map[gpr++] = 0xffff0000;
+ for (z = 0; z < 2; z++) {
+ A_OP(icode, &ptr, iMAC0, A_GPR(tmp + 2), A_FXBUS(FXBUS_PT_LEFT + z), A_C_00000000, A_C_00000000);
+ A_OP(icode, &ptr, iSKIP, A_GPR_COND, A_GPR_COND, A_GPR(gpr - 2), A_C_00000001);
+ A_OP(icode, &ptr, iACC3, A_GPR(tmp + 2), A_C_00000000, A_C_00010000, A_GPR(tmp + 2));
+ A_OP(icode, &ptr, iANDXOR, A_GPR(tmp + 2), A_GPR(tmp + 2), A_GPR(gpr - 1), A_C_00000000);
+ A_SWITCH(icode, &ptr, tmp + 0, tmp + 2, gpr + z);
+ A_SWITCH_NEG(icode, &ptr, tmp + 1, gpr + z);
+ A_SWITCH(icode, &ptr, tmp + 1, playback + SND_EMU10K1_PLAYBACK_CHANNELS + z, tmp + 1);
+ if ((z==1) && (emu->card_capabilities->spdif_bug)) {
+ /* Due to a SPDIF output bug on some Audigy cards, this code delays the Right channel by 1 sample */
+ snd_printk(KERN_INFO "Installing spdif_bug patch: %s\n", emu->card_capabilities->name);
+ A_OP(icode, &ptr, iACC3, A_EXTOUT(A_EXTOUT_FRONT_L + z), A_GPR(gpr - 3), A_C_00000000, A_C_00000000);
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 3), A_GPR(tmp + 0), A_GPR(tmp + 1), A_C_00000000);
+ } else {
+ A_OP(icode, &ptr, iACC3, A_EXTOUT(A_EXTOUT_FRONT_L + z), A_GPR(tmp + 0), A_GPR(tmp + 1), A_C_00000000);
+ }
+ }
+ snd_emu10k1_init_stereo_onoff_control(controls + nctl++, SNDRV_CTL_NAME_IEC958("Optical Raw ",PLAYBACK,SWITCH), gpr, 0);
+ gpr += 2;
+
+ A_PUT_STEREO_OUTPUT(A_EXTOUT_REAR_L, A_EXTOUT_REAR_R, playback+2 + SND_EMU10K1_PLAYBACK_CHANNELS);
+ A_PUT_OUTPUT(A_EXTOUT_CENTER, playback+4 + SND_EMU10K1_PLAYBACK_CHANNELS);
+ A_PUT_OUTPUT(A_EXTOUT_LFE, playback+5 + SND_EMU10K1_PLAYBACK_CHANNELS);
+
+ /* ADC buffer */
+#ifdef EMU10K1_CAPTURE_DIGITAL_OUT
+ A_PUT_STEREO_OUTPUT(A_EXTOUT_ADC_CAP_L, A_EXTOUT_ADC_CAP_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS);
+#else
+ A_PUT_OUTPUT(A_EXTOUT_ADC_CAP_L, capture);
+ A_PUT_OUTPUT(A_EXTOUT_ADC_CAP_R, capture+1);
+#endif
+
+ if (emu->card_capabilities->emu_model) {
+ if (emu->card_capabilities->ca0108_chip) {
+ snd_printk("EMU2 inputs on\n");
+ for (z = 0; z < 0x10; z++) {
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp,
+ bit_shifter16,
+ A3_EMU32IN(z),
+ A_FXBUS2(z*2) );
+ }
+ } else {
+ snd_printk("EMU inputs on\n");
+ /* Capture 16 (originally 8) channels of S32_LE sound */
+
+ /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
+ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
+ /* A_P16VIN(0) is delayed by one sample,
+ * so all other A_P16VIN channels will need to also be delayed
+ */
+ /* Left ADC in. 1 of 2 */
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
+ /* Right ADC in 1 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ /* Delaying by one sample: instead of copying the input
+ * value A_P16VIN to output A_FXBUS2 as in the first channel,
+ * we use an auxiliary register, delaying the value by one
+ * sample
+ */
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000);
+ /* For 96kHz mode */
+ /* Left ADC in. 2 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000);
+ /* Right ADC in 2 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
+ /* Pavel Hofman - we still have voices, A_FXBUS2s, and
+ * A_P16VINs available -
+ * let's add 8 more capture channels - total of 16
+ */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x10));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x12));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x14));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x16));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x18));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1a));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1c));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1e));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
+ A_C_00000000, A_C_00000000);
+ }
+
+#if 0
+ for (z = 4; z < 8; z++) {
+ A_OP(icode, &ptr, iACC3, A_FXBUS2(z), A_C_00000000, A_C_00000000, A_C_00000000);
+ }
+ for (z = 0xc; z < 0x10; z++) {
+ A_OP(icode, &ptr, iACC3, A_FXBUS2(z), A_C_00000000, A_C_00000000, A_C_00000000);
+ }
+#endif
+ } else {
+ /* EFX capture - capture the 16 EXTINs */
+ /* Capture 16 channels of S16_LE sound */
+ for (z = 0; z < 16; z++) {
+ A_OP(icode, &ptr, iACC3, A_FXBUS2(z), A_C_00000000, A_C_00000000, A_EXTIN(z));
+ }
+ }
+
+#endif /* JCD test */
+ /*
+ * ok, set up done..
+ */
+
+ if (gpr > tmp) {
+ snd_BUG();
+ err = -EIO;
+ goto __err;
+ }
+ /* clear remaining instruction memory */
+ while (ptr < 0x400)
+ A_OP(icode, &ptr, 0x0f, 0xc0, 0xc0, 0xcf, 0xc0);
+
+ seg = snd_enter_user();
+ icode->gpr_add_control_count = nctl;
+ icode->gpr_add_controls = (struct snd_emu10k1_fx8010_control_gpr __user *)controls;
+ emu->support_tlv = 1; /* support TLV */
+ err = snd_emu10k1_icode_poke(emu, icode);
+ emu->support_tlv = 0; /* clear again */
+ snd_leave_user(seg);
+
+ __err:
+ kfree(controls);
+ if (icode != NULL) {
+ kfree((void __force *)icode->gpr_map);
+ kfree(icode);
+ }
+ return err;
+}
+
+
+/*
+ * initial DSP configuration for Emu10k1
+ */
+
+/* when volume = max, then copy only to avoid volume modification */
+/* with iMAC0 (negative values) */
+static void __devinit _volume(struct snd_emu10k1_fx8010_code *icode, u32 *ptr, u32 dst, u32 src, u32 vol)
+{
+ OP(icode, ptr, iMAC0, dst, C_00000000, src, vol);
+ OP(icode, ptr, iANDXOR, C_00000000, vol, C_ffffffff, C_7fffffff);
+ OP(icode, ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, C_00000001);
+ OP(icode, ptr, iACC3, dst, src, C_00000000, C_00000000);
+}
+static void __devinit _volume_add(struct snd_emu10k1_fx8010_code *icode, u32 *ptr, u32 dst, u32 src, u32 vol)
+{
+ OP(icode, ptr, iANDXOR, C_00000000, vol, C_ffffffff, C_7fffffff);
+ OP(icode, ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, C_00000002);
+ OP(icode, ptr, iMACINT0, dst, dst, src, C_00000001);
+ OP(icode, ptr, iSKIP, C_00000000, C_7fffffff, C_7fffffff, C_00000001);
+ OP(icode, ptr, iMAC0, dst, dst, src, vol);
+}
+static void __devinit _volume_out(struct snd_emu10k1_fx8010_code *icode, u32 *ptr, u32 dst, u32 src, u32 vol)
+{
+ OP(icode, ptr, iANDXOR, C_00000000, vol, C_ffffffff, C_7fffffff);
+ OP(icode, ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, C_00000002);
+ OP(icode, ptr, iACC3, dst, src, C_00000000, C_00000000);
+ OP(icode, ptr, iSKIP, C_00000000, C_7fffffff, C_7fffffff, C_00000001);
+ OP(icode, ptr, iMAC0, dst, C_00000000, src, vol);
+}
+
+#define VOLUME(icode, ptr, dst, src, vol) \
+ _volume(icode, ptr, GPR(dst), GPR(src), GPR(vol))
+#define VOLUME_IN(icode, ptr, dst, src, vol) \
+ _volume(icode, ptr, GPR(dst), EXTIN(src), GPR(vol))
+#define VOLUME_ADD(icode, ptr, dst, src, vol) \
+ _volume_add(icode, ptr, GPR(dst), GPR(src), GPR(vol))
+#define VOLUME_ADDIN(icode, ptr, dst, src, vol) \
+ _volume_add(icode, ptr, GPR(dst), EXTIN(src), GPR(vol))
+#define VOLUME_OUT(icode, ptr, dst, src, vol) \
+ _volume_out(icode, ptr, EXTOUT(dst), GPR(src), GPR(vol))
+#define _SWITCH(icode, ptr, dst, src, sw) \
+ OP((icode), ptr, iMACINT0, dst, C_00000000, src, sw);
+#define SWITCH(icode, ptr, dst, src, sw) \
+ _SWITCH(icode, ptr, GPR(dst), GPR(src), GPR(sw))
+#define SWITCH_IN(icode, ptr, dst, src, sw) \
+ _SWITCH(icode, ptr, GPR(dst), EXTIN(src), GPR(sw))
+#define _SWITCH_NEG(icode, ptr, dst, src) \
+ OP((icode), ptr, iANDXOR, dst, src, C_00000001, C_00000001);
+#define SWITCH_NEG(icode, ptr, dst, src) \
+ _SWITCH_NEG(icode, ptr, GPR(dst), GPR(src))
+
+
+static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu)
+{
+ int err, i, z, gpr, tmp, playback, capture;
+ u32 ptr;
+ struct snd_emu10k1_fx8010_code *icode;
+ struct snd_emu10k1_fx8010_pcm_rec *ipcm = NULL;
+ struct snd_emu10k1_fx8010_control_gpr *controls = NULL, *ctl;
+ u32 *gpr_map;
+ mm_segment_t seg;
+
+ if ((icode = kzalloc(sizeof(*icode), GFP_KERNEL)) == NULL)
+ return -ENOMEM;
+ if ((icode->gpr_map = (u_int32_t __user *)
+ kcalloc(256 + 160 + 160 + 2 * 512, sizeof(u_int32_t),
+ GFP_KERNEL)) == NULL ||
+ (controls = kcalloc(SND_EMU10K1_GPR_CONTROLS,
+ sizeof(struct snd_emu10k1_fx8010_control_gpr),
+ GFP_KERNEL)) == NULL ||
+ (ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL)) == NULL) {
+ err = -ENOMEM;
+ goto __err;
+ }
+ gpr_map = (u32 __force *)icode->gpr_map;
+
+ icode->tram_data_map = icode->gpr_map + 256;
+ icode->tram_addr_map = icode->tram_data_map + 160;
+ icode->code = icode->tram_addr_map + 160;
+
+ /* clear free GPRs */
+ for (i = 0; i < 256; i++)
+ set_bit(i, icode->gpr_valid);
+
+ /* clear TRAM data & address lines */
+ for (i = 0; i < 160; i++)
+ set_bit(i, icode->tram_valid);
+
+ strcpy(icode->name, "SB Live! FX8010 code for ALSA v1.2 by Jaroslav Kysela");
+ ptr = 0; i = 0;
+ /* we have 12 inputs */
+ playback = SND_EMU10K1_INPUTS;
+ /* we have 6 playback channels and tone control doubles */
+ capture = playback + (SND_EMU10K1_PLAYBACK_CHANNELS * 2);
+ gpr = capture + SND_EMU10K1_CAPTURE_CHANNELS;
+ tmp = 0x88; /* we need 4 temporary GPR */
+ /* from 0x8c to 0xff is the area for tone control */
+
+ /* stop FX processor */
+ snd_emu10k1_ptr_write(emu, DBG, 0, (emu->fx8010.dbg = 0) | EMU10K1_DBG_SINGLE_STEP);
+
+ /*
+ * Process FX Buses
+ */
+ OP(icode, &ptr, iMACINT0, GPR(0), C_00000000, FXBUS(FXBUS_PCM_LEFT), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(1), C_00000000, FXBUS(FXBUS_PCM_RIGHT), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(2), C_00000000, FXBUS(FXBUS_MIDI_LEFT), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(3), C_00000000, FXBUS(FXBUS_MIDI_RIGHT), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(4), C_00000000, FXBUS(FXBUS_PCM_LEFT_REAR), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(5), C_00000000, FXBUS(FXBUS_PCM_RIGHT_REAR), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(6), C_00000000, FXBUS(FXBUS_PCM_CENTER), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(7), C_00000000, FXBUS(FXBUS_PCM_LFE), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(8), C_00000000, C_00000000, C_00000000); /* S/PDIF left */
+ OP(icode, &ptr, iMACINT0, GPR(9), C_00000000, C_00000000, C_00000000); /* S/PDIF right */
+ OP(icode, &ptr, iMACINT0, GPR(10), C_00000000, FXBUS(FXBUS_PCM_LEFT_FRONT), C_00000004);
+ OP(icode, &ptr, iMACINT0, GPR(11), C_00000000, FXBUS(FXBUS_PCM_RIGHT_FRONT), C_00000004);
+
+ /* Raw S/PDIF PCM */
+ ipcm->substream = 0;
+ ipcm->channels = 2;
+ ipcm->tram_start = 0;
+ ipcm->buffer_size = (64 * 1024) / 2;
+ ipcm->gpr_size = gpr++;
+ ipcm->gpr_ptr = gpr++;
+ ipcm->gpr_count = gpr++;
+ ipcm->gpr_tmpcount = gpr++;
+ ipcm->gpr_trigger = gpr++;
+ ipcm->gpr_running = gpr++;
+ ipcm->etram[0] = 0;
+ ipcm->etram[1] = 1;
+
+ gpr_map[gpr + 0] = 0xfffff000;
+ gpr_map[gpr + 1] = 0xffff0000;
+ gpr_map[gpr + 2] = 0x70000000;
+ gpr_map[gpr + 3] = 0x00000007;
+ gpr_map[gpr + 4] = 0x001f << 11;
+ gpr_map[gpr + 5] = 0x001c << 11;
+ gpr_map[gpr + 6] = (0x22 - 0x01) - 1; /* skip at 01 to 22 */
+ gpr_map[gpr + 7] = (0x22 - 0x06) - 1; /* skip at 06 to 22 */
+ gpr_map[gpr + 8] = 0x2000000 + (2<<11);
+ gpr_map[gpr + 9] = 0x4000000 + (2<<11);
+ gpr_map[gpr + 10] = 1<<11;
+ gpr_map[gpr + 11] = (0x24 - 0x0a) - 1; /* skip at 0a to 24 */
+ gpr_map[gpr + 12] = 0;
+
+ /* if the trigger flag is not set, skip */
+ /* 00: */ OP(icode, &ptr, iMAC0, C_00000000, GPR(ipcm->gpr_trigger), C_00000000, C_00000000);
+ /* 01: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_ZERO, GPR(gpr + 6));
+ /* if the running flag is set, we're running */
+ /* 02: */ OP(icode, &ptr, iMAC0, C_00000000, GPR(ipcm->gpr_running), C_00000000, C_00000000);
+ /* 03: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, C_00000004);
+ /* wait until ((GPR_DBAC>>11) & 0x1f) == 0x1c) */
+ /* 04: */ OP(icode, &ptr, iANDXOR, GPR(tmp + 0), GPR_DBAC, GPR(gpr + 4), C_00000000);
+ /* 05: */ OP(icode, &ptr, iMACINT0, C_00000000, GPR(tmp + 0), C_ffffffff, GPR(gpr + 5));
+ /* 06: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, GPR(gpr + 7));
+ /* 07: */ OP(icode, &ptr, iACC3, GPR(gpr + 12), C_00000010, C_00000001, C_00000000);
+
+ /* 08: */ OP(icode, &ptr, iANDXOR, GPR(ipcm->gpr_running), GPR(ipcm->gpr_running), C_00000000, C_00000001);
+ /* 09: */ OP(icode, &ptr, iACC3, GPR(gpr + 12), GPR(gpr + 12), C_ffffffff, C_00000000);
+ /* 0a: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, GPR(gpr + 11));
+ /* 0b: */ OP(icode, &ptr, iACC3, GPR(gpr + 12), C_00000001, C_00000000, C_00000000);
+
+ /* 0c: */ OP(icode, &ptr, iANDXOR, GPR(tmp + 0), ETRAM_DATA(ipcm->etram[0]), GPR(gpr + 0), C_00000000);
+ /* 0d: */ OP(icode, &ptr, iLOG, GPR(tmp + 0), GPR(tmp + 0), GPR(gpr + 3), C_00000000);
+ /* 0e: */ OP(icode, &ptr, iANDXOR, GPR(8), GPR(tmp + 0), GPR(gpr + 1), GPR(gpr + 2));
+ /* 0f: */ OP(icode, &ptr, iSKIP, C_00000000, GPR_COND, CC_REG_MINUS, C_00000001);
+ /* 10: */ OP(icode, &ptr, iANDXOR, GPR(8), GPR(8), GPR(gpr + 1), GPR(gpr + 2));
+
+ /* 11: */ OP(icode, &ptr, iANDXOR, GPR(tmp + 0), ETRAM_DATA(ipcm->etram[1]), GPR(gpr + 0), C_00000000);
+ /* 12: */ OP(icode, &ptr, iLOG, GPR(tmp + 0), GPR(tmp + 0), GPR(gpr + 3), C_00000000);
+ /* 13: */ OP(icode, &ptr, iANDXOR, GPR(9), GPR(tmp + 0), GPR(gpr + 1), GPR(gpr + 2));
+ /* 14: */ OP(icode, &ptr, iSKIP, C_00000000, GPR_COND, CC_REG_MINUS, C_00000001);
+ /* 15: */ OP(icode, &ptr, iANDXOR, GPR(9), GPR(9), GPR(gpr + 1), GPR(gpr + 2));
+
+ /* 16: */ OP(icode, &ptr, iACC3, GPR(tmp + 0), GPR(ipcm->gpr_ptr), C_00000001, C_00000000);
+ /* 17: */ OP(icode, &ptr, iMACINT0, C_00000000, GPR(tmp + 0), C_ffffffff, GPR(ipcm->gpr_size));
+ /* 18: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_MINUS, C_00000001);
+ /* 19: */ OP(icode, &ptr, iACC3, GPR(tmp + 0), C_00000000, C_00000000, C_00000000);
+ /* 1a: */ OP(icode, &ptr, iACC3, GPR(ipcm->gpr_ptr), GPR(tmp + 0), C_00000000, C_00000000);
+
+ /* 1b: */ OP(icode, &ptr, iACC3, GPR(ipcm->gpr_tmpcount), GPR(ipcm->gpr_tmpcount), C_ffffffff, C_00000000);
+ /* 1c: */ OP(icode, &ptr, iSKIP, GPR_COND, GPR_COND, CC_REG_NONZERO, C_00000002);
+ /* 1d: */ OP(icode, &ptr, iACC3, GPR(ipcm->gpr_tmpcount), GPR(ipcm->gpr_count), C_00000000, C_00000000);
+ /* 1e: */ OP(icode, &ptr, iACC3, GPR_IRQ, C_80000000, C_00000000, C_00000000);
+ /* 1f: */ OP(icode, &ptr, iANDXOR, GPR(ipcm->gpr_running), GPR(ipcm->gpr_running), C_00000001, C_00010000);
+
+ /* 20: */ OP(icode, &ptr, iANDXOR, GPR(ipcm->gpr_running), GPR(ipcm->gpr_running), C_00010000, C_00000001);
+ /* 21: */ OP(icode, &ptr, iSKIP, C_00000000, C_7fffffff, C_7fffffff, C_00000002);
+
+ /* 22: */ OP(icode, &ptr, iMACINT1, ETRAM_ADDR(ipcm->etram[0]), GPR(gpr + 8), GPR_DBAC, C_ffffffff);
+ /* 23: */ OP(icode, &ptr, iMACINT1, ETRAM_ADDR(ipcm->etram[1]), GPR(gpr + 9), GPR_DBAC, C_ffffffff);
+
+ /* 24: */
+ gpr += 13;
+
+ /* Wave Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME(icode, &ptr, playback + z, z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Wave Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* Wave Surround Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME(icode, &ptr, playback + 2 + z, z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Wave Surround Playback Volume", gpr, 0);
+ gpr += 2;
+
+ /* Wave Center/LFE Playback Volume */
+ OP(icode, &ptr, iACC3, GPR(tmp + 0), FXBUS(FXBUS_PCM_LEFT), FXBUS(FXBUS_PCM_RIGHT), C_00000000);
+ OP(icode, &ptr, iMACINT0, GPR(tmp + 0), C_00000000, GPR(tmp + 0), C_00000002);
+ VOLUME(icode, &ptr, playback + 4, tmp + 0, gpr);
+ snd_emu10k1_init_mono_control(controls + i++, "Wave Center Playback Volume", gpr++, 0);
+ VOLUME(icode, &ptr, playback + 5, tmp + 0, gpr);
+ snd_emu10k1_init_mono_control(controls + i++, "Wave LFE Playback Volume", gpr++, 0);
+
+ /* Wave Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH(icode, &ptr, tmp + 0, z, gpr + 2 + z);
+ VOLUME(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, "Wave Capture Volume", gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, "Wave Capture Switch", gpr + 2, 0);
+ gpr += 4;
+
+ /* Synth Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADD(icode, &ptr, playback + z, 2 + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Synth Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* Synth Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH(icode, &ptr, tmp + 0, 2 + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, "Synth Capture Volume", gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, "Synth Capture Switch", gpr + 2, 0);
+ gpr += 4;
+
+ /* Surround Digital Playback Volume (renamed later without Digital) */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADD(icode, &ptr, playback + 2 + z, 4 + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Surround Digital Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* Surround Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH(icode, &ptr, tmp + 0, 4 + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, "Surround Capture Volume", gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, "Surround Capture Switch", gpr + 2, 0);
+ gpr += 4;
+
+ /* Center Playback Volume (renamed later without Digital) */
+ VOLUME_ADD(icode, &ptr, playback + 4, 6, gpr);
+ snd_emu10k1_init_mono_control(controls + i++, "Center Digital Playback Volume", gpr++, 100);
+
+ /* LFE Playback Volume + Switch (renamed later without Digital) */
+ VOLUME_ADD(icode, &ptr, playback + 5, 7, gpr);
+ snd_emu10k1_init_mono_control(controls + i++, "LFE Digital Playback Volume", gpr++, 100);
+
+ /* Front Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADD(icode, &ptr, playback + z, 10 + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Front Playback Volume", gpr, 100);
+ gpr += 2;
+
+ /* Front Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH(icode, &ptr, tmp + 0, 10 + z, gpr + 2);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, "Front Capture Volume", gpr, 0);
+ snd_emu10k1_init_mono_onoff_control(controls + i++, "Front Capture Switch", gpr + 2, 0);
+ gpr += 3;
+
+ /*
+ * Process inputs
+ */
+
+ if (emu->fx8010.extin_mask & ((1<<EXTIN_AC97_L)|(1<<EXTIN_AC97_R))) {
+ /* AC'97 Playback Volume */
+ VOLUME_ADDIN(icode, &ptr, playback + 0, EXTIN_AC97_L, gpr); gpr++;
+ VOLUME_ADDIN(icode, &ptr, playback + 1, EXTIN_AC97_R, gpr); gpr++;
+ snd_emu10k1_init_stereo_control(controls + i++, "AC97 Playback Volume", gpr-2, 0);
+ /* AC'97 Capture Volume */
+ VOLUME_ADDIN(icode, &ptr, capture + 0, EXTIN_AC97_L, gpr); gpr++;
+ VOLUME_ADDIN(icode, &ptr, capture + 1, EXTIN_AC97_R, gpr); gpr++;
+ snd_emu10k1_init_stereo_control(controls + i++, "AC97 Capture Volume", gpr-2, 100);
+ }
+
+ if (emu->fx8010.extin_mask & ((1<<EXTIN_SPDIF_CD_L)|(1<<EXTIN_SPDIF_CD_R))) {
+ /* IEC958 TTL Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_SPDIF_CD_L + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",PLAYBACK,VOLUME), gpr, 0);
+ gpr += 2;
+
+ /* IEC958 TTL Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_SPDIF_CD_L + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",CAPTURE,VOLUME), gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("TTL ",CAPTURE,SWITCH), gpr + 2, 0);
+ gpr += 4;
+ }
+
+ if (emu->fx8010.extin_mask & ((1<<EXTIN_ZOOM_L)|(1<<EXTIN_ZOOM_R))) {
+ /* Zoom Video Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_ZOOM_L + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Zoom Video Playback Volume", gpr, 0);
+ gpr += 2;
+
+ /* Zoom Video Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_ZOOM_L + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, "Zoom Video Capture Volume", gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, "Zoom Video Capture Switch", gpr + 2, 0);
+ gpr += 4;
+ }
+
+ if (emu->fx8010.extin_mask & ((1<<EXTIN_TOSLINK_L)|(1<<EXTIN_TOSLINK_R))) {
+ /* IEC958 Optical Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_TOSLINK_L + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",PLAYBACK,VOLUME), gpr, 0);
+ gpr += 2;
+
+ /* IEC958 Optical Capture Volume */
+ for (z = 0; z < 2; z++) {
+ SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_TOSLINK_L + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",CAPTURE,VOLUME), gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("LiveDrive ",CAPTURE,SWITCH), gpr + 2, 0);
+ gpr += 4;
+ }
+
+ if (emu->fx8010.extin_mask & ((1<<EXTIN_LINE1_L)|(1<<EXTIN_LINE1_R))) {
+ /* Line LiveDrive Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_LINE1_L + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Line LiveDrive Playback Volume", gpr, 0);
+ gpr += 2;
+
+ /* Line LiveDrive Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_LINE1_L + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, "Line LiveDrive Capture Volume", gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, "Line LiveDrive Capture Switch", gpr + 2, 0);
+ gpr += 4;
+ }
+
+ if (emu->fx8010.extin_mask & ((1<<EXTIN_COAX_SPDIF_L)|(1<<EXTIN_COAX_SPDIF_R))) {
+ /* IEC958 Coax Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_COAX_SPDIF_L + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",PLAYBACK,VOLUME), gpr, 0);
+ gpr += 2;
+
+ /* IEC958 Coax Capture Volume + Switch */
+ for (z = 0; z < 2; z++) {
+ SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_COAX_SPDIF_L + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",CAPTURE,VOLUME), gpr, 0);
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("Coaxial ",CAPTURE,SWITCH), gpr + 2, 0);
+ gpr += 4;
+ }
+
+ if (emu->fx8010.extin_mask & ((1<<EXTIN_LINE2_L)|(1<<EXTIN_LINE2_R))) {
+ /* Line LiveDrive Playback Volume */
+ for (z = 0; z < 2; z++)
+ VOLUME_ADDIN(icode, &ptr, playback + z, EXTIN_LINE2_L + z, gpr + z);
+ snd_emu10k1_init_stereo_control(controls + i++, "Line2 LiveDrive Playback Volume", gpr, 0);
+ controls[i-1].id.index = 1;
+ gpr += 2;
+
+ /* Line LiveDrive Capture Volume */
+ for (z = 0; z < 2; z++) {
+ SWITCH_IN(icode, &ptr, tmp + 0, EXTIN_LINE2_L + z, gpr + 2 + z);
+ VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+ }
+ snd_emu10k1_init_stereo_control(controls + i++, "Line2 LiveDrive Capture Volume", gpr, 0);
+ controls[i-1].id.index = 1;
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, "Line2 LiveDrive Capture Switch", gpr + 2, 0);
+ controls[i-1].id.index = 1;
+ gpr += 4;
+ }
+
+ /*
+ * Process tone control
+ */
+ OP(icode, &ptr, iACC3, GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 0), GPR(playback + 0), C_00000000, C_00000000); /* left */
+ OP(icode, &ptr, iACC3, GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 1), GPR(playback + 1), C_00000000, C_00000000); /* right */
+ OP(icode, &ptr, iACC3, GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 2), GPR(playback + 2), C_00000000, C_00000000); /* rear left */
+ OP(icode, &ptr, iACC3, GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 3), GPR(playback + 3), C_00000000, C_00000000); /* rear right */
+ OP(icode, &ptr, iACC3, GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 4), GPR(playback + 4), C_00000000, C_00000000); /* center */
+ OP(icode, &ptr, iACC3, GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 5), GPR(playback + 5), C_00000000, C_00000000); /* LFE */
+
+ ctl = &controls[i + 0];
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, "Tone Control - Bass");
+ ctl->vcount = 2;
+ ctl->count = 10;
+ ctl->min = 0;
+ ctl->max = 40;
+ ctl->value[0] = ctl->value[1] = 20;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_BASS;
+ ctl = &controls[i + 1];
+ ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(ctl->id.name, "Tone Control - Treble");
+ ctl->vcount = 2;
+ ctl->count = 10;
+ ctl->min = 0;
+ ctl->max = 40;
+ ctl->value[0] = ctl->value[1] = 20;
+ ctl->translation = EMU10K1_GPR_TRANSLATION_TREBLE;
+
+#define BASS_GPR 0x8c
+#define TREBLE_GPR 0x96
+
+ for (z = 0; z < 5; z++) {
+ int j;
+ for (j = 0; j < 2; j++) {
+ controls[i + 0].gpr[z * 2 + j] = BASS_GPR + z * 2 + j;
+ controls[i + 1].gpr[z * 2 + j] = TREBLE_GPR + z * 2 + j;
+ }
+ }
+ for (z = 0; z < 3; z++) { /* front/rear/center-lfe */
+ int j, k, l, d;
+ for (j = 0; j < 2; j++) { /* left/right */
+ k = 0xa0 + (z * 8) + (j * 4);
+ l = 0xd0 + (z * 8) + (j * 4);
+ d = playback + SND_EMU10K1_PLAYBACK_CHANNELS + z * 2 + j;
+
+ OP(icode, &ptr, iMAC0, C_00000000, C_00000000, GPR(d), GPR(BASS_GPR + 0 + j));
+ OP(icode, &ptr, iMACMV, GPR(k+1), GPR(k), GPR(k+1), GPR(BASS_GPR + 4 + j));
+ OP(icode, &ptr, iMACMV, GPR(k), GPR(d), GPR(k), GPR(BASS_GPR + 2 + j));
+ OP(icode, &ptr, iMACMV, GPR(k+3), GPR(k+2), GPR(k+3), GPR(BASS_GPR + 8 + j));
+ OP(icode, &ptr, iMAC0, GPR(k+2), GPR_ACCU, GPR(k+2), GPR(BASS_GPR + 6 + j));
+ OP(icode, &ptr, iACC3, GPR(k+2), GPR(k+2), GPR(k+2), C_00000000);
+
+ OP(icode, &ptr, iMAC0, C_00000000, C_00000000, GPR(k+2), GPR(TREBLE_GPR + 0 + j));
+ OP(icode, &ptr, iMACMV, GPR(l+1), GPR(l), GPR(l+1), GPR(TREBLE_GPR + 4 + j));
+ OP(icode, &ptr, iMACMV, GPR(l), GPR(k+2), GPR(l), GPR(TREBLE_GPR + 2 + j));
+ OP(icode, &ptr, iMACMV, GPR(l+3), GPR(l+2), GPR(l+3), GPR(TREBLE_GPR + 8 + j));
+ OP(icode, &ptr, iMAC0, GPR(l+2), GPR_ACCU, GPR(l+2), GPR(TREBLE_GPR + 6 + j));
+ OP(icode, &ptr, iMACINT0, GPR(l+2), C_00000000, GPR(l+2), C_00000010);
+
+ OP(icode, &ptr, iACC3, GPR(d), GPR(l+2), C_00000000, C_00000000);
+
+ if (z == 2) /* center */
+ break;
+ }
+ }
+ i += 2;
+
+#undef BASS_GPR
+#undef TREBLE_GPR
+
+ for (z = 0; z < 6; z++) {
+ SWITCH(icode, &ptr, tmp + 0, playback + SND_EMU10K1_PLAYBACK_CHANNELS + z, gpr + 0);
+ SWITCH_NEG(icode, &ptr, tmp + 1, gpr + 0);
+ SWITCH(icode, &ptr, tmp + 1, playback + z, tmp + 1);
+ OP(icode, &ptr, iACC3, GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), GPR(tmp + 0), GPR(tmp + 1), C_00000000);
+ }
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, "Tone Control - Switch", gpr, 0);
+ gpr += 2;
+
+ /*
+ * Process outputs
+ */
+ if (emu->fx8010.extout_mask & ((1<<EXTOUT_AC97_L)|(1<<EXTOUT_AC97_R))) {
+ /* AC'97 Playback Volume */
+
+ for (z = 0; z < 2; z++)
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_AC97_L + z), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), C_00000000, C_00000000);
+ }
+
+ if (emu->fx8010.extout_mask & ((1<<EXTOUT_TOSLINK_L)|(1<<EXTOUT_TOSLINK_R))) {
+ /* IEC958 Optical Raw Playback Switch */
+
+ for (z = 0; z < 2; z++) {
+ SWITCH(icode, &ptr, tmp + 0, 8 + z, gpr + z);
+ SWITCH_NEG(icode, &ptr, tmp + 1, gpr + z);
+ SWITCH(icode, &ptr, tmp + 1, playback + SND_EMU10K1_PLAYBACK_CHANNELS + z, tmp + 1);
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_TOSLINK_L + z), GPR(tmp + 0), GPR(tmp + 1), C_00000000);
+#ifdef EMU10K1_CAPTURE_DIGITAL_OUT
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ADC_CAP_L + z), GPR(tmp + 0), GPR(tmp + 1), C_00000000);
+#endif
+ }
+
+ snd_emu10k1_init_stereo_onoff_control(controls + i++, SNDRV_CTL_NAME_IEC958("Optical Raw ",PLAYBACK,SWITCH), gpr, 0);
+ gpr += 2;
+ }
+
+ if (emu->fx8010.extout_mask & ((1<<EXTOUT_HEADPHONE_L)|(1<<EXTOUT_HEADPHONE_R))) {
+ /* Headphone Playback Volume */
+
+ for (z = 0; z < 2; z++) {
+ SWITCH(icode, &ptr, tmp + 0, playback + SND_EMU10K1_PLAYBACK_CHANNELS + 4 + z, gpr + 2 + z);
+ SWITCH_NEG(icode, &ptr, tmp + 1, gpr + 2 + z);
+ SWITCH(icode, &ptr, tmp + 1, playback + SND_EMU10K1_PLAYBACK_CHANNELS + z, tmp + 1);
+ OP(icode, &ptr, iACC3, GPR(tmp + 0), GPR(tmp + 0), GPR(tmp + 1), C_00000000);
+ VOLUME_OUT(icode, &ptr, EXTOUT_HEADPHONE_L + z, tmp + 0, gpr + z);
+ }
+
+ snd_emu10k1_init_stereo_control(controls + i++, "Headphone Playback Volume", gpr + 0, 0);
+ controls[i-1].id.index = 1; /* AC'97 can have also Headphone control */
+ snd_emu10k1_init_mono_onoff_control(controls + i++, "Headphone Center Playback Switch", gpr + 2, 0);
+ controls[i-1].id.index = 1;
+ snd_emu10k1_init_mono_onoff_control(controls + i++, "Headphone LFE Playback Switch", gpr + 3, 0);
+ controls[i-1].id.index = 1;
+
+ gpr += 4;
+ }
+
+ if (emu->fx8010.extout_mask & ((1<<EXTOUT_REAR_L)|(1<<EXTOUT_REAR_R)))
+ for (z = 0; z < 2; z++)
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_REAR_L + z), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 2 + z), C_00000000, C_00000000);
+
+ if (emu->fx8010.extout_mask & ((1<<EXTOUT_AC97_REAR_L)|(1<<EXTOUT_AC97_REAR_R)))
+ for (z = 0; z < 2; z++)
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_AC97_REAR_L + z), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 2 + z), C_00000000, C_00000000);
+
+ if (emu->fx8010.extout_mask & (1<<EXTOUT_AC97_CENTER)) {
+#ifndef EMU10K1_CENTER_LFE_FROM_FRONT
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_AC97_CENTER), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 4), C_00000000, C_00000000);
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ACENTER), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 4), C_00000000, C_00000000);
+#else
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_AC97_CENTER), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 0), C_00000000, C_00000000);
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ACENTER), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 0), C_00000000, C_00000000);
+#endif
+ }
+
+ if (emu->fx8010.extout_mask & (1<<EXTOUT_AC97_LFE)) {
+#ifndef EMU10K1_CENTER_LFE_FROM_FRONT
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_AC97_LFE), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 5), C_00000000, C_00000000);
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ALFE), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 5), C_00000000, C_00000000);
+#else
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_AC97_LFE), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 1), C_00000000, C_00000000);
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ALFE), GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 1), C_00000000, C_00000000);
+#endif
+ }
+
+#ifndef EMU10K1_CAPTURE_DIGITAL_OUT
+ for (z = 0; z < 2; z++)
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_ADC_CAP_L + z), GPR(capture + z), C_00000000, C_00000000);
+#endif
+
+ if (emu->fx8010.extout_mask & (1<<EXTOUT_MIC_CAP))
+ OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_MIC_CAP), GPR(capture + 2), C_00000000, C_00000000);
+
+ /* EFX capture - capture the 16 EXTINS */
+ if (emu->card_capabilities->sblive51) {
+ /* On the Live! 5.1, FXBUS2(1) and FXBUS(2) are shared with EXTOUT_ACENTER
+ * and EXTOUT_ALFE, so we can't connect inputs to them for multitrack recording.
+ *
+ * Since only 14 of the 16 EXTINs are used, this is not a big problem.
+ * We route AC97L and R to FX capture 14 and 15, SPDIF CD in to FX capture
+ * 0 and 3, then the rest of the EXTINs to the corresponding FX capture
+ * channel. Multitrack recorders will still see the center/lfe output signal
+ * on the second and third channels.
+ */
+ OP(icode, &ptr, iACC3, FXBUS2(14), C_00000000, C_00000000, EXTIN(0));
+ OP(icode, &ptr, iACC3, FXBUS2(15), C_00000000, C_00000000, EXTIN(1));
+ OP(icode, &ptr, iACC3, FXBUS2(0), C_00000000, C_00000000, EXTIN(2));
+ OP(icode, &ptr, iACC3, FXBUS2(3), C_00000000, C_00000000, EXTIN(3));
+ for (z = 4; z < 14; z++)
+ OP(icode, &ptr, iACC3, FXBUS2(z), C_00000000, C_00000000, EXTIN(z));
+ } else {
+ for (z = 0; z < 16; z++)
+ OP(icode, &ptr, iACC3, FXBUS2(z), C_00000000, C_00000000, EXTIN(z));
+ }
+
+
+ if (gpr > tmp) {
+ snd_BUG();
+ err = -EIO;
+ goto __err;
+ }
+ if (i > SND_EMU10K1_GPR_CONTROLS) {
+ snd_BUG();
+ err = -EIO;
+ goto __err;
+ }
+
+ /* clear remaining instruction memory */
+ while (ptr < 0x200)
+ OP(icode, &ptr, iACC3, C_00000000, C_00000000, C_00000000, C_00000000);
+
+ if ((err = snd_emu10k1_fx8010_tram_setup(emu, ipcm->buffer_size)) < 0)
+ goto __err;
+ seg = snd_enter_user();
+ icode->gpr_add_control_count = i;
+ icode->gpr_add_controls = (struct snd_emu10k1_fx8010_control_gpr __user *)controls;
+ emu->support_tlv = 1; /* support TLV */
+ err = snd_emu10k1_icode_poke(emu, icode);
+ emu->support_tlv = 0; /* clear again */
+ snd_leave_user(seg);
+ if (err >= 0)
+ err = snd_emu10k1_ipcm_poke(emu, ipcm);
+ __err:
+ kfree(ipcm);
+ kfree(controls);
+ if (icode != NULL) {
+ kfree((void __force *)icode->gpr_map);
+ kfree(icode);
+ }
+ return err;
+}
+
+int __devinit snd_emu10k1_init_efx(struct snd_emu10k1 *emu)
+{
+ spin_lock_init(&emu->fx8010.irq_lock);
+ INIT_LIST_HEAD(&emu->fx8010.gpr_ctl);
+ if (emu->audigy)
+ return _snd_emu10k1_audigy_init_efx(emu);
+ else
+ return _snd_emu10k1_init_efx(emu);
+}
+
+void snd_emu10k1_free_efx(struct snd_emu10k1 *emu)
+{
+ /* stop processor */
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg = A_DBG_SINGLE_STEP);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg = EMU10K1_DBG_SINGLE_STEP);
+}
+
+#if 0 /* FIXME: who use them? */
+int snd_emu10k1_fx8010_tone_control_activate(struct snd_emu10k1 *emu, int output)
+{
+ if (output < 0 || output >= 6)
+ return -EINVAL;
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + 0x94 + output, 0, 1);
+ return 0;
+}
+
+int snd_emu10k1_fx8010_tone_control_deactivate(struct snd_emu10k1 *emu, int output)
+{
+ if (output < 0 || output >= 6)
+ return -EINVAL;
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + 0x94 + output, 0, 0);
+ return 0;
+}
+#endif
+
+int snd_emu10k1_fx8010_tram_setup(struct snd_emu10k1 *emu, u32 size)
+{
+ u8 size_reg = 0;
+
+ /* size is in samples */
+ if (size != 0) {
+ size = (size - 1) >> 13;
+
+ while (size) {
+ size >>= 1;
+ size_reg++;
+ }
+ size = 0x2000 << size_reg;
+ }
+ if ((emu->fx8010.etram_pages.bytes / 2) == size)
+ return 0;
+ spin_lock_irq(&emu->emu_lock);
+ outl(HCFG_LOCKTANKCACHE_MASK | inl(emu->port + HCFG), emu->port + HCFG);
+ spin_unlock_irq(&emu->emu_lock);
+ snd_emu10k1_ptr_write(emu, TCB, 0, 0);
+ snd_emu10k1_ptr_write(emu, TCBS, 0, 0);
+ if (emu->fx8010.etram_pages.area != NULL) {
+ snd_dma_free_pages(&emu->fx8010.etram_pages);
+ emu->fx8010.etram_pages.area = NULL;
+ emu->fx8010.etram_pages.bytes = 0;
+ }
+
+ if (size > 0) {
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci),
+ size * 2, &emu->fx8010.etram_pages) < 0)
+ return -ENOMEM;
+ memset(emu->fx8010.etram_pages.area, 0, size * 2);
+ snd_emu10k1_ptr_write(emu, TCB, 0, emu->fx8010.etram_pages.addr);
+ snd_emu10k1_ptr_write(emu, TCBS, 0, size_reg);
+ spin_lock_irq(&emu->emu_lock);
+ outl(inl(emu->port + HCFG) & ~HCFG_LOCKTANKCACHE_MASK, emu->port + HCFG);
+ spin_unlock_irq(&emu->emu_lock);
+ }
+
+ return 0;
+}
+
+static int snd_emu10k1_fx8010_open(struct snd_hwdep * hw, struct file *file)
+{
+ return 0;
+}
+
+static void copy_string(char *dst, char *src, char *null, int idx)
+{
+ if (src == NULL)
+ sprintf(dst, "%s %02X", null, idx);
+ else
+ strcpy(dst, src);
+}
+
+static void snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_fx8010_info *info)
+{
+ char **fxbus, **extin, **extout;
+ unsigned short fxbus_mask, extin_mask, extout_mask;
+ int res;
+
+ info->internal_tram_size = emu->fx8010.itram_size;
+ info->external_tram_size = emu->fx8010.etram_pages.bytes / 2;
+ fxbus = fxbuses;
+ extin = emu->audigy ? audigy_ins : creative_ins;
+ extout = emu->audigy ? audigy_outs : creative_outs;
+ fxbus_mask = emu->fx8010.fxbus_mask;
+ extin_mask = emu->fx8010.extin_mask;
+ extout_mask = emu->fx8010.extout_mask;
+ for (res = 0; res < 16; res++, fxbus++, extin++, extout++) {
+ copy_string(info->fxbus_names[res], fxbus_mask & (1 << res) ? *fxbus : NULL, "FXBUS", res);
+ copy_string(info->extin_names[res], extin_mask & (1 << res) ? *extin : NULL, "Unused", res);
+ copy_string(info->extout_names[res], extout_mask & (1 << res) ? *extout : NULL, "Unused", res);
+ }
+ for (res = 16; res < 32; res++, extout++)
+ copy_string(info->extout_names[res], extout_mask & (1 << res) ? *extout : NULL, "Unused", res);
+ info->gpr_controls = emu->fx8010.gpr_count;
+}
+
+static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct snd_emu10k1 *emu = hw->private_data;
+ struct snd_emu10k1_fx8010_info *info;
+ struct snd_emu10k1_fx8010_code *icode;
+ struct snd_emu10k1_fx8010_pcm_rec *ipcm;
+ unsigned int addr;
+ void __user *argp = (void __user *)arg;
+ int res;
+
+ switch (cmd) {
+ case SNDRV_EMU10K1_IOCTL_PVERSION:
+ emu->support_tlv = 1;
+ return put_user(SNDRV_EMU10K1_VERSION, (int __user *)argp);
+ case SNDRV_EMU10K1_IOCTL_INFO:
+ info = kmalloc(sizeof(*info), GFP_KERNEL);
+ if (!info)
+ return -ENOMEM;
+ snd_emu10k1_fx8010_info(emu, info);
+ if (copy_to_user(argp, info, sizeof(*info))) {
+ kfree(info);
+ return -EFAULT;
+ }
+ kfree(info);
+ return 0;
+ case SNDRV_EMU10K1_IOCTL_CODE_POKE:
+ if (!capable(CAP_SYS_ADMIN))
+ return -EPERM;
+ icode = kmalloc(sizeof(*icode), GFP_KERNEL);
+ if (icode == NULL)
+ return -ENOMEM;
+ if (copy_from_user(icode, argp, sizeof(*icode))) {
+ kfree(icode);
+ return -EFAULT;
+ }
+ res = snd_emu10k1_icode_poke(emu, icode);
+ kfree(icode);
+ return res;
+ case SNDRV_EMU10K1_IOCTL_CODE_PEEK:
+ icode = kmalloc(sizeof(*icode), GFP_KERNEL);
+ if (icode == NULL)
+ return -ENOMEM;
+ if (copy_from_user(icode, argp, sizeof(*icode))) {
+ kfree(icode);
+ return -EFAULT;
+ }
+ res = snd_emu10k1_icode_peek(emu, icode);
+ if (res == 0 && copy_to_user(argp, icode, sizeof(*icode))) {
+ kfree(icode);
+ return -EFAULT;
+ }
+ kfree(icode);
+ return res;
+ case SNDRV_EMU10K1_IOCTL_PCM_POKE:
+ ipcm = kmalloc(sizeof(*ipcm), GFP_KERNEL);
+ if (ipcm == NULL)
+ return -ENOMEM;
+ if (copy_from_user(ipcm, argp, sizeof(*ipcm))) {
+ kfree(ipcm);
+ return -EFAULT;
+ }
+ res = snd_emu10k1_ipcm_poke(emu, ipcm);
+ kfree(ipcm);
+ return res;
+ case SNDRV_EMU10K1_IOCTL_PCM_PEEK:
+ ipcm = kzalloc(sizeof(*ipcm), GFP_KERNEL);
+ if (ipcm == NULL)
+ return -ENOMEM;
+ if (copy_from_user(ipcm, argp, sizeof(*ipcm))) {
+ kfree(ipcm);
+ return -EFAULT;
+ }
+ res = snd_emu10k1_ipcm_peek(emu, ipcm);
+ if (res == 0 && copy_to_user(argp, ipcm, sizeof(*ipcm))) {
+ kfree(ipcm);
+ return -EFAULT;
+ }
+ kfree(ipcm);
+ return res;
+ case SNDRV_EMU10K1_IOCTL_TRAM_SETUP:
+ if (!capable(CAP_SYS_ADMIN))
+ return -EPERM;
+ if (get_user(addr, (unsigned int __user *)argp))
+ return -EFAULT;
+ mutex_lock(&emu->fx8010.lock);
+ res = snd_emu10k1_fx8010_tram_setup(emu, addr);
+ mutex_unlock(&emu->fx8010.lock);
+ return res;
+ case SNDRV_EMU10K1_IOCTL_STOP:
+ if (!capable(CAP_SYS_ADMIN))
+ return -EPERM;
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg |= A_DBG_SINGLE_STEP);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg |= EMU10K1_DBG_SINGLE_STEP);
+ return 0;
+ case SNDRV_EMU10K1_IOCTL_CONTINUE:
+ if (!capable(CAP_SYS_ADMIN))
+ return -EPERM;
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg = 0);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg = 0);
+ return 0;
+ case SNDRV_EMU10K1_IOCTL_ZERO_TRAM_COUNTER:
+ if (!capable(CAP_SYS_ADMIN))
+ return -EPERM;
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg | A_DBG_ZC);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg | EMU10K1_DBG_ZC);
+ udelay(10);
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg);
+ return 0;
+ case SNDRV_EMU10K1_IOCTL_SINGLE_STEP:
+ if (!capable(CAP_SYS_ADMIN))
+ return -EPERM;
+ if (get_user(addr, (unsigned int __user *)argp))
+ return -EFAULT;
+ if (addr > 0x1ff)
+ return -EINVAL;
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg |= A_DBG_SINGLE_STEP | addr);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg |= EMU10K1_DBG_SINGLE_STEP | addr);
+ udelay(10);
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg |= A_DBG_SINGLE_STEP | A_DBG_STEP_ADDR | addr);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg |= EMU10K1_DBG_SINGLE_STEP | EMU10K1_DBG_STEP | addr);
+ return 0;
+ case SNDRV_EMU10K1_IOCTL_DBG_READ:
+ if (emu->audigy)
+ addr = snd_emu10k1_ptr_read(emu, A_DBG, 0);
+ else
+ addr = snd_emu10k1_ptr_read(emu, DBG, 0);
+ if (put_user(addr, (unsigned int __user *)argp))
+ return -EFAULT;
+ return 0;
+ }
+ return -ENOTTY;
+}
+
+static int snd_emu10k1_fx8010_release(struct snd_hwdep * hw, struct file *file)
+{
+ return 0;
+}
+
+int __devinit snd_emu10k1_fx8010_new(struct snd_emu10k1 *emu, int device, struct snd_hwdep ** rhwdep)
+{
+ struct snd_hwdep *hw;
+ int err;
+
+ if (rhwdep)
+ *rhwdep = NULL;
+ if ((err = snd_hwdep_new(emu->card, "FX8010", device, &hw)) < 0)
+ return err;
+ strcpy(hw->name, "EMU10K1 (FX8010)");
+ hw->iface = SNDRV_HWDEP_IFACE_EMU10K1;
+ hw->ops.open = snd_emu10k1_fx8010_open;
+ hw->ops.ioctl = snd_emu10k1_fx8010_ioctl;
+ hw->ops.release = snd_emu10k1_fx8010_release;
+ hw->private_data = emu;
+ if (rhwdep)
+ *rhwdep = hw;
+ return 0;
+}
+
+#ifdef CONFIG_PM
+int __devinit snd_emu10k1_efx_alloc_pm_buffer(struct snd_emu10k1 *emu)
+{
+ int len;
+
+ len = emu->audigy ? 0x200 : 0x100;
+ emu->saved_gpr = kmalloc(len * 4, GFP_KERNEL);
+ if (! emu->saved_gpr)
+ return -ENOMEM;
+ len = emu->audigy ? 0x100 : 0xa0;
+ emu->tram_val_saved = kmalloc(len * 4, GFP_KERNEL);
+ emu->tram_addr_saved = kmalloc(len * 4, GFP_KERNEL);
+ if (! emu->tram_val_saved || ! emu->tram_addr_saved)
+ return -ENOMEM;
+ len = emu->audigy ? 2 * 1024 : 2 * 512;
+ emu->saved_icode = vmalloc(len * 4);
+ if (! emu->saved_icode)
+ return -ENOMEM;
+ return 0;
+}
+
+void snd_emu10k1_efx_free_pm_buffer(struct snd_emu10k1 *emu)
+{
+ kfree(emu->saved_gpr);
+ kfree(emu->tram_val_saved);
+ kfree(emu->tram_addr_saved);
+ vfree(emu->saved_icode);
+}
+
+/*
+ * save/restore GPR, TRAM and codes
+ */
+void snd_emu10k1_efx_suspend(struct snd_emu10k1 *emu)
+{
+ int i, len;
+
+ len = emu->audigy ? 0x200 : 0x100;
+ for (i = 0; i < len; i++)
+ emu->saved_gpr[i] = snd_emu10k1_ptr_read(emu, emu->gpr_base + i, 0);
+
+ len = emu->audigy ? 0x100 : 0xa0;
+ for (i = 0; i < len; i++) {
+ emu->tram_val_saved[i] = snd_emu10k1_ptr_read(emu, TANKMEMDATAREGBASE + i, 0);
+ emu->tram_addr_saved[i] = snd_emu10k1_ptr_read(emu, TANKMEMADDRREGBASE + i, 0);
+ if (emu->audigy) {
+ emu->tram_addr_saved[i] >>= 12;
+ emu->tram_addr_saved[i] |=
+ snd_emu10k1_ptr_read(emu, A_TANKMEMCTLREGBASE + i, 0) << 20;
+ }
+ }
+
+ len = emu->audigy ? 2 * 1024 : 2 * 512;
+ for (i = 0; i < len; i++)
+ emu->saved_icode[i] = snd_emu10k1_efx_read(emu, i);
+}
+
+void snd_emu10k1_efx_resume(struct snd_emu10k1 *emu)
+{
+ int i, len;
+
+ /* set up TRAM */
+ if (emu->fx8010.etram_pages.bytes > 0) {
+ unsigned size, size_reg = 0;
+ size = emu->fx8010.etram_pages.bytes / 2;
+ size = (size - 1) >> 13;
+ while (size) {
+ size >>= 1;
+ size_reg++;
+ }
+ outl(HCFG_LOCKTANKCACHE_MASK | inl(emu->port + HCFG), emu->port + HCFG);
+ snd_emu10k1_ptr_write(emu, TCB, 0, emu->fx8010.etram_pages.addr);
+ snd_emu10k1_ptr_write(emu, TCBS, 0, size_reg);
+ outl(inl(emu->port + HCFG) & ~HCFG_LOCKTANKCACHE_MASK, emu->port + HCFG);
+ }
+
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg | A_DBG_SINGLE_STEP);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg | EMU10K1_DBG_SINGLE_STEP);
+
+ len = emu->audigy ? 0x200 : 0x100;
+ for (i = 0; i < len; i++)
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + i, 0, emu->saved_gpr[i]);
+
+ len = emu->audigy ? 0x100 : 0xa0;
+ for (i = 0; i < len; i++) {
+ snd_emu10k1_ptr_write(emu, TANKMEMDATAREGBASE + i, 0,
+ emu->tram_val_saved[i]);
+ if (! emu->audigy)
+ snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + i, 0,
+ emu->tram_addr_saved[i]);
+ else {
+ snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + i, 0,
+ emu->tram_addr_saved[i] << 12);
+ snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + i, 0,
+ emu->tram_addr_saved[i] >> 20);
+ }
+ }
+
+ len = emu->audigy ? 2 * 1024 : 2 * 512;
+ for (i = 0; i < len; i++)
+ snd_emu10k1_efx_write(emu, i, emu->saved_icode[i]);
+
+ /* start FX processor when the DSP code is updated */
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, A_DBG, 0, emu->fx8010.dbg);
+ else
+ snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg);
+}
+#endif
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
new file mode 100644
index 0000000..f34bbfb
--- /dev/null
+++ b/sound/pci/emu10k1/emumixer.c
@@ -0,0 +1,2091 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Takashi Iwai <tiwai@suse.de>
+ * Creative Labs, Inc.
+ * Routines for control of EMU10K1 chips / mixer routines
+ * Multichannel PCM support Copyright (c) Lee Revell <rlrevell@joe-job.com>
+ *
+ * Copyright (c) by James Courtier-Dutton <James@superbug.co.uk>
+ * Added EMU 1010 support.
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+#include <linux/delay.h>
+#include <sound/tlv.h>
+
+#include "p17v.h"
+
+#define AC97_ID_STAC9758 0x83847658
+
+static const DECLARE_TLV_DB_SCALE(snd_audigy_db_scale2, -10350, 50, 1); /* WM8775 gain scale */
+
+static int snd_emu10k1_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_emu10k1_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned long flags;
+
+ /* Limit: emu->spdif_bits */
+ if (idx >= 3)
+ return -EINVAL;
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff;
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+/*
+ * Items labels in enum mixer controls assigning source data to
+ * each destination
+ */
+static char *emu1010_src_texts[] = {
+ "Silence",
+ "Dock Mic A",
+ "Dock Mic B",
+ "Dock ADC1 Left",
+ "Dock ADC1 Right",
+ "Dock ADC2 Left",
+ "Dock ADC2 Right",
+ "Dock ADC3 Left",
+ "Dock ADC3 Right",
+ "0202 ADC Left",
+ "0202 ADC Right",
+ "0202 SPDIF Left",
+ "0202 SPDIF Right",
+ "ADAT 0",
+ "ADAT 1",
+ "ADAT 2",
+ "ADAT 3",
+ "ADAT 4",
+ "ADAT 5",
+ "ADAT 6",
+ "ADAT 7",
+ "DSP 0",
+ "DSP 1",
+ "DSP 2",
+ "DSP 3",
+ "DSP 4",
+ "DSP 5",
+ "DSP 6",
+ "DSP 7",
+ "DSP 8",
+ "DSP 9",
+ "DSP 10",
+ "DSP 11",
+ "DSP 12",
+ "DSP 13",
+ "DSP 14",
+ "DSP 15",
+ "DSP 16",
+ "DSP 17",
+ "DSP 18",
+ "DSP 19",
+ "DSP 20",
+ "DSP 21",
+ "DSP 22",
+ "DSP 23",
+ "DSP 24",
+ "DSP 25",
+ "DSP 26",
+ "DSP 27",
+ "DSP 28",
+ "DSP 29",
+ "DSP 30",
+ "DSP 31",
+};
+
+/* 1616(m) cardbus */
+
+static char *emu1616_src_texts[] = {
+ "Silence",
+ "Dock Mic A",
+ "Dock Mic B",
+ "Dock ADC1 Left",
+ "Dock ADC1 Right",
+ "Dock ADC2 Left",
+ "Dock ADC2 Right",
+ "Dock SPDIF Left",
+ "Dock SPDIF Right",
+ "ADAT 0",
+ "ADAT 1",
+ "ADAT 2",
+ "ADAT 3",
+ "ADAT 4",
+ "ADAT 5",
+ "ADAT 6",
+ "ADAT 7",
+ "DSP 0",
+ "DSP 1",
+ "DSP 2",
+ "DSP 3",
+ "DSP 4",
+ "DSP 5",
+ "DSP 6",
+ "DSP 7",
+ "DSP 8",
+ "DSP 9",
+ "DSP 10",
+ "DSP 11",
+ "DSP 12",
+ "DSP 13",
+ "DSP 14",
+ "DSP 15",
+ "DSP 16",
+ "DSP 17",
+ "DSP 18",
+ "DSP 19",
+ "DSP 20",
+ "DSP 21",
+ "DSP 22",
+ "DSP 23",
+ "DSP 24",
+ "DSP 25",
+ "DSP 26",
+ "DSP 27",
+ "DSP 28",
+ "DSP 29",
+ "DSP 30",
+ "DSP 31",
+};
+
+
+/*
+ * List of data sources available for each destination
+ */
+static unsigned int emu1010_src_regs[] = {
+ EMU_SRC_SILENCE,/* 0 */
+ EMU_SRC_DOCK_MIC_A1, /* 1 */
+ EMU_SRC_DOCK_MIC_B1, /* 2 */
+ EMU_SRC_DOCK_ADC1_LEFT1, /* 3 */
+ EMU_SRC_DOCK_ADC1_RIGHT1, /* 4 */
+ EMU_SRC_DOCK_ADC2_LEFT1, /* 5 */
+ EMU_SRC_DOCK_ADC2_RIGHT1, /* 6 */
+ EMU_SRC_DOCK_ADC3_LEFT1, /* 7 */
+ EMU_SRC_DOCK_ADC3_RIGHT1, /* 8 */
+ EMU_SRC_HAMOA_ADC_LEFT1, /* 9 */
+ EMU_SRC_HAMOA_ADC_RIGHT1, /* 10 */
+ EMU_SRC_HANA_SPDIF_LEFT1, /* 11 */
+ EMU_SRC_HANA_SPDIF_RIGHT1, /* 12 */
+ EMU_SRC_HANA_ADAT, /* 13 */
+ EMU_SRC_HANA_ADAT+1, /* 14 */
+ EMU_SRC_HANA_ADAT+2, /* 15 */
+ EMU_SRC_HANA_ADAT+3, /* 16 */
+ EMU_SRC_HANA_ADAT+4, /* 17 */
+ EMU_SRC_HANA_ADAT+5, /* 18 */
+ EMU_SRC_HANA_ADAT+6, /* 19 */
+ EMU_SRC_HANA_ADAT+7, /* 20 */
+ EMU_SRC_ALICE_EMU32A, /* 21 */
+ EMU_SRC_ALICE_EMU32A+1, /* 22 */
+ EMU_SRC_ALICE_EMU32A+2, /* 23 */
+ EMU_SRC_ALICE_EMU32A+3, /* 24 */
+ EMU_SRC_ALICE_EMU32A+4, /* 25 */
+ EMU_SRC_ALICE_EMU32A+5, /* 26 */
+ EMU_SRC_ALICE_EMU32A+6, /* 27 */
+ EMU_SRC_ALICE_EMU32A+7, /* 28 */
+ EMU_SRC_ALICE_EMU32A+8, /* 29 */
+ EMU_SRC_ALICE_EMU32A+9, /* 30 */
+ EMU_SRC_ALICE_EMU32A+0xa, /* 31 */
+ EMU_SRC_ALICE_EMU32A+0xb, /* 32 */
+ EMU_SRC_ALICE_EMU32A+0xc, /* 33 */
+ EMU_SRC_ALICE_EMU32A+0xd, /* 34 */
+ EMU_SRC_ALICE_EMU32A+0xe, /* 35 */
+ EMU_SRC_ALICE_EMU32A+0xf, /* 36 */
+ EMU_SRC_ALICE_EMU32B, /* 37 */
+ EMU_SRC_ALICE_EMU32B+1, /* 38 */
+ EMU_SRC_ALICE_EMU32B+2, /* 39 */
+ EMU_SRC_ALICE_EMU32B+3, /* 40 */
+ EMU_SRC_ALICE_EMU32B+4, /* 41 */
+ EMU_SRC_ALICE_EMU32B+5, /* 42 */
+ EMU_SRC_ALICE_EMU32B+6, /* 43 */
+ EMU_SRC_ALICE_EMU32B+7, /* 44 */
+ EMU_SRC_ALICE_EMU32B+8, /* 45 */
+ EMU_SRC_ALICE_EMU32B+9, /* 46 */
+ EMU_SRC_ALICE_EMU32B+0xa, /* 47 */
+ EMU_SRC_ALICE_EMU32B+0xb, /* 48 */
+ EMU_SRC_ALICE_EMU32B+0xc, /* 49 */
+ EMU_SRC_ALICE_EMU32B+0xd, /* 50 */
+ EMU_SRC_ALICE_EMU32B+0xe, /* 51 */
+ EMU_SRC_ALICE_EMU32B+0xf, /* 52 */
+};
+
+/* 1616(m) cardbus */
+static unsigned int emu1616_src_regs[] = {
+ EMU_SRC_SILENCE,
+ EMU_SRC_DOCK_MIC_A1,
+ EMU_SRC_DOCK_MIC_B1,
+ EMU_SRC_DOCK_ADC1_LEFT1,
+ EMU_SRC_DOCK_ADC1_RIGHT1,
+ EMU_SRC_DOCK_ADC2_LEFT1,
+ EMU_SRC_DOCK_ADC2_RIGHT1,
+ EMU_SRC_MDOCK_SPDIF_LEFT1,
+ EMU_SRC_MDOCK_SPDIF_RIGHT1,
+ EMU_SRC_MDOCK_ADAT,
+ EMU_SRC_MDOCK_ADAT+1,
+ EMU_SRC_MDOCK_ADAT+2,
+ EMU_SRC_MDOCK_ADAT+3,
+ EMU_SRC_MDOCK_ADAT+4,
+ EMU_SRC_MDOCK_ADAT+5,
+ EMU_SRC_MDOCK_ADAT+6,
+ EMU_SRC_MDOCK_ADAT+7,
+ EMU_SRC_ALICE_EMU32A,
+ EMU_SRC_ALICE_EMU32A+1,
+ EMU_SRC_ALICE_EMU32A+2,
+ EMU_SRC_ALICE_EMU32A+3,
+ EMU_SRC_ALICE_EMU32A+4,
+ EMU_SRC_ALICE_EMU32A+5,
+ EMU_SRC_ALICE_EMU32A+6,
+ EMU_SRC_ALICE_EMU32A+7,
+ EMU_SRC_ALICE_EMU32A+8,
+ EMU_SRC_ALICE_EMU32A+9,
+ EMU_SRC_ALICE_EMU32A+0xa,
+ EMU_SRC_ALICE_EMU32A+0xb,
+ EMU_SRC_ALICE_EMU32A+0xc,
+ EMU_SRC_ALICE_EMU32A+0xd,
+ EMU_SRC_ALICE_EMU32A+0xe,
+ EMU_SRC_ALICE_EMU32A+0xf,
+ EMU_SRC_ALICE_EMU32B,
+ EMU_SRC_ALICE_EMU32B+1,
+ EMU_SRC_ALICE_EMU32B+2,
+ EMU_SRC_ALICE_EMU32B+3,
+ EMU_SRC_ALICE_EMU32B+4,
+ EMU_SRC_ALICE_EMU32B+5,
+ EMU_SRC_ALICE_EMU32B+6,
+ EMU_SRC_ALICE_EMU32B+7,
+ EMU_SRC_ALICE_EMU32B+8,
+ EMU_SRC_ALICE_EMU32B+9,
+ EMU_SRC_ALICE_EMU32B+0xa,
+ EMU_SRC_ALICE_EMU32B+0xb,
+ EMU_SRC_ALICE_EMU32B+0xc,
+ EMU_SRC_ALICE_EMU32B+0xd,
+ EMU_SRC_ALICE_EMU32B+0xe,
+ EMU_SRC_ALICE_EMU32B+0xf,
+};
+
+/*
+ * Data destinations - physical EMU outputs.
+ * Each destination has an enum mixer control to choose a data source
+ */
+static unsigned int emu1010_output_dst[] = {
+ EMU_DST_DOCK_DAC1_LEFT1, /* 0 */
+ EMU_DST_DOCK_DAC1_RIGHT1, /* 1 */
+ EMU_DST_DOCK_DAC2_LEFT1, /* 2 */
+ EMU_DST_DOCK_DAC2_RIGHT1, /* 3 */
+ EMU_DST_DOCK_DAC3_LEFT1, /* 4 */
+ EMU_DST_DOCK_DAC3_RIGHT1, /* 5 */
+ EMU_DST_DOCK_DAC4_LEFT1, /* 6 */
+ EMU_DST_DOCK_DAC4_RIGHT1, /* 7 */
+ EMU_DST_DOCK_PHONES_LEFT1, /* 8 */
+ EMU_DST_DOCK_PHONES_RIGHT1, /* 9 */
+ EMU_DST_DOCK_SPDIF_LEFT1, /* 10 */
+ EMU_DST_DOCK_SPDIF_RIGHT1, /* 11 */
+ EMU_DST_HANA_SPDIF_LEFT1, /* 12 */
+ EMU_DST_HANA_SPDIF_RIGHT1, /* 13 */
+ EMU_DST_HAMOA_DAC_LEFT1, /* 14 */
+ EMU_DST_HAMOA_DAC_RIGHT1, /* 15 */
+ EMU_DST_HANA_ADAT, /* 16 */
+ EMU_DST_HANA_ADAT+1, /* 17 */
+ EMU_DST_HANA_ADAT+2, /* 18 */
+ EMU_DST_HANA_ADAT+3, /* 19 */
+ EMU_DST_HANA_ADAT+4, /* 20 */
+ EMU_DST_HANA_ADAT+5, /* 21 */
+ EMU_DST_HANA_ADAT+6, /* 22 */
+ EMU_DST_HANA_ADAT+7, /* 23 */
+};
+
+/* 1616(m) cardbus */
+static unsigned int emu1616_output_dst[] = {
+ EMU_DST_DOCK_DAC1_LEFT1,
+ EMU_DST_DOCK_DAC1_RIGHT1,
+ EMU_DST_DOCK_DAC2_LEFT1,
+ EMU_DST_DOCK_DAC2_RIGHT1,
+ EMU_DST_DOCK_DAC3_LEFT1,
+ EMU_DST_DOCK_DAC3_RIGHT1,
+ EMU_DST_MDOCK_SPDIF_LEFT1,
+ EMU_DST_MDOCK_SPDIF_RIGHT1,
+ EMU_DST_MDOCK_ADAT,
+ EMU_DST_MDOCK_ADAT+1,
+ EMU_DST_MDOCK_ADAT+2,
+ EMU_DST_MDOCK_ADAT+3,
+ EMU_DST_MDOCK_ADAT+4,
+ EMU_DST_MDOCK_ADAT+5,
+ EMU_DST_MDOCK_ADAT+6,
+ EMU_DST_MDOCK_ADAT+7,
+ EMU_DST_MANA_DAC_LEFT,
+ EMU_DST_MANA_DAC_RIGHT,
+};
+
+/*
+ * Data destinations - HANA outputs going to Alice2 (audigy) for
+ * capture (EMU32 + I2S links)
+ * Each destination has an enum mixer control to choose a data source
+ */
+static unsigned int emu1010_input_dst[] = {
+ EMU_DST_ALICE2_EMU32_0,
+ EMU_DST_ALICE2_EMU32_1,
+ EMU_DST_ALICE2_EMU32_2,
+ EMU_DST_ALICE2_EMU32_3,
+ EMU_DST_ALICE2_EMU32_4,
+ EMU_DST_ALICE2_EMU32_5,
+ EMU_DST_ALICE2_EMU32_6,
+ EMU_DST_ALICE2_EMU32_7,
+ EMU_DST_ALICE2_EMU32_8,
+ EMU_DST_ALICE2_EMU32_9,
+ EMU_DST_ALICE2_EMU32_A,
+ EMU_DST_ALICE2_EMU32_B,
+ EMU_DST_ALICE2_EMU32_C,
+ EMU_DST_ALICE2_EMU32_D,
+ EMU_DST_ALICE2_EMU32_E,
+ EMU_DST_ALICE2_EMU32_F,
+ EMU_DST_ALICE_I2S0_LEFT,
+ EMU_DST_ALICE_I2S0_RIGHT,
+ EMU_DST_ALICE_I2S1_LEFT,
+ EMU_DST_ALICE_I2S1_RIGHT,
+ EMU_DST_ALICE_I2S2_LEFT,
+ EMU_DST_ALICE_I2S2_RIGHT,
+};
+
+static int snd_emu1010_input_output_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ char **items;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) {
+ uinfo->value.enumerated.items = 49;
+ items = emu1616_src_texts;
+ } else {
+ uinfo->value.enumerated.items = 53;
+ items = emu1010_src_texts;
+ }
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ items[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_emu1010_output_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int channel;
+
+ channel = (kcontrol->private_value) & 0xff;
+ /* Limit: emu1010_output_dst, emu->emu1010.output_source */
+ if (channel >= 24 ||
+ (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 &&
+ channel >= 18))
+ return -EINVAL;
+ ucontrol->value.enumerated.item[0] = emu->emu1010.output_source[channel];
+ return 0;
+}
+
+static int snd_emu1010_output_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ unsigned int channel;
+
+ val = ucontrol->value.enumerated.item[0];
+ if (val >= 53 ||
+ (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 &&
+ val >= 49))
+ return -EINVAL;
+ channel = (kcontrol->private_value) & 0xff;
+ /* Limit: emu1010_output_dst, emu->emu1010.output_source */
+ if (channel >= 24 ||
+ (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 &&
+ channel >= 18))
+ return -EINVAL;
+ if (emu->emu1010.output_source[channel] == val)
+ return 0;
+ emu->emu1010.output_source[channel] = val;
+ if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616)
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ emu1616_output_dst[channel], emu1616_src_regs[val]);
+ else
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ emu1010_output_dst[channel], emu1010_src_regs[val]);
+ return 1;
+}
+
+static int snd_emu1010_input_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int channel;
+
+ channel = (kcontrol->private_value) & 0xff;
+ /* Limit: emu1010_input_dst, emu->emu1010.input_source */
+ if (channel >= 22)
+ return -EINVAL;
+ ucontrol->value.enumerated.item[0] = emu->emu1010.input_source[channel];
+ return 0;
+}
+
+static int snd_emu1010_input_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ unsigned int channel;
+
+ val = ucontrol->value.enumerated.item[0];
+ if (val >= 53 ||
+ (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616 &&
+ val >= 49))
+ return -EINVAL;
+ channel = (kcontrol->private_value) & 0xff;
+ /* Limit: emu1010_input_dst, emu->emu1010.input_source */
+ if (channel >= 22)
+ return -EINVAL;
+ if (emu->emu1010.input_source[channel] == val)
+ return 0;
+ emu->emu1010.input_source[channel] = val;
+ if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616)
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ emu1010_input_dst[channel], emu1616_src_regs[val]);
+ else
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ emu1010_input_dst[channel], emu1010_src_regs[val]);
+ return 1;
+}
+
+#define EMU1010_SOURCE_OUTPUT(xname,chid) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = snd_emu1010_input_output_source_info, \
+ .get = snd_emu1010_output_source_get, \
+ .put = snd_emu1010_output_source_put, \
+ .private_value = chid \
+}
+
+static struct snd_kcontrol_new snd_emu1010_output_enum_ctls[] __devinitdata = {
+ EMU1010_SOURCE_OUTPUT("Dock DAC1 Left Playback Enum", 0),
+ EMU1010_SOURCE_OUTPUT("Dock DAC1 Right Playback Enum", 1),
+ EMU1010_SOURCE_OUTPUT("Dock DAC2 Left Playback Enum", 2),
+ EMU1010_SOURCE_OUTPUT("Dock DAC2 Right Playback Enum", 3),
+ EMU1010_SOURCE_OUTPUT("Dock DAC3 Left Playback Enum", 4),
+ EMU1010_SOURCE_OUTPUT("Dock DAC3 Right Playback Enum", 5),
+ EMU1010_SOURCE_OUTPUT("Dock DAC4 Left Playback Enum", 6),
+ EMU1010_SOURCE_OUTPUT("Dock DAC4 Right Playback Enum", 7),
+ EMU1010_SOURCE_OUTPUT("Dock Phones Left Playback Enum", 8),
+ EMU1010_SOURCE_OUTPUT("Dock Phones Right Playback Enum", 9),
+ EMU1010_SOURCE_OUTPUT("Dock SPDIF Left Playback Enum", 0xa),
+ EMU1010_SOURCE_OUTPUT("Dock SPDIF Right Playback Enum", 0xb),
+ EMU1010_SOURCE_OUTPUT("1010 SPDIF Left Playback Enum", 0xc),
+ EMU1010_SOURCE_OUTPUT("1010 SPDIF Right Playback Enum", 0xd),
+ EMU1010_SOURCE_OUTPUT("0202 DAC Left Playback Enum", 0xe),
+ EMU1010_SOURCE_OUTPUT("0202 DAC Right Playback Enum", 0xf),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 0 Playback Enum", 0x10),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 1 Playback Enum", 0x11),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 2 Playback Enum", 0x12),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 3 Playback Enum", 0x13),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 4 Playback Enum", 0x14),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 5 Playback Enum", 0x15),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 6 Playback Enum", 0x16),
+ EMU1010_SOURCE_OUTPUT("1010 ADAT 7 Playback Enum", 0x17),
+};
+
+
+/* 1616(m) cardbus */
+static struct snd_kcontrol_new snd_emu1616_output_enum_ctls[] __devinitdata = {
+ EMU1010_SOURCE_OUTPUT("Dock DAC1 Left Playback Enum", 0),
+ EMU1010_SOURCE_OUTPUT("Dock DAC1 Right Playback Enum", 1),
+ EMU1010_SOURCE_OUTPUT("Dock DAC2 Left Playback Enum", 2),
+ EMU1010_SOURCE_OUTPUT("Dock DAC2 Right Playback Enum", 3),
+ EMU1010_SOURCE_OUTPUT("Dock DAC3 Left Playback Enum", 4),
+ EMU1010_SOURCE_OUTPUT("Dock DAC3 Right Playback Enum", 5),
+ EMU1010_SOURCE_OUTPUT("Dock SPDIF Left Playback Enum", 6),
+ EMU1010_SOURCE_OUTPUT("Dock SPDIF Right Playback Enum", 7),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 0 Playback Enum", 8),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 1 Playback Enum", 9),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 2 Playback Enum", 0xa),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 3 Playback Enum", 0xb),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 4 Playback Enum", 0xc),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 5 Playback Enum", 0xd),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 6 Playback Enum", 0xe),
+ EMU1010_SOURCE_OUTPUT("Dock ADAT 7 Playback Enum", 0xf),
+ EMU1010_SOURCE_OUTPUT("Mana DAC Left Playback Enum", 0x10),
+ EMU1010_SOURCE_OUTPUT("Mana DAC Right Playback Enum", 0x11),
+};
+
+
+#define EMU1010_SOURCE_INPUT(xname,chid) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = snd_emu1010_input_output_source_info, \
+ .get = snd_emu1010_input_source_get, \
+ .put = snd_emu1010_input_source_put, \
+ .private_value = chid \
+}
+
+static struct snd_kcontrol_new snd_emu1010_input_enum_ctls[] __devinitdata = {
+ EMU1010_SOURCE_INPUT("DSP 0 Capture Enum", 0),
+ EMU1010_SOURCE_INPUT("DSP 1 Capture Enum", 1),
+ EMU1010_SOURCE_INPUT("DSP 2 Capture Enum", 2),
+ EMU1010_SOURCE_INPUT("DSP 3 Capture Enum", 3),
+ EMU1010_SOURCE_INPUT("DSP 4 Capture Enum", 4),
+ EMU1010_SOURCE_INPUT("DSP 5 Capture Enum", 5),
+ EMU1010_SOURCE_INPUT("DSP 6 Capture Enum", 6),
+ EMU1010_SOURCE_INPUT("DSP 7 Capture Enum", 7),
+ EMU1010_SOURCE_INPUT("DSP 8 Capture Enum", 8),
+ EMU1010_SOURCE_INPUT("DSP 9 Capture Enum", 9),
+ EMU1010_SOURCE_INPUT("DSP A Capture Enum", 0xa),
+ EMU1010_SOURCE_INPUT("DSP B Capture Enum", 0xb),
+ EMU1010_SOURCE_INPUT("DSP C Capture Enum", 0xc),
+ EMU1010_SOURCE_INPUT("DSP D Capture Enum", 0xd),
+ EMU1010_SOURCE_INPUT("DSP E Capture Enum", 0xe),
+ EMU1010_SOURCE_INPUT("DSP F Capture Enum", 0xf),
+ EMU1010_SOURCE_INPUT("DSP 10 Capture Enum", 0x10),
+ EMU1010_SOURCE_INPUT("DSP 11 Capture Enum", 0x11),
+ EMU1010_SOURCE_INPUT("DSP 12 Capture Enum", 0x12),
+ EMU1010_SOURCE_INPUT("DSP 13 Capture Enum", 0x13),
+ EMU1010_SOURCE_INPUT("DSP 14 Capture Enum", 0x14),
+ EMU1010_SOURCE_INPUT("DSP 15 Capture Enum", 0x15),
+};
+
+
+
+#define snd_emu1010_adc_pads_info snd_ctl_boolean_mono_info
+
+static int snd_emu1010_adc_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int mask = kcontrol->private_value & 0xff;
+ ucontrol->value.integer.value[0] = (emu->emu1010.adc_pads & mask) ? 1 : 0;
+ return 0;
+}
+
+static int snd_emu1010_adc_pads_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int mask = kcontrol->private_value & 0xff;
+ unsigned int val, cache;
+ val = ucontrol->value.integer.value[0];
+ cache = emu->emu1010.adc_pads;
+ if (val == 1)
+ cache = cache | mask;
+ else
+ cache = cache & ~mask;
+ if (cache != emu->emu1010.adc_pads) {
+ snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, cache );
+ emu->emu1010.adc_pads = cache;
+ }
+
+ return 0;
+}
+
+
+
+#define EMU1010_ADC_PADS(xname,chid) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = snd_emu1010_adc_pads_info, \
+ .get = snd_emu1010_adc_pads_get, \
+ .put = snd_emu1010_adc_pads_put, \
+ .private_value = chid \
+}
+
+static struct snd_kcontrol_new snd_emu1010_adc_pads[] __devinitdata = {
+ EMU1010_ADC_PADS("ADC1 14dB PAD Audio Dock Capture Switch", EMU_HANA_DOCK_ADC_PAD1),
+ EMU1010_ADC_PADS("ADC2 14dB PAD Audio Dock Capture Switch", EMU_HANA_DOCK_ADC_PAD2),
+ EMU1010_ADC_PADS("ADC3 14dB PAD Audio Dock Capture Switch", EMU_HANA_DOCK_ADC_PAD3),
+ EMU1010_ADC_PADS("ADC1 14dB PAD 0202 Capture Switch", EMU_HANA_0202_ADC_PAD1),
+};
+
+#define snd_emu1010_dac_pads_info snd_ctl_boolean_mono_info
+
+static int snd_emu1010_dac_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int mask = kcontrol->private_value & 0xff;
+ ucontrol->value.integer.value[0] = (emu->emu1010.dac_pads & mask) ? 1 : 0;
+ return 0;
+}
+
+static int snd_emu1010_dac_pads_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int mask = kcontrol->private_value & 0xff;
+ unsigned int val, cache;
+ val = ucontrol->value.integer.value[0];
+ cache = emu->emu1010.dac_pads;
+ if (val == 1)
+ cache = cache | mask;
+ else
+ cache = cache & ~mask;
+ if (cache != emu->emu1010.dac_pads) {
+ snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, cache );
+ emu->emu1010.dac_pads = cache;
+ }
+
+ return 0;
+}
+
+
+
+#define EMU1010_DAC_PADS(xname,chid) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = snd_emu1010_dac_pads_info, \
+ .get = snd_emu1010_dac_pads_get, \
+ .put = snd_emu1010_dac_pads_put, \
+ .private_value = chid \
+}
+
+static struct snd_kcontrol_new snd_emu1010_dac_pads[] __devinitdata = {
+ EMU1010_DAC_PADS("DAC1 Audio Dock 14dB PAD Playback Switch", EMU_HANA_DOCK_DAC_PAD1),
+ EMU1010_DAC_PADS("DAC2 Audio Dock 14dB PAD Playback Switch", EMU_HANA_DOCK_DAC_PAD2),
+ EMU1010_DAC_PADS("DAC3 Audio Dock 14dB PAD Playback Switch", EMU_HANA_DOCK_DAC_PAD3),
+ EMU1010_DAC_PADS("DAC4 Audio Dock 14dB PAD Playback Switch", EMU_HANA_DOCK_DAC_PAD4),
+ EMU1010_DAC_PADS("DAC1 0202 14dB PAD Playback Switch", EMU_HANA_0202_DAC_PAD1),
+};
+
+
+static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {
+ "44100", "48000", "SPDIF", "ADAT"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+
+
+}
+
+static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->emu1010.internal_clock;
+ return 0;
+}
+
+static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ /* Limit: uinfo->value.enumerated.items = 4; */
+ if (val >= 4)
+ return -EINVAL;
+ change = (emu->emu1010.internal_clock != val);
+ if (change) {
+ emu->emu1010.internal_clock = val;
+ switch (val) {
+ case 0:
+ /* 44100 */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_44_1K );
+ /* Word Clock source, Internal 44.1kHz x1 */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_INT_44_1K | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2,
+ EMU_HANA_DOCK_LEDS_2_44K | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ break;
+ case 1:
+ /* 48000 */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K );
+ /* Word Clock source, Internal 48kHz x1 */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2,
+ EMU_HANA_DOCK_LEDS_2_48K | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ break;
+
+ case 2: /* Take clock from S/PDIF IN */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K );
+ /* Word Clock source, sync to S/PDIF input */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_HANA_SPDIF_IN | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2,
+ EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ break;
+
+ case 3:
+ /* Take clock from ADAT IN */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K );
+ /* Word Clock source, sync to ADAT input */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_HANA_ADAT_IN | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+
+
+ break;
+ }
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu1010_internal_clock =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Clock Internal Rate",
+ .count = 1,
+ .info = snd_emu1010_internal_clock_info,
+ .get = snd_emu1010_internal_clock_get,
+ .put = snd_emu1010_internal_clock_put
+};
+
+static int snd_audigy_i2c_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+#if 0
+ static char *texts[4] = {
+ "Unknown1", "Unknown2", "Mic", "Line"
+ };
+#endif
+ static char *texts[2] = {
+ "Mic", "Line"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_audigy_i2c_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->i2c_capture_source;
+ return 0;
+}
+
+static int snd_audigy_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int source_id;
+ unsigned int ngain, ogain;
+ u32 gpio;
+ int change = 0;
+ unsigned long flags;
+ u32 source;
+ /* If the capture source has changed,
+ * update the capture volume from the cached value
+ * for the particular source.
+ */
+ source_id = ucontrol->value.enumerated.item[0];
+ /* Limit: uinfo->value.enumerated.items = 2; */
+ /* emu->i2c_capture_volume */
+ if (source_id >= 2)
+ return -EINVAL;
+ change = (emu->i2c_capture_source != source_id);
+ if (change) {
+ snd_emu10k1_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ gpio = inl(emu->port + A_IOCFG);
+ if (source_id==0)
+ outl(gpio | 0x4, emu->port + A_IOCFG);
+ else
+ outl(gpio & ~0x4, emu->port + A_IOCFG);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+
+ ngain = emu->i2c_capture_volume[source_id][0]; /* Left */
+ ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
+ if (ngain != ogain)
+ snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff));
+ ngain = emu->i2c_capture_volume[source_id][1]; /* Right */
+ ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Right */
+ if (ngain != ogain)
+ snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+
+ source = 1 << (source_id + 2);
+ snd_emu10k1_i2c_write(emu, ADC_MUX, source); /* Set source */
+ emu->i2c_capture_source = source_id;
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new snd_audigy_i2c_capture_source =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_audigy_i2c_capture_source_info,
+ .get = snd_audigy_i2c_capture_source_get,
+ .put = snd_audigy_i2c_capture_source_put
+};
+
+static int snd_audigy_i2c_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_audigy_i2c_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int source_id;
+
+ source_id = kcontrol->private_value;
+ /* Limit: emu->i2c_capture_volume */
+ /* capture_source: uinfo->value.enumerated.items = 2 */
+ if (source_id >= 2)
+ return -EINVAL;
+
+ ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0];
+ ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1];
+ return 0;
+}
+
+static int snd_audigy_i2c_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int ogain;
+ unsigned int ngain;
+ unsigned int source_id;
+ int change = 0;
+
+ source_id = kcontrol->private_value;
+ /* Limit: emu->i2c_capture_volume */
+ /* capture_source: uinfo->value.enumerated.items = 2 */
+ if (source_id >= 2)
+ return -EINVAL;
+ ogain = emu->i2c_capture_volume[source_id][0]; /* Left */
+ ngain = ucontrol->value.integer.value[0];
+ if (ngain > 0xff)
+ return 0;
+ if (ogain != ngain) {
+ if (emu->i2c_capture_source == source_id)
+ snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) );
+ emu->i2c_capture_volume[source_id][0] = ngain;
+ change = 1;
+ }
+ ogain = emu->i2c_capture_volume[source_id][1]; /* Right */
+ ngain = ucontrol->value.integer.value[1];
+ if (ngain > 0xff)
+ return 0;
+ if (ogain != ngain) {
+ if (emu->i2c_capture_source == source_id)
+ snd_emu10k1_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+ emu->i2c_capture_volume[source_id][1] = ngain;
+ change = 1;
+ }
+
+ return change;
+}
+
+#define I2C_VOLUME(xname,chid) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = snd_audigy_i2c_volume_info, \
+ .get = snd_audigy_i2c_volume_get, \
+ .put = snd_audigy_i2c_volume_put, \
+ .tlv = { .p = snd_audigy_db_scale2 }, \
+ .private_value = chid \
+}
+
+
+static struct snd_kcontrol_new snd_audigy_i2c_volume_ctls[] __devinitdata = {
+ I2C_VOLUME("Mic Capture Volume", 0),
+ I2C_VOLUME("Line Capture Volume", 0)
+};
+
+#if 0
+static int snd_audigy_spdif_output_rate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"44100", "48000", "96000"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_audigy_spdif_output_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int tmp;
+ unsigned long flags;
+
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0);
+ switch (tmp & A_SPDIF_RATE_MASK) {
+ case A_SPDIF_44100:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ case A_SPDIF_48000:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ case A_SPDIF_96000:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 1;
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_audigy_spdif_output_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int reg, val, tmp;
+ unsigned long flags;
+
+ switch(ucontrol->value.enumerated.item[0]) {
+ case 0:
+ val = A_SPDIF_44100;
+ break;
+ case 1:
+ val = A_SPDIF_48000;
+ break;
+ case 2:
+ val = A_SPDIF_96000;
+ break;
+ default:
+ val = A_SPDIF_48000;
+ break;
+ }
+
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ reg = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0);
+ tmp = reg & ~A_SPDIF_RATE_MASK;
+ tmp |= val;
+ if ((change = (tmp != reg)))
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp);
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_audigy_spdif_output_rate =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Audigy SPDIF Output Sample Rate",
+ .count = 1,
+ .info = snd_audigy_spdif_output_rate_info,
+ .get = snd_audigy_spdif_output_rate_get,
+ .put = snd_audigy_spdif_output_rate_put
+};
+#endif
+
+static int snd_emu10k1_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ int change;
+ unsigned int val;
+ unsigned long flags;
+
+ /* Limit: emu->spdif_bits */
+ if (idx >= 3)
+ return -EINVAL;
+ val = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ change = val != emu->spdif_bits[idx];
+ if (change) {
+ snd_emu10k1_ptr_write(emu, SPCS0 + idx, 0, val);
+ emu->spdif_bits[idx] = val;
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_spdif_mask_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .count = 3,
+ .info = snd_emu10k1_spdif_info,
+ .get = snd_emu10k1_spdif_get_mask
+};
+
+static struct snd_kcontrol_new snd_emu10k1_spdif_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .count = 3,
+ .info = snd_emu10k1_spdif_info,
+ .get = snd_emu10k1_spdif_get,
+ .put = snd_emu10k1_spdif_put
+};
+
+
+static void update_emu10k1_fxrt(struct snd_emu10k1 *emu, int voice, unsigned char *route)
+{
+ if (emu->audigy) {
+ snd_emu10k1_ptr_write(emu, A_FXRT1, voice,
+ snd_emu10k1_compose_audigy_fxrt1(route));
+ snd_emu10k1_ptr_write(emu, A_FXRT2, voice,
+ snd_emu10k1_compose_audigy_fxrt2(route));
+ } else {
+ snd_emu10k1_ptr_write(emu, FXRT, voice,
+ snd_emu10k1_compose_send_routing(route));
+ }
+}
+
+static void update_emu10k1_send_volume(struct snd_emu10k1 *emu, int voice, unsigned char *volume)
+{
+ snd_emu10k1_ptr_write(emu, PTRX_FXSENDAMOUNT_A, voice, volume[0]);
+ snd_emu10k1_ptr_write(emu, PTRX_FXSENDAMOUNT_B, voice, volume[1]);
+ snd_emu10k1_ptr_write(emu, PSST_FXSENDAMOUNT_C, voice, volume[2]);
+ snd_emu10k1_ptr_write(emu, DSL_FXSENDAMOUNT_D, voice, volume[3]);
+ if (emu->audigy) {
+ unsigned int val = ((unsigned int)volume[4] << 24) |
+ ((unsigned int)volume[5] << 16) |
+ ((unsigned int)volume[6] << 8) |
+ (unsigned int)volume[7];
+ snd_emu10k1_ptr_write(emu, A_SENDAMOUNTS, voice, val);
+ }
+}
+
+/* PCM stream controls */
+
+static int snd_emu10k1_send_routing_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = emu->audigy ? 3*8 : 3*4;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = emu->audigy ? 0x3f : 0x0f;
+ return 0;
+}
+
+static int snd_emu10k1_send_routing_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ int voice, idx;
+ int num_efx = emu->audigy ? 8 : 4;
+ int mask = emu->audigy ? 0x3f : 0x0f;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (voice = 0; voice < 3; voice++)
+ for (idx = 0; idx < num_efx; idx++)
+ ucontrol->value.integer.value[(voice * num_efx) + idx] =
+ mix->send_routing[voice][idx] & mask;
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_send_routing_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ int change = 0, voice, idx, val;
+ int num_efx = emu->audigy ? 8 : 4;
+ int mask = emu->audigy ? 0x3f : 0x0f;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (voice = 0; voice < 3; voice++)
+ for (idx = 0; idx < num_efx; idx++) {
+ val = ucontrol->value.integer.value[(voice * num_efx) + idx] & mask;
+ if (mix->send_routing[voice][idx] != val) {
+ mix->send_routing[voice][idx] = val;
+ change = 1;
+ }
+ }
+ if (change && mix->epcm) {
+ if (mix->epcm->voices[0] && mix->epcm->voices[1]) {
+ update_emu10k1_fxrt(emu, mix->epcm->voices[0]->number,
+ &mix->send_routing[1][0]);
+ update_emu10k1_fxrt(emu, mix->epcm->voices[1]->number,
+ &mix->send_routing[2][0]);
+ } else if (mix->epcm->voices[0]) {
+ update_emu10k1_fxrt(emu, mix->epcm->voices[0]->number,
+ &mix->send_routing[0][0]);
+ }
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_send_routing_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "EMU10K1 PCM Send Routing",
+ .count = 32,
+ .info = snd_emu10k1_send_routing_info,
+ .get = snd_emu10k1_send_routing_get,
+ .put = snd_emu10k1_send_routing_put
+};
+
+static int snd_emu10k1_send_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = emu->audigy ? 3*8 : 3*4;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_emu10k1_send_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ int idx;
+ int num_efx = emu->audigy ? 8 : 4;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < 3*num_efx; idx++)
+ ucontrol->value.integer.value[idx] = mix->send_volume[idx/num_efx][idx%num_efx];
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_send_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ int change = 0, idx, val;
+ int num_efx = emu->audigy ? 8 : 4;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < 3*num_efx; idx++) {
+ val = ucontrol->value.integer.value[idx] & 255;
+ if (mix->send_volume[idx/num_efx][idx%num_efx] != val) {
+ mix->send_volume[idx/num_efx][idx%num_efx] = val;
+ change = 1;
+ }
+ }
+ if (change && mix->epcm) {
+ if (mix->epcm->voices[0] && mix->epcm->voices[1]) {
+ update_emu10k1_send_volume(emu, mix->epcm->voices[0]->number,
+ &mix->send_volume[1][0]);
+ update_emu10k1_send_volume(emu, mix->epcm->voices[1]->number,
+ &mix->send_volume[2][0]);
+ } else if (mix->epcm->voices[0]) {
+ update_emu10k1_send_volume(emu, mix->epcm->voices[0]->number,
+ &mix->send_volume[0][0]);
+ }
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_send_volume_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "EMU10K1 PCM Send Volume",
+ .count = 32,
+ .info = snd_emu10k1_send_volume_info,
+ .get = snd_emu10k1_send_volume_get,
+ .put = snd_emu10k1_send_volume_put
+};
+
+static int snd_emu10k1_attn_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xffff;
+ return 0;
+}
+
+static int snd_emu10k1_attn_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ unsigned long flags;
+ int idx;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < 3; idx++)
+ ucontrol->value.integer.value[idx] = mix->attn[idx];
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_attn_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ int change = 0, idx, val;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < 3; idx++) {
+ val = ucontrol->value.integer.value[idx] & 0xffff;
+ if (mix->attn[idx] != val) {
+ mix->attn[idx] = val;
+ change = 1;
+ }
+ }
+ if (change && mix->epcm) {
+ if (mix->epcm->voices[0] && mix->epcm->voices[1]) {
+ snd_emu10k1_ptr_write(emu, VTFT_VOLUMETARGET, mix->epcm->voices[0]->number, mix->attn[1]);
+ snd_emu10k1_ptr_write(emu, VTFT_VOLUMETARGET, mix->epcm->voices[1]->number, mix->attn[2]);
+ } else if (mix->epcm->voices[0]) {
+ snd_emu10k1_ptr_write(emu, VTFT_VOLUMETARGET, mix->epcm->voices[0]->number, mix->attn[0]);
+ }
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_attn_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "EMU10K1 PCM Volume",
+ .count = 32,
+ .info = snd_emu10k1_attn_info,
+ .get = snd_emu10k1_attn_get,
+ .put = snd_emu10k1_attn_put
+};
+
+/* Mutichannel PCM stream controls */
+
+static int snd_emu10k1_efx_send_routing_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = emu->audigy ? 8 : 4;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = emu->audigy ? 0x3f : 0x0f;
+ return 0;
+}
+
+static int snd_emu10k1_efx_send_routing_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ int idx;
+ int num_efx = emu->audigy ? 8 : 4;
+ int mask = emu->audigy ? 0x3f : 0x0f;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < num_efx; idx++)
+ ucontrol->value.integer.value[idx] =
+ mix->send_routing[0][idx] & mask;
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_efx_send_routing_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch];
+ int change = 0, idx, val;
+ int num_efx = emu->audigy ? 8 : 4;
+ int mask = emu->audigy ? 0x3f : 0x0f;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < num_efx; idx++) {
+ val = ucontrol->value.integer.value[idx] & mask;
+ if (mix->send_routing[0][idx] != val) {
+ mix->send_routing[0][idx] = val;
+ change = 1;
+ }
+ }
+
+ if (change && mix->epcm) {
+ if (mix->epcm->voices[ch]) {
+ update_emu10k1_fxrt(emu, mix->epcm->voices[ch]->number,
+ &mix->send_routing[0][0]);
+ }
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_efx_send_routing_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "Multichannel PCM Send Routing",
+ .count = 16,
+ .info = snd_emu10k1_efx_send_routing_info,
+ .get = snd_emu10k1_efx_send_routing_get,
+ .put = snd_emu10k1_efx_send_routing_put
+};
+
+static int snd_emu10k1_efx_send_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = emu->audigy ? 8 : 4;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_emu10k1_efx_send_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ int idx;
+ int num_efx = emu->audigy ? 8 : 4;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < num_efx; idx++)
+ ucontrol->value.integer.value[idx] = mix->send_volume[0][idx];
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_efx_send_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch];
+ int change = 0, idx, val;
+ int num_efx = emu->audigy ? 8 : 4;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ for (idx = 0; idx < num_efx; idx++) {
+ val = ucontrol->value.integer.value[idx] & 255;
+ if (mix->send_volume[0][idx] != val) {
+ mix->send_volume[0][idx] = val;
+ change = 1;
+ }
+ }
+ if (change && mix->epcm) {
+ if (mix->epcm->voices[ch]) {
+ update_emu10k1_send_volume(emu, mix->epcm->voices[ch]->number,
+ &mix->send_volume[0][0]);
+ }
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+
+static struct snd_kcontrol_new snd_emu10k1_efx_send_volume_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "Multichannel PCM Send Volume",
+ .count = 16,
+ .info = snd_emu10k1_efx_send_volume_info,
+ .get = snd_emu10k1_efx_send_volume_get,
+ .put = snd_emu10k1_efx_send_volume_put
+};
+
+static int snd_emu10k1_efx_attn_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xffff;
+ return 0;
+}
+
+static int snd_emu10k1_efx_attn_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ struct snd_emu10k1_pcm_mixer *mix =
+ &emu->efx_pcm_mixer[snd_ctl_get_ioffidx(kcontrol, &ucontrol->id)];
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ ucontrol->value.integer.value[0] = mix->attn[0];
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_efx_attn_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int ch = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ struct snd_emu10k1_pcm_mixer *mix = &emu->efx_pcm_mixer[ch];
+ int change = 0, val;
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ val = ucontrol->value.integer.value[0] & 0xffff;
+ if (mix->attn[0] != val) {
+ mix->attn[0] = val;
+ change = 1;
+ }
+ if (change && mix->epcm) {
+ if (mix->epcm->voices[ch]) {
+ snd_emu10k1_ptr_write(emu, VTFT_VOLUMETARGET, mix->epcm->voices[ch]->number, mix->attn[0]);
+ }
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_efx_attn_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "Multichannel PCM Volume",
+ .count = 16,
+ .info = snd_emu10k1_efx_attn_info,
+ .get = snd_emu10k1_efx_attn_get,
+ .put = snd_emu10k1_efx_attn_put
+};
+
+#define snd_emu10k1_shared_spdif_info snd_ctl_boolean_mono_info
+
+static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+
+ if (emu->audigy)
+ ucontrol->value.integer.value[0] = inl(emu->port + A_IOCFG) & A_IOCFG_GPOUT0 ? 1 : 0;
+ else
+ ucontrol->value.integer.value[0] = inl(emu->port + HCFG) & HCFG_GPOUT0 ? 1 : 0;
+ if (emu->card_capabilities->invert_shared_spdif)
+ ucontrol->value.integer.value[0] =
+ !ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned long flags;
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg, val, sw;
+ int change = 0;
+
+ sw = ucontrol->value.integer.value[0];
+ if (emu->card_capabilities->invert_shared_spdif)
+ sw = !sw;
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ if ( emu->card_capabilities->i2c_adc) {
+ /* Do nothing for Audigy 2 ZS Notebook */
+ } else if (emu->audigy) {
+ reg = inl(emu->port + A_IOCFG);
+ val = sw ? A_IOCFG_GPOUT0 : 0;
+ change = (reg & A_IOCFG_GPOUT0) != val;
+ if (change) {
+ reg &= ~A_IOCFG_GPOUT0;
+ reg |= val;
+ outl(reg | val, emu->port + A_IOCFG);
+ }
+ }
+ reg = inl(emu->port + HCFG);
+ val = sw ? HCFG_GPOUT0 : 0;
+ change |= (reg & HCFG_GPOUT0) != val;
+ if (change) {
+ reg &= ~HCFG_GPOUT0;
+ reg |= val;
+ outl(reg | val, emu->port + HCFG);
+ }
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_shared_spdif __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SB Live Analog/Digital Output Jack",
+ .info = snd_emu10k1_shared_spdif_info,
+ .get = snd_emu10k1_shared_spdif_get,
+ .put = snd_emu10k1_shared_spdif_put
+};
+
+static struct snd_kcontrol_new snd_audigy_shared_spdif __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Audigy Analog/Digital Output Jack",
+ .info = snd_emu10k1_shared_spdif_info,
+ .get = snd_emu10k1_shared_spdif_get,
+ .put = snd_emu10k1_shared_spdif_put
+};
+
+/*
+ */
+static void snd_emu10k1_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct snd_emu10k1 *emu = ac97->private_data;
+ emu->ac97 = NULL;
+}
+
+/*
+ */
+static int remove_ctl(struct snd_card *card, const char *name)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ strcpy(id.name, name);
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_remove_id(card, &id);
+}
+
+static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name)
+{
+ struct snd_ctl_elem_id sid;
+ memset(&sid, 0, sizeof(sid));
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_find_id(card, &sid);
+}
+
+static int rename_ctl(struct snd_card *card, const char *src, const char *dst)
+{
+ struct snd_kcontrol *kctl = ctl_find(card, src);
+ if (kctl) {
+ strcpy(kctl->id.name, dst);
+ return 0;
+ }
+ return -ENOENT;
+}
+
+int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
+ int pcm_device, int multi_device)
+{
+ int err, pcm;
+ struct snd_kcontrol *kctl;
+ struct snd_card *card = emu->card;
+ char **c;
+ static char *emu10k1_remove_ctls[] = {
+ /* no AC97 mono, surround, center/lfe */
+ "Master Mono Playback Switch",
+ "Master Mono Playback Volume",
+ "PCM Out Path & Mute",
+ "Mono Output Select",
+ "Front Playback Switch",
+ "Front Playback Volume",
+ "Surround Playback Switch",
+ "Surround Playback Volume",
+ "Center Playback Switch",
+ "Center Playback Volume",
+ "LFE Playback Switch",
+ "LFE Playback Volume",
+ NULL
+ };
+ static char *emu10k1_rename_ctls[] = {
+ "Surround Digital Playback Volume", "Surround Playback Volume",
+ "Center Digital Playback Volume", "Center Playback Volume",
+ "LFE Digital Playback Volume", "LFE Playback Volume",
+ NULL
+ };
+ static char *audigy_remove_ctls[] = {
+ /* Master/PCM controls on ac97 of Audigy has no effect */
+ /* On the Audigy2 the AC97 playback is piped into
+ * the Philips ADC for 24bit capture */
+ "PCM Playback Switch",
+ "PCM Playback Volume",
+ "Master Mono Playback Switch",
+ "Master Mono Playback Volume",
+ "Master Playback Switch",
+ "Master Playback Volume",
+ "PCM Out Path & Mute",
+ "Mono Output Select",
+ /* remove unused AC97 capture controls */
+ "Capture Source",
+ "Capture Switch",
+ "Capture Volume",
+ "Mic Select",
+ "Video Playback Switch",
+ "Video Playback Volume",
+ "Mic Playback Switch",
+ "Mic Playback Volume",
+ NULL
+ };
+ static char *audigy_rename_ctls[] = {
+ /* use conventional names */
+ "Wave Playback Volume", "PCM Playback Volume",
+ /* "Wave Capture Volume", "PCM Capture Volume", */
+ "Wave Master Playback Volume", "Master Playback Volume",
+ "AMic Playback Volume", "Mic Playback Volume",
+ NULL
+ };
+ static char *audigy_rename_ctls_i2c_adc[] = {
+ //"Analog Mix Capture Volume","OLD Analog Mix Capture Volume",
+ "Line Capture Volume", "Analog Mix Capture Volume",
+ "Wave Playback Volume", "OLD PCM Playback Volume",
+ "Wave Master Playback Volume", "Master Playback Volume",
+ "AMic Playback Volume", "Old Mic Playback Volume",
+ "CD Capture Volume", "IEC958 Optical Capture Volume",
+ NULL
+ };
+ static char *audigy_remove_ctls_i2c_adc[] = {
+ /* On the Audigy2 ZS Notebook
+ * Capture via WM8775 */
+ "Mic Capture Volume",
+ "Analog Mix Capture Volume",
+ "Aux Capture Volume",
+ "IEC958 Optical Capture Volume",
+ NULL
+ };
+ static char *audigy_remove_ctls_1361t_adc[] = {
+ /* On the Audigy2 the AC97 playback is piped into
+ * the Philips ADC for 24bit capture */
+ "PCM Playback Switch",
+ "PCM Playback Volume",
+ "Master Mono Playback Switch",
+ "Master Mono Playback Volume",
+ "Capture Source",
+ "Capture Switch",
+ "Capture Volume",
+ "Mic Capture Volume",
+ "Headphone Playback Switch",
+ "Headphone Playback Volume",
+ "3D Control - Center",
+ "3D Control - Depth",
+ "3D Control - Switch",
+ "Line2 Playback Volume",
+ "Line2 Capture Volume",
+ NULL
+ };
+ static char *audigy_rename_ctls_1361t_adc[] = {
+ "Master Playback Switch", "Master Capture Switch",
+ "Master Playback Volume", "Master Capture Volume",
+ "Wave Master Playback Volume", "Master Playback Volume",
+ "PC Speaker Playback Switch", "PC Speaker Capture Switch",
+ "PC Speaker Playback Volume", "PC Speaker Capture Volume",
+ "Phone Playback Switch", "Phone Capture Switch",
+ "Phone Playback Volume", "Phone Capture Volume",
+ "Mic Playback Switch", "Mic Capture Switch",
+ "Mic Playback Volume", "Mic Capture Volume",
+ "Line Playback Switch", "Line Capture Switch",
+ "Line Playback Volume", "Line Capture Volume",
+ "CD Playback Switch", "CD Capture Switch",
+ "CD Playback Volume", "CD Capture Volume",
+ "Aux Playback Switch", "Aux Capture Switch",
+ "Aux Playback Volume", "Aux Capture Volume",
+ "Video Playback Switch", "Video Capture Switch",
+ "Video Playback Volume", "Video Capture Volume",
+
+ NULL
+ };
+
+ if (emu->card_capabilities->ac97_chip) {
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_emu10k1_ac97_write,
+ .read = snd_emu10k1_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(emu->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+ pbus->no_vra = 1; /* we don't need VRA */
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = emu;
+ ac97.private_free = snd_emu10k1_mixer_free_ac97;
+ ac97.scaps = AC97_SCAP_NO_SPDIF;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &emu->ac97)) < 0) {
+ if (emu->card_capabilities->ac97_chip == 1)
+ return err;
+ snd_printd(KERN_INFO "emu10k1: AC97 is optional on this board\n");
+ snd_printd(KERN_INFO" Proceeding without ac97 mixers...\n");
+ snd_device_free(emu->card, pbus);
+ goto no_ac97; /* FIXME: get rid of ugly gotos.. */
+ }
+ if (emu->audigy) {
+ /* set master volume to 0 dB */
+ snd_ac97_write_cache(emu->ac97, AC97_MASTER, 0x0000);
+ /* set capture source to mic */
+ snd_ac97_write_cache(emu->ac97, AC97_REC_SEL, 0x0000);
+ if (emu->card_capabilities->adc_1361t)
+ c = audigy_remove_ctls_1361t_adc;
+ else
+ c = audigy_remove_ctls;
+ } else {
+ /*
+ * Credits for cards based on STAC9758:
+ * James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Voluspa <voluspa@comhem.se>
+ */
+ if (emu->ac97->id == AC97_ID_STAC9758) {
+ emu->rear_ac97 = 1;
+ snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE|AC97SLOT_REAR_LEFT|AC97SLOT_REAR_RIGHT);
+ snd_ac97_write_cache(emu->ac97, AC97_HEADPHONE, 0x0202);
+ }
+ /* remove unused AC97 controls */
+ snd_ac97_write_cache(emu->ac97, AC97_SURROUND_MASTER, 0x0202);
+ snd_ac97_write_cache(emu->ac97, AC97_CENTER_LFE_MASTER, 0x0202);
+ c = emu10k1_remove_ctls;
+ }
+ for (; *c; c++)
+ remove_ctl(card, *c);
+ } else if (emu->card_capabilities->i2c_adc) {
+ c = audigy_remove_ctls_i2c_adc;
+ for (; *c; c++)
+ remove_ctl(card, *c);
+ } else {
+ no_ac97:
+ if (emu->card_capabilities->ecard)
+ strcpy(emu->card->mixername, "EMU APS");
+ else if (emu->audigy)
+ strcpy(emu->card->mixername, "SB Audigy");
+ else
+ strcpy(emu->card->mixername, "Emu10k1");
+ }
+
+ if (emu->audigy)
+ if (emu->card_capabilities->adc_1361t)
+ c = audigy_rename_ctls_1361t_adc;
+ else if (emu->card_capabilities->i2c_adc)
+ c = audigy_rename_ctls_i2c_adc;
+ else
+ c = audigy_rename_ctls;
+ else
+ c = emu10k1_rename_ctls;
+ for (; *c; c += 2)
+ rename_ctl(card, c[0], c[1]);
+
+ if (emu->card_capabilities->subsystem == 0x20071102) { /* Audigy 4 Pro */
+ rename_ctl(card, "Line2 Capture Volume", "Line1/Mic Capture Volume");
+ rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume");
+ rename_ctl(card, "Aux2 Capture Volume", "Line3 Capture Volume");
+ rename_ctl(card, "Mic Capture Volume", "Unknown1 Capture Volume");
+ remove_ctl(card, "Headphone Playback Switch");
+ remove_ctl(card, "Headphone Playback Volume");
+ remove_ctl(card, "3D Control - Center");
+ remove_ctl(card, "3D Control - Depth");
+ remove_ctl(card, "3D Control - Switch");
+ }
+ if ((kctl = emu->ctl_send_routing = snd_ctl_new1(&snd_emu10k1_send_routing_control, emu)) == NULL)
+ return -ENOMEM;
+ kctl->id.device = pcm_device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = emu->ctl_send_volume = snd_ctl_new1(&snd_emu10k1_send_volume_control, emu)) == NULL)
+ return -ENOMEM;
+ kctl->id.device = pcm_device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = emu->ctl_attn = snd_ctl_new1(&snd_emu10k1_attn_control, emu)) == NULL)
+ return -ENOMEM;
+ kctl->id.device = pcm_device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+
+ if ((kctl = emu->ctl_efx_send_routing = snd_ctl_new1(&snd_emu10k1_efx_send_routing_control, emu)) == NULL)
+ return -ENOMEM;
+ kctl->id.device = multi_device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+
+ if ((kctl = emu->ctl_efx_send_volume = snd_ctl_new1(&snd_emu10k1_efx_send_volume_control, emu)) == NULL)
+ return -ENOMEM;
+ kctl->id.device = multi_device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+
+ if ((kctl = emu->ctl_efx_attn = snd_ctl_new1(&snd_emu10k1_efx_attn_control, emu)) == NULL)
+ return -ENOMEM;
+ kctl->id.device = multi_device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+
+ /* initialize the routing and volume table for each pcm playback stream */
+ for (pcm = 0; pcm < 32; pcm++) {
+ struct snd_emu10k1_pcm_mixer *mix;
+ int v;
+
+ mix = &emu->pcm_mixer[pcm];
+ mix->epcm = NULL;
+
+ for (v = 0; v < 4; v++)
+ mix->send_routing[0][v] =
+ mix->send_routing[1][v] =
+ mix->send_routing[2][v] = v;
+
+ memset(&mix->send_volume, 0, sizeof(mix->send_volume));
+ mix->send_volume[0][0] = mix->send_volume[0][1] =
+ mix->send_volume[1][0] = mix->send_volume[2][1] = 255;
+
+ mix->attn[0] = mix->attn[1] = mix->attn[2] = 0xffff;
+ }
+
+ /* initialize the routing and volume table for the multichannel playback stream */
+ for (pcm = 0; pcm < NUM_EFX_PLAYBACK; pcm++) {
+ struct snd_emu10k1_pcm_mixer *mix;
+ int v;
+
+ mix = &emu->efx_pcm_mixer[pcm];
+ mix->epcm = NULL;
+
+ mix->send_routing[0][0] = pcm;
+ mix->send_routing[0][1] = (pcm == 0) ? 1 : 0;
+ for (v = 0; v < 2; v++)
+ mix->send_routing[0][2+v] = 13+v;
+ if (emu->audigy)
+ for (v = 0; v < 4; v++)
+ mix->send_routing[0][4+v] = 60+v;
+
+ memset(&mix->send_volume, 0, sizeof(mix->send_volume));
+ mix->send_volume[0][0] = 255;
+
+ mix->attn[0] = 0xffff;
+ }
+
+ if (! emu->card_capabilities->ecard) { /* FIXME: APS has these controls? */
+ /* sb live! and audigy */
+ if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu)) == NULL)
+ return -ENOMEM;
+ if (!emu->audigy)
+ kctl->id.device = emu->pcm_efx->device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_control, emu)) == NULL)
+ return -ENOMEM;
+ if (!emu->audigy)
+ kctl->id.device = emu->pcm_efx->device;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ }
+
+ if (emu->card_capabilities->emu_model) {
+ ; /* Disable the snd_audigy_spdif_shared_spdif */
+ } else if (emu->audigy) {
+ if ((kctl = snd_ctl_new1(&snd_audigy_shared_spdif, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+#if 0
+ if ((kctl = snd_ctl_new1(&snd_audigy_spdif_output_rate, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+#endif
+ } else if (! emu->card_capabilities->ecard) {
+ /* sb live! */
+ if ((kctl = snd_ctl_new1(&snd_emu10k1_shared_spdif, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ }
+ if (emu->card_capabilities->ca0151_chip) { /* P16V */
+ if ((err = snd_p16v_mixer(emu)))
+ return err;
+ }
+
+ if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) {
+ /* 1616(m) cardbus */
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_emu1616_output_enum_ctls); i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1616_output_enum_ctls[i],
+ emu));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(snd_emu1010_input_enum_ctls); i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_input_enum_ctls[i],
+ emu));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(snd_emu1010_adc_pads) - 2; i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_adc_pads[i], emu));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(snd_emu1010_dac_pads) - 2; i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_dac_pads[i], emu));
+ if (err < 0)
+ return err;
+ }
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_internal_clock, emu));
+ if (err < 0)
+ return err;
+
+ } else if (emu->card_capabilities->emu_model) {
+ /* all other e-mu cards for now */
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_emu1010_output_enum_ctls); i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_output_enum_ctls[i],
+ emu));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(snd_emu1010_input_enum_ctls); i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_input_enum_ctls[i],
+ emu));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(snd_emu1010_adc_pads); i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_adc_pads[i], emu));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(snd_emu1010_dac_pads); i++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_dac_pads[i], emu));
+ if (err < 0)
+ return err;
+ }
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_emu1010_internal_clock, emu));
+ if (err < 0)
+ return err;
+ }
+
+ if ( emu->card_capabilities->i2c_adc) {
+ int i;
+
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_audigy_i2c_capture_source, emu));
+ if (err < 0)
+ return err;
+
+ for (i = 0; i < ARRAY_SIZE(snd_audigy_i2c_volume_ctls); i++) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_audigy_i2c_volume_ctls[i], emu));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c
new file mode 100644
index 0000000..8578c70
--- /dev/null
+++ b/sound/pci/emu10k1/emumpu401.c
@@ -0,0 +1,396 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of EMU10K1 MPU-401 in UART mode
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+
+#define EMU10K1_MIDI_MODE_INPUT (1<<0)
+#define EMU10K1_MIDI_MODE_OUTPUT (1<<1)
+
+static inline unsigned char mpu401_read(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_midi *mpu, int idx)
+{
+ if (emu->audigy)
+ return (unsigned char)snd_emu10k1_ptr_read(emu, mpu->port + idx, 0);
+ else
+ return inb(emu->port + mpu->port + idx);
+}
+
+static inline void mpu401_write(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_midi *mpu, int data, int idx)
+{
+ if (emu->audigy)
+ snd_emu10k1_ptr_write(emu, mpu->port + idx, 0, data);
+ else
+ outb(data, emu->port + mpu->port + idx);
+}
+
+#define mpu401_write_data(emu, mpu, data) mpu401_write(emu, mpu, data, 0)
+#define mpu401_write_cmd(emu, mpu, data) mpu401_write(emu, mpu, data, 1)
+#define mpu401_read_data(emu, mpu) mpu401_read(emu, mpu, 0)
+#define mpu401_read_stat(emu, mpu) mpu401_read(emu, mpu, 1)
+
+#define mpu401_input_avail(emu,mpu) (!(mpu401_read_stat(emu,mpu) & 0x80))
+#define mpu401_output_ready(emu,mpu) (!(mpu401_read_stat(emu,mpu) & 0x40))
+
+#define MPU401_RESET 0xff
+#define MPU401_ENTER_UART 0x3f
+#define MPU401_ACK 0xfe
+
+static void mpu401_clear_rx(struct snd_emu10k1 *emu, struct snd_emu10k1_midi *mpu)
+{
+ int timeout = 100000;
+ for (; timeout > 0 && mpu401_input_avail(emu, mpu); timeout--)
+ mpu401_read_data(emu, mpu);
+#ifdef CONFIG_SND_DEBUG
+ if (timeout <= 0)
+ snd_printk(KERN_ERR "cmd: clear rx timeout (status = 0x%x)\n", mpu401_read_stat(emu, mpu));
+#endif
+}
+
+/*
+
+ */
+
+static void do_emu10k1_midi_interrupt(struct snd_emu10k1 *emu, struct snd_emu10k1_midi *midi, unsigned int status)
+{
+ unsigned char byte;
+
+ if (midi->rmidi == NULL) {
+ snd_emu10k1_intr_disable(emu, midi->tx_enable | midi->rx_enable);
+ return;
+ }
+
+ spin_lock(&midi->input_lock);
+ if ((status & midi->ipr_rx) && mpu401_input_avail(emu, midi)) {
+ if (!(midi->midi_mode & EMU10K1_MIDI_MODE_INPUT)) {
+ mpu401_clear_rx(emu, midi);
+ } else {
+ byte = mpu401_read_data(emu, midi);
+ if (midi->substream_input)
+ snd_rawmidi_receive(midi->substream_input, &byte, 1);
+ }
+ }
+ spin_unlock(&midi->input_lock);
+
+ spin_lock(&midi->output_lock);
+ if ((status & midi->ipr_tx) && mpu401_output_ready(emu, midi)) {
+ if (midi->substream_output &&
+ snd_rawmidi_transmit(midi->substream_output, &byte, 1) == 1) {
+ mpu401_write_data(emu, midi, byte);
+ } else {
+ snd_emu10k1_intr_disable(emu, midi->tx_enable);
+ }
+ }
+ spin_unlock(&midi->output_lock);
+}
+
+static void snd_emu10k1_midi_interrupt(struct snd_emu10k1 *emu, unsigned int status)
+{
+ do_emu10k1_midi_interrupt(emu, &emu->midi, status);
+}
+
+static void snd_emu10k1_midi_interrupt2(struct snd_emu10k1 *emu, unsigned int status)
+{
+ do_emu10k1_midi_interrupt(emu, &emu->midi2, status);
+}
+
+static int snd_emu10k1_midi_cmd(struct snd_emu10k1 * emu, struct snd_emu10k1_midi *midi, unsigned char cmd, int ack)
+{
+ unsigned long flags;
+ int timeout, ok;
+
+ spin_lock_irqsave(&midi->input_lock, flags);
+ mpu401_write_data(emu, midi, 0x00);
+ /* mpu401_clear_rx(emu, midi); */
+
+ mpu401_write_cmd(emu, midi, cmd);
+ if (ack) {
+ ok = 0;
+ timeout = 10000;
+ while (!ok && timeout-- > 0) {
+ if (mpu401_input_avail(emu, midi)) {
+ if (mpu401_read_data(emu, midi) == MPU401_ACK)
+ ok = 1;
+ }
+ }
+ if (!ok && mpu401_read_data(emu, midi) == MPU401_ACK)
+ ok = 1;
+ } else {
+ ok = 1;
+ }
+ spin_unlock_irqrestore(&midi->input_lock, flags);
+ if (!ok) {
+ snd_printk(KERN_ERR "midi_cmd: 0x%x failed at 0x%lx (status = 0x%x, data = 0x%x)!!!\n",
+ cmd, emu->port,
+ mpu401_read_stat(emu, midi),
+ mpu401_read_data(emu, midi));
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_emu10k1_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_emu10k1 *emu;
+ struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)substream->rmidi->private_data;
+ unsigned long flags;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= EMU10K1_MIDI_MODE_INPUT;
+ midi->substream_input = substream;
+ if (!(midi->midi_mode & EMU10K1_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ if (snd_emu10k1_midi_cmd(emu, midi, MPU401_RESET, 1))
+ goto error_out;
+ if (snd_emu10k1_midi_cmd(emu, midi, MPU401_ENTER_UART, 1))
+ goto error_out;
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+
+error_out:
+ return -EIO;
+}
+
+static int snd_emu10k1_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_emu10k1 *emu;
+ struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)substream->rmidi->private_data;
+ unsigned long flags;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ midi->midi_mode |= EMU10K1_MIDI_MODE_OUTPUT;
+ midi->substream_output = substream;
+ if (!(midi->midi_mode & EMU10K1_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ if (snd_emu10k1_midi_cmd(emu, midi, MPU401_RESET, 1))
+ goto error_out;
+ if (snd_emu10k1_midi_cmd(emu, midi, MPU401_ENTER_UART, 1))
+ goto error_out;
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return 0;
+
+error_out:
+ return -EIO;
+}
+
+static int snd_emu10k1_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_emu10k1 *emu;
+ struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)substream->rmidi->private_data;
+ unsigned long flags;
+ int err = 0;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ snd_emu10k1_intr_disable(emu, midi->rx_enable);
+ midi->midi_mode &= ~EMU10K1_MIDI_MODE_INPUT;
+ midi->substream_input = NULL;
+ if (!(midi->midi_mode & EMU10K1_MIDI_MODE_OUTPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ err = snd_emu10k1_midi_cmd(emu, midi, MPU401_RESET, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return err;
+}
+
+static int snd_emu10k1_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_emu10k1 *emu;
+ struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)substream->rmidi->private_data;
+ unsigned long flags;
+ int err = 0;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+ spin_lock_irqsave(&midi->open_lock, flags);
+ snd_emu10k1_intr_disable(emu, midi->tx_enable);
+ midi->midi_mode &= ~EMU10K1_MIDI_MODE_OUTPUT;
+ midi->substream_output = NULL;
+ if (!(midi->midi_mode & EMU10K1_MIDI_MODE_INPUT)) {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ err = snd_emu10k1_midi_cmd(emu, midi, MPU401_RESET, 0);
+ } else {
+ spin_unlock_irqrestore(&midi->open_lock, flags);
+ }
+ return err;
+}
+
+static void snd_emu10k1_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_emu10k1 *emu;
+ struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)substream->rmidi->private_data;
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return;
+
+ if (up)
+ snd_emu10k1_intr_enable(emu, midi->rx_enable);
+ else
+ snd_emu10k1_intr_disable(emu, midi->rx_enable);
+}
+
+static void snd_emu10k1_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_emu10k1 *emu;
+ struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)substream->rmidi->private_data;
+ unsigned long flags;
+
+ emu = midi->emu;
+ if (snd_BUG_ON(!emu))
+ return;
+
+ if (up) {
+ int max = 4;
+ unsigned char byte;
+
+ /* try to send some amount of bytes here before interrupts */
+ spin_lock_irqsave(&midi->output_lock, flags);
+ while (max > 0) {
+ if (mpu401_output_ready(emu, midi)) {
+ if (!(midi->midi_mode & EMU10K1_MIDI_MODE_OUTPUT) ||
+ snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ /* no more data */
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ return;
+ }
+ mpu401_write_data(emu, midi, byte);
+ max--;
+ } else {
+ break;
+ }
+ }
+ spin_unlock_irqrestore(&midi->output_lock, flags);
+ snd_emu10k1_intr_enable(emu, midi->tx_enable);
+ } else {
+ snd_emu10k1_intr_disable(emu, midi->tx_enable);
+ }
+}
+
+/*
+
+ */
+
+static struct snd_rawmidi_ops snd_emu10k1_midi_output =
+{
+ .open = snd_emu10k1_midi_output_open,
+ .close = snd_emu10k1_midi_output_close,
+ .trigger = snd_emu10k1_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_emu10k1_midi_input =
+{
+ .open = snd_emu10k1_midi_input_open,
+ .close = snd_emu10k1_midi_input_close,
+ .trigger = snd_emu10k1_midi_input_trigger,
+};
+
+static void snd_emu10k1_midi_free(struct snd_rawmidi *rmidi)
+{
+ struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)rmidi->private_data;
+ midi->interrupt = NULL;
+ midi->rmidi = NULL;
+}
+
+static int __devinit emu10k1_midi_init(struct snd_emu10k1 *emu, struct snd_emu10k1_midi *midi, int device, char *name)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if ((err = snd_rawmidi_new(emu->card, name, device, 1, 1, &rmidi)) < 0)
+ return err;
+ midi->emu = emu;
+ spin_lock_init(&midi->open_lock);
+ spin_lock_init(&midi->input_lock);
+ spin_lock_init(&midi->output_lock);
+ strcpy(rmidi->name, name);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_emu10k1_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_emu10k1_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = midi;
+ rmidi->private_free = snd_emu10k1_midi_free;
+ midi->rmidi = rmidi;
+ return 0;
+}
+
+int __devinit snd_emu10k1_midi(struct snd_emu10k1 *emu)
+{
+ struct snd_emu10k1_midi *midi = &emu->midi;
+ int err;
+
+ if ((err = emu10k1_midi_init(emu, midi, 0, "EMU10K1 MPU-401 (UART)")) < 0)
+ return err;
+
+ midi->tx_enable = INTE_MIDITXENABLE;
+ midi->rx_enable = INTE_MIDIRXENABLE;
+ midi->port = MUDATA;
+ midi->ipr_tx = IPR_MIDITRANSBUFEMPTY;
+ midi->ipr_rx = IPR_MIDIRECVBUFEMPTY;
+ midi->interrupt = snd_emu10k1_midi_interrupt;
+ return 0;
+}
+
+int __devinit snd_emu10k1_audigy_midi(struct snd_emu10k1 *emu)
+{
+ struct snd_emu10k1_midi *midi;
+ int err;
+
+ midi = &emu->midi;
+ if ((err = emu10k1_midi_init(emu, midi, 0, "Audigy MPU-401 (UART)")) < 0)
+ return err;
+
+ midi->tx_enable = INTE_MIDITXENABLE;
+ midi->rx_enable = INTE_MIDIRXENABLE;
+ midi->port = A_MUDATA1;
+ midi->ipr_tx = IPR_MIDITRANSBUFEMPTY;
+ midi->ipr_rx = IPR_MIDIRECVBUFEMPTY;
+ midi->interrupt = snd_emu10k1_midi_interrupt;
+
+ midi = &emu->midi2;
+ if ((err = emu10k1_midi_init(emu, midi, 1, "Audigy MPU-401 #2")) < 0)
+ return err;
+
+ midi->tx_enable = INTE_A_MIDITXENABLE2;
+ midi->rx_enable = INTE_A_MIDIRXENABLE2;
+ midi->port = A_MUDATA2;
+ midi->ipr_tx = IPR_A_MIDITRANSBUFEMPTY2;
+ midi->ipr_rx = IPR_A_MIDIRECVBUFEMPTY2;
+ midi->interrupt = snd_emu10k1_midi_interrupt2;
+ return 0;
+}
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
new file mode 100644
index 0000000..cf9276d
--- /dev/null
+++ b/sound/pci/emu10k1/emupcm.c
@@ -0,0 +1,1820 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Creative Labs, Inc.
+ * Routines for control of EMU10K1 chips / PCM routines
+ * Multichannel PCM support Copyright (c) Lee Revell <rlrevell@joe-job.com>
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/pci.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+
+static void snd_emu10k1_pcm_interrupt(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_voice *voice)
+{
+ struct snd_emu10k1_pcm *epcm;
+
+ if ((epcm = voice->epcm) == NULL)
+ return;
+ if (epcm->substream == NULL)
+ return;
+#if 0
+ printk("IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n",
+ epcm->substream->runtime->hw->pointer(emu, epcm->substream),
+ snd_pcm_lib_period_bytes(epcm->substream),
+ snd_pcm_lib_buffer_bytes(epcm->substream));
+#endif
+ snd_pcm_period_elapsed(epcm->substream);
+}
+
+static void snd_emu10k1_pcm_ac97adc_interrupt(struct snd_emu10k1 *emu,
+ unsigned int status)
+{
+#if 0
+ if (status & IPR_ADCBUFHALFFULL) {
+ if (emu->pcm_capture_substream->runtime->mode == SNDRV_PCM_MODE_FRAME)
+ return;
+ }
+#endif
+ snd_pcm_period_elapsed(emu->pcm_capture_substream);
+}
+
+static void snd_emu10k1_pcm_ac97mic_interrupt(struct snd_emu10k1 *emu,
+ unsigned int status)
+{
+#if 0
+ if (status & IPR_MICBUFHALFFULL) {
+ if (emu->pcm_capture_mic_substream->runtime->mode == SNDRV_PCM_MODE_FRAME)
+ return;
+ }
+#endif
+ snd_pcm_period_elapsed(emu->pcm_capture_mic_substream);
+}
+
+static void snd_emu10k1_pcm_efx_interrupt(struct snd_emu10k1 *emu,
+ unsigned int status)
+{
+#if 0
+ if (status & IPR_EFXBUFHALFFULL) {
+ if (emu->pcm_capture_efx_substream->runtime->mode == SNDRV_PCM_MODE_FRAME)
+ return;
+ }
+#endif
+ snd_pcm_period_elapsed(emu->pcm_capture_efx_substream);
+}
+
+static snd_pcm_uframes_t snd_emu10k1_efx_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ unsigned int ptr;
+
+ if (!epcm->running)
+ return 0;
+ ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->voices[0]->number) & 0x00ffffff;
+ ptr += runtime->buffer_size;
+ ptr -= epcm->ccca_start_addr;
+ ptr %= runtime->buffer_size;
+
+ return ptr;
+}
+
+static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voices)
+{
+ int err, i;
+
+ if (epcm->voices[1] != NULL && voices < 2) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->voices[1]);
+ epcm->voices[1] = NULL;
+ }
+ for (i = 0; i < voices; i++) {
+ if (epcm->voices[i] == NULL)
+ break;
+ }
+ if (i == voices)
+ return 0; /* already allocated */
+
+ for (i = 0; i < ARRAY_SIZE(epcm->voices); i++) {
+ if (epcm->voices[i]) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]);
+ epcm->voices[i] = NULL;
+ }
+ }
+ err = snd_emu10k1_voice_alloc(epcm->emu,
+ epcm->type == PLAYBACK_EMUVOICE ? EMU10K1_PCM : EMU10K1_EFX,
+ voices,
+ &epcm->voices[0]);
+
+ if (err < 0)
+ return err;
+ epcm->voices[0]->epcm = epcm;
+ if (voices > 1) {
+ for (i = 1; i < voices; i++) {
+ epcm->voices[i] = &epcm->emu->voices[epcm->voices[0]->number + i];
+ epcm->voices[i]->epcm = epcm;
+ }
+ }
+ if (epcm->extra == NULL) {
+ err = snd_emu10k1_voice_alloc(epcm->emu,
+ epcm->type == PLAYBACK_EMUVOICE ? EMU10K1_PCM : EMU10K1_EFX,
+ 1,
+ &epcm->extra);
+ if (err < 0) {
+ /* printk("pcm_channel_alloc: failed extra: voices=%d, frame=%d\n", voices, frame); */
+ for (i = 0; i < voices; i++) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]);
+ epcm->voices[i] = NULL;
+ }
+ return err;
+ }
+ epcm->extra->epcm = epcm;
+ epcm->extra->interrupt = snd_emu10k1_pcm_interrupt;
+ }
+ return 0;
+}
+
+static unsigned int capture_period_sizes[31] = {
+ 384, 448, 512, 640,
+ 384*2, 448*2, 512*2, 640*2,
+ 384*4, 448*4, 512*4, 640*4,
+ 384*8, 448*8, 512*8, 640*8,
+ 384*16, 448*16, 512*16, 640*16,
+ 384*32, 448*32, 512*32, 640*32,
+ 384*64, 448*64, 512*64, 640*64,
+ 384*128,448*128,512*128
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_capture_period_sizes = {
+ .count = 31,
+ .list = capture_period_sizes,
+ .mask = 0
+};
+
+static unsigned int capture_rates[8] = {
+ 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_capture_rates = {
+ .count = 8,
+ .list = capture_rates,
+ .mask = 0
+};
+
+static unsigned int snd_emu10k1_capture_rate_reg(unsigned int rate)
+{
+ switch (rate) {
+ case 8000: return ADCCR_SAMPLERATE_8;
+ case 11025: return ADCCR_SAMPLERATE_11;
+ case 16000: return ADCCR_SAMPLERATE_16;
+ case 22050: return ADCCR_SAMPLERATE_22;
+ case 24000: return ADCCR_SAMPLERATE_24;
+ case 32000: return ADCCR_SAMPLERATE_32;
+ case 44100: return ADCCR_SAMPLERATE_44;
+ case 48000: return ADCCR_SAMPLERATE_48;
+ default:
+ snd_BUG();
+ return ADCCR_SAMPLERATE_8;
+ }
+}
+
+static unsigned int snd_emu10k1_audigy_capture_rate_reg(unsigned int rate)
+{
+ switch (rate) {
+ case 8000: return A_ADCCR_SAMPLERATE_8;
+ case 11025: return A_ADCCR_SAMPLERATE_11;
+ case 12000: return A_ADCCR_SAMPLERATE_12; /* really supported? */
+ case 16000: return ADCCR_SAMPLERATE_16;
+ case 22050: return ADCCR_SAMPLERATE_22;
+ case 24000: return ADCCR_SAMPLERATE_24;
+ case 32000: return ADCCR_SAMPLERATE_32;
+ case 44100: return ADCCR_SAMPLERATE_44;
+ case 48000: return ADCCR_SAMPLERATE_48;
+ default:
+ snd_BUG();
+ return A_ADCCR_SAMPLERATE_8;
+ }
+}
+
+static unsigned int emu10k1_calc_pitch_target(unsigned int rate)
+{
+ unsigned int pitch_target;
+
+ pitch_target = (rate << 8) / 375;
+ pitch_target = (pitch_target >> 1) + (pitch_target & 1);
+ return pitch_target;
+}
+
+#define PITCH_48000 0x00004000
+#define PITCH_96000 0x00008000
+#define PITCH_85000 0x00007155
+#define PITCH_80726 0x00006ba2
+#define PITCH_67882 0x00005a82
+#define PITCH_57081 0x00004c1c
+
+static unsigned int emu10k1_select_interprom(unsigned int pitch_target)
+{
+ if (pitch_target == PITCH_48000)
+ return CCCA_INTERPROM_0;
+ else if (pitch_target < PITCH_48000)
+ return CCCA_INTERPROM_1;
+ else if (pitch_target >= PITCH_96000)
+ return CCCA_INTERPROM_0;
+ else if (pitch_target >= PITCH_85000)
+ return CCCA_INTERPROM_6;
+ else if (pitch_target >= PITCH_80726)
+ return CCCA_INTERPROM_5;
+ else if (pitch_target >= PITCH_67882)
+ return CCCA_INTERPROM_4;
+ else if (pitch_target >= PITCH_57081)
+ return CCCA_INTERPROM_3;
+ else
+ return CCCA_INTERPROM_2;
+}
+
+/*
+ * calculate cache invalidate size
+ *
+ * stereo: channel is stereo
+ * w_16: using 16bit samples
+ *
+ * returns: cache invalidate size in samples
+ */
+static inline int emu10k1_ccis(int stereo, int w_16)
+{
+ if (w_16) {
+ return stereo ? 24 : 26;
+ } else {
+ return stereo ? 24*2 : 26*2;
+ }
+}
+
+static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
+ int master, int extra,
+ struct snd_emu10k1_voice *evoice,
+ unsigned int start_addr,
+ unsigned int end_addr,
+ struct snd_emu10k1_pcm_mixer *mix)
+{
+ struct snd_pcm_substream *substream = evoice->epcm->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int silent_page, tmp;
+ int voice, stereo, w_16;
+ unsigned char attn, send_amount[8];
+ unsigned char send_routing[8];
+ unsigned long flags;
+ unsigned int pitch_target;
+ unsigned int ccis;
+
+ voice = evoice->number;
+ stereo = runtime->channels == 2;
+ w_16 = snd_pcm_format_width(runtime->format) == 16;
+
+ if (!extra && stereo) {
+ start_addr >>= 1;
+ end_addr >>= 1;
+ }
+ if (w_16) {
+ start_addr >>= 1;
+ end_addr >>= 1;
+ }
+
+ spin_lock_irqsave(&emu->reg_lock, flags);
+
+ /* volume parameters */
+ if (extra) {
+ attn = 0;
+ memset(send_routing, 0, sizeof(send_routing));
+ send_routing[0] = 0;
+ send_routing[1] = 1;
+ send_routing[2] = 2;
+ send_routing[3] = 3;
+ memset(send_amount, 0, sizeof(send_amount));
+ } else {
+ /* mono, left, right (master voice = left) */
+ tmp = stereo ? (master ? 1 : 2) : 0;
+ memcpy(send_routing, &mix->send_routing[tmp][0], 8);
+ memcpy(send_amount, &mix->send_volume[tmp][0], 8);
+ }
+
+ ccis = emu10k1_ccis(stereo, w_16);
+
+ if (master) {
+ evoice->epcm->ccca_start_addr = start_addr + ccis;
+ if (extra) {
+ start_addr += ccis;
+ end_addr += ccis;
+ }
+ if (stereo && !extra) {
+ snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK);
+ snd_emu10k1_ptr_write(emu, CPF, (voice + 1), CPF_STEREO_MASK);
+ } else {
+ snd_emu10k1_ptr_write(emu, CPF, voice, 0);
+ }
+ }
+
+ /* setup routing */
+ if (emu->audigy) {
+ snd_emu10k1_ptr_write(emu, A_FXRT1, voice,
+ snd_emu10k1_compose_audigy_fxrt1(send_routing));
+ snd_emu10k1_ptr_write(emu, A_FXRT2, voice,
+ snd_emu10k1_compose_audigy_fxrt2(send_routing));
+ snd_emu10k1_ptr_write(emu, A_SENDAMOUNTS, voice,
+ ((unsigned int)send_amount[4] << 24) |
+ ((unsigned int)send_amount[5] << 16) |
+ ((unsigned int)send_amount[6] << 8) |
+ (unsigned int)send_amount[7]);
+ } else
+ snd_emu10k1_ptr_write(emu, FXRT, voice,
+ snd_emu10k1_compose_send_routing(send_routing));
+ /* Stop CA */
+ /* Assumption that PT is already 0 so no harm overwriting */
+ snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]);
+ snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24));
+ snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24));
+ if (emu->card_capabilities->emu_model)
+ pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */
+ else
+ pitch_target = emu10k1_calc_pitch_target(runtime->rate);
+ if (extra)
+ snd_emu10k1_ptr_write(emu, CCCA, voice, start_addr |
+ emu10k1_select_interprom(pitch_target) |
+ (w_16 ? 0 : CCCA_8BITSELECT));
+ else
+ snd_emu10k1_ptr_write(emu, CCCA, voice, (start_addr + ccis) |
+ emu10k1_select_interprom(pitch_target) |
+ (w_16 ? 0 : CCCA_8BITSELECT));
+ /* Clear filter delay memory */
+ snd_emu10k1_ptr_write(emu, Z1, voice, 0);
+ snd_emu10k1_ptr_write(emu, Z2, voice, 0);
+ /* invalidate maps */
+ silent_page = ((unsigned int)emu->silent_page.addr << 1) | MAP_PTI_MASK;
+ snd_emu10k1_ptr_write(emu, MAPA, voice, silent_page);
+ snd_emu10k1_ptr_write(emu, MAPB, voice, silent_page);
+ /* modulation envelope */
+ snd_emu10k1_ptr_write(emu, CVCF, voice, 0xffff);
+ snd_emu10k1_ptr_write(emu, VTFT, voice, 0xffff);
+ snd_emu10k1_ptr_write(emu, ATKHLDM, voice, 0);
+ snd_emu10k1_ptr_write(emu, DCYSUSM, voice, 0x007f);
+ snd_emu10k1_ptr_write(emu, LFOVAL1, voice, 0x8000);
+ snd_emu10k1_ptr_write(emu, LFOVAL2, voice, 0x8000);
+ snd_emu10k1_ptr_write(emu, FMMOD, voice, 0);
+ snd_emu10k1_ptr_write(emu, TREMFRQ, voice, 0);
+ snd_emu10k1_ptr_write(emu, FM2FRQ2, voice, 0);
+ snd_emu10k1_ptr_write(emu, ENVVAL, voice, 0x8000);
+ /* volume envelope */
+ snd_emu10k1_ptr_write(emu, ATKHLDV, voice, 0x7f7f);
+ snd_emu10k1_ptr_write(emu, ENVVOL, voice, 0x0000);
+ /* filter envelope */
+ snd_emu10k1_ptr_write(emu, PEFE_FILTERAMOUNT, voice, 0x7f);
+ /* pitch envelope */
+ snd_emu10k1_ptr_write(emu, PEFE_PITCHAMOUNT, voice, 0);
+
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+}
+
+static int snd_emu10k1_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ int err;
+
+ if ((err = snd_emu10k1_pcm_channel_alloc(epcm, params_channels(hw_params))) < 0)
+ return err;
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ if (err > 0) { /* change */
+ int mapped;
+ if (epcm->memblk != NULL)
+ snd_emu10k1_free_pages(emu, epcm->memblk);
+ epcm->memblk = snd_emu10k1_alloc_pages(emu, substream);
+ epcm->start_addr = 0;
+ if (! epcm->memblk)
+ return -ENOMEM;
+ mapped = ((struct snd_emu10k1_memblk *)epcm->memblk)->mapped_page;
+ if (mapped < 0)
+ return -ENOMEM;
+ epcm->start_addr = mapped << PAGE_SHIFT;
+ }
+ return 0;
+}
+
+static int snd_emu10k1_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm;
+
+ if (runtime->private_data == NULL)
+ return 0;
+ epcm = runtime->private_data;
+ if (epcm->extra) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->extra);
+ epcm->extra = NULL;
+ }
+ if (epcm->voices[1]) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->voices[1]);
+ epcm->voices[1] = NULL;
+ }
+ if (epcm->voices[0]) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->voices[0]);
+ epcm->voices[0] = NULL;
+ }
+ if (epcm->memblk) {
+ snd_emu10k1_free_pages(emu, epcm->memblk);
+ epcm->memblk = NULL;
+ epcm->start_addr = 0;
+ }
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_emu10k1_efx_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm;
+ int i;
+
+ if (runtime->private_data == NULL)
+ return 0;
+ epcm = runtime->private_data;
+ if (epcm->extra) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->extra);
+ epcm->extra = NULL;
+ }
+ for (i = 0; i < NUM_EFX_PLAYBACK; i++) {
+ if (epcm->voices[i]) {
+ snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]);
+ epcm->voices[i] = NULL;
+ }
+ }
+ if (epcm->memblk) {
+ snd_emu10k1_free_pages(emu, epcm->memblk);
+ epcm->memblk = NULL;
+ epcm->start_addr = 0;
+ }
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_emu10k1_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ unsigned int start_addr, end_addr;
+
+ start_addr = epcm->start_addr;
+ end_addr = snd_pcm_lib_period_bytes(substream);
+ if (runtime->channels == 2) {
+ start_addr >>= 1;
+ end_addr >>= 1;
+ }
+ end_addr += start_addr;
+ snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra,
+ start_addr, end_addr, NULL);
+ start_addr = epcm->start_addr;
+ end_addr = epcm->start_addr + snd_pcm_lib_buffer_bytes(substream);
+ snd_emu10k1_pcm_init_voice(emu, 1, 0, epcm->voices[0],
+ start_addr, end_addr,
+ &emu->pcm_mixer[substream->number]);
+ if (epcm->voices[1])
+ snd_emu10k1_pcm_init_voice(emu, 0, 0, epcm->voices[1],
+ start_addr, end_addr,
+ &emu->pcm_mixer[substream->number]);
+ return 0;
+}
+
+static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ unsigned int start_addr, end_addr;
+ unsigned int channel_size;
+ int i;
+
+ start_addr = epcm->start_addr;
+ end_addr = epcm->start_addr + snd_pcm_lib_buffer_bytes(substream);
+
+ /*
+ * the kX driver leaves some space between voices
+ */
+ channel_size = ( end_addr - start_addr ) / NUM_EFX_PLAYBACK;
+
+ snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra,
+ start_addr, start_addr + (channel_size / 2), NULL);
+
+ /* only difference with the master voice is we use it for the pointer */
+ snd_emu10k1_pcm_init_voice(emu, 1, 0, epcm->voices[0],
+ start_addr, start_addr + channel_size,
+ &emu->efx_pcm_mixer[0]);
+
+ start_addr += channel_size;
+ for (i = 1; i < NUM_EFX_PLAYBACK; i++) {
+ snd_emu10k1_pcm_init_voice(emu, 0, 0, epcm->voices[i],
+ start_addr, start_addr + channel_size,
+ &emu->efx_pcm_mixer[i]);
+ start_addr += channel_size;
+ }
+
+ return 0;
+}
+
+static struct snd_pcm_hardware snd_emu10k1_efx_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = NUM_EFX_PLAYBACK,
+ .channels_max = NUM_EFX_PLAYBACK,
+ .buffer_bytes_max = (64*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (64*1024),
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static int snd_emu10k1_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_emu10k1_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_emu10k1_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ int idx;
+
+ /* zeroing the buffer size will stop capture */
+ snd_emu10k1_ptr_write(emu, epcm->capture_bs_reg, 0, 0);
+ switch (epcm->type) {
+ case CAPTURE_AC97ADC:
+ snd_emu10k1_ptr_write(emu, ADCCR, 0, 0);
+ break;
+ case CAPTURE_EFX:
+ if (emu->audigy) {
+ snd_emu10k1_ptr_write(emu, A_FXWC1, 0, 0);
+ snd_emu10k1_ptr_write(emu, A_FXWC2, 0, 0);
+ } else
+ snd_emu10k1_ptr_write(emu, FXWC, 0, 0);
+ break;
+ default:
+ break;
+ }
+ snd_emu10k1_ptr_write(emu, epcm->capture_ba_reg, 0, runtime->dma_addr);
+ epcm->capture_bufsize = snd_pcm_lib_buffer_bytes(substream);
+ epcm->capture_bs_val = 0;
+ for (idx = 0; idx < 31; idx++) {
+ if (capture_period_sizes[idx] == epcm->capture_bufsize) {
+ epcm->capture_bs_val = idx + 1;
+ break;
+ }
+ }
+ if (epcm->capture_bs_val == 0) {
+ snd_BUG();
+ epcm->capture_bs_val++;
+ }
+ if (epcm->type == CAPTURE_AC97ADC) {
+ epcm->capture_cr_val = emu->audigy ? A_ADCCR_LCHANENABLE : ADCCR_LCHANENABLE;
+ if (runtime->channels > 1)
+ epcm->capture_cr_val |= emu->audigy ? A_ADCCR_RCHANENABLE : ADCCR_RCHANENABLE;
+ epcm->capture_cr_val |= emu->audigy ?
+ snd_emu10k1_audigy_capture_rate_reg(runtime->rate) :
+ snd_emu10k1_capture_rate_reg(runtime->rate);
+ }
+ return 0;
+}
+
+static void snd_emu10k1_playback_invalidate_cache(struct snd_emu10k1 *emu, int extra, struct snd_emu10k1_voice *evoice)
+{
+ struct snd_pcm_runtime *runtime;
+ unsigned int voice, stereo, i, ccis, cra = 64, cs, sample;
+
+ if (evoice == NULL)
+ return;
+ runtime = evoice->epcm->substream->runtime;
+ voice = evoice->number;
+ stereo = (!extra && runtime->channels == 2);
+ sample = snd_pcm_format_width(runtime->format) == 16 ? 0 : 0x80808080;
+ ccis = emu10k1_ccis(stereo, sample == 0);
+ /* set cs to 2 * number of cache registers beside the invalidated */
+ cs = (sample == 0) ? (32-ccis) : (64-ccis+1) >> 1;
+ if (cs > 16) cs = 16;
+ for (i = 0; i < cs; i++) {
+ snd_emu10k1_ptr_write(emu, CD0 + i, voice, sample);
+ if (stereo) {
+ snd_emu10k1_ptr_write(emu, CD0 + i, voice + 1, sample);
+ }
+ }
+ /* reset cache */
+ snd_emu10k1_ptr_write(emu, CCR_CACHEINVALIDSIZE, voice, 0);
+ snd_emu10k1_ptr_write(emu, CCR_READADDRESS, voice, cra);
+ if (stereo) {
+ snd_emu10k1_ptr_write(emu, CCR_CACHEINVALIDSIZE, voice + 1, 0);
+ snd_emu10k1_ptr_write(emu, CCR_READADDRESS, voice + 1, cra);
+ }
+ /* fill cache */
+ snd_emu10k1_ptr_write(emu, CCR_CACHEINVALIDSIZE, voice, ccis);
+ if (stereo) {
+ snd_emu10k1_ptr_write(emu, CCR_CACHEINVALIDSIZE, voice+1, ccis);
+ }
+}
+
+static void snd_emu10k1_playback_prepare_voice(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *evoice,
+ int master, int extra,
+ struct snd_emu10k1_pcm_mixer *mix)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ unsigned int attn, vattn;
+ unsigned int voice, tmp;
+
+ if (evoice == NULL) /* skip second voice for mono */
+ return;
+ substream = evoice->epcm->substream;
+ runtime = substream->runtime;
+ voice = evoice->number;
+
+ attn = extra ? 0 : 0x00ff;
+ tmp = runtime->channels == 2 ? (master ? 1 : 2) : 0;
+ vattn = mix != NULL ? (mix->attn[tmp] << 16) : 0;
+ snd_emu10k1_ptr_write(emu, IFATN, voice, attn);
+ snd_emu10k1_ptr_write(emu, VTFT, voice, vattn | 0xffff);
+ snd_emu10k1_ptr_write(emu, CVCF, voice, vattn | 0xffff);
+ snd_emu10k1_ptr_write(emu, DCYSUSV, voice, 0x7f7f);
+ snd_emu10k1_voice_clear_loop_stop(emu, voice);
+}
+
+static void snd_emu10k1_playback_trigger_voice(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *evoice, int master, int extra)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm_runtime *runtime;
+ unsigned int voice, pitch, pitch_target;
+
+ if (evoice == NULL) /* skip second voice for mono */
+ return;
+ substream = evoice->epcm->substream;
+ runtime = substream->runtime;
+ voice = evoice->number;
+
+ pitch = snd_emu10k1_rate_to_pitch(runtime->rate) >> 8;
+ if (emu->card_capabilities->emu_model)
+ pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */
+ else
+ pitch_target = emu10k1_calc_pitch_target(runtime->rate);
+ snd_emu10k1_ptr_write(emu, PTRX_PITCHTARGET, voice, pitch_target);
+ if (master || evoice->epcm->type == PLAYBACK_EFX)
+ snd_emu10k1_ptr_write(emu, CPF_CURRENTPITCH, voice, pitch_target);
+ snd_emu10k1_ptr_write(emu, IP, voice, pitch);
+ if (extra)
+ snd_emu10k1_voice_intr_enable(emu, voice);
+}
+
+static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *evoice)
+{
+ unsigned int voice;
+
+ if (evoice == NULL)
+ return;
+ voice = evoice->number;
+ snd_emu10k1_voice_intr_disable(emu, voice);
+ snd_emu10k1_ptr_write(emu, PTRX_PITCHTARGET, voice, 0);
+ snd_emu10k1_ptr_write(emu, CPF_CURRENTPITCH, voice, 0);
+ snd_emu10k1_ptr_write(emu, IFATN, voice, 0xffff);
+ snd_emu10k1_ptr_write(emu, VTFT, voice, 0xffff);
+ snd_emu10k1_ptr_write(emu, CVCF, voice, 0xffff);
+ snd_emu10k1_ptr_write(emu, IP, voice, 0);
+}
+
+static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ struct snd_emu10k1_pcm_mixer *mix;
+ int result = 0;
+
+ /* printk("trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", (int)emu, cmd, substream->ops->pointer(substream)); */
+ spin_lock(&emu->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */
+ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]);
+ /* follow thru */
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ mix = &emu->pcm_mixer[substream->number];
+ snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix);
+ snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix);
+ snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL);
+ snd_emu10k1_playback_trigger_voice(emu, epcm->voices[0], 1, 0);
+ snd_emu10k1_playback_trigger_voice(emu, epcm->voices[1], 0, 0);
+ snd_emu10k1_playback_trigger_voice(emu, epcm->extra, 1, 1);
+ epcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ epcm->running = 0;
+ snd_emu10k1_playback_stop_voice(emu, epcm->voices[0]);
+ snd_emu10k1_playback_stop_voice(emu, epcm->voices[1]);
+ snd_emu10k1_playback_stop_voice(emu, epcm->extra);
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ spin_unlock(&emu->reg_lock);
+ return result;
+}
+
+static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ int result = 0;
+
+ spin_lock(&emu->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /* hmm this should cause full and half full interrupt to be raised? */
+ outl(epcm->capture_ipr, emu->port + IPR);
+ snd_emu10k1_intr_enable(emu, epcm->capture_inte);
+ /* printk("adccr = 0x%x, adcbs = 0x%x\n", epcm->adccr, epcm->adcbs); */
+ switch (epcm->type) {
+ case CAPTURE_AC97ADC:
+ snd_emu10k1_ptr_write(emu, ADCCR, 0, epcm->capture_cr_val);
+ break;
+ case CAPTURE_EFX:
+ if (emu->audigy) {
+ snd_emu10k1_ptr_write(emu, A_FXWC1, 0, epcm->capture_cr_val);
+ snd_emu10k1_ptr_write(emu, A_FXWC2, 0, epcm->capture_cr_val2);
+ snd_printdd("cr_val=0x%x, cr_val2=0x%x\n", epcm->capture_cr_val, epcm->capture_cr_val2);
+ } else
+ snd_emu10k1_ptr_write(emu, FXWC, 0, epcm->capture_cr_val);
+ break;
+ default:
+ break;
+ }
+ snd_emu10k1_ptr_write(emu, epcm->capture_bs_reg, 0, epcm->capture_bs_val);
+ epcm->running = 1;
+ epcm->first_ptr = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ epcm->running = 0;
+ snd_emu10k1_intr_disable(emu, epcm->capture_inte);
+ outl(epcm->capture_ipr, emu->port + IPR);
+ snd_emu10k1_ptr_write(emu, epcm->capture_bs_reg, 0, 0);
+ switch (epcm->type) {
+ case CAPTURE_AC97ADC:
+ snd_emu10k1_ptr_write(emu, ADCCR, 0, 0);
+ break;
+ case CAPTURE_EFX:
+ if (emu->audigy) {
+ snd_emu10k1_ptr_write(emu, A_FXWC1, 0, 0);
+ snd_emu10k1_ptr_write(emu, A_FXWC2, 0, 0);
+ } else
+ snd_emu10k1_ptr_write(emu, FXWC, 0, 0);
+ break;
+ default:
+ break;
+ }
+ break;
+ default:
+ result = -EINVAL;
+ }
+ spin_unlock(&emu->reg_lock);
+ return result;
+}
+
+static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ unsigned int ptr;
+
+ if (!epcm->running)
+ return 0;
+ ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->voices[0]->number) & 0x00ffffff;
+#if 0 /* Perex's code */
+ ptr += runtime->buffer_size;
+ ptr -= epcm->ccca_start_addr;
+ ptr %= runtime->buffer_size;
+#else /* EMU10K1 Open Source code from Creative */
+ if (ptr < epcm->ccca_start_addr)
+ ptr += runtime->buffer_size - epcm->ccca_start_addr;
+ else {
+ ptr -= epcm->ccca_start_addr;
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+ }
+#endif
+ /* printk("ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", ptr, runtime->buffer_size, runtime->period_size); */
+ return ptr;
+}
+
+
+static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ int i;
+ int result = 0;
+
+ spin_lock(&emu->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* prepare voices */
+ for (i = 0; i < NUM_EFX_PLAYBACK; i++) {
+ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]);
+ }
+ snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra);
+
+ /* follow thru */
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL);
+ snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 0, 0,
+ &emu->efx_pcm_mixer[0]);
+ for (i = 1; i < NUM_EFX_PLAYBACK; i++)
+ snd_emu10k1_playback_prepare_voice(emu, epcm->voices[i], 0, 0,
+ &emu->efx_pcm_mixer[i]);
+ snd_emu10k1_playback_trigger_voice(emu, epcm->voices[0], 0, 0);
+ snd_emu10k1_playback_trigger_voice(emu, epcm->extra, 1, 1);
+ for (i = 1; i < NUM_EFX_PLAYBACK; i++)
+ snd_emu10k1_playback_trigger_voice(emu, epcm->voices[i], 0, 0);
+ epcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ epcm->running = 0;
+ for (i = 0; i < NUM_EFX_PLAYBACK; i++) {
+ snd_emu10k1_playback_stop_voice(emu, epcm->voices[i]);
+ }
+ snd_emu10k1_playback_stop_voice(emu, epcm->extra);
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ spin_unlock(&emu->reg_lock);
+ return result;
+}
+
+
+static snd_pcm_uframes_t snd_emu10k1_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ unsigned int ptr;
+
+ if (!epcm->running)
+ return 0;
+ if (epcm->first_ptr) {
+ udelay(50); /* hack, it takes awhile until capture is started */
+ epcm->first_ptr = 0;
+ }
+ ptr = snd_emu10k1_ptr_read(emu, epcm->capture_idx_reg, 0) & 0x0000ffff;
+ return bytes_to_frames(runtime, ptr);
+}
+
+/*
+ * Playback support device description
+ */
+
+static struct snd_pcm_hardware snd_emu10k1_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_96000,
+ .rate_min = 4000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+ * Capture support device description
+ */
+
+static struct snd_pcm_hardware snd_emu10k1_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (64*1024),
+ .period_bytes_min = 384,
+ .period_bytes_max = (64*1024),
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_emu10k1_capture_efx =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
+ .rate_min = 44100,
+ .rate_max = 192000,
+ .channels_min = 8,
+ .channels_max = 8,
+ .buffer_bytes_max = (64*1024),
+ .period_bytes_min = 384,
+ .period_bytes_max = (64*1024),
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+/*
+ *
+ */
+
+static void snd_emu10k1_pcm_mixer_notify1(struct snd_emu10k1 *emu, struct snd_kcontrol *kctl, int idx, int activate)
+{
+ struct snd_ctl_elem_id id;
+
+ if (! kctl)
+ return;
+ if (activate)
+ kctl->vd[idx].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ else
+ kctl->vd[idx].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(emu->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ snd_ctl_build_ioff(&id, kctl, idx));
+}
+
+static void snd_emu10k1_pcm_mixer_notify(struct snd_emu10k1 *emu, int idx, int activate)
+{
+ snd_emu10k1_pcm_mixer_notify1(emu, emu->ctl_send_routing, idx, activate);
+ snd_emu10k1_pcm_mixer_notify1(emu, emu->ctl_send_volume, idx, activate);
+ snd_emu10k1_pcm_mixer_notify1(emu, emu->ctl_attn, idx, activate);
+}
+
+static void snd_emu10k1_pcm_efx_mixer_notify(struct snd_emu10k1 *emu, int idx, int activate)
+{
+ snd_emu10k1_pcm_mixer_notify1(emu, emu->ctl_efx_send_routing, idx, activate);
+ snd_emu10k1_pcm_mixer_notify1(emu, emu->ctl_efx_send_volume, idx, activate);
+ snd_emu10k1_pcm_mixer_notify1(emu, emu->ctl_efx_attn, idx, activate);
+}
+
+static void snd_emu10k1_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static int snd_emu10k1_efx_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_pcm_mixer *mix;
+ int i;
+
+ for (i = 0; i < NUM_EFX_PLAYBACK; i++) {
+ mix = &emu->efx_pcm_mixer[i];
+ mix->epcm = NULL;
+ snd_emu10k1_pcm_efx_mixer_notify(emu, i, 0);
+ }
+ return 0;
+}
+
+static int snd_emu10k1_efx_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_pcm *epcm;
+ struct snd_emu10k1_pcm_mixer *mix;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int i;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->type = PLAYBACK_EFX;
+ epcm->substream = substream;
+
+ emu->pcm_playback_efx_substream = substream;
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_emu10k1_pcm_free_substream;
+ runtime->hw = snd_emu10k1_efx_playback;
+
+ for (i = 0; i < NUM_EFX_PLAYBACK; i++) {
+ mix = &emu->efx_pcm_mixer[i];
+ mix->send_routing[0][0] = i;
+ memset(&mix->send_volume, 0, sizeof(mix->send_volume));
+ mix->send_volume[0][0] = 255;
+ mix->attn[0] = 0xffff;
+ mix->epcm = epcm;
+ snd_emu10k1_pcm_efx_mixer_notify(emu, i, 1);
+ }
+ return 0;
+}
+
+static int snd_emu10k1_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_pcm *epcm;
+ struct snd_emu10k1_pcm_mixer *mix;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int i, err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->type = PLAYBACK_EMUVOICE;
+ epcm->substream = substream;
+ runtime->private_data = epcm;
+ runtime->private_free = snd_emu10k1_pcm_free_substream;
+ runtime->hw = snd_emu10k1_playback;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) {
+ kfree(epcm);
+ return err;
+ }
+ if ((err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX)) < 0) {
+ kfree(epcm);
+ return err;
+ }
+ mix = &emu->pcm_mixer[substream->number];
+ for (i = 0; i < 4; i++)
+ mix->send_routing[0][i] = mix->send_routing[1][i] = mix->send_routing[2][i] = i;
+ memset(&mix->send_volume, 0, sizeof(mix->send_volume));
+ mix->send_volume[0][0] = mix->send_volume[0][1] =
+ mix->send_volume[1][0] = mix->send_volume[2][1] = 255;
+ mix->attn[0] = mix->attn[1] = mix->attn[2] = 0xffff;
+ mix->epcm = epcm;
+ snd_emu10k1_pcm_mixer_notify(emu, substream->number, 1);
+ return 0;
+}
+
+static int snd_emu10k1_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_pcm_mixer *mix = &emu->pcm_mixer[substream->number];
+
+ mix->epcm = NULL;
+ snd_emu10k1_pcm_mixer_notify(emu, substream->number, 0);
+ return 0;
+}
+
+static int snd_emu10k1_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->type = CAPTURE_AC97ADC;
+ epcm->substream = substream;
+ epcm->capture_ipr = IPR_ADCBUFFULL|IPR_ADCBUFHALFFULL;
+ epcm->capture_inte = INTE_ADCBUFENABLE;
+ epcm->capture_ba_reg = ADCBA;
+ epcm->capture_bs_reg = ADCBS;
+ epcm->capture_idx_reg = emu->audigy ? A_ADCIDX : ADCIDX;
+ runtime->private_data = epcm;
+ runtime->private_free = snd_emu10k1_pcm_free_substream;
+ runtime->hw = snd_emu10k1_capture;
+ emu->capture_interrupt = snd_emu10k1_pcm_ac97adc_interrupt;
+ emu->pcm_capture_substream = substream;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_capture_rates);
+ return 0;
+}
+
+static int snd_emu10k1_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+
+ emu->capture_interrupt = NULL;
+ emu->pcm_capture_substream = NULL;
+ return 0;
+}
+
+static int snd_emu10k1_capture_mic_open(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->type = CAPTURE_AC97MIC;
+ epcm->substream = substream;
+ epcm->capture_ipr = IPR_MICBUFFULL|IPR_MICBUFHALFFULL;
+ epcm->capture_inte = INTE_MICBUFENABLE;
+ epcm->capture_ba_reg = MICBA;
+ epcm->capture_bs_reg = MICBS;
+ epcm->capture_idx_reg = emu->audigy ? A_MICIDX : MICIDX;
+ substream->runtime->private_data = epcm;
+ substream->runtime->private_free = snd_emu10k1_pcm_free_substream;
+ runtime->hw = snd_emu10k1_capture;
+ runtime->hw.rates = SNDRV_PCM_RATE_8000;
+ runtime->hw.rate_min = runtime->hw.rate_max = 8000;
+ runtime->hw.channels_min = 1;
+ emu->capture_mic_interrupt = snd_emu10k1_pcm_ac97mic_interrupt;
+ emu->pcm_capture_mic_substream = substream;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes);
+ return 0;
+}
+
+static int snd_emu10k1_capture_mic_close(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+
+ emu->capture_interrupt = NULL;
+ emu->pcm_capture_mic_substream = NULL;
+ return 0;
+}
+
+static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int nefx = emu->audigy ? 64 : 32;
+ int idx;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->type = CAPTURE_EFX;
+ epcm->substream = substream;
+ epcm->capture_ipr = IPR_EFXBUFFULL|IPR_EFXBUFHALFFULL;
+ epcm->capture_inte = INTE_EFXBUFENABLE;
+ epcm->capture_ba_reg = FXBA;
+ epcm->capture_bs_reg = FXBS;
+ epcm->capture_idx_reg = FXIDX;
+ substream->runtime->private_data = epcm;
+ substream->runtime->private_free = snd_emu10k1_pcm_free_substream;
+ runtime->hw = snd_emu10k1_capture_efx;
+ runtime->hw.rates = SNDRV_PCM_RATE_48000;
+ runtime->hw.rate_min = runtime->hw.rate_max = 48000;
+ spin_lock_irq(&emu->reg_lock);
+ if (emu->card_capabilities->emu_model) {
+ /* Nb. of channels has been increased to 16 */
+ /* TODO
+ * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE
+ * SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ * SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ * SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000
+ * rate_min = 44100,
+ * rate_max = 192000,
+ * channels_min = 16,
+ * channels_max = 16,
+ * Need to add mixer control to fix sample rate
+ *
+ * There are 32 mono channels of 16bits each.
+ * 24bit Audio uses 2x channels over 16bit
+ * 96kHz uses 2x channels over 48kHz
+ * 192kHz uses 4x channels over 48kHz
+ * So, for 48kHz 24bit, one has 16 channels
+ * for 96kHz 24bit, one has 8 channels
+ * for 192kHz 24bit, one has 4 channels
+ *
+ */
+#if 1
+ switch (emu->emu1010.internal_clock) {
+ case 0:
+ /* For 44.1kHz */
+ runtime->hw.rates = SNDRV_PCM_RATE_44100;
+ runtime->hw.rate_min = runtime->hw.rate_max = 44100;
+ runtime->hw.channels_min =
+ runtime->hw.channels_max = 16;
+ break;
+ case 1:
+ /* For 48kHz */
+ runtime->hw.rates = SNDRV_PCM_RATE_48000;
+ runtime->hw.rate_min = runtime->hw.rate_max = 48000;
+ runtime->hw.channels_min =
+ runtime->hw.channels_max = 16;
+ break;
+ };
+#endif
+#if 0
+ /* For 96kHz */
+ runtime->hw.rates = SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_min = runtime->hw.rate_max = 96000;
+ runtime->hw.channels_min = runtime->hw.channels_max = 4;
+#endif
+#if 0
+ /* For 192kHz */
+ runtime->hw.rates = SNDRV_PCM_RATE_192000;
+ runtime->hw.rate_min = runtime->hw.rate_max = 192000;
+ runtime->hw.channels_min = runtime->hw.channels_max = 2;
+#endif
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
+ /* efx_voices_mask[0] is expected to be zero
+ * efx_voices_mask[1] is expected to have 32bits set
+ */
+ } else {
+ runtime->hw.channels_min = runtime->hw.channels_max = 0;
+ for (idx = 0; idx < nefx; idx++) {
+ if (emu->efx_voices_mask[idx/32] & (1 << (idx%32))) {
+ runtime->hw.channels_min++;
+ runtime->hw.channels_max++;
+ }
+ }
+ }
+ epcm->capture_cr_val = emu->efx_voices_mask[0];
+ epcm->capture_cr_val2 = emu->efx_voices_mask[1];
+ spin_unlock_irq(&emu->reg_lock);
+ emu->capture_efx_interrupt = snd_emu10k1_pcm_efx_interrupt;
+ emu->pcm_capture_efx_substream = substream;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_capture_period_sizes);
+ return 0;
+}
+
+static int snd_emu10k1_capture_efx_close(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+
+ emu->capture_interrupt = NULL;
+ emu->pcm_capture_efx_substream = NULL;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_emu10k1_playback_ops = {
+ .open = snd_emu10k1_playback_open,
+ .close = snd_emu10k1_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1_playback_hw_params,
+ .hw_free = snd_emu10k1_playback_hw_free,
+ .prepare = snd_emu10k1_playback_prepare,
+ .trigger = snd_emu10k1_playback_trigger,
+ .pointer = snd_emu10k1_playback_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static struct snd_pcm_ops snd_emu10k1_capture_ops = {
+ .open = snd_emu10k1_capture_open,
+ .close = snd_emu10k1_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1_capture_hw_params,
+ .hw_free = snd_emu10k1_capture_hw_free,
+ .prepare = snd_emu10k1_capture_prepare,
+ .trigger = snd_emu10k1_capture_trigger,
+ .pointer = snd_emu10k1_capture_pointer,
+};
+
+/* EFX playback */
+static struct snd_pcm_ops snd_emu10k1_efx_playback_ops = {
+ .open = snd_emu10k1_efx_playback_open,
+ .close = snd_emu10k1_efx_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1_playback_hw_params,
+ .hw_free = snd_emu10k1_efx_playback_hw_free,
+ .prepare = snd_emu10k1_efx_playback_prepare,
+ .trigger = snd_emu10k1_efx_playback_trigger,
+ .pointer = snd_emu10k1_efx_playback_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+int __devinit snd_emu10k1_pcm(struct snd_emu10k1 * emu, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(emu->card, "emu10k1", device, 32, 1, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_emu10k1_capture_ops);
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strcpy(pcm->name, "ADC Capture/Standard PCM Playback");
+ emu->pcm = pcm;
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next)
+ if ((err = snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(emu->pci), 64*1024, 64*1024)) < 0)
+ return err;
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; substream; substream = substream->next)
+ snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), 64*1024, 64*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+int __devinit snd_emu10k1_pcm_multi(struct snd_emu10k1 * emu, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(emu->card, "emu10k1", device, 1, 0, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1_efx_playback_ops);
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strcpy(pcm->name, "Multichannel Playback");
+ emu->pcm_multi = pcm;
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next)
+ if ((err = snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(emu->pci), 64*1024, 64*1024)) < 0)
+ return err;
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+
+static struct snd_pcm_ops snd_emu10k1_capture_mic_ops = {
+ .open = snd_emu10k1_capture_mic_open,
+ .close = snd_emu10k1_capture_mic_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1_capture_hw_params,
+ .hw_free = snd_emu10k1_capture_hw_free,
+ .prepare = snd_emu10k1_capture_prepare,
+ .trigger = snd_emu10k1_capture_trigger,
+ .pointer = snd_emu10k1_capture_pointer,
+};
+
+int __devinit snd_emu10k1_pcm_mic(struct snd_emu10k1 * emu, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(emu->card, "emu10k1 mic", device, 0, 1, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_emu10k1_capture_mic_ops);
+
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "Mic Capture");
+ emu->pcm_mic = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), 64*1024, 64*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static int snd_emu10k1_pcm_efx_voices_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int nefx = emu->audigy ? 64 : 32;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = nefx;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_emu10k1_pcm_efx_voices_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int nefx = emu->audigy ? 64 : 32;
+ int idx;
+
+ spin_lock_irq(&emu->reg_lock);
+ for (idx = 0; idx < nefx; idx++)
+ ucontrol->value.integer.value[idx] = (emu->efx_voices_mask[idx / 32] & (1 << (idx % 32))) ? 1 : 0;
+ spin_unlock_irq(&emu->reg_lock);
+ return 0;
+}
+
+static int snd_emu10k1_pcm_efx_voices_mask_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int nval[2], bits;
+ int nefx = emu->audigy ? 64 : 32;
+ int nefxb = emu->audigy ? 7 : 6;
+ int change, idx;
+
+ nval[0] = nval[1] = 0;
+ for (idx = 0, bits = 0; idx < nefx; idx++)
+ if (ucontrol->value.integer.value[idx]) {
+ nval[idx / 32] |= 1 << (idx % 32);
+ bits++;
+ }
+
+ for (idx = 0; idx < nefxb; idx++)
+ if (1 << idx == bits)
+ break;
+
+ if (idx >= nefxb)
+ return -EINVAL;
+
+ spin_lock_irq(&emu->reg_lock);
+ change = (nval[0] != emu->efx_voices_mask[0]) ||
+ (nval[1] != emu->efx_voices_mask[1]);
+ emu->efx_voices_mask[0] = nval[0];
+ emu->efx_voices_mask[1] = nval[1];
+ spin_unlock_irq(&emu->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_emu10k1_pcm_efx_voices_mask = {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "Captured FX8010 Outputs",
+ .info = snd_emu10k1_pcm_efx_voices_mask_info,
+ .get = snd_emu10k1_pcm_efx_voices_mask_get,
+ .put = snd_emu10k1_pcm_efx_voices_mask_put
+};
+
+static struct snd_pcm_ops snd_emu10k1_capture_efx_ops = {
+ .open = snd_emu10k1_capture_efx_open,
+ .close = snd_emu10k1_capture_efx_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1_capture_hw_params,
+ .hw_free = snd_emu10k1_capture_hw_free,
+ .prepare = snd_emu10k1_capture_prepare,
+ .trigger = snd_emu10k1_capture_trigger,
+ .pointer = snd_emu10k1_capture_pointer,
+};
+
+
+/* EFX playback */
+
+#define INITIAL_TRAM_SHIFT 14
+#define INITIAL_TRAM_POS(size) ((((size) / 2) - INITIAL_TRAM_SHIFT) - 1)
+
+static void snd_emu10k1_fx8010_playback_irq(struct snd_emu10k1 *emu, void *private_data)
+{
+ struct snd_pcm_substream *substream = private_data;
+ snd_pcm_period_elapsed(substream);
+}
+
+static void snd_emu10k1_fx8010_playback_tram_poke1(unsigned short *dst_left,
+ unsigned short *dst_right,
+ unsigned short *src,
+ unsigned int count,
+ unsigned int tram_shift)
+{
+ /* printk("tram_poke1: dst_left = 0x%p, dst_right = 0x%p, src = 0x%p, count = 0x%x\n", dst_left, dst_right, src, count); */
+ if ((tram_shift & 1) == 0) {
+ while (count--) {
+ *dst_left-- = *src++;
+ *dst_right-- = *src++;
+ }
+ } else {
+ while (count--) {
+ *dst_right-- = *src++;
+ *dst_left-- = *src++;
+ }
+ }
+}
+
+static void fx8010_pb_trans_copy(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+ unsigned int tram_size = pcm->buffer_size;
+ unsigned short *src = (unsigned short *)(substream->runtime->dma_area + rec->sw_data);
+ unsigned int frames = bytes >> 2, count;
+ unsigned int tram_pos = pcm->tram_pos;
+ unsigned int tram_shift = pcm->tram_shift;
+
+ while (frames > tram_pos) {
+ count = tram_pos + 1;
+ snd_emu10k1_fx8010_playback_tram_poke1((unsigned short *)emu->fx8010.etram_pages.area + tram_pos,
+ (unsigned short *)emu->fx8010.etram_pages.area + tram_pos + tram_size / 2,
+ src, count, tram_shift);
+ src += count * 2;
+ frames -= count;
+ tram_pos = (tram_size / 2) - 1;
+ tram_shift++;
+ }
+ snd_emu10k1_fx8010_playback_tram_poke1((unsigned short *)emu->fx8010.etram_pages.area + tram_pos,
+ (unsigned short *)emu->fx8010.etram_pages.area + tram_pos + tram_size / 2,
+ src, frames, tram_shift);
+ tram_pos -= frames;
+ pcm->tram_pos = tram_pos;
+ pcm->tram_shift = tram_shift;
+}
+
+static int snd_emu10k1_fx8010_playback_transfer(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+
+ snd_pcm_indirect_playback_transfer(substream, &pcm->pcm_rec, fx8010_pb_trans_copy);
+ return 0;
+}
+
+static int snd_emu10k1_fx8010_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_emu10k1_fx8010_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+ unsigned int i;
+
+ for (i = 0; i < pcm->channels; i++)
+ snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + 0x80 + pcm->etram[i], 0, 0);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+ unsigned int i;
+
+ /* printk("prepare: etram_pages = 0x%p, dma_area = 0x%x, buffer_size = 0x%x (0x%x)\n", emu->fx8010.etram_pages, runtime->dma_area, runtime->buffer_size, runtime->buffer_size << 2); */
+ memset(&pcm->pcm_rec, 0, sizeof(pcm->pcm_rec));
+ pcm->pcm_rec.hw_buffer_size = pcm->buffer_size * 2; /* byte size */
+ pcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ pcm->tram_pos = INITIAL_TRAM_POS(pcm->buffer_size);
+ pcm->tram_shift = 0;
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_running, 0, 0); /* reset */
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 0); /* reset */
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_size, 0, runtime->buffer_size);
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_ptr, 0, 0); /* reset ptr number */
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_count, 0, runtime->period_size);
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_tmpcount, 0, runtime->period_size);
+ for (i = 0; i < pcm->channels; i++)
+ snd_emu10k1_ptr_write(emu, TANKMEMADDRREGBASE + 0x80 + pcm->etram[i], 0, (TANKMEMADDRREG_READ|TANKMEMADDRREG_ALIGN) + i * (runtime->buffer_size / pcm->channels));
+ return 0;
+}
+
+static int snd_emu10k1_fx8010_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+ int result = 0;
+
+ spin_lock(&emu->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* follow thru */
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+#ifdef EMU10K1_SET_AC3_IEC958
+ {
+ int i;
+ for (i = 0; i < 3; i++) {
+ unsigned int bits;
+ bits = SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS |
+ 0x00001200 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT | SPCS_NOTAUDIODATA;
+ snd_emu10k1_ptr_write(emu, SPCS0 + i, 0, bits);
+ }
+ }
+#endif
+ result = snd_emu10k1_fx8010_register_irq_handler(emu, snd_emu10k1_fx8010_playback_irq, pcm->gpr_running, substream, &pcm->irq);
+ if (result < 0)
+ goto __err;
+ snd_emu10k1_fx8010_playback_transfer(substream); /* roll the ball */
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ snd_emu10k1_fx8010_unregister_irq_handler(emu, pcm->irq); pcm->irq = NULL;
+ snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 0);
+ pcm->tram_pos = INITIAL_TRAM_POS(pcm->buffer_size);
+ pcm->tram_shift = 0;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ __err:
+ spin_unlock(&emu->reg_lock);
+ return result;
+}
+
+static snd_pcm_uframes_t snd_emu10k1_fx8010_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+ size_t ptr; /* byte pointer */
+
+ if (!snd_emu10k1_ptr_read(emu, emu->gpr_base + pcm->gpr_trigger, 0))
+ return 0;
+ ptr = snd_emu10k1_ptr_read(emu, emu->gpr_base + pcm->gpr_ptr, 0) << 2;
+ return snd_pcm_indirect_playback_pointer(substream, &pcm->pcm_rec, ptr);
+}
+
+static struct snd_pcm_hardware snd_emu10k1_fx8010_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_RESUME |
+ /* SNDRV_PCM_INFO_MMAP_VALID | */ SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 1024,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_emu10k1_fx8010_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+
+ runtime->hw = snd_emu10k1_fx8010_playback;
+ runtime->hw.channels_min = runtime->hw.channels_max = pcm->channels;
+ runtime->hw.period_bytes_max = (pcm->buffer_size * 2) / 2;
+ spin_lock_irq(&emu->reg_lock);
+ if (pcm->valid == 0) {
+ spin_unlock_irq(&emu->reg_lock);
+ return -ENODEV;
+ }
+ pcm->opened = 1;
+ spin_unlock_irq(&emu->reg_lock);
+ return 0;
+}
+
+static int snd_emu10k1_fx8010_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
+
+ spin_lock_irq(&emu->reg_lock);
+ pcm->opened = 0;
+ spin_unlock_irq(&emu->reg_lock);
+ return 0;
+}
+
+static struct snd_pcm_ops snd_emu10k1_fx8010_playback_ops = {
+ .open = snd_emu10k1_fx8010_playback_open,
+ .close = snd_emu10k1_fx8010_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_emu10k1_fx8010_playback_hw_params,
+ .hw_free = snd_emu10k1_fx8010_playback_hw_free,
+ .prepare = snd_emu10k1_fx8010_playback_prepare,
+ .trigger = snd_emu10k1_fx8010_playback_trigger,
+ .pointer = snd_emu10k1_fx8010_playback_pointer,
+ .ack = snd_emu10k1_fx8010_playback_transfer,
+};
+
+int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ struct snd_kcontrol *kctl;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(emu->card, "emu10k1 efx", device, 8, 1, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_emu10k1_fx8010_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_emu10k1_capture_efx_ops);
+
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "Multichannel Capture/PT Playback");
+ emu->pcm_efx = pcm;
+ if (rpcm)
+ *rpcm = pcm;
+
+ /* EFX capture - record the "FXBUS2" channels, by default we connect the EXTINs
+ * to these
+ */
+
+ /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */
+ if (emu->audigy) {
+ emu->efx_voices_mask[0] = 0;
+ if (emu->card_capabilities->emu_model)
+ /* Pavel Hofman - 32 voices will be used for
+ * capture (write mode) -
+ * each bit = corresponding voice
+ */
+ emu->efx_voices_mask[1] = 0xffffffff;
+ else
+ emu->efx_voices_mask[1] = 0xffff;
+ } else {
+ emu->efx_voices_mask[0] = 0xffff0000;
+ emu->efx_voices_mask[1] = 0;
+ }
+ /* For emu1010, the control has to set 32 upper bits (voices)
+ * out of the 64 bits (voices) to true for the 16-channels capture
+ * to work correctly. Correct A_FXWC2 initial value (0xffffffff)
+ * is already defined but the snd_emu10k1_pcm_efx_voices_mask
+ * control can override this register's value.
+ */
+ kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu);
+ if (!kctl)
+ return -ENOMEM;
+ kctl->id.device = device;
+ snd_ctl_add(emu->card, kctl);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), 64*1024, 64*1024);
+
+ return 0;
+}
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
new file mode 100644
index 0000000..216f974
--- /dev/null
+++ b/sound/pci/emu10k1/emuproc.c
@@ -0,0 +1,674 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Creative Labs, Inc.
+ * Routines for control of EMU10K1 chips / proc interface routines
+ *
+ * Copyright (c) by James Courtier-Dutton <James@superbug.co.uk>
+ * Added EMU 1010 support.
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+#include "p16v.h"
+
+#ifdef CONFIG_PROC_FS
+static void snd_emu10k1_proc_spdif_status(struct snd_emu10k1 * emu,
+ struct snd_info_buffer *buffer,
+ char *title,
+ int status_reg,
+ int rate_reg)
+{
+ static char *clkaccy[4] = { "1000ppm", "50ppm", "variable", "unknown" };
+ static int samplerate[16] = { 44100, 1, 48000, 32000, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15 };
+ static char *channel[16] = { "unspec", "left", "right", "3", "4", "5", "6", "7", "8", "9", "10", "11", "12", "13", "14", "15" };
+ static char *emphasis[8] = { "none", "50/15 usec 2 channel", "2", "3", "4", "5", "6", "7" };
+ unsigned int status, rate = 0;
+
+ status = snd_emu10k1_ptr_read(emu, status_reg, 0);
+
+ snd_iprintf(buffer, "\n%s\n", title);
+
+ if (status != 0xffffffff) {
+ snd_iprintf(buffer, "Professional Mode : %s\n", (status & SPCS_PROFESSIONAL) ? "yes" : "no");
+ snd_iprintf(buffer, "Not Audio Data : %s\n", (status & SPCS_NOTAUDIODATA) ? "yes" : "no");
+ snd_iprintf(buffer, "Copyright : %s\n", (status & SPCS_COPYRIGHT) ? "yes" : "no");
+ snd_iprintf(buffer, "Emphasis : %s\n", emphasis[(status & SPCS_EMPHASISMASK) >> 3]);
+ snd_iprintf(buffer, "Mode : %i\n", (status & SPCS_MODEMASK) >> 6);
+ snd_iprintf(buffer, "Category Code : 0x%x\n", (status & SPCS_CATEGORYCODEMASK) >> 8);
+ snd_iprintf(buffer, "Generation Status : %s\n", status & SPCS_GENERATIONSTATUS ? "original" : "copy");
+ snd_iprintf(buffer, "Source Mask : %i\n", (status & SPCS_SOURCENUMMASK) >> 16);
+ snd_iprintf(buffer, "Channel Number : %s\n", channel[(status & SPCS_CHANNELNUMMASK) >> 20]);
+ snd_iprintf(buffer, "Sample Rate : %iHz\n", samplerate[(status & SPCS_SAMPLERATEMASK) >> 24]);
+ snd_iprintf(buffer, "Clock Accuracy : %s\n", clkaccy[(status & SPCS_CLKACCYMASK) >> 28]);
+
+ if (rate_reg > 0) {
+ rate = snd_emu10k1_ptr_read(emu, rate_reg, 0);
+ snd_iprintf(buffer, "S/PDIF Valid : %s\n", rate & SRCS_SPDIFVALID ? "on" : "off");
+ snd_iprintf(buffer, "S/PDIF Locked : %s\n", rate & SRCS_SPDIFLOCKED ? "on" : "off");
+ snd_iprintf(buffer, "Rate Locked : %s\n", rate & SRCS_RATELOCKED ? "on" : "off");
+ /* From ((Rate * 48000 ) / 262144); */
+ snd_iprintf(buffer, "Estimated Sample Rate : %d\n", ((rate & 0xFFFFF ) * 375) >> 11);
+ }
+ } else {
+ snd_iprintf(buffer, "No signal detected.\n");
+ }
+
+}
+
+static void snd_emu10k1_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ /* FIXME - output names are in emufx.c too */
+ static char *creative_outs[32] = {
+ /* 00 */ "AC97 Left",
+ /* 01 */ "AC97 Right",
+ /* 02 */ "Optical IEC958 Left",
+ /* 03 */ "Optical IEC958 Right",
+ /* 04 */ "Center",
+ /* 05 */ "LFE",
+ /* 06 */ "Headphone Left",
+ /* 07 */ "Headphone Right",
+ /* 08 */ "Surround Left",
+ /* 09 */ "Surround Right",
+ /* 10 */ "PCM Capture Left",
+ /* 11 */ "PCM Capture Right",
+ /* 12 */ "MIC Capture",
+ /* 13 */ "AC97 Surround Left",
+ /* 14 */ "AC97 Surround Right",
+ /* 15 */ "???",
+ /* 16 */ "???",
+ /* 17 */ "Analog Center",
+ /* 18 */ "Analog LFE",
+ /* 19 */ "???",
+ /* 20 */ "???",
+ /* 21 */ "???",
+ /* 22 */ "???",
+ /* 23 */ "???",
+ /* 24 */ "???",
+ /* 25 */ "???",
+ /* 26 */ "???",
+ /* 27 */ "???",
+ /* 28 */ "???",
+ /* 29 */ "???",
+ /* 30 */ "???",
+ /* 31 */ "???"
+ };
+
+ static char *audigy_outs[64] = {
+ /* 00 */ "Digital Front Left",
+ /* 01 */ "Digital Front Right",
+ /* 02 */ "Digital Center",
+ /* 03 */ "Digital LEF",
+ /* 04 */ "Headphone Left",
+ /* 05 */ "Headphone Right",
+ /* 06 */ "Digital Rear Left",
+ /* 07 */ "Digital Rear Right",
+ /* 08 */ "Front Left",
+ /* 09 */ "Front Right",
+ /* 10 */ "Center",
+ /* 11 */ "LFE",
+ /* 12 */ "???",
+ /* 13 */ "???",
+ /* 14 */ "Rear Left",
+ /* 15 */ "Rear Right",
+ /* 16 */ "AC97 Front Left",
+ /* 17 */ "AC97 Front Right",
+ /* 18 */ "ADC Caputre Left",
+ /* 19 */ "ADC Capture Right",
+ /* 20 */ "???",
+ /* 21 */ "???",
+ /* 22 */ "???",
+ /* 23 */ "???",
+ /* 24 */ "???",
+ /* 25 */ "???",
+ /* 26 */ "???",
+ /* 27 */ "???",
+ /* 28 */ "???",
+ /* 29 */ "???",
+ /* 30 */ "???",
+ /* 31 */ "???",
+ /* 32 */ "FXBUS2_0",
+ /* 33 */ "FXBUS2_1",
+ /* 34 */ "FXBUS2_2",
+ /* 35 */ "FXBUS2_3",
+ /* 36 */ "FXBUS2_4",
+ /* 37 */ "FXBUS2_5",
+ /* 38 */ "FXBUS2_6",
+ /* 39 */ "FXBUS2_7",
+ /* 40 */ "FXBUS2_8",
+ /* 41 */ "FXBUS2_9",
+ /* 42 */ "FXBUS2_10",
+ /* 43 */ "FXBUS2_11",
+ /* 44 */ "FXBUS2_12",
+ /* 45 */ "FXBUS2_13",
+ /* 46 */ "FXBUS2_14",
+ /* 47 */ "FXBUS2_15",
+ /* 48 */ "FXBUS2_16",
+ /* 49 */ "FXBUS2_17",
+ /* 50 */ "FXBUS2_18",
+ /* 51 */ "FXBUS2_19",
+ /* 52 */ "FXBUS2_20",
+ /* 53 */ "FXBUS2_21",
+ /* 54 */ "FXBUS2_22",
+ /* 55 */ "FXBUS2_23",
+ /* 56 */ "FXBUS2_24",
+ /* 57 */ "FXBUS2_25",
+ /* 58 */ "FXBUS2_26",
+ /* 59 */ "FXBUS2_27",
+ /* 60 */ "FXBUS2_28",
+ /* 61 */ "FXBUS2_29",
+ /* 62 */ "FXBUS2_30",
+ /* 63 */ "FXBUS2_31"
+ };
+
+ struct snd_emu10k1 *emu = entry->private_data;
+ unsigned int val, val1;
+ int nefx = emu->audigy ? 64 : 32;
+ char **outputs = emu->audigy ? audigy_outs : creative_outs;
+ int idx;
+
+ snd_iprintf(buffer, "EMU10K1\n\n");
+ snd_iprintf(buffer, "Card : %s\n",
+ emu->audigy ? "Audigy" : (emu->card_capabilities->ecard ? "EMU APS" : "Creative"));
+ snd_iprintf(buffer, "Internal TRAM (words) : 0x%x\n", emu->fx8010.itram_size);
+ snd_iprintf(buffer, "External TRAM (words) : 0x%x\n", (int)emu->fx8010.etram_pages.bytes / 2);
+ snd_iprintf(buffer, "\n");
+ snd_iprintf(buffer, "Effect Send Routing :\n");
+ for (idx = 0; idx < NUM_G; idx++) {
+ val = emu->audigy ?
+ snd_emu10k1_ptr_read(emu, A_FXRT1, idx) :
+ snd_emu10k1_ptr_read(emu, FXRT, idx);
+ val1 = emu->audigy ?
+ snd_emu10k1_ptr_read(emu, A_FXRT2, idx) :
+ 0;
+ if (emu->audigy) {
+ snd_iprintf(buffer, "Ch%i: A=%i, B=%i, C=%i, D=%i, ",
+ idx,
+ val & 0x3f,
+ (val >> 8) & 0x3f,
+ (val >> 16) & 0x3f,
+ (val >> 24) & 0x3f);
+ snd_iprintf(buffer, "E=%i, F=%i, G=%i, H=%i\n",
+ val1 & 0x3f,
+ (val1 >> 8) & 0x3f,
+ (val1 >> 16) & 0x3f,
+ (val1 >> 24) & 0x3f);
+ } else {
+ snd_iprintf(buffer, "Ch%i: A=%i, B=%i, C=%i, D=%i\n",
+ idx,
+ (val >> 16) & 0x0f,
+ (val >> 20) & 0x0f,
+ (val >> 24) & 0x0f,
+ (val >> 28) & 0x0f);
+ }
+ }
+ snd_iprintf(buffer, "\nCaptured FX Outputs :\n");
+ for (idx = 0; idx < nefx; idx++) {
+ if (emu->efx_voices_mask[idx/32] & (1 << (idx%32)))
+ snd_iprintf(buffer, " Output %02i [%s]\n", idx, outputs[idx]);
+ }
+ snd_iprintf(buffer, "\nAll FX Outputs :\n");
+ for (idx = 0; idx < (emu->audigy ? 64 : 32); idx++)
+ snd_iprintf(buffer, " Output %02i [%s]\n", idx, outputs[idx]);
+}
+
+static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_emu10k1 *emu = entry->private_data;
+ u32 value;
+ u32 value2;
+ unsigned long flags;
+ u32 rate;
+
+ if (emu->card_capabilities->emu_model) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x38, &value);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ if ((value & 0x1) == 0) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x2a, &value);
+ snd_emu1010_fpga_read(emu, 0x2b, &value2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ rate = 0x1770000 / (((value << 5) | value2)+1);
+ snd_iprintf(buffer, "ADAT Locked : %u\n", rate);
+ } else {
+ snd_iprintf(buffer, "ADAT Unlocked\n");
+ }
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x20, &value);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ if ((value & 0x4) == 0) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x28, &value);
+ snd_emu1010_fpga_read(emu, 0x29, &value2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ rate = 0x1770000 / (((value << 5) | value2)+1);
+ snd_iprintf(buffer, "SPDIF Locked : %d\n", rate);
+ } else {
+ snd_iprintf(buffer, "SPDIF Unlocked\n");
+ }
+ } else {
+ snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS);
+ snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS);
+ }
+#if 0
+ val = snd_emu10k1_ptr_read(emu, ZVSRCS, 0);
+ snd_iprintf(buffer, "\nZoomed Video\n");
+ snd_iprintf(buffer, "Rate Locked : %s\n", val & SRCS_RATELOCKED ? "on" : "off");
+ snd_iprintf(buffer, "Estimated Sample Rate : 0x%x\n", val & SRCS_ESTSAMPLERATE);
+#endif
+}
+
+static void snd_emu10k1_proc_rates_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ static int samplerate[8] = { 44100, 48000, 96000, 192000, 4, 5, 6, 7 };
+ struct snd_emu10k1 *emu = entry->private_data;
+ unsigned int val, tmp, n;
+ val = snd_emu10k1_ptr20_read(emu, CAPTURE_RATE_STATUS, 0);
+ tmp = (val >> 16) & 0x8;
+ for (n = 0; n < 4; n++) {
+ tmp = val >> (16 + (n*4));
+ if (tmp & 0x8) snd_iprintf(buffer, "Channel %d: Rate=%d\n", n, samplerate[tmp & 0x7]);
+ else snd_iprintf(buffer, "Channel %d: No input\n", n);
+ }
+}
+
+static void snd_emu10k1_proc_acode_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ u32 pc;
+ struct snd_emu10k1 *emu = entry->private_data;
+
+ snd_iprintf(buffer, "FX8010 Instruction List '%s'\n", emu->fx8010.name);
+ snd_iprintf(buffer, " Code dump :\n");
+ for (pc = 0; pc < (emu->audigy ? 1024 : 512); pc++) {
+ u32 low, high;
+
+ low = snd_emu10k1_efx_read(emu, pc * 2);
+ high = snd_emu10k1_efx_read(emu, pc * 2 + 1);
+ if (emu->audigy)
+ snd_iprintf(buffer, " OP(0x%02x, 0x%03x, 0x%03x, 0x%03x, 0x%03x) /* 0x%04x: 0x%08x%08x */\n",
+ (high >> 24) & 0x0f,
+ (high >> 12) & 0x7ff,
+ (high >> 0) & 0x7ff,
+ (low >> 12) & 0x7ff,
+ (low >> 0) & 0x7ff,
+ pc,
+ high, low);
+ else
+ snd_iprintf(buffer, " OP(0x%02x, 0x%03x, 0x%03x, 0x%03x, 0x%03x) /* 0x%04x: 0x%08x%08x */\n",
+ (high >> 20) & 0x0f,
+ (high >> 10) & 0x3ff,
+ (high >> 0) & 0x3ff,
+ (low >> 10) & 0x3ff,
+ (low >> 0) & 0x3ff,
+ pc,
+ high, low);
+ }
+}
+
+#define TOTAL_SIZE_GPR (0x100*4)
+#define A_TOTAL_SIZE_GPR (0x200*4)
+#define TOTAL_SIZE_TANKMEM_DATA (0xa0*4)
+#define TOTAL_SIZE_TANKMEM_ADDR (0xa0*4)
+#define A_TOTAL_SIZE_TANKMEM_DATA (0x100*4)
+#define A_TOTAL_SIZE_TANKMEM_ADDR (0x100*4)
+#define TOTAL_SIZE_CODE (0x200*8)
+#define A_TOTAL_SIZE_CODE (0x400*8)
+
+static long snd_emu10k1_fx8010_read(struct snd_info_entry *entry,
+ void *file_private_data,
+ struct file *file, char __user *buf,
+ unsigned long count, unsigned long pos)
+{
+ long size;
+ struct snd_emu10k1 *emu = entry->private_data;
+ unsigned int offset;
+ int tram_addr = 0;
+
+ if (!strcmp(entry->name, "fx8010_tram_addr")) {
+ offset = TANKMEMADDRREGBASE;
+ tram_addr = 1;
+ } else if (!strcmp(entry->name, "fx8010_tram_data")) {
+ offset = TANKMEMDATAREGBASE;
+ } else if (!strcmp(entry->name, "fx8010_code")) {
+ offset = emu->audigy ? A_MICROCODEBASE : MICROCODEBASE;
+ } else {
+ offset = emu->audigy ? A_FXGPREGBASE : FXGPREGBASE;
+ }
+ size = count;
+ if (pos + size > entry->size)
+ size = (long)entry->size - pos;
+ if (size > 0) {
+ unsigned int *tmp;
+ long res;
+ unsigned int idx;
+ if ((tmp = kmalloc(size + 8, GFP_KERNEL)) == NULL)
+ return -ENOMEM;
+ for (idx = 0; idx < ((pos & 3) + size + 3) >> 2; idx++)
+ if (tram_addr && emu->audigy) {
+ tmp[idx] = snd_emu10k1_ptr_read(emu, offset + idx + (pos >> 2), 0) >> 11;
+ tmp[idx] |= snd_emu10k1_ptr_read(emu, 0x100 + idx + (pos >> 2), 0) << 20;
+ } else
+ tmp[idx] = snd_emu10k1_ptr_read(emu, offset + idx + (pos >> 2), 0);
+ if (copy_to_user(buf, ((char *)tmp) + (pos & 3), size))
+ res = -EFAULT;
+ else {
+ res = size;
+ }
+ kfree(tmp);
+ return res;
+ }
+ return 0;
+}
+
+static void snd_emu10k1_proc_voices_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_emu10k1 *emu = entry->private_data;
+ struct snd_emu10k1_voice *voice;
+ int idx;
+
+ snd_iprintf(buffer, "ch\tuse\tpcm\tefx\tsynth\tmidi\n");
+ for (idx = 0; idx < NUM_G; idx++) {
+ voice = &emu->voices[idx];
+ snd_iprintf(buffer, "%i\t%i\t%i\t%i\t%i\t%i\n",
+ idx,
+ voice->use,
+ voice->pcm,
+ voice->efx,
+ voice->synth,
+ voice->midi);
+ }
+}
+
+#ifdef CONFIG_SND_DEBUG
+static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_emu10k1 *emu = entry->private_data;
+ u32 value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "EMU1010 Registers:\n\n");
+
+ for(i = 0; i < 0x40; i+=1) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, i, &value);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "%02X: %08X, %02X\n", i, value, (value >> 8) & 0x7f);
+ }
+}
+
+static void snd_emu_proc_io_reg_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_emu10k1 *emu = entry->private_data;
+ unsigned long value;
+ unsigned long flags;
+ int i;
+ snd_iprintf(buffer, "IO Registers:\n\n");
+ for(i = 0; i < 0x40; i+=4) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ value = inl(emu->port + i);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ snd_iprintf(buffer, "%02X: %08lX\n", i, value);
+ }
+}
+
+static void snd_emu_proc_io_reg_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_emu10k1 *emu = entry->private_data;
+ unsigned long flags;
+ char line[64];
+ u32 reg, val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff) ) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(val, emu->port + (reg & 0xfffffffc));
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ }
+ }
+}
+
+static unsigned int snd_ptr_read(struct snd_emu10k1 * emu,
+ unsigned int iobase,
+ unsigned int reg,
+ unsigned int chn)
+{
+ unsigned long flags;
+ unsigned int regptr, val;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + iobase + PTR);
+ val = inl(emu->port + iobase + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+static void snd_ptr_write(struct snd_emu10k1 *emu,
+ unsigned int iobase,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data)
+{
+ unsigned int regptr;
+ unsigned long flags;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + iobase + PTR);
+ outl(data, emu->port + iobase + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+
+static void snd_emu_proc_ptr_reg_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer, int iobase, int offset, int length, int voices)
+{
+ struct snd_emu10k1 *emu = entry->private_data;
+ unsigned long value;
+ int i,j;
+ if (offset+length > 0xa0) {
+ snd_iprintf(buffer, "Input values out of range\n");
+ return;
+ }
+ snd_iprintf(buffer, "Registers 0x%x\n", iobase);
+ for(i = offset; i < offset+length; i++) {
+ snd_iprintf(buffer, "%02X: ",i);
+ for (j = 0; j < voices; j++) {
+ if(iobase == 0)
+ value = snd_ptr_read(emu, 0, i, j);
+ else
+ value = snd_ptr_read(emu, 0x20, i, j);
+ snd_iprintf(buffer, "%08lX ", value);
+ }
+ snd_iprintf(buffer, "\n");
+ }
+}
+
+static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer, int iobase)
+{
+ struct snd_emu10k1 *emu = entry->private_data;
+ char line[64];
+ unsigned int reg, channel_id , val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
+ continue;
+ if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) )
+ snd_ptr_write(emu, iobase, reg, channel_id, val);
+ }
+}
+
+static void snd_emu_proc_ptr_reg_write00(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_emu_proc_ptr_reg_write(entry, buffer, 0);
+}
+
+static void snd_emu_proc_ptr_reg_write20(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_emu_proc_ptr_reg_write(entry, buffer, 0x20);
+}
+
+
+static void snd_emu_proc_ptr_reg_read00a(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_emu_proc_ptr_reg_read(entry, buffer, 0, 0, 0x40, 64);
+}
+
+static void snd_emu_proc_ptr_reg_read00b(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_emu_proc_ptr_reg_read(entry, buffer, 0, 0x40, 0x40, 64);
+}
+
+static void snd_emu_proc_ptr_reg_read20a(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_emu_proc_ptr_reg_read(entry, buffer, 0x20, 0, 0x40, 4);
+}
+
+static void snd_emu_proc_ptr_reg_read20b(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ snd_emu_proc_ptr_reg_read(entry, buffer, 0x20, 0x40, 0x40, 4);
+}
+
+static void snd_emu_proc_ptr_reg_read20c(struct snd_info_entry *entry,
+ struct snd_info_buffer * buffer)
+{
+ snd_emu_proc_ptr_reg_read(entry, buffer, 0x20, 0x80, 0x20, 4);
+}
+#endif
+
+static struct snd_info_entry_ops snd_emu10k1_proc_ops_fx8010 = {
+ .read = snd_emu10k1_fx8010_read,
+};
+
+int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu)
+{
+ struct snd_info_entry *entry;
+#ifdef CONFIG_SND_DEBUG
+ if (emu->card_capabilities->emu_model) {
+ if (! snd_card_proc_new(emu->card, "emu1010_regs", &entry))
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read);
+ }
+ if (! snd_card_proc_new(emu->card, "io_regs", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read);
+ entry->c.text.write = snd_emu_proc_io_reg_write;
+ entry->mode |= S_IWUSR;
+ }
+ if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a);
+ entry->c.text.write = snd_emu_proc_ptr_reg_write00;
+ entry->mode |= S_IWUSR;
+ }
+ if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b);
+ entry->c.text.write = snd_emu_proc_ptr_reg_write00;
+ entry->mode |= S_IWUSR;
+ }
+ if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a);
+ entry->c.text.write = snd_emu_proc_ptr_reg_write20;
+ entry->mode |= S_IWUSR;
+ }
+ if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b);
+ entry->c.text.write = snd_emu_proc_ptr_reg_write20;
+ entry->mode |= S_IWUSR;
+ }
+ if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) {
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c);
+ entry->c.text.write = snd_emu_proc_ptr_reg_write20;
+ entry->mode |= S_IWUSR;
+ }
+#endif
+
+ if (! snd_card_proc_new(emu->card, "emu10k1", &entry))
+ snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_read);
+
+ if (emu->card_capabilities->emu10k2_chip) {
+ if (! snd_card_proc_new(emu->card, "spdif-in", &entry))
+ snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_spdif_read);
+ }
+ if (emu->card_capabilities->ca0151_chip) {
+ if (! snd_card_proc_new(emu->card, "capture-rates", &entry))
+ snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_rates_read);
+ }
+
+ if (! snd_card_proc_new(emu->card, "voices", &entry))
+ snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_voices_read);
+
+ if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = emu;
+ entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/;
+ entry->size = emu->audigy ? A_TOTAL_SIZE_GPR : TOTAL_SIZE_GPR;
+ entry->c.ops = &snd_emu10k1_proc_ops_fx8010;
+ }
+ if (! snd_card_proc_new(emu->card, "fx8010_tram_data", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = emu;
+ entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/;
+ entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_DATA : TOTAL_SIZE_TANKMEM_DATA ;
+ entry->c.ops = &snd_emu10k1_proc_ops_fx8010;
+ }
+ if (! snd_card_proc_new(emu->card, "fx8010_tram_addr", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = emu;
+ entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/;
+ entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_ADDR : TOTAL_SIZE_TANKMEM_ADDR ;
+ entry->c.ops = &snd_emu10k1_proc_ops_fx8010;
+ }
+ if (! snd_card_proc_new(emu->card, "fx8010_code", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = emu;
+ entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/;
+ entry->size = emu->audigy ? A_TOTAL_SIZE_CODE : TOTAL_SIZE_CODE;
+ entry->c.ops = &snd_emu10k1_proc_ops_fx8010;
+ }
+ if (! snd_card_proc_new(emu->card, "fx8010_acode", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = emu;
+ entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/;
+ entry->c.text.read = snd_emu10k1_proc_acode_read;
+ }
+ return 0;
+}
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
new file mode 100644
index 0000000..b5a802b
--- /dev/null
+++ b/sound/pci/emu10k1/io.c
@@ -0,0 +1,580 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Creative Labs, Inc.
+ * Routines for control of EMU10K1 chips
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+#include <linux/delay.h>
+#include "p17v.h"
+
+unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, unsigned int chn)
+{
+ unsigned long flags;
+ unsigned int regptr, val;
+ unsigned int mask;
+
+ mask = emu->audigy ? A_PTR_ADDRESS_MASK : PTR_ADDRESS_MASK;
+ regptr = ((reg << 16) & mask) | (chn & PTR_CHANNELNUM_MASK);
+
+ if (reg & 0xff000000) {
+ unsigned char size, offset;
+
+ size = (reg >> 24) & 0x3f;
+ offset = (reg >> 16) & 0x1f;
+ mask = ((1 << size) - 1) << offset;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+
+ return (val & mask) >> offset;
+ } else {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+ }
+}
+
+EXPORT_SYMBOL(snd_emu10k1_ptr_read);
+
+void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data)
+{
+ unsigned int regptr;
+ unsigned long flags;
+ unsigned int mask;
+
+ if (!emu) {
+ snd_printk(KERN_ERR "ptr_write: emu is null!\n");
+ dump_stack();
+ return;
+ }
+ mask = emu->audigy ? A_PTR_ADDRESS_MASK : PTR_ADDRESS_MASK;
+ regptr = ((reg << 16) & mask) | (chn & PTR_CHANNELNUM_MASK);
+
+ if (reg & 0xff000000) {
+ unsigned char size, offset;
+
+ size = (reg >> 24) & 0x3f;
+ offset = (reg >> 16) & 0x1f;
+ mask = ((1 << size) - 1) << offset;
+ data = (data << offset) & mask;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ data |= inl(emu->port + DATA) & ~mask;
+ outl(data, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ } else {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + PTR);
+ outl(data, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ }
+}
+
+EXPORT_SYMBOL(snd_emu10k1_ptr_write);
+
+unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu,
+ unsigned int reg,
+ unsigned int chn)
+{
+ unsigned long flags;
+ unsigned int regptr, val;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + 0x20 + PTR);
+ val = inl(emu->port + 0x20 + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data)
+{
+ unsigned int regptr;
+ unsigned long flags;
+
+ regptr = (reg << 16) | chn;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(regptr, emu->port + 0x20 + PTR);
+ outl(data, emu->port + 0x20 + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+int snd_emu10k1_spi_write(struct snd_emu10k1 * emu,
+ unsigned int data)
+{
+ unsigned int reset, set;
+ unsigned int reg, tmp;
+ int n, result;
+ int err = 0;
+
+ /* This function is not re-entrant, so protect against it. */
+ spin_lock(&emu->spi_lock);
+ if (emu->card_capabilities->ca0108_chip)
+ reg = 0x3c; /* PTR20, reg 0x3c */
+ else {
+ /* For other chip types the SPI register
+ * is currently unknown. */
+ err = 1;
+ goto spi_write_exit;
+ }
+ if (data > 0xffff) {
+ /* Only 16bit values allowed */
+ err = 1;
+ goto spi_write_exit;
+ }
+
+ tmp = snd_emu10k1_ptr20_read(emu, reg, 0);
+ reset = (tmp & ~0x3ffff) | 0x20000; /* Set xxx20000 */
+ set = reset | 0x10000; /* Set xxx1xxxx */
+ snd_emu10k1_ptr20_write(emu, reg, 0, reset | data);
+ tmp = snd_emu10k1_ptr20_read(emu, reg, 0); /* write post */
+ snd_emu10k1_ptr20_write(emu, reg, 0, set | data);
+ result = 1;
+ /* Wait for status bit to return to 0 */
+ for (n = 0; n < 100; n++) {
+ udelay(10);
+ tmp = snd_emu10k1_ptr20_read(emu, reg, 0);
+ if (!(tmp & 0x10000)) {
+ result = 0;
+ break;
+ }
+ }
+ if (result) {
+ /* Timed out */
+ err = 1;
+ goto spi_write_exit;
+ }
+ snd_emu10k1_ptr20_write(emu, reg, 0, reset | data);
+ tmp = snd_emu10k1_ptr20_read(emu, reg, 0); /* Write post */
+ err = 0;
+spi_write_exit:
+ spin_unlock(&emu->spi_lock);
+ return err;
+}
+
+/* The ADC does not support i2c read, so only write is implemented */
+int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu,
+ u32 reg,
+ u32 value)
+{
+ u32 tmp;
+ int timeout = 0;
+ int status;
+ int retry;
+ int err = 0;
+
+ if ((reg > 0x7f) || (value > 0x1ff)) {
+ snd_printk(KERN_ERR "i2c_write: invalid values.\n");
+ return -EINVAL;
+ }
+
+ /* This function is not re-entrant, so protect against it. */
+ spin_lock(&emu->i2c_lock);
+
+ tmp = reg << 25 | value << 16;
+
+ /* This controls the I2C connected to the WM8775 ADC Codec */
+ snd_emu10k1_ptr20_write(emu, P17V_I2C_1, 0, tmp);
+ tmp = snd_emu10k1_ptr20_read(emu, P17V_I2C_1, 0); /* write post */
+
+ for (retry = 0; retry < 10; retry++) {
+ /* Send the data to i2c */
+ tmp = 0;
+ tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD);
+ snd_emu10k1_ptr20_write(emu, P17V_I2C_ADDR, 0, tmp);
+
+ /* Wait till the transaction ends */
+ while (1) {
+ mdelay(1);
+ status = snd_emu10k1_ptr20_read(emu, P17V_I2C_ADDR, 0);
+ timeout++;
+ if ((status & I2C_A_ADC_START) == 0)
+ break;
+
+ if (timeout > 1000) {
+ snd_printk("emu10k1:I2C:timeout status=0x%x\n", status);
+ break;
+ }
+ }
+ //Read back and see if the transaction is successful
+ if ((status & I2C_A_ADC_ABORT) == 0)
+ break;
+ }
+
+ if (retry == 10) {
+ snd_printk(KERN_ERR "Writing to ADC failed!\n");
+ snd_printk(KERN_ERR "status=0x%x, reg=%d, value=%d\n",
+ status, reg, value);
+ /* dump_stack(); */
+ err = -EINVAL;
+ }
+
+ spin_unlock(&emu->i2c_lock);
+ return err;
+}
+
+int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value)
+{
+ unsigned long flags;
+
+ if (reg > 0x3f)
+ return 1;
+ reg += 0x40; /* 0x40 upwards are registers. */
+ if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */
+ return 1;
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(reg, emu->port + A_IOCFG);
+ udelay(10);
+ outl(reg | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */
+ udelay(10);
+ outl(value, emu->port + A_IOCFG);
+ udelay(10);
+ outl(value | 0x80 , emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+
+ return 0;
+}
+
+int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value)
+{
+ unsigned long flags;
+ if (reg > 0x3f)
+ return 1;
+ reg += 0x40; /* 0x40 upwards are registers. */
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outl(reg, emu->port + A_IOCFG);
+ udelay(10);
+ outl(reg | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */
+ udelay(10);
+ *value = ((inl(emu->port + A_IOCFG) >> 8) & 0x7f);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+
+ return 0;
+}
+
+/* Each Destination has one and only one Source,
+ * but one Source can feed any number of Destinations simultaneously.
+ */
+int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src)
+{
+ snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f) );
+ snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f) );
+ snd_emu1010_fpga_write(emu, 0x02, ((src >> 8) & 0x3f) );
+ snd_emu1010_fpga_write(emu, 0x03, (src & 0x3f) );
+
+ return 0;
+}
+
+void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ enable = inl(emu->port + INTE) | intrenb;
+ outl(enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ enable = inl(emu->port + INTE) & ~intrenb;
+ outl(enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_intr_enable(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+ unsigned int val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(CLIEH << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val |= 1 << (voicenum - 32);
+ } else {
+ outl(CLIEL << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val |= 1 << voicenum;
+ }
+ outl(val, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_intr_disable(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+ unsigned int val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(CLIEH << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val &= ~(1 << (voicenum - 32));
+ } else {
+ outl(CLIEL << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val &= ~(1 << voicenum);
+ }
+ outl(val, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_intr_ack(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(CLIPH << 16, emu->port + PTR);
+ voicenum = 1 << (voicenum - 32);
+ } else {
+ outl(CLIPL << 16, emu->port + PTR);
+ voicenum = 1 << voicenum;
+ }
+ outl(voicenum, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_half_loop_intr_enable(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+ unsigned int val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(HLIEH << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val |= 1 << (voicenum - 32);
+ } else {
+ outl(HLIEL << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val |= 1 << voicenum;
+ }
+ outl(val, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_half_loop_intr_disable(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+ unsigned int val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(HLIEH << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val &= ~(1 << (voicenum - 32));
+ } else {
+ outl(HLIEL << 16, emu->port + PTR);
+ val = inl(emu->port + DATA);
+ val &= ~(1 << voicenum);
+ }
+ outl(val, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_half_loop_intr_ack(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(HLIPH << 16, emu->port + PTR);
+ voicenum = 1 << (voicenum - 32);
+ } else {
+ outl(HLIPL << 16, emu->port + PTR);
+ voicenum = 1 << voicenum;
+ }
+ outl(voicenum, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_set_loop_stop(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+ unsigned int sol;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(SOLEH << 16, emu->port + PTR);
+ sol = inl(emu->port + DATA);
+ sol |= 1 << (voicenum - 32);
+ } else {
+ outl(SOLEL << 16, emu->port + PTR);
+ sol = inl(emu->port + DATA);
+ sol |= 1 << voicenum;
+ }
+ outl(sol, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_voice_clear_loop_stop(struct snd_emu10k1 *emu, unsigned int voicenum)
+{
+ unsigned long flags;
+ unsigned int sol;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ /* voice interrupt */
+ if (voicenum >= 32) {
+ outl(SOLEH << 16, emu->port + PTR);
+ sol = inl(emu->port + DATA);
+ sol &= ~(1 << (voicenum - 32));
+ } else {
+ outl(SOLEL << 16, emu->port + PTR);
+ sol = inl(emu->port + DATA);
+ sol &= ~(1 << voicenum);
+ }
+ outl(sol, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+void snd_emu10k1_wait(struct snd_emu10k1 *emu, unsigned int wait)
+{
+ volatile unsigned count;
+ unsigned int newtime = 0, curtime;
+
+ curtime = inl(emu->port + WC) >> 6;
+ while (wait-- > 0) {
+ count = 0;
+ while (count++ < 16384) {
+ newtime = inl(emu->port + WC) >> 6;
+ if (newtime != curtime)
+ break;
+ }
+ if (count >= 16384)
+ break;
+ curtime = newtime;
+ }
+}
+
+unsigned short snd_emu10k1_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct snd_emu10k1 *emu = ac97->private_data;
+ unsigned long flags;
+ unsigned short val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ val = inw(emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+void snd_emu10k1_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short data)
+{
+ struct snd_emu10k1 *emu = ac97->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ outw(data, emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+/*
+ * convert rate to pitch
+ */
+
+unsigned int snd_emu10k1_rate_to_pitch(unsigned int rate)
+{
+ static u32 logMagTable[128] = {
+ 0x00000, 0x02dfc, 0x05b9e, 0x088e6, 0x0b5d6, 0x0e26f, 0x10eb3, 0x13aa2,
+ 0x1663f, 0x1918a, 0x1bc84, 0x1e72e, 0x2118b, 0x23b9a, 0x2655d, 0x28ed5,
+ 0x2b803, 0x2e0e8, 0x30985, 0x331db, 0x359eb, 0x381b6, 0x3a93d, 0x3d081,
+ 0x3f782, 0x41e42, 0x444c1, 0x46b01, 0x49101, 0x4b6c4, 0x4dc49, 0x50191,
+ 0x5269e, 0x54b6f, 0x57006, 0x59463, 0x5b888, 0x5dc74, 0x60029, 0x623a7,
+ 0x646ee, 0x66a00, 0x68cdd, 0x6af86, 0x6d1fa, 0x6f43c, 0x7164b, 0x73829,
+ 0x759d4, 0x77b4f, 0x79c9a, 0x7bdb5, 0x7dea1, 0x7ff5e, 0x81fed, 0x8404e,
+ 0x86082, 0x88089, 0x8a064, 0x8c014, 0x8df98, 0x8fef1, 0x91e20, 0x93d26,
+ 0x95c01, 0x97ab4, 0x9993e, 0x9b79f, 0x9d5d9, 0x9f3ec, 0xa11d8, 0xa2f9d,
+ 0xa4d3c, 0xa6ab5, 0xa8808, 0xaa537, 0xac241, 0xadf26, 0xafbe7, 0xb1885,
+ 0xb3500, 0xb5157, 0xb6d8c, 0xb899f, 0xba58f, 0xbc15e, 0xbdd0c, 0xbf899,
+ 0xc1404, 0xc2f50, 0xc4a7b, 0xc6587, 0xc8073, 0xc9b3f, 0xcb5ed, 0xcd07c,
+ 0xceaec, 0xd053f, 0xd1f73, 0xd398a, 0xd5384, 0xd6d60, 0xd8720, 0xda0c3,
+ 0xdba4a, 0xdd3b4, 0xded03, 0xe0636, 0xe1f4e, 0xe384a, 0xe512c, 0xe69f3,
+ 0xe829f, 0xe9b31, 0xeb3a9, 0xecc08, 0xee44c, 0xefc78, 0xf148a, 0xf2c83,
+ 0xf4463, 0xf5c2a, 0xf73da, 0xf8b71, 0xfa2f0, 0xfba57, 0xfd1a7, 0xfe8df
+ };
+ static char logSlopeTable[128] = {
+ 0x5c, 0x5c, 0x5b, 0x5a, 0x5a, 0x59, 0x58, 0x58,
+ 0x57, 0x56, 0x56, 0x55, 0x55, 0x54, 0x53, 0x53,
+ 0x52, 0x52, 0x51, 0x51, 0x50, 0x50, 0x4f, 0x4f,
+ 0x4e, 0x4d, 0x4d, 0x4d, 0x4c, 0x4c, 0x4b, 0x4b,
+ 0x4a, 0x4a, 0x49, 0x49, 0x48, 0x48, 0x47, 0x47,
+ 0x47, 0x46, 0x46, 0x45, 0x45, 0x45, 0x44, 0x44,
+ 0x43, 0x43, 0x43, 0x42, 0x42, 0x42, 0x41, 0x41,
+ 0x41, 0x40, 0x40, 0x40, 0x3f, 0x3f, 0x3f, 0x3e,
+ 0x3e, 0x3e, 0x3d, 0x3d, 0x3d, 0x3c, 0x3c, 0x3c,
+ 0x3b, 0x3b, 0x3b, 0x3b, 0x3a, 0x3a, 0x3a, 0x39,
+ 0x39, 0x39, 0x39, 0x38, 0x38, 0x38, 0x38, 0x37,
+ 0x37, 0x37, 0x37, 0x36, 0x36, 0x36, 0x36, 0x35,
+ 0x35, 0x35, 0x35, 0x34, 0x34, 0x34, 0x34, 0x34,
+ 0x33, 0x33, 0x33, 0x33, 0x32, 0x32, 0x32, 0x32,
+ 0x32, 0x31, 0x31, 0x31, 0x31, 0x31, 0x30, 0x30,
+ 0x30, 0x30, 0x30, 0x2f, 0x2f, 0x2f, 0x2f, 0x2f
+ };
+ int i;
+
+ if (rate == 0)
+ return 0; /* Bail out if no leading "1" */
+ rate *= 11185; /* Scale 48000 to 0x20002380 */
+ for (i = 31; i > 0; i--) {
+ if (rate & 0x80000000) { /* Detect leading "1" */
+ return (((unsigned int) (i - 15) << 20) +
+ logMagTable[0x7f & (rate >> 24)] +
+ (0x7f & (rate >> 17)) *
+ logSlopeTable[0x7f & (rate >> 24)]);
+ }
+ rate <<= 1;
+ }
+
+ return 0; /* Should never reach this point */
+}
+
diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c
new file mode 100644
index 0000000..30bfed6
--- /dev/null
+++ b/sound/pci/emu10k1/irq.c
@@ -0,0 +1,208 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Creative Labs, Inc.
+ * Routines for IRQ control of EMU10K1 chips
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+
+irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id)
+{
+ struct snd_emu10k1 *emu = dev_id;
+ unsigned int status, status2, orig_status, orig_status2;
+ int handled = 0;
+ int timeout = 0;
+
+ while (((status = inl(emu->port + IPR)) != 0) && (timeout < 1000)) {
+ timeout++;
+ orig_status = status;
+ handled = 1;
+ if ((status & 0xffffffff) == 0xffffffff) {
+ snd_printk(KERN_INFO "snd-emu10k1: Suspected sound card removal\n");
+ break;
+ }
+ if (status & IPR_PCIERROR) {
+ snd_printk(KERN_ERR "interrupt: PCI error\n");
+ snd_emu10k1_intr_disable(emu, INTE_PCIERRORENABLE);
+ status &= ~IPR_PCIERROR;
+ }
+ if (status & (IPR_VOLINCR|IPR_VOLDECR|IPR_MUTE)) {
+ if (emu->hwvol_interrupt)
+ emu->hwvol_interrupt(emu, status);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_VOLINCRENABLE|INTE_VOLDECRENABLE|INTE_MUTEENABLE);
+ status &= ~(IPR_VOLINCR|IPR_VOLDECR|IPR_MUTE);
+ }
+ if (status & IPR_CHANNELLOOP) {
+ int voice;
+ int voice_max = status & IPR_CHANNELNUMBERMASK;
+ u32 val;
+ struct snd_emu10k1_voice *pvoice = emu->voices;
+
+ val = snd_emu10k1_ptr_read(emu, CLIPL, 0);
+ for (voice = 0; voice <= voice_max; voice++) {
+ if (voice == 0x20)
+ val = snd_emu10k1_ptr_read(emu, CLIPH, 0);
+ if (val & 1) {
+ if (pvoice->use && pvoice->interrupt != NULL) {
+ pvoice->interrupt(emu, pvoice);
+ snd_emu10k1_voice_intr_ack(emu, voice);
+ } else {
+ snd_emu10k1_voice_intr_disable(emu, voice);
+ }
+ }
+ val >>= 1;
+ pvoice++;
+ }
+ val = snd_emu10k1_ptr_read(emu, HLIPL, 0);
+ for (voice = 0; voice <= voice_max; voice++) {
+ if (voice == 0x20)
+ val = snd_emu10k1_ptr_read(emu, HLIPH, 0);
+ if (val & 1) {
+ if (pvoice->use && pvoice->interrupt != NULL) {
+ pvoice->interrupt(emu, pvoice);
+ snd_emu10k1_voice_half_loop_intr_ack(emu, voice);
+ } else {
+ snd_emu10k1_voice_half_loop_intr_disable(emu, voice);
+ }
+ }
+ val >>= 1;
+ pvoice++;
+ }
+ status &= ~IPR_CHANNELLOOP;
+ }
+ status &= ~IPR_CHANNELNUMBERMASK;
+ if (status & (IPR_ADCBUFFULL|IPR_ADCBUFHALFFULL)) {
+ if (emu->capture_interrupt)
+ emu->capture_interrupt(emu, status);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_ADCBUFENABLE);
+ status &= ~(IPR_ADCBUFFULL|IPR_ADCBUFHALFFULL);
+ }
+ if (status & (IPR_MICBUFFULL|IPR_MICBUFHALFFULL)) {
+ if (emu->capture_mic_interrupt)
+ emu->capture_mic_interrupt(emu, status);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_MICBUFENABLE);
+ status &= ~(IPR_MICBUFFULL|IPR_MICBUFHALFFULL);
+ }
+ if (status & (IPR_EFXBUFFULL|IPR_EFXBUFHALFFULL)) {
+ if (emu->capture_efx_interrupt)
+ emu->capture_efx_interrupt(emu, status);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_EFXBUFENABLE);
+ status &= ~(IPR_EFXBUFFULL|IPR_EFXBUFHALFFULL);
+ }
+ if (status & (IPR_MIDITRANSBUFEMPTY|IPR_MIDIRECVBUFEMPTY)) {
+ if (emu->midi.interrupt)
+ emu->midi.interrupt(emu, status);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_MIDITXENABLE|INTE_MIDIRXENABLE);
+ status &= ~(IPR_MIDITRANSBUFEMPTY|IPR_MIDIRECVBUFEMPTY);
+ }
+ if (status & (IPR_A_MIDITRANSBUFEMPTY2|IPR_A_MIDIRECVBUFEMPTY2)) {
+ if (emu->midi2.interrupt)
+ emu->midi2.interrupt(emu, status);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_A_MIDITXENABLE2|INTE_A_MIDIRXENABLE2);
+ status &= ~(IPR_A_MIDITRANSBUFEMPTY2|IPR_A_MIDIRECVBUFEMPTY2);
+ }
+ if (status & IPR_INTERVALTIMER) {
+ if (emu->timer)
+ snd_timer_interrupt(emu->timer, emu->timer->sticks);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_INTERVALTIMERENB);
+ status &= ~IPR_INTERVALTIMER;
+ }
+ if (status & (IPR_GPSPDIFSTATUSCHANGE|IPR_CDROMSTATUSCHANGE)) {
+ if (emu->spdif_interrupt)
+ emu->spdif_interrupt(emu, status);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_GPSPDIFENABLE|INTE_CDSPDIFENABLE);
+ status &= ~(IPR_GPSPDIFSTATUSCHANGE|IPR_CDROMSTATUSCHANGE);
+ }
+ if (status & IPR_FXDSP) {
+ if (emu->dsp_interrupt)
+ emu->dsp_interrupt(emu);
+ else
+ snd_emu10k1_intr_disable(emu, INTE_FXDSPENABLE);
+ status &= ~IPR_FXDSP;
+ }
+ if (status & IPR_P16V) {
+ while ((status2 = inl(emu->port + IPR2)) != 0) {
+ u32 mask = INTE2_PLAYBACK_CH_0_LOOP; /* Full Loop */
+ struct snd_emu10k1_voice *pvoice = &(emu->p16v_voices[0]);
+ struct snd_emu10k1_voice *cvoice = &(emu->p16v_capture_voice);
+
+ //printk(KERN_INFO "status2=0x%x\n", status2);
+ orig_status2 = status2;
+ if(status2 & mask) {
+ if(pvoice->use) {
+ snd_pcm_period_elapsed(pvoice->epcm->substream);
+ } else {
+ snd_printk(KERN_ERR "p16v: status: 0x%08x, mask=0x%08x, pvoice=%p, use=%d\n", status2, mask, pvoice, pvoice->use);
+ }
+ }
+ if(status2 & 0x110000) {
+ //printk(KERN_INFO "capture int found\n");
+ if(cvoice->use) {
+ //printk(KERN_INFO "capture period_elapsed\n");
+ snd_pcm_period_elapsed(cvoice->epcm->substream);
+ }
+ }
+ outl(orig_status2, emu->port + IPR2); /* ack all */
+ }
+ status &= ~IPR_P16V;
+ }
+
+ if (status) {
+ unsigned int bits;
+ snd_printk(KERN_ERR "emu10k1: unhandled interrupt: 0x%08x\n", status);
+ //make sure any interrupts we don't handle are disabled:
+ bits = INTE_FXDSPENABLE |
+ INTE_PCIERRORENABLE |
+ INTE_VOLINCRENABLE |
+ INTE_VOLDECRENABLE |
+ INTE_MUTEENABLE |
+ INTE_MICBUFENABLE |
+ INTE_ADCBUFENABLE |
+ INTE_EFXBUFENABLE |
+ INTE_GPSPDIFENABLE |
+ INTE_CDSPDIFENABLE |
+ INTE_INTERVALTIMERENB |
+ INTE_MIDITXENABLE |
+ INTE_MIDIRXENABLE;
+ if (emu->audigy)
+ bits |= INTE_A_MIDITXENABLE2 | INTE_A_MIDIRXENABLE2;
+ snd_emu10k1_intr_disable(emu, bits);
+ }
+ outl(orig_status, emu->port + IPR); /* ack all */
+ }
+ if (timeout == 1000)
+ snd_printk(KERN_INFO "emu10k1 irq routine failure\n");
+
+ return IRQ_RETVAL(handled);
+}
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
new file mode 100644
index 0000000..6a47672
--- /dev/null
+++ b/sound/pci/emu10k1/memory.c
@@ -0,0 +1,569 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * EMU10K1 memory page allocation (PTB area)
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/pci.h>
+#include <linux/time.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+
+/* page arguments of these two macros are Emu page (4096 bytes), not like
+ * aligned pages in others
+ */
+#define __set_ptb_entry(emu,page,addr) \
+ (((u32 *)(emu)->ptb_pages.area)[page] = cpu_to_le32(((addr) << 1) | (page)))
+
+#define UNIT_PAGES (PAGE_SIZE / EMUPAGESIZE)
+#define MAX_ALIGN_PAGES (MAXPAGES / UNIT_PAGES)
+/* get aligned page from offset address */
+#define get_aligned_page(offset) ((offset) >> PAGE_SHIFT)
+/* get offset address from aligned page */
+#define aligned_page_offset(page) ((page) << PAGE_SHIFT)
+
+#if PAGE_SIZE == 4096
+/* page size == EMUPAGESIZE */
+/* fill PTB entrie(s) corresponding to page with addr */
+#define set_ptb_entry(emu,page,addr) __set_ptb_entry(emu,page,addr)
+/* fill PTB entrie(s) corresponding to page with silence pointer */
+#define set_silent_ptb(emu,page) __set_ptb_entry(emu,page,emu->silent_page.addr)
+#else
+/* fill PTB entries -- we need to fill UNIT_PAGES entries */
+static inline void set_ptb_entry(struct snd_emu10k1 *emu, int page, dma_addr_t addr)
+{
+ int i;
+ page *= UNIT_PAGES;
+ for (i = 0; i < UNIT_PAGES; i++, page++) {
+ __set_ptb_entry(emu, page, addr);
+ addr += EMUPAGESIZE;
+ }
+}
+static inline void set_silent_ptb(struct snd_emu10k1 *emu, int page)
+{
+ int i;
+ page *= UNIT_PAGES;
+ for (i = 0; i < UNIT_PAGES; i++, page++)
+ /* do not increment ptr */
+ __set_ptb_entry(emu, page, emu->silent_page.addr);
+}
+#endif /* PAGE_SIZE */
+
+
+/*
+ */
+static int synth_alloc_pages(struct snd_emu10k1 *hw, struct snd_emu10k1_memblk *blk);
+static int synth_free_pages(struct snd_emu10k1 *hw, struct snd_emu10k1_memblk *blk);
+
+#define get_emu10k1_memblk(l,member) list_entry(l, struct snd_emu10k1_memblk, member)
+
+
+/* initialize emu10k1 part */
+static void emu10k1_memblk_init(struct snd_emu10k1_memblk *blk)
+{
+ blk->mapped_page = -1;
+ INIT_LIST_HEAD(&blk->mapped_link);
+ INIT_LIST_HEAD(&blk->mapped_order_link);
+ blk->map_locked = 0;
+
+ blk->first_page = get_aligned_page(blk->mem.offset);
+ blk->last_page = get_aligned_page(blk->mem.offset + blk->mem.size - 1);
+ blk->pages = blk->last_page - blk->first_page + 1;
+}
+
+/*
+ * search empty region on PTB with the given size
+ *
+ * if an empty region is found, return the page and store the next mapped block
+ * in nextp
+ * if not found, return a negative error code.
+ */
+static int search_empty_map_area(struct snd_emu10k1 *emu, int npages, struct list_head **nextp)
+{
+ int page = 0, found_page = -ENOMEM;
+ int max_size = npages;
+ int size;
+ struct list_head *candidate = &emu->mapped_link_head;
+ struct list_head *pos;
+
+ list_for_each (pos, &emu->mapped_link_head) {
+ struct snd_emu10k1_memblk *blk = get_emu10k1_memblk(pos, mapped_link);
+ if (blk->mapped_page < 0)
+ continue;
+ size = blk->mapped_page - page;
+ if (size == npages) {
+ *nextp = pos;
+ return page;
+ }
+ else if (size > max_size) {
+ /* we look for the maximum empty hole */
+ max_size = size;
+ candidate = pos;
+ found_page = page;
+ }
+ page = blk->mapped_page + blk->pages;
+ }
+ size = MAX_ALIGN_PAGES - page;
+ if (size >= max_size) {
+ *nextp = pos;
+ return page;
+ }
+ *nextp = candidate;
+ return found_page;
+}
+
+/*
+ * map a memory block onto emu10k1's PTB
+ *
+ * call with memblk_lock held
+ */
+static int map_memblk(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
+{
+ int page, pg;
+ struct list_head *next;
+
+ page = search_empty_map_area(emu, blk->pages, &next);
+ if (page < 0) /* not found */
+ return page;
+ /* insert this block in the proper position of mapped list */
+ list_add_tail(&blk->mapped_link, next);
+ /* append this as a newest block in order list */
+ list_add_tail(&blk->mapped_order_link, &emu->mapped_order_link_head);
+ blk->mapped_page = page;
+ /* fill PTB */
+ for (pg = blk->first_page; pg <= blk->last_page; pg++) {
+ set_ptb_entry(emu, page, emu->page_addr_table[pg]);
+ page++;
+ }
+ return 0;
+}
+
+/*
+ * unmap the block
+ * return the size of resultant empty pages
+ *
+ * call with memblk_lock held
+ */
+static int unmap_memblk(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
+{
+ int start_page, end_page, mpage, pg;
+ struct list_head *p;
+ struct snd_emu10k1_memblk *q;
+
+ /* calculate the expected size of empty region */
+ if ((p = blk->mapped_link.prev) != &emu->mapped_link_head) {
+ q = get_emu10k1_memblk(p, mapped_link);
+ start_page = q->mapped_page + q->pages;
+ } else
+ start_page = 0;
+ if ((p = blk->mapped_link.next) != &emu->mapped_link_head) {
+ q = get_emu10k1_memblk(p, mapped_link);
+ end_page = q->mapped_page;
+ } else
+ end_page = MAX_ALIGN_PAGES;
+
+ /* remove links */
+ list_del(&blk->mapped_link);
+ list_del(&blk->mapped_order_link);
+ /* clear PTB */
+ mpage = blk->mapped_page;
+ for (pg = blk->first_page; pg <= blk->last_page; pg++) {
+ set_silent_ptb(emu, mpage);
+ mpage++;
+ }
+ blk->mapped_page = -1;
+ return end_page - start_page; /* return the new empty size */
+}
+
+/*
+ * search empty pages with the given size, and create a memory block
+ *
+ * unlike synth_alloc the memory block is aligned to the page start
+ */
+static struct snd_emu10k1_memblk *
+search_empty(struct snd_emu10k1 *emu, int size)
+{
+ struct list_head *p;
+ struct snd_emu10k1_memblk *blk;
+ int page, psize;
+
+ psize = get_aligned_page(size + PAGE_SIZE -1);
+ page = 0;
+ list_for_each(p, &emu->memhdr->block) {
+ blk = get_emu10k1_memblk(p, mem.list);
+ if (page + psize <= blk->first_page)
+ goto __found_pages;
+ page = blk->last_page + 1;
+ }
+ if (page + psize > emu->max_cache_pages)
+ return NULL;
+
+__found_pages:
+ /* create a new memory block */
+ blk = (struct snd_emu10k1_memblk *)__snd_util_memblk_new(emu->memhdr, psize << PAGE_SHIFT, p->prev);
+ if (blk == NULL)
+ return NULL;
+ blk->mem.offset = aligned_page_offset(page); /* set aligned offset */
+ emu10k1_memblk_init(blk);
+ return blk;
+}
+
+
+/*
+ * check if the given pointer is valid for pages
+ */
+static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr)
+{
+ if (addr & ~emu->dma_mask) {
+ snd_printk(KERN_ERR "max memory size is 0x%lx (addr = 0x%lx)!!\n", emu->dma_mask, (unsigned long)addr);
+ return 0;
+ }
+ if (addr & (EMUPAGESIZE-1)) {
+ snd_printk(KERN_ERR "page is not aligned\n");
+ return 0;
+ }
+ return 1;
+}
+
+/*
+ * map the given memory block on PTB.
+ * if the block is already mapped, update the link order.
+ * if no empty pages are found, tries to release unsed memory blocks
+ * and retry the mapping.
+ */
+int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
+{
+ int err;
+ int size;
+ struct list_head *p, *nextp;
+ struct snd_emu10k1_memblk *deleted;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->memblk_lock, flags);
+ if (blk->mapped_page >= 0) {
+ /* update order link */
+ list_del(&blk->mapped_order_link);
+ list_add_tail(&blk->mapped_order_link, &emu->mapped_order_link_head);
+ spin_unlock_irqrestore(&emu->memblk_lock, flags);
+ return 0;
+ }
+ if ((err = map_memblk(emu, blk)) < 0) {
+ /* no enough page - try to unmap some blocks */
+ /* starting from the oldest block */
+ p = emu->mapped_order_link_head.next;
+ for (; p != &emu->mapped_order_link_head; p = nextp) {
+ nextp = p->next;
+ deleted = get_emu10k1_memblk(p, mapped_order_link);
+ if (deleted->map_locked)
+ continue;
+ size = unmap_memblk(emu, deleted);
+ if (size >= blk->pages) {
+ /* ok the empty region is enough large */
+ err = map_memblk(emu, blk);
+ break;
+ }
+ }
+ }
+ spin_unlock_irqrestore(&emu->memblk_lock, flags);
+ return err;
+}
+
+EXPORT_SYMBOL(snd_emu10k1_memblk_map);
+
+/*
+ * page allocation for DMA
+ */
+struct snd_util_memblk *
+snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_util_memhdr *hdr;
+ struct snd_emu10k1_memblk *blk;
+ int page, err, idx;
+
+ if (snd_BUG_ON(!emu))
+ return NULL;
+ if (snd_BUG_ON(runtime->dma_bytes <= 0 ||
+ runtime->dma_bytes >= MAXPAGES * EMUPAGESIZE))
+ return NULL;
+ hdr = emu->memhdr;
+ if (snd_BUG_ON(!hdr))
+ return NULL;
+
+ mutex_lock(&hdr->block_mutex);
+ blk = search_empty(emu, runtime->dma_bytes);
+ if (blk == NULL) {
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+ /* fill buffer addresses but pointers are not stored so that
+ * snd_free_pci_page() is not called in in synth_free()
+ */
+ idx = 0;
+ for (page = blk->first_page; page <= blk->last_page; page++, idx++) {
+ unsigned long ofs = idx << PAGE_SHIFT;
+ dma_addr_t addr;
+ addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+ if (! is_valid_page(emu, addr)) {
+ printk(KERN_ERR "emu: failure page = %d\n", idx);
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+ emu->page_addr_table[page] = addr;
+ emu->page_ptr_table[page] = NULL;
+ }
+
+ /* set PTB entries */
+ blk->map_locked = 1; /* do not unmap this block! */
+ err = snd_emu10k1_memblk_map(emu, blk);
+ if (err < 0) {
+ __snd_util_mem_free(hdr, (struct snd_util_memblk *)blk);
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+ mutex_unlock(&hdr->block_mutex);
+ return (struct snd_util_memblk *)blk;
+}
+
+
+/*
+ * release DMA buffer from page table
+ */
+int snd_emu10k1_free_pages(struct snd_emu10k1 *emu, struct snd_util_memblk *blk)
+{
+ if (snd_BUG_ON(!emu || !blk))
+ return -EINVAL;
+ return snd_emu10k1_synth_free(emu, blk);
+}
+
+
+/*
+ * memory allocation using multiple pages (for synth)
+ * Unlike the DMA allocation above, non-contiguous pages are assined.
+ */
+
+/*
+ * allocate a synth sample area
+ */
+struct snd_util_memblk *
+snd_emu10k1_synth_alloc(struct snd_emu10k1 *hw, unsigned int size)
+{
+ struct snd_emu10k1_memblk *blk;
+ struct snd_util_memhdr *hdr = hw->memhdr;
+
+ mutex_lock(&hdr->block_mutex);
+ blk = (struct snd_emu10k1_memblk *)__snd_util_mem_alloc(hdr, size);
+ if (blk == NULL) {
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+ if (synth_alloc_pages(hw, blk)) {
+ __snd_util_mem_free(hdr, (struct snd_util_memblk *)blk);
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+ snd_emu10k1_memblk_map(hw, blk);
+ mutex_unlock(&hdr->block_mutex);
+ return (struct snd_util_memblk *)blk;
+}
+
+EXPORT_SYMBOL(snd_emu10k1_synth_alloc);
+
+/*
+ * free a synth sample area
+ */
+int
+snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk)
+{
+ struct snd_util_memhdr *hdr = emu->memhdr;
+ struct snd_emu10k1_memblk *blk = (struct snd_emu10k1_memblk *)memblk;
+ unsigned long flags;
+
+ mutex_lock(&hdr->block_mutex);
+ spin_lock_irqsave(&emu->memblk_lock, flags);
+ if (blk->mapped_page >= 0)
+ unmap_memblk(emu, blk);
+ spin_unlock_irqrestore(&emu->memblk_lock, flags);
+ synth_free_pages(emu, blk);
+ __snd_util_mem_free(hdr, memblk);
+ mutex_unlock(&hdr->block_mutex);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_emu10k1_synth_free);
+
+/* check new allocation range */
+static void get_single_page_range(struct snd_util_memhdr *hdr,
+ struct snd_emu10k1_memblk *blk,
+ int *first_page_ret, int *last_page_ret)
+{
+ struct list_head *p;
+ struct snd_emu10k1_memblk *q;
+ int first_page, last_page;
+ first_page = blk->first_page;
+ if ((p = blk->mem.list.prev) != &hdr->block) {
+ q = get_emu10k1_memblk(p, mem.list);
+ if (q->last_page == first_page)
+ first_page++; /* first page was already allocated */
+ }
+ last_page = blk->last_page;
+ if ((p = blk->mem.list.next) != &hdr->block) {
+ q = get_emu10k1_memblk(p, mem.list);
+ if (q->first_page == last_page)
+ last_page--; /* last page was already allocated */
+ }
+ *first_page_ret = first_page;
+ *last_page_ret = last_page;
+}
+
+/* release allocated pages */
+static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page,
+ int last_page)
+{
+ int page;
+
+ for (page = first_page; page <= last_page; page++) {
+ free_page((unsigned long)emu->page_ptr_table[page]);
+ emu->page_addr_table[page] = 0;
+ emu->page_ptr_table[page] = NULL;
+ }
+}
+
+/*
+ * allocate kernel pages
+ */
+static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
+{
+ int page, first_page, last_page;
+
+ emu10k1_memblk_init(blk);
+ get_single_page_range(emu->memhdr, blk, &first_page, &last_page);
+ /* allocate kernel pages */
+ for (page = first_page; page <= last_page; page++) {
+ /* first try to allocate from <4GB zone */
+ struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 |
+ __GFP_NOWARN);
+ if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) {
+ if (p)
+ __free_page(p);
+ /* try to allocate from <16MB zone */
+ p = alloc_page(GFP_ATOMIC | GFP_DMA |
+ __GFP_NORETRY | /* no OOM-killer */
+ __GFP_NOWARN);
+ }
+ if (!p) {
+ __synth_free_pages(emu, first_page, page - 1);
+ return -ENOMEM;
+ }
+ emu->page_addr_table[page] = page_to_phys(p);
+ emu->page_ptr_table[page] = page_address(p);
+ }
+ return 0;
+}
+
+/*
+ * free pages
+ */
+static int synth_free_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk)
+{
+ int first_page, last_page;
+
+ get_single_page_range(emu->memhdr, blk, &first_page, &last_page);
+ __synth_free_pages(emu, first_page, last_page);
+ return 0;
+}
+
+/* calculate buffer pointer from offset address */
+static inline void *offset_ptr(struct snd_emu10k1 *emu, int page, int offset)
+{
+ char *ptr;
+ if (snd_BUG_ON(page < 0 || page >= emu->max_cache_pages))
+ return NULL;
+ ptr = emu->page_ptr_table[page];
+ if (! ptr) {
+ printk(KERN_ERR "emu10k1: access to NULL ptr: page = %d\n", page);
+ return NULL;
+ }
+ ptr += offset & (PAGE_SIZE - 1);
+ return (void*)ptr;
+}
+
+/*
+ * bzero(blk + offset, size)
+ */
+int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk,
+ int offset, int size)
+{
+ int page, nextofs, end_offset, temp, temp1;
+ void *ptr;
+ struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk;
+
+ offset += blk->offset & (PAGE_SIZE - 1);
+ end_offset = offset + size;
+ page = get_aligned_page(offset);
+ do {
+ nextofs = aligned_page_offset(page + 1);
+ temp = nextofs - offset;
+ temp1 = end_offset - offset;
+ if (temp1 < temp)
+ temp = temp1;
+ ptr = offset_ptr(emu, page + p->first_page, offset);
+ if (ptr)
+ memset(ptr, 0, temp);
+ offset = nextofs;
+ page++;
+ } while (offset < end_offset);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_emu10k1_synth_bzero);
+
+/*
+ * copy_from_user(blk + offset, data, size)
+ */
+int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk,
+ int offset, const char __user *data, int size)
+{
+ int page, nextofs, end_offset, temp, temp1;
+ void *ptr;
+ struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk;
+
+ offset += blk->offset & (PAGE_SIZE - 1);
+ end_offset = offset + size;
+ page = get_aligned_page(offset);
+ do {
+ nextofs = aligned_page_offset(page + 1);
+ temp = nextofs - offset;
+ temp1 = end_offset - offset;
+ if (temp1 < temp)
+ temp = temp1;
+ ptr = offset_ptr(emu, page + p->first_page, offset);
+ if (ptr && copy_from_user(ptr, data, temp))
+ return -EFAULT;
+ offset = nextofs;
+ data += temp;
+ page++;
+ } while (offset < end_offset);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_emu10k1_synth_copy_from_user);
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
new file mode 100644
index 0000000..749a21b
--- /dev/null
+++ b/sound/pci/emu10k1/p16v.c
@@ -0,0 +1,888 @@
+/*
+ * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver p16v chips
+ * Version: 0.25
+ *
+ * FEATURES currently supported:
+ * Output fixed at S32_LE, 2 channel to hw:0,0
+ * Rates: 44.1, 48, 96, 192.
+ *
+ * Changelog:
+ * 0.8
+ * Use separate card based buffer for periods table.
+ * 0.9
+ * Use 2 channel output streams instead of 8 channel.
+ * (8 channel output streams might be good for ASIO type output)
+ * Corrected speaker output, so Front -> Front etc.
+ * 0.10
+ * Fixed missed interrupts.
+ * 0.11
+ * Add Sound card model number and names.
+ * Add Analog volume controls.
+ * 0.12
+ * Corrected playback interrupts. Now interrupt per period, instead of half period.
+ * 0.13
+ * Use single trigger for multichannel.
+ * 0.14
+ * Mic capture now works at fixed: S32_LE, 96000Hz, Stereo.
+ * 0.15
+ * Force buffer_size / period_size == INTEGER.
+ * 0.16
+ * Update p16v.c to work with changed alsa api.
+ * 0.17
+ * Update p16v.c to work with changed alsa api. Removed boot_devs.
+ * 0.18
+ * Merging with snd-emu10k1 driver.
+ * 0.19
+ * One stereo channel at 24bit now works.
+ * 0.20
+ * Added better register defines.
+ * 0.21
+ * Integrated with snd-emu10k1 driver.
+ * 0.22
+ * Removed #if 0 ... #endif
+ * 0.23
+ * Implement different capture rates.
+ * 0.24
+ * Implement different capture source channels.
+ * e.g. When HD Capture source is set to SPDIF,
+ * setting HD Capture channel to 0 captures from CDROM digital input.
+ * setting HD Capture channel to 1 captures from SPDIF in.
+ * 0.25
+ * Include capture buffer sizes.
+ *
+ * BUGS:
+ * Some stability problems when unloading the snd-p16v kernel module.
+ * --
+ *
+ * TODO:
+ * SPDIF out.
+ * Find out how to change capture sample rates. E.g. To record SPDIF at 48000Hz.
+ * Currently capture fixed at 48000Hz.
+ *
+ * --
+ * GENERAL INFO:
+ * Model: SB0240
+ * P16V Chip: CA0151-DBS
+ * Audigy 2 Chip: CA0102-IAT
+ * AC97 Codec: STAC 9721
+ * ADC: Philips 1361T (Stereo 24bit)
+ * DAC: CS4382-K (8-channel, 24bit, 192Khz)
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+#include <sound/emu10k1.h>
+#include "p16v.h"
+
+#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
+#define PCM_FRONT_CHANNEL 0
+#define PCM_REAR_CHANNEL 1
+#define PCM_CENTER_LFE_CHANNEL 2
+#define PCM_SIDE_CHANNEL 3
+#define CONTROL_FRONT_CHANNEL 0
+#define CONTROL_REAR_CHANNEL 3
+#define CONTROL_CENTER_LFE_CHANNEL 1
+#define CONTROL_SIDE_CHANNEL 2
+
+/* Card IDs:
+ * Class 0401: 1102:0004 (rev 04) Subsystem: 1102:2002 -> Audigy2 ZS 7.1 Model:SB0350
+ * Class 0401: 1102:0004 (rev 04) Subsystem: 1102:1007 -> Audigy2 6.1 Model:SB0240
+ * Class 0401: 1102:0004 (rev 04) Subsystem: 1102:1002 -> Audigy2 Platinum Model:SB msb0240230009266
+ * Class 0401: 1102:0004 (rev 04) Subsystem: 1102:2007 -> Audigy4 Pro Model:SB0380 M1SB0380472001901E
+ *
+ */
+
+ /* hardware definition */
+static struct snd_pcm_hardware snd_p16v_playback_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only supports 24-bit samples padded to 32 bits. */
+ .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 192000,
+ .channels_min = 8,
+ .channels_max = 8,
+ .buffer_bytes_max = ((65536 - 64) * 8),
+ .period_bytes_min = 64,
+ .period_bytes_max = (65536 - 64),
+ .periods_min = 2,
+ .periods_max = 8,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_p16v_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (65536 - 64),
+ .period_bytes_min = 64,
+ .period_bytes_max = (65536 - 128) >> 1, /* size has to be N*64 bytes */
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+
+ if (epcm) {
+ //snd_printk("epcm free: %p\n", epcm);
+ kfree(epcm);
+ }
+}
+
+/* open_playback callback */
+static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substream, int channel_id)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_voice *channel = &(emu->p16v_voices[channel_id]);
+ struct snd_emu10k1_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ //snd_printk("epcm kcalloc: %p\n", epcm);
+
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->substream = substream;
+ //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id);
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_p16v_pcm_free_substream;
+
+ runtime->hw = snd_p16v_playback_hw;
+
+ channel->emu = emu;
+ channel->number = channel_id;
+
+ channel->use=1;
+ //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use);
+ //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+ //channel->interrupt = snd_p16v_pcm_channel_interrupt;
+ channel->epcm=epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ runtime->sync.id32[0] = substream->pcm->card->number;
+ runtime->sync.id32[1] = 'P';
+ runtime->sync.id32[2] = 16;
+ runtime->sync.id32[3] = 'V';
+
+ return 0;
+}
+/* open_capture callback */
+static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream, int channel_id)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_emu10k1_voice *channel = &(emu->p16v_capture_voice);
+ struct snd_emu10k1_pcm *epcm;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+ //snd_printk("epcm kcalloc: %p\n", epcm);
+
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->substream = substream;
+ //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id);
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_p16v_pcm_free_substream;
+
+ runtime->hw = snd_p16v_capture_hw;
+
+ channel->emu = emu;
+ channel->number = channel_id;
+
+ channel->use=1;
+ //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use);
+ //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+ //channel->interrupt = snd_p16v_pcm_channel_interrupt;
+ channel->epcm=epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/* close callback */
+static int snd_p16v_pcm_close_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ //struct snd_pcm_runtime *runtime = substream->runtime;
+ //struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ emu->p16v_voices[substream->pcm->device - emu->p16v_device_offset].use = 0;
+ /* FIXME: maybe zero others */
+ return 0;
+}
+
+/* close callback */
+static int snd_p16v_pcm_close_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ //struct snd_pcm_runtime *runtime = substream->runtime;
+ //struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ emu->p16v_capture_voice.use = 0;
+ /* FIXME: maybe zero others */
+ return 0;
+}
+
+static int snd_p16v_pcm_open_playback_front(struct snd_pcm_substream *substream)
+{
+ return snd_p16v_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
+}
+
+static int snd_p16v_pcm_open_capture(struct snd_pcm_substream *substream)
+{
+ // Only using channel 0 for now, but the card has 2 channels.
+ return snd_p16v_pcm_open_capture_channel(substream, 0);
+}
+
+/* hw_params callback */
+static int snd_p16v_pcm_hw_params_playback(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ int result;
+ result = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ return result;
+}
+
+/* hw_params callback */
+static int snd_p16v_pcm_hw_params_capture(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ int result;
+ result = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ return result;
+}
+
+
+/* hw_free callback */
+static int snd_p16v_pcm_hw_free_playback(struct snd_pcm_substream *substream)
+{
+ int result;
+ result = snd_pcm_lib_free_pages(substream);
+ return result;
+}
+
+/* hw_free callback */
+static int snd_p16v_pcm_hw_free_capture(struct snd_pcm_substream *substream)
+{
+ int result;
+ result = snd_pcm_lib_free_pages(substream);
+ return result;
+}
+
+
+/* prepare playback callback */
+static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int channel = substream->pcm->device - emu->p16v_device_offset;
+ u32 *table_base = (u32 *)(emu->p16v_buffer.area+(8*16*channel));
+ u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
+ int i;
+ u32 tmp;
+
+ //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
+ //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
+ //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes);
+ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
+ switch (runtime->rate) {
+ case 44100:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x8080);
+ break;
+ case 96000:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x4040);
+ break;
+ case 192000:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x2020);
+ break;
+ case 48000:
+ default:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x0000);
+ break;
+ }
+ /* FIXME: Check emu->buffer.size before actually writing to it. */
+ for(i = 0; i < runtime->periods; i++) {
+ table_base[i*2]=runtime->dma_addr+(i*period_size_bytes);
+ table_base[(i*2)+1]=period_size_bytes<<16;
+ }
+
+ snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_ADDR, channel, emu->p16v_buffer.addr+(8*16*channel));
+ snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19);
+ snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_PTR, channel, 0);
+ snd_emu10k1_ptr20_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr);
+ //snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
+ snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes
+ snd_emu10k1_ptr20_write(emu, PLAYBACK_POINTER, channel, 0);
+ snd_emu10k1_ptr20_write(emu, 0x07, channel, 0x0);
+ snd_emu10k1_ptr20_write(emu, 0x08, channel, 0);
+
+ return 0;
+}
+
+/* prepare capture callback */
+static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int channel = substream->pcm->device - emu->p16v_device_offset;
+ u32 tmp;
+ //printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1));
+ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
+ switch (runtime->rate) {
+ case 44100:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0800);
+ break;
+ case 96000:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0400);
+ break;
+ case 192000:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0200);
+ break;
+ case 48000:
+ default:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0000);
+ break;
+ }
+ /* FIXME: Check emu->buffer.size before actually writing to it. */
+ snd_emu10k1_ptr20_write(emu, 0x13, channel, 0);
+ snd_emu10k1_ptr20_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
+ snd_emu10k1_ptr20_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size) << 16); // buffer size in bytes
+ snd_emu10k1_ptr20_write(emu, CAPTURE_POINTER, channel, 0);
+ //snd_emu10k1_ptr20_write(emu, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC or Line in */
+ //snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<channel));
+
+ return 0;
+}
+
+static void snd_p16v_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ enable = inl(emu->port + INTE2) | intrenb;
+ outl(enable, emu->port + INTE2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_p16v_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int disable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ disable = inl(emu->port + INTE2) & (~intrenb);
+ outl(disable, emu->port + INTE2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+/* trigger_playback callback */
+static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime;
+ struct snd_emu10k1_pcm *epcm;
+ int channel;
+ int result = 0;
+ struct snd_pcm_substream *s;
+ u32 basic = 0;
+ u32 inte = 0;
+ int running = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ running=1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ default:
+ running = 0;
+ break;
+ }
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) != emu ||
+ s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ continue;
+ runtime = s->runtime;
+ epcm = runtime->private_data;
+ channel = substream->pcm->device-emu->p16v_device_offset;
+ //snd_printk("p16v channel=%d\n",channel);
+ epcm->running = running;
+ basic |= (0x1<<channel);
+ inte |= (INTE2_PLAYBACK_CH_0_LOOP<<channel);
+ snd_pcm_trigger_done(s, substream);
+ }
+ //snd_printk("basic=0x%x, inte=0x%x\n",basic, inte);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_p16v_intr_enable(emu, inte);
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0)| (basic));
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0) & ~(basic));
+ snd_p16v_intr_disable(emu, inte);
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* trigger_capture callback */
+static int snd_p16v_pcm_trigger_capture(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ int channel = 0;
+ int result = 0;
+ u32 inte = INTE2_CAPTURE_CH_0_LOOP | INTE2_CAPTURE_CH_0_HALF_LOOP;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_p16v_intr_enable(emu, inte);
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel));
+ epcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel));
+ snd_p16v_intr_disable(emu, inte);
+ //snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel));
+ epcm->running = 0;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* pointer_playback callback */
+static snd_pcm_uframes_t
+snd_p16v_pcm_pointer_playback(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0;
+ int channel = substream->pcm->device - emu->p16v_device_offset;
+ if (!epcm->running)
+ return 0;
+
+ ptr3 = snd_emu10k1_ptr20_read(emu, PLAYBACK_LIST_PTR, channel);
+ ptr1 = snd_emu10k1_ptr20_read(emu, PLAYBACK_POINTER, channel);
+ ptr4 = snd_emu10k1_ptr20_read(emu, PLAYBACK_LIST_PTR, channel);
+ if (ptr3 != ptr4) ptr1 = snd_emu10k1_ptr20_read(emu, PLAYBACK_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr2+= (ptr4 >> 3) * runtime->period_size;
+ ptr=ptr2;
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+
+ return ptr;
+}
+
+/* pointer_capture callback */
+static snd_pcm_uframes_t
+snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream)
+{
+ struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_emu10k1_pcm *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
+ int channel = 0;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr1 = snd_emu10k1_ptr20_read(emu, CAPTURE_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr=ptr2;
+ if (ptr >= runtime->buffer_size) {
+ ptr -= runtime->buffer_size;
+ printk(KERN_WARNING "buffer capture limited!\n");
+ }
+ //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
+
+ return ptr;
+}
+
+/* operators */
+static struct snd_pcm_ops snd_p16v_playback_front_ops = {
+ .open = snd_p16v_pcm_open_playback_front,
+ .close = snd_p16v_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_p16v_pcm_hw_params_playback,
+ .hw_free = snd_p16v_pcm_hw_free_playback,
+ .prepare = snd_p16v_pcm_prepare_playback,
+ .trigger = snd_p16v_pcm_trigger_playback,
+ .pointer = snd_p16v_pcm_pointer_playback,
+};
+
+static struct snd_pcm_ops snd_p16v_capture_ops = {
+ .open = snd_p16v_pcm_open_capture,
+ .close = snd_p16v_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_p16v_pcm_hw_params_capture,
+ .hw_free = snd_p16v_pcm_hw_free_capture,
+ .prepare = snd_p16v_pcm_prepare_capture,
+ .trigger = snd_p16v_pcm_trigger_capture,
+ .pointer = snd_p16v_pcm_pointer_capture,
+};
+
+
+int snd_p16v_free(struct snd_emu10k1 *chip)
+{
+ // release the data
+ if (chip->p16v_buffer.area) {
+ snd_dma_free_pages(&chip->p16v_buffer);
+ //snd_printk("period lables free: %p\n", &chip->p16v_buffer);
+ }
+ return 0;
+}
+
+int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ int err;
+ int capture=1;
+
+ //snd_printk("snd_p16v_pcm called. device=%d\n", device);
+ emu->p16v_device_offset = device;
+ if (rpcm)
+ *rpcm = NULL;
+
+ if ((err = snd_pcm_new(emu->card, "p16v", device, 1, capture, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+ // Single playback 8 channel device.
+ // Single capture 2 channel device.
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_p16v_playback_front_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_p16v_capture_ops);
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strcpy(pcm->name, "p16v");
+ emu->pcm_p16v = pcm;
+
+ for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0)
+ return err;
+ //snd_printk("preallocate playback substream: err=%d\n", err);
+ }
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 65536 - 64, 65536 - 64)) < 0)
+ return err;
+ //snd_printk("preallocate capture substream: err=%d\n", err);
+ }
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+static int snd_p16v_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_p16v_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int high_low = (kcontrol->private_value >> 8) & 0xff;
+ int reg = kcontrol->private_value & 0xff;
+ u32 value;
+
+ value = snd_emu10k1_ptr20_read(emu, reg, high_low);
+ if (high_low) {
+ ucontrol->value.integer.value[0] = 0xff - ((value >> 24) & 0xff); /* Left */
+ ucontrol->value.integer.value[1] = 0xff - ((value >> 16) & 0xff); /* Right */
+ } else {
+ ucontrol->value.integer.value[0] = 0xff - ((value >> 8) & 0xff); /* Left */
+ ucontrol->value.integer.value[1] = 0xff - ((value >> 0) & 0xff); /* Right */
+ }
+ return 0;
+}
+
+static int snd_p16v_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ int high_low = (kcontrol->private_value >> 8) & 0xff;
+ int reg = kcontrol->private_value & 0xff;
+ u32 value, oval;
+
+ oval = value = snd_emu10k1_ptr20_read(emu, reg, 0);
+ if (high_low == 1) {
+ value &= 0xffff;
+ value |= ((0xff - ucontrol->value.integer.value[0]) << 24) |
+ ((0xff - ucontrol->value.integer.value[1]) << 16);
+ } else {
+ value &= 0xffff0000;
+ value |= ((0xff - ucontrol->value.integer.value[0]) << 8) |
+ ((0xff - ucontrol->value.integer.value[1]) );
+ }
+ if (value != oval) {
+ snd_emu10k1_ptr20_write(emu, reg, 0, value);
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_p16v_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[8] = {
+ "SPDIF", "I2S", "SRC48", "SRCMulti_SPDIF", "SRCMulti_I2S",
+ "CDIF", "FX", "AC97"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item > 7)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_p16v_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->p16v_capture_source;
+ return 0;
+}
+
+static int snd_p16v_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 mask;
+ u32 source;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ if (val > 7)
+ return -EINVAL;
+ change = (emu->p16v_capture_source != val);
+ if (change) {
+ emu->p16v_capture_source = val;
+ source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
+ mask = snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0) & 0xffff;
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, source | mask);
+ }
+ return change;
+}
+
+static int snd_p16v_capture_channel_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = { "0", "1", "2", "3", };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_p16v_capture_channel_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->p16v_capture_channel;
+ return 0;
+}
+
+static int snd_p16v_capture_channel_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 tmp;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ if (val > 3)
+ return -EINVAL;
+ change = (emu->p16v_capture_channel != val);
+ if (change) {
+ emu->p16v_capture_channel = val;
+ tmp = snd_emu10k1_ptr20_read(emu, CAPTURE_P16V_SOURCE, 0) & 0xfffc;
+ snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, tmp | val);
+ }
+ return change;
+}
+static const DECLARE_TLV_DB_SCALE(snd_p16v_db_scale1, -5175, 25, 1);
+
+#define P16V_VOL(xname,xreg,xhl) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = snd_p16v_volume_info, \
+ .get = snd_p16v_volume_get, \
+ .put = snd_p16v_volume_put, \
+ .tlv = { .p = snd_p16v_db_scale1 }, \
+ .private_value = ((xreg) | ((xhl) << 8)) \
+}
+
+static struct snd_kcontrol_new p16v_mixer_controls[] __devinitdata = {
+ P16V_VOL("HD Analog Front Playback Volume", PLAYBACK_VOLUME_MIXER9, 0),
+ P16V_VOL("HD Analog Rear Playback Volume", PLAYBACK_VOLUME_MIXER10, 1),
+ P16V_VOL("HD Analog Center/LFE Playback Volume", PLAYBACK_VOLUME_MIXER9, 1),
+ P16V_VOL("HD Analog Side Playback Volume", PLAYBACK_VOLUME_MIXER10, 0),
+ P16V_VOL("HD SPDIF Front Playback Volume", PLAYBACK_VOLUME_MIXER7, 0),
+ P16V_VOL("HD SPDIF Rear Playback Volume", PLAYBACK_VOLUME_MIXER8, 1),
+ P16V_VOL("HD SPDIF Center/LFE Playback Volume", PLAYBACK_VOLUME_MIXER7, 1),
+ P16V_VOL("HD SPDIF Side Playback Volume", PLAYBACK_VOLUME_MIXER8, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "HD source Capture",
+ .info = snd_p16v_capture_source_info,
+ .get = snd_p16v_capture_source_get,
+ .put = snd_p16v_capture_source_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "HD channel Capture",
+ .info = snd_p16v_capture_channel_info,
+ .get = snd_p16v_capture_channel_get,
+ .put = snd_p16v_capture_channel_put
+ },
+};
+
+
+int __devinit snd_p16v_mixer(struct snd_emu10k1 *emu)
+{
+ int i, err;
+ struct snd_card *card = emu->card;
+
+ for (i = 0; i < ARRAY_SIZE(p16v_mixer_controls); i++) {
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&p16v_mixer_controls[i],
+ emu))) < 0)
+ return err;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_PM
+
+#define NUM_CHS 1 /* up to 4, but only first channel is used */
+
+int __devinit snd_p16v_alloc_pm_buffer(struct snd_emu10k1 *emu)
+{
+ emu->p16v_saved = vmalloc(NUM_CHS * 4 * 0x80);
+ if (! emu->p16v_saved)
+ return -ENOMEM;
+ return 0;
+}
+
+void snd_p16v_free_pm_buffer(struct snd_emu10k1 *emu)
+{
+ vfree(emu->p16v_saved);
+}
+
+void snd_p16v_suspend(struct snd_emu10k1 *emu)
+{
+ int i, ch;
+ unsigned int *val;
+
+ val = emu->p16v_saved;
+ for (ch = 0; ch < NUM_CHS; ch++)
+ for (i = 0; i < 0x80; i++, val++)
+ *val = snd_emu10k1_ptr20_read(emu, i, ch);
+}
+
+void snd_p16v_resume(struct snd_emu10k1 *emu)
+{
+ int i, ch;
+ unsigned int *val;
+
+ val = emu->p16v_saved;
+ for (ch = 0; ch < NUM_CHS; ch++)
+ for (i = 0; i < 0x80; i++, val++)
+ snd_emu10k1_ptr20_write(emu, i, ch, *val);
+}
+#endif
diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h
new file mode 100644
index 0000000..1532149
--- /dev/null
+++ b/sound/pci/emu10k1/p16v.h
@@ -0,0 +1,299 @@
+/*
+ * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver p16v chips
+ * Version: 0.21
+ *
+ * FEATURES currently supported:
+ * Output fixed at S32_LE, 2 channel to hw:0,0
+ * Rates: 44.1, 48, 96, 192.
+ *
+ * Changelog:
+ * 0.8
+ * Use separate card based buffer for periods table.
+ * 0.9
+ * Use 2 channel output streams instead of 8 channel.
+ * (8 channel output streams might be good for ASIO type output)
+ * Corrected speaker output, so Front -> Front etc.
+ * 0.10
+ * Fixed missed interrupts.
+ * 0.11
+ * Add Sound card model number and names.
+ * Add Analog volume controls.
+ * 0.12
+ * Corrected playback interrupts. Now interrupt per period, instead of half period.
+ * 0.13
+ * Use single trigger for multichannel.
+ * 0.14
+ * Mic capture now works at fixed: S32_LE, 96000Hz, Stereo.
+ * 0.15
+ * Force buffer_size / period_size == INTEGER.
+ * 0.16
+ * Update p16v.c to work with changed alsa api.
+ * 0.17
+ * Update p16v.c to work with changed alsa api. Removed boot_devs.
+ * 0.18
+ * Merging with snd-emu10k1 driver.
+ * 0.19
+ * One stereo channel at 24bit now works.
+ * 0.20
+ * Added better register defines.
+ * 0.21
+ * Split from p16v.c
+ *
+ *
+ * BUGS:
+ * Some stability problems when unloading the snd-p16v kernel module.
+ * --
+ *
+ * TODO:
+ * SPDIF out.
+ * Find out how to change capture sample rates. E.g. To record SPDIF at 48000Hz.
+ * Currently capture fixed at 48000Hz.
+ *
+ * --
+ * GENERAL INFO:
+ * Model: SB0240
+ * P16V Chip: CA0151-DBS
+ * Audigy 2 Chip: CA0102-IAT
+ * AC97 Codec: STAC 9721
+ * ADC: Philips 1361T (Stereo 24bit)
+ * DAC: CS4382-K (8-channel, 24bit, 192Khz)
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/********************************************************************************************************/
+/* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers */
+/********************************************************************************************************/
+
+/* The sample rate of the SPDIF outputs is set by modifying a register in the EMU10K2 PTR register A_SPDIF_SAMPLERATE.
+ * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters.
+ */
+
+/* Initally all registers from 0x00 to 0x3f have zero contents. */
+#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
+ /* One list entry: 4 bytes for DMA address,
+ * 4 bytes for period_size << 16.
+ * One list entry is 8 bytes long.
+ * One list entry for each period in the buffer.
+ */
+#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
+#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
+#define PLAYBACK_UNKNOWN3 0x03 /* Not used */
+#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
+#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
+#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
+#define PLAYBACK_FIFO_END_ADDRESS 0x07 /* Playback FIFO end address */
+#define PLAYBACK_FIFO_POINTER 0x08 /* Playback FIFO pointer and number of valid sound samples in cache */
+#define PLAYBACK_UNKNOWN9 0x09 /* Not used */
+#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
+#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
+#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
+#define CAPTURE_FIFO_POINTER 0x13 /* Capture FIFO pointer and number of valid sound samples in cache */
+#define CAPTURE_P16V_VOLUME1 0x14 /* Low: Capture volume 0xXXXX3030 */
+#define CAPTURE_P16V_VOLUME2 0x15 /* High:Has no effect on capture volume */
+#define CAPTURE_P16V_SOURCE 0x16 /* P16V source select. Set to 0x0700E4E5 for AC97 CAPTURE */
+ /* [0:1] Capture input 0 channel select. 0 = Capture output 0.
+ * 1 = Capture output 1.
+ * 2 = Capture output 2.
+ * 3 = Capture output 3.
+ * [3:2] Capture input 1 channel select. 0 = Capture output 0.
+ * 1 = Capture output 1.
+ * 2 = Capture output 2.
+ * 3 = Capture output 3.
+ * [5:4] Capture input 2 channel select. 0 = Capture output 0.
+ * 1 = Capture output 1.
+ * 2 = Capture output 2.
+ * 3 = Capture output 3.
+ * [7:6] Capture input 3 channel select. 0 = Capture output 0.
+ * 1 = Capture output 1.
+ * 2 = Capture output 2.
+ * 3 = Capture output 3.
+ * [9:8] Playback input 0 channel select. 0 = Play output 0.
+ * 1 = Play output 1.
+ * 2 = Play output 2.
+ * 3 = Play output 3.
+ * [11:10] Playback input 1 channel select. 0 = Play output 0.
+ * 1 = Play output 1.
+ * 2 = Play output 2.
+ * 3 = Play output 3.
+ * [13:12] Playback input 2 channel select. 0 = Play output 0.
+ * 1 = Play output 1.
+ * 2 = Play output 2.
+ * 3 = Play output 3.
+ * [15:14] Playback input 3 channel select. 0 = Play output 0.
+ * 1 = Play output 1.
+ * 2 = Play output 2.
+ * 3 = Play output 3.
+ * [19:16] Playback mixer output enable. 1 bit per channel.
+ * [23:20] Capture mixer output enable. 1 bit per channel.
+ * [26:24] FX engine channel capture 0 = 0x60-0x67.
+ * 1 = 0x68-0x6f.
+ * 2 = 0x70-0x77.
+ * 3 = 0x78-0x7f.
+ * 4 = 0x80-0x87.
+ * 5 = 0x88-0x8f.
+ * 6 = 0x90-0x97.
+ * 7 = 0x98-0x9f.
+ * [31:27] Not used.
+ */
+
+ /* 0x1 = capture on.
+ * 0x100 = capture off.
+ * 0x200 = capture off.
+ * 0x1000 = capture off.
+ */
+#define CAPTURE_RATE_STATUS 0x17 /* Capture sample rate. Read only */
+ /* [15:0] Not used.
+ * [18:16] Channel 0 Detected sample rate. 0 - 44.1khz
+ * 1 - 48 khz
+ * 2 - 96 khz
+ * 3 - 192 khz
+ * 7 - undefined rate.
+ * [19] Channel 0. 1 - Valid, 0 - Not Valid.
+ * [22:20] Channel 1 Detected sample rate.
+ * [23] Channel 1. 1 - Valid, 0 - Not Valid.
+ * [26:24] Channel 2 Detected sample rate.
+ * [27] Channel 2. 1 - Valid, 0 - Not Valid.
+ * [30:28] Channel 3 Detected sample rate.
+ * [31] Channel 3. 1 - Valid, 0 - Not Valid.
+ */
+/* 0x18 - 0x1f unused */
+#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played. Read only */
+/* 0x21 - 0x3f unused */
+#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
+ /* Playback (0x1<<channel_id) Don't touch high 16bits. */
+ /* Capture (0x100<<channel_id). not tested */
+ /* Start Playback [3:0] (one bit per channel)
+ * Start Capture [11:8] (one bit per channel)
+ * Record source select for channel 0 [18:16]
+ * Record source select for channel 1 [22:20]
+ * Record source select for channel 2 [26:24]
+ * Record source select for channel 3 [30:28]
+ * 0 - SPDIF channel.
+ * 1 - I2S channel.
+ * 2 - SRC48 channel.
+ * 3 - SRCMulti_SPDIF channel.
+ * 4 - SRCMulti_I2S channel.
+ * 5 - SPDIF channel.
+ * 6 - fxengine capture.
+ * 7 - AC97 capture.
+ */
+ /* Default 41110000.
+ * Writing 0xffffffff hangs the PC.
+ * Writing 0xffff0000 -> 77770000 so it must be some sort of route.
+ * bit 0x1 starts DMA playback on channel_id 0
+ */
+/* 0x41,42 take values from 0 - 0xffffffff, but have no effect on playback */
+/* 0x43,0x48 do not remember settings */
+/* 0x41-45 unused */
+#define WATERMARK 0x46 /* Test bit to indicate cache level usage */
+ /* Values it can have while playing on channel 0.
+ * 0000f000, 0000f004, 0000f008, 0000f00c.
+ * Readonly.
+ */
+/* 0x47-0x4f unused */
+/* 0x50-0x5f Capture cache data */
+#define SRCSel 0x60 /* SRCSel. Default 0x4. Bypass P16V 0x14 */
+ /* [0] 0 = 10K2 audio, 1 = SRC48 mixer output.
+ * [2] 0 = 10K2 audio, 1 = SRCMulti SPDIF mixer output.
+ * [4] 0 = 10K2 audio, 1 = SRCMulti I2S mixer output.
+ */
+ /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
+ /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
+ /* SRC48 and SRCMULTI sample rate select and output select. */
+ /* 0xffffffff -> 0xC0000015
+ * 0xXXXXXXX4 = Enable Front Left/Right
+ * Enable PCMs
+ */
+
+/* 0x61 -> 0x6c are Volume controls */
+#define PLAYBACK_VOLUME_MIXER1 0x61 /* SRC48 Low to mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER2 0x62 /* SRC48 High to mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER3 0x63 /* SRCMULTI SPDIF Low to mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER4 0x64 /* SRCMULTI SPDIF High to mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER5 0x65 /* SRCMULTI I2S Low to mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER6 0x66 /* SRCMULTI I2S High to mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER7 0x67 /* P16V Low to SRCMULTI SPDIF mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER8 0x68 /* P16V High to SRCMULTI SPDIF mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER9 0x69 /* P16V Low to SRCMULTI I2S mixer input volume control. */
+ /* 0xXXXX3030 = PCM0 Volume (Front).
+ * 0x3030XXXX = PCM1 Volume (Center)
+ */
+#define PLAYBACK_VOLUME_MIXER10 0x6a /* P16V High to SRCMULTI I2S mixer input volume control. */
+ /* 0x3030XXXX = PCM3 Volume (Rear). */
+#define PLAYBACK_VOLUME_MIXER11 0x6b /* E10K2 Low to SRC48 mixer input volume control. */
+#define PLAYBACK_VOLUME_MIXER12 0x6c /* E10K2 High to SRC48 mixer input volume control. */
+
+#define SRC48_ENABLE 0x6d /* SRC48 input audio enable */
+ /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
+ /* [23:16] The corresponding P16V channel to SRC48 enabled if == 1.
+ * [31:24] The corresponding E10K2 channel to SRC48 enabled.
+ */
+#define SRCMULTI_ENABLE 0x6e /* SRCMulti input audio enable. Default 0xffffffff */
+ /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
+ /* [7:0] The corresponding P16V channel to SRCMulti_I2S enabled if == 1.
+ * [15:8] The corresponding E10K2 channel to SRCMulti I2S enabled.
+ * [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled.
+ * [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled.
+ */
+ /* Bypass P16V 0xff00ff00
+ * Bitmap. 0 = Off, 1 = On.
+ * P16V playback outputs:
+ * 0xXXXXXXX1 = PCM0 Left. (Front)
+ * 0xXXXXXXX2 = PCM0 Right.
+ * 0xXXXXXXX4 = PCM1 Left. (Center/LFE)
+ * 0xXXXXXXX8 = PCM1 Right.
+ * 0xXXXXXX1X = PCM2 Left. (Unknown)
+ * 0xXXXXXX2X = PCM2 Right.
+ * 0xXXXXXX4X = PCM3 Left. (Rear)
+ * 0xXXXXXX8X = PCM3 Right.
+ */
+#define AUDIO_OUT_ENABLE 0x6f /* Default: 000100FF */
+ /* [3:0] Does something, but not documented. Probably capture enable.
+ * [7:4] Playback channels enable. not documented.
+ * [16] AC97 output enable if == 1
+ * [30] 0 = SRCMulti_I2S input from fxengine 0x68-0x6f.
+ * 1 = SRCMulti_I2S input from SRC48 output.
+ * [31] 0 = SRCMulti_SPDIF input from fxengine 0x60-0x67.
+ * 1 = SRCMulti_SPDIF input from SRC48 output.
+ */
+ /* 0xffffffff -> C00100FF */
+ /* 0 -> Not playback sound, irq still running */
+ /* 0xXXXXXX10 = PCM0 Left/Right On. (Front)
+ * 0xXXXXXX20 = PCM1 Left/Right On. (Center/LFE)
+ * 0xXXXXXX40 = PCM2 Left/Right On. (Unknown)
+ * 0xXXXXXX80 = PCM3 Left/Right On. (Rear)
+ */
+#define PLAYBACK_SPDIF_SELECT 0x70 /* Default: 12030F00 */
+ /* 0xffffffff -> 3FF30FFF */
+ /* 0x00000001 pauses stream/irq fail. */
+ /* All other bits do not effect playback */
+#define PLAYBACK_SPDIF_SRC_SELECT 0x71 /* Default: 0000E4E4 */
+ /* 0xffffffff -> F33FFFFF */
+ /* All bits do not effect playback */
+#define PLAYBACK_SPDIF_USER_DATA0 0x72 /* SPDIF out user data 0 */
+#define PLAYBACK_SPDIF_USER_DATA1 0x73 /* SPDIF out user data 1 */
+/* 0x74-0x75 unknown */
+#define CAPTURE_SPDIF_CONTROL 0x76 /* SPDIF in control setting */
+#define CAPTURE_SPDIF_STATUS 0x77 /* SPDIF in status */
+#define CAPURE_SPDIF_USER_DATA0 0x78 /* SPDIF in user data 0 */
+#define CAPURE_SPDIF_USER_DATA1 0x79 /* SPDIF in user data 1 */
+#define CAPURE_SPDIF_USER_DATA2 0x7a /* SPDIF in user data 2 */
+
diff --git a/sound/pci/emu10k1/p17v.h b/sound/pci/emu10k1/p17v.h
new file mode 100644
index 0000000..4ef5f68
--- /dev/null
+++ b/sound/pci/emu10k1/p17v.h
@@ -0,0 +1,158 @@
+/*
+ * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver p17v chips
+ * Version: 0.01
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/******************************************************************************/
+/* Audigy2Value Tina (P17V) pointer-offset register set,
+ * accessed through the PTR20 and DATA24 registers */
+/******************************************************************************/
+
+/* 00 - 07: Not used */
+#define P17V_PLAYBACK_FIFO_PTR 0x08 /* Current playback fifo pointer
+ * and number of sound samples in cache.
+ */
+/* 09 - 12: Not used */
+#define P17V_CAPTURE_FIFO_PTR 0x13 /* Current capture fifo pointer
+ * and number of sound samples in cache.
+ */
+/* 14 - 17: Not used */
+#define P17V_PB_CHN_SEL 0x18 /* P17v playback channel select */
+#define P17V_SE_SLOT_SEL_L 0x19 /* Sound Engine slot select low */
+#define P17V_SE_SLOT_SEL_H 0x1a /* Sound Engine slot select high */
+/* 1b - 1f: Not used */
+/* 20 - 2f: Not used */
+/* 30 - 3b: Not used */
+#define P17V_SPI 0x3c /* SPI interface register */
+#define P17V_I2C_ADDR 0x3d /* I2C Address */
+#define P17V_I2C_0 0x3e /* I2C Data */
+#define P17V_I2C_1 0x3f /* I2C Data */
+/* I2C values */
+#define I2C_A_ADC_ADD_MASK 0x000000fe /*The address is a 7 bit address */
+#define I2C_A_ADC_RW_MASK 0x00000001 /*bit mask for R/W */
+#define I2C_A_ADC_TRANS_MASK 0x00000010 /*Bit mask for I2c address DAC value */
+#define I2C_A_ADC_ABORT_MASK 0x00000020 /*Bit mask for I2C transaction abort flag */
+#define I2C_A_ADC_LAST_MASK 0x00000040 /*Bit mask for Last word transaction */
+#define I2C_A_ADC_BYTE_MASK 0x00000080 /*Bit mask for Byte Mode */
+
+#define I2C_A_ADC_ADD 0x00000034 /*This is the Device address for ADC */
+#define I2C_A_ADC_READ 0x00000001 /*To perform a read operation */
+#define I2C_A_ADC_START 0x00000100 /*Start I2C transaction */
+#define I2C_A_ADC_ABORT 0x00000200 /*I2C transaction abort */
+#define I2C_A_ADC_LAST 0x00000400 /*I2C last transaction */
+#define I2C_A_ADC_BYTE 0x00000800 /*I2C one byte mode */
+
+#define I2C_D_ADC_REG_MASK 0xfe000000 /*ADC address register */
+#define I2C_D_ADC_DAT_MASK 0x01ff0000 /*ADC data register */
+
+#define ADC_TIMEOUT 0x00000007 /*ADC Timeout Clock Disable */
+#define ADC_IFC_CTRL 0x0000000b /*ADC Interface Control */
+#define ADC_MASTER 0x0000000c /*ADC Master Mode Control */
+#define ADC_POWER 0x0000000d /*ADC PowerDown Control */
+#define ADC_ATTEN_ADCL 0x0000000e /*ADC Attenuation ADCL */
+#define ADC_ATTEN_ADCR 0x0000000f /*ADC Attenuation ADCR */
+#define ADC_ALC_CTRL1 0x00000010 /*ADC ALC Control 1 */
+#define ADC_ALC_CTRL2 0x00000011 /*ADC ALC Control 2 */
+#define ADC_ALC_CTRL3 0x00000012 /*ADC ALC Control 3 */
+#define ADC_NOISE_CTRL 0x00000013 /*ADC Noise Gate Control */
+#define ADC_LIMIT_CTRL 0x00000014 /*ADC Limiter Control */
+#define ADC_MUX 0x00000015 /*ADC Mux offset */
+#if 0
+/* FIXME: Not tested yet. */
+#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain
+#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB
+#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute
+#define ADC_MUTE 0x000000c0 //Value to mute ADC
+#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select
+#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock
+#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter
+#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window
+#endif
+
+#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux
+#define ADC_MUX_0 0x00000001 //Value to select Unknown at ADC Mux (Not used)
+#define ADC_MUX_1 0x00000002 //Value to select Unknown at ADC Mux (Not used)
+#define ADC_MUX_2 0x00000004 //Value to select Mic at ADC Mux
+#define ADC_MUX_3 0x00000008 //Value to select Line-In at ADC Mux
+
+#define P17V_START_AUDIO 0x40 /* Start Audio bit */
+/* 41 - 47: Reserved */
+#define P17V_START_CAPTURE 0x48 /* Start Capture bit */
+#define P17V_CAPTURE_FIFO_BASE 0x49 /* Record FIFO base address */
+#define P17V_CAPTURE_FIFO_SIZE 0x4a /* Record FIFO buffer size */
+#define P17V_CAPTURE_FIFO_INDEX 0x4b /* Record FIFO capture index */
+#define P17V_CAPTURE_VOL_H 0x4c /* P17v capture volume control */
+#define P17V_CAPTURE_VOL_L 0x4d /* P17v capture volume control */
+/* 4e - 4f: Not used */
+/* 50 - 5f: Not used */
+#define P17V_SRCSel 0x60 /* SRC48 and SRCMulti sample rate select
+ * and output select
+ */
+#define P17V_MIXER_AC97_10K1_VOL_L 0x61 /* 10K to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_10K1_VOL_H 0x62 /* 10K to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_P17V_VOL_L 0x63 /* P17V to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_P17V_VOL_H 0x64 /* P17V to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_SRP_REC_VOL_L 0x65 /* SRP Record to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_SRP_REC_VOL_H 0x66 /* SRP Record to Mixer_AC97 input volume control */
+/* 67 - 68: Reserved */
+#define P17V_MIXER_Spdif_10K1_VOL_L 0x69 /* 10K to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_10K1_VOL_H 0x6A /* 10K to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_P17V_VOL_L 0x6B /* P17V to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_P17V_VOL_H 0x6C /* P17V to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_SRP_REC_VOL_L 0x6D /* SRP Record to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_SRP_REC_VOL_H 0x6E /* SRP Record to Mixer_Spdif input volume control */
+/* 6f - 70: Reserved */
+#define P17V_MIXER_I2S_10K1_VOL_L 0x71 /* 10K to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_10K1_VOL_H 0x72 /* 10K to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_P17V_VOL_L 0x73 /* P17V to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_P17V_VOL_H 0x74 /* P17V to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_SRP_REC_VOL_L 0x75 /* SRP Record to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_SRP_REC_VOL_H 0x76 /* SRP Record to Mixer_I2S input volume control */
+/* 77 - 78: Reserved */
+#define P17V_MIXER_AC97_ENABLE 0x79 /* Mixer AC97 input audio enable */
+#define P17V_MIXER_SPDIF_ENABLE 0x7A /* Mixer SPDIF input audio enable */
+#define P17V_MIXER_I2S_ENABLE 0x7B /* Mixer I2S input audio enable */
+#define P17V_AUDIO_OUT_ENABLE 0x7C /* Audio out enable */
+#define P17V_MIXER_ATT 0x7D /* SRP Mixer Attenuation Select */
+#define P17V_SRP_RECORD_SRR 0x7E /* SRP Record channel source Select */
+#define P17V_SOFT_RESET_SRP_MIXER 0x7F /* SRP and mixer soft reset */
+
+#define P17V_AC97_OUT_MASTER_VOL_L 0x80 /* AC97 Output master volume control */
+#define P17V_AC97_OUT_MASTER_VOL_H 0x81 /* AC97 Output master volume control */
+#define P17V_SPDIF_OUT_MASTER_VOL_L 0x82 /* SPDIF Output master volume control */
+#define P17V_SPDIF_OUT_MASTER_VOL_H 0x83 /* SPDIF Output master volume control */
+#define P17V_I2S_OUT_MASTER_VOL_L 0x84 /* I2S Output master volume control */
+#define P17V_I2S_OUT_MASTER_VOL_H 0x85 /* I2S Output master volume control */
+/* 86 - 87: Not used */
+#define P17V_I2S_CHANNEL_SWAP_PHASE_INVERSE 0x88 /* I2S out mono channel swap
+ * and phase inverse */
+#define P17V_SPDIF_CHANNEL_SWAP_PHASE_INVERSE 0x89 /* SPDIF out mono channel swap
+ * and phase inverse */
+/* 8A: Not used */
+#define P17V_SRP_P17V_ESR 0x8B /* SRP_P17V estimated sample rate and rate lock */
+#define P17V_SRP_REC_ESR 0x8C /* SRP_REC estimated sample rate and rate lock */
+#define P17V_SRP_BYPASS 0x8D /* srps channel bypass and srps bypass */
+/* 8E - 92: Not used */
+#define P17V_I2S_SRC_SEL 0x93 /* I2SIN mode sel */
+
+
+
+
+
+
diff --git a/sound/pci/emu10k1/timer.c b/sound/pci/emu10k1/timer.c
new file mode 100644
index 0000000..72321e9
--- /dev/null
+++ b/sound/pci/emu10k1/timer.c
@@ -0,0 +1,96 @@
+/*
+ * Copyright (c) by Lee Revell <rlrevell@joe-job.com>
+ * Clemens Ladisch <clemens@ladisch.de>
+ * Routines for control of EMU10K1 chips
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+
+static int snd_emu10k1_timer_start(struct snd_timer *timer)
+{
+ struct snd_emu10k1 *emu;
+ unsigned long flags;
+ unsigned int delay;
+
+ emu = snd_timer_chip(timer);
+ delay = timer->sticks - 1;
+ if (delay < 5 ) /* minimum time is 5 ticks */
+ delay = 5;
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ snd_emu10k1_intr_enable(emu, INTE_INTERVALTIMERENB);
+ outw(delay & TIMER_RATE_MASK, emu->port + TIMER);
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_timer_stop(struct snd_timer *timer)
+{
+ struct snd_emu10k1 *emu;
+ unsigned long flags;
+
+ emu = snd_timer_chip(timer);
+ spin_lock_irqsave(&emu->reg_lock, flags);
+ snd_emu10k1_intr_disable(emu, INTE_INTERVALTIMERENB);
+ spin_unlock_irqrestore(&emu->reg_lock, flags);
+ return 0;
+}
+
+static int snd_emu10k1_timer_precise_resolution(struct snd_timer *timer,
+ unsigned long *num, unsigned long *den)
+{
+ *num = 1;
+ *den = 48000;
+ return 0;
+}
+
+static struct snd_timer_hardware snd_emu10k1_timer_hw = {
+ .flags = SNDRV_TIMER_HW_AUTO,
+ .resolution = 20833, /* 1 sample @ 48KHZ = 20.833...us */
+ .ticks = 1024,
+ .start = snd_emu10k1_timer_start,
+ .stop = snd_emu10k1_timer_stop,
+ .precise_resolution = snd_emu10k1_timer_precise_resolution,
+};
+
+int __devinit snd_emu10k1_timer(struct snd_emu10k1 *emu, int device)
+{
+ struct snd_timer *timer = NULL;
+ struct snd_timer_id tid;
+ int err;
+
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = emu->card->number;
+ tid.device = device;
+ tid.subdevice = 0;
+ if ((err = snd_timer_new(emu->card, "EMU10K1", &tid, &timer)) >= 0) {
+ strcpy(timer->name, "EMU10K1 timer");
+ timer->private_data = emu;
+ timer->hw = snd_emu10k1_timer_hw;
+ }
+ emu->timer = timer;
+ return err;
+}
diff --git a/sound/pci/emu10k1/tina2.h b/sound/pci/emu10k1/tina2.h
new file mode 100644
index 0000000..f2d8eb6
--- /dev/null
+++ b/sound/pci/emu10k1/tina2.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver tina2 chips
+ * Version: 0.1
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/********************************************************************************************************/
+/* Audigy2 Tina2 (notebook) pointer-offset register set, accessed through the PTR2 and DATA2 registers */
+/********************************************************************************************************/
+
+#define TINA2_VOLUME 0x71 /* Attenuate playback volume to prevent distortion. */
+ /* The windows driver does not use this register,
+ * so it must use some other attenuation method.
+ * Without this, the output is 12dB too loud,
+ * resulting in distortion.
+ */
+
diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c
new file mode 100644
index 0000000..d7300a1
--- /dev/null
+++ b/sound/pci/emu10k1/voice.c
@@ -0,0 +1,161 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Creative Labs, Inc.
+ * Lee Revell <rlrevell@joe-job.com>
+ * Routines for control of EMU10K1 chips - voice manager
+ *
+ * Rewrote voice allocator for multichannel support - rlrevell 12/2004
+ *
+ * BUGS:
+ * --
+ *
+ * TODO:
+ * --
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/emu10k1.h>
+
+/* Previously the voice allocator started at 0 every time. The new voice
+ * allocator uses a round robin scheme. The next free voice is tracked in
+ * the card record and each allocation begins where the last left off. The
+ * hardware requires stereo interleaved voices be aligned to an even/odd
+ * boundary. For multichannel voice allocation we ensure than the block of
+ * voices does not cross the 32 voice boundary. This simplifies the
+ * multichannel support and ensures we can use a single write to the
+ * (set|clear)_loop_stop registers. Otherwise (for example) the voices would
+ * get out of sync when pausing/resuming a stream.
+ * --rlrevell
+ */
+
+static int voice_alloc(struct snd_emu10k1 *emu, int type, int number,
+ struct snd_emu10k1_voice **rvoice)
+{
+ struct snd_emu10k1_voice *voice;
+ int i, j, k, first_voice, last_voice, skip;
+
+ *rvoice = NULL;
+ first_voice = last_voice = 0;
+ for (i = emu->next_free_voice, j = 0; j < NUM_G ; i += number, j += number) {
+ // printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice);
+ i %= NUM_G;
+
+ /* stereo voices must be even/odd */
+ if ((number == 2) && (i % 2)) {
+ i++;
+ continue;
+ }
+
+ skip = 0;
+ for (k = 0; k < number; k++) {
+ voice = &emu->voices[(i+k) % NUM_G];
+ if (voice->use) {
+ skip = 1;
+ break;
+ }
+ }
+ if (!skip) {
+ // printk("allocated voice %d\n", i);
+ first_voice = i;
+ last_voice = (i + number) % NUM_G;
+ emu->next_free_voice = last_voice;
+ break;
+ }
+ }
+
+ if (first_voice == last_voice)
+ return -ENOMEM;
+
+ for (i = 0; i < number; i++) {
+ voice = &emu->voices[(first_voice + i) % NUM_G];
+ // printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number);
+ voice->use = 1;
+ switch (type) {
+ case EMU10K1_PCM:
+ voice->pcm = 1;
+ break;
+ case EMU10K1_SYNTH:
+ voice->synth = 1;
+ break;
+ case EMU10K1_MIDI:
+ voice->midi = 1;
+ break;
+ case EMU10K1_EFX:
+ voice->efx = 1;
+ break;
+ }
+ }
+ *rvoice = &emu->voices[first_voice];
+ return 0;
+}
+
+int snd_emu10k1_voice_alloc(struct snd_emu10k1 *emu, int type, int number,
+ struct snd_emu10k1_voice **rvoice)
+{
+ unsigned long flags;
+ int result;
+
+ if (snd_BUG_ON(!rvoice))
+ return -EINVAL;
+ if (snd_BUG_ON(!number))
+ return -EINVAL;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (;;) {
+ result = voice_alloc(emu, type, number, rvoice);
+ if (result == 0 || type == EMU10K1_SYNTH || type == EMU10K1_MIDI)
+ break;
+
+ /* free a voice from synth */
+ if (emu->get_synth_voice) {
+ result = emu->get_synth_voice(emu);
+ if (result >= 0) {
+ struct snd_emu10k1_voice *pvoice = &emu->voices[result];
+ pvoice->interrupt = NULL;
+ pvoice->use = pvoice->pcm = pvoice->synth = pvoice->midi = pvoice->efx = 0;
+ pvoice->epcm = NULL;
+ }
+ }
+ if (result < 0)
+ break;
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+
+ return result;
+}
+
+EXPORT_SYMBOL(snd_emu10k1_voice_alloc);
+
+int snd_emu10k1_voice_free(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_voice *pvoice)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!pvoice))
+ return -EINVAL;
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ pvoice->interrupt = NULL;
+ pvoice->use = pvoice->pcm = pvoice->synth = pvoice->midi = pvoice->efx = 0;
+ pvoice->epcm = NULL;
+ snd_emu10k1_voice_init(emu, pvoice->number);
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_emu10k1_voice_free);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
new file mode 100644
index 0000000..9bf9536
--- /dev/null
+++ b/sound/pci/ens1370.c
@@ -0,0 +1,2497 @@
+/*
+ * Driver for Ensoniq ES1370/ES1371 AudioPCI soundcard
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
+ * Thomas Sailer <sailer@ife.ee.ethz.ch>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* Power-Management-Code ( CONFIG_PM )
+ * for ens1371 only ( FIXME )
+ * derived from cs4281.c, atiixp.c and via82xx.c
+ * using http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c1540.htm
+ * by Kurt J. Bosch
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#ifdef CHIP1371
+#include <sound/ac97_codec.h>
+#else
+#include <sound/ak4531_codec.h>
+#endif
+#include <sound/initval.h>
+#include <sound/asoundef.h>
+
+#ifndef CHIP1371
+#undef CHIP1370
+#define CHIP1370
+#endif
+
+#ifdef CHIP1370
+#define DRIVER_NAME "ENS1370"
+#else
+#define DRIVER_NAME "ENS1371"
+#endif
+
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Thomas Sailer <sailer@ife.ee.ethz.ch>");
+MODULE_LICENSE("GPL");
+#ifdef CHIP1370
+MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370");
+MODULE_SUPPORTED_DEVICE("{{Ensoniq,AudioPCI-97 ES1370},"
+ "{Creative Labs,SB PCI64/128 (ES1370)}}");
+#endif
+#ifdef CHIP1371
+MODULE_DESCRIPTION("Ensoniq/Creative AudioPCI ES1371+");
+MODULE_SUPPORTED_DEVICE("{{Ensoniq,AudioPCI ES1371/73},"
+ "{Ensoniq,AudioPCI ES1373},"
+ "{Creative Labs,Ectiva EV1938},"
+ "{Creative Labs,SB PCI64/128 (ES1371/73)},"
+ "{Creative Labs,Vibra PCI128},"
+ "{Ectiva,EV1938}}");
+#endif
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */
+#ifdef SUPPORT_JOYSTICK
+#ifdef CHIP1371
+static int joystick_port[SNDRV_CARDS];
+#else
+static int joystick[SNDRV_CARDS];
+#endif
+#endif
+#ifdef CHIP1371
+static int spdif[SNDRV_CARDS];
+static int lineio[SNDRV_CARDS];
+#endif
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Ensoniq AudioPCI soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Ensoniq AudioPCI soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Ensoniq AudioPCI soundcard.");
+#ifdef SUPPORT_JOYSTICK
+#ifdef CHIP1371
+module_param_array(joystick_port, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_port, "Joystick port address.");
+#else
+module_param_array(joystick, bool, NULL, 0444);
+MODULE_PARM_DESC(joystick, "Enable joystick.");
+#endif
+#endif /* SUPPORT_JOYSTICK */
+#ifdef CHIP1371
+module_param_array(spdif, int, NULL, 0444);
+MODULE_PARM_DESC(spdif, "S/PDIF output (-1 = none, 0 = auto, 1 = force).");
+module_param_array(lineio, int, NULL, 0444);
+MODULE_PARM_DESC(lineio, "Line In to Rear Out (0 = auto, 1 = force).");
+#endif
+
+/* ES1371 chip ID */
+/* This is a little confusing because all ES1371 compatible chips have the
+ same DEVICE_ID, the only thing differentiating them is the REV_ID field.
+ This is only significant if you want to enable features on the later parts.
+ Yes, I know it's stupid and why didn't we use the sub IDs?
+*/
+#define ES1371REV_ES1373_A 0x04
+#define ES1371REV_ES1373_B 0x06
+#define ES1371REV_CT5880_A 0x07
+#define CT5880REV_CT5880_C 0x02
+#define CT5880REV_CT5880_D 0x03 /* ??? -jk */
+#define CT5880REV_CT5880_E 0x04 /* mw */
+#define ES1371REV_ES1371_B 0x09
+#define EV1938REV_EV1938_A 0x00
+#define ES1371REV_ES1373_8 0x08
+
+/*
+ * Direct registers
+ */
+
+#define ES_REG(ensoniq, x) ((ensoniq)->port + ES_REG_##x)
+
+#define ES_REG_CONTROL 0x00 /* R/W: Interrupt/Chip select control register */
+#define ES_1370_ADC_STOP (1<<31) /* disable capture buffer transfers */
+#define ES_1370_XCTL1 (1<<30) /* general purpose output bit */
+#define ES_1373_BYPASS_P1 (1<<31) /* bypass SRC for PB1 */
+#define ES_1373_BYPASS_P2 (1<<30) /* bypass SRC for PB2 */
+#define ES_1373_BYPASS_R (1<<29) /* bypass SRC for REC */
+#define ES_1373_TEST_BIT (1<<28) /* should be set to 0 for normal operation */
+#define ES_1373_RECEN_B (1<<27) /* mix record with playback for I2S/SPDIF out */
+#define ES_1373_SPDIF_THRU (1<<26) /* 0 = SPDIF thru mode, 1 = SPDIF == dig out */
+#define ES_1371_JOY_ASEL(o) (((o)&0x03)<<24)/* joystick port mapping */
+#define ES_1371_JOY_ASELM (0x03<<24) /* mask for above */
+#define ES_1371_JOY_ASELI(i) (((i)>>24)&0x03)
+#define ES_1371_GPIO_IN(i) (((i)>>20)&0x0f)/* GPIO in [3:0] pins - R/O */
+#define ES_1370_PCLKDIVO(o) (((o)&0x1fff)<<16)/* clock divide ratio for DAC2 */
+#define ES_1370_PCLKDIVM ((0x1fff)<<16) /* mask for above */
+#define ES_1370_PCLKDIVI(i) (((i)>>16)&0x1fff)/* clock divide ratio for DAC2 */
+#define ES_1371_GPIO_OUT(o) (((o)&0x0f)<<16)/* GPIO out [3:0] pins - W/R */
+#define ES_1371_GPIO_OUTM (0x0f<<16) /* mask for above */
+#define ES_MSFMTSEL (1<<15) /* MPEG serial data format; 0 = SONY, 1 = I2S */
+#define ES_1370_M_SBB (1<<14) /* clock source for DAC - 0 = clock generator; 1 = MPEG clocks */
+#define ES_1371_SYNC_RES (1<<14) /* Warm AC97 reset */
+#define ES_1370_WTSRSEL(o) (((o)&0x03)<<12)/* fixed frequency clock for DAC1 */
+#define ES_1370_WTSRSELM (0x03<<12) /* mask for above */
+#define ES_1371_ADC_STOP (1<<13) /* disable CCB transfer capture information */
+#define ES_1371_PWR_INTRM (1<<12) /* power level change interrupts enable */
+#define ES_1370_DAC_SYNC (1<<11) /* DAC's are synchronous */
+#define ES_1371_M_CB (1<<11) /* capture clock source; 0 = AC'97 ADC; 1 = I2S */
+#define ES_CCB_INTRM (1<<10) /* CCB voice interrupts enable */
+#define ES_1370_M_CB (1<<9) /* capture clock source; 0 = ADC; 1 = MPEG */
+#define ES_1370_XCTL0 (1<<8) /* generap purpose output bit */
+#define ES_1371_PDLEV(o) (((o)&0x03)<<8) /* current power down level */
+#define ES_1371_PDLEVM (0x03<<8) /* mask for above */
+#define ES_BREQ (1<<7) /* memory bus request enable */
+#define ES_DAC1_EN (1<<6) /* DAC1 playback channel enable */
+#define ES_DAC2_EN (1<<5) /* DAC2 playback channel enable */
+#define ES_ADC_EN (1<<4) /* ADC capture channel enable */
+#define ES_UART_EN (1<<3) /* UART enable */
+#define ES_JYSTK_EN (1<<2) /* Joystick module enable */
+#define ES_1370_CDC_EN (1<<1) /* Codec interface enable */
+#define ES_1371_XTALCKDIS (1<<1) /* Xtal clock disable */
+#define ES_1370_SERR_DISABLE (1<<0) /* PCI serr signal disable */
+#define ES_1371_PCICLKDIS (1<<0) /* PCI clock disable */
+#define ES_REG_STATUS 0x04 /* R/O: Interrupt/Chip select status register */
+#define ES_INTR (1<<31) /* Interrupt is pending */
+#define ES_1371_ST_AC97_RST (1<<29) /* CT5880 AC'97 Reset bit */
+#define ES_1373_REAR_BIT27 (1<<27) /* rear bits: 000 - front, 010 - mirror, 101 - separate */
+#define ES_1373_REAR_BIT26 (1<<26)
+#define ES_1373_REAR_BIT24 (1<<24)
+#define ES_1373_GPIO_INT_EN(o)(((o)&0x0f)<<20)/* GPIO [3:0] pins - interrupt enable */
+#define ES_1373_SPDIF_EN (1<<18) /* SPDIF enable */
+#define ES_1373_SPDIF_TEST (1<<17) /* SPDIF test */
+#define ES_1371_TEST (1<<16) /* test ASIC */
+#define ES_1373_GPIO_INT(i) (((i)&0x0f)>>12)/* GPIO [3:0] pins - interrupt pending */
+#define ES_1370_CSTAT (1<<10) /* CODEC is busy or register write in progress */
+#define ES_1370_CBUSY (1<<9) /* CODEC is busy */
+#define ES_1370_CWRIP (1<<8) /* CODEC register write in progress */
+#define ES_1371_SYNC_ERR (1<<8) /* CODEC synchronization error occurred */
+#define ES_1371_VC(i) (((i)>>6)&0x03) /* voice code from CCB module */
+#define ES_1370_VC(i) (((i)>>5)&0x03) /* voice code from CCB module */
+#define ES_1371_MPWR (1<<5) /* power level interrupt pending */
+#define ES_MCCB (1<<4) /* CCB interrupt pending */
+#define ES_UART (1<<3) /* UART interrupt pending */
+#define ES_DAC1 (1<<2) /* DAC1 channel interrupt pending */
+#define ES_DAC2 (1<<1) /* DAC2 channel interrupt pending */
+#define ES_ADC (1<<0) /* ADC channel interrupt pending */
+#define ES_REG_UART_DATA 0x08 /* R/W: UART data register */
+#define ES_REG_UART_STATUS 0x09 /* R/O: UART status register */
+#define ES_RXINT (1<<7) /* RX interrupt occurred */
+#define ES_TXINT (1<<2) /* TX interrupt occurred */
+#define ES_TXRDY (1<<1) /* transmitter ready */
+#define ES_RXRDY (1<<0) /* receiver ready */
+#define ES_REG_UART_CONTROL 0x09 /* W/O: UART control register */
+#define ES_RXINTEN (1<<7) /* RX interrupt enable */
+#define ES_TXINTENO(o) (((o)&0x03)<<5) /* TX interrupt enable */
+#define ES_TXINTENM (0x03<<5) /* mask for above */
+#define ES_TXINTENI(i) (((i)>>5)&0x03)
+#define ES_CNTRL(o) (((o)&0x03)<<0) /* control */
+#define ES_CNTRLM (0x03<<0) /* mask for above */
+#define ES_REG_UART_RES 0x0a /* R/W: UART reserver register */
+#define ES_TEST_MODE (1<<0) /* test mode enabled */
+#define ES_REG_MEM_PAGE 0x0c /* R/W: Memory page register */
+#define ES_MEM_PAGEO(o) (((o)&0x0f)<<0) /* memory page select - out */
+#define ES_MEM_PAGEM (0x0f<<0) /* mask for above */
+#define ES_MEM_PAGEI(i) (((i)>>0)&0x0f) /* memory page select - in */
+#define ES_REG_1370_CODEC 0x10 /* W/O: Codec write register address */
+#define ES_1370_CODEC_WRITE(a,d) ((((a)&0xff)<<8)|(((d)&0xff)<<0))
+#define ES_REG_1371_CODEC 0x14 /* W/R: Codec Read/Write register address */
+#define ES_1371_CODEC_RDY (1<<31) /* codec ready */
+#define ES_1371_CODEC_WIP (1<<30) /* codec register access in progress */
+#define ES_1371_CODEC_PIRD (1<<23) /* codec read/write select register */
+#define ES_1371_CODEC_WRITE(a,d) ((((a)&0x7f)<<16)|(((d)&0xffff)<<0))
+#define ES_1371_CODEC_READS(a) ((((a)&0x7f)<<16)|ES_1371_CODEC_PIRD)
+#define ES_1371_CODEC_READ(i) (((i)>>0)&0xffff)
+
+#define ES_REG_1371_SMPRATE 0x10 /* W/R: Codec rate converter interface register */
+#define ES_1371_SRC_RAM_ADDRO(o) (((o)&0x7f)<<25)/* address of the sample rate converter */
+#define ES_1371_SRC_RAM_ADDRM (0x7f<<25) /* mask for above */
+#define ES_1371_SRC_RAM_ADDRI(i) (((i)>>25)&0x7f)/* address of the sample rate converter */
+#define ES_1371_SRC_RAM_WE (1<<24) /* R/W: read/write control for sample rate converter */
+#define ES_1371_SRC_RAM_BUSY (1<<23) /* R/O: sample rate memory is busy */
+#define ES_1371_SRC_DISABLE (1<<22) /* sample rate converter disable */
+#define ES_1371_DIS_P1 (1<<21) /* playback channel 1 accumulator update disable */
+#define ES_1371_DIS_P2 (1<<20) /* playback channel 1 accumulator update disable */
+#define ES_1371_DIS_R1 (1<<19) /* capture channel accumulator update disable */
+#define ES_1371_SRC_RAM_DATAO(o) (((o)&0xffff)<<0)/* current value of the sample rate converter */
+#define ES_1371_SRC_RAM_DATAM (0xffff<<0) /* mask for above */
+#define ES_1371_SRC_RAM_DATAI(i) (((i)>>0)&0xffff)/* current value of the sample rate converter */
+
+#define ES_REG_1371_LEGACY 0x18 /* W/R: Legacy control/status register */
+#define ES_1371_JFAST (1<<31) /* fast joystick timing */
+#define ES_1371_HIB (1<<30) /* host interrupt blocking enable */
+#define ES_1371_VSB (1<<29) /* SB; 0 = addr 0x220xH, 1 = 0x22FxH */
+#define ES_1371_VMPUO(o) (((o)&0x03)<<27)/* base register address; 0 = 0x320xH; 1 = 0x330xH; 2 = 0x340xH; 3 = 0x350xH */
+#define ES_1371_VMPUM (0x03<<27) /* mask for above */
+#define ES_1371_VMPUI(i) (((i)>>27)&0x03)/* base register address */
+#define ES_1371_VCDCO(o) (((o)&0x03)<<25)/* CODEC; 0 = 0x530xH; 1 = undefined; 2 = 0xe80xH; 3 = 0xF40xH */
+#define ES_1371_VCDCM (0x03<<25) /* mask for above */
+#define ES_1371_VCDCI(i) (((i)>>25)&0x03)/* CODEC address */
+#define ES_1371_FIRQ (1<<24) /* force an interrupt */
+#define ES_1371_SDMACAP (1<<23) /* enable event capture for slave DMA controller */
+#define ES_1371_SPICAP (1<<22) /* enable event capture for slave IRQ controller */
+#define ES_1371_MDMACAP (1<<21) /* enable event capture for master DMA controller */
+#define ES_1371_MPICAP (1<<20) /* enable event capture for master IRQ controller */
+#define ES_1371_ADCAP (1<<19) /* enable event capture for ADLIB register; 0x388xH */
+#define ES_1371_SVCAP (1<<18) /* enable event capture for SB registers */
+#define ES_1371_CDCCAP (1<<17) /* enable event capture for CODEC registers */
+#define ES_1371_BACAP (1<<16) /* enable event capture for SoundScape base address */
+#define ES_1371_EXI(i) (((i)>>8)&0x07) /* event number */
+#define ES_1371_AI(i) (((i)>>3)&0x1f) /* event significant I/O address */
+#define ES_1371_WR (1<<2) /* event capture; 0 = read; 1 = write */
+#define ES_1371_LEGINT (1<<0) /* interrupt for legacy events; 0 = interrupt did occur */
+
+#define ES_REG_CHANNEL_STATUS 0x1c /* R/W: first 32-bits from S/PDIF channel status block, es1373 */
+
+#define ES_REG_SERIAL 0x20 /* R/W: Serial interface control register */
+#define ES_1371_DAC_TEST (1<<22) /* DAC test mode enable */
+#define ES_P2_END_INCO(o) (((o)&0x07)<<19)/* binary offset value to increment / loop end */
+#define ES_P2_END_INCM (0x07<<19) /* mask for above */
+#define ES_P2_END_INCI(i) (((i)>>16)&0x07)/* binary offset value to increment / loop end */
+#define ES_P2_ST_INCO(o) (((o)&0x07)<<16)/* binary offset value to increment / start */
+#define ES_P2_ST_INCM (0x07<<16) /* mask for above */
+#define ES_P2_ST_INCI(i) (((i)<<16)&0x07)/* binary offset value to increment / start */
+#define ES_R1_LOOP_SEL (1<<15) /* ADC; 0 - loop mode; 1 = stop mode */
+#define ES_P2_LOOP_SEL (1<<14) /* DAC2; 0 - loop mode; 1 = stop mode */
+#define ES_P1_LOOP_SEL (1<<13) /* DAC1; 0 - loop mode; 1 = stop mode */
+#define ES_P2_PAUSE (1<<12) /* DAC2; 0 - play mode; 1 = pause mode */
+#define ES_P1_PAUSE (1<<11) /* DAC1; 0 - play mode; 1 = pause mode */
+#define ES_R1_INT_EN (1<<10) /* ADC interrupt enable */
+#define ES_P2_INT_EN (1<<9) /* DAC2 interrupt enable */
+#define ES_P1_INT_EN (1<<8) /* DAC1 interrupt enable */
+#define ES_P1_SCT_RLD (1<<7) /* force sample counter reload for DAC1 */
+#define ES_P2_DAC_SEN (1<<6) /* when stop mode: 0 - DAC2 play back zeros; 1 = DAC2 play back last sample */
+#define ES_R1_MODEO(o) (((o)&0x03)<<4) /* ADC mode; 0 = 8-bit mono; 1 = 8-bit stereo; 2 = 16-bit mono; 3 = 16-bit stereo */
+#define ES_R1_MODEM (0x03<<4) /* mask for above */
+#define ES_R1_MODEI(i) (((i)>>4)&0x03)
+#define ES_P2_MODEO(o) (((o)&0x03)<<2) /* DAC2 mode; -- '' -- */
+#define ES_P2_MODEM (0x03<<2) /* mask for above */
+#define ES_P2_MODEI(i) (((i)>>2)&0x03)
+#define ES_P1_MODEO(o) (((o)&0x03)<<0) /* DAC1 mode; -- '' -- */
+#define ES_P1_MODEM (0x03<<0) /* mask for above */
+#define ES_P1_MODEI(i) (((i)>>0)&0x03)
+
+#define ES_REG_DAC1_COUNT 0x24 /* R/W: DAC1 sample count register */
+#define ES_REG_DAC2_COUNT 0x28 /* R/W: DAC2 sample count register */
+#define ES_REG_ADC_COUNT 0x2c /* R/W: ADC sample count register */
+#define ES_REG_CURR_COUNT(i) (((i)>>16)&0xffff)
+#define ES_REG_COUNTO(o) (((o)&0xffff)<<0)
+#define ES_REG_COUNTM (0xffff<<0)
+#define ES_REG_COUNTI(i) (((i)>>0)&0xffff)
+
+#define ES_REG_DAC1_FRAME 0x30 /* R/W: PAGE 0x0c; DAC1 frame address */
+#define ES_REG_DAC1_SIZE 0x34 /* R/W: PAGE 0x0c; DAC1 frame size */
+#define ES_REG_DAC2_FRAME 0x38 /* R/W: PAGE 0x0c; DAC2 frame address */
+#define ES_REG_DAC2_SIZE 0x3c /* R/W: PAGE 0x0c; DAC2 frame size */
+#define ES_REG_ADC_FRAME 0x30 /* R/W: PAGE 0x0d; ADC frame address */
+#define ES_REG_ADC_SIZE 0x34 /* R/W: PAGE 0x0d; ADC frame size */
+#define ES_REG_FCURR_COUNTO(o) (((o)&0xffff)<<16)
+#define ES_REG_FCURR_COUNTM (0xffff<<16)
+#define ES_REG_FCURR_COUNTI(i) (((i)>>14)&0x3fffc)
+#define ES_REG_FSIZEO(o) (((o)&0xffff)<<0)
+#define ES_REG_FSIZEM (0xffff<<0)
+#define ES_REG_FSIZEI(i) (((i)>>0)&0xffff)
+#define ES_REG_PHANTOM_FRAME 0x38 /* R/W: PAGE 0x0d: phantom frame address */
+#define ES_REG_PHANTOM_COUNT 0x3c /* R/W: PAGE 0x0d: phantom frame count */
+
+#define ES_REG_UART_FIFO 0x30 /* R/W: PAGE 0x0e; UART FIFO register */
+#define ES_REG_UF_VALID (1<<8)
+#define ES_REG_UF_BYTEO(o) (((o)&0xff)<<0)
+#define ES_REG_UF_BYTEM (0xff<<0)
+#define ES_REG_UF_BYTEI(i) (((i)>>0)&0xff)
+
+
+/*
+ * Pages
+ */
+
+#define ES_PAGE_DAC 0x0c
+#define ES_PAGE_ADC 0x0d
+#define ES_PAGE_UART 0x0e
+#define ES_PAGE_UART1 0x0f
+
+/*
+ * Sample rate converter addresses
+ */
+
+#define ES_SMPREG_DAC1 0x70
+#define ES_SMPREG_DAC2 0x74
+#define ES_SMPREG_ADC 0x78
+#define ES_SMPREG_VOL_ADC 0x6c
+#define ES_SMPREG_VOL_DAC1 0x7c
+#define ES_SMPREG_VOL_DAC2 0x7e
+#define ES_SMPREG_TRUNC_N 0x00
+#define ES_SMPREG_INT_REGS 0x01
+#define ES_SMPREG_ACCUM_FRAC 0x02
+#define ES_SMPREG_VFREQ_FRAC 0x03
+
+/*
+ * Some contants
+ */
+
+#define ES_1370_SRCLOCK 1411200
+#define ES_1370_SRTODIV(x) (ES_1370_SRCLOCK/(x)-2)
+
+/*
+ * Open modes
+ */
+
+#define ES_MODE_PLAY1 0x0001
+#define ES_MODE_PLAY2 0x0002
+#define ES_MODE_CAPTURE 0x0004
+
+#define ES_MODE_OUTPUT 0x0001 /* for MIDI */
+#define ES_MODE_INPUT 0x0002 /* for MIDI */
+
+/*
+
+ */
+
+struct ensoniq {
+ spinlock_t reg_lock;
+ struct mutex src_mutex;
+
+ int irq;
+
+ unsigned long playback1size;
+ unsigned long playback2size;
+ unsigned long capture3size;
+
+ unsigned long port;
+ unsigned int mode;
+ unsigned int uartm; /* UART mode */
+
+ unsigned int ctrl; /* control register */
+ unsigned int sctrl; /* serial control register */
+ unsigned int cssr; /* control status register */
+ unsigned int uartc; /* uart control register */
+ unsigned int rev; /* chip revision */
+
+ union {
+#ifdef CHIP1371
+ struct {
+ struct snd_ac97 *ac97;
+ } es1371;
+#else
+ struct {
+ int pclkdiv_lock;
+ struct snd_ak4531 *ak4531;
+ } es1370;
+#endif
+ } u;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm1; /* DAC1/ADC PCM */
+ struct snd_pcm *pcm2; /* DAC2 PCM */
+ struct snd_pcm_substream *playback1_substream;
+ struct snd_pcm_substream *playback2_substream;
+ struct snd_pcm_substream *capture_substream;
+ unsigned int p1_dma_size;
+ unsigned int p2_dma_size;
+ unsigned int c_dma_size;
+ unsigned int p1_period_size;
+ unsigned int p2_period_size;
+ unsigned int c_period_size;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *midi_input;
+ struct snd_rawmidi_substream *midi_output;
+
+ unsigned int spdif;
+ unsigned int spdif_default;
+ unsigned int spdif_stream;
+
+#ifdef CHIP1370
+ struct snd_dma_buffer dma_bug;
+#endif
+
+#ifdef SUPPORT_JOYSTICK
+ struct gameport *gameport;
+#endif
+};
+
+static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id);
+
+static struct pci_device_id snd_audiopci_ids[] = {
+#ifdef CHIP1370
+ { 0x1274, 0x5000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1370 */
+#endif
+#ifdef CHIP1371
+ { 0x1274, 0x1371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1371 */
+ { 0x1274, 0x5880, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1373 - CT5880 */
+ { 0x1102, 0x8938, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Ectiva EV1938 */
+#endif
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_audiopci_ids);
+
+/*
+ * constants
+ */
+
+#define POLL_COUNT 0xa000
+
+#ifdef CHIP1370
+static unsigned int snd_es1370_fixed_rates[] =
+ {5512, 11025, 22050, 44100};
+static struct snd_pcm_hw_constraint_list snd_es1370_hw_constraints_rates = {
+ .count = 4,
+ .list = snd_es1370_fixed_rates,
+ .mask = 0,
+};
+static struct snd_ratnum es1370_clock = {
+ .num = ES_1370_SRCLOCK,
+ .den_min = 29,
+ .den_max = 353,
+ .den_step = 1,
+};
+static struct snd_pcm_hw_constraint_ratnums snd_es1370_hw_constraints_clock = {
+ .nrats = 1,
+ .rats = &es1370_clock,
+};
+#else
+static struct snd_ratden es1371_dac_clock = {
+ .num_min = 3000 * (1 << 15),
+ .num_max = 48000 * (1 << 15),
+ .num_step = 3000,
+ .den = 1 << 15,
+};
+static struct snd_pcm_hw_constraint_ratdens snd_es1371_hw_constraints_dac_clock = {
+ .nrats = 1,
+ .rats = &es1371_dac_clock,
+};
+static struct snd_ratnum es1371_adc_clock = {
+ .num = 48000 << 15,
+ .den_min = 32768,
+ .den_max = 393216,
+ .den_step = 1,
+};
+static struct snd_pcm_hw_constraint_ratnums snd_es1371_hw_constraints_adc_clock = {
+ .nrats = 1,
+ .rats = &es1371_adc_clock,
+};
+#endif
+static const unsigned int snd_ensoniq_sample_shift[] =
+ {0, 1, 1, 2};
+
+/*
+ * common I/O routines
+ */
+
+#ifdef CHIP1371
+
+static unsigned int snd_es1371_wait_src_ready(struct ensoniq * ensoniq)
+{
+ unsigned int t, r = 0;
+
+ for (t = 0; t < POLL_COUNT; t++) {
+ r = inl(ES_REG(ensoniq, 1371_SMPRATE));
+ if ((r & ES_1371_SRC_RAM_BUSY) == 0)
+ return r;
+ cond_resched();
+ }
+ snd_printk(KERN_ERR "wait src ready timeout 0x%lx [0x%x]\n",
+ ES_REG(ensoniq, 1371_SMPRATE), r);
+ return 0;
+}
+
+static unsigned int snd_es1371_src_read(struct ensoniq * ensoniq, unsigned short reg)
+{
+ unsigned int temp, i, orig, r;
+
+ /* wait for ready */
+ temp = orig = snd_es1371_wait_src_ready(ensoniq);
+
+ /* expose the SRC state bits */
+ r = temp & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 |
+ ES_1371_DIS_P2 | ES_1371_DIS_R1);
+ r |= ES_1371_SRC_RAM_ADDRO(reg) | 0x10000;
+ outl(r, ES_REG(ensoniq, 1371_SMPRATE));
+
+ /* now, wait for busy and the correct time to read */
+ temp = snd_es1371_wait_src_ready(ensoniq);
+
+ if ((temp & 0x00870000) != 0x00010000) {
+ /* wait for the right state */
+ for (i = 0; i < POLL_COUNT; i++) {
+ temp = inl(ES_REG(ensoniq, 1371_SMPRATE));
+ if ((temp & 0x00870000) == 0x00010000)
+ break;
+ }
+ }
+
+ /* hide the state bits */
+ r = orig & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 |
+ ES_1371_DIS_P2 | ES_1371_DIS_R1);
+ r |= ES_1371_SRC_RAM_ADDRO(reg);
+ outl(r, ES_REG(ensoniq, 1371_SMPRATE));
+
+ return temp;
+}
+
+static void snd_es1371_src_write(struct ensoniq * ensoniq,
+ unsigned short reg, unsigned short data)
+{
+ unsigned int r;
+
+ r = snd_es1371_wait_src_ready(ensoniq) &
+ (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 |
+ ES_1371_DIS_P2 | ES_1371_DIS_R1);
+ r |= ES_1371_SRC_RAM_ADDRO(reg) | ES_1371_SRC_RAM_DATAO(data);
+ outl(r | ES_1371_SRC_RAM_WE, ES_REG(ensoniq, 1371_SMPRATE));
+}
+
+#endif /* CHIP1371 */
+
+#ifdef CHIP1370
+
+static void snd_es1370_codec_write(struct snd_ak4531 *ak4531,
+ unsigned short reg, unsigned short val)
+{
+ struct ensoniq *ensoniq = ak4531->private_data;
+ unsigned long end_time = jiffies + HZ / 10;
+
+#if 0
+ printk("CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n",
+ reg, val, ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC));
+#endif
+ do {
+ if (!(inl(ES_REG(ensoniq, STATUS)) & ES_1370_CSTAT)) {
+ outw(ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC));
+ return;
+ }
+ schedule_timeout_uninterruptible(1);
+ } while (time_after(end_time, jiffies));
+ snd_printk(KERN_ERR "codec write timeout, status = 0x%x\n",
+ inl(ES_REG(ensoniq, STATUS)));
+}
+
+#endif /* CHIP1370 */
+
+#ifdef CHIP1371
+
+static void snd_es1371_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ struct ensoniq *ensoniq = ac97->private_data;
+ unsigned int t, x;
+
+ mutex_lock(&ensoniq->src_mutex);
+ for (t = 0; t < POLL_COUNT; t++) {
+ if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) {
+ /* save the current state for latter */
+ x = snd_es1371_wait_src_ready(ensoniq);
+ outl((x & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 |
+ ES_1371_DIS_P2 | ES_1371_DIS_R1)) | 0x00010000,
+ ES_REG(ensoniq, 1371_SMPRATE));
+ /* wait for not busy (state 0) first to avoid
+ transition states */
+ for (t = 0; t < POLL_COUNT; t++) {
+ if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) ==
+ 0x00000000)
+ break;
+ }
+ /* wait for a SAFE time to write addr/data and then do it, dammit */
+ for (t = 0; t < POLL_COUNT; t++) {
+ if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) ==
+ 0x00010000)
+ break;
+ }
+ outl(ES_1371_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1371_CODEC));
+ /* restore SRC reg */
+ snd_es1371_wait_src_ready(ensoniq);
+ outl(x, ES_REG(ensoniq, 1371_SMPRATE));
+ mutex_unlock(&ensoniq->src_mutex);
+ return;
+ }
+ }
+ mutex_unlock(&ensoniq->src_mutex);
+ snd_printk(KERN_ERR "codec write timeout at 0x%lx [0x%x]\n",
+ ES_REG(ensoniq, 1371_CODEC), inl(ES_REG(ensoniq, 1371_CODEC)));
+}
+
+static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct ensoniq *ensoniq = ac97->private_data;
+ unsigned int t, x, fail = 0;
+
+ __again:
+ mutex_lock(&ensoniq->src_mutex);
+ for (t = 0; t < POLL_COUNT; t++) {
+ if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) {
+ /* save the current state for latter */
+ x = snd_es1371_wait_src_ready(ensoniq);
+ outl((x & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 |
+ ES_1371_DIS_P2 | ES_1371_DIS_R1)) | 0x00010000,
+ ES_REG(ensoniq, 1371_SMPRATE));
+ /* wait for not busy (state 0) first to avoid
+ transition states */
+ for (t = 0; t < POLL_COUNT; t++) {
+ if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) ==
+ 0x00000000)
+ break;
+ }
+ /* wait for a SAFE time to write addr/data and then do it, dammit */
+ for (t = 0; t < POLL_COUNT; t++) {
+ if ((inl(ES_REG(ensoniq, 1371_SMPRATE)) & 0x00870000) ==
+ 0x00010000)
+ break;
+ }
+ outl(ES_1371_CODEC_READS(reg), ES_REG(ensoniq, 1371_CODEC));
+ /* restore SRC reg */
+ snd_es1371_wait_src_ready(ensoniq);
+ outl(x, ES_REG(ensoniq, 1371_SMPRATE));
+ /* wait for WIP again */
+ for (t = 0; t < POLL_COUNT; t++) {
+ if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP))
+ break;
+ }
+ /* now wait for the stinkin' data (RDY) */
+ for (t = 0; t < POLL_COUNT; t++) {
+ if ((x = inl(ES_REG(ensoniq, 1371_CODEC))) & ES_1371_CODEC_RDY) {
+ mutex_unlock(&ensoniq->src_mutex);
+ return ES_1371_CODEC_READ(x);
+ }
+ }
+ mutex_unlock(&ensoniq->src_mutex);
+ if (++fail > 10) {
+ snd_printk(KERN_ERR "codec read timeout (final) "
+ "at 0x%lx, reg = 0x%x [0x%x]\n",
+ ES_REG(ensoniq, 1371_CODEC), reg,
+ inl(ES_REG(ensoniq, 1371_CODEC)));
+ return 0;
+ }
+ goto __again;
+ }
+ }
+ mutex_unlock(&ensoniq->src_mutex);
+ snd_printk(KERN_ERR "es1371: codec read timeout at 0x%lx [0x%x]\n",
+ ES_REG(ensoniq, 1371_CODEC), inl(ES_REG(ensoniq, 1371_CODEC)));
+ return 0;
+}
+
+static void snd_es1371_codec_wait(struct snd_ac97 *ac97)
+{
+ msleep(750);
+ snd_es1371_codec_read(ac97, AC97_RESET);
+ snd_es1371_codec_read(ac97, AC97_VENDOR_ID1);
+ snd_es1371_codec_read(ac97, AC97_VENDOR_ID2);
+ msleep(50);
+}
+
+static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate)
+{
+ unsigned int n, truncm, freq, result;
+
+ mutex_lock(&ensoniq->src_mutex);
+ n = rate / 3000;
+ if ((1 << n) & ((1 << 15) | (1 << 13) | (1 << 11) | (1 << 9)))
+ n--;
+ truncm = (21 * n - 1) | 1;
+ freq = ((48000UL << 15) / rate) * n;
+ result = (48000UL << 15) / (freq / n);
+ if (rate >= 24000) {
+ if (truncm > 239)
+ truncm = 239;
+ snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_TRUNC_N,
+ (((239 - truncm) >> 1) << 9) | (n << 4));
+ } else {
+ if (truncm > 119)
+ truncm = 119;
+ snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_TRUNC_N,
+ 0x8000 | (((119 - truncm) >> 1) << 9) | (n << 4));
+ }
+ snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_INT_REGS,
+ (snd_es1371_src_read(ensoniq, ES_SMPREG_ADC +
+ ES_SMPREG_INT_REGS) & 0x00ff) |
+ ((freq >> 5) & 0xfc00));
+ snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_VFREQ_FRAC, freq & 0x7fff);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC, n << 8);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC + 1, n << 8);
+ mutex_unlock(&ensoniq->src_mutex);
+}
+
+static void snd_es1371_dac1_rate(struct ensoniq * ensoniq, unsigned int rate)
+{
+ unsigned int freq, r;
+
+ mutex_lock(&ensoniq->src_mutex);
+ freq = ((rate << 15) + 1500) / 3000;
+ r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE |
+ ES_1371_DIS_P2 | ES_1371_DIS_R1)) |
+ ES_1371_DIS_P1;
+ outl(r, ES_REG(ensoniq, 1371_SMPRATE));
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC1 + ES_SMPREG_INT_REGS,
+ (snd_es1371_src_read(ensoniq, ES_SMPREG_DAC1 +
+ ES_SMPREG_INT_REGS) & 0x00ff) |
+ ((freq >> 5) & 0xfc00));
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC1 + ES_SMPREG_VFREQ_FRAC, freq & 0x7fff);
+ r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE |
+ ES_1371_DIS_P2 | ES_1371_DIS_R1));
+ outl(r, ES_REG(ensoniq, 1371_SMPRATE));
+ mutex_unlock(&ensoniq->src_mutex);
+}
+
+static void snd_es1371_dac2_rate(struct ensoniq * ensoniq, unsigned int rate)
+{
+ unsigned int freq, r;
+
+ mutex_lock(&ensoniq->src_mutex);
+ freq = ((rate << 15) + 1500) / 3000;
+ r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE |
+ ES_1371_DIS_P1 | ES_1371_DIS_R1)) |
+ ES_1371_DIS_P2;
+ outl(r, ES_REG(ensoniq, 1371_SMPRATE));
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC2 + ES_SMPREG_INT_REGS,
+ (snd_es1371_src_read(ensoniq, ES_SMPREG_DAC2 +
+ ES_SMPREG_INT_REGS) & 0x00ff) |
+ ((freq >> 5) & 0xfc00));
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC2 + ES_SMPREG_VFREQ_FRAC,
+ freq & 0x7fff);
+ r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE |
+ ES_1371_DIS_P1 | ES_1371_DIS_R1));
+ outl(r, ES_REG(ensoniq, 1371_SMPRATE));
+ mutex_unlock(&ensoniq->src_mutex);
+}
+
+#endif /* CHIP1371 */
+
+static int snd_ensoniq_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ {
+ unsigned int what = 0;
+ struct snd_pcm_substream *s;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == ensoniq->playback1_substream) {
+ what |= ES_P1_PAUSE;
+ snd_pcm_trigger_done(s, substream);
+ } else if (s == ensoniq->playback2_substream) {
+ what |= ES_P2_PAUSE;
+ snd_pcm_trigger_done(s, substream);
+ } else if (s == ensoniq->capture_substream)
+ return -EINVAL;
+ }
+ spin_lock(&ensoniq->reg_lock);
+ if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH)
+ ensoniq->sctrl |= what;
+ else
+ ensoniq->sctrl &= ~what;
+ outl(ensoniq->sctrl, ES_REG(ensoniq, SERIAL));
+ spin_unlock(&ensoniq->reg_lock);
+ break;
+ }
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_STOP:
+ {
+ unsigned int what = 0;
+ struct snd_pcm_substream *s;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == ensoniq->playback1_substream) {
+ what |= ES_DAC1_EN;
+ snd_pcm_trigger_done(s, substream);
+ } else if (s == ensoniq->playback2_substream) {
+ what |= ES_DAC2_EN;
+ snd_pcm_trigger_done(s, substream);
+ } else if (s == ensoniq->capture_substream) {
+ what |= ES_ADC_EN;
+ snd_pcm_trigger_done(s, substream);
+ }
+ }
+ spin_lock(&ensoniq->reg_lock);
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ ensoniq->ctrl |= what;
+ else
+ ensoniq->ctrl &= ~what;
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ spin_unlock(&ensoniq->reg_lock);
+ break;
+ }
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * PCM part
+ */
+
+static int snd_ensoniq_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_ensoniq_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_ensoniq_playback1_prepare(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int mode = 0;
+
+ ensoniq->p1_dma_size = snd_pcm_lib_buffer_bytes(substream);
+ ensoniq->p1_period_size = snd_pcm_lib_period_bytes(substream);
+ if (snd_pcm_format_width(runtime->format) == 16)
+ mode |= 0x02;
+ if (runtime->channels > 1)
+ mode |= 0x01;
+ spin_lock_irq(&ensoniq->reg_lock);
+ ensoniq->ctrl &= ~ES_DAC1_EN;
+#ifdef CHIP1371
+ /* 48k doesn't need SRC (it breaks AC3-passthru) */
+ if (runtime->rate == 48000)
+ ensoniq->ctrl |= ES_1373_BYPASS_P1;
+ else
+ ensoniq->ctrl &= ~ES_1373_BYPASS_P1;
+#endif
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ outl(ES_MEM_PAGEO(ES_PAGE_DAC), ES_REG(ensoniq, MEM_PAGE));
+ outl(runtime->dma_addr, ES_REG(ensoniq, DAC1_FRAME));
+ outl((ensoniq->p1_dma_size >> 2) - 1, ES_REG(ensoniq, DAC1_SIZE));
+ ensoniq->sctrl &= ~(ES_P1_LOOP_SEL | ES_P1_PAUSE | ES_P1_SCT_RLD | ES_P1_MODEM);
+ ensoniq->sctrl |= ES_P1_INT_EN | ES_P1_MODEO(mode);
+ outl(ensoniq->sctrl, ES_REG(ensoniq, SERIAL));
+ outl((ensoniq->p1_period_size >> snd_ensoniq_sample_shift[mode]) - 1,
+ ES_REG(ensoniq, DAC1_COUNT));
+#ifdef CHIP1370
+ ensoniq->ctrl &= ~ES_1370_WTSRSELM;
+ switch (runtime->rate) {
+ case 5512: ensoniq->ctrl |= ES_1370_WTSRSEL(0); break;
+ case 11025: ensoniq->ctrl |= ES_1370_WTSRSEL(1); break;
+ case 22050: ensoniq->ctrl |= ES_1370_WTSRSEL(2); break;
+ case 44100: ensoniq->ctrl |= ES_1370_WTSRSEL(3); break;
+ default: snd_BUG();
+ }
+#endif
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ spin_unlock_irq(&ensoniq->reg_lock);
+#ifndef CHIP1370
+ snd_es1371_dac1_rate(ensoniq, runtime->rate);
+#endif
+ return 0;
+}
+
+static int snd_ensoniq_playback2_prepare(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int mode = 0;
+
+ ensoniq->p2_dma_size = snd_pcm_lib_buffer_bytes(substream);
+ ensoniq->p2_period_size = snd_pcm_lib_period_bytes(substream);
+ if (snd_pcm_format_width(runtime->format) == 16)
+ mode |= 0x02;
+ if (runtime->channels > 1)
+ mode |= 0x01;
+ spin_lock_irq(&ensoniq->reg_lock);
+ ensoniq->ctrl &= ~ES_DAC2_EN;
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ outl(ES_MEM_PAGEO(ES_PAGE_DAC), ES_REG(ensoniq, MEM_PAGE));
+ outl(runtime->dma_addr, ES_REG(ensoniq, DAC2_FRAME));
+ outl((ensoniq->p2_dma_size >> 2) - 1, ES_REG(ensoniq, DAC2_SIZE));
+ ensoniq->sctrl &= ~(ES_P2_LOOP_SEL | ES_P2_PAUSE | ES_P2_DAC_SEN |
+ ES_P2_END_INCM | ES_P2_ST_INCM | ES_P2_MODEM);
+ ensoniq->sctrl |= ES_P2_INT_EN | ES_P2_MODEO(mode) |
+ ES_P2_END_INCO(mode & 2 ? 2 : 1) | ES_P2_ST_INCO(0);
+ outl(ensoniq->sctrl, ES_REG(ensoniq, SERIAL));
+ outl((ensoniq->p2_period_size >> snd_ensoniq_sample_shift[mode]) - 1,
+ ES_REG(ensoniq, DAC2_COUNT));
+#ifdef CHIP1370
+ if (!(ensoniq->u.es1370.pclkdiv_lock & ES_MODE_CAPTURE)) {
+ ensoniq->ctrl &= ~ES_1370_PCLKDIVM;
+ ensoniq->ctrl |= ES_1370_PCLKDIVO(ES_1370_SRTODIV(runtime->rate));
+ ensoniq->u.es1370.pclkdiv_lock |= ES_MODE_PLAY2;
+ }
+#endif
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ spin_unlock_irq(&ensoniq->reg_lock);
+#ifndef CHIP1370
+ snd_es1371_dac2_rate(ensoniq, runtime->rate);
+#endif
+ return 0;
+}
+
+static int snd_ensoniq_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int mode = 0;
+
+ ensoniq->c_dma_size = snd_pcm_lib_buffer_bytes(substream);
+ ensoniq->c_period_size = snd_pcm_lib_period_bytes(substream);
+ if (snd_pcm_format_width(runtime->format) == 16)
+ mode |= 0x02;
+ if (runtime->channels > 1)
+ mode |= 0x01;
+ spin_lock_irq(&ensoniq->reg_lock);
+ ensoniq->ctrl &= ~ES_ADC_EN;
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ outl(ES_MEM_PAGEO(ES_PAGE_ADC), ES_REG(ensoniq, MEM_PAGE));
+ outl(runtime->dma_addr, ES_REG(ensoniq, ADC_FRAME));
+ outl((ensoniq->c_dma_size >> 2) - 1, ES_REG(ensoniq, ADC_SIZE));
+ ensoniq->sctrl &= ~(ES_R1_LOOP_SEL | ES_R1_MODEM);
+ ensoniq->sctrl |= ES_R1_INT_EN | ES_R1_MODEO(mode);
+ outl(ensoniq->sctrl, ES_REG(ensoniq, SERIAL));
+ outl((ensoniq->c_period_size >> snd_ensoniq_sample_shift[mode]) - 1,
+ ES_REG(ensoniq, ADC_COUNT));
+#ifdef CHIP1370
+ if (!(ensoniq->u.es1370.pclkdiv_lock & ES_MODE_PLAY2)) {
+ ensoniq->ctrl &= ~ES_1370_PCLKDIVM;
+ ensoniq->ctrl |= ES_1370_PCLKDIVO(ES_1370_SRTODIV(runtime->rate));
+ ensoniq->u.es1370.pclkdiv_lock |= ES_MODE_CAPTURE;
+ }
+#endif
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ spin_unlock_irq(&ensoniq->reg_lock);
+#ifndef CHIP1370
+ snd_es1371_adc_rate(ensoniq, runtime->rate);
+#endif
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_ensoniq_playback1_pointer(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ spin_lock(&ensoniq->reg_lock);
+ if (inl(ES_REG(ensoniq, CONTROL)) & ES_DAC1_EN) {
+ outl(ES_MEM_PAGEO(ES_PAGE_DAC), ES_REG(ensoniq, MEM_PAGE));
+ ptr = ES_REG_FCURR_COUNTI(inl(ES_REG(ensoniq, DAC1_SIZE)));
+ ptr = bytes_to_frames(substream->runtime, ptr);
+ } else {
+ ptr = 0;
+ }
+ spin_unlock(&ensoniq->reg_lock);
+ return ptr;
+}
+
+static snd_pcm_uframes_t snd_ensoniq_playback2_pointer(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ spin_lock(&ensoniq->reg_lock);
+ if (inl(ES_REG(ensoniq, CONTROL)) & ES_DAC2_EN) {
+ outl(ES_MEM_PAGEO(ES_PAGE_DAC), ES_REG(ensoniq, MEM_PAGE));
+ ptr = ES_REG_FCURR_COUNTI(inl(ES_REG(ensoniq, DAC2_SIZE)));
+ ptr = bytes_to_frames(substream->runtime, ptr);
+ } else {
+ ptr = 0;
+ }
+ spin_unlock(&ensoniq->reg_lock);
+ return ptr;
+}
+
+static snd_pcm_uframes_t snd_ensoniq_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ spin_lock(&ensoniq->reg_lock);
+ if (inl(ES_REG(ensoniq, CONTROL)) & ES_ADC_EN) {
+ outl(ES_MEM_PAGEO(ES_PAGE_ADC), ES_REG(ensoniq, MEM_PAGE));
+ ptr = ES_REG_FCURR_COUNTI(inl(ES_REG(ensoniq, ADC_SIZE)));
+ ptr = bytes_to_frames(substream->runtime, ptr);
+ } else {
+ ptr = 0;
+ }
+ spin_unlock(&ensoniq->reg_lock);
+ return ptr;
+}
+
+static struct snd_pcm_hardware snd_ensoniq_playback1 =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates =
+#ifndef CHIP1370
+ SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+#else
+ (SNDRV_PCM_RATE_KNOT | /* 5512Hz rate */
+ SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_22050 |
+ SNDRV_PCM_RATE_44100),
+#endif
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_ensoniq_playback2 =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_ensoniq_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_ensoniq_playback1_open(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ ensoniq->mode |= ES_MODE_PLAY1;
+ ensoniq->playback1_substream = substream;
+ runtime->hw = snd_ensoniq_playback1;
+ snd_pcm_set_sync(substream);
+ spin_lock_irq(&ensoniq->reg_lock);
+ if (ensoniq->spdif && ensoniq->playback2_substream == NULL)
+ ensoniq->spdif_stream = ensoniq->spdif_default;
+ spin_unlock_irq(&ensoniq->reg_lock);
+#ifdef CHIP1370
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_es1370_hw_constraints_rates);
+#else
+ snd_pcm_hw_constraint_ratdens(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_es1371_hw_constraints_dac_clock);
+#endif
+ return 0;
+}
+
+static int snd_ensoniq_playback2_open(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ ensoniq->mode |= ES_MODE_PLAY2;
+ ensoniq->playback2_substream = substream;
+ runtime->hw = snd_ensoniq_playback2;
+ snd_pcm_set_sync(substream);
+ spin_lock_irq(&ensoniq->reg_lock);
+ if (ensoniq->spdif && ensoniq->playback1_substream == NULL)
+ ensoniq->spdif_stream = ensoniq->spdif_default;
+ spin_unlock_irq(&ensoniq->reg_lock);
+#ifdef CHIP1370
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_es1370_hw_constraints_clock);
+#else
+ snd_pcm_hw_constraint_ratdens(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_es1371_hw_constraints_dac_clock);
+#endif
+ return 0;
+}
+
+static int snd_ensoniq_capture_open(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ ensoniq->mode |= ES_MODE_CAPTURE;
+ ensoniq->capture_substream = substream;
+ runtime->hw = snd_ensoniq_capture;
+ snd_pcm_set_sync(substream);
+#ifdef CHIP1370
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_es1370_hw_constraints_clock);
+#else
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_es1371_hw_constraints_adc_clock);
+#endif
+ return 0;
+}
+
+static int snd_ensoniq_playback1_close(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+
+ ensoniq->playback1_substream = NULL;
+ ensoniq->mode &= ~ES_MODE_PLAY1;
+ return 0;
+}
+
+static int snd_ensoniq_playback2_close(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+
+ ensoniq->playback2_substream = NULL;
+ spin_lock_irq(&ensoniq->reg_lock);
+#ifdef CHIP1370
+ ensoniq->u.es1370.pclkdiv_lock &= ~ES_MODE_PLAY2;
+#endif
+ ensoniq->mode &= ~ES_MODE_PLAY2;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_ensoniq_capture_close(struct snd_pcm_substream *substream)
+{
+ struct ensoniq *ensoniq = snd_pcm_substream_chip(substream);
+
+ ensoniq->capture_substream = NULL;
+ spin_lock_irq(&ensoniq->reg_lock);
+#ifdef CHIP1370
+ ensoniq->u.es1370.pclkdiv_lock &= ~ES_MODE_CAPTURE;
+#endif
+ ensoniq->mode &= ~ES_MODE_CAPTURE;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ensoniq_playback1_ops = {
+ .open = snd_ensoniq_playback1_open,
+ .close = snd_ensoniq_playback1_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ensoniq_hw_params,
+ .hw_free = snd_ensoniq_hw_free,
+ .prepare = snd_ensoniq_playback1_prepare,
+ .trigger = snd_ensoniq_trigger,
+ .pointer = snd_ensoniq_playback1_pointer,
+};
+
+static struct snd_pcm_ops snd_ensoniq_playback2_ops = {
+ .open = snd_ensoniq_playback2_open,
+ .close = snd_ensoniq_playback2_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ensoniq_hw_params,
+ .hw_free = snd_ensoniq_hw_free,
+ .prepare = snd_ensoniq_playback2_prepare,
+ .trigger = snd_ensoniq_trigger,
+ .pointer = snd_ensoniq_playback2_pointer,
+};
+
+static struct snd_pcm_ops snd_ensoniq_capture_ops = {
+ .open = snd_ensoniq_capture_open,
+ .close = snd_ensoniq_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ensoniq_hw_params,
+ .hw_free = snd_ensoniq_hw_free,
+ .prepare = snd_ensoniq_capture_prepare,
+ .trigger = snd_ensoniq_trigger,
+ .pointer = snd_ensoniq_capture_pointer,
+};
+
+static int __devinit snd_ensoniq_pcm(struct ensoniq * ensoniq, int device,
+ struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+#ifdef CHIP1370
+ err = snd_pcm_new(ensoniq->card, "ES1370/1", device, 1, 1, &pcm);
+#else
+ err = snd_pcm_new(ensoniq->card, "ES1371/1", device, 1, 1, &pcm);
+#endif
+ if (err < 0)
+ return err;
+
+#ifdef CHIP1370
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ensoniq_playback2_ops);
+#else
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ensoniq_playback1_ops);
+#endif
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ensoniq_capture_ops);
+
+ pcm->private_data = ensoniq;
+ pcm->info_flags = 0;
+#ifdef CHIP1370
+ strcpy(pcm->name, "ES1370 DAC2/ADC");
+#else
+ strcpy(pcm->name, "ES1371 DAC2/ADC");
+#endif
+ ensoniq->pcm1 = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ensoniq->pci), 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static int __devinit snd_ensoniq_pcm2(struct ensoniq * ensoniq, int device,
+ struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+#ifdef CHIP1370
+ err = snd_pcm_new(ensoniq->card, "ES1370/2", device, 1, 0, &pcm);
+#else
+ err = snd_pcm_new(ensoniq->card, "ES1371/2", device, 1, 0, &pcm);
+#endif
+ if (err < 0)
+ return err;
+
+#ifdef CHIP1370
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ensoniq_playback1_ops);
+#else
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ensoniq_playback2_ops);
+#endif
+ pcm->private_data = ensoniq;
+ pcm->info_flags = 0;
+#ifdef CHIP1370
+ strcpy(pcm->name, "ES1370 DAC1");
+#else
+ strcpy(pcm->name, "ES1371 DAC1");
+#endif
+ ensoniq->pcm2 = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ensoniq->pci), 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/*
+ * Mixer section
+ */
+
+/*
+ * ENS1371 mixer (including SPDIF interface)
+ */
+#ifdef CHIP1371
+static int snd_ens1373_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ens1373_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ spin_lock_irq(&ensoniq->reg_lock);
+ ucontrol->value.iec958.status[0] = (ensoniq->spdif_default >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (ensoniq->spdif_default >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (ensoniq->spdif_default >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (ensoniq->spdif_default >> 24) & 0xff;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_ens1373_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ((u32)ucontrol->value.iec958.status[0] << 0) |
+ ((u32)ucontrol->value.iec958.status[1] << 8) |
+ ((u32)ucontrol->value.iec958.status[2] << 16) |
+ ((u32)ucontrol->value.iec958.status[3] << 24);
+ spin_lock_irq(&ensoniq->reg_lock);
+ change = ensoniq->spdif_default != val;
+ ensoniq->spdif_default = val;
+ if (change && ensoniq->playback1_substream == NULL &&
+ ensoniq->playback2_substream == NULL)
+ outl(val, ES_REG(ensoniq, CHANNEL_STATUS));
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return change;
+}
+
+static int snd_ens1373_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static int snd_ens1373_spdif_stream_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ spin_lock_irq(&ensoniq->reg_lock);
+ ucontrol->value.iec958.status[0] = (ensoniq->spdif_stream >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (ensoniq->spdif_stream >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (ensoniq->spdif_stream >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (ensoniq->spdif_stream >> 24) & 0xff;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_ens1373_spdif_stream_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ((u32)ucontrol->value.iec958.status[0] << 0) |
+ ((u32)ucontrol->value.iec958.status[1] << 8) |
+ ((u32)ucontrol->value.iec958.status[2] << 16) |
+ ((u32)ucontrol->value.iec958.status[3] << 24);
+ spin_lock_irq(&ensoniq->reg_lock);
+ change = ensoniq->spdif_stream != val;
+ ensoniq->spdif_stream = val;
+ if (change && (ensoniq->playback1_substream != NULL ||
+ ensoniq->playback2_substream != NULL))
+ outl(val, ES_REG(ensoniq, CHANNEL_STATUS));
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return change;
+}
+
+#define ES1371_SPDIF(xname) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_es1371_spdif_info, \
+ .get = snd_es1371_spdif_get, .put = snd_es1371_spdif_put }
+
+#define snd_es1371_spdif_info snd_ctl_boolean_mono_info
+
+static int snd_es1371_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ ucontrol->value.integer.value[0] = ensoniq->ctrl & ES_1373_SPDIF_THRU ? 1 : 0;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_es1371_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ unsigned int nval1, nval2;
+ int change;
+
+ nval1 = ucontrol->value.integer.value[0] ? ES_1373_SPDIF_THRU : 0;
+ nval2 = ucontrol->value.integer.value[0] ? ES_1373_SPDIF_EN : 0;
+ spin_lock_irq(&ensoniq->reg_lock);
+ change = (ensoniq->ctrl & ES_1373_SPDIF_THRU) != nval1;
+ ensoniq->ctrl &= ~ES_1373_SPDIF_THRU;
+ ensoniq->ctrl |= nval1;
+ ensoniq->cssr &= ~ES_1373_SPDIF_EN;
+ ensoniq->cssr |= nval2;
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ outl(ensoniq->cssr, ES_REG(ensoniq, STATUS));
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return change;
+}
+
+
+/* spdif controls */
+static struct snd_kcontrol_new snd_es1371_mixer_spdif[] __devinitdata = {
+ ES1371_SPDIF(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH)),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_ens1373_spdif_info,
+ .get = snd_ens1373_spdif_default_get,
+ .put = snd_ens1373_spdif_default_put,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .info = snd_ens1373_spdif_info,
+ .get = snd_ens1373_spdif_mask_get
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_ens1373_spdif_info,
+ .get = snd_ens1373_spdif_stream_get,
+ .put = snd_ens1373_spdif_stream_put
+ },
+};
+
+
+#define snd_es1373_rear_info snd_ctl_boolean_mono_info
+
+static int snd_es1373_rear_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ int val = 0;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ if ((ensoniq->cssr & (ES_1373_REAR_BIT27|ES_1373_REAR_BIT26|
+ ES_1373_REAR_BIT24)) == ES_1373_REAR_BIT26)
+ val = 1;
+ ucontrol->value.integer.value[0] = val;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_es1373_rear_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ unsigned int nval1;
+ int change;
+
+ nval1 = ucontrol->value.integer.value[0] ?
+ ES_1373_REAR_BIT26 : (ES_1373_REAR_BIT27|ES_1373_REAR_BIT24);
+ spin_lock_irq(&ensoniq->reg_lock);
+ change = (ensoniq->cssr & (ES_1373_REAR_BIT27|
+ ES_1373_REAR_BIT26|ES_1373_REAR_BIT24)) != nval1;
+ ensoniq->cssr &= ~(ES_1373_REAR_BIT27|ES_1373_REAR_BIT26|ES_1373_REAR_BIT24);
+ ensoniq->cssr |= nval1;
+ outl(ensoniq->cssr, ES_REG(ensoniq, STATUS));
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ens1373_rear __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "AC97 2ch->4ch Copy Switch",
+ .info = snd_es1373_rear_info,
+ .get = snd_es1373_rear_get,
+ .put = snd_es1373_rear_put,
+};
+
+#define snd_es1373_line_info snd_ctl_boolean_mono_info
+
+static int snd_es1373_line_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ int val = 0;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ if ((ensoniq->ctrl & ES_1371_GPIO_OUTM) >= 4)
+ val = 1;
+ ucontrol->value.integer.value[0] = val;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_es1373_line_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ int changed;
+ unsigned int ctrl;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ ctrl = ensoniq->ctrl;
+ if (ucontrol->value.integer.value[0])
+ ensoniq->ctrl |= ES_1371_GPIO_OUT(4); /* switch line-in -> rear out */
+ else
+ ensoniq->ctrl &= ~ES_1371_GPIO_OUT(4);
+ changed = (ctrl != ensoniq->ctrl);
+ if (changed)
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_ens1373_line __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line In->Rear Out Switch",
+ .info = snd_es1373_line_info,
+ .get = snd_es1373_line_get,
+ .put = snd_es1373_line_put,
+};
+
+static void snd_ensoniq_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct ensoniq *ensoniq = ac97->private_data;
+ ensoniq->u.es1371.ac97 = NULL;
+}
+
+struct es1371_quirk {
+ unsigned short vid; /* vendor ID */
+ unsigned short did; /* device ID */
+ unsigned char rev; /* revision */
+};
+
+static int es1371_quirk_lookup(struct ensoniq *ensoniq,
+ struct es1371_quirk *list)
+{
+ while (list->vid != (unsigned short)PCI_ANY_ID) {
+ if (ensoniq->pci->vendor == list->vid &&
+ ensoniq->pci->device == list->did &&
+ ensoniq->rev == list->rev)
+ return 1;
+ list++;
+ }
+ return 0;
+}
+
+static struct es1371_quirk es1371_spdif_present[] __devinitdata = {
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_CT5880, .rev = CT5880REV_CT5880_C },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_CT5880, .rev = CT5880REV_CT5880_D },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_CT5880, .rev = CT5880REV_CT5880_E },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_ES1371, .rev = ES1371REV_CT5880_A },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_ES1371, .rev = ES1371REV_ES1373_8 },
+ { .vid = PCI_ANY_ID, .did = PCI_ANY_ID }
+};
+
+static struct snd_pci_quirk ens1373_line_quirk[] __devinitdata = {
+ SND_PCI_QUIRK_ID(0x1274, 0x2000), /* GA-7DXR */
+ SND_PCI_QUIRK_ID(0x1458, 0xa000), /* GA-8IEXP */
+ { } /* end */
+};
+
+static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq,
+ int has_spdif, int has_line)
+{
+ struct snd_card *card = ensoniq->card;
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_es1371_codec_write,
+ .read = snd_es1371_codec_read,
+ .wait = snd_es1371_codec_wait,
+ };
+
+ if ((err = snd_ac97_bus(card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = ensoniq;
+ ac97.private_free = snd_ensoniq_mixer_free_ac97;
+ ac97.pci = ensoniq->pci;
+ ac97.scaps = AC97_SCAP_AUDIO;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &ensoniq->u.es1371.ac97)) < 0)
+ return err;
+ if (has_spdif > 0 ||
+ (!has_spdif && es1371_quirk_lookup(ensoniq, es1371_spdif_present))) {
+ struct snd_kcontrol *kctl;
+ int i, is_spdif = 0;
+
+ ensoniq->spdif_default = ensoniq->spdif_stream =
+ SNDRV_PCM_DEFAULT_CON_SPDIF;
+ outl(ensoniq->spdif_default, ES_REG(ensoniq, CHANNEL_STATUS));
+
+ if (ensoniq->u.es1371.ac97->ext_id & AC97_EI_SPDIF)
+ is_spdif++;
+
+ for (i = 0; i < ARRAY_SIZE(snd_es1371_mixer_spdif); i++) {
+ kctl = snd_ctl_new1(&snd_es1371_mixer_spdif[i], ensoniq);
+ if (!kctl)
+ return -ENOMEM;
+ kctl->id.index = is_spdif;
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
+ return err;
+ }
+ }
+ if (ensoniq->u.es1371.ac97->ext_id & AC97_EI_SDAC) {
+ /* mirror rear to front speakers */
+ ensoniq->cssr &= ~(ES_1373_REAR_BIT27|ES_1373_REAR_BIT24);
+ ensoniq->cssr |= ES_1373_REAR_BIT26;
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ens1373_rear, ensoniq));
+ if (err < 0)
+ return err;
+ }
+ if (has_line > 0 ||
+ snd_pci_quirk_lookup(ensoniq->pci, ens1373_line_quirk)) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ens1373_line,
+ ensoniq));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+#endif /* CHIP1371 */
+
+/* generic control callbacks for ens1370 */
+#ifdef CHIP1370
+#define ENSONIQ_CONTROL(xname, mask) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, .info = snd_ensoniq_control_info, \
+ .get = snd_ensoniq_control_get, .put = snd_ensoniq_control_put, \
+ .private_value = mask }
+
+#define snd_ensoniq_control_info snd_ctl_boolean_mono_info
+
+static int snd_ensoniq_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ int mask = kcontrol->private_value;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ ucontrol->value.integer.value[0] = ensoniq->ctrl & mask ? 1 : 0;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_ensoniq_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ensoniq *ensoniq = snd_kcontrol_chip(kcontrol);
+ int mask = kcontrol->private_value;
+ unsigned int nval;
+ int change;
+
+ nval = ucontrol->value.integer.value[0] ? mask : 0;
+ spin_lock_irq(&ensoniq->reg_lock);
+ change = (ensoniq->ctrl & mask) != nval;
+ ensoniq->ctrl &= ~mask;
+ ensoniq->ctrl |= nval;
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return change;
+}
+
+/*
+ * ENS1370 mixer
+ */
+
+static struct snd_kcontrol_new snd_es1370_controls[2] __devinitdata = {
+ENSONIQ_CONTROL("PCM 0 Output also on Line-In Jack", ES_1370_XCTL0),
+ENSONIQ_CONTROL("Mic +5V bias", ES_1370_XCTL1)
+};
+
+#define ES1370_CONTROLS ARRAY_SIZE(snd_es1370_controls)
+
+static void snd_ensoniq_mixer_free_ak4531(struct snd_ak4531 *ak4531)
+{
+ struct ensoniq *ensoniq = ak4531->private_data;
+ ensoniq->u.es1370.ak4531 = NULL;
+}
+
+static int __devinit snd_ensoniq_1370_mixer(struct ensoniq * ensoniq)
+{
+ struct snd_card *card = ensoniq->card;
+ struct snd_ak4531 ak4531;
+ unsigned int idx;
+ int err;
+
+ /* try reset AK4531 */
+ outw(ES_1370_CODEC_WRITE(AK4531_RESET, 0x02), ES_REG(ensoniq, 1370_CODEC));
+ inw(ES_REG(ensoniq, 1370_CODEC));
+ udelay(100);
+ outw(ES_1370_CODEC_WRITE(AK4531_RESET, 0x03), ES_REG(ensoniq, 1370_CODEC));
+ inw(ES_REG(ensoniq, 1370_CODEC));
+ udelay(100);
+
+ memset(&ak4531, 0, sizeof(ak4531));
+ ak4531.write = snd_es1370_codec_write;
+ ak4531.private_data = ensoniq;
+ ak4531.private_free = snd_ensoniq_mixer_free_ak4531;
+ if ((err = snd_ak4531_mixer(card, &ak4531, &ensoniq->u.es1370.ak4531)) < 0)
+ return err;
+ for (idx = 0; idx < ES1370_CONTROLS; idx++) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_es1370_controls[idx], ensoniq));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+#endif /* CHIP1370 */
+
+#ifdef SUPPORT_JOYSTICK
+
+#ifdef CHIP1371
+static int __devinit snd_ensoniq_get_joystick_port(int dev)
+{
+ switch (joystick_port[dev]) {
+ case 0: /* disabled */
+ case 1: /* auto-detect */
+ case 0x200:
+ case 0x208:
+ case 0x210:
+ case 0x218:
+ return joystick_port[dev];
+
+ default:
+ printk(KERN_ERR "ens1371: invalid joystick port %#x", joystick_port[dev]);
+ return 0;
+ }
+}
+#else
+static inline int snd_ensoniq_get_joystick_port(int dev)
+{
+ return joystick[dev] ? 0x200 : 0;
+}
+#endif
+
+static int __devinit snd_ensoniq_create_gameport(struct ensoniq *ensoniq, int dev)
+{
+ struct gameport *gp;
+ int io_port;
+
+ io_port = snd_ensoniq_get_joystick_port(dev);
+
+ switch (io_port) {
+ case 0:
+ return -ENOSYS;
+
+ case 1: /* auto_detect */
+ for (io_port = 0x200; io_port <= 0x218; io_port += 8)
+ if (request_region(io_port, 8, "ens137x: gameport"))
+ break;
+ if (io_port > 0x218) {
+ printk(KERN_WARNING "ens137x: no gameport ports available\n");
+ return -EBUSY;
+ }
+ break;
+
+ default:
+ if (!request_region(io_port, 8, "ens137x: gameport")) {
+ printk(KERN_WARNING "ens137x: gameport io port 0x%#x in use\n",
+ io_port);
+ return -EBUSY;
+ }
+ break;
+ }
+
+ ensoniq->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "ens137x: cannot allocate memory for gameport\n");
+ release_region(io_port, 8);
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "ES137x");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(ensoniq->pci));
+ gameport_set_dev_parent(gp, &ensoniq->pci->dev);
+ gp->io = io_port;
+
+ ensoniq->ctrl |= ES_JYSTK_EN;
+#ifdef CHIP1371
+ ensoniq->ctrl &= ~ES_1371_JOY_ASELM;
+ ensoniq->ctrl |= ES_1371_JOY_ASEL((io_port - 0x200) / 8);
+#endif
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+
+ gameport_register_port(ensoniq->gameport);
+
+ return 0;
+}
+
+static void snd_ensoniq_free_gameport(struct ensoniq *ensoniq)
+{
+ if (ensoniq->gameport) {
+ int port = ensoniq->gameport->io;
+
+ gameport_unregister_port(ensoniq->gameport);
+ ensoniq->gameport = NULL;
+ ensoniq->ctrl &= ~ES_JYSTK_EN;
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ release_region(port, 8);
+ }
+}
+#else
+static inline int snd_ensoniq_create_gameport(struct ensoniq *ensoniq, long port) { return -ENOSYS; }
+static inline void snd_ensoniq_free_gameport(struct ensoniq *ensoniq) { }
+#endif /* SUPPORT_JOYSTICK */
+
+/*
+
+ */
+
+static void snd_ensoniq_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct ensoniq *ensoniq = entry->private_data;
+
+#ifdef CHIP1370
+ snd_iprintf(buffer, "Ensoniq AudioPCI ES1370\n\n");
+#else
+ snd_iprintf(buffer, "Ensoniq AudioPCI ES1371\n\n");
+#endif
+ snd_iprintf(buffer, "Joystick enable : %s\n",
+ ensoniq->ctrl & ES_JYSTK_EN ? "on" : "off");
+#ifdef CHIP1370
+ snd_iprintf(buffer, "MIC +5V bias : %s\n",
+ ensoniq->ctrl & ES_1370_XCTL1 ? "on" : "off");
+ snd_iprintf(buffer, "Line In to AOUT : %s\n",
+ ensoniq->ctrl & ES_1370_XCTL0 ? "on" : "off");
+#else
+ snd_iprintf(buffer, "Joystick port : 0x%x\n",
+ (ES_1371_JOY_ASELI(ensoniq->ctrl) * 8) + 0x200);
+#endif
+}
+
+static void __devinit snd_ensoniq_proc_init(struct ensoniq * ensoniq)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(ensoniq->card, "audiopci", &entry))
+ snd_info_set_text_ops(entry, ensoniq, snd_ensoniq_proc_read);
+}
+
+/*
+
+ */
+
+static int snd_ensoniq_free(struct ensoniq *ensoniq)
+{
+ snd_ensoniq_free_gameport(ensoniq);
+ if (ensoniq->irq < 0)
+ goto __hw_end;
+#ifdef CHIP1370
+ outl(ES_1370_SERR_DISABLE, ES_REG(ensoniq, CONTROL)); /* switch everything off */
+ outl(0, ES_REG(ensoniq, SERIAL)); /* clear serial interface */
+#else
+ outl(0, ES_REG(ensoniq, CONTROL)); /* switch everything off */
+ outl(0, ES_REG(ensoniq, SERIAL)); /* clear serial interface */
+#endif
+ if (ensoniq->irq >= 0)
+ synchronize_irq(ensoniq->irq);
+ pci_set_power_state(ensoniq->pci, 3);
+ __hw_end:
+#ifdef CHIP1370
+ if (ensoniq->dma_bug.area)
+ snd_dma_free_pages(&ensoniq->dma_bug);
+#endif
+ if (ensoniq->irq >= 0)
+ free_irq(ensoniq->irq, ensoniq);
+ pci_release_regions(ensoniq->pci);
+ pci_disable_device(ensoniq->pci);
+ kfree(ensoniq);
+ return 0;
+}
+
+static int snd_ensoniq_dev_free(struct snd_device *device)
+{
+ struct ensoniq *ensoniq = device->device_data;
+ return snd_ensoniq_free(ensoniq);
+}
+
+#ifdef CHIP1371
+static struct snd_pci_quirk es1371_amplifier_hack[] __devinitdata = {
+ SND_PCI_QUIRK_ID(0x107b, 0x2150), /* Gateway Solo 2150 */
+ SND_PCI_QUIRK_ID(0x13bd, 0x100c), /* EV1938 on Mebius PC-MJ100V */
+ SND_PCI_QUIRK_ID(0x1102, 0x5938), /* Targa Xtender300 */
+ SND_PCI_QUIRK_ID(0x1102, 0x8938), /* IPC Topnote G notebook */
+ { } /* end */
+};
+
+static struct es1371_quirk es1371_ac97_reset_hack[] = {
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_CT5880, .rev = CT5880REV_CT5880_C },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_CT5880, .rev = CT5880REV_CT5880_D },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_CT5880, .rev = CT5880REV_CT5880_E },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_ES1371, .rev = ES1371REV_CT5880_A },
+ { .vid = PCI_VENDOR_ID_ENSONIQ, .did = PCI_DEVICE_ID_ENSONIQ_ES1371, .rev = ES1371REV_ES1373_8 },
+ { .vid = PCI_ANY_ID, .did = PCI_ANY_ID }
+};
+#endif
+
+static void snd_ensoniq_chip_init(struct ensoniq *ensoniq)
+{
+#ifdef CHIP1371
+ int idx;
+#endif
+ /* this code was part of snd_ensoniq_create before intruduction
+ * of suspend/resume
+ */
+#ifdef CHIP1370
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ outl(ensoniq->sctrl, ES_REG(ensoniq, SERIAL));
+ outl(ES_MEM_PAGEO(ES_PAGE_ADC), ES_REG(ensoniq, MEM_PAGE));
+ outl(ensoniq->dma_bug.addr, ES_REG(ensoniq, PHANTOM_FRAME));
+ outl(0, ES_REG(ensoniq, PHANTOM_COUNT));
+#else
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ outl(ensoniq->sctrl, ES_REG(ensoniq, SERIAL));
+ outl(0, ES_REG(ensoniq, 1371_LEGACY));
+ if (es1371_quirk_lookup(ensoniq, es1371_ac97_reset_hack)) {
+ outl(ensoniq->cssr, ES_REG(ensoniq, STATUS));
+ /* need to delay around 20ms(bleech) to give
+ some CODECs enough time to wakeup */
+ msleep(20);
+ }
+ /* AC'97 warm reset to start the bitclk */
+ outl(ensoniq->ctrl | ES_1371_SYNC_RES, ES_REG(ensoniq, CONTROL));
+ inl(ES_REG(ensoniq, CONTROL));
+ udelay(20);
+ outl(ensoniq->ctrl, ES_REG(ensoniq, CONTROL));
+ /* Init the sample rate converter */
+ snd_es1371_wait_src_ready(ensoniq);
+ outl(ES_1371_SRC_DISABLE, ES_REG(ensoniq, 1371_SMPRATE));
+ for (idx = 0; idx < 0x80; idx++)
+ snd_es1371_src_write(ensoniq, idx, 0);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC1 + ES_SMPREG_TRUNC_N, 16 << 4);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC1 + ES_SMPREG_INT_REGS, 16 << 10);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC2 + ES_SMPREG_TRUNC_N, 16 << 4);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_DAC2 + ES_SMPREG_INT_REGS, 16 << 10);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC, 1 << 12);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC + 1, 1 << 12);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_DAC1, 1 << 12);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_DAC1 + 1, 1 << 12);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_DAC2, 1 << 12);
+ snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_DAC2 + 1, 1 << 12);
+ snd_es1371_adc_rate(ensoniq, 22050);
+ snd_es1371_dac1_rate(ensoniq, 22050);
+ snd_es1371_dac2_rate(ensoniq, 22050);
+ /* WARNING:
+ * enabling the sample rate converter without properly programming
+ * its parameters causes the chip to lock up (the SRC busy bit will
+ * be stuck high, and I've found no way to rectify this other than
+ * power cycle) - Thomas Sailer
+ */
+ snd_es1371_wait_src_ready(ensoniq);
+ outl(0, ES_REG(ensoniq, 1371_SMPRATE));
+ /* try reset codec directly */
+ outl(ES_1371_CODEC_WRITE(0, 0), ES_REG(ensoniq, 1371_CODEC));
+#endif
+ outb(ensoniq->uartc = 0x00, ES_REG(ensoniq, UART_CONTROL));
+ outb(0x00, ES_REG(ensoniq, UART_RES));
+ outl(ensoniq->cssr, ES_REG(ensoniq, STATUS));
+ synchronize_irq(ensoniq->irq);
+}
+
+#ifdef CONFIG_PM
+static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct ensoniq *ensoniq = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ snd_pcm_suspend_all(ensoniq->pcm1);
+ snd_pcm_suspend_all(ensoniq->pcm2);
+
+#ifdef CHIP1371
+ snd_ac97_suspend(ensoniq->u.es1371.ac97);
+#else
+ /* try to reset AK4531 */
+ outw(ES_1370_CODEC_WRITE(AK4531_RESET, 0x02), ES_REG(ensoniq, 1370_CODEC));
+ inw(ES_REG(ensoniq, 1370_CODEC));
+ udelay(100);
+ outw(ES_1370_CODEC_WRITE(AK4531_RESET, 0x03), ES_REG(ensoniq, 1370_CODEC));
+ inw(ES_REG(ensoniq, 1370_CODEC));
+ udelay(100);
+ snd_ak4531_suspend(ensoniq->u.es1370.ak4531);
+#endif
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_ensoniq_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct ensoniq *ensoniq = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR DRIVER_NAME ": pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_ensoniq_chip_init(ensoniq);
+
+#ifdef CHIP1371
+ snd_ac97_resume(ensoniq->u.es1371.ac97);
+#else
+ snd_ak4531_resume(ensoniq->u.es1370.ak4531);
+#endif
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+static int __devinit snd_ensoniq_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct ensoniq ** rensoniq)
+{
+ struct ensoniq *ensoniq;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ensoniq_dev_free,
+ };
+
+ *rensoniq = NULL;
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ ensoniq = kzalloc(sizeof(*ensoniq), GFP_KERNEL);
+ if (ensoniq == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&ensoniq->reg_lock);
+ mutex_init(&ensoniq->src_mutex);
+ ensoniq->card = card;
+ ensoniq->pci = pci;
+ ensoniq->irq = -1;
+ if ((err = pci_request_regions(pci, "Ensoniq AudioPCI")) < 0) {
+ kfree(ensoniq);
+ pci_disable_device(pci);
+ return err;
+ }
+ ensoniq->port = pci_resource_start(pci, 0);
+ if (request_irq(pci->irq, snd_audiopci_interrupt, IRQF_SHARED,
+ "Ensoniq AudioPCI", ensoniq)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_ensoniq_free(ensoniq);
+ return -EBUSY;
+ }
+ ensoniq->irq = pci->irq;
+#ifdef CHIP1370
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ 16, &ensoniq->dma_bug) < 0) {
+ snd_printk(KERN_ERR "unable to allocate space for phantom area - dma_bug\n");
+ snd_ensoniq_free(ensoniq);
+ return -EBUSY;
+ }
+#endif
+ pci_set_master(pci);
+ ensoniq->rev = pci->revision;
+#ifdef CHIP1370
+#if 0
+ ensoniq->ctrl = ES_1370_CDC_EN | ES_1370_SERR_DISABLE |
+ ES_1370_PCLKDIVO(ES_1370_SRTODIV(8000));
+#else /* get microphone working */
+ ensoniq->ctrl = ES_1370_CDC_EN | ES_1370_PCLKDIVO(ES_1370_SRTODIV(8000));
+#endif
+ ensoniq->sctrl = 0;
+#else
+ ensoniq->ctrl = 0;
+ ensoniq->sctrl = 0;
+ ensoniq->cssr = 0;
+ if (snd_pci_quirk_lookup(pci, es1371_amplifier_hack))
+ ensoniq->ctrl |= ES_1371_GPIO_OUT(1); /* turn amplifier on */
+
+ if (es1371_quirk_lookup(ensoniq, es1371_ac97_reset_hack))
+ ensoniq->cssr |= ES_1371_ST_AC97_RST;
+#endif
+
+ snd_ensoniq_chip_init(ensoniq);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ensoniq, &ops)) < 0) {
+ snd_ensoniq_free(ensoniq);
+ return err;
+ }
+
+ snd_ensoniq_proc_init(ensoniq);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rensoniq = ensoniq;
+ return 0;
+}
+
+/*
+ * MIDI section
+ */
+
+static void snd_ensoniq_midi_interrupt(struct ensoniq * ensoniq)
+{
+ struct snd_rawmidi *rmidi = ensoniq->rmidi;
+ unsigned char status, mask, byte;
+
+ if (rmidi == NULL)
+ return;
+ /* do Rx at first */
+ spin_lock(&ensoniq->reg_lock);
+ mask = ensoniq->uartm & ES_MODE_INPUT ? ES_RXRDY : 0;
+ while (mask) {
+ status = inb(ES_REG(ensoniq, UART_STATUS));
+ if ((status & mask) == 0)
+ break;
+ byte = inb(ES_REG(ensoniq, UART_DATA));
+ snd_rawmidi_receive(ensoniq->midi_input, &byte, 1);
+ }
+ spin_unlock(&ensoniq->reg_lock);
+
+ /* do Tx at second */
+ spin_lock(&ensoniq->reg_lock);
+ mask = ensoniq->uartm & ES_MODE_OUTPUT ? ES_TXRDY : 0;
+ while (mask) {
+ status = inb(ES_REG(ensoniq, UART_STATUS));
+ if ((status & mask) == 0)
+ break;
+ if (snd_rawmidi_transmit(ensoniq->midi_output, &byte, 1) != 1) {
+ ensoniq->uartc &= ~ES_TXINTENM;
+ outb(ensoniq->uartc, ES_REG(ensoniq, UART_CONTROL));
+ mask &= ~ES_TXRDY;
+ } else {
+ outb(byte, ES_REG(ensoniq, UART_DATA));
+ }
+ }
+ spin_unlock(&ensoniq->reg_lock);
+}
+
+static int snd_ensoniq_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct ensoniq *ensoniq = substream->rmidi->private_data;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ ensoniq->uartm |= ES_MODE_INPUT;
+ ensoniq->midi_input = substream;
+ if (!(ensoniq->uartm & ES_MODE_OUTPUT)) {
+ outb(ES_CNTRL(3), ES_REG(ensoniq, UART_CONTROL));
+ outb(ensoniq->uartc = 0, ES_REG(ensoniq, UART_CONTROL));
+ outl(ensoniq->ctrl |= ES_UART_EN, ES_REG(ensoniq, CONTROL));
+ }
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_ensoniq_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct ensoniq *ensoniq = substream->rmidi->private_data;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ if (!(ensoniq->uartm & ES_MODE_OUTPUT)) {
+ outb(ensoniq->uartc = 0, ES_REG(ensoniq, UART_CONTROL));
+ outl(ensoniq->ctrl &= ~ES_UART_EN, ES_REG(ensoniq, CONTROL));
+ } else {
+ outb(ensoniq->uartc &= ~ES_RXINTEN, ES_REG(ensoniq, UART_CONTROL));
+ }
+ ensoniq->midi_input = NULL;
+ ensoniq->uartm &= ~ES_MODE_INPUT;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_ensoniq_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct ensoniq *ensoniq = substream->rmidi->private_data;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ ensoniq->uartm |= ES_MODE_OUTPUT;
+ ensoniq->midi_output = substream;
+ if (!(ensoniq->uartm & ES_MODE_INPUT)) {
+ outb(ES_CNTRL(3), ES_REG(ensoniq, UART_CONTROL));
+ outb(ensoniq->uartc = 0, ES_REG(ensoniq, UART_CONTROL));
+ outl(ensoniq->ctrl |= ES_UART_EN, ES_REG(ensoniq, CONTROL));
+ }
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static int snd_ensoniq_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct ensoniq *ensoniq = substream->rmidi->private_data;
+
+ spin_lock_irq(&ensoniq->reg_lock);
+ if (!(ensoniq->uartm & ES_MODE_INPUT)) {
+ outb(ensoniq->uartc = 0, ES_REG(ensoniq, UART_CONTROL));
+ outl(ensoniq->ctrl &= ~ES_UART_EN, ES_REG(ensoniq, CONTROL));
+ } else {
+ outb(ensoniq->uartc &= ~ES_TXINTENM, ES_REG(ensoniq, UART_CONTROL));
+ }
+ ensoniq->midi_output = NULL;
+ ensoniq->uartm &= ~ES_MODE_OUTPUT;
+ spin_unlock_irq(&ensoniq->reg_lock);
+ return 0;
+}
+
+static void snd_ensoniq_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct ensoniq *ensoniq = substream->rmidi->private_data;
+ int idx;
+
+ spin_lock_irqsave(&ensoniq->reg_lock, flags);
+ if (up) {
+ if ((ensoniq->uartc & ES_RXINTEN) == 0) {
+ /* empty input FIFO */
+ for (idx = 0; idx < 32; idx++)
+ inb(ES_REG(ensoniq, UART_DATA));
+ ensoniq->uartc |= ES_RXINTEN;
+ outb(ensoniq->uartc, ES_REG(ensoniq, UART_CONTROL));
+ }
+ } else {
+ if (ensoniq->uartc & ES_RXINTEN) {
+ ensoniq->uartc &= ~ES_RXINTEN;
+ outb(ensoniq->uartc, ES_REG(ensoniq, UART_CONTROL));
+ }
+ }
+ spin_unlock_irqrestore(&ensoniq->reg_lock, flags);
+}
+
+static void snd_ensoniq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ unsigned long flags;
+ struct ensoniq *ensoniq = substream->rmidi->private_data;
+ unsigned char byte;
+
+ spin_lock_irqsave(&ensoniq->reg_lock, flags);
+ if (up) {
+ if (ES_TXINTENI(ensoniq->uartc) == 0) {
+ ensoniq->uartc |= ES_TXINTENO(1);
+ /* fill UART FIFO buffer at first, and turn Tx interrupts only if necessary */
+ while (ES_TXINTENI(ensoniq->uartc) == 1 &&
+ (inb(ES_REG(ensoniq, UART_STATUS)) & ES_TXRDY)) {
+ if (snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+ ensoniq->uartc &= ~ES_TXINTENM;
+ } else {
+ outb(byte, ES_REG(ensoniq, UART_DATA));
+ }
+ }
+ outb(ensoniq->uartc, ES_REG(ensoniq, UART_CONTROL));
+ }
+ } else {
+ if (ES_TXINTENI(ensoniq->uartc) == 1) {
+ ensoniq->uartc &= ~ES_TXINTENM;
+ outb(ensoniq->uartc, ES_REG(ensoniq, UART_CONTROL));
+ }
+ }
+ spin_unlock_irqrestore(&ensoniq->reg_lock, flags);
+}
+
+static struct snd_rawmidi_ops snd_ensoniq_midi_output =
+{
+ .open = snd_ensoniq_midi_output_open,
+ .close = snd_ensoniq_midi_output_close,
+ .trigger = snd_ensoniq_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_ensoniq_midi_input =
+{
+ .open = snd_ensoniq_midi_input_open,
+ .close = snd_ensoniq_midi_input_close,
+ .trigger = snd_ensoniq_midi_input_trigger,
+};
+
+static int __devinit snd_ensoniq_midi(struct ensoniq * ensoniq, int device,
+ struct snd_rawmidi **rrawmidi)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ if (rrawmidi)
+ *rrawmidi = NULL;
+ if ((err = snd_rawmidi_new(ensoniq->card, "ES1370/1", device, 1, 1, &rmidi)) < 0)
+ return err;
+#ifdef CHIP1370
+ strcpy(rmidi->name, "ES1370");
+#else
+ strcpy(rmidi->name, "ES1371");
+#endif
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_ensoniq_midi_output);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_ensoniq_midi_input);
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = ensoniq;
+ ensoniq->rmidi = rmidi;
+ if (rrawmidi)
+ *rrawmidi = rmidi;
+ return 0;
+}
+
+/*
+ * Interrupt handler
+ */
+
+static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id)
+{
+ struct ensoniq *ensoniq = dev_id;
+ unsigned int status, sctrl;
+
+ if (ensoniq == NULL)
+ return IRQ_NONE;
+
+ status = inl(ES_REG(ensoniq, STATUS));
+ if (!(status & ES_INTR))
+ return IRQ_NONE;
+
+ spin_lock(&ensoniq->reg_lock);
+ sctrl = ensoniq->sctrl;
+ if (status & ES_DAC1)
+ sctrl &= ~ES_P1_INT_EN;
+ if (status & ES_DAC2)
+ sctrl &= ~ES_P2_INT_EN;
+ if (status & ES_ADC)
+ sctrl &= ~ES_R1_INT_EN;
+ outl(sctrl, ES_REG(ensoniq, SERIAL));
+ outl(ensoniq->sctrl, ES_REG(ensoniq, SERIAL));
+ spin_unlock(&ensoniq->reg_lock);
+
+ if (status & ES_UART)
+ snd_ensoniq_midi_interrupt(ensoniq);
+ if ((status & ES_DAC2) && ensoniq->playback2_substream)
+ snd_pcm_period_elapsed(ensoniq->playback2_substream);
+ if ((status & ES_ADC) && ensoniq->capture_substream)
+ snd_pcm_period_elapsed(ensoniq->capture_substream);
+ if ((status & ES_DAC1) && ensoniq->playback1_substream)
+ snd_pcm_period_elapsed(ensoniq->playback1_substream);
+ return IRQ_HANDLED;
+}
+
+static int __devinit snd_audiopci_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct ensoniq *ensoniq;
+ int err, pcm_devs[2];
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_ensoniq_create(card, pci, &ensoniq)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = ensoniq;
+
+ pcm_devs[0] = 0; pcm_devs[1] = 1;
+#ifdef CHIP1370
+ if ((err = snd_ensoniq_1370_mixer(ensoniq)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#endif
+#ifdef CHIP1371
+ if ((err = snd_ensoniq_1371_mixer(ensoniq, spdif[dev], lineio[dev])) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+#endif
+ if ((err = snd_ensoniq_pcm(ensoniq, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ensoniq_pcm2(ensoniq, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ensoniq_midi(ensoniq, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_ensoniq_create_gameport(ensoniq, dev);
+
+ strcpy(card->driver, DRIVER_NAME);
+
+ strcpy(card->shortname, "Ensoniq AudioPCI");
+ sprintf(card->longname, "%s %s at 0x%lx, irq %i",
+ card->shortname,
+ card->driver,
+ ensoniq->port,
+ ensoniq->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_audiopci_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = DRIVER_NAME,
+ .id_table = snd_audiopci_ids,
+ .probe = snd_audiopci_probe,
+ .remove = __devexit_p(snd_audiopci_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_ensoniq_suspend,
+ .resume = snd_ensoniq_resume,
+#endif
+};
+
+static int __init alsa_card_ens137x_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_ens137x_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_ens137x_init)
+module_exit(alsa_card_ens137x_exit)
diff --git a/sound/pci/ens1371.c b/sound/pci/ens1371.c
new file mode 100644
index 0000000..ca0da0a
--- /dev/null
+++ b/sound/pci/ens1371.c
@@ -0,0 +1,2 @@
+#define CHIP1371
+#include "ens1370.c"
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
new file mode 100644
index 0000000..4cd9a1f
--- /dev/null
+++ b/sound/pci/es1938.c
@@ -0,0 +1,1899 @@
+/*
+ * Driver for ESS Solo-1 (ES1938, ES1946, ES1969) soundcard
+ * Copyright (c) by Jaromir Koutek <miri@punknet.cz>,
+ * Jaroslav Kysela <perex@perex.cz>,
+ * Thomas Sailer <sailer@ife.ee.ethz.ch>,
+ * Abramo Bagnara <abramo@alsa-project.org>,
+ * Markus Gruber <gruber@eikon.tum.de>
+ *
+ * Rewritten from sonicvibes.c source.
+ *
+ * TODO:
+ * Rewrite better spinlocks
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ NOTES:
+ - Capture data is written unaligned starting from dma_base + 1 so I need to
+ disable mmap and to add a copy callback.
+ - After several cycle of the following:
+ while : ; do arecord -d1 -f cd -t raw | aplay -f cd ; done
+ a "playback write error (DMA or IRQ trouble?)" may happen.
+ This is due to playback interrupts not generated.
+ I suspect a timing issue.
+ - Sometimes the interrupt handler is invoked wrongly during playback.
+ This generates some harmless "Unexpected hw_pointer: wrong interrupt
+ acknowledge".
+ I've seen that using small period sizes.
+ Reproducible with:
+ mpg123 test.mp3 &
+ hdparm -t -T /dev/hda
+*/
+
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <linux/delay.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/opl3.h>
+#include <sound/mpu401.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <asm/io.h>
+
+MODULE_AUTHOR("Jaromir Koutek <miri@punknet.cz>");
+MODULE_DESCRIPTION("ESS Solo-1");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ESS,ES1938},"
+ "{ESS,ES1946},"
+ "{ESS,ES1969},"
+ "{TerraTec,128i PCI}}");
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK 1
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for ESS Solo-1 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for ESS Solo-1 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable ESS Solo-1 soundcard.");
+
+#define SLIO_REG(chip, x) ((chip)->io_port + ESSIO_REG_##x)
+
+#define SLDM_REG(chip, x) ((chip)->ddma_port + ESSDM_REG_##x)
+
+#define SLSB_REG(chip, x) ((chip)->sb_port + ESSSB_REG_##x)
+
+#define SL_PCI_LEGACYCONTROL 0x40
+#define SL_PCI_CONFIG 0x50
+#define SL_PCI_DDMACONTROL 0x60
+
+#define ESSIO_REG_AUDIO2DMAADDR 0
+#define ESSIO_REG_AUDIO2DMACOUNT 4
+#define ESSIO_REG_AUDIO2MODE 6
+#define ESSIO_REG_IRQCONTROL 7
+
+#define ESSDM_REG_DMAADDR 0x00
+#define ESSDM_REG_DMACOUNT 0x04
+#define ESSDM_REG_DMACOMMAND 0x08
+#define ESSDM_REG_DMASTATUS 0x08
+#define ESSDM_REG_DMAMODE 0x0b
+#define ESSDM_REG_DMACLEAR 0x0d
+#define ESSDM_REG_DMAMASK 0x0f
+
+#define ESSSB_REG_FMLOWADDR 0x00
+#define ESSSB_REG_FMHIGHADDR 0x02
+#define ESSSB_REG_MIXERADDR 0x04
+#define ESSSB_REG_MIXERDATA 0x05
+
+#define ESSSB_IREG_AUDIO1 0x14
+#define ESSSB_IREG_MICMIX 0x1a
+#define ESSSB_IREG_RECSRC 0x1c
+#define ESSSB_IREG_MASTER 0x32
+#define ESSSB_IREG_FM 0x36
+#define ESSSB_IREG_AUXACD 0x38
+#define ESSSB_IREG_AUXB 0x3a
+#define ESSSB_IREG_PCSPEAKER 0x3c
+#define ESSSB_IREG_LINE 0x3e
+#define ESSSB_IREG_SPATCONTROL 0x50
+#define ESSSB_IREG_SPATLEVEL 0x52
+#define ESSSB_IREG_MASTER_LEFT 0x60
+#define ESSSB_IREG_MASTER_RIGHT 0x62
+#define ESSSB_IREG_MPU401CONTROL 0x64
+#define ESSSB_IREG_MICMIXRECORD 0x68
+#define ESSSB_IREG_AUDIO2RECORD 0x69
+#define ESSSB_IREG_AUXACDRECORD 0x6a
+#define ESSSB_IREG_FMRECORD 0x6b
+#define ESSSB_IREG_AUXBRECORD 0x6c
+#define ESSSB_IREG_MONO 0x6d
+#define ESSSB_IREG_LINERECORD 0x6e
+#define ESSSB_IREG_MONORECORD 0x6f
+#define ESSSB_IREG_AUDIO2SAMPLE 0x70
+#define ESSSB_IREG_AUDIO2MODE 0x71
+#define ESSSB_IREG_AUDIO2FILTER 0x72
+#define ESSSB_IREG_AUDIO2TCOUNTL 0x74
+#define ESSSB_IREG_AUDIO2TCOUNTH 0x76
+#define ESSSB_IREG_AUDIO2CONTROL1 0x78
+#define ESSSB_IREG_AUDIO2CONTROL2 0x7a
+#define ESSSB_IREG_AUDIO2 0x7c
+
+#define ESSSB_REG_RESET 0x06
+
+#define ESSSB_REG_READDATA 0x0a
+#define ESSSB_REG_WRITEDATA 0x0c
+#define ESSSB_REG_READSTATUS 0x0c
+
+#define ESSSB_REG_STATUS 0x0e
+
+#define ESS_CMD_EXTSAMPLERATE 0xa1
+#define ESS_CMD_FILTERDIV 0xa2
+#define ESS_CMD_DMACNTRELOADL 0xa4
+#define ESS_CMD_DMACNTRELOADH 0xa5
+#define ESS_CMD_ANALOGCONTROL 0xa8
+#define ESS_CMD_IRQCONTROL 0xb1
+#define ESS_CMD_DRQCONTROL 0xb2
+#define ESS_CMD_RECLEVEL 0xb4
+#define ESS_CMD_SETFORMAT 0xb6
+#define ESS_CMD_SETFORMAT2 0xb7
+#define ESS_CMD_DMACONTROL 0xb8
+#define ESS_CMD_DMATYPE 0xb9
+#define ESS_CMD_OFFSETLEFT 0xba
+#define ESS_CMD_OFFSETRIGHT 0xbb
+#define ESS_CMD_READREG 0xc0
+#define ESS_CMD_ENABLEEXT 0xc6
+#define ESS_CMD_PAUSEDMA 0xd0
+#define ESS_CMD_ENABLEAUDIO1 0xd1
+#define ESS_CMD_STOPAUDIO1 0xd3
+#define ESS_CMD_AUDIO1STATUS 0xd8
+#define ESS_CMD_CONTDMA 0xd4
+#define ESS_CMD_TESTIRQ 0xf2
+
+#define ESS_RECSRC_MIC 0
+#define ESS_RECSRC_AUXACD 2
+#define ESS_RECSRC_AUXB 5
+#define ESS_RECSRC_LINE 6
+#define ESS_RECSRC_NONE 7
+
+#define DAC1 0x01
+#define ADC1 0x02
+#define DAC2 0x04
+
+/*
+
+ */
+
+#define SAVED_REG_SIZE 32 /* max. number of registers to save */
+
+struct es1938 {
+ int irq;
+
+ unsigned long io_port;
+ unsigned long sb_port;
+ unsigned long vc_port;
+ unsigned long mpu_port;
+ unsigned long game_port;
+ unsigned long ddma_port;
+
+ unsigned char irqmask;
+ unsigned char revision;
+
+ struct snd_kcontrol *hw_volume;
+ struct snd_kcontrol *hw_switch;
+ struct snd_kcontrol *master_volume;
+ struct snd_kcontrol *master_switch;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *capture_substream;
+ struct snd_pcm_substream *playback1_substream;
+ struct snd_pcm_substream *playback2_substream;
+ struct snd_rawmidi *rmidi;
+
+ unsigned int dma1_size;
+ unsigned int dma2_size;
+ unsigned int dma1_start;
+ unsigned int dma2_start;
+ unsigned int dma1_shift;
+ unsigned int dma2_shift;
+ unsigned int last_capture_dmaaddr;
+ unsigned int active;
+
+ spinlock_t reg_lock;
+ spinlock_t mixer_lock;
+ struct snd_info_entry *proc_entry;
+
+#ifdef SUPPORT_JOYSTICK
+ struct gameport *gameport;
+#endif
+#ifdef CONFIG_PM
+ unsigned char saved_regs[SAVED_REG_SIZE];
+#endif
+};
+
+static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id);
+
+static struct pci_device_id snd_es1938_ids[] = {
+ { 0x125d, 0x1969, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Solo-1 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_es1938_ids);
+
+#define RESET_LOOP_TIMEOUT 0x10000
+#define WRITE_LOOP_TIMEOUT 0x10000
+#define GET_LOOP_TIMEOUT 0x01000
+
+#undef REG_DEBUG
+/* -----------------------------------------------------------------
+ * Write to a mixer register
+ * -----------------------------------------------------------------*/
+static void snd_es1938_mixer_write(struct es1938 *chip, unsigned char reg, unsigned char val)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ outb(reg, SLSB_REG(chip, MIXERADDR));
+ outb(val, SLSB_REG(chip, MIXERDATA));
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Mixer reg %02x set to %02x\n", reg, val);
+#endif
+}
+
+/* -----------------------------------------------------------------
+ * Read from a mixer register
+ * -----------------------------------------------------------------*/
+static int snd_es1938_mixer_read(struct es1938 *chip, unsigned char reg)
+{
+ int data;
+ unsigned long flags;
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ outb(reg, SLSB_REG(chip, MIXERADDR));
+ data = inb(SLSB_REG(chip, MIXERDATA));
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Mixer reg %02x now is %02x\n", reg, data);
+#endif
+ return data;
+}
+
+/* -----------------------------------------------------------------
+ * Write to some bits of a mixer register (return old value)
+ * -----------------------------------------------------------------*/
+static int snd_es1938_mixer_bits(struct es1938 *chip, unsigned char reg,
+ unsigned char mask, unsigned char val)
+{
+ unsigned long flags;
+ unsigned char old, new, oval;
+ spin_lock_irqsave(&chip->mixer_lock, flags);
+ outb(reg, SLSB_REG(chip, MIXERADDR));
+ old = inb(SLSB_REG(chip, MIXERDATA));
+ oval = old & mask;
+ if (val != oval) {
+ new = (old & ~mask) | (val & mask);
+ outb(new, SLSB_REG(chip, MIXERDATA));
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Mixer reg %02x was %02x, set to %02x\n",
+ reg, old, new);
+#endif
+ }
+ spin_unlock_irqrestore(&chip->mixer_lock, flags);
+ return oval;
+}
+
+/* -----------------------------------------------------------------
+ * Write command to Controller Registers
+ * -----------------------------------------------------------------*/
+static void snd_es1938_write_cmd(struct es1938 *chip, unsigned char cmd)
+{
+ int i;
+ unsigned char v;
+ for (i = 0; i < WRITE_LOOP_TIMEOUT; i++) {
+ if (!(v = inb(SLSB_REG(chip, READSTATUS)) & 0x80)) {
+ outb(cmd, SLSB_REG(chip, WRITEDATA));
+ return;
+ }
+ }
+ printk(KERN_ERR "snd_es1938_write_cmd timeout (0x02%x/0x02%x)\n", cmd, v);
+}
+
+/* -----------------------------------------------------------------
+ * Read the Read Data Buffer
+ * -----------------------------------------------------------------*/
+static int snd_es1938_get_byte(struct es1938 *chip)
+{
+ int i;
+ unsigned char v;
+ for (i = GET_LOOP_TIMEOUT; i; i--)
+ if ((v = inb(SLSB_REG(chip, STATUS))) & 0x80)
+ return inb(SLSB_REG(chip, READDATA));
+ snd_printk(KERN_ERR "get_byte timeout: status 0x02%x\n", v);
+ return -ENODEV;
+}
+
+/* -----------------------------------------------------------------
+ * Write value cmd register
+ * -----------------------------------------------------------------*/
+static void snd_es1938_write(struct es1938 *chip, unsigned char reg, unsigned char val)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1938_write_cmd(chip, reg);
+ snd_es1938_write_cmd(chip, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Reg %02x set to %02x\n", reg, val);
+#endif
+}
+
+/* -----------------------------------------------------------------
+ * Read data from cmd register and return it
+ * -----------------------------------------------------------------*/
+static unsigned char snd_es1938_read(struct es1938 *chip, unsigned char reg)
+{
+ unsigned char val;
+ unsigned long flags;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1938_write_cmd(chip, ESS_CMD_READREG);
+ snd_es1938_write_cmd(chip, reg);
+ val = snd_es1938_get_byte(chip);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Reg %02x now is %02x\n", reg, val);
+#endif
+ return val;
+}
+
+/* -----------------------------------------------------------------
+ * Write data to cmd register and return old value
+ * -----------------------------------------------------------------*/
+static int snd_es1938_bits(struct es1938 *chip, unsigned char reg, unsigned char mask,
+ unsigned char val)
+{
+ unsigned long flags;
+ unsigned char old, new, oval;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_es1938_write_cmd(chip, ESS_CMD_READREG);
+ snd_es1938_write_cmd(chip, reg);
+ old = snd_es1938_get_byte(chip);
+ oval = old & mask;
+ if (val != oval) {
+ snd_es1938_write_cmd(chip, reg);
+ new = (old & ~mask) | (val & mask);
+ snd_es1938_write_cmd(chip, new);
+#ifdef REG_DEBUG
+ snd_printk(KERN_DEBUG "Reg %02x was %02x, set to %02x\n",
+ reg, old, new);
+#endif
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return oval;
+}
+
+/* --------------------------------------------------------------------
+ * Reset the chip
+ * --------------------------------------------------------------------*/
+static void snd_es1938_reset(struct es1938 *chip)
+{
+ int i;
+
+ outb(3, SLSB_REG(chip, RESET));
+ inb(SLSB_REG(chip, RESET));
+ outb(0, SLSB_REG(chip, RESET));
+ for (i = 0; i < RESET_LOOP_TIMEOUT; i++) {
+ if (inb(SLSB_REG(chip, STATUS)) & 0x80) {
+ if (inb(SLSB_REG(chip, READDATA)) == 0xaa)
+ goto __next;
+ }
+ }
+ snd_printk(KERN_ERR "ESS Solo-1 reset failed\n");
+
+ __next:
+ snd_es1938_write_cmd(chip, ESS_CMD_ENABLEEXT);
+
+ /* Demand transfer DMA: 4 bytes per DMA request */
+ snd_es1938_write(chip, ESS_CMD_DMATYPE, 2);
+
+ /* Change behaviour of register A1
+ 4x oversampling
+ 2nd channel DAC asynchronous */
+ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2MODE, 0x32);
+ /* enable/select DMA channel and IRQ channel */
+ snd_es1938_bits(chip, ESS_CMD_IRQCONTROL, 0xf0, 0x50);
+ snd_es1938_bits(chip, ESS_CMD_DRQCONTROL, 0xf0, 0x50);
+ snd_es1938_write_cmd(chip, ESS_CMD_ENABLEAUDIO1);
+ /* Set spatializer parameters to recommended values */
+ snd_es1938_mixer_write(chip, 0x54, 0x8f);
+ snd_es1938_mixer_write(chip, 0x56, 0x95);
+ snd_es1938_mixer_write(chip, 0x58, 0x94);
+ snd_es1938_mixer_write(chip, 0x5a, 0x80);
+}
+
+/* --------------------------------------------------------------------
+ * Reset the FIFOs
+ * --------------------------------------------------------------------*/
+static void snd_es1938_reset_fifo(struct es1938 *chip)
+{
+ outb(2, SLSB_REG(chip, RESET));
+ outb(0, SLSB_REG(chip, RESET));
+}
+
+static struct snd_ratnum clocks[2] = {
+ {
+ .num = 793800,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ },
+ {
+ .num = 768000,
+ .den_min = 1,
+ .den_max = 128,
+ .den_step = 1,
+ }
+};
+
+static struct snd_pcm_hw_constraint_ratnums hw_constraints_clocks = {
+ .nrats = 2,
+ .rats = clocks,
+};
+
+
+static void snd_es1938_rate_set(struct es1938 *chip,
+ struct snd_pcm_substream *substream,
+ int mode)
+{
+ unsigned int bits, div0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (runtime->rate_num == clocks[0].num)
+ bits = 128 - runtime->rate_den;
+ else
+ bits = 256 - runtime->rate_den;
+
+ /* set filter register */
+ div0 = 256 - 7160000*20/(8*82*runtime->rate);
+
+ if (mode == DAC2) {
+ snd_es1938_mixer_write(chip, 0x70, bits);
+ snd_es1938_mixer_write(chip, 0x72, div0);
+ } else {
+ snd_es1938_write(chip, 0xA1, bits);
+ snd_es1938_write(chip, 0xA2, div0);
+ }
+}
+
+/* --------------------------------------------------------------------
+ * Configure Solo1 builtin DMA Controller
+ * --------------------------------------------------------------------*/
+
+static void snd_es1938_playback1_setdma(struct es1938 *chip)
+{
+ outb(0x00, SLIO_REG(chip, AUDIO2MODE));
+ outl(chip->dma2_start, SLIO_REG(chip, AUDIO2DMAADDR));
+ outw(0, SLIO_REG(chip, AUDIO2DMACOUNT));
+ outw(chip->dma2_size, SLIO_REG(chip, AUDIO2DMACOUNT));
+}
+
+static void snd_es1938_playback2_setdma(struct es1938 *chip)
+{
+ /* Enable DMA controller */
+ outb(0xc4, SLDM_REG(chip, DMACOMMAND));
+ /* 1. Master reset */
+ outb(0, SLDM_REG(chip, DMACLEAR));
+ /* 2. Mask DMA */
+ outb(1, SLDM_REG(chip, DMAMASK));
+ outb(0x18, SLDM_REG(chip, DMAMODE));
+ outl(chip->dma1_start, SLDM_REG(chip, DMAADDR));
+ outw(chip->dma1_size - 1, SLDM_REG(chip, DMACOUNT));
+ /* 3. Unmask DMA */
+ outb(0, SLDM_REG(chip, DMAMASK));
+}
+
+static void snd_es1938_capture_setdma(struct es1938 *chip)
+{
+ /* Enable DMA controller */
+ outb(0xc4, SLDM_REG(chip, DMACOMMAND));
+ /* 1. Master reset */
+ outb(0, SLDM_REG(chip, DMACLEAR));
+ /* 2. Mask DMA */
+ outb(1, SLDM_REG(chip, DMAMASK));
+ outb(0x14, SLDM_REG(chip, DMAMODE));
+ outl(chip->dma1_start, SLDM_REG(chip, DMAADDR));
+ chip->last_capture_dmaaddr = chip->dma1_start;
+ outw(chip->dma1_size - 1, SLDM_REG(chip, DMACOUNT));
+ /* 3. Unmask DMA */
+ outb(0, SLDM_REG(chip, DMAMASK));
+}
+
+/* ----------------------------------------------------------------------
+ *
+ * *** PCM part ***
+ */
+
+static int snd_es1938_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ int val;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ val = 0x0f;
+ chip->active |= ADC1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ val = 0x00;
+ chip->active &= ~ADC1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_es1938_write(chip, ESS_CMD_DMACONTROL, val);
+ return 0;
+}
+
+static int snd_es1938_playback1_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ /* According to the documentation this should be:
+ 0x13 but that value may randomly swap stereo channels */
+ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2CONTROL1, 0x92);
+ udelay(10);
+ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2CONTROL1, 0x93);
+ /* This two stage init gives the FIFO -> DAC connection time to
+ * settle before first data from DMA flows in. This should ensure
+ * no swapping of stereo channels. Report a bug if otherwise :-) */
+ outb(0x0a, SLIO_REG(chip, AUDIO2MODE));
+ chip->active |= DAC2;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ outb(0, SLIO_REG(chip, AUDIO2MODE));
+ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2CONTROL1, 0);
+ chip->active &= ~DAC2;
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int snd_es1938_playback2_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ int val;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ val = 5;
+ chip->active |= DAC1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ val = 0;
+ chip->active &= ~DAC1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_es1938_write(chip, ESS_CMD_DMACONTROL, val);
+ return 0;
+}
+
+static int snd_es1938_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ switch (substream->number) {
+ case 0:
+ return snd_es1938_playback1_trigger(substream, cmd);
+ case 1:
+ return snd_es1938_playback2_trigger(substream, cmd);
+ }
+ snd_BUG();
+ return -EINVAL;
+}
+
+/* --------------------------------------------------------------------
+ * First channel for Extended Mode Audio 1 ADC Operation
+ * --------------------------------------------------------------------*/
+static int snd_es1938_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int u, is8, mono;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma1_size = size;
+ chip->dma1_start = runtime->dma_addr;
+
+ mono = (runtime->channels > 1) ? 0 : 1;
+ is8 = snd_pcm_format_width(runtime->format) == 16 ? 0 : 1;
+ u = snd_pcm_format_unsigned(runtime->format);
+
+ chip->dma1_shift = 2 - mono - is8;
+
+ snd_es1938_reset_fifo(chip);
+
+ /* program type */
+ snd_es1938_bits(chip, ESS_CMD_ANALOGCONTROL, 0x03, (mono ? 2 : 1));
+
+ /* set clock and counters */
+ snd_es1938_rate_set(chip, substream, ADC1);
+
+ count = 0x10000 - count;
+ snd_es1938_write(chip, ESS_CMD_DMACNTRELOADL, count & 0xff);
+ snd_es1938_write(chip, ESS_CMD_DMACNTRELOADH, count >> 8);
+
+ /* initialize and configure ADC */
+ snd_es1938_write(chip, ESS_CMD_SETFORMAT2, u ? 0x51 : 0x71);
+ snd_es1938_write(chip, ESS_CMD_SETFORMAT2, 0x90 |
+ (u ? 0x00 : 0x20) |
+ (is8 ? 0x00 : 0x04) |
+ (mono ? 0x40 : 0x08));
+
+ // snd_es1938_reset_fifo(chip);
+
+ /* 11. configure system interrupt controller and DMA controller */
+ snd_es1938_capture_setdma(chip);
+
+ return 0;
+}
+
+
+/* ------------------------------------------------------------------------------
+ * Second Audio channel DAC Operation
+ * ------------------------------------------------------------------------------*/
+static int snd_es1938_playback1_prepare(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int u, is8, mono;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma2_size = size;
+ chip->dma2_start = runtime->dma_addr;
+
+ mono = (runtime->channels > 1) ? 0 : 1;
+ is8 = snd_pcm_format_width(runtime->format) == 16 ? 0 : 1;
+ u = snd_pcm_format_unsigned(runtime->format);
+
+ chip->dma2_shift = 2 - mono - is8;
+
+ snd_es1938_reset_fifo(chip);
+
+ /* set clock and counters */
+ snd_es1938_rate_set(chip, substream, DAC2);
+
+ count >>= 1;
+ count = 0x10000 - count;
+ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2TCOUNTL, count & 0xff);
+ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2TCOUNTH, count >> 8);
+
+ /* initialize and configure Audio 2 DAC */
+ snd_es1938_mixer_write(chip, ESSSB_IREG_AUDIO2CONTROL2, 0x40 | (u ? 0 : 4) |
+ (mono ? 0 : 2) | (is8 ? 0 : 1));
+
+ /* program DMA */
+ snd_es1938_playback1_setdma(chip);
+
+ return 0;
+}
+
+static int snd_es1938_playback2_prepare(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int u, is8, mono;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ chip->dma1_size = size;
+ chip->dma1_start = runtime->dma_addr;
+
+ mono = (runtime->channels > 1) ? 0 : 1;
+ is8 = snd_pcm_format_width(runtime->format) == 16 ? 0 : 1;
+ u = snd_pcm_format_unsigned(runtime->format);
+
+ chip->dma1_shift = 2 - mono - is8;
+
+ count = 0x10000 - count;
+
+ /* reset */
+ snd_es1938_reset_fifo(chip);
+
+ snd_es1938_bits(chip, ESS_CMD_ANALOGCONTROL, 0x03, (mono ? 2 : 1));
+
+ /* set clock and counters */
+ snd_es1938_rate_set(chip, substream, DAC1);
+ snd_es1938_write(chip, ESS_CMD_DMACNTRELOADL, count & 0xff);
+ snd_es1938_write(chip, ESS_CMD_DMACNTRELOADH, count >> 8);
+
+ /* initialized and configure DAC */
+ snd_es1938_write(chip, ESS_CMD_SETFORMAT, u ? 0x80 : 0x00);
+ snd_es1938_write(chip, ESS_CMD_SETFORMAT, u ? 0x51 : 0x71);
+ snd_es1938_write(chip, ESS_CMD_SETFORMAT2,
+ 0x90 | (mono ? 0x40 : 0x08) |
+ (is8 ? 0x00 : 0x04) | (u ? 0x00 : 0x20));
+
+ /* program DMA */
+ snd_es1938_playback2_setdma(chip);
+
+ return 0;
+}
+
+static int snd_es1938_playback_prepare(struct snd_pcm_substream *substream)
+{
+ switch (substream->number) {
+ case 0:
+ return snd_es1938_playback1_prepare(substream);
+ case 1:
+ return snd_es1938_playback2_prepare(substream);
+ }
+ snd_BUG();
+ return -EINVAL;
+}
+
+/* during the incrementing of dma counters the DMA register reads sometimes
+ returns garbage. To ensure a valid hw pointer, the following checks which
+ should be very unlikely to fail are used:
+ - is the current DMA address in the valid DMA range ?
+ - is the sum of DMA address and DMA counter pointing to the last DMA byte ?
+ One can argue this could differ by one byte depending on which register is
+ updated first, so the implementation below allows for that.
+*/
+static snd_pcm_uframes_t snd_es1938_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+#if 0
+ size_t old, new;
+ /* This stuff is *needed*, don't ask why - AB */
+ old = inw(SLDM_REG(chip, DMACOUNT));
+ while ((new = inw(SLDM_REG(chip, DMACOUNT))) != old)
+ old = new;
+ ptr = chip->dma1_size - 1 - new;
+#else
+ size_t count;
+ unsigned int diff;
+
+ ptr = inl(SLDM_REG(chip, DMAADDR));
+ count = inw(SLDM_REG(chip, DMACOUNT));
+ diff = chip->dma1_start + chip->dma1_size - ptr - count;
+
+ if (diff > 3 || ptr < chip->dma1_start
+ || ptr >= chip->dma1_start+chip->dma1_size)
+ ptr = chip->last_capture_dmaaddr; /* bad, use last saved */
+ else
+ chip->last_capture_dmaaddr = ptr; /* good, remember it */
+
+ ptr -= chip->dma1_start;
+#endif
+ return ptr >> chip->dma1_shift;
+}
+
+static snd_pcm_uframes_t snd_es1938_playback1_pointer(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+#if 1
+ ptr = chip->dma2_size - inw(SLIO_REG(chip, AUDIO2DMACOUNT));
+#else
+ ptr = inl(SLIO_REG(chip, AUDIO2DMAADDR)) - chip->dma2_start;
+#endif
+ return ptr >> chip->dma2_shift;
+}
+
+static snd_pcm_uframes_t snd_es1938_playback2_pointer(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+ size_t old, new;
+#if 1
+ /* This stuff is *needed*, don't ask why - AB */
+ old = inw(SLDM_REG(chip, DMACOUNT));
+ while ((new = inw(SLDM_REG(chip, DMACOUNT))) != old)
+ old = new;
+ ptr = chip->dma1_size - 1 - new;
+#else
+ ptr = inl(SLDM_REG(chip, DMAADDR)) - chip->dma1_start;
+#endif
+ return ptr >> chip->dma1_shift;
+}
+
+static snd_pcm_uframes_t snd_es1938_playback_pointer(struct snd_pcm_substream *substream)
+{
+ switch (substream->number) {
+ case 0:
+ return snd_es1938_playback1_pointer(substream);
+ case 1:
+ return snd_es1938_playback2_pointer(substream);
+ }
+ snd_BUG();
+ return -EINVAL;
+}
+
+static int snd_es1938_capture_copy(struct snd_pcm_substream *substream,
+ int channel,
+ snd_pcm_uframes_t pos,
+ void __user *dst,
+ snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ pos <<= chip->dma1_shift;
+ count <<= chip->dma1_shift;
+ if (snd_BUG_ON(pos + count > chip->dma1_size))
+ return -EINVAL;
+ if (pos + count < chip->dma1_size) {
+ if (copy_to_user(dst, runtime->dma_area + pos + 1, count))
+ return -EFAULT;
+ } else {
+ if (copy_to_user(dst, runtime->dma_area + pos + 1, count - 1))
+ return -EFAULT;
+ if (put_user(runtime->dma_area[0], ((unsigned char __user *)dst) + count - 1))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+/*
+ * buffer management
+ */
+static int snd_es1938_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+
+{
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_es1938_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* ----------------------------------------------------------------------
+ * Audio1 Capture (ADC)
+ * ----------------------------------------------------------------------*/
+static struct snd_pcm_hardware snd_es1938_capture =
+{
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 6000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 0x8000, /* DMA controller screws on higher values */
+ .period_bytes_min = 64,
+ .period_bytes_max = 0x8000,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 256,
+};
+
+/* -----------------------------------------------------------------------
+ * Audio2 Playback (DAC)
+ * -----------------------------------------------------------------------*/
+static struct snd_pcm_hardware snd_es1938_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 6000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 0x8000, /* DMA controller screws on higher values */
+ .period_bytes_min = 64,
+ .period_bytes_max = 0x8000,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 256,
+};
+
+static int snd_es1938_capture_open(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (chip->playback2_substream)
+ return -EAGAIN;
+ chip->capture_substream = substream;
+ runtime->hw = snd_es1938_capture;
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_clocks);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, 0xff00);
+ return 0;
+}
+
+static int snd_es1938_playback_open(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ switch (substream->number) {
+ case 0:
+ chip->playback1_substream = substream;
+ break;
+ case 1:
+ if (chip->capture_substream)
+ return -EAGAIN;
+ chip->playback2_substream = substream;
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+ runtime->hw = snd_es1938_playback;
+ snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_clocks);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 0, 0xff00);
+ return 0;
+}
+
+static int snd_es1938_capture_close(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+
+ chip->capture_substream = NULL;
+ return 0;
+}
+
+static int snd_es1938_playback_close(struct snd_pcm_substream *substream)
+{
+ struct es1938 *chip = snd_pcm_substream_chip(substream);
+
+ switch (substream->number) {
+ case 0:
+ chip->playback1_substream = NULL;
+ break;
+ case 1:
+ chip->playback2_substream = NULL;
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static struct snd_pcm_ops snd_es1938_playback_ops = {
+ .open = snd_es1938_playback_open,
+ .close = snd_es1938_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_es1938_pcm_hw_params,
+ .hw_free = snd_es1938_pcm_hw_free,
+ .prepare = snd_es1938_playback_prepare,
+ .trigger = snd_es1938_playback_trigger,
+ .pointer = snd_es1938_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_es1938_capture_ops = {
+ .open = snd_es1938_capture_open,
+ .close = snd_es1938_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_es1938_pcm_hw_params,
+ .hw_free = snd_es1938_pcm_hw_free,
+ .prepare = snd_es1938_capture_prepare,
+ .trigger = snd_es1938_capture_trigger,
+ .pointer = snd_es1938_capture_pointer,
+ .copy = snd_es1938_capture_copy,
+};
+
+static int __devinit snd_es1938_new_pcm(struct es1938 *chip, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(chip->card, "es-1938-1946", device, 2, 1, &pcm)) < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1938_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_es1938_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "ESS Solo-1");
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 64*1024);
+
+ chip->pcm = pcm;
+ return 0;
+}
+
+/* -------------------------------------------------------------------
+ *
+ * *** Mixer part ***
+ */
+
+static int snd_es1938_info_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[8] = {
+ "Mic", "Mic Master", "CD", "AOUT",
+ "Mic1", "Mix", "Line", "Master"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item > 7)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_es1938_get_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = snd_es1938_mixer_read(chip, 0x1c) & 0x07;
+ return 0;
+}
+
+static int snd_es1938_put_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char val = ucontrol->value.enumerated.item[0];
+
+ if (val > 7)
+ return -EINVAL;
+ return snd_es1938_mixer_bits(chip, 0x1c, 0x07, val) != val;
+}
+
+#define snd_es1938_info_spatializer_enable snd_ctl_boolean_mono_info
+
+static int snd_es1938_get_spatializer_enable(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char val = snd_es1938_mixer_read(chip, 0x50);
+ ucontrol->value.integer.value[0] = !!(val & 8);
+ return 0;
+}
+
+static int snd_es1938_put_spatializer_enable(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char oval, nval;
+ int change;
+ nval = ucontrol->value.integer.value[0] ? 0x0c : 0x04;
+ oval = snd_es1938_mixer_read(chip, 0x50) & 0x0c;
+ change = nval != oval;
+ if (change) {
+ snd_es1938_mixer_write(chip, 0x50, nval & ~0x04);
+ snd_es1938_mixer_write(chip, 0x50, nval);
+ }
+ return change;
+}
+
+static int snd_es1938_info_hw_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 63;
+ return 0;
+}
+
+static int snd_es1938_get_hw_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = snd_es1938_mixer_read(chip, 0x61) & 0x3f;
+ ucontrol->value.integer.value[1] = snd_es1938_mixer_read(chip, 0x63) & 0x3f;
+ return 0;
+}
+
+#define snd_es1938_info_hw_switch snd_ctl_boolean_stereo_info
+
+static int snd_es1938_get_hw_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = !(snd_es1938_mixer_read(chip, 0x61) & 0x40);
+ ucontrol->value.integer.value[1] = !(snd_es1938_mixer_read(chip, 0x63) & 0x40);
+ return 0;
+}
+
+static void snd_es1938_hwv_free(struct snd_kcontrol *kcontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ chip->master_volume = NULL;
+ chip->master_switch = NULL;
+ chip->hw_volume = NULL;
+ chip->hw_switch = NULL;
+}
+
+static int snd_es1938_reg_bits(struct es1938 *chip, unsigned char reg,
+ unsigned char mask, unsigned char val)
+{
+ if (reg < 0xa0)
+ return snd_es1938_mixer_bits(chip, reg, mask, val);
+ else
+ return snd_es1938_bits(chip, reg, mask, val);
+}
+
+static int snd_es1938_reg_read(struct es1938 *chip, unsigned char reg)
+{
+ if (reg < 0xa0)
+ return snd_es1938_mixer_read(chip, reg);
+ else
+ return snd_es1938_read(chip, reg);
+}
+
+#define ES1938_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,\
+ .name = xname, .index = xindex, \
+ .info = snd_es1938_info_single, \
+ .get = snd_es1938_get_single, .put = snd_es1938_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = xtlv } }
+#define ES1938_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_es1938_info_single, \
+ .get = snd_es1938_get_single, .put = snd_es1938_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_es1938_info_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_es1938_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int val;
+
+ val = snd_es1938_reg_read(chip, reg);
+ ucontrol->value.integer.value[0] = (val >> shift) & mask;
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_es1938_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ unsigned char val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ mask <<= shift;
+ val <<= shift;
+ return snd_es1938_reg_bits(chip, reg, mask, val) != val;
+}
+
+#define ES1938_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,\
+ .name = xname, .index = xindex, \
+ .info = snd_es1938_info_double, \
+ .get = snd_es1938_get_double, .put = snd_es1938_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22), \
+ .tlv = { .p = xtlv } }
+#define ES1938_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_es1938_info_double, \
+ .get = snd_es1938_get_double, .put = snd_es1938_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) }
+
+static int snd_es1938_info_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_es1938_get_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ unsigned char left, right;
+
+ left = snd_es1938_reg_read(chip, left_reg);
+ if (left_reg != right_reg)
+ right = snd_es1938_reg_read(chip, right_reg);
+ else
+ right = left;
+ ucontrol->value.integer.value[0] = (left >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (right >> shift_right) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_es1938_put_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct es1938 *chip = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned char val1, val2, mask1, mask2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ mask1 = mask << shift_left;
+ mask2 = mask << shift_right;
+ if (left_reg != right_reg) {
+ change = 0;
+ if (snd_es1938_reg_bits(chip, left_reg, mask1, val1) != val1)
+ change = 1;
+ if (snd_es1938_reg_bits(chip, right_reg, mask2, val2) != val2)
+ change = 1;
+ } else {
+ change = (snd_es1938_reg_bits(chip, left_reg, mask1 | mask2,
+ val1 | val2) != (val1 | val2));
+ }
+ return change;
+}
+
+static unsigned int db_scale_master[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1),
+ 54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0),
+};
+
+static unsigned int db_scale_audio1[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0),
+};
+
+static unsigned int db_scale_audio2[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0),
+};
+
+static unsigned int db_scale_mic[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(0, 150, 0),
+};
+
+static unsigned int db_scale_line[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0),
+};
+
+static const DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0);
+
+static struct snd_kcontrol_new snd_es1938_controls[] = {
+ES1938_DOUBLE_TLV("Master Playback Volume", 0, 0x60, 0x62, 0, 0, 63, 0,
+ db_scale_master),
+ES1938_DOUBLE("Master Playback Switch", 0, 0x60, 0x62, 6, 6, 1, 1),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Hardware Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_es1938_info_hw_volume,
+ .get = snd_es1938_get_hw_volume,
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Hardware Master Playback Switch",
+ .info = snd_es1938_info_hw_switch,
+ .get = snd_es1938_get_hw_switch,
+ .tlv = { .p = db_scale_master },
+},
+ES1938_SINGLE("Hardware Volume Split", 0, 0x64, 7, 1, 0),
+ES1938_DOUBLE_TLV("Line Playback Volume", 0, 0x3e, 0x3e, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE("CD Playback Volume", 0, 0x38, 0x38, 4, 0, 15, 0),
+ES1938_DOUBLE_TLV("FM Playback Volume", 0, 0x36, 0x36, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("Mic Playback Volume", 0, 0x1a, 0x1a, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0,
+ db_scale_capture),
+ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0),
+ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0),
+ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_es1938_info_mux,
+ .get = snd_es1938_get_mux,
+ .put = snd_es1938_put_mux,
+},
+ES1938_DOUBLE_TLV("Mono Input Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("PCM Capture Volume", 0, 0x69, 0x69, 4, 0, 15, 0,
+ db_scale_audio2),
+ES1938_DOUBLE_TLV("Mic Capture Volume", 0, 0x68, 0x68, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Line Capture Volume", 0, 0x6e, 0x6e, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("FM Capture Volume", 0, 0x6b, 0x6b, 4, 0, 15, 0,
+ db_scale_mic),
+ES1938_DOUBLE_TLV("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("CD Capture Volume", 0, 0x6a, 0x6a, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0,
+ db_scale_line),
+ES1938_DOUBLE_TLV("PCM Playback Volume", 0, 0x7c, 0x7c, 4, 0, 15, 0,
+ db_scale_audio2),
+ES1938_DOUBLE_TLV("PCM Playback Volume", 1, 0x14, 0x14, 4, 0, 15, 0,
+ db_scale_audio1),
+ES1938_SINGLE("3D Control - Level", 0, 0x52, 0, 63, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "3D Control - Switch",
+ .info = snd_es1938_info_spatializer_enable,
+ .get = snd_es1938_get_spatializer_enable,
+ .put = snd_es1938_put_spatializer_enable,
+},
+ES1938_SINGLE("Mic Boost (+26dB)", 0, 0x7d, 3, 1, 0)
+};
+
+
+/* ---------------------------------------------------------------------------- */
+/* ---------------------------------------------------------------------------- */
+
+/*
+ * initialize the chip - used by resume callback, too
+ */
+static void snd_es1938_chip_init(struct es1938 *chip)
+{
+ /* reset chip */
+ snd_es1938_reset(chip);
+
+ /* configure native mode */
+
+ /* enable bus master */
+ pci_set_master(chip->pci);
+
+ /* disable legacy audio */
+ pci_write_config_word(chip->pci, SL_PCI_LEGACYCONTROL, 0x805f);
+
+ /* set DDMA base */
+ pci_write_config_word(chip->pci, SL_PCI_DDMACONTROL, chip->ddma_port | 1);
+
+ /* set DMA/IRQ policy */
+ pci_write_config_dword(chip->pci, SL_PCI_CONFIG, 0);
+
+ /* enable Audio 1, Audio 2, MPU401 IRQ and HW volume IRQ*/
+ outb(0xf0, SLIO_REG(chip, IRQCONTROL));
+
+ /* reset DMA */
+ outb(0, SLDM_REG(chip, DMACLEAR));
+}
+
+#ifdef CONFIG_PM
+/*
+ * PM support
+ */
+
+static unsigned char saved_regs[SAVED_REG_SIZE+1] = {
+ 0x14, 0x1a, 0x1c, 0x3a, 0x3c, 0x3e, 0x36, 0x38,
+ 0x50, 0x52, 0x60, 0x61, 0x62, 0x63, 0x64, 0x68,
+ 0x69, 0x6a, 0x6b, 0x6d, 0x6e, 0x6f, 0x7c, 0x7d,
+ 0xa8, 0xb4,
+};
+
+
+static int es1938_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct es1938 *chip = card->private_data;
+ unsigned char *s, *d;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+
+ /* save mixer-related registers */
+ for (s = saved_regs, d = chip->saved_regs; *s; s++, d++)
+ *d = snd_es1938_reg_read(chip, *s);
+
+ outb(0x00, SLIO_REG(chip, IRQCONTROL)); /* disable irqs */
+ if (chip->irq >= 0) {
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ }
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int es1938_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct es1938 *chip = card->private_data;
+ unsigned char *s, *d;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "es1938: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+
+ if (request_irq(pci->irq, snd_es1938_interrupt,
+ IRQF_SHARED, "ES1938", chip)) {
+ printk(KERN_ERR "es1938: unable to grab IRQ %d, "
+ "disabling device\n", pci->irq);
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ chip->irq = pci->irq;
+ snd_es1938_chip_init(chip);
+
+ /* restore mixer-related registers */
+ for (s = saved_regs, d = chip->saved_regs; *s; s++, d++) {
+ if (*s < 0xa0)
+ snd_es1938_mixer_write(chip, *s, *d);
+ else
+ snd_es1938_write(chip, *s, *d);
+ }
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+#ifdef SUPPORT_JOYSTICK
+static int __devinit snd_es1938_create_gameport(struct es1938 *chip)
+{
+ struct gameport *gp;
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "es1938: cannot allocate memory for gameport\n");
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "ES1938");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+ gp->io = chip->game_port;
+
+ gameport_register_port(gp);
+
+ return 0;
+}
+
+static void snd_es1938_free_gameport(struct es1938 *chip)
+{
+ if (chip->gameport) {
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+ }
+}
+#else
+static inline int snd_es1938_create_gameport(struct es1938 *chip) { return -ENOSYS; }
+static inline void snd_es1938_free_gameport(struct es1938 *chip) { }
+#endif /* SUPPORT_JOYSTICK */
+
+static int snd_es1938_free(struct es1938 *chip)
+{
+ /* disable irqs */
+ outb(0x00, SLIO_REG(chip, IRQCONTROL));
+ if (chip->rmidi)
+ snd_es1938_mixer_bits(chip, ESSSB_IREG_MPU401CONTROL, 0x40, 0);
+
+ snd_es1938_free_gameport(chip);
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_es1938_dev_free(struct snd_device *device)
+{
+ struct es1938 *chip = device->device_data;
+ return snd_es1938_free(chip);
+}
+
+static int __devinit snd_es1938_create(struct snd_card *card,
+ struct pci_dev * pci,
+ struct es1938 ** rchip)
+{
+ struct es1938 *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_es1938_dev_free,
+ };
+
+ *rchip = NULL;
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ /* check, if we can restrict PCI DMA transfers to 24 bits */
+ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->mixer_lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ if ((err = pci_request_regions(pci, "ESS Solo-1")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->io_port = pci_resource_start(pci, 0);
+ chip->sb_port = pci_resource_start(pci, 1);
+ chip->vc_port = pci_resource_start(pci, 2);
+ chip->mpu_port = pci_resource_start(pci, 3);
+ chip->game_port = pci_resource_start(pci, 4);
+ if (request_irq(pci->irq, snd_es1938_interrupt, IRQF_SHARED,
+ "ES1938", chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_es1938_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+#ifdef ES1938_DDEBUG
+ snd_printk(KERN_DEBUG "create: io: 0x%lx, sb: 0x%lx, vc: 0x%lx, mpu: 0x%lx, game: 0x%lx\n",
+ chip->io_port, chip->sb_port, chip->vc_port, chip->mpu_port, chip->game_port);
+#endif
+
+ chip->ddma_port = chip->vc_port + 0x00; /* fix from Thomas Sailer */
+
+ snd_es1938_chip_init(chip);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_es1938_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+}
+
+/* --------------------------------------------------------------------
+ * Interrupt handler
+ * -------------------------------------------------------------------- */
+static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id)
+{
+ struct es1938 *chip = dev_id;
+ unsigned char status, audiostatus;
+ int handled = 0;
+
+ status = inb(SLIO_REG(chip, IRQCONTROL));
+#if 0
+ printk("Es1938debug - interrupt status: =0x%x\n", status);
+#endif
+
+ /* AUDIO 1 */
+ if (status & 0x10) {
+#if 0
+ printk("Es1938debug - AUDIO channel 1 interrupt\n");
+ printk("Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n",
+ inw(SLDM_REG(chip, DMACOUNT)));
+ printk("Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n",
+ inl(SLDM_REG(chip, DMAADDR)));
+ printk("Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n",
+ inl(SLDM_REG(chip, DMASTATUS)));
+#endif
+ /* clear irq */
+ handled = 1;
+ audiostatus = inb(SLSB_REG(chip, STATUS));
+ if (chip->active & ADC1)
+ snd_pcm_period_elapsed(chip->capture_substream);
+ else if (chip->active & DAC1)
+ snd_pcm_period_elapsed(chip->playback2_substream);
+ }
+
+ /* AUDIO 2 */
+ if (status & 0x20) {
+#if 0
+ printk("Es1938debug - AUDIO channel 2 interrupt\n");
+ printk("Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n",
+ inw(SLIO_REG(chip, AUDIO2DMACOUNT)));
+ printk("Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n",
+ inl(SLIO_REG(chip, AUDIO2DMAADDR)));
+
+#endif
+ /* clear irq */
+ handled = 1;
+ snd_es1938_mixer_bits(chip, ESSSB_IREG_AUDIO2CONTROL2, 0x80, 0);
+ if (chip->active & DAC2)
+ snd_pcm_period_elapsed(chip->playback1_substream);
+ }
+
+ /* Hardware volume */
+ if (status & 0x40) {
+ int split = snd_es1938_mixer_read(chip, 0x64) & 0x80;
+ handled = 1;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id);
+ if (!split) {
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_switch->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ }
+ /* ack interrupt */
+ snd_es1938_mixer_write(chip, 0x66, 0x00);
+ }
+
+ /* MPU401 */
+ if (status & 0x80) {
+ // the following line is evil! It switches off MIDI interrupt handling after the first interrupt received.
+ // replacing the last 0 by 0x40 works for ESS-Solo1, but just doing nothing works as well!
+ // andreas@flying-snail.de
+ // snd_es1938_mixer_bits(chip, ESSSB_IREG_MPU401CONTROL, 0x40, 0); /* ack? */
+ if (chip->rmidi) {
+ handled = 1;
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+ }
+ }
+ return IRQ_RETVAL(handled);
+}
+
+#define ES1938_DMA_SIZE 64
+
+static int __devinit snd_es1938_mixer(struct es1938 *chip)
+{
+ struct snd_card *card;
+ unsigned int idx;
+ int err;
+
+ card = chip->card;
+
+ strcpy(card->mixername, "ESS Solo-1");
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_es1938_controls); idx++) {
+ struct snd_kcontrol *kctl;
+ kctl = snd_ctl_new1(&snd_es1938_controls[idx], chip);
+ switch (idx) {
+ case 0:
+ chip->master_volume = kctl;
+ kctl->private_free = snd_es1938_hwv_free;
+ break;
+ case 1:
+ chip->master_switch = kctl;
+ kctl->private_free = snd_es1938_hwv_free;
+ break;
+ case 2:
+ chip->hw_volume = kctl;
+ kctl->private_free = snd_es1938_hwv_free;
+ break;
+ case 3:
+ chip->hw_switch = kctl;
+ kctl->private_free = snd_es1938_hwv_free;
+ break;
+ }
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ return err;
+ }
+ return 0;
+}
+
+
+static int __devinit snd_es1938_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct es1938 *chip;
+ struct snd_opl3 *opl3;
+ int idx, err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ for (idx = 0; idx < 5; idx++) {
+ if (pci_resource_start(pci, idx) == 0 ||
+ !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) {
+ snd_card_free(card);
+ return -ENODEV;
+ }
+ }
+ if ((err = snd_es1938_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ strcpy(card->driver, "ES1938");
+ strcpy(card->shortname, "ESS ES1938 (Solo-1)");
+ sprintf(card->longname, "%s rev %i, irq %i",
+ card->shortname,
+ chip->revision,
+ chip->irq);
+
+ if ((err = snd_es1938_new_pcm(chip, 0)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_es1938_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if (snd_opl3_create(card,
+ SLSB_REG(chip, FMLOWADDR),
+ SLSB_REG(chip, FMHIGHADDR),
+ OPL3_HW_OPL3, 1, &opl3) < 0) {
+ printk(KERN_ERR "es1938: OPL3 not detected at 0x%lx\n",
+ SLSB_REG(chip, FMLOWADDR));
+ } else {
+ if ((err = snd_opl3_timer_new(opl3, 0, 1)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+ if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
+ chip->mpu_port, MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi) < 0) {
+ printk(KERN_ERR "es1938: unable to initialize MPU-401\n");
+ } else {
+ // this line is vital for MIDI interrupt handling on ess-solo1
+ // andreas@flying-snail.de
+ snd_es1938_mixer_bits(chip, ESSSB_IREG_MPU401CONTROL, 0x40, 0x40);
+ }
+
+ snd_es1938_create_gameport(chip);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_es1938_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "ESS ES1938 (Solo-1)",
+ .id_table = snd_es1938_ids,
+ .probe = snd_es1938_probe,
+ .remove = __devexit_p(snd_es1938_remove),
+#ifdef CONFIG_PM
+ .suspend = es1938_suspend,
+ .resume = es1938_resume,
+#endif
+};
+
+static int __init alsa_card_es1938_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_es1938_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_es1938_init)
+module_exit(alsa_card_es1938_exit)
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
new file mode 100644
index 0000000..20ee759
--- /dev/null
+++ b/sound/pci/es1968.c
@@ -0,0 +1,2762 @@
+/*
+ * Driver for ESS Maestro 1/2/2E Sound Card (started 21.8.99)
+ * Copyright (c) by Matze Braun <MatzeBraun@gmx.de>.
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * Most of the driver code comes from Zach Brown(zab@redhat.com)
+ * Alan Cox OSS Driver
+ * Rewritted from card-es1938.c source.
+ *
+ * TODO:
+ * Perhaps Synth
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ * Notes from Zach Brown about the driver code
+ *
+ * Hardware Description
+ *
+ * A working Maestro setup contains the Maestro chip wired to a
+ * codec or 2. In the Maestro we have the APUs, the ASSP, and the
+ * Wavecache. The APUs can be though of as virtual audio routing
+ * channels. They can take data from a number of sources and perform
+ * basic encodings of the data. The wavecache is a storehouse for
+ * PCM data. Typically it deals with PCI and interracts with the
+ * APUs. The ASSP is a wacky DSP like device that ESS is loth
+ * to release docs on. Thankfully it isn't required on the Maestro
+ * until you start doing insane things like FM emulation and surround
+ * encoding. The codecs are almost always AC-97 compliant codecs,
+ * but it appears that early Maestros may have had PT101 (an ESS
+ * part?) wired to them. The only real difference in the Maestro
+ * families is external goop like docking capability, memory for
+ * the ASSP, and initialization differences.
+ *
+ * Driver Operation
+ *
+ * We only drive the APU/Wavecache as typical DACs and drive the
+ * mixers in the codecs. There are 64 APUs. We assign 6 to each
+ * /dev/dsp? device. 2 channels for output, and 4 channels for
+ * input.
+ *
+ * Each APU can do a number of things, but we only really use
+ * 3 basic functions. For playback we use them to convert PCM
+ * data fetched over PCI by the wavecahche into analog data that
+ * is handed to the codec. One APU for mono, and a pair for stereo.
+ * When in stereo, the combination of smarts in the APU and Wavecache
+ * decide which wavecache gets the left or right channel.
+ *
+ * For record we still use the old overly mono system. For each in
+ * coming channel the data comes in from the codec, through a 'input'
+ * APU, through another rate converter APU, and then into memory via
+ * the wavecache and PCI. If its stereo, we mash it back into LRLR in
+ * software. The pass between the 2 APUs is supposedly what requires us
+ * to have a 512 byte buffer sitting around in wavecache/memory.
+ *
+ * The wavecache makes our life even more fun. First off, it can
+ * only address the first 28 bits of PCI address space, making it
+ * useless on quite a few architectures. Secondly, its insane.
+ * It claims to fetch from 4 regions of PCI space, each 4 meg in length.
+ * But that doesn't really work. You can only use 1 region. So all our
+ * allocations have to be in 4meg of each other. Booo. Hiss.
+ * So we have a module parameter, dsps_order, that is the order of
+ * the number of dsps to provide. All their buffer space is allocated
+ * on open time. The sonicvibes OSS routines we inherited really want
+ * power of 2 buffers, so we have all those next to each other, then
+ * 512 byte regions for the recording wavecaches. This ends up
+ * wasting quite a bit of memory. The only fixes I can see would be
+ * getting a kernel allocator that could work in zones, or figuring out
+ * just how to coerce the WP into doing what we want.
+ *
+ * The indirection of the various registers means we have to spinlock
+ * nearly all register accesses. We have the main register indirection
+ * like the wave cache, maestro registers, etc. Then we have beasts
+ * like the APU interface that is indirect registers gotten at through
+ * the main maestro indirection. Ouch. We spinlock around the actual
+ * ports on a per card basis. This means spinlock activity at each IO
+ * operation, but the only IO operation clusters are in non critical
+ * paths and it makes the code far easier to follow. Interrupts are
+ * blocked while holding the locks because the int handler has to
+ * get at some of them :(. The mixer interface doesn't, however.
+ * We also have an OSS state lock that is thrown around in a few
+ * places.
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/mpu401.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+
+#define CARD_NAME "ESS Maestro1/2"
+#define DRIVER_NAME "ES1968"
+
+MODULE_DESCRIPTION("ESS Maestro");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ESS,Maestro 2e},"
+ "{ESS,Maestro 2},"
+ "{ESS,Maestro 1},"
+ "{TerraTec,DMX}}");
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK 1
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 1-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int total_bufsize[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1024 };
+static int pcm_substreams_p[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4 };
+static int pcm_substreams_c[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1 };
+static int clock[SNDRV_CARDS];
+static int use_pm[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
+#ifdef SUPPORT_JOYSTICK
+static int joystick[SNDRV_CARDS];
+#endif
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
+module_param_array(total_bufsize, int, NULL, 0444);
+MODULE_PARM_DESC(total_bufsize, "Total buffer size in kB.");
+module_param_array(pcm_substreams_p, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_substreams_p, "PCM Playback substreams for " CARD_NAME " soundcard.");
+module_param_array(pcm_substreams_c, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_substreams_c, "PCM Capture substreams for " CARD_NAME " soundcard.");
+module_param_array(clock, int, NULL, 0444);
+MODULE_PARM_DESC(clock, "Clock on " CARD_NAME " soundcard. (0 = auto-detect)");
+module_param_array(use_pm, int, NULL, 0444);
+MODULE_PARM_DESC(use_pm, "Toggle power-management. (0 = off, 1 = on, 2 = auto)");
+module_param_array(enable_mpu, int, NULL, 0444);
+MODULE_PARM_DESC(enable_mpu, "Enable MPU401. (0 = off, 1 = on, 2 = auto)");
+#ifdef SUPPORT_JOYSTICK
+module_param_array(joystick, bool, NULL, 0444);
+MODULE_PARM_DESC(joystick, "Enable joystick.");
+#endif
+
+
+#define NR_APUS 64
+#define NR_APU_REGS 16
+
+/* NEC Versas ? */
+#define NEC_VERSA_SUBID1 0x80581033
+#define NEC_VERSA_SUBID2 0x803c1033
+
+/* Mode Flags */
+#define ESS_FMT_STEREO 0x01
+#define ESS_FMT_16BIT 0x02
+
+#define DAC_RUNNING 1
+#define ADC_RUNNING 2
+
+/* Values for the ESM_LEGACY_AUDIO_CONTROL */
+
+#define ESS_DISABLE_AUDIO 0x8000
+#define ESS_ENABLE_SERIAL_IRQ 0x4000
+#define IO_ADRESS_ALIAS 0x0020
+#define MPU401_IRQ_ENABLE 0x0010
+#define MPU401_IO_ENABLE 0x0008
+#define GAME_IO_ENABLE 0x0004
+#define FM_IO_ENABLE 0x0002
+#define SB_IO_ENABLE 0x0001
+
+/* Values for the ESM_CONFIG_A */
+
+#define PIC_SNOOP1 0x4000
+#define PIC_SNOOP2 0x2000
+#define SAFEGUARD 0x0800
+#define DMA_CLEAR 0x0700
+#define DMA_DDMA 0x0000
+#define DMA_TDMA 0x0100
+#define DMA_PCPCI 0x0200
+#define POST_WRITE 0x0080
+#define PCI_TIMING 0x0040
+#define SWAP_LR 0x0020
+#define SUBTR_DECODE 0x0002
+
+/* Values for the ESM_CONFIG_B */
+
+#define SPDIF_CONFB 0x0100
+#define HWV_CONFB 0x0080
+#define DEBOUNCE 0x0040
+#define GPIO_CONFB 0x0020
+#define CHI_CONFB 0x0010
+#define IDMA_CONFB 0x0008 /*undoc */
+#define MIDI_FIX 0x0004 /*undoc */
+#define IRQ_TO_ISA 0x0001 /*undoc */
+
+/* Values for Ring Bus Control B */
+#define RINGB_2CODEC_ID_MASK 0x0003
+#define RINGB_DIS_VALIDATION 0x0008
+#define RINGB_EN_SPDIF 0x0010
+#define RINGB_EN_2CODEC 0x0020
+#define RINGB_SING_BIT_DUAL 0x0040
+
+/* ****Port Adresses**** */
+
+/* Write & Read */
+#define ESM_INDEX 0x02
+#define ESM_DATA 0x00
+
+/* AC97 + RingBus */
+#define ESM_AC97_INDEX 0x30
+#define ESM_AC97_DATA 0x32
+#define ESM_RING_BUS_DEST 0x34
+#define ESM_RING_BUS_CONTR_A 0x36
+#define ESM_RING_BUS_CONTR_B 0x38
+#define ESM_RING_BUS_SDO 0x3A
+
+/* WaveCache*/
+#define WC_INDEX 0x10
+#define WC_DATA 0x12
+#define WC_CONTROL 0x14
+
+/* ASSP*/
+#define ASSP_INDEX 0x80
+#define ASSP_MEMORY 0x82
+#define ASSP_DATA 0x84
+#define ASSP_CONTROL_A 0xA2
+#define ASSP_CONTROL_B 0xA4
+#define ASSP_CONTROL_C 0xA6
+#define ASSP_HOSTW_INDEX 0xA8
+#define ASSP_HOSTW_DATA 0xAA
+#define ASSP_HOSTW_IRQ 0xAC
+/* Midi */
+#define ESM_MPU401_PORT 0x98
+/* Others */
+#define ESM_PORT_HOST_IRQ 0x18
+
+#define IDR0_DATA_PORT 0x00
+#define IDR1_CRAM_POINTER 0x01
+#define IDR2_CRAM_DATA 0x02
+#define IDR3_WAVE_DATA 0x03
+#define IDR4_WAVE_PTR_LOW 0x04
+#define IDR5_WAVE_PTR_HI 0x05
+#define IDR6_TIMER_CTRL 0x06
+#define IDR7_WAVE_ROMRAM 0x07
+
+#define WRITEABLE_MAP 0xEFFFFF
+#define READABLE_MAP 0x64003F
+
+/* PCI Register */
+
+#define ESM_LEGACY_AUDIO_CONTROL 0x40
+#define ESM_ACPI_COMMAND 0x54
+#define ESM_CONFIG_A 0x50
+#define ESM_CONFIG_B 0x52
+#define ESM_DDMA 0x60
+
+/* Bob Bits */
+#define ESM_BOB_ENABLE 0x0001
+#define ESM_BOB_START 0x0001
+
+/* Host IRQ Control Bits */
+#define ESM_RESET_MAESTRO 0x8000
+#define ESM_RESET_DIRECTSOUND 0x4000
+#define ESM_HIRQ_ClkRun 0x0100
+#define ESM_HIRQ_HW_VOLUME 0x0040
+#define ESM_HIRQ_HARPO 0x0030 /* What's that? */
+#define ESM_HIRQ_ASSP 0x0010
+#define ESM_HIRQ_DSIE 0x0004
+#define ESM_HIRQ_MPU401 0x0002
+#define ESM_HIRQ_SB 0x0001
+
+/* Host IRQ Status Bits */
+#define ESM_MPU401_IRQ 0x02
+#define ESM_SB_IRQ 0x01
+#define ESM_SOUND_IRQ 0x04
+#define ESM_ASSP_IRQ 0x10
+#define ESM_HWVOL_IRQ 0x40
+
+#define ESS_SYSCLK 50000000
+#define ESM_BOB_FREQ 200
+#define ESM_BOB_FREQ_MAX 800
+
+#define ESM_FREQ_ESM1 (49152000L / 1024L) /* default rate 48000 */
+#define ESM_FREQ_ESM2 (50000000L / 1024L)
+
+/* APU Modes: reg 0x00, bit 4-7 */
+#define ESM_APU_MODE_SHIFT 4
+#define ESM_APU_MODE_MASK (0xf << 4)
+#define ESM_APU_OFF 0x00
+#define ESM_APU_16BITLINEAR 0x01 /* 16-Bit Linear Sample Player */
+#define ESM_APU_16BITSTEREO 0x02 /* 16-Bit Stereo Sample Player */
+#define ESM_APU_8BITLINEAR 0x03 /* 8-Bit Linear Sample Player */
+#define ESM_APU_8BITSTEREO 0x04 /* 8-Bit Stereo Sample Player */
+#define ESM_APU_8BITDIFF 0x05 /* 8-Bit Differential Sample Playrer */
+#define ESM_APU_DIGITALDELAY 0x06 /* Digital Delay Line */
+#define ESM_APU_DUALTAP 0x07 /* Dual Tap Reader */
+#define ESM_APU_CORRELATOR 0x08 /* Correlator */
+#define ESM_APU_INPUTMIXER 0x09 /* Input Mixer */
+#define ESM_APU_WAVETABLE 0x0A /* Wave Table Mode */
+#define ESM_APU_SRCONVERTOR 0x0B /* Sample Rate Convertor */
+#define ESM_APU_16BITPINGPONG 0x0C /* 16-Bit Ping-Pong Sample Player */
+#define ESM_APU_RESERVED1 0x0D /* Reserved 1 */
+#define ESM_APU_RESERVED2 0x0E /* Reserved 2 */
+#define ESM_APU_RESERVED3 0x0F /* Reserved 3 */
+
+/* reg 0x00 */
+#define ESM_APU_FILTER_Q_SHIFT 0
+#define ESM_APU_FILTER_Q_MASK (3 << 0)
+/* APU Filtey Q Control */
+#define ESM_APU_FILTER_LESSQ 0x00
+#define ESM_APU_FILTER_MOREQ 0x03
+
+#define ESM_APU_FILTER_TYPE_SHIFT 2
+#define ESM_APU_FILTER_TYPE_MASK (3 << 2)
+#define ESM_APU_ENV_TYPE_SHIFT 8
+#define ESM_APU_ENV_TYPE_MASK (3 << 8)
+#define ESM_APU_ENV_STATE_SHIFT 10
+#define ESM_APU_ENV_STATE_MASK (3 << 10)
+#define ESM_APU_END_CURVE (1 << 12)
+#define ESM_APU_INT_ON_LOOP (1 << 13)
+#define ESM_APU_DMA_ENABLE (1 << 14)
+
+/* reg 0x02 */
+#define ESM_APU_SUBMIX_GROUP_SHIRT 0
+#define ESM_APU_SUBMIX_GROUP_MASK (7 << 0)
+#define ESM_APU_SUBMIX_MODE (1 << 3)
+#define ESM_APU_6dB (1 << 4)
+#define ESM_APU_DUAL_EFFECT (1 << 5)
+#define ESM_APU_EFFECT_CHANNELS_SHIFT 6
+#define ESM_APU_EFFECT_CHANNELS_MASK (3 << 6)
+
+/* reg 0x03 */
+#define ESM_APU_STEP_SIZE_MASK 0x0fff
+
+/* reg 0x04 */
+#define ESM_APU_PHASE_SHIFT 0
+#define ESM_APU_PHASE_MASK (0xff << 0)
+#define ESM_APU_WAVE64K_PAGE_SHIFT 8 /* most 8bit of wave start offset */
+#define ESM_APU_WAVE64K_PAGE_MASK (0xff << 8)
+
+/* reg 0x05 - wave start offset */
+/* reg 0x06 - wave end offset */
+/* reg 0x07 - wave loop length */
+
+/* reg 0x08 */
+#define ESM_APU_EFFECT_GAIN_SHIFT 0
+#define ESM_APU_EFFECT_GAIN_MASK (0xff << 0)
+#define ESM_APU_TREMOLO_DEPTH_SHIFT 8
+#define ESM_APU_TREMOLO_DEPTH_MASK (0xf << 8)
+#define ESM_APU_TREMOLO_RATE_SHIFT 12
+#define ESM_APU_TREMOLO_RATE_MASK (0xf << 12)
+
+/* reg 0x09 */
+/* bit 0-7 amplitude dest? */
+#define ESM_APU_AMPLITUDE_NOW_SHIFT 8
+#define ESM_APU_AMPLITUDE_NOW_MASK (0xff << 8)
+
+/* reg 0x0a */
+#define ESM_APU_POLAR_PAN_SHIFT 0
+#define ESM_APU_POLAR_PAN_MASK (0x3f << 0)
+/* Polar Pan Control */
+#define ESM_APU_PAN_CENTER_CIRCLE 0x00
+#define ESM_APU_PAN_MIDDLE_RADIUS 0x01
+#define ESM_APU_PAN_OUTSIDE_RADIUS 0x02
+
+#define ESM_APU_FILTER_TUNING_SHIFT 8
+#define ESM_APU_FILTER_TUNING_MASK (0xff << 8)
+
+/* reg 0x0b */
+#define ESM_APU_DATA_SRC_A_SHIFT 0
+#define ESM_APU_DATA_SRC_A_MASK (0x7f << 0)
+#define ESM_APU_INV_POL_A (1 << 7)
+#define ESM_APU_DATA_SRC_B_SHIFT 8
+#define ESM_APU_DATA_SRC_B_MASK (0x7f << 8)
+#define ESM_APU_INV_POL_B (1 << 15)
+
+#define ESM_APU_VIBRATO_RATE_SHIFT 0
+#define ESM_APU_VIBRATO_RATE_MASK (0xf << 0)
+#define ESM_APU_VIBRATO_DEPTH_SHIFT 4
+#define ESM_APU_VIBRATO_DEPTH_MASK (0xf << 4)
+#define ESM_APU_VIBRATO_PHASE_SHIFT 8
+#define ESM_APU_VIBRATO_PHASE_MASK (0xff << 8)
+
+/* reg 0x0c */
+#define ESM_APU_RADIUS_SELECT (1 << 6)
+
+/* APU Filter Control */
+#define ESM_APU_FILTER_2POLE_LOPASS 0x00
+#define ESM_APU_FILTER_2POLE_BANDPASS 0x01
+#define ESM_APU_FILTER_2POLE_HIPASS 0x02
+#define ESM_APU_FILTER_1POLE_LOPASS 0x03
+#define ESM_APU_FILTER_1POLE_HIPASS 0x04
+#define ESM_APU_FILTER_OFF 0x05
+
+/* APU ATFP Type */
+#define ESM_APU_ATFP_AMPLITUDE 0x00
+#define ESM_APU_ATFP_TREMELO 0x01
+#define ESM_APU_ATFP_FILTER 0x02
+#define ESM_APU_ATFP_PAN 0x03
+
+/* APU ATFP Flags */
+#define ESM_APU_ATFP_FLG_OFF 0x00
+#define ESM_APU_ATFP_FLG_WAIT 0x01
+#define ESM_APU_ATFP_FLG_DONE 0x02
+#define ESM_APU_ATFP_FLG_INPROCESS 0x03
+
+
+/* capture mixing buffer size */
+#define ESM_MEM_ALIGN 0x1000
+#define ESM_MIXBUF_SIZE 0x400
+
+#define ESM_MODE_PLAY 0
+#define ESM_MODE_CAPTURE 1
+
+
+/* APU use in the driver */
+enum snd_enum_apu_type {
+ ESM_APU_PCM_PLAY,
+ ESM_APU_PCM_CAPTURE,
+ ESM_APU_PCM_RATECONV,
+ ESM_APU_FREE
+};
+
+/* chip type */
+enum {
+ TYPE_MAESTRO, TYPE_MAESTRO2, TYPE_MAESTRO2E
+};
+
+/* DMA Hack! */
+struct esm_memory {
+ struct snd_dma_buffer buf;
+ int empty; /* status */
+ struct list_head list;
+};
+
+/* Playback Channel */
+struct esschan {
+ int running;
+
+ u8 apu[4];
+ u8 apu_mode[4];
+
+ /* playback/capture pcm buffer */
+ struct esm_memory *memory;
+ /* capture mixer buffer */
+ struct esm_memory *mixbuf;
+
+ unsigned int hwptr; /* current hw pointer in bytes */
+ unsigned int count; /* sample counter in bytes */
+ unsigned int dma_size; /* total buffer size in bytes */
+ unsigned int frag_size; /* period size in bytes */
+ unsigned int wav_shift;
+ u16 base[4]; /* offset for ptr */
+
+ /* stereo/16bit flag */
+ unsigned char fmt;
+ int mode; /* playback / capture */
+
+ int bob_freq; /* required timer frequency */
+
+ struct snd_pcm_substream *substream;
+
+ /* linked list */
+ struct list_head list;
+
+#ifdef CONFIG_PM
+ u16 wc_map[4];
+#endif
+};
+
+struct es1968 {
+ /* Module Config */
+ int total_bufsize; /* in bytes */
+
+ int playback_streams, capture_streams;
+
+ unsigned int clock; /* clock */
+ /* for clock measurement */
+ unsigned int in_measurement: 1;
+ unsigned int measure_apu;
+ unsigned int measure_lastpos;
+ unsigned int measure_count;
+
+ /* buffer */
+ struct snd_dma_buffer dma;
+
+ /* Resources... */
+ int irq;
+ unsigned long io_port;
+ int type;
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ int do_pm; /* power-management enabled */
+
+ /* DMA memory block */
+ struct list_head buf_list;
+
+ /* ALSA Stuff */
+ struct snd_ac97 *ac97;
+ struct snd_kcontrol *master_switch; /* for h/w volume control */
+ struct snd_kcontrol *master_volume;
+
+ struct snd_rawmidi *rmidi;
+
+ spinlock_t reg_lock;
+ spinlock_t ac97_lock;
+ struct tasklet_struct hwvol_tq;
+ unsigned int in_suspend;
+
+ /* Maestro Stuff */
+ u16 maestro_map[32];
+ int bobclient; /* active timer instancs */
+ int bob_freq; /* timer frequency */
+ struct mutex memory_mutex; /* memory lock */
+
+ /* APU states */
+ unsigned char apu[NR_APUS];
+
+ /* active substreams */
+ struct list_head substream_list;
+ spinlock_t substream_lock;
+
+#ifdef CONFIG_PM
+ u16 apu_map[NR_APUS][NR_APU_REGS];
+#endif
+
+#ifdef SUPPORT_JOYSTICK
+ struct gameport *gameport;
+#endif
+};
+
+static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id);
+
+static struct pci_device_id snd_es1968_ids[] = {
+ /* Maestro 1 */
+ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO },
+ /* Maestro 2 */
+ { 0x125d, 0x1968, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO2 },
+ /* Maestro 2E */
+ { 0x125d, 0x1978, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO2E },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_es1968_ids);
+
+/* *********************
+ * Low Level Funcs! *
+ *********************/
+
+/* no spinlock */
+static void __maestro_write(struct es1968 *chip, u16 reg, u16 data)
+{
+ outw(reg, chip->io_port + ESM_INDEX);
+ outw(data, chip->io_port + ESM_DATA);
+ chip->maestro_map[reg] = data;
+}
+
+static inline void maestro_write(struct es1968 *chip, u16 reg, u16 data)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ __maestro_write(chip, reg, data);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+/* no spinlock */
+static u16 __maestro_read(struct es1968 *chip, u16 reg)
+{
+ if (READABLE_MAP & (1 << reg)) {
+ outw(reg, chip->io_port + ESM_INDEX);
+ chip->maestro_map[reg] = inw(chip->io_port + ESM_DATA);
+ }
+ return chip->maestro_map[reg];
+}
+
+static inline u16 maestro_read(struct es1968 *chip, u16 reg)
+{
+ unsigned long flags;
+ u16 result;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ result = __maestro_read(chip, reg);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return result;
+}
+
+/* Wait for the codec bus to be free */
+static int snd_es1968_ac97_wait(struct es1968 *chip)
+{
+ int timeout = 100000;
+
+ while (timeout-- > 0) {
+ if (!(inb(chip->io_port + ESM_AC97_INDEX) & 1))
+ return 0;
+ cond_resched();
+ }
+ snd_printd("es1968: ac97 timeout\n");
+ return 1; /* timeout */
+}
+
+static int snd_es1968_ac97_wait_poll(struct es1968 *chip)
+{
+ int timeout = 100000;
+
+ while (timeout-- > 0) {
+ if (!(inb(chip->io_port + ESM_AC97_INDEX) & 1))
+ return 0;
+ }
+ snd_printd("es1968: ac97 timeout\n");
+ return 1; /* timeout */
+}
+
+static void snd_es1968_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
+{
+ struct es1968 *chip = ac97->private_data;
+ unsigned long flags;
+
+ snd_es1968_ac97_wait(chip);
+
+ /* Write the bus */
+ spin_lock_irqsave(&chip->ac97_lock, flags);
+ outw(val, chip->io_port + ESM_AC97_DATA);
+ /*msleep(1);*/
+ outb(reg, chip->io_port + ESM_AC97_INDEX);
+ /*msleep(1);*/
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+}
+
+static unsigned short snd_es1968_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ u16 data = 0;
+ struct es1968 *chip = ac97->private_data;
+ unsigned long flags;
+
+ snd_es1968_ac97_wait(chip);
+
+ spin_lock_irqsave(&chip->ac97_lock, flags);
+ outb(reg | 0x80, chip->io_port + ESM_AC97_INDEX);
+ /*msleep(1);*/
+
+ if (!snd_es1968_ac97_wait_poll(chip)) {
+ data = inw(chip->io_port + ESM_AC97_DATA);
+ /*msleep(1);*/
+ }
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+
+ return data;
+}
+
+/* no spinlock */
+static void apu_index_set(struct es1968 *chip, u16 index)
+{
+ int i;
+ __maestro_write(chip, IDR1_CRAM_POINTER, index);
+ for (i = 0; i < 1000; i++)
+ if (__maestro_read(chip, IDR1_CRAM_POINTER) == index)
+ return;
+ snd_printd("es1968: APU register select failed. (Timeout)\n");
+}
+
+/* no spinlock */
+static void apu_data_set(struct es1968 *chip, u16 data)
+{
+ int i;
+ for (i = 0; i < 1000; i++) {
+ if (__maestro_read(chip, IDR0_DATA_PORT) == data)
+ return;
+ __maestro_write(chip, IDR0_DATA_PORT, data);
+ }
+ snd_printd("es1968: APU register set probably failed (Timeout)!\n");
+}
+
+/* no spinlock */
+static void __apu_set_register(struct es1968 *chip, u16 channel, u8 reg, u16 data)
+{
+ if (snd_BUG_ON(channel >= NR_APUS))
+ return;
+#ifdef CONFIG_PM
+ chip->apu_map[channel][reg] = data;
+#endif
+ reg |= (channel << 4);
+ apu_index_set(chip, reg);
+ apu_data_set(chip, data);
+}
+
+static void apu_set_register(struct es1968 *chip, u16 channel, u8 reg, u16 data)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ __apu_set_register(chip, channel, reg, data);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static u16 __apu_get_register(struct es1968 *chip, u16 channel, u8 reg)
+{
+ if (snd_BUG_ON(channel >= NR_APUS))
+ return 0;
+ reg |= (channel << 4);
+ apu_index_set(chip, reg);
+ return __maestro_read(chip, IDR0_DATA_PORT);
+}
+
+static u16 apu_get_register(struct es1968 *chip, u16 channel, u8 reg)
+{
+ unsigned long flags;
+ u16 v;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ v = __apu_get_register(chip, channel, reg);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return v;
+}
+
+#if 0 /* ASSP is not supported */
+
+static void assp_set_register(struct es1968 *chip, u32 reg, u32 value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ outl(reg, chip->io_port + ASSP_INDEX);
+ outl(value, chip->io_port + ASSP_DATA);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static u32 assp_get_register(struct es1968 *chip, u32 reg)
+{
+ unsigned long flags;
+ u32 value;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ outl(reg, chip->io_port + ASSP_INDEX);
+ value = inl(chip->io_port + ASSP_DATA);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ return value;
+}
+
+#endif
+
+static void wave_set_register(struct es1968 *chip, u16 reg, u16 value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ outw(reg, chip->io_port + WC_INDEX);
+ outw(value, chip->io_port + WC_DATA);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static u16 wave_get_register(struct es1968 *chip, u16 reg)
+{
+ unsigned long flags;
+ u16 value;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ outw(reg, chip->io_port + WC_INDEX);
+ value = inw(chip->io_port + WC_DATA);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ return value;
+}
+
+/* *******************
+ * Bob the Timer! *
+ *******************/
+
+static void snd_es1968_bob_stop(struct es1968 *chip)
+{
+ u16 reg;
+
+ reg = __maestro_read(chip, 0x11);
+ reg &= ~ESM_BOB_ENABLE;
+ __maestro_write(chip, 0x11, reg);
+ reg = __maestro_read(chip, 0x17);
+ reg &= ~ESM_BOB_START;
+ __maestro_write(chip, 0x17, reg);
+}
+
+static void snd_es1968_bob_start(struct es1968 *chip)
+{
+ int prescale;
+ int divide;
+
+ /* compute ideal interrupt frequency for buffer size & play rate */
+ /* first, find best prescaler value to match freq */
+ for (prescale = 5; prescale < 12; prescale++)
+ if (chip->bob_freq > (ESS_SYSCLK >> (prescale + 9)))
+ break;
+
+ /* next, back off prescaler whilst getting divider into optimum range */
+ divide = 1;
+ while ((prescale > 5) && (divide < 32)) {
+ prescale--;
+ divide <<= 1;
+ }
+ divide >>= 1;
+
+ /* now fine-tune the divider for best match */
+ for (; divide < 31; divide++)
+ if (chip->bob_freq >
+ ((ESS_SYSCLK >> (prescale + 9)) / (divide + 1))) break;
+
+ /* divide = 0 is illegal, but don't let prescale = 4! */
+ if (divide == 0) {
+ divide++;
+ if (prescale > 5)
+ prescale--;
+ } else if (divide > 1)
+ divide--;
+
+ __maestro_write(chip, 6, 0x9000 | (prescale << 5) | divide); /* set reg */
+
+ /* Now set IDR 11/17 */
+ __maestro_write(chip, 0x11, __maestro_read(chip, 0x11) | 1);
+ __maestro_write(chip, 0x17, __maestro_read(chip, 0x17) | 1);
+}
+
+/* call with substream spinlock */
+static void snd_es1968_bob_inc(struct es1968 *chip, int freq)
+{
+ chip->bobclient++;
+ if (chip->bobclient == 1) {
+ chip->bob_freq = freq;
+ snd_es1968_bob_start(chip);
+ } else if (chip->bob_freq < freq) {
+ snd_es1968_bob_stop(chip);
+ chip->bob_freq = freq;
+ snd_es1968_bob_start(chip);
+ }
+}
+
+/* call with substream spinlock */
+static void snd_es1968_bob_dec(struct es1968 *chip)
+{
+ chip->bobclient--;
+ if (chip->bobclient <= 0)
+ snd_es1968_bob_stop(chip);
+ else if (chip->bob_freq > ESM_BOB_FREQ) {
+ /* check reduction of timer frequency */
+ int max_freq = ESM_BOB_FREQ;
+ struct esschan *es;
+ list_for_each_entry(es, &chip->substream_list, list) {
+ if (max_freq < es->bob_freq)
+ max_freq = es->bob_freq;
+ }
+ if (max_freq != chip->bob_freq) {
+ snd_es1968_bob_stop(chip);
+ chip->bob_freq = max_freq;
+ snd_es1968_bob_start(chip);
+ }
+ }
+}
+
+static int
+snd_es1968_calc_bob_rate(struct es1968 *chip, struct esschan *es,
+ struct snd_pcm_runtime *runtime)
+{
+ /* we acquire 4 interrupts per period for precise control.. */
+ int freq = runtime->rate * 4;
+ if (es->fmt & ESS_FMT_STEREO)
+ freq <<= 1;
+ if (es->fmt & ESS_FMT_16BIT)
+ freq <<= 1;
+ freq /= es->frag_size;
+ if (freq < ESM_BOB_FREQ)
+ freq = ESM_BOB_FREQ;
+ else if (freq > ESM_BOB_FREQ_MAX)
+ freq = ESM_BOB_FREQ_MAX;
+ return freq;
+}
+
+
+/*************
+ * PCM Part *
+ *************/
+
+static u32 snd_es1968_compute_rate(struct es1968 *chip, u32 freq)
+{
+ u32 rate = (freq << 16) / chip->clock;
+#if 0 /* XXX: do we need this? */
+ if (rate > 0x10000)
+ rate = 0x10000;
+#endif
+ return rate;
+}
+
+/* get current pointer */
+static inline unsigned int
+snd_es1968_get_dma_ptr(struct es1968 *chip, struct esschan *es)
+{
+ unsigned int offset;
+
+ offset = apu_get_register(chip, es->apu[0], 5);
+
+ offset -= es->base[0];
+
+ return (offset & 0xFFFE); /* hardware is in words */
+}
+
+static void snd_es1968_apu_set_freq(struct es1968 *chip, int apu, int freq)
+{
+ apu_set_register(chip, apu, 2,
+ (apu_get_register(chip, apu, 2) & 0x00FF) |
+ ((freq & 0xff) << 8) | 0x10);
+ apu_set_register(chip, apu, 3, freq >> 8);
+}
+
+/* spin lock held */
+static inline void snd_es1968_trigger_apu(struct es1968 *esm, int apu, int mode)
+{
+ /* set the APU mode */
+ __apu_set_register(esm, apu, 0,
+ (__apu_get_register(esm, apu, 0) & 0xff0f) |
+ (mode << 4));
+}
+
+static void snd_es1968_pcm_start(struct es1968 *chip, struct esschan *es)
+{
+ spin_lock(&chip->reg_lock);
+ __apu_set_register(chip, es->apu[0], 5, es->base[0]);
+ snd_es1968_trigger_apu(chip, es->apu[0], es->apu_mode[0]);
+ if (es->mode == ESM_MODE_CAPTURE) {
+ __apu_set_register(chip, es->apu[2], 5, es->base[2]);
+ snd_es1968_trigger_apu(chip, es->apu[2], es->apu_mode[2]);
+ }
+ if (es->fmt & ESS_FMT_STEREO) {
+ __apu_set_register(chip, es->apu[1], 5, es->base[1]);
+ snd_es1968_trigger_apu(chip, es->apu[1], es->apu_mode[1]);
+ if (es->mode == ESM_MODE_CAPTURE) {
+ __apu_set_register(chip, es->apu[3], 5, es->base[3]);
+ snd_es1968_trigger_apu(chip, es->apu[3], es->apu_mode[3]);
+ }
+ }
+ spin_unlock(&chip->reg_lock);
+}
+
+static void snd_es1968_pcm_stop(struct es1968 *chip, struct esschan *es)
+{
+ spin_lock(&chip->reg_lock);
+ snd_es1968_trigger_apu(chip, es->apu[0], 0);
+ snd_es1968_trigger_apu(chip, es->apu[1], 0);
+ if (es->mode == ESM_MODE_CAPTURE) {
+ snd_es1968_trigger_apu(chip, es->apu[2], 0);
+ snd_es1968_trigger_apu(chip, es->apu[3], 0);
+ }
+ spin_unlock(&chip->reg_lock);
+}
+
+/* set the wavecache control reg */
+static void snd_es1968_program_wavecache(struct es1968 *chip, struct esschan *es,
+ int channel, u32 addr, int capture)
+{
+ u32 tmpval = (addr - 0x10) & 0xFFF8;
+
+ if (! capture) {
+ if (!(es->fmt & ESS_FMT_16BIT))
+ tmpval |= 4; /* 8bit */
+ if (es->fmt & ESS_FMT_STEREO)
+ tmpval |= 2; /* stereo */
+ }
+
+ /* set the wavecache control reg */
+ wave_set_register(chip, es->apu[channel] << 3, tmpval);
+
+#ifdef CONFIG_PM
+ es->wc_map[channel] = tmpval;
+#endif
+}
+
+
+static void snd_es1968_playback_setup(struct es1968 *chip, struct esschan *es,
+ struct snd_pcm_runtime *runtime)
+{
+ u32 pa;
+ int high_apu = 0;
+ int channel, apu;
+ int i, size;
+ unsigned long flags;
+ u32 freq;
+
+ size = es->dma_size >> es->wav_shift;
+
+ if (es->fmt & ESS_FMT_STEREO)
+ high_apu++;
+
+ for (channel = 0; channel <= high_apu; channel++) {
+ apu = es->apu[channel];
+
+ snd_es1968_program_wavecache(chip, es, channel, es->memory->buf.addr, 0);
+
+ /* Offset to PCMBAR */
+ pa = es->memory->buf.addr;
+ pa -= chip->dma.addr;
+ pa >>= 1; /* words */
+
+ pa |= 0x00400000; /* System RAM (Bit 22) */
+
+ if (es->fmt & ESS_FMT_STEREO) {
+ /* Enable stereo */
+ if (channel)
+ pa |= 0x00800000; /* (Bit 23) */
+ if (es->fmt & ESS_FMT_16BIT)
+ pa >>= 1;
+ }
+
+ /* base offset of dma calcs when reading the pointer
+ on this left one */
+ es->base[channel] = pa & 0xFFFF;
+
+ for (i = 0; i < 16; i++)
+ apu_set_register(chip, apu, i, 0x0000);
+
+ /* Load the buffer into the wave engine */
+ apu_set_register(chip, apu, 4, ((pa >> 16) & 0xFF) << 8);
+ apu_set_register(chip, apu, 5, pa & 0xFFFF);
+ apu_set_register(chip, apu, 6, (pa + size) & 0xFFFF);
+ /* setting loop == sample len */
+ apu_set_register(chip, apu, 7, size);
+
+ /* clear effects/env.. */
+ apu_set_register(chip, apu, 8, 0x0000);
+ /* set amp now to 0xd0 (?), low byte is 'amplitude dest'? */
+ apu_set_register(chip, apu, 9, 0xD000);
+
+ /* clear routing stuff */
+ apu_set_register(chip, apu, 11, 0x0000);
+ /* dma on, no envelopes, filter to all 1s) */
+ apu_set_register(chip, apu, 0, 0x400F);
+
+ if (es->fmt & ESS_FMT_16BIT)
+ es->apu_mode[channel] = ESM_APU_16BITLINEAR;
+ else
+ es->apu_mode[channel] = ESM_APU_8BITLINEAR;
+
+ if (es->fmt & ESS_FMT_STEREO) {
+ /* set panning: left or right */
+ /* Check: different panning. On my Canyon 3D Chipset the
+ Channels are swapped. I don't know, about the output
+ to the SPDif Link. Perhaps you have to change this
+ and not the APU Regs 4-5. */
+ apu_set_register(chip, apu, 10,
+ 0x8F00 | (channel ? 0 : 0x10));
+ es->apu_mode[channel] += 1; /* stereo */
+ } else
+ apu_set_register(chip, apu, 10, 0x8F08);
+ }
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ /* clear WP interrupts */
+ outw(1, chip->io_port + 0x04);
+ /* enable WP ints */
+ outw(inw(chip->io_port + ESM_PORT_HOST_IRQ) | ESM_HIRQ_DSIE, chip->io_port + ESM_PORT_HOST_IRQ);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ freq = runtime->rate;
+ /* set frequency */
+ if (freq > 48000)
+ freq = 48000;
+ if (freq < 4000)
+ freq = 4000;
+
+ /* hmmm.. */
+ if (!(es->fmt & ESS_FMT_16BIT) && !(es->fmt & ESS_FMT_STEREO))
+ freq >>= 1;
+
+ freq = snd_es1968_compute_rate(chip, freq);
+
+ /* Load the frequency, turn on 6dB */
+ snd_es1968_apu_set_freq(chip, es->apu[0], freq);
+ snd_es1968_apu_set_freq(chip, es->apu[1], freq);
+}
+
+
+static void init_capture_apu(struct es1968 *chip, struct esschan *es, int channel,
+ unsigned int pa, unsigned int bsize,
+ int mode, int route)
+{
+ int i, apu = es->apu[channel];
+
+ es->apu_mode[channel] = mode;
+
+ /* set the wavecache control reg */
+ snd_es1968_program_wavecache(chip, es, channel, pa, 1);
+
+ /* Offset to PCMBAR */
+ pa -= chip->dma.addr;
+ pa >>= 1; /* words */
+
+ /* base offset of dma calcs when reading the pointer
+ on this left one */
+ es->base[channel] = pa & 0xFFFF;
+ pa |= 0x00400000; /* bit 22 -> System RAM */
+
+ /* Begin loading the APU */
+ for (i = 0; i < 16; i++)
+ apu_set_register(chip, apu, i, 0x0000);
+
+ /* need to enable subgroups.. and we should probably
+ have different groups for different /dev/dsps.. */
+ apu_set_register(chip, apu, 2, 0x8);
+
+ /* Load the buffer into the wave engine */
+ apu_set_register(chip, apu, 4, ((pa >> 16) & 0xFF) << 8);
+ apu_set_register(chip, apu, 5, pa & 0xFFFF);
+ apu_set_register(chip, apu, 6, (pa + bsize) & 0xFFFF);
+ apu_set_register(chip, apu, 7, bsize);
+ /* clear effects/env.. */
+ apu_set_register(chip, apu, 8, 0x00F0);
+ /* amplitude now? sure. why not. */
+ apu_set_register(chip, apu, 9, 0x0000);
+ /* set filter tune, radius, polar pan */
+ apu_set_register(chip, apu, 10, 0x8F08);
+ /* route input */
+ apu_set_register(chip, apu, 11, route);
+ /* dma on, no envelopes, filter to all 1s) */
+ apu_set_register(chip, apu, 0, 0x400F);
+}
+
+static void snd_es1968_capture_setup(struct es1968 *chip, struct esschan *es,
+ struct snd_pcm_runtime *runtime)
+{
+ int size;
+ u32 freq;
+ unsigned long flags;
+
+ size = es->dma_size >> es->wav_shift;
+
+ /* APU assignments:
+ 0 = mono/left SRC
+ 1 = right SRC
+ 2 = mono/left Input Mixer
+ 3 = right Input Mixer
+ */
+ /* data seems to flow from the codec, through an apu into
+ the 'mixbuf' bit of page, then through the SRC apu
+ and out to the real 'buffer'. ok. sure. */
+
+ /* input mixer (left/mono) */
+ /* parallel in crap, see maestro reg 0xC [8-11] */
+ init_capture_apu(chip, es, 2,
+ es->mixbuf->buf.addr, ESM_MIXBUF_SIZE/4, /* in words */
+ ESM_APU_INPUTMIXER, 0x14);
+ /* SRC (left/mono); get input from inputing apu */
+ init_capture_apu(chip, es, 0, es->memory->buf.addr, size,
+ ESM_APU_SRCONVERTOR, es->apu[2]);
+ if (es->fmt & ESS_FMT_STEREO) {
+ /* input mixer (right) */
+ init_capture_apu(chip, es, 3,
+ es->mixbuf->buf.addr + ESM_MIXBUF_SIZE/2,
+ ESM_MIXBUF_SIZE/4, /* in words */
+ ESM_APU_INPUTMIXER, 0x15);
+ /* SRC (right) */
+ init_capture_apu(chip, es, 1,
+ es->memory->buf.addr + size*2, size,
+ ESM_APU_SRCONVERTOR, es->apu[3]);
+ }
+
+ freq = runtime->rate;
+ /* Sample Rate conversion APUs don't like 0x10000 for their rate */
+ if (freq > 47999)
+ freq = 47999;
+ if (freq < 4000)
+ freq = 4000;
+
+ freq = snd_es1968_compute_rate(chip, freq);
+
+ /* Load the frequency, turn on 6dB */
+ snd_es1968_apu_set_freq(chip, es->apu[0], freq);
+ snd_es1968_apu_set_freq(chip, es->apu[1], freq);
+
+ /* fix mixer rate at 48khz. and its _must_ be 0x10000. */
+ freq = 0x10000;
+ snd_es1968_apu_set_freq(chip, es->apu[2], freq);
+ snd_es1968_apu_set_freq(chip, es->apu[3], freq);
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ /* clear WP interrupts */
+ outw(1, chip->io_port + 0x04);
+ /* enable WP ints */
+ outw(inw(chip->io_port + ESM_PORT_HOST_IRQ) | ESM_HIRQ_DSIE, chip->io_port + ESM_PORT_HOST_IRQ);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+/*******************
+ * ALSA Interface *
+ *******************/
+
+static int snd_es1968_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct esschan *es = runtime->private_data;
+
+ es->dma_size = snd_pcm_lib_buffer_bytes(substream);
+ es->frag_size = snd_pcm_lib_period_bytes(substream);
+
+ es->wav_shift = 1; /* maestro handles always 16bit */
+ es->fmt = 0;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ es->fmt |= ESS_FMT_16BIT;
+ if (runtime->channels > 1) {
+ es->fmt |= ESS_FMT_STEREO;
+ if (es->fmt & ESS_FMT_16BIT) /* 8bit is already word shifted */
+ es->wav_shift++;
+ }
+ es->bob_freq = snd_es1968_calc_bob_rate(chip, es, runtime);
+
+ switch (es->mode) {
+ case ESM_MODE_PLAY:
+ snd_es1968_playback_setup(chip, es, runtime);
+ break;
+ case ESM_MODE_CAPTURE:
+ snd_es1968_capture_setup(chip, es, runtime);
+ break;
+ }
+
+ return 0;
+}
+
+static int snd_es1968_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct esschan *es = substream->runtime->private_data;
+
+ spin_lock(&chip->substream_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (es->running)
+ break;
+ snd_es1968_bob_inc(chip, es->bob_freq);
+ es->count = 0;
+ es->hwptr = 0;
+ snd_es1968_pcm_start(chip, es);
+ es->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (! es->running)
+ break;
+ snd_es1968_pcm_stop(chip, es);
+ es->running = 0;
+ snd_es1968_bob_dec(chip);
+ break;
+ }
+ spin_unlock(&chip->substream_lock);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_es1968_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct esschan *es = substream->runtime->private_data;
+ unsigned int ptr;
+
+ ptr = snd_es1968_get_dma_ptr(chip, es) << es->wav_shift;
+
+ return bytes_to_frames(substream->runtime, ptr % es->dma_size);
+}
+
+static struct snd_pcm_hardware snd_es1968_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ /*SNDRV_PCM_INFO_PAUSE |*/
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 256,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_es1968_capture = {
+ .info = (SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ /*SNDRV_PCM_INFO_PAUSE |*/
+ SNDRV_PCM_INFO_RESUME),
+ .formats = /*SNDRV_PCM_FMTBIT_U8 |*/ SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 65536,
+ .period_bytes_min = 256,
+ .period_bytes_max = 65536,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/* *************************
+ * DMA memory management *
+ *************************/
+
+/* Because the Maestro can only take addresses relative to the PCM base address
+ register :( */
+
+static int calc_available_memory_size(struct es1968 *chip)
+{
+ int max_size = 0;
+ struct esm_memory *buf;
+
+ mutex_lock(&chip->memory_mutex);
+ list_for_each_entry(buf, &chip->buf_list, list) {
+ if (buf->empty && buf->buf.bytes > max_size)
+ max_size = buf->buf.bytes;
+ }
+ mutex_unlock(&chip->memory_mutex);
+ if (max_size >= 128*1024)
+ max_size = 127*1024;
+ return max_size;
+}
+
+/* allocate a new memory chunk with the specified size */
+static struct esm_memory *snd_es1968_new_memory(struct es1968 *chip, int size)
+{
+ struct esm_memory *buf;
+
+ size = ALIGN(size, ESM_MEM_ALIGN);
+ mutex_lock(&chip->memory_mutex);
+ list_for_each_entry(buf, &chip->buf_list, list) {
+ if (buf->empty && buf->buf.bytes >= size)
+ goto __found;
+ }
+ mutex_unlock(&chip->memory_mutex);
+ return NULL;
+
+__found:
+ if (buf->buf.bytes > size) {
+ struct esm_memory *chunk = kmalloc(sizeof(*chunk), GFP_KERNEL);
+ if (chunk == NULL) {
+ mutex_unlock(&chip->memory_mutex);
+ return NULL;
+ }
+ chunk->buf = buf->buf;
+ chunk->buf.bytes -= size;
+ chunk->buf.area += size;
+ chunk->buf.addr += size;
+ chunk->empty = 1;
+ buf->buf.bytes = size;
+ list_add(&chunk->list, &buf->list);
+ }
+ buf->empty = 0;
+ mutex_unlock(&chip->memory_mutex);
+ return buf;
+}
+
+/* free a memory chunk */
+static void snd_es1968_free_memory(struct es1968 *chip, struct esm_memory *buf)
+{
+ struct esm_memory *chunk;
+
+ mutex_lock(&chip->memory_mutex);
+ buf->empty = 1;
+ if (buf->list.prev != &chip->buf_list) {
+ chunk = list_entry(buf->list.prev, struct esm_memory, list);
+ if (chunk->empty) {
+ chunk->buf.bytes += buf->buf.bytes;
+ list_del(&buf->list);
+ kfree(buf);
+ buf = chunk;
+ }
+ }
+ if (buf->list.next != &chip->buf_list) {
+ chunk = list_entry(buf->list.next, struct esm_memory, list);
+ if (chunk->empty) {
+ buf->buf.bytes += chunk->buf.bytes;
+ list_del(&chunk->list);
+ kfree(chunk);
+ }
+ }
+ mutex_unlock(&chip->memory_mutex);
+}
+
+static void snd_es1968_free_dmabuf(struct es1968 *chip)
+{
+ struct list_head *p;
+
+ if (! chip->dma.area)
+ return;
+ snd_dma_reserve_buf(&chip->dma, snd_dma_pci_buf_id(chip->pci));
+ while ((p = chip->buf_list.next) != &chip->buf_list) {
+ struct esm_memory *chunk = list_entry(p, struct esm_memory, list);
+ list_del(p);
+ kfree(chunk);
+ }
+}
+
+static int __devinit
+snd_es1968_init_dmabuf(struct es1968 *chip)
+{
+ int err;
+ struct esm_memory *chunk;
+
+ chip->dma.dev.type = SNDRV_DMA_TYPE_DEV;
+ chip->dma.dev.dev = snd_dma_pci_data(chip->pci);
+ if (! snd_dma_get_reserved_buf(&chip->dma, snd_dma_pci_buf_id(chip->pci))) {
+ err = snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ chip->total_bufsize, &chip->dma);
+ if (err < 0 || ! chip->dma.area) {
+ snd_printk(KERN_ERR "es1968: can't allocate dma pages for size %d\n",
+ chip->total_bufsize);
+ return -ENOMEM;
+ }
+ if ((chip->dma.addr + chip->dma.bytes - 1) & ~((1 << 28) - 1)) {
+ snd_dma_free_pages(&chip->dma);
+ snd_printk(KERN_ERR "es1968: DMA buffer beyond 256MB.\n");
+ return -ENOMEM;
+ }
+ }
+
+ INIT_LIST_HEAD(&chip->buf_list);
+ /* allocate an empty chunk */
+ chunk = kmalloc(sizeof(*chunk), GFP_KERNEL);
+ if (chunk == NULL) {
+ snd_es1968_free_dmabuf(chip);
+ return -ENOMEM;
+ }
+ memset(chip->dma.area, 0, ESM_MEM_ALIGN);
+ chunk->buf = chip->dma;
+ chunk->buf.area += ESM_MEM_ALIGN;
+ chunk->buf.addr += ESM_MEM_ALIGN;
+ chunk->buf.bytes -= ESM_MEM_ALIGN;
+ chunk->empty = 1;
+ list_add(&chunk->list, &chip->buf_list);
+
+ return 0;
+}
+
+/* setup the dma_areas */
+/* buffer is extracted from the pre-allocated memory chunk */
+static int snd_es1968_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct esschan *chan = runtime->private_data;
+ int size = params_buffer_bytes(hw_params);
+
+ if (chan->memory) {
+ if (chan->memory->buf.bytes >= size) {
+ runtime->dma_bytes = size;
+ return 0;
+ }
+ snd_es1968_free_memory(chip, chan->memory);
+ }
+ chan->memory = snd_es1968_new_memory(chip, size);
+ if (chan->memory == NULL) {
+ // snd_printd("cannot allocate dma buffer: size = %d\n", size);
+ return -ENOMEM;
+ }
+ snd_pcm_set_runtime_buffer(substream, &chan->memory->buf);
+ return 1; /* area was changed */
+}
+
+/* remove dma areas if allocated */
+static int snd_es1968_hw_free(struct snd_pcm_substream *substream)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct esschan *chan;
+
+ if (runtime->private_data == NULL)
+ return 0;
+ chan = runtime->private_data;
+ if (chan->memory) {
+ snd_es1968_free_memory(chip, chan->memory);
+ chan->memory = NULL;
+ }
+ return 0;
+}
+
+
+/*
+ * allocate APU pair
+ */
+static int snd_es1968_alloc_apu_pair(struct es1968 *chip, int type)
+{
+ int apu;
+
+ for (apu = 0; apu < NR_APUS; apu += 2) {
+ if (chip->apu[apu] == ESM_APU_FREE &&
+ chip->apu[apu + 1] == ESM_APU_FREE) {
+ chip->apu[apu] = chip->apu[apu + 1] = type;
+ return apu;
+ }
+ }
+ return -EBUSY;
+}
+
+/*
+ * release APU pair
+ */
+static void snd_es1968_free_apu_pair(struct es1968 *chip, int apu)
+{
+ chip->apu[apu] = chip->apu[apu + 1] = ESM_APU_FREE;
+}
+
+
+/******************
+ * PCM open/close *
+ ******************/
+
+static int snd_es1968_playback_open(struct snd_pcm_substream *substream)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct esschan *es;
+ int apu1;
+
+ /* search 2 APUs */
+ apu1 = snd_es1968_alloc_apu_pair(chip, ESM_APU_PCM_PLAY);
+ if (apu1 < 0)
+ return apu1;
+
+ es = kzalloc(sizeof(*es), GFP_KERNEL);
+ if (!es) {
+ snd_es1968_free_apu_pair(chip, apu1);
+ return -ENOMEM;
+ }
+
+ es->apu[0] = apu1;
+ es->apu[1] = apu1 + 1;
+ es->apu_mode[0] = 0;
+ es->apu_mode[1] = 0;
+ es->running = 0;
+ es->substream = substream;
+ es->mode = ESM_MODE_PLAY;
+
+ runtime->private_data = es;
+ runtime->hw = snd_es1968_playback;
+ runtime->hw.buffer_bytes_max = runtime->hw.period_bytes_max =
+ calc_available_memory_size(chip);
+
+ spin_lock_irq(&chip->substream_lock);
+ list_add(&es->list, &chip->substream_list);
+ spin_unlock_irq(&chip->substream_lock);
+
+ return 0;
+}
+
+static int snd_es1968_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct esschan *es;
+ int apu1, apu2;
+
+ apu1 = snd_es1968_alloc_apu_pair(chip, ESM_APU_PCM_CAPTURE);
+ if (apu1 < 0)
+ return apu1;
+ apu2 = snd_es1968_alloc_apu_pair(chip, ESM_APU_PCM_RATECONV);
+ if (apu2 < 0) {
+ snd_es1968_free_apu_pair(chip, apu1);
+ return apu2;
+ }
+
+ es = kzalloc(sizeof(*es), GFP_KERNEL);
+ if (!es) {
+ snd_es1968_free_apu_pair(chip, apu1);
+ snd_es1968_free_apu_pair(chip, apu2);
+ return -ENOMEM;
+ }
+
+ es->apu[0] = apu1;
+ es->apu[1] = apu1 + 1;
+ es->apu[2] = apu2;
+ es->apu[3] = apu2 + 1;
+ es->apu_mode[0] = 0;
+ es->apu_mode[1] = 0;
+ es->apu_mode[2] = 0;
+ es->apu_mode[3] = 0;
+ es->running = 0;
+ es->substream = substream;
+ es->mode = ESM_MODE_CAPTURE;
+
+ /* get mixbuffer */
+ if ((es->mixbuf = snd_es1968_new_memory(chip, ESM_MIXBUF_SIZE)) == NULL) {
+ snd_es1968_free_apu_pair(chip, apu1);
+ snd_es1968_free_apu_pair(chip, apu2);
+ kfree(es);
+ return -ENOMEM;
+ }
+ memset(es->mixbuf->buf.area, 0, ESM_MIXBUF_SIZE);
+
+ runtime->private_data = es;
+ runtime->hw = snd_es1968_capture;
+ runtime->hw.buffer_bytes_max = runtime->hw.period_bytes_max =
+ calc_available_memory_size(chip) - 1024; /* keep MIXBUF size */
+ snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES);
+
+ spin_lock_irq(&chip->substream_lock);
+ list_add(&es->list, &chip->substream_list);
+ spin_unlock_irq(&chip->substream_lock);
+
+ return 0;
+}
+
+static int snd_es1968_playback_close(struct snd_pcm_substream *substream)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct esschan *es;
+
+ if (substream->runtime->private_data == NULL)
+ return 0;
+ es = substream->runtime->private_data;
+ spin_lock_irq(&chip->substream_lock);
+ list_del(&es->list);
+ spin_unlock_irq(&chip->substream_lock);
+ snd_es1968_free_apu_pair(chip, es->apu[0]);
+ kfree(es);
+
+ return 0;
+}
+
+static int snd_es1968_capture_close(struct snd_pcm_substream *substream)
+{
+ struct es1968 *chip = snd_pcm_substream_chip(substream);
+ struct esschan *es;
+
+ if (substream->runtime->private_data == NULL)
+ return 0;
+ es = substream->runtime->private_data;
+ spin_lock_irq(&chip->substream_lock);
+ list_del(&es->list);
+ spin_unlock_irq(&chip->substream_lock);
+ snd_es1968_free_memory(chip, es->mixbuf);
+ snd_es1968_free_apu_pair(chip, es->apu[0]);
+ snd_es1968_free_apu_pair(chip, es->apu[2]);
+ kfree(es);
+
+ return 0;
+}
+
+static struct snd_pcm_ops snd_es1968_playback_ops = {
+ .open = snd_es1968_playback_open,
+ .close = snd_es1968_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_es1968_hw_params,
+ .hw_free = snd_es1968_hw_free,
+ .prepare = snd_es1968_pcm_prepare,
+ .trigger = snd_es1968_pcm_trigger,
+ .pointer = snd_es1968_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_es1968_capture_ops = {
+ .open = snd_es1968_capture_open,
+ .close = snd_es1968_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_es1968_hw_params,
+ .hw_free = snd_es1968_hw_free,
+ .prepare = snd_es1968_pcm_prepare,
+ .trigger = snd_es1968_pcm_trigger,
+ .pointer = snd_es1968_pcm_pointer,
+};
+
+
+/*
+ * measure clock
+ */
+#define CLOCK_MEASURE_BUFSIZE 16768 /* enough large for a single shot */
+
+static void __devinit es1968_measure_clock(struct es1968 *chip)
+{
+ int i, apu;
+ unsigned int pa, offset, t;
+ struct esm_memory *memory;
+ struct timeval start_time, stop_time;
+
+ if (chip->clock == 0)
+ chip->clock = 48000; /* default clock value */
+
+ /* search 2 APUs (although one apu is enough) */
+ if ((apu = snd_es1968_alloc_apu_pair(chip, ESM_APU_PCM_PLAY)) < 0) {
+ snd_printk(KERN_ERR "Hmm, cannot find empty APU pair!?\n");
+ return;
+ }
+ if ((memory = snd_es1968_new_memory(chip, CLOCK_MEASURE_BUFSIZE)) == NULL) {
+ snd_printk(KERN_ERR "cannot allocate dma buffer - using default clock %d\n", chip->clock);
+ snd_es1968_free_apu_pair(chip, apu);
+ return;
+ }
+
+ memset(memory->buf.area, 0, CLOCK_MEASURE_BUFSIZE);
+
+ wave_set_register(chip, apu << 3, (memory->buf.addr - 0x10) & 0xfff8);
+
+ pa = (unsigned int)((memory->buf.addr - chip->dma.addr) >> 1);
+ pa |= 0x00400000; /* System RAM (Bit 22) */
+
+ /* initialize apu */
+ for (i = 0; i < 16; i++)
+ apu_set_register(chip, apu, i, 0x0000);
+
+ apu_set_register(chip, apu, 0, 0x400f);
+ apu_set_register(chip, apu, 4, ((pa >> 16) & 0xff) << 8);
+ apu_set_register(chip, apu, 5, pa & 0xffff);
+ apu_set_register(chip, apu, 6, (pa + CLOCK_MEASURE_BUFSIZE/2) & 0xffff);
+ apu_set_register(chip, apu, 7, CLOCK_MEASURE_BUFSIZE/2);
+ apu_set_register(chip, apu, 8, 0x0000);
+ apu_set_register(chip, apu, 9, 0xD000);
+ apu_set_register(chip, apu, 10, 0x8F08);
+ apu_set_register(chip, apu, 11, 0x0000);
+ spin_lock_irq(&chip->reg_lock);
+ outw(1, chip->io_port + 0x04); /* clear WP interrupts */
+ outw(inw(chip->io_port + ESM_PORT_HOST_IRQ) | ESM_HIRQ_DSIE, chip->io_port + ESM_PORT_HOST_IRQ); /* enable WP ints */
+ spin_unlock_irq(&chip->reg_lock);
+
+ snd_es1968_apu_set_freq(chip, apu, ((unsigned int)48000 << 16) / chip->clock); /* 48000 Hz */
+
+ chip->in_measurement = 1;
+ chip->measure_apu = apu;
+ spin_lock_irq(&chip->reg_lock);
+ snd_es1968_bob_inc(chip, ESM_BOB_FREQ);
+ __apu_set_register(chip, apu, 5, pa & 0xffff);
+ snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR);
+ do_gettimeofday(&start_time);
+ spin_unlock_irq(&chip->reg_lock);
+ msleep(50);
+ spin_lock_irq(&chip->reg_lock);
+ offset = __apu_get_register(chip, apu, 5);
+ do_gettimeofday(&stop_time);
+ snd_es1968_trigger_apu(chip, apu, 0); /* stop */
+ snd_es1968_bob_dec(chip);
+ chip->in_measurement = 0;
+ spin_unlock_irq(&chip->reg_lock);
+
+ /* check the current position */
+ offset -= (pa & 0xffff);
+ offset &= 0xfffe;
+ offset += chip->measure_count * (CLOCK_MEASURE_BUFSIZE/2);
+
+ t = stop_time.tv_sec - start_time.tv_sec;
+ t *= 1000000;
+ if (stop_time.tv_usec < start_time.tv_usec)
+ t -= start_time.tv_usec - stop_time.tv_usec;
+ else
+ t += stop_time.tv_usec - start_time.tv_usec;
+ if (t == 0) {
+ snd_printk(KERN_ERR "?? calculation error..\n");
+ } else {
+ offset *= 1000;
+ offset = (offset / t) * 1000 + ((offset % t) * 1000) / t;
+ if (offset < 47500 || offset > 48500) {
+ if (offset >= 40000 && offset <= 50000)
+ chip->clock = (chip->clock * offset) / 48000;
+ }
+ printk(KERN_INFO "es1968: clocking to %d\n", chip->clock);
+ }
+ snd_es1968_free_memory(chip, memory);
+ snd_es1968_free_apu_pair(chip, apu);
+}
+
+
+/*
+ */
+
+static void snd_es1968_pcm_free(struct snd_pcm *pcm)
+{
+ struct es1968 *esm = pcm->private_data;
+ snd_es1968_free_dmabuf(esm);
+ esm->pcm = NULL;
+}
+
+static int __devinit
+snd_es1968_pcm(struct es1968 *chip, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ /* get DMA buffer */
+ if ((err = snd_es1968_init_dmabuf(chip)) < 0)
+ return err;
+
+ /* set PCMBAR */
+ wave_set_register(chip, 0x01FC, chip->dma.addr >> 12);
+ wave_set_register(chip, 0x01FD, chip->dma.addr >> 12);
+ wave_set_register(chip, 0x01FE, chip->dma.addr >> 12);
+ wave_set_register(chip, 0x01FF, chip->dma.addr >> 12);
+
+ if ((err = snd_pcm_new(chip->card, "ESS Maestro", device,
+ chip->playback_streams,
+ chip->capture_streams, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = chip;
+ pcm->private_free = snd_es1968_pcm_free;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_es1968_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_es1968_capture_ops);
+
+ pcm->info_flags = 0;
+
+ strcpy(pcm->name, "ESS Maestro");
+
+ chip->pcm = pcm;
+
+ return 0;
+}
+/*
+ * suppress jitter on some maestros when playing stereo
+ */
+static void snd_es1968_suppress_jitter(struct es1968 *chip, struct esschan *es)
+{
+ unsigned int cp1;
+ unsigned int cp2;
+ unsigned int diff;
+
+ cp1 = __apu_get_register(chip, 0, 5);
+ cp2 = __apu_get_register(chip, 1, 5);
+ diff = (cp1 > cp2 ? cp1 - cp2 : cp2 - cp1);
+
+ if (diff > 1)
+ __maestro_write(chip, IDR0_DATA_PORT, cp1);
+}
+
+/*
+ * update pointer
+ */
+static void snd_es1968_update_pcm(struct es1968 *chip, struct esschan *es)
+{
+ unsigned int hwptr;
+ unsigned int diff;
+ struct snd_pcm_substream *subs = es->substream;
+
+ if (subs == NULL || !es->running)
+ return;
+
+ hwptr = snd_es1968_get_dma_ptr(chip, es) << es->wav_shift;
+ hwptr %= es->dma_size;
+
+ diff = (es->dma_size + hwptr - es->hwptr) % es->dma_size;
+
+ es->hwptr = hwptr;
+ es->count += diff;
+
+ if (es->count > es->frag_size) {
+ spin_unlock(&chip->substream_lock);
+ snd_pcm_period_elapsed(subs);
+ spin_lock(&chip->substream_lock);
+ es->count %= es->frag_size;
+ }
+}
+
+/*
+ */
+static void es1968_update_hw_volume(unsigned long private_data)
+{
+ struct es1968 *chip = (struct es1968 *) private_data;
+ int x, val;
+ unsigned long flags;
+
+ /* Figure out which volume control button was pushed,
+ based on differences from the default register
+ values. */
+ x = inb(chip->io_port + 0x1c) & 0xee;
+ /* Reset the volume control registers. */
+ outb(0x88, chip->io_port + 0x1c);
+ outb(0x88, chip->io_port + 0x1d);
+ outb(0x88, chip->io_port + 0x1e);
+ outb(0x88, chip->io_port + 0x1f);
+
+ if (chip->in_suspend)
+ return;
+
+ if (! chip->master_switch || ! chip->master_volume)
+ return;
+
+ /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
+ spin_lock_irqsave(&chip->ac97_lock, flags);
+ val = chip->ac97->regs[AC97_MASTER];
+ switch (x) {
+ case 0x88:
+ /* mute */
+ val ^= 0x8000;
+ chip->ac97->regs[AC97_MASTER] = val;
+ outw(val, chip->io_port + ESM_AC97_DATA);
+ outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_switch->id);
+ break;
+ case 0xaa:
+ /* volume up */
+ if ((val & 0x7f) > 0)
+ val--;
+ if ((val & 0x7f00) > 0)
+ val -= 0x0100;
+ chip->ac97->regs[AC97_MASTER] = val;
+ outw(val, chip->io_port + ESM_AC97_DATA);
+ outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ break;
+ case 0x66:
+ /* volume down */
+ if ((val & 0x7f) < 0x1f)
+ val++;
+ if ((val & 0x7f00) < 0x1f00)
+ val += 0x0100;
+ chip->ac97->regs[AC97_MASTER] = val;
+ outw(val, chip->io_port + ESM_AC97_DATA);
+ outb(AC97_MASTER, chip->io_port + ESM_AC97_INDEX);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ break;
+ }
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+}
+
+/*
+ * interrupt handler
+ */
+static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
+{
+ struct es1968 *chip = dev_id;
+ u32 event;
+
+ if (!(event = inb(chip->io_port + 0x1A)))
+ return IRQ_NONE;
+
+ outw(inw(chip->io_port + 4) & 1, chip->io_port + 4);
+
+ if (event & ESM_HWVOL_IRQ)
+ tasklet_hi_schedule(&chip->hwvol_tq); /* we'll do this later */
+
+ /* else ack 'em all, i imagine */
+ outb(0xFF, chip->io_port + 0x1A);
+
+ if ((event & ESM_MPU401_IRQ) && chip->rmidi) {
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+ }
+
+ if (event & ESM_SOUND_IRQ) {
+ struct esschan *es;
+ spin_lock(&chip->substream_lock);
+ list_for_each_entry(es, &chip->substream_list, list) {
+ if (es->running) {
+ snd_es1968_update_pcm(chip, es);
+ if (es->fmt & ESS_FMT_STEREO)
+ snd_es1968_suppress_jitter(chip, es);
+ }
+ }
+ spin_unlock(&chip->substream_lock);
+ if (chip->in_measurement) {
+ unsigned int curp = __apu_get_register(chip, chip->measure_apu, 5);
+ if (curp < chip->measure_lastpos)
+ chip->measure_count++;
+ chip->measure_lastpos = curp;
+ }
+ }
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * Mixer stuff
+ */
+
+static int __devinit
+snd_es1968_mixer(struct es1968 *chip)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ struct snd_ctl_elem_id elem_id;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_es1968_ac97_write,
+ .read = snd_es1968_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+ pbus->no_vra = 1; /* ES1968 doesn't need VRA */
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97)) < 0)
+ return err;
+
+ /* attach master switch / volumes for h/w volume control */
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Switch");
+ chip->master_switch = snd_ctl_find_id(chip->card, &elem_id);
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Volume");
+ chip->master_volume = snd_ctl_find_id(chip->card, &elem_id);
+
+ return 0;
+}
+
+/*
+ * reset ac97 codec
+ */
+
+static void snd_es1968_ac97_reset(struct es1968 *chip)
+{
+ unsigned long ioaddr = chip->io_port;
+
+ unsigned short save_ringbus_a;
+ unsigned short save_68;
+ unsigned short w;
+ unsigned int vend;
+
+ /* save configuration */
+ save_ringbus_a = inw(ioaddr + 0x36);
+
+ //outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38); /* clear second codec id? */
+ /* set command/status address i/o to 1st codec */
+ outw(inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a);
+ outw(inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c);
+
+ /* disable ac link */
+ outw(0x0000, ioaddr + 0x36);
+ save_68 = inw(ioaddr + 0x68);
+ pci_read_config_word(chip->pci, 0x58, &w); /* something magical with gpio and bus arb. */
+ pci_read_config_dword(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, &vend);
+ if (w & 1)
+ save_68 |= 0x10;
+ outw(0xfffe, ioaddr + 0x64); /* unmask gpio 0 */
+ outw(0x0001, ioaddr + 0x68); /* gpio write */
+ outw(0x0000, ioaddr + 0x60); /* write 0 to gpio 0 */
+ udelay(20);
+ outw(0x0001, ioaddr + 0x60); /* write 1 to gpio 1 */
+ msleep(20);
+
+ outw(save_68 | 0x1, ioaddr + 0x68); /* now restore .. */
+ outw((inw(ioaddr + 0x38) & 0xfffc) | 0x1, ioaddr + 0x38);
+ outw((inw(ioaddr + 0x3a) & 0xfffc) | 0x1, ioaddr + 0x3a);
+ outw((inw(ioaddr + 0x3c) & 0xfffc) | 0x1, ioaddr + 0x3c);
+
+ /* now the second codec */
+ /* disable ac link */
+ outw(0x0000, ioaddr + 0x36);
+ outw(0xfff7, ioaddr + 0x64); /* unmask gpio 3 */
+ save_68 = inw(ioaddr + 0x68);
+ outw(0x0009, ioaddr + 0x68); /* gpio write 0 & 3 ?? */
+ outw(0x0001, ioaddr + 0x60); /* write 1 to gpio */
+ udelay(20);
+ outw(0x0009, ioaddr + 0x60); /* write 9 to gpio */
+ msleep(500);
+ //outw(inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38);
+ outw(inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a);
+ outw(inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c);
+
+#if 0 /* the loop here needs to be much better if we want it.. */
+ snd_printk(KERN_INFO "trying software reset\n");
+ /* try and do a software reset */
+ outb(0x80 | 0x7c, ioaddr + 0x30);
+ for (w = 0;; w++) {
+ if ((inw(ioaddr + 0x30) & 1) == 0) {
+ if (inb(ioaddr + 0x32) != 0)
+ break;
+
+ outb(0x80 | 0x7d, ioaddr + 0x30);
+ if (((inw(ioaddr + 0x30) & 1) == 0)
+ && (inb(ioaddr + 0x32) != 0))
+ break;
+ outb(0x80 | 0x7f, ioaddr + 0x30);
+ if (((inw(ioaddr + 0x30) & 1) == 0)
+ && (inb(ioaddr + 0x32) != 0))
+ break;
+ }
+
+ if (w > 10000) {
+ outb(inb(ioaddr + 0x37) | 0x08, ioaddr + 0x37); /* do a software reset */
+ msleep(500); /* oh my.. */
+ outb(inb(ioaddr + 0x37) & ~0x08,
+ ioaddr + 0x37);
+ udelay(1);
+ outw(0x80, ioaddr + 0x30);
+ for (w = 0; w < 10000; w++) {
+ if ((inw(ioaddr + 0x30) & 1) == 0)
+ break;
+ }
+ }
+ }
+#endif
+ if (vend == NEC_VERSA_SUBID1 || vend == NEC_VERSA_SUBID2) {
+ /* turn on external amp? */
+ outw(0xf9ff, ioaddr + 0x64);
+ outw(inw(ioaddr + 0x68) | 0x600, ioaddr + 0x68);
+ outw(0x0209, ioaddr + 0x60);
+ }
+
+ /* restore.. */
+ outw(save_ringbus_a, ioaddr + 0x36);
+
+ /* Turn on the 978 docking chip.
+ First frob the "master output enable" bit,
+ then set most of the playback volume control registers to max. */
+ outb(inb(ioaddr+0xc0)|(1<<5), ioaddr+0xc0);
+ outb(0xff, ioaddr+0xc3);
+ outb(0xff, ioaddr+0xc4);
+ outb(0xff, ioaddr+0xc6);
+ outb(0xff, ioaddr+0xc8);
+ outb(0x3f, ioaddr+0xcf);
+ outb(0x3f, ioaddr+0xd0);
+}
+
+static void snd_es1968_reset(struct es1968 *chip)
+{
+ /* Reset */
+ outw(ESM_RESET_MAESTRO | ESM_RESET_DIRECTSOUND,
+ chip->io_port + ESM_PORT_HOST_IRQ);
+ udelay(10);
+ outw(0x0000, chip->io_port + ESM_PORT_HOST_IRQ);
+ udelay(10);
+}
+
+/*
+ * initialize maestro chip
+ */
+static void snd_es1968_chip_init(struct es1968 *chip)
+{
+ struct pci_dev *pci = chip->pci;
+ int i;
+ unsigned long iobase = chip->io_port;
+ u16 w;
+ u32 n;
+
+ /* We used to muck around with pci config space that
+ * we had no business messing with. We don't know enough
+ * about the machine to know which DMA mode is appropriate,
+ * etc. We were guessing wrong on some machines and making
+ * them unhappy. We now trust in the BIOS to do things right,
+ * which almost certainly means a new host of problems will
+ * arise with broken BIOS implementations. screw 'em.
+ * We're already intolerant of machines that don't assign
+ * IRQs.
+ */
+
+ /* Config Reg A */
+ pci_read_config_word(pci, ESM_CONFIG_A, &w);
+
+ w &= ~DMA_CLEAR; /* Clear DMA bits */
+ w &= ~(PIC_SNOOP1 | PIC_SNOOP2); /* Clear Pic Snoop Mode Bits */
+ w &= ~SAFEGUARD; /* Safeguard off */
+ w |= POST_WRITE; /* Posted write */
+ w |= PCI_TIMING; /* PCI timing on */
+ /* XXX huh? claims to be reserved.. */
+ w &= ~SWAP_LR; /* swap left/right
+ seems to only have effect on SB
+ Emulation */
+ w &= ~SUBTR_DECODE; /* Subtractive decode off */
+
+ pci_write_config_word(pci, ESM_CONFIG_A, w);
+
+ /* Config Reg B */
+
+ pci_read_config_word(pci, ESM_CONFIG_B, &w);
+
+ w &= ~(1 << 15); /* Turn off internal clock multiplier */
+ /* XXX how do we know which to use? */
+ w &= ~(1 << 14); /* External clock */
+
+ w &= ~SPDIF_CONFB; /* disable S/PDIF output */
+ w |= HWV_CONFB; /* HWV on */
+ w |= DEBOUNCE; /* Debounce off: easier to push the HW buttons */
+ w &= ~GPIO_CONFB; /* GPIO 4:5 */
+ w |= CHI_CONFB; /* Disconnect from the CHI. Enabling this made a dell 7500 work. */
+ w &= ~IDMA_CONFB; /* IDMA off (undocumented) */
+ w &= ~MIDI_FIX; /* MIDI fix off (undoc) */
+ w &= ~(1 << 1); /* reserved, always write 0 */
+ w &= ~IRQ_TO_ISA; /* IRQ to ISA off (undoc) */
+
+ pci_write_config_word(pci, ESM_CONFIG_B, w);
+
+ /* DDMA off */
+
+ pci_read_config_word(pci, ESM_DDMA, &w);
+ w &= ~(1 << 0);
+ pci_write_config_word(pci, ESM_DDMA, w);
+
+ /*
+ * Legacy mode
+ */
+
+ pci_read_config_word(pci, ESM_LEGACY_AUDIO_CONTROL, &w);
+
+ w |= ESS_DISABLE_AUDIO; /* Disable Legacy Audio */
+ w &= ~ESS_ENABLE_SERIAL_IRQ; /* Disable SIRQ */
+ w &= ~(0x1f); /* disable mpu irq/io, game port, fm, SB */
+
+ pci_write_config_word(pci, ESM_LEGACY_AUDIO_CONTROL, w);
+
+ /* Set up 978 docking control chip. */
+ pci_read_config_word(pci, 0x58, &w);
+ w|=1<<2; /* Enable 978. */
+ w|=1<<3; /* Turn on 978 hardware volume control. */
+ w&=~(1<<11); /* Turn on 978 mixer volume control. */
+ pci_write_config_word(pci, 0x58, w);
+
+ /* Sound Reset */
+
+ snd_es1968_reset(chip);
+
+ /*
+ * Ring Bus Setup
+ */
+
+ /* setup usual 0x34 stuff.. 0x36 may be chip specific */
+ outw(0xC090, iobase + ESM_RING_BUS_DEST); /* direct sound, stereo */
+ udelay(20);
+ outw(0x3000, iobase + ESM_RING_BUS_CONTR_A); /* enable ringbus/serial */
+ udelay(20);
+
+ /*
+ * Reset the CODEC
+ */
+
+ snd_es1968_ac97_reset(chip);
+
+ /* Ring Bus Control B */
+
+ n = inl(iobase + ESM_RING_BUS_CONTR_B);
+ n &= ~RINGB_EN_SPDIF; /* SPDIF off */
+ //w |= RINGB_EN_2CODEC; /* enable 2nd codec */
+ outl(n, iobase + ESM_RING_BUS_CONTR_B);
+
+ /* Set hardware volume control registers to midpoints.
+ We can tell which button was pushed based on how they change. */
+ outb(0x88, iobase+0x1c);
+ outb(0x88, iobase+0x1d);
+ outb(0x88, iobase+0x1e);
+ outb(0x88, iobase+0x1f);
+
+ /* it appears some maestros (dell 7500) only work if these are set,
+ regardless of wether we use the assp or not. */
+
+ outb(0, iobase + ASSP_CONTROL_B);
+ outb(3, iobase + ASSP_CONTROL_A); /* M: Reserved bits... */
+ outb(0, iobase + ASSP_CONTROL_C); /* M: Disable ASSP, ASSP IRQ's and FM Port */
+
+ /*
+ * set up wavecache
+ */
+ for (i = 0; i < 16; i++) {
+ /* Write 0 into the buffer area 0x1E0->1EF */
+ outw(0x01E0 + i, iobase + WC_INDEX);
+ outw(0x0000, iobase + WC_DATA);
+
+ /* The 1.10 test program seem to write 0 into the buffer area
+ * 0x1D0-0x1DF too.*/
+ outw(0x01D0 + i, iobase + WC_INDEX);
+ outw(0x0000, iobase + WC_DATA);
+ }
+ wave_set_register(chip, IDR7_WAVE_ROMRAM,
+ (wave_get_register(chip, IDR7_WAVE_ROMRAM) & 0xFF00));
+ wave_set_register(chip, IDR7_WAVE_ROMRAM,
+ wave_get_register(chip, IDR7_WAVE_ROMRAM) | 0x100);
+ wave_set_register(chip, IDR7_WAVE_ROMRAM,
+ wave_get_register(chip, IDR7_WAVE_ROMRAM) & ~0x200);
+ wave_set_register(chip, IDR7_WAVE_ROMRAM,
+ wave_get_register(chip, IDR7_WAVE_ROMRAM) | ~0x400);
+
+
+ maestro_write(chip, IDR2_CRAM_DATA, 0x0000);
+ /* Now back to the DirectSound stuff */
+ /* audio serial configuration.. ? */
+ maestro_write(chip, 0x08, 0xB004);
+ maestro_write(chip, 0x09, 0x001B);
+ maestro_write(chip, 0x0A, 0x8000);
+ maestro_write(chip, 0x0B, 0x3F37);
+ maestro_write(chip, 0x0C, 0x0098);
+
+ /* parallel in, has something to do with recording :) */
+ maestro_write(chip, 0x0C,
+ (maestro_read(chip, 0x0C) & ~0xF000) | 0x8000);
+ /* parallel out */
+ maestro_write(chip, 0x0C,
+ (maestro_read(chip, 0x0C) & ~0x0F00) | 0x0500);
+
+ maestro_write(chip, 0x0D, 0x7632);
+
+ /* Wave cache control on - test off, sg off,
+ enable, enable extra chans 1Mb */
+
+ w = inw(iobase + WC_CONTROL);
+
+ w &= ~0xFA00; /* Seems to be reserved? I don't know */
+ w |= 0xA000; /* reserved... I don't know */
+ w &= ~0x0200; /* Channels 56,57,58,59 as Extra Play,Rec Channel enable
+ Seems to crash the Computer if enabled... */
+ w |= 0x0100; /* Wave Cache Operation Enabled */
+ w |= 0x0080; /* Channels 60/61 as Placback/Record enabled */
+ w &= ~0x0060; /* Clear Wavtable Size */
+ w |= 0x0020; /* Wavetable Size : 1MB */
+ /* Bit 4 is reserved */
+ w &= ~0x000C; /* DMA Stuff? I don't understand what the datasheet means */
+ /* Bit 1 is reserved */
+ w &= ~0x0001; /* Test Mode off */
+
+ outw(w, iobase + WC_CONTROL);
+
+ /* Now clear the APU control ram */
+ for (i = 0; i < NR_APUS; i++) {
+ for (w = 0; w < NR_APU_REGS; w++)
+ apu_set_register(chip, i, w, 0);
+
+ }
+}
+
+/* Enable IRQ's */
+static void snd_es1968_start_irq(struct es1968 *chip)
+{
+ unsigned short w;
+ w = ESM_HIRQ_DSIE | ESM_HIRQ_HW_VOLUME;
+ if (chip->rmidi)
+ w |= ESM_HIRQ_MPU401;
+ outw(w, chip->io_port + ESM_PORT_HOST_IRQ);
+}
+
+#ifdef CONFIG_PM
+/*
+ * PM support
+ */
+static int es1968_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct es1968 *chip = card->private_data;
+
+ if (! chip->do_pm)
+ return 0;
+
+ chip->in_suspend = 1;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_ac97_suspend(chip->ac97);
+ snd_es1968_bob_stop(chip);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int es1968_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct es1968 *chip = card->private_data;
+ struct esschan *es;
+
+ if (! chip->do_pm)
+ return 0;
+
+ /* restore all our config */
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "es1968: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_es1968_chip_init(chip);
+
+ /* need to restore the base pointers.. */
+ if (chip->dma.addr) {
+ /* set PCMBAR */
+ wave_set_register(chip, 0x01FC, chip->dma.addr >> 12);
+ }
+
+ snd_es1968_start_irq(chip);
+
+ /* restore ac97 state */
+ snd_ac97_resume(chip->ac97);
+
+ list_for_each_entry(es, &chip->substream_list, list) {
+ switch (es->mode) {
+ case ESM_MODE_PLAY:
+ snd_es1968_playback_setup(chip, es, es->substream->runtime);
+ break;
+ case ESM_MODE_CAPTURE:
+ snd_es1968_capture_setup(chip, es, es->substream->runtime);
+ break;
+ }
+ }
+
+ /* start timer again */
+ if (chip->bobclient)
+ snd_es1968_bob_start(chip);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ chip->in_suspend = 0;
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+#ifdef SUPPORT_JOYSTICK
+#define JOYSTICK_ADDR 0x200
+static int __devinit snd_es1968_create_gameport(struct es1968 *chip, int dev)
+{
+ struct gameport *gp;
+ struct resource *r;
+ u16 val;
+
+ if (!joystick[dev])
+ return -ENODEV;
+
+ r = request_region(JOYSTICK_ADDR, 8, "ES1968 gameport");
+ if (!r)
+ return -EBUSY;
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "es1968: cannot allocate memory for gameport\n");
+ release_and_free_resource(r);
+ return -ENOMEM;
+ }
+
+ pci_read_config_word(chip->pci, ESM_LEGACY_AUDIO_CONTROL, &val);
+ pci_write_config_word(chip->pci, ESM_LEGACY_AUDIO_CONTROL, val | 0x04);
+
+ gameport_set_name(gp, "ES1968 Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+ gp->io = JOYSTICK_ADDR;
+ gameport_set_port_data(gp, r);
+
+ gameport_register_port(gp);
+
+ return 0;
+}
+
+static void snd_es1968_free_gameport(struct es1968 *chip)
+{
+ if (chip->gameport) {
+ struct resource *r = gameport_get_port_data(chip->gameport);
+
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+
+ release_and_free_resource(r);
+ }
+}
+#else
+static inline int snd_es1968_create_gameport(struct es1968 *chip, int dev) { return -ENOSYS; }
+static inline void snd_es1968_free_gameport(struct es1968 *chip) { }
+#endif
+
+static int snd_es1968_free(struct es1968 *chip)
+{
+ if (chip->io_port) {
+ if (chip->irq >= 0)
+ synchronize_irq(chip->irq);
+ outw(1, chip->io_port + 0x04); /* clear WP interrupts */
+ outw(0, chip->io_port + ESM_PORT_HOST_IRQ); /* disable IRQ */
+ }
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ snd_es1968_free_gameport(chip);
+ chip->master_switch = NULL;
+ chip->master_volume = NULL;
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_es1968_dev_free(struct snd_device *device)
+{
+ struct es1968 *chip = device->device_data;
+ return snd_es1968_free(chip);
+}
+
+struct ess_device_list {
+ unsigned short type; /* chip type */
+ unsigned short vendor; /* subsystem vendor id */
+};
+
+static struct ess_device_list pm_whitelist[] __devinitdata = {
+ { TYPE_MAESTRO2E, 0x0e11 }, /* Compaq Armada */
+ { TYPE_MAESTRO2E, 0x1028 },
+ { TYPE_MAESTRO2E, 0x103c },
+ { TYPE_MAESTRO2E, 0x1179 },
+ { TYPE_MAESTRO2E, 0x14c0 }, /* HP omnibook 4150 */
+ { TYPE_MAESTRO2E, 0x1558 },
+};
+
+static struct ess_device_list mpu_blacklist[] __devinitdata = {
+ { TYPE_MAESTRO2, 0x125d },
+};
+
+static int __devinit snd_es1968_create(struct snd_card *card,
+ struct pci_dev *pci,
+ int total_bufsize,
+ int play_streams,
+ int capt_streams,
+ int chip_type,
+ int do_pm,
+ struct es1968 **chip_ret)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_es1968_dev_free,
+ };
+ struct es1968 *chip;
+ int i, err;
+
+ *chip_ret = NULL;
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ /* check, if we can restrict PCI DMA transfers to 28 bits */
+ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (! chip) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ /* Set Vars */
+ chip->type = chip_type;
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->substream_lock);
+ INIT_LIST_HEAD(&chip->buf_list);
+ INIT_LIST_HEAD(&chip->substream_list);
+ spin_lock_init(&chip->ac97_lock);
+ mutex_init(&chip->memory_mutex);
+ tasklet_init(&chip->hwvol_tq, es1968_update_hw_volume, (unsigned long)chip);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->total_bufsize = total_bufsize; /* in bytes */
+ chip->playback_streams = play_streams;
+ chip->capture_streams = capt_streams;
+
+ if ((err = pci_request_regions(pci, "ESS Maestro")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->io_port = pci_resource_start(pci, 0);
+ if (request_irq(pci->irq, snd_es1968_interrupt, IRQF_SHARED,
+ "ESS Maestro", chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_es1968_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ /* Clear Maestro_map */
+ for (i = 0; i < 32; i++)
+ chip->maestro_map[i] = 0;
+
+ /* Clear Apu Map */
+ for (i = 0; i < NR_APUS; i++)
+ chip->apu[i] = ESM_APU_FREE;
+
+ /* just to be sure */
+ pci_set_master(pci);
+
+ if (do_pm > 1) {
+ /* disable power-management if not on the whitelist */
+ unsigned short vend;
+ pci_read_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, &vend);
+ for (i = 0; i < (int)ARRAY_SIZE(pm_whitelist); i++) {
+ if (chip->type == pm_whitelist[i].type &&
+ vend == pm_whitelist[i].vendor) {
+ do_pm = 1;
+ break;
+ }
+ }
+ if (do_pm > 1) {
+ /* not matched; disabling pm */
+ printk(KERN_INFO "es1968: not attempting power management.\n");
+ do_pm = 0;
+ }
+ }
+ chip->do_pm = do_pm;
+
+ snd_es1968_chip_init(chip);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_es1968_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *chip_ret = chip;
+
+ return 0;
+}
+
+
+/*
+ */
+static int __devinit snd_es1968_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct es1968 *chip;
+ unsigned int i;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (!card)
+ return -ENOMEM;
+
+ if (total_bufsize[dev] < 128)
+ total_bufsize[dev] = 128;
+ if (total_bufsize[dev] > 4096)
+ total_bufsize[dev] = 4096;
+ if ((err = snd_es1968_create(card, pci,
+ total_bufsize[dev] * 1024, /* in bytes */
+ pcm_substreams_p[dev],
+ pcm_substreams_c[dev],
+ pci_id->driver_data,
+ use_pm[dev],
+ &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ switch (chip->type) {
+ case TYPE_MAESTRO2E:
+ strcpy(card->driver, "ES1978");
+ strcpy(card->shortname, "ESS ES1978 (Maestro 2E)");
+ break;
+ case TYPE_MAESTRO2:
+ strcpy(card->driver, "ES1968");
+ strcpy(card->shortname, "ESS ES1968 (Maestro 2)");
+ break;
+ case TYPE_MAESTRO:
+ strcpy(card->driver, "ESM1");
+ strcpy(card->shortname, "ESS Maestro 1");
+ break;
+ }
+
+ if ((err = snd_es1968_pcm(chip, 0)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_es1968_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if (enable_mpu[dev] == 2) {
+ /* check the black list */
+ unsigned short vend;
+ pci_read_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, &vend);
+ for (i = 0; i < ARRAY_SIZE(mpu_blacklist); i++) {
+ if (chip->type == mpu_blacklist[i].type &&
+ vend == mpu_blacklist[i].vendor) {
+ enable_mpu[dev] = 0;
+ break;
+ }
+ }
+ }
+ if (enable_mpu[dev]) {
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
+ chip->io_port + ESM_MPU401_PORT,
+ MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi)) < 0) {
+ printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n");
+ }
+ }
+
+ snd_es1968_create_gameport(chip, dev);
+
+ snd_es1968_start_irq(chip);
+
+ chip->clock = clock[dev];
+ if (! chip->clock)
+ es1968_measure_clock(chip);
+
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, chip->io_port, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_es1968_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "ES1968 (ESS Maestro)",
+ .id_table = snd_es1968_ids,
+ .probe = snd_es1968_probe,
+ .remove = __devexit_p(snd_es1968_remove),
+#ifdef CONFIG_PM
+ .suspend = es1968_suspend,
+ .resume = es1968_resume,
+#endif
+};
+
+static int __init alsa_card_es1968_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_es1968_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_es1968_init)
+module_exit(alsa_card_es1968_exit)
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
new file mode 100644
index 0000000..c129f9e
--- /dev/null
+++ b/sound/pci/fm801.c
@@ -0,0 +1,1608 @@
+/*
+ * The driver for the ForteMedia FM801 based soundcards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ * Support FM only card by Andy Shevchenko <andy@smile.org.ua>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include <sound/ac97_codec.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+
+#ifdef CONFIG_SND_FM801_TEA575X_BOOL
+#include <sound/tea575x-tuner.h>
+#define TEA575X_RADIO 1
+#endif
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("ForteMedia FM801");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801},"
+ "{Genius,SoundMaker Live 5.1}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+/*
+ * Enable TEA575x tuner
+ * 1 = MediaForte 256-PCS
+ * 2 = MediaForte 256-PCPR
+ * 3 = MediaForte 64-PCR
+ * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card
+ * High 16-bits are video (radio) device number + 1
+ */
+static int tea575x_tuner[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the FM801 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the FM801 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
+module_param_array(tea575x_tuner, int, NULL, 0444);
+MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner.");
+
+/*
+ * Direct registers
+ */
+
+#define FM801_REG(chip, reg) (chip->port + FM801_##reg)
+
+#define FM801_PCM_VOL 0x00 /* PCM Output Volume */
+#define FM801_FM_VOL 0x02 /* FM Output Volume */
+#define FM801_I2S_VOL 0x04 /* I2S Volume */
+#define FM801_REC_SRC 0x06 /* Record Source */
+#define FM801_PLY_CTRL 0x08 /* Playback Control */
+#define FM801_PLY_COUNT 0x0a /* Playback Count */
+#define FM801_PLY_BUF1 0x0c /* Playback Bufer I */
+#define FM801_PLY_BUF2 0x10 /* Playback Buffer II */
+#define FM801_CAP_CTRL 0x14 /* Capture Control */
+#define FM801_CAP_COUNT 0x16 /* Capture Count */
+#define FM801_CAP_BUF1 0x18 /* Capture Buffer I */
+#define FM801_CAP_BUF2 0x1c /* Capture Buffer II */
+#define FM801_CODEC_CTRL 0x22 /* Codec Control */
+#define FM801_I2S_MODE 0x24 /* I2S Mode Control */
+#define FM801_VOLUME 0x26 /* Volume Up/Down/Mute Status */
+#define FM801_I2C_CTRL 0x29 /* I2C Control */
+#define FM801_AC97_CMD 0x2a /* AC'97 Command */
+#define FM801_AC97_DATA 0x2c /* AC'97 Data */
+#define FM801_MPU401_DATA 0x30 /* MPU401 Data */
+#define FM801_MPU401_CMD 0x31 /* MPU401 Command */
+#define FM801_GPIO_CTRL 0x52 /* General Purpose I/O Control */
+#define FM801_GEN_CTRL 0x54 /* General Control */
+#define FM801_IRQ_MASK 0x56 /* Interrupt Mask */
+#define FM801_IRQ_STATUS 0x5a /* Interrupt Status */
+#define FM801_OPL3_BANK0 0x68 /* OPL3 Status Read / Bank 0 Write */
+#define FM801_OPL3_DATA0 0x69 /* OPL3 Data 0 Write */
+#define FM801_OPL3_BANK1 0x6a /* OPL3 Bank 1 Write */
+#define FM801_OPL3_DATA1 0x6b /* OPL3 Bank 1 Write */
+#define FM801_POWERDOWN 0x70 /* Blocks Power Down Control */
+
+/* codec access */
+#define FM801_AC97_READ (1<<7) /* read=1, write=0 */
+#define FM801_AC97_VALID (1<<8) /* port valid=1 */
+#define FM801_AC97_BUSY (1<<9) /* busy=1 */
+#define FM801_AC97_ADDR_SHIFT 10 /* codec id (2bit) */
+
+/* playback and record control register bits */
+#define FM801_BUF1_LAST (1<<1)
+#define FM801_BUF2_LAST (1<<2)
+#define FM801_START (1<<5)
+#define FM801_PAUSE (1<<6)
+#define FM801_IMMED_STOP (1<<7)
+#define FM801_RATE_SHIFT 8
+#define FM801_RATE_MASK (15 << FM801_RATE_SHIFT)
+#define FM801_CHANNELS_4 (1<<12) /* playback only */
+#define FM801_CHANNELS_6 (2<<12) /* playback only */
+#define FM801_CHANNELS_6MS (3<<12) /* playback only */
+#define FM801_CHANNELS_MASK (3<<12)
+#define FM801_16BIT (1<<14)
+#define FM801_STEREO (1<<15)
+
+/* IRQ status bits */
+#define FM801_IRQ_PLAYBACK (1<<8)
+#define FM801_IRQ_CAPTURE (1<<9)
+#define FM801_IRQ_VOLUME (1<<14)
+#define FM801_IRQ_MPU (1<<15)
+
+/* GPIO control register */
+#define FM801_GPIO_GP0 (1<<0) /* read/write */
+#define FM801_GPIO_GP1 (1<<1)
+#define FM801_GPIO_GP2 (1<<2)
+#define FM801_GPIO_GP3 (1<<3)
+#define FM801_GPIO_GP(x) (1<<(0+(x)))
+#define FM801_GPIO_GD0 (1<<8) /* directions: 1 = input, 0 = output*/
+#define FM801_GPIO_GD1 (1<<9)
+#define FM801_GPIO_GD2 (1<<10)
+#define FM801_GPIO_GD3 (1<<11)
+#define FM801_GPIO_GD(x) (1<<(8+(x)))
+#define FM801_GPIO_GS0 (1<<12) /* function select: */
+#define FM801_GPIO_GS1 (1<<13) /* 1 = GPIO */
+#define FM801_GPIO_GS2 (1<<14) /* 0 = other (S/PDIF, VOL) */
+#define FM801_GPIO_GS3 (1<<15)
+#define FM801_GPIO_GS(x) (1<<(12+(x)))
+
+/*
+
+ */
+
+struct fm801 {
+ int irq;
+
+ unsigned long port; /* I/O port number */
+ unsigned int multichannel: 1, /* multichannel support */
+ secondary: 1; /* secondary codec */
+ unsigned char secondary_addr; /* address of the secondary codec */
+ unsigned int tea575x_tuner; /* tuner flags */
+
+ unsigned short ply_ctrl; /* playback control */
+ unsigned short cap_ctrl; /* capture control */
+
+ unsigned long ply_buffer;
+ unsigned int ply_buf;
+ unsigned int ply_count;
+ unsigned int ply_size;
+ unsigned int ply_pos;
+
+ unsigned long cap_buffer;
+ unsigned int cap_buf;
+ unsigned int cap_count;
+ unsigned int cap_size;
+ unsigned int cap_pos;
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97;
+ struct snd_ac97 *ac97_sec;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_rawmidi *rmidi;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+ unsigned int p_dma_size;
+ unsigned int c_dma_size;
+
+ spinlock_t reg_lock;
+ struct snd_info_entry *proc_entry;
+
+#ifdef TEA575X_RADIO
+ struct snd_tea575x tea;
+#endif
+
+#ifdef CONFIG_PM
+ u16 saved_regs[0x20];
+#endif
+};
+
+static struct pci_device_id snd_fm801_ids[] = {
+ { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */
+ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_fm801_ids);
+
+/*
+ * common I/O routines
+ */
+
+static int snd_fm801_update_bits(struct fm801 *chip, unsigned short reg,
+ unsigned short mask, unsigned short value)
+{
+ int change;
+ unsigned long flags;
+ unsigned short old, new;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ old = inw(chip->port + reg);
+ new = (old & ~mask) | value;
+ change = old != new;
+ if (change)
+ outw(new, chip->port + reg);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return change;
+}
+
+static void snd_fm801_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct fm801 *chip = ac97->private_data;
+ int idx;
+
+ /*
+ * Wait until the codec interface is not ready..
+ */
+ for (idx = 0; idx < 100; idx++) {
+ if (!(inw(FM801_REG(chip, AC97_CMD)) & FM801_AC97_BUSY))
+ goto ok1;
+ udelay(10);
+ }
+ snd_printk(KERN_ERR "AC'97 interface is busy (1)\n");
+ return;
+
+ ok1:
+ /* write data and address */
+ outw(val, FM801_REG(chip, AC97_DATA));
+ outw(reg | (ac97->addr << FM801_AC97_ADDR_SHIFT), FM801_REG(chip, AC97_CMD));
+ /*
+ * Wait until the write command is not completed..
+ */
+ for (idx = 0; idx < 1000; idx++) {
+ if (!(inw(FM801_REG(chip, AC97_CMD)) & FM801_AC97_BUSY))
+ return;
+ udelay(10);
+ }
+ snd_printk(KERN_ERR "AC'97 interface #%d is busy (2)\n", ac97->num);
+}
+
+static unsigned short snd_fm801_codec_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct fm801 *chip = ac97->private_data;
+ int idx;
+
+ /*
+ * Wait until the codec interface is not ready..
+ */
+ for (idx = 0; idx < 100; idx++) {
+ if (!(inw(FM801_REG(chip, AC97_CMD)) & FM801_AC97_BUSY))
+ goto ok1;
+ udelay(10);
+ }
+ snd_printk(KERN_ERR "AC'97 interface is busy (1)\n");
+ return 0;
+
+ ok1:
+ /* read command */
+ outw(reg | (ac97->addr << FM801_AC97_ADDR_SHIFT) | FM801_AC97_READ,
+ FM801_REG(chip, AC97_CMD));
+ for (idx = 0; idx < 100; idx++) {
+ if (!(inw(FM801_REG(chip, AC97_CMD)) & FM801_AC97_BUSY))
+ goto ok2;
+ udelay(10);
+ }
+ snd_printk(KERN_ERR "AC'97 interface #%d is busy (2)\n", ac97->num);
+ return 0;
+
+ ok2:
+ for (idx = 0; idx < 1000; idx++) {
+ if (inw(FM801_REG(chip, AC97_CMD)) & FM801_AC97_VALID)
+ goto ok3;
+ udelay(10);
+ }
+ snd_printk(KERN_ERR "AC'97 interface #%d is not valid (2)\n", ac97->num);
+ return 0;
+
+ ok3:
+ return inw(FM801_REG(chip, AC97_DATA));
+}
+
+static unsigned int rates[] = {
+ 5500, 8000, 9600, 11025,
+ 16000, 19200, 22050, 32000,
+ 38400, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static unsigned int channels[] = {
+ 2, 4, 6
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+};
+
+/*
+ * Sample rate routines
+ */
+
+static unsigned short snd_fm801_rate_bits(unsigned int rate)
+{
+ unsigned int idx;
+
+ for (idx = 0; idx < ARRAY_SIZE(rates); idx++)
+ if (rates[idx] == rate)
+ return idx;
+ snd_BUG();
+ return ARRAY_SIZE(rates) - 1;
+}
+
+/*
+ * PCM part
+ */
+
+static int snd_fm801_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ chip->ply_ctrl &= ~(FM801_BUF1_LAST |
+ FM801_BUF2_LAST |
+ FM801_PAUSE);
+ chip->ply_ctrl |= FM801_START |
+ FM801_IMMED_STOP;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ chip->ply_ctrl &= ~(FM801_START | FM801_PAUSE);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ chip->ply_ctrl |= FM801_PAUSE;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ chip->ply_ctrl &= ~FM801_PAUSE;
+ break;
+ default:
+ spin_unlock(&chip->reg_lock);
+ snd_BUG();
+ return -EINVAL;
+ }
+ outw(chip->ply_ctrl, FM801_REG(chip, PLY_CTRL));
+ spin_unlock(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_fm801_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ chip->cap_ctrl &= ~(FM801_BUF1_LAST |
+ FM801_BUF2_LAST |
+ FM801_PAUSE);
+ chip->cap_ctrl |= FM801_START |
+ FM801_IMMED_STOP;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ chip->cap_ctrl &= ~(FM801_START | FM801_PAUSE);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ chip->cap_ctrl |= FM801_PAUSE;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ chip->cap_ctrl &= ~FM801_PAUSE;
+ break;
+ default:
+ spin_unlock(&chip->reg_lock);
+ snd_BUG();
+ return -EINVAL;
+ }
+ outw(chip->cap_ctrl, FM801_REG(chip, CAP_CTRL));
+ spin_unlock(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_fm801_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_fm801_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_fm801_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ chip->ply_size = snd_pcm_lib_buffer_bytes(substream);
+ chip->ply_count = snd_pcm_lib_period_bytes(substream);
+ spin_lock_irq(&chip->reg_lock);
+ chip->ply_ctrl &= ~(FM801_START | FM801_16BIT |
+ FM801_STEREO | FM801_RATE_MASK |
+ FM801_CHANNELS_MASK);
+ if (snd_pcm_format_width(runtime->format) == 16)
+ chip->ply_ctrl |= FM801_16BIT;
+ if (runtime->channels > 1) {
+ chip->ply_ctrl |= FM801_STEREO;
+ if (runtime->channels == 4)
+ chip->ply_ctrl |= FM801_CHANNELS_4;
+ else if (runtime->channels == 6)
+ chip->ply_ctrl |= FM801_CHANNELS_6;
+ }
+ chip->ply_ctrl |= snd_fm801_rate_bits(runtime->rate) << FM801_RATE_SHIFT;
+ chip->ply_buf = 0;
+ outw(chip->ply_ctrl, FM801_REG(chip, PLY_CTRL));
+ outw(chip->ply_count - 1, FM801_REG(chip, PLY_COUNT));
+ chip->ply_buffer = runtime->dma_addr;
+ chip->ply_pos = 0;
+ outl(chip->ply_buffer, FM801_REG(chip, PLY_BUF1));
+ outl(chip->ply_buffer + (chip->ply_count % chip->ply_size), FM801_REG(chip, PLY_BUF2));
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_fm801_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ chip->cap_size = snd_pcm_lib_buffer_bytes(substream);
+ chip->cap_count = snd_pcm_lib_period_bytes(substream);
+ spin_lock_irq(&chip->reg_lock);
+ chip->cap_ctrl &= ~(FM801_START | FM801_16BIT |
+ FM801_STEREO | FM801_RATE_MASK);
+ if (snd_pcm_format_width(runtime->format) == 16)
+ chip->cap_ctrl |= FM801_16BIT;
+ if (runtime->channels > 1)
+ chip->cap_ctrl |= FM801_STEREO;
+ chip->cap_ctrl |= snd_fm801_rate_bits(runtime->rate) << FM801_RATE_SHIFT;
+ chip->cap_buf = 0;
+ outw(chip->cap_ctrl, FM801_REG(chip, CAP_CTRL));
+ outw(chip->cap_count - 1, FM801_REG(chip, CAP_COUNT));
+ chip->cap_buffer = runtime->dma_addr;
+ chip->cap_pos = 0;
+ outl(chip->cap_buffer, FM801_REG(chip, CAP_BUF1));
+ outl(chip->cap_buffer + (chip->cap_count % chip->cap_size), FM801_REG(chip, CAP_BUF2));
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_fm801_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(chip->ply_ctrl & FM801_START))
+ return 0;
+ spin_lock(&chip->reg_lock);
+ ptr = chip->ply_pos + (chip->ply_count - 1) - inw(FM801_REG(chip, PLY_COUNT));
+ if (inw(FM801_REG(chip, IRQ_STATUS)) & FM801_IRQ_PLAYBACK) {
+ ptr += chip->ply_count;
+ ptr %= chip->ply_size;
+ }
+ spin_unlock(&chip->reg_lock);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_fm801_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(chip->cap_ctrl & FM801_START))
+ return 0;
+ spin_lock(&chip->reg_lock);
+ ptr = chip->cap_pos + (chip->cap_count - 1) - inw(FM801_REG(chip, CAP_COUNT));
+ if (inw(FM801_REG(chip, IRQ_STATUS)) & FM801_IRQ_CAPTURE) {
+ ptr += chip->cap_count;
+ ptr %= chip->cap_size;
+ }
+ spin_unlock(&chip->reg_lock);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static irqreturn_t snd_fm801_interrupt(int irq, void *dev_id)
+{
+ struct fm801 *chip = dev_id;
+ unsigned short status;
+ unsigned int tmp;
+
+ status = inw(FM801_REG(chip, IRQ_STATUS));
+ status &= FM801_IRQ_PLAYBACK|FM801_IRQ_CAPTURE|FM801_IRQ_MPU|FM801_IRQ_VOLUME;
+ if (! status)
+ return IRQ_NONE;
+ /* ack first */
+ outw(status, FM801_REG(chip, IRQ_STATUS));
+ if (chip->pcm && (status & FM801_IRQ_PLAYBACK) && chip->playback_substream) {
+ spin_lock(&chip->reg_lock);
+ chip->ply_buf++;
+ chip->ply_pos += chip->ply_count;
+ chip->ply_pos %= chip->ply_size;
+ tmp = chip->ply_pos + chip->ply_count;
+ tmp %= chip->ply_size;
+ outl(chip->ply_buffer + tmp,
+ (chip->ply_buf & 1) ?
+ FM801_REG(chip, PLY_BUF1) :
+ FM801_REG(chip, PLY_BUF2));
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(chip->playback_substream);
+ }
+ if (chip->pcm && (status & FM801_IRQ_CAPTURE) && chip->capture_substream) {
+ spin_lock(&chip->reg_lock);
+ chip->cap_buf++;
+ chip->cap_pos += chip->cap_count;
+ chip->cap_pos %= chip->cap_size;
+ tmp = chip->cap_pos + chip->cap_count;
+ tmp %= chip->cap_size;
+ outl(chip->cap_buffer + tmp,
+ (chip->cap_buf & 1) ?
+ FM801_REG(chip, CAP_BUF1) :
+ FM801_REG(chip, CAP_BUF2));
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(chip->capture_substream);
+ }
+ if (chip->rmidi && (status & FM801_IRQ_MPU))
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+ if (status & FM801_IRQ_VOLUME)
+ ;/* TODO */
+
+ return IRQ_HANDLED;
+}
+
+static struct snd_pcm_hardware snd_fm801_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5500,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_fm801_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5500,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_fm801_playback_open(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ chip->playback_substream = substream;
+ runtime->hw = snd_fm801_playback;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+ if (chip->multichannel) {
+ runtime->hw.channels_max = 6;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &hw_constraints_channels);
+ }
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_fm801_capture_open(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ chip->capture_substream = substream;
+ runtime->hw = snd_fm801_capture;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_fm801_playback_close(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+
+ chip->playback_substream = NULL;
+ return 0;
+}
+
+static int snd_fm801_capture_close(struct snd_pcm_substream *substream)
+{
+ struct fm801 *chip = snd_pcm_substream_chip(substream);
+
+ chip->capture_substream = NULL;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_fm801_playback_ops = {
+ .open = snd_fm801_playback_open,
+ .close = snd_fm801_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_fm801_hw_params,
+ .hw_free = snd_fm801_hw_free,
+ .prepare = snd_fm801_playback_prepare,
+ .trigger = snd_fm801_playback_trigger,
+ .pointer = snd_fm801_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_fm801_capture_ops = {
+ .open = snd_fm801_capture_open,
+ .close = snd_fm801_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_fm801_hw_params,
+ .hw_free = snd_fm801_hw_free,
+ .prepare = snd_fm801_capture_prepare,
+ .trigger = snd_fm801_capture_trigger,
+ .pointer = snd_fm801_capture_pointer,
+};
+
+static int __devinit snd_fm801_pcm(struct fm801 *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(chip->card, "FM801", device, 1, 1, &pcm)) < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_fm801_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_fm801_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "FM801");
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ chip->multichannel ? 128*1024 : 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/*
+ * TEA5757 radio
+ */
+
+#ifdef TEA575X_RADIO
+
+/* 256PCS GPIO numbers */
+#define TEA_256PCS_DATA 1
+#define TEA_256PCS_WRITE_ENABLE 2 /* inverted */
+#define TEA_256PCS_BUS_CLOCK 3
+
+static void snd_fm801_tea575x_256pcs_write(struct snd_tea575x *tea, unsigned int val)
+{
+ struct fm801 *chip = tea->private_data;
+ unsigned short reg;
+ int i = 25;
+
+ spin_lock_irq(&chip->reg_lock);
+ reg = inw(FM801_REG(chip, GPIO_CTRL));
+ /* use GPIO lines and set write enable bit */
+ reg |= FM801_GPIO_GS(TEA_256PCS_DATA) |
+ FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) |
+ FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK);
+ /* all of lines are in the write direction */
+ /* clear data and clock lines */
+ reg &= ~(FM801_GPIO_GD(TEA_256PCS_DATA) |
+ FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) |
+ FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_256PCS_DATA) |
+ FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE));
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+
+ while (i--) {
+ if (val & (1 << i))
+ reg |= FM801_GPIO_GP(TEA_256PCS_DATA);
+ else
+ reg &= ~FM801_GPIO_GP(TEA_256PCS_DATA);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ }
+
+ /* and reset the write enable bit */
+ reg |= FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE) |
+ FM801_GPIO_GP(TEA_256PCS_DATA);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ spin_unlock_irq(&chip->reg_lock);
+}
+
+static unsigned int snd_fm801_tea575x_256pcs_read(struct snd_tea575x *tea)
+{
+ struct fm801 *chip = tea->private_data;
+ unsigned short reg;
+ unsigned int val = 0;
+ int i;
+
+ spin_lock_irq(&chip->reg_lock);
+ reg = inw(FM801_REG(chip, GPIO_CTRL));
+ /* use GPIO lines, set data direction to input */
+ reg |= FM801_GPIO_GS(TEA_256PCS_DATA) |
+ FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) |
+ FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK) |
+ FM801_GPIO_GD(TEA_256PCS_DATA) |
+ FM801_GPIO_GP(TEA_256PCS_DATA) |
+ FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE);
+ /* all of lines are in the write direction, except data */
+ /* clear data, write enable and clock lines */
+ reg &= ~(FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) |
+ FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK));
+
+ for (i = 0; i < 24; i++) {
+ reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ val <<= 1;
+ if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCS_DATA))
+ val |= 1;
+ }
+
+ spin_unlock_irq(&chip->reg_lock);
+
+ return val;
+}
+
+/* 256PCPR GPIO numbers */
+#define TEA_256PCPR_BUS_CLOCK 0
+#define TEA_256PCPR_DATA 1
+#define TEA_256PCPR_WRITE_ENABLE 2 /* inverted */
+
+static void snd_fm801_tea575x_256pcpr_write(struct snd_tea575x *tea, unsigned int val)
+{
+ struct fm801 *chip = tea->private_data;
+ unsigned short reg;
+ int i = 25;
+
+ spin_lock_irq(&chip->reg_lock);
+ reg = inw(FM801_REG(chip, GPIO_CTRL));
+ /* use GPIO lines and set write enable bit */
+ reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) |
+ FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) |
+ FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK);
+ /* all of lines are in the write direction */
+ /* clear data and clock lines */
+ reg &= ~(FM801_GPIO_GD(TEA_256PCPR_DATA) |
+ FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) |
+ FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_256PCPR_DATA) |
+ FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE));
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+
+ while (i--) {
+ if (val & (1 << i))
+ reg |= FM801_GPIO_GP(TEA_256PCPR_DATA);
+ else
+ reg &= ~FM801_GPIO_GP(TEA_256PCPR_DATA);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ }
+
+ /* and reset the write enable bit */
+ reg |= FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE) |
+ FM801_GPIO_GP(TEA_256PCPR_DATA);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ spin_unlock_irq(&chip->reg_lock);
+}
+
+static unsigned int snd_fm801_tea575x_256pcpr_read(struct snd_tea575x *tea)
+{
+ struct fm801 *chip = tea->private_data;
+ unsigned short reg;
+ unsigned int val = 0;
+ int i;
+
+ spin_lock_irq(&chip->reg_lock);
+ reg = inw(FM801_REG(chip, GPIO_CTRL));
+ /* use GPIO lines, set data direction to input */
+ reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) |
+ FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) |
+ FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK) |
+ FM801_GPIO_GD(TEA_256PCPR_DATA) |
+ FM801_GPIO_GP(TEA_256PCPR_DATA) |
+ FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE);
+ /* all of lines are in the write direction, except data */
+ /* clear data, write enable and clock lines */
+ reg &= ~(FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) |
+ FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK));
+
+ for (i = 0; i < 24; i++) {
+ reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ val <<= 1;
+ if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCPR_DATA))
+ val |= 1;
+ }
+
+ spin_unlock_irq(&chip->reg_lock);
+
+ return val;
+}
+
+/* 64PCR GPIO numbers */
+#define TEA_64PCR_BUS_CLOCK 0
+#define TEA_64PCR_WRITE_ENABLE 1 /* inverted */
+#define TEA_64PCR_DATA 2
+
+static void snd_fm801_tea575x_64pcr_write(struct snd_tea575x *tea, unsigned int val)
+{
+ struct fm801 *chip = tea->private_data;
+ unsigned short reg;
+ int i = 25;
+
+ spin_lock_irq(&chip->reg_lock);
+ reg = inw(FM801_REG(chip, GPIO_CTRL));
+ /* use GPIO lines and set write enable bit */
+ reg |= FM801_GPIO_GS(TEA_64PCR_DATA) |
+ FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) |
+ FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK);
+ /* all of lines are in the write direction */
+ /* clear data and clock lines */
+ reg &= ~(FM801_GPIO_GD(TEA_64PCR_DATA) |
+ FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) |
+ FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_64PCR_DATA) |
+ FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE));
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+
+ while (i--) {
+ if (val & (1 << i))
+ reg |= FM801_GPIO_GP(TEA_64PCR_DATA);
+ else
+ reg &= ~FM801_GPIO_GP(TEA_64PCR_DATA);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ }
+
+ /* and reset the write enable bit */
+ reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE) |
+ FM801_GPIO_GP(TEA_64PCR_DATA);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ spin_unlock_irq(&chip->reg_lock);
+}
+
+static unsigned int snd_fm801_tea575x_64pcr_read(struct snd_tea575x *tea)
+{
+ struct fm801 *chip = tea->private_data;
+ unsigned short reg;
+ unsigned int val = 0;
+ int i;
+
+ spin_lock_irq(&chip->reg_lock);
+ reg = inw(FM801_REG(chip, GPIO_CTRL));
+ /* use GPIO lines, set data direction to input */
+ reg |= FM801_GPIO_GS(TEA_64PCR_DATA) |
+ FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) |
+ FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK) |
+ FM801_GPIO_GD(TEA_64PCR_DATA) |
+ FM801_GPIO_GP(TEA_64PCR_DATA) |
+ FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE);
+ /* all of lines are in the write direction, except data */
+ /* clear data, write enable and clock lines */
+ reg &= ~(FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) |
+ FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) |
+ FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK));
+
+ for (i = 0; i < 24; i++) {
+ reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+ val <<= 1;
+ if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_64PCR_DATA))
+ val |= 1;
+ }
+
+ spin_unlock_irq(&chip->reg_lock);
+
+ return val;
+}
+
+static void snd_fm801_tea575x_64pcr_mute(struct snd_tea575x *tea,
+ unsigned int mute)
+{
+ struct fm801 *chip = tea->private_data;
+ unsigned short reg;
+
+ spin_lock_irq(&chip->reg_lock);
+
+ reg = inw(FM801_REG(chip, GPIO_CTRL));
+ if (mute)
+ /* 0xf800 (mute) */
+ reg &= ~FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE);
+ else
+ /* 0xf802 (unmute) */
+ reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE);
+ outw(reg, FM801_REG(chip, GPIO_CTRL));
+ udelay(1);
+
+ spin_unlock_irq(&chip->reg_lock);
+}
+
+static struct snd_tea575x_ops snd_fm801_tea_ops[3] = {
+ {
+ /* 1 = MediaForte 256-PCS */
+ .write = snd_fm801_tea575x_256pcs_write,
+ .read = snd_fm801_tea575x_256pcs_read,
+ },
+ {
+ /* 2 = MediaForte 256-PCPR */
+ .write = snd_fm801_tea575x_256pcpr_write,
+ .read = snd_fm801_tea575x_256pcpr_read,
+ },
+ {
+ /* 3 = MediaForte 64-PCR */
+ .write = snd_fm801_tea575x_64pcr_write,
+ .read = snd_fm801_tea575x_64pcr_read,
+ .mute = snd_fm801_tea575x_64pcr_mute,
+ }
+};
+#endif
+
+/*
+ * Mixer routines
+ */
+
+#define FM801_SINGLE(xname, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_fm801_info_single, \
+ .get = snd_fm801_get_single, .put = snd_fm801_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_fm801_info_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_fm801_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct fm801 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ ucontrol->value.integer.value[0] = (inw(chip->port + reg) >> shift) & mask;
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_fm801_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct fm801 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ return snd_fm801_update_bits(chip, reg, mask << shift, val << shift);
+}
+
+#define FM801_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_fm801_info_double, \
+ .get = snd_fm801_get_double, .put = snd_fm801_put_double, \
+ .private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24) }
+#define FM801_DOUBLE_TLV(xname, reg, shift_left, shift_right, mask, invert, xtlv) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .name = xname, .info = snd_fm801_info_double, \
+ .get = snd_fm801_get_double, .put = snd_fm801_put_double, \
+ .private_value = reg | (shift_left << 8) | (shift_right << 12) | (mask << 16) | (invert << 24), \
+ .tlv = { .p = (xtlv) } }
+
+static int snd_fm801_info_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_fm801_get_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct fm801 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift_left = (kcontrol->private_value >> 8) & 0x0f;
+ int shift_right = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irq(&chip->reg_lock);
+ ucontrol->value.integer.value[0] = (inw(chip->port + reg) >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (inw(chip->port + reg) >> shift_right) & mask;
+ spin_unlock_irq(&chip->reg_lock);
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_fm801_put_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct fm801 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift_left = (kcontrol->private_value >> 8) & 0x0f;
+ int shift_right = (kcontrol->private_value >> 12) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ unsigned short val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ return snd_fm801_update_bits(chip, reg,
+ (mask << shift_left) | (mask << shift_right),
+ (val1 << shift_left ) | (val2 << shift_right));
+}
+
+static int snd_fm801_info_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[5] = {
+ "AC97 Primary", "FM", "I2S", "PCM", "AC97 Secondary"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 5;
+ if (uinfo->value.enumerated.item > 4)
+ uinfo->value.enumerated.item = 4;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_fm801_get_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct fm801 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ val = inw(FM801_REG(chip, REC_SRC)) & 7;
+ if (val > 4)
+ val = 4;
+ ucontrol->value.enumerated.item[0] = val;
+ return 0;
+}
+
+static int snd_fm801_put_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct fm801 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ if ((val = ucontrol->value.enumerated.item[0]) > 4)
+ return -EINVAL;
+ return snd_fm801_update_bits(chip, FM801_REC_SRC, 7, val);
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_dsp, -3450, 150, 0);
+
+#define FM801_CONTROLS ARRAY_SIZE(snd_fm801_controls)
+
+static struct snd_kcontrol_new snd_fm801_controls[] __devinitdata = {
+FM801_DOUBLE_TLV("Wave Playback Volume", FM801_PCM_VOL, 0, 8, 31, 1,
+ db_scale_dsp),
+FM801_SINGLE("Wave Playback Switch", FM801_PCM_VOL, 15, 1, 1),
+FM801_DOUBLE_TLV("I2S Playback Volume", FM801_I2S_VOL, 0, 8, 31, 1,
+ db_scale_dsp),
+FM801_SINGLE("I2S Playback Switch", FM801_I2S_VOL, 15, 1, 1),
+FM801_DOUBLE_TLV("FM Playback Volume", FM801_FM_VOL, 0, 8, 31, 1,
+ db_scale_dsp),
+FM801_SINGLE("FM Playback Switch", FM801_FM_VOL, 15, 1, 1),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Capture Source",
+ .info = snd_fm801_info_mux,
+ .get = snd_fm801_get_mux,
+ .put = snd_fm801_put_mux,
+}
+};
+
+#define FM801_CONTROLS_MULTI ARRAY_SIZE(snd_fm801_controls_multi)
+
+static struct snd_kcontrol_new snd_fm801_controls_multi[] __devinitdata = {
+FM801_SINGLE("AC97 2ch->4ch Copy Switch", FM801_CODEC_CTRL, 7, 1, 0),
+FM801_SINGLE("AC97 18-bit Switch", FM801_CODEC_CTRL, 10, 1, 0),
+FM801_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), FM801_I2S_MODE, 8, 1, 0),
+FM801_SINGLE(SNDRV_CTL_NAME_IEC958("Raw Data ",PLAYBACK,SWITCH), FM801_I2S_MODE, 9, 1, 0),
+FM801_SINGLE(SNDRV_CTL_NAME_IEC958("Raw Data ",CAPTURE,SWITCH), FM801_I2S_MODE, 10, 1, 0),
+FM801_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), FM801_GEN_CTRL, 2, 1, 0),
+};
+
+static void snd_fm801_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct fm801 *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void snd_fm801_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct fm801 *chip = ac97->private_data;
+ if (ac97->num == 0) {
+ chip->ac97 = NULL;
+ } else {
+ chip->ac97_sec = NULL;
+ }
+}
+
+static int __devinit snd_fm801_mixer(struct fm801 *chip)
+{
+ struct snd_ac97_template ac97;
+ unsigned int i;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_fm801_codec_write,
+ .read = snd_fm801_codec_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &chip->ac97_bus)) < 0)
+ return err;
+ chip->ac97_bus->private_free = snd_fm801_mixer_free_ac97_bus;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_fm801_mixer_free_ac97;
+ if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97)) < 0)
+ return err;
+ if (chip->secondary) {
+ ac97.num = 1;
+ ac97.addr = chip->secondary_addr;
+ if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97_sec)) < 0)
+ return err;
+ }
+ for (i = 0; i < FM801_CONTROLS; i++)
+ snd_ctl_add(chip->card, snd_ctl_new1(&snd_fm801_controls[i], chip));
+ if (chip->multichannel) {
+ for (i = 0; i < FM801_CONTROLS_MULTI; i++)
+ snd_ctl_add(chip->card, snd_ctl_new1(&snd_fm801_controls_multi[i], chip));
+ }
+ return 0;
+}
+
+/*
+ * initialization routines
+ */
+
+static int wait_for_codec(struct fm801 *chip, unsigned int codec_id,
+ unsigned short reg, unsigned long waits)
+{
+ unsigned long timeout = jiffies + waits;
+
+ outw(FM801_AC97_READ | (codec_id << FM801_AC97_ADDR_SHIFT) | reg,
+ FM801_REG(chip, AC97_CMD));
+ udelay(5);
+ do {
+ if ((inw(FM801_REG(chip, AC97_CMD)) & (FM801_AC97_VALID|FM801_AC97_BUSY))
+ == FM801_AC97_VALID)
+ return 0;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after(timeout, jiffies));
+ return -EIO;
+}
+
+static int snd_fm801_chip_init(struct fm801 *chip, int resume)
+{
+ unsigned short cmdw;
+
+ if (chip->tea575x_tuner & 0x0010)
+ goto __ac97_ok;
+
+ /* codec cold reset + AC'97 warm reset */
+ outw((1<<5) | (1<<6), FM801_REG(chip, CODEC_CTRL));
+ inw(FM801_REG(chip, CODEC_CTRL)); /* flush posting data */
+ udelay(100);
+ outw(0, FM801_REG(chip, CODEC_CTRL));
+
+ if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) {
+ snd_printk(KERN_ERR "Primary AC'97 codec not found\n");
+ if (! resume)
+ return -EIO;
+ }
+
+ if (chip->multichannel) {
+ if (chip->secondary_addr) {
+ wait_for_codec(chip, chip->secondary_addr,
+ AC97_VENDOR_ID1, msecs_to_jiffies(50));
+ } else {
+ /* my card has the secondary codec */
+ /* at address #3, so the loop is inverted */
+ int i;
+ for (i = 3; i > 0; i--) {
+ if (!wait_for_codec(chip, i, AC97_VENDOR_ID1,
+ msecs_to_jiffies(50))) {
+ cmdw = inw(FM801_REG(chip, AC97_DATA));
+ if (cmdw != 0xffff && cmdw != 0) {
+ chip->secondary = 1;
+ chip->secondary_addr = i;
+ break;
+ }
+ }
+ }
+ }
+
+ /* the recovery phase, it seems that probing for non-existing codec might */
+ /* cause timeout problems */
+ wait_for_codec(chip, 0, AC97_VENDOR_ID1, msecs_to_jiffies(750));
+ }
+
+ __ac97_ok:
+
+ /* init volume */
+ outw(0x0808, FM801_REG(chip, PCM_VOL));
+ outw(0x9f1f, FM801_REG(chip, FM_VOL));
+ outw(0x8808, FM801_REG(chip, I2S_VOL));
+
+ /* I2S control - I2S mode */
+ outw(0x0003, FM801_REG(chip, I2S_MODE));
+
+ /* interrupt setup */
+ cmdw = inw(FM801_REG(chip, IRQ_MASK));
+ if (chip->irq < 0)
+ cmdw |= 0x00c3; /* mask everything, no PCM nor MPU */
+ else
+ cmdw &= ~0x0083; /* unmask MPU, PLAYBACK & CAPTURE */
+ outw(cmdw, FM801_REG(chip, IRQ_MASK));
+
+ /* interrupt clear */
+ outw(FM801_IRQ_PLAYBACK|FM801_IRQ_CAPTURE|FM801_IRQ_MPU, FM801_REG(chip, IRQ_STATUS));
+
+ return 0;
+}
+
+
+static int snd_fm801_free(struct fm801 *chip)
+{
+ unsigned short cmdw;
+
+ if (chip->irq < 0)
+ goto __end_hw;
+
+ /* interrupt setup - mask everything */
+ cmdw = inw(FM801_REG(chip, IRQ_MASK));
+ cmdw |= 0x00c3;
+ outw(cmdw, FM801_REG(chip, IRQ_MASK));
+
+ __end_hw:
+#ifdef TEA575X_RADIO
+ snd_tea575x_exit(&chip->tea);
+#endif
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+
+ kfree(chip);
+ return 0;
+}
+
+static int snd_fm801_dev_free(struct snd_device *device)
+{
+ struct fm801 *chip = device->device_data;
+ return snd_fm801_free(chip);
+}
+
+static int __devinit snd_fm801_create(struct snd_card *card,
+ struct pci_dev * pci,
+ int tea575x_tuner,
+ struct fm801 ** rchip)
+{
+ struct fm801 *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_fm801_dev_free,
+ };
+
+ *rchip = NULL;
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&chip->reg_lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->tea575x_tuner = tea575x_tuner;
+ if ((err = pci_request_regions(pci, "FM801")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->port = pci_resource_start(pci, 0);
+ if ((tea575x_tuner & 0x0010) == 0) {
+ if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED,
+ "FM801", chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
+ snd_fm801_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ pci_set_master(pci);
+ }
+
+ if (pci->revision >= 0xb1) /* FM801-AU */
+ chip->multichannel = 1;
+
+ snd_fm801_chip_init(chip, 0);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_fm801_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+#ifdef TEA575X_RADIO
+ if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) {
+ chip->tea.dev_nr = tea575x_tuner >> 16;
+ chip->tea.card = card;
+ chip->tea.freq_fixup = 10700;
+ chip->tea.private_data = chip;
+ chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1];
+ snd_tea575x_init(&chip->tea);
+ }
+#endif
+
+ *rchip = chip;
+ return 0;
+}
+
+static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct fm801 *chip;
+ struct snd_opl3 *opl3;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ strcpy(card->driver, "FM801");
+ strcpy(card->shortname, "ForteMedia FM801-");
+ strcat(card->shortname, chip->multichannel ? "AU" : "AS");
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, chip->port, chip->irq);
+
+ if (tea575x_tuner[dev] & 0x0010)
+ goto __fm801_tuner_only;
+
+ if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_fm801_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801,
+ FM801_REG(chip, MPU401_DATA),
+ MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_opl3_create(card, FM801_REG(chip, OPL3_BANK0),
+ FM801_REG(chip, OPL3_BANK1),
+ OPL3_HW_OPL3_FM801, 1, &opl3)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ __fm801_tuner_only:
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_card_fm801_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+#ifdef CONFIG_PM
+static unsigned char saved_regs[] = {
+ FM801_PCM_VOL, FM801_I2S_VOL, FM801_FM_VOL, FM801_REC_SRC,
+ FM801_PLY_CTRL, FM801_PLY_COUNT, FM801_PLY_BUF1, FM801_PLY_BUF2,
+ FM801_CAP_CTRL, FM801_CAP_COUNT, FM801_CAP_BUF1, FM801_CAP_BUF2,
+ FM801_CODEC_CTRL, FM801_I2S_MODE, FM801_VOLUME, FM801_GEN_CTRL,
+};
+
+static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct fm801 *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_ac97_suspend(chip->ac97);
+ snd_ac97_suspend(chip->ac97_sec);
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ chip->saved_regs[i] = inw(chip->port + saved_regs[i]);
+ /* FIXME: tea575x suspend */
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_fm801_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct fm801 *chip = card->private_data;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "fm801: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_fm801_chip_init(chip, 1);
+ snd_ac97_resume(chip->ac97);
+ snd_ac97_resume(chip->ac97_sec);
+ for (i = 0; i < ARRAY_SIZE(saved_regs); i++)
+ outw(chip->saved_regs[i], chip->port + saved_regs[i]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
+
+static struct pci_driver driver = {
+ .name = "FM801",
+ .id_table = snd_fm801_ids,
+ .probe = snd_card_fm801_probe,
+ .remove = __devexit_p(snd_card_fm801_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_fm801_suspend,
+ .resume = snd_fm801_resume,
+#endif
+};
+
+static int __init alsa_card_fm801_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_fm801_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_fm801_init)
+module_exit(alsa_card_fm801_exit)
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
new file mode 100644
index 0000000..1980c6d
--- /dev/null
+++ b/sound/pci/hda/Makefile
@@ -0,0 +1,20 @@
+snd-hda-intel-y := hda_intel.o
+# since snd-hda-intel is the only driver using hda-codec,
+# merge it into a single module although it was originally
+# designed to be individual modules
+snd-hda-intel-y += hda_codec.o
+snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
+snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
+snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o
+snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_NVHDMI) += patch_nvhdmi.o
+
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
new file mode 100644
index 0000000..3ecd7e7
--- /dev/null
+++ b/sound/pci/hda/hda_beep.c
@@ -0,0 +1,142 @@
+/*
+ * Digital Beep Input Interface for HD-audio codec
+ *
+ * Author: Matthew Ranostay <mranostay@embeddedalley.com>
+ * Copyright (c) 2008 Embedded Alley Solutions Inc
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/input.h>
+#include <linux/pci.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include "hda_beep.h"
+
+enum {
+ DIGBEEP_HZ_STEP = 46875, /* 46.875 Hz */
+ DIGBEEP_HZ_MIN = 93750, /* 93.750 Hz */
+ DIGBEEP_HZ_MAX = 12000000, /* 12 KHz */
+};
+
+static void snd_hda_generate_beep(struct work_struct *work)
+{
+ struct hda_beep *beep =
+ container_of(work, struct hda_beep, beep_work);
+ struct hda_codec *codec = beep->codec;
+
+ if (!beep->enabled)
+ return;
+
+ /* generate tone */
+ snd_hda_codec_write_cache(codec, beep->nid, 0,
+ AC_VERB_SET_BEEP_CONTROL, beep->tone);
+}
+
+static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
+ unsigned int code, int hz)
+{
+ struct hda_beep *beep = input_get_drvdata(dev);
+
+ switch (code) {
+ case SND_BELL:
+ if (hz)
+ hz = 1000;
+ case SND_TONE:
+ hz *= 1000; /* fixed point */
+ hz = hz - DIGBEEP_HZ_MIN;
+ if (hz < 0)
+ hz = 0; /* turn off PC beep*/
+ else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN))
+ hz = 0xff;
+ else {
+ hz /= DIGBEEP_HZ_STEP;
+ hz++;
+ }
+ break;
+ default:
+ return -1;
+ }
+ beep->tone = hz;
+
+ /* schedule beep event */
+ schedule_work(&beep->beep_work);
+ return 0;
+}
+
+int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+ struct input_dev *input_dev;
+ struct hda_beep *beep;
+ int err;
+
+ beep = kzalloc(sizeof(*beep), GFP_KERNEL);
+ if (beep == NULL)
+ return -ENOMEM;
+ snprintf(beep->phys, sizeof(beep->phys),
+ "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
+ input_dev = input_allocate_device();
+ if (!input_dev) {
+ kfree(beep);
+ return -ENOMEM;
+ }
+
+ /* setup digital beep device */
+ input_dev->name = "HDA Digital PCBeep";
+ input_dev->phys = beep->phys;
+ input_dev->id.bustype = BUS_PCI;
+
+ input_dev->id.vendor = codec->vendor_id >> 16;
+ input_dev->id.product = codec->vendor_id & 0xffff;
+ input_dev->id.version = 0x01;
+
+ input_dev->evbit[0] = BIT_MASK(EV_SND);
+ input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE);
+ input_dev->event = snd_hda_beep_event;
+ input_dev->dev.parent = &codec->bus->pci->dev;
+ input_set_drvdata(input_dev, beep);
+
+ err = input_register_device(input_dev);
+ if (err < 0) {
+ input_free_device(input_dev);
+ kfree(beep);
+ return err;
+ }
+
+ /* enable linear scale */
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_2, 0x01);
+
+ beep->nid = nid;
+ beep->dev = input_dev;
+ beep->codec = codec;
+ beep->enabled = 1;
+ codec->beep = beep;
+
+ INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
+ return 0;
+}
+
+void snd_hda_detach_beep_device(struct hda_codec *codec)
+{
+ struct hda_beep *beep = codec->beep;
+ if (beep) {
+ cancel_work_sync(&beep->beep_work);
+ flush_scheduled_work();
+
+ input_unregister_device(beep->dev);
+ kfree(beep);
+ }
+}
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
new file mode 100644
index 0000000..b9679f0
--- /dev/null
+++ b/sound/pci/hda/hda_beep.h
@@ -0,0 +1,45 @@
+/*
+ * Digital Beep Input Interface for HD-audio codec
+ *
+ * Author: Matthew Ranostay <mranostay@embeddedalley.com>
+ * Copyright (c) 2008 Embedded Alley Solutions Inc
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_HDA_BEEP_H
+#define __SOUND_HDA_BEEP_H
+
+#include "hda_codec.h"
+
+/* beep information */
+struct hda_beep {
+ struct input_dev *dev;
+ struct hda_codec *codec;
+ char phys[32];
+ int tone;
+ int nid;
+ int enabled;
+ struct work_struct beep_work; /* scheduled task for beep event */
+};
+
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
+void snd_hda_detach_beep_device(struct hda_codec *codec);
+#else
+#define snd_hda_attach_beep_device(...)
+#define snd_hda_detach_beep_device(...)
+#endif
+#endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
new file mode 100644
index 0000000..eb91641
--- /dev/null
+++ b/sound/pci/hda/hda_codec.c
@@ -0,0 +1,3141 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include <sound/asoundef.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include "hda_local.h"
+#include <sound/hda_hwdep.h>
+#include "hda_patch.h" /* codec presets */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* define this option here to hide as static */
+static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
+module_param(power_save, int, 0644);
+MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
+ "(in second, 0 = disable).");
+#endif
+
+/*
+ * vendor / preset table
+ */
+
+struct hda_vendor_id {
+ unsigned int id;
+ const char *name;
+};
+
+/* codec vendor labels */
+static struct hda_vendor_id hda_vendor_ids[] = {
+ { 0x1002, "ATI" },
+ { 0x1057, "Motorola" },
+ { 0x1095, "Silicon Image" },
+ { 0x10ec, "Realtek" },
+ { 0x1106, "VIA" },
+ { 0x111d, "IDT" },
+ { 0x11c1, "LSI" },
+ { 0x11d4, "Analog Devices" },
+ { 0x13f6, "C-Media" },
+ { 0x14f1, "Conexant" },
+ { 0x17e8, "Chrontel" },
+ { 0x1854, "LG" },
+ { 0x1aec, "Wolfson Microelectronics" },
+ { 0x434d, "C-Media" },
+ { 0x8384, "SigmaTel" },
+ {} /* terminator */
+};
+
+static const struct hda_codec_preset *hda_preset_tables[] = {
+#ifdef CONFIG_SND_HDA_CODEC_REALTEK
+ snd_hda_preset_realtek,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_CMEDIA
+ snd_hda_preset_cmedia,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_ANALOG
+ snd_hda_preset_analog,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
+ snd_hda_preset_sigmatel,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_SI3054
+ snd_hda_preset_si3054,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
+ snd_hda_preset_atihdmi,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_CONEXANT
+ snd_hda_preset_conexant,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_VIA
+ snd_hda_preset_via,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_NVHDMI
+ snd_hda_preset_nvhdmi,
+#endif
+ NULL
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_power_work(struct work_struct *work);
+static void hda_keep_power_on(struct hda_codec *codec);
+#else
+static inline void hda_keep_power_on(struct hda_codec *codec) {}
+#endif
+
+/**
+ * snd_hda_codec_read - send a command and get the response
+ * @codec: the HDA codec
+ * @nid: NID to send the command
+ * @direct: direct flag
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command and read the corresponding response.
+ *
+ * Returns the obtained response value, or -1 for an error.
+ */
+unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
+ int direct,
+ unsigned int verb, unsigned int parm)
+{
+ unsigned int res;
+ snd_hda_power_up(codec);
+ mutex_lock(&codec->bus->cmd_mutex);
+ if (!codec->bus->ops.command(codec, nid, direct, verb, parm))
+ res = codec->bus->ops.get_response(codec);
+ else
+ res = (unsigned int)-1;
+ mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
+ return res;
+}
+
+/**
+ * snd_hda_codec_write - send a single command without waiting for response
+ * @codec: the HDA codec
+ * @nid: NID to send the command
+ * @direct: direct flag
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
+ unsigned int verb, unsigned int parm)
+{
+ int err;
+ snd_hda_power_up(codec);
+ mutex_lock(&codec->bus->cmd_mutex);
+ err = codec->bus->ops.command(codec, nid, direct, verb, parm);
+ mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
+ return err;
+}
+
+/**
+ * snd_hda_sequence_write - sequence writes
+ * @codec: the HDA codec
+ * @seq: VERB array to send
+ *
+ * Send the commands sequentially from the given array.
+ * The array must be terminated with NID=0.
+ */
+void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq)
+{
+ for (; seq->nid; seq++)
+ snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param);
+}
+
+/**
+ * snd_hda_get_sub_nodes - get the range of sub nodes
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @start_id: the pointer to store the start NID
+ *
+ * Parse the NID and store the start NID of its sub-nodes.
+ * Returns the number of sub-nodes.
+ */
+int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *start_id)
+{
+ unsigned int parm;
+
+ parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT);
+ if (parm == -1)
+ return 0;
+ *start_id = (parm >> 16) & 0x7fff;
+ return (int)(parm & 0x7fff);
+}
+
+/**
+ * snd_hda_get_connections - get connection list
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @conn_list: connection list array
+ * @max_conns: max. number of connections to store
+ *
+ * Parses the connection list of the given widget and stores the list
+ * of NIDs.
+ *
+ * Returns the number of connections, or a negative error code.
+ */
+int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *conn_list, int max_conns)
+{
+ unsigned int parm;
+ int i, conn_len, conns;
+ unsigned int shift, num_elems, mask;
+ hda_nid_t prev_nid;
+
+ if (snd_BUG_ON(!conn_list || max_conns <= 0))
+ return -EINVAL;
+
+ parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
+ if (parm & AC_CLIST_LONG) {
+ /* long form */
+ shift = 16;
+ num_elems = 2;
+ } else {
+ /* short form */
+ shift = 8;
+ num_elems = 4;
+ }
+ conn_len = parm & AC_CLIST_LENGTH;
+ mask = (1 << (shift-1)) - 1;
+
+ if (!conn_len)
+ return 0; /* no connection */
+
+ if (conn_len == 1) {
+ /* single connection */
+ parm = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_LIST, 0);
+ conn_list[0] = parm & mask;
+ return 1;
+ }
+
+ /* multi connection */
+ conns = 0;
+ prev_nid = 0;
+ for (i = 0; i < conn_len; i++) {
+ int range_val;
+ hda_nid_t val, n;
+
+ if (i % num_elems == 0)
+ parm = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_LIST, i);
+ range_val = !!(parm & (1 << (shift-1))); /* ranges */
+ val = parm & mask;
+ parm >>= shift;
+ if (range_val) {
+ /* ranges between the previous and this one */
+ if (!prev_nid || prev_nid >= val) {
+ snd_printk(KERN_WARNING "hda_codec: "
+ "invalid dep_range_val %x:%x\n",
+ prev_nid, val);
+ continue;
+ }
+ for (n = prev_nid + 1; n <= val; n++) {
+ if (conns >= max_conns) {
+ snd_printk(KERN_ERR
+ "Too many connections\n");
+ return -EINVAL;
+ }
+ conn_list[conns++] = n;
+ }
+ } else {
+ if (conns >= max_conns) {
+ snd_printk(KERN_ERR "Too many connections\n");
+ return -EINVAL;
+ }
+ conn_list[conns++] = val;
+ }
+ prev_nid = val;
+ }
+ return conns;
+}
+
+
+/**
+ * snd_hda_queue_unsol_event - add an unsolicited event to queue
+ * @bus: the BUS
+ * @res: unsolicited event (lower 32bit of RIRB entry)
+ * @res_ex: codec addr and flags (upper 32bit or RIRB entry)
+ *
+ * Adds the given event to the queue. The events are processed in
+ * the workqueue asynchronously. Call this function in the interrupt
+ * hanlder when RIRB receives an unsolicited event.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
+{
+ struct hda_bus_unsolicited *unsol;
+ unsigned int wp;
+
+ unsol = bus->unsol;
+ if (!unsol)
+ return 0;
+
+ wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE;
+ unsol->wp = wp;
+
+ wp <<= 1;
+ unsol->queue[wp] = res;
+ unsol->queue[wp + 1] = res_ex;
+
+ schedule_work(&unsol->work);
+
+ return 0;
+}
+
+/*
+ * process queued unsolicited events
+ */
+static void process_unsol_events(struct work_struct *work)
+{
+ struct hda_bus_unsolicited *unsol =
+ container_of(work, struct hda_bus_unsolicited, work);
+ struct hda_bus *bus = unsol->bus;
+ struct hda_codec *codec;
+ unsigned int rp, caddr, res;
+
+ while (unsol->rp != unsol->wp) {
+ rp = (unsol->rp + 1) % HDA_UNSOL_QUEUE_SIZE;
+ unsol->rp = rp;
+ rp <<= 1;
+ res = unsol->queue[rp];
+ caddr = unsol->queue[rp + 1];
+ if (!(caddr & (1 << 4))) /* no unsolicited event? */
+ continue;
+ codec = bus->caddr_tbl[caddr & 0x0f];
+ if (codec && codec->patch_ops.unsol_event)
+ codec->patch_ops.unsol_event(codec, res);
+ }
+}
+
+/*
+ * initialize unsolicited queue
+ */
+static int __devinit init_unsol_queue(struct hda_bus *bus)
+{
+ struct hda_bus_unsolicited *unsol;
+
+ if (bus->unsol) /* already initialized */
+ return 0;
+
+ unsol = kzalloc(sizeof(*unsol), GFP_KERNEL);
+ if (!unsol) {
+ snd_printk(KERN_ERR "hda_codec: "
+ "can't allocate unsolicited queue\n");
+ return -ENOMEM;
+ }
+ INIT_WORK(&unsol->work, process_unsol_events);
+ unsol->bus = bus;
+ bus->unsol = unsol;
+ return 0;
+}
+
+/*
+ * destructor
+ */
+static void snd_hda_codec_free(struct hda_codec *codec);
+
+static int snd_hda_bus_free(struct hda_bus *bus)
+{
+ struct hda_codec *codec, *n;
+
+ if (!bus)
+ return 0;
+ if (bus->unsol) {
+ flush_scheduled_work();
+ kfree(bus->unsol);
+ }
+ list_for_each_entry_safe(codec, n, &bus->codec_list, list) {
+ snd_hda_codec_free(codec);
+ }
+ if (bus->ops.private_free)
+ bus->ops.private_free(bus);
+ kfree(bus);
+ return 0;
+}
+
+static int snd_hda_bus_dev_free(struct snd_device *device)
+{
+ struct hda_bus *bus = device->device_data;
+ return snd_hda_bus_free(bus);
+}
+
+/**
+ * snd_hda_bus_new - create a HDA bus
+ * @card: the card entry
+ * @temp: the template for hda_bus information
+ * @busp: the pointer to store the created bus instance
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int __devinit snd_hda_bus_new(struct snd_card *card,
+ const struct hda_bus_template *temp,
+ struct hda_bus **busp)
+{
+ struct hda_bus *bus;
+ int err;
+ static struct snd_device_ops dev_ops = {
+ .dev_free = snd_hda_bus_dev_free,
+ };
+
+ if (snd_BUG_ON(!temp))
+ return -EINVAL;
+ if (snd_BUG_ON(!temp->ops.command || !temp->ops.get_response))
+ return -EINVAL;
+
+ if (busp)
+ *busp = NULL;
+
+ bus = kzalloc(sizeof(*bus), GFP_KERNEL);
+ if (bus == NULL) {
+ snd_printk(KERN_ERR "can't allocate struct hda_bus\n");
+ return -ENOMEM;
+ }
+
+ bus->card = card;
+ bus->private_data = temp->private_data;
+ bus->pci = temp->pci;
+ bus->modelname = temp->modelname;
+ bus->ops = temp->ops;
+
+ mutex_init(&bus->cmd_mutex);
+ INIT_LIST_HEAD(&bus->codec_list);
+
+ err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops);
+ if (err < 0) {
+ snd_hda_bus_free(bus);
+ return err;
+ }
+ if (busp)
+ *busp = bus;
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_GENERIC
+#define is_generic_config(codec) \
+ (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic"))
+#else
+#define is_generic_config(codec) 0
+#endif
+
+/*
+ * find a matching codec preset
+ */
+static const struct hda_codec_preset __devinit *
+find_codec_preset(struct hda_codec *codec)
+{
+ const struct hda_codec_preset **tbl, *preset;
+
+ if (is_generic_config(codec))
+ return NULL; /* use the generic parser */
+
+ for (tbl = hda_preset_tables; *tbl; tbl++) {
+ for (preset = *tbl; preset->id; preset++) {
+ u32 mask = preset->mask;
+ if (preset->afg && preset->afg != codec->afg)
+ continue;
+ if (preset->mfg && preset->mfg != codec->mfg)
+ continue;
+ if (!mask)
+ mask = ~0;
+ if (preset->id == (codec->vendor_id & mask) &&
+ (!preset->rev ||
+ preset->rev == codec->revision_id))
+ return preset;
+ }
+ }
+ return NULL;
+}
+
+/*
+ * snd_hda_get_codec_name - store the codec name
+ */
+void snd_hda_get_codec_name(struct hda_codec *codec,
+ char *name, int namelen)
+{
+ const struct hda_vendor_id *c;
+ const char *vendor = NULL;
+ u16 vendor_id = codec->vendor_id >> 16;
+ char tmp[16];
+
+ for (c = hda_vendor_ids; c->id; c++) {
+ if (c->id == vendor_id) {
+ vendor = c->name;
+ break;
+ }
+ }
+ if (!vendor) {
+ sprintf(tmp, "Generic %04x", vendor_id);
+ vendor = tmp;
+ }
+ if (codec->preset && codec->preset->name)
+ snprintf(name, namelen, "%s %s", vendor, codec->preset->name);
+ else
+ snprintf(name, namelen, "%s ID %x", vendor,
+ codec->vendor_id & 0xffff);
+}
+
+/*
+ * look for an AFG and MFG nodes
+ */
+static void __devinit setup_fg_nodes(struct hda_codec *codec)
+{
+ int i, total_nodes;
+ hda_nid_t nid;
+
+ total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid);
+ for (i = 0; i < total_nodes; i++, nid++) {
+ unsigned int func;
+ func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE);
+ switch (func & 0xff) {
+ case AC_GRP_AUDIO_FUNCTION:
+ codec->afg = nid;
+ break;
+ case AC_GRP_MODEM_FUNCTION:
+ codec->mfg = nid;
+ break;
+ default:
+ break;
+ }
+ }
+}
+
+/*
+ * read widget caps for each widget and store in cache
+ */
+static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node)
+{
+ int i;
+ hda_nid_t nid;
+
+ codec->num_nodes = snd_hda_get_sub_nodes(codec, fg_node,
+ &codec->start_nid);
+ codec->wcaps = kmalloc(codec->num_nodes * 4, GFP_KERNEL);
+ if (!codec->wcaps)
+ return -ENOMEM;
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++)
+ codec->wcaps[i] = snd_hda_param_read(codec, nid,
+ AC_PAR_AUDIO_WIDGET_CAP);
+ return 0;
+}
+
+
+static void init_hda_cache(struct hda_cache_rec *cache,
+ unsigned int record_size);
+static void free_hda_cache(struct hda_cache_rec *cache);
+
+/*
+ * codec destructor
+ */
+static void snd_hda_codec_free(struct hda_codec *codec)
+{
+ if (!codec)
+ return;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+ flush_scheduled_work();
+#endif
+ list_del(&codec->list);
+ codec->bus->caddr_tbl[codec->addr] = NULL;
+ if (codec->patch_ops.free)
+ codec->patch_ops.free(codec);
+ free_hda_cache(&codec->amp_cache);
+ free_hda_cache(&codec->cmd_cache);
+ kfree(codec->wcaps);
+ kfree(codec);
+}
+
+/**
+ * snd_hda_codec_new - create a HDA codec
+ * @bus: the bus to assign
+ * @codec_addr: the codec address
+ * @codecp: the pointer to store the generated codec
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
+ struct hda_codec **codecp)
+{
+ struct hda_codec *codec;
+ char component[31];
+ int err;
+
+ if (snd_BUG_ON(!bus))
+ return -EINVAL;
+ if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS))
+ return -EINVAL;
+
+ if (bus->caddr_tbl[codec_addr]) {
+ snd_printk(KERN_ERR "hda_codec: "
+ "address 0x%x is already occupied\n", codec_addr);
+ return -EBUSY;
+ }
+
+ codec = kzalloc(sizeof(*codec), GFP_KERNEL);
+ if (codec == NULL) {
+ snd_printk(KERN_ERR "can't allocate struct hda_codec\n");
+ return -ENOMEM;
+ }
+
+ codec->bus = bus;
+ codec->addr = codec_addr;
+ mutex_init(&codec->spdif_mutex);
+ init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
+ init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
+ /* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
+ * the caller has to power down appropriatley after initialization
+ * phase.
+ */
+ hda_keep_power_on(codec);
+#endif
+
+ list_add_tail(&codec->list, &bus->codec_list);
+ bus->caddr_tbl[codec_addr] = codec;
+
+ codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT,
+ AC_PAR_VENDOR_ID);
+ if (codec->vendor_id == -1)
+ /* read again, hopefully the access method was corrected
+ * in the last read...
+ */
+ codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT,
+ AC_PAR_VENDOR_ID);
+ codec->subsystem_id = snd_hda_param_read(codec, AC_NODE_ROOT,
+ AC_PAR_SUBSYSTEM_ID);
+ codec->revision_id = snd_hda_param_read(codec, AC_NODE_ROOT,
+ AC_PAR_REV_ID);
+
+ setup_fg_nodes(codec);
+ if (!codec->afg && !codec->mfg) {
+ snd_printdd("hda_codec: no AFG or MFG node found\n");
+ snd_hda_codec_free(codec);
+ return -ENODEV;
+ }
+
+ if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) {
+ snd_printk(KERN_ERR "hda_codec: cannot malloc\n");
+ snd_hda_codec_free(codec);
+ return -ENOMEM;
+ }
+
+ if (!codec->subsystem_id) {
+ hda_nid_t nid = codec->afg ? codec->afg : codec->mfg;
+ codec->subsystem_id =
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_SUBSYSTEM_ID, 0);
+ }
+
+ codec->preset = find_codec_preset(codec);
+ /* audio codec should override the mixer name */
+ if (codec->afg || !*bus->card->mixername)
+ snd_hda_get_codec_name(codec, bus->card->mixername,
+ sizeof(bus->card->mixername));
+
+ if (is_generic_config(codec)) {
+ err = snd_hda_parse_generic_codec(codec);
+ goto patched;
+ }
+ if (codec->preset && codec->preset->patch) {
+ err = codec->preset->patch(codec);
+ goto patched;
+ }
+
+ /* call the default parser */
+ err = snd_hda_parse_generic_codec(codec);
+ if (err < 0)
+ printk(KERN_ERR "hda-codec: No codec parser is available\n");
+
+ patched:
+ if (err < 0) {
+ snd_hda_codec_free(codec);
+ return err;
+ }
+
+ if (codec->patch_ops.unsol_event)
+ init_unsol_queue(bus);
+
+ snd_hda_codec_proc_new(codec);
+#ifdef CONFIG_SND_HDA_HWDEP
+ snd_hda_create_hwdep(codec);
+#endif
+
+ sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id);
+ snd_component_add(codec->bus->card, component);
+
+ if (codecp)
+ *codecp = codec;
+ return 0;
+}
+
+/**
+ * snd_hda_codec_setup_stream - set up the codec for streaming
+ * @codec: the CODEC to set up
+ * @nid: the NID to set up
+ * @stream_tag: stream tag to pass, it's between 0x1 and 0xf.
+ * @channel_id: channel id to pass, zero based.
+ * @format: stream format.
+ */
+void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
+ u32 stream_tag,
+ int channel_id, int format)
+{
+ if (!nid)
+ return;
+
+ snd_printdd("hda_codec_setup_stream: "
+ "NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
+ nid, stream_tag, channel_id, format);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID,
+ (stream_tag << 4) | channel_id);
+ msleep(1);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format);
+}
+
+void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
+{
+ if (!nid)
+ return;
+
+ snd_printdd("hda_codec_cleanup_stream: NID=0x%x\n", nid);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+#if 0 /* keep the format */
+ msleep(1);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
+#endif
+}
+
+/*
+ * amp access functions
+ */
+
+/* FIXME: more better hash key? */
+#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24))
+#define INFO_AMP_CAPS (1<<0)
+#define INFO_AMP_VOL(ch) (1 << (1 + (ch)))
+
+/* initialize the hash table */
+static void __devinit init_hda_cache(struct hda_cache_rec *cache,
+ unsigned int record_size)
+{
+ memset(cache, 0, sizeof(*cache));
+ memset(cache->hash, 0xff, sizeof(cache->hash));
+ cache->record_size = record_size;
+}
+
+static void free_hda_cache(struct hda_cache_rec *cache)
+{
+ kfree(cache->buffer);
+}
+
+/* query the hash. allocate an entry if not found. */
+static struct hda_cache_head *get_alloc_hash(struct hda_cache_rec *cache,
+ u32 key)
+{
+ u16 idx = key % (u16)ARRAY_SIZE(cache->hash);
+ u16 cur = cache->hash[idx];
+ struct hda_cache_head *info;
+
+ while (cur != 0xffff) {
+ info = (struct hda_cache_head *)(cache->buffer +
+ cur * cache->record_size);
+ if (info->key == key)
+ return info;
+ cur = info->next;
+ }
+
+ /* add a new hash entry */
+ if (cache->num_entries >= cache->size) {
+ /* reallocate the array */
+ unsigned int new_size = cache->size + 64;
+ void *new_buffer;
+ new_buffer = kcalloc(new_size, cache->record_size, GFP_KERNEL);
+ if (!new_buffer) {
+ snd_printk(KERN_ERR "hda_codec: "
+ "can't malloc amp_info\n");
+ return NULL;
+ }
+ if (cache->buffer) {
+ memcpy(new_buffer, cache->buffer,
+ cache->size * cache->record_size);
+ kfree(cache->buffer);
+ }
+ cache->size = new_size;
+ cache->buffer = new_buffer;
+ }
+ cur = cache->num_entries++;
+ info = (struct hda_cache_head *)(cache->buffer +
+ cur * cache->record_size);
+ info->key = key;
+ info->val = 0;
+ info->next = cache->hash[idx];
+ cache->hash[idx] = cur;
+
+ return info;
+}
+
+/* query and allocate an amp hash entry */
+static inline struct hda_amp_info *
+get_alloc_amp_hash(struct hda_codec *codec, u32 key)
+{
+ return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key);
+}
+
+/*
+ * query AMP capabilities for the given widget and direction
+ */
+u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
+{
+ struct hda_amp_info *info;
+
+ info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0));
+ if (!info)
+ return 0;
+ if (!(info->head.val & INFO_AMP_CAPS)) {
+ if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
+ nid = codec->afg;
+ info->amp_caps = snd_hda_param_read(codec, nid,
+ direction == HDA_OUTPUT ?
+ AC_PAR_AMP_OUT_CAP :
+ AC_PAR_AMP_IN_CAP);
+ if (info->amp_caps)
+ info->head.val |= INFO_AMP_CAPS;
+ }
+ return info->amp_caps;
+}
+
+int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
+ unsigned int caps)
+{
+ struct hda_amp_info *info;
+
+ info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, dir, 0));
+ if (!info)
+ return -EINVAL;
+ info->amp_caps = caps;
+ info->head.val |= INFO_AMP_CAPS;
+ return 0;
+}
+
+/*
+ * read the current volume to info
+ * if the cache exists, read the cache value.
+ */
+static unsigned int get_vol_mute(struct hda_codec *codec,
+ struct hda_amp_info *info, hda_nid_t nid,
+ int ch, int direction, int index)
+{
+ u32 val, parm;
+
+ if (info->head.val & INFO_AMP_VOL(ch))
+ return info->vol[ch];
+
+ parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
+ parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
+ parm |= index;
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE, parm);
+ info->vol[ch] = val & 0xff;
+ info->head.val |= INFO_AMP_VOL(ch);
+ return info->vol[ch];
+}
+
+/*
+ * write the current volume in info to the h/w and update the cache
+ */
+static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
+ hda_nid_t nid, int ch, int direction, int index,
+ int val)
+{
+ u32 parm;
+
+ parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
+ parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
+ parm |= index << AC_AMP_SET_INDEX_SHIFT;
+ parm |= val;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
+ info->vol[ch] = val;
+}
+
+/*
+ * read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit.
+ */
+int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
+ int direction, int index)
+{
+ struct hda_amp_info *info;
+ info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
+ if (!info)
+ return 0;
+ return get_vol_mute(codec, info, nid, ch, direction, index);
+}
+
+/*
+ * update the AMP value, mask = bit mask to set, val = the value
+ */
+int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
+ int direction, int idx, int mask, int val)
+{
+ struct hda_amp_info *info;
+
+ info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx));
+ if (!info)
+ return 0;
+ val &= mask;
+ val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask;
+ if (info->vol[ch] == val)
+ return 0;
+ put_vol_mute(codec, info, nid, ch, direction, idx, val);
+ return 1;
+}
+
+/*
+ * update the AMP stereo with the same mask and value
+ */
+int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
+ int direction, int idx, int mask, int val)
+{
+ int ch, ret = 0;
+ for (ch = 0; ch < 2; ch++)
+ ret |= snd_hda_codec_amp_update(codec, nid, ch, direction,
+ idx, mask, val);
+ return ret;
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+/* resume the all amp commands from the cache */
+void snd_hda_codec_resume_amp(struct hda_codec *codec)
+{
+ struct hda_amp_info *buffer = codec->amp_cache.buffer;
+ int i;
+
+ for (i = 0; i < codec->amp_cache.size; i++, buffer++) {
+ u32 key = buffer->head.key;
+ hda_nid_t nid;
+ unsigned int idx, dir, ch;
+ if (!key)
+ continue;
+ nid = key & 0xff;
+ idx = (key >> 16) & 0xff;
+ dir = (key >> 24) & 0xff;
+ for (ch = 0; ch < 2; ch++) {
+ if (!(buffer->head.val & INFO_AMP_VOL(ch)))
+ continue;
+ put_vol_mute(codec, buffer, nid, ch, dir, idx,
+ buffer->vol[ch]);
+ }
+ }
+}
+#endif /* SND_HDA_NEEDS_RESUME */
+
+/* volume */
+int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 nid = get_amp_nid(kcontrol);
+ u8 chs = get_amp_channels(kcontrol);
+ int dir = get_amp_direction(kcontrol);
+ u32 caps;
+
+ caps = query_amp_caps(codec, nid, dir);
+ /* num steps */
+ caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+ if (!caps) {
+ printk(KERN_WARNING "hda_codec: "
+ "num_steps = 0 for NID=0x%x (ctl = %s)\n", nid,
+ kcontrol->id.name);
+ return -EINVAL;
+ }
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = chs == 3 ? 2 : 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = caps;
+ return 0;
+}
+
+int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ int chs = get_amp_channels(kcontrol);
+ int dir = get_amp_direction(kcontrol);
+ int idx = get_amp_index(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+
+ if (chs & 1)
+ *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx)
+ & HDA_AMP_VOLMASK;
+ if (chs & 2)
+ *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx)
+ & HDA_AMP_VOLMASK;
+ return 0;
+}
+
+int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ int chs = get_amp_channels(kcontrol);
+ int dir = get_amp_direction(kcontrol);
+ int idx = get_amp_index(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change = 0;
+
+ snd_hda_power_up(codec);
+ if (chs & 1) {
+ change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+ 0x7f, *valp);
+ valp++;
+ }
+ if (chs & 2)
+ change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
+ 0x7f, *valp);
+ snd_hda_power_down(codec);
+ return change;
+}
+
+int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *_tlv)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ int dir = get_amp_direction(kcontrol);
+ u32 caps, val1, val2;
+
+ if (size < 4 * sizeof(unsigned int))
+ return -ENOMEM;
+ caps = query_amp_caps(codec, nid, dir);
+ val2 = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT;
+ val2 = (val2 + 1) * 25;
+ val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
+ val1 = ((int)val1) * ((int)val2);
+ if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
+ return -EFAULT;
+ if (put_user(2 * sizeof(unsigned int), _tlv + 1))
+ return -EFAULT;
+ if (put_user(val1, _tlv + 2))
+ return -EFAULT;
+ if (put_user(val2, _tlv + 3))
+ return -EFAULT;
+ return 0;
+}
+
+/*
+ * set (static) TLV for virtual master volume; recalculated as max 0dB
+ */
+void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
+ unsigned int *tlv)
+{
+ u32 caps;
+ int nums, step;
+
+ caps = query_amp_caps(codec, nid, dir);
+ nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+ step = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT;
+ step = (step + 1) * 25;
+ tlv[0] = SNDRV_CTL_TLVT_DB_SCALE;
+ tlv[1] = 2 * sizeof(unsigned int);
+ tlv[2] = -nums * step;
+ tlv[3] = step;
+}
+
+/* find a mixer control element with the given name */
+static struct snd_kcontrol *
+_snd_hda_find_mixer_ctl(struct hda_codec *codec,
+ const char *name, int idx)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ id.index = idx;
+ strcpy(id.name, name);
+ return snd_ctl_find_id(codec->bus->card, &id);
+}
+
+struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
+ const char *name)
+{
+ return _snd_hda_find_mixer_ctl(codec, name, 0);
+}
+
+/* create a virtual master control and add slaves */
+int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
+ unsigned int *tlv, const char **slaves)
+{
+ struct snd_kcontrol *kctl;
+ const char **s;
+ int err;
+
+ for (s = slaves; *s && !snd_hda_find_mixer_ctl(codec, *s); s++)
+ ;
+ if (!*s) {
+ snd_printdd("No slave found for %s\n", name);
+ return 0;
+ }
+ kctl = snd_ctl_make_virtual_master(name, tlv);
+ if (!kctl)
+ return -ENOMEM;
+ err = snd_ctl_add(codec->bus->card, kctl);
+ if (err < 0)
+ return err;
+
+ for (s = slaves; *s; s++) {
+ struct snd_kcontrol *sctl;
+
+ sctl = snd_hda_find_mixer_ctl(codec, *s);
+ if (!sctl) {
+ snd_printdd("Cannot find slave %s, skipped\n", *s);
+ continue;
+ }
+ err = snd_ctl_add_slave(kctl, sctl);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* switch */
+int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int chs = get_amp_channels(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = chs == 3 ? 2 : 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ int chs = get_amp_channels(kcontrol);
+ int dir = get_amp_direction(kcontrol);
+ int idx = get_amp_index(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+
+ if (chs & 1)
+ *valp++ = (snd_hda_codec_amp_read(codec, nid, 0, dir, idx) &
+ HDA_AMP_MUTE) ? 0 : 1;
+ if (chs & 2)
+ *valp = (snd_hda_codec_amp_read(codec, nid, 1, dir, idx) &
+ HDA_AMP_MUTE) ? 0 : 1;
+ return 0;
+}
+
+int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ int chs = get_amp_channels(kcontrol);
+ int dir = get_amp_direction(kcontrol);
+ int idx = get_amp_index(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change = 0;
+
+ snd_hda_power_up(codec);
+ if (chs & 1) {
+ change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+ valp++;
+ }
+ if (chs & 2)
+ change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (codec->patch_ops.check_power_status)
+ codec->patch_ops.check_power_status(codec, nid);
+#endif
+ snd_hda_power_down(codec);
+ return change;
+}
+
+/*
+ * bound volume controls
+ *
+ * bind multiple volumes (# indices, from 0)
+ */
+
+#define AMP_VAL_IDX_SHIFT 19
+#define AMP_VAL_IDX_MASK (0x0f<<19)
+
+int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned long pval;
+ int err;
+
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ pval = kcontrol->private_value;
+ kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */
+ err = snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
+ kcontrol->private_value = pval;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned long pval;
+ int i, indices, err = 0, change = 0;
+
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ pval = kcontrol->private_value;
+ indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT;
+ for (i = 0; i < indices; i++) {
+ kcontrol->private_value = (pval & ~AMP_VAL_IDX_MASK) |
+ (i << AMP_VAL_IDX_SHIFT);
+ err = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+ if (err < 0)
+ break;
+ change |= err;
+ }
+ kcontrol->private_value = pval;
+ mutex_unlock(&codec->spdif_mutex);
+ return err < 0 ? err : change;
+}
+
+/*
+ * generic bound volume/swtich controls
+ */
+int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ kcontrol->private_value = *c->values;
+ err = c->ops->info(kcontrol, uinfo);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ kcontrol->private_value = *c->values;
+ err = c->ops->get(kcontrol, ucontrol);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ unsigned long *vals;
+ int err = 0, change = 0;
+
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ for (vals = c->values; *vals; vals++) {
+ kcontrol->private_value = *vals;
+ err = c->ops->put(kcontrol, ucontrol);
+ if (err < 0)
+ break;
+ change |= err;
+ }
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err < 0 ? err : change;
+}
+
+int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ kcontrol->private_value = *c->values;
+ err = c->ops->tlv(kcontrol, op_flag, size, tlv);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+struct hda_ctl_ops snd_hda_bind_vol = {
+ .info = snd_hda_mixer_amp_volume_info,
+ .get = snd_hda_mixer_amp_volume_get,
+ .put = snd_hda_mixer_amp_volume_put,
+ .tlv = snd_hda_mixer_amp_tlv
+};
+
+struct hda_ctl_ops snd_hda_bind_sw = {
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = snd_hda_mixer_amp_switch_put,
+ .tlv = snd_hda_mixer_amp_tlv
+};
+
+/*
+ * SPDIF out controls
+ */
+
+static int snd_hda_spdif_mask_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_hda_spdif_cmask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_NONAUDIO |
+ IEC958_AES0_CON_EMPHASIS_5015 |
+ IEC958_AES0_CON_NOT_COPYRIGHT;
+ ucontrol->value.iec958.status[1] = IEC958_AES1_CON_CATEGORY |
+ IEC958_AES1_CON_ORIGINAL;
+ return 0;
+}
+
+static int snd_hda_spdif_pmask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PRO_EMPHASIS_5015;
+ return 0;
+}
+
+static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff;
+ ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff;
+
+ return 0;
+}
+
+/* convert from SPDIF status bits to HDA SPDIF bits
+ * bit 0 (DigEn) is always set zero (to be filled later)
+ */
+static unsigned short convert_from_spdif_status(unsigned int sbits)
+{
+ unsigned short val = 0;
+
+ if (sbits & IEC958_AES0_PROFESSIONAL)
+ val |= AC_DIG1_PROFESSIONAL;
+ if (sbits & IEC958_AES0_NONAUDIO)
+ val |= AC_DIG1_NONAUDIO;
+ if (sbits & IEC958_AES0_PROFESSIONAL) {
+ if ((sbits & IEC958_AES0_PRO_EMPHASIS) ==
+ IEC958_AES0_PRO_EMPHASIS_5015)
+ val |= AC_DIG1_EMPHASIS;
+ } else {
+ if ((sbits & IEC958_AES0_CON_EMPHASIS) ==
+ IEC958_AES0_CON_EMPHASIS_5015)
+ val |= AC_DIG1_EMPHASIS;
+ if (!(sbits & IEC958_AES0_CON_NOT_COPYRIGHT))
+ val |= AC_DIG1_COPYRIGHT;
+ if (sbits & (IEC958_AES1_CON_ORIGINAL << 8))
+ val |= AC_DIG1_LEVEL;
+ val |= sbits & (IEC958_AES1_CON_CATEGORY << 8);
+ }
+ return val;
+}
+
+/* convert to SPDIF status bits from HDA SPDIF bits
+ */
+static unsigned int convert_to_spdif_status(unsigned short val)
+{
+ unsigned int sbits = 0;
+
+ if (val & AC_DIG1_NONAUDIO)
+ sbits |= IEC958_AES0_NONAUDIO;
+ if (val & AC_DIG1_PROFESSIONAL)
+ sbits |= IEC958_AES0_PROFESSIONAL;
+ if (sbits & IEC958_AES0_PROFESSIONAL) {
+ if (sbits & AC_DIG1_EMPHASIS)
+ sbits |= IEC958_AES0_PRO_EMPHASIS_5015;
+ } else {
+ if (val & AC_DIG1_EMPHASIS)
+ sbits |= IEC958_AES0_CON_EMPHASIS_5015;
+ if (!(val & AC_DIG1_COPYRIGHT))
+ sbits |= IEC958_AES0_CON_NOT_COPYRIGHT;
+ if (val & AC_DIG1_LEVEL)
+ sbits |= (IEC958_AES1_CON_ORIGINAL << 8);
+ sbits |= val & (0x7f << 8);
+ }
+ return sbits;
+}
+
+/* set digital convert verbs both for the given NID and its slaves */
+static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
+ int verb, int val)
+{
+ hda_nid_t *d;
+
+ snd_hda_codec_write_cache(codec, nid, 0, verb, val);
+ d = codec->slave_dig_outs;
+ if (!d)
+ return;
+ for (; *d; d++)
+ snd_hda_codec_write_cache(codec, *d, 0, verb, val);
+}
+
+static inline void set_dig_out_convert(struct hda_codec *codec, hda_nid_t nid,
+ int dig1, int dig2)
+{
+ if (dig1 != -1)
+ set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_1, dig1);
+ if (dig2 != -1)
+ set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_2, dig2);
+}
+
+static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ unsigned short val;
+ int change;
+
+ mutex_lock(&codec->spdif_mutex);
+ codec->spdif_status = ucontrol->value.iec958.status[0] |
+ ((unsigned int)ucontrol->value.iec958.status[1] << 8) |
+ ((unsigned int)ucontrol->value.iec958.status[2] << 16) |
+ ((unsigned int)ucontrol->value.iec958.status[3] << 24);
+ val = convert_from_spdif_status(codec->spdif_status);
+ val |= codec->spdif_ctls & 1;
+ change = codec->spdif_ctls != val;
+ codec->spdif_ctls = val;
+
+ if (change)
+ set_dig_out_convert(codec, nid, val & 0xff, (val >> 8) & 0xff);
+
+ mutex_unlock(&codec->spdif_mutex);
+ return change;
+}
+
+#define snd_hda_spdif_out_switch_info snd_ctl_boolean_mono_info
+
+static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = codec->spdif_ctls & AC_DIG1_ENABLE;
+ return 0;
+}
+
+static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ unsigned short val;
+ int change;
+
+ mutex_lock(&codec->spdif_mutex);
+ val = codec->spdif_ctls & ~AC_DIG1_ENABLE;
+ if (ucontrol->value.integer.value[0])
+ val |= AC_DIG1_ENABLE;
+ change = codec->spdif_ctls != val;
+ if (change) {
+ codec->spdif_ctls = val;
+ set_dig_out_convert(codec, nid, val & 0xff, -1);
+ /* unmute amp switch (if any) */
+ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
+ (val & AC_DIG1_ENABLE))
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, 0);
+ }
+ mutex_unlock(&codec->spdif_mutex);
+ return change;
+}
+
+static struct snd_kcontrol_new dig_mixes[] = {
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_hda_spdif_mask_info,
+ .get = snd_hda_spdif_cmask_get,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+ .info = snd_hda_spdif_mask_info,
+ .get = snd_hda_spdif_pmask_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_hda_spdif_mask_info,
+ .get = snd_hda_spdif_default_get,
+ .put = snd_hda_spdif_default_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH),
+ .info = snd_hda_spdif_out_switch_info,
+ .get = snd_hda_spdif_out_switch_get,
+ .put = snd_hda_spdif_out_switch_put,
+ },
+ { } /* end */
+};
+
+#define SPDIF_MAX_IDX 4 /* 4 instances should be enough to probe */
+
+/**
+ * snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls
+ * @codec: the HDA codec
+ * @nid: audio out widget NID
+ *
+ * Creates controls related with the SPDIF output.
+ * Called from each patch supporting the SPDIF out.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
+{
+ int err;
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_new *dig_mix;
+ int idx;
+
+ for (idx = 0; idx < SPDIF_MAX_IDX; idx++) {
+ if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Playback Switch",
+ idx))
+ break;
+ }
+ if (idx >= SPDIF_MAX_IDX) {
+ printk(KERN_ERR "hda_codec: too many IEC958 outputs\n");
+ return -EBUSY;
+ }
+ for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
+ kctl = snd_ctl_new1(dig_mix, codec);
+ kctl->id.index = idx;
+ kctl->private_value = nid;
+ err = snd_ctl_add(codec->bus->card, kctl);
+ if (err < 0)
+ return err;
+ }
+ codec->spdif_ctls =
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1, 0);
+ codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls);
+ return 0;
+}
+
+/*
+ * SPDIF sharing with analog output
+ */
+static int spdif_share_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = mout->share_spdif;
+ return 0;
+}
+
+static int spdif_share_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol);
+ mout->share_spdif = !!ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static struct snd_kcontrol_new spdif_share_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Default PCM Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = spdif_share_sw_get,
+ .put = spdif_share_sw_put,
+};
+
+int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
+ struct hda_multi_out *mout)
+{
+ if (!mout->dig_out_nid)
+ return 0;
+ /* ATTENTION: here mout is passed as private_data, instead of codec */
+ return snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&spdif_share_sw, mout));
+}
+
+/*
+ * SPDIF input
+ */
+
+#define snd_hda_spdif_in_switch_info snd_hda_spdif_out_switch_info
+
+static int snd_hda_spdif_in_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = codec->spdif_in_enable;
+ return 0;
+}
+
+static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ unsigned int val = !!ucontrol->value.integer.value[0];
+ int change;
+
+ mutex_lock(&codec->spdif_mutex);
+ change = codec->spdif_in_enable != val;
+ if (change) {
+ codec->spdif_in_enable = val;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1, val);
+ }
+ mutex_unlock(&codec->spdif_mutex);
+ return change;
+}
+
+static int snd_hda_spdif_in_status_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ unsigned short val;
+ unsigned int sbits;
+
+ val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0);
+ sbits = convert_to_spdif_status(val);
+ ucontrol->value.iec958.status[0] = sbits;
+ ucontrol->value.iec958.status[1] = sbits >> 8;
+ ucontrol->value.iec958.status[2] = sbits >> 16;
+ ucontrol->value.iec958.status[3] = sbits >> 24;
+ return 0;
+}
+
+static struct snd_kcontrol_new dig_in_ctls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH),
+ .info = snd_hda_spdif_in_switch_info,
+ .get = snd_hda_spdif_in_switch_get,
+ .put = snd_hda_spdif_in_switch_put,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT),
+ .info = snd_hda_spdif_mask_info,
+ .get = snd_hda_spdif_in_status_get,
+ },
+ { } /* end */
+};
+
+/**
+ * snd_hda_create_spdif_in_ctls - create Input SPDIF-related controls
+ * @codec: the HDA codec
+ * @nid: audio in widget NID
+ *
+ * Creates controls related with the SPDIF input.
+ * Called from each patch supporting the SPDIF in.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
+{
+ int err;
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_new *dig_mix;
+ int idx;
+
+ for (idx = 0; idx < SPDIF_MAX_IDX; idx++) {
+ if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Capture Switch",
+ idx))
+ break;
+ }
+ if (idx >= SPDIF_MAX_IDX) {
+ printk(KERN_ERR "hda_codec: too many IEC958 inputs\n");
+ return -EBUSY;
+ }
+ for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) {
+ kctl = snd_ctl_new1(dig_mix, codec);
+ kctl->private_value = nid;
+ err = snd_ctl_add(codec->bus->card, kctl);
+ if (err < 0)
+ return err;
+ }
+ codec->spdif_in_enable =
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1, 0) &
+ AC_DIG1_ENABLE;
+ return 0;
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+/*
+ * command cache
+ */
+
+/* build a 32bit cache key with the widget id and the command parameter */
+#define build_cmd_cache_key(nid, verb) ((verb << 8) | nid)
+#define get_cmd_cache_nid(key) ((key) & 0xff)
+#define get_cmd_cache_cmd(key) (((key) >> 8) & 0xffff)
+
+/**
+ * snd_hda_codec_write_cache - send a single command with caching
+ * @codec: the HDA codec
+ * @nid: NID to send the command
+ * @direct: direct flag
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
+ int direct, unsigned int verb, unsigned int parm)
+{
+ int err;
+ snd_hda_power_up(codec);
+ mutex_lock(&codec->bus->cmd_mutex);
+ err = codec->bus->ops.command(codec, nid, direct, verb, parm);
+ if (!err) {
+ struct hda_cache_head *c;
+ u32 key = build_cmd_cache_key(nid, verb);
+ c = get_alloc_hash(&codec->cmd_cache, key);
+ if (c)
+ c->val = parm;
+ }
+ mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
+ return err;
+}
+
+/* resume the all commands from the cache */
+void snd_hda_codec_resume_cache(struct hda_codec *codec)
+{
+ struct hda_cache_head *buffer = codec->cmd_cache.buffer;
+ int i;
+
+ for (i = 0; i < codec->cmd_cache.size; i++, buffer++) {
+ u32 key = buffer->key;
+ if (!key)
+ continue;
+ snd_hda_codec_write(codec, get_cmd_cache_nid(key), 0,
+ get_cmd_cache_cmd(key), buffer->val);
+ }
+}
+
+/**
+ * snd_hda_sequence_write_cache - sequence writes with caching
+ * @codec: the HDA codec
+ * @seq: VERB array to send
+ *
+ * Send the commands sequentially from the given array.
+ * Thte commands are recorded on cache for power-save and resume.
+ * The array must be terminated with NID=0.
+ */
+void snd_hda_sequence_write_cache(struct hda_codec *codec,
+ const struct hda_verb *seq)
+{
+ for (; seq->nid; seq++)
+ snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb,
+ seq->param);
+}
+#endif /* SND_HDA_NEEDS_RESUME */
+
+/*
+ * set power state of the codec
+ */
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+ unsigned int power_state)
+{
+ hda_nid_t nid;
+ int i;
+
+ snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+ power_state);
+ msleep(10); /* partial workaround for "azx_get_response timeout" */
+
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ unsigned int wcaps = get_wcaps(codec, nid);
+ if (wcaps & AC_WCAP_POWER) {
+ unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
+ AC_WCAP_TYPE_SHIFT;
+ if (wid_type == AC_WID_PIN) {
+ unsigned int pincap;
+ /*
+ * don't power down the widget if it controls
+ * eapd and EAPD_BTLENABLE is set.
+ */
+ pincap = snd_hda_param_read(codec, nid,
+ AC_PAR_PIN_CAP);
+ if (pincap & AC_PINCAP_EAPD) {
+ int eapd = snd_hda_codec_read(codec,
+ nid, 0,
+ AC_VERB_GET_EAPD_BTLENABLE, 0);
+ eapd &= 0x02;
+ if (power_state == AC_PWRST_D3 && eapd)
+ continue;
+ }
+ }
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_POWER_STATE,
+ power_state);
+ }
+ }
+
+ if (power_state == AC_PWRST_D0) {
+ unsigned long end_time;
+ int state;
+ msleep(10);
+ /* wait until the codec reachs to D0 */
+ end_time = jiffies + msecs_to_jiffies(500);
+ do {
+ state = snd_hda_codec_read(codec, fg, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+ if (state == power_state)
+ break;
+ msleep(1);
+ } while (time_after_eq(end_time, jiffies));
+ }
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+/*
+ * call suspend and power-down; used both from PM and power-save
+ */
+static void hda_call_codec_suspend(struct hda_codec *codec)
+{
+ if (codec->patch_ops.suspend)
+ codec->patch_ops.suspend(codec, PMSG_SUSPEND);
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D3);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+ codec->power_on = 0;
+ codec->power_transition = 0;
+#endif
+}
+
+/*
+ * kick up codec; used both from PM and power-save
+ */
+static void hda_call_codec_resume(struct hda_codec *codec)
+{
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D0);
+ if (codec->patch_ops.resume)
+ codec->patch_ops.resume(codec);
+ else {
+ if (codec->patch_ops.init)
+ codec->patch_ops.init(codec);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ }
+}
+#endif /* SND_HDA_NEEDS_RESUME */
+
+
+/**
+ * snd_hda_build_controls - build mixer controls
+ * @bus: the BUS
+ *
+ * Creates mixer controls for each codec included in the bus.
+ *
+ * Returns 0 if successful, otherwise a negative error code.
+ */
+int __devinit snd_hda_build_controls(struct hda_bus *bus)
+{
+ struct hda_codec *codec;
+
+ list_for_each_entry(codec, &bus->codec_list, list) {
+ int err = 0;
+ /* fake as if already powered-on */
+ hda_keep_power_on(codec);
+ /* then fire up */
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D0);
+ /* continue to initialize... */
+ if (codec->patch_ops.init)
+ err = codec->patch_ops.init(codec);
+ if (!err && codec->patch_ops.build_controls)
+ err = codec->patch_ops.build_controls(codec);
+ snd_hda_power_down(codec);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/*
+ * stream formats
+ */
+struct hda_rate_tbl {
+ unsigned int hz;
+ unsigned int alsa_bits;
+ unsigned int hda_fmt;
+};
+
+static struct hda_rate_tbl rate_bits[] = {
+ /* rate in Hz, ALSA rate bitmask, HDA format value */
+
+ /* autodetected value used in snd_hda_query_supported_pcm */
+ { 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */
+ { 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */
+ { 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */
+ { 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */
+ { 32000, SNDRV_PCM_RATE_32000, 0x0a00 }, /* 2/3 x 48 */
+ { 44100, SNDRV_PCM_RATE_44100, 0x4000 }, /* 44 */
+ { 48000, SNDRV_PCM_RATE_48000, 0x0000 }, /* 48 */
+ { 88200, SNDRV_PCM_RATE_88200, 0x4800 }, /* 2 x 44 */
+ { 96000, SNDRV_PCM_RATE_96000, 0x0800 }, /* 2 x 48 */
+ { 176400, SNDRV_PCM_RATE_176400, 0x5800 },/* 4 x 44 */
+ { 192000, SNDRV_PCM_RATE_192000, 0x1800 }, /* 4 x 48 */
+#define AC_PAR_PCM_RATE_BITS 11
+ /* up to bits 10, 384kHZ isn't supported properly */
+
+ /* not autodetected value */
+ { 9600, SNDRV_PCM_RATE_KNOT, 0x0400 }, /* 1/5 x 48 */
+
+ { 0 } /* terminator */
+};
+
+/**
+ * snd_hda_calc_stream_format - calculate format bitset
+ * @rate: the sample rate
+ * @channels: the number of channels
+ * @format: the PCM format (SNDRV_PCM_FORMAT_XXX)
+ * @maxbps: the max. bps
+ *
+ * Calculate the format bitset from the given rate, channels and th PCM format.
+ *
+ * Return zero if invalid.
+ */
+unsigned int snd_hda_calc_stream_format(unsigned int rate,
+ unsigned int channels,
+ unsigned int format,
+ unsigned int maxbps)
+{
+ int i;
+ unsigned int val = 0;
+
+ for (i = 0; rate_bits[i].hz; i++)
+ if (rate_bits[i].hz == rate) {
+ val = rate_bits[i].hda_fmt;
+ break;
+ }
+ if (!rate_bits[i].hz) {
+ snd_printdd("invalid rate %d\n", rate);
+ return 0;
+ }
+
+ if (channels == 0 || channels > 8) {
+ snd_printdd("invalid channels %d\n", channels);
+ return 0;
+ }
+ val |= channels - 1;
+
+ switch (snd_pcm_format_width(format)) {
+ case 8: val |= 0x00; break;
+ case 16: val |= 0x10; break;
+ case 20:
+ case 24:
+ case 32:
+ if (maxbps >= 32)
+ val |= 0x40;
+ else if (maxbps >= 24)
+ val |= 0x30;
+ else
+ val |= 0x20;
+ break;
+ default:
+ snd_printdd("invalid format width %d\n",
+ snd_pcm_format_width(format));
+ return 0;
+ }
+
+ return val;
+}
+
+/**
+ * snd_hda_query_supported_pcm - query the supported PCM rates and formats
+ * @codec: the HDA codec
+ * @nid: NID to query
+ * @ratesp: the pointer to store the detected rate bitflags
+ * @formatsp: the pointer to store the detected formats
+ * @bpsp: the pointer to store the detected format widths
+ *
+ * Queries the supported PCM rates and formats. The NULL @ratesp, @formatsp
+ * or @bsps argument is ignored.
+ *
+ * Returns 0 if successful, otherwise a negative error code.
+ */
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+ u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
+{
+ int i;
+ unsigned int val, streams;
+
+ val = 0;
+ if (nid != codec->afg &&
+ (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) {
+ val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+ if (val == -1)
+ return -EIO;
+ }
+ if (!val)
+ val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+
+ if (ratesp) {
+ u32 rates = 0;
+ for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++) {
+ if (val & (1 << i))
+ rates |= rate_bits[i].alsa_bits;
+ }
+ *ratesp = rates;
+ }
+
+ if (formatsp || bpsp) {
+ u64 formats = 0;
+ unsigned int bps;
+ unsigned int wcaps;
+
+ wcaps = get_wcaps(codec, nid);
+ streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+ if (streams == -1)
+ return -EIO;
+ if (!streams) {
+ streams = snd_hda_param_read(codec, codec->afg,
+ AC_PAR_STREAM);
+ if (streams == -1)
+ return -EIO;
+ }
+
+ bps = 0;
+ if (streams & AC_SUPFMT_PCM) {
+ if (val & AC_SUPPCM_BITS_8) {
+ formats |= SNDRV_PCM_FMTBIT_U8;
+ bps = 8;
+ }
+ if (val & AC_SUPPCM_BITS_16) {
+ formats |= SNDRV_PCM_FMTBIT_S16_LE;
+ bps = 16;
+ }
+ if (wcaps & AC_WCAP_DIGITAL) {
+ if (val & AC_SUPPCM_BITS_32)
+ formats |= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE;
+ if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24))
+ formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ if (val & AC_SUPPCM_BITS_24)
+ bps = 24;
+ else if (val & AC_SUPPCM_BITS_20)
+ bps = 20;
+ } else if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24|
+ AC_SUPPCM_BITS_32)) {
+ formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ if (val & AC_SUPPCM_BITS_32)
+ bps = 32;
+ else if (val & AC_SUPPCM_BITS_24)
+ bps = 24;
+ else if (val & AC_SUPPCM_BITS_20)
+ bps = 20;
+ }
+ }
+ else if (streams == AC_SUPFMT_FLOAT32) {
+ /* should be exclusive */
+ formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
+ bps = 32;
+ } else if (streams == AC_SUPFMT_AC3) {
+ /* should be exclusive */
+ /* temporary hack: we have still no proper support
+ * for the direct AC3 stream...
+ */
+ formats |= SNDRV_PCM_FMTBIT_U8;
+ bps = 8;
+ }
+ if (formatsp)
+ *formatsp = formats;
+ if (bpsp)
+ *bpsp = bps;
+ }
+
+ return 0;
+}
+
+/**
+ * snd_hda_is_supported_format - check whether the given node supports
+ * the format val
+ *
+ * Returns 1 if supported, 0 if not.
+ */
+int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int format)
+{
+ int i;
+ unsigned int val = 0, rate, stream;
+
+ if (nid != codec->afg &&
+ (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) {
+ val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+ if (val == -1)
+ return 0;
+ }
+ if (!val) {
+ val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+ if (val == -1)
+ return 0;
+ }
+
+ rate = format & 0xff00;
+ for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++)
+ if (rate_bits[i].hda_fmt == rate) {
+ if (val & (1 << i))
+ break;
+ return 0;
+ }
+ if (i >= AC_PAR_PCM_RATE_BITS)
+ return 0;
+
+ stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+ if (stream == -1)
+ return 0;
+ if (!stream && nid != codec->afg)
+ stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM);
+ if (!stream || stream == -1)
+ return 0;
+
+ if (stream & AC_SUPFMT_PCM) {
+ switch (format & 0xf0) {
+ case 0x00:
+ if (!(val & AC_SUPPCM_BITS_8))
+ return 0;
+ break;
+ case 0x10:
+ if (!(val & AC_SUPPCM_BITS_16))
+ return 0;
+ break;
+ case 0x20:
+ if (!(val & AC_SUPPCM_BITS_20))
+ return 0;
+ break;
+ case 0x30:
+ if (!(val & AC_SUPPCM_BITS_24))
+ return 0;
+ break;
+ case 0x40:
+ if (!(val & AC_SUPPCM_BITS_32))
+ return 0;
+ break;
+ default:
+ return 0;
+ }
+ } else {
+ /* FIXME: check for float32 and AC3? */
+ }
+
+ return 1;
+}
+
+/*
+ * PCM stuff
+ */
+static int hda_pcm_default_open_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static int hda_pcm_default_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format);
+ return 0;
+}
+
+static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+ return 0;
+}
+
+static int __devinit set_pcm_default_values(struct hda_codec *codec,
+ struct hda_pcm_stream *info)
+{
+ /* query support PCM information from the given NID */
+ if (info->nid && (!info->rates || !info->formats)) {
+ snd_hda_query_supported_pcm(codec, info->nid,
+ info->rates ? NULL : &info->rates,
+ info->formats ? NULL : &info->formats,
+ info->maxbps ? NULL : &info->maxbps);
+ }
+ if (info->ops.open == NULL)
+ info->ops.open = hda_pcm_default_open_close;
+ if (info->ops.close == NULL)
+ info->ops.close = hda_pcm_default_open_close;
+ if (info->ops.prepare == NULL) {
+ if (snd_BUG_ON(!info->nid))
+ return -EINVAL;
+ info->ops.prepare = hda_pcm_default_prepare;
+ }
+ if (info->ops.cleanup == NULL) {
+ if (snd_BUG_ON(!info->nid))
+ return -EINVAL;
+ info->ops.cleanup = hda_pcm_default_cleanup;
+ }
+ return 0;
+}
+
+/**
+ * snd_hda_build_pcms - build PCM information
+ * @bus: the BUS
+ *
+ * Create PCM information for each codec included in the bus.
+ *
+ * The build_pcms codec patch is requested to set up codec->num_pcms and
+ * codec->pcm_info properly. The array is referred by the top-level driver
+ * to create its PCM instances.
+ * The allocated codec->pcm_info should be released in codec->patch_ops.free
+ * callback.
+ *
+ * At least, substreams, channels_min and channels_max must be filled for
+ * each stream. substreams = 0 indicates that the stream doesn't exist.
+ * When rates and/or formats are zero, the supported values are queried
+ * from the given nid. The nid is used also by the default ops.prepare
+ * and ops.cleanup callbacks.
+ *
+ * The driver needs to call ops.open in its open callback. Similarly,
+ * ops.close is supposed to be called in the close callback.
+ * ops.prepare should be called in the prepare or hw_params callback
+ * with the proper parameters for set up.
+ * ops.cleanup should be called in hw_free for clean up of streams.
+ *
+ * This function returns 0 if successfull, or a negative error code.
+ */
+int __devinit snd_hda_build_pcms(struct hda_bus *bus)
+{
+ struct hda_codec *codec;
+
+ list_for_each_entry(codec, &bus->codec_list, list) {
+ unsigned int pcm, s;
+ int err;
+ if (!codec->patch_ops.build_pcms)
+ continue;
+ err = codec->patch_ops.build_pcms(codec);
+ if (err < 0)
+ return err;
+ for (pcm = 0; pcm < codec->num_pcms; pcm++) {
+ for (s = 0; s < 2; s++) {
+ struct hda_pcm_stream *info;
+ info = &codec->pcm_info[pcm].stream[s];
+ if (!info->substreams)
+ continue;
+ err = set_pcm_default_values(codec, info);
+ if (err < 0)
+ return err;
+ }
+ }
+ }
+ return 0;
+}
+
+/**
+ * snd_hda_check_board_config - compare the current codec with the config table
+ * @codec: the HDA codec
+ * @num_configs: number of config enums
+ * @models: array of model name strings
+ * @tbl: configuration table, terminated by null entries
+ *
+ * Compares the modelname or PCI subsystem id of the current codec with the
+ * given configuration table. If a matching entry is found, returns its
+ * config value (supposed to be 0 or positive).
+ *
+ * If no entries are matching, the function returns a negative value.
+ */
+int snd_hda_check_board_config(struct hda_codec *codec,
+ int num_configs, const char **models,
+ const struct snd_pci_quirk *tbl)
+{
+ if (codec->bus->modelname && models) {
+ int i;
+ for (i = 0; i < num_configs; i++) {
+ if (models[i] &&
+ !strcmp(codec->bus->modelname, models[i])) {
+ snd_printd(KERN_INFO "hda_codec: model '%s' is "
+ "selected\n", models[i]);
+ return i;
+ }
+ }
+ }
+
+ if (!codec->bus->pci || !tbl)
+ return -1;
+
+ tbl = snd_pci_quirk_lookup(codec->bus->pci, tbl);
+ if (!tbl)
+ return -1;
+ if (tbl->value >= 0 && tbl->value < num_configs) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ char tmp[10];
+ const char *model = NULL;
+ if (models)
+ model = models[tbl->value];
+ if (!model) {
+ sprintf(tmp, "#%d", tbl->value);
+ model = tmp;
+ }
+ snd_printdd(KERN_INFO "hda_codec: model '%s' is selected "
+ "for config %x:%x (%s)\n",
+ model, tbl->subvendor, tbl->subdevice,
+ (tbl->name ? tbl->name : "Unknown device"));
+#endif
+ return tbl->value;
+ }
+ return -1;
+}
+
+/**
+ * snd_hda_add_new_ctls - create controls from the array
+ * @codec: the HDA codec
+ * @knew: the array of struct snd_kcontrol_new
+ *
+ * This helper function creates and add new controls in the given array.
+ * The array must be terminated with an empty entry as terminator.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
+{
+ int err;
+
+ for (; knew->name; knew++) {
+ struct snd_kcontrol *kctl;
+ kctl = snd_ctl_new1(knew, codec);
+ if (!kctl)
+ return -ENOMEM;
+ err = snd_ctl_add(codec->bus->card, kctl);
+ if (err < 0) {
+ if (!codec->addr)
+ return err;
+ kctl = snd_ctl_new1(knew, codec);
+ if (!kctl)
+ return -ENOMEM;
+ kctl->id.device = codec->addr;
+ err = snd_ctl_add(codec->bus->card, kctl);
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+ unsigned int power_state);
+
+static void hda_power_work(struct work_struct *work)
+{
+ struct hda_codec *codec =
+ container_of(work, struct hda_codec, power_work.work);
+
+ if (!codec->power_on || codec->power_count) {
+ codec->power_transition = 0;
+ return;
+ }
+
+ hda_call_codec_suspend(codec);
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+}
+
+static void hda_keep_power_on(struct hda_codec *codec)
+{
+ codec->power_count++;
+ codec->power_on = 1;
+}
+
+void snd_hda_power_up(struct hda_codec *codec)
+{
+ codec->power_count++;
+ if (codec->power_on || codec->power_transition)
+ return;
+
+ codec->power_on = 1;
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+ hda_call_codec_resume(codec);
+ cancel_delayed_work(&codec->power_work);
+ codec->power_transition = 0;
+}
+
+void snd_hda_power_down(struct hda_codec *codec)
+{
+ --codec->power_count;
+ if (!codec->power_on || codec->power_count || codec->power_transition)
+ return;
+ if (power_save) {
+ codec->power_transition = 1; /* avoid reentrance */
+ schedule_delayed_work(&codec->power_work,
+ msecs_to_jiffies(power_save * 1000));
+ }
+}
+
+int snd_hda_check_amp_list_power(struct hda_codec *codec,
+ struct hda_loopback_check *check,
+ hda_nid_t nid)
+{
+ struct hda_amp_list *p;
+ int ch, v;
+
+ if (!check->amplist)
+ return 0;
+ for (p = check->amplist; p->nid; p++) {
+ if (p->nid == nid)
+ break;
+ }
+ if (!p->nid)
+ return 0; /* nothing changed */
+
+ for (p = check->amplist; p->nid; p++) {
+ for (ch = 0; ch < 2; ch++) {
+ v = snd_hda_codec_amp_read(codec, p->nid, ch, p->dir,
+ p->idx);
+ if (!(v & HDA_AMP_MUTE) && v > 0) {
+ if (!check->power_on) {
+ check->power_on = 1;
+ snd_hda_power_up(codec);
+ }
+ return 1;
+ }
+ }
+ }
+ if (check->power_on) {
+ check->power_on = 0;
+ snd_hda_power_down(codec);
+ }
+ return 0;
+}
+#endif
+
+/*
+ * Channel mode helper
+ */
+int snd_hda_ch_mode_info(struct hda_codec *codec,
+ struct snd_ctl_elem_info *uinfo,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = num_chmodes;
+ if (uinfo->value.enumerated.item >= num_chmodes)
+ uinfo->value.enumerated.item = num_chmodes - 1;
+ sprintf(uinfo->value.enumerated.name, "%dch",
+ chmode[uinfo->value.enumerated.item].channels);
+ return 0;
+}
+
+int snd_hda_ch_mode_get(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes,
+ int max_channels)
+{
+ int i;
+
+ for (i = 0; i < num_chmodes; i++) {
+ if (max_channels == chmode[i].channels) {
+ ucontrol->value.enumerated.item[0] = i;
+ break;
+ }
+ }
+ return 0;
+}
+
+int snd_hda_ch_mode_put(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes,
+ int *max_channelsp)
+{
+ unsigned int mode;
+
+ mode = ucontrol->value.enumerated.item[0];
+ if (mode >= num_chmodes)
+ return -EINVAL;
+ if (*max_channelsp == chmode[mode].channels)
+ return 0;
+ /* change the current channel setting */
+ *max_channelsp = chmode[mode].channels;
+ if (chmode[mode].sequence)
+ snd_hda_sequence_write_cache(codec, chmode[mode].sequence);
+ return 1;
+}
+
+/*
+ * input MUX helper
+ */
+int snd_hda_input_mux_info(const struct hda_input_mux *imux,
+ struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int index;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = imux->num_items;
+ if (!imux->num_items)
+ return 0;
+ index = uinfo->value.enumerated.item;
+ if (index >= imux->num_items)
+ index = imux->num_items - 1;
+ strcpy(uinfo->value.enumerated.name, imux->items[index].label);
+ return 0;
+}
+
+int snd_hda_input_mux_put(struct hda_codec *codec,
+ const struct hda_input_mux *imux,
+ struct snd_ctl_elem_value *ucontrol,
+ hda_nid_t nid,
+ unsigned int *cur_val)
+{
+ unsigned int idx;
+
+ if (!imux->num_items)
+ return 0;
+ idx = ucontrol->value.enumerated.item[0];
+ if (idx >= imux->num_items)
+ idx = imux->num_items - 1;
+ if (*cur_val == idx)
+ return 0;
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
+ imux->items[idx].index);
+ *cur_val = idx;
+ return 1;
+}
+
+
+/*
+ * Multi-channel / digital-out PCM helper functions
+ */
+
+/* setup SPDIF output stream */
+static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int stream_tag, unsigned int format)
+{
+ /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
+ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+ set_dig_out_convert(codec, nid,
+ codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff,
+ -1);
+ snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+ if (codec->slave_dig_outs) {
+ hda_nid_t *d;
+ for (d = codec->slave_dig_outs; *d; d++)
+ snd_hda_codec_setup_stream(codec, *d, stream_tag, 0,
+ format);
+ }
+ /* turn on again (if needed) */
+ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+ set_dig_out_convert(codec, nid,
+ codec->spdif_ctls & 0xff, -1);
+}
+
+static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
+{
+ snd_hda_codec_cleanup_stream(codec, nid);
+ if (codec->slave_dig_outs) {
+ hda_nid_t *d;
+ for (d = codec->slave_dig_outs; *d; d++)
+ snd_hda_codec_cleanup_stream(codec, *d);
+ }
+}
+
+/*
+ * open the digital out in the exclusive mode
+ */
+int snd_hda_multi_out_dig_open(struct hda_codec *codec,
+ struct hda_multi_out *mout)
+{
+ mutex_lock(&codec->spdif_mutex);
+ if (mout->dig_out_used == HDA_DIG_ANALOG_DUP)
+ /* already opened as analog dup; reset it once */
+ cleanup_dig_out_stream(codec, mout->dig_out_nid);
+ mout->dig_out_used = HDA_DIG_EXCLUSIVE;
+ mutex_unlock(&codec->spdif_mutex);
+ return 0;
+}
+
+int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
+ struct hda_multi_out *mout,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ mutex_lock(&codec->spdif_mutex);
+ setup_dig_out_stream(codec, mout->dig_out_nid, stream_tag, format);
+ mutex_unlock(&codec->spdif_mutex);
+ return 0;
+}
+
+/*
+ * release the digital out
+ */
+int snd_hda_multi_out_dig_close(struct hda_codec *codec,
+ struct hda_multi_out *mout)
+{
+ mutex_lock(&codec->spdif_mutex);
+ mout->dig_out_used = 0;
+ mutex_unlock(&codec->spdif_mutex);
+ return 0;
+}
+
+/*
+ * set up more restrictions for analog out
+ */
+int snd_hda_multi_out_analog_open(struct hda_codec *codec,
+ struct hda_multi_out *mout,
+ struct snd_pcm_substream *substream,
+ struct hda_pcm_stream *hinfo)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ runtime->hw.channels_max = mout->max_channels;
+ if (mout->dig_out_nid) {
+ if (!mout->analog_rates) {
+ mout->analog_rates = hinfo->rates;
+ mout->analog_formats = hinfo->formats;
+ mout->analog_maxbps = hinfo->maxbps;
+ } else {
+ runtime->hw.rates = mout->analog_rates;
+ runtime->hw.formats = mout->analog_formats;
+ hinfo->maxbps = mout->analog_maxbps;
+ }
+ if (!mout->spdif_rates) {
+ snd_hda_query_supported_pcm(codec, mout->dig_out_nid,
+ &mout->spdif_rates,
+ &mout->spdif_formats,
+ &mout->spdif_maxbps);
+ }
+ mutex_lock(&codec->spdif_mutex);
+ if (mout->share_spdif) {
+ runtime->hw.rates &= mout->spdif_rates;
+ runtime->hw.formats &= mout->spdif_formats;
+ if (mout->spdif_maxbps < hinfo->maxbps)
+ hinfo->maxbps = mout->spdif_maxbps;
+ }
+ mutex_unlock(&codec->spdif_mutex);
+ }
+ return snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 2);
+}
+
+/*
+ * set up the i/o for analog out
+ * when the digital out is available, copy the front out to digital out, too.
+ */
+int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
+ struct hda_multi_out *mout,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ hda_nid_t *nids = mout->dac_nids;
+ int chs = substream->runtime->channels;
+ int i;
+
+ mutex_lock(&codec->spdif_mutex);
+ if (mout->dig_out_nid && mout->share_spdif &&
+ mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
+ if (chs == 2 &&
+ snd_hda_is_supported_format(codec, mout->dig_out_nid,
+ format) &&
+ !(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
+ mout->dig_out_used = HDA_DIG_ANALOG_DUP;
+ setup_dig_out_stream(codec, mout->dig_out_nid,
+ stream_tag, format);
+ } else {
+ mout->dig_out_used = 0;
+ cleanup_dig_out_stream(codec, mout->dig_out_nid);
+ }
+ }
+ mutex_unlock(&codec->spdif_mutex);
+
+ /* front */
+ snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag,
+ 0, format);
+ if (!mout->no_share_stream &&
+ mout->hp_nid && mout->hp_nid != nids[HDA_FRONT])
+ /* headphone out will just decode front left/right (stereo) */
+ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag,
+ 0, format);
+ /* extra outputs copied from front */
+ for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
+ if (!mout->no_share_stream && mout->extra_out_nid[i])
+ snd_hda_codec_setup_stream(codec,
+ mout->extra_out_nid[i],
+ stream_tag, 0, format);
+
+ /* surrounds */
+ for (i = 1; i < mout->num_dacs; i++) {
+ if (chs >= (i + 1) * 2) /* independent out */
+ snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
+ i * 2, format);
+ else if (!mout->no_share_stream) /* copy front */
+ snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
+ 0, format);
+ }
+ return 0;
+}
+
+/*
+ * clean up the setting for analog out
+ */
+int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
+ struct hda_multi_out *mout)
+{
+ hda_nid_t *nids = mout->dac_nids;
+ int i;
+
+ for (i = 0; i < mout->num_dacs; i++)
+ snd_hda_codec_cleanup_stream(codec, nids[i]);
+ if (mout->hp_nid)
+ snd_hda_codec_cleanup_stream(codec, mout->hp_nid);
+ for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
+ if (mout->extra_out_nid[i])
+ snd_hda_codec_cleanup_stream(codec,
+ mout->extra_out_nid[i]);
+ mutex_lock(&codec->spdif_mutex);
+ if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) {
+ cleanup_dig_out_stream(codec, mout->dig_out_nid);
+ mout->dig_out_used = 0;
+ }
+ mutex_unlock(&codec->spdif_mutex);
+ return 0;
+}
+
+/*
+ * Helper for automatic pin configuration
+ */
+
+static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)
+{
+ for (; *list; list++)
+ if (*list == nid)
+ return 1;
+ return 0;
+}
+
+
+/*
+ * Sort an associated group of pins according to their sequence numbers.
+ */
+static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences,
+ int num_pins)
+{
+ int i, j;
+ short seq;
+ hda_nid_t nid;
+
+ for (i = 0; i < num_pins; i++) {
+ for (j = i + 1; j < num_pins; j++) {
+ if (sequences[i] > sequences[j]) {
+ seq = sequences[i];
+ sequences[i] = sequences[j];
+ sequences[j] = seq;
+ nid = pins[i];
+ pins[i] = pins[j];
+ pins[j] = nid;
+ }
+ }
+ }
+}
+
+
+/*
+ * Parse all pin widgets and store the useful pin nids to cfg
+ *
+ * The number of line-outs or any primary output is stored in line_outs,
+ * and the corresponding output pins are assigned to line_out_pins[],
+ * in the order of front, rear, CLFE, side, ...
+ *
+ * If more extra outputs (speaker and headphone) are found, the pins are
+ * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack
+ * is detected, one of speaker of HP pins is assigned as the primary
+ * output, i.e. to line_out_pins[0]. So, line_outs is always positive
+ * if any analog output exists.
+ *
+ * The analog input pins are assigned to input_pins array.
+ * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
+ * respectively.
+ */
+int snd_hda_parse_pin_def_config(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ hda_nid_t *ignore_nids)
+{
+ hda_nid_t nid, end_nid;
+ short seq, assoc_line_out, assoc_speaker;
+ short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
+ short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
+ short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
+
+ memset(cfg, 0, sizeof(*cfg));
+
+ memset(sequences_line_out, 0, sizeof(sequences_line_out));
+ memset(sequences_speaker, 0, sizeof(sequences_speaker));
+ memset(sequences_hp, 0, sizeof(sequences_hp));
+ assoc_line_out = assoc_speaker = 0;
+
+ end_nid = codec->start_nid + codec->num_nodes;
+ for (nid = codec->start_nid; nid < end_nid; nid++) {
+ unsigned int wid_caps = get_wcaps(codec, nid);
+ unsigned int wid_type =
+ (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ unsigned int def_conf;
+ short assoc, loc;
+
+ /* read all default configuration for pin complex */
+ if (wid_type != AC_WID_PIN)
+ continue;
+ /* ignore the given nids (e.g. pc-beep returns error) */
+ if (ignore_nids && is_in_nid_list(nid, ignore_nids))
+ continue;
+
+ def_conf = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+ continue;
+ loc = get_defcfg_location(def_conf);
+ switch (get_defcfg_device(def_conf)) {
+ case AC_JACK_LINE_OUT:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+
+ if (!(wid_caps & AC_WCAP_STEREO))
+ if (!cfg->mono_out_pin)
+ cfg->mono_out_pin = nid;
+ if (!assoc)
+ continue;
+ if (!assoc_line_out)
+ assoc_line_out = assoc;
+ else if (assoc_line_out != assoc)
+ continue;
+ if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
+ continue;
+ cfg->line_out_pins[cfg->line_outs] = nid;
+ sequences_line_out[cfg->line_outs] = seq;
+ cfg->line_outs++;
+ break;
+ case AC_JACK_SPEAKER:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+ if (! assoc)
+ continue;
+ if (! assoc_speaker)
+ assoc_speaker = assoc;
+ else if (assoc_speaker != assoc)
+ continue;
+ if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
+ continue;
+ cfg->speaker_pins[cfg->speaker_outs] = nid;
+ sequences_speaker[cfg->speaker_outs] = seq;
+ cfg->speaker_outs++;
+ break;
+ case AC_JACK_HP_OUT:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+ if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
+ continue;
+ cfg->hp_pins[cfg->hp_outs] = nid;
+ sequences_hp[cfg->hp_outs] = (assoc << 4) | seq;
+ cfg->hp_outs++;
+ break;
+ case AC_JACK_MIC_IN: {
+ int preferred, alt;
+ if (loc == AC_JACK_LOC_FRONT) {
+ preferred = AUTO_PIN_FRONT_MIC;
+ alt = AUTO_PIN_MIC;
+ } else {
+ preferred = AUTO_PIN_MIC;
+ alt = AUTO_PIN_FRONT_MIC;
+ }
+ if (!cfg->input_pins[preferred])
+ cfg->input_pins[preferred] = nid;
+ else if (!cfg->input_pins[alt])
+ cfg->input_pins[alt] = nid;
+ break;
+ }
+ case AC_JACK_LINE_IN:
+ if (loc == AC_JACK_LOC_FRONT)
+ cfg->input_pins[AUTO_PIN_FRONT_LINE] = nid;
+ else
+ cfg->input_pins[AUTO_PIN_LINE] = nid;
+ break;
+ case AC_JACK_CD:
+ cfg->input_pins[AUTO_PIN_CD] = nid;
+ break;
+ case AC_JACK_AUX:
+ cfg->input_pins[AUTO_PIN_AUX] = nid;
+ break;
+ case AC_JACK_SPDIF_OUT:
+ cfg->dig_out_pin = nid;
+ break;
+ case AC_JACK_SPDIF_IN:
+ cfg->dig_in_pin = nid;
+ break;
+ }
+ }
+
+ /* FIX-UP:
+ * If no line-out is defined but multiple HPs are found,
+ * some of them might be the real line-outs.
+ */
+ if (!cfg->line_outs && cfg->hp_outs > 1) {
+ int i = 0;
+ while (i < cfg->hp_outs) {
+ /* The real HPs should have the sequence 0x0f */
+ if ((sequences_hp[i] & 0x0f) == 0x0f) {
+ i++;
+ continue;
+ }
+ /* Move it to the line-out table */
+ cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
+ sequences_line_out[cfg->line_outs] = sequences_hp[i];
+ cfg->line_outs++;
+ cfg->hp_outs--;
+ memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
+ sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
+ memmove(sequences_hp + i - 1, sequences_hp + i,
+ sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
+ }
+ }
+
+ /* sort by sequence */
+ sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
+ cfg->line_outs);
+ sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker,
+ cfg->speaker_outs);
+ sort_pins_by_sequence(cfg->hp_pins, sequences_hp,
+ cfg->hp_outs);
+
+ /* if we have only one mic, make it AUTO_PIN_MIC */
+ if (!cfg->input_pins[AUTO_PIN_MIC] &&
+ cfg->input_pins[AUTO_PIN_FRONT_MIC]) {
+ cfg->input_pins[AUTO_PIN_MIC] =
+ cfg->input_pins[AUTO_PIN_FRONT_MIC];
+ cfg->input_pins[AUTO_PIN_FRONT_MIC] = 0;
+ }
+ /* ditto for line-in */
+ if (!cfg->input_pins[AUTO_PIN_LINE] &&
+ cfg->input_pins[AUTO_PIN_FRONT_LINE]) {
+ cfg->input_pins[AUTO_PIN_LINE] =
+ cfg->input_pins[AUTO_PIN_FRONT_LINE];
+ cfg->input_pins[AUTO_PIN_FRONT_LINE] = 0;
+ }
+
+ /*
+ * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
+ * as a primary output
+ */
+ if (!cfg->line_outs) {
+ if (cfg->speaker_outs) {
+ cfg->line_outs = cfg->speaker_outs;
+ memcpy(cfg->line_out_pins, cfg->speaker_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->speaker_outs = 0;
+ memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
+ cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
+ } else if (cfg->hp_outs) {
+ cfg->line_outs = cfg->hp_outs;
+ memcpy(cfg->line_out_pins, cfg->hp_pins,
+ sizeof(cfg->hp_pins));
+ cfg->hp_outs = 0;
+ memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+ }
+ }
+
+ /* Reorder the surround channels
+ * ALSA sequence is front/surr/clfe/side
+ * HDA sequence is:
+ * 4-ch: front/surr => OK as it is
+ * 6-ch: front/clfe/surr
+ * 8-ch: front/clfe/rear/side|fc
+ */
+ switch (cfg->line_outs) {
+ case 3:
+ case 4:
+ nid = cfg->line_out_pins[1];
+ cfg->line_out_pins[1] = cfg->line_out_pins[2];
+ cfg->line_out_pins[2] = nid;
+ break;
+ }
+
+ /*
+ * debug prints of the parsed results
+ */
+ snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
+ cfg->line_out_pins[2], cfg->line_out_pins[3],
+ cfg->line_out_pins[4]);
+ snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->speaker_outs, cfg->speaker_pins[0],
+ cfg->speaker_pins[1], cfg->speaker_pins[2],
+ cfg->speaker_pins[3], cfg->speaker_pins[4]);
+ snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->hp_outs, cfg->hp_pins[0],
+ cfg->hp_pins[1], cfg->hp_pins[2],
+ cfg->hp_pins[3], cfg->hp_pins[4]);
+ snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin);
+ snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x,"
+ " cd=0x%x, aux=0x%x\n",
+ cfg->input_pins[AUTO_PIN_MIC],
+ cfg->input_pins[AUTO_PIN_FRONT_MIC],
+ cfg->input_pins[AUTO_PIN_LINE],
+ cfg->input_pins[AUTO_PIN_FRONT_LINE],
+ cfg->input_pins[AUTO_PIN_CD],
+ cfg->input_pins[AUTO_PIN_AUX]);
+
+ return 0;
+}
+
+/* labels for input pins */
+const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux"
+};
+
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+
+/**
+ * snd_hda_suspend - suspend the codecs
+ * @bus: the HDA bus
+ * @state: suspsend state
+ *
+ * Returns 0 if successful.
+ */
+int snd_hda_suspend(struct hda_bus *bus, pm_message_t state)
+{
+ struct hda_codec *codec;
+
+ list_for_each_entry(codec, &bus->codec_list, list) {
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!codec->power_on)
+ continue;
+#endif
+ hda_call_codec_suspend(codec);
+ }
+ return 0;
+}
+
+/**
+ * snd_hda_resume - resume the codecs
+ * @bus: the HDA bus
+ * @state: resume state
+ *
+ * Returns 0 if successful.
+ *
+ * This fucntion is defined only when POWER_SAVE isn't set.
+ * In the power-save mode, the codec is resumed dynamically.
+ */
+int snd_hda_resume(struct hda_bus *bus)
+{
+ struct hda_codec *codec;
+
+ list_for_each_entry(codec, &bus->codec_list, list) {
+ if (snd_hda_codec_needs_resume(codec))
+ hda_call_codec_resume(codec);
+ }
+ return 0;
+}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+int snd_hda_codecs_inuse(struct hda_bus *bus)
+{
+ struct hda_codec *codec;
+
+ list_for_each_entry(codec, &bus->codec_list, list) {
+ if (snd_hda_codec_needs_resume(codec))
+ return 1;
+ }
+ return 0;
+}
+#endif
+#endif
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
new file mode 100644
index 0000000..60468f5
--- /dev/null
+++ b/sound/pci/hda/hda_codec.h
@@ -0,0 +1,848 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the Free
+ * Software Foundation; either version 2 of the License, or (at your option)
+ * any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
+ * more details.
+ *
+ * You should have received a copy of the GNU General Public License along with
+ * this program; if not, write to the Free Software Foundation, Inc., 59
+ * Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __SOUND_HDA_CODEC_H
+#define __SOUND_HDA_CODEC_H
+
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/hwdep.h>
+
+#if defined(CONFIG_PM) || defined(CONFIG_SND_HDA_POWER_SAVE)
+#define SND_HDA_NEEDS_RESUME /* resume control code is required */
+#endif
+
+/*
+ * nodes
+ */
+#define AC_NODE_ROOT 0x00
+
+/*
+ * function group types
+ */
+enum {
+ AC_GRP_AUDIO_FUNCTION = 0x01,
+ AC_GRP_MODEM_FUNCTION = 0x02,
+};
+
+/*
+ * widget types
+ */
+enum {
+ AC_WID_AUD_OUT, /* Audio Out */
+ AC_WID_AUD_IN, /* Audio In */
+ AC_WID_AUD_MIX, /* Audio Mixer */
+ AC_WID_AUD_SEL, /* Audio Selector */
+ AC_WID_PIN, /* Pin Complex */
+ AC_WID_POWER, /* Power */
+ AC_WID_VOL_KNB, /* Volume Knob */
+ AC_WID_BEEP, /* Beep Generator */
+ AC_WID_VENDOR = 0x0f /* Vendor specific */
+};
+
+/*
+ * GET verbs
+ */
+#define AC_VERB_GET_STREAM_FORMAT 0x0a00
+#define AC_VERB_GET_AMP_GAIN_MUTE 0x0b00
+#define AC_VERB_GET_PROC_COEF 0x0c00
+#define AC_VERB_GET_COEF_INDEX 0x0d00
+#define AC_VERB_PARAMETERS 0x0f00
+#define AC_VERB_GET_CONNECT_SEL 0x0f01
+#define AC_VERB_GET_CONNECT_LIST 0x0f02
+#define AC_VERB_GET_PROC_STATE 0x0f03
+#define AC_VERB_GET_SDI_SELECT 0x0f04
+#define AC_VERB_GET_POWER_STATE 0x0f05
+#define AC_VERB_GET_CONV 0x0f06
+#define AC_VERB_GET_PIN_WIDGET_CONTROL 0x0f07
+#define AC_VERB_GET_UNSOLICITED_RESPONSE 0x0f08
+#define AC_VERB_GET_PIN_SENSE 0x0f09
+#define AC_VERB_GET_BEEP_CONTROL 0x0f0a
+#define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c
+#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d
+#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */
+#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f
+/* f10-f1a: GPIO */
+#define AC_VERB_GET_GPIO_DATA 0x0f15
+#define AC_VERB_GET_GPIO_MASK 0x0f16
+#define AC_VERB_GET_GPIO_DIRECTION 0x0f17
+#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18
+#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19
+#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a
+#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c
+/* f20: AFG/MFG */
+#define AC_VERB_GET_SUBSYSTEM_ID 0x0f20
+#define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d
+#define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e
+#define AC_VERB_GET_HDMI_ELDD 0x0f2f
+#define AC_VERB_GET_HDMI_DIP_INDEX 0x0f30
+#define AC_VERB_GET_HDMI_DIP_DATA 0x0f31
+#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32
+#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33
+#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34
+
+/*
+ * SET verbs
+ */
+#define AC_VERB_SET_STREAM_FORMAT 0x200
+#define AC_VERB_SET_AMP_GAIN_MUTE 0x300
+#define AC_VERB_SET_PROC_COEF 0x400
+#define AC_VERB_SET_COEF_INDEX 0x500
+#define AC_VERB_SET_CONNECT_SEL 0x701
+#define AC_VERB_SET_PROC_STATE 0x703
+#define AC_VERB_SET_SDI_SELECT 0x704
+#define AC_VERB_SET_POWER_STATE 0x705
+#define AC_VERB_SET_CHANNEL_STREAMID 0x706
+#define AC_VERB_SET_PIN_WIDGET_CONTROL 0x707
+#define AC_VERB_SET_UNSOLICITED_ENABLE 0x708
+#define AC_VERB_SET_PIN_SENSE 0x709
+#define AC_VERB_SET_BEEP_CONTROL 0x70a
+#define AC_VERB_SET_EAPD_BTLENABLE 0x70c
+#define AC_VERB_SET_DIGI_CONVERT_1 0x70d
+#define AC_VERB_SET_DIGI_CONVERT_2 0x70e
+#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f
+#define AC_VERB_SET_GPIO_DATA 0x715
+#define AC_VERB_SET_GPIO_MASK 0x716
+#define AC_VERB_SET_GPIO_DIRECTION 0x717
+#define AC_VERB_SET_GPIO_WAKE_MASK 0x718
+#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719
+#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f
+#define AC_VERB_SET_EAPD 0x788
+#define AC_VERB_SET_CODEC_RESET 0x7ff
+#define AC_VERB_SET_CVT_CHAN_COUNT 0x72d
+#define AC_VERB_SET_HDMI_DIP_INDEX 0x730
+#define AC_VERB_SET_HDMI_DIP_DATA 0x731
+#define AC_VERB_SET_HDMI_DIP_XMIT 0x732
+#define AC_VERB_SET_HDMI_CP_CTRL 0x733
+#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734
+
+/*
+ * Parameter IDs
+ */
+#define AC_PAR_VENDOR_ID 0x00
+#define AC_PAR_SUBSYSTEM_ID 0x01
+#define AC_PAR_REV_ID 0x02
+#define AC_PAR_NODE_COUNT 0x04
+#define AC_PAR_FUNCTION_TYPE 0x05
+#define AC_PAR_AUDIO_FG_CAP 0x08
+#define AC_PAR_AUDIO_WIDGET_CAP 0x09
+#define AC_PAR_PCM 0x0a
+#define AC_PAR_STREAM 0x0b
+#define AC_PAR_PIN_CAP 0x0c
+#define AC_PAR_AMP_IN_CAP 0x0d
+#define AC_PAR_CONNLIST_LEN 0x0e
+#define AC_PAR_POWER_STATE 0x0f
+#define AC_PAR_PROC_CAP 0x10
+#define AC_PAR_GPIO_CAP 0x11
+#define AC_PAR_AMP_OUT_CAP 0x12
+#define AC_PAR_VOL_KNB_CAP 0x13
+#define AC_PAR_HDMI_LPCM_CAP 0x20
+
+/*
+ * AC_VERB_PARAMETERS results (32bit)
+ */
+
+/* Function Group Type */
+#define AC_FGT_TYPE (0xff<<0)
+#define AC_FGT_TYPE_SHIFT 0
+#define AC_FGT_UNSOL_CAP (1<<8)
+
+/* Audio Function Group Capabilities */
+#define AC_AFG_OUT_DELAY (0xf<<0)
+#define AC_AFG_IN_DELAY (0xf<<8)
+#define AC_AFG_BEEP_GEN (1<<16)
+
+/* Audio Widget Capabilities */
+#define AC_WCAP_STEREO (1<<0) /* stereo I/O */
+#define AC_WCAP_IN_AMP (1<<1) /* AMP-in present */
+#define AC_WCAP_OUT_AMP (1<<2) /* AMP-out present */
+#define AC_WCAP_AMP_OVRD (1<<3) /* AMP-parameter override */
+#define AC_WCAP_FORMAT_OVRD (1<<4) /* format override */
+#define AC_WCAP_STRIPE (1<<5) /* stripe */
+#define AC_WCAP_PROC_WID (1<<6) /* Proc Widget */
+#define AC_WCAP_UNSOL_CAP (1<<7) /* Unsol capable */
+#define AC_WCAP_CONN_LIST (1<<8) /* connection list */
+#define AC_WCAP_DIGITAL (1<<9) /* digital I/O */
+#define AC_WCAP_POWER (1<<10) /* power control */
+#define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */
+#define AC_WCAP_CP_CAPS (1<<12) /* content protection */
+#define AC_WCAP_CHAN_CNT_EXT (7<<13) /* channel count ext */
+#define AC_WCAP_DELAY (0xf<<16)
+#define AC_WCAP_DELAY_SHIFT 16
+#define AC_WCAP_TYPE (0xf<<20)
+#define AC_WCAP_TYPE_SHIFT 20
+
+/* supported PCM rates and bits */
+#define AC_SUPPCM_RATES (0xfff << 0)
+#define AC_SUPPCM_BITS_8 (1<<16)
+#define AC_SUPPCM_BITS_16 (1<<17)
+#define AC_SUPPCM_BITS_20 (1<<18)
+#define AC_SUPPCM_BITS_24 (1<<19)
+#define AC_SUPPCM_BITS_32 (1<<20)
+
+/* supported PCM stream format */
+#define AC_SUPFMT_PCM (1<<0)
+#define AC_SUPFMT_FLOAT32 (1<<1)
+#define AC_SUPFMT_AC3 (1<<2)
+
+/* GP I/O count */
+#define AC_GPIO_IO_COUNT (0xff<<0)
+#define AC_GPIO_O_COUNT (0xff<<8)
+#define AC_GPIO_O_COUNT_SHIFT 8
+#define AC_GPIO_I_COUNT (0xff<<16)
+#define AC_GPIO_I_COUNT_SHIFT 16
+#define AC_GPIO_UNSOLICITED (1<<30)
+#define AC_GPIO_WAKE (1<<31)
+
+/* Converter stream, channel */
+#define AC_CONV_CHANNEL (0xf<<0)
+#define AC_CONV_STREAM (0xf<<4)
+#define AC_CONV_STREAM_SHIFT 4
+
+/* Input converter SDI select */
+#define AC_SDI_SELECT (0xf<<0)
+
+/* Unsolicited response control */
+#define AC_UNSOL_TAG (0x3f<<0)
+#define AC_UNSOL_ENABLED (1<<7)
+#define AC_USRSP_EN AC_UNSOL_ENABLED
+
+/* Unsolicited responses */
+#define AC_UNSOL_RES_TAG (0x3f<<26)
+#define AC_UNSOL_RES_TAG_SHIFT 26
+#define AC_UNSOL_RES_SUBTAG (0x1f<<21)
+#define AC_UNSOL_RES_SUBTAG_SHIFT 21
+#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */
+#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */
+#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */
+#define AC_UNSOL_RES_CP_READY (1<<0) /* content protection */
+
+/* Pin widget capabilies */
+#define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */
+#define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */
+#define AC_PINCAP_PRES_DETECT (1<<2) /* presence detect capable */
+#define AC_PINCAP_HP_DRV (1<<3) /* headphone drive capable */
+#define AC_PINCAP_OUT (1<<4) /* output capable */
+#define AC_PINCAP_IN (1<<5) /* input capable */
+#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */
+/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification,
+ * but is marked reserved in the Intel HDA specification.
+ */
+#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */
+/* Note: The same bit as LR_SWAP is newly defined as HDMI capability
+ * in HD-audio specification
+ */
+#define AC_PINCAP_HDMI (1<<7) /* HDMI pin */
+#define AC_PINCAP_VREF (0x37<<8)
+#define AC_PINCAP_VREF_SHIFT 8
+#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */
+/* Vref status (used in pin cap) */
+#define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */
+#define AC_PINCAP_VREF_50 (1<<1) /* 50% */
+#define AC_PINCAP_VREF_GRD (1<<2) /* ground */
+#define AC_PINCAP_VREF_80 (1<<4) /* 80% */
+#define AC_PINCAP_VREF_100 (1<<5) /* 100% */
+
+/* Amplifier capabilities */
+#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */
+#define AC_AMPCAP_OFFSET_SHIFT 0
+#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */
+#define AC_AMPCAP_NUM_STEPS_SHIFT 8
+#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB
+ * in 0.25dB
+ */
+#define AC_AMPCAP_STEP_SIZE_SHIFT 16
+#define AC_AMPCAP_MUTE (1<<31) /* mute capable */
+#define AC_AMPCAP_MUTE_SHIFT 31
+
+/* Connection list */
+#define AC_CLIST_LENGTH (0x7f<<0)
+#define AC_CLIST_LONG (1<<7)
+
+/* Supported power status */
+#define AC_PWRST_D0SUP (1<<0)
+#define AC_PWRST_D1SUP (1<<1)
+#define AC_PWRST_D2SUP (1<<2)
+#define AC_PWRST_D3SUP (1<<3)
+
+/* Power state values */
+#define AC_PWRST_SETTING (0xf<<0)
+#define AC_PWRST_ACTUAL (0xf<<4)
+#define AC_PWRST_ACTUAL_SHIFT 4
+#define AC_PWRST_D0 0x00
+#define AC_PWRST_D1 0x01
+#define AC_PWRST_D2 0x02
+#define AC_PWRST_D3 0x03
+
+/* Processing capabilies */
+#define AC_PCAP_BENIGN (1<<0)
+#define AC_PCAP_NUM_COEF (0xff<<8)
+#define AC_PCAP_NUM_COEF_SHIFT 8
+
+/* Volume knobs capabilities */
+#define AC_KNBCAP_NUM_STEPS (0x7f<<0)
+#define AC_KNBCAP_DELTA (1<<7)
+
+/* HDMI LPCM capabilities */
+#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */
+#define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */
+#define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */
+#define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */
+#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */
+#define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */
+#define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */
+#define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */
+#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */
+#define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */
+#define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */
+#define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */
+#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */
+#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */
+
+/*
+ * Control Parameters
+ */
+
+/* Amp gain/mute */
+#define AC_AMP_MUTE (1<<7)
+#define AC_AMP_GAIN (0x7f)
+#define AC_AMP_GET_INDEX (0xf<<0)
+
+#define AC_AMP_GET_LEFT (1<<13)
+#define AC_AMP_GET_RIGHT (0<<13)
+#define AC_AMP_GET_OUTPUT (1<<15)
+#define AC_AMP_GET_INPUT (0<<15)
+
+#define AC_AMP_SET_INDEX (0xf<<8)
+#define AC_AMP_SET_INDEX_SHIFT 8
+#define AC_AMP_SET_RIGHT (1<<12)
+#define AC_AMP_SET_LEFT (1<<13)
+#define AC_AMP_SET_INPUT (1<<14)
+#define AC_AMP_SET_OUTPUT (1<<15)
+
+/* DIGITAL1 bits */
+#define AC_DIG1_ENABLE (1<<0)
+#define AC_DIG1_V (1<<1)
+#define AC_DIG1_VCFG (1<<2)
+#define AC_DIG1_EMPHASIS (1<<3)
+#define AC_DIG1_COPYRIGHT (1<<4)
+#define AC_DIG1_NONAUDIO (1<<5)
+#define AC_DIG1_PROFESSIONAL (1<<6)
+#define AC_DIG1_LEVEL (1<<7)
+
+/* DIGITAL2 bits */
+#define AC_DIG2_CC (0x7f<<0)
+
+/* Pin widget control - 8bit */
+#define AC_PINCTL_VREFEN (0x7<<0)
+#define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */
+#define AC_PINCTL_VREF_50 1 /* 50% */
+#define AC_PINCTL_VREF_GRD 2 /* ground */
+#define AC_PINCTL_VREF_80 4 /* 80% */
+#define AC_PINCTL_VREF_100 5 /* 100% */
+#define AC_PINCTL_IN_EN (1<<5)
+#define AC_PINCTL_OUT_EN (1<<6)
+#define AC_PINCTL_HP_EN (1<<7)
+
+/* Pin sense - 32bit */
+#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff)
+#define AC_PINSENSE_PRESENCE (1<<31)
+#define AC_PINSENSE_ELDV (1<<30) /* ELD valid (HDMI) */
+
+/* EAPD/BTL enable - 32bit */
+#define AC_EAPDBTL_BALANCED (1<<0)
+#define AC_EAPDBTL_EAPD (1<<1)
+#define AC_EAPDBTL_LR_SWAP (1<<2)
+
+/* HDMI ELD data */
+#define AC_ELDD_ELD_VALID (1<<31)
+#define AC_ELDD_ELD_DATA 0xff
+
+/* HDMI DIP size */
+#define AC_DIPSIZE_ELD_BUF (1<<3) /* ELD buf size of packet size */
+#define AC_DIPSIZE_PACK_IDX (0x07<<0) /* packet index */
+
+/* HDMI DIP index */
+#define AC_DIPIDX_PACK_IDX (0x07<<5) /* packet idnex */
+#define AC_DIPIDX_BYTE_IDX (0x1f<<0) /* byte index */
+
+/* HDMI DIP xmit (transmit) control */
+#define AC_DIPXMIT_MASK (0x3<<6)
+#define AC_DIPXMIT_DISABLE (0x0<<6) /* disable xmit */
+#define AC_DIPXMIT_ONCE (0x2<<6) /* xmit once then disable */
+#define AC_DIPXMIT_BEST (0x3<<6) /* best effort */
+
+/* HDMI content protection (CP) control */
+#define AC_CPCTRL_CES (1<<9) /* current encryption state */
+#define AC_CPCTRL_READY (1<<8) /* ready bit */
+#define AC_CPCTRL_SUBTAG (0x1f<<3) /* subtag for unsol-resp */
+#define AC_CPCTRL_STATE (3<<0) /* current CP request state */
+
+/* Converter channel <-> HDMI slot mapping */
+#define AC_CVTMAP_HDMI_SLOT (0xf<<0) /* HDMI slot number */
+#define AC_CVTMAP_CHAN (0xf<<4) /* converter channel number */
+
+/* configuration default - 32bit */
+#define AC_DEFCFG_SEQUENCE (0xf<<0)
+#define AC_DEFCFG_DEF_ASSOC (0xf<<4)
+#define AC_DEFCFG_ASSOC_SHIFT 4
+#define AC_DEFCFG_MISC (0xf<<8)
+#define AC_DEFCFG_MISC_SHIFT 8
+#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0)
+#define AC_DEFCFG_COLOR (0xf<<12)
+#define AC_DEFCFG_COLOR_SHIFT 12
+#define AC_DEFCFG_CONN_TYPE (0xf<<16)
+#define AC_DEFCFG_CONN_TYPE_SHIFT 16
+#define AC_DEFCFG_DEVICE (0xf<<20)
+#define AC_DEFCFG_DEVICE_SHIFT 20
+#define AC_DEFCFG_LOCATION (0x3f<<24)
+#define AC_DEFCFG_LOCATION_SHIFT 24
+#define AC_DEFCFG_PORT_CONN (0x3<<30)
+#define AC_DEFCFG_PORT_CONN_SHIFT 30
+
+/* device device types (0x0-0xf) */
+enum {
+ AC_JACK_LINE_OUT,
+ AC_JACK_SPEAKER,
+ AC_JACK_HP_OUT,
+ AC_JACK_CD,
+ AC_JACK_SPDIF_OUT,
+ AC_JACK_DIG_OTHER_OUT,
+ AC_JACK_MODEM_LINE_SIDE,
+ AC_JACK_MODEM_HAND_SIDE,
+ AC_JACK_LINE_IN,
+ AC_JACK_AUX,
+ AC_JACK_MIC_IN,
+ AC_JACK_TELEPHONY,
+ AC_JACK_SPDIF_IN,
+ AC_JACK_DIG_OTHER_IN,
+ AC_JACK_OTHER = 0xf,
+};
+
+/* jack connection types (0x0-0xf) */
+enum {
+ AC_JACK_CONN_UNKNOWN,
+ AC_JACK_CONN_1_8,
+ AC_JACK_CONN_1_4,
+ AC_JACK_CONN_ATAPI,
+ AC_JACK_CONN_RCA,
+ AC_JACK_CONN_OPTICAL,
+ AC_JACK_CONN_OTHER_DIGITAL,
+ AC_JACK_CONN_OTHER_ANALOG,
+ AC_JACK_CONN_DIN,
+ AC_JACK_CONN_XLR,
+ AC_JACK_CONN_RJ11,
+ AC_JACK_CONN_COMB,
+ AC_JACK_CONN_OTHER = 0xf,
+};
+
+/* jack colors (0x0-0xf) */
+enum {
+ AC_JACK_COLOR_UNKNOWN,
+ AC_JACK_COLOR_BLACK,
+ AC_JACK_COLOR_GREY,
+ AC_JACK_COLOR_BLUE,
+ AC_JACK_COLOR_GREEN,
+ AC_JACK_COLOR_RED,
+ AC_JACK_COLOR_ORANGE,
+ AC_JACK_COLOR_YELLOW,
+ AC_JACK_COLOR_PURPLE,
+ AC_JACK_COLOR_PINK,
+ AC_JACK_COLOR_WHITE = 0xe,
+ AC_JACK_COLOR_OTHER,
+};
+
+/* Jack location (0x0-0x3f) */
+/* common case */
+enum {
+ AC_JACK_LOC_NONE,
+ AC_JACK_LOC_REAR,
+ AC_JACK_LOC_FRONT,
+ AC_JACK_LOC_LEFT,
+ AC_JACK_LOC_RIGHT,
+ AC_JACK_LOC_TOP,
+ AC_JACK_LOC_BOTTOM,
+};
+/* bits 4-5 */
+enum {
+ AC_JACK_LOC_EXTERNAL = 0x00,
+ AC_JACK_LOC_INTERNAL = 0x10,
+ AC_JACK_LOC_SEPARATE = 0x20,
+ AC_JACK_LOC_OTHER = 0x30,
+};
+enum {
+ /* external on primary chasis */
+ AC_JACK_LOC_REAR_PANEL = 0x07,
+ AC_JACK_LOC_DRIVE_BAY,
+ /* internal */
+ AC_JACK_LOC_RISER = 0x17,
+ AC_JACK_LOC_HDMI,
+ AC_JACK_LOC_ATAPI,
+ /* others */
+ AC_JACK_LOC_MOBILE_IN = 0x37,
+ AC_JACK_LOC_MOBILE_OUT,
+};
+
+/* Port connectivity (0-3) */
+enum {
+ AC_JACK_PORT_COMPLEX,
+ AC_JACK_PORT_NONE,
+ AC_JACK_PORT_FIXED,
+ AC_JACK_PORT_BOTH,
+};
+
+/* max. connections to a widget */
+#define HDA_MAX_CONNECTIONS 32
+
+/* max. codec address */
+#define HDA_MAX_CODEC_ADDRESS 0x0f
+
+/*
+ * Structures
+ */
+
+struct hda_bus;
+struct hda_beep;
+struct hda_codec;
+struct hda_pcm;
+struct hda_pcm_stream;
+struct hda_bus_unsolicited;
+
+/* NID type */
+typedef u16 hda_nid_t;
+
+/* bus operators */
+struct hda_bus_ops {
+ /* send a single command */
+ int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
+ unsigned int verb, unsigned int parm);
+ /* get a response from the last command */
+ unsigned int (*get_response)(struct hda_codec *codec);
+ /* free the private data */
+ void (*private_free)(struct hda_bus *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ /* notify power-up/down from codec to controller */
+ void (*pm_notify)(struct hda_codec *codec);
+#endif
+};
+
+/* template to pass to the bus constructor */
+struct hda_bus_template {
+ void *private_data;
+ struct pci_dev *pci;
+ const char *modelname;
+ struct hda_bus_ops ops;
+};
+
+/*
+ * codec bus
+ *
+ * each controller needs to creata a hda_bus to assign the accessor.
+ * A hda_bus contains several codecs in the list codec_list.
+ */
+struct hda_bus {
+ struct snd_card *card;
+
+ /* copied from template */
+ void *private_data;
+ struct pci_dev *pci;
+ const char *modelname;
+ struct hda_bus_ops ops;
+
+ /* codec linked list */
+ struct list_head codec_list;
+ /* link caddr -> codec */
+ struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1];
+
+ struct mutex cmd_mutex;
+
+ /* unsolicited event queue */
+ struct hda_bus_unsolicited *unsol;
+
+ struct snd_info_entry *proc;
+
+ /* misc op flags */
+ unsigned int needs_damn_long_delay :1;
+};
+
+/*
+ * codec preset
+ *
+ * Known codecs have the patch to build and set up the controls/PCMs
+ * better than the generic parser.
+ */
+struct hda_codec_preset {
+ unsigned int id;
+ unsigned int mask;
+ unsigned int subs;
+ unsigned int subs_mask;
+ unsigned int rev;
+ hda_nid_t afg, mfg;
+ const char *name;
+ int (*patch)(struct hda_codec *codec);
+};
+
+/* ops set by the preset patch */
+struct hda_codec_ops {
+ int (*build_controls)(struct hda_codec *codec);
+ int (*build_pcms)(struct hda_codec *codec);
+ int (*init)(struct hda_codec *codec);
+ void (*free)(struct hda_codec *codec);
+ void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+#ifdef SND_HDA_NEEDS_RESUME
+ int (*suspend)(struct hda_codec *codec, pm_message_t state);
+ int (*resume)(struct hda_codec *codec);
+#endif
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
+#endif
+};
+
+/* record for amp information cache */
+struct hda_cache_head {
+ u32 key; /* hash key */
+ u16 val; /* assigned value */
+ u16 next; /* next link; -1 = terminal */
+};
+
+struct hda_amp_info {
+ struct hda_cache_head head;
+ u32 amp_caps; /* amp capabilities */
+ u16 vol[2]; /* current volume & mute */
+};
+
+struct hda_cache_rec {
+ u16 hash[64]; /* hash table for index */
+ unsigned int num_entries; /* number of assigned entries */
+ unsigned int size; /* allocated size */
+ unsigned int record_size; /* record size (including header) */
+ void *buffer; /* hash table entries */
+};
+
+/* PCM callbacks */
+struct hda_pcm_ops {
+ int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ struct snd_pcm_substream *substream);
+ int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ struct snd_pcm_substream *substream);
+ int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ unsigned int stream_tag, unsigned int format,
+ struct snd_pcm_substream *substream);
+ int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ struct snd_pcm_substream *substream);
+};
+
+/* PCM information for each substream */
+struct hda_pcm_stream {
+ unsigned int substreams; /* number of substreams, 0 = not exist*/
+ unsigned int channels_min; /* min. number of channels */
+ unsigned int channels_max; /* max. number of channels */
+ hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */
+ u32 rates; /* supported rates */
+ u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */
+ unsigned int maxbps; /* supported max. bit per sample */
+ struct hda_pcm_ops ops;
+};
+
+/* PCM types */
+enum {
+ HDA_PCM_TYPE_AUDIO,
+ HDA_PCM_TYPE_SPDIF,
+ HDA_PCM_TYPE_HDMI,
+ HDA_PCM_TYPE_MODEM,
+ HDA_PCM_NTYPES
+};
+
+/* for PCM creation */
+struct hda_pcm {
+ char *name;
+ struct hda_pcm_stream stream[2];
+ unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */
+ int device; /* assigned device number */
+};
+
+/* codec information */
+struct hda_codec {
+ struct hda_bus *bus;
+ unsigned int addr; /* codec addr*/
+ struct list_head list; /* list point */
+
+ hda_nid_t afg; /* AFG node id */
+ hda_nid_t mfg; /* MFG node id */
+
+ /* ids */
+ u32 vendor_id;
+ u32 subsystem_id;
+ u32 revision_id;
+
+ /* detected preset */
+ const struct hda_codec_preset *preset;
+
+ /* set by patch */
+ struct hda_codec_ops patch_ops;
+
+ /* PCM to create, set by patch_ops.build_pcms callback */
+ unsigned int num_pcms;
+ struct hda_pcm *pcm_info;
+
+ /* codec specific info */
+ void *spec;
+
+ /* beep device */
+ struct hda_beep *beep;
+
+ /* widget capabilities cache */
+ unsigned int num_nodes;
+ hda_nid_t start_nid;
+ u32 *wcaps;
+
+ struct hda_cache_rec amp_cache; /* cache for amp access */
+ struct hda_cache_rec cmd_cache; /* cache for other commands */
+
+ struct mutex spdif_mutex;
+ unsigned int spdif_status; /* IEC958 status bits */
+ unsigned short spdif_ctls; /* SPDIF control bits */
+ unsigned int spdif_in_enable; /* SPDIF input enable? */
+ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
+
+ struct snd_hwdep *hwdep; /* assigned hwdep device */
+
+ /* misc flags */
+ unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each
+ * status change
+ * (e.g. Realtek codecs)
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ unsigned int power_on :1; /* current (global) power-state */
+ unsigned int power_transition :1; /* power-state in transition */
+ int power_count; /* current (global) power refcount */
+ struct delayed_work power_work; /* delayed task for powerdown */
+#endif
+};
+
+/* direction */
+enum {
+ HDA_INPUT, HDA_OUTPUT
+};
+
+
+/*
+ * constructors
+ */
+int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
+ struct hda_bus **busp);
+int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
+ struct hda_codec **codecp);
+
+/*
+ * low level functions
+ */
+unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
+ int direct,
+ unsigned int verb, unsigned int parm);
+int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
+ unsigned int verb, unsigned int parm);
+#define snd_hda_param_read(codec, nid, param) \
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param)
+int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *start_id);
+int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *conn_list, int max_conns);
+
+struct hda_verb {
+ hda_nid_t nid;
+ u32 verb;
+ u32 param;
+};
+
+void snd_hda_sequence_write(struct hda_codec *codec,
+ const struct hda_verb *seq);
+
+/* unsolicited event */
+int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex);
+
+/* cached write */
+#ifdef SND_HDA_NEEDS_RESUME
+int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
+ int direct, unsigned int verb, unsigned int parm);
+void snd_hda_sequence_write_cache(struct hda_codec *codec,
+ const struct hda_verb *seq);
+void snd_hda_codec_resume_cache(struct hda_codec *codec);
+#else
+#define snd_hda_codec_write_cache snd_hda_codec_write
+#define snd_hda_sequence_write_cache snd_hda_sequence_write
+#endif
+
+/*
+ * Mixer
+ */
+int snd_hda_build_controls(struct hda_bus *bus);
+
+/*
+ * PCM
+ */
+int snd_hda_build_pcms(struct hda_bus *bus);
+void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
+ u32 stream_tag,
+ int channel_id, int format);
+void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid);
+unsigned int snd_hda_calc_stream_format(unsigned int rate,
+ unsigned int channels,
+ unsigned int format,
+ unsigned int maxbps);
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+ u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
+int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int format);
+
+/*
+ * Misc
+ */
+void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen);
+
+/*
+ * power management
+ */
+#ifdef CONFIG_PM
+int snd_hda_suspend(struct hda_bus *bus, pm_message_t state);
+int snd_hda_resume(struct hda_bus *bus);
+#endif
+
+/*
+ * power saving
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+void snd_hda_power_up(struct hda_codec *codec);
+void snd_hda_power_down(struct hda_codec *codec);
+#define snd_hda_codec_needs_resume(codec) codec->power_count
+int snd_hda_codecs_inuse(struct hda_bus *bus);
+#else
+static inline void snd_hda_power_up(struct hda_codec *codec) {}
+static inline void snd_hda_power_down(struct hda_codec *codec) {}
+#define snd_hda_codec_needs_resume(codec) 1
+#define snd_hda_codecs_inuse(bus) 1
+#endif
+
+#endif /* __SOUND_HDA_CODEC_H */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
new file mode 100644
index 0000000..0ca3089
--- /dev/null
+++ b/sound/pci/hda/hda_generic.c
@@ -0,0 +1,1099 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Generic widget tree parser
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+/* widget node for parsing */
+struct hda_gnode {
+ hda_nid_t nid; /* NID of this widget */
+ unsigned short nconns; /* number of input connections */
+ hda_nid_t *conn_list;
+ hda_nid_t slist[2]; /* temporay list */
+ unsigned int wid_caps; /* widget capabilities */
+ unsigned char type; /* widget type */
+ unsigned char pin_ctl; /* pin controls */
+ unsigned char checked; /* the flag indicates that the node is already parsed */
+ unsigned int pin_caps; /* pin widget capabilities */
+ unsigned int def_cfg; /* default configuration */
+ unsigned int amp_out_caps; /* AMP out capabilities */
+ unsigned int amp_in_caps; /* AMP in capabilities */
+ struct list_head list;
+};
+
+/* patch-specific record */
+
+#define MAX_PCM_VOLS 2
+struct pcm_vol {
+ struct hda_gnode *node; /* Node for PCM volume */
+ unsigned int index; /* connection of PCM volume */
+};
+
+struct hda_gspec {
+ struct hda_gnode *dac_node[2]; /* DAC node */
+ struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */
+ struct pcm_vol pcm_vol[MAX_PCM_VOLS]; /* PCM volumes */
+ unsigned int pcm_vol_nodes; /* number of PCM volumes */
+
+ struct hda_gnode *adc_node; /* ADC node */
+ struct hda_gnode *cap_vol_node; /* Node for capture volume */
+ unsigned int cur_cap_src; /* current capture source */
+ struct hda_input_mux input_mux;
+ char cap_labels[HDA_MAX_NUM_INPUTS][16];
+
+ unsigned int def_amp_in_caps;
+ unsigned int def_amp_out_caps;
+
+ struct hda_pcm pcm_rec; /* PCM information */
+
+ struct list_head nid_list; /* list of widgets */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define MAX_LOOPBACK_AMPS 7
+ struct hda_loopback_check loopback;
+ int num_loopbacks;
+ struct hda_amp_list loopback_list[MAX_LOOPBACK_AMPS + 1];
+#endif
+};
+
+/*
+ * retrieve the default device type from the default config value
+ */
+#define defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> \
+ AC_DEFCFG_DEVICE_SHIFT)
+#define defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> \
+ AC_DEFCFG_LOCATION_SHIFT)
+#define defcfg_port_conn(node) (((node)->def_cfg & AC_DEFCFG_PORT_CONN) >> \
+ AC_DEFCFG_PORT_CONN_SHIFT)
+
+/*
+ * destructor
+ */
+static void snd_hda_generic_free(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_gnode *node, *n;
+
+ if (! spec)
+ return;
+ /* free all widgets */
+ list_for_each_entry_safe(node, n, &spec->nid_list, list) {
+ if (node->conn_list != node->slist)
+ kfree(node->conn_list);
+ kfree(node);
+ }
+ kfree(spec);
+}
+
+
+/*
+ * add a new widget node and read its attributes
+ */
+static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid_t nid)
+{
+ struct hda_gnode *node;
+ int nconns;
+ hda_nid_t conn_list[HDA_MAX_CONNECTIONS];
+
+ node = kzalloc(sizeof(*node), GFP_KERNEL);
+ if (node == NULL)
+ return -ENOMEM;
+ node->nid = nid;
+ nconns = snd_hda_get_connections(codec, nid, conn_list,
+ HDA_MAX_CONNECTIONS);
+ if (nconns < 0) {
+ kfree(node);
+ return nconns;
+ }
+ if (nconns <= ARRAY_SIZE(node->slist))
+ node->conn_list = node->slist;
+ else {
+ node->conn_list = kmalloc(sizeof(hda_nid_t) * nconns,
+ GFP_KERNEL);
+ if (! node->conn_list) {
+ snd_printk(KERN_ERR "hda-generic: cannot malloc\n");
+ kfree(node);
+ return -ENOMEM;
+ }
+ }
+ memcpy(node->conn_list, conn_list, nconns * sizeof(hda_nid_t));
+ node->nconns = nconns;
+ node->wid_caps = get_wcaps(codec, nid);
+ node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+
+ if (node->type == AC_WID_PIN) {
+ node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP);
+ node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ }
+
+ if (node->wid_caps & AC_WCAP_OUT_AMP) {
+ if (node->wid_caps & AC_WCAP_AMP_OVRD)
+ node->amp_out_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_OUT_CAP);
+ if (! node->amp_out_caps)
+ node->amp_out_caps = spec->def_amp_out_caps;
+ }
+ if (node->wid_caps & AC_WCAP_IN_AMP) {
+ if (node->wid_caps & AC_WCAP_AMP_OVRD)
+ node->amp_in_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_IN_CAP);
+ if (! node->amp_in_caps)
+ node->amp_in_caps = spec->def_amp_in_caps;
+ }
+ list_add_tail(&node->list, &spec->nid_list);
+ return 0;
+}
+
+/*
+ * build the AFG subtree
+ */
+static int build_afg_tree(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ int i, nodes, err;
+ hda_nid_t nid;
+
+ if (snd_BUG_ON(!spec))
+ return -EINVAL;
+
+ spec->def_amp_out_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_OUT_CAP);
+ spec->def_amp_in_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_IN_CAP);
+
+ nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
+ if (! nid || nodes < 0) {
+ printk(KERN_ERR "Invalid AFG subtree\n");
+ return -EINVAL;
+ }
+
+ /* parse all nodes belonging to the AFG */
+ for (i = 0; i < nodes; i++, nid++) {
+ if ((err = add_new_node(codec, spec, nid)) < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+
+/*
+ * look for the node record for the given NID
+ */
+/* FIXME: should avoid the braindead linear search */
+static struct hda_gnode *hda_get_node(struct hda_gspec *spec, hda_nid_t nid)
+{
+ struct hda_gnode *node;
+
+ list_for_each_entry(node, &spec->nid_list, list) {
+ if (node->nid == nid)
+ return node;
+ }
+ return NULL;
+}
+
+/*
+ * unmute (and set max vol) the output amplifier
+ */
+static int unmute_output(struct hda_codec *codec, struct hda_gnode *node)
+{
+ unsigned int val, ofs;
+ snd_printdd("UNMUTE OUT: NID=0x%x\n", node->nid);
+ val = (node->amp_out_caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+ ofs = (node->amp_out_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
+ if (val >= ofs)
+ val -= ofs;
+ snd_hda_codec_amp_stereo(codec, node->nid, HDA_OUTPUT, 0, 0xff, val);
+ return 0;
+}
+
+/*
+ * unmute (and set max vol) the input amplifier
+ */
+static int unmute_input(struct hda_codec *codec, struct hda_gnode *node, unsigned int index)
+{
+ unsigned int val, ofs;
+ snd_printdd("UNMUTE IN: NID=0x%x IDX=0x%x\n", node->nid, index);
+ val = (node->amp_in_caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+ ofs = (node->amp_in_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
+ if (val >= ofs)
+ val -= ofs;
+ snd_hda_codec_amp_stereo(codec, node->nid, HDA_INPUT, index, 0xff, val);
+ return 0;
+}
+
+/*
+ * select the input connection of the given node.
+ */
+static int select_input_connection(struct hda_codec *codec, struct hda_gnode *node,
+ unsigned int index)
+{
+ snd_printdd("CONNECT: NID=0x%x IDX=0x%x\n", node->nid, index);
+ return snd_hda_codec_write_cache(codec, node->nid, 0,
+ AC_VERB_SET_CONNECT_SEL, index);
+}
+
+/*
+ * clear checked flag of each node in the node list
+ */
+static void clear_check_flags(struct hda_gspec *spec)
+{
+ struct hda_gnode *node;
+
+ list_for_each_entry(node, &spec->nid_list, list) {
+ node->checked = 0;
+ }
+}
+
+/*
+ * parse the output path recursively until reach to an audio output widget
+ *
+ * returns 0 if not found, 1 if found, or a negative error code.
+ */
+static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec,
+ struct hda_gnode *node, int dac_idx)
+{
+ int i, err;
+ struct hda_gnode *child;
+
+ if (node->checked)
+ return 0;
+
+ node->checked = 1;
+ if (node->type == AC_WID_AUD_OUT) {
+ if (node->wid_caps & AC_WCAP_DIGITAL) {
+ snd_printdd("Skip Digital OUT node %x\n", node->nid);
+ return 0;
+ }
+ snd_printdd("AUD_OUT found %x\n", node->nid);
+ if (spec->dac_node[dac_idx]) {
+ /* already DAC node is assigned, just unmute & connect */
+ return node == spec->dac_node[dac_idx];
+ }
+ spec->dac_node[dac_idx] = node;
+ if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
+ spec->pcm_vol_nodes < MAX_PCM_VOLS) {
+ spec->pcm_vol[spec->pcm_vol_nodes].node = node;
+ spec->pcm_vol[spec->pcm_vol_nodes].index = 0;
+ spec->pcm_vol_nodes++;
+ }
+ return 1; /* found */
+ }
+
+ for (i = 0; i < node->nconns; i++) {
+ child = hda_get_node(spec, node->conn_list[i]);
+ if (! child)
+ continue;
+ err = parse_output_path(codec, spec, child, dac_idx);
+ if (err < 0)
+ return err;
+ else if (err > 0) {
+ /* found one,
+ * select the path, unmute both input and output
+ */
+ if (node->nconns > 1)
+ select_input_connection(codec, node, i);
+ unmute_input(codec, node, i);
+ unmute_output(codec, node);
+ if (spec->dac_node[dac_idx] &&
+ spec->pcm_vol_nodes < MAX_PCM_VOLS &&
+ !(spec->dac_node[dac_idx]->wid_caps &
+ AC_WCAP_OUT_AMP)) {
+ if ((node->wid_caps & AC_WCAP_IN_AMP) ||
+ (node->wid_caps & AC_WCAP_OUT_AMP)) {
+ int n = spec->pcm_vol_nodes;
+ spec->pcm_vol[n].node = node;
+ spec->pcm_vol[n].index = i;
+ spec->pcm_vol_nodes++;
+ }
+ }
+ return 1;
+ }
+ }
+ return 0;
+}
+
+/*
+ * Look for the output PIN widget with the given jack type
+ * and parse the output path to that PIN.
+ *
+ * Returns the PIN node when the path to DAC is established.
+ */
+static struct hda_gnode *parse_output_jack(struct hda_codec *codec,
+ struct hda_gspec *spec,
+ int jack_type)
+{
+ struct hda_gnode *node;
+ int err;
+
+ list_for_each_entry(node, &spec->nid_list, list) {
+ if (node->type != AC_WID_PIN)
+ continue;
+ /* output capable? */
+ if (! (node->pin_caps & AC_PINCAP_OUT))
+ continue;
+ if (defcfg_port_conn(node) == AC_JACK_PORT_NONE)
+ continue; /* unconnected */
+ if (jack_type >= 0) {
+ if (jack_type != defcfg_type(node))
+ continue;
+ if (node->wid_caps & AC_WCAP_DIGITAL)
+ continue; /* skip SPDIF */
+ } else {
+ /* output as default? */
+ if (! (node->pin_ctl & AC_PINCTL_OUT_EN))
+ continue;
+ }
+ clear_check_flags(spec);
+ err = parse_output_path(codec, spec, node, 0);
+ if (err < 0)
+ return NULL;
+ if (! err && spec->out_pin_node[0]) {
+ err = parse_output_path(codec, spec, node, 1);
+ if (err < 0)
+ return NULL;
+ }
+ if (err > 0) {
+ /* unmute the PIN output */
+ unmute_output(codec, node);
+ /* set PIN-Out enable */
+ snd_hda_codec_write_cache(codec, node->nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ AC_PINCTL_OUT_EN |
+ ((node->pin_caps & AC_PINCAP_HP_DRV) ?
+ AC_PINCTL_HP_EN : 0));
+ return node;
+ }
+ }
+ return NULL;
+}
+
+
+/*
+ * parse outputs
+ */
+static int parse_output(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_gnode *node;
+
+ /*
+ * Look for the output PIN widget
+ */
+ /* first, look for the line-out pin */
+ node = parse_output_jack(codec, spec, AC_JACK_LINE_OUT);
+ if (node) /* found, remember the PIN node */
+ spec->out_pin_node[0] = node;
+ else {
+ /* if no line-out is found, try speaker out */
+ node = parse_output_jack(codec, spec, AC_JACK_SPEAKER);
+ if (node)
+ spec->out_pin_node[0] = node;
+ }
+ /* look for the HP-out pin */
+ node = parse_output_jack(codec, spec, AC_JACK_HP_OUT);
+ if (node) {
+ if (! spec->out_pin_node[0])
+ spec->out_pin_node[0] = node;
+ else
+ spec->out_pin_node[1] = node;
+ }
+
+ if (! spec->out_pin_node[0]) {
+ /* no line-out or HP pins found,
+ * then choose for the first output pin
+ */
+ spec->out_pin_node[0] = parse_output_jack(codec, spec, -1);
+ if (! spec->out_pin_node[0])
+ snd_printd("hda_generic: no proper output path found\n");
+ }
+
+ return 0;
+}
+
+/*
+ * input MUX
+ */
+
+/* control callbacks */
+static int capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_gspec *spec = codec->spec;
+ return snd_hda_input_mux_info(&spec->input_mux, uinfo);
+}
+
+static int capture_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_gspec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->cur_cap_src;
+ return 0;
+}
+
+static int capture_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_gspec *spec = codec->spec;
+ return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol,
+ spec->adc_node->nid, &spec->cur_cap_src);
+}
+
+/*
+ * return the string name of the given input PIN widget
+ */
+static const char *get_input_type(struct hda_gnode *node, unsigned int *pinctl)
+{
+ unsigned int location = defcfg_location(node);
+ switch (defcfg_type(node)) {
+ case AC_JACK_LINE_IN:
+ if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+ return "Front Line";
+ return "Line";
+ case AC_JACK_CD:
+#if 0
+ if (pinctl)
+ *pinctl |= AC_PINCTL_VREF_GRD;
+#endif
+ return "CD";
+ case AC_JACK_AUX:
+ if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+ return "Front Aux";
+ return "Aux";
+ case AC_JACK_MIC_IN:
+ if (pinctl &&
+ (node->pin_caps &
+ (AC_PINCAP_VREF_80 << AC_PINCAP_VREF_SHIFT)))
+ *pinctl |= AC_PINCTL_VREF_80;
+ if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+ return "Front Mic";
+ return "Mic";
+ case AC_JACK_SPDIF_IN:
+ return "SPDIF";
+ case AC_JACK_DIG_OTHER_IN:
+ return "Digital";
+ }
+ return NULL;
+}
+
+/*
+ * parse the nodes recursively until reach to the input PIN
+ *
+ * returns 0 if not found, 1 if found, or a negative error code.
+ */
+static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec,
+ struct hda_gnode *node)
+{
+ int i, err;
+ unsigned int pinctl;
+ char *label;
+ const char *type;
+
+ if (node->checked)
+ return 0;
+
+ node->checked = 1;
+ if (node->type != AC_WID_PIN) {
+ for (i = 0; i < node->nconns; i++) {
+ struct hda_gnode *child;
+ child = hda_get_node(spec, node->conn_list[i]);
+ if (! child)
+ continue;
+ err = parse_adc_sub_nodes(codec, spec, child);
+ if (err < 0)
+ return err;
+ if (err > 0) {
+ /* found one,
+ * select the path, unmute both input and output
+ */
+ if (node->nconns > 1)
+ select_input_connection(codec, node, i);
+ unmute_input(codec, node, i);
+ unmute_output(codec, node);
+ return err;
+ }
+ }
+ return 0;
+ }
+
+ /* input capable? */
+ if (! (node->pin_caps & AC_PINCAP_IN))
+ return 0;
+
+ if (defcfg_port_conn(node) == AC_JACK_PORT_NONE)
+ return 0; /* unconnected */
+
+ if (node->wid_caps & AC_WCAP_DIGITAL)
+ return 0; /* skip SPDIF */
+
+ if (spec->input_mux.num_items >= HDA_MAX_NUM_INPUTS) {
+ snd_printk(KERN_ERR "hda_generic: Too many items for capture\n");
+ return -EINVAL;
+ }
+
+ pinctl = AC_PINCTL_IN_EN;
+ /* create a proper capture source label */
+ type = get_input_type(node, &pinctl);
+ if (! type) {
+ /* input as default? */
+ if (! (node->pin_ctl & AC_PINCTL_IN_EN))
+ return 0;
+ type = "Input";
+ }
+ label = spec->cap_labels[spec->input_mux.num_items];
+ strcpy(label, type);
+ spec->input_mux.items[spec->input_mux.num_items].label = label;
+
+ /* unmute the PIN external input */
+ unmute_input(codec, node, 0); /* index = 0? */
+ /* set PIN-In enable */
+ snd_hda_codec_write_cache(codec, node->nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+
+ return 1; /* found */
+}
+
+/* add a capture source element */
+static void add_cap_src(struct hda_gspec *spec, int idx)
+{
+ struct hda_input_mux_item *csrc;
+ char *buf;
+ int num, ocap;
+
+ num = spec->input_mux.num_items;
+ csrc = &spec->input_mux.items[num];
+ buf = spec->cap_labels[num];
+ for (ocap = 0; ocap < num; ocap++) {
+ if (! strcmp(buf, spec->cap_labels[ocap])) {
+ /* same label already exists,
+ * put the index number to be unique
+ */
+ sprintf(buf, "%s %d", spec->cap_labels[ocap], num);
+ break;
+ }
+ }
+ csrc->index = idx;
+ spec->input_mux.num_items++;
+}
+
+/*
+ * parse input
+ */
+static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_gnode *node;
+ int i, err;
+
+ snd_printdd("AUD_IN = %x\n", adc_node->nid);
+ clear_check_flags(spec);
+
+ // awk added - fixed no recording due to muted widget
+ unmute_input(codec, adc_node, 0);
+
+ /*
+ * check each connection of the ADC
+ * if it reaches to a proper input PIN, add the path as the
+ * input path.
+ */
+ /* first, check the direct connections to PIN widgets */
+ for (i = 0; i < adc_node->nconns; i++) {
+ node = hda_get_node(spec, adc_node->conn_list[i]);
+ if (node && node->type == AC_WID_PIN) {
+ err = parse_adc_sub_nodes(codec, spec, node);
+ if (err < 0)
+ return err;
+ else if (err > 0)
+ add_cap_src(spec, i);
+ }
+ }
+ /* ... then check the rests, more complicated connections */
+ for (i = 0; i < adc_node->nconns; i++) {
+ node = hda_get_node(spec, adc_node->conn_list[i]);
+ if (node && node->type != AC_WID_PIN) {
+ err = parse_adc_sub_nodes(codec, spec, node);
+ if (err < 0)
+ return err;
+ else if (err > 0)
+ add_cap_src(spec, i);
+ }
+ }
+
+ if (! spec->input_mux.num_items)
+ return 0; /* no input path found... */
+
+ snd_printdd("[Capture Source] NID=0x%x, #SRC=%d\n", adc_node->nid, spec->input_mux.num_items);
+ for (i = 0; i < spec->input_mux.num_items; i++)
+ snd_printdd(" [%s] IDX=0x%x\n", spec->input_mux.items[i].label,
+ spec->input_mux.items[i].index);
+
+ spec->adc_node = adc_node;
+ return 1;
+}
+
+/*
+ * parse input
+ */
+static int parse_input(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_gnode *node;
+ int err;
+
+ /*
+ * At first we look for an audio input widget.
+ * If it reaches to certain input PINs, we take it as the
+ * input path.
+ */
+ list_for_each_entry(node, &spec->nid_list, list) {
+ if (node->wid_caps & AC_WCAP_DIGITAL)
+ continue; /* skip SPDIF */
+ if (node->type == AC_WID_AUD_IN) {
+ err = parse_input_path(codec, node);
+ if (err < 0)
+ return err;
+ else if (err > 0)
+ return 0;
+ }
+ }
+ snd_printd("hda_generic: no proper input path found\n");
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void add_input_loopback(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_amp_list *p;
+
+ if (spec->num_loopbacks >= MAX_LOOPBACK_AMPS) {
+ snd_printk(KERN_ERR "hda_generic: Too many loopback ctls\n");
+ return;
+ }
+ p = &spec->loopback_list[spec->num_loopbacks++];
+ p->nid = nid;
+ p->dir = dir;
+ p->idx = idx;
+ spec->loopback.amplist = spec->loopback_list;
+}
+#else
+#define add_input_loopback(codec,nid,dir,idx)
+#endif
+
+/*
+ * create mixer controls if possible
+ */
+static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
+ unsigned int index, const char *type,
+ const char *dir_sfx, int is_loopback)
+{
+ char name[32];
+ int err;
+ int created = 0;
+ struct snd_kcontrol_new knew;
+
+ if (type)
+ sprintf(name, "%s %s Switch", type, dir_sfx);
+ else
+ sprintf(name, "%s Switch", dir_sfx);
+ if ((node->wid_caps & AC_WCAP_IN_AMP) &&
+ (node->amp_in_caps & AC_AMPCAP_MUTE)) {
+ knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT);
+ if (is_loopback)
+ add_input_loopback(codec, node->nid, HDA_INPUT, index);
+ snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
+ if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+ return err;
+ created = 1;
+ } else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
+ (node->amp_out_caps & AC_AMPCAP_MUTE)) {
+ knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT);
+ if (is_loopback)
+ add_input_loopback(codec, node->nid, HDA_OUTPUT, 0);
+ snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
+ if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+ return err;
+ created = 1;
+ }
+
+ if (type)
+ sprintf(name, "%s %s Volume", type, dir_sfx);
+ else
+ sprintf(name, "%s Volume", dir_sfx);
+ if ((node->wid_caps & AC_WCAP_IN_AMP) &&
+ (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) {
+ knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT);
+ snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
+ if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+ return err;
+ created = 1;
+ } else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
+ (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) {
+ knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT);
+ snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
+ if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+ return err;
+ created = 1;
+ }
+
+ return created;
+}
+
+/*
+ * check whether the controls with the given name and direction suffix already exist
+ */
+static int check_existing_control(struct hda_codec *codec, const char *type, const char *dir)
+{
+ struct snd_ctl_elem_id id;
+ memset(&id, 0, sizeof(id));
+ sprintf(id.name, "%s %s Volume", type, dir);
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ if (snd_ctl_find_id(codec->bus->card, &id))
+ return 1;
+ sprintf(id.name, "%s %s Switch", type, dir);
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ if (snd_ctl_find_id(codec->bus->card, &id))
+ return 1;
+ return 0;
+}
+
+/*
+ * build output mixer controls
+ */
+static int create_output_mixers(struct hda_codec *codec, const char **names)
+{
+ struct hda_gspec *spec = codec->spec;
+ int i, err;
+
+ for (i = 0; i < spec->pcm_vol_nodes; i++) {
+ err = create_mixer(codec, spec->pcm_vol[i].node,
+ spec->pcm_vol[i].index,
+ names[i], "Playback", 0);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int build_output_controls(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ static const char *types_speaker[] = { "Speaker", "Headphone" };
+ static const char *types_line[] = { "Front", "Headphone" };
+
+ switch (spec->pcm_vol_nodes) {
+ case 1:
+ return create_mixer(codec, spec->pcm_vol[0].node,
+ spec->pcm_vol[0].index,
+ "Master", "Playback", 0);
+ case 2:
+ if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER)
+ return create_output_mixers(codec, types_speaker);
+ else
+ return create_output_mixers(codec, types_line);
+ }
+ return 0;
+}
+
+/* create capture volume/switch */
+static int build_input_controls(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_gnode *adc_node = spec->adc_node;
+ int i, err;
+ static struct snd_kcontrol_new cap_sel = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = capture_source_info,
+ .get = capture_source_get,
+ .put = capture_source_put,
+ };
+
+ if (! adc_node || ! spec->input_mux.num_items)
+ return 0; /* not found */
+
+ spec->cur_cap_src = 0;
+ select_input_connection(codec, adc_node,
+ spec->input_mux.items[0].index);
+
+ /* create capture volume and switch controls if the ADC has an amp */
+ /* do we have only a single item? */
+ if (spec->input_mux.num_items == 1) {
+ err = create_mixer(codec, adc_node,
+ spec->input_mux.items[0].index,
+ NULL, "Capture", 0);
+ if (err < 0)
+ return err;
+ return 0;
+ }
+
+ /* create input MUX if multiple sources are available */
+ if ((err = snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&cap_sel, codec))) < 0)
+ return err;
+
+ /* no volume control? */
+ if (! (adc_node->wid_caps & AC_WCAP_IN_AMP) ||
+ ! (adc_node->amp_in_caps & AC_AMPCAP_NUM_STEPS))
+ return 0;
+
+ for (i = 0; i < spec->input_mux.num_items; i++) {
+ struct snd_kcontrol_new knew;
+ char name[32];
+ sprintf(name, "%s Capture Volume",
+ spec->input_mux.items[i].label);
+ knew = (struct snd_kcontrol_new)
+ HDA_CODEC_VOLUME(name, adc_node->nid,
+ spec->input_mux.items[i].index,
+ HDA_INPUT);
+ if ((err = snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&knew, codec))) < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+
+/*
+ * parse the nodes recursively until reach to the output PIN.
+ *
+ * returns 0 - if not found,
+ * 1 - if found, but no mixer is created
+ * 2 - if found and mixer was already created, (just skip)
+ * a negative error code
+ */
+static int parse_loopback_path(struct hda_codec *codec, struct hda_gspec *spec,
+ struct hda_gnode *node, struct hda_gnode *dest_node,
+ const char *type)
+{
+ int i, err;
+
+ if (node->checked)
+ return 0;
+
+ node->checked = 1;
+ if (node == dest_node) {
+ /* loopback connection found */
+ return 1;
+ }
+
+ for (i = 0; i < node->nconns; i++) {
+ struct hda_gnode *child = hda_get_node(spec, node->conn_list[i]);
+ if (! child)
+ continue;
+ err = parse_loopback_path(codec, spec, child, dest_node, type);
+ if (err < 0)
+ return err;
+ else if (err >= 1) {
+ if (err == 1) {
+ err = create_mixer(codec, node, i, type,
+ "Playback", 1);
+ if (err < 0)
+ return err;
+ if (err > 0)
+ return 2; /* ok, created */
+ /* not created, maybe in the lower path */
+ err = 1;
+ }
+ /* connect and unmute */
+ if (node->nconns > 1)
+ select_input_connection(codec, node, i);
+ unmute_input(codec, node, i);
+ unmute_output(codec, node);
+ return err;
+ }
+ }
+ return 0;
+}
+
+/*
+ * parse the tree and build the loopback controls
+ */
+static int build_loopback_controls(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_gnode *node;
+ int err;
+ const char *type;
+
+ if (! spec->out_pin_node[0])
+ return 0;
+
+ list_for_each_entry(node, &spec->nid_list, list) {
+ if (node->type != AC_WID_PIN)
+ continue;
+ /* input capable? */
+ if (! (node->pin_caps & AC_PINCAP_IN))
+ return 0;
+ type = get_input_type(node, NULL);
+ if (type) {
+ if (check_existing_control(codec, type, "Playback"))
+ continue;
+ clear_check_flags(spec);
+ err = parse_loopback_path(codec, spec,
+ spec->out_pin_node[0],
+ node, type);
+ if (err < 0)
+ return err;
+ if (! err)
+ continue;
+ }
+ }
+ return 0;
+}
+
+/*
+ * build mixer controls
+ */
+static int build_generic_controls(struct hda_codec *codec)
+{
+ int err;
+
+ if ((err = build_input_controls(codec)) < 0 ||
+ (err = build_output_controls(codec)) < 0 ||
+ (err = build_loopback_controls(codec)) < 0)
+ return err;
+
+ return 0;
+}
+
+/*
+ * PCM
+ */
+static struct hda_pcm_stream generic_pcm_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int generic_pcm2_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct hda_gspec *spec = codec->spec;
+
+ snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format);
+ snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid,
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int generic_pcm2_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct hda_gspec *spec = codec->spec;
+
+ snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+ snd_hda_codec_cleanup_stream(codec, spec->dac_node[1]->nid);
+ return 0;
+}
+
+static int build_generic_pcms(struct hda_codec *codec)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_pcm *info = &spec->pcm_rec;
+
+ if (! spec->dac_node[0] && ! spec->adc_node) {
+ snd_printd("hda_generic: no PCM found\n");
+ return 0;
+ }
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "HDA Generic";
+ if (spec->dac_node[0]) {
+ info->stream[0] = generic_pcm_playback;
+ info->stream[0].nid = spec->dac_node[0]->nid;
+ if (spec->dac_node[1]) {
+ info->stream[0].ops.prepare = generic_pcm2_prepare;
+ info->stream[0].ops.cleanup = generic_pcm2_cleanup;
+ }
+ }
+ if (spec->adc_node) {
+ info->stream[1] = generic_pcm_playback;
+ info->stream[1].nid = spec->adc_node->nid;
+ }
+
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int generic_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct hda_gspec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
+
+/*
+ */
+static struct hda_codec_ops generic_patch_ops = {
+ .build_controls = build_generic_controls,
+ .build_pcms = build_generic_pcms,
+ .free = snd_hda_generic_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = generic_check_power_status,
+#endif
+};
+
+/*
+ * the generic parser
+ */
+int snd_hda_parse_generic_codec(struct hda_codec *codec)
+{
+ struct hda_gspec *spec;
+ int err;
+
+ if(!codec->afg)
+ return 0;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL) {
+ printk(KERN_ERR "hda_generic: can't allocate spec\n");
+ return -ENOMEM;
+ }
+ codec->spec = spec;
+ INIT_LIST_HEAD(&spec->nid_list);
+
+ if ((err = build_afg_tree(codec)) < 0)
+ goto error;
+
+ if ((err = parse_input(codec)) < 0 ||
+ (err = parse_output(codec)) < 0)
+ goto error;
+
+ codec->patch_ops = generic_patch_ops;
+
+ return 0;
+
+ error:
+ snd_hda_generic_free(codec);
+ return err;
+}
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
new file mode 100644
index 0000000..6e18a42
--- /dev/null
+++ b/sound/pci/hda/hda_hwdep.c
@@ -0,0 +1,121 @@
+/*
+ * HWDEP Interface for HD-audio codec
+ *
+ * Copyright (c) 2007 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/compat.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include <sound/hda_hwdep.h>
+
+/*
+ * write/read an out-of-bound verb
+ */
+static int verb_write_ioctl(struct hda_codec *codec,
+ struct hda_verb_ioctl __user *arg)
+{
+ u32 verb, res;
+
+ if (get_user(verb, &arg->verb))
+ return -EFAULT;
+ res = snd_hda_codec_read(codec, verb >> 24, 0,
+ (verb >> 8) & 0xffff, verb & 0xff);
+ if (put_user(res, &arg->res))
+ return -EFAULT;
+ return 0;
+}
+
+static int get_wcap_ioctl(struct hda_codec *codec,
+ struct hda_verb_ioctl __user *arg)
+{
+ u32 verb, res;
+
+ if (get_user(verb, &arg->verb))
+ return -EFAULT;
+ res = get_wcaps(codec, verb >> 24);
+ if (put_user(res, &arg->res))
+ return -EFAULT;
+ return 0;
+}
+
+
+/*
+ */
+static int hda_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct hda_codec *codec = hw->private_data;
+ void __user *argp = (void __user *)arg;
+
+ switch (cmd) {
+ case HDA_IOCTL_PVERSION:
+ return put_user(HDA_HWDEP_VERSION, (int __user *)argp);
+ case HDA_IOCTL_VERB_WRITE:
+ return verb_write_ioctl(codec, argp);
+ case HDA_IOCTL_GET_WCAP:
+ return get_wcap_ioctl(codec, argp);
+ }
+ return -ENOIOCTLCMD;
+}
+
+#ifdef CONFIG_COMPAT
+static int hda_hwdep_ioctl_compat(struct snd_hwdep *hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hda_hwdep_ioctl(hw, file, cmd, (unsigned long)compat_ptr(arg));
+}
+#endif
+
+static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+#ifndef CONFIG_SND_DEBUG_VERBOSE
+ if (!capable(CAP_SYS_RAWIO))
+ return -EACCES;
+#endif
+ return 0;
+}
+
+int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
+{
+ char hwname[16];
+ struct snd_hwdep *hwdep;
+ int err;
+
+ sprintf(hwname, "HDA Codec %d", codec->addr);
+ err = snd_hwdep_new(codec->bus->card, hwname, codec->addr, &hwdep);
+ if (err < 0)
+ return err;
+ codec->hwdep = hwdep;
+ sprintf(hwdep->name, "HDA Codec %d", codec->addr);
+ hwdep->iface = SNDRV_HWDEP_IFACE_HDA;
+ hwdep->private_data = codec;
+ hwdep->exclusive = 1;
+
+ hwdep->ops.open = hda_hwdep_open;
+ hwdep->ops.ioctl = hda_hwdep_ioctl;
+#ifdef CONFIG_COMPAT
+ hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat;
+#endif
+
+ return 0;
+}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
new file mode 100644
index 0000000..34a5b69
--- /dev/null
+++ b/sound/pci/hda/hda_intel.c
@@ -0,0 +1,2496 @@
+/*
+ *
+ * hda_intel.c - Implementation of primary alsa driver code base
+ * for Intel HD Audio.
+ *
+ * Copyright(c) 2004 Intel Corporation. All rights reserved.
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ * PeiSen Hou <pshou@realtek.com.tw>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the Free
+ * Software Foundation; either version 2 of the License, or (at your option)
+ * any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
+ * more details.
+ *
+ * You should have received a copy of the GNU General Public License along with
+ * this program; if not, write to the Free Software Foundation, Inc., 59
+ * Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ *
+ * CONTACTS:
+ *
+ * Matt Jared matt.jared@intel.com
+ * Andy Kopp andy.kopp@intel.com
+ * Dan Kogan dan.d.kogan@intel.com
+ *
+ * CHANGES:
+ *
+ * 2004.12.01 Major rewrite by tiwai, merged the work of pshou
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/mutex.h>
+#include <linux/reboot.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include "hda_codec.h"
+
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+static char *model[SNDRV_CARDS];
+static int position_fix[SNDRV_CARDS];
+static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
+static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
+static int single_cmd;
+static int enable_msi;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Intel HD audio interface.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Intel HD audio interface.");
+module_param_array(model, charp, NULL, 0444);
+MODULE_PARM_DESC(model, "Use the given board model.");
+module_param_array(position_fix, int, NULL, 0444);
+MODULE_PARM_DESC(position_fix, "Fix DMA pointer "
+ "(0 = auto, 1 = none, 2 = POSBUF).");
+module_param_array(bdl_pos_adj, int, NULL, 0644);
+MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
+module_param_array(probe_mask, int, NULL, 0444);
+MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1).");
+module_param(single_cmd, bool, 0444);
+MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
+ "(for debugging only).");
+module_param(enable_msi, int, 0444);
+MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* power_save option is defined in hda_codec.c */
+
+/* reset the HD-audio controller in power save mode.
+ * this may give more power-saving, but will take longer time to
+ * wake up.
+ */
+static int power_save_controller = 1;
+module_param(power_save_controller, bool, 0644);
+MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
+#endif
+
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
+ "{Intel, ICH6M},"
+ "{Intel, ICH7},"
+ "{Intel, ESB2},"
+ "{Intel, ICH8},"
+ "{Intel, ICH9},"
+ "{Intel, ICH10},"
+ "{Intel, PCH},"
+ "{Intel, SCH},"
+ "{ATI, SB450},"
+ "{ATI, SB600},"
+ "{ATI, RS600},"
+ "{ATI, RS690},"
+ "{ATI, RS780},"
+ "{ATI, R600},"
+ "{ATI, RV630},"
+ "{ATI, RV610},"
+ "{ATI, RV670},"
+ "{ATI, RV635},"
+ "{ATI, RV620},"
+ "{ATI, RV770},"
+ "{VIA, VT8251},"
+ "{VIA, VT8237A},"
+ "{SiS, SIS966},"
+ "{ULI, M5461}}");
+MODULE_DESCRIPTION("Intel HDA driver");
+
+#define SFX "hda-intel: "
+
+
+/*
+ * registers
+ */
+#define ICH6_REG_GCAP 0x00
+#define ICH6_REG_VMIN 0x02
+#define ICH6_REG_VMAJ 0x03
+#define ICH6_REG_OUTPAY 0x04
+#define ICH6_REG_INPAY 0x06
+#define ICH6_REG_GCTL 0x08
+#define ICH6_REG_WAKEEN 0x0c
+#define ICH6_REG_STATESTS 0x0e
+#define ICH6_REG_GSTS 0x10
+#define ICH6_REG_INTCTL 0x20
+#define ICH6_REG_INTSTS 0x24
+#define ICH6_REG_WALCLK 0x30
+#define ICH6_REG_SYNC 0x34
+#define ICH6_REG_CORBLBASE 0x40
+#define ICH6_REG_CORBUBASE 0x44
+#define ICH6_REG_CORBWP 0x48
+#define ICH6_REG_CORBRP 0x4A
+#define ICH6_REG_CORBCTL 0x4c
+#define ICH6_REG_CORBSTS 0x4d
+#define ICH6_REG_CORBSIZE 0x4e
+
+#define ICH6_REG_RIRBLBASE 0x50
+#define ICH6_REG_RIRBUBASE 0x54
+#define ICH6_REG_RIRBWP 0x58
+#define ICH6_REG_RINTCNT 0x5a
+#define ICH6_REG_RIRBCTL 0x5c
+#define ICH6_REG_RIRBSTS 0x5d
+#define ICH6_REG_RIRBSIZE 0x5e
+
+#define ICH6_REG_IC 0x60
+#define ICH6_REG_IR 0x64
+#define ICH6_REG_IRS 0x68
+#define ICH6_IRS_VALID (1<<1)
+#define ICH6_IRS_BUSY (1<<0)
+
+#define ICH6_REG_DPLBASE 0x70
+#define ICH6_REG_DPUBASE 0x74
+#define ICH6_DPLBASE_ENABLE 0x1 /* Enable position buffer */
+
+/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
+enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
+
+/* stream register offsets from stream base */
+#define ICH6_REG_SD_CTL 0x00
+#define ICH6_REG_SD_STS 0x03
+#define ICH6_REG_SD_LPIB 0x04
+#define ICH6_REG_SD_CBL 0x08
+#define ICH6_REG_SD_LVI 0x0c
+#define ICH6_REG_SD_FIFOW 0x0e
+#define ICH6_REG_SD_FIFOSIZE 0x10
+#define ICH6_REG_SD_FORMAT 0x12
+#define ICH6_REG_SD_BDLPL 0x18
+#define ICH6_REG_SD_BDLPU 0x1c
+
+/* PCI space */
+#define ICH6_PCIREG_TCSEL 0x44
+
+/*
+ * other constants
+ */
+
+/* max number of SDs */
+/* ICH, ATI and VIA have 4 playback and 4 capture */
+#define ICH6_NUM_CAPTURE 4
+#define ICH6_NUM_PLAYBACK 4
+
+/* ULI has 6 playback and 5 capture */
+#define ULI_NUM_CAPTURE 5
+#define ULI_NUM_PLAYBACK 6
+
+/* ATI HDMI has 1 playback and 0 capture */
+#define ATIHDMI_NUM_CAPTURE 0
+#define ATIHDMI_NUM_PLAYBACK 1
+
+/* TERA has 4 playback and 3 capture */
+#define TERA_NUM_CAPTURE 3
+#define TERA_NUM_PLAYBACK 4
+
+/* this number is statically defined for simplicity */
+#define MAX_AZX_DEV 16
+
+/* max number of fragments - we may use more if allocating more pages for BDL */
+#define BDL_SIZE 4096
+#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16)
+#define AZX_MAX_FRAG 32
+/* max buffer size - no h/w limit, you can increase as you like */
+#define AZX_MAX_BUF_SIZE (1024*1024*1024)
+/* max number of PCM devics per card */
+#define AZX_MAX_PCMS 8
+
+/* RIRB int mask: overrun[2], response[0] */
+#define RIRB_INT_RESPONSE 0x01
+#define RIRB_INT_OVERRUN 0x04
+#define RIRB_INT_MASK 0x05
+
+/* STATESTS int mask: S3,SD2,SD1,SD0 */
+#define AZX_MAX_CODECS 4
+#define STATESTS_INT_MASK 0x0f
+
+/* SD_CTL bits */
+#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */
+#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */
+#define SD_CTL_STRIPE (3 << 16) /* stripe control */
+#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */
+#define SD_CTL_DIR (1 << 19) /* bi-directional stream */
+#define SD_CTL_STREAM_TAG_MASK (0xf << 20)
+#define SD_CTL_STREAM_TAG_SHIFT 20
+
+/* SD_CTL and SD_STS */
+#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */
+#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */
+#define SD_INT_COMPLETE 0x04 /* completion interrupt */
+#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\
+ SD_INT_COMPLETE)
+
+/* SD_STS */
+#define SD_STS_FIFO_READY 0x20 /* FIFO ready */
+
+/* INTCTL and INTSTS */
+#define ICH6_INT_ALL_STREAM 0xff /* all stream interrupts */
+#define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */
+#define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */
+
+/* GCTL unsolicited response enable bit */
+#define ICH6_GCTL_UREN (1<<8)
+
+/* GCTL reset bit */
+#define ICH6_GCTL_RESET (1<<0)
+
+/* CORB/RIRB control, read/write pointer */
+#define ICH6_RBCTL_DMA_EN 0x02 /* enable DMA */
+#define ICH6_RBCTL_IRQ_EN 0x01 /* enable IRQ */
+#define ICH6_RBRWP_CLR 0x8000 /* read/write pointer clear */
+/* below are so far hardcoded - should read registers in future */
+#define ICH6_MAX_CORB_ENTRIES 256
+#define ICH6_MAX_RIRB_ENTRIES 256
+
+/* position fix mode */
+enum {
+ POS_FIX_AUTO,
+ POS_FIX_LPIB,
+ POS_FIX_POSBUF,
+};
+
+/* Defines for ATI HD Audio support in SB450 south bridge */
+#define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42
+#define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02
+
+/* Defines for Nvidia HDA support */
+#define NVIDIA_HDA_TRANSREG_ADDR 0x4e
+#define NVIDIA_HDA_ENABLE_COHBITS 0x0f
+#define NVIDIA_HDA_ISTRM_COH 0x4d
+#define NVIDIA_HDA_OSTRM_COH 0x4c
+#define NVIDIA_HDA_ENABLE_COHBIT 0x01
+
+/* Defines for Intel SCH HDA snoop control */
+#define INTEL_SCH_HDA_DEVC 0x78
+#define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11)
+
+/* Define IN stream 0 FIFO size offset in VIA controller */
+#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90
+/* Define VIA HD Audio Device ID*/
+#define VIA_HDAC_DEVICE_ID 0x3288
+
+
+/*
+ */
+
+struct azx_dev {
+ struct snd_dma_buffer bdl; /* BDL buffer */
+ u32 *posbuf; /* position buffer pointer */
+
+ unsigned int bufsize; /* size of the play buffer in bytes */
+ unsigned int period_bytes; /* size of the period in bytes */
+ unsigned int frags; /* number for period in the play buffer */
+ unsigned int fifo_size; /* FIFO size */
+
+ void __iomem *sd_addr; /* stream descriptor pointer */
+
+ u32 sd_int_sta_mask; /* stream int status mask */
+
+ /* pcm support */
+ struct snd_pcm_substream *substream; /* assigned substream,
+ * set in PCM open
+ */
+ unsigned int format_val; /* format value to be set in the
+ * controller and the codec
+ */
+ unsigned char stream_tag; /* assigned stream */
+ unsigned char index; /* stream index */
+
+ unsigned int opened :1;
+ unsigned int running :1;
+ unsigned int irq_pending :1;
+ unsigned int irq_ignore :1;
+ /*
+ * For VIA:
+ * A flag to ensure DMA position is 0
+ * when link position is not greater than FIFO size
+ */
+ unsigned int insufficient :1;
+};
+
+/* CORB/RIRB */
+struct azx_rb {
+ u32 *buf; /* CORB/RIRB buffer
+ * Each CORB entry is 4byte, RIRB is 8byte
+ */
+ dma_addr_t addr; /* physical address of CORB/RIRB buffer */
+ /* for RIRB */
+ unsigned short rp, wp; /* read/write pointers */
+ int cmds; /* number of pending requests */
+ u32 res; /* last read value */
+};
+
+struct azx {
+ struct snd_card *card;
+ struct pci_dev *pci;
+ int dev_index;
+
+ /* chip type specific */
+ int driver_type;
+ int playback_streams;
+ int playback_index_offset;
+ int capture_streams;
+ int capture_index_offset;
+ int num_streams;
+
+ /* pci resources */
+ unsigned long addr;
+ void __iomem *remap_addr;
+ int irq;
+
+ /* locks */
+ spinlock_t reg_lock;
+ struct mutex open_mutex;
+
+ /* streams (x num_streams) */
+ struct azx_dev *azx_dev;
+
+ /* PCM */
+ struct snd_pcm *pcm[AZX_MAX_PCMS];
+
+ /* HD codec */
+ unsigned short codec_mask;
+ struct hda_bus *bus;
+
+ /* CORB/RIRB */
+ struct azx_rb corb;
+ struct azx_rb rirb;
+
+ /* CORB/RIRB and position buffers */
+ struct snd_dma_buffer rb;
+ struct snd_dma_buffer posbuf;
+
+ /* flags */
+ int position_fix;
+ unsigned int running :1;
+ unsigned int initialized :1;
+ unsigned int single_cmd :1;
+ unsigned int polling_mode :1;
+ unsigned int msi :1;
+ unsigned int irq_pending_warned :1;
+ unsigned int via_dmapos_patch :1; /* enable DMA-position fix for VIA */
+
+ /* for debugging */
+ unsigned int last_cmd; /* last issued command (to sync) */
+
+ /* for pending irqs */
+ struct work_struct irq_pending_work;
+
+ /* reboot notifier (for mysterious hangup problem at power-down) */
+ struct notifier_block reboot_notifier;
+};
+
+/* driver types */
+enum {
+ AZX_DRIVER_ICH,
+ AZX_DRIVER_SCH,
+ AZX_DRIVER_ATI,
+ AZX_DRIVER_ATIHDMI,
+ AZX_DRIVER_VIA,
+ AZX_DRIVER_SIS,
+ AZX_DRIVER_ULI,
+ AZX_DRIVER_NVIDIA,
+ AZX_DRIVER_TERA,
+ AZX_NUM_DRIVERS, /* keep this as last entry */
+};
+
+static char *driver_short_names[] __devinitdata = {
+ [AZX_DRIVER_ICH] = "HDA Intel",
+ [AZX_DRIVER_SCH] = "HDA Intel MID",
+ [AZX_DRIVER_ATI] = "HDA ATI SB",
+ [AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI",
+ [AZX_DRIVER_VIA] = "HDA VIA VT82xx",
+ [AZX_DRIVER_SIS] = "HDA SIS966",
+ [AZX_DRIVER_ULI] = "HDA ULI M5461",
+ [AZX_DRIVER_NVIDIA] = "HDA NVidia",
+ [AZX_DRIVER_TERA] = "HDA Teradici",
+};
+
+/*
+ * macros for easy use
+ */
+#define azx_writel(chip,reg,value) \
+ writel(value, (chip)->remap_addr + ICH6_REG_##reg)
+#define azx_readl(chip,reg) \
+ readl((chip)->remap_addr + ICH6_REG_##reg)
+#define azx_writew(chip,reg,value) \
+ writew(value, (chip)->remap_addr + ICH6_REG_##reg)
+#define azx_readw(chip,reg) \
+ readw((chip)->remap_addr + ICH6_REG_##reg)
+#define azx_writeb(chip,reg,value) \
+ writeb(value, (chip)->remap_addr + ICH6_REG_##reg)
+#define azx_readb(chip,reg) \
+ readb((chip)->remap_addr + ICH6_REG_##reg)
+
+#define azx_sd_writel(dev,reg,value) \
+ writel(value, (dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_readl(dev,reg) \
+ readl((dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_writew(dev,reg,value) \
+ writew(value, (dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_readw(dev,reg) \
+ readw((dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_writeb(dev,reg,value) \
+ writeb(value, (dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_readb(dev,reg) \
+ readb((dev)->sd_addr + ICH6_REG_##reg)
+
+/* for pcm support */
+#define get_azx_dev(substream) (substream->runtime->private_data)
+
+static int azx_acquire_irq(struct azx *chip, int do_disconnect);
+
+/*
+ * Interface for HD codec
+ */
+
+/*
+ * CORB / RIRB interface
+ */
+static int azx_alloc_cmd_io(struct azx *chip)
+{
+ int err;
+
+ /* single page (at least 4096 bytes) must suffice for both ringbuffes */
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ PAGE_SIZE, &chip->rb);
+ if (err < 0) {
+ snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n");
+ return err;
+ }
+ return 0;
+}
+
+static void azx_init_cmd_io(struct azx *chip)
+{
+ /* CORB set up */
+ chip->corb.addr = chip->rb.addr;
+ chip->corb.buf = (u32 *)chip->rb.area;
+ azx_writel(chip, CORBLBASE, (u32)chip->corb.addr);
+ azx_writel(chip, CORBUBASE, upper_32_bits(chip->corb.addr));
+
+ /* set the corb size to 256 entries (ULI requires explicitly) */
+ azx_writeb(chip, CORBSIZE, 0x02);
+ /* set the corb write pointer to 0 */
+ azx_writew(chip, CORBWP, 0);
+ /* reset the corb hw read pointer */
+ azx_writew(chip, CORBRP, ICH6_RBRWP_CLR);
+ /* enable corb dma */
+ azx_writeb(chip, CORBCTL, ICH6_RBCTL_DMA_EN);
+
+ /* RIRB set up */
+ chip->rirb.addr = chip->rb.addr + 2048;
+ chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
+ azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr);
+ azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr));
+
+ /* set the rirb size to 256 entries (ULI requires explicitly) */
+ azx_writeb(chip, RIRBSIZE, 0x02);
+ /* reset the rirb hw write pointer */
+ azx_writew(chip, RIRBWP, ICH6_RBRWP_CLR);
+ /* set N=1, get RIRB response interrupt for new entry */
+ azx_writew(chip, RINTCNT, 1);
+ /* enable rirb dma and response irq */
+ azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN);
+ chip->rirb.rp = chip->rirb.cmds = 0;
+}
+
+static void azx_free_cmd_io(struct azx *chip)
+{
+ /* disable ringbuffer DMAs */
+ azx_writeb(chip, RIRBCTL, 0);
+ azx_writeb(chip, CORBCTL, 0);
+}
+
+/* send a command */
+static int azx_corb_send_cmd(struct hda_codec *codec, u32 val)
+{
+ struct azx *chip = codec->bus->private_data;
+ unsigned int wp;
+
+ /* add command to corb */
+ wp = azx_readb(chip, CORBWP);
+ wp++;
+ wp %= ICH6_MAX_CORB_ENTRIES;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->rirb.cmds++;
+ chip->corb.buf[wp] = cpu_to_le32(val);
+ azx_writel(chip, CORBWP, wp);
+ spin_unlock_irq(&chip->reg_lock);
+
+ return 0;
+}
+
+#define ICH6_RIRB_EX_UNSOL_EV (1<<4)
+
+/* retrieve RIRB entry - called from interrupt handler */
+static void azx_update_rirb(struct azx *chip)
+{
+ unsigned int rp, wp;
+ u32 res, res_ex;
+
+ wp = azx_readb(chip, RIRBWP);
+ if (wp == chip->rirb.wp)
+ return;
+ chip->rirb.wp = wp;
+
+ while (chip->rirb.rp != wp) {
+ chip->rirb.rp++;
+ chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES;
+
+ rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */
+ res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]);
+ res = le32_to_cpu(chip->rirb.buf[rp]);
+ if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
+ snd_hda_queue_unsol_event(chip->bus, res, res_ex);
+ else if (chip->rirb.cmds) {
+ chip->rirb.res = res;
+ smp_wmb();
+ chip->rirb.cmds--;
+ }
+ }
+}
+
+/* receive a response */
+static unsigned int azx_rirb_get_response(struct hda_codec *codec)
+{
+ struct azx *chip = codec->bus->private_data;
+ unsigned long timeout;
+
+ again:
+ timeout = jiffies + msecs_to_jiffies(1000);
+ for (;;) {
+ if (chip->polling_mode) {
+ spin_lock_irq(&chip->reg_lock);
+ azx_update_rirb(chip);
+ spin_unlock_irq(&chip->reg_lock);
+ }
+ if (!chip->rirb.cmds) {
+ smp_rmb();
+ return chip->rirb.res; /* the last value */
+ }
+ if (time_after(jiffies, timeout))
+ break;
+ if (codec->bus->needs_damn_long_delay)
+ msleep(2); /* temporary workaround */
+ else {
+ udelay(10);
+ cond_resched();
+ }
+ }
+
+ if (chip->msi) {
+ snd_printk(KERN_WARNING "hda_intel: No response from codec, "
+ "disabling MSI: last cmd=0x%08x\n", chip->last_cmd);
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ pci_disable_msi(chip->pci);
+ chip->msi = 0;
+ if (azx_acquire_irq(chip, 1) < 0)
+ return -1;
+ goto again;
+ }
+
+ if (!chip->polling_mode) {
+ snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, "
+ "switching to polling mode: last cmd=0x%08x\n",
+ chip->last_cmd);
+ chip->polling_mode = 1;
+ goto again;
+ }
+
+ snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
+ "switching to single_cmd mode: last cmd=0x%08x\n",
+ chip->last_cmd);
+ chip->rirb.rp = azx_readb(chip, RIRBWP);
+ chip->rirb.cmds = 0;
+ /* switch to single_cmd mode */
+ chip->single_cmd = 1;
+ azx_free_cmd_io(chip);
+ return -1;
+}
+
+/*
+ * Use the single immediate command instead of CORB/RIRB for simplicity
+ *
+ * Note: according to Intel, this is not preferred use. The command was
+ * intended for the BIOS only, and may get confused with unsolicited
+ * responses. So, we shouldn't use it for normal operation from the
+ * driver.
+ * I left the codes, however, for debugging/testing purposes.
+ */
+
+/* send a command */
+static int azx_single_send_cmd(struct hda_codec *codec, u32 val)
+{
+ struct azx *chip = codec->bus->private_data;
+ int timeout = 50;
+
+ while (timeout--) {
+ /* check ICB busy bit */
+ if (!((azx_readw(chip, IRS) & ICH6_IRS_BUSY))) {
+ /* Clear IRV valid bit */
+ azx_writew(chip, IRS, azx_readw(chip, IRS) |
+ ICH6_IRS_VALID);
+ azx_writel(chip, IC, val);
+ azx_writew(chip, IRS, azx_readw(chip, IRS) |
+ ICH6_IRS_BUSY);
+ return 0;
+ }
+ udelay(1);
+ }
+ if (printk_ratelimit())
+ snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n",
+ azx_readw(chip, IRS), val);
+ return -EIO;
+}
+
+/* receive a response */
+static unsigned int azx_single_get_response(struct hda_codec *codec)
+{
+ struct azx *chip = codec->bus->private_data;
+ int timeout = 50;
+
+ while (timeout--) {
+ /* check IRV busy bit */
+ if (azx_readw(chip, IRS) & ICH6_IRS_VALID)
+ return azx_readl(chip, IR);
+ udelay(1);
+ }
+ if (printk_ratelimit())
+ snd_printd(SFX "get_response timeout: IRS=0x%x\n",
+ azx_readw(chip, IRS));
+ return (unsigned int)-1;
+}
+
+/*
+ * The below are the main callbacks from hda_codec.
+ *
+ * They are just the skeleton to call sub-callbacks according to the
+ * current setting of chip->single_cmd.
+ */
+
+/* send a command */
+static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid,
+ int direct, unsigned int verb,
+ unsigned int para)
+{
+ struct azx *chip = codec->bus->private_data;
+ u32 val;
+
+ val = (u32)(codec->addr & 0x0f) << 28;
+ val |= (u32)direct << 27;
+ val |= (u32)nid << 20;
+ val |= verb << 8;
+ val |= para;
+ chip->last_cmd = val;
+
+ if (chip->single_cmd)
+ return azx_single_send_cmd(codec, val);
+ else
+ return azx_corb_send_cmd(codec, val);
+}
+
+/* get a response */
+static unsigned int azx_get_response(struct hda_codec *codec)
+{
+ struct azx *chip = codec->bus->private_data;
+ if (chip->single_cmd)
+ return azx_single_get_response(codec);
+ else
+ return azx_rirb_get_response(codec);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void azx_power_notify(struct hda_codec *codec);
+#endif
+
+/* reset codec link */
+static int azx_reset(struct azx *chip)
+{
+ int count;
+
+ /* clear STATESTS */
+ azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+
+ /* reset controller */
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET);
+
+ count = 50;
+ while (azx_readb(chip, GCTL) && --count)
+ msleep(1);
+
+ /* delay for >= 100us for codec PLL to settle per spec
+ * Rev 0.9 section 5.5.1
+ */
+ msleep(1);
+
+ /* Bring controller out of reset */
+ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET);
+
+ count = 50;
+ while (!azx_readb(chip, GCTL) && --count)
+ msleep(1);
+
+ /* Brent Chartrand said to wait >= 540us for codecs to initialize */
+ msleep(1);
+
+ /* check to see if controller is ready */
+ if (!azx_readb(chip, GCTL)) {
+ snd_printd("azx_reset: controller not ready!\n");
+ return -EBUSY;
+ }
+
+ /* Accept unsolicited responses */
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN);
+
+ /* detect codecs */
+ if (!chip->codec_mask) {
+ chip->codec_mask = azx_readw(chip, STATESTS);
+ snd_printdd("codec_mask = 0x%x\n", chip->codec_mask);
+ }
+
+ return 0;
+}
+
+
+/*
+ * Lowlevel interface
+ */
+
+/* enable interrupts */
+static void azx_int_enable(struct azx *chip)
+{
+ /* enable controller CIE and GIE */
+ azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) |
+ ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN);
+}
+
+/* disable interrupts */
+static void azx_int_disable(struct azx *chip)
+{
+ int i;
+
+ /* disable interrupts in stream descriptor */
+ for (i = 0; i < chip->num_streams; i++) {
+ struct azx_dev *azx_dev = &chip->azx_dev[i];
+ azx_sd_writeb(azx_dev, SD_CTL,
+ azx_sd_readb(azx_dev, SD_CTL) & ~SD_INT_MASK);
+ }
+
+ /* disable SIE for all streams */
+ azx_writeb(chip, INTCTL, 0);
+
+ /* disable controller CIE and GIE */
+ azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) &
+ ~(ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN));
+}
+
+/* clear interrupts */
+static void azx_int_clear(struct azx *chip)
+{
+ int i;
+
+ /* clear stream status */
+ for (i = 0; i < chip->num_streams; i++) {
+ struct azx_dev *azx_dev = &chip->azx_dev[i];
+ azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
+ }
+
+ /* clear STATESTS */
+ azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+
+ /* clear rirb status */
+ azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
+
+ /* clear int status */
+ azx_writel(chip, INTSTS, ICH6_INT_CTRL_EN | ICH6_INT_ALL_STREAM);
+}
+
+/* start a stream */
+static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev)
+{
+ /*
+ * Before stream start, initialize parameter
+ */
+ azx_dev->insufficient = 1;
+
+ /* enable SIE */
+ azx_writeb(chip, INTCTL,
+ azx_readb(chip, INTCTL) | (1 << azx_dev->index));
+ /* set DMA start and interrupt mask */
+ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) |
+ SD_CTL_DMA_START | SD_INT_MASK);
+}
+
+/* stop a stream */
+static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev)
+{
+ /* stop DMA */
+ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
+ ~(SD_CTL_DMA_START | SD_INT_MASK));
+ azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
+ /* disable SIE */
+ azx_writeb(chip, INTCTL,
+ azx_readb(chip, INTCTL) & ~(1 << azx_dev->index));
+}
+
+
+/*
+ * reset and start the controller registers
+ */
+static void azx_init_chip(struct azx *chip)
+{
+ if (chip->initialized)
+ return;
+
+ /* reset controller */
+ azx_reset(chip);
+
+ /* initialize interrupts */
+ azx_int_clear(chip);
+ azx_int_enable(chip);
+
+ /* initialize the codec command I/O */
+ if (!chip->single_cmd)
+ azx_init_cmd_io(chip);
+
+ /* program the position buffer */
+ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
+ azx_writel(chip, DPUBASE, upper_32_bits(chip->posbuf.addr));
+
+ chip->initialized = 1;
+}
+
+/*
+ * initialize the PCI registers
+ */
+/* update bits in a PCI register byte */
+static void update_pci_byte(struct pci_dev *pci, unsigned int reg,
+ unsigned char mask, unsigned char val)
+{
+ unsigned char data;
+
+ pci_read_config_byte(pci, reg, &data);
+ data &= ~mask;
+ data |= (val & mask);
+ pci_write_config_byte(pci, reg, data);
+}
+
+static void azx_init_pci(struct azx *chip)
+{
+ unsigned short snoop;
+
+ /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
+ * TCSEL == Traffic Class Select Register, which sets PCI express QOS
+ * Ensuring these bits are 0 clears playback static on some HD Audio
+ * codecs
+ */
+ update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0);
+
+ switch (chip->driver_type) {
+ case AZX_DRIVER_ATI:
+ /* For ATI SB450 azalia HD audio, we need to enable snoop */
+ update_pci_byte(chip->pci,
+ ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
+ 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+ break;
+ case AZX_DRIVER_NVIDIA:
+ /* For NVIDIA HDA, enable snoop */
+ update_pci_byte(chip->pci,
+ NVIDIA_HDA_TRANSREG_ADDR,
+ 0x0f, NVIDIA_HDA_ENABLE_COHBITS);
+ update_pci_byte(chip->pci,
+ NVIDIA_HDA_ISTRM_COH,
+ 0x01, NVIDIA_HDA_ENABLE_COHBIT);
+ update_pci_byte(chip->pci,
+ NVIDIA_HDA_OSTRM_COH,
+ 0x01, NVIDIA_HDA_ENABLE_COHBIT);
+ break;
+ case AZX_DRIVER_SCH:
+ pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
+ if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) {
+ pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, \
+ snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP));
+ pci_read_config_word(chip->pci,
+ INTEL_SCH_HDA_DEVC, &snoop);
+ snd_printdd("HDA snoop disabled, enabling ... %s\n",\
+ (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) \
+ ? "Failed" : "OK");
+ }
+ break;
+
+ }
+}
+
+
+static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev);
+
+/*
+ * interrupt handler
+ */
+static irqreturn_t azx_interrupt(int irq, void *dev_id)
+{
+ struct azx *chip = dev_id;
+ struct azx_dev *azx_dev;
+ u32 status;
+ int i;
+
+ spin_lock(&chip->reg_lock);
+
+ status = azx_readl(chip, INTSTS);
+ if (status == 0) {
+ spin_unlock(&chip->reg_lock);
+ return IRQ_NONE;
+ }
+
+ for (i = 0; i < chip->num_streams; i++) {
+ azx_dev = &chip->azx_dev[i];
+ if (status & azx_dev->sd_int_sta_mask) {
+ azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
+ if (!azx_dev->substream || !azx_dev->running)
+ continue;
+ /* ignore the first dummy IRQ (due to pos_adj) */
+ if (azx_dev->irq_ignore) {
+ azx_dev->irq_ignore = 0;
+ continue;
+ }
+ /* check whether this IRQ is really acceptable */
+ if (azx_position_ok(chip, azx_dev)) {
+ azx_dev->irq_pending = 0;
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(azx_dev->substream);
+ spin_lock(&chip->reg_lock);
+ } else {
+ /* bogus IRQ, process it later */
+ azx_dev->irq_pending = 1;
+ schedule_work(&chip->irq_pending_work);
+ }
+ }
+ }
+
+ /* clear rirb int */
+ status = azx_readb(chip, RIRBSTS);
+ if (status & RIRB_INT_MASK) {
+ if (!chip->single_cmd && (status & RIRB_INT_RESPONSE))
+ azx_update_rirb(chip);
+ azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
+ }
+
+#if 0
+ /* clear state status int */
+ if (azx_readb(chip, STATESTS) & 0x04)
+ azx_writeb(chip, STATESTS, 0x04);
+#endif
+ spin_unlock(&chip->reg_lock);
+
+ return IRQ_HANDLED;
+}
+
+
+/*
+ * set up a BDL entry
+ */
+static int setup_bdle(struct snd_pcm_substream *substream,
+ struct azx_dev *azx_dev, u32 **bdlp,
+ int ofs, int size, int with_ioc)
+{
+ u32 *bdl = *bdlp;
+
+ while (size > 0) {
+ dma_addr_t addr;
+ int chunk;
+
+ if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES)
+ return -EINVAL;
+
+ addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+ /* program the address field of the BDL entry */
+ bdl[0] = cpu_to_le32((u32)addr);
+ bdl[1] = cpu_to_le32(upper_32_bits(addr));
+ /* program the size field of the BDL entry */
+ chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size);
+ bdl[2] = cpu_to_le32(chunk);
+ /* program the IOC to enable interrupt
+ * only when the whole fragment is processed
+ */
+ size -= chunk;
+ bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01);
+ bdl += 4;
+ azx_dev->frags++;
+ ofs += chunk;
+ }
+ *bdlp = bdl;
+ return ofs;
+}
+
+/*
+ * set up BDL entries
+ */
+static int azx_setup_periods(struct azx *chip,
+ struct snd_pcm_substream *substream,
+ struct azx_dev *azx_dev)
+{
+ u32 *bdl;
+ int i, ofs, periods, period_bytes;
+ int pos_adj;
+
+ /* reset BDL address */
+ azx_sd_writel(azx_dev, SD_BDLPL, 0);
+ azx_sd_writel(azx_dev, SD_BDLPU, 0);
+
+ period_bytes = snd_pcm_lib_period_bytes(substream);
+ azx_dev->period_bytes = period_bytes;
+ periods = azx_dev->bufsize / period_bytes;
+
+ /* program the initial BDL entries */
+ bdl = (u32 *)azx_dev->bdl.area;
+ ofs = 0;
+ azx_dev->frags = 0;
+ azx_dev->irq_ignore = 0;
+ pos_adj = bdl_pos_adj[chip->dev_index];
+ if (pos_adj > 0) {
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int pos_align = pos_adj;
+ pos_adj = (pos_adj * runtime->rate + 47999) / 48000;
+ if (!pos_adj)
+ pos_adj = pos_align;
+ else
+ pos_adj = ((pos_adj + pos_align - 1) / pos_align) *
+ pos_align;
+ pos_adj = frames_to_bytes(runtime, pos_adj);
+ if (pos_adj >= period_bytes) {
+ snd_printk(KERN_WARNING "Too big adjustment %d\n",
+ bdl_pos_adj[chip->dev_index]);
+ pos_adj = 0;
+ } else {
+ ofs = setup_bdle(substream, azx_dev,
+ &bdl, ofs, pos_adj, 1);
+ if (ofs < 0)
+ goto error;
+ azx_dev->irq_ignore = 1;
+ }
+ } else
+ pos_adj = 0;
+ for (i = 0; i < periods; i++) {
+ if (i == periods - 1 && pos_adj)
+ ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+ period_bytes - pos_adj, 0);
+ else
+ ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+ period_bytes, 1);
+ if (ofs < 0)
+ goto error;
+ }
+ return 0;
+
+ error:
+ snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n",
+ azx_dev->bufsize, period_bytes);
+ /* reset */
+ azx_sd_writel(azx_dev, SD_BDLPL, 0);
+ azx_sd_writel(azx_dev, SD_BDLPU, 0);
+ return -EINVAL;
+}
+
+/*
+ * set up the SD for streaming
+ */
+static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
+{
+ unsigned char val;
+ int timeout;
+
+ /* make sure the run bit is zero for SD */
+ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
+ ~SD_CTL_DMA_START);
+ /* reset stream */
+ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) |
+ SD_CTL_STREAM_RESET);
+ udelay(3);
+ timeout = 300;
+ while (!((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) &&
+ --timeout)
+ ;
+ val &= ~SD_CTL_STREAM_RESET;
+ azx_sd_writeb(azx_dev, SD_CTL, val);
+ udelay(3);
+
+ timeout = 300;
+ /* waiting for hardware to report that the stream is out of reset */
+ while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) &&
+ --timeout)
+ ;
+
+ /* program the stream_tag */
+ azx_sd_writel(azx_dev, SD_CTL,
+ (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)|
+ (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT));
+
+ /* program the length of samples in cyclic buffer */
+ azx_sd_writel(azx_dev, SD_CBL, azx_dev->bufsize);
+
+ /* program the stream format */
+ /* this value needs to be the same as the one programmed */
+ azx_sd_writew(azx_dev, SD_FORMAT, azx_dev->format_val);
+
+ /* program the stream LVI (last valid index) of the BDL */
+ azx_sd_writew(azx_dev, SD_LVI, azx_dev->frags - 1);
+
+ /* program the BDL address */
+ /* lower BDL address */
+ azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr);
+ /* upper BDL address */
+ azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr));
+
+ /* enable the position buffer */
+ if (chip->position_fix == POS_FIX_POSBUF ||
+ chip->position_fix == POS_FIX_AUTO ||
+ chip->via_dmapos_patch) {
+ if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
+ azx_writel(chip, DPLBASE,
+ (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
+ }
+
+ /* set the interrupt enable bits in the descriptor control register */
+ azx_sd_writel(azx_dev, SD_CTL,
+ azx_sd_readl(azx_dev, SD_CTL) | SD_INT_MASK);
+
+ return 0;
+}
+
+
+/*
+ * Codec initialization
+ */
+
+/* number of codec slots for each chipset: 0 = default slots (i.e. 4) */
+static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
+ [AZX_DRIVER_TERA] = 1,
+};
+
+/* number of slots to probe as default
+ * this can be different from azx_max_codecs[] -- e.g. some boards
+ * report wrongly the non-existing 4th slot availability
+ */
+static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = {
+ [AZX_DRIVER_ICH] = 3,
+ [AZX_DRIVER_ATI] = 3,
+};
+
+static int __devinit azx_codec_create(struct azx *chip, const char *model,
+ unsigned int codec_probe_mask)
+{
+ struct hda_bus_template bus_temp;
+ int c, codecs, audio_codecs, err;
+ int def_slots, max_slots;
+
+ memset(&bus_temp, 0, sizeof(bus_temp));
+ bus_temp.private_data = chip;
+ bus_temp.modelname = model;
+ bus_temp.pci = chip->pci;
+ bus_temp.ops.command = azx_send_cmd;
+ bus_temp.ops.get_response = azx_get_response;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ bus_temp.ops.pm_notify = azx_power_notify;
+#endif
+
+ err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus);
+ if (err < 0)
+ return err;
+
+ if (chip->driver_type == AZX_DRIVER_NVIDIA)
+ chip->bus->needs_damn_long_delay = 1;
+
+ codecs = audio_codecs = 0;
+ max_slots = azx_max_codecs[chip->driver_type];
+ if (!max_slots)
+ max_slots = AZX_MAX_CODECS;
+ def_slots = azx_default_codecs[chip->driver_type];
+ if (!def_slots)
+ def_slots = max_slots;
+ for (c = 0; c < def_slots; c++) {
+ if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
+ struct hda_codec *codec;
+ err = snd_hda_codec_new(chip->bus, c, &codec);
+ if (err < 0)
+ continue;
+ codecs++;
+ if (codec->afg)
+ audio_codecs++;
+ }
+ }
+ if (!audio_codecs) {
+ /* probe additional slots if no codec is found */
+ for (; c < max_slots; c++) {
+ if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
+ err = snd_hda_codec_new(chip->bus, c, NULL);
+ if (err < 0)
+ continue;
+ codecs++;
+ }
+ }
+ }
+ if (!codecs) {
+ snd_printk(KERN_ERR SFX "no codecs initialized\n");
+ return -ENXIO;
+ }
+
+ return 0;
+}
+
+
+/*
+ * PCM support
+ */
+
+/* assign a stream for the PCM */
+static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
+{
+ int dev, i, nums;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dev = chip->playback_index_offset;
+ nums = chip->playback_streams;
+ } else {
+ dev = chip->capture_index_offset;
+ nums = chip->capture_streams;
+ }
+ for (i = 0; i < nums; i++, dev++)
+ if (!chip->azx_dev[dev].opened) {
+ chip->azx_dev[dev].opened = 1;
+ return &chip->azx_dev[dev];
+ }
+ return NULL;
+}
+
+/* release the assigned stream */
+static inline void azx_release_device(struct azx_dev *azx_dev)
+{
+ azx_dev->opened = 0;
+}
+
+static struct snd_pcm_hardware azx_pcm_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ /* No full-resume yet implemented */
+ /* SNDRV_PCM_INFO_RESUME |*/
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = AZX_MAX_BUF_SIZE,
+ .period_bytes_min = 128,
+ .period_bytes_max = AZX_MAX_BUF_SIZE / 2,
+ .periods_min = 2,
+ .periods_max = AZX_MAX_FRAG,
+ .fifo_size = 0,
+};
+
+struct azx_pcm {
+ struct azx *chip;
+ struct hda_codec *codec;
+ struct hda_pcm_stream *hinfo[2];
+};
+
+static int azx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+ struct azx *chip = apcm->chip;
+ struct azx_dev *azx_dev;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long flags;
+ int err;
+
+ mutex_lock(&chip->open_mutex);
+ azx_dev = azx_assign_device(chip, substream->stream);
+ if (azx_dev == NULL) {
+ mutex_unlock(&chip->open_mutex);
+ return -EBUSY;
+ }
+ runtime->hw = azx_pcm_hw;
+ runtime->hw.channels_min = hinfo->channels_min;
+ runtime->hw.channels_max = hinfo->channels_max;
+ runtime->hw.formats = hinfo->formats;
+ runtime->hw.rates = hinfo->rates;
+ snd_pcm_limit_hw_rates(runtime);
+ snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ 128);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ 128);
+ snd_hda_power_up(apcm->codec);
+ err = hinfo->ops.open(hinfo, apcm->codec, substream);
+ if (err < 0) {
+ azx_release_device(azx_dev);
+ snd_hda_power_down(apcm->codec);
+ mutex_unlock(&chip->open_mutex);
+ return err;
+ }
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ azx_dev->substream = substream;
+ azx_dev->running = 0;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ runtime->private_data = azx_dev;
+ snd_pcm_set_sync(substream);
+ mutex_unlock(&chip->open_mutex);
+ return 0;
+}
+
+static int azx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+ struct azx *chip = apcm->chip;
+ struct azx_dev *azx_dev = get_azx_dev(substream);
+ unsigned long flags;
+
+ mutex_lock(&chip->open_mutex);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ azx_dev->substream = NULL;
+ azx_dev->running = 0;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ azx_release_device(azx_dev);
+ hinfo->ops.close(hinfo, apcm->codec, substream);
+ snd_hda_power_down(apcm->codec);
+ mutex_unlock(&chip->open_mutex);
+ return 0;
+}
+
+static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct azx_dev *azx_dev = get_azx_dev(substream);
+ struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+
+ /* reset BDL address */
+ azx_sd_writel(azx_dev, SD_BDLPL, 0);
+ azx_sd_writel(azx_dev, SD_BDLPU, 0);
+ azx_sd_writel(azx_dev, SD_CTL, 0);
+
+ hinfo->ops.cleanup(hinfo, apcm->codec, substream);
+
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int azx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct azx *chip = apcm->chip;
+ struct azx_dev *azx_dev = get_azx_dev(substream);
+ struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream);
+ azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate,
+ runtime->channels,
+ runtime->format,
+ hinfo->maxbps);
+ if (!azx_dev->format_val) {
+ snd_printk(KERN_ERR SFX
+ "invalid format_val, rate=%d, ch=%d, format=%d\n",
+ runtime->rate, runtime->channels, runtime->format);
+ return -EINVAL;
+ }
+
+ snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n",
+ azx_dev->bufsize, azx_dev->format_val);
+ if (azx_setup_periods(chip, substream, azx_dev) < 0)
+ return -EINVAL;
+ azx_setup_controller(chip, azx_dev);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1;
+ else
+ azx_dev->fifo_size = 0;
+
+ return hinfo->ops.prepare(hinfo, apcm->codec, azx_dev->stream_tag,
+ azx_dev->format_val, substream);
+}
+
+static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct azx *chip = apcm->chip;
+ struct azx_dev *azx_dev;
+ struct snd_pcm_substream *s;
+ int start, nsync = 0, sbits = 0;
+ int nwait, timeout;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_START:
+ start = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_STOP:
+ start = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ sbits |= 1 << azx_dev->index;
+ nsync++;
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ spin_lock(&chip->reg_lock);
+ if (nsync > 1) {
+ /* first, set SYNC bits of corresponding streams */
+ azx_writel(chip, SYNC, azx_readl(chip, SYNC) | sbits);
+ }
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ if (start)
+ azx_stream_start(chip, azx_dev);
+ else
+ azx_stream_stop(chip, azx_dev);
+ azx_dev->running = start;
+ }
+ spin_unlock(&chip->reg_lock);
+ if (start) {
+ if (nsync == 1)
+ return 0;
+ /* wait until all FIFOs get ready */
+ for (timeout = 5000; timeout; timeout--) {
+ nwait = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ if (!(azx_sd_readb(azx_dev, SD_STS) &
+ SD_STS_FIFO_READY))
+ nwait++;
+ }
+ if (!nwait)
+ break;
+ cpu_relax();
+ }
+ } else {
+ /* wait until all RUN bits are cleared */
+ for (timeout = 5000; timeout; timeout--) {
+ nwait = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s->pcm->card != substream->pcm->card)
+ continue;
+ azx_dev = get_azx_dev(s);
+ if (azx_sd_readb(azx_dev, SD_CTL) &
+ SD_CTL_DMA_START)
+ nwait++;
+ }
+ if (!nwait)
+ break;
+ cpu_relax();
+ }
+ }
+ if (nsync > 1) {
+ spin_lock(&chip->reg_lock);
+ /* reset SYNC bits */
+ azx_writel(chip, SYNC, azx_readl(chip, SYNC) & ~sbits);
+ spin_unlock(&chip->reg_lock);
+ }
+ return 0;
+}
+
+/* get the current DMA position with correction on VIA chips */
+static unsigned int azx_via_get_position(struct azx *chip,
+ struct azx_dev *azx_dev)
+{
+ unsigned int link_pos, mini_pos, bound_pos;
+ unsigned int mod_link_pos, mod_dma_pos, mod_mini_pos;
+ unsigned int fifo_size;
+
+ link_pos = azx_sd_readl(azx_dev, SD_LPIB);
+ if (azx_dev->index >= 4) {
+ /* Playback, no problem using link position */
+ return link_pos;
+ }
+
+ /* Capture */
+ /* For new chipset,
+ * use mod to get the DMA position just like old chipset
+ */
+ mod_dma_pos = le32_to_cpu(*azx_dev->posbuf);
+ mod_dma_pos %= azx_dev->period_bytes;
+
+ /* azx_dev->fifo_size can't get FIFO size of in stream.
+ * Get from base address + offset.
+ */
+ fifo_size = readw(chip->remap_addr + VIA_IN_STREAM0_FIFO_SIZE_OFFSET);
+
+ if (azx_dev->insufficient) {
+ /* Link position never gather than FIFO size */
+ if (link_pos <= fifo_size)
+ return 0;
+
+ azx_dev->insufficient = 0;
+ }
+
+ if (link_pos <= fifo_size)
+ mini_pos = azx_dev->bufsize + link_pos - fifo_size;
+ else
+ mini_pos = link_pos - fifo_size;
+
+ /* Find nearest previous boudary */
+ mod_mini_pos = mini_pos % azx_dev->period_bytes;
+ mod_link_pos = link_pos % azx_dev->period_bytes;
+ if (mod_link_pos >= fifo_size)
+ bound_pos = link_pos - mod_link_pos;
+ else if (mod_dma_pos >= mod_mini_pos)
+ bound_pos = mini_pos - mod_mini_pos;
+ else {
+ bound_pos = mini_pos - mod_mini_pos + azx_dev->period_bytes;
+ if (bound_pos >= azx_dev->bufsize)
+ bound_pos = 0;
+ }
+
+ /* Calculate real DMA position we want */
+ return bound_pos + mod_dma_pos;
+}
+
+static unsigned int azx_get_position(struct azx *chip,
+ struct azx_dev *azx_dev)
+{
+ unsigned int pos;
+
+ if (chip->via_dmapos_patch)
+ pos = azx_via_get_position(chip, azx_dev);
+ else if (chip->position_fix == POS_FIX_POSBUF ||
+ chip->position_fix == POS_FIX_AUTO) {
+ /* use the position buffer */
+ pos = le32_to_cpu(*azx_dev->posbuf);
+ } else {
+ /* read LPIB */
+ pos = azx_sd_readl(azx_dev, SD_LPIB);
+ }
+ if (pos >= azx_dev->bufsize)
+ pos = 0;
+ return pos;
+}
+
+static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ struct azx *chip = apcm->chip;
+ struct azx_dev *azx_dev = get_azx_dev(substream);
+ return bytes_to_frames(substream->runtime,
+ azx_get_position(chip, azx_dev));
+}
+
+/*
+ * Check whether the current DMA position is acceptable for updating
+ * periods. Returns non-zero if it's OK.
+ *
+ * Many HD-audio controllers appear pretty inaccurate about
+ * the update-IRQ timing. The IRQ is issued before actually the
+ * data is processed. So, we need to process it afterwords in a
+ * workqueue.
+ */
+static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
+{
+ unsigned int pos;
+
+ pos = azx_get_position(chip, azx_dev);
+ if (chip->position_fix == POS_FIX_AUTO) {
+ if (!pos) {
+ printk(KERN_WARNING
+ "hda-intel: Invalid position buffer, "
+ "using LPIB read method instead.\n");
+ chip->position_fix = POS_FIX_LPIB;
+ pos = azx_get_position(chip, azx_dev);
+ } else
+ chip->position_fix = POS_FIX_POSBUF;
+ }
+
+ if (!bdl_pos_adj[chip->dev_index])
+ return 1; /* no delayed ack */
+ if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
+ return 0; /* NG - it's below the period boundary */
+ return 1; /* OK, it's fine */
+}
+
+/*
+ * The work for pending PCM period updates.
+ */
+static void azx_irq_pending_work(struct work_struct *work)
+{
+ struct azx *chip = container_of(work, struct azx, irq_pending_work);
+ int i, pending;
+
+ if (!chip->irq_pending_warned) {
+ printk(KERN_WARNING
+ "hda-intel: IRQ timing workaround is activated "
+ "for card #%d. Suggest a bigger bdl_pos_adj.\n",
+ chip->card->number);
+ chip->irq_pending_warned = 1;
+ }
+
+ for (;;) {
+ pending = 0;
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < chip->num_streams; i++) {
+ struct azx_dev *azx_dev = &chip->azx_dev[i];
+ if (!azx_dev->irq_pending ||
+ !azx_dev->substream ||
+ !azx_dev->running)
+ continue;
+ if (azx_position_ok(chip, azx_dev)) {
+ azx_dev->irq_pending = 0;
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(azx_dev->substream);
+ spin_lock(&chip->reg_lock);
+ } else
+ pending++;
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ if (!pending)
+ return;
+ cond_resched();
+ }
+}
+
+/* clear irq_pending flags and assure no on-going workq */
+static void azx_clear_irq_pending(struct azx *chip)
+{
+ int i;
+
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < chip->num_streams; i++)
+ chip->azx_dev[i].irq_pending = 0;
+ spin_unlock_irq(&chip->reg_lock);
+ flush_scheduled_work();
+}
+
+static struct snd_pcm_ops azx_pcm_ops = {
+ .open = azx_pcm_open,
+ .close = azx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = azx_pcm_hw_params,
+ .hw_free = azx_pcm_hw_free,
+ .prepare = azx_pcm_prepare,
+ .trigger = azx_pcm_trigger,
+ .pointer = azx_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static void azx_pcm_free(struct snd_pcm *pcm)
+{
+ kfree(pcm->private_data);
+}
+
+static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
+ struct hda_pcm *cpcm)
+{
+ int err;
+ struct snd_pcm *pcm;
+ struct azx_pcm *apcm;
+
+ /* if no substreams are defined for both playback and capture,
+ * it's just a placeholder. ignore it.
+ */
+ if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
+ return 0;
+
+ if (snd_BUG_ON(!cpcm->name))
+ return -EINVAL;
+
+ err = snd_pcm_new(chip->card, cpcm->name, cpcm->device,
+ cpcm->stream[0].substreams,
+ cpcm->stream[1].substreams,
+ &pcm);
+ if (err < 0)
+ return err;
+ strcpy(pcm->name, cpcm->name);
+ apcm = kmalloc(sizeof(*apcm), GFP_KERNEL);
+ if (apcm == NULL)
+ return -ENOMEM;
+ apcm->chip = chip;
+ apcm->codec = codec;
+ apcm->hinfo[0] = &cpcm->stream[0];
+ apcm->hinfo[1] = &cpcm->stream[1];
+ pcm->private_data = apcm;
+ pcm->private_free = azx_pcm_free;
+ if (cpcm->stream[0].substreams)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops);
+ if (cpcm->stream[1].substreams)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops);
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 1024 * 64, 32 * 1024 * 1024);
+ chip->pcm[cpcm->device] = pcm;
+ return 0;
+}
+
+static int __devinit azx_pcm_create(struct azx *chip)
+{
+ static const char *dev_name[HDA_PCM_NTYPES] = {
+ "Audio", "SPDIF", "HDMI", "Modem"
+ };
+ /* starting device index for each PCM type */
+ static int dev_idx[HDA_PCM_NTYPES] = {
+ [HDA_PCM_TYPE_AUDIO] = 0,
+ [HDA_PCM_TYPE_SPDIF] = 1,
+ [HDA_PCM_TYPE_HDMI] = 3,
+ [HDA_PCM_TYPE_MODEM] = 6
+ };
+ /* normal audio device indices; not linear to keep compatibility */
+ static int audio_idx[4] = { 0, 2, 4, 5 };
+ struct hda_codec *codec;
+ int c, err;
+ int num_devs[HDA_PCM_NTYPES];
+
+ err = snd_hda_build_pcms(chip->bus);
+ if (err < 0)
+ return err;
+
+ /* create audio PCMs */
+ memset(num_devs, 0, sizeof(num_devs));
+ list_for_each_entry(codec, &chip->bus->codec_list, list) {
+ for (c = 0; c < codec->num_pcms; c++) {
+ struct hda_pcm *cpcm = &codec->pcm_info[c];
+ int type = cpcm->pcm_type;
+ switch (type) {
+ case HDA_PCM_TYPE_AUDIO:
+ if (num_devs[type] >= ARRAY_SIZE(audio_idx)) {
+ snd_printk(KERN_WARNING
+ "Too many audio devices\n");
+ continue;
+ }
+ cpcm->device = audio_idx[num_devs[type]];
+ break;
+ case HDA_PCM_TYPE_SPDIF:
+ case HDA_PCM_TYPE_HDMI:
+ case HDA_PCM_TYPE_MODEM:
+ if (num_devs[type]) {
+ snd_printk(KERN_WARNING
+ "%s already defined\n",
+ dev_name[type]);
+ continue;
+ }
+ cpcm->device = dev_idx[type];
+ break;
+ default:
+ snd_printk(KERN_WARNING
+ "Invalid PCM type %d\n", type);
+ continue;
+ }
+ num_devs[type]++;
+ err = create_codec_pcm(chip, codec, cpcm);
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+/*
+ * mixer creation - all stuff is implemented in hda module
+ */
+static int __devinit azx_mixer_create(struct azx *chip)
+{
+ return snd_hda_build_controls(chip->bus);
+}
+
+
+/*
+ * initialize SD streams
+ */
+static int __devinit azx_init_stream(struct azx *chip)
+{
+ int i;
+
+ /* initialize each stream (aka device)
+ * assign the starting bdl address to each stream (device)
+ * and initialize
+ */
+ for (i = 0; i < chip->num_streams; i++) {
+ struct azx_dev *azx_dev = &chip->azx_dev[i];
+ azx_dev->posbuf = (u32 __iomem *)(chip->posbuf.area + i * 8);
+ /* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
+ azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80);
+ /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */
+ azx_dev->sd_int_sta_mask = 1 << i;
+ /* stream tag: must be non-zero and unique */
+ azx_dev->index = i;
+ azx_dev->stream_tag = i + 1;
+ }
+
+ return 0;
+}
+
+static int azx_acquire_irq(struct azx *chip, int do_disconnect)
+{
+ if (request_irq(chip->pci->irq, azx_interrupt,
+ chip->msi ? 0 : IRQF_SHARED,
+ "HDA Intel", chip)) {
+ printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
+ "disabling device\n", chip->pci->irq);
+ if (do_disconnect)
+ snd_card_disconnect(chip->card);
+ return -1;
+ }
+ chip->irq = chip->pci->irq;
+ pci_intx(chip->pci, !chip->msi);
+ return 0;
+}
+
+
+static void azx_stop_chip(struct azx *chip)
+{
+ if (!chip->initialized)
+ return;
+
+ /* disable interrupts */
+ azx_int_disable(chip);
+ azx_int_clear(chip);
+
+ /* disable CORB/RIRB */
+ azx_free_cmd_io(chip);
+
+ /* disable position buffer */
+ azx_writel(chip, DPLBASE, 0);
+ azx_writel(chip, DPUBASE, 0);
+
+ chip->initialized = 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* power-up/down the controller */
+static void azx_power_notify(struct hda_codec *codec)
+{
+ struct azx *chip = codec->bus->private_data;
+ struct hda_codec *c;
+ int power_on = 0;
+
+ list_for_each_entry(c, &codec->bus->codec_list, list) {
+ if (c->power_on) {
+ power_on = 1;
+ break;
+ }
+ }
+ if (power_on)
+ azx_init_chip(chip);
+ else if (chip->running && power_save_controller)
+ azx_stop_chip(chip);
+}
+#endif /* CONFIG_SND_HDA_POWER_SAVE */
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int azx_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct azx *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ azx_clear_irq_pending(chip);
+ for (i = 0; i < AZX_MAX_PCMS; i++)
+ snd_pcm_suspend_all(chip->pcm[i]);
+ if (chip->initialized)
+ snd_hda_suspend(chip->bus, state);
+ azx_stop_chip(chip);
+ if (chip->irq >= 0) {
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ }
+ if (chip->msi)
+ pci_disable_msi(chip->pci);
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int azx_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct azx *chip = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "hda-intel: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+ if (chip->msi)
+ if (pci_enable_msi(pci) < 0)
+ chip->msi = 0;
+ if (azx_acquire_irq(chip, 1) < 0)
+ return -EIO;
+ azx_init_pci(chip);
+
+ if (snd_hda_codecs_inuse(chip->bus))
+ azx_init_chip(chip);
+
+ snd_hda_resume(chip->bus);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+/*
+ * reboot notifier for hang-up problem at power-down
+ */
+static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf)
+{
+ struct azx *chip = container_of(nb, struct azx, reboot_notifier);
+ azx_stop_chip(chip);
+ return NOTIFY_OK;
+}
+
+static void azx_notifier_register(struct azx *chip)
+{
+ chip->reboot_notifier.notifier_call = azx_halt;
+ register_reboot_notifier(&chip->reboot_notifier);
+}
+
+static void azx_notifier_unregister(struct azx *chip)
+{
+ if (chip->reboot_notifier.notifier_call)
+ unregister_reboot_notifier(&chip->reboot_notifier);
+}
+
+/*
+ * destructor
+ */
+static int azx_free(struct azx *chip)
+{
+ int i;
+
+ azx_notifier_unregister(chip);
+
+ if (chip->initialized) {
+ azx_clear_irq_pending(chip);
+ for (i = 0; i < chip->num_streams; i++)
+ azx_stream_stop(chip, &chip->azx_dev[i]);
+ azx_stop_chip(chip);
+ }
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void*)chip);
+ if (chip->msi)
+ pci_disable_msi(chip->pci);
+ if (chip->remap_addr)
+ iounmap(chip->remap_addr);
+
+ if (chip->azx_dev) {
+ for (i = 0; i < chip->num_streams; i++)
+ if (chip->azx_dev[i].bdl.area)
+ snd_dma_free_pages(&chip->azx_dev[i].bdl);
+ }
+ if (chip->rb.area)
+ snd_dma_free_pages(&chip->rb);
+ if (chip->posbuf.area)
+ snd_dma_free_pages(&chip->posbuf);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip->azx_dev);
+ kfree(chip);
+
+ return 0;
+}
+
+static int azx_dev_free(struct snd_device *device)
+{
+ return azx_free(device->device_data);
+}
+
+/*
+ * white/black-listing for position_fix
+ */
+static struct snd_pci_quirk position_fix_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
+ {}
+};
+
+static int __devinit check_position_fix(struct azx *chip, int fix)
+{
+ const struct snd_pci_quirk *q;
+
+ switch (fix) {
+ case POS_FIX_LPIB:
+ case POS_FIX_POSBUF:
+ return fix;
+ }
+
+ /* Check VIA/ATI HD Audio Controller exist */
+ switch (chip->driver_type) {
+ case AZX_DRIVER_VIA:
+ case AZX_DRIVER_ATI:
+ chip->via_dmapos_patch = 1;
+ /* Use link position directly, avoid any transfer problem. */
+ return POS_FIX_LPIB;
+ }
+ chip->via_dmapos_patch = 0;
+
+ q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
+ if (q) {
+ printk(KERN_INFO
+ "hda_intel: position_fix set to %d "
+ "for device %04x:%04x\n",
+ q->value, q->subvendor, q->subdevice);
+ return q->value;
+ }
+ return POS_FIX_AUTO;
+}
+
+/*
+ * black-lists for probe_mask
+ */
+static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
+ /* Thinkpad often breaks the controller communication when accessing
+ * to the non-working (or non-existing) modem codec slot.
+ */
+ SND_PCI_QUIRK(0x1014, 0x05b7, "Thinkpad Z60", 0x01),
+ SND_PCI_QUIRK(0x17aa, 0x2010, "Thinkpad X/T/R60", 0x01),
+ SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X/T/R61", 0x01),
+ {}
+};
+
+static void __devinit check_probe_mask(struct azx *chip, int dev)
+{
+ const struct snd_pci_quirk *q;
+
+ if (probe_mask[dev] == -1) {
+ q = snd_pci_quirk_lookup(chip->pci, probe_mask_list);
+ if (q) {
+ printk(KERN_INFO
+ "hda_intel: probe_mask set to 0x%x "
+ "for device %04x:%04x\n",
+ q->value, q->subvendor, q->subdevice);
+ probe_mask[dev] = q->value;
+ }
+ }
+}
+
+
+/*
+ * constructor
+ */
+static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
+ int dev, int driver_type,
+ struct azx **rchip)
+{
+ struct azx *chip;
+ int i, err;
+ unsigned short gcap;
+ static struct snd_device_ops ops = {
+ .dev_free = azx_dev_free,
+ };
+
+ *rchip = NULL;
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (!chip) {
+ snd_printk(KERN_ERR SFX "cannot allocate chip\n");
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&chip->reg_lock);
+ mutex_init(&chip->open_mutex);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->driver_type = driver_type;
+ chip->msi = enable_msi;
+ chip->dev_index = dev;
+ INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
+
+ chip->position_fix = check_position_fix(chip, position_fix[dev]);
+ check_probe_mask(chip, dev);
+
+ chip->single_cmd = single_cmd;
+
+ if (bdl_pos_adj[dev] < 0) {
+ switch (chip->driver_type) {
+ case AZX_DRIVER_ICH:
+ bdl_pos_adj[dev] = 1;
+ break;
+ default:
+ bdl_pos_adj[dev] = 32;
+ break;
+ }
+ }
+
+#if BITS_PER_LONG != 64
+ /* Fix up base address on ULI M5461 */
+ if (chip->driver_type == AZX_DRIVER_ULI) {
+ u16 tmp3;
+ pci_read_config_word(pci, 0x40, &tmp3);
+ pci_write_config_word(pci, 0x40, tmp3 | 0x10);
+ pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, 0);
+ }
+#endif
+
+ err = pci_request_regions(pci, "ICH HD audio");
+ if (err < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+
+ chip->addr = pci_resource_start(pci, 0);
+ chip->remap_addr = pci_ioremap_bar(pci, 0);
+ if (chip->remap_addr == NULL) {
+ snd_printk(KERN_ERR SFX "ioremap error\n");
+ err = -ENXIO;
+ goto errout;
+ }
+
+ if (chip->msi)
+ if (pci_enable_msi(pci) < 0)
+ chip->msi = 0;
+
+ if (azx_acquire_irq(chip, 0) < 0) {
+ err = -EBUSY;
+ goto errout;
+ }
+
+ pci_set_master(pci);
+ synchronize_irq(chip->irq);
+
+ gcap = azx_readw(chip, GCAP);
+ snd_printdd("chipset global capabilities = 0x%x\n", gcap);
+
+ /* ATI chips seems buggy about 64bit DMA addresses */
+ if (chip->driver_type == AZX_DRIVER_ATI)
+ gcap &= ~0x01;
+
+ /* allow 64bit DMA address if supported by H/W */
+ if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK))
+ pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK);
+ else {
+ pci_set_dma_mask(pci, DMA_32BIT_MASK);
+ pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK);
+ }
+
+ /* read number of streams from GCAP register instead of using
+ * hardcoded value
+ */
+ chip->capture_streams = (gcap >> 8) & 0x0f;
+ chip->playback_streams = (gcap >> 12) & 0x0f;
+ if (!chip->playback_streams && !chip->capture_streams) {
+ /* gcap didn't give any info, switching to old method */
+
+ switch (chip->driver_type) {
+ case AZX_DRIVER_ULI:
+ chip->playback_streams = ULI_NUM_PLAYBACK;
+ chip->capture_streams = ULI_NUM_CAPTURE;
+ break;
+ case AZX_DRIVER_ATIHDMI:
+ chip->playback_streams = ATIHDMI_NUM_PLAYBACK;
+ chip->capture_streams = ATIHDMI_NUM_CAPTURE;
+ break;
+ default:
+ chip->playback_streams = ICH6_NUM_PLAYBACK;
+ chip->capture_streams = ICH6_NUM_CAPTURE;
+ break;
+ }
+ }
+ chip->capture_index_offset = 0;
+ chip->playback_index_offset = chip->capture_streams;
+ chip->num_streams = chip->playback_streams + chip->capture_streams;
+ chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev),
+ GFP_KERNEL);
+ if (!chip->azx_dev) {
+ snd_printk(KERN_ERR "cannot malloc azx_dev\n");
+ goto errout;
+ }
+
+ for (i = 0; i < chip->num_streams; i++) {
+ /* allocate memory for the BDL for each stream */
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ BDL_SIZE, &chip->azx_dev[i].bdl);
+ if (err < 0) {
+ snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
+ goto errout;
+ }
+ }
+ /* allocate memory for the position buffer */
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ chip->num_streams * 8, &chip->posbuf);
+ if (err < 0) {
+ snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
+ goto errout;
+ }
+ /* allocate CORB/RIRB */
+ if (!chip->single_cmd) {
+ err = azx_alloc_cmd_io(chip);
+ if (err < 0)
+ goto errout;
+ }
+
+ /* initialize streams */
+ azx_init_stream(chip);
+
+ /* initialize chip */
+ azx_init_pci(chip);
+ azx_init_chip(chip);
+
+ /* codec detection */
+ if (!chip->codec_mask) {
+ snd_printk(KERN_ERR SFX "no codecs found!\n");
+ err = -ENODEV;
+ goto errout;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err <0) {
+ snd_printk(KERN_ERR SFX "Error creating device [card]!\n");
+ goto errout;
+ }
+
+ strcpy(card->driver, "HDA-Intel");
+ strcpy(card->shortname, driver_short_names[chip->driver_type]);
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->addr, chip->irq);
+
+ *rchip = chip;
+ return 0;
+
+ errout:
+ azx_free(chip);
+ return err;
+}
+
+static void power_down_all_codecs(struct azx *chip)
+{
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ /* The codecs were powered up in snd_hda_codec_new().
+ * Now all initialization done, so turn them down if possible
+ */
+ struct hda_codec *codec;
+ list_for_each_entry(codec, &chip->bus->codec_list, list) {
+ snd_hda_power_down(codec);
+ }
+#endif
+}
+
+static int __devinit azx_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct azx *chip;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (!card) {
+ snd_printk(KERN_ERR SFX "Error creating card!\n");
+ return -ENOMEM;
+ }
+
+ err = azx_create(card, pci, dev, pci_id->driver_data, &chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ /* create codec instances */
+ err = azx_codec_create(chip, model[dev], probe_mask[dev]);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* create PCM streams */
+ err = azx_pcm_create(chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* create mixer controls */
+ err = azx_mixer_create(chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ chip->running = 1;
+ power_down_all_codecs(chip);
+ azx_notifier_register(chip);
+
+ dev++;
+ return err;
+}
+
+static void __devexit azx_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+/* PCI IDs */
+static struct pci_device_id azx_ids[] = {
+ /* ICH 6..10 */
+ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x2911), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH },
+ { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH },
+ /* PCH */
+ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH },
+ /* SCH */
+ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH },
+ /* ATI SB 450/600 */
+ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI },
+ { PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATI },
+ /* ATI HDMI */
+ { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa18), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa20), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa28), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa30), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa38), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa40), .driver_data = AZX_DRIVER_ATIHDMI },
+ { PCI_DEVICE(0x1002, 0xaa48), .driver_data = AZX_DRIVER_ATIHDMI },
+ /* VIA VT8251/VT8237A */
+ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA },
+ /* SIS966 */
+ { PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS },
+ /* ULI M5461 */
+ { PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI },
+ /* NVIDIA MCP */
+ { PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA },
+ /* Teradici */
+ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
+ { 0, }
+};
+MODULE_DEVICE_TABLE(pci, azx_ids);
+
+/* pci_driver definition */
+static struct pci_driver driver = {
+ .name = "HDA Intel",
+ .id_table = azx_ids,
+ .probe = azx_probe,
+ .remove = __devexit_p(azx_remove),
+#ifdef CONFIG_PM
+ .suspend = azx_suspend,
+ .resume = azx_resume,
+#endif
+};
+
+static int __init alsa_card_azx_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_azx_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_azx_init)
+module_exit(alsa_card_azx_exit)
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
new file mode 100644
index 0000000..7957fef
--- /dev/null
+++ b/sound/pci/hda/hda_local.h
@@ -0,0 +1,433 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Local helper functions
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the Free
+ * Software Foundation; either version 2 of the License, or (at your option)
+ * any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
+ * more details.
+ *
+ * You should have received a copy of the GNU General Public License along with
+ * this program; if not, write to the Free Software Foundation, Inc., 59
+ * Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __SOUND_HDA_LOCAL_H
+#define __SOUND_HDA_LOCAL_H
+
+/*
+ * for mixer controls
+ */
+#define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \
+ ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19))
+/* mono volume with index (index=0,1,...) (channel=1,2) */
+#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+ .info = snd_hda_mixer_amp_volume_info, \
+ .get = snd_hda_mixer_amp_volume_get, \
+ .put = snd_hda_mixer_amp_volume_put, \
+ .tlv = { .c = snd_hda_mixer_amp_tlv }, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
+/* stereo volume with index */
+#define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \
+ HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, 3, xindex, direction)
+/* mono volume */
+#define HDA_CODEC_VOLUME_MONO(xname, nid, channel, xindex, direction) \
+ HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, channel, xindex, direction)
+/* stereo volume */
+#define HDA_CODEC_VOLUME(xname, nid, xindex, direction) \
+ HDA_CODEC_VOLUME_MONO(xname, nid, 3, xindex, direction)
+/* mono mute switch with index (index=0,1,...) (channel=1,2) */
+#define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
+ .info = snd_hda_mixer_amp_switch_info, \
+ .get = snd_hda_mixer_amp_switch_get, \
+ .put = snd_hda_mixer_amp_switch_put, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
+/* stereo mute switch with index */
+#define HDA_CODEC_MUTE_IDX(xname, xcidx, nid, xindex, direction) \
+ HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, 3, xindex, direction)
+/* mono mute switch */
+#define HDA_CODEC_MUTE_MONO(xname, nid, channel, xindex, direction) \
+ HDA_CODEC_MUTE_MONO_IDX(xname, 0, nid, channel, xindex, direction)
+/* stereo mute switch */
+#define HDA_CODEC_MUTE(xname, nid, xindex, direction) \
+ HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction)
+
+int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv);
+int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+/* lowlevel accessor with caching; use carefully */
+int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
+ int direction, int index);
+int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
+ int direction, int idx, int mask, int val);
+int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx, int mask, int val);
+#ifdef SND_HDA_NEEDS_RESUME
+void snd_hda_codec_resume_amp(struct hda_codec *codec);
+#endif
+
+void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
+ unsigned int *tlv);
+struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
+ const char *name);
+int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
+ unsigned int *tlv, const char **slaves);
+
+/* amp value bits */
+#define HDA_AMP_MUTE 0x80
+#define HDA_AMP_UNMUTE 0x00
+#define HDA_AMP_VOLMASK 0x7f
+
+/* mono switch binding multiple inputs */
+#define HDA_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = snd_hda_mixer_amp_switch_info, \
+ .get = snd_hda_mixer_bind_switch_get, \
+ .put = snd_hda_mixer_bind_switch_put, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, indices, direction) }
+
+/* stereo switch binding multiple inputs */
+#define HDA_BIND_MUTE(xname,nid,indices,dir) \
+ HDA_BIND_MUTE_MONO(xname,nid,3,indices,dir)
+
+int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+/* more generic bound controls */
+struct hda_ctl_ops {
+ snd_kcontrol_info_t *info;
+ snd_kcontrol_get_t *get;
+ snd_kcontrol_put_t *put;
+ snd_kcontrol_tlv_rw_t *tlv;
+};
+
+extern struct hda_ctl_ops snd_hda_bind_vol; /* for bind-volume with TLV */
+extern struct hda_ctl_ops snd_hda_bind_sw; /* for bind-switch */
+
+struct hda_bind_ctls {
+ struct hda_ctl_ops *ops;
+ long values[];
+};
+
+int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv);
+
+#define HDA_BIND_VOL(xname, bindrec) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |\
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,\
+ .info = snd_hda_mixer_bind_ctls_info,\
+ .get = snd_hda_mixer_bind_ctls_get,\
+ .put = snd_hda_mixer_bind_ctls_put,\
+ .tlv = { .c = snd_hda_mixer_bind_tlv },\
+ .private_value = (long) (bindrec) }
+#define HDA_BIND_SW(xname, bindrec) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,\
+ .name = xname, \
+ .info = snd_hda_mixer_bind_ctls_info,\
+ .get = snd_hda_mixer_bind_ctls_get,\
+ .put = snd_hda_mixer_bind_ctls_put,\
+ .private_value = (long) (bindrec) }
+
+/*
+ * SPDIF I/O
+ */
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
+
+/*
+ * input MUX helper
+ */
+#define HDA_MAX_NUM_INPUTS 16
+struct hda_input_mux_item {
+ const char *label;
+ unsigned int index;
+};
+struct hda_input_mux {
+ unsigned int num_items;
+ struct hda_input_mux_item items[HDA_MAX_NUM_INPUTS];
+};
+
+int snd_hda_input_mux_info(const struct hda_input_mux *imux,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_input_mux_put(struct hda_codec *codec,
+ const struct hda_input_mux *imux,
+ struct snd_ctl_elem_value *ucontrol, hda_nid_t nid,
+ unsigned int *cur_val);
+
+/*
+ * Channel mode helper
+ */
+struct hda_channel_mode {
+ int channels;
+ const struct hda_verb *sequence;
+};
+
+int snd_hda_ch_mode_info(struct hda_codec *codec,
+ struct snd_ctl_elem_info *uinfo,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes);
+int snd_hda_ch_mode_get(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes,
+ int max_channels);
+int snd_hda_ch_mode_put(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes,
+ int *max_channelsp);
+
+/*
+ * Multi-channel / digital-out PCM helper
+ */
+
+enum { HDA_FRONT, HDA_REAR, HDA_CLFE, HDA_SIDE }; /* index for dac_nidx */
+enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */
+
+struct hda_multi_out {
+ int num_dacs; /* # of DACs, must be more than 1 */
+ hda_nid_t *dac_nids; /* DAC list */
+ hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */
+ hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */
+ hda_nid_t dig_out_nid; /* digital out audio widget */
+ int max_channels; /* currently supported analog channels */
+ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */
+ int no_share_stream; /* don't share a stream with multiple pins */
+ int share_spdif; /* share SPDIF pin */
+ /* PCM information for both analog and SPDIF DACs */
+ unsigned int analog_rates;
+ unsigned int analog_maxbps;
+ u64 analog_formats;
+ unsigned int spdif_rates;
+ unsigned int spdif_maxbps;
+ u64 spdif_formats;
+};
+
+int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
+ struct hda_multi_out *mout);
+int snd_hda_multi_out_dig_open(struct hda_codec *codec,
+ struct hda_multi_out *mout);
+int snd_hda_multi_out_dig_close(struct hda_codec *codec,
+ struct hda_multi_out *mout);
+int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
+ struct hda_multi_out *mout,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream);
+int snd_hda_multi_out_analog_open(struct hda_codec *codec,
+ struct hda_multi_out *mout,
+ struct snd_pcm_substream *substream,
+ struct hda_pcm_stream *hinfo);
+int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
+ struct hda_multi_out *mout,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream);
+int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
+ struct hda_multi_out *mout);
+
+/*
+ * generic codec parser
+ */
+#ifdef CONFIG_SND_HDA_GENERIC
+int snd_hda_parse_generic_codec(struct hda_codec *codec);
+#else
+static inline int snd_hda_parse_generic_codec(struct hda_codec *codec)
+{
+ return -ENODEV;
+}
+#endif
+
+/*
+ * generic proc interface
+ */
+#ifdef CONFIG_PROC_FS
+int snd_hda_codec_proc_new(struct hda_codec *codec);
+#else
+static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
+#endif
+
+/*
+ * Misc
+ */
+int snd_hda_check_board_config(struct hda_codec *codec, int num_configs,
+ const char **modelnames,
+ const struct snd_pci_quirk *pci_list);
+int snd_hda_add_new_ctls(struct hda_codec *codec,
+ struct snd_kcontrol_new *knew);
+
+/*
+ * unsolicited event handler
+ */
+
+#define HDA_UNSOL_QUEUE_SIZE 64
+
+struct hda_bus_unsolicited {
+ /* ring buffer */
+ u32 queue[HDA_UNSOL_QUEUE_SIZE * 2];
+ unsigned int rp, wp;
+
+ /* workqueue */
+ struct work_struct work;
+ struct hda_bus *bus;
+};
+
+/*
+ * Helper for automatic ping configuration
+ */
+
+enum {
+ AUTO_PIN_MIC,
+ AUTO_PIN_FRONT_MIC,
+ AUTO_PIN_LINE,
+ AUTO_PIN_FRONT_LINE,
+ AUTO_PIN_CD,
+ AUTO_PIN_AUX,
+ AUTO_PIN_LAST
+};
+
+enum {
+ AUTO_PIN_LINE_OUT,
+ AUTO_PIN_SPEAKER_OUT,
+ AUTO_PIN_HP_OUT
+};
+
+extern const char *auto_pin_cfg_labels[AUTO_PIN_LAST];
+
+#define AUTO_CFG_MAX_OUTS 5
+
+struct auto_pin_cfg {
+ int line_outs;
+ /* sorted in the order of Front/Surr/CLFE/Side */
+ hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS];
+ int speaker_outs;
+ hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS];
+ int hp_outs;
+ int line_out_type; /* AUTO_PIN_XXX_OUT */
+ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
+ hda_nid_t input_pins[AUTO_PIN_LAST];
+ hda_nid_t dig_out_pin;
+ hda_nid_t dig_in_pin;
+ hda_nid_t mono_out_pin;
+};
+
+#define get_defcfg_connect(cfg) \
+ ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
+#define get_defcfg_association(cfg) \
+ ((cfg & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT)
+#define get_defcfg_location(cfg) \
+ ((cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
+#define get_defcfg_sequence(cfg) \
+ (cfg & AC_DEFCFG_SEQUENCE)
+#define get_defcfg_device(cfg) \
+ ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
+
+int snd_hda_parse_pin_def_config(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ hda_nid_t *ignore_nids);
+
+/* amp values */
+#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
+#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8))
+#define AMP_OUT_MUTE 0xb080
+#define AMP_OUT_UNMUTE 0xb000
+#define AMP_OUT_ZERO 0xb000
+/* pinctl values */
+#define PIN_IN (AC_PINCTL_IN_EN)
+#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ)
+#define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50)
+#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD)
+#define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80)
+#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100)
+#define PIN_OUT (AC_PINCTL_OUT_EN)
+#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN)
+#define PIN_HP_AMP (AC_PINCTL_HP_EN)
+
+/*
+ * get widget capabilities
+ */
+static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
+{
+ if (nid < codec->start_nid ||
+ nid >= codec->start_nid + codec->num_nodes)
+ return 0;
+ return codec->wcaps[nid - codec->start_nid];
+}
+
+u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
+int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
+ unsigned int caps);
+
+/*
+ * hwdep interface
+ */
+int snd_hda_create_hwdep(struct hda_codec *codec);
+
+/*
+ * power-management
+ */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+void snd_hda_schedule_power_save(struct hda_codec *codec);
+
+struct hda_amp_list {
+ hda_nid_t nid;
+ unsigned char dir;
+ unsigned char idx;
+};
+
+struct hda_loopback_check {
+ struct hda_amp_list *amplist;
+ int power_on;
+};
+
+int snd_hda_check_amp_list_power(struct hda_codec *codec,
+ struct hda_loopback_check *check,
+ hda_nid_t nid);
+#endif /* CONFIG_SND_HDA_POWER_SAVE */
+
+/*
+ * AMP control callbacks
+ */
+/* retrieve parameters from private_value */
+#define get_amp_nid(kc) ((kc)->private_value & 0xffff)
+#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3)
+#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1)
+#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf)
+
+#endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
new file mode 100644
index 0000000..dfbcfa8
--- /dev/null
+++ b/sound/pci/hda/hda_patch.h
@@ -0,0 +1,22 @@
+/*
+ * HDA Patches - included by hda_codec.c
+ */
+
+/* Realtek codecs */
+extern struct hda_codec_preset snd_hda_preset_realtek[];
+/* C-Media codecs */
+extern struct hda_codec_preset snd_hda_preset_cmedia[];
+/* Analog Devices codecs */
+extern struct hda_codec_preset snd_hda_preset_analog[];
+/* SigmaTel codecs */
+extern struct hda_codec_preset snd_hda_preset_sigmatel[];
+/* SiLabs 3054/3055 modem codecs */
+extern struct hda_codec_preset snd_hda_preset_si3054[];
+/* ATI HDMI codecs */
+extern struct hda_codec_preset snd_hda_preset_atihdmi[];
+/* Conexant audio codec */
+extern struct hda_codec_preset snd_hda_preset_conexant[];
+/* VIA codecs */
+extern struct hda_codec_preset snd_hda_preset_via[];
+/* NVIDIA HDMI codecs */
+extern struct hda_codec_preset snd_hda_preset_nvhdmi[];
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
new file mode 100644
index 0000000..c39af98
--- /dev/null
+++ b/sound/pci/hda/hda_proc.c
@@ -0,0 +1,676 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Generic proc interface
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+static const char *get_wid_type_name(unsigned int wid_value)
+{
+ static char *names[16] = {
+ [AC_WID_AUD_OUT] = "Audio Output",
+ [AC_WID_AUD_IN] = "Audio Input",
+ [AC_WID_AUD_MIX] = "Audio Mixer",
+ [AC_WID_AUD_SEL] = "Audio Selector",
+ [AC_WID_PIN] = "Pin Complex",
+ [AC_WID_POWER] = "Power Widget",
+ [AC_WID_VOL_KNB] = "Volume Knob Widget",
+ [AC_WID_BEEP] = "Beep Generator Widget",
+ [AC_WID_VENDOR] = "Vendor Defined Widget",
+ };
+ wid_value &= 0xf;
+ if (names[wid_value])
+ return names[wid_value];
+ else
+ return "UNKNOWN Widget";
+}
+
+static void print_amp_caps(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid, int dir)
+{
+ unsigned int caps;
+ caps = snd_hda_param_read(codec, nid,
+ dir == HDA_OUTPUT ?
+ AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP);
+ if (caps == -1 || caps == 0) {
+ snd_iprintf(buffer, "N/A\n");
+ return;
+ }
+ snd_iprintf(buffer, "ofs=0x%02x, nsteps=0x%02x, stepsize=0x%02x, "
+ "mute=%x\n",
+ caps & AC_AMPCAP_OFFSET,
+ (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT,
+ (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT,
+ (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT);
+}
+
+static void print_amp_vals(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid,
+ int dir, int stereo, int indices)
+{
+ unsigned int val;
+ int i;
+
+ dir = dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
+ for (i = 0; i < indices; i++) {
+ snd_iprintf(buffer, " [");
+ if (stereo) {
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_LEFT | dir | i);
+ snd_iprintf(buffer, "0x%02x ", val);
+ }
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_RIGHT | dir | i);
+ snd_iprintf(buffer, "0x%02x]", val);
+ }
+ snd_iprintf(buffer, "\n");
+}
+
+static void print_pcm_rates(struct snd_info_buffer *buffer, unsigned int pcm)
+{
+ static unsigned int rates[] = {
+ 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
+ 96000, 176400, 192000, 384000
+ };
+ int i;
+
+ pcm &= AC_SUPPCM_RATES;
+ snd_iprintf(buffer, " rates [0x%x]:", pcm);
+ for (i = 0; i < ARRAY_SIZE(rates); i++)
+ if (pcm & (1 << i))
+ snd_iprintf(buffer, " %d", rates[i]);
+ snd_iprintf(buffer, "\n");
+}
+
+static void print_pcm_bits(struct snd_info_buffer *buffer, unsigned int pcm)
+{
+ static unsigned int bits[] = { 8, 16, 20, 24, 32 };
+ int i;
+
+ pcm = (pcm >> 16) & 0xff;
+ snd_iprintf(buffer, " bits [0x%x]:", pcm);
+ for (i = 0; i < ARRAY_SIZE(bits); i++)
+ if (pcm & (1 << i))
+ snd_iprintf(buffer, " %d", bits[i]);
+ snd_iprintf(buffer, "\n");
+}
+
+static void print_pcm_formats(struct snd_info_buffer *buffer,
+ unsigned int streams)
+{
+ snd_iprintf(buffer, " formats [0x%x]:", streams & 0xf);
+ if (streams & AC_SUPFMT_PCM)
+ snd_iprintf(buffer, " PCM");
+ if (streams & AC_SUPFMT_FLOAT32)
+ snd_iprintf(buffer, " FLOAT");
+ if (streams & AC_SUPFMT_AC3)
+ snd_iprintf(buffer, " AC3");
+ snd_iprintf(buffer, "\n");
+}
+
+static void print_pcm_caps(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int pcm = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+ unsigned int stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+ if (pcm == -1 || stream == -1) {
+ snd_iprintf(buffer, "N/A\n");
+ return;
+ }
+ print_pcm_rates(buffer, pcm);
+ print_pcm_bits(buffer, pcm);
+ print_pcm_formats(buffer, stream);
+}
+
+static const char *get_jack_location(u32 cfg)
+{
+ static char *bases[7] = {
+ "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom",
+ };
+ static unsigned char specials_idx[] = {
+ 0x07, 0x08,
+ 0x17, 0x18, 0x19,
+ 0x37, 0x38
+ };
+ static char *specials[] = {
+ "Rear Panel", "Drive Bar",
+ "Riser", "HDMI", "ATAPI",
+ "Mobile-In", "Mobile-Out"
+ };
+ int i;
+ cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT;
+ if ((cfg & 0x0f) < 7)
+ return bases[cfg & 0x0f];
+ for (i = 0; i < ARRAY_SIZE(specials_idx); i++) {
+ if (cfg == specials_idx[i])
+ return specials[i];
+ }
+ return "UNKNOWN";
+}
+
+static const char *get_jack_connection(u32 cfg)
+{
+ static char *names[16] = {
+ "Unknown", "1/8", "1/4", "ATAPI",
+ "RCA", "Optical","Digital", "Analog",
+ "DIN", "XLR", "RJ11", "Comb",
+ NULL, NULL, NULL, "Other"
+ };
+ cfg = (cfg & AC_DEFCFG_CONN_TYPE) >> AC_DEFCFG_CONN_TYPE_SHIFT;
+ if (names[cfg])
+ return names[cfg];
+ else
+ return "UNKNOWN";
+}
+
+static const char *get_jack_color(u32 cfg)
+{
+ static char *names[16] = {
+ "Unknown", "Black", "Grey", "Blue",
+ "Green", "Red", "Orange", "Yellow",
+ "Purple", "Pink", NULL, NULL,
+ NULL, NULL, "White", "Other",
+ };
+ cfg = (cfg & AC_DEFCFG_COLOR) >> AC_DEFCFG_COLOR_SHIFT;
+ if (names[cfg])
+ return names[cfg];
+ else
+ return "UNKNOWN";
+}
+
+static void print_pin_caps(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid,
+ int *supports_vref)
+{
+ static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" };
+ static char *jack_types[16] = {
+ "Line Out", "Speaker", "HP Out", "CD",
+ "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
+ "Line In", "Aux", "Mic", "Telephony",
+ "SPDIF In", "Digitial In", "Reserved", "Other"
+ };
+ static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" };
+ unsigned int caps, val;
+
+ caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+ snd_iprintf(buffer, " Pincap 0x%08x:", caps);
+ if (caps & AC_PINCAP_IN)
+ snd_iprintf(buffer, " IN");
+ if (caps & AC_PINCAP_OUT)
+ snd_iprintf(buffer, " OUT");
+ if (caps & AC_PINCAP_HP_DRV)
+ snd_iprintf(buffer, " HP");
+ if (caps & AC_PINCAP_EAPD)
+ snd_iprintf(buffer, " EAPD");
+ if (caps & AC_PINCAP_PRES_DETECT)
+ snd_iprintf(buffer, " Detect");
+ if (caps & AC_PINCAP_BALANCE)
+ snd_iprintf(buffer, " Balanced");
+ if (caps & AC_PINCAP_HDMI) {
+ /* Realtek uses this bit as a different meaning */
+ if ((codec->vendor_id >> 16) == 0x10ec)
+ snd_iprintf(buffer, " R/L");
+ else
+ snd_iprintf(buffer, " HDMI");
+ }
+ if (caps & AC_PINCAP_TRIG_REQ)
+ snd_iprintf(buffer, " Trigger");
+ if (caps & AC_PINCAP_IMP_SENSE)
+ snd_iprintf(buffer, " ImpSense");
+ snd_iprintf(buffer, "\n");
+ if (caps & AC_PINCAP_VREF) {
+ unsigned int vref =
+ (caps & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+ snd_iprintf(buffer, " Vref caps:");
+ if (vref & AC_PINCAP_VREF_HIZ)
+ snd_iprintf(buffer, " HIZ");
+ if (vref & AC_PINCAP_VREF_50)
+ snd_iprintf(buffer, " 50");
+ if (vref & AC_PINCAP_VREF_GRD)
+ snd_iprintf(buffer, " GRD");
+ if (vref & AC_PINCAP_VREF_80)
+ snd_iprintf(buffer, " 80");
+ if (vref & AC_PINCAP_VREF_100)
+ snd_iprintf(buffer, " 100");
+ snd_iprintf(buffer, "\n");
+ *supports_vref = 1;
+ } else
+ *supports_vref = 0;
+ if (caps & AC_PINCAP_EAPD) {
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_EAPD_BTLENABLE, 0);
+ snd_iprintf(buffer, " EAPD 0x%x:", val);
+ if (val & AC_EAPDBTL_BALANCED)
+ snd_iprintf(buffer, " BALANCED");
+ if (val & AC_EAPDBTL_EAPD)
+ snd_iprintf(buffer, " EAPD");
+ if (val & AC_EAPDBTL_LR_SWAP)
+ snd_iprintf(buffer, " R/L");
+ snd_iprintf(buffer, "\n");
+ }
+ caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps,
+ jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT],
+ jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT],
+ jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3],
+ get_jack_location(caps));
+ snd_iprintf(buffer, " Conn = %s, Color = %s\n",
+ get_jack_connection(caps),
+ get_jack_color(caps));
+ /* Default association and sequence values refer to default grouping
+ * of pin complexes and their sequence within the group. This is used
+ * for priority and resource allocation.
+ */
+ snd_iprintf(buffer, " DefAssociation = 0x%x, Sequence = 0x%x\n",
+ (caps & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT,
+ caps & AC_DEFCFG_SEQUENCE);
+ if (((caps & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) &
+ AC_DEFCFG_MISC_NO_PRESENCE) {
+ /* Miscellaneous bit indicates external hardware does not
+ * support presence detection even if the pin complex
+ * indicates it is supported.
+ */
+ snd_iprintf(buffer, " Misc = NO_PRESENCE\n");
+ }
+}
+
+static void print_pin_ctls(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid,
+ int supports_vref)
+{
+ unsigned int pinctls;
+
+ pinctls = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ snd_iprintf(buffer, " Pin-ctls: 0x%02x:", pinctls);
+ if (pinctls & AC_PINCTL_IN_EN)
+ snd_iprintf(buffer, " IN");
+ if (pinctls & AC_PINCTL_OUT_EN)
+ snd_iprintf(buffer, " OUT");
+ if (pinctls & AC_PINCTL_HP_EN)
+ snd_iprintf(buffer, " HP");
+ if (supports_vref) {
+ int vref = pinctls & AC_PINCTL_VREFEN;
+ switch (vref) {
+ case AC_PINCTL_VREF_HIZ:
+ snd_iprintf(buffer, " VREF_HIZ");
+ break;
+ case AC_PINCTL_VREF_50:
+ snd_iprintf(buffer, " VREF_50");
+ break;
+ case AC_PINCTL_VREF_GRD:
+ snd_iprintf(buffer, " VREF_GRD");
+ break;
+ case AC_PINCTL_VREF_80:
+ snd_iprintf(buffer, " VREF_80");
+ break;
+ case AC_PINCTL_VREF_100:
+ snd_iprintf(buffer, " VREF_100");
+ break;
+ }
+ }
+ snd_iprintf(buffer, "\n");
+}
+
+static void print_vol_knob(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int cap = snd_hda_param_read(codec, nid,
+ AC_PAR_VOL_KNB_CAP);
+ snd_iprintf(buffer, " Volume-Knob: delta=%d, steps=%d, ",
+ (cap >> 7) & 1, cap & 0x7f);
+ cap = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
+ snd_iprintf(buffer, "direct=%d, val=%d\n",
+ (cap >> 7) & 1, cap & 0x7f);
+}
+
+static void print_audio_io(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid,
+ unsigned int wid_type)
+{
+ int conv = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
+ snd_iprintf(buffer,
+ " Converter: stream=%d, channel=%d\n",
+ (conv & AC_CONV_STREAM) >> AC_CONV_STREAM_SHIFT,
+ conv & AC_CONV_CHANNEL);
+
+ if (wid_type == AC_WID_AUD_IN && (conv & AC_CONV_CHANNEL) == 0) {
+ int sdi = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_SDI_SELECT, 0);
+ snd_iprintf(buffer, " SDI-Select: %d\n",
+ sdi & AC_SDI_SELECT);
+ }
+}
+
+static void print_digital_conv(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int digi1 = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1, 0);
+ snd_iprintf(buffer, " Digital:");
+ if (digi1 & AC_DIG1_ENABLE)
+ snd_iprintf(buffer, " Enabled");
+ if (digi1 & AC_DIG1_V)
+ snd_iprintf(buffer, " Validity");
+ if (digi1 & AC_DIG1_VCFG)
+ snd_iprintf(buffer, " ValidityCfg");
+ if (digi1 & AC_DIG1_EMPHASIS)
+ snd_iprintf(buffer, " Preemphasis");
+ if (digi1 & AC_DIG1_COPYRIGHT)
+ snd_iprintf(buffer, " Copyright");
+ if (digi1 & AC_DIG1_NONAUDIO)
+ snd_iprintf(buffer, " Non-Audio");
+ if (digi1 & AC_DIG1_PROFESSIONAL)
+ snd_iprintf(buffer, " Pro");
+ if (digi1 & AC_DIG1_LEVEL)
+ snd_iprintf(buffer, " GenLevel");
+ snd_iprintf(buffer, "\n");
+ snd_iprintf(buffer, " Digital category: 0x%x\n",
+ (digi1 >> 8) & AC_DIG2_CC);
+}
+
+static const char *get_pwr_state(u32 state)
+{
+ static const char *buf[4] = {
+ "D0", "D1", "D2", "D3"
+ };
+ if (state < 4)
+ return buf[state];
+ return "UNKNOWN";
+}
+
+static void print_power_state(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ int pwr = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+ snd_iprintf(buffer, " Power: setting=%s, actual=%s\n",
+ get_pwr_state(pwr & AC_PWRST_SETTING),
+ get_pwr_state((pwr & AC_PWRST_ACTUAL) >>
+ AC_PWRST_ACTUAL_SHIFT));
+}
+
+static void print_unsol_cap(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ int unsol = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_UNSOLICITED_RESPONSE, 0);
+ snd_iprintf(buffer,
+ " Unsolicited: tag=%02x, enabled=%d\n",
+ unsol & AC_UNSOL_TAG,
+ (unsol & AC_UNSOL_ENABLED) ? 1 : 0);
+}
+
+static void print_proc_caps(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int proc_caps = snd_hda_param_read(codec, nid,
+ AC_PAR_PROC_CAP);
+ snd_iprintf(buffer, " Processing caps: benign=%d, ncoeff=%d\n",
+ proc_caps & AC_PCAP_BENIGN,
+ (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT);
+}
+
+static void print_conn_list(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid,
+ unsigned int wid_type, hda_nid_t *conn,
+ int conn_len)
+{
+ int c, curr = -1;
+
+ if (conn_len > 1 && wid_type != AC_WID_AUD_MIX)
+ curr = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_SEL, 0);
+ snd_iprintf(buffer, " Connection: %d\n", conn_len);
+ if (conn_len > 0) {
+ snd_iprintf(buffer, " ");
+ for (c = 0; c < conn_len; c++) {
+ snd_iprintf(buffer, " 0x%02x", conn[c]);
+ if (c == curr)
+ snd_iprintf(buffer, "*");
+ }
+ snd_iprintf(buffer, "\n");
+ }
+}
+
+static void print_realtek_coef(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ int coeff = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_iprintf(buffer, " Processing Coefficient: 0x%02x\n", coeff);
+ coeff = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_COEF_INDEX, 0);
+ snd_iprintf(buffer, " Coefficient Index: 0x%02x\n", coeff);
+}
+
+static void print_gpio(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int gpio =
+ snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP);
+ unsigned int enable, direction, wake, unsol, sticky, data;
+ int i, max;
+ snd_iprintf(buffer, "GPIO: io=%d, o=%d, i=%d, "
+ "unsolicited=%d, wake=%d\n",
+ gpio & AC_GPIO_IO_COUNT,
+ (gpio & AC_GPIO_O_COUNT) >> AC_GPIO_O_COUNT_SHIFT,
+ (gpio & AC_GPIO_I_COUNT) >> AC_GPIO_I_COUNT_SHIFT,
+ (gpio & AC_GPIO_UNSOLICITED) ? 1 : 0,
+ (gpio & AC_GPIO_WAKE) ? 1 : 0);
+ max = gpio & AC_GPIO_IO_COUNT;
+ if (!max || max > 8)
+ return;
+ enable = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_MASK, 0);
+ direction = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_DIRECTION, 0);
+ wake = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_WAKE_MASK, 0);
+ unsol = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, 0);
+ sticky = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_STICKY_MASK, 0);
+ data = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_DATA, 0);
+ for (i = 0; i < max; ++i)
+ snd_iprintf(buffer,
+ " IO[%d]: enable=%d, dir=%d, wake=%d, "
+ "sticky=%d, data=%d\n", i,
+ (enable & (1<<i)) ? 1 : 0,
+ (direction & (1<<i)) ? 1 : 0,
+ (wake & (1<<i)) ? 1 : 0,
+ (sticky & (1<<i)) ? 1 : 0,
+ (data & (1<<i)) ? 1 : 0);
+ /* FIXME: add GPO and GPI pin information */
+}
+
+static void print_codec_info(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct hda_codec *codec = entry->private_data;
+ char buf[32];
+ hda_nid_t nid;
+ int i, nodes;
+
+ snd_hda_get_codec_name(codec, buf, sizeof(buf));
+ snd_iprintf(buffer, "Codec: %s\n", buf);
+ snd_iprintf(buffer, "Address: %d\n", codec->addr);
+ snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id);
+ snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id);
+ snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id);
+
+ if (codec->mfg)
+ snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg);
+ else
+ snd_iprintf(buffer, "No Modem Function Group found\n");
+
+ if (! codec->afg)
+ return;
+ snd_hda_power_up(codec);
+ snd_iprintf(buffer, "Default PCM:\n");
+ print_pcm_caps(buffer, codec, codec->afg);
+ snd_iprintf(buffer, "Default Amp-In caps: ");
+ print_amp_caps(buffer, codec, codec->afg, HDA_INPUT);
+ snd_iprintf(buffer, "Default Amp-Out caps: ");
+ print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT);
+
+ nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
+ if (! nid || nodes < 0) {
+ snd_iprintf(buffer, "Invalid AFG subtree\n");
+ snd_hda_power_down(codec);
+ return;
+ }
+
+ print_gpio(buffer, codec, codec->afg);
+
+ for (i = 0; i < nodes; i++, nid++) {
+ unsigned int wid_caps =
+ snd_hda_param_read(codec, nid,
+ AC_PAR_AUDIO_WIDGET_CAP);
+ unsigned int wid_type =
+ (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
+ int conn_len = 0;
+
+ snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid,
+ get_wid_type_name(wid_type), wid_caps);
+ if (wid_caps & AC_WCAP_STEREO) {
+ unsigned int chans;
+ chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13;
+ chans = ((chans << 1) | 1) + 1;
+ if (chans == 2)
+ snd_iprintf(buffer, " Stereo");
+ else
+ snd_iprintf(buffer, " %d-Channels", chans);
+ } else
+ snd_iprintf(buffer, " Mono");
+ if (wid_caps & AC_WCAP_DIGITAL)
+ snd_iprintf(buffer, " Digital");
+ if (wid_caps & AC_WCAP_IN_AMP)
+ snd_iprintf(buffer, " Amp-In");
+ if (wid_caps & AC_WCAP_OUT_AMP)
+ snd_iprintf(buffer, " Amp-Out");
+ if (wid_caps & AC_WCAP_STRIPE)
+ snd_iprintf(buffer, " Stripe");
+ if (wid_caps & AC_WCAP_LR_SWAP)
+ snd_iprintf(buffer, " R/L");
+ if (wid_caps & AC_WCAP_CP_CAPS)
+ snd_iprintf(buffer, " CP");
+ snd_iprintf(buffer, "\n");
+
+ /* volume knob is a special widget that always have connection
+ * list
+ */
+ if (wid_type == AC_WID_VOL_KNB)
+ wid_caps |= AC_WCAP_CONN_LIST;
+
+ if (wid_caps & AC_WCAP_CONN_LIST)
+ conn_len = snd_hda_get_connections(codec, nid, conn,
+ HDA_MAX_CONNECTIONS);
+
+ if (wid_caps & AC_WCAP_IN_AMP) {
+ snd_iprintf(buffer, " Amp-In caps: ");
+ print_amp_caps(buffer, codec, nid, HDA_INPUT);
+ snd_iprintf(buffer, " Amp-In vals: ");
+ print_amp_vals(buffer, codec, nid, HDA_INPUT,
+ wid_caps & AC_WCAP_STEREO,
+ wid_type == AC_WID_PIN ? 1 : conn_len);
+ }
+ if (wid_caps & AC_WCAP_OUT_AMP) {
+ snd_iprintf(buffer, " Amp-Out caps: ");
+ print_amp_caps(buffer, codec, nid, HDA_OUTPUT);
+ snd_iprintf(buffer, " Amp-Out vals: ");
+ print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
+ wid_caps & AC_WCAP_STEREO, 1);
+ }
+
+ switch (wid_type) {
+ case AC_WID_PIN: {
+ int supports_vref;
+ print_pin_caps(buffer, codec, nid, &supports_vref);
+ print_pin_ctls(buffer, codec, nid, supports_vref);
+ break;
+ }
+ case AC_WID_VOL_KNB:
+ print_vol_knob(buffer, codec, nid);
+ break;
+ case AC_WID_AUD_OUT:
+ case AC_WID_AUD_IN:
+ print_audio_io(buffer, codec, nid, wid_type);
+ if (wid_caps & AC_WCAP_DIGITAL)
+ print_digital_conv(buffer, codec, nid);
+ if (wid_caps & AC_WCAP_FORMAT_OVRD) {
+ snd_iprintf(buffer, " PCM:\n");
+ print_pcm_caps(buffer, codec, nid);
+ }
+ break;
+ }
+
+ if (wid_caps & AC_WCAP_UNSOL_CAP)
+ print_unsol_cap(buffer, codec, nid);
+
+ if (wid_caps & AC_WCAP_POWER)
+ print_power_state(buffer, codec, nid);
+
+ if (wid_caps & AC_WCAP_DELAY)
+ snd_iprintf(buffer, " Delay: %d samples\n",
+ (wid_caps & AC_WCAP_DELAY) >>
+ AC_WCAP_DELAY_SHIFT);
+
+ if (wid_caps & AC_WCAP_CONN_LIST)
+ print_conn_list(buffer, codec, nid, wid_type,
+ conn, conn_len);
+
+ if (wid_caps & AC_WCAP_PROC_WID)
+ print_proc_caps(buffer, codec, nid);
+
+ /* NID 0x20 == Realtek Define Registers */
+ if (codec->vendor_id == 0x10ec && nid == 0x20)
+ print_realtek_coef(buffer, codec, nid);
+ }
+ snd_hda_power_down(codec);
+}
+
+/*
+ * create a proc read
+ */
+int snd_hda_codec_proc_new(struct hda_codec *codec)
+{
+ char name[32];
+ struct snd_info_entry *entry;
+ int err;
+
+ snprintf(name, sizeof(name), "codec#%d", codec->addr);
+ err = snd_card_proc_new(codec->bus->card, name, &entry);
+ if (err < 0)
+ return err;
+
+ snd_info_set_text_ops(entry, codec, print_codec_info);
+ return 0;
+}
+
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
new file mode 100644
index 0000000..3495e83
--- /dev/null
+++ b/sound/pci/hda/patch_analog.c
@@ -0,0 +1,4331 @@
+/*
+ * HD audio interface patch for AD1882, AD1884, AD1981HD, AD1983, AD1984,
+ * AD1986A, AD1988
+ *
+ * Copyright (c) 2005-2007 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+
+struct ad198x_spec {
+ struct snd_kcontrol_new *mixers[5];
+ int num_mixers;
+
+ const struct hda_verb *init_verbs[5]; /* initialization verbs
+ * don't forget NULL termination!
+ */
+ unsigned int num_init_verbs;
+
+ /* playback */
+ struct hda_multi_out multiout; /* playback set-up
+ * max_channels, dacs must be set
+ * dig_out_nid and hp_nid are optional
+ */
+ unsigned int cur_eapd;
+ unsigned int need_dac_fix;
+
+ /* capture */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ hda_nid_t dig_in_nid; /* digital-in NID; optional */
+
+ /* capture source */
+ const struct hda_input_mux *input_mux;
+ hda_nid_t *capsrc_nids;
+ unsigned int cur_mux[3];
+
+ /* channel model */
+ const struct hda_channel_mode *channel_mode;
+ int num_channel_mode;
+
+ /* PCM information */
+ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
+
+ unsigned int spdif_route;
+
+ /* dynamic controls, init_verbs and input_mux */
+ struct auto_pin_cfg autocfg;
+ unsigned int num_kctl_alloc, num_kctl_used;
+ struct snd_kcontrol_new *kctl_alloc;
+ struct hda_input_mux private_imux;
+ hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+
+ unsigned int jack_present :1;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
+ /* for virtual master */
+ hda_nid_t vmaster_nid;
+ const char **slave_vols;
+ const char **slave_sws;
+};
+
+/*
+ * input MUX handling (common part)
+ */
+static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ return 0;
+}
+
+static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ spec->capsrc_nids[adc_idx],
+ &spec->cur_mux[adc_idx]);
+}
+
+/*
+ * initialization (common callbacks)
+ */
+static int ad198x_init(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_init_verbs; i++)
+ snd_hda_sequence_write(codec, spec->init_verbs[i]);
+ return 0;
+}
+
+static const char *ad_slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Mono Playback Volume",
+ "Speaker Playback Volume",
+ "IEC958 Playback Volume",
+ NULL
+};
+
+static const char *ad_slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Mono Playback Switch",
+ "Speaker Playback Switch",
+ "IEC958 Playback Switch",
+ NULL
+};
+
+static int ad198x_build_controls(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < spec->num_mixers; i++) {
+ err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+ if (err < 0)
+ return err;
+ }
+ if (spec->multiout.dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
+ }
+ if (spec->dig_in_nid) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+ if (err < 0)
+ return err;
+ }
+
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ unsigned int vmaster_tlv[4];
+ snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
+ HDA_OUTPUT, vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ vmaster_tlv,
+ (spec->slave_vols ?
+ spec->slave_vols : ad_slave_vols));
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL,
+ (spec->slave_sws ?
+ spec->slave_sws : ad_slave_sws));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
+/*
+ * Analog playback callbacks
+ */
+static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+/*
+ * Analog capture
+ */
+static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
+ return 0;
+}
+
+
+/*
+ */
+static struct hda_pcm_stream ad198x_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 6, /* changed later */
+ .nid = 0, /* fill later */
+ .ops = {
+ .open = ad198x_playback_pcm_open,
+ .prepare = ad198x_playback_pcm_prepare,
+ .cleanup = ad198x_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ad198x_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .prepare = ad198x_capture_pcm_prepare,
+ .cleanup = ad198x_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ad198x_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .open = ad198x_dig_playback_pcm_open,
+ .close = ad198x_dig_playback_pcm_close,
+ .prepare = ad198x_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream ad198x_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
+static int ad198x_build_pcms(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "AD198x Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+
+ if (spec->multiout.dig_out_nid) {
+ info++;
+ codec->num_pcms++;
+ info->name = "AD198x Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+ if (spec->dig_in_nid) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+ }
+ }
+
+ return 0;
+}
+
+static void ad198x_free(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int i;
+
+ if (spec->kctl_alloc) {
+ for (i = 0; i < spec->num_kctl_used; i++)
+ kfree(spec->kctl_alloc[i].name);
+ kfree(spec->kctl_alloc);
+ }
+ kfree(codec->spec);
+}
+
+static struct hda_codec_ops ad198x_patch_ops = {
+ .build_controls = ad198x_build_controls,
+ .build_pcms = ad198x_build_pcms,
+ .init = ad198x_init,
+ .free = ad198x_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = ad198x_check_power_status,
+#endif
+};
+
+
+/*
+ * EAPD control
+ * the private value = nid | (invert << 8)
+ */
+#define ad198x_eapd_info snd_ctl_boolean_mono_info
+
+static int ad198x_eapd_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ if (invert)
+ ucontrol->value.integer.value[0] = ! spec->cur_eapd;
+ else
+ ucontrol->value.integer.value[0] = spec->cur_eapd;
+ return 0;
+}
+
+static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+ unsigned int eapd;
+ eapd = !!ucontrol->value.integer.value[0];
+ if (invert)
+ eapd = !eapd;
+ if (eapd == spec->cur_eapd)
+ return 0;
+ spec->cur_eapd = eapd;
+ snd_hda_codec_write_cache(codec, nid,
+ 0, AC_VERB_SET_EAPD_BTLENABLE,
+ eapd ? 0x02 : 0x00);
+ return 1;
+}
+
+static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+
+/*
+ * AD1986A specific
+ */
+
+#define AD1986A_SPDIF_OUT 0x02
+#define AD1986A_FRONT_DAC 0x03
+#define AD1986A_SURR_DAC 0x04
+#define AD1986A_CLFE_DAC 0x05
+#define AD1986A_ADC 0x06
+
+static hda_nid_t ad1986a_dac_nids[3] = {
+ AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
+};
+static hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
+static hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
+
+static struct hda_input_mux ad1986a_capture_source = {
+ .num_items = 7,
+ .items = {
+ { "Mic", 0x0 },
+ { "CD", 0x1 },
+ { "Aux", 0x3 },
+ { "Line", 0x4 },
+ { "Mix", 0x5 },
+ { "Mono", 0x6 },
+ { "Phone", 0x7 },
+ },
+};
+
+
+static struct hda_bind_ctls ad1986a_bind_pcm_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls ad1986a_bind_pcm_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/*
+ * mixers
+ */
+static struct snd_kcontrol_new ad1986a_mixers[] = {
+ /*
+ * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
+ */
+ HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol),
+ HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/* additional mixers for 3stack mode */
+static struct snd_kcontrol_new ad1986a_3st_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = ad198x_ch_mode_info,
+ .get = ad198x_ch_mode_get,
+ .put = ad198x_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/* laptop model - 2ch only */
+static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
+
+/* master controls both pins 0x1a and 0x1b */
+static struct hda_bind_ctls ad1986a_laptop_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ 0,
+ },
+};
+
+static struct hda_bind_ctls ad1986a_laptop_master_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ 0,
+ },
+};
+
+static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+ /* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* laptop-eapd model - 2ch only */
+
+static struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x4 },
+ { "Mix", 0x5 },
+ },
+};
+
+static struct hda_input_mux ad1986a_automic_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Mix", 0x5 },
+ },
+};
+
+static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
+ HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "External Amplifier",
+ .info = ad198x_eapd_info,
+ .get = ad198x_eapd_get,
+ .put = ad198x_eapd_put,
+ .private_value = 0x1b | (1 << 8), /* port-D, inversed */
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1986a_samsung_mixers[] = {
+ HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "External Amplifier",
+ .info = ad198x_eapd_info,
+ .get = ad198x_eapd_get,
+ .put = ad198x_eapd_put,
+ .private_value = 0x1b | (1 << 8), /* port-D, inversed */
+ },
+ { } /* end */
+};
+
+/* re-connect the mic boost input according to the jack sensing */
+static void ad1986a_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+ present = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_PIN_SENSE, 0);
+ /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
+ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
+ (present & AC_PINSENSE_PRESENCE) ? 0 : 2);
+}
+
+#define AD1986A_MIC_EVENT 0x36
+
+static void ad1986a_automic_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != AD1986A_MIC_EVENT)
+ return;
+ ad1986a_automic(codec);
+}
+
+static int ad1986a_automic_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1986a_automic(codec);
+ return 0;
+}
+
+/* laptop-automute - 2ch only */
+
+static void ad1986a_update_hp(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int mute;
+
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE; /* mute internal speaker */
+ else
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+}
+
+static void ad1986a_hp_automute(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0);
+ /* Lenovo N100 seems to report the reversed bit for HP jack-sensing */
+ spec->jack_present = !(present & 0x80000000);
+ ad1986a_update_hp(codec);
+}
+
+#define AD1986A_HP_EVENT 0x37
+
+static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) != AD1986A_HP_EVENT)
+ return;
+ ad1986a_hp_automute(codec);
+}
+
+static int ad1986a_hp_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1986a_hp_automute(codec);
+ return 0;
+}
+
+/* bind hp and internal speaker mute (with plug check) */
+static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change)
+ ad1986a_update_hp(codec);
+ return change;
+}
+
+static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = {
+ HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = ad1986a_hp_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "External Amplifier",
+ .info = ad198x_eapd_info,
+ .get = ad198x_eapd_get,
+ .put = ad198x_eapd_put,
+ .private_value = 0x1b | (1 << 8), /* port-D, inversed */
+ },
+ { } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+static struct hda_verb ad1986a_init_verbs[] = {
+ /* Front, Surround, CLFE DAC; mute as default */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Downmix - off */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* HP, Line-Out, Surround, CLFE selectors */
+ {0x0a, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Mono selector */
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Mic selector: Mic 1/2 pin */
+ {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Line-in selector: Line-in */
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Mic 1/2 swap */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Record selector: mic */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Mic, Phone, CD, Aux, Line-In amp; mute as default */
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* PC beep */
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* HP Pin */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* Front, Surround, CLFE Pins */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Mono Pin */
+ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Mic Pin */
+ {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* Line, Aux, CD, Beep-In Pin */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ { } /* end */
+};
+
+static struct hda_verb ad1986a_ch2_init[] = {
+ /* Surround out -> Line In */
+ { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ /* Line-in selectors */
+ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 },
+ /* CLFE -> Mic in */
+ { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ /* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
+ { } /* end */
+};
+
+static struct hda_verb ad1986a_ch4_init[] = {
+ /* Surround out -> Surround */
+ { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ /* CLFE -> Mic in */
+ { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
+ { } /* end */
+};
+
+static struct hda_verb ad1986a_ch6_init[] = {
+ /* Surround out -> Surround out */
+ { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ /* CLFE -> CLFE */
+ { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ { } /* end */
+};
+
+static struct hda_channel_mode ad1986a_modes[3] = {
+ { 2, ad1986a_ch2_init },
+ { 4, ad1986a_ch4_init },
+ { 6, ad1986a_ch6_init },
+};
+
+/* eapd initialization */
+static struct hda_verb ad1986a_eapd_init_verbs[] = {
+ {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
+ {}
+};
+
+static struct hda_verb ad1986a_automic_verbs[] = {
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
+ {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT},
+ {}
+};
+
+/* Ultra initialization */
+static struct hda_verb ad1986a_ultra_init[] = {
+ /* eapd initialization */
+ { 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
+ /* CLFE -> Mic in */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 },
+ { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+ { } /* end */
+};
+
+/* pin sensing on HP jack */
+static struct hda_verb ad1986a_hp_init_verbs[] = {
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
+ {}
+};
+
+
+/* models */
+enum {
+ AD1986A_6STACK,
+ AD1986A_3STACK,
+ AD1986A_LAPTOP,
+ AD1986A_LAPTOP_EAPD,
+ AD1986A_LAPTOP_AUTOMUTE,
+ AD1986A_ULTRA,
+ AD1986A_SAMSUNG,
+ AD1986A_MODELS
+};
+
+static const char *ad1986a_models[AD1986A_MODELS] = {
+ [AD1986A_6STACK] = "6stack",
+ [AD1986A_3STACK] = "3stack",
+ [AD1986A_LAPTOP] = "laptop",
+ [AD1986A_LAPTOP_EAPD] = "laptop-eapd",
+ [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
+ [AD1986A_ULTRA] = "ultra",
+ [AD1986A_SAMSUNG] = "samsung",
+};
+
+static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
+ SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG),
+ SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG),
+ SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG),
+ SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
+ SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
+ SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
+ SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE),
+ SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP),
+ {}
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1986a_loopbacks[] = {
+ { 0x13, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x14, HDA_OUTPUT, 0 }, /* Phone */
+ { 0x15, HDA_OUTPUT, 0 }, /* CD */
+ { 0x16, HDA_OUTPUT, 0 }, /* Aux */
+ { 0x17, HDA_OUTPUT, 0 }, /* Line */
+ { } /* end */
+};
+#endif
+
+static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int conf = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
+}
+
+static int patch_ad1986a(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 6;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
+ spec->multiout.dac_nids = ad1986a_dac_nids;
+ spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = ad1986a_adc_nids;
+ spec->capsrc_nids = ad1986a_capsrc_nids;
+ spec->input_mux = &ad1986a_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1986a_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1986a_init_verbs;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1986a_loopbacks;
+#endif
+ spec->vmaster_nid = 0x1b;
+
+ codec->patch_ops = ad198x_patch_ops;
+
+ /* override some parameters */
+ board_config = snd_hda_check_board_config(codec, AD1986A_MODELS,
+ ad1986a_models,
+ ad1986a_cfg_tbl);
+ switch (board_config) {
+ case AD1986A_3STACK:
+ spec->num_mixers = 2;
+ spec->mixers[1] = ad1986a_3st_mixers;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = ad1986a_ch2_init;
+ spec->channel_mode = ad1986a_modes;
+ spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
+ spec->need_dac_fix = 1;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ break;
+ case AD1986A_LAPTOP:
+ spec->mixers[0] = ad1986a_laptop_mixers;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+ break;
+ case AD1986A_LAPTOP_EAPD:
+ spec->mixers[0] = ad1986a_laptop_eapd_mixers;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = ad1986a_eapd_init_verbs;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+ if (!is_jack_available(codec, 0x25))
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1986a_laptop_eapd_capture_source;
+ break;
+ case AD1986A_SAMSUNG:
+ spec->mixers[0] = ad1986a_samsung_mixers;
+ spec->num_init_verbs = 3;
+ spec->init_verbs[1] = ad1986a_eapd_init_verbs;
+ spec->init_verbs[2] = ad1986a_automic_verbs;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+ if (!is_jack_available(codec, 0x25))
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1986a_automic_capture_source;
+ codec->patch_ops.unsol_event = ad1986a_automic_unsol_event;
+ codec->patch_ops.init = ad1986a_automic_init;
+ break;
+ case AD1986A_LAPTOP_AUTOMUTE:
+ spec->mixers[0] = ad1986a_laptop_automute_mixers;
+ spec->num_init_verbs = 3;
+ spec->init_verbs[1] = ad1986a_eapd_init_verbs;
+ spec->init_verbs[2] = ad1986a_hp_init_verbs;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+ if (!is_jack_available(codec, 0x25))
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1986a_laptop_eapd_capture_source;
+ codec->patch_ops.unsol_event = ad1986a_hp_unsol_event;
+ codec->patch_ops.init = ad1986a_hp_init;
+ break;
+ case AD1986A_ULTRA:
+ spec->mixers[0] = ad1986a_laptop_eapd_mixers;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = ad1986a_ultra_init;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+ spec->multiout.dig_out_nid = 0;
+ break;
+ }
+
+ /* AD1986A has a hardware problem that it can't share a stream
+ * with multiple output pins. The copy of front to surrounds
+ * causes noisy or silent outputs at a certain timing, e.g.
+ * changing the volume.
+ * So, let's disable the shared stream.
+ */
+ spec->multiout.no_share_stream = 1;
+
+ return 0;
+}
+
+/*
+ * AD1983 specific
+ */
+
+#define AD1983_SPDIF_OUT 0x02
+#define AD1983_DAC 0x03
+#define AD1983_ADC 0x04
+
+static hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
+static hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
+static hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
+
+static struct hda_input_mux ad1983_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x1 },
+ { "Mix", 0x2 },
+ { "Mix Mono", 0x3 },
+ },
+};
+
+/*
+ * SPDIF playback route
+ */
+static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "PCM", "ADC" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->spdif_route;
+ return 0;
+}
+
+static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ if (ucontrol->value.enumerated.item[0] > 1)
+ return -EINVAL;
+ if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
+ spec->spdif_route = ucontrol->value.enumerated.item[0];
+ snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ spec->spdif_route);
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new ad1983_mixers[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb ad1983_init_verbs[] = {
+ /* Front, HP, Mono; mute as default */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Beep, PCM, Mic, Line-In: mute */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Front, HP selectors; from Mix */
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* Mono selector; from Mix */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic selector; Mic */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Line-in selector: Line-in */
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Mic boost: 0dB */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* Record selector: mic */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* SPDIF route: PCM */
+ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Front Pin */
+ {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* HP Pin */
+ {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* Mono Pin */
+ {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Mic Pin */
+ {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* Line Pin */
+ {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1983_loopbacks[] = {
+ { 0x12, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x13, HDA_OUTPUT, 0 }, /* Line */
+ { } /* end */
+};
+#endif
+
+static int patch_ad1983(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
+ spec->multiout.dac_nids = ad1983_dac_nids;
+ spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = ad1983_adc_nids;
+ spec->capsrc_nids = ad1983_capsrc_nids;
+ spec->input_mux = &ad1983_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1983_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1983_init_verbs;
+ spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1983_loopbacks;
+#endif
+ spec->vmaster_nid = 0x05;
+
+ codec->patch_ops = ad198x_patch_ops;
+
+ return 0;
+}
+
+
+/*
+ * AD1981 HD specific
+ */
+
+#define AD1981_SPDIF_OUT 0x02
+#define AD1981_DAC 0x03
+#define AD1981_ADC 0x04
+
+static hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
+static hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
+static hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
+
+/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
+static struct hda_input_mux ad1981_capture_source = {
+ .num_items = 7,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Line", 0x1 },
+ { "Mix", 0x2 },
+ { "Mix Mono", 0x3 },
+ { "CD", 0x4 },
+ { "Mic", 0x6 },
+ { "Aux", 0x7 },
+ },
+};
+
+static struct snd_kcontrol_new ad1981_mixers[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* identical with AD1983 */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb ad1981_init_verbs[] = {
+ /* Front, HP, Mono; mute as default */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Front, HP selectors; from Mix */
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* Mono selector; from Mix */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic Mixer; select Front Mic */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Mic boost: 0dB */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* Record selector: Front mic */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* SPDIF route: PCM */
+ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Front Pin */
+ {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* HP Pin */
+ {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* Mono Pin */
+ {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Front & Rear Mic Pins */
+ {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* Line Pin */
+ {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* Digital Beep */
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Line-Out as Input: disabled */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1981_loopbacks[] = {
+ { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
+ { 0x13, HDA_OUTPUT, 0 }, /* Line */
+ { 0x1b, HDA_OUTPUT, 0 }, /* Aux */
+ { 0x1c, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x1d, HDA_OUTPUT, 0 }, /* CD */
+ { } /* end */
+};
+#endif
+
+/*
+ * Patch for HP nx6320
+ *
+ * nx6320 uses EAPD in the reverse way - EAPD-on means the internal
+ * speaker output enabled _and_ mute-LED off.
+ */
+
+#define AD1981_HP_EVENT 0x37
+#define AD1981_MIC_EVENT 0x38
+
+static struct hda_verb ad1981_hp_init_verbs[] = {
+ {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */
+ /* pin sensing on HP and Mic jacks */
+ {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
+ {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
+ {}
+};
+
+/* turn on/off EAPD (+ mute HP) as a master switch */
+static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ if (! ad198x_eapd_put(kcontrol, ucontrol))
+ return 0;
+ /* change speaker pin appropriately */
+ snd_hda_codec_write(codec, 0x05, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->cur_eapd ? PIN_OUT : 0);
+ /* toggle HP mute appropriately */
+ snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ spec->cur_eapd ? 0 : HDA_AMP_MUTE);
+ return 1;
+}
+
+/* bind volumes of both NID 0x05 and 0x06 */
+static struct hda_bind_ctls ad1981_hp_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/* mute internal speaker if HP is plugged */
+static void ad1981_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x06, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+/* toggle input of built-in and mic jack appropriately */
+static void ad1981_hp_automic(struct hda_codec *codec)
+{
+ static struct hda_verb mic_jack_on[] = {
+ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {}
+ };
+ static struct hda_verb mic_jack_off[] = {
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {}
+ };
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x08, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ if (present)
+ snd_hda_sequence_write(codec, mic_jack_on);
+ else
+ snd_hda_sequence_write(codec, mic_jack_off);
+}
+
+/* unsolicited event for HP jack sensing */
+static void ad1981_hp_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ res >>= 26;
+ switch (res) {
+ case AD1981_HP_EVENT:
+ ad1981_hp_automute(codec);
+ break;
+ case AD1981_MIC_EVENT:
+ ad1981_hp_automic(codec);
+ break;
+ }
+}
+
+static struct hda_input_mux ad1981_hp_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Docking-Station", 0x1 },
+ { "Mix", 0x2 },
+ },
+};
+
+static struct snd_kcontrol_new ad1981_hp_mixers[] = {
+ HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = ad198x_eapd_info,
+ .get = ad198x_eapd_get,
+ .put = ad1981_hp_master_sw_put,
+ .private_value = 0x05,
+ },
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+#if 0
+ /* FIXME: analog mic/line loopback doesn't work with my tests...
+ * (although recording is OK)
+ */
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Docking-Station Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Docking-Station Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
+ /* FIXME: does this laptop have analog CD connection? */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
+#endif
+ HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* initialize jack-sensing, too */
+static int ad1981_hp_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1981_hp_automute(codec);
+ ad1981_hp_automic(codec);
+ return 0;
+}
+
+/* configuration for Toshiba Laptops */
+static struct hda_verb ad1981_toshiba_init_verbs[] = {
+ {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */
+ /* pin sensing on HP and Mic jacks */
+ {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
+ {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
+ {}
+};
+
+static struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
+ HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT),
+ { }
+};
+
+/* configuration for Lenovo Thinkpad T60 */
+static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* identical with AD1983 */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+static struct hda_input_mux ad1981_thinkpad_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Mix", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* models */
+enum {
+ AD1981_BASIC,
+ AD1981_HP,
+ AD1981_THINKPAD,
+ AD1981_TOSHIBA,
+ AD1981_MODELS
+};
+
+static const char *ad1981_models[AD1981_MODELS] = {
+ [AD1981_HP] = "hp",
+ [AD1981_THINKPAD] = "thinkpad",
+ [AD1981_BASIC] = "basic",
+ [AD1981_TOSHIBA] = "toshiba"
+};
+
+static struct snd_pci_quirk ad1981_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
+ SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
+ /* All HP models */
+ SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP),
+ SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
+ /* Lenovo Thinkpad T60/X60/Z6xx */
+ SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD),
+ /* HP nx6320 (reversed SSID, H/W bug) */
+ SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP),
+ {}
+};
+
+static int patch_ad1981(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
+ spec->multiout.dac_nids = ad1981_dac_nids;
+ spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = ad1981_adc_nids;
+ spec->capsrc_nids = ad1981_capsrc_nids;
+ spec->input_mux = &ad1981_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1981_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1981_init_verbs;
+ spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1981_loopbacks;
+#endif
+ spec->vmaster_nid = 0x05;
+
+ codec->patch_ops = ad198x_patch_ops;
+
+ /* override some parameters */
+ board_config = snd_hda_check_board_config(codec, AD1981_MODELS,
+ ad1981_models,
+ ad1981_cfg_tbl);
+ switch (board_config) {
+ case AD1981_HP:
+ spec->mixers[0] = ad1981_hp_mixers;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = ad1981_hp_init_verbs;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1981_hp_capture_source;
+
+ codec->patch_ops.init = ad1981_hp_init;
+ codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
+ break;
+ case AD1981_THINKPAD:
+ spec->mixers[0] = ad1981_thinkpad_mixers;
+ spec->input_mux = &ad1981_thinkpad_capture_source;
+ break;
+ case AD1981_TOSHIBA:
+ spec->mixers[0] = ad1981_hp_mixers;
+ spec->mixers[1] = ad1981_toshiba_mixers;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = ad1981_toshiba_init_verbs;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1981_hp_capture_source;
+ codec->patch_ops.init = ad1981_hp_init;
+ codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
+ break;
+ }
+ return 0;
+}
+
+
+/*
+ * AD1988
+ *
+ * Output pins and routes
+ *
+ * Pin Mix Sel DAC (*)
+ * port-A 0x11 (mute/hp) <- 0x22 <- 0x37 <- 03/04/06
+ * port-B 0x14 (mute/hp) <- 0x2b <- 0x30 <- 03/04/06
+ * port-C 0x15 (mute) <- 0x2c <- 0x31 <- 05/0a
+ * port-D 0x12 (mute/hp) <- 0x29 <- 04
+ * port-E 0x17 (mute/hp) <- 0x26 <- 0x32 <- 05/0a
+ * port-F 0x16 (mute) <- 0x2a <- 06
+ * port-G 0x24 (mute) <- 0x27 <- 05
+ * port-H 0x25 (mute) <- 0x28 <- 0a
+ * mono 0x13 (mute/amp)<- 0x1e <- 0x36 <- 03/04/06
+ *
+ * DAC0 = 03h, DAC1 = 04h, DAC2 = 05h, DAC3 = 06h, DAC4 = 0ah
+ * (*) DAC2/3/4 are swapped to DAC3/4/2 on AD198A rev.2 due to a h/w bug.
+ *
+ * Input pins and routes
+ *
+ * pin boost mix input # / adc input #
+ * port-A 0x11 -> 0x38 -> mix 2, ADC 0
+ * port-B 0x14 -> 0x39 -> mix 0, ADC 1
+ * port-C 0x15 -> 0x3a -> 33:0 - mix 1, ADC 2
+ * port-D 0x12 -> 0x3d -> mix 3, ADC 8
+ * port-E 0x17 -> 0x3c -> 34:0 - mix 4, ADC 4
+ * port-F 0x16 -> 0x3b -> mix 5, ADC 3
+ * port-G 0x24 -> N/A -> 33:1 - mix 1, 34:1 - mix 4, ADC 6
+ * port-H 0x25 -> N/A -> 33:2 - mix 1, 34:2 - mix 4, ADC 7
+ *
+ *
+ * DAC assignment
+ * 6stack - front/surr/CLFE/side/opt DACs - 04/06/05/0a/03
+ * 3stack - front/surr/CLFE/opt DACs - 04/05/0a/03
+ *
+ * Inputs of Analog Mix (0x20)
+ * 0:Port-B (front mic)
+ * 1:Port-C/G/H (line-in)
+ * 2:Port-A
+ * 3:Port-D (line-in/2)
+ * 4:Port-E/G/H (mic-in)
+ * 5:Port-F (mic2-in)
+ * 6:CD
+ * 7:Beep
+ *
+ * ADC selection
+ * 0:Port-A
+ * 1:Port-B (front mic-in)
+ * 2:Port-C (line-in)
+ * 3:Port-F (mic2-in)
+ * 4:Port-E (mic-in)
+ * 5:CD
+ * 6:Port-G
+ * 7:Port-H
+ * 8:Port-D (line-in/2)
+ * 9:Mix
+ *
+ * Proposed pin assignments by the datasheet
+ *
+ * 6-stack
+ * Port-A front headphone
+ * B front mic-in
+ * C rear line-in
+ * D rear front-out
+ * E rear mic-in
+ * F rear surround
+ * G rear CLFE
+ * H rear side
+ *
+ * 3-stack
+ * Port-A front headphone
+ * B front mic
+ * C rear line-in/surround
+ * D rear front-out
+ * E rear mic-in/CLFE
+ *
+ * laptop
+ * Port-A headphone
+ * B mic-in
+ * C docking station
+ * D internal speaker (with EAPD)
+ * E/F quad mic array
+ */
+
+
+/* models */
+enum {
+ AD1988_6STACK,
+ AD1988_6STACK_DIG,
+ AD1988_3STACK,
+ AD1988_3STACK_DIG,
+ AD1988_LAPTOP,
+ AD1988_LAPTOP_DIG,
+ AD1988_AUTO,
+ AD1988_MODEL_LAST,
+};
+
+/* reivision id to check workarounds */
+#define AD1988A_REV2 0x100200
+
+#define is_rev2(codec) \
+ ((codec)->vendor_id == 0x11d41988 && \
+ (codec)->revision_id == AD1988A_REV2)
+
+/*
+ * mixers
+ */
+
+static hda_nid_t ad1988_6stack_dac_nids[4] = {
+ 0x04, 0x06, 0x05, 0x0a
+};
+
+static hda_nid_t ad1988_3stack_dac_nids[3] = {
+ 0x04, 0x05, 0x0a
+};
+
+/* for AD1988A revision-2, DAC2-4 are swapped */
+static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
+ 0x04, 0x05, 0x0a, 0x06
+};
+
+static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
+ 0x04, 0x0a, 0x06
+};
+
+static hda_nid_t ad1988_adc_nids[3] = {
+ 0x08, 0x09, 0x0f
+};
+
+static hda_nid_t ad1988_capsrc_nids[3] = {
+ 0x0c, 0x0d, 0x0e
+};
+
+#define AD1988_SPDIF_OUT 0x02
+#define AD1988_SPDIF_OUT_HDMI 0x0b
+#define AD1988_SPDIF_IN 0x07
+
+static hda_nid_t ad1989b_slave_dig_outs[] = {
+ AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
+};
+
+static struct hda_input_mux ad1988_6stack_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Front Mic", 0x1 }, /* port-B */
+ { "Line", 0x2 }, /* port-C */
+ { "Mic", 0x4 }, /* port-E */
+ { "CD", 0x5 },
+ { "Mix", 0x9 },
+ },
+};
+
+static struct hda_input_mux ad1988_laptop_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic/Line", 0x1 }, /* port-B */
+ { "CD", 0x5 },
+ { "Mix", 0x9 },
+ },
+};
+
+/*
+ */
+static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
+ spec->num_channel_mode);
+}
+
+static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode, spec->multiout.max_channels);
+}
+
+static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+ int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ &spec->multiout.max_channels);
+ if (err >= 0 && spec->need_dac_fix)
+ spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+ return err;
+}
+
+/* 6-stack mode */
+static struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
+ HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT),
+
+ { } /* end */
+};
+
+/* 3-stack mode */
+static struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
+ HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = ad198x_ch_mode_info,
+ .get = ad198x_ch_mode_get,
+ .put = ad198x_ch_mode_put,
+ },
+
+ { } /* end */
+};
+
+/* laptop mode */
+static struct snd_kcontrol_new ad1988_laptop_mixers[] = {
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
+ HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost", 0x39, 0x0, HDA_OUTPUT),
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "External Amplifier",
+ .info = ad198x_eapd_info,
+ .get = ad198x_eapd_get,
+ .put = ad198x_eapd_put,
+ .private_value = 0x12 | (1 << 8), /* port-D, inversed */
+ },
+
+ { } /* end */
+};
+
+/* capture */
+static struct snd_kcontrol_new ad1988_capture_mixers[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 3,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {
+ "PCM", "ADC1", "ADC2", "ADC3"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= 4)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int sel;
+
+ sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_INPUT);
+ if (!(sel & 0x80))
+ ucontrol->value.enumerated.item[0] = 0;
+ else {
+ sel = snd_hda_codec_read(codec, 0x0b, 0,
+ AC_VERB_GET_CONNECT_SEL, 0);
+ if (sel < 3)
+ sel++;
+ else
+ sel = 0;
+ ucontrol->value.enumerated.item[0] = sel;
+ }
+ return 0;
+}
+
+static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int val, sel;
+ int change;
+
+ val = ucontrol->value.enumerated.item[0];
+ if (val > 3)
+ return -EINVAL;
+ if (!val) {
+ sel = snd_hda_codec_read(codec, 0x1d, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_INPUT);
+ change = sel & 0x80;
+ if (change) {
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(1));
+ }
+ } else {
+ sel = snd_hda_codec_read(codec, 0x1d, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_INPUT | 0x01);
+ change = sel & 0x80;
+ if (change) {
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(0));
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(1));
+ }
+ sel = snd_hda_codec_read(codec, 0x0b, 0,
+ AC_VERB_GET_CONNECT_SEL, 0) + 1;
+ change |= sel != val;
+ if (change)
+ snd_hda_codec_write_cache(codec, 0x0b, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ val - 1);
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Playback Source",
+ .info = ad1988_spdif_playback_source_info,
+ .get = ad1988_spdif_playback_source_get,
+ .put = ad1988_spdif_playback_source_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
+ HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+
+/*
+ * for 6-stack (+dig)
+ */
+static struct hda_verb ad1988_6stack_init_verbs[] = {
+ /* Front, Surround, CLFE, side DAC; unmute as default */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Port-A front headphon path */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Port-D line-out path */
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Port-F surround path */
+ {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Port-G CLFE path */
+ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Port-H side path */
+ {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Mono out path */
+ {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
+ /* Port-B front mic-in path */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Port-C line-in path */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Port-E mic-in path */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Analog CD Input */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
+
+ { }
+};
+
+static struct hda_verb ad1988_capture_init_verbs[] = {
+ /* mute analog mix */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* select ADCs - front-mic */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* ADCs; muted */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ { }
+};
+
+static struct hda_verb ad1988_spdif_init_verbs[] = {
+ /* SPDIF out sel */
+ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* SPDIF out pin */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+
+ { }
+};
+
+/* AD1989 has no ADC -> SPDIF route */
+static struct hda_verb ad1989_spdif_init_verbs[] = {
+ /* SPDIF-1 out pin */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ /* SPDIF-2/HDMI out pin */
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ { }
+};
+
+/*
+ * verbs for 3stack (+dig)
+ */
+static struct hda_verb ad1988_3stack_ch2_init[] = {
+ /* set port-C to line-in */
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ /* set port-E to mic-in */
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { } /* end */
+};
+
+static struct hda_verb ad1988_3stack_ch6_init[] = {
+ /* set port-C to surround out */
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ /* set port-E to CLFE out */
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static struct hda_channel_mode ad1988_3stack_modes[2] = {
+ { 2, ad1988_3stack_ch2_init },
+ { 6, ad1988_3stack_ch6_init },
+};
+
+static struct hda_verb ad1988_3stack_init_verbs[] = {
+ /* Front, Surround, CLFE, side DAC; unmute as default */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Port-A front headphon path */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Port-D line-out path */
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Mono out path */
+ {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
+ /* Port-B front mic-in path */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Port-C line-in/surround path - 6ch mode as default */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */
+ {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Port-E mic-in/CLFE path - 6ch mode as default */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */
+ {0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* mute analog mix */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* select ADCs - front-mic */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* ADCs; muted */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
+ { }
+};
+
+/*
+ * verbs for laptop mode (+dig)
+ */
+static struct hda_verb ad1988_laptop_hp_on[] = {
+ /* unmute port-A and mute port-D */
+ { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+static struct hda_verb ad1988_laptop_hp_off[] = {
+ /* mute port-A and unmute port-D */
+ { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+#define AD1988_HP_EVENT 0x01
+
+static struct hda_verb ad1988_laptop_init_verbs[] = {
+ /* Front, Surround, CLFE, side DAC; unmute as default */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Port-A front headphon path */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x01}, /* DAC1:04h */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* unsolicited event for pin-sense */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT },
+ /* Port-D line-out path + EAPD */
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */
+ /* Mono out path */
+ {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
+ /* Port-B mic-in path */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Port-C docking station - try to output */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* mute analog mix */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* select ADCs - mic */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* ADCs; muted */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
+ { }
+};
+
+static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) != AD1988_HP_EVENT)
+ return;
+ if (snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) & (1 << 31))
+ snd_hda_sequence_write(codec, ad1988_laptop_hp_on);
+ else
+ snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1988_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Line */
+ { 0x20, HDA_INPUT, 4 }, /* Mic */
+ { 0x20, HDA_INPUT, 6 }, /* CD */
+ { } /* end */
+};
+#endif
+
+/*
+ * Automatic parse of I/O pins from the BIOS configuration
+ */
+
+#define NUM_CONTROL_ALLOC 32
+#define NUM_VERB_ALLOC 32
+
+enum {
+ AD_CTL_WIDGET_VOL,
+ AD_CTL_WIDGET_MUTE,
+ AD_CTL_BIND_MUTE,
+};
+static struct snd_kcontrol_new ad1988_control_templates[] = {
+ HDA_CODEC_VOLUME(NULL, 0, 0, 0),
+ HDA_CODEC_MUTE(NULL, 0, 0, 0),
+ HDA_BIND_MUTE(NULL, 0, 0, 0),
+};
+
+/* add dynamic controls */
+static int add_control(struct ad198x_spec *spec, int type, const char *name,
+ unsigned long val)
+{
+ struct snd_kcontrol_new *knew;
+
+ if (spec->num_kctl_used >= spec->num_kctl_alloc) {
+ int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
+
+ knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
+ if (! knew)
+ return -ENOMEM;
+ if (spec->kctl_alloc) {
+ memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
+ kfree(spec->kctl_alloc);
+ }
+ spec->kctl_alloc = knew;
+ spec->num_kctl_alloc = num;
+ }
+
+ knew = &spec->kctl_alloc[spec->num_kctl_used];
+ *knew = ad1988_control_templates[type];
+ knew->name = kstrdup(name, GFP_KERNEL);
+ if (! knew->name)
+ return -ENOMEM;
+ knew->private_value = val;
+ spec->num_kctl_used++;
+ return 0;
+}
+
+#define AD1988_PIN_CD_NID 0x18
+#define AD1988_PIN_BEEP_NID 0x10
+
+static hda_nid_t ad1988_mixer_nids[8] = {
+ /* A B C D E F G H */
+ 0x22, 0x2b, 0x2c, 0x29, 0x26, 0x2a, 0x27, 0x28
+};
+
+static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx)
+{
+ static hda_nid_t idx_to_dac[8] = {
+ /* A B C D E F G H */
+ 0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a
+ };
+ static hda_nid_t idx_to_dac_rev2[8] = {
+ /* A B C D E F G H */
+ 0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06
+ };
+ if (is_rev2(codec))
+ return idx_to_dac_rev2[idx];
+ else
+ return idx_to_dac[idx];
+}
+
+static hda_nid_t ad1988_boost_nids[8] = {
+ 0x38, 0x39, 0x3a, 0x3d, 0x3c, 0x3b, 0, 0
+};
+
+static int ad1988_pin_idx(hda_nid_t nid)
+{
+ static hda_nid_t ad1988_io_pins[8] = {
+ 0x11, 0x14, 0x15, 0x12, 0x17, 0x16, 0x24, 0x25
+ };
+ int i;
+ for (i = 0; i < ARRAY_SIZE(ad1988_io_pins); i++)
+ if (ad1988_io_pins[i] == nid)
+ return i;
+ return 0; /* should be -1 */
+}
+
+static int ad1988_pin_to_loopback_idx(hda_nid_t nid)
+{
+ static int loopback_idx[8] = {
+ 2, 0, 1, 3, 4, 5, 1, 4
+ };
+ switch (nid) {
+ case AD1988_PIN_CD_NID:
+ return 6;
+ default:
+ return loopback_idx[ad1988_pin_idx(nid)];
+ }
+}
+
+static int ad1988_pin_to_adc_idx(hda_nid_t nid)
+{
+ static int adc_idx[8] = {
+ 0, 1, 2, 8, 4, 3, 6, 7
+ };
+ switch (nid) {
+ case AD1988_PIN_CD_NID:
+ return 5;
+ default:
+ return adc_idx[ad1988_pin_idx(nid)];
+ }
+}
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int ad1988_auto_fill_dac_nids(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct ad198x_spec *spec = codec->spec;
+ int i, idx;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ /* check the pins hardwired to audio widget */
+ for (i = 0; i < cfg->line_outs; i++) {
+ idx = ad1988_pin_idx(cfg->line_out_pins[i]);
+ spec->multiout.dac_nids[i] = ad1988_idx_to_dac(codec, idx);
+ }
+ spec->multiout.num_dacs = cfg->line_outs;
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int ad1988_auto_create_multi_out_ctls(struct ad198x_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
+ hda_nid_t nid;
+ int i, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t dac = spec->multiout.dac_nids[i];
+ if (! dac)
+ continue;
+ nid = ad1988_mixer_nids[ad1988_pin_idx(cfg->line_out_pins[i])];
+ if (i == 2) {
+ /* Center/LFE */
+ err = add_control(spec, AD_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(dac, 1, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, AD_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(dac, 2, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, AD_CTL_BIND_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 2, HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, AD_CTL_BIND_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 2, HDA_INPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = add_control(spec, AD_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = add_control(spec, AD_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+/* add playback controls for speaker and HP outputs */
+static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
+ const char *pfx)
+{
+ struct ad198x_spec *spec = codec->spec;
+ hda_nid_t nid;
+ int i, idx, err;
+ char name[32];
+
+ if (! pin)
+ return 0;
+
+ idx = ad1988_pin_idx(pin);
+ nid = ad1988_idx_to_dac(codec, idx);
+ /* check whether the corresponding DAC was already taken */
+ for (i = 0; i < spec->autocfg.line_outs; i++) {
+ hda_nid_t pin = spec->autocfg.line_out_pins[i];
+ hda_nid_t dac = ad1988_idx_to_dac(codec, ad1988_pin_idx(pin));
+ if (dac == nid)
+ break;
+ }
+ if (i >= spec->autocfg.line_outs) {
+ /* specify the DAC as the extra output */
+ if (!spec->multiout.hp_nid)
+ spec->multiout.hp_nid = nid;
+ else
+ spec->multiout.extra_out_nid[0] = nid;
+ /* control HP volume/switch on the output mixer amp */
+ sprintf(name, "%s Playback Volume", pfx);
+ err = add_control(spec, AD_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ nid = ad1988_mixer_nids[idx];
+ sprintf(name, "%s Playback Switch", pfx);
+ if ((err = add_control(spec, AD_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0)
+ return err;
+ return 0;
+}
+
+/* create input playback/capture controls for the given pin */
+static int new_analog_input(struct ad198x_spec *spec, hda_nid_t pin,
+ const char *ctlname, int boost)
+{
+ char name[32];
+ int err, idx;
+
+ sprintf(name, "%s Playback Volume", ctlname);
+ idx = ad1988_pin_to_loopback_idx(pin);
+ if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(0x20, 3, idx, HDA_INPUT))) < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", ctlname);
+ if ((err = add_control(spec, AD_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(0x20, 3, idx, HDA_INPUT))) < 0)
+ return err;
+ if (boost) {
+ hda_nid_t bnid;
+ idx = ad1988_pin_idx(pin);
+ bnid = ad1988_boost_nids[idx];
+ if (bnid) {
+ sprintf(name, "%s Boost", ctlname);
+ return add_control(spec, AD_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(bnid, 3, idx, HDA_OUTPUT));
+
+ }
+ }
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int ad1988_auto_create_analog_input_ctls(struct ad198x_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct hda_input_mux *imux = &spec->private_imux;
+ int i, err;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ err = new_analog_input(spec, cfg->input_pins[i],
+ auto_pin_cfg_labels[i],
+ i <= AUTO_PIN_FRONT_MIC);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index = ad1988_pin_to_adc_idx(cfg->input_pins[i]);
+ imux->num_items++;
+ }
+ imux->items[imux->num_items].label = "Mix";
+ imux->items[imux->num_items].index = 9;
+ imux->num_items++;
+
+ if ((err = add_control(spec, AD_CTL_WIDGET_VOL,
+ "Analog Mix Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0x0, HDA_OUTPUT))) < 0)
+ return err;
+ if ((err = add_control(spec, AD_CTL_WIDGET_MUTE,
+ "Analog Mix Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0x0, HDA_OUTPUT))) < 0)
+ return err;
+
+ return 0;
+}
+
+static void ad1988_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ /* set as output */
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
+ switch (nid) {
+ case 0x11: /* port-A - DAC 04 */
+ snd_hda_codec_write(codec, 0x37, 0, AC_VERB_SET_CONNECT_SEL, 0x01);
+ break;
+ case 0x14: /* port-B - DAC 06 */
+ snd_hda_codec_write(codec, 0x30, 0, AC_VERB_SET_CONNECT_SEL, 0x02);
+ break;
+ case 0x15: /* port-C - DAC 05 */
+ snd_hda_codec_write(codec, 0x31, 0, AC_VERB_SET_CONNECT_SEL, 0x00);
+ break;
+ case 0x17: /* port-E - DAC 0a */
+ snd_hda_codec_write(codec, 0x32, 0, AC_VERB_SET_CONNECT_SEL, 0x01);
+ break;
+ case 0x13: /* mono - DAC 04 */
+ snd_hda_codec_write(codec, 0x36, 0, AC_VERB_SET_CONNECT_SEL, 0x01);
+ break;
+ }
+}
+
+static void ad1988_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->autocfg.line_outs; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ ad1988_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
+ }
+}
+
+static void ad1988_auto_init_extra_out(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin) /* connect to front */
+ ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front */
+ ad1988_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+}
+
+static void ad1988_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ int i, idx;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (! nid)
+ continue;
+ switch (nid) {
+ case 0x15: /* port-C */
+ snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
+ break;
+ case 0x17: /* port-E */
+ snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
+ break;
+ }
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
+ if (nid != AD1988_PIN_CD_NID)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ idx = ad1988_pin_idx(nid);
+ if (ad1988_boost_nids[idx])
+ snd_hda_codec_write(codec, ad1988_boost_nids[idx], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_ZERO);
+ }
+}
+
+/* parse the BIOS configuration and set up the alc_spec */
+/* return 1 if successful, 0 if the proper config is not found, or a negative error code */
+static int ad1988_parse_auto_config(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ int err;
+
+ if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0)
+ return err;
+ if ((err = ad1988_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
+ return err;
+ if (! spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
+ if ((err = ad1988_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
+ (err = ad1988_auto_create_extra_out(codec,
+ spec->autocfg.speaker_pins[0],
+ "Speaker")) < 0 ||
+ (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
+ "Headphone")) < 0 ||
+ (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = AD1988_SPDIF_IN;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_init_verbs++] = ad1988_6stack_init_verbs;
+
+ spec->input_mux = &spec->private_imux;
+
+ return 1;
+}
+
+/* init callback for auto-configuration model -- overriding the default init */
+static int ad1988_auto_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1988_auto_init_multi_out(codec);
+ ad1988_auto_init_extra_out(codec);
+ ad1988_auto_init_analog_input(codec);
+ return 0;
+}
+
+
+/*
+ */
+
+static const char *ad1988_models[AD1988_MODEL_LAST] = {
+ [AD1988_6STACK] = "6stack",
+ [AD1988_6STACK_DIG] = "6stack-dig",
+ [AD1988_3STACK] = "3stack",
+ [AD1988_3STACK_DIG] = "3stack-dig",
+ [AD1988_LAPTOP] = "laptop",
+ [AD1988_LAPTOP_DIG] = "laptop-dig",
+ [AD1988_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk ad1988_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
+ SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
+ {}
+};
+
+static int patch_ad1988(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ if (is_rev2(codec))
+ snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n");
+
+ board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
+ ad1988_models, ad1988_cfg_tbl);
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: Unknown model for AD1988, trying auto-probe from BIOS...\n");
+ board_config = AD1988_AUTO;
+ }
+
+ if (board_config == AD1988_AUTO) {
+ /* automatic parse from the BIOS config */
+ int err = ad1988_parse_auto_config(codec);
+ if (err < 0) {
+ ad198x_free(codec);
+ return err;
+ } else if (! err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using 6-stack mode...\n");
+ board_config = AD1988_6STACK;
+ }
+ }
+
+ switch (board_config) {
+ case AD1988_6STACK:
+ case AD1988_6STACK_DIG:
+ spec->multiout.max_channels = 8;
+ spec->multiout.num_dacs = 4;
+ if (is_rev2(codec))
+ spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2;
+ else
+ spec->multiout.dac_nids = ad1988_6stack_dac_nids;
+ spec->input_mux = &ad1988_6stack_capture_source;
+ spec->num_mixers = 2;
+ if (is_rev2(codec))
+ spec->mixers[0] = ad1988_6stack_mixers1_rev2;
+ else
+ spec->mixers[0] = ad1988_6stack_mixers1;
+ spec->mixers[1] = ad1988_6stack_mixers2;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1988_6stack_init_verbs;
+ if (board_config == AD1988_6STACK_DIG) {
+ spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
+ spec->dig_in_nid = AD1988_SPDIF_IN;
+ }
+ break;
+ case AD1988_3STACK:
+ case AD1988_3STACK_DIG:
+ spec->multiout.max_channels = 6;
+ spec->multiout.num_dacs = 3;
+ if (is_rev2(codec))
+ spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2;
+ else
+ spec->multiout.dac_nids = ad1988_3stack_dac_nids;
+ spec->input_mux = &ad1988_6stack_capture_source;
+ spec->channel_mode = ad1988_3stack_modes;
+ spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes);
+ spec->num_mixers = 2;
+ if (is_rev2(codec))
+ spec->mixers[0] = ad1988_3stack_mixers1_rev2;
+ else
+ spec->mixers[0] = ad1988_3stack_mixers1;
+ spec->mixers[1] = ad1988_3stack_mixers2;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1988_3stack_init_verbs;
+ if (board_config == AD1988_3STACK_DIG)
+ spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
+ break;
+ case AD1988_LAPTOP:
+ case AD1988_LAPTOP_DIG:
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = ad1988_3stack_dac_nids;
+ spec->input_mux = &ad1988_laptop_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1988_laptop_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1988_laptop_init_verbs;
+ if (board_config == AD1988_LAPTOP_DIG)
+ spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
+ break;
+ }
+
+ spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
+ spec->adc_nids = ad1988_adc_nids;
+ spec->capsrc_nids = ad1988_capsrc_nids;
+ spec->mixers[spec->num_mixers++] = ad1988_capture_mixers;
+ spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs;
+ if (spec->multiout.dig_out_nid) {
+ if (codec->vendor_id >= 0x11d4989a) {
+ spec->mixers[spec->num_mixers++] =
+ ad1989_spdif_out_mixers;
+ spec->init_verbs[spec->num_init_verbs++] =
+ ad1989_spdif_init_verbs;
+ codec->slave_dig_outs = ad1989b_slave_dig_outs;
+ } else {
+ spec->mixers[spec->num_mixers++] =
+ ad1988_spdif_out_mixers;
+ spec->init_verbs[spec->num_init_verbs++] =
+ ad1988_spdif_init_verbs;
+ }
+ }
+ if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a)
+ spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers;
+
+ codec->patch_ops = ad198x_patch_ops;
+ switch (board_config) {
+ case AD1988_AUTO:
+ codec->patch_ops.init = ad1988_auto_init;
+ break;
+ case AD1988_LAPTOP:
+ case AD1988_LAPTOP_DIG:
+ codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
+ break;
+ }
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1988_loopbacks;
+#endif
+ spec->vmaster_nid = 0x04;
+
+ return 0;
+}
+
+
+/*
+ * AD1884 / AD1984
+ *
+ * port-B - front line/mic-in
+ * port-E - aux in/out
+ * port-F - aux in/out
+ * port-C - rear line/mic-in
+ * port-D - rear line/hp-out
+ * port-A - front line/hp-out
+ *
+ * AD1984 = AD1884 + two digital mic-ins
+ *
+ * FIXME:
+ * For simplicity, we share the single DAC for both HP and line-outs
+ * right now. The inidividual playbacks could be easily implemented,
+ * but no build-up framework is given, so far.
+ */
+
+static hda_nid_t ad1884_dac_nids[1] = {
+ 0x04,
+};
+
+static hda_nid_t ad1884_adc_nids[2] = {
+ 0x08, 0x09,
+};
+
+static hda_nid_t ad1884_capsrc_nids[2] = {
+ 0x0c, 0x0d,
+};
+
+#define AD1884_SPDIF_OUT 0x02
+
+static struct hda_input_mux ad1884_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Mic", 0x1 },
+ { "CD", 0x2 },
+ { "Mix", 0x3 },
+ },
+};
+
+static struct snd_kcontrol_new ad1884_base_mixers[] = {
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
+ /*
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ */
+ HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* SPDIF controls */
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ /* identical with ad1983 */
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1984_dmic_mixers[] = {
+ HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
+ HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0,
+ HDA_INPUT),
+ { } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+static struct hda_verb ad1884_init_verbs[] = {
+ /* DACs; mute as default */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Port-A (HP) mixer */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-A pin */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* HP selector - select DAC2 */
+ {0x22, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* Port-D (Line-out) mixer */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-D pin */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mono-out mixer */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Mono-out pin */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mono selector */
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* Port-B (front mic) pin */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-C (rear mic) pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog mixer; mute as default */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
+ /* SPDIF output selector */
+ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1884_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 2 }, /* CD */
+ { 0x20, HDA_INPUT, 4 }, /* Docking */
+ { } /* end */
+};
+#endif
+
+static const char *ad1884_slave_vols[] = {
+ "PCM Playback Volume",
+ "Mic Playback Volume",
+ "Mono Playback Volume",
+ "Front Mic Playback Volume",
+ "Mic Playback Volume",
+ "CD Playback Volume",
+ "Internal Mic Playback Volume",
+ "Docking Mic Playback Volume"
+ "Beep Playback Volume",
+ "IEC958 Playback Volume",
+ NULL
+};
+
+static int patch_ad1884(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
+ spec->multiout.dac_nids = ad1884_dac_nids;
+ spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
+ spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids);
+ spec->adc_nids = ad1884_adc_nids;
+ spec->capsrc_nids = ad1884_capsrc_nids;
+ spec->input_mux = &ad1884_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1884_base_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1884_init_verbs;
+ spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1884_loopbacks;
+#endif
+ spec->vmaster_nid = 0x04;
+ /* we need to cover all playback volumes */
+ spec->slave_vols = ad1884_slave_vols;
+
+ codec->patch_ops = ad198x_patch_ops;
+
+ return 0;
+}
+
+/*
+ * Lenovo Thinkpad T61/X61
+ */
+static struct hda_input_mux ad1984_thinkpad_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "Mix", 0x3 },
+ { "Docking-Station", 0x4 },
+ },
+};
+
+
+/*
+ * Dell Precision T3400
+ */
+static struct hda_input_mux ad1984_dell_desktop_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Line-In", 0x1 },
+ { "Mix", 0x3 },
+ },
+};
+
+
+static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* SPDIF controls */
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ /* identical with ad1983 */
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+/* additional verbs */
+static struct hda_verb ad1984_thinkpad_init_verbs[] = {
+ /* Port-E (docking station mic) pin */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* docking mic boost */
+ {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog mixer - docking mic; mute as default */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* enable EAPD bit */
+ {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ { } /* end */
+};
+
+/*
+ * Dell Precision T3400
+ */
+static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT),
+ /*
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT),
+ */
+ HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* Digial MIC ADC NID 0x05 + 0x06 */
+static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number);
+ return 0;
+}
+
+static struct hda_pcm_stream ad1984_pcm_dmic_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x05,
+ .ops = {
+ .prepare = ad1984_pcm_dmic_prepare,
+ .cleanup = ad1984_pcm_dmic_cleanup
+ },
+};
+
+static int ad1984_build_pcms(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ struct hda_pcm *info;
+ int err;
+
+ err = ad198x_build_pcms(codec);
+ if (err < 0)
+ return err;
+
+ info = spec->pcm_rec + codec->num_pcms;
+ codec->num_pcms++;
+ info->name = "AD1984 Digital Mic";
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture;
+ return 0;
+}
+
+/* models */
+enum {
+ AD1984_BASIC,
+ AD1984_THINKPAD,
+ AD1984_DELL_DESKTOP,
+ AD1984_MODELS
+};
+
+static const char *ad1984_models[AD1984_MODELS] = {
+ [AD1984_BASIC] = "basic",
+ [AD1984_THINKPAD] = "thinkpad",
+ [AD1984_DELL_DESKTOP] = "dell_desktop",
+};
+
+static struct snd_pci_quirk ad1984_cfg_tbl[] = {
+ /* Lenovo Thinkpad T61/X61 */
+ SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD),
+ SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
+ {}
+};
+
+static int patch_ad1984(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+ int board_config, err;
+
+ err = patch_ad1884(codec);
+ if (err < 0)
+ return err;
+ spec = codec->spec;
+ board_config = snd_hda_check_board_config(codec, AD1984_MODELS,
+ ad1984_models, ad1984_cfg_tbl);
+ switch (board_config) {
+ case AD1984_BASIC:
+ /* additional digital mics */
+ spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers;
+ codec->patch_ops.build_pcms = ad1984_build_pcms;
+ break;
+ case AD1984_THINKPAD:
+ spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
+ spec->input_mux = &ad1984_thinkpad_capture_source;
+ spec->mixers[0] = ad1984_thinkpad_mixers;
+ spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;
+ break;
+ case AD1984_DELL_DESKTOP:
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1984_dell_desktop_capture_source;
+ spec->mixers[0] = ad1984_dell_desktop_mixers;
+ break;
+ }
+ return 0;
+}
+
+
+/*
+ * AD1883 / AD1884A / AD1984A / AD1984B
+ *
+ * port-B (0x14) - front mic-in
+ * port-E (0x1c) - rear mic-in
+ * port-F (0x16) - CD / ext out
+ * port-C (0x15) - rear line-in
+ * port-D (0x12) - rear line-out
+ * port-A (0x11) - front hp-out
+ *
+ * AD1984A = AD1884A + digital-mic
+ * AD1883 = equivalent with AD1984A
+ * AD1984B = AD1984A + extra SPDIF-out
+ *
+ * FIXME:
+ * We share the single DAC for both HP and line-outs (see AD1884/1984).
+ */
+
+static hda_nid_t ad1884a_dac_nids[1] = {
+ 0x03,
+};
+
+#define ad1884a_adc_nids ad1884_adc_nids
+#define ad1884a_capsrc_nids ad1884_capsrc_nids
+
+#define AD1884A_SPDIF_OUT 0x02
+
+static struct hda_input_mux ad1884a_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Mic", 0x4 },
+ { "Line", 0x1 },
+ { "CD", 0x2 },
+ { "Mix", 0x3 },
+ },
+};
+
+static struct snd_kcontrol_new ad1884a_base_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* SPDIF controls */
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ /* identical with ad1983 */
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+static struct hda_verb ad1884a_init_verbs[] = {
+ /* DACs; unmute as default */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ /* Port-A (HP) mixer - route only from analog mixer */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-A pin */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-D (Line-out) mixer - route only from analog mixer */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-D pin */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mono-out mixer - route only from analog mixer */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Mono-out pin */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-B (front mic) pin */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-C (rear line-in) pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-E (rear mic) pin */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */
+ /* Port-F (CD) pin */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog mixer; mute as default */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* capture sources */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* SPDIF output amp */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1884a_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 2 }, /* CD */
+ { 0x20, HDA_INPUT, 4 }, /* Docking */
+ { } /* end */
+};
+#endif
+
+/*
+ * Laptop model
+ *
+ * Port A: Headphone jack
+ * Port B: MIC jack
+ * Port C: Internal MIC
+ * Port D: Dock Line Out (if enabled)
+ * Port E: Dock Line In (if enabled)
+ * Port F: Internal speakers
+ */
+
+static struct hda_input_mux ad1884a_laptop_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 }, /* port-B */
+ { "Internal Mic", 0x1 }, /* port-C */
+ { "Dock Mic", 0x4 }, /* port-E */
+ { "Mix", 0x3 },
+ },
+};
+
+static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/* mute internal speaker if HP is plugged */
+static void ad1884a_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x11, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ present ? 0x00 : 0x02);
+}
+
+/* switch to external mic if plugged */
+static void ad1884a_hp_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 0 : 1);
+}
+
+#define AD1884A_HP_EVENT 0x37
+#define AD1884A_MIC_EVENT 0x36
+
+/* unsolicited event for HP jack sensing */
+static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ switch (res >> 26) {
+ case AD1884A_HP_EVENT:
+ ad1884a_hp_automute(codec);
+ break;
+ case AD1884A_MIC_EVENT:
+ ad1884a_hp_automic(codec);
+ break;
+ }
+}
+
+/* initialize jack-sensing, too */
+static int ad1884a_hp_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1884a_hp_automute(codec);
+ ad1884a_hp_automic(codec);
+ return 0;
+}
+
+/* additional verbs for laptop model */
+static struct hda_verb ad1884a_laptop_verbs[] = {
+ /* Port-A (HP) pin - always unmuted */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Port-F (int speaker) mixer - route only from analog mixer */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-F pin */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-C pin - internal mic-in */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+ /* analog mix */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* unsolicited event for pin-sense */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+ { } /* end */
+};
+
+/*
+ * Thinkpad X300
+ * 0x11 - HP
+ * 0x12 - speaker
+ * 0x14 - mic-in
+ * 0x17 - built-in mic
+ */
+
+static struct hda_verb ad1984a_thinkpad_verbs[] = {
+ /* HP unmute */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* analog mix */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* turn on EAPD */
+ {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ /* unsolicited event for pin-sense */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ /* internal mic - dmic */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* set magic COEFs for dmic */
+ {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+ {0x01, AC_VERB_SET_PROC_COEF, 0x08},
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_input_mux ad1984a_thinkpad_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x5 },
+ { "Mix", 0x3 },
+ },
+};
+
+/* mute internal speaker if HP is plugged */
+static void ad1984a_thinkpad_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+/* unsolicited event for HP jack sensing */
+static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != AD1884A_HP_EVENT)
+ return;
+ ad1984a_thinkpad_automute(codec);
+}
+
+/* initialize jack-sensing, too */
+static int ad1984a_thinkpad_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1984a_thinkpad_automute(codec);
+ return 0;
+}
+
+/*
+ */
+
+enum {
+ AD1884A_DESKTOP,
+ AD1884A_LAPTOP,
+ AD1884A_MOBILE,
+ AD1884A_THINKPAD,
+ AD1884A_MODELS
+};
+
+static const char *ad1884a_models[AD1884A_MODELS] = {
+ [AD1884A_DESKTOP] = "desktop",
+ [AD1884A_LAPTOP] = "laptop",
+ [AD1884A_MOBILE] = "mobile",
+ [AD1884A_THINKPAD] = "thinkpad",
+};
+
+static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
+ SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
+ SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
+ SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
+ SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
+ SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
+ {}
+};
+
+static int patch_ad1884a(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
+ spec->multiout.dac_nids = ad1884a_dac_nids;
+ spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT;
+ spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids);
+ spec->adc_nids = ad1884a_adc_nids;
+ spec->capsrc_nids = ad1884a_capsrc_nids;
+ spec->input_mux = &ad1884a_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1884a_base_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1884a_init_verbs;
+ spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1884a_loopbacks;
+#endif
+ codec->patch_ops = ad198x_patch_ops;
+
+ /* override some parameters */
+ board_config = snd_hda_check_board_config(codec, AD1884A_MODELS,
+ ad1884a_models,
+ ad1884a_cfg_tbl);
+ switch (board_config) {
+ case AD1884A_LAPTOP:
+ spec->mixers[0] = ad1884a_laptop_mixers;
+ spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1884a_laptop_capture_source;
+ codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
+ codec->patch_ops.init = ad1884a_hp_init;
+ break;
+ case AD1884A_MOBILE:
+ spec->mixers[0] = ad1884a_mobile_mixers;
+ spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
+ spec->multiout.dig_out_nid = 0;
+ codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
+ codec->patch_ops.init = ad1884a_hp_init;
+ break;
+ case AD1884A_THINKPAD:
+ spec->mixers[0] = ad1984a_thinkpad_mixers;
+ spec->init_verbs[spec->num_init_verbs++] =
+ ad1984a_thinkpad_verbs;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1984a_thinkpad_capture_source;
+ codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
+ codec->patch_ops.init = ad1984a_thinkpad_init;
+ break;
+ }
+
+ return 0;
+}
+
+
+/*
+ * AD1882 / AD1882A
+ *
+ * port-A - front hp-out
+ * port-B - front mic-in
+ * port-C - rear line-in, shared surr-out (3stack)
+ * port-D - rear line-out
+ * port-E - rear mic-in, shared clfe-out (3stack)
+ * port-F - rear surr-out (6stack)
+ * port-G - rear clfe-out (6stack)
+ */
+
+static hda_nid_t ad1882_dac_nids[3] = {
+ 0x04, 0x03, 0x05
+};
+
+static hda_nid_t ad1882_adc_nids[2] = {
+ 0x08, 0x09,
+};
+
+static hda_nid_t ad1882_capsrc_nids[2] = {
+ 0x0c, 0x0d,
+};
+
+#define AD1882_SPDIF_OUT 0x02
+
+/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
+static struct hda_input_mux ad1882_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Front Mic", 0x1 },
+ { "Mic", 0x4 },
+ { "Line", 0x2 },
+ { "CD", 0x3 },
+ { "Mix", 0x7 },
+ },
+};
+
+/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
+static struct hda_input_mux ad1882a_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Front Mic", 0x1 },
+ { "Mic", 0x4},
+ { "Line", 0x2 },
+ { "Digital Mic", 0x06 },
+ { "Mix", 0x7 },
+ },
+};
+
+static struct snd_kcontrol_new ad1882_base_mixers[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line-In Boost", 0x3a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* SPDIF controls */
+ HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ /* identical with ad1983 */
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1882_loopback_mixers[] = {
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
+ HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1882_3stack_mixers[] = {
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = ad198x_ch_mode_info,
+ .get = ad198x_ch_mode_get,
+ .put = ad198x_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1882_6stack_mixers[] = {
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct hda_verb ad1882_ch2_init[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { } /* end */
+};
+
+static struct hda_verb ad1882_ch4_init[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { } /* end */
+};
+
+static struct hda_verb ad1882_ch6_init[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ { } /* end */
+};
+
+static struct hda_channel_mode ad1882_modes[3] = {
+ { 2, ad1882_ch2_init },
+ { 4, ad1882_ch4_init },
+ { 6, ad1882_ch6_init },
+};
+
+/*
+ * initialization verbs
+ */
+static struct hda_verb ad1882_init_verbs[] = {
+ /* DACs; mute as default */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Port-A (HP) mixer */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-A pin */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* HP selector - select DAC2 */
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* Port-D (Line-out) mixer */
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-D pin */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mono-out mixer */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Mono-out pin */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-B (front mic) pin */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
+ /* Port-C (line-in) pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
+ /* Port-C mixer - mute as input */
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Port-E (mic-in) pin */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
+ /* Port-E mixer - mute as input */
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Port-F (surround) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Port-G (CLFE) */
+ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog mixer; mute as default */
+ /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
+ /* SPDIF output selector */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1882_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 4 }, /* Line */
+ { 0x20, HDA_INPUT, 6 }, /* CD */
+ { } /* end */
+};
+#endif
+
+/* models */
+enum {
+ AD1882_3STACK,
+ AD1882_6STACK,
+ AD1882_MODELS
+};
+
+static const char *ad1882_models[AD1986A_MODELS] = {
+ [AD1882_3STACK] = "3stack",
+ [AD1882_6STACK] = "6stack",
+};
+
+
+static int patch_ad1882(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 6;
+ spec->multiout.num_dacs = 3;
+ spec->multiout.dac_nids = ad1882_dac_nids;
+ spec->multiout.dig_out_nid = AD1882_SPDIF_OUT;
+ spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
+ spec->adc_nids = ad1882_adc_nids;
+ spec->capsrc_nids = ad1882_capsrc_nids;
+ if (codec->vendor_id == 0x11d41882)
+ spec->input_mux = &ad1882_capture_source;
+ else
+ spec->input_mux = &ad1882a_capture_source;
+ spec->num_mixers = 2;
+ spec->mixers[0] = ad1882_base_mixers;
+ if (codec->vendor_id == 0x11d41882)
+ spec->mixers[1] = ad1882_loopback_mixers;
+ else
+ spec->mixers[1] = ad1882a_loopback_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1882_init_verbs;
+ spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1882_loopbacks;
+#endif
+ spec->vmaster_nid = 0x04;
+
+ codec->patch_ops = ad198x_patch_ops;
+
+ /* override some parameters */
+ board_config = snd_hda_check_board_config(codec, AD1882_MODELS,
+ ad1882_models, NULL);
+ switch (board_config) {
+ default:
+ case AD1882_3STACK:
+ spec->num_mixers = 3;
+ spec->mixers[2] = ad1882_3stack_mixers;
+ spec->channel_mode = ad1882_modes;
+ spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
+ spec->need_dac_fix = 1;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ break;
+ case AD1882_6STACK:
+ spec->num_mixers = 3;
+ spec->mixers[2] = ad1882_6stack_mixers;
+ break;
+ }
+ return 0;
+}
+
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_analog[] = {
+ { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
+ { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
+ { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
+ { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
+ { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a },
+ { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a },
+ { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
+ { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
+ { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
+ { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
+ { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
+ { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
+ { .id = 0x11d4882a, .name = "AD1882A", .patch = patch_ad1882 },
+ { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 },
+ { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
+ {} /* terminator */
+};
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
new file mode 100644
index 0000000..ba61575
--- /dev/null
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -0,0 +1,199 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for ATI HDMI codecs
+ *
+ * Copyright (c) 2006 ATI Technologies Inc.
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+
+struct atihdmi_spec {
+ struct hda_multi_out multiout;
+
+ struct hda_pcm pcm_rec;
+};
+
+#define CVT_NID 0x02 /* audio converter */
+#define PIN_NID 0x03 /* HDMI output pin */
+
+static struct hda_verb atihdmi_basic_init[] = {
+ /* enable digital output on pin widget */
+ { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ {} /* terminator */
+};
+
+/*
+ * Controls
+ */
+static int atihdmi_build_controls(struct hda_codec *codec)
+{
+ struct atihdmi_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int atihdmi_init(struct hda_codec *codec)
+{
+ snd_hda_sequence_write(codec, atihdmi_basic_init);
+ /* SI codec requires to unmute the pin */
+ if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, PIN_NID, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
+ return 0;
+}
+
+/*
+ * Digital out
+ */
+static int atihdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct atihdmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int atihdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct atihdmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int atihdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct atihdmi_spec *spec = codec->spec;
+ int chans = substream->runtime->channels;
+ int i, err;
+
+ err = snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+ if (err < 0)
+ return err;
+ snd_hda_codec_write(codec, CVT_NID, 0, AC_VERB_SET_CVT_CHAN_COUNT,
+ chans - 1);
+ /* FIXME: XXX */
+ for (i = 0; i < chans; i++) {
+ snd_hda_codec_write(codec, CVT_NID, 0,
+ AC_VERB_SET_HDMI_CHAN_SLOT,
+ (i << 4) | i);
+ }
+ return 0;
+}
+
+static struct hda_pcm_stream atihdmi_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = CVT_NID, /* NID to query formats and rates and setup streams */
+ .ops = {
+ .open = atihdmi_dig_playback_pcm_open,
+ .close = atihdmi_dig_playback_pcm_close,
+ .prepare = atihdmi_dig_playback_pcm_prepare
+ },
+};
+
+static int atihdmi_build_pcms(struct hda_codec *codec)
+{
+ struct atihdmi_spec *spec = codec->spec;
+ struct hda_pcm *info = &spec->pcm_rec;
+ unsigned int chans;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "ATI HDMI";
+ info->pcm_type = HDA_PCM_TYPE_HDMI;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback;
+
+ /* FIXME: we must check ELD and change the PCM parameters dynamically
+ */
+ chans = get_wcaps(codec, CVT_NID);
+ chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13;
+ chans = ((chans << 1) | 1) + 1;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans;
+
+ return 0;
+}
+
+static void atihdmi_free(struct hda_codec *codec)
+{
+ kfree(codec->spec);
+}
+
+static struct hda_codec_ops atihdmi_patch_ops = {
+ .build_controls = atihdmi_build_controls,
+ .build_pcms = atihdmi_build_pcms,
+ .init = atihdmi_init,
+ .free = atihdmi_free,
+};
+
+static int patch_atihdmi(struct hda_codec *codec)
+{
+ struct atihdmi_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.num_dacs = 0; /* no analog */
+ spec->multiout.max_channels = 2;
+ /* NID for copying analog to digital,
+ * seems to be unused in pure-digital
+ * case.
+ */
+ spec->multiout.dig_out_nid = CVT_NID;
+
+ codec->patch_ops = atihdmi_patch_ops;
+
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_atihdmi[] = {
+ { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
+ { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
+ { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi },
+ { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi },
+ { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi },
+ { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi },
+ { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi },
+ {} /* terminator */
+};
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
new file mode 100644
index 0000000..6ef57fb
--- /dev/null
+++ b/sound/pci/hda/patch_cmedia.c
@@ -0,0 +1,743 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for C-Media CMI9880
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+#define NUM_PINS 11
+
+
+/* board config type */
+enum {
+ CMI_MINIMAL, /* back 3-jack */
+ CMI_MIN_FP, /* back 3-jack + front-panel 2-jack */
+ CMI_FULL, /* back 6-jack + front-panel 2-jack */
+ CMI_FULL_DIG, /* back 6-jack + front-panel 2-jack + digital I/O */
+ CMI_ALLOUT, /* back 5-jack + front-panel 2-jack + digital out */
+ CMI_AUTO, /* let driver guess it */
+ CMI_MODELS
+};
+
+struct cmi_spec {
+ int board_config;
+ unsigned int no_line_in: 1; /* no line-in (5-jack) */
+ unsigned int front_panel: 1; /* has front-panel 2-jack */
+
+ /* playback */
+ struct hda_multi_out multiout;
+ hda_nid_t dac_nids[AUTO_CFG_MAX_OUTS]; /* NID for each DAC */
+ int num_dacs;
+
+ /* capture */
+ hda_nid_t *adc_nids;
+ hda_nid_t dig_in_nid;
+
+ /* capture source */
+ const struct hda_input_mux *input_mux;
+ unsigned int cur_mux[2];
+
+ /* channel mode */
+ int num_channel_modes;
+ const struct hda_channel_mode *channel_modes;
+
+ struct hda_pcm pcm_rec[2]; /* PCM information */
+
+ /* pin deafault configuration */
+ hda_nid_t pin_nid[NUM_PINS];
+ unsigned int def_conf[NUM_PINS];
+ unsigned int pin_def_confs;
+
+ /* multichannel pins */
+ struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */
+};
+
+/*
+ * input MUX
+ */
+static int cmi_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int cmi_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cmi_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ return 0;
+}
+
+static int cmi_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cmi_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
+}
+
+/*
+ * shared line-in, mic for surrounds
+ */
+
+/* 3-stack / 2 channel */
+static struct hda_verb cmi9880_ch2_init[] = {
+ /* set line-in PIN for input */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ /* set mic PIN for input, also enable vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ /* route front PCM (DAC1) to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ {}
+};
+
+/* 3-stack / 6 channel */
+static struct hda_verb cmi9880_ch6_init[] = {
+ /* set line-in PIN for output */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ /* set mic PIN for output */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ /* route front PCM (DAC1) to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ {}
+};
+
+/* 3-stack+front / 8 channel */
+static struct hda_verb cmi9880_ch8_init[] = {
+ /* set line-in PIN for output */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ /* set mic PIN for output */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ /* route rear-surround PCM (DAC4) to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ {}
+};
+
+static struct hda_channel_mode cmi9880_channel_modes[3] = {
+ { 2, cmi9880_ch2_init },
+ { 6, cmi9880_ch6_init },
+ { 8, cmi9880_ch8_init },
+};
+
+static int cmi_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_ch_mode_info(codec, uinfo, spec->channel_modes,
+ spec->num_channel_modes);
+}
+
+static int cmi_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_modes,
+ spec->num_channel_modes, spec->multiout.max_channels);
+}
+
+static int cmi_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_modes,
+ spec->num_channel_modes, &spec->multiout.max_channels);
+}
+
+/*
+ */
+static struct snd_kcontrol_new cmi9880_basic_mixer[] = {
+ /* CMI9880 has no playback volumes! */
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Side Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = cmi_mux_enum_info,
+ .get = cmi_mux_enum_get,
+ .put = cmi_mux_enum_put,
+ },
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x23, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x23, 0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/*
+ * shared I/O pins
+ */
+static struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = cmi_ch_mode_info,
+ .get = cmi_ch_mode_get,
+ .put = cmi_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/* AUD-in selections:
+ * 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x1f 0x20
+ */
+static struct hda_input_mux cmi9880_basic_mux = {
+ .num_items = 4,
+ .items = {
+ { "Front Mic", 0x5 },
+ { "Rear Mic", 0x2 },
+ { "Line", 0x1 },
+ { "CD", 0x7 },
+ }
+};
+
+static struct hda_input_mux cmi9880_no_line_mux = {
+ .num_items = 3,
+ .items = {
+ { "Front Mic", 0x5 },
+ { "Rear Mic", 0x2 },
+ { "CD", 0x7 },
+ }
+};
+
+/* front, rear, clfe, rear_surr */
+static hda_nid_t cmi9880_dac_nids[4] = {
+ 0x03, 0x04, 0x05, 0x06
+};
+/* ADC0, ADC1 */
+static hda_nid_t cmi9880_adc_nids[2] = {
+ 0x08, 0x09
+};
+
+#define CMI_DIG_OUT_NID 0x07
+#define CMI_DIG_IN_NID 0x0a
+
+/*
+ */
+static struct hda_verb cmi9880_basic_init[] = {
+ /* port-D for line out (rear panel) */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ /* port-E for HP out (front panel) */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ /* route front mic to ADC1/2 */
+ { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
+ { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
+ {} /* terminator */
+};
+
+static struct hda_verb cmi9880_allout_init[] = {
+ /* port-D for line out (rear panel) */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ /* port-E for HP out (front panel) */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-A for side (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ /* port-C for surround (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ /* route front mic to ADC1/2 */
+ { 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
+ { 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
+ {} /* terminator */
+};
+
+/*
+ */
+static int cmi9880_build_controls(struct hda_codec *codec)
+{
+ struct cmi_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer);
+ if (err < 0)
+ return err;
+ if (spec->channel_modes) {
+ err = snd_hda_add_new_ctls(codec, cmi9880_ch_mode_mixer);
+ if (err < 0)
+ return err;
+ }
+ if (spec->multiout.dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
+ }
+ if (spec->dig_in_nid) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* fill in the multi_dac_nids table, which will decide
+ which audio widget to use for each channel */
+static int cmi9880_fill_multi_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+{
+ struct cmi_spec *spec = codec->spec;
+ hda_nid_t nid;
+ int assigned[4];
+ int i, j;
+
+ /* clear the table, only one c-media dac assumed here */
+ memset(spec->dac_nids, 0, sizeof(spec->dac_nids));
+ memset(assigned, 0, sizeof(assigned));
+ /* check the pins we found */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ /* nid 0x0b~0x0e is hardwired to audio widget 0x3~0x6 */
+ if (nid >= 0x0b && nid <= 0x0e) {
+ spec->dac_nids[i] = (nid - 0x0b) + 0x03;
+ assigned[nid - 0x0b] = 1;
+ }
+ }
+ /* left pin can be connect to any audio widget */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid <= 0x0e)
+ continue;
+ /* search for an empty channel */
+ for (j = 0; j < cfg->line_outs; j++) {
+ if (! assigned[j]) {
+ spec->dac_nids[i] = j + 0x03;
+ assigned[j] = 1;
+ break;
+ }
+ }
+ }
+ spec->num_dacs = cfg->line_outs;
+ return 0;
+}
+
+/* create multi_init table, which is used for multichannel initialization */
+static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+{
+ struct cmi_spec *spec = codec->spec;
+ hda_nid_t nid;
+ int i, j, k, len;
+
+ /* clear the table, only one c-media dac assumed here */
+ memset(spec->multi_init, 0, sizeof(spec->multi_init));
+ for (j = 0, i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t conn[4];
+ nid = cfg->line_out_pins[i];
+ /* set as output */
+ spec->multi_init[j].nid = nid;
+ spec->multi_init[j].verb = AC_VERB_SET_PIN_WIDGET_CONTROL;
+ spec->multi_init[j].param = PIN_OUT;
+ j++;
+ if (nid > 0x0e) {
+ /* set connection */
+ spec->multi_init[j].nid = nid;
+ spec->multi_init[j].verb = AC_VERB_SET_CONNECT_SEL;
+ spec->multi_init[j].param = 0;
+ /* find the index in connect list */
+ len = snd_hda_get_connections(codec, nid, conn, 4);
+ for (k = 0; k < len; k++)
+ if (conn[k] == spec->dac_nids[i]) {
+ spec->multi_init[j].param = k;
+ break;
+ }
+ j++;
+ }
+ }
+ return 0;
+}
+
+static int cmi9880_init(struct hda_codec *codec)
+{
+ struct cmi_spec *spec = codec->spec;
+ if (spec->board_config == CMI_ALLOUT)
+ snd_hda_sequence_write(codec, cmi9880_allout_init);
+ else
+ snd_hda_sequence_write(codec, cmi9880_basic_init);
+ if (spec->board_config == CMI_AUTO)
+ snd_hda_sequence_write(codec, spec->multi_init);
+ return 0;
+}
+
+/*
+ * Analog playback callbacks
+ */
+static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+static int cmi9880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int cmi9880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int cmi9880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int cmi9880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+/*
+ * Analog capture
+ */
+static int cmi9880_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cmi_spec *spec = codec->spec;
+
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
+ return 0;
+}
+
+
+/*
+ */
+static struct hda_pcm_stream cmi9880_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 8,
+ .nid = 0x03, /* NID to query formats and rates */
+ .ops = {
+ .open = cmi9880_playback_pcm_open,
+ .prepare = cmi9880_playback_pcm_prepare,
+ .cleanup = cmi9880_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cmi9880_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x08, /* NID to query formats and rates */
+ .ops = {
+ .prepare = cmi9880_capture_pcm_prepare,
+ .cleanup = cmi9880_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cmi9880_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in cmi9880_build_pcms */
+ .ops = {
+ .open = cmi9880_dig_playback_pcm_open,
+ .close = cmi9880_dig_playback_pcm_close,
+ .prepare = cmi9880_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream cmi9880_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in cmi9880_build_pcms */
+};
+
+static int cmi9880_build_pcms(struct hda_codec *codec)
+{
+ struct cmi_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "CMI9880";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_analog_capture;
+
+ if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+ codec->num_pcms++;
+ info++;
+ info->name = "CMI9880 Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->multiout.dig_out_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+ }
+ if (spec->dig_in_nid) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+ }
+ }
+
+ return 0;
+}
+
+static void cmi9880_free(struct hda_codec *codec)
+{
+ kfree(codec->spec);
+}
+
+/*
+ */
+
+static const char *cmi9880_models[CMI_MODELS] = {
+ [CMI_MINIMAL] = "minimal",
+ [CMI_MIN_FP] = "min_fp",
+ [CMI_FULL] = "full",
+ [CMI_FULL_DIG] = "full_dig",
+ [CMI_ALLOUT] = "allout",
+ [CMI_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk cmi9880_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
+ SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL),
+ SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
+ {} /* terminator */
+};
+
+static struct hda_codec_ops cmi9880_patch_ops = {
+ .build_controls = cmi9880_build_controls,
+ .build_pcms = cmi9880_build_pcms,
+ .init = cmi9880_init,
+ .free = cmi9880_free,
+};
+
+static int patch_cmi9880(struct hda_codec *codec)
+{
+ struct cmi_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ spec->board_config = snd_hda_check_board_config(codec, CMI_MODELS,
+ cmi9880_models,
+ cmi9880_cfg_tbl);
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for CMI9880\n");
+ spec->board_config = CMI_AUTO; /* try everything */
+ }
+
+ /* copy default DAC NIDs */
+ memcpy(spec->dac_nids, cmi9880_dac_nids, sizeof(spec->dac_nids));
+ spec->num_dacs = 4;
+
+ switch (spec->board_config) {
+ case CMI_MINIMAL:
+ case CMI_MIN_FP:
+ spec->channel_modes = cmi9880_channel_modes;
+ if (spec->board_config == CMI_MINIMAL)
+ spec->num_channel_modes = 2;
+ else {
+ spec->front_panel = 1;
+ spec->num_channel_modes = 3;
+ }
+ spec->multiout.max_channels = cmi9880_channel_modes[0].channels;
+ spec->input_mux = &cmi9880_basic_mux;
+ break;
+ case CMI_FULL:
+ case CMI_FULL_DIG:
+ spec->front_panel = 1;
+ spec->multiout.max_channels = 8;
+ spec->input_mux = &cmi9880_basic_mux;
+ if (spec->board_config == CMI_FULL_DIG) {
+ spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
+ spec->dig_in_nid = CMI_DIG_IN_NID;
+ }
+ break;
+ case CMI_ALLOUT:
+ spec->front_panel = 1;
+ spec->multiout.max_channels = 8;
+ spec->no_line_in = 1;
+ spec->input_mux = &cmi9880_no_line_mux;
+ spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
+ break;
+ case CMI_AUTO:
+ {
+ unsigned int port_e, port_f, port_g, port_h;
+ unsigned int port_spdifi, port_spdifo;
+ struct auto_pin_cfg cfg;
+
+ /* collect pin default configuration */
+ port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ spec->front_panel = 1;
+ if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE ||
+ get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) {
+ port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ spec->channel_modes = cmi9880_channel_modes;
+ /* no front panel */
+ if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE ||
+ get_defcfg_connect(port_h) == AC_JACK_PORT_NONE) {
+ /* no optional rear panel */
+ spec->board_config = CMI_MINIMAL;
+ spec->front_panel = 0;
+ spec->num_channel_modes = 2;
+ } else {
+ spec->board_config = CMI_MIN_FP;
+ spec->num_channel_modes = 3;
+ }
+ spec->input_mux = &cmi9880_basic_mux;
+ spec->multiout.max_channels = cmi9880_channel_modes[0].channels;
+ } else {
+ spec->input_mux = &cmi9880_basic_mux;
+ port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE)
+ spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
+ if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE)
+ spec->dig_in_nid = CMI_DIG_IN_NID;
+ spec->multiout.max_channels = 8;
+ }
+ snd_hda_parse_pin_def_config(codec, &cfg, NULL);
+ if (cfg.line_outs) {
+ spec->multiout.max_channels = cfg.line_outs * 2;
+ cmi9880_fill_multi_dac_nids(codec, &cfg);
+ cmi9880_fill_multi_init(codec, &cfg);
+ } else
+ snd_printd("patch_cmedia: cannot detect association in defcfg\n");
+ break;
+ }
+ }
+
+ spec->multiout.num_dacs = spec->num_dacs;
+ spec->multiout.dac_nids = spec->dac_nids;
+
+ spec->adc_nids = cmi9880_adc_nids;
+
+ codec->patch_ops = cmi9880_patch_ops;
+
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_cmedia[] = {
+ { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
+ { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
+ {} /* terminator */
+};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
new file mode 100644
index 0000000..a50089f
--- /dev/null
+++ b/sound/pci/hda/patch_conexant.c
@@ -0,0 +1,1794 @@
+/*
+ * HD audio interface patch for Conexant HDA audio codec
+ *
+ * Copyright (c) 2006 Pototskiy Akex <alex.pototskiy@gmail.com>
+ * Takashi Iwai <tiwai@suse.de>
+ * Tobin Davis <tdavis@dsl-only.net>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+
+#define CXT_PIN_DIR_IN 0x00
+#define CXT_PIN_DIR_OUT 0x01
+#define CXT_PIN_DIR_INOUT 0x02
+#define CXT_PIN_DIR_IN_NOMICBIAS 0x03
+#define CXT_PIN_DIR_INOUT_NOMICBIAS 0x04
+
+#define CONEXANT_HP_EVENT 0x37
+#define CONEXANT_MIC_EVENT 0x38
+
+
+
+struct conexant_spec {
+
+ struct snd_kcontrol_new *mixers[5];
+ int num_mixers;
+
+ const struct hda_verb *init_verbs[5]; /* initialization verbs
+ * don't forget NULL
+ * termination!
+ */
+ unsigned int num_init_verbs;
+
+ /* playback */
+ struct hda_multi_out multiout; /* playback set-up
+ * max_channels, dacs must be set
+ * dig_out_nid and hp_nid are optional
+ */
+ unsigned int cur_eapd;
+ unsigned int hp_present;
+ unsigned int need_dac_fix;
+
+ /* capture */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ hda_nid_t dig_in_nid; /* digital-in NID; optional */
+
+ unsigned int cur_adc_idx;
+ hda_nid_t cur_adc;
+ unsigned int cur_adc_stream_tag;
+ unsigned int cur_adc_format;
+
+ /* capture source */
+ const struct hda_input_mux *input_mux;
+ hda_nid_t *capsrc_nids;
+ unsigned int cur_mux[3];
+
+ /* channel model */
+ const struct hda_channel_mode *channel_mode;
+ int num_channel_mode;
+
+ /* PCM information */
+ struct hda_pcm pcm_rec[2]; /* used in build_pcms() */
+
+ unsigned int spdif_route;
+
+ /* dynamic controls, init_verbs and input_mux */
+ struct auto_pin_cfg autocfg;
+ unsigned int num_kctl_alloc, num_kctl_used;
+ struct snd_kcontrol_new *kctl_alloc;
+ struct hda_input_mux private_imux;
+ hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+
+};
+
+static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int conexant_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag,
+ format, substream);
+}
+
+static int conexant_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int conexant_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int conexant_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int conexant_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+ stream_tag,
+ format, substream);
+}
+
+/*
+ * Analog capture
+ */
+static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
+ return 0;
+}
+
+
+
+static struct hda_pcm_stream conexant_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .open = conexant_playback_pcm_open,
+ .prepare = conexant_playback_pcm_prepare,
+ .cleanup = conexant_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream conexant_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .prepare = conexant_capture_pcm_prepare,
+ .cleanup = conexant_capture_pcm_cleanup
+ },
+};
+
+
+static struct hda_pcm_stream conexant_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .open = conexant_dig_playback_pcm_open,
+ .close = conexant_dig_playback_pcm_close,
+ .prepare = conexant_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream conexant_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
+static int cx5051_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ spec->cur_adc = spec->adc_nids[spec->cur_adc_idx];
+ spec->cur_adc_stream_tag = stream_tag;
+ spec->cur_adc_format = format;
+ snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+ return 0;
+}
+
+static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct conexant_spec *spec = codec->spec;
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = 0;
+ return 0;
+}
+
+static struct hda_pcm_stream cx5051_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .prepare = cx5051_capture_pcm_prepare,
+ .cleanup = cx5051_capture_pcm_cleanup
+ },
+};
+
+static int conexant_build_pcms(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "CONEXANT Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = conexant_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+ spec->multiout.max_channels;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dac_nids[0];
+ if (codec->vendor_id == 0x14f15051)
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ cx5051_pcm_analog_capture;
+ else
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ conexant_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+
+ if (spec->multiout.dig_out_nid) {
+ info++;
+ codec->num_pcms++;
+ info->name = "Conexant Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ conexant_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dig_out_nid;
+ if (spec->dig_in_nid) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ conexant_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+ spec->dig_in_nid;
+ }
+ }
+
+ return 0;
+}
+
+static int conexant_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+
+ return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int conexant_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ return 0;
+}
+
+static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ spec->capsrc_nids[adc_idx],
+ &spec->cur_mux[adc_idx]);
+}
+
+static int conexant_init(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_init_verbs; i++)
+ snd_hda_sequence_write(codec, spec->init_verbs[i]);
+ return 0;
+}
+
+static void conexant_free(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int i;
+
+ if (spec->kctl_alloc) {
+ for (i = 0; i < spec->num_kctl_used; i++)
+ kfree(spec->kctl_alloc[i].name);
+ kfree(spec->kctl_alloc);
+ }
+
+ kfree(codec->spec);
+}
+
+static int conexant_build_controls(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < spec->num_mixers; i++) {
+ err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+ if (err < 0)
+ return err;
+ }
+ if (spec->multiout.dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
+ }
+ if (spec->dig_in_nid) {
+ err = snd_hda_create_spdif_in_ctls(codec,spec->dig_in_nid);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static struct hda_codec_ops conexant_patch_ops = {
+ .build_controls = conexant_build_controls,
+ .build_pcms = conexant_build_pcms,
+ .init = conexant_init,
+ .free = conexant_free,
+};
+
+/*
+ * EAPD control
+ * the private value = nid | (invert << 8)
+ */
+
+#define cxt_eapd_info snd_ctl_boolean_mono_info
+
+static int cxt_eapd_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ if (invert)
+ ucontrol->value.integer.value[0] = !spec->cur_eapd;
+ else
+ ucontrol->value.integer.value[0] = spec->cur_eapd;
+ return 0;
+
+}
+
+static int cxt_eapd_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+ unsigned int eapd;
+
+ eapd = !!ucontrol->value.integer.value[0];
+ if (invert)
+ eapd = !eapd;
+ if (eapd == spec->cur_eapd)
+ return 0;
+
+ spec->cur_eapd = eapd;
+ snd_hda_codec_write_cache(codec, nid,
+ 0, AC_VERB_SET_EAPD_BTLENABLE,
+ eapd ? 0x02 : 0x00);
+ return 1;
+}
+
+/* controls for test mode */
+#ifdef CONFIG_SND_DEBUG
+
+#define CXT_EAPD_SWITCH(xname, nid, mask) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = cxt_eapd_info, \
+ .get = cxt_eapd_get, \
+ .put = cxt_eapd_put, \
+ .private_value = nid | (mask<<16) }
+
+
+
+static int conexant_ch_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
+ spec->num_channel_mode);
+}
+
+static int conexant_ch_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ spec->multiout.max_channels);
+}
+
+static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ &spec->multiout.max_channels);
+ if (err >= 0 && spec->need_dac_fix)
+ spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+ return err;
+}
+
+#define CXT_PIN_MODE(xname, nid, dir) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = conexant_ch_mode_info, \
+ .get = conexant_ch_mode_get, \
+ .put = conexant_ch_mode_put, \
+ .private_value = nid | (dir<<16) }
+
+#endif /* CONFIG_SND_DEBUG */
+
+/* Conexant 5045 specific */
+
+static hda_nid_t cxt5045_dac_nids[1] = { 0x19 };
+static hda_nid_t cxt5045_adc_nids[1] = { 0x1a };
+static hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a };
+#define CXT5045_SPDIF_OUT 0x18
+
+static struct hda_channel_mode cxt5045_modes[1] = {
+ { 2, NULL },
+};
+
+static struct hda_input_mux cxt5045_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "IntMic", 0x1 },
+ { "ExtMic", 0x2 },
+ }
+};
+
+static struct hda_input_mux cxt5045_capture_source_benq = {
+ .num_items = 3,
+ .items = {
+ { "IntMic", 0x1 },
+ { "ExtMic", 0x2 },
+ { "LineIn", 0x3 },
+ }
+};
+
+static struct hda_input_mux cxt5045_capture_source_hp530 = {
+ .num_items = 2,
+ .items = {
+ { "ExtMic", 0x1 },
+ { "IntMic", 0x2 },
+ }
+};
+
+/* turn on/off EAPD (+ mute HP) as a master switch */
+static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ unsigned int bits;
+
+ if (!cxt_eapd_put(kcontrol, ucontrol))
+ return 0;
+
+ /* toggle internal speakers mute depending of presence of
+ * the headphone jack
+ */
+ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+
+ bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x11, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ return 1;
+}
+
+/* bind volumes of both NID 0x10 and 0x11 */
+static struct hda_bind_ctls cxt5045_hp_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/* toggle input of built-in and mic jack appropriately */
+static void cxt5045_hp_automic(struct hda_codec *codec)
+{
+ static struct hda_verb mic_jack_on[] = {
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {}
+ };
+ static struct hda_verb mic_jack_off[] = {
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {}
+ };
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x12, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ if (present)
+ snd_hda_sequence_write(codec, mic_jack_on);
+ else
+ snd_hda_sequence_write(codec, mic_jack_off);
+}
+
+
+/* mute internal speaker if HP is plugged */
+static void cxt5045_hp_automute(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int bits;
+
+ spec->hp_present = snd_hda_codec_read(codec, 0x11, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+
+ bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+/* unsolicited event for HP jack sensing */
+static void cxt5045_hp_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ res >>= 26;
+ switch (res) {
+ case CONEXANT_HP_EVENT:
+ cxt5045_hp_automute(codec);
+ break;
+ case CONEXANT_MIC_EVENT:
+ cxt5045_hp_automic(codec);
+ break;
+
+ }
+}
+
+static struct snd_kcontrol_new cxt5045_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = conexant_mux_enum_info,
+ .get = conexant_mux_enum_get,
+ .put = conexant_mux_enum_put
+ },
+ HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x2, HDA_INPUT),
+ HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5045_hp_master_sw_put,
+ .private_value = 0x10,
+ },
+
+ {}
+};
+
+static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
+ HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
+
+ {}
+};
+
+static struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = conexant_mux_enum_info,
+ .get = conexant_mux_enum_get,
+ .put = conexant_mux_enum_put
+ },
+ HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x17, 0x1, HDA_INPUT),
+ HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5045_hp_master_sw_put,
+ .private_value = 0x10,
+ },
+
+ {}
+};
+
+static struct hda_verb cxt5045_init_verbs[] = {
+ /* Line in, Mic */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
+ /* HP, Amp */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Record selector: Int mic */
+ {0x1a, AC_VERB_SET_CONNECT_SEL,0x1},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
+ /* SPDIF route: PCM */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { 0x13, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ /* EAPD */
+ {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2 }, /* default on */
+ { } /* end */
+};
+
+static struct hda_verb cxt5045_benq_init_verbs[] = {
+ /* Int Mic, Mic */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
+ /* Line In,HP, Amp */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Record selector: Int mic */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
+ /* SPDIF route: PCM */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* EAPD */
+ {0x10, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+ { } /* end */
+};
+
+static struct hda_verb cxt5045_hp_sense_init_verbs[] = {
+ /* pin sensing on HP jack */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ { } /* end */
+};
+
+static struct hda_verb cxt5045_mic_sense_init_verbs[] = {
+ /* pin sensing on HP jack */
+ {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_DEBUG
+/* Test configuration for debugging, modelled after the ALC260 test
+ * configuration.
+ */
+static struct hda_input_mux cxt5045_test_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "MIXER", 0x0 },
+ { "MIC1 pin", 0x1 },
+ { "LINE1 pin", 0x2 },
+ { "HP-OUT pin", 0x3 },
+ { "CD pin", 0x4 },
+ },
+};
+
+static struct snd_kcontrol_new cxt5045_test_mixer[] = {
+
+ /* Output controls */
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+
+ /* Modes for retasking pin widgets */
+ CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT),
+ CXT_PIN_MODE("LINE1 pin mode", 0x12, CXT_PIN_DIR_INOUT),
+
+ /* EAPD Switch Control */
+ CXT_EAPD_SWITCH("External Amplifier", 0x10, 0x0),
+
+ /* Loopback mixer controls */
+
+ HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source",
+ .info = conexant_mux_enum_info,
+ .get = conexant_mux_enum_get,
+ .put = conexant_mux_enum_put,
+ },
+ /* Audio input controls */
+ HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb cxt5045_test_init_verbs[] = {
+ /* Set connections */
+ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ { 0x11, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ { 0x12, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ /* Enable retasking pins as output, initially without power amp */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* Disable digital (SPDIF) pins initially, but users can enable
+ * them via a mixer switch. In the case of SPDIF-out, this initverb
+ * payload also sets the generation to 0, output to be in "consumer"
+ * PCM format, copyright asserted, no pre-emphasis and no validity
+ * control.
+ */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* Unmute retasking pin widget output buffers since the default
+ * state appears to be output. As the pin mode is changed by the
+ * user the pin mode control will take care of enabling the pin's
+ * input/output buffers as needed.
+ */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mute capture amp left and right */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+ /* Set ADC connection select to match default mixer setting (mic1
+ * pin)
+ */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+
+ { }
+};
+#endif
+
+
+/* initialize jack-sensing, too */
+static int cxt5045_init(struct hda_codec *codec)
+{
+ conexant_init(codec);
+ cxt5045_hp_automute(codec);
+ return 0;
+}
+
+
+enum {
+ CXT5045_LAPTOP_HPSENSE,
+ CXT5045_LAPTOP_MICSENSE,
+ CXT5045_LAPTOP_HPMICSENSE,
+ CXT5045_BENQ,
+ CXT5045_LAPTOP_HP530,
+#ifdef CONFIG_SND_DEBUG
+ CXT5045_TEST,
+#endif
+ CXT5045_MODELS
+};
+
+static const char *cxt5045_models[CXT5045_MODELS] = {
+ [CXT5045_LAPTOP_HPSENSE] = "laptop-hpsense",
+ [CXT5045_LAPTOP_MICSENSE] = "laptop-micsense",
+ [CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense",
+ [CXT5045_BENQ] = "benq",
+ [CXT5045_LAPTOP_HP530] = "laptop-hp530",
+#ifdef CONFIG_SND_DEBUG
+ [CXT5045_TEST] = "test",
+#endif
+};
+
+static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530),
+ SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE),
+ SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE),
+ SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ),
+ SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE),
+ SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE),
+ SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505",
+ CXT5045_LAPTOP_HPMICSENSE),
+ SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE),
+ SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE),
+ SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE),
+ SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE),
+ SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE),
+ SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE),
+ {}
+};
+
+static int patch_cxt5045(struct hda_codec *codec)
+{
+ struct conexant_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
+ spec->multiout.dac_nids = cxt5045_dac_nids;
+ spec->multiout.dig_out_nid = CXT5045_SPDIF_OUT;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = cxt5045_adc_nids;
+ spec->capsrc_nids = cxt5045_capsrc_nids;
+ spec->input_mux = &cxt5045_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = cxt5045_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = cxt5045_init_verbs;
+ spec->spdif_route = 0;
+ spec->num_channel_mode = ARRAY_SIZE(cxt5045_modes),
+ spec->channel_mode = cxt5045_modes,
+
+
+ codec->patch_ops = conexant_patch_ops;
+
+ board_config = snd_hda_check_board_config(codec, CXT5045_MODELS,
+ cxt5045_models,
+ cxt5045_cfg_tbl);
+ switch (board_config) {
+ case CXT5045_LAPTOP_HPSENSE:
+ codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
+ spec->input_mux = &cxt5045_capture_source;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = cxt5045_hp_sense_init_verbs;
+ spec->mixers[0] = cxt5045_mixers;
+ codec->patch_ops.init = cxt5045_init;
+ break;
+ case CXT5045_LAPTOP_MICSENSE:
+ codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
+ spec->input_mux = &cxt5045_capture_source;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = cxt5045_mic_sense_init_verbs;
+ spec->mixers[0] = cxt5045_mixers;
+ codec->patch_ops.init = cxt5045_init;
+ break;
+ default:
+ case CXT5045_LAPTOP_HPMICSENSE:
+ codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
+ spec->input_mux = &cxt5045_capture_source;
+ spec->num_init_verbs = 3;
+ spec->init_verbs[1] = cxt5045_hp_sense_init_verbs;
+ spec->init_verbs[2] = cxt5045_mic_sense_init_verbs;
+ spec->mixers[0] = cxt5045_mixers;
+ codec->patch_ops.init = cxt5045_init;
+ break;
+ case CXT5045_BENQ:
+ codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
+ spec->input_mux = &cxt5045_capture_source_benq;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = cxt5045_benq_init_verbs;
+ spec->mixers[0] = cxt5045_mixers;
+ spec->mixers[1] = cxt5045_benq_mixers;
+ spec->num_mixers = 2;
+ codec->patch_ops.init = cxt5045_init;
+ break;
+ case CXT5045_LAPTOP_HP530:
+ codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
+ spec->input_mux = &cxt5045_capture_source_hp530;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = cxt5045_hp_sense_init_verbs;
+ spec->mixers[0] = cxt5045_mixers_hp530;
+ codec->patch_ops.init = cxt5045_init;
+ break;
+#ifdef CONFIG_SND_DEBUG
+ case CXT5045_TEST:
+ spec->input_mux = &cxt5045_test_capture_source;
+ spec->mixers[0] = cxt5045_test_mixer;
+ spec->init_verbs[0] = cxt5045_test_init_verbs;
+ break;
+
+#endif
+ }
+
+ switch (codec->subsystem_id >> 16) {
+ case 0x103c:
+ /* HP laptop has a really bad sound over 0dB on NID 0x17.
+ * Fix max PCM level to 0 dB
+ * (originall it has 0x2b steps with 0dB offset 0x14)
+ */
+ snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
+ (0x14 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
+ break;
+ }
+
+ return 0;
+}
+
+
+/* Conexant 5047 specific */
+#define CXT5047_SPDIF_OUT 0x11
+
+static hda_nid_t cxt5047_dac_nids[2] = { 0x10, 0x1c };
+static hda_nid_t cxt5047_adc_nids[1] = { 0x12 };
+static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a };
+
+static struct hda_channel_mode cxt5047_modes[1] = {
+ { 2, NULL },
+};
+
+static struct hda_input_mux cxt5047_capture_source = {
+ .num_items = 1,
+ .items = {
+ { "Mic", 0x2 },
+ }
+};
+
+static struct hda_input_mux cxt5047_hp_capture_source = {
+ .num_items = 1,
+ .items = {
+ { "ExtMic", 0x2 },
+ }
+};
+
+static struct hda_input_mux cxt5047_toshiba_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "ExtMic", 0x2 },
+ { "Line-In", 0x1 },
+ }
+};
+
+/* turn on/off EAPD (+ mute HP) as a master switch */
+static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ unsigned int bits;
+
+ if (!cxt_eapd_put(kcontrol, ucontrol))
+ return 0;
+
+ /* toggle internal speakers mute depending of presence of
+ * the headphone jack
+ */
+ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ return 1;
+}
+
+/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */
+static struct hda_bind_ctls cxt5047_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/* mute internal speaker if HP is plugged */
+static void cxt5047_hp_automute(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int bits;
+
+ spec->hp_present = snd_hda_codec_read(codec, 0x13, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+
+ bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ /* Mute/Unmute PCM 2 for good measure - some systems need this */
+ snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+/* mute internal speaker if HP is plugged */
+static void cxt5047_hp2_automute(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int bits;
+
+ spec->hp_present = snd_hda_codec_read(codec, 0x13, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+
+ bits = spec->hp_present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ /* Mute/Unmute PCM 2 for good measure - some systems need this */
+ snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+/* toggle input of built-in and mic jack appropriately */
+static void cxt5047_hp_automic(struct hda_codec *codec)
+{
+ static struct hda_verb mic_jack_on[] = {
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {}
+ };
+ static struct hda_verb mic_jack_off[] = {
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {}
+ };
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ if (present)
+ snd_hda_sequence_write(codec, mic_jack_on);
+ else
+ snd_hda_sequence_write(codec, mic_jack_off);
+}
+
+/* unsolicited event for HP jack sensing */
+static void cxt5047_hp_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case CONEXANT_HP_EVENT:
+ cxt5047_hp_automute(codec);
+ break;
+ case CONEXANT_MIC_EVENT:
+ cxt5047_hp_automic(codec);
+ break;
+ }
+}
+
+/* unsolicited event for HP jack sensing - non-EAPD systems */
+static void cxt5047_hp2_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ res >>= 26;
+ switch (res) {
+ case CONEXANT_HP_EVENT:
+ cxt5047_hp2_automute(codec);
+ break;
+ case CONEXANT_MIC_EVENT:
+ cxt5047_hp_automic(codec);
+ break;
+ }
+}
+
+static struct snd_kcontrol_new cxt5047_mixers[] = {
+ HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Gain Volume", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Gain Switch", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM-2 Volume", 0x1c, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM-2 Switch", 0x1c, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT),
+
+ {}
+};
+
+static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = conexant_mux_enum_info,
+ .get = conexant_mux_enum_get,
+ .put = conexant_mux_enum_put
+ },
+ HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
+ HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5047_hp_master_sw_put,
+ .private_value = 0x13,
+ },
+
+ {}
+};
+
+static struct snd_kcontrol_new cxt5047_hp_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = conexant_mux_enum_info,
+ .get = conexant_mux_enum_get,
+ .put = conexant_mux_enum_put
+ },
+ HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5047_hp_master_sw_put,
+ .private_value = 0x13,
+ },
+ { } /* end */
+};
+
+static struct hda_verb cxt5047_init_verbs[] = {
+ /* Line in, Mic, Built-in Mic */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 },
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 },
+ /* HP, Speaker */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ {0x13, AC_VERB_SET_CONNECT_SEL,0x1},
+ {0x1d, AC_VERB_SET_CONNECT_SEL,0x0},
+ /* Record selector: Mic */
+ {0x12, AC_VERB_SET_CONNECT_SEL,0x03},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
+ {0x1A, AC_VERB_SET_CONNECT_SEL,0x02},
+ {0x1A, AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x00},
+ {0x1A, AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_OUTPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x03},
+ /* SPDIF route: PCM */
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x0 },
+ /* Enable unsolicited events */
+ {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+ { } /* end */
+};
+
+/* configuration for Toshiba Laptops */
+static struct hda_verb cxt5047_toshiba_init_verbs[] = {
+ {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0 }, /* default on */
+ /* pin sensing on HP and Mic jacks */
+ {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+ /* Speaker routing */
+ {0x1d, AC_VERB_SET_CONNECT_SEL,0x1},
+ {}
+};
+
+/* configuration for HP Laptops */
+static struct hda_verb cxt5047_hp_init_verbs[] = {
+ /* pin sensing on HP jack */
+ {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ /* 0x13 is actually shared by both HP and speaker;
+ * setting the connection to 0 (=0x19) makes the master volume control
+ * working mysteriouslly...
+ */
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Record selector: Ext Mic */
+ {0x12, AC_VERB_SET_CONNECT_SEL,0x03},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE,
+ AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
+ /* Speaker routing */
+ {0x1d, AC_VERB_SET_CONNECT_SEL,0x1},
+ {}
+};
+
+/* Test configuration for debugging, modelled after the ALC260 test
+ * configuration.
+ */
+#ifdef CONFIG_SND_DEBUG
+static struct hda_input_mux cxt5047_test_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "LINE1 pin", 0x0 },
+ { "MIC1 pin", 0x1 },
+ { "MIC2 pin", 0x2 },
+ { "CD pin", 0x3 },
+ },
+};
+
+static struct snd_kcontrol_new cxt5047_test_mixer[] = {
+
+ /* Output only controls */
+ HDA_CODEC_VOLUME("OutAmp-1 Volume", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("OutAmp-1 Switch", 0x10,0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("OutAmp-2 Volume", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("OutAmp-2 Switch", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("HeadPhone Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("HeadPhone Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line1-Out Playback Volume", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line1-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line2-Out Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line2-Out Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+
+ /* Modes for retasking pin widgets */
+ CXT_PIN_MODE("LINE1 pin mode", 0x14, CXT_PIN_DIR_INOUT),
+ CXT_PIN_MODE("MIC1 pin mode", 0x15, CXT_PIN_DIR_INOUT),
+
+ /* EAPD Switch Control */
+ CXT_EAPD_SWITCH("External Amplifier", 0x13, 0x0),
+
+ /* Loopback mixer controls */
+ HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x12, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("MIC1 Playback Switch", 0x12, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x12, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("MIC2 Playback Switch", 0x12, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE Playback Volume", 0x12, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("LINE Playback Switch", 0x12, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x12, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x12, 0x04, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Capture-1 Volume", 0x19, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture-1 Switch", 0x19, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture-2 Volume", 0x19, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture-2 Switch", 0x19, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture-3 Volume", 0x19, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture-3 Switch", 0x19, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture-4 Volume", 0x19, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture-4 Switch", 0x19, 0x3, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source",
+ .info = conexant_mux_enum_info,
+ .get = conexant_mux_enum_get,
+ .put = conexant_mux_enum_put,
+ },
+ HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT),
+
+ { } /* end */
+};
+
+static struct hda_verb cxt5047_test_init_verbs[] = {
+ /* Enable retasking pins as output, initially without power amp */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* Disable digital (SPDIF) pins initially, but users can enable
+ * them via a mixer switch. In the case of SPDIF-out, this initverb
+ * payload also sets the generation to 0, output to be in "consumer"
+ * PCM format, copyright asserted, no pre-emphasis and no validity
+ * control.
+ */
+ {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure mic1, mic2, line1 pin widgets take input from the
+ * OUT1 sum bus when acting as an output.
+ */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* Unmute retasking pin widget output buffers since the default
+ * state appears to be output. As the pin mode is changed by the
+ * user the pin mode control will take care of enabling the pin's
+ * input/output buffers as needed.
+ */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mute capture amp left and right */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+ /* Set ADC connection select to match default mixer setting (mic1
+ * pin)
+ */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+#endif
+
+
+/* initialize jack-sensing, too */
+static int cxt5047_hp_init(struct hda_codec *codec)
+{
+ conexant_init(codec);
+ cxt5047_hp_automute(codec);
+ return 0;
+}
+
+
+enum {
+ CXT5047_LAPTOP, /* Laptops w/o EAPD support */
+ CXT5047_LAPTOP_HP, /* Some HP laptops */
+ CXT5047_LAPTOP_EAPD, /* Laptops with EAPD support */
+#ifdef CONFIG_SND_DEBUG
+ CXT5047_TEST,
+#endif
+ CXT5047_MODELS
+};
+
+static const char *cxt5047_models[CXT5047_MODELS] = {
+ [CXT5047_LAPTOP] = "laptop",
+ [CXT5047_LAPTOP_HP] = "laptop-hp",
+ [CXT5047_LAPTOP_EAPD] = "laptop-eapd",
+#ifdef CONFIG_SND_DEBUG
+ [CXT5047_TEST] = "test",
+#endif
+};
+
+static struct snd_pci_quirk cxt5047_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP),
+ SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP),
+ SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP),
+ SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP),
+ SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP),
+ SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD),
+ {}
+};
+
+static int patch_cxt5047(struct hda_codec *codec)
+{
+ struct conexant_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids);
+ spec->multiout.dac_nids = cxt5047_dac_nids;
+ spec->multiout.dig_out_nid = CXT5047_SPDIF_OUT;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = cxt5047_adc_nids;
+ spec->capsrc_nids = cxt5047_capsrc_nids;
+ spec->input_mux = &cxt5047_capture_source;
+ spec->num_mixers = 1;
+ spec->mixers[0] = cxt5047_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = cxt5047_init_verbs;
+ spec->spdif_route = 0;
+ spec->num_channel_mode = ARRAY_SIZE(cxt5047_modes),
+ spec->channel_mode = cxt5047_modes,
+
+ codec->patch_ops = conexant_patch_ops;
+
+ board_config = snd_hda_check_board_config(codec, CXT5047_MODELS,
+ cxt5047_models,
+ cxt5047_cfg_tbl);
+ switch (board_config) {
+ case CXT5047_LAPTOP:
+ codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event;
+ break;
+ case CXT5047_LAPTOP_HP:
+ spec->input_mux = &cxt5047_hp_capture_source;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = cxt5047_hp_init_verbs;
+ spec->mixers[0] = cxt5047_hp_mixers;
+ codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
+ codec->patch_ops.init = cxt5047_hp_init;
+ break;
+ case CXT5047_LAPTOP_EAPD:
+ spec->input_mux = &cxt5047_toshiba_capture_source;
+ spec->num_init_verbs = 2;
+ spec->init_verbs[1] = cxt5047_toshiba_init_verbs;
+ spec->mixers[0] = cxt5047_toshiba_mixers;
+ codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
+ break;
+#ifdef CONFIG_SND_DEBUG
+ case CXT5047_TEST:
+ spec->input_mux = &cxt5047_test_capture_source;
+ spec->mixers[0] = cxt5047_test_mixer;
+ spec->init_verbs[0] = cxt5047_test_init_verbs;
+ codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
+#endif
+ }
+ return 0;
+}
+
+/* Conexant 5051 specific */
+static hda_nid_t cxt5051_dac_nids[1] = { 0x10 };
+static hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 };
+#define CXT5051_SPDIF_OUT 0x1C
+#define CXT5051_PORTB_EVENT 0x38
+#define CXT5051_PORTC_EVENT 0x39
+
+static struct hda_channel_mode cxt5051_modes[1] = {
+ { 2, NULL },
+};
+
+static void cxt5051_update_speaker(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int pinctl;
+ pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinctl);
+}
+
+/* turn on/off EAPD (+ mute HP) as a master switch */
+static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (!cxt_eapd_put(kcontrol, ucontrol))
+ return 0;
+ cxt5051_update_speaker(codec);
+ return 1;
+}
+
+/* toggle input of built-in and mic jack appropriately */
+static void cxt5051_portb_automic(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x17, 0,
+ AC_VERB_GET_PIN_SENSE, 0) &
+ AC_PINSENSE_PRESENCE;
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ present ? 0x01 : 0x00);
+}
+
+/* switch the current ADC according to the jack state */
+static void cxt5051_portc_automic(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int present;
+ hda_nid_t new_adc;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) &
+ AC_PINSENSE_PRESENCE;
+ if (present)
+ spec->cur_adc_idx = 1;
+ else
+ spec->cur_adc_idx = 0;
+ new_adc = spec->adc_nids[spec->cur_adc_idx];
+ if (spec->cur_adc && spec->cur_adc != new_adc) {
+ /* stream is running, let's swap the current ADC */
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = new_adc;
+ snd_hda_codec_setup_stream(codec, new_adc,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+ }
+}
+
+/* mute internal speaker if HP is plugged */
+static void cxt5051_hp_automute(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+
+ spec->hp_present = snd_hda_codec_read(codec, 0x16, 0,
+ AC_VERB_GET_PIN_SENSE, 0) &
+ AC_PINSENSE_PRESENCE;
+ cxt5051_update_speaker(codec);
+}
+
+/* unsolicited event for HP jack sensing */
+static void cxt5051_hp_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case CONEXANT_HP_EVENT:
+ cxt5051_hp_automute(codec);
+ break;
+ case CXT5051_PORTB_EVENT:
+ cxt5051_portb_automic(codec);
+ break;
+ case CXT5051_PORTC_EVENT:
+ cxt5051_portc_automic(codec);
+ break;
+ }
+}
+
+static struct snd_kcontrol_new cxt5051_mixers[] = {
+ HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("External Mic Volume", 0x14, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("External Mic Switch", 0x14, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5051_hp_master_sw_put,
+ .private_value = 0x1a,
+ },
+
+ {}
+};
+
+static struct snd_kcontrol_new cxt5051_hp_mixers[] = {
+ HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("External Mic Volume", 0x15, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("External Mic Switch", 0x15, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = cxt_eapd_info,
+ .get = cxt_eapd_get,
+ .put = cxt5051_hp_master_sw_put,
+ .private_value = 0x1a,
+ },
+
+ {}
+};
+
+static struct hda_verb cxt5051_init_verbs[] = {
+ /* Line in, Mic */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+ /* SPK */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP, Amp */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* DAC1 */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Record selector: Int mic */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+ /* SPDIF route: PCM */
+ {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* EAPD */
+ {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+ {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT},
+ { } /* end */
+};
+
+/* initialize jack-sensing, too */
+static int cxt5051_init(struct hda_codec *codec)
+{
+ conexant_init(codec);
+ if (codec->patch_ops.unsol_event) {
+ cxt5051_hp_automute(codec);
+ cxt5051_portb_automic(codec);
+ cxt5051_portc_automic(codec);
+ }
+ return 0;
+}
+
+
+enum {
+ CXT5051_LAPTOP, /* Laptops w/ EAPD support */
+ CXT5051_HP, /* no docking */
+ CXT5051_MODELS
+};
+
+static const char *cxt5051_models[CXT5051_MODELS] = {
+ [CXT5051_LAPTOP] = "laptop",
+ [CXT5051_HP] = "hp",
+};
+
+static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
+ CXT5051_LAPTOP),
+ SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
+ {}
+};
+
+static int patch_cxt5051(struct hda_codec *codec)
+{
+ struct conexant_spec *spec;
+ int board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ codec->patch_ops = conexant_patch_ops;
+ codec->patch_ops.init = cxt5051_init;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(cxt5051_dac_nids);
+ spec->multiout.dac_nids = cxt5051_dac_nids;
+ spec->multiout.dig_out_nid = CXT5051_SPDIF_OUT;
+ spec->num_adc_nids = 1; /* not 2; via auto-mic switch */
+ spec->adc_nids = cxt5051_adc_nids;
+ spec->num_mixers = 1;
+ spec->mixers[0] = cxt5051_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = cxt5051_init_verbs;
+ spec->spdif_route = 0;
+ spec->num_channel_mode = ARRAY_SIZE(cxt5051_modes);
+ spec->channel_mode = cxt5051_modes;
+ spec->cur_adc = 0;
+ spec->cur_adc_idx = 0;
+
+ board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
+ cxt5051_models,
+ cxt5051_cfg_tbl);
+ switch (board_config) {
+ case CXT5051_HP:
+ codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
+ spec->mixers[0] = cxt5051_hp_mixers;
+ break;
+ default:
+ case CXT5051_LAPTOP:
+ codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
+ break;
+ }
+
+ return 0;
+}
+
+
+/*
+ */
+
+struct hda_codec_preset snd_hda_preset_conexant[] = {
+ { .id = 0x14f15045, .name = "CX20549 (Venice)",
+ .patch = patch_cxt5045 },
+ { .id = 0x14f15047, .name = "CX20551 (Waikiki)",
+ .patch = patch_cxt5047 },
+ { .id = 0x14f15051, .name = "CX20561 (Hermosa)",
+ .patch = patch_cxt5051 },
+ {} /* terminator */
+};
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
new file mode 100644
index 0000000..2eed2c8
--- /dev/null
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -0,0 +1,165 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for NVIDIA HDMI codecs
+ *
+ * Copyright (c) 2008 NVIDIA Corp. All rights reserved.
+ * Copyright (c) 2008 Wei Ni <wni@nvidia.com>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+struct nvhdmi_spec {
+ struct hda_multi_out multiout;
+
+ struct hda_pcm pcm_rec;
+};
+
+static struct hda_verb nvhdmi_basic_init[] = {
+ /* enable digital output on pin widget */
+ { 0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ {} /* terminator */
+};
+
+/*
+ * Controls
+ */
+static int nvhdmi_build_controls(struct hda_codec *codec)
+{
+ struct nvhdmi_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int nvhdmi_init(struct hda_codec *codec)
+{
+ snd_hda_sequence_write(codec, nvhdmi_basic_init);
+ return 0;
+}
+
+/*
+ * Digital out
+ */
+static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct nvhdmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int nvhdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct nvhdmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int nvhdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct nvhdmi_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+static struct hda_pcm_stream nvhdmi_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x4, /* NID to query formats and rates and setup streams */
+ .rates = SNDRV_PCM_RATE_48000,
+ .maxbps = 16,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .ops = {
+ .open = nvhdmi_dig_playback_pcm_open,
+ .close = nvhdmi_dig_playback_pcm_close,
+ .prepare = nvhdmi_dig_playback_pcm_prepare
+ },
+};
+
+static int nvhdmi_build_pcms(struct hda_codec *codec)
+{
+ struct nvhdmi_spec *spec = codec->spec;
+ struct hda_pcm *info = &spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "NVIDIA HDMI";
+ info->pcm_type = HDA_PCM_TYPE_HDMI;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback;
+
+ return 0;
+}
+
+static void nvhdmi_free(struct hda_codec *codec)
+{
+ kfree(codec->spec);
+}
+
+static struct hda_codec_ops nvhdmi_patch_ops = {
+ .build_controls = nvhdmi_build_controls,
+ .build_pcms = nvhdmi_build_pcms,
+ .init = nvhdmi_init,
+ .free = nvhdmi_free,
+};
+
+static int patch_nvhdmi(struct hda_codec *codec)
+{
+ struct nvhdmi_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.num_dacs = 0; /* no analog */
+ spec->multiout.max_channels = 2;
+ spec->multiout.dig_out_nid = 0x4; /* NID for copying analog to digital,
+ * seems to be unused in pure-digital
+ * case. */
+
+ codec->patch_ops = nvhdmi_patch_ops;
+
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
+ { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi },
+ { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi },
+ {} /* terminator */
+};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
new file mode 100644
index 0000000..7123a6e
--- /dev/null
+++ b/sound/pci/hda/patch_realtek.c
@@ -0,0 +1,16493 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for ALC 260/880/882 codecs
+ *
+ * Copyright (c) 2004 Kailang Yang <kailang@realtek.com.tw>
+ * PeiSen Hou <pshou@realtek.com.tw>
+ * Takashi Iwai <tiwai@suse.de>
+ * Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+
+#define ALC880_FRONT_EVENT 0x01
+#define ALC880_DCVOL_EVENT 0x02
+#define ALC880_HP_EVENT 0x04
+#define ALC880_MIC_EVENT 0x08
+
+/* ALC880 board config type */
+enum {
+ ALC880_3ST,
+ ALC880_3ST_DIG,
+ ALC880_5ST,
+ ALC880_5ST_DIG,
+ ALC880_W810,
+ ALC880_Z71V,
+ ALC880_6ST,
+ ALC880_6ST_DIG,
+ ALC880_F1734,
+ ALC880_ASUS,
+ ALC880_ASUS_DIG,
+ ALC880_ASUS_W1V,
+ ALC880_ASUS_DIG2,
+ ALC880_FUJITSU,
+ ALC880_UNIWILL_DIG,
+ ALC880_UNIWILL,
+ ALC880_UNIWILL_P53,
+ ALC880_CLEVO,
+ ALC880_TCL_S700,
+ ALC880_LG,
+ ALC880_LG_LW,
+ ALC880_MEDION_RIM,
+#ifdef CONFIG_SND_DEBUG
+ ALC880_TEST,
+#endif
+ ALC880_AUTO,
+ ALC880_MODEL_LAST /* last tag */
+};
+
+/* ALC260 models */
+enum {
+ ALC260_BASIC,
+ ALC260_HP,
+ ALC260_HP_DC7600,
+ ALC260_HP_3013,
+ ALC260_FUJITSU_S702X,
+ ALC260_ACER,
+ ALC260_WILL,
+ ALC260_REPLACER_672V,
+#ifdef CONFIG_SND_DEBUG
+ ALC260_TEST,
+#endif
+ ALC260_AUTO,
+ ALC260_MODEL_LAST /* last tag */
+};
+
+/* ALC262 models */
+enum {
+ ALC262_BASIC,
+ ALC262_HIPPO,
+ ALC262_HIPPO_1,
+ ALC262_FUJITSU,
+ ALC262_HP_BPC,
+ ALC262_HP_BPC_D7000_WL,
+ ALC262_HP_BPC_D7000_WF,
+ ALC262_HP_TC_T5735,
+ ALC262_HP_RP5700,
+ ALC262_BENQ_ED8,
+ ALC262_SONY_ASSAMD,
+ ALC262_BENQ_T31,
+ ALC262_ULTRA,
+ ALC262_LENOVO_3000,
+ ALC262_NEC,
+ ALC262_TOSHIBA_S06,
+ ALC262_TOSHIBA_RX1,
+ ALC262_AUTO,
+ ALC262_MODEL_LAST /* last tag */
+};
+
+/* ALC268 models */
+enum {
+ ALC267_QUANTA_IL1,
+ ALC268_3ST,
+ ALC268_TOSHIBA,
+ ALC268_ACER,
+ ALC268_ACER_ASPIRE_ONE,
+ ALC268_DELL,
+ ALC268_ZEPTO,
+#ifdef CONFIG_SND_DEBUG
+ ALC268_TEST,
+#endif
+ ALC268_AUTO,
+ ALC268_MODEL_LAST /* last tag */
+};
+
+/* ALC269 models */
+enum {
+ ALC269_BASIC,
+ ALC269_QUANTA_FL1,
+ ALC269_ASUS_EEEPC_P703,
+ ALC269_ASUS_EEEPC_P901,
+ ALC269_AUTO,
+ ALC269_MODEL_LAST /* last tag */
+};
+
+/* ALC861 models */
+enum {
+ ALC861_3ST,
+ ALC660_3ST,
+ ALC861_3ST_DIG,
+ ALC861_6ST_DIG,
+ ALC861_UNIWILL_M31,
+ ALC861_TOSHIBA,
+ ALC861_ASUS,
+ ALC861_ASUS_LAPTOP,
+ ALC861_AUTO,
+ ALC861_MODEL_LAST,
+};
+
+/* ALC861-VD models */
+enum {
+ ALC660VD_3ST,
+ ALC660VD_3ST_DIG,
+ ALC861VD_3ST,
+ ALC861VD_3ST_DIG,
+ ALC861VD_6ST_DIG,
+ ALC861VD_LENOVO,
+ ALC861VD_DALLAS,
+ ALC861VD_HP,
+ ALC861VD_AUTO,
+ ALC861VD_MODEL_LAST,
+};
+
+/* ALC662 models */
+enum {
+ ALC662_3ST_2ch_DIG,
+ ALC662_3ST_6ch_DIG,
+ ALC662_3ST_6ch,
+ ALC662_5ST_DIG,
+ ALC662_LENOVO_101E,
+ ALC662_ASUS_EEEPC_P701,
+ ALC662_ASUS_EEEPC_EP20,
+ ALC663_ASUS_M51VA,
+ ALC663_ASUS_G71V,
+ ALC663_ASUS_H13,
+ ALC663_ASUS_G50V,
+ ALC662_ECS,
+ ALC663_ASUS_MODE1,
+ ALC662_ASUS_MODE2,
+ ALC663_ASUS_MODE3,
+ ALC663_ASUS_MODE4,
+ ALC663_ASUS_MODE5,
+ ALC663_ASUS_MODE6,
+ ALC662_AUTO,
+ ALC662_MODEL_LAST,
+};
+
+/* ALC882 models */
+enum {
+ ALC882_3ST_DIG,
+ ALC882_6ST_DIG,
+ ALC882_ARIMA,
+ ALC882_W2JC,
+ ALC882_TARGA,
+ ALC882_ASUS_A7J,
+ ALC882_ASUS_A7M,
+ ALC885_MACPRO,
+ ALC885_MBP3,
+ ALC885_IMAC24,
+ ALC882_AUTO,
+ ALC882_MODEL_LAST,
+};
+
+/* ALC883 models */
+enum {
+ ALC883_3ST_2ch_DIG,
+ ALC883_3ST_6ch_DIG,
+ ALC883_3ST_6ch,
+ ALC883_6ST_DIG,
+ ALC883_TARGA_DIG,
+ ALC883_TARGA_2ch_DIG,
+ ALC883_ACER,
+ ALC883_ACER_ASPIRE,
+ ALC883_MEDION,
+ ALC883_MEDION_MD2,
+ ALC883_LAPTOP_EAPD,
+ ALC883_LENOVO_101E_2ch,
+ ALC883_LENOVO_NB0763,
+ ALC888_LENOVO_MS7195_DIG,
+ ALC888_LENOVO_SKY,
+ ALC883_HAIER_W66,
+ ALC888_3ST_HP,
+ ALC888_6ST_DELL,
+ ALC883_MITAC,
+ ALC883_CLEVO_M720,
+ ALC883_FUJITSU_PI2515,
+ ALC883_3ST_6ch_INTEL,
+ ALC888_ASUS_M90V,
+ ALC888_ASUS_EEE1601,
+ ALC883_AUTO,
+ ALC883_MODEL_LAST,
+};
+
+/* for GPIO Poll */
+#define GPIO_MASK 0x03
+
+struct alc_spec {
+ /* codec parameterization */
+ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
+ unsigned int num_mixers;
+
+ const struct hda_verb *init_verbs[5]; /* initialization verbs
+ * don't forget NULL
+ * termination!
+ */
+ unsigned int num_init_verbs;
+
+ char *stream_name_analog; /* analog PCM stream */
+ struct hda_pcm_stream *stream_analog_playback;
+ struct hda_pcm_stream *stream_analog_capture;
+ struct hda_pcm_stream *stream_analog_alt_playback;
+ struct hda_pcm_stream *stream_analog_alt_capture;
+
+ char *stream_name_digital; /* digital PCM stream */
+ struct hda_pcm_stream *stream_digital_playback;
+ struct hda_pcm_stream *stream_digital_capture;
+
+ /* playback */
+ struct hda_multi_out multiout; /* playback set-up
+ * max_channels, dacs must be set
+ * dig_out_nid and hp_nid are optional
+ */
+ hda_nid_t alt_dac_nid;
+
+ /* capture */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ hda_nid_t *capsrc_nids;
+ hda_nid_t dig_in_nid; /* digital-in NID; optional */
+
+ /* capture source */
+ unsigned int num_mux_defs;
+ const struct hda_input_mux *input_mux;
+ unsigned int cur_mux[3];
+
+ /* channel model */
+ const struct hda_channel_mode *channel_mode;
+ int num_channel_mode;
+ int need_dac_fix;
+
+ /* PCM information */
+ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
+
+ /* dynamic controls, init_verbs and input_mux */
+ struct auto_pin_cfg autocfg;
+ unsigned int num_kctl_alloc, num_kctl_used;
+ struct snd_kcontrol_new *kctl_alloc;
+ struct hda_input_mux private_imux;
+ hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+
+ /* hooks */
+ void (*init_hook)(struct hda_codec *codec);
+ void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+
+ /* for pin sensing */
+ unsigned int sense_updated: 1;
+ unsigned int jack_present: 1;
+ unsigned int master_sw: 1;
+
+ /* for virtual master */
+ hda_nid_t vmaster_nid;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
+
+ /* for PLL fix */
+ hda_nid_t pll_nid;
+ unsigned int pll_coef_idx, pll_coef_bit;
+
+#ifdef SND_HDA_NEEDS_RESUME
+#define ALC_MAX_PINS 16
+ unsigned int num_pins;
+ hda_nid_t pin_nids[ALC_MAX_PINS];
+ unsigned int pin_cfgs[ALC_MAX_PINS];
+#endif
+};
+
+/*
+ * configuration template - to be copied to the spec instance
+ */
+struct alc_config_preset {
+ struct snd_kcontrol_new *mixers[5]; /* should be identical size
+ * with spec
+ */
+ const struct hda_verb *init_verbs[5];
+ unsigned int num_dacs;
+ hda_nid_t *dac_nids;
+ hda_nid_t dig_out_nid; /* optional */
+ hda_nid_t hp_nid; /* optional */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ hda_nid_t *capsrc_nids;
+ hda_nid_t dig_in_nid;
+ unsigned int num_channel_mode;
+ const struct hda_channel_mode *channel_mode;
+ int need_dac_fix;
+ unsigned int num_mux_defs;
+ const struct hda_input_mux *input_mux;
+ void (*unsol_event)(struct hda_codec *, unsigned int);
+ void (*init_hook)(struct hda_codec *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_amp_list *loopbacks;
+#endif
+};
+
+
+/*
+ * input MUX handling
+ */
+static int alc_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ unsigned int mux_idx = snd_ctl_get_ioffidx(kcontrol, &uinfo->id);
+ if (mux_idx >= spec->num_mux_defs)
+ mux_idx = 0;
+ return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo);
+}
+
+static int alc_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ return 0;
+}
+
+static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
+ hda_nid_t nid = spec->capsrc_nids ?
+ spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
+ return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol,
+ nid, &spec->cur_mux[adc_idx]);
+}
+
+
+/*
+ * channel mode setting
+ */
+static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
+ spec->num_channel_mode);
+}
+
+static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ spec->multiout.max_channels);
+}
+
+static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+ spec->num_channel_mode,
+ &spec->multiout.max_channels);
+ if (err >= 0 && spec->need_dac_fix)
+ spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+ return err;
+}
+
+/*
+ * Control the mode of pin widget settings via the mixer. "pc" is used
+ * instead of "%" to avoid consequences of accidently treating the % as
+ * being part of a format specifier. Maximum allowed length of a value is
+ * 63 characters plus NULL terminator.
+ *
+ * Note: some retasking pin complexes seem to ignore requests for input
+ * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
+ * are requested. Therefore order this list so that this behaviour will not
+ * cause problems when mixer clients move through the enum sequentially.
+ * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
+ * March 2006.
+ */
+static char *alc_pin_mode_names[] = {
+ "Mic 50pc bias", "Mic 80pc bias",
+ "Line in", "Line out", "Headphone out",
+};
+static unsigned char alc_pin_mode_values[] = {
+ PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
+};
+/* The control can present all 5 options, or it can limit the options based
+ * in the pin being assumed to be exclusively an input or an output pin. In
+ * addition, "input" pins may or may not process the mic bias option
+ * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
+ * accept requests for bias as of chip versions up to March 2006) and/or
+ * wiring in the computer.
+ */
+#define ALC_PIN_DIR_IN 0x00
+#define ALC_PIN_DIR_OUT 0x01
+#define ALC_PIN_DIR_INOUT 0x02
+#define ALC_PIN_DIR_IN_NOMICBIAS 0x03
+#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
+
+/* Info about the pin modes supported by the different pin direction modes.
+ * For each direction the minimum and maximum values are given.
+ */
+static signed char alc_pin_mode_dir_info[5][2] = {
+ { 0, 2 }, /* ALC_PIN_DIR_IN */
+ { 3, 4 }, /* ALC_PIN_DIR_OUT */
+ { 0, 4 }, /* ALC_PIN_DIR_INOUT */
+ { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
+ { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
+};
+#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
+#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
+#define alc_pin_mode_n_items(_dir) \
+ (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
+
+static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int item_num = uinfo->value.enumerated.item;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
+
+ if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
+ item_num = alc_pin_mode_min(dir);
+ strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
+ return 0;
+}
+
+static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int i;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL,
+ 0x00);
+
+ /* Find enumerated value for current pinctl setting */
+ i = alc_pin_mode_min(dir);
+ while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir))
+ i++;
+ *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
+ return 0;
+}
+
+static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ signed int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL,
+ 0x00);
+
+ if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
+ val = alc_pin_mode_min(dir);
+
+ change = pinctl != alc_pin_mode_values[val];
+ if (change) {
+ /* Set pin mode to that requested */
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ alc_pin_mode_values[val]);
+
+ /* Also enable the retasking pin's input/output as required
+ * for the requested pin mode. Enum values of 2 or less are
+ * input modes.
+ *
+ * Dynamically switching the input/output buffers probably
+ * reduces noise slightly (particularly on input) so we'll
+ * do it. However, having both input and output buffers
+ * enabled simultaneously doesn't seem to be problematic if
+ * this turns out to be necessary in the future.
+ */
+ if (val <= 2) {
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+ HDA_AMP_MUTE, 0);
+ } else {
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, 0);
+ }
+ }
+ return change;
+}
+
+#define ALC_PIN_MODE(xname, nid, dir) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = alc_pin_mode_info, \
+ .get = alc_pin_mode_get, \
+ .put = alc_pin_mode_put, \
+ .private_value = nid | (dir<<16) }
+
+/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
+ * together using a mask with more than one bit set. This control is
+ * currently used only by the ALC260 test model. At this stage they are not
+ * needed for any "production" models.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_gpio_data_info snd_ctl_boolean_mono_info
+
+static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_DATA, 0x00);
+
+ *valp = (val & mask) != 0;
+ return 0;
+}
+static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ signed int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_GPIO_DATA,
+ 0x00);
+
+ /* Set/unset the masked GPIO bit(s) as needed */
+ change = (val == 0 ? 0 : mask) != (gpio_data & mask);
+ if (val == 0)
+ gpio_data &= ~mask;
+ else
+ gpio_data |= mask;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_GPIO_DATA, gpio_data);
+
+ return change;
+}
+#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = alc_gpio_data_info, \
+ .get = alc_gpio_data_get, \
+ .put = alc_gpio_data_put, \
+ .private_value = nid | (mask<<16) }
+#endif /* CONFIG_SND_DEBUG */
+
+/* A switch control to allow the enabling of the digital IO pins on the
+ * ALC260. This is incredibly simplistic; the intention of this control is
+ * to provide something in the test model allowing digital outputs to be
+ * identified if present. If models are found which can utilise these
+ * outputs a more complete mixer control can be devised for those models if
+ * necessary.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
+
+static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1, 0x00);
+
+ *valp = (val & mask) != 0;
+ return 0;
+}
+static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ signed int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_DIGI_CONVERT_1,
+ 0x00);
+
+ /* Set/unset the masked control bit(s) as needed */
+ change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
+ if (val==0)
+ ctrl_data &= ~mask;
+ else
+ ctrl_data |= mask;
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+ ctrl_data);
+
+ return change;
+}
+#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = alc_spdif_ctrl_info, \
+ .get = alc_spdif_ctrl_get, \
+ .put = alc_spdif_ctrl_put, \
+ .private_value = nid | (mask<<16) }
+#endif /* CONFIG_SND_DEBUG */
+
+/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
+ * Again, this is only used in the ALC26x test models to help identify when
+ * the EAPD line must be asserted for features to work.
+ */
+#ifdef CONFIG_SND_DEBUG
+#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
+
+static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_EAPD_BTLENABLE, 0x00);
+
+ *valp = (val & mask) != 0;
+ return 0;
+}
+
+static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int change;
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
+ long val = *ucontrol->value.integer.value;
+ unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_EAPD_BTLENABLE,
+ 0x00);
+
+ /* Set/unset the masked control bit(s) as needed */
+ change = (!val ? 0 : mask) != (ctrl_data & mask);
+ if (!val)
+ ctrl_data &= ~mask;
+ else
+ ctrl_data |= mask;
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ ctrl_data);
+
+ return change;
+}
+
+#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = alc_eapd_ctrl_info, \
+ .get = alc_eapd_ctrl_get, \
+ .put = alc_eapd_ctrl_put, \
+ .private_value = nid | (mask<<16) }
+#endif /* CONFIG_SND_DEBUG */
+
+/*
+ * set up from the preset table
+ */
+static void setup_preset(struct alc_spec *spec,
+ const struct alc_config_preset *preset)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
+ spec->mixers[spec->num_mixers++] = preset->mixers[i];
+ for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
+ i++)
+ spec->init_verbs[spec->num_init_verbs++] =
+ preset->init_verbs[i];
+
+ spec->channel_mode = preset->channel_mode;
+ spec->num_channel_mode = preset->num_channel_mode;
+ spec->need_dac_fix = preset->need_dac_fix;
+
+ spec->multiout.max_channels = spec->channel_mode[0].channels;
+
+ spec->multiout.num_dacs = preset->num_dacs;
+ spec->multiout.dac_nids = preset->dac_nids;
+ spec->multiout.dig_out_nid = preset->dig_out_nid;
+ spec->multiout.hp_nid = preset->hp_nid;
+
+ spec->num_mux_defs = preset->num_mux_defs;
+ if (!spec->num_mux_defs)
+ spec->num_mux_defs = 1;
+ spec->input_mux = preset->input_mux;
+
+ spec->num_adc_nids = preset->num_adc_nids;
+ spec->adc_nids = preset->adc_nids;
+ spec->capsrc_nids = preset->capsrc_nids;
+ spec->dig_in_nid = preset->dig_in_nid;
+
+ spec->unsol_event = preset->unsol_event;
+ spec->init_hook = preset->init_hook;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = preset->loopbacks;
+#endif
+}
+
+/* Enable GPIO mask and set output */
+static struct hda_verb alc_gpio1_init_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+ { }
+};
+
+static struct hda_verb alc_gpio2_init_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02},
+ { }
+};
+
+static struct hda_verb alc_gpio3_init_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x03},
+ { }
+};
+
+/*
+ * Fix hardware PLL issue
+ * On some codecs, the analog PLL gating control must be off while
+ * the default value is 1.
+ */
+static void alc_fix_pll(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int val;
+
+ if (!spec->pll_nid)
+ return;
+ snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
+ spec->pll_coef_idx);
+ val = snd_hda_codec_read(codec, spec->pll_nid, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX,
+ spec->pll_coef_idx);
+ snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF,
+ val & ~(1 << spec->pll_coef_bit));
+}
+
+static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int coef_idx, unsigned int coef_bit)
+{
+ struct alc_spec *spec = codec->spec;
+ spec->pll_nid = nid;
+ spec->pll_coef_idx = coef_idx;
+ spec->pll_coef_bit = coef_bit;
+ alc_fix_pll(codec);
+}
+
+static void alc_sku_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int present;
+ unsigned int hp_nid = spec->autocfg.hp_pins[0];
+ unsigned int sp_nid = spec->autocfg.speaker_pins[0];
+
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, hp_nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
+ snd_hda_codec_write(codec, sp_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->jack_present ? 0 : PIN_OUT);
+}
+
+#if 0 /* it's broken in some acses -- temporarily disabled */
+static void alc_mic_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int present;
+ unsigned int mic_nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
+ unsigned int fmic_nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
+ unsigned int mix_nid = spec->capsrc_nids[0];
+ unsigned int capsrc_idx_mic, capsrc_idx_fmic;
+
+ capsrc_idx_mic = mic_nid - 0x18;
+ capsrc_idx_fmic = fmic_nid - 0x18;
+ present = snd_hda_codec_read(codec, mic_nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (capsrc_idx_mic << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (capsrc_idx_fmic << 8) | (present ? 0x80 : 0));
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+#else
+#define alc_mic_automute(codec) /* NOP */
+#endif /* disabled */
+
+/* unsolicited event for HP jack sensing */
+static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if (codec->vendor_id == 0x10ec0880)
+ res >>= 28;
+ else
+ res >>= 26;
+ if (res == ALC880_HP_EVENT)
+ alc_sku_automute(codec);
+
+ if (res == ALC880_MIC_EVENT)
+ alc_mic_automute(codec);
+}
+
+static void alc_inithook(struct hda_codec *codec)
+{
+ alc_sku_automute(codec);
+ alc_mic_automute(codec);
+}
+
+/* additional initialization for ALC888 variants */
+static void alc888_coef_init(struct hda_codec *codec)
+{
+ unsigned int tmp;
+
+ snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0);
+ tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
+ if ((tmp & 0xf0) == 2)
+ /* alc888S-VC */
+ snd_hda_codec_read(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF, 0x830);
+ else
+ /* alc888-VB */
+ snd_hda_codec_read(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF, 0x3030);
+}
+
+/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
+ * 31 ~ 16 : Manufacture ID
+ * 15 ~ 8 : SKU ID
+ * 7 ~ 0 : Assembly ID
+ * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
+ */
+static void alc_subsystem_id(struct hda_codec *codec,
+ unsigned int porta, unsigned int porte,
+ unsigned int portd)
+{
+ unsigned int ass, tmp, i;
+ unsigned nid;
+ struct alc_spec *spec = codec->spec;
+
+ ass = codec->subsystem_id & 0xffff;
+ if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
+ goto do_sku;
+
+ /*
+ * 31~30 : port conetcivity
+ * 29~21 : reserve
+ * 20 : PCBEEP input
+ * 19~16 : Check sum (15:1)
+ * 15~1 : Custom
+ * 0 : override
+ */
+ nid = 0x1d;
+ if (codec->vendor_id == 0x10ec0260)
+ nid = 0x17;
+ ass = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ if (!(ass & 1) && !(ass & 0x100000))
+ return;
+ if ((ass >> 30) != 1) /* no physical connection */
+ return;
+
+ /* check sum */
+ tmp = 0;
+ for (i = 1; i < 16; i++) {
+ if ((ass >> i) & 1)
+ tmp++;
+ }
+ if (((ass >> 16) & 0xf) != tmp)
+ return;
+do_sku:
+ /*
+ * 0 : override
+ * 1 : Swap Jack
+ * 2 : 0 --> Desktop, 1 --> Laptop
+ * 3~5 : External Amplifier control
+ * 7~6 : Reserved
+ */
+ tmp = (ass & 0x38) >> 3; /* external Amp control */
+ switch (tmp) {
+ case 1:
+ snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
+ break;
+ case 3:
+ snd_hda_sequence_write(codec, alc_gpio2_init_verbs);
+ break;
+ case 7:
+ snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
+ break;
+ case 5: /* set EAPD output high */
+ switch (codec->vendor_id) {
+ case 0x10ec0260:
+ snd_hda_codec_write(codec, 0x0f, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 2);
+ snd_hda_codec_write(codec, 0x10, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 2);
+ break;
+ case 0x10ec0262:
+ case 0x10ec0267:
+ case 0x10ec0268:
+ case 0x10ec0269:
+ case 0x10ec0272:
+ case 0x10ec0660:
+ case 0x10ec0662:
+ case 0x10ec0663:
+ case 0x10ec0862:
+ case 0x10ec0889:
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 2);
+ snd_hda_codec_write(codec, 0x15, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 2);
+ break;
+ }
+ switch (codec->vendor_id) {
+ case 0x10ec0260:
+ snd_hda_codec_write(codec, 0x1a, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x1a, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x1a, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x1a, 0,
+ AC_VERB_SET_PROC_COEF,
+ tmp | 0x2010);
+ break;
+ case 0x10ec0262:
+ case 0x10ec0880:
+ case 0x10ec0882:
+ case 0x10ec0883:
+ case 0x10ec0885:
+ case 0x10ec0887:
+ case 0x10ec0889:
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x20, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF,
+ tmp | 0x2010);
+ break;
+ case 0x10ec0888:
+ /*alc888_coef_init(codec);*/ /* called in alc_init() */
+ break;
+ case 0x10ec0267:
+ case 0x10ec0268:
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x20, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF,
+ tmp | 0x3000);
+ break;
+ }
+ default:
+ break;
+ }
+
+ /* is laptop or Desktop and enable the function "Mute internal speaker
+ * when the external headphone out jack is plugged"
+ */
+ if (!(ass & 0x8000))
+ return;
+ /*
+ * 10~8 : Jack location
+ * 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered
+ * 14~13: Resvered
+ * 15 : 1 --> enable the function "Mute internal speaker
+ * when the external headphone out jack is plugged"
+ */
+ if (!spec->autocfg.speaker_pins[0]) {
+ if (spec->autocfg.line_out_pins[0])
+ spec->autocfg.speaker_pins[0] =
+ spec->autocfg.line_out_pins[0];
+ else
+ return;
+ }
+
+ if (!spec->autocfg.hp_pins[0]) {
+ tmp = (ass >> 11) & 0x3; /* HP to chassis */
+ if (tmp == 0)
+ spec->autocfg.hp_pins[0] = porta;
+ else if (tmp == 1)
+ spec->autocfg.hp_pins[0] = porte;
+ else if (tmp == 2)
+ spec->autocfg.hp_pins[0] = portd;
+ else
+ return;
+ }
+ if (spec->autocfg.hp_pins[0])
+ snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC880_HP_EVENT);
+
+#if 0 /* it's broken in some acses -- temporarily disabled */
+ if (spec->autocfg.input_pins[AUTO_PIN_MIC] &&
+ spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC])
+ snd_hda_codec_write(codec,
+ spec->autocfg.input_pins[AUTO_PIN_MIC], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC880_MIC_EVENT);
+#endif /* disabled */
+
+ spec->unsol_event = alc_sku_unsol_event;
+}
+
+/*
+ * Fix-up pin default configurations
+ */
+
+struct alc_pincfg {
+ hda_nid_t nid;
+ u32 val;
+};
+
+static void alc_fix_pincfg(struct hda_codec *codec,
+ const struct snd_pci_quirk *quirk,
+ const struct alc_pincfg **pinfix)
+{
+ const struct alc_pincfg *cfg;
+
+ quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
+ if (!quirk)
+ return;
+
+ cfg = pinfix[quirk->value];
+ for (; cfg->nid; cfg++) {
+ int i;
+ u32 val = cfg->val;
+ for (i = 0; i < 4; i++) {
+ snd_hda_codec_write(codec, cfg->nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
+ val & 0xff);
+ val >>= 8;
+ }
+ }
+}
+
+/*
+ * ALC880 3-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
+ * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
+ * F-Mic = 0x1b, HP = 0x19
+ */
+
+static hda_nid_t alc880_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x05, 0x04, 0x03
+};
+
+static hda_nid_t alc880_adc_nids[3] = {
+ /* ADC0-2 */
+ 0x07, 0x08, 0x09,
+};
+
+/* The datasheet says the node 0x07 is connected from inputs,
+ * but it shows zero connection in the real implementation on some devices.
+ * Note: this is a 915GAV bug, fixed on 915GLV
+ */
+static hda_nid_t alc880_adc_nids_alt[2] = {
+ /* ADC1-2 */
+ 0x08, 0x09,
+};
+
+#define ALC880_DIGOUT_NID 0x06
+#define ALC880_DIGIN_NID 0x0a
+
+static struct hda_input_mux alc880_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x3 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* channel source setting (2/6 channel selection for 3-stack) */
+/* 2ch mode */
+static struct hda_verb alc880_threestack_ch2_init[] = {
+ /* set line-in to input, mute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ /* set mic-in to input vref 80%, mute it */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/* 6ch mode */
+static struct hda_verb alc880_threestack_ch6_init[] = {
+ /* set line-in to output, unmute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ /* set mic-in to output, unmute it */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc880_threestack_modes[2] = {
+ { 2, alc880_threestack_ch2_init },
+ { 6, alc880_threestack_ch6_init },
+};
+
+static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc880_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 3,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* capture mixer elements (in case NID 0x07 not available) */
+static struct snd_kcontrol_new alc880_capture_alt_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+
+
+/*
+ * ALC880 5-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
+ * Side = 0x02 (0xd)
+ * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
+ * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
+ */
+
+/* additional mixers to alc880_three_stack_mixer */
+static struct snd_kcontrol_new alc880_five_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
+ { } /* end */
+};
+
+/* channel source setting (6/8 channel selection for 5-stack) */
+/* 6ch mode */
+static struct hda_verb alc880_fivestack_ch6_init[] = {
+ /* set line-in to input, mute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/* 8ch mode */
+static struct hda_verb alc880_fivestack_ch8_init[] = {
+ /* set line-in to output, unmute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc880_fivestack_modes[2] = {
+ { 6, alc880_fivestack_ch6_init },
+ { 8, alc880_fivestack_ch8_init },
+};
+
+
+/*
+ * ALC880 6-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
+ * Side = 0x05 (0x0f)
+ * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
+ * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
+ */
+
+static hda_nid_t alc880_6st_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+
+static struct hda_input_mux alc880_6stack_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* fixed 8-channels */
+static struct hda_channel_mode alc880_sixstack_modes[1] = {
+ { 8, NULL },
+};
+
+static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+
+/*
+ * ALC880 W810 model
+ *
+ * W810 has rear IO for:
+ * Front (DAC 02)
+ * Surround (DAC 03)
+ * Center/LFE (DAC 04)
+ * Digital out (06)
+ *
+ * The system also has a pair of internal speakers, and a headphone jack.
+ * These are both connected to Line2 on the codec, hence to DAC 02.
+ *
+ * There is a variable resistor to control the speaker or headphone
+ * volume. This is a hardware-only device without a software API.
+ *
+ * Plugging headphones in will disable the internal speakers. This is
+ * implemented in hardware, not via the driver using jack sense. In
+ * a similar fashion, plugging into the rear socket marked "front" will
+ * disable both the speakers and headphones.
+ *
+ * For input, there's a microphone jack, and an "audio in" jack.
+ * These may not do anything useful with this driver yet, because I
+ * haven't setup any initialization verbs for these yet...
+ */
+
+static hda_nid_t alc880_w810_dac_nids[3] = {
+ /* front, rear/surround, clfe */
+ 0x02, 0x03, 0x04
+};
+
+/* fixed 6 channels */
+static struct hda_channel_mode alc880_w810_modes[1] = {
+ { 6, NULL }
+};
+
+/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
+static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+
+/*
+ * Z710V model
+ *
+ * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
+ * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
+ * Line = 0x1a
+ */
+
+static hda_nid_t alc880_z71v_dac_nids[1] = {
+ 0x02
+};
+#define ALC880_Z71V_HP_DAC 0x03
+
+/* fixed 2 channels */
+static struct hda_channel_mode alc880_2_jack_modes[1] = {
+ { 2, NULL }
+};
+
+static struct snd_kcontrol_new alc880_z71v_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+/*
+ * ALC880 F1734 model
+ *
+ * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
+ * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
+ */
+
+static hda_nid_t alc880_f1734_dac_nids[1] = {
+ 0x03
+};
+#define ALC880_F1734_HP_DAC 0x02
+
+static struct snd_kcontrol_new alc880_f1734_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_input_mux alc880_f1734_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
+
+/*
+ * ALC880 ASUS model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ * Mic = 0x18, Line = 0x1a
+ */
+
+#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
+#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
+
+static struct snd_kcontrol_new alc880_asus_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/*
+ * ALC880 ASUS W1V model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ * Mic = 0x18, Line = 0x1a, Line2 = 0x1b
+ */
+
+/* additional mixers to alc880_asus_mixer */
+static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
+ HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
+ { } /* end */
+};
+
+/* additional mixers to alc880_asus_mixer */
+static struct snd_kcontrol_new alc880_pcbeep_mixer[] = {
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+/* TCL S700 */
+static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* Uniwill */
+static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+/*
+ * virtual master controls
+ */
+
+/*
+ * slave controls for virtual master
+ */
+static const char *alc_slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Speaker Playback Volume",
+ "Mono Playback Volume",
+ "Line-Out Playback Volume",
+ NULL,
+};
+
+static const char *alc_slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Speaker Playback Switch",
+ "Mono Playback Switch",
+ "IEC958 Playback Switch",
+ NULL,
+};
+
+/*
+ * build control elements
+ */
+static int alc_build_controls(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ int i;
+
+ for (i = 0; i < spec->num_mixers; i++) {
+ err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->multiout.dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
+ }
+ if (spec->dig_in_nid) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+ if (err < 0)
+ return err;
+ }
+
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ unsigned int vmaster_tlv[4];
+ snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
+ HDA_OUTPUT, vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ vmaster_tlv, alc_slave_vols);
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, alc_slave_sws);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+
+/*
+ * initialize the codec volumes, etc
+ */
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc880_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for front
+ * panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0f)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ { }
+};
+
+/*
+ * 3-stack pin configuration:
+ * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
+ */
+static struct hda_verb alc880_pin_3stack_init_verbs[] = {
+ /*
+ * preset connection lists of input pins
+ * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
+ */
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
+
+ /*
+ * Set pin mode and muting
+ */
+ /* set front pin widgets 0x14 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line2 (as front mic) pin widget for input and vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * 5-stack pin configuration:
+ * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
+ * line-in/side = 0x1a, f-mic = 0x1b
+ */
+static struct hda_verb alc880_pin_5stack_init_verbs[] = {
+ /*
+ * preset connection lists of input pins
+ * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
+ */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
+
+ /*
+ * Set pin mode and muting
+ */
+ /* set pin widgets 0x14-0x17 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* unmute pins for output (no gain on this amp) */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line2 (as front mic) pin widget for input and vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * W810 pin configuration:
+ * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
+ */
+static struct hda_verb alc880_pin_w810_init_verbs[] = {
+ /* hphone/speaker input selector: front DAC */
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ { }
+};
+
+/*
+ * Z71V pin configuration:
+ * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
+ */
+static struct hda_verb alc880_pin_z71v_init_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * 6-stack pin configuration:
+ * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
+ * f-mic = 0x19, line = 0x1a, HP = 0x1b
+ */
+static struct hda_verb alc880_pin_6stack_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * Uniwill pin configuration:
+ * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
+ * line = 0x1a
+ */
+static struct hda_verb alc880_uniwill_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */
+ /* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+
+ { }
+};
+
+/*
+* Uniwill P53
+* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
+ */
+static struct hda_verb alc880_uniwill_p53_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_DCVOL_EVENT},
+
+ { }
+};
+
+static struct hda_verb alc880_beep_init_verbs[] = {
+ { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_uniwill_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+/* auto-toggle front mic */
+static void alc880_uniwill_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
+}
+
+static void alc880_uniwill_automute(struct hda_codec *codec)
+{
+ alc880_uniwill_hp_automute(codec);
+ alc880_uniwill_mic_automute(codec);
+}
+
+static void alc880_uniwill_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ switch (res >> 28) {
+ case ALC880_HP_EVENT:
+ alc880_uniwill_hp_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc880_uniwill_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits);
+}
+
+static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
+ present &= HDA_AMP_VOLMASK;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
+ HDA_AMP_VOLMASK, present);
+ snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
+ HDA_AMP_VOLMASK, present);
+}
+
+static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ if ((res >> 28) == ALC880_HP_EVENT)
+ alc880_uniwill_p53_hp_automute(codec);
+ if ((res >> 28) == ALC880_DCVOL_EVENT)
+ alc880_uniwill_p53_dcvol_automute(codec);
+}
+
+/*
+ * F1734 pin configuration:
+ * HP = 0x14, speaker-out = 0x15, mic = 0x18
+ */
+static struct hda_verb alc880_pin_f1734_init_verbs[] = {
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_DCVOL_EVENT},
+
+ { }
+};
+
+/*
+ * ASUS pin configuration:
+ * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
+ */
+static struct hda_verb alc880_pin_asus_init_verbs[] = {
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/* Enable GPIO mask and set output */
+#define alc880_gpio1_init_verbs alc_gpio1_init_verbs
+#define alc880_gpio2_init_verbs alc_gpio2_init_verbs
+
+/* Clevo m520g init */
+static struct hda_verb alc880_pin_clevo_init_verbs[] = {
+ /* headphone output */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* line-out */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line-in */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* CD */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Mic1 (rear panel) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Mic2 (front panel) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* headphone */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
+ { }
+};
+
+static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
+ /* Headphone output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Front output*/
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
+
+ { }
+};
+
+/*
+ * LG m1 express dual
+ *
+ * Pin assignment:
+ * Rear Line-In/Out (blue): 0x14
+ * Build-in Mic-In: 0x15
+ * Speaker-out: 0x17
+ * HP-Out (green): 0x1b
+ * Mic-In/Out (red): 0x19
+ * SPDIF-Out: 0x1e
+ */
+
+/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
+static hda_nid_t alc880_lg_dac_nids[3] = {
+ 0x05, 0x02, 0x03
+};
+
+/* seems analog CD is not working */
+static struct hda_input_mux alc880_lg_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x5 },
+ { "Internal Mic", 0x6 },
+ },
+};
+
+/* 2,4,6 channel modes */
+static struct hda_verb alc880_lg_ch2_init[] = {
+ /* set line-in and mic-in to input */
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { }
+};
+
+static struct hda_verb alc880_lg_ch4_init[] = {
+ /* set line-in to out and mic-in to input */
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { }
+};
+
+static struct hda_verb alc880_lg_ch6_init[] = {
+ /* set line-in and mic-in to output */
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { }
+};
+
+static struct hda_channel_mode alc880_lg_ch_modes[3] = {
+ { 2, alc880_lg_ch2_init },
+ { 4, alc880_lg_ch4_init },
+ { 6, alc880_lg_ch6_init },
+};
+
+static struct snd_kcontrol_new alc880_lg_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc880_lg_init_verbs[] = {
+ /* set capture source to mic-in */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* mute all amp mixer inputs */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* line-in to input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* built-in mic */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* speaker-out */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* mic-in to input */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* HP-out */
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* jack sense */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ if ((res >> 28) == 0x01)
+ alc880_lg_automute(codec);
+}
+
+/*
+ * LG LW20
+ *
+ * Pin assignment:
+ * Speaker-out: 0x14
+ * Mic-In: 0x18
+ * Built-in Mic-In: 0x19
+ * Line-In: 0x1b
+ * HP-Out: 0x1a
+ * SPDIF-Out: 0x1e
+ */
+
+static struct hda_input_mux alc880_lg_lw_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "Line In", 0x2 },
+ },
+};
+
+#define alc880_lg_lw_modes alc880_threestack_modes
+
+static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc880_lg_lw_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
+
+ /* set capture source to mic-in */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /* speaker-out */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* HP-out */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* mic-in to input */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* built-in mic */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* jack sense */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_lw_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ if ((res >> 28) == 0x01)
+ alc880_lg_lw_automute(codec);
+}
+
+static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_input_mux alc880_medion_rim_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ },
+};
+
+static struct hda_verb alc880_medion_rim_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Internal Speaker */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_medion_rim_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ if (present)
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
+ else
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
+}
+
+static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 28 bit!
+ */
+ if ((res >> 28) == ALC880_HP_EVENT)
+ alc880_medion_rim_automute(codec);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc880_loopbacks[] = {
+ { 0x0b, HDA_INPUT, 0 },
+ { 0x0b, HDA_INPUT, 1 },
+ { 0x0b, HDA_INPUT, 2 },
+ { 0x0b, HDA_INPUT, 3 },
+ { 0x0b, HDA_INPUT, 4 },
+ { } /* end */
+};
+
+static struct hda_amp_list alc880_lg_loopbacks[] = {
+ { 0x0b, HDA_INPUT, 1 },
+ { 0x0b, HDA_INPUT, 6 },
+ { 0x0b, HDA_INPUT, 7 },
+ { } /* end */
+};
+#endif
+
+/*
+ * Common callbacks
+ */
+
+static int alc_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int i;
+
+ alc_fix_pll(codec);
+ if (codec->vendor_id == 0x10ec0888)
+ alc888_coef_init(codec);
+
+ for (i = 0; i < spec->num_init_verbs; i++)
+ snd_hda_sequence_write(codec, spec->init_verbs[i]);
+
+ if (spec->init_hook)
+ spec->init_hook(codec);
+
+ return 0;
+}
+
+static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (spec->unsol_event)
+ spec->unsol_event(codec, res);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
+/*
+ * Analog playback callbacks
+ */
+static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
+
+static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
+
+static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int alc880_alt_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number + 1],
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int alc880_alt_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+
+ snd_hda_codec_cleanup_stream(codec,
+ spec->adc_nids[substream->number + 1]);
+ return 0;
+}
+
+
+/*
+ */
+static struct hda_pcm_stream alc880_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 8,
+ /* NID is set in alc_build_pcms */
+ .ops = {
+ .open = alc880_playback_pcm_open,
+ .prepare = alc880_playback_pcm_prepare,
+ .cleanup = alc880_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream alc880_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
+static struct hda_pcm_stream alc880_pcm_analog_alt_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
+static struct hda_pcm_stream alc880_pcm_analog_alt_capture = {
+ .substreams = 2, /* can be overridden */
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+ .ops = {
+ .prepare = alc880_alt_capture_pcm_prepare,
+ .cleanup = alc880_alt_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream alc880_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+ .ops = {
+ .open = alc880_dig_playback_pcm_open,
+ .close = alc880_dig_playback_pcm_close,
+ .prepare = alc880_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream alc880_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
+/* Used by alc_build_pcms to flag that a PCM has no playback stream */
+static struct hda_pcm_stream alc_pcm_null_stream = {
+ .substreams = 0,
+ .channels_min = 0,
+ .channels_max = 0,
+};
+
+static int alc_build_pcms(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+ int i;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = spec->stream_name_analog;
+ if (spec->stream_analog_playback) {
+ if (snd_BUG_ON(!spec->multiout.dac_nids))
+ return -EINVAL;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
+ }
+ if (spec->stream_analog_capture) {
+ if (snd_BUG_ON(!spec->adc_nids))
+ return -EINVAL;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+ }
+
+ if (spec->channel_mode) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0;
+ for (i = 0; i < spec->num_channel_mode; i++) {
+ if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels;
+ }
+ }
+ }
+
+ /* SPDIF for stream index #1 */
+ if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+ codec->num_pcms = 2;
+ info = spec->pcm_rec + 1;
+ info->name = spec->stream_name_digital;
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->multiout.dig_out_nid &&
+ spec->stream_digital_playback) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+ }
+ if (spec->dig_in_nid &&
+ spec->stream_digital_capture) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture);
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+ }
+ /* FIXME: do we need this for all Realtek codec models? */
+ codec->spdif_status_reset = 1;
+ }
+
+ /* If the use of more than one ADC is requested for the current
+ * model, configure a second analog capture-only PCM.
+ */
+ /* Additional Analaog capture for index #2 */
+ if ((spec->alt_dac_nid && spec->stream_analog_alt_playback) ||
+ (spec->num_adc_nids > 1 && spec->stream_analog_alt_capture)) {
+ codec->num_pcms = 3;
+ info = spec->pcm_rec + 2;
+ info->name = spec->stream_name_analog;
+ if (spec->alt_dac_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ *spec->stream_analog_alt_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->alt_dac_nid;
+ } else {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ alc_pcm_null_stream;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0;
+ }
+ if (spec->num_adc_nids > 1) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ *spec->stream_analog_alt_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+ spec->adc_nids[1];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams =
+ spec->num_adc_nids - 1;
+ } else {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ alc_pcm_null_stream;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = 0;
+ }
+ }
+
+ return 0;
+}
+
+static void alc_free(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int i;
+
+ if (!spec)
+ return;
+
+ if (spec->kctl_alloc) {
+ for (i = 0; i < spec->num_kctl_used; i++)
+ kfree(spec->kctl_alloc[i].name);
+ kfree(spec->kctl_alloc);
+ }
+ kfree(spec);
+ codec->spec = NULL; /* to be sure */
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+static void store_pin_configs(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid, end_nid;
+
+ end_nid = codec->start_nid + codec->num_nodes;
+ for (nid = codec->start_nid; nid < end_nid; nid++) {
+ unsigned int wid_caps = get_wcaps(codec, nid);
+ unsigned int wid_type =
+ (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wid_type != AC_WID_PIN)
+ continue;
+ if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids))
+ break;
+ spec->pin_nids[spec->num_pins] = nid;
+ spec->pin_cfgs[spec->num_pins] =
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ spec->num_pins++;
+ }
+}
+
+static void resume_pin_configs(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_pins; i++) {
+ hda_nid_t pin_nid = spec->pin_nids[i];
+ unsigned int pin_config = spec->pin_cfgs[i];
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+ pin_config & 0x000000ff);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+ (pin_config & 0x0000ff00) >> 8);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+ (pin_config & 0x00ff0000) >> 16);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+ pin_config >> 24);
+ }
+}
+
+static int alc_resume(struct hda_codec *codec)
+{
+ resume_pin_configs(codec);
+ codec->patch_ops.init(codec);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ return 0;
+}
+#else
+#define store_pin_configs(codec)
+#endif
+
+/*
+ */
+static struct hda_codec_ops alc_patch_ops = {
+ .build_controls = alc_build_controls,
+ .build_pcms = alc_build_pcms,
+ .init = alc_init,
+ .free = alc_free,
+ .unsol_event = alc_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+ .resume = alc_resume,
+#endif
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = alc_check_power_status,
+#endif
+};
+
+
+/*
+ * Test configuration for debugging
+ *
+ * Almost all inputs/outputs are enabled. I/O pins can be configured via
+ * enum controls.
+ */
+#ifdef CONFIG_SND_DEBUG
+static hda_nid_t alc880_test_dac_nids[4] = {
+ 0x02, 0x03, 0x04, 0x05
+};
+
+static struct hda_input_mux alc880_test_capture_source = {
+ .num_items = 7,
+ .items = {
+ { "In-1", 0x0 },
+ { "In-2", 0x1 },
+ { "In-3", 0x2 },
+ { "In-4", 0x3 },
+ { "CD", 0x4 },
+ { "Front", 0x5 },
+ { "Surround", 0x6 },
+ },
+};
+
+static struct hda_channel_mode alc880_test_modes[4] = {
+ { 2, NULL },
+ { 4, NULL },
+ { 6, NULL },
+ { 8, NULL },
+};
+
+static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {
+ "N/A", "Line Out", "HP Out",
+ "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item >= 8)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int pin_ctl, item = 0;
+
+ pin_ctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (pin_ctl & AC_PINCTL_OUT_EN) {
+ if (pin_ctl & AC_PINCTL_HP_EN)
+ item = 2;
+ else
+ item = 1;
+ } else if (pin_ctl & AC_PINCTL_IN_EN) {
+ switch (pin_ctl & AC_PINCTL_VREFEN) {
+ case AC_PINCTL_VREF_HIZ: item = 3; break;
+ case AC_PINCTL_VREF_50: item = 4; break;
+ case AC_PINCTL_VREF_GRD: item = 5; break;
+ case AC_PINCTL_VREF_80: item = 6; break;
+ case AC_PINCTL_VREF_100: item = 7; break;
+ }
+ }
+ ucontrol->value.enumerated.item[0] = item;
+ return 0;
+}
+
+static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ static unsigned int ctls[] = {
+ 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
+ };
+ unsigned int old_ctl, new_ctl;
+
+ old_ctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ new_ctl = ctls[ucontrol->value.enumerated.item[0]];
+ if (old_ctl != new_ctl) {
+ int val;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ new_ctl);
+ val = ucontrol->value.enumerated.item[0] >= 3 ?
+ HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, val);
+ return 1;
+ }
+ return 0;
+}
+
+static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {
+ "Front", "Surround", "CLFE", "Side"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= 4)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int sel;
+
+ sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
+ ucontrol->value.enumerated.item[0] = sel & 3;
+ return 0;
+}
+
+static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int sel;
+
+ sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
+ if (ucontrol->value.enumerated.item[0] != sel) {
+ sel = ucontrol->value.enumerated.item[0] & 3;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, sel);
+ return 1;
+ }
+ return 0;
+}
+
+#define PIN_CTL_TEST(xname,nid) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .info = alc_test_pin_ctl_info, \
+ .get = alc_test_pin_ctl_get, \
+ .put = alc_test_pin_ctl_put, \
+ .private_value = nid \
+ }
+
+#define PIN_SRC_TEST(xname,nid) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .info = alc_test_pin_src_info, \
+ .get = alc_test_pin_src_get, \
+ .put = alc_test_pin_src_put, \
+ .private_value = nid \
+ }
+
+static struct snd_kcontrol_new alc880_test_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ PIN_CTL_TEST("Front Pin Mode", 0x14),
+ PIN_CTL_TEST("Surround Pin Mode", 0x15),
+ PIN_CTL_TEST("CLFE Pin Mode", 0x16),
+ PIN_CTL_TEST("Side Pin Mode", 0x17),
+ PIN_CTL_TEST("In-1 Pin Mode", 0x18),
+ PIN_CTL_TEST("In-2 Pin Mode", 0x19),
+ PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
+ PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
+ PIN_SRC_TEST("In-1 Pin Source", 0x18),
+ PIN_SRC_TEST("In-2 Pin Source", 0x19),
+ PIN_SRC_TEST("In-3 Pin Source", 0x1a),
+ PIN_SRC_TEST("In-4 Pin Source", 0x1b),
+ HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc880_test_init_verbs[] = {
+ /* Unmute inputs of 0x0c - 0x0f */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Vol output for 0x0c-0x0f */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Set output pins 0x14-0x17 */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Unmute output pins 0x14-0x17 */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Set input pins 0x18-0x1c */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Mute input pins 0x18-0x1b */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* ADC set up */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Analog input/passthru */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ { }
+};
+#endif
+
+/*
+ */
+
+static const char *alc880_models[ALC880_MODEL_LAST] = {
+ [ALC880_3ST] = "3stack",
+ [ALC880_TCL_S700] = "tcl",
+ [ALC880_3ST_DIG] = "3stack-digout",
+ [ALC880_CLEVO] = "clevo",
+ [ALC880_5ST] = "5stack",
+ [ALC880_5ST_DIG] = "5stack-digout",
+ [ALC880_W810] = "w810",
+ [ALC880_Z71V] = "z71v",
+ [ALC880_6ST] = "6stack",
+ [ALC880_6ST_DIG] = "6stack-digout",
+ [ALC880_ASUS] = "asus",
+ [ALC880_ASUS_W1V] = "asus-w1v",
+ [ALC880_ASUS_DIG] = "asus-dig",
+ [ALC880_ASUS_DIG2] = "asus-dig2",
+ [ALC880_UNIWILL_DIG] = "uniwill",
+ [ALC880_UNIWILL_P53] = "uniwill-p53",
+ [ALC880_FUJITSU] = "fujitsu",
+ [ALC880_F1734] = "F1734",
+ [ALC880_LG] = "lg",
+ [ALC880_LG_LW] = "lg-lw",
+ [ALC880_MEDION_RIM] = "medion",
+#ifdef CONFIG_SND_DEBUG
+ [ALC880_TEST] = "test",
+#endif
+ [ALC880_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc880_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810),
+ SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST),
+ SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
+ SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
+ SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
+ SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V),
+ /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
+ SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
+ SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
+ SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
+ SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */
+ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
+ SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
+ SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
+ SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST),
+ SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST),
+ SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO),
+ SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO),
+ SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2),
+ SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG),
+ SND_PCI_QUIRK(0x1584, 0x9054, "Uniwlll", ALC880_F1734),
+ SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
+ SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
+ SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
+ SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM),
+ SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
+ SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
+ SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL),
+ SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
+ SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW),
+ SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
+ SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
+ SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_LG_LW),
+ SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
+ SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */
+ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */
+ SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
+ SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
+ {}
+};
+
+/*
+ * ALC880 codec presets
+ */
+static struct alc_config_preset alc880_presets[] = {
+ [ALC880_3ST] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_3ST_DIG] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_TCL_S700] = {
+ .mixers = { alc880_tcl_s700_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_tcl_S700_init_verbs,
+ alc880_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_5ST] = {
+ .mixers = { alc880_three_stack_mixer,
+ alc880_five_stack_mixer},
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_5stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+ .channel_mode = alc880_fivestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_5ST_DIG] = {
+ .mixers = { alc880_three_stack_mixer,
+ alc880_five_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_5stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+ .channel_mode = alc880_fivestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_6ST] = {
+ .mixers = { alc880_six_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_6stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+ .dac_nids = alc880_6st_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+ .channel_mode = alc880_sixstack_modes,
+ .input_mux = &alc880_6stack_capture_source,
+ },
+ [ALC880_6ST_DIG] = {
+ .mixers = { alc880_six_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_6stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+ .dac_nids = alc880_6st_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+ .channel_mode = alc880_sixstack_modes,
+ .input_mux = &alc880_6stack_capture_source,
+ },
+ [ALC880_W810] = {
+ .mixers = { alc880_w810_base_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_w810_init_verbs,
+ alc880_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
+ .dac_nids = alc880_w810_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
+ .channel_mode = alc880_w810_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_Z71V] = {
+ .mixers = { alc880_z71v_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_z71v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
+ .dac_nids = alc880_z71v_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_F1734] = {
+ .mixers = { alc880_f1734_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_f1734_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
+ .dac_nids = alc880_f1734_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_f1734_capture_source,
+ .unsol_event = alc880_uniwill_p53_unsol_event,
+ .init_hook = alc880_uniwill_p53_hp_automute,
+ },
+ [ALC880_ASUS] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_DIG] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_DIG2] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio2_init_verbs }, /* use GPIO2 */
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_W1V] = {
+ .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_UNIWILL_DIG] = {
+ .mixers = { alc880_asus_mixer, alc880_pcbeep_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_asus_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_UNIWILL] = {
+ .mixers = { alc880_uniwill_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_uniwill_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ .unsol_event = alc880_uniwill_unsol_event,
+ .init_hook = alc880_uniwill_automute,
+ },
+ [ALC880_UNIWILL_P53] = {
+ .mixers = { alc880_uniwill_p53_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_uniwill_p53_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
+ .channel_mode = alc880_threestack_modes,
+ .input_mux = &alc880_capture_source,
+ .unsol_event = alc880_uniwill_p53_unsol_event,
+ .init_hook = alc880_uniwill_p53_hp_automute,
+ },
+ [ALC880_FUJITSU] = {
+ .mixers = { alc880_fujitsu_mixer,
+ alc880_pcbeep_mixer, },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_uniwill_p53_init_verbs,
+ alc880_beep_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ .unsol_event = alc880_uniwill_p53_unsol_event,
+ .init_hook = alc880_uniwill_p53_hp_automute,
+ },
+ [ALC880_CLEVO] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_pin_clevo_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_LG] = {
+ .mixers = { alc880_lg_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_lg_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
+ .dac_nids = alc880_lg_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
+ .channel_mode = alc880_lg_ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc880_lg_capture_source,
+ .unsol_event = alc880_lg_unsol_event,
+ .init_hook = alc880_lg_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .loopbacks = alc880_lg_loopbacks,
+#endif
+ },
+ [ALC880_LG_LW] = {
+ .mixers = { alc880_lg_lw_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_lg_lw_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes),
+ .channel_mode = alc880_lg_lw_modes,
+ .input_mux = &alc880_lg_lw_capture_source,
+ .unsol_event = alc880_lg_lw_unsol_event,
+ .init_hook = alc880_lg_lw_automute,
+ },
+ [ALC880_MEDION_RIM] = {
+ .mixers = { alc880_medion_rim_mixer },
+ .init_verbs = { alc880_volume_init_verbs,
+ alc880_medion_rim_init_verbs,
+ alc_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_medion_rim_capture_source,
+ .unsol_event = alc880_medion_rim_unsol_event,
+ .init_hook = alc880_medion_rim_automute,
+ },
+#ifdef CONFIG_SND_DEBUG
+ [ALC880_TEST] = {
+ .mixers = { alc880_test_mixer },
+ .init_verbs = { alc880_test_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
+ .dac_nids = alc880_test_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_test_modes),
+ .channel_mode = alc880_test_modes,
+ .input_mux = &alc880_test_capture_source,
+ },
+#endif
+};
+
+/*
+ * Automatic parse of I/O pins from the BIOS configuration
+ */
+
+#define NUM_CONTROL_ALLOC 32
+#define NUM_VERB_ALLOC 32
+
+enum {
+ ALC_CTL_WIDGET_VOL,
+ ALC_CTL_WIDGET_MUTE,
+ ALC_CTL_BIND_MUTE,
+};
+static struct snd_kcontrol_new alc880_control_templates[] = {
+ HDA_CODEC_VOLUME(NULL, 0, 0, 0),
+ HDA_CODEC_MUTE(NULL, 0, 0, 0),
+ HDA_BIND_MUTE(NULL, 0, 0, 0),
+};
+
+/* add dynamic controls */
+static int add_control(struct alc_spec *spec, int type, const char *name,
+ unsigned long val)
+{
+ struct snd_kcontrol_new *knew;
+
+ if (spec->num_kctl_used >= spec->num_kctl_alloc) {
+ int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
+
+ /* array + terminator */
+ knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL);
+ if (!knew)
+ return -ENOMEM;
+ if (spec->kctl_alloc) {
+ memcpy(knew, spec->kctl_alloc,
+ sizeof(*knew) * spec->num_kctl_alloc);
+ kfree(spec->kctl_alloc);
+ }
+ spec->kctl_alloc = knew;
+ spec->num_kctl_alloc = num;
+ }
+
+ knew = &spec->kctl_alloc[spec->num_kctl_used];
+ *knew = alc880_control_templates[type];
+ knew->name = kstrdup(name, GFP_KERNEL);
+ if (!knew->name)
+ return -ENOMEM;
+ knew->private_value = val;
+ spec->num_kctl_used++;
+ return 0;
+}
+
+#define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17)
+#define alc880_fixed_pin_idx(nid) ((nid) - 0x14)
+#define alc880_is_multi_pin(nid) ((nid) >= 0x18)
+#define alc880_multi_pin_idx(nid) ((nid) - 0x18)
+#define alc880_is_input_pin(nid) ((nid) >= 0x18)
+#define alc880_input_pin_idx(nid) ((nid) - 0x18)
+#define alc880_idx_to_dac(nid) ((nid) + 0x02)
+#define alc880_dac_to_idx(nid) ((nid) - 0x02)
+#define alc880_idx_to_mixer(nid) ((nid) + 0x0c)
+#define alc880_idx_to_selector(nid) ((nid) + 0x10)
+#define ALC880_PIN_CD_NID 0x1c
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int alc880_auto_fill_dac_nids(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ hda_nid_t nid;
+ int assigned[4];
+ int i, j;
+
+ memset(assigned, 0, sizeof(assigned));
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ /* check the pins hardwired to audio widget */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (alc880_is_fixed_pin(nid)) {
+ int idx = alc880_fixed_pin_idx(nid);
+ spec->multiout.dac_nids[i] = alc880_idx_to_dac(idx);
+ assigned[idx] = 1;
+ }
+ }
+ /* left pins can be connect to any audio widget */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (alc880_is_fixed_pin(nid))
+ continue;
+ /* search for an empty channel */
+ for (j = 0; j < cfg->line_outs; j++) {
+ if (!assigned[j]) {
+ spec->multiout.dac_nids[i] =
+ alc880_idx_to_dac(j);
+ assigned[j] = 1;
+ break;
+ }
+ }
+ }
+ spec->multiout.num_dacs = cfg->line_outs;
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = {
+ "Front", "Surround", NULL /*CLFE*/, "Side"
+ };
+ hda_nid_t nid;
+ int i, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (!spec->multiout.dac_nids[i])
+ continue;
+ nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i]));
+ if (i == 2) {
+ /* Center/LFE */
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_BIND_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 2,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_BIND_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 2,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 2,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+/* add playback controls for speaker and HP outputs */
+static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
+ const char *pfx)
+{
+ hda_nid_t nid;
+ int err;
+ char name[32];
+
+ if (!pin)
+ return 0;
+
+ if (alc880_is_fixed_pin(pin)) {
+ nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
+ /* specify the DAC as the extra output */
+ if (!spec->multiout.hp_nid)
+ spec->multiout.hp_nid = nid;
+ else
+ spec->multiout.extra_out_nid[0] = nid;
+ /* control HP volume/switch on the output mixer amp */
+ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin));
+ sprintf(name, "%s Playback Volume", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
+ if (err < 0)
+ return err;
+ } else if (alc880_is_multi_pin(pin)) {
+ /* set manual connection */
+ /* we have only a switch on HP-out PIN */
+ sprintf(name, "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* create input playback/capture controls for the given pin */
+static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
+ const char *ctlname,
+ int idx, hda_nid_t mix_nid)
+{
+ char name[32];
+ int err;
+
+ sprintf(name, "%s Playback Volume", ctlname);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", ctlname);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct hda_input_mux *imux = &spec->private_imux;
+ int i, err, idx;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (alc880_is_input_pin(cfg->input_pins[i])) {
+ idx = alc880_input_pin_idx(cfg->input_pins[i]);
+ err = new_analog_input(spec, cfg->input_pins[i],
+ auto_pin_cfg_labels[i],
+ idx, 0x0b);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label =
+ auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index =
+ alc880_input_pin_idx(cfg->input_pins[i]);
+ imux->num_items++;
+ }
+ }
+ return 0;
+}
+
+static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int pin_type)
+{
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pin_type);
+ /* unmute pin */
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
+}
+
+static void alc880_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ alc_set_pin_output(codec, nid, pin_type);
+ /* need the manual connection? */
+ if (alc880_is_multi_pin(nid)) {
+ struct alc_spec *spec = codec->spec;
+ int idx = alc880_multi_pin_idx(nid);
+ snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
+ AC_VERB_SET_CONNECT_SEL,
+ alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
+ }
+}
+
+static int get_pin_type(int line_out_type)
+{
+ if (line_out_type == AUTO_PIN_HP_OUT)
+ return PIN_HP;
+ else
+ return PIN_OUT;
+}
+
+static void alc880_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
+ for (i = 0; i < spec->autocfg.line_outs; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ alc880_auto_set_output_and_unmute(codec, nid, pin_type, i);
+ }
+}
+
+static void alc880_auto_init_extra_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin) /* connect to front */
+ alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front */
+ alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+}
+
+static void alc880_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (alc880_is_input_pin(nid)) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ i <= AUTO_PIN_FRONT_MIC ?
+ PIN_VREF80 : PIN_IN);
+ if (nid != ALC880_PIN_CD_NID)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+ }
+}
+
+/* parse the BIOS configuration and set up the alc_spec */
+/* return 1 if successful, 0 if the proper config is not found,
+ * or a negative error code
+ */
+static int alc880_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc880_ignore);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
+
+ err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_extra_out(spec,
+ spec->autocfg.speaker_pins[0],
+ "Speaker");
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+ "Headphone");
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = ALC880_DIGIN_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs;
+
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+/* additional initialization for auto-configuration model */
+static void alc880_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc880_auto_init_multi_out(codec);
+ alc880_auto_init_extra_out(codec);
+ alc880_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+/*
+ * OK, here we have finally the patch for ALC880
+ */
+
+static int patch_alc880(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ board_config = snd_hda_check_board_config(codec, ALC880_MODEL_LAST,
+ alc880_models,
+ alc880_cfg_tbl);
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC880, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC880_AUTO;
+ }
+
+ if (board_config == ALC880_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc880_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using 3-stack mode...\n");
+ board_config = ALC880_3ST;
+ }
+ }
+
+ if (board_config != ALC880_AUTO)
+ setup_preset(spec, &alc880_presets[board_config]);
+
+ spec->stream_name_analog = "ALC880 Analog";
+ spec->stream_analog_playback = &alc880_pcm_analog_playback;
+ spec->stream_analog_capture = &alc880_pcm_analog_capture;
+ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
+
+ spec->stream_name_digital = "ALC880 Digital";
+ spec->stream_digital_playback = &alc880_pcm_digital_playback;
+ spec->stream_digital_capture = &alc880_pcm_digital_capture;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ /* check whether NID 0x07 is valid */
+ unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]);
+ /* get type */
+ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wcap != AC_WID_AUD_IN) {
+ spec->adc_nids = alc880_adc_nids_alt;
+ spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
+ spec->mixers[spec->num_mixers] =
+ alc880_capture_alt_mixer;
+ spec->num_mixers++;
+ } else {
+ spec->adc_nids = alc880_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
+ spec->mixers[spec->num_mixers] = alc880_capture_mixer;
+ spec->num_mixers++;
+ }
+ }
+
+ spec->vmaster_nid = 0x0c;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC880_AUTO)
+ spec->init_hook = alc880_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc880_loopbacks;
+#endif
+
+ return 0;
+}
+
+
+/*
+ * ALC260 support
+ */
+
+static hda_nid_t alc260_dac_nids[1] = {
+ /* front */
+ 0x02,
+};
+
+static hda_nid_t alc260_adc_nids[1] = {
+ /* ADC0 */
+ 0x04,
+};
+
+static hda_nid_t alc260_adc_nids_alt[1] = {
+ /* ADC1 */
+ 0x05,
+};
+
+static hda_nid_t alc260_hp_adc_nids[2] = {
+ /* ADC1, 0 */
+ 0x05, 0x04
+};
+
+/* NIDs used when simultaneous access to both ADCs makes sense. Note that
+ * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
+ */
+static hda_nid_t alc260_dual_adc_nids[2] = {
+ /* ADC0, ADC1 */
+ 0x04, 0x05
+};
+
+#define ALC260_DIGOUT_NID 0x03
+#define ALC260_DIGIN_NID 0x06
+
+static struct hda_input_mux alc260_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
+ * headphone jack and the internal CD lines since these are the only pins at
+ * which audio can appear. For flexibility, also allow the option of
+ * recording the mixer output on the second ADC (ADC0 doesn't have a
+ * connection to the mixer output).
+ */
+static struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
+ {
+ .num_items = 3,
+ .items = {
+ { "Mic/Line", 0x0 },
+ { "CD", 0x4 },
+ { "Headphone", 0x2 },
+ },
+ },
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic/Line", 0x0 },
+ { "CD", 0x4 },
+ { "Headphone", 0x2 },
+ { "Mixer", 0x5 },
+ },
+ },
+
+};
+
+/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
+ * the Fujitsu S702x, but jacks are marked differently.
+ */
+static struct hda_input_mux alc260_acer_capture_sources[2] = {
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Headphone", 0x5 },
+ },
+ },
+ {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Headphone", 0x6 },
+ { "Mixer", 0x5 },
+ },
+ },
+};
+/*
+ * This is just place-holder, so there's something for alc_build_pcms to look
+ * at when it calculates the maximum number of channels. ALC260 has no mixer
+ * element which allows changing the channel mode, so the verb list is
+ * never used.
+ */
+static struct hda_channel_mode alc260_modes[1] = {
+ { 2, NULL },
+};
+
+
+/* Mixer combinations
+ *
+ * basic: base_output + input + pc_beep + capture
+ * HP: base_output + input + capture_alt
+ * HP_3013: hp_3013 + input + capture
+ * fujitsu: fujitsu + capture
+ * acer: acer + capture
+ */
+
+static struct snd_kcontrol_new alc260_base_output_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc260_input_mixer[] = {
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc260_pc_beep_mixer[] = {
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+/* update HP, line and mono out pins according to the master switch */
+static void alc260_hp_master_update(struct hda_codec *codec,
+ hda_nid_t hp, hda_nid_t line,
+ hda_nid_t mono)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int val = spec->master_sw ? PIN_HP : 0;
+ /* change HP and line-out pins */
+ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val);
+ snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val);
+ /* mono (speaker) depending on the HP jack sense */
+ val = (val && !spec->jack_present) ? PIN_OUT : 0;
+ snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val);
+}
+
+static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ *ucontrol->value.integer.value = spec->master_sw;
+ return 0;
+}
+
+static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int val = !!*ucontrol->value.integer.value;
+ hda_nid_t hp, line, mono;
+
+ if (val == spec->master_sw)
+ return 0;
+ spec->master_sw = val;
+ hp = (kcontrol->private_value >> 16) & 0xff;
+ line = (kcontrol->private_value >> 8) & 0xff;
+ mono = kcontrol->private_value & 0xff;
+ alc260_hp_master_update(codec, hp, line, mono);
+ return 1;
+}
+
+static struct snd_kcontrol_new alc260_hp_output_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ .private_value = (0x0f << 16) | (0x10 << 8) | 0x11
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc260_hp_unsol_verbs[] = {
+ {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x10, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ alc260_hp_master_update(codec, 0x0f, 0x10, 0x11);
+}
+
+static void alc260_hp_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc260_hp_automute(codec);
+}
+
+static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc260_hp_master_sw_get,
+ .put = alc260_hp_master_sw_put,
+ .private_value = (0x10 << 16) | (0x15 << 8) | 0x11
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Aux-In Playback Volume", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("Aux-In Playback Switch", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct hda_bind_ctls alc260_dc7600_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc260_dc7600_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol),
+ HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc260_hp_3013_unsol_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {},
+};
+
+static void alc260_hp_3013_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ alc260_hp_master_update(codec, 0x10, 0x15, 0x11);
+}
+
+static void alc260_hp_3013_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc260_hp_3013_automute(codec);
+}
+
+static void alc260_hp_3012_automute(struct hda_codec *codec)
+{
+ unsigned int present, bits;
+
+ present = snd_hda_codec_read(codec, 0x10, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE;
+
+ bits = present ? 0 : PIN_OUT;
+ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ bits);
+ snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ bits);
+ snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ bits);
+}
+
+static void alc260_hp_3012_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc260_hp_3012_automute(codec);
+}
+
+/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
+ * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
+ */
+static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
+ ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+ { } /* end */
+};
+
+/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
+ * versions of the ALC260 don't act on requests to enable mic bias from NID
+ * 0x0f (used to drive the headphone jack in these laptops). The ALC260
+ * datasheet doesn't mention this restriction. At this stage it's not clear
+ * whether this behaviour is intentional or is a hardware bug in chip
+ * revisions available in early 2006. Therefore for now allow the
+ * "Headphone Jack Mode" control to span all choices, but if it turns out
+ * that the lack of mic bias for this NID is intentional we could change the
+ * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ *
+ * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
+ * don't appear to make the mic bias available from the "line" jack, even
+ * though the NID used for this jack (0x14) can supply it. The theory is
+ * that perhaps Acer have included blocking capacitors between the ALC260
+ * and the output jack. If this turns out to be the case for all such
+ * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
+ * to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ *
+ * The C20x Tablet series have a mono internal speaker which is controlled
+ * via the chip's Mono sum widget and pin complex, so include the necessary
+ * controls for such models. On models without a "mono speaker" the control
+ * won't do anything.
+ */
+static struct snd_kcontrol_new alc260_acer_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
+ ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
+ HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
+ * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
+ */
+static struct snd_kcontrol_new alc260_will_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
+ * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
+ */
+static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
+ { } /* end */
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc260_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x05, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x05, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc260_capture_alt_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+static struct hda_verb alc260_init_verbs[] = {
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /* LINE-2 is used for line-out in rear */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* select line-out */
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* LINE-OUT pin */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* enable HP */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* enable Mono */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* mute capture amp left and right */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* set connection select to line in (default select for this ADC) */
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* set vol=0 Line-Out mixer amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* set vol=0 HP mixer amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* set vol=0 Mono mixer amp left and right */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* unmute LINE-2 out pin */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* mute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* mute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* mute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { }
+};
+
+#if 0 /* should be identical with alc260_init_verbs? */
+static struct hda_verb alc260_hp_init_verbs[] = {
+ /* Headphone and output */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ /* mono output */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* Line-2 pin widget for output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* unmute amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* unmute Line-Out mixer amp left and right (volume = 0) */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* unmute HP mixer amp left and right (volume = 0) */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* Unmute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ { }
+};
+#endif
+
+static struct hda_verb alc260_hp_3013_init_verbs[] = {
+ /* Line out and output */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* mono output */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Mic2 (front panel) pin widget for input and vref at 80% */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ /* Line In pin widget for input */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* Headphone pin widget for output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ /* CD pin widget for input */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ /* unmute amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+ /* set connection select to line in (default select for this ADC) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* unmute Line-Out mixer amp left and right (volume = 0) */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* unmute HP mixer amp left and right (volume = 0) */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* mute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
+ * Line In 2 = 0x03
+ */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+ /* Unmute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* Unmute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ { }
+};
+
+/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
+ * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
+ * audio = 0x16, internal speaker = 0x10.
+ */
+static struct hda_verb alc260_fujitsu_init_verbs[] = {
+ /* Disable all GPIOs */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0},
+ /* Internal speaker is connected to headphone pin */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Headphone/Line-out jack connects to Line1 pin; make it an output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Mic/Line-in jack is connected to mic1 pin, so make it an input */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Ensure all other unused pins are disabled and muted. */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+ /* Disable digital (SPDIF) pins */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure Line1 pin widget takes its input from the OUT1 sum bus
+ * when acting as an output.
+ */
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Line1 pin widget output buffer since it starts as an output.
+ * If the pin mode is changed by the user the pin mode control will
+ * take care of enabling the pin's input/output buffers as needed.
+ * Therefore there's no need to enable the input buffer at this
+ * stage.
+ */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute input buffer of pin widget used for Line-in (no equiv
+ * mixer ctrl)
+ */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to match default mixer setting - line
+ * in (on mic1 pin)
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Do the same for the second ADC: mute capture input amp and
+ * set ADC connection to line in (on mic1 pin)
+ */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+
+/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
+ * similar laptops (adapted from Fujitsu init verbs).
+ */
+static struct hda_verb alc260_acer_init_verbs[] = {
+ /* On TravelMate laptops, GPIO 0 enables the internal speaker and
+ * the headphone jack. Turn this on and rely on the standard mute
+ * methods whenever the user wants to turn these outputs off.
+ */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+ /* Internal speaker/Headphone jack is connected to Line-out pin */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Internal microphone/Mic jack is connected to Mic1 pin */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ /* Line In jack is connected to Line1 pin */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Ensure all other unused pins are disabled and muted. */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Disable digital (SPDIF) pins */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
+ * bus when acting as outputs.
+ */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Unmute Line-out pin widget amp left and right
+ * (no equiv mixer ctrl)
+ */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute mono pin widget amp output (no equiv mixer ctrl) */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Mic1 and Line1 pin widget input buffers since they start as
+ * inputs. If the pin mode is changed by the user the pin mode control
+ * will take care of enabling the pin's input/output buffers as needed.
+ * Therefore there's no need to enable the input buffer at this
+ * stage.
+ */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to match default mixer setting - mic
+ * (on mic1 pin)
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Do similar with the second ADC: mute capture input amp and
+ * set ADC connection to mic to match ALSA's default state.
+ */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+
+static struct hda_verb alc260_will_verbs[] = {
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
+ {}
+};
+
+static struct hda_verb alc260_replacer_672v_verbs[] = {
+ {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
+
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
+
+ {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc260_replacer_672v_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
+ present = snd_hda_codec_read(codec, 0x0f, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ if (present) {
+ snd_hda_codec_write_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 1);
+ snd_hda_codec_write_cache(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_HP);
+ } else {
+ snd_hda_codec_write_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 0);
+ snd_hda_codec_write_cache(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_OUT);
+ }
+}
+
+static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc260_replacer_672v_automute(codec);
+}
+
+static struct hda_verb alc260_hp_dc7600_verbs[] = {
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+/* Test configuration for debugging, modelled after the ALC880 test
+ * configuration.
+ */
+#ifdef CONFIG_SND_DEBUG
+static hda_nid_t alc260_test_dac_nids[1] = {
+ 0x02,
+};
+static hda_nid_t alc260_test_adc_nids[2] = {
+ 0x04, 0x05,
+};
+/* For testing the ALC260, each input MUX needs its own definition since
+ * the signal assignments are different. This assumes that the first ADC
+ * is NID 0x04.
+ */
+static struct hda_input_mux alc260_test_capture_sources[2] = {
+ {
+ .num_items = 7,
+ .items = {
+ { "MIC1 pin", 0x0 },
+ { "MIC2 pin", 0x1 },
+ { "LINE1 pin", 0x2 },
+ { "LINE2 pin", 0x3 },
+ { "CD pin", 0x4 },
+ { "LINE-OUT pin", 0x5 },
+ { "HP-OUT pin", 0x6 },
+ },
+ },
+ {
+ .num_items = 8,
+ .items = {
+ { "MIC1 pin", 0x0 },
+ { "MIC2 pin", 0x1 },
+ { "LINE1 pin", 0x2 },
+ { "LINE2 pin", 0x3 },
+ { "CD pin", 0x4 },
+ { "Mixer", 0x5 },
+ { "LINE-OUT pin", 0x6 },
+ { "HP-OUT pin", 0x7 },
+ },
+ },
+};
+static struct snd_kcontrol_new alc260_test_mixer[] = {
+ /* Output driver widgets */
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
+
+ /* Modes for retasking pin widgets
+ * Note: the ALC260 doesn't seem to act on requests to enable mic
+ * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
+ * mention this restriction. At this stage it's not clear whether
+ * this behaviour is intentional or is a hardware bug in chip
+ * revisions available at least up until early 2006. Therefore for
+ * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
+ * choices, but if it turns out that the lack of mic bias for these
+ * NIDs is intentional we could change their modes from
+ * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ */
+ ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
+
+ /* Loopback mixer controls */
+ HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
+ HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
+ HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
+
+ /* Controls for GPIO pins, assuming they are configured as outputs */
+ ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
+
+ /* Switches to allow the digital IO pins to be enabled. The datasheet
+ * is ambigious as to which NID is which; testing on laptops which
+ * make this output available should provide clarification.
+ */
+ ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
+ ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
+
+ /* A switch allowing EAPD to be enabled. Some laptops seem to use
+ * this output to turn on an external amplifier.
+ */
+ ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
+ ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
+
+ { } /* end */
+};
+static struct hda_verb alc260_test_init_verbs[] = {
+ /* Enable all GPIOs as outputs with an initial value of 0 */
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
+
+ /* Enable retasking pins as output, initially without power amp */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* Disable digital (SPDIF) pins initially, but users can enable
+ * them via a mixer switch. In the case of SPDIF-out, this initverb
+ * payload also sets the generation to 0, output to be in "consumer"
+ * PCM format, copyright asserted, no pre-emphasis and no validity
+ * control.
+ */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
+ * OUT1 sum bus when acting as an output.
+ */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Start with output sum widgets muted and their output gains at min */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Unmute retasking pin widget output buffers since the default
+ * state appears to be output. As the pin mode is changed by the
+ * user the pin mode control will take care of enabling the pin's
+ * input/output buffers as needed.
+ */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Also unmute the mono-out pin widget */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to match default mixer setting (mic1
+ * pin)
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Do the same for the second ADC: mute capture input amp and
+ * set ADC connection to mic1 pin
+ */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+ { }
+};
+#endif
+
+#define alc260_pcm_analog_playback alc880_pcm_analog_alt_playback
+#define alc260_pcm_analog_capture alc880_pcm_analog_capture
+
+#define alc260_pcm_digital_playback alc880_pcm_digital_playback
+#define alc260_pcm_digital_capture alc880_pcm_digital_capture
+
+/*
+ * for BIOS auto-configuration
+ */
+
+static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
+ const char *pfx, int *vol_bits)
+{
+ hda_nid_t nid_vol;
+ unsigned long vol_val, sw_val;
+ char name[32];
+ int err;
+
+ if (nid >= 0x0f && nid < 0x11) {
+ nid_vol = nid - 0x7;
+ vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
+ sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+ } else if (nid == 0x11) {
+ nid_vol = nid - 0x7;
+ vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT);
+ sw_val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
+ } else if (nid >= 0x12 && nid <= 0x15) {
+ nid_vol = 0x08;
+ vol_val = HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT);
+ sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+ } else
+ return 0; /* N/A */
+
+ if (!(*vol_bits & (1 << nid_vol))) {
+ /* first control for the volume widget */
+ snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
+ if (err < 0)
+ return err;
+ *vol_bits |= (1 << nid_vol);
+ }
+ snprintf(name, sizeof(name), "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
+ if (err < 0)
+ return err;
+ return 1;
+}
+
+/* add playback controls from the parsed DAC table */
+static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ hda_nid_t nid;
+ int err;
+ int vols = 0;
+
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ spec->multiout.dac_nids[0] = 0x02;
+
+ nid = cfg->line_out_pins[0];
+ if (nid) {
+ err = alc260_add_playback_controls(spec, nid, "Front", &vols);
+ if (err < 0)
+ return err;
+ }
+
+ nid = cfg->speaker_pins[0];
+ if (nid) {
+ err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
+ if (err < 0)
+ return err;
+ }
+
+ nid = cfg->hp_pins[0];
+ if (nid) {
+ err = alc260_add_playback_controls(spec, nid, "Headphone",
+ &vols);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct hda_input_mux *imux = &spec->private_imux;
+ int i, err, idx;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (cfg->input_pins[i] >= 0x12) {
+ idx = cfg->input_pins[i] - 0x12;
+ err = new_analog_input(spec, cfg->input_pins[i],
+ auto_pin_cfg_labels[i], idx,
+ 0x07);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label =
+ auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+ if (cfg->input_pins[i] >= 0x0f && cfg->input_pins[i] <= 0x10){
+ idx = cfg->input_pins[i] - 0x09;
+ err = new_analog_input(spec, cfg->input_pins[i],
+ auto_pin_cfg_labels[i], idx,
+ 0x07);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label =
+ auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+ }
+ return 0;
+}
+
+static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int sel_idx)
+{
+ alc_set_pin_output(codec, nid, pin_type);
+ /* need the manual connection? */
+ if (nid >= 0x12) {
+ int idx = nid - 0x12;
+ snd_hda_codec_write(codec, idx + 0x0b, 0,
+ AC_VERB_SET_CONNECT_SEL, sel_idx);
+ }
+}
+
+static void alc260_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid;
+
+ alc_subsystem_id(codec, 0x10, 0x15, 0x0f);
+ nid = spec->autocfg.line_out_pins[0];
+ if (nid) {
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ alc260_auto_set_output_and_unmute(codec, nid, pin_type, 0);
+ }
+
+ nid = spec->autocfg.speaker_pins[0];
+ if (nid)
+ alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
+
+ nid = spec->autocfg.hp_pins[0];
+ if (nid)
+ alc260_auto_set_output_and_unmute(codec, nid, PIN_HP, 0);
+}
+
+#define ALC260_PIN_CD_NID 0x16
+static void alc260_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (nid >= 0x12) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ i <= AUTO_PIN_FRONT_MIC ?
+ PIN_VREF80 : PIN_IN);
+ if (nid != ALC260_PIN_CD_NID)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+ }
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc260_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x08 - 0x0a)
+ */
+ /* set vol=0 to output mixers */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ { }
+};
+
+static int alc260_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int wcap;
+ int err;
+ static hda_nid_t alc260_ignore[] = { 0x17, 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc260_ignore);
+ if (err < 0)
+ return err;
+ err = alc260_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->kctl_alloc)
+ return 0; /* can't find valid BIOS pin config */
+ err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs;
+
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+
+ /* check whether NID 0x04 is valid */
+ wcap = get_wcaps(codec, 0x04);
+ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
+ if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
+ spec->adc_nids = alc260_adc_nids_alt;
+ spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids_alt);
+ spec->mixers[spec->num_mixers] = alc260_capture_alt_mixer;
+ } else {
+ spec->adc_nids = alc260_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
+ spec->mixers[spec->num_mixers] = alc260_capture_mixer;
+ }
+ spec->num_mixers++;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+/* additional initialization for auto-configuration model */
+static void alc260_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc260_auto_init_multi_out(codec);
+ alc260_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc260_loopbacks[] = {
+ { 0x07, HDA_INPUT, 0 },
+ { 0x07, HDA_INPUT, 1 },
+ { 0x07, HDA_INPUT, 2 },
+ { 0x07, HDA_INPUT, 3 },
+ { 0x07, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+/*
+ * ALC260 configurations
+ */
+static const char *alc260_models[ALC260_MODEL_LAST] = {
+ [ALC260_BASIC] = "basic",
+ [ALC260_HP] = "hp",
+ [ALC260_HP_3013] = "hp-3013",
+ [ALC260_HP_DC7600] = "hp-dc7600",
+ [ALC260_FUJITSU_S702X] = "fujitsu",
+ [ALC260_ACER] = "acer",
+ [ALC260_WILL] = "will",
+ [ALC260_REPLACER_672V] = "replacer",
+#ifdef CONFIG_SND_DEBUG
+ [ALC260_TEST] = "test",
+#endif
+ [ALC260_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc260_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
+ SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
+ SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
+ SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x103c, 0x3016, "HP", ALC260_HP),
+ SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
+ SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
+ SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
+ SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
+ SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
+ SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
+ SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
+ {}
+};
+
+static struct alc_config_preset alc260_presets[] = {
+ [ALC260_BASIC] = {
+ .mixers = { alc260_base_output_mixer,
+ alc260_input_mixer,
+ alc260_pc_beep_mixer,
+ alc260_capture_mixer },
+ .init_verbs = { alc260_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+ .adc_nids = alc260_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ },
+ [ALC260_HP] = {
+ .mixers = { alc260_hp_output_mixer,
+ alc260_input_mixer,
+ alc260_capture_alt_mixer },
+ .init_verbs = { alc260_init_verbs,
+ alc260_hp_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
+ .adc_nids = alc260_hp_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc260_hp_unsol_event,
+ .init_hook = alc260_hp_automute,
+ },
+ [ALC260_HP_DC7600] = {
+ .mixers = { alc260_hp_dc7600_mixer,
+ alc260_input_mixer,
+ alc260_capture_alt_mixer },
+ .init_verbs = { alc260_init_verbs,
+ alc260_hp_dc7600_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
+ .adc_nids = alc260_hp_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc260_hp_3012_unsol_event,
+ .init_hook = alc260_hp_3012_automute,
+ },
+ [ALC260_HP_3013] = {
+ .mixers = { alc260_hp_3013_mixer,
+ alc260_input_mixer,
+ alc260_capture_alt_mixer },
+ .init_verbs = { alc260_hp_3013_init_verbs,
+ alc260_hp_3013_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
+ .adc_nids = alc260_hp_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc260_hp_3013_unsol_event,
+ .init_hook = alc260_hp_3013_automute,
+ },
+ [ALC260_FUJITSU_S702X] = {
+ .mixers = { alc260_fujitsu_mixer,
+ alc260_capture_mixer },
+ .init_verbs = { alc260_fujitsu_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+ .adc_nids = alc260_dual_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
+ .input_mux = alc260_fujitsu_capture_sources,
+ },
+ [ALC260_ACER] = {
+ .mixers = { alc260_acer_mixer,
+ alc260_capture_mixer },
+ .init_verbs = { alc260_acer_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+ .adc_nids = alc260_dual_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
+ .input_mux = alc260_acer_capture_sources,
+ },
+ [ALC260_WILL] = {
+ .mixers = { alc260_will_mixer,
+ alc260_capture_mixer },
+ .init_verbs = { alc260_init_verbs, alc260_will_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+ .adc_nids = alc260_adc_nids,
+ .dig_out_nid = ALC260_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ },
+ [ALC260_REPLACER_672V] = {
+ .mixers = { alc260_replacer_672v_mixer,
+ alc260_capture_mixer },
+ .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_dac_nids),
+ .dac_nids = alc260_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
+ .adc_nids = alc260_adc_nids,
+ .dig_out_nid = ALC260_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .input_mux = &alc260_capture_source,
+ .unsol_event = alc260_replacer_672v_unsol_event,
+ .init_hook = alc260_replacer_672v_automute,
+ },
+#ifdef CONFIG_SND_DEBUG
+ [ALC260_TEST] = {
+ .mixers = { alc260_test_mixer,
+ alc260_capture_mixer },
+ .init_verbs = { alc260_test_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
+ .dac_nids = alc260_test_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
+ .adc_nids = alc260_test_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc260_modes),
+ .channel_mode = alc260_modes,
+ .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
+ .input_mux = alc260_test_capture_sources,
+ },
+#endif
+};
+
+static int patch_alc260(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ board_config = snd_hda_check_board_config(codec, ALC260_MODEL_LAST,
+ alc260_models,
+ alc260_cfg_tbl);
+ if (board_config < 0) {
+ snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC260_AUTO;
+ }
+
+ if (board_config == ALC260_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc260_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC260_BASIC;
+ }
+ }
+
+ if (board_config != ALC260_AUTO)
+ setup_preset(spec, &alc260_presets[board_config]);
+
+ spec->stream_name_analog = "ALC260 Analog";
+ spec->stream_analog_playback = &alc260_pcm_analog_playback;
+ spec->stream_analog_capture = &alc260_pcm_analog_capture;
+
+ spec->stream_name_digital = "ALC260 Digital";
+ spec->stream_digital_playback = &alc260_pcm_digital_playback;
+ spec->stream_digital_capture = &alc260_pcm_digital_capture;
+
+ spec->vmaster_nid = 0x08;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC260_AUTO)
+ spec->init_hook = alc260_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc260_loopbacks;
+#endif
+
+ return 0;
+}
+
+
+/*
+ * ALC882 support
+ *
+ * ALC882 is almost identical with ALC880 but has cleaner and more flexible
+ * configuration. Each pin widget can choose any input DACs and a mixer.
+ * Each ADC is connected from a mixer of all inputs. This makes possible
+ * 6-channel independent captures.
+ *
+ * In addition, an independent DAC for the multi-playback (not used in this
+ * driver yet).
+ */
+#define ALC882_DIGOUT_NID 0x06
+#define ALC882_DIGIN_NID 0x0a
+
+static struct hda_channel_mode alc882_ch_modes[1] = {
+ { 8, NULL }
+};
+
+static hda_nid_t alc882_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+
+/* identical with ALC880 */
+#define alc882_adc_nids alc880_adc_nids
+#define alc882_adc_nids_alt alc880_adc_nids_alt
+
+static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
+static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+
+static struct hda_input_mux alc882_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+#define alc882_mux_enum_info alc_mux_enum_info
+#define alc882_mux_enum_get alc_mux_enum_get
+
+static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ const struct hda_input_mux *imux = spec->input_mux;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ hda_nid_t nid = spec->capsrc_nids ?
+ spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx];
+ unsigned int *cur_val = &spec->cur_mux[adc_idx];
+ unsigned int i, idx;
+
+ idx = ucontrol->value.enumerated.item[0];
+ if (idx >= imux->num_items)
+ idx = imux->num_items - 1;
+ if (*cur_val == idx)
+ return 0;
+ for (i = 0; i < imux->num_items; i++) {
+ unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+ imux->items[i].index,
+ HDA_AMP_MUTE, v);
+ }
+ *cur_val = idx;
+ return 1;
+}
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc882_3ST_ch2_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc882_3ST_ch6_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc882_3ST_6ch_modes[2] = {
+ { 2, alc882_3ST_ch2_init },
+ { 6, alc882_3ST_ch6_init },
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc882_sixstack_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc882_sixstack_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc882_sixstack_modes[2] = {
+ { 6, alc882_sixstack_ch6_init },
+ { 8, alc882_sixstack_ch8_init },
+};
+
+/*
+ * macbook pro ALC885 can switch LineIn to LineOut without loosing Mic
+ */
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc885_mbp_ch2_init[] = {
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc885_mbp_ch6_init[] = {
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ { } /* end */
+};
+
+static struct hda_channel_mode alc885_mbp_6ch_modes[2] = {
+ { 2, alc885_mbp_ch2_init },
+ { 6, alc885_mbp_ch6_init },
+};
+
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+static struct snd_kcontrol_new alc882_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
+ { } /* end */
+};
+static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc882_targa_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ???
+ * Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c
+ */
+static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mobile Front Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mobile Line Playback Volume", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc882_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc882_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* CLFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Side mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* Side Pin: output 3 (0x0f) */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ { }
+};
+
+static struct hda_verb alc882_eapd_verbs[] = {
+ /* change to EAPD mode */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
+ { }
+};
+
+/* Mac Pro test */
+static struct snd_kcontrol_new alc882_macpro_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc882_macpro_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Speaker: output */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x04},
+ /* Headphone output (output 0 - 0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ { }
+};
+
+/* Macbook Pro rev3 */
+static struct hda_verb alc885_mbp3_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP Pin: output 0 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: use output 1 when in LineOut mode */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ { }
+};
+
+/* iMac 24 mixer. */
+static struct snd_kcontrol_new alc885_imac24_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT),
+ { } /* end */
+};
+
+/* iMac 24 init verbs. */
+static struct hda_verb alc885_imac24_init_verbs[] = {
+ /* Internal speakers: output 0 (0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Internal speakers: output 0 (0x0c) */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Headphone: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ /* Front Mic: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ { }
+};
+
+/* Toggle speaker-output according to the hp-jack state */
+static void alc885_imac24_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+/* Processes unsolicited events. */
+static void alc885_imac24_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Headphone insertion or removal. */
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc885_imac24_automute(codec);
+}
+
+static void alc885_mbp3_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
+
+}
+static void alc885_mbp3_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Headphone insertion or removal. */
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc885_mbp3_automute(codec);
+}
+
+
+static struct hda_verb alc882_targa_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x03},
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc882_targa_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+ present ? 1 : 3);
+}
+
+static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ /* Looks like the unsol event is incompatible with the standard
+ * definition. 4bit tag is placed at 26 bit!
+ */
+ if (((res >> 26) == ALC880_HP_EVENT)) {
+ alc882_targa_automute(codec);
+ }
+}
+
+static struct hda_verb alc882_asus_a7j_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ { } /* end */
+};
+
+static struct hda_verb alc882_asus_a7m_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ { } /* end */
+};
+
+static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
+{
+ unsigned int gpiostate, gpiomask, gpiodir;
+
+ gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DATA, 0);
+
+ if (!muted)
+ gpiostate |= (1 << pin);
+ else
+ gpiostate &= ~(1 << pin);
+
+ gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_MASK, 0);
+ gpiomask |= (1 << pin);
+
+ gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DIRECTION, 0);
+ gpiodir |= (1 << pin);
+
+
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_MASK, gpiomask);
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DIRECTION, gpiodir);
+
+ msleep(1);
+
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DATA, gpiostate);
+}
+
+/* set up GPIO at initialization */
+static void alc885_macpro_init_hook(struct hda_codec *codec)
+{
+ alc882_gpio_mute(codec, 0, 0);
+ alc882_gpio_mute(codec, 1, 0);
+}
+
+/* set up GPIO and update auto-muting at initialization */
+static void alc885_imac24_init_hook(struct hda_codec *codec)
+{
+ alc885_macpro_init_hook(codec);
+ alc885_imac24_automute(codec);
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc882_auto_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0f)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+
+ { }
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc882_capture_alt_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc882_mux_enum_info,
+ .get = alc882_mux_enum_get,
+ .put = alc882_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc882_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 3,
+ .info = alc882_mux_enum_info,
+ .get = alc882_mux_enum_get,
+ .put = alc882_mux_enum_put,
+ },
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc882_loopbacks alc880_loopbacks
+#endif
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc882_pcm_analog_playback alc880_pcm_analog_playback
+#define alc882_pcm_analog_capture alc880_pcm_analog_capture
+#define alc882_pcm_digital_playback alc880_pcm_digital_playback
+#define alc882_pcm_digital_capture alc880_pcm_digital_capture
+
+/*
+ * configuration and preset
+ */
+static const char *alc882_models[ALC882_MODEL_LAST] = {
+ [ALC882_3ST_DIG] = "3stack-dig",
+ [ALC882_6ST_DIG] = "6stack-dig",
+ [ALC882_ARIMA] = "arima",
+ [ALC882_W2JC] = "w2jc",
+ [ALC882_TARGA] = "targa",
+ [ALC882_ASUS_A7J] = "asus-a7j",
+ [ALC882_ASUS_A7M] = "asus-a7m",
+ [ALC885_MACPRO] = "macpro",
+ [ALC885_MBP3] = "mbp3",
+ [ALC885_IMAC24] = "imac24",
+ [ALC882_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc882_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
+ SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
+ SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
+ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
+ SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
+ {}
+};
+
+static struct alc_config_preset alc882_presets[] = {
+ [ALC882_3ST_DIG] = {
+ .mixers = { alc882_base_mixer },
+ .init_verbs = { alc882_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_6ST_DIG] = {
+ .mixers = { alc882_base_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+ .channel_mode = alc882_sixstack_modes,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_ARIMA] = {
+ .mixers = { alc882_base_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_init_verbs, alc882_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+ .channel_mode = alc882_sixstack_modes,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_W2JC] = {
+ .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ },
+ [ALC885_MBP3] = {
+ .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_mbp3_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_mbp_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes),
+ .input_mux = &alc882_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc885_mbp3_unsol_event,
+ .init_hook = alc885_mbp3_automute,
+ },
+ [ALC885_MACPRO] = {
+ .mixers = { alc882_macpro_mixer },
+ .init_verbs = { alc882_macpro_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .input_mux = &alc882_capture_source,
+ .init_hook = alc885_macpro_init_hook,
+ },
+ [ALC885_IMAC24] = {
+ .mixers = { alc885_imac24_mixer },
+ .init_verbs = { alc885_imac24_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+ .channel_mode = alc882_ch_modes,
+ .input_mux = &alc882_capture_source,
+ .unsol_event = alc885_imac24_unsol_event,
+ .init_hook = alc885_imac24_init_hook,
+ },
+ [ALC882_TARGA] = {
+ .mixers = { alc882_targa_mixer, alc882_chmode_mixer,
+ alc882_capture_mixer },
+ .init_verbs = { alc882_init_verbs, alc882_targa_verbs},
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+ .adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+ .channel_mode = alc882_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ .unsol_event = alc882_targa_unsol_event,
+ .init_hook = alc882_targa_automute,
+ },
+ [ALC882_ASUS_A7J] = {
+ .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer,
+ alc882_capture_mixer },
+ .init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs},
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+ .adc_nids = alc882_adc_nids,
+ .capsrc_nids = alc882_capsrc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+ .channel_mode = alc882_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+ [ALC882_ASUS_A7M] = {
+ .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
+ alc880_gpio1_init_verbs,
+ alc882_asus_a7m_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
+};
+
+
+/*
+ * Pin config fixes
+ */
+enum {
+ PINFIX_ABIT_AW9D_MAX
+};
+
+static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
+ { 0x15, 0x01080104 }, /* side */
+ { 0x16, 0x01011012 }, /* rear */
+ { 0x17, 0x01016011 }, /* clfe */
+ { }
+};
+
+static const struct alc_pincfg *alc882_pin_fixes[] = {
+ [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+};
+
+static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+ SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
+ {}
+};
+
+/*
+ * BIOS auto configuration
+ */
+static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ /* set as output */
+ struct alc_spec *spec = codec->spec;
+ int idx;
+
+ alc_set_pin_output(codec, nid, pin_type);
+ if (spec->multiout.dac_nids[dac_idx] == 0x25)
+ idx = 4;
+ else
+ idx = spec->multiout.dac_nids[dac_idx] - 2;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
+
+}
+
+static void alc882_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
+ for (i = 0; i <= HDA_SIDE; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ if (nid)
+ alc882_auto_set_output_and_unmute(codec, nid, pin_type,
+ i);
+ }
+}
+
+static void alc882_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front */
+ /* use dac 0 */
+ alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+}
+
+#define alc882_is_input_pin(nid) alc880_is_input_pin(nid)
+#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID
+
+static void alc882_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ unsigned int vref;
+ if (!nid)
+ continue;
+ vref = PIN_IN;
+ if (1 /*i <= AUTO_PIN_FRONT_MIC*/) {
+ unsigned int pincap;
+ pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+ if ((pincap >> AC_PINCAP_VREF_SHIFT) &
+ AC_PINCAP_VREF_80)
+ vref = PIN_VREF80;
+ }
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, vref);
+ if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+}
+
+static void alc882_auto_init_input_src(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ const struct hda_input_mux *imux = spec->input_mux;
+ int c;
+
+ for (c = 0; c < spec->num_adc_nids; c++) {
+ hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
+ hda_nid_t nid = spec->capsrc_nids[c];
+ int conns, mute, idx, item;
+
+ conns = snd_hda_get_connections(codec, nid, conn_list,
+ ARRAY_SIZE(conn_list));
+ if (conns < 0)
+ continue;
+ for (idx = 0; idx < conns; idx++) {
+ /* if the current connection is the selected one,
+ * unmute it as default - otherwise mute it
+ */
+ mute = AMP_IN_MUTE(idx);
+ for (item = 0; item < imux->num_items; item++) {
+ if (imux->items[item].index == idx) {
+ if (spec->cur_mux[c] == item)
+ mute = AMP_IN_UNMUTE(idx);
+ break;
+ }
+ }
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, mute);
+ }
+ }
+}
+
+/* add mic boosts if needed */
+static int alc_auto_add_mic_boost(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ hda_nid_t nid;
+
+ nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
+ if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Mic Boost",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
+ if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Front Mic Boost",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* almost identical with ALC880 parser... */
+static int alc882_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err = alc880_parse_auto_config(codec);
+
+ if (err < 0)
+ return err;
+ else if (!err)
+ return 0; /* no config found */
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ /* hack - override the init verbs */
+ spec->init_verbs[0] = alc882_auto_init_verbs;
+
+ return 1; /* config found */
+}
+
+/* additional initialization for auto-configuration model */
+static void alc882_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc882_auto_init_multi_out(codec);
+ alc882_auto_init_hp_out(codec);
+ alc882_auto_init_analog_input(codec);
+ alc882_auto_init_input_src(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */
+
+static int patch_alc882(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
+ alc882_models,
+ alc882_cfg_tbl);
+
+ if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
+ /* Pick up systems that don't supply PCI SSID */
+ switch (codec->subsystem_id) {
+ case 0x106b0c00: /* Mac Pro */
+ board_config = ALC885_MACPRO;
+ break;
+ case 0x106b1000: /* iMac 24 */
+ case 0x106b2800: /* AppleTV */
+ board_config = ALC885_IMAC24;
+ break;
+ case 0x106b00a0: /* MacBookPro3,1 - Another revision */
+ case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
+ case 0x106b00a4: /* MacbookPro4,1 */
+ case 0x106b2c00: /* Macbook Pro rev3 */
+ case 0x106b3600: /* Macbook 3.1 */
+ case 0x106b3800: /* MacbookPro4,1 - latter revision */
+ board_config = ALC885_MBP3;
+ break;
+ default:
+ /* ALC889A is handled better as ALC888-compatible */
+ if (codec->revision_id == 0x100101 ||
+ codec->revision_id == 0x100103) {
+ alc_free(codec);
+ return patch_alc883(codec);
+ }
+ printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC882_AUTO;
+ }
+ }
+
+ alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+
+ if (board_config == ALC882_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc882_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC882_3ST_DIG;
+ }
+ }
+
+ if (board_config != ALC882_AUTO)
+ setup_preset(spec, &alc882_presets[board_config]);
+
+ if (codec->vendor_id == 0x10ec0885) {
+ spec->stream_name_analog = "ALC885 Analog";
+ spec->stream_name_digital = "ALC885 Digital";
+ } else {
+ spec->stream_name_analog = "ALC882 Analog";
+ spec->stream_name_digital = "ALC882 Digital";
+ }
+
+ spec->stream_analog_playback = &alc882_pcm_analog_playback;
+ spec->stream_analog_capture = &alc882_pcm_analog_capture;
+ /* FIXME: setup DAC5 */
+ /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
+ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
+
+ spec->stream_digital_playback = &alc882_pcm_digital_playback;
+ spec->stream_digital_capture = &alc882_pcm_digital_capture;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ /* check whether NID 0x07 is valid */
+ unsigned int wcap = get_wcaps(codec, 0x07);
+ /* get type */
+ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wcap != AC_WID_AUD_IN) {
+ spec->adc_nids = alc882_adc_nids_alt;
+ spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt);
+ spec->capsrc_nids = alc882_capsrc_nids_alt;
+ spec->mixers[spec->num_mixers] =
+ alc882_capture_alt_mixer;
+ spec->num_mixers++;
+ } else {
+ spec->adc_nids = alc882_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
+ spec->capsrc_nids = alc882_capsrc_nids;
+ spec->mixers[spec->num_mixers] = alc882_capture_mixer;
+ spec->num_mixers++;
+ }
+ }
+
+ spec->vmaster_nid = 0x0c;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC882_AUTO)
+ spec->init_hook = alc882_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc882_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC883 support
+ *
+ * ALC883 is almost identical with ALC880 but has cleaner and more flexible
+ * configuration. Each pin widget can choose any input DACs and a mixer.
+ * Each ADC is connected from a mixer of all inputs. This makes possible
+ * 6-channel independent captures.
+ *
+ * In addition, an independent DAC for the multi-playback (not used in this
+ * driver yet).
+ */
+#define ALC883_DIGOUT_NID 0x06
+#define ALC883_DIGIN_NID 0x0a
+
+static hda_nid_t alc883_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+
+static hda_nid_t alc883_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x08, 0x09,
+};
+
+static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 };
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+
+static struct hda_input_mux alc883_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc883_3stack_6ch_intel = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x1 },
+ { "Front Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc883_lenovo_101e_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "iMic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Int Mic", 0x1 },
+ },
+};
+
+static struct hda_input_mux alc883_lenovo_sky_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc883_asus_eee1601_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ },
+};
+
+#define alc883_mux_enum_info alc_mux_enum_info
+#define alc883_mux_enum_get alc_mux_enum_get
+/* ALC883 has the ALC882-type input selection */
+#define alc883_mux_enum_put alc882_mux_enum_put
+
+/*
+ * 2ch mode
+ */
+static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
+ { 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_3ST_6ch_modes[3] = {
+ { 2, alc883_3ST_ch2_init },
+ { 4, alc883_3ST_ch4_init },
+ { 6, alc883_3ST_ch6_init },
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_intel_init[] = {
+ { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
+ { 2, alc883_3ST_ch2_intel_init },
+ { 4, alc883_3ST_ch4_intel_init },
+ { 6, alc883_3ST_ch6_intel_init },
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_sixstack_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc883_sixstack_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_sixstack_modes[2] = {
+ { 6, alc883_sixstack_ch6_init },
+ { 8, alc883_sixstack_ch8_init },
+};
+
+static struct hda_verb alc883_medion_eapd_verbs[] = {
+ /* eanable EAPD on medion laptop */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
+ { }
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+
+static struct snd_kcontrol_new alc883_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_mitac_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_tagra_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume",
+ 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_bind_ctls alc883_bind_cap_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc883_bind_cap_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol),
+ HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc883_init_verbs[] = {
+ /* ADC1: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* CLFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Side mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* mute analog input loopbacks */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* Side Pin: output 3 (0x0f) */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_mitac_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+/* auto-toggle front mic */
+/*
+static void alc883_mitac_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
+}
+*/
+
+static void alc883_mitac_automute(struct hda_codec *codec)
+{
+ alc883_mitac_hp_automute(codec);
+ /* alc883_mitac_mic_automute(codec); */
+}
+
+static void alc883_mitac_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc883_mitac_hp_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ /* alc883_mitac_mic_automute(codec); */
+ break;
+ }
+}
+
+static struct hda_verb alc883_mitac_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Subwoofer */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, */
+
+ { } /* end */
+};
+
+static struct hda_verb alc883_clevo_m720_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Int speaker */
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
+ /* HP */
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Subwoofer */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
+ { } /* end */
+};
+
+static struct hda_verb alc883_tagra_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x03},
+
+ { } /* end */
+};
+
+static struct hda_verb alc883_lenovo_101e_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT|AC_USRSP_EN},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT|AC_USRSP_EN},
+ { } /* end */
+};
+
+static struct hda_verb alc883_lenovo_nb0763_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ { } /* end */
+};
+
+static struct hda_verb alc888_lenovo_ms7195_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT | AC_USRSP_EN},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static struct hda_verb alc883_haier_w66_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ { } /* end */
+};
+
+static struct hda_verb alc888_lenovo_sky_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static struct hda_verb alc888_3st_hp_verbs[] = {
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */
+ { }
+};
+
+static struct hda_verb alc888_6st_dell_verbs[] = {
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static struct hda_verb alc888_3st_hp_2ch_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { }
+};
+
+static struct hda_verb alc888_3st_hp_6ch_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { }
+};
+
+static struct hda_channel_mode alc888_3st_hp_modes[2] = {
+ { 2, alc888_3st_hp_2ch_init },
+ { 6, alc888_3st_hp_6ch_init },
+};
+
+/* toggle front-jack and RCA according to the hp-jack state */
+static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+/* toggle RCA according to the front-jack state */
+static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc883_lenovo_ms7195_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc888_lenovo_ms7195_front_automute(codec);
+ if ((res >> 26) == ALC880_FRONT_EVENT)
+ alc888_lenovo_ms7195_rca_automute(codec);
+}
+
+static struct hda_verb alc883_medion_md2_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_medion_md2_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc883_medion_md2_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_medion_md2_automute(codec);
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_tagra_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+ present ? 1 : 3);
+}
+
+static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_tagra_automute(codec);
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_clevo_m720_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc883_clevo_m720_automute(struct hda_codec *codec)
+{
+ alc883_clevo_m720_hp_automute(codec);
+ alc883_clevo_m720_mic_automute(codec);
+}
+
+static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc883_clevo_m720_hp_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc883_clevo_m720_mic_automute(codec);
+ break;
+ }
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_2ch_fujitsu_pi2515_automute(codec);
+}
+
+static void alc883_haier_w66_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? 0x80 : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ 0x80, bits);
+}
+
+static void alc883_haier_w66_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_haier_w66_automute(codec);
+}
+
+static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc883_lenovo_101e_all_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_lenovo_101e_all_automute(codec);
+ if ((res >> 26) == ALC880_FRONT_EVENT)
+ alc883_lenovo_101e_ispeaker_automute(codec);
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_acer_aspire_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc883_acer_aspire_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_acer_aspire_automute(codec);
+}
+
+static struct hda_verb alc883_acer_eapd_verbs[] = {
+ /* HP Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front Pin: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* eanable EAPD on medion laptop */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static void alc888_6st_dell_front_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc888_6st_dell_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ printk("hp_event\n");
+ alc888_6st_dell_front_automute(codec);
+ break;
+ }
+}
+
+static void alc888_lenovo_sky_front_automute(struct hda_codec *codec)
+{
+ unsigned int mute;
+ unsigned int present;
+
+ snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ present = (present & 0x80000000) != 0;
+ if (present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+}
+
+static void alc883_lenovo_sky_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc888_lenovo_sky_front_automute(codec);
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc883_auto_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0f)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ /* Input mixer2 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ { }
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc883_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc882_mux_enum_info,
+ .get = alc882_mux_enum_get,
+ .put = alc882_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc888_asus_m90v_verbs[] = {
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* enable unsolicited event */
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static void alc883_nb_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
+}
+
+static void alc883_M90V_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? 0 : PIN_OUT;
+ snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ bits);
+ snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ bits);
+ snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ bits);
+}
+
+static void alc883_mode2_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc883_M90V_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc883_nb_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc883_mode2_inithook(struct hda_codec *codec)
+{
+ alc883_M90V_speaker_automute(codec);
+ alc883_nb_mic_automute(codec);
+}
+
+static struct hda_verb alc888_asus_eee1601_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0838},
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static void alc883_eee1601_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? 0 : PIN_OUT;
+ snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ bits);
+}
+
+static void alc883_eee1601_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc883_eee1601_speaker_automute(codec);
+ break;
+ }
+}
+
+static void alc883_eee1601_inithook(struct hda_codec *codec)
+{
+ alc883_eee1601_speaker_automute(codec);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc883_loopbacks alc880_loopbacks
+#endif
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc883_pcm_analog_playback alc880_pcm_analog_playback
+#define alc883_pcm_analog_capture alc880_pcm_analog_capture
+#define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
+#define alc883_pcm_digital_playback alc880_pcm_digital_playback
+#define alc883_pcm_digital_capture alc880_pcm_digital_capture
+
+/*
+ * configuration and preset
+ */
+static const char *alc883_models[ALC883_MODEL_LAST] = {
+ [ALC883_3ST_2ch_DIG] = "3stack-dig",
+ [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig",
+ [ALC883_3ST_6ch] = "3stack-6ch",
+ [ALC883_6ST_DIG] = "6stack-dig",
+ [ALC883_TARGA_DIG] = "targa-dig",
+ [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
+ [ALC883_ACER] = "acer",
+ [ALC883_ACER_ASPIRE] = "acer-aspire",
+ [ALC883_MEDION] = "medion",
+ [ALC883_MEDION_MD2] = "medion-md2",
+ [ALC883_LAPTOP_EAPD] = "laptop-eapd",
+ [ALC883_LENOVO_101E_2ch] = "lenovo-101e",
+ [ALC883_LENOVO_NB0763] = "lenovo-nb0763",
+ [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
+ [ALC888_LENOVO_SKY] = "lenovo-sky",
+ [ALC883_HAIER_W66] = "haier-w66",
+ [ALC888_3ST_HP] = "3stack-hp",
+ [ALC888_6ST_DELL] = "6stack-dell",
+ [ALC883_MITAC] = "mitac",
+ [ALC883_CLEVO_M720] = "clevo-m720",
+ [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
+ [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel",
+ [ALC883_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc883_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
+ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
+ SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
+ SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
+ SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
+ SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
+ SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
+ SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL),
+ SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch),
+ SND_PCI_QUIRK(0x1458, 0xa002, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
+ SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
+ SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
+ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
+ SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550",
+ ALC883_FUJITSU_PI2515),
+ SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515),
+ SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
+ SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
+ SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+ SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
+ SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
+ SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
+ SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
+ SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
+ {}
+};
+
+static struct alc_config_preset alc883_presets[] = {
+ [ALC883_3ST_2ch_DIG] = {
+ .mixers = { alc883_3ST_2ch_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_3ST_6ch_DIG] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_3ST_6ch] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_3ST_6ch_INTEL] = {
+ .mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
+ .channel_mode = alc883_3ST_6ch_intel_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_3stack_6ch_intel,
+ },
+ [ALC883_6ST_DIG] = {
+ .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_TARGA_DIG] = {
+ .mixers = { alc883_tagra_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_tagra_unsol_event,
+ .init_hook = alc883_tagra_automute,
+ },
+ [ALC883_TARGA_2ch_DIG] = {
+ .mixers = { alc883_tagra_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_tagra_unsol_event,
+ .init_hook = alc883_tagra_automute,
+ },
+ [ALC883_ACER] = {
+ .mixers = { alc883_base_mixer },
+ /* On TravelMate laptops, GPIO 0 enables the internal speaker
+ * and the headphone jack. Turn this on and rely on the
+ * standard mute methods whenever the user wants to turn
+ * these outputs off.
+ */
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_ACER_ASPIRE] = {
+ .mixers = { alc883_acer_aspire_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_acer_aspire_unsol_event,
+ .init_hook = alc883_acer_aspire_automute,
+ },
+ [ALC883_MEDION] = {
+ .mixers = { alc883_fivestack_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs,
+ alc883_medion_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_MEDION_MD2] = {
+ .mixers = { alc883_medion_md2_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_medion_md2_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_medion_md2_unsol_event,
+ .init_hook = alc883_medion_md2_automute,
+ },
+ [ALC883_LAPTOP_EAPD] = {
+ .mixers = { alc883_base_mixer },
+ .init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC883_CLEVO_M720] = {
+ .mixers = { alc883_clevo_m720_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_clevo_m720_unsol_event,
+ .init_hook = alc883_clevo_m720_automute,
+ },
+ [ALC883_LENOVO_101E_2ch] = {
+ .mixers = { alc883_lenovo_101e_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_lenovo_101e_capture_source,
+ .unsol_event = alc883_lenovo_101e_unsol_event,
+ .init_hook = alc883_lenovo_101e_all_automute,
+ },
+ [ALC883_LENOVO_NB0763] = {
+ .mixers = { alc883_lenovo_nb0763_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_lenovo_nb0763_capture_source,
+ .unsol_event = alc883_medion_md2_unsol_event,
+ .init_hook = alc883_medion_md2_automute,
+ },
+ [ALC888_LENOVO_MS7195_DIG] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_lenovo_ms7195_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_lenovo_ms7195_unsol_event,
+ .init_hook = alc888_lenovo_ms7195_front_automute,
+ },
+ [ALC883_HAIER_W66] = {
+ .mixers = { alc883_tagra_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_haier_w66_unsol_event,
+ .init_hook = alc883_haier_w66_automute,
+ },
+ [ALC888_3ST_HP] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
+ .channel_mode = alc888_3st_hp_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ },
+ [ALC888_6ST_DELL] = {
+ .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc888_6st_dell_unsol_event,
+ .init_hook = alc888_6st_dell_front_automute,
+ },
+ [ALC883_MITAC] = {
+ .mixers = { alc883_mitac_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_mitac_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_mitac_unsol_event,
+ .init_hook = alc883_mitac_automute,
+ },
+ [ALC883_FUJITSU_PI2515] = {
+ .mixers = { alc883_2ch_fujitsu_pi2515_mixer },
+ .init_verbs = { alc883_init_verbs,
+ alc883_2ch_fujitsu_pi2515_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_fujitsu_pi2515_capture_source,
+ .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event,
+ .init_hook = alc883_2ch_fujitsu_pi2515_automute,
+ },
+ [ALC888_LENOVO_SKY] = {
+ .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_lenovo_sky_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .adc_nids = alc883_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+ .channel_mode = alc883_sixstack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_lenovo_sky_capture_source,
+ .unsol_event = alc883_lenovo_sky_unsol_event,
+ .init_hook = alc888_lenovo_sky_front_automute,
+ },
+ [ALC888_ASUS_M90V] = {
+ .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_asus_m90v_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_fujitsu_pi2515_capture_source,
+ .unsol_event = alc883_mode2_unsol_event,
+ .init_hook = alc883_mode2_inithook,
+ },
+ [ALC888_ASUS_EEE1601] = {
+ .mixers = { alc883_asus_eee1601_mixer },
+ .init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_asus_eee1601_capture_source,
+ .unsol_event = alc883_eee1601_unsol_event,
+ .init_hook = alc883_eee1601_inithook,
+ },
+};
+
+
+/*
+ * BIOS auto configuration
+ */
+static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ /* set as output */
+ struct alc_spec *spec = codec->spec;
+ int idx;
+
+ alc_set_pin_output(codec, nid, pin_type);
+ if (spec->multiout.dac_nids[dac_idx] == 0x25)
+ idx = 4;
+ else
+ idx = spec->multiout.dac_nids[dac_idx] - 2;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
+
+}
+
+static void alc883_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
+ for (i = 0; i <= HDA_SIDE; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ if (nid)
+ alc883_auto_set_output_and_unmute(codec, nid, pin_type,
+ i);
+ }
+}
+
+static void alc883_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front */
+ /* use dac 0 */
+ alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+}
+
+#define alc883_is_input_pin(nid) alc880_is_input_pin(nid)
+#define ALC883_PIN_CD_NID ALC880_PIN_CD_NID
+
+static void alc883_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (alc883_is_input_pin(nid)) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ (i <= AUTO_PIN_FRONT_MIC ?
+ PIN_VREF80 : PIN_IN));
+ if (nid != ALC883_PIN_CD_NID)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+ }
+}
+
+#define alc883_auto_init_input_src alc882_auto_init_input_src
+
+/* almost identical with ALC880 parser... */
+static int alc883_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err = alc880_parse_auto_config(codec);
+
+ if (err < 0)
+ return err;
+ else if (!err)
+ return 0; /* no config found */
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ /* hack - override the init verbs */
+ spec->init_verbs[0] = alc883_auto_init_verbs;
+ spec->mixers[spec->num_mixers] = alc883_capture_mixer;
+ spec->num_mixers++;
+
+ return 1; /* config found */
+}
+
+/* additional initialization for auto-configuration model */
+static void alc883_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc883_auto_init_multi_out(codec);
+ alc883_auto_init_hp_out(codec);
+ alc883_auto_init_analog_input(codec);
+ alc883_auto_init_input_src(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+static int patch_alc883(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+
+ board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST,
+ alc883_models,
+ alc883_cfg_tbl);
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC883, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC883_AUTO;
+ }
+
+ if (board_config == ALC883_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc883_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC883_3ST_2ch_DIG;
+ }
+ }
+
+ if (board_config != ALC883_AUTO)
+ setup_preset(spec, &alc883_presets[board_config]);
+
+ switch (codec->vendor_id) {
+ case 0x10ec0888:
+ if (codec->revision_id == 0x100101) {
+ spec->stream_name_analog = "ALC1200 Analog";
+ spec->stream_name_digital = "ALC1200 Digital";
+ } else {
+ spec->stream_name_analog = "ALC888 Analog";
+ spec->stream_name_digital = "ALC888 Digital";
+ }
+ break;
+ case 0x10ec0889:
+ spec->stream_name_analog = "ALC889 Analog";
+ spec->stream_name_digital = "ALC889 Digital";
+ break;
+ default:
+ spec->stream_name_analog = "ALC883 Analog";
+ spec->stream_name_digital = "ALC883 Digital";
+ break;
+ }
+
+ spec->stream_analog_playback = &alc883_pcm_analog_playback;
+ spec->stream_analog_capture = &alc883_pcm_analog_capture;
+ spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture;
+
+ spec->stream_digital_playback = &alc883_pcm_digital_playback;
+ spec->stream_digital_capture = &alc883_pcm_digital_capture;
+
+ spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+ spec->adc_nids = alc883_adc_nids;
+ spec->capsrc_nids = alc883_capsrc_nids;
+
+ spec->vmaster_nid = 0x0c;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC883_AUTO)
+ spec->init_hook = alc883_auto_init;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc883_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC262 support
+ */
+
+#define ALC262_DIGOUT_NID ALC880_DIGOUT_NID
+#define ALC262_DIGIN_NID ALC880_DIGIN_NID
+
+#define alc262_dac_nids alc260_dac_nids
+#define alc262_adc_nids alc882_adc_nids
+#define alc262_adc_nids_alt alc882_adc_nids_alt
+#define alc262_capsrc_nids alc882_capsrc_nids
+#define alc262_capsrc_nids_alt alc882_capsrc_nids_alt
+
+#define alc262_modes alc260_modes
+#define alc262_capture_source alc882_capture_source
+
+static hda_nid_t alc262_dmic_adc_nids[1] = {
+ /* ADC0 */
+ 0x09
+};
+
+static hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 };
+
+static struct snd_kcontrol_new alc262_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+ /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/* update HP, line and mono-out pins according to the master switch */
+static void alc262_hp_master_update(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int val = spec->master_sw;
+
+ /* HP & line-out */
+ snd_hda_codec_write_cache(codec, 0x1b, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val ? PIN_HP : 0);
+ snd_hda_codec_write_cache(codec, 0x15, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val ? PIN_HP : 0);
+ /* mono (speaker) depending on the HP jack sense */
+ val = val && !spec->jack_present;
+ snd_hda_codec_write_cache(codec, 0x16, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ val ? PIN_OUT : 0);
+}
+
+static void alc262_hp_bpc_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int presence;
+ presence = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
+ alc262_hp_master_update(codec);
+}
+
+static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hp_bpc_automute(codec);
+}
+
+static void alc262_hp_wildwest_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int presence;
+ presence = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
+ alc262_hp_master_update(codec);
+}
+
+static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hp_wildwest_automute(codec);
+}
+
+static int alc262_hp_master_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ *ucontrol->value.integer.value = spec->master_sw;
+ return 0;
+}
+
+static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int val = !!*ucontrol->value.integer.value;
+
+ if (val == spec->master_sw)
+ return 0;
+ spec->master_sw = val;
+ alc262_hp_master_update(codec);
+ return 1;
+}
+
+static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc262_hp_master_sw_get,
+ .put = alc262_hp_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
+ HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = alc262_hp_master_sw_get,
+ .put = alc262_hp_master_sw_put,
+ },
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x1a, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
+ HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Rear Mic Boost", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_hp_t5735_automute(struct hda_codec *codec, int force)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (force || !spec->sense_updated) {
+ unsigned int present;
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ spec->sense_updated = 1;
+ }
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ spec->jack_present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc262_hp_t5735_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hp_t5735_automute(codec, 1);
+}
+
+static void alc262_hp_t5735_init_hook(struct hda_codec *codec)
+{
+ alc262_hp_t5735_automute(codec, 1);
+}
+
+static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc262_hp_t5735_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc262_hp_rp5700_verbs[] = {
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x00 << 8))},
+ {}
+};
+
+static struct hda_input_mux alc262_hp_rp5700_capture_source = {
+ .num_items = 1,
+ .items = {
+ { "Line", 0x1 },
+ },
+};
+
+/* bind hp and internal speaker mute (with plug check) */
+static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ /* change hp mute */
+ change = snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change) {
+ /* change speaker according to HP jack state */
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE;
+ else
+ mute = snd_hda_codec_amp_read(codec, 0x15, 0,
+ HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new alc262_sony_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc262_sony_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+#define alc262_capture_mixer alc882_capture_mixer
+#define alc262_capture_alt_mixer alc882_capture_alt_mixer
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc262_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+
+ { }
+};
+
+static struct hda_verb alc262_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static struct hda_verb alc262_hippo_unsol_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static struct hda_verb alc262_hippo1_unsol_verbs[] = {
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static struct hda_verb alc262_sony_unsol_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, // Front Mic
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static struct hda_input_mux alc262_dmic_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Int DMic", 0x9 },
+ { "Mic", 0x0 },
+ },
+};
+
+static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc262_toshiba_s06_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x22, AC_VERB_SET_CONNECT_SEL, 0x09},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static void alc262_dmic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x22, 0,
+ AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09);
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc262_toshiba_s06_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? 0 : PIN_OUT;
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
+}
+
+
+
+/* unsolicited event for HP jack sensing */
+static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc262_toshiba_s06_speaker_automute(codec);
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc262_dmic_automute(codec);
+
+}
+
+static void alc262_toshiba_s06_init_hook(struct hda_codec *codec)
+{
+ alc262_toshiba_s06_speaker_automute(codec);
+ alc262_dmic_automute(codec);
+}
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_hippo_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+ unsigned int present;
+
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
+ if (spec->jack_present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_hippo_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hippo_automute(codec);
+}
+
+static void alc262_hippo1_automute(struct hda_codec *codec)
+{
+ unsigned int mute;
+ unsigned int present;
+
+ snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ present = (present & 0x80000000) != 0;
+ if (present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_hippo1_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hippo1_automute(codec);
+}
+
+/*
+ * nec model
+ * 0x15 = headphone
+ * 0x16 = internal speaker
+ * 0x18 = external mic
+ */
+
+static struct snd_kcontrol_new alc262_nec_mixer[] = {
+ HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc262_nec_verbs[] = {
+ /* Unmute Speaker */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Headphone */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ /* External mic to headphone */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* External mic to speaker */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {}
+};
+
+/*
+ * fujitsu model
+ * 0x14 = headphone/spdif-out, 0x15 = internal speaker,
+ * 0x1b = port replicator headphone out
+ */
+
+#define ALC_HP_EVENT 0x37
+
+static struct hda_verb alc262_fujitsu_unsol_verbs[] = {
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = {
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {}
+};
+
+static struct hda_input_mux alc262_fujitsu_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Int Mic", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc262_HP_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "AUX IN", 0x6 },
+ },
+};
+
+static struct hda_input_mux alc262_HP_D7000_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x2 },
+ { "Line", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
+/* mute/unmute internal speaker according to the hp jacks and mute state */
+static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+
+ if (force || !spec->sense_updated) {
+ unsigned int present;
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
+ /* check laptop HP jack */
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+ /* check docking HP jack */
+ present |= snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ if (present & AC_PINSENSE_PRESENCE)
+ spec->jack_present = 1;
+ else
+ spec->jack_present = 0;
+ spec->sense_updated = 1;
+ }
+ /* unmute internal speaker only if both HPs are unplugged and
+ * master switch is on
+ */
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE;
+ else
+ mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_fujitsu_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC_HP_EVENT)
+ return;
+ alc262_fujitsu_automute(codec, 1);
+}
+
+static void alc262_fujitsu_init_hook(struct hda_codec *codec)
+{
+ alc262_fujitsu_automute(codec, 1);
+}
+
+/* bind volumes of both NID 0x0c and 0x0d */
+static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+
+ if (force || !spec->sense_updated) {
+ unsigned int present_int_hp;
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present_int_hp = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present_int_hp & 0x80000000) != 0;
+ spec->sense_updated = 1;
+ }
+ if (spec->jack_present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC_HP_EVENT)
+ return;
+ alc262_lenovo_3000_automute(codec, 1);
+}
+
+/* bind hp and internal speaker mute (with plug check) */
+static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp ? 0 : HDA_AMP_MUTE);
+
+ if (change)
+ alc262_fujitsu_automute(codec, 0);
+ return change;
+}
+
+static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc262_fujitsu_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+/* bind hp and internal speaker mute (with plug check) */
+static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp ? 0 : HDA_AMP_MUTE);
+
+ if (change)
+ alc262_lenovo_3000_automute(codec, 0);
+ return change;
+}
+
+static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc262_lenovo_3000_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc262_sony_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* additional init verbs for Benq laptops */
+static struct hda_verb alc262_EAPD_verbs[] = {
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
+ {}
+};
+
+static struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+ {}
+};
+
+/* Samsung Q1 Ultra Vista model setup */
+static struct snd_kcontrol_new alc262_ultra_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Mic Boost", 0x15, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc262_ultra_verbs[] = {
+ /* output mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* speaker */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ /* internal mic */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* ADC, choose mic */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(8)},
+ {}
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_ultra_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+
+ mute = 0;
+ /* auto-mute only when HP is used as HP */
+ if (!spec->cur_mux[0]) {
+ unsigned int present;
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE;
+ }
+ /* mute/unmute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ /* mute/unmute HP */
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute ? 0 : HDA_AMP_MUTE);
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc262_ultra_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_ultra_automute(codec);
+}
+
+static struct hda_input_mux alc262_ultra_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Headphone", 0x7 },
+ },
+};
+
+static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ int ret;
+
+ ret = alc882_mux_enum_put(kcontrol, ucontrol);
+ if (!ret)
+ return 0;
+ /* reprogram the HP pin as mic or HP according to the input source */
+ snd_hda_codec_write_cache(codec, 0x15, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->cur_mux[0] ? PIN_VREF80 : PIN_HP);
+ alc262_ultra_automute(codec); /* mute/unmute HP */
+ return ret;
+}
+
+static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = alc882_mux_enum_info,
+ .get = alc882_mux_enum_get,
+ .put = alc262_ultra_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* add playback controls from the parsed DAC table */
+static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ hda_nid_t nid;
+ int err;
+
+ spec->multiout.num_dacs = 1; /* only use one dac */
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ spec->multiout.dac_nids[0] = 2;
+
+ nid = cfg->line_out_pins[0];
+ if (nid) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+
+ nid = cfg->speaker_pins[0];
+ if (nid) {
+ if (nid == 0x16) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Speaker Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Speaker Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Speaker Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ nid = cfg->hp_pins[0];
+ if (nid) {
+ /* spec->multiout.hp_nid = 2; */
+ if (nid == 0x16) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+/* identical with ALC880 */
+#define alc262_auto_create_analog_input_ctls \
+ alc880_auto_create_analog_input_ctls
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc262_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0f)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+
+ { }
+};
+
+static struct hda_verb alc262_HP_BPC_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
+ { }
+};
+
+static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for front
+ * panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Mono */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* rear MIC */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* Line in */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Line out */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD in */
+
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 }, */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, /*rear MIC*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, /*Line in*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, /*F MIC*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, /*Front*/
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))}, /*CD*/
+ /* {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))}, /*HP*/
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x06 << 8))}, */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x07 << 8))},
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+
+ { }
+};
+
+static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) },
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc262_loopbacks alc880_loopbacks
+#endif
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc262_pcm_analog_playback alc880_pcm_analog_playback
+#define alc262_pcm_analog_capture alc880_pcm_analog_capture
+#define alc262_pcm_digital_playback alc880_pcm_digital_playback
+#define alc262_pcm_digital_capture alc880_pcm_digital_capture
+
+/*
+ * BIOS auto configuration
+ */
+static int alc262_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t alc262_ignore[] = { 0x1d, 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc262_ignore);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
+ err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = ALC262_DIGIN_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs;
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+#define alc262_auto_init_multi_out alc882_auto_init_multi_out
+#define alc262_auto_init_hp_out alc882_auto_init_hp_out
+#define alc262_auto_init_analog_input alc882_auto_init_analog_input
+#define alc262_auto_init_input_src alc882_auto_init_input_src
+
+
+/* init callback for auto-configuration model -- overriding the default init */
+static void alc262_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc262_auto_init_multi_out(codec);
+ alc262_auto_init_hp_out(codec);
+ alc262_auto_init_analog_input(codec);
+ alc262_auto_init_input_src(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+/*
+ * configuration and preset
+ */
+static const char *alc262_models[ALC262_MODEL_LAST] = {
+ [ALC262_BASIC] = "basic",
+ [ALC262_HIPPO] = "hippo",
+ [ALC262_HIPPO_1] = "hippo_1",
+ [ALC262_FUJITSU] = "fujitsu",
+ [ALC262_HP_BPC] = "hp-bpc",
+ [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
+ [ALC262_HP_TC_T5735] = "hp-tc-t5735",
+ [ALC262_HP_RP5700] = "hp-rp5700",
+ [ALC262_BENQ_ED8] = "benq",
+ [ALC262_BENQ_T31] = "benq-t31",
+ [ALC262_SONY_ASSAMD] = "sony-assamd",
+ [ALC262_TOSHIBA_S06] = "toshiba-s06",
+ [ALC262_TOSHIBA_RX1] = "toshiba-rx1",
+ [ALC262_ULTRA] = "ultra",
+ [ALC262_LENOVO_3000] = "lenovo-3000",
+ [ALC262_NEC] = "nec",
+ [ALC262_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc262_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
+ SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
+ SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2803, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2805, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x2806, "HP D7000", ALC262_HP_BPC_D7000_WL),
+ SND_PCI_QUIRK(0x103c, 0x2807, "HP D7000", ALC262_HP_BPC_D7000_WF),
+ SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC),
+ SND_PCI_QUIRK(0x103c, 0x302f, "HP Thin Client T5735",
+ ALC262_HP_TC_T5735),
+ SND_PCI_QUIRK(0x103c, 0x2817, "HP RP5700", ALC262_HP_RP5700),
+ SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
+ SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
+ SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
+ ALC262_TOSHIBA_RX1),
+ SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
+ SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
+ SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
+ SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
+ SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA),
+ SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
+ SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
+ SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
+ SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
+ SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
+ {}
+};
+
+static struct alc_config_preset alc262_presets[] = {
+ [ALC262_BASIC] = {
+ .mixers = { alc262_base_mixer },
+ .init_verbs = { alc262_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ },
+ [ALC262_HIPPO] = {
+ .mixers = { alc262_base_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo_automute,
+ },
+ [ALC262_HIPPO_1] = {
+ .mixers = { alc262_hippo1_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hippo1_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x02,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc262_hippo1_unsol_event,
+ .init_hook = alc262_hippo1_automute,
+ },
+ [ALC262_FUJITSU] = {
+ .mixers = { alc262_fujitsu_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+ alc262_fujitsu_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_fujitsu_capture_source,
+ .unsol_event = alc262_fujitsu_unsol_event,
+ .init_hook = alc262_fujitsu_init_hook,
+ },
+ [ALC262_HP_BPC] = {
+ .mixers = { alc262_HP_BPC_mixer },
+ .init_verbs = { alc262_HP_BPC_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_capture_source,
+ .unsol_event = alc262_hp_bpc_unsol_event,
+ .init_hook = alc262_hp_bpc_automute,
+ },
+ [ALC262_HP_BPC_D7000_WF] = {
+ .mixers = { alc262_HP_BPC_WildWest_mixer },
+ .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc262_hp_wildwest_unsol_event,
+ .init_hook = alc262_hp_wildwest_automute,
+ },
+ [ALC262_HP_BPC_D7000_WL] = {
+ .mixers = { alc262_HP_BPC_WildWest_mixer,
+ alc262_HP_BPC_WildWest_option_mixer },
+ .init_verbs = { alc262_HP_BPC_WildWest_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_HP_D7000_capture_source,
+ .unsol_event = alc262_hp_wildwest_unsol_event,
+ .init_hook = alc262_hp_wildwest_automute,
+ },
+ [ALC262_HP_TC_T5735] = {
+ .mixers = { alc262_hp_t5735_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hp_t5735_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc262_hp_t5735_unsol_event,
+ .init_hook = alc262_hp_t5735_init_hook,
+ },
+ [ALC262_HP_RP5700] = {
+ .mixers = { alc262_hp_rp5700_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_hp_rp5700_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_hp_rp5700_capture_source,
+ },
+ [ALC262_BENQ_ED8] = {
+ .mixers = { alc262_base_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ },
+ [ALC262_SONY_ASSAMD] = {
+ .mixers = { alc262_sony_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_sony_unsol_verbs},
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo_automute,
+ },
+ [ALC262_BENQ_T31] = {
+ .mixers = { alc262_benq_t31_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo_automute,
+ },
+ [ALC262_ULTRA] = {
+ .mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer },
+ .init_verbs = { alc262_ultra_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_ultra_capture_source,
+ .adc_nids = alc262_adc_nids, /* ADC0 */
+ .capsrc_nids = alc262_capsrc_nids,
+ .num_adc_nids = 1, /* single ADC */
+ .unsol_event = alc262_ultra_unsol_event,
+ .init_hook = alc262_ultra_automute,
+ },
+ [ALC262_LENOVO_3000] = {
+ .mixers = { alc262_lenovo_3000_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+ alc262_lenovo_3000_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_fujitsu_capture_source,
+ .unsol_event = alc262_lenovo_3000_unsol_event,
+ },
+ [ALC262_NEC] = {
+ .mixers = { alc262_nec_mixer },
+ .init_verbs = { alc262_nec_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ },
+ [ALC262_TOSHIBA_S06] = {
+ .mixers = { alc262_toshiba_s06_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_toshiba_s06_verbs,
+ alc262_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .capsrc_nids = alc262_dmic_capsrc_nids,
+ .dac_nids = alc262_dac_nids,
+ .adc_nids = alc262_dmic_adc_nids, /* ADC0 */
+ .dig_out_nid = ALC262_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_dmic_capture_source,
+ .unsol_event = alc262_toshiba_s06_unsol_event,
+ .init_hook = alc262_toshiba_s06_init_hook,
+ },
+ [ALC262_TOSHIBA_RX1] = {
+ .mixers = { alc262_toshiba_rx1_mixer },
+ .init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+ .dac_nids = alc262_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc262_modes),
+ .channel_mode = alc262_modes,
+ .input_mux = &alc262_capture_source,
+ .unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo_automute,
+ },
+};
+
+static int patch_alc262(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+#if 0
+ /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is
+ * under-run
+ */
+ {
+ int tmp;
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
+ }
+#endif
+
+ alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+
+ board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST,
+ alc262_models,
+ alc262_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC262, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC262_AUTO;
+ }
+
+ if (board_config == ALC262_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc262_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC262_BASIC;
+ }
+ }
+
+ if (board_config != ALC262_AUTO)
+ setup_preset(spec, &alc262_presets[board_config]);
+
+ spec->stream_name_analog = "ALC262 Analog";
+ spec->stream_analog_playback = &alc262_pcm_analog_playback;
+ spec->stream_analog_capture = &alc262_pcm_analog_capture;
+
+ spec->stream_name_digital = "ALC262 Digital";
+ spec->stream_digital_playback = &alc262_pcm_digital_playback;
+ spec->stream_digital_capture = &alc262_pcm_digital_capture;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ /* check whether NID 0x07 is valid */
+ unsigned int wcap = get_wcaps(codec, 0x07);
+
+ /* get type */
+ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wcap != AC_WID_AUD_IN) {
+ spec->adc_nids = alc262_adc_nids_alt;
+ spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids_alt);
+ spec->capsrc_nids = alc262_capsrc_nids_alt;
+ spec->mixers[spec->num_mixers] =
+ alc262_capture_alt_mixer;
+ spec->num_mixers++;
+ } else {
+ spec->adc_nids = alc262_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc262_adc_nids);
+ spec->capsrc_nids = alc262_capsrc_nids;
+ spec->mixers[spec->num_mixers] = alc262_capture_mixer;
+ spec->num_mixers++;
+ }
+ }
+
+ spec->vmaster_nid = 0x0c;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC262_AUTO)
+ spec->init_hook = alc262_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc262_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC268 channel source setting (2 channel)
+ */
+#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID
+#define alc268_modes alc260_modes
+
+static hda_nid_t alc268_dac_nids[2] = {
+ /* front, hp */
+ 0x02, 0x03
+};
+
+static hda_nid_t alc268_adc_nids[2] = {
+ /* ADC0-1 */
+ 0x08, 0x07
+};
+
+static hda_nid_t alc268_adc_nids_alt[1] = {
+ /* ADC0 */
+ 0x08
+};
+
+static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 };
+
+static struct snd_kcontrol_new alc268_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
+/* bind Beep switches of both NID 0x0f and 0x10 */
+static struct hda_bind_ctls alc268_bind_beep_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x10, 3, 1, HDA_INPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc268_beep_mixer[] = {
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT),
+ HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw),
+ { }
+};
+
+static struct hda_verb alc268_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/* Toshiba specific */
+#define alc268_toshiba_automute alc262_hippo_automute
+
+static struct hda_verb alc268_toshiba_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+static struct hda_input_mux alc268_acer_lc_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "i-Mic", 0x6 },
+ { "E-Mic", 0x0 },
+ },
+};
+
+/* Acer specific */
+/* bind volumes of both NID 0x02 and 0x03 */
+static struct hda_bind_ctls alc268_acer_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc268_acer_automute(struct hda_codec *codec, int force)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+
+ if (force || !spec->sense_updated) {
+ unsigned int present;
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
+ spec->sense_updated = 1;
+ }
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE; /* mute internal speaker */
+ else /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+}
+
+
+/* bind hp and internal speaker mute (with plug check) */
+static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change)
+ alc268_acer_automute(codec, 0);
+ return change;
+}
+
+static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc268_acer_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT),
+ { }
+};
+
+static struct snd_kcontrol_new alc268_acer_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc268_acer_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
+static struct hda_verb alc268_acer_aspire_one_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x06},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017},
+ { }
+};
+
+static struct hda_verb alc268_acer_verbs[] = {
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+/* unsolicited event for HP jack sensing */
+static void alc268_toshiba_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc268_toshiba_automute(codec);
+}
+
+static void alc268_acer_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc268_acer_automute(codec, 1);
+}
+
+static void alc268_acer_init_hook(struct hda_codec *codec)
+{
+ alc268_acer_automute(codec, 1);
+}
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? AMP_IN_MUTE(0) : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+}
+
+
+static void alc268_acer_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL,
+ present ? 0x0 : 0x6);
+}
+
+static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc268_aspire_one_speaker_automute(codec);
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc268_acer_mic_automute(codec);
+}
+
+static void alc268_acer_lc_init_hook(struct hda_codec *codec)
+{
+ alc268_aspire_one_speaker_automute(codec);
+ alc268_acer_mic_automute(codec);
+}
+
+static struct snd_kcontrol_new alc268_dell_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static struct hda_verb alc268_dell_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc268_dell_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned int mute;
+
+ present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0);
+ if (present & 0x80000000)
+ mute = HDA_AMP_MUTE;
+ else
+ mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+}
+
+static void alc268_dell_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc268_dell_automute(codec);
+}
+
+#define alc268_dell_init_hook alc268_dell_automute
+
+static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Mic Capture Switch", 0x23, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ { }
+};
+
+static struct hda_verb alc267_quanta_il1_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+ { }
+};
+
+static void alc267_quanta_il1_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ present ? 0 : PIN_OUT);
+}
+
+static void alc267_quanta_il1_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x23, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ present ? 0x00 : 0x01);
+}
+
+static void alc267_quanta_il1_automute(struct hda_codec *codec)
+{
+ alc267_quanta_il1_hp_automute(codec);
+ alc267_quanta_il1_mic_automute(codec);
+}
+
+static void alc267_quanta_il1_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc267_quanta_il1_hp_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc267_quanta_il1_mic_automute(codec);
+ break;
+ }
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc268_base_init_verbs[] = {
+ /* Unmute DAC0-1 and set vol = 0 */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ /* set PCBEEP vol = 0, mute connections */
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ /* Unmute Selector 23h,24h and set the default input to mic-in */
+
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ { }
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc268_volume_init_verbs[] = {
+ /* set output DAC */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ /* set PCBEEP vol = 0, mute connections */
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ { }
+};
+
+#define alc268_mux_enum_info alc_mux_enum_info
+#define alc268_mux_enum_get alc_mux_enum_get
+#define alc268_mux_enum_put alc_mux_enum_put
+
+static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc268_mux_enum_info,
+ .get = alc268_mux_enum_get,
+ .put = alc268_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc268_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc268_mux_enum_info,
+ .get = alc268_mux_enum_get,
+ .put = alc268_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_input_mux alc268_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x3 },
+ },
+};
+
+static struct hda_input_mux alc268_acer_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x6 },
+ { "Line", 0x2 },
+ },
+};
+
+#ifdef CONFIG_SND_DEBUG
+static struct snd_kcontrol_new alc268_test_mixer[] = {
+ /* Volume widgets */
+ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Mono sum Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE("LINE-OUT sum Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_BIND_MUTE("HP-OUT sum Playback Switch", 0x10, 2, HDA_INPUT),
+ HDA_BIND_MUTE("LINE-OUT Playback Switch", 0x14, 2, HDA_OUTPUT),
+ HDA_BIND_MUTE("HP-OUT Playback Switch", 0x15, 2, HDA_OUTPUT),
+ HDA_BIND_MUTE("Mono Playback Switch", 0x16, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("MIC1 Capture Volume", 0x18, 0x0, HDA_INPUT),
+ HDA_BIND_MUTE("MIC1 Capture Switch", 0x18, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("MIC2 Capture Volume", 0x19, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("LINE1 Capture Volume", 0x1a, 0x0, HDA_INPUT),
+ HDA_BIND_MUTE("LINE1 Capture Switch", 0x1a, 2, HDA_OUTPUT),
+ /* The below appears problematic on some hardwares */
+ /*HDA_CODEC_VOLUME("PCBEEP Playback Volume", 0x1d, 0x0, HDA_INPUT),*/
+ HDA_CODEC_VOLUME("PCM-IN1 Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("PCM-IN1 Capture Switch", 0x23, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM-IN2 Capture Volume", 0x24, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("PCM-IN2 Capture Switch", 0x24, 2, HDA_OUTPUT),
+
+ /* Modes for retasking pin widgets */
+ ALC_PIN_MODE("LINE-OUT pin mode", 0x14, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("HP-OUT pin mode", 0x15, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("MIC1 pin mode", 0x18, ALC_PIN_DIR_INOUT),
+ ALC_PIN_MODE("LINE1 pin mode", 0x1a, ALC_PIN_DIR_INOUT),
+
+ /* Controls for GPIO pins, assuming they are configured as outputs */
+ ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
+ ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
+
+ /* Switches to allow the digital SPDIF output pin to be enabled.
+ * The ALC268 does not have an SPDIF input.
+ */
+ ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x06, 0x01),
+
+ /* A switch allowing EAPD to be enabled. Some laptops seem to use
+ * this output to turn on an external amplifier.
+ */
+ ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
+ ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
+
+ { } /* end */
+};
+#endif
+
+/* create input playback/capture controls for the given pin */
+static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
+ const char *ctlname, int idx)
+{
+ char name[32];
+ int err;
+
+ sprintf(name, "%s Playback Volume", ctlname);
+ if (nid == 0x14) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(0x02, 3, idx,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (nid == 0x15) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(0x03, 3, idx,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else
+ return -1;
+ sprintf(name, "%s Playback Switch", ctlname);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ hda_nid_t nid;
+ int err;
+
+ spec->multiout.num_dacs = 2; /* only use one dac */
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ spec->multiout.dac_nids[0] = 2;
+ spec->multiout.dac_nids[1] = 3;
+
+ nid = cfg->line_out_pins[0];
+ if (nid)
+ alc268_new_analog_output(spec, nid, "Front", 0);
+
+ nid = cfg->speaker_pins[0];
+ if (nid == 0x1d) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Speaker Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ nid = cfg->hp_pins[0];
+ if (nid)
+ alc268_new_analog_output(spec, nid, "Headphone", 0);
+
+ nid = cfg->line_out_pins[1] | cfg->line_out_pins[2];
+ if (nid == 0x16) {
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Mono Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct hda_input_mux *imux = &spec->private_imux;
+ int i, idx1;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ switch(cfg->input_pins[i]) {
+ case 0x18:
+ idx1 = 0; /* Mic 1 */
+ break;
+ case 0x19:
+ idx1 = 1; /* Mic 2 */
+ break;
+ case 0x1a:
+ idx1 = 2; /* Line In */
+ break;
+ case 0x1c:
+ idx1 = 3; /* CD */
+ break;
+ case 0x12:
+ case 0x13:
+ idx1 = 6; /* digital mics */
+ break;
+ default:
+ continue;
+ }
+ imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index = idx1;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0];
+ hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
+ hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
+ unsigned int dac_vol1, dac_vol2;
+
+ if (speaker_nid) {
+ snd_hda_codec_write(codec, speaker_nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_codec_write(codec, 0x0f, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(1));
+ snd_hda_codec_write(codec, 0x10, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(1));
+ } else {
+ snd_hda_codec_write(codec, 0x0f, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
+ snd_hda_codec_write(codec, 0x10, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
+ }
+
+ dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */
+ if (line_nid == 0x14)
+ dac_vol2 = AMP_OUT_ZERO;
+ else if (line_nid == 0x15)
+ dac_vol1 = AMP_OUT_ZERO;
+ if (hp_nid == 0x14)
+ dac_vol2 = AMP_OUT_ZERO;
+ else if (hp_nid == 0x15)
+ dac_vol1 = AMP_OUT_ZERO;
+ if (line_nid != 0x16 || hp_nid != 0x16 ||
+ spec->autocfg.line_out_pins[1] != 0x16 ||
+ spec->autocfg.line_out_pins[2] != 0x16)
+ dac_vol1 = dac_vol2 = AMP_OUT_ZERO;
+
+ snd_hda_codec_write(codec, 0x02, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, dac_vol1);
+ snd_hda_codec_write(codec, 0x03, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2);
+}
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc268_pcm_analog_playback alc880_pcm_analog_playback
+#define alc268_pcm_analog_capture alc880_pcm_analog_capture
+#define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
+#define alc268_pcm_digital_playback alc880_pcm_digital_playback
+
+/*
+ * BIOS auto configuration
+ */
+static int alc268_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t alc268_ignore[] = { 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc268_ignore);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
+
+ err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = 2;
+
+ /* digital only support output */
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ if (spec->autocfg.speaker_pins[0] != 0x1d)
+ spec->mixers[spec->num_mixers++] = alc268_beep_mixer;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs;
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+#define alc268_auto_init_multi_out alc882_auto_init_multi_out
+#define alc268_auto_init_hp_out alc882_auto_init_hp_out
+#define alc268_auto_init_analog_input alc882_auto_init_analog_input
+
+/* init callback for auto-configuration model -- overriding the default init */
+static void alc268_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc268_auto_init_multi_out(codec);
+ alc268_auto_init_hp_out(codec);
+ alc268_auto_init_mono_speaker_out(codec);
+ alc268_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+/*
+ * configuration and preset
+ */
+static const char *alc268_models[ALC268_MODEL_LAST] = {
+ [ALC267_QUANTA_IL1] = "quanta-il1",
+ [ALC268_3ST] = "3stack",
+ [ALC268_TOSHIBA] = "toshiba",
+ [ALC268_ACER] = "acer",
+ [ALC268_ACER_ASPIRE_ONE] = "acer-aspire",
+ [ALC268_DELL] = "dell",
+ [ALC268_ZEPTO] = "zepto",
+#ifdef CONFIG_SND_DEBUG
+ [ALC268_TEST] = "test",
+#endif
+ [ALC268_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc268_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
+ ALC268_ACER_ASPIRE_ONE),
+ SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
+ SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
+ SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x1179, 0xff64, "TOSHIBA L305", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
+ SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
+ SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
+ {}
+};
+
+static struct alc_config_preset alc268_presets[] = {
+ [ALC267_QUANTA_IL1] = {
+ .mixers = { alc267_quanta_il1_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc267_quanta_il1_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc267_quanta_il1_unsol_event,
+ .init_hook = alc267_quanta_il1_automute,
+ },
+ [ALC268_3ST] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ },
+ [ALC268_TOSHIBA] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_toshiba_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc268_toshiba_unsol_event,
+ .init_hook = alc268_toshiba_automute,
+ },
+ [ALC268_ACER] = {
+ .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_acer_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_acer_capture_source,
+ .unsol_event = alc268_acer_unsol_event,
+ .init_hook = alc268_acer_init_hook,
+ },
+ [ALC268_ACER_ASPIRE_ONE] = {
+ .mixers = { alc268_acer_aspire_one_mixer,
+ alc268_capture_alt_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_acer_aspire_one_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_acer_lc_capture_source,
+ .unsol_event = alc268_acer_lc_unsol_event,
+ .init_hook = alc268_acer_lc_init_hook,
+ },
+ [ALC268_DELL] = {
+ .mixers = { alc268_dell_mixer, alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_dell_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .unsol_event = alc268_dell_unsol_event,
+ .init_hook = alc268_dell_init_hook,
+ .input_mux = &alc268_capture_source,
+ },
+ [ALC268_ZEPTO] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ alc268_beep_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_toshiba_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc268_toshiba_unsol_event,
+ .init_hook = alc268_toshiba_automute
+ },
+#ifdef CONFIG_SND_DEBUG
+ [ALC268_TEST] = {
+ .mixers = { alc268_test_mixer, alc268_capture_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_volume_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .capsrc_nids = alc268_capsrc_nids,
+ .hp_nid = 0x03,
+ .dig_out_nid = ALC268_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ },
+#endif
+};
+
+static int patch_alc268(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ board_config = snd_hda_check_board_config(codec, ALC268_MODEL_LAST,
+ alc268_models,
+ alc268_cfg_tbl);
+
+ if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC268, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC268_AUTO;
+ }
+
+ if (board_config == ALC268_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc268_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC268_3ST;
+ }
+ }
+
+ if (board_config != ALC268_AUTO)
+ setup_preset(spec, &alc268_presets[board_config]);
+
+ if (codec->vendor_id == 0x10ec0267) {
+ spec->stream_name_analog = "ALC267 Analog";
+ spec->stream_name_digital = "ALC267 Digital";
+ } else {
+ spec->stream_name_analog = "ALC268 Analog";
+ spec->stream_name_digital = "ALC268 Digital";
+ }
+
+ spec->stream_analog_playback = &alc268_pcm_analog_playback;
+ spec->stream_analog_capture = &alc268_pcm_analog_capture;
+ spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture;
+
+ spec->stream_digital_playback = &alc268_pcm_digital_playback;
+
+ if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
+ /* override the amp caps for beep generator */
+ snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
+ (0x0c << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (0 << AC_AMPCAP_MUTE_SHIFT));
+
+ if (!spec->adc_nids && spec->input_mux) {
+ /* check whether NID 0x07 is valid */
+ unsigned int wcap = get_wcaps(codec, 0x07);
+ int i;
+
+ /* get type */
+ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
+ spec->adc_nids = alc268_adc_nids_alt;
+ spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
+ spec->mixers[spec->num_mixers] =
+ alc268_capture_alt_mixer;
+ spec->num_mixers++;
+ } else {
+ spec->adc_nids = alc268_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
+ spec->mixers[spec->num_mixers] =
+ alc268_capture_mixer;
+ spec->num_mixers++;
+ }
+ spec->capsrc_nids = alc268_capsrc_nids;
+ /* set default input source */
+ for (i = 0; i < spec->num_adc_nids; i++)
+ snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
+ 0, AC_VERB_SET_CONNECT_SEL,
+ spec->input_mux->items[0].index);
+ }
+
+ spec->vmaster_nid = 0x02;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC268_AUTO)
+ spec->init_hook = alc268_auto_init;
+
+ return 0;
+}
+
+/*
+ * ALC269 channel source setting (2 channel)
+ */
+#define ALC269_DIGOUT_NID ALC880_DIGOUT_NID
+
+#define alc269_dac_nids alc260_dac_nids
+
+static hda_nid_t alc269_adc_nids[1] = {
+ /* ADC1 */
+ 0x08,
+};
+
+static hda_nid_t alc269_capsrc_nids[1] = {
+ 0x23,
+};
+
+/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24),
+ * not a mux!
+ */
+
+static struct hda_input_mux alc269_eeepc_dmic_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "i-Mic", 0x5 },
+ { "e-Mic", 0x0 },
+ },
+};
+
+static struct hda_input_mux alc269_eeepc_amic_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "i-Mic", 0x1 },
+ { "e-Mic", 0x0 },
+ },
+};
+
+#define alc269_modes alc260_modes
+#define alc269_capture_source alc880_lg_lw_capture_source
+
+static struct snd_kcontrol_new alc269_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc268_acer_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ { }
+};
+
+/* bind volumes of both NID 0x0c and 0x0d */
+static struct hda_bind_ctls alc269_epc_bind_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
+ HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
+ HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc269_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+/* beep control */
+static struct snd_kcontrol_new alc269_beep_mixer[] = {
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_verb alc269_quanta_fl1_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? AMP_IN_MUTE(0) : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 0x0c);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF, 0x680);
+
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 0x0c);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF, 0x480);
+}
+
+static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x23, 0,
+ AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1);
+}
+
+static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc269_quanta_fl1_speaker_automute(codec);
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc269_quanta_fl1_mic_automute(codec);
+}
+
+static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
+{
+ alc269_quanta_fl1_speaker_automute(codec);
+ alc269_quanta_fl1_mic_automute(codec);
+}
+
+static struct hda_verb alc269_eeepc_dmic_init_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x05},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7019 | (0x00 << 8))},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x23, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 },
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x701b | (0x00 << 8))},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc269_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? AMP_IN_MUTE(0) : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+}
+
+static void alc269_eeepc_dmic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x23, 0,
+ AC_VERB_SET_CONNECT_SEL, (present ? 0 : 5));
+}
+
+static void alc269_eeepc_amic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc269_speaker_automute(codec);
+
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc269_eeepc_dmic_automute(codec);
+}
+
+static void alc269_eeepc_dmic_inithook(struct hda_codec *codec)
+{
+ alc269_speaker_automute(codec);
+ alc269_eeepc_dmic_automute(codec);
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc269_speaker_automute(codec);
+
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc269_eeepc_amic_automute(codec);
+}
+
+static void alc269_eeepc_amic_inithook(struct hda_codec *codec)
+{
+ alc269_speaker_automute(codec);
+ alc269_eeepc_amic_automute(codec);
+}
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc269_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the
+ * analog-loopback mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for
+ * front panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0e)
+ */
+ /* set vol=0 to output mixers */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+
+ /* set EAPD */
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/* add playback controls from the parsed DAC table */
+static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ hda_nid_t nid;
+ int err;
+
+ spec->multiout.num_dacs = 1; /* only use one dac */
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ spec->multiout.dac_nids[0] = 2;
+
+ nid = cfg->line_out_pins[0];
+ if (nid) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+
+ nid = cfg->speaker_pins[0];
+ if (nid) {
+ if (!cfg->line_out_pins[0]) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Speaker Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ if (nid == 0x16) {
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Speaker Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Speaker Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ nid = cfg->hp_pins[0];
+ if (nid) {
+ /* spec->multiout.hp_nid = 2; */
+ if (!cfg->line_out_pins[0] && !cfg->speaker_pins[0]) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ if (nid == 0x16) {
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int err;
+
+ err = alc880_auto_create_analog_input_ctls(spec, cfg);
+ if (err < 0)
+ return err;
+ /* digital-mic input pin is excluded in alc880_auto_create..()
+ * because it's under 0x18
+ */
+ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
+ cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
+ struct hda_input_mux *imux = &spec->private_imux;
+ imux->items[imux->num_items].label = "Int Mic";
+ imux->items[imux->num_items].index = 0x05;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc269_loopbacks alc880_loopbacks
+#endif
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc269_pcm_analog_playback alc880_pcm_analog_playback
+#define alc269_pcm_analog_capture alc880_pcm_analog_capture
+#define alc269_pcm_digital_playback alc880_pcm_digital_playback
+#define alc269_pcm_digital_capture alc880_pcm_digital_capture
+
+/*
+ * BIOS auto configuration
+ */
+static int alc269_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i, err;
+ static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc269_ignore);
+ if (err < 0)
+ return err;
+
+ err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc269_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC269_DIGOUT_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ /* create a beep mixer control if the pin 0x1d isn't assigned */
+ for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++)
+ if (spec->autocfg.input_pins[i] == 0x1d)
+ break;
+ if (i >= ARRAY_SIZE(spec->autocfg.input_pins))
+ spec->mixers[spec->num_mixers++] = alc269_beep_mixer;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc269_init_verbs;
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+ /* set default input source */
+ snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0],
+ 0, AC_VERB_SET_CONNECT_SEL,
+ spec->input_mux->items[0].index);
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ spec->mixers[spec->num_mixers] = alc269_capture_mixer;
+ spec->num_mixers++;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+#define alc269_auto_init_multi_out alc882_auto_init_multi_out
+#define alc269_auto_init_hp_out alc882_auto_init_hp_out
+#define alc269_auto_init_analog_input alc882_auto_init_analog_input
+
+
+/* init callback for auto-configuration model -- overriding the default init */
+static void alc269_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc269_auto_init_multi_out(codec);
+ alc269_auto_init_hp_out(codec);
+ alc269_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+/*
+ * configuration and preset
+ */
+static const char *alc269_models[ALC269_MODEL_LAST] = {
+ [ALC269_BASIC] = "basic",
+ [ALC269_QUANTA_FL1] = "quanta",
+ [ALC269_ASUS_EEEPC_P703] = "eeepc-p703",
+ [ALC269_ASUS_EEEPC_P901] = "eeepc-p901"
+};
+
+static struct snd_pci_quirk alc269_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
+ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
+ ALC269_ASUS_EEEPC_P703),
+ SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
+ ALC269_ASUS_EEEPC_P901),
+ SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
+ ALC269_ASUS_EEEPC_P901),
+ {}
+};
+
+static struct alc_config_preset alc269_presets[] = {
+ [ALC269_BASIC] = {
+ .mixers = { alc269_base_mixer, alc269_capture_mixer },
+ .init_verbs = { alc269_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_capture_source,
+ },
+ [ALC269_QUANTA_FL1] = {
+ .mixers = { alc269_quanta_fl1_mixer },
+ .init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_capture_source,
+ .unsol_event = alc269_quanta_fl1_unsol_event,
+ .init_hook = alc269_quanta_fl1_init_hook,
+ },
+ [ALC269_ASUS_EEEPC_P703] = {
+ .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer },
+ .init_verbs = { alc269_init_verbs,
+ alc269_eeepc_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_eeepc_amic_capture_source,
+ .unsol_event = alc269_eeepc_amic_unsol_event,
+ .init_hook = alc269_eeepc_amic_inithook,
+ },
+ [ALC269_ASUS_EEEPC_P901] = {
+ .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer},
+ .init_verbs = { alc269_init_verbs,
+ alc269_eeepc_dmic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc269_dac_nids),
+ .dac_nids = alc269_dac_nids,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc269_modes),
+ .channel_mode = alc269_modes,
+ .input_mux = &alc269_eeepc_dmic_capture_source,
+ .unsol_event = alc269_eeepc_dmic_unsol_event,
+ .init_hook = alc269_eeepc_dmic_inithook,
+ },
+};
+
+static int patch_alc269(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ alc_fix_pll_init(codec, 0x20, 0x04, 15);
+
+ board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST,
+ alc269_models,
+ alc269_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC269, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC269_AUTO;
+ }
+
+ if (board_config == ALC269_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc269_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC269_BASIC;
+ }
+ }
+
+ if (board_config != ALC269_AUTO)
+ setup_preset(spec, &alc269_presets[board_config]);
+
+ spec->stream_name_analog = "ALC269 Analog";
+ spec->stream_analog_playback = &alc269_pcm_analog_playback;
+ spec->stream_analog_capture = &alc269_pcm_analog_capture;
+
+ spec->stream_name_digital = "ALC269 Digital";
+ spec->stream_digital_playback = &alc269_pcm_digital_playback;
+ spec->stream_digital_capture = &alc269_pcm_digital_capture;
+
+ spec->adc_nids = alc269_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
+ spec->capsrc_nids = alc269_capsrc_nids;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC269_AUTO)
+ spec->init_hook = alc269_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc269_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC861 channel source setting (2/6 channel selection for 3-stack)
+ */
+
+/*
+ * set the path ways for 2 channel output
+ * need to set the codec line out and mic 1 pin widgets to inputs
+ */
+static struct hda_verb alc861_threestack_ch2_init[] = {
+ /* set pin widget 1Ah (line in) for input */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* set pin widget 18h (mic1/2) for input, for mic also enable
+ * the vref
+ */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+#endif
+ { } /* end */
+};
+/*
+ * 6ch mode
+ * need to set the codec line out and mic 1 pin widgets to outputs
+ */
+static struct hda_verb alc861_threestack_ch6_init[] = {
+ /* set pin widget 1Ah (line in) for output (Back Surround)*/
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* set pin widget 18h (mic1) for output (CLFE)*/
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+
+ { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
+#endif
+ { } /* end */
+};
+
+static struct hda_channel_mode alc861_threestack_modes[2] = {
+ { 2, alc861_threestack_ch2_init },
+ { 6, alc861_threestack_ch6_init },
+};
+/* Set mic1 as input and unmute the mixer */
+static struct hda_verb alc861_uniwill_m31_ch2_init[] = {
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+ { } /* end */
+};
+/* Set mic1 as output and mute mixer */
+static struct hda_verb alc861_uniwill_m31_ch4_init[] = {
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+ { } /* end */
+};
+
+static struct hda_channel_mode alc861_uniwill_m31_modes[2] = {
+ { 2, alc861_uniwill_m31_ch2_init },
+ { 4, alc861_uniwill_m31_ch4_init },
+};
+
+/* Set mic1 and line-in as input and unmute the mixer */
+static struct hda_verb alc861_asus_ch2_init[] = {
+ /* set pin widget 1Ah (line in) for input */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* set pin widget 18h (mic1/2) for input, for mic also enable
+ * the vref
+ */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+#endif
+ { } /* end */
+};
+/* Set mic1 nad line-in as output and mute mixer */
+static struct hda_verb alc861_asus_ch6_init[] = {
+ /* set pin widget 1Ah (line in) for output (Back Surround)*/
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
+ /* set pin widget 18h (mic1) for output (CLFE)*/
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */
+ { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
+
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+#if 0
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
+#endif
+ { } /* end */
+};
+
+static struct hda_channel_mode alc861_asus_modes[2] = {
+ { 2, alc861_asus_ch2_init },
+ { 6, alc861_asus_ch6_init },
+};
+
+/* patch-ALC861 */
+
+static struct snd_kcontrol_new alc861_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+
+ /*Input mixer control */
+ /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+ /* Capture mixer control */
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc861_3ST_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
+
+ /* Input mixer control */
+ /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+ /* Capture mixer control */
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ .private_value = ARRAY_SIZE(alc861_threestack_modes),
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc861_toshiba_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+
+ /*Capture mixer control */
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ /*HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT), */
+
+ /* Input mixer control */
+ /* HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT), */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_INPUT),
+
+ /* Capture mixer control */
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ .private_value = ARRAY_SIZE(alc861_uniwill_m31_modes),
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc861_asus_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Side Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+
+ /* Input mixer control */
+ HDA_CODEC_VOLUME("Input Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Input Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x15, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x10, 0x01, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x03, HDA_OUTPUT),
+
+ /* Capture mixer control */
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ .private_value = ARRAY_SIZE(alc861_asus_modes),
+ },
+ { }
+};
+
+/* additional mixer */
+static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT),
+ { }
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc861_base_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+
+ { }
+};
+
+static struct hda_verb alc861_threestack_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ { }
+};
+
+static struct hda_verb alc861_uniwill_m31_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel) */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ /* this has to be set to VREF80 */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ { }
+};
+
+static struct hda_verb alc861_asus_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* port-A for surround (rear panel)
+ * according to codec#0 this is the HP jack
+ */
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, /* was 0x00 */
+ /* route front PCM to HP */
+ { 0x0e, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ /* port-B for mic-in (rear panel) with vref */
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-C for line-in (rear panel) */
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* port-D for Front */
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-E for HP out (front panel) */
+ /* this has to be set to VREF80 */
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* route front PCM to HP */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ /* port-F for mic-in (front panel) with vref */
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* port-G for CLFE (rear panel) */
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* CD-in */
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* route front mic to ADC1*/
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, /* Output 0~12 step */
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* hp used DAC 3 (Front) */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ { }
+};
+
+/* additional init verbs for ASUS laptops */
+static struct hda_verb alc861_asus_laptop_init_verbs[] = {
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */
+ { }
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc861_auto_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ /* {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute DAC0~3 & spdif out*/
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Unmute Mixer 14 (mic) 1c (Line in)*/
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Unmute Stereo Mixer 15 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c},
+
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, /* set Mic 1 */
+
+ { }
+};
+
+static struct hda_verb alc861_toshiba_init_verbs[] = {
+ {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+
+ { }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc861_toshiba_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x0f, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
+ HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
+}
+
+static void alc861_toshiba_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc861_toshiba_automute(codec);
+}
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc861_pcm_analog_playback alc880_pcm_analog_playback
+#define alc861_pcm_analog_capture alc880_pcm_analog_capture
+#define alc861_pcm_digital_playback alc880_pcm_digital_playback
+#define alc861_pcm_digital_capture alc880_pcm_digital_capture
+
+
+#define ALC861_DIGOUT_NID 0x07
+
+static struct hda_channel_mode alc861_8ch_modes[1] = {
+ { 8, NULL }
+};
+
+static hda_nid_t alc861_dac_nids[4] = {
+ /* front, surround, clfe, side */
+ 0x03, 0x06, 0x05, 0x04
+};
+
+static hda_nid_t alc660_dac_nids[3] = {
+ /* front, clfe, surround */
+ 0x03, 0x05, 0x06
+};
+
+static hda_nid_t alc861_adc_nids[1] = {
+ /* ADC0-2 */
+ 0x08,
+};
+
+static struct hda_input_mux alc861_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x3 },
+ { "Line", 0x1 },
+ { "CD", 0x4 },
+ { "Mixer", 0x5 },
+ },
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int alc861_auto_fill_dac_nids(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int i;
+ hda_nid_t nid;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ if (i >= ARRAY_SIZE(alc861_dac_nids))
+ continue;
+ spec->multiout.dac_nids[i] = alc861_dac_nids[i];
+ }
+ }
+ spec->multiout.num_dacs = cfg->line_outs;
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = {
+ "Front", "Surround", NULL /*CLFE*/, "Side"
+ };
+ hda_nid_t nid;
+ int i, idx, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = spec->multiout.dac_nids[i];
+ if (!nid)
+ continue;
+ if (nid == 0x05) {
+ /* Center/LFE */
+ err = add_control(spec, ALC_CTL_BIND_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_BIND_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ for (idx = 0; idx < ARRAY_SIZE(alc861_dac_nids) - 1;
+ idx++)
+ if (nid == alc861_dac_nids[idx])
+ break;
+ sprintf(name, "%s Playback Switch", chname[idx]);
+ err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
+{
+ int err;
+ hda_nid_t nid;
+
+ if (!pin)
+ return 0;
+
+ if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) {
+ nid = 0x03;
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ spec->multiout.hp_nid = nid;
+ }
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct hda_input_mux *imux = &spec->private_imux;
+ int i, err, idx, idx1;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ switch (cfg->input_pins[i]) {
+ case 0x0c:
+ idx1 = 1;
+ idx = 2; /* Line In */
+ break;
+ case 0x0f:
+ idx1 = 2;
+ idx = 2; /* Line In */
+ break;
+ case 0x0d:
+ idx1 = 0;
+ idx = 1; /* Mic In */
+ break;
+ case 0x10:
+ idx1 = 3;
+ idx = 1; /* Mic In */
+ break;
+ case 0x11:
+ idx1 = 4;
+ idx = 0; /* CD */
+ break;
+ default:
+ continue;
+ }
+
+ err = new_analog_input(spec, cfg->input_pins[i],
+ auto_pin_cfg_labels[i], idx, 0x15);
+ if (err < 0)
+ return err;
+
+ imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index = idx1;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new alc861_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid,
+ int pin_type, int dac_idx)
+{
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pin_type);
+ snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
+}
+
+static void alc861_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ alc_subsystem_id(codec, 0x0e, 0x0f, 0x0b);
+ for (i = 0; i < spec->autocfg.line_outs; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ if (nid)
+ alc861_auto_set_output_and_unmute(codec, nid, pin_type,
+ spec->multiout.dac_nids[i]);
+ }
+}
+
+static void alc861_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front */
+ alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
+ spec->multiout.dac_nids[0]);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+}
+
+static void alc861_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (nid >= 0x0c && nid <= 0x11) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ i <= AUTO_PIN_FRONT_MIC ?
+ PIN_VREF80 : PIN_IN);
+ }
+ }
+}
+
+/* parse the BIOS configuration and set up the alc_spec */
+/* return 1 if successful, 0 if the proper config is not found,
+ * or a negative error code
+ */
+static int alc861_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t alc861_ignore[] = { 0x1d, 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc861_ignore);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
+
+ err = alc861_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs;
+
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+
+ spec->adc_nids = alc861_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
+ spec->mixers[spec->num_mixers] = alc861_capture_mixer;
+ spec->num_mixers++;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+/* additional initialization for auto-configuration model */
+static void alc861_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc861_auto_init_multi_out(codec);
+ alc861_auto_init_hp_out(codec);
+ alc861_auto_init_analog_input(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc861_loopbacks[] = {
+ { 0x15, HDA_INPUT, 0 },
+ { 0x15, HDA_INPUT, 1 },
+ { 0x15, HDA_INPUT, 2 },
+ { 0x15, HDA_INPUT, 3 },
+ { } /* end */
+};
+#endif
+
+
+/*
+ * configuration and preset
+ */
+static const char *alc861_models[ALC861_MODEL_LAST] = {
+ [ALC861_3ST] = "3stack",
+ [ALC660_3ST] = "3stack-660",
+ [ALC861_3ST_DIG] = "3stack-dig",
+ [ALC861_6ST_DIG] = "6stack-dig",
+ [ALC861_UNIWILL_M31] = "uniwill-m31",
+ [ALC861_TOSHIBA] = "toshiba",
+ [ALC861_ASUS] = "asus",
+ [ALC861_ASUS_LAPTOP] = "asus-laptop",
+ [ALC861_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc861_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST),
+ SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
+ SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
+ SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
+ SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
+ /* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
+ * Any other models that need this preset?
+ */
+ /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
+ SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
+ SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
+ SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
+ SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
+ SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
+ /* FIXME: the below seems conflict */
+ /* SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), */
+ SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
+ SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
+ {}
+};
+
+static struct alc_config_preset alc861_presets[] = {
+ [ALC861_3ST] = {
+ .mixers = { alc861_3ST_mixer },
+ .init_verbs = { alc861_threestack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+ .channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_3ST_DIG] = {
+ .mixers = { alc861_base_mixer },
+ .init_verbs = { alc861_threestack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+ .channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_6ST_DIG] = {
+ .mixers = { alc861_base_mixer },
+ .init_verbs = { alc861_base_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_8ch_modes),
+ .channel_mode = alc861_8ch_modes,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC660_3ST] = {
+ .mixers = { alc861_3ST_mixer },
+ .init_verbs = { alc861_threestack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc660_dac_nids),
+ .dac_nids = alc660_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+ .channel_mode = alc861_threestack_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_UNIWILL_M31] = {
+ .mixers = { alc861_uniwill_m31_mixer },
+ .init_verbs = { alc861_uniwill_m31_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_uniwill_m31_modes),
+ .channel_mode = alc861_uniwill_m31_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_TOSHIBA] = {
+ .mixers = { alc861_toshiba_mixer },
+ .init_verbs = { alc861_base_init_verbs,
+ alc861_toshiba_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ .unsol_event = alc861_toshiba_unsol_event,
+ .init_hook = alc861_toshiba_automute,
+ },
+ [ALC861_ASUS] = {
+ .mixers = { alc861_asus_mixer },
+ .init_verbs = { alc861_asus_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861_asus_modes),
+ .channel_mode = alc861_asus_modes,
+ .need_dac_fix = 1,
+ .hp_nid = 0x06,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+ [ALC861_ASUS_LAPTOP] = {
+ .mixers = { alc861_toshiba_mixer, alc861_asus_laptop_mixer },
+ .init_verbs = { alc861_asus_init_verbs,
+ alc861_asus_laptop_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861_dac_nids),
+ .dac_nids = alc861_dac_nids,
+ .dig_out_nid = ALC861_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .need_dac_fix = 1,
+ .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+ .adc_nids = alc861_adc_nids,
+ .input_mux = &alc861_capture_source,
+ },
+};
+
+
+static int patch_alc861(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int board_config;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ board_config = snd_hda_check_board_config(codec, ALC861_MODEL_LAST,
+ alc861_models,
+ alc861_cfg_tbl);
+
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC861, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC861_AUTO;
+ }
+
+ if (board_config == ALC861_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc861_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC861_3ST_DIG;
+ }
+ }
+
+ if (board_config != ALC861_AUTO)
+ setup_preset(spec, &alc861_presets[board_config]);
+
+ spec->stream_name_analog = "ALC861 Analog";
+ spec->stream_analog_playback = &alc861_pcm_analog_playback;
+ spec->stream_analog_capture = &alc861_pcm_analog_capture;
+
+ spec->stream_name_digital = "ALC861 Digital";
+ spec->stream_digital_playback = &alc861_pcm_digital_playback;
+ spec->stream_digital_capture = &alc861_pcm_digital_capture;
+
+ spec->vmaster_nid = 0x03;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC861_AUTO)
+ spec->init_hook = alc861_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC861-VD support
+ *
+ * Based on ALC882
+ *
+ * In addition, an independent DAC
+ */
+#define ALC861VD_DIGOUT_NID 0x06
+
+static hda_nid_t alc861vd_dac_nids[4] = {
+ /* front, surr, clfe, side surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+
+/* dac_nids for ALC660vd are in a different order - according to
+ * Realtek's driver.
+ * This should probably tesult in a different mixer for 6stack models
+ * of ALC660vd codecs, but for now there is only 3stack mixer
+ * - and it is the same as in 861vd.
+ * adc_nids in ALC660vd are (is) the same as in 861vd
+ */
+static hda_nid_t alc660vd_dac_nids[3] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x04, 0x03
+};
+
+static hda_nid_t alc861vd_adc_nids[1] = {
+ /* ADC0 */
+ 0x09,
+};
+
+static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 };
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+static struct hda_input_mux alc861vd_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc861vd_dallas_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Ext Mic", 0x0 },
+ { "Int Mic", 0x1 },
+ },
+};
+
+static struct hda_input_mux alc861vd_hp_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "ATAPI Mic", 0x1 },
+ },
+};
+
+#define alc861vd_mux_enum_info alc_mux_enum_info
+#define alc861vd_mux_enum_get alc_mux_enum_get
+/* ALC861VD has the ALC882-type input selection (but has only one ADC) */
+#define alc861vd_mux_enum_put alc882_mux_enum_put
+
+/*
+ * 2ch mode
+ */
+static struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = {
+ { 2, NULL }
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc861vd_6stack_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc861vd_6stack_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc861vd_6stack_modes[2] = {
+ { 6, alc861vd_6stack_ch6_init },
+ { 8, alc861vd_6stack_ch8_init },
+};
+
+static struct snd_kcontrol_new alc861vd_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc861vd_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc861vd_mux_enum_info,
+ .get = alc861vd_mux_enum_get,
+ .put = alc861vd_mux_enum_put,
+ },
+ { } /* end */
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+static struct snd_kcontrol_new alc861vd_6st_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0,
+ HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc861vd_3st_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ /*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/
+ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+
+ { } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, HP = 0x15,
+ * Ext Mic=0x18, Int Mic = 0x19, CD = 0x1c, PC Beep = 0x1d
+ */
+static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, Line-out = 0x15,
+ * Front Mic=0x18, ATAPI Mic = 0x19,
+ */
+static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ { } /* end */
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc861vd_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0 and set the default input to mic-in
+ */
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of
+ * the analog-loopback mixer widget
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /*
+ * Set up output mixers (0x02 - 0x05)
+ */
+ /* set vol=0 to output mixers */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ { }
+};
+
+/*
+ * 3-stack pin configuration:
+ * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
+ */
+static struct hda_verb alc861vd_3stack_init_verbs[] = {
+ /*
+ * Set pin mode and muting
+ */
+ /* set front pin widgets 0x14 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * 6-stack pin configuration:
+ */
+static struct hda_verb alc861vd_6stack_init_verbs[] = {
+ /*
+ * Set pin mode and muting
+ */
+ /* set front pin widgets 0x14 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* Side Pin: output 3 (0x0f) */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+static struct hda_verb alc861vd_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static struct hda_verb alc660vd_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+static struct hda_verb alc861vd_lenovo_unsol_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {}
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc861vd_lenovo_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc861vd_lenovo_automute(struct hda_codec *codec)
+{
+ alc861vd_lenovo_hp_automute(codec);
+ alc861vd_lenovo_mic_automute(codec);
+}
+
+static void alc861vd_lenovo_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc861vd_lenovo_hp_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc861vd_lenovo_mic_automute(codec);
+ break;
+ }
+}
+
+static struct hda_verb alc861vd_dallas_verbs[] = {
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+
+ { } /* end */
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc861vd_dallas_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc861vd_dallas_automute(codec);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc861vd_loopbacks alc880_loopbacks
+#endif
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc861vd_pcm_analog_playback alc880_pcm_analog_playback
+#define alc861vd_pcm_analog_capture alc880_pcm_analog_capture
+#define alc861vd_pcm_digital_playback alc880_pcm_digital_playback
+#define alc861vd_pcm_digital_capture alc880_pcm_digital_capture
+
+/*
+ * configuration and preset
+ */
+static const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
+ [ALC660VD_3ST] = "3stack-660",
+ [ALC660VD_3ST_DIG] = "3stack-660-digout",
+ [ALC861VD_3ST] = "3stack",
+ [ALC861VD_3ST_DIG] = "3stack-digout",
+ [ALC861VD_6ST_DIG] = "6stack-digout",
+ [ALC861VD_LENOVO] = "lenovo",
+ [ALC861VD_DALLAS] = "dallas",
+ [ALC861VD_HP] = "hp",
+ [ALC861VD_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
+ SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
+ SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
+ SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
+ SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
+ SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
+ SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
+ /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
+ SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
+ SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
+ SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
+ SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
+ {}
+};
+
+static struct alc_config_preset alc861vd_presets[] = {
+ [ALC660VD_3ST] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+ .dac_nids = alc660vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC660VD_3ST_DIG] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+ .dac_nids = alc660vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_3ST] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_3ST_DIG] = {
+ .mixers = { alc861vd_3st_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_6ST_DIG] = {
+ .mixers = { alc861vd_6st_mixer, alc861vd_chmode_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_6stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_6stack_modes),
+ .channel_mode = alc861vd_6stack_modes,
+ .input_mux = &alc861vd_capture_source,
+ },
+ [ALC861VD_LENOVO] = {
+ .mixers = { alc861vd_lenovo_mixer },
+ .init_verbs = { alc861vd_volume_init_verbs,
+ alc861vd_3stack_init_verbs,
+ alc861vd_eapd_verbs,
+ alc861vd_lenovo_unsol_verbs },
+ .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+ .dac_nids = alc660vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_capture_source,
+ .unsol_event = alc861vd_lenovo_unsol_event,
+ .init_hook = alc861vd_lenovo_automute,
+ },
+ [ALC861VD_DALLAS] = {
+ .mixers = { alc861vd_dallas_mixer },
+ .init_verbs = { alc861vd_dallas_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_dallas_capture_source,
+ .unsol_event = alc861vd_dallas_unsol_event,
+ .init_hook = alc861vd_dallas_automute,
+ },
+ [ALC861VD_HP] = {
+ .mixers = { alc861vd_hp_mixer },
+ .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_hp_capture_source,
+ .unsol_event = alc861vd_dallas_unsol_event,
+ .init_hook = alc861vd_dallas_automute,
+ },
+};
+
+/*
+ * BIOS auto configuration
+ */
+static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type, int dac_idx)
+{
+ alc_set_pin_output(codec, nid, pin_type);
+}
+
+static void alc861vd_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
+ for (i = 0; i <= HDA_SIDE; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ if (nid)
+ alc861vd_auto_set_output_and_unmute(codec, nid,
+ pin_type, i);
+ }
+}
+
+
+static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front and use dac 0 */
+ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+}
+
+#define alc861vd_is_input_pin(nid) alc880_is_input_pin(nid)
+#define ALC861VD_PIN_CD_NID ALC880_PIN_CD_NID
+
+static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (alc861vd_is_input_pin(nid)) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ i <= AUTO_PIN_FRONT_MIC ?
+ PIN_VREF80 : PIN_IN);
+ if (nid != ALC861VD_PIN_CD_NID)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+ }
+}
+
+#define alc861vd_auto_init_input_src alc882_auto_init_input_src
+
+#define alc861vd_idx_to_mixer_vol(nid) ((nid) + 0x02)
+#define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c)
+
+/* add playback controls from the parsed DAC table */
+/* Based on ALC880 version. But ALC861VD has separate,
+ * different NIDs for mute/unmute switch and volume control */
+static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"};
+ hda_nid_t nid_v, nid_s;
+ int i, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (!spec->multiout.dac_nids[i])
+ continue;
+ nid_v = alc861vd_idx_to_mixer_vol(
+ alc880_dac_to_idx(
+ spec->multiout.dac_nids[i]));
+ nid_s = alc861vd_idx_to_mixer_switch(
+ alc880_dac_to_idx(
+ spec->multiout.dac_nids[i]));
+
+ if (i == 2) {
+ /* Center/LFE */
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_v, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_v, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_BIND_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_s, 1, 2,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_BIND_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_s, 2, 2,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_v, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_s, 3, 2,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+/* add playback controls for speaker and HP outputs */
+/* Based on ALC880 version. But ALC861VD has separate,
+ * different NIDs for mute/unmute switch and volume control */
+static int alc861vd_auto_create_extra_out(struct alc_spec *spec,
+ hda_nid_t pin, const char *pfx)
+{
+ hda_nid_t nid_v, nid_s;
+ int err;
+ char name[32];
+
+ if (!pin)
+ return 0;
+
+ if (alc880_is_fixed_pin(pin)) {
+ nid_v = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
+ /* specify the DAC as the extra output */
+ if (!spec->multiout.hp_nid)
+ spec->multiout.hp_nid = nid_v;
+ else
+ spec->multiout.extra_out_nid[0] = nid_v;
+ /* control HP volume/switch on the output mixer amp */
+ nid_v = alc861vd_idx_to_mixer_vol(
+ alc880_fixed_pin_idx(pin));
+ nid_s = alc861vd_idx_to_mixer_switch(
+ alc880_fixed_pin_idx(pin));
+
+ sprintf(name, "%s Playback Volume", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT));
+ if (err < 0)
+ return err;
+ } else if (alc880_is_multi_pin(pin)) {
+ /* set manual connection */
+ /* we have only a switch on HP-out PIN */
+ sprintf(name, "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* parse the BIOS configuration and set up the alc_spec
+ * return 1 if successful, 0 if the proper config is not found,
+ * or a negative error code
+ * Based on ALC880 version - had to change it to override
+ * alc880_auto_create_extra_out and alc880_auto_create_multi_out_ctls */
+static int alc861vd_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t alc861vd_ignore[] = { 0x1d, 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc861vd_ignore);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
+
+ err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc861vd_auto_create_extra_out(spec,
+ spec->autocfg.speaker_pins[0],
+ "Speaker");
+ if (err < 0)
+ return err;
+ err = alc861vd_auto_create_extra_out(spec,
+ spec->autocfg.hp_pins[0],
+ "Headphone");
+ if (err < 0)
+ return err;
+ err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_init_verbs++]
+ = alc861vd_volume_init_verbs;
+
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+/* additional initialization for auto-configuration model */
+static void alc861vd_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc861vd_auto_init_multi_out(codec);
+ alc861vd_auto_init_hp_out(codec);
+ alc861vd_auto_init_analog_input(codec);
+ alc861vd_auto_init_input_src(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+static int patch_alc861vd(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ board_config = snd_hda_check_board_config(codec, ALC861VD_MODEL_LAST,
+ alc861vd_models,
+ alc861vd_cfg_tbl);
+
+ if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC660VD/"
+ "ALC861VD, trying auto-probe from BIOS...\n");
+ board_config = ALC861VD_AUTO;
+ }
+
+ if (board_config == ALC861VD_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc861vd_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC861VD_3ST;
+ }
+ }
+
+ if (board_config != ALC861VD_AUTO)
+ setup_preset(spec, &alc861vd_presets[board_config]);
+
+ if (codec->vendor_id == 0x10ec0660) {
+ spec->stream_name_analog = "ALC660-VD Analog";
+ spec->stream_name_digital = "ALC660-VD Digital";
+ /* always turn on EAPD */
+ spec->init_verbs[spec->num_init_verbs++] = alc660vd_eapd_verbs;
+ } else {
+ spec->stream_name_analog = "ALC861VD Analog";
+ spec->stream_name_digital = "ALC861VD Digital";
+ }
+
+ spec->stream_analog_playback = &alc861vd_pcm_analog_playback;
+ spec->stream_analog_capture = &alc861vd_pcm_analog_capture;
+
+ spec->stream_digital_playback = &alc861vd_pcm_digital_playback;
+ spec->stream_digital_capture = &alc861vd_pcm_digital_capture;
+
+ spec->adc_nids = alc861vd_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
+ spec->capsrc_nids = alc861vd_capsrc_nids;
+
+ spec->mixers[spec->num_mixers] = alc861vd_capture_mixer;
+ spec->num_mixers++;
+
+ spec->vmaster_nid = 0x02;
+
+ codec->patch_ops = alc_patch_ops;
+
+ if (board_config == ALC861VD_AUTO)
+ spec->init_hook = alc861vd_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861vd_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * ALC662 support
+ *
+ * ALC662 is almost identical with ALC880 but has cleaner and more flexible
+ * configuration. Each pin widget can choose any input DACs and a mixer.
+ * Each ADC is connected from a mixer of all inputs. This makes possible
+ * 6-channel independent captures.
+ *
+ * In addition, an independent DAC for the multi-playback (not used in this
+ * driver yet).
+ */
+#define ALC662_DIGOUT_NID 0x06
+#define ALC662_DIGIN_NID 0x0a
+
+static hda_nid_t alc662_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x03, 0x04
+};
+
+static hda_nid_t alc662_adc_nids[1] = {
+ /* ADC1-2 */
+ 0x09,
+};
+
+static hda_nid_t alc662_capsrc_nids[1] = { 0x22 };
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+static struct hda_input_mux alc662_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct hda_input_mux alc662_lenovo_101e_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static struct hda_input_mux alc662_eeepc_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "i-Mic", 0x1 },
+ { "e-Mic", 0x0 },
+ },
+};
+
+static struct hda_input_mux alc663_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ },
+};
+
+static struct hda_input_mux alc663_m51va_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Ext-Mic", 0x0 },
+ { "D-Mic", 0x9 },
+ },
+};
+
+#define alc662_mux_enum_info alc_mux_enum_info
+#define alc662_mux_enum_get alc_mux_enum_get
+#define alc662_mux_enum_put alc882_mux_enum_put
+
+/*
+ * 2ch mode
+ */
+static struct hda_channel_mode alc662_3ST_2ch_modes[1] = {
+ { 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc662_3ST_ch2_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc662_3ST_ch6_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc662_3ST_6ch_modes[2] = {
+ { 2, alc662_3ST_ch2_init },
+ { 6, alc662_3ST_ch6_init },
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc662_sixstack_ch6_init[] = {
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc662_sixstack_ch8_init[] = {
+ { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc662_5stack_modes[2] = {
+ { 2, alc662_sixstack_ch6_init },
+ { 6, alc662_sixstack_ch8_init },
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+
+static struct snd_kcontrol_new alc662_base_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ /*Input mixer control */
+ HDA_CODEC_VOLUME("CD Playback Volume", 0xb, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0xb, 0x4, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0xb, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0xb, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0xb, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0xb, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0xb, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0xb, 0x01, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x0d, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Speaker Playback Switch", 0x03, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("e-Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
+ HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_bind_ctls alc663_asus_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc663_asus_one_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc663_m51va_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_bind_ctls alc663_asus_tree_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
+ { } /* end */
+};
+
+static struct hda_bind_ctls alc663_asus_four_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_1bjd_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct hda_bind_ctls alc663_asus_two_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc663_asus_two_bind_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume",
+ &alc663_asus_two_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc663_g71v_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc663_g50v_mixer[] = {
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_chmode_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc_ch_mode_info,
+ .get = alc_ch_mode_get,
+ .put = alc_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc662_init_verbs[] = {
+ /* ADC: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Rear Pin: output 1 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* CLFE Pin: output 2 (0x0e) */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /* always trun on EAPD */
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+
+ { }
+};
+
+static struct hda_verb alc662_sue_init_verbs[] = {
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc662_eeepc_sue_init_verbs[] = {
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+/* Set Unsolicited Event*/
+static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = {
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc662_auto_init_verbs[] = {
+ /*
+ * Unmute ADC and set the default input to mic-in
+ */
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ * Note: PASD motherboards uses the Line In 2 as the input for front
+ * panel mic (mic 2)
+ */
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /*
+ * Set up output mixers (0x0c - 0x0f)
+ */
+ /* set vol=0 to output mixers */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { }
+};
+
+/* additional verbs for ALC663 */
+static struct hda_verb alc663_auto_init_verbs[] = {
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ { }
+};
+
+static struct hda_verb alc663_m51va_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_21jd_amic_init_verbs[] = {
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc662_1bjd_amic_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_15jd_amic_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = {
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_g71v_init_verbs[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
+ /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */
+
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
+
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc663_g50v_init_verbs[] = {
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */
+
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+static struct hda_verb alc662_ecs_init_verbs[] = {
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc662_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = alc662_mux_enum_info,
+ .get = alc662_mux_enum_get,
+ .put = alc662_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc662_auto_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc662_lenovo_101e_all_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc662_lenovo_101e_all_automute(codec);
+ if ((res >> 26) == ALC880_FRONT_EVENT)
+ alc662_lenovo_101e_ispeaker_automute(codec);
+}
+
+static void alc662_eeepc_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc662_eeepc_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc262_hippo1_automute( codec );
+
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc662_eeepc_mic_automute(codec);
+}
+
+static void alc662_eeepc_inithook(struct hda_codec *codec)
+{
+ alc262_hippo1_automute( codec );
+ alc662_eeepc_mic_automute(codec);
+}
+
+static void alc662_eeepc_ep20_automute(struct hda_codec *codec)
+{
+ unsigned int mute;
+ unsigned int present;
+
+ snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ present = (present & 0x80000000) != 0;
+ if (present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc662_eeepc_ep20_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc662_eeepc_ep20_automute(codec);
+}
+
+static void alc662_eeepc_ep20_inithook(struct hda_codec *codec)
+{
+ alc662_eeepc_ep20_automute(codec);
+}
+
+static void alc663_m51va_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+}
+
+static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+}
+
+static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), bits);
+ snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), bits);
+}
+
+static void alc662_f5z_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? 0 : PIN_OUT;
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
+}
+
+static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present1, present2;
+
+ present1 = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ present2 = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+
+ if (present1 || present2) {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ } else {
+ snd_hda_codec_write_cache(codec, 0x14, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ }
+}
+
+static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int present1, present2;
+
+ present1 = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ present2 = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+
+ if (present1 || present2) {
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+ } else {
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
+ AMP_IN_MUTE(0), 0);
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
+ AMP_IN_MUTE(0), 0);
+ }
+}
+
+static void alc663_m51va_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
+ snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x09 << 8) | (present ? 0x80 : 0));
+}
+
+static void alc663_m51va_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_m51va_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc663_m51va_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_m51va_inithook(struct hda_codec *codec)
+{
+ alc663_m51va_speaker_automute(codec);
+ alc663_m51va_mic_automute(codec);
+}
+
+/* ***************** Mode1 ******************************/
+static void alc663_mode1_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_m51va_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_mode1_inithook(struct hda_codec *codec)
+{
+ alc663_m51va_speaker_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode2 ******************************/
+static void alc662_mode2_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc662_f5z_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc662_mode2_inithook(struct hda_codec *codec)
+{
+ alc662_f5z_speaker_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode3 ******************************/
+static void alc663_mode3_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_two_hp_m1_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_mode3_inithook(struct hda_codec *codec)
+{
+ alc663_two_hp_m1_speaker_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode4 ******************************/
+static void alc663_mode4_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_21jd_two_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_mode4_inithook(struct hda_codec *codec)
+{
+ alc663_21jd_two_speaker_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode5 ******************************/
+static void alc663_mode5_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_15jd_two_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_mode5_inithook(struct hda_codec *codec)
+{
+ alc663_15jd_two_speaker_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+/* ***************** Mode6 ******************************/
+static void alc663_mode6_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_two_hp_m2_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_mode6_inithook(struct hda_codec *codec)
+{
+ alc663_two_hp_m2_speaker_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+
+static void alc663_g71v_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x21, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc663_g71v_front_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+}
+
+static void alc663_g71v_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_g71v_hp_automute(codec);
+ break;
+ case ALC880_FRONT_EVENT:
+ alc663_g71v_front_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_g71v_inithook(struct hda_codec *codec)
+{
+ alc663_g71v_front_automute(codec);
+ alc663_g71v_hp_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+
+static void alc663_g50v_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case ALC880_HP_EVENT:
+ alc663_m51va_speaker_automute(codec);
+ break;
+ case ALC880_MIC_EVENT:
+ alc662_eeepc_mic_automute(codec);
+ break;
+ }
+}
+
+static void alc663_g50v_inithook(struct hda_codec *codec)
+{
+ alc663_m51va_speaker_automute(codec);
+ alc662_eeepc_mic_automute(codec);
+}
+
+/* bind hp and internal speaker mute (with plug check) */
+static int alc662_ecs_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ change = snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change)
+ alc262_hippo1_automute(codec);
+ return change;
+}
+
+static struct snd_kcontrol_new alc662_ecs_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc662_ecs_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ },
+
+ HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("e-Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc662_loopbacks alc880_loopbacks
+#endif
+
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc662_pcm_analog_playback alc880_pcm_analog_playback
+#define alc662_pcm_analog_capture alc880_pcm_analog_capture
+#define alc662_pcm_digital_playback alc880_pcm_digital_playback
+#define alc662_pcm_digital_capture alc880_pcm_digital_capture
+
+/*
+ * configuration and preset
+ */
+static const char *alc662_models[ALC662_MODEL_LAST] = {
+ [ALC662_3ST_2ch_DIG] = "3stack-dig",
+ [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig",
+ [ALC662_3ST_6ch] = "3stack-6ch",
+ [ALC662_5ST_DIG] = "6stack-dig",
+ [ALC662_LENOVO_101E] = "lenovo-101e",
+ [ALC662_ASUS_EEEPC_P701] = "eeepc-p701",
+ [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20",
+ [ALC662_ECS] = "ecs",
+ [ALC663_ASUS_M51VA] = "m51va",
+ [ALC663_ASUS_G71V] = "g71v",
+ [ALC663_ASUS_H13] = "h13",
+ [ALC663_ASUS_G50V] = "g50v",
+ [ALC663_ASUS_MODE1] = "asus-mode1",
+ [ALC662_ASUS_MODE2] = "asus-mode2",
+ [ALC663_ASUS_MODE3] = "asus-mode3",
+ [ALC663_ASUS_MODE4] = "asus-mode4",
+ [ALC663_ASUS_MODE5] = "asus-mode5",
+ [ALC663_ASUS_MODE6] = "asus-mode6",
+ [ALC662_AUTO] = "auto",
+};
+
+static struct snd_pci_quirk alc662_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
+ SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
+ SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
+ SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
+ SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
+ SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+ SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+ SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
+ SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
+ SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+ SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
+ SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
+ SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
+ ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13),
+ SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13),
+ {}
+};
+
+static struct alc_config_preset alc662_presets[] = {
+ [ALC662_3ST_2ch_DIG] = {
+ .mixers = { alc662_3ST_2ch_mixer, alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_3ST_6ch_DIG] = {
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer,
+ alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_3ST_6ch] = {
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer,
+ alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_5ST_DIG] = {
+ .mixers = { alc662_base_mixer, alc662_chmode_mixer,
+ alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .dig_in_nid = ALC662_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_5stack_modes),
+ .channel_mode = alc662_5stack_modes,
+ .input_mux = &alc662_capture_source,
+ },
+ [ALC662_LENOVO_101E] = {
+ .mixers = { alc662_lenovo_101e_mixer, alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs, alc662_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_lenovo_101e_capture_source,
+ .unsol_event = alc662_lenovo_101e_unsol_event,
+ .init_hook = alc662_lenovo_101e_all_automute,
+ },
+ [ALC662_ASUS_EEEPC_P701] = {
+ .mixers = { alc662_eeepc_p701_mixer, alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eeepc_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc662_eeepc_unsol_event,
+ .init_hook = alc662_eeepc_inithook,
+ },
+ [ALC662_ASUS_EEEPC_EP20] = {
+ .mixers = { alc662_eeepc_ep20_mixer, alc662_capture_mixer,
+ alc662_chmode_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eeepc_ep20_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .input_mux = &alc662_lenovo_101e_capture_source,
+ .unsol_event = alc662_eeepc_ep20_unsol_event,
+ .init_hook = alc662_eeepc_ep20_inithook,
+ },
+ [ALC662_ECS] = {
+ .mixers = { alc662_ecs_mixer, alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_ecs_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc662_eeepc_unsol_event,
+ .init_hook = alc662_eeepc_inithook,
+ },
+ [ALC663_ASUS_M51VA] = {
+ .mixers = { alc663_m51va_mixer, alc662_capture_mixer},
+ .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc663_m51va_capture_source,
+ .unsol_event = alc663_m51va_unsol_event,
+ .init_hook = alc663_m51va_inithook,
+ },
+ [ALC663_ASUS_G71V] = {
+ .mixers = { alc663_g71v_mixer, alc662_capture_mixer},
+ .init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc663_g71v_unsol_event,
+ .init_hook = alc663_g71v_inithook,
+ },
+ [ALC663_ASUS_H13] = {
+ .mixers = { alc663_m51va_mixer, alc662_capture_mixer},
+ .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc663_m51va_capture_source,
+ .unsol_event = alc663_m51va_unsol_event,
+ .init_hook = alc663_m51va_inithook,
+ },
+ [ALC663_ASUS_G50V] = {
+ .mixers = { alc663_g50v_mixer, alc662_capture_mixer},
+ .init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
+ .channel_mode = alc662_3ST_6ch_modes,
+ .input_mux = &alc663_capture_source,
+ .unsol_event = alc663_g50v_unsol_event,
+ .init_hook = alc663_g50v_inithook,
+ },
+ [ALC663_ASUS_MODE1] = {
+ .mixers = { alc663_m51va_mixer, alc662_auto_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc663_21jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc663_mode1_unsol_event,
+ .init_hook = alc663_mode1_inithook,
+ },
+ [ALC662_ASUS_MODE2] = {
+ .mixers = { alc662_1bjd_mixer, alc662_auto_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_1bjd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc662_mode2_unsol_event,
+ .init_hook = alc662_mode2_inithook,
+ },
+ [ALC663_ASUS_MODE3] = {
+ .mixers = { alc663_two_hp_m1_mixer, alc662_auto_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc663_two_hp_amic_m1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc663_mode3_unsol_event,
+ .init_hook = alc663_mode3_inithook,
+ },
+ [ALC663_ASUS_MODE4] = {
+ .mixers = { alc663_asus_21jd_clfe_mixer,
+ alc662_auto_capture_mixer},
+ .init_verbs = { alc662_init_verbs,
+ alc663_21jd_amic_init_verbs},
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc663_mode4_unsol_event,
+ .init_hook = alc663_mode4_inithook,
+ },
+ [ALC663_ASUS_MODE5] = {
+ .mixers = { alc663_asus_15jd_clfe_mixer,
+ alc662_auto_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc663_15jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc663_mode5_unsol_event,
+ .init_hook = alc663_mode5_inithook,
+ },
+ [ALC663_ASUS_MODE6] = {
+ .mixers = { alc663_two_hp_m2_mixer, alc662_auto_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc663_two_hp_amic_m2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .hp_nid = 0x03,
+ .dac_nids = alc662_dac_nids,
+ .dig_out_nid = ALC662_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc663_mode6_unsol_event,
+ .init_hook = alc663_mode6_inithook,
+ },
+};
+
+
+/*
+ * BIOS auto configuration
+ */
+
+/* add playback controls from the parsed DAC table */
+static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = {
+ "Front", "Surround", NULL /*CLFE*/, "Side"
+ };
+ hda_nid_t nid;
+ int i, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (!spec->multiout.dac_nids[i])
+ continue;
+ nid = alc880_idx_to_dac(i);
+ if (i == 2) {
+ /* Center/LFE */
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
+ 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+/* add playback controls for speaker and HP outputs */
+static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
+ const char *pfx)
+{
+ hda_nid_t nid;
+ int err;
+ char name[32];
+
+ if (!pin)
+ return 0;
+
+ if (pin == 0x17) {
+ /* ALC663 has a mono output pin on 0x17 */
+ sprintf(name, "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
+ return err;
+ }
+
+ if (alc880_is_fixed_pin(pin)) {
+ nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
+ /* printk("DAC nid=%x\n",nid); */
+ /* specify the DAC as the extra output */
+ if (!spec->multiout.hp_nid)
+ spec->multiout.hp_nid = nid;
+ else
+ spec->multiout.extra_out_nid[0] = nid;
+ /* control HP volume/switch on the output mixer amp */
+ nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
+ sprintf(name, "%s Playback Volume", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
+ if (err < 0)
+ return err;
+ } else if (alc880_is_multi_pin(pin)) {
+ /* set manual connection */
+ /* we have only a switch on HP-out PIN */
+ sprintf(name, "%s Playback Switch", pfx);
+ err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct hda_input_mux *imux = &spec->private_imux;
+ int i, err, idx;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (alc880_is_input_pin(cfg->input_pins[i])) {
+ idx = alc880_input_pin_idx(cfg->input_pins[i]);
+ err = new_analog_input(spec, cfg->input_pins[i],
+ auto_pin_cfg_labels[i],
+ idx, 0x0b);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label =
+ auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index =
+ alc880_input_pin_idx(cfg->input_pins[i]);
+ imux->num_items++;
+ }
+ }
+ return 0;
+}
+
+static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ alc_set_pin_output(codec, nid, pin_type);
+ /* need the manual connection? */
+ if (alc880_is_multi_pin(nid)) {
+ struct alc_spec *spec = codec->spec;
+ int idx = alc880_multi_pin_idx(nid);
+ snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
+ AC_VERB_SET_CONNECT_SEL,
+ alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
+ }
+}
+
+static void alc662_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
+ for (i = 0; i <= HDA_SIDE; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
+ if (nid)
+ alc662_auto_set_output_and_unmute(codec, nid, pin_type,
+ i);
+ }
+}
+
+static void alc662_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front */
+ /* use dac 0 */
+ alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ pin = spec->autocfg.speaker_pins[0];
+ if (pin)
+ alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+}
+
+#define alc662_is_input_pin(nid) alc880_is_input_pin(nid)
+#define ALC662_PIN_CD_NID ALC880_PIN_CD_NID
+
+static void alc662_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (alc662_is_input_pin(nid)) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ (i <= AUTO_PIN_FRONT_MIC ?
+ PIN_VREF80 : PIN_IN));
+ if (nid != ALC662_PIN_CD_NID)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+ }
+}
+
+#define alc662_auto_init_input_src alc882_auto_init_input_src
+
+static int alc662_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t alc662_ignore[] = { 0x1d, 0 };
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ alc662_ignore);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs)
+ return 0; /* can't find valid BIOS pin config */
+
+ err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = alc662_auto_create_extra_out(spec,
+ spec->autocfg.speaker_pins[0],
+ "Speaker");
+ if (err < 0)
+ return err;
+ err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+ "Headphone");
+ if (err < 0)
+ return err;
+ err = alc662_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->num_mux_defs = 1;
+ spec->input_mux = &spec->private_imux;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs;
+ if (codec->vendor_id == 0x10ec0663)
+ spec->init_verbs[spec->num_init_verbs++] =
+ alc663_auto_init_verbs;
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ spec->mixers[spec->num_mixers] = alc662_capture_mixer;
+ spec->num_mixers++;
+
+ store_pin_configs(codec);
+ return 1;
+}
+
+/* additional initialization for auto-configuration model */
+static void alc662_auto_init(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ alc662_auto_init_multi_out(codec);
+ alc662_auto_init_hp_out(codec);
+ alc662_auto_init_analog_input(codec);
+ alc662_auto_init_input_src(codec);
+ if (spec->unsol_event)
+ alc_inithook(codec);
+}
+
+static int patch_alc662(struct hda_codec *codec)
+{
+ struct alc_spec *spec;
+ int err, board_config;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ alc_fix_pll_init(codec, 0x20, 0x04, 15);
+
+ board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST,
+ alc662_models,
+ alc662_cfg_tbl);
+ if (board_config < 0) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC662, "
+ "trying auto-probe from BIOS...\n");
+ board_config = ALC662_AUTO;
+ }
+
+ if (board_config == ALC662_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc662_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO
+ "hda_codec: Cannot set up configuration "
+ "from BIOS. Using base mode...\n");
+ board_config = ALC662_3ST_2ch_DIG;
+ }
+ }
+
+ if (board_config != ALC662_AUTO)
+ setup_preset(spec, &alc662_presets[board_config]);
+
+ if (codec->vendor_id == 0x10ec0663) {
+ spec->stream_name_analog = "ALC663 Analog";
+ spec->stream_name_digital = "ALC663 Digital";
+ } else if (codec->vendor_id == 0x10ec0272) {
+ spec->stream_name_analog = "ALC272 Analog";
+ spec->stream_name_digital = "ALC272 Digital";
+ } else {
+ spec->stream_name_analog = "ALC662 Analog";
+ spec->stream_name_digital = "ALC662 Digital";
+ }
+
+ spec->stream_analog_playback = &alc662_pcm_analog_playback;
+ spec->stream_analog_capture = &alc662_pcm_analog_capture;
+
+ spec->stream_digital_playback = &alc662_pcm_digital_playback;
+ spec->stream_digital_capture = &alc662_pcm_digital_capture;
+
+ spec->adc_nids = alc662_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
+ spec->capsrc_nids = alc662_capsrc_nids;
+
+ spec->vmaster_nid = 0x02;
+
+ codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC662_AUTO)
+ spec->init_hook = alc662_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc662_loopbacks;
+#endif
+
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_realtek[] = {
+ { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
+ { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
+ { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
+ { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
+ { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
+ { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
+ { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
+ .patch = patch_alc861 },
+ { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
+ { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
+ { .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd },
+ { .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2",
+ .patch = patch_alc883 },
+ { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
+ .patch = patch_alc662 },
+ { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
+ { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
+ { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
+ { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
+ { .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
+ .patch = patch_alc882 }, /* should be patch_alc883() in future */
+ { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
+ .patch = patch_alc882 }, /* should be patch_alc883() in future */
+ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
+ { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 },
+ { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
+ { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
+ .patch = patch_alc883 },
+ { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 },
+ {} /* terminator */
+};
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
new file mode 100644
index 0000000..9332b63
--- /dev/null
+++ b/sound/pci/hda/patch_si3054.c
@@ -0,0 +1,303 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for Silicon Labs 3054/5 modem codec
+ *
+ * Copyright (c) 2005 Sasha Khapyorsky <sashak@alsa-project.org>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+
+/* si3054 verbs */
+#define SI3054_VERB_READ_NODE 0x900
+#define SI3054_VERB_WRITE_NODE 0x100
+
+/* si3054 nodes (registers) */
+#define SI3054_EXTENDED_MID 2
+#define SI3054_LINE_RATE 3
+#define SI3054_LINE_LEVEL 4
+#define SI3054_GPIO_CFG 5
+#define SI3054_GPIO_POLARITY 6
+#define SI3054_GPIO_STICKY 7
+#define SI3054_GPIO_WAKEUP 8
+#define SI3054_GPIO_STATUS 9
+#define SI3054_GPIO_CONTROL 10
+#define SI3054_MISC_AFE 11
+#define SI3054_CHIPID 12
+#define SI3054_LINE_CFG1 13
+#define SI3054_LINE_STATUS 14
+#define SI3054_DC_TERMINATION 15
+#define SI3054_LINE_CONFIG 16
+#define SI3054_CALLPROG_ATT 17
+#define SI3054_SQ_CONTROL 18
+#define SI3054_MISC_CONTROL 19
+#define SI3054_RING_CTRL1 20
+#define SI3054_RING_CTRL2 21
+
+/* extended MID */
+#define SI3054_MEI_READY 0xf
+
+/* line level */
+#define SI3054_ATAG_MASK 0x00f0
+#define SI3054_DTAG_MASK 0xf000
+
+/* GPIO bits */
+#define SI3054_GPIO_OH 0x0001
+#define SI3054_GPIO_CID 0x0002
+
+/* chipid and revisions */
+#define SI3054_CHIPID_CODEC_REV_MASK 0x000f
+#define SI3054_CHIPID_DAA_REV_MASK 0x00f0
+#define SI3054_CHIPID_INTERNATIONAL 0x0100
+#define SI3054_CHIPID_DAA_ID 0x0f00
+#define SI3054_CHIPID_CODEC_ID (1<<12)
+
+/* si3054 codec registers (nodes) access macros */
+#define GET_REG(codec,reg) (snd_hda_codec_read(codec,reg,0,SI3054_VERB_READ_NODE,0))
+#define SET_REG(codec,reg,val) (snd_hda_codec_write(codec,reg,0,SI3054_VERB_WRITE_NODE,val))
+#define SET_REG_CACHE(codec,reg,val) \
+ snd_hda_codec_write_cache(codec,reg,0,SI3054_VERB_WRITE_NODE,val)
+
+
+struct si3054_spec {
+ unsigned international;
+ struct hda_pcm pcm;
+};
+
+
+/*
+ * Modem mixer
+ */
+
+#define PRIVATE_VALUE(reg,mask) ((reg<<16)|(mask&0xffff))
+#define PRIVATE_REG(val) ((val>>16)&0xffff)
+#define PRIVATE_MASK(val) (val&0xffff)
+
+#define si3054_switch_info snd_ctl_boolean_mono_info
+
+static int si3054_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uvalue)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 reg = PRIVATE_REG(kcontrol->private_value);
+ u16 mask = PRIVATE_MASK(kcontrol->private_value);
+ uvalue->value.integer.value[0] = (GET_REG(codec, reg)) & mask ? 1 : 0 ;
+ return 0;
+}
+
+static int si3054_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *uvalue)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 reg = PRIVATE_REG(kcontrol->private_value);
+ u16 mask = PRIVATE_MASK(kcontrol->private_value);
+ if (uvalue->value.integer.value[0])
+ SET_REG_CACHE(codec, reg, (GET_REG(codec, reg)) | mask);
+ else
+ SET_REG_CACHE(codec, reg, (GET_REG(codec, reg)) & ~mask);
+ return 0;
+}
+
+#define SI3054_KCONTROL(kname,reg,mask) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = kname, \
+ .info = si3054_switch_info, \
+ .get = si3054_switch_get, \
+ .put = si3054_switch_put, \
+ .private_value = PRIVATE_VALUE(reg,mask), \
+}
+
+
+static struct snd_kcontrol_new si3054_modem_mixer[] = {
+ SI3054_KCONTROL("Off-hook Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_OH),
+ SI3054_KCONTROL("Caller ID Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_CID),
+ {}
+};
+
+static int si3054_build_controls(struct hda_codec *codec)
+{
+ return snd_hda_add_new_ctls(codec, si3054_modem_mixer);
+}
+
+
+/*
+ * PCM callbacks
+ */
+
+static int si3054_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ u16 val;
+
+ SET_REG(codec, SI3054_LINE_RATE, substream->runtime->rate);
+ val = GET_REG(codec, SI3054_LINE_LEVEL);
+ val &= 0xff << (8 * (substream->stream != SNDRV_PCM_STREAM_PLAYBACK));
+ val |= ((stream_tag & 0xf) << 4) << (8 * (substream->stream == SNDRV_PCM_STREAM_PLAYBACK));
+ SET_REG(codec, SI3054_LINE_LEVEL, val);
+
+ snd_hda_codec_setup_stream(codec, hinfo->nid,
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int si3054_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ static unsigned int rates[] = { 8000, 9600, 16000 };
+ static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+ substream->runtime->hw.period_bytes_min = 80;
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates);
+}
+
+
+static struct hda_pcm_stream si3054_pcm = {
+ .substreams = 1,
+ .channels_min = 1,
+ .channels_max = 1,
+ .nid = 0x1,
+ .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_KNOT,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .maxbps = 16,
+ .ops = {
+ .open = si3054_pcm_open,
+ .prepare = si3054_pcm_prepare,
+ },
+};
+
+
+static int si3054_build_pcms(struct hda_codec *codec)
+{
+ struct si3054_spec *spec = codec->spec;
+ struct hda_pcm *info = &spec->pcm;
+ si3054_pcm.nid = codec->mfg;
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+ info->name = "Si3054 Modem";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm;
+ info->pcm_type = HDA_PCM_TYPE_MODEM;
+ return 0;
+}
+
+
+/*
+ * Init part
+ */
+
+static int si3054_init(struct hda_codec *codec)
+{
+ struct si3054_spec *spec = codec->spec;
+ unsigned wait_count;
+ u16 val;
+
+ snd_hda_codec_write(codec, AC_NODE_ROOT, 0, AC_VERB_SET_CODEC_RESET, 0);
+ snd_hda_codec_write(codec, codec->mfg, 0, AC_VERB_SET_STREAM_FORMAT, 0);
+ SET_REG(codec, SI3054_LINE_RATE, 9600);
+ SET_REG(codec, SI3054_LINE_LEVEL, SI3054_DTAG_MASK|SI3054_ATAG_MASK);
+ SET_REG(codec, SI3054_EXTENDED_MID, 0);
+
+ wait_count = 10;
+ do {
+ msleep(2);
+ val = GET_REG(codec, SI3054_EXTENDED_MID);
+ } while ((val & SI3054_MEI_READY) != SI3054_MEI_READY && wait_count--);
+
+ if((val&SI3054_MEI_READY) != SI3054_MEI_READY) {
+ snd_printk(KERN_ERR "si3054: cannot initialize. EXT MID = %04x\n", val);
+ /* let's pray that this is no fatal error */
+ /* return -EACCES; */
+ }
+
+ SET_REG(codec, SI3054_GPIO_POLARITY, 0xffff);
+ SET_REG(codec, SI3054_GPIO_CFG, 0x0);
+ SET_REG(codec, SI3054_MISC_AFE, 0);
+ SET_REG(codec, SI3054_LINE_CFG1,0x200);
+
+ if((GET_REG(codec,SI3054_LINE_STATUS) & (1<<6)) == 0) {
+ snd_printd("Link Frame Detect(FDT) is not ready (line status: %04x)\n",
+ GET_REG(codec,SI3054_LINE_STATUS));
+ }
+
+ spec->international = GET_REG(codec, SI3054_CHIPID) & SI3054_CHIPID_INTERNATIONAL;
+
+ return 0;
+}
+
+static void si3054_free(struct hda_codec *codec)
+{
+ kfree(codec->spec);
+}
+
+
+/*
+ */
+
+static struct hda_codec_ops si3054_patch_ops = {
+ .build_controls = si3054_build_controls,
+ .build_pcms = si3054_build_pcms,
+ .init = si3054_init,
+ .free = si3054_free,
+};
+
+static int patch_si3054(struct hda_codec *codec)
+{
+ struct si3054_spec *spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+ codec->spec = spec;
+ codec->patch_ops = si3054_patch_ops;
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_si3054[] = {
+ { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x11c13055, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x11c13155, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 },
+ { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 },
+ /* VIA HDA on Clevo m540 */
+ { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 },
+ /* Asus A8J Modem (SM56) */
+ { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 },
+ /* LG LW20 modem */
+ { .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 },
+ {}
+};
+
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
new file mode 100644
index 0000000..96f4c8d
--- /dev/null
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -0,0 +1,5355 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for SigmaTel STAC92xx
+ *
+ * Copyright (c) 2005 Embedded Alley Solutions, Inc.
+ * Matt Porter <mporter@embeddedalley.com>
+ *
+ * Based on patch_cmedia.c and patch_realtek.c
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include <sound/asoundef.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+#include "hda_beep.h"
+
+#define NUM_CONTROL_ALLOC 32
+
+#define STAC_VREF_EVENT 0x00
+#define STAC_INSERT_EVENT 0x10
+#define STAC_PWR_EVENT 0x20
+#define STAC_HP_EVENT 0x30
+
+enum {
+ STAC_REF,
+ STAC_9200_OQO,
+ STAC_9200_DELL_D21,
+ STAC_9200_DELL_D22,
+ STAC_9200_DELL_D23,
+ STAC_9200_DELL_M21,
+ STAC_9200_DELL_M22,
+ STAC_9200_DELL_M23,
+ STAC_9200_DELL_M24,
+ STAC_9200_DELL_M25,
+ STAC_9200_DELL_M26,
+ STAC_9200_DELL_M27,
+ STAC_9200_GATEWAY,
+ STAC_9200_PANASONIC,
+ STAC_9200_MODELS
+};
+
+enum {
+ STAC_9205_REF,
+ STAC_9205_DELL_M42,
+ STAC_9205_DELL_M43,
+ STAC_9205_DELL_M44,
+ STAC_9205_MODELS
+};
+
+enum {
+ STAC_92HD73XX_NO_JD, /* no jack-detection */
+ STAC_92HD73XX_REF,
+ STAC_DELL_M6_AMIC,
+ STAC_DELL_M6_DMIC,
+ STAC_DELL_M6_BOTH,
+ STAC_DELL_EQ,
+ STAC_92HD73XX_MODELS
+};
+
+enum {
+ STAC_92HD83XXX_REF,
+ STAC_92HD83XXX_MODELS
+};
+
+enum {
+ STAC_92HD71BXX_REF,
+ STAC_DELL_M4_1,
+ STAC_DELL_M4_2,
+ STAC_DELL_M4_3,
+ STAC_HP_M4,
+ STAC_HP_DV5,
+ STAC_92HD71BXX_MODELS
+};
+
+enum {
+ STAC_925x_REF,
+ STAC_M2_2,
+ STAC_MA6,
+ STAC_PA6,
+ STAC_925x_MODELS
+};
+
+enum {
+ STAC_D945_REF,
+ STAC_D945GTP3,
+ STAC_D945GTP5,
+ STAC_INTEL_MAC_V1,
+ STAC_INTEL_MAC_V2,
+ STAC_INTEL_MAC_V3,
+ STAC_INTEL_MAC_V4,
+ STAC_INTEL_MAC_V5,
+ STAC_INTEL_MAC_AUTO, /* This model is selected if no module parameter
+ * is given, one of the above models will be
+ * chosen according to the subsystem id. */
+ /* for backward compatibility */
+ STAC_MACMINI,
+ STAC_MACBOOK,
+ STAC_MACBOOK_PRO_V1,
+ STAC_MACBOOK_PRO_V2,
+ STAC_IMAC_INTEL,
+ STAC_IMAC_INTEL_20,
+ STAC_ECS_202,
+ STAC_922X_DELL_D81,
+ STAC_922X_DELL_D82,
+ STAC_922X_DELL_M81,
+ STAC_922X_DELL_M82,
+ STAC_922X_MODELS
+};
+
+enum {
+ STAC_D965_REF_NO_JD, /* no jack-detection */
+ STAC_D965_REF,
+ STAC_D965_3ST,
+ STAC_D965_5ST,
+ STAC_DELL_3ST,
+ STAC_DELL_BIOS,
+ STAC_927X_MODELS
+};
+
+struct sigmatel_spec {
+ struct snd_kcontrol_new *mixers[4];
+ unsigned int num_mixers;
+
+ int board_config;
+ unsigned int eapd_switch: 1;
+ unsigned int surr_switch: 1;
+ unsigned int line_switch: 1;
+ unsigned int mic_switch: 1;
+ unsigned int alt_switch: 1;
+ unsigned int hp_detect: 1;
+ unsigned int spdif_mute: 1;
+
+ /* gpio lines */
+ unsigned int eapd_mask;
+ unsigned int gpio_mask;
+ unsigned int gpio_dir;
+ unsigned int gpio_data;
+ unsigned int gpio_mute;
+
+ /* stream */
+ unsigned int stream_delay;
+
+ /* analog loopback */
+ unsigned char aloopback_mask;
+ unsigned char aloopback_shift;
+
+ /* power management */
+ unsigned int num_pwrs;
+ unsigned int *pwr_mapping;
+ hda_nid_t *pwr_nids;
+ hda_nid_t *dac_list;
+
+ /* playback */
+ struct hda_input_mux *mono_mux;
+ struct hda_input_mux *amp_mux;
+ unsigned int cur_mmux;
+ struct hda_multi_out multiout;
+ hda_nid_t dac_nids[5];
+
+ /* capture */
+ hda_nid_t *adc_nids;
+ unsigned int num_adcs;
+ hda_nid_t *mux_nids;
+ unsigned int num_muxes;
+ hda_nid_t *dmic_nids;
+ unsigned int num_dmics;
+ hda_nid_t *dmux_nids;
+ unsigned int num_dmuxes;
+ hda_nid_t *smux_nids;
+ unsigned int num_smuxes;
+ const char **spdif_labels;
+
+ hda_nid_t dig_in_nid;
+ hda_nid_t mono_nid;
+ hda_nid_t anabeep_nid;
+ hda_nid_t digbeep_nid;
+
+ /* pin widgets */
+ hda_nid_t *pin_nids;
+ unsigned int num_pins;
+ unsigned int *pin_configs;
+ unsigned int *bios_pin_configs;
+
+ /* codec specific stuff */
+ struct hda_verb *init;
+ struct snd_kcontrol_new *mixer;
+
+ /* capture source */
+ struct hda_input_mux *dinput_mux;
+ unsigned int cur_dmux[2];
+ struct hda_input_mux *input_mux;
+ unsigned int cur_mux[3];
+ struct hda_input_mux *sinput_mux;
+ unsigned int cur_smux[2];
+ unsigned int cur_amux;
+ hda_nid_t *amp_nids;
+ unsigned int num_amps;
+ unsigned int powerdown_adcs;
+
+ /* i/o switches */
+ unsigned int io_switch[2];
+ unsigned int clfe_swap;
+ unsigned int hp_switch; /* NID of HP as line-out */
+ unsigned int aloopback;
+
+ struct hda_pcm pcm_rec[2]; /* PCM information */
+
+ /* dynamic controls and input_mux */
+ struct auto_pin_cfg autocfg;
+ unsigned int num_kctl_alloc, num_kctl_used;
+ struct snd_kcontrol_new *kctl_alloc;
+ struct hda_input_mux private_dimux;
+ struct hda_input_mux private_imux;
+ struct hda_input_mux private_smux;
+ struct hda_input_mux private_amp_mux;
+ struct hda_input_mux private_mono_mux;
+};
+
+static hda_nid_t stac9200_adc_nids[1] = {
+ 0x03,
+};
+
+static hda_nid_t stac9200_mux_nids[1] = {
+ 0x0c,
+};
+
+static hda_nid_t stac9200_dac_nids[1] = {
+ 0x02,
+};
+
+static hda_nid_t stac92hd73xx_pwr_nids[8] = {
+ 0x0a, 0x0b, 0x0c, 0xd, 0x0e,
+ 0x0f, 0x10, 0x11
+};
+
+static hda_nid_t stac92hd73xx_slave_dig_outs[2] = {
+ 0x26, 0,
+};
+
+static hda_nid_t stac92hd73xx_adc_nids[2] = {
+ 0x1a, 0x1b
+};
+
+#define DELL_M6_AMP 2
+static hda_nid_t stac92hd73xx_amp_nids[3] = {
+ 0x0b, 0x0c, 0x0e
+};
+
+#define STAC92HD73XX_NUM_DMICS 2
+static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
+ 0x13, 0x14, 0
+};
+
+#define STAC92HD73_DAC_COUNT 5
+static hda_nid_t stac92hd73xx_dac_nids[STAC92HD73_DAC_COUNT] = {
+ 0x15, 0x16, 0x17, 0x18, 0x19,
+};
+
+static hda_nid_t stac92hd73xx_mux_nids[4] = {
+ 0x28, 0x29, 0x2a, 0x2b,
+};
+
+static hda_nid_t stac92hd73xx_dmux_nids[2] = {
+ 0x20, 0x21,
+};
+
+static hda_nid_t stac92hd73xx_smux_nids[2] = {
+ 0x22, 0x23,
+};
+
+#define STAC92HD83XXX_NUM_DMICS 2
+static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
+ 0x11, 0x12, 0
+};
+
+#define STAC92HD81_DAC_COUNT 2
+#define STAC92HD83_DAC_COUNT 3
+static hda_nid_t stac92hd83xxx_dac_nids[STAC92HD73_DAC_COUNT] = {
+ 0x13, 0x14, 0x22,
+};
+
+static hda_nid_t stac92hd83xxx_dmux_nids[2] = {
+ 0x17, 0x18,
+};
+
+static hda_nid_t stac92hd83xxx_adc_nids[2] = {
+ 0x15, 0x16,
+};
+
+static hda_nid_t stac92hd83xxx_pwr_nids[4] = {
+ 0xa, 0xb, 0xd, 0xe,
+};
+
+static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = {
+ 0x1e, 0,
+};
+
+static unsigned int stac92hd83xxx_pwr_mapping[4] = {
+ 0x03, 0x0c, 0x10, 0x40,
+};
+
+static hda_nid_t stac92hd71bxx_pwr_nids[3] = {
+ 0x0a, 0x0d, 0x0f
+};
+
+static hda_nid_t stac92hd71bxx_adc_nids[2] = {
+ 0x12, 0x13,
+};
+
+static hda_nid_t stac92hd71bxx_mux_nids[2] = {
+ 0x1a, 0x1b
+};
+
+static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
+ 0x1c, 0x1d,
+};
+
+static hda_nid_t stac92hd71bxx_smux_nids[2] = {
+ 0x24, 0x25,
+};
+
+static hda_nid_t stac92hd71bxx_dac_nids[1] = {
+ 0x10, /*0x11, */
+};
+
+#define STAC92HD71BXX_NUM_DMICS 2
+static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = {
+ 0x18, 0x19, 0
+};
+
+static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = {
+ 0x22, 0
+};
+
+static hda_nid_t stac925x_adc_nids[1] = {
+ 0x03,
+};
+
+static hda_nid_t stac925x_mux_nids[1] = {
+ 0x0f,
+};
+
+static hda_nid_t stac925x_dac_nids[1] = {
+ 0x02,
+};
+
+#define STAC925X_NUM_DMICS 1
+static hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = {
+ 0x15, 0
+};
+
+static hda_nid_t stac925x_dmux_nids[1] = {
+ 0x14,
+};
+
+static hda_nid_t stac922x_adc_nids[2] = {
+ 0x06, 0x07,
+};
+
+static hda_nid_t stac922x_mux_nids[2] = {
+ 0x12, 0x13,
+};
+
+static hda_nid_t stac927x_adc_nids[3] = {
+ 0x07, 0x08, 0x09
+};
+
+static hda_nid_t stac927x_mux_nids[3] = {
+ 0x15, 0x16, 0x17
+};
+
+static hda_nid_t stac927x_smux_nids[1] = {
+ 0x21,
+};
+
+static hda_nid_t stac927x_dac_nids[6] = {
+ 0x02, 0x03, 0x04, 0x05, 0x06, 0
+};
+
+static hda_nid_t stac927x_dmux_nids[1] = {
+ 0x1b,
+};
+
+#define STAC927X_NUM_DMICS 2
+static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = {
+ 0x13, 0x14, 0
+};
+
+static const char *stac927x_spdif_labels[5] = {
+ "Digital Playback", "ADAT", "Analog Mux 1",
+ "Analog Mux 2", "Analog Mux 3"
+};
+
+static hda_nid_t stac9205_adc_nids[2] = {
+ 0x12, 0x13
+};
+
+static hda_nid_t stac9205_mux_nids[2] = {
+ 0x19, 0x1a
+};
+
+static hda_nid_t stac9205_dmux_nids[1] = {
+ 0x1d,
+};
+
+static hda_nid_t stac9205_smux_nids[1] = {
+ 0x21,
+};
+
+#define STAC9205_NUM_DMICS 2
+static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = {
+ 0x17, 0x18, 0
+};
+
+static hda_nid_t stac9200_pin_nids[8] = {
+ 0x08, 0x09, 0x0d, 0x0e,
+ 0x0f, 0x10, 0x11, 0x12,
+};
+
+static hda_nid_t stac925x_pin_nids[8] = {
+ 0x07, 0x08, 0x0a, 0x0b,
+ 0x0c, 0x0d, 0x10, 0x11,
+};
+
+static hda_nid_t stac922x_pin_nids[10] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x10, 0x11, 0x15, 0x1b,
+};
+
+static hda_nid_t stac92hd73xx_pin_nids[13] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x10, 0x11, 0x12, 0x13,
+ 0x14, 0x22, 0x23
+};
+
+static hda_nid_t stac92hd83xxx_pin_nids[14] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x10, 0x11, 0x12, 0x13,
+ 0x1d, 0x1e, 0x1f, 0x20
+};
+static hda_nid_t stac92hd71bxx_pin_nids[11] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x14, 0x18, 0x19, 0x1e,
+ 0x1f,
+};
+
+static hda_nid_t stac927x_pin_nids[14] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x10, 0x11, 0x12, 0x13,
+ 0x14, 0x21, 0x22, 0x23,
+};
+
+static hda_nid_t stac9205_pin_nids[12] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x14, 0x16, 0x17, 0x18,
+ 0x21, 0x22,
+};
+
+#define stac92xx_amp_volume_info snd_hda_mixer_amp_volume_info
+
+static int stac92xx_amp_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = spec->amp_nids[spec->cur_amux];
+
+ kcontrol->private_value ^= get_amp_nid(kcontrol);
+ kcontrol->private_value |= nid;
+
+ return snd_hda_mixer_amp_volume_get(kcontrol, ucontrol);
+}
+
+static int stac92xx_amp_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = spec->amp_nids[spec->cur_amux];
+
+ kcontrol->private_value ^= get_amp_nid(kcontrol);
+ kcontrol->private_value |= nid;
+
+ return snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
+}
+
+static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->dinput_mux, uinfo);
+}
+
+static int stac92xx_dmux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_dmux[dmux_idx];
+ return 0;
+}
+
+static int stac92xx_dmux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ return snd_hda_input_mux_put(codec, spec->dinput_mux, ucontrol,
+ spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]);
+}
+
+static int stac92xx_smux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->sinput_mux, uinfo);
+}
+
+static int stac92xx_smux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_smux[smux_idx];
+ return 0;
+}
+
+static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_input_mux *smux = &spec->private_smux;
+ unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ int err, val;
+ hda_nid_t nid;
+
+ err = snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol,
+ spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]);
+ if (err < 0)
+ return err;
+
+ if (spec->spdif_mute) {
+ if (smux_idx == 0)
+ nid = spec->multiout.dig_out_nid;
+ else
+ nid = codec->slave_dig_outs[smux_idx - 1];
+ if (spec->cur_smux[smux_idx] == smux->num_items - 1)
+ val = AMP_OUT_MUTE;
+ else
+ val = AMP_OUT_UNMUTE;
+ /* un/mute SPDIF out */
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, val);
+ }
+ return 0;
+}
+
+static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int stac92xx_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ return 0;
+}
+
+static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);
+}
+
+static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->mono_mux, uinfo);
+}
+
+static int stac92xx_mono_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mmux;
+ return 0;
+}
+
+static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ return snd_hda_input_mux_put(codec, spec->mono_mux, ucontrol,
+ spec->mono_nid, &spec->cur_mmux);
+}
+
+static int stac92xx_amp_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->amp_mux, uinfo);
+}
+
+static int stac92xx_amp_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->cur_amux;
+ return 0;
+}
+
+static int stac92xx_amp_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ struct snd_kcontrol *ctl =
+ snd_hda_find_mixer_ctl(codec, "Amp Capture Volume");
+ if (!ctl)
+ return -EINVAL;
+
+ snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &ctl->id);
+
+ return snd_hda_input_mux_put(codec, spec->amp_mux, ucontrol,
+ 0, &spec->cur_amux);
+}
+
+#define stac92xx_aloopback_info snd_ctl_boolean_mono_info
+
+static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.integer.value[0] = !!(spec->aloopback &
+ (spec->aloopback_mask << idx));
+ return 0;
+}
+
+static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int dac_mode;
+ unsigned int val, idx_val;
+
+ idx_val = spec->aloopback_mask << idx;
+ if (ucontrol->value.integer.value[0])
+ val = spec->aloopback | idx_val;
+ else
+ val = spec->aloopback & ~idx_val;
+ if (spec->aloopback == val)
+ return 0;
+
+ spec->aloopback = val;
+
+ /* Only return the bits defined by the shift value of the
+ * first two bytes of the mask
+ */
+ dac_mode = snd_hda_codec_read(codec, codec->afg, 0,
+ kcontrol->private_value & 0xFFFF, 0x0);
+ dac_mode >>= spec->aloopback_shift;
+
+ if (spec->aloopback & idx_val) {
+ snd_hda_power_up(codec);
+ dac_mode |= idx_val;
+ } else {
+ snd_hda_power_down(codec);
+ dac_mode &= ~idx_val;
+ }
+
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ kcontrol->private_value >> 16, dac_mode);
+
+ return 1;
+}
+
+static struct hda_verb stac9200_core_init[] = {
+ /* set dac0mux for dac converter */
+ { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {}
+};
+
+static struct hda_verb stac9200_eapd_init[] = {
+ /* set dac0mux for dac converter */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ {}
+};
+
+static struct hda_verb stac92hd73xx_6ch_core_init[] = {
+ /* set master volume and direct control */
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* setup audio connections */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* setup adcs to point to mixer */
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* setup import muxs */
+ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {}
+};
+
+static struct hda_verb dell_eq_core_init[] = {
+ /* set master volume to max value without distortion
+ * and direct control */
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
+ /* setup audio connections */
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02},
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* setup adcs to point to mixer */
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ /* setup import muxs */
+ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {}
+};
+
+static struct hda_verb dell_m6_core_init[] = {
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* setup audio connections */
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* setup adcs to point to mixer */
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ /* setup import muxs */
+ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {}
+};
+
+static struct hda_verb stac92hd73xx_8ch_core_init[] = {
+ /* set master volume and direct control */
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* setup audio connections */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* connect hp ports to dac3 */
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x03},
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* setup adcs to point to mixer */
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* setup import muxs */
+ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {}
+};
+
+static struct hda_verb stac92hd73xx_10ch_core_init[] = {
+ /* set master volume and direct control */
+ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* setup audio connections */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ /* dac3 is connected to import3 mux */
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f},
+ /* connect hp ports to dac4 */
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x04},
+ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x04},
+ /* setup adcs to point to mixer */
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* setup import muxs */
+ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03},
+ {}
+};
+
+static struct hda_verb stac92hd83xxx_core_init[] = {
+ /* start of config #1 */
+ { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3},
+
+ /* start of config #2 */
+ { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
+ { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
+ { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
+
+ /* power state controls amps */
+ { 0x01, AC_VERB_SET_EAPD, 1 << 2},
+ {}
+};
+
+static struct hda_verb stac92hd71bxx_core_init[] = {
+ /* set master volume and direct control */
+ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* connect headphone jack to dac1 */
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
+ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {}
+};
+
+#define HD_DISABLE_PORTF 2
+static struct hda_verb stac92hd71bxx_analog_core_init[] = {
+ /* start of config #1 */
+
+ /* connect port 0f to audio mixer */
+ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
+ /* unmute right and left channels for node 0x0f */
+ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* start of config #2 */
+
+ /* set master volume and direct control */
+ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* connect headphone jack to dac1 */
+ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* unmute right and left channels for nodes 0x0a, 0xd */
+ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {}
+};
+
+static struct hda_verb stac925x_core_init[] = {
+ /* set dac0mux for dac converter */
+ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {}
+};
+
+static struct hda_verb stac922x_core_init[] = {
+ /* set master volume and direct control */
+ { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ {}
+};
+
+static struct hda_verb d965_core_init[] = {
+ /* set master volume and direct control */
+ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* unmute node 0x1b */
+ { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* select node 0x03 as DAC */
+ { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {}
+};
+
+static struct hda_verb stac927x_core_init[] = {
+ /* set master volume and direct control */
+ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* enable analog pc beep path */
+ { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
+ {}
+};
+
+static struct hda_verb stac9205_core_init[] = {
+ /* set master volume and direct control */
+ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+ /* enable analog pc beep path */
+ { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
+ {}
+};
+
+#define STAC_MONO_MUX \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Mono Mux", \
+ .count = 1, \
+ .info = stac92xx_mono_mux_enum_info, \
+ .get = stac92xx_mono_mux_enum_get, \
+ .put = stac92xx_mono_mux_enum_put, \
+ }
+
+#define STAC_AMP_MUX \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Amp Selector Capture Switch", \
+ .count = 1, \
+ .info = stac92xx_amp_mux_enum_info, \
+ .get = stac92xx_amp_mux_enum_get, \
+ .put = stac92xx_amp_mux_enum_put, \
+ }
+
+#define STAC_AMP_VOL(xname, nid, chs, idx, dir) \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = 0, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+ .info = stac92xx_amp_volume_info, \
+ .get = stac92xx_amp_volume_get, \
+ .put = stac92xx_amp_volume_put, \
+ .tlv = { .c = snd_hda_mixer_amp_tlv }, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \
+ }
+
+#define STAC_INPUT_SOURCE(cnt) \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Input Source", \
+ .count = cnt, \
+ .info = stac92xx_mux_enum_info, \
+ .get = stac92xx_mux_enum_get, \
+ .put = stac92xx_mux_enum_put, \
+ }
+
+#define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Analog Loopback", \
+ .count = cnt, \
+ .info = stac92xx_aloopback_info, \
+ .get = stac92xx_aloopback_get, \
+ .put = stac92xx_aloopback_put, \
+ .private_value = verb_read | (verb_write << 16), \
+ }
+
+static struct snd_kcontrol_new stac9200_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
+ STAC_INPUT_SOURCE(1),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
+ { } /* end */
+};
+
+#define DELL_M6_MIXER 6
+static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
+ /* start of config #1 */
+ HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
+
+ /* start of config #2 */
+ HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
+
+ STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
+
+ { } /* end */
+};
+
+static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
+ STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
+ STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT),
+ { } /* end */
+};
+
+
+static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT),
+
+ /*
+ HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT),
+ */
+ { } /* end */
+};
+
+static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
+ STAC_INPUT_SOURCE(2),
+ STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
+ /* analog pc-beep replaced with digital beep support */
+ /*
+ HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
+ */
+
+ HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT),
+
+ HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT),
+
+ HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),
+
+ HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
+ HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
+ STAC_INPUT_SOURCE(2),
+ STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new stac925x_mixer[] = {
+ STAC_INPUT_SOURCE(1),
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new stac9205_mixer[] = {
+ STAC_INPUT_SOURCE(2),
+ STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+/* This needs to be generated dynamically based on sequence */
+static struct snd_kcontrol_new stac922x_mixer[] = {
+ STAC_INPUT_SOURCE(2),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+static struct snd_kcontrol_new stac927x_mixer[] = {
+ STAC_INPUT_SOURCE(3),
+ STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new stac_dmux_mixer = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Input Source",
+ /* count set later */
+ .info = stac92xx_dmux_enum_info,
+ .get = stac92xx_dmux_enum_get,
+ .put = stac92xx_dmux_enum_put,
+};
+
+static struct snd_kcontrol_new stac_smux_mixer = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Playback Source",
+ /* count set later */
+ .info = stac92xx_smux_enum_info,
+ .get = stac92xx_smux_enum_get,
+ .put = stac92xx_smux_enum_put,
+};
+
+static const char *slave_vols[] = {
+ "Front Playback Volume",
+ "Surround Playback Volume",
+ "Center Playback Volume",
+ "LFE Playback Volume",
+ "Side Playback Volume",
+ "Headphone Playback Volume",
+ "Headphone Playback Volume",
+ "Speaker Playback Volume",
+ "External Speaker Playback Volume",
+ "Speaker2 Playback Volume",
+ NULL
+};
+
+static const char *slave_sws[] = {
+ "Front Playback Switch",
+ "Surround Playback Switch",
+ "Center Playback Switch",
+ "LFE Playback Switch",
+ "Side Playback Switch",
+ "Headphone Playback Switch",
+ "Headphone Playback Switch",
+ "Speaker Playback Switch",
+ "External Speaker Playback Switch",
+ "Speaker2 Playback Switch",
+ "IEC958 Playback Switch",
+ NULL
+};
+
+static int stac92xx_build_controls(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int err;
+ int i;
+
+ err = snd_hda_add_new_ctls(codec, spec->mixer);
+ if (err < 0)
+ return err;
+
+ for (i = 0; i < spec->num_mixers; i++) {
+ err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+ if (err < 0)
+ return err;
+ }
+ if (spec->num_dmuxes > 0) {
+ stac_dmux_mixer.count = spec->num_dmuxes;
+ err = snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&stac_dmux_mixer, codec));
+ if (err < 0)
+ return err;
+ }
+ if (spec->num_smuxes > 0) {
+ int wcaps = get_wcaps(codec, spec->multiout.dig_out_nid);
+ struct hda_input_mux *smux = &spec->private_smux;
+ /* check for mute support on SPDIF out */
+ if (wcaps & AC_WCAP_OUT_AMP) {
+ smux->items[smux->num_items].label = "Off";
+ smux->items[smux->num_items].index = 0;
+ smux->num_items++;
+ spec->spdif_mute = 1;
+ }
+ stac_smux_mixer.count = spec->num_smuxes;
+ err = snd_ctl_add(codec->bus->card,
+ snd_ctl_new1(&stac_smux_mixer, codec));
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->multiout.dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
+ }
+ if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+ if (err < 0)
+ return err;
+ }
+
+ /* if we have no master control, let's create it */
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
+ unsigned int vmaster_tlv[4];
+ snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
+ HDA_OUTPUT, vmaster_tlv);
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ vmaster_tlv, slave_vols);
+ if (err < 0)
+ return err;
+ }
+ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+ err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, slave_sws);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static unsigned int ref9200_pin_configs[8] = {
+ 0x01c47010, 0x01447010, 0x0221401f, 0x01114010,
+ 0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
+};
+
+/*
+ STAC 9200 pin configs for
+ 102801A8
+ 102801DE
+ 102801E8
+*/
+static unsigned int dell9200_d21_pin_configs[8] = {
+ 0x400001f0, 0x400001f1, 0x02214030, 0x01014010,
+ 0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
+};
+
+/*
+ STAC 9200 pin configs for
+ 102801C0
+ 102801C1
+*/
+static unsigned int dell9200_d22_pin_configs[8] = {
+ 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010,
+ 0x01813020, 0x02a19021, 0x90100140, 0x400001f2,
+};
+
+/*
+ STAC 9200 pin configs for
+ 102801C4 (Dell Dimension E310)
+ 102801C5
+ 102801C7
+ 102801D9
+ 102801DA
+ 102801E3
+*/
+static unsigned int dell9200_d23_pin_configs[8] = {
+ 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010,
+ 0x01813020, 0x01a19021, 0x90100140, 0x400001f2,
+};
+
+
+/*
+ STAC 9200-32 pin configs for
+ 102801B5 (Dell Inspiron 630m)
+ 102801D8 (Dell Inspiron 640m)
+*/
+static unsigned int dell9200_m21_pin_configs[8] = {
+ 0x40c003fa, 0x03441340, 0x0321121f, 0x90170310,
+ 0x408003fb, 0x03a11020, 0x401003fc, 0x403003fd,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801C2 (Dell Latitude D620)
+ 102801C8
+ 102801CC (Dell Latitude D820)
+ 102801D4
+ 102801D6
+*/
+static unsigned int dell9200_m22_pin_configs[8] = {
+ 0x40c003fa, 0x0144131f, 0x0321121f, 0x90170310,
+ 0x90a70321, 0x03a11020, 0x401003fb, 0x40f000fc,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801CE (Dell XPS M1710)
+ 102801CF (Dell Precision M90)
+*/
+static unsigned int dell9200_m23_pin_configs[8] = {
+ 0x40c003fa, 0x01441340, 0x0421421f, 0x90170310,
+ 0x408003fb, 0x04a1102e, 0x90170311, 0x403003fc,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801C9
+ 102801CA
+ 102801CB (Dell Latitude 120L)
+ 102801D3
+*/
+static unsigned int dell9200_m24_pin_configs[8] = {
+ 0x40c003fa, 0x404003fb, 0x0321121f, 0x90170310,
+ 0x408003fc, 0x03a11020, 0x401003fd, 0x403003fe,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801BD (Dell Inspiron E1505n)
+ 102801EE
+ 102801EF
+*/
+static unsigned int dell9200_m25_pin_configs[8] = {
+ 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310,
+ 0x408003fb, 0x04a11020, 0x401003fc, 0x403003fd,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801F5 (Dell Inspiron 1501)
+ 102801F6
+*/
+static unsigned int dell9200_m26_pin_configs[8] = {
+ 0x40c003fa, 0x404003fb, 0x0421121f, 0x90170310,
+ 0x408003fc, 0x04a11020, 0x401003fd, 0x403003fe,
+};
+
+/*
+ STAC 9200-32
+ 102801CD (Dell Inspiron E1705/9400)
+*/
+static unsigned int dell9200_m27_pin_configs[8] = {
+ 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310,
+ 0x90170310, 0x04a11020, 0x90170310, 0x40f003fc,
+};
+
+static unsigned int oqo9200_pin_configs[8] = {
+ 0x40c000f0, 0x404000f1, 0x0221121f, 0x02211210,
+ 0x90170111, 0x90a70120, 0x400000f2, 0x400000f3,
+};
+
+
+static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
+ [STAC_REF] = ref9200_pin_configs,
+ [STAC_9200_OQO] = oqo9200_pin_configs,
+ [STAC_9200_DELL_D21] = dell9200_d21_pin_configs,
+ [STAC_9200_DELL_D22] = dell9200_d22_pin_configs,
+ [STAC_9200_DELL_D23] = dell9200_d23_pin_configs,
+ [STAC_9200_DELL_M21] = dell9200_m21_pin_configs,
+ [STAC_9200_DELL_M22] = dell9200_m22_pin_configs,
+ [STAC_9200_DELL_M23] = dell9200_m23_pin_configs,
+ [STAC_9200_DELL_M24] = dell9200_m24_pin_configs,
+ [STAC_9200_DELL_M25] = dell9200_m25_pin_configs,
+ [STAC_9200_DELL_M26] = dell9200_m26_pin_configs,
+ [STAC_9200_DELL_M27] = dell9200_m27_pin_configs,
+ [STAC_9200_PANASONIC] = ref9200_pin_configs,
+};
+
+static const char *stac9200_models[STAC_9200_MODELS] = {
+ [STAC_REF] = "ref",
+ [STAC_9200_OQO] = "oqo",
+ [STAC_9200_DELL_D21] = "dell-d21",
+ [STAC_9200_DELL_D22] = "dell-d22",
+ [STAC_9200_DELL_D23] = "dell-d23",
+ [STAC_9200_DELL_M21] = "dell-m21",
+ [STAC_9200_DELL_M22] = "dell-m22",
+ [STAC_9200_DELL_M23] = "dell-m23",
+ [STAC_9200_DELL_M24] = "dell-m24",
+ [STAC_9200_DELL_M25] = "dell-m25",
+ [STAC_9200_DELL_M26] = "dell-m26",
+ [STAC_9200_DELL_M27] = "dell-m27",
+ [STAC_9200_GATEWAY] = "gateway",
+ [STAC_9200_PANASONIC] = "panasonic",
+};
+
+static struct snd_pci_quirk stac9200_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ "DFI LanParty", STAC_REF),
+ /* Dell laptops have BIOS problem */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8,
+ "unknown Dell", STAC_9200_DELL_D21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01b5,
+ "Dell Inspiron 630m", STAC_9200_DELL_M21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01bd,
+ "Dell Inspiron E1505n", STAC_9200_DELL_M25),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c0,
+ "unknown Dell", STAC_9200_DELL_D22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c1,
+ "unknown Dell", STAC_9200_DELL_D22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c2,
+ "Dell Latitude D620", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c5,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c7,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c8,
+ "unknown Dell", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c9,
+ "unknown Dell", STAC_9200_DELL_M24),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ca,
+ "unknown Dell", STAC_9200_DELL_M24),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cb,
+ "Dell Latitude 120L", STAC_9200_DELL_M24),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cc,
+ "Dell Latitude D820", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cd,
+ "Dell Inspiron E1705/9400", STAC_9200_DELL_M27),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ce,
+ "Dell XPS M1710", STAC_9200_DELL_M23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cf,
+ "Dell Precision M90", STAC_9200_DELL_M23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d3,
+ "unknown Dell", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d4,
+ "unknown Dell", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d6,
+ "unknown Dell", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d8,
+ "Dell Inspiron 640m", STAC_9200_DELL_M21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d9,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01da,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01de,
+ "unknown Dell", STAC_9200_DELL_D21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01e3,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01e8,
+ "unknown Dell", STAC_9200_DELL_D21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ee,
+ "unknown Dell", STAC_9200_DELL_M25),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ef,
+ "unknown Dell", STAC_9200_DELL_M25),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f5,
+ "Dell Inspiron 1501", STAC_9200_DELL_M26),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f6,
+ "unknown Dell", STAC_9200_DELL_M26),
+ /* Panasonic */
+ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC),
+ /* Gateway machines needs EAPD to be set on resume */
+ SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_GATEWAY),
+ SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*",
+ STAC_9200_GATEWAY),
+ SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707",
+ STAC_9200_GATEWAY),
+ /* OQO Mobile */
+ SND_PCI_QUIRK(0x1106, 0x3288, "OQO Model 2", STAC_9200_OQO),
+ {} /* terminator */
+};
+
+static unsigned int ref925x_pin_configs[8] = {
+ 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021,
+ 0x90a70320, 0x02214210, 0x01019020, 0x9033032e,
+};
+
+static unsigned int stac925x_MA6_pin_configs[8] = {
+ 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021,
+ 0x90a70320, 0x90100211, 0x400003f1, 0x9033032e,
+};
+
+static unsigned int stac925x_PA6_pin_configs[8] = {
+ 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021,
+ 0x50a103f0, 0x90100211, 0x400003f1, 0x9033032e,
+};
+
+static unsigned int stac925xM2_2_pin_configs[8] = {
+ 0x40c003f3, 0x424503f2, 0x04180011, 0x02a19020,
+ 0x50a103f0, 0x90100212, 0x400003f1, 0x9033032e,
+};
+
+static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = {
+ [STAC_REF] = ref925x_pin_configs,
+ [STAC_M2_2] = stac925xM2_2_pin_configs,
+ [STAC_MA6] = stac925x_MA6_pin_configs,
+ [STAC_PA6] = stac925x_PA6_pin_configs,
+};
+
+static const char *stac925x_models[STAC_925x_MODELS] = {
+ [STAC_REF] = "ref",
+ [STAC_M2_2] = "m2-2",
+ [STAC_MA6] = "m6",
+ [STAC_PA6] = "pa6",
+};
+
+static struct snd_pci_quirk stac925x_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF),
+ SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF),
+ SND_PCI_QUIRK(0x107b, 0x0316, "Gateway M255", STAC_REF),
+ SND_PCI_QUIRK(0x107b, 0x0366, "Gateway MP6954", STAC_REF),
+ SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_MA6),
+ SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_PA6),
+ SND_PCI_QUIRK(0x1002, 0x437b, "Gateway MX6453", STAC_M2_2),
+ {} /* terminator */
+};
+
+static unsigned int ref92hd73xx_pin_configs[13] = {
+ 0x02214030, 0x02a19040, 0x01a19020, 0x02214030,
+ 0x0181302e, 0x01014010, 0x01014020, 0x01014030,
+ 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050,
+ 0x01452050,
+};
+
+static unsigned int dell_m6_pin_configs[13] = {
+ 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110,
+ 0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0,
+ 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0,
+ 0x4f0000f0,
+};
+
+static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
+ [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs,
+ [STAC_DELL_M6_AMIC] = dell_m6_pin_configs,
+ [STAC_DELL_M6_DMIC] = dell_m6_pin_configs,
+ [STAC_DELL_M6_BOTH] = dell_m6_pin_configs,
+ [STAC_DELL_EQ] = dell_m6_pin_configs,
+};
+
+static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
+ [STAC_92HD73XX_NO_JD] = "no-jd",
+ [STAC_92HD73XX_REF] = "ref",
+ [STAC_DELL_M6_AMIC] = "dell-m6-amic",
+ [STAC_DELL_M6_DMIC] = "dell-m6-dmic",
+ [STAC_DELL_M6_BOTH] = "dell-m6",
+ [STAC_DELL_EQ] = "dell-eq",
+};
+
+static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ "DFI LanParty", STAC_92HD73XX_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254,
+ "Dell Studio 1535", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255,
+ "unknown Dell", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0256,
+ "unknown Dell", STAC_DELL_M6_BOTH),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0257,
+ "unknown Dell", STAC_DELL_M6_BOTH),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025e,
+ "unknown Dell", STAC_DELL_M6_AMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025f,
+ "unknown Dell", STAC_DELL_M6_AMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271,
+ "unknown Dell", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0272,
+ "unknown Dell", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x029f,
+ "Dell Studio 1537", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0,
+ "Dell Studio 17", STAC_DELL_M6_DMIC),
+ {} /* terminator */
+};
+
+static unsigned int ref92hd83xxx_pin_configs[14] = {
+ 0x02214030, 0x02211010, 0x02a19020, 0x02170130,
+ 0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
+ 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0,
+ 0x01451160, 0x98560170,
+};
+
+static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
+ [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
+};
+
+static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
+ [STAC_92HD83XXX_REF] = "ref",
+};
+
+static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ "DFI LanParty", STAC_92HD71BXX_REF),
+ {} /* terminator */
+};
+
+static unsigned int ref92hd71bxx_pin_configs[11] = {
+ 0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
+ 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
+ 0x90a000f0, 0x01452050, 0x01452050,
+};
+
+static unsigned int dell_m4_1_pin_configs[11] = {
+ 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110,
+ 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0,
+ 0x40f000f0, 0x4f0000f0, 0x4f0000f0,
+};
+
+static unsigned int dell_m4_2_pin_configs[11] = {
+ 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
+ 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0,
+ 0x40f000f0, 0x044413b0, 0x044413b0,
+};
+
+static unsigned int dell_m4_3_pin_configs[11] = {
+ 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
+ 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0,
+ 0x40f000f0, 0x044413b0, 0x044413b0,
+};
+
+static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
+ [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs,
+ [STAC_DELL_M4_1] = dell_m4_1_pin_configs,
+ [STAC_DELL_M4_2] = dell_m4_2_pin_configs,
+ [STAC_DELL_M4_3] = dell_m4_3_pin_configs,
+ [STAC_HP_M4] = NULL,
+ [STAC_HP_DV5] = NULL,
+};
+
+static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
+ [STAC_92HD71BXX_REF] = "ref",
+ [STAC_DELL_M4_1] = "dell-m4-1",
+ [STAC_DELL_M4_2] = "dell-m4-2",
+ [STAC_DELL_M4_3] = "dell-m4-3",
+ [STAC_HP_M4] = "hp-m4",
+ [STAC_HP_DV5] = "hp-dv5",
+};
+
+static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ "DFI LanParty", STAC_92HD71BXX_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
+ "HP dv5", STAC_HP_M4),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
+ "HP dv7", STAC_HP_M4),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
+ "HP dv7", STAC_HP_M4),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603,
+ "HP dv5", STAC_HP_DV5),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
+ "unknown HP", STAC_HP_M4),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0250,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024f,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024d,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0251,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0277,
+ "unknown Dell", STAC_DELL_M4_1),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0263,
+ "unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0265,
+ "unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0262,
+ "unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264,
+ "unknown Dell", STAC_DELL_M4_2),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02aa,
+ "unknown Dell", STAC_DELL_M4_3),
+ {} /* terminator */
+};
+
+static unsigned int ref922x_pin_configs[10] = {
+ 0x01014010, 0x01016011, 0x01012012, 0x0221401f,
+ 0x01813122, 0x01011014, 0x01441030, 0x01c41030,
+ 0x40000100, 0x40000100,
+};
+
+/*
+ STAC 922X pin configs for
+ 102801A7
+ 102801AB
+ 102801A9
+ 102801D1
+ 102801D2
+*/
+static unsigned int dell_922x_d81_pin_configs[10] = {
+ 0x02214030, 0x01a19021, 0x01111012, 0x01114010,
+ 0x02a19020, 0x01117011, 0x400001f0, 0x400001f1,
+ 0x01813122, 0x400001f2,
+};
+
+/*
+ STAC 922X pin configs for
+ 102801AC
+ 102801D0
+*/
+static unsigned int dell_922x_d82_pin_configs[10] = {
+ 0x02214030, 0x01a19021, 0x01111012, 0x01114010,
+ 0x02a19020, 0x01117011, 0x01451140, 0x400001f0,
+ 0x01813122, 0x400001f1,
+};
+
+/*
+ STAC 922X pin configs for
+ 102801BF
+*/
+static unsigned int dell_922x_m81_pin_configs[10] = {
+ 0x0321101f, 0x01112024, 0x01111222, 0x91174220,
+ 0x03a11050, 0x01116221, 0x90a70330, 0x01452340,
+ 0x40C003f1, 0x405003f0,
+};
+
+/*
+ STAC 9221 A1 pin configs for
+ 102801D7 (Dell XPS M1210)
+*/
+static unsigned int dell_922x_m82_pin_configs[10] = {
+ 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310,
+ 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2,
+ 0x508003f3, 0x405003f4,
+};
+
+static unsigned int d945gtp3_pin_configs[10] = {
+ 0x0221401f, 0x01a19022, 0x01813021, 0x01014010,
+ 0x40000100, 0x40000100, 0x40000100, 0x40000100,
+ 0x02a19120, 0x40000100,
+};
+
+static unsigned int d945gtp5_pin_configs[10] = {
+ 0x0221401f, 0x01011012, 0x01813024, 0x01014010,
+ 0x01a19021, 0x01016011, 0x01452130, 0x40000100,
+ 0x02a19320, 0x40000100,
+};
+
+static unsigned int intel_mac_v1_pin_configs[10] = {
+ 0x0121e21f, 0x400000ff, 0x9017e110, 0x400000fd,
+ 0x400000fe, 0x0181e020, 0x1145e030, 0x11c5e240,
+ 0x400000fc, 0x400000fb,
+};
+
+static unsigned int intel_mac_v2_pin_configs[10] = {
+ 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd,
+ 0x400000fe, 0x0181e020, 0x1145e230, 0x500000fa,
+ 0x400000fc, 0x400000fb,
+};
+
+static unsigned int intel_mac_v3_pin_configs[10] = {
+ 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd,
+ 0x400000fe, 0x0181e020, 0x1145e230, 0x11c5e240,
+ 0x400000fc, 0x400000fb,
+};
+
+static unsigned int intel_mac_v4_pin_configs[10] = {
+ 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f,
+ 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240,
+ 0x400000fc, 0x400000fb,
+};
+
+static unsigned int intel_mac_v5_pin_configs[10] = {
+ 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f,
+ 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240,
+ 0x400000fc, 0x400000fb,
+};
+
+static unsigned int ecs202_pin_configs[10] = {
+ 0x0221401f, 0x02a19020, 0x01a19020, 0x01114010,
+ 0x408000f0, 0x01813022, 0x074510a0, 0x40c400f1,
+ 0x9037012e, 0x40e000f2,
+};
+
+static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
+ [STAC_D945_REF] = ref922x_pin_configs,
+ [STAC_D945GTP3] = d945gtp3_pin_configs,
+ [STAC_D945GTP5] = d945gtp5_pin_configs,
+ [STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs,
+ [STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs,
+ [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs,
+ [STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs,
+ [STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs,
+ [STAC_INTEL_MAC_AUTO] = intel_mac_v3_pin_configs,
+ /* for backward compatibility */
+ [STAC_MACMINI] = intel_mac_v3_pin_configs,
+ [STAC_MACBOOK] = intel_mac_v5_pin_configs,
+ [STAC_MACBOOK_PRO_V1] = intel_mac_v3_pin_configs,
+ [STAC_MACBOOK_PRO_V2] = intel_mac_v3_pin_configs,
+ [STAC_IMAC_INTEL] = intel_mac_v2_pin_configs,
+ [STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs,
+ [STAC_ECS_202] = ecs202_pin_configs,
+ [STAC_922X_DELL_D81] = dell_922x_d81_pin_configs,
+ [STAC_922X_DELL_D82] = dell_922x_d82_pin_configs,
+ [STAC_922X_DELL_M81] = dell_922x_m81_pin_configs,
+ [STAC_922X_DELL_M82] = dell_922x_m82_pin_configs,
+};
+
+static const char *stac922x_models[STAC_922X_MODELS] = {
+ [STAC_D945_REF] = "ref",
+ [STAC_D945GTP5] = "5stack",
+ [STAC_D945GTP3] = "3stack",
+ [STAC_INTEL_MAC_V1] = "intel-mac-v1",
+ [STAC_INTEL_MAC_V2] = "intel-mac-v2",
+ [STAC_INTEL_MAC_V3] = "intel-mac-v3",
+ [STAC_INTEL_MAC_V4] = "intel-mac-v4",
+ [STAC_INTEL_MAC_V5] = "intel-mac-v5",
+ [STAC_INTEL_MAC_AUTO] = "intel-mac-auto",
+ /* for backward compatibility */
+ [STAC_MACMINI] = "macmini",
+ [STAC_MACBOOK] = "macbook",
+ [STAC_MACBOOK_PRO_V1] = "macbook-pro-v1",
+ [STAC_MACBOOK_PRO_V2] = "macbook-pro",
+ [STAC_IMAC_INTEL] = "imac-intel",
+ [STAC_IMAC_INTEL_20] = "imac-intel-20",
+ [STAC_ECS_202] = "ecs202",
+ [STAC_922X_DELL_D81] = "dell-d81",
+ [STAC_922X_DELL_D82] = "dell-d82",
+ [STAC_922X_DELL_M81] = "dell-m81",
+ [STAC_922X_DELL_M82] = "dell-m82",
+};
+
+static struct snd_pci_quirk stac922x_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ "DFI LanParty", STAC_D945_REF),
+ /* Intel 945G based systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0202,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0606,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0601,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0111,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x1115,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x1116,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x1117,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x1118,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x1119,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x8826,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5049,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5055,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5048,
+ "Intel D945G", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0110,
+ "Intel D945G", STAC_D945GTP3),
+ /* Intel D945G 5-stack systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0404,
+ "Intel D945G", STAC_D945GTP5),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0303,
+ "Intel D945G", STAC_D945GTP5),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0013,
+ "Intel D945G", STAC_D945GTP5),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0417,
+ "Intel D945G", STAC_D945GTP5),
+ /* Intel 945P based systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0b0b,
+ "Intel D945P", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0112,
+ "Intel D945P", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0d0d,
+ "Intel D945P", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0909,
+ "Intel D945P", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0505,
+ "Intel D945P", STAC_D945GTP3),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707,
+ "Intel D945P", STAC_D945GTP5),
+ /* other systems */
+ /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */
+ SND_PCI_QUIRK(0x8384, 0x7680,
+ "Mac", STAC_INTEL_MAC_AUTO),
+ /* Dell systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a7,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a9,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ab,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ac,
+ "unknown Dell", STAC_922X_DELL_D82),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01bf,
+ "unknown Dell", STAC_922X_DELL_M81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d0,
+ "unknown Dell", STAC_922X_DELL_D82),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d1,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d2,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7,
+ "Dell XPS M1210", STAC_922X_DELL_M82),
+ /* ECS/PC Chips boards */
+ SND_PCI_QUIRK(0x1019, 0x2144,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2608,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2633,
+ "ECS/PC chips P17G/1333", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2811,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2812,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2813,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2814,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2815,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2816,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2817,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2818,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2819,
+ "ECS/PC chips", STAC_ECS_202),
+ SND_PCI_QUIRK(0x1019, 0x2820,
+ "ECS/PC chips", STAC_ECS_202),
+ {} /* terminator */
+};
+
+static unsigned int ref927x_pin_configs[14] = {
+ 0x02214020, 0x02a19080, 0x0181304e, 0x01014010,
+ 0x01a19040, 0x01011012, 0x01016011, 0x0101201f,
+ 0x183301f0, 0x18a001f0, 0x18a001f0, 0x01442070,
+ 0x01c42190, 0x40000100,
+};
+
+static unsigned int d965_3st_pin_configs[14] = {
+ 0x0221401f, 0x02a19120, 0x40000100, 0x01014011,
+ 0x01a19021, 0x01813024, 0x40000100, 0x40000100,
+ 0x40000100, 0x40000100, 0x40000100, 0x40000100,
+ 0x40000100, 0x40000100
+};
+
+static unsigned int d965_5st_pin_configs[14] = {
+ 0x02214020, 0x02a19080, 0x0181304e, 0x01014010,
+ 0x01a19040, 0x01011012, 0x01016011, 0x40000100,
+ 0x40000100, 0x40000100, 0x40000100, 0x01442070,
+ 0x40000100, 0x40000100
+};
+
+static unsigned int dell_3st_pin_configs[14] = {
+ 0x02211230, 0x02a11220, 0x01a19040, 0x01114210,
+ 0x01111212, 0x01116211, 0x01813050, 0x01112214,
+ 0x403003fa, 0x90a60040, 0x90a60040, 0x404003fb,
+ 0x40c003fc, 0x40000100
+};
+
+static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
+ [STAC_D965_REF_NO_JD] = ref927x_pin_configs,
+ [STAC_D965_REF] = ref927x_pin_configs,
+ [STAC_D965_3ST] = d965_3st_pin_configs,
+ [STAC_D965_5ST] = d965_5st_pin_configs,
+ [STAC_DELL_3ST] = dell_3st_pin_configs,
+ [STAC_DELL_BIOS] = NULL,
+};
+
+static const char *stac927x_models[STAC_927X_MODELS] = {
+ [STAC_D965_REF_NO_JD] = "ref-no-jd",
+ [STAC_D965_REF] = "ref",
+ [STAC_D965_3ST] = "3stack",
+ [STAC_D965_5ST] = "5stack",
+ [STAC_DELL_3ST] = "dell-3stack",
+ [STAC_DELL_BIOS] = "dell-bios",
+};
+
+static struct snd_pci_quirk stac927x_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ "DFI LanParty", STAC_D965_REF),
+ /* Intel 946 based systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST),
+ /* 965 based 3 stack systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2116, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2115, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2114, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2113, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2112, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2111, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2110, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2009, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2008, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2007, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2006, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2005, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2004, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST),
+ /* Dell 3 stack systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST),
+ /* Dell 3 stack systems with verb table in BIOS */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_BIOS),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS),
+ /* 965 based 5 stack systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2304, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2305, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2501, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2502, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2503, "Intel D965", STAC_D965_5ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2504, "Intel D965", STAC_D965_5ST),
+ {} /* terminator */
+};
+
+static unsigned int ref9205_pin_configs[12] = {
+ 0x40000100, 0x40000100, 0x01016011, 0x01014010,
+ 0x01813122, 0x01a19021, 0x01019020, 0x40000100,
+ 0x90a000f0, 0x90a000f0, 0x01441030, 0x01c41030
+};
+
+/*
+ STAC 9205 pin configs for
+ 102801F1
+ 102801F2
+ 102801FC
+ 102801FD
+ 10280204
+ 1028021F
+ 10280228 (Dell Vostro 1500)
+*/
+static unsigned int dell_9205_m42_pin_configs[12] = {
+ 0x0321101F, 0x03A11020, 0x400003FA, 0x90170310,
+ 0x400003FB, 0x400003FC, 0x400003FD, 0x40F000F9,
+ 0x90A60330, 0x400003FF, 0x0144131F, 0x40C003FE,
+};
+
+/*
+ STAC 9205 pin configs for
+ 102801F9
+ 102801FA
+ 102801FE
+ 102801FF (Dell Precision M4300)
+ 10280206
+ 10280200
+ 10280201
+*/
+static unsigned int dell_9205_m43_pin_configs[12] = {
+ 0x0321101f, 0x03a11020, 0x90a70330, 0x90170310,
+ 0x400000fe, 0x400000ff, 0x400000fd, 0x40f000f9,
+ 0x400000fa, 0x400000fc, 0x0144131f, 0x40c003f8,
+};
+
+static unsigned int dell_9205_m44_pin_configs[12] = {
+ 0x0421101f, 0x04a11020, 0x400003fa, 0x90170310,
+ 0x400003fb, 0x400003fc, 0x400003fd, 0x400003f9,
+ 0x90a60330, 0x400003ff, 0x01441340, 0x40c003fe,
+};
+
+static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
+ [STAC_9205_REF] = ref9205_pin_configs,
+ [STAC_9205_DELL_M42] = dell_9205_m42_pin_configs,
+ [STAC_9205_DELL_M43] = dell_9205_m43_pin_configs,
+ [STAC_9205_DELL_M44] = dell_9205_m44_pin_configs,
+};
+
+static const char *stac9205_models[STAC_9205_MODELS] = {
+ [STAC_9205_REF] = "ref",
+ [STAC_9205_DELL_M42] = "dell-m42",
+ [STAC_9205_DELL_M43] = "dell-m43",
+ [STAC_9205_DELL_M44] = "dell-m44",
+};
+
+static struct snd_pci_quirk stac9205_cfg_tbl[] = {
+ /* SigmaTel reference board */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
+ "DFI LanParty", STAC_9205_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f8,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fd,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fe,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ff,
+ "Dell Precision M4300", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0204,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f,
+ "Dell Inspiron", STAC_9205_DELL_M44),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
+ "Dell Vostro 1500", STAC_9205_DELL_M42),
+ {} /* terminator */
+};
+
+static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
+{
+ int i;
+ struct sigmatel_spec *spec = codec->spec;
+
+ if (! spec->bios_pin_configs) {
+ spec->bios_pin_configs = kcalloc(spec->num_pins,
+ sizeof(*spec->bios_pin_configs), GFP_KERNEL);
+ if (! spec->bios_pin_configs)
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < spec->num_pins; i++) {
+ hda_nid_t nid = spec->pin_nids[i];
+ unsigned int pin_cfg;
+
+ pin_cfg = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0x00);
+ snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n",
+ nid, pin_cfg);
+ spec->bios_pin_configs[i] = pin_cfg;
+ }
+
+ return 0;
+}
+
+static void stac92xx_set_config_reg(struct hda_codec *codec,
+ hda_nid_t pin_nid, unsigned int pin_config)
+{
+ int i;
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+ pin_config & 0x000000ff);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+ (pin_config & 0x0000ff00) >> 8);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+ (pin_config & 0x00ff0000) >> 16);
+ snd_hda_codec_write(codec, pin_nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+ pin_config >> 24);
+ i = snd_hda_codec_read(codec, pin_nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT,
+ 0x00);
+ snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n",
+ pin_nid, i);
+}
+
+static void stac92xx_set_config_regs(struct hda_codec *codec)
+{
+ int i;
+ struct sigmatel_spec *spec = codec->spec;
+
+ if (!spec->pin_configs)
+ return;
+
+ for (i = 0; i < spec->num_pins; i++)
+ stac92xx_set_config_reg(codec, spec->pin_nids[i],
+ spec->pin_configs[i]);
+}
+
+/*
+ * Analog playback callbacks
+ */
+static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ if (spec->stream_delay)
+ msleep(spec->stream_delay);
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream);
+}
+
+static int stac92xx_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital playback callbacks
+ */
+static int stac92xx_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int stac92xx_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
+
+
+/*
+ * Analog capture callbacks
+ */
+static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = spec->adc_nids[substream->number];
+
+ if (spec->powerdown_adcs) {
+ msleep(40);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ }
+ snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+ return 0;
+}
+
+static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = spec->adc_nids[substream->number];
+
+ snd_hda_codec_cleanup_stream(codec, nid);
+ if (spec->powerdown_adcs)
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ return 0;
+}
+
+static struct hda_pcm_stream stac92xx_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in stac92xx_build_pcms */
+ .ops = {
+ .open = stac92xx_dig_playback_pcm_open,
+ .close = stac92xx_dig_playback_pcm_close,
+ .prepare = stac92xx_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream stac92xx_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in stac92xx_build_pcms */
+};
+
+static struct hda_pcm_stream stac92xx_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 8,
+ .nid = 0x02, /* NID to query formats and rates */
+ .ops = {
+ .open = stac92xx_playback_pcm_open,
+ .prepare = stac92xx_playback_pcm_prepare,
+ .cleanup = stac92xx_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x06, /* NID to query formats and rates */
+ .ops = {
+ .open = stac92xx_playback_pcm_open,
+ .prepare = stac92xx_playback_pcm_prepare,
+ .cleanup = stac92xx_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream stac92xx_pcm_analog_capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID + .substreams is set in stac92xx_build_pcms */
+ .ops = {
+ .prepare = stac92xx_capture_pcm_prepare,
+ .cleanup = stac92xx_capture_pcm_cleanup
+ },
+};
+
+static int stac92xx_build_pcms(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "STAC92xx Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dac_nids[0];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs;
+
+ if (spec->alt_switch) {
+ codec->num_pcms++;
+ info++;
+ info->name = "STAC92xx Analog Alt";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_alt_playback;
+ }
+
+ if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+ codec->num_pcms++;
+ info++;
+ info->name = "STAC92xx Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->multiout.dig_out_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+ }
+ if (spec->dig_in_nid) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+ }
+ }
+
+ return 0;
+}
+
+static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int pincap = snd_hda_param_read(codec, nid,
+ AC_PAR_PIN_CAP);
+ pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+ if (pincap & AC_PINCAP_VREF_100)
+ return AC_PINCTL_VREF_100;
+ if (pincap & AC_PINCAP_VREF_80)
+ return AC_PINCTL_VREF_80;
+ if (pincap & AC_PINCAP_VREF_50)
+ return AC_PINCTL_VREF_50;
+ if (pincap & AC_PINCAP_VREF_GRD)
+ return AC_PINCTL_VREF_GRD;
+ return 0;
+}
+
+static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type)
+
+{
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+}
+
+#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.integer.value[0] = !!spec->hp_switch;
+ return 0;
+}
+
+static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ int nid = kcontrol->private_value;
+
+ spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0;
+
+ /* check to be sure that the ports are upto date with
+ * switch changes
+ */
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+
+ return 1;
+}
+
+#define stac92xx_io_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ int io_idx = kcontrol-> private_value & 0xff;
+
+ ucontrol->value.integer.value[0] = spec->io_switch[io_idx];
+ return 0;
+}
+
+static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = kcontrol->private_value >> 8;
+ int io_idx = kcontrol-> private_value & 0xff;
+ unsigned short val = !!ucontrol->value.integer.value[0];
+
+ spec->io_switch[io_idx] = val;
+
+ if (val)
+ stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+ else {
+ unsigned int pinctl = AC_PINCTL_IN_EN;
+ if (io_idx) /* set VREF for mic */
+ pinctl |= stac92xx_get_vref(codec, nid);
+ stac92xx_auto_set_pinctl(codec, nid, pinctl);
+ }
+
+ /* check the auto-mute again: we need to mute/unmute the speaker
+ * appropriately according to the pin direction
+ */
+ if (spec->hp_detect)
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+
+ return 1;
+}
+
+#define stac92xx_clfe_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_clfe_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.integer.value[0] = spec->clfe_swap;
+ return 0;
+}
+
+static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+ unsigned int val = !!ucontrol->value.integer.value[0];
+
+ if (spec->clfe_swap == val)
+ return 0;
+
+ spec->clfe_swap = val;
+
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ spec->clfe_swap ? 0x4 : 0x0);
+
+ return 1;
+}
+
+#define STAC_CODEC_HP_SWITCH(xname) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = 0, \
+ .info = stac92xx_hp_switch_info, \
+ .get = stac92xx_hp_switch_get, \
+ .put = stac92xx_hp_switch_put, \
+ }
+
+#define STAC_CODEC_IO_SWITCH(xname, xpval) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = 0, \
+ .info = stac92xx_io_switch_info, \
+ .get = stac92xx_io_switch_get, \
+ .put = stac92xx_io_switch_put, \
+ .private_value = xpval, \
+ }
+
+#define STAC_CODEC_CLFE_SWITCH(xname, xpval) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = 0, \
+ .info = stac92xx_clfe_switch_info, \
+ .get = stac92xx_clfe_switch_get, \
+ .put = stac92xx_clfe_switch_put, \
+ .private_value = xpval, \
+ }
+
+enum {
+ STAC_CTL_WIDGET_VOL,
+ STAC_CTL_WIDGET_MUTE,
+ STAC_CTL_WIDGET_MONO_MUX,
+ STAC_CTL_WIDGET_AMP_MUX,
+ STAC_CTL_WIDGET_AMP_VOL,
+ STAC_CTL_WIDGET_HP_SWITCH,
+ STAC_CTL_WIDGET_IO_SWITCH,
+ STAC_CTL_WIDGET_CLFE_SWITCH
+};
+
+static struct snd_kcontrol_new stac92xx_control_templates[] = {
+ HDA_CODEC_VOLUME(NULL, 0, 0, 0),
+ HDA_CODEC_MUTE(NULL, 0, 0, 0),
+ STAC_MONO_MUX,
+ STAC_AMP_MUX,
+ STAC_AMP_VOL(NULL, 0, 0, 0, 0),
+ STAC_CODEC_HP_SWITCH(NULL),
+ STAC_CODEC_IO_SWITCH(NULL, 0),
+ STAC_CODEC_CLFE_SWITCH(NULL, 0),
+};
+
+/* add dynamic controls */
+static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
+ struct snd_kcontrol_new *ktemp,
+ int idx, const char *name,
+ unsigned long val)
+{
+ struct snd_kcontrol_new *knew;
+
+ if (spec->num_kctl_used >= spec->num_kctl_alloc) {
+ int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
+
+ knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
+ if (! knew)
+ return -ENOMEM;
+ if (spec->kctl_alloc) {
+ memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
+ kfree(spec->kctl_alloc);
+ }
+ spec->kctl_alloc = knew;
+ spec->num_kctl_alloc = num;
+ }
+
+ knew = &spec->kctl_alloc[spec->num_kctl_used];
+ *knew = *ktemp;
+ knew->index = idx;
+ knew->name = kstrdup(name, GFP_KERNEL);
+ if (!knew->name)
+ return -ENOMEM;
+ knew->private_value = val;
+ spec->num_kctl_used++;
+ return 0;
+}
+
+static inline int stac92xx_add_control_idx(struct sigmatel_spec *spec,
+ int type, int idx, const char *name,
+ unsigned long val)
+{
+ return stac92xx_add_control_temp(spec,
+ &stac92xx_control_templates[type],
+ idx, name, val);
+}
+
+
+/* add dynamic controls */
+static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type,
+ const char *name, unsigned long val)
+{
+ return stac92xx_add_control_idx(spec, type, 0, name, val);
+}
+
+/* flag inputs as additional dynamic lineouts */
+static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int wcaps, wtype;
+ int i, num_dacs = 0;
+
+ /* use the wcaps cache to count all DACs available for line-outs */
+ for (i = 0; i < codec->num_nodes; i++) {
+ wcaps = codec->wcaps[i];
+ wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+
+ if (wtype == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL))
+ num_dacs++;
+ }
+
+ snd_printdd("%s: total dac count=%d\n", __func__, num_dacs);
+
+ switch (cfg->line_outs) {
+ case 3:
+ /* add line-in as side */
+ if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 3) {
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_LINE];
+ spec->line_switch = 1;
+ cfg->line_outs++;
+ }
+ break;
+ case 2:
+ /* add line-in as clfe and mic as side */
+ if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 2) {
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_LINE];
+ spec->line_switch = 1;
+ cfg->line_outs++;
+ }
+ if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 3) {
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_MIC];
+ spec->mic_switch = 1;
+ cfg->line_outs++;
+ }
+ break;
+ case 1:
+ /* add line-in as surr and mic as clfe */
+ if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 1) {
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_LINE];
+ spec->line_switch = 1;
+ cfg->line_outs++;
+ }
+ if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 2) {
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_MIC];
+ spec->mic_switch = 1;
+ cfg->line_outs++;
+ }
+ break;
+ }
+
+ return 0;
+}
+
+
+static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+ int i;
+
+ for (i = 0; i < spec->multiout.num_dacs; i++) {
+ if (spec->multiout.dac_nids[i] == nid)
+ return 1;
+ }
+
+ return 0;
+}
+
+/*
+ * Fill in the dac_nids table from the parsed pin configuration
+ * This function only works when every pin in line_out_pins[]
+ * contains atleast one DAC in its connection list. Some 92xx
+ * codecs are not connected directly to a DAC, such as the 9200
+ * and 9202/925x. For those, dac_nids[] must be hard-coded.
+ */
+static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i, j, conn_len = 0;
+ hda_nid_t nid, conn[HDA_MAX_CONNECTIONS];
+ unsigned int wcaps, wtype;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ conn_len = snd_hda_get_connections(codec, nid, conn,
+ HDA_MAX_CONNECTIONS);
+ for (j = 0; j < conn_len; j++) {
+ wcaps = snd_hda_param_read(codec, conn[j],
+ AC_PAR_AUDIO_WIDGET_CAP);
+ wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ if (wtype != AC_WID_AUD_OUT ||
+ (wcaps & AC_WCAP_DIGITAL))
+ continue;
+ /* conn[j] is a DAC routed to this line-out */
+ if (!is_in_dac_nids(spec, conn[j]))
+ break;
+ }
+
+ if (j == conn_len) {
+ if (spec->multiout.num_dacs > 0) {
+ /* we have already working output pins,
+ * so let's drop the broken ones again
+ */
+ cfg->line_outs = spec->multiout.num_dacs;
+ break;
+ }
+ /* error out, no available DAC found */
+ snd_printk(KERN_ERR
+ "%s: No available DAC for pin 0x%x\n",
+ __func__, nid);
+ return -ENODEV;
+ }
+
+ spec->multiout.dac_nids[i] = conn[j];
+ spec->multiout.num_dacs++;
+ if (conn_len > 1) {
+ /* select this DAC in the pin's input mux */
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, j);
+
+ }
+ }
+
+ snd_printd("dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ spec->multiout.num_dacs,
+ spec->multiout.dac_nids[0],
+ spec->multiout.dac_nids[1],
+ spec->multiout.dac_nids[2],
+ spec->multiout.dac_nids[3],
+ spec->multiout.dac_nids[4]);
+ return 0;
+}
+
+/* create volume control/switch for the given prefx type */
+static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs)
+{
+ char name[32];
+ int err;
+
+ sprintf(name, "%s Playback Volume", pfx);
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", pfx);
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+ if (!spec->multiout.hp_nid)
+ spec->multiout.hp_nid = nid;
+ else if (spec->multiout.num_dacs > 4) {
+ printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid);
+ return 1;
+ } else {
+ spec->multiout.dac_nids[spec->multiout.num_dacs] = nid;
+ spec->multiout.num_dacs++;
+ }
+ return 0;
+}
+
+static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+ if (is_in_dac_nids(spec, nid))
+ return 1;
+ if (spec->multiout.hp_nid == nid)
+ return 1;
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ static const char *chname[4] = {
+ "Front", "Surround", NULL /*CLFE*/, "Side"
+ };
+ hda_nid_t nid = 0;
+ int i, err;
+
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int wid_caps, pincap;
+
+
+ for (i = 0; i < cfg->line_outs && i < spec->multiout.num_dacs; i++) {
+ if (!spec->multiout.dac_nids[i])
+ continue;
+
+ nid = spec->multiout.dac_nids[i];
+
+ if (i == 2) {
+ /* Center/LFE */
+ err = create_controls(spec, "Center", nid, 1);
+ if (err < 0)
+ return err;
+ err = create_controls(spec, "LFE", nid, 2);
+ if (err < 0)
+ return err;
+
+ wid_caps = get_wcaps(codec, nid);
+
+ if (wid_caps & AC_WCAP_LR_SWAP) {
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_CLFE_SWITCH,
+ "Swap Center/LFE Playback Switch", nid);
+
+ if (err < 0)
+ return err;
+ }
+
+ } else {
+ err = create_controls(spec, chname[i], nid, 3);
+ if (err < 0)
+ return err;
+ }
+ }
+
+ if ((spec->multiout.num_dacs - cfg->line_outs) > 0 &&
+ cfg->hp_outs == 1 && !spec->multiout.hp_nid)
+ spec->multiout.hp_nid = nid;
+
+ if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) {
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_HP_SWITCH,
+ "Headphone as Line Out Switch",
+ cfg->hp_pins[cfg->hp_outs - 1]);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->line_switch) {
+ nid = cfg->input_pins[AUTO_PIN_LINE];
+ pincap = snd_hda_param_read(codec, nid,
+ AC_PAR_PIN_CAP);
+ if (pincap & AC_PINCAP_OUT) {
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_IO_SWITCH,
+ "Line In as Output Switch", nid << 8);
+ if (err < 0)
+ return err;
+ }
+ }
+
+ if (spec->mic_switch) {
+ unsigned int def_conf;
+ unsigned int mic_pin = AUTO_PIN_MIC;
+again:
+ nid = cfg->input_pins[mic_pin];
+ def_conf = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ /* some laptops have an internal analog microphone
+ * which can't be used as a output */
+ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) {
+ pincap = snd_hda_param_read(codec, nid,
+ AC_PAR_PIN_CAP);
+ if (pincap & AC_PINCAP_OUT) {
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_IO_SWITCH,
+ "Mic as Output Switch", (nid << 8) | 1);
+ nid = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+ if (!check_in_dac_nids(spec, nid))
+ add_spec_dacs(spec, nid);
+ if (err < 0)
+ return err;
+ }
+ } else if (mic_pin == AUTO_PIN_MIC) {
+ mic_pin = AUTO_PIN_FRONT_MIC;
+ goto again;
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls for Speaker and HP outputs */
+static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid;
+ int i, old_num_dacs, err;
+
+ old_num_dacs = spec->multiout.num_dacs;
+ for (i = 0; i < cfg->hp_outs; i++) {
+ unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]);
+ if (wid_caps & AC_WCAP_UNSOL_CAP)
+ spec->hp_detect = 1;
+ nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+ if (check_in_dac_nids(spec, nid))
+ nid = 0;
+ if (! nid)
+ continue;
+ add_spec_dacs(spec, nid);
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ nid = snd_hda_codec_read(codec, cfg->speaker_pins[i], 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+ if (check_in_dac_nids(spec, nid))
+ nid = 0;
+ if (! nid)
+ continue;
+ add_spec_dacs(spec, nid);
+ }
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+ if (check_in_dac_nids(spec, nid))
+ nid = 0;
+ if (! nid)
+ continue;
+ add_spec_dacs(spec, nid);
+ }
+ for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) {
+ static const char *pfxs[] = {
+ "Speaker", "External Speaker", "Speaker2",
+ };
+ err = create_controls(spec, pfxs[i - old_num_dacs],
+ spec->multiout.dac_nids[i], 3);
+ if (err < 0)
+ return err;
+ }
+ if (spec->multiout.hp_nid) {
+ err = create_controls(spec, "Headphone",
+ spec->multiout.hp_nid, 3);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* labels for mono mux outputs */
+static const char *stac92xx_mono_labels[4] = {
+ "DAC0", "DAC1", "Mixer", "DAC2"
+};
+
+/* create mono mux for mono out on capable codecs */
+static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_input_mux *mono_mux = &spec->private_mono_mux;
+ int i, num_cons;
+ hda_nid_t con_lst[ARRAY_SIZE(stac92xx_mono_labels)];
+
+ num_cons = snd_hda_get_connections(codec,
+ spec->mono_nid,
+ con_lst,
+ HDA_MAX_NUM_INPUTS);
+ if (!num_cons || num_cons > ARRAY_SIZE(stac92xx_mono_labels))
+ return -EINVAL;
+
+ for (i = 0; i < num_cons; i++) {
+ mono_mux->items[mono_mux->num_items].label =
+ stac92xx_mono_labels[i];
+ mono_mux->items[mono_mux->num_items].index = i;
+ mono_mux->num_items++;
+ }
+
+ return stac92xx_add_control(spec, STAC_CTL_WIDGET_MONO_MUX,
+ "Mono Mux", spec->mono_nid);
+}
+
+/* labels for amp mux outputs */
+static const char *stac92xx_amp_labels[3] = {
+ "Front Microphone", "Microphone", "Line In",
+};
+
+/* create amp out controls mux on capable codecs */
+static int stac92xx_auto_create_amp_output_ctls(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_input_mux *amp_mux = &spec->private_amp_mux;
+ int i, err;
+
+ for (i = 0; i < spec->num_amps; i++) {
+ amp_mux->items[amp_mux->num_items].label =
+ stac92xx_amp_labels[i];
+ amp_mux->items[amp_mux->num_items].index = i;
+ amp_mux->num_items++;
+ }
+
+ if (spec->num_amps > 1) {
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX,
+ "Amp Selector Capture Switch", 0);
+ if (err < 0)
+ return err;
+ }
+ return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL,
+ "Amp Capture Volume",
+ HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT));
+}
+
+
+/* create PC beep volume controls */
+static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
+ hda_nid_t nid)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+ int err;
+
+ /* check for mute support for the the amp */
+ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
+ "PC Beep Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+
+ /* check to see if there is volume support for the amp */
+ if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) {
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL,
+ "PC Beep Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+#define stac92xx_dig_beep_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_dig_beep_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = codec->beep->enabled;
+ return 0;
+}
+
+static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int enabled = !!ucontrol->value.integer.value[0];
+ if (codec->beep->enabled != enabled) {
+ codec->beep->enabled = enabled;
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = stac92xx_dig_beep_switch_info,
+ .get = stac92xx_dig_beep_switch_get,
+ .put = stac92xx_dig_beep_switch_put,
+};
+
+static int stac92xx_beep_switch_ctl(struct hda_codec *codec)
+{
+ return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl,
+ 0, "PC Beep Playback Switch", 0);
+}
+#endif
+
+static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int wcaps, nid, i, err = 0;
+
+ for (i = 0; i < spec->num_muxes; i++) {
+ nid = spec->mux_nids[i];
+ wcaps = get_wcaps(codec, nid);
+
+ if (wcaps & AC_WCAP_OUT_AMP) {
+ err = stac92xx_add_control_idx(spec,
+ STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+};
+
+static const char *stac92xx_spdif_labels[3] = {
+ "Digital Playback", "Analog Mux 1", "Analog Mux 2",
+};
+
+static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_input_mux *spdif_mux = &spec->private_smux;
+ const char **labels = spec->spdif_labels;
+ int i, num_cons;
+ hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
+
+ num_cons = snd_hda_get_connections(codec,
+ spec->smux_nids[0],
+ con_lst,
+ HDA_MAX_NUM_INPUTS);
+ if (!num_cons)
+ return -EINVAL;
+
+ if (!labels)
+ labels = stac92xx_spdif_labels;
+
+ for (i = 0; i < num_cons; i++) {
+ spdif_mux->items[spdif_mux->num_items].label = labels[i];
+ spdif_mux->items[spdif_mux->num_items].index = i;
+ spdif_mux->num_items++;
+ }
+
+ return 0;
+}
+
+/* labels for dmic mux inputs */
+static const char *stac92xx_dmic_labels[5] = {
+ "Analog Inputs", "Digital Mic 1", "Digital Mic 2",
+ "Digital Mic 3", "Digital Mic 4"
+};
+
+/* create playback/capture controls for input pins on dmic capable codecs */
+static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_input_mux *dimux = &spec->private_dimux;
+ hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
+ int err, i, j;
+ char name[32];
+
+ dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0];
+ dimux->items[dimux->num_items].index = 0;
+ dimux->num_items++;
+
+ for (i = 0; i < spec->num_dmics; i++) {
+ hda_nid_t nid;
+ int index;
+ int num_cons;
+ unsigned int wcaps;
+ unsigned int def_conf;
+
+ def_conf = snd_hda_codec_read(codec,
+ spec->dmic_nids[i],
+ 0,
+ AC_VERB_GET_CONFIG_DEFAULT,
+ 0);
+ if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+ continue;
+
+ nid = spec->dmic_nids[i];
+ num_cons = snd_hda_get_connections(codec,
+ spec->dmux_nids[0],
+ con_lst,
+ HDA_MAX_NUM_INPUTS);
+ for (j = 0; j < num_cons; j++)
+ if (con_lst[j] == nid) {
+ index = j;
+ goto found;
+ }
+ continue;
+found:
+ wcaps = get_wcaps(codec, nid) &
+ (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
+
+ if (wcaps) {
+ sprintf(name, "%s Capture Volume",
+ stac92xx_dmic_labels[dimux->num_items]);
+
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_VOL,
+ name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ (wcaps & AC_WCAP_OUT_AMP) ?
+ HDA_OUTPUT : HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+
+ dimux->items[dimux->num_items].label =
+ stac92xx_dmic_labels[dimux->num_items];
+ dimux->items[dimux->num_items].index = index;
+ dimux->num_items++;
+ }
+
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_input_mux *imux = &spec->private_imux;
+ hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
+ int i, j, k;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ int index;
+
+ if (!cfg->input_pins[i])
+ continue;
+ index = -1;
+ for (j = 0; j < spec->num_muxes; j++) {
+ int num_cons;
+ num_cons = snd_hda_get_connections(codec,
+ spec->mux_nids[j],
+ con_lst,
+ HDA_MAX_NUM_INPUTS);
+ for (k = 0; k < num_cons; k++)
+ if (con_lst[k] == cfg->input_pins[i]) {
+ index = k;
+ goto found;
+ }
+ }
+ continue;
+ found:
+ imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
+ imux->items[imux->num_items].index = index;
+ imux->num_items++;
+ }
+
+ if (imux->num_items) {
+ /*
+ * Set the current input for the muxes.
+ * The STAC9221 has two input muxes with identical source
+ * NID lists. Hopefully this won't get confused.
+ */
+ for (i = 0; i < spec->num_muxes; i++) {
+ snd_hda_codec_write_cache(codec, spec->mux_nids[i], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ imux->items[0].index);
+ }
+ }
+
+ return 0;
+}
+
+static void stac92xx_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->autocfg.line_outs; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+ }
+}
+
+static void stac92xx_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->autocfg.hp_outs; i++) {
+ hda_nid_t pin;
+ pin = spec->autocfg.hp_pins[i];
+ if (pin) /* connect to front */
+ stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ }
+ for (i = 0; i < spec->autocfg.speaker_outs; i++) {
+ hda_nid_t pin;
+ pin = spec->autocfg.speaker_pins[i];
+ if (pin) /* connect to front */
+ stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN);
+ }
+}
+
+static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int err;
+ int hp_speaker_swap = 0;
+
+ if ((err = snd_hda_parse_pin_def_config(codec,
+ &spec->autocfg,
+ spec->dmic_nids)) < 0)
+ return err;
+ if (! spec->autocfg.line_outs)
+ return 0; /* can't find valid pin config */
+
+ /* If we have no real line-out pin and multiple hp-outs, HPs should
+ * be set up as multi-channel outputs.
+ */
+ if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT &&
+ spec->autocfg.hp_outs > 1) {
+ /* Copy hp_outs to line_outs, backup line_outs in
+ * speaker_outs so that the following routines can handle
+ * HP pins as primary outputs.
+ */
+ memcpy(spec->autocfg.speaker_pins, spec->autocfg.line_out_pins,
+ sizeof(spec->autocfg.line_out_pins));
+ spec->autocfg.speaker_outs = spec->autocfg.line_outs;
+ memcpy(spec->autocfg.line_out_pins, spec->autocfg.hp_pins,
+ sizeof(spec->autocfg.hp_pins));
+ spec->autocfg.line_outs = spec->autocfg.hp_outs;
+ hp_speaker_swap = 1;
+ }
+ if (spec->autocfg.mono_out_pin) {
+ int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
+ (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
+ u32 caps = query_amp_caps(codec,
+ spec->autocfg.mono_out_pin, dir);
+ hda_nid_t conn_list[1];
+
+ /* get the mixer node and then the mono mux if it exists */
+ if (snd_hda_get_connections(codec,
+ spec->autocfg.mono_out_pin, conn_list, 1) &&
+ snd_hda_get_connections(codec, conn_list[0],
+ conn_list, 1)) {
+
+ int wcaps = get_wcaps(codec, conn_list[0]);
+ int wid_type = (wcaps & AC_WCAP_TYPE)
+ >> AC_WCAP_TYPE_SHIFT;
+ /* LR swap check, some stac925x have a mux that
+ * changes the DACs output path instead of the
+ * mono-mux path.
+ */
+ if (wid_type == AC_WID_AUD_SEL &&
+ !(wcaps & AC_WCAP_LR_SWAP))
+ spec->mono_nid = conn_list[0];
+ }
+ if (dir) {
+ hda_nid_t nid = spec->autocfg.mono_out_pin;
+
+ /* most mono outs have a least a mute/unmute switch */
+ dir = (dir & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT;
+ err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
+ "Mono Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir));
+ if (err < 0)
+ return err;
+ /* check for volume support for the amp */
+ if ((caps & AC_AMPCAP_NUM_STEPS)
+ >> AC_AMPCAP_NUM_STEPS_SHIFT) {
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_VOL,
+ "Mono Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin,
+ AC_PINCTL_OUT_EN);
+ }
+
+ if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0)
+ return err;
+ if (spec->multiout.num_dacs == 0)
+ if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
+ return err;
+
+ err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg);
+
+ if (err < 0)
+ return err;
+
+ /* setup analog beep controls */
+ if (spec->anabeep_nid > 0) {
+ err = stac92xx_auto_create_beep_ctls(codec,
+ spec->anabeep_nid);
+ if (err < 0)
+ return err;
+ }
+
+ /* setup digital beep controls and input device */
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+ if (spec->digbeep_nid > 0) {
+ hda_nid_t nid = spec->digbeep_nid;
+ unsigned int caps;
+
+ err = stac92xx_auto_create_beep_ctls(codec, nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_attach_beep_device(codec, nid);
+ if (err < 0)
+ return err;
+ /* if no beep switch is available, make its own one */
+ caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+ if (codec->beep &&
+ !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) {
+ err = stac92xx_beep_switch_ctl(codec);
+ if (err < 0)
+ return err;
+ }
+ }
+#endif
+
+ if (hp_speaker_swap == 1) {
+ /* Restore the hp_outs and line_outs */
+ memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins,
+ sizeof(spec->autocfg.line_out_pins));
+ spec->autocfg.hp_outs = spec->autocfg.line_outs;
+ memcpy(spec->autocfg.line_out_pins, spec->autocfg.speaker_pins,
+ sizeof(spec->autocfg.speaker_pins));
+ spec->autocfg.line_outs = spec->autocfg.speaker_outs;
+ memset(spec->autocfg.speaker_pins, 0,
+ sizeof(spec->autocfg.speaker_pins));
+ spec->autocfg.speaker_outs = 0;
+ }
+
+ err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg);
+
+ if (err < 0)
+ return err;
+
+ err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
+
+ if (err < 0)
+ return err;
+
+ if (spec->mono_nid > 0) {
+ err = stac92xx_auto_create_mono_output_ctls(codec);
+ if (err < 0)
+ return err;
+ }
+ if (spec->num_amps > 0) {
+ err = stac92xx_auto_create_amp_output_ctls(codec);
+ if (err < 0)
+ return err;
+ }
+ if (spec->num_dmics > 0 && !spec->dinput_mux)
+ if ((err = stac92xx_auto_create_dmic_input_ctls(codec,
+ &spec->autocfg)) < 0)
+ return err;
+ if (spec->num_muxes > 0) {
+ err = stac92xx_auto_create_mux_input_ctls(codec);
+ if (err < 0)
+ return err;
+ }
+ if (spec->num_smuxes > 0) {
+ err = stac92xx_auto_create_spdif_mux_ctls(codec);
+ if (err < 0)
+ return err;
+ }
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+ if (spec->multiout.max_channels > 2)
+ spec->surr_switch = 1;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = dig_out;
+ if (dig_in && spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = dig_in;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->input_mux = &spec->private_imux;
+ spec->dinput_mux = &spec->private_dimux;
+ spec->sinput_mux = &spec->private_smux;
+ spec->mono_mux = &spec->private_mono_mux;
+ spec->amp_mux = &spec->private_amp_mux;
+ return 1;
+}
+
+/* add playback controls for HP output */
+static int stac9200_auto_create_hp_ctls(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t pin = cfg->hp_pins[0];
+ unsigned int wid_caps;
+
+ if (! pin)
+ return 0;
+
+ wid_caps = get_wcaps(codec, pin);
+ if (wid_caps & AC_WCAP_UNSOL_CAP)
+ spec->hp_detect = 1;
+
+ return 0;
+}
+
+/* add playback controls for LFE output */
+static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int err;
+ hda_nid_t lfe_pin = 0x0;
+ int i;
+
+ /*
+ * search speaker outs and line outs for a mono speaker pin
+ * with an amp. If one is found, add LFE controls
+ * for it.
+ */
+ for (i = 0; i < spec->autocfg.speaker_outs && lfe_pin == 0x0; i++) {
+ hda_nid_t pin = spec->autocfg.speaker_pins[i];
+ unsigned int wcaps = get_wcaps(codec, pin);
+ wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP);
+ if (wcaps == AC_WCAP_OUT_AMP)
+ /* found a mono speaker with an amp, must be lfe */
+ lfe_pin = pin;
+ }
+
+ /* if speaker_outs is 0, then speakers may be in line_outs */
+ if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) {
+ for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) {
+ hda_nid_t pin = spec->autocfg.line_out_pins[i];
+ unsigned int defcfg;
+ defcfg = snd_hda_codec_read(codec, pin, 0,
+ AC_VERB_GET_CONFIG_DEFAULT,
+ 0x00);
+ if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) {
+ unsigned int wcaps = get_wcaps(codec, pin);
+ wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP);
+ if (wcaps == AC_WCAP_OUT_AMP)
+ /* found a mono speaker with an amp,
+ must be lfe */
+ lfe_pin = pin;
+ }
+ }
+ }
+
+ if (lfe_pin) {
+ err = create_controls(spec, "LFE", lfe_pin, 1);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static int stac9200_parse_auto_config(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int err;
+
+ if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0)
+ return err;
+
+ if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0)
+ return err;
+
+ if ((err = stac9200_auto_create_hp_ctls(codec, &spec->autocfg)) < 0)
+ return err;
+
+ if ((err = stac9200_auto_create_lfe_ctls(codec, &spec->autocfg)) < 0)
+ return err;
+
+ if (spec->num_muxes > 0) {
+ err = stac92xx_auto_create_mux_input_ctls(codec);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = 0x05;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = 0x04;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->input_mux = &spec->private_imux;
+ spec->dinput_mux = &spec->private_dimux;
+
+ return 1;
+}
+
+/*
+ * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a
+ * funky external mute control using GPIO pins.
+ */
+
+static void stac_gpio_set(struct hda_codec *codec, unsigned int mask,
+ unsigned int dir_mask, unsigned int data)
+{
+ unsigned int gpiostate, gpiomask, gpiodir;
+
+ gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DATA, 0);
+ gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask);
+
+ gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_MASK, 0);
+ gpiomask |= mask;
+
+ gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DIRECTION, 0);
+ gpiodir |= dir_mask;
+
+ /* Configure GPIOx as CMOS */
+ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0);
+
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_MASK, gpiomask);
+ snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */
+
+ msleep(1);
+
+ snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */
+}
+
+static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int event)
+{
+ if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ (AC_USRSP_EN | event));
+}
+
+static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
+{
+ int i;
+ for (i = 0; i < cfg->hp_outs; i++)
+ if (cfg->hp_pins[i] == nid)
+ return 1; /* nid is a HP-Out */
+
+ return 0; /* nid is not a HP-Out */
+};
+
+static void stac92xx_power_down(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+
+ /* power down inactive DACs */
+ hda_nid_t *dac;
+ for (dac = spec->dac_list; *dac; dac++)
+ if (!is_in_dac_nids(spec, *dac) &&
+ spec->multiout.hp_nid != *dac)
+ snd_hda_codec_write_cache(codec, *dac, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+}
+
+static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
+ int enable);
+
+static int stac92xx_init(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int gpio;
+ int i;
+
+ snd_hda_sequence_write(codec, spec->init);
+
+ /* power down adcs initially */
+ if (spec->powerdown_adcs)
+ for (i = 0; i < spec->num_adcs; i++)
+ snd_hda_codec_write_cache(codec,
+ spec->adc_nids[i], 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+
+ /* set up GPIO */
+ gpio = spec->gpio_data;
+ /* turn on EAPD statically when spec->eapd_switch isn't set.
+ * otherwise, unsol event will turn it on/off dynamically
+ */
+ if (!spec->eapd_switch)
+ gpio |= spec->eapd_mask;
+ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio);
+
+ /* set up pins */
+ if (spec->hp_detect) {
+ /* Enable unsolicited responses on the HP widget */
+ for (i = 0; i < cfg->hp_outs; i++)
+ enable_pin_detect(codec, cfg->hp_pins[i],
+ STAC_HP_EVENT);
+ /* force to enable the first line-out; the others are set up
+ * in unsol_event
+ */
+ stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0],
+ AC_PINCTL_OUT_EN);
+ stac92xx_auto_init_hp_out(codec);
+ /* fake event to set up pins */
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+ } else {
+ stac92xx_auto_init_multi_out(codec);
+ stac92xx_auto_init_hp_out(codec);
+ }
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = cfg->input_pins[i];
+ if (nid) {
+ unsigned int pinctl;
+ if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) {
+ /* for mic pins, force to initialize */
+ pinctl = stac92xx_get_vref(codec, nid);
+ } else {
+ pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ /* if PINCTL already set then skip */
+ if (pinctl & AC_PINCTL_IN_EN)
+ continue;
+ }
+ pinctl |= AC_PINCTL_IN_EN;
+ stac92xx_auto_set_pinctl(codec, nid, pinctl);
+ }
+ }
+ for (i = 0; i < spec->num_dmics; i++)
+ stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i],
+ AC_PINCTL_IN_EN);
+ if (cfg->dig_out_pin)
+ stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin,
+ AC_PINCTL_OUT_EN);
+ if (cfg->dig_in_pin)
+ stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin,
+ AC_PINCTL_IN_EN);
+ for (i = 0; i < spec->num_pwrs; i++) {
+ hda_nid_t nid = spec->pwr_nids[i];
+ int pinctl, def_conf;
+ int event = STAC_PWR_EVENT;
+
+ if (is_nid_hp_pin(cfg, nid) && spec->hp_detect)
+ continue; /* already has an unsol event */
+
+ pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ /* outputs are only ports capable of power management
+ * any attempts on powering down a input port cause the
+ * referenced VREF to act quirky.
+ */
+ if (pinctl & AC_PINCTL_IN_EN)
+ continue;
+ def_conf = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ def_conf = get_defcfg_connect(def_conf);
+ /* skip any ports that don't have jacks since presence
+ * detection is useless */
+ if (def_conf != AC_JACK_PORT_COMPLEX) {
+ if (def_conf != AC_JACK_PORT_NONE)
+ stac_toggle_power_map(codec, nid, 1);
+ continue;
+ }
+ enable_pin_detect(codec, spec->pwr_nids[i], event | i);
+ codec->patch_ops.unsol_event(codec, (event | i) << 26);
+ }
+ if (spec->dac_list)
+ stac92xx_power_down(codec);
+ return 0;
+}
+
+static void stac92xx_free(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i;
+
+ if (! spec)
+ return;
+
+ if (spec->kctl_alloc) {
+ for (i = 0; i < spec->num_kctl_used; i++)
+ kfree(spec->kctl_alloc[i].name);
+ kfree(spec->kctl_alloc);
+ }
+
+ if (spec->bios_pin_configs)
+ kfree(spec->bios_pin_configs);
+
+ kfree(spec);
+ snd_hda_detach_beep_device(codec);
+}
+
+static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int flag)
+{
+ unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
+ 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
+
+ if (pin_ctl & AC_PINCTL_IN_EN) {
+ /*
+ * we need to check the current set-up direction of
+ * shared input pins since they can be switched via
+ * "xxx as Output" mixer switch
+ */
+ struct sigmatel_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ if ((nid == cfg->input_pins[AUTO_PIN_LINE] &&
+ spec->line_switch) ||
+ (nid == cfg->input_pins[AUTO_PIN_MIC] &&
+ spec->mic_switch))
+ return;
+ }
+
+ /* if setting pin direction bits, clear the current
+ direction bits first */
+ if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))
+ pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
+
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pin_ctl | flag);
+}
+
+static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int flag)
+{
+ unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
+ 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pin_ctl & ~flag);
+}
+
+static int get_hp_pin_presence(struct hda_codec *codec, hda_nid_t nid)
+{
+ if (!nid)
+ return 0;
+ if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00)
+ & (1 << 31)) {
+ unsigned int pinctl;
+ pinctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (pinctl & AC_PINCTL_IN_EN)
+ return 0; /* mic- or line-input */
+ else
+ return 1; /* HP-output */
+ }
+ return 0;
+}
+
+/* return non-zero if the hp-pin of the given array index isn't
+ * a jack-detection target
+ */
+static int no_hp_sensing(struct sigmatel_spec *spec, int i)
+{
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+
+ /* ignore sensing of shared line and mic jacks */
+ if (spec->line_switch &&
+ cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_LINE])
+ return 1;
+ if (spec->mic_switch &&
+ cfg->hp_pins[i] == cfg->input_pins[AUTO_PIN_MIC])
+ return 1;
+ /* ignore if the pin is set as line-out */
+ if (cfg->hp_pins[i] == spec->hp_switch)
+ return 1;
+ return 0;
+}
+
+static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, presence;
+
+ presence = 0;
+ if (spec->gpio_mute)
+ presence = !(snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute);
+
+ for (i = 0; i < cfg->hp_outs; i++) {
+ if (presence)
+ break;
+ if (no_hp_sensing(spec, i))
+ continue;
+ presence = get_hp_pin_presence(codec, cfg->hp_pins[i]);
+ }
+
+ if (presence) {
+ /* disable lineouts */
+ if (spec->hp_switch)
+ stac92xx_reset_pinctl(codec, spec->hp_switch,
+ AC_PINCTL_OUT_EN);
+ for (i = 0; i < cfg->line_outs; i++)
+ stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
+ AC_PINCTL_OUT_EN);
+ for (i = 0; i < cfg->speaker_outs; i++)
+ stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
+ AC_PINCTL_OUT_EN);
+ if (spec->eapd_mask && spec->eapd_switch)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data &
+ ~spec->eapd_mask);
+ } else {
+ /* enable lineouts */
+ if (spec->hp_switch)
+ stac92xx_set_pinctl(codec, spec->hp_switch,
+ AC_PINCTL_OUT_EN);
+ for (i = 0; i < cfg->line_outs; i++)
+ stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
+ AC_PINCTL_OUT_EN);
+ for (i = 0; i < cfg->speaker_outs; i++)
+ stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
+ AC_PINCTL_OUT_EN);
+ if (spec->eapd_mask && spec->eapd_switch)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data |
+ spec->eapd_mask);
+ }
+ /* toggle hp outs */
+ for (i = 0; i < cfg->hp_outs; i++) {
+ unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
+ if (no_hp_sensing(spec, i))
+ continue;
+ if (presence)
+ stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
+#if 0 /* FIXME */
+/* Resetting the pinctl like below may lead to (a sort of) regressions
+ * on some devices since they use the HP pin actually for line/speaker
+ * outs although the default pin config shows a different pin (that is
+ * wrong and useless).
+ *
+ * So, it's basically a problem of default pin configs, likely a BIOS issue.
+ * But, disabling the code below just works around it, and I'm too tired of
+ * bug reports with such devices...
+ */
+ else
+ stac92xx_reset_pinctl(codec, cfg->hp_pins[i], val);
+#endif /* FIXME */
+ }
+}
+
+static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
+ int enable)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int idx, val;
+
+ for (idx = 0; idx < spec->num_pwrs; idx++) {
+ if (spec->pwr_nids[idx] == nid)
+ break;
+ }
+ if (idx >= spec->num_pwrs)
+ return;
+
+ /* several codecs have two power down bits */
+ if (spec->pwr_mapping)
+ idx = spec->pwr_mapping[idx];
+ else
+ idx = 1 << idx;
+
+ val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff;
+ if (enable)
+ val &= ~idx;
+ else
+ val |= idx;
+
+ /* power down unused output ports */
+ snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val);
+}
+
+static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid)
+{
+ stac_toggle_power_map(codec, nid, get_hp_pin_presence(codec, nid));
+}
+
+static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int idx = res >> 26 & 0x0f;
+
+ switch ((res >> 26) & 0x70) {
+ case STAC_HP_EVENT:
+ stac92xx_hp_detect(codec, res);
+ /* fallthru */
+ case STAC_PWR_EVENT:
+ if (spec->num_pwrs > 0)
+ stac92xx_pin_sense(codec, idx);
+ break;
+ case STAC_VREF_EVENT: {
+ int data = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DATA, 0);
+ /* toggle VREF state based on GPIOx status */
+ snd_hda_codec_write(codec, codec->afg, 0, 0x7e0,
+ !!(data & (1 << idx)));
+ break;
+ }
+ }
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+static int stac92xx_resume(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+
+ stac92xx_set_config_regs(codec);
+ snd_hda_sequence_write(codec, spec->init);
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ /* power down inactive DACs */
+ if (spec->dac_list)
+ stac92xx_power_down(codec);
+ /* invoke unsolicited event to reset the HP state */
+ if (spec->hp_detect)
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+ return 0;
+}
+#endif
+
+static struct hda_codec_ops stac92xx_patch_ops = {
+ .build_controls = stac92xx_build_controls,
+ .build_pcms = stac92xx_build_pcms,
+ .init = stac92xx_init,
+ .free = stac92xx_free,
+ .unsol_event = stac92xx_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+ .resume = stac92xx_resume,
+#endif
+};
+
+static int patch_stac9200(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac9200_pin_nids);
+ spec->pin_nids = stac9200_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS,
+ stac9200_models,
+ stac9200_cfg_tbl);
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
+ spec->pin_configs = stac9200_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = stac9200_dac_nids;
+ spec->adc_nids = stac9200_adc_nids;
+ spec->mux_nids = stac9200_mux_nids;
+ spec->num_muxes = 1;
+ spec->num_dmics = 0;
+ spec->num_adcs = 1;
+ spec->num_pwrs = 0;
+
+ if (spec->board_config == STAC_9200_GATEWAY ||
+ spec->board_config == STAC_9200_OQO)
+ spec->init = stac9200_eapd_init;
+ else
+ spec->init = stac9200_core_init;
+ spec->mixer = stac9200_mixer;
+
+ if (spec->board_config == STAC_9200_PANASONIC) {
+ spec->gpio_mask = spec->gpio_dir = 0x09;
+ spec->gpio_data = 0x00;
+ }
+
+ err = stac9200_parse_auto_config(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+static int patch_stac925x(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac925x_pin_nids);
+ spec->pin_nids = stac925x_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec, STAC_925x_MODELS,
+ stac925x_models,
+ stac925x_cfg_tbl);
+ again:
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x,"
+ "using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else if (stac925x_brd_tbl[spec->board_config] != NULL){
+ spec->pin_configs = stac925x_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = stac925x_dac_nids;
+ spec->adc_nids = stac925x_adc_nids;
+ spec->mux_nids = stac925x_mux_nids;
+ spec->num_muxes = 1;
+ spec->num_adcs = 1;
+ spec->num_pwrs = 0;
+ switch (codec->vendor_id) {
+ case 0x83847632: /* STAC9202 */
+ case 0x83847633: /* STAC9202D */
+ case 0x83847636: /* STAC9251 */
+ case 0x83847637: /* STAC9251D */
+ spec->num_dmics = STAC925X_NUM_DMICS;
+ spec->dmic_nids = stac925x_dmic_nids;
+ spec->num_dmuxes = ARRAY_SIZE(stac925x_dmux_nids);
+ spec->dmux_nids = stac925x_dmux_nids;
+ break;
+ default:
+ spec->num_dmics = 0;
+ break;
+ }
+
+ spec->init = stac925x_core_init;
+ spec->mixer = stac925x_mixer;
+
+ err = stac92xx_parse_auto_config(codec, 0x8, 0x7);
+ if (!err) {
+ if (spec->board_config < 0) {
+ printk(KERN_WARNING "hda_codec: No auto-config is "
+ "available, default to model=ref\n");
+ spec->board_config = STAC_925x_REF;
+ goto again;
+ }
+ err = -EINVAL;
+ }
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+static struct hda_input_mux stac92hd73xx_dmux = {
+ .num_items = 4,
+ .items = {
+ { "Analog Inputs", 0x0b },
+ { "Digital Mic 1", 0x09 },
+ { "Digital Mic 2", 0x0a },
+ { "CD", 0x08 },
+ }
+};
+
+static int patch_stac92hd73xx(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ hda_nid_t conn[STAC92HD73_DAC_COUNT + 2];
+ int err = 0;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ codec->slave_dig_outs = stac92hd73xx_slave_dig_outs;
+ spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids);
+ spec->pin_nids = stac92hd73xx_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec,
+ STAC_92HD73XX_MODELS,
+ stac92hd73xx_models,
+ stac92hd73xx_cfg_tbl);
+again:
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+ " STAC92HD73XX, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
+ spec->pin_configs = stac92hd73xx_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ spec->multiout.num_dacs = snd_hda_get_connections(codec, 0x0a,
+ conn, STAC92HD73_DAC_COUNT + 2) - 1;
+
+ if (spec->multiout.num_dacs < 0) {
+ printk(KERN_WARNING "hda_codec: Could not determine "
+ "number of channels defaulting to DAC count\n");
+ spec->multiout.num_dacs = STAC92HD73_DAC_COUNT;
+ }
+
+ switch (spec->multiout.num_dacs) {
+ case 0x3: /* 6 Channel */
+ spec->multiout.hp_nid = 0x17;
+ spec->mixer = stac92hd73xx_6ch_mixer;
+ spec->init = stac92hd73xx_6ch_core_init;
+ break;
+ case 0x4: /* 8 Channel */
+ spec->multiout.hp_nid = 0x18;
+ spec->mixer = stac92hd73xx_8ch_mixer;
+ spec->init = stac92hd73xx_8ch_core_init;
+ break;
+ case 0x5: /* 10 Channel */
+ spec->multiout.hp_nid = 0x19;
+ spec->mixer = stac92hd73xx_10ch_mixer;
+ spec->init = stac92hd73xx_10ch_core_init;
+ };
+
+ spec->multiout.dac_nids = stac92hd73xx_dac_nids;
+ spec->aloopback_mask = 0x01;
+ spec->aloopback_shift = 8;
+
+ spec->digbeep_nid = 0x1c;
+ spec->mux_nids = stac92hd73xx_mux_nids;
+ spec->adc_nids = stac92hd73xx_adc_nids;
+ spec->dmic_nids = stac92hd73xx_dmic_nids;
+ spec->dmux_nids = stac92hd73xx_dmux_nids;
+ spec->smux_nids = stac92hd73xx_smux_nids;
+ spec->amp_nids = stac92hd73xx_amp_nids;
+ spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids);
+
+ spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids);
+ spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids);
+ memcpy(&spec->private_dimux, &stac92hd73xx_dmux,
+ sizeof(stac92hd73xx_dmux));
+
+ switch (spec->board_config) {
+ case STAC_DELL_EQ:
+ spec->init = dell_eq_core_init;
+ /* fallthru */
+ case STAC_DELL_M6_AMIC:
+ case STAC_DELL_M6_DMIC:
+ case STAC_DELL_M6_BOTH:
+ spec->num_smuxes = 0;
+ spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
+ spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
+ spec->eapd_switch = 0;
+ spec->num_amps = 1;
+ spec->multiout.hp_nid = 0; /* dual HPs */
+
+ if (!spec->init)
+ spec->init = dell_m6_core_init;
+ switch (spec->board_config) {
+ case STAC_DELL_M6_AMIC: /* Analog Mics */
+ stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
+ spec->num_dmics = 0;
+ spec->private_dimux.num_items = 1;
+ break;
+ case STAC_DELL_M6_DMIC: /* Digital Mics */
+ stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+ spec->num_dmics = 1;
+ spec->private_dimux.num_items = 2;
+ break;
+ case STAC_DELL_M6_BOTH: /* Both */
+ stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
+ stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+ spec->num_dmics = 1;
+ spec->private_dimux.num_items = 2;
+ break;
+ }
+ break;
+ default:
+ spec->num_dmics = STAC92HD73XX_NUM_DMICS;
+ spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
+ spec->eapd_switch = 1;
+ }
+ if (spec->board_config > STAC_92HD73XX_REF) {
+ /* GPIO0 High = Enable EAPD */
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
+ spec->gpio_data = 0x01;
+ }
+ spec->dinput_mux = &spec->private_dimux;
+
+ spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids);
+ spec->pwr_nids = stac92hd73xx_pwr_nids;
+
+ err = stac92xx_parse_auto_config(codec, 0x25, 0x27);
+
+ if (!err) {
+ if (spec->board_config < 0) {
+ printk(KERN_WARNING "hda_codec: No auto-config is "
+ "available, default to model=ref\n");
+ spec->board_config = STAC_92HD73XX_REF;
+ goto again;
+ }
+ err = -EINVAL;
+ }
+
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ if (spec->board_config == STAC_92HD73XX_NO_JD)
+ spec->hp_detect = 0;
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+static struct hda_input_mux stac92hd83xxx_dmux = {
+ .num_items = 3,
+ .items = {
+ { "Analog Inputs", 0x03 },
+ { "Digital Mic 1", 0x04 },
+ { "Digital Mic 2", 0x05 },
+ }
+};
+
+static int patch_stac92hd83xxx(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
+ spec->mono_nid = 0x19;
+ spec->digbeep_nid = 0x21;
+ spec->dmic_nids = stac92hd83xxx_dmic_nids;
+ spec->dmux_nids = stac92hd83xxx_dmux_nids;
+ spec->adc_nids = stac92hd83xxx_adc_nids;
+ spec->pwr_nids = stac92hd83xxx_pwr_nids;
+ spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
+ spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
+ spec->multiout.dac_nids = stac92hd83xxx_dac_nids;
+
+ spec->init = stac92hd83xxx_core_init;
+ switch (codec->vendor_id) {
+ case 0x111d7605:
+ spec->multiout.num_dacs = STAC92HD81_DAC_COUNT;
+ break;
+ default:
+ spec->num_pwrs--;
+ spec->init++; /* switch to config #2 */
+ spec->multiout.num_dacs = STAC92HD83_DAC_COUNT;
+ }
+
+ spec->mixer = stac92hd83xxx_mixer;
+ spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
+ spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids);
+ spec->num_dmics = STAC92HD83XXX_NUM_DMICS;
+ spec->dinput_mux = &stac92hd83xxx_dmux;
+ spec->pin_nids = stac92hd83xxx_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec,
+ STAC_92HD83XXX_MODELS,
+ stac92hd83xxx_models,
+ stac92hd83xxx_cfg_tbl);
+again:
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+ " STAC92HD83XXX, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
+ spec->pin_configs = stac92hd83xxx_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ err = stac92xx_parse_auto_config(codec, 0x1d, 0);
+ if (!err) {
+ if (spec->board_config < 0) {
+ printk(KERN_WARNING "hda_codec: No auto-config is "
+ "available, default to model=ref\n");
+ spec->board_config = STAC_92HD83XXX_REF;
+ goto again;
+ }
+ err = -EINVAL;
+ }
+
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i;
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_POWER_STATE, pwr);
+
+ msleep(1);
+ for (i = 0; i < spec->num_adcs; i++) {
+ snd_hda_codec_write_cache(codec,
+ spec->adc_nids[i], 0,
+ AC_VERB_SET_POWER_STATE, pwr);
+ }
+};
+
+static int stac92hd71xx_resume(struct hda_codec *codec)
+{
+ stac92hd71xx_set_power_state(codec, AC_PWRST_D0);
+ return stac92xx_resume(codec);
+}
+
+static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state)
+{
+ struct sigmatel_spec *spec = codec->spec;
+
+ stac92hd71xx_set_power_state(codec, AC_PWRST_D3);
+ if (spec->eapd_mask)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data &
+ ~spec->eapd_mask);
+ return 0;
+};
+
+#endif
+
+static struct hda_codec_ops stac92hd71bxx_patch_ops = {
+ .build_controls = stac92xx_build_controls,
+ .build_pcms = stac92xx_build_pcms,
+ .init = stac92xx_init,
+ .free = stac92xx_free,
+ .unsol_event = stac92xx_unsol_event,
+#ifdef SND_HDA_NEEDS_RESUME
+ .resume = stac92hd71xx_resume,
+ .suspend = stac92hd71xx_suspend,
+#endif
+};
+
+static struct hda_input_mux stac92hd71bxx_dmux = {
+ .num_items = 4,
+ .items = {
+ { "Analog Inputs", 0x00 },
+ { "Mixer", 0x01 },
+ { "Digital Mic 1", 0x02 },
+ { "Digital Mic 2", 0x03 },
+ }
+};
+
+static int patch_stac92hd71bxx(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err = 0;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ codec->patch_ops = stac92xx_patch_ops;
+ spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
+ spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
+ spec->pin_nids = stac92hd71bxx_pin_nids;
+ memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
+ sizeof(stac92hd71bxx_dmux));
+ spec->board_config = snd_hda_check_board_config(codec,
+ STAC_92HD71BXX_MODELS,
+ stac92hd71bxx_models,
+ stac92hd71bxx_cfg_tbl);
+again:
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+ " STAC92HD71BXX, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
+ spec->pin_configs = stac92hd71bxx_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ if (spec->board_config > STAC_92HD71BXX_REF) {
+ /* GPIO0 = EAPD */
+ spec->gpio_mask = 0x01;
+ spec->gpio_dir = 0x01;
+ spec->gpio_data = 0x01;
+ }
+
+ switch (codec->vendor_id) {
+ case 0x111d76b6: /* 4 Port without Analog Mixer */
+ case 0x111d76b7:
+ case 0x111d76b4: /* 6 Port without Analog Mixer */
+ case 0x111d76b5:
+ spec->mixer = stac92hd71bxx_mixer;
+ spec->init = stac92hd71bxx_core_init;
+ codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
+ break;
+ case 0x111d7608: /* 5 Port with Analog Mixer */
+ switch (spec->board_config) {
+ case STAC_HP_M4:
+ /* Enable VREF power saving on GPIO1 detect */
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02);
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ (AC_USRSP_EN | STAC_VREF_EVENT | 0x01));
+ spec->gpio_mask |= 0x02;
+ break;
+ }
+ if ((codec->revision_id & 0xf) == 0 ||
+ (codec->revision_id & 0xf) == 1) {
+#ifdef SND_HDA_NEEDS_RESUME
+ codec->patch_ops = stac92hd71bxx_patch_ops;
+#endif
+ spec->stream_delay = 40; /* 40 milliseconds */
+ }
+
+ /* no output amps */
+ spec->num_pwrs = 0;
+ spec->mixer = stac92hd71bxx_analog_mixer;
+ spec->dinput_mux = &spec->private_dimux;
+
+ /* disable VSW */
+ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
+ stac92xx_set_config_reg(codec, 0xf, 0x40f000f0);
+ break;
+ case 0x111d7603: /* 6 Port with Analog Mixer */
+ if ((codec->revision_id & 0xf) == 1) {
+#ifdef SND_HDA_NEEDS_RESUME
+ codec->patch_ops = stac92hd71bxx_patch_ops;
+#endif
+ spec->stream_delay = 40; /* 40 milliseconds */
+ }
+
+ /* no output amps */
+ spec->num_pwrs = 0;
+ /* fallthru */
+ default:
+ spec->dinput_mux = &spec->private_dimux;
+ spec->mixer = stac92hd71bxx_analog_mixer;
+ spec->init = stac92hd71bxx_analog_core_init;
+ codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
+ }
+
+ spec->aloopback_mask = 0x50;
+ spec->aloopback_shift = 0;
+
+ spec->powerdown_adcs = 1;
+ spec->digbeep_nid = 0x26;
+ spec->mux_nids = stac92hd71bxx_mux_nids;
+ spec->adc_nids = stac92hd71bxx_adc_nids;
+ spec->dmic_nids = stac92hd71bxx_dmic_nids;
+ spec->dmux_nids = stac92hd71bxx_dmux_nids;
+ spec->smux_nids = stac92hd71bxx_smux_nids;
+ spec->pwr_nids = stac92hd71bxx_pwr_nids;
+
+ spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids);
+
+ switch (spec->board_config) {
+ case STAC_HP_M4:
+ /* enable internal microphone */
+ stac92xx_set_config_reg(codec, 0x0e, 0x01813040);
+ stac92xx_auto_set_pinctl(codec, 0x0e,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_80);
+ /* fallthru */
+ case STAC_DELL_M4_2:
+ spec->num_dmics = 0;
+ spec->num_smuxes = 0;
+ spec->num_dmuxes = 0;
+ break;
+ case STAC_DELL_M4_1:
+ case STAC_DELL_M4_3:
+ spec->num_dmics = 1;
+ spec->num_smuxes = 0;
+ spec->num_dmuxes = 1;
+ break;
+ default:
+ spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
+ spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids);
+ spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+ };
+
+ spec->multiout.num_dacs = 1;
+ spec->multiout.hp_nid = 0x11;
+ spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
+ if (spec->dinput_mux)
+ spec->private_dimux.num_items +=
+ spec->num_dmics -
+ (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
+
+ err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
+ if (!err) {
+ if (spec->board_config < 0) {
+ printk(KERN_WARNING "hda_codec: No auto-config is "
+ "available, default to model=ref\n");
+ spec->board_config = STAC_92HD71BXX_REF;
+ goto again;
+ }
+ err = -EINVAL;
+ }
+
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ return 0;
+};
+
+static int patch_stac922x(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac922x_pin_nids);
+ spec->pin_nids = stac922x_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS,
+ stac922x_models,
+ stac922x_cfg_tbl);
+ if (spec->board_config == STAC_INTEL_MAC_AUTO) {
+ spec->gpio_mask = spec->gpio_dir = 0x03;
+ spec->gpio_data = 0x03;
+ /* Intel Macs have all same PCI SSID, so we need to check
+ * codec SSID to distinguish the exact models
+ */
+ printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id);
+ switch (codec->subsystem_id) {
+
+ case 0x106b0800:
+ spec->board_config = STAC_INTEL_MAC_V1;
+ break;
+ case 0x106b0600:
+ case 0x106b0700:
+ spec->board_config = STAC_INTEL_MAC_V2;
+ break;
+ case 0x106b0e00:
+ case 0x106b0f00:
+ case 0x106b1600:
+ case 0x106b1700:
+ case 0x106b0200:
+ case 0x106b1e00:
+ spec->board_config = STAC_INTEL_MAC_V3;
+ break;
+ case 0x106b1a00:
+ case 0x00000100:
+ spec->board_config = STAC_INTEL_MAC_V4;
+ break;
+ case 0x106b0a00:
+ case 0x106b2200:
+ spec->board_config = STAC_INTEL_MAC_V5;
+ break;
+ default:
+ spec->board_config = STAC_INTEL_MAC_V3;
+ break;
+ }
+ }
+
+ again:
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
+ "using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else if (stac922x_brd_tbl[spec->board_config] != NULL) {
+ spec->pin_configs = stac922x_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ spec->adc_nids = stac922x_adc_nids;
+ spec->mux_nids = stac922x_mux_nids;
+ spec->num_muxes = ARRAY_SIZE(stac922x_mux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac922x_adc_nids);
+ spec->num_dmics = 0;
+ spec->num_pwrs = 0;
+
+ spec->init = stac922x_core_init;
+ spec->mixer = stac922x_mixer;
+
+ spec->multiout.dac_nids = spec->dac_nids;
+
+ err = stac92xx_parse_auto_config(codec, 0x08, 0x09);
+ if (!err) {
+ if (spec->board_config < 0) {
+ printk(KERN_WARNING "hda_codec: No auto-config is "
+ "available, default to model=ref\n");
+ spec->board_config = STAC_D945_REF;
+ goto again;
+ }
+ err = -EINVAL;
+ }
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ /* Fix Mux capture level; max to 2 */
+ snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT,
+ (0 << AC_AMPCAP_OFFSET_SHIFT) |
+ (2 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (0 << AC_AMPCAP_MUTE_SHIFT));
+
+ return 0;
+}
+
+static int patch_stac927x(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
+ spec->pin_nids = stac927x_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS,
+ stac927x_models,
+ stac927x_cfg_tbl);
+ again:
+ if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) {
+ if (spec->board_config < 0)
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+ "STAC927x, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
+ spec->pin_configs = stac927x_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ spec->digbeep_nid = 0x23;
+ spec->adc_nids = stac927x_adc_nids;
+ spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids);
+ spec->mux_nids = stac927x_mux_nids;
+ spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids);
+ spec->smux_nids = stac927x_smux_nids;
+ spec->num_smuxes = ARRAY_SIZE(stac927x_smux_nids);
+ spec->spdif_labels = stac927x_spdif_labels;
+ spec->dac_list = stac927x_dac_nids;
+ spec->multiout.dac_nids = spec->dac_nids;
+
+ switch (spec->board_config) {
+ case STAC_D965_3ST:
+ case STAC_D965_5ST:
+ /* GPIO0 High = Enable EAPD */
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01;
+ spec->gpio_data = 0x01;
+ spec->num_dmics = 0;
+
+ spec->init = d965_core_init;
+ spec->mixer = stac927x_mixer;
+ break;
+ case STAC_DELL_BIOS:
+ switch (codec->subsystem_id) {
+ case 0x10280209:
+ case 0x1028022e:
+ /* correct the device field to SPDIF out */
+ stac92xx_set_config_reg(codec, 0x21, 0x01442070);
+ break;
+ };
+ /* configure the analog microphone on some laptops */
+ stac92xx_set_config_reg(codec, 0x0c, 0x90a79130);
+ /* correct the front output jack as a hp out */
+ stac92xx_set_config_reg(codec, 0x0f, 0x0227011f);
+ /* correct the front input jack as a mic */
+ stac92xx_set_config_reg(codec, 0x0e, 0x02a79130);
+ /* fallthru */
+ case STAC_DELL_3ST:
+ /* GPIO2 High = Enable EAPD */
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04;
+ spec->gpio_data = 0x04;
+ spec->dmic_nids = stac927x_dmic_nids;
+ spec->num_dmics = STAC927X_NUM_DMICS;
+
+ spec->init = d965_core_init;
+ spec->mixer = stac927x_mixer;
+ spec->dmux_nids = stac927x_dmux_nids;
+ spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
+ break;
+ default:
+ if (spec->board_config > STAC_D965_REF) {
+ /* GPIO0 High = Enable EAPD */
+ spec->eapd_mask = spec->gpio_mask = 0x01;
+ spec->gpio_dir = spec->gpio_data = 0x01;
+ }
+ spec->num_dmics = 0;
+
+ spec->init = stac927x_core_init;
+ spec->mixer = stac927x_mixer;
+ }
+
+ spec->num_pwrs = 0;
+ spec->aloopback_mask = 0x40;
+ spec->aloopback_shift = 0;
+ spec->eapd_switch = 1;
+
+ err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
+ if (!err) {
+ if (spec->board_config < 0) {
+ printk(KERN_WARNING "hda_codec: No auto-config is "
+ "available, default to model=ref\n");
+ spec->board_config = STAC_D965_REF;
+ goto again;
+ }
+ err = -EINVAL;
+ }
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ /*
+ * !!FIXME!!
+ * The STAC927x seem to require fairly long delays for certain
+ * command sequences. With too short delays (even if the answer
+ * is set to RIRB properly), it results in the silence output
+ * on some hardwares like Dell.
+ *
+ * The below flag enables the longer delay (see get_response
+ * in hda_intel.c).
+ */
+ codec->bus->needs_damn_long_delay = 1;
+
+ /* no jack detecion for ref-no-jd model */
+ if (spec->board_config == STAC_D965_REF_NO_JD)
+ spec->hp_detect = 0;
+
+ return 0;
+}
+
+static int patch_stac9205(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ spec->num_pins = ARRAY_SIZE(stac9205_pin_nids);
+ spec->pin_nids = stac9205_pin_nids;
+ spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS,
+ stac9205_models,
+ stac9205_cfg_tbl);
+ again:
+ if (spec->board_config < 0) {
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n");
+ err = stac92xx_save_bios_config_regs(codec);
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+ spec->pin_configs = spec->bios_pin_configs;
+ } else {
+ spec->pin_configs = stac9205_brd_tbl[spec->board_config];
+ stac92xx_set_config_regs(codec);
+ }
+
+ spec->digbeep_nid = 0x23;
+ spec->adc_nids = stac9205_adc_nids;
+ spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids);
+ spec->mux_nids = stac9205_mux_nids;
+ spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids);
+ spec->smux_nids = stac9205_smux_nids;
+ spec->num_smuxes = ARRAY_SIZE(stac9205_smux_nids);
+ spec->dmic_nids = stac9205_dmic_nids;
+ spec->num_dmics = STAC9205_NUM_DMICS;
+ spec->dmux_nids = stac9205_dmux_nids;
+ spec->num_dmuxes = ARRAY_SIZE(stac9205_dmux_nids);
+ spec->num_pwrs = 0;
+
+ spec->init = stac9205_core_init;
+ spec->mixer = stac9205_mixer;
+
+ spec->aloopback_mask = 0x40;
+ spec->aloopback_shift = 0;
+ spec->eapd_switch = 1;
+ spec->multiout.dac_nids = spec->dac_nids;
+
+ switch (spec->board_config){
+ case STAC_9205_DELL_M43:
+ /* Enable SPDIF in/out */
+ stac92xx_set_config_reg(codec, 0x1f, 0x01441030);
+ stac92xx_set_config_reg(codec, 0x20, 0x1c410030);
+
+ /* Enable unsol response for GPIO4/Dock HP connection */
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10);
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ (AC_USRSP_EN | STAC_HP_EVENT));
+
+ spec->gpio_dir = 0x0b;
+ spec->eapd_mask = 0x01;
+ spec->gpio_mask = 0x1b;
+ spec->gpio_mute = 0x10;
+ /* GPIO0 High = EAPD, GPIO1 Low = Headphone Mute,
+ * GPIO3 Low = DRM
+ */
+ spec->gpio_data = 0x01;
+ break;
+ case STAC_9205_REF:
+ /* SPDIF-In enabled */
+ break;
+ default:
+ /* GPIO0 High = EAPD */
+ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
+ spec->gpio_data = 0x01;
+ break;
+ }
+
+ err = stac92xx_parse_auto_config(codec, 0x1f, 0x20);
+ if (!err) {
+ if (spec->board_config < 0) {
+ printk(KERN_WARNING "hda_codec: No auto-config is "
+ "available, default to model=ref\n");
+ spec->board_config = STAC_9205_REF;
+ goto again;
+ }
+ err = -EINVAL;
+ }
+ if (err < 0) {
+ stac92xx_free(codec);
+ return err;
+ }
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+/*
+ * STAC9872 hack
+ */
+
+/* static config for Sony VAIO FE550G and Sony VAIO AR */
+static hda_nid_t vaio_dacs[] = { 0x2 };
+#define VAIO_HP_DAC 0x5
+static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ };
+static hda_nid_t vaio_mux_nids[] = { 0x15 };
+
+static struct hda_input_mux vaio_mux = {
+ .num_items = 3,
+ .items = {
+ /* { "HP", 0x0 }, */
+ { "Mic Jack", 0x1 },
+ { "Internal Mic", 0x2 },
+ { "PCM", 0x3 },
+ }
+};
+
+static struct hda_verb vaio_init[] = {
+ {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
+ {0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT},
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
+ {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
+ {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
+ {}
+};
+
+static struct hda_verb vaio_ar_init[] = {
+ {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
+ {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
+ {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
+/* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
+/* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
+ {}
+};
+
+/* bind volumes of both NID 0x02 and 0x05 */
+static struct hda_bind_ctls vaio_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/* bind volumes of both NID 0x02 and 0x05 */
+static struct hda_bind_ctls vaio_bind_master_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
+ 0,
+ },
+};
+
+static struct snd_kcontrol_new vaio_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw),
+ /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = stac92xx_mux_enum_info,
+ .get = stac92xx_mux_enum_get,
+ .put = stac92xx_mux_enum_put,
+ },
+ {}
+};
+
+static struct snd_kcontrol_new vaio_ar_mixer[] = {
+ HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw),
+ /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
+ HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
+ /*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .count = 1,
+ .info = stac92xx_mux_enum_info,
+ .get = stac92xx_mux_enum_get,
+ .put = stac92xx_mux_enum_put,
+ },
+ {}
+};
+
+static struct hda_codec_ops stac9872_patch_ops = {
+ .build_controls = stac92xx_build_controls,
+ .build_pcms = stac92xx_build_pcms,
+ .init = stac92xx_init,
+ .free = stac92xx_free,
+#ifdef SND_HDA_NEEDS_RESUME
+ .resume = stac92xx_resume,
+#endif
+};
+
+static int stac9872_vaio_init(struct hda_codec *codec)
+{
+ int err;
+
+ err = stac92xx_init(codec);
+ if (err < 0)
+ return err;
+ if (codec->patch_ops.unsol_event)
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+ return 0;
+}
+
+static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res)
+{
+ if (get_hp_pin_presence(codec, 0x0a)) {
+ stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
+ stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
+ } else {
+ stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
+ stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
+ }
+}
+
+static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ switch (res >> 26) {
+ case STAC_HP_EVENT:
+ stac9872_vaio_hp_detect(codec, res);
+ break;
+ }
+}
+
+static struct hda_codec_ops stac9872_vaio_patch_ops = {
+ .build_controls = stac92xx_build_controls,
+ .build_pcms = stac92xx_build_pcms,
+ .init = stac9872_vaio_init,
+ .free = stac92xx_free,
+ .unsol_event = stac9872_vaio_unsol_event,
+#ifdef CONFIG_PM
+ .resume = stac92xx_resume,
+#endif
+};
+
+enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */
+ CXD9872RD_VAIO,
+ /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */
+ STAC9872AK_VAIO,
+ /* Unknown. id=0x83847661 and subsys=0x104D1200. */
+ STAC9872K_VAIO,
+ /* AR Series. id=0x83847664 and subsys=104D1300 */
+ CXD9872AKD_VAIO,
+ STAC_9872_MODELS,
+};
+
+static const char *stac9872_models[STAC_9872_MODELS] = {
+ [CXD9872RD_VAIO] = "vaio",
+ [CXD9872AKD_VAIO] = "vaio-ar",
+};
+
+static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x104d, 0x81e6, "Sony VAIO F/S", CXD9872RD_VAIO),
+ SND_PCI_QUIRK(0x104d, 0x81ef, "Sony VAIO F/S", CXD9872RD_VAIO),
+ SND_PCI_QUIRK(0x104d, 0x81fd, "Sony VAIO AR", CXD9872AKD_VAIO),
+ SND_PCI_QUIRK(0x104d, 0x8205, "Sony VAIO AR", CXD9872AKD_VAIO),
+ {}
+};
+
+static int patch_stac9872(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+ int board_config;
+
+ board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
+ stac9872_models,
+ stac9872_cfg_tbl);
+ if (board_config < 0)
+ /* unknown config, let generic-parser do its job... */
+ return snd_hda_parse_generic_codec(codec);
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+ switch (board_config) {
+ case CXD9872RD_VAIO:
+ case STAC9872AK_VAIO:
+ case STAC9872K_VAIO:
+ spec->mixer = vaio_mixer;
+ spec->init = vaio_init;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs);
+ spec->multiout.dac_nids = vaio_dacs;
+ spec->multiout.hp_nid = VAIO_HP_DAC;
+ spec->num_adcs = ARRAY_SIZE(vaio_adcs);
+ spec->adc_nids = vaio_adcs;
+ spec->num_pwrs = 0;
+ spec->input_mux = &vaio_mux;
+ spec->mux_nids = vaio_mux_nids;
+ codec->patch_ops = stac9872_vaio_patch_ops;
+ break;
+
+ case CXD9872AKD_VAIO:
+ spec->mixer = vaio_ar_mixer;
+ spec->init = vaio_ar_init;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs);
+ spec->multiout.dac_nids = vaio_dacs;
+ spec->multiout.hp_nid = VAIO_HP_DAC;
+ spec->num_adcs = ARRAY_SIZE(vaio_adcs);
+ spec->num_pwrs = 0;
+ spec->adc_nids = vaio_adcs;
+ spec->input_mux = &vaio_mux;
+ spec->mux_nids = vaio_mux_nids;
+ codec->patch_ops = stac9872_patch_ops;
+ break;
+ }
+
+ return 0;
+}
+
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_sigmatel[] = {
+ { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 },
+ { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x },
+ { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x },
+ { .id = 0x83847880, .name = "STAC9220 A2", .patch = patch_stac922x },
+ { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x },
+ { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x },
+ { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x },
+ { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac927x },
+ { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac927x },
+ { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac927x },
+ { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac927x },
+ { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac927x },
+ { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac927x },
+ { .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x },
+ { .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x },
+ { .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x },
+ { .id = 0x83847623, .name = "STAC9273D", .patch = patch_stac927x },
+ { .id = 0x83847624, .name = "STAC9272X", .patch = patch_stac927x },
+ { .id = 0x83847625, .name = "STAC9272D", .patch = patch_stac927x },
+ { .id = 0x83847626, .name = "STAC9271X", .patch = patch_stac927x },
+ { .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x },
+ { .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x },
+ { .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x },
+ { .id = 0x83847632, .name = "STAC9202", .patch = patch_stac925x },
+ { .id = 0x83847633, .name = "STAC9202D", .patch = patch_stac925x },
+ { .id = 0x83847634, .name = "STAC9250", .patch = patch_stac925x },
+ { .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x },
+ { .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x },
+ { .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x },
+ { .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x },
+ { .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x },
+ /* The following does not take into account .id=0x83847661 when subsys =
+ * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
+ * currently not fully supported.
+ */
+ { .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 },
+ { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 },
+ { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 },
+ { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 },
+ { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 },
+ { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 },
+ { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },
+ { .id = 0x838476a4, .name = "STAC9255", .patch = patch_stac9205 },
+ { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 },
+ { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
+ { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
+ { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
+ { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
+ { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
+ { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
+ { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
+ { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
+ {} /* terminator */
+};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
new file mode 100644
index 0000000..63e4871
--- /dev/null
+++ b/sound/pci/hda/patch_via.c
@@ -0,0 +1,3335 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec
+ *
+ * Copyright (c) 2006-2008 Lydia Wang <lydiawang@viatech.com>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */
+/* */
+/* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */
+/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */
+/* 2006-08-02 Lydia Wang Add support to VT1709 codec */
+/* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */
+/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */
+/* 2007-09-17 Lydia Wang Add VT1708B codec support */
+/* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */
+/* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */
+/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */
+/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */
+/* 2008-04-09 Lydia Wang Add Independent HP feature */
+/* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */
+/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */
+/* */
+/* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */
+
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/asoundef.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_patch.h"
+
+/* amp values */
+#define AMP_VAL_IDX_SHIFT 19
+#define AMP_VAL_IDX_MASK (0x0f<<19)
+
+#define NUM_CONTROL_ALLOC 32
+#define NUM_VERB_ALLOC 32
+
+/* Pin Widget NID */
+#define VT1708_HP_NID 0x13
+#define VT1708_DIGOUT_NID 0x14
+#define VT1708_DIGIN_NID 0x16
+#define VT1708_DIGIN_PIN 0x26
+#define VT1708_HP_PIN_NID 0x20
+#define VT1708_CD_PIN_NID 0x24
+
+#define VT1709_HP_DAC_NID 0x28
+#define VT1709_DIGOUT_NID 0x13
+#define VT1709_DIGIN_NID 0x17
+#define VT1709_DIGIN_PIN 0x25
+
+#define VT1708B_HP_NID 0x25
+#define VT1708B_DIGOUT_NID 0x12
+#define VT1708B_DIGIN_NID 0x15
+#define VT1708B_DIGIN_PIN 0x21
+
+#define VT1708S_HP_NID 0x25
+#define VT1708S_DIGOUT_NID 0x12
+
+#define VT1702_HP_NID 0x17
+#define VT1702_DIGOUT_NID 0x11
+
+#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b)
+#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713)
+#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717)
+#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723)
+#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727)
+#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397)
+#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398)
+
+enum VIA_HDA_CODEC {
+ UNKNOWN = -1,
+ VT1708,
+ VT1709_10CH,
+ VT1709_6CH,
+ VT1708B_8CH,
+ VT1708B_4CH,
+ VT1708S,
+ VT1702,
+ CODEC_TYPES,
+};
+
+static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id)
+{
+ u16 ven_id = vendor_id >> 16;
+ u16 dev_id = vendor_id & 0xffff;
+ enum VIA_HDA_CODEC codec_type;
+
+ /* get codec type */
+ if (ven_id != 0x1106)
+ codec_type = UNKNOWN;
+ else if (dev_id >= 0x1708 && dev_id <= 0x170b)
+ codec_type = VT1708;
+ else if (dev_id >= 0xe710 && dev_id <= 0xe713)
+ codec_type = VT1709_10CH;
+ else if (dev_id >= 0xe714 && dev_id <= 0xe717)
+ codec_type = VT1709_6CH;
+ else if (dev_id >= 0xe720 && dev_id <= 0xe723)
+ codec_type = VT1708B_8CH;
+ else if (dev_id >= 0xe724 && dev_id <= 0xe727)
+ codec_type = VT1708B_4CH;
+ else if ((dev_id & 0xfff) == 0x397
+ && (dev_id >> 12) < 8)
+ codec_type = VT1708S;
+ else if ((dev_id & 0xfff) == 0x398
+ && (dev_id >> 12) < 8)
+ codec_type = VT1702;
+ else
+ codec_type = UNKNOWN;
+ return codec_type;
+};
+
+#define VIA_HP_EVENT 0x01
+#define VIA_GPIO_EVENT 0x02
+
+enum {
+ VIA_CTL_WIDGET_VOL,
+ VIA_CTL_WIDGET_MUTE,
+};
+
+enum {
+ AUTO_SEQ_FRONT = 0,
+ AUTO_SEQ_SURROUND,
+ AUTO_SEQ_CENLFE,
+ AUTO_SEQ_SIDE
+};
+
+#define get_amp_nid(kc) ((kc)->private_value & 0xffff)
+
+/* Some VT1708S based boards gets the micboost setting wrong, so we have
+ * to apply some brute-force and re-write the TLV's by software. */
+static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *_tlv)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+
+ if (get_codec_type(codec->vendor_id) == VT1708S
+ && (nid == 0x1a || nid == 0x1e)) {
+ if (size < 4 * sizeof(unsigned int))
+ return -ENOMEM;
+ if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */
+ return -EFAULT;
+ if (put_user(2 * sizeof(unsigned int), _tlv + 1))
+ return -EFAULT;
+ if (put_user(0, _tlv + 2)) /* offset = 0 */
+ return -EFAULT;
+ if (put_user(1000, _tlv + 3)) /* step size = 10 dB */
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static int mic_boost_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = get_amp_nid(kcontrol);
+
+ if (get_codec_type(codec->vendor_id) == VT1708S
+ && (nid == 0x1a || nid == 0x1e)) {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 3;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new vt1708_control_templates[] = {
+ HDA_CODEC_VOLUME(NULL, 0, 0, 0),
+ HDA_CODEC_MUTE(NULL, 0, 0, 0),
+};
+
+
+struct via_spec {
+ /* codec parameterization */
+ struct snd_kcontrol_new *mixers[3];
+ unsigned int num_mixers;
+
+ struct hda_verb *init_verbs[5];
+ unsigned int num_iverbs;
+
+ char *stream_name_analog;
+ struct hda_pcm_stream *stream_analog_playback;
+ struct hda_pcm_stream *stream_analog_capture;
+
+ char *stream_name_digital;
+ struct hda_pcm_stream *stream_digital_playback;
+ struct hda_pcm_stream *stream_digital_capture;
+
+ /* playback */
+ struct hda_multi_out multiout;
+ hda_nid_t extra_dig_out_nid;
+
+ /* capture */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ hda_nid_t dig_in_nid;
+
+ /* capture source */
+ const struct hda_input_mux *input_mux;
+ unsigned int cur_mux[3];
+
+ /* PCM information */
+ struct hda_pcm pcm_rec[3];
+
+ /* dynamic controls, init_verbs and input_mux */
+ struct auto_pin_cfg autocfg;
+ unsigned int num_kctl_alloc, num_kctl_used;
+ struct snd_kcontrol_new *kctl_alloc;
+ struct hda_input_mux private_imux[2];
+ hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+
+ /* HP mode source */
+ const struct hda_input_mux *hp_mux;
+ unsigned int hp_independent_mode;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
+};
+
+static hda_nid_t vt1708_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x15, 0x27
+};
+
+static hda_nid_t vt1709_adc_nids[3] = {
+ /* ADC1-2 */
+ 0x14, 0x15, 0x16
+};
+
+static hda_nid_t vt1708B_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x13, 0x14
+};
+
+static hda_nid_t vt1708S_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x13, 0x14
+};
+
+static hda_nid_t vt1702_adc_nids[3] = {
+ /* ADC1-2 */
+ 0x12, 0x20, 0x1F
+};
+
+/* add dynamic controls */
+static int via_add_control(struct via_spec *spec, int type, const char *name,
+ unsigned long val)
+{
+ struct snd_kcontrol_new *knew;
+
+ if (spec->num_kctl_used >= spec->num_kctl_alloc) {
+ int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
+
+ /* array + terminator */
+ knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL);
+ if (!knew)
+ return -ENOMEM;
+ if (spec->kctl_alloc) {
+ memcpy(knew, spec->kctl_alloc,
+ sizeof(*knew) * spec->num_kctl_alloc);
+ kfree(spec->kctl_alloc);
+ }
+ spec->kctl_alloc = knew;
+ spec->num_kctl_alloc = num;
+ }
+
+ knew = &spec->kctl_alloc[spec->num_kctl_used];
+ *knew = vt1708_control_templates[type];
+ knew->name = kstrdup(name, GFP_KERNEL);
+
+ if (!knew->name)
+ return -ENOMEM;
+ knew->private_value = val;
+ spec->num_kctl_used++;
+ return 0;
+}
+
+/* create input playback/capture controls for the given pin */
+static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin,
+ const char *ctlname, int idx, int mix_nid)
+{
+ char name[32];
+ int err;
+
+ sprintf(name, "%s Playback Volume", ctlname);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", ctlname);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static void via_auto_set_output_and_unmute(struct hda_codec *codec,
+ hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ /* set as output */
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pin_type);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
+}
+
+
+static void via_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ if (nid)
+ via_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
+ }
+}
+
+static void via_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pins[0];
+ if (pin) /* connect to front */
+ via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+}
+
+static void via_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ (i <= AUTO_PIN_FRONT_MIC ?
+ PIN_VREF50 : PIN_IN));
+
+ }
+}
+/*
+ * input MUX handling
+ */
+static int via_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int via_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ return 0;
+}
+
+static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int vendor_id = codec->vendor_id;
+
+ /* AIW0 lydia 060801 add for correct sw0 input select */
+ if (IS_VT1708_VENDORID(vendor_id) && (adc_idx == 0))
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ 0x18, &spec->cur_mux[adc_idx]);
+ else if ((IS_VT1709_10CH_VENDORID(vendor_id) ||
+ IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0))
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ 0x19, &spec->cur_mux[adc_idx]);
+ else if ((IS_VT1708B_8CH_VENDORID(vendor_id) ||
+ IS_VT1708B_4CH_VENDORID(vendor_id)) && (adc_idx == 0))
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ 0x17, &spec->cur_mux[adc_idx]);
+ else if (IS_VT1702_VENDORID(vendor_id) && (adc_idx == 0))
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ 0x13, &spec->cur_mux[adc_idx]);
+ else
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ spec->adc_nids[adc_idx],
+ &spec->cur_mux[adc_idx]);
+}
+
+static int via_independent_hp_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(spec->hp_mux, uinfo);
+}
+
+static int via_independent_hp_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ hda_nid_t nid = spec->autocfg.hp_pins[0];
+ unsigned int pinsel = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_SEL,
+ 0x00);
+
+ ucontrol->value.enumerated.item[0] = pinsel;
+
+ return 0;
+}
+
+static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ hda_nid_t nid = spec->autocfg.hp_pins[0];
+ unsigned int pinsel = ucontrol->value.enumerated.item[0];
+ unsigned int con_nid = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+
+ if (con_nid == spec->multiout.hp_nid) {
+ if (pinsel == 0) {
+ if (!spec->hp_independent_mode) {
+ if (spec->multiout.num_dacs > 1)
+ spec->multiout.num_dacs -= 1;
+ spec->hp_independent_mode = 1;
+ }
+ } else if (pinsel == 1) {
+ if (spec->hp_independent_mode) {
+ if (spec->multiout.num_dacs > 1)
+ spec->multiout.num_dacs += 1;
+ spec->hp_independent_mode = 0;
+ }
+ }
+ } else {
+ if (pinsel == 0) {
+ if (spec->hp_independent_mode) {
+ if (spec->multiout.num_dacs > 1)
+ spec->multiout.num_dacs += 1;
+ spec->hp_independent_mode = 0;
+ }
+ } else if (pinsel == 1) {
+ if (!spec->hp_independent_mode) {
+ if (spec->multiout.num_dacs > 1)
+ spec->multiout.num_dacs -= 1;
+ spec->hp_independent_mode = 1;
+ }
+ }
+ }
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
+ pinsel);
+
+ if (spec->multiout.hp_nid &&
+ spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT])
+ snd_hda_codec_setup_stream(codec,
+ spec->multiout.hp_nid,
+ 0, 0, 0);
+
+ return 0;
+}
+
+static struct snd_kcontrol_new via_hp_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Independent HP",
+ .count = 1,
+ .info = via_independent_hp_info,
+ .get = via_independent_hp_get,
+ .put = via_independent_hp_put,
+ },
+ { } /* end */
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1708_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x27, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x27, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb vt1708_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ */
+ /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /*
+ * Set up output mixers (0x19 - 0x1b)
+ */
+ /* set vol=0 to output mixers */
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Setup default input to PW4 */
+ {0x20, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* PW9 Output enable */
+ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ { }
+};
+
+static int via_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
+
+static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+
+static void playback_multi_pcm_prep_0(struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ struct hda_multi_out *mout = &spec->multiout;
+ hda_nid_t *nids = mout->dac_nids;
+ int chs = substream->runtime->channels;
+ int i;
+
+ mutex_lock(&codec->spdif_mutex);
+ if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
+ if (chs == 2 &&
+ snd_hda_is_supported_format(codec, mout->dig_out_nid,
+ format) &&
+ !(codec->spdif_status & IEC958_AES0_NONAUDIO)) {
+ mout->dig_out_used = HDA_DIG_ANALOG_DUP;
+ /* turn off SPDIF once; otherwise the IEC958 bits won't
+ * be updated */
+ if (codec->spdif_ctls & AC_DIG1_ENABLE)
+ snd_hda_codec_write(codec, mout->dig_out_nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls &
+ ~AC_DIG1_ENABLE & 0xff);
+ snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+ stream_tag, 0, format);
+ /* turn on again (if needed) */
+ if (codec->spdif_ctls & AC_DIG1_ENABLE)
+ snd_hda_codec_write(codec, mout->dig_out_nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls & 0xff);
+ } else {
+ mout->dig_out_used = 0;
+ snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+ 0, 0, 0);
+ }
+ }
+ mutex_unlock(&codec->spdif_mutex);
+
+ /* front */
+ snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag,
+ 0, format);
+
+ if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
+ !spec->hp_independent_mode)
+ /* headphone out will just decode front left/right (stereo) */
+ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag,
+ 0, format);
+
+ /* extra outputs copied from front */
+ for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
+ if (mout->extra_out_nid[i])
+ snd_hda_codec_setup_stream(codec,
+ mout->extra_out_nid[i],
+ stream_tag, 0, format);
+
+ /* surrounds */
+ for (i = 1; i < mout->num_dacs; i++) {
+ if (chs >= (i + 1) * 2) /* independent out */
+ snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
+ i * 2, format);
+ else /* copy front */
+ snd_hda_codec_setup_stream(codec, nids[i], stream_tag,
+ 0, format);
+ }
+}
+
+static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ struct hda_multi_out *mout = &spec->multiout;
+ hda_nid_t *nids = mout->dac_nids;
+
+ if (substream->number == 0)
+ playback_multi_pcm_prep_0(codec, stream_tag, format,
+ substream);
+ else {
+ if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
+ spec->hp_independent_mode)
+ snd_hda_codec_setup_stream(codec, mout->hp_nid,
+ stream_tag, 0, format);
+ }
+
+ return 0;
+}
+
+static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ struct hda_multi_out *mout = &spec->multiout;
+ hda_nid_t *nids = mout->dac_nids;
+ int i;
+
+ if (substream->number == 0) {
+ for (i = 0; i < mout->num_dacs; i++)
+ snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0);
+
+ if (mout->hp_nid && !spec->hp_independent_mode)
+ snd_hda_codec_setup_stream(codec, mout->hp_nid,
+ 0, 0, 0);
+
+ for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++)
+ if (mout->extra_out_nid[i])
+ snd_hda_codec_setup_stream(codec,
+ mout->extra_out_nid[i],
+ 0, 0, 0);
+ mutex_lock(&codec->spdif_mutex);
+ if (mout->dig_out_nid &&
+ mout->dig_out_used == HDA_DIG_ANALOG_DUP) {
+ snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+ 0, 0, 0);
+ mout->dig_out_used = 0;
+ }
+ mutex_unlock(&codec->spdif_mutex);
+ } else {
+ if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
+ spec->hp_independent_mode)
+ snd_hda_codec_setup_stream(codec, mout->hp_nid,
+ 0, 0, 0);
+ }
+
+ return 0;
+}
+
+/*
+ * Digital out
+ */
+static int via_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int via_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+/* setup SPDIF output stream */
+static void setup_dig_playback_stream(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int stream_tag, unsigned int format)
+{
+ /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
+ if (codec->spdif_ctls & AC_DIG1_ENABLE)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
+ snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+ /* turn on again (if needed) */
+ if (codec->spdif_ctls & AC_DIG1_ENABLE)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls & 0xff);
+}
+
+static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ hda_nid_t nid;
+
+ /* 1st or 2nd S/PDIF */
+ if (substream->number == 0)
+ nid = spec->multiout.dig_out_nid;
+ else if (substream->number == 1)
+ nid = spec->extra_dig_out_nid;
+ else
+ return -1;
+
+ mutex_lock(&codec->spdif_mutex);
+ setup_dig_playback_stream(codec, nid, stream_tag, format);
+ mutex_unlock(&codec->spdif_mutex);
+ return 0;
+}
+
+/*
+ * Analog capture
+ */
+static int via_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
+ return 0;
+}
+
+static struct hda_pcm_stream vt1708_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 8,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 8,
+ .nid = 0x10, /* NID to query formats and rates */
+ /* We got noisy outputs on the right channel on VT1708 when
+ * 24bit samples are used. Until any workaround is found,
+ * disable the 24bit format, so far.
+ */
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_pcm_prepare,
+ .cleanup = via_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x15, /* NID to query formats and rates */
+ .ops = {
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream vt1708_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int via_build_controls(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+ int i;
+
+ for (i = 0; i < spec->num_mixers; i++) {
+ err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->multiout.dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec,
+ &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
+
+ if (spec->extra_dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec,
+ spec->extra_dig_out_nid);
+ if (err < 0)
+ return err;
+ }
+ }
+ if (spec->dig_in_nid) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int via_build_pcms(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = spec->stream_name_analog;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+ spec->multiout.max_channels;
+
+ if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+ codec->num_pcms++;
+ info++;
+ info->name = spec->stream_name_digital;
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->multiout.dig_out_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ *(spec->stream_digital_playback);
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dig_out_nid;
+ }
+ if (spec->dig_in_nid) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ *(spec->stream_digital_capture);
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+ spec->dig_in_nid;
+ }
+ }
+
+ return 0;
+}
+
+static void via_free(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ unsigned int i;
+
+ if (!spec)
+ return;
+
+ if (spec->kctl_alloc) {
+ for (i = 0; i < spec->num_kctl_used; i++)
+ kfree(spec->kctl_alloc[i].name);
+ kfree(spec->kctl_alloc);
+ }
+
+ kfree(codec->spec);
+}
+
+/* mute internal speaker if HP is plugged */
+static void via_hp_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ struct via_spec *spec = codec->spec;
+
+ present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0],
+ HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ present ? HDA_AMP_MUTE : 0);
+}
+
+static void via_gpio_control(struct hda_codec *codec)
+{
+ unsigned int gpio_data;
+ unsigned int vol_counter;
+ unsigned int vol;
+ unsigned int master_vol;
+
+ struct via_spec *spec = codec->spec;
+
+ gpio_data = snd_hda_codec_read(codec, codec->afg, 0,
+ AC_VERB_GET_GPIO_DATA, 0) & 0x03;
+
+ vol_counter = (snd_hda_codec_read(codec, codec->afg, 0,
+ 0xF84, 0) & 0x3F0000) >> 16;
+
+ vol = vol_counter & 0x1F;
+ master_vol = snd_hda_codec_read(codec, 0x1A, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_INPUT);
+
+ if (gpio_data == 0x02) {
+ /* unmute line out */
+ snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0],
+ HDA_OUTPUT, 0, HDA_AMP_MUTE, 0);
+
+ if (vol_counter & 0x20) {
+ /* decrease volume */
+ if (vol > master_vol)
+ vol = master_vol;
+ snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT,
+ 0, HDA_AMP_VOLMASK,
+ master_vol-vol);
+ } else {
+ /* increase volume */
+ snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, 0,
+ HDA_AMP_VOLMASK,
+ ((master_vol+vol) > 0x2A) ? 0x2A :
+ (master_vol+vol));
+ }
+ } else if (!(gpio_data & 0x02)) {
+ /* mute line out */
+ snd_hda_codec_amp_stereo(codec,
+ spec->autocfg.line_out_pins[0],
+ HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ HDA_AMP_MUTE);
+ }
+}
+
+/* unsolicited event for jack sensing */
+static void via_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ res >>= 26;
+ if (res == VIA_HP_EVENT)
+ via_hp_automute(codec);
+ else if (res == VIA_GPIO_EVENT)
+ via_gpio_control(codec);
+}
+
+static hda_nid_t slave_dig_outs[] = {
+ 0,
+};
+
+static int via_init(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int i;
+ for (i = 0; i < spec->num_iverbs; i++)
+ snd_hda_sequence_write(codec, spec->init_verbs[i]);
+
+ /* Lydia Add for EAPD enable */
+ if (!spec->dig_in_nid) { /* No Digital In connection */
+ if (IS_VT1708_VENDORID(codec->vendor_id)) {
+ snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_OUT);
+ snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 0x02);
+ } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) ||
+ IS_VT1709_6CH_VENDORID(codec->vendor_id)) {
+ snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_OUT);
+ snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 0x02);
+ } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) ||
+ IS_VT1708B_4CH_VENDORID(codec->vendor_id)) {
+ snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_OUT);
+ snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 0x02);
+ }
+ } else /* enable SPDIF-input pin */
+ snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+
+ /* no slave outs */
+ codec->slave_dig_outs = slave_dig_outs;
+
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct via_spec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
+/*
+ */
+static struct hda_codec_ops via_patch_ops = {
+ .build_controls = via_build_controls,
+ .build_pcms = via_build_pcms,
+ .init = via_init,
+ .free = via_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = via_check_power_status,
+#endif
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1708_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int i;
+ hda_nid_t nid;
+
+ spec->multiout.num_dacs = cfg->line_outs;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ for(i = 0; i < 4; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ /* config dac list */
+ switch (i) {
+ case AUTO_SEQ_FRONT:
+ spec->multiout.dac_nids[i] = 0x10;
+ break;
+ case AUTO_SEQ_CENLFE:
+ spec->multiout.dac_nids[i] = 0x12;
+ break;
+ case AUTO_SEQ_SURROUND:
+ spec->multiout.dac_nids[i] = 0x11;
+ break;
+ case AUTO_SEQ_SIDE:
+ spec->multiout.dac_nids[i] = 0x13;
+ break;
+ }
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+ hda_nid_t nid, nid_vol = 0;
+ int i, err;
+
+ for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
+ nid = cfg->line_out_pins[i];
+
+ if (!nid)
+ continue;
+
+ if (i != AUTO_SEQ_FRONT)
+ nid_vol = 0x18 + i;
+
+ if (i == AUTO_SEQ_CENLFE) {
+ /* Center/LFE */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_FRONT){
+ /* add control to mixer index 0 */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x17, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+
+ /* add control to PW3 */
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static void create_hp_imux(struct via_spec *spec)
+{
+ int i;
+ struct hda_input_mux *imux = &spec->private_imux[1];
+ static const char *texts[] = { "OFF", "ON", NULL};
+
+ /* for hp mode select */
+ i = 0;
+ while (texts[i] != NULL) {
+ imux->items[imux->num_items].label = texts[i];
+ imux->items[imux->num_items].index = i;
+ imux->num_items++;
+ i++;
+ }
+
+ spec->hp_mux = &spec->private_imux[1];
+}
+
+static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x1d: /* Mic */
+ idx = 2;
+ break;
+
+ case 0x1e: /* Line In */
+ idx = 3;
+ break;
+
+ case 0x21: /* Front Mic */
+ idx = 4;
+ break;
+
+ case 0x24: /* CD */
+ idx = 1;
+ break;
+ }
+ err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
+ idx, 0x17);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1708_loopbacks[] = {
+ { 0x17, HDA_INPUT, 1 },
+ { 0x17, HDA_INPUT, 2 },
+ { 0x17, HDA_INPUT, 3 },
+ { 0x17, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int def_conf;
+ unsigned char seqassoc;
+
+ def_conf = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ seqassoc = (unsigned char) get_defcfg_association(def_conf);
+ seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf);
+ if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) {
+ if (seqassoc == 0xff) {
+ def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30));
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+ def_conf >> 24);
+ }
+ }
+
+ return;
+}
+
+static int vt1708_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+ /* Add HP and CD pin config connect bit re-config action */
+ vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID);
+ vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID);
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+ err = vt1708_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1708_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1708_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = VT1708_DIGOUT_NID;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = VT1708_DIGIN_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_iverbs++] = vt1708_volume_init_verbs;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ return 1;
+}
+
+/* init callback for auto-configuration model -- overriding the default init */
+static int via_auto_init(struct hda_codec *codec)
+{
+ via_init(codec);
+ via_auto_init_multi_out(codec);
+ via_auto_init_hp_out(codec);
+ via_auto_init_analog_input(codec);
+ return 0;
+}
+
+static int patch_vt1708(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1708_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+
+ spec->stream_name_analog = "VT1708 Analog";
+ spec->stream_analog_playback = &vt1708_pcm_analog_playback;
+ /* disable 32bit format on VT1708 */
+ if (codec->vendor_id == 0x11061708)
+ spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback;
+ spec->stream_analog_capture = &vt1708_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1708 Digital";
+ spec->stream_digital_playback = &vt1708_pcm_digital_playback;
+ spec->stream_digital_capture = &vt1708_pcm_digital_capture;
+
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1708_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids);
+ spec->mixers[spec->num_mixers] = vt1708_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1708_loopbacks;
+#endif
+
+ return 0;
+}
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1709_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x15, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x16, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x16, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb vt1709_uniwill_init_verbs[] = {
+ {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ { }
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb vt1709_10ch_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ */
+ /* Amp Indices: AOW0=0, CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /*
+ * Set up output selector (0x1a, 0x1b, 0x29)
+ */
+ /* set vol=0 to output mixers */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /*
+ * Unmute PW3 and PW4
+ */
+ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Set input of PW4 as AOW4 */
+ {0x20, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* PW9 Output enable */
+ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ { }
+};
+
+static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 10,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_pcm_prepare,
+ .cleanup = via_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 6,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_pcm_prepare,
+ .cleanup = via_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1709_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x14, /* NID to query formats and rates */
+ .ops = {
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1709_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close
+ },
+};
+
+static struct hda_pcm_stream vt1709_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int vt1709_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int i;
+ hda_nid_t nid;
+
+ if (cfg->line_outs == 4) /* 10 channels */
+ spec->multiout.num_dacs = cfg->line_outs+1; /* AOW0~AOW4 */
+ else if (cfg->line_outs == 3) /* 6 channels */
+ spec->multiout.num_dacs = cfg->line_outs; /* AOW0~AOW2 */
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ if (cfg->line_outs == 4) { /* 10 channels */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ /* config dac list */
+ switch (i) {
+ case AUTO_SEQ_FRONT:
+ /* AOW0 */
+ spec->multiout.dac_nids[i] = 0x10;
+ break;
+ case AUTO_SEQ_CENLFE:
+ /* AOW2 */
+ spec->multiout.dac_nids[i] = 0x12;
+ break;
+ case AUTO_SEQ_SURROUND:
+ /* AOW3 */
+ spec->multiout.dac_nids[i] = 0x11;
+ break;
+ case AUTO_SEQ_SIDE:
+ /* AOW1 */
+ spec->multiout.dac_nids[i] = 0x27;
+ break;
+ default:
+ break;
+ }
+ }
+ }
+ spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */
+
+ } else if (cfg->line_outs == 3) { /* 6 channels */
+ for(i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ /* config dac list */
+ switch(i) {
+ case AUTO_SEQ_FRONT:
+ /* AOW0 */
+ spec->multiout.dac_nids[i] = 0x10;
+ break;
+ case AUTO_SEQ_CENLFE:
+ /* AOW2 */
+ spec->multiout.dac_nids[i] = 0x12;
+ break;
+ case AUTO_SEQ_SURROUND:
+ /* AOW1 */
+ spec->multiout.dac_nids[i] = 0x11;
+ break;
+ default:
+ break;
+ }
+ }
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+ hda_nid_t nid = 0;
+ int i, err;
+
+ for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
+ nid = cfg->line_out_pins[i];
+
+ if (!nid)
+ continue;
+
+ if (i == AUTO_SEQ_CENLFE) {
+ /* Center/LFE */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x1b, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x1b, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x1b, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x1b, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_FRONT){
+ /* add control to mixer index 0 */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x18, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x18, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+
+ /* add control to PW3 */
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_SURROUND) {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_SIDE) {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ if (spec->multiout.num_dacs == 5) /* 10 channels */
+ spec->multiout.hp_nid = VT1709_HP_DAC_NID;
+ else if (spec->multiout.num_dacs == 3) /* 6 channels */
+ spec->multiout.hp_nid = 0;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x1d: /* Mic */
+ idx = 2;
+ break;
+
+ case 0x1e: /* Line In */
+ idx = 3;
+ break;
+
+ case 0x21: /* Front Mic */
+ idx = 4;
+ break;
+
+ case 0x23: /* CD */
+ idx = 1;
+ break;
+ }
+ err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
+ idx, 0x18);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static int vt1709_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+ err = vt1709_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1709_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1709_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1709_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = VT1709_DIGOUT_NID;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = VT1709_DIGIN_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1709_loopbacks[] = {
+ { 0x18, HDA_INPUT, 1 },
+ { 0x18, HDA_INPUT, 2 },
+ { 0x18, HDA_INPUT, 3 },
+ { 0x18, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+static int patch_vt1709_10ch(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ err = vt1709_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration. "
+ "Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1709_10ch_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1709 Analog";
+ spec->stream_analog_playback = &vt1709_10ch_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1709_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1709 Digital";
+ spec->stream_digital_playback = &vt1709_pcm_digital_playback;
+ spec->stream_digital_capture = &vt1709_pcm_digital_capture;
+
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1709_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
+ spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1709_loopbacks;
+#endif
+
+ return 0;
+}
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb vt1709_6ch_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ */
+ /* Amp Indices: AOW0=0, CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /*
+ * Set up output selector (0x1a, 0x1b, 0x29)
+ */
+ /* set vol=0 to output mixers */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /*
+ * Unmute PW3 and PW4
+ */
+ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Set input of PW4 as MW0 */
+ {0x20, AC_VERB_SET_CONNECT_SEL, 0},
+ /* PW9 Output enable */
+ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ { }
+};
+
+static int patch_vt1709_6ch(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ err = vt1709_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration. "
+ "Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1709_6ch_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1709 Analog";
+ spec->stream_analog_playback = &vt1709_6ch_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1709_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1709 Digital";
+ spec->stream_digital_playback = &vt1709_pcm_digital_playback;
+ spec->stream_digital_capture = &vt1709_pcm_digital_capture;
+
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1709_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
+ spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1709_loopbacks;
+#endif
+ return 0;
+}
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1708B_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb vt1708B_8ch_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ */
+ /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /*
+ * Set up output mixers
+ */
+ /* set vol=0 to output mixers */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Setup default input to PW4 */
+ {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* PW9 Output enable */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* PW10 Input enable */
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ { }
+};
+
+static struct hda_verb vt1708B_4ch_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ */
+ /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /*
+ * Set up output mixers
+ */
+ /* set vol=0 to output mixers */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+ /* Setup default input of PW4 to MW0 */
+ {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* PW9 Output enable */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* PW10 Input enable */
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ { }
+};
+
+static struct hda_verb vt1708B_uniwill_init_verbs[] = {
+ {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ { }
+};
+
+static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 8,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 4,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708B_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x13, /* NID to query formats and rates */
+ .ops = {
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708B_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream vt1708B_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1708B_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int i;
+ hda_nid_t nid;
+
+ spec->multiout.num_dacs = cfg->line_outs;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ for (i = 0; i < 4; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ /* config dac list */
+ switch (i) {
+ case AUTO_SEQ_FRONT:
+ spec->multiout.dac_nids[i] = 0x10;
+ break;
+ case AUTO_SEQ_CENLFE:
+ spec->multiout.dac_nids[i] = 0x24;
+ break;
+ case AUTO_SEQ_SURROUND:
+ spec->multiout.dac_nids[i] = 0x11;
+ break;
+ case AUTO_SEQ_SIDE:
+ spec->multiout.dac_nids[i] = 0x25;
+ break;
+ }
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+ hda_nid_t nid_vols[] = {0x16, 0x18, 0x26, 0x27};
+ hda_nid_t nid, nid_vol = 0;
+ int i, err;
+
+ for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
+ nid = cfg->line_out_pins[i];
+
+ if (!nid)
+ continue;
+
+ nid_vol = nid_vols[i];
+
+ if (i == AUTO_SEQ_CENLFE) {
+ /* Center/LFE */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_FRONT) {
+ /* add control to mixer index 0 */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+
+ /* add control to PW3 */
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x1a: /* Mic */
+ idx = 2;
+ break;
+
+ case 0x1b: /* Line In */
+ idx = 3;
+ break;
+
+ case 0x1e: /* Front Mic */
+ idx = 4;
+ break;
+
+ case 0x1f: /* CD */
+ idx = 1;
+ break;
+ }
+ err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
+ idx, 0x16);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static int vt1708B_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+ err = vt1708B_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1708B_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1708B_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1708B_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = VT1708B_DIGIN_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1708B_loopbacks[] = {
+ { 0x16, HDA_INPUT, 1 },
+ { 0x16, HDA_INPUT, 2 },
+ { 0x16, HDA_INPUT, 3 },
+ { 0x16, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+static int patch_vt1708B_8ch(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1708B_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1708B_8ch_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1708B Analog";
+ spec->stream_analog_playback = &vt1708B_8ch_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1708B_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1708B Digital";
+ spec->stream_digital_playback = &vt1708B_pcm_digital_playback;
+ spec->stream_digital_capture = &vt1708B_pcm_digital_capture;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1708B_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
+ spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1708B_loopbacks;
+#endif
+
+ return 0;
+}
+
+static int patch_vt1708B_4ch(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1708B_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1708B_4ch_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1708B Analog";
+ spec->stream_analog_playback = &vt1708B_4ch_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1708B_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1708B Digital";
+ spec->stream_digital_playback = &vt1708B_pcm_digital_playback;
+ spec->stream_digital_capture = &vt1708B_pcm_digital_capture;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1708B_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
+ spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1708B_loopbacks;
+#endif
+
+ return 0;
+}
+
+/* Patch for VT1708S */
+
+/* VT1708S software backdoor based override for buggy hardware micboost
+ * setting */
+#define MIC_BOOST_VOLUME(xname, nid) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = 0, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+ .info = mic_boost_volume_info, \
+ .get = snd_hda_mixer_amp_volume_get, \
+ .put = snd_hda_mixer_amp_volume_put, \
+ .tlv = { .c = mic_boost_tlv }, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) }
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1708S_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
+ MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A),
+ MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb vt1708S_volume_init_verbs[] = {
+ /* Unmute ADC0-1 and set the default input to mic-in */
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the
+ * analog-loopback mixer widget */
+ /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ /* Setup default input of PW4 to MW0 */
+ {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* PW9, PW10 Output enable */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Enable Mic Boost Volume backdoor */
+ {0x1, 0xf98, 0x1},
+ { }
+};
+
+static struct hda_verb vt1708S_uniwill_init_verbs[] = {
+ {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ { }
+};
+
+static struct hda_pcm_stream vt1708S_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 8,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_pcm_prepare,
+ .cleanup = via_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708S_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x13, /* NID to query formats and rates */
+ .ops = {
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1708S_pcm_digital_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare
+ },
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1708S_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int i;
+ hda_nid_t nid;
+
+ spec->multiout.num_dacs = cfg->line_outs;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ for (i = 0; i < 4; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ /* config dac list */
+ switch (i) {
+ case AUTO_SEQ_FRONT:
+ spec->multiout.dac_nids[i] = 0x10;
+ break;
+ case AUTO_SEQ_CENLFE:
+ spec->multiout.dac_nids[i] = 0x24;
+ break;
+ case AUTO_SEQ_SURROUND:
+ spec->multiout.dac_nids[i] = 0x11;
+ break;
+ case AUTO_SEQ_SIDE:
+ spec->multiout.dac_nids[i] = 0x25;
+ break;
+ }
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+ hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25};
+ hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27};
+ hda_nid_t nid, nid_vol, nid_mute;
+ int i, err;
+
+ for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
+ nid = cfg->line_out_pins[i];
+
+ if (!nid)
+ continue;
+
+ nid_vol = nid_vols[i];
+ nid_mute = nid_mutes[i];
+
+ if (i == AUTO_SEQ_CENLFE) {
+ /* Center/LFE */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_mute,
+ 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_mute,
+ 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_FRONT) {
+ /* add control to mixer index 0 */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0,
+ HDA_INPUT));
+ if (err < 0)
+ return err;
+
+ /* Front */
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_mute,
+ 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_mute,
+ 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = 5;
+ imux->num_items++;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x1a: /* Mic */
+ idx = 2;
+ break;
+
+ case 0x1b: /* Line In */
+ idx = 3;
+ break;
+
+ case 0x1e: /* Front Mic */
+ idx = 4;
+ break;
+
+ case 0x1f: /* CD */
+ idx = 1;
+ break;
+ }
+ err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
+ idx, 0x16);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx-1;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static int vt1708S_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t vt1708s_ignore[] = {0x21, 0};
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ vt1708s_ignore);
+ if (err < 0)
+ return err;
+ err = vt1708S_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1708S_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID;
+
+ spec->extra_dig_out_nid = 0x15;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1708S_loopbacks[] = {
+ { 0x16, HDA_INPUT, 1 },
+ { 0x16, HDA_INPUT, 2 },
+ { 0x16, HDA_INPUT, 3 },
+ { 0x16, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+static int patch_vt1708S(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1708S_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1708S Analog";
+ spec->stream_analog_playback = &vt1708S_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1708S_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1708S Digital";
+ spec->stream_digital_playback = &vt1708S_pcm_digital_playback;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1708S_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids);
+ spec->mixers[spec->num_mixers] = vt1708S_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1708S_loopbacks;
+#endif
+
+ return 0;
+}
+
+/* Patch for VT1702 */
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1702_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x20, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x1F, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x1F, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Digital Mic Boost Capture Volume", 0x1E, 0x0,
+ HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 1,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb vt1702_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1F, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
+ */
+ /* Amp Indices: Mic1 = 1, Line = 1, Mic2 = 3 */
+ {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* Setup default input of PW4 to MW0 */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* PW6 PW7 Output enable */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ { }
+};
+
+static struct hda_verb vt1702_uniwill_init_verbs[] = {
+ {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT},
+ {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ { }
+};
+
+static struct hda_pcm_stream vt1702_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1702_pcm_analog_capture = {
+ .substreams = 3,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x12, /* NID to query formats and rates */
+ .ops = {
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1702_pcm_digital_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare
+ },
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1702_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ if (cfg->line_out_pins[0]) {
+ /* config dac list */
+ spec->multiout.dac_nids[0] = 0x10;
+ }
+
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1702_auto_create_line_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int err;
+
+ if (!cfg->line_out_pins[0])
+ return -1;
+
+ /* add control to mixer index 0 */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+
+ /* Front */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = 0x1D;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x1D, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = 3;
+ imux->num_items++;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x14: /* Mic */
+ idx = 1;
+ break;
+
+ case 0x15: /* Line In */
+ idx = 2;
+ break;
+
+ case 0x18: /* Front Mic */
+ idx = 3;
+ break;
+ }
+ err = via_new_analog_input(spec, cfg->input_pins[i],
+ labels[i], idx, 0x1A);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx-1;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static int vt1702_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+ static hda_nid_t vt1702_ignore[] = {0x1C, 0};
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+ vt1702_ignore);
+ if (err < 0)
+ return err;
+ err = vt1702_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1702_auto_create_line_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = VT1702_DIGOUT_NID;
+
+ spec->extra_dig_out_nid = 0x1B;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1702_loopbacks[] = {
+ { 0x1A, HDA_INPUT, 1 },
+ { 0x1A, HDA_INPUT, 2 },
+ { 0x1A, HDA_INPUT, 3 },
+ { 0x1A, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+static int patch_vt1702(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+ unsigned int response;
+ unsigned char control;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1702_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1702_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1702_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1702 Analog";
+ spec->stream_analog_playback = &vt1702_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1702_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1702 Digital";
+ spec->stream_digital_playback = &vt1702_pcm_digital_playback;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1702_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids);
+ spec->mixers[spec->num_mixers] = vt1702_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1702_loopbacks;
+#endif
+
+ /* Open backdoor */
+ response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0);
+ control = (unsigned char)(response & 0xff);
+ control |= 0x3;
+ snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control);
+
+ /* Enable GPIO 0&1 for volume&mute control */
+ /* Enable GPIO 2 for DMIC-DATA */
+ response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0);
+ control = (unsigned char)((response >> 16) & 0x3f);
+ snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control);
+
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_via[] = {
+ { .id = 0x11061708, .name = "VIA VT1708", .patch = patch_vt1708},
+ { .id = 0x11061709, .name = "VIA VT1708", .patch = patch_vt1708},
+ { .id = 0x1106170A, .name = "VIA VT1708", .patch = patch_vt1708},
+ { .id = 0x1106170B, .name = "VIA VT1708", .patch = patch_vt1708},
+ { .id = 0x1106E710, .name = "VIA VT1709 10-Ch",
+ .patch = patch_vt1709_10ch},
+ { .id = 0x1106E711, .name = "VIA VT1709 10-Ch",
+ .patch = patch_vt1709_10ch},
+ { .id = 0x1106E712, .name = "VIA VT1709 10-Ch",
+ .patch = patch_vt1709_10ch},
+ { .id = 0x1106E713, .name = "VIA VT1709 10-Ch",
+ .patch = patch_vt1709_10ch},
+ { .id = 0x1106E714, .name = "VIA VT1709 6-Ch",
+ .patch = patch_vt1709_6ch},
+ { .id = 0x1106E715, .name = "VIA VT1709 6-Ch",
+ .patch = patch_vt1709_6ch},
+ { .id = 0x1106E716, .name = "VIA VT1709 6-Ch",
+ .patch = patch_vt1709_6ch},
+ { .id = 0x1106E717, .name = "VIA VT1709 6-Ch",
+ .patch = patch_vt1709_6ch},
+ { .id = 0x1106E720, .name = "VIA VT1708B 8-Ch",
+ .patch = patch_vt1708B_8ch},
+ { .id = 0x1106E721, .name = "VIA VT1708B 8-Ch",
+ .patch = patch_vt1708B_8ch},
+ { .id = 0x1106E722, .name = "VIA VT1708B 8-Ch",
+ .patch = patch_vt1708B_8ch},
+ { .id = 0x1106E723, .name = "VIA VT1708B 8-Ch",
+ .patch = patch_vt1708B_8ch},
+ { .id = 0x1106E724, .name = "VIA VT1708B 4-Ch",
+ .patch = patch_vt1708B_4ch},
+ { .id = 0x1106E725, .name = "VIA VT1708B 4-Ch",
+ .patch = patch_vt1708B_4ch},
+ { .id = 0x1106E726, .name = "VIA VT1708B 4-Ch",
+ .patch = patch_vt1708B_4ch},
+ { .id = 0x1106E727, .name = "VIA VT1708B 4-Ch",
+ .patch = patch_vt1708B_4ch},
+ { .id = 0x11060397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11061397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11062397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11063397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11064397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11065397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11066397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11067397, .name = "VIA VT1708S",
+ .patch = patch_vt1708S},
+ { .id = 0x11060398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ { .id = 0x11061398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ { .id = 0x11062398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ { .id = 0x11063398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ { .id = 0x11064398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ { .id = 0x11065398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ { .id = 0x11066398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ { .id = 0x11067398, .name = "VIA VT1702",
+ .patch = patch_vt1702},
+ {} /* terminator */
+};
diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile
new file mode 100644
index 0000000..f99fe08
--- /dev/null
+++ b/sound/pci/ice1712/Makefile
@@ -0,0 +1,12 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ice17xx-ak4xxx-objs := ak4xxx.o
+snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o
+snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o
+obj-$(CONFIG_SND_ICE1724) += snd-ice1724.o snd-ice17xx-ak4xxx.o
diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c
new file mode 100644
index 0000000..03391da
--- /dev/null
+++ b/sound/pci/ice1712/ak4xxx.c
@@ -0,0 +1,194 @@
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * AK4524 / AK4528 / AK4529 / AK4355 / AK4381 interface
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include "ice1712.h"
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("ICEnsemble ICE17xx <-> AK4xxx AD/DA chip interface");
+MODULE_LICENSE("GPL");
+
+static void snd_ice1712_akm4xxx_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ snd_ice1712_save_gpio_status(ice);
+}
+
+static void snd_ice1712_akm4xxx_unlock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ snd_ice1712_restore_gpio_status(ice);
+}
+
+/*
+ * write AK4xxx register
+ */
+static void snd_ice1712_akm4xxx_write(struct snd_akm4xxx *ak, int chip,
+ unsigned char addr, unsigned char data)
+{
+ unsigned int tmp;
+ int idx;
+ unsigned int addrdata;
+ struct snd_ak4xxx_private *priv = (void *)ak->private_value[0];
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ if (snd_BUG_ON(chip < 0 || chip >= 4))
+ return;
+
+ tmp = snd_ice1712_gpio_read(ice);
+ tmp |= priv->add_flags;
+ tmp &= ~priv->mask_flags;
+ if (priv->cs_mask == priv->cs_addr) {
+ if (priv->cif) {
+ tmp |= priv->cs_mask; /* start without chip select */
+ } else {
+ tmp &= ~priv->cs_mask; /* chip select low */
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+ } else {
+ /* doesn't handle cf=1 yet */
+ tmp &= ~priv->cs_mask;
+ tmp |= priv->cs_addr;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+
+ /* build I2C address + data byte */
+ addrdata = (priv->caddr << 6) | 0x20 | (addr & 0x1f);
+ addrdata = (addrdata << 8) | data;
+ for (idx = 15; idx >= 0; idx--) {
+ /* drop clock */
+ tmp &= ~priv->clk_mask;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ /* set data */
+ if (addrdata & (1 << idx))
+ tmp |= priv->data_mask;
+ else
+ tmp &= ~priv->data_mask;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ /* raise clock */
+ tmp |= priv->clk_mask;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+
+ if (priv->cs_mask == priv->cs_addr) {
+ if (priv->cif) {
+ /* assert a cs pulse to trigger */
+ tmp &= ~priv->cs_mask;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+ tmp |= priv->cs_mask; /* chip select high to trigger */
+ } else {
+ tmp &= ~priv->cs_mask;
+ tmp |= priv->cs_none; /* deselect address */
+ }
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+}
+
+/*
+ * initialize the struct snd_akm4xxx record with the template
+ */
+int snd_ice1712_akm4xxx_init(struct snd_akm4xxx *ak, const struct snd_akm4xxx *temp,
+ const struct snd_ak4xxx_private *_priv, struct snd_ice1712 *ice)
+{
+ struct snd_ak4xxx_private *priv;
+
+ if (_priv != NULL) {
+ priv = kmalloc(sizeof(*priv), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+ *priv = *_priv;
+ } else {
+ priv = NULL;
+ }
+ *ak = *temp;
+ ak->card = ice->card;
+ ak->private_value[0] = (unsigned long)priv;
+ ak->private_data[0] = ice;
+ if (ak->ops.lock == NULL)
+ ak->ops.lock = snd_ice1712_akm4xxx_lock;
+ if (ak->ops.unlock == NULL)
+ ak->ops.unlock = snd_ice1712_akm4xxx_unlock;
+ if (ak->ops.write == NULL)
+ ak->ops.write = snd_ice1712_akm4xxx_write;
+ snd_akm4xxx_init(ak);
+ return 0;
+}
+
+void snd_ice1712_akm4xxx_free(struct snd_ice1712 *ice)
+{
+ unsigned int akidx;
+ if (ice->akm == NULL)
+ return;
+ for (akidx = 0; akidx < ice->akm_codecs; akidx++) {
+ struct snd_akm4xxx *ak = &ice->akm[akidx];
+ kfree((void*)ak->private_value[0]);
+ }
+ kfree(ice->akm);
+}
+
+/*
+ * build AK4xxx controls
+ */
+int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice)
+{
+ unsigned int akidx;
+ int err;
+
+ for (akidx = 0; akidx < ice->akm_codecs; akidx++) {
+ struct snd_akm4xxx *ak = &ice->akm[akidx];
+ err = snd_akm4xxx_build_controls(ak);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int __init alsa_ice1712_akm4xxx_module_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_ice1712_akm4xxx_module_exit(void)
+{
+}
+
+module_init(alsa_ice1712_akm4xxx_module_init)
+module_exit(alsa_ice1712_akm4xxx_module_exit)
+
+EXPORT_SYMBOL(snd_ice1712_akm4xxx_init);
+EXPORT_SYMBOL(snd_ice1712_akm4xxx_free);
+EXPORT_SYMBOL(snd_ice1712_akm4xxx_build_controls);
diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
new file mode 100644
index 0000000..3756430
--- /dev/null
+++ b/sound/pci/ice1712/amp.c
@@ -0,0 +1,94 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Advanced Micro Peripherals Ltd AUDIO2000
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "amp.h"
+
+static void wm_put(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ unsigned short cval;
+ cval = (reg << 9) | val;
+ snd_vt1724_write_i2c(ice, WM_DEV, cval >> 8, cval & 0xff);
+}
+
+static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice)
+{
+ static const unsigned short wm_inits[] = {
+ WM_ATTEN_L, 0x0000, /* 0 db */
+ WM_ATTEN_R, 0x0000, /* 0 db */
+ WM_DAC_CTRL, 0x0008, /* 24bit I2S */
+ WM_INT_CTRL, 0x0001, /* 24bit I2S */
+ };
+
+ unsigned int i;
+
+ /* only use basic functionality for now */
+
+ ice->num_total_dacs = 2; /* only PSDOUT0 is connected */
+ ice->num_total_adcs = 2;
+
+ /* Chaintech AV-710 has another codecs, which need initialization */
+ /* initialize WM8728 codec */
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) {
+ for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
+ wm_put(ice, wm_inits[i], wm_inits[i+1]);
+ }
+
+ return 0;
+}
+
+static int __devinit snd_vt1724_amp_add_controls(struct snd_ice1712 *ice)
+{
+ /* we use pins 39 and 41 of the VT1616 for left and right read outputs */
+ snd_ac97_write_cache(ice->ac97, 0x5a, snd_ac97_read(ice->ac97, 0x5a) & ~0x8000);
+ return 0;
+}
+
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1724_amp_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_AV710,
+ .name = "Chaintech AV-710",
+ .model = "av710",
+ .chip_init = snd_vt1724_amp_init,
+ .build_controls = snd_vt1724_amp_add_controls,
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_AUDIO2000,
+ .name = "AMP Ltd AUDIO2000",
+ .model = "amp2000",
+ .chip_init = snd_vt1724_amp_init,
+ .build_controls = snd_vt1724_amp_add_controls,
+ },
+ { } /* terminator */
+};
+
diff --git a/sound/pci/ice1712/amp.h b/sound/pci/ice1712/amp.h
new file mode 100644
index 0000000..bf81d30
--- /dev/null
+++ b/sound/pci/ice1712/amp.h
@@ -0,0 +1,48 @@
+#ifndef __SOUND_AMP_H
+#define __SOUND_AMP_H
+
+/*
+ * ALSA driver for VIA VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Advanced Micro Peripherals Ltd AUDIO2000
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define AMP_AUDIO2000_DEVICE_DESC "{AMP Ltd,AUDIO2000},"\
+ "{Chaintech,AV-710},"
+
+#if 0
+#define VT1724_SUBDEVICE_AUDIO2000 0x12142417 /* Advanced Micro Peripherals Ltd AUDIO2000 */
+#else
+#define VT1724_SUBDEVICE_AUDIO2000 0x00030003 /* a dummy ID for AMP Audio2000 */
+#endif
+#define VT1724_SUBDEVICE_AV710 0x12142417 /* AV710 - the same ID with Audio2000! */
+
+/* WM8728 on I2C for AV710 */
+#define WM_DEV 0x36
+
+#define WM_ATTEN_L 0x00
+#define WM_ATTEN_R 0x01
+#define WM_DAC_CTRL 0x02
+#define WM_INT_CTRL 0x03
+
+extern struct snd_ice1712_card_info snd_vt1724_amp_cards[];
+
+
+#endif /* __SOUND_AMP_H */
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
new file mode 100644
index 0000000..110d16e
--- /dev/null
+++ b/sound/pci/ice1712/aureon.c
@@ -0,0 +1,2284 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Terratec Aureon cards
+ *
+ * Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ * NOTES:
+ *
+ * - we reuse the struct snd_akm4xxx record for storing the wm8770 codec data.
+ * both wm and akm codecs are pretty similar, so we can integrate
+ * both controls in the future, once if wm codecs are reused in
+ * many boards.
+ *
+ * - DAC digital volumes are not implemented in the mixer.
+ * if they show better response than DAC analog volumes, we can use them
+ * instead.
+ *
+ * Lowlevel functions for AudioTrak Prodigy 7.1 (and possibly 192) cards
+ * Copyright (c) 2003 Dimitromanolakis Apostolos <apostol@cs.utoronto.ca>
+ *
+ * version 0.82: Stable / not all features work yet (no communication with AC97 secondary)
+ * added 64x/128x oversampling switch (should be 64x only for 96khz)
+ * fixed some recording labels (still need to check the rest)
+ * recording is working probably thanks to correct wm8770 initialization
+ *
+ * version 0.5: Initial release:
+ * working: analog output, mixer, headphone amplifier switch
+ * not working: prety much everything else, at least i could verify that
+ * we have no digital output, no capture, pretty bad clicks and poops
+ * on mixer switch and other coll stuff.
+ */
+
+#include <linux/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "aureon.h"
+#include <sound/tlv.h>
+
+/* AC97 register cache for Aureon */
+struct aureon_spec {
+ unsigned short stac9744[64];
+ unsigned int cs8415_mux;
+ unsigned short master[2];
+ unsigned short vol[8];
+ unsigned char pca9554_out;
+};
+
+/* WM8770 registers */
+#define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */
+#define WM_DAC_MASTER_ATTEN 0x08 /* DAC master analog attenuation */
+#define WM_DAC_DIG_ATTEN 0x09 /* DAC1-8 digital attenuation */
+#define WM_DAC_DIG_MASTER_ATTEN 0x11 /* DAC master digital attenuation */
+#define WM_PHASE_SWAP 0x12 /* DAC phase */
+#define WM_DAC_CTRL1 0x13 /* DAC control bits */
+#define WM_MUTE 0x14 /* mute controls */
+#define WM_DAC_CTRL2 0x15 /* de-emphasis and zefo-flag */
+#define WM_INT_CTRL 0x16 /* interface control */
+#define WM_MASTER 0x17 /* master clock and mode */
+#define WM_POWERDOWN 0x18 /* power-down controls */
+#define WM_ADC_GAIN 0x19 /* ADC gain L(19)/R(1a) */
+#define WM_ADC_MUX 0x1b /* input MUX */
+#define WM_OUT_MUX1 0x1c /* output MUX */
+#define WM_OUT_MUX2 0x1e /* output MUX */
+#define WM_RESET 0x1f /* software reset */
+
+/* CS8415A registers */
+#define CS8415_CTRL1 0x01
+#define CS8415_CTRL2 0x02
+#define CS8415_QSUB 0x14
+#define CS8415_RATIO 0x1E
+#define CS8415_C_BUFFER 0x20
+#define CS8415_ID 0x7F
+
+/* PCA9554 registers */
+#define PCA9554_DEV 0x40 /* I2C device address */
+#define PCA9554_IN 0x00 /* input port */
+#define PCA9554_OUT 0x01 /* output port */
+#define PCA9554_INVERT 0x02 /* input invert */
+#define PCA9554_DIR 0x03 /* port directions */
+
+/*
+ * Aureon Universe additional controls using PCA9554
+ */
+
+/*
+ * Send data to pca9554
+ */
+static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg,
+ unsigned char data)
+{
+ unsigned int tmp;
+ int i, j;
+ unsigned char dev = PCA9554_DEV; /* ID 0100000, write */
+ unsigned char val = 0;
+
+ tmp = snd_ice1712_gpio_read(ice);
+
+ snd_ice1712_gpio_set_mask(ice, ~(AUREON_SPI_MOSI|AUREON_SPI_CLK|
+ AUREON_WM_RW|AUREON_WM_CS|
+ AUREON_CS8415_CS));
+ tmp |= AUREON_WM_RW;
+ tmp |= AUREON_CS8415_CS | AUREON_WM_CS; /* disable SPI devices */
+
+ tmp &= ~AUREON_SPI_MOSI;
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(50);
+
+ /*
+ * send i2c stop condition and start condition
+ * to obtain sane state
+ */
+ tmp |= AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(50);
+ tmp |= AUREON_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(100);
+ tmp &= ~AUREON_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(50);
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(100);
+ /*
+ * send device address, command and value,
+ * skipping ack cycles inbetween
+ */
+ for (j = 0; j < 3; j++) {
+ switch (j) {
+ case 0:
+ val = dev;
+ break;
+ case 1:
+ val = reg;
+ break;
+ case 2:
+ val = data;
+ break;
+ }
+ for (i = 7; i >= 0; i--) {
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ if (val & (1 << i))
+ tmp |= AUREON_SPI_MOSI;
+ else
+ tmp &= ~AUREON_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ tmp |= AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ }
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ tmp |= AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ }
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ tmp &= ~AUREON_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(40);
+ tmp |= AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(50);
+ tmp |= AUREON_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(100);
+}
+
+static int aureon_universe_inmux_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ char *texts[3] = {"Internal Aux", "Wavetable", "Rear Line-In"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int aureon_universe_inmux_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ ucontrol->value.enumerated.item[0] = spec->pca9554_out;
+ return 0;
+}
+
+static int aureon_universe_inmux_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ unsigned char oval, nval;
+ int change;
+
+ nval = ucontrol->value.enumerated.item[0];
+ if (nval >= 3)
+ return -EINVAL;
+ snd_ice1712_save_gpio_status(ice);
+ oval = spec->pca9554_out;
+ change = (oval != nval);
+ if (change) {
+ aureon_pca9554_write(ice, PCA9554_OUT, nval);
+ spec->pca9554_out = nval;
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+
+static void aureon_ac97_write(struct snd_ice1712 *ice, unsigned short reg,
+ unsigned short val)
+{
+ struct aureon_spec *spec = ice->spec;
+ unsigned int tmp;
+
+ /* Send address to XILINX chip */
+ tmp = (snd_ice1712_gpio_read(ice) & ~0xFF) | (reg & 0x7F);
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+ tmp |= AUREON_AC97_ADDR;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+ tmp &= ~AUREON_AC97_ADDR;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+
+ /* Send low-order byte to XILINX chip */
+ tmp &= ~AUREON_AC97_DATA_MASK;
+ tmp |= val & AUREON_AC97_DATA_MASK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+ tmp |= AUREON_AC97_DATA_LOW;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+ tmp &= ~AUREON_AC97_DATA_LOW;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+
+ /* Send high-order byte to XILINX chip */
+ tmp &= ~AUREON_AC97_DATA_MASK;
+ tmp |= (val >> 8) & AUREON_AC97_DATA_MASK;
+
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+ tmp |= AUREON_AC97_DATA_HIGH;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+ tmp &= ~AUREON_AC97_DATA_HIGH;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+
+ /* Instruct XILINX chip to parse the data to the STAC9744 chip */
+ tmp |= AUREON_AC97_COMMIT;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+ tmp &= ~AUREON_AC97_COMMIT;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(10);
+
+ /* Store the data in out private buffer */
+ spec->stac9744[(reg & 0x7F) >> 1] = val;
+}
+
+static unsigned short aureon_ac97_read(struct snd_ice1712 *ice, unsigned short reg)
+{
+ struct aureon_spec *spec = ice->spec;
+ return spec->stac9744[(reg & 0x7F) >> 1];
+}
+
+/*
+ * Initialize STAC9744 chip
+ */
+static int aureon_ac97_init(struct snd_ice1712 *ice)
+{
+ struct aureon_spec *spec = ice->spec;
+ int i;
+ static const unsigned short ac97_defaults[] = {
+ 0x00, 0x9640,
+ 0x02, 0x8000,
+ 0x04, 0x8000,
+ 0x06, 0x8000,
+ 0x0C, 0x8008,
+ 0x0E, 0x8008,
+ 0x10, 0x8808,
+ 0x12, 0x8808,
+ 0x14, 0x8808,
+ 0x16, 0x8808,
+ 0x18, 0x8808,
+ 0x1C, 0x8000,
+ 0x26, 0x000F,
+ 0x28, 0x0201,
+ 0x2C, 0xBB80,
+ 0x32, 0xBB80,
+ 0x7C, 0x8384,
+ 0x7E, 0x7644,
+ (unsigned short)-1
+ };
+ unsigned int tmp;
+
+ /* Cold reset */
+ tmp = (snd_ice1712_gpio_read(ice) | AUREON_AC97_RESET) & ~AUREON_AC97_DATA_MASK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(3);
+
+ tmp &= ~AUREON_AC97_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(3);
+
+ tmp |= AUREON_AC97_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(3);
+
+ memset(&spec->stac9744, 0, sizeof(spec->stac9744));
+ for (i = 0; ac97_defaults[i] != (unsigned short)-1; i += 2)
+ spec->stac9744[(ac97_defaults[i]) >> 1] = ac97_defaults[i+1];
+
+ /* Unmute AC'97 master volume permanently - muting is done by WM8770 */
+ aureon_ac97_write(ice, AC97_MASTER, 0x0000);
+
+ return 0;
+}
+
+#define AUREON_AC97_STEREO 0x80
+
+/*
+ * AC'97 volume controls
+ */
+static int aureon_ac97_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = kcontrol->private_value & AUREON_AC97_STEREO ? 2 : 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 31;
+ return 0;
+}
+
+static int aureon_ac97_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short vol;
+
+ mutex_lock(&ice->gpio_mutex);
+
+ vol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F);
+ ucontrol->value.integer.value[0] = 0x1F - (vol & 0x1F);
+ if (kcontrol->private_value & AUREON_AC97_STEREO)
+ ucontrol->value.integer.value[1] = 0x1F - ((vol >> 8) & 0x1F);
+
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int aureon_ac97_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+
+ ovol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F);
+ nvol = (0x1F - ucontrol->value.integer.value[0]) & 0x001F;
+ if (kcontrol->private_value & AUREON_AC97_STEREO)
+ nvol |= ((0x1F - ucontrol->value.integer.value[1]) << 8) & 0x1F00;
+ nvol |= ovol & ~0x1F1F;
+
+ change = (ovol != nvol);
+ if (change)
+ aureon_ac97_write(ice, kcontrol->private_value & 0x7F, nvol);
+
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * AC'97 mute controls
+ */
+#define aureon_ac97_mute_info snd_ctl_boolean_mono_info
+
+static int aureon_ac97_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+
+ ucontrol->value.integer.value[0] = aureon_ac97_read(ice,
+ kcontrol->private_value & 0x7F) & 0x8000 ? 0 : 1;
+
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int aureon_ac97_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+
+ ovol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F);
+ nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x8000) | (ovol & ~0x8000);
+
+ change = (ovol != nvol);
+ if (change)
+ aureon_ac97_write(ice, kcontrol->private_value & 0x7F, nvol);
+
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * AC'97 mute controls
+ */
+#define aureon_ac97_micboost_info snd_ctl_boolean_mono_info
+
+static int aureon_ac97_micboost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+
+ ucontrol->value.integer.value[0] = aureon_ac97_read(ice, AC97_MIC) & 0x0020 ? 0 : 1;
+
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int aureon_ac97_micboost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+
+ ovol = aureon_ac97_read(ice, AC97_MIC);
+ nvol = (ucontrol->value.integer.value[0] ? 0x0000 : 0x0020) | (ovol & ~0x0020);
+
+ change = (ovol != nvol);
+ if (change)
+ aureon_ac97_write(ice, AC97_MIC, nvol);
+
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * write data in the SPI mode
+ */
+static void aureon_spi_write(struct snd_ice1712 *ice, unsigned int cs, unsigned int data, int bits)
+{
+ unsigned int tmp;
+ int i;
+ unsigned int mosi, clk;
+
+ tmp = snd_ice1712_gpio_read(ice);
+
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT ||
+ ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71XT) {
+ snd_ice1712_gpio_set_mask(ice, ~(PRODIGY_SPI_MOSI|PRODIGY_SPI_CLK|PRODIGY_WM_CS));
+ mosi = PRODIGY_SPI_MOSI;
+ clk = PRODIGY_SPI_CLK;
+ } else {
+ snd_ice1712_gpio_set_mask(ice, ~(AUREON_WM_RW|AUREON_SPI_MOSI|AUREON_SPI_CLK|
+ AUREON_WM_CS|AUREON_CS8415_CS));
+ mosi = AUREON_SPI_MOSI;
+ clk = AUREON_SPI_CLK;
+
+ tmp |= AUREON_WM_RW;
+ }
+
+ tmp &= ~cs;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ for (i = bits - 1; i >= 0; i--) {
+ tmp &= ~clk;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ if (data & (1 << i))
+ tmp |= mosi;
+ else
+ tmp &= ~mosi;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= clk;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+
+ tmp &= ~clk;
+ tmp |= cs;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= clk;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+}
+
+/*
+ * Read data in SPI mode
+ */
+static void aureon_spi_read(struct snd_ice1712 *ice, unsigned int cs,
+ unsigned int data, int bits, unsigned char *buffer, int size)
+{
+ int i, j;
+ unsigned int tmp;
+
+ tmp = (snd_ice1712_gpio_read(ice) & ~AUREON_SPI_CLK) | AUREON_CS8415_CS|AUREON_WM_CS;
+ snd_ice1712_gpio_write(ice, tmp);
+ tmp &= ~cs;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ for (i = bits-1; i >= 0; i--) {
+ if (data & (1 << i))
+ tmp |= AUREON_SPI_MOSI;
+ else
+ tmp &= ~AUREON_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ tmp |= AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+
+ for (j = 0; j < size; j++) {
+ unsigned char outdata = 0;
+ for (i = 7; i >= 0; i--) {
+ tmp = snd_ice1712_gpio_read(ice);
+ outdata <<= 1;
+ outdata |= (tmp & AUREON_SPI_MISO) ? 1 : 0;
+ udelay(1);
+
+ tmp |= AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ tmp &= ~AUREON_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+ buffer[j] = outdata;
+ }
+
+ tmp |= cs;
+ snd_ice1712_gpio_write(ice, tmp);
+}
+
+static unsigned char aureon_cs8415_get(struct snd_ice1712 *ice, int reg)
+{
+ unsigned char val;
+ aureon_spi_write(ice, AUREON_CS8415_CS, 0x2000 | reg, 16);
+ aureon_spi_read(ice, AUREON_CS8415_CS, 0x21, 8, &val, 1);
+ return val;
+}
+
+static void aureon_cs8415_read(struct snd_ice1712 *ice, int reg,
+ unsigned char *buffer, int size)
+{
+ aureon_spi_write(ice, AUREON_CS8415_CS, 0x2000 | reg, 16);
+ aureon_spi_read(ice, AUREON_CS8415_CS, 0x21, 8, buffer, size);
+}
+
+static void aureon_cs8415_put(struct snd_ice1712 *ice, int reg,
+ unsigned char val)
+{
+ aureon_spi_write(ice, AUREON_CS8415_CS, 0x200000 | (reg << 8) | val, 24);
+}
+
+/*
+ * get the current register value of WM codec
+ */
+static unsigned short wm_get(struct snd_ice1712 *ice, int reg)
+{
+ reg <<= 1;
+ return ((unsigned short)ice->akm[0].images[reg] << 8) |
+ ice->akm[0].images[reg + 1];
+}
+
+/*
+ * set the register value of WM codec
+ */
+static void wm_put_nocache(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ aureon_spi_write(ice,
+ ((ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT ||
+ ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71XT) ?
+ PRODIGY_WM_CS : AUREON_WM_CS),
+ (reg << 9) | (val & 0x1ff), 16);
+}
+
+/*
+ * set the register value of WM codec and remember it
+ */
+static void wm_put(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ wm_put_nocache(ice, reg, val);
+ reg <<= 1;
+ ice->akm[0].images[reg] = val >> 8;
+ ice->akm[0].images[reg + 1] = val;
+}
+
+/*
+ */
+#define aureon_mono_bool_info snd_ctl_boolean_mono_info
+
+/*
+ * AC'97 master playback mute controls (Mute on WM8770 chip)
+ */
+#define aureon_ac97_mmute_info snd_ctl_boolean_mono_info
+
+static int aureon_ac97_mmute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_OUT_MUX1) >> 1) & 0x01;
+
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+
+ ovol = wm_get(ice, WM_OUT_MUX1);
+ nvol = (ovol & ~0x02) | (ucontrol->value.integer.value[0] ? 0x02 : 0x00);
+ change = (ovol != nvol);
+ if (change)
+ wm_put(ice, WM_OUT_MUX1, nvol);
+
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0);
+
+/*
+ * Logarithmic volume values for WM8770
+ * Computed as 20 * Log10(255 / x)
+ */
+static const unsigned char wm_vol[256] = {
+ 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23,
+ 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17,
+ 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13,
+ 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11,
+ 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8,
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6,
+ 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+ 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3,
+ 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+ 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0
+};
+
+#define WM_VOL_MAX (sizeof(wm_vol) - 1)
+#define WM_VOL_MUTE 0x8000
+
+static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master)
+{
+ unsigned char nvol;
+
+ if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+ nvol = 0;
+ else
+ nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX];
+
+ wm_put(ice, index, nvol);
+ wm_put_nocache(ice, index, 0x180 | nvol);
+}
+
+/*
+ * DAC mute control
+ */
+#define wm_pcm_mute_info snd_ctl_boolean_mono_info
+
+static int wm_pcm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_MUTE) & 0x10) ? 0 : 1;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_pcm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short nval, oval;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+ oval = wm_get(ice, WM_MUTE);
+ nval = (oval & ~0x10) | (ucontrol->value.integer.value[0] ? 0 : 0x10);
+ change = (oval != nval);
+ if (change)
+ wm_put(ice, WM_MUTE, nval);
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * Master volume attenuation mixer control
+ */
+static int wm_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = WM_VOL_MAX;
+ return 0;
+}
+
+static int wm_master_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ int i;
+ for (i = 0; i < 2; i++)
+ ucontrol->value.integer.value[i] =
+ spec->master[i] & ~WM_VOL_MUTE;
+ return 0;
+}
+
+static int wm_master_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ int ch, change = 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (ch = 0; ch < 2; ch++) {
+ unsigned int vol = ucontrol->value.integer.value[ch];
+ if (vol > WM_VOL_MAX)
+ continue;
+ vol |= spec->master[ch] & WM_VOL_MUTE;
+ if (vol != spec->master[ch]) {
+ int dac;
+ spec->master[ch] = vol;
+ for (dac = 0; dac < ice->num_total_dacs; dac += 2)
+ wm_set_vol(ice, WM_DAC_ATTEN + dac + ch,
+ spec->vol[dac + ch],
+ spec->master[ch]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * DAC volume attenuation mixer control
+ */
+static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int voices = kcontrol->private_value >> 8;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = voices;
+ uinfo->value.integer.min = 0; /* mute (-101dB) */
+ uinfo->value.integer.max = 0x7F; /* 0dB */
+ return 0;
+}
+
+static int wm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ int i, ofs, voices;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ for (i = 0; i < voices; i++)
+ ucontrol->value.integer.value[i] =
+ spec->vol[ofs+i] & ~WM_VOL_MUTE;
+ return 0;
+}
+
+static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ int i, idx, ofs, voices;
+ int change = 0;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < voices; i++) {
+ unsigned int vol = ucontrol->value.integer.value[i];
+ if (vol > 0x7f)
+ continue;
+ vol |= spec->vol[ofs+i];
+ if (vol != spec->vol[ofs+i]) {
+ spec->vol[ofs+i] = vol;
+ idx = WM_DAC_ATTEN + ofs + i;
+ wm_set_vol(ice, idx, spec->vol[ofs + i],
+ spec->master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * WM8770 mute control
+ */
+static int wm_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = kcontrol->private_value >> 8;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int wm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ int voices, ofs, i;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xFF;
+
+ for (i = 0; i < voices; i++)
+ ucontrol->value.integer.value[i] =
+ (spec->vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1;
+ return 0;
+}
+
+static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ int change = 0, voices, ofs, i;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xFF;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < voices; i++) {
+ int val = (spec->vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1;
+ if (ucontrol->value.integer.value[i] != val) {
+ spec->vol[ofs + i] &= ~WM_VOL_MUTE;
+ spec->vol[ofs + i] |=
+ ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE;
+ wm_set_vol(ice, ofs + i, spec->vol[ofs + i],
+ spec->master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * WM8770 master mute control
+ */
+#define wm_master_mute_info snd_ctl_boolean_stereo_info
+
+static int wm_master_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+
+ ucontrol->value.integer.value[0] =
+ (spec->master[0] & WM_VOL_MUTE) ? 0 : 1;
+ ucontrol->value.integer.value[1] =
+ (spec->master[1] & WM_VOL_MUTE) ? 0 : 1;
+ return 0;
+}
+
+static int wm_master_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ int change = 0, i;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < 2; i++) {
+ int val = (spec->master[i] & WM_VOL_MUTE) ? 0 : 1;
+ if (ucontrol->value.integer.value[i] != val) {
+ int dac;
+ spec->master[i] &= ~WM_VOL_MUTE;
+ spec->master[i] |=
+ ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE;
+ for (dac = 0; dac < ice->num_total_dacs; dac += 2)
+ wm_set_vol(ice, WM_DAC_ATTEN + dac + i,
+ spec->vol[dac + i],
+ spec->master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/* digital master volume */
+#define PCM_0dB 0xff
+#define PCM_RES 128 /* -64dB */
+#define PCM_MIN (PCM_0dB - PCM_RES)
+static int wm_pcm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0; /* mute (-64dB) */
+ uinfo->value.integer.max = PCM_RES; /* 0dB */
+ return 0;
+}
+
+static int wm_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ mutex_lock(&ice->gpio_mutex);
+ val = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff;
+ val = val > PCM_MIN ? (val - PCM_MIN) : 0;
+ ucontrol->value.integer.value[0] = val;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int change = 0;
+
+ nvol = ucontrol->value.integer.value[0];
+ if (nvol > PCM_RES)
+ return -EINVAL;
+ snd_ice1712_save_gpio_status(ice);
+ nvol = (nvol ? (nvol + PCM_MIN) : 0) & 0xff;
+ ovol = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff;
+ if (ovol != nvol) {
+ wm_put(ice, WM_DAC_DIG_MASTER_ATTEN, nvol); /* prelatch */
+ wm_put_nocache(ice, WM_DAC_DIG_MASTER_ATTEN, nvol | 0x100); /* update */
+ change = 1;
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * ADC mute control
+ */
+#define wm_adc_mute_info snd_ctl_boolean_stereo_info
+
+static int wm_adc_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+ int i;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ val = wm_get(ice, WM_ADC_GAIN + i);
+ ucontrol->value.integer.value[i] = ~val>>5 & 0x1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short new, old;
+ int i, change = 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < 2; i++) {
+ old = wm_get(ice, WM_ADC_GAIN + i);
+ new = (~ucontrol->value.integer.value[i]<<5&0x20) | (old&~0x20);
+ if (new != old) {
+ wm_put(ice, WM_ADC_GAIN + i, new);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * ADC gain mixer control
+ */
+static int wm_adc_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* -12dB */
+ uinfo->value.integer.max = 0x1f; /* 19dB */
+ return 0;
+}
+
+static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int i, idx;
+ unsigned short vol;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ idx = WM_ADC_GAIN + i;
+ vol = wm_get(ice, idx) & 0x1f;
+ ucontrol->value.integer.value[i] = vol;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int i, idx;
+ unsigned short ovol, nvol;
+ int change = 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < 2; i++) {
+ idx = WM_ADC_GAIN + i;
+ nvol = ucontrol->value.integer.value[i] & 0x1f;
+ ovol = wm_get(ice, idx);
+ if ((ovol & 0x1f) != nvol) {
+ wm_put(ice, idx, nvol | (ovol & ~0x1f));
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * ADC input mux mixer control
+ */
+static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "CD", /* AIN1 */
+ "Aux", /* AIN2 */
+ "Line", /* AIN3 */
+ "Mic", /* AIN4 */
+ "AC97" /* AIN5 */
+ };
+ static const char * const universe_texts[] = {
+ "Aux1", /* AIN1 */
+ "CD", /* AIN2 */
+ "Phono", /* AIN3 */
+ "Line", /* AIN4 */
+ "Aux2", /* AIN5 */
+ "Mic", /* AIN6 */
+ "Aux3", /* AIN7 */
+ "AC97" /* AIN8 */
+ };
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 2;
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE) {
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, universe_texts[uinfo->value.enumerated.item]);
+ } else {
+ uinfo->value.enumerated.items = 5;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ }
+ return 0;
+}
+
+static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ mutex_lock(&ice->gpio_mutex);
+ val = wm_get(ice, WM_ADC_MUX);
+ ucontrol->value.enumerated.item[0] = val & 7;
+ ucontrol->value.enumerated.item[1] = (val >> 4) & 7;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short oval, nval;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+ oval = wm_get(ice, WM_ADC_MUX);
+ nval = oval & ~0x77;
+ nval |= ucontrol->value.enumerated.item[0] & 7;
+ nval |= (ucontrol->value.enumerated.item[1] & 7) << 4;
+ change = (oval != nval);
+ if (change)
+ wm_put(ice, WM_ADC_MUX, nval);
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * CS8415 Input mux
+ */
+static int aureon_cs8415_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ static const char * const aureon_texts[] = {
+ "CD", /* RXP0 */
+ "Optical" /* RXP1 */
+ };
+ static const char * const prodigy_texts[] = {
+ "CD",
+ "Coax"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71)
+ strcpy(uinfo->value.enumerated.name, prodigy_texts[uinfo->value.enumerated.item]);
+ else
+ strcpy(uinfo->value.enumerated.name, aureon_texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+
+ /* snd_ice1712_save_gpio_status(ice); */
+ /* val = aureon_cs8415_get(ice, CS8415_CTRL2); */
+ ucontrol->value.enumerated.item[0] = spec->cs8415_mux;
+ /* snd_ice1712_restore_gpio_status(ice); */
+ return 0;
+}
+
+static int aureon_cs8415_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct aureon_spec *spec = ice->spec;
+ unsigned short oval, nval;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+ oval = aureon_cs8415_get(ice, CS8415_CTRL2);
+ nval = oval & ~0x07;
+ nval |= ucontrol->value.enumerated.item[0] & 7;
+ change = (oval != nval);
+ if (change)
+ aureon_cs8415_put(ice, CS8415_CTRL2, nval);
+ snd_ice1712_restore_gpio_status(ice);
+ spec->cs8415_mux = ucontrol->value.enumerated.item[0];
+ return change;
+}
+
+static int aureon_cs8415_rate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 192000;
+ return 0;
+}
+
+static int aureon_cs8415_rate_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char ratio;
+ ratio = aureon_cs8415_get(ice, CS8415_RATIO);
+ ucontrol->value.integer.value[0] = (int)((unsigned int)ratio * 750);
+ return 0;
+}
+
+/*
+ * CS8415A Mute
+ */
+#define aureon_cs8415_mute_info snd_ctl_boolean_mono_info
+
+static int aureon_cs8415_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ snd_ice1712_save_gpio_status(ice);
+ ucontrol->value.integer.value[0] = (aureon_cs8415_get(ice, CS8415_CTRL1) & 0x20) ? 0 : 1;
+ snd_ice1712_restore_gpio_status(ice);
+ return 0;
+}
+
+static int aureon_cs8415_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char oval, nval;
+ int change;
+ snd_ice1712_save_gpio_status(ice);
+ oval = aureon_cs8415_get(ice, CS8415_CTRL1);
+ if (ucontrol->value.integer.value[0])
+ nval = oval & ~0x20;
+ else
+ nval = oval | 0x20;
+ change = (oval != nval);
+ if (change)
+ aureon_cs8415_put(ice, CS8415_CTRL1, nval);
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * CS8415A Q-Sub info
+ */
+static int aureon_cs8415_qsub_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = 10;
+ return 0;
+}
+
+static int aureon_cs8415_qsub_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ snd_ice1712_save_gpio_status(ice);
+ aureon_cs8415_read(ice, CS8415_QSUB, ucontrol->value.bytes.data, 10);
+ snd_ice1712_restore_gpio_status(ice);
+
+ return 0;
+}
+
+static int aureon_cs8415_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int aureon_cs8415_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ memset(ucontrol->value.iec958.status, 0xFF, 24);
+ return 0;
+}
+
+static int aureon_cs8415_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ snd_ice1712_save_gpio_status(ice);
+ aureon_cs8415_read(ice, CS8415_C_BUFFER, ucontrol->value.iec958.status, 24);
+ snd_ice1712_restore_gpio_status(ice);
+ return 0;
+}
+
+/*
+ * Headphone Amplifier
+ */
+static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable)
+{
+ unsigned int tmp, tmp2;
+
+ tmp2 = tmp = snd_ice1712_gpio_read(ice);
+ if (enable)
+ if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT &&
+ ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT)
+ tmp |= AUREON_HP_SEL;
+ else
+ tmp |= PRODIGY_HP_SEL;
+ else
+ if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT &&
+ ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT)
+ tmp &= ~AUREON_HP_SEL;
+ else
+ tmp &= ~PRODIGY_HP_SEL;
+ if (tmp != tmp2) {
+ snd_ice1712_gpio_write(ice, tmp);
+ return 1;
+ }
+ return 0;
+}
+
+static int aureon_get_headphone_amp(struct snd_ice1712 *ice)
+{
+ unsigned int tmp = snd_ice1712_gpio_read(ice);
+
+ return (tmp & AUREON_HP_SEL) != 0;
+}
+
+#define aureon_hpamp_info snd_ctl_boolean_mono_info
+
+static int aureon_hpamp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = aureon_get_headphone_amp(ice);
+ return 0;
+}
+
+
+static int aureon_hpamp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ return aureon_set_headphone_amp(ice, ucontrol->value.integer.value[0]);
+}
+
+/*
+ * Deemphasis
+ */
+
+#define aureon_deemp_info snd_ctl_boolean_mono_info
+
+static int aureon_deemp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL2) & 0xf) == 0xf;
+ return 0;
+}
+
+static int aureon_deemp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int temp, temp2;
+ temp2 = temp = wm_get(ice, WM_DAC_CTRL2);
+ if (ucontrol->value.integer.value[0])
+ temp |= 0xf;
+ else
+ temp &= ~0xf;
+ if (temp != temp2) {
+ wm_put(ice, WM_DAC_CTRL2, temp);
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * ADC Oversampling
+ */
+static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[2] = { "128x", "64x" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = (wm_get(ice, WM_MASTER) & 0x8) == 0x8;
+ return 0;
+}
+
+static int aureon_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ int temp, temp2;
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ temp2 = temp = wm_get(ice, WM_MASTER);
+
+ if (ucontrol->value.enumerated.item[0])
+ temp |= 0x8;
+ else
+ temp &= ~0x8;
+
+ if (temp != temp2) {
+ wm_put(ice, WM_MASTER, temp);
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * mixers
+ */
+
+static struct snd_kcontrol_new aureon_dac_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = wm_master_mute_info,
+ .get = wm_master_mute_get,
+ .put = wm_master_mute_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Master Playback Volume",
+ .info = wm_master_vol_info,
+ .get = wm_master_vol_get,
+ .put = wm_master_vol_put,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Front Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 0
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Front Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 0,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Rear Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 2
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Rear Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 2,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Center Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (1 << 8) | 4
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Center Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (1 << 8) | 4,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "LFE Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (1 << 8) | 5
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "LFE Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (1 << 8) | 5,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Side Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 6
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Side Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 6,
+ .tlv = { .p = db_scale_wm_dac }
+ }
+};
+
+static struct snd_kcontrol_new wm_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .info = wm_pcm_mute_info,
+ .get = wm_pcm_mute_get,
+ .put = wm_pcm_mute_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "PCM Playback Volume",
+ .info = wm_pcm_vol_info,
+ .get = wm_pcm_vol_get,
+ .put = wm_pcm_vol_put,
+ .tlv = { .p = db_scale_wm_pcm }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Switch",
+ .info = wm_adc_mute_info,
+ .get = wm_adc_mute_get,
+ .put = wm_adc_mute_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Capture Volume",
+ .info = wm_adc_vol_info,
+ .get = wm_adc_vol_get,
+ .put = wm_adc_vol_put,
+ .tlv = { .p = db_scale_wm_adc }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = wm_adc_mux_info,
+ .get = wm_adc_mux_get,
+ .put = wm_adc_mux_put,
+ .private_value = 5
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "External Amplifier",
+ .info = aureon_hpamp_info,
+ .get = aureon_hpamp_get,
+ .put = aureon_hpamp_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Deemphasis Switch",
+ .info = aureon_deemp_info,
+ .get = aureon_deemp_get,
+ .put = aureon_deemp_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Oversampling",
+ .info = aureon_oversampling_info,
+ .get = aureon_oversampling_get,
+ .put = aureon_oversampling_put
+ }
+};
+
+static struct snd_kcontrol_new ac97_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "AC97 Playback Switch",
+ .info = aureon_ac97_mmute_info,
+ .get = aureon_ac97_mmute_get,
+ .put = aureon_ac97_mmute_put,
+ .private_value = AC97_MASTER
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "AC97 Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_MASTER|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_master }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "CD Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_CD
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "CD Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_CD|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Aux Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_AUX,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Aux Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_AUX|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_LINE
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Line Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_LINE|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_MIC
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Mic Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_MIC,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Boost (+20dB)",
+ .info = aureon_ac97_micboost_info,
+ .get = aureon_ac97_micboost_get,
+ .put = aureon_ac97_micboost_put
+ }
+};
+
+static struct snd_kcontrol_new universe_ac97_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "AC97 Playback Switch",
+ .info = aureon_ac97_mmute_info,
+ .get = aureon_ac97_mmute_get,
+ .put = aureon_ac97_mmute_put,
+ .private_value = AC97_MASTER
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "AC97 Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_MASTER|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_master }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "CD Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_AUX
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "CD Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_AUX|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Phono Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_CD
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Phono Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_CD|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_LINE
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Line Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_LINE|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_MIC
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Mic Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_MIC,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Boost (+20dB)",
+ .info = aureon_ac97_micboost_info,
+ .get = aureon_ac97_micboost_get,
+ .put = aureon_ac97_micboost_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Aux Playback Switch",
+ .info = aureon_ac97_mute_info,
+ .get = aureon_ac97_mute_get,
+ .put = aureon_ac97_mute_put,
+ .private_value = AC97_VIDEO,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Aux Playback Volume",
+ .info = aureon_ac97_vol_info,
+ .get = aureon_ac97_vol_get,
+ .put = aureon_ac97_vol_put,
+ .private_value = AC97_VIDEO|AUREON_AC97_STEREO,
+ .tlv = { .p = db_scale_ac97_gain }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Aux Source",
+ .info = aureon_universe_inmux_info,
+ .get = aureon_universe_inmux_get,
+ .put = aureon_universe_inmux_put
+ }
+
+};
+
+static struct snd_kcontrol_new cs8415_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, SWITCH),
+ .info = aureon_cs8415_mute_info,
+ .get = aureon_cs8415_mute_get,
+ .put = aureon_cs8415_mute_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE) "Source",
+ .info = aureon_cs8415_mux_info,
+ .get = aureon_cs8415_mux_get,
+ .put = aureon_cs8415_mux_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("Q-subcode ", CAPTURE, DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = aureon_cs8415_qsub_info,
+ .get = aureon_cs8415_qsub_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = aureon_cs8415_spdif_info,
+ .get = aureon_cs8415_mask_get
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = aureon_cs8415_spdif_info,
+ .get = aureon_cs8415_spdif_get
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, NONE) "Rate",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = aureon_cs8415_rate_info,
+ .get = aureon_cs8415_rate_get
+ }
+};
+
+static int __devinit aureon_add_controls(struct snd_ice1712 *ice)
+{
+ unsigned int i, counts;
+ int err;
+
+ counts = ARRAY_SIZE(aureon_dac_controls);
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY)
+ counts -= 2; /* no side */
+ for (i = 0; i < counts; i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&aureon_dac_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm_controls); i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&wm_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE) {
+ for (i = 0; i < ARRAY_SIZE(universe_ac97_controls); i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&universe_ac97_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+ } else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT &&
+ ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) {
+ for (i = 0; i < ARRAY_SIZE(ac97_controls); i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&ac97_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT &&
+ ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) {
+ unsigned char id;
+ snd_ice1712_save_gpio_status(ice);
+ id = aureon_cs8415_get(ice, CS8415_ID);
+ if (id != 0x41)
+ snd_printk(KERN_INFO "No CS8415 chip. Skipping CS8415 controls.\n");
+ else if ((id & 0x0F) != 0x01)
+ snd_printk(KERN_INFO "Detected unsupported CS8415 rev. (%c)\n", (char)((id & 0x0F) + 'A' - 1));
+ else {
+ for (i = 0; i < ARRAY_SIZE(cs8415_controls); i++) {
+ struct snd_kcontrol *kctl;
+ err = snd_ctl_add(ice->card, (kctl = snd_ctl_new1(&cs8415_controls[i], ice)));
+ if (err < 0)
+ return err;
+ if (i > 1)
+ kctl->id.device = ice->pcm->device;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ }
+
+ return 0;
+}
+
+
+/*
+ * initialize the chip
+ */
+static int __devinit aureon_init(struct snd_ice1712 *ice)
+{
+ static const unsigned short wm_inits_aureon[] = {
+ /* These come first to reduce init pop noise */
+ 0x1b, 0x044, /* ADC Mux (AC'97 source) */
+ 0x1c, 0x00B, /* Out Mux1 (VOUT1 = DAC+AUX, VOUT2 = DAC) */
+ 0x1d, 0x009, /* Out Mux2 (VOUT2 = DAC, VOUT3 = DAC) */
+
+ 0x18, 0x000, /* All power-up */
+
+ 0x16, 0x122, /* I2S, normal polarity, 24bit */
+ 0x17, 0x022, /* 256fs, slave mode */
+ 0x00, 0, /* DAC1 analog mute */
+ 0x01, 0, /* DAC2 analog mute */
+ 0x02, 0, /* DAC3 analog mute */
+ 0x03, 0, /* DAC4 analog mute */
+ 0x04, 0, /* DAC5 analog mute */
+ 0x05, 0, /* DAC6 analog mute */
+ 0x06, 0, /* DAC7 analog mute */
+ 0x07, 0, /* DAC8 analog mute */
+ 0x08, 0x100, /* master analog mute */
+ 0x09, 0xff, /* DAC1 digital full */
+ 0x0a, 0xff, /* DAC2 digital full */
+ 0x0b, 0xff, /* DAC3 digital full */
+ 0x0c, 0xff, /* DAC4 digital full */
+ 0x0d, 0xff, /* DAC5 digital full */
+ 0x0e, 0xff, /* DAC6 digital full */
+ 0x0f, 0xff, /* DAC7 digital full */
+ 0x10, 0xff, /* DAC8 digital full */
+ 0x11, 0x1ff, /* master digital full */
+ 0x12, 0x000, /* phase normal */
+ 0x13, 0x090, /* unmute DAC L/R */
+ 0x14, 0x000, /* all unmute */
+ 0x15, 0x000, /* no deemphasis, no ZFLG */
+ 0x19, 0x000, /* -12dB ADC/L */
+ 0x1a, 0x000, /* -12dB ADC/R */
+ (unsigned short)-1
+ };
+ static const unsigned short wm_inits_prodigy[] = {
+
+ /* These come first to reduce init pop noise */
+ 0x1b, 0x000, /* ADC Mux */
+ 0x1c, 0x009, /* Out Mux1 */
+ 0x1d, 0x009, /* Out Mux2 */
+
+ 0x18, 0x000, /* All power-up */
+
+ 0x16, 0x022, /* I2S, normal polarity, 24bit, high-pass on */
+ 0x17, 0x006, /* 128fs, slave mode */
+
+ 0x00, 0, /* DAC1 analog mute */
+ 0x01, 0, /* DAC2 analog mute */
+ 0x02, 0, /* DAC3 analog mute */
+ 0x03, 0, /* DAC4 analog mute */
+ 0x04, 0, /* DAC5 analog mute */
+ 0x05, 0, /* DAC6 analog mute */
+ 0x06, 0, /* DAC7 analog mute */
+ 0x07, 0, /* DAC8 analog mute */
+ 0x08, 0x100, /* master analog mute */
+
+ 0x09, 0x7f, /* DAC1 digital full */
+ 0x0a, 0x7f, /* DAC2 digital full */
+ 0x0b, 0x7f, /* DAC3 digital full */
+ 0x0c, 0x7f, /* DAC4 digital full */
+ 0x0d, 0x7f, /* DAC5 digital full */
+ 0x0e, 0x7f, /* DAC6 digital full */
+ 0x0f, 0x7f, /* DAC7 digital full */
+ 0x10, 0x7f, /* DAC8 digital full */
+ 0x11, 0x1FF, /* master digital full */
+
+ 0x12, 0x000, /* phase normal */
+ 0x13, 0x090, /* unmute DAC L/R */
+ 0x14, 0x000, /* all unmute */
+ 0x15, 0x000, /* no deemphasis, no ZFLG */
+
+ 0x19, 0x000, /* -12dB ADC/L */
+ 0x1a, 0x000, /* -12dB ADC/R */
+ (unsigned short)-1
+
+ };
+ static const unsigned short cs_inits[] = {
+ 0x0441, /* RUN */
+ 0x0180, /* no mute, OMCK output on RMCK pin */
+ 0x0201, /* S/PDIF source on RXP1 */
+ 0x0605, /* slave, 24bit, MSB on second OSCLK, SDOUT for right channel when OLRCK is high */
+ (unsigned short)-1
+ };
+ struct aureon_spec *spec;
+ unsigned int tmp;
+ const unsigned short *p;
+ int err, i;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) {
+ ice->num_total_dacs = 6;
+ ice->num_total_adcs = 2;
+ } else {
+ /* aureon 7.1 and prodigy 7.1 */
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 2;
+ }
+
+ /* to remeber the register values of CS8415 */
+ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (!ice->akm)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ err = aureon_ac97_init(ice);
+ if (err != 0)
+ return err;
+
+ snd_ice1712_gpio_set_dir(ice, 0x5fffff); /* fix this for the time being */
+
+ /* reset the wm codec as the SPI mode */
+ snd_ice1712_save_gpio_status(ice);
+ snd_ice1712_gpio_set_mask(ice, ~(AUREON_WM_RESET|AUREON_WM_CS|AUREON_CS8415_CS|AUREON_HP_SEL));
+
+ tmp = snd_ice1712_gpio_read(ice);
+ tmp &= ~AUREON_WM_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= AUREON_WM_CS | AUREON_CS8415_CS;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= AUREON_WM_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ /* initialize WM8770 codec */
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71 ||
+ ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT ||
+ ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71XT)
+ p = wm_inits_prodigy;
+ else
+ p = wm_inits_aureon;
+ for (; *p != (unsigned short)-1; p += 2)
+ wm_put(ice, p[0], p[1]);
+
+ /* initialize CS8415A codec */
+ if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT &&
+ ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71XT) {
+ for (p = cs_inits; *p != (unsigned short)-1; p++)
+ aureon_spi_write(ice, AUREON_CS8415_CS, *p | 0x200000, 24);
+ spec->cs8415_mux = 1;
+
+ aureon_set_headphone_amp(ice, 1);
+ }
+
+ snd_ice1712_restore_gpio_status(ice);
+
+ /* initialize PCA9554 pin directions & set default input */
+ aureon_pca9554_write(ice, PCA9554_DIR, 0x00);
+ aureon_pca9554_write(ice, PCA9554_OUT, 0x00); /* internal AUX */
+
+ spec->master[0] = WM_VOL_MUTE;
+ spec->master[1] = WM_VOL_MUTE;
+ for (i = 0; i < ice->num_total_dacs; i++) {
+ spec->vol[i] = WM_VOL_MUTE;
+ wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]);
+ }
+
+ return 0;
+}
+
+
+/*
+ * Aureon boards don't provide the EEPROM data except for the vendor IDs.
+ * hence the driver needs to sets up it properly.
+ */
+
+static unsigned char aureon51_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x0a, /* clock 512, spdif-in/ADC, 3DACs */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xfc, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x5f,
+ [ICE_EEP2_GPIO_MASK] = 0x00,
+ [ICE_EEP2_GPIO_MASK1] = 0x00,
+ [ICE_EEP2_GPIO_MASK2] = 0x00,
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+
+static unsigned char aureon71_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x0b, /* clock 512, spdif-in/ADC, 4DACs */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xfc, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x5f,
+ [ICE_EEP2_GPIO_MASK] = 0x00,
+ [ICE_EEP2_GPIO_MASK1] = 0x00,
+ [ICE_EEP2_GPIO_MASK2] = 0x00,
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+#define prodigy71_eeprom aureon71_eeprom
+
+static unsigned char aureon71_universe_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401, spdif-in/ADC,
+ * 4DACs
+ */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xfc, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x5f,
+ [ICE_EEP2_GPIO_MASK] = 0x00,
+ [ICE_EEP2_GPIO_MASK1] = 0x00,
+ [ICE_EEP2_GPIO_MASK2] = 0x00,
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+
+static unsigned char prodigy71lt_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x4b, /* clock 384, spdif-in/ADC, 4DACs */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xfc, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x5f,
+ [ICE_EEP2_GPIO_MASK] = 0x00,
+ [ICE_EEP2_GPIO_MASK1] = 0x00,
+ [ICE_EEP2_GPIO_MASK2] = 0x00,
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+#define prodigy71xt_eeprom prodigy71lt_eeprom
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1724_aureon_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_AUREON51_SKY,
+ .name = "Terratec Aureon 5.1-Sky",
+ .model = "aureon51",
+ .chip_init = aureon_init,
+ .build_controls = aureon_add_controls,
+ .eeprom_size = sizeof(aureon51_eeprom),
+ .eeprom_data = aureon51_eeprom,
+ .driver = "Aureon51",
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_AUREON71_SPACE,
+ .name = "Terratec Aureon 7.1-Space",
+ .model = "aureon71",
+ .chip_init = aureon_init,
+ .build_controls = aureon_add_controls,
+ .eeprom_size = sizeof(aureon71_eeprom),
+ .eeprom_data = aureon71_eeprom,
+ .driver = "Aureon71",
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_AUREON71_UNIVERSE,
+ .name = "Terratec Aureon 7.1-Universe",
+ .model = "universe",
+ .chip_init = aureon_init,
+ .build_controls = aureon_add_controls,
+ .eeprom_size = sizeof(aureon71_universe_eeprom),
+ .eeprom_data = aureon71_universe_eeprom,
+ .driver = "Aureon71Univ", /* keep in 15 letters */
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_PRODIGY71,
+ .name = "Audiotrak Prodigy 7.1",
+ .model = "prodigy71",
+ .chip_init = aureon_init,
+ .build_controls = aureon_add_controls,
+ .eeprom_size = sizeof(prodigy71_eeprom),
+ .eeprom_data = prodigy71_eeprom,
+ .driver = "Prodigy71", /* should be identical with Aureon71 */
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_PRODIGY71LT,
+ .name = "Audiotrak Prodigy 7.1 LT",
+ .model = "prodigy71lt",
+ .chip_init = aureon_init,
+ .build_controls = aureon_add_controls,
+ .eeprom_size = sizeof(prodigy71lt_eeprom),
+ .eeprom_data = prodigy71lt_eeprom,
+ .driver = "Prodigy71LT",
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_PRODIGY71XT,
+ .name = "Audiotrak Prodigy 7.1 XT",
+ .model = "prodigy71xt",
+ .chip_init = aureon_init,
+ .build_controls = aureon_add_controls,
+ .eeprom_size = sizeof(prodigy71xt_eeprom),
+ .eeprom_data = prodigy71xt_eeprom,
+ .driver = "Prodigy71LT",
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/aureon.h b/sound/pci/ice1712/aureon.h
new file mode 100644
index 0000000..c253b8e
--- /dev/null
+++ b/sound/pci/ice1712/aureon.h
@@ -0,0 +1,65 @@
+#ifndef __SOUND_AUREON_H
+#define __SOUND_AUREON_H
+
+/*
+ * ALSA driver for VIA VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Terratec Aureon cards
+ *
+ * Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define AUREON_DEVICE_DESC "{Terratec,Aureon 5.1 Sky},"\
+ "{Terratec,Aureon 7.1 Space},"\
+ "{Terratec,Aureon 7.1 Universe}," \
+ "{AudioTrak,Prodigy 7.1}," \
+ "{AudioTrak,Prodigy 7.1 LT},"\
+ "{AudioTrak,Prodigy 7.1 XT},"
+
+#define VT1724_SUBDEVICE_AUREON51_SKY 0x3b154711 /* Aureon 5.1 Sky */
+#define VT1724_SUBDEVICE_AUREON71_SPACE 0x3b154511 /* Aureon 7.1 Space */
+#define VT1724_SUBDEVICE_AUREON71_UNIVERSE 0x3b155311 /* Aureon 7.1 Universe */
+#define VT1724_SUBDEVICE_PRODIGY71 0x33495345 /* PRODIGY 7.1 */
+#define VT1724_SUBDEVICE_PRODIGY71LT 0x32315441 /* PRODIGY 7.1 LT */
+#define VT1724_SUBDEVICE_PRODIGY71XT 0x36315441 /* PRODIGY 7.1 XT*/
+
+extern struct snd_ice1712_card_info snd_vt1724_aureon_cards[];
+
+/* GPIO bits */
+#define AUREON_CS8415_CS (1 << 22)
+#define AUREON_SPI_MISO (1 << 21)
+#define AUREON_WM_RESET (1 << 20)
+#define AUREON_SPI_CLK (1 << 19)
+#define AUREON_SPI_MOSI (1 << 18)
+#define AUREON_WM_RW (1 << 17)
+#define AUREON_AC97_RESET (1 << 16)
+#define AUREON_DIGITAL_SEL1 (1 << 15)
+#define AUREON_HP_SEL (1 << 14)
+#define AUREON_WM_CS (1 << 12)
+#define AUREON_AC97_COMMIT (1 << 11)
+#define AUREON_AC97_ADDR (1 << 10)
+#define AUREON_AC97_DATA_LOW (1 << 9)
+#define AUREON_AC97_DATA_HIGH (1 << 8)
+#define AUREON_AC97_DATA_MASK 0xFF
+
+#define PRODIGY_WM_CS (1 << 8)
+#define PRODIGY_SPI_MOSI (1 << 10)
+#define PRODIGY_SPI_CLK (1 << 9)
+#define PRODIGY_HP_SEL (1 << 5)
+
+#endif /* __SOUND_AUREON_H */
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
new file mode 100644
index 0000000..d216362
--- /dev/null
+++ b/sound/pci/ice1712/delta.c
@@ -0,0 +1,817 @@
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for M-Audio Delta 1010, 1010E, 44, 66, 66E, Dio2496,
+ * Audiophile, Digigram VX442
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/cs8427.h>
+#include <sound/asoundef.h>
+
+#include "ice1712.h"
+#include "delta.h"
+
+#define SND_CS8403
+#include <sound/cs8403.h>
+
+
+/*
+ * CS8427 via SPI mode (for Audiophile), emulated I2C
+ */
+
+/* send 8 bits */
+static void ap_cs8427_write_byte(struct snd_ice1712 *ice, unsigned char data, unsigned char tmp)
+{
+ int idx;
+
+ for (idx = 7; idx >= 0; idx--) {
+ tmp &= ~(ICE1712_DELTA_AP_DOUT|ICE1712_DELTA_AP_CCLK);
+ if (data & (1 << idx))
+ tmp |= ICE1712_DELTA_AP_DOUT;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+ tmp |= ICE1712_DELTA_AP_CCLK;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+ }
+}
+
+/* read 8 bits */
+static unsigned char ap_cs8427_read_byte(struct snd_ice1712 *ice, unsigned char tmp)
+{
+ unsigned char data = 0;
+ int idx;
+
+ for (idx = 7; idx >= 0; idx--) {
+ tmp &= ~ICE1712_DELTA_AP_CCLK;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+ if (snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA) & ICE1712_DELTA_AP_DIN)
+ data |= 1 << idx;
+ tmp |= ICE1712_DELTA_AP_CCLK;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+ }
+ return data;
+}
+
+/* assert chip select */
+static unsigned char ap_cs8427_codec_select(struct snd_ice1712 *ice)
+{
+ unsigned char tmp;
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010E:
+ case ICE1712_SUBDEVICE_DELTA1010LT:
+ tmp &= ~ICE1712_DELTA_1010LT_CS;
+ tmp |= ICE1712_DELTA_1010LT_CCLK | ICE1712_DELTA_1010LT_CS_CS8427;
+ break;
+ case ICE1712_SUBDEVICE_AUDIOPHILE:
+ case ICE1712_SUBDEVICE_DELTA410:
+ tmp |= ICE1712_DELTA_AP_CCLK | ICE1712_DELTA_AP_CS_CODEC;
+ tmp &= ~ICE1712_DELTA_AP_CS_DIGITAL;
+ break;
+ case ICE1712_SUBDEVICE_VX442:
+ tmp |= ICE1712_VX442_CCLK | ICE1712_VX442_CODEC_CHIP_A | ICE1712_VX442_CODEC_CHIP_B;
+ tmp &= ~ICE1712_VX442_CS_DIGITAL;
+ break;
+ }
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+ return tmp;
+}
+
+/* deassert chip select */
+static void ap_cs8427_codec_deassert(struct snd_ice1712 *ice, unsigned char tmp)
+{
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010E:
+ case ICE1712_SUBDEVICE_DELTA1010LT:
+ tmp &= ~ICE1712_DELTA_1010LT_CS;
+ tmp |= ICE1712_DELTA_1010LT_CS_NONE;
+ break;
+ case ICE1712_SUBDEVICE_AUDIOPHILE:
+ case ICE1712_SUBDEVICE_DELTA410:
+ tmp |= ICE1712_DELTA_AP_CS_DIGITAL;
+ break;
+ case ICE1712_SUBDEVICE_VX442:
+ tmp |= ICE1712_VX442_CS_DIGITAL;
+ break;
+ }
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+}
+
+/* sequential write */
+static int ap_cs8427_sendbytes(struct snd_i2c_device *device, unsigned char *bytes, int count)
+{
+ struct snd_ice1712 *ice = device->bus->private_data;
+ int res = count;
+ unsigned char tmp;
+
+ mutex_lock(&ice->gpio_mutex);
+ tmp = ap_cs8427_codec_select(ice);
+ ap_cs8427_write_byte(ice, (device->addr << 1) | 0, tmp); /* address + write mode */
+ while (count-- > 0)
+ ap_cs8427_write_byte(ice, *bytes++, tmp);
+ ap_cs8427_codec_deassert(ice, tmp);
+ mutex_unlock(&ice->gpio_mutex);
+ return res;
+}
+
+/* sequential read */
+static int ap_cs8427_readbytes(struct snd_i2c_device *device, unsigned char *bytes, int count)
+{
+ struct snd_ice1712 *ice = device->bus->private_data;
+ int res = count;
+ unsigned char tmp;
+
+ mutex_lock(&ice->gpio_mutex);
+ tmp = ap_cs8427_codec_select(ice);
+ ap_cs8427_write_byte(ice, (device->addr << 1) | 1, tmp); /* address + read mode */
+ while (count-- > 0)
+ *bytes++ = ap_cs8427_read_byte(ice, tmp);
+ ap_cs8427_codec_deassert(ice, tmp);
+ mutex_unlock(&ice->gpio_mutex);
+ return res;
+}
+
+static int ap_cs8427_probeaddr(struct snd_i2c_bus *bus, unsigned short addr)
+{
+ if (addr == 0x10)
+ return 1;
+ return -ENOENT;
+}
+
+static struct snd_i2c_ops ap_cs8427_i2c_ops = {
+ .sendbytes = ap_cs8427_sendbytes,
+ .readbytes = ap_cs8427_readbytes,
+ .probeaddr = ap_cs8427_probeaddr,
+};
+
+/*
+ */
+
+static void snd_ice1712_delta_cs8403_spdif_write(struct snd_ice1712 *ice, unsigned char bits)
+{
+ unsigned char tmp, mask1, mask2;
+ int idx;
+ /* send byte to transmitter */
+ mask1 = ICE1712_DELTA_SPDIF_OUT_STAT_CLOCK;
+ mask2 = ICE1712_DELTA_SPDIF_OUT_STAT_DATA;
+ mutex_lock(&ice->gpio_mutex);
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+ for (idx = 7; idx >= 0; idx--) {
+ tmp &= ~(mask1 | mask2);
+ if (bits & (1 << idx))
+ tmp |= mask2;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(100);
+ tmp |= mask1;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(100);
+ }
+ tmp &= ~mask1;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+
+static void delta_spdif_default_get(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ snd_cs8403_decode_spdif_bits(&ucontrol->value.iec958, ice->spdif.cs8403_bits);
+}
+
+static int delta_spdif_default_put(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int val;
+ int change;
+
+ val = snd_cs8403_encode_spdif_bits(&ucontrol->value.iec958);
+ spin_lock_irq(&ice->reg_lock);
+ change = ice->spdif.cs8403_bits != val;
+ ice->spdif.cs8403_bits = val;
+ if (change && ice->playback_pro_substream == NULL) {
+ spin_unlock_irq(&ice->reg_lock);
+ snd_ice1712_delta_cs8403_spdif_write(ice, val);
+ } else {
+ spin_unlock_irq(&ice->reg_lock);
+ }
+ return change;
+}
+
+static void delta_spdif_stream_get(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ snd_cs8403_decode_spdif_bits(&ucontrol->value.iec958, ice->spdif.cs8403_stream_bits);
+}
+
+static int delta_spdif_stream_put(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int val;
+ int change;
+
+ val = snd_cs8403_encode_spdif_bits(&ucontrol->value.iec958);
+ spin_lock_irq(&ice->reg_lock);
+ change = ice->spdif.cs8403_stream_bits != val;
+ ice->spdif.cs8403_stream_bits = val;
+ if (change && ice->playback_pro_substream != NULL) {
+ spin_unlock_irq(&ice->reg_lock);
+ snd_ice1712_delta_cs8403_spdif_write(ice, val);
+ } else {
+ spin_unlock_irq(&ice->reg_lock);
+ }
+ return change;
+}
+
+
+/*
+ * AK4524 on Delta 44 and 66 to choose the chip mask
+ */
+static void delta_ak4524_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ak4xxx_private *priv = (void *)ak->private_value[0];
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ snd_ice1712_save_gpio_status(ice);
+ priv->cs_mask =
+ priv->cs_addr = chip == 0 ? ICE1712_DELTA_CODEC_CHIP_A :
+ ICE1712_DELTA_CODEC_CHIP_B;
+}
+
+/*
+ * AK4524 on Delta1010LT to choose the chip address
+ */
+static void delta1010lt_ak4524_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ak4xxx_private *priv = (void *)ak->private_value[0];
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ snd_ice1712_save_gpio_status(ice);
+ priv->cs_mask = ICE1712_DELTA_1010LT_CS;
+ priv->cs_addr = chip << 4;
+}
+
+/*
+ * AK4528 on VX442 to choose the chip mask
+ */
+static void vx442_ak4524_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ak4xxx_private *priv = (void *)ak->private_value[0];
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ snd_ice1712_save_gpio_status(ice);
+ priv->cs_mask =
+ priv->cs_addr = chip == 0 ? ICE1712_VX442_CODEC_CHIP_A :
+ ICE1712_VX442_CODEC_CHIP_B;
+}
+
+/*
+ * change the DFS bit according rate for Delta1010
+ */
+static void delta_1010_set_rate_val(struct snd_ice1712 *ice, unsigned int rate)
+{
+ unsigned char tmp, tmp2;
+
+ if (rate == 0) /* no hint - S/PDIF input is master, simply return */
+ return;
+
+ mutex_lock(&ice->gpio_mutex);
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+ tmp2 = tmp & ~ICE1712_DELTA_DFS;
+ if (rate > 48000)
+ tmp2 |= ICE1712_DELTA_DFS;
+ if (tmp != tmp2)
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp2);
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+/*
+ * change the rate of AK4524 on Delta 44/66, AP, 1010LT
+ */
+static void delta_ak4524_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
+{
+ unsigned char tmp, tmp2;
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ if (rate == 0) /* no hint - S/PDIF input is master, simply return */
+ return;
+
+ /* check before reset ak4524 to avoid unnecessary clicks */
+ mutex_lock(&ice->gpio_mutex);
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+ mutex_unlock(&ice->gpio_mutex);
+ tmp2 = tmp & ~ICE1712_DELTA_DFS;
+ if (rate > 48000)
+ tmp2 |= ICE1712_DELTA_DFS;
+ if (tmp == tmp2)
+ return;
+
+ /* do it again */
+ snd_akm4xxx_reset(ak, 1);
+ mutex_lock(&ice->gpio_mutex);
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA) & ~ICE1712_DELTA_DFS;
+ if (rate > 48000)
+ tmp |= ICE1712_DELTA_DFS;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ mutex_unlock(&ice->gpio_mutex);
+ snd_akm4xxx_reset(ak, 0);
+}
+
+/*
+ * change the rate of AK4524 on VX442
+ */
+static void vx442_ak4524_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
+{
+ unsigned char val;
+
+ val = (rate > 48000) ? 0x65 : 0x60;
+ if (snd_akm4xxx_get(ak, 0, 0x02) != val ||
+ snd_akm4xxx_get(ak, 1, 0x02) != val) {
+ snd_akm4xxx_reset(ak, 1);
+ snd_akm4xxx_write(ak, 0, 0x02, val);
+ snd_akm4xxx_write(ak, 1, 0x02, val);
+ snd_akm4xxx_reset(ak, 0);
+ }
+}
+
+
+/*
+ * SPDIF ops for Delta 1010, Dio, 66
+ */
+
+/* open callback */
+static void delta_open_spdif(struct snd_ice1712 *ice, struct snd_pcm_substream *substream)
+{
+ ice->spdif.cs8403_stream_bits = ice->spdif.cs8403_bits;
+}
+
+/* set up */
+static void delta_setup_spdif(struct snd_ice1712 *ice, int rate)
+{
+ unsigned long flags;
+ unsigned int tmp;
+ int change;
+
+ spin_lock_irqsave(&ice->reg_lock, flags);
+ tmp = ice->spdif.cs8403_stream_bits;
+ if (tmp & 0x01) /* consumer */
+ tmp &= (tmp & 0x01) ? ~0x06 : ~0x18;
+ switch (rate) {
+ case 32000: tmp |= (tmp & 0x01) ? 0x04 : 0x00; break;
+ case 44100: tmp |= (tmp & 0x01) ? 0x00 : 0x10; break;
+ case 48000: tmp |= (tmp & 0x01) ? 0x02 : 0x08; break;
+ default: tmp |= (tmp & 0x01) ? 0x00 : 0x18; break;
+ }
+ change = ice->spdif.cs8403_stream_bits != tmp;
+ ice->spdif.cs8403_stream_bits = tmp;
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+ if (change)
+ snd_ctl_notify(ice->card, SNDRV_CTL_EVENT_MASK_VALUE, &ice->spdif.stream_ctl->id);
+ snd_ice1712_delta_cs8403_spdif_write(ice, tmp);
+}
+
+#define snd_ice1712_delta1010lt_wordclock_status_info \
+ snd_ctl_boolean_mono_info
+
+static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ char reg = 0x10; /* CS8427 receiver error register */
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ if (snd_i2c_sendbytes(ice->cs8427, &reg, 1) != 1)
+ snd_printk(KERN_ERR "unable to send register 0x%x byte to CS8427\n", reg);
+ snd_i2c_readbytes(ice->cs8427, &reg, 1);
+ ucontrol->value.integer.value[0] = (reg & CS8427_UNLOCK) ? 1 : 0;
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_status __devinitdata =
+{
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ),
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Word Clock Status",
+ .info = snd_ice1712_delta1010lt_wordclock_status_info,
+ .get = snd_ice1712_delta1010lt_wordclock_status_get,
+};
+
+/*
+ * initialize the chips on M-Audio cards
+ */
+
+static struct snd_akm4xxx akm_audiophile __devinitdata = {
+ .type = SND_AK4528,
+ .num_adcs = 2,
+ .num_dacs = 2,
+ .ops = {
+ .set_rate_val = delta_ak4524_set_rate_val
+ }
+};
+
+static struct snd_ak4xxx_private akm_audiophile_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0,
+ .data_mask = ICE1712_DELTA_AP_DOUT,
+ .clk_mask = ICE1712_DELTA_AP_CCLK,
+ .cs_mask = ICE1712_DELTA_AP_CS_CODEC,
+ .cs_addr = ICE1712_DELTA_AP_CS_CODEC,
+ .cs_none = 0,
+ .add_flags = ICE1712_DELTA_AP_CS_DIGITAL,
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_delta410 __devinitdata = {
+ .type = SND_AK4529,
+ .num_adcs = 2,
+ .num_dacs = 8,
+ .ops = {
+ .set_rate_val = delta_ak4524_set_rate_val
+ }
+};
+
+static struct snd_ak4xxx_private akm_delta410_priv __devinitdata = {
+ .caddr = 0,
+ .cif = 0,
+ .data_mask = ICE1712_DELTA_AP_DOUT,
+ .clk_mask = ICE1712_DELTA_AP_CCLK,
+ .cs_mask = ICE1712_DELTA_AP_CS_CODEC,
+ .cs_addr = ICE1712_DELTA_AP_CS_CODEC,
+ .cs_none = 0,
+ .add_flags = ICE1712_DELTA_AP_CS_DIGITAL,
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_delta1010lt __devinitdata = {
+ .type = SND_AK4524,
+ .num_adcs = 8,
+ .num_dacs = 8,
+ .ops = {
+ .lock = delta1010lt_ak4524_lock,
+ .set_rate_val = delta_ak4524_set_rate_val
+ }
+};
+
+static struct snd_ak4xxx_private akm_delta1010lt_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0, /* the default level of the CIF pin from AK4524 */
+ .data_mask = ICE1712_DELTA_1010LT_DOUT,
+ .clk_mask = ICE1712_DELTA_1010LT_CCLK,
+ .cs_mask = 0,
+ .cs_addr = 0, /* set later */
+ .cs_none = ICE1712_DELTA_1010LT_CS_NONE,
+ .add_flags = 0,
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_delta44 __devinitdata = {
+ .type = SND_AK4524,
+ .num_adcs = 4,
+ .num_dacs = 4,
+ .ops = {
+ .lock = delta_ak4524_lock,
+ .set_rate_val = delta_ak4524_set_rate_val
+ }
+};
+
+static struct snd_ak4xxx_private akm_delta44_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0, /* the default level of the CIF pin from AK4524 */
+ .data_mask = ICE1712_DELTA_CODEC_SERIAL_DATA,
+ .clk_mask = ICE1712_DELTA_CODEC_SERIAL_CLOCK,
+ .cs_mask = 0,
+ .cs_addr = 0, /* set later */
+ .cs_none = 0,
+ .add_flags = 0,
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_vx442 __devinitdata = {
+ .type = SND_AK4524,
+ .num_adcs = 4,
+ .num_dacs = 4,
+ .ops = {
+ .lock = vx442_ak4524_lock,
+ .set_rate_val = vx442_ak4524_set_rate_val
+ }
+};
+
+static struct snd_ak4xxx_private akm_vx442_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0,
+ .data_mask = ICE1712_VX442_DOUT,
+ .clk_mask = ICE1712_VX442_CCLK,
+ .cs_mask = 0,
+ .cs_addr = 0, /* set later */
+ .cs_none = 0,
+ .add_flags = 0,
+ .mask_flags = 0,
+};
+
+static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
+{
+ int err;
+ struct snd_akm4xxx *ak;
+
+ if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA1010 &&
+ ice->eeprom.gpiodir == 0x7b)
+ ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA1010E;
+
+ if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA66 &&
+ ice->eeprom.gpiodir == 0xfb)
+ ice->eeprom.subvendor = ICE1712_SUBDEVICE_DELTA66E;
+
+ /* determine I2C, DACs and ADCs */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_AUDIOPHILE:
+ ice->num_total_dacs = 2;
+ ice->num_total_adcs = 2;
+ break;
+ case ICE1712_SUBDEVICE_DELTA410:
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 2;
+ break;
+ case ICE1712_SUBDEVICE_DELTA44:
+ case ICE1712_SUBDEVICE_DELTA66:
+ ice->num_total_dacs = ice->omni ? 8 : 4;
+ ice->num_total_adcs = ice->omni ? 8 : 4;
+ break;
+ case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_DELTA1010E:
+ case ICE1712_SUBDEVICE_DELTA1010LT:
+ case ICE1712_SUBDEVICE_MEDIASTATION:
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 8;
+ break;
+ case ICE1712_SUBDEVICE_DELTADIO2496:
+ ice->num_total_dacs = 4; /* two AK4324 codecs */
+ break;
+ case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E: /* omni not suported yet */
+ ice->num_total_dacs = 4;
+ ice->num_total_adcs = 4;
+ break;
+ }
+
+ /* initialize spdif */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_AUDIOPHILE:
+ case ICE1712_SUBDEVICE_DELTA410:
+ case ICE1712_SUBDEVICE_DELTA1010E:
+ case ICE1712_SUBDEVICE_DELTA1010LT:
+ case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E:
+ if ((err = snd_i2c_bus_create(ice->card, "ICE1712 GPIO 1", NULL, &ice->i2c)) < 0) {
+ snd_printk(KERN_ERR "unable to create I2C bus\n");
+ return err;
+ }
+ ice->i2c->private_data = ice;
+ ice->i2c->ops = &ap_cs8427_i2c_ops;
+ if ((err = snd_ice1712_init_cs8427(ice, CS8427_BASE_ADDR)) < 0)
+ return err;
+ break;
+ case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_MEDIASTATION:
+ ice->gpio.set_pro_rate = delta_1010_set_rate_val;
+ break;
+ case ICE1712_SUBDEVICE_DELTADIO2496:
+ ice->gpio.set_pro_rate = delta_1010_set_rate_val;
+ /* fall thru */
+ case ICE1712_SUBDEVICE_DELTA66:
+ ice->spdif.ops.open = delta_open_spdif;
+ ice->spdif.ops.setup_rate = delta_setup_spdif;
+ ice->spdif.ops.default_get = delta_spdif_default_get;
+ ice->spdif.ops.default_put = delta_spdif_default_put;
+ ice->spdif.ops.stream_get = delta_spdif_stream_get;
+ ice->spdif.ops.stream_put = delta_spdif_stream_put;
+ /* Set spdif defaults */
+ snd_ice1712_delta_cs8403_spdif_write(ice, ice->spdif.cs8403_bits);
+ break;
+ }
+
+ /* no analog? */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_DELTA1010E:
+ case ICE1712_SUBDEVICE_DELTADIO2496:
+ case ICE1712_SUBDEVICE_MEDIASTATION:
+ return 0;
+ }
+
+ /* second stage of initialization, analog parts and others */
+ ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (! ak)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_AUDIOPHILE:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_audiophile, &akm_audiophile_priv, ice);
+ break;
+ case ICE1712_SUBDEVICE_DELTA410:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_delta410, &akm_delta410_priv, ice);
+ break;
+ case ICE1712_SUBDEVICE_DELTA1010LT:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_delta1010lt, &akm_delta1010lt_priv, ice);
+ break;
+ case ICE1712_SUBDEVICE_DELTA66:
+ case ICE1712_SUBDEVICE_DELTA44:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_delta44, &akm_delta44_priv, ice);
+ break;
+ case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_vx442, &akm_vx442_priv, ice);
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+
+ return err;
+}
+
+
+/*
+ * additional controls for M-Audio cards
+ */
+
+static struct snd_kcontrol_new snd_ice1712_delta1010_wordclock_select __devinitdata =
+ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Sync", 0, ICE1712_DELTA_WORD_CLOCK_SELECT, 1, 0);
+static struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_select __devinitdata =
+ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Sync", 0, ICE1712_DELTA_1010LT_WORDCLOCK, 0, 0);
+static struct snd_kcontrol_new snd_ice1712_delta1010_wordclock_status __devinitdata =
+ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Status", 0, ICE1712_DELTA_WORD_CLOCK_STATUS, 1, SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE);
+static struct snd_kcontrol_new snd_ice1712_deltadio2496_spdif_in_select __devinitdata =
+ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "IEC958 Input Optical", 0, ICE1712_DELTA_SPDIF_INPUT_SELECT, 0, 0);
+static struct snd_kcontrol_new snd_ice1712_delta_spdif_in_status __devinitdata =
+ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Delta IEC958 Input Status", 0, ICE1712_DELTA_SPDIF_IN_STAT, 1, SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE);
+
+
+static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice)
+{
+ int err;
+
+ /* 1010 and dio specific controls */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_MEDIASTATION:
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010_wordclock_select, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010_wordclock_status, ice));
+ if (err < 0)
+ return err;
+ break;
+ case ICE1712_SUBDEVICE_DELTADIO2496:
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_deltadio2496_spdif_in_select, ice));
+ if (err < 0)
+ return err;
+ break;
+ case ICE1712_SUBDEVICE_DELTA1010E:
+ case ICE1712_SUBDEVICE_DELTA1010LT:
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010lt_wordclock_select, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010lt_wordclock_status, ice));
+ if (err < 0)
+ return err;
+ break;
+ }
+
+ /* normal spdif controls */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_DELTADIO2496:
+ case ICE1712_SUBDEVICE_DELTA66:
+ case ICE1712_SUBDEVICE_MEDIASTATION:
+ err = snd_ice1712_spdif_build_controls(ice);
+ if (err < 0)
+ return err;
+ break;
+ }
+
+ /* spdif status in */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010:
+ case ICE1712_SUBDEVICE_DELTADIO2496:
+ case ICE1712_SUBDEVICE_DELTA66:
+ case ICE1712_SUBDEVICE_MEDIASTATION:
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta_spdif_in_status, ice));
+ if (err < 0)
+ return err;
+ break;
+ }
+
+ /* ak4524 controls */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DELTA1010LT:
+ case ICE1712_SUBDEVICE_AUDIOPHILE:
+ case ICE1712_SUBDEVICE_DELTA410:
+ case ICE1712_SUBDEVICE_DELTA44:
+ case ICE1712_SUBDEVICE_DELTA66:
+ case ICE1712_SUBDEVICE_VX442:
+ case ICE1712_SUBDEVICE_DELTA66E:
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+ break;
+ }
+
+ return 0;
+}
+
+
+/* entry point */
+struct snd_ice1712_card_info snd_ice1712_delta_cards[] __devinitdata = {
+ {
+ .subvendor = ICE1712_SUBDEVICE_DELTA1010,
+ .name = "M Audio Delta 1010",
+ .model = "delta1010",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_DELTADIO2496,
+ .name = "M Audio Delta DiO 2496",
+ .model = "dio2496",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ .no_mpu401 = 1,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_DELTA66,
+ .name = "M Audio Delta 66",
+ .model = "delta66",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ .no_mpu401 = 1,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_DELTA44,
+ .name = "M Audio Delta 44",
+ .model = "delta44",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ .no_mpu401 = 1,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_AUDIOPHILE,
+ .name = "M Audio Audiophile 24/96",
+ .model = "audiophile",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_DELTA410,
+ .name = "M Audio Delta 410",
+ .model = "delta410",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_DELTA1010LT,
+ .name = "M Audio Delta 1010LT",
+ .model = "delta1010lt",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_VX442,
+ .name = "Digigram VX442",
+ .model = "vx442",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ .no_mpu401 = 1,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_MEDIASTATION,
+ .name = "Lionstracs Mediastation",
+ .model = "mediastation",
+ .chip_init = snd_ice1712_delta_init,
+ .build_controls = snd_ice1712_delta_add_controls,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h
new file mode 100644
index 0000000..f7f14df
--- /dev/null
+++ b/sound/pci/ice1712/delta.h
@@ -0,0 +1,153 @@
+#ifndef __SOUND_DELTA_H
+#define __SOUND_DELTA_H
+
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for M-Audio Delta 1010, 44, 66, Dio2496, Audiophile
+ * Digigram VX442
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define DELTA_DEVICE_DESC \
+ "{MidiMan M Audio,Delta 1010},"\
+ "{MidiMan M Audio,Delta 1010LT},"\
+ "{MidiMan M Audio,Delta DiO 2496},"\
+ "{MidiMan M Audio,Delta 66},"\
+ "{MidiMan M Audio,Delta 44},"\
+ "{MidiMan M Audio,Delta 410},"\
+ "{MidiMan M Audio,Audiophile 24/96},"\
+ "{Digigram,VX442},"\
+ "{Lionstracs,Mediastation},"
+
+#define ICE1712_SUBDEVICE_DELTA1010 0x121430d6
+#define ICE1712_SUBDEVICE_DELTA1010E 0xff1430d6
+#define ICE1712_SUBDEVICE_DELTADIO2496 0x121431d6
+#define ICE1712_SUBDEVICE_DELTA66 0x121432d6
+#define ICE1712_SUBDEVICE_DELTA66E 0xff1432d6
+#define ICE1712_SUBDEVICE_DELTA44 0x121433d6
+#define ICE1712_SUBDEVICE_AUDIOPHILE 0x121434d6
+#define ICE1712_SUBDEVICE_DELTA410 0x121438d6
+#define ICE1712_SUBDEVICE_DELTA1010LT 0x12143bd6
+#define ICE1712_SUBDEVICE_VX442 0x12143cd6
+#define ICE1712_SUBDEVICE_MEDIASTATION 0x694c0100
+
+/* entry point */
+extern struct snd_ice1712_card_info snd_ice1712_delta_cards[];
+
+
+/*
+ * MidiMan M-Audio Delta GPIO definitions
+ */
+
+/* MidiMan M-Audio Delta shared pins */
+#define ICE1712_DELTA_DFS 0x01 /* fast/slow sample rate mode */
+ /* (>48kHz must be 1) */
+#define ICE1712_DELTA_SPDIF_IN_STAT 0x02
+ /* S/PDIF input status */
+ /* 0 = valid signal is present */
+ /* all except Delta44 */
+ /* look to CS8414 datasheet */
+#define ICE1712_DELTA_SPDIF_OUT_STAT_CLOCK 0x04
+ /* S/PDIF output status clock */
+ /* (writing on rising edge - 0->1) */
+ /* all except Delta44 */
+ /* look to CS8404A datasheet */
+#define ICE1712_DELTA_SPDIF_OUT_STAT_DATA 0x08
+ /* S/PDIF output status data */
+ /* all except Delta44 */
+ /* look to CS8404A datasheet */
+/* MidiMan M-Audio DeltaDiO */
+/* 0x01 = DFS */
+/* 0x02 = SPDIF_IN_STAT */
+/* 0x04 = SPDIF_OUT_STAT_CLOCK */
+/* 0x08 = SPDIF_OUT_STAT_DATA */
+#define ICE1712_DELTA_SPDIF_INPUT_SELECT 0x10
+ /* coaxial (0), optical (1) */
+ /* S/PDIF input select*/
+
+/* MidiMan M-Audio Delta1010 */
+/* 0x01 = DFS */
+/* 0x02 = SPDIF_IN_STAT */
+/* 0x04 = SPDIF_OUT_STAT_CLOCK */
+/* 0x08 = SPDIF_OUT_STAT_DATA */
+#define ICE1712_DELTA_WORD_CLOCK_SELECT 0x10
+ /* 1 - clock are taken from S/PDIF input */
+ /* 0 - clock are taken from Word Clock input */
+ /* affected SPMCLKIN pin of Envy24 */
+#define ICE1712_DELTA_WORD_CLOCK_STATUS 0x20
+ /* 0 = valid word clock signal is present */
+
+/* MidiMan M-Audio Delta66 */
+/* 0x01 = DFS */
+/* 0x02 = SPDIF_IN_STAT */
+/* 0x04 = SPDIF_OUT_STAT_CLOCK */
+/* 0x08 = SPDIF_OUT_STAT_DATA */
+#define ICE1712_DELTA_CODEC_SERIAL_DATA 0x10
+ /* AKM4524 serial data */
+#define ICE1712_DELTA_CODEC_SERIAL_CLOCK 0x20
+ /* AKM4524 serial clock */
+ /* (writing on rising edge - 0->1 */
+#define ICE1712_DELTA_CODEC_CHIP_A 0x40
+#define ICE1712_DELTA_CODEC_CHIP_B 0x80
+ /* 1 - select chip A or B */
+
+/* MidiMan M-Audio Delta44 */
+/* 0x01 = DFS */
+/* 0x10 = CODEC_SERIAL_DATA */
+/* 0x20 = CODEC_SERIAL_CLOCK */
+/* 0x40 = CODEC_CHIP_A */
+/* 0x80 = CODEC_CHIP_B */
+
+/* MidiMan M-Audio Audiophile/Delta410 definitions */
+/* thanks to Kristof Pelckmans <Kristof.Pelckmans@antwerpen.be> for Delta410 info */
+/* 0x01 = DFS */
+#define ICE1712_DELTA_AP_CCLK 0x02 /* SPI clock */
+ /* (clocking on rising edge - 0->1) */
+#define ICE1712_DELTA_AP_DIN 0x04 /* data input */
+#define ICE1712_DELTA_AP_DOUT 0x08 /* data output */
+#define ICE1712_DELTA_AP_CS_DIGITAL 0x10 /* CS8427 chip select */
+ /* low signal = select */
+#define ICE1712_DELTA_AP_CS_CODEC 0x20 /* AK4528 (audiophile), AK4529 (Delta410) chip select */
+ /* low signal = select */
+
+/* MidiMan M-Audio Delta1010LT definitions */
+/* thanks to Anders Johansson <ajh@watri.uwa.edu.au> */
+/* 0x01 = DFS */
+#define ICE1712_DELTA_1010LT_CCLK 0x02 /* SPI clock (AK4524 + CS8427) */
+#define ICE1712_DELTA_1010LT_DIN 0x04 /* data input (CS8427) */
+#define ICE1712_DELTA_1010LT_DOUT 0x08 /* data output (AK4524 + CS8427) */
+#define ICE1712_DELTA_1010LT_CS 0x70 /* mask for CS address */
+#define ICE1712_DELTA_1010LT_CS_CHIP_A 0x00 /* AK4524 #0 */
+#define ICE1712_DELTA_1010LT_CS_CHIP_B 0x10 /* AK4524 #1 */
+#define ICE1712_DELTA_1010LT_CS_CHIP_C 0x20 /* AK4524 #2 */
+#define ICE1712_DELTA_1010LT_CS_CHIP_D 0x30 /* AK4524 #3 */
+#define ICE1712_DELTA_1010LT_CS_CS8427 0x40 /* CS8427 */
+#define ICE1712_DELTA_1010LT_CS_NONE 0x50 /* nothing */
+#define ICE1712_DELTA_1010LT_WORDCLOCK 0x80 /* sample clock source: 0 = Word Clock Input, 1 = S/PDIF Input ??? */
+
+/* Digigram VX442 definitions */
+#define ICE1712_VX442_CCLK 0x02 /* SPI clock */
+#define ICE1712_VX442_DIN 0x04 /* data input */
+#define ICE1712_VX442_DOUT 0x08 /* data output */
+#define ICE1712_VX442_CS_DIGITAL 0x10 /* chip select, low = CS8427 */
+#define ICE1712_VX442_CODEC_CHIP_A 0x20 /* select chip A */
+#define ICE1712_VX442_CODEC_CHIP_B 0x40 /* select chip B */
+
+#endif /* __SOUND_DELTA_H */
diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h
new file mode 100644
index 0000000..a0c5e00
--- /dev/null
+++ b/sound/pci/ice1712/envy24ht.h
@@ -0,0 +1,219 @@
+#ifndef __SOUND_VT1724_H
+#define __SOUND_VT1724_H
+
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24)
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/control.h>
+#include <sound/ac97_codec.h>
+#include <sound/rawmidi.h>
+#include <sound/i2c.h>
+#include <sound/pcm.h>
+
+#include "ice1712.h"
+
+enum {
+ ICE_EEP2_SYSCONF = 0, /* 06 */
+ ICE_EEP2_ACLINK, /* 07 */
+ ICE_EEP2_I2S, /* 08 */
+ ICE_EEP2_SPDIF, /* 09 */
+ ICE_EEP2_GPIO_DIR, /* 0a */
+ ICE_EEP2_GPIO_DIR1, /* 0b */
+ ICE_EEP2_GPIO_DIR2, /* 0c */
+ ICE_EEP2_GPIO_MASK, /* 0d */
+ ICE_EEP2_GPIO_MASK1, /* 0e */
+ ICE_EEP2_GPIO_MASK2, /* 0f */
+ ICE_EEP2_GPIO_STATE, /* 10 */
+ ICE_EEP2_GPIO_STATE1, /* 11 */
+ ICE_EEP2_GPIO_STATE2 /* 12 */
+};
+
+/*
+ * Direct registers
+ */
+
+#define ICEREG1724(ice, x) ((ice)->port + VT1724_REG_##x)
+
+#define VT1724_REG_CONTROL 0x00 /* byte */
+#define VT1724_RESET 0x80 /* reset whole chip */
+#define VT1724_REG_IRQMASK 0x01 /* byte */
+#define VT1724_IRQ_MPU_RX 0x80
+#define VT1724_IRQ_MPU_TX 0x20
+#define VT1724_IRQ_MTPCM 0x10
+#define VT1724_REG_IRQSTAT 0x02 /* byte */
+/* look to VT1724_IRQ_* */
+#define VT1724_REG_SYS_CFG 0x04 /* byte - system configuration PCI60 on Envy24*/
+#define VT1724_CFG_CLOCK 0xc0
+#define VT1724_CFG_CLOCK512 0x00 /* 22.5692Mhz, 44.1kHz*512 */
+#define VT1724_CFG_CLOCK384 0x40 /* 16.9344Mhz, 44.1kHz*384 */
+#define VT1724_CFG_MPU401 0x20 /* MPU401 UARTs */
+#define VT1724_CFG_ADC_MASK 0x0c /* one, two or one and S/PDIF, stereo ADCs */
+#define VT1724_CFG_DAC_MASK 0x03 /* one, two, three, four stereo DACs */
+
+#define VT1724_REG_AC97_CFG 0x05 /* byte */
+#define VT1724_CFG_PRO_I2S 0x80 /* multitrack converter: I2S or AC'97 */
+#define VT1724_CFG_AC97_PACKED 0x01 /* split or packed mode - AC'97 */
+
+#define VT1724_REG_I2S_FEATURES 0x06 /* byte */
+#define VT1724_CFG_I2S_VOLUME 0x80 /* volume/mute capability */
+#define VT1724_CFG_I2S_96KHZ 0x40 /* supports 96kHz sampling */
+#define VT1724_CFG_I2S_RESMASK 0x30 /* resolution mask, 16,18,20,24-bit */
+#define VT1724_CFG_I2S_192KHZ 0x08 /* supports 192kHz sampling */
+#define VT1724_CFG_I2S_OTHER 0x07 /* other I2S IDs */
+
+#define VT1724_REG_SPDIF_CFG 0x07 /* byte */
+#define VT1724_CFG_SPDIF_OUT_EN 0x80 /*Internal S/PDIF output is enabled*/
+#define VT1724_CFG_SPDIF_OUT_INT 0x40 /*Internal S/PDIF output is implemented*/
+#define VT1724_CFG_I2S_CHIPID 0x3c /* I2S chip ID */
+#define VT1724_CFG_SPDIF_IN 0x02 /* S/PDIF input is present */
+#define VT1724_CFG_SPDIF_OUT 0x01 /* External S/PDIF output is present */
+
+/*there is no consumer AC97 codec with the VT1724*/
+//#define VT1724_REG_AC97_INDEX 0x08 /* byte */
+//#define VT1724_REG_AC97_CMD 0x09 /* byte */
+
+#define VT1724_REG_MPU_TXFIFO 0x0a /*byte ro. number of bytes in TX fifo*/
+#define VT1724_REG_MPU_RXFIFO 0x0b /*byte ro. number of bytes in RX fifo*/
+
+#define VT1724_REG_MPU_DATA 0x0c /* byte */
+#define VT1724_REG_MPU_CTRL 0x0d /* byte */
+#define VT1724_MPU_UART 0x01
+#define VT1724_MPU_TX_EMPTY 0x02
+#define VT1724_MPU_TX_FULL 0x04
+#define VT1724_MPU_RX_EMPTY 0x08
+#define VT1724_MPU_RX_FULL 0x10
+
+#define VT1724_REG_MPU_FIFO_WM 0x0e /*byte set the high/low watermarks for RX/TX fifos*/
+#define VT1724_MPU_RX_FIFO 0x20 //1=rx fifo watermark 0=tx fifo watermark
+#define VT1724_MPU_FIFO_MASK 0x1f
+
+#define VT1724_REG_I2C_DEV_ADDR 0x10 /* byte */
+#define VT1724_I2C_WRITE 0x01 /* write direction */
+#define VT1724_REG_I2C_BYTE_ADDR 0x11 /* byte */
+#define VT1724_REG_I2C_DATA 0x12 /* byte */
+#define VT1724_REG_I2C_CTRL 0x13 /* byte */
+#define VT1724_I2C_EEPROM 0x80 /* 1 = EEPROM exists */
+#define VT1724_I2C_BUSY 0x01 /* busy bit */
+
+#define VT1724_REG_GPIO_DATA 0x14 /* word */
+#define VT1724_REG_GPIO_WRITE_MASK 0x16 /* word */
+#define VT1724_REG_GPIO_DIRECTION 0x18 /* dword? (3 bytes) 0=input 1=output.
+ bit3 - during reset used for Eeprom power-on strapping
+ if TESTEN# pin active, bit 2 always input*/
+#define VT1724_REG_POWERDOWN 0x1c
+#define VT1724_REG_GPIO_DATA_22 0x1e /* byte direction for GPIO 16:22 */
+#define VT1724_REG_GPIO_WRITE_MASK_22 0x1f /* byte write mask for GPIO 16:22 */
+
+
+/*
+ * Professional multi-track direct control registers
+ */
+
+#define ICEMT1724(ice, x) ((ice)->profi_port + VT1724_MT_##x)
+
+#define VT1724_MT_IRQ 0x00 /* byte - interrupt mask */
+#define VT1724_MULTI_PDMA4 0x80 /* SPDIF Out / PDMA4 */
+#define VT1724_MULTI_PDMA3 0x40 /* PDMA3 */
+#define VT1724_MULTI_PDMA2 0x20 /* PDMA2 */
+#define VT1724_MULTI_PDMA1 0x10 /* PDMA1 */
+#define VT1724_MULTI_FIFO_ERR 0x08 /* DMA FIFO underrun/overrun. */
+#define VT1724_MULTI_RDMA1 0x04 /* RDMA1 (S/PDIF input) */
+#define VT1724_MULTI_RDMA0 0x02 /* RMDA0 */
+#define VT1724_MULTI_PDMA0 0x01 /* MC Interleave/PDMA0 */
+
+#define VT1724_MT_RATE 0x01 /* byte - sampling rate select */
+#define VT1724_SPDIF_MASTER 0x10 /* S/PDIF input is master clock */
+#define VT1724_MT_I2S_FORMAT 0x02 /* byte - I2S data format */
+#define VT1724_MT_I2S_MCLK_128X 0x08
+#define VT1724_MT_I2S_FORMAT_MASK 0x03
+#define VT1724_MT_I2S_FORMAT_I2S 0x00
+#define VT1724_MT_DMA_INT_MASK 0x03 /* byte -DMA Interrupt Mask */
+/* lool to VT1724_MULTI_* */
+#define VT1724_MT_AC97_INDEX 0x04 /* byte - AC'97 index */
+#define VT1724_MT_AC97_CMD 0x05 /* byte - AC'97 command & status */
+#define VT1724_AC97_COLD 0x80 /* cold reset */
+#define VT1724_AC97_WARM 0x40 /* warm reset */
+#define VT1724_AC97_WRITE 0x20 /* W: write, R: write in progress */
+#define VT1724_AC97_READ 0x10 /* W: read, R: read in progress */
+#define VT1724_AC97_READY 0x08 /* codec ready status bit */
+#define VT1724_AC97_ID_MASK 0x03 /* codec id mask */
+#define VT1724_MT_AC97_DATA 0x06 /* word - AC'97 data */
+#define VT1724_MT_PLAYBACK_ADDR 0x10 /* dword - playback address */
+#define VT1724_MT_PLAYBACK_SIZE 0x14 /* dword - playback size */
+#define VT1724_MT_DMA_CONTROL 0x18 /* byte - control */
+#define VT1724_PDMA4_START 0x80 /* SPDIF out / PDMA4 start */
+#define VT1724_PDMA3_START 0x40 /* PDMA3 start */
+#define VT1724_PDMA2_START 0x20 /* PDMA2 start */
+#define VT1724_PDMA1_START 0x10 /* PDMA1 start */
+#define VT1724_RDMA1_START 0x04 /* RDMA1 start */
+#define VT1724_RDMA0_START 0x02 /* RMDA0 start */
+#define VT1724_PDMA0_START 0x01 /* MC Interleave / PDMA0 start */
+#define VT1724_MT_BURST 0x19 /* Interleaved playback DMA Active streams / PCI burst size */
+#define VT1724_MT_DMA_FIFO_ERR 0x1a /*Global playback and record DMA FIFO Underrun/Overrun */
+#define VT1724_PDMA4_UNDERRUN 0x80
+#define VT1724_PDMA2_UNDERRUN 0x40
+#define VT1724_PDMA3_UNDERRUN 0x20
+#define VT1724_PDMA1_UNDERRUN 0x10
+#define VT1724_RDMA1_UNDERRUN 0x04
+#define VT1724_RDMA0_UNDERRUN 0x02
+#define VT1724_PDMA0_UNDERRUN 0x01
+#define VT1724_MT_DMA_PAUSE 0x1b /*Global playback and record DMA FIFO pause/resume */
+#define VT1724_PDMA4_PAUSE 0x80
+#define VT1724_PDMA3_PAUSE 0x40
+#define VT1724_PDMA2_PAUSE 0x20
+#define VT1724_PDMA1_PAUSE 0x10
+#define VT1724_RDMA1_PAUSE 0x04
+#define VT1724_RDMA0_PAUSE 0x02
+#define VT1724_PDMA0_PAUSE 0x01
+#define VT1724_MT_PLAYBACK_COUNT 0x1c /* word - playback count */
+#define VT1724_MT_CAPTURE_ADDR 0x20 /* dword - capture address */
+#define VT1724_MT_CAPTURE_SIZE 0x24 /* word - capture size */
+#define VT1724_MT_CAPTURE_COUNT 0x26 /* word - capture count */
+
+#define VT1724_MT_ROUTE_PLAYBACK 0x2c /* word */
+
+#define VT1724_MT_RDMA1_ADDR 0x30 /* dword - RDMA1 capture address */
+#define VT1724_MT_RDMA1_SIZE 0x34 /* word - RDMA1 capture size */
+#define VT1724_MT_RDMA1_COUNT 0x36 /* word - RDMA1 capture count */
+
+#define VT1724_MT_SPDIF_CTRL 0x3c /* word */
+#define VT1724_MT_MONITOR_PEAKINDEX 0x3e /* byte */
+#define VT1724_MT_MONITOR_PEAKDATA 0x3f /* byte */
+
+/* concurrent stereo channels */
+#define VT1724_MT_PDMA4_ADDR 0x40 /* dword */
+#define VT1724_MT_PDMA4_SIZE 0x44 /* word */
+#define VT1724_MT_PDMA4_COUNT 0x46 /* word */
+#define VT1724_MT_PDMA3_ADDR 0x50 /* dword */
+#define VT1724_MT_PDMA3_SIZE 0x54 /* word */
+#define VT1724_MT_PDMA3_COUNT 0x56 /* word */
+#define VT1724_MT_PDMA2_ADDR 0x60 /* dword */
+#define VT1724_MT_PDMA2_SIZE 0x64 /* word */
+#define VT1724_MT_PDMA2_COUNT 0x66 /* word */
+#define VT1724_MT_PDMA1_ADDR 0x70 /* dword */
+#define VT1724_MT_PDMA1_SIZE 0x74 /* word */
+#define VT1724_MT_PDMA1_COUNT 0x76 /* word */
+
+
+unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, unsigned char dev, unsigned char addr);
+void snd_vt1724_write_i2c(struct snd_ice1712 *ice, unsigned char dev, unsigned char addr, unsigned char data);
+
+#endif /* __SOUND_VT1724_H */
diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c
new file mode 100644
index 0000000..6fe35b8
--- /dev/null
+++ b/sound/pci/ice1712/ews.c
@@ -0,0 +1,1087 @@
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for Terratec EWS88MT/D, EWX24/96, DMX 6Fire
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ * 2002 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/cs8427.h>
+#include <sound/asoundef.h>
+
+#include "ice1712.h"
+#include "ews.h"
+
+#define SND_CS8404
+#include <sound/cs8403.h>
+
+enum {
+ EWS_I2C_CS8404 = 0, EWS_I2C_PCF1, EWS_I2C_PCF2,
+ EWS_I2C_88D = 0,
+ EWS_I2C_6FIRE = 0
+};
+
+
+/* additional i2c devices for EWS boards */
+struct ews_spec {
+ struct snd_i2c_device *i2cdevs[3];
+};
+
+/*
+ * access via i2c mode (for EWX 24/96, EWS 88MT&D)
+ */
+
+/* send SDA and SCL */
+static void ewx_i2c_setlines(struct snd_i2c_bus *bus, int clk, int data)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ unsigned char tmp = 0;
+ if (clk)
+ tmp |= ICE1712_EWX2496_SERIAL_CLOCK;
+ if (data)
+ tmp |= ICE1712_EWX2496_SERIAL_DATA;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+ udelay(5);
+}
+
+static int ewx_i2c_getclock(struct snd_i2c_bus *bus)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ return snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA) & ICE1712_EWX2496_SERIAL_CLOCK ? 1 : 0;
+}
+
+static int ewx_i2c_getdata(struct snd_i2c_bus *bus, int ack)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ int bit;
+ /* set RW pin to low */
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ~ICE1712_EWX2496_RW);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, 0);
+ if (ack)
+ udelay(5);
+ bit = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA) & ICE1712_EWX2496_SERIAL_DATA ? 1 : 0;
+ /* set RW pin to high */
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, ICE1712_EWX2496_RW);
+ /* reset write mask */
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ~ICE1712_EWX2496_SERIAL_CLOCK);
+ return bit;
+}
+
+static void ewx_i2c_start(struct snd_i2c_bus *bus)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ unsigned char mask;
+
+ snd_ice1712_save_gpio_status(ice);
+ /* set RW high */
+ mask = ICE1712_EWX2496_RW;
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWX2496:
+ mask |= ICE1712_EWX2496_AK4524_CS; /* CS high also */
+ break;
+ case ICE1712_SUBDEVICE_DMX6FIRE:
+ mask |= ICE1712_6FIRE_AK4524_CS_MASK; /* CS high also */
+ break;
+ }
+ snd_ice1712_gpio_write_bits(ice, mask, mask);
+}
+
+static void ewx_i2c_stop(struct snd_i2c_bus *bus)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ snd_ice1712_restore_gpio_status(ice);
+}
+
+static void ewx_i2c_direction(struct snd_i2c_bus *bus, int clock, int data)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ unsigned char mask = 0;
+
+ if (clock)
+ mask |= ICE1712_EWX2496_SERIAL_CLOCK; /* write SCL */
+ if (data)
+ mask |= ICE1712_EWX2496_SERIAL_DATA; /* write SDA */
+ ice->gpio.direction &= ~(ICE1712_EWX2496_SERIAL_CLOCK|ICE1712_EWX2496_SERIAL_DATA);
+ ice->gpio.direction |= mask;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION, ice->gpio.direction);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ~mask);
+}
+
+static struct snd_i2c_bit_ops snd_ice1712_ewx_cs8427_bit_ops = {
+ .start = ewx_i2c_start,
+ .stop = ewx_i2c_stop,
+ .direction = ewx_i2c_direction,
+ .setlines = ewx_i2c_setlines,
+ .getclock = ewx_i2c_getclock,
+ .getdata = ewx_i2c_getdata,
+};
+
+
+/*
+ * AK4524 access
+ */
+
+/* AK4524 chip select; address 0x48 bit 0-3 */
+static int snd_ice1712_ews88mt_chip_select(struct snd_ice1712 *ice, int chip_mask)
+{
+ struct ews_spec *spec = ice->spec;
+ unsigned char data, ndata;
+
+ if (snd_BUG_ON(chip_mask < 0 || chip_mask > 0x0f))
+ return -EINVAL;
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF2], &data, 1) != 1)
+ goto __error;
+ ndata = (data & 0xf0) | chip_mask;
+ if (ndata != data)
+ if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_PCF2], &ndata, 1)
+ != 1)
+ goto __error;
+ snd_i2c_unlock(ice->i2c);
+ return 0;
+
+ __error:
+ snd_i2c_unlock(ice->i2c);
+ snd_printk(KERN_ERR "AK4524 chip select failed, check cable to the front module\n");
+ return -EIO;
+}
+
+/* start callback for EWS88MT, needs to select a certain chip mask */
+static void ews88mt_ak4524_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+ unsigned char tmp;
+ /* assert AK4524 CS */
+ if (snd_ice1712_ews88mt_chip_select(ice, ~(1 << chip) & 0x0f) < 0)
+ snd_printk(KERN_ERR "fatal error (ews88mt chip select)\n");
+ snd_ice1712_save_gpio_status(ice);
+ tmp = ICE1712_EWS88_SERIAL_DATA |
+ ICE1712_EWS88_SERIAL_CLOCK |
+ ICE1712_EWS88_RW;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION,
+ ice->gpio.direction | tmp);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ~tmp);
+}
+
+/* stop callback for EWS88MT, needs to deselect chip mask */
+static void ews88mt_ak4524_unlock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+ snd_ice1712_restore_gpio_status(ice);
+ udelay(1);
+ snd_ice1712_ews88mt_chip_select(ice, 0x0f);
+}
+
+/* start callback for EWX24/96 */
+static void ewx2496_ak4524_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+ unsigned char tmp;
+ snd_ice1712_save_gpio_status(ice);
+ tmp = ICE1712_EWX2496_SERIAL_DATA |
+ ICE1712_EWX2496_SERIAL_CLOCK |
+ ICE1712_EWX2496_AK4524_CS |
+ ICE1712_EWX2496_RW;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION,
+ ice->gpio.direction | tmp);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ~tmp);
+}
+
+/* start callback for DMX 6fire */
+static void dmx6fire_ak4524_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ak4xxx_private *priv = (void *)ak->private_value[0];
+ struct snd_ice1712 *ice = ak->private_data[0];
+ unsigned char tmp;
+ snd_ice1712_save_gpio_status(ice);
+ tmp = priv->cs_mask = priv->cs_addr = (1 << chip) & ICE1712_6FIRE_AK4524_CS_MASK;
+ tmp |= ICE1712_6FIRE_SERIAL_DATA |
+ ICE1712_6FIRE_SERIAL_CLOCK |
+ ICE1712_6FIRE_RW;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION,
+ ice->gpio.direction | tmp);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ~tmp);
+}
+
+/*
+ * CS8404 interface on EWS88MT/D
+ */
+
+static void snd_ice1712_ews_cs8404_spdif_write(struct snd_ice1712 *ice, unsigned char bits)
+{
+ struct ews_spec *spec = ice->spec;
+ unsigned char bytes[2];
+
+ snd_i2c_lock(ice->i2c);
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWS88MT:
+ case ICE1712_SUBDEVICE_EWS88MT_NEW:
+ case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+ if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_CS8404], &bits, 1)
+ != 1)
+ goto _error;
+ break;
+ case ICE1712_SUBDEVICE_EWS88D:
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_88D], bytes, 2)
+ != 2)
+ goto _error;
+ if (bits != bytes[1]) {
+ bytes[1] = bits;
+ if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_88D],
+ bytes, 2) != 2)
+ goto _error;
+ }
+ break;
+ }
+ _error:
+ snd_i2c_unlock(ice->i2c);
+}
+
+/*
+ */
+
+static void ews88_spdif_default_get(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ snd_cs8404_decode_spdif_bits(&ucontrol->value.iec958, ice->spdif.cs8403_bits);
+}
+
+static int ews88_spdif_default_put(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int val;
+ int change;
+
+ val = snd_cs8404_encode_spdif_bits(&ucontrol->value.iec958);
+ spin_lock_irq(&ice->reg_lock);
+ change = ice->spdif.cs8403_bits != val;
+ ice->spdif.cs8403_bits = val;
+ if (change && ice->playback_pro_substream == NULL) {
+ spin_unlock_irq(&ice->reg_lock);
+ snd_ice1712_ews_cs8404_spdif_write(ice, val);
+ } else {
+ spin_unlock_irq(&ice->reg_lock);
+ }
+ return change;
+}
+
+static void ews88_spdif_stream_get(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ snd_cs8404_decode_spdif_bits(&ucontrol->value.iec958, ice->spdif.cs8403_stream_bits);
+}
+
+static int ews88_spdif_stream_put(struct snd_ice1712 *ice, struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int val;
+ int change;
+
+ val = snd_cs8404_encode_spdif_bits(&ucontrol->value.iec958);
+ spin_lock_irq(&ice->reg_lock);
+ change = ice->spdif.cs8403_stream_bits != val;
+ ice->spdif.cs8403_stream_bits = val;
+ if (change && ice->playback_pro_substream != NULL) {
+ spin_unlock_irq(&ice->reg_lock);
+ snd_ice1712_ews_cs8404_spdif_write(ice, val);
+ } else {
+ spin_unlock_irq(&ice->reg_lock);
+ }
+ return change;
+}
+
+
+/* open callback */
+static void ews88_open_spdif(struct snd_ice1712 *ice, struct snd_pcm_substream *substream)
+{
+ ice->spdif.cs8403_stream_bits = ice->spdif.cs8403_bits;
+}
+
+/* set up SPDIF for EWS88MT / EWS88D */
+static void ews88_setup_spdif(struct snd_ice1712 *ice, int rate)
+{
+ unsigned long flags;
+ unsigned char tmp;
+ int change;
+
+ spin_lock_irqsave(&ice->reg_lock, flags);
+ tmp = ice->spdif.cs8403_stream_bits;
+ if (tmp & 0x10) /* consumer */
+ tmp &= (tmp & 0x01) ? ~0x06 : ~0x60;
+ switch (rate) {
+ case 32000: tmp |= (tmp & 0x01) ? 0x02 : 0x00; break;
+ case 44100: tmp |= (tmp & 0x01) ? 0x06 : 0x40; break;
+ case 48000: tmp |= (tmp & 0x01) ? 0x04 : 0x20; break;
+ default: tmp |= (tmp & 0x01) ? 0x06 : 0x40; break;
+ }
+ change = ice->spdif.cs8403_stream_bits != tmp;
+ ice->spdif.cs8403_stream_bits = tmp;
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+ if (change)
+ snd_ctl_notify(ice->card, SNDRV_CTL_EVENT_MASK_VALUE, &ice->spdif.stream_ctl->id);
+ snd_ice1712_ews_cs8404_spdif_write(ice, tmp);
+}
+
+
+/*
+ */
+static struct snd_akm4xxx akm_ews88mt __devinitdata = {
+ .num_adcs = 8,
+ .num_dacs = 8,
+ .type = SND_AK4524,
+ .ops = {
+ .lock = ews88mt_ak4524_lock,
+ .unlock = ews88mt_ak4524_unlock
+ }
+};
+
+static struct snd_ak4xxx_private akm_ews88mt_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 1, /* CIF high */
+ .data_mask = ICE1712_EWS88_SERIAL_DATA,
+ .clk_mask = ICE1712_EWS88_SERIAL_CLOCK,
+ .cs_mask = 0,
+ .cs_addr = 0,
+ .cs_none = 0, /* no chip select on gpio */
+ .add_flags = ICE1712_EWS88_RW, /* set rw bit high */
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_ewx2496 __devinitdata = {
+ .num_adcs = 2,
+ .num_dacs = 2,
+ .type = SND_AK4524,
+ .ops = {
+ .lock = ewx2496_ak4524_lock
+ }
+};
+
+static struct snd_ak4xxx_private akm_ewx2496_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 1, /* CIF high */
+ .data_mask = ICE1712_EWS88_SERIAL_DATA,
+ .clk_mask = ICE1712_EWS88_SERIAL_CLOCK,
+ .cs_mask = ICE1712_EWX2496_AK4524_CS,
+ .cs_addr = ICE1712_EWX2496_AK4524_CS,
+ .cs_none = 0,
+ .add_flags = ICE1712_EWS88_RW, /* set rw bit high */
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_6fire __devinitdata = {
+ .num_adcs = 6,
+ .num_dacs = 6,
+ .type = SND_AK4524,
+ .ops = {
+ .lock = dmx6fire_ak4524_lock
+ }
+};
+
+static struct snd_ak4xxx_private akm_6fire_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 1, /* CIF high */
+ .data_mask = ICE1712_6FIRE_SERIAL_DATA,
+ .clk_mask = ICE1712_6FIRE_SERIAL_CLOCK,
+ .cs_mask = 0,
+ .cs_addr = 0, /* set later */
+ .cs_none = 0,
+ .add_flags = ICE1712_6FIRE_RW, /* set rw bit high */
+ .mask_flags = 0,
+};
+
+/*
+ * initialize the chip
+ */
+
+/* 6fire specific */
+#define PCF9554_REG_INPUT 0
+#define PCF9554_REG_OUTPUT 1
+#define PCF9554_REG_POLARITY 2
+#define PCF9554_REG_CONFIG 3
+
+static int snd_ice1712_6fire_write_pca(struct snd_ice1712 *ice, unsigned char reg, unsigned char data);
+
+static int __devinit snd_ice1712_ews_init(struct snd_ice1712 *ice)
+{
+ int err;
+ struct snd_akm4xxx *ak;
+ struct ews_spec *spec;
+
+ /* set the analog DACs */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWX2496:
+ ice->num_total_dacs = 2;
+ ice->num_total_adcs = 2;
+ break;
+ case ICE1712_SUBDEVICE_EWS88MT:
+ case ICE1712_SUBDEVICE_EWS88MT_NEW:
+ case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 8;
+ break;
+ case ICE1712_SUBDEVICE_EWS88D:
+ /* Note: not analog but ADAT I/O */
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 8;
+ break;
+ case ICE1712_SUBDEVICE_DMX6FIRE:
+ ice->num_total_dacs = 6;
+ ice->num_total_adcs = 6;
+ break;
+ }
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ /* create i2c */
+ if ((err = snd_i2c_bus_create(ice->card, "ICE1712 GPIO 1", NULL, &ice->i2c)) < 0) {
+ snd_printk(KERN_ERR "unable to create I2C bus\n");
+ return err;
+ }
+ ice->i2c->private_data = ice;
+ ice->i2c->hw_ops.bit = &snd_ice1712_ewx_cs8427_bit_ops;
+
+ /* create i2c devices */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_DMX6FIRE:
+ err = snd_i2c_device_create(ice->i2c, "PCF9554",
+ ICE1712_6FIRE_PCF9554_ADDR,
+ &spec->i2cdevs[EWS_I2C_6FIRE]);
+ if (err < 0) {
+ snd_printk(KERN_ERR "PCF9554 initialization failed\n");
+ return err;
+ }
+ snd_ice1712_6fire_write_pca(ice, PCF9554_REG_CONFIG, 0x80);
+ break;
+ case ICE1712_SUBDEVICE_EWS88MT:
+ case ICE1712_SUBDEVICE_EWS88MT_NEW:
+ case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+
+ err = snd_i2c_device_create(ice->i2c, "CS8404",
+ ICE1712_EWS88MT_CS8404_ADDR,
+ &spec->i2cdevs[EWS_I2C_CS8404]);
+ if (err < 0)
+ return err;
+ err = snd_i2c_device_create(ice->i2c, "PCF8574 (1st)",
+ ICE1712_EWS88MT_INPUT_ADDR,
+ &spec->i2cdevs[EWS_I2C_PCF1]);
+ if (err < 0)
+ return err;
+ err = snd_i2c_device_create(ice->i2c, "PCF8574 (2nd)",
+ ICE1712_EWS88MT_OUTPUT_ADDR,
+ &spec->i2cdevs[EWS_I2C_PCF2]);
+ if (err < 0)
+ return err;
+ /* Check if the front module is connected */
+ if ((err = snd_ice1712_ews88mt_chip_select(ice, 0x0f)) < 0)
+ return err;
+ break;
+ case ICE1712_SUBDEVICE_EWS88D:
+ err = snd_i2c_device_create(ice->i2c, "PCF8575",
+ ICE1712_EWS88D_PCF_ADDR,
+ &spec->i2cdevs[EWS_I2C_88D]);
+ if (err < 0)
+ return err;
+ break;
+ }
+
+ /* set up SPDIF interface */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWX2496:
+ if ((err = snd_ice1712_init_cs8427(ice, CS8427_BASE_ADDR)) < 0)
+ return err;
+ snd_cs8427_reg_write(ice->cs8427, CS8427_REG_RECVERRMASK, CS8427_UNLOCK | CS8427_CONF | CS8427_BIP | CS8427_PAR);
+ break;
+ case ICE1712_SUBDEVICE_DMX6FIRE:
+ if ((err = snd_ice1712_init_cs8427(ice, ICE1712_6FIRE_CS8427_ADDR)) < 0)
+ return err;
+ snd_cs8427_reg_write(ice->cs8427, CS8427_REG_RECVERRMASK, CS8427_UNLOCK | CS8427_CONF | CS8427_BIP | CS8427_PAR);
+ break;
+ case ICE1712_SUBDEVICE_EWS88MT:
+ case ICE1712_SUBDEVICE_EWS88MT_NEW:
+ case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+ case ICE1712_SUBDEVICE_EWS88D:
+ /* set up CS8404 */
+ ice->spdif.ops.open = ews88_open_spdif;
+ ice->spdif.ops.setup_rate = ews88_setup_spdif;
+ ice->spdif.ops.default_get = ews88_spdif_default_get;
+ ice->spdif.ops.default_put = ews88_spdif_default_put;
+ ice->spdif.ops.stream_get = ews88_spdif_stream_get;
+ ice->spdif.ops.stream_put = ews88_spdif_stream_put;
+ /* Set spdif defaults */
+ snd_ice1712_ews_cs8404_spdif_write(ice, ice->spdif.cs8403_bits);
+ break;
+ }
+
+ /* no analog? */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWS88D:
+ return 0;
+ }
+
+ /* analog section */
+ ak = ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (! ak)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWS88MT:
+ case ICE1712_SUBDEVICE_EWS88MT_NEW:
+ case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_ews88mt, &akm_ews88mt_priv, ice);
+ break;
+ case ICE1712_SUBDEVICE_EWX2496:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_ewx2496, &akm_ewx2496_priv, ice);
+ break;
+ case ICE1712_SUBDEVICE_DMX6FIRE:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_6fire, &akm_6fire_priv, ice);
+ break;
+ default:
+ err = 0;
+ }
+
+ return err;
+}
+
+/*
+ * EWX 24/96 specific controls
+ */
+
+/* i/o sensitivity - this callback is shared among other devices, too */
+static int snd_ice1712_ewx_io_sense_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo){
+
+ static char *texts[2] = {
+ "+4dBu", "-10dBV",
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= 2)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ice1712_ewx_io_sense_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char mask = kcontrol->private_value & 0xff;
+
+ snd_ice1712_save_gpio_status(ice);
+ ucontrol->value.enumerated.item[0] = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA) & mask ? 1 : 0;
+ snd_ice1712_restore_gpio_status(ice);
+ return 0;
+}
+
+static int snd_ice1712_ewx_io_sense_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char mask = kcontrol->private_value & 0xff;
+ int val, nval;
+
+ if (kcontrol->private_value & (1 << 31))
+ return -EPERM;
+ nval = ucontrol->value.enumerated.item[0] ? mask : 0;
+ snd_ice1712_save_gpio_status(ice);
+ val = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+ nval |= val & ~mask;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, nval);
+ snd_ice1712_restore_gpio_status(ice);
+ return val != nval;
+}
+
+static struct snd_kcontrol_new snd_ice1712_ewx2496_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Sensitivity Switch",
+ .info = snd_ice1712_ewx_io_sense_info,
+ .get = snd_ice1712_ewx_io_sense_get,
+ .put = snd_ice1712_ewx_io_sense_put,
+ .private_value = ICE1712_EWX2496_AIN_SEL,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Output Sensitivity Switch",
+ .info = snd_ice1712_ewx_io_sense_info,
+ .get = snd_ice1712_ewx_io_sense_get,
+ .put = snd_ice1712_ewx_io_sense_put,
+ .private_value = ICE1712_EWX2496_AOUT_SEL,
+ },
+};
+
+
+/*
+ * EWS88MT specific controls
+ */
+/* analog output sensitivity;; address 0x48 bit 6 */
+static int snd_ice1712_ews88mt_output_sense_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct ews_spec *spec = ice->spec;
+ unsigned char data;
+
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ snd_i2c_unlock(ice->i2c);
+ ucontrol->value.enumerated.item[0] = data & ICE1712_EWS88MT_OUTPUT_SENSE ? 1 : 0; /* high = -10dBV, low = +4dBu */
+ return 0;
+}
+
+/* analog output sensitivity;; address 0x48 bit 6 */
+static int snd_ice1712_ews88mt_output_sense_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct ews_spec *spec = ice->spec;
+ unsigned char data, ndata;
+
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF2], &data, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ ndata = (data & ~ICE1712_EWS88MT_OUTPUT_SENSE) | (ucontrol->value.enumerated.item[0] ? ICE1712_EWS88MT_OUTPUT_SENSE : 0);
+ if (ndata != data && snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_PCF2],
+ &ndata, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ snd_i2c_unlock(ice->i2c);
+ return ndata != data;
+}
+
+/* analog input sensitivity; address 0x46 */
+static int snd_ice1712_ews88mt_input_sense_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct ews_spec *spec = ice->spec;
+ int channel = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned char data;
+
+ if (snd_BUG_ON(channel < 0 || channel > 7))
+ return 0;
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF1], &data, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ /* reversed; high = +4dBu, low = -10dBV */
+ ucontrol->value.enumerated.item[0] = data & (1 << channel) ? 0 : 1;
+ snd_i2c_unlock(ice->i2c);
+ return 0;
+}
+
+/* analog output sensitivity; address 0x46 */
+static int snd_ice1712_ews88mt_input_sense_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct ews_spec *spec = ice->spec;
+ int channel = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned char data, ndata;
+
+ if (snd_BUG_ON(channel < 0 || channel > 7))
+ return 0;
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_PCF1], &data, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ ndata = (data & ~(1 << channel)) | (ucontrol->value.enumerated.item[0] ? 0 : (1 << channel));
+ if (ndata != data && snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_PCF1],
+ &ndata, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ snd_i2c_unlock(ice->i2c);
+ return ndata != data;
+}
+
+static struct snd_kcontrol_new snd_ice1712_ews88mt_input_sense __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Sensitivity Switch",
+ .info = snd_ice1712_ewx_io_sense_info,
+ .get = snd_ice1712_ews88mt_input_sense_get,
+ .put = snd_ice1712_ews88mt_input_sense_put,
+ .count = 8,
+};
+
+static struct snd_kcontrol_new snd_ice1712_ews88mt_output_sense __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Output Sensitivity Switch",
+ .info = snd_ice1712_ewx_io_sense_info,
+ .get = snd_ice1712_ews88mt_output_sense_get,
+ .put = snd_ice1712_ews88mt_output_sense_put,
+};
+
+
+/*
+ * EWS88D specific controls
+ */
+
+#define snd_ice1712_ews88d_control_info snd_ctl_boolean_mono_info
+
+static int snd_ice1712_ews88d_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct ews_spec *spec = ice->spec;
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ unsigned char data[2];
+
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_88D], data, 2) != 2) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ snd_i2c_unlock(ice->i2c);
+ data[0] = (data[shift >> 3] >> (shift & 7)) & 0x01;
+ if (invert)
+ data[0] ^= 0x01;
+ ucontrol->value.integer.value[0] = data[0];
+ return 0;
+}
+
+static int snd_ice1712_ews88d_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct ews_spec *spec = ice->spec;
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ unsigned char data[2], ndata[2];
+ int change;
+
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_88D], data, 2) != 2) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ ndata[shift >> 3] = data[shift >> 3] & ~(1 << (shift & 7));
+ if (invert) {
+ if (! ucontrol->value.integer.value[0])
+ ndata[shift >> 3] |= (1 << (shift & 7));
+ } else {
+ if (ucontrol->value.integer.value[0])
+ ndata[shift >> 3] |= (1 << (shift & 7));
+ }
+ change = (data[shift >> 3] != ndata[shift >> 3]);
+ if (change &&
+ snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_88D], data, 2) != 2) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ snd_i2c_unlock(ice->i2c);
+ return change;
+}
+
+#define EWS88D_CONTROL(xiface, xname, xshift, xinvert, xaccess) \
+{ .iface = xiface,\
+ .name = xname,\
+ .access = xaccess,\
+ .info = snd_ice1712_ews88d_control_info,\
+ .get = snd_ice1712_ews88d_control_get,\
+ .put = snd_ice1712_ews88d_control_put,\
+ .private_value = xshift | (xinvert << 8),\
+}
+
+static struct snd_kcontrol_new snd_ice1712_ews88d_controls[] __devinitdata = {
+ EWS88D_CONTROL(SNDRV_CTL_ELEM_IFACE_MIXER, "IEC958 Input Optical", 0, 1, 0), /* inverted */
+ EWS88D_CONTROL(SNDRV_CTL_ELEM_IFACE_MIXER, "ADAT Output Optical", 1, 0, 0),
+ EWS88D_CONTROL(SNDRV_CTL_ELEM_IFACE_MIXER, "ADAT External Master Clock", 2, 0, 0),
+ EWS88D_CONTROL(SNDRV_CTL_ELEM_IFACE_MIXER, "Enable ADAT", 3, 0, 0),
+ EWS88D_CONTROL(SNDRV_CTL_ELEM_IFACE_MIXER, "ADAT Through", 4, 1, 0),
+};
+
+
+/*
+ * DMX 6Fire specific controls
+ */
+
+static int snd_ice1712_6fire_read_pca(struct snd_ice1712 *ice, unsigned char reg)
+{
+ unsigned char byte;
+ struct ews_spec *spec = ice->spec;
+
+ snd_i2c_lock(ice->i2c);
+ byte = reg;
+ snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1);
+ byte = 0;
+ if (snd_i2c_readbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ printk(KERN_ERR "cannot read pca\n");
+ return -EIO;
+ }
+ snd_i2c_unlock(ice->i2c);
+ return byte;
+}
+
+static int snd_ice1712_6fire_write_pca(struct snd_ice1712 *ice, unsigned char reg, unsigned char data)
+{
+ unsigned char bytes[2];
+ struct ews_spec *spec = ice->spec;
+
+ snd_i2c_lock(ice->i2c);
+ bytes[0] = reg;
+ bytes[1] = data;
+ if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], bytes, 2) != 2) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ snd_i2c_unlock(ice->i2c);
+ return 0;
+}
+
+#define snd_ice1712_6fire_control_info snd_ctl_boolean_mono_info
+
+static int snd_ice1712_6fire_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ int data;
+
+ if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0)
+ return data;
+ data = (data >> shift) & 1;
+ if (invert)
+ data ^= 1;
+ ucontrol->value.integer.value[0] = data;
+ return 0;
+}
+
+static int snd_ice1712_6fire_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value >> 8) & 1;
+ int data, ndata;
+
+ if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0)
+ return data;
+ ndata = data & ~(1 << shift);
+ if (ucontrol->value.integer.value[0])
+ ndata |= (1 << shift);
+ if (invert)
+ ndata ^= (1 << shift);
+ if (data != ndata) {
+ snd_ice1712_6fire_write_pca(ice, PCF9554_REG_OUTPUT, (unsigned char)ndata);
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_ice1712_6fire_select_input_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {
+ "Internal", "Front Input", "Rear Input", "Wave Table"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= 4)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ice1712_6fire_select_input_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int data;
+
+ if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0)
+ return data;
+ ucontrol->value.integer.value[0] = data & 3;
+ return 0;
+}
+
+static int snd_ice1712_6fire_select_input_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int data, ndata;
+
+ if ((data = snd_ice1712_6fire_read_pca(ice, PCF9554_REG_OUTPUT)) < 0)
+ return data;
+ ndata = data & ~3;
+ ndata |= (ucontrol->value.integer.value[0] & 3);
+ if (data != ndata) {
+ snd_ice1712_6fire_write_pca(ice, PCF9554_REG_OUTPUT, (unsigned char)ndata);
+ return 1;
+ }
+ return 0;
+}
+
+
+#define DMX6FIRE_CONTROL(xname, xshift, xinvert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,\
+ .name = xname,\
+ .info = snd_ice1712_6fire_control_info,\
+ .get = snd_ice1712_6fire_control_get,\
+ .put = snd_ice1712_6fire_control_put,\
+ .private_value = xshift | (xinvert << 8),\
+}
+
+static struct snd_kcontrol_new snd_ice1712_6fire_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Select",
+ .info = snd_ice1712_6fire_select_input_info,
+ .get = snd_ice1712_6fire_select_input_get,
+ .put = snd_ice1712_6fire_select_input_put,
+ },
+ DMX6FIRE_CONTROL("Front Digital Input Switch", 2, 1),
+ // DMX6FIRE_CONTROL("Master Clock Select", 3, 0),
+ DMX6FIRE_CONTROL("Optical Digital Input Switch", 4, 0),
+ DMX6FIRE_CONTROL("Phono Analog Input Switch", 5, 0),
+ DMX6FIRE_CONTROL("Breakbox LED", 6, 0),
+};
+
+
+static int __devinit snd_ice1712_ews_add_controls(struct snd_ice1712 *ice)
+{
+ unsigned int idx;
+ int err;
+
+ /* all terratec cards have spdif, but cs8427 module builds it's own controls */
+ if (ice->cs8427 == NULL) {
+ err = snd_ice1712_spdif_build_controls(ice);
+ if (err < 0)
+ return err;
+ }
+
+ /* ak4524 controls */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWX2496:
+ case ICE1712_SUBDEVICE_EWS88MT:
+ case ICE1712_SUBDEVICE_EWS88MT_NEW:
+ case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+ case ICE1712_SUBDEVICE_DMX6FIRE:
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+ break;
+ }
+
+ /* card specific controls */
+ switch (ice->eeprom.subvendor) {
+ case ICE1712_SUBDEVICE_EWX2496:
+ for (idx = 0; idx < ARRAY_SIZE(snd_ice1712_ewx2496_controls); idx++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_ewx2496_controls[idx], ice));
+ if (err < 0)
+ return err;
+ }
+ break;
+ case ICE1712_SUBDEVICE_EWS88MT:
+ case ICE1712_SUBDEVICE_EWS88MT_NEW:
+ case ICE1712_SUBDEVICE_PHASE88:
+ case ICE1712_SUBDEVICE_TS88:
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_ews88mt_input_sense, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_ews88mt_output_sense, ice));
+ if (err < 0)
+ return err;
+ break;
+ case ICE1712_SUBDEVICE_EWS88D:
+ for (idx = 0; idx < ARRAY_SIZE(snd_ice1712_ews88d_controls); idx++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_ews88d_controls[idx], ice));
+ if (err < 0)
+ return err;
+ }
+ break;
+ case ICE1712_SUBDEVICE_DMX6FIRE:
+ for (idx = 0; idx < ARRAY_SIZE(snd_ice1712_6fire_controls); idx++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_6fire_controls[idx], ice));
+ if (err < 0)
+ return err;
+ }
+ break;
+ }
+ return 0;
+}
+
+
+/* entry point */
+struct snd_ice1712_card_info snd_ice1712_ews_cards[] __devinitdata = {
+ {
+ .subvendor = ICE1712_SUBDEVICE_EWX2496,
+ .name = "TerraTec EWX24/96",
+ .model = "ewx2496",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_EWS88MT,
+ .name = "TerraTec EWS88MT",
+ .model = "ews88mt",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_EWS88MT_NEW,
+ .name = "TerraTec EWS88MT",
+ .model = "ews88mt_new",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_PHASE88,
+ .name = "TerraTec Phase88",
+ .model = "phase88",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_TS88,
+ .name = "terrasoniq TS88",
+ .model = "phase88",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_EWS88D,
+ .name = "TerraTec EWS88D",
+ .model = "ews88d",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_DMX6FIRE,
+ .name = "TerraTec DMX6Fire",
+ .model = "dmx6fire",
+ .chip_init = snd_ice1712_ews_init,
+ .build_controls = snd_ice1712_ews_add_controls,
+ .mpu401_1_name = "MIDI-Front DMX6fire",
+ .mpu401_2_name = "Wavetable DMX6fire",
+ .mpu401_2_info_flags = MPU401_INFO_OUTPUT,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/ews.h b/sound/pci/ice1712/ews.h
new file mode 100644
index 0000000..1c44371
--- /dev/null
+++ b/sound/pci/ice1712/ews.h
@@ -0,0 +1,86 @@
+#ifndef __SOUND_EWS_H
+#define __SOUND_EWS_H
+
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for Terratec EWS88MT/D, EWX24/96, DMX 6Fire
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ * 2002 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define EWS_DEVICE_DESC \
+ "{TerraTec,EWX 24/96},"\
+ "{TerraTec,EWS 88MT},"\
+ "{TerraTec,EWS 88D},"\
+ "{TerraTec,DMX 6Fire},"\
+ "{TerraTec,Phase 88}," \
+ "{terrasoniq,TS 88},"
+
+#define ICE1712_SUBDEVICE_EWX2496 0x3b153011
+#define ICE1712_SUBDEVICE_EWS88MT 0x3b151511
+#define ICE1712_SUBDEVICE_EWS88MT_NEW 0x3b152511
+#define ICE1712_SUBDEVICE_EWS88D 0x3b152b11
+#define ICE1712_SUBDEVICE_DMX6FIRE 0x3b153811
+#define ICE1712_SUBDEVICE_PHASE88 0x3b155111
+#define ICE1712_SUBDEVICE_TS88 0x3b157c11
+
+/* entry point */
+extern struct snd_ice1712_card_info snd_ice1712_ews_cards[];
+
+
+/* TerraTec EWX 24/96 configuration definitions */
+
+#define ICE1712_EWX2496_AK4524_CS 0x01 /* AK4524 chip select; low = active */
+#define ICE1712_EWX2496_AIN_SEL 0x02 /* input sensitivity switch; high = louder */
+#define ICE1712_EWX2496_AOUT_SEL 0x04 /* output sensitivity switch; high = louder */
+#define ICE1712_EWX2496_RW 0x08 /* read/write switch for i2c; high = write */
+#define ICE1712_EWX2496_SERIAL_DATA 0x10 /* i2c & ak4524 data */
+#define ICE1712_EWX2496_SERIAL_CLOCK 0x20 /* i2c & ak4524 clock */
+#define ICE1712_EWX2496_TX2 0x40 /* MIDI2 (not used) */
+#define ICE1712_EWX2496_RX2 0x80 /* MIDI2 (not used) */
+
+/* TerraTec EWS 88MT/D configuration definitions */
+/* RW, SDA snd SCLK are identical with EWX24/96 */
+#define ICE1712_EWS88_CS8414_RATE 0x07 /* CS8414 sample rate: gpio 0-2 */
+#define ICE1712_EWS88_RW 0x08 /* read/write switch for i2c; high = write */
+#define ICE1712_EWS88_SERIAL_DATA 0x10 /* i2c & ak4524 data */
+#define ICE1712_EWS88_SERIAL_CLOCK 0x20 /* i2c & ak4524 clock */
+#define ICE1712_EWS88_TX2 0x40 /* MIDI2 (only on 88D) */
+#define ICE1712_EWS88_RX2 0x80 /* MIDI2 (only on 88D) */
+
+/* i2c address */
+#define ICE1712_EWS88MT_CS8404_ADDR (0x40>>1)
+#define ICE1712_EWS88MT_INPUT_ADDR (0x46>>1)
+#define ICE1712_EWS88MT_OUTPUT_ADDR (0x48>>1)
+#define ICE1712_EWS88MT_OUTPUT_SENSE 0x40 /* mask */
+#define ICE1712_EWS88D_PCF_ADDR (0x40>>1)
+
+/* TerraTec DMX 6Fire configuration definitions */
+#define ICE1712_6FIRE_AK4524_CS_MASK 0x07 /* AK4524 chip select #1-#3 */
+#define ICE1712_6FIRE_RW 0x08 /* read/write switch for i2c; high = write */
+#define ICE1712_6FIRE_SERIAL_DATA 0x10 /* i2c & ak4524 data */
+#define ICE1712_6FIRE_SERIAL_CLOCK 0x20 /* i2c & ak4524 clock */
+#define ICE1712_6FIRE_TX2 0x40 /* MIDI2 */
+#define ICE1712_6FIRE_RX2 0x80 /* MIDI2 */
+
+#define ICE1712_6FIRE_PCF9554_ADDR (0x40>>1)
+#define ICE1712_6FIRE_CS8427_ADDR (0x22)
+
+#endif /* __SOUND_EWS_H */
diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c
new file mode 100644
index 0000000..6914189
--- /dev/null
+++ b/sound/pci/ice1712/hoontech.c
@@ -0,0 +1,360 @@
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for Hoontech STDSP24
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "hoontech.h"
+
+/* Hoontech-specific setting */
+struct hoontech_spec {
+ unsigned char boxbits[4];
+ unsigned int config;
+ unsigned short boxconfig[4];
+};
+
+static void __devinit snd_ice1712_stdsp24_gpio_write(struct snd_ice1712 *ice, unsigned char byte)
+{
+ byte |= ICE1712_STDSP24_CLOCK_BIT;
+ udelay(100);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, byte);
+ byte &= ~ICE1712_STDSP24_CLOCK_BIT;
+ udelay(100);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, byte);
+ byte |= ICE1712_STDSP24_CLOCK_BIT;
+ udelay(100);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, byte);
+}
+
+static void __devinit snd_ice1712_stdsp24_darear(struct snd_ice1712 *ice, int activate)
+{
+ struct hoontech_spec *spec = ice->spec;
+ mutex_lock(&ice->gpio_mutex);
+ ICE1712_STDSP24_0_DAREAR(spec->boxbits, activate);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[0]);
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void __devinit snd_ice1712_stdsp24_mute(struct snd_ice1712 *ice, int activate)
+{
+ struct hoontech_spec *spec = ice->spec;
+ mutex_lock(&ice->gpio_mutex);
+ ICE1712_STDSP24_3_MUTE(spec->boxbits, activate);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]);
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void __devinit snd_ice1712_stdsp24_insel(struct snd_ice1712 *ice, int activate)
+{
+ struct hoontech_spec *spec = ice->spec;
+ mutex_lock(&ice->gpio_mutex);
+ ICE1712_STDSP24_3_INSEL(spec->boxbits, activate);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]);
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void __devinit snd_ice1712_stdsp24_box_channel(struct snd_ice1712 *ice, int box, int chn, int activate)
+{
+ struct hoontech_spec *spec = ice->spec;
+
+ mutex_lock(&ice->gpio_mutex);
+
+ /* select box */
+ ICE1712_STDSP24_0_BOX(spec->boxbits, box);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[0]);
+
+ /* prepare for write */
+ if (chn == 3)
+ ICE1712_STDSP24_2_CHN4(spec->boxbits, 0);
+ ICE1712_STDSP24_2_MIDI1(spec->boxbits, activate);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]);
+
+ ICE1712_STDSP24_1_CHN1(spec->boxbits, 1);
+ ICE1712_STDSP24_1_CHN2(spec->boxbits, 1);
+ ICE1712_STDSP24_1_CHN3(spec->boxbits, 1);
+ ICE1712_STDSP24_2_CHN4(spec->boxbits, 1);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[1]);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+ udelay(100);
+ if (chn == 3) {
+ ICE1712_STDSP24_2_CHN4(spec->boxbits, 0);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+ } else {
+ switch (chn) {
+ case 0: ICE1712_STDSP24_1_CHN1(spec->boxbits, 0); break;
+ case 1: ICE1712_STDSP24_1_CHN2(spec->boxbits, 0); break;
+ case 2: ICE1712_STDSP24_1_CHN3(spec->boxbits, 0); break;
+ }
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[1]);
+ }
+ udelay(100);
+ ICE1712_STDSP24_1_CHN1(spec->boxbits, 1);
+ ICE1712_STDSP24_1_CHN2(spec->boxbits, 1);
+ ICE1712_STDSP24_1_CHN3(spec->boxbits, 1);
+ ICE1712_STDSP24_2_CHN4(spec->boxbits, 1);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[1]);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+ udelay(100);
+
+ ICE1712_STDSP24_2_MIDI1(spec->boxbits, 0);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void __devinit snd_ice1712_stdsp24_box_midi(struct snd_ice1712 *ice, int box, int master)
+{
+ struct hoontech_spec *spec = ice->spec;
+
+ mutex_lock(&ice->gpio_mutex);
+
+ /* select box */
+ ICE1712_STDSP24_0_BOX(spec->boxbits, box);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[0]);
+
+ ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 1);
+ ICE1712_STDSP24_2_MIDI1(spec->boxbits, master);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]);
+
+ udelay(100);
+
+ ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 0);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+
+ mdelay(10);
+
+ ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 1);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[2]);
+
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void __devinit snd_ice1712_stdsp24_midi2(struct snd_ice1712 *ice, int activate)
+{
+ struct hoontech_spec *spec = ice->spec;
+ mutex_lock(&ice->gpio_mutex);
+ ICE1712_STDSP24_3_MIDI2(spec->boxbits, activate);
+ snd_ice1712_stdsp24_gpio_write(ice, spec->boxbits[3]);
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice)
+{
+ struct hoontech_spec *spec;
+ int box, chn;
+
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 8;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ ICE1712_STDSP24_SET_ADDR(spec->boxbits, 0);
+ ICE1712_STDSP24_CLOCK(spec->boxbits, 0, 1);
+ ICE1712_STDSP24_0_BOX(spec->boxbits, 0);
+ ICE1712_STDSP24_0_DAREAR(spec->boxbits, 0);
+
+ ICE1712_STDSP24_SET_ADDR(spec->boxbits, 1);
+ ICE1712_STDSP24_CLOCK(spec->boxbits, 1, 1);
+ ICE1712_STDSP24_1_CHN1(spec->boxbits, 1);
+ ICE1712_STDSP24_1_CHN2(spec->boxbits, 1);
+ ICE1712_STDSP24_1_CHN3(spec->boxbits, 1);
+
+ ICE1712_STDSP24_SET_ADDR(spec->boxbits, 2);
+ ICE1712_STDSP24_CLOCK(spec->boxbits, 2, 1);
+ ICE1712_STDSP24_2_CHN4(spec->boxbits, 1);
+ ICE1712_STDSP24_2_MIDIIN(spec->boxbits, 1);
+ ICE1712_STDSP24_2_MIDI1(spec->boxbits, 0);
+
+ ICE1712_STDSP24_SET_ADDR(spec->boxbits, 3);
+ ICE1712_STDSP24_CLOCK(spec->boxbits, 3, 1);
+ ICE1712_STDSP24_3_MIDI2(spec->boxbits, 0);
+ ICE1712_STDSP24_3_MUTE(spec->boxbits, 1);
+ ICE1712_STDSP24_3_INSEL(spec->boxbits, 0);
+
+ /* let's go - activate only functions in first box */
+ spec->config = 0;
+ /* ICE1712_STDSP24_MUTE |
+ ICE1712_STDSP24_INSEL |
+ ICE1712_STDSP24_DAREAR; */
+ /* These boxconfigs have caused problems in the past.
+ * The code is not optimal, but should now enable a working config to
+ * be achieved.
+ * ** MIDI IN can only be configured on one box **
+ * ICE1712_STDSP24_BOX_MIDI1 needs to be set for that box.
+ * Tests on a ADAC2000 box suggest the box config flags do not
+ * work as would be expected, and the inputs are crossed.
+ * Setting ICE1712_STDSP24_BOX_MIDI1 and ICE1712_STDSP24_BOX_MIDI2
+ * on the same box connects MIDI-In to both 401 uarts; both outputs
+ * are then active on all boxes.
+ * The default config here sets up everything on the first box.
+ * Alan Horstmann 5.2.2008
+ */
+ spec->boxconfig[0] = ICE1712_STDSP24_BOX_CHN1 |
+ ICE1712_STDSP24_BOX_CHN2 |
+ ICE1712_STDSP24_BOX_CHN3 |
+ ICE1712_STDSP24_BOX_CHN4 |
+ ICE1712_STDSP24_BOX_MIDI1 |
+ ICE1712_STDSP24_BOX_MIDI2;
+ spec->boxconfig[1] =
+ spec->boxconfig[2] =
+ spec->boxconfig[3] = 0;
+ snd_ice1712_stdsp24_darear(ice,
+ (spec->config & ICE1712_STDSP24_DAREAR) ? 1 : 0);
+ snd_ice1712_stdsp24_mute(ice,
+ (spec->config & ICE1712_STDSP24_MUTE) ? 1 : 0);
+ snd_ice1712_stdsp24_insel(ice,
+ (spec->config & ICE1712_STDSP24_INSEL) ? 1 : 0);
+ for (box = 0; box < 4; box++) {
+ if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI2)
+ snd_ice1712_stdsp24_midi2(ice, 1);
+ for (chn = 0; chn < 4; chn++)
+ snd_ice1712_stdsp24_box_channel(ice, box, chn,
+ (spec->boxconfig[box] & (1 << chn)) ? 1 : 0);
+ if (spec->boxconfig[box] & ICE1712_STDSP24_BOX_MIDI1)
+ snd_ice1712_stdsp24_box_midi(ice, box, 1);
+ }
+
+ return 0;
+}
+
+/*
+ * AK4524 access
+ */
+
+/* start callback for STDSP24 with modified hardware */
+static void stdsp24_ak4524_lock(struct snd_akm4xxx *ak, int chip)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+ unsigned char tmp;
+ snd_ice1712_save_gpio_status(ice);
+ tmp = ICE1712_STDSP24_SERIAL_DATA |
+ ICE1712_STDSP24_SERIAL_CLOCK |
+ ICE1712_STDSP24_AK4524_CS;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION,
+ ice->gpio.direction | tmp);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ~tmp);
+}
+
+static int __devinit snd_ice1712_value_init(struct snd_ice1712 *ice)
+{
+ /* Hoontech STDSP24 with modified hardware */
+ static struct snd_akm4xxx akm_stdsp24_mv __devinitdata = {
+ .num_adcs = 2,
+ .num_dacs = 2,
+ .type = SND_AK4524,
+ .ops = {
+ .lock = stdsp24_ak4524_lock
+ }
+ };
+
+ static struct snd_ak4xxx_private akm_stdsp24_mv_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 1, /* CIF high */
+ .data_mask = ICE1712_STDSP24_SERIAL_DATA,
+ .clk_mask = ICE1712_STDSP24_SERIAL_CLOCK,
+ .cs_mask = ICE1712_STDSP24_AK4524_CS,
+ .cs_addr = ICE1712_STDSP24_AK4524_CS,
+ .cs_none = 0,
+ .add_flags = 0,
+ };
+
+ int err;
+ struct snd_akm4xxx *ak;
+
+ /* set the analog DACs */
+ ice->num_total_dacs = 2;
+
+ /* set the analog ADCs */
+ ice->num_total_adcs = 2;
+
+ /* analog section */
+ ak = ice->akm = kmalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (! ak)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ err = snd_ice1712_akm4xxx_init(ak, &akm_stdsp24_mv, &akm_stdsp24_mv_priv, ice);
+ if (err < 0)
+ return err;
+
+ /* ak4524 controls */
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int __devinit snd_ice1712_ez8_init(struct snd_ice1712 *ice)
+{
+ ice->gpio.write_mask = ice->eeprom.gpiomask;
+ ice->gpio.direction = ice->eeprom.gpiodir;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, ice->eeprom.gpiomask);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION, ice->eeprom.gpiodir);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, ice->eeprom.gpiostate);
+ return 0;
+}
+
+
+/* entry point */
+struct snd_ice1712_card_info snd_ice1712_hoontech_cards[] __devinitdata = {
+ {
+ .subvendor = ICE1712_SUBDEVICE_STDSP24,
+ .name = "Hoontech SoundTrack Audio DSP24",
+ .model = "dsp24",
+ .chip_init = snd_ice1712_hoontech_init,
+ .mpu401_1_name = "MIDI-1 Hoontech/STA DSP24",
+ .mpu401_2_name = "MIDI-2 Hoontech/STA DSP24",
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_STDSP24_VALUE, /* a dummy id */
+ .name = "Hoontech SoundTrack Audio DSP24 Value",
+ .model = "dsp24_value",
+ .chip_init = snd_ice1712_value_init,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_STDSP24_MEDIA7_1,
+ .name = "Hoontech STA DSP24 Media 7.1",
+ .model = "dsp24_71",
+ .chip_init = snd_ice1712_hoontech_init,
+ },
+ {
+ .subvendor = ICE1712_SUBDEVICE_EVENT_EZ8, /* a dummy id */
+ .name = "Event Electronics EZ8",
+ .model = "ez8",
+ .chip_init = snd_ice1712_ez8_init,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/hoontech.h b/sound/pci/ice1712/hoontech.h
new file mode 100644
index 0000000..cc1da1e
--- /dev/null
+++ b/sound/pci/ice1712/hoontech.h
@@ -0,0 +1,77 @@
+#ifndef __SOUND_HOONTECH_H
+#define __SOUND_HOONTECH_H
+
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for Hoontech STDSP24
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define HOONTECH_DEVICE_DESC \
+ "{Hoontech,SoundTrack DSP 24}," \
+ "{Hoontech,SoundTrack DSP 24 Value}," \
+ "{Hoontech,SoundTrack DSP 24 Media 7.1}," \
+ "{Event Electronics,EZ8},"
+
+#define ICE1712_SUBDEVICE_STDSP24 0x12141217 /* Hoontech SoundTrack Audio DSP 24 */
+#define ICE1712_SUBDEVICE_STDSP24_VALUE 0x00010010 /* A dummy id for Hoontech SoundTrack Audio DSP 24 Value */
+#define ICE1712_SUBDEVICE_STDSP24_MEDIA7_1 0x16141217 /* Hoontech ST Audio DSP24 Media 7.1 */
+#define ICE1712_SUBDEVICE_EVENT_EZ8 0x00010001 /* A dummy id for EZ8 */
+
+extern struct snd_ice1712_card_info snd_ice1712_hoontech_cards[];
+
+
+/* Hoontech SoundTrack Audio DSP 24 GPIO definitions */
+
+#define ICE1712_STDSP24_0_BOX(r, x) r[0] = ((r[0] & ~3) | ((x)&3))
+#define ICE1712_STDSP24_0_DAREAR(r, x) r[0] = ((r[0] & ~4) | (((x)&1)<<2))
+#define ICE1712_STDSP24_1_CHN1(r, x) r[1] = ((r[1] & ~1) | ((x)&1))
+#define ICE1712_STDSP24_1_CHN2(r, x) r[1] = ((r[1] & ~2) | (((x)&1)<<1))
+#define ICE1712_STDSP24_1_CHN3(r, x) r[1] = ((r[1] & ~4) | (((x)&1)<<2))
+#define ICE1712_STDSP24_2_CHN4(r, x) r[2] = ((r[2] & ~1) | ((x)&1))
+#define ICE1712_STDSP24_2_MIDIIN(r, x) r[2] = ((r[2] & ~2) | (((x)&1)<<1))
+#define ICE1712_STDSP24_2_MIDI1(r, x) r[2] = ((r[2] & ~4) | (((x)&1)<<2))
+#define ICE1712_STDSP24_3_MIDI2(r, x) r[3] = ((r[3] & ~1) | ((x)&1))
+#define ICE1712_STDSP24_3_MUTE(r, x) r[3] = ((r[3] & ~2) | (((x)&1)<<1))
+#define ICE1712_STDSP24_3_INSEL(r, x) r[3] = ((r[3] & ~4) | (((x)&1)<<2))
+#define ICE1712_STDSP24_SET_ADDR(r, a) r[a&3] = ((r[a&3] & ~0x18) | (((a)&3)<<3))
+#define ICE1712_STDSP24_CLOCK(r, a, c) r[a&3] = ((r[a&3] & ~0x20) | (((c)&1)<<5))
+#define ICE1712_STDSP24_CLOCK_BIT (1<<5)
+
+/* Hoontech SoundTrack Audio DSP 24 box configuration definitions */
+
+#define ICE1712_STDSP24_DAREAR (1<<0)
+#define ICE1712_STDSP24_MUTE (1<<1)
+#define ICE1712_STDSP24_INSEL (1<<2)
+
+#define ICE1712_STDSP24_BOX_CHN1 (1<<0) /* input channel 1 */
+#define ICE1712_STDSP24_BOX_CHN2 (1<<1) /* input channel 2 */
+#define ICE1712_STDSP24_BOX_CHN3 (1<<2) /* input channel 3 */
+#define ICE1712_STDSP24_BOX_CHN4 (1<<3) /* input channel 4 */
+#define ICE1712_STDSP24_BOX_MIDI1 (1<<8)
+#define ICE1712_STDSP24_BOX_MIDI2 (1<<9)
+
+/* Hoontech SoundTrack Audio DSP 24 Value definitions for modified hardware */
+
+#define ICE1712_STDSP24_AK4524_CS 0x03 /* AK4524 chip select; low = active */
+#define ICE1712_STDSP24_SERIAL_DATA 0x0c /* ak4524 data */
+#define ICE1712_STDSP24_SERIAL_CLOCK 0x30 /* ak4524 clock */
+
+#endif /* __SOUND_HOONTECH_H */
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
new file mode 100644
index 0000000..58d7cda
--- /dev/null
+++ b/sound/pci/ice1712/ice1712.c
@@ -0,0 +1,2801 @@
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ NOTES:
+ - spdif nonaudio consumer mode does not work (at least with my
+ Sony STR-DB830)
+*/
+
+/*
+ * Changes:
+ *
+ * 2002.09.09 Takashi Iwai <tiwai@suse.de>
+ * split the code to several files. each low-level routine
+ * is stored in the local file and called from registration
+ * function from card_info struct.
+ *
+ * 2002.11.26 James Stafford <jstafford@ampltd.com>
+ * Added support for VT1724 (Envy24HT)
+ * I have left out support for 176.4 and 192 KHz for the moment.
+ * I also haven't done anything with the internal S/PDIF transmitter or the MPU-401
+ *
+ * 2003.02.20 Taksahi Iwai <tiwai@suse.de>
+ * Split vt1724 part to an independent driver.
+ * The GPIO is accessed through the callback functions now.
+ *
+ * 2004.03.31 Doug McLain <nostar@comcast.net>
+ * Added support for Event Electronics EZ8 card to hoontech.c.
+ */
+
+
+#include <linux/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/cs8427.h>
+#include <sound/info.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <sound/asoundef.h>
+
+#include "ice1712.h"
+
+/* lowlevel routines */
+#include "delta.h"
+#include "ews.h"
+#include "hoontech.h"
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("ICEnsemble ICE1712 (Envy24)");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{"
+ HOONTECH_DEVICE_DESC
+ DELTA_DEVICE_DESC
+ EWS_DEVICE_DESC
+ "{ICEnsemble,Generic ICE1712},"
+ "{ICEnsemble,Generic Envy24}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */
+static char *model[SNDRV_CARDS];
+static int omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */
+static int cs8427_timeout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 500}; /* CS8427 S/PDIF transciever reset timeout value in msec */
+static int dxr_enable[SNDRV_CARDS]; /* DXR enable for DMX6FIRE */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for ICE1712 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for ICE1712 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable ICE1712 soundcard.");
+module_param_array(omni, bool, NULL, 0444);
+MODULE_PARM_DESC(omni, "Enable Midiman M-Audio Delta Omni I/O support.");
+module_param_array(cs8427_timeout, int, NULL, 0444);
+MODULE_PARM_DESC(cs8427_timeout, "Define reset timeout for cs8427 chip in msec resolution.");
+module_param_array(model, charp, NULL, 0444);
+MODULE_PARM_DESC(model, "Use the given board model.");
+module_param_array(dxr_enable, int, NULL, 0444);
+MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE.");
+
+
+static const struct pci_device_id snd_ice1712_ids[] = {
+ { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_ICE_1712, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICE1712 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_ice1712_ids);
+
+static int snd_ice1712_build_pro_mixer(struct snd_ice1712 *ice);
+static int snd_ice1712_build_controls(struct snd_ice1712 *ice);
+
+static int PRO_RATE_LOCKED;
+static int PRO_RATE_RESET = 1;
+static unsigned int PRO_RATE_DEFAULT = 44100;
+
+/*
+ * Basic I/O
+ */
+
+/* check whether the clock mode is spdif-in */
+static inline int is_spdif_master(struct snd_ice1712 *ice)
+{
+ return (inb(ICEMT(ice, RATE)) & ICE1712_SPDIF_MASTER) ? 1 : 0;
+}
+
+static inline int is_pro_rate_locked(struct snd_ice1712 *ice)
+{
+ return is_spdif_master(ice) || PRO_RATE_LOCKED;
+}
+
+static inline void snd_ice1712_ds_write(struct snd_ice1712 *ice, u8 channel, u8 addr, u32 data)
+{
+ outb((channel << 4) | addr, ICEDS(ice, INDEX));
+ outl(data, ICEDS(ice, DATA));
+}
+
+static inline u32 snd_ice1712_ds_read(struct snd_ice1712 *ice, u8 channel, u8 addr)
+{
+ outb((channel << 4) | addr, ICEDS(ice, INDEX));
+ return inl(ICEDS(ice, DATA));
+}
+
+static void snd_ice1712_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct snd_ice1712 *ice = ac97->private_data;
+ int tm;
+ unsigned char old_cmd = 0;
+
+ for (tm = 0; tm < 0x10000; tm++) {
+ old_cmd = inb(ICEREG(ice, AC97_CMD));
+ if (old_cmd & (ICE1712_AC97_WRITE | ICE1712_AC97_READ))
+ continue;
+ if (!(old_cmd & ICE1712_AC97_READY))
+ continue;
+ break;
+ }
+ outb(reg, ICEREG(ice, AC97_INDEX));
+ outw(val, ICEREG(ice, AC97_DATA));
+ old_cmd &= ~(ICE1712_AC97_PBK_VSR | ICE1712_AC97_CAP_VSR);
+ outb(old_cmd | ICE1712_AC97_WRITE, ICEREG(ice, AC97_CMD));
+ for (tm = 0; tm < 0x10000; tm++)
+ if ((inb(ICEREG(ice, AC97_CMD)) & ICE1712_AC97_WRITE) == 0)
+ break;
+}
+
+static unsigned short snd_ice1712_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct snd_ice1712 *ice = ac97->private_data;
+ int tm;
+ unsigned char old_cmd = 0;
+
+ for (tm = 0; tm < 0x10000; tm++) {
+ old_cmd = inb(ICEREG(ice, AC97_CMD));
+ if (old_cmd & (ICE1712_AC97_WRITE | ICE1712_AC97_READ))
+ continue;
+ if (!(old_cmd & ICE1712_AC97_READY))
+ continue;
+ break;
+ }
+ outb(reg, ICEREG(ice, AC97_INDEX));
+ outb(old_cmd | ICE1712_AC97_READ, ICEREG(ice, AC97_CMD));
+ for (tm = 0; tm < 0x10000; tm++)
+ if ((inb(ICEREG(ice, AC97_CMD)) & ICE1712_AC97_READ) == 0)
+ break;
+ if (tm >= 0x10000) /* timeout */
+ return ~0;
+ return inw(ICEREG(ice, AC97_DATA));
+}
+
+/*
+ * pro ac97 section
+ */
+
+static void snd_ice1712_pro_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct snd_ice1712 *ice = ac97->private_data;
+ int tm;
+ unsigned char old_cmd = 0;
+
+ for (tm = 0; tm < 0x10000; tm++) {
+ old_cmd = inb(ICEMT(ice, AC97_CMD));
+ if (old_cmd & (ICE1712_AC97_WRITE | ICE1712_AC97_READ))
+ continue;
+ if (!(old_cmd & ICE1712_AC97_READY))
+ continue;
+ break;
+ }
+ outb(reg, ICEMT(ice, AC97_INDEX));
+ outw(val, ICEMT(ice, AC97_DATA));
+ old_cmd &= ~(ICE1712_AC97_PBK_VSR | ICE1712_AC97_CAP_VSR);
+ outb(old_cmd | ICE1712_AC97_WRITE, ICEMT(ice, AC97_CMD));
+ for (tm = 0; tm < 0x10000; tm++)
+ if ((inb(ICEMT(ice, AC97_CMD)) & ICE1712_AC97_WRITE) == 0)
+ break;
+}
+
+
+static unsigned short snd_ice1712_pro_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct snd_ice1712 *ice = ac97->private_data;
+ int tm;
+ unsigned char old_cmd = 0;
+
+ for (tm = 0; tm < 0x10000; tm++) {
+ old_cmd = inb(ICEMT(ice, AC97_CMD));
+ if (old_cmd & (ICE1712_AC97_WRITE | ICE1712_AC97_READ))
+ continue;
+ if (!(old_cmd & ICE1712_AC97_READY))
+ continue;
+ break;
+ }
+ outb(reg, ICEMT(ice, AC97_INDEX));
+ outb(old_cmd | ICE1712_AC97_READ, ICEMT(ice, AC97_CMD));
+ for (tm = 0; tm < 0x10000; tm++)
+ if ((inb(ICEMT(ice, AC97_CMD)) & ICE1712_AC97_READ) == 0)
+ break;
+ if (tm >= 0x10000) /* timeout */
+ return ~0;
+ return inw(ICEMT(ice, AC97_DATA));
+}
+
+/*
+ * consumer ac97 digital mix
+ */
+#define snd_ice1712_digmix_route_ac97_info snd_ctl_boolean_mono_info
+
+static int snd_ice1712_digmix_route_ac97_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = inb(ICEMT(ice, MONITOR_ROUTECTRL)) & ICE1712_ROUTE_AC97 ? 1 : 0;
+ return 0;
+}
+
+static int snd_ice1712_digmix_route_ac97_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val, nval;
+
+ spin_lock_irq(&ice->reg_lock);
+ val = inb(ICEMT(ice, MONITOR_ROUTECTRL));
+ nval = val & ~ICE1712_ROUTE_AC97;
+ if (ucontrol->value.integer.value[0])
+ nval |= ICE1712_ROUTE_AC97;
+ outb(nval, ICEMT(ice, MONITOR_ROUTECTRL));
+ spin_unlock_irq(&ice->reg_lock);
+ return val != nval;
+}
+
+static struct snd_kcontrol_new snd_ice1712_mixer_digmix_route_ac97 __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Mixer To AC97",
+ .info = snd_ice1712_digmix_route_ac97_info,
+ .get = snd_ice1712_digmix_route_ac97_get,
+ .put = snd_ice1712_digmix_route_ac97_put,
+};
+
+
+/*
+ * gpio operations
+ */
+static void snd_ice1712_set_gpio_dir(struct snd_ice1712 *ice, unsigned int data)
+{
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION, data);
+ inb(ICEREG(ice, DATA)); /* dummy read for pci-posting */
+}
+
+static void snd_ice1712_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data)
+{
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, data);
+ inb(ICEREG(ice, DATA)); /* dummy read for pci-posting */
+}
+
+static unsigned int snd_ice1712_get_gpio_data(struct snd_ice1712 *ice)
+{
+ return snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+}
+
+static void snd_ice1712_set_gpio_data(struct snd_ice1712 *ice, unsigned int val)
+{
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, val);
+ inb(ICEREG(ice, DATA)); /* dummy read for pci-posting */
+}
+
+/*
+ *
+ * CS8427 interface
+ *
+ */
+
+/*
+ * change the input clock selection
+ * spdif_clock = 1 - IEC958 input, 0 - Envy24
+ */
+static int snd_ice1712_cs8427_set_input_clock(struct snd_ice1712 *ice, int spdif_clock)
+{
+ unsigned char reg[2] = { 0x80 | 4, 0 }; /* CS8427 auto increment | register number 4 + data */
+ unsigned char val, nval;
+ int res = 0;
+
+ snd_i2c_lock(ice->i2c);
+ if (snd_i2c_sendbytes(ice->cs8427, reg, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ if (snd_i2c_readbytes(ice->cs8427, &val, 1) != 1) {
+ snd_i2c_unlock(ice->i2c);
+ return -EIO;
+ }
+ nval = val & 0xf0;
+ if (spdif_clock)
+ nval |= 0x01;
+ else
+ nval |= 0x04;
+ if (val != nval) {
+ reg[1] = nval;
+ if (snd_i2c_sendbytes(ice->cs8427, reg, 2) != 2) {
+ res = -EIO;
+ } else {
+ res++;
+ }
+ }
+ snd_i2c_unlock(ice->i2c);
+ return res;
+}
+
+/*
+ * spdif callbacks
+ */
+static void open_cs8427(struct snd_ice1712 *ice, struct snd_pcm_substream *substream)
+{
+ snd_cs8427_iec958_active(ice->cs8427, 1);
+}
+
+static void close_cs8427(struct snd_ice1712 *ice, struct snd_pcm_substream *substream)
+{
+ snd_cs8427_iec958_active(ice->cs8427, 0);
+}
+
+static void setup_cs8427(struct snd_ice1712 *ice, int rate)
+{
+ snd_cs8427_iec958_pcm(ice->cs8427, rate);
+}
+
+/*
+ * create and initialize callbacks for cs8427 interface
+ */
+int __devinit snd_ice1712_init_cs8427(struct snd_ice1712 *ice, int addr)
+{
+ int err;
+
+ err = snd_cs8427_create(ice->i2c, addr,
+ (ice->cs8427_timeout * HZ) / 1000, &ice->cs8427);
+ if (err < 0) {
+ snd_printk(KERN_ERR "CS8427 initialization failed\n");
+ return err;
+ }
+ ice->spdif.ops.open = open_cs8427;
+ ice->spdif.ops.close = close_cs8427;
+ ice->spdif.ops.setup_rate = setup_cs8427;
+ return 0;
+}
+
+static void snd_ice1712_set_input_clock_source(struct snd_ice1712 *ice, int spdif_is_master)
+{
+ /* change CS8427 clock source too */
+ if (ice->cs8427)
+ snd_ice1712_cs8427_set_input_clock(ice, spdif_is_master);
+ /* notify ak4524 chip as well */
+ if (spdif_is_master) {
+ unsigned int i;
+ for (i = 0; i < ice->akm_codecs; i++) {
+ if (ice->akm[i].ops.set_rate_val)
+ ice->akm[i].ops.set_rate_val(&ice->akm[i], 0);
+ }
+ }
+}
+
+/*
+ * Interrupt handler
+ */
+
+static irqreturn_t snd_ice1712_interrupt(int irq, void *dev_id)
+{
+ struct snd_ice1712 *ice = dev_id;
+ unsigned char status;
+ int handled = 0;
+
+ while (1) {
+ status = inb(ICEREG(ice, IRQSTAT));
+ if (status == 0)
+ break;
+ handled = 1;
+ if (status & ICE1712_IRQ_MPU1) {
+ if (ice->rmidi[0])
+ snd_mpu401_uart_interrupt(irq, ice->rmidi[0]->private_data);
+ outb(ICE1712_IRQ_MPU1, ICEREG(ice, IRQSTAT));
+ status &= ~ICE1712_IRQ_MPU1;
+ }
+ if (status & ICE1712_IRQ_TIMER)
+ outb(ICE1712_IRQ_TIMER, ICEREG(ice, IRQSTAT));
+ if (status & ICE1712_IRQ_MPU2) {
+ if (ice->rmidi[1])
+ snd_mpu401_uart_interrupt(irq, ice->rmidi[1]->private_data);
+ outb(ICE1712_IRQ_MPU2, ICEREG(ice, IRQSTAT));
+ status &= ~ICE1712_IRQ_MPU2;
+ }
+ if (status & ICE1712_IRQ_PROPCM) {
+ unsigned char mtstat = inb(ICEMT(ice, IRQ));
+ if (mtstat & ICE1712_MULTI_PBKSTATUS) {
+ if (ice->playback_pro_substream)
+ snd_pcm_period_elapsed(ice->playback_pro_substream);
+ outb(ICE1712_MULTI_PBKSTATUS, ICEMT(ice, IRQ));
+ }
+ if (mtstat & ICE1712_MULTI_CAPSTATUS) {
+ if (ice->capture_pro_substream)
+ snd_pcm_period_elapsed(ice->capture_pro_substream);
+ outb(ICE1712_MULTI_CAPSTATUS, ICEMT(ice, IRQ));
+ }
+ }
+ if (status & ICE1712_IRQ_FM)
+ outb(ICE1712_IRQ_FM, ICEREG(ice, IRQSTAT));
+ if (status & ICE1712_IRQ_PBKDS) {
+ u32 idx;
+ u16 pbkstatus;
+ struct snd_pcm_substream *substream;
+ pbkstatus = inw(ICEDS(ice, INTSTAT));
+ /* printk("pbkstatus = 0x%x\n", pbkstatus); */
+ for (idx = 0; idx < 6; idx++) {
+ if ((pbkstatus & (3 << (idx * 2))) == 0)
+ continue;
+ substream = ice->playback_con_substream_ds[idx];
+ if (substream != NULL)
+ snd_pcm_period_elapsed(substream);
+ outw(3 << (idx * 2), ICEDS(ice, INTSTAT));
+ }
+ outb(ICE1712_IRQ_PBKDS, ICEREG(ice, IRQSTAT));
+ }
+ if (status & ICE1712_IRQ_CONCAP) {
+ if (ice->capture_con_substream)
+ snd_pcm_period_elapsed(ice->capture_con_substream);
+ outb(ICE1712_IRQ_CONCAP, ICEREG(ice, IRQSTAT));
+ }
+ if (status & ICE1712_IRQ_CONPBK) {
+ if (ice->playback_con_substream)
+ snd_pcm_period_elapsed(ice->playback_con_substream);
+ outb(ICE1712_IRQ_CONPBK, ICEREG(ice, IRQSTAT));
+ }
+ }
+ return IRQ_RETVAL(handled);
+}
+
+
+/*
+ * PCM part - misc
+ */
+
+static int snd_ice1712_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_ice1712_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/*
+ * PCM part - consumer I/O
+ */
+
+static int snd_ice1712_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ int result = 0;
+ u32 tmp;
+
+ spin_lock(&ice->reg_lock);
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_PBK_CTRL);
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ tmp |= 1;
+ } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ tmp &= ~1;
+ } else if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH) {
+ tmp |= 2;
+ } else if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) {
+ tmp &= ~2;
+ } else {
+ result = -EINVAL;
+ }
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_CTRL, tmp);
+ spin_unlock(&ice->reg_lock);
+ return result;
+}
+
+static int snd_ice1712_playback_ds_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ int result = 0;
+ u32 tmp;
+
+ spin_lock(&ice->reg_lock);
+ tmp = snd_ice1712_ds_read(ice, substream->number * 2, ICE1712_DSC_CONTROL);
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ tmp |= 1;
+ } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ tmp &= ~1;
+ } else if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH) {
+ tmp |= 2;
+ } else if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) {
+ tmp &= ~2;
+ } else {
+ result = -EINVAL;
+ }
+ snd_ice1712_ds_write(ice, substream->number * 2, ICE1712_DSC_CONTROL, tmp);
+ spin_unlock(&ice->reg_lock);
+ return result;
+}
+
+static int snd_ice1712_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ int result = 0;
+ u8 tmp;
+
+ spin_lock(&ice->reg_lock);
+ tmp = snd_ice1712_read(ice, ICE1712_IREG_CAP_CTRL);
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ tmp |= 1;
+ } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ tmp &= ~1;
+ } else {
+ result = -EINVAL;
+ }
+ snd_ice1712_write(ice, ICE1712_IREG_CAP_CTRL, tmp);
+ spin_unlock(&ice->reg_lock);
+ return result;
+}
+
+static int snd_ice1712_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ u32 period_size, buf_size, rate, tmp;
+
+ period_size = (snd_pcm_lib_period_bytes(substream) >> 2) - 1;
+ buf_size = snd_pcm_lib_buffer_bytes(substream) - 1;
+ tmp = 0x0000;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ tmp |= 0x10;
+ if (runtime->channels == 2)
+ tmp |= 0x08;
+ rate = (runtime->rate * 8192) / 375;
+ if (rate > 0x000fffff)
+ rate = 0x000fffff;
+ spin_lock_irq(&ice->reg_lock);
+ outb(0, ice->ddma_port + 15);
+ outb(ICE1712_DMA_MODE_WRITE | ICE1712_DMA_AUTOINIT, ice->ddma_port + 0x0b);
+ outl(runtime->dma_addr, ice->ddma_port + 0);
+ outw(buf_size, ice->ddma_port + 4);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_RATE_LO, rate & 0xff);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_RATE_MID, (rate >> 8) & 0xff);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_RATE_HI, (rate >> 16) & 0xff);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_CTRL, tmp);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_COUNT_LO, period_size & 0xff);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_COUNT_HI, period_size >> 8);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_LEFT, 0);
+ snd_ice1712_write(ice, ICE1712_IREG_PBK_RIGHT, 0);
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static int snd_ice1712_playback_ds_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ u32 period_size, buf_size, rate, tmp, chn;
+
+ period_size = snd_pcm_lib_period_bytes(substream) - 1;
+ buf_size = snd_pcm_lib_buffer_bytes(substream) - 1;
+ tmp = 0x0064;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ tmp &= ~0x04;
+ if (runtime->channels == 2)
+ tmp |= 0x08;
+ rate = (runtime->rate * 8192) / 375;
+ if (rate > 0x000fffff)
+ rate = 0x000fffff;
+ ice->playback_con_active_buf[substream->number] = 0;
+ ice->playback_con_virt_addr[substream->number] = runtime->dma_addr;
+ chn = substream->number * 2;
+ spin_lock_irq(&ice->reg_lock);
+ snd_ice1712_ds_write(ice, chn, ICE1712_DSC_ADDR0, runtime->dma_addr);
+ snd_ice1712_ds_write(ice, chn, ICE1712_DSC_COUNT0, period_size);
+ snd_ice1712_ds_write(ice, chn, ICE1712_DSC_ADDR1, runtime->dma_addr + (runtime->periods > 1 ? period_size + 1 : 0));
+ snd_ice1712_ds_write(ice, chn, ICE1712_DSC_COUNT1, period_size);
+ snd_ice1712_ds_write(ice, chn, ICE1712_DSC_RATE, rate);
+ snd_ice1712_ds_write(ice, chn, ICE1712_DSC_VOLUME, 0);
+ snd_ice1712_ds_write(ice, chn, ICE1712_DSC_CONTROL, tmp);
+ if (runtime->channels == 2) {
+ snd_ice1712_ds_write(ice, chn + 1, ICE1712_DSC_RATE, rate);
+ snd_ice1712_ds_write(ice, chn + 1, ICE1712_DSC_VOLUME, 0);
+ }
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static int snd_ice1712_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ u32 period_size, buf_size;
+ u8 tmp;
+
+ period_size = (snd_pcm_lib_period_bytes(substream) >> 2) - 1;
+ buf_size = snd_pcm_lib_buffer_bytes(substream) - 1;
+ tmp = 0x06;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ tmp &= ~0x04;
+ if (runtime->channels == 2)
+ tmp &= ~0x02;
+ spin_lock_irq(&ice->reg_lock);
+ outl(ice->capture_con_virt_addr = runtime->dma_addr, ICEREG(ice, CONCAP_ADDR));
+ outw(buf_size, ICEREG(ice, CONCAP_COUNT));
+ snd_ice1712_write(ice, ICE1712_IREG_CAP_COUNT_HI, period_size >> 8);
+ snd_ice1712_write(ice, ICE1712_IREG_CAP_COUNT_LO, period_size & 0xff);
+ snd_ice1712_write(ice, ICE1712_IREG_CAP_CTRL, tmp);
+ spin_unlock_irq(&ice->reg_lock);
+ snd_ac97_set_rate(ice->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_ice1712_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ size_t ptr;
+
+ if (!(snd_ice1712_read(ice, ICE1712_IREG_PBK_CTRL) & 1))
+ return 0;
+ ptr = runtime->buffer_size - inw(ice->ddma_port + 4);
+ if (ptr == runtime->buffer_size)
+ ptr = 0;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ u8 addr;
+ size_t ptr;
+
+ if (!(snd_ice1712_ds_read(ice, substream->number * 2, ICE1712_DSC_CONTROL) & 1))
+ return 0;
+ if (ice->playback_con_active_buf[substream->number])
+ addr = ICE1712_DSC_ADDR1;
+ else
+ addr = ICE1712_DSC_ADDR0;
+ ptr = snd_ice1712_ds_read(ice, substream->number * 2, addr) -
+ ice->playback_con_virt_addr[substream->number];
+ if (ptr == substream->runtime->buffer_size)
+ ptr = 0;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(snd_ice1712_read(ice, ICE1712_IREG_CAP_CTRL) & 1))
+ return 0;
+ ptr = inl(ICEREG(ice, CONCAP_ADDR)) - ice->capture_con_virt_addr;
+ if (ptr == substream->runtime->buffer_size)
+ ptr = 0;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static const struct snd_pcm_hardware snd_ice1712_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (64*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (64*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static const struct snd_pcm_hardware snd_ice1712_playback_ds = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static const struct snd_pcm_hardware snd_ice1712_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (64*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (64*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_ice1712_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ ice->playback_con_substream = substream;
+ runtime->hw = snd_ice1712_playback;
+ return 0;
+}
+
+static int snd_ice1712_playback_ds_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ u32 tmp;
+
+ ice->playback_con_substream_ds[substream->number] = substream;
+ runtime->hw = snd_ice1712_playback_ds;
+ spin_lock_irq(&ice->reg_lock);
+ tmp = inw(ICEDS(ice, INTMASK)) & ~(1 << (substream->number * 2));
+ outw(tmp, ICEDS(ice, INTMASK));
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static int snd_ice1712_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ ice->capture_con_substream = substream;
+ runtime->hw = snd_ice1712_capture;
+ runtime->hw.rates = ice->ac97->rates[AC97_RATES_ADC];
+ if (!(runtime->hw.rates & SNDRV_PCM_RATE_8000))
+ runtime->hw.rate_min = 48000;
+ return 0;
+}
+
+static int snd_ice1712_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ ice->playback_con_substream = NULL;
+ return 0;
+}
+
+static int snd_ice1712_playback_ds_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ u32 tmp;
+
+ spin_lock_irq(&ice->reg_lock);
+ tmp = inw(ICEDS(ice, INTMASK)) | (3 << (substream->number * 2));
+ outw(tmp, ICEDS(ice, INTMASK));
+ spin_unlock_irq(&ice->reg_lock);
+ ice->playback_con_substream_ds[substream->number] = NULL;
+ return 0;
+}
+
+static int snd_ice1712_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ ice->capture_con_substream = NULL;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ice1712_playback_ops = {
+ .open = snd_ice1712_playback_open,
+ .close = snd_ice1712_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ice1712_hw_params,
+ .hw_free = snd_ice1712_hw_free,
+ .prepare = snd_ice1712_playback_prepare,
+ .trigger = snd_ice1712_playback_trigger,
+ .pointer = snd_ice1712_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_ice1712_playback_ds_ops = {
+ .open = snd_ice1712_playback_ds_open,
+ .close = snd_ice1712_playback_ds_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ice1712_hw_params,
+ .hw_free = snd_ice1712_hw_free,
+ .prepare = snd_ice1712_playback_ds_prepare,
+ .trigger = snd_ice1712_playback_ds_trigger,
+ .pointer = snd_ice1712_playback_ds_pointer,
+};
+
+static struct snd_pcm_ops snd_ice1712_capture_ops = {
+ .open = snd_ice1712_capture_open,
+ .close = snd_ice1712_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ice1712_hw_params,
+ .hw_free = snd_ice1712_hw_free,
+ .prepare = snd_ice1712_capture_prepare,
+ .trigger = snd_ice1712_capture_trigger,
+ .pointer = snd_ice1712_capture_pointer,
+};
+
+static int __devinit snd_ice1712_pcm(struct snd_ice1712 *ice, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ err = snd_pcm_new(ice->card, "ICE1712 consumer", device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ice1712_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ice1712_capture_ops);
+
+ pcm->private_data = ice;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "ICE1712 consumer");
+ ice->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ice->pci), 64*1024, 64*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ printk(KERN_WARNING "Consumer PCM code does not work well at the moment --jk\n");
+
+ return 0;
+}
+
+static int __devinit snd_ice1712_pcm_ds(struct snd_ice1712 *ice, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ err = snd_pcm_new(ice->card, "ICE1712 consumer (DS)", device, 6, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ice1712_playback_ds_ops);
+
+ pcm->private_data = ice;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "ICE1712 consumer (DS)");
+ ice->pcm_ds = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ice->pci), 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+/*
+ * PCM code - professional part (multitrack)
+ */
+
+static unsigned int rates[] = { 8000, 9600, 11025, 12000, 16000, 22050, 24000,
+ 32000, 44100, 48000, 64000, 88200, 96000 };
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static int snd_ice1712_pro_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ {
+ unsigned int what;
+ unsigned int old;
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -EINVAL;
+ what = ICE1712_PLAYBACK_PAUSE;
+ snd_pcm_trigger_done(substream, substream);
+ spin_lock(&ice->reg_lock);
+ old = inl(ICEMT(ice, PLAYBACK_CONTROL));
+ if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH)
+ old |= what;
+ else
+ old &= ~what;
+ outl(old, ICEMT(ice, PLAYBACK_CONTROL));
+ spin_unlock(&ice->reg_lock);
+ break;
+ }
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_STOP:
+ {
+ unsigned int what = 0;
+ unsigned int old;
+ struct snd_pcm_substream *s;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == ice->playback_pro_substream) {
+ what |= ICE1712_PLAYBACK_START;
+ snd_pcm_trigger_done(s, substream);
+ } else if (s == ice->capture_pro_substream) {
+ what |= ICE1712_CAPTURE_START_SHADOW;
+ snd_pcm_trigger_done(s, substream);
+ }
+ }
+ spin_lock(&ice->reg_lock);
+ old = inl(ICEMT(ice, PLAYBACK_CONTROL));
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ old |= what;
+ else
+ old &= ~what;
+ outl(old, ICEMT(ice, PLAYBACK_CONTROL));
+ spin_unlock(&ice->reg_lock);
+ break;
+ }
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ */
+static void snd_ice1712_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, int force)
+{
+ unsigned long flags;
+ unsigned char val, old;
+ unsigned int i;
+
+ switch (rate) {
+ case 8000: val = 6; break;
+ case 9600: val = 3; break;
+ case 11025: val = 10; break;
+ case 12000: val = 2; break;
+ case 16000: val = 5; break;
+ case 22050: val = 9; break;
+ case 24000: val = 1; break;
+ case 32000: val = 4; break;
+ case 44100: val = 8; break;
+ case 48000: val = 0; break;
+ case 64000: val = 15; break;
+ case 88200: val = 11; break;
+ case 96000: val = 7; break;
+ default:
+ snd_BUG();
+ val = 0;
+ rate = 48000;
+ break;
+ }
+
+ spin_lock_irqsave(&ice->reg_lock, flags);
+ if (inb(ICEMT(ice, PLAYBACK_CONTROL)) & (ICE1712_CAPTURE_START_SHADOW|
+ ICE1712_PLAYBACK_PAUSE|
+ ICE1712_PLAYBACK_START)) {
+__out:
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+ return;
+ }
+ if (!force && is_pro_rate_locked(ice))
+ goto __out;
+
+ old = inb(ICEMT(ice, RATE));
+ if (!force && old == val)
+ goto __out;
+ outb(val, ICEMT(ice, RATE));
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+
+ if (ice->gpio.set_pro_rate)
+ ice->gpio.set_pro_rate(ice, rate);
+ for (i = 0; i < ice->akm_codecs; i++) {
+ if (ice->akm[i].ops.set_rate_val)
+ ice->akm[i].ops.set_rate_val(&ice->akm[i], rate);
+ }
+ if (ice->spdif.ops.setup_rate)
+ ice->spdif.ops.setup_rate(ice, rate);
+}
+
+static int snd_ice1712_playback_pro_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ ice->playback_pro_size = snd_pcm_lib_buffer_bytes(substream);
+ spin_lock_irq(&ice->reg_lock);
+ outl(substream->runtime->dma_addr, ICEMT(ice, PLAYBACK_ADDR));
+ outw((ice->playback_pro_size >> 2) - 1, ICEMT(ice, PLAYBACK_SIZE));
+ outw((snd_pcm_lib_period_bytes(substream) >> 2) - 1, ICEMT(ice, PLAYBACK_COUNT));
+ spin_unlock_irq(&ice->reg_lock);
+
+ return 0;
+}
+
+static int snd_ice1712_playback_pro_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ snd_ice1712_set_pro_rate(ice, params_rate(hw_params), 0);
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_ice1712_capture_pro_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ ice->capture_pro_size = snd_pcm_lib_buffer_bytes(substream);
+ spin_lock_irq(&ice->reg_lock);
+ outl(substream->runtime->dma_addr, ICEMT(ice, CAPTURE_ADDR));
+ outw((ice->capture_pro_size >> 2) - 1, ICEMT(ice, CAPTURE_SIZE));
+ outw((snd_pcm_lib_period_bytes(substream) >> 2) - 1, ICEMT(ice, CAPTURE_COUNT));
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static int snd_ice1712_capture_pro_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ snd_ice1712_set_pro_rate(ice, params_rate(hw_params), 0);
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static snd_pcm_uframes_t snd_ice1712_playback_pro_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_PLAYBACK_START))
+ return 0;
+ ptr = ice->playback_pro_size - (inw(ICEMT(ice, PLAYBACK_SIZE)) << 2);
+ if (ptr == substream->runtime->buffer_size)
+ ptr = 0;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_CAPTURE_START_SHADOW))
+ return 0;
+ ptr = ice->capture_pro_size - (inw(ICEMT(ice, CAPTURE_SIZE)) << 2);
+ if (ptr == substream->runtime->buffer_size)
+ ptr = 0;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static const struct snd_pcm_hardware snd_ice1712_playback_pro = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_96000,
+ .rate_min = 4000,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 10,
+ .buffer_bytes_max = (256*1024),
+ .period_bytes_min = 10 * 4 * 2,
+ .period_bytes_max = 131040,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static const struct snd_pcm_hardware snd_ice1712_capture_pro = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_96000,
+ .rate_min = 4000,
+ .rate_max = 96000,
+ .channels_min = 12,
+ .channels_max = 12,
+ .buffer_bytes_max = (256*1024),
+ .period_bytes_min = 12 * 4 * 2,
+ .period_bytes_max = 131040,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_ice1712_playback_pro_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ ice->playback_pro_substream = substream;
+ runtime->hw = snd_ice1712_playback_pro;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates);
+
+ if (ice->spdif.ops.open)
+ ice->spdif.ops.open(ice, substream);
+
+ return 0;
+}
+
+static int snd_ice1712_capture_pro_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ ice->capture_pro_substream = substream;
+ runtime->hw = snd_ice1712_capture_pro;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates);
+ return 0;
+}
+
+static int snd_ice1712_playback_pro_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ if (PRO_RATE_RESET)
+ snd_ice1712_set_pro_rate(ice, PRO_RATE_DEFAULT, 0);
+ ice->playback_pro_substream = NULL;
+ if (ice->spdif.ops.close)
+ ice->spdif.ops.close(ice, substream);
+
+ return 0;
+}
+
+static int snd_ice1712_capture_pro_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ if (PRO_RATE_RESET)
+ snd_ice1712_set_pro_rate(ice, PRO_RATE_DEFAULT, 0);
+ ice->capture_pro_substream = NULL;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ice1712_playback_pro_ops = {
+ .open = snd_ice1712_playback_pro_open,
+ .close = snd_ice1712_playback_pro_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ice1712_playback_pro_hw_params,
+ .hw_free = snd_ice1712_hw_free,
+ .prepare = snd_ice1712_playback_pro_prepare,
+ .trigger = snd_ice1712_pro_trigger,
+ .pointer = snd_ice1712_playback_pro_pointer,
+};
+
+static struct snd_pcm_ops snd_ice1712_capture_pro_ops = {
+ .open = snd_ice1712_capture_pro_open,
+ .close = snd_ice1712_capture_pro_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ice1712_capture_pro_hw_params,
+ .hw_free = snd_ice1712_hw_free,
+ .prepare = snd_ice1712_capture_pro_prepare,
+ .trigger = snd_ice1712_pro_trigger,
+ .pointer = snd_ice1712_capture_pro_pointer,
+};
+
+static int __devinit snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ err = snd_pcm_new(ice->card, "ICE1712 multi", device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ice1712_playback_pro_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ice1712_capture_pro_ops);
+
+ pcm->private_data = ice;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "ICE1712 multi");
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ice->pci), 256*1024, 256*1024);
+
+ ice->pcm_pro = pcm;
+ if (rpcm)
+ *rpcm = pcm;
+
+ if (ice->cs8427) {
+ /* assign channels to iec958 */
+ err = snd_cs8427_iec958_build(ice->cs8427,
+ pcm->streams[0].substream,
+ pcm->streams[1].substream);
+ if (err < 0)
+ return err;
+ }
+
+ err = snd_ice1712_build_pro_mixer(ice);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * Mixer section
+ */
+
+static void snd_ice1712_update_volume(struct snd_ice1712 *ice, int index)
+{
+ unsigned int vol = ice->pro_volumes[index];
+ unsigned short val = 0;
+
+ val |= (vol & 0x8000) == 0 ? (96 - (vol & 0x7f)) : 0x7f;
+ val |= ((vol & 0x80000000) == 0 ? (96 - ((vol >> 16) & 0x7f)) : 0x7f) << 8;
+ outb(index, ICEMT(ice, MONITOR_INDEX));
+ outw(val, ICEMT(ice, MONITOR_VOLUME));
+}
+
+#define snd_ice1712_pro_mixer_switch_info snd_ctl_boolean_stereo_info
+
+static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
+
+ spin_lock_irq(&ice->reg_lock);
+ ucontrol->value.integer.value[0] =
+ !((ice->pro_volumes[priv_idx] >> 15) & 1);
+ ucontrol->value.integer.value[1] =
+ !((ice->pro_volumes[priv_idx] >> 31) & 1);
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static int snd_ice1712_pro_mixer_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
+ unsigned int nval, change;
+
+ nval = (ucontrol->value.integer.value[0] ? 0 : 0x00008000) |
+ (ucontrol->value.integer.value[1] ? 0 : 0x80000000);
+ spin_lock_irq(&ice->reg_lock);
+ nval |= ice->pro_volumes[priv_idx] & ~0x80008000;
+ change = nval != ice->pro_volumes[priv_idx];
+ ice->pro_volumes[priv_idx] = nval;
+ snd_ice1712_update_volume(ice, priv_idx);
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static int snd_ice1712_pro_mixer_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 96;
+ return 0;
+}
+
+static int snd_ice1712_pro_mixer_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
+
+ spin_lock_irq(&ice->reg_lock);
+ ucontrol->value.integer.value[0] =
+ (ice->pro_volumes[priv_idx] >> 0) & 127;
+ ucontrol->value.integer.value[1] =
+ (ice->pro_volumes[priv_idx] >> 16) & 127;
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static int snd_ice1712_pro_mixer_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int priv_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) +
+ kcontrol->private_value;
+ unsigned int nval, change;
+
+ nval = (ucontrol->value.integer.value[0] & 127) |
+ ((ucontrol->value.integer.value[1] & 127) << 16);
+ spin_lock_irq(&ice->reg_lock);
+ nval |= ice->pro_volumes[priv_idx] & ~0x007f007f;
+ change = nval != ice->pro_volumes[priv_idx];
+ ice->pro_volumes[priv_idx] = nval;
+ snd_ice1712_update_volume(ice, priv_idx);
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_playback, -14400, 150, 0);
+
+static struct snd_kcontrol_new snd_ice1712_multi_playback_ctrls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Playback Switch",
+ .info = snd_ice1712_pro_mixer_switch_info,
+ .get = snd_ice1712_pro_mixer_switch_get,
+ .put = snd_ice1712_pro_mixer_switch_put,
+ .private_value = 0,
+ .count = 10,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Multi Playback Volume",
+ .info = snd_ice1712_pro_mixer_volume_info,
+ .get = snd_ice1712_pro_mixer_volume_get,
+ .put = snd_ice1712_pro_mixer_volume_put,
+ .private_value = 0,
+ .count = 10,
+ .tlv = { .p = db_scale_playback }
+ },
+};
+
+static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_switch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "H/W Multi Capture Switch",
+ .info = snd_ice1712_pro_mixer_switch_info,
+ .get = snd_ice1712_pro_mixer_switch_get,
+ .put = snd_ice1712_pro_mixer_switch_put,
+ .private_value = 10,
+};
+
+static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, SWITCH),
+ .info = snd_ice1712_pro_mixer_switch_info,
+ .get = snd_ice1712_pro_mixer_switch_get,
+ .put = snd_ice1712_pro_mixer_switch_put,
+ .private_value = 18,
+ .count = 2,
+};
+
+static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "H/W Multi Capture Volume",
+ .info = snd_ice1712_pro_mixer_volume_info,
+ .get = snd_ice1712_pro_mixer_volume_get,
+ .put = snd_ice1712_pro_mixer_volume_put,
+ .private_value = 10,
+ .tlv = { .p = db_scale_playback }
+};
+
+static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, VOLUME),
+ .info = snd_ice1712_pro_mixer_volume_info,
+ .get = snd_ice1712_pro_mixer_volume_get,
+ .put = snd_ice1712_pro_mixer_volume_put,
+ .private_value = 18,
+ .count = 2,
+};
+
+static int __devinit snd_ice1712_build_pro_mixer(struct snd_ice1712 *ice)
+{
+ struct snd_card *card = ice->card;
+ unsigned int idx;
+ int err;
+
+ /* multi-channel mixer */
+ for (idx = 0; idx < ARRAY_SIZE(snd_ice1712_multi_playback_ctrls); idx++) {
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ice1712_multi_playback_ctrls[idx], ice));
+ if (err < 0)
+ return err;
+ }
+
+ if (ice->num_total_adcs > 0) {
+ struct snd_kcontrol_new tmp = snd_ice1712_multi_capture_analog_switch;
+ tmp.count = ice->num_total_adcs;
+ err = snd_ctl_add(card, snd_ctl_new1(&tmp, ice));
+ if (err < 0)
+ return err;
+ }
+
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ice1712_multi_capture_spdif_switch, ice));
+ if (err < 0)
+ return err;
+
+ if (ice->num_total_adcs > 0) {
+ struct snd_kcontrol_new tmp = snd_ice1712_multi_capture_analog_volume;
+ tmp.count = ice->num_total_adcs;
+ err = snd_ctl_add(card, snd_ctl_new1(&tmp, ice));
+ if (err < 0)
+ return err;
+ }
+
+ err = snd_ctl_add(card, snd_ctl_new1(&snd_ice1712_multi_capture_spdif_volume, ice));
+ if (err < 0)
+ return err;
+
+ /* initialize volumes */
+ for (idx = 0; idx < 10; idx++) {
+ ice->pro_volumes[idx] = 0x80008000; /* mute */
+ snd_ice1712_update_volume(ice, idx);
+ }
+ for (idx = 10; idx < 10 + ice->num_total_adcs; idx++) {
+ ice->pro_volumes[idx] = 0x80008000; /* mute */
+ snd_ice1712_update_volume(ice, idx);
+ }
+ for (idx = 18; idx < 20; idx++) {
+ ice->pro_volumes[idx] = 0x80008000; /* mute */
+ snd_ice1712_update_volume(ice, idx);
+ }
+ return 0;
+}
+
+static void snd_ice1712_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct snd_ice1712 *ice = ac97->private_data;
+ ice->ac97 = NULL;
+}
+
+static int __devinit snd_ice1712_ac97_mixer(struct snd_ice1712 *ice)
+{
+ int err, bus_num = 0;
+ struct snd_ac97_template ac97;
+ struct snd_ac97_bus *pbus;
+ static struct snd_ac97_bus_ops con_ops = {
+ .write = snd_ice1712_ac97_write,
+ .read = snd_ice1712_ac97_read,
+ };
+ static struct snd_ac97_bus_ops pro_ops = {
+ .write = snd_ice1712_pro_ac97_write,
+ .read = snd_ice1712_pro_ac97_read,
+ };
+
+ if (ice_has_con_ac97(ice)) {
+ err = snd_ac97_bus(ice->card, bus_num++, &con_ops, NULL, &pbus);
+ if (err < 0)
+ return err;
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = ice;
+ ac97.private_free = snd_ice1712_mixer_free_ac97;
+ err = snd_ac97_mixer(pbus, &ac97, &ice->ac97);
+ if (err < 0)
+ printk(KERN_WARNING "ice1712: cannot initialize ac97 for consumer, skipped\n");
+ else {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice));
+ if (err < 0)
+ return err;
+ return 0;
+ }
+ }
+
+ if (!(ice->eeprom.data[ICE_EEP1_ACLINK] & ICE1712_CFG_PRO_I2S)) {
+ err = snd_ac97_bus(ice->card, bus_num, &pro_ops, NULL, &pbus);
+ if (err < 0)
+ return err;
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = ice;
+ ac97.private_free = snd_ice1712_mixer_free_ac97;
+ err = snd_ac97_mixer(pbus, &ac97, &ice->ac97);
+ if (err < 0)
+ printk(KERN_WARNING "ice1712: cannot initialize pro ac97, skipped\n");
+ else
+ return 0;
+ }
+ /* I2S mixer only */
+ strcat(ice->card->mixername, "ICE1712 - multitrack");
+ return 0;
+}
+
+/*
+ *
+ */
+
+static inline unsigned int eeprom_double(struct snd_ice1712 *ice, int idx)
+{
+ return (unsigned int)ice->eeprom.data[idx] | ((unsigned int)ice->eeprom.data[idx + 1] << 8);
+}
+
+static void snd_ice1712_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = entry->private_data;
+ unsigned int idx;
+
+ snd_iprintf(buffer, "%s\n\n", ice->card->longname);
+ snd_iprintf(buffer, "EEPROM:\n");
+
+ snd_iprintf(buffer, " Subvendor : 0x%x\n", ice->eeprom.subvendor);
+ snd_iprintf(buffer, " Size : %i bytes\n", ice->eeprom.size);
+ snd_iprintf(buffer, " Version : %i\n", ice->eeprom.version);
+ snd_iprintf(buffer, " Codec : 0x%x\n", ice->eeprom.data[ICE_EEP1_CODEC]);
+ snd_iprintf(buffer, " ACLink : 0x%x\n", ice->eeprom.data[ICE_EEP1_ACLINK]);
+ snd_iprintf(buffer, " I2S ID : 0x%x\n", ice->eeprom.data[ICE_EEP1_I2SID]);
+ snd_iprintf(buffer, " S/PDIF : 0x%x\n", ice->eeprom.data[ICE_EEP1_SPDIF]);
+ snd_iprintf(buffer, " GPIO mask : 0x%x\n", ice->eeprom.gpiomask);
+ snd_iprintf(buffer, " GPIO state : 0x%x\n", ice->eeprom.gpiostate);
+ snd_iprintf(buffer, " GPIO direction : 0x%x\n", ice->eeprom.gpiodir);
+ snd_iprintf(buffer, " AC'97 main : 0x%x\n", eeprom_double(ice, ICE_EEP1_AC97_MAIN_LO));
+ snd_iprintf(buffer, " AC'97 pcm : 0x%x\n", eeprom_double(ice, ICE_EEP1_AC97_PCM_LO));
+ snd_iprintf(buffer, " AC'97 record : 0x%x\n", eeprom_double(ice, ICE_EEP1_AC97_REC_LO));
+ snd_iprintf(buffer, " AC'97 record src : 0x%x\n", ice->eeprom.data[ICE_EEP1_AC97_RECSRC]);
+ for (idx = 0; idx < 4; idx++)
+ snd_iprintf(buffer, " DAC ID #%i : 0x%x\n", idx, ice->eeprom.data[ICE_EEP1_DAC_ID + idx]);
+ for (idx = 0; idx < 4; idx++)
+ snd_iprintf(buffer, " ADC ID #%i : 0x%x\n", idx, ice->eeprom.data[ICE_EEP1_ADC_ID + idx]);
+ for (idx = 0x1c; idx < ice->eeprom.size; idx++)
+ snd_iprintf(buffer, " Extra #%02i : 0x%x\n", idx, ice->eeprom.data[idx]);
+
+ snd_iprintf(buffer, "\nRegisters:\n");
+ snd_iprintf(buffer, " PSDOUT03 : 0x%04x\n", (unsigned)inw(ICEMT(ice, ROUTE_PSDOUT03)));
+ snd_iprintf(buffer, " CAPTURE : 0x%08x\n", inl(ICEMT(ice, ROUTE_CAPTURE)));
+ snd_iprintf(buffer, " SPDOUT : 0x%04x\n", (unsigned)inw(ICEMT(ice, ROUTE_SPDOUT)));
+ snd_iprintf(buffer, " RATE : 0x%02x\n", (unsigned)inb(ICEMT(ice, RATE)));
+ snd_iprintf(buffer, " GPIO_DATA : 0x%02x\n", (unsigned)snd_ice1712_get_gpio_data(ice));
+ snd_iprintf(buffer, " GPIO_WRITE_MASK : 0x%02x\n", (unsigned)snd_ice1712_read(ice, ICE1712_IREG_GPIO_WRITE_MASK));
+ snd_iprintf(buffer, " GPIO_DIRECTION : 0x%02x\n", (unsigned)snd_ice1712_read(ice, ICE1712_IREG_GPIO_DIRECTION));
+}
+
+static void __devinit snd_ice1712_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(ice->card, "ice1712", &entry))
+ snd_info_set_text_ops(entry, ice, snd_ice1712_proc_read);
+}
+
+/*
+ *
+ */
+
+static int snd_ice1712_eeprom_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = sizeof(struct snd_ice1712_eeprom);
+ return 0;
+}
+
+static int snd_ice1712_eeprom_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ memcpy(ucontrol->value.bytes.data, &ice->eeprom, sizeof(ice->eeprom));
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ice1712_eeprom __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "ICE1712 EEPROM",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_ice1712_eeprom_info,
+ .get = snd_ice1712_eeprom_get
+};
+
+/*
+ */
+static int snd_ice1712_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ice1712_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ if (ice->spdif.ops.default_get)
+ ice->spdif.ops.default_get(ice, ucontrol);
+ return 0;
+}
+
+static int snd_ice1712_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ if (ice->spdif.ops.default_put)
+ return ice->spdif.ops.default_put(ice, ucontrol);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ice1712_spdif_default __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .info = snd_ice1712_spdif_info,
+ .get = snd_ice1712_spdif_default_get,
+ .put = snd_ice1712_spdif_default_put
+};
+
+static int snd_ice1712_spdif_maskc_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ if (ice->spdif.ops.default_get) {
+ ucontrol->value.iec958.status[0] = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_CON_NOT_COPYRIGHT |
+ IEC958_AES0_CON_EMPHASIS;
+ ucontrol->value.iec958.status[1] = IEC958_AES1_CON_ORIGINAL |
+ IEC958_AES1_CON_CATEGORY;
+ ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS;
+ } else {
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ ucontrol->value.iec958.status[4] = 0xff;
+ }
+ return 0;
+}
+
+static int snd_ice1712_spdif_maskp_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ if (ice->spdif.ops.default_get) {
+ ucontrol->value.iec958.status[0] = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_PRO_FS |
+ IEC958_AES0_PRO_EMPHASIS;
+ ucontrol->value.iec958.status[1] = IEC958_AES1_PRO_MODE;
+ } else {
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ ucontrol->value.iec958.status[4] = 0xff;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ice1712_spdif_maskc __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK),
+ .info = snd_ice1712_spdif_info,
+ .get = snd_ice1712_spdif_maskc_get,
+};
+
+static struct snd_kcontrol_new snd_ice1712_spdif_maskp __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK),
+ .info = snd_ice1712_spdif_info,
+ .get = snd_ice1712_spdif_maskp_get,
+};
+
+static int snd_ice1712_spdif_stream_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ if (ice->spdif.ops.stream_get)
+ ice->spdif.ops.stream_get(ice, ucontrol);
+ return 0;
+}
+
+static int snd_ice1712_spdif_stream_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ if (ice->spdif.ops.stream_put)
+ return ice->spdif.ops.stream_put(ice, ucontrol);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ice1712_spdif_stream __devinitdata =
+{
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE),
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM),
+ .info = snd_ice1712_spdif_info,
+ .get = snd_ice1712_spdif_stream_get,
+ .put = snd_ice1712_spdif_stream_put
+};
+
+int snd_ice1712_gpio_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char mask = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value & (1<<24)) ? 1 : 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ ucontrol->value.integer.value[0] =
+ (snd_ice1712_gpio_read(ice) & mask ? 1 : 0) ^ invert;
+ snd_ice1712_restore_gpio_status(ice);
+ return 0;
+}
+
+int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char mask = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value & (1<<24)) ? mask : 0;
+ unsigned int val, nval;
+
+ if (kcontrol->private_value & (1 << 31))
+ return -EPERM;
+ nval = (ucontrol->value.integer.value[0] ? mask : 0) ^ invert;
+ snd_ice1712_save_gpio_status(ice);
+ val = snd_ice1712_gpio_read(ice);
+ nval |= val & ~mask;
+ if (val != nval)
+ snd_ice1712_gpio_write(ice, nval);
+ snd_ice1712_restore_gpio_status(ice);
+ return val != nval;
+}
+
+/*
+ * rate
+ */
+static int snd_ice1712_pro_internal_clock_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "8000", /* 0: 6 */
+ "9600", /* 1: 3 */
+ "11025", /* 2: 10 */
+ "12000", /* 3: 2 */
+ "16000", /* 4: 5 */
+ "22050", /* 5: 9 */
+ "24000", /* 6: 1 */
+ "32000", /* 7: 4 */
+ "44100", /* 8: 8 */
+ "48000", /* 9: 0 */
+ "64000", /* 10: 15 */
+ "88200", /* 11: 11 */
+ "96000", /* 12: 7 */
+ "IEC958 Input", /* 13: -- */
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 14;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ice1712_pro_internal_clock_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ static const unsigned char xlate[16] = {
+ 9, 6, 3, 1, 7, 4, 0, 12, 8, 5, 2, 11, 255, 255, 255, 10
+ };
+ unsigned char val;
+
+ spin_lock_irq(&ice->reg_lock);
+ if (is_spdif_master(ice)) {
+ ucontrol->value.enumerated.item[0] = 13;
+ } else {
+ val = xlate[inb(ICEMT(ice, RATE)) & 15];
+ if (val == 255) {
+ snd_BUG();
+ val = 0;
+ }
+ ucontrol->value.enumerated.item[0] = val;
+ }
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static int snd_ice1712_pro_internal_clock_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ static const unsigned int xrate[13] = {
+ 8000, 9600, 11025, 12000, 16000, 22050, 24000,
+ 32000, 44100, 48000, 64000, 88200, 96000
+ };
+ unsigned char oval;
+ int change = 0;
+
+ spin_lock_irq(&ice->reg_lock);
+ oval = inb(ICEMT(ice, RATE));
+ if (ucontrol->value.enumerated.item[0] == 13) {
+ outb(oval | ICE1712_SPDIF_MASTER, ICEMT(ice, RATE));
+ } else {
+ PRO_RATE_DEFAULT = xrate[ucontrol->value.integer.value[0] % 13];
+ spin_unlock_irq(&ice->reg_lock);
+ snd_ice1712_set_pro_rate(ice, PRO_RATE_DEFAULT, 1);
+ spin_lock_irq(&ice->reg_lock);
+ }
+ change = inb(ICEMT(ice, RATE)) != oval;
+ spin_unlock_irq(&ice->reg_lock);
+
+ if ((oval & ICE1712_SPDIF_MASTER) !=
+ (inb(ICEMT(ice, RATE)) & ICE1712_SPDIF_MASTER))
+ snd_ice1712_set_input_clock_source(ice, is_spdif_master(ice));
+
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ice1712_pro_internal_clock __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Internal Clock",
+ .info = snd_ice1712_pro_internal_clock_info,
+ .get = snd_ice1712_pro_internal_clock_get,
+ .put = snd_ice1712_pro_internal_clock_put
+};
+
+static int snd_ice1712_pro_internal_clock_default_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "8000", /* 0: 6 */
+ "9600", /* 1: 3 */
+ "11025", /* 2: 10 */
+ "12000", /* 3: 2 */
+ "16000", /* 4: 5 */
+ "22050", /* 5: 9 */
+ "24000", /* 6: 1 */
+ "32000", /* 7: 4 */
+ "44100", /* 8: 8 */
+ "48000", /* 9: 0 */
+ "64000", /* 10: 15 */
+ "88200", /* 11: 11 */
+ "96000", /* 12: 7 */
+ /* "IEC958 Input", 13: -- */
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 13;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ice1712_pro_internal_clock_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int val;
+ static const unsigned int xrate[13] = {
+ 8000, 9600, 11025, 12000, 16000, 22050, 24000,
+ 32000, 44100, 48000, 64000, 88200, 96000
+ };
+
+ for (val = 0; val < 13; val++) {
+ if (xrate[val] == PRO_RATE_DEFAULT)
+ break;
+ }
+
+ ucontrol->value.enumerated.item[0] = val;
+ return 0;
+}
+
+static int snd_ice1712_pro_internal_clock_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ static const unsigned int xrate[13] = {
+ 8000, 9600, 11025, 12000, 16000, 22050, 24000,
+ 32000, 44100, 48000, 64000, 88200, 96000
+ };
+ unsigned char oval;
+ int change = 0;
+
+ oval = PRO_RATE_DEFAULT;
+ PRO_RATE_DEFAULT = xrate[ucontrol->value.integer.value[0] % 13];
+ change = PRO_RATE_DEFAULT != oval;
+
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ice1712_pro_internal_clock_default __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Internal Clock Default",
+ .info = snd_ice1712_pro_internal_clock_default_info,
+ .get = snd_ice1712_pro_internal_clock_default_get,
+ .put = snd_ice1712_pro_internal_clock_default_put
+};
+
+#define snd_ice1712_pro_rate_locking_info snd_ctl_boolean_mono_info
+
+static int snd_ice1712_pro_rate_locking_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = PRO_RATE_LOCKED;
+ return 0;
+}
+
+static int snd_ice1712_pro_rate_locking_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int change = 0, nval;
+
+ nval = ucontrol->value.integer.value[0] ? 1 : 0;
+ spin_lock_irq(&ice->reg_lock);
+ change = PRO_RATE_LOCKED != nval;
+ PRO_RATE_LOCKED = nval;
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ice1712_pro_rate_locking __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Rate Locking",
+ .info = snd_ice1712_pro_rate_locking_info,
+ .get = snd_ice1712_pro_rate_locking_get,
+ .put = snd_ice1712_pro_rate_locking_put
+};
+
+#define snd_ice1712_pro_rate_reset_info snd_ctl_boolean_mono_info
+
+static int snd_ice1712_pro_rate_reset_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = PRO_RATE_RESET;
+ return 0;
+}
+
+static int snd_ice1712_pro_rate_reset_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int change = 0, nval;
+
+ nval = ucontrol->value.integer.value[0] ? 1 : 0;
+ spin_lock_irq(&ice->reg_lock);
+ change = PRO_RATE_RESET != nval;
+ PRO_RATE_RESET = nval;
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ice1712_pro_rate_reset __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Rate Reset",
+ .info = snd_ice1712_pro_rate_reset_info,
+ .get = snd_ice1712_pro_rate_reset_get,
+ .put = snd_ice1712_pro_rate_reset_put
+};
+
+/*
+ * routing
+ */
+static int snd_ice1712_pro_route_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "PCM Out", /* 0 */
+ "H/W In 0", "H/W In 1", "H/W In 2", "H/W In 3", /* 1-4 */
+ "H/W In 4", "H/W In 5", "H/W In 6", "H/W In 7", /* 5-8 */
+ "IEC958 In L", "IEC958 In R", /* 9-10 */
+ "Digital Mixer", /* 11 - optional */
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items =
+ snd_ctl_get_ioffidx(kcontrol, &uinfo->id) < 2 ? 12 : 11;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ice1712_pro_route_analog_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int val, cval;
+
+ spin_lock_irq(&ice->reg_lock);
+ val = inw(ICEMT(ice, ROUTE_PSDOUT03));
+ cval = inl(ICEMT(ice, ROUTE_CAPTURE));
+ spin_unlock_irq(&ice->reg_lock);
+
+ val >>= ((idx % 2) * 8) + ((idx / 2) * 2);
+ val &= 3;
+ cval >>= ((idx / 2) * 8) + ((idx % 2) * 4);
+ if (val == 1 && idx < 2)
+ ucontrol->value.enumerated.item[0] = 11;
+ else if (val == 2)
+ ucontrol->value.enumerated.item[0] = (cval & 7) + 1;
+ else if (val == 3)
+ ucontrol->value.enumerated.item[0] = ((cval >> 3) & 1) + 9;
+ else
+ ucontrol->value.enumerated.item[0] = 0;
+ return 0;
+}
+
+static int snd_ice1712_pro_route_analog_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int change, shift;
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int val, old_val, nval;
+
+ /* update PSDOUT */
+ if (ucontrol->value.enumerated.item[0] >= 11)
+ nval = idx < 2 ? 1 : 0; /* dig mixer (or pcm) */
+ else if (ucontrol->value.enumerated.item[0] >= 9)
+ nval = 3; /* spdif in */
+ else if (ucontrol->value.enumerated.item[0] >= 1)
+ nval = 2; /* analog in */
+ else
+ nval = 0; /* pcm */
+ shift = ((idx % 2) * 8) + ((idx / 2) * 2);
+ spin_lock_irq(&ice->reg_lock);
+ val = old_val = inw(ICEMT(ice, ROUTE_PSDOUT03));
+ val &= ~(0x03 << shift);
+ val |= nval << shift;
+ change = val != old_val;
+ if (change)
+ outw(val, ICEMT(ice, ROUTE_PSDOUT03));
+ spin_unlock_irq(&ice->reg_lock);
+ if (nval < 2) /* dig mixer of pcm */
+ return change;
+
+ /* update CAPTURE */
+ spin_lock_irq(&ice->reg_lock);
+ val = old_val = inl(ICEMT(ice, ROUTE_CAPTURE));
+ shift = ((idx / 2) * 8) + ((idx % 2) * 4);
+ if (nval == 2) { /* analog in */
+ nval = ucontrol->value.enumerated.item[0] - 1;
+ val &= ~(0x07 << shift);
+ val |= nval << shift;
+ } else { /* spdif in */
+ nval = (ucontrol->value.enumerated.item[0] - 9) << 3;
+ val &= ~(0x08 << shift);
+ val |= nval << shift;
+ }
+ if (val != old_val) {
+ change = 1;
+ outl(val, ICEMT(ice, ROUTE_CAPTURE));
+ }
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static int snd_ice1712_pro_route_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int val, cval;
+ val = inw(ICEMT(ice, ROUTE_SPDOUT));
+ cval = (val >> (idx * 4 + 8)) & 0x0f;
+ val = (val >> (idx * 2)) & 0x03;
+ if (val == 1)
+ ucontrol->value.enumerated.item[0] = 11;
+ else if (val == 2)
+ ucontrol->value.enumerated.item[0] = (cval & 7) + 1;
+ else if (val == 3)
+ ucontrol->value.enumerated.item[0] = ((cval >> 3) & 1) + 9;
+ else
+ ucontrol->value.enumerated.item[0] = 0;
+ return 0;
+}
+
+static int snd_ice1712_pro_route_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int change, shift;
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ unsigned int val, old_val, nval;
+
+ /* update SPDOUT */
+ spin_lock_irq(&ice->reg_lock);
+ val = old_val = inw(ICEMT(ice, ROUTE_SPDOUT));
+ if (ucontrol->value.enumerated.item[0] >= 11)
+ nval = 1;
+ else if (ucontrol->value.enumerated.item[0] >= 9)
+ nval = 3;
+ else if (ucontrol->value.enumerated.item[0] >= 1)
+ nval = 2;
+ else
+ nval = 0;
+ shift = idx * 2;
+ val &= ~(0x03 << shift);
+ val |= nval << shift;
+ shift = idx * 4 + 8;
+ if (nval == 2) {
+ nval = ucontrol->value.enumerated.item[0] - 1;
+ val &= ~(0x07 << shift);
+ val |= nval << shift;
+ } else if (nval == 3) {
+ nval = (ucontrol->value.enumerated.item[0] - 9) << 3;
+ val &= ~(0x08 << shift);
+ val |= nval << shift;
+ }
+ change = val != old_val;
+ if (change)
+ outw(val, ICEMT(ice, ROUTE_SPDOUT));
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ice1712_mixer_pro_analog_route __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "H/W Playback Route",
+ .info = snd_ice1712_pro_route_info,
+ .get = snd_ice1712_pro_route_analog_get,
+ .put = snd_ice1712_pro_route_analog_put,
+};
+
+static struct snd_kcontrol_new snd_ice1712_mixer_pro_spdif_route __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route",
+ .info = snd_ice1712_pro_route_info,
+ .get = snd_ice1712_pro_route_spdif_get,
+ .put = snd_ice1712_pro_route_spdif_put,
+ .count = 2,
+};
+
+
+static int snd_ice1712_pro_volume_rate_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_ice1712_pro_volume_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = inb(ICEMT(ice, MONITOR_RATE));
+ return 0;
+}
+
+static int snd_ice1712_pro_volume_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int change;
+
+ spin_lock_irq(&ice->reg_lock);
+ change = inb(ICEMT(ice, MONITOR_RATE)) != ucontrol->value.integer.value[0];
+ outb(ucontrol->value.integer.value[0], ICEMT(ice, MONITOR_RATE));
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ice1712_mixer_pro_volume_rate __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Volume Rate",
+ .info = snd_ice1712_pro_volume_rate_info,
+ .get = snd_ice1712_pro_volume_rate_get,
+ .put = snd_ice1712_pro_volume_rate_put
+};
+
+static int snd_ice1712_pro_peak_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 22;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx;
+
+ spin_lock_irq(&ice->reg_lock);
+ for (idx = 0; idx < 22; idx++) {
+ outb(idx, ICEMT(ice, MONITOR_PEAKINDEX));
+ ucontrol->value.integer.value[idx] = inb(ICEMT(ice, MONITOR_PEAKDATA));
+ }
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Peak",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_ice1712_pro_peak_info,
+ .get = snd_ice1712_pro_peak_get
+};
+
+/*
+ *
+ */
+
+/*
+ * list of available boards
+ */
+static struct snd_ice1712_card_info *card_tables[] __devinitdata = {
+ snd_ice1712_hoontech_cards,
+ snd_ice1712_delta_cards,
+ snd_ice1712_ews_cards,
+ NULL,
+};
+
+static unsigned char __devinit snd_ice1712_read_i2c(struct snd_ice1712 *ice,
+ unsigned char dev,
+ unsigned char addr)
+{
+ long t = 0x10000;
+
+ outb(addr, ICEREG(ice, I2C_BYTE_ADDR));
+ outb(dev & ~ICE1712_I2C_WRITE, ICEREG(ice, I2C_DEV_ADDR));
+ while (t-- > 0 && (inb(ICEREG(ice, I2C_CTRL)) & ICE1712_I2C_BUSY)) ;
+ return inb(ICEREG(ice, I2C_DATA));
+}
+
+static int __devinit snd_ice1712_read_eeprom(struct snd_ice1712 *ice,
+ const char *modelname)
+{
+ int dev = 0xa0; /* EEPROM device address */
+ unsigned int i, size;
+ struct snd_ice1712_card_info * const *tbl, *c;
+
+ if (!modelname || !*modelname) {
+ ice->eeprom.subvendor = 0;
+ if ((inb(ICEREG(ice, I2C_CTRL)) & ICE1712_I2C_EEPROM) != 0)
+ ice->eeprom.subvendor = (snd_ice1712_read_i2c(ice, dev, 0x00) << 0) |
+ (snd_ice1712_read_i2c(ice, dev, 0x01) << 8) |
+ (snd_ice1712_read_i2c(ice, dev, 0x02) << 16) |
+ (snd_ice1712_read_i2c(ice, dev, 0x03) << 24);
+ if (ice->eeprom.subvendor == 0 ||
+ ice->eeprom.subvendor == (unsigned int)-1) {
+ /* invalid subvendor from EEPROM, try the PCI subststem ID instead */
+ u16 vendor, device;
+ pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID, &vendor);
+ pci_read_config_word(ice->pci, PCI_SUBSYSTEM_ID, &device);
+ ice->eeprom.subvendor = ((unsigned int)swab16(vendor) << 16) | swab16(device);
+ if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) {
+ printk(KERN_ERR "ice1712: No valid ID is found\n");
+ return -ENXIO;
+ }
+ }
+ }
+ for (tbl = card_tables; *tbl; tbl++) {
+ for (c = *tbl; c->subvendor; c++) {
+ if (modelname && c->model && !strcmp(modelname, c->model)) {
+ printk(KERN_INFO "ice1712: Using board model %s\n", c->name);
+ ice->eeprom.subvendor = c->subvendor;
+ } else if (c->subvendor != ice->eeprom.subvendor)
+ continue;
+ if (!c->eeprom_size || !c->eeprom_data)
+ goto found;
+ /* if the EEPROM is given by the driver, use it */
+ snd_printdd("using the defined eeprom..\n");
+ ice->eeprom.version = 1;
+ ice->eeprom.size = c->eeprom_size + 6;
+ memcpy(ice->eeprom.data, c->eeprom_data, c->eeprom_size);
+ goto read_skipped;
+ }
+ }
+ printk(KERN_WARNING "ice1712: No matching model found for ID 0x%x\n",
+ ice->eeprom.subvendor);
+
+ found:
+ ice->eeprom.size = snd_ice1712_read_i2c(ice, dev, 0x04);
+ if (ice->eeprom.size < 6)
+ ice->eeprom.size = 32; /* FIXME: any cards without the correct size? */
+ else if (ice->eeprom.size > 32) {
+ snd_printk(KERN_ERR "invalid EEPROM (size = %i)\n", ice->eeprom.size);
+ return -EIO;
+ }
+ ice->eeprom.version = snd_ice1712_read_i2c(ice, dev, 0x05);
+ if (ice->eeprom.version != 1) {
+ snd_printk(KERN_ERR "invalid EEPROM version %i\n",
+ ice->eeprom.version);
+ /* return -EIO; */
+ }
+ size = ice->eeprom.size - 6;
+ for (i = 0; i < size; i++)
+ ice->eeprom.data[i] = snd_ice1712_read_i2c(ice, dev, i + 6);
+
+ read_skipped:
+ ice->eeprom.gpiomask = ice->eeprom.data[ICE_EEP1_GPIO_MASK];
+ ice->eeprom.gpiostate = ice->eeprom.data[ICE_EEP1_GPIO_STATE];
+ ice->eeprom.gpiodir = ice->eeprom.data[ICE_EEP1_GPIO_DIR];
+
+ return 0;
+}
+
+
+
+static int __devinit snd_ice1712_chip_init(struct snd_ice1712 *ice)
+{
+ outb(ICE1712_RESET | ICE1712_NATIVE, ICEREG(ice, CONTROL));
+ udelay(200);
+ outb(ICE1712_NATIVE, ICEREG(ice, CONTROL));
+ udelay(200);
+ if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE &&
+ !ice->dxr_enable)
+ /* Set eeprom value to limit active ADCs and DACs to 6;
+ * Also disable AC97 as no hardware in standard 6fire card/box
+ * Note: DXR extensions are not currently supported
+ */
+ ice->eeprom.data[ICE_EEP1_CODEC] = 0x3a;
+ pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]);
+ pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]);
+ pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]);
+ pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]);
+ if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24) {
+ ice->gpio.write_mask = ice->eeprom.gpiomask;
+ ice->gpio.direction = ice->eeprom.gpiodir;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK,
+ ice->eeprom.gpiomask);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION,
+ ice->eeprom.gpiodir);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA,
+ ice->eeprom.gpiostate);
+ } else {
+ ice->gpio.write_mask = 0xc0;
+ ice->gpio.direction = 0xff;
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, 0xc0);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DIRECTION, 0xff);
+ snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA,
+ ICE1712_STDSP24_CLOCK_BIT);
+ }
+ snd_ice1712_write(ice, ICE1712_IREG_PRO_POWERDOWN, 0);
+ if (!(ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97)) {
+ outb(ICE1712_AC97_WARM, ICEREG(ice, AC97_CMD));
+ udelay(100);
+ outb(0, ICEREG(ice, AC97_CMD));
+ udelay(200);
+ snd_ice1712_write(ice, ICE1712_IREG_CONSUMER_POWERDOWN, 0);
+ }
+ snd_ice1712_set_pro_rate(ice, 48000, 1);
+
+ return 0;
+}
+
+int __devinit snd_ice1712_spdif_build_controls(struct snd_ice1712 *ice)
+{
+ int err;
+ struct snd_kcontrol *kctl;
+
+ if (snd_BUG_ON(!ice->pcm_pro))
+ return -EIO;
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_default, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm_pro->device;
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_maskc, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm_pro->device;
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_maskp, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm_pro->device;
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_ice1712_spdif_stream, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm_pro->device;
+ ice->spdif.stream_ctl = kctl;
+ return 0;
+}
+
+
+static int __devinit snd_ice1712_build_controls(struct snd_ice1712 *ice)
+{
+ int err;
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_eeprom, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_pro_internal_clock, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_pro_internal_clock_default, ice));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_pro_rate_locking, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_pro_rate_reset, ice));
+ if (err < 0)
+ return err;
+
+ if (ice->num_total_dacs > 0) {
+ struct snd_kcontrol_new tmp = snd_ice1712_mixer_pro_analog_route;
+ tmp.count = ice->num_total_dacs;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&tmp, ice));
+ if (err < 0)
+ return err;
+ }
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_spdif_route, ice));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_volume_rate, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_ice1712_free(struct snd_ice1712 *ice)
+{
+ if (!ice->port)
+ goto __hw_end;
+ /* mask all interrupts */
+ outb(0xc0, ICEMT(ice, IRQ));
+ outb(0xff, ICEREG(ice, IRQMASK));
+ /* --- */
+__hw_end:
+ if (ice->irq >= 0)
+ free_irq(ice->irq, ice);
+
+ if (ice->port)
+ pci_release_regions(ice->pci);
+ snd_ice1712_akm4xxx_free(ice);
+ pci_disable_device(ice->pci);
+ kfree(ice->spec);
+ kfree(ice);
+ return 0;
+}
+
+static int snd_ice1712_dev_free(struct snd_device *device)
+{
+ struct snd_ice1712 *ice = device->device_data;
+ return snd_ice1712_free(ice);
+}
+
+static int __devinit snd_ice1712_create(struct snd_card *card,
+ struct pci_dev *pci,
+ const char *modelname,
+ int omni,
+ int cs8427_timeout,
+ int dxr_enable,
+ struct snd_ice1712 **r_ice1712)
+{
+ struct snd_ice1712 *ice;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ice1712_dev_free,
+ };
+
+ *r_ice1712 = NULL;
+
+ /* enable PCI device */
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+ /* check, if we can restrict PCI DMA transfers to 28 bits */
+ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ ice = kzalloc(sizeof(*ice), GFP_KERNEL);
+ if (ice == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ ice->omni = omni ? 1 : 0;
+ if (cs8427_timeout < 1)
+ cs8427_timeout = 1;
+ else if (cs8427_timeout > 1000)
+ cs8427_timeout = 1000;
+ ice->cs8427_timeout = cs8427_timeout;
+ ice->dxr_enable = dxr_enable;
+ spin_lock_init(&ice->reg_lock);
+ mutex_init(&ice->gpio_mutex);
+ mutex_init(&ice->i2c_mutex);
+ mutex_init(&ice->open_mutex);
+ ice->gpio.set_mask = snd_ice1712_set_gpio_mask;
+ ice->gpio.set_dir = snd_ice1712_set_gpio_dir;
+ ice->gpio.set_data = snd_ice1712_set_gpio_data;
+ ice->gpio.get_data = snd_ice1712_get_gpio_data;
+
+ ice->spdif.cs8403_bits =
+ ice->spdif.cs8403_stream_bits = (0x01 | /* consumer format */
+ 0x10 | /* no emphasis */
+ 0x20); /* PCM encoder/decoder */
+ ice->card = card;
+ ice->pci = pci;
+ ice->irq = -1;
+ pci_set_master(pci);
+ pci_write_config_word(ice->pci, 0x40, 0x807f);
+ pci_write_config_word(ice->pci, 0x42, 0x0006);
+ snd_ice1712_proc_init(ice);
+ synchronize_irq(pci->irq);
+
+ err = pci_request_regions(pci, "ICE1712");
+ if (err < 0) {
+ kfree(ice);
+ pci_disable_device(pci);
+ return err;
+ }
+ ice->port = pci_resource_start(pci, 0);
+ ice->ddma_port = pci_resource_start(pci, 1);
+ ice->dmapath_port = pci_resource_start(pci, 2);
+ ice->profi_port = pci_resource_start(pci, 3);
+
+ if (request_irq(pci->irq, snd_ice1712_interrupt, IRQF_SHARED,
+ "ICE1712", ice)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_ice1712_free(ice);
+ return -EIO;
+ }
+
+ ice->irq = pci->irq;
+
+ if (snd_ice1712_read_eeprom(ice, modelname) < 0) {
+ snd_ice1712_free(ice);
+ return -EIO;
+ }
+ if (snd_ice1712_chip_init(ice) < 0) {
+ snd_ice1712_free(ice);
+ return -EIO;
+ }
+
+ /* unmask used interrupts */
+ outb(((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) == 0 ?
+ ICE1712_IRQ_MPU2 : 0) |
+ ((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97) ?
+ ICE1712_IRQ_PBKDS | ICE1712_IRQ_CONCAP | ICE1712_IRQ_CONPBK : 0),
+ ICEREG(ice, IRQMASK));
+ outb(0x00, ICEMT(ice, IRQ));
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops);
+ if (err < 0) {
+ snd_ice1712_free(ice);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_ice1712 = ice;
+ return 0;
+}
+
+
+/*
+ *
+ * Registration
+ *
+ */
+
+static struct snd_ice1712_card_info no_matched __devinitdata;
+
+static int __devinit snd_ice1712_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_ice1712 *ice;
+ int pcm_dev = 0, err;
+ struct snd_ice1712_card_info * const *tbl, *c;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "ICE1712");
+ strcpy(card->shortname, "ICEnsemble ICE1712");
+
+ err = snd_ice1712_create(card, pci, model[dev], omni[dev],
+ cs8427_timeout[dev], dxr_enable[dev], &ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ for (tbl = card_tables; *tbl; tbl++) {
+ for (c = *tbl; c->subvendor; c++) {
+ if (c->subvendor == ice->eeprom.subvendor) {
+ strcpy(card->shortname, c->name);
+ if (c->driver) /* specific driver? */
+ strcpy(card->driver, c->driver);
+ if (c->chip_init) {
+ err = c->chip_init(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+ goto __found;
+ }
+ }
+ }
+ c = &no_matched;
+ __found:
+
+ err = snd_ice1712_pcm_profi(ice, pcm_dev++, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if (ice_has_con_ac97(ice)) {
+ err = snd_ice1712_pcm(ice, pcm_dev++, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+
+ err = snd_ice1712_ac97_mixer(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ err = snd_ice1712_build_controls(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if (c->build_controls) {
+ err = c->build_controls(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+
+ if (ice_has_con_ac97(ice)) {
+ err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+
+ if (!c->no_mpu401) {
+ err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
+ ICEREG(ice, MPU1_CTRL),
+ (c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED),
+ ice->irq, 0, &ice->rmidi[0]);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if (c->mpu401_1_name)
+ /* Prefered name available in card_info */
+ snprintf(ice->rmidi[0]->name,
+ sizeof(ice->rmidi[0]->name),
+ "%s %d", c->mpu401_1_name, card->number);
+
+ if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) {
+ /* 2nd port used */
+ err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712,
+ ICEREG(ice, MPU2_CTRL),
+ (c->mpu401_2_info_flags | MPU401_INFO_INTEGRATED),
+ ice->irq, 0, &ice->rmidi[1]);
+
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if (c->mpu401_2_name)
+ /* Prefered name available in card_info */
+ snprintf(ice->rmidi[1]->name,
+ sizeof(ice->rmidi[1]->name),
+ "%s %d", c->mpu401_2_name,
+ card->number);
+ }
+ }
+
+ snd_ice1712_set_input_clock_source(ice, 0);
+
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, ice->port, ice->irq);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_ice1712_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "ICE1712",
+ .id_table = snd_ice1712_ids,
+ .probe = snd_ice1712_probe,
+ .remove = __devexit_p(snd_ice1712_remove),
+};
+
+static int __init alsa_card_ice1712_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_ice1712_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_ice1712_init)
+module_exit(alsa_card_ice1712_exit)
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
new file mode 100644
index 0000000..fdae6de
--- /dev/null
+++ b/sound/pci/ice1712/ice1712.h
@@ -0,0 +1,504 @@
+#ifndef __SOUND_ICE1712_H
+#define __SOUND_ICE1712_H
+
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/control.h>
+#include <sound/ac97_codec.h>
+#include <sound/rawmidi.h>
+#include <sound/i2c.h>
+#include <sound/ak4xxx-adda.h>
+#include <sound/ak4114.h>
+#include <sound/pt2258.h>
+#include <sound/pcm.h>
+#include <sound/mpu401.h>
+
+
+/*
+ * Direct registers
+ */
+
+#define ICEREG(ice, x) ((ice)->port + ICE1712_REG_##x)
+
+#define ICE1712_REG_CONTROL 0x00 /* byte */
+#define ICE1712_RESET 0x80 /* reset whole chip */
+#define ICE1712_SERR_LEVEL 0x04 /* SERR# level otherwise edge */
+#define ICE1712_NATIVE 0x01 /* native mode otherwise SB */
+#define ICE1712_REG_IRQMASK 0x01 /* byte */
+#define ICE1712_IRQ_MPU1 0x80
+#define ICE1712_IRQ_TIMER 0x40
+#define ICE1712_IRQ_MPU2 0x20
+#define ICE1712_IRQ_PROPCM 0x10
+#define ICE1712_IRQ_FM 0x08 /* FM/MIDI - legacy */
+#define ICE1712_IRQ_PBKDS 0x04 /* playback DS channels */
+#define ICE1712_IRQ_CONCAP 0x02 /* consumer capture */
+#define ICE1712_IRQ_CONPBK 0x01 /* consumer playback */
+#define ICE1712_REG_IRQSTAT 0x02 /* byte */
+/* look to ICE1712_IRQ_* */
+#define ICE1712_REG_INDEX 0x03 /* byte - indirect CCIxx regs */
+#define ICE1712_REG_DATA 0x04 /* byte - indirect CCIxx regs */
+#define ICE1712_REG_NMI_STAT1 0x05 /* byte */
+#define ICE1712_REG_NMI_DATA 0x06 /* byte */
+#define ICE1712_REG_NMI_INDEX 0x07 /* byte */
+#define ICE1712_REG_AC97_INDEX 0x08 /* byte */
+#define ICE1712_REG_AC97_CMD 0x09 /* byte */
+#define ICE1712_AC97_COLD 0x80 /* cold reset */
+#define ICE1712_AC97_WARM 0x40 /* warm reset */
+#define ICE1712_AC97_WRITE 0x20 /* W: write, R: write in progress */
+#define ICE1712_AC97_READ 0x10 /* W: read, R: read in progress */
+#define ICE1712_AC97_READY 0x08 /* codec ready status bit */
+#define ICE1712_AC97_PBK_VSR 0x02 /* playback VSR */
+#define ICE1712_AC97_CAP_VSR 0x01 /* capture VSR */
+#define ICE1712_REG_AC97_DATA 0x0a /* word (little endian) */
+#define ICE1712_REG_MPU1_CTRL 0x0c /* byte */
+#define ICE1712_REG_MPU1_DATA 0x0d /* byte */
+#define ICE1712_REG_I2C_DEV_ADDR 0x10 /* byte */
+#define ICE1712_I2C_WRITE 0x01 /* write direction */
+#define ICE1712_REG_I2C_BYTE_ADDR 0x11 /* byte */
+#define ICE1712_REG_I2C_DATA 0x12 /* byte */
+#define ICE1712_REG_I2C_CTRL 0x13 /* byte */
+#define ICE1712_I2C_EEPROM 0x80 /* EEPROM exists */
+#define ICE1712_I2C_BUSY 0x01 /* busy bit */
+#define ICE1712_REG_CONCAP_ADDR 0x14 /* dword - consumer capture */
+#define ICE1712_REG_CONCAP_COUNT 0x18 /* word - current/base count */
+#define ICE1712_REG_SERR_SHADOW 0x1b /* byte */
+#define ICE1712_REG_MPU2_CTRL 0x1c /* byte */
+#define ICE1712_REG_MPU2_DATA 0x1d /* byte */
+#define ICE1712_REG_TIMER 0x1e /* word */
+
+/*
+ * Indirect registers
+ */
+
+#define ICE1712_IREG_PBK_COUNT_LO 0x00
+#define ICE1712_IREG_PBK_COUNT_HI 0x01
+#define ICE1712_IREG_PBK_CTRL 0x02
+#define ICE1712_IREG_PBK_LEFT 0x03 /* left volume */
+#define ICE1712_IREG_PBK_RIGHT 0x04 /* right volume */
+#define ICE1712_IREG_PBK_SOFT 0x05 /* soft volume */
+#define ICE1712_IREG_PBK_RATE_LO 0x06
+#define ICE1712_IREG_PBK_RATE_MID 0x07
+#define ICE1712_IREG_PBK_RATE_HI 0x08
+#define ICE1712_IREG_CAP_COUNT_LO 0x10
+#define ICE1712_IREG_CAP_COUNT_HI 0x11
+#define ICE1712_IREG_CAP_CTRL 0x12
+#define ICE1712_IREG_GPIO_DATA 0x20
+#define ICE1712_IREG_GPIO_WRITE_MASK 0x21
+#define ICE1712_IREG_GPIO_DIRECTION 0x22
+#define ICE1712_IREG_CONSUMER_POWERDOWN 0x30
+#define ICE1712_IREG_PRO_POWERDOWN 0x31
+
+/*
+ * Consumer section direct DMA registers
+ */
+
+#define ICEDS(ice, x) ((ice)->dmapath_port + ICE1712_DS_##x)
+
+#define ICE1712_DS_INTMASK 0x00 /* word - interrupt mask */
+#define ICE1712_DS_INTSTAT 0x02 /* word - interrupt status */
+#define ICE1712_DS_DATA 0x04 /* dword - channel data */
+#define ICE1712_DS_INDEX 0x08 /* dword - channel index */
+
+/*
+ * Consumer section channel registers
+ */
+
+#define ICE1712_DSC_ADDR0 0x00 /* dword - base address 0 */
+#define ICE1712_DSC_COUNT0 0x01 /* word - count 0 */
+#define ICE1712_DSC_ADDR1 0x02 /* dword - base address 1 */
+#define ICE1712_DSC_COUNT1 0x03 /* word - count 1 */
+#define ICE1712_DSC_CONTROL 0x04 /* byte - control & status */
+#define ICE1712_BUFFER1 0x80 /* buffer1 is active */
+#define ICE1712_BUFFER1_AUTO 0x40 /* buffer1 auto init */
+#define ICE1712_BUFFER0_AUTO 0x20 /* buffer0 auto init */
+#define ICE1712_FLUSH 0x10 /* flush FIFO */
+#define ICE1712_STEREO 0x08 /* stereo */
+#define ICE1712_16BIT 0x04 /* 16-bit data */
+#define ICE1712_PAUSE 0x02 /* pause */
+#define ICE1712_START 0x01 /* start */
+#define ICE1712_DSC_RATE 0x05 /* dword - rate */
+#define ICE1712_DSC_VOLUME 0x06 /* word - volume control */
+
+/*
+ * Professional multi-track direct control registers
+ */
+
+#define ICEMT(ice, x) ((ice)->profi_port + ICE1712_MT_##x)
+
+#define ICE1712_MT_IRQ 0x00 /* byte - interrupt mask */
+#define ICE1712_MULTI_CAPTURE 0x80 /* capture IRQ */
+#define ICE1712_MULTI_PLAYBACK 0x40 /* playback IRQ */
+#define ICE1712_MULTI_CAPSTATUS 0x02 /* capture IRQ status */
+#define ICE1712_MULTI_PBKSTATUS 0x01 /* playback IRQ status */
+#define ICE1712_MT_RATE 0x01 /* byte - sampling rate select */
+#define ICE1712_SPDIF_MASTER 0x10 /* S/PDIF input is master clock */
+#define ICE1712_MT_I2S_FORMAT 0x02 /* byte - I2S data format */
+#define ICE1712_MT_AC97_INDEX 0x04 /* byte - AC'97 index */
+#define ICE1712_MT_AC97_CMD 0x05 /* byte - AC'97 command & status */
+/* look to ICE1712_AC97_* */
+#define ICE1712_MT_AC97_DATA 0x06 /* word - AC'97 data */
+#define ICE1712_MT_PLAYBACK_ADDR 0x10 /* dword - playback address */
+#define ICE1712_MT_PLAYBACK_SIZE 0x14 /* word - playback size */
+#define ICE1712_MT_PLAYBACK_COUNT 0x16 /* word - playback count */
+#define ICE1712_MT_PLAYBACK_CONTROL 0x18 /* byte - control */
+#define ICE1712_CAPTURE_START_SHADOW 0x04 /* capture start */
+#define ICE1712_PLAYBACK_PAUSE 0x02 /* playback pause */
+#define ICE1712_PLAYBACK_START 0x01 /* playback start */
+#define ICE1712_MT_CAPTURE_ADDR 0x20 /* dword - capture address */
+#define ICE1712_MT_CAPTURE_SIZE 0x24 /* word - capture size */
+#define ICE1712_MT_CAPTURE_COUNT 0x26 /* word - capture count */
+#define ICE1712_MT_CAPTURE_CONTROL 0x28 /* byte - control */
+#define ICE1712_CAPTURE_START 0x01 /* capture start */
+#define ICE1712_MT_ROUTE_PSDOUT03 0x30 /* word */
+#define ICE1712_MT_ROUTE_SPDOUT 0x32 /* word */
+#define ICE1712_MT_ROUTE_CAPTURE 0x34 /* dword */
+#define ICE1712_MT_MONITOR_VOLUME 0x38 /* word */
+#define ICE1712_MT_MONITOR_INDEX 0x3a /* byte */
+#define ICE1712_MT_MONITOR_RATE 0x3b /* byte */
+#define ICE1712_MT_MONITOR_ROUTECTRL 0x3c /* byte */
+#define ICE1712_ROUTE_AC97 0x01 /* route digital mixer output to AC'97 */
+#define ICE1712_MT_MONITOR_PEAKINDEX 0x3e /* byte */
+#define ICE1712_MT_MONITOR_PEAKDATA 0x3f /* byte */
+
+/*
+ * Codec configuration bits
+ */
+
+/* PCI[60] System Configuration */
+#define ICE1712_CFG_CLOCK 0xc0
+#define ICE1712_CFG_CLOCK512 0x00 /* 22.5692Mhz, 44.1kHz*512 */
+#define ICE1712_CFG_CLOCK384 0x40 /* 16.9344Mhz, 44.1kHz*384 */
+#define ICE1712_CFG_EXT 0x80 /* external clock */
+#define ICE1712_CFG_2xMPU401 0x20 /* two MPU401 UARTs */
+#define ICE1712_CFG_NO_CON_AC97 0x10 /* consumer AC'97 codec is not present */
+#define ICE1712_CFG_ADC_MASK 0x0c /* one, two, three, four stereo ADCs */
+#define ICE1712_CFG_DAC_MASK 0x03 /* one, two, three, four stereo DACs */
+/* PCI[61] AC-Link Configuration */
+#define ICE1712_CFG_PRO_I2S 0x80 /* multitrack converter: I2S or AC'97 */
+#define ICE1712_CFG_AC97_PACKED 0x01 /* split or packed mode - AC'97 */
+/* PCI[62] I2S Features */
+#define ICE1712_CFG_I2S_VOLUME 0x80 /* volume/mute capability */
+#define ICE1712_CFG_I2S_96KHZ 0x40 /* supports 96kHz sampling */
+#define ICE1712_CFG_I2S_RESMASK 0x30 /* resolution mask, 16,18,20,24-bit */
+#define ICE1712_CFG_I2S_OTHER 0x0f /* other I2S IDs */
+/* PCI[63] S/PDIF Configuration */
+#define ICE1712_CFG_I2S_CHIPID 0xfc /* I2S chip ID */
+#define ICE1712_CFG_SPDIF_IN 0x02 /* S/PDIF input is present */
+#define ICE1712_CFG_SPDIF_OUT 0x01 /* S/PDIF output is present */
+
+/*
+ * DMA mode values
+ * identical with DMA_XXX on i386 architecture.
+ */
+#define ICE1712_DMA_MODE_WRITE 0x48
+#define ICE1712_DMA_AUTOINIT 0x10
+
+
+/*
+ *
+ */
+
+struct snd_ice1712;
+
+struct snd_ice1712_eeprom {
+ unsigned int subvendor; /* PCI[2c-2f] */
+ unsigned char size; /* size of EEPROM image in bytes */
+ unsigned char version; /* must be 1 (or 2 for vt1724) */
+ unsigned char data[32];
+ unsigned int gpiomask;
+ unsigned int gpiostate;
+ unsigned int gpiodir;
+};
+
+enum {
+ ICE_EEP1_CODEC = 0, /* 06 */
+ ICE_EEP1_ACLINK, /* 07 */
+ ICE_EEP1_I2SID, /* 08 */
+ ICE_EEP1_SPDIF, /* 09 */
+ ICE_EEP1_GPIO_MASK, /* 0a */
+ ICE_EEP1_GPIO_STATE, /* 0b */
+ ICE_EEP1_GPIO_DIR, /* 0c */
+ ICE_EEP1_AC97_MAIN_LO, /* 0d */
+ ICE_EEP1_AC97_MAIN_HI, /* 0e */
+ ICE_EEP1_AC97_PCM_LO, /* 0f */
+ ICE_EEP1_AC97_PCM_HI, /* 10 */
+ ICE_EEP1_AC97_REC_LO, /* 11 */
+ ICE_EEP1_AC97_REC_HI, /* 12 */
+ ICE_EEP1_AC97_RECSRC, /* 13 */
+ ICE_EEP1_DAC_ID, /* 14 */
+ ICE_EEP1_DAC_ID1,
+ ICE_EEP1_DAC_ID2,
+ ICE_EEP1_DAC_ID3,
+ ICE_EEP1_ADC_ID, /* 18 */
+ ICE_EEP1_ADC_ID1,
+ ICE_EEP1_ADC_ID2,
+ ICE_EEP1_ADC_ID3
+};
+
+#define ice_has_con_ac97(ice) (!((ice)->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97))
+
+
+struct snd_ak4xxx_private {
+ unsigned int cif:1; /* CIF mode */
+ unsigned char caddr; /* C0 and C1 bits */
+ unsigned int data_mask; /* DATA gpio bit */
+ unsigned int clk_mask; /* CLK gpio bit */
+ unsigned int cs_mask; /* bit mask for select/deselect address */
+ unsigned int cs_addr; /* bits to select address */
+ unsigned int cs_none; /* bits to deselect address */
+ unsigned int add_flags; /* additional bits at init */
+ unsigned int mask_flags; /* total mask bits */
+ struct snd_akm4xxx_ops {
+ void (*set_rate_val)(struct snd_akm4xxx *ak, unsigned int rate);
+ } ops;
+};
+
+struct snd_ice1712_spdif {
+ unsigned char cs8403_bits;
+ unsigned char cs8403_stream_bits;
+ struct snd_kcontrol *stream_ctl;
+
+ struct snd_ice1712_spdif_ops {
+ void (*open)(struct snd_ice1712 *, struct snd_pcm_substream *);
+ void (*setup_rate)(struct snd_ice1712 *, int rate);
+ void (*close)(struct snd_ice1712 *, struct snd_pcm_substream *);
+ void (*default_get)(struct snd_ice1712 *, struct snd_ctl_elem_value *ucontrol);
+ int (*default_put)(struct snd_ice1712 *, struct snd_ctl_elem_value *ucontrol);
+ void (*stream_get)(struct snd_ice1712 *, struct snd_ctl_elem_value *ucontrol);
+ int (*stream_put)(struct snd_ice1712 *, struct snd_ctl_elem_value *ucontrol);
+ } ops;
+};
+
+
+struct snd_ice1712 {
+ unsigned long conp_dma_size;
+ unsigned long conc_dma_size;
+ unsigned long prop_dma_size;
+ unsigned long proc_dma_size;
+ int irq;
+
+ unsigned long port;
+ unsigned long ddma_port;
+ unsigned long dmapath_port;
+ unsigned long profi_port;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm *pcm_ds;
+ struct snd_pcm *pcm_pro;
+ struct snd_pcm_substream *playback_con_substream;
+ struct snd_pcm_substream *playback_con_substream_ds[6];
+ struct snd_pcm_substream *capture_con_substream;
+ struct snd_pcm_substream *playback_pro_substream;
+ struct snd_pcm_substream *capture_pro_substream;
+ unsigned int playback_pro_size;
+ unsigned int capture_pro_size;
+ unsigned int playback_con_virt_addr[6];
+ unsigned int playback_con_active_buf[6];
+ unsigned int capture_con_virt_addr;
+ unsigned int ac97_ext_id;
+ struct snd_ac97 *ac97;
+ struct snd_rawmidi *rmidi[2];
+
+ spinlock_t reg_lock;
+ struct snd_info_entry *proc_entry;
+
+ struct snd_ice1712_eeprom eeprom;
+
+ unsigned int pro_volumes[20];
+ unsigned int omni:1; /* Delta Omni I/O */
+ unsigned int dxr_enable:1; /* Terratec DXR enable for DMX6FIRE */
+ unsigned int vt1724:1;
+ unsigned int vt1720:1;
+ unsigned int has_spdif:1; /* VT1720/4 - has SPDIF I/O */
+ unsigned int force_pdma4:1; /* VT1720/4 - PDMA4 as non-spdif */
+ unsigned int force_rdma1:1; /* VT1720/4 - RDMA1 as non-spdif */
+ unsigned int midi_output:1; /* VT1720/4: MIDI output triggered */
+ unsigned int midi_input:1; /* VT1720/4: MIDI input triggered */
+ unsigned int num_total_dacs; /* total DACs */
+ unsigned int num_total_adcs; /* total ADCs */
+ unsigned int cur_rate; /* current rate */
+
+ struct mutex open_mutex;
+ struct snd_pcm_substream *pcm_reserved[4];
+ struct snd_pcm_hw_constraint_list *hw_rates; /* card-specific rate constraints */
+
+ unsigned int akm_codecs;
+ struct snd_akm4xxx *akm;
+ struct snd_ice1712_spdif spdif;
+
+ struct mutex i2c_mutex; /* I2C mutex for ICE1724 registers */
+ struct snd_i2c_bus *i2c; /* I2C bus */
+ struct snd_i2c_device *cs8427; /* CS8427 I2C device */
+ unsigned int cs8427_timeout; /* CS8427 reset timeout in HZ/100 */
+
+ struct ice1712_gpio {
+ unsigned int direction; /* current direction bits */
+ unsigned int write_mask; /* current mask bits */
+ unsigned int saved[2]; /* for ewx_i2c */
+ /* operators */
+ void (*set_mask)(struct snd_ice1712 *ice, unsigned int data);
+ void (*set_dir)(struct snd_ice1712 *ice, unsigned int data);
+ void (*set_data)(struct snd_ice1712 *ice, unsigned int data);
+ unsigned int (*get_data)(struct snd_ice1712 *ice);
+ /* misc operators - move to another place? */
+ void (*set_pro_rate)(struct snd_ice1712 *ice, unsigned int rate);
+ void (*i2s_mclk_changed)(struct snd_ice1712 *ice);
+ } gpio;
+ struct mutex gpio_mutex;
+
+ /* other board-specific data */
+ void *spec;
+
+ /* VT172x specific */
+ int pro_rate_default;
+ int (*is_spdif_master)(struct snd_ice1712 *ice);
+ unsigned int (*get_rate)(struct snd_ice1712 *ice);
+ void (*set_rate)(struct snd_ice1712 *ice, unsigned int rate);
+ unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate);
+ void (*set_spdif_clock)(struct snd_ice1712 *ice);
+
+};
+
+
+/*
+ * gpio access functions
+ */
+static inline void snd_ice1712_gpio_set_dir(struct snd_ice1712 *ice, unsigned int bits)
+{
+ ice->gpio.set_dir(ice, bits);
+}
+
+static inline void snd_ice1712_gpio_set_mask(struct snd_ice1712 *ice, unsigned int bits)
+{
+ ice->gpio.set_mask(ice, bits);
+}
+
+static inline void snd_ice1712_gpio_write(struct snd_ice1712 *ice, unsigned int val)
+{
+ ice->gpio.set_data(ice, val);
+}
+
+static inline unsigned int snd_ice1712_gpio_read(struct snd_ice1712 *ice)
+{
+ return ice->gpio.get_data(ice);
+}
+
+/*
+ * save and restore gpio status
+ * The access to gpio will be protected by mutex, so don't forget to
+ * restore!
+ */
+static inline void snd_ice1712_save_gpio_status(struct snd_ice1712 *ice)
+{
+ mutex_lock(&ice->gpio_mutex);
+ ice->gpio.saved[0] = ice->gpio.direction;
+ ice->gpio.saved[1] = ice->gpio.write_mask;
+}
+
+static inline void snd_ice1712_restore_gpio_status(struct snd_ice1712 *ice)
+{
+ ice->gpio.set_dir(ice, ice->gpio.saved[0]);
+ ice->gpio.set_mask(ice, ice->gpio.saved[1]);
+ ice->gpio.direction = ice->gpio.saved[0];
+ ice->gpio.write_mask = ice->gpio.saved[1];
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+/* for bit controls */
+#define ICE1712_GPIO(xiface, xname, xindex, mask, invert, xaccess) \
+{ .iface = xiface, .name = xname, .access = xaccess, .info = snd_ctl_boolean_mono_info, \
+ .get = snd_ice1712_gpio_get, .put = snd_ice1712_gpio_put, \
+ .private_value = mask | (invert << 24) }
+
+int snd_ice1712_gpio_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
+int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
+
+/*
+ * set gpio direction, write mask and data
+ */
+static inline void snd_ice1712_gpio_write_bits(struct snd_ice1712 *ice,
+ unsigned int mask, unsigned int bits)
+{
+ unsigned val;
+
+ ice->gpio.direction |= mask;
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+ val = snd_ice1712_gpio_read(ice);
+ val &= ~mask;
+ val |= mask & bits;
+ snd_ice1712_gpio_write(ice, val);
+}
+
+static inline int snd_ice1712_gpio_read_bits(struct snd_ice1712 *ice,
+ unsigned int mask)
+{
+ ice->gpio.direction &= ~mask;
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+ return snd_ice1712_gpio_read(ice) & mask;
+}
+
+int snd_ice1712_spdif_build_controls(struct snd_ice1712 *ice);
+
+int snd_ice1712_akm4xxx_init(struct snd_akm4xxx *ak, const struct snd_akm4xxx *template,
+ const struct snd_ak4xxx_private *priv, struct snd_ice1712 *ice);
+void snd_ice1712_akm4xxx_free(struct snd_ice1712 *ice);
+int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice);
+
+int snd_ice1712_init_cs8427(struct snd_ice1712 *ice, int addr);
+
+static inline void snd_ice1712_write(struct snd_ice1712 *ice, u8 addr, u8 data)
+{
+ outb(addr, ICEREG(ice, INDEX));
+ outb(data, ICEREG(ice, DATA));
+}
+
+static inline u8 snd_ice1712_read(struct snd_ice1712 *ice, u8 addr)
+{
+ outb(addr, ICEREG(ice, INDEX));
+ return inb(ICEREG(ice, DATA));
+}
+
+
+/*
+ * entry pointer
+ */
+
+struct snd_ice1712_card_info {
+ unsigned int subvendor;
+ char *name;
+ char *model;
+ char *driver;
+ int (*chip_init)(struct snd_ice1712 *);
+ int (*build_controls)(struct snd_ice1712 *);
+ unsigned int no_mpu401:1;
+ unsigned int mpu401_1_info_flags;
+ unsigned int mpu401_2_info_flags;
+ const char *mpu401_1_name;
+ const char *mpu401_2_name;
+ const unsigned int eeprom_size;
+ const unsigned char *eeprom_data;
+};
+
+
+#endif /* __SOUND_ICE1712_H */
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
new file mode 100644
index 0000000..1b3f117
--- /dev/null
+++ b/sound/pci/ice1712/ice1724.c
@@ -0,0 +1,2624 @@
+/*
+ * ALSA driver for VT1724 ICEnsemble ICE1724 / VIA VT1724 (Envy24HT)
+ * VIA VT1720 (Envy24PT)
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ * 2002 James Stafford <jstafford@ampltd.com>
+ * 2003 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/rawmidi.h>
+#include <sound/initval.h>
+
+#include <sound/asoundef.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+
+/* lowlevel routines */
+#include "amp.h"
+#include "revo.h"
+#include "aureon.h"
+#include "vt1720_mobo.h"
+#include "pontis.h"
+#include "prodigy192.h"
+#include "prodigy_hifi.h"
+#include "juli.h"
+#include "phase.h"
+#include "wtm.h"
+#include "se.h"
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{"
+ REVO_DEVICE_DESC
+ AMP_AUDIO2000_DEVICE_DESC
+ AUREON_DEVICE_DESC
+ VT1720_MOBO_DEVICE_DESC
+ PONTIS_DEVICE_DESC
+ PRODIGY192_DEVICE_DESC
+ PRODIGY_HIFI_DEVICE_DESC
+ JULI_DEVICE_DESC
+ PHASE_DEVICE_DESC
+ WTM_DEVICE_DESC
+ SE_DEVICE_DESC
+ "{VIA,VT1720},"
+ "{VIA,VT1724},"
+ "{ICEnsemble,Generic ICE1724},"
+ "{ICEnsemble,Generic Envy24HT}"
+ "{ICEnsemble,Generic Envy24PT}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static char *model[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for ICE1724 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for ICE1724 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable ICE1724 soundcard.");
+module_param_array(model, charp, NULL, 0444);
+MODULE_PARM_DESC(model, "Use the given board model.");
+
+
+/* Both VT1720 and VT1724 have the same PCI IDs */
+static const struct pci_device_id snd_vt1724_ids[] = {
+ { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_VT1724, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_vt1724_ids);
+
+
+static int PRO_RATE_LOCKED;
+static int PRO_RATE_RESET = 1;
+static unsigned int PRO_RATE_DEFAULT = 44100;
+
+/*
+ * Basic I/O
+ */
+
+/*
+ * default rates, default clock routines
+ */
+
+/* check whether the clock mode is spdif-in */
+static inline int stdclock_is_spdif_master(struct snd_ice1712 *ice)
+{
+ return (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER) ? 1 : 0;
+}
+
+static inline int is_pro_rate_locked(struct snd_ice1712 *ice)
+{
+ return ice->is_spdif_master(ice) || PRO_RATE_LOCKED;
+}
+
+/*
+ * ac97 section
+ */
+
+static unsigned char snd_vt1724_ac97_ready(struct snd_ice1712 *ice)
+{
+ unsigned char old_cmd;
+ int tm;
+ for (tm = 0; tm < 0x10000; tm++) {
+ old_cmd = inb(ICEMT1724(ice, AC97_CMD));
+ if (old_cmd & (VT1724_AC97_WRITE | VT1724_AC97_READ))
+ continue;
+ if (!(old_cmd & VT1724_AC97_READY))
+ continue;
+ return old_cmd;
+ }
+ snd_printd(KERN_ERR "snd_vt1724_ac97_ready: timeout\n");
+ return old_cmd;
+}
+
+static int snd_vt1724_ac97_wait_bit(struct snd_ice1712 *ice, unsigned char bit)
+{
+ int tm;
+ for (tm = 0; tm < 0x10000; tm++)
+ if ((inb(ICEMT1724(ice, AC97_CMD)) & bit) == 0)
+ return 0;
+ snd_printd(KERN_ERR "snd_vt1724_ac97_wait_bit: timeout\n");
+ return -EIO;
+}
+
+static void snd_vt1724_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct snd_ice1712 *ice = ac97->private_data;
+ unsigned char old_cmd;
+
+ old_cmd = snd_vt1724_ac97_ready(ice);
+ old_cmd &= ~VT1724_AC97_ID_MASK;
+ old_cmd |= ac97->num;
+ outb(reg, ICEMT1724(ice, AC97_INDEX));
+ outw(val, ICEMT1724(ice, AC97_DATA));
+ outb(old_cmd | VT1724_AC97_WRITE, ICEMT1724(ice, AC97_CMD));
+ snd_vt1724_ac97_wait_bit(ice, VT1724_AC97_WRITE);
+}
+
+static unsigned short snd_vt1724_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct snd_ice1712 *ice = ac97->private_data;
+ unsigned char old_cmd;
+
+ old_cmd = snd_vt1724_ac97_ready(ice);
+ old_cmd &= ~VT1724_AC97_ID_MASK;
+ old_cmd |= ac97->num;
+ outb(reg, ICEMT1724(ice, AC97_INDEX));
+ outb(old_cmd | VT1724_AC97_READ, ICEMT1724(ice, AC97_CMD));
+ if (snd_vt1724_ac97_wait_bit(ice, VT1724_AC97_READ) < 0)
+ return ~0;
+ return inw(ICEMT1724(ice, AC97_DATA));
+}
+
+
+/*
+ * GPIO operations
+ */
+
+/* set gpio direction 0 = read, 1 = write */
+static void snd_vt1724_set_gpio_dir(struct snd_ice1712 *ice, unsigned int data)
+{
+ outl(data, ICEREG1724(ice, GPIO_DIRECTION));
+ inw(ICEREG1724(ice, GPIO_DIRECTION)); /* dummy read for pci-posting */
+}
+
+/* set the gpio mask (0 = writable) */
+static void snd_vt1724_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data)
+{
+ outw(data, ICEREG1724(ice, GPIO_WRITE_MASK));
+ if (!ice->vt1720) /* VT1720 supports only 16 GPIO bits */
+ outb((data >> 16) & 0xff, ICEREG1724(ice, GPIO_WRITE_MASK_22));
+ inw(ICEREG1724(ice, GPIO_WRITE_MASK)); /* dummy read for pci-posting */
+}
+
+static void snd_vt1724_set_gpio_data(struct snd_ice1712 *ice, unsigned int data)
+{
+ outw(data, ICEREG1724(ice, GPIO_DATA));
+ if (!ice->vt1720)
+ outb(data >> 16, ICEREG1724(ice, GPIO_DATA_22));
+ inw(ICEREG1724(ice, GPIO_DATA)); /* dummy read for pci-posting */
+}
+
+static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice)
+{
+ unsigned int data;
+ if (!ice->vt1720)
+ data = (unsigned int)inb(ICEREG1724(ice, GPIO_DATA_22));
+ else
+ data = 0;
+ data = (data << 16) | inw(ICEREG1724(ice, GPIO_DATA));
+ return data;
+}
+
+/*
+ * MIDI
+ */
+
+static void vt1724_midi_clear_rx(struct snd_ice1712 *ice)
+{
+ unsigned int count;
+
+ for (count = inb(ICEREG1724(ice, MPU_RXFIFO)); count > 0; --count)
+ inb(ICEREG1724(ice, MPU_DATA));
+}
+
+static inline struct snd_rawmidi_substream *
+get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream)
+{
+ return list_first_entry(&ice->rmidi[0]->streams[stream].substreams,
+ struct snd_rawmidi_substream, list);
+}
+
+static void vt1724_midi_write(struct snd_ice1712 *ice)
+{
+ struct snd_rawmidi_substream *s;
+ int count, i;
+ u8 buffer[32];
+
+ s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_OUTPUT);
+ count = 31 - inb(ICEREG1724(ice, MPU_TXFIFO));
+ if (count > 0) {
+ count = snd_rawmidi_transmit(s, buffer, count);
+ for (i = 0; i < count; ++i)
+ outb(buffer[i], ICEREG1724(ice, MPU_DATA));
+ }
+}
+
+static void vt1724_midi_read(struct snd_ice1712 *ice)
+{
+ struct snd_rawmidi_substream *s;
+ int count, i;
+ u8 buffer[32];
+
+ s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_INPUT);
+ count = inb(ICEREG1724(ice, MPU_RXFIFO));
+ if (count > 0) {
+ count = min(count, 32);
+ for (i = 0; i < count; ++i)
+ buffer[i] = inb(ICEREG1724(ice, MPU_DATA));
+ snd_rawmidi_receive(s, buffer, count);
+ }
+}
+
+static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream,
+ u8 flag, int enable)
+{
+ struct snd_ice1712 *ice = substream->rmidi->private_data;
+ u8 mask;
+
+ spin_lock_irq(&ice->reg_lock);
+ mask = inb(ICEREG1724(ice, IRQMASK));
+ if (enable)
+ mask &= ~flag;
+ else
+ mask |= flag;
+ outb(mask, ICEREG1724(ice, IRQMASK));
+ spin_unlock_irq(&ice->reg_lock);
+}
+
+static int vt1724_midi_output_open(struct snd_rawmidi_substream *s)
+{
+ vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1);
+ return 0;
+}
+
+static int vt1724_midi_output_close(struct snd_rawmidi_substream *s)
+{
+ vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0);
+ return 0;
+}
+
+static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up)
+{
+ struct snd_ice1712 *ice = s->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ice->reg_lock, flags);
+ if (up) {
+ ice->midi_output = 1;
+ vt1724_midi_write(ice);
+ } else {
+ ice->midi_output = 0;
+ }
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+}
+
+static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s)
+{
+ struct snd_ice1712 *ice = s->rmidi->private_data;
+ unsigned long timeout;
+
+ /* 32 bytes should be transmitted in less than about 12 ms */
+ timeout = jiffies + msecs_to_jiffies(15);
+ do {
+ if (inb(ICEREG1724(ice, MPU_CTRL)) & VT1724_MPU_TX_EMPTY)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after(timeout, jiffies));
+}
+
+static struct snd_rawmidi_ops vt1724_midi_output_ops = {
+ .open = vt1724_midi_output_open,
+ .close = vt1724_midi_output_close,
+ .trigger = vt1724_midi_output_trigger,
+ .drain = vt1724_midi_output_drain,
+};
+
+static int vt1724_midi_input_open(struct snd_rawmidi_substream *s)
+{
+ vt1724_midi_clear_rx(s->rmidi->private_data);
+ vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 1);
+ return 0;
+}
+
+static int vt1724_midi_input_close(struct snd_rawmidi_substream *s)
+{
+ vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 0);
+ return 0;
+}
+
+static void vt1724_midi_input_trigger(struct snd_rawmidi_substream *s, int up)
+{
+ struct snd_ice1712 *ice = s->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ice->reg_lock, flags);
+ if (up) {
+ ice->midi_input = 1;
+ vt1724_midi_read(ice);
+ } else {
+ ice->midi_input = 0;
+ }
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+}
+
+static struct snd_rawmidi_ops vt1724_midi_input_ops = {
+ .open = vt1724_midi_input_open,
+ .close = vt1724_midi_input_close,
+ .trigger = vt1724_midi_input_trigger,
+};
+
+
+/*
+ * Interrupt handler
+ */
+
+static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
+{
+ struct snd_ice1712 *ice = dev_id;
+ unsigned char status;
+ unsigned char status_mask =
+ VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX | VT1724_IRQ_MTPCM;
+ int handled = 0;
+#ifdef CONFIG_SND_DEBUG
+ int timeout = 0;
+#endif
+
+ while (1) {
+ status = inb(ICEREG1724(ice, IRQSTAT));
+ status &= status_mask;
+ if (status == 0)
+ break;
+#ifdef CONFIG_SND_DEBUG
+ if (++timeout > 10) {
+ printk(KERN_ERR
+ "ice1724: Too long irq loop, status = 0x%x\n",
+ status);
+ break;
+ }
+#endif
+ handled = 1;
+ if (status & VT1724_IRQ_MPU_TX) {
+ spin_lock(&ice->reg_lock);
+ if (ice->midi_output)
+ vt1724_midi_write(ice);
+ spin_unlock(&ice->reg_lock);
+ /* Due to mysterical reasons, MPU_TX is always
+ * generated (and can't be cleared) when a PCM
+ * playback is going. So let's ignore at the
+ * next loop.
+ */
+ status_mask &= ~VT1724_IRQ_MPU_TX;
+ }
+ if (status & VT1724_IRQ_MPU_RX) {
+ spin_lock(&ice->reg_lock);
+ if (ice->midi_input)
+ vt1724_midi_read(ice);
+ else
+ vt1724_midi_clear_rx(ice);
+ spin_unlock(&ice->reg_lock);
+ }
+ /* ack MPU irq */
+ outb(status, ICEREG1724(ice, IRQSTAT));
+ if (status & VT1724_IRQ_MTPCM) {
+ /*
+ * Multi-track PCM
+ * PCM assignment are:
+ * Playback DMA0 (M/C) = playback_pro_substream
+ * Playback DMA1 = playback_con_substream_ds[0]
+ * Playback DMA2 = playback_con_substream_ds[1]
+ * Playback DMA3 = playback_con_substream_ds[2]
+ * Playback DMA4 (SPDIF) = playback_con_substream
+ * Record DMA0 = capture_pro_substream
+ * Record DMA1 = capture_con_substream
+ */
+ unsigned char mtstat = inb(ICEMT1724(ice, IRQ));
+ if (mtstat & VT1724_MULTI_PDMA0) {
+ if (ice->playback_pro_substream)
+ snd_pcm_period_elapsed(ice->playback_pro_substream);
+ }
+ if (mtstat & VT1724_MULTI_RDMA0) {
+ if (ice->capture_pro_substream)
+ snd_pcm_period_elapsed(ice->capture_pro_substream);
+ }
+ if (mtstat & VT1724_MULTI_PDMA1) {
+ if (ice->playback_con_substream_ds[0])
+ snd_pcm_period_elapsed(ice->playback_con_substream_ds[0]);
+ }
+ if (mtstat & VT1724_MULTI_PDMA2) {
+ if (ice->playback_con_substream_ds[1])
+ snd_pcm_period_elapsed(ice->playback_con_substream_ds[1]);
+ }
+ if (mtstat & VT1724_MULTI_PDMA3) {
+ if (ice->playback_con_substream_ds[2])
+ snd_pcm_period_elapsed(ice->playback_con_substream_ds[2]);
+ }
+ if (mtstat & VT1724_MULTI_PDMA4) {
+ if (ice->playback_con_substream)
+ snd_pcm_period_elapsed(ice->playback_con_substream);
+ }
+ if (mtstat & VT1724_MULTI_RDMA1) {
+ if (ice->capture_con_substream)
+ snd_pcm_period_elapsed(ice->capture_con_substream);
+ }
+ /* ack anyway to avoid freeze */
+ outb(mtstat, ICEMT1724(ice, IRQ));
+ /* ought to really handle this properly */
+ if (mtstat & VT1724_MULTI_FIFO_ERR) {
+ unsigned char fstat = inb(ICEMT1724(ice, DMA_FIFO_ERR));
+ outb(fstat, ICEMT1724(ice, DMA_FIFO_ERR));
+ outb(VT1724_MULTI_FIFO_ERR | inb(ICEMT1724(ice, DMA_INT_MASK)), ICEMT1724(ice, DMA_INT_MASK));
+ /* If I don't do this, I get machine lockup due to continual interrupts */
+ }
+
+ }
+ }
+ return IRQ_RETVAL(handled);
+}
+
+/*
+ * PCM code - professional part (multitrack)
+ */
+
+static unsigned int rates[] = {
+ 8000, 9600, 11025, 12000, 16000, 22050, 24000,
+ 32000, 44100, 48000, 64000, 88200, 96000,
+ 176400, 192000,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates_96 = {
+ .count = ARRAY_SIZE(rates) - 2, /* up to 96000 */
+ .list = rates,
+ .mask = 0,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates_48 = {
+ .count = ARRAY_SIZE(rates) - 5, /* up to 48000 */
+ .list = rates,
+ .mask = 0,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates_192 = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+struct vt1724_pcm_reg {
+ unsigned int addr; /* ADDR register offset */
+ unsigned int size; /* SIZE register offset */
+ unsigned int count; /* COUNT register offset */
+ unsigned int start; /* start & pause bit */
+};
+
+static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ unsigned char what;
+ unsigned char old;
+ struct snd_pcm_substream *s;
+
+ what = 0;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) == ice) {
+ const struct vt1724_pcm_reg *reg;
+ reg = s->runtime->private_data;
+ what |= reg->start;
+ snd_pcm_trigger_done(s, substream);
+ }
+ }
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ spin_lock(&ice->reg_lock);
+ old = inb(ICEMT1724(ice, DMA_PAUSE));
+ if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH)
+ old |= what;
+ else
+ old &= ~what;
+ outb(old, ICEMT1724(ice, DMA_PAUSE));
+ spin_unlock(&ice->reg_lock);
+ break;
+
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_STOP:
+ spin_lock(&ice->reg_lock);
+ old = inb(ICEMT1724(ice, DMA_CONTROL));
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ old |= what;
+ else
+ old &= ~what;
+ outb(old, ICEMT1724(ice, DMA_CONTROL));
+ spin_unlock(&ice->reg_lock);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ */
+
+#define DMA_STARTS (VT1724_RDMA0_START|VT1724_PDMA0_START|VT1724_RDMA1_START|\
+ VT1724_PDMA1_START|VT1724_PDMA2_START|VT1724_PDMA3_START|VT1724_PDMA4_START)
+#define DMA_PAUSES (VT1724_RDMA0_PAUSE|VT1724_PDMA0_PAUSE|VT1724_RDMA1_PAUSE|\
+ VT1724_PDMA1_PAUSE|VT1724_PDMA2_PAUSE|VT1724_PDMA3_PAUSE|VT1724_PDMA4_PAUSE)
+
+static const unsigned int stdclock_rate_list[16] = {
+ 48000, 24000, 12000, 9600, 32000, 16000, 8000, 96000, 44100,
+ 22050, 11025, 88200, 176400, 0, 192000, 64000
+};
+
+static unsigned int stdclock_get_rate(struct snd_ice1712 *ice)
+{
+ unsigned int rate;
+ rate = stdclock_rate_list[inb(ICEMT1724(ice, RATE)) & 15];
+ return rate;
+}
+
+static void stdclock_set_rate(struct snd_ice1712 *ice, unsigned int rate)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(stdclock_rate_list); i++) {
+ if (stdclock_rate_list[i] == rate) {
+ outb(i, ICEMT1724(ice, RATE));
+ return;
+ }
+ }
+}
+
+static unsigned char stdclock_set_mclk(struct snd_ice1712 *ice,
+ unsigned int rate)
+{
+ unsigned char val, old;
+ /* check MT02 */
+ if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) {
+ val = old = inb(ICEMT1724(ice, I2S_FORMAT));
+ if (rate > 96000)
+ val |= VT1724_MT_I2S_MCLK_128X; /* 128x MCLK */
+ else
+ val &= ~VT1724_MT_I2S_MCLK_128X; /* 256x MCLK */
+ if (val != old) {
+ outb(val, ICEMT1724(ice, I2S_FORMAT));
+ /* master clock changed */
+ return 1;
+ }
+ }
+ /* no change in master clock */
+ return 0;
+}
+
+static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
+ int force)
+{
+ unsigned long flags;
+ unsigned char mclk_change;
+ unsigned int i, old_rate;
+
+ if (rate > ice->hw_rates->list[ice->hw_rates->count - 1])
+ return;
+ spin_lock_irqsave(&ice->reg_lock, flags);
+ if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) ||
+ (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) {
+ /* running? we cannot change the rate now... */
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+ return;
+ }
+ if (!force && is_pro_rate_locked(ice)) {
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+ return;
+ }
+
+ old_rate = ice->get_rate(ice);
+ if (force || (old_rate != rate))
+ ice->set_rate(ice, rate);
+ else if (rate == ice->cur_rate) {
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+ return;
+ }
+
+ ice->cur_rate = rate;
+
+ /* setting master clock */
+ mclk_change = ice->set_mclk(ice, rate);
+
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+
+ if (mclk_change && ice->gpio.i2s_mclk_changed)
+ ice->gpio.i2s_mclk_changed(ice);
+ if (ice->gpio.set_pro_rate)
+ ice->gpio.set_pro_rate(ice, rate);
+
+ /* set up codecs */
+ for (i = 0; i < ice->akm_codecs; i++) {
+ if (ice->akm[i].ops.set_rate_val)
+ ice->akm[i].ops.set_rate_val(&ice->akm[i], rate);
+ }
+ if (ice->spdif.ops.setup_rate)
+ ice->spdif.ops.setup_rate(ice, rate);
+}
+
+static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ int i, chs;
+
+ chs = params_channels(hw_params);
+ mutex_lock(&ice->open_mutex);
+ /* mark surround channels */
+ if (substream == ice->playback_pro_substream) {
+ /* PDMA0 can be multi-channel up to 8 */
+ chs = chs / 2 - 1;
+ for (i = 0; i < chs; i++) {
+ if (ice->pcm_reserved[i] &&
+ ice->pcm_reserved[i] != substream) {
+ mutex_unlock(&ice->open_mutex);
+ return -EBUSY;
+ }
+ ice->pcm_reserved[i] = substream;
+ }
+ for (; i < 3; i++) {
+ if (ice->pcm_reserved[i] == substream)
+ ice->pcm_reserved[i] = NULL;
+ }
+ } else {
+ for (i = 0; i < 3; i++) {
+ /* check individual playback stream */
+ if (ice->playback_con_substream_ds[i] == substream) {
+ if (ice->pcm_reserved[i] &&
+ ice->pcm_reserved[i] != substream) {
+ mutex_unlock(&ice->open_mutex);
+ return -EBUSY;
+ }
+ ice->pcm_reserved[i] = substream;
+ break;
+ }
+ }
+ }
+ mutex_unlock(&ice->open_mutex);
+ snd_vt1724_set_pro_rate(ice, params_rate(hw_params), 0);
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_vt1724_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ int i;
+
+ mutex_lock(&ice->open_mutex);
+ /* unmark surround channels */
+ for (i = 0; i < 3; i++)
+ if (ice->pcm_reserved[i] == substream)
+ ice->pcm_reserved[i] = NULL;
+ mutex_unlock(&ice->open_mutex);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_vt1724_playback_pro_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ unsigned char val;
+ unsigned int size;
+
+ spin_lock_irq(&ice->reg_lock);
+ val = (8 - substream->runtime->channels) >> 1;
+ outb(val, ICEMT1724(ice, BURST));
+
+ outl(substream->runtime->dma_addr, ICEMT1724(ice, PLAYBACK_ADDR));
+
+ size = (snd_pcm_lib_buffer_bytes(substream) >> 2) - 1;
+ /* outl(size, ICEMT1724(ice, PLAYBACK_SIZE)); */
+ outw(size, ICEMT1724(ice, PLAYBACK_SIZE));
+ outb(size >> 16, ICEMT1724(ice, PLAYBACK_SIZE) + 2);
+ size = (snd_pcm_lib_period_bytes(substream) >> 2) - 1;
+ /* outl(size, ICEMT1724(ice, PLAYBACK_COUNT)); */
+ outw(size, ICEMT1724(ice, PLAYBACK_COUNT));
+ outb(size >> 16, ICEMT1724(ice, PLAYBACK_COUNT) + 2);
+
+ spin_unlock_irq(&ice->reg_lock);
+
+ /* printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); */
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_vt1724_playback_pro_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(inl(ICEMT1724(ice, DMA_CONTROL)) & VT1724_PDMA0_START))
+ return 0;
+#if 0 /* read PLAYBACK_ADDR */
+ ptr = inl(ICEMT1724(ice, PLAYBACK_ADDR));
+ if (ptr < substream->runtime->dma_addr) {
+ snd_printd("ice1724: invalid negative ptr\n");
+ return 0;
+ }
+ ptr -= substream->runtime->dma_addr;
+ ptr = bytes_to_frames(substream->runtime, ptr);
+ if (ptr >= substream->runtime->buffer_size) {
+ snd_printd("ice1724: invalid ptr %d (size=%d)\n",
+ (int)ptr, (int)substream->runtime->period_size);
+ return 0;
+ }
+#else /* read PLAYBACK_SIZE */
+ ptr = inl(ICEMT1724(ice, PLAYBACK_SIZE)) & 0xffffff;
+ ptr = (ptr + 1) << 2;
+ ptr = bytes_to_frames(substream->runtime, ptr);
+ if (!ptr)
+ ;
+ else if (ptr <= substream->runtime->buffer_size)
+ ptr = substream->runtime->buffer_size - ptr;
+ else {
+ snd_printd("ice1724: invalid ptr %d (size=%d)\n",
+ (int)ptr, (int)substream->runtime->buffer_size);
+ ptr = 0;
+ }
+#endif
+ return ptr;
+}
+
+static int snd_vt1724_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ const struct vt1724_pcm_reg *reg = substream->runtime->private_data;
+
+ spin_lock_irq(&ice->reg_lock);
+ outl(substream->runtime->dma_addr, ice->profi_port + reg->addr);
+ outw((snd_pcm_lib_buffer_bytes(substream) >> 2) - 1,
+ ice->profi_port + reg->size);
+ outw((snd_pcm_lib_period_bytes(substream) >> 2) - 1,
+ ice->profi_port + reg->count);
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_vt1724_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ const struct vt1724_pcm_reg *reg = substream->runtime->private_data;
+ size_t ptr;
+
+ if (!(inl(ICEMT1724(ice, DMA_CONTROL)) & reg->start))
+ return 0;
+#if 0 /* use ADDR register */
+ ptr = inl(ice->profi_port + reg->addr);
+ ptr -= substream->runtime->dma_addr;
+ return bytes_to_frames(substream->runtime, ptr);
+#else /* use SIZE register */
+ ptr = inw(ice->profi_port + reg->size);
+ ptr = (ptr + 1) << 2;
+ ptr = bytes_to_frames(substream->runtime, ptr);
+ if (!ptr)
+ ;
+ else if (ptr <= substream->runtime->buffer_size)
+ ptr = substream->runtime->buffer_size - ptr;
+ else {
+ snd_printd("ice1724: invalid ptr %d (size=%d)\n",
+ (int)ptr, (int)substream->runtime->buffer_size);
+ ptr = 0;
+ }
+ return ptr;
+#endif
+}
+
+static const struct vt1724_pcm_reg vt1724_playback_pro_reg = {
+ .addr = VT1724_MT_PLAYBACK_ADDR,
+ .size = VT1724_MT_PLAYBACK_SIZE,
+ .count = VT1724_MT_PLAYBACK_COUNT,
+ .start = VT1724_PDMA0_START,
+};
+
+static const struct vt1724_pcm_reg vt1724_capture_pro_reg = {
+ .addr = VT1724_MT_CAPTURE_ADDR,
+ .size = VT1724_MT_CAPTURE_SIZE,
+ .count = VT1724_MT_CAPTURE_COUNT,
+ .start = VT1724_RDMA0_START,
+};
+
+static const struct snd_pcm_hardware snd_vt1724_playback_pro = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_192000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 8,
+ .buffer_bytes_max = (1UL << 21), /* 19bits dword */
+ .period_bytes_min = 8 * 4 * 2, /* FIXME: constraints needed */
+ .period_bytes_max = (1UL << 21),
+ .periods_min = 2,
+ .periods_max = 1024,
+};
+
+static const struct snd_pcm_hardware snd_vt1724_spdif = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_32000|SNDRV_PCM_RATE_44100|
+ SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_88200|
+ SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_176400|
+ SNDRV_PCM_RATE_192000),
+ .rate_min = 32000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (1UL << 18), /* 16bits dword */
+ .period_bytes_min = 2 * 4 * 2,
+ .period_bytes_max = (1UL << 18),
+ .periods_min = 2,
+ .periods_max = 1024,
+};
+
+static const struct snd_pcm_hardware snd_vt1724_2ch_stereo = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_192000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (1UL << 18), /* 16bits dword */
+ .period_bytes_min = 2 * 4 * 2,
+ .period_bytes_max = (1UL << 18),
+ .periods_min = 2,
+ .periods_max = 1024,
+};
+
+/*
+ * set rate constraints
+ */
+static void set_std_hw_rates(struct snd_ice1712 *ice)
+{
+ if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) {
+ /* I2S */
+ /* VT1720 doesn't support more than 96kHz */
+ if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720)
+ ice->hw_rates = &hw_constraints_rates_192;
+ else
+ ice->hw_rates = &hw_constraints_rates_96;
+ } else {
+ /* ACLINK */
+ ice->hw_rates = &hw_constraints_rates_48;
+ }
+}
+
+static int set_rate_constraints(struct snd_ice1712 *ice,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.rate_min = ice->hw_rates->list[0];
+ runtime->hw.rate_max = ice->hw_rates->list[ice->hw_rates->count - 1];
+ runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+ return snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ ice->hw_rates);
+}
+
+/* multi-channel playback needs alignment 8x32bit regardless of the channels
+ * actually used
+ */
+#define VT1724_BUFFER_ALIGN 0x20
+
+static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ int chs, num_indeps;
+
+ runtime->private_data = (void *)&vt1724_playback_pro_reg;
+ ice->playback_pro_substream = substream;
+ runtime->hw = snd_vt1724_playback_pro;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ set_rate_constraints(ice, substream);
+ mutex_lock(&ice->open_mutex);
+ /* calculate the currently available channels */
+ num_indeps = ice->num_total_dacs / 2 - 1;
+ for (chs = 0; chs < num_indeps; chs++) {
+ if (ice->pcm_reserved[chs])
+ break;
+ }
+ chs = (chs + 1) * 2;
+ runtime->hw.channels_max = chs;
+ if (chs > 2) /* channels must be even */
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
+ mutex_unlock(&ice->open_mutex);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ VT1724_BUFFER_ALIGN);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ VT1724_BUFFER_ALIGN);
+ return 0;
+}
+
+static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->private_data = (void *)&vt1724_capture_pro_reg;
+ ice->capture_pro_substream = substream;
+ runtime->hw = snd_vt1724_2ch_stereo;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ set_rate_constraints(ice, substream);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ VT1724_BUFFER_ALIGN);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ VT1724_BUFFER_ALIGN);
+ return 0;
+}
+
+static int snd_vt1724_playback_pro_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ if (PRO_RATE_RESET)
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
+ ice->playback_pro_substream = NULL;
+
+ return 0;
+}
+
+static int snd_vt1724_capture_pro_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ if (PRO_RATE_RESET)
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
+ ice->capture_pro_substream = NULL;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_vt1724_playback_pro_ops = {
+ .open = snd_vt1724_playback_pro_open,
+ .close = snd_vt1724_playback_pro_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_vt1724_pcm_hw_params,
+ .hw_free = snd_vt1724_pcm_hw_free,
+ .prepare = snd_vt1724_playback_pro_prepare,
+ .trigger = snd_vt1724_pcm_trigger,
+ .pointer = snd_vt1724_playback_pro_pointer,
+};
+
+static struct snd_pcm_ops snd_vt1724_capture_pro_ops = {
+ .open = snd_vt1724_capture_pro_open,
+ .close = snd_vt1724_capture_pro_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_vt1724_pcm_hw_params,
+ .hw_free = snd_vt1724_pcm_hw_free,
+ .prepare = snd_vt1724_pcm_prepare,
+ .trigger = snd_vt1724_pcm_trigger,
+ .pointer = snd_vt1724_pcm_pointer,
+};
+
+static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(ice->card, "ICE1724", device, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_vt1724_playback_pro_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vt1724_capture_pro_ops);
+
+ pcm->private_data = ice;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "ICE1724");
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ice->pci),
+ 256*1024, 256*1024);
+
+ ice->pcm_pro = pcm;
+
+ return 0;
+}
+
+
+/*
+ * SPDIF PCM
+ */
+
+static const struct vt1724_pcm_reg vt1724_playback_spdif_reg = {
+ .addr = VT1724_MT_PDMA4_ADDR,
+ .size = VT1724_MT_PDMA4_SIZE,
+ .count = VT1724_MT_PDMA4_COUNT,
+ .start = VT1724_PDMA4_START,
+};
+
+static const struct vt1724_pcm_reg vt1724_capture_spdif_reg = {
+ .addr = VT1724_MT_RDMA1_ADDR,
+ .size = VT1724_MT_RDMA1_SIZE,
+ .count = VT1724_MT_RDMA1_COUNT,
+ .start = VT1724_RDMA1_START,
+};
+
+/* update spdif control bits; call with reg_lock */
+static void update_spdif_bits(struct snd_ice1712 *ice, unsigned int val)
+{
+ unsigned char cbit, disabled;
+
+ cbit = inb(ICEREG1724(ice, SPDIF_CFG));
+ disabled = cbit & ~VT1724_CFG_SPDIF_OUT_EN;
+ if (cbit != disabled)
+ outb(disabled, ICEREG1724(ice, SPDIF_CFG));
+ outw(val, ICEMT1724(ice, SPDIF_CTRL));
+ if (cbit != disabled)
+ outb(cbit, ICEREG1724(ice, SPDIF_CFG));
+ outw(val, ICEMT1724(ice, SPDIF_CTRL));
+}
+
+/* update SPDIF control bits according to the given rate */
+static void update_spdif_rate(struct snd_ice1712 *ice, unsigned int rate)
+{
+ unsigned int val, nval;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ice->reg_lock, flags);
+ nval = val = inw(ICEMT1724(ice, SPDIF_CTRL));
+ nval &= ~(7 << 12);
+ switch (rate) {
+ case 44100: break;
+ case 48000: nval |= 2 << 12; break;
+ case 32000: nval |= 3 << 12; break;
+ case 88200: nval |= 4 << 12; break;
+ case 96000: nval |= 5 << 12; break;
+ case 192000: nval |= 6 << 12; break;
+ case 176400: nval |= 7 << 12; break;
+ }
+ if (val != nval)
+ update_spdif_bits(ice, nval);
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+}
+
+static int snd_vt1724_playback_spdif_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ if (!ice->force_pdma4)
+ update_spdif_rate(ice, substream->runtime->rate);
+ return snd_vt1724_pcm_prepare(substream);
+}
+
+static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->private_data = (void *)&vt1724_playback_spdif_reg;
+ ice->playback_con_substream = substream;
+ if (ice->force_pdma4) {
+ runtime->hw = snd_vt1724_2ch_stereo;
+ set_rate_constraints(ice, substream);
+ } else
+ runtime->hw = snd_vt1724_spdif;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ VT1724_BUFFER_ALIGN);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ VT1724_BUFFER_ALIGN);
+ if (ice->spdif.ops.open)
+ ice->spdif.ops.open(ice, substream);
+ return 0;
+}
+
+static int snd_vt1724_playback_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ if (PRO_RATE_RESET)
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
+ ice->playback_con_substream = NULL;
+ if (ice->spdif.ops.close)
+ ice->spdif.ops.close(ice, substream);
+
+ return 0;
+}
+
+static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->private_data = (void *)&vt1724_capture_spdif_reg;
+ ice->capture_con_substream = substream;
+ if (ice->force_rdma1) {
+ runtime->hw = snd_vt1724_2ch_stereo;
+ set_rate_constraints(ice, substream);
+ } else
+ runtime->hw = snd_vt1724_spdif;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ VT1724_BUFFER_ALIGN);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ VT1724_BUFFER_ALIGN);
+ if (ice->spdif.ops.open)
+ ice->spdif.ops.open(ice, substream);
+ return 0;
+}
+
+static int snd_vt1724_capture_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ if (PRO_RATE_RESET)
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
+ ice->capture_con_substream = NULL;
+ if (ice->spdif.ops.close)
+ ice->spdif.ops.close(ice, substream);
+
+ return 0;
+}
+
+static struct snd_pcm_ops snd_vt1724_playback_spdif_ops = {
+ .open = snd_vt1724_playback_spdif_open,
+ .close = snd_vt1724_playback_spdif_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_vt1724_pcm_hw_params,
+ .hw_free = snd_vt1724_pcm_hw_free,
+ .prepare = snd_vt1724_playback_spdif_prepare,
+ .trigger = snd_vt1724_pcm_trigger,
+ .pointer = snd_vt1724_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_vt1724_capture_spdif_ops = {
+ .open = snd_vt1724_capture_spdif_open,
+ .close = snd_vt1724_capture_spdif_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_vt1724_pcm_hw_params,
+ .hw_free = snd_vt1724_pcm_hw_free,
+ .prepare = snd_vt1724_pcm_prepare,
+ .trigger = snd_vt1724_pcm_trigger,
+ .pointer = snd_vt1724_pcm_pointer,
+};
+
+
+static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device)
+{
+ char *name;
+ struct snd_pcm *pcm;
+ int play, capt;
+ int err;
+
+ if (ice->force_pdma4 ||
+ (ice->eeprom.data[ICE_EEP2_SPDIF] & VT1724_CFG_SPDIF_OUT_INT)) {
+ play = 1;
+ ice->has_spdif = 1;
+ } else
+ play = 0;
+ if (ice->force_rdma1 ||
+ (ice->eeprom.data[ICE_EEP2_SPDIF] & VT1724_CFG_SPDIF_IN)) {
+ capt = 1;
+ ice->has_spdif = 1;
+ } else
+ capt = 0;
+ if (!play && !capt)
+ return 0; /* no spdif device */
+
+ if (ice->force_pdma4 || ice->force_rdma1)
+ name = "ICE1724 Secondary";
+ else
+ name = "IEC1724 IEC958";
+ err = snd_pcm_new(ice->card, name, device, play, capt, &pcm);
+ if (err < 0)
+ return err;
+
+ if (play)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_vt1724_playback_spdif_ops);
+ if (capt)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_vt1724_capture_spdif_ops);
+
+ pcm->private_data = ice;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, name);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ice->pci),
+ 64*1024, 64*1024);
+
+ ice->pcm = pcm;
+
+ return 0;
+}
+
+
+/*
+ * independent surround PCMs
+ */
+
+static const struct vt1724_pcm_reg vt1724_playback_dma_regs[3] = {
+ {
+ .addr = VT1724_MT_PDMA1_ADDR,
+ .size = VT1724_MT_PDMA1_SIZE,
+ .count = VT1724_MT_PDMA1_COUNT,
+ .start = VT1724_PDMA1_START,
+ },
+ {
+ .addr = VT1724_MT_PDMA2_ADDR,
+ .size = VT1724_MT_PDMA2_SIZE,
+ .count = VT1724_MT_PDMA2_COUNT,
+ .start = VT1724_PDMA2_START,
+ },
+ {
+ .addr = VT1724_MT_PDMA3_ADDR,
+ .size = VT1724_MT_PDMA3_SIZE,
+ .count = VT1724_MT_PDMA3_COUNT,
+ .start = VT1724_PDMA3_START,
+ },
+};
+
+static int snd_vt1724_playback_indep_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ unsigned char val;
+
+ spin_lock_irq(&ice->reg_lock);
+ val = 3 - substream->number;
+ if (inb(ICEMT1724(ice, BURST)) < val)
+ outb(val, ICEMT1724(ice, BURST));
+ spin_unlock_irq(&ice->reg_lock);
+ return snd_vt1724_pcm_prepare(substream);
+}
+
+static int snd_vt1724_playback_indep_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ mutex_lock(&ice->open_mutex);
+ /* already used by PDMA0? */
+ if (ice->pcm_reserved[substream->number]) {
+ mutex_unlock(&ice->open_mutex);
+ return -EBUSY; /* FIXME: should handle blocking mode properly */
+ }
+ mutex_unlock(&ice->open_mutex);
+ runtime->private_data = (void *)&vt1724_playback_dma_regs[substream->number];
+ ice->playback_con_substream_ds[substream->number] = substream;
+ runtime->hw = snd_vt1724_2ch_stereo;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ set_rate_constraints(ice, substream);
+ return 0;
+}
+
+static int snd_vt1724_playback_indep_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+
+ if (PRO_RATE_RESET)
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 0);
+ ice->playback_con_substream_ds[substream->number] = NULL;
+ ice->pcm_reserved[substream->number] = NULL;
+
+ return 0;
+}
+
+static struct snd_pcm_ops snd_vt1724_playback_indep_ops = {
+ .open = snd_vt1724_playback_indep_open,
+ .close = snd_vt1724_playback_indep_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_vt1724_pcm_hw_params,
+ .hw_free = snd_vt1724_pcm_hw_free,
+ .prepare = snd_vt1724_playback_indep_prepare,
+ .trigger = snd_vt1724_pcm_trigger,
+ .pointer = snd_vt1724_pcm_pointer,
+};
+
+
+static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device)
+{
+ struct snd_pcm *pcm;
+ int play;
+ int err;
+
+ play = ice->num_total_dacs / 2 - 1;
+ if (play <= 0)
+ return 0;
+
+ err = snd_pcm_new(ice->card, "ICE1724 Surrounds", device, play, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_vt1724_playback_indep_ops);
+
+ pcm->private_data = ice;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "ICE1724 Surround PCM");
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(ice->pci),
+ 64*1024, 64*1024);
+
+ ice->pcm_ds = pcm;
+
+ return 0;
+}
+
+
+/*
+ * Mixer section
+ */
+
+static int __devinit snd_vt1724_ac97_mixer(struct snd_ice1712 *ice)
+{
+ int err;
+
+ if (!(ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S)) {
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_vt1724_ac97_write,
+ .read = snd_vt1724_ac97_read,
+ };
+
+ /* cold reset */
+ outb(inb(ICEMT1724(ice, AC97_CMD)) | 0x80, ICEMT1724(ice, AC97_CMD));
+ mdelay(5); /* FIXME */
+ outb(inb(ICEMT1724(ice, AC97_CMD)) & ~0x80, ICEMT1724(ice, AC97_CMD));
+
+ err = snd_ac97_bus(ice->card, 0, &ops, NULL, &pbus);
+ if (err < 0)
+ return err;
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = ice;
+ err = snd_ac97_mixer(pbus, &ac97, &ice->ac97);
+ if (err < 0)
+ printk(KERN_WARNING "ice1712: cannot initialize pro ac97, skipped\n");
+ else
+ return 0;
+ }
+ /* I2S mixer only */
+ strcat(ice->card->mixername, "ICE1724 - multitrack");
+ return 0;
+}
+
+/*
+ *
+ */
+
+static inline unsigned int eeprom_triple(struct snd_ice1712 *ice, int idx)
+{
+ return (unsigned int)ice->eeprom.data[idx] | \
+ ((unsigned int)ice->eeprom.data[idx + 1] << 8) | \
+ ((unsigned int)ice->eeprom.data[idx + 2] << 16);
+}
+
+static void snd_vt1724_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = entry->private_data;
+ unsigned int idx;
+
+ snd_iprintf(buffer, "%s\n\n", ice->card->longname);
+ snd_iprintf(buffer, "EEPROM:\n");
+
+ snd_iprintf(buffer, " Subvendor : 0x%x\n", ice->eeprom.subvendor);
+ snd_iprintf(buffer, " Size : %i bytes\n", ice->eeprom.size);
+ snd_iprintf(buffer, " Version : %i\n", ice->eeprom.version);
+ snd_iprintf(buffer, " System Config : 0x%x\n",
+ ice->eeprom.data[ICE_EEP2_SYSCONF]);
+ snd_iprintf(buffer, " ACLink : 0x%x\n",
+ ice->eeprom.data[ICE_EEP2_ACLINK]);
+ snd_iprintf(buffer, " I2S : 0x%x\n",
+ ice->eeprom.data[ICE_EEP2_I2S]);
+ snd_iprintf(buffer, " S/PDIF : 0x%x\n",
+ ice->eeprom.data[ICE_EEP2_SPDIF]);
+ snd_iprintf(buffer, " GPIO direction : 0x%x\n",
+ ice->eeprom.gpiodir);
+ snd_iprintf(buffer, " GPIO mask : 0x%x\n",
+ ice->eeprom.gpiomask);
+ snd_iprintf(buffer, " GPIO state : 0x%x\n",
+ ice->eeprom.gpiostate);
+ for (idx = 0x12; idx < ice->eeprom.size; idx++)
+ snd_iprintf(buffer, " Extra #%02i : 0x%x\n",
+ idx, ice->eeprom.data[idx]);
+
+ snd_iprintf(buffer, "\nRegisters:\n");
+
+ snd_iprintf(buffer, " PSDOUT03 : 0x%08x\n",
+ (unsigned)inl(ICEMT1724(ice, ROUTE_PLAYBACK)));
+ for (idx = 0x0; idx < 0x20 ; idx++)
+ snd_iprintf(buffer, " CCS%02x : 0x%02x\n",
+ idx, inb(ice->port+idx));
+ for (idx = 0x0; idx < 0x30 ; idx++)
+ snd_iprintf(buffer, " MT%02x : 0x%02x\n",
+ idx, inb(ice->profi_port+idx));
+}
+
+static void __devinit snd_vt1724_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(ice->card, "ice1724", &entry))
+ snd_info_set_text_ops(entry, ice, snd_vt1724_proc_read);
+}
+
+/*
+ *
+ */
+
+static int snd_vt1724_eeprom_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = sizeof(struct snd_ice1712_eeprom);
+ return 0;
+}
+
+static int snd_vt1724_eeprom_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ memcpy(ucontrol->value.bytes.data, &ice->eeprom, sizeof(ice->eeprom));
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_vt1724_eeprom __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "ICE1724 EEPROM",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_vt1724_eeprom_info,
+ .get = snd_vt1724_eeprom_get
+};
+
+/*
+ */
+static int snd_vt1724_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static unsigned int encode_spdif_bits(struct snd_aes_iec958 *diga)
+{
+ unsigned int val, rbits;
+
+ val = diga->status[0] & 0x03; /* professional, non-audio */
+ if (val & 0x01) {
+ /* professional */
+ if ((diga->status[0] & IEC958_AES0_PRO_EMPHASIS) ==
+ IEC958_AES0_PRO_EMPHASIS_5015)
+ val |= 1U << 3;
+ rbits = (diga->status[4] >> 3) & 0x0f;
+ if (rbits) {
+ switch (rbits) {
+ case 2: val |= 5 << 12; break; /* 96k */
+ case 3: val |= 6 << 12; break; /* 192k */
+ case 10: val |= 4 << 12; break; /* 88.2k */
+ case 11: val |= 7 << 12; break; /* 176.4k */
+ }
+ } else {
+ switch (diga->status[0] & IEC958_AES0_PRO_FS) {
+ case IEC958_AES0_PRO_FS_44100:
+ break;
+ case IEC958_AES0_PRO_FS_32000:
+ val |= 3U << 12;
+ break;
+ default:
+ val |= 2U << 12;
+ break;
+ }
+ }
+ } else {
+ /* consumer */
+ val |= diga->status[1] & 0x04; /* copyright */
+ if ((diga->status[0] & IEC958_AES0_CON_EMPHASIS) ==
+ IEC958_AES0_CON_EMPHASIS_5015)
+ val |= 1U << 3;
+ val |= (unsigned int)(diga->status[1] & 0x3f) << 4; /* category */
+ val |= (unsigned int)(diga->status[3] & IEC958_AES3_CON_FS) << 12; /* fs */
+ }
+ return val;
+}
+
+static void decode_spdif_bits(struct snd_aes_iec958 *diga, unsigned int val)
+{
+ memset(diga->status, 0, sizeof(diga->status));
+ diga->status[0] = val & 0x03; /* professional, non-audio */
+ if (val & 0x01) {
+ /* professional */
+ if (val & (1U << 3))
+ diga->status[0] |= IEC958_AES0_PRO_EMPHASIS_5015;
+ switch ((val >> 12) & 0x7) {
+ case 0:
+ break;
+ case 2:
+ diga->status[0] |= IEC958_AES0_PRO_FS_32000;
+ break;
+ default:
+ diga->status[0] |= IEC958_AES0_PRO_FS_48000;
+ break;
+ }
+ } else {
+ /* consumer */
+ diga->status[0] |= val & (1U << 2); /* copyright */
+ if (val & (1U << 3))
+ diga->status[0] |= IEC958_AES0_CON_EMPHASIS_5015;
+ diga->status[1] |= (val >> 4) & 0x3f; /* category */
+ diga->status[3] |= (val >> 12) & 0x07; /* fs */
+ }
+}
+
+static int snd_vt1724_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ val = inw(ICEMT1724(ice, SPDIF_CTRL));
+ decode_spdif_bits(&ucontrol->value.iec958, val);
+ return 0;
+}
+
+static int snd_vt1724_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int val, old;
+
+ val = encode_spdif_bits(&ucontrol->value.iec958);
+ spin_lock_irq(&ice->reg_lock);
+ old = inw(ICEMT1724(ice, SPDIF_CTRL));
+ if (val != old)
+ update_spdif_bits(ice, val);
+ spin_unlock_irq(&ice->reg_lock);
+ return val != old;
+}
+
+static struct snd_kcontrol_new snd_vt1724_spdif_default __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .info = snd_vt1724_spdif_info,
+ .get = snd_vt1724_spdif_default_get,
+ .put = snd_vt1724_spdif_default_put
+};
+
+static int snd_vt1724_spdif_maskc_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_CON_NOT_COPYRIGHT |
+ IEC958_AES0_CON_EMPHASIS;
+ ucontrol->value.iec958.status[1] = IEC958_AES1_CON_ORIGINAL |
+ IEC958_AES1_CON_CATEGORY;
+ ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS;
+ return 0;
+}
+
+static int snd_vt1724_spdif_maskp_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_PRO_FS |
+ IEC958_AES0_PRO_EMPHASIS;
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_vt1724_spdif_maskc __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK),
+ .info = snd_vt1724_spdif_info,
+ .get = snd_vt1724_spdif_maskc_get,
+};
+
+static struct snd_kcontrol_new snd_vt1724_spdif_maskp __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK),
+ .info = snd_vt1724_spdif_info,
+ .get = snd_vt1724_spdif_maskp_get,
+};
+
+#define snd_vt1724_spdif_sw_info snd_ctl_boolean_mono_info
+
+static int snd_vt1724_spdif_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = inb(ICEREG1724(ice, SPDIF_CFG)) &
+ VT1724_CFG_SPDIF_OUT_EN ? 1 : 0;
+ return 0;
+}
+
+static int snd_vt1724_spdif_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char old, val;
+
+ spin_lock_irq(&ice->reg_lock);
+ old = val = inb(ICEREG1724(ice, SPDIF_CFG));
+ val &= ~VT1724_CFG_SPDIF_OUT_EN;
+ if (ucontrol->value.integer.value[0])
+ val |= VT1724_CFG_SPDIF_OUT_EN;
+ if (old != val)
+ outb(val, ICEREG1724(ice, SPDIF_CFG));
+ spin_unlock_irq(&ice->reg_lock);
+ return old != val;
+}
+
+static struct snd_kcontrol_new snd_vt1724_spdif_switch __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* FIXME: the following conflict with IEC958 Playback Route */
+ /* .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), */
+ .name = SNDRV_CTL_NAME_IEC958("Output ", NONE, SWITCH),
+ .info = snd_vt1724_spdif_sw_info,
+ .get = snd_vt1724_spdif_sw_get,
+ .put = snd_vt1724_spdif_sw_put
+};
+
+
+#if 0 /* NOT USED YET */
+/*
+ * GPIO access from extern
+ */
+
+#define snd_vt1724_gpio_info snd_ctl_boolean_mono_info
+
+int snd_vt1724_gpio_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value & (1<<24)) ? 1 : 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ ucontrol->value.integer.value[0] =
+ (snd_ice1712_gpio_read(ice) & (1 << shift) ? 1 : 0) ^ invert;
+ snd_ice1712_restore_gpio_status(ice);
+ return 0;
+}
+
+int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int shift = kcontrol->private_value & 0xff;
+ int invert = (kcontrol->private_value & (1<<24)) ? mask : 0;
+ unsigned int val, nval;
+
+ if (kcontrol->private_value & (1 << 31))
+ return -EPERM;
+ nval = (ucontrol->value.integer.value[0] ? (1 << shift) : 0) ^ invert;
+ snd_ice1712_save_gpio_status(ice);
+ val = snd_ice1712_gpio_read(ice);
+ nval |= val & ~(1 << shift);
+ if (val != nval)
+ snd_ice1712_gpio_write(ice, nval);
+ snd_ice1712_restore_gpio_status(ice);
+ return val != nval;
+}
+#endif /* NOT USED YET */
+
+/*
+ * rate
+ */
+static int snd_vt1724_pro_internal_clock_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = ice->hw_rates->count + 1;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ if (uinfo->value.enumerated.item == uinfo->value.enumerated.items - 1)
+ strcpy(uinfo->value.enumerated.name, "IEC958 Input");
+ else
+ sprintf(uinfo->value.enumerated.name, "%d",
+ ice->hw_rates->list[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int i, rate;
+
+ spin_lock_irq(&ice->reg_lock);
+ if (ice->is_spdif_master(ice)) {
+ ucontrol->value.enumerated.item[0] = ice->hw_rates->count;
+ } else {
+ rate = ice->get_rate(ice);
+ ucontrol->value.enumerated.item[0] = 0;
+ for (i = 0; i < ice->hw_rates->count; i++) {
+ if (ice->hw_rates->list[i] == rate) {
+ ucontrol->value.enumerated.item[0] = i;
+ break;
+ }
+ }
+ }
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+/* setting clock to external - SPDIF */
+static void stdclock_set_spdif_clock(struct snd_ice1712 *ice)
+{
+ unsigned char oval;
+ unsigned char i2s_oval;
+ oval = inb(ICEMT1724(ice, RATE));
+ outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE));
+ /* setting 256fs */
+ i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT));
+ outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT));
+}
+
+static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int old_rate, new_rate;
+ unsigned int item = ucontrol->value.enumerated.item[0];
+ unsigned int spdif = ice->hw_rates->count;
+
+ if (item > spdif)
+ return -EINVAL;
+
+ spin_lock_irq(&ice->reg_lock);
+ if (ice->is_spdif_master(ice))
+ old_rate = 0;
+ else
+ old_rate = ice->get_rate(ice);
+ if (item == spdif) {
+ /* switching to external clock via SPDIF */
+ ice->set_spdif_clock(ice);
+ new_rate = 0;
+ } else {
+ /* internal on-card clock */
+ new_rate = ice->hw_rates->list[item];
+ ice->pro_rate_default = new_rate;
+ spin_unlock_irq(&ice->reg_lock);
+ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1);
+ spin_lock_irq(&ice->reg_lock);
+ }
+ spin_unlock_irq(&ice->reg_lock);
+
+ /* the first reset to the SPDIF master mode? */
+ if (old_rate != new_rate && !new_rate) {
+ /* notify akm chips as well */
+ unsigned int i;
+ if (ice->gpio.set_pro_rate)
+ ice->gpio.set_pro_rate(ice, 0);
+ for (i = 0; i < ice->akm_codecs; i++) {
+ if (ice->akm[i].ops.set_rate_val)
+ ice->akm[i].ops.set_rate_val(&ice->akm[i], 0);
+ }
+ }
+ return old_rate != new_rate;
+}
+
+static struct snd_kcontrol_new snd_vt1724_pro_internal_clock __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Internal Clock",
+ .info = snd_vt1724_pro_internal_clock_info,
+ .get = snd_vt1724_pro_internal_clock_get,
+ .put = snd_vt1724_pro_internal_clock_put
+};
+
+#define snd_vt1724_pro_rate_locking_info snd_ctl_boolean_mono_info
+
+static int snd_vt1724_pro_rate_locking_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = PRO_RATE_LOCKED;
+ return 0;
+}
+
+static int snd_vt1724_pro_rate_locking_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int change = 0, nval;
+
+ nval = ucontrol->value.integer.value[0] ? 1 : 0;
+ spin_lock_irq(&ice->reg_lock);
+ change = PRO_RATE_LOCKED != nval;
+ PRO_RATE_LOCKED = nval;
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_vt1724_pro_rate_locking __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Rate Locking",
+ .info = snd_vt1724_pro_rate_locking_info,
+ .get = snd_vt1724_pro_rate_locking_get,
+ .put = snd_vt1724_pro_rate_locking_put
+};
+
+#define snd_vt1724_pro_rate_reset_info snd_ctl_boolean_mono_info
+
+static int snd_vt1724_pro_rate_reset_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = PRO_RATE_RESET ? 1 : 0;
+ return 0;
+}
+
+static int snd_vt1724_pro_rate_reset_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int change = 0, nval;
+
+ nval = ucontrol->value.integer.value[0] ? 1 : 0;
+ spin_lock_irq(&ice->reg_lock);
+ change = PRO_RATE_RESET != nval;
+ PRO_RATE_RESET = nval;
+ spin_unlock_irq(&ice->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_vt1724_pro_rate_reset __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Rate Reset",
+ .info = snd_vt1724_pro_rate_reset_info,
+ .get = snd_vt1724_pro_rate_reset_get,
+ .put = snd_vt1724_pro_rate_reset_put
+};
+
+
+/*
+ * routing
+ */
+static int snd_vt1724_pro_route_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {
+ "PCM Out", /* 0 */
+ "H/W In 0", "H/W In 1", /* 1-2 */
+ "IEC958 In L", "IEC958 In R", /* 3-4 */
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 5;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static inline int analog_route_shift(int idx)
+{
+ return (idx % 2) * 12 + ((idx / 2) * 3) + 8;
+}
+
+static inline int digital_route_shift(int idx)
+{
+ return idx * 3;
+}
+
+static int get_route_val(struct snd_ice1712 *ice, int shift)
+{
+ unsigned long val;
+ unsigned char eitem;
+ static const unsigned char xlate[8] = {
+ 0, 255, 1, 2, 255, 255, 3, 4,
+ };
+
+ val = inl(ICEMT1724(ice, ROUTE_PLAYBACK));
+ val >>= shift;
+ val &= 7; /* we now have 3 bits per output */
+ eitem = xlate[val];
+ if (eitem == 255) {
+ snd_BUG();
+ return 0;
+ }
+ return eitem;
+}
+
+static int put_route_val(struct snd_ice1712 *ice, unsigned int val, int shift)
+{
+ unsigned int old_val, nval;
+ int change;
+ static const unsigned char xroute[8] = {
+ 0, /* PCM */
+ 2, /* PSDIN0 Left */
+ 3, /* PSDIN0 Right */
+ 6, /* SPDIN Left */
+ 7, /* SPDIN Right */
+ };
+
+ nval = xroute[val % 5];
+ val = old_val = inl(ICEMT1724(ice, ROUTE_PLAYBACK));
+ val &= ~(0x07 << shift);
+ val |= nval << shift;
+ change = val != old_val;
+ if (change)
+ outl(val, ICEMT1724(ice, ROUTE_PLAYBACK));
+ return change;
+}
+
+static int snd_vt1724_pro_route_analog_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ ucontrol->value.enumerated.item[0] =
+ get_route_val(ice, analog_route_shift(idx));
+ return 0;
+}
+
+static int snd_vt1724_pro_route_analog_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ return put_route_val(ice, ucontrol->value.enumerated.item[0],
+ analog_route_shift(idx));
+}
+
+static int snd_vt1724_pro_route_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ ucontrol->value.enumerated.item[0] =
+ get_route_val(ice, digital_route_shift(idx));
+ return 0;
+}
+
+static int snd_vt1724_pro_route_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ return put_route_val(ice, ucontrol->value.enumerated.item[0],
+ digital_route_shift(idx));
+}
+
+static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "H/W Playback Route",
+ .info = snd_vt1724_pro_route_info,
+ .get = snd_vt1724_pro_route_analog_get,
+ .put = snd_vt1724_pro_route_analog_put,
+};
+
+static struct snd_kcontrol_new snd_vt1724_mixer_pro_spdif_route __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route",
+ .info = snd_vt1724_pro_route_info,
+ .get = snd_vt1724_pro_route_spdif_get,
+ .put = snd_vt1724_pro_route_spdif_put,
+ .count = 2,
+};
+
+
+static int snd_vt1724_pro_peak_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 22; /* FIXME: for compatibility with ice1712... */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx;
+
+ spin_lock_irq(&ice->reg_lock);
+ for (idx = 0; idx < 22; idx++) {
+ outb(idx, ICEMT1724(ice, MONITOR_PEAKINDEX));
+ ucontrol->value.integer.value[idx] =
+ inb(ICEMT1724(ice, MONITOR_PEAKDATA));
+ }
+ spin_unlock_irq(&ice->reg_lock);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Multi Track Peak",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+ .info = snd_vt1724_pro_peak_info,
+ .get = snd_vt1724_pro_peak_get
+};
+
+/*
+ *
+ */
+
+static struct snd_ice1712_card_info no_matched __devinitdata;
+
+static struct snd_ice1712_card_info *card_tables[] __devinitdata = {
+ snd_vt1724_revo_cards,
+ snd_vt1724_amp_cards,
+ snd_vt1724_aureon_cards,
+ snd_vt1720_mobo_cards,
+ snd_vt1720_pontis_cards,
+ snd_vt1724_prodigy_hifi_cards,
+ snd_vt1724_prodigy192_cards,
+ snd_vt1724_juli_cards,
+ snd_vt1724_phase_cards,
+ snd_vt1724_wtm_cards,
+ snd_vt1724_se_cards,
+ NULL,
+};
+
+
+/*
+ */
+
+static void wait_i2c_busy(struct snd_ice1712 *ice)
+{
+ int t = 0x10000;
+ while ((inb(ICEREG1724(ice, I2C_CTRL)) & VT1724_I2C_BUSY) && t--)
+ ;
+ if (t == -1)
+ printk(KERN_ERR "ice1724: i2c busy timeout\n");
+}
+
+unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice,
+ unsigned char dev, unsigned char addr)
+{
+ unsigned char val;
+
+ mutex_lock(&ice->i2c_mutex);
+ wait_i2c_busy(ice);
+ outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR));
+ outb(dev & ~VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR));
+ wait_i2c_busy(ice);
+ val = inb(ICEREG1724(ice, I2C_DATA));
+ mutex_unlock(&ice->i2c_mutex);
+ /* printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); */
+ return val;
+}
+
+void snd_vt1724_write_i2c(struct snd_ice1712 *ice,
+ unsigned char dev, unsigned char addr, unsigned char data)
+{
+ mutex_lock(&ice->i2c_mutex);
+ wait_i2c_busy(ice);
+ /* printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); */
+ outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR));
+ outb(data, ICEREG1724(ice, I2C_DATA));
+ outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR));
+ wait_i2c_busy(ice);
+ mutex_unlock(&ice->i2c_mutex);
+}
+
+static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice,
+ const char *modelname)
+{
+ const int dev = 0xa0; /* EEPROM device address */
+ unsigned int i, size;
+ struct snd_ice1712_card_info * const *tbl, *c;
+
+ if (!modelname || !*modelname) {
+ ice->eeprom.subvendor = 0;
+ if ((inb(ICEREG1724(ice, I2C_CTRL)) & VT1724_I2C_EEPROM) != 0)
+ ice->eeprom.subvendor =
+ (snd_vt1724_read_i2c(ice, dev, 0x00) << 0) |
+ (snd_vt1724_read_i2c(ice, dev, 0x01) << 8) |
+ (snd_vt1724_read_i2c(ice, dev, 0x02) << 16) |
+ (snd_vt1724_read_i2c(ice, dev, 0x03) << 24);
+ if (ice->eeprom.subvendor == 0 ||
+ ice->eeprom.subvendor == (unsigned int)-1) {
+ /* invalid subvendor from EEPROM, try the PCI
+ * subststem ID instead
+ */
+ u16 vendor, device;
+ pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID,
+ &vendor);
+ pci_read_config_word(ice->pci, PCI_SUBSYSTEM_ID, &device);
+ ice->eeprom.subvendor =
+ ((unsigned int)swab16(vendor) << 16) | swab16(device);
+ if (ice->eeprom.subvendor == 0 ||
+ ice->eeprom.subvendor == (unsigned int)-1) {
+ printk(KERN_ERR "ice1724: No valid ID is found\n");
+ return -ENXIO;
+ }
+ }
+ }
+ for (tbl = card_tables; *tbl; tbl++) {
+ for (c = *tbl; c->subvendor; c++) {
+ if (modelname && c->model &&
+ !strcmp(modelname, c->model)) {
+ printk(KERN_INFO "ice1724: Using board model %s\n",
+ c->name);
+ ice->eeprom.subvendor = c->subvendor;
+ } else if (c->subvendor != ice->eeprom.subvendor)
+ continue;
+ if (!c->eeprom_size || !c->eeprom_data)
+ goto found;
+ /* if the EEPROM is given by the driver, use it */
+ snd_printdd("using the defined eeprom..\n");
+ ice->eeprom.version = 2;
+ ice->eeprom.size = c->eeprom_size + 6;
+ memcpy(ice->eeprom.data, c->eeprom_data, c->eeprom_size);
+ goto read_skipped;
+ }
+ }
+ printk(KERN_WARNING "ice1724: No matching model found for ID 0x%x\n",
+ ice->eeprom.subvendor);
+
+ found:
+ ice->eeprom.size = snd_vt1724_read_i2c(ice, dev, 0x04);
+ if (ice->eeprom.size < 6)
+ ice->eeprom.size = 32;
+ else if (ice->eeprom.size > 32) {
+ printk(KERN_ERR "ice1724: Invalid EEPROM (size = %i)\n",
+ ice->eeprom.size);
+ return -EIO;
+ }
+ ice->eeprom.version = snd_vt1724_read_i2c(ice, dev, 0x05);
+ if (ice->eeprom.version != 2)
+ printk(KERN_WARNING "ice1724: Invalid EEPROM version %i\n",
+ ice->eeprom.version);
+ size = ice->eeprom.size - 6;
+ for (i = 0; i < size; i++)
+ ice->eeprom.data[i] = snd_vt1724_read_i2c(ice, dev, i + 6);
+
+ read_skipped:
+ ice->eeprom.gpiomask = eeprom_triple(ice, ICE_EEP2_GPIO_MASK);
+ ice->eeprom.gpiostate = eeprom_triple(ice, ICE_EEP2_GPIO_STATE);
+ ice->eeprom.gpiodir = eeprom_triple(ice, ICE_EEP2_GPIO_DIR);
+
+ return 0;
+}
+
+
+
+static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice)
+{
+ outb(VT1724_RESET , ICEREG1724(ice, CONTROL));
+ msleep(10);
+ outb(0, ICEREG1724(ice, CONTROL));
+ msleep(10);
+}
+
+static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice)
+{
+ outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG));
+ outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG));
+ outb(ice->eeprom.data[ICE_EEP2_I2S], ICEREG1724(ice, I2S_FEATURES));
+ outb(ice->eeprom.data[ICE_EEP2_SPDIF], ICEREG1724(ice, SPDIF_CFG));
+
+ ice->gpio.write_mask = ice->eeprom.gpiomask;
+ ice->gpio.direction = ice->eeprom.gpiodir;
+ snd_vt1724_set_gpio_mask(ice, ice->eeprom.gpiomask);
+ snd_vt1724_set_gpio_dir(ice, ice->eeprom.gpiodir);
+ snd_vt1724_set_gpio_data(ice, ice->eeprom.gpiostate);
+
+ outb(0, ICEREG1724(ice, POWERDOWN));
+
+ return 0;
+}
+
+static int __devinit snd_vt1724_spdif_build_controls(struct snd_ice1712 *ice)
+{
+ int err;
+ struct snd_kcontrol *kctl;
+
+ if (snd_BUG_ON(!ice->pcm))
+ return -EIO;
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_spdif_route, ice));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_spdif_switch, ice));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_default, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm->device;
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_maskc, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm->device;
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_maskp, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm->device;
+#if 0 /* use default only */
+ err = snd_ctl_add(ice->card, kctl = snd_ctl_new1(&snd_vt1724_spdif_stream, ice));
+ if (err < 0)
+ return err;
+ kctl->id.device = ice->pcm->device;
+ ice->spdif.stream_ctl = kctl;
+#endif
+ return 0;
+}
+
+
+static int __devinit snd_vt1724_build_controls(struct snd_ice1712 *ice)
+{
+ int err;
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_eeprom, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_pro_internal_clock, ice));
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_pro_rate_locking, ice));
+ if (err < 0)
+ return err;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_pro_rate_reset, ice));
+ if (err < 0)
+ return err;
+
+ if (ice->num_total_dacs > 0) {
+ struct snd_kcontrol_new tmp = snd_vt1724_mixer_pro_analog_route;
+ tmp.count = ice->num_total_dacs;
+ if (ice->vt1720 && tmp.count > 2)
+ tmp.count = 2;
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&tmp, ice));
+ if (err < 0)
+ return err;
+ }
+
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_vt1724_free(struct snd_ice1712 *ice)
+{
+ if (!ice->port)
+ goto __hw_end;
+ /* mask all interrupts */
+ outb(0xff, ICEMT1724(ice, DMA_INT_MASK));
+ outb(0xff, ICEREG1724(ice, IRQMASK));
+ /* --- */
+__hw_end:
+ if (ice->irq >= 0)
+ free_irq(ice->irq, ice);
+ pci_release_regions(ice->pci);
+ snd_ice1712_akm4xxx_free(ice);
+ pci_disable_device(ice->pci);
+ kfree(ice->spec);
+ kfree(ice);
+ return 0;
+}
+
+static int snd_vt1724_dev_free(struct snd_device *device)
+{
+ struct snd_ice1712 *ice = device->device_data;
+ return snd_vt1724_free(ice);
+}
+
+static int __devinit snd_vt1724_create(struct snd_card *card,
+ struct pci_dev *pci,
+ const char *modelname,
+ struct snd_ice1712 **r_ice1712)
+{
+ struct snd_ice1712 *ice;
+ int err;
+ unsigned char mask;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_vt1724_dev_free,
+ };
+
+ *r_ice1712 = NULL;
+
+ /* enable PCI device */
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ ice = kzalloc(sizeof(*ice), GFP_KERNEL);
+ if (ice == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ ice->vt1724 = 1;
+ spin_lock_init(&ice->reg_lock);
+ mutex_init(&ice->gpio_mutex);
+ mutex_init(&ice->open_mutex);
+ mutex_init(&ice->i2c_mutex);
+ ice->gpio.set_mask = snd_vt1724_set_gpio_mask;
+ ice->gpio.set_dir = snd_vt1724_set_gpio_dir;
+ ice->gpio.set_data = snd_vt1724_set_gpio_data;
+ ice->gpio.get_data = snd_vt1724_get_gpio_data;
+ ice->card = card;
+ ice->pci = pci;
+ ice->irq = -1;
+ pci_set_master(pci);
+ snd_vt1724_proc_init(ice);
+ synchronize_irq(pci->irq);
+
+ err = pci_request_regions(pci, "ICE1724");
+ if (err < 0) {
+ kfree(ice);
+ pci_disable_device(pci);
+ return err;
+ }
+ ice->port = pci_resource_start(pci, 0);
+ ice->profi_port = pci_resource_start(pci, 1);
+
+ if (request_irq(pci->irq, snd_vt1724_interrupt,
+ IRQF_SHARED, "ICE1724", ice)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_vt1724_free(ice);
+ return -EIO;
+ }
+
+ ice->irq = pci->irq;
+
+ snd_vt1724_chip_reset(ice);
+ if (snd_vt1724_read_eeprom(ice, modelname) < 0) {
+ snd_vt1724_free(ice);
+ return -EIO;
+ }
+ if (snd_vt1724_chip_init(ice) < 0) {
+ snd_vt1724_free(ice);
+ return -EIO;
+ }
+
+ /* unmask used interrupts */
+ mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX;
+ outb(mask, ICEREG1724(ice, IRQMASK));
+ /* don't handle FIFO overrun/underruns (just yet),
+ * since they cause machine lockups
+ */
+ outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK));
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops);
+ if (err < 0) {
+ snd_vt1724_free(ice);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_ice1712 = ice;
+ return 0;
+}
+
+
+/*
+ *
+ * Registration
+ *
+ */
+
+static int __devinit snd_vt1724_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_ice1712 *ice;
+ int pcm_dev = 0, err;
+ struct snd_ice1712_card_info * const *tbl, *c;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "ICE1724");
+ strcpy(card->shortname, "ICEnsemble ICE1724");
+
+ err = snd_vt1724_create(card, pci, model[dev], &ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ for (tbl = card_tables; *tbl; tbl++) {
+ for (c = *tbl; c->subvendor; c++) {
+ if (c->subvendor == ice->eeprom.subvendor) {
+ strcpy(card->shortname, c->name);
+ if (c->driver) /* specific driver? */
+ strcpy(card->driver, c->driver);
+ if (c->chip_init) {
+ err = c->chip_init(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+ goto __found;
+ }
+ }
+ }
+ c = &no_matched;
+__found:
+ /*
+ * VT1724 has separate DMAs for the analog and the SPDIF streams while
+ * ICE1712 has only one for both (mixed up).
+ *
+ * Confusingly the analog PCM is named "professional" here because it
+ * was called so in ice1712 driver, and vt1724 driver is derived from
+ * ice1712 driver.
+ */
+ ice->pro_rate_default = PRO_RATE_DEFAULT;
+ if (!ice->is_spdif_master)
+ ice->is_spdif_master = stdclock_is_spdif_master;
+ if (!ice->get_rate)
+ ice->get_rate = stdclock_get_rate;
+ if (!ice->set_rate)
+ ice->set_rate = stdclock_set_rate;
+ if (!ice->set_mclk)
+ ice->set_mclk = stdclock_set_mclk;
+ if (!ice->set_spdif_clock)
+ ice->set_spdif_clock = stdclock_set_spdif_clock;
+ if (!ice->hw_rates)
+ set_std_hw_rates(ice);
+
+ err = snd_vt1724_pcm_profi(ice, pcm_dev++);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ err = snd_vt1724_pcm_spdif(ice, pcm_dev++);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ err = snd_vt1724_pcm_indep(ice, pcm_dev++);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ err = snd_vt1724_ac97_mixer(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ err = snd_vt1724_build_controls(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if (ice->pcm && ice->has_spdif) { /* has SPDIF I/O */
+ err = snd_vt1724_spdif_build_controls(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+
+ if (c->build_controls) {
+ err = c->build_controls(ice);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+
+ if (!c->no_mpu401) {
+ if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) {
+ struct snd_rawmidi *rmidi;
+
+ err = snd_rawmidi_new(card, "MIDI", 0, 1, 1, &rmidi);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ ice->rmidi[0] = rmidi;
+ rmidi->private_data = ice;
+ strcpy(rmidi->name, "ICE1724 MIDI");
+ rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &vt1724_midi_output_ops);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &vt1724_midi_input_ops);
+
+ /* set watermarks */
+ outb(VT1724_MPU_RX_FIFO | 0x1,
+ ICEREG1724(ice, MPU_FIFO_WM));
+ outb(0x1, ICEREG1724(ice, MPU_FIFO_WM));
+ /* set UART mode */
+ outb(VT1724_MPU_UART, ICEREG1724(ice, MPU_CTRL));
+ }
+ }
+
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname, ice->port, ice->irq);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_vt1724_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "ICE1724",
+ .id_table = snd_vt1724_ids,
+ .probe = snd_vt1724_probe,
+ .remove = __devexit_p(snd_vt1724_remove),
+};
+
+static int __init alsa_card_ice1724_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_ice1724_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_ice1724_init)
+module_exit(alsa_card_ice1724_exit)
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
new file mode 100644
index 0000000..c51659b
--- /dev/null
+++ b/sound/pci/ice1712/juli.c
@@ -0,0 +1,687 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for ESI Juli@ cards
+ *
+ * Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz>
+ * 2008 Pavel Hofman <dustin@seznam.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/tlv.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "juli.h"
+
+struct juli_spec {
+ struct ak4114 *ak4114;
+ unsigned int analog:1;
+};
+
+/*
+ * chip addresses on I2C bus
+ */
+#define AK4114_ADDR 0x20 /* S/PDIF receiver */
+#define AK4358_ADDR 0x22 /* DAC */
+
+/*
+ * Juli does not use the standard ICE1724 clock scheme. Juli's ice1724 chip is
+ * supplied by external clock provided by Xilinx array and MK73-1 PLL frequency
+ * multiplier. Actual frequency is set by ice1724 GPIOs hooked to the Xilinx.
+ *
+ * The clock circuitry is supplied by the two ice1724 crystals. This
+ * arrangement allows to generate independent clock signal for AK4114's input
+ * rate detection circuit. As a result, Juli, unlike most other
+ * ice1724+ak4114-based cards, detects spdif input rate correctly.
+ * This fact is applied in the driver, allowing to modify PCM stream rate
+ * parameter according to the actual input rate.
+ *
+ * Juli uses the remaining three stereo-channels of its DAC to optionally
+ * monitor analog input, digital input, and digital output. The corresponding
+ * I2S signals are routed by Xilinx, controlled by GPIOs.
+ *
+ * The master mute is implemented using output muting transistors (GPIO) in
+ * combination with smuting the DAC.
+ *
+ * The card itself has no HW master volume control, implemented using the
+ * vmaster control.
+ *
+ * TODO:
+ * researching and fixing the input monitors
+ */
+
+/*
+ * GPIO pins
+ */
+#define GPIO_FREQ_MASK (3<<0)
+#define GPIO_FREQ_32KHZ (0<<0)
+#define GPIO_FREQ_44KHZ (1<<0)
+#define GPIO_FREQ_48KHZ (2<<0)
+#define GPIO_MULTI_MASK (3<<2)
+#define GPIO_MULTI_4X (0<<2)
+#define GPIO_MULTI_2X (1<<2)
+#define GPIO_MULTI_1X (2<<2) /* also external */
+#define GPIO_MULTI_HALF (3<<2)
+#define GPIO_INTERNAL_CLOCK (1<<4) /* 0 = external, 1 = internal */
+#define GPIO_CLOCK_MASK (1<<4)
+#define GPIO_ANALOG_PRESENT (1<<5) /* RO only: 0 = present */
+#define GPIO_RXMCLK_SEL (1<<7) /* must be 0 */
+#define GPIO_AK5385A_CKS0 (1<<8)
+#define GPIO_AK5385A_DFS1 (1<<9)
+#define GPIO_AK5385A_DFS0 (1<<10)
+#define GPIO_DIGOUT_MONITOR (1<<11) /* 1 = active */
+#define GPIO_DIGIN_MONITOR (1<<12) /* 1 = active */
+#define GPIO_ANAIN_MONITOR (1<<13) /* 1 = active */
+#define GPIO_AK5385A_CKS1 (1<<14) /* must be 0 */
+#define GPIO_MUTE_CONTROL (1<<15) /* output mute, 1 = muted */
+
+#define GPIO_RATE_MASK (GPIO_FREQ_MASK | GPIO_MULTI_MASK | \
+ GPIO_CLOCK_MASK)
+#define GPIO_AK5385A_MASK (GPIO_AK5385A_CKS0 | GPIO_AK5385A_DFS0 | \
+ GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS1)
+
+#define JULI_PCM_RATE (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+#define GPIO_RATE_16000 (GPIO_FREQ_32KHZ | GPIO_MULTI_HALF | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_22050 (GPIO_FREQ_44KHZ | GPIO_MULTI_HALF | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_24000 (GPIO_FREQ_48KHZ | GPIO_MULTI_HALF | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_32000 (GPIO_FREQ_32KHZ | GPIO_MULTI_1X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_44100 (GPIO_FREQ_44KHZ | GPIO_MULTI_1X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_48000 (GPIO_FREQ_48KHZ | GPIO_MULTI_1X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_64000 (GPIO_FREQ_32KHZ | GPIO_MULTI_2X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_88200 (GPIO_FREQ_44KHZ | GPIO_MULTI_2X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_96000 (GPIO_FREQ_48KHZ | GPIO_MULTI_2X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_176400 (GPIO_FREQ_44KHZ | GPIO_MULTI_4X | \
+ GPIO_INTERNAL_CLOCK)
+#define GPIO_RATE_192000 (GPIO_FREQ_48KHZ | GPIO_MULTI_4X | \
+ GPIO_INTERNAL_CLOCK)
+
+/*
+ * Initial setup of the conversion array GPIO <-> rate
+ */
+static unsigned int juli_rates[] = {
+ 16000, 22050, 24000, 32000,
+ 44100, 48000, 64000, 88200,
+ 96000, 176400, 192000,
+};
+
+static unsigned int gpio_vals[] = {
+ GPIO_RATE_16000, GPIO_RATE_22050, GPIO_RATE_24000, GPIO_RATE_32000,
+ GPIO_RATE_44100, GPIO_RATE_48000, GPIO_RATE_64000, GPIO_RATE_88200,
+ GPIO_RATE_96000, GPIO_RATE_176400, GPIO_RATE_192000,
+};
+
+static struct snd_pcm_hw_constraint_list juli_rates_info = {
+ .count = ARRAY_SIZE(juli_rates),
+ .list = juli_rates,
+ .mask = 0,
+};
+
+static int get_gpio_val(int rate)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(juli_rates); i++)
+ if (juli_rates[i] == rate)
+ return gpio_vals[i];
+ return 0;
+}
+
+static void juli_ak4114_write(void *private_data, unsigned char reg,
+ unsigned char val)
+{
+ snd_vt1724_write_i2c((struct snd_ice1712 *)private_data, AK4114_ADDR,
+ reg, val);
+}
+
+static unsigned char juli_ak4114_read(void *private_data, unsigned char reg)
+{
+ return snd_vt1724_read_i2c((struct snd_ice1712 *)private_data,
+ AK4114_ADDR, reg);
+}
+
+/*
+ * If SPDIF capture and slaved to SPDIF-IN, setting runtime rate
+ * to the external rate
+ */
+static void juli_spdif_in_open(struct snd_ice1712 *ice,
+ struct snd_pcm_substream *substream)
+{
+ struct juli_spec *spec = ice->spec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int rate;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
+ !ice->is_spdif_master(ice))
+ return;
+ rate = snd_ak4114_external_rate(spec->ak4114);
+ if (rate >= runtime->hw.rate_min && rate <= runtime->hw.rate_max) {
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+}
+
+/*
+ * AK4358 section
+ */
+
+static void juli_akm_lock(struct snd_akm4xxx *ak, int chip)
+{
+}
+
+static void juli_akm_unlock(struct snd_akm4xxx *ak, int chip)
+{
+}
+
+static void juli_akm_write(struct snd_akm4xxx *ak, int chip,
+ unsigned char addr, unsigned char data)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+
+ if (snd_BUG_ON(chip))
+ return;
+ snd_vt1724_write_i2c(ice, AK4358_ADDR, addr, data);
+}
+
+/*
+ * change the rate of envy24HT, AK4358, AK5385
+ */
+static void juli_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
+{
+ unsigned char old, tmp, ak4358_dfs;
+ unsigned int ak5385_pins, old_gpio, new_gpio;
+ struct snd_ice1712 *ice = ak->private_data[0];
+ struct juli_spec *spec = ice->spec;
+
+ if (rate == 0) /* no hint - S/PDIF input is master or the new spdif
+ input rate undetected, simply return */
+ return;
+
+ /* adjust DFS on codecs */
+ if (rate > 96000) {
+ ak4358_dfs = 2;
+ ak5385_pins = GPIO_AK5385A_DFS1 | GPIO_AK5385A_CKS0;
+ } else if (rate > 48000) {
+ ak4358_dfs = 1;
+ ak5385_pins = GPIO_AK5385A_DFS0;
+ } else {
+ ak4358_dfs = 0;
+ ak5385_pins = 0;
+ }
+ /* AK5385 first, since it requires cold reset affecting both codecs */
+ old_gpio = ice->gpio.get_data(ice);
+ new_gpio = (old_gpio & ~GPIO_AK5385A_MASK) | ak5385_pins;
+ /* printk(KERN_DEBUG "JULI - ak5385 set_rate_val: new gpio 0x%x\n",
+ new_gpio); */
+ ice->gpio.set_data(ice, new_gpio);
+
+ /* cold reset */
+ old = inb(ICEMT1724(ice, AC97_CMD));
+ outb(old | VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD));
+ udelay(1);
+ outb(old & ~VT1724_AC97_COLD, ICEMT1724(ice, AC97_CMD));
+
+ /* AK4358 */
+ /* set new value, reset DFS */
+ tmp = snd_akm4xxx_get(ak, 0, 2);
+ snd_akm4xxx_reset(ak, 1);
+ tmp = snd_akm4xxx_get(ak, 0, 2);
+ tmp &= ~(0x03 << 4);
+ tmp |= ak4358_dfs << 4;
+ snd_akm4xxx_set(ak, 0, 2, tmp);
+ snd_akm4xxx_reset(ak, 0);
+
+ /* reinit ak4114 */
+ snd_ak4114_reinit(spec->ak4114);
+}
+
+#define AK_DAC(xname, xch) { .name = xname, .num_channels = xch }
+#define PCM_VOLUME "PCM Playback Volume"
+#define MONITOR_AN_IN_VOLUME "Monitor Analog In Volume"
+#define MONITOR_DIG_IN_VOLUME "Monitor Digital In Volume"
+#define MONITOR_DIG_OUT_VOLUME "Monitor Digital Out Volume"
+
+static const struct snd_akm4xxx_dac_channel juli_dac[] = {
+ AK_DAC(PCM_VOLUME, 2),
+ AK_DAC(MONITOR_AN_IN_VOLUME, 2),
+ AK_DAC(MONITOR_DIG_OUT_VOLUME, 2),
+ AK_DAC(MONITOR_DIG_IN_VOLUME, 2),
+};
+
+
+static struct snd_akm4xxx akm_juli_dac __devinitdata = {
+ .type = SND_AK4358,
+ .num_dacs = 8, /* DAC1 - analog out
+ DAC2 - analog in monitor
+ DAC3 - digital out monitor
+ DAC4 - digital in monitor
+ */
+ .ops = {
+ .lock = juli_akm_lock,
+ .unlock = juli_akm_unlock,
+ .write = juli_akm_write,
+ .set_rate_val = juli_akm_set_rate_val
+ },
+ .dac_info = juli_dac,
+};
+
+#define juli_mute_info snd_ctl_boolean_mono_info
+
+static int juli_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ val = ice->gpio.get_data(ice) & (unsigned int) kcontrol->private_value;
+ if (kcontrol->private_value == GPIO_MUTE_CONTROL)
+ /* val 0 = signal on */
+ ucontrol->value.integer.value[0] = (val) ? 0 : 1;
+ else
+ /* val 1 = signal on */
+ ucontrol->value.integer.value[0] = (val) ? 1 : 0;
+ return 0;
+}
+
+static int juli_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int old_gpio, new_gpio;
+ old_gpio = ice->gpio.get_data(ice);
+ if (ucontrol->value.integer.value[0]) {
+ /* unmute */
+ if (kcontrol->private_value == GPIO_MUTE_CONTROL) {
+ /* 0 = signal on */
+ new_gpio = old_gpio & ~GPIO_MUTE_CONTROL;
+ /* un-smuting DAC */
+ snd_akm4xxx_write(ice->akm, 0, 0x01, 0x01);
+ } else
+ /* 1 = signal on */
+ new_gpio = old_gpio |
+ (unsigned int) kcontrol->private_value;
+ } else {
+ /* mute */
+ if (kcontrol->private_value == GPIO_MUTE_CONTROL) {
+ /* 1 = signal off */
+ new_gpio = old_gpio | GPIO_MUTE_CONTROL;
+ /* smuting DAC */
+ snd_akm4xxx_write(ice->akm, 0, 0x01, 0x03);
+ } else
+ /* 0 = signal off */
+ new_gpio = old_gpio &
+ ~((unsigned int) kcontrol->private_value);
+ }
+ /* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \
+ new_gpio 0x%x\n",
+ (unsigned int)ucontrol->value.integer.value[0], old_gpio,
+ new_gpio); */
+ if (old_gpio != new_gpio) {
+ ice->gpio.set_data(ice, new_gpio);
+ return 1;
+ }
+ /* no change */
+ return 0;
+}
+
+static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_MUTE_CONTROL,
+ },
+ /* Although the following functionality respects the succint NDA'd
+ * documentation from the card manufacturer, and the same way of
+ * operation is coded in OSS Juli driver, only Digital Out monitor
+ * seems to work. Surprisingly, Analog input monitor outputs Digital
+ * output data. The two are independent, as enabling both doubles
+ * volume of the monitor sound.
+ *
+ * Checking traces on the board suggests the functionality described
+ * by the manufacturer is correct - I2S from ADC and AK4114
+ * go to ICE as well as to Xilinx, I2S inputs of DAC2,3,4 (the monitor
+ * inputs) are fed from Xilinx.
+ *
+ * I even checked traces on board and coded a support in driver for
+ * an alternative possiblity - the unused I2S ICE output channels
+ * switched to HW-IN/SPDIF-IN and providing the monitoring signal to
+ * the DAC - to no avail. The I2S outputs seem to be unconnected.
+ *
+ * The windows driver supports the monitoring correctly.
+ */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Analog In Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_ANAIN_MONITOR,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Digital Out Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_DIGOUT_MONITOR,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Digital In Switch",
+ .info = juli_mute_info,
+ .get = juli_mute_get,
+ .put = juli_mute_put,
+ .private_value = GPIO_DIGIN_MONITOR,
+ },
+};
+
+
+static void ak4358_proc_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data;
+ int reg, val;
+ for (reg = 0; reg <= 0xf; reg++) {
+ val = snd_akm4xxx_get(ice->akm, 0, reg);
+ snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val);
+ }
+}
+
+static void ak4358_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+ if (!snd_card_proc_new(ice->card, "ak4358_codec", &entry))
+ snd_info_set_text_ops(entry, ice, ak4358_proc_regs_read);
+}
+
+static char *slave_vols[] __devinitdata = {
+ PCM_VOLUME,
+ MONITOR_AN_IN_VOLUME,
+ MONITOR_DIG_IN_VOLUME,
+ MONITOR_DIG_OUT_VOLUME,
+ NULL
+};
+
+static __devinitdata
+DECLARE_TLV_DB_SCALE(juli_master_db_scale, -6350, 50, 1);
+
+static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card,
+ const char *name)
+{
+ struct snd_ctl_elem_id sid;
+ memset(&sid, 0, sizeof(sid));
+ /* FIXME: strcpy is bad. */
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_find_id(card, &sid);
+}
+
+static void __devinit add_slaves(struct snd_card *card,
+ struct snd_kcontrol *master, char **list)
+{
+ for (; *list; list++) {
+ struct snd_kcontrol *slave = ctl_find(card, *list);
+ /* printk(KERN_DEBUG "add_slaves - %s\n", *list); */
+ if (slave) {
+ /* printk(KERN_DEBUG "slave %s found\n", *list); */
+ snd_ctl_add_slave(master, slave);
+ }
+ }
+}
+
+static int __devinit juli_add_controls(struct snd_ice1712 *ice)
+{
+ struct juli_spec *spec = ice->spec;
+ int err;
+ unsigned int i;
+ struct snd_kcontrol *vmaster;
+
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+
+ for (i = 0; i < ARRAY_SIZE(juli_mute_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&juli_mute_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+ /* Create virtual master control */
+ vmaster = snd_ctl_make_virtual_master("Master Playback Volume",
+ juli_master_db_scale);
+ if (!vmaster)
+ return -ENOMEM;
+ add_slaves(ice->card, vmaster, slave_vols);
+ err = snd_ctl_add(ice->card, vmaster);
+ if (err < 0)
+ return err;
+
+ /* only capture SPDIF over AK4114 */
+ err = snd_ak4114_build(spec->ak4114, NULL,
+ ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
+
+ ak4358_proc_init(ice);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * initialize the chip
+ */
+
+static inline int juli_is_spdif_master(struct snd_ice1712 *ice)
+{
+ return (ice->gpio.get_data(ice) & GPIO_INTERNAL_CLOCK) ? 0 : 1;
+}
+
+static unsigned int juli_get_rate(struct snd_ice1712 *ice)
+{
+ int i;
+ unsigned char result;
+
+ result = ice->gpio.get_data(ice) & GPIO_RATE_MASK;
+ for (i = 0; i < ARRAY_SIZE(gpio_vals); i++)
+ if (gpio_vals[i] == result)
+ return juli_rates[i];
+ return 0;
+}
+
+/* setting new rate */
+static void juli_set_rate(struct snd_ice1712 *ice, unsigned int rate)
+{
+ unsigned int old, new;
+ unsigned char val;
+
+ old = ice->gpio.get_data(ice);
+ new = (old & ~GPIO_RATE_MASK) | get_gpio_val(rate);
+ /* printk(KERN_DEBUG "JULI - set_rate: old %x, new %x\n",
+ old & GPIO_RATE_MASK,
+ new & GPIO_RATE_MASK); */
+
+ ice->gpio.set_data(ice, new);
+ /* switching to external clock - supplied by external circuits */
+ val = inb(ICEMT1724(ice, RATE));
+ outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE));
+}
+
+static inline unsigned char juli_set_mclk(struct snd_ice1712 *ice,
+ unsigned int rate)
+{
+ /* no change in master clock */
+ return 0;
+}
+
+/* setting clock to external - SPDIF */
+static void juli_set_spdif_clock(struct snd_ice1712 *ice)
+{
+ unsigned int old;
+ old = ice->gpio.get_data(ice);
+ /* external clock (= 0), multiply 1x, 48kHz */
+ ice->gpio.set_data(ice, (old & ~GPIO_RATE_MASK) | GPIO_MULTI_1X |
+ GPIO_FREQ_48KHZ);
+}
+
+/* Called when ak4114 detects change in the input SPDIF stream */
+static void juli_ak4114_change(struct ak4114 *ak4114, unsigned char c0,
+ unsigned char c1)
+{
+ struct snd_ice1712 *ice = ak4114->change_callback_private;
+ int rate;
+ if (ice->is_spdif_master(ice) && c1) {
+ /* only for SPDIF master mode, rate was changed */
+ rate = snd_ak4114_external_rate(ak4114);
+ /* printk(KERN_DEBUG "ak4114 - input rate changed to %d\n",
+ rate); */
+ juli_akm_set_rate_val(ice->akm, rate);
+ }
+}
+
+static int __devinit juli_init(struct snd_ice1712 *ice)
+{
+ static const unsigned char ak4114_init_vals[] = {
+ /* AK4117_REG_PWRDN */ AK4114_RST | AK4114_PWN |
+ AK4114_OCKS0 | AK4114_OCKS1,
+ /* AK4114_REQ_FORMAT */ AK4114_DIF_I24I2S,
+ /* AK4114_REG_IO0 */ AK4114_TX1E,
+ /* AK4114_REG_IO1 */ AK4114_EFH_1024 | AK4114_DIT |
+ AK4114_IPS(1),
+ /* AK4114_REG_INT0_MASK */ 0,
+ /* AK4114_REG_INT1_MASK */ 0
+ };
+ static const unsigned char ak4114_init_txcsb[] = {
+ 0x41, 0x02, 0x2c, 0x00, 0x00
+ };
+ int err;
+ struct juli_spec *spec;
+ struct snd_akm4xxx *ak;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ err = snd_ak4114_create(ice->card,
+ juli_ak4114_read,
+ juli_ak4114_write,
+ ak4114_init_vals, ak4114_init_txcsb,
+ ice, &spec->ak4114);
+ if (err < 0)
+ return err;
+ /* callback for codecs rate setting */
+ spec->ak4114->change_callback = juli_ak4114_change;
+ spec->ak4114->change_callback_private = ice;
+ /* AK4114 in Juli can detect external rate correctly */
+ spec->ak4114->check_flags = 0;
+
+#if 0
+/*
+ * it seems that the analog doughter board detection does not work reliably, so
+ * force the analog flag; it should be very rare (if ever) to come at Juli@
+ * used without the analog daughter board
+ */
+ spec->analog = (ice->gpio.get_data(ice) & GPIO_ANALOG_PRESENT) ? 0 : 1;
+#else
+ spec->analog = 1;
+#endif
+
+ if (spec->analog) {
+ printk(KERN_INFO "juli@: analog I/O detected\n");
+ ice->num_total_dacs = 2;
+ ice->num_total_adcs = 2;
+
+ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ ak = ice->akm;
+ if (!ak)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+ err = snd_ice1712_akm4xxx_init(ak, &akm_juli_dac, NULL, ice);
+ if (err < 0)
+ return err;
+ }
+
+ /* juli is clocked by Xilinx array */
+ ice->hw_rates = &juli_rates_info;
+ ice->is_spdif_master = juli_is_spdif_master;
+ ice->get_rate = juli_get_rate;
+ ice->set_rate = juli_set_rate;
+ ice->set_mclk = juli_set_mclk;
+ ice->set_spdif_clock = juli_set_spdif_clock;
+
+ ice->spdif.ops.open = juli_spdif_in_open;
+ return 0;
+}
+
+
+/*
+ * Juli@ boards don't provide the EEPROM data except for the vendor IDs.
+ * hence the driver needs to sets up it properly.
+ */
+
+static unsigned char juli_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401, 1xADC, 1xDACs,
+ SPDIF in */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0x9f, /* 5, 6:inputs; 7, 4-0 outputs*/
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x7f,
+ [ICE_EEP2_GPIO_MASK] = 0x60, /* 5, 6: locked; 7, 4-0 writable */
+ [ICE_EEP2_GPIO_MASK1] = 0x00, /* 0-7 writable */
+ [ICE_EEP2_GPIO_MASK2] = 0x7f,
+ [ICE_EEP2_GPIO_STATE] = GPIO_FREQ_48KHZ | GPIO_MULTI_1X |
+ GPIO_INTERNAL_CLOCK, /* internal clock, multiple 1x, 48kHz*/
+ [ICE_EEP2_GPIO_STATE1] = 0x00, /* unmuted */
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1724_juli_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_JULI,
+ .name = "ESI Juli@",
+ .model = "juli",
+ .chip_init = juli_init,
+ .build_controls = juli_add_controls,
+ .eeprom_size = sizeof(juli_eeprom),
+ .eeprom_data = juli_eeprom,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/juli.h b/sound/pci/ice1712/juli.h
new file mode 100644
index 0000000..d9f8534
--- /dev/null
+++ b/sound/pci/ice1712/juli.h
@@ -0,0 +1,10 @@
+#ifndef __SOUND_JULI_H
+#define __SOUND_JULI_H
+
+#define JULI_DEVICE_DESC "{ESI,Juli@},"
+
+#define VT1724_SUBDEVICE_JULI 0x31305345 /* Juli@ */
+
+extern struct snd_ice1712_card_info snd_vt1724_juli_cards[];
+
+#endif /* __SOUND_JULI_H */
diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c
new file mode 100644
index 0000000..de29be8
--- /dev/null
+++ b/sound/pci/ice1712/phase.c
@@ -0,0 +1,975 @@
+/*
+ * ALSA driver for ICEnsemble ICE1724 (Envy24)
+ *
+ * Lowlevel functions for Terratec PHASE 22
+ *
+ * Copyright (c) 2005 Misha Zhilin <misha@epiphan.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* PHASE 22 overview:
+ * Audio controller: VIA Envy24HT-S (slightly trimmed down Envy24HT, 4in/4out)
+ * Analog chip: AK4524 (partially via Philip's 74HCT125)
+ * Digital receiver: CS8414-CS (supported in this release)
+ * PHASE 22 revision 2.0 and Terrasoniq/Musonik TS22PCI have CS8416
+ * (support status unknown, please test and report)
+ *
+ * Envy connects to AK4524
+ * - CS directly from GPIO 10
+ * - CCLK via 74HCT125's gate #4 from GPIO 4
+ * - CDTI via 74HCT125's gate #2 from GPIO 5
+ * CDTI may be completely blocked by 74HCT125's gate #1
+ * controlled by GPIO 3
+ */
+
+/* PHASE 28 overview:
+ * Audio controller: VIA Envy24HT (full untrimmed version, 4in/8out)
+ * Analog chip: WM8770 (8 channel 192k DAC, 2 channel 96k ADC)
+ * Digital receiver: CS8414-CS (supported in this release)
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "phase.h"
+#include <sound/tlv.h>
+
+/* AC97 register cache for Phase28 */
+struct phase28_spec {
+ unsigned short master[2];
+ unsigned short vol[8];
+};
+
+/* WM8770 registers */
+#define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */
+#define WM_DAC_MASTER_ATTEN 0x08 /* DAC master analog attenuation */
+#define WM_DAC_DIG_ATTEN 0x09 /* DAC1-8 digital attenuation */
+#define WM_DAC_DIG_MASTER_ATTEN 0x11 /* DAC master digital attenuation */
+#define WM_PHASE_SWAP 0x12 /* DAC phase */
+#define WM_DAC_CTRL1 0x13 /* DAC control bits */
+#define WM_MUTE 0x14 /* mute controls */
+#define WM_DAC_CTRL2 0x15 /* de-emphasis and zefo-flag */
+#define WM_INT_CTRL 0x16 /* interface control */
+#define WM_MASTER 0x17 /* master clock and mode */
+#define WM_POWERDOWN 0x18 /* power-down controls */
+#define WM_ADC_GAIN 0x19 /* ADC gain L(19)/R(1a) */
+#define WM_ADC_MUX 0x1b /* input MUX */
+#define WM_OUT_MUX1 0x1c /* output MUX */
+#define WM_OUT_MUX2 0x1e /* output MUX */
+#define WM_RESET 0x1f /* software reset */
+
+
+/*
+ * Logarithmic volume values for WM8770
+ * Computed as 20 * Log10(255 / x)
+ */
+static const unsigned char wm_vol[256] = {
+ 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24,
+ 24, 23, 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18,
+ 17, 17, 17, 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14,
+ 14, 13, 13, 13, 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11,
+ 11, 11, 11, 11, 11, 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9,
+ 9, 9, 9, 9, 9, 9, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7,
+ 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5,
+ 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
+ 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
+ 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+ 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
+};
+
+#define WM_VOL_MAX (sizeof(wm_vol) - 1)
+#define WM_VOL_MUTE 0x8000
+
+static struct snd_akm4xxx akm_phase22 __devinitdata = {
+ .type = SND_AK4524,
+ .num_dacs = 2,
+ .num_adcs = 2,
+};
+
+static struct snd_ak4xxx_private akm_phase22_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 1,
+ .data_mask = 1 << 4,
+ .clk_mask = 1 << 5,
+ .cs_mask = 1 << 10,
+ .cs_addr = 1 << 10,
+ .cs_none = 0,
+ .add_flags = 1 << 3,
+ .mask_flags = 0,
+};
+
+static int __devinit phase22_init(struct snd_ice1712 *ice)
+{
+ struct snd_akm4xxx *ak;
+ int err;
+
+ /* Configure DAC/ADC description for generic part of ice1724 */
+ switch (ice->eeprom.subvendor) {
+ case VT1724_SUBDEVICE_PHASE22:
+ case VT1724_SUBDEVICE_TS22:
+ ice->num_total_dacs = 2;
+ ice->num_total_adcs = 2;
+ ice->vt1720 = 1; /* Envy24HT-S have 16 bit wide GPIO */
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+
+ /* Initialize analog chips */
+ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ ak = ice->akm;
+ if (!ak)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+ switch (ice->eeprom.subvendor) {
+ case VT1724_SUBDEVICE_PHASE22:
+ case VT1724_SUBDEVICE_TS22:
+ err = snd_ice1712_akm4xxx_init(ak, &akm_phase22,
+ &akm_phase22_priv, ice);
+ if (err < 0)
+ return err;
+ break;
+ }
+
+ return 0;
+}
+
+static int __devinit phase22_add_controls(struct snd_ice1712 *ice)
+{
+ int err = 0;
+
+ switch (ice->eeprom.subvendor) {
+ case VT1724_SUBDEVICE_PHASE22:
+ case VT1724_SUBDEVICE_TS22:
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static unsigned char phase22_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x28, /* clock 512, mpu 401,
+ spdif-in/1xADC, 1xDACs */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xf0, /* vol, 96k, 24bit */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0xff,
+ [ICE_EEP2_GPIO_MASK] = 0x00,
+ [ICE_EEP2_GPIO_MASK1] = 0x00,
+ [ICE_EEP2_GPIO_MASK2] = 0x00,
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+
+static unsigned char phase28_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401,
+ spdif-in/1xADC, 4xDACs */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xfc, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x5f,
+ [ICE_EEP2_GPIO_MASK] = 0x00,
+ [ICE_EEP2_GPIO_MASK1] = 0x00,
+ [ICE_EEP2_GPIO_MASK2] = 0x00,
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+
+/*
+ * write data in the SPI mode
+ */
+static void phase28_spi_write(struct snd_ice1712 *ice, unsigned int cs,
+ unsigned int data, int bits)
+{
+ unsigned int tmp;
+ int i;
+
+ tmp = snd_ice1712_gpio_read(ice);
+
+ snd_ice1712_gpio_set_mask(ice, ~(PHASE28_WM_RW|PHASE28_SPI_MOSI|
+ PHASE28_SPI_CLK|PHASE28_WM_CS));
+ tmp |= PHASE28_WM_RW;
+ tmp &= ~cs;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ for (i = bits - 1; i >= 0; i--) {
+ tmp &= ~PHASE28_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ if (data & (1 << i))
+ tmp |= PHASE28_SPI_MOSI;
+ else
+ tmp &= ~PHASE28_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+
+ tmp &= ~PHASE28_SPI_CLK;
+ tmp |= cs;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+}
+
+/*
+ * get the current register value of WM codec
+ */
+static unsigned short wm_get(struct snd_ice1712 *ice, int reg)
+{
+ reg <<= 1;
+ return ((unsigned short)ice->akm[0].images[reg] << 8) |
+ ice->akm[0].images[reg + 1];
+}
+
+/*
+ * set the register value of WM codec
+ */
+static void wm_put_nocache(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ phase28_spi_write(ice, PHASE28_WM_CS, (reg << 9) | (val & 0x1ff), 16);
+}
+
+/*
+ * set the register value of WM codec and remember it
+ */
+static void wm_put(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ wm_put_nocache(ice, reg, val);
+ reg <<= 1;
+ ice->akm[0].images[reg] = val >> 8;
+ ice->akm[0].images[reg + 1] = val;
+}
+
+static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index,
+ unsigned short vol, unsigned short master)
+{
+ unsigned char nvol;
+
+ if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+ nvol = 0;
+ else
+ nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) *
+ (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX];
+
+ wm_put(ice, index, nvol);
+ wm_put_nocache(ice, index, 0x180 | nvol);
+}
+
+/*
+ * DAC mute control
+ */
+#define wm_pcm_mute_info snd_ctl_boolean_mono_info
+
+static int wm_pcm_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_MUTE) & 0x10) ?
+ 0 : 1;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_pcm_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short nval, oval;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+ oval = wm_get(ice, WM_MUTE);
+ nval = (oval & ~0x10) | (ucontrol->value.integer.value[0] ? 0 : 0x10);
+ change = (nval != oval);
+ if (change)
+ wm_put(ice, WM_MUTE, nval);
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * Master volume attenuation mixer control
+ */
+static int wm_master_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = WM_VOL_MAX;
+ return 0;
+}
+
+static int wm_master_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+ int i;
+ for (i = 0; i < 2; i++)
+ ucontrol->value.integer.value[i] = spec->master[i] &
+ ~WM_VOL_MUTE;
+ return 0;
+}
+
+static int wm_master_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+ int ch, change = 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (ch = 0; ch < 2; ch++) {
+ unsigned int vol = ucontrol->value.integer.value[ch];
+ if (vol > WM_VOL_MAX)
+ continue;
+ vol |= spec->master[ch] & WM_VOL_MUTE;
+ if (vol != spec->master[ch]) {
+ int dac;
+ spec->master[ch] = vol;
+ for (dac = 0; dac < ice->num_total_dacs; dac += 2)
+ wm_set_vol(ice, WM_DAC_ATTEN + dac + ch,
+ spec->vol[dac + ch],
+ spec->master[ch]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+static int __devinit phase28_init(struct snd_ice1712 *ice)
+{
+ static const unsigned short wm_inits_phase28[] = {
+ /* These come first to reduce init pop noise */
+ 0x1b, 0x044, /* ADC Mux (AC'97 source) */
+ 0x1c, 0x00B, /* Out Mux1 (VOUT1 = DAC+AUX, VOUT2 = DAC) */
+ 0x1d, 0x009, /* Out Mux2 (VOUT2 = DAC, VOUT3 = DAC) */
+
+ 0x18, 0x000, /* All power-up */
+
+ 0x16, 0x122, /* I2S, normal polarity, 24bit */
+ 0x17, 0x022, /* 256fs, slave mode */
+ 0x00, 0, /* DAC1 analog mute */
+ 0x01, 0, /* DAC2 analog mute */
+ 0x02, 0, /* DAC3 analog mute */
+ 0x03, 0, /* DAC4 analog mute */
+ 0x04, 0, /* DAC5 analog mute */
+ 0x05, 0, /* DAC6 analog mute */
+ 0x06, 0, /* DAC7 analog mute */
+ 0x07, 0, /* DAC8 analog mute */
+ 0x08, 0x100, /* master analog mute */
+ 0x09, 0xff, /* DAC1 digital full */
+ 0x0a, 0xff, /* DAC2 digital full */
+ 0x0b, 0xff, /* DAC3 digital full */
+ 0x0c, 0xff, /* DAC4 digital full */
+ 0x0d, 0xff, /* DAC5 digital full */
+ 0x0e, 0xff, /* DAC6 digital full */
+ 0x0f, 0xff, /* DAC7 digital full */
+ 0x10, 0xff, /* DAC8 digital full */
+ 0x11, 0x1ff, /* master digital full */
+ 0x12, 0x000, /* phase normal */
+ 0x13, 0x090, /* unmute DAC L/R */
+ 0x14, 0x000, /* all unmute */
+ 0x15, 0x000, /* no deemphasis, no ZFLG */
+ 0x19, 0x000, /* -12dB ADC/L */
+ 0x1a, 0x000, /* -12dB ADC/R */
+ (unsigned short)-1
+ };
+
+ unsigned int tmp;
+ struct snd_akm4xxx *ak;
+ struct phase28_spec *spec;
+ const unsigned short *p;
+ int i;
+
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 2;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ /* Initialize analog chips */
+ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ ak = ice->akm;
+ if (!ak)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ snd_ice1712_gpio_set_dir(ice, 0x5fffff); /* fix this for time being */
+
+ /* reset the wm codec as the SPI mode */
+ snd_ice1712_save_gpio_status(ice);
+ snd_ice1712_gpio_set_mask(ice, ~(PHASE28_WM_RESET|PHASE28_WM_CS|
+ PHASE28_HP_SEL));
+
+ tmp = snd_ice1712_gpio_read(ice);
+ tmp &= ~PHASE28_WM_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_WM_CS;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_WM_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ p = wm_inits_phase28;
+ for (; *p != (unsigned short)-1; p += 2)
+ wm_put(ice, p[0], p[1]);
+
+ snd_ice1712_restore_gpio_status(ice);
+
+ spec->master[0] = WM_VOL_MUTE;
+ spec->master[1] = WM_VOL_MUTE;
+ for (i = 0; i < ice->num_total_dacs; i++) {
+ spec->vol[i] = WM_VOL_MUTE;
+ wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]);
+ }
+
+ return 0;
+}
+
+/*
+ * DAC volume attenuation mixer control
+ */
+static int wm_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int voices = kcontrol->private_value >> 8;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = voices;
+ uinfo->value.integer.min = 0; /* mute (-101dB) */
+ uinfo->value.integer.max = 0x7F; /* 0dB */
+ return 0;
+}
+
+static int wm_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+ int i, ofs, voices;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ for (i = 0; i < voices; i++)
+ ucontrol->value.integer.value[i] =
+ spec->vol[ofs+i] & ~WM_VOL_MUTE;
+ return 0;
+}
+
+static int wm_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+ int i, idx, ofs, voices;
+ int change = 0;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < voices; i++) {
+ unsigned int vol;
+ vol = ucontrol->value.integer.value[i];
+ if (vol > 0x7f)
+ continue;
+ vol |= spec->vol[ofs+i] & WM_VOL_MUTE;
+ if (vol != spec->vol[ofs+i]) {
+ spec->vol[ofs+i] = vol;
+ idx = WM_DAC_ATTEN + ofs + i;
+ wm_set_vol(ice, idx, spec->vol[ofs+i],
+ spec->master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * WM8770 mute control
+ */
+static int wm_mute_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo) {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = kcontrol->private_value >> 8;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int wm_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+ int voices, ofs, i;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xFF;
+
+ for (i = 0; i < voices; i++)
+ ucontrol->value.integer.value[i] =
+ (spec->vol[ofs+i] & WM_VOL_MUTE) ? 0 : 1;
+ return 0;
+}
+
+static int wm_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+ int change = 0, voices, ofs, i;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xFF;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < voices; i++) {
+ int val = (spec->vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1;
+ if (ucontrol->value.integer.value[i] != val) {
+ spec->vol[ofs + i] &= ~WM_VOL_MUTE;
+ spec->vol[ofs + i] |=
+ ucontrol->value.integer.value[i] ? 0 :
+ WM_VOL_MUTE;
+ wm_set_vol(ice, ofs + i, spec->vol[ofs + i],
+ spec->master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * WM8770 master mute control
+ */
+#define wm_master_mute_info snd_ctl_boolean_stereo_info
+
+static int wm_master_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+
+ ucontrol->value.integer.value[0] =
+ (spec->master[0] & WM_VOL_MUTE) ? 0 : 1;
+ ucontrol->value.integer.value[1] =
+ (spec->master[1] & WM_VOL_MUTE) ? 0 : 1;
+ return 0;
+}
+
+static int wm_master_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct phase28_spec *spec = ice->spec;
+ int change = 0, i;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < 2; i++) {
+ int val = (spec->master[i] & WM_VOL_MUTE) ? 0 : 1;
+ if (ucontrol->value.integer.value[i] != val) {
+ int dac;
+ spec->master[i] &= ~WM_VOL_MUTE;
+ spec->master[i] |=
+ ucontrol->value.integer.value[i] ? 0 :
+ WM_VOL_MUTE;
+ for (dac = 0; dac < ice->num_total_dacs; dac += 2)
+ wm_set_vol(ice, WM_DAC_ATTEN + dac + i,
+ spec->vol[dac + i],
+ spec->master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/* digital master volume */
+#define PCM_0dB 0xff
+#define PCM_RES 128 /* -64dB */
+#define PCM_MIN (PCM_0dB - PCM_RES)
+static int wm_pcm_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0; /* mute (-64dB) */
+ uinfo->value.integer.max = PCM_RES; /* 0dB */
+ return 0;
+}
+
+static int wm_pcm_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ mutex_lock(&ice->gpio_mutex);
+ val = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff;
+ val = val > PCM_MIN ? (val - PCM_MIN) : 0;
+ ucontrol->value.integer.value[0] = val;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_pcm_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int change = 0;
+
+ nvol = ucontrol->value.integer.value[0];
+ if (nvol > PCM_RES)
+ return -EINVAL;
+ snd_ice1712_save_gpio_status(ice);
+ nvol = (nvol ? (nvol + PCM_MIN) : 0) & 0xff;
+ ovol = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff;
+ if (ovol != nvol) {
+ wm_put(ice, WM_DAC_DIG_MASTER_ATTEN, nvol); /* prelatch */
+ /* update */
+ wm_put_nocache(ice, WM_DAC_DIG_MASTER_ATTEN, nvol | 0x100);
+ change = 1;
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * Deemphasis
+ */
+#define phase28_deemp_info snd_ctl_boolean_mono_info
+
+static int phase28_deemp_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL2) & 0xf) ==
+ 0xf;
+ return 0;
+}
+
+static int phase28_deemp_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int temp, temp2;
+ temp = wm_get(ice, WM_DAC_CTRL2);
+ temp2 = temp;
+ if (ucontrol->value.integer.value[0])
+ temp |= 0xf;
+ else
+ temp &= ~0xf;
+ if (temp != temp2) {
+ wm_put(ice, WM_DAC_CTRL2, temp);
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * ADC Oversampling
+ */
+static int phase28_oversampling_info(struct snd_kcontrol *k,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "128x", "64x" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items -
+ 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int phase28_oversampling_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = (wm_get(ice, WM_MASTER) & 0x8) ==
+ 0x8;
+ return 0;
+}
+
+static int phase28_oversampling_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int temp, temp2;
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ temp = wm_get(ice, WM_MASTER);
+ temp2 = temp;
+
+ if (ucontrol->value.enumerated.item[0])
+ temp |= 0x8;
+ else
+ temp &= ~0x8;
+
+ if (temp != temp2) {
+ wm_put(ice, WM_MASTER, temp);
+ return 1;
+ }
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1);
+
+static struct snd_kcontrol_new phase28_dac_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = wm_master_mute_info,
+ .get = wm_master_mute_get,
+ .put = wm_master_mute_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Master Playback Volume",
+ .info = wm_master_vol_info,
+ .get = wm_master_vol_get,
+ .put = wm_master_vol_put,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Front Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 0
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Front Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 0,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Rear Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 2
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Rear Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 2,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Center Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (1 << 8) | 4
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Center Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (1 << 8) | 4,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "LFE Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (1 << 8) | 5
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "LFE Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (1 << 8) | 5,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Side Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 6
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Side Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 6,
+ .tlv = { .p = db_scale_wm_dac }
+ }
+};
+
+static struct snd_kcontrol_new wm_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .info = wm_pcm_mute_info,
+ .get = wm_pcm_mute_get,
+ .put = wm_pcm_mute_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "PCM Playback Volume",
+ .info = wm_pcm_vol_info,
+ .get = wm_pcm_vol_get,
+ .put = wm_pcm_vol_put,
+ .tlv = { .p = db_scale_wm_pcm }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Deemphasis Switch",
+ .info = phase28_deemp_info,
+ .get = phase28_deemp_get,
+ .put = phase28_deemp_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Oversampling",
+ .info = phase28_oversampling_info,
+ .get = phase28_oversampling_get,
+ .put = phase28_oversampling_put
+ }
+};
+
+static int __devinit phase28_add_controls(struct snd_ice1712 *ice)
+{
+ unsigned int i, counts;
+ int err;
+
+ counts = ARRAY_SIZE(phase28_dac_controls);
+ for (i = 0; i < counts; i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&phase28_dac_controls[i],
+ ice));
+ if (err < 0)
+ return err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&wm_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+struct snd_ice1712_card_info snd_vt1724_phase_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_PHASE22,
+ .name = "Terratec PHASE 22",
+ .model = "phase22",
+ .chip_init = phase22_init,
+ .build_controls = phase22_add_controls,
+ .eeprom_size = sizeof(phase22_eeprom),
+ .eeprom_data = phase22_eeprom,
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_PHASE28,
+ .name = "Terratec PHASE 28",
+ .model = "phase28",
+ .chip_init = phase28_init,
+ .build_controls = phase28_add_controls,
+ .eeprom_size = sizeof(phase28_eeprom),
+ .eeprom_data = phase28_eeprom,
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_TS22,
+ .name = "Terrasoniq TS22 PCI",
+ .model = "TS22",
+ .chip_init = phase22_init,
+ .build_controls = phase22_add_controls,
+ .eeprom_size = sizeof(phase22_eeprom),
+ .eeprom_data = phase22_eeprom,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/phase.h b/sound/pci/ice1712/phase.h
new file mode 100644
index 0000000..7fc22d9
--- /dev/null
+++ b/sound/pci/ice1712/phase.h
@@ -0,0 +1,53 @@
+#ifndef __SOUND_PHASE_H
+#define __SOUND_PHASE_H
+
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for Terratec PHASE 22
+ *
+ * Copyright (c) 2005 Misha Zhilin <misha@epiphan.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define PHASE_DEVICE_DESC "{Terratec,Phase 22},"\
+ "{Terratec,Phase 28},"\
+ "{Terrasoniq,TS22},"
+
+#define VT1724_SUBDEVICE_PHASE22 0x3b155011
+#define VT1724_SUBDEVICE_PHASE28 0x3b154911
+#define VT1724_SUBDEVICE_TS22 0x3b157b11
+
+/* entry point */
+extern struct snd_ice1712_card_info snd_vt1724_phase_cards[];
+
+/* PHASE28 GPIO bits */
+#define PHASE28_SPI_MISO (1 << 21)
+#define PHASE28_WM_RESET (1 << 20)
+#define PHASE28_SPI_CLK (1 << 19)
+#define PHASE28_SPI_MOSI (1 << 18)
+#define PHASE28_WM_RW (1 << 17)
+#define PHASE28_AC97_RESET (1 << 16)
+#define PHASE28_DIGITAL_SEL1 (1 << 15)
+#define PHASE28_HP_SEL (1 << 14)
+#define PHASE28_WM_CS (1 << 12)
+#define PHASE28_AC97_COMMIT (1 << 11)
+#define PHASE28_AC97_ADDR (1 << 10)
+#define PHASE28_AC97_DATA_LOW (1 << 9)
+#define PHASE28_AC97_DATA_HIGH (1 << 8)
+#define PHASE28_AC97_DATA_MASK 0xFF
+#endif /* __SOUND_PHASE */
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
new file mode 100644
index 0000000..6bc3f91
--- /dev/null
+++ b/sound/pci/ice1712/pontis.c
@@ -0,0 +1,836 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Pontis MS300
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "pontis.h"
+
+/* I2C addresses */
+#define WM_DEV 0x34
+#define CS_DEV 0x20
+
+/* WM8776 registers */
+#define WM_HP_ATTEN_L 0x00 /* headphone left attenuation */
+#define WM_HP_ATTEN_R 0x01 /* headphone left attenuation */
+#define WM_HP_MASTER 0x02 /* headphone master (both channels) */
+ /* override LLR */
+#define WM_DAC_ATTEN_L 0x03 /* digital left attenuation */
+#define WM_DAC_ATTEN_R 0x04
+#define WM_DAC_MASTER 0x05
+#define WM_PHASE_SWAP 0x06 /* DAC phase swap */
+#define WM_DAC_CTRL1 0x07
+#define WM_DAC_MUTE 0x08
+#define WM_DAC_CTRL2 0x09
+#define WM_DAC_INT 0x0a
+#define WM_ADC_INT 0x0b
+#define WM_MASTER_CTRL 0x0c
+#define WM_POWERDOWN 0x0d
+#define WM_ADC_ATTEN_L 0x0e
+#define WM_ADC_ATTEN_R 0x0f
+#define WM_ALC_CTRL1 0x10
+#define WM_ALC_CTRL2 0x11
+#define WM_ALC_CTRL3 0x12
+#define WM_NOISE_GATE 0x13
+#define WM_LIMITER 0x14
+#define WM_ADC_MUX 0x15
+#define WM_OUT_MUX 0x16
+#define WM_RESET 0x17
+
+/*
+ * GPIO
+ */
+#define PONTIS_CS_CS (1<<4) /* CS */
+#define PONTIS_CS_CLK (1<<5) /* CLK */
+#define PONTIS_CS_RDATA (1<<6) /* CS8416 -> VT1720 */
+#define PONTIS_CS_WDATA (1<<7) /* VT1720 -> CS8416 */
+
+
+/*
+ * get the current register value of WM codec
+ */
+static unsigned short wm_get(struct snd_ice1712 *ice, int reg)
+{
+ reg <<= 1;
+ return ((unsigned short)ice->akm[0].images[reg] << 8) |
+ ice->akm[0].images[reg + 1];
+}
+
+/*
+ * set the register value of WM codec and remember it
+ */
+static void wm_put_nocache(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ unsigned short cval;
+ cval = (reg << 9) | val;
+ snd_vt1724_write_i2c(ice, WM_DEV, cval >> 8, cval & 0xff);
+}
+
+static void wm_put(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ wm_put_nocache(ice, reg, val);
+ reg <<= 1;
+ ice->akm[0].images[reg] = val >> 8;
+ ice->akm[0].images[reg + 1] = val;
+}
+
+/*
+ * DAC volume attenuation mixer control (-64dB to 0dB)
+ */
+
+#define DAC_0dB 0xff
+#define DAC_RES 128
+#define DAC_MIN (DAC_0dB - DAC_RES)
+
+static int wm_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* mute */
+ uinfo->value.integer.max = DAC_RES; /* 0dB, 0.5dB step */
+ return 0;
+}
+
+static int wm_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+ int i;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ val = wm_get(ice, WM_DAC_ATTEN_L + i) & 0xff;
+ val = val > DAC_MIN ? (val - DAC_MIN) : 0;
+ ucontrol->value.integer.value[i] = val;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short oval, nval;
+ int i, idx, change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ nval = ucontrol->value.integer.value[i];
+ nval = (nval ? (nval + DAC_MIN) : 0) & 0xff;
+ idx = WM_DAC_ATTEN_L + i;
+ oval = wm_get(ice, idx) & 0xff;
+ if (oval != nval) {
+ wm_put(ice, idx, nval);
+ wm_put_nocache(ice, idx, nval | 0x100);
+ change = 1;
+ }
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * ADC gain mixer control (-64dB to 0dB)
+ */
+
+#define ADC_0dB 0xcf
+#define ADC_RES 128
+#define ADC_MIN (ADC_0dB - ADC_RES)
+
+static int wm_adc_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* mute (-64dB) */
+ uinfo->value.integer.max = ADC_RES; /* 0dB, 0.5dB step */
+ return 0;
+}
+
+static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+ int i;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ val = wm_get(ice, WM_ADC_ATTEN_L + i) & 0xff;
+ val = val > ADC_MIN ? (val - ADC_MIN) : 0;
+ ucontrol->value.integer.value[i] = val;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int i, idx, change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ nvol = ucontrol->value.integer.value[i];
+ nvol = nvol ? (nvol + ADC_MIN) : 0;
+ idx = WM_ADC_ATTEN_L + i;
+ ovol = wm_get(ice, idx) & 0xff;
+ if (ovol != nvol) {
+ wm_put(ice, idx, nvol);
+ change = 1;
+ }
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * ADC input mux mixer control
+ */
+#define wm_adc_mux_info snd_ctl_boolean_mono_info
+
+static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int bit = kcontrol->private_value;
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_ADC_MUX) & (1 << bit)) ? 1 : 0;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int bit = kcontrol->private_value;
+ unsigned short oval, nval;
+ int change;
+
+ mutex_lock(&ice->gpio_mutex);
+ nval = oval = wm_get(ice, WM_ADC_MUX);
+ if (ucontrol->value.integer.value[0])
+ nval |= (1 << bit);
+ else
+ nval &= ~(1 << bit);
+ change = nval != oval;
+ if (change) {
+ wm_put(ice, WM_ADC_MUX, nval);
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * Analog bypass (In -> Out)
+ */
+#define wm_bypass_info snd_ctl_boolean_mono_info
+
+static int wm_bypass_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_OUT_MUX) & 0x04) ? 1 : 0;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_bypass_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val, oval;
+ int change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ val = oval = wm_get(ice, WM_OUT_MUX);
+ if (ucontrol->value.integer.value[0])
+ val |= 0x04;
+ else
+ val &= ~0x04;
+ if (val != oval) {
+ wm_put(ice, WM_OUT_MUX, val);
+ change = 1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * Left/Right swap
+ */
+#define wm_chswap_info snd_ctl_boolean_mono_info
+
+static int wm_chswap_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL1) & 0xf0) != 0x90;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_chswap_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val, oval;
+ int change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ oval = wm_get(ice, WM_DAC_CTRL1);
+ val = oval & 0x0f;
+ if (ucontrol->value.integer.value[0])
+ val |= 0x60;
+ else
+ val |= 0x90;
+ if (val != oval) {
+ wm_put(ice, WM_DAC_CTRL1, val);
+ wm_put_nocache(ice, WM_DAC_CTRL1, val);
+ change = 1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * write data in the SPI mode
+ */
+static void set_gpio_bit(struct snd_ice1712 *ice, unsigned int bit, int val)
+{
+ unsigned int tmp = snd_ice1712_gpio_read(ice);
+ if (val)
+ tmp |= bit;
+ else
+ tmp &= ~bit;
+ snd_ice1712_gpio_write(ice, tmp);
+}
+
+static void spi_send_byte(struct snd_ice1712 *ice, unsigned char data)
+{
+ int i;
+ for (i = 0; i < 8; i++) {
+ set_gpio_bit(ice, PONTIS_CS_CLK, 0);
+ udelay(1);
+ set_gpio_bit(ice, PONTIS_CS_WDATA, data & 0x80);
+ udelay(1);
+ set_gpio_bit(ice, PONTIS_CS_CLK, 1);
+ udelay(1);
+ data <<= 1;
+ }
+}
+
+static unsigned int spi_read_byte(struct snd_ice1712 *ice)
+{
+ int i;
+ unsigned int val = 0;
+
+ for (i = 0; i < 8; i++) {
+ val <<= 1;
+ set_gpio_bit(ice, PONTIS_CS_CLK, 0);
+ udelay(1);
+ if (snd_ice1712_gpio_read(ice) & PONTIS_CS_RDATA)
+ val |= 1;
+ udelay(1);
+ set_gpio_bit(ice, PONTIS_CS_CLK, 1);
+ udelay(1);
+ }
+ return val;
+}
+
+
+static void spi_write(struct snd_ice1712 *ice, unsigned int dev, unsigned int reg, unsigned int data)
+{
+ snd_ice1712_gpio_set_dir(ice, PONTIS_CS_CS|PONTIS_CS_WDATA|PONTIS_CS_CLK);
+ snd_ice1712_gpio_set_mask(ice, ~(PONTIS_CS_CS|PONTIS_CS_WDATA|PONTIS_CS_CLK));
+ set_gpio_bit(ice, PONTIS_CS_CS, 0);
+ spi_send_byte(ice, dev & ~1); /* WRITE */
+ spi_send_byte(ice, reg); /* MAP */
+ spi_send_byte(ice, data); /* DATA */
+ /* trigger */
+ set_gpio_bit(ice, PONTIS_CS_CS, 1);
+ udelay(1);
+ /* restore */
+ snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask);
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+}
+
+static unsigned int spi_read(struct snd_ice1712 *ice, unsigned int dev, unsigned int reg)
+{
+ unsigned int val;
+ snd_ice1712_gpio_set_dir(ice, PONTIS_CS_CS|PONTIS_CS_WDATA|PONTIS_CS_CLK);
+ snd_ice1712_gpio_set_mask(ice, ~(PONTIS_CS_CS|PONTIS_CS_WDATA|PONTIS_CS_CLK));
+ set_gpio_bit(ice, PONTIS_CS_CS, 0);
+ spi_send_byte(ice, dev & ~1); /* WRITE */
+ spi_send_byte(ice, reg); /* MAP */
+ /* trigger */
+ set_gpio_bit(ice, PONTIS_CS_CS, 1);
+ udelay(1);
+ set_gpio_bit(ice, PONTIS_CS_CS, 0);
+ spi_send_byte(ice, dev | 1); /* READ */
+ val = spi_read_byte(ice);
+ /* trigger */
+ set_gpio_bit(ice, PONTIS_CS_CS, 1);
+ udelay(1);
+ /* restore */
+ snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask);
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+ return val;
+}
+
+
+/*
+ * SPDIF input source
+ */
+static int cs_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "Coax", /* RXP0 */
+ "Optical", /* RXP1 */
+ "CD", /* RXP2 */
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int cs_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.enumerated.item[0] = ice->gpio.saved[0];
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int cs_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ if (ucontrol->value.enumerated.item[0] != ice->gpio.saved[0]) {
+ ice->gpio.saved[0] = ucontrol->value.enumerated.item[0] & 3;
+ val = 0x80 | (ice->gpio.saved[0] << 3);
+ spi_write(ice, CS_DEV, 0x04, val);
+ change = 1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+
+/*
+ * GPIO controls
+ */
+static int pontis_gpio_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xffff; /* 16bit */
+ return 0;
+}
+
+static int pontis_gpio_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&ice->gpio_mutex);
+ /* 4-7 reserved */
+ ucontrol->value.integer.value[0] = (~ice->gpio.write_mask & 0xffff) | 0x00f0;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int pontis_gpio_mask_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int changed;
+ mutex_lock(&ice->gpio_mutex);
+ /* 4-7 reserved */
+ val = (~ucontrol->value.integer.value[0] & 0xffff) | 0x00f0;
+ changed = val != ice->gpio.write_mask;
+ ice->gpio.write_mask = val;
+ mutex_unlock(&ice->gpio_mutex);
+ return changed;
+}
+
+static int pontis_gpio_dir_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&ice->gpio_mutex);
+ /* 4-7 reserved */
+ ucontrol->value.integer.value[0] = ice->gpio.direction & 0xff0f;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int pontis_gpio_dir_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int changed;
+ mutex_lock(&ice->gpio_mutex);
+ /* 4-7 reserved */
+ val = ucontrol->value.integer.value[0] & 0xff0f;
+ changed = (val != ice->gpio.direction);
+ ice->gpio.direction = val;
+ mutex_unlock(&ice->gpio_mutex);
+ return changed;
+}
+
+static int pontis_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&ice->gpio_mutex);
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+ snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask);
+ ucontrol->value.integer.value[0] = snd_ice1712_gpio_read(ice) & 0xffff;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int pontis_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned int val, nval;
+ int changed = 0;
+ mutex_lock(&ice->gpio_mutex);
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+ snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask);
+ val = snd_ice1712_gpio_read(ice) & 0xffff;
+ nval = ucontrol->value.integer.value[0] & 0xffff;
+ if (val != nval) {
+ snd_ice1712_gpio_write(ice, nval);
+ changed = 1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return changed;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_volume, -6400, 50, 1);
+
+/*
+ * mixers
+ */
+
+static struct snd_kcontrol_new pontis_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "PCM Playback Volume",
+ .info = wm_dac_vol_info,
+ .get = wm_dac_vol_get,
+ .put = wm_dac_vol_put,
+ .tlv = { .p = db_scale_volume },
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Capture Volume",
+ .info = wm_adc_vol_info,
+ .get = wm_adc_vol_get,
+ .put = wm_adc_vol_put,
+ .tlv = { .p = db_scale_volume },
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "CD Capture Switch",
+ .info = wm_adc_mux_info,
+ .get = wm_adc_mux_get,
+ .put = wm_adc_mux_put,
+ .private_value = 0,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Capture Switch",
+ .info = wm_adc_mux_info,
+ .get = wm_adc_mux_get,
+ .put = wm_adc_mux_put,
+ .private_value = 1,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Bypass Switch",
+ .info = wm_bypass_info,
+ .get = wm_bypass_get,
+ .put = wm_bypass_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Swap Output Channels",
+ .info = wm_chswap_info,
+ .get = wm_chswap_get,
+ .put = wm_chswap_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Input Source",
+ .info = cs_source_info,
+ .get = cs_source_get,
+ .put = cs_source_put,
+ },
+ /* FIXME: which interface? */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "GPIO Mask",
+ .info = pontis_gpio_mask_info,
+ .get = pontis_gpio_mask_get,
+ .put = pontis_gpio_mask_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "GPIO Direction",
+ .info = pontis_gpio_mask_info,
+ .get = pontis_gpio_dir_get,
+ .put = pontis_gpio_dir_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "GPIO Data",
+ .info = pontis_gpio_mask_info,
+ .get = pontis_gpio_data_get,
+ .put = pontis_gpio_data_put,
+ },
+};
+
+
+/*
+ * WM codec registers
+ */
+static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data;
+ char line[64];
+ unsigned int reg, val;
+ mutex_lock(&ice->gpio_mutex);
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ if (reg <= 0x17 && val <= 0xffff)
+ wm_put(ice, reg, val);
+ }
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data;
+ int reg, val;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (reg = 0; reg <= 0x17; reg++) {
+ val = wm_get(ice, reg);
+ snd_iprintf(buffer, "%02x = %04x\n", reg, val);
+ }
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void wm_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+ if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) {
+ snd_info_set_text_ops(entry, ice, wm_proc_regs_read);
+ entry->mode |= S_IWUSR;
+ entry->c.text.write = wm_proc_regs_write;
+ }
+}
+
+static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data;
+ int reg, val;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (reg = 0; reg <= 0x26; reg++) {
+ val = spi_read(ice, CS_DEV, reg);
+ snd_iprintf(buffer, "%02x = %02x\n", reg, val);
+ }
+ val = spi_read(ice, CS_DEV, 0x7f);
+ snd_iprintf(buffer, "%02x = %02x\n", 0x7f, val);
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void cs_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+ if (! snd_card_proc_new(ice->card, "cs_codec", &entry))
+ snd_info_set_text_ops(entry, ice, cs_proc_regs_read);
+}
+
+
+static int __devinit pontis_add_controls(struct snd_ice1712 *ice)
+{
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < ARRAY_SIZE(pontis_controls); i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&pontis_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ wm_proc_init(ice);
+ cs_proc_init(ice);
+
+ return 0;
+}
+
+
+/*
+ * initialize the chip
+ */
+static int __devinit pontis_init(struct snd_ice1712 *ice)
+{
+ static const unsigned short wm_inits[] = {
+ /* These come first to reduce init pop noise */
+ WM_ADC_MUX, 0x00c0, /* ADC mute */
+ WM_DAC_MUTE, 0x0001, /* DAC softmute */
+ WM_DAC_CTRL1, 0x0000, /* DAC mute */
+
+ WM_POWERDOWN, 0x0008, /* All power-up except HP */
+ WM_RESET, 0x0000, /* reset */
+ };
+ static const unsigned short wm_inits2[] = {
+ WM_MASTER_CTRL, 0x0022, /* 256fs, slave mode */
+ WM_DAC_INT, 0x0022, /* I2S, normal polarity, 24bit */
+ WM_ADC_INT, 0x0022, /* I2S, normal polarity, 24bit */
+ WM_DAC_CTRL1, 0x0090, /* DAC L/R */
+ WM_OUT_MUX, 0x0001, /* OUT DAC */
+ WM_HP_ATTEN_L, 0x0179, /* HP 0dB */
+ WM_HP_ATTEN_R, 0x0179, /* HP 0dB */
+ WM_DAC_ATTEN_L, 0x0000, /* DAC 0dB */
+ WM_DAC_ATTEN_L, 0x0100, /* DAC 0dB */
+ WM_DAC_ATTEN_R, 0x0000, /* DAC 0dB */
+ WM_DAC_ATTEN_R, 0x0100, /* DAC 0dB */
+ /* WM_DAC_MASTER, 0x0100, */ /* DAC master muted */
+ WM_PHASE_SWAP, 0x0000, /* phase normal */
+ WM_DAC_CTRL2, 0x0000, /* no deemphasis, no ZFLG */
+ WM_ADC_ATTEN_L, 0x0000, /* ADC muted */
+ WM_ADC_ATTEN_R, 0x0000, /* ADC muted */
+#if 0
+ WM_ALC_CTRL1, 0x007b, /* */
+ WM_ALC_CTRL2, 0x0000, /* */
+ WM_ALC_CTRL3, 0x0000, /* */
+ WM_NOISE_GATE, 0x0000, /* */
+#endif
+ WM_DAC_MUTE, 0x0000, /* DAC unmute */
+ WM_ADC_MUX, 0x0003, /* ADC unmute, both CD/Line On */
+ };
+ static const unsigned char cs_inits[] = {
+ 0x04, 0x80, /* RUN, RXP0 */
+ 0x05, 0x05, /* slave, 24bit */
+ 0x01, 0x00,
+ 0x02, 0x00,
+ 0x03, 0x00,
+ };
+ unsigned int i;
+
+ ice->vt1720 = 1;
+ ice->num_total_dacs = 2;
+ ice->num_total_adcs = 2;
+
+ /* to remeber the register values */
+ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (! ice->akm)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ /* HACK - use this as the SPDIF source.
+ * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
+ */
+ ice->gpio.saved[0] = 0;
+
+ /* initialize WM8776 codec */
+ for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
+ wm_put(ice, wm_inits[i], wm_inits[i+1]);
+ schedule_timeout_uninterruptible(1);
+ for (i = 0; i < ARRAY_SIZE(wm_inits2); i += 2)
+ wm_put(ice, wm_inits2[i], wm_inits2[i+1]);
+
+ /* initialize CS8416 codec */
+ /* assert PRST#; MT05 bit 7 */
+ outb(inb(ICEMT1724(ice, AC97_CMD)) | 0x80, ICEMT1724(ice, AC97_CMD));
+ mdelay(5);
+ /* deassert PRST# */
+ outb(inb(ICEMT1724(ice, AC97_CMD)) & ~0x80, ICEMT1724(ice, AC97_CMD));
+
+ for (i = 0; i < ARRAY_SIZE(cs_inits); i += 2)
+ spi_write(ice, CS_DEV, cs_inits[i], cs_inits[i+1]);
+
+ return 0;
+}
+
+
+/*
+ * Pontis boards don't provide the EEPROM data at all.
+ * hence the driver needs to sets up it properly.
+ */
+
+static unsigned char pontis_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x08, /* clock 256, mpu401, spdif-in/ADC, 1DAC */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0x07,
+ [ICE_EEP2_GPIO_DIR1] = 0x00,
+ [ICE_EEP2_GPIO_DIR2] = 0x00, /* ignored */
+ [ICE_EEP2_GPIO_MASK] = 0x0f, /* 4-7 reserved for CS8416 */
+ [ICE_EEP2_GPIO_MASK1] = 0xff,
+ [ICE_EEP2_GPIO_MASK2] = 0x00, /* ignored */
+ [ICE_EEP2_GPIO_STATE] = 0x06, /* 0-low, 1-high, 2-high */
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00, /* ignored */
+};
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1720_pontis_cards[] __devinitdata = {
+ {
+ .subvendor = VT1720_SUBDEVICE_PONTIS_MS300,
+ .name = "Pontis MS300",
+ .model = "ms300",
+ .chip_init = pontis_init,
+ .build_controls = pontis_add_controls,
+ .eeprom_size = sizeof(pontis_eeprom),
+ .eeprom_data = pontis_eeprom,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/pontis.h b/sound/pci/ice1712/pontis.h
new file mode 100644
index 0000000..d0d1378
--- /dev/null
+++ b/sound/pci/ice1712/pontis.h
@@ -0,0 +1,33 @@
+#ifndef __SOUND_PONTIS_H
+#define __SOUND_PONTIS_H
+
+/*
+ * ALSA driver for VIA VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Pontis MS300 boards
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define PONTIS_DEVICE_DESC "{Pontis,MS300},"
+
+#define VT1720_SUBDEVICE_PONTIS_MS300 0x00020002 /* a dummy id for MS300 */
+
+extern struct snd_ice1712_card_info snd_vt1720_pontis_cards[];
+
+#endif /* __SOUND_PONTIS_H */
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
new file mode 100644
index 0000000..48d3679
--- /dev/null
+++ b/sound/pci/ice1712/prodigy192.c
@@ -0,0 +1,817 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for AudioTrak Prodigy 192 cards
+ * Supported IEC958 input from optional MI/ODI/O add-on card.
+ *
+ * Specifics (SW, HW):
+ * -------------------
+ * * 49.5MHz crystal
+ * * SPDIF-OUT on the card:
+ * - coax (through isolation transformer)/toslink supplied by
+ * 74HC04 gates - 3 in parallel
+ * - output switched between on-board CD drive dig-out connector
+ * and ice1724 SPDTX pin, using 74HC02 NOR gates, controlled
+ * by GPIO20 (0 = CD dig-out, 1 = SPDTX)
+ * * SPDTX goes straight to MI/ODI/O card's SPDIF-OUT coax
+ *
+ * * MI/ODI/O card: AK4114 based, used for iec958 input only
+ * - toslink input -> RX0
+ * - coax input -> RX1
+ * - 4wire protocol:
+ * AK4114 ICE1724
+ * ------------------------------
+ * CDTO (pin 32) -- GPIO11 pin 86
+ * CDTI (pin 33) -- GPIO10 pin 77
+ * CCLK (pin 34) -- GPIO9 pin 76
+ * CSN (pin 35) -- GPIO8 pin 75
+ * - output data Mode 7 (24bit, I2S, slave)
+ * - both MCKO1 and MCKO2 of ak4114 are fed to FPGA, which
+ * outputs master clock to SPMCLKIN of ice1724.
+ * Experimentally I found out that only a combination of
+ * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 -
+ * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct
+ * sampling rate. That means the the FPGA doubles the
+ * MCK01 rate.
+ *
+ * Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
+ * Copyright (c) 2003 Dimitromanolakis Apostolos <apostol@cs.utoronto.ca>
+ * Copyright (c) 2004 Kouichi ONO <co2b@ceres.dti.ne.jp>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "prodigy192.h"
+#include "stac946x.h"
+#include <sound/tlv.h>
+
+struct prodigy192_spec {
+ struct ak4114 *ak4114;
+ /* rate change needs atomic mute/unmute of all dacs*/
+ struct mutex mute_mutex;
+};
+
+static inline void stac9460_put(struct snd_ice1712 *ice, int reg, unsigned char val)
+{
+ snd_vt1724_write_i2c(ice, PRODIGY192_STAC9460_ADDR, reg, val);
+}
+
+static inline unsigned char stac9460_get(struct snd_ice1712 *ice, int reg)
+{
+ return snd_vt1724_read_i2c(ice, PRODIGY192_STAC9460_ADDR, reg);
+}
+
+/*
+ * DAC mute control
+ */
+
+/*
+ * idx = STAC9460 volume register number, mute: 0 = mute, 1 = unmute
+ */
+static int stac9460_dac_mute(struct snd_ice1712 *ice, int idx,
+ unsigned char mute)
+{
+ unsigned char new, old;
+ int change;
+ old = stac9460_get(ice, idx);
+ new = (~mute << 7 & 0x80) | (old & ~0x80);
+ change = (new != old);
+ if (change)
+ /*printk ("Volume register 0x%02x: 0x%02x\n", idx, new);*/
+ stac9460_put(ice, idx, new);
+ return change;
+}
+
+#define stac9460_dac_mute_info snd_ctl_boolean_mono_info
+
+static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int idx;
+
+ if (kcontrol->private_value)
+ idx = STAC946X_MASTER_VOLUME;
+ else
+ idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME;
+ val = stac9460_get(ice, idx);
+ ucontrol->value.integer.value[0] = (~val >> 7) & 0x1;
+ return 0;
+}
+
+static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy192_spec *spec = ice->spec;
+ int idx, change;
+
+ if (kcontrol->private_value)
+ idx = STAC946X_MASTER_VOLUME;
+ else
+ idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME;
+ /* due to possible conflicts with stac9460_set_rate_val, mutexing */
+ mutex_lock(&spec->mute_mutex);
+ /*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx,
+ ucontrol->value.integer.value[0]);*/
+ change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]);
+ mutex_unlock(&spec->mute_mutex);
+ return change;
+}
+
+/*
+ * DAC volume attenuation mixer control
+ */
+static int stac9460_dac_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0; /* mute */
+ uinfo->value.integer.max = 0x7f; /* 0dB */
+ return 0;
+}
+
+static int stac9460_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx;
+ unsigned char vol;
+
+ if (kcontrol->private_value)
+ idx = STAC946X_MASTER_VOLUME;
+ else
+ idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME;
+ vol = stac9460_get(ice, idx) & 0x7f;
+ ucontrol->value.integer.value[0] = 0x7f - vol;
+
+ return 0;
+}
+
+static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx;
+ unsigned char tmp, ovol, nvol;
+ int change;
+
+ if (kcontrol->private_value)
+ idx = STAC946X_MASTER_VOLUME;
+ else
+ idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME;
+ nvol = ucontrol->value.integer.value[0];
+ tmp = stac9460_get(ice, idx);
+ ovol = 0x7f - (tmp & 0x7f);
+ change = (ovol != nvol);
+ if (change) {
+ ovol = (0x7f - nvol) | (tmp & 0x80);
+ /*printk("DAC Volume: reg 0x%02x: 0x%02x\n", idx, ovol);*/
+ stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80));
+ }
+ return change;
+}
+
+/*
+ * ADC mute control
+ */
+#define stac9460_adc_mute_info snd_ctl_boolean_stereo_info
+
+static int stac9460_adc_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int i;
+
+ for (i = 0; i < 2; ++i) {
+ val = stac9460_get(ice, STAC946X_MIC_L_VOLUME + i);
+ ucontrol->value.integer.value[i] = ~val>>7 & 0x1;
+ }
+
+ return 0;
+}
+
+static int stac9460_adc_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char new, old;
+ int i, reg;
+ int change;
+
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ old = stac9460_get(ice, reg);
+ new = (~ucontrol->value.integer.value[i]<<7&0x80) | (old&~0x80);
+ change = (new != old);
+ if (change)
+ stac9460_put(ice, reg, new);
+ }
+
+ return change;
+}
+
+/*
+ * ADC gain mixer control
+ */
+static int stac9460_adc_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* 0dB */
+ uinfo->value.integer.max = 0x0f; /* 22.5dB */
+ return 0;
+}
+
+static int stac9460_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int i, reg;
+ unsigned char vol;
+
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ vol = stac9460_get(ice, reg) & 0x0f;
+ ucontrol->value.integer.value[i] = 0x0f - vol;
+ }
+
+ return 0;
+}
+
+static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int i, reg;
+ unsigned char ovol, nvol;
+ int change;
+
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ nvol = ucontrol->value.integer.value[i] & 0x0f;
+ ovol = 0x0f - stac9460_get(ice, reg);
+ change = ((ovol & 0x0f) != nvol);
+ if (change)
+ stac9460_put(ice, reg, (0x0f - nvol) | (ovol & ~0x0f));
+ }
+
+ return change;
+}
+
+static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "Line In", "Mic" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+
+static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+
+ val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE);
+ ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1;
+ return 0;
+}
+
+static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char new, old;
+ int change;
+ old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE);
+ new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80);
+ change = (new != old);
+ if (change)
+ stac9460_put(ice, STAC946X_GENERAL_PURPOSE, new);
+ return change;
+}
+/*
+ * Handler for setting correct codec rate - called when rate change is detected
+ */
+static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate)
+{
+ unsigned char old, new;
+ int idx;
+ unsigned char changed[7];
+ struct prodigy192_spec *spec = ice->spec;
+
+ if (rate == 0) /* no hint - S/PDIF input is master, simply return */
+ return;
+ else if (rate <= 48000)
+ new = 0x08; /* 256x, base rate mode */
+ else if (rate <= 96000)
+ new = 0x11; /* 256x, mid rate mode */
+ else
+ new = 0x12; /* 128x, high rate mode */
+ old = stac9460_get(ice, STAC946X_MASTER_CLOCKING);
+ if (old == new)
+ return;
+ /* change detected, setting master clock, muting first */
+ /* due to possible conflicts with mute controls - mutexing */
+ mutex_lock(&spec->mute_mutex);
+ /* we have to remember current mute status for each DAC */
+ for (idx = 0; idx < 7 ; ++idx)
+ changed[idx] = stac9460_dac_mute(ice,
+ STAC946X_MASTER_VOLUME + idx, 0);
+ /*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/
+ stac9460_put(ice, STAC946X_MASTER_CLOCKING, new);
+ udelay(10);
+ /* unmuting - only originally unmuted dacs -
+ * i.e. those changed when muting */
+ for (idx = 0; idx < 7 ; ++idx) {
+ if (changed[idx])
+ stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 1);
+ }
+ mutex_unlock(&spec->mute_mutex);
+}
+
+
+static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0);
+
+/*
+ * mixers
+ */
+
+static struct snd_kcontrol_new stac_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = stac9460_dac_mute_info,
+ .get = stac9460_dac_mute_get,
+ .put = stac9460_dac_mute_put,
+ .private_value = 1,
+ .tlv = { .p = db_scale_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Master Playback Volume",
+ .info = stac9460_dac_vol_info,
+ .get = stac9460_dac_vol_get,
+ .put = stac9460_dac_vol_put,
+ .private_value = 1,
+ .tlv = { .p = db_scale_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Switch",
+ .count = 6,
+ .info = stac9460_dac_mute_info,
+ .get = stac9460_dac_mute_get,
+ .put = stac9460_dac_mute_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "DAC Volume",
+ .count = 6,
+ .info = stac9460_dac_vol_info,
+ .get = stac9460_dac_vol_get,
+ .put = stac9460_dac_vol_put,
+ .tlv = { .p = db_scale_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Capture Switch",
+ .count = 1,
+ .info = stac9460_adc_mute_info,
+ .get = stac9460_adc_mute_get,
+ .put = stac9460_adc_mute_put,
+
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "ADC Capture Volume",
+ .count = 1,
+ .info = stac9460_adc_vol_info,
+ .get = stac9460_adc_vol_get,
+ .put = stac9460_adc_vol_put,
+ .tlv = { .p = db_scale_adc }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Capture Input",
+ .info = stac9460_mic_sw_info,
+ .get = stac9460_mic_sw_get,
+ .put = stac9460_mic_sw_put,
+
+ },
+};
+
+/* AK4114 - ICE1724 connections on Prodigy192 + MI/ODI/O */
+/* CDTO (pin 32) -- GPIO11 pin 86
+ * CDTI (pin 33) -- GPIO10 pin 77
+ * CCLK (pin 34) -- GPIO9 pin 76
+ * CSN (pin 35) -- GPIO8 pin 75
+ */
+#define AK4114_ADDR 0x00 /* C1-C0: Chip Address
+ * (According to datasheet fixed to “00”)
+ */
+
+/*
+ * 4wire ak4114 protocol - writing data
+ */
+static void write_data(struct snd_ice1712 *ice, unsigned int gpio,
+ unsigned int data, int idx)
+{
+ for (; idx >= 0; idx--) {
+ /* drop clock */
+ gpio &= ~VT1724_PRODIGY192_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ /* set data */
+ if (data & (1 << idx))
+ gpio |= VT1724_PRODIGY192_CDOUT;
+ else
+ gpio &= ~VT1724_PRODIGY192_CDOUT;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ /* raise clock */
+ gpio |= VT1724_PRODIGY192_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ }
+}
+
+/*
+ * 4wire ak4114 protocol - reading data
+ */
+static unsigned char read_data(struct snd_ice1712 *ice, unsigned int gpio,
+ int idx)
+{
+ unsigned char data = 0;
+
+ for (; idx >= 0; idx--) {
+ /* drop clock */
+ gpio &= ~VT1724_PRODIGY192_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ /* read data */
+ if (snd_ice1712_gpio_read(ice) & VT1724_PRODIGY192_CDIN)
+ data |= (1 << idx);
+ udelay(1);
+ /* raise clock */
+ gpio |= VT1724_PRODIGY192_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ }
+ return data;
+}
+/*
+ * 4wire ak4114 protocol - starting sequence
+ */
+static unsigned int prodigy192_4wire_start(struct snd_ice1712 *ice)
+{
+ unsigned int tmp;
+
+ snd_ice1712_save_gpio_status(ice);
+ tmp = snd_ice1712_gpio_read(ice);
+
+ tmp |= VT1724_PRODIGY192_CCLK; /* high at init */
+ tmp &= ~VT1724_PRODIGY192_CS; /* drop chip select */
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ return tmp;
+}
+
+/*
+ * 4wire ak4114 protocol - final sequence
+ */
+static void prodigy192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp)
+{
+ tmp |= VT1724_PRODIGY192_CS; /* raise chip select */
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ snd_ice1712_restore_gpio_status(ice);
+}
+
+/*
+ * Write data to addr register of ak4114
+ */
+static void prodigy192_ak4114_write(void *private_data, unsigned char addr,
+ unsigned char data)
+{
+ struct snd_ice1712 *ice = private_data;
+ unsigned int tmp, addrdata;
+ tmp = prodigy192_4wire_start(ice);
+ addrdata = (AK4114_ADDR << 6) | 0x20 | (addr & 0x1f);
+ addrdata = (addrdata << 8) | data;
+ write_data(ice, tmp, addrdata, 15);
+ prodigy192_4wire_finish(ice, tmp);
+}
+
+/*
+ * Read data from addr register of ak4114
+ */
+static unsigned char prodigy192_ak4114_read(void *private_data,
+ unsigned char addr)
+{
+ struct snd_ice1712 *ice = private_data;
+ unsigned int tmp;
+ unsigned char data;
+
+ tmp = prodigy192_4wire_start(ice);
+ write_data(ice, tmp, (AK4114_ADDR << 6) | (addr & 0x1f), 7);
+ data = read_data(ice, tmp, 7);
+ prodigy192_4wire_finish(ice, tmp);
+ return data;
+}
+
+
+static int ak4114_input_sw_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = { "Toslink", "Coax" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+
+static int ak4114_input_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+
+ val = prodigy192_ak4114_read(ice, AK4114_REG_IO1);
+ /* AK4114_IPS0 bit = 0 -> RX0 = Toslink
+ * AK4114_IPS0 bit = 1 -> RX1 = Coax
+ */
+ ucontrol->value.enumerated.item[0] = (val & AK4114_IPS0) ? 1 : 0;
+ return 0;
+}
+
+static int ak4114_input_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char new, old, itemvalue;
+ int change;
+
+ old = prodigy192_ak4114_read(ice, AK4114_REG_IO1);
+ /* AK4114_IPS0 could be any bit */
+ itemvalue = (ucontrol->value.enumerated.item[0]) ? 0xff : 0x00;
+
+ new = (itemvalue & AK4114_IPS0) | (old & ~AK4114_IPS0);
+ change = (new != old);
+ if (change)
+ prodigy192_ak4114_write(ice, AK4114_REG_IO1, new);
+ return change;
+}
+
+
+static struct snd_kcontrol_new ak4114_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "MIODIO IEC958 Capture Input",
+ .info = ak4114_input_sw_info,
+ .get = ak4114_input_sw_get,
+ .put = ak4114_input_sw_put,
+
+ }
+};
+
+
+static int prodigy192_ak4114_init(struct snd_ice1712 *ice)
+{
+ static const unsigned char ak4114_init_vals[] = {
+ AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1,
+ /* ice1724 expects I2S and provides clock,
+ * DEM0 disables the deemphasis filter
+ */
+ AK4114_DIF_I24I2S | AK4114_DEM0 ,
+ AK4114_TX1E,
+ AK4114_EFH_1024 | AK4114_DIT, /* default input RX0 */
+ 0,
+ 0
+ };
+ static const unsigned char ak4114_init_txcsb[] = {
+ 0x41, 0x02, 0x2c, 0x00, 0x00
+ };
+ struct prodigy192_spec *spec = ice->spec;
+ int err;
+
+ err = snd_ak4114_create(ice->card,
+ prodigy192_ak4114_read,
+ prodigy192_ak4114_write,
+ ak4114_init_vals, ak4114_init_txcsb,
+ ice, &spec->ak4114);
+ if (err < 0)
+ return err;
+ /* AK4114 in Prodigy192 cannot detect external rate correctly.
+ * No reason to stop capture stream due to incorrect checks */
+ spec->ak4114->check_flags = AK4114_CHECK_NO_RATE;
+ return 0;
+}
+
+static void stac9460_proc_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data;
+ int reg, val;
+ /* registers 0x0 - 0x14 */
+ for (reg = 0; reg <= 0x15; reg++) {
+ val = stac9460_get(ice, reg);
+ snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val);
+ }
+}
+
+
+static void stac9460_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+ if (!snd_card_proc_new(ice->card, "stac9460_codec", &entry))
+ snd_info_set_text_ops(entry, ice, stac9460_proc_regs_read);
+}
+
+
+static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice)
+{
+ struct prodigy192_spec *spec = ice->spec;
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < ARRAY_SIZE(stac_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&stac_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+ if (spec->ak4114) {
+ /* ak4114 is connected */
+ for (i = 0; i < ARRAY_SIZE(ak4114_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&ak4114_controls[i],
+ ice));
+ if (err < 0)
+ return err;
+ }
+ err = snd_ak4114_build(spec->ak4114,
+ NULL, /* ak4114 in MIO/DI/O handles no IEC958 output */
+ ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
+ if (err < 0)
+ return err;
+ }
+ stac9460_proc_init(ice);
+ return 0;
+}
+
+/*
+ * check for presence of MI/ODI/O add-on card with digital inputs
+ */
+static int prodigy192_miodio_exists(struct snd_ice1712 *ice)
+{
+
+ unsigned char orig_value;
+ const unsigned char test_data = 0xd1; /* random value */
+ unsigned char addr = AK4114_REG_INT0_MASK; /* random SAFE address */
+ int exists = 0;
+
+ orig_value = prodigy192_ak4114_read(ice, addr);
+ prodigy192_ak4114_write(ice, addr, test_data);
+ if (prodigy192_ak4114_read(ice, addr) == test_data) {
+ /* ak4114 seems to communicate, apparently exists */
+ /* writing back original value */
+ prodigy192_ak4114_write(ice, addr, orig_value);
+ exists = 1;
+ }
+ return exists;
+}
+
+/*
+ * initialize the chip
+ */
+static int __devinit prodigy192_init(struct snd_ice1712 *ice)
+{
+ static const unsigned short stac_inits_prodigy[] = {
+ STAC946X_RESET, 0,
+ STAC946X_MASTER_CLOCKING, 0x11,
+/* STAC946X_MASTER_VOLUME, 0,
+ STAC946X_LF_VOLUME, 0,
+ STAC946X_RF_VOLUME, 0,
+ STAC946X_LR_VOLUME, 0,
+ STAC946X_RR_VOLUME, 0,
+ STAC946X_CENTER_VOLUME, 0,
+ STAC946X_LFE_VOLUME, 0,*/
+ (unsigned short)-1
+ };
+ const unsigned short *p;
+ int err = 0;
+ struct prodigy192_spec *spec;
+
+ /* prodigy 192 */
+ ice->num_total_dacs = 6;
+ ice->num_total_adcs = 2;
+ ice->vt1720 = 0; /* ice1724, e.g. 23 GPIOs */
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+ mutex_init(&spec->mute_mutex);
+
+ /* initialize codec */
+ p = stac_inits_prodigy;
+ for (; *p != (unsigned short)-1; p += 2)
+ stac9460_put(ice, p[0], p[1]);
+ ice->gpio.set_pro_rate = stac9460_set_rate_val;
+
+ /* MI/ODI/O add on card with AK4114 */
+ if (prodigy192_miodio_exists(ice)) {
+ err = prodigy192_ak4114_init(ice);
+ /* from this moment if err = 0 then
+ * spec->ak4114 should not be null
+ */
+ snd_printdd("AK4114 initialized with status %d\n", err);
+ } else
+ snd_printdd("AK4114 not found\n");
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * Aureon boards don't provide the EEPROM data except for the vendor IDs.
+ * hence the driver needs to sets up it properly.
+ */
+
+static unsigned char prodigy71_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x6a, /* 49MHz crystal, mpu401,
+ * spdif-in+ 1 stereo ADC,
+ * 3 stereo DACs
+ */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = ~(VT1724_PRODIGY192_CDIN >> 8) ,
+ [ICE_EEP2_GPIO_DIR2] = 0xbf,
+ [ICE_EEP2_GPIO_MASK] = 0x00,
+ [ICE_EEP2_GPIO_MASK1] = 0x00,
+ [ICE_EEP2_GPIO_MASK2] = 0x00,
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x10, /* GPIO20: 0 = CD drive dig. input
+ * passthrough,
+ * 1 = SPDIF-OUT from ice1724
+ */
+};
+
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1724_prodigy192_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_PRODIGY192VE,
+ .name = "Audiotrak Prodigy 192",
+ .model = "prodigy192",
+ .chip_init = prodigy192_init,
+ .build_controls = prodigy192_add_controls,
+ .eeprom_size = sizeof(prodigy71_eeprom),
+ .eeprom_data = prodigy71_eeprom,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/prodigy192.h b/sound/pci/ice1712/prodigy192.h
new file mode 100644
index 0000000..16a53b4
--- /dev/null
+++ b/sound/pci/ice1712/prodigy192.h
@@ -0,0 +1,19 @@
+#ifndef __SOUND_PRODIGY192_H
+#define __SOUND_PRODIGY192_H
+
+#define PRODIGY192_DEVICE_DESC "{AudioTrak,Prodigy 192},"
+#define PRODIGY192_STAC9460_ADDR 0x54
+
+#define VT1724_SUBDEVICE_PRODIGY192VE 0x34495345 /* PRODIGY 192 VE */
+/*
+ * AudioTrak Prodigy192 GPIO definitions for MI/ODI/O card with
+ * AK4114 (SPDIF-IN)
+ */
+#define VT1724_PRODIGY192_CS (1 << 8) /* GPIO8, pin 75 */
+#define VT1724_PRODIGY192_CCLK (1 << 9) /* GPIO9, pin 76 */
+#define VT1724_PRODIGY192_CDOUT (1 << 10) /* GPIO10, pin 77 */
+#define VT1724_PRODIGY192_CDIN (1 << 11) /* GPIO11, pin 86 */
+
+extern struct snd_ice1712_card_info snd_vt1724_prodigy192_cards[];
+
+#endif /* __SOUND_PRODIGY192_H */
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
new file mode 100644
index 0000000..043a938
--- /dev/null
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -0,0 +1,1210 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Audiotrak Prodigy 7.1 Hifi
+ * based on pontis.c
+ *
+ * Copyright (c) 2007 Julian Scheel <julian@jusst.de>
+ * Copyright (c) 2007 allank
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "prodigy_hifi.h"
+
+struct prodigy_hifi_spec {
+ unsigned short master[2];
+ unsigned short vol[8];
+};
+
+/* I2C addresses */
+#define WM_DEV 0x34
+
+/* WM8776 registers */
+#define WM_HP_ATTEN_L 0x00 /* headphone left attenuation */
+#define WM_HP_ATTEN_R 0x01 /* headphone left attenuation */
+#define WM_HP_MASTER 0x02 /* headphone master (both channels),
+ override LLR */
+#define WM_DAC_ATTEN_L 0x03 /* digital left attenuation */
+#define WM_DAC_ATTEN_R 0x04
+#define WM_DAC_MASTER 0x05
+#define WM_PHASE_SWAP 0x06 /* DAC phase swap */
+#define WM_DAC_CTRL1 0x07
+#define WM_DAC_MUTE 0x08
+#define WM_DAC_CTRL2 0x09
+#define WM_DAC_INT 0x0a
+#define WM_ADC_INT 0x0b
+#define WM_MASTER_CTRL 0x0c
+#define WM_POWERDOWN 0x0d
+#define WM_ADC_ATTEN_L 0x0e
+#define WM_ADC_ATTEN_R 0x0f
+#define WM_ALC_CTRL1 0x10
+#define WM_ALC_CTRL2 0x11
+#define WM_ALC_CTRL3 0x12
+#define WM_NOISE_GATE 0x13
+#define WM_LIMITER 0x14
+#define WM_ADC_MUX 0x15
+#define WM_OUT_MUX 0x16
+#define WM_RESET 0x17
+
+/* Analog Recording Source :- Mic, LineIn, CD/Video, */
+
+/* implement capture source select control for WM8776 */
+
+#define WM_AIN1 "AIN1"
+#define WM_AIN2 "AIN2"
+#define WM_AIN3 "AIN3"
+#define WM_AIN4 "AIN4"
+#define WM_AIN5 "AIN5"
+
+/* GPIO pins of envy24ht connected to wm8766 */
+#define WM8766_SPI_CLK (1<<17) /* CLK, Pin97 on ICE1724 */
+#define WM8766_SPI_MD (1<<16) /* DATA VT1724 -> WM8766, Pin96 */
+#define WM8766_SPI_ML (1<<18) /* Latch, Pin98 */
+
+/* WM8766 registers */
+#define WM8766_DAC_CTRL 0x02 /* DAC Control */
+#define WM8766_INT_CTRL 0x03 /* Interface Control */
+#define WM8766_DAC_CTRL2 0x09
+#define WM8766_DAC_CTRL3 0x0a
+#define WM8766_RESET 0x1f
+#define WM8766_LDA1 0x00
+#define WM8766_LDA2 0x04
+#define WM8766_LDA3 0x06
+#define WM8766_RDA1 0x01
+#define WM8766_RDA2 0x05
+#define WM8766_RDA3 0x07
+#define WM8766_MUTE1 0x0C
+#define WM8766_MUTE2 0x0F
+
+
+/*
+ * Prodigy HD2
+ */
+#define AK4396_ADDR 0x00
+#define AK4396_CSN (1 << 8) /* CSN->GPIO8, pin 75 */
+#define AK4396_CCLK (1 << 9) /* CCLK->GPIO9, pin 76 */
+#define AK4396_CDTI (1 << 10) /* CDTI->GPIO10, pin 77 */
+
+/* ak4396 registers */
+#define AK4396_CTRL1 0x00
+#define AK4396_CTRL2 0x01
+#define AK4396_CTRL3 0x02
+#define AK4396_LCH_ATT 0x03
+#define AK4396_RCH_ATT 0x04
+
+
+/*
+ * get the current register value of WM codec
+ */
+static unsigned short wm_get(struct snd_ice1712 *ice, int reg)
+{
+ reg <<= 1;
+ return ((unsigned short)ice->akm[0].images[reg] << 8) |
+ ice->akm[0].images[reg + 1];
+}
+
+/*
+ * set the register value of WM codec and remember it
+ */
+static void wm_put_nocache(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ unsigned short cval;
+ cval = (reg << 9) | val;
+ snd_vt1724_write_i2c(ice, WM_DEV, cval >> 8, cval & 0xff);
+}
+
+static void wm_put(struct snd_ice1712 *ice, int reg, unsigned short val)
+{
+ wm_put_nocache(ice, reg, val);
+ reg <<= 1;
+ ice->akm[0].images[reg] = val >> 8;
+ ice->akm[0].images[reg + 1] = val;
+}
+
+/*
+ * write data in the SPI mode
+ */
+
+static void set_gpio_bit(struct snd_ice1712 *ice, unsigned int bit, int val)
+{
+ unsigned int tmp = snd_ice1712_gpio_read(ice);
+ if (val)
+ tmp |= bit;
+ else
+ tmp &= ~bit;
+ snd_ice1712_gpio_write(ice, tmp);
+}
+
+/*
+ * SPI implementation for WM8766 codec - only writing supported, no readback
+ */
+
+static void wm8766_spi_send_word(struct snd_ice1712 *ice, unsigned int data)
+{
+ int i;
+ for (i = 0; i < 16; i++) {
+ set_gpio_bit(ice, WM8766_SPI_CLK, 0);
+ udelay(1);
+ set_gpio_bit(ice, WM8766_SPI_MD, data & 0x8000);
+ udelay(1);
+ set_gpio_bit(ice, WM8766_SPI_CLK, 1);
+ udelay(1);
+ data <<= 1;
+ }
+}
+
+static void wm8766_spi_write(struct snd_ice1712 *ice, unsigned int reg,
+ unsigned int data)
+{
+ unsigned int block;
+
+ snd_ice1712_gpio_set_dir(ice, WM8766_SPI_MD|
+ WM8766_SPI_CLK|WM8766_SPI_ML);
+ snd_ice1712_gpio_set_mask(ice, ~(WM8766_SPI_MD|
+ WM8766_SPI_CLK|WM8766_SPI_ML));
+ /* latch must be low when writing */
+ set_gpio_bit(ice, WM8766_SPI_ML, 0);
+ block = (reg << 9) | (data & 0x1ff);
+ wm8766_spi_send_word(ice, block); /* REGISTER ADDRESS */
+ /* release latch */
+ set_gpio_bit(ice, WM8766_SPI_ML, 1);
+ udelay(1);
+ /* restore */
+ snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask);
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+}
+
+
+/*
+ * serial interface for ak4396 - only writing supported, no readback
+ */
+
+static void ak4396_send_word(struct snd_ice1712 *ice, unsigned int data)
+{
+ int i;
+ for (i = 0; i < 16; i++) {
+ set_gpio_bit(ice, AK4396_CCLK, 0);
+ udelay(1);
+ set_gpio_bit(ice, AK4396_CDTI, data & 0x8000);
+ udelay(1);
+ set_gpio_bit(ice, AK4396_CCLK, 1);
+ udelay(1);
+ data <<= 1;
+ }
+}
+
+static void ak4396_write(struct snd_ice1712 *ice, unsigned int reg,
+ unsigned int data)
+{
+ unsigned int block;
+
+ snd_ice1712_gpio_set_dir(ice, AK4396_CSN|AK4396_CCLK|AK4396_CDTI);
+ snd_ice1712_gpio_set_mask(ice, ~(AK4396_CSN|AK4396_CCLK|AK4396_CDTI));
+ /* latch must be low when writing */
+ set_gpio_bit(ice, AK4396_CSN, 0);
+ block = ((AK4396_ADDR & 0x03) << 14) | (1 << 13) |
+ ((reg & 0x1f) << 8) | (data & 0xff);
+ ak4396_send_word(ice, block); /* REGISTER ADDRESS */
+ /* release latch */
+ set_gpio_bit(ice, AK4396_CSN, 1);
+ udelay(1);
+ /* restore */
+ snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask);
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+}
+
+
+/*
+ * ak4396 mixers
+ */
+
+
+
+/*
+ * DAC volume attenuation mixer control (-64dB to 0dB)
+ */
+
+static int ak4396_dac_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* mute */
+ uinfo->value.integer.max = 0xFF; /* linear */
+ return 0;
+}
+
+static int ak4396_dac_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i;
+
+ for (i = 0; i < 2; i++)
+ ucontrol->value.integer.value[i] = spec->vol[i];
+
+ return 0;
+}
+
+static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i;
+ int change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ if (ucontrol->value.integer.value[i] != spec->vol[i]) {
+ spec->vol[i] = ucontrol->value.integer.value[i];
+ ak4396_write(ice, AK4396_LCH_ATT + i,
+ spec->vol[i] & 0xff);
+ change = 1;
+ }
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+
+static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Front Playback Volume",
+ .info = ak4396_dac_vol_info,
+ .get = ak4396_dac_vol_get,
+ .put = ak4396_dac_vol_put,
+ .tlv = { .p = db_scale_wm_dac },
+ },
+};
+
+
+/* --------------- */
+
+/*
+ * Logarithmic volume values for WM87*6
+ * Computed as 20 * Log10(255 / x)
+ */
+static const unsigned char wm_vol[256] = {
+ 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23,
+ 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17,
+ 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13,
+ 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11,
+ 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8,
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6,
+ 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+ 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3,
+ 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+ 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0
+};
+
+#define WM_VOL_MAX (sizeof(wm_vol) - 1)
+#define WM_VOL_MUTE 0x8000
+
+
+#define DAC_0dB 0xff
+#define DAC_RES 128
+#define DAC_MIN (DAC_0dB - DAC_RES)
+
+
+static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index,
+ unsigned short vol, unsigned short master)
+{
+ unsigned char nvol;
+
+ if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+ nvol = 0;
+ else {
+ nvol = (((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 128)
+ & WM_VOL_MAX;
+ nvol = (nvol ? (nvol + DAC_MIN) : 0) & 0xff;
+ }
+
+ wm_put(ice, index, nvol);
+ wm_put_nocache(ice, index, 0x100 | nvol);
+}
+
+static void wm8766_set_vol(struct snd_ice1712 *ice, unsigned int index,
+ unsigned short vol, unsigned short master)
+{
+ unsigned char nvol;
+
+ if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+ nvol = 0;
+ else {
+ nvol = (((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 128)
+ & WM_VOL_MAX;
+ nvol = (nvol ? (nvol + DAC_MIN) : 0) & 0xff;
+ }
+
+ wm8766_spi_write(ice, index, (0x0100 | nvol));
+}
+
+
+/*
+ * DAC volume attenuation mixer control (-64dB to 0dB)
+ */
+
+static int wm_dac_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* mute */
+ uinfo->value.integer.max = DAC_RES; /* 0dB, 0.5dB step */
+ return 0;
+}
+
+static int wm_dac_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i;
+
+ for (i = 0; i < 2; i++)
+ ucontrol->value.integer.value[i] =
+ spec->vol[2 + i] & ~WM_VOL_MUTE;
+ return 0;
+}
+
+static int wm_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i, idx, change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ if (ucontrol->value.integer.value[i] != spec->vol[2 + i]) {
+ idx = WM_DAC_ATTEN_L + i;
+ spec->vol[2 + i] &= WM_VOL_MUTE;
+ spec->vol[2 + i] |= ucontrol->value.integer.value[i];
+ wm_set_vol(ice, idx, spec->vol[2 + i], spec->master[i]);
+ change = 1;
+ }
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+
+/*
+ * WM8766 DAC volume attenuation mixer control
+ */
+static int wm8766_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int voices = kcontrol->private_value >> 8;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = voices;
+ uinfo->value.integer.min = 0; /* mute */
+ uinfo->value.integer.max = DAC_RES; /* 0dB */
+ return 0;
+}
+
+static int wm8766_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i, ofs, voices;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ for (i = 0; i < voices; i++)
+ ucontrol->value.integer.value[i] = spec->vol[ofs + i];
+ return 0;
+}
+
+static int wm8766_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i, idx, ofs, voices;
+ int change = 0;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < voices; i++) {
+ if (ucontrol->value.integer.value[i] != spec->vol[ofs + i]) {
+ idx = WM8766_LDA1 + ofs + i;
+ spec->vol[ofs + i] &= WM_VOL_MUTE;
+ spec->vol[ofs + i] |= ucontrol->value.integer.value[i];
+ wm8766_set_vol(ice, idx,
+ spec->vol[ofs + i], spec->master[i]);
+ change = 1;
+ }
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * Master volume attenuation mixer control / applied to WM8776+WM8766
+ */
+static int wm_master_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = DAC_RES;
+ return 0;
+}
+
+static int wm_master_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int i;
+ for (i = 0; i < 2; i++)
+ ucontrol->value.integer.value[i] = spec->master[i];
+ return 0;
+}
+
+static int wm_master_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ struct prodigy_hifi_spec *spec = ice->spec;
+ int ch, change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (ch = 0; ch < 2; ch++) {
+ if (ucontrol->value.integer.value[ch] != spec->master[ch]) {
+ spec->master[ch] = ucontrol->value.integer.value[ch];
+
+ /* Apply to front DAC */
+ wm_set_vol(ice, WM_DAC_ATTEN_L + ch,
+ spec->vol[2 + ch], spec->master[ch]);
+
+ wm8766_set_vol(ice, WM8766_LDA1 + ch,
+ spec->vol[0 + ch], spec->master[ch]);
+
+ wm8766_set_vol(ice, WM8766_LDA2 + ch,
+ spec->vol[4 + ch], spec->master[ch]);
+
+ wm8766_set_vol(ice, WM8766_LDA3 + ch,
+ spec->vol[6 + ch], spec->master[ch]);
+ change = 1;
+ }
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+
+/* KONSTI */
+
+static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char* texts[32] = {
+ "NULL", WM_AIN1, WM_AIN2, WM_AIN1 "+" WM_AIN2,
+ WM_AIN3, WM_AIN1 "+" WM_AIN3, WM_AIN2 "+" WM_AIN3,
+ WM_AIN1 "+" WM_AIN2 "+" WM_AIN3,
+ WM_AIN4, WM_AIN1 "+" WM_AIN4, WM_AIN2 "+" WM_AIN4,
+ WM_AIN1 "+" WM_AIN2 "+" WM_AIN4,
+ WM_AIN3 "+" WM_AIN4, WM_AIN1 "+" WM_AIN3 "+" WM_AIN4,
+ WM_AIN2 "+" WM_AIN3 "+" WM_AIN4,
+ WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4,
+ WM_AIN5, WM_AIN1 "+" WM_AIN5, WM_AIN2 "+" WM_AIN5,
+ WM_AIN1 "+" WM_AIN2 "+" WM_AIN5,
+ WM_AIN3 "+" WM_AIN5, WM_AIN1 "+" WM_AIN3 "+" WM_AIN5,
+ WM_AIN2 "+" WM_AIN3 "+" WM_AIN5,
+ WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN5,
+ WM_AIN4 "+" WM_AIN5, WM_AIN1 "+" WM_AIN4 "+" WM_AIN5,
+ WM_AIN2 "+" WM_AIN4 "+" WM_AIN5,
+ WM_AIN1 "+" WM_AIN2 "+" WM_AIN4 "+" WM_AIN5,
+ WM_AIN3 "+" WM_AIN4 "+" WM_AIN5,
+ WM_AIN1 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5,
+ WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5,
+ WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 32;
+ if (uinfo->value.enumerated.item > 31)
+ uinfo->value.enumerated.item = 31;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short oval, nval;
+ int change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ oval = wm_get(ice, WM_ADC_MUX);
+ nval = (oval & 0xe0) | ucontrol->value.integer.value[0];
+ if (nval != oval) {
+ wm_put(ice, WM_ADC_MUX, nval);
+ change = 1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/* KONSTI */
+
+/*
+ * ADC gain mixer control (-64dB to 0dB)
+ */
+
+#define ADC_0dB 0xcf
+#define ADC_RES 128
+#define ADC_MIN (ADC_0dB - ADC_RES)
+
+static int wm_adc_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* mute (-64dB) */
+ uinfo->value.integer.max = ADC_RES; /* 0dB, 0.5dB step */
+ return 0;
+}
+
+static int wm_adc_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+ int i;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ val = wm_get(ice, WM_ADC_ATTEN_L + i) & 0xff;
+ val = val > ADC_MIN ? (val - ADC_MIN) : 0;
+ ucontrol->value.integer.value[i] = val;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int i, idx, change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (i = 0; i < 2; i++) {
+ nvol = ucontrol->value.integer.value[i];
+ nvol = nvol ? (nvol + ADC_MIN) : 0;
+ idx = WM_ADC_ATTEN_L + i;
+ ovol = wm_get(ice, idx) & 0xff;
+ if (ovol != nvol) {
+ wm_put(ice, idx, nvol);
+ change = 1;
+ }
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * ADC input mux mixer control
+ */
+#define wm_adc_mux_info snd_ctl_boolean_mono_info
+
+static int wm_adc_mux_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int bit = kcontrol->private_value;
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] =
+ (wm_get(ice, WM_ADC_MUX) & (1 << bit)) ? 1 : 0;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_adc_mux_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int bit = kcontrol->private_value;
+ unsigned short oval, nval;
+ int change;
+
+ mutex_lock(&ice->gpio_mutex);
+ nval = oval = wm_get(ice, WM_ADC_MUX);
+ if (ucontrol->value.integer.value[0])
+ nval |= (1 << bit);
+ else
+ nval &= ~(1 << bit);
+ change = nval != oval;
+ if (change) {
+ wm_put(ice, WM_ADC_MUX, nval);
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+/*
+ * Analog bypass (In -> Out)
+ */
+#define wm_bypass_info snd_ctl_boolean_mono_info
+
+static int wm_bypass_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] =
+ (wm_get(ice, WM_OUT_MUX) & 0x04) ? 1 : 0;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_bypass_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val, oval;
+ int change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ val = oval = wm_get(ice, WM_OUT_MUX);
+ if (ucontrol->value.integer.value[0])
+ val |= 0x04;
+ else
+ val &= ~0x04;
+ if (val != oval) {
+ wm_put(ice, WM_OUT_MUX, val);
+ change = 1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+/*
+ * Left/Right swap
+ */
+#define wm_chswap_info snd_ctl_boolean_mono_info
+
+static int wm_chswap_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] =
+ (wm_get(ice, WM_DAC_CTRL1) & 0xf0) != 0x90;
+ mutex_unlock(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_chswap_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val, oval;
+ int change = 0;
+
+ mutex_lock(&ice->gpio_mutex);
+ oval = wm_get(ice, WM_DAC_CTRL1);
+ val = oval & 0x0f;
+ if (ucontrol->value.integer.value[0])
+ val |= 0x60;
+ else
+ val |= 0x90;
+ if (val != oval) {
+ wm_put(ice, WM_DAC_CTRL1, val);
+ wm_put_nocache(ice, WM_DAC_CTRL1, val);
+ change = 1;
+ }
+ mutex_unlock(&ice->gpio_mutex);
+ return change;
+}
+
+
+/*
+ * mixers
+ */
+
+static struct snd_kcontrol_new prodigy_hifi_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Master Playback Volume",
+ .info = wm_master_vol_info,
+ .get = wm_master_vol_get,
+ .put = wm_master_vol_put,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Front Playback Volume",
+ .info = wm_dac_vol_info,
+ .get = wm_dac_vol_get,
+ .put = wm_dac_vol_put,
+ .tlv = { .p = db_scale_wm_dac },
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Rear Playback Volume",
+ .info = wm8766_vol_info,
+ .get = wm8766_vol_get,
+ .put = wm8766_vol_put,
+ .private_value = (2 << 8) | 0,
+ .tlv = { .p = db_scale_wm_dac },
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Center Playback Volume",
+ .info = wm8766_vol_info,
+ .get = wm8766_vol_get,
+ .put = wm8766_vol_put,
+ .private_value = (1 << 8) | 4,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "LFE Playback Volume",
+ .info = wm8766_vol_info,
+ .get = wm8766_vol_get,
+ .put = wm8766_vol_put,
+ .private_value = (1 << 8) | 5,
+ .tlv = { .p = db_scale_wm_dac }
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Side Playback Volume",
+ .info = wm8766_vol_info,
+ .get = wm8766_vol_get,
+ .put = wm8766_vol_put,
+ .private_value = (2 << 8) | 6,
+ .tlv = { .p = db_scale_wm_dac },
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Capture Volume",
+ .info = wm_adc_vol_info,
+ .get = wm_adc_vol_get,
+ .put = wm_adc_vol_put,
+ .tlv = { .p = db_scale_wm_dac },
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "CD Capture Switch",
+ .info = wm_adc_mux_info,
+ .get = wm_adc_mux_get,
+ .put = wm_adc_mux_put,
+ .private_value = 0,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Capture Switch",
+ .info = wm_adc_mux_info,
+ .get = wm_adc_mux_get,
+ .put = wm_adc_mux_put,
+ .private_value = 1,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Bypass Switch",
+ .info = wm_bypass_info,
+ .get = wm_bypass_get,
+ .put = wm_bypass_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Swap Output Channels",
+ .info = wm_chswap_info,
+ .get = wm_chswap_get,
+ .put = wm_chswap_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Capture Source",
+ .info = wm_adc_mux_enum_info,
+ .get = wm_adc_mux_enum_get,
+ .put = wm_adc_mux_enum_put,
+ },
+};
+
+/*
+ * WM codec registers
+ */
+static void wm_proc_regs_write(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = entry->private_data;
+ char line[64];
+ unsigned int reg, val;
+ mutex_lock(&ice->gpio_mutex);
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", &reg, &val) != 2)
+ continue;
+ if (reg <= 0x17 && val <= 0xffff)
+ wm_put(ice, reg, val);
+ }
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void wm_proc_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ice1712 *ice = entry->private_data;
+ int reg, val;
+
+ mutex_lock(&ice->gpio_mutex);
+ for (reg = 0; reg <= 0x17; reg++) {
+ val = wm_get(ice, reg);
+ snd_iprintf(buffer, "%02x = %04x\n", reg, val);
+ }
+ mutex_unlock(&ice->gpio_mutex);
+}
+
+static void wm_proc_init(struct snd_ice1712 *ice)
+{
+ struct snd_info_entry *entry;
+ if (!snd_card_proc_new(ice->card, "wm_codec", &entry)) {
+ snd_info_set_text_ops(entry, ice, wm_proc_regs_read);
+ entry->mode |= S_IWUSR;
+ entry->c.text.write = wm_proc_regs_write;
+ }
+}
+
+static int __devinit prodigy_hifi_add_controls(struct snd_ice1712 *ice)
+{
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < ARRAY_SIZE(prodigy_hifi_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&prodigy_hifi_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ wm_proc_init(ice);
+
+ return 0;
+}
+
+static int __devinit prodigy_hd2_add_controls(struct snd_ice1712 *ice)
+{
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < ARRAY_SIZE(prodigy_hd2_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&prodigy_hd2_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ wm_proc_init(ice);
+
+ return 0;
+}
+
+
+/*
+ * initialize the chip
+ */
+static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice)
+{
+ static unsigned short wm_inits[] = {
+ /* These come first to reduce init pop noise */
+ WM_ADC_MUX, 0x0003, /* ADC mute */
+ /* 0x00c0 replaced by 0x0003 */
+
+ WM_DAC_MUTE, 0x0001, /* DAC softmute */
+ WM_DAC_CTRL1, 0x0000, /* DAC mute */
+
+ WM_POWERDOWN, 0x0008, /* All power-up except HP */
+ WM_RESET, 0x0000, /* reset */
+ };
+ static unsigned short wm_inits2[] = {
+ WM_MASTER_CTRL, 0x0022, /* 256fs, slave mode */
+ WM_DAC_INT, 0x0022, /* I2S, normal polarity, 24bit */
+ WM_ADC_INT, 0x0022, /* I2S, normal polarity, 24bit */
+ WM_DAC_CTRL1, 0x0090, /* DAC L/R */
+ WM_OUT_MUX, 0x0001, /* OUT DAC */
+ WM_HP_ATTEN_L, 0x0179, /* HP 0dB */
+ WM_HP_ATTEN_R, 0x0179, /* HP 0dB */
+ WM_DAC_ATTEN_L, 0x0000, /* DAC 0dB */
+ WM_DAC_ATTEN_L, 0x0100, /* DAC 0dB */
+ WM_DAC_ATTEN_R, 0x0000, /* DAC 0dB */
+ WM_DAC_ATTEN_R, 0x0100, /* DAC 0dB */
+ WM_PHASE_SWAP, 0x0000, /* phase normal */
+#if 0
+ WM_DAC_MASTER, 0x0100, /* DAC master muted */
+#endif
+ WM_DAC_CTRL2, 0x0000, /* no deemphasis, no ZFLG */
+ WM_ADC_ATTEN_L, 0x0000, /* ADC muted */
+ WM_ADC_ATTEN_R, 0x0000, /* ADC muted */
+#if 1
+ WM_ALC_CTRL1, 0x007b, /* */
+ WM_ALC_CTRL2, 0x0000, /* */
+ WM_ALC_CTRL3, 0x0000, /* */
+ WM_NOISE_GATE, 0x0000, /* */
+#endif
+ WM_DAC_MUTE, 0x0000, /* DAC unmute */
+ WM_ADC_MUX, 0x0003, /* ADC unmute, both CD/Line On */
+ };
+ static unsigned short wm8766_inits[] = {
+ WM8766_RESET, 0x0000,
+ WM8766_DAC_CTRL, 0x0120,
+ WM8766_INT_CTRL, 0x0022, /* I2S Normal Mode, 24 bit */
+ WM8766_DAC_CTRL2, 0x0001,
+ WM8766_DAC_CTRL3, 0x0080,
+ WM8766_LDA1, 0x0100,
+ WM8766_LDA2, 0x0100,
+ WM8766_LDA3, 0x0100,
+ WM8766_RDA1, 0x0100,
+ WM8766_RDA2, 0x0100,
+ WM8766_RDA3, 0x0100,
+ WM8766_MUTE1, 0x0000,
+ WM8766_MUTE2, 0x0000,
+ };
+
+ struct prodigy_hifi_spec *spec;
+ unsigned int i;
+
+ ice->vt1720 = 0;
+ ice->vt1724 = 1;
+
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 1;
+
+ /* HACK - use this as the SPDIF source.
+ * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
+ */
+ ice->gpio.saved[0] = 0;
+ /* to remeber the register values */
+
+ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (! ice->akm)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ /* initialize WM8776 codec */
+ for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
+ wm_put(ice, wm_inits[i], wm_inits[i+1]);
+ schedule_timeout_uninterruptible(1);
+ for (i = 0; i < ARRAY_SIZE(wm_inits2); i += 2)
+ wm_put(ice, wm_inits2[i], wm_inits2[i+1]);
+
+ /* initialize WM8766 codec */
+ for (i = 0; i < ARRAY_SIZE(wm8766_inits); i += 2)
+ wm8766_spi_write(ice, wm8766_inits[i], wm8766_inits[i+1]);
+
+
+ return 0;
+}
+
+
+/*
+ * initialize the chip
+ */
+static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice)
+{
+ static unsigned short ak4396_inits[] = {
+ AK4396_CTRL1, 0x87, /* I2S Normal Mode, 24 bit */
+ AK4396_CTRL2, 0x02,
+ AK4396_CTRL3, 0x00,
+ AK4396_LCH_ATT, 0x00,
+ AK4396_RCH_ATT, 0x00,
+ };
+
+ struct prodigy_hifi_spec *spec;
+ unsigned int i;
+
+ ice->vt1720 = 0;
+ ice->vt1724 = 1;
+
+ ice->num_total_dacs = 1;
+ ice->num_total_adcs = 1;
+
+ /* HACK - use this as the SPDIF source.
+ * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten
+ */
+ ice->gpio.saved[0] = 0;
+ /* to remeber the register values */
+
+ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (! ice->akm)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ /* initialize ak4396 codec */
+ /* reset codec */
+ ak4396_write(ice, AK4396_CTRL1, 0x86);
+ msleep(100);
+ ak4396_write(ice, AK4396_CTRL1, 0x87);
+
+ for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2)
+ ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]);
+
+ return 0;
+}
+
+
+static unsigned char prodigy71hifi_eeprom[] __devinitdata = {
+ 0x4b, /* SYSCONF: clock 512, spdif-in/ADC, 4DACs */
+ 0x80, /* ACLINK: I2S */
+ 0xfc, /* I2S: vol, 96k, 24bit, 192k */
+ 0xc3, /* SPDIF: out-en, out-int, spdif-in */
+ 0xff, /* GPIO_DIR */
+ 0xff, /* GPIO_DIR1 */
+ 0x5f, /* GPIO_DIR2 */
+ 0x00, /* GPIO_MASK */
+ 0x00, /* GPIO_MASK1 */
+ 0x00, /* GPIO_MASK2 */
+ 0x00, /* GPIO_STATE */
+ 0x00, /* GPIO_STATE1 */
+ 0x00, /* GPIO_STATE2 */
+};
+
+static unsigned char prodigyhd2_eeprom[] __devinitdata = {
+ 0x4b, /* SYSCONF: clock 512, spdif-in/ADC, 4DACs */
+ 0x80, /* ACLINK: I2S */
+ 0xfc, /* I2S: vol, 96k, 24bit, 192k */
+ 0xc3, /* SPDIF: out-en, out-int, spdif-in */
+ 0xff, /* GPIO_DIR */
+ 0xff, /* GPIO_DIR1 */
+ 0x5f, /* GPIO_DIR2 */
+ 0x00, /* GPIO_MASK */
+ 0x00, /* GPIO_MASK1 */
+ 0x00, /* GPIO_MASK2 */
+ 0x00, /* GPIO_STATE */
+ 0x00, /* GPIO_STATE1 */
+ 0x00, /* GPIO_STATE2 */
+};
+
+static unsigned char fortissimo4_eeprom[] __devinitdata = {
+ 0x43, /* SYSCONF: clock 512, ADC, 4DACs */
+ 0x80, /* ACLINK: I2S */
+ 0xfc, /* I2S: vol, 96k, 24bit, 192k */
+ 0xc1, /* SPDIF: out-en, out-int */
+ 0xff, /* GPIO_DIR */
+ 0xff, /* GPIO_DIR1 */
+ 0x5f, /* GPIO_DIR2 */
+ 0x00, /* GPIO_MASK */
+ 0x00, /* GPIO_MASK1 */
+ 0x00, /* GPIO_MASK2 */
+ 0x00, /* GPIO_STATE */
+ 0x00, /* GPIO_STATE1 */
+ 0x00, /* GPIO_STATE2 */
+};
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1724_prodigy_hifi_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_PRODIGY_HIFI,
+ .name = "Audiotrak Prodigy 7.1 HiFi",
+ .model = "prodigy71hifi",
+ .chip_init = prodigy_hifi_init,
+ .build_controls = prodigy_hifi_add_controls,
+ .eeprom_size = sizeof(prodigy71hifi_eeprom),
+ .eeprom_data = prodigy71hifi_eeprom,
+ .driver = "Prodigy71HIFI",
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_PRODIGY_HD2,
+ .name = "Audiotrak Prodigy HD2",
+ .model = "prodigyhd2",
+ .chip_init = prodigy_hd2_init,
+ .build_controls = prodigy_hd2_add_controls,
+ .eeprom_size = sizeof(prodigyhd2_eeprom),
+ .eeprom_data = prodigyhd2_eeprom,
+ .driver = "Prodigy71HD2",
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_FORTISSIMO4,
+ .name = "Hercules Fortissimo IV",
+ .model = "fortissimo4",
+ .chip_init = prodigy_hifi_init,
+ .build_controls = prodigy_hifi_add_controls,
+ .eeprom_size = sizeof(fortissimo4_eeprom),
+ .eeprom_data = fortissimo4_eeprom,
+ .driver = "Fortissimo4",
+ },
+ { } /* terminator */
+};
+
diff --git a/sound/pci/ice1712/prodigy_hifi.h b/sound/pci/ice1712/prodigy_hifi.h
new file mode 100644
index 0000000..a4415d4
--- /dev/null
+++ b/sound/pci/ice1712/prodigy_hifi.h
@@ -0,0 +1,38 @@
+#ifndef __SOUND_PRODIGY_HIFI_H
+#define __SOUND_PRODIGY_HIFI_H
+
+/*
+ * ALSA driver for VIA VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Audiotrak Prodigy Hifi
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define PRODIGY_HIFI_DEVICE_DESC "{Audiotrak,Prodigy 7.1 HIFI},"\
+ "{Audiotrak Prodigy HD2},"\
+ "{Hercules Fortissimo IV},"
+
+#define VT1724_SUBDEVICE_PRODIGY_HIFI 0x38315441 /* PRODIGY 7.1 HIFI */
+#define VT1724_SUBDEVICE_PRODIGY_HD2 0x37315441 /* PRODIGY HD2 */
+#define VT1724_SUBDEVICE_FORTISSIMO4 0x81160100 /* Fortissimo IV */
+
+
+extern struct snd_ice1712_card_info snd_vt1724_prodigy_hifi_cards[];
+
+#endif /* __SOUND_PRODIGY_HIFI_H */
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
new file mode 100644
index 0000000..b508bb3
--- /dev/null
+++ b/sound/pci/ice1712/revo.c
@@ -0,0 +1,633 @@
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for M-Audio Audiophile 192, Revolution 7.1 and 5.1
+ *
+ * Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "revo.h"
+
+/* a non-standard I2C device for revo51 */
+struct revo51_spec {
+ struct snd_i2c_device *dev;
+ struct snd_pt2258 *pt2258;
+};
+
+static void revo_i2s_mclk_changed(struct snd_ice1712 *ice)
+{
+ /* assert PRST# to converters; MT05 bit 7 */
+ outb(inb(ICEMT1724(ice, AC97_CMD)) | 0x80, ICEMT1724(ice, AC97_CMD));
+ mdelay(5);
+ /* deassert PRST# */
+ outb(inb(ICEMT1724(ice, AC97_CMD)) & ~0x80, ICEMT1724(ice, AC97_CMD));
+}
+
+/*
+ * change the rate of Envy24HT, AK4355 and AK4381
+ */
+static void revo_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
+{
+ unsigned char old, tmp, dfs;
+ int reg, shift;
+
+ if (rate == 0) /* no hint - S/PDIF input is master, simply return */
+ return;
+
+ /* adjust DFS on codecs */
+ if (rate > 96000)
+ dfs = 2;
+ else if (rate > 48000)
+ dfs = 1;
+ else
+ dfs = 0;
+
+ if (ak->type == SND_AK4355 || ak->type == SND_AK4358) {
+ reg = 2;
+ shift = 4;
+ } else {
+ reg = 1;
+ shift = 3;
+ }
+ tmp = snd_akm4xxx_get(ak, 0, reg);
+ old = (tmp >> shift) & 0x03;
+ if (old == dfs)
+ return;
+
+ /* reset DFS */
+ snd_akm4xxx_reset(ak, 1);
+ tmp = snd_akm4xxx_get(ak, 0, reg);
+ tmp &= ~(0x03 << shift);
+ tmp |= dfs << shift;
+ /* snd_akm4xxx_write(ak, 0, reg, tmp); */
+ snd_akm4xxx_set(ak, 0, reg, tmp); /* value is written in reset(0) */
+ snd_akm4xxx_reset(ak, 0);
+}
+
+/*
+ * I2C access to the PT2258 volume controller on GPIO 6/7 (Revolution 5.1)
+ */
+
+static void revo_i2c_start(struct snd_i2c_bus *bus)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ snd_ice1712_save_gpio_status(ice);
+}
+
+static void revo_i2c_stop(struct snd_i2c_bus *bus)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ snd_ice1712_restore_gpio_status(ice);
+}
+
+static void revo_i2c_direction(struct snd_i2c_bus *bus, int clock, int data)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ unsigned int mask, val;
+
+ val = 0;
+ if (clock)
+ val |= VT1724_REVO_I2C_CLOCK; /* write SCL */
+ if (data)
+ val |= VT1724_REVO_I2C_DATA; /* write SDA */
+ mask = VT1724_REVO_I2C_CLOCK | VT1724_REVO_I2C_DATA;
+ ice->gpio.direction &= ~mask;
+ ice->gpio.direction |= val;
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction);
+ snd_ice1712_gpio_set_mask(ice, ~mask);
+}
+
+static void revo_i2c_setlines(struct snd_i2c_bus *bus, int clk, int data)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ unsigned int val = 0;
+
+ if (clk)
+ val |= VT1724_REVO_I2C_CLOCK;
+ if (data)
+ val |= VT1724_REVO_I2C_DATA;
+ snd_ice1712_gpio_write_bits(ice,
+ VT1724_REVO_I2C_DATA |
+ VT1724_REVO_I2C_CLOCK, val);
+ udelay(5);
+}
+
+static int revo_i2c_getdata(struct snd_i2c_bus *bus, int ack)
+{
+ struct snd_ice1712 *ice = bus->private_data;
+ int bit;
+
+ if (ack)
+ udelay(5);
+ bit = snd_ice1712_gpio_read_bits(ice, VT1724_REVO_I2C_DATA) ? 1 : 0;
+ return bit;
+}
+
+static struct snd_i2c_bit_ops revo51_bit_ops = {
+ .start = revo_i2c_start,
+ .stop = revo_i2c_stop,
+ .direction = revo_i2c_direction,
+ .setlines = revo_i2c_setlines,
+ .getdata = revo_i2c_getdata,
+};
+
+static int revo51_i2c_init(struct snd_ice1712 *ice,
+ struct snd_pt2258 *pt)
+{
+ struct revo51_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ /* create the I2C bus */
+ err = snd_i2c_bus_create(ice->card, "ICE1724 GPIO6", NULL, &ice->i2c);
+ if (err < 0)
+ return err;
+
+ ice->i2c->private_data = ice;
+ ice->i2c->hw_ops.bit = &revo51_bit_ops;
+
+ /* create the I2C device */
+ err = snd_i2c_device_create(ice->i2c, "PT2258", 0x40, &spec->dev);
+ if (err < 0)
+ return err;
+
+ pt->card = ice->card;
+ pt->i2c_bus = ice->i2c;
+ pt->i2c_dev = spec->dev;
+ spec->pt2258 = pt;
+
+ snd_pt2258_reset(pt);
+
+ return 0;
+}
+
+/*
+ * initialize the chips on M-Audio Revolution cards
+ */
+
+#define AK_DAC(xname,xch) { .name = xname, .num_channels = xch }
+
+static const struct snd_akm4xxx_dac_channel revo71_front[] = {
+ {
+ .name = "PCM Playback Volume",
+ .num_channels = 2,
+ /* front channels DAC supports muting */
+ .switch_name = "PCM Playback Switch",
+ },
+};
+
+static const struct snd_akm4xxx_dac_channel revo71_surround[] = {
+ AK_DAC("PCM Center Playback Volume", 1),
+ AK_DAC("PCM LFE Playback Volume", 1),
+ AK_DAC("PCM Side Playback Volume", 2),
+ AK_DAC("PCM Rear Playback Volume", 2),
+};
+
+static const struct snd_akm4xxx_dac_channel revo51_dac[] = {
+ AK_DAC("PCM Playback Volume", 2),
+ AK_DAC("PCM Center Playback Volume", 1),
+ AK_DAC("PCM LFE Playback Volume", 1),
+ AK_DAC("PCM Rear Playback Volume", 2),
+ AK_DAC("PCM Headphone Volume", 2),
+};
+
+static const char *revo51_adc_input_names[] = {
+ "Mic",
+ "Line",
+ "CD",
+ NULL
+};
+
+static const struct snd_akm4xxx_adc_channel revo51_adc[] = {
+ {
+ .name = "PCM Capture Volume",
+ .switch_name = "PCM Capture Switch",
+ .num_channels = 2,
+ .input_names = revo51_adc_input_names
+ },
+};
+
+static struct snd_akm4xxx akm_revo_front __devinitdata = {
+ .type = SND_AK4381,
+ .num_dacs = 2,
+ .ops = {
+ .set_rate_val = revo_set_rate_val
+ },
+ .dac_info = revo71_front,
+};
+
+static struct snd_ak4xxx_private akm_revo_front_priv __devinitdata = {
+ .caddr = 1,
+ .cif = 0,
+ .data_mask = VT1724_REVO_CDOUT,
+ .clk_mask = VT1724_REVO_CCLK,
+ .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
+ .cs_addr = VT1724_REVO_CS0 | VT1724_REVO_CS2,
+ .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
+ .add_flags = VT1724_REVO_CCLK, /* high at init */
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_revo_surround __devinitdata = {
+ .type = SND_AK4355,
+ .idx_offset = 1,
+ .num_dacs = 6,
+ .ops = {
+ .set_rate_val = revo_set_rate_val
+ },
+ .dac_info = revo71_surround,
+};
+
+static struct snd_ak4xxx_private akm_revo_surround_priv __devinitdata = {
+ .caddr = 3,
+ .cif = 0,
+ .data_mask = VT1724_REVO_CDOUT,
+ .clk_mask = VT1724_REVO_CCLK,
+ .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
+ .cs_addr = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1 | VT1724_REVO_CS2,
+ .add_flags = VT1724_REVO_CCLK, /* high at init */
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_revo51 __devinitdata = {
+ .type = SND_AK4358,
+ .num_dacs = 8,
+ .ops = {
+ .set_rate_val = revo_set_rate_val
+ },
+ .dac_info = revo51_dac,
+};
+
+static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0,
+ .data_mask = VT1724_REVO_CDOUT,
+ .clk_mask = VT1724_REVO_CCLK,
+ .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .cs_addr = VT1724_REVO_CS1,
+ .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .add_flags = VT1724_REVO_CCLK, /* high at init */
+ .mask_flags = 0,
+};
+
+static struct snd_akm4xxx akm_revo51_adc __devinitdata = {
+ .type = SND_AK5365,
+ .num_adcs = 2,
+ .adc_info = revo51_adc,
+};
+
+static struct snd_ak4xxx_private akm_revo51_adc_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0,
+ .data_mask = VT1724_REVO_CDOUT,
+ .clk_mask = VT1724_REVO_CCLK,
+ .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .cs_addr = VT1724_REVO_CS0,
+ .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .add_flags = VT1724_REVO_CCLK, /* high at init */
+ .mask_flags = 0,
+};
+
+static struct snd_pt2258 ptc_revo51_volume;
+
+/* AK4358 for AP192 DAC, AK5385A for ADC */
+static void ap192_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
+{
+ struct snd_ice1712 *ice = ak->private_data[0];
+ int dfs;
+
+ revo_set_rate_val(ak, rate);
+
+ /* reset CKS */
+ snd_ice1712_gpio_write_bits(ice, 1 << 8, rate > 96000 ? 1 << 8 : 0);
+ /* reset DFS pins of AK5385A for ADC, too */
+ if (rate > 96000)
+ dfs = 2;
+ else if (rate > 48000)
+ dfs = 1;
+ else
+ dfs = 0;
+ snd_ice1712_gpio_write_bits(ice, 3 << 9, dfs << 9);
+ /* reset ADC */
+ snd_ice1712_gpio_write_bits(ice, 1 << 11, 0);
+ snd_ice1712_gpio_write_bits(ice, 1 << 11, 1 << 11);
+}
+
+static const struct snd_akm4xxx_dac_channel ap192_dac[] = {
+ AK_DAC("PCM Playback Volume", 2)
+};
+
+static struct snd_akm4xxx akm_ap192 __devinitdata = {
+ .type = SND_AK4358,
+ .num_dacs = 2,
+ .ops = {
+ .set_rate_val = ap192_set_rate_val
+ },
+ .dac_info = ap192_dac,
+};
+
+static struct snd_ak4xxx_private akm_ap192_priv __devinitdata = {
+ .caddr = 2,
+ .cif = 0,
+ .data_mask = VT1724_REVO_CDOUT,
+ .clk_mask = VT1724_REVO_CCLK,
+ .cs_mask = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .cs_addr = VT1724_REVO_CS1,
+ .cs_none = VT1724_REVO_CS0 | VT1724_REVO_CS1,
+ .add_flags = VT1724_REVO_CCLK, /* high at init */
+ .mask_flags = 0,
+};
+
+/* AK4114 support on Audiophile 192 */
+/* CDTO (pin 32) -- GPIO2 pin 52
+ * CDTI (pin 33) -- GPIO3 pin 53 (shared with AK4358)
+ * CCLK (pin 34) -- GPIO1 pin 51 (shared with AK4358)
+ * CSN (pin 35) -- GPIO7 pin 59
+ */
+#define AK4114_ADDR 0x02
+
+static void write_data(struct snd_ice1712 *ice, unsigned int gpio,
+ unsigned int data, int idx)
+{
+ for (; idx >= 0; idx--) {
+ /* drop clock */
+ gpio &= ~VT1724_REVO_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ /* set data */
+ if (data & (1 << idx))
+ gpio |= VT1724_REVO_CDOUT;
+ else
+ gpio &= ~VT1724_REVO_CDOUT;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ /* raise clock */
+ gpio |= VT1724_REVO_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ }
+}
+
+static unsigned char read_data(struct snd_ice1712 *ice, unsigned int gpio,
+ int idx)
+{
+ unsigned char data = 0;
+
+ for (; idx >= 0; idx--) {
+ /* drop clock */
+ gpio &= ~VT1724_REVO_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ /* read data */
+ if (snd_ice1712_gpio_read(ice) & VT1724_REVO_CDIN)
+ data |= (1 << idx);
+ udelay(1);
+ /* raise clock */
+ gpio |= VT1724_REVO_CCLK;
+ snd_ice1712_gpio_write(ice, gpio);
+ udelay(1);
+ }
+ return data;
+}
+
+static unsigned int ap192_4wire_start(struct snd_ice1712 *ice)
+{
+ unsigned int tmp;
+
+ snd_ice1712_save_gpio_status(ice);
+ tmp = snd_ice1712_gpio_read(ice);
+ tmp |= VT1724_REVO_CCLK; /* high at init */
+ tmp |= VT1724_REVO_CS0;
+ tmp &= ~VT1724_REVO_CS1;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ return tmp;
+}
+
+static void ap192_4wire_finish(struct snd_ice1712 *ice, unsigned int tmp)
+{
+ tmp |= VT1724_REVO_CS1;
+ tmp |= VT1724_REVO_CS0;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ snd_ice1712_restore_gpio_status(ice);
+}
+
+static void ap192_ak4114_write(void *private_data, unsigned char addr,
+ unsigned char data)
+{
+ struct snd_ice1712 *ice = private_data;
+ unsigned int tmp, addrdata;
+
+ tmp = ap192_4wire_start(ice);
+ addrdata = (AK4114_ADDR << 6) | 0x20 | (addr & 0x1f);
+ addrdata = (addrdata << 8) | data;
+ write_data(ice, tmp, addrdata, 15);
+ ap192_4wire_finish(ice, tmp);
+}
+
+static unsigned char ap192_ak4114_read(void *private_data, unsigned char addr)
+{
+ struct snd_ice1712 *ice = private_data;
+ unsigned int tmp;
+ unsigned char data;
+
+ tmp = ap192_4wire_start(ice);
+ write_data(ice, tmp, (AK4114_ADDR << 6) | (addr & 0x1f), 7);
+ data = read_data(ice, tmp, 7);
+ ap192_4wire_finish(ice, tmp);
+ return data;
+}
+
+static int __devinit ap192_ak4114_init(struct snd_ice1712 *ice)
+{
+ static const unsigned char ak4114_init_vals[] = {
+ AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1,
+ AK4114_DIF_I24I2S,
+ AK4114_TX1E,
+ AK4114_EFH_1024 | AK4114_DIT | AK4114_IPS(1),
+ 0,
+ 0
+ };
+ static const unsigned char ak4114_init_txcsb[] = {
+ 0x41, 0x02, 0x2c, 0x00, 0x00
+ };
+ struct ak4114 *ak;
+ int err;
+
+ err = snd_ak4114_create(ice->card,
+ ap192_ak4114_read,
+ ap192_ak4114_write,
+ ak4114_init_vals, ak4114_init_txcsb,
+ ice, &ak);
+ /* AK4114 in Revo cannot detect external rate correctly.
+ * No reason to stop capture stream due to incorrect checks */
+ ak->check_flags = AK4114_CHECK_NO_RATE;
+
+ return 0; /* error ignored; it's no fatal error */
+}
+
+static int __devinit revo_init(struct snd_ice1712 *ice)
+{
+ struct snd_akm4xxx *ak;
+ int err;
+
+ /* determine I2C, DACs and ADCs */
+ switch (ice->eeprom.subvendor) {
+ case VT1724_SUBDEVICE_REVOLUTION71:
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 2;
+ ice->gpio.i2s_mclk_changed = revo_i2s_mclk_changed;
+ break;
+ case VT1724_SUBDEVICE_REVOLUTION51:
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 2;
+ break;
+ case VT1724_SUBDEVICE_AUDIOPHILE192:
+ ice->num_total_dacs = 2;
+ ice->num_total_adcs = 2;
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+
+ /* second stage of initialization, analog parts and others */
+ ak = ice->akm = kcalloc(2, sizeof(struct snd_akm4xxx), GFP_KERNEL);
+ if (! ak)
+ return -ENOMEM;
+ switch (ice->eeprom.subvendor) {
+ case VT1724_SUBDEVICE_REVOLUTION71:
+ ice->akm_codecs = 2;
+ err = snd_ice1712_akm4xxx_init(ak, &akm_revo_front,
+ &akm_revo_front_priv, ice);
+ if (err < 0)
+ return err;
+ err = snd_ice1712_akm4xxx_init(ak+1, &akm_revo_surround,
+ &akm_revo_surround_priv, ice);
+ if (err < 0)
+ return err;
+ /* unmute all codecs */
+ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE,
+ VT1724_REVO_MUTE);
+ break;
+ case VT1724_SUBDEVICE_REVOLUTION51:
+ ice->akm_codecs = 2;
+ err = snd_ice1712_akm4xxx_init(ak, &akm_revo51,
+ &akm_revo51_priv, ice);
+ if (err < 0)
+ return err;
+ err = snd_ice1712_akm4xxx_init(ak+1, &akm_revo51_adc,
+ &akm_revo51_adc_priv, ice);
+ if (err < 0)
+ return err;
+ err = revo51_i2c_init(ice, &ptc_revo51_volume);
+ if (err < 0)
+ return err;
+ /* unmute all codecs */
+ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE,
+ VT1724_REVO_MUTE);
+ break;
+ case VT1724_SUBDEVICE_AUDIOPHILE192:
+ ice->akm_codecs = 1;
+ err = snd_ice1712_akm4xxx_init(ak, &akm_ap192, &akm_ap192_priv,
+ ice);
+ if (err < 0)
+ return err;
+
+ /* unmute all codecs */
+ snd_ice1712_gpio_write_bits(ice, VT1724_REVO_MUTE,
+ VT1724_REVO_MUTE);
+ break;
+ }
+
+ return 0;
+}
+
+
+static int __devinit revo_add_controls(struct snd_ice1712 *ice)
+{
+ struct revo51_spec *spec;
+ int err;
+
+ switch (ice->eeprom.subvendor) {
+ case VT1724_SUBDEVICE_REVOLUTION71:
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+ break;
+ case VT1724_SUBDEVICE_REVOLUTION51:
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+ spec = ice->spec;
+ err = snd_pt2258_build_controls(spec->pt2258);
+ if (err < 0)
+ return err;
+ break;
+ case VT1724_SUBDEVICE_AUDIOPHILE192:
+ err = snd_ice1712_akm4xxx_build_controls(ice);
+ if (err < 0)
+ return err;
+ err = ap192_ak4114_init(ice);
+ if (err < 0)
+ return err;
+ break;
+ }
+ return 0;
+}
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1724_revo_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_REVOLUTION71,
+ .name = "M Audio Revolution-7.1",
+ .model = "revo71",
+ .chip_init = revo_init,
+ .build_controls = revo_add_controls,
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_REVOLUTION51,
+ .name = "M Audio Revolution-5.1",
+ .model = "revo51",
+ .chip_init = revo_init,
+ .build_controls = revo_add_controls,
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_AUDIOPHILE192,
+ .name = "M Audio Audiophile192",
+ .model = "ap192",
+ .chip_init = revo_init,
+ .build_controls = revo_add_controls,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/revo.h b/sound/pci/ice1712/revo.h
new file mode 100644
index 0000000..a3ba425
--- /dev/null
+++ b/sound/pci/ice1712/revo.h
@@ -0,0 +1,55 @@
+#ifndef __SOUND_REVO_H
+#define __SOUND_REVO_H
+
+/*
+ * ALSA driver for ICEnsemble ICE1712 (Envy24)
+ *
+ * Lowlevel functions for M-Audio Revolution 7.1
+ *
+ * Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define REVO_DEVICE_DESC \
+ "{MidiMan M Audio,Revolution 7.1},"\
+ "{MidiMan M Audio,Revolution 5.1},"\
+ "{MidiMan M Audio,Audiophile 192},"
+
+#define VT1724_SUBDEVICE_REVOLUTION71 0x12143036
+#define VT1724_SUBDEVICE_REVOLUTION51 0x12143136
+#define VT1724_SUBDEVICE_AUDIOPHILE192 0x12143236
+
+/* entry point */
+extern struct snd_ice1712_card_info snd_vt1724_revo_cards[];
+
+
+/*
+ * MidiMan M-Audio Revolution GPIO definitions
+ */
+
+#define VT1724_REVO_CCLK 0x02
+#define VT1724_REVO_CDIN 0x04 /* not used */
+#define VT1724_REVO_CDOUT 0x08
+#define VT1724_REVO_CS0 0x10 /* AK5365 chipselect for (revo51) */
+#define VT1724_REVO_CS1 0x20 /* front AKM4381 chipselect */
+#define VT1724_REVO_CS2 0x40 /* surround AKM4355 CS (revo71) */
+#define VT1724_REVO_I2C_DATA 0x40 /* I2C: PT 2258 SDA (on revo51) */
+#define VT1724_REVO_I2C_CLOCK 0x80 /* I2C: PT 2258 SCL (on revo51) */
+#define VT1724_REVO_CS3 0x80 /* AK4114 for AP192 */
+#define VT1724_REVO_MUTE (1<<22) /* 0 = all mute, 1 = normal operation */
+
+#endif /* __SOUND_REVO_H */
diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c
new file mode 100644
index 0000000..69673b9
--- /dev/null
+++ b/sound/pci/ice1712/se.c
@@ -0,0 +1,774 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for ONKYO WAVIO SE-90PCI and SE-200PCI
+ *
+ * Copyright (c) 2007 Shin-ya Okada sh_okada(at)d4.dion.ne.jp
+ * (at) -> @
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/tlv.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "se.h"
+
+struct se_spec {
+ struct {
+ unsigned char ch1, ch2;
+ } vol[8];
+};
+
+/****************************************************************************/
+/* ONKYO WAVIO SE-200PCI */
+/****************************************************************************/
+/*
+ * system configuration ICE_EEP2_SYSCONF=0x4b
+ * XIN1 49.152MHz
+ * not have UART
+ * one stereo ADC and a S/PDIF receiver connected
+ * four stereo DACs connected
+ *
+ * AC-Link configuration ICE_EEP2_ACLINK=0x80
+ * use I2C, not use AC97
+ *
+ * I2S converters feature ICE_EEP2_I2S=0x78
+ * I2S codec has no volume/mute control feature
+ * I2S codec supports 96KHz and 192KHz
+ * I2S codec 24bits
+ *
+ * S/PDIF configuration ICE_EEP2_SPDIF=0xc3
+ * Enable integrated S/PDIF transmitter
+ * internal S/PDIF out implemented
+ * S/PDIF is stereo
+ * External S/PDIF out implemented
+ *
+ *
+ * ** connected chips **
+ *
+ * WM8740
+ * A 2ch-DAC of main outputs.
+ * It setuped as I2S mode by wire, so no way to setup from software.
+ * The sample-rate are automatically changed.
+ * ML/I2S (28pin) --------+
+ * MC/DM1 (27pin) -- 5V |
+ * MD/DM0 (26pin) -- GND |
+ * MUTEB (25pin) -- NC |
+ * MODE (24pin) -- GND |
+ * CSBIW (23pin) --------+
+ * |
+ * RSTB (22pin) --R(1K)-+
+ * Probably it reduce the noise from the control line.
+ *
+ * WM8766
+ * A 6ch-DAC for surrounds.
+ * It's control wire was connected to GPIOxx (3-wire serial interface)
+ * ML/I2S (11pin) -- GPIO18
+ * MC/IWL (12pin) -- GPIO17
+ * MD/DM (13pin) -- GPIO16
+ * MUTE (14pin) -- GPIO01
+ *
+ * WM8776
+ * A 2ch-ADC(with 10ch-selector) plus 2ch-DAC.
+ * It's control wire was connected to SDA/SCLK (2-wire serial interface)
+ * MODE (16pin) -- R(1K) -- GND
+ * CE (17pin) -- R(1K) -- GND 2-wire mode (address=0x34)
+ * DI (18pin) -- SDA
+ * CL (19pin) -- SCLK
+ *
+ *
+ * ** output pins and device names **
+ *
+ * 7.1ch name -- output connector color -- device (-D option)
+ *
+ * FRONT 2ch -- green -- plughw:0,0
+ * CENTER(Lch) SUBWOOFER(Rch) -- black -- plughw:0,2,0
+ * SURROUND 2ch -- orange -- plughw:0,2,1
+ * SURROUND BACK 2ch -- white -- plughw:0,2,2
+ *
+ */
+
+
+/****************************************************************************/
+/* WM8740 interface */
+/****************************************************************************/
+
+static void __devinit se200pci_WM8740_init(struct snd_ice1712 *ice)
+{
+ /* nothing to do */
+}
+
+
+static void se200pci_WM8740_set_pro_rate(struct snd_ice1712 *ice,
+ unsigned int rate)
+{
+ /* nothing to do */
+}
+
+
+/****************************************************************************/
+/* WM8766 interface */
+/****************************************************************************/
+
+static void se200pci_WM8766_write(struct snd_ice1712 *ice,
+ unsigned int addr, unsigned int data)
+{
+ unsigned int st;
+ unsigned int bits;
+ int i;
+ const unsigned int DATA = 0x010000;
+ const unsigned int CLOCK = 0x020000;
+ const unsigned int LOAD = 0x040000;
+ const unsigned int ALL_MASK = (DATA | CLOCK | LOAD);
+
+ snd_ice1712_save_gpio_status(ice);
+
+ st = ((addr & 0x7f) << 9) | (data & 0x1ff);
+ snd_ice1712_gpio_set_dir(ice, ice->gpio.direction | ALL_MASK);
+ snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask & ~ALL_MASK);
+ bits = snd_ice1712_gpio_read(ice) & ~ALL_MASK;
+
+ snd_ice1712_gpio_write(ice, bits);
+ for (i = 0; i < 16; i++) {
+ udelay(1);
+ bits &= ~CLOCK;
+ st = (st << 1);
+ if (st & 0x10000)
+ bits |= DATA;
+ else
+ bits &= ~DATA;
+
+ snd_ice1712_gpio_write(ice, bits);
+
+ udelay(1);
+ bits |= CLOCK;
+ snd_ice1712_gpio_write(ice, bits);
+ }
+
+ udelay(1);
+ bits |= LOAD;
+ snd_ice1712_gpio_write(ice, bits);
+
+ udelay(1);
+ bits |= (DATA | CLOCK);
+ snd_ice1712_gpio_write(ice, bits);
+
+ snd_ice1712_restore_gpio_status(ice);
+}
+
+static void se200pci_WM8766_set_volume(struct snd_ice1712 *ice, int ch,
+ unsigned int vol1, unsigned int vol2)
+{
+ switch (ch) {
+ case 0:
+ se200pci_WM8766_write(ice, 0x000, vol1);
+ se200pci_WM8766_write(ice, 0x001, vol2 | 0x100);
+ break;
+ case 1:
+ se200pci_WM8766_write(ice, 0x004, vol1);
+ se200pci_WM8766_write(ice, 0x005, vol2 | 0x100);
+ break;
+ case 2:
+ se200pci_WM8766_write(ice, 0x006, vol1);
+ se200pci_WM8766_write(ice, 0x007, vol2 | 0x100);
+ break;
+ }
+}
+
+static void __devinit se200pci_WM8766_init(struct snd_ice1712 *ice)
+{
+ se200pci_WM8766_write(ice, 0x1f, 0x000); /* RESET ALL */
+ udelay(10);
+
+ se200pci_WM8766_set_volume(ice, 0, 0, 0); /* volume L=0 R=0 */
+ se200pci_WM8766_set_volume(ice, 1, 0, 0); /* volume L=0 R=0 */
+ se200pci_WM8766_set_volume(ice, 2, 0, 0); /* volume L=0 R=0 */
+
+ se200pci_WM8766_write(ice, 0x03, 0x022); /* serial mode I2S-24bits */
+ se200pci_WM8766_write(ice, 0x0a, 0x080); /* MCLK=256fs */
+ se200pci_WM8766_write(ice, 0x12, 0x000); /* MDP=0 */
+ se200pci_WM8766_write(ice, 0x15, 0x000); /* MDP=0 */
+ se200pci_WM8766_write(ice, 0x09, 0x000); /* demp=off mute=off */
+
+ se200pci_WM8766_write(ice, 0x02, 0x124); /* ch-assign L=L R=R RESET */
+ se200pci_WM8766_write(ice, 0x02, 0x120); /* ch-assign L=L R=R */
+}
+
+static void se200pci_WM8766_set_pro_rate(struct snd_ice1712 *ice,
+ unsigned int rate)
+{
+ if (rate > 96000)
+ se200pci_WM8766_write(ice, 0x0a, 0x000); /* MCLK=128fs */
+ else
+ se200pci_WM8766_write(ice, 0x0a, 0x080); /* MCLK=256fs */
+}
+
+
+/****************************************************************************/
+/* WM8776 interface */
+/****************************************************************************/
+
+static void se200pci_WM8776_write(struct snd_ice1712 *ice,
+ unsigned int addr, unsigned int data)
+{
+ unsigned int val;
+
+ val = (addr << 9) | data;
+ snd_vt1724_write_i2c(ice, 0x34, val >> 8, val & 0xff);
+}
+
+
+static void se200pci_WM8776_set_output_volume(struct snd_ice1712 *ice,
+ unsigned int vol1, unsigned int vol2)
+{
+ se200pci_WM8776_write(ice, 0x03, vol1);
+ se200pci_WM8776_write(ice, 0x04, vol2 | 0x100);
+}
+
+static void se200pci_WM8776_set_input_volume(struct snd_ice1712 *ice,
+ unsigned int vol1, unsigned int vol2)
+{
+ se200pci_WM8776_write(ice, 0x0e, vol1);
+ se200pci_WM8776_write(ice, 0x0f, vol2 | 0x100);
+}
+
+static const char *se200pci_sel[] = {
+ "LINE-IN", "CD-IN", "MIC-IN", "ALL-MIX", NULL
+};
+
+static void se200pci_WM8776_set_input_selector(struct snd_ice1712 *ice,
+ unsigned int sel)
+{
+ static unsigned char vals[] = {
+ /* LINE, CD, MIC, ALL, GND */
+ 0x10, 0x04, 0x08, 0x1c, 0x03
+ };
+ if (sel > 4)
+ sel = 4;
+ se200pci_WM8776_write(ice, 0x15, vals[sel]);
+}
+
+static void se200pci_WM8776_set_afl(struct snd_ice1712 *ice, unsigned int afl)
+{
+ /* AFL -- After Fader Listening */
+ if (afl)
+ se200pci_WM8776_write(ice, 0x16, 0x005);
+ else
+ se200pci_WM8776_write(ice, 0x16, 0x001);
+}
+
+static const char *se200pci_agc[] = {
+ "Off", "LimiterMode", "ALCMode", NULL
+};
+
+static void se200pci_WM8776_set_agc(struct snd_ice1712 *ice, unsigned int agc)
+{
+ /* AGC -- Auto Gain Control of the input */
+ switch (agc) {
+ case 0:
+ se200pci_WM8776_write(ice, 0x11, 0x000); /* Off */
+ break;
+ case 1:
+ se200pci_WM8776_write(ice, 0x10, 0x07b);
+ se200pci_WM8776_write(ice, 0x11, 0x100); /* LimiterMode */
+ break;
+ case 2:
+ se200pci_WM8776_write(ice, 0x10, 0x1fb);
+ se200pci_WM8776_write(ice, 0x11, 0x100); /* ALCMode */
+ break;
+ }
+}
+
+static void __devinit se200pci_WM8776_init(struct snd_ice1712 *ice)
+{
+ int i;
+ static unsigned short __devinitdata default_values[] = {
+ 0x100, 0x100, 0x100,
+ 0x100, 0x100, 0x100,
+ 0x000, 0x090, 0x000, 0x000,
+ 0x022, 0x022, 0x022,
+ 0x008, 0x0cf, 0x0cf, 0x07b, 0x000,
+ 0x032, 0x000, 0x0a6, 0x001, 0x001
+ };
+
+ se200pci_WM8776_write(ice, 0x17, 0x000); /* reset all */
+ /* ADC and DAC interface is I2S 24bits mode */
+ /* The sample-rate are automatically changed */
+ udelay(10);
+ /* BUT my board can not do reset all, so I load all by manually. */
+ for (i = 0; i < ARRAY_SIZE(default_values); i++)
+ se200pci_WM8776_write(ice, i, default_values[i]);
+
+ se200pci_WM8776_set_input_selector(ice, 0);
+ se200pci_WM8776_set_afl(ice, 0);
+ se200pci_WM8776_set_agc(ice, 0);
+ se200pci_WM8776_set_input_volume(ice, 0, 0);
+ se200pci_WM8776_set_output_volume(ice, 0, 0);
+
+ /* head phone mute and power down */
+ se200pci_WM8776_write(ice, 0x00, 0);
+ se200pci_WM8776_write(ice, 0x01, 0);
+ se200pci_WM8776_write(ice, 0x02, 0x100);
+ se200pci_WM8776_write(ice, 0x0d, 0x080);
+}
+
+static void se200pci_WM8776_set_pro_rate(struct snd_ice1712 *ice,
+ unsigned int rate)
+{
+ /* nothing to do */
+}
+
+
+/****************************************************************************/
+/* runtime interface */
+/****************************************************************************/
+
+static void se200pci_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate)
+{
+ se200pci_WM8740_set_pro_rate(ice, rate);
+ se200pci_WM8766_set_pro_rate(ice, rate);
+ se200pci_WM8776_set_pro_rate(ice, rate);
+}
+
+struct se200pci_control {
+ char *name;
+ enum {
+ WM8766,
+ WM8776in,
+ WM8776out,
+ WM8776sel,
+ WM8776agc,
+ WM8776afl
+ } target;
+ enum { VOLUME1, VOLUME2, BOOLEAN, ENUM } type;
+ int ch;
+ const char **member;
+ const char *comment;
+};
+
+static const struct se200pci_control se200pci_cont[] = {
+ {
+ .name = "Front Playback Volume",
+ .target = WM8776out,
+ .type = VOLUME1,
+ .comment = "Front(green)"
+ },
+ {
+ .name = "Side Playback Volume",
+ .target = WM8766,
+ .type = VOLUME1,
+ .ch = 1,
+ .comment = "Surround(orange)"
+ },
+ {
+ .name = "Surround Playback Volume",
+ .target = WM8766,
+ .type = VOLUME1,
+ .ch = 2,
+ .comment = "SurroundBack(white)"
+ },
+ {
+ .name = "CLFE Playback Volume",
+ .target = WM8766,
+ .type = VOLUME1,
+ .ch = 0,
+ .comment = "Center(Lch)&SubWoofer(Rch)(black)"
+ },
+ {
+ .name = "Capture Volume",
+ .target = WM8776in,
+ .type = VOLUME2
+ },
+ {
+ .name = "Capture Select",
+ .target = WM8776sel,
+ .type = ENUM,
+ .member = se200pci_sel
+ },
+ {
+ .name = "AGC Capture Mode",
+ .target = WM8776agc,
+ .type = ENUM,
+ .member = se200pci_agc
+ },
+ {
+ .name = "AFL Bypass Playback Switch",
+ .target = WM8776afl,
+ .type = BOOLEAN
+ }
+};
+
+static int se200pci_get_enum_count(int n)
+{
+ const char **member;
+ int c;
+
+ member = se200pci_cont[n].member;
+ if (!member)
+ return 0;
+ for (c = 0; member[c]; c++)
+ ;
+ return c;
+}
+
+static int se200pci_cont_volume_info(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* mute */
+ uinfo->value.integer.max = 0xff; /* 0dB */
+ return 0;
+}
+
+#define se200pci_cont_boolean_info snd_ctl_boolean_mono_info
+
+static int se200pci_cont_enum_info(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int n, c;
+
+ n = kc->private_value;
+ c = se200pci_get_enum_count(n);
+ if (!c)
+ return -EINVAL;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = c;
+ if (uinfo->value.enumerated.item >= c)
+ uinfo->value.enumerated.item = c - 1;
+ strcpy(uinfo->value.enumerated.name,
+ se200pci_cont[n].member[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int se200pci_cont_volume_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *uc)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kc);
+ struct se_spec *spec = ice->spec;
+ int n = kc->private_value;
+ uc->value.integer.value[0] = spec->vol[n].ch1;
+ uc->value.integer.value[1] = spec->vol[n].ch2;
+ return 0;
+}
+
+static int se200pci_cont_boolean_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *uc)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kc);
+ struct se_spec *spec = ice->spec;
+ int n = kc->private_value;
+ uc->value.integer.value[0] = spec->vol[n].ch1;
+ return 0;
+}
+
+static int se200pci_cont_enum_get(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *uc)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kc);
+ struct se_spec *spec = ice->spec;
+ int n = kc->private_value;
+ uc->value.enumerated.item[0] = spec->vol[n].ch1;
+ return 0;
+}
+
+static void se200pci_cont_update(struct snd_ice1712 *ice, int n)
+{
+ struct se_spec *spec = ice->spec;
+ switch (se200pci_cont[n].target) {
+ case WM8766:
+ se200pci_WM8766_set_volume(ice,
+ se200pci_cont[n].ch,
+ spec->vol[n].ch1,
+ spec->vol[n].ch2);
+ break;
+
+ case WM8776in:
+ se200pci_WM8776_set_input_volume(ice,
+ spec->vol[n].ch1,
+ spec->vol[n].ch2);
+ break;
+
+ case WM8776out:
+ se200pci_WM8776_set_output_volume(ice,
+ spec->vol[n].ch1,
+ spec->vol[n].ch2);
+ break;
+
+ case WM8776sel:
+ se200pci_WM8776_set_input_selector(ice,
+ spec->vol[n].ch1);
+ break;
+
+ case WM8776agc:
+ se200pci_WM8776_set_agc(ice, spec->vol[n].ch1);
+ break;
+
+ case WM8776afl:
+ se200pci_WM8776_set_afl(ice, spec->vol[n].ch1);
+ break;
+
+ default:
+ break;
+ }
+}
+
+static int se200pci_cont_volume_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *uc)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kc);
+ struct se_spec *spec = ice->spec;
+ int n = kc->private_value;
+ unsigned int vol1, vol2;
+ int changed;
+
+ changed = 0;
+ vol1 = uc->value.integer.value[0] & 0xff;
+ vol2 = uc->value.integer.value[1] & 0xff;
+ if (spec->vol[n].ch1 != vol1) {
+ spec->vol[n].ch1 = vol1;
+ changed = 1;
+ }
+ if (spec->vol[n].ch2 != vol2) {
+ spec->vol[n].ch2 = vol2;
+ changed = 1;
+ }
+ if (changed)
+ se200pci_cont_update(ice, n);
+
+ return changed;
+}
+
+static int se200pci_cont_boolean_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *uc)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kc);
+ struct se_spec *spec = ice->spec;
+ int n = kc->private_value;
+ unsigned int vol1;
+
+ vol1 = !!uc->value.integer.value[0];
+ if (spec->vol[n].ch1 != vol1) {
+ spec->vol[n].ch1 = vol1;
+ se200pci_cont_update(ice, n);
+ return 1;
+ }
+ return 0;
+}
+
+static int se200pci_cont_enum_put(struct snd_kcontrol *kc,
+ struct snd_ctl_elem_value *uc)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kc);
+ struct se_spec *spec = ice->spec;
+ int n = kc->private_value;
+ unsigned int vol1;
+
+ vol1 = uc->value.enumerated.item[0];
+ if (vol1 >= se200pci_get_enum_count(n))
+ return -EINVAL;
+ if (spec->vol[n].ch1 != vol1) {
+ spec->vol[n].ch1 = vol1;
+ se200pci_cont_update(ice, n);
+ return 1;
+ }
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_gain1, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_gain2, -10350, 50, 1);
+
+static int __devinit se200pci_add_controls(struct snd_ice1712 *ice)
+{
+ int i;
+ struct snd_kcontrol_new cont;
+ int err;
+
+ memset(&cont, 0, sizeof(cont));
+ cont.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ for (i = 0; i < ARRAY_SIZE(se200pci_cont); i++) {
+ cont.private_value = i;
+ cont.name = se200pci_cont[i].name;
+ cont.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ cont.tlv.p = NULL;
+ switch (se200pci_cont[i].type) {
+ case VOLUME1:
+ case VOLUME2:
+ cont.info = se200pci_cont_volume_info;
+ cont.get = se200pci_cont_volume_get;
+ cont.put = se200pci_cont_volume_put;
+ cont.access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ if (se200pci_cont[i].type == VOLUME1)
+ cont.tlv.p = db_scale_gain1;
+ else
+ cont.tlv.p = db_scale_gain2;
+ break;
+ case BOOLEAN:
+ cont.info = se200pci_cont_boolean_info;
+ cont.get = se200pci_cont_boolean_get;
+ cont.put = se200pci_cont_boolean_put;
+ break;
+ case ENUM:
+ cont.info = se200pci_cont_enum_info;
+ cont.get = se200pci_cont_enum_get;
+ cont.put = se200pci_cont_enum_put;
+ break;
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&cont, ice));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+
+/****************************************************************************/
+/* ONKYO WAVIO SE-90PCI */
+/****************************************************************************/
+/*
+ * system configuration ICE_EEP2_SYSCONF=0x4b
+ * AC-Link configuration ICE_EEP2_ACLINK=0x80
+ * I2S converters feature ICE_EEP2_I2S=0x78
+ * S/PDIF configuration ICE_EEP2_SPDIF=0xc3
+ *
+ * ** connected chip **
+ *
+ * WM8716
+ * A 2ch-DAC of main outputs.
+ * It setuped as I2S mode by wire, so no way to setup from software.
+ * ML/I2S (28pin) -- +5V
+ * MC/DM1 (27pin) -- GND
+ * MC/DM0 (26pin) -- GND
+ * MUTEB (25pin) -- open (internal pull-up)
+ * MODE (24pin) -- GND
+ * CSBIWO (23pin) -- +5V
+ *
+ */
+
+ /* Nothing to do for this chip. */
+
+
+/****************************************************************************/
+/* probe/initialize/setup */
+/****************************************************************************/
+
+static int __devinit se_init(struct snd_ice1712 *ice)
+{
+ struct se_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ ice->spec = spec;
+
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_SE90PCI) {
+ ice->num_total_dacs = 2;
+ ice->num_total_adcs = 0;
+ ice->vt1720 = 1;
+ return 0;
+
+ } else if (ice->eeprom.subvendor == VT1724_SUBDEVICE_SE200PCI) {
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 2;
+ se200pci_WM8740_init(ice);
+ se200pci_WM8766_init(ice);
+ se200pci_WM8776_init(ice);
+ ice->gpio.set_pro_rate = se200pci_set_pro_rate;
+ return 0;
+ }
+
+ return -ENOENT;
+}
+
+static int __devinit se_add_controls(struct snd_ice1712 *ice)
+{
+ int err;
+
+ err = 0;
+ /* nothing to do for VT1724_SUBDEVICE_SE90PCI */
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_SE200PCI)
+ err = se200pci_add_controls(ice);
+
+ return err;
+}
+
+
+/****************************************************************************/
+/* entry point */
+/****************************************************************************/
+
+static unsigned char se200pci_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x4b, /* 49.152Hz, spdif-in/ADC, 4DACs */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0x78, /* 96k-ok, 24bit, 192k-ok */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+
+ [ICE_EEP2_GPIO_DIR] = 0x02, /* WM8766 mute 1=output */
+ [ICE_EEP2_GPIO_DIR1] = 0x00, /* not used */
+ [ICE_EEP2_GPIO_DIR2] = 0x07, /* WM8766 ML/MC/MD 1=output */
+
+ [ICE_EEP2_GPIO_MASK] = 0x00, /* 0=writable */
+ [ICE_EEP2_GPIO_MASK1] = 0x00, /* 0=writable */
+ [ICE_EEP2_GPIO_MASK2] = 0x00, /* 0=writable */
+
+ [ICE_EEP2_GPIO_STATE] = 0x00, /* WM8766 mute=0 */
+ [ICE_EEP2_GPIO_STATE1] = 0x00, /* not used */
+ [ICE_EEP2_GPIO_STATE2] = 0x07, /* WM8766 ML/MC/MD */
+};
+
+static unsigned char se90pci_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x4b, /* 49.152Hz, spdif-in/ADC, 4DACs */
+ [ICE_EEP2_ACLINK] = 0x80, /* I2S */
+ [ICE_EEP2_I2S] = 0x78, /* 96k-ok, 24bit, 192k-ok */
+ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */
+
+ /* ALL GPIO bits are in input mode */
+};
+
+struct snd_ice1712_card_info snd_vt1724_se_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_SE200PCI,
+ .name = "ONKYO SE200PCI",
+ .model = "se200pci",
+ .chip_init = se_init,
+ .build_controls = se_add_controls,
+ .eeprom_size = sizeof(se200pci_eeprom),
+ .eeprom_data = se200pci_eeprom,
+ },
+ {
+ .subvendor = VT1724_SUBDEVICE_SE90PCI,
+ .name = "ONKYO SE90PCI",
+ .model = "se90pci",
+ .chip_init = se_init,
+ .build_controls = se_add_controls,
+ .eeprom_size = sizeof(se90pci_eeprom),
+ .eeprom_data = se90pci_eeprom,
+ },
+ {} /*terminator*/
+};
diff --git a/sound/pci/ice1712/se.h b/sound/pci/ice1712/se.h
new file mode 100644
index 0000000..0b0a9da
--- /dev/null
+++ b/sound/pci/ice1712/se.h
@@ -0,0 +1,15 @@
+#ifndef __SOUND_SE_H
+#define __SOUND_SE_H
+
+/* ID */
+#define SE_DEVICE_DESC \
+ "{ONKYO INC,SE-90PCI},"\
+ "{ONKYO INC,SE-200PCI},"
+
+#define VT1724_SUBDEVICE_SE90PCI 0xb161000
+#define VT1724_SUBDEVICE_SE200PCI 0xb160100
+
+/* entry struct */
+extern struct snd_ice1712_card_info snd_vt1724_se_cards[];
+
+#endif /* __SOUND_SE_H */
diff --git a/sound/pci/ice1712/stac946x.h b/sound/pci/ice1712/stac946x.h
new file mode 100644
index 0000000..5b39095
--- /dev/null
+++ b/sound/pci/ice1712/stac946x.h
@@ -0,0 +1,25 @@
+#ifndef __SOUND_STAC946X_H
+#define __SOUND_STAC946X_H
+
+#define STAC946X_RESET 0x00
+#define STAC946X_STATUS 0x01
+#define STAC946X_MASTER_VOLUME 0x02
+#define STAC946X_LF_VOLUME 0x03
+#define STAC946X_RF_VOLUME 0x04
+#define STAC946X_LR_VOLUME 0x05
+#define STAC946X_RR_VOLUME 0x06
+#define STAC946X_CENTER_VOLUME 0x07
+#define STAC946X_LFE_VOLUME 0x08
+#define STAC946X_MIC_L_VOLUME 0x09
+#define STAC946X_MIC_R_VOLUME 0x0a
+#define STAC946X_DEEMPHASIS 0x0c
+#define STAC946X_GENERAL_PURPOSE 0x0d
+#define STAC946X_AUDIO_PORT_CONTROL 0x0e
+#define STAC946X_MASTER_CLOCKING 0x0f
+#define STAC946X_POWERDOWN_CTRL1 0x10
+#define STAC946X_POWERDOWN_CTRL2 0x11
+#define STAC946X_REVISION_CODE 0x12
+#define STAC946X_ADDRESS_CONTROL 0x13
+#define STAC946X_ADDRESS 0x14
+
+#endif /* __SOUND_STAC946X_H */
diff --git a/sound/pci/ice1712/vt1720_mobo.c b/sound/pci/ice1712/vt1720_mobo.c
new file mode 100644
index 0000000..7f9674b
--- /dev/null
+++ b/sound/pci/ice1712/vt1720_mobo.c
@@ -0,0 +1,140 @@
+/*
+ * ALSA driver for VT1720/VT1724 (Envy24PT/Envy24HT)
+ *
+ * Lowlevel functions for VT1720-based motherboards
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "vt1720_mobo.h"
+
+
+static int __devinit k8x800_init(struct snd_ice1712 *ice)
+{
+ ice->vt1720 = 1;
+
+ /* VT1616 codec */
+ ice->num_total_dacs = 6;
+ ice->num_total_adcs = 2;
+
+ /* WM8728 codec */
+ /* FIXME: TODO */
+
+ return 0;
+}
+
+static int __devinit k8x800_add_controls(struct snd_ice1712 *ice)
+{
+ /* FIXME: needs some quirks for VT1616? */
+ return 0;
+}
+
+/* EEPROM image */
+
+static unsigned char k8x800_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x01, /* clock 256, 1ADC, 2DACs */
+ [ICE_EEP2_ACLINK] = 0x02, /* ACLINK, packed */
+ [ICE_EEP2_I2S] = 0x00, /* - */
+ [ICE_EEP2_SPDIF] = 0x00, /* - */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x00, /* - */
+ [ICE_EEP2_GPIO_MASK] = 0xff,
+ [ICE_EEP2_GPIO_MASK1] = 0xff,
+ [ICE_EEP2_GPIO_MASK2] = 0x00, /* - */
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00, /* - */
+};
+
+static unsigned char sn25p_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x01, /* clock 256, 1ADC, 2DACs */
+ [ICE_EEP2_ACLINK] = 0x02, /* ACLINK, packed */
+ [ICE_EEP2_I2S] = 0x00, /* - */
+ [ICE_EEP2_SPDIF] = 0x41, /* - */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0x00, /* - */
+ [ICE_EEP2_GPIO_MASK] = 0xff,
+ [ICE_EEP2_GPIO_MASK1] = 0xff,
+ [ICE_EEP2_GPIO_MASK2] = 0x00, /* - */
+ [ICE_EEP2_GPIO_STATE] = 0x00,
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00, /* - */
+};
+
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1720_mobo_cards[] __devinitdata = {
+ {
+ .subvendor = VT1720_SUBDEVICE_K8X800,
+ .name = "Albatron K8X800 Pro II",
+ .model = "k8x800",
+ .chip_init = k8x800_init,
+ .build_controls = k8x800_add_controls,
+ .eeprom_size = sizeof(k8x800_eeprom),
+ .eeprom_data = k8x800_eeprom,
+ },
+ {
+ .subvendor = VT1720_SUBDEVICE_ZNF3_150,
+ .name = "Chaintech ZNF3-150",
+ /* identical with k8x800 */
+ .chip_init = k8x800_init,
+ .build_controls = k8x800_add_controls,
+ .eeprom_size = sizeof(k8x800_eeprom),
+ .eeprom_data = k8x800_eeprom,
+ },
+ {
+ .subvendor = VT1720_SUBDEVICE_ZNF3_250,
+ .name = "Chaintech ZNF3-250",
+ /* identical with k8x800 */
+ .chip_init = k8x800_init,
+ .build_controls = k8x800_add_controls,
+ .eeprom_size = sizeof(k8x800_eeprom),
+ .eeprom_data = k8x800_eeprom,
+ },
+ {
+ .subvendor = VT1720_SUBDEVICE_9CJS,
+ .name = "Chaintech 9CJS",
+ /* identical with k8x800 */
+ .chip_init = k8x800_init,
+ .build_controls = k8x800_add_controls,
+ .eeprom_size = sizeof(k8x800_eeprom),
+ .eeprom_data = k8x800_eeprom,
+ },
+ {
+ .subvendor = VT1720_SUBDEVICE_SN25P,
+ .name = "Shuttle SN25P",
+ .model = "sn25p",
+ .chip_init = k8x800_init,
+ .build_controls = k8x800_add_controls,
+ .eeprom_size = sizeof(k8x800_eeprom),
+ .eeprom_data = sn25p_eeprom,
+ },
+ { } /* terminator */
+};
+
diff --git a/sound/pci/ice1712/vt1720_mobo.h b/sound/pci/ice1712/vt1720_mobo.h
new file mode 100644
index 0000000..0b1b0ee
--- /dev/null
+++ b/sound/pci/ice1712/vt1720_mobo.h
@@ -0,0 +1,41 @@
+#ifndef __SOUND_VT1720_MOBO_H
+#define __SOUND_VT1720_MOBO_H
+
+/*
+ * ALSA driver for VT1720/VT1724 (Envy24PT/Envy24HT)
+ *
+ * Lowlevel functions for VT1720-based motherboards
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#define VT1720_MOBO_DEVICE_DESC "{Albatron,K8X800 Pro II},"\
+ "{Chaintech,ZNF3-150},"\
+ "{Chaintech,ZNF3-250},"\
+ "{Chaintech,9CJS},"\
+ "{Shuttle,SN25P},"
+
+#define VT1720_SUBDEVICE_K8X800 0xf217052c
+#define VT1720_SUBDEVICE_ZNF3_150 0x0f2741f6
+#define VT1720_SUBDEVICE_ZNF3_250 0x0f2745f6
+#define VT1720_SUBDEVICE_9CJS 0x0f272327
+#define VT1720_SUBDEVICE_SN25P 0x97123650
+
+extern struct snd_ice1712_card_info snd_vt1720_mobo_cards[];
+
+#endif /* __SOUND_VT1720_MOBO_H */
diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c
new file mode 100644
index 0000000..5af9e84
--- /dev/null
+++ b/sound/pci/ice1712/wtm.c
@@ -0,0 +1,518 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for Ego Sys Waveterminal 192M
+ *
+ * Copyright (c) 2006 Guedez Clement <klem.dev@gmail.com>
+ * Some functions are taken from the Prodigy192 driver
+ * source
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+
+
+#include <linux/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "wtm.h"
+#include "stac946x.h"
+
+
+/*
+ * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus
+ */
+static inline void stac9460_put(struct snd_ice1712 *ice, int reg,
+ unsigned char val)
+{
+ snd_vt1724_write_i2c(ice, STAC9460_I2C_ADDR, reg, val);
+}
+
+static inline unsigned char stac9460_get(struct snd_ice1712 *ice, int reg)
+{
+ return snd_vt1724_read_i2c(ice, STAC9460_I2C_ADDR, reg);
+}
+
+/*
+ * 2*ADC 2*DAC no2 ringbuffer r/w on i2c bus
+ */
+static inline void stac9460_2_put(struct snd_ice1712 *ice, int reg,
+ unsigned char val)
+{
+ snd_vt1724_write_i2c(ice, STAC9460_2_I2C_ADDR, reg, val);
+}
+
+static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg)
+{
+ return snd_vt1724_read_i2c(ice, STAC9460_2_I2C_ADDR, reg);
+}
+
+
+/*
+ * DAC mute control
+ */
+#define stac9460_dac_mute_info snd_ctl_boolean_mono_info
+
+static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int idx, id;
+
+ if (kcontrol->private_value) {
+ idx = STAC946X_MASTER_VOLUME;
+ id = 0;
+ } else {
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ idx = id + STAC946X_LF_VOLUME;
+ }
+ if (id < 6)
+ val = stac9460_get(ice, idx);
+ else
+ val = stac9460_2_get(ice, idx - 6);
+ ucontrol->value.integer.value[0] = (~val >> 7) & 0x1;
+ return 0;
+}
+
+static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char new, old;
+ int id, idx;
+ int change;
+
+ if (kcontrol->private_value) {
+ idx = STAC946X_MASTER_VOLUME;
+ old = stac9460_get(ice, idx);
+ new = (~ucontrol->value.integer.value[0] << 7 & 0x80) |
+ (old & ~0x80);
+ change = (new != old);
+ if (change) {
+ stac9460_put(ice, idx, new);
+ stac9460_2_put(ice, idx, new);
+ }
+ } else {
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ idx = id + STAC946X_LF_VOLUME;
+ if (id < 6)
+ old = stac9460_get(ice, idx);
+ else
+ old = stac9460_2_get(ice, idx - 6);
+ new = (~ucontrol->value.integer.value[0] << 7 & 0x80) |
+ (old & ~0x80);
+ change = (new != old);
+ if (change) {
+ if (id < 6)
+ stac9460_put(ice, idx, new);
+ else
+ stac9460_2_put(ice, idx - 6, new);
+ }
+ }
+ return change;
+}
+
+/*
+ * DAC volume attenuation mixer control
+ */
+static int stac9460_dac_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0; /* mute */
+ uinfo->value.integer.max = 0x7f; /* 0dB */
+ return 0;
+}
+
+static int stac9460_dac_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx, id;
+ unsigned char vol;
+
+ if (kcontrol->private_value) {
+ idx = STAC946X_MASTER_VOLUME;
+ id = 0;
+ } else {
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ idx = id + STAC946X_LF_VOLUME;
+ }
+ if (id < 6)
+ vol = stac9460_get(ice, idx) & 0x7f;
+ else
+ vol = stac9460_2_get(ice, idx - 6) & 0x7f;
+ ucontrol->value.integer.value[0] = 0x7f - vol;
+ return 0;
+}
+
+static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int idx, id;
+ unsigned char tmp, ovol, nvol;
+ int change;
+
+ if (kcontrol->private_value) {
+ idx = STAC946X_MASTER_VOLUME;
+ nvol = ucontrol->value.integer.value[0] & 0x7f;
+ tmp = stac9460_get(ice, idx);
+ ovol = 0x7f - (tmp & 0x7f);
+ change = (ovol != nvol);
+ if (change) {
+ stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80));
+ stac9460_2_put(ice, idx, (0x7f - nvol) | (tmp & 0x80));
+ }
+ } else {
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ idx = id + STAC946X_LF_VOLUME;
+ nvol = ucontrol->value.integer.value[0] & 0x7f;
+ if (id < 6)
+ tmp = stac9460_get(ice, idx);
+ else
+ tmp = stac9460_2_get(ice, idx - 6);
+ ovol = 0x7f - (tmp & 0x7f);
+ change = (ovol != nvol);
+ if (change) {
+ if (id < 6)
+ stac9460_put(ice, idx, (0x7f - nvol) |
+ (tmp & 0x80));
+ else
+ stac9460_2_put(ice, idx-6, (0x7f - nvol) |
+ (tmp & 0x80));
+ }
+ }
+ return change;
+}
+
+/*
+ * ADC mute control
+ */
+#define stac9460_adc_mute_info snd_ctl_boolean_stereo_info
+
+static int stac9460_adc_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int i, id;
+
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ if (id == 0) {
+ for (i = 0; i < 2; ++i) {
+ val = stac9460_get(ice, STAC946X_MIC_L_VOLUME + i);
+ ucontrol->value.integer.value[i] = ~val>>7 & 0x1;
+ }
+ } else {
+ for (i = 0; i < 2; ++i) {
+ val = stac9460_2_get(ice, STAC946X_MIC_L_VOLUME + i);
+ ucontrol->value.integer.value[i] = ~val>>7 & 0x1;
+ }
+ }
+ return 0;
+}
+
+static int stac9460_adc_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char new, old;
+ int i, reg, id;
+ int change;
+
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ if (id == 0) {
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ old = stac9460_get(ice, reg);
+ new = (~ucontrol->value.integer.value[i]<<7&0x80) |
+ (old&~0x80);
+ change = (new != old);
+ if (change)
+ stac9460_put(ice, reg, new);
+ }
+ } else {
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ old = stac9460_2_get(ice, reg);
+ new = (~ucontrol->value.integer.value[i]<<7&0x80) |
+ (old&~0x80);
+ change = (new != old);
+ if (change)
+ stac9460_2_put(ice, reg, new);
+ }
+ }
+ return change;
+}
+
+/*
+ *ADC gain mixer control
+ */
+static int stac9460_adc_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0; /* 0dB */
+ uinfo->value.integer.max = 0x0f; /* 22.5dB */
+ return 0;
+}
+
+static int stac9460_adc_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int i, reg, id;
+ unsigned char vol;
+
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ if (id == 0) {
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ vol = stac9460_get(ice, reg) & 0x0f;
+ ucontrol->value.integer.value[i] = 0x0f - vol;
+ }
+ } else {
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ vol = stac9460_2_get(ice, reg) & 0x0f;
+ ucontrol->value.integer.value[i] = 0x0f - vol;
+ }
+ }
+ return 0;
+}
+
+static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ int i, reg, id;
+ unsigned char ovol, nvol;
+ int change;
+
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ if (id == 0) {
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ nvol = ucontrol->value.integer.value[i] & 0x0f;
+ ovol = 0x0f - stac9460_get(ice, reg);
+ change = ((ovol & 0x0f) != nvol);
+ if (change)
+ stac9460_put(ice, reg, (0x0f - nvol) |
+ (ovol & ~0x0f));
+ }
+ } else {
+ for (i = 0; i < 2; ++i) {
+ reg = STAC946X_MIC_L_VOLUME + i;
+ nvol = ucontrol->value.integer.value[i] & 0x0f;
+ ovol = 0x0f - stac9460_2_get(ice, reg);
+ change = ((ovol & 0x0f) != nvol);
+ if (change)
+ stac9460_2_put(ice, reg, (0x0f - nvol) |
+ (ovol & ~0x0f));
+ }
+ }
+ return change;
+}
+
+/*
+ * MIC / LINE switch fonction
+ */
+
+#define stac9460_mic_sw_info snd_ctl_boolean_mono_info
+
+static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int id;
+
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ if (id == 0)
+ val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE);
+ else
+ val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE);
+ ucontrol->value.integer.value[0] = ~val>>7 & 0x1;
+ return 0;
+}
+
+static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
+ unsigned char new, old;
+ int change, id;
+
+ id = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ if (id == 0)
+ old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE);
+ else
+ old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE);
+ new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80);
+ change = (new != old);
+ if (change) {
+ if (id == 0)
+ stac9460_put(ice, STAC946X_GENERAL_PURPOSE, new);
+ else
+ stac9460_2_put(ice, STAC946X_GENERAL_PURPOSE, new);
+ }
+ return change;
+}
+
+/*
+ * Control tabs
+ */
+static struct snd_kcontrol_new stac9640_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = stac9460_dac_mute_info,
+ .get = stac9460_dac_mute_get,
+ .put = stac9460_dac_mute_put,
+ .private_value = 1
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .info = stac9460_dac_vol_info,
+ .get = stac9460_dac_vol_get,
+ .put = stac9460_dac_vol_put,
+ .private_value = 1,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "MIC/Line switch",
+ .count = 2,
+ .info = stac9460_mic_sw_info,
+ .get = stac9460_mic_sw_get,
+ .put = stac9460_mic_sw_put,
+
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Switch",
+ .count = 8,
+ .info = stac9460_dac_mute_info,
+ .get = stac9460_dac_mute_get,
+ .put = stac9460_dac_mute_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Volume",
+ .count = 8,
+ .info = stac9460_dac_vol_info,
+ .get = stac9460_dac_vol_get,
+ .put = stac9460_dac_vol_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Switch",
+ .count = 2,
+ .info = stac9460_adc_mute_info,
+ .get = stac9460_adc_mute_get,
+ .put = stac9460_adc_mute_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Volume",
+ .count = 2,
+ .info = stac9460_adc_vol_info,
+ .get = stac9460_adc_vol_get,
+ .put = stac9460_adc_vol_put,
+
+ }
+};
+
+
+
+/*INIT*/
+static int __devinit wtm_add_controls(struct snd_ice1712 *ice)
+{
+ unsigned int i;
+ int err;
+
+ for (i = 0; i < ARRAY_SIZE(stac9640_controls); i++) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&stac9640_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int __devinit wtm_init(struct snd_ice1712 *ice)
+{
+ static unsigned short stac_inits_prodigy[] = {
+ STAC946X_RESET, 0,
+ (unsigned short)-1
+ };
+ unsigned short *p;
+
+ /*WTM 192M*/
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 4;
+ ice->force_rdma1 = 1;
+
+ /*initialize codec*/
+ p = stac_inits_prodigy;
+ for (; *p != (unsigned short)-1; p += 2) {
+ stac9460_put(ice, p[0], p[1]);
+ stac9460_2_put(ice, p[0], p[1]);
+ }
+ return 0;
+}
+
+
+static unsigned char wtm_eeprom[] __devinitdata = {
+ 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */
+ 0x80, /* ACLINK : I2S */
+ 0xf8, /* I2S: vol; 96k, 24bit, 192k */
+ 0xc1 /*SPDIF: out-en, spidf ext out*/,
+ 0x9f, /* GPIO_DIR */
+ 0xff, /* GPIO_DIR1 */
+ 0x7f, /* GPIO_DIR2 */
+ 0x9f, /* GPIO_MASK */
+ 0xff, /* GPIO_MASK1 */
+ 0x7f, /* GPIO_MASK2 */
+ 0x16, /* GPIO_STATE */
+ 0x80, /* GPIO_STATE1 */
+ 0x00, /* GPIO_STATE2 */
+};
+
+
+/*entry point*/
+struct snd_ice1712_card_info snd_vt1724_wtm_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_WTM,
+ .name = "ESI Waveterminal 192M",
+ .model = "WT192M",
+ .chip_init = wtm_init,
+ .build_controls = wtm_add_controls,
+ .eeprom_size = sizeof(wtm_eeprom),
+ .eeprom_data = wtm_eeprom,
+ },
+ {} /*terminator*/
+};
diff --git a/sound/pci/ice1712/wtm.h b/sound/pci/ice1712/wtm.h
new file mode 100644
index 0000000..423c1a2
--- /dev/null
+++ b/sound/pci/ice1712/wtm.h
@@ -0,0 +1,20 @@
+#ifndef __SOUND_WTM_H
+#define __SOUND_WTM_H
+
+/* ID */
+#define WTM_DEVICE_DESC "{EGO SYS INC,WaveTerminal 192M},"
+#define VT1724_SUBDEVICE_WTM 0x36495345 /* WT192M ver1.0 */
+
+/*
+ *chip addresses on I2C bus
+ */
+
+#define AK4114_ADDR 0x20 /*S/PDIF receiver*/
+#define STAC9460_I2C_ADDR 0x54 /* ADC*2 | DAC*6 */
+#define STAC9460_2_I2C_ADDR 0x56 /* ADC|DAC *2 */
+
+
+extern struct snd_ice1712_card_info snd_vt1724_wtm_cards[];
+
+#endif /* __SOUND_WTM_H */
+
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
new file mode 100644
index 0000000..19d3391
--- /dev/null
+++ b/sound/pci/intel8x0.c
@@ -0,0 +1,3165 @@
+/*
+ * ALSA driver for Intel ICH (i8x0) chipsets
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This code also contains alpha support for SiS 735 chipsets provided
+ * by Mike Pieper <mptei@users.sourceforge.net>. We have no datasheet
+ * for SiS735, so the code is not fully functional.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/initval.h>
+/* for 440MX workaround */
+#include <asm/pgtable.h>
+#include <asm/cacheflush.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH},"
+ "{Intel,82901AB-ICH0},"
+ "{Intel,82801BA-ICH2},"
+ "{Intel,82801CA-ICH3},"
+ "{Intel,82801DB-ICH4},"
+ "{Intel,ICH5},"
+ "{Intel,ICH6},"
+ "{Intel,ICH7},"
+ "{Intel,6300ESB},"
+ "{Intel,ESB2},"
+ "{Intel,MX440},"
+ "{SiS,SI7012},"
+ "{NVidia,nForce Audio},"
+ "{NVidia,nForce2 Audio},"
+ "{NVidia,nForce3 Audio},"
+ "{NVidia,MCP04},"
+ "{NVidia,MCP501},"
+ "{NVidia,CK804},"
+ "{NVidia,CK8},"
+ "{NVidia,CK8S},"
+ "{AMD,AMD768},"
+ "{AMD,AMD8111},"
+ "{ALI,M5455}}");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int ac97_clock;
+static char *ac97_quirk;
+static int buggy_semaphore;
+static int buggy_irq = -1; /* auto-check */
+static int xbox;
+static int spdif_aclink = -1;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for Intel i8x0 soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for Intel i8x0 soundcard.");
+module_param(ac97_clock, int, 0444);
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = whitelist + auto-detect, 1 = force autodetect).");
+module_param(ac97_quirk, charp, 0444);
+MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
+module_param(buggy_semaphore, bool, 0444);
+MODULE_PARM_DESC(buggy_semaphore, "Enable workaround for hardwares with problematic codec semaphores.");
+module_param(buggy_irq, bool, 0444);
+MODULE_PARM_DESC(buggy_irq, "Enable workaround for buggy interrupts on some motherboards.");
+module_param(xbox, bool, 0444);
+MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection.");
+module_param(spdif_aclink, int, 0444);
+MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link.");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+static int joystick;
+module_param(joystick, int, 0444);
+
+/*
+ * Direct registers
+ */
+enum { DEVICE_INTEL, DEVICE_INTEL_ICH4, DEVICE_SIS, DEVICE_ALI, DEVICE_NFORCE };
+
+#define ICHREG(x) ICH_REG_##x
+
+#define DEFINE_REGSET(name,base) \
+enum { \
+ ICH_REG_##name##_BDBAR = base + 0x0, /* dword - buffer descriptor list base address */ \
+ ICH_REG_##name##_CIV = base + 0x04, /* byte - current index value */ \
+ ICH_REG_##name##_LVI = base + 0x05, /* byte - last valid index */ \
+ ICH_REG_##name##_SR = base + 0x06, /* byte - status register */ \
+ ICH_REG_##name##_PICB = base + 0x08, /* word - position in current buffer */ \
+ ICH_REG_##name##_PIV = base + 0x0a, /* byte - prefetched index value */ \
+ ICH_REG_##name##_CR = base + 0x0b, /* byte - control register */ \
+};
+
+/* busmaster blocks */
+DEFINE_REGSET(OFF, 0); /* offset */
+DEFINE_REGSET(PI, 0x00); /* PCM in */
+DEFINE_REGSET(PO, 0x10); /* PCM out */
+DEFINE_REGSET(MC, 0x20); /* Mic in */
+
+/* ICH4 busmaster blocks */
+DEFINE_REGSET(MC2, 0x40); /* Mic in 2 */
+DEFINE_REGSET(PI2, 0x50); /* PCM in 2 */
+DEFINE_REGSET(SP, 0x60); /* SPDIF out */
+
+/* values for each busmaster block */
+
+/* LVI */
+#define ICH_REG_LVI_MASK 0x1f
+
+/* SR */
+#define ICH_FIFOE 0x10 /* FIFO error */
+#define ICH_BCIS 0x08 /* buffer completion interrupt status */
+#define ICH_LVBCI 0x04 /* last valid buffer completion interrupt */
+#define ICH_CELV 0x02 /* current equals last valid */
+#define ICH_DCH 0x01 /* DMA controller halted */
+
+/* PIV */
+#define ICH_REG_PIV_MASK 0x1f /* mask */
+
+/* CR */
+#define ICH_IOCE 0x10 /* interrupt on completion enable */
+#define ICH_FEIE 0x08 /* fifo error interrupt enable */
+#define ICH_LVBIE 0x04 /* last valid buffer interrupt enable */
+#define ICH_RESETREGS 0x02 /* reset busmaster registers */
+#define ICH_STARTBM 0x01 /* start busmaster operation */
+
+
+/* global block */
+#define ICH_REG_GLOB_CNT 0x2c /* dword - global control */
+#define ICH_PCM_SPDIF_MASK 0xc0000000 /* s/pdif pcm slot mask (ICH4) */
+#define ICH_PCM_SPDIF_NONE 0x00000000 /* reserved - undefined */
+#define ICH_PCM_SPDIF_78 0x40000000 /* s/pdif pcm on slots 7&8 */
+#define ICH_PCM_SPDIF_69 0x80000000 /* s/pdif pcm on slots 6&9 */
+#define ICH_PCM_SPDIF_1011 0xc0000000 /* s/pdif pcm on slots 10&11 */
+#define ICH_PCM_20BIT 0x00400000 /* 20-bit samples (ICH4) */
+#define ICH_PCM_246_MASK 0x00300000 /* chan mask (not all chips) */
+#define ICH_PCM_8 0x00300000 /* 8 channels (not all chips) */
+#define ICH_PCM_6 0x00200000 /* 6 channels (not all chips) */
+#define ICH_PCM_4 0x00100000 /* 4 channels (not all chips) */
+#define ICH_PCM_2 0x00000000 /* 2 channels (stereo) */
+#define ICH_SIS_PCM_246_MASK 0x000000c0 /* 6 channels (SIS7012) */
+#define ICH_SIS_PCM_6 0x00000080 /* 6 channels (SIS7012) */
+#define ICH_SIS_PCM_4 0x00000040 /* 4 channels (SIS7012) */
+#define ICH_SIS_PCM_2 0x00000000 /* 2 channels (SIS7012) */
+#define ICH_TRIE 0x00000040 /* tertiary resume interrupt enable */
+#define ICH_SRIE 0x00000020 /* secondary resume interrupt enable */
+#define ICH_PRIE 0x00000010 /* primary resume interrupt enable */
+#define ICH_ACLINK 0x00000008 /* AClink shut off */
+#define ICH_AC97WARM 0x00000004 /* AC'97 warm reset */
+#define ICH_AC97COLD 0x00000002 /* AC'97 cold reset */
+#define ICH_GIE 0x00000001 /* GPI interrupt enable */
+#define ICH_REG_GLOB_STA 0x30 /* dword - global status */
+#define ICH_TRI 0x20000000 /* ICH4: tertiary (AC_SDIN2) resume interrupt */
+#define ICH_TCR 0x10000000 /* ICH4: tertiary (AC_SDIN2) codec ready */
+#define ICH_BCS 0x08000000 /* ICH4: bit clock stopped */
+#define ICH_SPINT 0x04000000 /* ICH4: S/PDIF interrupt */
+#define ICH_P2INT 0x02000000 /* ICH4: PCM2-In interrupt */
+#define ICH_M2INT 0x01000000 /* ICH4: Mic2-In interrupt */
+#define ICH_SAMPLE_CAP 0x00c00000 /* ICH4: sample capability bits (RO) */
+#define ICH_SAMPLE_16_20 0x00400000 /* ICH4: 16- and 20-bit samples */
+#define ICH_MULTICHAN_CAP 0x00300000 /* ICH4: multi-channel capability bits (RO) */
+#define ICH_SIS_TRI 0x00080000 /* SIS: tertiary resume irq */
+#define ICH_SIS_TCR 0x00040000 /* SIS: tertiary codec ready */
+#define ICH_MD3 0x00020000 /* modem power down semaphore */
+#define ICH_AD3 0x00010000 /* audio power down semaphore */
+#define ICH_RCS 0x00008000 /* read completion status */
+#define ICH_BIT3 0x00004000 /* bit 3 slot 12 */
+#define ICH_BIT2 0x00002000 /* bit 2 slot 12 */
+#define ICH_BIT1 0x00001000 /* bit 1 slot 12 */
+#define ICH_SRI 0x00000800 /* secondary (AC_SDIN1) resume interrupt */
+#define ICH_PRI 0x00000400 /* primary (AC_SDIN0) resume interrupt */
+#define ICH_SCR 0x00000200 /* secondary (AC_SDIN1) codec ready */
+#define ICH_PCR 0x00000100 /* primary (AC_SDIN0) codec ready */
+#define ICH_MCINT 0x00000080 /* MIC capture interrupt */
+#define ICH_POINT 0x00000040 /* playback interrupt */
+#define ICH_PIINT 0x00000020 /* capture interrupt */
+#define ICH_NVSPINT 0x00000010 /* nforce spdif interrupt */
+#define ICH_MOINT 0x00000004 /* modem playback interrupt */
+#define ICH_MIINT 0x00000002 /* modem capture interrupt */
+#define ICH_GSCI 0x00000001 /* GPI status change interrupt */
+#define ICH_REG_ACC_SEMA 0x34 /* byte - codec write semaphore */
+#define ICH_CAS 0x01 /* codec access semaphore */
+#define ICH_REG_SDM 0x80
+#define ICH_DI2L_MASK 0x000000c0 /* PCM In 2, Mic In 2 data in line */
+#define ICH_DI2L_SHIFT 6
+#define ICH_DI1L_MASK 0x00000030 /* PCM In 1, Mic In 1 data in line */
+#define ICH_DI1L_SHIFT 4
+#define ICH_SE 0x00000008 /* steer enable */
+#define ICH_LDI_MASK 0x00000003 /* last codec read data input */
+
+#define ICH_MAX_FRAGS 32 /* max hw frags */
+
+
+/*
+ * registers for Ali5455
+ */
+
+/* ALi 5455 busmaster blocks */
+DEFINE_REGSET(AL_PI, 0x40); /* ALi PCM in */
+DEFINE_REGSET(AL_PO, 0x50); /* Ali PCM out */
+DEFINE_REGSET(AL_MC, 0x60); /* Ali Mic in */
+DEFINE_REGSET(AL_CDC_SPO, 0x70); /* Ali Codec SPDIF out */
+DEFINE_REGSET(AL_CENTER, 0x80); /* Ali center out */
+DEFINE_REGSET(AL_LFE, 0x90); /* Ali center out */
+DEFINE_REGSET(AL_CLR_SPI, 0xa0); /* Ali Controller SPDIF in */
+DEFINE_REGSET(AL_CLR_SPO, 0xb0); /* Ali Controller SPDIF out */
+DEFINE_REGSET(AL_I2S, 0xc0); /* Ali I2S in */
+DEFINE_REGSET(AL_PI2, 0xd0); /* Ali PCM2 in */
+DEFINE_REGSET(AL_MC2, 0xe0); /* Ali Mic2 in */
+
+enum {
+ ICH_REG_ALI_SCR = 0x00, /* System Control Register */
+ ICH_REG_ALI_SSR = 0x04, /* System Status Register */
+ ICH_REG_ALI_DMACR = 0x08, /* DMA Control Register */
+ ICH_REG_ALI_FIFOCR1 = 0x0c, /* FIFO Control Register 1 */
+ ICH_REG_ALI_INTERFACECR = 0x10, /* Interface Control Register */
+ ICH_REG_ALI_INTERRUPTCR = 0x14, /* Interrupt control Register */
+ ICH_REG_ALI_INTERRUPTSR = 0x18, /* Interrupt Status Register */
+ ICH_REG_ALI_FIFOCR2 = 0x1c, /* FIFO Control Register 2 */
+ ICH_REG_ALI_CPR = 0x20, /* Command Port Register */
+ ICH_REG_ALI_CPR_ADDR = 0x22, /* ac97 addr write */
+ ICH_REG_ALI_SPR = 0x24, /* Status Port Register */
+ ICH_REG_ALI_SPR_ADDR = 0x26, /* ac97 addr read */
+ ICH_REG_ALI_FIFOCR3 = 0x2c, /* FIFO Control Register 3 */
+ ICH_REG_ALI_TTSR = 0x30, /* Transmit Tag Slot Register */
+ ICH_REG_ALI_RTSR = 0x34, /* Receive Tag Slot Register */
+ ICH_REG_ALI_CSPSR = 0x38, /* Command/Status Port Status Register */
+ ICH_REG_ALI_CAS = 0x3c, /* Codec Write Semaphore Register */
+ ICH_REG_ALI_HWVOL = 0xf0, /* hardware volume control/status */
+ ICH_REG_ALI_I2SCR = 0xf4, /* I2S control/status */
+ ICH_REG_ALI_SPDIFCSR = 0xf8, /* spdif channel status register */
+ ICH_REG_ALI_SPDIFICS = 0xfc, /* spdif interface control/status */
+};
+
+#define ALI_CAS_SEM_BUSY 0x80000000
+#define ALI_CPR_ADDR_SECONDARY 0x100
+#define ALI_CPR_ADDR_READ 0x80
+#define ALI_CSPSR_CODEC_READY 0x08
+#define ALI_CSPSR_READ_OK 0x02
+#define ALI_CSPSR_WRITE_OK 0x01
+
+/* interrupts for the whole chip by interrupt status register finish */
+
+#define ALI_INT_MICIN2 (1<<26)
+#define ALI_INT_PCMIN2 (1<<25)
+#define ALI_INT_I2SIN (1<<24)
+#define ALI_INT_SPDIFOUT (1<<23) /* controller spdif out INTERRUPT */
+#define ALI_INT_SPDIFIN (1<<22)
+#define ALI_INT_LFEOUT (1<<21)
+#define ALI_INT_CENTEROUT (1<<20)
+#define ALI_INT_CODECSPDIFOUT (1<<19)
+#define ALI_INT_MICIN (1<<18)
+#define ALI_INT_PCMOUT (1<<17)
+#define ALI_INT_PCMIN (1<<16)
+#define ALI_INT_CPRAIS (1<<7) /* command port available */
+#define ALI_INT_SPRAIS (1<<5) /* status port available */
+#define ALI_INT_GPIO (1<<1)
+#define ALI_INT_MASK (ALI_INT_SPDIFOUT|ALI_INT_CODECSPDIFOUT|\
+ ALI_INT_MICIN|ALI_INT_PCMOUT|ALI_INT_PCMIN)
+
+#define ICH_ALI_SC_RESET (1<<31) /* master reset */
+#define ICH_ALI_SC_AC97_DBL (1<<30)
+#define ICH_ALI_SC_CODEC_SPDF (3<<20) /* 1=7/8, 2=6/9, 3=10/11 */
+#define ICH_ALI_SC_IN_BITS (3<<18)
+#define ICH_ALI_SC_OUT_BITS (3<<16)
+#define ICH_ALI_SC_6CH_CFG (3<<14)
+#define ICH_ALI_SC_PCM_4 (1<<8)
+#define ICH_ALI_SC_PCM_6 (2<<8)
+#define ICH_ALI_SC_PCM_246_MASK (3<<8)
+
+#define ICH_ALI_SS_SEC_ID (3<<5)
+#define ICH_ALI_SS_PRI_ID (3<<3)
+
+#define ICH_ALI_IF_AC97SP (1<<21)
+#define ICH_ALI_IF_MC (1<<20)
+#define ICH_ALI_IF_PI (1<<19)
+#define ICH_ALI_IF_MC2 (1<<18)
+#define ICH_ALI_IF_PI2 (1<<17)
+#define ICH_ALI_IF_LINE_SRC (1<<15) /* 0/1 = slot 3/6 */
+#define ICH_ALI_IF_MIC_SRC (1<<14) /* 0/1 = slot 3/6 */
+#define ICH_ALI_IF_SPDF_SRC (3<<12) /* 00 = PCM, 01 = AC97-in, 10 = spdif-in, 11 = i2s */
+#define ICH_ALI_IF_AC97_OUT (3<<8) /* 00 = PCM, 10 = spdif-in, 11 = i2s */
+#define ICH_ALI_IF_PO_SPDF (1<<3)
+#define ICH_ALI_IF_PO (1<<1)
+
+/*
+ *
+ */
+
+enum {
+ ICHD_PCMIN,
+ ICHD_PCMOUT,
+ ICHD_MIC,
+ ICHD_MIC2,
+ ICHD_PCM2IN,
+ ICHD_SPBAR,
+ ICHD_LAST = ICHD_SPBAR
+};
+enum {
+ NVD_PCMIN,
+ NVD_PCMOUT,
+ NVD_MIC,
+ NVD_SPBAR,
+ NVD_LAST = NVD_SPBAR
+};
+enum {
+ ALID_PCMIN,
+ ALID_PCMOUT,
+ ALID_MIC,
+ ALID_AC97SPDIFOUT,
+ ALID_SPDIFIN,
+ ALID_SPDIFOUT,
+ ALID_LAST = ALID_SPDIFOUT
+};
+
+#define get_ichdev(substream) (substream->runtime->private_data)
+
+struct ichdev {
+ unsigned int ichd; /* ich device number */
+ unsigned long reg_offset; /* offset to bmaddr */
+ u32 *bdbar; /* CPU address (32bit) */
+ unsigned int bdbar_addr; /* PCI bus address (32bit) */
+ struct snd_pcm_substream *substream;
+ unsigned int physbuf; /* physical address (32bit) */
+ unsigned int size;
+ unsigned int fragsize;
+ unsigned int fragsize1;
+ unsigned int position;
+ unsigned int pos_shift;
+ int frags;
+ int lvi;
+ int lvi_frag;
+ int civ;
+ int ack;
+ int ack_reload;
+ unsigned int ack_bit;
+ unsigned int roff_sr;
+ unsigned int roff_picb;
+ unsigned int int_sta_mask; /* interrupt status mask */
+ unsigned int ali_slot; /* ALI DMA slot */
+ struct ac97_pcm *pcm;
+ int pcm_open_flag;
+ unsigned int page_attr_changed: 1;
+ unsigned int suspended: 1;
+};
+
+struct intel8x0 {
+ unsigned int device_type;
+
+ int irq;
+
+ void __iomem *addr;
+ void __iomem *bmaddr;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+
+ int pcm_devs;
+ struct snd_pcm *pcm[6];
+ struct ichdev ichd[6];
+
+ unsigned multi4: 1,
+ multi6: 1,
+ multi8 :1,
+ dra: 1,
+ smp20bit: 1;
+ unsigned in_ac97_init: 1,
+ in_sdin_init: 1;
+ unsigned in_measurement: 1; /* during ac97 clock measurement */
+ unsigned fix_nocache: 1; /* workaround for 440MX */
+ unsigned buggy_irq: 1; /* workaround for buggy mobos */
+ unsigned xbox: 1; /* workaround for Xbox AC'97 detection */
+ unsigned buggy_semaphore: 1; /* workaround for buggy codec semaphore */
+
+ int spdif_idx; /* SPDIF BAR index; *_SPBAR or -1 if use PCMOUT */
+ unsigned int sdm_saved; /* SDM reg value */
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97[3];
+ unsigned int ac97_sdin[3];
+ unsigned int max_codecs, ncodecs;
+ unsigned int *codec_bit;
+ unsigned int codec_isr_bits;
+ unsigned int codec_ready_bits;
+
+ spinlock_t reg_lock;
+
+ u32 bdbars_count;
+ struct snd_dma_buffer bdbars;
+ u32 int_sta_reg; /* interrupt status register */
+ u32 int_sta_mask; /* interrupt status mask */
+};
+
+static struct pci_device_id snd_intel8x0_ids[] = {
+ { 0x8086, 0x2415, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */
+ { 0x8086, 0x2425, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */
+ { 0x8086, 0x2445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */
+ { 0x8086, 0x2485, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */
+ { 0x8086, 0x24c5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH4 */
+ { 0x8086, 0x24d5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH5 */
+ { 0x8086, 0x25a6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB */
+ { 0x8086, 0x266e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH6 */
+ { 0x8086, 0x27de, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH7 */
+ { 0x8086, 0x2698, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB2 */
+ { 0x8086, 0x7195, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */
+ { 0x1039, 0x7012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7012 */
+ { 0x10de, 0x01b1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */
+ { 0x10de, 0x003a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP04 */
+ { 0x10de, 0x006a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */
+ { 0x10de, 0x0059, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK804 */
+ { 0x10de, 0x008a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8 */
+ { 0x10de, 0x00da, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */
+ { 0x10de, 0x00ea, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8S */
+ { 0x10de, 0x026b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP51 */
+ { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */
+ { 0x1022, 0x7445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */
+ { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_intel8x0_ids);
+
+/*
+ * Lowlevel I/O - busmaster
+ */
+
+static inline u8 igetbyte(struct intel8x0 *chip, u32 offset)
+{
+ return ioread8(chip->bmaddr + offset);
+}
+
+static inline u16 igetword(struct intel8x0 *chip, u32 offset)
+{
+ return ioread16(chip->bmaddr + offset);
+}
+
+static inline u32 igetdword(struct intel8x0 *chip, u32 offset)
+{
+ return ioread32(chip->bmaddr + offset);
+}
+
+static inline void iputbyte(struct intel8x0 *chip, u32 offset, u8 val)
+{
+ iowrite8(val, chip->bmaddr + offset);
+}
+
+static inline void iputword(struct intel8x0 *chip, u32 offset, u16 val)
+{
+ iowrite16(val, chip->bmaddr + offset);
+}
+
+static inline void iputdword(struct intel8x0 *chip, u32 offset, u32 val)
+{
+ iowrite32(val, chip->bmaddr + offset);
+}
+
+/*
+ * Lowlevel I/O - AC'97 registers
+ */
+
+static inline u16 iagetword(struct intel8x0 *chip, u32 offset)
+{
+ return ioread16(chip->addr + offset);
+}
+
+static inline void iaputword(struct intel8x0 *chip, u32 offset, u16 val)
+{
+ iowrite16(val, chip->addr + offset);
+}
+
+/*
+ * Basic I/O
+ */
+
+/*
+ * access to AC97 codec via normal i/o (for ICH and SIS7012)
+ */
+
+static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int codec)
+{
+ int time;
+
+ if (codec > 2)
+ return -EIO;
+ if (chip->in_sdin_init) {
+ /* we don't know the ready bit assignment at the moment */
+ /* so we check any */
+ codec = chip->codec_isr_bits;
+ } else {
+ codec = chip->codec_bit[chip->ac97_sdin[codec]];
+ }
+
+ /* codec ready ? */
+ if ((igetdword(chip, ICHREG(GLOB_STA)) & codec) == 0)
+ return -EIO;
+
+ if (chip->buggy_semaphore)
+ return 0; /* just ignore ... */
+
+ /* Anyone holding a semaphore for 1 msec should be shot... */
+ time = 100;
+ do {
+ if (!(igetbyte(chip, ICHREG(ACC_SEMA)) & ICH_CAS))
+ return 0;
+ udelay(10);
+ } while (time--);
+
+ /* access to some forbidden (non existant) ac97 registers will not
+ * reset the semaphore. So even if you don't get the semaphore, still
+ * continue the access. We don't need the semaphore anyway. */
+ snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
+ igetbyte(chip, ICHREG(ACC_SEMA)), igetdword(chip, ICHREG(GLOB_STA)));
+ iagetword(chip, 0); /* clear semaphore flag */
+ /* I don't care about the semaphore */
+ return -EBUSY;
+}
+
+static void snd_intel8x0_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct intel8x0 *chip = ac97->private_data;
+
+ if (snd_intel8x0_codec_semaphore(chip, ac97->num) < 0) {
+ if (! chip->in_ac97_init)
+ snd_printk(KERN_ERR "codec_write %d: semaphore is not ready for register 0x%x\n", ac97->num, reg);
+ }
+ iaputword(chip, reg + ac97->num * 0x80, val);
+}
+
+static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct intel8x0 *chip = ac97->private_data;
+ unsigned short res;
+ unsigned int tmp;
+
+ if (snd_intel8x0_codec_semaphore(chip, ac97->num) < 0) {
+ if (! chip->in_ac97_init)
+ snd_printk(KERN_ERR "codec_read %d: semaphore is not ready for register 0x%x\n", ac97->num, reg);
+ res = 0xffff;
+ } else {
+ res = iagetword(chip, reg + ac97->num * 0x80);
+ if ((tmp = igetdword(chip, ICHREG(GLOB_STA))) & ICH_RCS) {
+ /* reset RCS and preserve other R/WC bits */
+ iputdword(chip, ICHREG(GLOB_STA), tmp &
+ ~(chip->codec_ready_bits | ICH_GSCI));
+ if (! chip->in_ac97_init)
+ snd_printk(KERN_ERR "codec_read %d: read timeout for register 0x%x\n", ac97->num, reg);
+ res = 0xffff;
+ }
+ }
+ return res;
+}
+
+static void __devinit snd_intel8x0_codec_read_test(struct intel8x0 *chip,
+ unsigned int codec)
+{
+ unsigned int tmp;
+
+ if (snd_intel8x0_codec_semaphore(chip, codec) >= 0) {
+ iagetword(chip, codec * 0x80);
+ if ((tmp = igetdword(chip, ICHREG(GLOB_STA))) & ICH_RCS) {
+ /* reset RCS and preserve other R/WC bits */
+ iputdword(chip, ICHREG(GLOB_STA), tmp &
+ ~(chip->codec_ready_bits | ICH_GSCI));
+ }
+ }
+}
+
+/*
+ * access to AC97 for Ali5455
+ */
+static int snd_intel8x0_ali_codec_ready(struct intel8x0 *chip, int mask)
+{
+ int count = 0;
+ for (count = 0; count < 0x7f; count++) {
+ int val = igetbyte(chip, ICHREG(ALI_CSPSR));
+ if (val & mask)
+ return 0;
+ }
+ if (! chip->in_ac97_init)
+ snd_printd(KERN_WARNING "intel8x0: AC97 codec ready timeout.\n");
+ return -EBUSY;
+}
+
+static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip)
+{
+ int time = 100;
+ if (chip->buggy_semaphore)
+ return 0; /* just ignore ... */
+ while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY))
+ udelay(1);
+ if (! time && ! chip->in_ac97_init)
+ snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n");
+ return snd_intel8x0_ali_codec_ready(chip, ALI_CSPSR_CODEC_READY);
+}
+
+static unsigned short snd_intel8x0_ali_codec_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct intel8x0 *chip = ac97->private_data;
+ unsigned short data = 0xffff;
+
+ if (snd_intel8x0_ali_codec_semaphore(chip))
+ goto __err;
+ reg |= ALI_CPR_ADDR_READ;
+ if (ac97->num)
+ reg |= ALI_CPR_ADDR_SECONDARY;
+ iputword(chip, ICHREG(ALI_CPR_ADDR), reg);
+ if (snd_intel8x0_ali_codec_ready(chip, ALI_CSPSR_READ_OK))
+ goto __err;
+ data = igetword(chip, ICHREG(ALI_SPR));
+ __err:
+ return data;
+}
+
+static void snd_intel8x0_ali_codec_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct intel8x0 *chip = ac97->private_data;
+
+ if (snd_intel8x0_ali_codec_semaphore(chip))
+ return;
+ iputword(chip, ICHREG(ALI_CPR), val);
+ if (ac97->num)
+ reg |= ALI_CPR_ADDR_SECONDARY;
+ iputword(chip, ICHREG(ALI_CPR_ADDR), reg);
+ snd_intel8x0_ali_codec_ready(chip, ALI_CSPSR_WRITE_OK);
+}
+
+
+/*
+ * DMA I/O
+ */
+static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ichdev)
+{
+ int idx;
+ u32 *bdbar = ichdev->bdbar;
+ unsigned long port = ichdev->reg_offset;
+
+ iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr);
+ if (ichdev->size == ichdev->fragsize) {
+ ichdev->ack_reload = ichdev->ack = 2;
+ ichdev->fragsize1 = ichdev->fragsize >> 1;
+ for (idx = 0; idx < (ICH_REG_LVI_MASK + 1) * 2; idx += 4) {
+ bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf);
+ bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
+ ichdev->fragsize1 >> ichdev->pos_shift);
+ bdbar[idx + 2] = cpu_to_le32(ichdev->physbuf + (ichdev->size >> 1));
+ bdbar[idx + 3] = cpu_to_le32(0x80000000 | /* interrupt on completion */
+ ichdev->fragsize1 >> ichdev->pos_shift);
+ }
+ ichdev->frags = 2;
+ } else {
+ ichdev->ack_reload = ichdev->ack = 1;
+ ichdev->fragsize1 = ichdev->fragsize;
+ for (idx = 0; idx < (ICH_REG_LVI_MASK + 1) * 2; idx += 2) {
+ bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf +
+ (((idx >> 1) * ichdev->fragsize) %
+ ichdev->size));
+ bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
+ ichdev->fragsize >> ichdev->pos_shift);
+#if 0
+ printk("bdbar[%i] = 0x%x [0x%x]\n",
+ idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
+#endif
+ }
+ ichdev->frags = ichdev->size / ichdev->fragsize;
+ }
+ iputbyte(chip, port + ICH_REG_OFF_LVI, ichdev->lvi = ICH_REG_LVI_MASK);
+ ichdev->civ = 0;
+ iputbyte(chip, port + ICH_REG_OFF_CIV, 0);
+ ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags;
+ ichdev->position = 0;
+#if 0
+ printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n",
+ ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1);
+#endif
+ /* clear interrupts */
+ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
+}
+
+#ifdef __i386__
+/*
+ * Intel 82443MX running a 100MHz processor system bus has a hardware bug,
+ * which aborts PCI busmaster for audio transfer. A workaround is to set
+ * the pages as non-cached. For details, see the errata in
+ * http://www.intel.com/design/chipsets/specupdt/245051.htm
+ */
+static void fill_nocache(void *buf, int size, int nocache)
+{
+ size = (size + PAGE_SIZE - 1) >> PAGE_SHIFT;
+ if (nocache)
+ set_pages_uc(virt_to_page(buf), size);
+ else
+ set_pages_wb(virt_to_page(buf), size);
+}
+#else
+#define fill_nocache(buf, size, nocache) do { ; } while (0)
+#endif
+
+/*
+ * Interrupt handler
+ */
+
+static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ichdev)
+{
+ unsigned long port = ichdev->reg_offset;
+ unsigned long flags;
+ int status, civ, i, step;
+ int ack = 0;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ status = igetbyte(chip, port + ichdev->roff_sr);
+ civ = igetbyte(chip, port + ICH_REG_OFF_CIV);
+ if (!(status & ICH_BCIS)) {
+ step = 0;
+ } else if (civ == ichdev->civ) {
+ // snd_printd("civ same %d\n", civ);
+ step = 1;
+ ichdev->civ++;
+ ichdev->civ &= ICH_REG_LVI_MASK;
+ } else {
+ step = civ - ichdev->civ;
+ if (step < 0)
+ step += ICH_REG_LVI_MASK + 1;
+ // if (step != 1)
+ // snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ);
+ ichdev->civ = civ;
+ }
+
+ ichdev->position += step * ichdev->fragsize1;
+ if (! chip->in_measurement)
+ ichdev->position %= ichdev->size;
+ ichdev->lvi += step;
+ ichdev->lvi &= ICH_REG_LVI_MASK;
+ iputbyte(chip, port + ICH_REG_OFF_LVI, ichdev->lvi);
+ for (i = 0; i < step; i++) {
+ ichdev->lvi_frag++;
+ ichdev->lvi_frag %= ichdev->frags;
+ ichdev->bdbar[ichdev->lvi * 2] = cpu_to_le32(ichdev->physbuf + ichdev->lvi_frag * ichdev->fragsize1);
+#if 0
+ printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n",
+ ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2],
+ ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port),
+ inl(port + 4), inb(port + ICH_REG_OFF_CR));
+#endif
+ if (--ichdev->ack == 0) {
+ ichdev->ack = ichdev->ack_reload;
+ ack = 1;
+ }
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (ack && ichdev->substream) {
+ snd_pcm_period_elapsed(ichdev->substream);
+ }
+ iputbyte(chip, port + ichdev->roff_sr,
+ status & (ICH_FIFOE | ICH_BCIS | ICH_LVBCI));
+}
+
+static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id)
+{
+ struct intel8x0 *chip = dev_id;
+ struct ichdev *ichdev;
+ unsigned int status;
+ unsigned int i;
+
+ status = igetdword(chip, chip->int_sta_reg);
+ if (status == 0xffffffff) /* we are not yet resumed */
+ return IRQ_NONE;
+
+ if ((status & chip->int_sta_mask) == 0) {
+ if (status) {
+ /* ack */
+ iputdword(chip, chip->int_sta_reg, status);
+ if (! chip->buggy_irq)
+ status = 0;
+ }
+ return IRQ_RETVAL(status);
+ }
+
+ for (i = 0; i < chip->bdbars_count; i++) {
+ ichdev = &chip->ichd[i];
+ if (status & ichdev->int_sta_mask)
+ snd_intel8x0_update(chip, ichdev);
+ }
+
+ /* ack them */
+ iputdword(chip, chip->int_sta_reg, status & chip->int_sta_mask);
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * PCM part
+ */
+
+static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct ichdev *ichdev = get_ichdev(substream);
+ unsigned char val = 0;
+ unsigned long port = ichdev->reg_offset;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_RESUME:
+ ichdev->suspended = 0;
+ /* fallthru */
+ case SNDRV_PCM_TRIGGER_START:
+ val = ICH_IOCE | ICH_STARTBM;
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ichdev->suspended = 1;
+ /* fallthru */
+ case SNDRV_PCM_TRIGGER_STOP:
+ val = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ val = ICH_IOCE;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ val = ICH_IOCE | ICH_STARTBM;
+ break;
+ default:
+ return -EINVAL;
+ }
+ iputbyte(chip, port + ICH_REG_OFF_CR, val);
+ if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ /* wait until DMA stopped */
+ while (!(igetbyte(chip, port + ichdev->roff_sr) & ICH_DCH)) ;
+ /* reset whole DMA things */
+ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_RESETREGS);
+ }
+ return 0;
+}
+
+static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct ichdev *ichdev = get_ichdev(substream);
+ unsigned long port = ichdev->reg_offset;
+ static int fiforeg[] = {
+ ICHREG(ALI_FIFOCR1), ICHREG(ALI_FIFOCR2), ICHREG(ALI_FIFOCR3)
+ };
+ unsigned int val, fifo;
+
+ val = igetdword(chip, ICHREG(ALI_DMACR));
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_RESUME:
+ ichdev->suspended = 0;
+ /* fallthru */
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* clear FIFO for synchronization of channels */
+ fifo = igetdword(chip, fiforeg[ichdev->ali_slot / 4]);
+ fifo &= ~(0xff << (ichdev->ali_slot % 4));
+ fifo |= 0x83 << (ichdev->ali_slot % 4);
+ iputdword(chip, fiforeg[ichdev->ali_slot / 4], fifo);
+ }
+ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_IOCE);
+ val &= ~(1 << (ichdev->ali_slot + 16)); /* clear PAUSE flag */
+ /* start DMA */
+ iputdword(chip, ICHREG(ALI_DMACR), val | (1 << ichdev->ali_slot));
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ichdev->suspended = 1;
+ /* fallthru */
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ /* pause */
+ iputdword(chip, ICHREG(ALI_DMACR), val | (1 << (ichdev->ali_slot + 16)));
+ iputbyte(chip, port + ICH_REG_OFF_CR, 0);
+ while (igetbyte(chip, port + ICH_REG_OFF_CR))
+ ;
+ if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH)
+ break;
+ /* reset whole DMA things */
+ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_RESETREGS);
+ /* clear interrupts */
+ iputbyte(chip, port + ICH_REG_OFF_SR,
+ igetbyte(chip, port + ICH_REG_OFF_SR) | 0x1e);
+ iputdword(chip, ICHREG(ALI_INTERRUPTSR),
+ igetdword(chip, ICHREG(ALI_INTERRUPTSR)) & ichdev->int_sta_mask);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct ichdev *ichdev = get_ichdev(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int dbl = params_rate(hw_params) > 48000;
+ int err;
+
+ if (chip->fix_nocache && ichdev->page_attr_changed) {
+ fill_nocache(runtime->dma_area, runtime->dma_bytes, 0); /* clear */
+ ichdev->page_attr_changed = 0;
+ }
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ if (chip->fix_nocache) {
+ if (runtime->dma_area && ! ichdev->page_attr_changed) {
+ fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
+ ichdev->page_attr_changed = 1;
+ }
+ }
+ if (ichdev->pcm_open_flag) {
+ snd_ac97_pcm_close(ichdev->pcm);
+ ichdev->pcm_open_flag = 0;
+ }
+ err = snd_ac97_pcm_open(ichdev->pcm, params_rate(hw_params),
+ params_channels(hw_params),
+ ichdev->pcm->r[dbl].slots);
+ if (err >= 0) {
+ ichdev->pcm_open_flag = 1;
+ /* Force SPDIF setting */
+ if (ichdev->ichd == ICHD_PCMOUT && chip->spdif_idx < 0)
+ snd_ac97_set_rate(ichdev->pcm->r[0].codec[0], AC97_SPDIF,
+ params_rate(hw_params));
+ }
+ return err;
+}
+
+static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct ichdev *ichdev = get_ichdev(substream);
+
+ if (ichdev->pcm_open_flag) {
+ snd_ac97_pcm_close(ichdev->pcm);
+ ichdev->pcm_open_flag = 0;
+ }
+ if (chip->fix_nocache && ichdev->page_attr_changed) {
+ fill_nocache(substream->runtime->dma_area, substream->runtime->dma_bytes, 0);
+ ichdev->page_attr_changed = 0;
+ }
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static void snd_intel8x0_setup_pcm_out(struct intel8x0 *chip,
+ struct snd_pcm_runtime *runtime)
+{
+ unsigned int cnt;
+ int dbl = runtime->rate > 48000;
+
+ spin_lock_irq(&chip->reg_lock);
+ switch (chip->device_type) {
+ case DEVICE_ALI:
+ cnt = igetdword(chip, ICHREG(ALI_SCR));
+ cnt &= ~ICH_ALI_SC_PCM_246_MASK;
+ if (runtime->channels == 4 || dbl)
+ cnt |= ICH_ALI_SC_PCM_4;
+ else if (runtime->channels == 6)
+ cnt |= ICH_ALI_SC_PCM_6;
+ iputdword(chip, ICHREG(ALI_SCR), cnt);
+ break;
+ case DEVICE_SIS:
+ cnt = igetdword(chip, ICHREG(GLOB_CNT));
+ cnt &= ~ICH_SIS_PCM_246_MASK;
+ if (runtime->channels == 4 || dbl)
+ cnt |= ICH_SIS_PCM_4;
+ else if (runtime->channels == 6)
+ cnt |= ICH_SIS_PCM_6;
+ iputdword(chip, ICHREG(GLOB_CNT), cnt);
+ break;
+ default:
+ cnt = igetdword(chip, ICHREG(GLOB_CNT));
+ cnt &= ~(ICH_PCM_246_MASK | ICH_PCM_20BIT);
+ if (runtime->channels == 4 || dbl)
+ cnt |= ICH_PCM_4;
+ else if (runtime->channels == 6)
+ cnt |= ICH_PCM_6;
+ else if (runtime->channels == 8)
+ cnt |= ICH_PCM_8;
+ if (chip->device_type == DEVICE_NFORCE) {
+ /* reset to 2ch once to keep the 6 channel data in alignment,
+ * to start from Front Left always
+ */
+ if (cnt & ICH_PCM_246_MASK) {
+ iputdword(chip, ICHREG(GLOB_CNT), cnt & ~ICH_PCM_246_MASK);
+ spin_unlock_irq(&chip->reg_lock);
+ msleep(50); /* grrr... */
+ spin_lock_irq(&chip->reg_lock);
+ }
+ } else if (chip->device_type == DEVICE_INTEL_ICH4) {
+ if (runtime->sample_bits > 16)
+ cnt |= ICH_PCM_20BIT;
+ }
+ iputdword(chip, ICHREG(GLOB_CNT), cnt);
+ break;
+ }
+ spin_unlock_irq(&chip->reg_lock);
+}
+
+static int snd_intel8x0_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ichdev *ichdev = get_ichdev(substream);
+
+ ichdev->physbuf = runtime->dma_addr;
+ ichdev->size = snd_pcm_lib_buffer_bytes(substream);
+ ichdev->fragsize = snd_pcm_lib_period_bytes(substream);
+ if (ichdev->ichd == ICHD_PCMOUT) {
+ snd_intel8x0_setup_pcm_out(chip, runtime);
+ if (chip->device_type == DEVICE_INTEL_ICH4)
+ ichdev->pos_shift = (runtime->sample_bits > 16) ? 2 : 1;
+ }
+ snd_intel8x0_setup_periods(chip, ichdev);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct ichdev *ichdev = get_ichdev(substream);
+ size_t ptr1, ptr;
+ int civ, timeout = 100;
+ unsigned int position;
+
+ spin_lock(&chip->reg_lock);
+ do {
+ civ = igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV);
+ ptr1 = igetword(chip, ichdev->reg_offset + ichdev->roff_picb);
+ position = ichdev->position;
+ if (ptr1 == 0) {
+ udelay(10);
+ continue;
+ }
+ if (civ == igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV) &&
+ ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb))
+ break;
+ } while (timeout--);
+ ptr1 <<= ichdev->pos_shift;
+ ptr = ichdev->fragsize1 - ptr1;
+ ptr += position;
+ spin_unlock(&chip->reg_lock);
+ if (ptr >= ichdev->size)
+ return 0;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static struct snd_pcm_hardware snd_intel8x0_stream =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 128 * 1024,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static unsigned int channels4[] = {
+ 2, 4,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_channels4 = {
+ .count = ARRAY_SIZE(channels4),
+ .list = channels4,
+ .mask = 0,
+};
+
+static unsigned int channels6[] = {
+ 2, 4, 6,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_channels6 = {
+ .count = ARRAY_SIZE(channels6),
+ .list = channels6,
+ .mask = 0,
+};
+
+static unsigned int channels8[] = {
+ 2, 4, 6, 8,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_channels8 = {
+ .count = ARRAY_SIZE(channels8),
+ .list = channels8,
+ .mask = 0,
+};
+
+static int snd_intel8x0_pcm_open(struct snd_pcm_substream *substream, struct ichdev *ichdev)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ ichdev->substream = substream;
+ runtime->hw = snd_intel8x0_stream;
+ runtime->hw.rates = ichdev->pcm->rates;
+ snd_pcm_limit_hw_rates(runtime);
+ if (chip->device_type == DEVICE_SIS) {
+ runtime->hw.buffer_bytes_max = 64*1024;
+ runtime->hw.period_bytes_max = 64*1024;
+ }
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ runtime->private_data = ichdev;
+ return 0;
+}
+
+static int snd_intel8x0_playback_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ err = snd_intel8x0_pcm_open(substream, &chip->ichd[ICHD_PCMOUT]);
+ if (err < 0)
+ return err;
+
+ if (chip->multi8) {
+ runtime->hw.channels_max = 8;
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &hw_constraints_channels8);
+ } else if (chip->multi6) {
+ runtime->hw.channels_max = 6;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &hw_constraints_channels6);
+ } else if (chip->multi4) {
+ runtime->hw.channels_max = 4;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ &hw_constraints_channels4);
+ }
+ if (chip->dra) {
+ snd_ac97_pcm_double_rate_rules(runtime);
+ }
+ if (chip->smp20bit) {
+ runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 20);
+ }
+ return 0;
+}
+
+static int snd_intel8x0_playback_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ICHD_PCMOUT].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0_capture_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[ICHD_PCMIN]);
+}
+
+static int snd_intel8x0_capture_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ICHD_PCMIN].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0_mic_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[ICHD_MIC]);
+}
+
+static int snd_intel8x0_mic_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ICHD_MIC].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0_mic2_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[ICHD_MIC2]);
+}
+
+static int snd_intel8x0_mic2_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ICHD_MIC2].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0_capture2_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[ICHD_PCM2IN]);
+}
+
+static int snd_intel8x0_capture2_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ICHD_PCM2IN].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ int idx = chip->device_type == DEVICE_NFORCE ? NVD_SPBAR : ICHD_SPBAR;
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[idx]);
+}
+
+static int snd_intel8x0_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ int idx = chip->device_type == DEVICE_NFORCE ? NVD_SPBAR : ICHD_SPBAR;
+
+ chip->ichd[idx].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0_ali_ac97spdifout_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ unsigned int val;
+
+ spin_lock_irq(&chip->reg_lock);
+ val = igetdword(chip, ICHREG(ALI_INTERFACECR));
+ val |= ICH_ALI_IF_AC97SP;
+ iputdword(chip, ICHREG(ALI_INTERFACECR), val);
+ /* also needs to set ALI_SC_CODEC_SPDF correctly */
+ spin_unlock_irq(&chip->reg_lock);
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[ALID_AC97SPDIFOUT]);
+}
+
+static int snd_intel8x0_ali_ac97spdifout_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+ unsigned int val;
+
+ chip->ichd[ALID_AC97SPDIFOUT].substream = NULL;
+ spin_lock_irq(&chip->reg_lock);
+ val = igetdword(chip, ICHREG(ALI_INTERFACECR));
+ val &= ~ICH_ALI_IF_AC97SP;
+ iputdword(chip, ICHREG(ALI_INTERFACECR), val);
+ spin_unlock_irq(&chip->reg_lock);
+
+ return 0;
+}
+
+#if 0 // NYI
+static int snd_intel8x0_ali_spdifin_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[ALID_SPDIFIN]);
+}
+
+static int snd_intel8x0_ali_spdifin_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ALID_SPDIFIN].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0_ali_spdifout_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0_pcm_open(substream, &chip->ichd[ALID_SPDIFOUT]);
+}
+
+static int snd_intel8x0_ali_spdifout_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0 *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ALID_SPDIFOUT].substream = NULL;
+ return 0;
+}
+#endif
+
+static struct snd_pcm_ops snd_intel8x0_playback_ops = {
+ .open = snd_intel8x0_playback_open,
+ .close = snd_intel8x0_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_capture_ops = {
+ .open = snd_intel8x0_capture_open,
+ .close = snd_intel8x0_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_capture_mic_ops = {
+ .open = snd_intel8x0_mic_open,
+ .close = snd_intel8x0_mic_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_capture_mic2_ops = {
+ .open = snd_intel8x0_mic2_open,
+ .close = snd_intel8x0_mic2_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_capture2_ops = {
+ .open = snd_intel8x0_capture2_open,
+ .close = snd_intel8x0_capture2_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_spdif_ops = {
+ .open = snd_intel8x0_spdif_open,
+ .close = snd_intel8x0_spdif_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_ali_playback_ops = {
+ .open = snd_intel8x0_playback_open,
+ .close = snd_intel8x0_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_ali_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_ali_capture_ops = {
+ .open = snd_intel8x0_capture_open,
+ .close = snd_intel8x0_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_ali_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_ali_capture_mic_ops = {
+ .open = snd_intel8x0_mic_open,
+ .close = snd_intel8x0_mic_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_ali_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_ali_ac97spdifout_ops = {
+ .open = snd_intel8x0_ali_ac97spdifout_open,
+ .close = snd_intel8x0_ali_ac97spdifout_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_ali_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+#if 0 // NYI
+static struct snd_pcm_ops snd_intel8x0_ali_spdifin_ops = {
+ .open = snd_intel8x0_ali_spdifin_open,
+ .close = snd_intel8x0_ali_spdifin_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0_ali_spdifout_ops = {
+ .open = snd_intel8x0_ali_spdifout_open,
+ .close = snd_intel8x0_ali_spdifout_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+#endif // NYI
+
+struct ich_pcm_table {
+ char *suffix;
+ struct snd_pcm_ops *playback_ops;
+ struct snd_pcm_ops *capture_ops;
+ size_t prealloc_size;
+ size_t prealloc_max_size;
+ int ac97_idx;
+};
+
+static int __devinit snd_intel8x0_pcm1(struct intel8x0 *chip, int device,
+ struct ich_pcm_table *rec)
+{
+ struct snd_pcm *pcm;
+ int err;
+ char name[32];
+
+ if (rec->suffix)
+ sprintf(name, "Intel ICH - %s", rec->suffix);
+ else
+ strcpy(name, "Intel ICH");
+ err = snd_pcm_new(chip->card, name, device,
+ rec->playback_ops ? 1 : 0,
+ rec->capture_ops ? 1 : 0, &pcm);
+ if (err < 0)
+ return err;
+
+ if (rec->playback_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, rec->playback_ops);
+ if (rec->capture_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, rec->capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ if (rec->suffix)
+ sprintf(pcm->name, "%s - %s", chip->card->shortname, rec->suffix);
+ else
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcm[device] = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ rec->prealloc_size, rec->prealloc_max_size);
+
+ return 0;
+}
+
+static struct ich_pcm_table intel_pcms[] __devinitdata = {
+ {
+ .playback_ops = &snd_intel8x0_playback_ops,
+ .capture_ops = &snd_intel8x0_capture_ops,
+ .prealloc_size = 64 * 1024,
+ .prealloc_max_size = 128 * 1024,
+ },
+ {
+ .suffix = "MIC ADC",
+ .capture_ops = &snd_intel8x0_capture_mic_ops,
+ .prealloc_size = 0,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = ICHD_MIC,
+ },
+ {
+ .suffix = "MIC2 ADC",
+ .capture_ops = &snd_intel8x0_capture_mic2_ops,
+ .prealloc_size = 0,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = ICHD_MIC2,
+ },
+ {
+ .suffix = "ADC2",
+ .capture_ops = &snd_intel8x0_capture2_ops,
+ .prealloc_size = 0,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = ICHD_PCM2IN,
+ },
+ {
+ .suffix = "IEC958",
+ .playback_ops = &snd_intel8x0_spdif_ops,
+ .prealloc_size = 64 * 1024,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = ICHD_SPBAR,
+ },
+};
+
+static struct ich_pcm_table nforce_pcms[] __devinitdata = {
+ {
+ .playback_ops = &snd_intel8x0_playback_ops,
+ .capture_ops = &snd_intel8x0_capture_ops,
+ .prealloc_size = 64 * 1024,
+ .prealloc_max_size = 128 * 1024,
+ },
+ {
+ .suffix = "MIC ADC",
+ .capture_ops = &snd_intel8x0_capture_mic_ops,
+ .prealloc_size = 0,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = NVD_MIC,
+ },
+ {
+ .suffix = "IEC958",
+ .playback_ops = &snd_intel8x0_spdif_ops,
+ .prealloc_size = 64 * 1024,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = NVD_SPBAR,
+ },
+};
+
+static struct ich_pcm_table ali_pcms[] __devinitdata = {
+ {
+ .playback_ops = &snd_intel8x0_ali_playback_ops,
+ .capture_ops = &snd_intel8x0_ali_capture_ops,
+ .prealloc_size = 64 * 1024,
+ .prealloc_max_size = 128 * 1024,
+ },
+ {
+ .suffix = "MIC ADC",
+ .capture_ops = &snd_intel8x0_ali_capture_mic_ops,
+ .prealloc_size = 0,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = ALID_MIC,
+ },
+ {
+ .suffix = "IEC958",
+ .playback_ops = &snd_intel8x0_ali_ac97spdifout_ops,
+ /* .capture_ops = &snd_intel8x0_ali_spdifin_ops, */
+ .prealloc_size = 64 * 1024,
+ .prealloc_max_size = 128 * 1024,
+ .ac97_idx = ALID_AC97SPDIFOUT,
+ },
+#if 0 // NYI
+ {
+ .suffix = "HW IEC958",
+ .playback_ops = &snd_intel8x0_ali_spdifout_ops,
+ .prealloc_size = 64 * 1024,
+ .prealloc_max_size = 128 * 1024,
+ },
+#endif
+};
+
+static int __devinit snd_intel8x0_pcm(struct intel8x0 *chip)
+{
+ int i, tblsize, device, err;
+ struct ich_pcm_table *tbl, *rec;
+
+ switch (chip->device_type) {
+ case DEVICE_INTEL_ICH4:
+ tbl = intel_pcms;
+ tblsize = ARRAY_SIZE(intel_pcms);
+ if (spdif_aclink)
+ tblsize--;
+ break;
+ case DEVICE_NFORCE:
+ tbl = nforce_pcms;
+ tblsize = ARRAY_SIZE(nforce_pcms);
+ if (spdif_aclink)
+ tblsize--;
+ break;
+ case DEVICE_ALI:
+ tbl = ali_pcms;
+ tblsize = ARRAY_SIZE(ali_pcms);
+ break;
+ default:
+ tbl = intel_pcms;
+ tblsize = 2;
+ break;
+ }
+
+ device = 0;
+ for (i = 0; i < tblsize; i++) {
+ rec = tbl + i;
+ if (i > 0 && rec->ac97_idx) {
+ /* activate PCM only when associated AC'97 codec */
+ if (! chip->ichd[rec->ac97_idx].pcm)
+ continue;
+ }
+ err = snd_intel8x0_pcm1(chip, device, rec);
+ if (err < 0)
+ return err;
+ device++;
+ }
+
+ chip->pcm_devs = device;
+ return 0;
+}
+
+
+/*
+ * Mixer part
+ */
+
+static void snd_intel8x0_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct intel8x0 *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void snd_intel8x0_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct intel8x0 *chip = ac97->private_data;
+ chip->ac97[ac97->num] = NULL;
+}
+
+static struct ac97_pcm ac97_pcm_defs[] __devinitdata = {
+ /* front PCM */
+ {
+ .exclusive = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT) |
+ (1 << AC97_SLOT_PCM_CENTER) |
+ (1 << AC97_SLOT_PCM_SLEFT) |
+ (1 << AC97_SLOT_PCM_SRIGHT) |
+ (1 << AC97_SLOT_LFE)
+ },
+ {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT) |
+ (1 << AC97_SLOT_PCM_LEFT_0) |
+ (1 << AC97_SLOT_PCM_RIGHT_0)
+ }
+ }
+ },
+ /* PCM IN #1 */
+ {
+ .stream = 1,
+ .exclusive = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT)
+ }
+ }
+ },
+ /* MIC IN #1 */
+ {
+ .stream = 1,
+ .exclusive = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_MIC)
+ }
+ }
+ },
+ /* S/PDIF PCM */
+ {
+ .exclusive = 1,
+ .spdif = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_SPDIF_LEFT2) |
+ (1 << AC97_SLOT_SPDIF_RIGHT2)
+ }
+ }
+ },
+ /* PCM IN #2 */
+ {
+ .stream = 1,
+ .exclusive = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT)
+ }
+ }
+ },
+ /* MIC IN #2 */
+ {
+ .stream = 1,
+ .exclusive = 1,
+ .r = { {
+ .slots = (1 << AC97_SLOT_MIC)
+ }
+ }
+ },
+};
+
+static struct ac97_quirk ac97_quirks[] __devinitdata = {
+ {
+ .subvendor = 0x0e11,
+ .subdevice = 0x000e,
+ .name = "Compaq Deskpro EN", /* AD1885 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x0e11,
+ .subdevice = 0x008a,
+ .name = "Compaq Evo W4000", /* AD1885 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x0e11,
+ .subdevice = 0x00b8,
+ .name = "Compaq Evo D510C",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x0e11,
+ .subdevice = 0x0860,
+ .name = "HP/Compaq nx7010",
+ .type = AC97_TUNE_MUTE_LED
+ },
+ {
+ .subvendor = 0x1014,
+ .subdevice = 0x1f00,
+ .name = "MS-9128",
+ .type = AC97_TUNE_ALC_JACK
+ },
+ {
+ .subvendor = 0x1014,
+ .subdevice = 0x0267,
+ .name = "IBM NetVista A30p", /* AD1981B */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1025,
+ .subdevice = 0x0082,
+ .name = "Acer Travelmate 2310",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1025,
+ .subdevice = 0x0083,
+ .name = "Acer Aspire 3003LCi",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x00d8,
+ .name = "Dell Precision 530", /* AD1885 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x010d,
+ .name = "Dell", /* which model? AD1885 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x0126,
+ .name = "Dell Optiplex GX260", /* AD1981A */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x012c,
+ .name = "Dell Precision 650", /* AD1981A */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x012d,
+ .name = "Dell Precision 450", /* AD1981B*/
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x0147,
+ .name = "Dell", /* which model? AD1981B*/
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x0151,
+ .name = "Dell Optiplex GX270", /* AD1981B */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x014e,
+ .name = "Dell D800", /* STAC9750/51 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x0163,
+ .name = "Dell Unknown", /* STAC9750/51 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x0186,
+ .name = "Dell Latitude D810", /* cf. Malone #41015 */
+ .type = AC97_TUNE_HP_MUTE_LED
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x0188,
+ .name = "Dell Inspiron 6000",
+ .type = AC97_TUNE_HP_MUTE_LED /* cf. Malone #41015 */
+ },
+ {
+ .subvendor = 0x1028,
+ .subdevice = 0x0191,
+ .name = "Dell Inspiron 8600",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x006d,
+ .name = "HP zv5000",
+ .type = AC97_TUNE_MUTE_LED /*AD1981B*/
+ },
+ { /* FIXME: which codec? */
+ .subvendor = 0x103c,
+ .subdevice = 0x00c3,
+ .name = "HP xw6000",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x088c,
+ .name = "HP nc8000",
+ .type = AC97_TUNE_HP_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x0890,
+ .name = "HP nc6000",
+ .type = AC97_TUNE_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x0934,
+ .name = "HP nx8220",
+ .type = AC97_TUNE_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x129d,
+ .name = "HP xw8000",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x0938,
+ .name = "HP nc4200",
+ .type = AC97_TUNE_HP_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x099c,
+ .name = "HP nx6110/nc6120",
+ .type = AC97_TUNE_HP_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x0944,
+ .name = "HP nc6220",
+ .type = AC97_TUNE_HP_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x0934,
+ .name = "HP nc8220",
+ .type = AC97_TUNE_HP_MUTE_LED
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x12f1,
+ .name = "HP xw8200", /* AD1981B*/
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x12f2,
+ .name = "HP xw6200",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x103c,
+ .subdevice = 0x3008,
+ .name = "HP xw4200", /* AD1981B*/
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x104d,
+ .subdevice = 0x8197,
+ .name = "Sony S1XP",
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x1043,
+ .subdevice = 0x80f3,
+ .name = "ASUS ICH5/AD1985",
+ .type = AC97_TUNE_AD_SHARING
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x11c3,
+ .name = "Fujitsu-Siemens E4010",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x1225,
+ .name = "Fujitsu-Siemens T3010",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x1253,
+ .name = "Fujitsu S6210", /* STAC9750/51 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x127d,
+ .name = "Fujitsu Lifebook P7010",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x127e,
+ .name = "Fujitsu Lifebook C1211D",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x12ec,
+ .name = "Fujitsu-Siemens 4010",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x12f2,
+ .name = "Fujitsu-Siemens Celsius H320",
+ .type = AC97_TUNE_SWAP_HP
+ },
+ {
+ .subvendor = 0x10f1,
+ .subdevice = 0x2665,
+ .name = "Fujitsu-Siemens Celsius", /* AD1981? */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10f1,
+ .subdevice = 0x2885,
+ .name = "AMD64 Mobo", /* ALC650 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10f1,
+ .subdevice = 0x2895,
+ .name = "Tyan Thunder K8WE",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10f7,
+ .subdevice = 0x834c,
+ .name = "Panasonic CF-R4",
+ .type = AC97_TUNE_HP_ONLY,
+ },
+ {
+ .subvendor = 0x110a,
+ .subdevice = 0x0056,
+ .name = "Fujitsu-Siemens Scenic", /* AD1981? */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x11d4,
+ .subdevice = 0x5375,
+ .name = "ADI AD1985 (discrete)",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1462,
+ .subdevice = 0x5470,
+ .name = "MSI P4 ATX 645 Ultra",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1734,
+ .subdevice = 0x0088,
+ .name = "Fujitsu-Siemens D1522", /* AD1981 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0x2000,
+ .mask = 0xfff0,
+ .name = "Intel ICH5/AD1985",
+ .type = AC97_TUNE_AD_SHARING
+ },
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0x4000,
+ .mask = 0xfff0,
+ .name = "Intel ICH5/AD1985",
+ .type = AC97_TUNE_AD_SHARING
+ },
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0x4856,
+ .name = "Intel D845WN (82801BA)",
+ .type = AC97_TUNE_SWAP_HP
+ },
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0x4d44,
+ .name = "Intel D850EMV2", /* AD1885 */
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0x4d56,
+ .name = "Intel ICH/AD1885",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0x6000,
+ .mask = 0xfff0,
+ .name = "Intel ICH5/AD1985",
+ .type = AC97_TUNE_AD_SHARING
+ },
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0xe000,
+ .mask = 0xfff0,
+ .name = "Intel ICH5/AD1985",
+ .type = AC97_TUNE_AD_SHARING
+ },
+#if 0 /* FIXME: this seems wrong on most boards */
+ {
+ .subvendor = 0x8086,
+ .subdevice = 0xa000,
+ .mask = 0xfff0,
+ .name = "Intel ICH5/AD1985",
+ .type = AC97_TUNE_HP_ONLY
+ },
+#endif
+ { } /* terminator */
+};
+
+static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock,
+ const char *quirk_override)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ unsigned int i, codecs;
+ unsigned int glob_sta = 0;
+ struct snd_ac97_bus_ops *ops;
+ static struct snd_ac97_bus_ops standard_bus_ops = {
+ .write = snd_intel8x0_codec_write,
+ .read = snd_intel8x0_codec_read,
+ };
+ static struct snd_ac97_bus_ops ali_bus_ops = {
+ .write = snd_intel8x0_ali_codec_write,
+ .read = snd_intel8x0_ali_codec_read,
+ };
+
+ chip->spdif_idx = -1; /* use PCMOUT (or disabled) */
+ if (!spdif_aclink) {
+ switch (chip->device_type) {
+ case DEVICE_NFORCE:
+ chip->spdif_idx = NVD_SPBAR;
+ break;
+ case DEVICE_ALI:
+ chip->spdif_idx = ALID_AC97SPDIFOUT;
+ break;
+ case DEVICE_INTEL_ICH4:
+ chip->spdif_idx = ICHD_SPBAR;
+ break;
+ };
+ }
+
+ chip->in_ac97_init = 1;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_intel8x0_mixer_free_ac97;
+ ac97.scaps = AC97_SCAP_SKIP_MODEM | AC97_SCAP_POWER_SAVE;
+ if (chip->xbox)
+ ac97.scaps |= AC97_SCAP_DETECT_BY_VENDOR;
+ if (chip->device_type != DEVICE_ALI) {
+ glob_sta = igetdword(chip, ICHREG(GLOB_STA));
+ ops = &standard_bus_ops;
+ chip->in_sdin_init = 1;
+ codecs = 0;
+ for (i = 0; i < chip->max_codecs; i++) {
+ if (! (glob_sta & chip->codec_bit[i]))
+ continue;
+ if (chip->device_type == DEVICE_INTEL_ICH4) {
+ snd_intel8x0_codec_read_test(chip, codecs);
+ chip->ac97_sdin[codecs] =
+ igetbyte(chip, ICHREG(SDM)) & ICH_LDI_MASK;
+ if (snd_BUG_ON(chip->ac97_sdin[codecs] >= 3))
+ chip->ac97_sdin[codecs] = 0;
+ } else
+ chip->ac97_sdin[codecs] = i;
+ codecs++;
+ }
+ chip->in_sdin_init = 0;
+ if (! codecs)
+ codecs = 1;
+ } else {
+ ops = &ali_bus_ops;
+ codecs = 1;
+ /* detect the secondary codec */
+ for (i = 0; i < 100; i++) {
+ unsigned int reg = igetdword(chip, ICHREG(ALI_RTSR));
+ if (reg & 0x40) {
+ codecs = 2;
+ break;
+ }
+ iputdword(chip, ICHREG(ALI_RTSR), reg | 0x40);
+ udelay(1);
+ }
+ }
+ if ((err = snd_ac97_bus(chip->card, 0, ops, chip, &pbus)) < 0)
+ goto __err;
+ pbus->private_free = snd_intel8x0_mixer_free_ac97_bus;
+ if (ac97_clock >= 8000 && ac97_clock <= 48000)
+ pbus->clock = ac97_clock;
+ /* FIXME: my test board doesn't work well with VRA... */
+ if (chip->device_type == DEVICE_ALI)
+ pbus->no_vra = 1;
+ else
+ pbus->dra = 1;
+ chip->ac97_bus = pbus;
+ chip->ncodecs = codecs;
+
+ ac97.pci = chip->pci;
+ for (i = 0; i < codecs; i++) {
+ ac97.num = i;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97[i])) < 0) {
+ if (err != -EACCES)
+ snd_printk(KERN_ERR "Unable to initialize codec #%d\n", i);
+ if (i == 0)
+ goto __err;
+ }
+ }
+ /* tune up the primary codec */
+ snd_ac97_tune_hardware(chip->ac97[0], ac97_quirks, quirk_override);
+ /* enable separate SDINs for ICH4 */
+ if (chip->device_type == DEVICE_INTEL_ICH4)
+ pbus->isdin = 1;
+ /* find the available PCM streams */
+ i = ARRAY_SIZE(ac97_pcm_defs);
+ if (chip->device_type != DEVICE_INTEL_ICH4)
+ i -= 2; /* do not allocate PCM2IN and MIC2 */
+ if (chip->spdif_idx < 0)
+ i--; /* do not allocate S/PDIF */
+ err = snd_ac97_pcm_assign(pbus, i, ac97_pcm_defs);
+ if (err < 0)
+ goto __err;
+ chip->ichd[ICHD_PCMOUT].pcm = &pbus->pcms[0];
+ chip->ichd[ICHD_PCMIN].pcm = &pbus->pcms[1];
+ chip->ichd[ICHD_MIC].pcm = &pbus->pcms[2];
+ if (chip->spdif_idx >= 0)
+ chip->ichd[chip->spdif_idx].pcm = &pbus->pcms[3];
+ if (chip->device_type == DEVICE_INTEL_ICH4) {
+ chip->ichd[ICHD_PCM2IN].pcm = &pbus->pcms[4];
+ chip->ichd[ICHD_MIC2].pcm = &pbus->pcms[5];
+ }
+ /* enable separate SDINs for ICH4 */
+ if (chip->device_type == DEVICE_INTEL_ICH4) {
+ struct ac97_pcm *pcm = chip->ichd[ICHD_PCM2IN].pcm;
+ u8 tmp = igetbyte(chip, ICHREG(SDM));
+ tmp &= ~(ICH_DI2L_MASK|ICH_DI1L_MASK);
+ if (pcm) {
+ tmp |= ICH_SE; /* steer enable for multiple SDINs */
+ tmp |= chip->ac97_sdin[0] << ICH_DI1L_SHIFT;
+ for (i = 1; i < 4; i++) {
+ if (pcm->r[0].codec[i]) {
+ tmp |= chip->ac97_sdin[pcm->r[0].codec[1]->num] << ICH_DI2L_SHIFT;
+ break;
+ }
+ }
+ } else {
+ tmp &= ~ICH_SE; /* steer disable */
+ }
+ iputbyte(chip, ICHREG(SDM), tmp);
+ }
+ if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_PCM_SLEFT)) {
+ chip->multi4 = 1;
+ if (pbus->pcms[0].r[0].slots & (1 << AC97_SLOT_LFE)) {
+ chip->multi6 = 1;
+ if (chip->ac97[0]->flags & AC97_HAS_8CH)
+ chip->multi8 = 1;
+ }
+ }
+ if (pbus->pcms[0].r[1].rslots[0]) {
+ chip->dra = 1;
+ }
+ if (chip->device_type == DEVICE_INTEL_ICH4) {
+ if ((igetdword(chip, ICHREG(GLOB_STA)) & ICH_SAMPLE_CAP) == ICH_SAMPLE_16_20)
+ chip->smp20bit = 1;
+ }
+ if (chip->device_type == DEVICE_NFORCE && !spdif_aclink) {
+ /* 48kHz only */
+ chip->ichd[chip->spdif_idx].pcm->rates = SNDRV_PCM_RATE_48000;
+ }
+ if (chip->device_type == DEVICE_INTEL_ICH4 && !spdif_aclink) {
+ /* use slot 10/11 for SPDIF */
+ u32 val;
+ val = igetdword(chip, ICHREG(GLOB_CNT)) & ~ICH_PCM_SPDIF_MASK;
+ val |= ICH_PCM_SPDIF_1011;
+ iputdword(chip, ICHREG(GLOB_CNT), val);
+ snd_ac97_update_bits(chip->ac97[0], AC97_EXTENDED_STATUS, 0x03 << 4, 0x03 << 4);
+ }
+ chip->in_ac97_init = 0;
+ return 0;
+
+ __err:
+ /* clear the cold-reset bit for the next chance */
+ if (chip->device_type != DEVICE_ALI)
+ iputdword(chip, ICHREG(GLOB_CNT),
+ igetdword(chip, ICHREG(GLOB_CNT)) & ~ICH_AC97COLD);
+ return err;
+}
+
+
+/*
+ *
+ */
+
+static void do_ali_reset(struct intel8x0 *chip)
+{
+ iputdword(chip, ICHREG(ALI_SCR), ICH_ALI_SC_RESET);
+ iputdword(chip, ICHREG(ALI_FIFOCR1), 0x83838383);
+ iputdword(chip, ICHREG(ALI_FIFOCR2), 0x83838383);
+ iputdword(chip, ICHREG(ALI_FIFOCR3), 0x83838383);
+ iputdword(chip, ICHREG(ALI_INTERFACECR),
+ ICH_ALI_IF_PI|ICH_ALI_IF_PO);
+ iputdword(chip, ICHREG(ALI_INTERRUPTCR), 0x00000000);
+ iputdword(chip, ICHREG(ALI_INTERRUPTSR), 0x00000000);
+}
+
+static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
+{
+ unsigned long end_time;
+ unsigned int cnt, status, nstatus;
+
+ /* put logic to right state */
+ /* first clear status bits */
+ status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT;
+ if (chip->device_type == DEVICE_NFORCE)
+ status |= ICH_NVSPINT;
+ cnt = igetdword(chip, ICHREG(GLOB_STA));
+ iputdword(chip, ICHREG(GLOB_STA), cnt & status);
+
+ /* ACLink on, 2 channels */
+ cnt = igetdword(chip, ICHREG(GLOB_CNT));
+ cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+ /* do cold reset - the full ac97 powerdown may leave the controller
+ * in a warm state but actually it cannot communicate with the codec.
+ */
+ iputdword(chip, ICHREG(GLOB_CNT), cnt & ~ICH_AC97COLD);
+ cnt = igetdword(chip, ICHREG(GLOB_CNT));
+ udelay(10);
+ iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD);
+ msleep(1);
+#else
+ /* finish cold or do warm reset */
+ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM;
+ iputdword(chip, ICHREG(GLOB_CNT), cnt);
+ end_time = (jiffies + (HZ / 4)) + 1;
+ do {
+ if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0)
+ goto __ok;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n",
+ igetdword(chip, ICHREG(GLOB_CNT)));
+ return -EIO;
+
+ __ok:
+#endif
+ if (probing) {
+ /* wait for any codec ready status.
+ * Once it becomes ready it should remain ready
+ * as long as we do not disable the ac97 link.
+ */
+ end_time = jiffies + HZ;
+ do {
+ status = igetdword(chip, ICHREG(GLOB_STA)) &
+ chip->codec_isr_bits;
+ if (status)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ if (! status) {
+ /* no codec is found */
+ snd_printk(KERN_ERR "codec_ready: codec is not ready [0x%x]\n",
+ igetdword(chip, ICHREG(GLOB_STA)));
+ return -EIO;
+ }
+
+ /* wait for other codecs ready status. */
+ end_time = jiffies + HZ / 4;
+ while (status != chip->codec_isr_bits &&
+ time_after_eq(end_time, jiffies)) {
+ schedule_timeout_uninterruptible(1);
+ status |= igetdword(chip, ICHREG(GLOB_STA)) &
+ chip->codec_isr_bits;
+ }
+
+ } else {
+ /* resume phase */
+ int i;
+ status = 0;
+ for (i = 0; i < chip->ncodecs; i++)
+ if (chip->ac97[i])
+ status |= chip->codec_bit[chip->ac97_sdin[i]];
+ /* wait until all the probed codecs are ready */
+ end_time = jiffies + HZ;
+ do {
+ nstatus = igetdword(chip, ICHREG(GLOB_STA)) &
+ chip->codec_isr_bits;
+ if (status == nstatus)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ }
+
+ if (chip->device_type == DEVICE_SIS) {
+ /* unmute the output on SIS7012 */
+ iputword(chip, 0x4c, igetword(chip, 0x4c) | 1);
+ }
+ if (chip->device_type == DEVICE_NFORCE && !spdif_aclink) {
+ /* enable SPDIF interrupt */
+ unsigned int val;
+ pci_read_config_dword(chip->pci, 0x4c, &val);
+ val |= 0x1000000;
+ pci_write_config_dword(chip->pci, 0x4c, val);
+ }
+ return 0;
+}
+
+static int snd_intel8x0_ali_chip_init(struct intel8x0 *chip, int probing)
+{
+ u32 reg;
+ int i = 0;
+
+ reg = igetdword(chip, ICHREG(ALI_SCR));
+ if ((reg & 2) == 0) /* Cold required */
+ reg |= 2;
+ else
+ reg |= 1; /* Warm */
+ reg &= ~0x80000000; /* ACLink on */
+ iputdword(chip, ICHREG(ALI_SCR), reg);
+
+ for (i = 0; i < HZ / 2; i++) {
+ if (! (igetdword(chip, ICHREG(ALI_INTERRUPTSR)) & ALI_INT_GPIO))
+ goto __ok;
+ schedule_timeout_uninterruptible(1);
+ }
+ snd_printk(KERN_ERR "AC'97 reset failed.\n");
+ if (probing)
+ return -EIO;
+
+ __ok:
+ for (i = 0; i < HZ / 2; i++) {
+ reg = igetdword(chip, ICHREG(ALI_RTSR));
+ if (reg & 0x80) /* primary codec */
+ break;
+ iputdword(chip, ICHREG(ALI_RTSR), reg | 0x80);
+ schedule_timeout_uninterruptible(1);
+ }
+
+ do_ali_reset(chip);
+ return 0;
+}
+
+static int snd_intel8x0_chip_init(struct intel8x0 *chip, int probing)
+{
+ unsigned int i, timeout;
+ int err;
+
+ if (chip->device_type != DEVICE_ALI) {
+ if ((err = snd_intel8x0_ich_chip_init(chip, probing)) < 0)
+ return err;
+ iagetword(chip, 0); /* clear semaphore flag */
+ } else {
+ if ((err = snd_intel8x0_ali_chip_init(chip, probing)) < 0)
+ return err;
+ }
+
+ /* disable interrupts */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, 0x00);
+ /* reset channels */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS);
+ for (i = 0; i < chip->bdbars_count; i++) {
+ timeout = 100000;
+ while (--timeout != 0) {
+ if ((igetbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset) & ICH_RESETREGS) == 0)
+ break;
+ }
+ if (timeout == 0)
+ printk(KERN_ERR "intel8x0: reset of registers failed?\n");
+ }
+ /* initialize Buffer Descriptor Lists */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputdword(chip, ICH_REG_OFF_BDBAR + chip->ichd[i].reg_offset,
+ chip->ichd[i].bdbar_addr);
+ return 0;
+}
+
+static int snd_intel8x0_free(struct intel8x0 *chip)
+{
+ unsigned int i;
+
+ if (chip->irq < 0)
+ goto __hw_end;
+ /* disable interrupts */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, 0x00);
+ /* reset channels */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS);
+ if (chip->device_type == DEVICE_NFORCE && !spdif_aclink) {
+ /* stop the spdif interrupt */
+ unsigned int val;
+ pci_read_config_dword(chip->pci, 0x4c, &val);
+ val &= ~0x1000000;
+ pci_write_config_dword(chip->pci, 0x4c, val);
+ }
+ /* --- */
+
+ __hw_end:
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (chip->bdbars.area) {
+ if (chip->fix_nocache)
+ fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 0);
+ snd_dma_free_pages(&chip->bdbars);
+ }
+ if (chip->addr)
+ pci_iounmap(chip->pci, chip->addr);
+ if (chip->bmaddr)
+ pci_iounmap(chip->pci, chip->bmaddr);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct intel8x0 *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < chip->pcm_devs; i++)
+ snd_pcm_suspend_all(chip->pcm[i]);
+ /* clear nocache */
+ if (chip->fix_nocache) {
+ for (i = 0; i < chip->bdbars_count; i++) {
+ struct ichdev *ichdev = &chip->ichd[i];
+ if (ichdev->substream && ichdev->page_attr_changed) {
+ struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
+ if (runtime->dma_area)
+ fill_nocache(runtime->dma_area, runtime->dma_bytes, 0);
+ }
+ }
+ }
+ for (i = 0; i < chip->ncodecs; i++)
+ snd_ac97_suspend(chip->ac97[i]);
+ if (chip->device_type == DEVICE_INTEL_ICH4)
+ chip->sdm_saved = igetbyte(chip, ICHREG(SDM));
+
+ if (chip->irq >= 0) {
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ }
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ /* The call below may disable built-in speaker on some laptops
+ * after S2RAM. So, don't touch it.
+ */
+ /* pci_set_power_state(pci, pci_choose_state(pci, state)); */
+ return 0;
+}
+
+static int intel8x0_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct intel8x0 *chip = card->private_data;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "intel8x0: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+ snd_intel8x0_chip_init(chip, 0);
+ if (request_irq(pci->irq, snd_intel8x0_interrupt,
+ IRQF_SHARED, card->shortname, chip)) {
+ printk(KERN_ERR "intel8x0: unable to grab IRQ %d, "
+ "disabling device\n", pci->irq);
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ chip->irq = pci->irq;
+ synchronize_irq(chip->irq);
+
+ /* re-initialize mixer stuff */
+ if (chip->device_type == DEVICE_INTEL_ICH4 && !spdif_aclink) {
+ /* enable separate SDINs for ICH4 */
+ iputbyte(chip, ICHREG(SDM), chip->sdm_saved);
+ /* use slot 10/11 for SPDIF */
+ iputdword(chip, ICHREG(GLOB_CNT),
+ (igetdword(chip, ICHREG(GLOB_CNT)) & ~ICH_PCM_SPDIF_MASK) |
+ ICH_PCM_SPDIF_1011);
+ }
+
+ /* refill nocache */
+ if (chip->fix_nocache)
+ fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
+
+ for (i = 0; i < chip->ncodecs; i++)
+ snd_ac97_resume(chip->ac97[i]);
+
+ /* refill nocache */
+ if (chip->fix_nocache) {
+ for (i = 0; i < chip->bdbars_count; i++) {
+ struct ichdev *ichdev = &chip->ichd[i];
+ if (ichdev->substream && ichdev->page_attr_changed) {
+ struct snd_pcm_runtime *runtime = ichdev->substream->runtime;
+ if (runtime->dma_area)
+ fill_nocache(runtime->dma_area, runtime->dma_bytes, 1);
+ }
+ }
+ }
+
+ /* resume status */
+ for (i = 0; i < chip->bdbars_count; i++) {
+ struct ichdev *ichdev = &chip->ichd[i];
+ unsigned long port = ichdev->reg_offset;
+ if (! ichdev->substream || ! ichdev->suspended)
+ continue;
+ if (ichdev->ichd == ICHD_PCMOUT)
+ snd_intel8x0_setup_pcm_out(chip, ichdev->substream->runtime);
+ iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr);
+ iputbyte(chip, port + ICH_REG_OFF_LVI, ichdev->lvi);
+ iputbyte(chip, port + ICH_REG_OFF_CIV, ichdev->civ);
+ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
+ }
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+#define INTEL8X0_TESTBUF_SIZE 32768 /* enough large for one shot */
+
+static void __devinit intel8x0_measure_ac97_clock(struct intel8x0 *chip)
+{
+ struct snd_pcm_substream *subs;
+ struct ichdev *ichdev;
+ unsigned long port;
+ unsigned long pos, t;
+ struct timeval start_time, stop_time;
+
+ if (chip->ac97_bus->clock != 48000)
+ return; /* specified in module option */
+
+ subs = chip->pcm[0]->streams[0].substream;
+ if (! subs || subs->dma_buffer.bytes < INTEL8X0_TESTBUF_SIZE) {
+ snd_printk(KERN_WARNING "no playback buffer allocated - aborting measure ac97 clock\n");
+ return;
+ }
+ ichdev = &chip->ichd[ICHD_PCMOUT];
+ ichdev->physbuf = subs->dma_buffer.addr;
+ ichdev->size = chip->ichd[ICHD_PCMOUT].fragsize = INTEL8X0_TESTBUF_SIZE;
+ ichdev->substream = NULL; /* don't process interrupts */
+
+ /* set rate */
+ if (snd_ac97_set_rate(chip->ac97[0], AC97_PCM_FRONT_DAC_RATE, 48000) < 0) {
+ snd_printk(KERN_ERR "cannot set ac97 rate: clock = %d\n", chip->ac97_bus->clock);
+ return;
+ }
+ snd_intel8x0_setup_periods(chip, ichdev);
+ port = ichdev->reg_offset;
+ spin_lock_irq(&chip->reg_lock);
+ chip->in_measurement = 1;
+ /* trigger */
+ if (chip->device_type != DEVICE_ALI)
+ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_IOCE | ICH_STARTBM);
+ else {
+ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_IOCE);
+ iputdword(chip, ICHREG(ALI_DMACR), 1 << ichdev->ali_slot);
+ }
+ do_gettimeofday(&start_time);
+ spin_unlock_irq(&chip->reg_lock);
+ msleep(50);
+ spin_lock_irq(&chip->reg_lock);
+ /* check the position */
+ pos = ichdev->fragsize1;
+ pos -= igetword(chip, ichdev->reg_offset + ichdev->roff_picb) << ichdev->pos_shift;
+ pos += ichdev->position;
+ chip->in_measurement = 0;
+ do_gettimeofday(&stop_time);
+ /* stop */
+ if (chip->device_type == DEVICE_ALI) {
+ iputdword(chip, ICHREG(ALI_DMACR), 1 << (ichdev->ali_slot + 16));
+ iputbyte(chip, port + ICH_REG_OFF_CR, 0);
+ while (igetbyte(chip, port + ICH_REG_OFF_CR))
+ ;
+ } else {
+ iputbyte(chip, port + ICH_REG_OFF_CR, 0);
+ while (!(igetbyte(chip, port + ichdev->roff_sr) & ICH_DCH))
+ ;
+ }
+ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_RESETREGS);
+ spin_unlock_irq(&chip->reg_lock);
+
+ t = stop_time.tv_sec - start_time.tv_sec;
+ t *= 1000000;
+ t += stop_time.tv_usec - start_time.tv_usec;
+ printk(KERN_INFO "%s: measured %lu usecs\n", __func__, t);
+ if (t == 0) {
+ snd_printk(KERN_ERR "?? calculation error..\n");
+ return;
+ }
+ pos = (pos / 4) * 1000;
+ pos = (pos / t) * 1000 + ((pos % t) * 1000) / t;
+ if (pos < 40000 || pos >= 60000)
+ /* abnormal value. hw problem? */
+ printk(KERN_INFO "intel8x0: measured clock %ld rejected\n", pos);
+ else if (pos < 47500 || pos > 48500)
+ /* not 48000Hz, tuning the clock.. */
+ chip->ac97_bus->clock = (chip->ac97_bus->clock * 48000) / pos;
+ printk(KERN_INFO "intel8x0: clocking to %d\n", chip->ac97_bus->clock);
+ snd_ac97_update_power(chip->ac97[0], AC97_PCM_FRONT_DAC_RATE, 0);
+}
+
+static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000),
+ SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100),
+ SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000),
+ SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),
+ SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000),
+ { } /* terminator */
+};
+
+static int __devinit intel8x0_in_clock_list(struct intel8x0 *chip)
+{
+ struct pci_dev *pci = chip->pci;
+ const struct snd_pci_quirk *wl;
+
+ wl = snd_pci_quirk_lookup(pci, intel8x0_clock_list);
+ if (!wl)
+ return 0;
+ printk(KERN_INFO "intel8x0: white list rate for %04x:%04x is %i\n",
+ pci->subsystem_vendor, pci->subsystem_device, wl->value);
+ chip->ac97_bus->clock = wl->value;
+ return 1;
+}
+
+#ifdef CONFIG_PROC_FS
+static void snd_intel8x0_proc_read(struct snd_info_entry * entry,
+ struct snd_info_buffer *buffer)
+{
+ struct intel8x0 *chip = entry->private_data;
+ unsigned int tmp;
+
+ snd_iprintf(buffer, "Intel8x0\n\n");
+ if (chip->device_type == DEVICE_ALI)
+ return;
+ tmp = igetdword(chip, ICHREG(GLOB_STA));
+ snd_iprintf(buffer, "Global control : 0x%08x\n", igetdword(chip, ICHREG(GLOB_CNT)));
+ snd_iprintf(buffer, "Global status : 0x%08x\n", tmp);
+ if (chip->device_type == DEVICE_INTEL_ICH4)
+ snd_iprintf(buffer, "SDM : 0x%08x\n", igetdword(chip, ICHREG(SDM)));
+ snd_iprintf(buffer, "AC'97 codecs ready :");
+ if (tmp & chip->codec_isr_bits) {
+ int i;
+ static const char *codecs[3] = {
+ "primary", "secondary", "tertiary"
+ };
+ for (i = 0; i < chip->max_codecs; i++)
+ if (tmp & chip->codec_bit[i])
+ snd_iprintf(buffer, " %s", codecs[i]);
+ } else
+ snd_iprintf(buffer, " none");
+ snd_iprintf(buffer, "\n");
+ if (chip->device_type == DEVICE_INTEL_ICH4 ||
+ chip->device_type == DEVICE_SIS)
+ snd_iprintf(buffer, "AC'97 codecs SDIN : %i %i %i\n",
+ chip->ac97_sdin[0],
+ chip->ac97_sdin[1],
+ chip->ac97_sdin[2]);
+}
+
+static void __devinit snd_intel8x0_proc_init(struct intel8x0 * chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "intel8x0", &entry))
+ snd_info_set_text_ops(entry, chip, snd_intel8x0_proc_read);
+}
+#else
+#define snd_intel8x0_proc_init(x)
+#endif
+
+static int snd_intel8x0_dev_free(struct snd_device *device)
+{
+ struct intel8x0 *chip = device->device_data;
+ return snd_intel8x0_free(chip);
+}
+
+struct ich_reg_info {
+ unsigned int int_sta_mask;
+ unsigned int offset;
+};
+
+static unsigned int ich_codec_bits[3] = {
+ ICH_PCR, ICH_SCR, ICH_TCR
+};
+static unsigned int sis_codec_bits[3] = {
+ ICH_PCR, ICH_SCR, ICH_SIS_TCR
+};
+
+static int __devinit snd_intel8x0_create(struct snd_card *card,
+ struct pci_dev *pci,
+ unsigned long device_type,
+ struct intel8x0 ** r_intel8x0)
+{
+ struct intel8x0 *chip;
+ int err;
+ unsigned int i;
+ unsigned int int_sta_masks;
+ struct ichdev *ichdev;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_intel8x0_dev_free,
+ };
+
+ static unsigned int bdbars[] = {
+ 3, /* DEVICE_INTEL */
+ 6, /* DEVICE_INTEL_ICH4 */
+ 3, /* DEVICE_SIS */
+ 6, /* DEVICE_ALI */
+ 4, /* DEVICE_NFORCE */
+ };
+ static struct ich_reg_info intel_regs[6] = {
+ { ICH_PIINT, 0 },
+ { ICH_POINT, 0x10 },
+ { ICH_MCINT, 0x20 },
+ { ICH_M2INT, 0x40 },
+ { ICH_P2INT, 0x50 },
+ { ICH_SPINT, 0x60 },
+ };
+ static struct ich_reg_info nforce_regs[4] = {
+ { ICH_PIINT, 0 },
+ { ICH_POINT, 0x10 },
+ { ICH_MCINT, 0x20 },
+ { ICH_NVSPINT, 0x70 },
+ };
+ static struct ich_reg_info ali_regs[6] = {
+ { ALI_INT_PCMIN, 0x40 },
+ { ALI_INT_PCMOUT, 0x50 },
+ { ALI_INT_MICIN, 0x60 },
+ { ALI_INT_CODECSPDIFOUT, 0x70 },
+ { ALI_INT_SPDIFIN, 0xa0 },
+ { ALI_INT_SPDIFOUT, 0xb0 },
+ };
+ struct ich_reg_info *tbl;
+
+ *r_intel8x0 = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&chip->reg_lock);
+ chip->device_type = device_type;
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* module parameters */
+ chip->buggy_irq = buggy_irq;
+ chip->buggy_semaphore = buggy_semaphore;
+ if (xbox)
+ chip->xbox = 1;
+
+ if (pci->vendor == PCI_VENDOR_ID_INTEL &&
+ pci->device == PCI_DEVICE_ID_INTEL_440MX)
+ chip->fix_nocache = 1; /* enable workaround */
+
+ if ((err = pci_request_regions(pci, card->shortname)) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+
+ if (device_type == DEVICE_ALI) {
+ /* ALI5455 has no ac97 region */
+ chip->bmaddr = pci_iomap(pci, 0, 0);
+ goto port_inited;
+ }
+
+ if (pci_resource_flags(pci, 2) & IORESOURCE_MEM) /* ICH4 and Nforce */
+ chip->addr = pci_iomap(pci, 2, 0);
+ else
+ chip->addr = pci_iomap(pci, 0, 0);
+ if (!chip->addr) {
+ snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
+ snd_intel8x0_free(chip);
+ return -EIO;
+ }
+ if (pci_resource_flags(pci, 3) & IORESOURCE_MEM) /* ICH4 */
+ chip->bmaddr = pci_iomap(pci, 3, 0);
+ else
+ chip->bmaddr = pci_iomap(pci, 1, 0);
+ if (!chip->bmaddr) {
+ snd_printk(KERN_ERR "Controller space ioremap problem\n");
+ snd_intel8x0_free(chip);
+ return -EIO;
+ }
+
+ port_inited:
+ chip->bdbars_count = bdbars[device_type];
+
+ /* initialize offsets */
+ switch (device_type) {
+ case DEVICE_NFORCE:
+ tbl = nforce_regs;
+ break;
+ case DEVICE_ALI:
+ tbl = ali_regs;
+ break;
+ default:
+ tbl = intel_regs;
+ break;
+ }
+ for (i = 0; i < chip->bdbars_count; i++) {
+ ichdev = &chip->ichd[i];
+ ichdev->ichd = i;
+ ichdev->reg_offset = tbl[i].offset;
+ ichdev->int_sta_mask = tbl[i].int_sta_mask;
+ if (device_type == DEVICE_SIS) {
+ /* SiS 7012 swaps the registers */
+ ichdev->roff_sr = ICH_REG_OFF_PICB;
+ ichdev->roff_picb = ICH_REG_OFF_SR;
+ } else {
+ ichdev->roff_sr = ICH_REG_OFF_SR;
+ ichdev->roff_picb = ICH_REG_OFF_PICB;
+ }
+ if (device_type == DEVICE_ALI)
+ ichdev->ali_slot = (ichdev->reg_offset - 0x40) / 0x10;
+ /* SIS7012 handles the pcm data in bytes, others are in samples */
+ ichdev->pos_shift = (device_type == DEVICE_SIS) ? 0 : 1;
+ }
+
+ /* allocate buffer descriptor lists */
+ /* the start of each lists must be aligned to 8 bytes */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2,
+ &chip->bdbars) < 0) {
+ snd_intel8x0_free(chip);
+ snd_printk(KERN_ERR "intel8x0: cannot allocate buffer descriptors\n");
+ return -ENOMEM;
+ }
+ /* tables must be aligned to 8 bytes here, but the kernel pages
+ are much bigger, so we don't care (on i386) */
+ /* workaround for 440MX */
+ if (chip->fix_nocache)
+ fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1);
+ int_sta_masks = 0;
+ for (i = 0; i < chip->bdbars_count; i++) {
+ ichdev = &chip->ichd[i];
+ ichdev->bdbar = ((u32 *)chip->bdbars.area) +
+ (i * ICH_MAX_FRAGS * 2);
+ ichdev->bdbar_addr = chip->bdbars.addr +
+ (i * sizeof(u32) * ICH_MAX_FRAGS * 2);
+ int_sta_masks |= ichdev->int_sta_mask;
+ }
+ chip->int_sta_reg = device_type == DEVICE_ALI ?
+ ICH_REG_ALI_INTERRUPTSR : ICH_REG_GLOB_STA;
+ chip->int_sta_mask = int_sta_masks;
+
+ pci_set_master(pci);
+
+ switch(chip->device_type) {
+ case DEVICE_INTEL_ICH4:
+ /* ICH4 can have three codecs */
+ chip->max_codecs = 3;
+ chip->codec_bit = ich_codec_bits;
+ chip->codec_ready_bits = ICH_PRI | ICH_SRI | ICH_TRI;
+ break;
+ case DEVICE_SIS:
+ /* recent SIS7012 can have three codecs */
+ chip->max_codecs = 3;
+ chip->codec_bit = sis_codec_bits;
+ chip->codec_ready_bits = ICH_PRI | ICH_SRI | ICH_SIS_TRI;
+ break;
+ default:
+ /* others up to two codecs */
+ chip->max_codecs = 2;
+ chip->codec_bit = ich_codec_bits;
+ chip->codec_ready_bits = ICH_PRI | ICH_SRI;
+ break;
+ }
+ for (i = 0; i < chip->max_codecs; i++)
+ chip->codec_isr_bits |= chip->codec_bit[i];
+
+ if ((err = snd_intel8x0_chip_init(chip, 1)) < 0) {
+ snd_intel8x0_free(chip);
+ return err;
+ }
+
+ /* request irq after initializaing int_sta_mask, etc */
+ if (request_irq(pci->irq, snd_intel8x0_interrupt,
+ IRQF_SHARED, card->shortname, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_intel8x0_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_intel8x0_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_intel8x0 = chip;
+ return 0;
+}
+
+static struct shortname_table {
+ unsigned int id;
+ const char *s;
+} shortnames[] __devinitdata = {
+ { PCI_DEVICE_ID_INTEL_82801AA_5, "Intel 82801AA-ICH" },
+ { PCI_DEVICE_ID_INTEL_82801AB_5, "Intel 82901AB-ICH0" },
+ { PCI_DEVICE_ID_INTEL_82801BA_4, "Intel 82801BA-ICH2" },
+ { PCI_DEVICE_ID_INTEL_440MX, "Intel 440MX" },
+ { PCI_DEVICE_ID_INTEL_82801CA_5, "Intel 82801CA-ICH3" },
+ { PCI_DEVICE_ID_INTEL_82801DB_5, "Intel 82801DB-ICH4" },
+ { PCI_DEVICE_ID_INTEL_82801EB_5, "Intel ICH5" },
+ { PCI_DEVICE_ID_INTEL_ESB_5, "Intel 6300ESB" },
+ { PCI_DEVICE_ID_INTEL_ICH6_18, "Intel ICH6" },
+ { PCI_DEVICE_ID_INTEL_ICH7_20, "Intel ICH7" },
+ { PCI_DEVICE_ID_INTEL_ESB2_14, "Intel ESB2" },
+ { PCI_DEVICE_ID_SI_7012, "SiS SI7012" },
+ { PCI_DEVICE_ID_NVIDIA_MCP1_AUDIO, "NVidia nForce" },
+ { PCI_DEVICE_ID_NVIDIA_MCP2_AUDIO, "NVidia nForce2" },
+ { PCI_DEVICE_ID_NVIDIA_MCP3_AUDIO, "NVidia nForce3" },
+ { PCI_DEVICE_ID_NVIDIA_CK8S_AUDIO, "NVidia CK8S" },
+ { PCI_DEVICE_ID_NVIDIA_CK804_AUDIO, "NVidia CK804" },
+ { PCI_DEVICE_ID_NVIDIA_CK8_AUDIO, "NVidia CK8" },
+ { 0x003a, "NVidia MCP04" },
+ { 0x746d, "AMD AMD8111" },
+ { 0x7445, "AMD AMD768" },
+ { 0x5455, "ALi M5455" },
+ { 0, NULL },
+};
+
+static struct snd_pci_quirk spdif_aclink_defaults[] __devinitdata = {
+ SND_PCI_QUIRK(0x147b, 0x1c1a, "ASUS KN8", 1),
+ { } /* end */
+};
+
+/* look up white/black list for SPDIF over ac-link */
+static int __devinit check_default_spdif_aclink(struct pci_dev *pci)
+{
+ const struct snd_pci_quirk *w;
+
+ w = snd_pci_quirk_lookup(pci, spdif_aclink_defaults);
+ if (w) {
+ if (w->value)
+ snd_printdd(KERN_INFO "intel8x0: Using SPDIF over "
+ "AC-Link for %s\n", w->name);
+ else
+ snd_printdd(KERN_INFO "intel8x0: Using integrated "
+ "SPDIF DMA for %s\n", w->name);
+ return w->value;
+ }
+ return 0;
+}
+
+static int __devinit snd_intel8x0_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct intel8x0 *chip;
+ int err;
+ struct shortname_table *name;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if (spdif_aclink < 0)
+ spdif_aclink = check_default_spdif_aclink(pci);
+
+ strcpy(card->driver, "ICH");
+ if (!spdif_aclink) {
+ switch (pci_id->driver_data) {
+ case DEVICE_NFORCE:
+ strcpy(card->driver, "NFORCE");
+ break;
+ case DEVICE_INTEL_ICH4:
+ strcpy(card->driver, "ICH4");
+ }
+ }
+
+ strcpy(card->shortname, "Intel ICH");
+ for (name = shortnames; name->id; name++) {
+ if (pci->device == name->id) {
+ strcpy(card->shortname, name->s);
+ break;
+ }
+ }
+
+ if (buggy_irq < 0) {
+ /* some Nforce[2] and ICH boards have problems with IRQ handling.
+ * Needs to return IRQ_HANDLED for unknown irqs.
+ */
+ if (pci_id->driver_data == DEVICE_NFORCE)
+ buggy_irq = 1;
+ else
+ buggy_irq = 0;
+ }
+
+ if ((err = snd_intel8x0_create(card, pci, pci_id->driver_data,
+ &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ if ((err = snd_intel8x0_mixer(chip, ac97_clock, ac97_quirk)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_intel8x0_pcm(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_intel8x0_proc_init(chip);
+
+ snprintf(card->longname, sizeof(card->longname),
+ "%s with %s at irq %i", card->shortname,
+ snd_ac97_get_short_name(chip->ac97[0]), chip->irq);
+
+ if (ac97_clock == 0 || ac97_clock == 1) {
+ if (ac97_clock == 0) {
+ if (intel8x0_in_clock_list(chip) == 0)
+ intel8x0_measure_ac97_clock(chip);
+ } else {
+ intel8x0_measure_ac97_clock(chip);
+ }
+ }
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ return 0;
+}
+
+static void __devexit snd_intel8x0_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Intel ICH",
+ .id_table = snd_intel8x0_ids,
+ .probe = snd_intel8x0_probe,
+ .remove = __devexit_p(snd_intel8x0_remove),
+#ifdef CONFIG_PM
+ .suspend = intel8x0_suspend,
+ .resume = intel8x0_resume,
+#endif
+};
+
+
+static int __init alsa_card_intel8x0_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_intel8x0_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_intel8x0_init)
+module_exit(alsa_card_intel8x0_exit)
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
new file mode 100644
index 0000000..93449e4
--- /dev/null
+++ b/sound/pci/intel8x0m.c
@@ -0,0 +1,1343 @@
+/*
+ * ALSA modem driver for Intel ICH (i8x0) chipsets
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ *
+ * This is modified (by Sasha Khapyorsky <sashak@alsa-project.org>) version
+ * of ALSA ICH sound driver intel8x0.c .
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; "
+ "SiS 7013; NVidia MCP/2/2S/3 modems");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH},"
+ "{Intel,82901AB-ICH0},"
+ "{Intel,82801BA-ICH2},"
+ "{Intel,82801CA-ICH3},"
+ "{Intel,82801DB-ICH4},"
+ "{Intel,ICH5},"
+ "{Intel,ICH6},"
+ "{Intel,ICH7},"
+ "{Intel,MX440},"
+ "{SiS,7013},"
+ "{NVidia,NForce Modem},"
+ "{NVidia,NForce2 Modem},"
+ "{NVidia,NForce2s Modem},"
+ "{NVidia,NForce3 Modem},"
+ "{AMD,AMD768}}");
+
+static int index = -2; /* Exclude the first card */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int ac97_clock;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for Intel i8x0 modemcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for Intel i8x0 modemcard.");
+module_param(ac97_clock, int, 0444);
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = auto-detect).");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
+/*
+ * Direct registers
+ */
+enum { DEVICE_INTEL, DEVICE_SIS, DEVICE_ALI, DEVICE_NFORCE };
+
+#define ICHREG(x) ICH_REG_##x
+
+#define DEFINE_REGSET(name,base) \
+enum { \
+ ICH_REG_##name##_BDBAR = base + 0x0, /* dword - buffer descriptor list base address */ \
+ ICH_REG_##name##_CIV = base + 0x04, /* byte - current index value */ \
+ ICH_REG_##name##_LVI = base + 0x05, /* byte - last valid index */ \
+ ICH_REG_##name##_SR = base + 0x06, /* byte - status register */ \
+ ICH_REG_##name##_PICB = base + 0x08, /* word - position in current buffer */ \
+ ICH_REG_##name##_PIV = base + 0x0a, /* byte - prefetched index value */ \
+ ICH_REG_##name##_CR = base + 0x0b, /* byte - control register */ \
+};
+
+/* busmaster blocks */
+DEFINE_REGSET(OFF, 0); /* offset */
+
+/* values for each busmaster block */
+
+/* LVI */
+#define ICH_REG_LVI_MASK 0x1f
+
+/* SR */
+#define ICH_FIFOE 0x10 /* FIFO error */
+#define ICH_BCIS 0x08 /* buffer completion interrupt status */
+#define ICH_LVBCI 0x04 /* last valid buffer completion interrupt */
+#define ICH_CELV 0x02 /* current equals last valid */
+#define ICH_DCH 0x01 /* DMA controller halted */
+
+/* PIV */
+#define ICH_REG_PIV_MASK 0x1f /* mask */
+
+/* CR */
+#define ICH_IOCE 0x10 /* interrupt on completion enable */
+#define ICH_FEIE 0x08 /* fifo error interrupt enable */
+#define ICH_LVBIE 0x04 /* last valid buffer interrupt enable */
+#define ICH_RESETREGS 0x02 /* reset busmaster registers */
+#define ICH_STARTBM 0x01 /* start busmaster operation */
+
+
+/* global block */
+#define ICH_REG_GLOB_CNT 0x3c /* dword - global control */
+#define ICH_TRIE 0x00000040 /* tertiary resume interrupt enable */
+#define ICH_SRIE 0x00000020 /* secondary resume interrupt enable */
+#define ICH_PRIE 0x00000010 /* primary resume interrupt enable */
+#define ICH_ACLINK 0x00000008 /* AClink shut off */
+#define ICH_AC97WARM 0x00000004 /* AC'97 warm reset */
+#define ICH_AC97COLD 0x00000002 /* AC'97 cold reset */
+#define ICH_GIE 0x00000001 /* GPI interrupt enable */
+#define ICH_REG_GLOB_STA 0x40 /* dword - global status */
+#define ICH_TRI 0x20000000 /* ICH4: tertiary (AC_SDIN2) resume interrupt */
+#define ICH_TCR 0x10000000 /* ICH4: tertiary (AC_SDIN2) codec ready */
+#define ICH_BCS 0x08000000 /* ICH4: bit clock stopped */
+#define ICH_SPINT 0x04000000 /* ICH4: S/PDIF interrupt */
+#define ICH_P2INT 0x02000000 /* ICH4: PCM2-In interrupt */
+#define ICH_M2INT 0x01000000 /* ICH4: Mic2-In interrupt */
+#define ICH_SAMPLE_CAP 0x00c00000 /* ICH4: sample capability bits (RO) */
+#define ICH_MULTICHAN_CAP 0x00300000 /* ICH4: multi-channel capability bits (RO) */
+#define ICH_MD3 0x00020000 /* modem power down semaphore */
+#define ICH_AD3 0x00010000 /* audio power down semaphore */
+#define ICH_RCS 0x00008000 /* read completion status */
+#define ICH_BIT3 0x00004000 /* bit 3 slot 12 */
+#define ICH_BIT2 0x00002000 /* bit 2 slot 12 */
+#define ICH_BIT1 0x00001000 /* bit 1 slot 12 */
+#define ICH_SRI 0x00000800 /* secondary (AC_SDIN1) resume interrupt */
+#define ICH_PRI 0x00000400 /* primary (AC_SDIN0) resume interrupt */
+#define ICH_SCR 0x00000200 /* secondary (AC_SDIN1) codec ready */
+#define ICH_PCR 0x00000100 /* primary (AC_SDIN0) codec ready */
+#define ICH_MCINT 0x00000080 /* MIC capture interrupt */
+#define ICH_POINT 0x00000040 /* playback interrupt */
+#define ICH_PIINT 0x00000020 /* capture interrupt */
+#define ICH_NVSPINT 0x00000010 /* nforce spdif interrupt */
+#define ICH_MOINT 0x00000004 /* modem playback interrupt */
+#define ICH_MIINT 0x00000002 /* modem capture interrupt */
+#define ICH_GSCI 0x00000001 /* GPI status change interrupt */
+#define ICH_REG_ACC_SEMA 0x44 /* byte - codec write semaphore */
+#define ICH_CAS 0x01 /* codec access semaphore */
+
+#define ICH_MAX_FRAGS 32 /* max hw frags */
+
+
+/*
+ *
+ */
+
+enum { ICHD_MDMIN, ICHD_MDMOUT, ICHD_MDMLAST = ICHD_MDMOUT };
+enum { ALID_MDMIN, ALID_MDMOUT, ALID_MDMLAST = ALID_MDMOUT };
+
+#define get_ichdev(substream) (substream->runtime->private_data)
+
+struct ichdev {
+ unsigned int ichd; /* ich device number */
+ unsigned long reg_offset; /* offset to bmaddr */
+ u32 *bdbar; /* CPU address (32bit) */
+ unsigned int bdbar_addr; /* PCI bus address (32bit) */
+ struct snd_pcm_substream *substream;
+ unsigned int physbuf; /* physical address (32bit) */
+ unsigned int size;
+ unsigned int fragsize;
+ unsigned int fragsize1;
+ unsigned int position;
+ int frags;
+ int lvi;
+ int lvi_frag;
+ int civ;
+ int ack;
+ int ack_reload;
+ unsigned int ack_bit;
+ unsigned int roff_sr;
+ unsigned int roff_picb;
+ unsigned int int_sta_mask; /* interrupt status mask */
+ unsigned int ali_slot; /* ALI DMA slot */
+ struct snd_ac97 *ac97;
+};
+
+struct intel8x0m {
+ unsigned int device_type;
+
+ int irq;
+
+ void __iomem *addr;
+ void __iomem *bmaddr;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+
+ int pcm_devs;
+ struct snd_pcm *pcm[2];
+ struct ichdev ichd[2];
+
+ unsigned int in_ac97_init: 1;
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97;
+
+ spinlock_t reg_lock;
+
+ struct snd_dma_buffer bdbars;
+ u32 bdbars_count;
+ u32 int_sta_reg; /* interrupt status register */
+ u32 int_sta_mask; /* interrupt status mask */
+ unsigned int pcm_pos_shift;
+};
+
+static struct pci_device_id snd_intel8x0m_ids[] = {
+ { 0x8086, 0x2416, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */
+ { 0x8086, 0x2426, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */
+ { 0x8086, 0x2446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */
+ { 0x8086, 0x2486, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */
+ { 0x8086, 0x24c6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH4 */
+ { 0x8086, 0x24d6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH5 */
+ { 0x8086, 0x266d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH6 */
+ { 0x8086, 0x27dd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH7 */
+ { 0x8086, 0x7196, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */
+ { 0x1022, 0x7446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */
+ { 0x1039, 0x7013, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7013 */
+ { 0x10de, 0x01c1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */
+ { 0x10de, 0x0069, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */
+ { 0x10de, 0x0089, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2s */
+ { 0x10de, 0x00d9, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */
+#if 0
+ { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */
+ { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */
+#endif
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_intel8x0m_ids);
+
+/*
+ * Lowlevel I/O - busmaster
+ */
+
+static inline u8 igetbyte(struct intel8x0m *chip, u32 offset)
+{
+ return ioread8(chip->bmaddr + offset);
+}
+
+static inline u16 igetword(struct intel8x0m *chip, u32 offset)
+{
+ return ioread16(chip->bmaddr + offset);
+}
+
+static inline u32 igetdword(struct intel8x0m *chip, u32 offset)
+{
+ return ioread32(chip->bmaddr + offset);
+}
+
+static inline void iputbyte(struct intel8x0m *chip, u32 offset, u8 val)
+{
+ iowrite8(val, chip->bmaddr + offset);
+}
+
+static inline void iputword(struct intel8x0m *chip, u32 offset, u16 val)
+{
+ iowrite16(val, chip->bmaddr + offset);
+}
+
+static inline void iputdword(struct intel8x0m *chip, u32 offset, u32 val)
+{
+ iowrite32(val, chip->bmaddr + offset);
+}
+
+/*
+ * Lowlevel I/O - AC'97 registers
+ */
+
+static inline u16 iagetword(struct intel8x0m *chip, u32 offset)
+{
+ return ioread16(chip->addr + offset);
+}
+
+static inline void iaputword(struct intel8x0m *chip, u32 offset, u16 val)
+{
+ iowrite16(val, chip->addr + offset);
+}
+
+/*
+ * Basic I/O
+ */
+
+/*
+ * access to AC97 codec via normal i/o (for ICH and SIS7013)
+ */
+
+/* return the GLOB_STA bit for the corresponding codec */
+static unsigned int get_ich_codec_bit(struct intel8x0m *chip, unsigned int codec)
+{
+ static unsigned int codec_bit[3] = {
+ ICH_PCR, ICH_SCR, ICH_TCR
+ };
+ if (snd_BUG_ON(codec >= 3))
+ return ICH_PCR;
+ return codec_bit[codec];
+}
+
+static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int codec)
+{
+ int time;
+
+ if (codec > 1)
+ return -EIO;
+ codec = get_ich_codec_bit(chip, codec);
+
+ /* codec ready ? */
+ if ((igetdword(chip, ICHREG(GLOB_STA)) & codec) == 0)
+ return -EIO;
+
+ /* Anyone holding a semaphore for 1 msec should be shot... */
+ time = 100;
+ do {
+ if (!(igetbyte(chip, ICHREG(ACC_SEMA)) & ICH_CAS))
+ return 0;
+ udelay(10);
+ } while (time--);
+
+ /* access to some forbidden (non existant) ac97 registers will not
+ * reset the semaphore. So even if you don't get the semaphore, still
+ * continue the access. We don't need the semaphore anyway. */
+ snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n",
+ igetbyte(chip, ICHREG(ACC_SEMA)), igetdword(chip, ICHREG(GLOB_STA)));
+ iagetword(chip, 0); /* clear semaphore flag */
+ /* I don't care about the semaphore */
+ return -EBUSY;
+}
+
+static void snd_intel8x0_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct intel8x0m *chip = ac97->private_data;
+
+ if (snd_intel8x0m_codec_semaphore(chip, ac97->num) < 0) {
+ if (! chip->in_ac97_init)
+ snd_printk(KERN_ERR "codec_write %d: semaphore is not ready for register 0x%x\n", ac97->num, reg);
+ }
+ iaputword(chip, reg + ac97->num * 0x80, val);
+}
+
+static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct intel8x0m *chip = ac97->private_data;
+ unsigned short res;
+ unsigned int tmp;
+
+ if (snd_intel8x0m_codec_semaphore(chip, ac97->num) < 0) {
+ if (! chip->in_ac97_init)
+ snd_printk(KERN_ERR "codec_read %d: semaphore is not ready for register 0x%x\n", ac97->num, reg);
+ res = 0xffff;
+ } else {
+ res = iagetword(chip, reg + ac97->num * 0x80);
+ if ((tmp = igetdword(chip, ICHREG(GLOB_STA))) & ICH_RCS) {
+ /* reset RCS and preserve other R/WC bits */
+ iputdword(chip, ICHREG(GLOB_STA),
+ tmp & ~(ICH_SRI|ICH_PRI|ICH_TRI|ICH_GSCI));
+ if (! chip->in_ac97_init)
+ snd_printk(KERN_ERR "codec_read %d: read timeout for register 0x%x\n", ac97->num, reg);
+ res = 0xffff;
+ }
+ }
+ if (reg == AC97_GPIO_STATUS)
+ iagetword(chip, 0); /* clear semaphore */
+ return res;
+}
+
+
+/*
+ * DMA I/O
+ */
+static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev)
+{
+ int idx;
+ u32 *bdbar = ichdev->bdbar;
+ unsigned long port = ichdev->reg_offset;
+
+ iputdword(chip, port + ICH_REG_OFF_BDBAR, ichdev->bdbar_addr);
+ if (ichdev->size == ichdev->fragsize) {
+ ichdev->ack_reload = ichdev->ack = 2;
+ ichdev->fragsize1 = ichdev->fragsize >> 1;
+ for (idx = 0; idx < (ICH_REG_LVI_MASK + 1) * 2; idx += 4) {
+ bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf);
+ bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
+ ichdev->fragsize1 >> chip->pcm_pos_shift);
+ bdbar[idx + 2] = cpu_to_le32(ichdev->physbuf + (ichdev->size >> 1));
+ bdbar[idx + 3] = cpu_to_le32(0x80000000 | /* interrupt on completion */
+ ichdev->fragsize1 >> chip->pcm_pos_shift);
+ }
+ ichdev->frags = 2;
+ } else {
+ ichdev->ack_reload = ichdev->ack = 1;
+ ichdev->fragsize1 = ichdev->fragsize;
+ for (idx = 0; idx < (ICH_REG_LVI_MASK + 1) * 2; idx += 2) {
+ bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf + (((idx >> 1) * ichdev->fragsize) % ichdev->size));
+ bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
+ ichdev->fragsize >> chip->pcm_pos_shift);
+ // printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
+ }
+ ichdev->frags = ichdev->size / ichdev->fragsize;
+ }
+ iputbyte(chip, port + ICH_REG_OFF_LVI, ichdev->lvi = ICH_REG_LVI_MASK);
+ ichdev->civ = 0;
+ iputbyte(chip, port + ICH_REG_OFF_CIV, 0);
+ ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags;
+ ichdev->position = 0;
+#if 0
+ printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n",
+ ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1);
+#endif
+ /* clear interrupts */
+ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
+}
+
+/*
+ * Interrupt handler
+ */
+
+static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ichdev)
+{
+ unsigned long port = ichdev->reg_offset;
+ int civ, i, step;
+ int ack = 0;
+
+ civ = igetbyte(chip, port + ICH_REG_OFF_CIV);
+ if (civ == ichdev->civ) {
+ // snd_printd("civ same %d\n", civ);
+ step = 1;
+ ichdev->civ++;
+ ichdev->civ &= ICH_REG_LVI_MASK;
+ } else {
+ step = civ - ichdev->civ;
+ if (step < 0)
+ step += ICH_REG_LVI_MASK + 1;
+ // if (step != 1)
+ // snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ);
+ ichdev->civ = civ;
+ }
+
+ ichdev->position += step * ichdev->fragsize1;
+ ichdev->position %= ichdev->size;
+ ichdev->lvi += step;
+ ichdev->lvi &= ICH_REG_LVI_MASK;
+ iputbyte(chip, port + ICH_REG_OFF_LVI, ichdev->lvi);
+ for (i = 0; i < step; i++) {
+ ichdev->lvi_frag++;
+ ichdev->lvi_frag %= ichdev->frags;
+ ichdev->bdbar[ichdev->lvi * 2] = cpu_to_le32(ichdev->physbuf +
+ ichdev->lvi_frag *
+ ichdev->fragsize1);
+#if 0
+ printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n",
+ ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2],
+ ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port),
+ inl(port + 4), inb(port + ICH_REG_OFF_CR));
+#endif
+ if (--ichdev->ack == 0) {
+ ichdev->ack = ichdev->ack_reload;
+ ack = 1;
+ }
+ }
+ if (ack && ichdev->substream) {
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(ichdev->substream);
+ spin_lock(&chip->reg_lock);
+ }
+ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
+}
+
+static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id)
+{
+ struct intel8x0m *chip = dev_id;
+ struct ichdev *ichdev;
+ unsigned int status;
+ unsigned int i;
+
+ spin_lock(&chip->reg_lock);
+ status = igetdword(chip, chip->int_sta_reg);
+ if (status == 0xffffffff) { /* we are not yet resumed */
+ spin_unlock(&chip->reg_lock);
+ return IRQ_NONE;
+ }
+ if ((status & chip->int_sta_mask) == 0) {
+ if (status)
+ iputdword(chip, chip->int_sta_reg, status);
+ spin_unlock(&chip->reg_lock);
+ return IRQ_NONE;
+ }
+
+ for (i = 0; i < chip->bdbars_count; i++) {
+ ichdev = &chip->ichd[i];
+ if (status & ichdev->int_sta_mask)
+ snd_intel8x0_update(chip, ichdev);
+ }
+
+ /* ack them */
+ iputdword(chip, chip->int_sta_reg, status & chip->int_sta_mask);
+ spin_unlock(&chip->reg_lock);
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * PCM part
+ */
+
+static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct intel8x0m *chip = snd_pcm_substream_chip(substream);
+ struct ichdev *ichdev = get_ichdev(substream);
+ unsigned char val = 0;
+ unsigned long port = ichdev->reg_offset;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ val = ICH_IOCE | ICH_STARTBM;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ val = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ val = ICH_IOCE;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ val = ICH_IOCE | ICH_STARTBM;
+ break;
+ default:
+ return -EINVAL;
+ }
+ iputbyte(chip, port + ICH_REG_OFF_CR, val);
+ if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ /* wait until DMA stopped */
+ while (!(igetbyte(chip, port + ichdev->roff_sr) & ICH_DCH)) ;
+ /* reset whole DMA things */
+ iputbyte(chip, port + ICH_REG_OFF_CR, ICH_RESETREGS);
+ }
+ return 0;
+}
+
+static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct intel8x0m *chip = snd_pcm_substream_chip(substream);
+ struct ichdev *ichdev = get_ichdev(substream);
+ size_t ptr1, ptr;
+
+ ptr1 = igetword(chip, ichdev->reg_offset + ichdev->roff_picb) << chip->pcm_pos_shift;
+ if (ptr1 != 0)
+ ptr = ichdev->fragsize1 - ptr1;
+ else
+ ptr = 0;
+ ptr += ichdev->position;
+ if (ptr >= ichdev->size)
+ return 0;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static int snd_intel8x0m_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct intel8x0m *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ichdev *ichdev = get_ichdev(substream);
+
+ ichdev->physbuf = runtime->dma_addr;
+ ichdev->size = snd_pcm_lib_buffer_bytes(substream);
+ ichdev->fragsize = snd_pcm_lib_period_bytes(substream);
+ snd_ac97_write(ichdev->ac97, AC97_LINE1_RATE, runtime->rate);
+ snd_ac97_write(ichdev->ac97, AC97_LINE1_LEVEL, 0);
+ snd_intel8x0_setup_periods(chip, ichdev);
+ return 0;
+}
+
+static struct snd_pcm_hardware snd_intel8x0m_stream =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 64 * 1024,
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+
+static int snd_intel8x0m_pcm_open(struct snd_pcm_substream *substream, struct ichdev *ichdev)
+{
+ static unsigned int rates[] = { 8000, 9600, 12000, 16000 };
+ static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ ichdev->substream = substream;
+ runtime->hw = snd_intel8x0m_stream;
+ err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+ if ( err < 0 )
+ return err;
+ runtime->private_data = ichdev;
+ return 0;
+}
+
+static int snd_intel8x0m_playback_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0m *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0m_pcm_open(substream, &chip->ichd[ICHD_MDMOUT]);
+}
+
+static int snd_intel8x0m_playback_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0m *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ICHD_MDMOUT].substream = NULL;
+ return 0;
+}
+
+static int snd_intel8x0m_capture_open(struct snd_pcm_substream *substream)
+{
+ struct intel8x0m *chip = snd_pcm_substream_chip(substream);
+
+ return snd_intel8x0m_pcm_open(substream, &chip->ichd[ICHD_MDMIN]);
+}
+
+static int snd_intel8x0m_capture_close(struct snd_pcm_substream *substream)
+{
+ struct intel8x0m *chip = snd_pcm_substream_chip(substream);
+
+ chip->ichd[ICHD_MDMIN].substream = NULL;
+ return 0;
+}
+
+
+static struct snd_pcm_ops snd_intel8x0m_playback_ops = {
+ .open = snd_intel8x0m_playback_open,
+ .close = snd_intel8x0m_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0m_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_intel8x0m_capture_ops = {
+ .open = snd_intel8x0m_capture_open,
+ .close = snd_intel8x0m_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_intel8x0_hw_params,
+ .hw_free = snd_intel8x0_hw_free,
+ .prepare = snd_intel8x0m_pcm_prepare,
+ .trigger = snd_intel8x0_pcm_trigger,
+ .pointer = snd_intel8x0_pcm_pointer,
+};
+
+
+struct ich_pcm_table {
+ char *suffix;
+ struct snd_pcm_ops *playback_ops;
+ struct snd_pcm_ops *capture_ops;
+ size_t prealloc_size;
+ size_t prealloc_max_size;
+ int ac97_idx;
+};
+
+static int __devinit snd_intel8x0_pcm1(struct intel8x0m *chip, int device,
+ struct ich_pcm_table *rec)
+{
+ struct snd_pcm *pcm;
+ int err;
+ char name[32];
+
+ if (rec->suffix)
+ sprintf(name, "Intel ICH - %s", rec->suffix);
+ else
+ strcpy(name, "Intel ICH");
+ err = snd_pcm_new(chip->card, name, device,
+ rec->playback_ops ? 1 : 0,
+ rec->capture_ops ? 1 : 0, &pcm);
+ if (err < 0)
+ return err;
+
+ if (rec->playback_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, rec->playback_ops);
+ if (rec->capture_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, rec->capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
+ if (rec->suffix)
+ sprintf(pcm->name, "%s - %s", chip->card->shortname, rec->suffix);
+ else
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcm[device] = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ rec->prealloc_size,
+ rec->prealloc_max_size);
+
+ return 0;
+}
+
+static struct ich_pcm_table intel_pcms[] __devinitdata = {
+ {
+ .suffix = "Modem",
+ .playback_ops = &snd_intel8x0m_playback_ops,
+ .capture_ops = &snd_intel8x0m_capture_ops,
+ .prealloc_size = 32 * 1024,
+ .prealloc_max_size = 64 * 1024,
+ },
+};
+
+static int __devinit snd_intel8x0_pcm(struct intel8x0m *chip)
+{
+ int i, tblsize, device, err;
+ struct ich_pcm_table *tbl, *rec;
+
+#if 1
+ tbl = intel_pcms;
+ tblsize = 1;
+#else
+ switch (chip->device_type) {
+ case DEVICE_NFORCE:
+ tbl = nforce_pcms;
+ tblsize = ARRAY_SIZE(nforce_pcms);
+ break;
+ case DEVICE_ALI:
+ tbl = ali_pcms;
+ tblsize = ARRAY_SIZE(ali_pcms);
+ break;
+ default:
+ tbl = intel_pcms;
+ tblsize = 2;
+ break;
+ }
+#endif
+ device = 0;
+ for (i = 0; i < tblsize; i++) {
+ rec = tbl + i;
+ if (i > 0 && rec->ac97_idx) {
+ /* activate PCM only when associated AC'97 codec */
+ if (! chip->ichd[rec->ac97_idx].ac97)
+ continue;
+ }
+ err = snd_intel8x0_pcm1(chip, device, rec);
+ if (err < 0)
+ return err;
+ device++;
+ }
+
+ chip->pcm_devs = device;
+ return 0;
+}
+
+
+/*
+ * Mixer part
+ */
+
+static void snd_intel8x0_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct intel8x0m *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void snd_intel8x0_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct intel8x0m *chip = ac97->private_data;
+ chip->ac97 = NULL;
+}
+
+
+static int __devinit snd_intel8x0_mixer(struct intel8x0m *chip, int ac97_clock)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ struct snd_ac97 *x97;
+ int err;
+ unsigned int glob_sta = 0;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_intel8x0_codec_write,
+ .read = snd_intel8x0_codec_read,
+ };
+
+ chip->in_ac97_init = 1;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_intel8x0_mixer_free_ac97;
+ ac97.scaps = AC97_SCAP_SKIP_AUDIO | AC97_SCAP_POWER_SAVE;
+
+ glob_sta = igetdword(chip, ICHREG(GLOB_STA));
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &pbus)) < 0)
+ goto __err;
+ pbus->private_free = snd_intel8x0_mixer_free_ac97_bus;
+ if (ac97_clock >= 8000 && ac97_clock <= 48000)
+ pbus->clock = ac97_clock;
+ chip->ac97_bus = pbus;
+
+ ac97.pci = chip->pci;
+ ac97.num = glob_sta & ICH_SCR ? 1 : 0;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &x97)) < 0) {
+ snd_printk(KERN_ERR "Unable to initialize codec #%d\n", ac97.num);
+ if (ac97.num == 0)
+ goto __err;
+ return err;
+ }
+ chip->ac97 = x97;
+ if(ac97_is_modem(x97) && !chip->ichd[ICHD_MDMIN].ac97) {
+ chip->ichd[ICHD_MDMIN].ac97 = x97;
+ chip->ichd[ICHD_MDMOUT].ac97 = x97;
+ }
+
+ chip->in_ac97_init = 0;
+ return 0;
+
+ __err:
+ /* clear the cold-reset bit for the next chance */
+ if (chip->device_type != DEVICE_ALI)
+ iputdword(chip, ICHREG(GLOB_CNT),
+ igetdword(chip, ICHREG(GLOB_CNT)) & ~ICH_AC97COLD);
+ return err;
+}
+
+
+/*
+ *
+ */
+
+static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing)
+{
+ unsigned long end_time;
+ unsigned int cnt, status, nstatus;
+
+ /* put logic to right state */
+ /* first clear status bits */
+ status = ICH_RCS | ICH_MIINT | ICH_MOINT;
+ cnt = igetdword(chip, ICHREG(GLOB_STA));
+ iputdword(chip, ICHREG(GLOB_STA), cnt & status);
+
+ /* ACLink on, 2 channels */
+ cnt = igetdword(chip, ICHREG(GLOB_CNT));
+ cnt &= ~(ICH_ACLINK);
+ /* finish cold or do warm reset */
+ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM;
+ iputdword(chip, ICHREG(GLOB_CNT), cnt);
+ end_time = (jiffies + (HZ / 4)) + 1;
+ do {
+ if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0)
+ goto __ok;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n",
+ igetdword(chip, ICHREG(GLOB_CNT)));
+ return -EIO;
+
+ __ok:
+ if (probing) {
+ /* wait for any codec ready status.
+ * Once it becomes ready it should remain ready
+ * as long as we do not disable the ac97 link.
+ */
+ end_time = jiffies + HZ;
+ do {
+ status = igetdword(chip, ICHREG(GLOB_STA)) &
+ (ICH_PCR | ICH_SCR | ICH_TCR);
+ if (status)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ if (! status) {
+ /* no codec is found */
+ snd_printk(KERN_ERR "codec_ready: codec is not ready [0x%x]\n",
+ igetdword(chip, ICHREG(GLOB_STA)));
+ return -EIO;
+ }
+
+ /* up to two codecs (modem cannot be tertiary with ICH4) */
+ nstatus = ICH_PCR | ICH_SCR;
+
+ /* wait for other codecs ready status. */
+ end_time = jiffies + HZ / 4;
+ while (status != nstatus && time_after_eq(end_time, jiffies)) {
+ schedule_timeout_uninterruptible(1);
+ status |= igetdword(chip, ICHREG(GLOB_STA)) & nstatus;
+ }
+
+ } else {
+ /* resume phase */
+ status = 0;
+ if (chip->ac97)
+ status |= get_ich_codec_bit(chip, chip->ac97->num);
+ /* wait until all the probed codecs are ready */
+ end_time = jiffies + HZ;
+ do {
+ nstatus = igetdword(chip, ICHREG(GLOB_STA)) &
+ (ICH_PCR | ICH_SCR | ICH_TCR);
+ if (status == nstatus)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_after_eq(end_time, jiffies));
+ }
+
+ if (chip->device_type == DEVICE_SIS) {
+ /* unmute the output on SIS7012 */
+ iputword(chip, 0x4c, igetword(chip, 0x4c) | 1);
+ }
+
+ return 0;
+}
+
+static int snd_intel8x0_chip_init(struct intel8x0m *chip, int probing)
+{
+ unsigned int i;
+ int err;
+
+ if ((err = snd_intel8x0m_ich_chip_init(chip, probing)) < 0)
+ return err;
+ iagetword(chip, 0); /* clear semaphore flag */
+
+ /* disable interrupts */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, 0x00);
+ /* reset channels */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS);
+ /* initialize Buffer Descriptor Lists */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputdword(chip, ICH_REG_OFF_BDBAR + chip->ichd[i].reg_offset, chip->ichd[i].bdbar_addr);
+ return 0;
+}
+
+static int snd_intel8x0_free(struct intel8x0m *chip)
+{
+ unsigned int i;
+
+ if (chip->irq < 0)
+ goto __hw_end;
+ /* disable interrupts */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, 0x00);
+ /* reset channels */
+ for (i = 0; i < chip->bdbars_count; i++)
+ iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS);
+ __hw_end:
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (chip->bdbars.area)
+ snd_dma_free_pages(&chip->bdbars);
+ if (chip->addr)
+ pci_iounmap(chip->pci, chip->addr);
+ if (chip->bmaddr)
+ pci_iounmap(chip->pci, chip->bmaddr);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct intel8x0m *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < chip->pcm_devs; i++)
+ snd_pcm_suspend_all(chip->pcm[i]);
+ snd_ac97_suspend(chip->ac97);
+ if (chip->irq >= 0) {
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ }
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int intel8x0m_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct intel8x0m *chip = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "intel8x0m: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+ if (request_irq(pci->irq, snd_intel8x0_interrupt,
+ IRQF_SHARED, card->shortname, chip)) {
+ printk(KERN_ERR "intel8x0m: unable to grab IRQ %d, "
+ "disabling device\n", pci->irq);
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ chip->irq = pci->irq;
+ snd_intel8x0_chip_init(chip, 0);
+ snd_ac97_resume(chip->ac97);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+#ifdef CONFIG_PROC_FS
+static void snd_intel8x0m_proc_read(struct snd_info_entry * entry,
+ struct snd_info_buffer *buffer)
+{
+ struct intel8x0m *chip = entry->private_data;
+ unsigned int tmp;
+
+ snd_iprintf(buffer, "Intel8x0m\n\n");
+ if (chip->device_type == DEVICE_ALI)
+ return;
+ tmp = igetdword(chip, ICHREG(GLOB_STA));
+ snd_iprintf(buffer, "Global control : 0x%08x\n",
+ igetdword(chip, ICHREG(GLOB_CNT)));
+ snd_iprintf(buffer, "Global status : 0x%08x\n", tmp);
+ snd_iprintf(buffer, "AC'97 codecs ready :%s%s%s%s\n",
+ tmp & ICH_PCR ? " primary" : "",
+ tmp & ICH_SCR ? " secondary" : "",
+ tmp & ICH_TCR ? " tertiary" : "",
+ (tmp & (ICH_PCR | ICH_SCR | ICH_TCR)) == 0 ? " none" : "");
+}
+
+static void __devinit snd_intel8x0m_proc_init(struct intel8x0m * chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "intel8x0m", &entry))
+ snd_info_set_text_ops(entry, chip, snd_intel8x0m_proc_read);
+}
+#else /* !CONFIG_PROC_FS */
+#define snd_intel8x0m_proc_init(chip)
+#endif /* CONFIG_PROC_FS */
+
+
+static int snd_intel8x0_dev_free(struct snd_device *device)
+{
+ struct intel8x0m *chip = device->device_data;
+ return snd_intel8x0_free(chip);
+}
+
+struct ich_reg_info {
+ unsigned int int_sta_mask;
+ unsigned int offset;
+};
+
+static int __devinit snd_intel8x0m_create(struct snd_card *card,
+ struct pci_dev *pci,
+ unsigned long device_type,
+ struct intel8x0m ** r_intel8x0)
+{
+ struct intel8x0m *chip;
+ int err;
+ unsigned int i;
+ unsigned int int_sta_masks;
+ struct ichdev *ichdev;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_intel8x0_dev_free,
+ };
+ static struct ich_reg_info intel_regs[2] = {
+ { ICH_MIINT, 0 },
+ { ICH_MOINT, 0x10 },
+ };
+ struct ich_reg_info *tbl;
+
+ *r_intel8x0 = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&chip->reg_lock);
+ chip->device_type = device_type;
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ if ((err = pci_request_regions(pci, card->shortname)) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+
+ if (device_type == DEVICE_ALI) {
+ /* ALI5455 has no ac97 region */
+ chip->bmaddr = pci_iomap(pci, 0, 0);
+ goto port_inited;
+ }
+
+ if (pci_resource_flags(pci, 2) & IORESOURCE_MEM) /* ICH4 and Nforce */
+ chip->addr = pci_iomap(pci, 2, 0);
+ else
+ chip->addr = pci_iomap(pci, 0, 0);
+ if (!chip->addr) {
+ snd_printk(KERN_ERR "AC'97 space ioremap problem\n");
+ snd_intel8x0_free(chip);
+ return -EIO;
+ }
+ if (pci_resource_flags(pci, 3) & IORESOURCE_MEM) /* ICH4 */
+ chip->bmaddr = pci_iomap(pci, 3, 0);
+ else
+ chip->bmaddr = pci_iomap(pci, 1, 0);
+ if (!chip->bmaddr) {
+ snd_printk(KERN_ERR "Controller space ioremap problem\n");
+ snd_intel8x0_free(chip);
+ return -EIO;
+ }
+
+ port_inited:
+ if (request_irq(pci->irq, snd_intel8x0_interrupt, IRQF_SHARED,
+ card->shortname, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_intel8x0_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ pci_set_master(pci);
+ synchronize_irq(chip->irq);
+
+ /* initialize offsets */
+ chip->bdbars_count = 2;
+ tbl = intel_regs;
+
+ for (i = 0; i < chip->bdbars_count; i++) {
+ ichdev = &chip->ichd[i];
+ ichdev->ichd = i;
+ ichdev->reg_offset = tbl[i].offset;
+ ichdev->int_sta_mask = tbl[i].int_sta_mask;
+ if (device_type == DEVICE_SIS) {
+ /* SiS 7013 swaps the registers */
+ ichdev->roff_sr = ICH_REG_OFF_PICB;
+ ichdev->roff_picb = ICH_REG_OFF_SR;
+ } else {
+ ichdev->roff_sr = ICH_REG_OFF_SR;
+ ichdev->roff_picb = ICH_REG_OFF_PICB;
+ }
+ if (device_type == DEVICE_ALI)
+ ichdev->ali_slot = (ichdev->reg_offset - 0x40) / 0x10;
+ }
+ /* SIS7013 handles the pcm data in bytes, others are in words */
+ chip->pcm_pos_shift = (device_type == DEVICE_SIS) ? 0 : 1;
+
+ /* allocate buffer descriptor lists */
+ /* the start of each lists must be aligned to 8 bytes */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2,
+ &chip->bdbars) < 0) {
+ snd_intel8x0_free(chip);
+ return -ENOMEM;
+ }
+ /* tables must be aligned to 8 bytes here, but the kernel pages
+ are much bigger, so we don't care (on i386) */
+ int_sta_masks = 0;
+ for (i = 0; i < chip->bdbars_count; i++) {
+ ichdev = &chip->ichd[i];
+ ichdev->bdbar = ((u32 *)chip->bdbars.area) + (i * ICH_MAX_FRAGS * 2);
+ ichdev->bdbar_addr = chip->bdbars.addr + (i * sizeof(u32) * ICH_MAX_FRAGS * 2);
+ int_sta_masks |= ichdev->int_sta_mask;
+ }
+ chip->int_sta_reg = ICH_REG_GLOB_STA;
+ chip->int_sta_mask = int_sta_masks;
+
+ if ((err = snd_intel8x0_chip_init(chip, 1)) < 0) {
+ snd_intel8x0_free(chip);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_intel8x0_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_intel8x0 = chip;
+ return 0;
+}
+
+static struct shortname_table {
+ unsigned int id;
+ const char *s;
+} shortnames[] __devinitdata = {
+ { PCI_DEVICE_ID_INTEL_82801AA_6, "Intel 82801AA-ICH" },
+ { PCI_DEVICE_ID_INTEL_82801AB_6, "Intel 82901AB-ICH0" },
+ { PCI_DEVICE_ID_INTEL_82801BA_6, "Intel 82801BA-ICH2" },
+ { PCI_DEVICE_ID_INTEL_440MX_6, "Intel 440MX" },
+ { PCI_DEVICE_ID_INTEL_82801CA_6, "Intel 82801CA-ICH3" },
+ { PCI_DEVICE_ID_INTEL_82801DB_6, "Intel 82801DB-ICH4" },
+ { PCI_DEVICE_ID_INTEL_82801EB_6, "Intel ICH5" },
+ { PCI_DEVICE_ID_INTEL_ICH6_17, "Intel ICH6" },
+ { PCI_DEVICE_ID_INTEL_ICH7_19, "Intel ICH7" },
+ { 0x7446, "AMD AMD768" },
+ { PCI_DEVICE_ID_SI_7013, "SiS SI7013" },
+ { PCI_DEVICE_ID_NVIDIA_MCP1_MODEM, "NVidia nForce" },
+ { PCI_DEVICE_ID_NVIDIA_MCP2_MODEM, "NVidia nForce2" },
+ { PCI_DEVICE_ID_NVIDIA_MCP2S_MODEM, "NVidia nForce2s" },
+ { PCI_DEVICE_ID_NVIDIA_MCP3_MODEM, "NVidia nForce3" },
+#if 0
+ { 0x5455, "ALi M5455" },
+ { 0x746d, "AMD AMD8111" },
+#endif
+ { 0 },
+};
+
+static int __devinit snd_intel8x0m_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct intel8x0m *chip;
+ int err;
+ struct shortname_table *name;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "ICH-MODEM");
+ strcpy(card->shortname, "Intel ICH");
+ for (name = shortnames; name->id; name++) {
+ if (pci->device == name->id) {
+ strcpy(card->shortname, name->s);
+ break;
+ }
+ }
+ strcat(card->shortname," Modem");
+
+ if ((err = snd_intel8x0m_create(card, pci, pci_id->driver_data, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ if ((err = snd_intel8x0_mixer(chip, ac97_clock)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_intel8x0_pcm(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_intel8x0m_proc_init(chip);
+
+ sprintf(card->longname, "%s at irq %i",
+ card->shortname, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ return 0;
+}
+
+static void __devexit snd_intel8x0m_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Intel ICH Modem",
+ .id_table = snd_intel8x0m_ids,
+ .probe = snd_intel8x0m_probe,
+ .remove = __devexit_p(snd_intel8x0m_remove),
+#ifdef CONFIG_PM
+ .suspend = intel8x0m_suspend,
+ .resume = intel8x0m_resume,
+#endif
+};
+
+
+static int __init alsa_card_intel8x0m_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_intel8x0m_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_intel8x0m_init)
+module_exit(alsa_card_intel8x0m_exit)
diff --git a/sound/pci/korg1212/Makefile b/sound/pci/korg1212/Makefile
new file mode 100644
index 0000000..f11ce1b
--- /dev/null
+++ b/sound/pci/korg1212/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-korg1212-objs := korg1212.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_KORG1212) += snd-korg1212.o
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
new file mode 100644
index 0000000..5f8006b
--- /dev/null
+++ b/sound/pci/korg1212/korg1212.c
@@ -0,0 +1,2495 @@
+/*
+ * Driver for the Korg 1212 IO PCI card
+ *
+ * Copyright (c) 2001 Haroldo Gamal <gamal@alternex.com.br>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <linux/firmware.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+
+// ----------------------------------------------------------------------------
+// Debug Stuff
+// ----------------------------------------------------------------------------
+#define K1212_DEBUG_LEVEL 0
+#if K1212_DEBUG_LEVEL > 0
+#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args)
+#else
+#define K1212_DEBUG_PRINTK(fmt,...)
+#endif
+#if K1212_DEBUG_LEVEL > 1
+#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args)
+#else
+#define K1212_DEBUG_PRINTK_VERBOSE(fmt,...)
+#endif
+
+// ----------------------------------------------------------------------------
+// Record/Play Buffer Allocation Method. If K1212_LARGEALLOC is defined all
+// buffers are alocated as a large piece inside KorgSharedBuffer.
+// ----------------------------------------------------------------------------
+//#define K1212_LARGEALLOC 1
+
+// ----------------------------------------------------------------------------
+// Valid states of the Korg 1212 I/O card.
+// ----------------------------------------------------------------------------
+enum CardState {
+ K1212_STATE_NONEXISTENT, // there is no card here
+ K1212_STATE_UNINITIALIZED, // the card is awaiting DSP download
+ K1212_STATE_DSP_IN_PROCESS, // the card is currently downloading its DSP code
+ K1212_STATE_DSP_COMPLETE, // the card has finished the DSP download
+ K1212_STATE_READY, // the card can be opened by an application. Any application
+ // requests prior to this state should fail. Only an open
+ // request can be made at this state.
+ K1212_STATE_OPEN, // an application has opened the card
+ K1212_STATE_SETUP, // the card has been setup for play
+ K1212_STATE_PLAYING, // the card is playing
+ K1212_STATE_MONITOR, // the card is in the monitor mode
+ K1212_STATE_CALIBRATING, // the card is currently calibrating
+ K1212_STATE_ERRORSTOP, // the card has stopped itself because of an error and we
+ // are in the process of cleaning things up.
+ K1212_STATE_MAX_STATE // state values of this and beyond are invalid
+};
+
+// ----------------------------------------------------------------------------
+// The following enumeration defines the constants written to the card's
+// host-to-card doorbell to initiate a command.
+// ----------------------------------------------------------------------------
+enum korg1212_dbcnst {
+ K1212_DB_RequestForData = 0, // sent by the card to request a buffer fill.
+ K1212_DB_TriggerPlay = 1, // starts playback/record on the card.
+ K1212_DB_SelectPlayMode = 2, // select monitor, playback setup, or stop.
+ K1212_DB_ConfigureBufferMemory = 3, // tells card where the host audio buffers are.
+ K1212_DB_RequestAdatTimecode = 4, // asks the card for the latest ADAT timecode value.
+ K1212_DB_SetClockSourceRate = 5, // sets the clock source and rate for the card.
+ K1212_DB_ConfigureMiscMemory = 6, // tells card where other buffers are.
+ K1212_DB_TriggerFromAdat = 7, // tells card to trigger from Adat at a specific
+ // timecode value.
+ K1212_DB_DMAERROR = 0x80, // DMA Error - the PCI bus is congestioned.
+ K1212_DB_CARDSTOPPED = 0x81, // Card has stopped by user request.
+ K1212_DB_RebootCard = 0xA0, // instructs the card to reboot.
+ K1212_DB_BootFromDSPPage4 = 0xA4, // instructs the card to boot from the DSP microcode
+ // on page 4 (local page to card).
+ K1212_DB_DSPDownloadDone = 0xAE, // sent by the card to indicate the download has
+ // completed.
+ K1212_DB_StartDSPDownload = 0xAF // tells the card to download its DSP firmware.
+};
+
+
+// ----------------------------------------------------------------------------
+// The following enumeration defines return codes
+// to the Korg 1212 I/O driver.
+// ----------------------------------------------------------------------------
+enum snd_korg1212rc {
+ K1212_CMDRET_Success = 0, // command was successfully placed
+ K1212_CMDRET_DIOCFailure, // the DeviceIoControl call failed
+ K1212_CMDRET_PMFailure, // the protected mode call failed
+ K1212_CMDRET_FailUnspecified, // unspecified failure
+ K1212_CMDRET_FailBadState, // the specified command can not be given in
+ // the card's current state. (or the wave device's
+ // state)
+ K1212_CMDRET_CardUninitialized, // the card is uninitialized and cannot be used
+ K1212_CMDRET_BadIndex, // an out of range card index was specified
+ K1212_CMDRET_BadHandle, // an invalid card handle was specified
+ K1212_CMDRET_NoFillRoutine, // a play request has been made before a fill routine set
+ K1212_CMDRET_FillRoutineInUse, // can't set a new fill routine while one is in use
+ K1212_CMDRET_NoAckFromCard, // the card never acknowledged a command
+ K1212_CMDRET_BadParams, // bad parameters were provided by the caller
+
+ K1212_CMDRET_BadDevice, // the specified wave device was out of range
+ K1212_CMDRET_BadFormat // the specified wave format is unsupported
+};
+
+// ----------------------------------------------------------------------------
+// The following enumeration defines the constants used to select the play
+// mode for the card in the SelectPlayMode command.
+// ----------------------------------------------------------------------------
+enum PlayModeSelector {
+ K1212_MODE_SetupPlay = 0x00000001, // provides card with pre-play information
+ K1212_MODE_MonitorOn = 0x00000002, // tells card to turn on monitor mode
+ K1212_MODE_MonitorOff = 0x00000004, // tells card to turn off monitor mode
+ K1212_MODE_StopPlay = 0x00000008 // stops playback on the card
+};
+
+// ----------------------------------------------------------------------------
+// The following enumeration defines the constants used to select the monitor
+// mode for the card in the SetMonitorMode command.
+// ----------------------------------------------------------------------------
+enum MonitorModeSelector {
+ K1212_MONMODE_Off = 0, // tells card to turn off monitor mode
+ K1212_MONMODE_On // tells card to turn on monitor mode
+};
+
+#define MAILBOX0_OFFSET 0x40 // location of mailbox 0 relative to base address
+#define MAILBOX1_OFFSET 0x44 // location of mailbox 1 relative to base address
+#define MAILBOX2_OFFSET 0x48 // location of mailbox 2 relative to base address
+#define MAILBOX3_OFFSET 0x4c // location of mailbox 3 relative to base address
+#define OUT_DOORBELL_OFFSET 0x60 // location of PCI to local doorbell
+#define IN_DOORBELL_OFFSET 0x64 // location of local to PCI doorbell
+#define STATUS_REG_OFFSET 0x68 // location of interrupt control/status register
+#define PCI_CONTROL_OFFSET 0x6c // location of the EEPROM, PCI, User I/O, init control
+ // register
+#define SENS_CONTROL_OFFSET 0x6e // location of the input sensitivity setting register.
+ // this is the upper word of the PCI control reg.
+#define DEV_VEND_ID_OFFSET 0x70 // location of the device and vendor ID register
+
+#define MAX_COMMAND_RETRIES 5 // maximum number of times the driver will attempt
+ // to send a command before giving up.
+#define COMMAND_ACK_MASK 0x8000 // the MSB is set in the command acknowledgment from
+ // the card.
+#define DOORBELL_VAL_MASK 0x00FF // the doorbell value is one byte
+
+#define CARD_BOOT_DELAY_IN_MS 10
+#define CARD_BOOT_TIMEOUT 10
+#define DSP_BOOT_DELAY_IN_MS 200
+
+#define kNumBuffers 8
+#define k1212MaxCards 4
+#define k1212NumWaveDevices 6
+#define k16BitChannels 10
+#define k32BitChannels 2
+#define kAudioChannels (k16BitChannels + k32BitChannels)
+#define kPlayBufferFrames 1024
+
+#define K1212_ANALOG_CHANNELS 2
+#define K1212_SPDIF_CHANNELS 2
+#define K1212_ADAT_CHANNELS 8
+#define K1212_CHANNELS (K1212_ADAT_CHANNELS + K1212_ANALOG_CHANNELS)
+#define K1212_MIN_CHANNELS 1
+#define K1212_MAX_CHANNELS K1212_CHANNELS
+#define K1212_FRAME_SIZE (sizeof(struct KorgAudioFrame))
+#define K1212_MAX_SAMPLES (kPlayBufferFrames*kNumBuffers)
+#define K1212_PERIODS (kNumBuffers)
+#define K1212_PERIOD_BYTES (K1212_FRAME_SIZE*kPlayBufferFrames)
+#define K1212_BUF_SIZE (K1212_PERIOD_BYTES*kNumBuffers)
+#define K1212_ANALOG_BUF_SIZE (K1212_ANALOG_CHANNELS * 2 * kPlayBufferFrames * kNumBuffers)
+#define K1212_SPDIF_BUF_SIZE (K1212_SPDIF_CHANNELS * 3 * kPlayBufferFrames * kNumBuffers)
+#define K1212_ADAT_BUF_SIZE (K1212_ADAT_CHANNELS * 2 * kPlayBufferFrames * kNumBuffers)
+#define K1212_MAX_BUF_SIZE (K1212_ANALOG_BUF_SIZE + K1212_ADAT_BUF_SIZE)
+
+#define k1212MinADCSens 0x7f
+#define k1212MaxADCSens 0x00
+#define k1212MaxVolume 0x7fff
+#define k1212MaxWaveVolume 0xffff
+#define k1212MinVolume 0x0000
+#define k1212MaxVolInverted 0x8000
+
+// -----------------------------------------------------------------
+// the following bits are used for controlling interrupts in the
+// interrupt control/status reg
+// -----------------------------------------------------------------
+#define PCI_INT_ENABLE_BIT 0x00000100
+#define PCI_DOORBELL_INT_ENABLE_BIT 0x00000200
+#define LOCAL_INT_ENABLE_BIT 0x00010000
+#define LOCAL_DOORBELL_INT_ENABLE_BIT 0x00020000
+#define LOCAL_DMA1_INT_ENABLE_BIT 0x00080000
+
+// -----------------------------------------------------------------
+// the following bits are defined for the PCI command register
+// -----------------------------------------------------------------
+#define PCI_CMD_MEM_SPACE_ENABLE_BIT 0x0002
+#define PCI_CMD_IO_SPACE_ENABLE_BIT 0x0001
+#define PCI_CMD_BUS_MASTER_ENABLE_BIT 0x0004
+
+// -----------------------------------------------------------------
+// the following bits are defined for the PCI status register
+// -----------------------------------------------------------------
+#define PCI_STAT_PARITY_ERROR_BIT 0x8000
+#define PCI_STAT_SYSTEM_ERROR_BIT 0x4000
+#define PCI_STAT_MASTER_ABORT_RCVD_BIT 0x2000
+#define PCI_STAT_TARGET_ABORT_RCVD_BIT 0x1000
+#define PCI_STAT_TARGET_ABORT_SENT_BIT 0x0800
+
+// ------------------------------------------------------------------------
+// the following constants are used in setting the 1212 I/O card's input
+// sensitivity.
+// ------------------------------------------------------------------------
+#define SET_SENS_LOCALINIT_BITPOS 15
+#define SET_SENS_DATA_BITPOS 10
+#define SET_SENS_CLOCK_BITPOS 8
+#define SET_SENS_LOADSHIFT_BITPOS 0
+
+#define SET_SENS_LEFTCHANID 0x00
+#define SET_SENS_RIGHTCHANID 0x01
+
+#define K1212SENSUPDATE_DELAY_IN_MS 50
+
+// --------------------------------------------------------------------------
+// WaitRTCTicks
+//
+// This function waits the specified number of real time clock ticks.
+// According to the DDK, each tick is ~0.8 microseconds.
+// The defines following the function declaration can be used for the
+// numTicksToWait parameter.
+// --------------------------------------------------------------------------
+#define ONE_RTC_TICK 1
+#define SENSCLKPULSE_WIDTH 4
+#define LOADSHIFT_DELAY 4
+#define INTERCOMMAND_DELAY 40
+#define STOPCARD_DELAY 300 // max # RTC ticks for the card to stop once we write
+ // the command register. (could be up to 180 us)
+#define COMMAND_ACK_DELAY 13 // number of RTC ticks to wait for an acknowledgement
+ // from the card after sending a command.
+
+enum ClockSourceIndex {
+ K1212_CLKIDX_AdatAt44_1K = 0, // selects source as ADAT at 44.1 kHz
+ K1212_CLKIDX_AdatAt48K, // selects source as ADAT at 48 kHz
+ K1212_CLKIDX_WordAt44_1K, // selects source as S/PDIF at 44.1 kHz
+ K1212_CLKIDX_WordAt48K, // selects source as S/PDIF at 48 kHz
+ K1212_CLKIDX_LocalAt44_1K, // selects source as local clock at 44.1 kHz
+ K1212_CLKIDX_LocalAt48K, // selects source as local clock at 48 kHz
+ K1212_CLKIDX_Invalid // used to check validity of the index
+};
+
+enum ClockSourceType {
+ K1212_CLKIDX_Adat = 0, // selects source as ADAT
+ K1212_CLKIDX_Word, // selects source as S/PDIF
+ K1212_CLKIDX_Local // selects source as local clock
+};
+
+struct KorgAudioFrame {
+ u16 frameData16[k16BitChannels]; /* channels 0-9 use 16 bit samples */
+ u32 frameData32[k32BitChannels]; /* channels 10-11 use 32 bits - only 20 are sent across S/PDIF */
+ u32 timeCodeVal; /* holds the ADAT timecode value */
+};
+
+struct KorgAudioBuffer {
+ struct KorgAudioFrame bufferData[kPlayBufferFrames]; /* buffer definition */
+};
+
+struct KorgSharedBuffer {
+#ifdef K1212_LARGEALLOC
+ struct KorgAudioBuffer playDataBufs[kNumBuffers];
+ struct KorgAudioBuffer recordDataBufs[kNumBuffers];
+#endif
+ short volumeData[kAudioChannels];
+ u32 cardCommand;
+ u16 routeData [kAudioChannels];
+ u32 AdatTimeCode; // ADAT timecode value
+};
+
+struct SensBits {
+ union {
+ struct {
+ unsigned int leftChanVal:8;
+ unsigned int leftChanId:8;
+ } v;
+ u16 leftSensBits;
+ } l;
+ union {
+ struct {
+ unsigned int rightChanVal:8;
+ unsigned int rightChanId:8;
+ } v;
+ u16 rightSensBits;
+ } r;
+};
+
+struct snd_korg1212 {
+ struct snd_card *card;
+ struct pci_dev *pci;
+ struct snd_pcm *pcm;
+ int irq;
+
+ spinlock_t lock;
+ struct mutex open_mutex;
+
+ struct timer_list timer; /* timer callback for checking ack of stop request */
+ int stop_pending_cnt; /* counter for stop pending check */
+
+ wait_queue_head_t wait;
+
+ unsigned long iomem;
+ unsigned long ioport;
+ unsigned long iomem2;
+ unsigned long irqcount;
+ unsigned long inIRQ;
+ void __iomem *iobase;
+
+ struct snd_dma_buffer dma_dsp;
+ struct snd_dma_buffer dma_play;
+ struct snd_dma_buffer dma_rec;
+ struct snd_dma_buffer dma_shared;
+
+ u32 DataBufsSize;
+
+ struct KorgAudioBuffer * playDataBufsPtr;
+ struct KorgAudioBuffer * recordDataBufsPtr;
+
+ struct KorgSharedBuffer * sharedBufferPtr;
+
+ u32 RecDataPhy;
+ u32 PlayDataPhy;
+ unsigned long sharedBufferPhy;
+ u32 VolumeTablePhy;
+ u32 RoutingTablePhy;
+ u32 AdatTimeCodePhy;
+
+ u32 __iomem * statusRegPtr; // address of the interrupt status/control register
+ u32 __iomem * outDoorbellPtr; // address of the host->card doorbell register
+ u32 __iomem * inDoorbellPtr; // address of the card->host doorbell register
+ u32 __iomem * mailbox0Ptr; // address of mailbox 0 on the card
+ u32 __iomem * mailbox1Ptr; // address of mailbox 1 on the card
+ u32 __iomem * mailbox2Ptr; // address of mailbox 2 on the card
+ u32 __iomem * mailbox3Ptr; // address of mailbox 3 on the card
+ u32 __iomem * controlRegPtr; // address of the EEPROM, PCI, I/O, Init ctrl reg
+ u16 __iomem * sensRegPtr; // address of the sensitivity setting register
+ u32 __iomem * idRegPtr; // address of the device and vendor ID registers
+
+ size_t periodsize;
+ int channels;
+ int currentBuffer;
+
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ pid_t capture_pid;
+ pid_t playback_pid;
+
+ enum CardState cardState;
+ int running;
+ int idleMonitorOn; // indicates whether the card is in idle monitor mode.
+ u32 cmdRetryCount; // tracks how many times we have retried sending to the card.
+
+ enum ClockSourceIndex clkSrcRate; // sample rate and clock source
+
+ enum ClockSourceType clkSource; // clock source
+ int clkRate; // clock rate
+
+ int volumePhase[kAudioChannels];
+
+ u16 leftADCInSens; // ADC left channel input sensitivity
+ u16 rightADCInSens; // ADC right channel input sensitivity
+
+ int opencnt; // Open/Close count
+ int setcnt; // SetupForPlay count
+ int playcnt; // TriggerPlay count
+ int errorcnt; // Error Count
+ unsigned long totalerrorcnt; // Total Error Count
+
+ int dsp_is_loaded;
+ int dsp_stop_is_processed;
+
+};
+
+MODULE_DESCRIPTION("korg1212");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{KORG,korg1212}}");
+MODULE_FIRMWARE("korg/k1212.dsp");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Korg 1212 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Korg 1212 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard.");
+MODULE_AUTHOR("Haroldo Gamal <gamal@alternex.com.br>");
+
+static struct pci_device_id snd_korg1212_ids[] = {
+ {
+ .vendor = 0x10b5,
+ .device = 0x906d,
+ .subvendor = PCI_ANY_ID,
+ .subdevice = PCI_ANY_ID,
+ },
+ { 0, },
+};
+
+MODULE_DEVICE_TABLE(pci, snd_korg1212_ids);
+
+static char *stateName[] = {
+ "Non-existent",
+ "Uninitialized",
+ "DSP download in process",
+ "DSP download complete",
+ "Ready",
+ "Open",
+ "Setup for play",
+ "Playing",
+ "Monitor mode on",
+ "Calibrating",
+ "Invalid"
+};
+
+static char *clockSourceTypeName[] = { "ADAT", "S/PDIF", "local" };
+
+static char *clockSourceName[] = {
+ "ADAT at 44.1 kHz",
+ "ADAT at 48 kHz",
+ "S/PDIF at 44.1 kHz",
+ "S/PDIF at 48 kHz",
+ "local clock at 44.1 kHz",
+ "local clock at 48 kHz"
+};
+
+static char *channelName[] = {
+ "ADAT-1",
+ "ADAT-2",
+ "ADAT-3",
+ "ADAT-4",
+ "ADAT-5",
+ "ADAT-6",
+ "ADAT-7",
+ "ADAT-8",
+ "Analog-L",
+ "Analog-R",
+ "SPDIF-L",
+ "SPDIF-R",
+};
+
+static u16 ClockSourceSelector[] = {
+ 0x8000, // selects source as ADAT at 44.1 kHz
+ 0x0000, // selects source as ADAT at 48 kHz
+ 0x8001, // selects source as S/PDIF at 44.1 kHz
+ 0x0001, // selects source as S/PDIF at 48 kHz
+ 0x8002, // selects source as local clock at 44.1 kHz
+ 0x0002 // selects source as local clock at 48 kHz
+};
+
+union swap_u32 { unsigned char c[4]; u32 i; };
+
+#ifdef SNDRV_BIG_ENDIAN
+static u32 LowerWordSwap(u32 swappee)
+#else
+static u32 UpperWordSwap(u32 swappee)
+#endif
+{
+ union swap_u32 retVal, swapper;
+
+ swapper.i = swappee;
+ retVal.c[2] = swapper.c[3];
+ retVal.c[3] = swapper.c[2];
+ retVal.c[1] = swapper.c[1];
+ retVal.c[0] = swapper.c[0];
+
+ return retVal.i;
+}
+
+#ifdef SNDRV_BIG_ENDIAN
+static u32 UpperWordSwap(u32 swappee)
+#else
+static u32 LowerWordSwap(u32 swappee)
+#endif
+{
+ union swap_u32 retVal, swapper;
+
+ swapper.i = swappee;
+ retVal.c[2] = swapper.c[2];
+ retVal.c[3] = swapper.c[3];
+ retVal.c[1] = swapper.c[0];
+ retVal.c[0] = swapper.c[1];
+
+ return retVal.i;
+}
+
+#define SetBitInWord(theWord,bitPosition) (*theWord) |= (0x0001 << bitPosition)
+#define SetBitInDWord(theWord,bitPosition) (*theWord) |= (0x00000001 << bitPosition)
+#define ClearBitInWord(theWord,bitPosition) (*theWord) &= ~(0x0001 << bitPosition)
+#define ClearBitInDWord(theWord,bitPosition) (*theWord) &= ~(0x00000001 << bitPosition)
+
+static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212,
+ enum korg1212_dbcnst doorbellVal,
+ u32 mailBox0Val, u32 mailBox1Val,
+ u32 mailBox2Val, u32 mailBox3Val)
+{
+ u32 retryCount;
+ u16 mailBox3Lo;
+ int rc = K1212_CMDRET_Success;
+
+ if (!korg1212->outDoorbellPtr) {
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: CardUninitialized\n");
+ return K1212_CMDRET_CardUninitialized;
+ }
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Card <- 0x%08x 0x%08x [%s]\n",
+ doorbellVal, mailBox0Val, stateName[korg1212->cardState]);
+ for (retryCount = 0; retryCount < MAX_COMMAND_RETRIES; retryCount++) {
+ writel(mailBox3Val, korg1212->mailbox3Ptr);
+ writel(mailBox2Val, korg1212->mailbox2Ptr);
+ writel(mailBox1Val, korg1212->mailbox1Ptr);
+ writel(mailBox0Val, korg1212->mailbox0Ptr);
+ writel(doorbellVal, korg1212->outDoorbellPtr); // interrupt the card
+
+ // --------------------------------------------------------------
+ // the reboot command will not give an acknowledgement.
+ // --------------------------------------------------------------
+ if ( doorbellVal == K1212_DB_RebootCard ||
+ doorbellVal == K1212_DB_BootFromDSPPage4 ||
+ doorbellVal == K1212_DB_StartDSPDownload ) {
+ rc = K1212_CMDRET_Success;
+ break;
+ }
+
+ // --------------------------------------------------------------
+ // See if the card acknowledged the command. Wait a bit, then
+ // read in the low word of mailbox3. If the MSB is set and the
+ // low byte is equal to the doorbell value, then it ack'd.
+ // --------------------------------------------------------------
+ udelay(COMMAND_ACK_DELAY);
+ mailBox3Lo = readl(korg1212->mailbox3Ptr);
+ if (mailBox3Lo & COMMAND_ACK_MASK) {
+ if ((mailBox3Lo & DOORBELL_VAL_MASK) == (doorbellVal & DOORBELL_VAL_MASK)) {
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- Success\n");
+ rc = K1212_CMDRET_Success;
+ break;
+ }
+ }
+ }
+ korg1212->cmdRetryCount += retryCount;
+
+ if (retryCount >= MAX_COMMAND_RETRIES) {
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- NoAckFromCard\n");
+ rc = K1212_CMDRET_NoAckFromCard;
+ }
+
+ return rc;
+}
+
+/* spinlock already held */
+static void snd_korg1212_SendStop(struct snd_korg1212 *korg1212)
+{
+ if (! korg1212->stop_pending_cnt) {
+ korg1212->sharedBufferPtr->cardCommand = 0xffffffff;
+ /* program the timer */
+ korg1212->stop_pending_cnt = HZ;
+ korg1212->timer.expires = jiffies + 1;
+ add_timer(&korg1212->timer);
+ }
+}
+
+static void snd_korg1212_SendStopAndWait(struct snd_korg1212 *korg1212)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&korg1212->lock, flags);
+ korg1212->dsp_stop_is_processed = 0;
+ snd_korg1212_SendStop(korg1212);
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+ wait_event_timeout(korg1212->wait, korg1212->dsp_stop_is_processed, (HZ * 3) / 2);
+}
+
+/* timer callback for checking the ack of stop request */
+static void snd_korg1212_timer_func(unsigned long data)
+{
+ struct snd_korg1212 *korg1212 = (struct snd_korg1212 *) data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&korg1212->lock, flags);
+ if (korg1212->sharedBufferPtr->cardCommand == 0) {
+ /* ack'ed */
+ korg1212->stop_pending_cnt = 0;
+ korg1212->dsp_stop_is_processed = 1;
+ wake_up(&korg1212->wait);
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Stop ack'ed [%s]\n",
+ stateName[korg1212->cardState]);
+ } else {
+ if (--korg1212->stop_pending_cnt > 0) {
+ /* reprogram timer */
+ korg1212->timer.expires = jiffies + 1;
+ add_timer(&korg1212->timer);
+ } else {
+ snd_printd("korg1212_timer_func timeout\n");
+ korg1212->sharedBufferPtr->cardCommand = 0;
+ korg1212->dsp_stop_is_processed = 1;
+ wake_up(&korg1212->wait);
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Stop timeout [%s]\n",
+ stateName[korg1212->cardState]);
+ }
+ }
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+}
+
+static int snd_korg1212_TurnOnIdleMonitor(struct snd_korg1212 *korg1212)
+{
+ unsigned long flags;
+ int rc;
+
+ udelay(INTERCOMMAND_DELAY);
+ spin_lock_irqsave(&korg1212->lock, flags);
+ korg1212->idleMonitorOn = 1;
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode,
+ K1212_MODE_MonitorOn, 0, 0, 0);
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+ return rc;
+}
+
+static void snd_korg1212_TurnOffIdleMonitor(struct snd_korg1212 *korg1212)
+{
+ if (korg1212->idleMonitorOn) {
+ snd_korg1212_SendStopAndWait(korg1212);
+ korg1212->idleMonitorOn = 0;
+ }
+}
+
+static inline void snd_korg1212_setCardState(struct snd_korg1212 * korg1212, enum CardState csState)
+{
+ korg1212->cardState = csState;
+}
+
+static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212)
+{
+ K1212_DEBUG_PRINTK("K1212_DEBUG: OpenCard [%s] %d\n",
+ stateName[korg1212->cardState], korg1212->opencnt);
+ mutex_lock(&korg1212->open_mutex);
+ if (korg1212->opencnt++ == 0) {
+ snd_korg1212_TurnOffIdleMonitor(korg1212);
+ snd_korg1212_setCardState(korg1212, K1212_STATE_OPEN);
+ }
+
+ mutex_unlock(&korg1212->open_mutex);
+ return 1;
+}
+
+static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212)
+{
+ K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard [%s] %d\n",
+ stateName[korg1212->cardState], korg1212->opencnt);
+
+ mutex_lock(&korg1212->open_mutex);
+ if (--(korg1212->opencnt)) {
+ mutex_unlock(&korg1212->open_mutex);
+ return 0;
+ }
+
+ if (korg1212->cardState == K1212_STATE_SETUP) {
+ int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode,
+ K1212_MODE_StopPlay, 0, 0, 0);
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+ if (rc != K1212_CMDRET_Success) {
+ mutex_unlock(&korg1212->open_mutex);
+ return 0;
+ }
+ } else if (korg1212->cardState > K1212_STATE_SETUP) {
+ snd_korg1212_SendStopAndWait(korg1212);
+ }
+
+ if (korg1212->cardState > K1212_STATE_READY) {
+ snd_korg1212_TurnOnIdleMonitor(korg1212);
+ snd_korg1212_setCardState(korg1212, K1212_STATE_READY);
+ }
+
+ mutex_unlock(&korg1212->open_mutex);
+ return 0;
+}
+
+/* spinlock already held */
+static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212)
+{
+ int rc;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay [%s] %d\n",
+ stateName[korg1212->cardState], korg1212->setcnt);
+
+ if (korg1212->setcnt++)
+ return 0;
+
+ snd_korg1212_setCardState(korg1212, K1212_STATE_SETUP);
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode,
+ K1212_MODE_SetupPlay, 0, 0, 0);
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+ if (rc != K1212_CMDRET_Success) {
+ return 1;
+ }
+ return 0;
+}
+
+/* spinlock already held */
+static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212)
+{
+ int rc;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay [%s] %d\n",
+ stateName[korg1212->cardState], korg1212->playcnt);
+
+ if (korg1212->playcnt++)
+ return 0;
+
+ snd_korg1212_setCardState(korg1212, K1212_STATE_PLAYING);
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_TriggerPlay, 0, 0, 0, 0);
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+ if (rc != K1212_CMDRET_Success) {
+ return 1;
+ }
+ return 0;
+}
+
+/* spinlock already held */
+static int snd_korg1212_StopPlay(struct snd_korg1212 * korg1212)
+{
+ K1212_DEBUG_PRINTK("K1212_DEBUG: StopPlay [%s] %d\n",
+ stateName[korg1212->cardState], korg1212->playcnt);
+
+ if (--(korg1212->playcnt))
+ return 0;
+
+ korg1212->setcnt = 0;
+
+ if (korg1212->cardState != K1212_STATE_ERRORSTOP)
+ snd_korg1212_SendStop(korg1212);
+
+ snd_korg1212_setCardState(korg1212, K1212_STATE_OPEN);
+ return 0;
+}
+
+static void snd_korg1212_EnableCardInterrupts(struct snd_korg1212 * korg1212)
+{
+ writel(PCI_INT_ENABLE_BIT |
+ PCI_DOORBELL_INT_ENABLE_BIT |
+ LOCAL_INT_ENABLE_BIT |
+ LOCAL_DOORBELL_INT_ENABLE_BIT |
+ LOCAL_DMA1_INT_ENABLE_BIT,
+ korg1212->statusRegPtr);
+}
+
+#if 0 /* not used */
+
+static int snd_korg1212_SetMonitorMode(struct snd_korg1212 *korg1212,
+ enum MonitorModeSelector mode)
+{
+ K1212_DEBUG_PRINTK("K1212_DEBUG: SetMonitorMode [%s]\n",
+ stateName[korg1212->cardState]);
+
+ switch (mode) {
+ case K1212_MONMODE_Off:
+ if (korg1212->cardState != K1212_STATE_MONITOR)
+ return 0;
+ else {
+ snd_korg1212_SendStopAndWait(korg1212);
+ snd_korg1212_setCardState(korg1212, K1212_STATE_OPEN);
+ }
+ break;
+
+ case K1212_MONMODE_On:
+ if (korg1212->cardState != K1212_STATE_OPEN)
+ return 0;
+ else {
+ int rc;
+ snd_korg1212_setCardState(korg1212, K1212_STATE_MONITOR);
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode,
+ K1212_MODE_MonitorOn, 0, 0, 0);
+ if (rc != K1212_CMDRET_Success)
+ return 0;
+ }
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 1;
+}
+
+#endif /* not used */
+
+static inline int snd_korg1212_use_is_exclusive(struct snd_korg1212 *korg1212)
+{
+ if (korg1212->playback_pid != korg1212->capture_pid &&
+ korg1212->playback_pid >= 0 && korg1212->capture_pid >= 0)
+ return 0;
+
+ return 1;
+}
+
+static int snd_korg1212_SetRate(struct snd_korg1212 *korg1212, int rate)
+{
+ static enum ClockSourceIndex s44[] = {
+ K1212_CLKIDX_AdatAt44_1K,
+ K1212_CLKIDX_WordAt44_1K,
+ K1212_CLKIDX_LocalAt44_1K
+ };
+ static enum ClockSourceIndex s48[] = {
+ K1212_CLKIDX_AdatAt48K,
+ K1212_CLKIDX_WordAt48K,
+ K1212_CLKIDX_LocalAt48K
+ };
+ int parm, rc;
+
+ if (!snd_korg1212_use_is_exclusive (korg1212))
+ return -EBUSY;
+
+ switch (rate) {
+ case 44100:
+ parm = s44[korg1212->clkSource];
+ break;
+
+ case 48000:
+ parm = s48[korg1212->clkSource];
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ korg1212->clkSrcRate = parm;
+ korg1212->clkRate = rate;
+
+ udelay(INTERCOMMAND_DELAY);
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SetClockSourceRate,
+ ClockSourceSelector[korg1212->clkSrcRate],
+ 0, 0, 0);
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+
+ return 0;
+}
+
+static int snd_korg1212_SetClockSource(struct snd_korg1212 *korg1212, int source)
+{
+
+ if (source < 0 || source > 2)
+ return -EINVAL;
+
+ korg1212->clkSource = source;
+
+ snd_korg1212_SetRate(korg1212, korg1212->clkRate);
+
+ return 0;
+}
+
+static void snd_korg1212_DisableCardInterrupts(struct snd_korg1212 *korg1212)
+{
+ writel(0, korg1212->statusRegPtr);
+}
+
+static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212)
+{
+ struct SensBits sensVals;
+ int bitPosition;
+ int channel;
+ int clkIs48K;
+ int monModeSet;
+ u16 controlValue; // this keeps the current value to be written to
+ // the card's eeprom control register.
+ u16 count;
+ unsigned long flags;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity [%s]\n",
+ stateName[korg1212->cardState]);
+
+ // ----------------------------------------------------------------------------
+ // initialize things. The local init bit is always set when writing to the
+ // card's control register.
+ // ----------------------------------------------------------------------------
+ controlValue = 0;
+ SetBitInWord(&controlValue, SET_SENS_LOCALINIT_BITPOS); // init the control value
+
+ // ----------------------------------------------------------------------------
+ // make sure the card is not in monitor mode when we do this update.
+ // ----------------------------------------------------------------------------
+ if (korg1212->cardState == K1212_STATE_MONITOR || korg1212->idleMonitorOn) {
+ monModeSet = 1;
+ snd_korg1212_SendStopAndWait(korg1212);
+ } else
+ monModeSet = 0;
+
+ spin_lock_irqsave(&korg1212->lock, flags);
+
+ // ----------------------------------------------------------------------------
+ // we are about to send new values to the card, so clear the new values queued
+ // flag. Also, clear out mailbox 3, so we don't lockup.
+ // ----------------------------------------------------------------------------
+ writel(0, korg1212->mailbox3Ptr);
+ udelay(LOADSHIFT_DELAY);
+
+ // ----------------------------------------------------------------------------
+ // determine whether we are running a 48K or 44.1K clock. This info is used
+ // later when setting the SPDIF FF after the volume has been shifted in.
+ // ----------------------------------------------------------------------------
+ switch (korg1212->clkSrcRate) {
+ case K1212_CLKIDX_AdatAt44_1K:
+ case K1212_CLKIDX_WordAt44_1K:
+ case K1212_CLKIDX_LocalAt44_1K:
+ clkIs48K = 0;
+ break;
+
+ case K1212_CLKIDX_WordAt48K:
+ case K1212_CLKIDX_AdatAt48K:
+ case K1212_CLKIDX_LocalAt48K:
+ default:
+ clkIs48K = 1;
+ break;
+ }
+
+ // ----------------------------------------------------------------------------
+ // start the update. Setup the bit structure and then shift the bits.
+ // ----------------------------------------------------------------------------
+ sensVals.l.v.leftChanId = SET_SENS_LEFTCHANID;
+ sensVals.r.v.rightChanId = SET_SENS_RIGHTCHANID;
+ sensVals.l.v.leftChanVal = korg1212->leftADCInSens;
+ sensVals.r.v.rightChanVal = korg1212->rightADCInSens;
+
+ // ----------------------------------------------------------------------------
+ // now start shifting the bits in. Start with the left channel then the right.
+ // ----------------------------------------------------------------------------
+ for (channel = 0; channel < 2; channel++) {
+
+ // ----------------------------------------------------------------------------
+ // Bring the load/shift line low, then wait - the spec says >150ns from load/
+ // shift low to the first rising edge of the clock.
+ // ----------------------------------------------------------------------------
+ ClearBitInWord(&controlValue, SET_SENS_LOADSHIFT_BITPOS);
+ ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS);
+ writew(controlValue, korg1212->sensRegPtr); // load/shift goes low
+ udelay(LOADSHIFT_DELAY);
+
+ for (bitPosition = 15; bitPosition >= 0; bitPosition--) { // for all the bits
+ if (channel == 0) {
+ if (sensVals.l.leftSensBits & (0x0001 << bitPosition))
+ SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set high
+ else
+ ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set low
+ } else {
+ if (sensVals.r.rightSensBits & (0x0001 << bitPosition))
+ SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set high
+ else
+ ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS); // data bit set low
+ }
+
+ ClearBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS);
+ writew(controlValue, korg1212->sensRegPtr); // clock goes low
+ udelay(SENSCLKPULSE_WIDTH);
+ SetBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS);
+ writew(controlValue, korg1212->sensRegPtr); // clock goes high
+ udelay(SENSCLKPULSE_WIDTH);
+ }
+
+ // ----------------------------------------------------------------------------
+ // finish up SPDIF for left. Bring the load/shift line high, then write a one
+ // bit if the clock rate is 48K otherwise write 0.
+ // ----------------------------------------------------------------------------
+ ClearBitInWord(&controlValue, SET_SENS_DATA_BITPOS);
+ ClearBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS);
+ SetBitInWord(&controlValue, SET_SENS_LOADSHIFT_BITPOS);
+ writew(controlValue, korg1212->sensRegPtr); // load shift goes high - clk low
+ udelay(SENSCLKPULSE_WIDTH);
+
+ if (clkIs48K)
+ SetBitInWord(&controlValue, SET_SENS_DATA_BITPOS);
+
+ writew(controlValue, korg1212->sensRegPtr); // set/clear data bit
+ udelay(ONE_RTC_TICK);
+ SetBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS);
+ writew(controlValue, korg1212->sensRegPtr); // clock goes high
+ udelay(SENSCLKPULSE_WIDTH);
+ ClearBitInWord(&controlValue, SET_SENS_CLOCK_BITPOS);
+ writew(controlValue, korg1212->sensRegPtr); // clock goes low
+ udelay(SENSCLKPULSE_WIDTH);
+ }
+
+ // ----------------------------------------------------------------------------
+ // The update is complete. Set a timeout. This is the inter-update delay.
+ // Also, if the card was in monitor mode, restore it.
+ // ----------------------------------------------------------------------------
+ for (count = 0; count < 10; count++)
+ udelay(SENSCLKPULSE_WIDTH);
+
+ if (monModeSet) {
+ int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode,
+ K1212_MODE_MonitorOn, 0, 0, 0);
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+ }
+
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+
+ return 1;
+}
+
+static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212)
+{
+ int channel, rc;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is complete. [%s]\n",
+ stateName[korg1212->cardState]);
+
+ // ----------------------------------------------------
+ // tell the card to boot
+ // ----------------------------------------------------
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_BootFromDSPPage4, 0, 0, 0, 0);
+
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Boot from Page 4 - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+ msleep(DSP_BOOT_DELAY_IN_MS);
+
+ // --------------------------------------------------------------------------------
+ // Let the card know where all the buffers are.
+ // --------------------------------------------------------------------------------
+ rc = snd_korg1212_Send1212Command(korg1212,
+ K1212_DB_ConfigureBufferMemory,
+ LowerWordSwap(korg1212->PlayDataPhy),
+ LowerWordSwap(korg1212->RecDataPhy),
+ ((kNumBuffers * kPlayBufferFrames) / 2), // size given to the card
+ // is based on 2 buffers
+ 0
+ );
+
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Buffer Memory - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+
+ udelay(INTERCOMMAND_DELAY);
+
+ rc = snd_korg1212_Send1212Command(korg1212,
+ K1212_DB_ConfigureMiscMemory,
+ LowerWordSwap(korg1212->VolumeTablePhy),
+ LowerWordSwap(korg1212->RoutingTablePhy),
+ LowerWordSwap(korg1212->AdatTimeCodePhy),
+ 0
+ );
+
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Misc Memory - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+
+ // --------------------------------------------------------------------------------
+ // Initialize the routing and volume tables, then update the card's state.
+ // --------------------------------------------------------------------------------
+ udelay(INTERCOMMAND_DELAY);
+
+ for (channel = 0; channel < kAudioChannels; channel++) {
+ korg1212->sharedBufferPtr->volumeData[channel] = k1212MaxVolume;
+ //korg1212->sharedBufferPtr->routeData[channel] = channel;
+ korg1212->sharedBufferPtr->routeData[channel] = 8 + (channel & 1);
+ }
+
+ snd_korg1212_WriteADCSensitivity(korg1212);
+
+ udelay(INTERCOMMAND_DELAY);
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SetClockSourceRate,
+ ClockSourceSelector[korg1212->clkSrcRate],
+ 0, 0, 0);
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+
+ rc = snd_korg1212_TurnOnIdleMonitor(korg1212);
+ snd_korg1212_setCardState(korg1212, K1212_STATE_READY);
+
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Set Monitor On - RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+
+ snd_korg1212_setCardState(korg1212, K1212_STATE_DSP_COMPLETE);
+}
+
+static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id)
+{
+ u32 doorbellValue;
+ struct snd_korg1212 *korg1212 = dev_id;
+
+ doorbellValue = readl(korg1212->inDoorbellPtr);
+
+ if (!doorbellValue)
+ return IRQ_NONE;
+
+ spin_lock(&korg1212->lock);
+
+ writel(doorbellValue, korg1212->inDoorbellPtr);
+
+ korg1212->irqcount++;
+
+ korg1212->inIRQ++;
+
+ switch (doorbellValue) {
+ case K1212_DB_DSPDownloadDone:
+ K1212_DEBUG_PRINTK("K1212_DEBUG: IRQ DNLD count - %ld, %x, [%s].\n",
+ korg1212->irqcount, doorbellValue,
+ stateName[korg1212->cardState]);
+ if (korg1212->cardState == K1212_STATE_DSP_IN_PROCESS) {
+ korg1212->dsp_is_loaded = 1;
+ wake_up(&korg1212->wait);
+ }
+ break;
+
+ // ------------------------------------------------------------------------
+ // an error occurred - stop the card
+ // ------------------------------------------------------------------------
+ case K1212_DB_DMAERROR:
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DMAE count - %ld, %x, [%s].\n",
+ korg1212->irqcount, doorbellValue,
+ stateName[korg1212->cardState]);
+ snd_printk(KERN_ERR "korg1212: DMA Error\n");
+ korg1212->errorcnt++;
+ korg1212->totalerrorcnt++;
+ korg1212->sharedBufferPtr->cardCommand = 0;
+ snd_korg1212_setCardState(korg1212, K1212_STATE_ERRORSTOP);
+ break;
+
+ // ------------------------------------------------------------------------
+ // the card has stopped by our request. Clear the command word and signal
+ // the semaphore in case someone is waiting for this.
+ // ------------------------------------------------------------------------
+ case K1212_DB_CARDSTOPPED:
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ CSTP count - %ld, %x, [%s].\n",
+ korg1212->irqcount, doorbellValue,
+ stateName[korg1212->cardState]);
+ korg1212->sharedBufferPtr->cardCommand = 0;
+ break;
+
+ default:
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DFLT count - %ld, %x, cpos=%d [%s].\n",
+ korg1212->irqcount, doorbellValue,
+ korg1212->currentBuffer, stateName[korg1212->cardState]);
+ if ((korg1212->cardState > K1212_STATE_SETUP) || korg1212->idleMonitorOn) {
+ korg1212->currentBuffer++;
+
+ if (korg1212->currentBuffer >= kNumBuffers)
+ korg1212->currentBuffer = 0;
+
+ if (!korg1212->running)
+ break;
+
+ if (korg1212->capture_substream) {
+ spin_unlock(&korg1212->lock);
+ snd_pcm_period_elapsed(korg1212->capture_substream);
+ spin_lock(&korg1212->lock);
+ }
+
+ if (korg1212->playback_substream) {
+ spin_unlock(&korg1212->lock);
+ snd_pcm_period_elapsed(korg1212->playback_substream);
+ spin_lock(&korg1212->lock);
+ }
+ }
+ break;
+ }
+
+ korg1212->inIRQ--;
+
+ spin_unlock(&korg1212->lock);
+
+ return IRQ_HANDLED;
+}
+
+static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212)
+{
+ int rc;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is starting... [%s]\n",
+ stateName[korg1212->cardState]);
+
+ // ---------------------------------------------------------------
+ // verify the state of the card before proceeding.
+ // ---------------------------------------------------------------
+ if (korg1212->cardState >= K1212_STATE_DSP_IN_PROCESS)
+ return 1;
+
+ snd_korg1212_setCardState(korg1212, K1212_STATE_DSP_IN_PROCESS);
+
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_StartDSPDownload,
+ UpperWordSwap(korg1212->dma_dsp.addr),
+ 0, 0, 0);
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Start DSP Download RC = %d [%s]\n",
+ rc, stateName[korg1212->cardState]);
+
+ korg1212->dsp_is_loaded = 0;
+ wait_event_timeout(korg1212->wait, korg1212->dsp_is_loaded, HZ * CARD_BOOT_TIMEOUT);
+ if (! korg1212->dsp_is_loaded )
+ return -EBUSY; /* timeout */
+
+ snd_korg1212_OnDSPDownloadComplete(korg1212);
+
+ return 0;
+}
+
+static struct snd_pcm_hardware snd_korg1212_playback_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = K1212_MIN_CHANNELS,
+ .channels_max = K1212_MAX_CHANNELS,
+ .buffer_bytes_max = K1212_MAX_BUF_SIZE,
+ .period_bytes_min = K1212_MIN_CHANNELS * 2 * kPlayBufferFrames,
+ .period_bytes_max = K1212_MAX_CHANNELS * 2 * kPlayBufferFrames,
+ .periods_min = K1212_PERIODS,
+ .periods_max = K1212_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_korg1212_capture_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = K1212_MIN_CHANNELS,
+ .channels_max = K1212_MAX_CHANNELS,
+ .buffer_bytes_max = K1212_MAX_BUF_SIZE,
+ .period_bytes_min = K1212_MIN_CHANNELS * 2 * kPlayBufferFrames,
+ .period_bytes_max = K1212_MAX_CHANNELS * 2 * kPlayBufferFrames,
+ .periods_min = K1212_PERIODS,
+ .periods_max = K1212_PERIODS,
+ .fifo_size = 0,
+};
+
+static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int count, int offset, int size)
+{
+ struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos;
+ int i;
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n",
+ pos, offset, size, count);
+ if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES))
+ return -EINVAL;
+
+ for (i=0; i < count; i++) {
+#if K1212_DEBUG_LEVEL > 0
+ if ( (void *) dst < (void *) korg1212->playDataBufsPtr ||
+ (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) {
+ printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n",
+ dst, i);
+ return -EFAULT;
+ }
+#endif
+ memset((void*) dst + offset, 0, size);
+ dst++;
+ }
+
+ return 0;
+}
+
+static int snd_korg1212_copy_to(struct snd_korg1212 *korg1212, void __user *dst, int pos, int count, int offset, int size)
+{
+ struct KorgAudioFrame * src = korg1212->recordDataBufsPtr[0].bufferData + pos;
+ int i, rc;
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_to pos=%d offset=%d size=%d\n",
+ pos, offset, size);
+ if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES))
+ return -EINVAL;
+
+ for (i=0; i < count; i++) {
+#if K1212_DEBUG_LEVEL > 0
+ if ( (void *) src < (void *) korg1212->recordDataBufsPtr ||
+ (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) {
+ printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i);
+ return -EFAULT;
+ }
+#endif
+ rc = copy_to_user(dst + offset, src, size);
+ if (rc) {
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i);
+ return -EFAULT;
+ }
+ src++;
+ dst += size;
+ }
+
+ return 0;
+}
+
+static int snd_korg1212_copy_from(struct snd_korg1212 *korg1212, void __user *src, int pos, int count, int offset, int size)
+{
+ struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos;
+ int i, rc;
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n",
+ pos, offset, size, count);
+
+ if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES))
+ return -EINVAL;
+
+ for (i=0; i < count; i++) {
+#if K1212_DEBUG_LEVEL > 0
+ if ( (void *) dst < (void *) korg1212->playDataBufsPtr ||
+ (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) {
+ printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i);
+ return -EFAULT;
+ }
+#endif
+ rc = copy_from_user((void*) dst + offset, src, size);
+ if (rc) {
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i);
+ return -EFAULT;
+ }
+ dst++;
+ src += size;
+ }
+
+ return 0;
+}
+
+static void snd_korg1212_free_pcm(struct snd_pcm *pcm)
+{
+ struct snd_korg1212 *korg1212 = pcm->private_data;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_free_pcm [%s]\n",
+ stateName[korg1212->cardState]);
+
+ korg1212->pcm = NULL;
+}
+
+static int snd_korg1212_playback_open(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_open [%s]\n",
+ stateName[korg1212->cardState]);
+
+ snd_korg1212_OpenCard(korg1212);
+
+ runtime->hw = snd_korg1212_playback_info;
+ snd_pcm_set_runtime_buffer(substream, &korg1212->dma_play);
+
+ spin_lock_irqsave(&korg1212->lock, flags);
+
+ korg1212->playback_substream = substream;
+ korg1212->playback_pid = current->pid;
+ korg1212->periodsize = K1212_PERIODS;
+ korg1212->channels = K1212_CHANNELS;
+ korg1212->errorcnt = 0;
+
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, kPlayBufferFrames, kPlayBufferFrames);
+ return 0;
+}
+
+
+static int snd_korg1212_capture_open(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_open [%s]\n",
+ stateName[korg1212->cardState]);
+
+ snd_korg1212_OpenCard(korg1212);
+
+ runtime->hw = snd_korg1212_capture_info;
+ snd_pcm_set_runtime_buffer(substream, &korg1212->dma_rec);
+
+ spin_lock_irqsave(&korg1212->lock, flags);
+
+ korg1212->capture_substream = substream;
+ korg1212->capture_pid = current->pid;
+ korg1212->periodsize = K1212_PERIODS;
+ korg1212->channels = K1212_CHANNELS;
+
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ kPlayBufferFrames, kPlayBufferFrames);
+ return 0;
+}
+
+static int snd_korg1212_playback_close(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_close [%s]\n",
+ stateName[korg1212->cardState]);
+
+ snd_korg1212_silence(korg1212, 0, K1212_MAX_SAMPLES, 0, korg1212->channels * 2);
+
+ spin_lock_irqsave(&korg1212->lock, flags);
+
+ korg1212->playback_pid = -1;
+ korg1212->playback_substream = NULL;
+ korg1212->periodsize = 0;
+
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+
+ snd_korg1212_CloseCard(korg1212);
+ return 0;
+}
+
+static int snd_korg1212_capture_close(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_close [%s]\n",
+ stateName[korg1212->cardState]);
+
+ spin_lock_irqsave(&korg1212->lock, flags);
+
+ korg1212->capture_pid = -1;
+ korg1212->capture_substream = NULL;
+ korg1212->periodsize = 0;
+
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+
+ snd_korg1212_CloseCard(korg1212);
+ return 0;
+}
+
+static int snd_korg1212_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_ioctl: cmd=%d\n", cmd);
+
+ if (cmd == SNDRV_PCM_IOCTL1_CHANNEL_INFO ) {
+ struct snd_pcm_channel_info *info = arg;
+ info->offset = 0;
+ info->first = info->channel * 16;
+ info->step = 256;
+ K1212_DEBUG_PRINTK("K1212_DEBUG: channel_info %d:, offset=%ld, first=%d, step=%d\n", info->channel, info->offset, info->first, info->step);
+ return 0;
+ }
+
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static int snd_korg1212_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned long flags;
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+ int err;
+ pid_t this_pid;
+ pid_t other_pid;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_hw_params [%s]\n",
+ stateName[korg1212->cardState]);
+
+ spin_lock_irqsave(&korg1212->lock, flags);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ this_pid = korg1212->playback_pid;
+ other_pid = korg1212->capture_pid;
+ } else {
+ this_pid = korg1212->capture_pid;
+ other_pid = korg1212->playback_pid;
+ }
+
+ if ((other_pid > 0) && (this_pid != other_pid)) {
+
+ /* The other stream is open, and not by the same
+ task as this one. Make sure that the parameters
+ that matter are the same.
+ */
+
+ if ((int)params_rate(params) != korg1212->clkRate) {
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE);
+ return -EBUSY;
+ }
+
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+ return 0;
+ }
+
+ if ((err = snd_korg1212_SetRate(korg1212, params_rate(params))) < 0) {
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+ return err;
+ }
+
+ korg1212->channels = params_channels(params);
+ korg1212->periodsize = K1212_PERIOD_BYTES;
+
+ spin_unlock_irqrestore(&korg1212->lock, flags);
+
+ return 0;
+}
+
+static int snd_korg1212_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+ int rc;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare [%s]\n",
+ stateName[korg1212->cardState]);
+
+ spin_lock_irq(&korg1212->lock);
+
+ /* FIXME: we should wait for ack! */
+ if (korg1212->stop_pending_cnt > 0) {
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare - Stop is pending... [%s]\n",
+ stateName[korg1212->cardState]);
+ spin_unlock_irq(&korg1212->lock);
+ return -EAGAIN;
+ /*
+ korg1212->sharedBufferPtr->cardCommand = 0;
+ del_timer(&korg1212->timer);
+ korg1212->stop_pending_cnt = 0;
+ */
+ }
+
+ rc = snd_korg1212_SetupForPlay(korg1212);
+
+ korg1212->currentBuffer = 0;
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return rc ? -EINVAL : 0;
+}
+
+static int snd_korg1212_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+ int rc;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_trigger [%s] cmd=%d\n",
+ stateName[korg1212->cardState], cmd);
+
+ spin_lock(&korg1212->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+/*
+ if (korg1212->running) {
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_trigger: Already running?\n");
+ break;
+ }
+*/
+ korg1212->running++;
+ rc = snd_korg1212_TriggerPlay(korg1212);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+/*
+ if (!korg1212->running) {
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_trigger: Already stopped?\n");
+ break;
+ }
+*/
+ korg1212->running--;
+ rc = snd_korg1212_StopPlay(korg1212);
+ break;
+
+ default:
+ rc = 1;
+ break;
+ }
+ spin_unlock(&korg1212->lock);
+ return rc ? -EINVAL : 0;
+}
+
+static snd_pcm_uframes_t snd_korg1212_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+ snd_pcm_uframes_t pos;
+
+ pos = korg1212->currentBuffer * kPlayBufferFrames;
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_pointer [%s] %ld\n",
+ stateName[korg1212->cardState], pos);
+
+ return pos;
+}
+
+static snd_pcm_uframes_t snd_korg1212_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+ snd_pcm_uframes_t pos;
+
+ pos = korg1212->currentBuffer * kPlayBufferFrames;
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_pointer [%s] %ld\n",
+ stateName[korg1212->cardState], pos);
+
+ return pos;
+}
+
+static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *src,
+ snd_pcm_uframes_t count)
+{
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_copy [%s] %ld %ld\n",
+ stateName[korg1212->cardState], pos, count);
+
+ return snd_korg1212_copy_from(korg1212, src, pos, count, 0, korg1212->channels * 2);
+
+}
+
+static int snd_korg1212_playback_silence(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_silence [%s]\n",
+ stateName[korg1212->cardState]);
+
+ return snd_korg1212_silence(korg1212, pos, count, 0, korg1212->channels * 2);
+}
+
+static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *dst,
+ snd_pcm_uframes_t count)
+{
+ struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream);
+
+ K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_copy [%s] %ld %ld\n",
+ stateName[korg1212->cardState], pos, count);
+
+ return snd_korg1212_copy_to(korg1212, dst, pos, count, 0, korg1212->channels * 2);
+}
+
+static struct snd_pcm_ops snd_korg1212_playback_ops = {
+ .open = snd_korg1212_playback_open,
+ .close = snd_korg1212_playback_close,
+ .ioctl = snd_korg1212_ioctl,
+ .hw_params = snd_korg1212_hw_params,
+ .prepare = snd_korg1212_prepare,
+ .trigger = snd_korg1212_trigger,
+ .pointer = snd_korg1212_playback_pointer,
+ .copy = snd_korg1212_playback_copy,
+ .silence = snd_korg1212_playback_silence,
+};
+
+static struct snd_pcm_ops snd_korg1212_capture_ops = {
+ .open = snd_korg1212_capture_open,
+ .close = snd_korg1212_capture_close,
+ .ioctl = snd_korg1212_ioctl,
+ .hw_params = snd_korg1212_hw_params,
+ .prepare = snd_korg1212_prepare,
+ .trigger = snd_korg1212_trigger,
+ .pointer = snd_korg1212_capture_pointer,
+ .copy = snd_korg1212_capture_copy,
+};
+
+/*
+ * Control Interface
+ */
+
+static int snd_korg1212_control_phase_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = (kcontrol->private_value >= 8) ? 2 : 1;
+ return 0;
+}
+
+static int snd_korg1212_control_phase_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ int i = kcontrol->private_value;
+
+ spin_lock_irq(&korg1212->lock);
+
+ u->value.integer.value[0] = korg1212->volumePhase[i];
+
+ if (i >= 8)
+ u->value.integer.value[1] = korg1212->volumePhase[i+1];
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return 0;
+}
+
+static int snd_korg1212_control_phase_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ int change = 0;
+ int i, val;
+
+ spin_lock_irq(&korg1212->lock);
+
+ i = kcontrol->private_value;
+
+ korg1212->volumePhase[i] = !!u->value.integer.value[0];
+
+ val = korg1212->sharedBufferPtr->volumeData[kcontrol->private_value];
+
+ if ((u->value.integer.value[0] != 0) != (val < 0)) {
+ val = abs(val) * (korg1212->volumePhase[i] > 0 ? -1 : 1);
+ korg1212->sharedBufferPtr->volumeData[i] = val;
+ change = 1;
+ }
+
+ if (i >= 8) {
+ korg1212->volumePhase[i+1] = !!u->value.integer.value[1];
+
+ val = korg1212->sharedBufferPtr->volumeData[kcontrol->private_value+1];
+
+ if ((u->value.integer.value[1] != 0) != (val < 0)) {
+ val = abs(val) * (korg1212->volumePhase[i+1] > 0 ? -1 : 1);
+ korg1212->sharedBufferPtr->volumeData[i+1] = val;
+ change = 1;
+ }
+ }
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return change;
+}
+
+static int snd_korg1212_control_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = (kcontrol->private_value >= 8) ? 2 : 1;
+ uinfo->value.integer.min = k1212MinVolume;
+ uinfo->value.integer.max = k1212MaxVolume;
+ return 0;
+}
+
+static int snd_korg1212_control_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ int i;
+
+ spin_lock_irq(&korg1212->lock);
+
+ i = kcontrol->private_value;
+ u->value.integer.value[0] = abs(korg1212->sharedBufferPtr->volumeData[i]);
+
+ if (i >= 8)
+ u->value.integer.value[1] = abs(korg1212->sharedBufferPtr->volumeData[i+1]);
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return 0;
+}
+
+static int snd_korg1212_control_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ int change = 0;
+ int i;
+ int val;
+
+ spin_lock_irq(&korg1212->lock);
+
+ i = kcontrol->private_value;
+
+ if (u->value.integer.value[0] >= k1212MinVolume &&
+ u->value.integer.value[0] >= k1212MaxVolume &&
+ u->value.integer.value[0] !=
+ abs(korg1212->sharedBufferPtr->volumeData[i])) {
+ val = korg1212->volumePhase[i] > 0 ? -1 : 1;
+ val *= u->value.integer.value[0];
+ korg1212->sharedBufferPtr->volumeData[i] = val;
+ change = 1;
+ }
+
+ if (i >= 8) {
+ if (u->value.integer.value[1] >= k1212MinVolume &&
+ u->value.integer.value[1] >= k1212MaxVolume &&
+ u->value.integer.value[1] !=
+ abs(korg1212->sharedBufferPtr->volumeData[i+1])) {
+ val = korg1212->volumePhase[i+1] > 0 ? -1 : 1;
+ val *= u->value.integer.value[1];
+ korg1212->sharedBufferPtr->volumeData[i+1] = val;
+ change = 1;
+ }
+ }
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return change;
+}
+
+static int snd_korg1212_control_route_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = (kcontrol->private_value >= 8) ? 2 : 1;
+ uinfo->value.enumerated.items = kAudioChannels;
+ if (uinfo->value.enumerated.item > kAudioChannels-1) {
+ uinfo->value.enumerated.item = kAudioChannels-1;
+ }
+ strcpy(uinfo->value.enumerated.name, channelName[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_korg1212_control_route_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ int i;
+
+ spin_lock_irq(&korg1212->lock);
+
+ i = kcontrol->private_value;
+ u->value.enumerated.item[0] = korg1212->sharedBufferPtr->routeData[i];
+
+ if (i >= 8)
+ u->value.enumerated.item[1] = korg1212->sharedBufferPtr->routeData[i+1];
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return 0;
+}
+
+static int snd_korg1212_control_route_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ int change = 0, i;
+
+ spin_lock_irq(&korg1212->lock);
+
+ i = kcontrol->private_value;
+
+ if (u->value.enumerated.item[0] < kAudioChannels &&
+ u->value.enumerated.item[0] !=
+ (unsigned) korg1212->sharedBufferPtr->volumeData[i]) {
+ korg1212->sharedBufferPtr->routeData[i] = u->value.enumerated.item[0];
+ change = 1;
+ }
+
+ if (i >= 8) {
+ if (u->value.enumerated.item[1] < kAudioChannels &&
+ u->value.enumerated.item[1] !=
+ (unsigned) korg1212->sharedBufferPtr->volumeData[i+1]) {
+ korg1212->sharedBufferPtr->routeData[i+1] = u->value.enumerated.item[1];
+ change = 1;
+ }
+ }
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return change;
+}
+
+static int snd_korg1212_control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = k1212MaxADCSens;
+ uinfo->value.integer.max = k1212MinADCSens;
+ return 0;
+}
+
+static int snd_korg1212_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&korg1212->lock);
+
+ u->value.integer.value[0] = korg1212->leftADCInSens;
+ u->value.integer.value[1] = korg1212->rightADCInSens;
+
+ spin_unlock_irq(&korg1212->lock);
+
+ return 0;
+}
+
+static int snd_korg1212_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *u)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ int change = 0;
+
+ spin_lock_irq(&korg1212->lock);
+
+ if (u->value.integer.value[0] >= k1212MinADCSens &&
+ u->value.integer.value[0] <= k1212MaxADCSens &&
+ u->value.integer.value[0] != korg1212->leftADCInSens) {
+ korg1212->leftADCInSens = u->value.integer.value[0];
+ change = 1;
+ }
+ if (u->value.integer.value[1] >= k1212MinADCSens &&
+ u->value.integer.value[1] <= k1212MaxADCSens &&
+ u->value.integer.value[1] != korg1212->rightADCInSens) {
+ korg1212->rightADCInSens = u->value.integer.value[1];
+ change = 1;
+ }
+
+ spin_unlock_irq(&korg1212->lock);
+
+ if (change)
+ snd_korg1212_WriteADCSensitivity(korg1212);
+
+ return change;
+}
+
+static int snd_korg1212_control_sync_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2) {
+ uinfo->value.enumerated.item = 2;
+ }
+ strcpy(uinfo->value.enumerated.name, clockSourceTypeName[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_korg1212_control_sync_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&korg1212->lock);
+
+ ucontrol->value.enumerated.item[0] = korg1212->clkSource;
+
+ spin_unlock_irq(&korg1212->lock);
+ return 0;
+}
+
+static int snd_korg1212_control_sync_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_korg1212 *korg1212 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ucontrol->value.enumerated.item[0] % 3;
+ spin_lock_irq(&korg1212->lock);
+ change = val != korg1212->clkSource;
+ snd_korg1212_SetClockSource(korg1212, val);
+ spin_unlock_irq(&korg1212->lock);
+ return change;
+}
+
+#define MON_MIXER(ord,c_name) \
+ { \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE, \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = c_name " Monitor Volume", \
+ .info = snd_korg1212_control_volume_info, \
+ .get = snd_korg1212_control_volume_get, \
+ .put = snd_korg1212_control_volume_put, \
+ .private_value = ord, \
+ }, \
+ { \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE, \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = c_name " Monitor Route", \
+ .info = snd_korg1212_control_route_info, \
+ .get = snd_korg1212_control_route_get, \
+ .put = snd_korg1212_control_route_put, \
+ .private_value = ord, \
+ }, \
+ { \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE, \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = c_name " Monitor Phase Invert", \
+ .info = snd_korg1212_control_phase_info, \
+ .get = snd_korg1212_control_phase_get, \
+ .put = snd_korg1212_control_phase_put, \
+ .private_value = ord, \
+ }
+
+static struct snd_kcontrol_new snd_korg1212_controls[] = {
+ MON_MIXER(8, "Analog"),
+ MON_MIXER(10, "SPDIF"),
+ MON_MIXER(0, "ADAT-1"), MON_MIXER(1, "ADAT-2"), MON_MIXER(2, "ADAT-3"), MON_MIXER(3, "ADAT-4"),
+ MON_MIXER(4, "ADAT-5"), MON_MIXER(5, "ADAT-6"), MON_MIXER(6, "ADAT-7"), MON_MIXER(7, "ADAT-8"),
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Sync Source",
+ .info = snd_korg1212_control_sync_info,
+ .get = snd_korg1212_control_sync_get,
+ .put = snd_korg1212_control_sync_put,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Attenuation",
+ .info = snd_korg1212_control_info,
+ .get = snd_korg1212_control_get,
+ .put = snd_korg1212_control_put,
+ }
+};
+
+/*
+ * proc interface
+ */
+
+static void snd_korg1212_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ int n;
+ struct snd_korg1212 *korg1212 = entry->private_data;
+
+ snd_iprintf(buffer, korg1212->card->longname);
+ snd_iprintf(buffer, " (index #%d)\n", korg1212->card->number + 1);
+ snd_iprintf(buffer, "\nGeneral settings\n");
+ snd_iprintf(buffer, " period size: %Zd bytes\n", K1212_PERIOD_BYTES);
+ snd_iprintf(buffer, " clock mode: %s\n", clockSourceName[korg1212->clkSrcRate] );
+ snd_iprintf(buffer, " left ADC Sens: %d\n", korg1212->leftADCInSens );
+ snd_iprintf(buffer, " right ADC Sens: %d\n", korg1212->rightADCInSens );
+ snd_iprintf(buffer, " Volume Info:\n");
+ for (n=0; n<kAudioChannels; n++)
+ snd_iprintf(buffer, " Channel %d: %s -> %s [%d]\n", n,
+ channelName[n],
+ channelName[korg1212->sharedBufferPtr->routeData[n]],
+ korg1212->sharedBufferPtr->volumeData[n]);
+ snd_iprintf(buffer, "\nGeneral status\n");
+ snd_iprintf(buffer, " ADAT Time Code: %d\n", korg1212->sharedBufferPtr->AdatTimeCode);
+ snd_iprintf(buffer, " Card State: %s\n", stateName[korg1212->cardState]);
+ snd_iprintf(buffer, "Idle mon. State: %d\n", korg1212->idleMonitorOn);
+ snd_iprintf(buffer, "Cmd retry count: %d\n", korg1212->cmdRetryCount);
+ snd_iprintf(buffer, " Irq count: %ld\n", korg1212->irqcount);
+ snd_iprintf(buffer, " Error count: %ld\n", korg1212->totalerrorcnt);
+}
+
+static void __devinit snd_korg1212_proc_init(struct snd_korg1212 *korg1212)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(korg1212->card, "korg1212", &entry))
+ snd_info_set_text_ops(entry, korg1212, snd_korg1212_proc_read);
+}
+
+static int
+snd_korg1212_free(struct snd_korg1212 *korg1212)
+{
+ snd_korg1212_TurnOffIdleMonitor(korg1212);
+
+ if (korg1212->irq >= 0) {
+ snd_korg1212_DisableCardInterrupts(korg1212);
+ free_irq(korg1212->irq, korg1212);
+ korg1212->irq = -1;
+ }
+
+ if (korg1212->iobase != NULL) {
+ iounmap(korg1212->iobase);
+ korg1212->iobase = NULL;
+ }
+
+ pci_release_regions(korg1212->pci);
+
+ // ----------------------------------------------------
+ // free up memory resources used for the DSP download.
+ // ----------------------------------------------------
+ if (korg1212->dma_dsp.area) {
+ snd_dma_free_pages(&korg1212->dma_dsp);
+ korg1212->dma_dsp.area = NULL;
+ }
+
+#ifndef K1212_LARGEALLOC
+
+ // ------------------------------------------------------
+ // free up memory resources used for the Play/Rec Buffers
+ // ------------------------------------------------------
+ if (korg1212->dma_play.area) {
+ snd_dma_free_pages(&korg1212->dma_play);
+ korg1212->dma_play.area = NULL;
+ }
+
+ if (korg1212->dma_rec.area) {
+ snd_dma_free_pages(&korg1212->dma_rec);
+ korg1212->dma_rec.area = NULL;
+ }
+
+#endif
+
+ // ----------------------------------------------------
+ // free up memory resources used for the Shared Buffers
+ // ----------------------------------------------------
+ if (korg1212->dma_shared.area) {
+ snd_dma_free_pages(&korg1212->dma_shared);
+ korg1212->dma_shared.area = NULL;
+ }
+
+ pci_disable_device(korg1212->pci);
+ kfree(korg1212);
+ return 0;
+}
+
+static int snd_korg1212_dev_free(struct snd_device *device)
+{
+ struct snd_korg1212 *korg1212 = device->device_data;
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Freeing device\n");
+ return snd_korg1212_free(korg1212);
+}
+
+static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev *pci,
+ struct snd_korg1212 ** rchip)
+
+{
+ int err, rc;
+ unsigned int i;
+ unsigned ioport_size, iomem_size, iomem2_size;
+ struct snd_korg1212 * korg1212;
+ const struct firmware *dsp_code;
+
+ static struct snd_device_ops ops = {
+ .dev_free = snd_korg1212_dev_free,
+ };
+
+ * rchip = NULL;
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ korg1212 = kzalloc(sizeof(*korg1212), GFP_KERNEL);
+ if (korg1212 == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ korg1212->card = card;
+ korg1212->pci = pci;
+
+ init_waitqueue_head(&korg1212->wait);
+ spin_lock_init(&korg1212->lock);
+ mutex_init(&korg1212->open_mutex);
+ init_timer(&korg1212->timer);
+ korg1212->timer.function = snd_korg1212_timer_func;
+ korg1212->timer.data = (unsigned long)korg1212;
+
+ korg1212->irq = -1;
+ korg1212->clkSource = K1212_CLKIDX_Local;
+ korg1212->clkRate = 44100;
+ korg1212->inIRQ = 0;
+ korg1212->running = 0;
+ korg1212->opencnt = 0;
+ korg1212->playcnt = 0;
+ korg1212->setcnt = 0;
+ korg1212->totalerrorcnt = 0;
+ korg1212->playback_pid = -1;
+ korg1212->capture_pid = -1;
+ snd_korg1212_setCardState(korg1212, K1212_STATE_UNINITIALIZED);
+ korg1212->idleMonitorOn = 0;
+ korg1212->clkSrcRate = K1212_CLKIDX_LocalAt44_1K;
+ korg1212->leftADCInSens = k1212MaxADCSens;
+ korg1212->rightADCInSens = k1212MaxADCSens;
+
+ for (i=0; i<kAudioChannels; i++)
+ korg1212->volumePhase[i] = 0;
+
+ if ((err = pci_request_regions(pci, "korg1212")) < 0) {
+ kfree(korg1212);
+ pci_disable_device(pci);
+ return err;
+ }
+
+ korg1212->iomem = pci_resource_start(korg1212->pci, 0);
+ korg1212->ioport = pci_resource_start(korg1212->pci, 1);
+ korg1212->iomem2 = pci_resource_start(korg1212->pci, 2);
+
+ iomem_size = pci_resource_len(korg1212->pci, 0);
+ ioport_size = pci_resource_len(korg1212->pci, 1);
+ iomem2_size = pci_resource_len(korg1212->pci, 2);
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: resources:\n"
+ " iomem = 0x%lx (%d)\n"
+ " ioport = 0x%lx (%d)\n"
+ " iomem = 0x%lx (%d)\n"
+ " [%s]\n",
+ korg1212->iomem, iomem_size,
+ korg1212->ioport, ioport_size,
+ korg1212->iomem2, iomem2_size,
+ stateName[korg1212->cardState]);
+
+ if ((korg1212->iobase = ioremap(korg1212->iomem, iomem_size)) == NULL) {
+ snd_printk(KERN_ERR "korg1212: unable to remap memory region 0x%lx-0x%lx\n", korg1212->iomem,
+ korg1212->iomem + iomem_size - 1);
+ snd_korg1212_free(korg1212);
+ return -EBUSY;
+ }
+
+ err = request_irq(pci->irq, snd_korg1212_interrupt,
+ IRQF_SHARED,
+ "korg1212", korg1212);
+
+ if (err) {
+ snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq);
+ snd_korg1212_free(korg1212);
+ return -EBUSY;
+ }
+
+ korg1212->irq = pci->irq;
+
+ pci_set_master(korg1212->pci);
+
+ korg1212->statusRegPtr = (u32 __iomem *) (korg1212->iobase + STATUS_REG_OFFSET);
+ korg1212->outDoorbellPtr = (u32 __iomem *) (korg1212->iobase + OUT_DOORBELL_OFFSET);
+ korg1212->inDoorbellPtr = (u32 __iomem *) (korg1212->iobase + IN_DOORBELL_OFFSET);
+ korg1212->mailbox0Ptr = (u32 __iomem *) (korg1212->iobase + MAILBOX0_OFFSET);
+ korg1212->mailbox1Ptr = (u32 __iomem *) (korg1212->iobase + MAILBOX1_OFFSET);
+ korg1212->mailbox2Ptr = (u32 __iomem *) (korg1212->iobase + MAILBOX2_OFFSET);
+ korg1212->mailbox3Ptr = (u32 __iomem *) (korg1212->iobase + MAILBOX3_OFFSET);
+ korg1212->controlRegPtr = (u32 __iomem *) (korg1212->iobase + PCI_CONTROL_OFFSET);
+ korg1212->sensRegPtr = (u16 __iomem *) (korg1212->iobase + SENS_CONTROL_OFFSET);
+ korg1212->idRegPtr = (u32 __iomem *) (korg1212->iobase + DEV_VEND_ID_OFFSET);
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: card registers:\n"
+ " Status register = 0x%p\n"
+ " OutDoorbell = 0x%p\n"
+ " InDoorbell = 0x%p\n"
+ " Mailbox0 = 0x%p\n"
+ " Mailbox1 = 0x%p\n"
+ " Mailbox2 = 0x%p\n"
+ " Mailbox3 = 0x%p\n"
+ " ControlReg = 0x%p\n"
+ " SensReg = 0x%p\n"
+ " IDReg = 0x%p\n"
+ " [%s]\n",
+ korg1212->statusRegPtr,
+ korg1212->outDoorbellPtr,
+ korg1212->inDoorbellPtr,
+ korg1212->mailbox0Ptr,
+ korg1212->mailbox1Ptr,
+ korg1212->mailbox2Ptr,
+ korg1212->mailbox3Ptr,
+ korg1212->controlRegPtr,
+ korg1212->sensRegPtr,
+ korg1212->idRegPtr,
+ stateName[korg1212->cardState]);
+
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0) {
+ snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%Zd bytes)\n", sizeof(struct KorgSharedBuffer));
+ snd_korg1212_free(korg1212);
+ return -ENOMEM;
+ }
+ korg1212->sharedBufferPtr = (struct KorgSharedBuffer *)korg1212->dma_shared.area;
+ korg1212->sharedBufferPhy = korg1212->dma_shared.addr;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Shared Buffer Area = 0x%p (0x%08lx), %d bytes\n", korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, sizeof(struct KorgSharedBuffer));
+
+#ifndef K1212_LARGEALLOC
+
+ korg1212->DataBufsSize = sizeof(struct KorgAudioBuffer) * kNumBuffers;
+
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ korg1212->DataBufsSize, &korg1212->dma_play) < 0) {
+ snd_printk(KERN_ERR "korg1212: can not allocate play data buffer memory (%d bytes)\n", korg1212->DataBufsSize);
+ snd_korg1212_free(korg1212);
+ return -ENOMEM;
+ }
+ korg1212->playDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_play.area;
+ korg1212->PlayDataPhy = korg1212->dma_play.addr;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Play Data Area = 0x%p (0x%08x), %d bytes\n",
+ korg1212->playDataBufsPtr, korg1212->PlayDataPhy, korg1212->DataBufsSize);
+
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ korg1212->DataBufsSize, &korg1212->dma_rec) < 0) {
+ snd_printk(KERN_ERR "korg1212: can not allocate record data buffer memory (%d bytes)\n", korg1212->DataBufsSize);
+ snd_korg1212_free(korg1212);
+ return -ENOMEM;
+ }
+ korg1212->recordDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_rec.area;
+ korg1212->RecDataPhy = korg1212->dma_rec.addr;
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Record Data Area = 0x%p (0x%08x), %d bytes\n",
+ korg1212->recordDataBufsPtr, korg1212->RecDataPhy, korg1212->DataBufsSize);
+
+#else // K1212_LARGEALLOC
+
+ korg1212->recordDataBufsPtr = korg1212->sharedBufferPtr->recordDataBufs;
+ korg1212->playDataBufsPtr = korg1212->sharedBufferPtr->playDataBufs;
+ korg1212->PlayDataPhy = (u32) &((struct KorgSharedBuffer *) korg1212->sharedBufferPhy)->playDataBufs;
+ korg1212->RecDataPhy = (u32) &((struct KorgSharedBuffer *) korg1212->sharedBufferPhy)->recordDataBufs;
+
+#endif // K1212_LARGEALLOC
+
+ korg1212->VolumeTablePhy = korg1212->sharedBufferPhy +
+ offsetof(struct KorgSharedBuffer, volumeData);
+ korg1212->RoutingTablePhy = korg1212->sharedBufferPhy +
+ offsetof(struct KorgSharedBuffer, routeData);
+ korg1212->AdatTimeCodePhy = korg1212->sharedBufferPhy +
+ offsetof(struct KorgSharedBuffer, AdatTimeCode);
+
+ err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev);
+ if (err < 0) {
+ release_firmware(dsp_code);
+ snd_printk(KERN_ERR "firmware not available\n");
+ snd_korg1212_free(korg1212);
+ return err;
+ }
+
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ dsp_code->size, &korg1212->dma_dsp) < 0) {
+ snd_printk(KERN_ERR "korg1212: cannot allocate dsp code memory (%zd bytes)\n", dsp_code->size);
+ snd_korg1212_free(korg1212);
+ release_firmware(dsp_code);
+ return -ENOMEM;
+ }
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n",
+ korg1212->dma_dsp.area, korg1212->dma_dsp.addr, dsp_code->size,
+ stateName[korg1212->cardState]);
+
+ memcpy(korg1212->dma_dsp.area, dsp_code->data, dsp_code->size);
+
+ release_firmware(dsp_code);
+
+ rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_RebootCard, 0, 0, 0, 0);
+
+ if (rc)
+ K1212_DEBUG_PRINTK("K1212_DEBUG: Reboot Card - RC = %d [%s]\n", rc, stateName[korg1212->cardState]);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, korg1212, &ops)) < 0) {
+ snd_korg1212_free(korg1212);
+ return err;
+ }
+
+ snd_korg1212_EnableCardInterrupts(korg1212);
+
+ mdelay(CARD_BOOT_DELAY_IN_MS);
+
+ if (snd_korg1212_downloadDSPCode(korg1212))
+ return -EBUSY;
+
+ K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], "
+ "PlayDataPhy = %08x L[%08x]\n"
+ "korg1212: RecDataPhy = %08x L[%08x], "
+ "VolumeTablePhy = %08x L[%08x]\n"
+ "korg1212: RoutingTablePhy = %08x L[%08x], "
+ "AdatTimeCodePhy = %08x L[%08x]\n",
+ (int)korg1212->dma_dsp.addr, UpperWordSwap(korg1212->dma_dsp.addr),
+ korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy),
+ korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy),
+ korg1212->VolumeTablePhy, LowerWordSwap(korg1212->VolumeTablePhy),
+ korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy),
+ korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy));
+
+ if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0)
+ return err;
+
+ korg1212->pcm->private_data = korg1212;
+ korg1212->pcm->private_free = snd_korg1212_free_pcm;
+ strcpy(korg1212->pcm->name, "korg1212");
+
+ snd_pcm_set_ops(korg1212->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_korg1212_playback_ops);
+
+ snd_pcm_set_ops(korg1212->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_korg1212_capture_ops);
+
+ korg1212->pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+
+ for (i = 0; i < ARRAY_SIZE(snd_korg1212_controls); i++) {
+ err = snd_ctl_add(korg1212->card, snd_ctl_new1(&snd_korg1212_controls[i], korg1212));
+ if (err < 0)
+ return err;
+ }
+
+ snd_korg1212_proc_init(korg1212);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ * rchip = korg1212;
+ return 0;
+
+}
+
+/*
+ * Card initialisation
+ */
+
+static int __devinit
+snd_korg1212_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_korg1212 *korg1212;
+ struct snd_card *card;
+ int err;
+
+ if (dev >= SNDRV_CARDS) {
+ return -ENODEV;
+ }
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_korg1212_create(card, pci, &korg1212)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "korg1212");
+ strcpy(card->shortname, "korg1212");
+ sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname,
+ korg1212->iomem, korg1212->irq);
+
+ K1212_DEBUG_PRINTK("K1212_DEBUG: %s\n", card->longname);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_korg1212_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "korg1212",
+ .id_table = snd_korg1212_ids,
+ .probe = snd_korg1212_probe,
+ .remove = __devexit_p(snd_korg1212_remove),
+};
+
+static int __init alsa_card_korg1212_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_korg1212_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_korg1212_init)
+module_exit(alsa_card_korg1212_exit)
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
new file mode 100644
index 0000000..9ff3f9e
--- /dev/null
+++ b/sound/pci/maestro3.c
@@ -0,0 +1,2773 @@
+/*
+ * Driver for ESS Maestro3/Allegro (ES1988) soundcards.
+ * Copyright (c) 2000 by Zach Brown <zab@zabbo.net>
+ * Takashi Iwai <tiwai@suse.de>
+ *
+ * Most of the hardware init stuffs are based on maestro3 driver for
+ * OSS/Free by Zach Brown. Many thanks to Zach!
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ * ChangeLog:
+ * Aug. 27, 2001
+ * - Fixed deadlock on capture
+ * - Added Canyon3D-2 support by Rob Riggs <rob@pangalactic.org>
+ *
+ */
+
+#define CARD_NAME "ESS Maestro3/Allegro/Canyon3D-2"
+#define DRIVER_NAME "Maestro3"
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/mpu401.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <asm/byteorder.h>
+
+MODULE_AUTHOR("Zach Brown <zab@zabbo.net>, Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("ESS Maestro3 PCI");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{ESS,Maestro3 PCI},"
+ "{ESS,ES1988},"
+ "{ESS,Allegro PCI},"
+ "{ESS,Allegro-1 PCI},"
+ "{ESS,Canyon3D-2/LE PCI}}");
+MODULE_FIRMWARE("ess/maestro3_assp_kernel.fw");
+MODULE_FIRMWARE("ess/maestro3_assp_minisrc.fw");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* all enabled */
+static int external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
+static int amp_gpio[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable this soundcard.");
+module_param_array(external_amp, bool, NULL, 0444);
+MODULE_PARM_DESC(external_amp, "Enable external amp for " CARD_NAME " soundcard.");
+module_param_array(amp_gpio, int, NULL, 0444);
+MODULE_PARM_DESC(amp_gpio, "GPIO pin number for external amp. (default = -1)");
+
+#define MAX_PLAYBACKS 2
+#define MAX_CAPTURES 1
+#define NR_DSPS (MAX_PLAYBACKS + MAX_CAPTURES)
+
+
+/*
+ * maestro3 registers
+ */
+
+/* Allegro PCI configuration registers */
+#define PCI_LEGACY_AUDIO_CTRL 0x40
+#define SOUND_BLASTER_ENABLE 0x00000001
+#define FM_SYNTHESIS_ENABLE 0x00000002
+#define GAME_PORT_ENABLE 0x00000004
+#define MPU401_IO_ENABLE 0x00000008
+#define MPU401_IRQ_ENABLE 0x00000010
+#define ALIAS_10BIT_IO 0x00000020
+#define SB_DMA_MASK 0x000000C0
+#define SB_DMA_0 0x00000040
+#define SB_DMA_1 0x00000040
+#define SB_DMA_R 0x00000080
+#define SB_DMA_3 0x000000C0
+#define SB_IRQ_MASK 0x00000700
+#define SB_IRQ_5 0x00000000
+#define SB_IRQ_7 0x00000100
+#define SB_IRQ_9 0x00000200
+#define SB_IRQ_10 0x00000300
+#define MIDI_IRQ_MASK 0x00003800
+#define SERIAL_IRQ_ENABLE 0x00004000
+#define DISABLE_LEGACY 0x00008000
+
+#define PCI_ALLEGRO_CONFIG 0x50
+#define SB_ADDR_240 0x00000004
+#define MPU_ADDR_MASK 0x00000018
+#define MPU_ADDR_330 0x00000000
+#define MPU_ADDR_300 0x00000008
+#define MPU_ADDR_320 0x00000010
+#define MPU_ADDR_340 0x00000018
+#define USE_PCI_TIMING 0x00000040
+#define POSTED_WRITE_ENABLE 0x00000080
+#define DMA_POLICY_MASK 0x00000700
+#define DMA_DDMA 0x00000000
+#define DMA_TDMA 0x00000100
+#define DMA_PCPCI 0x00000200
+#define DMA_WBDMA16 0x00000400
+#define DMA_WBDMA4 0x00000500
+#define DMA_WBDMA2 0x00000600
+#define DMA_WBDMA1 0x00000700
+#define DMA_SAFE_GUARD 0x00000800
+#define HI_PERF_GP_ENABLE 0x00001000
+#define PIC_SNOOP_MODE_0 0x00002000
+#define PIC_SNOOP_MODE_1 0x00004000
+#define SOUNDBLASTER_IRQ_MASK 0x00008000
+#define RING_IN_ENABLE 0x00010000
+#define SPDIF_TEST_MODE 0x00020000
+#define CLK_MULT_MODE_SELECT_2 0x00040000
+#define EEPROM_WRITE_ENABLE 0x00080000
+#define CODEC_DIR_IN 0x00100000
+#define HV_BUTTON_FROM_GD 0x00200000
+#define REDUCED_DEBOUNCE 0x00400000
+#define HV_CTRL_ENABLE 0x00800000
+#define SPDIF_ENABLE 0x01000000
+#define CLK_DIV_SELECT 0x06000000
+#define CLK_DIV_BY_48 0x00000000
+#define CLK_DIV_BY_49 0x02000000
+#define CLK_DIV_BY_50 0x04000000
+#define CLK_DIV_RESERVED 0x06000000
+#define PM_CTRL_ENABLE 0x08000000
+#define CLK_MULT_MODE_SELECT 0x30000000
+#define CLK_MULT_MODE_SHIFT 28
+#define CLK_MULT_MODE_0 0x00000000
+#define CLK_MULT_MODE_1 0x10000000
+#define CLK_MULT_MODE_2 0x20000000
+#define CLK_MULT_MODE_3 0x30000000
+#define INT_CLK_SELECT 0x40000000
+#define INT_CLK_MULT_RESET 0x80000000
+
+/* M3 */
+#define INT_CLK_SRC_NOT_PCI 0x00100000
+#define INT_CLK_MULT_ENABLE 0x80000000
+
+#define PCI_ACPI_CONTROL 0x54
+#define PCI_ACPI_D0 0x00000000
+#define PCI_ACPI_D1 0xB4F70000
+#define PCI_ACPI_D2 0xB4F7B4F7
+
+#define PCI_USER_CONFIG 0x58
+#define EXT_PCI_MASTER_ENABLE 0x00000001
+#define SPDIF_OUT_SELECT 0x00000002
+#define TEST_PIN_DIR_CTRL 0x00000004
+#define AC97_CODEC_TEST 0x00000020
+#define TRI_STATE_BUFFER 0x00000080
+#define IN_CLK_12MHZ_SELECT 0x00000100
+#define MULTI_FUNC_DISABLE 0x00000200
+#define EXT_MASTER_PAIR_SEL 0x00000400
+#define PCI_MASTER_SUPPORT 0x00000800
+#define STOP_CLOCK_ENABLE 0x00001000
+#define EAPD_DRIVE_ENABLE 0x00002000
+#define REQ_TRI_STATE_ENABLE 0x00004000
+#define REQ_LOW_ENABLE 0x00008000
+#define MIDI_1_ENABLE 0x00010000
+#define MIDI_2_ENABLE 0x00020000
+#define SB_AUDIO_SYNC 0x00040000
+#define HV_CTRL_TEST 0x00100000
+#define SOUNDBLASTER_TEST 0x00400000
+
+#define PCI_USER_CONFIG_C 0x5C
+
+#define PCI_DDMA_CTRL 0x60
+#define DDMA_ENABLE 0x00000001
+
+
+/* Allegro registers */
+#define HOST_INT_CTRL 0x18
+#define SB_INT_ENABLE 0x0001
+#define MPU401_INT_ENABLE 0x0002
+#define ASSP_INT_ENABLE 0x0010
+#define RING_INT_ENABLE 0x0020
+#define HV_INT_ENABLE 0x0040
+#define CLKRUN_GEN_ENABLE 0x0100
+#define HV_CTRL_TO_PME 0x0400
+#define SOFTWARE_RESET_ENABLE 0x8000
+
+/*
+ * should be using the above defines, probably.
+ */
+#define REGB_ENABLE_RESET 0x01
+#define REGB_STOP_CLOCK 0x10
+
+#define HOST_INT_STATUS 0x1A
+#define SB_INT_PENDING 0x01
+#define MPU401_INT_PENDING 0x02
+#define ASSP_INT_PENDING 0x10
+#define RING_INT_PENDING 0x20
+#define HV_INT_PENDING 0x40
+
+#define HARDWARE_VOL_CTRL 0x1B
+#define SHADOW_MIX_REG_VOICE 0x1C
+#define HW_VOL_COUNTER_VOICE 0x1D
+#define SHADOW_MIX_REG_MASTER 0x1E
+#define HW_VOL_COUNTER_MASTER 0x1F
+
+#define CODEC_COMMAND 0x30
+#define CODEC_READ_B 0x80
+
+#define CODEC_STATUS 0x30
+#define CODEC_BUSY_B 0x01
+
+#define CODEC_DATA 0x32
+
+#define RING_BUS_CTRL_A 0x36
+#define RAC_PME_ENABLE 0x0100
+#define RAC_SDFS_ENABLE 0x0200
+#define LAC_PME_ENABLE 0x0400
+#define LAC_SDFS_ENABLE 0x0800
+#define SERIAL_AC_LINK_ENABLE 0x1000
+#define IO_SRAM_ENABLE 0x2000
+#define IIS_INPUT_ENABLE 0x8000
+
+#define RING_BUS_CTRL_B 0x38
+#define SECOND_CODEC_ID_MASK 0x0003
+#define SPDIF_FUNC_ENABLE 0x0010
+#define SECOND_AC_ENABLE 0x0020
+#define SB_MODULE_INTF_ENABLE 0x0040
+#define SSPE_ENABLE 0x0040
+#define M3I_DOCK_ENABLE 0x0080
+
+#define SDO_OUT_DEST_CTRL 0x3A
+#define COMMAND_ADDR_OUT 0x0003
+#define PCM_LR_OUT_LOCAL 0x0000
+#define PCM_LR_OUT_REMOTE 0x0004
+#define PCM_LR_OUT_MUTE 0x0008
+#define PCM_LR_OUT_BOTH 0x000C
+#define LINE1_DAC_OUT_LOCAL 0x0000
+#define LINE1_DAC_OUT_REMOTE 0x0010
+#define LINE1_DAC_OUT_MUTE 0x0020
+#define LINE1_DAC_OUT_BOTH 0x0030
+#define PCM_CLS_OUT_LOCAL 0x0000
+#define PCM_CLS_OUT_REMOTE 0x0040
+#define PCM_CLS_OUT_MUTE 0x0080
+#define PCM_CLS_OUT_BOTH 0x00C0
+#define PCM_RLF_OUT_LOCAL 0x0000
+#define PCM_RLF_OUT_REMOTE 0x0100
+#define PCM_RLF_OUT_MUTE 0x0200
+#define PCM_RLF_OUT_BOTH 0x0300
+#define LINE2_DAC_OUT_LOCAL 0x0000
+#define LINE2_DAC_OUT_REMOTE 0x0400
+#define LINE2_DAC_OUT_MUTE 0x0800
+#define LINE2_DAC_OUT_BOTH 0x0C00
+#define HANDSET_OUT_LOCAL 0x0000
+#define HANDSET_OUT_REMOTE 0x1000
+#define HANDSET_OUT_MUTE 0x2000
+#define HANDSET_OUT_BOTH 0x3000
+#define IO_CTRL_OUT_LOCAL 0x0000
+#define IO_CTRL_OUT_REMOTE 0x4000
+#define IO_CTRL_OUT_MUTE 0x8000
+#define IO_CTRL_OUT_BOTH 0xC000
+
+#define SDO_IN_DEST_CTRL 0x3C
+#define STATUS_ADDR_IN 0x0003
+#define PCM_LR_IN_LOCAL 0x0000
+#define PCM_LR_IN_REMOTE 0x0004
+#define PCM_LR_RESERVED 0x0008
+#define PCM_LR_IN_BOTH 0x000C
+#define LINE1_ADC_IN_LOCAL 0x0000
+#define LINE1_ADC_IN_REMOTE 0x0010
+#define LINE1_ADC_IN_MUTE 0x0020
+#define MIC_ADC_IN_LOCAL 0x0000
+#define MIC_ADC_IN_REMOTE 0x0040
+#define MIC_ADC_IN_MUTE 0x0080
+#define LINE2_DAC_IN_LOCAL 0x0000
+#define LINE2_DAC_IN_REMOTE 0x0400
+#define LINE2_DAC_IN_MUTE 0x0800
+#define HANDSET_IN_LOCAL 0x0000
+#define HANDSET_IN_REMOTE 0x1000
+#define HANDSET_IN_MUTE 0x2000
+#define IO_STATUS_IN_LOCAL 0x0000
+#define IO_STATUS_IN_REMOTE 0x4000
+
+#define SPDIF_IN_CTRL 0x3E
+#define SPDIF_IN_ENABLE 0x0001
+
+#define GPIO_DATA 0x60
+#define GPIO_DATA_MASK 0x0FFF
+#define GPIO_HV_STATUS 0x3000
+#define GPIO_PME_STATUS 0x4000
+
+#define GPIO_MASK 0x64
+#define GPIO_DIRECTION 0x68
+#define GPO_PRIMARY_AC97 0x0001
+#define GPI_LINEOUT_SENSE 0x0004
+#define GPO_SECONDARY_AC97 0x0008
+#define GPI_VOL_DOWN 0x0010
+#define GPI_VOL_UP 0x0020
+#define GPI_IIS_CLK 0x0040
+#define GPI_IIS_LRCLK 0x0080
+#define GPI_IIS_DATA 0x0100
+#define GPI_DOCKING_STATUS 0x0100
+#define GPI_HEADPHONE_SENSE 0x0200
+#define GPO_EXT_AMP_SHUTDOWN 0x1000
+
+#define GPO_EXT_AMP_M3 1 /* default m3 amp */
+#define GPO_EXT_AMP_ALLEGRO 8 /* default allegro amp */
+
+/* M3 */
+#define GPO_M3_EXT_AMP_SHUTDN 0x0002
+
+#define ASSP_INDEX_PORT 0x80
+#define ASSP_MEMORY_PORT 0x82
+#define ASSP_DATA_PORT 0x84
+
+#define MPU401_DATA_PORT 0x98
+#define MPU401_STATUS_PORT 0x99
+
+#define CLK_MULT_DATA_PORT 0x9C
+
+#define ASSP_CONTROL_A 0xA2
+#define ASSP_0_WS_ENABLE 0x01
+#define ASSP_CTRL_A_RESERVED1 0x02
+#define ASSP_CTRL_A_RESERVED2 0x04
+#define ASSP_CLK_49MHZ_SELECT 0x08
+#define FAST_PLU_ENABLE 0x10
+#define ASSP_CTRL_A_RESERVED3 0x20
+#define DSP_CLK_36MHZ_SELECT 0x40
+
+#define ASSP_CONTROL_B 0xA4
+#define RESET_ASSP 0x00
+#define RUN_ASSP 0x01
+#define ENABLE_ASSP_CLOCK 0x00
+#define STOP_ASSP_CLOCK 0x10
+#define RESET_TOGGLE 0x40
+
+#define ASSP_CONTROL_C 0xA6
+#define ASSP_HOST_INT_ENABLE 0x01
+#define FM_ADDR_REMAP_DISABLE 0x02
+#define HOST_WRITE_PORT_ENABLE 0x08
+
+#define ASSP_HOST_INT_STATUS 0xAC
+#define DSP2HOST_REQ_PIORECORD 0x01
+#define DSP2HOST_REQ_I2SRATE 0x02
+#define DSP2HOST_REQ_TIMER 0x04
+
+/* AC97 registers */
+/* XXX fix this crap up */
+/*#define AC97_RESET 0x00*/
+
+#define AC97_VOL_MUTE_B 0x8000
+#define AC97_VOL_M 0x1F
+#define AC97_LEFT_VOL_S 8
+
+#define AC97_MASTER_VOL 0x02
+#define AC97_LINE_LEVEL_VOL 0x04
+#define AC97_MASTER_MONO_VOL 0x06
+#define AC97_PC_BEEP_VOL 0x0A
+#define AC97_PC_BEEP_VOL_M 0x0F
+#define AC97_SROUND_MASTER_VOL 0x38
+#define AC97_PC_BEEP_VOL_S 1
+
+/*#define AC97_PHONE_VOL 0x0C
+#define AC97_MIC_VOL 0x0E*/
+#define AC97_MIC_20DB_ENABLE 0x40
+
+/*#define AC97_LINEIN_VOL 0x10
+#define AC97_CD_VOL 0x12
+#define AC97_VIDEO_VOL 0x14
+#define AC97_AUX_VOL 0x16*/
+#define AC97_PCM_OUT_VOL 0x18
+/*#define AC97_RECORD_SELECT 0x1A*/
+#define AC97_RECORD_MIC 0x00
+#define AC97_RECORD_CD 0x01
+#define AC97_RECORD_VIDEO 0x02
+#define AC97_RECORD_AUX 0x03
+#define AC97_RECORD_MONO_MUX 0x02
+#define AC97_RECORD_DIGITAL 0x03
+#define AC97_RECORD_LINE 0x04
+#define AC97_RECORD_STEREO 0x05
+#define AC97_RECORD_MONO 0x06
+#define AC97_RECORD_PHONE 0x07
+
+/*#define AC97_RECORD_GAIN 0x1C*/
+#define AC97_RECORD_VOL_M 0x0F
+
+/*#define AC97_GENERAL_PURPOSE 0x20*/
+#define AC97_POWER_DOWN_CTRL 0x26
+#define AC97_ADC_READY 0x0001
+#define AC97_DAC_READY 0x0002
+#define AC97_ANALOG_READY 0x0004
+#define AC97_VREF_ON 0x0008
+#define AC97_PR0 0x0100
+#define AC97_PR1 0x0200
+#define AC97_PR2 0x0400
+#define AC97_PR3 0x0800
+#define AC97_PR4 0x1000
+
+#define AC97_RESERVED1 0x28
+
+#define AC97_VENDOR_TEST 0x5A
+
+#define AC97_CLOCK_DELAY 0x5C
+#define AC97_LINEOUT_MUX_SEL 0x0001
+#define AC97_MONO_MUX_SEL 0x0002
+#define AC97_CLOCK_DELAY_SEL 0x1F
+#define AC97_DAC_CDS_SHIFT 6
+#define AC97_ADC_CDS_SHIFT 11
+
+#define AC97_MULTI_CHANNEL_SEL 0x74
+
+/*#define AC97_VENDOR_ID1 0x7C
+#define AC97_VENDOR_ID2 0x7E*/
+
+/*
+ * ASSP control regs
+ */
+#define DSP_PORT_TIMER_COUNT 0x06
+
+#define DSP_PORT_MEMORY_INDEX 0x80
+
+#define DSP_PORT_MEMORY_TYPE 0x82
+#define MEMTYPE_INTERNAL_CODE 0x0002
+#define MEMTYPE_INTERNAL_DATA 0x0003
+#define MEMTYPE_MASK 0x0003
+
+#define DSP_PORT_MEMORY_DATA 0x84
+
+#define DSP_PORT_CONTROL_REG_A 0xA2
+#define DSP_PORT_CONTROL_REG_B 0xA4
+#define DSP_PORT_CONTROL_REG_C 0xA6
+
+#define REV_A_CODE_MEMORY_BEGIN 0x0000
+#define REV_A_CODE_MEMORY_END 0x0FFF
+#define REV_A_CODE_MEMORY_UNIT_LENGTH 0x0040
+#define REV_A_CODE_MEMORY_LENGTH (REV_A_CODE_MEMORY_END - REV_A_CODE_MEMORY_BEGIN + 1)
+
+#define REV_B_CODE_MEMORY_BEGIN 0x0000
+#define REV_B_CODE_MEMORY_END 0x0BFF
+#define REV_B_CODE_MEMORY_UNIT_LENGTH 0x0040
+#define REV_B_CODE_MEMORY_LENGTH (REV_B_CODE_MEMORY_END - REV_B_CODE_MEMORY_BEGIN + 1)
+
+#define REV_A_DATA_MEMORY_BEGIN 0x1000
+#define REV_A_DATA_MEMORY_END 0x2FFF
+#define REV_A_DATA_MEMORY_UNIT_LENGTH 0x0080
+#define REV_A_DATA_MEMORY_LENGTH (REV_A_DATA_MEMORY_END - REV_A_DATA_MEMORY_BEGIN + 1)
+
+#define REV_B_DATA_MEMORY_BEGIN 0x1000
+#define REV_B_DATA_MEMORY_END 0x2BFF
+#define REV_B_DATA_MEMORY_UNIT_LENGTH 0x0080
+#define REV_B_DATA_MEMORY_LENGTH (REV_B_DATA_MEMORY_END - REV_B_DATA_MEMORY_BEGIN + 1)
+
+
+#define NUM_UNITS_KERNEL_CODE 16
+#define NUM_UNITS_KERNEL_DATA 2
+
+#define NUM_UNITS_KERNEL_CODE_WITH_HSP 16
+#define NUM_UNITS_KERNEL_DATA_WITH_HSP 5
+
+/*
+ * Kernel data layout
+ */
+
+#define DP_SHIFT_COUNT 7
+
+#define KDATA_BASE_ADDR 0x1000
+#define KDATA_BASE_ADDR2 0x1080
+
+#define KDATA_TASK0 (KDATA_BASE_ADDR + 0x0000)
+#define KDATA_TASK1 (KDATA_BASE_ADDR + 0x0001)
+#define KDATA_TASK2 (KDATA_BASE_ADDR + 0x0002)
+#define KDATA_TASK3 (KDATA_BASE_ADDR + 0x0003)
+#define KDATA_TASK4 (KDATA_BASE_ADDR + 0x0004)
+#define KDATA_TASK5 (KDATA_BASE_ADDR + 0x0005)
+#define KDATA_TASK6 (KDATA_BASE_ADDR + 0x0006)
+#define KDATA_TASK7 (KDATA_BASE_ADDR + 0x0007)
+#define KDATA_TASK_ENDMARK (KDATA_BASE_ADDR + 0x0008)
+
+#define KDATA_CURRENT_TASK (KDATA_BASE_ADDR + 0x0009)
+#define KDATA_TASK_SWITCH (KDATA_BASE_ADDR + 0x000A)
+
+#define KDATA_INSTANCE0_POS3D (KDATA_BASE_ADDR + 0x000B)
+#define KDATA_INSTANCE1_POS3D (KDATA_BASE_ADDR + 0x000C)
+#define KDATA_INSTANCE2_POS3D (KDATA_BASE_ADDR + 0x000D)
+#define KDATA_INSTANCE3_POS3D (KDATA_BASE_ADDR + 0x000E)
+#define KDATA_INSTANCE4_POS3D (KDATA_BASE_ADDR + 0x000F)
+#define KDATA_INSTANCE5_POS3D (KDATA_BASE_ADDR + 0x0010)
+#define KDATA_INSTANCE6_POS3D (KDATA_BASE_ADDR + 0x0011)
+#define KDATA_INSTANCE7_POS3D (KDATA_BASE_ADDR + 0x0012)
+#define KDATA_INSTANCE8_POS3D (KDATA_BASE_ADDR + 0x0013)
+#define KDATA_INSTANCE_POS3D_ENDMARK (KDATA_BASE_ADDR + 0x0014)
+
+#define KDATA_INSTANCE0_SPKVIRT (KDATA_BASE_ADDR + 0x0015)
+#define KDATA_INSTANCE_SPKVIRT_ENDMARK (KDATA_BASE_ADDR + 0x0016)
+
+#define KDATA_INSTANCE0_SPDIF (KDATA_BASE_ADDR + 0x0017)
+#define KDATA_INSTANCE_SPDIF_ENDMARK (KDATA_BASE_ADDR + 0x0018)
+
+#define KDATA_INSTANCE0_MODEM (KDATA_BASE_ADDR + 0x0019)
+#define KDATA_INSTANCE_MODEM_ENDMARK (KDATA_BASE_ADDR + 0x001A)
+
+#define KDATA_INSTANCE0_SRC (KDATA_BASE_ADDR + 0x001B)
+#define KDATA_INSTANCE1_SRC (KDATA_BASE_ADDR + 0x001C)
+#define KDATA_INSTANCE_SRC_ENDMARK (KDATA_BASE_ADDR + 0x001D)
+
+#define KDATA_INSTANCE0_MINISRC (KDATA_BASE_ADDR + 0x001E)
+#define KDATA_INSTANCE1_MINISRC (KDATA_BASE_ADDR + 0x001F)
+#define KDATA_INSTANCE2_MINISRC (KDATA_BASE_ADDR + 0x0020)
+#define KDATA_INSTANCE3_MINISRC (KDATA_BASE_ADDR + 0x0021)
+#define KDATA_INSTANCE_MINISRC_ENDMARK (KDATA_BASE_ADDR + 0x0022)
+
+#define KDATA_INSTANCE0_CPYTHRU (KDATA_BASE_ADDR + 0x0023)
+#define KDATA_INSTANCE1_CPYTHRU (KDATA_BASE_ADDR + 0x0024)
+#define KDATA_INSTANCE_CPYTHRU_ENDMARK (KDATA_BASE_ADDR + 0x0025)
+
+#define KDATA_CURRENT_DMA (KDATA_BASE_ADDR + 0x0026)
+#define KDATA_DMA_SWITCH (KDATA_BASE_ADDR + 0x0027)
+#define KDATA_DMA_ACTIVE (KDATA_BASE_ADDR + 0x0028)
+
+#define KDATA_DMA_XFER0 (KDATA_BASE_ADDR + 0x0029)
+#define KDATA_DMA_XFER1 (KDATA_BASE_ADDR + 0x002A)
+#define KDATA_DMA_XFER2 (KDATA_BASE_ADDR + 0x002B)
+#define KDATA_DMA_XFER3 (KDATA_BASE_ADDR + 0x002C)
+#define KDATA_DMA_XFER4 (KDATA_BASE_ADDR + 0x002D)
+#define KDATA_DMA_XFER5 (KDATA_BASE_ADDR + 0x002E)
+#define KDATA_DMA_XFER6 (KDATA_BASE_ADDR + 0x002F)
+#define KDATA_DMA_XFER7 (KDATA_BASE_ADDR + 0x0030)
+#define KDATA_DMA_XFER8 (KDATA_BASE_ADDR + 0x0031)
+#define KDATA_DMA_XFER_ENDMARK (KDATA_BASE_ADDR + 0x0032)
+
+#define KDATA_I2S_SAMPLE_COUNT (KDATA_BASE_ADDR + 0x0033)
+#define KDATA_I2S_INT_METER (KDATA_BASE_ADDR + 0x0034)
+#define KDATA_I2S_ACTIVE (KDATA_BASE_ADDR + 0x0035)
+
+#define KDATA_TIMER_COUNT_RELOAD (KDATA_BASE_ADDR + 0x0036)
+#define KDATA_TIMER_COUNT_CURRENT (KDATA_BASE_ADDR + 0x0037)
+
+#define KDATA_HALT_SYNCH_CLIENT (KDATA_BASE_ADDR + 0x0038)
+#define KDATA_HALT_SYNCH_DMA (KDATA_BASE_ADDR + 0x0039)
+#define KDATA_HALT_ACKNOWLEDGE (KDATA_BASE_ADDR + 0x003A)
+
+#define KDATA_ADC1_XFER0 (KDATA_BASE_ADDR + 0x003B)
+#define KDATA_ADC1_XFER_ENDMARK (KDATA_BASE_ADDR + 0x003C)
+#define KDATA_ADC1_LEFT_VOLUME (KDATA_BASE_ADDR + 0x003D)
+#define KDATA_ADC1_RIGHT_VOLUME (KDATA_BASE_ADDR + 0x003E)
+#define KDATA_ADC1_LEFT_SUR_VOL (KDATA_BASE_ADDR + 0x003F)
+#define KDATA_ADC1_RIGHT_SUR_VOL (KDATA_BASE_ADDR + 0x0040)
+
+#define KDATA_ADC2_XFER0 (KDATA_BASE_ADDR + 0x0041)
+#define KDATA_ADC2_XFER_ENDMARK (KDATA_BASE_ADDR + 0x0042)
+#define KDATA_ADC2_LEFT_VOLUME (KDATA_BASE_ADDR + 0x0043)
+#define KDATA_ADC2_RIGHT_VOLUME (KDATA_BASE_ADDR + 0x0044)
+#define KDATA_ADC2_LEFT_SUR_VOL (KDATA_BASE_ADDR + 0x0045)
+#define KDATA_ADC2_RIGHT_SUR_VOL (KDATA_BASE_ADDR + 0x0046)
+
+#define KDATA_CD_XFER0 (KDATA_BASE_ADDR + 0x0047)
+#define KDATA_CD_XFER_ENDMARK (KDATA_BASE_ADDR + 0x0048)
+#define KDATA_CD_LEFT_VOLUME (KDATA_BASE_ADDR + 0x0049)
+#define KDATA_CD_RIGHT_VOLUME (KDATA_BASE_ADDR + 0x004A)
+#define KDATA_CD_LEFT_SUR_VOL (KDATA_BASE_ADDR + 0x004B)
+#define KDATA_CD_RIGHT_SUR_VOL (KDATA_BASE_ADDR + 0x004C)
+
+#define KDATA_MIC_XFER0 (KDATA_BASE_ADDR + 0x004D)
+#define KDATA_MIC_XFER_ENDMARK (KDATA_BASE_ADDR + 0x004E)
+#define KDATA_MIC_VOLUME (KDATA_BASE_ADDR + 0x004F)
+#define KDATA_MIC_SUR_VOL (KDATA_BASE_ADDR + 0x0050)
+
+#define KDATA_I2S_XFER0 (KDATA_BASE_ADDR + 0x0051)
+#define KDATA_I2S_XFER_ENDMARK (KDATA_BASE_ADDR + 0x0052)
+
+#define KDATA_CHI_XFER0 (KDATA_BASE_ADDR + 0x0053)
+#define KDATA_CHI_XFER_ENDMARK (KDATA_BASE_ADDR + 0x0054)
+
+#define KDATA_SPDIF_XFER (KDATA_BASE_ADDR + 0x0055)
+#define KDATA_SPDIF_CURRENT_FRAME (KDATA_BASE_ADDR + 0x0056)
+#define KDATA_SPDIF_FRAME0 (KDATA_BASE_ADDR + 0x0057)
+#define KDATA_SPDIF_FRAME1 (KDATA_BASE_ADDR + 0x0058)
+#define KDATA_SPDIF_FRAME2 (KDATA_BASE_ADDR + 0x0059)
+
+#define KDATA_SPDIF_REQUEST (KDATA_BASE_ADDR + 0x005A)
+#define KDATA_SPDIF_TEMP (KDATA_BASE_ADDR + 0x005B)
+
+#define KDATA_SPDIFIN_XFER0 (KDATA_BASE_ADDR + 0x005C)
+#define KDATA_SPDIFIN_XFER_ENDMARK (KDATA_BASE_ADDR + 0x005D)
+#define KDATA_SPDIFIN_INT_METER (KDATA_BASE_ADDR + 0x005E)
+
+#define KDATA_DSP_RESET_COUNT (KDATA_BASE_ADDR + 0x005F)
+#define KDATA_DEBUG_OUTPUT (KDATA_BASE_ADDR + 0x0060)
+
+#define KDATA_KERNEL_ISR_LIST (KDATA_BASE_ADDR + 0x0061)
+
+#define KDATA_KERNEL_ISR_CBSR1 (KDATA_BASE_ADDR + 0x0062)
+#define KDATA_KERNEL_ISR_CBER1 (KDATA_BASE_ADDR + 0x0063)
+#define KDATA_KERNEL_ISR_CBCR (KDATA_BASE_ADDR + 0x0064)
+#define KDATA_KERNEL_ISR_AR0 (KDATA_BASE_ADDR + 0x0065)
+#define KDATA_KERNEL_ISR_AR1 (KDATA_BASE_ADDR + 0x0066)
+#define KDATA_KERNEL_ISR_AR2 (KDATA_BASE_ADDR + 0x0067)
+#define KDATA_KERNEL_ISR_AR3 (KDATA_BASE_ADDR + 0x0068)
+#define KDATA_KERNEL_ISR_AR4 (KDATA_BASE_ADDR + 0x0069)
+#define KDATA_KERNEL_ISR_AR5 (KDATA_BASE_ADDR + 0x006A)
+#define KDATA_KERNEL_ISR_BRCR (KDATA_BASE_ADDR + 0x006B)
+#define KDATA_KERNEL_ISR_PASR (KDATA_BASE_ADDR + 0x006C)
+#define KDATA_KERNEL_ISR_PAER (KDATA_BASE_ADDR + 0x006D)
+
+#define KDATA_CLIENT_SCRATCH0 (KDATA_BASE_ADDR + 0x006E)
+#define KDATA_CLIENT_SCRATCH1 (KDATA_BASE_ADDR + 0x006F)
+#define KDATA_KERNEL_SCRATCH (KDATA_BASE_ADDR + 0x0070)
+#define KDATA_KERNEL_ISR_SCRATCH (KDATA_BASE_ADDR + 0x0071)
+
+#define KDATA_OUEUE_LEFT (KDATA_BASE_ADDR + 0x0072)
+#define KDATA_QUEUE_RIGHT (KDATA_BASE_ADDR + 0x0073)
+
+#define KDATA_ADC1_REQUEST (KDATA_BASE_ADDR + 0x0074)
+#define KDATA_ADC2_REQUEST (KDATA_BASE_ADDR + 0x0075)
+#define KDATA_CD_REQUEST (KDATA_BASE_ADDR + 0x0076)
+#define KDATA_MIC_REQUEST (KDATA_BASE_ADDR + 0x0077)
+
+#define KDATA_ADC1_MIXER_REQUEST (KDATA_BASE_ADDR + 0x0078)
+#define KDATA_ADC2_MIXER_REQUEST (KDATA_BASE_ADDR + 0x0079)
+#define KDATA_CD_MIXER_REQUEST (KDATA_BASE_ADDR + 0x007A)
+#define KDATA_MIC_MIXER_REQUEST (KDATA_BASE_ADDR + 0x007B)
+#define KDATA_MIC_SYNC_COUNTER (KDATA_BASE_ADDR + 0x007C)
+
+/*
+ * second 'segment' (?) reserved for mixer
+ * buffers..
+ */
+
+#define KDATA_MIXER_WORD0 (KDATA_BASE_ADDR2 + 0x0000)
+#define KDATA_MIXER_WORD1 (KDATA_BASE_ADDR2 + 0x0001)
+#define KDATA_MIXER_WORD2 (KDATA_BASE_ADDR2 + 0x0002)
+#define KDATA_MIXER_WORD3 (KDATA_BASE_ADDR2 + 0x0003)
+#define KDATA_MIXER_WORD4 (KDATA_BASE_ADDR2 + 0x0004)
+#define KDATA_MIXER_WORD5 (KDATA_BASE_ADDR2 + 0x0005)
+#define KDATA_MIXER_WORD6 (KDATA_BASE_ADDR2 + 0x0006)
+#define KDATA_MIXER_WORD7 (KDATA_BASE_ADDR2 + 0x0007)
+#define KDATA_MIXER_WORD8 (KDATA_BASE_ADDR2 + 0x0008)
+#define KDATA_MIXER_WORD9 (KDATA_BASE_ADDR2 + 0x0009)
+#define KDATA_MIXER_WORDA (KDATA_BASE_ADDR2 + 0x000A)
+#define KDATA_MIXER_WORDB (KDATA_BASE_ADDR2 + 0x000B)
+#define KDATA_MIXER_WORDC (KDATA_BASE_ADDR2 + 0x000C)
+#define KDATA_MIXER_WORDD (KDATA_BASE_ADDR2 + 0x000D)
+#define KDATA_MIXER_WORDE (KDATA_BASE_ADDR2 + 0x000E)
+#define KDATA_MIXER_WORDF (KDATA_BASE_ADDR2 + 0x000F)
+
+#define KDATA_MIXER_XFER0 (KDATA_BASE_ADDR2 + 0x0010)
+#define KDATA_MIXER_XFER1 (KDATA_BASE_ADDR2 + 0x0011)
+#define KDATA_MIXER_XFER2 (KDATA_BASE_ADDR2 + 0x0012)
+#define KDATA_MIXER_XFER3 (KDATA_BASE_ADDR2 + 0x0013)
+#define KDATA_MIXER_XFER4 (KDATA_BASE_ADDR2 + 0x0014)
+#define KDATA_MIXER_XFER5 (KDATA_BASE_ADDR2 + 0x0015)
+#define KDATA_MIXER_XFER6 (KDATA_BASE_ADDR2 + 0x0016)
+#define KDATA_MIXER_XFER7 (KDATA_BASE_ADDR2 + 0x0017)
+#define KDATA_MIXER_XFER8 (KDATA_BASE_ADDR2 + 0x0018)
+#define KDATA_MIXER_XFER9 (KDATA_BASE_ADDR2 + 0x0019)
+#define KDATA_MIXER_XFER_ENDMARK (KDATA_BASE_ADDR2 + 0x001A)
+
+#define KDATA_MIXER_TASK_NUMBER (KDATA_BASE_ADDR2 + 0x001B)
+#define KDATA_CURRENT_MIXER (KDATA_BASE_ADDR2 + 0x001C)
+#define KDATA_MIXER_ACTIVE (KDATA_BASE_ADDR2 + 0x001D)
+#define KDATA_MIXER_BANK_STATUS (KDATA_BASE_ADDR2 + 0x001E)
+#define KDATA_DAC_LEFT_VOLUME (KDATA_BASE_ADDR2 + 0x001F)
+#define KDATA_DAC_RIGHT_VOLUME (KDATA_BASE_ADDR2 + 0x0020)
+
+#define MAX_INSTANCE_MINISRC (KDATA_INSTANCE_MINISRC_ENDMARK - KDATA_INSTANCE0_MINISRC)
+#define MAX_VIRTUAL_DMA_CHANNELS (KDATA_DMA_XFER_ENDMARK - KDATA_DMA_XFER0)
+#define MAX_VIRTUAL_MIXER_CHANNELS (KDATA_MIXER_XFER_ENDMARK - KDATA_MIXER_XFER0)
+#define MAX_VIRTUAL_ADC1_CHANNELS (KDATA_ADC1_XFER_ENDMARK - KDATA_ADC1_XFER0)
+
+/*
+ * client data area offsets
+ */
+#define CDATA_INSTANCE_READY 0x00
+
+#define CDATA_HOST_SRC_ADDRL 0x01
+#define CDATA_HOST_SRC_ADDRH 0x02
+#define CDATA_HOST_SRC_END_PLUS_1L 0x03
+#define CDATA_HOST_SRC_END_PLUS_1H 0x04
+#define CDATA_HOST_SRC_CURRENTL 0x05
+#define CDATA_HOST_SRC_CURRENTH 0x06
+
+#define CDATA_IN_BUF_CONNECT 0x07
+#define CDATA_OUT_BUF_CONNECT 0x08
+
+#define CDATA_IN_BUF_BEGIN 0x09
+#define CDATA_IN_BUF_END_PLUS_1 0x0A
+#define CDATA_IN_BUF_HEAD 0x0B
+#define CDATA_IN_BUF_TAIL 0x0C
+#define CDATA_OUT_BUF_BEGIN 0x0D
+#define CDATA_OUT_BUF_END_PLUS_1 0x0E
+#define CDATA_OUT_BUF_HEAD 0x0F
+#define CDATA_OUT_BUF_TAIL 0x10
+
+#define CDATA_DMA_CONTROL 0x11
+#define CDATA_RESERVED 0x12
+
+#define CDATA_FREQUENCY 0x13
+#define CDATA_LEFT_VOLUME 0x14
+#define CDATA_RIGHT_VOLUME 0x15
+#define CDATA_LEFT_SUR_VOL 0x16
+#define CDATA_RIGHT_SUR_VOL 0x17
+
+#define CDATA_HEADER_LEN 0x18
+
+#define SRC3_DIRECTION_OFFSET CDATA_HEADER_LEN
+#define SRC3_MODE_OFFSET (CDATA_HEADER_LEN + 1)
+#define SRC3_WORD_LENGTH_OFFSET (CDATA_HEADER_LEN + 2)
+#define SRC3_PARAMETER_OFFSET (CDATA_HEADER_LEN + 3)
+#define SRC3_COEFF_ADDR_OFFSET (CDATA_HEADER_LEN + 8)
+#define SRC3_FILTAP_ADDR_OFFSET (CDATA_HEADER_LEN + 10)
+#define SRC3_TEMP_INBUF_ADDR_OFFSET (CDATA_HEADER_LEN + 16)
+#define SRC3_TEMP_OUTBUF_ADDR_OFFSET (CDATA_HEADER_LEN + 17)
+
+#define MINISRC_IN_BUFFER_SIZE ( 0x50 * 2 )
+#define MINISRC_OUT_BUFFER_SIZE ( 0x50 * 2 * 2)
+#define MINISRC_TMP_BUFFER_SIZE ( 112 + ( MINISRC_BIQUAD_STAGE * 3 + 4 ) * 2 * 2 )
+#define MINISRC_BIQUAD_STAGE 2
+#define MINISRC_COEF_LOC 0x175
+
+#define DMACONTROL_BLOCK_MASK 0x000F
+#define DMAC_BLOCK0_SELECTOR 0x0000
+#define DMAC_BLOCK1_SELECTOR 0x0001
+#define DMAC_BLOCK2_SELECTOR 0x0002
+#define DMAC_BLOCK3_SELECTOR 0x0003
+#define DMAC_BLOCK4_SELECTOR 0x0004
+#define DMAC_BLOCK5_SELECTOR 0x0005
+#define DMAC_BLOCK6_SELECTOR 0x0006
+#define DMAC_BLOCK7_SELECTOR 0x0007
+#define DMAC_BLOCK8_SELECTOR 0x0008
+#define DMAC_BLOCK9_SELECTOR 0x0009
+#define DMAC_BLOCKA_SELECTOR 0x000A
+#define DMAC_BLOCKB_SELECTOR 0x000B
+#define DMAC_BLOCKC_SELECTOR 0x000C
+#define DMAC_BLOCKD_SELECTOR 0x000D
+#define DMAC_BLOCKE_SELECTOR 0x000E
+#define DMAC_BLOCKF_SELECTOR 0x000F
+#define DMACONTROL_PAGE_MASK 0x00F0
+#define DMAC_PAGE0_SELECTOR 0x0030
+#define DMAC_PAGE1_SELECTOR 0x0020
+#define DMAC_PAGE2_SELECTOR 0x0010
+#define DMAC_PAGE3_SELECTOR 0x0000
+#define DMACONTROL_AUTOREPEAT 0x1000
+#define DMACONTROL_STOPPED 0x2000
+#define DMACONTROL_DIRECTION 0x0100
+
+/*
+ * an arbitrary volume we set the internal
+ * volume settings to so that the ac97 volume
+ * range is a little less insane. 0x7fff is
+ * max.
+ */
+#define ARB_VOLUME ( 0x6800 )
+
+/*
+ */
+
+struct m3_list {
+ int curlen;
+ int mem_addr;
+ int max;
+};
+
+struct m3_dma {
+
+ int number;
+ struct snd_pcm_substream *substream;
+
+ struct assp_instance {
+ unsigned short code, data;
+ } inst;
+
+ int running;
+ int opened;
+
+ unsigned long buffer_addr;
+ int dma_size;
+ int period_size;
+ unsigned int hwptr;
+ int count;
+
+ int index[3];
+ struct m3_list *index_list[3];
+
+ int in_lists;
+
+ struct list_head list;
+
+};
+
+struct snd_m3 {
+
+ struct snd_card *card;
+
+ unsigned long iobase;
+
+ int irq;
+ unsigned int allegro_flag : 1;
+
+ struct snd_ac97 *ac97;
+
+ struct snd_pcm *pcm;
+
+ struct pci_dev *pci;
+
+ int dacs_active;
+ int timer_users;
+
+ struct m3_list msrc_list;
+ struct m3_list mixer_list;
+ struct m3_list adc1_list;
+ struct m3_list dma_list;
+
+ /* for storing reset state..*/
+ u8 reset_state;
+
+ int external_amp;
+ int amp_gpio; /* gpio pin # for external amp, -1 = default */
+ unsigned int hv_config; /* hardware-volume config bits */
+ unsigned irda_workaround :1; /* avoid to touch 0x10 on GPIO_DIRECTION
+ (e.g. for IrDA on Dell Inspirons) */
+ unsigned is_omnibook :1; /* Do HP OmniBook GPIO magic? */
+
+ /* midi */
+ struct snd_rawmidi *rmidi;
+
+ /* pcm streams */
+ int num_substreams;
+ struct m3_dma *substreams;
+
+ spinlock_t reg_lock;
+ spinlock_t ac97_lock;
+
+ struct snd_kcontrol *master_switch;
+ struct snd_kcontrol *master_volume;
+ struct tasklet_struct hwvol_tq;
+
+#ifdef CONFIG_PM
+ u16 *suspend_mem;
+#endif
+
+ const struct firmware *assp_kernel_image;
+ const struct firmware *assp_minisrc_image;
+};
+
+/*
+ * pci ids
+ */
+static struct pci_device_id snd_m3_ids[] = {
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_CANYON3D_2LE, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_CANYON3D_2, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_MAESTRO3, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_MAESTRO3_1, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_MAESTRO3_HW, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_MAESTRO3_2, PCI_ANY_ID, PCI_ANY_ID,
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {0,},
+};
+
+MODULE_DEVICE_TABLE(pci, snd_m3_ids);
+
+static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d),
+ SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d),
+ SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03),
+ SND_PCI_QUIRK(0x1509, 0x1740, "LEGEND ZhaoYang 3100CF", 0x03),
+ { } /* END */
+};
+
+static struct snd_pci_quirk m3_irda_quirk_list[] __devinitdata = {
+ SND_PCI_QUIRK(0x1028, 0x00b0, "Dell Inspiron 4000", 1),
+ SND_PCI_QUIRK(0x1028, 0x00a4, "Dell Inspiron 8000", 1),
+ SND_PCI_QUIRK(0x1028, 0x00e6, "Dell Inspiron 8100", 1),
+ { } /* END */
+};
+
+/* hardware volume quirks */
+static struct snd_pci_quirk m3_hv_quirk_list[] __devinitdata = {
+ /* Allegro chips */
+ SND_PCI_QUIRK(0x0E11, 0x002E, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x0E11, 0x0094, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x0E11, 0xB112, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x0E11, 0xB114, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x103C, 0x0012, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x103C, 0x0018, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x103C, 0x001C, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x103C, 0x001D, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x103C, 0x001E, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x107B, 0x3350, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x10F7, 0x8338, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x10F7, 0x833C, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x10F7, 0x833D, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x10F7, 0x833E, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x10F7, 0x833F, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x13BD, 0x1018, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x13BD, 0x1019, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x13BD, 0x101A, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x14FF, 0x0F03, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x14FF, 0x0F04, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x14FF, 0x0F05, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x156D, 0xB400, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x156D, 0xB795, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x156D, 0xB797, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x156D, 0xC700, NULL, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD),
+ SND_PCI_QUIRK(0x1033, 0x80F1, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x103C, 0x001A, NULL, /* HP OmniBook 6100 */
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x107B, 0x340A, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x107B, 0x3450, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x109F, 0x3134, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x109F, 0x3161, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x144D, 0x3280, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x144D, 0x3281, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x144D, 0xC002, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x144D, 0xC003, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x1509, 0x1740, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x1610, 0x0010, NULL,
+ HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x1042, 0x1042, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x107B, 0x9500, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x14FF, 0x0F06, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x1558, 0x8586, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x161F, 0x2011, NULL, HV_CTRL_ENABLE),
+ /* Maestro3 chips */
+ SND_PCI_QUIRK(0x103C, 0x000E, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x103C, 0x0010, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x103C, 0x0011, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x103C, 0x001B, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x104D, 0x80A6, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x104D, 0x80AA, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x107B, 0x5300, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x110A, 0x1998, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x13BD, 0x1015, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x13BD, 0x101C, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x13BD, 0x1802, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x1599, 0x0715, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x5643, 0x5643, NULL, HV_CTRL_ENABLE),
+ SND_PCI_QUIRK(0x144D, 0x3260, NULL, HV_CTRL_ENABLE | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x144D, 0x3261, NULL, HV_CTRL_ENABLE | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x144D, 0xC000, NULL, HV_CTRL_ENABLE | REDUCED_DEBOUNCE),
+ SND_PCI_QUIRK(0x144D, 0xC001, NULL, HV_CTRL_ENABLE | REDUCED_DEBOUNCE),
+ { } /* END */
+};
+
+/* HP Omnibook quirks */
+static struct snd_pci_quirk m3_omnibook_quirk_list[] __devinitdata = {
+ SND_PCI_QUIRK_ID(0x103c, 0x0010), /* HP OmniBook 6000 */
+ SND_PCI_QUIRK_ID(0x103c, 0x0011), /* HP OmniBook 500 */
+ { } /* END */
+};
+
+/*
+ * lowlevel functions
+ */
+
+static inline void snd_m3_outw(struct snd_m3 *chip, u16 value, unsigned long reg)
+{
+ outw(value, chip->iobase + reg);
+}
+
+static inline u16 snd_m3_inw(struct snd_m3 *chip, unsigned long reg)
+{
+ return inw(chip->iobase + reg);
+}
+
+static inline void snd_m3_outb(struct snd_m3 *chip, u8 value, unsigned long reg)
+{
+ outb(value, chip->iobase + reg);
+}
+
+static inline u8 snd_m3_inb(struct snd_m3 *chip, unsigned long reg)
+{
+ return inb(chip->iobase + reg);
+}
+
+/*
+ * access 16bit words to the code or data regions of the dsp's memory.
+ * index addresses 16bit words.
+ */
+static u16 snd_m3_assp_read(struct snd_m3 *chip, u16 region, u16 index)
+{
+ snd_m3_outw(chip, region & MEMTYPE_MASK, DSP_PORT_MEMORY_TYPE);
+ snd_m3_outw(chip, index, DSP_PORT_MEMORY_INDEX);
+ return snd_m3_inw(chip, DSP_PORT_MEMORY_DATA);
+}
+
+static void snd_m3_assp_write(struct snd_m3 *chip, u16 region, u16 index, u16 data)
+{
+ snd_m3_outw(chip, region & MEMTYPE_MASK, DSP_PORT_MEMORY_TYPE);
+ snd_m3_outw(chip, index, DSP_PORT_MEMORY_INDEX);
+ snd_m3_outw(chip, data, DSP_PORT_MEMORY_DATA);
+}
+
+static void snd_m3_assp_halt(struct snd_m3 *chip)
+{
+ chip->reset_state = snd_m3_inb(chip, DSP_PORT_CONTROL_REG_B) & ~REGB_STOP_CLOCK;
+ msleep(10);
+ snd_m3_outb(chip, chip->reset_state & ~REGB_ENABLE_RESET, DSP_PORT_CONTROL_REG_B);
+}
+
+static void snd_m3_assp_continue(struct snd_m3 *chip)
+{
+ snd_m3_outb(chip, chip->reset_state | REGB_ENABLE_RESET, DSP_PORT_CONTROL_REG_B);
+}
+
+
+/*
+ * This makes me sad. the maestro3 has lists
+ * internally that must be packed.. 0 terminates,
+ * apparently, or maybe all unused entries have
+ * to be 0, the lists have static lengths set
+ * by the binary code images.
+ */
+
+static int snd_m3_add_list(struct snd_m3 *chip, struct m3_list *list, u16 val)
+{
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ list->mem_addr + list->curlen,
+ val);
+ return list->curlen++;
+}
+
+static void snd_m3_remove_list(struct snd_m3 *chip, struct m3_list *list, int index)
+{
+ u16 val;
+ int lastindex = list->curlen - 1;
+
+ if (index != lastindex) {
+ val = snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA,
+ list->mem_addr + lastindex);
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ list->mem_addr + index,
+ val);
+ }
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ list->mem_addr + lastindex,
+ 0);
+
+ list->curlen--;
+}
+
+static void snd_m3_inc_timer_users(struct snd_m3 *chip)
+{
+ chip->timer_users++;
+ if (chip->timer_users != 1)
+ return;
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_TIMER_COUNT_RELOAD,
+ 240);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_TIMER_COUNT_CURRENT,
+ 240);
+
+ snd_m3_outw(chip,
+ snd_m3_inw(chip, HOST_INT_CTRL) | CLKRUN_GEN_ENABLE,
+ HOST_INT_CTRL);
+}
+
+static void snd_m3_dec_timer_users(struct snd_m3 *chip)
+{
+ chip->timer_users--;
+ if (chip->timer_users > 0)
+ return;
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_TIMER_COUNT_RELOAD,
+ 0);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_TIMER_COUNT_CURRENT,
+ 0);
+
+ snd_m3_outw(chip,
+ snd_m3_inw(chip, HOST_INT_CTRL) & ~CLKRUN_GEN_ENABLE,
+ HOST_INT_CTRL);
+}
+
+/*
+ * start/stop
+ */
+
+/* spinlock held! */
+static int snd_m3_pcm_start(struct snd_m3 *chip, struct m3_dma *s,
+ struct snd_pcm_substream *subs)
+{
+ if (! s || ! subs)
+ return -EINVAL;
+
+ snd_m3_inc_timer_users(chip);
+ switch (subs->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ chip->dacs_active++;
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_INSTANCE_READY, 1);
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_MIXER_TASK_NUMBER,
+ chip->dacs_active);
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_ADC1_REQUEST, 1);
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_INSTANCE_READY, 1);
+ break;
+ }
+ return 0;
+}
+
+/* spinlock held! */
+static int snd_m3_pcm_stop(struct snd_m3 *chip, struct m3_dma *s,
+ struct snd_pcm_substream *subs)
+{
+ if (! s || ! subs)
+ return -EINVAL;
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_INSTANCE_READY, 0);
+ snd_m3_dec_timer_users(chip);
+ switch (subs->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ chip->dacs_active--;
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_MIXER_TASK_NUMBER,
+ chip->dacs_active);
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_ADC1_REQUEST, 0);
+ break;
+ }
+ return 0;
+}
+
+static int
+snd_m3_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
+{
+ struct snd_m3 *chip = snd_pcm_substream_chip(subs);
+ struct m3_dma *s = subs->runtime->private_data;
+ int err = -EINVAL;
+
+ if (snd_BUG_ON(!s))
+ return -ENXIO;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (s->running)
+ err = -EBUSY;
+ else {
+ s->running = 1;
+ err = snd_m3_pcm_start(chip, s, subs);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (! s->running)
+ err = 0; /* should return error? */
+ else {
+ s->running = 0;
+ err = snd_m3_pcm_stop(chip, s, subs);
+ }
+ break;
+ }
+ spin_unlock(&chip->reg_lock);
+ return err;
+}
+
+/*
+ * setup
+ */
+static void
+snd_m3_pcm_setup1(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substream *subs)
+{
+ int dsp_in_size, dsp_out_size, dsp_in_buffer, dsp_out_buffer;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dsp_in_size = MINISRC_IN_BUFFER_SIZE - (0x20 * 2);
+ dsp_out_size = MINISRC_OUT_BUFFER_SIZE - (0x20 * 2);
+ } else {
+ dsp_in_size = MINISRC_IN_BUFFER_SIZE - (0x10 * 2);
+ dsp_out_size = MINISRC_OUT_BUFFER_SIZE - (0x10 * 2);
+ }
+ dsp_in_buffer = s->inst.data + (MINISRC_TMP_BUFFER_SIZE / 2);
+ dsp_out_buffer = dsp_in_buffer + (dsp_in_size / 2) + 1;
+
+ s->dma_size = frames_to_bytes(runtime, runtime->buffer_size);
+ s->period_size = frames_to_bytes(runtime, runtime->period_size);
+ s->hwptr = 0;
+ s->count = 0;
+
+#define LO(x) ((x) & 0xffff)
+#define HI(x) LO((x) >> 16)
+
+ /* host dma buffer pointers */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_ADDRL,
+ LO(s->buffer_addr));
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_ADDRH,
+ HI(s->buffer_addr));
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_END_PLUS_1L,
+ LO(s->buffer_addr + s->dma_size));
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_END_PLUS_1H,
+ HI(s->buffer_addr + s->dma_size));
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_CURRENTL,
+ LO(s->buffer_addr));
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_CURRENTH,
+ HI(s->buffer_addr));
+#undef LO
+#undef HI
+
+ /* dsp buffers */
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_IN_BUF_BEGIN,
+ dsp_in_buffer);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_IN_BUF_END_PLUS_1,
+ dsp_in_buffer + (dsp_in_size / 2));
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_IN_BUF_HEAD,
+ dsp_in_buffer);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_IN_BUF_TAIL,
+ dsp_in_buffer);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_OUT_BUF_BEGIN,
+ dsp_out_buffer);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_OUT_BUF_END_PLUS_1,
+ dsp_out_buffer + (dsp_out_size / 2));
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_OUT_BUF_HEAD,
+ dsp_out_buffer);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_OUT_BUF_TAIL,
+ dsp_out_buffer);
+}
+
+static void snd_m3_pcm_setup2(struct snd_m3 *chip, struct m3_dma *s,
+ struct snd_pcm_runtime *runtime)
+{
+ u32 freq;
+
+ /*
+ * put us in the lists if we're not already there
+ */
+ if (! s->in_lists) {
+ s->index[0] = snd_m3_add_list(chip, s->index_list[0],
+ s->inst.data >> DP_SHIFT_COUNT);
+ s->index[1] = snd_m3_add_list(chip, s->index_list[1],
+ s->inst.data >> DP_SHIFT_COUNT);
+ s->index[2] = snd_m3_add_list(chip, s->index_list[2],
+ s->inst.data >> DP_SHIFT_COUNT);
+ s->in_lists = 1;
+ }
+
+ /* write to 'mono' word */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + SRC3_DIRECTION_OFFSET + 1,
+ runtime->channels == 2 ? 0 : 1);
+ /* write to '8bit' word */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + SRC3_DIRECTION_OFFSET + 2,
+ snd_pcm_format_width(runtime->format) == 16 ? 0 : 1);
+
+ /* set up dac/adc rate */
+ freq = ((runtime->rate << 15) + 24000 ) / 48000;
+ if (freq)
+ freq--;
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_FREQUENCY,
+ freq);
+}
+
+
+static const struct play_vals {
+ u16 addr, val;
+} pv[] = {
+ {CDATA_LEFT_VOLUME, ARB_VOLUME},
+ {CDATA_RIGHT_VOLUME, ARB_VOLUME},
+ {SRC3_DIRECTION_OFFSET, 0} ,
+ /* +1, +2 are stereo/16 bit */
+ {SRC3_DIRECTION_OFFSET + 3, 0x0000}, /* fraction? */
+ {SRC3_DIRECTION_OFFSET + 4, 0}, /* first l */
+ {SRC3_DIRECTION_OFFSET + 5, 0}, /* first r */
+ {SRC3_DIRECTION_OFFSET + 6, 0}, /* second l */
+ {SRC3_DIRECTION_OFFSET + 7, 0}, /* second r */
+ {SRC3_DIRECTION_OFFSET + 8, 0}, /* delta l */
+ {SRC3_DIRECTION_OFFSET + 9, 0}, /* delta r */
+ {SRC3_DIRECTION_OFFSET + 10, 0x8000}, /* round */
+ {SRC3_DIRECTION_OFFSET + 11, 0xFF00}, /* higher bute mark */
+ {SRC3_DIRECTION_OFFSET + 13, 0}, /* temp0 */
+ {SRC3_DIRECTION_OFFSET + 14, 0}, /* c fraction */
+ {SRC3_DIRECTION_OFFSET + 15, 0}, /* counter */
+ {SRC3_DIRECTION_OFFSET + 16, 8}, /* numin */
+ {SRC3_DIRECTION_OFFSET + 17, 50*2}, /* numout */
+ {SRC3_DIRECTION_OFFSET + 18, MINISRC_BIQUAD_STAGE - 1}, /* numstage */
+ {SRC3_DIRECTION_OFFSET + 20, 0}, /* filtertap */
+ {SRC3_DIRECTION_OFFSET + 21, 0} /* booster */
+};
+
+
+/* the mode passed should be already shifted and masked */
+static void
+snd_m3_playback_setup(struct snd_m3 *chip, struct m3_dma *s,
+ struct snd_pcm_substream *subs)
+{
+ unsigned int i;
+
+ /*
+ * some per client initializers
+ */
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + SRC3_DIRECTION_OFFSET + 12,
+ s->inst.data + 40 + 8);
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + SRC3_DIRECTION_OFFSET + 19,
+ s->inst.code + MINISRC_COEF_LOC);
+
+ /* enable or disable low pass filter? */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + SRC3_DIRECTION_OFFSET + 22,
+ subs->runtime->rate > 45000 ? 0xff : 0);
+
+ /* tell it which way dma is going? */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_DMA_CONTROL,
+ DMACONTROL_AUTOREPEAT + DMAC_PAGE3_SELECTOR + DMAC_BLOCKF_SELECTOR);
+
+ /*
+ * set an armload of static initializers
+ */
+ for (i = 0; i < ARRAY_SIZE(pv); i++)
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + pv[i].addr, pv[i].val);
+}
+
+/*
+ * Native record driver
+ */
+static const struct rec_vals {
+ u16 addr, val;
+} rv[] = {
+ {CDATA_LEFT_VOLUME, ARB_VOLUME},
+ {CDATA_RIGHT_VOLUME, ARB_VOLUME},
+ {SRC3_DIRECTION_OFFSET, 1} ,
+ /* +1, +2 are stereo/16 bit */
+ {SRC3_DIRECTION_OFFSET + 3, 0x0000}, /* fraction? */
+ {SRC3_DIRECTION_OFFSET + 4, 0}, /* first l */
+ {SRC3_DIRECTION_OFFSET + 5, 0}, /* first r */
+ {SRC3_DIRECTION_OFFSET + 6, 0}, /* second l */
+ {SRC3_DIRECTION_OFFSET + 7, 0}, /* second r */
+ {SRC3_DIRECTION_OFFSET + 8, 0}, /* delta l */
+ {SRC3_DIRECTION_OFFSET + 9, 0}, /* delta r */
+ {SRC3_DIRECTION_OFFSET + 10, 0x8000}, /* round */
+ {SRC3_DIRECTION_OFFSET + 11, 0xFF00}, /* higher bute mark */
+ {SRC3_DIRECTION_OFFSET + 13, 0}, /* temp0 */
+ {SRC3_DIRECTION_OFFSET + 14, 0}, /* c fraction */
+ {SRC3_DIRECTION_OFFSET + 15, 0}, /* counter */
+ {SRC3_DIRECTION_OFFSET + 16, 50},/* numin */
+ {SRC3_DIRECTION_OFFSET + 17, 8}, /* numout */
+ {SRC3_DIRECTION_OFFSET + 18, 0}, /* numstage */
+ {SRC3_DIRECTION_OFFSET + 19, 0}, /* coef */
+ {SRC3_DIRECTION_OFFSET + 20, 0}, /* filtertap */
+ {SRC3_DIRECTION_OFFSET + 21, 0}, /* booster */
+ {SRC3_DIRECTION_OFFSET + 22, 0xff} /* skip lpf */
+};
+
+static void
+snd_m3_capture_setup(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substream *subs)
+{
+ unsigned int i;
+
+ /*
+ * some per client initializers
+ */
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + SRC3_DIRECTION_OFFSET + 12,
+ s->inst.data + 40 + 8);
+
+ /* tell it which way dma is going? */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_DMA_CONTROL,
+ DMACONTROL_DIRECTION + DMACONTROL_AUTOREPEAT +
+ DMAC_PAGE3_SELECTOR + DMAC_BLOCKF_SELECTOR);
+
+ /*
+ * set an armload of static initializers
+ */
+ for (i = 0; i < ARRAY_SIZE(rv); i++)
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + rv[i].addr, rv[i].val);
+}
+
+static int snd_m3_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct m3_dma *s = substream->runtime->private_data;
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ /* set buffer address */
+ s->buffer_addr = substream->runtime->dma_addr;
+ if (s->buffer_addr & 0x3) {
+ snd_printk(KERN_ERR "oh my, not aligned\n");
+ s->buffer_addr = s->buffer_addr & ~0x3;
+ }
+ return 0;
+}
+
+static int snd_m3_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct m3_dma *s;
+
+ if (substream->runtime->private_data == NULL)
+ return 0;
+ s = substream->runtime->private_data;
+ snd_pcm_lib_free_pages(substream);
+ s->buffer_addr = 0;
+ return 0;
+}
+
+static int
+snd_m3_pcm_prepare(struct snd_pcm_substream *subs)
+{
+ struct snd_m3 *chip = snd_pcm_substream_chip(subs);
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct m3_dma *s = runtime->private_data;
+
+ if (snd_BUG_ON(!s))
+ return -ENXIO;
+
+ if (runtime->format != SNDRV_PCM_FORMAT_U8 &&
+ runtime->format != SNDRV_PCM_FORMAT_S16_LE)
+ return -EINVAL;
+ if (runtime->rate > 48000 ||
+ runtime->rate < 8000)
+ return -EINVAL;
+
+ spin_lock_irq(&chip->reg_lock);
+
+ snd_m3_pcm_setup1(chip, s, subs);
+
+ if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_m3_playback_setup(chip, s, subs);
+ else
+ snd_m3_capture_setup(chip, s, subs);
+
+ snd_m3_pcm_setup2(chip, s, runtime);
+
+ spin_unlock_irq(&chip->reg_lock);
+
+ return 0;
+}
+
+/*
+ * get current pointer
+ */
+static unsigned int
+snd_m3_get_pointer(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substream *subs)
+{
+ u16 hi = 0, lo = 0;
+ int retry = 10;
+ u32 addr;
+
+ /*
+ * try and get a valid answer
+ */
+ while (retry--) {
+ hi = snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_CURRENTH);
+
+ lo = snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_CURRENTL);
+
+ if (hi == snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA,
+ s->inst.data + CDATA_HOST_SRC_CURRENTH))
+ break;
+ }
+ addr = lo | ((u32)hi<<16);
+ return (unsigned int)(addr - s->buffer_addr);
+}
+
+static snd_pcm_uframes_t
+snd_m3_pcm_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_m3 *chip = snd_pcm_substream_chip(subs);
+ unsigned int ptr;
+ struct m3_dma *s = subs->runtime->private_data;
+
+ if (snd_BUG_ON(!s))
+ return 0;
+
+ spin_lock(&chip->reg_lock);
+ ptr = snd_m3_get_pointer(chip, s, subs);
+ spin_unlock(&chip->reg_lock);
+ return bytes_to_frames(subs->runtime, ptr);
+}
+
+
+/* update pointer */
+/* spinlock held! */
+static void snd_m3_update_ptr(struct snd_m3 *chip, struct m3_dma *s)
+{
+ struct snd_pcm_substream *subs = s->substream;
+ unsigned int hwptr;
+ int diff;
+
+ if (! s->running)
+ return;
+
+ hwptr = snd_m3_get_pointer(chip, s, subs);
+
+ /* try to avoid expensive modulo divisions */
+ if (hwptr >= s->dma_size)
+ hwptr %= s->dma_size;
+
+ diff = s->dma_size + hwptr - s->hwptr;
+ if (diff >= s->dma_size)
+ diff %= s->dma_size;
+
+ s->hwptr = hwptr;
+ s->count += diff;
+
+ if (s->count >= (signed)s->period_size) {
+
+ if (s->count < 2 * (signed)s->period_size)
+ s->count -= (signed)s->period_size;
+ else
+ s->count %= s->period_size;
+
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(subs);
+ spin_lock(&chip->reg_lock);
+ }
+}
+
+static void snd_m3_update_hw_volume(unsigned long private_data)
+{
+ struct snd_m3 *chip = (struct snd_m3 *) private_data;
+ int x, val;
+ unsigned long flags;
+
+ /* Figure out which volume control button was pushed,
+ based on differences from the default register
+ values. */
+ x = inb(chip->iobase + SHADOW_MIX_REG_VOICE) & 0xee;
+
+ /* Reset the volume control registers. */
+ outb(0x88, chip->iobase + SHADOW_MIX_REG_VOICE);
+ outb(0x88, chip->iobase + HW_VOL_COUNTER_VOICE);
+ outb(0x88, chip->iobase + SHADOW_MIX_REG_MASTER);
+ outb(0x88, chip->iobase + HW_VOL_COUNTER_MASTER);
+
+ if (!chip->master_switch || !chip->master_volume)
+ return;
+
+ /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
+ spin_lock_irqsave(&chip->ac97_lock, flags);
+
+ val = chip->ac97->regs[AC97_MASTER_VOL];
+ switch (x) {
+ case 0x88:
+ /* mute */
+ val ^= 0x8000;
+ chip->ac97->regs[AC97_MASTER_VOL] = val;
+ outw(val, chip->iobase + CODEC_DATA);
+ outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_switch->id);
+ break;
+ case 0xaa:
+ /* volume up */
+ if ((val & 0x7f) > 0)
+ val--;
+ if ((val & 0x7f00) > 0)
+ val -= 0x0100;
+ chip->ac97->regs[AC97_MASTER_VOL] = val;
+ outw(val, chip->iobase + CODEC_DATA);
+ outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ break;
+ case 0x66:
+ /* volume down */
+ if ((val & 0x7f) < 0x1f)
+ val++;
+ if ((val & 0x7f00) < 0x1f00)
+ val += 0x0100;
+ chip->ac97->regs[AC97_MASTER_VOL] = val;
+ outw(val, chip->iobase + CODEC_DATA);
+ outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ break;
+ }
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+}
+
+static irqreturn_t snd_m3_interrupt(int irq, void *dev_id)
+{
+ struct snd_m3 *chip = dev_id;
+ u8 status;
+ int i;
+
+ status = inb(chip->iobase + HOST_INT_STATUS);
+
+ if (status == 0xff)
+ return IRQ_NONE;
+
+ if (status & HV_INT_PENDING)
+ tasklet_hi_schedule(&chip->hwvol_tq);
+
+ /*
+ * ack an assp int if its running
+ * and has an int pending
+ */
+ if (status & ASSP_INT_PENDING) {
+ u8 ctl = inb(chip->iobase + ASSP_CONTROL_B);
+ if (!(ctl & STOP_ASSP_CLOCK)) {
+ ctl = inb(chip->iobase + ASSP_HOST_INT_STATUS);
+ if (ctl & DSP2HOST_REQ_TIMER) {
+ outb(DSP2HOST_REQ_TIMER, chip->iobase + ASSP_HOST_INT_STATUS);
+ /* update adc/dac info if it was a timer int */
+ spin_lock(&chip->reg_lock);
+ for (i = 0; i < chip->num_substreams; i++) {
+ struct m3_dma *s = &chip->substreams[i];
+ if (s->running)
+ snd_m3_update_ptr(chip, s);
+ }
+ spin_unlock(&chip->reg_lock);
+ }
+ }
+ }
+
+#if 0 /* TODO: not supported yet */
+ if ((status & MPU401_INT_PENDING) && chip->rmidi)
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data, regs);
+#endif
+
+ /* ack ints */
+ outb(status, chip->iobase + HOST_INT_STATUS);
+
+ return IRQ_HANDLED;
+}
+
+
+/*
+ */
+
+static struct snd_pcm_hardware snd_m3_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ /*SNDRV_PCM_INFO_PAUSE |*/
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (512*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (512*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static struct snd_pcm_hardware snd_m3_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ /*SNDRV_PCM_INFO_PAUSE |*/
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (512*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (512*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+
+/*
+ */
+
+static int
+snd_m3_substream_open(struct snd_m3 *chip, struct snd_pcm_substream *subs)
+{
+ int i;
+ struct m3_dma *s;
+
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < chip->num_substreams; i++) {
+ s = &chip->substreams[i];
+ if (! s->opened)
+ goto __found;
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return -ENOMEM;
+__found:
+ s->opened = 1;
+ s->running = 0;
+ spin_unlock_irq(&chip->reg_lock);
+
+ subs->runtime->private_data = s;
+ s->substream = subs;
+
+ /* set list owners */
+ if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ s->index_list[0] = &chip->mixer_list;
+ } else
+ s->index_list[0] = &chip->adc1_list;
+ s->index_list[1] = &chip->msrc_list;
+ s->index_list[2] = &chip->dma_list;
+
+ return 0;
+}
+
+static void
+snd_m3_substream_close(struct snd_m3 *chip, struct snd_pcm_substream *subs)
+{
+ struct m3_dma *s = subs->runtime->private_data;
+
+ if (s == NULL)
+ return; /* not opened properly */
+
+ spin_lock_irq(&chip->reg_lock);
+ if (s->substream && s->running)
+ snd_m3_pcm_stop(chip, s, s->substream); /* does this happen? */
+ if (s->in_lists) {
+ snd_m3_remove_list(chip, s->index_list[0], s->index[0]);
+ snd_m3_remove_list(chip, s->index_list[1], s->index[1]);
+ snd_m3_remove_list(chip, s->index_list[2], s->index[2]);
+ s->in_lists = 0;
+ }
+ s->running = 0;
+ s->opened = 0;
+ spin_unlock_irq(&chip->reg_lock);
+}
+
+static int
+snd_m3_playback_open(struct snd_pcm_substream *subs)
+{
+ struct snd_m3 *chip = snd_pcm_substream_chip(subs);
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ int err;
+
+ if ((err = snd_m3_substream_open(chip, subs)) < 0)
+ return err;
+
+ runtime->hw = snd_m3_playback;
+
+ return 0;
+}
+
+static int
+snd_m3_playback_close(struct snd_pcm_substream *subs)
+{
+ struct snd_m3 *chip = snd_pcm_substream_chip(subs);
+
+ snd_m3_substream_close(chip, subs);
+ return 0;
+}
+
+static int
+snd_m3_capture_open(struct snd_pcm_substream *subs)
+{
+ struct snd_m3 *chip = snd_pcm_substream_chip(subs);
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ int err;
+
+ if ((err = snd_m3_substream_open(chip, subs)) < 0)
+ return err;
+
+ runtime->hw = snd_m3_capture;
+
+ return 0;
+}
+
+static int
+snd_m3_capture_close(struct snd_pcm_substream *subs)
+{
+ struct snd_m3 *chip = snd_pcm_substream_chip(subs);
+
+ snd_m3_substream_close(chip, subs);
+ return 0;
+}
+
+/*
+ * create pcm instance
+ */
+
+static struct snd_pcm_ops snd_m3_playback_ops = {
+ .open = snd_m3_playback_open,
+ .close = snd_m3_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_m3_pcm_hw_params,
+ .hw_free = snd_m3_pcm_hw_free,
+ .prepare = snd_m3_pcm_prepare,
+ .trigger = snd_m3_pcm_trigger,
+ .pointer = snd_m3_pcm_pointer,
+};
+
+static struct snd_pcm_ops snd_m3_capture_ops = {
+ .open = snd_m3_capture_open,
+ .close = snd_m3_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_m3_pcm_hw_params,
+ .hw_free = snd_m3_pcm_hw_free,
+ .prepare = snd_m3_pcm_prepare,
+ .trigger = snd_m3_pcm_trigger,
+ .pointer = snd_m3_pcm_pointer,
+};
+
+static int __devinit
+snd_m3_pcm(struct snd_m3 * chip, int device)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(chip->card, chip->card->driver, device,
+ MAX_PLAYBACKS, MAX_CAPTURES, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_m3_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_m3_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, chip->card->driver);
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 64*1024);
+
+ return 0;
+}
+
+
+/*
+ * ac97 interface
+ */
+
+/*
+ * Wait for the ac97 serial bus to be free.
+ * return nonzero if the bus is still busy.
+ */
+static int snd_m3_ac97_wait(struct snd_m3 *chip)
+{
+ int i = 10000;
+
+ do {
+ if (! (snd_m3_inb(chip, 0x30) & 1))
+ return 0;
+ cpu_relax();
+ } while (i-- > 0);
+
+ snd_printk(KERN_ERR "ac97 serial bus busy\n");
+ return 1;
+}
+
+static unsigned short
+snd_m3_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct snd_m3 *chip = ac97->private_data;
+ unsigned long flags;
+ unsigned short data = 0xffff;
+
+ if (snd_m3_ac97_wait(chip))
+ goto fail;
+ spin_lock_irqsave(&chip->ac97_lock, flags);
+ snd_m3_outb(chip, 0x80 | (reg & 0x7f), CODEC_COMMAND);
+ if (snd_m3_ac97_wait(chip))
+ goto fail_unlock;
+ data = snd_m3_inw(chip, CODEC_DATA);
+fail_unlock:
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+fail:
+ return data;
+}
+
+static void
+snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
+{
+ struct snd_m3 *chip = ac97->private_data;
+ unsigned long flags;
+
+ if (snd_m3_ac97_wait(chip))
+ return;
+ spin_lock_irqsave(&chip->ac97_lock, flags);
+ snd_m3_outw(chip, val, CODEC_DATA);
+ snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+}
+
+
+static void snd_m3_remote_codec_config(int io, int isremote)
+{
+ isremote = isremote ? 1 : 0;
+
+ outw((inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK) | isremote,
+ io + RING_BUS_CTRL_B);
+ outw((inw(io + SDO_OUT_DEST_CTRL) & ~COMMAND_ADDR_OUT) | isremote,
+ io + SDO_OUT_DEST_CTRL);
+ outw((inw(io + SDO_IN_DEST_CTRL) & ~STATUS_ADDR_IN) | isremote,
+ io + SDO_IN_DEST_CTRL);
+}
+
+/*
+ * hack, returns non zero on err
+ */
+static int snd_m3_try_read_vendor(struct snd_m3 *chip)
+{
+ u16 ret;
+
+ if (snd_m3_ac97_wait(chip))
+ return 1;
+
+ snd_m3_outb(chip, 0x80 | (AC97_VENDOR_ID1 & 0x7f), 0x30);
+
+ if (snd_m3_ac97_wait(chip))
+ return 1;
+
+ ret = snd_m3_inw(chip, 0x32);
+
+ return (ret == 0) || (ret == 0xffff);
+}
+
+static void snd_m3_ac97_reset(struct snd_m3 *chip)
+{
+ u16 dir;
+ int delay1 = 0, delay2 = 0, i;
+ int io = chip->iobase;
+
+ if (chip->allegro_flag) {
+ /*
+ * the onboard codec on the allegro seems
+ * to want to wait a very long time before
+ * coming back to life
+ */
+ delay1 = 50;
+ delay2 = 800;
+ } else {
+ /* maestro3 */
+ delay1 = 20;
+ delay2 = 500;
+ }
+
+ for (i = 0; i < 5; i++) {
+ dir = inw(io + GPIO_DIRECTION);
+ if (!chip->irda_workaround)
+ dir |= 0x10; /* assuming pci bus master? */
+
+ snd_m3_remote_codec_config(io, 0);
+
+ outw(IO_SRAM_ENABLE, io + RING_BUS_CTRL_A);
+ udelay(20);
+
+ outw(dir & ~GPO_PRIMARY_AC97 , io + GPIO_DIRECTION);
+ outw(~GPO_PRIMARY_AC97 , io + GPIO_MASK);
+ outw(0, io + GPIO_DATA);
+ outw(dir | GPO_PRIMARY_AC97, io + GPIO_DIRECTION);
+
+ schedule_timeout_uninterruptible(msecs_to_jiffies(delay1));
+
+ outw(GPO_PRIMARY_AC97, io + GPIO_DATA);
+ udelay(5);
+ /* ok, bring back the ac-link */
+ outw(IO_SRAM_ENABLE | SERIAL_AC_LINK_ENABLE, io + RING_BUS_CTRL_A);
+ outw(~0, io + GPIO_MASK);
+
+ schedule_timeout_uninterruptible(msecs_to_jiffies(delay2));
+
+ if (! snd_m3_try_read_vendor(chip))
+ break;
+
+ delay1 += 10;
+ delay2 += 100;
+
+ snd_printd("maestro3: retrying codec reset with delays of %d and %d ms\n",
+ delay1, delay2);
+ }
+
+#if 0
+ /* more gung-ho reset that doesn't
+ * seem to work anywhere :)
+ */
+ tmp = inw(io + RING_BUS_CTRL_A);
+ outw(RAC_SDFS_ENABLE|LAC_SDFS_ENABLE, io + RING_BUS_CTRL_A);
+ msleep(20);
+ outw(tmp, io + RING_BUS_CTRL_A);
+ msleep(50);
+#endif
+}
+
+static int __devinit snd_m3_mixer(struct snd_m3 *chip)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ struct snd_ctl_elem_id elem_id;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_m3_ac97_write,
+ .read = snd_m3_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97)) < 0)
+ return err;
+
+ /* seems ac97 PCM needs initialization.. hack hack.. */
+ snd_ac97_write(chip->ac97, AC97_PCM, 0x8000 | (15 << 8) | 15);
+ schedule_timeout_uninterruptible(msecs_to_jiffies(100));
+ snd_ac97_write(chip->ac97, AC97_PCM, 0);
+
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Switch");
+ chip->master_switch = snd_ctl_find_id(chip->card, &elem_id);
+ memset(&elem_id, 0, sizeof(elem_id));
+ elem_id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(elem_id.name, "Master Playback Volume");
+ chip->master_volume = snd_ctl_find_id(chip->card, &elem_id);
+
+ return 0;
+}
+
+
+/*
+ * initialize ASSP
+ */
+
+#define MINISRC_LPF_LEN 10
+static const u16 minisrc_lpf[MINISRC_LPF_LEN] = {
+ 0X0743, 0X1104, 0X0A4C, 0XF88D, 0X242C,
+ 0X1023, 0X1AA9, 0X0B60, 0XEFDD, 0X186F
+};
+
+static void snd_m3_assp_init(struct snd_m3 *chip)
+{
+ unsigned int i;
+ const u16 *data;
+
+ /* zero kernel data */
+ for (i = 0; i < (REV_B_DATA_MEMORY_UNIT_LENGTH * NUM_UNITS_KERNEL_DATA) / 2; i++)
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_BASE_ADDR + i, 0);
+
+ /* zero mixer data? */
+ for (i = 0; i < (REV_B_DATA_MEMORY_UNIT_LENGTH * NUM_UNITS_KERNEL_DATA) / 2; i++)
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_BASE_ADDR2 + i, 0);
+
+ /* init dma pointer */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_CURRENT_DMA,
+ KDATA_DMA_XFER0);
+
+ /* write kernel into code memory.. */
+ data = (const u16 *)chip->assp_kernel_image->data;
+ for (i = 0 ; i * 2 < chip->assp_kernel_image->size; i++) {
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE,
+ REV_B_CODE_MEMORY_BEGIN + i,
+ le16_to_cpu(data[i]));
+ }
+
+ /*
+ * We only have this one client and we know that 0x400
+ * is free in our kernel's mem map, so lets just
+ * drop it there. It seems that the minisrc doesn't
+ * need vectors, so we won't bother with them..
+ */
+ data = (const u16 *)chip->assp_minisrc_image->data;
+ for (i = 0; i * 2 < chip->assp_minisrc_image->size; i++) {
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE,
+ 0x400 + i, le16_to_cpu(data[i]));
+ }
+
+ /*
+ * write the coefficients for the low pass filter?
+ */
+ for (i = 0; i < MINISRC_LPF_LEN ; i++) {
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE,
+ 0x400 + MINISRC_COEF_LOC + i,
+ minisrc_lpf[i]);
+ }
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE,
+ 0x400 + MINISRC_COEF_LOC + MINISRC_LPF_LEN,
+ 0x8000);
+
+ /*
+ * the minisrc is the only thing on
+ * our task list..
+ */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_TASK0,
+ 0x400);
+
+ /*
+ * init the mixer number..
+ */
+
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_MIXER_TASK_NUMBER,0);
+
+ /*
+ * EXTREME KERNEL MASTER VOLUME
+ */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_DAC_LEFT_VOLUME, ARB_VOLUME);
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_DAC_RIGHT_VOLUME, ARB_VOLUME);
+
+ chip->mixer_list.curlen = 0;
+ chip->mixer_list.mem_addr = KDATA_MIXER_XFER0;
+ chip->mixer_list.max = MAX_VIRTUAL_MIXER_CHANNELS;
+ chip->adc1_list.curlen = 0;
+ chip->adc1_list.mem_addr = KDATA_ADC1_XFER0;
+ chip->adc1_list.max = MAX_VIRTUAL_ADC1_CHANNELS;
+ chip->dma_list.curlen = 0;
+ chip->dma_list.mem_addr = KDATA_DMA_XFER0;
+ chip->dma_list.max = MAX_VIRTUAL_DMA_CHANNELS;
+ chip->msrc_list.curlen = 0;
+ chip->msrc_list.mem_addr = KDATA_INSTANCE0_MINISRC;
+ chip->msrc_list.max = MAX_INSTANCE_MINISRC;
+}
+
+
+static int __devinit snd_m3_assp_client_init(struct snd_m3 *chip, struct m3_dma *s, int index)
+{
+ int data_bytes = 2 * ( MINISRC_TMP_BUFFER_SIZE / 2 +
+ MINISRC_IN_BUFFER_SIZE / 2 +
+ 1 + MINISRC_OUT_BUFFER_SIZE / 2 + 1 );
+ int address, i;
+
+ /*
+ * the revb memory map has 0x1100 through 0x1c00
+ * free.
+ */
+
+ /*
+ * align instance address to 256 bytes so that its
+ * shifted list address is aligned.
+ * list address = (mem address >> 1) >> 7;
+ */
+ data_bytes = ALIGN(data_bytes, 256);
+ address = 0x1100 + ((data_bytes/2) * index);
+
+ if ((address + (data_bytes/2)) >= 0x1c00) {
+ snd_printk(KERN_ERR "no memory for %d bytes at ind %d (addr 0x%x)\n",
+ data_bytes, index, address);
+ return -ENOMEM;
+ }
+
+ s->number = index;
+ s->inst.code = 0x400;
+ s->inst.data = address;
+
+ for (i = data_bytes / 2; i > 0; address++, i--) {
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ address, 0);
+ }
+
+ return 0;
+}
+
+
+/*
+ * this works for the reference board, have to find
+ * out about others
+ *
+ * this needs more magic for 4 speaker, but..
+ */
+static void
+snd_m3_amp_enable(struct snd_m3 *chip, int enable)
+{
+ int io = chip->iobase;
+ u16 gpo, polarity;
+
+ if (! chip->external_amp)
+ return;
+
+ polarity = enable ? 0 : 1;
+ polarity = polarity << chip->amp_gpio;
+ gpo = 1 << chip->amp_gpio;
+
+ outw(~gpo, io + GPIO_MASK);
+
+ outw(inw(io + GPIO_DIRECTION) | gpo,
+ io + GPIO_DIRECTION);
+
+ outw((GPO_SECONDARY_AC97 | GPO_PRIMARY_AC97 | polarity),
+ io + GPIO_DATA);
+
+ outw(0xffff, io + GPIO_MASK);
+}
+
+static void
+snd_m3_hv_init(struct snd_m3 *chip)
+{
+ unsigned long io = chip->iobase;
+ u16 val = GPI_VOL_DOWN | GPI_VOL_UP;
+
+ if (!chip->is_omnibook)
+ return;
+
+ /*
+ * Volume buttons on some HP OmniBook laptops
+ * require some GPIO magic to work correctly.
+ */
+ outw(0xffff, io + GPIO_MASK);
+ outw(0x0000, io + GPIO_DATA);
+
+ outw(~val, io + GPIO_MASK);
+ outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION);
+ outw(val, io + GPIO_MASK);
+
+ outw(0xffff, io + GPIO_MASK);
+}
+
+static int
+snd_m3_chip_init(struct snd_m3 *chip)
+{
+ struct pci_dev *pcidev = chip->pci;
+ unsigned long io = chip->iobase;
+ u32 n;
+ u16 w;
+ u8 t; /* makes as much sense as 'n', no? */
+
+ pci_read_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, &w);
+ w &= ~(SOUND_BLASTER_ENABLE|FM_SYNTHESIS_ENABLE|
+ MPU401_IO_ENABLE|MPU401_IRQ_ENABLE|ALIAS_10BIT_IO|
+ DISABLE_LEGACY);
+ pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w);
+
+ pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n);
+ n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD);
+ n |= chip->hv_config;
+ /* For some reason we must always use reduced debounce. */
+ n |= REDUCED_DEBOUNCE;
+ n |= PM_CTRL_ENABLE | CLK_DIV_BY_49 | USE_PCI_TIMING;
+ pci_write_config_dword(pcidev, PCI_ALLEGRO_CONFIG, n);
+
+ outb(RESET_ASSP, chip->iobase + ASSP_CONTROL_B);
+ pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n);
+ n &= ~INT_CLK_SELECT;
+ if (!chip->allegro_flag) {
+ n &= ~INT_CLK_MULT_ENABLE;
+ n |= INT_CLK_SRC_NOT_PCI;
+ }
+ n &= ~( CLK_MULT_MODE_SELECT | CLK_MULT_MODE_SELECT_2 );
+ pci_write_config_dword(pcidev, PCI_ALLEGRO_CONFIG, n);
+
+ if (chip->allegro_flag) {
+ pci_read_config_dword(pcidev, PCI_USER_CONFIG, &n);
+ n |= IN_CLK_12MHZ_SELECT;
+ pci_write_config_dword(pcidev, PCI_USER_CONFIG, n);
+ }
+
+ t = inb(chip->iobase + ASSP_CONTROL_A);
+ t &= ~( DSP_CLK_36MHZ_SELECT | ASSP_CLK_49MHZ_SELECT);
+ t |= ASSP_CLK_49MHZ_SELECT;
+ t |= ASSP_0_WS_ENABLE;
+ outb(t, chip->iobase + ASSP_CONTROL_A);
+
+ snd_m3_assp_init(chip); /* download DSP code before starting ASSP below */
+ outb(RUN_ASSP, chip->iobase + ASSP_CONTROL_B);
+
+ outb(0x00, io + HARDWARE_VOL_CTRL);
+ outb(0x88, io + SHADOW_MIX_REG_VOICE);
+ outb(0x88, io + HW_VOL_COUNTER_VOICE);
+ outb(0x88, io + SHADOW_MIX_REG_MASTER);
+ outb(0x88, io + HW_VOL_COUNTER_MASTER);
+
+ return 0;
+}
+
+static void
+snd_m3_enable_ints(struct snd_m3 *chip)
+{
+ unsigned long io = chip->iobase;
+ unsigned short val;
+
+ /* TODO: MPU401 not supported yet */
+ val = ASSP_INT_ENABLE /*| MPU401_INT_ENABLE*/;
+ if (chip->hv_config & HV_CTRL_ENABLE)
+ val |= HV_INT_ENABLE;
+ outw(val, io + HOST_INT_CTRL);
+ outb(inb(io + ASSP_CONTROL_C) | ASSP_HOST_INT_ENABLE,
+ io + ASSP_CONTROL_C);
+}
+
+
+/*
+ */
+
+static int snd_m3_free(struct snd_m3 *chip)
+{
+ struct m3_dma *s;
+ int i;
+
+ if (chip->substreams) {
+ spin_lock_irq(&chip->reg_lock);
+ for (i = 0; i < chip->num_substreams; i++) {
+ s = &chip->substreams[i];
+ /* check surviving pcms; this should not happen though.. */
+ if (s->substream && s->running)
+ snd_m3_pcm_stop(chip, s, s->substream);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ kfree(chip->substreams);
+ }
+ if (chip->iobase) {
+ outw(0, chip->iobase + HOST_INT_CTRL); /* disable ints */
+ }
+
+#ifdef CONFIG_PM
+ vfree(chip->suspend_mem);
+#endif
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+
+ if (chip->iobase)
+ pci_release_regions(chip->pci);
+
+ release_firmware(chip->assp_kernel_image);
+ release_firmware(chip->assp_minisrc_image);
+
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+
+/*
+ * APM support
+ */
+#ifdef CONFIG_PM
+static int m3_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_m3 *chip = card->private_data;
+ int i, dsp_index;
+
+ if (chip->suspend_mem == NULL)
+ return 0;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_ac97_suspend(chip->ac97);
+
+ msleep(10); /* give the assp a chance to idle.. */
+
+ snd_m3_assp_halt(chip);
+
+ /* save dsp image */
+ dsp_index = 0;
+ for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++)
+ chip->suspend_mem[dsp_index++] =
+ snd_m3_assp_read(chip, MEMTYPE_INTERNAL_CODE, i);
+ for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++)
+ chip->suspend_mem[dsp_index++] =
+ snd_m3_assp_read(chip, MEMTYPE_INTERNAL_DATA, i);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int m3_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_m3 *chip = card->private_data;
+ int i, dsp_index;
+
+ if (chip->suspend_mem == NULL)
+ return 0;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "maestor3: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ /* first lets just bring everything back. .*/
+ snd_m3_outw(chip, 0, 0x54);
+ snd_m3_outw(chip, 0, 0x56);
+
+ snd_m3_chip_init(chip);
+ snd_m3_assp_halt(chip);
+ snd_m3_ac97_reset(chip);
+
+ /* restore dsp image */
+ dsp_index = 0;
+ for (i = REV_B_CODE_MEMORY_BEGIN; i <= REV_B_CODE_MEMORY_END; i++)
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, i,
+ chip->suspend_mem[dsp_index++]);
+ for (i = REV_B_DATA_MEMORY_BEGIN ; i <= REV_B_DATA_MEMORY_END; i++)
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA, i,
+ chip->suspend_mem[dsp_index++]);
+
+ /* tell the dma engine to restart itself */
+ snd_m3_assp_write(chip, MEMTYPE_INTERNAL_DATA,
+ KDATA_DMA_ACTIVE, 0);
+
+ /* restore ac97 registers */
+ snd_ac97_resume(chip->ac97);
+
+ snd_m3_assp_continue(chip);
+ snd_m3_enable_ints(chip);
+ snd_m3_amp_enable(chip, 1);
+
+ snd_m3_hv_init(chip);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+
+/*
+ */
+
+static int snd_m3_dev_free(struct snd_device *device)
+{
+ struct snd_m3 *chip = device->device_data;
+ return snd_m3_free(chip);
+}
+
+static int __devinit
+snd_m3_create(struct snd_card *card, struct pci_dev *pci,
+ int enable_amp,
+ int amp_gpio,
+ struct snd_m3 **chip_ret)
+{
+ struct snd_m3 *chip;
+ int i, err;
+ const struct snd_pci_quirk *quirk;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_m3_dev_free,
+ };
+
+ *chip_ret = NULL;
+
+ if (pci_enable_device(pci))
+ return -EIO;
+
+ /* check, if we can restrict PCI DMA transfers to 28 bits */
+ if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->ac97_lock);
+
+ switch (pci->device) {
+ case PCI_DEVICE_ID_ESS_ALLEGRO:
+ case PCI_DEVICE_ID_ESS_ALLEGRO_1:
+ case PCI_DEVICE_ID_ESS_CANYON3D_2LE:
+ case PCI_DEVICE_ID_ESS_CANYON3D_2:
+ chip->allegro_flag = 1;
+ break;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ chip->external_amp = enable_amp;
+ if (amp_gpio >= 0 && amp_gpio <= 0x0f)
+ chip->amp_gpio = amp_gpio;
+ else {
+ quirk = snd_pci_quirk_lookup(pci, m3_amp_quirk_list);
+ if (quirk) {
+ snd_printdd(KERN_INFO "maestro3: set amp-gpio "
+ "for '%s'\n", quirk->name);
+ chip->amp_gpio = quirk->value;
+ } else if (chip->allegro_flag)
+ chip->amp_gpio = GPO_EXT_AMP_ALLEGRO;
+ else /* presumably this is for all 'maestro3's.. */
+ chip->amp_gpio = GPO_EXT_AMP_M3;
+ }
+
+ quirk = snd_pci_quirk_lookup(pci, m3_irda_quirk_list);
+ if (quirk) {
+ snd_printdd(KERN_INFO "maestro3: enabled irda workaround "
+ "for '%s'\n", quirk->name);
+ chip->irda_workaround = 1;
+ }
+ quirk = snd_pci_quirk_lookup(pci, m3_hv_quirk_list);
+ if (quirk)
+ chip->hv_config = quirk->value;
+ if (snd_pci_quirk_lookup(pci, m3_omnibook_quirk_list))
+ chip->is_omnibook = 1;
+
+ chip->num_substreams = NR_DSPS;
+ chip->substreams = kcalloc(chip->num_substreams, sizeof(struct m3_dma),
+ GFP_KERNEL);
+ if (chip->substreams == NULL) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ err = request_firmware(&chip->assp_kernel_image,
+ "ess/maestro3_assp_kernel.fw", &pci->dev);
+ if (err < 0) {
+ snd_m3_free(chip);
+ return err;
+ }
+
+ err = request_firmware(&chip->assp_minisrc_image,
+ "ess/maestro3_assp_minisrc.fw", &pci->dev);
+ if (err < 0) {
+ snd_m3_free(chip);
+ return err;
+ }
+
+ if ((err = pci_request_regions(pci, card->driver)) < 0) {
+ snd_m3_free(chip);
+ return err;
+ }
+ chip->iobase = pci_resource_start(pci, 0);
+
+ /* just to be sure */
+ pci_set_master(pci);
+
+ snd_m3_chip_init(chip);
+ snd_m3_assp_halt(chip);
+
+ snd_m3_ac97_reset(chip);
+
+ snd_m3_amp_enable(chip, 1);
+
+ snd_m3_hv_init(chip);
+
+ tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip);
+
+ if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED,
+ card->driver, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_m3_free(chip);
+ return -ENOMEM;
+ }
+ chip->irq = pci->irq;
+
+#ifdef CONFIG_PM
+ chip->suspend_mem = vmalloc(sizeof(u16) * (REV_B_CODE_MEMORY_LENGTH + REV_B_DATA_MEMORY_LENGTH));
+ if (chip->suspend_mem == NULL)
+ snd_printk(KERN_WARNING "can't allocate apm buffer\n");
+#endif
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_m3_free(chip);
+ return err;
+ }
+
+ if ((err = snd_m3_mixer(chip)) < 0)
+ return err;
+
+ for (i = 0; i < chip->num_substreams; i++) {
+ struct m3_dma *s = &chip->substreams[i];
+ if ((err = snd_m3_assp_client_init(chip, s, i)) < 0)
+ return err;
+ }
+
+ if ((err = snd_m3_pcm(chip, 0)) < 0)
+ return err;
+
+ snd_m3_enable_ints(chip);
+ snd_m3_assp_continue(chip);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *chip_ret = chip;
+
+ return 0;
+}
+
+/*
+ */
+static int __devinit
+snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_m3 *chip;
+ int err;
+
+ /* don't pick up modems */
+ if (((pci->class >> 8) & 0xffff) != PCI_CLASS_MULTIMEDIA_AUDIO)
+ return -ENODEV;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ switch (pci->device) {
+ case PCI_DEVICE_ID_ESS_ALLEGRO:
+ case PCI_DEVICE_ID_ESS_ALLEGRO_1:
+ strcpy(card->driver, "Allegro");
+ break;
+ case PCI_DEVICE_ID_ESS_CANYON3D_2LE:
+ case PCI_DEVICE_ID_ESS_CANYON3D_2:
+ strcpy(card->driver, "Canyon3D-2");
+ break;
+ default:
+ strcpy(card->driver, "Maestro3");
+ break;
+ }
+
+ if ((err = snd_m3_create(card, pci,
+ external_amp[dev],
+ amp_gpio[dev],
+ &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ sprintf(card->shortname, "ESS %s PCI", card->driver);
+ sprintf(card->longname, "%s at 0x%lx, irq %d",
+ card->shortname, chip->iobase, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+#if 0 /* TODO: not supported yet */
+ /* TODO enable MIDI IRQ and I/O */
+ err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401,
+ chip->iobase + MPU401_DATA_PORT,
+ MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi);
+ if (err < 0)
+ printk(KERN_WARNING "maestro3: no MIDI support.\n");
+#endif
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_m3_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Maestro3",
+ .id_table = snd_m3_ids,
+ .probe = snd_m3_probe,
+ .remove = __devexit_p(snd_m3_remove),
+#ifdef CONFIG_PM
+ .suspend = m3_suspend,
+ .resume = m3_resume,
+#endif
+};
+
+static int __init alsa_card_m3_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_m3_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_m3_init)
+module_exit(alsa_card_m3_exit)
diff --git a/sound/pci/mixart/Makefile b/sound/pci/mixart/Makefile
new file mode 100644
index 0000000..cce159e
--- /dev/null
+++ b/sound/pci/mixart/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-mixart-objs := mixart.o mixart_core.o mixart_hwdep.o mixart_mixer.o
+
+obj-$(CONFIG_SND_MIXART) += snd-mixart.o
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
new file mode 100644
index 0000000..f342def
--- /dev/null
+++ b/sound/pci/mixart/mixart.c
@@ -0,0 +1,1457 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * main file with alsa callbacks
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "mixart.h"
+#include "mixart_hwdep.h"
+#include "mixart_core.h"
+#include "mixart_mixer.h"
+
+#define CARD_NAME "miXart"
+
+MODULE_AUTHOR("Digigram <alsa@digigram.com>");
+MODULE_DESCRIPTION("Digigram " CARD_NAME);
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Digigram " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Digigram " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard.");
+
+/*
+ */
+
+static struct pci_device_id snd_mixart_ids[] = {
+ { 0x1057, 0x0003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* MC8240 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_mixart_ids);
+
+
+static int mixart_set_pipe_state(struct mixart_mgr *mgr,
+ struct mixart_pipe *pipe, int start)
+{
+ struct mixart_group_state_req group_state;
+ struct mixart_group_state_resp group_state_resp;
+ struct mixart_msg request;
+ int err;
+ u32 system_msg_uid;
+
+ switch(pipe->status) {
+ case PIPE_RUNNING:
+ case PIPE_CLOCK_SET:
+ if(start) return 0; /* already started */
+ break;
+ case PIPE_STOPPED:
+ if(!start) return 0; /* already stopped */
+ break;
+ default:
+ snd_printk(KERN_ERR "error mixart_set_pipe_state called with wrong pipe->status!\n");
+ return -EINVAL; /* function called with wrong pipe status */
+ }
+
+ system_msg_uid = 0x12345678; /* the event ! (take care: the MSB and two LSB's have to be 0) */
+
+ /* wait on the last MSG_SYSTEM_SEND_SYNCHRO_CMD command to be really finished */
+
+ request.message_id = MSG_SYSTEM_WAIT_SYNCHRO_CMD;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = &system_msg_uid;
+ request.size = sizeof(system_msg_uid);
+
+ err = snd_mixart_send_msg_wait_notif(mgr, &request, system_msg_uid);
+ if(err) {
+ snd_printk(KERN_ERR "error : MSG_SYSTEM_WAIT_SYNCHRO_CMD was not notified !\n");
+ return err;
+ }
+
+ /* start or stop the pipe (1 pipe) */
+
+ memset(&group_state, 0, sizeof(group_state));
+ group_state.pipe_count = 1;
+ group_state.pipe_uid[0] = pipe->group_uid;
+
+ if(start)
+ request.message_id = MSG_STREAM_START_STREAM_GRP_PACKET;
+ else
+ request.message_id = MSG_STREAM_STOP_STREAM_GRP_PACKET;
+
+ request.uid = pipe->group_uid; /*(struct mixart_uid){0,0};*/
+ request.data = &group_state;
+ request.size = sizeof(group_state);
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(group_state_resp), &group_state_resp);
+ if (err < 0 || group_state_resp.txx_status != 0) {
+ snd_printk(KERN_ERR "error MSG_STREAM_ST***_STREAM_GRP_PACKET err=%x stat=%x !\n", err, group_state_resp.txx_status);
+ return -EINVAL;
+ }
+
+ if(start) {
+ u32 stat;
+
+ group_state.pipe_count = 0; /* in case of start same command once again with pipe_count=0 */
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(group_state_resp), &group_state_resp);
+ if (err < 0 || group_state_resp.txx_status != 0) {
+ snd_printk(KERN_ERR "error MSG_STREAM_START_STREAM_GRP_PACKET err=%x stat=%x !\n", err, group_state_resp.txx_status);
+ return -EINVAL;
+ }
+
+ /* in case of start send a synchro top */
+
+ request.message_id = MSG_SYSTEM_SEND_SYNCHRO_CMD;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = NULL;
+ request.size = 0;
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(stat), &stat);
+ if (err < 0 || stat != 0) {
+ snd_printk(KERN_ERR "error MSG_SYSTEM_SEND_SYNCHRO_CMD err=%x stat=%x !\n", err, stat);
+ return -EINVAL;
+ }
+
+ pipe->status = PIPE_RUNNING;
+ }
+ else /* !start */
+ pipe->status = PIPE_STOPPED;
+
+ return 0;
+}
+
+
+static int mixart_set_clock(struct mixart_mgr *mgr,
+ struct mixart_pipe *pipe, unsigned int rate)
+{
+ struct mixart_msg request;
+ struct mixart_clock_properties clock_properties;
+ struct mixart_clock_properties_resp clock_prop_resp;
+ int err;
+
+ switch(pipe->status) {
+ case PIPE_CLOCK_SET:
+ break;
+ case PIPE_RUNNING:
+ if(rate != 0)
+ break;
+ default:
+ if(rate == 0)
+ return 0; /* nothing to do */
+ else {
+ snd_printk(KERN_ERR "error mixart_set_clock(%d) called with wrong pipe->status !\n", rate);
+ return -EINVAL;
+ }
+ }
+
+ memset(&clock_properties, 0, sizeof(clock_properties));
+ clock_properties.clock_generic_type = (rate != 0) ? CGT_INTERNAL_CLOCK : CGT_NO_CLOCK;
+ clock_properties.clock_mode = CM_STANDALONE;
+ clock_properties.frequency = rate;
+ clock_properties.nb_callers = 1; /* only one entry in uid_caller ! */
+ clock_properties.uid_caller[0] = pipe->group_uid;
+
+ snd_printdd("mixart_set_clock to %d kHz\n", rate);
+
+ request.message_id = MSG_CLOCK_SET_PROPERTIES;
+ request.uid = mgr->uid_console_manager;
+ request.data = &clock_properties;
+ request.size = sizeof(clock_properties);
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(clock_prop_resp), &clock_prop_resp);
+ if (err < 0 || clock_prop_resp.status != 0 || clock_prop_resp.clock_mode != CM_STANDALONE) {
+ snd_printk(KERN_ERR "error MSG_CLOCK_SET_PROPERTIES err=%x stat=%x mod=%x !\n", err, clock_prop_resp.status, clock_prop_resp.clock_mode);
+ return -EINVAL;
+ }
+
+ if(rate) pipe->status = PIPE_CLOCK_SET;
+ else pipe->status = PIPE_RUNNING;
+
+ return 0;
+}
+
+
+/*
+ * Allocate or reference output pipe for analog IOs (pcmp0/1)
+ */
+struct mixart_pipe *
+snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture,
+ int monitoring)
+{
+ int stream_count;
+ struct mixart_pipe *pipe;
+ struct mixart_msg request;
+
+ if(capture) {
+ if (pcm_number == MIXART_PCM_ANALOG) {
+ pipe = &(chip->pipe_in_ana); /* analog inputs */
+ } else {
+ pipe = &(chip->pipe_in_dig); /* digital inputs */
+ }
+ request.message_id = MSG_STREAM_ADD_OUTPUT_GROUP;
+ stream_count = MIXART_CAPTURE_STREAMS;
+ } else {
+ if (pcm_number == MIXART_PCM_ANALOG) {
+ pipe = &(chip->pipe_out_ana); /* analog outputs */
+ } else {
+ pipe = &(chip->pipe_out_dig); /* digital outputs */
+ }
+ request.message_id = MSG_STREAM_ADD_INPUT_GROUP;
+ stream_count = MIXART_PLAYBACK_STREAMS;
+ }
+
+ /* a new stream is opened and there are already all streams in use */
+ if( (monitoring == 0) && (pipe->references >= stream_count) ) {
+ return NULL;
+ }
+
+ /* pipe is not yet defined */
+ if( pipe->status == PIPE_UNDEFINED ) {
+ int err, i;
+ struct {
+ struct mixart_streaming_group_req sgroup_req;
+ struct mixart_streaming_group sgroup_resp;
+ } *buf;
+
+ snd_printdd("add_ref_pipe audio chip(%d) pcm(%d)\n", chip->chip_idx, pcm_number);
+
+ buf = kmalloc(sizeof(*buf), GFP_KERNEL);
+ if (!buf)
+ return NULL;
+
+ request.uid = (struct mixart_uid){0,0}; /* should be StreamManagerUID, but zero is OK if there is only one ! */
+ request.data = &buf->sgroup_req;
+ request.size = sizeof(buf->sgroup_req);
+
+ memset(&buf->sgroup_req, 0, sizeof(buf->sgroup_req));
+
+ buf->sgroup_req.stream_count = stream_count;
+ buf->sgroup_req.channel_count = 2;
+ buf->sgroup_req.latency = 256;
+ buf->sgroup_req.connector = pipe->uid_left_connector; /* the left connector */
+
+ for (i=0; i<stream_count; i++) {
+ int j;
+ struct mixart_flowinfo *flowinfo;
+ struct mixart_bufferinfo *bufferinfo;
+
+ /* we don't yet know the format, so config 16 bit pcm audio for instance */
+ buf->sgroup_req.stream_info[i].size_max_byte_frame = 1024;
+ buf->sgroup_req.stream_info[i].size_max_sample_frame = 256;
+ buf->sgroup_req.stream_info[i].nb_bytes_max_per_sample = MIXART_FLOAT_P__4_0_TO_HEX; /* is 4.0f */
+
+ /* find the right bufferinfo_array */
+ j = (chip->chip_idx * MIXART_MAX_STREAM_PER_CARD) + (pcm_number * (MIXART_PLAYBACK_STREAMS + MIXART_CAPTURE_STREAMS)) + i;
+ if(capture) j += MIXART_PLAYBACK_STREAMS; /* in the array capture is behind playback */
+
+ buf->sgroup_req.flow_entry[i] = j;
+
+ flowinfo = (struct mixart_flowinfo *)chip->mgr->flowinfo.area;
+ flowinfo[j].bufferinfo_array_phy_address = (u32)chip->mgr->bufferinfo.addr + (j * sizeof(struct mixart_bufferinfo));
+ flowinfo[j].bufferinfo_count = 1; /* 1 will set the miXart to ring-buffer mode ! */
+
+ bufferinfo = (struct mixart_bufferinfo *)chip->mgr->bufferinfo.area;
+ bufferinfo[j].buffer_address = 0; /* buffer is not yet allocated */
+ bufferinfo[j].available_length = 0; /* buffer is not yet allocated */
+
+ /* construct the identifier of the stream buffer received in the interrupts ! */
+ bufferinfo[j].buffer_id = (chip->chip_idx << MIXART_NOTIFY_CARD_OFFSET) + (pcm_number << MIXART_NOTIFY_PCM_OFFSET ) + i;
+ if(capture) {
+ bufferinfo[j].buffer_id |= MIXART_NOTIFY_CAPT_MASK;
+ }
+ }
+
+ err = snd_mixart_send_msg(chip->mgr, &request, sizeof(buf->sgroup_resp), &buf->sgroup_resp);
+ if((err < 0) || (buf->sgroup_resp.status != 0)) {
+ snd_printk(KERN_ERR "error MSG_STREAM_ADD_**PUT_GROUP err=%x stat=%x !\n", err, buf->sgroup_resp.status);
+ kfree(buf);
+ return NULL;
+ }
+
+ pipe->group_uid = buf->sgroup_resp.group; /* id of the pipe, as returned by embedded */
+ pipe->stream_count = buf->sgroup_resp.stream_count;
+ /* pipe->stream_uid[i] = buf->sgroup_resp.stream[i].stream_uid; */
+
+ pipe->status = PIPE_STOPPED;
+ kfree(buf);
+ }
+
+ if(monitoring) pipe->monitoring = 1;
+ else pipe->references++;
+
+ return pipe;
+}
+
+
+int snd_mixart_kill_ref_pipe(struct mixart_mgr *mgr,
+ struct mixart_pipe *pipe, int monitoring)
+{
+ int err = 0;
+
+ if(pipe->status == PIPE_UNDEFINED)
+ return 0;
+
+ if(monitoring)
+ pipe->monitoring = 0;
+ else
+ pipe->references--;
+
+ if((pipe->references <= 0) && (pipe->monitoring == 0)) {
+
+ struct mixart_msg request;
+ struct mixart_delete_group_resp delete_resp;
+
+ /* release the clock */
+ err = mixart_set_clock( mgr, pipe, 0);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "mixart_set_clock(0) return error!\n");
+ }
+
+ /* stop the pipe */
+ err = mixart_set_pipe_state(mgr, pipe, 0);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "error stopping pipe!\n");
+ }
+
+ request.message_id = MSG_STREAM_DELETE_GROUP;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = &pipe->group_uid; /* the streaming group ! */
+ request.size = sizeof(pipe->group_uid);
+
+ /* delete the pipe */
+ err = snd_mixart_send_msg(mgr, &request, sizeof(delete_resp), &delete_resp);
+ if ((err < 0) || (delete_resp.status != 0)) {
+ snd_printk(KERN_ERR "error MSG_STREAM_DELETE_GROUP err(%x), status(%x)\n", err, delete_resp.status);
+ }
+
+ pipe->group_uid = (struct mixart_uid){0,0};
+ pipe->stream_count = 0;
+ pipe->status = PIPE_UNDEFINED;
+ }
+
+ return err;
+}
+
+static int mixart_set_stream_state(struct mixart_stream *stream, int start)
+{
+ struct snd_mixart *chip;
+ struct mixart_stream_state_req stream_state_req;
+ struct mixart_msg request;
+
+ if(!stream->substream)
+ return -EINVAL;
+
+ memset(&stream_state_req, 0, sizeof(stream_state_req));
+ stream_state_req.stream_count = 1;
+ stream_state_req.stream_info.stream_desc.uid_pipe = stream->pipe->group_uid;
+ stream_state_req.stream_info.stream_desc.stream_idx = stream->substream->number;
+
+ if (stream->substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ request.message_id = start ? MSG_STREAM_START_INPUT_STAGE_PACKET : MSG_STREAM_STOP_INPUT_STAGE_PACKET;
+ else
+ request.message_id = start ? MSG_STREAM_START_OUTPUT_STAGE_PACKET : MSG_STREAM_STOP_OUTPUT_STAGE_PACKET;
+
+ request.uid = (struct mixart_uid){0,0};
+ request.data = &stream_state_req;
+ request.size = sizeof(stream_state_req);
+
+ stream->abs_period_elapsed = 0; /* reset stream pos */
+ stream->buf_periods = 0;
+ stream->buf_period_frag = 0;
+
+ chip = snd_pcm_substream_chip(stream->substream);
+
+ return snd_mixart_send_msg_nonblock(chip->mgr, &request);
+}
+
+/*
+ * Trigger callback
+ */
+
+static int snd_mixart_trigger(struct snd_pcm_substream *subs, int cmd)
+{
+ struct mixart_stream *stream = subs->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+
+ snd_printdd("SNDRV_PCM_TRIGGER_START\n");
+
+ /* START_STREAM */
+ if( mixart_set_stream_state(stream, 1) )
+ return -EINVAL;
+
+ stream->status = MIXART_STREAM_STATUS_RUNNING;
+
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+
+ /* STOP_STREAM */
+ if( mixart_set_stream_state(stream, 0) )
+ return -EINVAL;
+
+ stream->status = MIXART_STREAM_STATUS_OPEN;
+
+ snd_printdd("SNDRV_PCM_TRIGGER_STOP\n");
+
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ /* TODO */
+ stream->status = MIXART_STREAM_STATUS_PAUSE;
+ snd_printdd("SNDRV_PCM_PAUSE_PUSH\n");
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /* TODO */
+ stream->status = MIXART_STREAM_STATUS_RUNNING;
+ snd_printdd("SNDRV_PCM_PAUSE_RELEASE\n");
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int mixart_sync_nonblock_events(struct mixart_mgr *mgr)
+{
+ unsigned long timeout = jiffies + HZ;
+ while (atomic_read(&mgr->msg_processed) > 0) {
+ if (time_after(jiffies, timeout)) {
+ snd_printk(KERN_ERR "mixart: cannot process nonblock events!\n");
+ return -EBUSY;
+ }
+ schedule_timeout_uninterruptible(1);
+ }
+ return 0;
+}
+
+/*
+ * prepare callback for all pcms
+ */
+static int snd_mixart_prepare(struct snd_pcm_substream *subs)
+{
+ struct snd_mixart *chip = snd_pcm_substream_chip(subs);
+ struct mixart_stream *stream = subs->runtime->private_data;
+
+ /* TODO de façon non bloquante, réappliquer les hw_params (rate, bits, codec) */
+
+ snd_printdd("snd_mixart_prepare\n");
+
+ mixart_sync_nonblock_events(chip->mgr);
+
+ /* only the first stream can choose the sample rate */
+ /* the further opened streams will be limited to its frequency (see open) */
+ if(chip->mgr->ref_count_rate == 1)
+ chip->mgr->sample_rate = subs->runtime->rate;
+
+ /* set the clock only once (first stream) on the same pipe */
+ if(stream->pipe->references == 1) {
+ if( mixart_set_clock(chip->mgr, stream->pipe, subs->runtime->rate) )
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+static int mixart_set_format(struct mixart_stream *stream, snd_pcm_format_t format)
+{
+ int err;
+ struct snd_mixart *chip;
+ struct mixart_msg request;
+ struct mixart_stream_param_desc stream_param;
+ struct mixart_return_uid resp;
+
+ chip = snd_pcm_substream_chip(stream->substream);
+
+ memset(&stream_param, 0, sizeof(stream_param));
+
+ stream_param.coding_type = CT_LINEAR;
+ stream_param.number_of_channel = stream->channels;
+
+ stream_param.sampling_freq = chip->mgr->sample_rate;
+ if(stream_param.sampling_freq == 0)
+ stream_param.sampling_freq = 44100; /* if frequency not yet defined, use some default */
+
+ switch(format){
+ case SNDRV_PCM_FORMAT_U8:
+ stream_param.sample_type = ST_INTEGER_8;
+ stream_param.sample_size = 8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ stream_param.sample_type = ST_INTEGER_16LE;
+ stream_param.sample_size = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ stream_param.sample_type = ST_INTEGER_16BE;
+ stream_param.sample_size = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ stream_param.sample_type = ST_INTEGER_24LE;
+ stream_param.sample_size = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3BE:
+ stream_param.sample_type = ST_INTEGER_24BE;
+ stream_param.sample_size = 24;
+ break;
+ case SNDRV_PCM_FORMAT_FLOAT_LE:
+ stream_param.sample_type = ST_FLOATING_POINT_32LE;
+ stream_param.sample_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_FLOAT_BE:
+ stream_param.sample_type = ST_FLOATING_POINT_32BE;
+ stream_param.sample_size = 32;
+ break;
+ default:
+ snd_printk(KERN_ERR "error mixart_set_format() : unknown format\n");
+ return -EINVAL;
+ }
+
+ snd_printdd("set SNDRV_PCM_FORMAT sample_type(%d) sample_size(%d) freq(%d) channels(%d)\n",
+ stream_param.sample_type, stream_param.sample_size, stream_param.sampling_freq, stream->channels);
+
+ /* TODO: what else to configure ? */
+ /* stream_param.samples_per_frame = 2; */
+ /* stream_param.bytes_per_frame = 4; */
+ /* stream_param.bytes_per_sample = 2; */
+
+ stream_param.pipe_count = 1; /* set to 1 */
+ stream_param.stream_count = 1; /* set to 1 */
+ stream_param.stream_desc[0].uid_pipe = stream->pipe->group_uid;
+ stream_param.stream_desc[0].stream_idx = stream->substream->number;
+
+ request.message_id = MSG_STREAM_SET_INPUT_STAGE_PARAM;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = &stream_param;
+ request.size = sizeof(stream_param);
+
+ err = snd_mixart_send_msg(chip->mgr, &request, sizeof(resp), &resp);
+ if((err < 0) || resp.error_code) {
+ snd_printk(KERN_ERR "MSG_STREAM_SET_INPUT_STAGE_PARAM err=%x; resp=%x\n", err, resp.error_code);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/*
+ * HW_PARAMS callback for all pcms
+ */
+static int snd_mixart_hw_params(struct snd_pcm_substream *subs,
+ struct snd_pcm_hw_params *hw)
+{
+ struct snd_mixart *chip = snd_pcm_substream_chip(subs);
+ struct mixart_mgr *mgr = chip->mgr;
+ struct mixart_stream *stream = subs->runtime->private_data;
+ snd_pcm_format_t format;
+ int err;
+ int channels;
+
+ /* set up channels */
+ channels = params_channels(hw);
+
+ /* set up format for the stream */
+ format = params_format(hw);
+
+ mutex_lock(&mgr->setup_mutex);
+
+ /* update the stream levels */
+ if( stream->pcm_number <= MIXART_PCM_DIGITAL ) {
+ int is_aes = stream->pcm_number > MIXART_PCM_ANALOG;
+ if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK )
+ mixart_update_playback_stream_level(chip, is_aes, subs->number);
+ else
+ mixart_update_capture_stream_level( chip, is_aes);
+ }
+
+ stream->channels = channels;
+
+ /* set the format to the board */
+ err = mixart_set_format(stream, format);
+ if(err < 0) {
+ mutex_unlock(&mgr->setup_mutex);
+ return err;
+ }
+
+ /* allocate buffer */
+ err = snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw));
+
+ if (err > 0) {
+ struct mixart_bufferinfo *bufferinfo;
+ int i = (chip->chip_idx * MIXART_MAX_STREAM_PER_CARD) + (stream->pcm_number * (MIXART_PLAYBACK_STREAMS+MIXART_CAPTURE_STREAMS)) + subs->number;
+ if( subs->stream == SNDRV_PCM_STREAM_CAPTURE ) {
+ i += MIXART_PLAYBACK_STREAMS; /* in array capture is behind playback */
+ }
+
+ bufferinfo = (struct mixart_bufferinfo *)chip->mgr->bufferinfo.area;
+ bufferinfo[i].buffer_address = subs->runtime->dma_addr;
+ bufferinfo[i].available_length = subs->runtime->dma_bytes;
+ /* bufferinfo[i].buffer_id is already defined */
+
+ snd_printdd("snd_mixart_hw_params(pcm %d) : dma_addr(%x) dma_bytes(%x) subs-number(%d)\n", i,
+ bufferinfo[i].buffer_address,
+ bufferinfo[i].available_length,
+ subs->number);
+ }
+ mutex_unlock(&mgr->setup_mutex);
+
+ return err;
+}
+
+static int snd_mixart_hw_free(struct snd_pcm_substream *subs)
+{
+ struct snd_mixart *chip = snd_pcm_substream_chip(subs);
+ snd_pcm_lib_free_pages(subs);
+ mixart_sync_nonblock_events(chip->mgr);
+ return 0;
+}
+
+
+
+/*
+ * TODO CONFIGURATION SPACE for all pcms, mono pcm must update channels_max
+ */
+static struct snd_pcm_hardware snd_mixart_analog_caps =
+{
+ .info = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = ( SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
+ SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE ),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (32*1024),
+ .period_bytes_min = 256, /* 256 frames U8 mono*/
+ .period_bytes_max = (16*1024),
+ .periods_min = 2,
+ .periods_max = (32*1024/256),
+};
+
+static struct snd_pcm_hardware snd_mixart_digital_caps =
+{
+ .info = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = ( SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
+ SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE ),
+ .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 32000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (32*1024),
+ .period_bytes_min = 256, /* 256 frames U8 mono*/
+ .period_bytes_max = (16*1024),
+ .periods_min = 2,
+ .periods_max = (32*1024/256),
+};
+
+
+static int snd_mixart_playback_open(struct snd_pcm_substream *subs)
+{
+ struct snd_mixart *chip = snd_pcm_substream_chip(subs);
+ struct mixart_mgr *mgr = chip->mgr;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct snd_pcm *pcm = subs->pcm;
+ struct mixart_stream *stream;
+ struct mixart_pipe *pipe;
+ int err = 0;
+ int pcm_number;
+
+ mutex_lock(&mgr->setup_mutex);
+
+ if ( pcm == chip->pcm ) {
+ pcm_number = MIXART_PCM_ANALOG;
+ runtime->hw = snd_mixart_analog_caps;
+ } else {
+ snd_BUG_ON(pcm != chip->pcm_dig);
+ pcm_number = MIXART_PCM_DIGITAL;
+ runtime->hw = snd_mixart_digital_caps;
+ }
+ snd_printdd("snd_mixart_playback_open C%d/P%d/Sub%d\n", chip->chip_idx, pcm_number, subs->number);
+
+ /* get stream info */
+ stream = &(chip->playback_stream[pcm_number][subs->number]);
+
+ if (stream->status != MIXART_STREAM_STATUS_FREE){
+ /* streams in use */
+ snd_printk(KERN_ERR "snd_mixart_playback_open C%d/P%d/Sub%d in use\n", chip->chip_idx, pcm_number, subs->number);
+ err = -EBUSY;
+ goto _exit_open;
+ }
+
+ /* get pipe pointer (out pipe) */
+ pipe = snd_mixart_add_ref_pipe(chip, pcm_number, 0, 0);
+
+ if (pipe == NULL) {
+ err = -EINVAL;
+ goto _exit_open;
+ }
+
+ /* start the pipe if necessary */
+ err = mixart_set_pipe_state(chip->mgr, pipe, 1);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "error starting pipe!\n");
+ snd_mixart_kill_ref_pipe(chip->mgr, pipe, 0);
+ err = -EINVAL;
+ goto _exit_open;
+ }
+
+ stream->pipe = pipe;
+ stream->pcm_number = pcm_number;
+ stream->status = MIXART_STREAM_STATUS_OPEN;
+ stream->substream = subs;
+ stream->channels = 0; /* not configured yet */
+
+ runtime->private_data = stream;
+
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64);
+
+ /* if a sample rate is already used, another stream cannot change */
+ if(mgr->ref_count_rate++) {
+ if(mgr->sample_rate) {
+ runtime->hw.rate_min = runtime->hw.rate_max = mgr->sample_rate;
+ }
+ }
+
+ _exit_open:
+ mutex_unlock(&mgr->setup_mutex);
+
+ return err;
+}
+
+
+static int snd_mixart_capture_open(struct snd_pcm_substream *subs)
+{
+ struct snd_mixart *chip = snd_pcm_substream_chip(subs);
+ struct mixart_mgr *mgr = chip->mgr;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct snd_pcm *pcm = subs->pcm;
+ struct mixart_stream *stream;
+ struct mixart_pipe *pipe;
+ int err = 0;
+ int pcm_number;
+
+ mutex_lock(&mgr->setup_mutex);
+
+ if ( pcm == chip->pcm ) {
+ pcm_number = MIXART_PCM_ANALOG;
+ runtime->hw = snd_mixart_analog_caps;
+ } else {
+ snd_BUG_ON(pcm != chip->pcm_dig);
+ pcm_number = MIXART_PCM_DIGITAL;
+ runtime->hw = snd_mixart_digital_caps;
+ }
+
+ runtime->hw.channels_min = 2; /* for instance, no mono */
+
+ snd_printdd("snd_mixart_capture_open C%d/P%d/Sub%d\n", chip->chip_idx, pcm_number, subs->number);
+
+ /* get stream info */
+ stream = &(chip->capture_stream[pcm_number]);
+
+ if (stream->status != MIXART_STREAM_STATUS_FREE){
+ /* streams in use */
+ snd_printk(KERN_ERR "snd_mixart_capture_open C%d/P%d/Sub%d in use\n", chip->chip_idx, pcm_number, subs->number);
+ err = -EBUSY;
+ goto _exit_open;
+ }
+
+ /* get pipe pointer (in pipe) */
+ pipe = snd_mixart_add_ref_pipe(chip, pcm_number, 1, 0);
+
+ if (pipe == NULL) {
+ err = -EINVAL;
+ goto _exit_open;
+ }
+
+ /* start the pipe if necessary */
+ err = mixart_set_pipe_state(chip->mgr, pipe, 1);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "error starting pipe!\n");
+ snd_mixart_kill_ref_pipe(chip->mgr, pipe, 0);
+ err = -EINVAL;
+ goto _exit_open;
+ }
+
+ stream->pipe = pipe;
+ stream->pcm_number = pcm_number;
+ stream->status = MIXART_STREAM_STATUS_OPEN;
+ stream->substream = subs;
+ stream->channels = 0; /* not configured yet */
+
+ runtime->private_data = stream;
+
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 64);
+
+ /* if a sample rate is already used, another stream cannot change */
+ if(mgr->ref_count_rate++) {
+ if(mgr->sample_rate) {
+ runtime->hw.rate_min = runtime->hw.rate_max = mgr->sample_rate;
+ }
+ }
+
+ _exit_open:
+ mutex_unlock(&mgr->setup_mutex);
+
+ return err;
+}
+
+
+
+static int snd_mixart_close(struct snd_pcm_substream *subs)
+{
+ struct snd_mixart *chip = snd_pcm_substream_chip(subs);
+ struct mixart_mgr *mgr = chip->mgr;
+ struct mixart_stream *stream = subs->runtime->private_data;
+
+ mutex_lock(&mgr->setup_mutex);
+
+ snd_printdd("snd_mixart_close C%d/P%d/Sub%d\n", chip->chip_idx, stream->pcm_number, subs->number);
+
+ /* sample rate released */
+ if(--mgr->ref_count_rate == 0) {
+ mgr->sample_rate = 0;
+ }
+
+ /* delete pipe */
+ if (snd_mixart_kill_ref_pipe(mgr, stream->pipe, 0 ) < 0) {
+
+ snd_printk(KERN_ERR "error snd_mixart_kill_ref_pipe C%dP%d\n", chip->chip_idx, stream->pcm_number);
+ }
+
+ stream->pipe = NULL;
+ stream->status = MIXART_STREAM_STATUS_FREE;
+ stream->substream = NULL;
+
+ mutex_unlock(&mgr->setup_mutex);
+ return 0;
+}
+
+
+static snd_pcm_uframes_t snd_mixart_stream_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct mixart_stream *stream = runtime->private_data;
+
+ return (snd_pcm_uframes_t)((stream->buf_periods * runtime->period_size) + stream->buf_period_frag);
+}
+
+
+
+static struct snd_pcm_ops snd_mixart_playback_ops = {
+ .open = snd_mixart_playback_open,
+ .close = snd_mixart_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = snd_mixart_prepare,
+ .hw_params = snd_mixart_hw_params,
+ .hw_free = snd_mixart_hw_free,
+ .trigger = snd_mixart_trigger,
+ .pointer = snd_mixart_stream_pointer,
+};
+
+static struct snd_pcm_ops snd_mixart_capture_ops = {
+ .open = snd_mixart_capture_open,
+ .close = snd_mixart_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = snd_mixart_prepare,
+ .hw_params = snd_mixart_hw_params,
+ .hw_free = snd_mixart_hw_free,
+ .trigger = snd_mixart_trigger,
+ .pointer = snd_mixart_stream_pointer,
+};
+
+static void preallocate_buffers(struct snd_mixart *chip, struct snd_pcm *pcm)
+{
+#if 0
+ struct snd_pcm_substream *subs;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ int idx = 0;
+ for (subs = pcm->streams[stream].substream; subs; subs = subs->next, idx++)
+ /* set up the unique device id with the chip index */
+ subs->dma_device.id = subs->pcm->device << 16 |
+ subs->stream << 8 | (subs->number + 1) |
+ (chip->chip_idx + 1) << 24;
+ }
+#endif
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->mgr->pci), 32*1024, 32*1024);
+}
+
+/*
+ */
+static int snd_mixart_pcm_analog(struct snd_mixart *chip)
+{
+ int err;
+ struct snd_pcm *pcm;
+ char name[32];
+
+ sprintf(name, "miXart analog %d", chip->chip_idx);
+ if ((err = snd_pcm_new(chip->card, name, MIXART_PCM_ANALOG,
+ MIXART_PLAYBACK_STREAMS,
+ MIXART_CAPTURE_STREAMS, &pcm)) < 0) {
+ snd_printk(KERN_ERR "cannot create the analog pcm %d\n", chip->chip_idx);
+ return err;
+ }
+
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_mixart_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_mixart_capture_ops);
+
+ pcm->info_flags = 0;
+ strcpy(pcm->name, name);
+
+ preallocate_buffers(chip, pcm);
+
+ chip->pcm = pcm;
+ return 0;
+}
+
+
+/*
+ */
+static int snd_mixart_pcm_digital(struct snd_mixart *chip)
+{
+ int err;
+ struct snd_pcm *pcm;
+ char name[32];
+
+ sprintf(name, "miXart AES/EBU %d", chip->chip_idx);
+ if ((err = snd_pcm_new(chip->card, name, MIXART_PCM_DIGITAL,
+ MIXART_PLAYBACK_STREAMS,
+ MIXART_CAPTURE_STREAMS, &pcm)) < 0) {
+ snd_printk(KERN_ERR "cannot create the digital pcm %d\n", chip->chip_idx);
+ return err;
+ }
+
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_mixart_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_mixart_capture_ops);
+
+ pcm->info_flags = 0;
+ strcpy(pcm->name, name);
+
+ preallocate_buffers(chip, pcm);
+
+ chip->pcm_dig = pcm;
+ return 0;
+}
+
+static int snd_mixart_chip_free(struct snd_mixart *chip)
+{
+ kfree(chip);
+ return 0;
+}
+
+static int snd_mixart_chip_dev_free(struct snd_device *device)
+{
+ struct snd_mixart *chip = device->device_data;
+ return snd_mixart_chip_free(chip);
+}
+
+
+/*
+ */
+static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card *card, int idx)
+{
+ int err;
+ struct snd_mixart *chip;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_mixart_chip_dev_free,
+ };
+
+ mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (! chip) {
+ snd_printk(KERN_ERR "cannot allocate chip\n");
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->chip_idx = idx;
+ chip->mgr = mgr;
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_mixart_chip_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &mgr->pci->dev);
+
+ return 0;
+}
+
+int snd_mixart_create_pcm(struct snd_mixart* chip)
+{
+ int err;
+
+ err = snd_mixart_pcm_analog(chip);
+ if (err < 0)
+ return err;
+
+ if(chip->mgr->board_type == MIXART_DAUGHTER_TYPE_AES) {
+
+ err = snd_mixart_pcm_digital(chip);
+ if (err < 0)
+ return err;
+ }
+ return err;
+}
+
+
+/*
+ * release all the cards assigned to a manager instance
+ */
+static int snd_mixart_free(struct mixart_mgr *mgr)
+{
+ unsigned int i;
+
+ for (i = 0; i < mgr->num_cards; i++) {
+ if (mgr->chip[i])
+ snd_card_free(mgr->chip[i]->card);
+ }
+
+ /* stop mailbox */
+ snd_mixart_exit_mailbox(mgr);
+
+ /* release irq */
+ if (mgr->irq >= 0)
+ free_irq(mgr->irq, mgr);
+
+ /* reset board if some firmware was loaded */
+ if(mgr->dsp_loaded) {
+ snd_mixart_reset_board(mgr);
+ snd_printdd("reset miXart !\n");
+ }
+
+ /* release the i/o ports */
+ for (i = 0; i < 2; i++) {
+ if (mgr->mem[i].virt)
+ iounmap(mgr->mem[i].virt);
+ }
+ pci_release_regions(mgr->pci);
+
+ /* free flowarray */
+ if(mgr->flowinfo.area) {
+ snd_dma_free_pages(&mgr->flowinfo);
+ mgr->flowinfo.area = NULL;
+ }
+ /* free bufferarray */
+ if(mgr->bufferinfo.area) {
+ snd_dma_free_pages(&mgr->bufferinfo);
+ mgr->bufferinfo.area = NULL;
+ }
+
+ pci_disable_device(mgr->pci);
+ kfree(mgr);
+ return 0;
+}
+
+/*
+ * proc interface
+ */
+static long long snd_mixart_BA0_llseek(struct snd_info_entry *entry,
+ void *private_file_data,
+ struct file *file,
+ long long offset,
+ int orig)
+{
+ offset = offset & ~3; /* 4 bytes aligned */
+
+ switch(orig) {
+ case SEEK_SET:
+ file->f_pos = offset;
+ break;
+ case SEEK_CUR:
+ file->f_pos += offset;
+ break;
+ case SEEK_END: /* offset is negative */
+ file->f_pos = MIXART_BA0_SIZE + offset;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if(file->f_pos > MIXART_BA0_SIZE)
+ file->f_pos = MIXART_BA0_SIZE;
+ return file->f_pos;
+}
+
+static long long snd_mixart_BA1_llseek(struct snd_info_entry *entry,
+ void *private_file_data,
+ struct file *file,
+ long long offset,
+ int orig)
+{
+ offset = offset & ~3; /* 4 bytes aligned */
+
+ switch(orig) {
+ case SEEK_SET:
+ file->f_pos = offset;
+ break;
+ case SEEK_CUR:
+ file->f_pos += offset;
+ break;
+ case SEEK_END: /* offset is negative */
+ file->f_pos = MIXART_BA1_SIZE + offset;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if(file->f_pos > MIXART_BA1_SIZE)
+ file->f_pos = MIXART_BA1_SIZE;
+ return file->f_pos;
+}
+
+/*
+ mixart_BA0 proc interface for BAR 0 - read callback
+ */
+static long snd_mixart_BA0_read(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, char __user *buf,
+ unsigned long count, unsigned long pos)
+{
+ struct mixart_mgr *mgr = entry->private_data;
+
+ count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+ if(count <= 0)
+ return 0;
+ if(pos + count > MIXART_BA0_SIZE)
+ count = (long)(MIXART_BA0_SIZE - pos);
+ if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count))
+ return -EFAULT;
+ return count;
+}
+
+/*
+ mixart_BA1 proc interface for BAR 1 - read callback
+ */
+static long snd_mixart_BA1_read(struct snd_info_entry *entry, void *file_private_data,
+ struct file *file, char __user *buf,
+ unsigned long count, unsigned long pos)
+{
+ struct mixart_mgr *mgr = entry->private_data;
+
+ count = count & ~3; /* make sure the read size is a multiple of 4 bytes */
+ if(count <= 0)
+ return 0;
+ if(pos + count > MIXART_BA1_SIZE)
+ count = (long)(MIXART_BA1_SIZE - pos);
+ if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count))
+ return -EFAULT;
+ return count;
+}
+
+static struct snd_info_entry_ops snd_mixart_proc_ops_BA0 = {
+ .read = snd_mixart_BA0_read,
+ .llseek = snd_mixart_BA0_llseek
+};
+
+static struct snd_info_entry_ops snd_mixart_proc_ops_BA1 = {
+ .read = snd_mixart_BA1_read,
+ .llseek = snd_mixart_BA1_llseek
+};
+
+
+static void snd_mixart_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_mixart *chip = entry->private_data;
+ u32 ref;
+
+ snd_iprintf(buffer, "Digigram miXart (alsa card %d)\n\n", chip->chip_idx);
+
+ /* stats available when embedded OS is running */
+ if (chip->mgr->dsp_loaded & ( 1 << MIXART_MOTHERBOARD_ELF_INDEX)) {
+ snd_iprintf(buffer, "- hardware -\n");
+ switch (chip->mgr->board_type ) {
+ case MIXART_DAUGHTER_TYPE_NONE : snd_iprintf(buffer, "\tmiXart8 (no daughter board)\n\n"); break;
+ case MIXART_DAUGHTER_TYPE_AES : snd_iprintf(buffer, "\tmiXart8 AES/EBU\n\n"); break;
+ case MIXART_DAUGHTER_TYPE_COBRANET : snd_iprintf(buffer, "\tmiXart8 Cobranet\n\n"); break;
+ default: snd_iprintf(buffer, "\tUNKNOWN!\n\n"); break;
+ }
+
+ snd_iprintf(buffer, "- system load -\n");
+
+ /* get perf reference */
+
+ ref = readl_be( MIXART_MEM( chip->mgr, MIXART_PSEUDOREG_PERF_SYSTEM_LOAD_OFFSET));
+
+ if (ref) {
+ u32 mailbox = 100 * readl_be( MIXART_MEM( chip->mgr, MIXART_PSEUDOREG_PERF_MAILBX_LOAD_OFFSET)) / ref;
+ u32 streaming = 100 * readl_be( MIXART_MEM( chip->mgr, MIXART_PSEUDOREG_PERF_STREAM_LOAD_OFFSET)) / ref;
+ u32 interr = 100 * readl_be( MIXART_MEM( chip->mgr, MIXART_PSEUDOREG_PERF_INTERR_LOAD_OFFSET)) / ref;
+
+ snd_iprintf(buffer, "\tstreaming : %d\n", streaming);
+ snd_iprintf(buffer, "\tmailbox : %d\n", mailbox);
+ snd_iprintf(buffer, "\tinterrups handling : %d\n\n", interr);
+ }
+ } /* endif elf loaded */
+}
+
+static void __devinit snd_mixart_proc_init(struct snd_mixart *chip)
+{
+ struct snd_info_entry *entry;
+
+ /* text interface to read perf and temp meters */
+ if (! snd_card_proc_new(chip->card, "board_info", &entry)) {
+ entry->private_data = chip;
+ entry->c.text.read = snd_mixart_proc_read;
+ }
+
+ if (! snd_card_proc_new(chip->card, "mixart_BA0", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = chip->mgr;
+ entry->c.ops = &snd_mixart_proc_ops_BA0;
+ entry->size = MIXART_BA0_SIZE;
+ }
+ if (! snd_card_proc_new(chip->card, "mixart_BA1", &entry)) {
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = chip->mgr;
+ entry->c.ops = &snd_mixart_proc_ops_BA1;
+ entry->size = MIXART_BA1_SIZE;
+ }
+}
+/* end of proc interface */
+
+
+/*
+ * probe function - creates the card manager
+ */
+static int __devinit snd_mixart_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct mixart_mgr *mgr;
+ unsigned int i;
+ int err;
+ size_t size;
+
+ /*
+ */
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (! enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ pci_set_master(pci);
+
+ /* check if we can restrict PCI DMA transfers to 32 bits */
+ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ /*
+ */
+ mgr = kzalloc(sizeof(*mgr), GFP_KERNEL);
+ if (! mgr) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ mgr->pci = pci;
+ mgr->irq = -1;
+
+ /* resource assignment */
+ if ((err = pci_request_regions(pci, CARD_NAME)) < 0) {
+ kfree(mgr);
+ pci_disable_device(pci);
+ return err;
+ }
+ for (i = 0; i < 2; i++) {
+ mgr->mem[i].phys = pci_resource_start(pci, i);
+ mgr->mem[i].virt = pci_ioremap_bar(pci, i);
+ if (!mgr->mem[i].virt) {
+ printk(KERN_ERR "unable to remap resource 0x%lx\n",
+ mgr->mem[i].phys);
+ snd_mixart_free(mgr);
+ return -EBUSY;
+ }
+ }
+
+ if (request_irq(pci->irq, snd_mixart_interrupt, IRQF_SHARED,
+ CARD_NAME, mgr)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_mixart_free(mgr);
+ return -EBUSY;
+ }
+ mgr->irq = pci->irq;
+
+ sprintf(mgr->shortname, "Digigram miXart");
+ sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, irq %i", mgr->shortname, mgr->mem[0].phys, mgr->mem[1].phys, mgr->irq);
+
+ /* ISR spinlock */
+ spin_lock_init(&mgr->lock);
+
+ /* init mailbox */
+ mgr->msg_fifo_readptr = 0;
+ mgr->msg_fifo_writeptr = 0;
+
+ spin_lock_init(&mgr->msg_lock);
+ mutex_init(&mgr->msg_mutex);
+ init_waitqueue_head(&mgr->msg_sleep);
+ atomic_set(&mgr->msg_processed, 0);
+
+ /* init setup mutex*/
+ mutex_init(&mgr->setup_mutex);
+
+ /* init message taslket */
+ tasklet_init(&mgr->msg_taskq, snd_mixart_msg_tasklet, (unsigned long) mgr);
+
+ /* card assignment */
+ mgr->num_cards = MIXART_MAX_CARDS; /* 4 FIXME: configurable? */
+ for (i = 0; i < mgr->num_cards; i++) {
+ struct snd_card *card;
+ char tmpid[16];
+ int idx;
+
+ if (index[dev] < 0)
+ idx = index[dev];
+ else
+ idx = index[dev] + i;
+ snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : "MIXART", i);
+ card = snd_card_new(idx, tmpid, THIS_MODULE, 0);
+
+ if (! card) {
+ snd_printk(KERN_ERR "cannot allocate the card %d\n", i);
+ snd_mixart_free(mgr);
+ return -ENOMEM;
+ }
+
+ strcpy(card->driver, CARD_NAME);
+ sprintf(card->shortname, "%s [PCM #%d]", mgr->shortname, i);
+ sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i);
+
+ if ((err = snd_mixart_create(mgr, card, i)) < 0) {
+ snd_mixart_free(mgr);
+ return err;
+ }
+
+ if(i==0) {
+ /* init proc interface only for chip0 */
+ snd_mixart_proc_init(mgr->chip[i]);
+ }
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_mixart_free(mgr);
+ return err;
+ }
+ }
+
+ /* init firmware status (mgr->dsp_loaded reset in hwdep_new) */
+ mgr->board_type = MIXART_DAUGHTER_TYPE_NONE;
+
+ /* create array of streaminfo */
+ size = PAGE_ALIGN( (MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS *
+ sizeof(struct mixart_flowinfo)) );
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, &mgr->flowinfo) < 0) {
+ snd_mixart_free(mgr);
+ return -ENOMEM;
+ }
+ /* init streaminfo_array */
+ memset(mgr->flowinfo.area, 0, size);
+
+ /* create array of bufferinfo */
+ size = PAGE_ALIGN( (MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS *
+ sizeof(struct mixart_bufferinfo)) );
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, &mgr->bufferinfo) < 0) {
+ snd_mixart_free(mgr);
+ return -ENOMEM;
+ }
+ /* init bufferinfo_array */
+ memset(mgr->bufferinfo.area, 0, size);
+
+ /* set up firmware */
+ err = snd_mixart_setup_firmware(mgr);
+ if (err < 0) {
+ snd_mixart_free(mgr);
+ return err;
+ }
+
+ pci_set_drvdata(pci, mgr);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_mixart_remove(struct pci_dev *pci)
+{
+ snd_mixart_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Digigram miXart",
+ .id_table = snd_mixart_ids,
+ .probe = snd_mixart_probe,
+ .remove = __devexit_p(snd_mixart_remove),
+};
+
+static int __init alsa_card_mixart_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_mixart_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_mixart_init)
+module_exit(alsa_card_mixart_exit)
diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h
new file mode 100644
index 0000000..561634d
--- /dev/null
+++ b/sound/pci/mixart/mixart.h
@@ -0,0 +1,228 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * main header file
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_MIXART_H
+#define __SOUND_MIXART_H
+
+#include <linux/interrupt.h>
+#include <linux/mutex.h>
+#include <sound/pcm.h>
+
+#define MIXART_DRIVER_VERSION 0x000100 /* 0.1.0 */
+
+
+/*
+ */
+
+struct mixart_uid {
+ u32 object_id;
+ u32 desc;
+};
+
+struct mem_area {
+ unsigned long phys;
+ void __iomem *virt;
+ struct resource *res;
+};
+
+
+struct mixart_route {
+ unsigned char connected;
+ unsigned char phase_inv;
+ int volume;
+};
+
+
+/* firmware status codes */
+#define MIXART_MOTHERBOARD_XLX_INDEX 0
+#define MIXART_MOTHERBOARD_ELF_INDEX 1
+#define MIXART_AESEBUBOARD_XLX_INDEX 2
+#define MIXART_HARDW_FILES_MAX_INDEX 3 /* xilinx, elf, AESEBU xilinx */
+
+#define MIXART_MAX_CARDS 4
+#define MSG_FIFO_SIZE 16
+
+#define MIXART_MAX_PHYS_CONNECTORS (MIXART_MAX_CARDS * 2 * 2) /* 4 * stereo * (analog+digital) */
+
+struct mixart_mgr {
+ unsigned int num_cards;
+ struct snd_mixart *chip[MIXART_MAX_CARDS];
+
+ struct pci_dev *pci;
+
+ int irq;
+
+ /* memory-maps */
+ struct mem_area mem[2];
+
+ /* share the name */
+ char shortname[32]; /* short name of this soundcard */
+ char longname[80]; /* name of this soundcard */
+
+ /* message tasklet */
+ struct tasklet_struct msg_taskq;
+
+ /* one and only blocking message or notification may be pending */
+ u32 pending_event;
+ wait_queue_head_t msg_sleep;
+
+ /* messages stored for tasklet */
+ u32 msg_fifo[MSG_FIFO_SIZE];
+ int msg_fifo_readptr;
+ int msg_fifo_writeptr;
+ atomic_t msg_processed; /* number of messages to be processed in takslet */
+
+ spinlock_t lock; /* interrupt spinlock */
+ spinlock_t msg_lock; /* mailbox spinlock */
+ struct mutex msg_mutex; /* mutex for blocking_requests */
+
+ struct mutex setup_mutex; /* mutex used in hw_params, open and close */
+
+ /* hardware interface */
+ unsigned int dsp_loaded; /* bit flags of loaded dsp indices */
+ unsigned int board_type; /* read from embedded once elf file is loaded, 250 = miXart8, 251 = with AES, 252 = with Cobranet */
+
+ struct snd_dma_buffer flowinfo;
+ struct snd_dma_buffer bufferinfo;
+
+ struct mixart_uid uid_console_manager;
+ int sample_rate;
+ int ref_count_rate;
+
+ struct mutex mixer_mutex; /* mutex for mixer */
+
+};
+
+
+#define MIXART_STREAM_STATUS_FREE 0
+#define MIXART_STREAM_STATUS_OPEN 1
+#define MIXART_STREAM_STATUS_RUNNING 2
+#define MIXART_STREAM_STATUS_DRAINING 3
+#define MIXART_STREAM_STATUS_PAUSE 4
+
+#define MIXART_PLAYBACK_STREAMS 4
+#define MIXART_CAPTURE_STREAMS 1
+
+#define MIXART_PCM_ANALOG 0
+#define MIXART_PCM_DIGITAL 1
+#define MIXART_PCM_TOTAL 2
+
+#define MIXART_MAX_STREAM_PER_CARD (MIXART_PCM_TOTAL * (MIXART_PLAYBACK_STREAMS + MIXART_CAPTURE_STREAMS) )
+
+
+#define MIXART_NOTIFY_CARD_MASK 0xF000
+#define MIXART_NOTIFY_CARD_OFFSET 12
+#define MIXART_NOTIFY_PCM_MASK 0x0F00
+#define MIXART_NOTIFY_PCM_OFFSET 8
+#define MIXART_NOTIFY_CAPT_MASK 0x0080
+#define MIXART_NOTIFY_SUBS_MASK 0x007F
+
+
+struct mixart_stream {
+ struct snd_pcm_substream *substream;
+ struct mixart_pipe *pipe;
+ int pcm_number;
+
+ int status; /* nothing, running, draining */
+
+ u64 abs_period_elapsed; /* last absolute stream position where period_elapsed was called (multiple of runtime->period_size) */
+ u32 buf_periods; /* periods counter in the buffer (< runtime->periods) */
+ u32 buf_period_frag; /* defines with buf_period_pos the exact position in the buffer (< runtime->period_size) */
+
+ int channels;
+};
+
+
+enum mixart_pipe_status {
+ PIPE_UNDEFINED,
+ PIPE_STOPPED,
+ PIPE_RUNNING,
+ PIPE_CLOCK_SET
+};
+
+struct mixart_pipe {
+ struct mixart_uid group_uid; /* id of the pipe, as returned by embedded */
+ int stream_count;
+ struct mixart_uid uid_left_connector; /* UID's for the audio connectors */
+ struct mixart_uid uid_right_connector;
+ enum mixart_pipe_status status;
+ int references; /* number of subs openned */
+ int monitoring; /* pipe used for monitoring issue */
+};
+
+
+struct snd_mixart {
+ struct snd_card *card;
+ struct mixart_mgr *mgr;
+ int chip_idx; /* zero based */
+ struct snd_hwdep *hwdep; /* DSP loader, only for the first card */
+
+ struct snd_pcm *pcm; /* PCM analog i/o */
+ struct snd_pcm *pcm_dig; /* PCM digital i/o */
+
+ /* allocate stereo pipe for instance */
+ struct mixart_pipe pipe_in_ana;
+ struct mixart_pipe pipe_out_ana;
+
+ /* if AES/EBU daughter board is available, additional pipes possible on pcm_dig */
+ struct mixart_pipe pipe_in_dig;
+ struct mixart_pipe pipe_out_dig;
+
+ struct mixart_stream playback_stream[MIXART_PCM_TOTAL][MIXART_PLAYBACK_STREAMS]; /* 0 = pcm, 1 = pcm_dig */
+ struct mixart_stream capture_stream[MIXART_PCM_TOTAL]; /* 0 = pcm, 1 = pcm_dig */
+
+ /* UID's for the physical io's */
+ struct mixart_uid uid_out_analog_physio;
+ struct mixart_uid uid_in_analog_physio;
+
+ int analog_playback_active[2]; /* Mixer : Master Playback active (!mute) */
+ int analog_playback_volume[2]; /* Mixer : Master Playback Volume */
+ int analog_capture_volume[2]; /* Mixer : Master Capture Volume */
+ int digital_playback_active[2*MIXART_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Active [(analog+AES output)*streams][stereo]*/
+ int digital_playback_volume[2*MIXART_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Volume [(analog+AES output)*streams][stereo]*/
+ int digital_capture_volume[2][2]; /* Mixer : Digital Capture Volume [analog+AES output][stereo] */
+ int monitoring_active[2]; /* Mixer : Monitoring Active */
+ int monitoring_volume[2]; /* Mixer : Monitoring Volume */
+};
+
+struct mixart_bufferinfo
+{
+ u32 buffer_address;
+ u32 reserved[5];
+ u32 available_length;
+ u32 buffer_id;
+};
+
+struct mixart_flowinfo
+{
+ u32 bufferinfo_array_phy_address;
+ u32 reserved[11];
+ u32 bufferinfo_count;
+ u32 capture;
+};
+
+/* exported */
+int snd_mixart_create_pcm(struct snd_mixart * chip);
+struct mixart_pipe *snd_mixart_add_ref_pipe(struct snd_mixart *chip, int pcm_number, int capture, int monitoring);
+int snd_mixart_kill_ref_pipe(struct mixart_mgr *mgr, struct mixart_pipe *pipe, int monitoring);
+
+#endif /* __SOUND_MIXART_H */
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
new file mode 100644
index 0000000..b9a06c2
--- /dev/null
+++ b/sound/pci/mixart/mixart_core.c
@@ -0,0 +1,598 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * low level interface with interrupt handling and mail box implementation
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/interrupt.h>
+#include <linux/mutex.h>
+
+#include <asm/io.h>
+#include <sound/core.h>
+#include "mixart.h"
+#include "mixart_hwdep.h"
+#include "mixart_core.h"
+
+
+#define MSG_TIMEOUT_JIFFIES (400 * HZ) / 1000 /* 400 ms */
+
+#define MSG_DESCRIPTOR_SIZE 0x24
+#define MSG_HEADER_SIZE (MSG_DESCRIPTOR_SIZE + 4)
+
+#define MSG_DEFAULT_SIZE 512
+
+#define MSG_TYPE_MASK 0x00000003 /* mask for following types */
+#define MSG_TYPE_NOTIFY 0 /* embedded -> driver (only notification, do not get_msg() !) */
+#define MSG_TYPE_COMMAND 1 /* driver <-> embedded (a command has no answer) */
+#define MSG_TYPE_REQUEST 2 /* driver -> embedded (request will get an answer back) */
+#define MSG_TYPE_ANSWER 3 /* embedded -> driver */
+#define MSG_CANCEL_NOTIFY_MASK 0x80000000 /* this bit is set for a notification that has been canceled */
+
+
+static int retrieve_msg_frame(struct mixart_mgr *mgr, u32 *msg_frame)
+{
+ /* read the message frame fifo */
+ u32 headptr, tailptr;
+
+ tailptr = readl_be(MIXART_MEM(mgr, MSG_OUTBOUND_POST_TAIL));
+ headptr = readl_be(MIXART_MEM(mgr, MSG_OUTBOUND_POST_HEAD));
+
+ if (tailptr == headptr)
+ return 0; /* no message posted */
+
+ if (tailptr < MSG_OUTBOUND_POST_STACK)
+ return 0; /* error */
+ if (tailptr >= MSG_OUTBOUND_POST_STACK + MSG_BOUND_STACK_SIZE)
+ return 0; /* error */
+
+ *msg_frame = readl_be(MIXART_MEM(mgr, tailptr));
+
+ /* increment the tail index */
+ tailptr += 4;
+ if( tailptr >= (MSG_OUTBOUND_POST_STACK+MSG_BOUND_STACK_SIZE) )
+ tailptr = MSG_OUTBOUND_POST_STACK;
+ writel_be(tailptr, MIXART_MEM(mgr, MSG_OUTBOUND_POST_TAIL));
+
+ return 1;
+}
+
+static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp,
+ u32 msg_frame_address )
+{
+ unsigned long flags;
+ u32 headptr;
+ u32 size;
+ int err;
+#ifndef __BIG_ENDIAN
+ unsigned int i;
+#endif
+
+ spin_lock_irqsave(&mgr->msg_lock, flags);
+ err = 0;
+
+ /* copy message descriptor from miXart to driver */
+ size = readl_be(MIXART_MEM(mgr, msg_frame_address)); /* size of descriptor + response */
+ resp->message_id = readl_be(MIXART_MEM(mgr, msg_frame_address + 4)); /* dwMessageID */
+ resp->uid.object_id = readl_be(MIXART_MEM(mgr, msg_frame_address + 8)); /* uidDest */
+ resp->uid.desc = readl_be(MIXART_MEM(mgr, msg_frame_address + 12)); /* */
+
+ if( (size < MSG_DESCRIPTOR_SIZE) || (resp->size < (size - MSG_DESCRIPTOR_SIZE))) {
+ err = -EINVAL;
+ snd_printk(KERN_ERR "problem with response size = %d\n", size);
+ goto _clean_exit;
+ }
+ size -= MSG_DESCRIPTOR_SIZE;
+
+ memcpy_fromio(resp->data, MIXART_MEM(mgr, msg_frame_address + MSG_HEADER_SIZE ), size);
+ resp->size = size;
+
+ /* swap if necessary */
+#ifndef __BIG_ENDIAN
+ size /= 4; /* u32 size */
+ for(i=0; i < size; i++) {
+ ((u32*)resp->data)[i] = be32_to_cpu(((u32*)resp->data)[i]);
+ }
+#endif
+
+ /*
+ * free message frame address
+ */
+ headptr = readl_be(MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD));
+
+ if( (headptr < MSG_OUTBOUND_FREE_STACK) || ( headptr >= (MSG_OUTBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE))) {
+ err = -EINVAL;
+ goto _clean_exit;
+ }
+
+ /* give address back to outbound fifo */
+ writel_be(msg_frame_address, MIXART_MEM(mgr, headptr));
+
+ /* increment the outbound free head */
+ headptr += 4;
+ if( headptr >= (MSG_OUTBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE) )
+ headptr = MSG_OUTBOUND_FREE_STACK;
+
+ writel_be(headptr, MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD));
+
+ _clean_exit:
+ spin_unlock_irqrestore(&mgr->msg_lock, flags);
+
+ return err;
+}
+
+
+/*
+ * send a message to miXart. return: the msg_frame used for this message
+ */
+/* call with mgr->msg_lock held! */
+static int send_msg( struct mixart_mgr *mgr,
+ struct mixart_msg *msg,
+ int max_answersize,
+ int mark_pending,
+ u32 *msg_event)
+{
+ u32 headptr, tailptr;
+ u32 msg_frame_address;
+ int err, i;
+
+ if (snd_BUG_ON(msg->size % 4))
+ return -EINVAL;
+
+ err = 0;
+
+ /* get message frame address */
+ tailptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_FREE_TAIL));
+ headptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_FREE_HEAD));
+
+ if (tailptr == headptr) {
+ snd_printk(KERN_ERR "error: no message frame available\n");
+ return -EBUSY;
+ }
+
+ if( (tailptr < MSG_INBOUND_FREE_STACK) || (tailptr >= (MSG_INBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE))) {
+ return -EINVAL;
+ }
+
+ msg_frame_address = readl_be(MIXART_MEM(mgr, tailptr));
+ writel(0, MIXART_MEM(mgr, tailptr)); /* set address to zero on this fifo position */
+
+ /* increment the inbound free tail */
+ tailptr += 4;
+ if( tailptr >= (MSG_INBOUND_FREE_STACK+MSG_BOUND_STACK_SIZE) )
+ tailptr = MSG_INBOUND_FREE_STACK;
+
+ writel_be(tailptr, MIXART_MEM(mgr, MSG_INBOUND_FREE_TAIL));
+
+ /* TODO : use memcpy_toio() with intermediate buffer to copy the message */
+
+ /* copy message descriptor to card memory */
+ writel_be( msg->size + MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address) ); /* size of descriptor + request */
+ writel_be( msg->message_id , MIXART_MEM(mgr, msg_frame_address + 4) ); /* dwMessageID */
+ writel_be( msg->uid.object_id, MIXART_MEM(mgr, msg_frame_address + 8) ); /* uidDest */
+ writel_be( msg->uid.desc, MIXART_MEM(mgr, msg_frame_address + 12) ); /* */
+ writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 16) ); /* SizeHeader */
+ writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 20) ); /* OffsetDLL_T16 */
+ writel_be( msg->size, MIXART_MEM(mgr, msg_frame_address + 24) ); /* SizeDLL_T16 */
+ writel_be( MSG_DESCRIPTOR_SIZE, MIXART_MEM(mgr, msg_frame_address + 28) ); /* OffsetDLL_DRV */
+ writel_be( 0, MIXART_MEM(mgr, msg_frame_address + 32) ); /* SizeDLL_DRV */
+ writel_be( MSG_DESCRIPTOR_SIZE + max_answersize, MIXART_MEM(mgr, msg_frame_address + 36) ); /* dwExpectedAnswerSize */
+
+ /* copy message data to card memory */
+ for( i=0; i < msg->size; i+=4 ) {
+ writel_be( *(u32*)(msg->data + i), MIXART_MEM(mgr, MSG_HEADER_SIZE + msg_frame_address + i) );
+ }
+
+ if( mark_pending ) {
+ if( *msg_event ) {
+ /* the pending event is the notification we wait for ! */
+ mgr->pending_event = *msg_event;
+ }
+ else {
+ /* the pending event is the answer we wait for (same address than the request)! */
+ mgr->pending_event = msg_frame_address;
+
+ /* copy address back to caller */
+ *msg_event = msg_frame_address;
+ }
+ }
+
+ /* mark the frame as a request (will have an answer) */
+ msg_frame_address |= MSG_TYPE_REQUEST;
+
+ /* post the frame */
+ headptr = readl_be(MIXART_MEM(mgr, MSG_INBOUND_POST_HEAD));
+
+ if( (headptr < MSG_INBOUND_POST_STACK) || (headptr >= (MSG_INBOUND_POST_STACK+MSG_BOUND_STACK_SIZE))) {
+ return -EINVAL;
+ }
+
+ writel_be(msg_frame_address, MIXART_MEM(mgr, headptr));
+
+ /* increment the inbound post head */
+ headptr += 4;
+ if( headptr >= (MSG_INBOUND_POST_STACK+MSG_BOUND_STACK_SIZE) )
+ headptr = MSG_INBOUND_POST_STACK;
+
+ writel_be(headptr, MIXART_MEM(mgr, MSG_INBOUND_POST_HEAD));
+
+ return 0;
+}
+
+
+int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int max_resp_size, void *resp_data)
+{
+ struct mixart_msg resp;
+ u32 msg_frame = 0; /* set to 0, so it's no notification to wait for, but the answer */
+ int err;
+ wait_queue_t wait;
+ long timeout;
+
+ mutex_lock(&mgr->msg_mutex);
+
+ init_waitqueue_entry(&wait, current);
+
+ spin_lock_irq(&mgr->msg_lock);
+ /* send the message */
+ err = send_msg(mgr, request, max_resp_size, 1, &msg_frame); /* send and mark the answer pending */
+ if (err) {
+ spin_unlock_irq(&mgr->msg_lock);
+ mutex_unlock(&mgr->msg_mutex);
+ return err;
+ }
+
+ set_current_state(TASK_UNINTERRUPTIBLE);
+ add_wait_queue(&mgr->msg_sleep, &wait);
+ spin_unlock_irq(&mgr->msg_lock);
+ timeout = schedule_timeout(MSG_TIMEOUT_JIFFIES);
+ remove_wait_queue(&mgr->msg_sleep, &wait);
+
+ if (! timeout) {
+ /* error - no ack */
+ mutex_unlock(&mgr->msg_mutex);
+ snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame);
+ return -EIO;
+ }
+
+ /* retrieve the answer into the same struct mixart_msg */
+ resp.message_id = 0;
+ resp.uid = (struct mixart_uid){0,0};
+ resp.data = resp_data;
+ resp.size = max_resp_size;
+
+ err = get_msg(mgr, &resp, msg_frame);
+
+ if( request->message_id != resp.message_id )
+ snd_printk(KERN_ERR "REPONSE ERROR!\n");
+
+ mutex_unlock(&mgr->msg_mutex);
+ return err;
+}
+
+
+int snd_mixart_send_msg_wait_notif(struct mixart_mgr *mgr,
+ struct mixart_msg *request, u32 notif_event)
+{
+ int err;
+ wait_queue_t wait;
+ long timeout;
+
+ if (snd_BUG_ON(!notif_event))
+ return -EINVAL;
+ if (snd_BUG_ON((notif_event & MSG_TYPE_MASK) != MSG_TYPE_NOTIFY))
+ return -EINVAL;
+ if (snd_BUG_ON(notif_event & MSG_CANCEL_NOTIFY_MASK))
+ return -EINVAL;
+
+ mutex_lock(&mgr->msg_mutex);
+
+ init_waitqueue_entry(&wait, current);
+
+ spin_lock_irq(&mgr->msg_lock);
+ /* send the message */
+ err = send_msg(mgr, request, MSG_DEFAULT_SIZE, 1, &notif_event); /* send and mark the notification event pending */
+ if(err) {
+ spin_unlock_irq(&mgr->msg_lock);
+ mutex_unlock(&mgr->msg_mutex);
+ return err;
+ }
+
+ set_current_state(TASK_UNINTERRUPTIBLE);
+ add_wait_queue(&mgr->msg_sleep, &wait);
+ spin_unlock_irq(&mgr->msg_lock);
+ timeout = schedule_timeout(MSG_TIMEOUT_JIFFIES);
+ remove_wait_queue(&mgr->msg_sleep, &wait);
+
+ if (! timeout) {
+ /* error - no ack */
+ mutex_unlock(&mgr->msg_mutex);
+ snd_printk(KERN_ERR "error: notification %x not received\n", notif_event);
+ return -EIO;
+ }
+
+ mutex_unlock(&mgr->msg_mutex);
+ return 0;
+}
+
+
+int snd_mixart_send_msg_nonblock(struct mixart_mgr *mgr, struct mixart_msg *request)
+{
+ u32 message_frame;
+ unsigned long flags;
+ int err;
+
+ /* just send the message (do not mark it as a pending one) */
+ spin_lock_irqsave(&mgr->msg_lock, flags);
+ err = send_msg(mgr, request, MSG_DEFAULT_SIZE, 0, &message_frame);
+ spin_unlock_irqrestore(&mgr->msg_lock, flags);
+
+ /* the answer will be handled by snd_struct mixart_msgasklet() */
+ atomic_inc(&mgr->msg_processed);
+
+ return err;
+}
+
+
+/* common buffer of tasklet and interrupt to send/receive messages */
+static u32 mixart_msg_data[MSG_DEFAULT_SIZE / 4];
+
+
+void snd_mixart_msg_tasklet(unsigned long arg)
+{
+ struct mixart_mgr *mgr = ( struct mixart_mgr*)(arg);
+ struct mixart_msg resp;
+ u32 msg, addr, type;
+ int err;
+
+ spin_lock(&mgr->lock);
+
+ while (mgr->msg_fifo_readptr != mgr->msg_fifo_writeptr) {
+ msg = mgr->msg_fifo[mgr->msg_fifo_readptr];
+ mgr->msg_fifo_readptr++;
+ mgr->msg_fifo_readptr %= MSG_FIFO_SIZE;
+
+ /* process the message ... */
+ addr = msg & ~MSG_TYPE_MASK;
+ type = msg & MSG_TYPE_MASK;
+
+ switch (type) {
+ case MSG_TYPE_ANSWER:
+ /* answer to a message on that we did not wait for (send_msg_nonblock) */
+ resp.message_id = 0;
+ resp.data = mixart_msg_data;
+ resp.size = sizeof(mixart_msg_data);
+ err = get_msg(mgr, &resp, addr);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "tasklet: error(%d) reading mf %x\n", err, msg);
+ break;
+ }
+
+ switch(resp.message_id) {
+ case MSG_STREAM_START_INPUT_STAGE_PACKET:
+ case MSG_STREAM_START_OUTPUT_STAGE_PACKET:
+ case MSG_STREAM_STOP_INPUT_STAGE_PACKET:
+ case MSG_STREAM_STOP_OUTPUT_STAGE_PACKET:
+ if(mixart_msg_data[0])
+ snd_printk(KERN_ERR "tasklet : error MSG_STREAM_ST***_***PUT_STAGE_PACKET status=%x\n", mixart_msg_data[0]);
+ break;
+ default:
+ snd_printdd("tasklet received mf(%x) : msg_id(%x) uid(%x, %x) size(%zd)\n",
+ msg, resp.message_id, resp.uid.object_id, resp.uid.desc, resp.size);
+ break;
+ }
+ break;
+ case MSG_TYPE_NOTIFY:
+ /* msg contains no address ! do not get_msg() ! */
+ case MSG_TYPE_COMMAND:
+ /* get_msg() necessary */
+ default:
+ snd_printk(KERN_ERR "tasklet doesn't know what to do with message %x\n", msg);
+ } /* switch type */
+
+ /* decrement counter */
+ atomic_dec(&mgr->msg_processed);
+
+ } /* while there is a msg in fifo */
+
+ spin_unlock(&mgr->lock);
+}
+
+
+irqreturn_t snd_mixart_interrupt(int irq, void *dev_id)
+{
+ struct mixart_mgr *mgr = dev_id;
+ int err;
+ struct mixart_msg resp;
+
+ u32 msg;
+ u32 it_reg;
+
+ spin_lock(&mgr->lock);
+
+ it_reg = readl_le(MIXART_REG(mgr, MIXART_PCI_OMISR_OFFSET));
+ if( !(it_reg & MIXART_OIDI) ) {
+ /* this device did not cause the interrupt */
+ spin_unlock(&mgr->lock);
+ return IRQ_NONE;
+ }
+
+ /* mask all interrupts */
+ writel_le(MIXART_HOST_ALL_INTERRUPT_MASKED, MIXART_REG(mgr, MIXART_PCI_OMIMR_OFFSET));
+
+ /* outdoorbell register clear */
+ it_reg = readl(MIXART_REG(mgr, MIXART_PCI_ODBR_OFFSET));
+ writel(it_reg, MIXART_REG(mgr, MIXART_PCI_ODBR_OFFSET));
+
+ /* clear interrupt */
+ writel_le( MIXART_OIDI, MIXART_REG(mgr, MIXART_PCI_OMISR_OFFSET) );
+
+ /* process interrupt */
+ while (retrieve_msg_frame(mgr, &msg)) {
+
+ switch (msg & MSG_TYPE_MASK) {
+ case MSG_TYPE_COMMAND:
+ resp.message_id = 0;
+ resp.data = mixart_msg_data;
+ resp.size = sizeof(mixart_msg_data);
+ err = get_msg(mgr, &resp, msg & ~MSG_TYPE_MASK);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "interrupt: error(%d) reading mf %x\n", err, msg);
+ break;
+ }
+
+ if(resp.message_id == MSG_SERVICES_TIMER_NOTIFY) {
+ int i;
+ struct mixart_timer_notify *notify;
+ notify = (struct mixart_timer_notify *)mixart_msg_data;
+
+ for(i=0; i<notify->stream_count; i++) {
+
+ u32 buffer_id = notify->streams[i].buffer_id;
+ unsigned int chip_number = (buffer_id & MIXART_NOTIFY_CARD_MASK) >> MIXART_NOTIFY_CARD_OFFSET; /* card0 to 3 */
+ unsigned int pcm_number = (buffer_id & MIXART_NOTIFY_PCM_MASK ) >> MIXART_NOTIFY_PCM_OFFSET; /* pcm0 to 3 */
+ unsigned int sub_number = buffer_id & MIXART_NOTIFY_SUBS_MASK; /* 0 to MIXART_PLAYBACK_STREAMS */
+ unsigned int is_capture = ((buffer_id & MIXART_NOTIFY_CAPT_MASK) != 0); /* playback == 0 / capture == 1 */
+
+ struct snd_mixart *chip = mgr->chip[chip_number];
+ struct mixart_stream *stream;
+
+ if ((chip_number >= mgr->num_cards) || (pcm_number >= MIXART_PCM_TOTAL) || (sub_number >= MIXART_PLAYBACK_STREAMS)) {
+ snd_printk(KERN_ERR "error MSG_SERVICES_TIMER_NOTIFY buffer_id (%x) pos(%d)\n",
+ buffer_id, notify->streams[i].sample_pos_low_part);
+ break;
+ }
+
+ if (is_capture)
+ stream = &chip->capture_stream[pcm_number];
+ else
+ stream = &chip->playback_stream[pcm_number][sub_number];
+
+ if (stream->substream && (stream->status == MIXART_STREAM_STATUS_RUNNING)) {
+ struct snd_pcm_runtime *runtime = stream->substream->runtime;
+ int elapsed = 0;
+ u64 sample_count = ((u64)notify->streams[i].sample_pos_high_part) << 32;
+ sample_count |= notify->streams[i].sample_pos_low_part;
+
+ while (1) {
+ u64 new_elapse_pos = stream->abs_period_elapsed + runtime->period_size;
+
+ if (new_elapse_pos > sample_count) {
+ break; /* while */
+ }
+ else {
+ elapsed = 1;
+ stream->buf_periods++;
+ if (stream->buf_periods >= runtime->periods)
+ stream->buf_periods = 0;
+
+ stream->abs_period_elapsed = new_elapse_pos;
+ }
+ }
+ stream->buf_period_frag = (u32)( sample_count - stream->abs_period_elapsed );
+
+ if(elapsed) {
+ spin_unlock(&mgr->lock);
+ snd_pcm_period_elapsed(stream->substream);
+ spin_lock(&mgr->lock);
+ }
+ }
+ }
+ break;
+ }
+ if(resp.message_id == MSG_SERVICES_REPORT_TRACES) {
+ if(resp.size > 1) {
+#ifndef __BIG_ENDIAN
+ /* Traces are text: the swapped msg_data has to be swapped back ! */
+ int i;
+ for(i=0; i<(resp.size/4); i++) {
+ (mixart_msg_data)[i] = cpu_to_be32((mixart_msg_data)[i]);
+ }
+#endif
+ ((char*)mixart_msg_data)[resp.size - 1] = 0;
+ snd_printdd("MIXART TRACE : %s\n", (char*)mixart_msg_data);
+ }
+ break;
+ }
+
+ snd_printdd("command %x not handled\n", resp.message_id);
+ break;
+
+ case MSG_TYPE_NOTIFY:
+ if(msg & MSG_CANCEL_NOTIFY_MASK) {
+ msg &= ~MSG_CANCEL_NOTIFY_MASK;
+ snd_printk(KERN_ERR "canceled notification %x !\n", msg);
+ }
+ /* no break, continue ! */
+ case MSG_TYPE_ANSWER:
+ /* answer or notification to a message we are waiting for*/
+ spin_lock(&mgr->msg_lock);
+ if( (msg & ~MSG_TYPE_MASK) == mgr->pending_event ) {
+ wake_up(&mgr->msg_sleep);
+ mgr->pending_event = 0;
+ }
+ /* answer to a message we did't want to wait for */
+ else {
+ mgr->msg_fifo[mgr->msg_fifo_writeptr] = msg;
+ mgr->msg_fifo_writeptr++;
+ mgr->msg_fifo_writeptr %= MSG_FIFO_SIZE;
+ tasklet_hi_schedule(&mgr->msg_taskq);
+ }
+ spin_unlock(&mgr->msg_lock);
+ break;
+ case MSG_TYPE_REQUEST:
+ default:
+ snd_printdd("interrupt received request %x\n", msg);
+ /* TODO : are there things to do here ? */
+ break;
+ } /* switch on msg type */
+ } /* while there are msgs */
+
+ /* allow interrupt again */
+ writel_le( MIXART_ALLOW_OUTBOUND_DOORBELL, MIXART_REG( mgr, MIXART_PCI_OMIMR_OFFSET));
+
+ spin_unlock(&mgr->lock);
+
+ return IRQ_HANDLED;
+}
+
+
+void snd_mixart_init_mailbox(struct mixart_mgr *mgr)
+{
+ writel( 0, MIXART_MEM( mgr, MSG_HOST_RSC_PROTECTION ) );
+ writel( 0, MIXART_MEM( mgr, MSG_AGENT_RSC_PROTECTION ) );
+
+ /* allow outbound messagebox to generate interrupts */
+ if(mgr->irq >= 0) {
+ writel_le( MIXART_ALLOW_OUTBOUND_DOORBELL, MIXART_REG( mgr, MIXART_PCI_OMIMR_OFFSET));
+ }
+ return;
+}
+
+void snd_mixart_exit_mailbox(struct mixart_mgr *mgr)
+{
+ /* no more interrupts on outbound messagebox */
+ writel_le( MIXART_HOST_ALL_INTERRUPT_MASKED, MIXART_REG( mgr, MIXART_PCI_OMIMR_OFFSET));
+ return;
+}
+
+void snd_mixart_reset_board(struct mixart_mgr *mgr)
+{
+ /* reset miXart */
+ writel_be( 1, MIXART_REG(mgr, MIXART_BA1_BRUTAL_RESET_OFFSET) );
+ return;
+}
diff --git a/sound/pci/mixart/mixart_core.h b/sound/pci/mixart/mixart_core.h
new file mode 100644
index 0000000..c919b73
--- /dev/null
+++ b/sound/pci/mixart/mixart_core.h
@@ -0,0 +1,571 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * low level interface with interrupt handling and mail box implementation
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_MIXART_CORE_H
+#define __SOUND_MIXART_CORE_H
+
+
+enum mixart_message_id {
+ MSG_CONNECTOR_GET_AUDIO_INFO = 0x050008,
+ MSG_CONNECTOR_GET_OUT_AUDIO_LEVEL = 0x050009,
+ MSG_CONNECTOR_SET_OUT_AUDIO_LEVEL = 0x05000A,
+
+ MSG_CONSOLE_MANAGER = 0x070000,
+ MSG_CONSOLE_GET_CLOCK_UID = 0x070003,
+
+ MSG_PHYSICALIO_SET_LEVEL = 0x0F0008,
+
+ MSG_STREAM_ADD_INPUT_GROUP = 0x130000,
+ MSG_STREAM_ADD_OUTPUT_GROUP = 0x130001,
+ MSG_STREAM_DELETE_GROUP = 0x130004,
+ MSG_STREAM_START_STREAM_GRP_PACKET = 0x130006,
+ MSG_STREAM_START_INPUT_STAGE_PACKET = 0x130007,
+ MSG_STREAM_START_OUTPUT_STAGE_PACKET = 0x130008,
+ MSG_STREAM_STOP_STREAM_GRP_PACKET = 0x130009,
+ MSG_STREAM_STOP_INPUT_STAGE_PACKET = 0x13000A,
+ MSG_STREAM_STOP_OUTPUT_STAGE_PACKET = 0x13000B,
+ MSG_STREAM_SET_INPUT_STAGE_PARAM = 0x13000F,
+ MSG_STREAM_SET_OUTPUT_STAGE_PARAM = 0x130010,
+ MSG_STREAM_SET_IN_AUDIO_LEVEL = 0x130015,
+ MSG_STREAM_SET_OUT_STREAM_LEVEL = 0x130017,
+
+ MSG_SYSTEM_FIRST_ID = 0x160000,
+ MSG_SYSTEM_ENUM_PHYSICAL_IO = 0x16000E,
+ MSG_SYSTEM_ENUM_PLAY_CONNECTOR = 0x160017,
+ MSG_SYSTEM_ENUM_RECORD_CONNECTOR = 0x160018,
+ MSG_SYSTEM_WAIT_SYNCHRO_CMD = 0x16002C,
+ MSG_SYSTEM_SEND_SYNCHRO_CMD = 0x16002D,
+
+ MSG_SERVICES_TIMER_NOTIFY = 0x1D0404,
+ MSG_SERVICES_REPORT_TRACES = 0x1D0700,
+
+ MSG_CLOCK_CHECK_PROPERTIES = 0x200001,
+ MSG_CLOCK_SET_PROPERTIES = 0x200002,
+};
+
+
+struct mixart_msg
+{
+ u32 message_id;
+ struct mixart_uid uid;
+ void* data;
+ size_t size;
+};
+
+/* structs used to communicate with miXart */
+
+struct mixart_enum_connector_resp
+{
+ u32 error_code;
+ u32 first_uid_offset;
+ u32 uid_count;
+ u32 current_uid_index;
+ struct mixart_uid uid[MIXART_MAX_PHYS_CONNECTORS];
+} __attribute__((packed));
+
+
+/* used for following struct */
+#define MIXART_FLOAT_P_22_0_TO_HEX 0x41b00000 /* 22.0f */
+#define MIXART_FLOAT_M_20_0_TO_HEX 0xc1a00000 /* -20.0f */
+#define MIXART_FLOAT____0_0_TO_HEX 0x00000000 /* 0.0f */
+
+struct mixart_audio_info_req
+{
+ u32 line_max_level; /* float */
+ u32 micro_max_level; /* float */
+ u32 cd_max_level; /* float */
+} __attribute__((packed));
+
+struct mixart_analog_hw_info
+{
+ u32 is_present;
+ u32 hw_connection_type;
+ u32 max_level; /* float */
+ u32 min_var_level; /* float */
+ u32 max_var_level; /* float */
+ u32 step_var_level; /* float */
+ u32 fix_gain; /* float */
+ u32 zero_var; /* float */
+} __attribute__((packed));
+
+struct mixart_digital_hw_info
+{
+ u32 hw_connection_type;
+ u32 presence;
+ u32 clock;
+ u32 reserved;
+} __attribute__((packed));
+
+struct mixart_analog_info
+{
+ u32 type_mask;
+ struct mixart_analog_hw_info micro_info;
+ struct mixart_analog_hw_info line_info;
+ struct mixart_analog_hw_info cd_info;
+ u32 analog_level_present;
+} __attribute__((packed));
+
+struct mixart_digital_info
+{
+ u32 type_mask;
+ struct mixart_digital_hw_info aes_info;
+ struct mixart_digital_hw_info adat_info;
+} __attribute__((packed));
+
+struct mixart_audio_info
+{
+ u32 clock_type_mask;
+ struct mixart_analog_info analog_info;
+ struct mixart_digital_info digital_info;
+} __attribute__((packed));
+
+struct mixart_audio_info_resp
+{
+ u32 txx_status;
+ struct mixart_audio_info info;
+} __attribute__((packed));
+
+
+/* used for nb_bytes_max_per_sample */
+#define MIXART_FLOAT_P__4_0_TO_HEX 0x40800000 /* +4.0f */
+#define MIXART_FLOAT_P__8_0_TO_HEX 0x41000000 /* +8.0f */
+
+struct mixart_stream_info
+{
+ u32 size_max_byte_frame;
+ u32 size_max_sample_frame;
+ u32 nb_bytes_max_per_sample; /* float */
+} __attribute__((packed));
+
+/* MSG_STREAM_ADD_INPUT_GROUP */
+/* MSG_STREAM_ADD_OUTPUT_GROUP */
+
+struct mixart_streaming_group_req
+{
+ u32 stream_count;
+ u32 channel_count;
+ u32 user_grp_number;
+ u32 first_phys_audio;
+ u32 latency;
+ struct mixart_stream_info stream_info[32];
+ struct mixart_uid connector;
+ u32 flow_entry[32];
+} __attribute__((packed));
+
+struct mixart_stream_desc
+{
+ struct mixart_uid stream_uid;
+ u32 stream_desc;
+} __attribute__((packed));
+
+struct mixart_streaming_group
+{
+ u32 status;
+ struct mixart_uid group;
+ u32 pipe_desc;
+ u32 stream_count;
+ struct mixart_stream_desc stream[32];
+} __attribute__((packed));
+
+/* MSG_STREAM_DELETE_GROUP */
+
+/* request : mixart_uid_t group */
+
+struct mixart_delete_group_resp
+{
+ u32 status;
+ u32 unused[2];
+} __attribute__((packed));
+
+
+/* MSG_STREAM_START_INPUT_STAGE_PACKET = 0x130000 + 7,
+ MSG_STREAM_START_OUTPUT_STAGE_PACKET = 0x130000 + 8,
+ MSG_STREAM_STOP_INPUT_STAGE_PACKET = 0x130000 + 10,
+ MSG_STREAM_STOP_OUTPUT_STAGE_PACKET = 0x130000 + 11,
+ */
+
+struct mixart_fx_couple_uid
+{
+ struct mixart_uid uid_fx_code;
+ struct mixart_uid uid_fx_data;
+} __attribute__((packed));
+
+struct mixart_txx_stream_desc
+{
+ struct mixart_uid uid_pipe;
+ u32 stream_idx;
+ u32 fx_number;
+ struct mixart_fx_couple_uid uid_fx[4];
+} __attribute__((packed));
+
+struct mixart_flow_info
+{
+ struct mixart_txx_stream_desc stream_desc;
+ u32 flow_entry;
+ u32 flow_phy_addr;
+} __attribute__((packed));
+
+struct mixart_stream_state_req
+{
+ u32 delayed;
+ u64 scheduler;
+ u32 reserved4np[3];
+ u32 stream_count; /* set to 1 for instance */
+ struct mixart_flow_info stream_info; /* could be an array[stream_count] */
+} __attribute__((packed));
+
+/* MSG_STREAM_START_STREAM_GRP_PACKET = 0x130000 + 6
+ MSG_STREAM_STOP_STREAM_GRP_PACKET = 0x130000 + 9
+ */
+
+struct mixart_group_state_req
+{
+ u32 delayed;
+ u64 scheduler;
+ u32 reserved4np[2];
+ u32 pipe_count; /* set to 1 for instance */
+ struct mixart_uid pipe_uid[1]; /* could be an array[pipe_count] */
+} __attribute__((packed));
+
+struct mixart_group_state_resp
+{
+ u32 txx_status;
+ u64 scheduler;
+} __attribute__((packed));
+
+
+
+/* Structures used by the MSG_SERVICES_TIMER_NOTIFY command */
+
+struct mixart_sample_pos
+{
+ u32 buffer_id;
+ u32 validity;
+ u32 sample_pos_high_part;
+ u32 sample_pos_low_part;
+} __attribute__((packed));
+
+struct mixart_timer_notify
+{
+ u32 stream_count;
+ struct mixart_sample_pos streams[MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS];
+} __attribute__((packed));
+
+
+/* MSG_CONSOLE_GET_CLOCK_UID = 0x070003,
+ */
+
+/* request is a uid with desc = MSG_CONSOLE_MANAGER | cardindex */
+
+struct mixart_return_uid
+{
+ u32 error_code;
+ struct mixart_uid uid;
+} __attribute__((packed));
+
+/* MSG_CLOCK_CHECK_PROPERTIES = 0x200001,
+ MSG_CLOCK_SET_PROPERTIES = 0x200002,
+*/
+
+enum mixart_clock_generic_type {
+ CGT_NO_CLOCK,
+ CGT_INTERNAL_CLOCK,
+ CGT_PROGRAMMABLE_CLOCK,
+ CGT_INTERNAL_ENSLAVED_CLOCK,
+ CGT_EXTERNAL_CLOCK,
+ CGT_CURRENT_CLOCK
+};
+
+enum mixart_clock_mode {
+ CM_UNDEFINED,
+ CM_MASTER,
+ CM_SLAVE,
+ CM_STANDALONE,
+ CM_NOT_CONCERNED
+};
+
+
+struct mixart_clock_properties
+{
+ u32 error_code;
+ u32 validation_mask;
+ u32 frequency;
+ u32 reference_frequency;
+ u32 clock_generic_type;
+ u32 clock_mode;
+ struct mixart_uid uid_clock_source;
+ struct mixart_uid uid_event_source;
+ u32 event_mode;
+ u32 synchro_signal_presence;
+ u32 format;
+ u32 board_mask;
+ u32 nb_callers; /* set to 1 (see below) */
+ struct mixart_uid uid_caller[1];
+} __attribute__((packed));
+
+struct mixart_clock_properties_resp
+{
+ u32 status;
+ u32 clock_mode;
+} __attribute__((packed));
+
+
+/* MSG_STREAM_SET_INPUT_STAGE_PARAM = 0x13000F */
+/* MSG_STREAM_SET_OUTPUT_STAGE_PARAM = 0x130010 */
+
+enum mixart_coding_type {
+ CT_NOT_DEFINED,
+ CT_LINEAR,
+ CT_MPEG_L1,
+ CT_MPEG_L2,
+ CT_MPEG_L3,
+ CT_MPEG_L3_LSF,
+ CT_GSM
+};
+enum mixart_sample_type {
+ ST_NOT_DEFINED,
+ ST_FLOATING_POINT_32BE,
+ ST_FLOATING_POINT_32LE,
+ ST_FLOATING_POINT_64BE,
+ ST_FLOATING_POINT_64LE,
+ ST_FIXED_POINT_8,
+ ST_FIXED_POINT_16BE,
+ ST_FIXED_POINT_16LE,
+ ST_FIXED_POINT_24BE,
+ ST_FIXED_POINT_24LE,
+ ST_FIXED_POINT_32BE,
+ ST_FIXED_POINT_32LE,
+ ST_INTEGER_8,
+ ST_INTEGER_16BE,
+ ST_INTEGER_16LE,
+ ST_INTEGER_24BE,
+ ST_INTEGER_24LE,
+ ST_INTEGER_32BE,
+ ST_INTEGER_32LE
+};
+
+struct mixart_stream_param_desc
+{
+ u32 coding_type; /* use enum mixart_coding_type */
+ u32 sample_type; /* use enum mixart_sample_type */
+
+ union {
+ struct {
+ u32 linear_endian_ness;
+ u32 linear_bits;
+ u32 is_signed;
+ u32 is_float;
+ } linear_format_info;
+
+ struct {
+ u32 mpeg_layer;
+ u32 mpeg_mode;
+ u32 mpeg_mode_extension;
+ u32 mpeg_pre_emphasis;
+ u32 mpeg_has_padding_bit;
+ u32 mpeg_has_crc;
+ u32 mpeg_has_extension;
+ u32 mpeg_is_original;
+ u32 mpeg_has_copyright;
+ } mpeg_format_info;
+ } format_info;
+
+ u32 delayed;
+ u64 scheduler;
+ u32 sample_size;
+ u32 has_header;
+ u32 has_suffix;
+ u32 has_bitrate;
+ u32 samples_per_frame;
+ u32 bytes_per_frame;
+ u32 bytes_per_sample;
+ u32 sampling_freq;
+ u32 number_of_channel;
+ u32 stream_number;
+ u32 buffer_size;
+ u32 differed_time;
+ u32 reserved4np[3];
+ u32 pipe_count; /* set to 1 (array size !) */
+ u32 stream_count; /* set to 1 (array size !) */
+ struct mixart_txx_stream_desc stream_desc[1]; /* only one stream per command, but this could be an array */
+
+} __attribute__((packed));
+
+
+/* MSG_CONNECTOR_GET_OUT_AUDIO_LEVEL = 0x050009,
+ */
+
+
+struct mixart_get_out_audio_level
+{
+ u32 txx_status;
+ u32 digital_level; /* float */
+ u32 analog_level; /* float */
+ u32 monitor_level; /* float */
+ u32 mute;
+ u32 monitor_mute1;
+ u32 monitor_mute2;
+} __attribute__((packed));
+
+
+/* MSG_CONNECTOR_SET_OUT_AUDIO_LEVEL = 0x05000A,
+ */
+
+/* used for valid_mask below */
+#define MIXART_AUDIO_LEVEL_ANALOG_MASK 0x01
+#define MIXART_AUDIO_LEVEL_DIGITAL_MASK 0x02
+#define MIXART_AUDIO_LEVEL_MONITOR_MASK 0x04
+#define MIXART_AUDIO_LEVEL_MUTE_MASK 0x08
+#define MIXART_AUDIO_LEVEL_MUTE_M1_MASK 0x10
+#define MIXART_AUDIO_LEVEL_MUTE_M2_MASK 0x20
+
+struct mixart_set_out_audio_level
+{
+ u32 delayed;
+ u64 scheduler;
+ u32 valid_mask1;
+ u32 valid_mask2;
+ u32 digital_level; /* float */
+ u32 analog_level; /* float */
+ u32 monitor_level; /* float */
+ u32 mute;
+ u32 monitor_mute1;
+ u32 monitor_mute2;
+ u32 reserved4np;
+} __attribute__((packed));
+
+
+/* MSG_SYSTEM_ENUM_PHYSICAL_IO = 0x16000E,
+ */
+
+#define MIXART_MAX_PHYS_IO (MIXART_MAX_CARDS * 2 * 2) /* 4 * (analog+digital) * (playback+capture) */
+
+struct mixart_uid_enumeration
+{
+ u32 error_code;
+ u32 first_uid_offset;
+ u32 nb_uid;
+ u32 current_uid_index;
+ struct mixart_uid uid[MIXART_MAX_PHYS_IO];
+} __attribute__((packed));
+
+
+/* MSG_PHYSICALIO_SET_LEVEL = 0x0F0008,
+ MSG_PHYSICALIO_GET_LEVEL = 0x0F000C,
+*/
+
+struct mixart_io_channel_level
+{
+ u32 analog_level; /* float */
+ u32 unused[2];
+} __attribute__((packed));
+
+struct mixart_io_level
+{
+ s32 channel; /* 0=left, 1=right, -1=both, -2=both same */
+ struct mixart_io_channel_level level[2];
+} __attribute__((packed));
+
+
+/* MSG_STREAM_SET_IN_AUDIO_LEVEL = 0x130015,
+ */
+
+struct mixart_in_audio_level_info
+{
+ struct mixart_uid connector;
+ u32 valid_mask1;
+ u32 valid_mask2;
+ u32 digital_level;
+ u32 analog_level;
+} __attribute__((packed));
+
+struct mixart_set_in_audio_level_req
+{
+ u32 delayed;
+ u64 scheduler;
+ u32 audio_count; /* set to <= 2 */
+ u32 reserved4np;
+ struct mixart_in_audio_level_info level[2];
+} __attribute__((packed));
+
+/* response is a 32 bit status */
+
+
+/* MSG_STREAM_SET_OUT_STREAM_LEVEL = 0x130017,
+ */
+
+/* defines used for valid_mask1 */
+#define MIXART_OUT_STREAM_SET_LEVEL_LEFT_AUDIO1 0x01
+#define MIXART_OUT_STREAM_SET_LEVEL_LEFT_AUDIO2 0x02
+#define MIXART_OUT_STREAM_SET_LEVEL_RIGHT_AUDIO1 0x04
+#define MIXART_OUT_STREAM_SET_LEVEL_RIGHT_AUDIO2 0x08
+#define MIXART_OUT_STREAM_SET_LEVEL_STREAM_1 0x10
+#define MIXART_OUT_STREAM_SET_LEVEL_STREAM_2 0x20
+#define MIXART_OUT_STREAM_SET_LEVEL_MUTE_1 0x40
+#define MIXART_OUT_STREAM_SET_LEVEL_MUTE_2 0x80
+
+struct mixart_out_stream_level_info
+{
+ u32 valid_mask1;
+ u32 valid_mask2;
+ u32 left_to_out1_level;
+ u32 left_to_out2_level;
+ u32 right_to_out1_level;
+ u32 right_to_out2_level;
+ u32 digital_level1;
+ u32 digital_level2;
+ u32 mute1;
+ u32 mute2;
+} __attribute__((packed));
+
+struct mixart_set_out_stream_level
+{
+ struct mixart_txx_stream_desc desc;
+ struct mixart_out_stream_level_info out_level;
+} __attribute__((packed));
+
+struct mixart_set_out_stream_level_req
+{
+ u32 delayed;
+ u64 scheduler;
+ u32 reserved4np[2];
+ u32 nb_of_stream; /* set to 1 */
+ struct mixart_set_out_stream_level stream_level; /* could be an array */
+} __attribute__((packed));
+
+/* response to this request is a u32 status value */
+
+
+/* exported */
+void snd_mixart_init_mailbox(struct mixart_mgr *mgr);
+void snd_mixart_exit_mailbox(struct mixart_mgr *mgr);
+
+int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int max_resp_size, void *resp_data);
+int snd_mixart_send_msg_wait_notif(struct mixart_mgr *mgr, struct mixart_msg *request, u32 notif_event);
+int snd_mixart_send_msg_nonblock(struct mixart_mgr *mgr, struct mixart_msg *request);
+
+irqreturn_t snd_mixart_interrupt(int irq, void *dev_id);
+void snd_mixart_msg_tasklet(unsigned long arg);
+
+void snd_mixart_reset_board(struct mixart_mgr *mgr);
+
+#endif /* __SOUND_MIXART_CORE_H */
diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c
new file mode 100644
index 0000000..3782b52
--- /dev/null
+++ b/sound/pci/mixart/mixart_hwdep.c
@@ -0,0 +1,657 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * DSP firmware management
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/firmware.h>
+#include <linux/vmalloc.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include "mixart.h"
+#include "mixart_mixer.h"
+#include "mixart_core.h"
+#include "mixart_hwdep.h"
+
+
+/**
+ * wait for a value on a peudo register, exit with a timeout
+ *
+ * @param mgr pointer to miXart manager structure
+ * @param offset unsigned pseudo_register base + offset of value
+ * @param value value
+ * @param timeout timeout in centisenconds
+ */
+static int mixart_wait_nice_for_register_value(struct mixart_mgr *mgr,
+ u32 offset, int is_egal,
+ u32 value, unsigned long timeout)
+{
+ unsigned long end_time = jiffies + (timeout * HZ / 100);
+ u32 read;
+
+ do { /* we may take too long time in this loop.
+ * so give controls back to kernel if needed.
+ */
+ cond_resched();
+
+ read = readl_be( MIXART_MEM( mgr, offset ));
+ if(is_egal) {
+ if(read == value) return 0;
+ }
+ else { /* wait for different value */
+ if(read != value) return 0;
+ }
+ } while ( time_after_eq(end_time, jiffies) );
+
+ return -EBUSY;
+}
+
+
+/*
+ structures needed to upload elf code packets
+ */
+struct snd_mixart_elf32_ehdr {
+ u8 e_ident[16];
+ u16 e_type;
+ u16 e_machine;
+ u32 e_version;
+ u32 e_entry;
+ u32 e_phoff;
+ u32 e_shoff;
+ u32 e_flags;
+ u16 e_ehsize;
+ u16 e_phentsize;
+ u16 e_phnum;
+ u16 e_shentsize;
+ u16 e_shnum;
+ u16 e_shstrndx;
+};
+
+struct snd_mixart_elf32_phdr {
+ u32 p_type;
+ u32 p_offset;
+ u32 p_vaddr;
+ u32 p_paddr;
+ u32 p_filesz;
+ u32 p_memsz;
+ u32 p_flags;
+ u32 p_align;
+};
+
+static int mixart_load_elf(struct mixart_mgr *mgr, const struct firmware *dsp )
+{
+ char elf32_magic_number[4] = {0x7f,'E','L','F'};
+ struct snd_mixart_elf32_ehdr *elf_header;
+ int i;
+
+ elf_header = (struct snd_mixart_elf32_ehdr *)dsp->data;
+ for( i=0; i<4; i++ )
+ if ( elf32_magic_number[i] != elf_header->e_ident[i] )
+ return -EINVAL;
+
+ if( elf_header->e_phoff != 0 ) {
+ struct snd_mixart_elf32_phdr elf_programheader;
+
+ for( i=0; i < be16_to_cpu(elf_header->e_phnum); i++ ) {
+ u32 pos = be32_to_cpu(elf_header->e_phoff) + (u32)(i * be16_to_cpu(elf_header->e_phentsize));
+
+ memcpy( &elf_programheader, dsp->data + pos, sizeof(elf_programheader) );
+
+ if(elf_programheader.p_type != 0) {
+ if( elf_programheader.p_filesz != 0 ) {
+ memcpy_toio( MIXART_MEM( mgr, be32_to_cpu(elf_programheader.p_vaddr)),
+ dsp->data + be32_to_cpu( elf_programheader.p_offset ),
+ be32_to_cpu( elf_programheader.p_filesz ));
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+/*
+ * get basic information and init miXart
+ */
+
+/* audio IDs for request to the board */
+#define MIXART_FIRST_ANA_AUDIO_ID 0
+#define MIXART_FIRST_DIG_AUDIO_ID 8
+
+static int mixart_enum_connectors(struct mixart_mgr *mgr)
+{
+ u32 k;
+ int err;
+ struct mixart_msg request;
+ struct mixart_enum_connector_resp *connector;
+ struct mixart_audio_info_req *audio_info_req;
+ struct mixart_audio_info_resp *audio_info;
+
+ connector = kmalloc(sizeof(*connector), GFP_KERNEL);
+ audio_info_req = kmalloc(sizeof(*audio_info_req), GFP_KERNEL);
+ audio_info = kmalloc(sizeof(*audio_info), GFP_KERNEL);
+ if (! connector || ! audio_info_req || ! audio_info) {
+ err = -ENOMEM;
+ goto __error;
+ }
+
+ audio_info_req->line_max_level = MIXART_FLOAT_P_22_0_TO_HEX;
+ audio_info_req->micro_max_level = MIXART_FLOAT_M_20_0_TO_HEX;
+ audio_info_req->cd_max_level = MIXART_FLOAT____0_0_TO_HEX;
+
+ request.message_id = MSG_SYSTEM_ENUM_PLAY_CONNECTOR;
+ request.uid = (struct mixart_uid){0,0}; /* board num = 0 */
+ request.data = NULL;
+ request.size = 0;
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(*connector), connector);
+ if((err < 0) || (connector->error_code) || (connector->uid_count > MIXART_MAX_PHYS_CONNECTORS)) {
+ snd_printk(KERN_ERR "error MSG_SYSTEM_ENUM_PLAY_CONNECTOR\n");
+ err = -EINVAL;
+ goto __error;
+ }
+
+ for(k=0; k < connector->uid_count; k++) {
+ struct mixart_pipe *pipe;
+
+ if(k < MIXART_FIRST_DIG_AUDIO_ID) {
+ pipe = &mgr->chip[k/2]->pipe_out_ana;
+ } else {
+ pipe = &mgr->chip[(k-MIXART_FIRST_DIG_AUDIO_ID)/2]->pipe_out_dig;
+ }
+ if(k & 1) {
+ pipe->uid_right_connector = connector->uid[k]; /* odd */
+ } else {
+ pipe->uid_left_connector = connector->uid[k]; /* even */
+ }
+
+ /* snd_printk(KERN_DEBUG "playback connector[%d].object_id = %x\n", k, connector->uid[k].object_id); */
+
+ /* TODO: really need send_msg MSG_CONNECTOR_GET_AUDIO_INFO for each connector ? perhaps for analog level caps ? */
+ request.message_id = MSG_CONNECTOR_GET_AUDIO_INFO;
+ request.uid = connector->uid[k];
+ request.data = audio_info_req;
+ request.size = sizeof(*audio_info_req);
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(*audio_info), audio_info);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "error MSG_CONNECTOR_GET_AUDIO_INFO\n");
+ goto __error;
+ }
+ /*snd_printk(KERN_DEBUG "play analog_info.analog_level_present = %x\n", audio_info->info.analog_info.analog_level_present);*/
+ }
+
+ request.message_id = MSG_SYSTEM_ENUM_RECORD_CONNECTOR;
+ request.uid = (struct mixart_uid){0,0}; /* board num = 0 */
+ request.data = NULL;
+ request.size = 0;
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(*connector), connector);
+ if((err < 0) || (connector->error_code) || (connector->uid_count > MIXART_MAX_PHYS_CONNECTORS)) {
+ snd_printk(KERN_ERR "error MSG_SYSTEM_ENUM_RECORD_CONNECTOR\n");
+ err = -EINVAL;
+ goto __error;
+ }
+
+ for(k=0; k < connector->uid_count; k++) {
+ struct mixart_pipe *pipe;
+
+ if(k < MIXART_FIRST_DIG_AUDIO_ID) {
+ pipe = &mgr->chip[k/2]->pipe_in_ana;
+ } else {
+ pipe = &mgr->chip[(k-MIXART_FIRST_DIG_AUDIO_ID)/2]->pipe_in_dig;
+ }
+ if(k & 1) {
+ pipe->uid_right_connector = connector->uid[k]; /* odd */
+ } else {
+ pipe->uid_left_connector = connector->uid[k]; /* even */
+ }
+
+ /* snd_printk(KERN_DEBUG "capture connector[%d].object_id = %x\n", k, connector->uid[k].object_id); */
+
+ /* TODO: really need send_msg MSG_CONNECTOR_GET_AUDIO_INFO for each connector ? perhaps for analog level caps ? */
+ request.message_id = MSG_CONNECTOR_GET_AUDIO_INFO;
+ request.uid = connector->uid[k];
+ request.data = audio_info_req;
+ request.size = sizeof(*audio_info_req);
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(*audio_info), audio_info);
+ if( err < 0 ) {
+ snd_printk(KERN_ERR "error MSG_CONNECTOR_GET_AUDIO_INFO\n");
+ goto __error;
+ }
+ /*snd_printk(KERN_DEBUG "rec analog_info.analog_level_present = %x\n", audio_info->info.analog_info.analog_level_present);*/
+ }
+ err = 0;
+
+ __error:
+ kfree(connector);
+ kfree(audio_info_req);
+ kfree(audio_info);
+
+ return err;
+}
+
+static int mixart_enum_physio(struct mixart_mgr *mgr)
+{
+ u32 k;
+ int err;
+ struct mixart_msg request;
+ struct mixart_uid get_console_mgr;
+ struct mixart_return_uid console_mgr;
+ struct mixart_uid_enumeration phys_io;
+
+ /* get the uid for the console manager */
+ get_console_mgr.object_id = 0;
+ get_console_mgr.desc = MSG_CONSOLE_MANAGER | 0; /* cardindex = 0 */
+
+ request.message_id = MSG_CONSOLE_GET_CLOCK_UID;
+ request.uid = get_console_mgr;
+ request.data = &get_console_mgr;
+ request.size = sizeof(get_console_mgr);
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(console_mgr), &console_mgr);
+
+ if( (err < 0) || (console_mgr.error_code != 0) ) {
+ snd_printk(KERN_DEBUG "error MSG_CONSOLE_GET_CLOCK_UID : err=%x\n", console_mgr.error_code);
+ return -EINVAL;
+ }
+
+ /* used later for clock issues ! */
+ mgr->uid_console_manager = console_mgr.uid;
+
+ request.message_id = MSG_SYSTEM_ENUM_PHYSICAL_IO;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = &console_mgr.uid;
+ request.size = sizeof(console_mgr.uid);
+
+ err = snd_mixart_send_msg(mgr, &request, sizeof(phys_io), &phys_io);
+ if( (err < 0) || ( phys_io.error_code != 0 ) ) {
+ snd_printk(KERN_ERR "error MSG_SYSTEM_ENUM_PHYSICAL_IO err(%x) error_code(%x)\n", err, phys_io.error_code );
+ return -EINVAL;
+ }
+
+ /* min 2 phys io per card (analog in + analog out) */
+ if (phys_io.nb_uid < MIXART_MAX_CARDS * 2)
+ return -EINVAL;
+
+ for(k=0; k<mgr->num_cards; k++) {
+ mgr->chip[k]->uid_in_analog_physio = phys_io.uid[k];
+ mgr->chip[k]->uid_out_analog_physio = phys_io.uid[phys_io.nb_uid/2 + k];
+ }
+
+ return 0;
+}
+
+
+static int mixart_first_init(struct mixart_mgr *mgr)
+{
+ u32 k;
+ int err;
+ struct mixart_msg request;
+
+ if((err = mixart_enum_connectors(mgr)) < 0) return err;
+
+ if((err = mixart_enum_physio(mgr)) < 0) return err;
+
+ /* send a synchro command to card (necessary to do this before first MSG_STREAM_START_STREAM_GRP_PACKET) */
+ /* though why not here */
+ request.message_id = MSG_SYSTEM_SEND_SYNCHRO_CMD;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = NULL;
+ request.size = 0;
+ /* this command has no data. response is a 32 bit status */
+ err = snd_mixart_send_msg(mgr, &request, sizeof(k), &k);
+ if( (err < 0) || (k != 0) ) {
+ snd_printk(KERN_ERR "error MSG_SYSTEM_SEND_SYNCHRO_CMD\n");
+ return err == 0 ? -EINVAL : err;
+ }
+
+ return 0;
+}
+
+
+/* firmware base addresses (when hard coded) */
+#define MIXART_MOTHERBOARD_XLX_BASE_ADDRESS 0x00600000
+
+static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmware *dsp)
+{
+ int err, card_index;
+ u32 status_xilinx, status_elf, status_daught;
+ u32 val;
+
+ /* read motherboard xilinx status */
+ status_xilinx = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_MXLX_STATUS_OFFSET ));
+ /* read elf status */
+ status_elf = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_ELF_STATUS_OFFSET ));
+ /* read daughterboard xilinx status */
+ status_daught = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_DXLX_STATUS_OFFSET ));
+
+ /* motherboard xilinx status 5 will say that the board is performing a reset */
+ if( status_xilinx == 5 ) {
+ snd_printk( KERN_ERR "miXart is resetting !\n");
+ return -EAGAIN; /* try again later */
+ }
+
+ switch (index) {
+ case MIXART_MOTHERBOARD_XLX_INDEX:
+
+ /* xilinx already loaded ? */
+ if( status_xilinx == 4 ) {
+ snd_printk( KERN_DEBUG "xilinx is already loaded !\n");
+ return 0;
+ }
+ /* the status should be 0 == "idle" */
+ if( status_xilinx != 0 ) {
+ snd_printk( KERN_ERR "xilinx load error ! status = %d\n", status_xilinx);
+ return -EIO; /* modprob -r may help ? */
+ }
+
+ /* check xilinx validity */
+ if (((u32*)(dsp->data))[0] == 0xffffffff)
+ return -EINVAL;
+ if (dsp->size % 4)
+ return -EINVAL;
+
+ /* set xilinx status to copying */
+ writel_be( 1, MIXART_MEM( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET ));
+
+ /* setup xilinx base address */
+ writel_be( MIXART_MOTHERBOARD_XLX_BASE_ADDRESS, MIXART_MEM( mgr,MIXART_PSEUDOREG_MXLX_BASE_ADDR_OFFSET ));
+ /* setup code size for xilinx file */
+ writel_be( dsp->size, MIXART_MEM( mgr, MIXART_PSEUDOREG_MXLX_SIZE_OFFSET ));
+
+ /* copy xilinx code */
+ memcpy_toio( MIXART_MEM( mgr, MIXART_MOTHERBOARD_XLX_BASE_ADDRESS), dsp->data, dsp->size);
+
+ /* set xilinx status to copy finished */
+ writel_be( 2, MIXART_MEM( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET ));
+
+ /* return, because no further processing needed */
+ return 0;
+
+ case MIXART_MOTHERBOARD_ELF_INDEX:
+
+ if( status_elf == 4 ) {
+ snd_printk( KERN_DEBUG "elf file already loaded !\n");
+ return 0;
+ }
+
+ /* the status should be 0 == "idle" */
+ if( status_elf != 0 ) {
+ snd_printk( KERN_ERR "elf load error ! status = %d\n", status_elf);
+ return -EIO; /* modprob -r may help ? */
+ }
+
+ /* wait for xilinx status == 4 */
+ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET, 1, 4, 500); /* 5sec */
+ if (err < 0) {
+ snd_printk( KERN_ERR "xilinx was not loaded or could not be started\n");
+ return err;
+ }
+
+ /* init some data on the card */
+ writel_be( 0, MIXART_MEM( mgr, MIXART_PSEUDOREG_BOARDNUMBER ) ); /* set miXart boardnumber to 0 */
+ writel_be( 0, MIXART_MEM( mgr, MIXART_FLOWTABLE_PTR ) ); /* reset pointer to flow table on miXart */
+
+ /* set elf status to copying */
+ writel_be( 1, MIXART_MEM( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET ));
+
+ /* process the copying of the elf packets */
+ err = mixart_load_elf( mgr, dsp );
+ if (err < 0) return err;
+
+ /* set elf status to copy finished */
+ writel_be( 2, MIXART_MEM( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET ));
+
+ /* wait for elf status == 4 */
+ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET, 1, 4, 300); /* 3sec */
+ if (err < 0) {
+ snd_printk( KERN_ERR "elf could not be started\n");
+ return err;
+ }
+
+ /* miXart waits at this point on the pointer to the flow table */
+ writel_be( (u32)mgr->flowinfo.addr, MIXART_MEM( mgr, MIXART_FLOWTABLE_PTR ) ); /* give pointer of flow table to miXart */
+
+ return 0; /* return, another xilinx file has to be loaded before */
+
+ case MIXART_AESEBUBOARD_XLX_INDEX:
+ default:
+
+ /* elf and xilinx should be loaded */
+ if( (status_elf != 4) || (status_xilinx != 4) ) {
+ printk( KERN_ERR "xilinx or elf not successfully loaded\n");
+ return -EIO; /* modprob -r may help ? */
+ }
+
+ /* wait for daughter detection != 0 */
+ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET, 0, 0, 30); /* 300msec */
+ if (err < 0) {
+ snd_printk( KERN_ERR "error starting elf file\n");
+ return err;
+ }
+
+ /* the board type can now be retrieved */
+ mgr->board_type = (DAUGHTER_TYPE_MASK & readl_be( MIXART_MEM( mgr, MIXART_PSEUDOREG_DBRD_TYPE_OFFSET)));
+
+ if (mgr->board_type == MIXART_DAUGHTER_TYPE_NONE)
+ break; /* no daughter board; the file does not have to be loaded, continue after the switch */
+
+ /* only if aesebu daughter board presence (elf code must run) */
+ if (mgr->board_type != MIXART_DAUGHTER_TYPE_AES )
+ return -EINVAL;
+
+ /* daughter should be idle */
+ if( status_daught != 0 ) {
+ printk( KERN_ERR "daughter load error ! status = %d\n", status_daught);
+ return -EIO; /* modprob -r may help ? */
+ }
+
+ /* check daughterboard xilinx validity */
+ if (((u32*)(dsp->data))[0] == 0xffffffff)
+ return -EINVAL;
+ if (dsp->size % 4)
+ return -EINVAL;
+
+ /* inform mixart about the size of the file */
+ writel_be( dsp->size, MIXART_MEM( mgr, MIXART_PSEUDOREG_DXLX_SIZE_OFFSET ));
+
+ /* set daughterboard status to 1 */
+ writel_be( 1, MIXART_MEM( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET ));
+
+ /* wait for status == 2 */
+ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 2, 30); /* 300msec */
+ if (err < 0) {
+ snd_printk( KERN_ERR "daughter board load error\n");
+ return err;
+ }
+
+ /* get the address where to write the file */
+ val = readl_be( MIXART_MEM( mgr, MIXART_PSEUDOREG_DXLX_BASE_ADDR_OFFSET ));
+ if (!val)
+ return -EINVAL;
+
+ /* copy daughterboard xilinx code */
+ memcpy_toio( MIXART_MEM( mgr, val), dsp->data, dsp->size);
+
+ /* set daughterboard status to 4 */
+ writel_be( 4, MIXART_MEM( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET ));
+
+ /* continue with init */
+ break;
+ } /* end of switch file index*/
+
+ /* wait for daughter status == 3 */
+ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 3, 300); /* 3sec */
+ if (err < 0) {
+ snd_printk( KERN_ERR "daughter board could not be initialised\n");
+ return err;
+ }
+
+ /* init mailbox (communication with embedded) */
+ snd_mixart_init_mailbox(mgr);
+
+ /* first communication with embedded */
+ err = mixart_first_init(mgr);
+ if (err < 0) {
+ snd_printk( KERN_ERR "miXart could not be set up\n");
+ return err;
+ }
+
+ /* create devices and mixer in accordance with HW options*/
+ for (card_index = 0; card_index < mgr->num_cards; card_index++) {
+ struct snd_mixart *chip = mgr->chip[card_index];
+
+ if ((err = snd_mixart_create_pcm(chip)) < 0)
+ return err;
+
+ if (card_index == 0) {
+ if ((err = snd_mixart_create_mixer(chip->mgr)) < 0)
+ return err;
+ }
+
+ if ((err = snd_card_register(chip->card)) < 0)
+ return err;
+ };
+
+ snd_printdd("miXart firmware downloaded and successfully set up\n");
+
+ return 0;
+}
+
+
+#if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE)
+#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */
+#define SND_MIXART_FW_LOADER /* use the standard firmware loader */
+#endif
+#endif
+
+#ifdef SND_MIXART_FW_LOADER
+
+int snd_mixart_setup_firmware(struct mixart_mgr *mgr)
+{
+ static char *fw_files[3] = {
+ "miXart8.xlx", "miXart8.elf", "miXart8AES.xlx"
+ };
+ char path[32];
+
+ const struct firmware *fw_entry;
+ int i, err;
+
+ for (i = 0; i < 3; i++) {
+ sprintf(path, "mixart/%s", fw_files[i]);
+ if (request_firmware(&fw_entry, path, &mgr->pci->dev)) {
+ snd_printk(KERN_ERR "miXart: can't load firmware %s\n", path);
+ return -ENOENT;
+ }
+ /* fake hwdep dsp record */
+ err = mixart_dsp_load(mgr, i, fw_entry);
+ release_firmware(fw_entry);
+ if (err < 0)
+ return err;
+ mgr->dsp_loaded |= 1 << i;
+ }
+ return 0;
+}
+
+MODULE_FIRMWARE("mixart/miXart8.xlx");
+MODULE_FIRMWARE("mixart/miXart8.elf");
+MODULE_FIRMWARE("mixart/miXart8AES.xlx");
+
+#else /* old style firmware loading */
+
+/* miXart hwdep interface id string */
+#define SND_MIXART_HWDEP_ID "miXart Loader"
+
+static int mixart_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+static int mixart_hwdep_release(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+static int mixart_hwdep_dsp_status(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_status *info)
+{
+ struct mixart_mgr *mgr = hw->private_data;
+
+ strcpy(info->id, "miXart");
+ info->num_dsps = MIXART_HARDW_FILES_MAX_INDEX;
+
+ if (mgr->dsp_loaded & (1 << MIXART_MOTHERBOARD_ELF_INDEX))
+ info->chip_ready = 1;
+
+ info->version = MIXART_DRIVER_VERSION;
+ return 0;
+}
+
+static int mixart_hwdep_dsp_load(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_image *dsp)
+{
+ struct mixart_mgr* mgr = hw->private_data;
+ struct firmware fw;
+ int err;
+
+ fw.size = dsp->length;
+ fw.data = vmalloc(dsp->length);
+ if (! fw.data) {
+ snd_printk(KERN_ERR "miXart: cannot allocate image size %d\n",
+ (int)dsp->length);
+ return -ENOMEM;
+ }
+ if (copy_from_user((void *) fw.data, dsp->image, dsp->length)) {
+ vfree(fw.data);
+ return -EFAULT;
+ }
+ err = mixart_dsp_load(mgr, dsp->index, &fw);
+ vfree(fw.data);
+ if (err < 0)
+ return err;
+ mgr->dsp_loaded |= 1 << dsp->index;
+ return err;
+}
+
+int snd_mixart_setup_firmware(struct mixart_mgr *mgr)
+{
+ int err;
+ struct snd_hwdep *hw;
+
+ /* only create hwdep interface for first cardX (see "index" module parameter)*/
+ if ((err = snd_hwdep_new(mgr->chip[0]->card, SND_MIXART_HWDEP_ID, 0, &hw)) < 0)
+ return err;
+
+ hw->iface = SNDRV_HWDEP_IFACE_MIXART;
+ hw->private_data = mgr;
+ hw->ops.open = mixart_hwdep_open;
+ hw->ops.release = mixart_hwdep_release;
+ hw->ops.dsp_status = mixart_hwdep_dsp_status;
+ hw->ops.dsp_load = mixart_hwdep_dsp_load;
+ hw->exclusive = 1;
+ sprintf(hw->name, SND_MIXART_HWDEP_ID);
+ mgr->dsp_loaded = 0;
+
+ return snd_card_register(mgr->chip[0]->card);
+}
+
+#endif /* SND_MIXART_FW_LOADER */
diff --git a/sound/pci/mixart/mixart_hwdep.h b/sound/pci/mixart/mixart_hwdep.h
new file mode 100644
index 0000000..a46f508
--- /dev/null
+++ b/sound/pci/mixart/mixart_hwdep.h
@@ -0,0 +1,145 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * definitions and makros for basic card access
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_MIXART_HWDEP_H
+#define __SOUND_MIXART_HWDEP_H
+
+#include <sound/hwdep.h>
+
+#define readl_be(x) be32_to_cpu(__raw_readl(x))
+#define writel_be(data,addr) __raw_writel(cpu_to_be32(data),addr)
+
+#define readl_le(x) le32_to_cpu(__raw_readl(x))
+#define writel_le(data,addr) __raw_writel(cpu_to_le32(data),addr)
+
+#define MIXART_MEM(mgr,x) ((mgr)->mem[0].virt + (x))
+#define MIXART_REG(mgr,x) ((mgr)->mem[1].virt + (x))
+
+
+/* Daughter board Type */
+#define DAUGHTER_TYPE_MASK 0x0F
+#define DAUGHTER_VER_MASK 0xF0
+#define DAUGHTER_TYPEVER_MASK (DAUGHTER_TYPE_MASK|DAUGHTER_VER_MASK)
+
+#define MIXART_DAUGHTER_TYPE_NONE 0x00
+#define MIXART_DAUGHTER_TYPE_COBRANET 0x08
+#define MIXART_DAUGHTER_TYPE_AES 0x0E
+
+
+
+#define MIXART_BA0_SIZE (16 * 1024 * 1024) /* 16M */
+#define MIXART_BA1_SIZE (4 * 1024) /* 4k */
+
+/*
+ * -----------BAR 0 --------------------------------------------------------------------------------------------------------
+ */
+#define MIXART_PSEUDOREG 0x2000 /* base address for pseudoregister */
+
+#define MIXART_PSEUDOREG_BOARDNUMBER MIXART_PSEUDOREG+0 /* board number */
+
+/* perfmeter (available when elf loaded)*/
+#define MIXART_PSEUDOREG_PERF_STREAM_LOAD_OFFSET MIXART_PSEUDOREG+0x70 /* streaming load */
+#define MIXART_PSEUDOREG_PERF_SYSTEM_LOAD_OFFSET MIXART_PSEUDOREG+0x78 /* system load (reference)*/
+#define MIXART_PSEUDOREG_PERF_MAILBX_LOAD_OFFSET MIXART_PSEUDOREG+0x7C /* mailbox load */
+#define MIXART_PSEUDOREG_PERF_INTERR_LOAD_OFFSET MIXART_PSEUDOREG+0x74 /* interrupt handling load */
+
+/* motherboard xilinx loader info */
+#define MIXART_PSEUDOREG_MXLX_BASE_ADDR_OFFSET MIXART_PSEUDOREG+0x9C /* 0x00600000 */
+#define MIXART_PSEUDOREG_MXLX_SIZE_OFFSET MIXART_PSEUDOREG+0xA0 /* xilinx size in bytes */
+#define MIXART_PSEUDOREG_MXLX_STATUS_OFFSET MIXART_PSEUDOREG+0xA4 /* status = EMBEBBED_STAT_XXX */
+
+/* elf loader info */
+#define MIXART_PSEUDOREG_ELF_STATUS_OFFSET MIXART_PSEUDOREG+0xB0 /* status = EMBEBBED_STAT_XXX */
+
+/*
+* after the elf code is loaded, and the flowtable info was passed to it,
+* the driver polls on this address, until it shows 1 (presence) or 2 (absence)
+* once it is non-zero, the daughter board type may be read
+*/
+#define MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET MIXART_PSEUDOREG+0x990
+
+/* Global info structure */
+#define MIXART_PSEUDOREG_DBRD_TYPE_OFFSET MIXART_PSEUDOREG+0x994 /* Type and version of daughterboard */
+
+
+/* daughterboard xilinx loader info */
+#define MIXART_PSEUDOREG_DXLX_BASE_ADDR_OFFSET MIXART_PSEUDOREG+0x998 /* get the address here where to write the file */
+#define MIXART_PSEUDOREG_DXLX_SIZE_OFFSET MIXART_PSEUDOREG+0x99C /* xilinx size in bytes */
+#define MIXART_PSEUDOREG_DXLX_STATUS_OFFSET MIXART_PSEUDOREG+0x9A0 /* status = EMBEBBED_STAT_XXX */
+
+/* */
+#define MIXART_FLOWTABLE_PTR 0x3000 /* pointer to flow table */
+
+/* mailbox addresses */
+
+/* message DRV -> EMB */
+#define MSG_INBOUND_POST_HEAD 0x010008 /* DRV posts MF + increment4 */
+#define MSG_INBOUND_POST_TAIL 0x01000C /* EMB gets MF + increment4 */
+/* message EMB -> DRV */
+#define MSG_OUTBOUND_POST_TAIL 0x01001C /* DRV gets MF + increment4 */
+#define MSG_OUTBOUND_POST_HEAD 0x010018 /* EMB posts MF + increment4 */
+/* Get Free Frames */
+#define MSG_INBOUND_FREE_TAIL 0x010004 /* DRV gets MFA + increment4 */
+#define MSG_OUTBOUND_FREE_TAIL 0x010014 /* EMB gets MFA + increment4 */
+/* Put Free Frames */
+#define MSG_OUTBOUND_FREE_HEAD 0x010010 /* DRV puts MFA + increment4 */
+#define MSG_INBOUND_FREE_HEAD 0x010000 /* EMB puts MFA + increment4 */
+
+/* firmware addresses of the message fifos */
+#define MSG_BOUND_STACK_SIZE 0x004000 /* size of each following stack */
+/* posted messages */
+#define MSG_OUTBOUND_POST_STACK 0x108000 /* stack of messages to the DRV */
+#define MSG_INBOUND_POST_STACK 0x104000 /* stack of messages to the EMB */
+/* available empty messages */
+#define MSG_OUTBOUND_FREE_STACK 0x10C000 /* stack of free enveloped for EMB */
+#define MSG_INBOUND_FREE_STACK 0x100000 /* stack of free enveloped for DRV */
+
+
+/* defines for mailbox message frames */
+#define MSG_FRAME_OFFSET 0x64
+#define MSG_FRAME_SIZE 0x6400
+#define MSG_FRAME_NUMBER 32
+#define MSG_FROM_AGENT_ITMF_OFFSET (MSG_FRAME_OFFSET + (MSG_FRAME_SIZE * MSG_FRAME_NUMBER))
+#define MSG_TO_AGENT_ITMF_OFFSET (MSG_FROM_AGENT_ITMF_OFFSET + MSG_FRAME_SIZE)
+#define MSG_HOST_RSC_PROTECTION (MSG_TO_AGENT_ITMF_OFFSET + MSG_FRAME_SIZE)
+#define MSG_AGENT_RSC_PROTECTION (MSG_HOST_RSC_PROTECTION + 4)
+
+
+/*
+ * -----------BAR 1 --------------------------------------------------------------------------------------------------------
+ */
+
+/* interrupt addresses and constants */
+#define MIXART_PCI_OMIMR_OFFSET 0x34 /* outbound message interrupt mask register */
+#define MIXART_PCI_OMISR_OFFSET 0x30 /* outbound message interrupt status register */
+#define MIXART_PCI_ODBR_OFFSET 0x60 /* outbound doorbell register */
+
+#define MIXART_BA1_BRUTAL_RESET_OFFSET 0x68 /* write 1 in LSBit to reset board */
+
+#define MIXART_HOST_ALL_INTERRUPT_MASKED 0x02B /* 0000 0010 1011 */
+#define MIXART_ALLOW_OUTBOUND_DOORBELL 0x023 /* 0000 0010 0011 */
+#define MIXART_OIDI 0x008 /* 0000 0000 1000 */
+
+
+int snd_mixart_setup_firmware(struct mixart_mgr *mgr);
+
+#endif /* __SOUND_MIXART_HWDEP_H */
diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c
new file mode 100644
index 0000000..3ba6174
--- /dev/null
+++ b/sound/pci/mixart/mixart_mixer.c
@@ -0,0 +1,1186 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * mixer callbacks
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/time.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include "mixart.h"
+#include "mixart_core.h"
+#include "mixart_hwdep.h"
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include "mixart_mixer.h"
+
+static u32 mixart_analog_level[256] = {
+ 0xc2c00000, /* [000] -96.0 dB */
+ 0xc2bf0000, /* [001] -95.5 dB */
+ 0xc2be0000, /* [002] -95.0 dB */
+ 0xc2bd0000, /* [003] -94.5 dB */
+ 0xc2bc0000, /* [004] -94.0 dB */
+ 0xc2bb0000, /* [005] -93.5 dB */
+ 0xc2ba0000, /* [006] -93.0 dB */
+ 0xc2b90000, /* [007] -92.5 dB */
+ 0xc2b80000, /* [008] -92.0 dB */
+ 0xc2b70000, /* [009] -91.5 dB */
+ 0xc2b60000, /* [010] -91.0 dB */
+ 0xc2b50000, /* [011] -90.5 dB */
+ 0xc2b40000, /* [012] -90.0 dB */
+ 0xc2b30000, /* [013] -89.5 dB */
+ 0xc2b20000, /* [014] -89.0 dB */
+ 0xc2b10000, /* [015] -88.5 dB */
+ 0xc2b00000, /* [016] -88.0 dB */
+ 0xc2af0000, /* [017] -87.5 dB */
+ 0xc2ae0000, /* [018] -87.0 dB */
+ 0xc2ad0000, /* [019] -86.5 dB */
+ 0xc2ac0000, /* [020] -86.0 dB */
+ 0xc2ab0000, /* [021] -85.5 dB */
+ 0xc2aa0000, /* [022] -85.0 dB */
+ 0xc2a90000, /* [023] -84.5 dB */
+ 0xc2a80000, /* [024] -84.0 dB */
+ 0xc2a70000, /* [025] -83.5 dB */
+ 0xc2a60000, /* [026] -83.0 dB */
+ 0xc2a50000, /* [027] -82.5 dB */
+ 0xc2a40000, /* [028] -82.0 dB */
+ 0xc2a30000, /* [029] -81.5 dB */
+ 0xc2a20000, /* [030] -81.0 dB */
+ 0xc2a10000, /* [031] -80.5 dB */
+ 0xc2a00000, /* [032] -80.0 dB */
+ 0xc29f0000, /* [033] -79.5 dB */
+ 0xc29e0000, /* [034] -79.0 dB */
+ 0xc29d0000, /* [035] -78.5 dB */
+ 0xc29c0000, /* [036] -78.0 dB */
+ 0xc29b0000, /* [037] -77.5 dB */
+ 0xc29a0000, /* [038] -77.0 dB */
+ 0xc2990000, /* [039] -76.5 dB */
+ 0xc2980000, /* [040] -76.0 dB */
+ 0xc2970000, /* [041] -75.5 dB */
+ 0xc2960000, /* [042] -75.0 dB */
+ 0xc2950000, /* [043] -74.5 dB */
+ 0xc2940000, /* [044] -74.0 dB */
+ 0xc2930000, /* [045] -73.5 dB */
+ 0xc2920000, /* [046] -73.0 dB */
+ 0xc2910000, /* [047] -72.5 dB */
+ 0xc2900000, /* [048] -72.0 dB */
+ 0xc28f0000, /* [049] -71.5 dB */
+ 0xc28e0000, /* [050] -71.0 dB */
+ 0xc28d0000, /* [051] -70.5 dB */
+ 0xc28c0000, /* [052] -70.0 dB */
+ 0xc28b0000, /* [053] -69.5 dB */
+ 0xc28a0000, /* [054] -69.0 dB */
+ 0xc2890000, /* [055] -68.5 dB */
+ 0xc2880000, /* [056] -68.0 dB */
+ 0xc2870000, /* [057] -67.5 dB */
+ 0xc2860000, /* [058] -67.0 dB */
+ 0xc2850000, /* [059] -66.5 dB */
+ 0xc2840000, /* [060] -66.0 dB */
+ 0xc2830000, /* [061] -65.5 dB */
+ 0xc2820000, /* [062] -65.0 dB */
+ 0xc2810000, /* [063] -64.5 dB */
+ 0xc2800000, /* [064] -64.0 dB */
+ 0xc27e0000, /* [065] -63.5 dB */
+ 0xc27c0000, /* [066] -63.0 dB */
+ 0xc27a0000, /* [067] -62.5 dB */
+ 0xc2780000, /* [068] -62.0 dB */
+ 0xc2760000, /* [069] -61.5 dB */
+ 0xc2740000, /* [070] -61.0 dB */
+ 0xc2720000, /* [071] -60.5 dB */
+ 0xc2700000, /* [072] -60.0 dB */
+ 0xc26e0000, /* [073] -59.5 dB */
+ 0xc26c0000, /* [074] -59.0 dB */
+ 0xc26a0000, /* [075] -58.5 dB */
+ 0xc2680000, /* [076] -58.0 dB */
+ 0xc2660000, /* [077] -57.5 dB */
+ 0xc2640000, /* [078] -57.0 dB */
+ 0xc2620000, /* [079] -56.5 dB */
+ 0xc2600000, /* [080] -56.0 dB */
+ 0xc25e0000, /* [081] -55.5 dB */
+ 0xc25c0000, /* [082] -55.0 dB */
+ 0xc25a0000, /* [083] -54.5 dB */
+ 0xc2580000, /* [084] -54.0 dB */
+ 0xc2560000, /* [085] -53.5 dB */
+ 0xc2540000, /* [086] -53.0 dB */
+ 0xc2520000, /* [087] -52.5 dB */
+ 0xc2500000, /* [088] -52.0 dB */
+ 0xc24e0000, /* [089] -51.5 dB */
+ 0xc24c0000, /* [090] -51.0 dB */
+ 0xc24a0000, /* [091] -50.5 dB */
+ 0xc2480000, /* [092] -50.0 dB */
+ 0xc2460000, /* [093] -49.5 dB */
+ 0xc2440000, /* [094] -49.0 dB */
+ 0xc2420000, /* [095] -48.5 dB */
+ 0xc2400000, /* [096] -48.0 dB */
+ 0xc23e0000, /* [097] -47.5 dB */
+ 0xc23c0000, /* [098] -47.0 dB */
+ 0xc23a0000, /* [099] -46.5 dB */
+ 0xc2380000, /* [100] -46.0 dB */
+ 0xc2360000, /* [101] -45.5 dB */
+ 0xc2340000, /* [102] -45.0 dB */
+ 0xc2320000, /* [103] -44.5 dB */
+ 0xc2300000, /* [104] -44.0 dB */
+ 0xc22e0000, /* [105] -43.5 dB */
+ 0xc22c0000, /* [106] -43.0 dB */
+ 0xc22a0000, /* [107] -42.5 dB */
+ 0xc2280000, /* [108] -42.0 dB */
+ 0xc2260000, /* [109] -41.5 dB */
+ 0xc2240000, /* [110] -41.0 dB */
+ 0xc2220000, /* [111] -40.5 dB */
+ 0xc2200000, /* [112] -40.0 dB */
+ 0xc21e0000, /* [113] -39.5 dB */
+ 0xc21c0000, /* [114] -39.0 dB */
+ 0xc21a0000, /* [115] -38.5 dB */
+ 0xc2180000, /* [116] -38.0 dB */
+ 0xc2160000, /* [117] -37.5 dB */
+ 0xc2140000, /* [118] -37.0 dB */
+ 0xc2120000, /* [119] -36.5 dB */
+ 0xc2100000, /* [120] -36.0 dB */
+ 0xc20e0000, /* [121] -35.5 dB */
+ 0xc20c0000, /* [122] -35.0 dB */
+ 0xc20a0000, /* [123] -34.5 dB */
+ 0xc2080000, /* [124] -34.0 dB */
+ 0xc2060000, /* [125] -33.5 dB */
+ 0xc2040000, /* [126] -33.0 dB */
+ 0xc2020000, /* [127] -32.5 dB */
+ 0xc2000000, /* [128] -32.0 dB */
+ 0xc1fc0000, /* [129] -31.5 dB */
+ 0xc1f80000, /* [130] -31.0 dB */
+ 0xc1f40000, /* [131] -30.5 dB */
+ 0xc1f00000, /* [132] -30.0 dB */
+ 0xc1ec0000, /* [133] -29.5 dB */
+ 0xc1e80000, /* [134] -29.0 dB */
+ 0xc1e40000, /* [135] -28.5 dB */
+ 0xc1e00000, /* [136] -28.0 dB */
+ 0xc1dc0000, /* [137] -27.5 dB */
+ 0xc1d80000, /* [138] -27.0 dB */
+ 0xc1d40000, /* [139] -26.5 dB */
+ 0xc1d00000, /* [140] -26.0 dB */
+ 0xc1cc0000, /* [141] -25.5 dB */
+ 0xc1c80000, /* [142] -25.0 dB */
+ 0xc1c40000, /* [143] -24.5 dB */
+ 0xc1c00000, /* [144] -24.0 dB */
+ 0xc1bc0000, /* [145] -23.5 dB */
+ 0xc1b80000, /* [146] -23.0 dB */
+ 0xc1b40000, /* [147] -22.5 dB */
+ 0xc1b00000, /* [148] -22.0 dB */
+ 0xc1ac0000, /* [149] -21.5 dB */
+ 0xc1a80000, /* [150] -21.0 dB */
+ 0xc1a40000, /* [151] -20.5 dB */
+ 0xc1a00000, /* [152] -20.0 dB */
+ 0xc19c0000, /* [153] -19.5 dB */
+ 0xc1980000, /* [154] -19.0 dB */
+ 0xc1940000, /* [155] -18.5 dB */
+ 0xc1900000, /* [156] -18.0 dB */
+ 0xc18c0000, /* [157] -17.5 dB */
+ 0xc1880000, /* [158] -17.0 dB */
+ 0xc1840000, /* [159] -16.5 dB */
+ 0xc1800000, /* [160] -16.0 dB */
+ 0xc1780000, /* [161] -15.5 dB */
+ 0xc1700000, /* [162] -15.0 dB */
+ 0xc1680000, /* [163] -14.5 dB */
+ 0xc1600000, /* [164] -14.0 dB */
+ 0xc1580000, /* [165] -13.5 dB */
+ 0xc1500000, /* [166] -13.0 dB */
+ 0xc1480000, /* [167] -12.5 dB */
+ 0xc1400000, /* [168] -12.0 dB */
+ 0xc1380000, /* [169] -11.5 dB */
+ 0xc1300000, /* [170] -11.0 dB */
+ 0xc1280000, /* [171] -10.5 dB */
+ 0xc1200000, /* [172] -10.0 dB */
+ 0xc1180000, /* [173] -9.5 dB */
+ 0xc1100000, /* [174] -9.0 dB */
+ 0xc1080000, /* [175] -8.5 dB */
+ 0xc1000000, /* [176] -8.0 dB */
+ 0xc0f00000, /* [177] -7.5 dB */
+ 0xc0e00000, /* [178] -7.0 dB */
+ 0xc0d00000, /* [179] -6.5 dB */
+ 0xc0c00000, /* [180] -6.0 dB */
+ 0xc0b00000, /* [181] -5.5 dB */
+ 0xc0a00000, /* [182] -5.0 dB */
+ 0xc0900000, /* [183] -4.5 dB */
+ 0xc0800000, /* [184] -4.0 dB */
+ 0xc0600000, /* [185] -3.5 dB */
+ 0xc0400000, /* [186] -3.0 dB */
+ 0xc0200000, /* [187] -2.5 dB */
+ 0xc0000000, /* [188] -2.0 dB */
+ 0xbfc00000, /* [189] -1.5 dB */
+ 0xbf800000, /* [190] -1.0 dB */
+ 0xbf000000, /* [191] -0.5 dB */
+ 0x00000000, /* [192] 0.0 dB */
+ 0x3f000000, /* [193] 0.5 dB */
+ 0x3f800000, /* [194] 1.0 dB */
+ 0x3fc00000, /* [195] 1.5 dB */
+ 0x40000000, /* [196] 2.0 dB */
+ 0x40200000, /* [197] 2.5 dB */
+ 0x40400000, /* [198] 3.0 dB */
+ 0x40600000, /* [199] 3.5 dB */
+ 0x40800000, /* [200] 4.0 dB */
+ 0x40900000, /* [201] 4.5 dB */
+ 0x40a00000, /* [202] 5.0 dB */
+ 0x40b00000, /* [203] 5.5 dB */
+ 0x40c00000, /* [204] 6.0 dB */
+ 0x40d00000, /* [205] 6.5 dB */
+ 0x40e00000, /* [206] 7.0 dB */
+ 0x40f00000, /* [207] 7.5 dB */
+ 0x41000000, /* [208] 8.0 dB */
+ 0x41080000, /* [209] 8.5 dB */
+ 0x41100000, /* [210] 9.0 dB */
+ 0x41180000, /* [211] 9.5 dB */
+ 0x41200000, /* [212] 10.0 dB */
+ 0x41280000, /* [213] 10.5 dB */
+ 0x41300000, /* [214] 11.0 dB */
+ 0x41380000, /* [215] 11.5 dB */
+ 0x41400000, /* [216] 12.0 dB */
+ 0x41480000, /* [217] 12.5 dB */
+ 0x41500000, /* [218] 13.0 dB */
+ 0x41580000, /* [219] 13.5 dB */
+ 0x41600000, /* [220] 14.0 dB */
+ 0x41680000, /* [221] 14.5 dB */
+ 0x41700000, /* [222] 15.0 dB */
+ 0x41780000, /* [223] 15.5 dB */
+ 0x41800000, /* [224] 16.0 dB */
+ 0x41840000, /* [225] 16.5 dB */
+ 0x41880000, /* [226] 17.0 dB */
+ 0x418c0000, /* [227] 17.5 dB */
+ 0x41900000, /* [228] 18.0 dB */
+ 0x41940000, /* [229] 18.5 dB */
+ 0x41980000, /* [230] 19.0 dB */
+ 0x419c0000, /* [231] 19.5 dB */
+ 0x41a00000, /* [232] 20.0 dB */
+ 0x41a40000, /* [233] 20.5 dB */
+ 0x41a80000, /* [234] 21.0 dB */
+ 0x41ac0000, /* [235] 21.5 dB */
+ 0x41b00000, /* [236] 22.0 dB */
+ 0x41b40000, /* [237] 22.5 dB */
+ 0x41b80000, /* [238] 23.0 dB */
+ 0x41bc0000, /* [239] 23.5 dB */
+ 0x41c00000, /* [240] 24.0 dB */
+ 0x41c40000, /* [241] 24.5 dB */
+ 0x41c80000, /* [242] 25.0 dB */
+ 0x41cc0000, /* [243] 25.5 dB */
+ 0x41d00000, /* [244] 26.0 dB */
+ 0x41d40000, /* [245] 26.5 dB */
+ 0x41d80000, /* [246] 27.0 dB */
+ 0x41dc0000, /* [247] 27.5 dB */
+ 0x41e00000, /* [248] 28.0 dB */
+ 0x41e40000, /* [249] 28.5 dB */
+ 0x41e80000, /* [250] 29.0 dB */
+ 0x41ec0000, /* [251] 29.5 dB */
+ 0x41f00000, /* [252] 30.0 dB */
+ 0x41f40000, /* [253] 30.5 dB */
+ 0x41f80000, /* [254] 31.0 dB */
+ 0x41fc0000, /* [255] 31.5 dB */
+};
+
+#define MIXART_ANALOG_CAPTURE_LEVEL_MIN 0 /* -96.0 dB + 8.0 dB = -88.0 dB */
+#define MIXART_ANALOG_CAPTURE_LEVEL_MAX 255 /* 31.5 dB + 8.0 dB = 39.5 dB */
+#define MIXART_ANALOG_CAPTURE_ZERO_LEVEL 176 /* -8.0 dB + 8.0 dB = 0.0 dB */
+
+#define MIXART_ANALOG_PLAYBACK_LEVEL_MIN 0 /* -96.0 dB + 1.5 dB = -94.5 dB (possible is down to (-114.0+1.5)dB) */
+#define MIXART_ANALOG_PLAYBACK_LEVEL_MAX 192 /* 0.0 dB + 1.5 dB = 1.5 dB */
+#define MIXART_ANALOG_PLAYBACK_ZERO_LEVEL 189 /* -1.5 dB + 1.5 dB = 0.0 dB */
+
+static int mixart_update_analog_audio_level(struct snd_mixart* chip, int is_capture)
+{
+ int i, err;
+ struct mixart_msg request;
+ struct mixart_io_level io_level;
+ struct mixart_return_uid resp;
+
+ memset(&io_level, 0, sizeof(io_level));
+ io_level.channel = -1; /* left and right */
+
+ for(i=0; i<2; i++) {
+ if(is_capture) {
+ io_level.level[i].analog_level = mixart_analog_level[chip->analog_capture_volume[i]];
+ } else {
+ if(chip->analog_playback_active[i])
+ io_level.level[i].analog_level = mixart_analog_level[chip->analog_playback_volume[i]];
+ else
+ io_level.level[i].analog_level = mixart_analog_level[MIXART_ANALOG_PLAYBACK_LEVEL_MIN];
+ }
+ }
+
+ if(is_capture) request.uid = chip->uid_in_analog_physio;
+ else request.uid = chip->uid_out_analog_physio;
+ request.message_id = MSG_PHYSICALIO_SET_LEVEL;
+ request.data = &io_level;
+ request.size = sizeof(io_level);
+
+ err = snd_mixart_send_msg(chip->mgr, &request, sizeof(resp), &resp);
+ if((err<0) || (resp.error_code)) {
+ snd_printk(KERN_DEBUG "error MSG_PHYSICALIO_SET_LEVEL card(%d) is_capture(%d) error_code(%x)\n", chip->chip_idx, is_capture, resp.error_code);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * analog level control
+ */
+static int mixart_analog_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ if(kcontrol->private_value == 0) { /* playback */
+ uinfo->value.integer.min = MIXART_ANALOG_PLAYBACK_LEVEL_MIN; /* -96 dB */
+ uinfo->value.integer.max = MIXART_ANALOG_PLAYBACK_LEVEL_MAX; /* 0 dB */
+ } else { /* capture */
+ uinfo->value.integer.min = MIXART_ANALOG_CAPTURE_LEVEL_MIN; /* -96 dB */
+ uinfo->value.integer.max = MIXART_ANALOG_CAPTURE_LEVEL_MAX; /* 31.5 dB */
+ }
+ return 0;
+}
+
+static int mixart_analog_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if(kcontrol->private_value == 0) { /* playback */
+ ucontrol->value.integer.value[0] = chip->analog_playback_volume[0];
+ ucontrol->value.integer.value[1] = chip->analog_playback_volume[1];
+ } else { /* capture */
+ ucontrol->value.integer.value[0] = chip->analog_capture_volume[0];
+ ucontrol->value.integer.value[1] = chip->analog_capture_volume[1];
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int is_capture, i;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ is_capture = (kcontrol->private_value != 0);
+ for (i = 0; i < 2; i++) {
+ int new_volume = ucontrol->value.integer.value[i];
+ int *stored_volume = is_capture ?
+ &chip->analog_capture_volume[i] :
+ &chip->analog_playback_volume[i];
+ if (is_capture) {
+ if (new_volume < MIXART_ANALOG_CAPTURE_LEVEL_MIN ||
+ new_volume > MIXART_ANALOG_CAPTURE_LEVEL_MAX)
+ continue;
+ } else {
+ if (new_volume < MIXART_ANALOG_PLAYBACK_LEVEL_MIN ||
+ new_volume > MIXART_ANALOG_PLAYBACK_LEVEL_MAX)
+ continue;
+ }
+ if (*stored_volume != new_volume) {
+ *stored_volume = new_volume;
+ changed = 1;
+ }
+ }
+ if (changed)
+ mixart_update_analog_audio_level(chip, is_capture);
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_analog, -9600, 50, 0);
+
+static struct snd_kcontrol_new mixart_control_analog_level = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ /* name will be filled later */
+ .info = mixart_analog_vol_info,
+ .get = mixart_analog_vol_get,
+ .put = mixart_analog_vol_put,
+ .tlv = { .p = db_scale_analog },
+};
+
+/* shared */
+#define mixart_sw_info snd_ctl_boolean_stereo_info
+
+static int mixart_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->analog_playback_active[0];
+ ucontrol->value.integer.value[1] = chip->analog_playback_active[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int mixart_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int i, changed = 0;
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for (i = 0; i < 2; i++) {
+ if (chip->analog_playback_active[i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->analog_playback_active[i] =
+ !!ucontrol->value.integer.value[i];
+ changed = 1;
+ }
+ }
+ if (changed) /* update playback levels */
+ mixart_update_analog_audio_level(chip, 0);
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new mixart_control_output_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = mixart_sw_info, /* shared */
+ .get = mixart_audio_sw_get,
+ .put = mixart_audio_sw_put
+};
+
+static u32 mixart_digital_level[256] = {
+ 0x00000000, /* [000] = 0.00e+000 = mute if <= -109.5dB */
+ 0x366e1c7a, /* [001] = 3.55e-006 = pow(10.0, 0.05 * -109.0dB) */
+ 0x367c3860, /* [002] = 3.76e-006 = pow(10.0, 0.05 * -108.5dB) */
+ 0x36859525, /* [003] = 3.98e-006 = pow(10.0, 0.05 * -108.0dB) */
+ 0x368d7f74, /* [004] = 4.22e-006 = pow(10.0, 0.05 * -107.5dB) */
+ 0x3695e1d4, /* [005] = 4.47e-006 = pow(10.0, 0.05 * -107.0dB) */
+ 0x369ec362, /* [006] = 4.73e-006 = pow(10.0, 0.05 * -106.5dB) */
+ 0x36a82ba8, /* [007] = 5.01e-006 = pow(10.0, 0.05 * -106.0dB) */
+ 0x36b222a0, /* [008] = 5.31e-006 = pow(10.0, 0.05 * -105.5dB) */
+ 0x36bcb0c1, /* [009] = 5.62e-006 = pow(10.0, 0.05 * -105.0dB) */
+ 0x36c7defd, /* [010] = 5.96e-006 = pow(10.0, 0.05 * -104.5dB) */
+ 0x36d3b6d3, /* [011] = 6.31e-006 = pow(10.0, 0.05 * -104.0dB) */
+ 0x36e0424e, /* [012] = 6.68e-006 = pow(10.0, 0.05 * -103.5dB) */
+ 0x36ed8c14, /* [013] = 7.08e-006 = pow(10.0, 0.05 * -103.0dB) */
+ 0x36fb9f6c, /* [014] = 7.50e-006 = pow(10.0, 0.05 * -102.5dB) */
+ 0x37054423, /* [015] = 7.94e-006 = pow(10.0, 0.05 * -102.0dB) */
+ 0x370d29a5, /* [016] = 8.41e-006 = pow(10.0, 0.05 * -101.5dB) */
+ 0x371586f0, /* [017] = 8.91e-006 = pow(10.0, 0.05 * -101.0dB) */
+ 0x371e631b, /* [018] = 9.44e-006 = pow(10.0, 0.05 * -100.5dB) */
+ 0x3727c5ac, /* [019] = 1.00e-005 = pow(10.0, 0.05 * -100.0dB) */
+ 0x3731b69a, /* [020] = 1.06e-005 = pow(10.0, 0.05 * -99.5dB) */
+ 0x373c3e53, /* [021] = 1.12e-005 = pow(10.0, 0.05 * -99.0dB) */
+ 0x374765c8, /* [022] = 1.19e-005 = pow(10.0, 0.05 * -98.5dB) */
+ 0x3753366f, /* [023] = 1.26e-005 = pow(10.0, 0.05 * -98.0dB) */
+ 0x375fba4f, /* [024] = 1.33e-005 = pow(10.0, 0.05 * -97.5dB) */
+ 0x376cfc07, /* [025] = 1.41e-005 = pow(10.0, 0.05 * -97.0dB) */
+ 0x377b06d5, /* [026] = 1.50e-005 = pow(10.0, 0.05 * -96.5dB) */
+ 0x3784f352, /* [027] = 1.58e-005 = pow(10.0, 0.05 * -96.0dB) */
+ 0x378cd40b, /* [028] = 1.68e-005 = pow(10.0, 0.05 * -95.5dB) */
+ 0x37952c42, /* [029] = 1.78e-005 = pow(10.0, 0.05 * -95.0dB) */
+ 0x379e030e, /* [030] = 1.88e-005 = pow(10.0, 0.05 * -94.5dB) */
+ 0x37a75fef, /* [031] = 2.00e-005 = pow(10.0, 0.05 * -94.0dB) */
+ 0x37b14ad5, /* [032] = 2.11e-005 = pow(10.0, 0.05 * -93.5dB) */
+ 0x37bbcc2c, /* [033] = 2.24e-005 = pow(10.0, 0.05 * -93.0dB) */
+ 0x37c6ecdd, /* [034] = 2.37e-005 = pow(10.0, 0.05 * -92.5dB) */
+ 0x37d2b65a, /* [035] = 2.51e-005 = pow(10.0, 0.05 * -92.0dB) */
+ 0x37df32a3, /* [036] = 2.66e-005 = pow(10.0, 0.05 * -91.5dB) */
+ 0x37ec6c50, /* [037] = 2.82e-005 = pow(10.0, 0.05 * -91.0dB) */
+ 0x37fa6e9b, /* [038] = 2.99e-005 = pow(10.0, 0.05 * -90.5dB) */
+ 0x3804a2b3, /* [039] = 3.16e-005 = pow(10.0, 0.05 * -90.0dB) */
+ 0x380c7ea4, /* [040] = 3.35e-005 = pow(10.0, 0.05 * -89.5dB) */
+ 0x3814d1cc, /* [041] = 3.55e-005 = pow(10.0, 0.05 * -89.0dB) */
+ 0x381da33c, /* [042] = 3.76e-005 = pow(10.0, 0.05 * -88.5dB) */
+ 0x3826fa6f, /* [043] = 3.98e-005 = pow(10.0, 0.05 * -88.0dB) */
+ 0x3830df51, /* [044] = 4.22e-005 = pow(10.0, 0.05 * -87.5dB) */
+ 0x383b5a49, /* [045] = 4.47e-005 = pow(10.0, 0.05 * -87.0dB) */
+ 0x3846743b, /* [046] = 4.73e-005 = pow(10.0, 0.05 * -86.5dB) */
+ 0x38523692, /* [047] = 5.01e-005 = pow(10.0, 0.05 * -86.0dB) */
+ 0x385eab48, /* [048] = 5.31e-005 = pow(10.0, 0.05 * -85.5dB) */
+ 0x386bdcf1, /* [049] = 5.62e-005 = pow(10.0, 0.05 * -85.0dB) */
+ 0x3879d6bc, /* [050] = 5.96e-005 = pow(10.0, 0.05 * -84.5dB) */
+ 0x38845244, /* [051] = 6.31e-005 = pow(10.0, 0.05 * -84.0dB) */
+ 0x388c2971, /* [052] = 6.68e-005 = pow(10.0, 0.05 * -83.5dB) */
+ 0x3894778d, /* [053] = 7.08e-005 = pow(10.0, 0.05 * -83.0dB) */
+ 0x389d43a4, /* [054] = 7.50e-005 = pow(10.0, 0.05 * -82.5dB) */
+ 0x38a6952c, /* [055] = 7.94e-005 = pow(10.0, 0.05 * -82.0dB) */
+ 0x38b0740f, /* [056] = 8.41e-005 = pow(10.0, 0.05 * -81.5dB) */
+ 0x38bae8ac, /* [057] = 8.91e-005 = pow(10.0, 0.05 * -81.0dB) */
+ 0x38c5fbe2, /* [058] = 9.44e-005 = pow(10.0, 0.05 * -80.5dB) */
+ 0x38d1b717, /* [059] = 1.00e-004 = pow(10.0, 0.05 * -80.0dB) */
+ 0x38de2440, /* [060] = 1.06e-004 = pow(10.0, 0.05 * -79.5dB) */
+ 0x38eb4de8, /* [061] = 1.12e-004 = pow(10.0, 0.05 * -79.0dB) */
+ 0x38f93f3a, /* [062] = 1.19e-004 = pow(10.0, 0.05 * -78.5dB) */
+ 0x39040206, /* [063] = 1.26e-004 = pow(10.0, 0.05 * -78.0dB) */
+ 0x390bd472, /* [064] = 1.33e-004 = pow(10.0, 0.05 * -77.5dB) */
+ 0x39141d84, /* [065] = 1.41e-004 = pow(10.0, 0.05 * -77.0dB) */
+ 0x391ce445, /* [066] = 1.50e-004 = pow(10.0, 0.05 * -76.5dB) */
+ 0x39263027, /* [067] = 1.58e-004 = pow(10.0, 0.05 * -76.0dB) */
+ 0x3930090d, /* [068] = 1.68e-004 = pow(10.0, 0.05 * -75.5dB) */
+ 0x393a7753, /* [069] = 1.78e-004 = pow(10.0, 0.05 * -75.0dB) */
+ 0x394583d2, /* [070] = 1.88e-004 = pow(10.0, 0.05 * -74.5dB) */
+ 0x395137ea, /* [071] = 2.00e-004 = pow(10.0, 0.05 * -74.0dB) */
+ 0x395d9d8a, /* [072] = 2.11e-004 = pow(10.0, 0.05 * -73.5dB) */
+ 0x396abf37, /* [073] = 2.24e-004 = pow(10.0, 0.05 * -73.0dB) */
+ 0x3978a814, /* [074] = 2.37e-004 = pow(10.0, 0.05 * -72.5dB) */
+ 0x3983b1f8, /* [075] = 2.51e-004 = pow(10.0, 0.05 * -72.0dB) */
+ 0x398b7fa6, /* [076] = 2.66e-004 = pow(10.0, 0.05 * -71.5dB) */
+ 0x3993c3b2, /* [077] = 2.82e-004 = pow(10.0, 0.05 * -71.0dB) */
+ 0x399c8521, /* [078] = 2.99e-004 = pow(10.0, 0.05 * -70.5dB) */
+ 0x39a5cb5f, /* [079] = 3.16e-004 = pow(10.0, 0.05 * -70.0dB) */
+ 0x39af9e4d, /* [080] = 3.35e-004 = pow(10.0, 0.05 * -69.5dB) */
+ 0x39ba063f, /* [081] = 3.55e-004 = pow(10.0, 0.05 * -69.0dB) */
+ 0x39c50c0b, /* [082] = 3.76e-004 = pow(10.0, 0.05 * -68.5dB) */
+ 0x39d0b90a, /* [083] = 3.98e-004 = pow(10.0, 0.05 * -68.0dB) */
+ 0x39dd1726, /* [084] = 4.22e-004 = pow(10.0, 0.05 * -67.5dB) */
+ 0x39ea30db, /* [085] = 4.47e-004 = pow(10.0, 0.05 * -67.0dB) */
+ 0x39f81149, /* [086] = 4.73e-004 = pow(10.0, 0.05 * -66.5dB) */
+ 0x3a03621b, /* [087] = 5.01e-004 = pow(10.0, 0.05 * -66.0dB) */
+ 0x3a0b2b0d, /* [088] = 5.31e-004 = pow(10.0, 0.05 * -65.5dB) */
+ 0x3a136a16, /* [089] = 5.62e-004 = pow(10.0, 0.05 * -65.0dB) */
+ 0x3a1c2636, /* [090] = 5.96e-004 = pow(10.0, 0.05 * -64.5dB) */
+ 0x3a2566d5, /* [091] = 6.31e-004 = pow(10.0, 0.05 * -64.0dB) */
+ 0x3a2f33cd, /* [092] = 6.68e-004 = pow(10.0, 0.05 * -63.5dB) */
+ 0x3a399570, /* [093] = 7.08e-004 = pow(10.0, 0.05 * -63.0dB) */
+ 0x3a44948c, /* [094] = 7.50e-004 = pow(10.0, 0.05 * -62.5dB) */
+ 0x3a503a77, /* [095] = 7.94e-004 = pow(10.0, 0.05 * -62.0dB) */
+ 0x3a5c9112, /* [096] = 8.41e-004 = pow(10.0, 0.05 * -61.5dB) */
+ 0x3a69a2d7, /* [097] = 8.91e-004 = pow(10.0, 0.05 * -61.0dB) */
+ 0x3a777ada, /* [098] = 9.44e-004 = pow(10.0, 0.05 * -60.5dB) */
+ 0x3a83126f, /* [099] = 1.00e-003 = pow(10.0, 0.05 * -60.0dB) */
+ 0x3a8ad6a8, /* [100] = 1.06e-003 = pow(10.0, 0.05 * -59.5dB) */
+ 0x3a9310b1, /* [101] = 1.12e-003 = pow(10.0, 0.05 * -59.0dB) */
+ 0x3a9bc784, /* [102] = 1.19e-003 = pow(10.0, 0.05 * -58.5dB) */
+ 0x3aa50287, /* [103] = 1.26e-003 = pow(10.0, 0.05 * -58.0dB) */
+ 0x3aaec98e, /* [104] = 1.33e-003 = pow(10.0, 0.05 * -57.5dB) */
+ 0x3ab924e5, /* [105] = 1.41e-003 = pow(10.0, 0.05 * -57.0dB) */
+ 0x3ac41d56, /* [106] = 1.50e-003 = pow(10.0, 0.05 * -56.5dB) */
+ 0x3acfbc31, /* [107] = 1.58e-003 = pow(10.0, 0.05 * -56.0dB) */
+ 0x3adc0b51, /* [108] = 1.68e-003 = pow(10.0, 0.05 * -55.5dB) */
+ 0x3ae91528, /* [109] = 1.78e-003 = pow(10.0, 0.05 * -55.0dB) */
+ 0x3af6e4c6, /* [110] = 1.88e-003 = pow(10.0, 0.05 * -54.5dB) */
+ 0x3b02c2f2, /* [111] = 2.00e-003 = pow(10.0, 0.05 * -54.0dB) */
+ 0x3b0a8276, /* [112] = 2.11e-003 = pow(10.0, 0.05 * -53.5dB) */
+ 0x3b12b782, /* [113] = 2.24e-003 = pow(10.0, 0.05 * -53.0dB) */
+ 0x3b1b690d, /* [114] = 2.37e-003 = pow(10.0, 0.05 * -52.5dB) */
+ 0x3b249e76, /* [115] = 2.51e-003 = pow(10.0, 0.05 * -52.0dB) */
+ 0x3b2e5f8f, /* [116] = 2.66e-003 = pow(10.0, 0.05 * -51.5dB) */
+ 0x3b38b49f, /* [117] = 2.82e-003 = pow(10.0, 0.05 * -51.0dB) */
+ 0x3b43a669, /* [118] = 2.99e-003 = pow(10.0, 0.05 * -50.5dB) */
+ 0x3b4f3e37, /* [119] = 3.16e-003 = pow(10.0, 0.05 * -50.0dB) */
+ 0x3b5b85e0, /* [120] = 3.35e-003 = pow(10.0, 0.05 * -49.5dB) */
+ 0x3b6887cf, /* [121] = 3.55e-003 = pow(10.0, 0.05 * -49.0dB) */
+ 0x3b764f0e, /* [122] = 3.76e-003 = pow(10.0, 0.05 * -48.5dB) */
+ 0x3b8273a6, /* [123] = 3.98e-003 = pow(10.0, 0.05 * -48.0dB) */
+ 0x3b8a2e77, /* [124] = 4.22e-003 = pow(10.0, 0.05 * -47.5dB) */
+ 0x3b925e89, /* [125] = 4.47e-003 = pow(10.0, 0.05 * -47.0dB) */
+ 0x3b9b0ace, /* [126] = 4.73e-003 = pow(10.0, 0.05 * -46.5dB) */
+ 0x3ba43aa2, /* [127] = 5.01e-003 = pow(10.0, 0.05 * -46.0dB) */
+ 0x3badf5d1, /* [128] = 5.31e-003 = pow(10.0, 0.05 * -45.5dB) */
+ 0x3bb8449c, /* [129] = 5.62e-003 = pow(10.0, 0.05 * -45.0dB) */
+ 0x3bc32fc3, /* [130] = 5.96e-003 = pow(10.0, 0.05 * -44.5dB) */
+ 0x3bcec08a, /* [131] = 6.31e-003 = pow(10.0, 0.05 * -44.0dB) */
+ 0x3bdb00c0, /* [132] = 6.68e-003 = pow(10.0, 0.05 * -43.5dB) */
+ 0x3be7facc, /* [133] = 7.08e-003 = pow(10.0, 0.05 * -43.0dB) */
+ 0x3bf5b9b0, /* [134] = 7.50e-003 = pow(10.0, 0.05 * -42.5dB) */
+ 0x3c02248a, /* [135] = 7.94e-003 = pow(10.0, 0.05 * -42.0dB) */
+ 0x3c09daac, /* [136] = 8.41e-003 = pow(10.0, 0.05 * -41.5dB) */
+ 0x3c1205c6, /* [137] = 8.91e-003 = pow(10.0, 0.05 * -41.0dB) */
+ 0x3c1aacc8, /* [138] = 9.44e-003 = pow(10.0, 0.05 * -40.5dB) */
+ 0x3c23d70a, /* [139] = 1.00e-002 = pow(10.0, 0.05 * -40.0dB) */
+ 0x3c2d8c52, /* [140] = 1.06e-002 = pow(10.0, 0.05 * -39.5dB) */
+ 0x3c37d4dd, /* [141] = 1.12e-002 = pow(10.0, 0.05 * -39.0dB) */
+ 0x3c42b965, /* [142] = 1.19e-002 = pow(10.0, 0.05 * -38.5dB) */
+ 0x3c4e4329, /* [143] = 1.26e-002 = pow(10.0, 0.05 * -38.0dB) */
+ 0x3c5a7bf1, /* [144] = 1.33e-002 = pow(10.0, 0.05 * -37.5dB) */
+ 0x3c676e1e, /* [145] = 1.41e-002 = pow(10.0, 0.05 * -37.0dB) */
+ 0x3c7524ac, /* [146] = 1.50e-002 = pow(10.0, 0.05 * -36.5dB) */
+ 0x3c81d59f, /* [147] = 1.58e-002 = pow(10.0, 0.05 * -36.0dB) */
+ 0x3c898712, /* [148] = 1.68e-002 = pow(10.0, 0.05 * -35.5dB) */
+ 0x3c91ad39, /* [149] = 1.78e-002 = pow(10.0, 0.05 * -35.0dB) */
+ 0x3c9a4efc, /* [150] = 1.88e-002 = pow(10.0, 0.05 * -34.5dB) */
+ 0x3ca373af, /* [151] = 2.00e-002 = pow(10.0, 0.05 * -34.0dB) */
+ 0x3cad2314, /* [152] = 2.11e-002 = pow(10.0, 0.05 * -33.5dB) */
+ 0x3cb76563, /* [153] = 2.24e-002 = pow(10.0, 0.05 * -33.0dB) */
+ 0x3cc24350, /* [154] = 2.37e-002 = pow(10.0, 0.05 * -32.5dB) */
+ 0x3ccdc614, /* [155] = 2.51e-002 = pow(10.0, 0.05 * -32.0dB) */
+ 0x3cd9f773, /* [156] = 2.66e-002 = pow(10.0, 0.05 * -31.5dB) */
+ 0x3ce6e1c6, /* [157] = 2.82e-002 = pow(10.0, 0.05 * -31.0dB) */
+ 0x3cf49003, /* [158] = 2.99e-002 = pow(10.0, 0.05 * -30.5dB) */
+ 0x3d0186e2, /* [159] = 3.16e-002 = pow(10.0, 0.05 * -30.0dB) */
+ 0x3d0933ac, /* [160] = 3.35e-002 = pow(10.0, 0.05 * -29.5dB) */
+ 0x3d1154e1, /* [161] = 3.55e-002 = pow(10.0, 0.05 * -29.0dB) */
+ 0x3d19f169, /* [162] = 3.76e-002 = pow(10.0, 0.05 * -28.5dB) */
+ 0x3d231090, /* [163] = 3.98e-002 = pow(10.0, 0.05 * -28.0dB) */
+ 0x3d2cba15, /* [164] = 4.22e-002 = pow(10.0, 0.05 * -27.5dB) */
+ 0x3d36f62b, /* [165] = 4.47e-002 = pow(10.0, 0.05 * -27.0dB) */
+ 0x3d41cd81, /* [166] = 4.73e-002 = pow(10.0, 0.05 * -26.5dB) */
+ 0x3d4d494a, /* [167] = 5.01e-002 = pow(10.0, 0.05 * -26.0dB) */
+ 0x3d597345, /* [168] = 5.31e-002 = pow(10.0, 0.05 * -25.5dB) */
+ 0x3d6655c3, /* [169] = 5.62e-002 = pow(10.0, 0.05 * -25.0dB) */
+ 0x3d73fbb4, /* [170] = 5.96e-002 = pow(10.0, 0.05 * -24.5dB) */
+ 0x3d813856, /* [171] = 6.31e-002 = pow(10.0, 0.05 * -24.0dB) */
+ 0x3d88e078, /* [172] = 6.68e-002 = pow(10.0, 0.05 * -23.5dB) */
+ 0x3d90fcbf, /* [173] = 7.08e-002 = pow(10.0, 0.05 * -23.0dB) */
+ 0x3d99940e, /* [174] = 7.50e-002 = pow(10.0, 0.05 * -22.5dB) */
+ 0x3da2adad, /* [175] = 7.94e-002 = pow(10.0, 0.05 * -22.0dB) */
+ 0x3dac5156, /* [176] = 8.41e-002 = pow(10.0, 0.05 * -21.5dB) */
+ 0x3db68738, /* [177] = 8.91e-002 = pow(10.0, 0.05 * -21.0dB) */
+ 0x3dc157fb, /* [178] = 9.44e-002 = pow(10.0, 0.05 * -20.5dB) */
+ 0x3dcccccd, /* [179] = 1.00e-001 = pow(10.0, 0.05 * -20.0dB) */
+ 0x3dd8ef67, /* [180] = 1.06e-001 = pow(10.0, 0.05 * -19.5dB) */
+ 0x3de5ca15, /* [181] = 1.12e-001 = pow(10.0, 0.05 * -19.0dB) */
+ 0x3df367bf, /* [182] = 1.19e-001 = pow(10.0, 0.05 * -18.5dB) */
+ 0x3e00e9f9, /* [183] = 1.26e-001 = pow(10.0, 0.05 * -18.0dB) */
+ 0x3e088d77, /* [184] = 1.33e-001 = pow(10.0, 0.05 * -17.5dB) */
+ 0x3e10a4d3, /* [185] = 1.41e-001 = pow(10.0, 0.05 * -17.0dB) */
+ 0x3e1936ec, /* [186] = 1.50e-001 = pow(10.0, 0.05 * -16.5dB) */
+ 0x3e224b06, /* [187] = 1.58e-001 = pow(10.0, 0.05 * -16.0dB) */
+ 0x3e2be8d7, /* [188] = 1.68e-001 = pow(10.0, 0.05 * -15.5dB) */
+ 0x3e361887, /* [189] = 1.78e-001 = pow(10.0, 0.05 * -15.0dB) */
+ 0x3e40e2bb, /* [190] = 1.88e-001 = pow(10.0, 0.05 * -14.5dB) */
+ 0x3e4c509b, /* [191] = 2.00e-001 = pow(10.0, 0.05 * -14.0dB) */
+ 0x3e586bd9, /* [192] = 2.11e-001 = pow(10.0, 0.05 * -13.5dB) */
+ 0x3e653ebb, /* [193] = 2.24e-001 = pow(10.0, 0.05 * -13.0dB) */
+ 0x3e72d424, /* [194] = 2.37e-001 = pow(10.0, 0.05 * -12.5dB) */
+ 0x3e809bcc, /* [195] = 2.51e-001 = pow(10.0, 0.05 * -12.0dB) */
+ 0x3e883aa8, /* [196] = 2.66e-001 = pow(10.0, 0.05 * -11.5dB) */
+ 0x3e904d1c, /* [197] = 2.82e-001 = pow(10.0, 0.05 * -11.0dB) */
+ 0x3e98da02, /* [198] = 2.99e-001 = pow(10.0, 0.05 * -10.5dB) */
+ 0x3ea1e89b, /* [199] = 3.16e-001 = pow(10.0, 0.05 * -10.0dB) */
+ 0x3eab8097, /* [200] = 3.35e-001 = pow(10.0, 0.05 * -9.5dB) */
+ 0x3eb5aa1a, /* [201] = 3.55e-001 = pow(10.0, 0.05 * -9.0dB) */
+ 0x3ec06dc3, /* [202] = 3.76e-001 = pow(10.0, 0.05 * -8.5dB) */
+ 0x3ecbd4b4, /* [203] = 3.98e-001 = pow(10.0, 0.05 * -8.0dB) */
+ 0x3ed7e89b, /* [204] = 4.22e-001 = pow(10.0, 0.05 * -7.5dB) */
+ 0x3ee4b3b6, /* [205] = 4.47e-001 = pow(10.0, 0.05 * -7.0dB) */
+ 0x3ef240e2, /* [206] = 4.73e-001 = pow(10.0, 0.05 * -6.5dB) */
+ 0x3f004dce, /* [207] = 5.01e-001 = pow(10.0, 0.05 * -6.0dB) */
+ 0x3f07e80b, /* [208] = 5.31e-001 = pow(10.0, 0.05 * -5.5dB) */
+ 0x3f0ff59a, /* [209] = 5.62e-001 = pow(10.0, 0.05 * -5.0dB) */
+ 0x3f187d50, /* [210] = 5.96e-001 = pow(10.0, 0.05 * -4.5dB) */
+ 0x3f21866c, /* [211] = 6.31e-001 = pow(10.0, 0.05 * -4.0dB) */
+ 0x3f2b1896, /* [212] = 6.68e-001 = pow(10.0, 0.05 * -3.5dB) */
+ 0x3f353bef, /* [213] = 7.08e-001 = pow(10.0, 0.05 * -3.0dB) */
+ 0x3f3ff911, /* [214] = 7.50e-001 = pow(10.0, 0.05 * -2.5dB) */
+ 0x3f4b5918, /* [215] = 7.94e-001 = pow(10.0, 0.05 * -2.0dB) */
+ 0x3f5765ac, /* [216] = 8.41e-001 = pow(10.0, 0.05 * -1.5dB) */
+ 0x3f642905, /* [217] = 8.91e-001 = pow(10.0, 0.05 * -1.0dB) */
+ 0x3f71adf9, /* [218] = 9.44e-001 = pow(10.0, 0.05 * -0.5dB) */
+ 0x3f800000, /* [219] = 1.00e+000 = pow(10.0, 0.05 * 0.0dB) */
+ 0x3f8795a0, /* [220] = 1.06e+000 = pow(10.0, 0.05 * 0.5dB) */
+ 0x3f8f9e4d, /* [221] = 1.12e+000 = pow(10.0, 0.05 * 1.0dB) */
+ 0x3f9820d7, /* [222] = 1.19e+000 = pow(10.0, 0.05 * 1.5dB) */
+ 0x3fa12478, /* [223] = 1.26e+000 = pow(10.0, 0.05 * 2.0dB) */
+ 0x3faab0d5, /* [224] = 1.33e+000 = pow(10.0, 0.05 * 2.5dB) */
+ 0x3fb4ce08, /* [225] = 1.41e+000 = pow(10.0, 0.05 * 3.0dB) */
+ 0x3fbf84a6, /* [226] = 1.50e+000 = pow(10.0, 0.05 * 3.5dB) */
+ 0x3fcaddc8, /* [227] = 1.58e+000 = pow(10.0, 0.05 * 4.0dB) */
+ 0x3fd6e30d, /* [228] = 1.68e+000 = pow(10.0, 0.05 * 4.5dB) */
+ 0x3fe39ea9, /* [229] = 1.78e+000 = pow(10.0, 0.05 * 5.0dB) */
+ 0x3ff11b6a, /* [230] = 1.88e+000 = pow(10.0, 0.05 * 5.5dB) */
+ 0x3fff64c1, /* [231] = 2.00e+000 = pow(10.0, 0.05 * 6.0dB) */
+ 0x40074368, /* [232] = 2.11e+000 = pow(10.0, 0.05 * 6.5dB) */
+ 0x400f4735, /* [233] = 2.24e+000 = pow(10.0, 0.05 * 7.0dB) */
+ 0x4017c496, /* [234] = 2.37e+000 = pow(10.0, 0.05 * 7.5dB) */
+ 0x4020c2bf, /* [235] = 2.51e+000 = pow(10.0, 0.05 * 8.0dB) */
+ 0x402a4952, /* [236] = 2.66e+000 = pow(10.0, 0.05 * 8.5dB) */
+ 0x40346063, /* [237] = 2.82e+000 = pow(10.0, 0.05 * 9.0dB) */
+ 0x403f1082, /* [238] = 2.99e+000 = pow(10.0, 0.05 * 9.5dB) */
+ 0x404a62c2, /* [239] = 3.16e+000 = pow(10.0, 0.05 * 10.0dB) */
+ 0x405660bd, /* [240] = 3.35e+000 = pow(10.0, 0.05 * 10.5dB) */
+ 0x406314a0, /* [241] = 3.55e+000 = pow(10.0, 0.05 * 11.0dB) */
+ 0x40708933, /* [242] = 3.76e+000 = pow(10.0, 0.05 * 11.5dB) */
+ 0x407ec9e1, /* [243] = 3.98e+000 = pow(10.0, 0.05 * 12.0dB) */
+ 0x4086f161, /* [244] = 4.22e+000 = pow(10.0, 0.05 * 12.5dB) */
+ 0x408ef052, /* [245] = 4.47e+000 = pow(10.0, 0.05 * 13.0dB) */
+ 0x4097688d, /* [246] = 4.73e+000 = pow(10.0, 0.05 * 13.5dB) */
+ 0x40a06142, /* [247] = 5.01e+000 = pow(10.0, 0.05 * 14.0dB) */
+ 0x40a9e20e, /* [248] = 5.31e+000 = pow(10.0, 0.05 * 14.5dB) */
+ 0x40b3f300, /* [249] = 5.62e+000 = pow(10.0, 0.05 * 15.0dB) */
+ 0x40be9ca5, /* [250] = 5.96e+000 = pow(10.0, 0.05 * 15.5dB) */
+ 0x40c9e807, /* [251] = 6.31e+000 = pow(10.0, 0.05 * 16.0dB) */
+ 0x40d5debc, /* [252] = 6.68e+000 = pow(10.0, 0.05 * 16.5dB) */
+ 0x40e28aeb, /* [253] = 7.08e+000 = pow(10.0, 0.05 * 17.0dB) */
+ 0x40eff755, /* [254] = 7.50e+000 = pow(10.0, 0.05 * 17.5dB) */
+ 0x40fe2f5e, /* [255] = 7.94e+000 = pow(10.0, 0.05 * 18.0dB) */
+};
+
+#define MIXART_DIGITAL_LEVEL_MIN 0 /* -109.5 dB */
+#define MIXART_DIGITAL_LEVEL_MAX 255 /* 18.0 dB */
+#define MIXART_DIGITAL_ZERO_LEVEL 219 /* 0.0 dB */
+
+
+int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int idx)
+{
+ int err, i;
+ int volume[2];
+ struct mixart_msg request;
+ struct mixart_set_out_stream_level_req set_level;
+ u32 status;
+ struct mixart_pipe *pipe;
+
+ memset(&set_level, 0, sizeof(set_level));
+ set_level.nb_of_stream = 1;
+ set_level.stream_level.desc.stream_idx = idx;
+
+ if(is_aes) {
+ pipe = &chip->pipe_out_dig; /* AES playback */
+ idx += MIXART_PLAYBACK_STREAMS;
+ } else {
+ pipe = &chip->pipe_out_ana; /* analog playback */
+ }
+
+ /* only when pipe exists ! */
+ if(pipe->status == PIPE_UNDEFINED)
+ return 0;
+
+ set_level.stream_level.desc.uid_pipe = pipe->group_uid;
+
+ for(i=0; i<2; i++) {
+ if(chip->digital_playback_active[idx][i])
+ volume[i] = chip->digital_playback_volume[idx][i];
+ else
+ volume[i] = MIXART_DIGITAL_LEVEL_MIN;
+ }
+
+ set_level.stream_level.out_level.valid_mask1 = MIXART_OUT_STREAM_SET_LEVEL_LEFT_AUDIO1 | MIXART_OUT_STREAM_SET_LEVEL_RIGHT_AUDIO2;
+ set_level.stream_level.out_level.left_to_out1_level = mixart_digital_level[volume[0]];
+ set_level.stream_level.out_level.right_to_out2_level = mixart_digital_level[volume[1]];
+
+ request.message_id = MSG_STREAM_SET_OUT_STREAM_LEVEL;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = &set_level;
+ request.size = sizeof(set_level);
+
+ err = snd_mixart_send_msg(chip->mgr, &request, sizeof(status), &status);
+ if((err<0) || status) {
+ snd_printk(KERN_DEBUG "error MSG_STREAM_SET_OUT_STREAM_LEVEL card(%d) status(%x)\n", chip->chip_idx, status);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes)
+{
+ int err, i, idx;
+ struct mixart_pipe *pipe;
+ struct mixart_msg request;
+ struct mixart_set_in_audio_level_req set_level;
+ u32 status;
+
+ if(is_aes) {
+ idx = 1;
+ pipe = &chip->pipe_in_dig;
+ } else {
+ idx = 0;
+ pipe = &chip->pipe_in_ana;
+ }
+
+ /* only when pipe exists ! */
+ if(pipe->status == PIPE_UNDEFINED)
+ return 0;
+
+ memset(&set_level, 0, sizeof(set_level));
+ set_level.audio_count = 2;
+ set_level.level[0].connector = pipe->uid_left_connector;
+ set_level.level[1].connector = pipe->uid_right_connector;
+
+ for(i=0; i<2; i++) {
+ set_level.level[i].valid_mask1 = MIXART_AUDIO_LEVEL_DIGITAL_MASK;
+ set_level.level[i].digital_level = mixart_digital_level[chip->digital_capture_volume[idx][i]];
+ }
+
+ request.message_id = MSG_STREAM_SET_IN_AUDIO_LEVEL;
+ request.uid = (struct mixart_uid){0,0};
+ request.data = &set_level;
+ request.size = sizeof(set_level);
+
+ err = snd_mixart_send_msg(chip->mgr, &request, sizeof(status), &status);
+ if((err<0) || status) {
+ snd_printk(KERN_DEBUG "error MSG_STREAM_SET_IN_AUDIO_LEVEL card(%d) status(%x)\n", chip->chip_idx, status);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/* shared */
+static int mixart_digital_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = MIXART_DIGITAL_LEVEL_MIN; /* -109.5 dB */
+ uinfo->value.integer.max = MIXART_DIGITAL_LEVEL_MAX; /* 18.0 dB */
+ return 0;
+}
+
+#define MIXART_VOL_REC_MASK 1
+#define MIXART_VOL_AES_MASK 2
+
+static int mixart_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+ int *stored_volume;
+ int is_capture = kcontrol->private_value & MIXART_VOL_REC_MASK;
+ int is_aes = kcontrol->private_value & MIXART_VOL_AES_MASK;
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if(is_capture) {
+ if(is_aes) stored_volume = chip->digital_capture_volume[1]; /* AES capture */
+ else stored_volume = chip->digital_capture_volume[0]; /* analog capture */
+ } else {
+ snd_BUG_ON(idx >= MIXART_PLAYBACK_STREAMS);
+ if(is_aes) stored_volume = chip->digital_playback_volume[MIXART_PLAYBACK_STREAMS + idx]; /* AES playback */
+ else stored_volume = chip->digital_playback_volume[idx]; /* analog playback */
+ }
+ ucontrol->value.integer.value[0] = stored_volume[0];
+ ucontrol->value.integer.value[1] = stored_volume[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+ int changed = 0;
+ int is_capture = kcontrol->private_value & MIXART_VOL_REC_MASK;
+ int is_aes = kcontrol->private_value & MIXART_VOL_AES_MASK;
+ int* stored_volume;
+ int i;
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if (is_capture) {
+ if (is_aes) /* AES capture */
+ stored_volume = chip->digital_capture_volume[1];
+ else /* analog capture */
+ stored_volume = chip->digital_capture_volume[0];
+ } else {
+ snd_BUG_ON(idx >= MIXART_PLAYBACK_STREAMS);
+ if (is_aes) /* AES playback */
+ stored_volume = chip->digital_playback_volume[MIXART_PLAYBACK_STREAMS + idx];
+ else /* analog playback */
+ stored_volume = chip->digital_playback_volume[idx];
+ }
+ for (i = 0; i < 2; i++) {
+ int vol = ucontrol->value.integer.value[i];
+ if (vol < MIXART_DIGITAL_LEVEL_MIN ||
+ vol > MIXART_DIGITAL_LEVEL_MAX)
+ continue;
+ if (stored_volume[i] != vol) {
+ stored_volume[i] = vol;
+ changed = 1;
+ }
+ }
+ if (changed) {
+ if (is_capture)
+ mixart_update_capture_stream_level(chip, is_aes);
+ else
+ mixart_update_playback_stream_level(chip, is_aes, idx);
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0);
+
+static struct snd_kcontrol_new snd_mixart_pcm_vol =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ /* name will be filled later */
+ /* count will be filled later */
+ .info = mixart_digital_vol_info, /* shared */
+ .get = mixart_pcm_vol_get,
+ .put = mixart_pcm_vol_put,
+ .tlv = { .p = db_scale_digital },
+};
+
+
+static int mixart_pcm_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+ snd_BUG_ON(idx >= MIXART_PLAYBACK_STREAMS);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if(kcontrol->private_value & MIXART_VOL_AES_MASK) /* AES playback */
+ idx += MIXART_PLAYBACK_STREAMS;
+ ucontrol->value.integer.value[0] = chip->digital_playback_active[idx][0];
+ ucontrol->value.integer.value[1] = chip->digital_playback_active[idx][1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int mixart_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int is_aes = kcontrol->private_value & MIXART_VOL_AES_MASK;
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+ int i, j;
+ snd_BUG_ON(idx >= MIXART_PLAYBACK_STREAMS);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ j = idx;
+ if (is_aes)
+ j += MIXART_PLAYBACK_STREAMS;
+ for (i = 0; i < 2; i++) {
+ if (chip->digital_playback_active[j][i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->digital_playback_active[j][i] =
+ !!ucontrol->value.integer.value[i];
+ changed = 1;
+ }
+ }
+ if (changed)
+ mixart_update_playback_stream_level(chip, is_aes, idx);
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new mixart_control_pcm_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* name will be filled later */
+ .count = MIXART_PLAYBACK_STREAMS,
+ .info = mixart_sw_info, /* shared */
+ .get = mixart_pcm_sw_get,
+ .put = mixart_pcm_sw_put
+};
+
+static int mixart_update_monitoring(struct snd_mixart* chip, int channel)
+{
+ int err;
+ struct mixart_msg request;
+ struct mixart_set_out_audio_level audio_level;
+ u32 resp;
+
+ if(chip->pipe_out_ana.status == PIPE_UNDEFINED)
+ return -EINVAL; /* no pipe defined */
+
+ if(!channel) request.uid = chip->pipe_out_ana.uid_left_connector;
+ else request.uid = chip->pipe_out_ana.uid_right_connector;
+ request.message_id = MSG_CONNECTOR_SET_OUT_AUDIO_LEVEL;
+ request.data = &audio_level;
+ request.size = sizeof(audio_level);
+
+ memset(&audio_level, 0, sizeof(audio_level));
+ audio_level.valid_mask1 = MIXART_AUDIO_LEVEL_MONITOR_MASK | MIXART_AUDIO_LEVEL_MUTE_M1_MASK;
+ audio_level.monitor_level = mixart_digital_level[chip->monitoring_volume[channel!=0]];
+ audio_level.monitor_mute1 = !chip->monitoring_active[channel!=0];
+
+ err = snd_mixart_send_msg(chip->mgr, &request, sizeof(resp), &resp);
+ if((err<0) || resp) {
+ snd_printk(KERN_DEBUG "error MSG_CONNECTOR_SET_OUT_AUDIO_LEVEL card(%d) resp(%x)\n", chip->chip_idx, resp);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * monitoring level control
+ */
+
+static int mixart_monitor_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->monitoring_volume[0];
+ ucontrol->value.integer.value[1] = chip->monitoring_volume[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int i;
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for (i = 0; i < 2; i++) {
+ if (chip->monitoring_volume[i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->monitoring_volume[i] =
+ !!ucontrol->value.integer.value[i];
+ mixart_update_monitoring(chip, i);
+ changed = 1;
+ }
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new mixart_control_monitor_vol = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Monitoring Volume",
+ .info = mixart_digital_vol_info, /* shared */
+ .get = mixart_monitor_vol_get,
+ .put = mixart_monitor_vol_put,
+ .tlv = { .p = db_scale_digital },
+};
+
+/*
+ * monitoring switch control
+ */
+
+static int mixart_monitor_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->monitoring_active[0];
+ ucontrol->value.integer.value[1] = chip->monitoring_active[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int mixart_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int i;
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for (i = 0; i < 2; i++) {
+ if (chip->monitoring_active[i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->monitoring_active[i] =
+ !!ucontrol->value.integer.value[i];
+ changed |= (1<<i); /* mask 0x01 ans 0x02 */
+ }
+ }
+ if (changed) {
+ /* allocate or release resources for monitoring */
+ int allocate = chip->monitoring_active[0] ||
+ chip->monitoring_active[1];
+ if (allocate) {
+ /* allocate the playback pipe for monitoring */
+ snd_mixart_add_ref_pipe(chip, MIXART_PCM_ANALOG, 0, 1);
+ /* allocate the capture pipe for monitoring */
+ snd_mixart_add_ref_pipe(chip, MIXART_PCM_ANALOG, 1, 1);
+ }
+ if (changed & 0x01)
+ mixart_update_monitoring(chip, 0);
+ if (changed & 0x02)
+ mixart_update_monitoring(chip, 1);
+ if (!allocate) {
+ /* release the capture pipe for monitoring */
+ snd_mixart_kill_ref_pipe(chip->mgr,
+ &chip->pipe_in_ana, 1);
+ /* release the playback pipe for monitoring */
+ snd_mixart_kill_ref_pipe(chip->mgr,
+ &chip->pipe_out_ana, 1);
+ }
+ }
+
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return (changed != 0);
+}
+
+static struct snd_kcontrol_new mixart_control_monitor_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitoring Switch",
+ .info = mixart_sw_info, /* shared */
+ .get = mixart_monitor_sw_get,
+ .put = mixart_monitor_sw_put
+};
+
+
+static void mixart_reset_audio_levels(struct snd_mixart *chip)
+{
+ /* analog volumes can be set even if there is no pipe */
+ mixart_update_analog_audio_level(chip, 0);
+ /* analog levels for capture only on the first two chips */
+ if(chip->chip_idx < 2) {
+ mixart_update_analog_audio_level(chip, 1);
+ }
+ return;
+}
+
+
+int snd_mixart_create_mixer(struct mixart_mgr *mgr)
+{
+ struct snd_mixart *chip;
+ int err, i;
+
+ mutex_init(&mgr->mixer_mutex); /* can be in another place */
+
+ for(i=0; i<mgr->num_cards; i++) {
+ struct snd_kcontrol_new temp;
+ chip = mgr->chip[i];
+
+ /* analog output level control */
+ temp = mixart_control_analog_level;
+ temp.name = "Master Playback Volume";
+ temp.private_value = 0; /* playback */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ /* output mute controls */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&mixart_control_output_switch, chip))) < 0)
+ return err;
+
+ /* analog input level control only on first two chips !*/
+ if(i<2) {
+ temp = mixart_control_analog_level;
+ temp.name = "Master Capture Volume";
+ temp.private_value = 1; /* capture */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+
+ temp = snd_mixart_pcm_vol;
+ temp.name = "PCM Playback Volume";
+ temp.count = MIXART_PLAYBACK_STREAMS;
+ temp.private_value = 0; /* playback analog */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+
+ temp.name = "PCM Capture Volume";
+ temp.count = 1;
+ temp.private_value = MIXART_VOL_REC_MASK; /* capture analog */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+
+ if(mgr->board_type == MIXART_DAUGHTER_TYPE_AES) {
+ temp.name = "AES Playback Volume";
+ temp.count = MIXART_PLAYBACK_STREAMS;
+ temp.private_value = MIXART_VOL_AES_MASK; /* playback AES/EBU */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+
+ temp.name = "AES Capture Volume";
+ temp.count = 0;
+ temp.private_value = MIXART_VOL_REC_MASK | MIXART_VOL_AES_MASK; /* capture AES/EBU */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+ temp = mixart_control_pcm_switch;
+ temp.name = "PCM Playback Switch";
+ temp.private_value = 0; /* playback analog */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+
+ if(mgr->board_type == MIXART_DAUGHTER_TYPE_AES) {
+ temp.name = "AES Playback Switch";
+ temp.private_value = MIXART_VOL_AES_MASK; /* playback AES/EBU */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ }
+
+ /* monitoring */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&mixart_control_monitor_vol, chip))) < 0)
+ return err;
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&mixart_control_monitor_sw, chip))) < 0)
+ return err;
+
+ /* init all mixer data and program the master volumes/switches */
+ mixart_reset_audio_levels(chip);
+ }
+ return 0;
+}
diff --git a/sound/pci/mixart/mixart_mixer.h b/sound/pci/mixart/mixart_mixer.h
new file mode 100644
index 0000000..04aa24e
--- /dev/null
+++ b/sound/pci/mixart/mixart_mixer.h
@@ -0,0 +1,31 @@
+/*
+ * Driver for Digigram miXart soundcards
+ *
+ * include file for mixer
+ *
+ * Copyright (c) 2003 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_MIXART_MIXER_H
+#define __SOUND_MIXART_MIXER_H
+
+/* exported */
+int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int idx);
+int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes);
+int snd_mixart_create_mixer(struct mixart_mgr* mgr);
+
+#endif /* __SOUND_MIXART_MIXER_H */
diff --git a/sound/pci/nm256/Makefile b/sound/pci/nm256/Makefile
new file mode 100644
index 0000000..a1bd44f
--- /dev/null
+++ b/sound/pci/nm256/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-nm256-objs := nm256.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_NM256) += snd-nm256.o
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
new file mode 100644
index 0000000..50c9f8a
--- /dev/null
+++ b/sound/pci/nm256/nm256.c
@@ -0,0 +1,1768 @@
+/*
+ * Driver for NeoMagic 256AV and 256ZX chipsets.
+ * Copyright (c) 2000 by Takashi Iwai <tiwai@suse.de>
+ *
+ * Based on nm256_audio.c OSS driver in linux kernel.
+ * The original author of OSS nm256 driver wishes to remain anonymous,
+ * so I just put my acknoledgment to him/her here.
+ * The original author's web page is found at
+ * http://www.uglx.org/sony.html
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+
+#define CARD_NAME "NeoMagic 256AV/ZX"
+#define DRIVER_NAME "NM256"
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("NeoMagic NM256AV/ZX");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{NeoMagic,NM256AV},"
+ "{NeoMagic,NM256ZX}}");
+
+/*
+ * some compile conditions.
+ */
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int playback_bufsize = 16;
+static int capture_bufsize = 16;
+static int force_ac97; /* disabled as default */
+static int buffer_top; /* not specified */
+static int use_cache; /* disabled */
+static int vaio_hack; /* disabled */
+static int reset_workaround;
+static int reset_workaround_2;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param(playback_bufsize, int, 0444);
+MODULE_PARM_DESC(playback_bufsize, "DAC frame size in kB for " CARD_NAME " soundcard.");
+module_param(capture_bufsize, int, 0444);
+MODULE_PARM_DESC(capture_bufsize, "ADC frame size in kB for " CARD_NAME " soundcard.");
+module_param(force_ac97, bool, 0444);
+MODULE_PARM_DESC(force_ac97, "Force to use AC97 codec for " CARD_NAME " soundcard.");
+module_param(buffer_top, int, 0444);
+MODULE_PARM_DESC(buffer_top, "Set the top address of audio buffer for " CARD_NAME " soundcard.");
+module_param(use_cache, bool, 0444);
+MODULE_PARM_DESC(use_cache, "Enable the cache for coefficient table access.");
+module_param(vaio_hack, bool, 0444);
+MODULE_PARM_DESC(vaio_hack, "Enable workaround for Sony VAIO notebooks.");
+module_param(reset_workaround, bool, 0444);
+MODULE_PARM_DESC(reset_workaround, "Enable AC97 RESET workaround for some laptops.");
+module_param(reset_workaround_2, bool, 0444);
+MODULE_PARM_DESC(reset_workaround_2, "Enable extended AC97 RESET workaround for some other laptops.");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
+
+
+/*
+ * hw definitions
+ */
+
+/* The BIOS signature. */
+#define NM_SIGNATURE 0x4e4d0000
+/* Signature mask. */
+#define NM_SIG_MASK 0xffff0000
+
+/* Size of the second memory area. */
+#define NM_PORT2_SIZE 4096
+
+/* The base offset of the mixer in the second memory area. */
+#define NM_MIXER_OFFSET 0x600
+
+/* The maximum size of a coefficient entry. */
+#define NM_MAX_PLAYBACK_COEF_SIZE 0x5000
+#define NM_MAX_RECORD_COEF_SIZE 0x1260
+
+/* The interrupt register. */
+#define NM_INT_REG 0xa04
+/* And its bits. */
+#define NM_PLAYBACK_INT 0x40
+#define NM_RECORD_INT 0x100
+#define NM_MISC_INT_1 0x4000
+#define NM_MISC_INT_2 0x1
+#define NM_ACK_INT(chip, X) snd_nm256_writew(chip, NM_INT_REG, (X) << 1)
+
+/* The AV's "mixer ready" status bit and location. */
+#define NM_MIXER_STATUS_OFFSET 0xa04
+#define NM_MIXER_READY_MASK 0x0800
+#define NM_MIXER_PRESENCE 0xa06
+#define NM_PRESENCE_MASK 0x0050
+#define NM_PRESENCE_VALUE 0x0040
+
+/*
+ * For the ZX. It uses the same interrupt register, but it holds 32
+ * bits instead of 16.
+ */
+#define NM2_PLAYBACK_INT 0x10000
+#define NM2_RECORD_INT 0x80000
+#define NM2_MISC_INT_1 0x8
+#define NM2_MISC_INT_2 0x2
+#define NM2_ACK_INT(chip, X) snd_nm256_writel(chip, NM_INT_REG, (X))
+
+/* The ZX's "mixer ready" status bit and location. */
+#define NM2_MIXER_STATUS_OFFSET 0xa06
+#define NM2_MIXER_READY_MASK 0x0800
+
+/* The playback registers start from here. */
+#define NM_PLAYBACK_REG_OFFSET 0x0
+/* The record registers start from here. */
+#define NM_RECORD_REG_OFFSET 0x200
+
+/* The rate register is located 2 bytes from the start of the register area. */
+#define NM_RATE_REG_OFFSET 2
+
+/* Mono/stereo flag, number of bits on playback, and rate mask. */
+#define NM_RATE_STEREO 1
+#define NM_RATE_BITS_16 2
+#define NM_RATE_MASK 0xf0
+
+/* Playback enable register. */
+#define NM_PLAYBACK_ENABLE_REG (NM_PLAYBACK_REG_OFFSET + 0x1)
+#define NM_PLAYBACK_ENABLE_FLAG 1
+#define NM_PLAYBACK_ONESHOT 2
+#define NM_PLAYBACK_FREERUN 4
+
+/* Mutes the audio output. */
+#define NM_AUDIO_MUTE_REG (NM_PLAYBACK_REG_OFFSET + 0x18)
+#define NM_AUDIO_MUTE_LEFT 0x8000
+#define NM_AUDIO_MUTE_RIGHT 0x0080
+
+/* Recording enable register. */
+#define NM_RECORD_ENABLE_REG (NM_RECORD_REG_OFFSET + 0)
+#define NM_RECORD_ENABLE_FLAG 1
+#define NM_RECORD_FREERUN 2
+
+/* coefficient buffer pointer */
+#define NM_COEFF_START_OFFSET 0x1c
+#define NM_COEFF_END_OFFSET 0x20
+
+/* DMA buffer offsets */
+#define NM_RBUFFER_START (NM_RECORD_REG_OFFSET + 0x4)
+#define NM_RBUFFER_END (NM_RECORD_REG_OFFSET + 0x10)
+#define NM_RBUFFER_WMARK (NM_RECORD_REG_OFFSET + 0xc)
+#define NM_RBUFFER_CURRP (NM_RECORD_REG_OFFSET + 0x8)
+
+#define NM_PBUFFER_START (NM_PLAYBACK_REG_OFFSET + 0x4)
+#define NM_PBUFFER_END (NM_PLAYBACK_REG_OFFSET + 0x14)
+#define NM_PBUFFER_WMARK (NM_PLAYBACK_REG_OFFSET + 0xc)
+#define NM_PBUFFER_CURRP (NM_PLAYBACK_REG_OFFSET + 0x8)
+
+struct nm256_stream {
+
+ struct nm256 *chip;
+ struct snd_pcm_substream *substream;
+ int running;
+ int suspended;
+
+ u32 buf; /* offset from chip->buffer */
+ int bufsize; /* buffer size in bytes */
+ void __iomem *bufptr; /* mapped pointer */
+ unsigned long bufptr_addr; /* physical address of the mapped pointer */
+
+ int dma_size; /* buffer size of the substream in bytes */
+ int period_size; /* period size in bytes */
+ int periods; /* # of periods */
+ int shift; /* bit shifts */
+ int cur_period; /* current period # */
+
+};
+
+struct nm256 {
+
+ struct snd_card *card;
+
+ void __iomem *cport; /* control port */
+ struct resource *res_cport; /* its resource */
+ unsigned long cport_addr; /* physical address */
+
+ void __iomem *buffer; /* buffer */
+ struct resource *res_buffer; /* its resource */
+ unsigned long buffer_addr; /* buffer phyiscal address */
+
+ u32 buffer_start; /* start offset from pci resource 0 */
+ u32 buffer_end; /* end offset */
+ u32 buffer_size; /* total buffer size */
+
+ u32 all_coeff_buf; /* coefficient buffer */
+ u32 coeff_buf[2]; /* coefficient buffer for each stream */
+
+ unsigned int coeffs_current: 1; /* coeff. table is loaded? */
+ unsigned int use_cache: 1; /* use one big coef. table */
+ unsigned int reset_workaround: 1; /* Workaround for some laptops to avoid freeze */
+ unsigned int reset_workaround_2: 1; /* Extended workaround for some other laptops to avoid freeze */
+ unsigned int in_resume: 1;
+
+ int mixer_base; /* register offset of ac97 mixer */
+ int mixer_status_offset; /* offset of mixer status reg. */
+ int mixer_status_mask; /* bit mask to test the mixer status */
+
+ int irq;
+ int irq_acks;
+ irq_handler_t interrupt;
+ int badintrcount; /* counter to check bogus interrupts */
+ struct mutex irq_mutex;
+
+ struct nm256_stream streams[2];
+
+ struct snd_ac97 *ac97;
+ unsigned short *ac97_regs; /* register caches, only for valid regs */
+
+ struct snd_pcm *pcm;
+
+ struct pci_dev *pci;
+
+ spinlock_t reg_lock;
+
+};
+
+
+/*
+ * include coefficient table
+ */
+#include "nm256_coef.c"
+
+
+/*
+ * PCI ids
+ */
+static struct pci_device_id snd_nm256_ids[] = {
+ {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0},
+ {0,},
+};
+
+MODULE_DEVICE_TABLE(pci, snd_nm256_ids);
+
+
+/*
+ * lowlvel stuffs
+ */
+
+static inline u8
+snd_nm256_readb(struct nm256 *chip, int offset)
+{
+ return readb(chip->cport + offset);
+}
+
+static inline u16
+snd_nm256_readw(struct nm256 *chip, int offset)
+{
+ return readw(chip->cport + offset);
+}
+
+static inline u32
+snd_nm256_readl(struct nm256 *chip, int offset)
+{
+ return readl(chip->cport + offset);
+}
+
+static inline void
+snd_nm256_writeb(struct nm256 *chip, int offset, u8 val)
+{
+ writeb(val, chip->cport + offset);
+}
+
+static inline void
+snd_nm256_writew(struct nm256 *chip, int offset, u16 val)
+{
+ writew(val, chip->cport + offset);
+}
+
+static inline void
+snd_nm256_writel(struct nm256 *chip, int offset, u32 val)
+{
+ writel(val, chip->cport + offset);
+}
+
+static inline void
+snd_nm256_write_buffer(struct nm256 *chip, void *src, int offset, int size)
+{
+ offset -= chip->buffer_start;
+#ifdef CONFIG_SND_DEBUG
+ if (offset < 0 || offset >= chip->buffer_size) {
+ snd_printk(KERN_ERR "write_buffer invalid offset = %d size = %d\n",
+ offset, size);
+ return;
+ }
+#endif
+ memcpy_toio(chip->buffer + offset, src, size);
+}
+
+/*
+ * coefficient handlers -- what a magic!
+ */
+
+static u16
+snd_nm256_get_start_offset(int which)
+{
+ u16 offset = 0;
+ while (which-- > 0)
+ offset += coefficient_sizes[which];
+ return offset;
+}
+
+static void
+snd_nm256_load_one_coefficient(struct nm256 *chip, int stream, u32 port, int which)
+{
+ u32 coeff_buf = chip->coeff_buf[stream];
+ u16 offset = snd_nm256_get_start_offset(which);
+ u16 size = coefficient_sizes[which];
+
+ snd_nm256_write_buffer(chip, coefficients + offset, coeff_buf, size);
+ snd_nm256_writel(chip, port, coeff_buf);
+ /* ??? Record seems to behave differently than playback. */
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ size--;
+ snd_nm256_writel(chip, port + 4, coeff_buf + size);
+}
+
+static void
+snd_nm256_load_coefficient(struct nm256 *chip, int stream, int number)
+{
+ /* The enable register for the specified engine. */
+ u32 poffset = (stream == SNDRV_PCM_STREAM_CAPTURE ?
+ NM_RECORD_ENABLE_REG : NM_PLAYBACK_ENABLE_REG);
+ u32 addr = NM_COEFF_START_OFFSET;
+
+ addr += (stream == SNDRV_PCM_STREAM_CAPTURE ?
+ NM_RECORD_REG_OFFSET : NM_PLAYBACK_REG_OFFSET);
+
+ if (snd_nm256_readb(chip, poffset) & 1) {
+ snd_printd("NM256: Engine was enabled while loading coefficients!\n");
+ return;
+ }
+
+ /* The recording engine uses coefficient values 8-15. */
+ number &= 7;
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
+ number += 8;
+
+ if (! chip->use_cache) {
+ snd_nm256_load_one_coefficient(chip, stream, addr, number);
+ return;
+ }
+ if (! chip->coeffs_current) {
+ snd_nm256_write_buffer(chip, coefficients, chip->all_coeff_buf,
+ NM_TOTAL_COEFF_COUNT * 4);
+ chip->coeffs_current = 1;
+ } else {
+ u32 base = chip->all_coeff_buf;
+ u32 offset = snd_nm256_get_start_offset(number);
+ u32 end_offset = offset + coefficient_sizes[number];
+ snd_nm256_writel(chip, addr, base + offset);
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ end_offset--;
+ snd_nm256_writel(chip, addr + 4, base + end_offset);
+ }
+}
+
+
+/* The actual rates supported by the card. */
+static unsigned int samplerates[8] = {
+ 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000,
+};
+static struct snd_pcm_hw_constraint_list constraints_rates = {
+ .count = ARRAY_SIZE(samplerates),
+ .list = samplerates,
+ .mask = 0,
+};
+
+/*
+ * return the index of the target rate
+ */
+static int
+snd_nm256_fixed_rate(unsigned int rate)
+{
+ unsigned int i;
+ for (i = 0; i < ARRAY_SIZE(samplerates); i++) {
+ if (rate == samplerates[i])
+ return i;
+ }
+ snd_BUG();
+ return 0;
+}
+
+/*
+ * set sample rate and format
+ */
+static void
+snd_nm256_set_format(struct nm256 *chip, struct nm256_stream *s,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int rate_index = snd_nm256_fixed_rate(runtime->rate);
+ unsigned char ratebits = (rate_index << 4) & NM_RATE_MASK;
+
+ s->shift = 0;
+ if (snd_pcm_format_width(runtime->format) == 16) {
+ ratebits |= NM_RATE_BITS_16;
+ s->shift++;
+ }
+ if (runtime->channels > 1) {
+ ratebits |= NM_RATE_STEREO;
+ s->shift++;
+ }
+
+ runtime->rate = samplerates[rate_index];
+
+ switch (substream->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ snd_nm256_load_coefficient(chip, 0, rate_index); /* 0 = playback */
+ snd_nm256_writeb(chip,
+ NM_PLAYBACK_REG_OFFSET + NM_RATE_REG_OFFSET,
+ ratebits);
+ break;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ snd_nm256_load_coefficient(chip, 1, rate_index); /* 1 = record */
+ snd_nm256_writeb(chip,
+ NM_RECORD_REG_OFFSET + NM_RATE_REG_OFFSET,
+ ratebits);
+ break;
+ }
+}
+
+/* acquire interrupt */
+static int snd_nm256_acquire_irq(struct nm256 *chip)
+{
+ mutex_lock(&chip->irq_mutex);
+ if (chip->irq < 0) {
+ if (request_irq(chip->pci->irq, chip->interrupt, IRQF_SHARED,
+ chip->card->driver, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->pci->irq);
+ mutex_unlock(&chip->irq_mutex);
+ return -EBUSY;
+ }
+ chip->irq = chip->pci->irq;
+ }
+ chip->irq_acks++;
+ mutex_unlock(&chip->irq_mutex);
+ return 0;
+}
+
+/* release interrupt */
+static void snd_nm256_release_irq(struct nm256 *chip)
+{
+ mutex_lock(&chip->irq_mutex);
+ if (chip->irq_acks > 0)
+ chip->irq_acks--;
+ if (chip->irq_acks == 0 && chip->irq >= 0) {
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ }
+ mutex_unlock(&chip->irq_mutex);
+}
+
+/*
+ * start / stop
+ */
+
+/* update the watermark (current period) */
+static void snd_nm256_pcm_mark(struct nm256 *chip, struct nm256_stream *s, int reg)
+{
+ s->cur_period++;
+ s->cur_period %= s->periods;
+ snd_nm256_writel(chip, reg, s->buf + s->cur_period * s->period_size);
+}
+
+#define snd_nm256_playback_mark(chip, s) snd_nm256_pcm_mark(chip, s, NM_PBUFFER_WMARK)
+#define snd_nm256_capture_mark(chip, s) snd_nm256_pcm_mark(chip, s, NM_RBUFFER_WMARK)
+
+static void
+snd_nm256_playback_start(struct nm256 *chip, struct nm256_stream *s,
+ struct snd_pcm_substream *substream)
+{
+ /* program buffer pointers */
+ snd_nm256_writel(chip, NM_PBUFFER_START, s->buf);
+ snd_nm256_writel(chip, NM_PBUFFER_END, s->buf + s->dma_size - (1 << s->shift));
+ snd_nm256_writel(chip, NM_PBUFFER_CURRP, s->buf);
+ snd_nm256_playback_mark(chip, s);
+
+ /* Enable playback engine and interrupts. */
+ snd_nm256_writeb(chip, NM_PLAYBACK_ENABLE_REG,
+ NM_PLAYBACK_ENABLE_FLAG | NM_PLAYBACK_FREERUN);
+ /* Enable both channels. */
+ snd_nm256_writew(chip, NM_AUDIO_MUTE_REG, 0x0);
+}
+
+static void
+snd_nm256_capture_start(struct nm256 *chip, struct nm256_stream *s,
+ struct snd_pcm_substream *substream)
+{
+ /* program buffer pointers */
+ snd_nm256_writel(chip, NM_RBUFFER_START, s->buf);
+ snd_nm256_writel(chip, NM_RBUFFER_END, s->buf + s->dma_size);
+ snd_nm256_writel(chip, NM_RBUFFER_CURRP, s->buf);
+ snd_nm256_capture_mark(chip, s);
+
+ /* Enable playback engine and interrupts. */
+ snd_nm256_writeb(chip, NM_RECORD_ENABLE_REG,
+ NM_RECORD_ENABLE_FLAG | NM_RECORD_FREERUN);
+}
+
+/* Stop the play engine. */
+static void
+snd_nm256_playback_stop(struct nm256 *chip)
+{
+ /* Shut off sound from both channels. */
+ snd_nm256_writew(chip, NM_AUDIO_MUTE_REG,
+ NM_AUDIO_MUTE_LEFT | NM_AUDIO_MUTE_RIGHT);
+ /* Disable play engine. */
+ snd_nm256_writeb(chip, NM_PLAYBACK_ENABLE_REG, 0);
+}
+
+static void
+snd_nm256_capture_stop(struct nm256 *chip)
+{
+ /* Disable recording engine. */
+ snd_nm256_writeb(chip, NM_RECORD_ENABLE_REG, 0);
+}
+
+static int
+snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+ struct nm256_stream *s = substream->runtime->private_data;
+ int err = 0;
+
+ if (snd_BUG_ON(!s))
+ return -ENXIO;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_RESUME:
+ s->suspended = 0;
+ /* fallthru */
+ case SNDRV_PCM_TRIGGER_START:
+ if (! s->running) {
+ snd_nm256_playback_start(chip, s, substream);
+ s->running = 1;
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ s->suspended = 1;
+ /* fallthru */
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (s->running) {
+ snd_nm256_playback_stop(chip);
+ s->running = 0;
+ }
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ spin_unlock(&chip->reg_lock);
+ return err;
+}
+
+static int
+snd_nm256_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+ struct nm256_stream *s = substream->runtime->private_data;
+ int err = 0;
+
+ if (snd_BUG_ON(!s))
+ return -ENXIO;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (! s->running) {
+ snd_nm256_capture_start(chip, s, substream);
+ s->running = 1;
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (s->running) {
+ snd_nm256_capture_stop(chip);
+ s->running = 0;
+ }
+ break;
+ default:
+ err = -EINVAL;
+ break;
+ }
+ spin_unlock(&chip->reg_lock);
+ return err;
+}
+
+
+/*
+ * prepare playback/capture channel
+ */
+static int snd_nm256_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct nm256_stream *s = runtime->private_data;
+
+ if (snd_BUG_ON(!s))
+ return -ENXIO;
+ s->dma_size = frames_to_bytes(runtime, substream->runtime->buffer_size);
+ s->period_size = frames_to_bytes(runtime, substream->runtime->period_size);
+ s->periods = substream->runtime->periods;
+ s->cur_period = 0;
+
+ spin_lock_irq(&chip->reg_lock);
+ s->running = 0;
+ snd_nm256_set_format(chip, s, substream);
+ spin_unlock_irq(&chip->reg_lock);
+
+ return 0;
+}
+
+
+/*
+ * get the current pointer
+ */
+static snd_pcm_uframes_t
+snd_nm256_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+ struct nm256_stream *s = substream->runtime->private_data;
+ unsigned long curp;
+
+ if (snd_BUG_ON(!s))
+ return 0;
+ curp = snd_nm256_readl(chip, NM_PBUFFER_CURRP) - (unsigned long)s->buf;
+ curp %= s->dma_size;
+ return bytes_to_frames(substream->runtime, curp);
+}
+
+static snd_pcm_uframes_t
+snd_nm256_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+ struct nm256_stream *s = substream->runtime->private_data;
+ unsigned long curp;
+
+ if (snd_BUG_ON(!s))
+ return 0;
+ curp = snd_nm256_readl(chip, NM_RBUFFER_CURRP) - (unsigned long)s->buf;
+ curp %= s->dma_size;
+ return bytes_to_frames(substream->runtime, curp);
+}
+
+/* Remapped I/O space can be accessible as pointer on i386 */
+/* This might be changed in the future */
+#ifndef __i386__
+/*
+ * silence / copy for playback
+ */
+static int
+snd_nm256_playback_silence(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct nm256_stream *s = runtime->private_data;
+ count = frames_to_bytes(runtime, count);
+ pos = frames_to_bytes(runtime, pos);
+ memset_io(s->bufptr + pos, 0, count);
+ return 0;
+}
+
+static int
+snd_nm256_playback_copy(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *src,
+ snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct nm256_stream *s = runtime->private_data;
+ count = frames_to_bytes(runtime, count);
+ pos = frames_to_bytes(runtime, pos);
+ if (copy_from_user_toio(s->bufptr + pos, src, count))
+ return -EFAULT;
+ return 0;
+}
+
+/*
+ * copy to user
+ */
+static int
+snd_nm256_capture_copy(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *dst,
+ snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct nm256_stream *s = runtime->private_data;
+ count = frames_to_bytes(runtime, count);
+ pos = frames_to_bytes(runtime, pos);
+ if (copy_to_user_fromio(dst, s->bufptr + pos, count))
+ return -EFAULT;
+ return 0;
+}
+
+#endif /* !__i386__ */
+
+
+/*
+ * update playback/capture watermarks
+ */
+
+/* spinlock held! */
+static void
+snd_nm256_playback_update(struct nm256 *chip)
+{
+ struct nm256_stream *s;
+
+ s = &chip->streams[SNDRV_PCM_STREAM_PLAYBACK];
+ if (s->running && s->substream) {
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(s->substream);
+ spin_lock(&chip->reg_lock);
+ snd_nm256_playback_mark(chip, s);
+ }
+}
+
+/* spinlock held! */
+static void
+snd_nm256_capture_update(struct nm256 *chip)
+{
+ struct nm256_stream *s;
+
+ s = &chip->streams[SNDRV_PCM_STREAM_CAPTURE];
+ if (s->running && s->substream) {
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(s->substream);
+ spin_lock(&chip->reg_lock);
+ snd_nm256_capture_mark(chip, s);
+ }
+}
+
+/*
+ * hardware info
+ */
+static struct snd_pcm_hardware snd_nm256_playback =
+{
+ .info = SNDRV_PCM_INFO_MMAP_IOMEM |SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ /*SNDRV_PCM_INFO_PAUSE |*/
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_KNOT/*24k*/ | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .periods_min = 2,
+ .periods_max = 1024,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 256,
+ .period_bytes_max = 128 * 1024,
+};
+
+static struct snd_pcm_hardware snd_nm256_capture =
+{
+ .info = SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ /*SNDRV_PCM_INFO_PAUSE |*/
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_KNOT/*24k*/ | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .periods_min = 2,
+ .periods_max = 1024,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 256,
+ .period_bytes_max = 128 * 1024,
+};
+
+
+/* set dma transfer size */
+static int snd_nm256_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ /* area and addr are already set and unchanged */
+ substream->runtime->dma_bytes = params_buffer_bytes(hw_params);
+ return 0;
+}
+
+/*
+ * open
+ */
+static void snd_nm256_setup_stream(struct nm256 *chip, struct nm256_stream *s,
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hardware *hw_ptr)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ s->running = 0;
+ runtime->hw = *hw_ptr;
+ runtime->hw.buffer_bytes_max = s->bufsize;
+ runtime->hw.period_bytes_max = s->bufsize / 2;
+ runtime->dma_area = (void __force *) s->bufptr;
+ runtime->dma_addr = s->bufptr_addr;
+ runtime->dma_bytes = s->bufsize;
+ runtime->private_data = s;
+ s->substream = substream;
+
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+}
+
+static int
+snd_nm256_playback_open(struct snd_pcm_substream *substream)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+
+ if (snd_nm256_acquire_irq(chip) < 0)
+ return -EBUSY;
+ snd_nm256_setup_stream(chip, &chip->streams[SNDRV_PCM_STREAM_PLAYBACK],
+ substream, &snd_nm256_playback);
+ return 0;
+}
+
+static int
+snd_nm256_capture_open(struct snd_pcm_substream *substream)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+
+ if (snd_nm256_acquire_irq(chip) < 0)
+ return -EBUSY;
+ snd_nm256_setup_stream(chip, &chip->streams[SNDRV_PCM_STREAM_CAPTURE],
+ substream, &snd_nm256_capture);
+ return 0;
+}
+
+/*
+ * close - we don't have to do special..
+ */
+static int
+snd_nm256_playback_close(struct snd_pcm_substream *substream)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+
+ snd_nm256_release_irq(chip);
+ return 0;
+}
+
+
+static int
+snd_nm256_capture_close(struct snd_pcm_substream *substream)
+{
+ struct nm256 *chip = snd_pcm_substream_chip(substream);
+
+ snd_nm256_release_irq(chip);
+ return 0;
+}
+
+/*
+ * create a pcm instance
+ */
+static struct snd_pcm_ops snd_nm256_playback_ops = {
+ .open = snd_nm256_playback_open,
+ .close = snd_nm256_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_nm256_pcm_hw_params,
+ .prepare = snd_nm256_pcm_prepare,
+ .trigger = snd_nm256_playback_trigger,
+ .pointer = snd_nm256_playback_pointer,
+#ifndef __i386__
+ .copy = snd_nm256_playback_copy,
+ .silence = snd_nm256_playback_silence,
+#endif
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static struct snd_pcm_ops snd_nm256_capture_ops = {
+ .open = snd_nm256_capture_open,
+ .close = snd_nm256_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_nm256_pcm_hw_params,
+ .prepare = snd_nm256_pcm_prepare,
+ .trigger = snd_nm256_capture_trigger,
+ .pointer = snd_nm256_capture_pointer,
+#ifndef __i386__
+ .copy = snd_nm256_capture_copy,
+#endif
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static int __devinit
+snd_nm256_pcm(struct nm256 *chip, int device)
+{
+ struct snd_pcm *pcm;
+ int i, err;
+
+ for (i = 0; i < 2; i++) {
+ struct nm256_stream *s = &chip->streams[i];
+ s->bufptr = chip->buffer + (s->buf - chip->buffer_start);
+ s->bufptr_addr = chip->buffer_addr + (s->buf - chip->buffer_start);
+ }
+
+ err = snd_pcm_new(chip->card, chip->card->driver, device,
+ 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_nm256_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_nm256_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ chip->pcm = pcm;
+
+ return 0;
+}
+
+
+/*
+ * Initialize the hardware.
+ */
+static void
+snd_nm256_init_chip(struct nm256 *chip)
+{
+ /* Reset everything. */
+ snd_nm256_writeb(chip, 0x0, 0x11);
+ snd_nm256_writew(chip, 0x214, 0);
+ /* stop sounds.. */
+ //snd_nm256_playback_stop(chip);
+ //snd_nm256_capture_stop(chip);
+}
+
+
+static irqreturn_t
+snd_nm256_intr_check(struct nm256 *chip)
+{
+ if (chip->badintrcount++ > 1000) {
+ /*
+ * I'm not sure if the best thing is to stop the card from
+ * playing or just release the interrupt (after all, we're in
+ * a bad situation, so doing fancy stuff may not be such a good
+ * idea).
+ *
+ * I worry about the card engine continuing to play noise
+ * over and over, however--that could become a very
+ * obnoxious problem. And we know that when this usually
+ * happens things are fairly safe, it just means the user's
+ * inserted a PCMCIA card and someone's spamming us with IRQ 9s.
+ */
+ if (chip->streams[SNDRV_PCM_STREAM_PLAYBACK].running)
+ snd_nm256_playback_stop(chip);
+ if (chip->streams[SNDRV_PCM_STREAM_CAPTURE].running)
+ snd_nm256_capture_stop(chip);
+ chip->badintrcount = 0;
+ return IRQ_HANDLED;
+ }
+ return IRQ_NONE;
+}
+
+/*
+ * Handle a potential interrupt for the device referred to by DEV_ID.
+ *
+ * I don't like the cut-n-paste job here either between the two routines,
+ * but there are sufficient differences between the two interrupt handlers
+ * that parameterizing it isn't all that great either. (Could use a macro,
+ * I suppose...yucky bleah.)
+ */
+
+static irqreturn_t
+snd_nm256_interrupt(int irq, void *dev_id)
+{
+ struct nm256 *chip = dev_id;
+ u16 status;
+ u8 cbyte;
+
+ status = snd_nm256_readw(chip, NM_INT_REG);
+
+ /* Not ours. */
+ if (status == 0)
+ return snd_nm256_intr_check(chip);
+
+ chip->badintrcount = 0;
+
+ /* Rather boring; check for individual interrupts and process them. */
+
+ spin_lock(&chip->reg_lock);
+ if (status & NM_PLAYBACK_INT) {
+ status &= ~NM_PLAYBACK_INT;
+ NM_ACK_INT(chip, NM_PLAYBACK_INT);
+ snd_nm256_playback_update(chip);
+ }
+
+ if (status & NM_RECORD_INT) {
+ status &= ~NM_RECORD_INT;
+ NM_ACK_INT(chip, NM_RECORD_INT);
+ snd_nm256_capture_update(chip);
+ }
+
+ if (status & NM_MISC_INT_1) {
+ status &= ~NM_MISC_INT_1;
+ NM_ACK_INT(chip, NM_MISC_INT_1);
+ snd_printd("NM256: Got misc interrupt #1\n");
+ snd_nm256_writew(chip, NM_INT_REG, 0x8000);
+ cbyte = snd_nm256_readb(chip, 0x400);
+ snd_nm256_writeb(chip, 0x400, cbyte | 2);
+ }
+
+ if (status & NM_MISC_INT_2) {
+ status &= ~NM_MISC_INT_2;
+ NM_ACK_INT(chip, NM_MISC_INT_2);
+ snd_printd("NM256: Got misc interrupt #2\n");
+ cbyte = snd_nm256_readb(chip, 0x400);
+ snd_nm256_writeb(chip, 0x400, cbyte & ~2);
+ }
+
+ /* Unknown interrupt. */
+ if (status) {
+ snd_printd("NM256: Fire in the hole! Unknown status 0x%x\n",
+ status);
+ /* Pray. */
+ NM_ACK_INT(chip, status);
+ }
+
+ spin_unlock(&chip->reg_lock);
+ return IRQ_HANDLED;
+}
+
+/*
+ * Handle a potential interrupt for the device referred to by DEV_ID.
+ * This handler is for the 256ZX, and is very similar to the non-ZX
+ * routine.
+ */
+
+static irqreturn_t
+snd_nm256_interrupt_zx(int irq, void *dev_id)
+{
+ struct nm256 *chip = dev_id;
+ u32 status;
+ u8 cbyte;
+
+ status = snd_nm256_readl(chip, NM_INT_REG);
+
+ /* Not ours. */
+ if (status == 0)
+ return snd_nm256_intr_check(chip);
+
+ chip->badintrcount = 0;
+
+ /* Rather boring; check for individual interrupts and process them. */
+
+ spin_lock(&chip->reg_lock);
+ if (status & NM2_PLAYBACK_INT) {
+ status &= ~NM2_PLAYBACK_INT;
+ NM2_ACK_INT(chip, NM2_PLAYBACK_INT);
+ snd_nm256_playback_update(chip);
+ }
+
+ if (status & NM2_RECORD_INT) {
+ status &= ~NM2_RECORD_INT;
+ NM2_ACK_INT(chip, NM2_RECORD_INT);
+ snd_nm256_capture_update(chip);
+ }
+
+ if (status & NM2_MISC_INT_1) {
+ status &= ~NM2_MISC_INT_1;
+ NM2_ACK_INT(chip, NM2_MISC_INT_1);
+ snd_printd("NM256: Got misc interrupt #1\n");
+ cbyte = snd_nm256_readb(chip, 0x400);
+ snd_nm256_writeb(chip, 0x400, cbyte | 2);
+ }
+
+ if (status & NM2_MISC_INT_2) {
+ status &= ~NM2_MISC_INT_2;
+ NM2_ACK_INT(chip, NM2_MISC_INT_2);
+ snd_printd("NM256: Got misc interrupt #2\n");
+ cbyte = snd_nm256_readb(chip, 0x400);
+ snd_nm256_writeb(chip, 0x400, cbyte & ~2);
+ }
+
+ /* Unknown interrupt. */
+ if (status) {
+ snd_printd("NM256: Fire in the hole! Unknown status 0x%x\n",
+ status);
+ /* Pray. */
+ NM2_ACK_INT(chip, status);
+ }
+
+ spin_unlock(&chip->reg_lock);
+ return IRQ_HANDLED;
+}
+
+/*
+ * AC97 interface
+ */
+
+/*
+ * Waits for the mixer to become ready to be written; returns a zero value
+ * if it timed out.
+ */
+static int
+snd_nm256_ac97_ready(struct nm256 *chip)
+{
+ int timeout = 10;
+ u32 testaddr;
+ u16 testb;
+
+ testaddr = chip->mixer_status_offset;
+ testb = chip->mixer_status_mask;
+
+ /*
+ * Loop around waiting for the mixer to become ready.
+ */
+ while (timeout-- > 0) {
+ if ((snd_nm256_readw(chip, testaddr) & testb) == 0)
+ return 1;
+ udelay(100);
+ }
+ return 0;
+}
+
+/*
+ * Initial register values to be written to the AC97 mixer.
+ * While most of these are identical to the reset values, we do this
+ * so that we have most of the register contents cached--this avoids
+ * reading from the mixer directly (which seems to be problematic,
+ * probably due to ignorance).
+ */
+
+struct initialValues {
+ unsigned short reg;
+ unsigned short value;
+};
+
+static struct initialValues nm256_ac97_init_val[] =
+{
+ { AC97_MASTER, 0x8000 },
+ { AC97_HEADPHONE, 0x8000 },
+ { AC97_MASTER_MONO, 0x8000 },
+ { AC97_PC_BEEP, 0x8000 },
+ { AC97_PHONE, 0x8008 },
+ { AC97_MIC, 0x8000 },
+ { AC97_LINE, 0x8808 },
+ { AC97_CD, 0x8808 },
+ { AC97_VIDEO, 0x8808 },
+ { AC97_AUX, 0x8808 },
+ { AC97_PCM, 0x8808 },
+ { AC97_REC_SEL, 0x0000 },
+ { AC97_REC_GAIN, 0x0B0B },
+ { AC97_GENERAL_PURPOSE, 0x0000 },
+ { AC97_3D_CONTROL, 0x8000 },
+ { AC97_VENDOR_ID1, 0x8384 },
+ { AC97_VENDOR_ID2, 0x7609 },
+};
+
+static int nm256_ac97_idx(unsigned short reg)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(nm256_ac97_init_val); i++)
+ if (nm256_ac97_init_val[i].reg == reg)
+ return i;
+ return -1;
+}
+
+/*
+ * some nm256 easily crash when reading from mixer registers
+ * thus we're treating it as a write-only mixer and cache the
+ * written values
+ */
+static unsigned short
+snd_nm256_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct nm256 *chip = ac97->private_data;
+ int idx = nm256_ac97_idx(reg);
+
+ if (idx < 0)
+ return 0;
+ return chip->ac97_regs[idx];
+}
+
+/*
+ */
+static void
+snd_nm256_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ struct nm256 *chip = ac97->private_data;
+ int tries = 2;
+ int idx = nm256_ac97_idx(reg);
+ u32 base;
+
+ if (idx < 0)
+ return;
+
+ base = chip->mixer_base;
+
+ snd_nm256_ac97_ready(chip);
+
+ /* Wait for the write to take, too. */
+ while (tries-- > 0) {
+ snd_nm256_writew(chip, base + reg, val);
+ msleep(1); /* a little delay here seems better.. */
+ if (snd_nm256_ac97_ready(chip)) {
+ /* successful write: set cache */
+ chip->ac97_regs[idx] = val;
+ return;
+ }
+ }
+ snd_printd("nm256: ac97 codec not ready..\n");
+}
+
+/* static resolution table */
+static struct snd_ac97_res_table nm256_res_table[] = {
+ { AC97_MASTER, 0x1f1f },
+ { AC97_HEADPHONE, 0x1f1f },
+ { AC97_MASTER_MONO, 0x001f },
+ { AC97_PC_BEEP, 0x001f },
+ { AC97_PHONE, 0x001f },
+ { AC97_MIC, 0x001f },
+ { AC97_LINE, 0x1f1f },
+ { AC97_CD, 0x1f1f },
+ { AC97_VIDEO, 0x1f1f },
+ { AC97_AUX, 0x1f1f },
+ { AC97_PCM, 0x1f1f },
+ { AC97_REC_GAIN, 0x0f0f },
+ { } /* terminator */
+};
+
+/* initialize the ac97 into a known state */
+static void
+snd_nm256_ac97_reset(struct snd_ac97 *ac97)
+{
+ struct nm256 *chip = ac97->private_data;
+
+ /* Reset the mixer. 'Tis magic! */
+ snd_nm256_writeb(chip, 0x6c0, 1);
+ if (! chip->reset_workaround) {
+ /* Dell latitude LS will lock up by this */
+ snd_nm256_writeb(chip, 0x6cc, 0x87);
+ }
+ if (! chip->reset_workaround_2) {
+ /* Dell latitude CSx will lock up by this */
+ snd_nm256_writeb(chip, 0x6cc, 0x80);
+ snd_nm256_writeb(chip, 0x6cc, 0x0);
+ }
+ if (! chip->in_resume) {
+ int i;
+ for (i = 0; i < ARRAY_SIZE(nm256_ac97_init_val); i++) {
+ /* preload the cache, so as to avoid even a single
+ * read of the mixer regs
+ */
+ snd_nm256_ac97_write(ac97, nm256_ac97_init_val[i].reg,
+ nm256_ac97_init_val[i].value);
+ }
+ }
+}
+
+/* create an ac97 mixer interface */
+static int __devinit
+snd_nm256_mixer(struct nm256 *chip)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .reset = snd_nm256_ac97_reset,
+ .write = snd_nm256_ac97_write,
+ .read = snd_nm256_ac97_read,
+ };
+
+ chip->ac97_regs = kcalloc(ARRAY_SIZE(nm256_ac97_init_val),
+ sizeof(short), GFP_KERNEL);
+ if (! chip->ac97_regs)
+ return -ENOMEM;
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.scaps = AC97_SCAP_AUDIO; /* we support audio! */
+ ac97.private_data = chip;
+ ac97.res_table = nm256_res_table;
+ pbus->no_vra = 1;
+ err = snd_ac97_mixer(pbus, &ac97, &chip->ac97);
+ if (err < 0)
+ return err;
+ if (! (chip->ac97->id & (0xf0000000))) {
+ /* looks like an invalid id */
+ sprintf(chip->card->mixername, "%s AC97", chip->card->driver);
+ }
+ return 0;
+}
+
+/*
+ * See if the signature left by the NM256 BIOS is intact; if so, we use
+ * the associated address as the end of our audio buffer in the video
+ * RAM.
+ */
+
+static int __devinit
+snd_nm256_peek_for_sig(struct nm256 *chip)
+{
+ /* The signature is located 1K below the end of video RAM. */
+ void __iomem *temp;
+ /* Default buffer end is 5120 bytes below the top of RAM. */
+ unsigned long pointer_found = chip->buffer_end - 0x1400;
+ u32 sig;
+
+ temp = ioremap_nocache(chip->buffer_addr + chip->buffer_end - 0x400, 16);
+ if (temp == NULL) {
+ snd_printk(KERN_ERR "Unable to scan for card signature in video RAM\n");
+ return -EBUSY;
+ }
+
+ sig = readl(temp);
+ if ((sig & NM_SIG_MASK) == NM_SIGNATURE) {
+ u32 pointer = readl(temp + 4);
+
+ /*
+ * If it's obviously invalid, don't use it
+ */
+ if (pointer == 0xffffffff ||
+ pointer < chip->buffer_size ||
+ pointer > chip->buffer_end) {
+ snd_printk(KERN_ERR "invalid signature found: 0x%x\n", pointer);
+ iounmap(temp);
+ return -ENODEV;
+ } else {
+ pointer_found = pointer;
+ printk(KERN_INFO "nm256: found card signature in video RAM: 0x%x\n",
+ pointer);
+ }
+ }
+
+ iounmap(temp);
+ chip->buffer_end = pointer_found;
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * APM event handler, so the card is properly reinitialized after a power
+ * event.
+ */
+static int nm256_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct nm256 *chip = card->private_data;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_ac97_suspend(chip->ac97);
+ chip->coeffs_current = 0;
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int nm256_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct nm256 *chip = card->private_data;
+ int i;
+
+ /* Perform a full reset on the hardware */
+ chip->in_resume = 1;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "nm256: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_nm256_init_chip(chip);
+
+ /* restore ac97 */
+ snd_ac97_resume(chip->ac97);
+
+ for (i = 0; i < 2; i++) {
+ struct nm256_stream *s = &chip->streams[i];
+ if (s->substream && s->suspended) {
+ spin_lock_irq(&chip->reg_lock);
+ snd_nm256_set_format(chip, s, s->substream);
+ spin_unlock_irq(&chip->reg_lock);
+ }
+ }
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ chip->in_resume = 0;
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static int snd_nm256_free(struct nm256 *chip)
+{
+ if (chip->streams[SNDRV_PCM_STREAM_PLAYBACK].running)
+ snd_nm256_playback_stop(chip);
+ if (chip->streams[SNDRV_PCM_STREAM_CAPTURE].running)
+ snd_nm256_capture_stop(chip);
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+
+ if (chip->cport)
+ iounmap(chip->cport);
+ if (chip->buffer)
+ iounmap(chip->buffer);
+ release_and_free_resource(chip->res_cport);
+ release_and_free_resource(chip->res_buffer);
+
+ pci_disable_device(chip->pci);
+ kfree(chip->ac97_regs);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_nm256_dev_free(struct snd_device *device)
+{
+ struct nm256 *chip = device->device_data;
+ return snd_nm256_free(chip);
+}
+
+static int __devinit
+snd_nm256_create(struct snd_card *card, struct pci_dev *pci,
+ struct nm256 **chip_ret)
+{
+ struct nm256 *chip;
+ int err, pval;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_nm256_dev_free,
+ };
+ u32 addr;
+
+ *chip_ret = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->use_cache = use_cache;
+ spin_lock_init(&chip->reg_lock);
+ chip->irq = -1;
+ mutex_init(&chip->irq_mutex);
+
+ /* store buffer sizes in bytes */
+ chip->streams[SNDRV_PCM_STREAM_PLAYBACK].bufsize = playback_bufsize * 1024;
+ chip->streams[SNDRV_PCM_STREAM_CAPTURE].bufsize = capture_bufsize * 1024;
+
+ /*
+ * The NM256 has two memory ports. The first port is nothing
+ * more than a chunk of video RAM, which is used as the I/O ring
+ * buffer. The second port has the actual juicy stuff (like the
+ * mixer and the playback engine control registers).
+ */
+
+ chip->buffer_addr = pci_resource_start(pci, 0);
+ chip->cport_addr = pci_resource_start(pci, 1);
+
+ /* Init the memory port info. */
+ /* remap control port (#2) */
+ chip->res_cport = request_mem_region(chip->cport_addr, NM_PORT2_SIZE,
+ card->driver);
+ if (chip->res_cport == NULL) {
+ snd_printk(KERN_ERR "memory region 0x%lx (size 0x%x) busy\n",
+ chip->cport_addr, NM_PORT2_SIZE);
+ err = -EBUSY;
+ goto __error;
+ }
+ chip->cport = ioremap_nocache(chip->cport_addr, NM_PORT2_SIZE);
+ if (chip->cport == NULL) {
+ snd_printk(KERN_ERR "unable to map control port %lx\n", chip->cport_addr);
+ err = -ENOMEM;
+ goto __error;
+ }
+
+ if (!strcmp(card->driver, "NM256AV")) {
+ /* Ok, try to see if this is a non-AC97 version of the hardware. */
+ pval = snd_nm256_readw(chip, NM_MIXER_PRESENCE);
+ if ((pval & NM_PRESENCE_MASK) != NM_PRESENCE_VALUE) {
+ if (! force_ac97) {
+ printk(KERN_ERR "nm256: no ac97 is found!\n");
+ printk(KERN_ERR " force the driver to load by "
+ "passing in the module parameter\n");
+ printk(KERN_ERR " force_ac97=1\n");
+ printk(KERN_ERR " or try sb16, opl3sa2, or "
+ "cs423x drivers instead.\n");
+ err = -ENXIO;
+ goto __error;
+ }
+ }
+ chip->buffer_end = 2560 * 1024;
+ chip->interrupt = snd_nm256_interrupt;
+ chip->mixer_status_offset = NM_MIXER_STATUS_OFFSET;
+ chip->mixer_status_mask = NM_MIXER_READY_MASK;
+ } else {
+ /* Not sure if there is any relevant detect for the ZX or not. */
+ if (snd_nm256_readb(chip, 0xa0b) != 0)
+ chip->buffer_end = 6144 * 1024;
+ else
+ chip->buffer_end = 4096 * 1024;
+
+ chip->interrupt = snd_nm256_interrupt_zx;
+ chip->mixer_status_offset = NM2_MIXER_STATUS_OFFSET;
+ chip->mixer_status_mask = NM2_MIXER_READY_MASK;
+ }
+
+ chip->buffer_size = chip->streams[SNDRV_PCM_STREAM_PLAYBACK].bufsize +
+ chip->streams[SNDRV_PCM_STREAM_CAPTURE].bufsize;
+ if (chip->use_cache)
+ chip->buffer_size += NM_TOTAL_COEFF_COUNT * 4;
+ else
+ chip->buffer_size += NM_MAX_PLAYBACK_COEF_SIZE + NM_MAX_RECORD_COEF_SIZE;
+
+ if (buffer_top >= chip->buffer_size && buffer_top < chip->buffer_end)
+ chip->buffer_end = buffer_top;
+ else {
+ /* get buffer end pointer from signature */
+ if ((err = snd_nm256_peek_for_sig(chip)) < 0)
+ goto __error;
+ }
+
+ chip->buffer_start = chip->buffer_end - chip->buffer_size;
+ chip->buffer_addr += chip->buffer_start;
+
+ printk(KERN_INFO "nm256: Mapping port 1 from 0x%x - 0x%x\n",
+ chip->buffer_start, chip->buffer_end);
+
+ chip->res_buffer = request_mem_region(chip->buffer_addr,
+ chip->buffer_size,
+ card->driver);
+ if (chip->res_buffer == NULL) {
+ snd_printk(KERN_ERR "nm256: buffer 0x%lx (size 0x%x) busy\n",
+ chip->buffer_addr, chip->buffer_size);
+ err = -EBUSY;
+ goto __error;
+ }
+ chip->buffer = ioremap_nocache(chip->buffer_addr, chip->buffer_size);
+ if (chip->buffer == NULL) {
+ err = -ENOMEM;
+ snd_printk(KERN_ERR "unable to map ring buffer at %lx\n", chip->buffer_addr);
+ goto __error;
+ }
+
+ /* set offsets */
+ addr = chip->buffer_start;
+ chip->streams[SNDRV_PCM_STREAM_PLAYBACK].buf = addr;
+ addr += chip->streams[SNDRV_PCM_STREAM_PLAYBACK].bufsize;
+ chip->streams[SNDRV_PCM_STREAM_CAPTURE].buf = addr;
+ addr += chip->streams[SNDRV_PCM_STREAM_CAPTURE].bufsize;
+ if (chip->use_cache) {
+ chip->all_coeff_buf = addr;
+ } else {
+ chip->coeff_buf[SNDRV_PCM_STREAM_PLAYBACK] = addr;
+ addr += NM_MAX_PLAYBACK_COEF_SIZE;
+ chip->coeff_buf[SNDRV_PCM_STREAM_CAPTURE] = addr;
+ }
+
+ /* Fixed setting. */
+ chip->mixer_base = NM_MIXER_OFFSET;
+
+ chip->coeffs_current = 0;
+
+ snd_nm256_init_chip(chip);
+
+ // pci_set_master(pci); /* needed? */
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0)
+ goto __error;
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *chip_ret = chip;
+ return 0;
+
+__error:
+ snd_nm256_free(chip);
+ return err;
+}
+
+
+enum { NM_BLACKLISTED, NM_RESET_WORKAROUND, NM_RESET_WORKAROUND_2 };
+
+static struct snd_pci_quirk nm256_quirks[] __devinitdata = {
+ /* HP omnibook 4150 has cs4232 codec internally */
+ SND_PCI_QUIRK(0x103c, 0x0007, "HP omnibook 4150", NM_BLACKLISTED),
+ /* Reset workarounds to avoid lock-ups */
+ SND_PCI_QUIRK(0x104d, 0x8041, "Sony PCG-F305", NM_RESET_WORKAROUND),
+ SND_PCI_QUIRK(0x1028, 0x0080, "Dell Latitude LS", NM_RESET_WORKAROUND),
+ SND_PCI_QUIRK(0x1028, 0x0091, "Dell Latitude CSx", NM_RESET_WORKAROUND_2),
+ { } /* terminator */
+};
+
+
+static int __devinit snd_nm256_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct nm256 *chip;
+ int err;
+ const struct snd_pci_quirk *q;
+
+ q = snd_pci_quirk_lookup(pci, nm256_quirks);
+ if (q) {
+ snd_printdd(KERN_INFO "nm256: Enabled quirk for %s.\n", q->name);
+ switch (q->value) {
+ case NM_BLACKLISTED:
+ printk(KERN_INFO "nm256: The device is blacklisted. "
+ "Loading stopped\n");
+ return -ENODEV;
+ case NM_RESET_WORKAROUND_2:
+ reset_workaround_2 = 1;
+ /* Fall-through */
+ case NM_RESET_WORKAROUND:
+ reset_workaround = 1;
+ break;
+ }
+ }
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ switch (pci->device) {
+ case PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO:
+ strcpy(card->driver, "NM256AV");
+ break;
+ case PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO:
+ strcpy(card->driver, "NM256ZX");
+ break;
+ case PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO:
+ strcpy(card->driver, "NM256XL+");
+ break;
+ default:
+ snd_printk(KERN_ERR "invalid device id 0x%x\n", pci->device);
+ snd_card_free(card);
+ return -EINVAL;
+ }
+
+ if (vaio_hack)
+ buffer_top = 0x25a800; /* this avoids conflicts with XFree86 server */
+
+ if (playback_bufsize < 4)
+ playback_bufsize = 4;
+ if (playback_bufsize > 128)
+ playback_bufsize = 128;
+ if (capture_bufsize < 4)
+ capture_bufsize = 4;
+ if (capture_bufsize > 128)
+ capture_bufsize = 128;
+ if ((err = snd_nm256_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+
+ if (reset_workaround) {
+ snd_printdd(KERN_INFO "nm256: reset_workaround activated\n");
+ chip->reset_workaround = 1;
+ }
+
+ if (reset_workaround_2) {
+ snd_printdd(KERN_INFO "nm256: reset_workaround_2 activated\n");
+ chip->reset_workaround_2 = 1;
+ }
+
+ if ((err = snd_nm256_pcm(chip, 0)) < 0 ||
+ (err = snd_nm256_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ sprintf(card->shortname, "NeoMagic %s", card->driver);
+ sprintf(card->longname, "%s at 0x%lx & 0x%lx, irq %d",
+ card->shortname,
+ chip->buffer_addr, chip->cport_addr, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ return 0;
+}
+
+static void __devexit snd_nm256_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+
+static struct pci_driver driver = {
+ .name = "NeoMagic 256",
+ .id_table = snd_nm256_ids,
+ .probe = snd_nm256_probe,
+ .remove = __devexit_p(snd_nm256_remove),
+#ifdef CONFIG_PM
+ .suspend = nm256_suspend,
+ .resume = nm256_resume,
+#endif
+};
+
+
+static int __init alsa_card_nm256_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_nm256_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_nm256_init)
+module_exit(alsa_card_nm256_exit)
diff --git a/sound/pci/nm256/nm256_coef.c b/sound/pci/nm256/nm256_coef.c
new file mode 100644
index 0000000..747d5d6
--- /dev/null
+++ b/sound/pci/nm256/nm256_coef.c
@@ -0,0 +1,4607 @@
+#define NM_TOTAL_COEFF_COUNT 0x3158
+
+static char coefficients[NM_TOTAL_COEFF_COUNT * 4] = {
+ 0xFF, 0xFF, 0x2F, 0x00, 0x4B, 0xFF, 0xA5, 0x01, 0xEF, 0xFC, 0x21,
+ 0x05, 0x87, 0xF7, 0x62, 0x11, 0xE9, 0x45, 0x5E, 0xF9, 0xB5, 0x01,
+ 0xDE, 0xFF, 0xA4, 0xFF, 0x60, 0x00, 0xCA, 0xFF, 0x0D, 0x00, 0xFD,
+ 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3D, 0xFC, 0xD6, 0x06,
+ 0x4C, 0xF3, 0xED, 0x20, 0x3D, 0x3D, 0x4A, 0xF3, 0x4E, 0x05, 0xB1,
+ 0xFD, 0xE1, 0x00, 0xC3, 0xFF, 0x05, 0x00, 0x02, 0x00, 0xFD, 0xFF,
+ 0x2A, 0x00, 0x5C, 0xFF, 0xAA, 0x01, 0x71, 0xFC, 0x07, 0x07, 0x7E,
+ 0xF1, 0x44, 0x30, 0x44, 0x30, 0x7E, 0xF1, 0x07, 0x07, 0x71, 0xFC,
+ 0xAA, 0x01, 0x5C, 0xFF, 0x2A, 0x00, 0xFD, 0xFF, 0x02, 0x00, 0x05,
+ 0x00, 0xC3, 0xFF, 0xE1, 0x00, 0xB1, 0xFD, 0x4E, 0x05, 0x4A, 0xF3,
+ 0x3D, 0x3D, 0xED, 0x20, 0x4C, 0xF3, 0xD6, 0x06, 0x3D, 0xFC, 0xE6,
+ 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x0D, 0x00, 0xCA, 0xFF,
+ 0x60, 0x00, 0xA4, 0xFF, 0xDE, 0xFF, 0xB5, 0x01, 0x5E, 0xF9, 0xE9,
+ 0x45, 0x62, 0x11, 0x87, 0xF7, 0x21, 0x05, 0xEF, 0xFC, 0xA5, 0x01,
+ 0x4B, 0xFF, 0x2F, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x1E, 0x00, 0x84,
+ 0xFF, 0x11, 0x01, 0x34, 0xFE, 0x8F, 0x02, 0xC7, 0xFC, 0xAE, 0x03,
+ 0xF7, 0x48, 0xAE, 0x03, 0xC7, 0xFC, 0x8F, 0x02, 0x34, 0xFE, 0x11,
+ 0x01, 0x84, 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3D, 0xFF,
+ 0xCA, 0x01, 0x95, 0xFC, 0xEA, 0x05, 0xBB, 0xF5, 0x25, 0x17, 0x3C,
+ 0x43, 0x8D, 0xF6, 0x43, 0x03, 0xF5, 0xFE, 0x26, 0x00, 0x20, 0x00,
+ 0xE2, 0xFF, 0x08, 0x00, 0xFD, 0xFF, 0x30, 0x00, 0x4D, 0xFF, 0xC5,
+ 0x01, 0x4C, 0xFC, 0x26, 0x07, 0xA3, 0xF1, 0xAB, 0x2C, 0xBB, 0x33,
+ 0x8F, 0xF1, 0xCA, 0x06, 0xA6, 0xFC, 0x85, 0x01, 0x6F, 0xFF, 0x24,
+ 0x00, 0xFD, 0xFF, 0x03, 0x00, 0xFE, 0xFF, 0xD5, 0xFF, 0xBC, 0x00,
+ 0xF0, 0xFD, 0xEC, 0x04, 0xD9, 0xF3, 0xB1, 0x3E, 0xCD, 0x1E, 0xC1,
+ 0xF3, 0xAF, 0x06, 0x49, 0xFC, 0xE4, 0x01, 0x36, 0xFF, 0x36, 0x00,
+ 0xFE, 0xFF, 0x16, 0x00, 0xA6, 0xFF, 0xBB, 0x00, 0xE9, 0xFE, 0x38,
+ 0x01, 0x4B, 0xFF, 0x28, 0xFE, 0x3A, 0x48, 0x04, 0x0A, 0x2E, 0xFA,
+ 0xDF, 0x03, 0x8A, 0xFD, 0x60, 0x01, 0x65, 0xFF, 0x27, 0x00, 0x00,
+ 0x00, 0xFF, 0xFF, 0x2E, 0x00, 0x50, 0xFF, 0x98, 0x01, 0x0D, 0xFD,
+ 0xE0, 0x04, 0x14, 0xF8, 0xC3, 0x0F, 0x89, 0x46, 0x4C, 0xFA, 0x38,
+ 0x01, 0x25, 0x00, 0x7D, 0xFF, 0x73, 0x00, 0xC2, 0xFF, 0x0F, 0x00,
+ 0xFD, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xE3, 0x01, 0x31, 0xFC, 0x0F,
+ 0x07, 0x84, 0xF2, 0x29, 0x25, 0x1A, 0x3A, 0x67, 0xF2, 0xF6, 0x05,
+ 0x41, 0xFD, 0x24, 0x01, 0xA1, 0xFF, 0x12, 0x00, 0x00, 0x00, 0xFF,
+ 0xFF, 0x15, 0x00, 0x97, 0xFF, 0x37, 0x01, 0x22, 0xFD, 0x23, 0x06,
+ 0x2F, 0xF2, 0x11, 0x39, 0x7B, 0x26, 0x50, 0xF2, 0x1B, 0x07, 0x32,
+ 0xFC, 0xE1, 0x01, 0x3C, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x0E, 0x00,
+ 0xC8, 0xFF, 0x64, 0x00, 0x9B, 0xFF, 0xEE, 0xFF, 0x98, 0x01, 0x93,
+ 0xF9, 0x10, 0x46, 0x03, 0x11, 0xA7, 0xF7, 0x12, 0x05, 0xF6, 0xFC,
+ 0xA2, 0x01, 0x4C, 0xFF, 0x2F, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x26,
+ 0x00, 0x6A, 0xFF, 0x53, 0x01, 0xA6, 0xFD, 0xA6, 0x03, 0xA1, 0xFA,
+ 0xDE, 0x08, 0x76, 0x48, 0x0C, 0xFF, 0xDE, 0xFE, 0x73, 0x01, 0xC9,
+ 0xFE, 0xCA, 0x00, 0xA0, 0xFF, 0x17, 0x00, 0xFE, 0xFF, 0x36, 0x00,
+ 0x36, 0xFF, 0xE1, 0x01, 0x52, 0xFC, 0x93, 0x06, 0x10, 0xF4, 0x78,
+ 0x1D, 0x90, 0x3F, 0x3E, 0xF4, 0xAA, 0x04, 0x19, 0xFE, 0xA4, 0x00,
+ 0xE2, 0xFF, 0xFA, 0xFF, 0x03, 0x00, 0xFD, 0xFF, 0x26, 0x00, 0x68,
+ 0xFF, 0x93, 0x01, 0x92, 0xFC, 0xE2, 0x06, 0x83, 0xF1, 0x8C, 0x32,
+ 0xED, 0x2D, 0x90, 0xF1, 0x1E, 0x07, 0x57, 0xFC, 0xBD, 0x01, 0x51,
+ 0xFF, 0x2E, 0x00, 0xFD, 0xFF, 0x07, 0x00, 0xE8, 0xFF, 0x12, 0x00,
+ 0x42, 0x00, 0xC4, 0xFE, 0x94, 0x03, 0x02, 0xF6, 0x89, 0x42, 0x76,
+ 0x18, 0x5C, 0xF5, 0x12, 0x06, 0x84, 0xFC, 0xD1, 0x01, 0x3B, 0xFF,
+ 0x34, 0x00, 0xFE, 0xFF, 0x1D, 0x00, 0x8A, 0xFF, 0x03, 0x01, 0x53,
+ 0xFE, 0x53, 0x02, 0x39, 0xFD, 0xA9, 0x02, 0xF2, 0x48, 0xB9, 0x04,
+ 0x54, 0xFC, 0xCA, 0x02, 0x16, 0xFE, 0x20, 0x01, 0x7F, 0xFF, 0x20,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x40, 0xFF, 0xC3, 0x01,
+ 0xA7, 0xFC, 0xC0, 0x05, 0x1E, 0xF6, 0xD8, 0x15, 0xE7, 0x43, 0x20,
+ 0xF7, 0xEF, 0x02, 0x27, 0xFF, 0x0A, 0x00, 0x2E, 0x00, 0xDD, 0xFF,
+ 0x09, 0x00, 0xFD, 0xFF, 0x31, 0x00, 0x48, 0xFF, 0xCD, 0x01, 0x43,
+ 0xFC, 0x2A, 0x07, 0xBC, 0xF1, 0x64, 0x2B, 0xE3, 0x34, 0xA3, 0xF1,
+ 0xAE, 0x06, 0xBD, 0xFC, 0x77, 0x01, 0x77, 0xFF, 0x21, 0x00, 0xFE,
+ 0xFF, 0x02, 0x00, 0x03, 0x00, 0xCA, 0xFF, 0xD4, 0x00, 0xC8, 0xFD,
+ 0x2A, 0x05, 0x7D, 0xF3, 0xCA, 0x3D, 0x22, 0x20, 0x76, 0xF3, 0xC8,
+ 0x06, 0x41, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0x14, 0x00, 0xAC, 0xFF, 0xAC, 0x00, 0x08, 0xFF, 0xFD, 0x00, 0xB5,
+ 0xFF, 0x4B, 0xFD, 0xF4, 0x47, 0x30, 0x0B, 0xBC, 0xF9, 0x17, 0x04,
+ 0x6E, 0xFD, 0x6D, 0x01, 0x60, 0xFF, 0x29, 0x00, 0x00, 0x00, 0xFF,
+ 0xFF, 0x2C, 0x00, 0x54, 0xFF, 0x8D, 0x01, 0x26, 0xFD, 0xAD, 0x04,
+ 0x82, 0xF8, 0x87, 0x0E, 0xF9, 0x46, 0x0C, 0xFB, 0xD4, 0x00, 0x5D,
+ 0x00, 0x5E, 0xFF, 0x82, 0x00, 0xBD, 0xFF, 0x10, 0x00, 0xFD, 0xFF,
+ 0x36, 0x00, 0x38, 0xFF, 0xE5, 0x01, 0x33, 0xFC, 0x01, 0x07, 0xBE,
+ 0xF2, 0xD6, 0x23, 0x1F, 0x3B, 0xA5, 0xF2, 0xC5, 0x05, 0x62, 0xFD,
+ 0x10, 0x01, 0xAB, 0xFF, 0x0E, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x19,
+ 0x00, 0x8E, 0xFF, 0x49, 0x01, 0x04, 0xFD, 0x4D, 0x06, 0x00, 0xF2,
+ 0xFE, 0x37, 0xCB, 0x27, 0x21, 0xF2, 0x23, 0x07, 0x34, 0xFC, 0xDD,
+ 0x01, 0x3F, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x0C, 0x00, 0xCE, 0xFF,
+ 0x56, 0x00, 0xB9, 0xFF, 0xB8, 0xFF, 0xF7, 0x01, 0xE2, 0xF8, 0x8D,
+ 0x45, 0x46, 0x12, 0x3C, 0xF7, 0x43, 0x05, 0xDF, 0xFC, 0xAC, 0x01,
+ 0x48, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x24, 0x00, 0x70,
+ 0xFF, 0x46, 0x01, 0xC3, 0xFD, 0x6D, 0x03, 0x14, 0xFB, 0xBE, 0x07,
+ 0xA6, 0x48, 0xF8, 0xFF, 0x70, 0xFE, 0xAE, 0x01, 0xAA, 0xFE, 0xD9,
+ 0x00, 0x9A, 0xFF, 0x19, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF,
+ 0xDE, 0x01, 0x5D, 0xFC, 0x74, 0x06, 0x63, 0xF4, 0x23, 0x1C, 0x66,
+ 0x40, 0xAA, 0xF4, 0x65, 0x04, 0x44, 0xFE, 0x8B, 0x00, 0xEE, 0xFF,
+ 0xF5, 0xFF, 0x04, 0x00, 0xFD, 0xFF, 0x29, 0x00, 0x61, 0xFF, 0x9F,
+ 0x01, 0x80, 0xFC, 0xF7, 0x06, 0x7D, 0xF1, 0x5A, 0x31, 0x2C, 0x2F,
+ 0x83, 0xF1, 0x13, 0x07, 0x64, 0xFC, 0xB3, 0x01, 0x57, 0xFF, 0x2C,
+ 0x00, 0xFD, 0xFF, 0x06, 0x00, 0xED, 0xFF, 0x05, 0x00, 0x5D, 0x00,
+ 0x95, 0xFE, 0xE2, 0x03, 0x7F, 0xF5, 0xCC, 0x41, 0xC7, 0x19, 0xFF,
+ 0xF4, 0x37, 0x06, 0x75, 0xFC, 0xD6, 0x01, 0x39, 0xFF, 0x35, 0x00,
+ 0xFE, 0xFF, 0x1B, 0x00, 0x90, 0xFF, 0xF4, 0x00, 0x72, 0xFE, 0x18,
+ 0x02, 0xAA, 0xFD, 0xAB, 0x01, 0xDF, 0x48, 0xCA, 0x05, 0xE1, 0xFB,
+ 0x05, 0x03, 0xF7, 0xFD, 0x2E, 0x01, 0x79, 0xFF, 0x21, 0x00, 0x00,
+ 0x00, 0xFF, 0xFF, 0x32, 0x00, 0x43, 0xFF, 0xBB, 0x01, 0xBA, 0xFC,
+ 0x95, 0x05, 0x83, 0xF6, 0x8C, 0x14, 0x87, 0x44, 0xBB, 0xF7, 0x98,
+ 0x02, 0x5A, 0xFF, 0xEE, 0xFF, 0x3C, 0x00, 0xD8, 0xFF, 0x0A, 0x00,
+ 0xFD, 0xFF, 0x32, 0x00, 0x44, 0xFF, 0xD3, 0x01, 0x3C, 0xFC, 0x2A,
+ 0x07, 0xDC, 0xF1, 0x1A, 0x2A, 0x06, 0x36, 0xBE, 0xF1, 0x8E, 0x06,
+ 0xD5, 0xFC, 0x67, 0x01, 0x7F, 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0x01,
+ 0x00, 0x07, 0x00, 0xBE, 0xFF, 0xEA, 0x00, 0xA2, 0xFD, 0x65, 0x05,
+ 0x28, 0xF3, 0xDB, 0x3C, 0x78, 0x21, 0x30, 0xF3, 0xDF, 0x06, 0x3A,
+ 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x13, 0x00,
+ 0xB2, 0xFF, 0x9D, 0x00, 0x27, 0xFF, 0xC3, 0x00, 0x1F, 0x00, 0x76,
+ 0xFC, 0xA3, 0x47, 0x60, 0x0C, 0x4A, 0xF9, 0x4E, 0x04, 0x53, 0xFD,
+ 0x79, 0x01, 0x5C, 0xFF, 0x2A, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2B,
+ 0x00, 0x58, 0xFF, 0x82, 0x01, 0x3F, 0xFD, 0x78, 0x04, 0xF2, 0xF8,
+ 0x50, 0x0D, 0x5E, 0x47, 0xD5, 0xFB, 0x6F, 0x00, 0x96, 0x00, 0x40,
+ 0xFF, 0x91, 0x00, 0xB7, 0xFF, 0x12, 0x00, 0xFD, 0xFF, 0x36, 0x00,
+ 0x37, 0xFF, 0xE6, 0x01, 0x36, 0xFC, 0xEF, 0x06, 0xFC, 0xF2, 0x81,
+ 0x22, 0x1C, 0x3C, 0xEC, 0xF2, 0x90, 0x05, 0x85, 0xFD, 0xFB, 0x00,
+ 0xB6, 0xFF, 0x0A, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x1C, 0x00, 0x85,
+ 0xFF, 0x5B, 0x01, 0xE9, 0xFC, 0x73, 0x06, 0xD8, 0xF1, 0xE5, 0x36,
+ 0x19, 0x29, 0xF8, 0xF1, 0x29, 0x07, 0x37, 0xFC, 0xD8, 0x01, 0x42,
+ 0xFF, 0x33, 0x00, 0xFD, 0xFF, 0x0B, 0x00, 0xD3, 0xFF, 0x47, 0x00,
+ 0xD7, 0xFF, 0x82, 0xFF, 0x53, 0x02, 0x39, 0xF8, 0xFD, 0x44, 0x8D,
+ 0x13, 0xD3, 0xF6, 0x72, 0x05, 0xCA, 0xFC, 0xB5, 0x01, 0x45, 0xFF,
+ 0x31, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x23, 0x00, 0x75, 0xFF, 0x39,
+ 0x01, 0xE0, 0xFD, 0x33, 0x03, 0x87, 0xFB, 0xA2, 0x06, 0xCB, 0x48,
+ 0xEA, 0x00, 0x01, 0xFE, 0xE9, 0x01, 0x8A, 0xFE, 0xE8, 0x00, 0x95,
+ 0xFF, 0x1A, 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x38, 0xFF, 0xDA, 0x01,
+ 0x6A, 0xFC, 0x53, 0x06, 0xBA, 0xF4, 0xCE, 0x1A, 0x32, 0x41, 0x1F,
+ 0xF5, 0x1D, 0x04, 0x71, 0xFE, 0x71, 0x00, 0xFB, 0xFF, 0xF0, 0xFF,
+ 0x05, 0x00, 0xFD, 0xFF, 0x2B, 0x00, 0x5B, 0xFF, 0xAB, 0x01, 0x6F,
+ 0xFC, 0x08, 0x07, 0x7E, 0xF1, 0x21, 0x30, 0x67, 0x30, 0x7D, 0xF1,
+ 0x05, 0x07, 0x73, 0xFC, 0xA8, 0x01, 0x5C, 0xFF, 0x2A, 0x00, 0xFD,
+ 0xFF, 0x05, 0x00, 0xF2, 0xFF, 0xF8, 0xFF, 0x77, 0x00, 0x67, 0xFE,
+ 0x2D, 0x04, 0x04, 0xF5, 0x07, 0x41, 0x1B, 0x1B, 0xA6, 0xF4, 0x5A,
+ 0x06, 0x67, 0xFC, 0xDB, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFE, 0xFF,
+ 0x1A, 0x00, 0x96, 0xFF, 0xE5, 0x00, 0x91, 0xFE, 0xDC, 0x01, 0x1A,
+ 0xFE, 0xB3, 0x00, 0xC3, 0x48, 0xE1, 0x06, 0x6E, 0xFB, 0x40, 0x03,
+ 0xDA, 0xFD, 0x3C, 0x01, 0x74, 0xFF, 0x23, 0x00, 0x00, 0x00, 0xFF,
+ 0xFF, 0x31, 0x00, 0x46, 0xFF, 0xB3, 0x01, 0xCF, 0xFC, 0x67, 0x05,
+ 0xEA, 0xF6, 0x44, 0x13, 0x1E, 0x45, 0x5E, 0xF8, 0x3F, 0x02, 0x8E,
+ 0xFF, 0xD0, 0xFF, 0x4A, 0x00, 0xD2, 0xFF, 0x0B, 0x00, 0xFD, 0xFF,
+ 0x33, 0x00, 0x41, 0xFF, 0xD9, 0x01, 0x36, 0xFC, 0x28, 0x07, 0x01,
+ 0xF2, 0xCE, 0x28, 0x23, 0x37, 0xE0, 0xF1, 0x6B, 0x06, 0xEF, 0xFC,
+ 0x57, 0x01, 0x87, 0xFF, 0x1B, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x0B,
+ 0x00, 0xB4, 0xFF, 0x00, 0x01, 0x7E, 0xFD, 0x9C, 0x05, 0xDC, 0xF2,
+ 0xE4, 0x3B, 0xCD, 0x22, 0xEE, 0xF2, 0xF3, 0x06, 0x35, 0xFC, 0xE6,
+ 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x11, 0x00, 0xB8, 0xFF,
+ 0x8E, 0x00, 0x46, 0xFF, 0x8A, 0x00, 0x86, 0x00, 0xA7, 0xFB, 0x48,
+ 0x47, 0x95, 0x0D, 0xD9, 0xF8, 0x84, 0x04, 0x39, 0xFD, 0x85, 0x01,
+ 0x57, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2A, 0x00, 0x5D,
+ 0xFF, 0x76, 0x01, 0x59, 0xFD, 0x42, 0x04, 0x63, 0xF9, 0x1C, 0x0C,
+ 0xB6, 0x47, 0xA4, 0xFC, 0x07, 0x00, 0xD0, 0x00, 0x20, 0xFF, 0xA0,
+ 0x00, 0xB1, 0xFF, 0x13, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF,
+ 0xE6, 0x01, 0x3B, 0xFC, 0xDA, 0x06, 0x3F, 0xF3, 0x2C, 0x21, 0x11,
+ 0x3D, 0x3A, 0xF3, 0x58, 0x05, 0xAA, 0xFD, 0xE5, 0x00, 0xC1, 0xFF,
+ 0x06, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x1F, 0x00, 0x7D, 0xFF, 0x6B,
+ 0x01, 0xCF, 0xFC, 0x96, 0x06, 0xB7, 0xF1, 0xC6, 0x35, 0x64, 0x2A,
+ 0xD4, 0xF1, 0x2B, 0x07, 0x3D, 0xFC, 0xD2, 0x01, 0x45, 0xFF, 0x32,
+ 0x00, 0xFD, 0xFF, 0x0A, 0x00, 0xD9, 0xFF, 0x39, 0x00, 0xF4, 0xFF,
+ 0x4E, 0xFF, 0xAC, 0x02, 0x98, 0xF7, 0x65, 0x44, 0xD6, 0x14, 0x6C,
+ 0xF6, 0x9F, 0x05, 0xB6, 0xFC, 0xBD, 0x01, 0x42, 0xFF, 0x32, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0x21, 0x00, 0x7A, 0xFF, 0x2B, 0x01, 0xFE,
+ 0xFD, 0xF8, 0x02, 0xFB, 0xFB, 0x8D, 0x05, 0xE5, 0x48, 0xE3, 0x01,
+ 0x91, 0xFD, 0x25, 0x02, 0x6B, 0xFE, 0xF7, 0x00, 0x8F, 0xFF, 0x1C,
+ 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xD5, 0x01, 0x78, 0xFC,
+ 0x2F, 0x06, 0x13, 0xF5, 0x7C, 0x19, 0xF7, 0x41, 0x9B, 0xF5, 0xD1,
+ 0x03, 0x9F, 0xFE, 0x57, 0x00, 0x08, 0x00, 0xEC, 0xFF, 0x06, 0x00,
+ 0xFD, 0xFF, 0x2D, 0x00, 0x55, 0xFF, 0xB5, 0x01, 0x61, 0xFC, 0x16,
+ 0x07, 0x85, 0xF1, 0xE6, 0x2E, 0x9E, 0x31, 0x7D, 0xF1, 0xF3, 0x06,
+ 0x84, 0xFC, 0x9D, 0x01, 0x63, 0xFF, 0x28, 0x00, 0xFD, 0xFF, 0x04,
+ 0x00, 0xF6, 0xFF, 0xEB, 0xFF, 0x91, 0x00, 0x3B, 0xFE, 0x75, 0x04,
+ 0x92, 0xF4, 0x36, 0x40, 0x6E, 0x1C, 0x50, 0xF4, 0x7B, 0x06, 0x5B,
+ 0xFC, 0xDF, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x18, 0x00,
+ 0x9C, 0xFF, 0xD6, 0x00, 0xB1, 0xFE, 0xA1, 0x01, 0x89, 0xFE, 0xC3,
+ 0xFF, 0x9C, 0x48, 0xFD, 0x07, 0xFA, 0xFA, 0x7A, 0x03, 0xBC, 0xFD,
+ 0x49, 0x01, 0x6E, 0xFF, 0x24, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x30,
+ 0x00, 0x49, 0xFF, 0xAA, 0x01, 0xE4, 0xFC, 0x38, 0x05, 0x54, 0xF7,
+ 0xFE, 0x11, 0xAA, 0x45, 0x09, 0xF9, 0xE2, 0x01, 0xC4, 0xFF, 0xB3,
+ 0xFF, 0x59, 0x00, 0xCD, 0xFF, 0x0D, 0x00, 0xFD, 0xFF, 0x34, 0x00,
+ 0x3E, 0xFF, 0xDE, 0x01, 0x33, 0xFC, 0x22, 0x07, 0x2B, 0xF2, 0x80,
+ 0x27, 0x3B, 0x38, 0x0A, 0xF2, 0x44, 0x06, 0x0B, 0xFD, 0x45, 0x01,
+ 0x90, 0xFF, 0x18, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x0F, 0x00, 0xA9,
+ 0xFF, 0x15, 0x01, 0x5B, 0xFD, 0xD0, 0x05, 0x97, 0xF2, 0xE6, 0x3A,
+ 0x21, 0x24, 0xB1, 0xF2, 0x04, 0x07, 0x33, 0xFC, 0xE5, 0x01, 0x39,
+ 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x10, 0x00, 0xBE, 0xFF, 0x7F, 0x00,
+ 0x65, 0xFF, 0x51, 0x00, 0xEB, 0x00, 0xE1, 0xFA, 0xE1, 0x46, 0xCD,
+ 0x0E, 0x6A, 0xF8, 0xB8, 0x04, 0x20, 0xFD, 0x90, 0x01, 0x53, 0xFF,
+ 0x2D, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x28, 0x00, 0x62, 0xFF, 0x6A,
+ 0x01, 0x74, 0xFD, 0x0A, 0x04, 0xD5, 0xF9, 0xED, 0x0A, 0x03, 0x48,
+ 0x7C, 0xFD, 0x9E, 0xFF, 0x0A, 0x01, 0x01, 0xFF, 0xAF, 0x00, 0xAB,
+ 0xFF, 0x14, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE5, 0x01,
+ 0x42, 0xFC, 0xC3, 0x06, 0x87, 0xF3, 0xD7, 0x1F, 0xFE, 0x3D, 0x91,
+ 0xF3, 0x1D, 0x05, 0xD1, 0xFD, 0xCE, 0x00, 0xCC, 0xFF, 0x02, 0x00,
+ 0x02, 0x00, 0xFE, 0xFF, 0x22, 0x00, 0x75, 0xFF, 0x7A, 0x01, 0xB8,
+ 0xFC, 0xB4, 0x06, 0x9E, 0xF1, 0xA2, 0x34, 0xAD, 0x2B, 0xB6, 0xF1,
+ 0x29, 0x07, 0x45, 0xFC, 0xCB, 0x01, 0x49, 0xFF, 0x31, 0x00, 0xFD,
+ 0xFF, 0x09, 0x00, 0xDE, 0xFF, 0x2B, 0x00, 0x11, 0x00, 0x1B, 0xFF,
+ 0x02, 0x03, 0xFE, 0xF6, 0xC3, 0x43, 0x22, 0x16, 0x07, 0xF6, 0xCA,
+ 0x05, 0xA3, 0xFC, 0xC5, 0x01, 0x3F, 0xFF, 0x33, 0x00, 0xFF, 0xFF,
+ 0x00, 0x00, 0x20, 0x00, 0x80, 0xFF, 0x1C, 0x01, 0x1C, 0xFE, 0xBD,
+ 0x02, 0x6E, 0xFC, 0x7D, 0x04, 0xF3, 0x48, 0xE2, 0x02, 0x1F, 0xFD,
+ 0x60, 0x02, 0x4C, 0xFE, 0x06, 0x01, 0x89, 0xFF, 0x1D, 0x00, 0xFE,
+ 0xFF, 0x34, 0x00, 0x3C, 0xFF, 0xCF, 0x01, 0x88, 0xFC, 0x09, 0x06,
+ 0x71, 0xF5, 0x2B, 0x18, 0xB2, 0x42, 0x20, 0xF6, 0x83, 0x03, 0xCF,
+ 0xFE, 0x3C, 0x00, 0x15, 0x00, 0xE6, 0xFF, 0x07, 0x00, 0xFD, 0xFF,
+ 0x2E, 0x00, 0x50, 0xFF, 0xBF, 0x01, 0x54, 0xFC, 0x20, 0x07, 0x94,
+ 0xF1, 0xA6, 0x2D, 0xD0, 0x32, 0x85, 0xF1, 0xDD, 0x06, 0x96, 0xFC,
+ 0x90, 0x01, 0x69, 0xFF, 0x26, 0x00, 0xFD, 0xFF, 0x03, 0x00, 0xFB,
+ 0xFF, 0xDF, 0xFF, 0xA9, 0x00, 0x10, 0xFE, 0xB9, 0x04, 0x27, 0xF4,
+ 0x5E, 0x3F, 0xC3, 0x1D, 0xFE, 0xF3, 0x99, 0x06, 0x50, 0xFC, 0xE2,
+ 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x17, 0x00, 0xA2, 0xFF,
+ 0xC7, 0x00, 0xD0, 0xFE, 0x65, 0x01, 0xF6, 0xFE, 0xD9, 0xFE, 0x6A,
+ 0x48, 0x1F, 0x09, 0x87, 0xFA, 0xB3, 0x03, 0xA0, 0xFD, 0x56, 0x01,
+ 0x69, 0xFF, 0x26, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2F, 0x00, 0x4D,
+ 0xFF, 0xA0, 0x01, 0xFB, 0xFC, 0x07, 0x05, 0xBF, 0xF7, 0xBB, 0x10,
+ 0x2B, 0x46, 0xBB, 0xF9, 0x83, 0x01, 0xFA, 0xFF, 0x95, 0xFF, 0x68,
+ 0x00, 0xC7, 0xFF, 0x0E, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3C, 0xFF,
+ 0xE1, 0x01, 0x31, 0xFC, 0x19, 0x07, 0x5B, 0xF2, 0x30, 0x26, 0x4B,
+ 0x39, 0x3B, 0xF2, 0x1A, 0x06, 0x29, 0xFD, 0x33, 0x01, 0x99, 0xFF,
+ 0x15, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x13, 0x00, 0x9F, 0xFF, 0x28,
+ 0x01, 0x3A, 0xFD, 0x00, 0x06, 0x5A, 0xF2, 0xDF, 0x39, 0x73, 0x25,
+ 0x79, 0xF2, 0x12, 0x07, 0x31, 0xFC, 0xE3, 0x01, 0x3B, 0xFF, 0x35,
+ 0x00, 0xFD, 0xFF, 0x0F, 0x00, 0xC4, 0xFF, 0x70, 0x00, 0x84, 0xFF,
+ 0x19, 0x00, 0x4D, 0x01, 0x22, 0xFA, 0x70, 0x46, 0x0A, 0x10, 0xFC,
+ 0xF7, 0xEB, 0x04, 0x08, 0xFD, 0x9A, 0x01, 0x4F, 0xFF, 0x2E, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0x27, 0x00, 0x66, 0xFF, 0x5E, 0x01, 0x90,
+ 0xFD, 0xD2, 0x03, 0x47, 0xFA, 0xC3, 0x09, 0x48, 0x48, 0x5A, 0xFE,
+ 0x33, 0xFF, 0x45, 0x01, 0xE2, 0xFE, 0xBE, 0x00, 0xA5, 0xFF, 0x16,
+ 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE3, 0x01, 0x4B, 0xFC,
+ 0xA9, 0x06, 0xD2, 0xF3, 0x81, 0x1E, 0xE4, 0x3E, 0xEF, 0xF3, 0xDE,
+ 0x04, 0xF9, 0xFD, 0xB7, 0x00, 0xD8, 0xFF, 0xFD, 0xFF, 0x03, 0x00,
+ 0xFD, 0xFF, 0x24, 0x00, 0x6D, 0xFF, 0x88, 0x01, 0xA2, 0xFC, 0xD0,
+ 0x06, 0x8C, 0xF1, 0x78, 0x33, 0xF2, 0x2C, 0x9E, 0xF1, 0x24, 0x07,
+ 0x4E, 0xFC, 0xC3, 0x01, 0x4E, 0xFF, 0x2F, 0x00, 0xFD, 0xFF, 0x08,
+ 0x00, 0xE4, 0xFF, 0x1D, 0x00, 0x2D, 0x00, 0xEA, 0xFE, 0x56, 0x03,
+ 0x6D, 0xF6, 0x17, 0x43, 0x70, 0x17, 0xA6, 0xF5, 0xF3, 0x05, 0x91,
+ 0xFC, 0xCC, 0x01, 0x3D, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0x1E, 0x00,
+ 0x86, 0xFF, 0x0E, 0x01, 0x3B, 0xFE, 0x82, 0x02, 0xE0, 0xFC, 0x73,
+ 0x03, 0xF6, 0x48, 0xE9, 0x03, 0xAD, 0xFC, 0x9C, 0x02, 0x2D, 0xFE,
+ 0x14, 0x01, 0x83, 0xFF, 0x1F, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x33,
+ 0x00, 0x3E, 0xFF, 0xC9, 0x01, 0x99, 0xFC, 0xE1, 0x05, 0xD1, 0xF5,
+ 0xDC, 0x16, 0x65, 0x43, 0xAD, 0xF6, 0x31, 0x03, 0x00, 0xFF, 0x20,
+ 0x00, 0x23, 0x00, 0xE1, 0xFF, 0x08, 0x00, 0xFD, 0xFF, 0x30, 0x00,
+ 0x4C, 0xFF, 0xC7, 0x01, 0x4A, 0xFC, 0x27, 0x07, 0xA8, 0xF1, 0x62,
+ 0x2C, 0xFD, 0x33, 0x93, 0xF1, 0xC4, 0x06, 0xAB, 0xFC, 0x82, 0x01,
+ 0x71, 0xFF, 0x23, 0x00, 0xFE, 0xFF, 0x02, 0x00, 0xFF, 0xFF, 0xD3,
+ 0xFF, 0xC1, 0x00, 0xE7, 0xFD, 0xFA, 0x04, 0xC4, 0xF3, 0x7E, 0x3E,
+ 0x19, 0x1F, 0xB0, 0xF3, 0xB5, 0x06, 0x47, 0xFC, 0xE4, 0x01, 0x36,
+ 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x15, 0x00, 0xA8, 0xFF, 0xB8, 0x00,
+ 0xF0, 0xFE, 0x2B, 0x01, 0x63, 0xFF, 0xF6, 0xFD, 0x2C, 0x48, 0x47,
+ 0x0A, 0x14, 0xFA, 0xEB, 0x03, 0x84, 0xFD, 0x63, 0x01, 0x64, 0xFF,
+ 0x27, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2D, 0x00, 0x51, 0xFF, 0x96,
+ 0x01, 0x13, 0xFD, 0xD5, 0x04, 0x2C, 0xF8, 0x7D, 0x0F, 0xA3, 0x46,
+ 0x76, 0xFA, 0x22, 0x01, 0x32, 0x00, 0x76, 0xFF, 0x76, 0x00, 0xC1,
+ 0xFF, 0x0F, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x3A, 0xFF, 0xE4, 0x01,
+ 0x32, 0xFC, 0x0C, 0x07, 0x91, 0xF2, 0xDD, 0x24, 0x54, 0x3A, 0x74,
+ 0xF2, 0xEB, 0x05, 0x49, 0xFD, 0x20, 0x01, 0xA3, 0xFF, 0x11, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x16, 0x00, 0x95, 0xFF, 0x3B, 0x01, 0x1B,
+ 0xFD, 0x2D, 0x06, 0x24, 0xF2, 0xD3, 0x38, 0xC6, 0x26, 0x45, 0xF2,
+ 0x1D, 0x07, 0x32, 0xFC, 0xE0, 0x01, 0x3D, 0xFF, 0x35, 0x00, 0xFD,
+ 0xFF, 0x0D, 0x00, 0xC9, 0xFF, 0x61, 0x00, 0xA2, 0xFF, 0xE2, 0xFF,
+ 0xAE, 0x01, 0x6B, 0xF9, 0xF2, 0x45, 0x4A, 0x11, 0x8F, 0xF7, 0x1D,
+ 0x05, 0xF1, 0xFC, 0xA4, 0x01, 0x4B, 0xFF, 0x2F, 0x00, 0xFF, 0xFF,
+ 0x00, 0x00, 0x25, 0x00, 0x6C, 0xFF, 0x51, 0x01, 0xAC, 0xFD, 0x9A,
+ 0x03, 0xBA, 0xFA, 0x9E, 0x08, 0x81, 0x48, 0x40, 0xFF, 0xC6, 0xFE,
+ 0x80, 0x01, 0xC2, 0xFE, 0xCE, 0x00, 0x9F, 0xFF, 0x17, 0x00, 0xFE,
+ 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE1, 0x01, 0x55, 0xFC, 0x8C, 0x06,
+ 0x22, 0xF4, 0x2C, 0x1D, 0xC0, 0x3F, 0x55, 0xF4, 0x9B, 0x04, 0x23,
+ 0xFE, 0x9F, 0x00, 0xE4, 0xFF, 0xF9, 0xFF, 0x04, 0x00, 0xFD, 0xFF,
+ 0x27, 0x00, 0x66, 0xFF, 0x96, 0x01, 0x8E, 0xFC, 0xE7, 0x06, 0x81,
+ 0xF1, 0x48, 0x32, 0x34, 0x2E, 0x8D, 0xF1, 0x1C, 0x07, 0x5A, 0xFC,
+ 0xBB, 0x01, 0x53, 0xFF, 0x2E, 0x00, 0xFD, 0xFF, 0x07, 0x00, 0xE9,
+ 0xFF, 0x0F, 0x00, 0x48, 0x00, 0xB9, 0xFE, 0xA6, 0x03, 0xE4, 0xF5,
+ 0x60, 0x42, 0xC1, 0x18, 0x47, 0xF5, 0x1A, 0x06, 0x81, 0xFC, 0xD2,
+ 0x01, 0x3B, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1C, 0x00, 0x8B, 0xFF,
+ 0xFF, 0x00, 0x5A, 0xFE, 0x46, 0x02, 0x52, 0xFD, 0x70, 0x02, 0xED,
+ 0x48, 0xF5, 0x04, 0x3B, 0xFC, 0xD7, 0x02, 0x0F, 0xFE, 0x23, 0x01,
+ 0x7E, 0xFF, 0x20, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x40,
+ 0xFF, 0xC1, 0x01, 0xAB, 0xFC, 0xB7, 0x05, 0x34, 0xF6, 0x8E, 0x15,
+ 0x0B, 0x44, 0x42, 0xF7, 0xDC, 0x02, 0x32, 0xFF, 0x04, 0x00, 0x31,
+ 0x00, 0xDC, 0xFF, 0x09, 0x00, 0xFD, 0xFF, 0x31, 0x00, 0x47, 0xFF,
+ 0xCE, 0x01, 0x41, 0xFC, 0x2A, 0x07, 0xC2, 0xF1, 0x1B, 0x2B, 0x25,
+ 0x35, 0xA8, 0xF1, 0xA7, 0x06, 0xC2, 0xFC, 0x74, 0x01, 0x78, 0xFF,
+ 0x20, 0x00, 0xFE, 0xFF, 0x02, 0x00, 0x04, 0x00, 0xC7, 0xFF, 0xD9,
+ 0x00, 0xBF, 0xFD, 0x38, 0x05, 0x69, 0xF3, 0x96, 0x3D, 0x6F, 0x20,
+ 0x66, 0xF3, 0xCE, 0x06, 0x3F, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0x14, 0x00, 0xAE, 0xFF, 0xA9, 0x00, 0x0F, 0xFF,
+ 0xF0, 0x00, 0xCD, 0xFF, 0x1B, 0xFD, 0xE4, 0x47, 0x73, 0x0B, 0xA2,
+ 0xF9, 0x23, 0x04, 0x68, 0xFD, 0x70, 0x01, 0x5F, 0xFF, 0x29, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x2C, 0x00, 0x55, 0xFF, 0x8B, 0x01, 0x2B,
+ 0xFD, 0xA1, 0x04, 0x9B, 0xF8, 0x42, 0x0E, 0x0F, 0x47, 0x38, 0xFB,
+ 0xBE, 0x00, 0x6A, 0x00, 0x58, 0xFF, 0x85, 0x00, 0xBB, 0xFF, 0x10,
+ 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xE6, 0x01, 0x34, 0xFC,
+ 0xFD, 0x06, 0xCB, 0xF2, 0x8A, 0x23, 0x58, 0x3B, 0xB4, 0xF2, 0xBA,
+ 0x05, 0x6A, 0xFD, 0x0B, 0x01, 0xAE, 0xFF, 0x0D, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x19, 0x00, 0x8C, 0xFF, 0x4D, 0x01, 0xFE, 0xFC, 0x56,
+ 0x06, 0xF7, 0xF1, 0xBF, 0x37, 0x15, 0x28, 0x18, 0xF2, 0x25, 0x07,
+ 0x34, 0xFC, 0xDC, 0x01, 0x3F, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x0C,
+ 0x00, 0xCF, 0xFF, 0x52, 0x00, 0xC0, 0xFF, 0xAC, 0xFF, 0x0C, 0x02,
+ 0xBC, 0xF8, 0x6D, 0x45, 0x8E, 0x12, 0x24, 0xF7, 0x4D, 0x05, 0xDB,
+ 0xFC, 0xAE, 0x01, 0x48, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0x24, 0x00, 0x71, 0xFF, 0x43, 0x01, 0xC9, 0xFD, 0x60, 0x03, 0x2E,
+ 0xFB, 0x7E, 0x07, 0xAF, 0x48, 0x2D, 0x00, 0x58, 0xFE, 0xBB, 0x01,
+ 0xA3, 0xFE, 0xDD, 0x00, 0x99, 0xFF, 0x19, 0x00, 0xFE, 0xFF, 0x36,
+ 0x00, 0x37, 0xFF, 0xDD, 0x01, 0x60, 0xFC, 0x6D, 0x06, 0x76, 0xF4,
+ 0xD8, 0x1B, 0x95, 0x40, 0xC3, 0xF4, 0x56, 0x04, 0x4E, 0xFE, 0x85,
+ 0x00, 0xF1, 0xFF, 0xF4, 0xFF, 0x04, 0x00, 0xFD, 0xFF, 0x29, 0x00,
+ 0x60, 0xFF, 0xA2, 0x01, 0x7C, 0xFC, 0xFB, 0x06, 0x7C, 0xF1, 0x15,
+ 0x31, 0x73, 0x2F, 0x81, 0xF1, 0x10, 0x07, 0x67, 0xFC, 0xB1, 0x01,
+ 0x58, 0xFF, 0x2C, 0x00, 0xFD, 0xFF, 0x06, 0x00, 0xEE, 0xFF, 0x02,
+ 0x00, 0x63, 0x00, 0x8A, 0xFE, 0xF3, 0x03, 0x63, 0xF5, 0xA1, 0x41,
+ 0x12, 0x1A, 0xEB, 0xF4, 0x3F, 0x06, 0x72, 0xFC, 0xD7, 0x01, 0x39,
+ 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1B, 0x00, 0x91, 0xFF, 0xF1, 0x00,
+ 0x79, 0xFE, 0x0A, 0x02, 0xC3, 0xFD, 0x73, 0x01, 0xDB, 0x48, 0x07,
+ 0x06, 0xC7, 0xFB, 0x12, 0x03, 0xF1, 0xFD, 0x31, 0x01, 0x78, 0xFF,
+ 0x22, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x32, 0x00, 0x43, 0xFF, 0xBA,
+ 0x01, 0xBF, 0xFC, 0x8B, 0x05, 0x99, 0xF6, 0x43, 0x14, 0xA9, 0x44,
+ 0xDE, 0xF7, 0x85, 0x02, 0x65, 0xFF, 0xE7, 0xFF, 0x3F, 0x00, 0xD6,
+ 0xFF, 0x0A, 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x44, 0xFF, 0xD5, 0x01,
+ 0x3A, 0xFC, 0x2A, 0x07, 0xE3, 0xF1, 0xD1, 0x29, 0x46, 0x36, 0xC5,
+ 0xF1, 0x87, 0x06, 0xDA, 0xFC, 0x64, 0x01, 0x80, 0xFF, 0x1E, 0x00,
+ 0xFE, 0xFF, 0x01, 0x00, 0x08, 0x00, 0xBC, 0xFF, 0xEF, 0x00, 0x9A,
+ 0xFD, 0x72, 0x05, 0x16, 0xF3, 0xA5, 0x3C, 0xC4, 0x21, 0x21, 0xF3,
+ 0xE4, 0x06, 0x39, 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0x12, 0x00, 0xB3, 0xFF, 0x99, 0x00, 0x2E, 0xFF, 0xB6, 0x00,
+ 0x36, 0x00, 0x47, 0xFC, 0x90, 0x47, 0xA4, 0x0C, 0x31, 0xF9, 0x5A,
+ 0x04, 0x4E, 0xFD, 0x7C, 0x01, 0x5B, 0xFF, 0x2A, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x2B, 0x00, 0x59, 0xFF, 0x80, 0x01, 0x45, 0xFD, 0x6C,
+ 0x04, 0x0B, 0xF9, 0x0B, 0x0D, 0x73, 0x47, 0x02, 0xFC, 0x58, 0x00,
+ 0xA3, 0x00, 0x39, 0xFF, 0x94, 0x00, 0xB5, 0xFF, 0x12, 0x00, 0xFD,
+ 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x37, 0xFC, 0xEB, 0x06,
+ 0x0B, 0xF3, 0x35, 0x22, 0x52, 0x3C, 0xFD, 0xF2, 0x84, 0x05, 0x8D,
+ 0xFD, 0xF6, 0x00, 0xB8, 0xFF, 0x09, 0x00, 0x01, 0x00, 0xFE, 0xFF,
+ 0x1D, 0x00, 0x83, 0xFF, 0x5E, 0x01, 0xE3, 0xFC, 0x7B, 0x06, 0xD0,
+ 0xF1, 0xA5, 0x36, 0x62, 0x29, 0xEF, 0xF1, 0x29, 0x07, 0x39, 0xFC,
+ 0xD7, 0x01, 0x42, 0xFF, 0x33, 0x00, 0xFD, 0xFF, 0x0B, 0x00, 0xD5,
+ 0xFF, 0x44, 0x00, 0xDD, 0xFF, 0x77, 0xFF, 0x67, 0x02, 0x14, 0xF8,
+ 0xDC, 0x44, 0xD5, 0x13, 0xBC, 0xF6, 0x7C, 0x05, 0xC5, 0xFC, 0xB7,
+ 0x01, 0x44, 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x22, 0x00,
+ 0x76, 0xFF, 0x35, 0x01, 0xE7, 0xFD, 0x26, 0x03, 0xA1, 0xFB, 0x64,
+ 0x06, 0xD2, 0x48, 0x21, 0x01, 0xE8, 0xFD, 0xF7, 0x01, 0x83, 0xFE,
+ 0xEC, 0x00, 0x93, 0xFF, 0x1A, 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x39,
+ 0xFF, 0xD9, 0x01, 0x6D, 0xFC, 0x4B, 0x06, 0xCD, 0xF4, 0x83, 0x1A,
+ 0x5F, 0x41, 0x3A, 0xF5, 0x0C, 0x04, 0x7B, 0xFE, 0x6C, 0x00, 0xFE,
+ 0xFF, 0xEF, 0xFF, 0x05, 0x00, 0xFD, 0xFF, 0x2B, 0x00, 0x5A, 0xFF,
+ 0xAD, 0x01, 0x6C, 0xFC, 0x0C, 0x07, 0x7F, 0xF1, 0xDC, 0x2F, 0xAD,
+ 0x30, 0x7D, 0xF1, 0x01, 0x07, 0x76, 0xFC, 0xA6, 0x01, 0x5E, 0xFF,
+ 0x2A, 0x00, 0xFD, 0xFF, 0x05, 0x00, 0xF3, 0xFF, 0xF5, 0xFF, 0x7D,
+ 0x00, 0x5D, 0xFE, 0x3E, 0x04, 0xEA, 0xF4, 0xD9, 0x40, 0x66, 0x1B,
+ 0x93, 0xF4, 0x62, 0x06, 0x64, 0xFC, 0xDC, 0x01, 0x38, 0xFF, 0x36,
+ 0x00, 0xFE, 0xFF, 0x19, 0x00, 0x97, 0xFF, 0xE2, 0x00, 0x98, 0xFE,
+ 0xCF, 0x01, 0x33, 0xFE, 0x7D, 0x00, 0xBB, 0x48, 0x1F, 0x07, 0x54,
+ 0xFB, 0x4C, 0x03, 0xD3, 0xFD, 0x3F, 0x01, 0x73, 0xFF, 0x23, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x31, 0x00, 0x46, 0xFF, 0xB1, 0x01, 0xD3,
+ 0xFC, 0x5D, 0x05, 0x01, 0xF7, 0xFB, 0x12, 0x3F, 0x45, 0x83, 0xF8,
+ 0x2A, 0x02, 0x9A, 0xFF, 0xCA, 0xFF, 0x4E, 0x00, 0xD1, 0xFF, 0x0C,
+ 0x00, 0xFD, 0xFF, 0x34, 0x00, 0x40, 0xFF, 0xDA, 0x01, 0x35, 0xFC,
+ 0x27, 0x07, 0x09, 0xF2, 0x85, 0x28, 0x63, 0x37, 0xE9, 0xF1, 0x63,
+ 0x06, 0xF5, 0xFC, 0x53, 0x01, 0x89, 0xFF, 0x1A, 0x00, 0xFE, 0xFF,
+ 0x00, 0x00, 0x0C, 0x00, 0xB1, 0xFF, 0x04, 0x01, 0x76, 0xFD, 0xA8,
+ 0x05, 0xCC, 0xF2, 0xAB, 0x3B, 0x18, 0x23, 0xE0, 0xF2, 0xF7, 0x06,
+ 0x35, 0xFC, 0xE6, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x11,
+ 0x00, 0xB9, 0xFF, 0x8A, 0x00, 0x4D, 0xFF, 0x7D, 0x00, 0x9C, 0x00,
+ 0x7B, 0xFB, 0x31, 0x47, 0xD9, 0x0D, 0xC0, 0xF8, 0x8F, 0x04, 0x34,
+ 0xFD, 0x87, 0x01, 0x56, 0xFF, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x29, 0x00, 0x5E, 0xFF, 0x74, 0x01, 0x5F, 0xFD, 0x35, 0x04, 0x7C,
+ 0xF9, 0xD8, 0x0B, 0xC9, 0x47, 0xD4, 0xFC, 0xF0, 0xFF, 0xDD, 0x00,
+ 0x19, 0xFF, 0xA4, 0x00, 0xAF, 0xFF, 0x13, 0x00, 0xFD, 0xFF, 0x36,
+ 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3D, 0xFC, 0xD5, 0x06, 0x4F, 0xF3,
+ 0xE0, 0x20, 0x45, 0x3D, 0x4D, 0xF3, 0x4B, 0x05, 0xB3, 0xFD, 0xE0,
+ 0x00, 0xC3, 0xFF, 0x05, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x20, 0x00,
+ 0x7B, 0xFF, 0x6E, 0x01, 0xCA, 0xFC, 0x9D, 0x06, 0xB1, 0xF1, 0x86,
+ 0x35, 0xAE, 0x2A, 0xCD, 0xF1, 0x2B, 0x07, 0x3F, 0xFC, 0xD1, 0x01,
+ 0x46, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x0A, 0x00, 0xDA, 0xFF, 0x36,
+ 0x00, 0xFA, 0xFF, 0x43, 0xFF, 0xBF, 0x02, 0x75, 0xF7, 0x42, 0x44,
+ 0x20, 0x15, 0x55, 0xF6, 0xA9, 0x05, 0xB2, 0xFC, 0xBF, 0x01, 0x41,
+ 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x21, 0x00, 0x7C, 0xFF,
+ 0x27, 0x01, 0x05, 0xFE, 0xEB, 0x02, 0x14, 0xFC, 0x50, 0x05, 0xEA,
+ 0x48, 0x1B, 0x02, 0x78, 0xFD, 0x32, 0x02, 0x64, 0xFE, 0xFA, 0x00,
+ 0x8D, 0xFF, 0x1C, 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xD4,
+ 0x01, 0x7C, 0xFC, 0x27, 0x06, 0x28, 0xF5, 0x31, 0x19, 0x21, 0x42,
+ 0xB8, 0xF5, 0xC0, 0x03, 0xAA, 0xFE, 0x51, 0x00, 0x0B, 0x00, 0xEA,
+ 0xFF, 0x06, 0x00, 0xFD, 0xFF, 0x2D, 0x00, 0x54, 0xFF, 0xB7, 0x01,
+ 0x5E, 0xFC, 0x19, 0x07, 0x88, 0xF1, 0x9F, 0x2E, 0xE3, 0x31, 0x7E,
+ 0xF1, 0xEE, 0x06, 0x88, 0xFC, 0x9A, 0x01, 0x64, 0xFF, 0x28, 0x00,
+ 0xFD, 0xFF, 0x04, 0x00, 0xF7, 0xFF, 0xE8, 0xFF, 0x96, 0x00, 0x31,
+ 0xFE, 0x84, 0x04, 0x79, 0xF4, 0x07, 0x40, 0xBA, 0x1C, 0x3E, 0xF4,
+ 0x82, 0x06, 0x58, 0xFC, 0xE0, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE,
+ 0xFF, 0x18, 0x00, 0x9D, 0xFF, 0xD3, 0x00, 0xB8, 0xFE, 0x93, 0x01,
+ 0xA1, 0xFE, 0x8E, 0xFF, 0x92, 0x48, 0x3D, 0x08, 0xE1, 0xFA, 0x86,
+ 0x03, 0xB6, 0xFD, 0x4C, 0x01, 0x6D, 0xFF, 0x25, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x30, 0x00, 0x4A, 0xFF, 0xA8, 0x01, 0xE9, 0xFC, 0x2D,
+ 0x05, 0x6B, 0xF7, 0xB6, 0x11, 0xC8, 0x45, 0x30, 0xF9, 0xCD, 0x01,
+ 0xD0, 0xFF, 0xAC, 0xFF, 0x5C, 0x00, 0xCB, 0xFF, 0x0D, 0x00, 0xFD,
+ 0xFF, 0x34, 0x00, 0x3E, 0xFF, 0xDF, 0x01, 0x33, 0xFC, 0x20, 0x07,
+ 0x35, 0xF2, 0x36, 0x27, 0x78, 0x38, 0x14, 0xF2, 0x3B, 0x06, 0x11,
+ 0xFD, 0x41, 0x01, 0x92, 0xFF, 0x17, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0x10, 0x00, 0xA7, 0xFF, 0x19, 0x01, 0x53, 0xFD, 0xDB, 0x05, 0x88,
+ 0xF2, 0xAD, 0x3A, 0x6D, 0x24, 0xA4, 0xF2, 0x08, 0x07, 0x32, 0xFC,
+ 0xE5, 0x01, 0x39, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x10, 0x00, 0xBF,
+ 0xFF, 0x7B, 0x00, 0x6C, 0xFF, 0x44, 0x00, 0x01, 0x01, 0xB6, 0xFA,
+ 0xC8, 0x46, 0x13, 0x0F, 0x51, 0xF8, 0xC4, 0x04, 0x1B, 0xFD, 0x92,
+ 0x01, 0x52, 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x28, 0x00,
+ 0x63, 0xFF, 0x67, 0x01, 0x7A, 0xFD, 0xFE, 0x03, 0xEE, 0xF9, 0xAA,
+ 0x0A, 0x16, 0x48, 0xAC, 0xFD, 0x86, 0xFF, 0x17, 0x01, 0xFA, 0xFE,
+ 0xB3, 0x00, 0xAA, 0xFF, 0x15, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36,
+ 0xFF, 0xE5, 0x01, 0x44, 0xFC, 0xBD, 0x06, 0x97, 0xF3, 0x8A, 0x1F,
+ 0x31, 0x3E, 0xA5, 0xF3, 0x0F, 0x05, 0xDA, 0xFD, 0xC9, 0x00, 0xCF,
+ 0xFF, 0x01, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x22, 0x00, 0x73, 0xFF,
+ 0x7D, 0x01, 0xB3, 0xFC, 0xBB, 0x06, 0x9A, 0xF1, 0x60, 0x34, 0xF5,
+ 0x2B, 0xB0, 0xF1, 0x28, 0x07, 0x47, 0xFC, 0xCA, 0x01, 0x4A, 0xFF,
+ 0x30, 0x00, 0xFD, 0xFF, 0x09, 0x00, 0xDF, 0xFF, 0x28, 0x00, 0x17,
+ 0x00, 0x10, 0xFF, 0x15, 0x03, 0xDD, 0xF6, 0x9E, 0x43, 0x6C, 0x16,
+ 0xF1, 0xF5, 0xD3, 0x05, 0x9F, 0xFC, 0xC6, 0x01, 0x3F, 0xFF, 0x33,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x1F, 0x00, 0x81, 0xFF, 0x19, 0x01,
+ 0x23, 0xFE, 0xB0, 0x02, 0x87, 0xFC, 0x41, 0x04, 0xF4, 0x48, 0x1C,
+ 0x03, 0x06, 0xFD, 0x6E, 0x02, 0x45, 0xFE, 0x09, 0x01, 0x88, 0xFF,
+ 0x1D, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3C, 0xFF, 0xCE, 0x01, 0x8C,
+ 0xFC, 0x00, 0x06, 0x86, 0xF5, 0xE0, 0x17, 0xDB, 0x42, 0x3F, 0xF6,
+ 0x71, 0x03, 0xD9, 0xFE, 0x36, 0x00, 0x18, 0x00, 0xE5, 0xFF, 0x07,
+ 0x00, 0xFD, 0xFF, 0x2F, 0x00, 0x4F, 0xFF, 0xC1, 0x01, 0x52, 0xFC,
+ 0x22, 0x07, 0x98, 0xF1, 0x5E, 0x2D, 0x13, 0x33, 0x87, 0xF1, 0xD8,
+ 0x06, 0x9B, 0xFC, 0x8D, 0x01, 0x6B, 0xFF, 0x25, 0x00, 0xFD, 0xFF,
+ 0x03, 0x00, 0xFC, 0xFF, 0xDC, 0xFF, 0xAF, 0x00, 0x07, 0xFE, 0xC8,
+ 0x04, 0x10, 0xF4, 0x2D, 0x3F, 0x0F, 0x1E, 0xED, 0xF3, 0xA0, 0x06,
+ 0x4E, 0xFC, 0xE3, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x16,
+ 0x00, 0xA3, 0xFF, 0xC3, 0x00, 0xD7, 0xFE, 0x58, 0x01, 0x0F, 0xFF,
+ 0xA6, 0xFE, 0x5D, 0x48, 0x61, 0x09, 0x6E, 0xFA, 0xC0, 0x03, 0x99,
+ 0xFD, 0x59, 0x01, 0x68, 0xFF, 0x26, 0x00, 0x00, 0x00, 0xFF, 0xFF,
+ 0x2E, 0x00, 0x4E, 0xFF, 0x9E, 0x01, 0x00, 0xFD, 0xFC, 0x04, 0xD7,
+ 0xF7, 0x75, 0x10, 0x48, 0x46, 0xE4, 0xF9, 0x6E, 0x01, 0x06, 0x00,
+ 0x8E, 0xFF, 0x6B, 0x00, 0xC6, 0xFF, 0x0E, 0x00, 0xFD, 0xFF, 0x35,
+ 0x00, 0x3B, 0xFF, 0xE2, 0x01, 0x31, 0xFC, 0x16, 0x07, 0x67, 0xF2,
+ 0xE5, 0x25, 0x87, 0x39, 0x47, 0xF2, 0x10, 0x06, 0x30, 0xFD, 0x2F,
+ 0x01, 0x9C, 0xFF, 0x14, 0x00, 0xFF, 0xFF, 0xFF, 0xFF, 0x13, 0x00,
+ 0x9D, 0xFF, 0x2D, 0x01, 0x33, 0xFD, 0x0B, 0x06, 0x4D, 0xF2, 0xA5,
+ 0x39, 0xBF, 0x25, 0x6D, 0xF2, 0x15, 0x07, 0x31, 0xFC, 0xE2, 0x01,
+ 0x3B, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x0E, 0x00, 0xC5, 0xFF, 0x6D,
+ 0x00, 0x8B, 0xFF, 0x0D, 0x00, 0x63, 0x01, 0xF9, 0xF9, 0x55, 0x46,
+ 0x51, 0x10, 0xE3, 0xF7, 0xF7, 0x04, 0x03, 0xFD, 0x9D, 0x01, 0x4E,
+ 0xFF, 0x2E, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x26, 0x00, 0x68, 0xFF,
+ 0x5B, 0x01, 0x96, 0xFD, 0xC6, 0x03, 0x61, 0xFA, 0x81, 0x09, 0x57,
+ 0x48, 0x8D, 0xFE, 0x1B, 0xFF, 0x52, 0x01, 0xDB, 0xFE, 0xC2, 0x00,
+ 0xA4, 0xFF, 0x16, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE3,
+ 0x01, 0x4D, 0xFC, 0xA3, 0x06, 0xE4, 0xF3, 0x36, 0x1E, 0x16, 0x3F,
+ 0x05, 0xF4, 0xCF, 0x04, 0x02, 0xFE, 0xB2, 0x00, 0xDB, 0xFF, 0xFC,
+ 0xFF, 0x03, 0x00, 0xFD, 0xFF, 0x25, 0x00, 0x6C, 0xFF, 0x8B, 0x01,
+ 0x9D, 0xFC, 0xD5, 0x06, 0x89, 0xF1, 0x35, 0x33, 0x3A, 0x2D, 0x9A,
+ 0xF1, 0x23, 0x07, 0x51, 0xFC, 0xC2, 0x01, 0x4F, 0xFF, 0x2F, 0x00,
+ 0xFD, 0xFF, 0x07, 0x00, 0xE5, 0xFF, 0x1A, 0x00, 0x33, 0x00, 0xDF,
+ 0xFE, 0x68, 0x03, 0x4E, 0xF6, 0xEE, 0x42, 0xBB, 0x17, 0x90, 0xF5,
+ 0xFC, 0x05, 0x8E, 0xFC, 0xCD, 0x01, 0x3C, 0xFF, 0x34, 0x00, 0xFE,
+ 0xFF, 0x1E, 0x00, 0x87, 0xFF, 0x0B, 0x01, 0x42, 0xFE, 0x74, 0x02,
+ 0xF9, 0xFC, 0x39, 0x03, 0xF5, 0x48, 0x24, 0x04, 0x94, 0xFC, 0xA9,
+ 0x02, 0x27, 0xFE, 0x18, 0x01, 0x82, 0xFF, 0x1F, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x33, 0x00, 0x3E, 0xFF, 0xC7, 0x01, 0x9D, 0xFC, 0xD8,
+ 0x05, 0xE7, 0xF5, 0x91, 0x16, 0x89, 0x43, 0xCD, 0xF6, 0x1E, 0x03,
+ 0x0B, 0xFF, 0x1A, 0x00, 0x26, 0x00, 0xE0, 0xFF, 0x08, 0x00, 0xFD,
+ 0xFF, 0x30, 0x00, 0x4B, 0xFF, 0xC9, 0x01, 0x48, 0xFC, 0x28, 0x07,
+ 0xAD, 0xF1, 0x19, 0x2C, 0x3F, 0x34, 0x97, 0xF1, 0xBE, 0x06, 0xB0,
+ 0xFC, 0x7F, 0x01, 0x72, 0xFF, 0x23, 0x00, 0xFE, 0xFF, 0x02, 0x00,
+ 0x00, 0x00, 0xD0, 0xFF, 0xC7, 0x00, 0xDE, 0xFD, 0x08, 0x05, 0xB0,
+ 0xF3, 0x4A, 0x3E, 0x64, 0x1F, 0xA0, 0xF3, 0xBB, 0x06, 0x45, 0xFC,
+ 0xE5, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x15, 0x00, 0xA9,
+ 0xFF, 0xB4, 0x00, 0xF7, 0xFE, 0x1D, 0x01, 0x7A, 0xFF, 0xC5, 0xFD,
+ 0x1D, 0x48, 0x89, 0x0A, 0xFB, 0xF9, 0xF8, 0x03, 0x7D, 0xFD, 0x66,
+ 0x01, 0x63, 0xFF, 0x28, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2D, 0x00,
+ 0x52, 0xFF, 0x93, 0x01, 0x18, 0xFD, 0xC9, 0x04, 0x45, 0xF8, 0x36,
+ 0x0F, 0xBB, 0x46, 0xA1, 0xFA, 0x0C, 0x01, 0x3E, 0x00, 0x70, 0xFF,
+ 0x7A, 0x00, 0xC0, 0xFF, 0x0F, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x39,
+ 0xFF, 0xE4, 0x01, 0x32, 0xFC, 0x09, 0x07, 0x9D, 0xF2, 0x92, 0x24,
+ 0x8F, 0x3A, 0x82, 0xF2, 0xE1, 0x05, 0x50, 0xFD, 0x1B, 0x01, 0xA6,
+ 0xFF, 0x10, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x17, 0x00, 0x93, 0xFF,
+ 0x3F, 0x01, 0x15, 0xFD, 0x36, 0x06, 0x19, 0xF2, 0x97, 0x38, 0x11,
+ 0x27, 0x3B, 0xF2, 0x1F, 0x07, 0x32, 0xFC, 0xDF, 0x01, 0x3D, 0xFF,
+ 0x34, 0x00, 0xFD, 0xFF, 0x0D, 0x00, 0xCB, 0xFF, 0x5E, 0x00, 0xA9,
+ 0xFF, 0xD6, 0xFF, 0xC3, 0x01, 0x43, 0xF9, 0xD7, 0x45, 0x92, 0x11,
+ 0x77, 0xF7, 0x28, 0x05, 0xEC, 0xFC, 0xA7, 0x01, 0x4A, 0xFF, 0x2F,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x25, 0x00, 0x6D, 0xFF, 0x4E, 0x01,
+ 0xB3, 0xFD, 0x8D, 0x03, 0xD4, 0xFA, 0x5D, 0x08, 0x8D, 0x48, 0x74,
+ 0xFF, 0xAE, 0xFE, 0x8D, 0x01, 0xBB, 0xFE, 0xD1, 0x00, 0x9E, 0xFF,
+ 0x18, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE0, 0x01, 0x57,
+ 0xFC, 0x85, 0x06, 0x34, 0xF4, 0xE0, 0x1C, 0xF0, 0x3F, 0x6D, 0xF4,
+ 0x8C, 0x04, 0x2C, 0xFE, 0x99, 0x00, 0xE7, 0xFF, 0xF8, 0xFF, 0x04,
+ 0x00, 0xFD, 0xFF, 0x27, 0x00, 0x65, 0xFF, 0x98, 0x01, 0x8A, 0xFC,
+ 0xEC, 0x06, 0x7F, 0xF1, 0x04, 0x32, 0x7B, 0x2E, 0x8A, 0xF1, 0x1A,
+ 0x07, 0x5D, 0xFC, 0xB8, 0x01, 0x54, 0xFF, 0x2D, 0x00, 0xFD, 0xFF,
+ 0x06, 0x00, 0xEA, 0xFF, 0x0C, 0x00, 0x4E, 0x00, 0xAF, 0xFE, 0xB8,
+ 0x03, 0xC7, 0xF5, 0x38, 0x42, 0x0C, 0x19, 0x32, 0xF5, 0x23, 0x06,
+ 0x7D, 0xFC, 0xD3, 0x01, 0x3A, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1C,
+ 0x00, 0x8D, 0xFF, 0xFC, 0x00, 0x61, 0xFE, 0x39, 0x02, 0x6B, 0xFD,
+ 0x37, 0x02, 0xEB, 0x48, 0x31, 0x05, 0x21, 0xFC, 0xE4, 0x02, 0x08,
+ 0xFE, 0x26, 0x01, 0x7C, 0xFF, 0x21, 0x00, 0x00, 0x00, 0xFF, 0xFF,
+ 0x32, 0x00, 0x41, 0xFF, 0xC0, 0x01, 0xAF, 0xFC, 0xAD, 0x05, 0x4A,
+ 0xF6, 0x44, 0x15, 0x2F, 0x44, 0x64, 0xF7, 0xC9, 0x02, 0x3D, 0xFF,
+ 0xFE, 0xFF, 0x34, 0x00, 0xDB, 0xFF, 0x09, 0x00, 0xFD, 0xFF, 0x32,
+ 0x00, 0x47, 0xFF, 0xD0, 0x01, 0x40, 0xFC, 0x2A, 0x07, 0xCA, 0xF1,
+ 0xD1, 0x2A, 0x65, 0x35, 0xAE, 0xF1, 0xA0, 0x06, 0xC7, 0xFC, 0x70,
+ 0x01, 0x7A, 0xFF, 0x20, 0x00, 0xFE, 0xFF, 0x02, 0x00, 0x05, 0x00,
+ 0xC5, 0xFF, 0xDE, 0x00, 0xB7, 0xFD, 0x45, 0x05, 0x56, 0xF3, 0x61,
+ 0x3D, 0xBA, 0x20, 0x56, 0xF3, 0xD3, 0x06, 0x3E, 0xFC, 0xE6, 0x01,
+ 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x13, 0x00, 0xAF, 0xFF, 0xA5,
+ 0x00, 0x16, 0xFF, 0xE3, 0x00, 0xE4, 0xFF, 0xEB, 0xFC, 0xD2, 0x47,
+ 0xB6, 0x0B, 0x89, 0xF9, 0x2F, 0x04, 0x62, 0xFD, 0x72, 0x01, 0x5E,
+ 0xFF, 0x29, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2C, 0x00, 0x56, 0xFF,
+ 0x88, 0x01, 0x31, 0xFD, 0x95, 0x04, 0xB4, 0xF8, 0xFC, 0x0D, 0x26,
+ 0x47, 0x64, 0xFB, 0xA7, 0x00, 0x77, 0x00, 0x51, 0xFF, 0x89, 0x00,
+ 0xBA, 0xFF, 0x11, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xE6,
+ 0x01, 0x34, 0xFC, 0xF9, 0x06, 0xD9, 0xF2, 0x3F, 0x23, 0x90, 0x3B,
+ 0xC4, 0xF2, 0xAE, 0x05, 0x72, 0xFD, 0x07, 0x01, 0xB0, 0xFF, 0x0C,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x1A, 0x00, 0x8A, 0xFF, 0x51, 0x01,
+ 0xF8, 0xFC, 0x5E, 0x06, 0xED, 0xF1, 0x82, 0x37, 0x60, 0x28, 0x0E,
+ 0xF2, 0x26, 0x07, 0x35, 0xFC, 0xDB, 0x01, 0x40, 0xFF, 0x34, 0x00,
+ 0xFD, 0xFF, 0x0C, 0x00, 0xD0, 0xFF, 0x4F, 0x00, 0xC7, 0xFF, 0xA0,
+ 0xFF, 0x20, 0x02, 0x96, 0xF8, 0x4E, 0x45, 0xD7, 0x12, 0x0D, 0xF7,
+ 0x58, 0x05, 0xD6, 0xFC, 0xB0, 0x01, 0x47, 0xFF, 0x30, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x23, 0x00, 0x72, 0xFF, 0x40, 0x01, 0xD0, 0xFD,
+ 0x53, 0x03, 0x47, 0xFB, 0x3F, 0x07, 0xB8, 0x48, 0x62, 0x00, 0x3F,
+ 0xFE, 0xC8, 0x01, 0x9C, 0xFE, 0xE0, 0x00, 0x98, 0xFF, 0x19, 0x00,
+ 0xFE, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xDC, 0x01, 0x63, 0xFC, 0x66,
+ 0x06, 0x89, 0xF4, 0x8C, 0x1B, 0xC3, 0x40, 0xDD, 0xF4, 0x46, 0x04,
+ 0x58, 0xFE, 0x80, 0x00, 0xF4, 0xFF, 0xF3, 0xFF, 0x05, 0x00, 0xFD,
+ 0xFF, 0x29, 0x00, 0x5F, 0xFF, 0xA5, 0x01, 0x78, 0xFC, 0xFF, 0x06,
+ 0x7D, 0xF1, 0xCF, 0x30, 0xB8, 0x2F, 0x80, 0xF1, 0x0D, 0x07, 0x6A,
+ 0xFC, 0xAE, 0x01, 0x59, 0xFF, 0x2B, 0x00, 0xFD, 0xFF, 0x05, 0x00,
+ 0xEF, 0xFF, 0xFF, 0xFF, 0x69, 0x00, 0x80, 0xFE, 0x04, 0x04, 0x48,
+ 0xF5, 0x74, 0x41, 0x5D, 0x1A, 0xD7, 0xF4, 0x47, 0x06, 0x6F, 0xFC,
+ 0xD8, 0x01, 0x39, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1B, 0x00, 0x93,
+ 0xFF, 0xED, 0x00, 0x80, 0xFE, 0xFD, 0x01, 0xDC, 0xFD, 0x3C, 0x01,
+ 0xD5, 0x48, 0x45, 0x06, 0xAE, 0xFB, 0x1F, 0x03, 0xEA, 0xFD, 0x34,
+ 0x01, 0x77, 0xFF, 0x22, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x31, 0x00,
+ 0x44, 0xFF, 0xB8, 0x01, 0xC3, 0xFC, 0x81, 0x05, 0xB0, 0xF6, 0xFA,
+ 0x13, 0xCC, 0x44, 0x02, 0xF8, 0x71, 0x02, 0x71, 0xFF, 0xE1, 0xFF,
+ 0x42, 0x00, 0xD5, 0xFF, 0x0B, 0x00, 0xFD, 0xFF, 0x33, 0x00, 0x43,
+ 0xFF, 0xD6, 0x01, 0x39, 0xFC, 0x2A, 0x07, 0xEB, 0xF1, 0x87, 0x29,
+ 0x85, 0x36, 0xCC, 0xF1, 0x7F, 0x06, 0xE0, 0xFC, 0x60, 0x01, 0x82,
+ 0xFF, 0x1D, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x09, 0x00, 0xBA, 0xFF,
+ 0xF4, 0x00, 0x91, 0xFD, 0x7E, 0x05, 0x05, 0xF3, 0x6E, 0x3C, 0x10,
+ 0x22, 0x12, 0xF3, 0xE9, 0x06, 0x38, 0xFC, 0xE6, 0x01, 0x37, 0xFF,
+ 0x36, 0x00, 0xFD, 0xFF, 0x12, 0x00, 0xB5, 0xFF, 0x96, 0x00, 0x35,
+ 0xFF, 0xA9, 0x00, 0x4D, 0x00, 0x19, 0xFC, 0x7C, 0x47, 0xE8, 0x0C,
+ 0x18, 0xF9, 0x66, 0x04, 0x48, 0xFD, 0x7E, 0x01, 0x5A, 0xFF, 0x2B,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x2A, 0x00, 0x5A, 0xFF, 0x7D, 0x01,
+ 0x4B, 0xFD, 0x60, 0x04, 0x24, 0xF9, 0xC6, 0x0C, 0x86, 0x47, 0x30,
+ 0xFC, 0x41, 0x00, 0xB0, 0x00, 0x32, 0xFF, 0x98, 0x00, 0xB4, 0xFF,
+ 0x12, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x38,
+ 0xFC, 0xE6, 0x06, 0x19, 0xF3, 0xEA, 0x21, 0x8A, 0x3C, 0x0E, 0xF3,
+ 0x78, 0x05, 0x96, 0xFD, 0xF1, 0x00, 0xBB, 0xFF, 0x08, 0x00, 0x01,
+ 0x00, 0xFE, 0xFF, 0x1D, 0x00, 0x81, 0xFF, 0x62, 0x01, 0xDD, 0xFC,
+ 0x83, 0x06, 0xC9, 0xF1, 0x66, 0x36, 0xAC, 0x29, 0xE7, 0xF1, 0x2A,
+ 0x07, 0x3A, 0xFC, 0xD5, 0x01, 0x43, 0xFF, 0x33, 0x00, 0xFD, 0xFF,
+ 0x0B, 0x00, 0xD6, 0xFF, 0x41, 0x00, 0xE4, 0xFF, 0x6B, 0xFF, 0x7B,
+ 0x02, 0xF0, 0xF7, 0xBA, 0x44, 0x1E, 0x14, 0xA5, 0xF6, 0x86, 0x05,
+ 0xC1, 0xFC, 0xB9, 0x01, 0x44, 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0x22, 0x00, 0x77, 0xFF, 0x32, 0x01, 0xED, 0xFD, 0x19, 0x03,
+ 0xBB, 0xFB, 0x26, 0x06, 0xD7, 0x48, 0x58, 0x01, 0xCF, 0xFD, 0x04,
+ 0x02, 0x7D, 0xFE, 0xEF, 0x00, 0x92, 0xFF, 0x1B, 0x00, 0xFE, 0xFF,
+ 0x35, 0x00, 0x39, 0xFF, 0xD8, 0x01, 0x70, 0xFC, 0x43, 0x06, 0xE1,
+ 0xF4, 0x38, 0x1A, 0x8C, 0x41, 0x55, 0xF5, 0xFC, 0x03, 0x85, 0xFE,
+ 0x66, 0x00, 0x01, 0x00, 0xEE, 0xFF, 0x06, 0x00, 0xFD, 0xFF, 0x2B,
+ 0x00, 0x59, 0xFF, 0xB0, 0x01, 0x69, 0xFC, 0x0F, 0x07, 0x80, 0xF1,
+ 0x96, 0x2F, 0xF2, 0x30, 0x7C, 0xF1, 0xFD, 0x06, 0x7A, 0xFC, 0xA3,
+ 0x01, 0x5F, 0xFF, 0x29, 0x00, 0xFD, 0xFF, 0x05, 0x00, 0xF4, 0xFF,
+ 0xF2, 0xFF, 0x83, 0x00, 0x53, 0xFE, 0x4E, 0x04, 0xD0, 0xF4, 0xAB,
+ 0x40, 0xB2, 0x1B, 0x7F, 0xF4, 0x69, 0x06, 0x62, 0xFC, 0xDD, 0x01,
+ 0x38, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x19, 0x00, 0x98, 0xFF, 0xDE,
+ 0x00, 0x9F, 0xFE, 0xC2, 0x01, 0x4B, 0xFE, 0x48, 0x00, 0xB3, 0x48,
+ 0x5E, 0x07, 0x3B, 0xFB, 0x59, 0x03, 0xCD, 0xFD, 0x42, 0x01, 0x71,
+ 0xFF, 0x24, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x30, 0x00, 0x47, 0xFF,
+ 0xAF, 0x01, 0xD8, 0xFC, 0x52, 0x05, 0x19, 0xF7, 0xB2, 0x12, 0x5C,
+ 0x45, 0xA9, 0xF8, 0x16, 0x02, 0xA6, 0xFF, 0xC3, 0xFF, 0x51, 0x00,
+ 0xD0, 0xFF, 0x0C, 0x00, 0xFD, 0xFF, 0x34, 0x00, 0x40, 0xFF, 0xDB,
+ 0x01, 0x35, 0xFC, 0x25, 0x07, 0x13, 0xF2, 0x3A, 0x28, 0xA0, 0x37,
+ 0xF2, 0xF1, 0x5A, 0x06, 0xFB, 0xFC, 0x4F, 0x01, 0x8B, 0xFF, 0x1A,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x0D, 0x00, 0xAF, 0xFF, 0x09, 0x01,
+ 0x6E, 0xFD, 0xB4, 0x05, 0xBC, 0xF2, 0x73, 0x3B, 0x64, 0x23, 0xD2,
+ 0xF2, 0xFB, 0x06, 0x34, 0xFC, 0xE6, 0x01, 0x38, 0xFF, 0x36, 0x00,
+ 0xFD, 0xFF, 0x11, 0x00, 0xBB, 0xFF, 0x87, 0x00, 0x54, 0xFF, 0x70,
+ 0x00, 0xB3, 0x00, 0x4E, 0xFB, 0x1A, 0x47, 0x1F, 0x0E, 0xA8, 0xF8,
+ 0x9B, 0x04, 0x2E, 0xFD, 0x8A, 0x01, 0x55, 0xFF, 0x2C, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x29, 0x00, 0x5F, 0xFF, 0x71, 0x01, 0x65, 0xFD,
+ 0x29, 0x04, 0x96, 0xF9, 0x95, 0x0B, 0xDC, 0x47, 0x03, 0xFD, 0xD9,
+ 0xFF, 0xEA, 0x00, 0x12, 0xFF, 0xA7, 0x00, 0xAE, 0xFF, 0x14, 0x00,
+ 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3E, 0xFC, 0xD0,
+ 0x06, 0x5E, 0xF3, 0x94, 0x20, 0x7B, 0x3D, 0x60, 0xF3, 0x3E, 0x05,
+ 0xBB, 0xFD, 0xDB, 0x00, 0xC6, 0xFF, 0x04, 0x00, 0x02, 0x00, 0xFE,
+ 0xFF, 0x20, 0x00, 0x79, 0xFF, 0x72, 0x01, 0xC4, 0xFC, 0xA4, 0x06,
+ 0xAB, 0xF1, 0x46, 0x35, 0xF7, 0x2A, 0xC6, 0xF1, 0x2A, 0x07, 0x40,
+ 0xFC, 0xCF, 0x01, 0x47, 0xFF, 0x31, 0x00, 0xFD, 0xFF, 0x09, 0x00,
+ 0xDB, 0xFF, 0x33, 0x00, 0x01, 0x00, 0x38, 0xFF, 0xD3, 0x02, 0x53,
+ 0xF7, 0x1F, 0x44, 0x69, 0x15, 0x3F, 0xF6, 0xB2, 0x05, 0xAD, 0xFC,
+ 0xC1, 0x01, 0x41, 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x20,
+ 0x00, 0x7D, 0xFF, 0x24, 0x01, 0x0C, 0xFE, 0xDE, 0x02, 0x2E, 0xFC,
+ 0x13, 0x05, 0xEC, 0x48, 0x54, 0x02, 0x5E, 0xFD, 0x3F, 0x02, 0x5D,
+ 0xFE, 0xFE, 0x00, 0x8C, 0xFF, 0x1C, 0x00, 0xFE, 0xFF, 0x35, 0x00,
+ 0x3B, 0xFF, 0xD3, 0x01, 0x7F, 0xFC, 0x1F, 0x06, 0x3C, 0xF5, 0xE6,
+ 0x18, 0x4D, 0x42, 0xD5, 0xF5, 0xAF, 0x03, 0xB4, 0xFE, 0x4B, 0x00,
+ 0x0E, 0x00, 0xE9, 0xFF, 0x07, 0x00, 0xFD, 0xFF, 0x2D, 0x00, 0x53,
+ 0xFF, 0xBA, 0x01, 0x5B, 0xFC, 0x1B, 0x07, 0x8B, 0xF1, 0x58, 0x2E,
+ 0x26, 0x32, 0x80, 0xF1, 0xEA, 0x06, 0x8C, 0xFC, 0x97, 0x01, 0x66,
+ 0xFF, 0x27, 0x00, 0xFD, 0xFF, 0x04, 0x00, 0xF8, 0xFF, 0xE6, 0xFF,
+ 0x9C, 0x00, 0x27, 0xFE, 0x94, 0x04, 0x61, 0xF4, 0xD7, 0x3F, 0x06,
+ 0x1D, 0x2B, 0xF4, 0x89, 0x06, 0x56, 0xFC, 0xE0, 0x01, 0x37, 0xFF,
+ 0x36, 0x00, 0xFE, 0xFF, 0x17, 0x00, 0x9E, 0xFF, 0xCF, 0x00, 0xBF,
+ 0xFE, 0x86, 0x01, 0xBA, 0xFE, 0x5A, 0xFF, 0x86, 0x48, 0x7D, 0x08,
+ 0xC7, 0xFA, 0x93, 0x03, 0xB0, 0xFD, 0x4F, 0x01, 0x6C, 0xFF, 0x25,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2F, 0x00, 0x4B, 0xFF, 0xA6, 0x01,
+ 0xEE, 0xFC, 0x23, 0x05, 0x83, 0xF7, 0x6E, 0x11, 0xE5, 0x45, 0x57,
+ 0xF9, 0xB8, 0x01, 0xDC, 0xFF, 0xA5, 0xFF, 0x5F, 0x00, 0xCA, 0xFF,
+ 0x0D, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3D, 0xFF, 0xDF, 0x01, 0x32,
+ 0xFC, 0x1E, 0x07, 0x40, 0xF2, 0xEB, 0x26, 0xB5, 0x38, 0x1F, 0xF2,
+ 0x32, 0x06, 0x18, 0xFD, 0x3D, 0x01, 0x94, 0xFF, 0x16, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x11, 0x00, 0xA4, 0xFF, 0x1D, 0x01, 0x4C, 0xFD,
+ 0xE6, 0x05, 0x7B, 0xF2, 0x71, 0x3A, 0xB8, 0x24, 0x97, 0xF2, 0x0B,
+ 0x07, 0x32, 0xFC, 0xE4, 0x01, 0x39, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0x0F, 0x00, 0xC0, 0xFF, 0x78, 0x00, 0x73, 0xFF, 0x38, 0x00, 0x17,
+ 0x01, 0x8B, 0xFA, 0xAF, 0x46, 0x59, 0x0F, 0x39, 0xF8, 0xCF, 0x04,
+ 0x15, 0xFD, 0x95, 0x01, 0x51, 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0x28, 0x00, 0x64, 0xFF, 0x65, 0x01, 0x81, 0xFD, 0xF2, 0x03,
+ 0x08, 0xFA, 0x68, 0x0A, 0x25, 0x48, 0xDE, 0xFD, 0x6E, 0xFF, 0x24,
+ 0x01, 0xF3, 0xFE, 0xB6, 0x00, 0xA8, 0xFF, 0x15, 0x00, 0xFD, 0xFF,
+ 0x36, 0x00, 0x36, 0xFF, 0xE5, 0x01, 0x46, 0xFC, 0xB8, 0x06, 0xA8,
+ 0xF3, 0x3F, 0x1F, 0x64, 0x3E, 0xBA, 0xF3, 0x01, 0x05, 0xE2, 0xFD,
+ 0xC4, 0x00, 0xD2, 0xFF, 0x00, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x23,
+ 0x00, 0x71, 0xFF, 0x81, 0x01, 0xAE, 0xFC, 0xC1, 0x06, 0x95, 0xF1,
+ 0x1E, 0x34, 0x3E, 0x2C, 0xAB, 0xF1, 0x27, 0x07, 0x49, 0xFC, 0xC8,
+ 0x01, 0x4B, 0xFF, 0x30, 0x00, 0xFD, 0xFF, 0x08, 0x00, 0xE1, 0xFF,
+ 0x25, 0x00, 0x1D, 0x00, 0x05, 0xFF, 0x28, 0x03, 0xBD, 0xF6, 0x77,
+ 0x43, 0xB6, 0x16, 0xDC, 0xF5, 0xDD, 0x05, 0x9B, 0xFC, 0xC8, 0x01,
+ 0x3E, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x1F, 0x00, 0x83,
+ 0xFF, 0x16, 0x01, 0x2A, 0xFE, 0xA3, 0x02, 0xA1, 0xFC, 0x06, 0x04,
+ 0xF5, 0x48, 0x56, 0x03, 0xED, 0xFC, 0x7B, 0x02, 0x3E, 0xFE, 0x0C,
+ 0x01, 0x86, 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3D, 0xFF,
+ 0xCC, 0x01, 0x8F, 0xFC, 0xF8, 0x05, 0x9B, 0xF5, 0x96, 0x17, 0x02,
+ 0x43, 0x5E, 0xF6, 0x5F, 0x03, 0xE4, 0xFE, 0x30, 0x00, 0x1B, 0x00,
+ 0xE4, 0xFF, 0x08, 0x00, 0xFD, 0xFF, 0x2F, 0x00, 0x4E, 0xFF, 0xC3,
+ 0x01, 0x4F, 0xFC, 0x24, 0x07, 0x9C, 0xF1, 0x17, 0x2D, 0x57, 0x33,
+ 0x8A, 0xF1, 0xD3, 0x06, 0x9F, 0xFC, 0x8A, 0x01, 0x6D, 0xFF, 0x25,
+ 0x00, 0xFD, 0xFF, 0x03, 0x00, 0xFD, 0xFF, 0xD9, 0xFF, 0xB4, 0x00,
+ 0xFD, 0xFD, 0xD7, 0x04, 0xFA, 0xF3, 0xFC, 0x3E, 0x5B, 0x1E, 0xDB,
+ 0xF3, 0xA6, 0x06, 0x4C, 0xFC, 0xE3, 0x01, 0x36, 0xFF, 0x36, 0x00,
+ 0xFE, 0xFF, 0x16, 0x00, 0xA4, 0xFF, 0xC0, 0x00, 0xDE, 0xFE, 0x4B,
+ 0x01, 0x27, 0xFF, 0x73, 0xFE, 0x4F, 0x48, 0xA2, 0x09, 0x54, 0xFA,
+ 0xCC, 0x03, 0x93, 0xFD, 0x5C, 0x01, 0x67, 0xFF, 0x27, 0x00, 0x00,
+ 0x00, 0xFF, 0xFF, 0x2E, 0x00, 0x4E, 0xFF, 0x9C, 0x01, 0x05, 0xFD,
+ 0xF1, 0x04, 0xF0, 0xF7, 0x2D, 0x10, 0x61, 0x46, 0x0D, 0xFA, 0x58,
+ 0x01, 0x13, 0x00, 0x87, 0xFF, 0x6E, 0x00, 0xC4, 0xFF, 0x0E, 0x00,
+ 0xFD, 0xFF, 0x35, 0x00, 0x3B, 0xFF, 0xE3, 0x01, 0x31, 0xFC, 0x14,
+ 0x07, 0x73, 0xF2, 0x99, 0x25, 0xC2, 0x39, 0x54, 0xF2, 0x05, 0x06,
+ 0x37, 0xFD, 0x2B, 0x01, 0x9E, 0xFF, 0x13, 0x00, 0xFF, 0xFF, 0xFF,
+ 0xFF, 0x14, 0x00, 0x9B, 0xFF, 0x31, 0x01, 0x2C, 0xFD, 0x15, 0x06,
+ 0x41, 0xF2, 0x6A, 0x39, 0x0A, 0x26, 0x61, 0xF2, 0x17, 0x07, 0x31,
+ 0xFC, 0xE2, 0x01, 0x3B, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x0E, 0x00,
+ 0xC6, 0xFF, 0x69, 0x00, 0x91, 0xFF, 0x00, 0x00, 0x78, 0x01, 0xD0,
+ 0xF9, 0x39, 0x46, 0x98, 0x10, 0xCB, 0xF7, 0x02, 0x05, 0xFE, 0xFC,
+ 0x9F, 0x01, 0x4D, 0xFF, 0x2E, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x26,
+ 0x00, 0x69, 0xFF, 0x58, 0x01, 0x9D, 0xFD, 0xB9, 0x03, 0x7B, 0xFA,
+ 0x40, 0x09, 0x63, 0x48, 0xBF, 0xFE, 0x03, 0xFF, 0x5F, 0x01, 0xD4,
+ 0xFE, 0xC5, 0x00, 0xA2, 0xFF, 0x16, 0x00, 0xFE, 0xFF, 0x36, 0x00,
+ 0x36, 0xFF, 0xE2, 0x01, 0x4F, 0xFC, 0x9C, 0x06, 0xF5, 0xF3, 0xEA,
+ 0x1D, 0x47, 0x3F, 0x1B, 0xF4, 0xC1, 0x04, 0x0B, 0xFE, 0xAC, 0x00,
+ 0xDE, 0xFF, 0xFB, 0xFF, 0x03, 0x00, 0xFD, 0xFF, 0x25, 0x00, 0x6A,
+ 0xFF, 0x8E, 0x01, 0x99, 0xFC, 0xDB, 0x06, 0x86, 0xF1, 0xF2, 0x32,
+ 0x82, 0x2D, 0x96, 0xF1, 0x21, 0x07, 0x53, 0xFC, 0xC0, 0x01, 0x50,
+ 0xFF, 0x2E, 0x00, 0xFD, 0xFF, 0x07, 0x00, 0xE6, 0xFF, 0x17, 0x00,
+ 0x39, 0x00, 0xD4, 0xFE, 0x7A, 0x03, 0x2F, 0xF6, 0xC7, 0x42, 0x06,
+ 0x18, 0x7B, 0xF5, 0x05, 0x06, 0x8A, 0xFC, 0xCF, 0x01, 0x3C, 0xFF,
+ 0x34, 0x00, 0xFE, 0xFF, 0x1D, 0x00, 0x88, 0xFF, 0x07, 0x01, 0x49,
+ 0xFE, 0x67, 0x02, 0x13, 0xFD, 0xFF, 0x02, 0xF4, 0x48, 0x5F, 0x04,
+ 0x7A, 0xFC, 0xB6, 0x02, 0x20, 0xFE, 0x1B, 0x01, 0x81, 0xFF, 0x1F,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x3F, 0xFF, 0xC6, 0x01,
+ 0xA1, 0xFC, 0xCF, 0x05, 0xFC, 0xF5, 0x47, 0x16, 0xB0, 0x43, 0xEE,
+ 0xF6, 0x0C, 0x03, 0x16, 0xFF, 0x14, 0x00, 0x29, 0x00, 0xDF, 0xFF,
+ 0x09, 0x00, 0xFD, 0xFF, 0x30, 0x00, 0x4A, 0xFF, 0xCA, 0x01, 0x46,
+ 0xFC, 0x29, 0x07, 0xB3, 0xF1, 0xD1, 0x2B, 0x81, 0x34, 0x9C, 0xF1,
+ 0xB8, 0x06, 0xB5, 0xFC, 0x7C, 0x01, 0x74, 0xFF, 0x22, 0x00, 0xFE,
+ 0xFF, 0x02, 0x00, 0x01, 0x00, 0xCE, 0xFF, 0xCC, 0x00, 0xD5, 0xFD,
+ 0x16, 0x05, 0x9B, 0xF3, 0x18, 0x3E, 0xB1, 0x1F, 0x8F, 0xF3, 0xC0,
+ 0x06, 0x43, 0xFC, 0xE5, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0x15, 0x00, 0xAA, 0xFF, 0xB1, 0x00, 0xFE, 0xFE, 0x10, 0x01, 0x92,
+ 0xFF, 0x94, 0xFD, 0x0D, 0x48, 0xCB, 0x0A, 0xE2, 0xF9, 0x04, 0x04,
+ 0x77, 0xFD, 0x69, 0x01, 0x62, 0xFF, 0x28, 0x00, 0x00, 0x00, 0xFF,
+ 0xFF, 0x2D, 0x00, 0x52, 0xFF, 0x91, 0x01, 0x1E, 0xFD, 0xBE, 0x04,
+ 0x5E, 0xF8, 0xF0, 0x0E, 0xD3, 0x46, 0xCB, 0xFA, 0xF6, 0x00, 0x4B,
+ 0x00, 0x69, 0xFF, 0x7D, 0x00, 0xBE, 0xFF, 0x10, 0x00, 0xFD, 0xFF,
+ 0x36, 0x00, 0x39, 0xFF, 0xE5, 0x01, 0x32, 0xFC, 0x06, 0x07, 0xAA,
+ 0xF2, 0x46, 0x24, 0xC8, 0x3A, 0x90, 0xF2, 0xD6, 0x05, 0x57, 0xFD,
+ 0x17, 0x01, 0xA8, 0xFF, 0x0F, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x18,
+ 0x00, 0x91, 0xFF, 0x43, 0x01, 0x0E, 0xFD, 0x40, 0x06, 0x0F, 0xF2,
+ 0x5B, 0x38, 0x5C, 0x27, 0x30, 0xF2, 0x21, 0x07, 0x33, 0xFC, 0xDE,
+ 0x01, 0x3E, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x0D, 0x00, 0xCC, 0xFF,
+ 0x5A, 0x00, 0xAF, 0xFF, 0xCA, 0xFF, 0xD8, 0x01, 0x1C, 0xF9, 0xB8,
+ 0x45, 0xDA, 0x11, 0x60, 0xF7, 0x33, 0x05, 0xE7, 0xFC, 0xA9, 0x01,
+ 0x4A, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x25, 0x00, 0x6E,
+ 0xFF, 0x4B, 0x01, 0xB9, 0xFD, 0x80, 0x03, 0xEE, 0xFA, 0x1D, 0x08,
+ 0x98, 0x48, 0xA8, 0xFF, 0x95, 0xFE, 0x9A, 0x01, 0xB4, 0xFE, 0xD4,
+ 0x00, 0x9C, 0xFF, 0x18, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF,
+ 0xDF, 0x01, 0x5A, 0xFC, 0x7E, 0x06, 0x47, 0xF4, 0x94, 0x1C, 0x1F,
+ 0x40, 0x85, 0xF4, 0x7D, 0x04, 0x36, 0xFE, 0x93, 0x00, 0xEA, 0xFF,
+ 0xF7, 0xFF, 0x04, 0x00, 0xFD, 0xFF, 0x28, 0x00, 0x63, 0xFF, 0x9B,
+ 0x01, 0x86, 0xFC, 0xF1, 0x06, 0x7E, 0xF1, 0xC0, 0x31, 0xC2, 0x2E,
+ 0x87, 0xF1, 0x17, 0x07, 0x5F, 0xFC, 0xB6, 0x01, 0x55, 0xFF, 0x2D,
+ 0x00, 0xFD, 0xFF, 0x06, 0x00, 0xEB, 0xFF, 0x09, 0x00, 0x54, 0x00,
+ 0xA4, 0xFE, 0xC9, 0x03, 0xAA, 0xF5, 0x0C, 0x42, 0x56, 0x19, 0x1E,
+ 0xF5, 0x2B, 0x06, 0x7A, 0xFC, 0xD4, 0x01, 0x3A, 0xFF, 0x35, 0x00,
+ 0xFE, 0xFF, 0x1C, 0x00, 0x8E, 0xFF, 0xF9, 0x00, 0x68, 0xFE, 0x2C,
+ 0x02, 0x84, 0xFD, 0xFF, 0x01, 0xE6, 0x48, 0x6E, 0x05, 0x07, 0xFC,
+ 0xF1, 0x02, 0x01, 0xFE, 0x29, 0x01, 0x7B, 0xFF, 0x21, 0x00, 0x00,
+ 0x00, 0xFF, 0xFF, 0x32, 0x00, 0x42, 0xFF, 0xBE, 0x01, 0xB4, 0xFC,
+ 0xA4, 0x05, 0x61, 0xF6, 0xFB, 0x14, 0x53, 0x44, 0x86, 0xF7, 0xB6,
+ 0x02, 0x49, 0xFF, 0xF7, 0xFF, 0x37, 0x00, 0xD9, 0xFF, 0x0A, 0x00,
+ 0xFD, 0xFF, 0x32, 0x00, 0x46, 0xFF, 0xD1, 0x01, 0x3E, 0xFC, 0x2B,
+ 0x07, 0xD0, 0xF1, 0x89, 0x2A, 0xA6, 0x35, 0xB4, 0xF1, 0x99, 0x06,
+ 0xCD, 0xFC, 0x6D, 0x01, 0x7C, 0xFF, 0x1F, 0x00, 0xFE, 0xFF, 0x01,
+ 0x00, 0x06, 0x00, 0xC2, 0xFF, 0xE3, 0x00, 0xAE, 0xFD, 0x52, 0x05,
+ 0x44, 0xF3, 0x2A, 0x3D, 0x06, 0x21, 0x47, 0xF3, 0xD8, 0x06, 0x3C,
+ 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x13, 0x00,
+ 0xB0, 0xFF, 0xA2, 0x00, 0x1D, 0xFF, 0xD6, 0x00, 0xFC, 0xFF, 0xBC,
+ 0xFC, 0xC0, 0x47, 0xFA, 0x0B, 0x70, 0xF9, 0x3C, 0x04, 0x5C, 0xFD,
+ 0x75, 0x01, 0x5D, 0xFF, 0x2A, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2B,
+ 0x00, 0x57, 0xFF, 0x86, 0x01, 0x36, 0xFD, 0x89, 0x04, 0xCD, 0xF8,
+ 0xB7, 0x0D, 0x3D, 0x47, 0x91, 0xFB, 0x91, 0x00, 0x83, 0x00, 0x4A,
+ 0xFF, 0x8C, 0x00, 0xB9, 0xFF, 0x11, 0x00, 0xFD, 0xFF, 0x36, 0x00,
+ 0x38, 0xFF, 0xE6, 0x01, 0x35, 0xFC, 0xF5, 0x06, 0xE7, 0xF2, 0xF2,
+ 0x22, 0xC7, 0x3B, 0xD4, 0xF2, 0xA2, 0x05, 0x7A, 0xFD, 0x02, 0x01,
+ 0xB2, 0xFF, 0x0B, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x1B, 0x00, 0x88,
+ 0xFF, 0x55, 0x01, 0xF2, 0xFC, 0x67, 0x06, 0xE4, 0xF1, 0x44, 0x37,
+ 0xAA, 0x28, 0x05, 0xF2, 0x27, 0x07, 0x36, 0xFC, 0xDA, 0x01, 0x41,
+ 0xFF, 0x33, 0x00, 0xFD, 0xFF, 0x0B, 0x00, 0xD2, 0xFF, 0x4C, 0x00,
+ 0xCD, 0xFF, 0x94, 0xFF, 0x34, 0x02, 0x70, 0xF8, 0x2E, 0x45, 0x20,
+ 0x13, 0xF6, 0xF6, 0x62, 0x05, 0xD1, 0xFC, 0xB2, 0x01, 0x46, 0xFF,
+ 0x31, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x23, 0x00, 0x73, 0xFF, 0x3D,
+ 0x01, 0xD6, 0xFD, 0x46, 0x03, 0x61, 0xFB, 0x00, 0x07, 0xBF, 0x48,
+ 0x98, 0x00, 0x26, 0xFE, 0xD5, 0x01, 0x95, 0xFE, 0xE3, 0x00, 0x96,
+ 0xFF, 0x1A, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xDB, 0x01,
+ 0x66, 0xFC, 0x5E, 0x06, 0x9C, 0xF4, 0x40, 0x1B, 0xEF, 0x40, 0xF7,
+ 0xF4, 0x35, 0x04, 0x62, 0xFE, 0x7A, 0x00, 0xF7, 0xFF, 0xF2, 0xFF,
+ 0x05, 0x00, 0xFD, 0xFF, 0x2A, 0x00, 0x5D, 0xFF, 0xA7, 0x01, 0x75,
+ 0xFC, 0x03, 0x07, 0x7D, 0xF1, 0x8A, 0x30, 0xFF, 0x2F, 0x7E, 0xF1,
+ 0x0A, 0x07, 0x6E, 0xFC, 0xAC, 0x01, 0x5A, 0xFF, 0x2B, 0x00, 0xFD,
+ 0xFF, 0x05, 0x00, 0xF0, 0xFF, 0xFC, 0xFF, 0x6E, 0x00, 0x76, 0xFE,
+ 0x15, 0x04, 0x2C, 0xF5, 0x49, 0x41, 0xA9, 0x1A, 0xC3, 0xF4, 0x4F,
+ 0x06, 0x6C, 0xFC, 0xD9, 0x01, 0x38, 0xFF, 0x35, 0x00, 0xFE, 0xFF,
+ 0x1A, 0x00, 0x94, 0xFF, 0xEA, 0x00, 0x87, 0xFE, 0xF0, 0x01, 0xF5,
+ 0xFD, 0x05, 0x01, 0xCE, 0x48, 0x83, 0x06, 0x94, 0xFB, 0x2C, 0x03,
+ 0xE4, 0xFD, 0x37, 0x01, 0x76, 0xFF, 0x22, 0x00, 0x00, 0x00, 0xFF,
+ 0xFF, 0x31, 0x00, 0x45, 0xFF, 0xB6, 0x01, 0xC8, 0xFC, 0x77, 0x05,
+ 0xC7, 0xF6, 0xB1, 0x13, 0xED, 0x44, 0x26, 0xF8, 0x5D, 0x02, 0x7D,
+ 0xFF, 0xDA, 0xFF, 0x46, 0x00, 0xD4, 0xFF, 0x0B, 0x00, 0xFD, 0xFF,
+ 0x33, 0x00, 0x42, 0xFF, 0xD7, 0x01, 0x38, 0xFC, 0x29, 0x07, 0xF3,
+ 0xF1, 0x3E, 0x29, 0xC6, 0x36, 0xD4, 0xF1, 0x77, 0x06, 0xE6, 0xFC,
+ 0x5C, 0x01, 0x84, 0xFF, 0x1C, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x0A,
+ 0x00, 0xB7, 0xFF, 0xF9, 0x00, 0x89, 0xFD, 0x8A, 0x05, 0xF4, 0xF2,
+ 0x37, 0x3C, 0x5B, 0x22, 0x03, 0xF3, 0xED, 0x06, 0x37, 0xFC, 0xE6,
+ 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x12, 0x00, 0xB6, 0xFF,
+ 0x93, 0x00, 0x3C, 0xFF, 0x9D, 0x00, 0x63, 0x00, 0xEB, 0xFB, 0x69,
+ 0x47, 0x2D, 0x0D, 0xFF, 0xF8, 0x72, 0x04, 0x42, 0xFD, 0x81, 0x01,
+ 0x59, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2A, 0x00, 0x5B,
+ 0xFF, 0x7A, 0x01, 0x50, 0xFD, 0x54, 0x04, 0x3D, 0xF9, 0x82, 0x0C,
+ 0x9A, 0x47, 0x5E, 0xFC, 0x2A, 0x00, 0xBD, 0x00, 0x2B, 0xFF, 0x9B,
+ 0x00, 0xB3, 0xFF, 0x12, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF,
+ 0xE6, 0x01, 0x3A, 0xFC, 0xE2, 0x06, 0x28, 0xF3, 0x9E, 0x21, 0xC0,
+ 0x3C, 0x1F, 0xF3, 0x6C, 0x05, 0x9E, 0xFD, 0xED, 0x00, 0xBD, 0xFF,
+ 0x07, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x1E, 0x00, 0x80, 0xFF, 0x66,
+ 0x01, 0xD8, 0xFC, 0x8B, 0x06, 0xC1, 0xF1, 0x27, 0x36, 0xF6, 0x29,
+ 0xDF, 0xF1, 0x2A, 0x07, 0x3B, 0xFC, 0xD4, 0x01, 0x44, 0xFF, 0x32,
+ 0x00, 0xFD, 0xFF, 0x0A, 0x00, 0xD7, 0xFF, 0x3E, 0x00, 0xEA, 0xFF,
+ 0x60, 0xFF, 0x8F, 0x02, 0xCD, 0xF7, 0x99, 0x44, 0x68, 0x14, 0x8E,
+ 0xF6, 0x90, 0x05, 0xBC, 0xFC, 0xBA, 0x01, 0x43, 0xFF, 0x32, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0x22, 0x00, 0x79, 0xFF, 0x2F, 0x01, 0xF4,
+ 0xFD, 0x0C, 0x03, 0xD4, 0xFB, 0xE9, 0x05, 0xDE, 0x48, 0x8F, 0x01,
+ 0xB6, 0xFD, 0x11, 0x02, 0x76, 0xFE, 0xF2, 0x00, 0x91, 0xFF, 0x1B,
+ 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x39, 0xFF, 0xD7, 0x01, 0x73, 0xFC,
+ 0x3B, 0x06, 0xF5, 0xF4, 0xED, 0x19, 0xB7, 0x41, 0x71, 0xF5, 0xEB,
+ 0x03, 0x90, 0xFE, 0x60, 0x00, 0x04, 0x00, 0xED, 0xFF, 0x06, 0x00,
+ 0xFD, 0xFF, 0x2C, 0x00, 0x57, 0xFF, 0xB2, 0x01, 0x65, 0xFC, 0x12,
+ 0x07, 0x82, 0xF1, 0x50, 0x2F, 0x38, 0x31, 0x7C, 0xF1, 0xF9, 0x06,
+ 0x7E, 0xFC, 0xA1, 0x01, 0x61, 0xFF, 0x29, 0x00, 0xFD, 0xFF, 0x04,
+ 0x00, 0xF5, 0xFF, 0xEF, 0xFF, 0x88, 0x00, 0x49, 0xFE, 0x5D, 0x04,
+ 0xB7, 0xF4, 0x7D, 0x40, 0xFD, 0x1B, 0x6C, 0xF4, 0x70, 0x06, 0x5F,
+ 0xFC, 0xDE, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x19, 0x00,
+ 0x9A, 0xFF, 0xDB, 0x00, 0xA6, 0xFE, 0xB4, 0x01, 0x64, 0xFE, 0x12,
+ 0x00, 0xAA, 0x48, 0x9E, 0x07, 0x21, 0xFB, 0x66, 0x03, 0xC6, 0xFD,
+ 0x45, 0x01, 0x70, 0xFF, 0x24, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x30,
+ 0x00, 0x48, 0xFF, 0xAD, 0x01, 0xDD, 0xFC, 0x48, 0x05, 0x30, 0xF7,
+ 0x6B, 0x12, 0x7D, 0x45, 0xCF, 0xF8, 0x01, 0x02, 0xB2, 0xFF, 0xBD,
+ 0xFF, 0x54, 0x00, 0xCE, 0xFF, 0x0C, 0x00, 0xFD, 0xFF, 0x34, 0x00,
+ 0x3F, 0xFF, 0xDC, 0x01, 0x34, 0xFC, 0x24, 0x07, 0x1C, 0xF2, 0xF0,
+ 0x27, 0xDF, 0x37, 0xFB, 0xF1, 0x51, 0x06, 0x01, 0xFD, 0x4B, 0x01,
+ 0x8D, 0xFF, 0x19, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x0E, 0x00, 0xAC,
+ 0xFF, 0x0E, 0x01, 0x66, 0xFD, 0xBF, 0x05, 0xAD, 0xF2, 0x3B, 0x3B,
+ 0xB0, 0x23, 0xC4, 0xF2, 0xFF, 0x06, 0x33, 0xFC, 0xE5, 0x01, 0x38,
+ 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x10, 0x00, 0xBC, 0xFF, 0x84, 0x00,
+ 0x5B, 0xFF, 0x64, 0x00, 0xC9, 0x00, 0x22, 0xFB, 0x02, 0x47, 0x64,
+ 0x0E, 0x8F, 0xF8, 0xA7, 0x04, 0x29, 0xFD, 0x8C, 0x01, 0x54, 0xFF,
+ 0x2C, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x29, 0x00, 0x60, 0xFF, 0x6E,
+ 0x01, 0x6B, 0xFD, 0x1D, 0x04, 0xAF, 0xF9, 0x51, 0x0B, 0xEC, 0x47,
+ 0x33, 0xFD, 0xC1, 0xFF, 0xF7, 0x00, 0x0C, 0xFF, 0xAA, 0x00, 0xAD,
+ 0xFF, 0x14, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01,
+ 0x40, 0xFC, 0xCB, 0x06, 0x6E, 0xF3, 0x49, 0x20, 0xB0, 0x3D, 0x73,
+ 0xF3, 0x31, 0x05, 0xC4, 0xFD, 0xD6, 0x00, 0xC8, 0xFF, 0x03, 0x00,
+ 0x02, 0x00, 0xFE, 0xFF, 0x21, 0x00, 0x77, 0xFF, 0x75, 0x01, 0xBF,
+ 0xFC, 0xAB, 0x06, 0xA6, 0xF1, 0x05, 0x35, 0x40, 0x2B, 0xBF, 0xF1,
+ 0x2A, 0x07, 0x42, 0xFC, 0xCE, 0x01, 0x48, 0xFF, 0x31, 0x00, 0xFD,
+ 0xFF, 0x09, 0x00, 0xDC, 0xFF, 0x2F, 0x00, 0x07, 0x00, 0x2C, 0xFF,
+ 0xE6, 0x02, 0x31, 0xF7, 0xFA, 0x43, 0xB3, 0x15, 0x29, 0xF6, 0xBC,
+ 0x05, 0xA9, 0xFC, 0xC2, 0x01, 0x40, 0xFF, 0x33, 0x00, 0xFF, 0xFF,
+ 0x00, 0x00, 0x20, 0x00, 0x7E, 0xFF, 0x21, 0x01, 0x12, 0xFE, 0xD1,
+ 0x02, 0x47, 0xFC, 0xD7, 0x04, 0xF0, 0x48, 0x8D, 0x02, 0x45, 0xFD,
+ 0x4D, 0x02, 0x56, 0xFE, 0x01, 0x01, 0x8B, 0xFF, 0x1D, 0x00, 0xFE,
+ 0xFF, 0x34, 0x00, 0x3B, 0xFF, 0xD1, 0x01, 0x83, 0xFC, 0x16, 0x06,
+ 0x51, 0xF5, 0x9B, 0x18, 0x75, 0x42, 0xF3, 0xF5, 0x9D, 0x03, 0xBF,
+ 0xFE, 0x45, 0x00, 0x11, 0x00, 0xE8, 0xFF, 0x07, 0x00, 0xFD, 0xFF,
+ 0x2E, 0x00, 0x52, 0xFF, 0xBC, 0x01, 0x58, 0xFC, 0x1D, 0x07, 0x8E,
+ 0xF1, 0x11, 0x2E, 0x6B, 0x32, 0x81, 0xF1, 0xE5, 0x06, 0x90, 0xFC,
+ 0x94, 0x01, 0x67, 0xFF, 0x26, 0x00, 0xFD, 0xFF, 0x04, 0x00, 0xF9,
+ 0xFF, 0xE3, 0xFF, 0xA1, 0x00, 0x1E, 0xFE, 0xA3, 0x04, 0x49, 0xF4,
+ 0xA8, 0x3F, 0x52, 0x1D, 0x19, 0xF4, 0x90, 0x06, 0x53, 0xFC, 0xE1,
+ 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x17, 0x00, 0xA0, 0xFF,
+ 0xCC, 0x00, 0xC6, 0xFE, 0x79, 0x01, 0xD2, 0xFE, 0x26, 0xFF, 0x7C,
+ 0x48, 0xBE, 0x08, 0xAE, 0xFA, 0xA0, 0x03, 0xA9, 0xFD, 0x52, 0x01,
+ 0x6B, 0xFF, 0x25, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2F, 0x00, 0x4C,
+ 0xFF, 0xA3, 0x01, 0xF3, 0xFC, 0x18, 0x05, 0x9B, 0xF7, 0x27, 0x11,
+ 0x02, 0x46, 0x7F, 0xF9, 0xA3, 0x01, 0xE8, 0xFF, 0x9F, 0xFF, 0x63,
+ 0x00, 0xC9, 0xFF, 0x0D, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3C, 0xFF,
+ 0xE0, 0x01, 0x32, 0xFC, 0x1C, 0x07, 0x4B, 0xF2, 0xA0, 0x26, 0xF2,
+ 0x38, 0x2A, 0xF2, 0x28, 0x06, 0x1F, 0xFD, 0x39, 0x01, 0x96, 0xFF,
+ 0x16, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x11, 0x00, 0xA2, 0xFF, 0x22,
+ 0x01, 0x45, 0xFD, 0xF1, 0x05, 0x6D, 0xF2, 0x38, 0x3A, 0x03, 0x25,
+ 0x8B, 0xF2, 0x0E, 0x07, 0x32, 0xFC, 0xE4, 0x01, 0x3A, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0x0F, 0x00, 0xC2, 0xFF, 0x75, 0x00, 0x7A, 0xFF,
+ 0x2B, 0x00, 0x2D, 0x01, 0x61, 0xFA, 0x97, 0x46, 0xA0, 0x0F, 0x20,
+ 0xF8, 0xDA, 0x04, 0x10, 0xFD, 0x97, 0x01, 0x50, 0xFF, 0x2E, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0x27, 0x00, 0x65, 0xFF, 0x62, 0x01, 0x87,
+ 0xFD, 0xE5, 0x03, 0x21, 0xFA, 0x25, 0x0A, 0x33, 0x48, 0x0F, 0xFE,
+ 0x57, 0xFF, 0x31, 0x01, 0xEC, 0xFE, 0xB9, 0x00, 0xA7, 0xFF, 0x15,
+ 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE4, 0x01, 0x48, 0xFC,
+ 0xB2, 0x06, 0xB9, 0xF3, 0xF3, 0x1E, 0x98, 0x3E, 0xCF, 0xF3, 0xF3,
+ 0x04, 0xEB, 0xFD, 0xBF, 0x00, 0xD4, 0xFF, 0xFF, 0xFF, 0x03, 0x00,
+ 0xFE, 0xFF, 0x23, 0x00, 0x70, 0xFF, 0x84, 0x01, 0xA9, 0xFC, 0xC7,
+ 0x06, 0x91, 0xF1, 0xDC, 0x33, 0x87, 0x2C, 0xA5, 0xF1, 0x26, 0x07,
+ 0x4B, 0xFC, 0xC6, 0x01, 0x4C, 0xFF, 0x30, 0x00, 0xFD, 0xFF, 0x08,
+ 0x00, 0xE2, 0xFF, 0x21, 0x00, 0x23, 0x00, 0xFA, 0xFE, 0x3A, 0x03,
+ 0x9D, 0xF6, 0x50, 0x43, 0x00, 0x17, 0xC6, 0xF5, 0xE6, 0x05, 0x97,
+ 0xFC, 0xC9, 0x01, 0x3E, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0x00, 0x00,
+ 0x1E, 0x00, 0x84, 0xFF, 0x13, 0x01, 0x31, 0xFE, 0x95, 0x02, 0xBA,
+ 0xFC, 0xCB, 0x03, 0xF7, 0x48, 0x91, 0x03, 0xD3, 0xFC, 0x88, 0x02,
+ 0x38, 0xFE, 0x10, 0x01, 0x85, 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0x34,
+ 0x00, 0x3D, 0xFF, 0xCB, 0x01, 0x93, 0xFC, 0xEF, 0x05, 0xB0, 0xF5,
+ 0x4B, 0x17, 0x2A, 0x43, 0x7D, 0xF6, 0x4D, 0x03, 0xEF, 0xFE, 0x2A,
+ 0x00, 0x1E, 0x00, 0xE3, 0xFF, 0x08, 0x00, 0xFD, 0xFF, 0x2F, 0x00,
+ 0x4D, 0xFF, 0xC4, 0x01, 0x4D, 0xFC, 0x25, 0x07, 0xA1, 0xF1, 0xCE,
+ 0x2C, 0x99, 0x33, 0x8E, 0xF1, 0xCD, 0x06, 0xA4, 0xFC, 0x87, 0x01,
+ 0x6E, 0xFF, 0x24, 0x00, 0xFD, 0xFF, 0x03, 0x00, 0xFE, 0xFF, 0xD7,
+ 0xFF, 0xBA, 0x00, 0xF4, 0xFD, 0xE5, 0x04, 0xE4, 0xF3, 0xCA, 0x3E,
+ 0xA7, 0x1E, 0xCA, 0xF3, 0xAC, 0x06, 0x4A, 0xFC, 0xE4, 0x01, 0x36,
+ 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x16, 0x00, 0xA6, 0xFF, 0xBD, 0x00,
+ 0xE5, 0xFE, 0x3E, 0x01, 0x3F, 0xFF, 0x41, 0xFE, 0x41, 0x48, 0xE4,
+ 0x09, 0x3B, 0xFA, 0xD9, 0x03, 0x8D, 0xFD, 0x5F, 0x01, 0x66, 0xFF,
+ 0x27, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2E, 0x00, 0x4F, 0xFF, 0x99,
+ 0x01, 0x0B, 0xFD, 0xE6, 0x04, 0x08, 0xF8, 0xE7, 0x0F, 0x7C, 0x46,
+ 0x37, 0xFA, 0x42, 0x01, 0x1F, 0x00, 0x81, 0xFF, 0x71, 0x00, 0xC3,
+ 0xFF, 0x0F, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xE3, 0x01,
+ 0x31, 0xFC, 0x11, 0x07, 0x7F, 0xF2, 0x4E, 0x25, 0xFD, 0x39, 0x60,
+ 0xF2, 0xFB, 0x05, 0x3E, 0xFD, 0x26, 0x01, 0xA0, 0xFF, 0x12, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x15, 0x00, 0x98, 0xFF, 0x35, 0x01, 0x25,
+ 0xFD, 0x1E, 0x06, 0x35, 0xF2, 0x2E, 0x39, 0x55, 0x26, 0x56, 0xF2,
+ 0x1A, 0x07, 0x31, 0xFC, 0xE1, 0x01, 0x3C, 0xFF, 0x35, 0x00, 0xFD,
+ 0xFF, 0x0E, 0x00, 0xC7, 0xFF, 0x66, 0x00, 0x98, 0xFF, 0xF4, 0xFF,
+ 0x8E, 0x01, 0xA7, 0xF9, 0x1D, 0x46, 0xDF, 0x10, 0xB3, 0xF7, 0x0D,
+ 0x05, 0xF8, 0xFC, 0xA1, 0x01, 0x4C, 0xFF, 0x2F, 0x00, 0xFF, 0xFF,
+ 0x00, 0x00, 0x26, 0x00, 0x6A, 0xFF, 0x55, 0x01, 0xA3, 0xFD, 0xAD,
+ 0x03, 0x94, 0xFA, 0xFF, 0x08, 0x70, 0x48, 0xF3, 0xFE, 0xEA, 0xFE,
+ 0x6C, 0x01, 0xCD, 0xFE, 0xC9, 0x00, 0xA1, 0xFF, 0x17, 0x00, 0xFE,
+ 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE2, 0x01, 0x51, 0xFC, 0x96, 0x06,
+ 0x07, 0xF4, 0x9E, 0x1D, 0x77, 0x3F, 0x32, 0xF4, 0xB2, 0x04, 0x15,
+ 0xFE, 0xA7, 0x00, 0xE0, 0xFF, 0xFA, 0xFF, 0x03, 0x00, 0xFD, 0xFF,
+ 0x26, 0x00, 0x69, 0xFF, 0x91, 0x01, 0x94, 0xFC, 0xE0, 0x06, 0x84,
+ 0xF1, 0xAF, 0x32, 0xCA, 0x2D, 0x92, 0xF1, 0x1F, 0x07, 0x56, 0xFC,
+ 0xBE, 0x01, 0x51, 0xFF, 0x2E, 0x00, 0xFD, 0xFF, 0x07, 0x00, 0xE7,
+ 0xFF, 0x14, 0x00, 0x3F, 0x00, 0xC9, 0xFE, 0x8C, 0x03, 0x11, 0xF6,
+ 0x9E, 0x42, 0x50, 0x18, 0x66, 0xF5, 0x0D, 0x06, 0x86, 0xFC, 0xD0,
+ 0x01, 0x3B, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0x1D, 0x00, 0x8A, 0xFF,
+ 0x04, 0x01, 0x50, 0xFE, 0x5A, 0x02, 0x2C, 0xFD, 0xC6, 0x02, 0xF2,
+ 0x48, 0x9B, 0x04, 0x61, 0xFC, 0xC3, 0x02, 0x19, 0xFE, 0x1E, 0x01,
+ 0x7F, 0xFF, 0x20, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x40,
+ 0xFF, 0xC4, 0x01, 0xA5, 0xFC, 0xC5, 0x05, 0x13, 0xF6, 0xFD, 0x15,
+ 0xD4, 0x43, 0x0F, 0xF7, 0xF9, 0x02, 0x21, 0xFF, 0x0D, 0x00, 0x2C,
+ 0x00, 0xDE, 0xFF, 0x09, 0x00, 0xFD, 0xFF, 0x31, 0x00, 0x49, 0xFF,
+ 0xCC, 0x01, 0x44, 0xFC, 0x29, 0x07, 0xB9, 0xF1, 0x89, 0x2B, 0xC3,
+ 0x34, 0xA0, 0xF1, 0xB1, 0x06, 0xBA, 0xFC, 0x79, 0x01, 0x76, 0xFF,
+ 0x21, 0x00, 0xFE, 0xFF, 0x02, 0x00, 0x02, 0x00, 0xCB, 0xFF, 0xD1,
+ 0x00, 0xCC, 0xFD, 0x24, 0x05, 0x87, 0xF3, 0xE4, 0x3D, 0xFD, 0x1F,
+ 0x7F, 0xF3, 0xC6, 0x06, 0x41, 0xFC, 0xE5, 0x01, 0x36, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0x14, 0x00, 0xAC, 0xFF, 0xAE, 0x00, 0x05, 0xFF,
+ 0x03, 0x01, 0xAA, 0xFF, 0x63, 0xFD, 0xFD, 0x47, 0x0E, 0x0B, 0xC8,
+ 0xF9, 0x11, 0x04, 0x71, 0xFD, 0x6C, 0x01, 0x61, 0xFF, 0x28, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x2D, 0x00, 0x53, 0xFF, 0x8F, 0x01, 0x23,
+ 0xFD, 0xB2, 0x04, 0x76, 0xF8, 0xAA, 0x0E, 0xED, 0x46, 0xF7, 0xFA,
+ 0xDF, 0x00, 0x57, 0x00, 0x62, 0xFF, 0x80, 0x00, 0xBD, 0xFF, 0x10,
+ 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x39, 0xFF, 0xE5, 0x01, 0x33, 0xFC,
+ 0x03, 0x07, 0xB7, 0xF2, 0xFC, 0x23, 0x03, 0x3B, 0x9E, 0xF2, 0xCB,
+ 0x05, 0x5F, 0xFD, 0x12, 0x01, 0xAA, 0xFF, 0x0E, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x18, 0x00, 0x8F, 0xFF, 0x47, 0x01, 0x08, 0xFD, 0x49,
+ 0x06, 0x05, 0xF2, 0x1D, 0x38, 0xA6, 0x27, 0x26, 0xF2, 0x23, 0x07,
+ 0x33, 0xFC, 0xDD, 0x01, 0x3E, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x0C,
+ 0x00, 0xCD, 0xFF, 0x57, 0x00, 0xB6, 0xFF, 0xBE, 0xFF, 0xED, 0x01,
+ 0xF5, 0xF8, 0x9B, 0x45, 0x22, 0x12, 0x48, 0xF7, 0x3D, 0x05, 0xE2,
+ 0xFC, 0xAB, 0x01, 0x49, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0x24, 0x00, 0x6F, 0xFF, 0x48, 0x01, 0xC0, 0xFD, 0x73, 0x03, 0x07,
+ 0xFB, 0xDD, 0x07, 0xA1, 0x48, 0xDD, 0xFF, 0x7D, 0xFE, 0xA7, 0x01,
+ 0xAD, 0xFE, 0xD8, 0x00, 0x9B, 0xFF, 0x18, 0x00, 0xFE, 0xFF, 0x36,
+ 0x00, 0x37, 0xFF, 0xDF, 0x01, 0x5C, 0xFC, 0x78, 0x06, 0x5A, 0xF4,
+ 0x49, 0x1C, 0x4E, 0x40, 0x9E, 0xF4, 0x6D, 0x04, 0x3F, 0xFE, 0x8E,
+ 0x00, 0xED, 0xFF, 0xF6, 0xFF, 0x04, 0x00, 0xFD, 0xFF, 0x28, 0x00,
+ 0x62, 0xFF, 0x9E, 0x01, 0x82, 0xFC, 0xF5, 0x06, 0x7D, 0xF1, 0x7B,
+ 0x31, 0x09, 0x2F, 0x84, 0xF1, 0x15, 0x07, 0x62, 0xFC, 0xB4, 0x01,
+ 0x56, 0xFF, 0x2C, 0x00, 0xFD, 0xFF, 0x06, 0x00, 0xEC, 0xFF, 0x06,
+ 0x00, 0x5A, 0x00, 0x9A, 0xFE, 0xDA, 0x03, 0x8D, 0xF5, 0xE1, 0x41,
+ 0xA1, 0x19, 0x09, 0xF5, 0x33, 0x06, 0x77, 0xFC, 0xD6, 0x01, 0x3A,
+ 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1B, 0x00, 0x8F, 0xFF, 0xF5, 0x00,
+ 0x6F, 0xFE, 0x1E, 0x02, 0x9D, 0xFD, 0xC7, 0x01, 0xE1, 0x48, 0xAB,
+ 0x05, 0xEE, 0xFB, 0xFE, 0x02, 0xFB, 0xFD, 0x2C, 0x01, 0x7A, 0xFF,
+ 0x21, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x32, 0x00, 0x42, 0xFF, 0xBC,
+ 0x01, 0xB8, 0xFC, 0x9A, 0x05, 0x77, 0xF6, 0xB1, 0x14, 0x77, 0x44,
+ 0xA9, 0xF7, 0xA2, 0x02, 0x54, 0xFF, 0xF1, 0xFF, 0x3A, 0x00, 0xD8,
+ 0xFF, 0x0A, 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x45, 0xFF, 0xD3, 0x01,
+ 0x3C, 0xFC, 0x2A, 0x07, 0xD8, 0xF1, 0x3F, 0x2A, 0xE6, 0x35, 0xBB,
+ 0xF1, 0x92, 0x06, 0xD2, 0xFC, 0x69, 0x01, 0x7E, 0xFF, 0x1F, 0x00,
+ 0xFE, 0xFF, 0x01, 0x00, 0x07, 0x00, 0xC0, 0xFF, 0xE8, 0x00, 0xA6,
+ 0xFD, 0x5F, 0x05, 0x31, 0xF3, 0xF6, 0x3C, 0x52, 0x21, 0x37, 0xF3,
+ 0xDD, 0x06, 0x3B, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0x13, 0x00, 0xB1, 0xFF, 0x9F, 0x00, 0x24, 0xFF, 0xC9, 0x00,
+ 0x13, 0x00, 0x8D, 0xFC, 0xAE, 0x47, 0x3E, 0x0C, 0x56, 0xF9, 0x48,
+ 0x04, 0x56, 0xFD, 0x78, 0x01, 0x5C, 0xFF, 0x2A, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x2B, 0x00, 0x58, 0xFF, 0x83, 0x01, 0x3C, 0xFD, 0x7E,
+ 0x04, 0xE6, 0xF8, 0x72, 0x0D, 0x52, 0x47, 0xBE, 0xFB, 0x7A, 0x00,
+ 0x90, 0x00, 0x43, 0xFF, 0x8F, 0x00, 0xB7, 0xFF, 0x11, 0x00, 0xFD,
+ 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x36, 0xFC, 0xF1, 0x06,
+ 0xF5, 0xF2, 0xA7, 0x22, 0xFF, 0x3B, 0xE4, 0xF2, 0x96, 0x05, 0x81,
+ 0xFD, 0xFD, 0x00, 0xB5, 0xFF, 0x0B, 0x00, 0x01, 0x00, 0xFE, 0xFF,
+ 0x1C, 0x00, 0x86, 0xFF, 0x59, 0x01, 0xEC, 0xFC, 0x6F, 0x06, 0xDC,
+ 0xF1, 0x04, 0x37, 0xF3, 0x28, 0xFC, 0xF1, 0x28, 0x07, 0x37, 0xFC,
+ 0xD8, 0x01, 0x41, 0xFF, 0x33, 0x00, 0xFD, 0xFF, 0x0B, 0x00, 0xD3,
+ 0xFF, 0x49, 0x00, 0xD4, 0xFF, 0x88, 0xFF, 0x49, 0x02, 0x4B, 0xF8,
+ 0x0D, 0x45, 0x68, 0x13, 0xDF, 0xF6, 0x6C, 0x05, 0xCC, 0xFC, 0xB4,
+ 0x01, 0x45, 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x23, 0x00,
+ 0x74, 0xFF, 0x3A, 0x01, 0xDD, 0xFD, 0x39, 0x03, 0x7B, 0xFB, 0xC1,
+ 0x06, 0xC7, 0x48, 0xCF, 0x00, 0x0D, 0xFE, 0xE3, 0x01, 0x8E, 0xFE,
+ 0xE7, 0x00, 0x95, 0xFF, 0x1A, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x38,
+ 0xFF, 0xDA, 0x01, 0x69, 0xFC, 0x57, 0x06, 0xAF, 0xF4, 0xF5, 0x1A,
+ 0x1D, 0x41, 0x11, 0xF5, 0x25, 0x04, 0x6C, 0xFE, 0x74, 0x00, 0xF9,
+ 0xFF, 0xF1, 0xFF, 0x05, 0x00, 0xFD, 0xFF, 0x2A, 0x00, 0x5C, 0xFF,
+ 0xAA, 0x01, 0x71, 0xFC, 0x07, 0x07, 0x7E, 0xF1, 0x44, 0x30, 0x44,
+ 0x30, 0x7E, 0xF1, 0x07, 0x07, 0x71, 0xFC, 0xAA, 0x01, 0x5C, 0xFF,
+ 0x2A, 0x00, 0xFD, 0xFF, 0x05, 0x00, 0xF1, 0xFF, 0xF9, 0xFF, 0x74,
+ 0x00, 0x6C, 0xFE, 0x25, 0x04, 0x11, 0xF5, 0x1D, 0x41, 0xF5, 0x1A,
+ 0xAF, 0xF4, 0x57, 0x06, 0x69, 0xFC, 0xDA, 0x01, 0x38, 0xFF, 0x36,
+ 0x00, 0xFE, 0xFF, 0x1A, 0x00, 0x95, 0xFF, 0xE7, 0x00, 0x8E, 0xFE,
+ 0xE3, 0x01, 0x0D, 0xFE, 0xCF, 0x00, 0xC7, 0x48, 0xC1, 0x06, 0x7B,
+ 0xFB, 0x39, 0x03, 0xDD, 0xFD, 0x3A, 0x01, 0x74, 0xFF, 0x23, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x31, 0x00, 0x45, 0xFF, 0xB4, 0x01, 0xCC,
+ 0xFC, 0x6C, 0x05, 0xDF, 0xF6, 0x68, 0x13, 0x0D, 0x45, 0x4B, 0xF8,
+ 0x49, 0x02, 0x88, 0xFF, 0xD4, 0xFF, 0x49, 0x00, 0xD3, 0xFF, 0x0B,
+ 0x00, 0xFD, 0xFF, 0x33, 0x00, 0x41, 0xFF, 0xD8, 0x01, 0x37, 0xFC,
+ 0x28, 0x07, 0xFC, 0xF1, 0xF3, 0x28, 0x04, 0x37, 0xDC, 0xF1, 0x6F,
+ 0x06, 0xEC, 0xFC, 0x59, 0x01, 0x86, 0xFF, 0x1C, 0x00, 0xFE, 0xFF,
+ 0x01, 0x00, 0x0B, 0x00, 0xB5, 0xFF, 0xFD, 0x00, 0x81, 0xFD, 0x96,
+ 0x05, 0xE4, 0xF2, 0xFF, 0x3B, 0xA7, 0x22, 0xF5, 0xF2, 0xF1, 0x06,
+ 0x36, 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x11,
+ 0x00, 0xB7, 0xFF, 0x8F, 0x00, 0x43, 0xFF, 0x90, 0x00, 0x7A, 0x00,
+ 0xBE, 0xFB, 0x52, 0x47, 0x72, 0x0D, 0xE6, 0xF8, 0x7E, 0x04, 0x3C,
+ 0xFD, 0x83, 0x01, 0x58, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x2A, 0x00, 0x5C, 0xFF, 0x78, 0x01, 0x56, 0xFD, 0x48, 0x04, 0x56,
+ 0xF9, 0x3E, 0x0C, 0xAE, 0x47, 0x8D, 0xFC, 0x13, 0x00, 0xC9, 0x00,
+ 0x24, 0xFF, 0x9F, 0x00, 0xB1, 0xFF, 0x13, 0x00, 0xFD, 0xFF, 0x36,
+ 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3B, 0xFC, 0xDD, 0x06, 0x37, 0xF3,
+ 0x52, 0x21, 0xF6, 0x3C, 0x31, 0xF3, 0x5F, 0x05, 0xA6, 0xFD, 0xE8,
+ 0x00, 0xC0, 0xFF, 0x07, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x1F, 0x00,
+ 0x7E, 0xFF, 0x69, 0x01, 0xD2, 0xFC, 0x92, 0x06, 0xBB, 0xF1, 0xE6,
+ 0x35, 0x3F, 0x2A, 0xD8, 0xF1, 0x2A, 0x07, 0x3C, 0xFC, 0xD3, 0x01,
+ 0x45, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x0A, 0x00, 0xD8, 0xFF, 0x3A,
+ 0x00, 0xF1, 0xFF, 0x54, 0xFF, 0xA2, 0x02, 0xA9, 0xF7, 0x77, 0x44,
+ 0xB1, 0x14, 0x77, 0xF6, 0x9A, 0x05, 0xB8, 0xFC, 0xBC, 0x01, 0x42,
+ 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x21, 0x00, 0x7A, 0xFF,
+ 0x2C, 0x01, 0xFB, 0xFD, 0xFE, 0x02, 0xEE, 0xFB, 0xAB, 0x05, 0xE1,
+ 0x48, 0xC7, 0x01, 0x9D, 0xFD, 0x1E, 0x02, 0x6F, 0xFE, 0xF5, 0x00,
+ 0x8F, 0xFF, 0x1B, 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xD6,
+ 0x01, 0x77, 0xFC, 0x33, 0x06, 0x09, 0xF5, 0xA1, 0x19, 0xE1, 0x41,
+ 0x8D, 0xF5, 0xDA, 0x03, 0x9A, 0xFE, 0x5A, 0x00, 0x06, 0x00, 0xEC,
+ 0xFF, 0x06, 0x00, 0xFD, 0xFF, 0x2C, 0x00, 0x56, 0xFF, 0xB4, 0x01,
+ 0x62, 0xFC, 0x15, 0x07, 0x84, 0xF1, 0x09, 0x2F, 0x7B, 0x31, 0x7D,
+ 0xF1, 0xF5, 0x06, 0x82, 0xFC, 0x9E, 0x01, 0x62, 0xFF, 0x28, 0x00,
+ 0xFD, 0xFF, 0x04, 0x00, 0xF6, 0xFF, 0xED, 0xFF, 0x8E, 0x00, 0x3F,
+ 0xFE, 0x6D, 0x04, 0x9E, 0xF4, 0x4E, 0x40, 0x49, 0x1C, 0x5A, 0xF4,
+ 0x78, 0x06, 0x5C, 0xFC, 0xDF, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE,
+ 0xFF, 0x18, 0x00, 0x9B, 0xFF, 0xD8, 0x00, 0xAD, 0xFE, 0xA7, 0x01,
+ 0x7D, 0xFE, 0xDD, 0xFF, 0xA1, 0x48, 0xDD, 0x07, 0x07, 0xFB, 0x73,
+ 0x03, 0xC0, 0xFD, 0x48, 0x01, 0x6F, 0xFF, 0x24, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x30, 0x00, 0x49, 0xFF, 0xAB, 0x01, 0xE2, 0xFC, 0x3D,
+ 0x05, 0x48, 0xF7, 0x22, 0x12, 0x9B, 0x45, 0xF5, 0xF8, 0xED, 0x01,
+ 0xBE, 0xFF, 0xB6, 0xFF, 0x57, 0x00, 0xCD, 0xFF, 0x0C, 0x00, 0xFD,
+ 0xFF, 0x34, 0x00, 0x3E, 0xFF, 0xDD, 0x01, 0x33, 0xFC, 0x23, 0x07,
+ 0x26, 0xF2, 0xA6, 0x27, 0x1D, 0x38, 0x05, 0xF2, 0x49, 0x06, 0x08,
+ 0xFD, 0x47, 0x01, 0x8F, 0xFF, 0x18, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0x0E, 0x00, 0xAA, 0xFF, 0x12, 0x01, 0x5F, 0xFD, 0xCB, 0x05, 0x9E,
+ 0xF2, 0x03, 0x3B, 0xFC, 0x23, 0xB7, 0xF2, 0x03, 0x07, 0x33, 0xFC,
+ 0xE5, 0x01, 0x39, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x10, 0x00, 0xBD,
+ 0xFF, 0x80, 0x00, 0x62, 0xFF, 0x57, 0x00, 0xDF, 0x00, 0xF7, 0xFA,
+ 0xED, 0x46, 0xAA, 0x0E, 0x76, 0xF8, 0xB2, 0x04, 0x23, 0xFD, 0x8F,
+ 0x01, 0x53, 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x28, 0x00,
+ 0x61, 0xFF, 0x6C, 0x01, 0x71, 0xFD, 0x11, 0x04, 0xC8, 0xF9, 0x0E,
+ 0x0B, 0xFD, 0x47, 0x63, 0xFD, 0xAA, 0xFF, 0x03, 0x01, 0x05, 0xFF,
+ 0xAE, 0x00, 0xAC, 0xFF, 0x14, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36,
+ 0xFF, 0xE5, 0x01, 0x41, 0xFC, 0xC6, 0x06, 0x7F, 0xF3, 0xFD, 0x1F,
+ 0xE4, 0x3D, 0x87, 0xF3, 0x24, 0x05, 0xCC, 0xFD, 0xD1, 0x00, 0xCB,
+ 0xFF, 0x02, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x21, 0x00, 0x76, 0xFF,
+ 0x79, 0x01, 0xBA, 0xFC, 0xB1, 0x06, 0xA0, 0xF1, 0xC3, 0x34, 0x89,
+ 0x2B, 0xB9, 0xF1, 0x29, 0x07, 0x44, 0xFC, 0xCC, 0x01, 0x49, 0xFF,
+ 0x31, 0x00, 0xFD, 0xFF, 0x09, 0x00, 0xDE, 0xFF, 0x2C, 0x00, 0x0D,
+ 0x00, 0x21, 0xFF, 0xF9, 0x02, 0x0F, 0xF7, 0xD4, 0x43, 0xFD, 0x15,
+ 0x13, 0xF6, 0xC5, 0x05, 0xA5, 0xFC, 0xC4, 0x01, 0x40, 0xFF, 0x33,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x20, 0x00, 0x7F, 0xFF, 0x1E, 0x01,
+ 0x19, 0xFE, 0xC3, 0x02, 0x61, 0xFC, 0x9B, 0x04, 0xF2, 0x48, 0xC6,
+ 0x02, 0x2C, 0xFD, 0x5A, 0x02, 0x50, 0xFE, 0x04, 0x01, 0x8A, 0xFF,
+ 0x1D, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3B, 0xFF, 0xD0, 0x01, 0x86,
+ 0xFC, 0x0D, 0x06, 0x66, 0xF5, 0x50, 0x18, 0x9E, 0x42, 0x11, 0xF6,
+ 0x8C, 0x03, 0xC9, 0xFE, 0x3F, 0x00, 0x14, 0x00, 0xE7, 0xFF, 0x07,
+ 0x00, 0xFD, 0xFF, 0x2E, 0x00, 0x51, 0xFF, 0xBE, 0x01, 0x56, 0xFC,
+ 0x1F, 0x07, 0x92, 0xF1, 0xCA, 0x2D, 0xAF, 0x32, 0x84, 0xF1, 0xE0,
+ 0x06, 0x94, 0xFC, 0x91, 0x01, 0x69, 0xFF, 0x26, 0x00, 0xFD, 0xFF,
+ 0x03, 0x00, 0xFA, 0xFF, 0xE0, 0xFF, 0xA7, 0x00, 0x15, 0xFE, 0xB2,
+ 0x04, 0x32, 0xF4, 0x77, 0x3F, 0x9E, 0x1D, 0x07, 0xF4, 0x96, 0x06,
+ 0x51, 0xFC, 0xE2, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x17,
+ 0x00, 0xA1, 0xFF, 0xC9, 0x00, 0xCD, 0xFE, 0x6C, 0x01, 0xEA, 0xFE,
+ 0xF3, 0xFE, 0x70, 0x48, 0xFF, 0x08, 0x94, 0xFA, 0xAD, 0x03, 0xA3,
+ 0xFD, 0x55, 0x01, 0x6A, 0xFF, 0x26, 0x00, 0x00, 0x00, 0xFF, 0xFF,
+ 0x2F, 0x00, 0x4C, 0xFF, 0xA1, 0x01, 0xF8, 0xFC, 0x0D, 0x05, 0xB3,
+ 0xF7, 0xDF, 0x10, 0x1D, 0x46, 0xA7, 0xF9, 0x8E, 0x01, 0xF4, 0xFF,
+ 0x98, 0xFF, 0x66, 0x00, 0xC7, 0xFF, 0x0E, 0x00, 0xFD, 0xFF, 0x35,
+ 0x00, 0x3C, 0xFF, 0xE1, 0x01, 0x31, 0xFC, 0x1A, 0x07, 0x56, 0xF2,
+ 0x55, 0x26, 0x2E, 0x39, 0x35, 0xF2, 0x1E, 0x06, 0x25, 0xFD, 0x35,
+ 0x01, 0x98, 0xFF, 0x15, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x12, 0x00,
+ 0xA0, 0xFF, 0x26, 0x01, 0x3E, 0xFD, 0xFB, 0x05, 0x60, 0xF2, 0xFD,
+ 0x39, 0x4E, 0x25, 0x7F, 0xF2, 0x11, 0x07, 0x31, 0xFC, 0xE3, 0x01,
+ 0x3A, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x0F, 0x00, 0xC3, 0xFF, 0x71,
+ 0x00, 0x81, 0xFF, 0x1F, 0x00, 0x42, 0x01, 0x37, 0xFA, 0x7C, 0x46,
+ 0xE7, 0x0F, 0x08, 0xF8, 0xE6, 0x04, 0x0B, 0xFD, 0x99, 0x01, 0x4F,
+ 0xFF, 0x2E, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x27, 0x00, 0x66, 0xFF,
+ 0x5F, 0x01, 0x8D, 0xFD, 0xD9, 0x03, 0x3B, 0xFA, 0xE4, 0x09, 0x41,
+ 0x48, 0x41, 0xFE, 0x3F, 0xFF, 0x3E, 0x01, 0xE5, 0xFE, 0xBD, 0x00,
+ 0xA6, 0xFF, 0x16, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE4,
+ 0x01, 0x4A, 0xFC, 0xAC, 0x06, 0xCA, 0xF3, 0xA7, 0x1E, 0xCA, 0x3E,
+ 0xE4, 0xF3, 0xE5, 0x04, 0xF4, 0xFD, 0xBA, 0x00, 0xD7, 0xFF, 0xFE,
+ 0xFF, 0x03, 0x00, 0xFD, 0xFF, 0x24, 0x00, 0x6E, 0xFF, 0x87, 0x01,
+ 0xA4, 0xFC, 0xCD, 0x06, 0x8E, 0xF1, 0x99, 0x33, 0xCE, 0x2C, 0xA1,
+ 0xF1, 0x25, 0x07, 0x4D, 0xFC, 0xC4, 0x01, 0x4D, 0xFF, 0x2F, 0x00,
+ 0xFD, 0xFF, 0x08, 0x00, 0xE3, 0xFF, 0x1E, 0x00, 0x2A, 0x00, 0xEF,
+ 0xFE, 0x4D, 0x03, 0x7D, 0xF6, 0x2A, 0x43, 0x4B, 0x17, 0xB0, 0xF5,
+ 0xEF, 0x05, 0x93, 0xFC, 0xCB, 0x01, 0x3D, 0xFF, 0x34, 0x00, 0xFE,
+ 0xFF, 0x1E, 0x00, 0x85, 0xFF, 0x10, 0x01, 0x38, 0xFE, 0x88, 0x02,
+ 0xD3, 0xFC, 0x91, 0x03, 0xF7, 0x48, 0xCB, 0x03, 0xBA, 0xFC, 0x95,
+ 0x02, 0x31, 0xFE, 0x13, 0x01, 0x84, 0xFF, 0x1E, 0x00, 0x00, 0x00,
+ 0xFE, 0xFF, 0x34, 0x00, 0x3E, 0xFF, 0xC9, 0x01, 0x97, 0xFC, 0xE6,
+ 0x05, 0xC6, 0xF5, 0x00, 0x17, 0x50, 0x43, 0x9D, 0xF6, 0x3A, 0x03,
+ 0xFA, 0xFE, 0x23, 0x00, 0x21, 0x00, 0xE2, 0xFF, 0x08, 0x00, 0xFD,
+ 0xFF, 0x30, 0x00, 0x4C, 0xFF, 0xC6, 0x01, 0x4B, 0xFC, 0x26, 0x07,
+ 0xA5, 0xF1, 0x87, 0x2C, 0xDC, 0x33, 0x91, 0xF1, 0xC7, 0x06, 0xA9,
+ 0xFC, 0x84, 0x01, 0x70, 0xFF, 0x23, 0x00, 0xFE, 0xFF, 0x03, 0x00,
+ 0xFF, 0xFF, 0xD4, 0xFF, 0xBF, 0x00, 0xEB, 0xFD, 0xF3, 0x04, 0xCF,
+ 0xF3, 0x98, 0x3E, 0xF3, 0x1E, 0xB9, 0xF3, 0xB2, 0x06, 0x48, 0xFC,
+ 0xE4, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x15, 0x00, 0xA7,
+ 0xFF, 0xB9, 0x00, 0xEC, 0xFE, 0x31, 0x01, 0x57, 0xFF, 0x0F, 0xFE,
+ 0x33, 0x48, 0x25, 0x0A, 0x21, 0xFA, 0xE5, 0x03, 0x87, 0xFD, 0x62,
+ 0x01, 0x65, 0xFF, 0x27, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2E, 0x00,
+ 0x50, 0xFF, 0x97, 0x01, 0x10, 0xFD, 0xDA, 0x04, 0x20, 0xF8, 0xA0,
+ 0x0F, 0x97, 0x46, 0x61, 0xFA, 0x2D, 0x01, 0x2B, 0x00, 0x7A, 0xFF,
+ 0x75, 0x00, 0xC2, 0xFF, 0x0F, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x3A,
+ 0xFF, 0xE4, 0x01, 0x32, 0xFC, 0x0E, 0x07, 0x8B, 0xF2, 0x03, 0x25,
+ 0x38, 0x3A, 0x6D, 0xF2, 0xF1, 0x05, 0x45, 0xFD, 0x22, 0x01, 0xA2,
+ 0xFF, 0x11, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x16, 0x00, 0x96, 0xFF,
+ 0x39, 0x01, 0x1F, 0xFD, 0x28, 0x06, 0x2A, 0xF2, 0xF2, 0x38, 0xA0,
+ 0x26, 0x4B, 0xF2, 0x1C, 0x07, 0x32, 0xFC, 0xE0, 0x01, 0x3C, 0xFF,
+ 0x35, 0x00, 0xFD, 0xFF, 0x0D, 0x00, 0xC9, 0xFF, 0x63, 0x00, 0x9F,
+ 0xFF, 0xE8, 0xFF, 0xA3, 0x01, 0x7F, 0xF9, 0x02, 0x46, 0x27, 0x11,
+ 0x9B, 0xF7, 0x18, 0x05, 0xF3, 0xFC, 0xA3, 0x01, 0x4C, 0xFF, 0x2F,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x25, 0x00, 0x6B, 0xFF, 0x52, 0x01,
+ 0xA9, 0xFD, 0xA0, 0x03, 0xAE, 0xFA, 0xBE, 0x08, 0x7C, 0x48, 0x26,
+ 0xFF, 0xD2, 0xFE, 0x79, 0x01, 0xC6, 0xFE, 0xCC, 0x00, 0xA0, 0xFF,
+ 0x17, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE1, 0x01, 0x53,
+ 0xFC, 0x90, 0x06, 0x19, 0xF4, 0x52, 0x1D, 0xA8, 0x3F, 0x49, 0xF4,
+ 0xA3, 0x04, 0x1E, 0xFE, 0xA1, 0x00, 0xE3, 0xFF, 0xF9, 0xFF, 0x04,
+ 0x00, 0xFD, 0xFF, 0x26, 0x00, 0x67, 0xFF, 0x94, 0x01, 0x90, 0xFC,
+ 0xE5, 0x06, 0x81, 0xF1, 0x6B, 0x32, 0x11, 0x2E, 0x8E, 0xF1, 0x1D,
+ 0x07, 0x58, 0xFC, 0xBC, 0x01, 0x52, 0xFF, 0x2E, 0x00, 0xFD, 0xFF,
+ 0x07, 0x00, 0xE8, 0xFF, 0x11, 0x00, 0x45, 0x00, 0xBF, 0xFE, 0x9D,
+ 0x03, 0xF3, 0xF5, 0x75, 0x42, 0x9B, 0x18, 0x51, 0xF5, 0x16, 0x06,
+ 0x83, 0xFC, 0xD1, 0x01, 0x3B, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0x1D,
+ 0x00, 0x8B, 0xFF, 0x01, 0x01, 0x56, 0xFE, 0x4D, 0x02, 0x45, 0xFD,
+ 0x8D, 0x02, 0xF0, 0x48, 0xD7, 0x04, 0x47, 0xFC, 0xD1, 0x02, 0x12,
+ 0xFE, 0x21, 0x01, 0x7E, 0xFF, 0x20, 0x00, 0x00, 0x00, 0xFF, 0xFF,
+ 0x33, 0x00, 0x40, 0xFF, 0xC2, 0x01, 0xA9, 0xFC, 0xBC, 0x05, 0x29,
+ 0xF6, 0xB3, 0x15, 0xFA, 0x43, 0x31, 0xF7, 0xE6, 0x02, 0x2C, 0xFF,
+ 0x07, 0x00, 0x2F, 0x00, 0xDC, 0xFF, 0x09, 0x00, 0xFD, 0xFF, 0x31,
+ 0x00, 0x48, 0xFF, 0xCE, 0x01, 0x42, 0xFC, 0x2A, 0x07, 0xBF, 0xF1,
+ 0x40, 0x2B, 0x05, 0x35, 0xA6, 0xF1, 0xAB, 0x06, 0xBF, 0xFC, 0x75,
+ 0x01, 0x77, 0xFF, 0x21, 0x00, 0xFE, 0xFF, 0x02, 0x00, 0x03, 0x00,
+ 0xC8, 0xFF, 0xD6, 0x00, 0xC4, 0xFD, 0x31, 0x05, 0x73, 0xF3, 0xB0,
+ 0x3D, 0x49, 0x20, 0x6E, 0xF3, 0xCB, 0x06, 0x40, 0xFC, 0xE6, 0x01,
+ 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x14, 0x00, 0xAD, 0xFF, 0xAA,
+ 0x00, 0x0C, 0xFF, 0xF7, 0x00, 0xC1, 0xFF, 0x33, 0xFD, 0xEC, 0x47,
+ 0x51, 0x0B, 0xAF, 0xF9, 0x1D, 0x04, 0x6B, 0xFD, 0x6E, 0x01, 0x60,
+ 0xFF, 0x29, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2C, 0x00, 0x54, 0xFF,
+ 0x8C, 0x01, 0x29, 0xFD, 0xA7, 0x04, 0x8F, 0xF8, 0x64, 0x0E, 0x02,
+ 0x47, 0x22, 0xFB, 0xC9, 0x00, 0x64, 0x00, 0x5B, 0xFF, 0x84, 0x00,
+ 0xBC, 0xFF, 0x10, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xE5,
+ 0x01, 0x33, 0xFC, 0xFF, 0x06, 0xC4, 0xF2, 0xB0, 0x23, 0x3B, 0x3B,
+ 0xAD, 0xF2, 0xBF, 0x05, 0x66, 0xFD, 0x0E, 0x01, 0xAC, 0xFF, 0x0E,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x19, 0x00, 0x8D, 0xFF, 0x4B, 0x01,
+ 0x01, 0xFD, 0x51, 0x06, 0xFB, 0xF1, 0xDF, 0x37, 0xF0, 0x27, 0x1C,
+ 0xF2, 0x24, 0x07, 0x34, 0xFC, 0xDC, 0x01, 0x3F, 0xFF, 0x34, 0x00,
+ 0xFD, 0xFF, 0x0C, 0x00, 0xCE, 0xFF, 0x54, 0x00, 0xBD, 0xFF, 0xB2,
+ 0xFF, 0x01, 0x02, 0xCF, 0xF8, 0x7D, 0x45, 0x6B, 0x12, 0x30, 0xF7,
+ 0x48, 0x05, 0xDD, 0xFC, 0xAD, 0x01, 0x48, 0xFF, 0x30, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x24, 0x00, 0x70, 0xFF, 0x45, 0x01, 0xC6, 0xFD,
+ 0x66, 0x03, 0x21, 0xFB, 0x9E, 0x07, 0xAA, 0x48, 0x12, 0x00, 0x64,
+ 0xFE, 0xB4, 0x01, 0xA6, 0xFE, 0xDB, 0x00, 0x9A, 0xFF, 0x19, 0x00,
+ 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xDE, 0x01, 0x5F, 0xFC, 0x70,
+ 0x06, 0x6C, 0xF4, 0xFD, 0x1B, 0x7D, 0x40, 0xB7, 0xF4, 0x5D, 0x04,
+ 0x49, 0xFE, 0x88, 0x00, 0xEF, 0xFF, 0xF5, 0xFF, 0x04, 0x00, 0xFD,
+ 0xFF, 0x29, 0x00, 0x61, 0xFF, 0xA1, 0x01, 0x7E, 0xFC, 0xF9, 0x06,
+ 0x7C, 0xF1, 0x38, 0x31, 0x50, 0x2F, 0x82, 0xF1, 0x12, 0x07, 0x65,
+ 0xFC, 0xB2, 0x01, 0x57, 0xFF, 0x2C, 0x00, 0xFD, 0xFF, 0x06, 0x00,
+ 0xED, 0xFF, 0x04, 0x00, 0x60, 0x00, 0x90, 0xFE, 0xEB, 0x03, 0x71,
+ 0xF5, 0xB7, 0x41, 0xED, 0x19, 0xF5, 0xF4, 0x3B, 0x06, 0x73, 0xFC,
+ 0xD7, 0x01, 0x39, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1B, 0x00, 0x91,
+ 0xFF, 0xF2, 0x00, 0x76, 0xFE, 0x11, 0x02, 0xB6, 0xFD, 0x8F, 0x01,
+ 0xDE, 0x48, 0xE9, 0x05, 0xD4, 0xFB, 0x0C, 0x03, 0xF4, 0xFD, 0x2F,
+ 0x01, 0x79, 0xFF, 0x22, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x32, 0x00,
+ 0x43, 0xFF, 0xBA, 0x01, 0xBC, 0xFC, 0x90, 0x05, 0x8E, 0xF6, 0x68,
+ 0x14, 0x99, 0x44, 0xCD, 0xF7, 0x8F, 0x02, 0x60, 0xFF, 0xEA, 0xFF,
+ 0x3E, 0x00, 0xD7, 0xFF, 0x0A, 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x44,
+ 0xFF, 0xD4, 0x01, 0x3B, 0xFC, 0x2A, 0x07, 0xDF, 0xF1, 0xF6, 0x29,
+ 0x27, 0x36, 0xC1, 0xF1, 0x8B, 0x06, 0xD8, 0xFC, 0x66, 0x01, 0x80,
+ 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x07, 0x00, 0xBD, 0xFF,
+ 0xED, 0x00, 0x9E, 0xFD, 0x6C, 0x05, 0x1F, 0xF3, 0xC0, 0x3C, 0x9E,
+ 0x21, 0x28, 0xF3, 0xE2, 0x06, 0x3A, 0xFC, 0xE6, 0x01, 0x37, 0xFF,
+ 0x36, 0x00, 0xFD, 0xFF, 0x12, 0x00, 0xB3, 0xFF, 0x9B, 0x00, 0x2B,
+ 0xFF, 0xBD, 0x00, 0x2A, 0x00, 0x5E, 0xFC, 0x9A, 0x47, 0x82, 0x0C,
+ 0x3D, 0xF9, 0x54, 0x04, 0x50, 0xFD, 0x7A, 0x01, 0x5B, 0xFF, 0x2A,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x2B, 0x00, 0x59, 0xFF, 0x81, 0x01,
+ 0x42, 0xFD, 0x72, 0x04, 0xFF, 0xF8, 0x2D, 0x0D, 0x69, 0x47, 0xEB,
+ 0xFB, 0x63, 0x00, 0x9D, 0x00, 0x3C, 0xFF, 0x93, 0x00, 0xB6, 0xFF,
+ 0x12, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x37,
+ 0xFC, 0xED, 0x06, 0x03, 0xF3, 0x5B, 0x22, 0x37, 0x3C, 0xF4, 0xF2,
+ 0x8A, 0x05, 0x89, 0xFD, 0xF9, 0x00, 0xB7, 0xFF, 0x0A, 0x00, 0x01,
+ 0x00, 0xFE, 0xFF, 0x1C, 0x00, 0x84, 0xFF, 0x5C, 0x01, 0xE6, 0xFC,
+ 0x77, 0x06, 0xD4, 0xF1, 0xC6, 0x36, 0x3E, 0x29, 0xF3, 0xF1, 0x29,
+ 0x07, 0x38, 0xFC, 0xD7, 0x01, 0x42, 0xFF, 0x33, 0x00, 0xFD, 0xFF,
+ 0x0B, 0x00, 0xD4, 0xFF, 0x46, 0x00, 0xDA, 0xFF, 0x7D, 0xFF, 0x5D,
+ 0x02, 0x26, 0xF8, 0xED, 0x44, 0xB1, 0x13, 0xC7, 0xF6, 0x77, 0x05,
+ 0xC8, 0xFC, 0xB6, 0x01, 0x45, 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0x22, 0x00, 0x76, 0xFF, 0x37, 0x01, 0xE4, 0xFD, 0x2C, 0x03,
+ 0x94, 0xFB, 0x83, 0x06, 0xCE, 0x48, 0x05, 0x01, 0xF5, 0xFD, 0xF0,
+ 0x01, 0x87, 0xFE, 0xEA, 0x00, 0x94, 0xFF, 0x1A, 0x00, 0xFE, 0xFF,
+ 0x35, 0x00, 0x38, 0xFF, 0xD9, 0x01, 0x6C, 0xFC, 0x4F, 0x06, 0xC3,
+ 0xF4, 0xA9, 0x1A, 0x49, 0x41, 0x2C, 0xF5, 0x15, 0x04, 0x76, 0xFE,
+ 0x6E, 0x00, 0xFC, 0xFF, 0xF0, 0xFF, 0x05, 0x00, 0xFD, 0xFF, 0x2B,
+ 0x00, 0x5A, 0xFF, 0xAC, 0x01, 0x6E, 0xFC, 0x0A, 0x07, 0x7E, 0xF1,
+ 0xFF, 0x2F, 0x8A, 0x30, 0x7D, 0xF1, 0x03, 0x07, 0x75, 0xFC, 0xA7,
+ 0x01, 0x5D, 0xFF, 0x2A, 0x00, 0xFD, 0xFF, 0x05, 0x00, 0xF2, 0xFF,
+ 0xF7, 0xFF, 0x7A, 0x00, 0x62, 0xFE, 0x35, 0x04, 0xF7, 0xF4, 0xEF,
+ 0x40, 0x40, 0x1B, 0x9C, 0xF4, 0x5E, 0x06, 0x66, 0xFC, 0xDB, 0x01,
+ 0x38, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x1A, 0x00, 0x96, 0xFF, 0xE3,
+ 0x00, 0x95, 0xFE, 0xD5, 0x01, 0x26, 0xFE, 0x98, 0x00, 0xBF, 0x48,
+ 0x00, 0x07, 0x61, 0xFB, 0x46, 0x03, 0xD6, 0xFD, 0x3D, 0x01, 0x73,
+ 0xFF, 0x23, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x31, 0x00, 0x46, 0xFF,
+ 0xB2, 0x01, 0xD1, 0xFC, 0x62, 0x05, 0xF6, 0xF6, 0x20, 0x13, 0x2E,
+ 0x45, 0x70, 0xF8, 0x34, 0x02, 0x94, 0xFF, 0xCD, 0xFF, 0x4C, 0x00,
+ 0xD2, 0xFF, 0x0B, 0x00, 0xFD, 0xFF, 0x33, 0x00, 0x41, 0xFF, 0xDA,
+ 0x01, 0x36, 0xFC, 0x27, 0x07, 0x05, 0xF2, 0xAA, 0x28, 0x44, 0x37,
+ 0xE4, 0xF1, 0x67, 0x06, 0xF2, 0xFC, 0x55, 0x01, 0x88, 0xFF, 0x1B,
+ 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x0B, 0x00, 0xB2, 0xFF, 0x02, 0x01,
+ 0x7A, 0xFD, 0xA2, 0x05, 0xD4, 0xF2, 0xC7, 0x3B, 0xF2, 0x22, 0xE7,
+ 0xF2, 0xF5, 0x06, 0x35, 0xFC, 0xE6, 0x01, 0x38, 0xFF, 0x36, 0x00,
+ 0xFD, 0xFF, 0x11, 0x00, 0xB9, 0xFF, 0x8C, 0x00, 0x4A, 0xFF, 0x83,
+ 0x00, 0x91, 0x00, 0x91, 0xFB, 0x3D, 0x47, 0xB7, 0x0D, 0xCD, 0xF8,
+ 0x89, 0x04, 0x36, 0xFD, 0x86, 0x01, 0x57, 0xFF, 0x2B, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x2A, 0x00, 0x5D, 0xFF, 0x75, 0x01, 0x5C, 0xFD,
+ 0x3C, 0x04, 0x70, 0xF9, 0xFA, 0x0B, 0xC0, 0x47, 0xBC, 0xFC, 0xFC,
+ 0xFF, 0xD6, 0x00, 0x1D, 0xFF, 0xA2, 0x00, 0xB0, 0xFF, 0x13, 0x00,
+ 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3C, 0xFC, 0xD8,
+ 0x06, 0x47, 0xF3, 0x06, 0x21, 0x2A, 0x3D, 0x44, 0xF3, 0x52, 0x05,
+ 0xAE, 0xFD, 0xE3, 0x00, 0xC2, 0xFF, 0x06, 0x00, 0x01, 0x00, 0xFE,
+ 0xFF, 0x1F, 0x00, 0x7C, 0xFF, 0x6D, 0x01, 0xCD, 0xFC, 0x99, 0x06,
+ 0xB4, 0xF1, 0xA6, 0x35, 0x89, 0x2A, 0xD0, 0xF1, 0x2B, 0x07, 0x3E,
+ 0xFC, 0xD1, 0x01, 0x46, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x0A, 0x00,
+ 0xD9, 0xFF, 0x37, 0x00, 0xF7, 0xFF, 0x49, 0xFF, 0xB6, 0x02, 0x86,
+ 0xF7, 0x53, 0x44, 0xFB, 0x14, 0x61, 0xF6, 0xA4, 0x05, 0xB4, 0xFC,
+ 0xBE, 0x01, 0x42, 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x21,
+ 0x00, 0x7B, 0xFF, 0x29, 0x01, 0x01, 0xFE, 0xF1, 0x02, 0x07, 0xFC,
+ 0x6E, 0x05, 0xE6, 0x48, 0xFF, 0x01, 0x84, 0xFD, 0x2C, 0x02, 0x68,
+ 0xFE, 0xF9, 0x00, 0x8E, 0xFF, 0x1C, 0x00, 0xFE, 0xFF, 0x35, 0x00,
+ 0x3A, 0xFF, 0xD4, 0x01, 0x7A, 0xFC, 0x2B, 0x06, 0x1E, 0xF5, 0x56,
+ 0x19, 0x0C, 0x42, 0xAA, 0xF5, 0xC9, 0x03, 0xA4, 0xFE, 0x54, 0x00,
+ 0x09, 0x00, 0xEB, 0xFF, 0x06, 0x00, 0xFD, 0xFF, 0x2D, 0x00, 0x55,
+ 0xFF, 0xB6, 0x01, 0x5F, 0xFC, 0x17, 0x07, 0x87, 0xF1, 0xC2, 0x2E,
+ 0xC0, 0x31, 0x7E, 0xF1, 0xF1, 0x06, 0x86, 0xFC, 0x9B, 0x01, 0x63,
+ 0xFF, 0x28, 0x00, 0xFD, 0xFF, 0x04, 0x00, 0xF7, 0xFF, 0xEA, 0xFF,
+ 0x93, 0x00, 0x36, 0xFE, 0x7D, 0x04, 0x85, 0xF4, 0x1F, 0x40, 0x94,
+ 0x1C, 0x47, 0xF4, 0x7E, 0x06, 0x5A, 0xFC, 0xDF, 0x01, 0x37, 0xFF,
+ 0x36, 0x00, 0xFE, 0xFF, 0x18, 0x00, 0x9C, 0xFF, 0xD4, 0x00, 0xB4,
+ 0xFE, 0x9A, 0x01, 0x95, 0xFE, 0xA8, 0xFF, 0x98, 0x48, 0x1D, 0x08,
+ 0xEE, 0xFA, 0x80, 0x03, 0xB9, 0xFD, 0x4B, 0x01, 0x6E, 0xFF, 0x25,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x30, 0x00, 0x4A, 0xFF, 0xA9, 0x01,
+ 0xE7, 0xFC, 0x33, 0x05, 0x60, 0xF7, 0xDA, 0x11, 0xB8, 0x45, 0x1C,
+ 0xF9, 0xD8, 0x01, 0xCA, 0xFF, 0xAF, 0xFF, 0x5A, 0x00, 0xCC, 0xFF,
+ 0x0D, 0x00, 0xFD, 0xFF, 0x34, 0x00, 0x3E, 0xFF, 0xDE, 0x01, 0x33,
+ 0xFC, 0x21, 0x07, 0x30, 0xF2, 0x5C, 0x27, 0x5B, 0x38, 0x0F, 0xF2,
+ 0x40, 0x06, 0x0E, 0xFD, 0x43, 0x01, 0x91, 0xFF, 0x18, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x0F, 0x00, 0xA8, 0xFF, 0x17, 0x01, 0x57, 0xFD,
+ 0xD6, 0x05, 0x90, 0xF2, 0xC8, 0x3A, 0x46, 0x24, 0xAA, 0xF2, 0x06,
+ 0x07, 0x32, 0xFC, 0xE5, 0x01, 0x39, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0x10, 0x00, 0xBE, 0xFF, 0x7D, 0x00, 0x69, 0xFF, 0x4B, 0x00, 0xF6,
+ 0x00, 0xCB, 0xFA, 0xD3, 0x46, 0xF0, 0x0E, 0x5E, 0xF8, 0xBE, 0x04,
+ 0x1E, 0xFD, 0x91, 0x01, 0x52, 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0x28, 0x00, 0x62, 0xFF, 0x69, 0x01, 0x77, 0xFD, 0x04, 0x04,
+ 0xE2, 0xF9, 0xCB, 0x0A, 0x0D, 0x48, 0x94, 0xFD, 0x92, 0xFF, 0x10,
+ 0x01, 0xFE, 0xFE, 0xB1, 0x00, 0xAA, 0xFF, 0x15, 0x00, 0xFD, 0xFF,
+ 0x36, 0x00, 0x36, 0xFF, 0xE5, 0x01, 0x43, 0xFC, 0xC0, 0x06, 0x8F,
+ 0xF3, 0xB1, 0x1F, 0x18, 0x3E, 0x9B, 0xF3, 0x16, 0x05, 0xD5, 0xFD,
+ 0xCC, 0x00, 0xCE, 0xFF, 0x01, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x22,
+ 0x00, 0x74, 0xFF, 0x7C, 0x01, 0xB5, 0xFC, 0xB8, 0x06, 0x9C, 0xF1,
+ 0x81, 0x34, 0xD1, 0x2B, 0xB3, 0xF1, 0x29, 0x07, 0x46, 0xFC, 0xCA,
+ 0x01, 0x4A, 0xFF, 0x30, 0x00, 0xFD, 0xFF, 0x09, 0x00, 0xDF, 0xFF,
+ 0x29, 0x00, 0x14, 0x00, 0x16, 0xFF, 0x0C, 0x03, 0xEE, 0xF6, 0xB0,
+ 0x43, 0x47, 0x16, 0xFC, 0xF5, 0xCF, 0x05, 0xA1, 0xFC, 0xC6, 0x01,
+ 0x3F, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x1F, 0x00, 0x81,
+ 0xFF, 0x1B, 0x01, 0x20, 0xFE, 0xB6, 0x02, 0x7A, 0xFC, 0x5F, 0x04,
+ 0xF4, 0x48, 0xFF, 0x02, 0x13, 0xFD, 0x67, 0x02, 0x49, 0xFE, 0x07,
+ 0x01, 0x88, 0xFF, 0x1D, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3C, 0xFF,
+ 0xCF, 0x01, 0x8A, 0xFC, 0x05, 0x06, 0x7B, 0xF5, 0x06, 0x18, 0xC7,
+ 0x42, 0x2F, 0xF6, 0x7A, 0x03, 0xD4, 0xFE, 0x39, 0x00, 0x17, 0x00,
+ 0xE6, 0xFF, 0x07, 0x00, 0xFD, 0xFF, 0x2E, 0x00, 0x50, 0xFF, 0xC0,
+ 0x01, 0x53, 0xFC, 0x21, 0x07, 0x96, 0xF1, 0x82, 0x2D, 0xF2, 0x32,
+ 0x86, 0xF1, 0xDB, 0x06, 0x99, 0xFC, 0x8E, 0x01, 0x6A, 0xFF, 0x25,
+ 0x00, 0xFD, 0xFF, 0x03, 0x00, 0xFB, 0xFF, 0xDE, 0xFF, 0xAC, 0x00,
+ 0x0B, 0xFE, 0xC1, 0x04, 0x1B, 0xF4, 0x47, 0x3F, 0xEA, 0x1D, 0xF5,
+ 0xF3, 0x9C, 0x06, 0x4F, 0xFC, 0xE2, 0x01, 0x36, 0xFF, 0x36, 0x00,
+ 0xFE, 0xFF, 0x16, 0x00, 0xA2, 0xFF, 0xC5, 0x00, 0xD4, 0xFE, 0x5F,
+ 0x01, 0x03, 0xFF, 0xBF, 0xFE, 0x63, 0x48, 0x40, 0x09, 0x7B, 0xFA,
+ 0xB9, 0x03, 0x9D, 0xFD, 0x58, 0x01, 0x69, 0xFF, 0x26, 0x00, 0x00,
+ 0x00, 0xFF, 0xFF, 0x2E, 0x00, 0x4D, 0xFF, 0x9F, 0x01, 0xFE, 0xFC,
+ 0x02, 0x05, 0xCB, 0xF7, 0x98, 0x10, 0x39, 0x46, 0xD0, 0xF9, 0x78,
+ 0x01, 0x00, 0x00, 0x91, 0xFF, 0x69, 0x00, 0xC6, 0xFF, 0x0E, 0x00,
+ 0xFD, 0xFF, 0x35, 0x00, 0x3B, 0xFF, 0xE2, 0x01, 0x31, 0xFC, 0x17,
+ 0x07, 0x61, 0xF2, 0x0A, 0x26, 0x6A, 0x39, 0x41, 0xF2, 0x15, 0x06,
+ 0x2C, 0xFD, 0x31, 0x01, 0x9B, 0xFF, 0x14, 0x00, 0xFF, 0xFF, 0xFF,
+ 0xFF, 0x13, 0x00, 0x9E, 0xFF, 0x2B, 0x01, 0x37, 0xFD, 0x05, 0x06,
+ 0x54, 0xF2, 0xC2, 0x39, 0x99, 0x25, 0x73, 0xF2, 0x14, 0x07, 0x31,
+ 0xFC, 0xE3, 0x01, 0x3B, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x0E, 0x00,
+ 0xC4, 0xFF, 0x6E, 0x00, 0x87, 0xFF, 0x13, 0x00, 0x58, 0x01, 0x0D,
+ 0xFA, 0x61, 0x46, 0x2D, 0x10, 0xF0, 0xF7, 0xF1, 0x04, 0x05, 0xFD,
+ 0x9C, 0x01, 0x4E, 0xFF, 0x2E, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x27,
+ 0x00, 0x67, 0xFF, 0x5C, 0x01, 0x93, 0xFD, 0xCC, 0x03, 0x54, 0xFA,
+ 0xA2, 0x09, 0x4F, 0x48, 0x73, 0xFE, 0x27, 0xFF, 0x4B, 0x01, 0xDE,
+ 0xFE, 0xC0, 0x00, 0xA4, 0xFF, 0x16, 0x00, 0xFE, 0xFF, 0x36, 0x00,
+ 0x36, 0xFF, 0xE3, 0x01, 0x4C, 0xFC, 0xA6, 0x06, 0xDB, 0xF3, 0x5B,
+ 0x1E, 0xFC, 0x3E, 0xFA, 0xF3, 0xD7, 0x04, 0xFD, 0xFD, 0xB4, 0x00,
+ 0xD9, 0xFF, 0xFD, 0xFF, 0x03, 0x00, 0xFD, 0xFF, 0x25, 0x00, 0x6D,
+ 0xFF, 0x8A, 0x01, 0x9F, 0xFC, 0xD3, 0x06, 0x8A, 0xF1, 0x57, 0x33,
+ 0x17, 0x2D, 0x9C, 0xF1, 0x24, 0x07, 0x4F, 0xFC, 0xC3, 0x01, 0x4E,
+ 0xFF, 0x2F, 0x00, 0xFD, 0xFF, 0x08, 0x00, 0xE4, 0xFF, 0x1B, 0x00,
+ 0x30, 0x00, 0xE4, 0xFE, 0x5F, 0x03, 0x5E, 0xF6, 0x02, 0x43, 0x96,
+ 0x17, 0x9B, 0xF5, 0xF8, 0x05, 0x8F, 0xFC, 0xCC, 0x01, 0x3D, 0xFF,
+ 0x34, 0x00, 0xFE, 0xFF, 0x1E, 0x00, 0x86, 0xFF, 0x0C, 0x01, 0x3E,
+ 0xFE, 0x7B, 0x02, 0xED, 0xFC, 0x56, 0x03, 0xF5, 0x48, 0x06, 0x04,
+ 0xA1, 0xFC, 0xA3, 0x02, 0x2A, 0xFE, 0x16, 0x01, 0x83, 0xFF, 0x1F,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x3E, 0xFF, 0xC8, 0x01,
+ 0x9B, 0xFC, 0xDD, 0x05, 0xDC, 0xF5, 0xB6, 0x16, 0x77, 0x43, 0xBD,
+ 0xF6, 0x28, 0x03, 0x05, 0xFF, 0x1D, 0x00, 0x25, 0x00, 0xE1, 0xFF,
+ 0x08, 0x00, 0xFD, 0xFF, 0x30, 0x00, 0x4B, 0xFF, 0xC8, 0x01, 0x49,
+ 0xFC, 0x27, 0x07, 0xAB, 0xF1, 0x3E, 0x2C, 0x1E, 0x34, 0x95, 0xF1,
+ 0xC1, 0x06, 0xAE, 0xFC, 0x81, 0x01, 0x71, 0xFF, 0x23, 0x00, 0xFE,
+ 0xFF, 0x02, 0x00, 0x00, 0x00, 0xD2, 0xFF, 0xC4, 0x00, 0xE2, 0xFD,
+ 0x01, 0x05, 0xBA, 0xF3, 0x64, 0x3E, 0x3F, 0x1F, 0xA8, 0xF3, 0xB8,
+ 0x06, 0x46, 0xFC, 0xE5, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0x15, 0x00, 0xA8, 0xFF, 0xB6, 0x00, 0xF3, 0xFE, 0x24, 0x01, 0x6E,
+ 0xFF, 0xDE, 0xFD, 0x25, 0x48, 0x68, 0x0A, 0x08, 0xFA, 0xF2, 0x03,
+ 0x81, 0xFD, 0x65, 0x01, 0x64, 0xFF, 0x28, 0x00, 0x00, 0x00, 0xFF,
+ 0xFF, 0x2D, 0x00, 0x51, 0xFF, 0x95, 0x01, 0x15, 0xFD, 0xCF, 0x04,
+ 0x39, 0xF8, 0x59, 0x0F, 0xAF, 0x46, 0x8B, 0xFA, 0x17, 0x01, 0x38,
+ 0x00, 0x73, 0xFF, 0x78, 0x00, 0xC0, 0xFF, 0x0F, 0x00, 0xFD, 0xFF,
+ 0x36, 0x00, 0x39, 0xFF, 0xE4, 0x01, 0x32, 0xFC, 0x0B, 0x07, 0x97,
+ 0xF2, 0xB8, 0x24, 0x71, 0x3A, 0x7B, 0xF2, 0xE6, 0x05, 0x4C, 0xFD,
+ 0x1D, 0x01, 0xA4, 0xFF, 0x11, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x16,
+ 0x00, 0x94, 0xFF, 0x3D, 0x01, 0x18, 0xFD, 0x32, 0x06, 0x1F, 0xF2,
+ 0xB5, 0x38, 0xEB, 0x26, 0x40, 0xF2, 0x1E, 0x07, 0x32, 0xFC, 0xDF,
+ 0x01, 0x3D, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x0D, 0x00, 0xCA, 0xFF,
+ 0x5F, 0x00, 0xA5, 0xFF, 0xDC, 0xFF, 0xB8, 0x01, 0x57, 0xF9, 0xE5,
+ 0x45, 0x6E, 0x11, 0x83, 0xF7, 0x23, 0x05, 0xEE, 0xFC, 0xA6, 0x01,
+ 0x4B, 0xFF, 0x2F, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x25, 0x00, 0x6C,
+ 0xFF, 0x4F, 0x01, 0xB0, 0xFD, 0x93, 0x03, 0xC7, 0xFA, 0x7D, 0x08,
+ 0x86, 0x48, 0x5A, 0xFF, 0xBA, 0xFE, 0x86, 0x01, 0xBF, 0xFE, 0xCF,
+ 0x00, 0x9E, 0xFF, 0x17, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF,
+ 0xE0, 0x01, 0x56, 0xFC, 0x89, 0x06, 0x2B, 0xF4, 0x06, 0x1D, 0xD7,
+ 0x3F, 0x61, 0xF4, 0x94, 0x04, 0x27, 0xFE, 0x9C, 0x00, 0xE6, 0xFF,
+ 0xF8, 0xFF, 0x04, 0x00, 0xFD, 0xFF, 0x27, 0x00, 0x66, 0xFF, 0x97,
+ 0x01, 0x8C, 0xFC, 0xEA, 0x06, 0x80, 0xF1, 0x26, 0x32, 0x58, 0x2E,
+ 0x8B, 0xF1, 0x1B, 0x07, 0x5B, 0xFC, 0xBA, 0x01, 0x53, 0xFF, 0x2D,
+ 0x00, 0xFD, 0xFF, 0x07, 0x00, 0xE9, 0xFF, 0x0E, 0x00, 0x4B, 0x00,
+ 0xB4, 0xFE, 0xAF, 0x03, 0xD5, 0xF5, 0x4D, 0x42, 0xE6, 0x18, 0x3C,
+ 0xF5, 0x1F, 0x06, 0x7F, 0xFC, 0xD3, 0x01, 0x3B, 0xFF, 0x35, 0x00,
+ 0xFE, 0xFF, 0x1C, 0x00, 0x8C, 0xFF, 0xFE, 0x00, 0x5D, 0xFE, 0x3F,
+ 0x02, 0x5E, 0xFD, 0x54, 0x02, 0xEC, 0x48, 0x13, 0x05, 0x2E, 0xFC,
+ 0xDE, 0x02, 0x0C, 0xFE, 0x24, 0x01, 0x7D, 0xFF, 0x20, 0x00, 0x00,
+ 0x00, 0xFF, 0xFF, 0x32, 0x00, 0x41, 0xFF, 0xC1, 0x01, 0xAD, 0xFC,
+ 0xB2, 0x05, 0x3F, 0xF6, 0x69, 0x15, 0x1F, 0x44, 0x53, 0xF7, 0xD3,
+ 0x02, 0x38, 0xFF, 0x01, 0x00, 0x33, 0x00, 0xDB, 0xFF, 0x09, 0x00,
+ 0xFD, 0xFF, 0x31, 0x00, 0x47, 0xFF, 0xCF, 0x01, 0x40, 0xFC, 0x2A,
+ 0x07, 0xC6, 0xF1, 0xF7, 0x2A, 0x46, 0x35, 0xAB, 0xF1, 0xA4, 0x06,
+ 0xC4, 0xFC, 0x72, 0x01, 0x79, 0xFF, 0x20, 0x00, 0xFE, 0xFF, 0x02,
+ 0x00, 0x04, 0x00, 0xC6, 0xFF, 0xDB, 0x00, 0xBB, 0xFD, 0x3E, 0x05,
+ 0x60, 0xF3, 0x7B, 0x3D, 0x94, 0x20, 0x5E, 0xF3, 0xD0, 0x06, 0x3E,
+ 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x14, 0x00,
+ 0xAE, 0xFF, 0xA7, 0x00, 0x12, 0xFF, 0xEA, 0x00, 0xD9, 0xFF, 0x03,
+ 0xFD, 0xDC, 0x47, 0x95, 0x0B, 0x96, 0xF9, 0x29, 0x04, 0x65, 0xFD,
+ 0x71, 0x01, 0x5F, 0xFF, 0x29, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2C,
+ 0x00, 0x55, 0xFF, 0x8A, 0x01, 0x2E, 0xFD, 0x9B, 0x04, 0xA8, 0xF8,
+ 0x1F, 0x0E, 0x1A, 0x47, 0x4E, 0xFB, 0xB3, 0x00, 0x70, 0x00, 0x54,
+ 0xFF, 0x87, 0x00, 0xBB, 0xFF, 0x11, 0x00, 0xFD, 0xFF, 0x36, 0x00,
+ 0x38, 0xFF, 0xE6, 0x01, 0x34, 0xFC, 0xFB, 0x06, 0xD2, 0xF2, 0x64,
+ 0x23, 0x73, 0x3B, 0xBC, 0xF2, 0xB4, 0x05, 0x6E, 0xFD, 0x09, 0x01,
+ 0xAF, 0xFF, 0x0D, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x1A, 0x00, 0x8B,
+ 0xFF, 0x4F, 0x01, 0xFB, 0xFC, 0x5A, 0x06, 0xF2, 0xF1, 0xA0, 0x37,
+ 0x3A, 0x28, 0x13, 0xF2, 0x25, 0x07, 0x35, 0xFC, 0xDB, 0x01, 0x40,
+ 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x0C, 0x00, 0xD0, 0xFF, 0x51, 0x00,
+ 0xC3, 0xFF, 0xA6, 0xFF, 0x16, 0x02, 0xA9, 0xF8, 0x5C, 0x45, 0xB2,
+ 0x12, 0x19, 0xF7, 0x52, 0x05, 0xD8, 0xFC, 0xAF, 0x01, 0x47, 0xFF,
+ 0x30, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x24, 0x00, 0x71, 0xFF, 0x42,
+ 0x01, 0xCD, 0xFD, 0x59, 0x03, 0x3B, 0xFB, 0x5E, 0x07, 0xB3, 0x48,
+ 0x48, 0x00, 0x4B, 0xFE, 0xC2, 0x01, 0x9F, 0xFE, 0xDE, 0x00, 0x98,
+ 0xFF, 0x19, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xDD, 0x01,
+ 0x62, 0xFC, 0x69, 0x06, 0x7F, 0xF4, 0xB2, 0x1B, 0xAB, 0x40, 0xD0,
+ 0xF4, 0x4E, 0x04, 0x53, 0xFE, 0x83, 0x00, 0xF2, 0xFF, 0xF4, 0xFF,
+ 0x05, 0x00, 0xFD, 0xFF, 0x29, 0x00, 0x5F, 0xFF, 0xA3, 0x01, 0x7A,
+ 0xFC, 0xFD, 0x06, 0x7C, 0xF1, 0xF2, 0x30, 0x96, 0x2F, 0x80, 0xF1,
+ 0x0F, 0x07, 0x69, 0xFC, 0xB0, 0x01, 0x59, 0xFF, 0x2B, 0x00, 0xFD,
+ 0xFF, 0x06, 0x00, 0xEE, 0xFF, 0x01, 0x00, 0x66, 0x00, 0x85, 0xFE,
+ 0xFC, 0x03, 0x55, 0xF5, 0x8C, 0x41, 0x38, 0x1A, 0xE1, 0xF4, 0x43,
+ 0x06, 0x70, 0xFC, 0xD8, 0x01, 0x39, 0xFF, 0x35, 0x00, 0xFE, 0xFF,
+ 0x1B, 0x00, 0x92, 0xFF, 0xEF, 0x00, 0x7D, 0xFE, 0x04, 0x02, 0xCF,
+ 0xFD, 0x58, 0x01, 0xD7, 0x48, 0x26, 0x06, 0xBB, 0xFB, 0x19, 0x03,
+ 0xED, 0xFD, 0x32, 0x01, 0x77, 0xFF, 0x22, 0x00, 0x00, 0x00, 0xFF,
+ 0xFF, 0x32, 0x00, 0x44, 0xFF, 0xB9, 0x01, 0xC1, 0xFC, 0x86, 0x05,
+ 0xA5, 0xF6, 0x1E, 0x14, 0xBA, 0x44, 0xF0, 0xF7, 0x7B, 0x02, 0x6B,
+ 0xFF, 0xE4, 0xFF, 0x41, 0x00, 0xD6, 0xFF, 0x0B, 0x00, 0xFD, 0xFF,
+ 0x33, 0x00, 0x43, 0xFF, 0xD5, 0x01, 0x3A, 0xFC, 0x2A, 0x07, 0xE7,
+ 0xF1, 0xAC, 0x29, 0x66, 0x36, 0xC9, 0xF1, 0x83, 0x06, 0xDD, 0xFC,
+ 0x62, 0x01, 0x81, 0xFF, 0x1D, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x08,
+ 0x00, 0xBB, 0xFF, 0xF1, 0x00, 0x96, 0xFD, 0x78, 0x05, 0x0E, 0xF3,
+ 0x8A, 0x3C, 0xEA, 0x21, 0x19, 0xF3, 0xE6, 0x06, 0x38, 0xFC, 0xE6,
+ 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x12, 0x00, 0xB4, 0xFF,
+ 0x98, 0x00, 0x32, 0xFF, 0xB0, 0x00, 0x41, 0x00, 0x30, 0xFC, 0x86,
+ 0x47, 0xC6, 0x0C, 0x24, 0xF9, 0x60, 0x04, 0x4B, 0xFD, 0x7D, 0x01,
+ 0x5A, 0xFF, 0x2A, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2B, 0x00, 0x5A,
+ 0xFF, 0x7E, 0x01, 0x48, 0xFD, 0x66, 0x04, 0x18, 0xF9, 0xE8, 0x0C,
+ 0x7C, 0x47, 0x19, 0xFC, 0x4D, 0x00, 0xA9, 0x00, 0x35, 0xFF, 0x96,
+ 0x00, 0xB5, 0xFF, 0x12, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF,
+ 0xE6, 0x01, 0x38, 0xFC, 0xE9, 0x06, 0x12, 0xF3, 0x10, 0x22, 0x6E,
+ 0x3C, 0x05, 0xF3, 0x7E, 0x05, 0x91, 0xFD, 0xF4, 0x00, 0xBA, 0xFF,
+ 0x09, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x1D, 0x00, 0x82, 0xFF, 0x60,
+ 0x01, 0xE0, 0xFC, 0x7F, 0x06, 0xCC, 0xF1, 0x85, 0x36, 0x87, 0x29,
+ 0xEB, 0xF1, 0x2A, 0x07, 0x39, 0xFC, 0xD6, 0x01, 0x43, 0xFF, 0x33,
+ 0x00, 0xFD, 0xFF, 0x0B, 0x00, 0xD5, 0xFF, 0x42, 0x00, 0xE1, 0xFF,
+ 0x71, 0xFF, 0x71, 0x02, 0x02, 0xF8, 0xCC, 0x44, 0xFA, 0x13, 0xB0,
+ 0xF6, 0x81, 0x05, 0xC3, 0xFC, 0xB8, 0x01, 0x44, 0xFF, 0x31, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0x22, 0x00, 0x77, 0xFF, 0x34, 0x01, 0xEA,
+ 0xFD, 0x1F, 0x03, 0xAE, 0xFB, 0x45, 0x06, 0xD5, 0x48, 0x3C, 0x01,
+ 0xDC, 0xFD, 0xFD, 0x01, 0x80, 0xFE, 0xED, 0x00, 0x93, 0xFF, 0x1B,
+ 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x39, 0xFF, 0xD8, 0x01, 0x6F, 0xFC,
+ 0x47, 0x06, 0xD7, 0xF4, 0x5D, 0x1A, 0x74, 0x41, 0x48, 0xF5, 0x04,
+ 0x04, 0x80, 0xFE, 0x69, 0x00, 0xFF, 0xFF, 0xEF, 0xFF, 0x05, 0x00,
+ 0xFD, 0xFF, 0x2B, 0x00, 0x59, 0xFF, 0xAE, 0x01, 0x6A, 0xFC, 0x0D,
+ 0x07, 0x80, 0xF1, 0xB8, 0x2F, 0xCF, 0x30, 0x7D, 0xF1, 0xFF, 0x06,
+ 0x78, 0xFC, 0xA5, 0x01, 0x5F, 0xFF, 0x29, 0x00, 0xFD, 0xFF, 0x05,
+ 0x00, 0xF3, 0xFF, 0xF4, 0xFF, 0x80, 0x00, 0x58, 0xFE, 0x46, 0x04,
+ 0xDD, 0xF4, 0xC3, 0x40, 0x8C, 0x1B, 0x89, 0xF4, 0x66, 0x06, 0x63,
+ 0xFC, 0xDC, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x19, 0x00,
+ 0x98, 0xFF, 0xE0, 0x00, 0x9C, 0xFE, 0xC8, 0x01, 0x3F, 0xFE, 0x62,
+ 0x00, 0xB8, 0x48, 0x3F, 0x07, 0x47, 0xFB, 0x53, 0x03, 0xD0, 0xFD,
+ 0x40, 0x01, 0x72, 0xFF, 0x23, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x30,
+ 0x00, 0x47, 0xFF, 0xB0, 0x01, 0xD6, 0xFC, 0x58, 0x05, 0x0D, 0xF7,
+ 0xD7, 0x12, 0x4E, 0x45, 0x96, 0xF8, 0x20, 0x02, 0xA0, 0xFF, 0xC7,
+ 0xFF, 0x4F, 0x00, 0xD0, 0xFF, 0x0C, 0x00, 0xFD, 0xFF, 0x34, 0x00,
+ 0x40, 0xFF, 0xDB, 0x01, 0x35, 0xFC, 0x26, 0x07, 0x0E, 0xF2, 0x60,
+ 0x28, 0x82, 0x37, 0xED, 0xF1, 0x5E, 0x06, 0xF8, 0xFC, 0x51, 0x01,
+ 0x8A, 0xFF, 0x1A, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x0C, 0x00, 0xB0,
+ 0xFF, 0x07, 0x01, 0x72, 0xFD, 0xAE, 0x05, 0xC4, 0xF2, 0x90, 0x3B,
+ 0x3F, 0x23, 0xD9, 0xF2, 0xF9, 0x06, 0x34, 0xFC, 0xE6, 0x01, 0x38,
+ 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x11, 0x00, 0xBA, 0xFF, 0x89, 0x00,
+ 0x51, 0xFF, 0x77, 0x00, 0xA7, 0x00, 0x64, 0xFB, 0x26, 0x47, 0xFC,
+ 0x0D, 0xB4, 0xF8, 0x95, 0x04, 0x31, 0xFD, 0x88, 0x01, 0x56, 0xFF,
+ 0x2C, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x29, 0x00, 0x5E, 0xFF, 0x72,
+ 0x01, 0x62, 0xFD, 0x2F, 0x04, 0x89, 0xF9, 0xB6, 0x0B, 0xD2, 0x47,
+ 0xEB, 0xFC, 0xE4, 0xFF, 0xE3, 0x00, 0x16, 0xFF, 0xA5, 0x00, 0xAF,
+ 0xFF, 0x13, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01,
+ 0x3E, 0xFC, 0xD3, 0x06, 0x56, 0xF3, 0xBA, 0x20, 0x61, 0x3D, 0x56,
+ 0xF3, 0x45, 0x05, 0xB7, 0xFD, 0xDE, 0x00, 0xC5, 0xFF, 0x05, 0x00,
+ 0x02, 0x00, 0xFE, 0xFF, 0x20, 0x00, 0x7A, 0xFF, 0x70, 0x01, 0xC7,
+ 0xFC, 0xA0, 0x06, 0xAE, 0xF1, 0x65, 0x35, 0xD1, 0x2A, 0xCA, 0xF1,
+ 0x2A, 0x07, 0x40, 0xFC, 0xD0, 0x01, 0x47, 0xFF, 0x32, 0x00, 0xFD,
+ 0xFF, 0x09, 0x00, 0xDB, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0x3D, 0xFF,
+ 0xC9, 0x02, 0x64, 0xF7, 0x2F, 0x44, 0x44, 0x15, 0x4A, 0xF6, 0xAD,
+ 0x05, 0xAF, 0xFC, 0xC0, 0x01, 0x41, 0xFF, 0x32, 0x00, 0xFF, 0xFF,
+ 0x00, 0x00, 0x21, 0x00, 0x7C, 0xFF, 0x26, 0x01, 0x08, 0xFE, 0xE4,
+ 0x02, 0x21, 0xFC, 0x31, 0x05, 0xEB, 0x48, 0x37, 0x02, 0x6B, 0xFD,
+ 0x39, 0x02, 0x61, 0xFE, 0xFC, 0x00, 0x8D, 0xFF, 0x1C, 0x00, 0xFE,
+ 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xD3, 0x01, 0x7D, 0xFC, 0x23, 0x06,
+ 0x32, 0xF5, 0x0C, 0x19, 0x38, 0x42, 0xC7, 0xF5, 0xB8, 0x03, 0xAF,
+ 0xFE, 0x4E, 0x00, 0x0C, 0x00, 0xEA, 0xFF, 0x06, 0x00, 0xFD, 0xFF,
+ 0x2D, 0x00, 0x54, 0xFF, 0xB8, 0x01, 0x5D, 0xFC, 0x1A, 0x07, 0x8A,
+ 0xF1, 0x7B, 0x2E, 0x04, 0x32, 0x7F, 0xF1, 0xEC, 0x06, 0x8A, 0xFC,
+ 0x98, 0x01, 0x65, 0xFF, 0x27, 0x00, 0xFD, 0xFF, 0x04, 0x00, 0xF8,
+ 0xFF, 0xE7, 0xFF, 0x99, 0x00, 0x2C, 0xFE, 0x8C, 0x04, 0x6D, 0xF4,
+ 0xF0, 0x3F, 0xE0, 0x1C, 0x34, 0xF4, 0x85, 0x06, 0x57, 0xFC, 0xE0,
+ 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x18, 0x00, 0x9E, 0xFF,
+ 0xD1, 0x00, 0xBB, 0xFE, 0x8D, 0x01, 0xAE, 0xFE, 0x74, 0xFF, 0x8D,
+ 0x48, 0x5D, 0x08, 0xD4, 0xFA, 0x8D, 0x03, 0xB3, 0xFD, 0x4E, 0x01,
+ 0x6D, 0xFF, 0x25, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2F, 0x00, 0x4A,
+ 0xFF, 0xA7, 0x01, 0xEC, 0xFC, 0x28, 0x05, 0x77, 0xF7, 0x92, 0x11,
+ 0xD7, 0x45, 0x43, 0xF9, 0xC3, 0x01, 0xD6, 0xFF, 0xA9, 0xFF, 0x5E,
+ 0x00, 0xCB, 0xFF, 0x0D, 0x00, 0xFD, 0xFF, 0x34, 0x00, 0x3D, 0xFF,
+ 0xDF, 0x01, 0x32, 0xFC, 0x1F, 0x07, 0x3B, 0xF2, 0x11, 0x27, 0x97,
+ 0x38, 0x19, 0xF2, 0x36, 0x06, 0x15, 0xFD, 0x3F, 0x01, 0x93, 0xFF,
+ 0x17, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x10, 0x00, 0xA6, 0xFF, 0x1B,
+ 0x01, 0x50, 0xFD, 0xE1, 0x05, 0x82, 0xF2, 0x8F, 0x3A, 0x92, 0x24,
+ 0x9D, 0xF2, 0x09, 0x07, 0x32, 0xFC, 0xE4, 0x01, 0x39, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0x0F, 0x00, 0xC0, 0xFF, 0x7A, 0x00, 0x70, 0xFF,
+ 0x3E, 0x00, 0x0C, 0x01, 0xA1, 0xFA, 0xBB, 0x46, 0x36, 0x0F, 0x45,
+ 0xF8, 0xC9, 0x04, 0x18, 0xFD, 0x93, 0x01, 0x52, 0xFF, 0x2D, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0x28, 0x00, 0x63, 0xFF, 0x66, 0x01, 0x7D,
+ 0xFD, 0xF8, 0x03, 0xFB, 0xF9, 0x89, 0x0A, 0x1D, 0x48, 0xC5, 0xFD,
+ 0x7A, 0xFF, 0x1D, 0x01, 0xF7, 0xFE, 0xB4, 0x00, 0xA9, 0xFF, 0x15,
+ 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE5, 0x01, 0x45, 0xFC,
+ 0xBB, 0x06, 0xA0, 0xF3, 0x64, 0x1F, 0x4A, 0x3E, 0xB0, 0xF3, 0x08,
+ 0x05, 0xDE, 0xFD, 0xC7, 0x00, 0xD0, 0xFF, 0x00, 0x00, 0x02, 0x00,
+ 0xFE, 0xFF, 0x23, 0x00, 0x72, 0xFF, 0x7F, 0x01, 0xB0, 0xFC, 0xBE,
+ 0x06, 0x97, 0xF1, 0x3F, 0x34, 0x19, 0x2C, 0xAD, 0xF1, 0x28, 0x07,
+ 0x48, 0xFC, 0xC9, 0x01, 0x4B, 0xFF, 0x30, 0x00, 0xFD, 0xFF, 0x08,
+ 0x00, 0xE0, 0xFF, 0x26, 0x00, 0x1A, 0x00, 0x0B, 0xFF, 0x1E, 0x03,
+ 0xCD, 0xF6, 0x89, 0x43, 0x91, 0x16, 0xE7, 0xF5, 0xD8, 0x05, 0x9D,
+ 0xFC, 0xC7, 0x01, 0x3E, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0x1F, 0x00, 0x82, 0xFF, 0x18, 0x01, 0x27, 0xFE, 0xA9, 0x02, 0x94,
+ 0xFC, 0x24, 0x04, 0xF5, 0x48, 0x39, 0x03, 0xF9, 0xFC, 0x74, 0x02,
+ 0x42, 0xFE, 0x0B, 0x01, 0x87, 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0x34,
+ 0x00, 0x3C, 0xFF, 0xCD, 0x01, 0x8E, 0xFC, 0xFC, 0x05, 0x90, 0xF5,
+ 0xBB, 0x17, 0xEE, 0x42, 0x4E, 0xF6, 0x68, 0x03, 0xDF, 0xFE, 0x33,
+ 0x00, 0x1A, 0x00, 0xE5, 0xFF, 0x07, 0x00, 0xFD, 0xFF, 0x2F, 0x00,
+ 0x4F, 0xFF, 0xC2, 0x01, 0x51, 0xFC, 0x23, 0x07, 0x9A, 0xF1, 0x3A,
+ 0x2D, 0x35, 0x33, 0x89, 0xF1, 0xD5, 0x06, 0x9D, 0xFC, 0x8B, 0x01,
+ 0x6C, 0xFF, 0x25, 0x00, 0xFD, 0xFF, 0x03, 0x00, 0xFC, 0xFF, 0xDB,
+ 0xFF, 0xB2, 0x00, 0x02, 0xFE, 0xCF, 0x04, 0x05, 0xF4, 0x16, 0x3F,
+ 0x36, 0x1E, 0xE4, 0xF3, 0xA3, 0x06, 0x4D, 0xFC, 0xE3, 0x01, 0x36,
+ 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x16, 0x00, 0xA4, 0xFF, 0xC2, 0x00,
+ 0xDB, 0xFE, 0x52, 0x01, 0x1B, 0xFF, 0x8D, 0xFE, 0x57, 0x48, 0x81,
+ 0x09, 0x61, 0xFA, 0xC6, 0x03, 0x96, 0xFD, 0x5B, 0x01, 0x68, 0xFF,
+ 0x26, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2E, 0x00, 0x4E, 0xFF, 0x9D,
+ 0x01, 0x03, 0xFD, 0xF7, 0x04, 0xE3, 0xF7, 0x51, 0x10, 0x55, 0x46,
+ 0xF9, 0xF9, 0x63, 0x01, 0x0D, 0x00, 0x8B, 0xFF, 0x6D, 0x00, 0xC5,
+ 0xFF, 0x0E, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3B, 0xFF, 0xE2, 0x01,
+ 0x31, 0xFC, 0x15, 0x07, 0x6D, 0xF2, 0xBF, 0x25, 0xA5, 0x39, 0x4D,
+ 0xF2, 0x0B, 0x06, 0x33, 0xFD, 0x2D, 0x01, 0x9D, 0xFF, 0x13, 0x00,
+ 0xFF, 0xFF, 0xFF, 0xFF, 0x14, 0x00, 0x9C, 0xFF, 0x2F, 0x01, 0x30,
+ 0xFD, 0x10, 0x06, 0x47, 0xF2, 0x87, 0x39, 0xE5, 0x25, 0x67, 0xF2,
+ 0x16, 0x07, 0x31, 0xFC, 0xE2, 0x01, 0x3B, 0xFF, 0x35, 0x00, 0xFD,
+ 0xFF, 0x0E, 0x00, 0xC6, 0xFF, 0x6B, 0x00, 0x8E, 0xFF, 0x06, 0x00,
+ 0x6E, 0x01, 0xE4, 0xF9, 0x48, 0x46, 0x75, 0x10, 0xD7, 0xF7, 0xFC,
+ 0x04, 0x00, 0xFD, 0x9E, 0x01, 0x4E, 0xFF, 0x2E, 0x00, 0xFF, 0xFF,
+ 0x00, 0x00, 0x26, 0x00, 0x68, 0xFF, 0x59, 0x01, 0x99, 0xFD, 0xC0,
+ 0x03, 0x6E, 0xFA, 0x61, 0x09, 0x5D, 0x48, 0xA6, 0xFE, 0x0F, 0xFF,
+ 0x58, 0x01, 0xD7, 0xFE, 0xC3, 0x00, 0xA3, 0xFF, 0x16, 0x00, 0xFE,
+ 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE3, 0x01, 0x4E, 0xFC, 0xA0, 0x06,
+ 0xED, 0xF3, 0x0F, 0x1E, 0x2D, 0x3F, 0x10, 0xF4, 0xC8, 0x04, 0x07,
+ 0xFE, 0xAF, 0x00, 0xDC, 0xFF, 0xFC, 0xFF, 0x03, 0x00, 0xFD, 0xFF,
+ 0x25, 0x00, 0x6B, 0xFF, 0x8D, 0x01, 0x9B, 0xFC, 0xD8, 0x06, 0x87,
+ 0xF1, 0x13, 0x33, 0x5E, 0x2D, 0x98, 0xF1, 0x22, 0x07, 0x52, 0xFC,
+ 0xC1, 0x01, 0x4F, 0xFF, 0x2F, 0x00, 0xFD, 0xFF, 0x07, 0x00, 0xE5,
+ 0xFF, 0x18, 0x00, 0x36, 0x00, 0xD9, 0xFE, 0x71, 0x03, 0x3F, 0xF6,
+ 0xDB, 0x42, 0xE0, 0x17, 0x86, 0xF5, 0x00, 0x06, 0x8C, 0xFC, 0xCE,
+ 0x01, 0x3C, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0x1D, 0x00, 0x88, 0xFF,
+ 0x09, 0x01, 0x45, 0xFE, 0x6E, 0x02, 0x06, 0xFD, 0x1C, 0x03, 0xF4,
+ 0x48, 0x41, 0x04, 0x87, 0xFC, 0xB0, 0x02, 0x23, 0xFE, 0x19, 0x01,
+ 0x81, 0xFF, 0x1F, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x3F,
+ 0xFF, 0xC6, 0x01, 0x9F, 0xFC, 0xD3, 0x05, 0xF1, 0xF5, 0x6C, 0x16,
+ 0x9E, 0x43, 0xDD, 0xF6, 0x15, 0x03, 0x10, 0xFF, 0x17, 0x00, 0x28,
+ 0x00, 0xDF, 0xFF, 0x09, 0x00, 0xFD, 0xFF, 0x30, 0x00, 0x4A, 0xFF,
+ 0xCA, 0x01, 0x47, 0xFC, 0x28, 0x07, 0xB0, 0xF1, 0xF5, 0x2B, 0x60,
+ 0x34, 0x9A, 0xF1, 0xBB, 0x06, 0xB3, 0xFC, 0x7D, 0x01, 0x73, 0xFF,
+ 0x22, 0x00, 0xFE, 0xFF, 0x02, 0x00, 0x01, 0x00, 0xCF, 0xFF, 0xC9,
+ 0x00, 0xDA, 0xFD, 0x0F, 0x05, 0xA5, 0xF3, 0x31, 0x3E, 0x8A, 0x1F,
+ 0x97, 0xF3, 0xBD, 0x06, 0x44, 0xFC, 0xE5, 0x01, 0x36, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0x15, 0x00, 0xAA, 0xFF, 0xB3, 0x00, 0xFA, 0xFE,
+ 0x17, 0x01, 0x86, 0xFF, 0xAC, 0xFD, 0x16, 0x48, 0xAA, 0x0A, 0xEE,
+ 0xF9, 0xFE, 0x03, 0x7A, 0xFD, 0x67, 0x01, 0x63, 0xFF, 0x28, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x2D, 0x00, 0x52, 0xFF, 0x92, 0x01, 0x1B,
+ 0xFD, 0xC4, 0x04, 0x51, 0xF8, 0x13, 0x0F, 0xC8, 0x46, 0xB6, 0xFA,
+ 0x01, 0x01, 0x44, 0x00, 0x6C, 0xFF, 0x7B, 0x00, 0xBF, 0xFF, 0x10,
+ 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x39, 0xFF, 0xE5, 0x01, 0x32, 0xFC,
+ 0x08, 0x07, 0xA4, 0xF2, 0x6D, 0x24, 0xAD, 0x3A, 0x88, 0xF2, 0xDB,
+ 0x05, 0x53, 0xFD, 0x19, 0x01, 0xA7, 0xFF, 0x10, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x17, 0x00, 0x92, 0xFF, 0x41, 0x01, 0x11, 0xFD, 0x3B,
+ 0x06, 0x14, 0xF2, 0x78, 0x38, 0x36, 0x27, 0x35, 0xF2, 0x20, 0x07,
+ 0x33, 0xFC, 0xDF, 0x01, 0x3E, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x0D,
+ 0x00, 0xCB, 0xFF, 0x5C, 0x00, 0xAC, 0xFF, 0xD0, 0xFF, 0xCD, 0x01,
+ 0x30, 0xF9, 0xC8, 0x45, 0xB6, 0x11, 0x6B, 0xF7, 0x2D, 0x05, 0xE9,
+ 0xFC, 0xA8, 0x01, 0x4A, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0x25, 0x00, 0x6D, 0xFF, 0x4C, 0x01, 0xB6, 0xFD, 0x86, 0x03, 0xE1,
+ 0xFA, 0x3D, 0x08, 0x92, 0x48, 0x8E, 0xFF, 0xA1, 0xFE, 0x93, 0x01,
+ 0xB8, 0xFE, 0xD3, 0x00, 0x9D, 0xFF, 0x18, 0x00, 0xFE, 0xFF, 0x36,
+ 0x00, 0x37, 0xFF, 0xE0, 0x01, 0x58, 0xFC, 0x82, 0x06, 0x3E, 0xF4,
+ 0xBA, 0x1C, 0x07, 0x40, 0x79, 0xF4, 0x84, 0x04, 0x31, 0xFE, 0x96,
+ 0x00, 0xE8, 0xFF, 0xF7, 0xFF, 0x04, 0x00, 0xFD, 0xFF, 0x28, 0x00,
+ 0x64, 0xFF, 0x9A, 0x01, 0x88, 0xFC, 0xEE, 0x06, 0x7E, 0xF1, 0xE3,
+ 0x31, 0x9F, 0x2E, 0x88, 0xF1, 0x19, 0x07, 0x5E, 0xFC, 0xB7, 0x01,
+ 0x54, 0xFF, 0x2D, 0x00, 0xFD, 0xFF, 0x06, 0x00, 0xEA, 0xFF, 0x0B,
+ 0x00, 0x51, 0x00, 0xAA, 0xFE, 0xC0, 0x03, 0xB8, 0xF5, 0x21, 0x42,
+ 0x31, 0x19, 0x28, 0xF5, 0x27, 0x06, 0x7C, 0xFC, 0xD4, 0x01, 0x3A,
+ 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1C, 0x00, 0x8D, 0xFF, 0xFA, 0x00,
+ 0x64, 0xFE, 0x32, 0x02, 0x78, 0xFD, 0x1B, 0x02, 0xEA, 0x48, 0x50,
+ 0x05, 0x14, 0xFC, 0xEB, 0x02, 0x05, 0xFE, 0x27, 0x01, 0x7C, 0xFF,
+ 0x21, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x32, 0x00, 0x41, 0xFF, 0xBF,
+ 0x01, 0xB2, 0xFC, 0xA9, 0x05, 0x55, 0xF6, 0x20, 0x15, 0x42, 0x44,
+ 0x75, 0xF7, 0xBF, 0x02, 0x43, 0xFF, 0xFA, 0xFF, 0x36, 0x00, 0xDA,
+ 0xFF, 0x0A, 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x46, 0xFF, 0xD1, 0x01,
+ 0x3F, 0xFC, 0x2B, 0x07, 0xCD, 0xF1, 0xAE, 0x2A, 0x86, 0x35, 0xB1,
+ 0xF1, 0x9D, 0x06, 0xCA, 0xFC, 0x6E, 0x01, 0x7B, 0xFF, 0x20, 0x00,
+ 0xFE, 0xFF, 0x02, 0x00, 0x05, 0x00, 0xC3, 0xFF, 0xE0, 0x00, 0xB3,
+ 0xFD, 0x4B, 0x05, 0x4D, 0xF3, 0x45, 0x3D, 0xE0, 0x20, 0x4F, 0xF3,
+ 0xD5, 0x06, 0x3D, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0x13, 0x00, 0xAF, 0xFF, 0xA4, 0x00, 0x19, 0xFF, 0xDD, 0x00,
+ 0xF0, 0xFF, 0xD4, 0xFC, 0xC9, 0x47, 0xD8, 0x0B, 0x7C, 0xF9, 0x35,
+ 0x04, 0x5F, 0xFD, 0x74, 0x01, 0x5E, 0xFF, 0x29, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x2C, 0x00, 0x56, 0xFF, 0x87, 0x01, 0x34, 0xFD, 0x8F,
+ 0x04, 0xC0, 0xF8, 0xD9, 0x0D, 0x31, 0x47, 0x7B, 0xFB, 0x9C, 0x00,
+ 0x7D, 0x00, 0x4D, 0xFF, 0x8A, 0x00, 0xB9, 0xFF, 0x11, 0x00, 0xFD,
+ 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xE6, 0x01, 0x35, 0xFC, 0xF7, 0x06,
+ 0xE0, 0xF2, 0x18, 0x23, 0xAB, 0x3B, 0xCC, 0xF2, 0xA8, 0x05, 0x76,
+ 0xFD, 0x04, 0x01, 0xB1, 0xFF, 0x0C, 0x00, 0x00, 0x00, 0xFE, 0xFF,
+ 0x1A, 0x00, 0x89, 0xFF, 0x53, 0x01, 0xF5, 0xFC, 0x63, 0x06, 0xE9,
+ 0xF1, 0x63, 0x37, 0x85, 0x28, 0x09, 0xF2, 0x27, 0x07, 0x35, 0xFC,
+ 0xDA, 0x01, 0x40, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x0C, 0x00, 0xD1,
+ 0xFF, 0x4E, 0x00, 0xCA, 0xFF, 0x9A, 0xFF, 0x2A, 0x02, 0x83, 0xF8,
+ 0x3F, 0x45, 0xFB, 0x12, 0x01, 0xF7, 0x5D, 0x05, 0xD3, 0xFC, 0xB1,
+ 0x01, 0x46, 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x23, 0x00,
+ 0x73, 0xFF, 0x3F, 0x01, 0xD3, 0xFD, 0x4C, 0x03, 0x54, 0xFB, 0x1F,
+ 0x07, 0xBB, 0x48, 0x7D, 0x00, 0x33, 0xFE, 0xCF, 0x01, 0x98, 0xFE,
+ 0xE2, 0x00, 0x97, 0xFF, 0x19, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x38,
+ 0xFF, 0xDC, 0x01, 0x64, 0xFC, 0x62, 0x06, 0x93, 0xF4, 0x66, 0x1B,
+ 0xD9, 0x40, 0xEA, 0xF4, 0x3E, 0x04, 0x5D, 0xFE, 0x7D, 0x00, 0xF5,
+ 0xFF, 0xF3, 0xFF, 0x05, 0x00, 0xFD, 0xFF, 0x2A, 0x00, 0x5E, 0xFF,
+ 0xA6, 0x01, 0x76, 0xFC, 0x01, 0x07, 0x7D, 0xF1, 0xAD, 0x30, 0xDC,
+ 0x2F, 0x7F, 0xF1, 0x0C, 0x07, 0x6C, 0xFC, 0xAD, 0x01, 0x5A, 0xFF,
+ 0x2B, 0x00, 0xFD, 0xFF, 0x05, 0x00, 0xEF, 0xFF, 0xFE, 0xFF, 0x6C,
+ 0x00, 0x7B, 0xFE, 0x0C, 0x04, 0x3A, 0xF5, 0x5F, 0x41, 0x83, 0x1A,
+ 0xCD, 0xF4, 0x4B, 0x06, 0x6D, 0xFC, 0xD9, 0x01, 0x39, 0xFF, 0x35,
+ 0x00, 0xFE, 0xFF, 0x1A, 0x00, 0x93, 0xFF, 0xEC, 0x00, 0x83, 0xFE,
+ 0xF7, 0x01, 0xE8, 0xFD, 0x21, 0x01, 0xD2, 0x48, 0x64, 0x06, 0xA1,
+ 0xFB, 0x26, 0x03, 0xE7, 0xFD, 0x35, 0x01, 0x76, 0xFF, 0x22, 0x00,
+ 0x00, 0x00, 0xFF, 0xFF, 0x31, 0x00, 0x44, 0xFF, 0xB7, 0x01, 0xC5,
+ 0xFC, 0x7C, 0x05, 0xBC, 0xF6, 0xD5, 0x13, 0xDC, 0x44, 0x14, 0xF8,
+ 0x67, 0x02, 0x77, 0xFF, 0xDD, 0xFF, 0x44, 0x00, 0xD5, 0xFF, 0x0B,
+ 0x00, 0xFD, 0xFF, 0x33, 0x00, 0x42, 0xFF, 0xD7, 0x01, 0x39, 0xFC,
+ 0x29, 0x07, 0xEF, 0xF1, 0x62, 0x29, 0xA5, 0x36, 0xD0, 0xF1, 0x7B,
+ 0x06, 0xE3, 0xFC, 0x5E, 0x01, 0x83, 0xFF, 0x1D, 0x00, 0xFE, 0xFF,
+ 0x01, 0x00, 0x09, 0x00, 0xB8, 0xFF, 0xF6, 0x00, 0x8D, 0xFD, 0x84,
+ 0x05, 0xFD, 0xF2, 0x52, 0x3C, 0x35, 0x22, 0x0B, 0xF3, 0xEB, 0x06,
+ 0x37, 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x12,
+ 0x00, 0xB5, 0xFF, 0x94, 0x00, 0x39, 0xFF, 0xA3, 0x00, 0x58, 0x00,
+ 0x02, 0xFC, 0x73, 0x47, 0x0B, 0x0D, 0x0B, 0xF9, 0x6C, 0x04, 0x45,
+ 0xFD, 0x80, 0x01, 0x59, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x2A, 0x00, 0x5B, 0xFF, 0x7C, 0x01, 0x4E, 0xFD, 0x5A, 0x04, 0x31,
+ 0xF9, 0xA4, 0x0C, 0x90, 0x47, 0x47, 0xFC, 0x36, 0x00, 0xB6, 0x00,
+ 0x2E, 0xFF, 0x99, 0x00, 0xB3, 0xFF, 0x12, 0x00, 0xFD, 0xFF, 0x36,
+ 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x39, 0xFC, 0xE4, 0x06, 0x21, 0xF3,
+ 0xC4, 0x21, 0xA5, 0x3C, 0x16, 0xF3, 0x72, 0x05, 0x9A, 0xFD, 0xEF,
+ 0x00, 0xBC, 0xFF, 0x08, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x1E, 0x00,
+ 0x80, 0xFF, 0x64, 0x01, 0xDA, 0xFC, 0x87, 0x06, 0xC5, 0xF1, 0x46,
+ 0x36, 0xD1, 0x29, 0xE3, 0xF1, 0x2A, 0x07, 0x3A, 0xFC, 0xD5, 0x01,
+ 0x44, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x0A, 0x00, 0xD6, 0xFF, 0x3F,
+ 0x00, 0xE7, 0xFF, 0x65, 0xFF, 0x85, 0x02, 0xDE, 0xF7, 0xA9, 0x44,
+ 0x43, 0x14, 0x99, 0xF6, 0x8B, 0x05, 0xBF, 0xFC, 0xBA, 0x01, 0x43,
+ 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x22, 0x00, 0x78, 0xFF,
+ 0x31, 0x01, 0xF1, 0xFD, 0x12, 0x03, 0xC7, 0xFB, 0x07, 0x06, 0xDB,
+ 0x48, 0x73, 0x01, 0xC3, 0xFD, 0x0A, 0x02, 0x79, 0xFE, 0xF1, 0x00,
+ 0x91, 0xFF, 0x1B, 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x39, 0xFF, 0xD7,
+ 0x01, 0x72, 0xFC, 0x3F, 0x06, 0xEB, 0xF4, 0x12, 0x1A, 0xA1, 0x41,
+ 0x63, 0xF5, 0xF3, 0x03, 0x8A, 0xFE, 0x63, 0x00, 0x02, 0x00, 0xEE,
+ 0xFF, 0x06, 0x00, 0xFD, 0xFF, 0x2C, 0x00, 0x58, 0xFF, 0xB1, 0x01,
+ 0x67, 0xFC, 0x10, 0x07, 0x81, 0xF1, 0x73, 0x2F, 0x15, 0x31, 0x7C,
+ 0xF1, 0xFB, 0x06, 0x7C, 0xFC, 0xA2, 0x01, 0x60, 0xFF, 0x29, 0x00,
+ 0xFD, 0xFF, 0x04, 0x00, 0xF4, 0xFF, 0xF1, 0xFF, 0x85, 0x00, 0x4E,
+ 0xFE, 0x56, 0x04, 0xC3, 0xF4, 0x95, 0x40, 0xD8, 0x1B, 0x76, 0xF4,
+ 0x6D, 0x06, 0x60, 0xFC, 0xDD, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE,
+ 0xFF, 0x19, 0x00, 0x99, 0xFF, 0xDD, 0x00, 0xA3, 0xFE, 0xBB, 0x01,
+ 0x58, 0xFE, 0x2D, 0x00, 0xAF, 0x48, 0x7E, 0x07, 0x2E, 0xFB, 0x60,
+ 0x03, 0xC9, 0xFD, 0x43, 0x01, 0x71, 0xFF, 0x24, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x30, 0x00, 0x48, 0xFF, 0xAE, 0x01, 0xDB, 0xFC, 0x4D,
+ 0x05, 0x24, 0xF7, 0x8E, 0x12, 0x6D, 0x45, 0xBC, 0xF8, 0x0C, 0x02,
+ 0xAC, 0xFF, 0xC0, 0xFF, 0x52, 0x00, 0xCF, 0xFF, 0x0C, 0x00, 0xFD,
+ 0xFF, 0x34, 0x00, 0x3F, 0xFF, 0xDC, 0x01, 0x34, 0xFC, 0x25, 0x07,
+ 0x18, 0xF2, 0x15, 0x28, 0xBF, 0x37, 0xF7, 0xF1, 0x56, 0x06, 0xFE,
+ 0xFC, 0x4D, 0x01, 0x8C, 0xFF, 0x19, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0x0D, 0x00, 0xAE, 0xFF, 0x0B, 0x01, 0x6A, 0xFD, 0xBA, 0x05, 0xB4,
+ 0xF2, 0x58, 0x3B, 0x8A, 0x23, 0xCB, 0xF2, 0xFD, 0x06, 0x34, 0xFC,
+ 0xE6, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x10, 0x00, 0xBB,
+ 0xFF, 0x85, 0x00, 0x58, 0xFF, 0x6A, 0x00, 0xBE, 0x00, 0x38, 0xFB,
+ 0x0F, 0x47, 0x42, 0x0E, 0x9B, 0xF8, 0xA1, 0x04, 0x2B, 0xFD, 0x8B,
+ 0x01, 0x55, 0xFF, 0x2C, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x29, 0x00,
+ 0x5F, 0xFF, 0x70, 0x01, 0x68, 0xFD, 0x23, 0x04, 0xA2, 0xF9, 0x73,
+ 0x0B, 0xE4, 0x47, 0x1B, 0xFD, 0xCD, 0xFF, 0xF0, 0x00, 0x0F, 0xFF,
+ 0xA9, 0x00, 0xAE, 0xFF, 0x14, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36,
+ 0xFF, 0xE6, 0x01, 0x3F, 0xFC, 0xCE, 0x06, 0x66, 0xF3, 0x6F, 0x20,
+ 0x96, 0x3D, 0x69, 0xF3, 0x38, 0x05, 0xBF, 0xFD, 0xD9, 0x00, 0xC7,
+ 0xFF, 0x04, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x20, 0x00, 0x78, 0xFF,
+ 0x74, 0x01, 0xC2, 0xFC, 0xA7, 0x06, 0xA8, 0xF1, 0x25, 0x35, 0x1B,
+ 0x2B, 0xC2, 0xF1, 0x2A, 0x07, 0x41, 0xFC, 0xCE, 0x01, 0x47, 0xFF,
+ 0x31, 0x00, 0xFD, 0xFF, 0x09, 0x00, 0xDC, 0xFF, 0x31, 0x00, 0x04,
+ 0x00, 0x32, 0xFF, 0xDC, 0x02, 0x42, 0xF7, 0x0B, 0x44, 0x8E, 0x15,
+ 0x34, 0xF6, 0xB7, 0x05, 0xAB, 0xFC, 0xC1, 0x01, 0x40, 0xFF, 0x33,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x20, 0x00, 0x7E, 0xFF, 0x23, 0x01,
+ 0x0F, 0xFE, 0xD7, 0x02, 0x3B, 0xFC, 0xF5, 0x04, 0xED, 0x48, 0x70,
+ 0x02, 0x52, 0xFD, 0x46, 0x02, 0x5A, 0xFE, 0xFF, 0x00, 0x8B, 0xFF,
+ 0x1C, 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x3B, 0xFF, 0xD2, 0x01, 0x81,
+ 0xFC, 0x1A, 0x06, 0x47, 0xF5, 0xC1, 0x18, 0x60, 0x42, 0xE4, 0xF5,
+ 0xA6, 0x03, 0xB9, 0xFE, 0x48, 0x00, 0x0F, 0x00, 0xE9, 0xFF, 0x07,
+ 0x00, 0xFD, 0xFF, 0x2E, 0x00, 0x53, 0xFF, 0xBB, 0x01, 0x5A, 0xFC,
+ 0x1C, 0x07, 0x8D, 0xF1, 0x34, 0x2E, 0x48, 0x32, 0x81, 0xF1, 0xE7,
+ 0x06, 0x8E, 0xFC, 0x96, 0x01, 0x66, 0xFF, 0x27, 0x00, 0xFD, 0xFF,
+ 0x04, 0x00, 0xF9, 0xFF, 0xE4, 0xFF, 0x9F, 0x00, 0x23, 0xFE, 0x9B,
+ 0x04, 0x55, 0xF4, 0xC0, 0x3F, 0x2C, 0x1D, 0x22, 0xF4, 0x8C, 0x06,
+ 0x55, 0xFC, 0xE1, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x17,
+ 0x00, 0x9F, 0xFF, 0xCE, 0x00, 0xC2, 0xFE, 0x80, 0x01, 0xC6, 0xFE,
+ 0x40, 0xFF, 0x81, 0x48, 0x9E, 0x08, 0xBA, 0xFA, 0x9A, 0x03, 0xAC,
+ 0xFD, 0x51, 0x01, 0x6C, 0xFF, 0x25, 0x00, 0x00, 0x00, 0xFF, 0xFF,
+ 0x2F, 0x00, 0x4B, 0xFF, 0xA4, 0x01, 0xF1, 0xFC, 0x1D, 0x05, 0x8F,
+ 0xF7, 0x4A, 0x11, 0xF2, 0x45, 0x6B, 0xF9, 0xAE, 0x01, 0xE2, 0xFF,
+ 0xA2, 0xFF, 0x61, 0x00, 0xC9, 0xFF, 0x0D, 0x00, 0xFD, 0xFF, 0x35,
+ 0x00, 0x3D, 0xFF, 0xE0, 0x01, 0x32, 0xFC, 0x1D, 0x07, 0x45, 0xF2,
+ 0xC6, 0x26, 0xD3, 0x38, 0x24, 0xF2, 0x2D, 0x06, 0x1B, 0xFD, 0x3B,
+ 0x01, 0x95, 0xFF, 0x16, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x11, 0x00,
+ 0xA3, 0xFF, 0x20, 0x01, 0x49, 0xFD, 0xEB, 0x05, 0x74, 0xF2, 0x54,
+ 0x3A, 0xDD, 0x24, 0x91, 0xF2, 0x0C, 0x07, 0x32, 0xFC, 0xE4, 0x01,
+ 0x3A, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x0F, 0x00, 0xC1, 0xFF, 0x76,
+ 0x00, 0x76, 0xFF, 0x32, 0x00, 0x22, 0x01, 0x76, 0xFA, 0xA3, 0x46,
+ 0x7D, 0x0F, 0x2C, 0xF8, 0xD5, 0x04, 0x13, 0xFD, 0x96, 0x01, 0x51,
+ 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x27, 0x00, 0x64, 0xFF,
+ 0x63, 0x01, 0x84, 0xFD, 0xEB, 0x03, 0x14, 0xFA, 0x47, 0x0A, 0x2C,
+ 0x48, 0xF6, 0xFD, 0x63, 0xFF, 0x2B, 0x01, 0xF0, 0xFE, 0xB8, 0x00,
+ 0xA8, 0xFF, 0x15, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE4,
+ 0x01, 0x47, 0xFC, 0xB5, 0x06, 0xB0, 0xF3, 0x19, 0x1F, 0x7E, 0x3E,
+ 0xC4, 0xF3, 0xFA, 0x04, 0xE7, 0xFD, 0xC1, 0x00, 0xD3, 0xFF, 0xFF,
+ 0xFF, 0x02, 0x00, 0xFE, 0xFF, 0x23, 0x00, 0x71, 0xFF, 0x82, 0x01,
+ 0xAB, 0xFC, 0xC4, 0x06, 0x93, 0xF1, 0xFD, 0x33, 0x62, 0x2C, 0xA8,
+ 0xF1, 0x27, 0x07, 0x4A, 0xFC, 0xC7, 0x01, 0x4C, 0xFF, 0x30, 0x00,
+ 0xFD, 0xFF, 0x08, 0x00, 0xE1, 0xFF, 0x23, 0x00, 0x20, 0x00, 0x00,
+ 0xFF, 0x31, 0x03, 0xAD, 0xF6, 0x65, 0x43, 0xDC, 0x16, 0xD1, 0xF5,
+ 0xE1, 0x05, 0x99, 0xFC, 0xC9, 0x01, 0x3E, 0xFF, 0x33, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x1F, 0x00, 0x83, 0xFF, 0x14, 0x01, 0x2D, 0xFE,
+ 0x9C, 0x02, 0xAD, 0xFC, 0xE9, 0x03, 0xF6, 0x48, 0x73, 0x03, 0xE0,
+ 0xFC, 0x82, 0x02, 0x3B, 0xFE, 0x0E, 0x01, 0x86, 0xFF, 0x1E, 0x00,
+ 0xFE, 0xFF, 0x34, 0x00, 0x3D, 0xFF, 0xCC, 0x01, 0x91, 0xFC, 0xF3,
+ 0x05, 0xA6, 0xF5, 0x70, 0x17, 0x17, 0x43, 0x6D, 0xF6, 0x56, 0x03,
+ 0xEA, 0xFE, 0x2D, 0x00, 0x1D, 0x00, 0xE4, 0xFF, 0x08, 0x00, 0xFD,
+ 0xFF, 0x2F, 0x00, 0x4E, 0xFF, 0xC3, 0x01, 0x4E, 0xFC, 0x24, 0x07,
+ 0x9E, 0xF1, 0xF2, 0x2C, 0x78, 0x33, 0x8C, 0xF1, 0xD0, 0x06, 0xA2,
+ 0xFC, 0x88, 0x01, 0x6D, 0xFF, 0x24, 0x00, 0xFD, 0xFF, 0x03, 0x00,
+ 0xFD, 0xFF, 0xD8, 0xFF, 0xB7, 0x00, 0xF9, 0xFD, 0xDE, 0x04, 0xEF,
+ 0xF3, 0xE4, 0x3E, 0x81, 0x1E, 0xD2, 0xF3, 0xA9, 0x06, 0x4B, 0xFC,
+ 0xE3, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x16, 0x00, 0xA5,
+ 0xFF, 0xBE, 0x00, 0xE2, 0xFE, 0x45, 0x01, 0x33, 0xFF, 0x5A, 0xFE,
+ 0x48, 0x48, 0xC3, 0x09, 0x47, 0xFA, 0xD2, 0x03, 0x90, 0xFD, 0x5E,
+ 0x01, 0x66, 0xFF, 0x27, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2E, 0x00,
+ 0x4F, 0xFF, 0x9A, 0x01, 0x08, 0xFD, 0xEB, 0x04, 0xFC, 0xF7, 0x0A,
+ 0x10, 0x70, 0x46, 0x22, 0xFA, 0x4D, 0x01, 0x19, 0x00, 0x84, 0xFF,
+ 0x70, 0x00, 0xC4, 0xFF, 0x0F, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3B,
+ 0xFF, 0xE3, 0x01, 0x31, 0xFC, 0x12, 0x07, 0x79, 0xF2, 0x73, 0x25,
+ 0xDF, 0x39, 0x5A, 0xF2, 0x00, 0x06, 0x3A, 0xFD, 0x28, 0x01, 0x9F,
+ 0xFF, 0x13, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x15, 0x00, 0x99, 0xFF,
+ 0x33, 0x01, 0x29, 0xFD, 0x1A, 0x06, 0x3B, 0xF2, 0x4B, 0x39, 0x30,
+ 0x26, 0x5B, 0xF2, 0x19, 0x07, 0x31, 0xFC, 0xE1, 0x01, 0x3C, 0xFF,
+ 0x35, 0x00, 0xFD, 0xFF, 0x0E, 0x00, 0xC7, 0xFF, 0x68, 0x00, 0x95,
+ 0xFF, 0xFA, 0xFF, 0x83, 0x01, 0xBB, 0xF9, 0x2B, 0x46, 0xBB, 0x10,
+ 0xBF, 0xF7, 0x07, 0x05, 0xFB, 0xFC, 0xA0, 0x01, 0x4D, 0xFF, 0x2F,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x26, 0x00, 0x69, 0xFF, 0x56, 0x01,
+ 0xA0, 0xFD, 0xB3, 0x03, 0x87, 0xFA, 0x1F, 0x09, 0x6A, 0x48, 0xD9,
+ 0xFE, 0xF6, 0xFE, 0x65, 0x01, 0xD0, 0xFE, 0xC7, 0x00, 0xA2, 0xFF,
+ 0x17, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE2, 0x01, 0x50,
+ 0xFC, 0x99, 0x06, 0xFE, 0xF3, 0xC3, 0x1D, 0x5E, 0x3F, 0x27, 0xF4,
+ 0xB9, 0x04, 0x10, 0xFE, 0xA9, 0x00, 0xDF, 0xFF, 0xFB, 0xFF, 0x03,
+ 0x00, 0xFD, 0xFF, 0x26, 0x00, 0x69, 0xFF, 0x90, 0x01, 0x96, 0xFC,
+ 0xDD, 0x06, 0x85, 0xF1, 0xD0, 0x32, 0xA6, 0x2D, 0x94, 0xF1, 0x20,
+ 0x07, 0x54, 0xFC, 0xBF, 0x01, 0x50, 0xFF, 0x2E, 0x00, 0xFD, 0xFF,
+ 0x07, 0x00, 0xE6, 0xFF, 0x15, 0x00, 0x3C, 0x00, 0xCF, 0xFE, 0x83,
+ 0x03, 0x20, 0xF6, 0xB2, 0x42, 0x2B, 0x18, 0x71, 0xF5, 0x09, 0x06,
+ 0x88, 0xFC, 0xCF, 0x01, 0x3C, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0x1D,
+ 0x00, 0x89, 0xFF, 0x06, 0x01, 0x4C, 0xFE, 0x60, 0x02, 0x1F, 0xFD,
+ 0xE2, 0x02, 0xF3, 0x48, 0x7D, 0x04, 0x6E, 0xFC, 0xBD, 0x02, 0x1C,
+ 0xFE, 0x1C, 0x01, 0x80, 0xFF, 0x20, 0x00, 0x00, 0x00, 0xFF, 0xFF,
+ 0x33, 0x00, 0x3F, 0xFF, 0xC5, 0x01, 0xA3, 0xFC, 0xCA, 0x05, 0x07,
+ 0xF6, 0x22, 0x16, 0xC3, 0x43, 0xFE, 0xF6, 0x02, 0x03, 0x1B, 0xFF,
+ 0x11, 0x00, 0x2B, 0x00, 0xDE, 0xFF, 0x09, 0x00, 0xFD, 0xFF, 0x31,
+ 0x00, 0x49, 0xFF, 0xCB, 0x01, 0x45, 0xFC, 0x29, 0x07, 0xB6, 0xF1,
+ 0xAD, 0x2B, 0xA2, 0x34, 0x9E, 0xF1, 0xB4, 0x06, 0xB8, 0xFC, 0x7A,
+ 0x01, 0x75, 0xFF, 0x22, 0x00, 0xFE, 0xFF, 0x02, 0x00, 0x02, 0x00,
+ 0xCC, 0xFF, 0xCE, 0x00, 0xD1, 0xFD, 0x1D, 0x05, 0x91, 0xF3, 0xFE,
+ 0x3D, 0xD7, 0x1F, 0x87, 0xF3, 0xC3, 0x06, 0x42, 0xFC, 0xE5, 0x01,
+ 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x14, 0x00, 0xAB, 0xFF, 0xAF,
+ 0x00, 0x01, 0xFF, 0x0A, 0x01, 0x9E, 0xFF, 0x7C, 0xFD, 0x03, 0x48,
+ 0xED, 0x0A, 0xD5, 0xF9, 0x0A, 0x04, 0x74, 0xFD, 0x6A, 0x01, 0x62,
+ 0xFF, 0x28, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x2D, 0x00, 0x53, 0xFF,
+ 0x90, 0x01, 0x20, 0xFD, 0xB8, 0x04, 0x6A, 0xF8, 0xCD, 0x0E, 0xE1,
+ 0x46, 0xE1, 0xFA, 0xEB, 0x00, 0x51, 0x00, 0x65, 0xFF, 0x7F, 0x00,
+ 0xBE, 0xFF, 0x10, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x39, 0xFF, 0xE5,
+ 0x01, 0x33, 0xFC, 0x04, 0x07, 0xB1, 0xF2, 0x21, 0x24, 0xE6, 0x3A,
+ 0x97, 0xF2, 0xD0, 0x05, 0x5B, 0xFD, 0x15, 0x01, 0xA9, 0xFF, 0x0F,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x18, 0x00, 0x90, 0xFF, 0x45, 0x01,
+ 0x0B, 0xFD, 0x44, 0x06, 0x0A, 0xF2, 0x3B, 0x38, 0x80, 0x27, 0x2B,
+ 0xF2, 0x22, 0x07, 0x33, 0xFC, 0xDE, 0x01, 0x3E, 0xFF, 0x34, 0x00,
+ 0xFD, 0xFF, 0x0D, 0x00, 0xCD, 0xFF, 0x59, 0x00, 0xB3, 0xFF, 0xC4,
+ 0xFF, 0xE2, 0x01, 0x09, 0xF9, 0xAA, 0x45, 0xFE, 0x11, 0x54, 0xF7,
+ 0x38, 0x05, 0xE4, 0xFC, 0xAA, 0x01, 0x49, 0xFF, 0x30, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x24, 0x00, 0x6E, 0xFF, 0x49, 0x01, 0xBC, 0xFD,
+ 0x7A, 0x03, 0xFA, 0xFA, 0xFD, 0x07, 0x9C, 0x48, 0xC3, 0xFF, 0x89,
+ 0xFE, 0xA1, 0x01, 0xB1, 0xFE, 0xD6, 0x00, 0x9C, 0xFF, 0x18, 0x00,
+ 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xDF, 0x01, 0x5B, 0xFC, 0x7B,
+ 0x06, 0x50, 0xF4, 0x6E, 0x1C, 0x36, 0x40, 0x92, 0xF4, 0x75, 0x04,
+ 0x3B, 0xFE, 0x91, 0x00, 0xEB, 0xFF, 0xF6, 0xFF, 0x04, 0x00, 0xFD,
+ 0xFF, 0x28, 0x00, 0x63, 0xFF, 0x9D, 0x01, 0x84, 0xFC, 0xF3, 0x06,
+ 0x7D, 0xF1, 0x9E, 0x31, 0xE6, 0x2E, 0x85, 0xF1, 0x16, 0x07, 0x61,
+ 0xFC, 0xB5, 0x01, 0x55, 0xFF, 0x2D, 0x00, 0xFD, 0xFF, 0x06, 0x00,
+ 0xEC, 0xFF, 0x08, 0x00, 0x57, 0x00, 0x9F, 0xFE, 0xD1, 0x03, 0x9B,
+ 0xF5, 0xF7, 0x41, 0x7C, 0x19, 0x13, 0xF5, 0x2F, 0x06, 0x78, 0xFC,
+ 0xD5, 0x01, 0x3A, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1C, 0x00, 0x8F,
+ 0xFF, 0xF7, 0x00, 0x6B, 0xFE, 0x25, 0x02, 0x91, 0xFD, 0xE3, 0x01,
+ 0xE5, 0x48, 0x8D, 0x05, 0xFB, 0xFB, 0xF8, 0x02, 0xFE, 0xFD, 0x2B,
+ 0x01, 0x7A, 0xFF, 0x21, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x32, 0x00,
+ 0x42, 0xFF, 0xBD, 0x01, 0xB6, 0xFC, 0x9F, 0x05, 0x6C, 0xF6, 0xD6,
+ 0x14, 0x65, 0x44, 0x98, 0xF7, 0xAC, 0x02, 0x4E, 0xFF, 0xF4, 0xFF,
+ 0x39, 0x00, 0xD9, 0xFF, 0x0A, 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x45,
+ 0xFF, 0xD2, 0x01, 0x3D, 0xFC, 0x2B, 0x07, 0xD4, 0xF1, 0x64, 0x2A,
+ 0xC6, 0x35, 0xB7, 0xF1, 0x96, 0x06, 0xCF, 0xFC, 0x6B, 0x01, 0x7D,
+ 0xFF, 0x1F, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x06, 0x00, 0xC1, 0xFF,
+ 0xE5, 0x00, 0xAA, 0xFD, 0x58, 0x05, 0x3A, 0xF3, 0x11, 0x3D, 0x2C,
+ 0x21, 0x3F, 0xF3, 0xDA, 0x06, 0x3B, 0xFC, 0xE6, 0x01, 0x36, 0xFF,
+ 0x36, 0x00, 0xFD, 0xFF, 0x13, 0x00, 0xB1, 0xFF, 0xA0, 0x00, 0x20,
+ 0xFF, 0xD0, 0x00, 0x07, 0x00, 0xA4, 0xFC, 0xB6, 0x47, 0x1C, 0x0C,
+ 0x63, 0xF9, 0x42, 0x04, 0x59, 0xFD, 0x76, 0x01, 0x5D, 0xFF, 0x2A,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x2B, 0x00, 0x57, 0xFF, 0x85, 0x01,
+ 0x39, 0xFD, 0x84, 0x04, 0xD9, 0xF8, 0x95, 0x0D, 0x48, 0x47, 0xA7,
+ 0xFB, 0x86, 0x00, 0x8A, 0x00, 0x46, 0xFF, 0x8E, 0x00, 0xB8, 0xFF,
+ 0x11, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x35,
+ 0xFC, 0xF3, 0x06, 0xEE, 0xF2, 0xCD, 0x22, 0xE4, 0x3B, 0xDC, 0xF2,
+ 0x9C, 0x05, 0x7E, 0xFD, 0x00, 0x01, 0xB4, 0xFF, 0x0B, 0x00, 0x01,
+ 0x00, 0xFE, 0xFF, 0x1B, 0x00, 0x87, 0xFF, 0x57, 0x01, 0xEF, 0xFC,
+ 0x6B, 0x06, 0xE0, 0xF1, 0x23, 0x37, 0xCE, 0x28, 0x01, 0xF2, 0x28,
+ 0x07, 0x36, 0xFC, 0xD9, 0x01, 0x41, 0xFF, 0x33, 0x00, 0xFD, 0xFF,
+ 0x0B, 0x00, 0xD2, 0xFF, 0x4A, 0x00, 0xD0, 0xFF, 0x8E, 0xFF, 0x3F,
+ 0x02, 0x5E, 0xF8, 0x1E, 0x45, 0x44, 0x13, 0xEA, 0xF6, 0x67, 0x05,
+ 0xCF, 0xFC, 0xB3, 0x01, 0x46, 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0x23, 0x00, 0x74, 0xFF, 0x3C, 0x01, 0xDA, 0xFD, 0x40, 0x03,
+ 0x6E, 0xFB, 0xE1, 0x06, 0xC3, 0x48, 0xB3, 0x00, 0x1A, 0xFE, 0xDC,
+ 0x01, 0x91, 0xFE, 0xE5, 0x00, 0x96, 0xFF, 0x1A, 0x00, 0xFE, 0xFF,
+ 0x36, 0x00, 0x38, 0xFF, 0xDB, 0x01, 0x67, 0xFC, 0x5A, 0x06, 0xA6,
+ 0xF4, 0x1B, 0x1B, 0x07, 0x41, 0x04, 0xF5, 0x2D, 0x04, 0x67, 0xFE,
+ 0x77, 0x00, 0xF8, 0xFF, 0xF2, 0xFF, 0x05, 0x00, 0xFD, 0xFF, 0x2A,
+ 0x00, 0x5C, 0xFF, 0xA8, 0x01, 0x73, 0xFC, 0x05, 0x07, 0x7D, 0xF1,
+ 0x67, 0x30, 0x21, 0x30, 0x7E, 0xF1, 0x08, 0x07, 0x6F, 0xFC, 0xAB,
+ 0x01, 0x5B, 0xFF, 0x2B, 0x00, 0xFD, 0xFF, 0x05, 0x00, 0xF0, 0xFF,
+ 0xFB, 0xFF, 0x71, 0x00, 0x71, 0xFE, 0x1D, 0x04, 0x1F, 0xF5, 0x32,
+ 0x41, 0xCE, 0x1A, 0xBA, 0xF4, 0x53, 0x06, 0x6A, 0xFC, 0xDA, 0x01,
+ 0x38, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x1A, 0x00, 0x95, 0xFF, 0xE8,
+ 0x00, 0x8A, 0xFE, 0xE9, 0x01, 0x01, 0xFE, 0xEA, 0x00, 0xCB, 0x48,
+ 0xA2, 0x06, 0x87, 0xFB, 0x33, 0x03, 0xE0, 0xFD, 0x39, 0x01, 0x75,
+ 0xFF, 0x23, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x31, 0x00, 0x45, 0xFF,
+ 0xB5, 0x01, 0xCA, 0xFC, 0x72, 0x05, 0xD3, 0xF6, 0x8D, 0x13, 0xFD,
+ 0x44, 0x39, 0xF8, 0x53, 0x02, 0x82, 0xFF, 0xD7, 0xFF, 0x47, 0x00,
+ 0xD3, 0xFF, 0x0B, 0x00, 0xFD, 0xFF, 0x33, 0x00, 0x42, 0xFF, 0xD8,
+ 0x01, 0x37, 0xFC, 0x29, 0x07, 0xF8, 0xF1, 0x19, 0x29, 0xE5, 0x36,
+ 0xD8, 0xF1, 0x73, 0x06, 0xE9, 0xFC, 0x5B, 0x01, 0x85, 0xFF, 0x1C,
+ 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x0A, 0x00, 0xB6, 0xFF, 0xFB, 0x00,
+ 0x85, 0xFD, 0x90, 0x05, 0xEC, 0xF2, 0x1C, 0x3C, 0x81, 0x22, 0xFC,
+ 0xF2, 0xEF, 0x06, 0x36, 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00,
+ 0xFD, 0xFF, 0x12, 0x00, 0xB7, 0xFF, 0x91, 0x00, 0x40, 0xFF, 0x96,
+ 0x00, 0x6F, 0x00, 0xD5, 0xFB, 0x5E, 0x47, 0x50, 0x0D, 0xF2, 0xF8,
+ 0x78, 0x04, 0x3F, 0xFD, 0x82, 0x01, 0x58, 0xFF, 0x2B, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x2A, 0x00, 0x5C, 0xFF, 0x79, 0x01, 0x53, 0xFD,
+ 0x4E, 0x04, 0x4A, 0xF9, 0x60, 0x0C, 0xA3, 0x47, 0x76, 0xFC, 0x1F,
+ 0x00, 0xC3, 0x00, 0x27, 0xFF, 0x9D, 0x00, 0xB2, 0xFF, 0x13, 0x00,
+ 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x3A, 0xFC, 0xDF,
+ 0x06, 0x30, 0xF3, 0x78, 0x21, 0xDB, 0x3C, 0x28, 0xF3, 0x65, 0x05,
+ 0xA2, 0xFD, 0xEA, 0x00, 0xBE, 0xFF, 0x07, 0x00, 0x01, 0x00, 0xFE,
+ 0xFF, 0x1E, 0x00, 0x7F, 0xFF, 0x67, 0x01, 0xD5, 0xFC, 0x8E, 0x06,
+ 0xBE, 0xF1, 0x06, 0x36, 0x1A, 0x2A, 0xDC, 0xF1, 0x2A, 0x07, 0x3C,
+ 0xFC, 0xD3, 0x01, 0x44, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x0A, 0x00,
+ 0xD8, 0xFF, 0x3C, 0x00, 0xEE, 0xFF, 0x5A, 0xFF, 0x98, 0x02, 0xBB,
+ 0xF7, 0x87, 0x44, 0x8C, 0x14, 0x83, 0xF6, 0x95, 0x05, 0xBA, 0xFC,
+ 0xBB, 0x01, 0x43, 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x21,
+ 0x00, 0x79, 0xFF, 0x2E, 0x01, 0xF7, 0xFD, 0x05, 0x03, 0xE1, 0xFB,
+ 0xCA, 0x05, 0xDF, 0x48, 0xAB, 0x01, 0xAA, 0xFD, 0x18, 0x02, 0x72,
+ 0xFE, 0xF4, 0x00, 0x90, 0xFF, 0x1B, 0x00, 0xFE, 0xFF, 0x35, 0x00,
+ 0x39, 0xFF, 0xD6, 0x01, 0x75, 0xFC, 0x37, 0x06, 0xFF, 0xF4, 0xC7,
+ 0x19, 0xCC, 0x41, 0x7F, 0xF5, 0xE2, 0x03, 0x95, 0xFE, 0x5D, 0x00,
+ 0x05, 0x00, 0xED, 0xFF, 0x06, 0x00, 0xFD, 0xFF, 0x2C, 0x00, 0x57,
+ 0xFF, 0xB3, 0x01, 0x64, 0xFC, 0x13, 0x07, 0x83, 0xF1, 0x2C, 0x2F,
+ 0x5A, 0x31, 0x7D, 0xF1, 0xF7, 0x06, 0x80, 0xFC, 0x9F, 0x01, 0x61,
+ 0xFF, 0x29, 0x00, 0xFD, 0xFF, 0x04, 0x00, 0xF5, 0xFF, 0xEE, 0xFF,
+ 0x8B, 0x00, 0x44, 0xFE, 0x65, 0x04, 0xAA, 0xF4, 0x66, 0x40, 0x23,
+ 0x1C, 0x63, 0xF4, 0x74, 0x06, 0x5D, 0xFC, 0xDE, 0x01, 0x37, 0xFF,
+ 0x36, 0x00, 0xFE, 0xFF, 0x19, 0x00, 0x9A, 0xFF, 0xD9, 0x00, 0xAA,
+ 0xFE, 0xAE, 0x01, 0x70, 0xFE, 0xF8, 0xFF, 0xA6, 0x48, 0xBE, 0x07,
+ 0x14, 0xFB, 0x6D, 0x03, 0xC3, 0xFD, 0x46, 0x01, 0x70, 0xFF, 0x24,
+ 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x30, 0x00, 0x48, 0xFF, 0xAC, 0x01,
+ 0xDF, 0xFC, 0x43, 0x05, 0x3C, 0xF7, 0x46, 0x12, 0x8D, 0x45, 0xE2,
+ 0xF8, 0xF7, 0x01, 0xB8, 0xFF, 0xB9, 0xFF, 0x56, 0x00, 0xCE, 0xFF,
+ 0x0C, 0x00, 0xFD, 0xFF, 0x34, 0x00, 0x3F, 0xFF, 0xDD, 0x01, 0x34,
+ 0xFC, 0x23, 0x07, 0x21, 0xF2, 0xCB, 0x27, 0xFE, 0x37, 0x00, 0xF2,
+ 0x4D, 0x06, 0x04, 0xFD, 0x49, 0x01, 0x8E, 0xFF, 0x19, 0x00, 0xFF,
+ 0xFF, 0x00, 0x00, 0x0E, 0x00, 0xAB, 0xFF, 0x10, 0x01, 0x62, 0xFD,
+ 0xC5, 0x05, 0xA5, 0xF2, 0x1F, 0x3B, 0xD6, 0x23, 0xBE, 0xF2, 0x01,
+ 0x07, 0x33, 0xFC, 0xE5, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0x10, 0x00, 0xBD, 0xFF, 0x82, 0x00, 0x5E, 0xFF, 0x5D, 0x00, 0xD4,
+ 0x00, 0x0C, 0xFB, 0xF9, 0x46, 0x87, 0x0E, 0x82, 0xF8, 0xAD, 0x04,
+ 0x26, 0xFD, 0x8D, 0x01, 0x54, 0xFF, 0x2C, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0x29, 0x00, 0x60, 0xFF, 0x6D, 0x01, 0x6E, 0xFD, 0x17, 0x04,
+ 0xBC, 0xF9, 0x30, 0x0B, 0xF4, 0x47, 0x4B, 0xFD, 0xB5, 0xFF, 0xFD,
+ 0x00, 0x08, 0xFF, 0xAC, 0x00, 0xAC, 0xFF, 0x14, 0x00, 0xFD, 0xFF,
+ 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x41, 0xFC, 0xC8, 0x06, 0x76,
+ 0xF3, 0x22, 0x20, 0xCA, 0x3D, 0x7D, 0xF3, 0x2A, 0x05, 0xC8, 0xFD,
+ 0xD4, 0x00, 0xCA, 0xFF, 0x03, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x21,
+ 0x00, 0x77, 0xFF, 0x77, 0x01, 0xBD, 0xFC, 0xAE, 0x06, 0xA3, 0xF1,
+ 0xE3, 0x34, 0x64, 0x2B, 0xBC, 0xF1, 0x2A, 0x07, 0x43, 0xFC, 0xCD,
+ 0x01, 0x48, 0xFF, 0x31, 0x00, 0xFD, 0xFF, 0x09, 0x00, 0xDD, 0xFF,
+ 0x2E, 0x00, 0x0A, 0x00, 0x27, 0xFF, 0xEF, 0x02, 0x20, 0xF7, 0xE7,
+ 0x43, 0xD8, 0x15, 0x1E, 0xF6, 0xC0, 0x05, 0xA7, 0xFC, 0xC3, 0x01,
+ 0x40, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x20, 0x00, 0x7F,
+ 0xFF, 0x20, 0x01, 0x16, 0xFE, 0xCA, 0x02, 0x54, 0xFC, 0xB9, 0x04,
+ 0xF2, 0x48, 0xA9, 0x02, 0x39, 0xFD, 0x53, 0x02, 0x53, 0xFE, 0x03,
+ 0x01, 0x8A, 0xFF, 0x1D, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3B, 0xFF,
+ 0xD1, 0x01, 0x84, 0xFC, 0x12, 0x06, 0x5C, 0xF5, 0x76, 0x18, 0x89,
+ 0x42, 0x02, 0xF6, 0x94, 0x03, 0xC4, 0xFE, 0x42, 0x00, 0x12, 0x00,
+ 0xE8, 0xFF, 0x07, 0x00, 0xFD, 0xFF, 0x2E, 0x00, 0x51, 0xFF, 0xBD,
+ 0x01, 0x57, 0xFC, 0x1E, 0x07, 0x90, 0xF1, 0xED, 0x2D, 0x8C, 0x32,
+ 0x83, 0xF1, 0xE2, 0x06, 0x92, 0xFC, 0x93, 0x01, 0x68, 0xFF, 0x26,
+ 0x00, 0xFD, 0xFF, 0x03, 0x00, 0xFA, 0xFF, 0xE2, 0xFF, 0xA4, 0x00,
+ 0x19, 0xFE, 0xAA, 0x04, 0x3E, 0xF4, 0x90, 0x3F, 0x78, 0x1D, 0x10,
+ 0xF4, 0x93, 0x06, 0x52, 0xFC, 0xE1, 0x01, 0x36, 0xFF, 0x36, 0x00,
+ 0xFE, 0xFF, 0x17, 0x00, 0xA0, 0xFF, 0xCA, 0x00, 0xC9, 0xFE, 0x73,
+ 0x01, 0xDE, 0xFE, 0x0C, 0xFF, 0x76, 0x48, 0xDE, 0x08, 0xA1, 0xFA,
+ 0xA6, 0x03, 0xA6, 0xFD, 0x53, 0x01, 0x6A, 0xFF, 0x26, 0x00, 0x00,
+ 0x00, 0xFF, 0xFF, 0x2F, 0x00, 0x4C, 0xFF, 0xA2, 0x01, 0xF6, 0xFC,
+ 0x12, 0x05, 0xA7, 0xF7, 0x03, 0x11, 0x10, 0x46, 0x93, 0xF9, 0x98,
+ 0x01, 0xEE, 0xFF, 0x9B, 0xFF, 0x64, 0x00, 0xC8, 0xFF, 0x0E, 0x00,
+ 0xFD, 0xFF, 0x35, 0x00, 0x3C, 0xFF, 0xE1, 0x01, 0x32, 0xFC, 0x1B,
+ 0x07, 0x50, 0xF2, 0x7B, 0x26, 0x11, 0x39, 0x2F, 0xF2, 0x23, 0x06,
+ 0x22, 0xFD, 0x37, 0x01, 0x97, 0xFF, 0x15, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0x12, 0x00, 0xA1, 0xFF, 0x24, 0x01, 0x41, 0xFD, 0xF6, 0x05,
+ 0x67, 0xF2, 0x1A, 0x3A, 0x29, 0x25, 0x84, 0xF2, 0x0F, 0x07, 0x31,
+ 0xFC, 0xE3, 0x01, 0x3A, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x0F, 0x00,
+ 0xC2, 0xFF, 0x73, 0x00, 0x7D, 0xFF, 0x25, 0x00, 0x38, 0x01, 0x4C,
+ 0xFA, 0x89, 0x46, 0xC3, 0x0F, 0x14, 0xF8, 0xE0, 0x04, 0x0D, 0xFD,
+ 0x98, 0x01, 0x50, 0xFF, 0x2E, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x27,
+ 0x00, 0x65, 0xFF, 0x60, 0x01, 0x8A, 0xFD, 0xDF, 0x03, 0x2E, 0xFA,
+ 0x04, 0x0A, 0x3A, 0x48, 0x28, 0xFE, 0x4B, 0xFF, 0x38, 0x01, 0xE9,
+ 0xFE, 0xBB, 0x00, 0xA6, 0xFF, 0x16, 0x00, 0xFE, 0xFF, 0x36, 0x00,
+ 0x36, 0xFF, 0xE4, 0x01, 0x49, 0xFC, 0xAF, 0x06, 0xC1, 0xF3, 0xCD,
+ 0x1E, 0xB1, 0x3E, 0xD9, 0xF3, 0xEC, 0x04, 0xF0, 0xFD, 0xBC, 0x00,
+ 0xD5, 0xFF, 0xFE, 0xFF, 0x03, 0x00, 0xFD, 0xFF, 0x24, 0x00, 0x6F,
+ 0xFF, 0x85, 0x01, 0xA6, 0xFC, 0xCA, 0x06, 0x8F, 0xF1, 0xBB, 0x33,
+ 0xAB, 0x2C, 0xA3, 0xF1, 0x26, 0x07, 0x4C, 0xFC, 0xC5, 0x01, 0x4D,
+ 0xFF, 0x30, 0x00, 0xFD, 0xFF, 0x08, 0x00, 0xE2, 0xFF, 0x20, 0x00,
+ 0x26, 0x00, 0xF5, 0xFE, 0x43, 0x03, 0x8D, 0xF6, 0x3C, 0x43, 0x25,
+ 0x17, 0xBB, 0xF5, 0xEA, 0x05, 0x95, 0xFC, 0xCA, 0x01, 0x3D, 0xFF,
+ 0x34, 0x00, 0xFE, 0xFF, 0x00, 0x00, 0x1E, 0x00, 0x84, 0xFF, 0x11,
+ 0x01, 0x34, 0xFE, 0x8F, 0x02, 0xC7, 0xFC, 0xAE, 0x03, 0xF7, 0x48,
+ 0xAE, 0x03, 0xC7, 0xFC, 0x8F, 0x02, 0x34, 0xFE, 0x11, 0x01, 0x84,
+ 0xFF, 0x1E, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01,
+ 0x3D, 0xFC, 0xD6, 0x06, 0x4C, 0xF3, 0xED, 0x20, 0x3D, 0x3D, 0x4A,
+ 0xF3, 0x4E, 0x05, 0xB1, 0xFD, 0xE1, 0x00, 0xC3, 0xFF, 0x05, 0x00,
+ 0x02, 0x00, 0x02, 0x00, 0x05, 0x00, 0xC3, 0xFF, 0xE1, 0x00, 0xB1,
+ 0xFD, 0x4E, 0x05, 0x4A, 0xF3, 0x3D, 0x3D, 0xED, 0x20, 0x4C, 0xF3,
+ 0xD6, 0x06, 0x3D, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0x00, 0x00, 0x1E, 0x00, 0x84, 0xFF, 0x11, 0x01, 0x34, 0xFE,
+ 0x8F, 0x02, 0xC7, 0xFC, 0xAE, 0x03, 0xF7, 0x48, 0xAE, 0x03, 0xC7,
+ 0xFC, 0x8F, 0x02, 0x34, 0xFE, 0x11, 0x01, 0x84, 0xFF, 0x1E, 0x00,
+ 0xFD, 0xFF, 0x30, 0x00, 0x4D, 0xFF, 0xC5, 0x01, 0x4C, 0xFC, 0x26,
+ 0x07, 0xA3, 0xF1, 0xAB, 0x2C, 0xBB, 0x33, 0x8F, 0xF1, 0xCA, 0x06,
+ 0xA6, 0xFC, 0x85, 0x01, 0x6F, 0xFF, 0x24, 0x00, 0xFD, 0xFF, 0x16,
+ 0x00, 0xA6, 0xFF, 0xBB, 0x00, 0xE9, 0xFE, 0x38, 0x01, 0x4B, 0xFF,
+ 0x28, 0xFE, 0x3A, 0x48, 0x04, 0x0A, 0x2E, 0xFA, 0xDF, 0x03, 0x8A,
+ 0xFD, 0x60, 0x01, 0x65, 0xFF, 0x27, 0x00, 0x00, 0x00, 0xFD, 0xFF,
+ 0x35, 0x00, 0x3A, 0xFF, 0xE3, 0x01, 0x31, 0xFC, 0x0F, 0x07, 0x84,
+ 0xF2, 0x29, 0x25, 0x1A, 0x3A, 0x67, 0xF2, 0xF6, 0x05, 0x41, 0xFD,
+ 0x24, 0x01, 0xA1, 0xFF, 0x12, 0x00, 0x00, 0x00, 0x0E, 0x00, 0xC8,
+ 0xFF, 0x64, 0x00, 0x9B, 0xFF, 0xEE, 0xFF, 0x98, 0x01, 0x93, 0xF9,
+ 0x10, 0x46, 0x03, 0x11, 0xA7, 0xF7, 0x12, 0x05, 0xF6, 0xFC, 0xA2,
+ 0x01, 0x4C, 0xFF, 0x2F, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00,
+ 0x36, 0xFF, 0xE1, 0x01, 0x52, 0xFC, 0x93, 0x06, 0x10, 0xF4, 0x78,
+ 0x1D, 0x90, 0x3F, 0x3E, 0xF4, 0xAA, 0x04, 0x19, 0xFE, 0xA4, 0x00,
+ 0xE2, 0xFF, 0xFA, 0xFF, 0x03, 0x00, 0x07, 0x00, 0xE8, 0xFF, 0x12,
+ 0x00, 0x42, 0x00, 0xC4, 0xFE, 0x94, 0x03, 0x02, 0xF6, 0x89, 0x42,
+ 0x76, 0x18, 0x5C, 0xF5, 0x12, 0x06, 0x84, 0xFC, 0xD1, 0x01, 0x3B,
+ 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x33, 0x00, 0x40, 0xFF,
+ 0xC3, 0x01, 0xA7, 0xFC, 0xC0, 0x05, 0x1E, 0xF6, 0xD8, 0x15, 0xE7,
+ 0x43, 0x20, 0xF7, 0xEF, 0x02, 0x27, 0xFF, 0x0A, 0x00, 0x2E, 0x00,
+ 0xDD, 0xFF, 0x09, 0x00, 0x02, 0x00, 0x03, 0x00, 0xCA, 0xFF, 0xD4,
+ 0x00, 0xC8, 0xFD, 0x2A, 0x05, 0x7D, 0xF3, 0xCA, 0x3D, 0x22, 0x20,
+ 0x76, 0xF3, 0xC8, 0x06, 0x41, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0xFF, 0xFF, 0x2C, 0x00, 0x54, 0xFF, 0x8D, 0x01,
+ 0x26, 0xFD, 0xAD, 0x04, 0x82, 0xF8, 0x87, 0x0E, 0xF9, 0x46, 0x0C,
+ 0xFB, 0xD4, 0x00, 0x5D, 0x00, 0x5E, 0xFF, 0x82, 0x00, 0xBD, 0xFF,
+ 0x10, 0x00, 0xFF, 0xFF, 0x19, 0x00, 0x8E, 0xFF, 0x49, 0x01, 0x04,
+ 0xFD, 0x4D, 0x06, 0x00, 0xF2, 0xFE, 0x37, 0xCB, 0x27, 0x21, 0xF2,
+ 0x23, 0x07, 0x34, 0xFC, 0xDD, 0x01, 0x3F, 0xFF, 0x34, 0x00, 0xFD,
+ 0xFF, 0x00, 0x00, 0x24, 0x00, 0x70, 0xFF, 0x46, 0x01, 0xC3, 0xFD,
+ 0x6D, 0x03, 0x14, 0xFB, 0xBE, 0x07, 0xA6, 0x48, 0xF8, 0xFF, 0x70,
+ 0xFE, 0xAE, 0x01, 0xAA, 0xFE, 0xD9, 0x00, 0x9A, 0xFF, 0x19, 0x00,
+ 0xFD, 0xFF, 0x29, 0x00, 0x61, 0xFF, 0x9F, 0x01, 0x80, 0xFC, 0xF7,
+ 0x06, 0x7D, 0xF1, 0x5A, 0x31, 0x2C, 0x2F, 0x83, 0xF1, 0x13, 0x07,
+ 0x64, 0xFC, 0xB3, 0x01, 0x57, 0xFF, 0x2C, 0x00, 0xFD, 0xFF, 0x1B,
+ 0x00, 0x90, 0xFF, 0xF4, 0x00, 0x72, 0xFE, 0x18, 0x02, 0xAA, 0xFD,
+ 0xAB, 0x01, 0xDF, 0x48, 0xCA, 0x05, 0xE1, 0xFB, 0x05, 0x03, 0xF7,
+ 0xFD, 0x2E, 0x01, 0x79, 0xFF, 0x21, 0x00, 0x00, 0x00, 0xFD, 0xFF,
+ 0x32, 0x00, 0x44, 0xFF, 0xD3, 0x01, 0x3C, 0xFC, 0x2A, 0x07, 0xDC,
+ 0xF1, 0x1A, 0x2A, 0x06, 0x36, 0xBE, 0xF1, 0x8E, 0x06, 0xD5, 0xFC,
+ 0x67, 0x01, 0x7F, 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0x13, 0x00, 0xB2,
+ 0xFF, 0x9D, 0x00, 0x27, 0xFF, 0xC3, 0x00, 0x1F, 0x00, 0x76, 0xFC,
+ 0xA3, 0x47, 0x60, 0x0C, 0x4A, 0xF9, 0x4E, 0x04, 0x53, 0xFD, 0x79,
+ 0x01, 0x5C, 0xFF, 0x2A, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00,
+ 0x37, 0xFF, 0xE6, 0x01, 0x36, 0xFC, 0xEF, 0x06, 0xFC, 0xF2, 0x81,
+ 0x22, 0x1C, 0x3C, 0xEC, 0xF2, 0x90, 0x05, 0x85, 0xFD, 0xFB, 0x00,
+ 0xB6, 0xFF, 0x0A, 0x00, 0x01, 0x00, 0x0B, 0x00, 0xD3, 0xFF, 0x47,
+ 0x00, 0xD7, 0xFF, 0x82, 0xFF, 0x53, 0x02, 0x39, 0xF8, 0xFD, 0x44,
+ 0x8D, 0x13, 0xD3, 0xF6, 0x72, 0x05, 0xCA, 0xFC, 0xB5, 0x01, 0x45,
+ 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x35, 0x00, 0x38, 0xFF,
+ 0xDA, 0x01, 0x6A, 0xFC, 0x53, 0x06, 0xBA, 0xF4, 0xCE, 0x1A, 0x32,
+ 0x41, 0x1F, 0xF5, 0x1D, 0x04, 0x71, 0xFE, 0x71, 0x00, 0xFB, 0xFF,
+ 0xF0, 0xFF, 0x05, 0x00, 0x05, 0x00, 0xF2, 0xFF, 0xF8, 0xFF, 0x77,
+ 0x00, 0x67, 0xFE, 0x2D, 0x04, 0x04, 0xF5, 0x07, 0x41, 0x1B, 0x1B,
+ 0xA6, 0xF4, 0x5A, 0x06, 0x67, 0xFC, 0xDB, 0x01, 0x38, 0xFF, 0x36,
+ 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x31, 0x00, 0x46, 0xFF, 0xB3, 0x01,
+ 0xCF, 0xFC, 0x67, 0x05, 0xEA, 0xF6, 0x44, 0x13, 0x1E, 0x45, 0x5E,
+ 0xF8, 0x3F, 0x02, 0x8E, 0xFF, 0xD0, 0xFF, 0x4A, 0x00, 0xD2, 0xFF,
+ 0x0B, 0x00, 0x01, 0x00, 0x0B, 0x00, 0xB4, 0xFF, 0x00, 0x01, 0x7E,
+ 0xFD, 0x9C, 0x05, 0xDC, 0xF2, 0xE4, 0x3B, 0xCD, 0x22, 0xEE, 0xF2,
+ 0xF3, 0x06, 0x35, 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0x00, 0x00, 0x2A, 0x00, 0x5D, 0xFF, 0x76, 0x01, 0x59, 0xFD,
+ 0x42, 0x04, 0x63, 0xF9, 0x1C, 0x0C, 0xB6, 0x47, 0xA4, 0xFC, 0x07,
+ 0x00, 0xD0, 0x00, 0x20, 0xFF, 0xA0, 0x00, 0xB1, 0xFF, 0x13, 0x00,
+ 0xFE, 0xFF, 0x1F, 0x00, 0x7D, 0xFF, 0x6B, 0x01, 0xCF, 0xFC, 0x96,
+ 0x06, 0xB7, 0xF1, 0xC6, 0x35, 0x64, 0x2A, 0xD4, 0xF1, 0x2B, 0x07,
+ 0x3D, 0xFC, 0xD2, 0x01, 0x45, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x00,
+ 0x00, 0x21, 0x00, 0x7A, 0xFF, 0x2B, 0x01, 0xFE, 0xFD, 0xF8, 0x02,
+ 0xFB, 0xFB, 0x8D, 0x05, 0xE5, 0x48, 0xE3, 0x01, 0x91, 0xFD, 0x25,
+ 0x02, 0x6B, 0xFE, 0xF7, 0x00, 0x8F, 0xFF, 0x1C, 0x00, 0xFD, 0xFF,
+ 0x2D, 0x00, 0x55, 0xFF, 0xB5, 0x01, 0x61, 0xFC, 0x16, 0x07, 0x85,
+ 0xF1, 0xE6, 0x2E, 0x9E, 0x31, 0x7D, 0xF1, 0xF3, 0x06, 0x84, 0xFC,
+ 0x9D, 0x01, 0x63, 0xFF, 0x28, 0x00, 0xFD, 0xFF, 0x18, 0x00, 0x9C,
+ 0xFF, 0xD6, 0x00, 0xB1, 0xFE, 0xA1, 0x01, 0x89, 0xFE, 0xC3, 0xFF,
+ 0x9C, 0x48, 0xFD, 0x07, 0xFA, 0xFA, 0x7A, 0x03, 0xBC, 0xFD, 0x49,
+ 0x01, 0x6E, 0xFF, 0x24, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x34, 0x00,
+ 0x3E, 0xFF, 0xDE, 0x01, 0x33, 0xFC, 0x22, 0x07, 0x2B, 0xF2, 0x80,
+ 0x27, 0x3B, 0x38, 0x0A, 0xF2, 0x44, 0x06, 0x0B, 0xFD, 0x45, 0x01,
+ 0x90, 0xFF, 0x18, 0x00, 0xFF, 0xFF, 0x10, 0x00, 0xBE, 0xFF, 0x7F,
+ 0x00, 0x65, 0xFF, 0x51, 0x00, 0xEB, 0x00, 0xE1, 0xFA, 0xE1, 0x46,
+ 0xCD, 0x0E, 0x6A, 0xF8, 0xB8, 0x04, 0x20, 0xFD, 0x90, 0x01, 0x53,
+ 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF,
+ 0xE5, 0x01, 0x42, 0xFC, 0xC3, 0x06, 0x87, 0xF3, 0xD7, 0x1F, 0xFE,
+ 0x3D, 0x91, 0xF3, 0x1D, 0x05, 0xD1, 0xFD, 0xCE, 0x00, 0xCC, 0xFF,
+ 0x02, 0x00, 0x02, 0x00, 0x09, 0x00, 0xDE, 0xFF, 0x2B, 0x00, 0x11,
+ 0x00, 0x1B, 0xFF, 0x02, 0x03, 0xFE, 0xF6, 0xC3, 0x43, 0x22, 0x16,
+ 0x07, 0xF6, 0xCA, 0x05, 0xA3, 0xFC, 0xC5, 0x01, 0x3F, 0xFF, 0x33,
+ 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x34, 0x00, 0x3C, 0xFF, 0xCF, 0x01,
+ 0x88, 0xFC, 0x09, 0x06, 0x71, 0xF5, 0x2B, 0x18, 0xB2, 0x42, 0x20,
+ 0xF6, 0x83, 0x03, 0xCF, 0xFE, 0x3C, 0x00, 0x15, 0x00, 0xE6, 0xFF,
+ 0x07, 0x00, 0x03, 0x00, 0xFB, 0xFF, 0xDF, 0xFF, 0xA9, 0x00, 0x10,
+ 0xFE, 0xB9, 0x04, 0x27, 0xF4, 0x5E, 0x3F, 0xC3, 0x1D, 0xFE, 0xF3,
+ 0x99, 0x06, 0x50, 0xFC, 0xE2, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE,
+ 0xFF, 0xFF, 0xFF, 0x2F, 0x00, 0x4D, 0xFF, 0xA0, 0x01, 0xFB, 0xFC,
+ 0x07, 0x05, 0xBF, 0xF7, 0xBB, 0x10, 0x2B, 0x46, 0xBB, 0xF9, 0x83,
+ 0x01, 0xFA, 0xFF, 0x95, 0xFF, 0x68, 0x00, 0xC7, 0xFF, 0x0E, 0x00,
+ 0x00, 0x00, 0x13, 0x00, 0x9F, 0xFF, 0x28, 0x01, 0x3A, 0xFD, 0x00,
+ 0x06, 0x5A, 0xF2, 0xDF, 0x39, 0x73, 0x25, 0x79, 0xF2, 0x12, 0x07,
+ 0x31, 0xFC, 0xE3, 0x01, 0x3B, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x00,
+ 0x00, 0x27, 0x00, 0x66, 0xFF, 0x5E, 0x01, 0x90, 0xFD, 0xD2, 0x03,
+ 0x47, 0xFA, 0xC3, 0x09, 0x48, 0x48, 0x5A, 0xFE, 0x33, 0xFF, 0x45,
+ 0x01, 0xE2, 0xFE, 0xBE, 0x00, 0xA5, 0xFF, 0x16, 0x00, 0xFD, 0xFF,
+ 0x24, 0x00, 0x6D, 0xFF, 0x88, 0x01, 0xA2, 0xFC, 0xD0, 0x06, 0x8C,
+ 0xF1, 0x78, 0x33, 0xF2, 0x2C, 0x9E, 0xF1, 0x24, 0x07, 0x4E, 0xFC,
+ 0xC3, 0x01, 0x4E, 0xFF, 0x2F, 0x00, 0xFD, 0xFF, 0x1E, 0x00, 0x86,
+ 0xFF, 0x0E, 0x01, 0x3B, 0xFE, 0x82, 0x02, 0xE0, 0xFC, 0x73, 0x03,
+ 0xF6, 0x48, 0xE9, 0x03, 0xAD, 0xFC, 0x9C, 0x02, 0x2D, 0xFE, 0x14,
+ 0x01, 0x83, 0xFF, 0x1F, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x30, 0x00,
+ 0x4C, 0xFF, 0xC7, 0x01, 0x4A, 0xFC, 0x27, 0x07, 0xA8, 0xF1, 0x62,
+ 0x2C, 0xFD, 0x33, 0x93, 0xF1, 0xC4, 0x06, 0xAB, 0xFC, 0x82, 0x01,
+ 0x71, 0xFF, 0x23, 0x00, 0xFE, 0xFF, 0x15, 0x00, 0xA8, 0xFF, 0xB8,
+ 0x00, 0xF0, 0xFE, 0x2B, 0x01, 0x63, 0xFF, 0xF6, 0xFD, 0x2C, 0x48,
+ 0x47, 0x0A, 0x14, 0xFA, 0xEB, 0x03, 0x84, 0xFD, 0x63, 0x01, 0x64,
+ 0xFF, 0x27, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x3A, 0xFF,
+ 0xE4, 0x01, 0x32, 0xFC, 0x0C, 0x07, 0x91, 0xF2, 0xDD, 0x24, 0x54,
+ 0x3A, 0x74, 0xF2, 0xEB, 0x05, 0x49, 0xFD, 0x20, 0x01, 0xA3, 0xFF,
+ 0x11, 0x00, 0x00, 0x00, 0x0D, 0x00, 0xC9, 0xFF, 0x61, 0x00, 0xA2,
+ 0xFF, 0xE2, 0xFF, 0xAE, 0x01, 0x6B, 0xF9, 0xF2, 0x45, 0x4A, 0x11,
+ 0x8F, 0xF7, 0x1D, 0x05, 0xF1, 0xFC, 0xA4, 0x01, 0x4B, 0xFF, 0x2F,
+ 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE1, 0x01,
+ 0x55, 0xFC, 0x8C, 0x06, 0x22, 0xF4, 0x2C, 0x1D, 0xC0, 0x3F, 0x55,
+ 0xF4, 0x9B, 0x04, 0x23, 0xFE, 0x9F, 0x00, 0xE4, 0xFF, 0xF9, 0xFF,
+ 0x04, 0x00, 0x07, 0x00, 0xE9, 0xFF, 0x0F, 0x00, 0x48, 0x00, 0xB9,
+ 0xFE, 0xA6, 0x03, 0xE4, 0xF5, 0x60, 0x42, 0xC1, 0x18, 0x47, 0xF5,
+ 0x1A, 0x06, 0x81, 0xFC, 0xD2, 0x01, 0x3B, 0xFF, 0x35, 0x00, 0xFE,
+ 0xFF, 0xFF, 0xFF, 0x33, 0x00, 0x40, 0xFF, 0xC1, 0x01, 0xAB, 0xFC,
+ 0xB7, 0x05, 0x34, 0xF6, 0x8E, 0x15, 0x0B, 0x44, 0x42, 0xF7, 0xDC,
+ 0x02, 0x32, 0xFF, 0x04, 0x00, 0x31, 0x00, 0xDC, 0xFF, 0x09, 0x00,
+ 0x02, 0x00, 0x04, 0x00, 0xC7, 0xFF, 0xD9, 0x00, 0xBF, 0xFD, 0x38,
+ 0x05, 0x69, 0xF3, 0x96, 0x3D, 0x6F, 0x20, 0x66, 0xF3, 0xCE, 0x06,
+ 0x3F, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFF,
+ 0xFF, 0x2C, 0x00, 0x55, 0xFF, 0x8B, 0x01, 0x2B, 0xFD, 0xA1, 0x04,
+ 0x9B, 0xF8, 0x42, 0x0E, 0x0F, 0x47, 0x38, 0xFB, 0xBE, 0x00, 0x6A,
+ 0x00, 0x58, 0xFF, 0x85, 0x00, 0xBB, 0xFF, 0x10, 0x00, 0xFF, 0xFF,
+ 0x19, 0x00, 0x8C, 0xFF, 0x4D, 0x01, 0xFE, 0xFC, 0x56, 0x06, 0xF7,
+ 0xF1, 0xBF, 0x37, 0x15, 0x28, 0x18, 0xF2, 0x25, 0x07, 0x34, 0xFC,
+ 0xDC, 0x01, 0x3F, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x24,
+ 0x00, 0x71, 0xFF, 0x43, 0x01, 0xC9, 0xFD, 0x60, 0x03, 0x2E, 0xFB,
+ 0x7E, 0x07, 0xAF, 0x48, 0x2D, 0x00, 0x58, 0xFE, 0xBB, 0x01, 0xA3,
+ 0xFE, 0xDD, 0x00, 0x99, 0xFF, 0x19, 0x00, 0xFD, 0xFF, 0x29, 0x00,
+ 0x60, 0xFF, 0xA2, 0x01, 0x7C, 0xFC, 0xFB, 0x06, 0x7C, 0xF1, 0x15,
+ 0x31, 0x73, 0x2F, 0x81, 0xF1, 0x10, 0x07, 0x67, 0xFC, 0xB1, 0x01,
+ 0x58, 0xFF, 0x2C, 0x00, 0xFD, 0xFF, 0x1B, 0x00, 0x91, 0xFF, 0xF1,
+ 0x00, 0x79, 0xFE, 0x0A, 0x02, 0xC3, 0xFD, 0x73, 0x01, 0xDB, 0x48,
+ 0x07, 0x06, 0xC7, 0xFB, 0x12, 0x03, 0xF1, 0xFD, 0x31, 0x01, 0x78,
+ 0xFF, 0x22, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x44, 0xFF,
+ 0xD5, 0x01, 0x3A, 0xFC, 0x2A, 0x07, 0xE3, 0xF1, 0xD1, 0x29, 0x46,
+ 0x36, 0xC5, 0xF1, 0x87, 0x06, 0xDA, 0xFC, 0x64, 0x01, 0x80, 0xFF,
+ 0x1E, 0x00, 0xFE, 0xFF, 0x12, 0x00, 0xB3, 0xFF, 0x99, 0x00, 0x2E,
+ 0xFF, 0xB6, 0x00, 0x36, 0x00, 0x47, 0xFC, 0x90, 0x47, 0xA4, 0x0C,
+ 0x31, 0xF9, 0x5A, 0x04, 0x4E, 0xFD, 0x7C, 0x01, 0x5B, 0xFF, 0x2A,
+ 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01,
+ 0x37, 0xFC, 0xEB, 0x06, 0x0B, 0xF3, 0x35, 0x22, 0x52, 0x3C, 0xFD,
+ 0xF2, 0x84, 0x05, 0x8D, 0xFD, 0xF6, 0x00, 0xB8, 0xFF, 0x09, 0x00,
+ 0x01, 0x00, 0x0B, 0x00, 0xD5, 0xFF, 0x44, 0x00, 0xDD, 0xFF, 0x77,
+ 0xFF, 0x67, 0x02, 0x14, 0xF8, 0xDC, 0x44, 0xD5, 0x13, 0xBC, 0xF6,
+ 0x7C, 0x05, 0xC5, 0xFC, 0xB7, 0x01, 0x44, 0xFF, 0x31, 0x00, 0xFF,
+ 0xFF, 0xFE, 0xFF, 0x35, 0x00, 0x39, 0xFF, 0xD9, 0x01, 0x6D, 0xFC,
+ 0x4B, 0x06, 0xCD, 0xF4, 0x83, 0x1A, 0x5F, 0x41, 0x3A, 0xF5, 0x0C,
+ 0x04, 0x7B, 0xFE, 0x6C, 0x00, 0xFE, 0xFF, 0xEF, 0xFF, 0x05, 0x00,
+ 0x05, 0x00, 0xF3, 0xFF, 0xF5, 0xFF, 0x7D, 0x00, 0x5D, 0xFE, 0x3E,
+ 0x04, 0xEA, 0xF4, 0xD9, 0x40, 0x66, 0x1B, 0x93, 0xF4, 0x62, 0x06,
+ 0x64, 0xFC, 0xDC, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF,
+ 0xFF, 0x31, 0x00, 0x46, 0xFF, 0xB1, 0x01, 0xD3, 0xFC, 0x5D, 0x05,
+ 0x01, 0xF7, 0xFB, 0x12, 0x3F, 0x45, 0x83, 0xF8, 0x2A, 0x02, 0x9A,
+ 0xFF, 0xCA, 0xFF, 0x4E, 0x00, 0xD1, 0xFF, 0x0C, 0x00, 0x00, 0x00,
+ 0x0C, 0x00, 0xB1, 0xFF, 0x04, 0x01, 0x76, 0xFD, 0xA8, 0x05, 0xCC,
+ 0xF2, 0xAB, 0x3B, 0x18, 0x23, 0xE0, 0xF2, 0xF7, 0x06, 0x35, 0xFC,
+ 0xE6, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x29,
+ 0x00, 0x5E, 0xFF, 0x74, 0x01, 0x5F, 0xFD, 0x35, 0x04, 0x7C, 0xF9,
+ 0xD8, 0x0B, 0xC9, 0x47, 0xD4, 0xFC, 0xF0, 0xFF, 0xDD, 0x00, 0x19,
+ 0xFF, 0xA4, 0x00, 0xAF, 0xFF, 0x13, 0x00, 0xFE, 0xFF, 0x20, 0x00,
+ 0x7B, 0xFF, 0x6E, 0x01, 0xCA, 0xFC, 0x9D, 0x06, 0xB1, 0xF1, 0x86,
+ 0x35, 0xAE, 0x2A, 0xCD, 0xF1, 0x2B, 0x07, 0x3F, 0xFC, 0xD1, 0x01,
+ 0x46, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x21, 0x00, 0x7C,
+ 0xFF, 0x27, 0x01, 0x05, 0xFE, 0xEB, 0x02, 0x14, 0xFC, 0x50, 0x05,
+ 0xEA, 0x48, 0x1B, 0x02, 0x78, 0xFD, 0x32, 0x02, 0x64, 0xFE, 0xFA,
+ 0x00, 0x8D, 0xFF, 0x1C, 0x00, 0xFD, 0xFF, 0x2D, 0x00, 0x54, 0xFF,
+ 0xB7, 0x01, 0x5E, 0xFC, 0x19, 0x07, 0x88, 0xF1, 0x9F, 0x2E, 0xE3,
+ 0x31, 0x7E, 0xF1, 0xEE, 0x06, 0x88, 0xFC, 0x9A, 0x01, 0x64, 0xFF,
+ 0x28, 0x00, 0xFD, 0xFF, 0x18, 0x00, 0x9D, 0xFF, 0xD3, 0x00, 0xB8,
+ 0xFE, 0x93, 0x01, 0xA1, 0xFE, 0x8E, 0xFF, 0x92, 0x48, 0x3D, 0x08,
+ 0xE1, 0xFA, 0x86, 0x03, 0xB6, 0xFD, 0x4C, 0x01, 0x6D, 0xFF, 0x25,
+ 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x34, 0x00, 0x3E, 0xFF, 0xDF, 0x01,
+ 0x33, 0xFC, 0x20, 0x07, 0x35, 0xF2, 0x36, 0x27, 0x78, 0x38, 0x14,
+ 0xF2, 0x3B, 0x06, 0x11, 0xFD, 0x41, 0x01, 0x92, 0xFF, 0x17, 0x00,
+ 0xFF, 0xFF, 0x10, 0x00, 0xBF, 0xFF, 0x7B, 0x00, 0x6C, 0xFF, 0x44,
+ 0x00, 0x01, 0x01, 0xB6, 0xFA, 0xC8, 0x46, 0x13, 0x0F, 0x51, 0xF8,
+ 0xC4, 0x04, 0x1B, 0xFD, 0x92, 0x01, 0x52, 0xFF, 0x2D, 0x00, 0xFF,
+ 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE5, 0x01, 0x44, 0xFC,
+ 0xBD, 0x06, 0x97, 0xF3, 0x8A, 0x1F, 0x31, 0x3E, 0xA5, 0xF3, 0x0F,
+ 0x05, 0xDA, 0xFD, 0xC9, 0x00, 0xCF, 0xFF, 0x01, 0x00, 0x02, 0x00,
+ 0x09, 0x00, 0xDF, 0xFF, 0x28, 0x00, 0x17, 0x00, 0x10, 0xFF, 0x15,
+ 0x03, 0xDD, 0xF6, 0x9E, 0x43, 0x6C, 0x16, 0xF1, 0xF5, 0xD3, 0x05,
+ 0x9F, 0xFC, 0xC6, 0x01, 0x3F, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0xFE,
+ 0xFF, 0x34, 0x00, 0x3C, 0xFF, 0xCE, 0x01, 0x8C, 0xFC, 0x00, 0x06,
+ 0x86, 0xF5, 0xE0, 0x17, 0xDB, 0x42, 0x3F, 0xF6, 0x71, 0x03, 0xD9,
+ 0xFE, 0x36, 0x00, 0x18, 0x00, 0xE5, 0xFF, 0x07, 0x00, 0x03, 0x00,
+ 0xFC, 0xFF, 0xDC, 0xFF, 0xAF, 0x00, 0x07, 0xFE, 0xC8, 0x04, 0x10,
+ 0xF4, 0x2D, 0x3F, 0x0F, 0x1E, 0xED, 0xF3, 0xA0, 0x06, 0x4E, 0xFC,
+ 0xE3, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x2E,
+ 0x00, 0x4E, 0xFF, 0x9E, 0x01, 0x00, 0xFD, 0xFC, 0x04, 0xD7, 0xF7,
+ 0x75, 0x10, 0x48, 0x46, 0xE4, 0xF9, 0x6E, 0x01, 0x06, 0x00, 0x8E,
+ 0xFF, 0x6B, 0x00, 0xC6, 0xFF, 0x0E, 0x00, 0xFF, 0xFF, 0x13, 0x00,
+ 0x9D, 0xFF, 0x2D, 0x01, 0x33, 0xFD, 0x0B, 0x06, 0x4D, 0xF2, 0xA5,
+ 0x39, 0xBF, 0x25, 0x6D, 0xF2, 0x15, 0x07, 0x31, 0xFC, 0xE2, 0x01,
+ 0x3B, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x26, 0x00, 0x68,
+ 0xFF, 0x5B, 0x01, 0x96, 0xFD, 0xC6, 0x03, 0x61, 0xFA, 0x81, 0x09,
+ 0x57, 0x48, 0x8D, 0xFE, 0x1B, 0xFF, 0x52, 0x01, 0xDB, 0xFE, 0xC2,
+ 0x00, 0xA4, 0xFF, 0x16, 0x00, 0xFD, 0xFF, 0x25, 0x00, 0x6C, 0xFF,
+ 0x8B, 0x01, 0x9D, 0xFC, 0xD5, 0x06, 0x89, 0xF1, 0x35, 0x33, 0x3A,
+ 0x2D, 0x9A, 0xF1, 0x23, 0x07, 0x51, 0xFC, 0xC2, 0x01, 0x4F, 0xFF,
+ 0x2F, 0x00, 0xFD, 0xFF, 0x1E, 0x00, 0x87, 0xFF, 0x0B, 0x01, 0x42,
+ 0xFE, 0x74, 0x02, 0xF9, 0xFC, 0x39, 0x03, 0xF5, 0x48, 0x24, 0x04,
+ 0x94, 0xFC, 0xA9, 0x02, 0x27, 0xFE, 0x18, 0x01, 0x82, 0xFF, 0x1F,
+ 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x30, 0x00, 0x4B, 0xFF, 0xC9, 0x01,
+ 0x48, 0xFC, 0x28, 0x07, 0xAD, 0xF1, 0x19, 0x2C, 0x3F, 0x34, 0x97,
+ 0xF1, 0xBE, 0x06, 0xB0, 0xFC, 0x7F, 0x01, 0x72, 0xFF, 0x23, 0x00,
+ 0xFE, 0xFF, 0x15, 0x00, 0xA9, 0xFF, 0xB4, 0x00, 0xF7, 0xFE, 0x1D,
+ 0x01, 0x7A, 0xFF, 0xC5, 0xFD, 0x1D, 0x48, 0x89, 0x0A, 0xFB, 0xF9,
+ 0xF8, 0x03, 0x7D, 0xFD, 0x66, 0x01, 0x63, 0xFF, 0x28, 0x00, 0x00,
+ 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x39, 0xFF, 0xE4, 0x01, 0x32, 0xFC,
+ 0x09, 0x07, 0x9D, 0xF2, 0x92, 0x24, 0x8F, 0x3A, 0x82, 0xF2, 0xE1,
+ 0x05, 0x50, 0xFD, 0x1B, 0x01, 0xA6, 0xFF, 0x10, 0x00, 0x00, 0x00,
+ 0x0D, 0x00, 0xCB, 0xFF, 0x5E, 0x00, 0xA9, 0xFF, 0xD6, 0xFF, 0xC3,
+ 0x01, 0x43, 0xF9, 0xD7, 0x45, 0x92, 0x11, 0x77, 0xF7, 0x28, 0x05,
+ 0xEC, 0xFC, 0xA7, 0x01, 0x4A, 0xFF, 0x2F, 0x00, 0xFF, 0xFF, 0xFE,
+ 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE0, 0x01, 0x57, 0xFC, 0x85, 0x06,
+ 0x34, 0xF4, 0xE0, 0x1C, 0xF0, 0x3F, 0x6D, 0xF4, 0x8C, 0x04, 0x2C,
+ 0xFE, 0x99, 0x00, 0xE7, 0xFF, 0xF8, 0xFF, 0x04, 0x00, 0x06, 0x00,
+ 0xEA, 0xFF, 0x0C, 0x00, 0x4E, 0x00, 0xAF, 0xFE, 0xB8, 0x03, 0xC7,
+ 0xF5, 0x38, 0x42, 0x0C, 0x19, 0x32, 0xF5, 0x23, 0x06, 0x7D, 0xFC,
+ 0xD3, 0x01, 0x3A, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x32,
+ 0x00, 0x41, 0xFF, 0xC0, 0x01, 0xAF, 0xFC, 0xAD, 0x05, 0x4A, 0xF6,
+ 0x44, 0x15, 0x2F, 0x44, 0x64, 0xF7, 0xC9, 0x02, 0x3D, 0xFF, 0xFE,
+ 0xFF, 0x34, 0x00, 0xDB, 0xFF, 0x09, 0x00, 0x02, 0x00, 0x05, 0x00,
+ 0xC5, 0xFF, 0xDE, 0x00, 0xB7, 0xFD, 0x45, 0x05, 0x56, 0xF3, 0x61,
+ 0x3D, 0xBA, 0x20, 0x56, 0xF3, 0xD3, 0x06, 0x3E, 0xFC, 0xE6, 0x01,
+ 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFF, 0xFF, 0x2C, 0x00, 0x56,
+ 0xFF, 0x88, 0x01, 0x31, 0xFD, 0x95, 0x04, 0xB4, 0xF8, 0xFC, 0x0D,
+ 0x26, 0x47, 0x64, 0xFB, 0xA7, 0x00, 0x77, 0x00, 0x51, 0xFF, 0x89,
+ 0x00, 0xBA, 0xFF, 0x11, 0x00, 0xFF, 0xFF, 0x1A, 0x00, 0x8A, 0xFF,
+ 0x51, 0x01, 0xF8, 0xFC, 0x5E, 0x06, 0xED, 0xF1, 0x82, 0x37, 0x60,
+ 0x28, 0x0E, 0xF2, 0x26, 0x07, 0x35, 0xFC, 0xDB, 0x01, 0x40, 0xFF,
+ 0x34, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x23, 0x00, 0x72, 0xFF, 0x40,
+ 0x01, 0xD0, 0xFD, 0x53, 0x03, 0x47, 0xFB, 0x3F, 0x07, 0xB8, 0x48,
+ 0x62, 0x00, 0x3F, 0xFE, 0xC8, 0x01, 0x9C, 0xFE, 0xE0, 0x00, 0x98,
+ 0xFF, 0x19, 0x00, 0xFD, 0xFF, 0x29, 0x00, 0x5F, 0xFF, 0xA5, 0x01,
+ 0x78, 0xFC, 0xFF, 0x06, 0x7D, 0xF1, 0xCF, 0x30, 0xB8, 0x2F, 0x80,
+ 0xF1, 0x0D, 0x07, 0x6A, 0xFC, 0xAE, 0x01, 0x59, 0xFF, 0x2B, 0x00,
+ 0xFD, 0xFF, 0x1B, 0x00, 0x93, 0xFF, 0xED, 0x00, 0x80, 0xFE, 0xFD,
+ 0x01, 0xDC, 0xFD, 0x3C, 0x01, 0xD5, 0x48, 0x45, 0x06, 0xAE, 0xFB,
+ 0x1F, 0x03, 0xEA, 0xFD, 0x34, 0x01, 0x77, 0xFF, 0x22, 0x00, 0x00,
+ 0x00, 0xFD, 0xFF, 0x33, 0x00, 0x43, 0xFF, 0xD6, 0x01, 0x39, 0xFC,
+ 0x2A, 0x07, 0xEB, 0xF1, 0x87, 0x29, 0x85, 0x36, 0xCC, 0xF1, 0x7F,
+ 0x06, 0xE0, 0xFC, 0x60, 0x01, 0x82, 0xFF, 0x1D, 0x00, 0xFE, 0xFF,
+ 0x12, 0x00, 0xB5, 0xFF, 0x96, 0x00, 0x35, 0xFF, 0xA9, 0x00, 0x4D,
+ 0x00, 0x19, 0xFC, 0x7C, 0x47, 0xE8, 0x0C, 0x18, 0xF9, 0x66, 0x04,
+ 0x48, 0xFD, 0x7E, 0x01, 0x5A, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0xFD,
+ 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x38, 0xFC, 0xE6, 0x06,
+ 0x19, 0xF3, 0xEA, 0x21, 0x8A, 0x3C, 0x0E, 0xF3, 0x78, 0x05, 0x96,
+ 0xFD, 0xF1, 0x00, 0xBB, 0xFF, 0x08, 0x00, 0x01, 0x00, 0x0B, 0x00,
+ 0xD6, 0xFF, 0x41, 0x00, 0xE4, 0xFF, 0x6B, 0xFF, 0x7B, 0x02, 0xF0,
+ 0xF7, 0xBA, 0x44, 0x1E, 0x14, 0xA5, 0xF6, 0x86, 0x05, 0xC1, 0xFC,
+ 0xB9, 0x01, 0x44, 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x35,
+ 0x00, 0x39, 0xFF, 0xD8, 0x01, 0x70, 0xFC, 0x43, 0x06, 0xE1, 0xF4,
+ 0x38, 0x1A, 0x8C, 0x41, 0x55, 0xF5, 0xFC, 0x03, 0x85, 0xFE, 0x66,
+ 0x00, 0x01, 0x00, 0xEE, 0xFF, 0x06, 0x00, 0x05, 0x00, 0xF4, 0xFF,
+ 0xF2, 0xFF, 0x83, 0x00, 0x53, 0xFE, 0x4E, 0x04, 0xD0, 0xF4, 0xAB,
+ 0x40, 0xB2, 0x1B, 0x7F, 0xF4, 0x69, 0x06, 0x62, 0xFC, 0xDD, 0x01,
+ 0x38, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x30, 0x00, 0x47,
+ 0xFF, 0xAF, 0x01, 0xD8, 0xFC, 0x52, 0x05, 0x19, 0xF7, 0xB2, 0x12,
+ 0x5C, 0x45, 0xA9, 0xF8, 0x16, 0x02, 0xA6, 0xFF, 0xC3, 0xFF, 0x51,
+ 0x00, 0xD0, 0xFF, 0x0C, 0x00, 0x00, 0x00, 0x0D, 0x00, 0xAF, 0xFF,
+ 0x09, 0x01, 0x6E, 0xFD, 0xB4, 0x05, 0xBC, 0xF2, 0x73, 0x3B, 0x64,
+ 0x23, 0xD2, 0xF2, 0xFB, 0x06, 0x34, 0xFC, 0xE6, 0x01, 0x38, 0xFF,
+ 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x29, 0x00, 0x5F, 0xFF, 0x71,
+ 0x01, 0x65, 0xFD, 0x29, 0x04, 0x96, 0xF9, 0x95, 0x0B, 0xDC, 0x47,
+ 0x03, 0xFD, 0xD9, 0xFF, 0xEA, 0x00, 0x12, 0xFF, 0xA7, 0x00, 0xAE,
+ 0xFF, 0x14, 0x00, 0xFE, 0xFF, 0x20, 0x00, 0x79, 0xFF, 0x72, 0x01,
+ 0xC4, 0xFC, 0xA4, 0x06, 0xAB, 0xF1, 0x46, 0x35, 0xF7, 0x2A, 0xC6,
+ 0xF1, 0x2A, 0x07, 0x40, 0xFC, 0xCF, 0x01, 0x47, 0xFF, 0x31, 0x00,
+ 0xFD, 0xFF, 0x00, 0x00, 0x20, 0x00, 0x7D, 0xFF, 0x24, 0x01, 0x0C,
+ 0xFE, 0xDE, 0x02, 0x2E, 0xFC, 0x13, 0x05, 0xEC, 0x48, 0x54, 0x02,
+ 0x5E, 0xFD, 0x3F, 0x02, 0x5D, 0xFE, 0xFE, 0x00, 0x8C, 0xFF, 0x1C,
+ 0x00, 0xFD, 0xFF, 0x2D, 0x00, 0x53, 0xFF, 0xBA, 0x01, 0x5B, 0xFC,
+ 0x1B, 0x07, 0x8B, 0xF1, 0x58, 0x2E, 0x26, 0x32, 0x80, 0xF1, 0xEA,
+ 0x06, 0x8C, 0xFC, 0x97, 0x01, 0x66, 0xFF, 0x27, 0x00, 0xFD, 0xFF,
+ 0x17, 0x00, 0x9E, 0xFF, 0xCF, 0x00, 0xBF, 0xFE, 0x86, 0x01, 0xBA,
+ 0xFE, 0x5A, 0xFF, 0x86, 0x48, 0x7D, 0x08, 0xC7, 0xFA, 0x93, 0x03,
+ 0xB0, 0xFD, 0x4F, 0x01, 0x6C, 0xFF, 0x25, 0x00, 0x00, 0x00, 0xFD,
+ 0xFF, 0x35, 0x00, 0x3D, 0xFF, 0xDF, 0x01, 0x32, 0xFC, 0x1E, 0x07,
+ 0x40, 0xF2, 0xEB, 0x26, 0xB5, 0x38, 0x1F, 0xF2, 0x32, 0x06, 0x18,
+ 0xFD, 0x3D, 0x01, 0x94, 0xFF, 0x16, 0x00, 0xFF, 0xFF, 0x0F, 0x00,
+ 0xC0, 0xFF, 0x78, 0x00, 0x73, 0xFF, 0x38, 0x00, 0x17, 0x01, 0x8B,
+ 0xFA, 0xAF, 0x46, 0x59, 0x0F, 0x39, 0xF8, 0xCF, 0x04, 0x15, 0xFD,
+ 0x95, 0x01, 0x51, 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0xFD, 0xFF, 0x36,
+ 0x00, 0x36, 0xFF, 0xE5, 0x01, 0x46, 0xFC, 0xB8, 0x06, 0xA8, 0xF3,
+ 0x3F, 0x1F, 0x64, 0x3E, 0xBA, 0xF3, 0x01, 0x05, 0xE2, 0xFD, 0xC4,
+ 0x00, 0xD2, 0xFF, 0x00, 0x00, 0x02, 0x00, 0x08, 0x00, 0xE1, 0xFF,
+ 0x25, 0x00, 0x1D, 0x00, 0x05, 0xFF, 0x28, 0x03, 0xBD, 0xF6, 0x77,
+ 0x43, 0xB6, 0x16, 0xDC, 0xF5, 0xDD, 0x05, 0x9B, 0xFC, 0xC8, 0x01,
+ 0x3E, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x34, 0x00, 0x3D,
+ 0xFF, 0xCC, 0x01, 0x8F, 0xFC, 0xF8, 0x05, 0x9B, 0xF5, 0x96, 0x17,
+ 0x02, 0x43, 0x5E, 0xF6, 0x5F, 0x03, 0xE4, 0xFE, 0x30, 0x00, 0x1B,
+ 0x00, 0xE4, 0xFF, 0x08, 0x00, 0x03, 0x00, 0xFD, 0xFF, 0xD9, 0xFF,
+ 0xB4, 0x00, 0xFD, 0xFD, 0xD7, 0x04, 0xFA, 0xF3, 0xFC, 0x3E, 0x5B,
+ 0x1E, 0xDB, 0xF3, 0xA6, 0x06, 0x4C, 0xFC, 0xE3, 0x01, 0x36, 0xFF,
+ 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x2E, 0x00, 0x4E, 0xFF, 0x9C,
+ 0x01, 0x05, 0xFD, 0xF1, 0x04, 0xF0, 0xF7, 0x2D, 0x10, 0x61, 0x46,
+ 0x0D, 0xFA, 0x58, 0x01, 0x13, 0x00, 0x87, 0xFF, 0x6E, 0x00, 0xC4,
+ 0xFF, 0x0E, 0x00, 0xFF, 0xFF, 0x14, 0x00, 0x9B, 0xFF, 0x31, 0x01,
+ 0x2C, 0xFD, 0x15, 0x06, 0x41, 0xF2, 0x6A, 0x39, 0x0A, 0x26, 0x61,
+ 0xF2, 0x17, 0x07, 0x31, 0xFC, 0xE2, 0x01, 0x3B, 0xFF, 0x35, 0x00,
+ 0xFD, 0xFF, 0x00, 0x00, 0x26, 0x00, 0x69, 0xFF, 0x58, 0x01, 0x9D,
+ 0xFD, 0xB9, 0x03, 0x7B, 0xFA, 0x40, 0x09, 0x63, 0x48, 0xBF, 0xFE,
+ 0x03, 0xFF, 0x5F, 0x01, 0xD4, 0xFE, 0xC5, 0x00, 0xA2, 0xFF, 0x16,
+ 0x00, 0xFD, 0xFF, 0x25, 0x00, 0x6A, 0xFF, 0x8E, 0x01, 0x99, 0xFC,
+ 0xDB, 0x06, 0x86, 0xF1, 0xF2, 0x32, 0x82, 0x2D, 0x96, 0xF1, 0x21,
+ 0x07, 0x53, 0xFC, 0xC0, 0x01, 0x50, 0xFF, 0x2E, 0x00, 0xFD, 0xFF,
+ 0x1D, 0x00, 0x88, 0xFF, 0x07, 0x01, 0x49, 0xFE, 0x67, 0x02, 0x13,
+ 0xFD, 0xFF, 0x02, 0xF4, 0x48, 0x5F, 0x04, 0x7A, 0xFC, 0xB6, 0x02,
+ 0x20, 0xFE, 0x1B, 0x01, 0x81, 0xFF, 0x1F, 0x00, 0x00, 0x00, 0xFD,
+ 0xFF, 0x30, 0x00, 0x4A, 0xFF, 0xCA, 0x01, 0x46, 0xFC, 0x29, 0x07,
+ 0xB3, 0xF1, 0xD1, 0x2B, 0x81, 0x34, 0x9C, 0xF1, 0xB8, 0x06, 0xB5,
+ 0xFC, 0x7C, 0x01, 0x74, 0xFF, 0x22, 0x00, 0xFE, 0xFF, 0x15, 0x00,
+ 0xAA, 0xFF, 0xB1, 0x00, 0xFE, 0xFE, 0x10, 0x01, 0x92, 0xFF, 0x94,
+ 0xFD, 0x0D, 0x48, 0xCB, 0x0A, 0xE2, 0xF9, 0x04, 0x04, 0x77, 0xFD,
+ 0x69, 0x01, 0x62, 0xFF, 0x28, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36,
+ 0x00, 0x39, 0xFF, 0xE5, 0x01, 0x32, 0xFC, 0x06, 0x07, 0xAA, 0xF2,
+ 0x46, 0x24, 0xC8, 0x3A, 0x90, 0xF2, 0xD6, 0x05, 0x57, 0xFD, 0x17,
+ 0x01, 0xA8, 0xFF, 0x0F, 0x00, 0x00, 0x00, 0x0D, 0x00, 0xCC, 0xFF,
+ 0x5A, 0x00, 0xAF, 0xFF, 0xCA, 0xFF, 0xD8, 0x01, 0x1C, 0xF9, 0xB8,
+ 0x45, 0xDA, 0x11, 0x60, 0xF7, 0x33, 0x05, 0xE7, 0xFC, 0xA9, 0x01,
+ 0x4A, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x37,
+ 0xFF, 0xDF, 0x01, 0x5A, 0xFC, 0x7E, 0x06, 0x47, 0xF4, 0x94, 0x1C,
+ 0x1F, 0x40, 0x85, 0xF4, 0x7D, 0x04, 0x36, 0xFE, 0x93, 0x00, 0xEA,
+ 0xFF, 0xF7, 0xFF, 0x04, 0x00, 0x06, 0x00, 0xEB, 0xFF, 0x09, 0x00,
+ 0x54, 0x00, 0xA4, 0xFE, 0xC9, 0x03, 0xAA, 0xF5, 0x0C, 0x42, 0x56,
+ 0x19, 0x1E, 0xF5, 0x2B, 0x06, 0x7A, 0xFC, 0xD4, 0x01, 0x3A, 0xFF,
+ 0x35, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x32, 0x00, 0x42, 0xFF, 0xBE,
+ 0x01, 0xB4, 0xFC, 0xA4, 0x05, 0x61, 0xF6, 0xFB, 0x14, 0x53, 0x44,
+ 0x86, 0xF7, 0xB6, 0x02, 0x49, 0xFF, 0xF7, 0xFF, 0x37, 0x00, 0xD9,
+ 0xFF, 0x0A, 0x00, 0x01, 0x00, 0x06, 0x00, 0xC2, 0xFF, 0xE3, 0x00,
+ 0xAE, 0xFD, 0x52, 0x05, 0x44, 0xF3, 0x2A, 0x3D, 0x06, 0x21, 0x47,
+ 0xF3, 0xD8, 0x06, 0x3C, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00,
+ 0xFD, 0xFF, 0x00, 0x00, 0x2B, 0x00, 0x57, 0xFF, 0x86, 0x01, 0x36,
+ 0xFD, 0x89, 0x04, 0xCD, 0xF8, 0xB7, 0x0D, 0x3D, 0x47, 0x91, 0xFB,
+ 0x91, 0x00, 0x83, 0x00, 0x4A, 0xFF, 0x8C, 0x00, 0xB9, 0xFF, 0x11,
+ 0x00, 0xFE, 0xFF, 0x1B, 0x00, 0x88, 0xFF, 0x55, 0x01, 0xF2, 0xFC,
+ 0x67, 0x06, 0xE4, 0xF1, 0x44, 0x37, 0xAA, 0x28, 0x05, 0xF2, 0x27,
+ 0x07, 0x36, 0xFC, 0xDA, 0x01, 0x41, 0xFF, 0x33, 0x00, 0xFD, 0xFF,
+ 0x00, 0x00, 0x23, 0x00, 0x73, 0xFF, 0x3D, 0x01, 0xD6, 0xFD, 0x46,
+ 0x03, 0x61, 0xFB, 0x00, 0x07, 0xBF, 0x48, 0x98, 0x00, 0x26, 0xFE,
+ 0xD5, 0x01, 0x95, 0xFE, 0xE3, 0x00, 0x96, 0xFF, 0x1A, 0x00, 0xFD,
+ 0xFF, 0x2A, 0x00, 0x5D, 0xFF, 0xA7, 0x01, 0x75, 0xFC, 0x03, 0x07,
+ 0x7D, 0xF1, 0x8A, 0x30, 0xFF, 0x2F, 0x7E, 0xF1, 0x0A, 0x07, 0x6E,
+ 0xFC, 0xAC, 0x01, 0x5A, 0xFF, 0x2B, 0x00, 0xFD, 0xFF, 0x1A, 0x00,
+ 0x94, 0xFF, 0xEA, 0x00, 0x87, 0xFE, 0xF0, 0x01, 0xF5, 0xFD, 0x05,
+ 0x01, 0xCE, 0x48, 0x83, 0x06, 0x94, 0xFB, 0x2C, 0x03, 0xE4, 0xFD,
+ 0x37, 0x01, 0x76, 0xFF, 0x22, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x33,
+ 0x00, 0x42, 0xFF, 0xD7, 0x01, 0x38, 0xFC, 0x29, 0x07, 0xF3, 0xF1,
+ 0x3E, 0x29, 0xC6, 0x36, 0xD4, 0xF1, 0x77, 0x06, 0xE6, 0xFC, 0x5C,
+ 0x01, 0x84, 0xFF, 0x1C, 0x00, 0xFE, 0xFF, 0x12, 0x00, 0xB6, 0xFF,
+ 0x93, 0x00, 0x3C, 0xFF, 0x9D, 0x00, 0x63, 0x00, 0xEB, 0xFB, 0x69,
+ 0x47, 0x2D, 0x0D, 0xFF, 0xF8, 0x72, 0x04, 0x42, 0xFD, 0x81, 0x01,
+ 0x59, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x37,
+ 0xFF, 0xE6, 0x01, 0x3A, 0xFC, 0xE2, 0x06, 0x28, 0xF3, 0x9E, 0x21,
+ 0xC0, 0x3C, 0x1F, 0xF3, 0x6C, 0x05, 0x9E, 0xFD, 0xED, 0x00, 0xBD,
+ 0xFF, 0x07, 0x00, 0x01, 0x00, 0x0A, 0x00, 0xD7, 0xFF, 0x3E, 0x00,
+ 0xEA, 0xFF, 0x60, 0xFF, 0x8F, 0x02, 0xCD, 0xF7, 0x99, 0x44, 0x68,
+ 0x14, 0x8E, 0xF6, 0x90, 0x05, 0xBC, 0xFC, 0xBA, 0x01, 0x43, 0xFF,
+ 0x32, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x35, 0x00, 0x39, 0xFF, 0xD7,
+ 0x01, 0x73, 0xFC, 0x3B, 0x06, 0xF5, 0xF4, 0xED, 0x19, 0xB7, 0x41,
+ 0x71, 0xF5, 0xEB, 0x03, 0x90, 0xFE, 0x60, 0x00, 0x04, 0x00, 0xED,
+ 0xFF, 0x06, 0x00, 0x04, 0x00, 0xF5, 0xFF, 0xEF, 0xFF, 0x88, 0x00,
+ 0x49, 0xFE, 0x5D, 0x04, 0xB7, 0xF4, 0x7D, 0x40, 0xFD, 0x1B, 0x6C,
+ 0xF4, 0x70, 0x06, 0x5F, 0xFC, 0xDE, 0x01, 0x37, 0xFF, 0x36, 0x00,
+ 0xFE, 0xFF, 0xFF, 0xFF, 0x30, 0x00, 0x48, 0xFF, 0xAD, 0x01, 0xDD,
+ 0xFC, 0x48, 0x05, 0x30, 0xF7, 0x6B, 0x12, 0x7D, 0x45, 0xCF, 0xF8,
+ 0x01, 0x02, 0xB2, 0xFF, 0xBD, 0xFF, 0x54, 0x00, 0xCE, 0xFF, 0x0C,
+ 0x00, 0x00, 0x00, 0x0E, 0x00, 0xAC, 0xFF, 0x0E, 0x01, 0x66, 0xFD,
+ 0xBF, 0x05, 0xAD, 0xF2, 0x3B, 0x3B, 0xB0, 0x23, 0xC4, 0xF2, 0xFF,
+ 0x06, 0x33, 0xFC, 0xE5, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0x00, 0x00, 0x29, 0x00, 0x60, 0xFF, 0x6E, 0x01, 0x6B, 0xFD, 0x1D,
+ 0x04, 0xAF, 0xF9, 0x51, 0x0B, 0xEC, 0x47, 0x33, 0xFD, 0xC1, 0xFF,
+ 0xF7, 0x00, 0x0C, 0xFF, 0xAA, 0x00, 0xAD, 0xFF, 0x14, 0x00, 0xFE,
+ 0xFF, 0x21, 0x00, 0x77, 0xFF, 0x75, 0x01, 0xBF, 0xFC, 0xAB, 0x06,
+ 0xA6, 0xF1, 0x05, 0x35, 0x40, 0x2B, 0xBF, 0xF1, 0x2A, 0x07, 0x42,
+ 0xFC, 0xCE, 0x01, 0x48, 0xFF, 0x31, 0x00, 0xFD, 0xFF, 0x00, 0x00,
+ 0x20, 0x00, 0x7E, 0xFF, 0x21, 0x01, 0x12, 0xFE, 0xD1, 0x02, 0x47,
+ 0xFC, 0xD7, 0x04, 0xF0, 0x48, 0x8D, 0x02, 0x45, 0xFD, 0x4D, 0x02,
+ 0x56, 0xFE, 0x01, 0x01, 0x8B, 0xFF, 0x1D, 0x00, 0xFD, 0xFF, 0x2E,
+ 0x00, 0x52, 0xFF, 0xBC, 0x01, 0x58, 0xFC, 0x1D, 0x07, 0x8E, 0xF1,
+ 0x11, 0x2E, 0x6B, 0x32, 0x81, 0xF1, 0xE5, 0x06, 0x90, 0xFC, 0x94,
+ 0x01, 0x67, 0xFF, 0x26, 0x00, 0xFD, 0xFF, 0x17, 0x00, 0xA0, 0xFF,
+ 0xCC, 0x00, 0xC6, 0xFE, 0x79, 0x01, 0xD2, 0xFE, 0x26, 0xFF, 0x7C,
+ 0x48, 0xBE, 0x08, 0xAE, 0xFA, 0xA0, 0x03, 0xA9, 0xFD, 0x52, 0x01,
+ 0x6B, 0xFF, 0x25, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3C,
+ 0xFF, 0xE0, 0x01, 0x32, 0xFC, 0x1C, 0x07, 0x4B, 0xF2, 0xA0, 0x26,
+ 0xF2, 0x38, 0x2A, 0xF2, 0x28, 0x06, 0x1F, 0xFD, 0x39, 0x01, 0x96,
+ 0xFF, 0x16, 0x00, 0xFF, 0xFF, 0x0F, 0x00, 0xC2, 0xFF, 0x75, 0x00,
+ 0x7A, 0xFF, 0x2B, 0x00, 0x2D, 0x01, 0x61, 0xFA, 0x97, 0x46, 0xA0,
+ 0x0F, 0x20, 0xF8, 0xDA, 0x04, 0x10, 0xFD, 0x97, 0x01, 0x50, 0xFF,
+ 0x2E, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE4,
+ 0x01, 0x48, 0xFC, 0xB2, 0x06, 0xB9, 0xF3, 0xF3, 0x1E, 0x98, 0x3E,
+ 0xCF, 0xF3, 0xF3, 0x04, 0xEB, 0xFD, 0xBF, 0x00, 0xD4, 0xFF, 0xFF,
+ 0xFF, 0x03, 0x00, 0x08, 0x00, 0xE2, 0xFF, 0x21, 0x00, 0x23, 0x00,
+ 0xFA, 0xFE, 0x3A, 0x03, 0x9D, 0xF6, 0x50, 0x43, 0x00, 0x17, 0xC6,
+ 0xF5, 0xE6, 0x05, 0x97, 0xFC, 0xC9, 0x01, 0x3E, 0xFF, 0x34, 0x00,
+ 0xFE, 0xFF, 0xFE, 0xFF, 0x34, 0x00, 0x3D, 0xFF, 0xCB, 0x01, 0x93,
+ 0xFC, 0xEF, 0x05, 0xB0, 0xF5, 0x4B, 0x17, 0x2A, 0x43, 0x7D, 0xF6,
+ 0x4D, 0x03, 0xEF, 0xFE, 0x2A, 0x00, 0x1E, 0x00, 0xE3, 0xFF, 0x08,
+ 0x00, 0x03, 0x00, 0xFE, 0xFF, 0xD7, 0xFF, 0xBA, 0x00, 0xF4, 0xFD,
+ 0xE5, 0x04, 0xE4, 0xF3, 0xCA, 0x3E, 0xA7, 0x1E, 0xCA, 0xF3, 0xAC,
+ 0x06, 0x4A, 0xFC, 0xE4, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF,
+ 0xFF, 0xFF, 0x2E, 0x00, 0x4F, 0xFF, 0x99, 0x01, 0x0B, 0xFD, 0xE6,
+ 0x04, 0x08, 0xF8, 0xE7, 0x0F, 0x7C, 0x46, 0x37, 0xFA, 0x42, 0x01,
+ 0x1F, 0x00, 0x81, 0xFF, 0x71, 0x00, 0xC3, 0xFF, 0x0F, 0x00, 0xFF,
+ 0xFF, 0x15, 0x00, 0x98, 0xFF, 0x35, 0x01, 0x25, 0xFD, 0x1E, 0x06,
+ 0x35, 0xF2, 0x2E, 0x39, 0x55, 0x26, 0x56, 0xF2, 0x1A, 0x07, 0x31,
+ 0xFC, 0xE1, 0x01, 0x3C, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x00, 0x00,
+ 0x26, 0x00, 0x6A, 0xFF, 0x55, 0x01, 0xA3, 0xFD, 0xAD, 0x03, 0x94,
+ 0xFA, 0xFF, 0x08, 0x70, 0x48, 0xF3, 0xFE, 0xEA, 0xFE, 0x6C, 0x01,
+ 0xCD, 0xFE, 0xC9, 0x00, 0xA1, 0xFF, 0x17, 0x00, 0xFD, 0xFF, 0x26,
+ 0x00, 0x69, 0xFF, 0x91, 0x01, 0x94, 0xFC, 0xE0, 0x06, 0x84, 0xF1,
+ 0xAF, 0x32, 0xCA, 0x2D, 0x92, 0xF1, 0x1F, 0x07, 0x56, 0xFC, 0xBE,
+ 0x01, 0x51, 0xFF, 0x2E, 0x00, 0xFD, 0xFF, 0x1D, 0x00, 0x8A, 0xFF,
+ 0x04, 0x01, 0x50, 0xFE, 0x5A, 0x02, 0x2C, 0xFD, 0xC6, 0x02, 0xF2,
+ 0x48, 0x9B, 0x04, 0x61, 0xFC, 0xC3, 0x02, 0x19, 0xFE, 0x1E, 0x01,
+ 0x7F, 0xFF, 0x20, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x31, 0x00, 0x49,
+ 0xFF, 0xCC, 0x01, 0x44, 0xFC, 0x29, 0x07, 0xB9, 0xF1, 0x89, 0x2B,
+ 0xC3, 0x34, 0xA0, 0xF1, 0xB1, 0x06, 0xBA, 0xFC, 0x79, 0x01, 0x76,
+ 0xFF, 0x21, 0x00, 0xFE, 0xFF, 0x14, 0x00, 0xAC, 0xFF, 0xAE, 0x00,
+ 0x05, 0xFF, 0x03, 0x01, 0xAA, 0xFF, 0x63, 0xFD, 0xFD, 0x47, 0x0E,
+ 0x0B, 0xC8, 0xF9, 0x11, 0x04, 0x71, 0xFD, 0x6C, 0x01, 0x61, 0xFF,
+ 0x28, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x39, 0xFF, 0xE5,
+ 0x01, 0x33, 0xFC, 0x03, 0x07, 0xB7, 0xF2, 0xFC, 0x23, 0x03, 0x3B,
+ 0x9E, 0xF2, 0xCB, 0x05, 0x5F, 0xFD, 0x12, 0x01, 0xAA, 0xFF, 0x0E,
+ 0x00, 0x00, 0x00, 0x0C, 0x00, 0xCD, 0xFF, 0x57, 0x00, 0xB6, 0xFF,
+ 0xBE, 0xFF, 0xED, 0x01, 0xF5, 0xF8, 0x9B, 0x45, 0x22, 0x12, 0x48,
+ 0xF7, 0x3D, 0x05, 0xE2, 0xFC, 0xAB, 0x01, 0x49, 0xFF, 0x30, 0x00,
+ 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xDF, 0x01, 0x5C,
+ 0xFC, 0x78, 0x06, 0x5A, 0xF4, 0x49, 0x1C, 0x4E, 0x40, 0x9E, 0xF4,
+ 0x6D, 0x04, 0x3F, 0xFE, 0x8E, 0x00, 0xED, 0xFF, 0xF6, 0xFF, 0x04,
+ 0x00, 0x06, 0x00, 0xEC, 0xFF, 0x06, 0x00, 0x5A, 0x00, 0x9A, 0xFE,
+ 0xDA, 0x03, 0x8D, 0xF5, 0xE1, 0x41, 0xA1, 0x19, 0x09, 0xF5, 0x33,
+ 0x06, 0x77, 0xFC, 0xD6, 0x01, 0x3A, 0xFF, 0x35, 0x00, 0xFE, 0xFF,
+ 0xFF, 0xFF, 0x32, 0x00, 0x42, 0xFF, 0xBC, 0x01, 0xB8, 0xFC, 0x9A,
+ 0x05, 0x77, 0xF6, 0xB1, 0x14, 0x77, 0x44, 0xA9, 0xF7, 0xA2, 0x02,
+ 0x54, 0xFF, 0xF1, 0xFF, 0x3A, 0x00, 0xD8, 0xFF, 0x0A, 0x00, 0x01,
+ 0x00, 0x07, 0x00, 0xC0, 0xFF, 0xE8, 0x00, 0xA6, 0xFD, 0x5F, 0x05,
+ 0x31, 0xF3, 0xF6, 0x3C, 0x52, 0x21, 0x37, 0xF3, 0xDD, 0x06, 0x3B,
+ 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00,
+ 0x2B, 0x00, 0x58, 0xFF, 0x83, 0x01, 0x3C, 0xFD, 0x7E, 0x04, 0xE6,
+ 0xF8, 0x72, 0x0D, 0x52, 0x47, 0xBE, 0xFB, 0x7A, 0x00, 0x90, 0x00,
+ 0x43, 0xFF, 0x8F, 0x00, 0xB7, 0xFF, 0x11, 0x00, 0xFE, 0xFF, 0x1C,
+ 0x00, 0x86, 0xFF, 0x59, 0x01, 0xEC, 0xFC, 0x6F, 0x06, 0xDC, 0xF1,
+ 0x04, 0x37, 0xF3, 0x28, 0xFC, 0xF1, 0x28, 0x07, 0x37, 0xFC, 0xD8,
+ 0x01, 0x41, 0xFF, 0x33, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x23, 0x00,
+ 0x74, 0xFF, 0x3A, 0x01, 0xDD, 0xFD, 0x39, 0x03, 0x7B, 0xFB, 0xC1,
+ 0x06, 0xC7, 0x48, 0xCF, 0x00, 0x0D, 0xFE, 0xE3, 0x01, 0x8E, 0xFE,
+ 0xE7, 0x00, 0x95, 0xFF, 0x1A, 0x00, 0xFD, 0xFF, 0x2A, 0x00, 0x5C,
+ 0xFF, 0xAA, 0x01, 0x71, 0xFC, 0x07, 0x07, 0x7E, 0xF1, 0x44, 0x30,
+ 0x44, 0x30, 0x7E, 0xF1, 0x07, 0x07, 0x71, 0xFC, 0xAA, 0x01, 0x5C,
+ 0xFF, 0x2A, 0x00, 0xFD, 0xFF, 0x1A, 0x00, 0x95, 0xFF, 0xE7, 0x00,
+ 0x8E, 0xFE, 0xE3, 0x01, 0x0D, 0xFE, 0xCF, 0x00, 0xC7, 0x48, 0xC1,
+ 0x06, 0x7B, 0xFB, 0x39, 0x03, 0xDD, 0xFD, 0x3A, 0x01, 0x74, 0xFF,
+ 0x23, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x33, 0x00, 0x41, 0xFF, 0xD8,
+ 0x01, 0x37, 0xFC, 0x28, 0x07, 0xFC, 0xF1, 0xF3, 0x28, 0x04, 0x37,
+ 0xDC, 0xF1, 0x6F, 0x06, 0xEC, 0xFC, 0x59, 0x01, 0x86, 0xFF, 0x1C,
+ 0x00, 0xFE, 0xFF, 0x11, 0x00, 0xB7, 0xFF, 0x8F, 0x00, 0x43, 0xFF,
+ 0x90, 0x00, 0x7A, 0x00, 0xBE, 0xFB, 0x52, 0x47, 0x72, 0x0D, 0xE6,
+ 0xF8, 0x7E, 0x04, 0x3C, 0xFD, 0x83, 0x01, 0x58, 0xFF, 0x2B, 0x00,
+ 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3B,
+ 0xFC, 0xDD, 0x06, 0x37, 0xF3, 0x52, 0x21, 0xF6, 0x3C, 0x31, 0xF3,
+ 0x5F, 0x05, 0xA6, 0xFD, 0xE8, 0x00, 0xC0, 0xFF, 0x07, 0x00, 0x01,
+ 0x00, 0x0A, 0x00, 0xD8, 0xFF, 0x3A, 0x00, 0xF1, 0xFF, 0x54, 0xFF,
+ 0xA2, 0x02, 0xA9, 0xF7, 0x77, 0x44, 0xB1, 0x14, 0x77, 0xF6, 0x9A,
+ 0x05, 0xB8, 0xFC, 0xBC, 0x01, 0x42, 0xFF, 0x32, 0x00, 0xFF, 0xFF,
+ 0xFE, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xD6, 0x01, 0x77, 0xFC, 0x33,
+ 0x06, 0x09, 0xF5, 0xA1, 0x19, 0xE1, 0x41, 0x8D, 0xF5, 0xDA, 0x03,
+ 0x9A, 0xFE, 0x5A, 0x00, 0x06, 0x00, 0xEC, 0xFF, 0x06, 0x00, 0x04,
+ 0x00, 0xF6, 0xFF, 0xED, 0xFF, 0x8E, 0x00, 0x3F, 0xFE, 0x6D, 0x04,
+ 0x9E, 0xF4, 0x4E, 0x40, 0x49, 0x1C, 0x5A, 0xF4, 0x78, 0x06, 0x5C,
+ 0xFC, 0xDF, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF,
+ 0x30, 0x00, 0x49, 0xFF, 0xAB, 0x01, 0xE2, 0xFC, 0x3D, 0x05, 0x48,
+ 0xF7, 0x22, 0x12, 0x9B, 0x45, 0xF5, 0xF8, 0xED, 0x01, 0xBE, 0xFF,
+ 0xB6, 0xFF, 0x57, 0x00, 0xCD, 0xFF, 0x0C, 0x00, 0x00, 0x00, 0x0E,
+ 0x00, 0xAA, 0xFF, 0x12, 0x01, 0x5F, 0xFD, 0xCB, 0x05, 0x9E, 0xF2,
+ 0x03, 0x3B, 0xFC, 0x23, 0xB7, 0xF2, 0x03, 0x07, 0x33, 0xFC, 0xE5,
+ 0x01, 0x39, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x28, 0x00,
+ 0x61, 0xFF, 0x6C, 0x01, 0x71, 0xFD, 0x11, 0x04, 0xC8, 0xF9, 0x0E,
+ 0x0B, 0xFD, 0x47, 0x63, 0xFD, 0xAA, 0xFF, 0x03, 0x01, 0x05, 0xFF,
+ 0xAE, 0x00, 0xAC, 0xFF, 0x14, 0x00, 0xFE, 0xFF, 0x21, 0x00, 0x76,
+ 0xFF, 0x79, 0x01, 0xBA, 0xFC, 0xB1, 0x06, 0xA0, 0xF1, 0xC3, 0x34,
+ 0x89, 0x2B, 0xB9, 0xF1, 0x29, 0x07, 0x44, 0xFC, 0xCC, 0x01, 0x49,
+ 0xFF, 0x31, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x20, 0x00, 0x7F, 0xFF,
+ 0x1E, 0x01, 0x19, 0xFE, 0xC3, 0x02, 0x61, 0xFC, 0x9B, 0x04, 0xF2,
+ 0x48, 0xC6, 0x02, 0x2C, 0xFD, 0x5A, 0x02, 0x50, 0xFE, 0x04, 0x01,
+ 0x8A, 0xFF, 0x1D, 0x00, 0xFD, 0xFF, 0x2E, 0x00, 0x51, 0xFF, 0xBE,
+ 0x01, 0x56, 0xFC, 0x1F, 0x07, 0x92, 0xF1, 0xCA, 0x2D, 0xAF, 0x32,
+ 0x84, 0xF1, 0xE0, 0x06, 0x94, 0xFC, 0x91, 0x01, 0x69, 0xFF, 0x26,
+ 0x00, 0xFD, 0xFF, 0x17, 0x00, 0xA1, 0xFF, 0xC9, 0x00, 0xCD, 0xFE,
+ 0x6C, 0x01, 0xEA, 0xFE, 0xF3, 0xFE, 0x70, 0x48, 0xFF, 0x08, 0x94,
+ 0xFA, 0xAD, 0x03, 0xA3, 0xFD, 0x55, 0x01, 0x6A, 0xFF, 0x26, 0x00,
+ 0x00, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3C, 0xFF, 0xE1, 0x01, 0x31,
+ 0xFC, 0x1A, 0x07, 0x56, 0xF2, 0x55, 0x26, 0x2E, 0x39, 0x35, 0xF2,
+ 0x1E, 0x06, 0x25, 0xFD, 0x35, 0x01, 0x98, 0xFF, 0x15, 0x00, 0xFF,
+ 0xFF, 0x0F, 0x00, 0xC3, 0xFF, 0x71, 0x00, 0x81, 0xFF, 0x1F, 0x00,
+ 0x42, 0x01, 0x37, 0xFA, 0x7C, 0x46, 0xE7, 0x0F, 0x08, 0xF8, 0xE6,
+ 0x04, 0x0B, 0xFD, 0x99, 0x01, 0x4F, 0xFF, 0x2E, 0x00, 0xFF, 0xFF,
+ 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE4, 0x01, 0x4A, 0xFC, 0xAC,
+ 0x06, 0xCA, 0xF3, 0xA7, 0x1E, 0xCA, 0x3E, 0xE4, 0xF3, 0xE5, 0x04,
+ 0xF4, 0xFD, 0xBA, 0x00, 0xD7, 0xFF, 0xFE, 0xFF, 0x03, 0x00, 0x08,
+ 0x00, 0xE3, 0xFF, 0x1E, 0x00, 0x2A, 0x00, 0xEF, 0xFE, 0x4D, 0x03,
+ 0x7D, 0xF6, 0x2A, 0x43, 0x4B, 0x17, 0xB0, 0xF5, 0xEF, 0x05, 0x93,
+ 0xFC, 0xCB, 0x01, 0x3D, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0xFE, 0xFF,
+ 0x34, 0x00, 0x3E, 0xFF, 0xC9, 0x01, 0x97, 0xFC, 0xE6, 0x05, 0xC6,
+ 0xF5, 0x00, 0x17, 0x50, 0x43, 0x9D, 0xF6, 0x3A, 0x03, 0xFA, 0xFE,
+ 0x23, 0x00, 0x21, 0x00, 0xE2, 0xFF, 0x08, 0x00, 0x03, 0x00, 0xFF,
+ 0xFF, 0xD4, 0xFF, 0xBF, 0x00, 0xEB, 0xFD, 0xF3, 0x04, 0xCF, 0xF3,
+ 0x98, 0x3E, 0xF3, 0x1E, 0xB9, 0xF3, 0xB2, 0x06, 0x48, 0xFC, 0xE4,
+ 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x2E, 0x00,
+ 0x50, 0xFF, 0x97, 0x01, 0x10, 0xFD, 0xDA, 0x04, 0x20, 0xF8, 0xA0,
+ 0x0F, 0x97, 0x46, 0x61, 0xFA, 0x2D, 0x01, 0x2B, 0x00, 0x7A, 0xFF,
+ 0x75, 0x00, 0xC2, 0xFF, 0x0F, 0x00, 0xFF, 0xFF, 0x16, 0x00, 0x96,
+ 0xFF, 0x39, 0x01, 0x1F, 0xFD, 0x28, 0x06, 0x2A, 0xF2, 0xF2, 0x38,
+ 0xA0, 0x26, 0x4B, 0xF2, 0x1C, 0x07, 0x32, 0xFC, 0xE0, 0x01, 0x3C,
+ 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x25, 0x00, 0x6B, 0xFF,
+ 0x52, 0x01, 0xA9, 0xFD, 0xA0, 0x03, 0xAE, 0xFA, 0xBE, 0x08, 0x7C,
+ 0x48, 0x26, 0xFF, 0xD2, 0xFE, 0x79, 0x01, 0xC6, 0xFE, 0xCC, 0x00,
+ 0xA0, 0xFF, 0x17, 0x00, 0xFD, 0xFF, 0x26, 0x00, 0x67, 0xFF, 0x94,
+ 0x01, 0x90, 0xFC, 0xE5, 0x06, 0x81, 0xF1, 0x6B, 0x32, 0x11, 0x2E,
+ 0x8E, 0xF1, 0x1D, 0x07, 0x58, 0xFC, 0xBC, 0x01, 0x52, 0xFF, 0x2E,
+ 0x00, 0xFD, 0xFF, 0x1D, 0x00, 0x8B, 0xFF, 0x01, 0x01, 0x56, 0xFE,
+ 0x4D, 0x02, 0x45, 0xFD, 0x8D, 0x02, 0xF0, 0x48, 0xD7, 0x04, 0x47,
+ 0xFC, 0xD1, 0x02, 0x12, 0xFE, 0x21, 0x01, 0x7E, 0xFF, 0x20, 0x00,
+ 0x00, 0x00, 0xFD, 0xFF, 0x31, 0x00, 0x48, 0xFF, 0xCE, 0x01, 0x42,
+ 0xFC, 0x2A, 0x07, 0xBF, 0xF1, 0x40, 0x2B, 0x05, 0x35, 0xA6, 0xF1,
+ 0xAB, 0x06, 0xBF, 0xFC, 0x75, 0x01, 0x77, 0xFF, 0x21, 0x00, 0xFE,
+ 0xFF, 0x14, 0x00, 0xAD, 0xFF, 0xAA, 0x00, 0x0C, 0xFF, 0xF7, 0x00,
+ 0xC1, 0xFF, 0x33, 0xFD, 0xEC, 0x47, 0x51, 0x0B, 0xAF, 0xF9, 0x1D,
+ 0x04, 0x6B, 0xFD, 0x6E, 0x01, 0x60, 0xFF, 0x29, 0x00, 0x00, 0x00,
+ 0xFD, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xE5, 0x01, 0x33, 0xFC, 0xFF,
+ 0x06, 0xC4, 0xF2, 0xB0, 0x23, 0x3B, 0x3B, 0xAD, 0xF2, 0xBF, 0x05,
+ 0x66, 0xFD, 0x0E, 0x01, 0xAC, 0xFF, 0x0E, 0x00, 0x00, 0x00, 0x0C,
+ 0x00, 0xCE, 0xFF, 0x54, 0x00, 0xBD, 0xFF, 0xB2, 0xFF, 0x01, 0x02,
+ 0xCF, 0xF8, 0x7D, 0x45, 0x6B, 0x12, 0x30, 0xF7, 0x48, 0x05, 0xDD,
+ 0xFC, 0xAD, 0x01, 0x48, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0xFE, 0xFF,
+ 0x36, 0x00, 0x37, 0xFF, 0xDE, 0x01, 0x5F, 0xFC, 0x70, 0x06, 0x6C,
+ 0xF4, 0xFD, 0x1B, 0x7D, 0x40, 0xB7, 0xF4, 0x5D, 0x04, 0x49, 0xFE,
+ 0x88, 0x00, 0xEF, 0xFF, 0xF5, 0xFF, 0x04, 0x00, 0x06, 0x00, 0xED,
+ 0xFF, 0x04, 0x00, 0x60, 0x00, 0x90, 0xFE, 0xEB, 0x03, 0x71, 0xF5,
+ 0xB7, 0x41, 0xED, 0x19, 0xF5, 0xF4, 0x3B, 0x06, 0x73, 0xFC, 0xD7,
+ 0x01, 0x39, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x32, 0x00,
+ 0x43, 0xFF, 0xBA, 0x01, 0xBC, 0xFC, 0x90, 0x05, 0x8E, 0xF6, 0x68,
+ 0x14, 0x99, 0x44, 0xCD, 0xF7, 0x8F, 0x02, 0x60, 0xFF, 0xEA, 0xFF,
+ 0x3E, 0x00, 0xD7, 0xFF, 0x0A, 0x00, 0x01, 0x00, 0x07, 0x00, 0xBD,
+ 0xFF, 0xED, 0x00, 0x9E, 0xFD, 0x6C, 0x05, 0x1F, 0xF3, 0xC0, 0x3C,
+ 0x9E, 0x21, 0x28, 0xF3, 0xE2, 0x06, 0x3A, 0xFC, 0xE6, 0x01, 0x37,
+ 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x2B, 0x00, 0x59, 0xFF,
+ 0x81, 0x01, 0x42, 0xFD, 0x72, 0x04, 0xFF, 0xF8, 0x2D, 0x0D, 0x69,
+ 0x47, 0xEB, 0xFB, 0x63, 0x00, 0x9D, 0x00, 0x3C, 0xFF, 0x93, 0x00,
+ 0xB6, 0xFF, 0x12, 0x00, 0xFE, 0xFF, 0x1C, 0x00, 0x84, 0xFF, 0x5C,
+ 0x01, 0xE6, 0xFC, 0x77, 0x06, 0xD4, 0xF1, 0xC6, 0x36, 0x3E, 0x29,
+ 0xF3, 0xF1, 0x29, 0x07, 0x38, 0xFC, 0xD7, 0x01, 0x42, 0xFF, 0x33,
+ 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x22, 0x00, 0x76, 0xFF, 0x37, 0x01,
+ 0xE4, 0xFD, 0x2C, 0x03, 0x94, 0xFB, 0x83, 0x06, 0xCE, 0x48, 0x05,
+ 0x01, 0xF5, 0xFD, 0xF0, 0x01, 0x87, 0xFE, 0xEA, 0x00, 0x94, 0xFF,
+ 0x1A, 0x00, 0xFD, 0xFF, 0x2B, 0x00, 0x5A, 0xFF, 0xAC, 0x01, 0x6E,
+ 0xFC, 0x0A, 0x07, 0x7E, 0xF1, 0xFF, 0x2F, 0x8A, 0x30, 0x7D, 0xF1,
+ 0x03, 0x07, 0x75, 0xFC, 0xA7, 0x01, 0x5D, 0xFF, 0x2A, 0x00, 0xFD,
+ 0xFF, 0x1A, 0x00, 0x96, 0xFF, 0xE3, 0x00, 0x95, 0xFE, 0xD5, 0x01,
+ 0x26, 0xFE, 0x98, 0x00, 0xBF, 0x48, 0x00, 0x07, 0x61, 0xFB, 0x46,
+ 0x03, 0xD6, 0xFD, 0x3D, 0x01, 0x73, 0xFF, 0x23, 0x00, 0x00, 0x00,
+ 0xFD, 0xFF, 0x33, 0x00, 0x41, 0xFF, 0xDA, 0x01, 0x36, 0xFC, 0x27,
+ 0x07, 0x05, 0xF2, 0xAA, 0x28, 0x44, 0x37, 0xE4, 0xF1, 0x67, 0x06,
+ 0xF2, 0xFC, 0x55, 0x01, 0x88, 0xFF, 0x1B, 0x00, 0xFE, 0xFF, 0x11,
+ 0x00, 0xB9, 0xFF, 0x8C, 0x00, 0x4A, 0xFF, 0x83, 0x00, 0x91, 0x00,
+ 0x91, 0xFB, 0x3D, 0x47, 0xB7, 0x0D, 0xCD, 0xF8, 0x89, 0x04, 0x36,
+ 0xFD, 0x86, 0x01, 0x57, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0xFD, 0xFF,
+ 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3C, 0xFC, 0xD8, 0x06, 0x47,
+ 0xF3, 0x06, 0x21, 0x2A, 0x3D, 0x44, 0xF3, 0x52, 0x05, 0xAE, 0xFD,
+ 0xE3, 0x00, 0xC2, 0xFF, 0x06, 0x00, 0x01, 0x00, 0x0A, 0x00, 0xD9,
+ 0xFF, 0x37, 0x00, 0xF7, 0xFF, 0x49, 0xFF, 0xB6, 0x02, 0x86, 0xF7,
+ 0x53, 0x44, 0xFB, 0x14, 0x61, 0xF6, 0xA4, 0x05, 0xB4, 0xFC, 0xBE,
+ 0x01, 0x42, 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x35, 0x00,
+ 0x3A, 0xFF, 0xD4, 0x01, 0x7A, 0xFC, 0x2B, 0x06, 0x1E, 0xF5, 0x56,
+ 0x19, 0x0C, 0x42, 0xAA, 0xF5, 0xC9, 0x03, 0xA4, 0xFE, 0x54, 0x00,
+ 0x09, 0x00, 0xEB, 0xFF, 0x06, 0x00, 0x04, 0x00, 0xF7, 0xFF, 0xEA,
+ 0xFF, 0x93, 0x00, 0x36, 0xFE, 0x7D, 0x04, 0x85, 0xF4, 0x1F, 0x40,
+ 0x94, 0x1C, 0x47, 0xF4, 0x7E, 0x06, 0x5A, 0xFC, 0xDF, 0x01, 0x37,
+ 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x30, 0x00, 0x4A, 0xFF,
+ 0xA9, 0x01, 0xE7, 0xFC, 0x33, 0x05, 0x60, 0xF7, 0xDA, 0x11, 0xB8,
+ 0x45, 0x1C, 0xF9, 0xD8, 0x01, 0xCA, 0xFF, 0xAF, 0xFF, 0x5A, 0x00,
+ 0xCC, 0xFF, 0x0D, 0x00, 0x00, 0x00, 0x0F, 0x00, 0xA8, 0xFF, 0x17,
+ 0x01, 0x57, 0xFD, 0xD6, 0x05, 0x90, 0xF2, 0xC8, 0x3A, 0x46, 0x24,
+ 0xAA, 0xF2, 0x06, 0x07, 0x32, 0xFC, 0xE5, 0x01, 0x39, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x28, 0x00, 0x62, 0xFF, 0x69, 0x01,
+ 0x77, 0xFD, 0x04, 0x04, 0xE2, 0xF9, 0xCB, 0x0A, 0x0D, 0x48, 0x94,
+ 0xFD, 0x92, 0xFF, 0x10, 0x01, 0xFE, 0xFE, 0xB1, 0x00, 0xAA, 0xFF,
+ 0x15, 0x00, 0xFE, 0xFF, 0x22, 0x00, 0x74, 0xFF, 0x7C, 0x01, 0xB5,
+ 0xFC, 0xB8, 0x06, 0x9C, 0xF1, 0x81, 0x34, 0xD1, 0x2B, 0xB3, 0xF1,
+ 0x29, 0x07, 0x46, 0xFC, 0xCA, 0x01, 0x4A, 0xFF, 0x30, 0x00, 0xFD,
+ 0xFF, 0x00, 0x00, 0x1F, 0x00, 0x81, 0xFF, 0x1B, 0x01, 0x20, 0xFE,
+ 0xB6, 0x02, 0x7A, 0xFC, 0x5F, 0x04, 0xF4, 0x48, 0xFF, 0x02, 0x13,
+ 0xFD, 0x67, 0x02, 0x49, 0xFE, 0x07, 0x01, 0x88, 0xFF, 0x1D, 0x00,
+ 0xFD, 0xFF, 0x2E, 0x00, 0x50, 0xFF, 0xC0, 0x01, 0x53, 0xFC, 0x21,
+ 0x07, 0x96, 0xF1, 0x82, 0x2D, 0xF2, 0x32, 0x86, 0xF1, 0xDB, 0x06,
+ 0x99, 0xFC, 0x8E, 0x01, 0x6A, 0xFF, 0x25, 0x00, 0xFD, 0xFF, 0x16,
+ 0x00, 0xA2, 0xFF, 0xC5, 0x00, 0xD4, 0xFE, 0x5F, 0x01, 0x03, 0xFF,
+ 0xBF, 0xFE, 0x63, 0x48, 0x40, 0x09, 0x7B, 0xFA, 0xB9, 0x03, 0x9D,
+ 0xFD, 0x58, 0x01, 0x69, 0xFF, 0x26, 0x00, 0x00, 0x00, 0xFD, 0xFF,
+ 0x35, 0x00, 0x3B, 0xFF, 0xE2, 0x01, 0x31, 0xFC, 0x17, 0x07, 0x61,
+ 0xF2, 0x0A, 0x26, 0x6A, 0x39, 0x41, 0xF2, 0x15, 0x06, 0x2C, 0xFD,
+ 0x31, 0x01, 0x9B, 0xFF, 0x14, 0x00, 0xFF, 0xFF, 0x0E, 0x00, 0xC4,
+ 0xFF, 0x6E, 0x00, 0x87, 0xFF, 0x13, 0x00, 0x58, 0x01, 0x0D, 0xFA,
+ 0x61, 0x46, 0x2D, 0x10, 0xF0, 0xF7, 0xF1, 0x04, 0x05, 0xFD, 0x9C,
+ 0x01, 0x4E, 0xFF, 0x2E, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00,
+ 0x36, 0xFF, 0xE3, 0x01, 0x4C, 0xFC, 0xA6, 0x06, 0xDB, 0xF3, 0x5B,
+ 0x1E, 0xFC, 0x3E, 0xFA, 0xF3, 0xD7, 0x04, 0xFD, 0xFD, 0xB4, 0x00,
+ 0xD9, 0xFF, 0xFD, 0xFF, 0x03, 0x00, 0x08, 0x00, 0xE4, 0xFF, 0x1B,
+ 0x00, 0x30, 0x00, 0xE4, 0xFE, 0x5F, 0x03, 0x5E, 0xF6, 0x02, 0x43,
+ 0x96, 0x17, 0x9B, 0xF5, 0xF8, 0x05, 0x8F, 0xFC, 0xCC, 0x01, 0x3D,
+ 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x33, 0x00, 0x3E, 0xFF,
+ 0xC8, 0x01, 0x9B, 0xFC, 0xDD, 0x05, 0xDC, 0xF5, 0xB6, 0x16, 0x77,
+ 0x43, 0xBD, 0xF6, 0x28, 0x03, 0x05, 0xFF, 0x1D, 0x00, 0x25, 0x00,
+ 0xE1, 0xFF, 0x08, 0x00, 0x02, 0x00, 0x00, 0x00, 0xD2, 0xFF, 0xC4,
+ 0x00, 0xE2, 0xFD, 0x01, 0x05, 0xBA, 0xF3, 0x64, 0x3E, 0x3F, 0x1F,
+ 0xA8, 0xF3, 0xB8, 0x06, 0x46, 0xFC, 0xE5, 0x01, 0x36, 0xFF, 0x36,
+ 0x00, 0xFD, 0xFF, 0xFF, 0xFF, 0x2D, 0x00, 0x51, 0xFF, 0x95, 0x01,
+ 0x15, 0xFD, 0xCF, 0x04, 0x39, 0xF8, 0x59, 0x0F, 0xAF, 0x46, 0x8B,
+ 0xFA, 0x17, 0x01, 0x38, 0x00, 0x73, 0xFF, 0x78, 0x00, 0xC0, 0xFF,
+ 0x0F, 0x00, 0xFF, 0xFF, 0x16, 0x00, 0x94, 0xFF, 0x3D, 0x01, 0x18,
+ 0xFD, 0x32, 0x06, 0x1F, 0xF2, 0xB5, 0x38, 0xEB, 0x26, 0x40, 0xF2,
+ 0x1E, 0x07, 0x32, 0xFC, 0xDF, 0x01, 0x3D, 0xFF, 0x35, 0x00, 0xFD,
+ 0xFF, 0x00, 0x00, 0x25, 0x00, 0x6C, 0xFF, 0x4F, 0x01, 0xB0, 0xFD,
+ 0x93, 0x03, 0xC7, 0xFA, 0x7D, 0x08, 0x86, 0x48, 0x5A, 0xFF, 0xBA,
+ 0xFE, 0x86, 0x01, 0xBF, 0xFE, 0xCF, 0x00, 0x9E, 0xFF, 0x17, 0x00,
+ 0xFD, 0xFF, 0x27, 0x00, 0x66, 0xFF, 0x97, 0x01, 0x8C, 0xFC, 0xEA,
+ 0x06, 0x80, 0xF1, 0x26, 0x32, 0x58, 0x2E, 0x8B, 0xF1, 0x1B, 0x07,
+ 0x5B, 0xFC, 0xBA, 0x01, 0x53, 0xFF, 0x2D, 0x00, 0xFD, 0xFF, 0x1C,
+ 0x00, 0x8C, 0xFF, 0xFE, 0x00, 0x5D, 0xFE, 0x3F, 0x02, 0x5E, 0xFD,
+ 0x54, 0x02, 0xEC, 0x48, 0x13, 0x05, 0x2E, 0xFC, 0xDE, 0x02, 0x0C,
+ 0xFE, 0x24, 0x01, 0x7D, 0xFF, 0x20, 0x00, 0x00, 0x00, 0xFD, 0xFF,
+ 0x31, 0x00, 0x47, 0xFF, 0xCF, 0x01, 0x40, 0xFC, 0x2A, 0x07, 0xC6,
+ 0xF1, 0xF7, 0x2A, 0x46, 0x35, 0xAB, 0xF1, 0xA4, 0x06, 0xC4, 0xFC,
+ 0x72, 0x01, 0x79, 0xFF, 0x20, 0x00, 0xFE, 0xFF, 0x14, 0x00, 0xAE,
+ 0xFF, 0xA7, 0x00, 0x12, 0xFF, 0xEA, 0x00, 0xD9, 0xFF, 0x03, 0xFD,
+ 0xDC, 0x47, 0x95, 0x0B, 0x96, 0xF9, 0x29, 0x04, 0x65, 0xFD, 0x71,
+ 0x01, 0x5F, 0xFF, 0x29, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00,
+ 0x38, 0xFF, 0xE6, 0x01, 0x34, 0xFC, 0xFB, 0x06, 0xD2, 0xF2, 0x64,
+ 0x23, 0x73, 0x3B, 0xBC, 0xF2, 0xB4, 0x05, 0x6E, 0xFD, 0x09, 0x01,
+ 0xAF, 0xFF, 0x0D, 0x00, 0x00, 0x00, 0x0C, 0x00, 0xD0, 0xFF, 0x51,
+ 0x00, 0xC3, 0xFF, 0xA6, 0xFF, 0x16, 0x02, 0xA9, 0xF8, 0x5C, 0x45,
+ 0xB2, 0x12, 0x19, 0xF7, 0x52, 0x05, 0xD8, 0xFC, 0xAF, 0x01, 0x47,
+ 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x38, 0xFF,
+ 0xDD, 0x01, 0x62, 0xFC, 0x69, 0x06, 0x7F, 0xF4, 0xB2, 0x1B, 0xAB,
+ 0x40, 0xD0, 0xF4, 0x4E, 0x04, 0x53, 0xFE, 0x83, 0x00, 0xF2, 0xFF,
+ 0xF4, 0xFF, 0x05, 0x00, 0x06, 0x00, 0xEE, 0xFF, 0x01, 0x00, 0x66,
+ 0x00, 0x85, 0xFE, 0xFC, 0x03, 0x55, 0xF5, 0x8C, 0x41, 0x38, 0x1A,
+ 0xE1, 0xF4, 0x43, 0x06, 0x70, 0xFC, 0xD8, 0x01, 0x39, 0xFF, 0x35,
+ 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x32, 0x00, 0x44, 0xFF, 0xB9, 0x01,
+ 0xC1, 0xFC, 0x86, 0x05, 0xA5, 0xF6, 0x1E, 0x14, 0xBA, 0x44, 0xF0,
+ 0xF7, 0x7B, 0x02, 0x6B, 0xFF, 0xE4, 0xFF, 0x41, 0x00, 0xD6, 0xFF,
+ 0x0B, 0x00, 0x01, 0x00, 0x08, 0x00, 0xBB, 0xFF, 0xF1, 0x00, 0x96,
+ 0xFD, 0x78, 0x05, 0x0E, 0xF3, 0x8A, 0x3C, 0xEA, 0x21, 0x19, 0xF3,
+ 0xE6, 0x06, 0x38, 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0x00, 0x00, 0x2B, 0x00, 0x5A, 0xFF, 0x7E, 0x01, 0x48, 0xFD,
+ 0x66, 0x04, 0x18, 0xF9, 0xE8, 0x0C, 0x7C, 0x47, 0x19, 0xFC, 0x4D,
+ 0x00, 0xA9, 0x00, 0x35, 0xFF, 0x96, 0x00, 0xB5, 0xFF, 0x12, 0x00,
+ 0xFE, 0xFF, 0x1D, 0x00, 0x82, 0xFF, 0x60, 0x01, 0xE0, 0xFC, 0x7F,
+ 0x06, 0xCC, 0xF1, 0x85, 0x36, 0x87, 0x29, 0xEB, 0xF1, 0x2A, 0x07,
+ 0x39, 0xFC, 0xD6, 0x01, 0x43, 0xFF, 0x33, 0x00, 0xFD, 0xFF, 0x00,
+ 0x00, 0x22, 0x00, 0x77, 0xFF, 0x34, 0x01, 0xEA, 0xFD, 0x1F, 0x03,
+ 0xAE, 0xFB, 0x45, 0x06, 0xD5, 0x48, 0x3C, 0x01, 0xDC, 0xFD, 0xFD,
+ 0x01, 0x80, 0xFE, 0xED, 0x00, 0x93, 0xFF, 0x1B, 0x00, 0xFD, 0xFF,
+ 0x2B, 0x00, 0x59, 0xFF, 0xAE, 0x01, 0x6A, 0xFC, 0x0D, 0x07, 0x80,
+ 0xF1, 0xB8, 0x2F, 0xCF, 0x30, 0x7D, 0xF1, 0xFF, 0x06, 0x78, 0xFC,
+ 0xA5, 0x01, 0x5F, 0xFF, 0x29, 0x00, 0xFD, 0xFF, 0x19, 0x00, 0x98,
+ 0xFF, 0xE0, 0x00, 0x9C, 0xFE, 0xC8, 0x01, 0x3F, 0xFE, 0x62, 0x00,
+ 0xB8, 0x48, 0x3F, 0x07, 0x47, 0xFB, 0x53, 0x03, 0xD0, 0xFD, 0x40,
+ 0x01, 0x72, 0xFF, 0x23, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x34, 0x00,
+ 0x40, 0xFF, 0xDB, 0x01, 0x35, 0xFC, 0x26, 0x07, 0x0E, 0xF2, 0x60,
+ 0x28, 0x82, 0x37, 0xED, 0xF1, 0x5E, 0x06, 0xF8, 0xFC, 0x51, 0x01,
+ 0x8A, 0xFF, 0x1A, 0x00, 0xFF, 0xFF, 0x11, 0x00, 0xBA, 0xFF, 0x89,
+ 0x00, 0x51, 0xFF, 0x77, 0x00, 0xA7, 0x00, 0x64, 0xFB, 0x26, 0x47,
+ 0xFC, 0x0D, 0xB4, 0xF8, 0x95, 0x04, 0x31, 0xFD, 0x88, 0x01, 0x56,
+ 0xFF, 0x2C, 0x00, 0xFF, 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF,
+ 0xE6, 0x01, 0x3E, 0xFC, 0xD3, 0x06, 0x56, 0xF3, 0xBA, 0x20, 0x61,
+ 0x3D, 0x56, 0xF3, 0x45, 0x05, 0xB7, 0xFD, 0xDE, 0x00, 0xC5, 0xFF,
+ 0x05, 0x00, 0x02, 0x00, 0x09, 0x00, 0xDB, 0xFF, 0x34, 0x00, 0xFE,
+ 0xFF, 0x3D, 0xFF, 0xC9, 0x02, 0x64, 0xF7, 0x2F, 0x44, 0x44, 0x15,
+ 0x4A, 0xF6, 0xAD, 0x05, 0xAF, 0xFC, 0xC0, 0x01, 0x41, 0xFF, 0x32,
+ 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xD3, 0x01,
+ 0x7D, 0xFC, 0x23, 0x06, 0x32, 0xF5, 0x0C, 0x19, 0x38, 0x42, 0xC7,
+ 0xF5, 0xB8, 0x03, 0xAF, 0xFE, 0x4E, 0x00, 0x0C, 0x00, 0xEA, 0xFF,
+ 0x06, 0x00, 0x04, 0x00, 0xF8, 0xFF, 0xE7, 0xFF, 0x99, 0x00, 0x2C,
+ 0xFE, 0x8C, 0x04, 0x6D, 0xF4, 0xF0, 0x3F, 0xE0, 0x1C, 0x34, 0xF4,
+ 0x85, 0x06, 0x57, 0xFC, 0xE0, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE,
+ 0xFF, 0xFF, 0xFF, 0x2F, 0x00, 0x4A, 0xFF, 0xA7, 0x01, 0xEC, 0xFC,
+ 0x28, 0x05, 0x77, 0xF7, 0x92, 0x11, 0xD7, 0x45, 0x43, 0xF9, 0xC3,
+ 0x01, 0xD6, 0xFF, 0xA9, 0xFF, 0x5E, 0x00, 0xCB, 0xFF, 0x0D, 0x00,
+ 0x00, 0x00, 0x10, 0x00, 0xA6, 0xFF, 0x1B, 0x01, 0x50, 0xFD, 0xE1,
+ 0x05, 0x82, 0xF2, 0x8F, 0x3A, 0x92, 0x24, 0x9D, 0xF2, 0x09, 0x07,
+ 0x32, 0xFC, 0xE4, 0x01, 0x39, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x00,
+ 0x00, 0x28, 0x00, 0x63, 0xFF, 0x66, 0x01, 0x7D, 0xFD, 0xF8, 0x03,
+ 0xFB, 0xF9, 0x89, 0x0A, 0x1D, 0x48, 0xC5, 0xFD, 0x7A, 0xFF, 0x1D,
+ 0x01, 0xF7, 0xFE, 0xB4, 0x00, 0xA9, 0xFF, 0x15, 0x00, 0xFE, 0xFF,
+ 0x23, 0x00, 0x72, 0xFF, 0x7F, 0x01, 0xB0, 0xFC, 0xBE, 0x06, 0x97,
+ 0xF1, 0x3F, 0x34, 0x19, 0x2C, 0xAD, 0xF1, 0x28, 0x07, 0x48, 0xFC,
+ 0xC9, 0x01, 0x4B, 0xFF, 0x30, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x1F,
+ 0x00, 0x82, 0xFF, 0x18, 0x01, 0x27, 0xFE, 0xA9, 0x02, 0x94, 0xFC,
+ 0x24, 0x04, 0xF5, 0x48, 0x39, 0x03, 0xF9, 0xFC, 0x74, 0x02, 0x42,
+ 0xFE, 0x0B, 0x01, 0x87, 0xFF, 0x1E, 0x00, 0xFD, 0xFF, 0x2F, 0x00,
+ 0x4F, 0xFF, 0xC2, 0x01, 0x51, 0xFC, 0x23, 0x07, 0x9A, 0xF1, 0x3A,
+ 0x2D, 0x35, 0x33, 0x89, 0xF1, 0xD5, 0x06, 0x9D, 0xFC, 0x8B, 0x01,
+ 0x6C, 0xFF, 0x25, 0x00, 0xFD, 0xFF, 0x16, 0x00, 0xA4, 0xFF, 0xC2,
+ 0x00, 0xDB, 0xFE, 0x52, 0x01, 0x1B, 0xFF, 0x8D, 0xFE, 0x57, 0x48,
+ 0x81, 0x09, 0x61, 0xFA, 0xC6, 0x03, 0x96, 0xFD, 0x5B, 0x01, 0x68,
+ 0xFF, 0x26, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3B, 0xFF,
+ 0xE2, 0x01, 0x31, 0xFC, 0x15, 0x07, 0x6D, 0xF2, 0xBF, 0x25, 0xA5,
+ 0x39, 0x4D, 0xF2, 0x0B, 0x06, 0x33, 0xFD, 0x2D, 0x01, 0x9D, 0xFF,
+ 0x13, 0x00, 0xFF, 0xFF, 0x0E, 0x00, 0xC6, 0xFF, 0x6B, 0x00, 0x8E,
+ 0xFF, 0x06, 0x00, 0x6E, 0x01, 0xE4, 0xF9, 0x48, 0x46, 0x75, 0x10,
+ 0xD7, 0xF7, 0xFC, 0x04, 0x00, 0xFD, 0x9E, 0x01, 0x4E, 0xFF, 0x2E,
+ 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE3, 0x01,
+ 0x4E, 0xFC, 0xA0, 0x06, 0xED, 0xF3, 0x0F, 0x1E, 0x2D, 0x3F, 0x10,
+ 0xF4, 0xC8, 0x04, 0x07, 0xFE, 0xAF, 0x00, 0xDC, 0xFF, 0xFC, 0xFF,
+ 0x03, 0x00, 0x07, 0x00, 0xE5, 0xFF, 0x18, 0x00, 0x36, 0x00, 0xD9,
+ 0xFE, 0x71, 0x03, 0x3F, 0xF6, 0xDB, 0x42, 0xE0, 0x17, 0x86, 0xF5,
+ 0x00, 0x06, 0x8C, 0xFC, 0xCE, 0x01, 0x3C, 0xFF, 0x34, 0x00, 0xFE,
+ 0xFF, 0xFF, 0xFF, 0x33, 0x00, 0x3F, 0xFF, 0xC6, 0x01, 0x9F, 0xFC,
+ 0xD3, 0x05, 0xF1, 0xF5, 0x6C, 0x16, 0x9E, 0x43, 0xDD, 0xF6, 0x15,
+ 0x03, 0x10, 0xFF, 0x17, 0x00, 0x28, 0x00, 0xDF, 0xFF, 0x09, 0x00,
+ 0x02, 0x00, 0x01, 0x00, 0xCF, 0xFF, 0xC9, 0x00, 0xDA, 0xFD, 0x0F,
+ 0x05, 0xA5, 0xF3, 0x31, 0x3E, 0x8A, 0x1F, 0x97, 0xF3, 0xBD, 0x06,
+ 0x44, 0xFC, 0xE5, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFF,
+ 0xFF, 0x2D, 0x00, 0x52, 0xFF, 0x92, 0x01, 0x1B, 0xFD, 0xC4, 0x04,
+ 0x51, 0xF8, 0x13, 0x0F, 0xC8, 0x46, 0xB6, 0xFA, 0x01, 0x01, 0x44,
+ 0x00, 0x6C, 0xFF, 0x7B, 0x00, 0xBF, 0xFF, 0x10, 0x00, 0xFF, 0xFF,
+ 0x17, 0x00, 0x92, 0xFF, 0x41, 0x01, 0x11, 0xFD, 0x3B, 0x06, 0x14,
+ 0xF2, 0x78, 0x38, 0x36, 0x27, 0x35, 0xF2, 0x20, 0x07, 0x33, 0xFC,
+ 0xDF, 0x01, 0x3E, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x25,
+ 0x00, 0x6D, 0xFF, 0x4C, 0x01, 0xB6, 0xFD, 0x86, 0x03, 0xE1, 0xFA,
+ 0x3D, 0x08, 0x92, 0x48, 0x8E, 0xFF, 0xA1, 0xFE, 0x93, 0x01, 0xB8,
+ 0xFE, 0xD3, 0x00, 0x9D, 0xFF, 0x18, 0x00, 0xFD, 0xFF, 0x28, 0x00,
+ 0x64, 0xFF, 0x9A, 0x01, 0x88, 0xFC, 0xEE, 0x06, 0x7E, 0xF1, 0xE3,
+ 0x31, 0x9F, 0x2E, 0x88, 0xF1, 0x19, 0x07, 0x5E, 0xFC, 0xB7, 0x01,
+ 0x54, 0xFF, 0x2D, 0x00, 0xFD, 0xFF, 0x1C, 0x00, 0x8D, 0xFF, 0xFA,
+ 0x00, 0x64, 0xFE, 0x32, 0x02, 0x78, 0xFD, 0x1B, 0x02, 0xEA, 0x48,
+ 0x50, 0x05, 0x14, 0xFC, 0xEB, 0x02, 0x05, 0xFE, 0x27, 0x01, 0x7C,
+ 0xFF, 0x21, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x46, 0xFF,
+ 0xD1, 0x01, 0x3F, 0xFC, 0x2B, 0x07, 0xCD, 0xF1, 0xAE, 0x2A, 0x86,
+ 0x35, 0xB1, 0xF1, 0x9D, 0x06, 0xCA, 0xFC, 0x6E, 0x01, 0x7B, 0xFF,
+ 0x20, 0x00, 0xFE, 0xFF, 0x13, 0x00, 0xAF, 0xFF, 0xA4, 0x00, 0x19,
+ 0xFF, 0xDD, 0x00, 0xF0, 0xFF, 0xD4, 0xFC, 0xC9, 0x47, 0xD8, 0x0B,
+ 0x7C, 0xF9, 0x35, 0x04, 0x5F, 0xFD, 0x74, 0x01, 0x5E, 0xFF, 0x29,
+ 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xE6, 0x01,
+ 0x35, 0xFC, 0xF7, 0x06, 0xE0, 0xF2, 0x18, 0x23, 0xAB, 0x3B, 0xCC,
+ 0xF2, 0xA8, 0x05, 0x76, 0xFD, 0x04, 0x01, 0xB1, 0xFF, 0x0C, 0x00,
+ 0x00, 0x00, 0x0C, 0x00, 0xD1, 0xFF, 0x4E, 0x00, 0xCA, 0xFF, 0x9A,
+ 0xFF, 0x2A, 0x02, 0x83, 0xF8, 0x3F, 0x45, 0xFB, 0x12, 0x01, 0xF7,
+ 0x5D, 0x05, 0xD3, 0xFC, 0xB1, 0x01, 0x46, 0xFF, 0x31, 0x00, 0xFF,
+ 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xDC, 0x01, 0x64, 0xFC,
+ 0x62, 0x06, 0x93, 0xF4, 0x66, 0x1B, 0xD9, 0x40, 0xEA, 0xF4, 0x3E,
+ 0x04, 0x5D, 0xFE, 0x7D, 0x00, 0xF5, 0xFF, 0xF3, 0xFF, 0x05, 0x00,
+ 0x05, 0x00, 0xEF, 0xFF, 0xFE, 0xFF, 0x6C, 0x00, 0x7B, 0xFE, 0x0C,
+ 0x04, 0x3A, 0xF5, 0x5F, 0x41, 0x83, 0x1A, 0xCD, 0xF4, 0x4B, 0x06,
+ 0x6D, 0xFC, 0xD9, 0x01, 0x39, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0xFF,
+ 0xFF, 0x31, 0x00, 0x44, 0xFF, 0xB7, 0x01, 0xC5, 0xFC, 0x7C, 0x05,
+ 0xBC, 0xF6, 0xD5, 0x13, 0xDC, 0x44, 0x14, 0xF8, 0x67, 0x02, 0x77,
+ 0xFF, 0xDD, 0xFF, 0x44, 0x00, 0xD5, 0xFF, 0x0B, 0x00, 0x01, 0x00,
+ 0x09, 0x00, 0xB8, 0xFF, 0xF6, 0x00, 0x8D, 0xFD, 0x84, 0x05, 0xFD,
+ 0xF2, 0x52, 0x3C, 0x35, 0x22, 0x0B, 0xF3, 0xEB, 0x06, 0x37, 0xFC,
+ 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x2A,
+ 0x00, 0x5B, 0xFF, 0x7C, 0x01, 0x4E, 0xFD, 0x5A, 0x04, 0x31, 0xF9,
+ 0xA4, 0x0C, 0x90, 0x47, 0x47, 0xFC, 0x36, 0x00, 0xB6, 0x00, 0x2E,
+ 0xFF, 0x99, 0x00, 0xB3, 0xFF, 0x12, 0x00, 0xFE, 0xFF, 0x1E, 0x00,
+ 0x80, 0xFF, 0x64, 0x01, 0xDA, 0xFC, 0x87, 0x06, 0xC5, 0xF1, 0x46,
+ 0x36, 0xD1, 0x29, 0xE3, 0xF1, 0x2A, 0x07, 0x3A, 0xFC, 0xD5, 0x01,
+ 0x44, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x22, 0x00, 0x78,
+ 0xFF, 0x31, 0x01, 0xF1, 0xFD, 0x12, 0x03, 0xC7, 0xFB, 0x07, 0x06,
+ 0xDB, 0x48, 0x73, 0x01, 0xC3, 0xFD, 0x0A, 0x02, 0x79, 0xFE, 0xF1,
+ 0x00, 0x91, 0xFF, 0x1B, 0x00, 0xFD, 0xFF, 0x2C, 0x00, 0x58, 0xFF,
+ 0xB1, 0x01, 0x67, 0xFC, 0x10, 0x07, 0x81, 0xF1, 0x73, 0x2F, 0x15,
+ 0x31, 0x7C, 0xF1, 0xFB, 0x06, 0x7C, 0xFC, 0xA2, 0x01, 0x60, 0xFF,
+ 0x29, 0x00, 0xFD, 0xFF, 0x19, 0x00, 0x99, 0xFF, 0xDD, 0x00, 0xA3,
+ 0xFE, 0xBB, 0x01, 0x58, 0xFE, 0x2D, 0x00, 0xAF, 0x48, 0x7E, 0x07,
+ 0x2E, 0xFB, 0x60, 0x03, 0xC9, 0xFD, 0x43, 0x01, 0x71, 0xFF, 0x24,
+ 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x34, 0x00, 0x3F, 0xFF, 0xDC, 0x01,
+ 0x34, 0xFC, 0x25, 0x07, 0x18, 0xF2, 0x15, 0x28, 0xBF, 0x37, 0xF7,
+ 0xF1, 0x56, 0x06, 0xFE, 0xFC, 0x4D, 0x01, 0x8C, 0xFF, 0x19, 0x00,
+ 0xFF, 0xFF, 0x10, 0x00, 0xBB, 0xFF, 0x85, 0x00, 0x58, 0xFF, 0x6A,
+ 0x00, 0xBE, 0x00, 0x38, 0xFB, 0x0F, 0x47, 0x42, 0x0E, 0x9B, 0xF8,
+ 0xA1, 0x04, 0x2B, 0xFD, 0x8B, 0x01, 0x55, 0xFF, 0x2C, 0x00, 0xFF,
+ 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3F, 0xFC,
+ 0xCE, 0x06, 0x66, 0xF3, 0x6F, 0x20, 0x96, 0x3D, 0x69, 0xF3, 0x38,
+ 0x05, 0xBF, 0xFD, 0xD9, 0x00, 0xC7, 0xFF, 0x04, 0x00, 0x02, 0x00,
+ 0x09, 0x00, 0xDC, 0xFF, 0x31, 0x00, 0x04, 0x00, 0x32, 0xFF, 0xDC,
+ 0x02, 0x42, 0xF7, 0x0B, 0x44, 0x8E, 0x15, 0x34, 0xF6, 0xB7, 0x05,
+ 0xAB, 0xFC, 0xC1, 0x01, 0x40, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0xFE,
+ 0xFF, 0x35, 0x00, 0x3B, 0xFF, 0xD2, 0x01, 0x81, 0xFC, 0x1A, 0x06,
+ 0x47, 0xF5, 0xC1, 0x18, 0x60, 0x42, 0xE4, 0xF5, 0xA6, 0x03, 0xB9,
+ 0xFE, 0x48, 0x00, 0x0F, 0x00, 0xE9, 0xFF, 0x07, 0x00, 0x04, 0x00,
+ 0xF9, 0xFF, 0xE4, 0xFF, 0x9F, 0x00, 0x23, 0xFE, 0x9B, 0x04, 0x55,
+ 0xF4, 0xC0, 0x3F, 0x2C, 0x1D, 0x22, 0xF4, 0x8C, 0x06, 0x55, 0xFC,
+ 0xE1, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x2F,
+ 0x00, 0x4B, 0xFF, 0xA4, 0x01, 0xF1, 0xFC, 0x1D, 0x05, 0x8F, 0xF7,
+ 0x4A, 0x11, 0xF2, 0x45, 0x6B, 0xF9, 0xAE, 0x01, 0xE2, 0xFF, 0xA2,
+ 0xFF, 0x61, 0x00, 0xC9, 0xFF, 0x0D, 0x00, 0x00, 0x00, 0x11, 0x00,
+ 0xA3, 0xFF, 0x20, 0x01, 0x49, 0xFD, 0xEB, 0x05, 0x74, 0xF2, 0x54,
+ 0x3A, 0xDD, 0x24, 0x91, 0xF2, 0x0C, 0x07, 0x32, 0xFC, 0xE4, 0x01,
+ 0x3A, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x27, 0x00, 0x64,
+ 0xFF, 0x63, 0x01, 0x84, 0xFD, 0xEB, 0x03, 0x14, 0xFA, 0x47, 0x0A,
+ 0x2C, 0x48, 0xF6, 0xFD, 0x63, 0xFF, 0x2B, 0x01, 0xF0, 0xFE, 0xB8,
+ 0x00, 0xA8, 0xFF, 0x15, 0x00, 0xFE, 0xFF, 0x23, 0x00, 0x71, 0xFF,
+ 0x82, 0x01, 0xAB, 0xFC, 0xC4, 0x06, 0x93, 0xF1, 0xFD, 0x33, 0x62,
+ 0x2C, 0xA8, 0xF1, 0x27, 0x07, 0x4A, 0xFC, 0xC7, 0x01, 0x4C, 0xFF,
+ 0x30, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x1F, 0x00, 0x83, 0xFF, 0x14,
+ 0x01, 0x2D, 0xFE, 0x9C, 0x02, 0xAD, 0xFC, 0xE9, 0x03, 0xF6, 0x48,
+ 0x73, 0x03, 0xE0, 0xFC, 0x82, 0x02, 0x3B, 0xFE, 0x0E, 0x01, 0x86,
+ 0xFF, 0x1E, 0x00, 0xFD, 0xFF, 0x2F, 0x00, 0x4E, 0xFF, 0xC3, 0x01,
+ 0x4E, 0xFC, 0x24, 0x07, 0x9E, 0xF1, 0xF2, 0x2C, 0x78, 0x33, 0x8C,
+ 0xF1, 0xD0, 0x06, 0xA2, 0xFC, 0x88, 0x01, 0x6D, 0xFF, 0x24, 0x00,
+ 0xFD, 0xFF, 0x16, 0x00, 0xA5, 0xFF, 0xBE, 0x00, 0xE2, 0xFE, 0x45,
+ 0x01, 0x33, 0xFF, 0x5A, 0xFE, 0x48, 0x48, 0xC3, 0x09, 0x47, 0xFA,
+ 0xD2, 0x03, 0x90, 0xFD, 0x5E, 0x01, 0x66, 0xFF, 0x27, 0x00, 0x00,
+ 0x00, 0xFD, 0xFF, 0x35, 0x00, 0x3B, 0xFF, 0xE3, 0x01, 0x31, 0xFC,
+ 0x12, 0x07, 0x79, 0xF2, 0x73, 0x25, 0xDF, 0x39, 0x5A, 0xF2, 0x00,
+ 0x06, 0x3A, 0xFD, 0x28, 0x01, 0x9F, 0xFF, 0x13, 0x00, 0x00, 0x00,
+ 0x0E, 0x00, 0xC7, 0xFF, 0x68, 0x00, 0x95, 0xFF, 0xFA, 0xFF, 0x83,
+ 0x01, 0xBB, 0xF9, 0x2B, 0x46, 0xBB, 0x10, 0xBF, 0xF7, 0x07, 0x05,
+ 0xFB, 0xFC, 0xA0, 0x01, 0x4D, 0xFF, 0x2F, 0x00, 0xFF, 0xFF, 0xFE,
+ 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE2, 0x01, 0x50, 0xFC, 0x99, 0x06,
+ 0xFE, 0xF3, 0xC3, 0x1D, 0x5E, 0x3F, 0x27, 0xF4, 0xB9, 0x04, 0x10,
+ 0xFE, 0xA9, 0x00, 0xDF, 0xFF, 0xFB, 0xFF, 0x03, 0x00, 0x07, 0x00,
+ 0xE6, 0xFF, 0x15, 0x00, 0x3C, 0x00, 0xCF, 0xFE, 0x83, 0x03, 0x20,
+ 0xF6, 0xB2, 0x42, 0x2B, 0x18, 0x71, 0xF5, 0x09, 0x06, 0x88, 0xFC,
+ 0xCF, 0x01, 0x3C, 0xFF, 0x34, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x33,
+ 0x00, 0x3F, 0xFF, 0xC5, 0x01, 0xA3, 0xFC, 0xCA, 0x05, 0x07, 0xF6,
+ 0x22, 0x16, 0xC3, 0x43, 0xFE, 0xF6, 0x02, 0x03, 0x1B, 0xFF, 0x11,
+ 0x00, 0x2B, 0x00, 0xDE, 0xFF, 0x09, 0x00, 0x02, 0x00, 0x02, 0x00,
+ 0xCC, 0xFF, 0xCE, 0x00, 0xD1, 0xFD, 0x1D, 0x05, 0x91, 0xF3, 0xFE,
+ 0x3D, 0xD7, 0x1F, 0x87, 0xF3, 0xC3, 0x06, 0x42, 0xFC, 0xE5, 0x01,
+ 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFF, 0xFF, 0x2D, 0x00, 0x53,
+ 0xFF, 0x90, 0x01, 0x20, 0xFD, 0xB8, 0x04, 0x6A, 0xF8, 0xCD, 0x0E,
+ 0xE1, 0x46, 0xE1, 0xFA, 0xEB, 0x00, 0x51, 0x00, 0x65, 0xFF, 0x7F,
+ 0x00, 0xBE, 0xFF, 0x10, 0x00, 0xFF, 0xFF, 0x18, 0x00, 0x90, 0xFF,
+ 0x45, 0x01, 0x0B, 0xFD, 0x44, 0x06, 0x0A, 0xF2, 0x3B, 0x38, 0x80,
+ 0x27, 0x2B, 0xF2, 0x22, 0x07, 0x33, 0xFC, 0xDE, 0x01, 0x3E, 0xFF,
+ 0x34, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x24, 0x00, 0x6E, 0xFF, 0x49,
+ 0x01, 0xBC, 0xFD, 0x7A, 0x03, 0xFA, 0xFA, 0xFD, 0x07, 0x9C, 0x48,
+ 0xC3, 0xFF, 0x89, 0xFE, 0xA1, 0x01, 0xB1, 0xFE, 0xD6, 0x00, 0x9C,
+ 0xFF, 0x18, 0x00, 0xFD, 0xFF, 0x28, 0x00, 0x63, 0xFF, 0x9D, 0x01,
+ 0x84, 0xFC, 0xF3, 0x06, 0x7D, 0xF1, 0x9E, 0x31, 0xE6, 0x2E, 0x85,
+ 0xF1, 0x16, 0x07, 0x61, 0xFC, 0xB5, 0x01, 0x55, 0xFF, 0x2D, 0x00,
+ 0xFD, 0xFF, 0x1C, 0x00, 0x8F, 0xFF, 0xF7, 0x00, 0x6B, 0xFE, 0x25,
+ 0x02, 0x91, 0xFD, 0xE3, 0x01, 0xE5, 0x48, 0x8D, 0x05, 0xFB, 0xFB,
+ 0xF8, 0x02, 0xFE, 0xFD, 0x2B, 0x01, 0x7A, 0xFF, 0x21, 0x00, 0x00,
+ 0x00, 0xFD, 0xFF, 0x32, 0x00, 0x45, 0xFF, 0xD2, 0x01, 0x3D, 0xFC,
+ 0x2B, 0x07, 0xD4, 0xF1, 0x64, 0x2A, 0xC6, 0x35, 0xB7, 0xF1, 0x96,
+ 0x06, 0xCF, 0xFC, 0x6B, 0x01, 0x7D, 0xFF, 0x1F, 0x00, 0xFE, 0xFF,
+ 0x13, 0x00, 0xB1, 0xFF, 0xA0, 0x00, 0x20, 0xFF, 0xD0, 0x00, 0x07,
+ 0x00, 0xA4, 0xFC, 0xB6, 0x47, 0x1C, 0x0C, 0x63, 0xF9, 0x42, 0x04,
+ 0x59, 0xFD, 0x76, 0x01, 0x5D, 0xFF, 0x2A, 0x00, 0x00, 0x00, 0xFD,
+ 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x35, 0xFC, 0xF3, 0x06,
+ 0xEE, 0xF2, 0xCD, 0x22, 0xE4, 0x3B, 0xDC, 0xF2, 0x9C, 0x05, 0x7E,
+ 0xFD, 0x00, 0x01, 0xB4, 0xFF, 0x0B, 0x00, 0x01, 0x00, 0x0B, 0x00,
+ 0xD2, 0xFF, 0x4A, 0x00, 0xD0, 0xFF, 0x8E, 0xFF, 0x3F, 0x02, 0x5E,
+ 0xF8, 0x1E, 0x45, 0x44, 0x13, 0xEA, 0xF6, 0x67, 0x05, 0xCF, 0xFC,
+ 0xB3, 0x01, 0x46, 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x36,
+ 0x00, 0x38, 0xFF, 0xDB, 0x01, 0x67, 0xFC, 0x5A, 0x06, 0xA6, 0xF4,
+ 0x1B, 0x1B, 0x07, 0x41, 0x04, 0xF5, 0x2D, 0x04, 0x67, 0xFE, 0x77,
+ 0x00, 0xF8, 0xFF, 0xF2, 0xFF, 0x05, 0x00, 0x05, 0x00, 0xF0, 0xFF,
+ 0xFB, 0xFF, 0x71, 0x00, 0x71, 0xFE, 0x1D, 0x04, 0x1F, 0xF5, 0x32,
+ 0x41, 0xCE, 0x1A, 0xBA, 0xF4, 0x53, 0x06, 0x6A, 0xFC, 0xDA, 0x01,
+ 0x38, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x31, 0x00, 0x45,
+ 0xFF, 0xB5, 0x01, 0xCA, 0xFC, 0x72, 0x05, 0xD3, 0xF6, 0x8D, 0x13,
+ 0xFD, 0x44, 0x39, 0xF8, 0x53, 0x02, 0x82, 0xFF, 0xD7, 0xFF, 0x47,
+ 0x00, 0xD3, 0xFF, 0x0B, 0x00, 0x01, 0x00, 0x0A, 0x00, 0xB6, 0xFF,
+ 0xFB, 0x00, 0x85, 0xFD, 0x90, 0x05, 0xEC, 0xF2, 0x1C, 0x3C, 0x81,
+ 0x22, 0xFC, 0xF2, 0xEF, 0x06, 0x36, 0xFC, 0xE6, 0x01, 0x37, 0xFF,
+ 0x36, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x2A, 0x00, 0x5C, 0xFF, 0x79,
+ 0x01, 0x53, 0xFD, 0x4E, 0x04, 0x4A, 0xF9, 0x60, 0x0C, 0xA3, 0x47,
+ 0x76, 0xFC, 0x1F, 0x00, 0xC3, 0x00, 0x27, 0xFF, 0x9D, 0x00, 0xB2,
+ 0xFF, 0x13, 0x00, 0xFE, 0xFF, 0x1E, 0x00, 0x7F, 0xFF, 0x67, 0x01,
+ 0xD5, 0xFC, 0x8E, 0x06, 0xBE, 0xF1, 0x06, 0x36, 0x1A, 0x2A, 0xDC,
+ 0xF1, 0x2A, 0x07, 0x3C, 0xFC, 0xD3, 0x01, 0x44, 0xFF, 0x32, 0x00,
+ 0xFD, 0xFF, 0x00, 0x00, 0x21, 0x00, 0x79, 0xFF, 0x2E, 0x01, 0xF7,
+ 0xFD, 0x05, 0x03, 0xE1, 0xFB, 0xCA, 0x05, 0xDF, 0x48, 0xAB, 0x01,
+ 0xAA, 0xFD, 0x18, 0x02, 0x72, 0xFE, 0xF4, 0x00, 0x90, 0xFF, 0x1B,
+ 0x00, 0xFD, 0xFF, 0x2C, 0x00, 0x57, 0xFF, 0xB3, 0x01, 0x64, 0xFC,
+ 0x13, 0x07, 0x83, 0xF1, 0x2C, 0x2F, 0x5A, 0x31, 0x7D, 0xF1, 0xF7,
+ 0x06, 0x80, 0xFC, 0x9F, 0x01, 0x61, 0xFF, 0x29, 0x00, 0xFD, 0xFF,
+ 0x19, 0x00, 0x9A, 0xFF, 0xD9, 0x00, 0xAA, 0xFE, 0xAE, 0x01, 0x70,
+ 0xFE, 0xF8, 0xFF, 0xA6, 0x48, 0xBE, 0x07, 0x14, 0xFB, 0x6D, 0x03,
+ 0xC3, 0xFD, 0x46, 0x01, 0x70, 0xFF, 0x24, 0x00, 0x00, 0x00, 0xFD,
+ 0xFF, 0x34, 0x00, 0x3F, 0xFF, 0xDD, 0x01, 0x34, 0xFC, 0x23, 0x07,
+ 0x21, 0xF2, 0xCB, 0x27, 0xFE, 0x37, 0x00, 0xF2, 0x4D, 0x06, 0x04,
+ 0xFD, 0x49, 0x01, 0x8E, 0xFF, 0x19, 0x00, 0xFF, 0xFF, 0x10, 0x00,
+ 0xBD, 0xFF, 0x82, 0x00, 0x5E, 0xFF, 0x5D, 0x00, 0xD4, 0x00, 0x0C,
+ 0xFB, 0xF9, 0x46, 0x87, 0x0E, 0x82, 0xF8, 0xAD, 0x04, 0x26, 0xFD,
+ 0x8D, 0x01, 0x54, 0xFF, 0x2C, 0x00, 0xFF, 0xFF, 0xFD, 0xFF, 0x36,
+ 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x41, 0xFC, 0xC8, 0x06, 0x76, 0xF3,
+ 0x22, 0x20, 0xCA, 0x3D, 0x7D, 0xF3, 0x2A, 0x05, 0xC8, 0xFD, 0xD4,
+ 0x00, 0xCA, 0xFF, 0x03, 0x00, 0x02, 0x00, 0x09, 0x00, 0xDD, 0xFF,
+ 0x2E, 0x00, 0x0A, 0x00, 0x27, 0xFF, 0xEF, 0x02, 0x20, 0xF7, 0xE7,
+ 0x43, 0xD8, 0x15, 0x1E, 0xF6, 0xC0, 0x05, 0xA7, 0xFC, 0xC3, 0x01,
+ 0x40, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0xFE, 0xFF, 0x34, 0x00, 0x3B,
+ 0xFF, 0xD1, 0x01, 0x84, 0xFC, 0x12, 0x06, 0x5C, 0xF5, 0x76, 0x18,
+ 0x89, 0x42, 0x02, 0xF6, 0x94, 0x03, 0xC4, 0xFE, 0x42, 0x00, 0x12,
+ 0x00, 0xE8, 0xFF, 0x07, 0x00, 0x03, 0x00, 0xFA, 0xFF, 0xE2, 0xFF,
+ 0xA4, 0x00, 0x19, 0xFE, 0xAA, 0x04, 0x3E, 0xF4, 0x90, 0x3F, 0x78,
+ 0x1D, 0x10, 0xF4, 0x93, 0x06, 0x52, 0xFC, 0xE1, 0x01, 0x36, 0xFF,
+ 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x2F, 0x00, 0x4C, 0xFF, 0xA2,
+ 0x01, 0xF6, 0xFC, 0x12, 0x05, 0xA7, 0xF7, 0x03, 0x11, 0x10, 0x46,
+ 0x93, 0xF9, 0x98, 0x01, 0xEE, 0xFF, 0x9B, 0xFF, 0x64, 0x00, 0xC8,
+ 0xFF, 0x0E, 0x00, 0x00, 0x00, 0x12, 0x00, 0xA1, 0xFF, 0x24, 0x01,
+ 0x41, 0xFD, 0xF6, 0x05, 0x67, 0xF2, 0x1A, 0x3A, 0x29, 0x25, 0x84,
+ 0xF2, 0x0F, 0x07, 0x31, 0xFC, 0xE3, 0x01, 0x3A, 0xFF, 0x35, 0x00,
+ 0xFD, 0xFF, 0x00, 0x00, 0x27, 0x00, 0x65, 0xFF, 0x60, 0x01, 0x8A,
+ 0xFD, 0xDF, 0x03, 0x2E, 0xFA, 0x04, 0x0A, 0x3A, 0x48, 0x28, 0xFE,
+ 0x4B, 0xFF, 0x38, 0x01, 0xE9, 0xFE, 0xBB, 0x00, 0xA6, 0xFF, 0x16,
+ 0x00, 0xFD, 0xFF, 0x24, 0x00, 0x6F, 0xFF, 0x85, 0x01, 0xA6, 0xFC,
+ 0xCA, 0x06, 0x8F, 0xF1, 0xBB, 0x33, 0xAB, 0x2C, 0xA3, 0xF1, 0x26,
+ 0x07, 0x4C, 0xFC, 0xC5, 0x01, 0x4D, 0xFF, 0x30, 0x00, 0xFD, 0xFF,
+ 0x00, 0x00, 0x1E, 0x00, 0x84, 0xFF, 0x11, 0x01, 0x34, 0xFE, 0x8F,
+ 0x02, 0xC7, 0xFC, 0xAE, 0x03, 0xF7, 0x48, 0xAE, 0x03, 0xC7, 0xFC,
+ 0x8F, 0x02, 0x34, 0xFE, 0x11, 0x01, 0x84, 0xFF, 0x1E, 0x00, 0xFD,
+ 0xFF, 0x2A, 0x00, 0x5C, 0xFF, 0xAA, 0x01, 0x71, 0xFC, 0x07, 0x07,
+ 0x7E, 0xF1, 0x44, 0x30, 0x44, 0x30, 0x7E, 0xF1, 0x07, 0x07, 0x71,
+ 0xFC, 0xAA, 0x01, 0x5C, 0xFF, 0x2A, 0x00, 0xFD, 0xFF, 0x00, 0x00,
+ 0x1E, 0x00, 0x84, 0xFF, 0x11, 0x01, 0x34, 0xFE, 0x8F, 0x02, 0xC7,
+ 0xFC, 0xAE, 0x03, 0xF7, 0x48, 0xAE, 0x03, 0xC7, 0xFC, 0x8F, 0x02,
+ 0x34, 0xFE, 0x11, 0x01, 0x84, 0xFF, 0x1E, 0x00, 0x02, 0x00, 0x05,
+ 0x00, 0xC3, 0xFF, 0xE1, 0x00, 0xB1, 0xFD, 0x4E, 0x05, 0x4A, 0xF3,
+ 0x3D, 0x3D, 0xED, 0x20, 0x4C, 0xF3, 0xD6, 0x06, 0x3D, 0xFC, 0xE6,
+ 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x36, 0x00,
+ 0x36, 0xFF, 0xE6, 0x01, 0x3D, 0xFC, 0xD6, 0x06, 0x4C, 0xF3, 0xED,
+ 0x20, 0x3D, 0x3D, 0x4A, 0xF3, 0x4E, 0x05, 0xB1, 0xFD, 0xE1, 0x00,
+ 0xC3, 0xFF, 0x05, 0x00, 0x02, 0x00, 0x00, 0x00, 0x1E, 0x00, 0x84,
+ 0xFF, 0x11, 0x01, 0x34, 0xFE, 0x8F, 0x02, 0xC7, 0xFC, 0xAE, 0x03,
+ 0xF7, 0x48, 0xAE, 0x03, 0xC7, 0xFC, 0x8F, 0x02, 0x34, 0xFE, 0x11,
+ 0x01, 0x84, 0xFF, 0x1E, 0x00, 0x16, 0x00, 0xA6, 0xFF, 0xBB, 0x00,
+ 0xE9, 0xFE, 0x38, 0x01, 0x4B, 0xFF, 0x28, 0xFE, 0x3A, 0x48, 0x04,
+ 0x0A, 0x2E, 0xFA, 0xDF, 0x03, 0x8A, 0xFD, 0x60, 0x01, 0x65, 0xFF,
+ 0x27, 0x00, 0x00, 0x00, 0x0E, 0x00, 0xC8, 0xFF, 0x64, 0x00, 0x9B,
+ 0xFF, 0xEE, 0xFF, 0x98, 0x01, 0x93, 0xF9, 0x10, 0x46, 0x03, 0x11,
+ 0xA7, 0xF7, 0x12, 0x05, 0xF6, 0xFC, 0xA2, 0x01, 0x4C, 0xFF, 0x2F,
+ 0x00, 0xFF, 0xFF, 0x07, 0x00, 0xE8, 0xFF, 0x12, 0x00, 0x42, 0x00,
+ 0xC4, 0xFE, 0x94, 0x03, 0x02, 0xF6, 0x89, 0x42, 0x76, 0x18, 0x5C,
+ 0xF5, 0x12, 0x06, 0x84, 0xFC, 0xD1, 0x01, 0x3B, 0xFF, 0x34, 0x00,
+ 0xFE, 0xFF, 0x02, 0x00, 0x03, 0x00, 0xCA, 0xFF, 0xD4, 0x00, 0xC8,
+ 0xFD, 0x2A, 0x05, 0x7D, 0xF3, 0xCA, 0x3D, 0x22, 0x20, 0x76, 0xF3,
+ 0xC8, 0x06, 0x41, 0xFC, 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0xFF, 0xFF, 0x19, 0x00, 0x8E, 0xFF, 0x49, 0x01, 0x04, 0xFD,
+ 0x4D, 0x06, 0x00, 0xF2, 0xFE, 0x37, 0xCB, 0x27, 0x21, 0xF2, 0x23,
+ 0x07, 0x34, 0xFC, 0xDD, 0x01, 0x3F, 0xFF, 0x34, 0x00, 0xFD, 0xFF,
+ 0xFD, 0xFF, 0x29, 0x00, 0x61, 0xFF, 0x9F, 0x01, 0x80, 0xFC, 0xF7,
+ 0x06, 0x7D, 0xF1, 0x5A, 0x31, 0x2C, 0x2F, 0x83, 0xF1, 0x13, 0x07,
+ 0x64, 0xFC, 0xB3, 0x01, 0x57, 0xFF, 0x2C, 0x00, 0xFD, 0xFF, 0xFD,
+ 0xFF, 0x32, 0x00, 0x44, 0xFF, 0xD3, 0x01, 0x3C, 0xFC, 0x2A, 0x07,
+ 0xDC, 0xF1, 0x1A, 0x2A, 0x06, 0x36, 0xBE, 0xF1, 0x8E, 0x06, 0xD5,
+ 0xFC, 0x67, 0x01, 0x7F, 0xFF, 0x1E, 0x00, 0xFE, 0xFF, 0xFD, 0xFF,
+ 0x36, 0x00, 0x37, 0xFF, 0xE6, 0x01, 0x36, 0xFC, 0xEF, 0x06, 0xFC,
+ 0xF2, 0x81, 0x22, 0x1C, 0x3C, 0xEC, 0xF2, 0x90, 0x05, 0x85, 0xFD,
+ 0xFB, 0x00, 0xB6, 0xFF, 0x0A, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x35,
+ 0x00, 0x38, 0xFF, 0xDA, 0x01, 0x6A, 0xFC, 0x53, 0x06, 0xBA, 0xF4,
+ 0xCE, 0x1A, 0x32, 0x41, 0x1F, 0xF5, 0x1D, 0x04, 0x71, 0xFE, 0x71,
+ 0x00, 0xFB, 0xFF, 0xF0, 0xFF, 0x05, 0x00, 0xFF, 0xFF, 0x31, 0x00,
+ 0x46, 0xFF, 0xB3, 0x01, 0xCF, 0xFC, 0x67, 0x05, 0xEA, 0xF6, 0x44,
+ 0x13, 0x1E, 0x45, 0x5E, 0xF8, 0x3F, 0x02, 0x8E, 0xFF, 0xD0, 0xFF,
+ 0x4A, 0x00, 0xD2, 0xFF, 0x0B, 0x00, 0x00, 0x00, 0x2A, 0x00, 0x5D,
+ 0xFF, 0x76, 0x01, 0x59, 0xFD, 0x42, 0x04, 0x63, 0xF9, 0x1C, 0x0C,
+ 0xB6, 0x47, 0xA4, 0xFC, 0x07, 0x00, 0xD0, 0x00, 0x20, 0xFF, 0xA0,
+ 0x00, 0xB1, 0xFF, 0x13, 0x00, 0x00, 0x00, 0x21, 0x00, 0x7A, 0xFF,
+ 0x2B, 0x01, 0xFE, 0xFD, 0xF8, 0x02, 0xFB, 0xFB, 0x8D, 0x05, 0xE5,
+ 0x48, 0xE3, 0x01, 0x91, 0xFD, 0x25, 0x02, 0x6B, 0xFE, 0xF7, 0x00,
+ 0x8F, 0xFF, 0x1C, 0x00, 0x18, 0x00, 0x9C, 0xFF, 0xD6, 0x00, 0xB1,
+ 0xFE, 0xA1, 0x01, 0x89, 0xFE, 0xC3, 0xFF, 0x9C, 0x48, 0xFD, 0x07,
+ 0xFA, 0xFA, 0x7A, 0x03, 0xBC, 0xFD, 0x49, 0x01, 0x6E, 0xFF, 0x24,
+ 0x00, 0x00, 0x00, 0x10, 0x00, 0xBE, 0xFF, 0x7F, 0x00, 0x65, 0xFF,
+ 0x51, 0x00, 0xEB, 0x00, 0xE1, 0xFA, 0xE1, 0x46, 0xCD, 0x0E, 0x6A,
+ 0xF8, 0xB8, 0x04, 0x20, 0xFD, 0x90, 0x01, 0x53, 0xFF, 0x2D, 0x00,
+ 0xFF, 0xFF, 0x09, 0x00, 0xDE, 0xFF, 0x2B, 0x00, 0x11, 0x00, 0x1B,
+ 0xFF, 0x02, 0x03, 0xFE, 0xF6, 0xC3, 0x43, 0x22, 0x16, 0x07, 0xF6,
+ 0xCA, 0x05, 0xA3, 0xFC, 0xC5, 0x01, 0x3F, 0xFF, 0x33, 0x00, 0xFF,
+ 0xFF, 0x03, 0x00, 0xFB, 0xFF, 0xDF, 0xFF, 0xA9, 0x00, 0x10, 0xFE,
+ 0xB9, 0x04, 0x27, 0xF4, 0x5E, 0x3F, 0xC3, 0x1D, 0xFE, 0xF3, 0x99,
+ 0x06, 0x50, 0xFC, 0xE2, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF,
+ 0x00, 0x00, 0x13, 0x00, 0x9F, 0xFF, 0x28, 0x01, 0x3A, 0xFD, 0x00,
+ 0x06, 0x5A, 0xF2, 0xDF, 0x39, 0x73, 0x25, 0x79, 0xF2, 0x12, 0x07,
+ 0x31, 0xFC, 0xE3, 0x01, 0x3B, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0xFD,
+ 0xFF, 0x24, 0x00, 0x6D, 0xFF, 0x88, 0x01, 0xA2, 0xFC, 0xD0, 0x06,
+ 0x8C, 0xF1, 0x78, 0x33, 0xF2, 0x2C, 0x9E, 0xF1, 0x24, 0x07, 0x4E,
+ 0xFC, 0xC3, 0x01, 0x4E, 0xFF, 0x2F, 0x00, 0xFD, 0xFF, 0xFD, 0xFF,
+ 0x30, 0x00, 0x4C, 0xFF, 0xC7, 0x01, 0x4A, 0xFC, 0x27, 0x07, 0xA8,
+ 0xF1, 0x62, 0x2C, 0xFD, 0x33, 0x93, 0xF1, 0xC4, 0x06, 0xAB, 0xFC,
+ 0x82, 0x01, 0x71, 0xFF, 0x23, 0x00, 0xFE, 0xFF, 0xFD, 0xFF, 0x36,
+ 0x00, 0x3A, 0xFF, 0xE4, 0x01, 0x32, 0xFC, 0x0C, 0x07, 0x91, 0xF2,
+ 0xDD, 0x24, 0x54, 0x3A, 0x74, 0xF2, 0xEB, 0x05, 0x49, 0xFD, 0x20,
+ 0x01, 0xA3, 0xFF, 0x11, 0x00, 0x00, 0x00, 0xFE, 0xFF, 0x36, 0x00,
+ 0x37, 0xFF, 0xE1, 0x01, 0x55, 0xFC, 0x8C, 0x06, 0x22, 0xF4, 0x2C,
+ 0x1D, 0xC0, 0x3F, 0x55, 0xF4, 0x9B, 0x04, 0x23, 0xFE, 0x9F, 0x00,
+ 0xE4, 0xFF, 0xF9, 0xFF, 0x04, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x40,
+ 0xFF, 0xC1, 0x01, 0xAB, 0xFC, 0xB7, 0x05, 0x34, 0xF6, 0x8E, 0x15,
+ 0x0B, 0x44, 0x42, 0xF7, 0xDC, 0x02, 0x32, 0xFF, 0x04, 0x00, 0x31,
+ 0x00, 0xDC, 0xFF, 0x09, 0x00, 0xFF, 0xFF, 0x2C, 0x00, 0x55, 0xFF,
+ 0x8B, 0x01, 0x2B, 0xFD, 0xA1, 0x04, 0x9B, 0xF8, 0x42, 0x0E, 0x0F,
+ 0x47, 0x38, 0xFB, 0xBE, 0x00, 0x6A, 0x00, 0x58, 0xFF, 0x85, 0x00,
+ 0xBB, 0xFF, 0x10, 0x00, 0x00, 0x00, 0x24, 0x00, 0x71, 0xFF, 0x43,
+ 0x01, 0xC9, 0xFD, 0x60, 0x03, 0x2E, 0xFB, 0x7E, 0x07, 0xAF, 0x48,
+ 0x2D, 0x00, 0x58, 0xFE, 0xBB, 0x01, 0xA3, 0xFE, 0xDD, 0x00, 0x99,
+ 0xFF, 0x19, 0x00, 0x1B, 0x00, 0x91, 0xFF, 0xF1, 0x00, 0x79, 0xFE,
+ 0x0A, 0x02, 0xC3, 0xFD, 0x73, 0x01, 0xDB, 0x48, 0x07, 0x06, 0xC7,
+ 0xFB, 0x12, 0x03, 0xF1, 0xFD, 0x31, 0x01, 0x78, 0xFF, 0x22, 0x00,
+ 0x00, 0x00, 0x12, 0x00, 0xB3, 0xFF, 0x99, 0x00, 0x2E, 0xFF, 0xB6,
+ 0x00, 0x36, 0x00, 0x47, 0xFC, 0x90, 0x47, 0xA4, 0x0C, 0x31, 0xF9,
+ 0x5A, 0x04, 0x4E, 0xFD, 0x7C, 0x01, 0x5B, 0xFF, 0x2A, 0x00, 0x00,
+ 0x00, 0x0B, 0x00, 0xD5, 0xFF, 0x44, 0x00, 0xDD, 0xFF, 0x77, 0xFF,
+ 0x67, 0x02, 0x14, 0xF8, 0xDC, 0x44, 0xD5, 0x13, 0xBC, 0xF6, 0x7C,
+ 0x05, 0xC5, 0xFC, 0xB7, 0x01, 0x44, 0xFF, 0x31, 0x00, 0xFF, 0xFF,
+ 0x05, 0x00, 0xF3, 0xFF, 0xF5, 0xFF, 0x7D, 0x00, 0x5D, 0xFE, 0x3E,
+ 0x04, 0xEA, 0xF4, 0xD9, 0x40, 0x66, 0x1B, 0x93, 0xF4, 0x62, 0x06,
+ 0x64, 0xFC, 0xDC, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x00,
+ 0x00, 0x0C, 0x00, 0xB1, 0xFF, 0x04, 0x01, 0x76, 0xFD, 0xA8, 0x05,
+ 0xCC, 0xF2, 0xAB, 0x3B, 0x18, 0x23, 0xE0, 0xF2, 0xF7, 0x06, 0x35,
+ 0xFC, 0xE6, 0x01, 0x38, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFE, 0xFF,
+ 0x20, 0x00, 0x7B, 0xFF, 0x6E, 0x01, 0xCA, 0xFC, 0x9D, 0x06, 0xB1,
+ 0xF1, 0x86, 0x35, 0xAE, 0x2A, 0xCD, 0xF1, 0x2B, 0x07, 0x3F, 0xFC,
+ 0xD1, 0x01, 0x46, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x2D,
+ 0x00, 0x54, 0xFF, 0xB7, 0x01, 0x5E, 0xFC, 0x19, 0x07, 0x88, 0xF1,
+ 0x9F, 0x2E, 0xE3, 0x31, 0x7E, 0xF1, 0xEE, 0x06, 0x88, 0xFC, 0x9A,
+ 0x01, 0x64, 0xFF, 0x28, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x34, 0x00,
+ 0x3E, 0xFF, 0xDF, 0x01, 0x33, 0xFC, 0x20, 0x07, 0x35, 0xF2, 0x36,
+ 0x27, 0x78, 0x38, 0x14, 0xF2, 0x3B, 0x06, 0x11, 0xFD, 0x41, 0x01,
+ 0x92, 0xFF, 0x17, 0x00, 0xFF, 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x36,
+ 0xFF, 0xE5, 0x01, 0x44, 0xFC, 0xBD, 0x06, 0x97, 0xF3, 0x8A, 0x1F,
+ 0x31, 0x3E, 0xA5, 0xF3, 0x0F, 0x05, 0xDA, 0xFD, 0xC9, 0x00, 0xCF,
+ 0xFF, 0x01, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3C, 0xFF,
+ 0xCE, 0x01, 0x8C, 0xFC, 0x00, 0x06, 0x86, 0xF5, 0xE0, 0x17, 0xDB,
+ 0x42, 0x3F, 0xF6, 0x71, 0x03, 0xD9, 0xFE, 0x36, 0x00, 0x18, 0x00,
+ 0xE5, 0xFF, 0x07, 0x00, 0xFF, 0xFF, 0x2E, 0x00, 0x4E, 0xFF, 0x9E,
+ 0x01, 0x00, 0xFD, 0xFC, 0x04, 0xD7, 0xF7, 0x75, 0x10, 0x48, 0x46,
+ 0xE4, 0xF9, 0x6E, 0x01, 0x06, 0x00, 0x8E, 0xFF, 0x6B, 0x00, 0xC6,
+ 0xFF, 0x0E, 0x00, 0x00, 0x00, 0x26, 0x00, 0x68, 0xFF, 0x5B, 0x01,
+ 0x96, 0xFD, 0xC6, 0x03, 0x61, 0xFA, 0x81, 0x09, 0x57, 0x48, 0x8D,
+ 0xFE, 0x1B, 0xFF, 0x52, 0x01, 0xDB, 0xFE, 0xC2, 0x00, 0xA4, 0xFF,
+ 0x16, 0x00, 0x1E, 0x00, 0x87, 0xFF, 0x0B, 0x01, 0x42, 0xFE, 0x74,
+ 0x02, 0xF9, 0xFC, 0x39, 0x03, 0xF5, 0x48, 0x24, 0x04, 0x94, 0xFC,
+ 0xA9, 0x02, 0x27, 0xFE, 0x18, 0x01, 0x82, 0xFF, 0x1F, 0x00, 0x00,
+ 0x00, 0x15, 0x00, 0xA9, 0xFF, 0xB4, 0x00, 0xF7, 0xFE, 0x1D, 0x01,
+ 0x7A, 0xFF, 0xC5, 0xFD, 0x1D, 0x48, 0x89, 0x0A, 0xFB, 0xF9, 0xF8,
+ 0x03, 0x7D, 0xFD, 0x66, 0x01, 0x63, 0xFF, 0x28, 0x00, 0x00, 0x00,
+ 0x0D, 0x00, 0xCB, 0xFF, 0x5E, 0x00, 0xA9, 0xFF, 0xD6, 0xFF, 0xC3,
+ 0x01, 0x43, 0xF9, 0xD7, 0x45, 0x92, 0x11, 0x77, 0xF7, 0x28, 0x05,
+ 0xEC, 0xFC, 0xA7, 0x01, 0x4A, 0xFF, 0x2F, 0x00, 0xFF, 0xFF, 0x06,
+ 0x00, 0xEA, 0xFF, 0x0C, 0x00, 0x4E, 0x00, 0xAF, 0xFE, 0xB8, 0x03,
+ 0xC7, 0xF5, 0x38, 0x42, 0x0C, 0x19, 0x32, 0xF5, 0x23, 0x06, 0x7D,
+ 0xFC, 0xD3, 0x01, 0x3A, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x02, 0x00,
+ 0x05, 0x00, 0xC5, 0xFF, 0xDE, 0x00, 0xB7, 0xFD, 0x45, 0x05, 0x56,
+ 0xF3, 0x61, 0x3D, 0xBA, 0x20, 0x56, 0xF3, 0xD3, 0x06, 0x3E, 0xFC,
+ 0xE6, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFF, 0xFF, 0x1A,
+ 0x00, 0x8A, 0xFF, 0x51, 0x01, 0xF8, 0xFC, 0x5E, 0x06, 0xED, 0xF1,
+ 0x82, 0x37, 0x60, 0x28, 0x0E, 0xF2, 0x26, 0x07, 0x35, 0xFC, 0xDB,
+ 0x01, 0x40, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x29, 0x00,
+ 0x5F, 0xFF, 0xA5, 0x01, 0x78, 0xFC, 0xFF, 0x06, 0x7D, 0xF1, 0xCF,
+ 0x30, 0xB8, 0x2F, 0x80, 0xF1, 0x0D, 0x07, 0x6A, 0xFC, 0xAE, 0x01,
+ 0x59, 0xFF, 0x2B, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x33, 0x00, 0x43,
+ 0xFF, 0xD6, 0x01, 0x39, 0xFC, 0x2A, 0x07, 0xEB, 0xF1, 0x87, 0x29,
+ 0x85, 0x36, 0xCC, 0xF1, 0x7F, 0x06, 0xE0, 0xFC, 0x60, 0x01, 0x82,
+ 0xFF, 0x1D, 0x00, 0xFE, 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x37, 0xFF,
+ 0xE6, 0x01, 0x38, 0xFC, 0xE6, 0x06, 0x19, 0xF3, 0xEA, 0x21, 0x8A,
+ 0x3C, 0x0E, 0xF3, 0x78, 0x05, 0x96, 0xFD, 0xF1, 0x00, 0xBB, 0xFF,
+ 0x08, 0x00, 0x01, 0x00, 0xFE, 0xFF, 0x35, 0x00, 0x39, 0xFF, 0xD8,
+ 0x01, 0x70, 0xFC, 0x43, 0x06, 0xE1, 0xF4, 0x38, 0x1A, 0x8C, 0x41,
+ 0x55, 0xF5, 0xFC, 0x03, 0x85, 0xFE, 0x66, 0x00, 0x01, 0x00, 0xEE,
+ 0xFF, 0x06, 0x00, 0xFF, 0xFF, 0x30, 0x00, 0x47, 0xFF, 0xAF, 0x01,
+ 0xD8, 0xFC, 0x52, 0x05, 0x19, 0xF7, 0xB2, 0x12, 0x5C, 0x45, 0xA9,
+ 0xF8, 0x16, 0x02, 0xA6, 0xFF, 0xC3, 0xFF, 0x51, 0x00, 0xD0, 0xFF,
+ 0x0C, 0x00, 0x00, 0x00, 0x29, 0x00, 0x5F, 0xFF, 0x71, 0x01, 0x65,
+ 0xFD, 0x29, 0x04, 0x96, 0xF9, 0x95, 0x0B, 0xDC, 0x47, 0x03, 0xFD,
+ 0xD9, 0xFF, 0xEA, 0x00, 0x12, 0xFF, 0xA7, 0x00, 0xAE, 0xFF, 0x14,
+ 0x00, 0x00, 0x00, 0x20, 0x00, 0x7D, 0xFF, 0x24, 0x01, 0x0C, 0xFE,
+ 0xDE, 0x02, 0x2E, 0xFC, 0x13, 0x05, 0xEC, 0x48, 0x54, 0x02, 0x5E,
+ 0xFD, 0x3F, 0x02, 0x5D, 0xFE, 0xFE, 0x00, 0x8C, 0xFF, 0x1C, 0x00,
+ 0x17, 0x00, 0x9E, 0xFF, 0xCF, 0x00, 0xBF, 0xFE, 0x86, 0x01, 0xBA,
+ 0xFE, 0x5A, 0xFF, 0x86, 0x48, 0x7D, 0x08, 0xC7, 0xFA, 0x93, 0x03,
+ 0xB0, 0xFD, 0x4F, 0x01, 0x6C, 0xFF, 0x25, 0x00, 0x00, 0x00, 0x0F,
+ 0x00, 0xC0, 0xFF, 0x78, 0x00, 0x73, 0xFF, 0x38, 0x00, 0x17, 0x01,
+ 0x8B, 0xFA, 0xAF, 0x46, 0x59, 0x0F, 0x39, 0xF8, 0xCF, 0x04, 0x15,
+ 0xFD, 0x95, 0x01, 0x51, 0xFF, 0x2D, 0x00, 0xFF, 0xFF, 0x08, 0x00,
+ 0xE1, 0xFF, 0x25, 0x00, 0x1D, 0x00, 0x05, 0xFF, 0x28, 0x03, 0xBD,
+ 0xF6, 0x77, 0x43, 0xB6, 0x16, 0xDC, 0xF5, 0xDD, 0x05, 0x9B, 0xFC,
+ 0xC8, 0x01, 0x3E, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0x03, 0x00, 0xFD,
+ 0xFF, 0xD9, 0xFF, 0xB4, 0x00, 0xFD, 0xFD, 0xD7, 0x04, 0xFA, 0xF3,
+ 0xFC, 0x3E, 0x5B, 0x1E, 0xDB, 0xF3, 0xA6, 0x06, 0x4C, 0xFC, 0xE3,
+ 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x14, 0x00,
+ 0x9B, 0xFF, 0x31, 0x01, 0x2C, 0xFD, 0x15, 0x06, 0x41, 0xF2, 0x6A,
+ 0x39, 0x0A, 0x26, 0x61, 0xF2, 0x17, 0x07, 0x31, 0xFC, 0xE2, 0x01,
+ 0x3B, 0xFF, 0x35, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x25, 0x00, 0x6A,
+ 0xFF, 0x8E, 0x01, 0x99, 0xFC, 0xDB, 0x06, 0x86, 0xF1, 0xF2, 0x32,
+ 0x82, 0x2D, 0x96, 0xF1, 0x21, 0x07, 0x53, 0xFC, 0xC0, 0x01, 0x50,
+ 0xFF, 0x2E, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x30, 0x00, 0x4A, 0xFF,
+ 0xCA, 0x01, 0x46, 0xFC, 0x29, 0x07, 0xB3, 0xF1, 0xD1, 0x2B, 0x81,
+ 0x34, 0x9C, 0xF1, 0xB8, 0x06, 0xB5, 0xFC, 0x7C, 0x01, 0x74, 0xFF,
+ 0x22, 0x00, 0xFE, 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x39, 0xFF, 0xE5,
+ 0x01, 0x32, 0xFC, 0x06, 0x07, 0xAA, 0xF2, 0x46, 0x24, 0xC8, 0x3A,
+ 0x90, 0xF2, 0xD6, 0x05, 0x57, 0xFD, 0x17, 0x01, 0xA8, 0xFF, 0x0F,
+ 0x00, 0x00, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xDF, 0x01,
+ 0x5A, 0xFC, 0x7E, 0x06, 0x47, 0xF4, 0x94, 0x1C, 0x1F, 0x40, 0x85,
+ 0xF4, 0x7D, 0x04, 0x36, 0xFE, 0x93, 0x00, 0xEA, 0xFF, 0xF7, 0xFF,
+ 0x04, 0x00, 0xFF, 0xFF, 0x32, 0x00, 0x42, 0xFF, 0xBE, 0x01, 0xB4,
+ 0xFC, 0xA4, 0x05, 0x61, 0xF6, 0xFB, 0x14, 0x53, 0x44, 0x86, 0xF7,
+ 0xB6, 0x02, 0x49, 0xFF, 0xF7, 0xFF, 0x37, 0x00, 0xD9, 0xFF, 0x0A,
+ 0x00, 0x00, 0x00, 0x2B, 0x00, 0x57, 0xFF, 0x86, 0x01, 0x36, 0xFD,
+ 0x89, 0x04, 0xCD, 0xF8, 0xB7, 0x0D, 0x3D, 0x47, 0x91, 0xFB, 0x91,
+ 0x00, 0x83, 0x00, 0x4A, 0xFF, 0x8C, 0x00, 0xB9, 0xFF, 0x11, 0x00,
+ 0x00, 0x00, 0x23, 0x00, 0x73, 0xFF, 0x3D, 0x01, 0xD6, 0xFD, 0x46,
+ 0x03, 0x61, 0xFB, 0x00, 0x07, 0xBF, 0x48, 0x98, 0x00, 0x26, 0xFE,
+ 0xD5, 0x01, 0x95, 0xFE, 0xE3, 0x00, 0x96, 0xFF, 0x1A, 0x00, 0x1A,
+ 0x00, 0x94, 0xFF, 0xEA, 0x00, 0x87, 0xFE, 0xF0, 0x01, 0xF5, 0xFD,
+ 0x05, 0x01, 0xCE, 0x48, 0x83, 0x06, 0x94, 0xFB, 0x2C, 0x03, 0xE4,
+ 0xFD, 0x37, 0x01, 0x76, 0xFF, 0x22, 0x00, 0x00, 0x00, 0x12, 0x00,
+ 0xB6, 0xFF, 0x93, 0x00, 0x3C, 0xFF, 0x9D, 0x00, 0x63, 0x00, 0xEB,
+ 0xFB, 0x69, 0x47, 0x2D, 0x0D, 0xFF, 0xF8, 0x72, 0x04, 0x42, 0xFD,
+ 0x81, 0x01, 0x59, 0xFF, 0x2B, 0x00, 0x00, 0x00, 0x0A, 0x00, 0xD7,
+ 0xFF, 0x3E, 0x00, 0xEA, 0xFF, 0x60, 0xFF, 0x8F, 0x02, 0xCD, 0xF7,
+ 0x99, 0x44, 0x68, 0x14, 0x8E, 0xF6, 0x90, 0x05, 0xBC, 0xFC, 0xBA,
+ 0x01, 0x43, 0xFF, 0x32, 0x00, 0xFF, 0xFF, 0x04, 0x00, 0xF5, 0xFF,
+ 0xEF, 0xFF, 0x88, 0x00, 0x49, 0xFE, 0x5D, 0x04, 0xB7, 0xF4, 0x7D,
+ 0x40, 0xFD, 0x1B, 0x6C, 0xF4, 0x70, 0x06, 0x5F, 0xFC, 0xDE, 0x01,
+ 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x00, 0x00, 0x0E, 0x00, 0xAC,
+ 0xFF, 0x0E, 0x01, 0x66, 0xFD, 0xBF, 0x05, 0xAD, 0xF2, 0x3B, 0x3B,
+ 0xB0, 0x23, 0xC4, 0xF2, 0xFF, 0x06, 0x33, 0xFC, 0xE5, 0x01, 0x38,
+ 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFE, 0xFF, 0x21, 0x00, 0x77, 0xFF,
+ 0x75, 0x01, 0xBF, 0xFC, 0xAB, 0x06, 0xA6, 0xF1, 0x05, 0x35, 0x40,
+ 0x2B, 0xBF, 0xF1, 0x2A, 0x07, 0x42, 0xFC, 0xCE, 0x01, 0x48, 0xFF,
+ 0x31, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x2E, 0x00, 0x52, 0xFF, 0xBC,
+ 0x01, 0x58, 0xFC, 0x1D, 0x07, 0x8E, 0xF1, 0x11, 0x2E, 0x6B, 0x32,
+ 0x81, 0xF1, 0xE5, 0x06, 0x90, 0xFC, 0x94, 0x01, 0x67, 0xFF, 0x26,
+ 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x35, 0x00, 0x3C, 0xFF, 0xE0, 0x01,
+ 0x32, 0xFC, 0x1C, 0x07, 0x4B, 0xF2, 0xA0, 0x26, 0xF2, 0x38, 0x2A,
+ 0xF2, 0x28, 0x06, 0x1F, 0xFD, 0x39, 0x01, 0x96, 0xFF, 0x16, 0x00,
+ 0xFF, 0xFF, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE4, 0x01, 0x48,
+ 0xFC, 0xB2, 0x06, 0xB9, 0xF3, 0xF3, 0x1E, 0x98, 0x3E, 0xCF, 0xF3,
+ 0xF3, 0x04, 0xEB, 0xFD, 0xBF, 0x00, 0xD4, 0xFF, 0xFF, 0xFF, 0x03,
+ 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3D, 0xFF, 0xCB, 0x01, 0x93, 0xFC,
+ 0xEF, 0x05, 0xB0, 0xF5, 0x4B, 0x17, 0x2A, 0x43, 0x7D, 0xF6, 0x4D,
+ 0x03, 0xEF, 0xFE, 0x2A, 0x00, 0x1E, 0x00, 0xE3, 0xFF, 0x08, 0x00,
+ 0xFF, 0xFF, 0x2E, 0x00, 0x4F, 0xFF, 0x99, 0x01, 0x0B, 0xFD, 0xE6,
+ 0x04, 0x08, 0xF8, 0xE7, 0x0F, 0x7C, 0x46, 0x37, 0xFA, 0x42, 0x01,
+ 0x1F, 0x00, 0x81, 0xFF, 0x71, 0x00, 0xC3, 0xFF, 0x0F, 0x00, 0x00,
+ 0x00, 0x26, 0x00, 0x6A, 0xFF, 0x55, 0x01, 0xA3, 0xFD, 0xAD, 0x03,
+ 0x94, 0xFA, 0xFF, 0x08, 0x70, 0x48, 0xF3, 0xFE, 0xEA, 0xFE, 0x6C,
+ 0x01, 0xCD, 0xFE, 0xC9, 0x00, 0xA1, 0xFF, 0x17, 0x00, 0x1D, 0x00,
+ 0x8A, 0xFF, 0x04, 0x01, 0x50, 0xFE, 0x5A, 0x02, 0x2C, 0xFD, 0xC6,
+ 0x02, 0xF2, 0x48, 0x9B, 0x04, 0x61, 0xFC, 0xC3, 0x02, 0x19, 0xFE,
+ 0x1E, 0x01, 0x7F, 0xFF, 0x20, 0x00, 0x00, 0x00, 0x14, 0x00, 0xAC,
+ 0xFF, 0xAE, 0x00, 0x05, 0xFF, 0x03, 0x01, 0xAA, 0xFF, 0x63, 0xFD,
+ 0xFD, 0x47, 0x0E, 0x0B, 0xC8, 0xF9, 0x11, 0x04, 0x71, 0xFD, 0x6C,
+ 0x01, 0x61, 0xFF, 0x28, 0x00, 0x00, 0x00, 0x0C, 0x00, 0xCD, 0xFF,
+ 0x57, 0x00, 0xB6, 0xFF, 0xBE, 0xFF, 0xED, 0x01, 0xF5, 0xF8, 0x9B,
+ 0x45, 0x22, 0x12, 0x48, 0xF7, 0x3D, 0x05, 0xE2, 0xFC, 0xAB, 0x01,
+ 0x49, 0xFF, 0x30, 0x00, 0xFF, 0xFF, 0x06, 0x00, 0xEC, 0xFF, 0x06,
+ 0x00, 0x5A, 0x00, 0x9A, 0xFE, 0xDA, 0x03, 0x8D, 0xF5, 0xE1, 0x41,
+ 0xA1, 0x19, 0x09, 0xF5, 0x33, 0x06, 0x77, 0xFC, 0xD6, 0x01, 0x3A,
+ 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x07, 0x00, 0xC0, 0xFF,
+ 0xE8, 0x00, 0xA6, 0xFD, 0x5F, 0x05, 0x31, 0xF3, 0xF6, 0x3C, 0x52,
+ 0x21, 0x37, 0xF3, 0xDD, 0x06, 0x3B, 0xFC, 0xE6, 0x01, 0x36, 0xFF,
+ 0x36, 0x00, 0xFD, 0xFF, 0xFE, 0xFF, 0x1C, 0x00, 0x86, 0xFF, 0x59,
+ 0x01, 0xEC, 0xFC, 0x6F, 0x06, 0xDC, 0xF1, 0x04, 0x37, 0xF3, 0x28,
+ 0xFC, 0xF1, 0x28, 0x07, 0x37, 0xFC, 0xD8, 0x01, 0x41, 0xFF, 0x33,
+ 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x2A, 0x00, 0x5C, 0xFF, 0xAA, 0x01,
+ 0x71, 0xFC, 0x07, 0x07, 0x7E, 0xF1, 0x44, 0x30, 0x44, 0x30, 0x7E,
+ 0xF1, 0x07, 0x07, 0x71, 0xFC, 0xAA, 0x01, 0x5C, 0xFF, 0x2A, 0x00,
+ 0xFD, 0xFF, 0xFD, 0xFF, 0x33, 0x00, 0x41, 0xFF, 0xD8, 0x01, 0x37,
+ 0xFC, 0x28, 0x07, 0xFC, 0xF1, 0xF3, 0x28, 0x04, 0x37, 0xDC, 0xF1,
+ 0x6F, 0x06, 0xEC, 0xFC, 0x59, 0x01, 0x86, 0xFF, 0x1C, 0x00, 0xFE,
+ 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3B, 0xFC,
+ 0xDD, 0x06, 0x37, 0xF3, 0x52, 0x21, 0xF6, 0x3C, 0x31, 0xF3, 0x5F,
+ 0x05, 0xA6, 0xFD, 0xE8, 0x00, 0xC0, 0xFF, 0x07, 0x00, 0x01, 0x00,
+ 0xFE, 0xFF, 0x35, 0x00, 0x3A, 0xFF, 0xD6, 0x01, 0x77, 0xFC, 0x33,
+ 0x06, 0x09, 0xF5, 0xA1, 0x19, 0xE1, 0x41, 0x8D, 0xF5, 0xDA, 0x03,
+ 0x9A, 0xFE, 0x5A, 0x00, 0x06, 0x00, 0xEC, 0xFF, 0x06, 0x00, 0xFF,
+ 0xFF, 0x30, 0x00, 0x49, 0xFF, 0xAB, 0x01, 0xE2, 0xFC, 0x3D, 0x05,
+ 0x48, 0xF7, 0x22, 0x12, 0x9B, 0x45, 0xF5, 0xF8, 0xED, 0x01, 0xBE,
+ 0xFF, 0xB6, 0xFF, 0x57, 0x00, 0xCD, 0xFF, 0x0C, 0x00, 0x00, 0x00,
+ 0x28, 0x00, 0x61, 0xFF, 0x6C, 0x01, 0x71, 0xFD, 0x11, 0x04, 0xC8,
+ 0xF9, 0x0E, 0x0B, 0xFD, 0x47, 0x63, 0xFD, 0xAA, 0xFF, 0x03, 0x01,
+ 0x05, 0xFF, 0xAE, 0x00, 0xAC, 0xFF, 0x14, 0x00, 0x00, 0x00, 0x20,
+ 0x00, 0x7F, 0xFF, 0x1E, 0x01, 0x19, 0xFE, 0xC3, 0x02, 0x61, 0xFC,
+ 0x9B, 0x04, 0xF2, 0x48, 0xC6, 0x02, 0x2C, 0xFD, 0x5A, 0x02, 0x50,
+ 0xFE, 0x04, 0x01, 0x8A, 0xFF, 0x1D, 0x00, 0x17, 0x00, 0xA1, 0xFF,
+ 0xC9, 0x00, 0xCD, 0xFE, 0x6C, 0x01, 0xEA, 0xFE, 0xF3, 0xFE, 0x70,
+ 0x48, 0xFF, 0x08, 0x94, 0xFA, 0xAD, 0x03, 0xA3, 0xFD, 0x55, 0x01,
+ 0x6A, 0xFF, 0x26, 0x00, 0x00, 0x00, 0x0F, 0x00, 0xC3, 0xFF, 0x71,
+ 0x00, 0x81, 0xFF, 0x1F, 0x00, 0x42, 0x01, 0x37, 0xFA, 0x7C, 0x46,
+ 0xE7, 0x0F, 0x08, 0xF8, 0xE6, 0x04, 0x0B, 0xFD, 0x99, 0x01, 0x4F,
+ 0xFF, 0x2E, 0x00, 0xFF, 0xFF, 0x08, 0x00, 0xE3, 0xFF, 0x1E, 0x00,
+ 0x2A, 0x00, 0xEF, 0xFE, 0x4D, 0x03, 0x7D, 0xF6, 0x2A, 0x43, 0x4B,
+ 0x17, 0xB0, 0xF5, 0xEF, 0x05, 0x93, 0xFC, 0xCB, 0x01, 0x3D, 0xFF,
+ 0x34, 0x00, 0xFE, 0xFF, 0x03, 0x00, 0xFF, 0xFF, 0xD4, 0xFF, 0xBF,
+ 0x00, 0xEB, 0xFD, 0xF3, 0x04, 0xCF, 0xF3, 0x98, 0x3E, 0xF3, 0x1E,
+ 0xB9, 0xF3, 0xB2, 0x06, 0x48, 0xFC, 0xE4, 0x01, 0x36, 0xFF, 0x36,
+ 0x00, 0xFE, 0xFF, 0xFF, 0xFF, 0x16, 0x00, 0x96, 0xFF, 0x39, 0x01,
+ 0x1F, 0xFD, 0x28, 0x06, 0x2A, 0xF2, 0xF2, 0x38, 0xA0, 0x26, 0x4B,
+ 0xF2, 0x1C, 0x07, 0x32, 0xFC, 0xE0, 0x01, 0x3C, 0xFF, 0x35, 0x00,
+ 0xFD, 0xFF, 0xFD, 0xFF, 0x26, 0x00, 0x67, 0xFF, 0x94, 0x01, 0x90,
+ 0xFC, 0xE5, 0x06, 0x81, 0xF1, 0x6B, 0x32, 0x11, 0x2E, 0x8E, 0xF1,
+ 0x1D, 0x07, 0x58, 0xFC, 0xBC, 0x01, 0x52, 0xFF, 0x2E, 0x00, 0xFD,
+ 0xFF, 0xFD, 0xFF, 0x31, 0x00, 0x48, 0xFF, 0xCE, 0x01, 0x42, 0xFC,
+ 0x2A, 0x07, 0xBF, 0xF1, 0x40, 0x2B, 0x05, 0x35, 0xA6, 0xF1, 0xAB,
+ 0x06, 0xBF, 0xFC, 0x75, 0x01, 0x77, 0xFF, 0x21, 0x00, 0xFE, 0xFF,
+ 0xFD, 0xFF, 0x36, 0x00, 0x38, 0xFF, 0xE5, 0x01, 0x33, 0xFC, 0xFF,
+ 0x06, 0xC4, 0xF2, 0xB0, 0x23, 0x3B, 0x3B, 0xAD, 0xF2, 0xBF, 0x05,
+ 0x66, 0xFD, 0x0E, 0x01, 0xAC, 0xFF, 0x0E, 0x00, 0x00, 0x00, 0xFE,
+ 0xFF, 0x36, 0x00, 0x37, 0xFF, 0xDE, 0x01, 0x5F, 0xFC, 0x70, 0x06,
+ 0x6C, 0xF4, 0xFD, 0x1B, 0x7D, 0x40, 0xB7, 0xF4, 0x5D, 0x04, 0x49,
+ 0xFE, 0x88, 0x00, 0xEF, 0xFF, 0xF5, 0xFF, 0x04, 0x00, 0xFF, 0xFF,
+ 0x32, 0x00, 0x43, 0xFF, 0xBA, 0x01, 0xBC, 0xFC, 0x90, 0x05, 0x8E,
+ 0xF6, 0x68, 0x14, 0x99, 0x44, 0xCD, 0xF7, 0x8F, 0x02, 0x60, 0xFF,
+ 0xEA, 0xFF, 0x3E, 0x00, 0xD7, 0xFF, 0x0A, 0x00, 0x00, 0x00, 0x2B,
+ 0x00, 0x59, 0xFF, 0x81, 0x01, 0x42, 0xFD, 0x72, 0x04, 0xFF, 0xF8,
+ 0x2D, 0x0D, 0x69, 0x47, 0xEB, 0xFB, 0x63, 0x00, 0x9D, 0x00, 0x3C,
+ 0xFF, 0x93, 0x00, 0xB6, 0xFF, 0x12, 0x00, 0x00, 0x00, 0x22, 0x00,
+ 0x76, 0xFF, 0x37, 0x01, 0xE4, 0xFD, 0x2C, 0x03, 0x94, 0xFB, 0x83,
+ 0x06, 0xCE, 0x48, 0x05, 0x01, 0xF5, 0xFD, 0xF0, 0x01, 0x87, 0xFE,
+ 0xEA, 0x00, 0x94, 0xFF, 0x1A, 0x00, 0x1A, 0x00, 0x96, 0xFF, 0xE3,
+ 0x00, 0x95, 0xFE, 0xD5, 0x01, 0x26, 0xFE, 0x98, 0x00, 0xBF, 0x48,
+ 0x00, 0x07, 0x61, 0xFB, 0x46, 0x03, 0xD6, 0xFD, 0x3D, 0x01, 0x73,
+ 0xFF, 0x23, 0x00, 0x00, 0x00, 0x11, 0x00, 0xB9, 0xFF, 0x8C, 0x00,
+ 0x4A, 0xFF, 0x83, 0x00, 0x91, 0x00, 0x91, 0xFB, 0x3D, 0x47, 0xB7,
+ 0x0D, 0xCD, 0xF8, 0x89, 0x04, 0x36, 0xFD, 0x86, 0x01, 0x57, 0xFF,
+ 0x2B, 0x00, 0x00, 0x00, 0x0A, 0x00, 0xD9, 0xFF, 0x37, 0x00, 0xF7,
+ 0xFF, 0x49, 0xFF, 0xB6, 0x02, 0x86, 0xF7, 0x53, 0x44, 0xFB, 0x14,
+ 0x61, 0xF6, 0xA4, 0x05, 0xB4, 0xFC, 0xBE, 0x01, 0x42, 0xFF, 0x32,
+ 0x00, 0xFF, 0xFF, 0x04, 0x00, 0xF7, 0xFF, 0xEA, 0xFF, 0x93, 0x00,
+ 0x36, 0xFE, 0x7D, 0x04, 0x85, 0xF4, 0x1F, 0x40, 0x94, 0x1C, 0x47,
+ 0xF4, 0x7E, 0x06, 0x5A, 0xFC, 0xDF, 0x01, 0x37, 0xFF, 0x36, 0x00,
+ 0xFE, 0xFF, 0x00, 0x00, 0x0F, 0x00, 0xA8, 0xFF, 0x17, 0x01, 0x57,
+ 0xFD, 0xD6, 0x05, 0x90, 0xF2, 0xC8, 0x3A, 0x46, 0x24, 0xAA, 0xF2,
+ 0x06, 0x07, 0x32, 0xFC, 0xE5, 0x01, 0x39, 0xFF, 0x36, 0x00, 0xFD,
+ 0xFF, 0xFE, 0xFF, 0x22, 0x00, 0x74, 0xFF, 0x7C, 0x01, 0xB5, 0xFC,
+ 0xB8, 0x06, 0x9C, 0xF1, 0x81, 0x34, 0xD1, 0x2B, 0xB3, 0xF1, 0x29,
+ 0x07, 0x46, 0xFC, 0xCA, 0x01, 0x4A, 0xFF, 0x30, 0x00, 0xFD, 0xFF,
+ 0xFD, 0xFF, 0x2E, 0x00, 0x50, 0xFF, 0xC0, 0x01, 0x53, 0xFC, 0x21,
+ 0x07, 0x96, 0xF1, 0x82, 0x2D, 0xF2, 0x32, 0x86, 0xF1, 0xDB, 0x06,
+ 0x99, 0xFC, 0x8E, 0x01, 0x6A, 0xFF, 0x25, 0x00, 0xFD, 0xFF, 0xFD,
+ 0xFF, 0x35, 0x00, 0x3B, 0xFF, 0xE2, 0x01, 0x31, 0xFC, 0x17, 0x07,
+ 0x61, 0xF2, 0x0A, 0x26, 0x6A, 0x39, 0x41, 0xF2, 0x15, 0x06, 0x2C,
+ 0xFD, 0x31, 0x01, 0x9B, 0xFF, 0x14, 0x00, 0xFF, 0xFF, 0xFE, 0xFF,
+ 0x36, 0x00, 0x36, 0xFF, 0xE3, 0x01, 0x4C, 0xFC, 0xA6, 0x06, 0xDB,
+ 0xF3, 0x5B, 0x1E, 0xFC, 0x3E, 0xFA, 0xF3, 0xD7, 0x04, 0xFD, 0xFD,
+ 0xB4, 0x00, 0xD9, 0xFF, 0xFD, 0xFF, 0x03, 0x00, 0xFF, 0xFF, 0x33,
+ 0x00, 0x3E, 0xFF, 0xC8, 0x01, 0x9B, 0xFC, 0xDD, 0x05, 0xDC, 0xF5,
+ 0xB6, 0x16, 0x77, 0x43, 0xBD, 0xF6, 0x28, 0x03, 0x05, 0xFF, 0x1D,
+ 0x00, 0x25, 0x00, 0xE1, 0xFF, 0x08, 0x00, 0xFF, 0xFF, 0x2D, 0x00,
+ 0x51, 0xFF, 0x95, 0x01, 0x15, 0xFD, 0xCF, 0x04, 0x39, 0xF8, 0x59,
+ 0x0F, 0xAF, 0x46, 0x8B, 0xFA, 0x17, 0x01, 0x38, 0x00, 0x73, 0xFF,
+ 0x78, 0x00, 0xC0, 0xFF, 0x0F, 0x00, 0x00, 0x00, 0x25, 0x00, 0x6C,
+ 0xFF, 0x4F, 0x01, 0xB0, 0xFD, 0x93, 0x03, 0xC7, 0xFA, 0x7D, 0x08,
+ 0x86, 0x48, 0x5A, 0xFF, 0xBA, 0xFE, 0x86, 0x01, 0xBF, 0xFE, 0xCF,
+ 0x00, 0x9E, 0xFF, 0x17, 0x00, 0x1C, 0x00, 0x8C, 0xFF, 0xFE, 0x00,
+ 0x5D, 0xFE, 0x3F, 0x02, 0x5E, 0xFD, 0x54, 0x02, 0xEC, 0x48, 0x13,
+ 0x05, 0x2E, 0xFC, 0xDE, 0x02, 0x0C, 0xFE, 0x24, 0x01, 0x7D, 0xFF,
+ 0x20, 0x00, 0x00, 0x00, 0x14, 0x00, 0xAE, 0xFF, 0xA7, 0x00, 0x12,
+ 0xFF, 0xEA, 0x00, 0xD9, 0xFF, 0x03, 0xFD, 0xDC, 0x47, 0x95, 0x0B,
+ 0x96, 0xF9, 0x29, 0x04, 0x65, 0xFD, 0x71, 0x01, 0x5F, 0xFF, 0x29,
+ 0x00, 0x00, 0x00, 0x0C, 0x00, 0xD0, 0xFF, 0x51, 0x00, 0xC3, 0xFF,
+ 0xA6, 0xFF, 0x16, 0x02, 0xA9, 0xF8, 0x5C, 0x45, 0xB2, 0x12, 0x19,
+ 0xF7, 0x52, 0x05, 0xD8, 0xFC, 0xAF, 0x01, 0x47, 0xFF, 0x30, 0x00,
+ 0xFF, 0xFF, 0x06, 0x00, 0xEE, 0xFF, 0x01, 0x00, 0x66, 0x00, 0x85,
+ 0xFE, 0xFC, 0x03, 0x55, 0xF5, 0x8C, 0x41, 0x38, 0x1A, 0xE1, 0xF4,
+ 0x43, 0x06, 0x70, 0xFC, 0xD8, 0x01, 0x39, 0xFF, 0x35, 0x00, 0xFE,
+ 0xFF, 0x01, 0x00, 0x08, 0x00, 0xBB, 0xFF, 0xF1, 0x00, 0x96, 0xFD,
+ 0x78, 0x05, 0x0E, 0xF3, 0x8A, 0x3C, 0xEA, 0x21, 0x19, 0xF3, 0xE6,
+ 0x06, 0x38, 0xFC, 0xE6, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF,
+ 0xFE, 0xFF, 0x1D, 0x00, 0x82, 0xFF, 0x60, 0x01, 0xE0, 0xFC, 0x7F,
+ 0x06, 0xCC, 0xF1, 0x85, 0x36, 0x87, 0x29, 0xEB, 0xF1, 0x2A, 0x07,
+ 0x39, 0xFC, 0xD6, 0x01, 0x43, 0xFF, 0x33, 0x00, 0xFD, 0xFF, 0xFD,
+ 0xFF, 0x2B, 0x00, 0x59, 0xFF, 0xAE, 0x01, 0x6A, 0xFC, 0x0D, 0x07,
+ 0x80, 0xF1, 0xB8, 0x2F, 0xCF, 0x30, 0x7D, 0xF1, 0xFF, 0x06, 0x78,
+ 0xFC, 0xA5, 0x01, 0x5F, 0xFF, 0x29, 0x00, 0xFD, 0xFF, 0xFD, 0xFF,
+ 0x34, 0x00, 0x40, 0xFF, 0xDB, 0x01, 0x35, 0xFC, 0x26, 0x07, 0x0E,
+ 0xF2, 0x60, 0x28, 0x82, 0x37, 0xED, 0xF1, 0x5E, 0x06, 0xF8, 0xFC,
+ 0x51, 0x01, 0x8A, 0xFF, 0x1A, 0x00, 0xFF, 0xFF, 0xFD, 0xFF, 0x36,
+ 0x00, 0x36, 0xFF, 0xE6, 0x01, 0x3E, 0xFC, 0xD3, 0x06, 0x56, 0xF3,
+ 0xBA, 0x20, 0x61, 0x3D, 0x56, 0xF3, 0x45, 0x05, 0xB7, 0xFD, 0xDE,
+ 0x00, 0xC5, 0xFF, 0x05, 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x35, 0x00,
+ 0x3A, 0xFF, 0xD3, 0x01, 0x7D, 0xFC, 0x23, 0x06, 0x32, 0xF5, 0x0C,
+ 0x19, 0x38, 0x42, 0xC7, 0xF5, 0xB8, 0x03, 0xAF, 0xFE, 0x4E, 0x00,
+ 0x0C, 0x00, 0xEA, 0xFF, 0x06, 0x00, 0xFF, 0xFF, 0x2F, 0x00, 0x4A,
+ 0xFF, 0xA7, 0x01, 0xEC, 0xFC, 0x28, 0x05, 0x77, 0xF7, 0x92, 0x11,
+ 0xD7, 0x45, 0x43, 0xF9, 0xC3, 0x01, 0xD6, 0xFF, 0xA9, 0xFF, 0x5E,
+ 0x00, 0xCB, 0xFF, 0x0D, 0x00, 0x00, 0x00, 0x28, 0x00, 0x63, 0xFF,
+ 0x66, 0x01, 0x7D, 0xFD, 0xF8, 0x03, 0xFB, 0xF9, 0x89, 0x0A, 0x1D,
+ 0x48, 0xC5, 0xFD, 0x7A, 0xFF, 0x1D, 0x01, 0xF7, 0xFE, 0xB4, 0x00,
+ 0xA9, 0xFF, 0x15, 0x00, 0x00, 0x00, 0x1F, 0x00, 0x82, 0xFF, 0x18,
+ 0x01, 0x27, 0xFE, 0xA9, 0x02, 0x94, 0xFC, 0x24, 0x04, 0xF5, 0x48,
+ 0x39, 0x03, 0xF9, 0xFC, 0x74, 0x02, 0x42, 0xFE, 0x0B, 0x01, 0x87,
+ 0xFF, 0x1E, 0x00, 0x16, 0x00, 0xA4, 0xFF, 0xC2, 0x00, 0xDB, 0xFE,
+ 0x52, 0x01, 0x1B, 0xFF, 0x8D, 0xFE, 0x57, 0x48, 0x81, 0x09, 0x61,
+ 0xFA, 0xC6, 0x03, 0x96, 0xFD, 0x5B, 0x01, 0x68, 0xFF, 0x26, 0x00,
+ 0x00, 0x00, 0x0E, 0x00, 0xC6, 0xFF, 0x6B, 0x00, 0x8E, 0xFF, 0x06,
+ 0x00, 0x6E, 0x01, 0xE4, 0xF9, 0x48, 0x46, 0x75, 0x10, 0xD7, 0xF7,
+ 0xFC, 0x04, 0x00, 0xFD, 0x9E, 0x01, 0x4E, 0xFF, 0x2E, 0x00, 0xFF,
+ 0xFF, 0x07, 0x00, 0xE5, 0xFF, 0x18, 0x00, 0x36, 0x00, 0xD9, 0xFE,
+ 0x71, 0x03, 0x3F, 0xF6, 0xDB, 0x42, 0xE0, 0x17, 0x86, 0xF5, 0x00,
+ 0x06, 0x8C, 0xFC, 0xCE, 0x01, 0x3C, 0xFF, 0x34, 0x00, 0xFE, 0xFF,
+ 0x02, 0x00, 0x01, 0x00, 0xCF, 0xFF, 0xC9, 0x00, 0xDA, 0xFD, 0x0F,
+ 0x05, 0xA5, 0xF3, 0x31, 0x3E, 0x8A, 0x1F, 0x97, 0xF3, 0xBD, 0x06,
+ 0x44, 0xFC, 0xE5, 0x01, 0x36, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFF,
+ 0xFF, 0x17, 0x00, 0x92, 0xFF, 0x41, 0x01, 0x11, 0xFD, 0x3B, 0x06,
+ 0x14, 0xF2, 0x78, 0x38, 0x36, 0x27, 0x35, 0xF2, 0x20, 0x07, 0x33,
+ 0xFC, 0xDF, 0x01, 0x3E, 0xFF, 0x34, 0x00, 0xFD, 0xFF, 0xFD, 0xFF,
+ 0x28, 0x00, 0x64, 0xFF, 0x9A, 0x01, 0x88, 0xFC, 0xEE, 0x06, 0x7E,
+ 0xF1, 0xE3, 0x31, 0x9F, 0x2E, 0x88, 0xF1, 0x19, 0x07, 0x5E, 0xFC,
+ 0xB7, 0x01, 0x54, 0xFF, 0x2D, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x32,
+ 0x00, 0x46, 0xFF, 0xD1, 0x01, 0x3F, 0xFC, 0x2B, 0x07, 0xCD, 0xF1,
+ 0xAE, 0x2A, 0x86, 0x35, 0xB1, 0xF1, 0x9D, 0x06, 0xCA, 0xFC, 0x6E,
+ 0x01, 0x7B, 0xFF, 0x20, 0x00, 0xFE, 0xFF, 0xFD, 0xFF, 0x36, 0x00,
+ 0x38, 0xFF, 0xE6, 0x01, 0x35, 0xFC, 0xF7, 0x06, 0xE0, 0xF2, 0x18,
+ 0x23, 0xAB, 0x3B, 0xCC, 0xF2, 0xA8, 0x05, 0x76, 0xFD, 0x04, 0x01,
+ 0xB1, 0xFF, 0x0C, 0x00, 0x00, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x38,
+ 0xFF, 0xDC, 0x01, 0x64, 0xFC, 0x62, 0x06, 0x93, 0xF4, 0x66, 0x1B,
+ 0xD9, 0x40, 0xEA, 0xF4, 0x3E, 0x04, 0x5D, 0xFE, 0x7D, 0x00, 0xF5,
+ 0xFF, 0xF3, 0xFF, 0x05, 0x00, 0xFF, 0xFF, 0x31, 0x00, 0x44, 0xFF,
+ 0xB7, 0x01, 0xC5, 0xFC, 0x7C, 0x05, 0xBC, 0xF6, 0xD5, 0x13, 0xDC,
+ 0x44, 0x14, 0xF8, 0x67, 0x02, 0x77, 0xFF, 0xDD, 0xFF, 0x44, 0x00,
+ 0xD5, 0xFF, 0x0B, 0x00, 0x00, 0x00, 0x2A, 0x00, 0x5B, 0xFF, 0x7C,
+ 0x01, 0x4E, 0xFD, 0x5A, 0x04, 0x31, 0xF9, 0xA4, 0x0C, 0x90, 0x47,
+ 0x47, 0xFC, 0x36, 0x00, 0xB6, 0x00, 0x2E, 0xFF, 0x99, 0x00, 0xB3,
+ 0xFF, 0x12, 0x00, 0x00, 0x00, 0x22, 0x00, 0x78, 0xFF, 0x31, 0x01,
+ 0xF1, 0xFD, 0x12, 0x03, 0xC7, 0xFB, 0x07, 0x06, 0xDB, 0x48, 0x73,
+ 0x01, 0xC3, 0xFD, 0x0A, 0x02, 0x79, 0xFE, 0xF1, 0x00, 0x91, 0xFF,
+ 0x1B, 0x00, 0x19, 0x00, 0x99, 0xFF, 0xDD, 0x00, 0xA3, 0xFE, 0xBB,
+ 0x01, 0x58, 0xFE, 0x2D, 0x00, 0xAF, 0x48, 0x7E, 0x07, 0x2E, 0xFB,
+ 0x60, 0x03, 0xC9, 0xFD, 0x43, 0x01, 0x71, 0xFF, 0x24, 0x00, 0x00,
+ 0x00, 0x10, 0x00, 0xBB, 0xFF, 0x85, 0x00, 0x58, 0xFF, 0x6A, 0x00,
+ 0xBE, 0x00, 0x38, 0xFB, 0x0F, 0x47, 0x42, 0x0E, 0x9B, 0xF8, 0xA1,
+ 0x04, 0x2B, 0xFD, 0x8B, 0x01, 0x55, 0xFF, 0x2C, 0x00, 0xFF, 0xFF,
+ 0x09, 0x00, 0xDC, 0xFF, 0x31, 0x00, 0x04, 0x00, 0x32, 0xFF, 0xDC,
+ 0x02, 0x42, 0xF7, 0x0B, 0x44, 0x8E, 0x15, 0x34, 0xF6, 0xB7, 0x05,
+ 0xAB, 0xFC, 0xC1, 0x01, 0x40, 0xFF, 0x33, 0x00, 0xFF, 0xFF, 0x04,
+ 0x00, 0xF9, 0xFF, 0xE4, 0xFF, 0x9F, 0x00, 0x23, 0xFE, 0x9B, 0x04,
+ 0x55, 0xF4, 0xC0, 0x3F, 0x2C, 0x1D, 0x22, 0xF4, 0x8C, 0x06, 0x55,
+ 0xFC, 0xE1, 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFE, 0xFF, 0x00, 0x00,
+ 0x11, 0x00, 0xA3, 0xFF, 0x20, 0x01, 0x49, 0xFD, 0xEB, 0x05, 0x74,
+ 0xF2, 0x54, 0x3A, 0xDD, 0x24, 0x91, 0xF2, 0x0C, 0x07, 0x32, 0xFC,
+ 0xE4, 0x01, 0x3A, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFE, 0xFF, 0x23,
+ 0x00, 0x71, 0xFF, 0x82, 0x01, 0xAB, 0xFC, 0xC4, 0x06, 0x93, 0xF1,
+ 0xFD, 0x33, 0x62, 0x2C, 0xA8, 0xF1, 0x27, 0x07, 0x4A, 0xFC, 0xC7,
+ 0x01, 0x4C, 0xFF, 0x30, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x2F, 0x00,
+ 0x4E, 0xFF, 0xC3, 0x01, 0x4E, 0xFC, 0x24, 0x07, 0x9E, 0xF1, 0xF2,
+ 0x2C, 0x78, 0x33, 0x8C, 0xF1, 0xD0, 0x06, 0xA2, 0xFC, 0x88, 0x01,
+ 0x6D, 0xFF, 0x24, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x35, 0x00, 0x3B,
+ 0xFF, 0xE3, 0x01, 0x31, 0xFC, 0x12, 0x07, 0x79, 0xF2, 0x73, 0x25,
+ 0xDF, 0x39, 0x5A, 0xF2, 0x00, 0x06, 0x3A, 0xFD, 0x28, 0x01, 0x9F,
+ 0xFF, 0x13, 0x00, 0x00, 0x00, 0xFE, 0xFF, 0x36, 0x00, 0x36, 0xFF,
+ 0xE2, 0x01, 0x50, 0xFC, 0x99, 0x06, 0xFE, 0xF3, 0xC3, 0x1D, 0x5E,
+ 0x3F, 0x27, 0xF4, 0xB9, 0x04, 0x10, 0xFE, 0xA9, 0x00, 0xDF, 0xFF,
+ 0xFB, 0xFF, 0x03, 0x00, 0xFF, 0xFF, 0x33, 0x00, 0x3F, 0xFF, 0xC5,
+ 0x01, 0xA3, 0xFC, 0xCA, 0x05, 0x07, 0xF6, 0x22, 0x16, 0xC3, 0x43,
+ 0xFE, 0xF6, 0x02, 0x03, 0x1B, 0xFF, 0x11, 0x00, 0x2B, 0x00, 0xDE,
+ 0xFF, 0x09, 0x00, 0xFF, 0xFF, 0x2D, 0x00, 0x53, 0xFF, 0x90, 0x01,
+ 0x20, 0xFD, 0xB8, 0x04, 0x6A, 0xF8, 0xCD, 0x0E, 0xE1, 0x46, 0xE1,
+ 0xFA, 0xEB, 0x00, 0x51, 0x00, 0x65, 0xFF, 0x7F, 0x00, 0xBE, 0xFF,
+ 0x10, 0x00, 0x00, 0x00, 0x24, 0x00, 0x6E, 0xFF, 0x49, 0x01, 0xBC,
+ 0xFD, 0x7A, 0x03, 0xFA, 0xFA, 0xFD, 0x07, 0x9C, 0x48, 0xC3, 0xFF,
+ 0x89, 0xFE, 0xA1, 0x01, 0xB1, 0xFE, 0xD6, 0x00, 0x9C, 0xFF, 0x18,
+ 0x00, 0x1C, 0x00, 0x8F, 0xFF, 0xF7, 0x00, 0x6B, 0xFE, 0x25, 0x02,
+ 0x91, 0xFD, 0xE3, 0x01, 0xE5, 0x48, 0x8D, 0x05, 0xFB, 0xFB, 0xF8,
+ 0x02, 0xFE, 0xFD, 0x2B, 0x01, 0x7A, 0xFF, 0x21, 0x00, 0x00, 0x00,
+ 0x13, 0x00, 0xB1, 0xFF, 0xA0, 0x00, 0x20, 0xFF, 0xD0, 0x00, 0x07,
+ 0x00, 0xA4, 0xFC, 0xB6, 0x47, 0x1C, 0x0C, 0x63, 0xF9, 0x42, 0x04,
+ 0x59, 0xFD, 0x76, 0x01, 0x5D, 0xFF, 0x2A, 0x00, 0x00, 0x00, 0x0B,
+ 0x00, 0xD2, 0xFF, 0x4A, 0x00, 0xD0, 0xFF, 0x8E, 0xFF, 0x3F, 0x02,
+ 0x5E, 0xF8, 0x1E, 0x45, 0x44, 0x13, 0xEA, 0xF6, 0x67, 0x05, 0xCF,
+ 0xFC, 0xB3, 0x01, 0x46, 0xFF, 0x31, 0x00, 0xFF, 0xFF, 0x05, 0x00,
+ 0xF0, 0xFF, 0xFB, 0xFF, 0x71, 0x00, 0x71, 0xFE, 0x1D, 0x04, 0x1F,
+ 0xF5, 0x32, 0x41, 0xCE, 0x1A, 0xBA, 0xF4, 0x53, 0x06, 0x6A, 0xFC,
+ 0xDA, 0x01, 0x38, 0xFF, 0x35, 0x00, 0xFE, 0xFF, 0x01, 0x00, 0x0A,
+ 0x00, 0xB6, 0xFF, 0xFB, 0x00, 0x85, 0xFD, 0x90, 0x05, 0xEC, 0xF2,
+ 0x1C, 0x3C, 0x81, 0x22, 0xFC, 0xF2, 0xEF, 0x06, 0x36, 0xFC, 0xE6,
+ 0x01, 0x37, 0xFF, 0x36, 0x00, 0xFD, 0xFF, 0xFE, 0xFF, 0x1E, 0x00,
+ 0x7F, 0xFF, 0x67, 0x01, 0xD5, 0xFC, 0x8E, 0x06, 0xBE, 0xF1, 0x06,
+ 0x36, 0x1A, 0x2A, 0xDC, 0xF1, 0x2A, 0x07, 0x3C, 0xFC, 0xD3, 0x01,
+ 0x44, 0xFF, 0x32, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x2C, 0x00, 0x57,
+ 0xFF, 0xB3, 0x01, 0x64, 0xFC, 0x13, 0x07, 0x83, 0xF1, 0x2C, 0x2F,
+ 0x5A, 0x31, 0x7D, 0xF1, 0xF7, 0x06, 0x80, 0xFC, 0x9F, 0x01, 0x61,
+ 0xFF, 0x29, 0x00, 0xFD, 0xFF, 0xFD, 0xFF, 0x34, 0x00, 0x3F, 0xFF,
+ 0xDD, 0x01, 0x34, 0xFC, 0x23, 0x07, 0x21, 0xF2, 0xCB, 0x27, 0xFE,
+ 0x37, 0x00, 0xF2, 0x4D, 0x06, 0x04, 0xFD, 0x49, 0x01, 0x8E, 0xFF,
+ 0x19, 0x00, 0xFF, 0xFF, 0xFD, 0xFF, 0x36, 0x00, 0x36, 0xFF, 0xE6,
+ 0x01, 0x41, 0xFC, 0xC8, 0x06, 0x76, 0xF3, 0x22, 0x20, 0xCA, 0x3D,
+ 0x7D, 0xF3, 0x2A, 0x05, 0xC8, 0xFD, 0xD4, 0x00, 0xCA, 0xFF, 0x03,
+ 0x00, 0x02, 0x00, 0xFE, 0xFF, 0x34, 0x00, 0x3B, 0xFF, 0xD1, 0x01,
+ 0x84, 0xFC, 0x12, 0x06, 0x5C, 0xF5, 0x76, 0x18, 0x89, 0x42, 0x02,
+ 0xF6, 0x94, 0x03, 0xC4, 0xFE, 0x42, 0x00, 0x12, 0x00, 0xE8, 0xFF,
+ 0x07, 0x00, 0xFF, 0xFF, 0x2F, 0x00, 0x4C, 0xFF, 0xA2, 0x01, 0xF6,
+ 0xFC, 0x12, 0x05, 0xA7, 0xF7, 0x03, 0x11, 0x10, 0x46, 0x93, 0xF9,
+ 0x98, 0x01, 0xEE, 0xFF, 0x9B, 0xFF, 0x64, 0x00, 0xC8, 0xFF, 0x0E,
+ 0x00, 0x00, 0x00, 0x27, 0x00, 0x65, 0xFF, 0x60, 0x01, 0x8A, 0xFD,
+ 0xDF, 0x03, 0x2E, 0xFA, 0x04, 0x0A, 0x3A, 0x48, 0x28, 0xFE, 0x4B,
+ 0xFF, 0x38, 0x01, 0xE9, 0xFE, 0xBB, 0x00, 0xA6, 0xFF, 0x16, 0x00,
+ 0x00, 0x00, 0x1E, 0x00, 0x84, 0xFF, 0x11, 0x01, 0x34, 0xFE, 0x8F,
+ 0x02, 0xC7, 0xFC, 0xAE, 0x03, 0xF7, 0x48, 0xAE, 0x03, 0xC7, 0xFC,
+ 0x8F, 0x02, 0x34, 0xFE, 0x11, 0x01, 0x84, 0xFF, 0x1E, 0x00, 0x00,
+ 0x00, 0xF4, 0xFF, 0x1A, 0x00, 0xFF, 0x00, 0x07, 0x03, 0x16, 0x06,
+ 0x7C, 0x09, 0x2A, 0x0C, 0x2E, 0x0D, 0x2A, 0x0C, 0x7C, 0x09, 0x16,
+ 0x06, 0x07, 0x03, 0xFF, 0x00, 0x1A, 0x00, 0xF4, 0xFF, 0xF2, 0xFF,
+ 0xA0, 0xFF, 0x71, 0xFF, 0x71, 0x00, 0x86, 0x03, 0x73, 0x08, 0x88,
+ 0x0D, 0x78, 0x10, 0xC9, 0x0F, 0xD5, 0x0B, 0x8B, 0x06, 0x28, 0x02,
+ 0xDF, 0xFF, 0x6F, 0xFF, 0xC3, 0xFF, 0xFD, 0xFF, 0x00, 0x00, 0xDC,
+ 0xFF, 0x80, 0xFF, 0x9A, 0xFF, 0x46, 0x01, 0x1E, 0x05, 0x5A, 0x0A,
+ 0xED, 0x0E, 0xAA, 0x10, 0xAF, 0x0E, 0xFD, 0x09, 0xCB, 0x04, 0x18,
+ 0x01, 0x8E, 0xFF, 0x85, 0xFF, 0xE1, 0xFF, 0xFC, 0xFF, 0xBD, 0xFF,
+ 0x6D, 0xFF, 0xF6, 0xFF, 0x65, 0x02, 0xE5, 0x06, 0x2B, 0x0C, 0xF3,
+ 0x0F, 0x60, 0x10, 0x3B, 0x0D, 0x16, 0x08, 0x3F, 0x03, 0x50, 0x00,
+ 0x6E, 0xFF, 0xA7, 0xFF, 0xF5, 0xFF, 0xEF, 0xFF, 0x9A, 0xFF, 0x75,
+ 0xFF, 0x91, 0x00, 0xC9, 0x03, 0xC8, 0x08, 0xCC, 0x0D, 0x89, 0x10,
+ 0x9F, 0x0F, 0x85, 0x0B, 0x3B, 0x06, 0xF4, 0x01, 0xCD, 0xFF, 0x72,
+ 0xFF, 0xC9, 0xFF, 0xFE, 0xFF, 0x00, 0x00, 0xD7, 0xFF, 0x7B, 0xFF,
+ 0xA5, 0xFF, 0x73, 0x01, 0x6A, 0x05, 0xAD, 0x0A, 0x21, 0x0F, 0xA6,
+ 0x10, 0x74, 0x0E, 0xA9, 0x09, 0x83, 0x04, 0xF0, 0x00, 0x85, 0xFF,
+ 0x8B, 0xFF, 0xE5, 0xFF, 0xFA, 0xFF, 0xB7, 0xFF, 0x6C, 0xFF, 0x0C,
+ 0x00, 0x9D, 0x02, 0x37, 0x07, 0x78, 0x0C, 0x15, 0x10, 0x47, 0x10,
+ 0xF3, 0x0C, 0xC2, 0x07, 0x01, 0x03, 0x35, 0x00, 0x6D, 0xFF, 0xAD,
+ 0xFF, 0xF7, 0xFF, 0xEB, 0xFF, 0x94, 0xFF, 0x7A, 0xFF, 0xB3, 0x00,
+ 0x0D, 0x04, 0x1C, 0x09, 0x0D, 0x0E, 0x97, 0x10, 0x73, 0x0F, 0x35,
+ 0x0B, 0xEB, 0x05, 0xC1, 0x01, 0xBD, 0xFF, 0x75, 0xFF, 0xCE, 0xFF,
+ 0xFF, 0xFF, 0xFF, 0xFF, 0xD2, 0xFF, 0x77, 0xFF, 0xB3, 0xFF, 0xA1,
+ 0x01, 0xB7, 0x05, 0xFF, 0x0A, 0x53, 0x0F, 0x9E, 0x10, 0x37, 0x0E,
+ 0x55, 0x09, 0x3B, 0x04, 0xCB, 0x00, 0x7E, 0xFF, 0x90, 0xFF, 0xE9,
+ 0xFF, 0xF8, 0xFF, 0xB1, 0xFF, 0x6C, 0xFF, 0x24, 0x00, 0xD8, 0x02,
+ 0x8A, 0x07, 0xC2, 0x0C, 0x34, 0x10, 0x2A, 0x10, 0xAA, 0x0C, 0x6F,
+ 0x07, 0xC4, 0x02, 0x1C, 0x00, 0x6C, 0xFF, 0xB3, 0xFF, 0xF9, 0xFF,
+ 0xE8, 0xFF, 0x8E, 0xFF, 0x80, 0xFF, 0xD7, 0x00, 0x53, 0x04, 0x71,
+ 0x09, 0x4C, 0x0E, 0xA1, 0x10, 0x43, 0x0F, 0xE3, 0x0A, 0x9D, 0x05,
+ 0x91, 0x01, 0xAE, 0xFF, 0x79, 0xFF, 0xD4, 0xFF, 0x00, 0x00, 0xFF,
+ 0xFF, 0xCD, 0xFF, 0x74, 0xFF, 0xC2, 0xFF, 0xD2, 0x01, 0x06, 0x06,
+ 0x50, 0x0B, 0x82, 0x0F, 0x93, 0x10, 0xF8, 0x0D, 0x00, 0x09, 0xF6,
+ 0x03, 0xA7, 0x00, 0x78, 0xFF, 0x96, 0xFF, 0xEC, 0xFF, 0xF6, 0xFF,
+ 0xAB, 0xFF, 0x6D, 0xFF, 0x3E, 0x00, 0x15, 0x03, 0xDE, 0x07, 0x0B,
+ 0x0D, 0x50, 0x10, 0x0A, 0x10, 0x5E, 0x0C, 0x1C, 0x07, 0x8A, 0x02,
+ 0x04, 0x00, 0x6C, 0xFF, 0xB9, 0xFF, 0xFB, 0xFF, 0xE4, 0xFF, 0x89,
+ 0xFF, 0x88, 0xFF, 0xFD, 0x00, 0x9B, 0x04, 0xC5, 0x09, 0x88, 0x0E,
+ 0xA8, 0x10, 0x10, 0x0F, 0x91, 0x0A, 0x50, 0x05, 0x64, 0x01, 0xA1,
+ 0xFF, 0x7D, 0xFF, 0xD9, 0xFF, 0x00, 0x00, 0xFE, 0xFF, 0xC7, 0xFF,
+ 0x71, 0xFF, 0xD3, 0xFF, 0x05, 0x02, 0x55, 0x06, 0xA0, 0x0B, 0xAD,
+ 0x0F, 0x84, 0x10, 0xB6, 0x0D, 0xAC, 0x08, 0xB3, 0x03, 0x86, 0x00,
+ 0x74, 0xFF, 0x9C, 0xFF, 0xF0, 0xFF, 0xF4, 0xFF, 0xA5, 0xFF, 0x6F,
+ 0xFF, 0x5A, 0x00, 0x54, 0x03, 0x32, 0x08, 0x52, 0x0D, 0x68, 0x10,
+ 0xE6, 0x0F, 0x11, 0x0C, 0xCA, 0x06, 0x52, 0x02, 0xEF, 0xFF, 0x6E,
+ 0xFF, 0xBF, 0xFF, 0xFC, 0xFF, 0xDF, 0xFF, 0x84, 0xFF, 0x91, 0xFF,
+ 0x25, 0x01, 0xE4, 0x04, 0x19, 0x0A, 0xC2, 0x0E, 0xAA, 0x10, 0xDA,
+ 0x0E, 0x3E, 0x0A, 0x05, 0x05, 0x38, 0x01, 0x96, 0xFF, 0x81, 0xFF,
+ 0xDD, 0xFF, 0x00, 0x00, 0xFD, 0xFF, 0xC1, 0xFF, 0x6E, 0xFF, 0xE6,
+ 0xFF, 0x3A, 0x02, 0xA6, 0x06, 0xEF, 0x0B, 0xD6, 0x0F, 0x71, 0x10,
+ 0x71, 0x0D, 0x57, 0x08, 0x71, 0x03, 0x67, 0x00, 0x70, 0xFF, 0xA2,
+ 0xFF, 0xF3, 0xFF, 0xF1, 0xFF, 0x9F, 0xFF, 0x72, 0xFF, 0x78, 0x00,
+ 0x95, 0x03, 0x86, 0x08, 0x98, 0x0D, 0x7C, 0x10, 0xC0, 0x0F, 0xC3,
+ 0x0B, 0x79, 0x06, 0x1C, 0x02, 0xDB, 0xFF, 0x70, 0xFF, 0xC5, 0xFF,
+ 0xFE, 0xFF, 0x00, 0x00, 0xDB, 0xFF, 0x7F, 0xFF, 0x9C, 0xFF, 0x50,
+ 0x01, 0x2F, 0x05, 0x6C, 0x0A, 0xF9, 0x0E, 0xA9, 0x10, 0xA2, 0x0E,
+ 0xEA, 0x09, 0xBB, 0x04, 0x0F, 0x01, 0x8C, 0xFF, 0x87, 0xFF, 0xE2,
+ 0xFF, 0xFC, 0xFF, 0xBC, 0xFF, 0x6D, 0xFF, 0xFA, 0xFF, 0x71, 0x02,
+ 0xF7, 0x06, 0x3C, 0x0C, 0xFB, 0x0F, 0x5B, 0x10, 0x2B, 0x0D, 0x03,
+ 0x08, 0x31, 0x03, 0x4A, 0x00, 0x6E, 0xFF, 0xA8, 0xFF, 0xF5, 0xFF,
+ 0xEE, 0xFF, 0x99, 0xFF, 0x76, 0xFF, 0x98, 0x00, 0xD8, 0x03, 0xDB,
+ 0x08, 0xDB, 0x0D, 0x8D, 0x10, 0x96, 0x0F, 0x73, 0x0B, 0x29, 0x06,
+ 0xE8, 0x01, 0xC9, 0xFF, 0x72, 0xFF, 0xCA, 0xFF, 0xFE, 0xFF, 0x00,
+ 0x00, 0xD6, 0xFF, 0x7A, 0xFF, 0xA8, 0xFF, 0x7D, 0x01, 0x7B, 0x05,
+ 0xBF, 0x0A, 0x2D, 0x0F, 0xA5, 0x10, 0x67, 0x0E, 0x96, 0x09, 0x73,
+ 0x04, 0xE7, 0x00, 0x84, 0xFF, 0x8C, 0xFF, 0xE6, 0xFF, 0xFA, 0xFF,
+ 0xB6, 0xFF, 0x6C, 0xFF, 0x11, 0x00, 0xAA, 0x02, 0x4A, 0x07, 0x88,
+ 0x0C, 0x1C, 0x10, 0x41, 0x10, 0xE3, 0x0C, 0xAF, 0x07, 0xF3, 0x02,
+ 0x2F, 0x00, 0x6C, 0xFF, 0xAE, 0xFF, 0xF7, 0xFF, 0xEA, 0xFF, 0x93,
+ 0xFF, 0x7B, 0xFF, 0xBB, 0x00, 0x1C, 0x04, 0x2F, 0x09, 0x1B, 0x0E,
+ 0x9A, 0x10, 0x68, 0x0F, 0x23, 0x0B, 0xDA, 0x05, 0xB7, 0x01, 0xB9,
+ 0xFF, 0x76, 0xFF, 0xD0, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xD1, 0xFF,
+ 0x76, 0xFF, 0xB6, 0xFF, 0xAC, 0x01, 0xC8, 0x05, 0x11, 0x0B, 0x5E,
+ 0x0F, 0x9C, 0x10, 0x29, 0x0E, 0x42, 0x09, 0x2C, 0x04, 0xC2, 0x00,
+ 0x7D, 0xFF, 0x92, 0xFF, 0xEA, 0xFF, 0xF8, 0xFF, 0xB0, 0xFF, 0x6C,
+ 0xFF, 0x29, 0x00, 0xE6, 0x02, 0x9D, 0x07, 0xD3, 0x0C, 0x3B, 0x10,
+ 0x23, 0x10, 0x99, 0x0C, 0x5C, 0x07, 0xB7, 0x02, 0x16, 0x00, 0x6C,
+ 0xFF, 0xB4, 0xFF, 0xF9, 0xFF, 0xE7, 0xFF, 0x8D, 0xFF, 0x82, 0xFF,
+ 0xDF, 0x00, 0x63, 0x04, 0x84, 0x09, 0x59, 0x0E, 0xA3, 0x10, 0x38,
+ 0x0F, 0xD1, 0x0A, 0x8C, 0x05, 0x87, 0x01, 0xAB, 0xFF, 0x79, 0xFF,
+ 0xD5, 0xFF, 0x00, 0x00, 0xFF, 0xFF, 0xCB, 0xFF, 0x73, 0xFF, 0xC6,
+ 0xFF, 0xDD, 0x01, 0x17, 0x06, 0x62, 0x0B, 0x8C, 0x0F, 0x90, 0x10,
+ 0xE9, 0x0D, 0xED, 0x08, 0xE7, 0x03, 0xA0, 0x00, 0x77, 0xFF, 0x97,
+ 0xFF, 0xED, 0xFF, 0xF6, 0xFF, 0xA9, 0xFF, 0x6D, 0xFF, 0x44, 0x00,
+ 0x23, 0x03, 0xF1, 0x07, 0x1B, 0x0D, 0x55, 0x10, 0x02, 0x10, 0x4D,
+ 0x0C, 0x0A, 0x07, 0x7E, 0x02, 0xFF, 0xFF, 0x6D, 0xFF, 0xBA, 0xFF,
+ 0xFB, 0xFF, 0xE3, 0xFF, 0x88, 0xFF, 0x8A, 0xFF, 0x06, 0x01, 0xAB,
+ 0x04, 0xD8, 0x09, 0x95, 0x0E, 0xA9, 0x10, 0x05, 0x0F, 0x7F, 0x0A,
+ 0x40, 0x05, 0x5A, 0x01, 0x9F, 0xFF, 0x7E, 0xFF, 0xDA, 0xFF, 0x00,
+ 0x00, 0xFE, 0xFF, 0xC6, 0xFF, 0x70, 0xFF, 0xD7, 0xFF, 0x10, 0x02,
+ 0x67, 0x06, 0xB1, 0x0B, 0xB7, 0x0F, 0x80, 0x10, 0xA7, 0x0D, 0x99,
+ 0x08, 0xA4, 0x03, 0x7F, 0x00, 0x73, 0xFF, 0x9D, 0xFF, 0xF0, 0xFF,
+ 0xF3, 0xFF, 0xA3, 0xFF, 0x70, 0xFF, 0x60, 0x00, 0x62, 0x03, 0x45,
+ 0x08, 0x62, 0x0D, 0x6C, 0x10, 0xDE, 0x0F, 0x00, 0x0C, 0xB8, 0x06,
+ 0x46, 0x02, 0xEA, 0xFF, 0x6E, 0xFF, 0xC0, 0xFF, 0xFD, 0xFF, 0x00,
+ 0x00, 0xDE, 0xFF, 0x83, 0xFF, 0x94, 0xFF, 0x2F, 0x01, 0xF4, 0x04,
+ 0x2B, 0x0A, 0xCE, 0x0E, 0xAA, 0x10, 0xCE, 0x0E, 0x2B, 0x0A, 0xF4,
+ 0x04, 0x2F, 0x01, 0x94, 0xFF, 0x83, 0xFF, 0xDE, 0xFF, 0xFD, 0xFF,
+ 0xC0, 0xFF, 0x6E, 0xFF, 0xEA, 0xFF, 0x46, 0x02, 0xB8, 0x06, 0x00,
+ 0x0C, 0xDE, 0x0F, 0x6C, 0x10, 0x62, 0x0D, 0x45, 0x08, 0x62, 0x03,
+ 0x60, 0x00, 0x70, 0xFF, 0xA3, 0xFF, 0xF3, 0xFF, 0xF0, 0xFF, 0x9D,
+ 0xFF, 0x73, 0xFF, 0x7F, 0x00, 0xA4, 0x03, 0x99, 0x08, 0xA7, 0x0D,
+ 0x80, 0x10, 0xB7, 0x0F, 0xB1, 0x0B, 0x67, 0x06, 0x10, 0x02, 0xD7,
+ 0xFF, 0x70, 0xFF, 0xC6, 0xFF, 0xFE, 0xFF, 0x00, 0x00, 0xDA, 0xFF,
+ 0x7E, 0xFF, 0x9F, 0xFF, 0x5A, 0x01, 0x40, 0x05, 0x7F, 0x0A, 0x05,
+ 0x0F, 0xA9, 0x10, 0x95, 0x0E, 0xD8, 0x09, 0xAB, 0x04, 0x06, 0x01,
+ 0x8A, 0xFF, 0x88, 0xFF, 0xE3, 0xFF, 0xFB, 0xFF, 0xBA, 0xFF, 0x6D,
+ 0xFF, 0xFF, 0xFF, 0x7E, 0x02, 0x0A, 0x07, 0x4D, 0x0C, 0x02, 0x10,
+ 0x55, 0x10, 0x1B, 0x0D, 0xF1, 0x07, 0x23, 0x03, 0x44, 0x00, 0x6D,
+ 0xFF, 0xA9, 0xFF, 0xF6, 0xFF, 0xED, 0xFF, 0x97, 0xFF, 0x77, 0xFF,
+ 0xA0, 0x00, 0xE7, 0x03, 0xED, 0x08, 0xE9, 0x0D, 0x90, 0x10, 0x8C,
+ 0x0F, 0x62, 0x0B, 0x17, 0x06, 0xDD, 0x01, 0xC6, 0xFF, 0x73, 0xFF,
+ 0xCB, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0xD5, 0xFF, 0x79, 0xFF, 0xAB,
+ 0xFF, 0x87, 0x01, 0x8C, 0x05, 0xD1, 0x0A, 0x38, 0x0F, 0xA3, 0x10,
+ 0x59, 0x0E, 0x84, 0x09, 0x63, 0x04, 0xDF, 0x00, 0x82, 0xFF, 0x8D,
+ 0xFF, 0xE7, 0xFF, 0xF9, 0xFF, 0xB4, 0xFF, 0x6C, 0xFF, 0x16, 0x00,
+ 0xB7, 0x02, 0x5C, 0x07, 0x99, 0x0C, 0x23, 0x10, 0x3B, 0x10, 0xD3,
+ 0x0C, 0x9D, 0x07, 0xE6, 0x02, 0x29, 0x00, 0x6C, 0xFF, 0xB0, 0xFF,
+ 0xF8, 0xFF, 0xEA, 0xFF, 0x92, 0xFF, 0x7D, 0xFF, 0xC2, 0x00, 0x2C,
+ 0x04, 0x42, 0x09, 0x29, 0x0E, 0x9C, 0x10, 0x5E, 0x0F, 0x11, 0x0B,
+ 0xC8, 0x05, 0xAC, 0x01, 0xB6, 0xFF, 0x76, 0xFF, 0xD1, 0xFF, 0xFF,
+ 0xFF, 0xFF, 0xFF, 0xD0, 0xFF, 0x76, 0xFF, 0xB9, 0xFF, 0xB7, 0x01,
+ 0xDA, 0x05, 0x23, 0x0B, 0x68, 0x0F, 0x9A, 0x10, 0x1B, 0x0E, 0x2F,
+ 0x09, 0x1C, 0x04, 0xBB, 0x00, 0x7B, 0xFF, 0x93, 0xFF, 0xEA, 0xFF,
+ 0xF7, 0xFF, 0xAE, 0xFF, 0x6C, 0xFF, 0x2F, 0x00, 0xF3, 0x02, 0xAF,
+ 0x07, 0xE3, 0x0C, 0x41, 0x10, 0x1C, 0x10, 0x88, 0x0C, 0x4A, 0x07,
+ 0xAA, 0x02, 0x11, 0x00, 0x6C, 0xFF, 0xB6, 0xFF, 0xFA, 0xFF, 0xE6,
+ 0xFF, 0x8C, 0xFF, 0x84, 0xFF, 0xE7, 0x00, 0x73, 0x04, 0x96, 0x09,
+ 0x67, 0x0E, 0xA5, 0x10, 0x2D, 0x0F, 0xBF, 0x0A, 0x7B, 0x05, 0x7D,
+ 0x01, 0xA8, 0xFF, 0x7A, 0xFF, 0xD6, 0xFF, 0x00, 0x00, 0xFE, 0xFF,
+ 0xCA, 0xFF, 0x72, 0xFF, 0xC9, 0xFF, 0xE8, 0x01, 0x29, 0x06, 0x73,
+ 0x0B, 0x96, 0x0F, 0x8D, 0x10, 0xDB, 0x0D, 0xDB, 0x08, 0xD8, 0x03,
+ 0x98, 0x00, 0x76, 0xFF, 0x99, 0xFF, 0xEE, 0xFF, 0xF5, 0xFF, 0xA8,
+ 0xFF, 0x6E, 0xFF, 0x4A, 0x00, 0x31, 0x03, 0x03, 0x08, 0x2B, 0x0D,
+ 0x5B, 0x10, 0xFB, 0x0F, 0x3C, 0x0C, 0xF7, 0x06, 0x71, 0x02, 0xFA,
+ 0xFF, 0x6D, 0xFF, 0xBC, 0xFF, 0xFC, 0xFF, 0xE2, 0xFF, 0x87, 0xFF,
+ 0x8C, 0xFF, 0x0F, 0x01, 0xBB, 0x04, 0xEA, 0x09, 0xA2, 0x0E, 0xA9,
+ 0x10, 0xF9, 0x0E, 0x6C, 0x0A, 0x2F, 0x05, 0x50, 0x01, 0x9C, 0xFF,
+ 0x7F, 0xFF, 0xDB, 0xFF, 0x00, 0x00, 0xFE, 0xFF, 0xC5, 0xFF, 0x70,
+ 0xFF, 0xDB, 0xFF, 0x1C, 0x02, 0x79, 0x06, 0xC3, 0x0B, 0xC0, 0x0F,
+ 0x7C, 0x10, 0x98, 0x0D, 0x86, 0x08, 0x95, 0x03, 0x78, 0x00, 0x72,
+ 0xFF, 0x9F, 0xFF, 0xF1, 0xFF, 0xF3, 0xFF, 0xA2, 0xFF, 0x70, 0xFF,
+ 0x67, 0x00, 0x71, 0x03, 0x57, 0x08, 0x71, 0x0D, 0x71, 0x10, 0xD6,
+ 0x0F, 0xEF, 0x0B, 0xA6, 0x06, 0x3A, 0x02, 0xE6, 0xFF, 0x6E, 0xFF,
+ 0xC1, 0xFF, 0xFD, 0xFF, 0x00, 0x00, 0xDD, 0xFF, 0x81, 0xFF, 0x96,
+ 0xFF, 0x38, 0x01, 0x05, 0x05, 0x3E, 0x0A, 0xDA, 0x0E, 0xAA, 0x10,
+ 0xC2, 0x0E, 0x19, 0x0A, 0xE4, 0x04, 0x25, 0x01, 0x91, 0xFF, 0x84,
+ 0xFF, 0xDF, 0xFF, 0xFC, 0xFF, 0xBF, 0xFF, 0x6E, 0xFF, 0xEF, 0xFF,
+ 0x52, 0x02, 0xCA, 0x06, 0x11, 0x0C, 0xE6, 0x0F, 0x68, 0x10, 0x52,
+ 0x0D, 0x32, 0x08, 0x54, 0x03, 0x5A, 0x00, 0x6F, 0xFF, 0xA5, 0xFF,
+ 0xF4, 0xFF, 0xF0, 0xFF, 0x9C, 0xFF, 0x74, 0xFF, 0x86, 0x00, 0xB3,
+ 0x03, 0xAC, 0x08, 0xB6, 0x0D, 0x84, 0x10, 0xAD, 0x0F, 0xA0, 0x0B,
+ 0x55, 0x06, 0x05, 0x02, 0xD3, 0xFF, 0x71, 0xFF, 0xC7, 0xFF, 0xFE,
+ 0xFF, 0x00, 0x00, 0xD9, 0xFF, 0x7D, 0xFF, 0xA1, 0xFF, 0x64, 0x01,
+ 0x50, 0x05, 0x91, 0x0A, 0x10, 0x0F, 0xA8, 0x10, 0x88, 0x0E, 0xC5,
+ 0x09, 0x9B, 0x04, 0xFD, 0x00, 0x88, 0xFF, 0x89, 0xFF, 0xE4, 0xFF,
+ 0xFB, 0xFF, 0xB9, 0xFF, 0x6C, 0xFF, 0x04, 0x00, 0x8A, 0x02, 0x1C,
+ 0x07, 0x5E, 0x0C, 0x0A, 0x10, 0x50, 0x10, 0x0B, 0x0D, 0xDE, 0x07,
+ 0x15, 0x03, 0x3E, 0x00, 0x6D, 0xFF, 0xAB, 0xFF, 0xF6, 0xFF, 0xEC,
+ 0xFF, 0x96, 0xFF, 0x78, 0xFF, 0xA7, 0x00, 0xF6, 0x03, 0x00, 0x09,
+ 0xF8, 0x0D, 0x93, 0x10, 0x82, 0x0F, 0x50, 0x0B, 0x06, 0x06, 0xD2,
+ 0x01, 0xC2, 0xFF, 0x74, 0xFF, 0xCD, 0xFF, 0xFF, 0xFF, 0x00, 0x00,
+ 0xD4, 0xFF, 0x79, 0xFF, 0xAE, 0xFF, 0x91, 0x01, 0x9D, 0x05, 0xE3,
+ 0x0A, 0x43, 0x0F, 0xA1, 0x10, 0x4C, 0x0E, 0x71, 0x09, 0x53, 0x04,
+ 0xD7, 0x00, 0x80, 0xFF, 0x8E, 0xFF, 0xE8, 0xFF, 0xF9, 0xFF, 0xB3,
+ 0xFF, 0x6C, 0xFF, 0x1C, 0x00, 0xC4, 0x02, 0x6F, 0x07, 0xAA, 0x0C,
+ 0x2A, 0x10, 0x34, 0x10, 0xC2, 0x0C, 0x8A, 0x07, 0xD8, 0x02, 0x24,
+ 0x00, 0x6C, 0xFF, 0xB1, 0xFF, 0xF8, 0xFF, 0xE9, 0xFF, 0x90, 0xFF,
+ 0x7E, 0xFF, 0xCB, 0x00, 0x3B, 0x04, 0x55, 0x09, 0x37, 0x0E, 0x9E,
+ 0x10, 0x53, 0x0F, 0xFF, 0x0A, 0xB7, 0x05, 0xA1, 0x01, 0xB3, 0xFF,
+ 0x77, 0xFF, 0xD2, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xCE, 0xFF, 0x75,
+ 0xFF, 0xBD, 0xFF, 0xC1, 0x01, 0xEB, 0x05, 0x35, 0x0B, 0x73, 0x0F,
+ 0x97, 0x10, 0x0D, 0x0E, 0x1C, 0x09, 0x0D, 0x04, 0xB3, 0x00, 0x7A,
+ 0xFF, 0x94, 0xFF, 0xEB, 0xFF, 0xF7, 0xFF, 0xAD, 0xFF, 0x6D, 0xFF,
+ 0x35, 0x00, 0x01, 0x03, 0xC2, 0x07, 0xF3, 0x0C, 0x47, 0x10, 0x15,
+ 0x10, 0x78, 0x0C, 0x37, 0x07, 0x9D, 0x02, 0x0C, 0x00, 0x6C, 0xFF,
+ 0xB7, 0xFF, 0xFA, 0xFF, 0xE5, 0xFF, 0x8B, 0xFF, 0x85, 0xFF, 0xF0,
+ 0x00, 0x83, 0x04, 0xA9, 0x09, 0x74, 0x0E, 0xA6, 0x10, 0x21, 0x0F,
+ 0xAD, 0x0A, 0x6A, 0x05, 0x73, 0x01, 0xA5, 0xFF, 0x7B, 0xFF, 0xD7,
+ 0xFF, 0x00, 0x00, 0xFE, 0xFF, 0xC9, 0xFF, 0x72, 0xFF, 0xCD, 0xFF,
+ 0xF4, 0x01, 0x3B, 0x06, 0x85, 0x0B, 0x9F, 0x0F, 0x89, 0x10, 0xCC,
+ 0x0D, 0xC8, 0x08, 0xC9, 0x03, 0x91, 0x00, 0x75, 0xFF, 0x9A, 0xFF,
+ 0xEF, 0xFF, 0xF5, 0xFF, 0xA7, 0xFF, 0x6E, 0xFF, 0x50, 0x00, 0x3F,
+ 0x03, 0x16, 0x08, 0x3B, 0x0D, 0x60, 0x10, 0xF3, 0x0F, 0x2B, 0x0C,
+ 0xE5, 0x06, 0x65, 0x02, 0xF6, 0xFF, 0x6D, 0xFF, 0xBD, 0xFF, 0xFC,
+ 0xFF, 0xE1, 0xFF, 0x85, 0xFF, 0x8E, 0xFF, 0x18, 0x01, 0xCB, 0x04,
+ 0xFD, 0x09, 0xAF, 0x0E, 0xAA, 0x10, 0xED, 0x0E, 0x5A, 0x0A, 0x1E,
+ 0x05, 0x46, 0x01, 0x9A, 0xFF, 0x80, 0xFF, 0xDC, 0xFF, 0x00, 0x00,
+ 0xFD, 0xFF, 0xC3, 0xFF, 0x6F, 0xFF, 0xDF, 0xFF, 0x28, 0x02, 0x8B,
+ 0x06, 0xD5, 0x0B, 0xC9, 0x0F, 0x78, 0x10, 0x88, 0x0D, 0x73, 0x08,
+ 0x86, 0x03, 0x71, 0x00, 0x71, 0xFF, 0xA0, 0xFF, 0xF2, 0xFF, 0xF2,
+ 0xFF, 0xA1, 0xFF, 0x71, 0xFF, 0x6E, 0x00, 0x7F, 0x03, 0x6A, 0x08,
+ 0x81, 0x0D, 0x76, 0x10, 0xCD, 0x0F, 0xDD, 0x0B, 0x94, 0x06, 0x2E,
+ 0x02, 0xE1, 0xFF, 0x6F, 0xFF, 0xC3, 0xFF, 0xFD, 0xFF, 0x00, 0x00,
+ 0xDC, 0xFF, 0x80, 0xFF, 0x98, 0xFF, 0x42, 0x01, 0x16, 0x05, 0x50,
+ 0x0A, 0xE7, 0x0E, 0xAA, 0x10, 0xB5, 0x0E, 0x06, 0x0A, 0xD3, 0x04,
+ 0x1C, 0x01, 0x8F, 0xFF, 0x85, 0xFF, 0xE0, 0xFF, 0xFC, 0xFF, 0xBE,
+ 0xFF, 0x6D, 0xFF, 0xF3, 0xFF, 0x5E, 0x02, 0xDC, 0x06, 0x23, 0x0C,
+ 0xEF, 0x0F, 0x63, 0x10, 0x43, 0x0D, 0x1F, 0x08, 0x46, 0x03, 0x53,
+ 0x00, 0x6E, 0xFF, 0xA6, 0xFF, 0xF4, 0xFF, 0xEF, 0xFF, 0x9B, 0xFF,
+ 0x75, 0xFF, 0x8D, 0x00, 0xC1, 0x03, 0xBE, 0x08, 0xC4, 0x0D, 0x88,
+ 0x10, 0xA4, 0x0F, 0x8E, 0x0B, 0x43, 0x06, 0xF9, 0x01, 0xCF, 0xFF,
+ 0x71, 0xFF, 0xC8, 0xFF, 0xFE, 0xFF, 0x00, 0x00, 0xD8, 0xFF, 0x7C,
+ 0xFF, 0xA4, 0xFF, 0x6E, 0x01, 0x61, 0x05, 0xA3, 0x0A, 0x1C, 0x0F,
+ 0xA7, 0x10, 0x7B, 0x0E, 0xB2, 0x09, 0x8B, 0x04, 0xF4, 0x00, 0x86,
+ 0xFF, 0x8A, 0xFF, 0xE4, 0xFF, 0xFA, 0xFF, 0xB8, 0xFF, 0x6C, 0xFF,
+ 0x09, 0x00, 0x97, 0x02, 0x2E, 0x07, 0x6F, 0x0C, 0x11, 0x10, 0x4A,
+ 0x10, 0xFB, 0x0C, 0xCB, 0x07, 0x07, 0x03, 0x38, 0x00, 0x6D, 0xFF,
+ 0xAC, 0xFF, 0xF7, 0xFF, 0xEC, 0xFF, 0x95, 0xFF, 0x79, 0xFF, 0xAF,
+ 0x00, 0x05, 0x04, 0x13, 0x09, 0x06, 0x0E, 0x96, 0x10, 0x78, 0x0F,
+ 0x3E, 0x0B, 0xF4, 0x05, 0xC7, 0x01, 0xBF, 0xFF, 0x74, 0xFF, 0xCE,
+ 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xD2, 0xFF, 0x78, 0xFF, 0xB1, 0xFF,
+ 0x9C, 0x01, 0xAE, 0x05, 0xF6, 0x0A, 0x4E, 0x0F, 0x9F, 0x10, 0x3E,
+ 0x0E, 0x5E, 0x09, 0x43, 0x04, 0xCF, 0x00, 0x7F, 0xFF, 0x90, 0xFF,
+ 0xE8, 0xFF, 0xF9, 0xFF, 0xB2, 0xFF, 0x6C, 0xFF, 0x21, 0x00, 0xD2,
+ 0x02, 0x81, 0x07, 0xBA, 0x0C, 0x31, 0x10, 0x2E, 0x10, 0xB2, 0x0C,
+ 0x78, 0x07, 0xCB, 0x02, 0x1E, 0x00, 0x6C, 0xFF, 0xB2, 0xFF, 0xF9,
+ 0xFF, 0xE8, 0xFF, 0x8F, 0xFF, 0x80, 0xFF, 0xD3, 0x00, 0x4B, 0x04,
+ 0x67, 0x09, 0x45, 0x0E, 0xA0, 0x10, 0x48, 0x0F, 0xEC, 0x0A, 0xA6,
+ 0x05, 0x97, 0x01, 0xB0, 0xFF, 0x78, 0xFF, 0xD3, 0xFF, 0x00, 0x00,
+ 0xFF, 0xFF, 0xCD, 0xFF, 0x74, 0xFF, 0xC0, 0xFF, 0xCC, 0x01, 0xFD,
+ 0x05, 0x47, 0x0B, 0x7D, 0x0F, 0x94, 0x10, 0xFF, 0x0D, 0x0A, 0x09,
+ 0xFE, 0x03, 0xAB, 0x00, 0x79, 0xFF, 0x95, 0xFF, 0xEC, 0xFF, 0xF7,
+ 0xFF, 0xAC, 0xFF, 0x6D, 0xFF, 0x3B, 0x00, 0x0E, 0x03, 0xD5, 0x07,
+ 0x03, 0x0D, 0x4D, 0x10, 0x0E, 0x10, 0x67, 0x0C, 0x25, 0x07, 0x91,
+ 0x02, 0x07, 0x00, 0x6C, 0xFF, 0xB8, 0xFF, 0xFB, 0xFF, 0xE4, 0xFF,
+ 0x89, 0xFF, 0x87, 0xFF, 0xF9, 0x00, 0x93, 0x04, 0xBC, 0x09, 0x82,
+ 0x0E, 0xA7, 0x10, 0x16, 0x0F, 0x9A, 0x0A, 0x59, 0x05, 0x69, 0x01,
+ 0xA3, 0xFF, 0x7C, 0xFF, 0xD8, 0xFF, 0x00, 0x00, 0xFE, 0xFF, 0xC8,
+ 0xFF, 0x71, 0xFF, 0xD1, 0xFF, 0xFF, 0x01, 0x4C, 0x06, 0x97, 0x0B,
+ 0xA9, 0x0F, 0x86, 0x10, 0xBD, 0x0D, 0xB5, 0x08, 0xBA, 0x03, 0x8A,
+ 0x00, 0x74, 0xFF, 0x9B, 0xFF, 0xEF, 0xFF, 0xF4, 0xFF, 0xA5, 0xFF,
+ 0x6F, 0xFF, 0x57, 0x00, 0x4D, 0x03, 0x29, 0x08, 0x4B, 0x0D, 0x65,
+ 0x10, 0xEB, 0x0F, 0x1A, 0x0C, 0xD3, 0x06, 0x58, 0x02, 0xF1, 0xFF,
+ 0x6D, 0xFF, 0xBE, 0xFF, 0xFC, 0xFF, 0xE0, 0xFF, 0x84, 0xFF, 0x90,
+ 0xFF, 0x21, 0x01, 0xDC, 0x04, 0x10, 0x0A, 0xBB, 0x0E, 0xAA, 0x10,
+ 0xE1, 0x0E, 0x47, 0x0A, 0x0D, 0x05, 0x3D, 0x01, 0x97, 0xFF, 0x81,
+ 0xFF, 0xDD, 0xFF, 0x00, 0x00, 0xFD, 0xFF, 0xC2, 0xFF, 0x6F, 0xFF,
+ 0xE4, 0xFF, 0x34, 0x02, 0x9D, 0x06, 0xE6, 0x0B, 0xD1, 0x0F, 0x73,
+ 0x10, 0x79, 0x0D, 0x61, 0x08, 0x78, 0x03, 0x6A, 0x00, 0x70, 0xFF,
+ 0xA1, 0xFF, 0xF2, 0xFF, 0xF1, 0xFF, 0x9F, 0xFF, 0x72, 0xFF, 0x74,
+ 0x00, 0x8E, 0x03, 0x7D, 0x08, 0x90, 0x0D, 0x7A, 0x10, 0xC4, 0x0F,
+ 0xCC, 0x0B, 0x82, 0x06, 0x22, 0x02, 0xDD, 0xFF, 0x6F, 0xFF, 0xC4,
+ 0xFF, 0xFD, 0xFF, 0x00, 0x00, 0xDB, 0xFF, 0x7F, 0xFF, 0x9B, 0xFF,
+ 0x4B, 0x01, 0x26, 0x05, 0x63, 0x0A, 0xF3, 0x0E, 0xAA, 0x10, 0xA8,
+ 0x0E, 0xF4, 0x09, 0xC3, 0x04, 0x13, 0x01, 0x8D, 0xFF, 0x86, 0xFF,
+ 0xE1, 0xFF, 0xFC, 0xFF, 0xBC, 0xFF, 0x6D, 0xFF, 0xF8, 0xFF, 0x6B,
+ 0x02, 0xEE, 0x06, 0x34, 0x0C, 0xF7, 0x0F, 0x5D, 0x10, 0x33, 0x0D,
+ 0x0D, 0x08, 0x38, 0x03, 0x4D, 0x00, 0x6E, 0xFF, 0xA7, 0xFF, 0xF5,
+ 0xFF, 0xEE, 0xFF, 0x99, 0xFF, 0x76, 0xFF, 0x94, 0x00, 0xD0, 0x03,
+ 0xD1, 0x08, 0xD3, 0x0D, 0x8B, 0x10, 0x9A, 0x0F, 0x7C, 0x0B, 0x32,
+ 0x06, 0xEE, 0x01, 0xCB, 0xFF, 0x72, 0xFF, 0xCA, 0xFF, 0xFE, 0xFF,
+ 0x00, 0x00, 0xD6, 0xFF, 0x7B, 0xFF, 0xA7, 0xFF, 0x78, 0x01, 0x72,
+ 0x05, 0xB6, 0x0A, 0x27, 0x0F, 0xA5, 0x10, 0x6E, 0x0E, 0xA0, 0x09,
+ 0x7B, 0x04, 0xEC, 0x00, 0x85, 0xFF, 0x8B, 0xFF, 0xE5, 0xFF, 0xFA,
+ 0xFF, 0xB6, 0xFF, 0x6C, 0xFF, 0x0E, 0x00, 0xA4, 0x02, 0x41, 0x07,
+ 0x80, 0x0C, 0x19, 0x10, 0x44, 0x10, 0xEB, 0x0C, 0xB9, 0x07, 0xFA,
+ 0x02, 0x32, 0x00, 0x6D, 0xFF, 0xAE, 0xFF, 0xF7, 0xFF, 0xEB, 0xFF,
+ 0x93, 0xFF, 0x7B, 0xFF, 0xB7, 0x00, 0x15, 0x04, 0x26, 0x09, 0x14,
+ 0x0E, 0x98, 0x10, 0x6D, 0x0F, 0x2C, 0x0B, 0xE3, 0x05, 0xBC, 0x01,
+ 0xBB, 0xFF, 0x75, 0xFF, 0xCF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xD1,
+ 0xFF, 0x77, 0xFF, 0xB5, 0xFF, 0xA6, 0x01, 0xC0, 0x05, 0x08, 0x0B,
+ 0x58, 0x0F, 0x9D, 0x10, 0x30, 0x0E, 0x4B, 0x09, 0x34, 0x04, 0xC6,
+ 0x00, 0x7D, 0xFF, 0x91, 0xFF, 0xE9, 0xFF, 0xF8, 0xFF, 0xB0, 0xFF,
+ 0x6C, 0xFF, 0x27, 0x00, 0xDF, 0x02, 0x94, 0x07, 0xCA, 0x0C, 0x37,
+ 0x10, 0x27, 0x10, 0xA1, 0x0C, 0x65, 0x07, 0xBE, 0x02, 0x19, 0x00,
+ 0x6C, 0xFF, 0xB4, 0xFF, 0xF9, 0xFF, 0xE7, 0xFF, 0x8E, 0xFF, 0x81,
+ 0xFF, 0xDB, 0x00, 0x5B, 0x04, 0x7A, 0x09, 0x53, 0x0E, 0xA2, 0x10,
+ 0x3D, 0x0F, 0xDA, 0x0A, 0x95, 0x05, 0x8C, 0x01, 0xAD, 0xFF, 0x79,
+ 0xFF, 0xD4, 0xFF, 0x00, 0x00, 0xFF, 0xFF, 0xCC, 0xFF, 0x73, 0xFF,
+ 0xC4, 0xFF, 0xD7, 0x01, 0x0E, 0x06, 0x59, 0x0B, 0x87, 0x0F, 0x91,
+ 0x10, 0xF0, 0x0D, 0xF7, 0x08, 0xEF, 0x03, 0xA3, 0x00, 0x78, 0xFF,
+ 0x97, 0xFF, 0xED, 0xFF, 0xF6, 0xFF, 0xAA, 0xFF, 0x6D, 0xFF, 0x41,
+ 0x00, 0x1C, 0x03, 0xE7, 0x07, 0x13, 0x0D, 0x52, 0x10, 0x06, 0x10,
+ 0x56, 0x0C, 0x13, 0x07, 0x84, 0x02, 0x02, 0x00, 0x6D, 0xFF, 0xBA,
+ 0xFF, 0xFB, 0xFF, 0xE3, 0xFF, 0x88, 0xFF, 0x89, 0xFF, 0x01, 0x01,
+ 0xA3, 0x04, 0xCE, 0x09, 0x8F, 0x0E, 0xA8, 0x10, 0x0A, 0x0F, 0x88,
+ 0x0A, 0x48, 0x05, 0x5F, 0x01, 0xA0, 0xFF, 0x7D, 0xFF, 0xD9, 0xFF,
+ 0x00, 0x00, 0xFE, 0xFF, 0xC7, 0xFF, 0x70, 0xFF, 0xD5, 0xFF, 0x0B,
+ 0x02, 0x5E, 0x06, 0xA9, 0x0B, 0xB2, 0x0F, 0x82, 0x10, 0xAE, 0x0D,
+ 0xA2, 0x08, 0xAB, 0x03, 0x82, 0x00, 0x73, 0xFF, 0x9D, 0xFF, 0xF0,
+ 0xFF, 0xF3, 0xFF, 0xA4, 0xFF, 0x6F, 0xFF, 0x5D, 0x00, 0x5B, 0x03,
+ 0x3B, 0x08, 0x5A, 0x0D, 0x6A, 0x10, 0xE2, 0x0F, 0x09, 0x0C, 0xC1,
+ 0x06, 0x4C, 0x02, 0xEC, 0xFF, 0x6E, 0xFF, 0xC0, 0xFF, 0xFC, 0xFF,
+ 0xDF, 0xFF, 0x83, 0xFF, 0x93, 0xFF, 0x2A, 0x01, 0xEC, 0x04, 0x22,
+ 0x0A, 0xC8, 0x0E, 0xAB, 0x10, 0xD4, 0x0E, 0x35, 0x0A, 0xFD, 0x04,
+ 0x33, 0x01, 0x95, 0xFF, 0x82, 0xFF, 0xDE, 0xFF, 0x00, 0x00, 0xFD,
+ 0xFF, 0xC1, 0xFF, 0x6E, 0xFF, 0xE8, 0xFF, 0x40, 0x02, 0xAF, 0x06,
+ 0xF7, 0x0B, 0xDA, 0x0F, 0x6F, 0x10, 0x6A, 0x0D, 0x4E, 0x08, 0x6A,
+ 0x03, 0x64, 0x00, 0x70, 0xFF, 0xA3, 0xFF, 0xF3, 0xFF, 0xF1, 0xFF,
+ 0x9E, 0xFF, 0x72, 0xFF, 0x7B, 0x00, 0x9C, 0x03, 0x90, 0x08, 0x9F,
+ 0x0D, 0x7E, 0x10, 0xBB, 0x0F, 0xBA, 0x0B, 0x70, 0x06, 0x16, 0x02,
+ 0xD9, 0xFF, 0x70, 0xFF, 0xC5, 0xFF, 0xFE, 0xFF, 0x00, 0x00, 0xDA,
+ 0xFF, 0x7E, 0xFF, 0x9D, 0xFF, 0x55, 0x01, 0x37, 0x05, 0x75, 0x0A,
+ 0xFF, 0x0E, 0xA9, 0x10, 0x9C, 0x0E, 0xE1, 0x09, 0xB3, 0x04, 0x0A,
+ 0x01, 0x8B, 0xFF, 0x87, 0xFF, 0xE2, 0xFF, 0xFB, 0xFF, 0xBB, 0xFF,
+ 0x6D, 0xFF, 0xFD, 0xFF, 0x77, 0x02, 0x01, 0x07, 0x45, 0x0C, 0xFF,
+ 0x0F, 0x58, 0x10, 0x23, 0x0D, 0xFA, 0x07, 0x2A, 0x03, 0x47, 0x00,
+ 0x6E, 0xFF, 0xA9, 0xFF, 0xF5, 0xFF, 0xED, 0xFF, 0x98, 0xFF, 0x77,
+ 0xFF, 0x9C, 0x00, 0xDF, 0x03, 0xE4, 0x08, 0xE2, 0x0D, 0x8E, 0x10,
+ 0x91, 0x0F, 0x6B, 0x0B, 0x20, 0x06, 0xE3, 0x01, 0xC8, 0xFF, 0x73,
+ 0xFF, 0xCB, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0xD5, 0xFF, 0x7A, 0xFF,
+ 0xAA, 0xFF, 0x82, 0x01, 0x83, 0x05, 0xC8, 0x0A, 0x32, 0x0F, 0xA4,
+ 0x10, 0x60, 0x0E, 0x8D, 0x09, 0x6B, 0x04, 0xE3, 0x00, 0x83, 0xFF,
+ 0x8D, 0xFF, 0xE6, 0xFF, 0xFA, 0xFF, 0xB5, 0xFF, 0x6C, 0xFF, 0x14,
+ 0x00, 0xB1, 0x02, 0x53, 0x07, 0x91, 0x0C, 0x20, 0x10, 0x3E, 0x10,
+ 0xDB, 0x0C, 0xA6, 0x07, 0xEC, 0x02, 0x2C, 0x00, 0x6C, 0xFF, 0xAF,
+ 0xFF, 0xF8, 0xFF, 0xEA, 0xFF, 0x92, 0xFF, 0x7C, 0xFF, 0xBE, 0x00,
+ 0x24, 0x04, 0x38, 0x09, 0x22, 0x0E, 0x9B, 0x10, 0x63, 0x0F, 0x1A,
+ 0x0B, 0xD1, 0x05, 0xB1, 0x01, 0xB8, 0xFF, 0x76, 0xFF, 0xD0, 0xFF,
+ 0xFF, 0xFF, 0xFF, 0xFF, 0xD0, 0xFF, 0x76, 0xFF, 0xB8, 0xFF, 0xB1,
+ 0x01, 0xD1, 0x05, 0x1A, 0x0B, 0x63, 0x0F, 0x9B, 0x10, 0x22, 0x0E,
+ 0x38, 0x09, 0x24, 0x04, 0xBE, 0x00, 0x7C, 0xFF, 0x92, 0xFF, 0xEA,
+ 0xFF, 0xF8, 0xFF, 0xAF, 0xFF, 0x6C, 0xFF, 0x2C, 0x00, 0xEC, 0x02,
+ 0xA6, 0x07, 0xDB, 0x0C, 0x3E, 0x10, 0x20, 0x10, 0x91, 0x0C, 0x53,
+ 0x07, 0xB1, 0x02, 0x14, 0x00, 0x6C, 0xFF, 0xB5, 0xFF, 0xFA, 0xFF,
+ 0xE6, 0xFF, 0x8D, 0xFF, 0x83, 0xFF, 0xE3, 0x00, 0x6B, 0x04, 0x8D,
+ 0x09, 0x60, 0x0E, 0xA4, 0x10, 0x32, 0x0F, 0xC8, 0x0A, 0x83, 0x05,
+ 0x82, 0x01, 0xAA, 0xFF, 0x7A, 0xFF, 0xD5, 0xFF, 0x00, 0x00, 0xFF,
+ 0xFF, 0xCB, 0xFF, 0x73, 0xFF, 0xC8, 0xFF, 0xE3, 0x01, 0x20, 0x06,
+ 0x6B, 0x0B, 0x91, 0x0F, 0x8E, 0x10, 0xE2, 0x0D, 0xE4, 0x08, 0xDF,
+ 0x03, 0x9C, 0x00, 0x77, 0xFF, 0x98, 0xFF, 0xED, 0xFF, 0xF5, 0xFF,
+ 0xA9, 0xFF, 0x6E, 0xFF, 0x47, 0x00, 0x2A, 0x03, 0xFA, 0x07, 0x23,
+ 0x0D, 0x58, 0x10, 0xFF, 0x0F, 0x45, 0x0C, 0x01, 0x07, 0x77, 0x02,
+ 0xFD, 0xFF, 0x6D, 0xFF, 0xBB, 0xFF, 0xFB, 0xFF, 0xE2, 0xFF, 0x87,
+ 0xFF, 0x8B, 0xFF, 0x0A, 0x01, 0xB3, 0x04, 0xE1, 0x09, 0x9C, 0x0E,
+ 0xA9, 0x10, 0xFF, 0x0E, 0x75, 0x0A, 0x37, 0x05, 0x55, 0x01, 0x9D,
+ 0xFF, 0x7E, 0xFF, 0xDA, 0xFF, 0x00, 0x00, 0xFE, 0xFF, 0xC5, 0xFF,
+ 0x70, 0xFF, 0xD9, 0xFF, 0x16, 0x02, 0x70, 0x06, 0xBA, 0x0B, 0xBB,
+ 0x0F, 0x7E, 0x10, 0x9F, 0x0D, 0x90, 0x08, 0x9C, 0x03, 0x7B, 0x00,
+ 0x72, 0xFF, 0x9E, 0xFF, 0xF1, 0xFF, 0xF3, 0xFF, 0xA3, 0xFF, 0x70,
+ 0xFF, 0x64, 0x00, 0x6A, 0x03, 0x4E, 0x08, 0x6A, 0x0D, 0x6F, 0x10,
+ 0xDA, 0x0F, 0xF7, 0x0B, 0xAF, 0x06, 0x40, 0x02, 0xE8, 0xFF, 0x6E,
+ 0xFF, 0xC1, 0xFF, 0xFD, 0xFF, 0x00, 0x00, 0xDE, 0xFF, 0x82, 0xFF,
+ 0x95, 0xFF, 0x33, 0x01, 0xFD, 0x04, 0x35, 0x0A, 0xD4, 0x0E, 0xAB,
+ 0x10, 0xC8, 0x0E, 0x22, 0x0A, 0xEC, 0x04, 0x2A, 0x01, 0x93, 0xFF,
+ 0x83, 0xFF, 0xDF, 0xFF, 0xFC, 0xFF, 0xC0, 0xFF, 0x6E, 0xFF, 0xEC,
+ 0xFF, 0x4C, 0x02, 0xC1, 0x06, 0x09, 0x0C, 0xE2, 0x0F, 0x6A, 0x10,
+ 0x5A, 0x0D, 0x3B, 0x08, 0x5B, 0x03, 0x5D, 0x00, 0x6F, 0xFF, 0xA4,
+ 0xFF, 0xF3, 0xFF, 0xF0, 0xFF, 0x9D, 0xFF, 0x73, 0xFF, 0x82, 0x00,
+ 0xAB, 0x03, 0xA2, 0x08, 0xAE, 0x0D, 0x82, 0x10, 0xB2, 0x0F, 0xA9,
+ 0x0B, 0x5E, 0x06, 0x0B, 0x02, 0xD5, 0xFF, 0x70, 0xFF, 0xC7, 0xFF,
+ 0xFE, 0xFF, 0x00, 0x00, 0xD9, 0xFF, 0x7D, 0xFF, 0xA0, 0xFF, 0x5F,
+ 0x01, 0x48, 0x05, 0x88, 0x0A, 0x0A, 0x0F, 0xA8, 0x10, 0x8F, 0x0E,
+ 0xCE, 0x09, 0xA3, 0x04, 0x01, 0x01, 0x89, 0xFF, 0x88, 0xFF, 0xE3,
+ 0xFF, 0xFB, 0xFF, 0xBA, 0xFF, 0x6D, 0xFF, 0x02, 0x00, 0x84, 0x02,
+ 0x13, 0x07, 0x56, 0x0C, 0x06, 0x10, 0x52, 0x10, 0x13, 0x0D, 0xE7,
+ 0x07, 0x1C, 0x03, 0x41, 0x00, 0x6D, 0xFF, 0xAA, 0xFF, 0xF6, 0xFF,
+ 0xED, 0xFF, 0x97, 0xFF, 0x78, 0xFF, 0xA3, 0x00, 0xEF, 0x03, 0xF7,
+ 0x08, 0xF0, 0x0D, 0x91, 0x10, 0x87, 0x0F, 0x59, 0x0B, 0x0E, 0x06,
+ 0xD7, 0x01, 0xC4, 0xFF, 0x73, 0xFF, 0xCC, 0xFF, 0xFF, 0xFF, 0x00,
+ 0x00, 0xD4, 0xFF, 0x79, 0xFF, 0xAD, 0xFF, 0x8C, 0x01, 0x95, 0x05,
+ 0xDA, 0x0A, 0x3D, 0x0F, 0xA2, 0x10, 0x53, 0x0E, 0x7A, 0x09, 0x5B,
+ 0x04, 0xDB, 0x00, 0x81, 0xFF, 0x8E, 0xFF, 0xE7, 0xFF, 0xF9, 0xFF,
+ 0xB4, 0xFF, 0x6C, 0xFF, 0x19, 0x00, 0xBE, 0x02, 0x65, 0x07, 0xA1,
+ 0x0C, 0x27, 0x10, 0x37, 0x10, 0xCA, 0x0C, 0x94, 0x07, 0xDF, 0x02,
+ 0x27, 0x00, 0x6C, 0xFF, 0xB0, 0xFF, 0xF8, 0xFF, 0xE9, 0xFF, 0x91,
+ 0xFF, 0x7D, 0xFF, 0xC6, 0x00, 0x34, 0x04, 0x4B, 0x09, 0x30, 0x0E,
+ 0x9D, 0x10, 0x58, 0x0F, 0x08, 0x0B, 0xC0, 0x05, 0xA6, 0x01, 0xB5,
+ 0xFF, 0x77, 0xFF, 0xD1, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xCF, 0xFF,
+ 0x75, 0xFF, 0xBB, 0xFF, 0xBC, 0x01, 0xE3, 0x05, 0x2C, 0x0B, 0x6D,
+ 0x0F, 0x98, 0x10, 0x14, 0x0E, 0x26, 0x09, 0x15, 0x04, 0xB7, 0x00,
+ 0x7B, 0xFF, 0x93, 0xFF, 0xEB, 0xFF, 0xF7, 0xFF, 0xAE, 0xFF, 0x6D,
+ 0xFF, 0x32, 0x00, 0xFA, 0x02, 0xB9, 0x07, 0xEB, 0x0C, 0x44, 0x10,
+ 0x19, 0x10, 0x80, 0x0C, 0x41, 0x07, 0xA4, 0x02, 0x0E, 0x00, 0x6C,
+ 0xFF, 0xB6, 0xFF, 0xFA, 0xFF, 0xE5, 0xFF, 0x8B, 0xFF, 0x85, 0xFF,
+ 0xEC, 0x00, 0x7B, 0x04, 0xA0, 0x09, 0x6E, 0x0E, 0xA5, 0x10, 0x27,
+ 0x0F, 0xB6, 0x0A, 0x72, 0x05, 0x78, 0x01, 0xA7, 0xFF, 0x7B, 0xFF,
+ 0xD6, 0xFF, 0x00, 0x00, 0xFE, 0xFF, 0xCA, 0xFF, 0x72, 0xFF, 0xCB,
+ 0xFF, 0xEE, 0x01, 0x32, 0x06, 0x7C, 0x0B, 0x9A, 0x0F, 0x8B, 0x10,
+ 0xD3, 0x0D, 0xD1, 0x08, 0xD0, 0x03, 0x94, 0x00, 0x76, 0xFF, 0x99,
+ 0xFF, 0xEE, 0xFF, 0xF5, 0xFF, 0xA7, 0xFF, 0x6E, 0xFF, 0x4D, 0x00,
+ 0x38, 0x03, 0x0D, 0x08, 0x33, 0x0D, 0x5D, 0x10, 0xF7, 0x0F, 0x34,
+ 0x0C, 0xEE, 0x06, 0x6B, 0x02, 0xF8, 0xFF, 0x6D, 0xFF, 0xBC, 0xFF,
+ 0xFC, 0xFF, 0xE1, 0xFF, 0x86, 0xFF, 0x8D, 0xFF, 0x13, 0x01, 0xC3,
+ 0x04, 0xF4, 0x09, 0xA8, 0x0E, 0xAA, 0x10, 0xF3, 0x0E, 0x63, 0x0A,
+ 0x26, 0x05, 0x4B, 0x01, 0x9B, 0xFF, 0x7F, 0xFF, 0xDB, 0xFF, 0x00,
+ 0x00, 0xFD, 0xFF, 0xC4, 0xFF, 0x6F, 0xFF, 0xDD, 0xFF, 0x22, 0x02,
+ 0x82, 0x06, 0xCC, 0x0B, 0xC4, 0x0F, 0x7A, 0x10, 0x90, 0x0D, 0x7D,
+ 0x08, 0x8E, 0x03, 0x74, 0x00, 0x72, 0xFF, 0x9F, 0xFF, 0xF1, 0xFF,
+ 0xF2, 0xFF, 0xA1, 0xFF, 0x70, 0xFF, 0x6A, 0x00, 0x78, 0x03, 0x61,
+ 0x08, 0x79, 0x0D, 0x73, 0x10, 0xD1, 0x0F, 0xE6, 0x0B, 0x9D, 0x06,
+ 0x34, 0x02, 0xE4, 0xFF, 0x6F, 0xFF, 0xC2, 0xFF, 0xFD, 0xFF, 0x00,
+ 0x00, 0xDD, 0xFF, 0x81, 0xFF, 0x97, 0xFF, 0x3D, 0x01, 0x0D, 0x05,
+ 0x47, 0x0A, 0xE1, 0x0E, 0xAA, 0x10, 0xBB, 0x0E, 0x10, 0x0A, 0xDC,
+ 0x04, 0x21, 0x01, 0x90, 0xFF, 0x84, 0xFF, 0xE0, 0xFF, 0xFC, 0xFF,
+ 0xBE, 0xFF, 0x6D, 0xFF, 0xF1, 0xFF, 0x58, 0x02, 0xD3, 0x06, 0x1A,
+ 0x0C, 0xEB, 0x0F, 0x65, 0x10, 0x4B, 0x0D, 0x29, 0x08, 0x4D, 0x03,
+ 0x57, 0x00, 0x6F, 0xFF, 0xA5, 0xFF, 0xF4, 0xFF, 0xEF, 0xFF, 0x9B,
+ 0xFF, 0x74, 0xFF, 0x8A, 0x00, 0xBA, 0x03, 0xB5, 0x08, 0xBD, 0x0D,
+ 0x86, 0x10, 0xA9, 0x0F, 0x97, 0x0B, 0x4C, 0x06, 0xFF, 0x01, 0xD1,
+ 0xFF, 0x71, 0xFF, 0xC8, 0xFF, 0xFE, 0xFF, 0x00, 0x00, 0xD8, 0xFF,
+ 0x7C, 0xFF, 0xA3, 0xFF, 0x69, 0x01, 0x59, 0x05, 0x9A, 0x0A, 0x16,
+ 0x0F, 0xA7, 0x10, 0x82, 0x0E, 0xBC, 0x09, 0x93, 0x04, 0xF9, 0x00,
+ 0x87, 0xFF, 0x89, 0xFF, 0xE4, 0xFF, 0xFB, 0xFF, 0xB8, 0xFF, 0x6C,
+ 0xFF, 0x07, 0x00, 0x91, 0x02, 0x25, 0x07, 0x67, 0x0C, 0x0E, 0x10,
+ 0x4D, 0x10, 0x03, 0x0D, 0xD5, 0x07, 0x0E, 0x03, 0x3B, 0x00, 0x6D,
+ 0xFF, 0xAC, 0xFF, 0xF7, 0xFF, 0xEC, 0xFF, 0x95, 0xFF, 0x79, 0xFF,
+ 0xAB, 0x00, 0xFE, 0x03, 0x0A, 0x09, 0xFF, 0x0D, 0x94, 0x10, 0x7D,
+ 0x0F, 0x47, 0x0B, 0xFD, 0x05, 0xCC, 0x01, 0xC0, 0xFF, 0x74, 0xFF,
+ 0xCD, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0xD3, 0xFF, 0x78, 0xFF, 0xB0,
+ 0xFF, 0x97, 0x01, 0xA6, 0x05, 0xEC, 0x0A, 0x48, 0x0F, 0xA0, 0x10,
+ 0x45, 0x0E, 0x67, 0x09, 0x4B, 0x04, 0xD3, 0x00, 0x80, 0xFF, 0x8F,
+ 0xFF, 0xE8, 0xFF, 0xF9, 0xFF, 0xB2, 0xFF, 0x6C, 0xFF, 0x1E, 0x00,
+ 0xCB, 0x02, 0x78, 0x07, 0xB2, 0x0C, 0x2E, 0x10, 0x31, 0x10, 0xBA,
+ 0x0C, 0x81, 0x07, 0xD2, 0x02, 0x21, 0x00, 0x6C, 0xFF, 0xB2, 0xFF,
+ 0xF9, 0xFF, 0xE8, 0xFF, 0x90, 0xFF, 0x7F, 0xFF, 0xCF, 0x00, 0x43,
+ 0x04, 0x5E, 0x09, 0x3E, 0x0E, 0x9F, 0x10, 0x4E, 0x0F, 0xF6, 0x0A,
+ 0xAE, 0x05, 0x9C, 0x01, 0xB1, 0xFF, 0x78, 0xFF, 0xD2, 0xFF, 0xFF,
+ 0xFF, 0xFF, 0xFF, 0xCE, 0xFF, 0x74, 0xFF, 0xBF, 0xFF, 0xC7, 0x01,
+ 0xF4, 0x05, 0x3E, 0x0B, 0x78, 0x0F, 0x96, 0x10, 0x06, 0x0E, 0x13,
+ 0x09, 0x05, 0x04, 0xAF, 0x00, 0x79, 0xFF, 0x95, 0xFF, 0xEC, 0xFF,
+ 0xF7, 0xFF, 0xAC, 0xFF, 0x6D, 0xFF, 0x38, 0x00, 0x07, 0x03, 0xCB,
+ 0x07, 0xFB, 0x0C, 0x4A, 0x10, 0x11, 0x10, 0x6F, 0x0C, 0x2E, 0x07,
+ 0x97, 0x02, 0x09, 0x00, 0x6C, 0xFF, 0xB8, 0xFF, 0xFA, 0xFF, 0xE4,
+ 0xFF, 0x8A, 0xFF, 0x86, 0xFF, 0xF4, 0x00, 0x8B, 0x04, 0xB2, 0x09,
+ 0x7B, 0x0E, 0xA7, 0x10, 0x1C, 0x0F, 0xA3, 0x0A, 0x61, 0x05, 0x6E,
+ 0x01, 0xA4, 0xFF, 0x7C, 0xFF, 0xD8, 0xFF, 0x00, 0x00, 0xFE, 0xFF,
+ 0xC8, 0xFF, 0x71, 0xFF, 0xCF, 0xFF, 0xF9, 0x01, 0x43, 0x06, 0x8E,
+ 0x0B, 0xA4, 0x0F, 0x88, 0x10, 0xC4, 0x0D, 0xBE, 0x08, 0xC1, 0x03,
+ 0x8D, 0x00, 0x75, 0xFF, 0x9B, 0xFF, 0xEF, 0xFF, 0xF4, 0xFF, 0xA6,
+ 0xFF, 0x6E, 0xFF, 0x53, 0x00, 0x46, 0x03, 0x1F, 0x08, 0x43, 0x0D,
+ 0x63, 0x10, 0xEF, 0x0F, 0x23, 0x0C, 0xDC, 0x06, 0x5E, 0x02, 0xF3,
+ 0xFF, 0x6D, 0xFF, 0xBE, 0xFF, 0xFC, 0xFF, 0xE0, 0xFF, 0x85, 0xFF,
+ 0x8F, 0xFF, 0x1C, 0x01, 0xD3, 0x04, 0x06, 0x0A, 0xB5, 0x0E, 0xAA,
+ 0x10, 0xE7, 0x0E, 0x50, 0x0A, 0x16, 0x05, 0x42, 0x01, 0x98, 0xFF,
+ 0x80, 0xFF, 0xDC, 0xFF, 0x00, 0x00, 0xFD, 0xFF, 0xC3, 0xFF, 0x6F,
+ 0xFF, 0xE1, 0xFF, 0x2E, 0x02, 0x94, 0x06, 0xDD, 0x0B, 0xCD, 0x0F,
+ 0x76, 0x10, 0x81, 0x0D, 0x6A, 0x08, 0x7F, 0x03, 0x6E, 0x00, 0x71,
+ 0xFF, 0xA1, 0xFF, 0xF2, 0xFF, 0x00, 0x00, 0x15, 0x00, 0xD1, 0xFF,
+ 0x8B, 0xFE, 0xBC, 0xFD, 0xE1, 0x00, 0x84, 0x09, 0xB0, 0x13, 0x47,
+ 0x18, 0xB0, 0x13, 0x84, 0x09, 0xE1, 0x00, 0xBC, 0xFD, 0x8B, 0xFE,
+ 0xD1, 0xFF, 0x15, 0x00, 0xFD, 0xFF, 0x13, 0x00, 0xDA, 0x00, 0x30,
+ 0x00, 0x5D, 0xFC, 0xB3, 0xFC, 0x35, 0x0A, 0xC2, 0x1C, 0x24, 0x20,
+ 0x48, 0x10, 0x5D, 0xFF, 0x74, 0xFB, 0x3A, 0xFF, 0xFB, 0x00, 0x42,
+ 0x00, 0xF8, 0xFF, 0xFA, 0xFF, 0x2C, 0x00, 0xF3, 0x00, 0xAD, 0xFF,
+ 0xC5, 0xFB, 0x11, 0xFE, 0xAF, 0x0D, 0xEF, 0x1E, 0x68, 0x1E, 0xBC,
+ 0x0C, 0xA7, 0xFD, 0xEA, 0xFB, 0xD3, 0xFF, 0xEE, 0x00, 0x24, 0x00,
+ 0xFA, 0xFF, 0xF7, 0xFF, 0x4C, 0x00, 0xFB, 0x00, 0x0C, 0xFF, 0x5F,
+ 0xFB, 0xE8, 0xFF, 0x3D, 0x11, 0x7E, 0x20, 0x13, 0x1C, 0x4C, 0x09,
+ 0x6A, 0xFC, 0x8C, 0xFC, 0x4E, 0x00, 0xD1, 0x00, 0x0E, 0x00, 0xFD,
+ 0xFF, 0xF7, 0xFF, 0x72, 0x00, 0xEC, 0x00, 0x55, 0xFE, 0x3D, 0xFB,
+ 0x37, 0x02, 0xBE, 0x14, 0x5D, 0x21, 0x40, 0x19, 0x18, 0x06, 0xA2,
+ 0xFB, 0x47, 0xFD, 0xA7, 0x00, 0xAB, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0xFC, 0xFF, 0x9B, 0x00, 0xC0, 0x00, 0x92, 0xFD, 0x73,
+ 0xFB, 0xF2, 0x04, 0x0E, 0x18, 0x81, 0x21, 0x0C, 0x16, 0x37, 0x03,
+ 0x47, 0xFB, 0x0B, 0xFE, 0xDF, 0x00, 0x82, 0x00, 0xF9, 0xFF, 0xFE,
+ 0xFF, 0x08, 0x00, 0xC3, 0x00, 0x74, 0x00, 0xD2, 0xFC, 0x10, 0xFC,
+ 0x08, 0x08, 0x0A, 0x1B, 0xE9, 0x20, 0x9A, 0x12, 0xBE, 0x00, 0x49,
+ 0xFB, 0xC8, 0xFE, 0xF9, 0x00, 0x5A, 0x00, 0xF7, 0xFF, 0xFC, 0xFF,
+ 0x1B, 0x00, 0xE4, 0x00, 0x06, 0x00, 0x24, 0xFC, 0x1E, 0xFD, 0x65,
+ 0x0B, 0x94, 0x1D, 0x9D, 0x1F, 0x0D, 0x0F, 0xB8, 0xFE, 0x96, 0xFB,
+ 0x72, 0xFF, 0xF9, 0x00, 0x37, 0x00, 0xF8, 0xFF, 0xF9, 0xFF, 0x36,
+ 0x00, 0xF8, 0x00, 0x78, 0xFF, 0x9B, 0xFB, 0xA6, 0xFE, 0xE9, 0x0E,
+ 0x8D, 0x1F, 0xAA, 0x1D, 0x87, 0x0B, 0x2B, 0xFD, 0x1E, 0xFC, 0x02,
+ 0x00, 0xE5, 0x00, 0x1C, 0x00, 0xFB, 0xFF, 0xF7, 0xFF, 0x58, 0x00,
+ 0xF9, 0x00, 0xCF, 0xFE, 0x4A, 0xFB, 0xA7, 0x00, 0x77, 0x12, 0xE0,
+ 0x20, 0x26, 0x1B, 0x28, 0x08, 0x18, 0xFC, 0xCB, 0xFC, 0x71, 0x00,
+ 0xC5, 0x00, 0x08, 0x00, 0xFE, 0xFF, 0xF8, 0xFF, 0x80, 0x00, 0xE1,
+ 0x00, 0x13, 0xFE, 0x45, 0xFB, 0x1D, 0x03, 0xEB, 0x15, 0x7F, 0x21,
+ 0x2D, 0x18, 0x0E, 0x05, 0x77, 0xFB, 0x8B, 0xFD, 0xBE, 0x00, 0x9D,
+ 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0xA9, 0x00,
+ 0xAA, 0x00, 0x4F, 0xFD, 0x9D, 0xFB, 0xFA, 0x05, 0x22, 0x19, 0x62,
+ 0x21, 0xE0, 0x14, 0x50, 0x02, 0x3E, 0xFB, 0x4E, 0xFE, 0xEB, 0x00,
+ 0x73, 0x00, 0xF7, 0xFF, 0xFE, 0xFF, 0x0D, 0x00, 0xD0, 0x00, 0x52,
+ 0x00, 0x93, 0xFC, 0x60, 0xFC, 0x2C, 0x09, 0xFA, 0x1B, 0x8A, 0x20,
+ 0x60, 0x11, 0xFD, 0xFF, 0x5C, 0xFB, 0x06, 0xFF, 0xFB, 0x00, 0x4D,
+ 0x00, 0xF7, 0xFF, 0xFA, 0xFF, 0x23, 0x00, 0xED, 0x00, 0xD9, 0xFF,
+ 0xEF, 0xFB, 0x98, 0xFD, 0x99, 0x0C, 0x54, 0x1E, 0x02, 0x1F, 0xD2,
+ 0x0D, 0x20, 0xFE, 0xC0, 0xFB, 0xA7, 0xFF, 0xF4, 0x00, 0x2D, 0x00,
+ 0xF9, 0xFF, 0xF8, 0xFF, 0x41, 0x00, 0xFB, 0x00, 0x41, 0xFF, 0x78,
+ 0xFB, 0x4A, 0xFF, 0x25, 0x10, 0x16, 0x20, 0xDA, 0x1C, 0x56, 0x0A,
+ 0xBE, 0xFC, 0x56, 0xFC, 0x2C, 0x00, 0xDB, 0x00, 0x14, 0x00, 0xFD,
+ 0xFF, 0xF7, 0xFF, 0x66, 0x00, 0xF4, 0x00, 0x8F, 0xFE, 0x3F, 0xFB,
+ 0x75, 0x01, 0xAE, 0x13, 0x2C, 0x21, 0x2A, 0x1A, 0x0D, 0x07, 0xD4,
+ 0xFB, 0x0C, 0xFD, 0x8F, 0x00, 0xB7, 0x00, 0x03, 0x00, 0xFF, 0xFF,
+ 0x00, 0x00, 0xFA, 0xFF, 0x8E, 0x00, 0xD1, 0x00, 0xCF, 0xFD, 0x58,
+ 0xFB, 0x10, 0x04, 0x10, 0x17, 0x8A, 0x21, 0x10, 0x17, 0x10, 0x04,
+ 0x58, 0xFB, 0xCF, 0xFD, 0xD1, 0x00, 0x8E, 0x00, 0xFA, 0xFF, 0xFF,
+ 0xFF, 0x03, 0x00, 0xB7, 0x00, 0x8F, 0x00, 0x0C, 0xFD, 0xD4, 0xFB,
+ 0x0D, 0x07, 0x2A, 0x1A, 0x2C, 0x21, 0xAE, 0x13, 0x75, 0x01, 0x3F,
+ 0xFB, 0x8F, 0xFE, 0xF4, 0x00, 0x66, 0x00, 0xF7, 0xFF, 0xFD, 0xFF,
+ 0x14, 0x00, 0xDB, 0x00, 0x2C, 0x00, 0x56, 0xFC, 0xBE, 0xFC, 0x56,
+ 0x0A, 0xDA, 0x1C, 0x16, 0x20, 0x25, 0x10, 0x4A, 0xFF, 0x78, 0xFB,
+ 0x41, 0xFF, 0xFB, 0x00, 0x41, 0x00, 0xF8, 0xFF, 0xF9, 0xFF, 0x2D,
+ 0x00, 0xF4, 0x00, 0xA7, 0xFF, 0xC0, 0xFB, 0x20, 0xFE, 0xD2, 0x0D,
+ 0x02, 0x1F, 0x54, 0x1E, 0x99, 0x0C, 0x98, 0xFD, 0xEF, 0xFB, 0xD9,
+ 0xFF, 0xED, 0x00, 0x23, 0x00, 0xFA, 0xFF, 0xF7, 0xFF, 0x4D, 0x00,
+ 0xFB, 0x00, 0x06, 0xFF, 0x5C, 0xFB, 0xFD, 0xFF, 0x60, 0x11, 0x8A,
+ 0x20, 0xFA, 0x1B, 0x2C, 0x09, 0x60, 0xFC, 0x93, 0xFC, 0x52, 0x00,
+ 0xD0, 0x00, 0x0D, 0x00, 0xFE, 0xFF, 0xF7, 0xFF, 0x73, 0x00, 0xEB,
+ 0x00, 0x4E, 0xFE, 0x3E, 0xFB, 0x50, 0x02, 0xE0, 0x14, 0x62, 0x21,
+ 0x22, 0x19, 0xFA, 0x05, 0x9D, 0xFB, 0x4F, 0xFD, 0xAA, 0x00, 0xA9,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0x9D, 0x00,
+ 0xBE, 0x00, 0x8B, 0xFD, 0x77, 0xFB, 0x0E, 0x05, 0x2D, 0x18, 0x7F,
+ 0x21, 0xEB, 0x15, 0x1D, 0x03, 0x45, 0xFB, 0x13, 0xFE, 0xE1, 0x00,
+ 0x80, 0x00, 0xF8, 0xFF, 0xFE, 0xFF, 0x08, 0x00, 0xC5, 0x00, 0x71,
+ 0x00, 0xCB, 0xFC, 0x18, 0xFC, 0x28, 0x08, 0x26, 0x1B, 0xE0, 0x20,
+ 0x77, 0x12, 0xA7, 0x00, 0x4A, 0xFB, 0xCF, 0xFE, 0xF9, 0x00, 0x58,
+ 0x00, 0xF7, 0xFF, 0xFB, 0xFF, 0x1C, 0x00, 0xE5, 0x00, 0x02, 0x00,
+ 0x1E, 0xFC, 0x2B, 0xFD, 0x87, 0x0B, 0xAA, 0x1D, 0x8D, 0x1F, 0xE9,
+ 0x0E, 0xA6, 0xFE, 0x9B, 0xFB, 0x78, 0xFF, 0xF8, 0x00, 0x36, 0x00,
+ 0xF9, 0xFF, 0xF8, 0xFF, 0x37, 0x00, 0xF9, 0x00, 0x72, 0xFF, 0x96,
+ 0xFB, 0xB8, 0xFE, 0x0D, 0x0F, 0x9D, 0x1F, 0x94, 0x1D, 0x65, 0x0B,
+ 0x1E, 0xFD, 0x24, 0xFC, 0x06, 0x00, 0xE4, 0x00, 0x1B, 0x00, 0xFC,
+ 0xFF, 0xF7, 0xFF, 0x5A, 0x00, 0xF9, 0x00, 0xC8, 0xFE, 0x49, 0xFB,
+ 0xBE, 0x00, 0x9A, 0x12, 0xE9, 0x20, 0x0A, 0x1B, 0x08, 0x08, 0x10,
+ 0xFC, 0xD2, 0xFC, 0x74, 0x00, 0xC3, 0x00, 0x08, 0x00, 0xFE, 0xFF,
+ 0xF9, 0xFF, 0x82, 0x00, 0xDF, 0x00, 0x0B, 0xFE, 0x47, 0xFB, 0x37,
+ 0x03, 0x0C, 0x16, 0x81, 0x21, 0x0E, 0x18, 0xF2, 0x04, 0x73, 0xFB,
+ 0x92, 0xFD, 0xC0, 0x00, 0x9B, 0x00, 0xFC, 0xFF, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0xAB, 0x00, 0xA7, 0x00, 0x47, 0xFD, 0xA2, 0xFB,
+ 0x18, 0x06, 0x40, 0x19, 0x5D, 0x21, 0xBE, 0x14, 0x37, 0x02, 0x3D,
+ 0xFB, 0x55, 0xFE, 0xEC, 0x00, 0x72, 0x00, 0xF7, 0xFF, 0xFD, 0xFF,
+ 0x0E, 0x00, 0xD1, 0x00, 0x4E, 0x00, 0x8C, 0xFC, 0x6A, 0xFC, 0x4C,
+ 0x09, 0x13, 0x1C, 0x7E, 0x20, 0x3D, 0x11, 0xE8, 0xFF, 0x5F, 0xFB,
+ 0x0C, 0xFF, 0xFB, 0x00, 0x4C, 0x00, 0xF7, 0xFF, 0xFA, 0xFF, 0x24,
+ 0x00, 0xEE, 0x00, 0xD3, 0xFF, 0xEA, 0xFB, 0xA7, 0xFD, 0xBC, 0x0C,
+ 0x68, 0x1E, 0xEF, 0x1E, 0xAF, 0x0D, 0x11, 0xFE, 0xC5, 0xFB, 0xAD,
+ 0xFF, 0xF3, 0x00, 0x2C, 0x00, 0xFA, 0xFF, 0xF8, 0xFF, 0x42, 0x00,
+ 0xFB, 0x00, 0x3A, 0xFF, 0x74, 0xFB, 0x5D, 0xFF, 0x48, 0x10, 0x24,
+ 0x20, 0xC2, 0x1C, 0x35, 0x0A, 0xB3, 0xFC, 0x5D, 0xFC, 0x30, 0x00,
+ 0xDA, 0x00, 0x13, 0x00, 0xFD, 0xFF, 0xF7, 0xFF, 0x67, 0x00, 0xF3,
+ 0x00, 0x88, 0xFE, 0x3E, 0xFB, 0x8C, 0x01, 0xD0, 0x13, 0x33, 0x21,
+ 0x0D, 0x1A, 0xEE, 0x06, 0xCD, 0xFB, 0x13, 0xFD, 0x92, 0x00, 0xB6,
+ 0x00, 0x03, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0xFA, 0xFF, 0x90, 0x00,
+ 0xCF, 0x00, 0xC7, 0xFD, 0x5B, 0xFB, 0x2B, 0x04, 0x31, 0x17, 0x8A,
+ 0x21, 0xF0, 0x16, 0xF4, 0x03, 0x56, 0xFB, 0xD6, 0xFD, 0xD3, 0x00,
+ 0x8D, 0x00, 0xFA, 0xFF, 0xFF, 0xFF, 0x04, 0x00, 0xB9, 0x00, 0x8C,
+ 0x00, 0x05, 0xFD, 0xDB, 0xFB, 0x2C, 0x07, 0x47, 0x1A, 0x25, 0x21,
+ 0x8B, 0x13, 0x5D, 0x01, 0x40, 0xFB, 0x97, 0xFE, 0xF5, 0x00, 0x64,
+ 0x00, 0xF7, 0xFF, 0xFC, 0xFF, 0x15, 0x00, 0xDC, 0x00, 0x27, 0x00,
+ 0x50, 0xFC, 0xCA, 0xFC, 0x78, 0x0A, 0xF2, 0x1C, 0x07, 0x20, 0x02,
+ 0x10, 0x37, 0xFF, 0x7B, 0xFB, 0x47, 0xFF, 0xFB, 0x00, 0x40, 0x00,
+ 0xF8, 0xFF, 0xF9, 0xFF, 0x2E, 0x00, 0xF5, 0x00, 0xA2, 0xFF, 0xBB,
+ 0xFB, 0x31, 0xFE, 0xF5, 0x0D, 0x14, 0x1F, 0x3F, 0x1E, 0x77, 0x0C,
+ 0x8A, 0xFD, 0xF5, 0xFB, 0xDE, 0xFF, 0xEC, 0x00, 0x22, 0x00, 0xFB,
+ 0xFF, 0xF7, 0xFF, 0x4E, 0x00, 0xFB, 0x00, 0xFF, 0xFE, 0x59, 0xFB,
+ 0x11, 0x00, 0x83, 0x11, 0x96, 0x20, 0xE0, 0x1B, 0x0B, 0x09, 0x56,
+ 0xFC, 0x99, 0xFC, 0x56, 0x00, 0xCE, 0x00, 0x0D, 0x00, 0xFE, 0xFF,
+ 0xF8, 0xFF, 0x75, 0x00, 0xEA, 0x00, 0x47, 0xFE, 0x3E, 0xFB, 0x69,
+ 0x02, 0x02, 0x15, 0x66, 0x21, 0x04, 0x19, 0xDC, 0x05, 0x98, 0xFB,
+ 0x56, 0xFD, 0xAD, 0x00, 0xA8, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0x00,
+ 0x00, 0xFD, 0xFF, 0x9E, 0x00, 0xBC, 0x00, 0x83, 0xFD, 0x7B, 0xFB,
+ 0x2B, 0x05, 0x4C, 0x18, 0x7C, 0x21, 0xCA, 0x15, 0x03, 0x03, 0x44,
+ 0xFB, 0x1A, 0xFE, 0xE2, 0x00, 0x7E, 0x00, 0xF8, 0xFF, 0xFE, 0xFF,
+ 0x09, 0x00, 0xC6, 0x00, 0x6D, 0x00, 0xC3, 0xFC, 0x20, 0xFC, 0x49,
+ 0x08, 0x41, 0x1B, 0xD6, 0x20, 0x54, 0x12, 0x92, 0x00, 0x4C, 0xFB,
+ 0xD6, 0xFE, 0xFA, 0x00, 0x57, 0x00, 0xF7, 0xFF, 0xFB, 0xFF, 0x1D,
+ 0x00, 0xE6, 0x00, 0xFD, 0xFF, 0x18, 0xFC, 0x38, 0xFD, 0xA9, 0x0B,
+ 0xC0, 0x1D, 0x7C, 0x1F, 0xC6, 0x0E, 0x95, 0xFE, 0x9F, 0xFB, 0x7E,
+ 0xFF, 0xF8, 0x00, 0x35, 0x00, 0xF9, 0xFF, 0xF8, 0xFF, 0x38, 0x00,
+ 0xF9, 0x00, 0x6C, 0xFF, 0x92, 0xFB, 0xC9, 0xFE, 0x2F, 0x0F, 0xAD,
+ 0x1F, 0x7D, 0x1D, 0x42, 0x0B, 0x12, 0xFD, 0x2A, 0xFC, 0x0B, 0x00,
+ 0xE3, 0x00, 0x1A, 0x00, 0xFC, 0xFF, 0xF7, 0xFF, 0x5B, 0x00, 0xF8,
+ 0x00, 0xC1, 0xFE, 0x47, 0xFB, 0xD4, 0x00, 0xBC, 0x12, 0xF3, 0x20,
+ 0xEF, 0x1A, 0xE9, 0x07, 0x08, 0xFC, 0xD9, 0xFC, 0x78, 0x00, 0xC2,
+ 0x00, 0x07, 0x00, 0xFF, 0xFF, 0xF9, 0xFF, 0x83, 0x00, 0xDD, 0x00,
+ 0x04, 0xFE, 0x49, 0xFB, 0x52, 0x03, 0x2D, 0x16, 0x83, 0x21, 0xEF,
+ 0x17, 0xD5, 0x04, 0x6F, 0xFB, 0x9A, 0xFD, 0xC3, 0x00, 0x9A, 0x00,
+ 0xFC, 0xFF, 0x00, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0xAD, 0x00, 0xA4,
+ 0x00, 0x40, 0xFD, 0xA8, 0xFB, 0x36, 0x06, 0x5E, 0x19, 0x58, 0x21,
+ 0x9C, 0x14, 0x1E, 0x02, 0x3D, 0xFB, 0x5D, 0xFE, 0xED, 0x00, 0x70,
+ 0x00, 0xF7, 0xFF, 0xFD, 0xFF, 0x0F, 0x00, 0xD2, 0x00, 0x4A, 0x00,
+ 0x85, 0xFC, 0x74, 0xFC, 0x6D, 0x09, 0x2D, 0x1C, 0x72, 0x20, 0x1A,
+ 0x11, 0xD4, 0xFF, 0x61, 0xFB, 0x13, 0xFF, 0xFC, 0x00, 0x4A, 0x00,
+ 0xF7, 0xFF, 0xFA, 0xFF, 0x25, 0x00, 0xEF, 0x00, 0xCE, 0xFF, 0xE4,
+ 0xFB, 0xB5, 0xFD, 0xDE, 0x0C, 0x7C, 0x1E, 0xDD, 0x1E, 0x8C, 0x0D,
+ 0x01, 0xFE, 0xCA, 0xFB, 0xB3, 0xFF, 0xF3, 0x00, 0x2B, 0x00, 0xFA,
+ 0xFF, 0xF8, 0xFF, 0x44, 0x00, 0xFB, 0x00, 0x34, 0xFF, 0x71, 0xFB,
+ 0x71, 0xFF, 0x6B, 0x10, 0x32, 0x20, 0xA9, 0x1C, 0x13, 0x0A, 0xA8,
+ 0xFC, 0x63, 0xFC, 0x35, 0x00, 0xD9, 0x00, 0x12, 0x00, 0xFD, 0xFF,
+ 0xF7, 0xFF, 0x69, 0x00, 0xF2, 0x00, 0x81, 0xFE, 0x3E, 0xFB, 0xA4,
+ 0x01, 0xF2, 0x13, 0x3A, 0x21, 0xF0, 0x19, 0xCF, 0x06, 0xC7, 0xFB,
+ 0x1B, 0xFD, 0x96, 0x00, 0xB4, 0x00, 0x02, 0x00, 0xFF, 0xFF, 0x00,
+ 0x00, 0xFB, 0xFF, 0x92, 0x00, 0xCD, 0x00, 0xC0, 0xFD, 0x5E, 0xFB,
+ 0x47, 0x04, 0x51, 0x17, 0x8A, 0x21, 0xD0, 0x16, 0xD9, 0x03, 0x53,
+ 0xFB, 0xDE, 0xFD, 0xD5, 0x00, 0x8B, 0x00, 0xFA, 0xFF, 0xFF, 0xFF,
+ 0x04, 0x00, 0xBA, 0x00, 0x89, 0x00, 0xFD, 0xFC, 0xE2, 0xFB, 0x4B,
+ 0x07, 0x63, 0x1A, 0x1D, 0x21, 0x69, 0x13, 0x46, 0x01, 0x41, 0xFB,
+ 0x9E, 0xFE, 0xF5, 0x00, 0x63, 0x00, 0xF7, 0xFF, 0xFC, 0xFF, 0x16,
+ 0x00, 0xDD, 0x00, 0x23, 0x00, 0x49, 0xFC, 0xD5, 0xFC, 0x99, 0x0A,
+ 0x09, 0x1D, 0xF9, 0x1F, 0xDF, 0x0F, 0x24, 0xFF, 0x7F, 0xFB, 0x4D,
+ 0xFF, 0xFB, 0x00, 0x3F, 0x00, 0xF8, 0xFF, 0xF9, 0xFF, 0x2F, 0x00,
+ 0xF5, 0x00, 0x9C, 0xFF, 0xB6, 0xFB, 0x41, 0xFE, 0x17, 0x0E, 0x26,
+ 0x1F, 0x2B, 0x1E, 0x54, 0x0C, 0x7C, 0xFD, 0xFA, 0xFB, 0xE3, 0xFF,
+ 0xEB, 0x00, 0x21, 0x00, 0xFB, 0xFF, 0xF7, 0xFF, 0x50, 0x00, 0xFB,
+ 0x00, 0xF8, 0xFE, 0x57, 0xFB, 0x26, 0x00, 0xA6, 0x11, 0xA1, 0x20,
+ 0xC6, 0x1B, 0xEA, 0x08, 0x4D, 0xFC, 0xA0, 0xFC, 0x5A, 0x00, 0xCD,
+ 0x00, 0x0C, 0x00, 0xFE, 0xFF, 0xF8, 0xFF, 0x77, 0x00, 0xE9, 0x00,
+ 0x3F, 0xFE, 0x3F, 0xFB, 0x82, 0x02, 0x23, 0x15, 0x6B, 0x21, 0xE5,
+ 0x18, 0xBE, 0x05, 0x93, 0xFB, 0x5E, 0xFD, 0xAF, 0x00, 0xA6, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFD, 0xFF, 0xA0, 0x00, 0xB9,
+ 0x00, 0x7C, 0xFD, 0x80, 0xFB, 0x48, 0x05, 0x6B, 0x18, 0x79, 0x21,
+ 0xA9, 0x15, 0xE9, 0x02, 0x43, 0xFB, 0x21, 0xFE, 0xE3, 0x00, 0x7D,
+ 0x00, 0xF8, 0xFF, 0xFE, 0xFF, 0x09, 0x00, 0xC7, 0x00, 0x69, 0x00,
+ 0xBC, 0xFC, 0x29, 0xFC, 0x69, 0x08, 0x5C, 0x1B, 0xCC, 0x20, 0x32,
+ 0x12, 0x7C, 0x00, 0x4E, 0xFB, 0xDD, 0xFE, 0xFA, 0x00, 0x56, 0x00,
+ 0xF7, 0xFF, 0xFB, 0xFF, 0x1D, 0x00, 0xE7, 0x00, 0xF8, 0xFF, 0x12,
+ 0xFC, 0x45, 0xFD, 0xCB, 0x0B, 0xD6, 0x1D, 0x6C, 0x1F, 0xA3, 0x0E,
+ 0x84, 0xFE, 0xA4, 0xFB, 0x84, 0xFF, 0xF7, 0x00, 0x34, 0x00, 0xF9,
+ 0xFF, 0xF8, 0xFF, 0x3A, 0x00, 0xFA, 0x00, 0x66, 0xFF, 0x8E, 0xFB,
+ 0xDB, 0xFE, 0x53, 0x0F, 0xBD, 0x1F, 0x66, 0x1D, 0x21, 0x0B, 0x05,
+ 0xFD, 0x30, 0xFC, 0x10, 0x00, 0xE2, 0x00, 0x19, 0x00, 0xFC, 0xFF,
+ 0xF7, 0xFF, 0x5D, 0x00, 0xF8, 0x00, 0xBA, 0xFE, 0x46, 0xFB, 0xEA,
+ 0x00, 0xDF, 0x12, 0xFC, 0x20, 0xD3, 0x1A, 0xC9, 0x07, 0x00, 0xFC,
+ 0xE0, 0xFC, 0x7B, 0x00, 0xC0, 0x00, 0x07, 0x00, 0xFF, 0xFF, 0xF9,
+ 0xFF, 0x85, 0x00, 0xDC, 0x00, 0xFC, 0xFD, 0x4A, 0xFB, 0x6C, 0x03,
+ 0x4E, 0x16, 0x85, 0x21, 0xCF, 0x17, 0xB8, 0x04, 0x6C, 0xFB, 0xA2,
+ 0xFD, 0xC5, 0x00, 0x98, 0x00, 0xFC, 0xFF, 0x00, 0x00, 0xFF, 0xFF,
+ 0x01, 0x00, 0xAE, 0x00, 0xA1, 0x00, 0x38, 0xFD, 0xAE, 0xFB, 0x54,
+ 0x06, 0x7C, 0x19, 0x53, 0x21, 0x7B, 0x14, 0x05, 0x02, 0x3D, 0xFB,
+ 0x64, 0xFE, 0xEE, 0x00, 0x6F, 0x00, 0xF7, 0xFF, 0xFD, 0xFF, 0x0F,
+ 0x00, 0xD4, 0x00, 0x46, 0x00, 0x7E, 0xFC, 0x7E, 0xFC, 0x8E, 0x09,
+ 0x46, 0x1C, 0x66, 0x20, 0xF7, 0x10, 0xC0, 0xFF, 0x64, 0xFB, 0x1A,
+ 0xFF, 0xFC, 0x00, 0x49, 0x00, 0xF7, 0xFF, 0xFA, 0xFF, 0x26, 0x00,
+ 0xF0, 0x00, 0xC9, 0xFF, 0xDF, 0xFB, 0xC4, 0xFD, 0x01, 0x0D, 0x90,
+ 0x1E, 0xCA, 0x1E, 0x69, 0x0D, 0xF1, 0xFD, 0xCF, 0xFB, 0xB8, 0xFF,
+ 0xF2, 0x00, 0x29, 0x00, 0xFA, 0xFF, 0xF7, 0xFF, 0x45, 0x00, 0xFC,
+ 0x00, 0x2D, 0xFF, 0x6D, 0xFB, 0x84, 0xFF, 0x8E, 0x10, 0x3F, 0x20,
+ 0x91, 0x1C, 0xF2, 0x09, 0x9D, 0xFC, 0x6A, 0xFC, 0x39, 0x00, 0xD7,
+ 0x00, 0x12, 0x00, 0xFD, 0xFF, 0xF7, 0xFF, 0x6A, 0x00, 0xF1, 0x00,
+ 0x7A, 0xFE, 0x3D, 0xFB, 0xBC, 0x01, 0x14, 0x14, 0x41, 0x21, 0xD4,
+ 0x19, 0xB0, 0x06, 0xC0, 0xFB, 0x22, 0xFD, 0x99, 0x00, 0xB3, 0x00,
+ 0x02, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0xFB, 0xFF, 0x93, 0x00, 0xCB,
+ 0x00, 0xB8, 0xFD, 0x61, 0xFB, 0x63, 0x04, 0x71, 0x17, 0x89, 0x21,
+ 0xB0, 0x16, 0xBD, 0x03, 0x51, 0xFB, 0xE6, 0xFD, 0xD7, 0x00, 0x8A,
+ 0x00, 0xFA, 0xFF, 0xFF, 0xFF, 0x05, 0x00, 0xBC, 0x00, 0x86, 0x00,
+ 0xF6, 0xFC, 0xE9, 0xFB, 0x6A, 0x07, 0x80, 0x1A, 0x15, 0x21, 0x47,
+ 0x13, 0x2F, 0x01, 0x42, 0xFB, 0xA5, 0xFE, 0xF6, 0x00, 0x61, 0x00,
+ 0xF7, 0xFF, 0xFC, 0xFF, 0x16, 0x00, 0xDF, 0x00, 0x1E, 0x00, 0x43,
+ 0xFC, 0xE1, 0xFC, 0xBB, 0x0A, 0x21, 0x1D, 0xEA, 0x1F, 0xBC, 0x0F,
+ 0x12, 0xFF, 0x82, 0xFB, 0x54, 0xFF, 0xFA, 0x00, 0x3D, 0x00, 0xF8,
+ 0xFF, 0xF9, 0xFF, 0x30, 0x00, 0xF6, 0x00, 0x96, 0xFF, 0xB1, 0xFB,
+ 0x51, 0xFE, 0x3A, 0x0E, 0x38, 0x1F, 0x16, 0x1E, 0x32, 0x0C, 0x6E,
+ 0xFD, 0x00, 0xFC, 0xE8, 0xFF, 0xEA, 0x00, 0x20, 0x00, 0xFB, 0xFF,
+ 0xF7, 0xFF, 0x51, 0x00, 0xFB, 0x00, 0xF1, 0xFE, 0x54, 0xFB, 0x3B,
+ 0x00, 0xC9, 0x11, 0xAD, 0x20, 0xAC, 0x1B, 0xCA, 0x08, 0x44, 0xFC,
+ 0xA7, 0xFC, 0x5E, 0x00, 0xCC, 0x00, 0x0B, 0x00, 0xFE, 0xFF, 0xF8,
+ 0xFF, 0x78, 0x00, 0xE7, 0x00, 0x38, 0xFE, 0x40, 0xFB, 0x9B, 0x02,
+ 0x45, 0x15, 0x6F, 0x21, 0xC7, 0x18, 0xA1, 0x05, 0x8E, 0xFB, 0x65,
+ 0xFD, 0xB2, 0x00, 0xA5, 0x00, 0xFE, 0xFF, 0x00, 0x00, 0x00, 0x00,
+ 0xFE, 0xFF, 0xA2, 0x00, 0xB7, 0x00, 0x74, 0xFD, 0x84, 0xFB, 0x66,
+ 0x05, 0x8A, 0x18, 0x76, 0x21, 0x87, 0x15, 0xCF, 0x02, 0x41, 0xFB,
+ 0x29, 0xFE, 0xE5, 0x00, 0x7B, 0x00, 0xF8, 0xFF, 0xFE, 0xFF, 0x0A,
+ 0x00, 0xC9, 0x00, 0x66, 0x00, 0xB5, 0xFC, 0x32, 0xFC, 0x89, 0x08,
+ 0x77, 0x1B, 0xC2, 0x20, 0x0F, 0x12, 0x66, 0x00, 0x50, 0xFB, 0xE4,
+ 0xFE, 0xFA, 0x00, 0x54, 0x00, 0xF7, 0xFF, 0xFB, 0xFF, 0x1E, 0x00,
+ 0xE8, 0x00, 0xF3, 0xFF, 0x0C, 0xFC, 0x53, 0xFD, 0xED, 0x0B, 0xEB,
+ 0x1D, 0x5A, 0x1F, 0x80, 0x0E, 0x73, 0xFE, 0xA8, 0xFB, 0x8A, 0xFF,
+ 0xF7, 0x00, 0x32, 0x00, 0xF9, 0xFF, 0xF8, 0xFF, 0x3B, 0x00, 0xFA,
+ 0x00, 0x60, 0xFF, 0x8A, 0xFB, 0xED, 0xFE, 0x76, 0x0F, 0xCC, 0x1F,
+ 0x4F, 0x1D, 0xFF, 0x0A, 0xF9, 0xFC, 0x36, 0xFC, 0x15, 0x00, 0xE1,
+ 0x00, 0x18, 0x00, 0xFC, 0xFF, 0xF7, 0xFF, 0x5E, 0x00, 0xF7, 0x00,
+ 0xB3, 0xFE, 0x44, 0xFB, 0x01, 0x01, 0x02, 0x13, 0x04, 0x21, 0xB8,
+ 0x1A, 0xA9, 0x07, 0xF8, 0xFB, 0xE7, 0xFC, 0x7F, 0x00, 0xBF, 0x00,
+ 0x06, 0x00, 0xFF, 0xFF, 0xF9, 0xFF, 0x86, 0x00, 0xDA, 0x00, 0xF5,
+ 0xFD, 0x4C, 0xFB, 0x87, 0x03, 0x6E, 0x16, 0x86, 0x21, 0xB0, 0x17,
+ 0x9C, 0x04, 0x68, 0xFB, 0xA9, 0xFD, 0xC7, 0x00, 0x96, 0x00, 0xFB,
+ 0xFF, 0x00, 0x00, 0xFF, 0xFF, 0x01, 0x00, 0xB0, 0x00, 0x9F, 0x00,
+ 0x31, 0xFD, 0xB4, 0xFB, 0x73, 0x06, 0x99, 0x19, 0x4D, 0x21, 0x59,
+ 0x14, 0xED, 0x01, 0x3D, 0xFB, 0x6B, 0xFE, 0xEF, 0x00, 0x6D, 0x00,
+ 0xF7, 0xFF, 0xFD, 0xFF, 0x10, 0x00, 0xD5, 0x00, 0x42, 0x00, 0x77,
+ 0xFC, 0x88, 0xFC, 0xAF, 0x09, 0x5F, 0x1C, 0x59, 0x20, 0xD4, 0x10,
+ 0xAC, 0xFF, 0x67, 0xFB, 0x20, 0xFF, 0xFC, 0x00, 0x48, 0x00, 0xF7,
+ 0xFF, 0xFA, 0xFF, 0x27, 0x00, 0xF0, 0x00, 0xC3, 0xFF, 0xD9, 0xFB,
+ 0xD3, 0xFD, 0x24, 0x0D, 0xA3, 0x1E, 0xB7, 0x1E, 0x46, 0x0D, 0xE2,
+ 0xFD, 0xD4, 0xFB, 0xBE, 0xFF, 0xF1, 0x00, 0x28, 0x00, 0xFA, 0xFF,
+ 0xF7, 0xFF, 0x46, 0x00, 0xFC, 0x00, 0x27, 0xFF, 0x6A, 0xFB, 0x98,
+ 0xFF, 0xB1, 0x10, 0x4C, 0x20, 0x78, 0x1C, 0xD1, 0x09, 0x93, 0xFC,
+ 0x71, 0xFC, 0x3D, 0x00, 0xD6, 0x00, 0x11, 0x00, 0xFD, 0xFF, 0xF7,
+ 0xFF, 0x6C, 0x00, 0xF0, 0x00, 0x72, 0xFE, 0x3D, 0xFB, 0xD4, 0x01,
+ 0x36, 0x14, 0x47, 0x21, 0xB6, 0x19, 0x91, 0x06, 0xBA, 0xFB, 0x29,
+ 0xFD, 0x9C, 0x00, 0xB1, 0x00, 0x02, 0x00, 0xFF, 0xFF, 0x00, 0x00,
+ 0xFB, 0xFF, 0x95, 0x00, 0xC9, 0x00, 0xB1, 0xFD, 0x65, 0xFB, 0x80,
+ 0x04, 0x90, 0x17, 0x88, 0x21, 0x8F, 0x16, 0xA2, 0x03, 0x4E, 0xFB,
+ 0xED, 0xFD, 0xD9, 0x00, 0x88, 0x00, 0xF9, 0xFF, 0xFF, 0xFF, 0x05,
+ 0x00, 0xBD, 0x00, 0x82, 0x00, 0xEF, 0xFC, 0xF0, 0xFB, 0x8A, 0x07,
+ 0x9C, 0x1A, 0x0D, 0x21, 0x24, 0x13, 0x18, 0x01, 0x43, 0xFB, 0xAC,
+ 0xFE, 0xF7, 0x00, 0x60, 0x00, 0xF7, 0xFF, 0xFC, 0xFF, 0x17, 0x00,
+ 0xE0, 0x00, 0x1A, 0x00, 0x3D, 0xFC, 0xED, 0xFC, 0xDD, 0x0A, 0x38,
+ 0x1D, 0xDB, 0x1F, 0x99, 0x0F, 0xFF, 0xFE, 0x86, 0xFB, 0x5A, 0xFF,
+ 0xFA, 0x00, 0x3C, 0x00, 0xF8, 0xFF, 0xF9, 0xFF, 0x31, 0x00, 0xF6,
+ 0x00, 0x90, 0xFF, 0xAD, 0xFB, 0x62, 0xFE, 0x5D, 0x0E, 0x49, 0x1F,
+ 0x01, 0x1E, 0x10, 0x0C, 0x60, 0xFD, 0x06, 0xFC, 0xEE, 0xFF, 0xE9,
+ 0x00, 0x1F, 0x00, 0xFB, 0xFF, 0xF7, 0xFF, 0x53, 0x00, 0xFB, 0x00,
+ 0xEB, 0xFE, 0x52, 0xFB, 0x51, 0x00, 0xEC, 0x11, 0xB7, 0x20, 0x91,
+ 0x1B, 0xA9, 0x08, 0x3B, 0xFC, 0xAE, 0xFC, 0x62, 0x00, 0xCA, 0x00,
+ 0x0B, 0x00, 0xFE, 0xFF, 0xF8, 0xFF, 0x7A, 0x00, 0xE6, 0x00, 0x30,
+ 0xFE, 0x40, 0xFB, 0xB5, 0x02, 0x66, 0x15, 0x73, 0x21, 0xA9, 0x18,
+ 0x83, 0x05, 0x89, 0xFB, 0x6D, 0xFD, 0xB4, 0x00, 0xA3, 0x00, 0xFE,
+ 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFE, 0xFF, 0xA3, 0x00, 0xB4, 0x00,
+ 0x6D, 0xFD, 0x89, 0xFB, 0x83, 0x05, 0xA9, 0x18, 0x73, 0x21, 0x66,
+ 0x15, 0xB5, 0x02, 0x40, 0xFB, 0x30, 0xFE, 0xE6, 0x00, 0x7A, 0x00,
+ 0xF8, 0xFF, 0xFE, 0xFF, 0x0B, 0x00, 0xCA, 0x00, 0x62, 0x00, 0xAE,
+ 0xFC, 0x3B, 0xFC, 0xA9, 0x08, 0x91, 0x1B, 0xB7, 0x20, 0xEC, 0x11,
+ 0x51, 0x00, 0x52, 0xFB, 0xEB, 0xFE, 0xFB, 0x00, 0x53, 0x00, 0xF7,
+ 0xFF, 0xFB, 0xFF, 0x1F, 0x00, 0xE9, 0x00, 0xEE, 0xFF, 0x06, 0xFC,
+ 0x60, 0xFD, 0x10, 0x0C, 0x01, 0x1E, 0x49, 0x1F, 0x5D, 0x0E, 0x62,
+ 0xFE, 0xAD, 0xFB, 0x90, 0xFF, 0xF6, 0x00, 0x31, 0x00, 0xF9, 0xFF,
+ 0xF8, 0xFF, 0x3C, 0x00, 0xFA, 0x00, 0x5A, 0xFF, 0x86, 0xFB, 0xFF,
+ 0xFE, 0x99, 0x0F, 0xDB, 0x1F, 0x38, 0x1D, 0xDD, 0x0A, 0xED, 0xFC,
+ 0x3D, 0xFC, 0x1A, 0x00, 0xE0, 0x00, 0x17, 0x00, 0xFC, 0xFF, 0xF7,
+ 0xFF, 0x60, 0x00, 0xF7, 0x00, 0xAC, 0xFE, 0x43, 0xFB, 0x18, 0x01,
+ 0x24, 0x13, 0x0D, 0x21, 0x9C, 0x1A, 0x8A, 0x07, 0xF0, 0xFB, 0xEF,
+ 0xFC, 0x82, 0x00, 0xBD, 0x00, 0x05, 0x00, 0xFF, 0xFF, 0xF9, 0xFF,
+ 0x88, 0x00, 0xD9, 0x00, 0xED, 0xFD, 0x4E, 0xFB, 0xA2, 0x03, 0x8F,
+ 0x16, 0x88, 0x21, 0x90, 0x17, 0x80, 0x04, 0x65, 0xFB, 0xB1, 0xFD,
+ 0xC9, 0x00, 0x95, 0x00, 0xFB, 0xFF, 0x00, 0x00, 0xFF, 0xFF, 0x02,
+ 0x00, 0xB1, 0x00, 0x9C, 0x00, 0x29, 0xFD, 0xBA, 0xFB, 0x91, 0x06,
+ 0xB6, 0x19, 0x47, 0x21, 0x36, 0x14, 0xD4, 0x01, 0x3D, 0xFB, 0x72,
+ 0xFE, 0xF0, 0x00, 0x6C, 0x00, 0xF7, 0xFF, 0xFD, 0xFF, 0x11, 0x00,
+ 0xD6, 0x00, 0x3D, 0x00, 0x71, 0xFC, 0x93, 0xFC, 0xD1, 0x09, 0x78,
+ 0x1C, 0x4C, 0x20, 0xB1, 0x10, 0x98, 0xFF, 0x6A, 0xFB, 0x27, 0xFF,
+ 0xFC, 0x00, 0x46, 0x00, 0xF7, 0xFF, 0xFA, 0xFF, 0x28, 0x00, 0xF1,
+ 0x00, 0xBE, 0xFF, 0xD4, 0xFB, 0xE2, 0xFD, 0x46, 0x0D, 0xB7, 0x1E,
+ 0xA3, 0x1E, 0x24, 0x0D, 0xD3, 0xFD, 0xD9, 0xFB, 0xC3, 0xFF, 0xF0,
+ 0x00, 0x27, 0x00, 0xFA, 0xFF, 0xF7, 0xFF, 0x48, 0x00, 0xFC, 0x00,
+ 0x20, 0xFF, 0x67, 0xFB, 0xAC, 0xFF, 0xD4, 0x10, 0x59, 0x20, 0x5F,
+ 0x1C, 0xAF, 0x09, 0x88, 0xFC, 0x77, 0xFC, 0x42, 0x00, 0xD5, 0x00,
+ 0x10, 0x00, 0xFD, 0xFF, 0xF7, 0xFF, 0x6D, 0x00, 0xEF, 0x00, 0x6B,
+ 0xFE, 0x3D, 0xFB, 0xED, 0x01, 0x59, 0x14, 0x4D, 0x21, 0x99, 0x19,
+ 0x73, 0x06, 0xB4, 0xFB, 0x31, 0xFD, 0x9F, 0x00, 0xB0, 0x00, 0x01,
+ 0x00, 0xFF, 0xFF, 0x00, 0x00, 0xFB, 0xFF, 0x96, 0x00, 0xC7, 0x00,
+ 0xA9, 0xFD, 0x68, 0xFB, 0x9C, 0x04, 0xB0, 0x17, 0x86, 0x21, 0x6E,
+ 0x16, 0x87, 0x03, 0x4C, 0xFB, 0xF5, 0xFD, 0xDA, 0x00, 0x86, 0x00,
+ 0xF9, 0xFF, 0xFF, 0xFF, 0x06, 0x00, 0xBF, 0x00, 0x7F, 0x00, 0xE7,
+ 0xFC, 0xF8, 0xFB, 0xA9, 0x07, 0xB8, 0x1A, 0x04, 0x21, 0x02, 0x13,
+ 0x01, 0x01, 0x44, 0xFB, 0xB3, 0xFE, 0xF7, 0x00, 0x5E, 0x00, 0xF7,
+ 0xFF, 0xFC, 0xFF, 0x18, 0x00, 0xE1, 0x00, 0x15, 0x00, 0x36, 0xFC,
+ 0xF9, 0xFC, 0xFF, 0x0A, 0x4F, 0x1D, 0xCC, 0x1F, 0x76, 0x0F, 0xED,
+ 0xFE, 0x8A, 0xFB, 0x60, 0xFF, 0xFA, 0x00, 0x3B, 0x00, 0xF8, 0xFF,
+ 0xF9, 0xFF, 0x32, 0x00, 0xF7, 0x00, 0x8A, 0xFF, 0xA8, 0xFB, 0x73,
+ 0xFE, 0x80, 0x0E, 0x5A, 0x1F, 0xEB, 0x1D, 0xED, 0x0B, 0x53, 0xFD,
+ 0x0C, 0xFC, 0xF3, 0xFF, 0xE8, 0x00, 0x1E, 0x00, 0xFB, 0xFF, 0xF7,
+ 0xFF, 0x54, 0x00, 0xFA, 0x00, 0xE4, 0xFE, 0x50, 0xFB, 0x66, 0x00,
+ 0x0F, 0x12, 0xC2, 0x20, 0x77, 0x1B, 0x89, 0x08, 0x32, 0xFC, 0xB5,
+ 0xFC, 0x66, 0x00, 0xC9, 0x00, 0x0A, 0x00, 0xFE, 0xFF, 0xF8, 0xFF,
+ 0x7B, 0x00, 0xE5, 0x00, 0x29, 0xFE, 0x41, 0xFB, 0xCF, 0x02, 0x87,
+ 0x15, 0x76, 0x21, 0x8A, 0x18, 0x66, 0x05, 0x84, 0xFB, 0x74, 0xFD,
+ 0xB7, 0x00, 0xA2, 0x00, 0xFE, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFE,
+ 0xFF, 0xA5, 0x00, 0xB2, 0x00, 0x65, 0xFD, 0x8E, 0xFB, 0xA1, 0x05,
+ 0xC7, 0x18, 0x6F, 0x21, 0x45, 0x15, 0x9B, 0x02, 0x40, 0xFB, 0x38,
+ 0xFE, 0xE7, 0x00, 0x78, 0x00, 0xF8, 0xFF, 0xFE, 0xFF, 0x0B, 0x00,
+ 0xCC, 0x00, 0x5E, 0x00, 0xA7, 0xFC, 0x44, 0xFC, 0xCA, 0x08, 0xAC,
+ 0x1B, 0xAD, 0x20, 0xC9, 0x11, 0x3B, 0x00, 0x54, 0xFB, 0xF1, 0xFE,
+ 0xFB, 0x00, 0x51, 0x00, 0xF7, 0xFF, 0xFB, 0xFF, 0x20, 0x00, 0xEA,
+ 0x00, 0xE8, 0xFF, 0x00, 0xFC, 0x6E, 0xFD, 0x32, 0x0C, 0x16, 0x1E,
+ 0x38, 0x1F, 0x3A, 0x0E, 0x51, 0xFE, 0xB1, 0xFB, 0x96, 0xFF, 0xF6,
+ 0x00, 0x30, 0x00, 0xF9, 0xFF, 0xF8, 0xFF, 0x3D, 0x00, 0xFA, 0x00,
+ 0x54, 0xFF, 0x82, 0xFB, 0x12, 0xFF, 0xBC, 0x0F, 0xEA, 0x1F, 0x21,
+ 0x1D, 0xBB, 0x0A, 0xE1, 0xFC, 0x43, 0xFC, 0x1E, 0x00, 0xDF, 0x00,
+ 0x16, 0x00, 0xFC, 0xFF, 0xF7, 0xFF, 0x61, 0x00, 0xF6, 0x00, 0xA5,
+ 0xFE, 0x42, 0xFB, 0x2F, 0x01, 0x47, 0x13, 0x15, 0x21, 0x80, 0x1A,
+ 0x6A, 0x07, 0xE9, 0xFB, 0xF6, 0xFC, 0x86, 0x00, 0xBC, 0x00, 0x05,
+ 0x00, 0xFF, 0xFF, 0xFA, 0xFF, 0x8A, 0x00, 0xD7, 0x00, 0xE6, 0xFD,
+ 0x51, 0xFB, 0xBD, 0x03, 0xB0, 0x16, 0x89, 0x21, 0x71, 0x17, 0x63,
+ 0x04, 0x61, 0xFB, 0xB8, 0xFD, 0xCB, 0x00, 0x93, 0x00, 0xFB, 0xFF,
+ 0x00, 0x00, 0xFF, 0xFF, 0x02, 0x00, 0xB3, 0x00, 0x99, 0x00, 0x22,
+ 0xFD, 0xC0, 0xFB, 0xB0, 0x06, 0xD4, 0x19, 0x41, 0x21, 0x14, 0x14,
+ 0xBC, 0x01, 0x3D, 0xFB, 0x7A, 0xFE, 0xF1, 0x00, 0x6A, 0x00, 0xF7,
+ 0xFF, 0xFD, 0xFF, 0x12, 0x00, 0xD7, 0x00, 0x39, 0x00, 0x6A, 0xFC,
+ 0x9D, 0xFC, 0xF2, 0x09, 0x91, 0x1C, 0x3F, 0x20, 0x8E, 0x10, 0x84,
+ 0xFF, 0x6D, 0xFB, 0x2D, 0xFF, 0xFC, 0x00, 0x45, 0x00, 0xF7, 0xFF,
+ 0xFA, 0xFF, 0x29, 0x00, 0xF2, 0x00, 0xB8, 0xFF, 0xCF, 0xFB, 0xF1,
+ 0xFD, 0x69, 0x0D, 0xCA, 0x1E, 0x90, 0x1E, 0x01, 0x0D, 0xC4, 0xFD,
+ 0xDF, 0xFB, 0xC9, 0xFF, 0xF0, 0x00, 0x26, 0x00, 0xFA, 0xFF, 0xF7,
+ 0xFF, 0x49, 0x00, 0xFC, 0x00, 0x1A, 0xFF, 0x64, 0xFB, 0xC0, 0xFF,
+ 0xF7, 0x10, 0x66, 0x20, 0x46, 0x1C, 0x8E, 0x09, 0x7E, 0xFC, 0x7E,
+ 0xFC, 0x46, 0x00, 0xD4, 0x00, 0x0F, 0x00, 0xFD, 0xFF, 0xF7, 0xFF,
+ 0x6F, 0x00, 0xEE, 0x00, 0x64, 0xFE, 0x3D, 0xFB, 0x05, 0x02, 0x7B,
+ 0x14, 0x53, 0x21, 0x7C, 0x19, 0x54, 0x06, 0xAE, 0xFB, 0x38, 0xFD,
+ 0xA1, 0x00, 0xAE, 0x00, 0x01, 0x00, 0xFF, 0xFF, 0x00, 0x00, 0xFC,
+ 0xFF, 0x98, 0x00, 0xC5, 0x00, 0xA2, 0xFD, 0x6C, 0xFB, 0xB8, 0x04,
+ 0xCF, 0x17, 0x85, 0x21, 0x4E, 0x16, 0x6C, 0x03, 0x4A, 0xFB, 0xFC,
+ 0xFD, 0xDC, 0x00, 0x85, 0x00, 0xF9, 0xFF, 0xFF, 0xFF, 0x07, 0x00,
+ 0xC0, 0x00, 0x7B, 0x00, 0xE0, 0xFC, 0x00, 0xFC, 0xC9, 0x07, 0xD3,
+ 0x1A, 0xFC, 0x20, 0xDF, 0x12, 0xEA, 0x00, 0x46, 0xFB, 0xBA, 0xFE,
+ 0xF8, 0x00, 0x5D, 0x00, 0xF7, 0xFF, 0xFC, 0xFF, 0x19, 0x00, 0xE2,
+ 0x00, 0x10, 0x00, 0x30, 0xFC, 0x05, 0xFD, 0x21, 0x0B, 0x66, 0x1D,
+ 0xBD, 0x1F, 0x53, 0x0F, 0xDB, 0xFE, 0x8E, 0xFB, 0x66, 0xFF, 0xFA,
+ 0x00, 0x3A, 0x00, 0xF8, 0xFF, 0xF9, 0xFF, 0x34, 0x00, 0xF7, 0x00,
+ 0x84, 0xFF, 0xA4, 0xFB, 0x84, 0xFE, 0xA3, 0x0E, 0x6C, 0x1F, 0xD6,
+ 0x1D, 0xCB, 0x0B, 0x45, 0xFD, 0x12, 0xFC, 0xF8, 0xFF, 0xE7, 0x00,
+ 0x1D, 0x00, 0xFB, 0xFF, 0xF7, 0xFF, 0x56, 0x00, 0xFA, 0x00, 0xDD,
+ 0xFE, 0x4E, 0xFB, 0x7C, 0x00, 0x32, 0x12, 0xCC, 0x20, 0x5C, 0x1B,
+ 0x69, 0x08, 0x29, 0xFC, 0xBC, 0xFC, 0x69, 0x00, 0xC7, 0x00, 0x09,
+ 0x00, 0xFE, 0xFF, 0xF8, 0xFF, 0x7D, 0x00, 0xE3, 0x00, 0x21, 0xFE,
+ 0x43, 0xFB, 0xE9, 0x02, 0xA9, 0x15, 0x79, 0x21, 0x6B, 0x18, 0x48,
+ 0x05, 0x80, 0xFB, 0x7C, 0xFD, 0xB9, 0x00, 0xA0, 0x00, 0xFD, 0xFF,
+ 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0xA6, 0x00, 0xAF, 0x00, 0x5E,
+ 0xFD, 0x93, 0xFB, 0xBE, 0x05, 0xE5, 0x18, 0x6B, 0x21, 0x23, 0x15,
+ 0x82, 0x02, 0x3F, 0xFB, 0x3F, 0xFE, 0xE9, 0x00, 0x77, 0x00, 0xF8,
+ 0xFF, 0xFE, 0xFF, 0x0C, 0x00, 0xCD, 0x00, 0x5A, 0x00, 0xA0, 0xFC,
+ 0x4D, 0xFC, 0xEA, 0x08, 0xC6, 0x1B, 0xA1, 0x20, 0xA6, 0x11, 0x26,
+ 0x00, 0x57, 0xFB, 0xF8, 0xFE, 0xFB, 0x00, 0x50, 0x00, 0xF7, 0xFF,
+ 0xFB, 0xFF, 0x21, 0x00, 0xEB, 0x00, 0xE3, 0xFF, 0xFA, 0xFB, 0x7C,
+ 0xFD, 0x54, 0x0C, 0x2B, 0x1E, 0x26, 0x1F, 0x17, 0x0E, 0x41, 0xFE,
+ 0xB6, 0xFB, 0x9C, 0xFF, 0xF5, 0x00, 0x2F, 0x00, 0xF9, 0xFF, 0xF8,
+ 0xFF, 0x3F, 0x00, 0xFB, 0x00, 0x4D, 0xFF, 0x7F, 0xFB, 0x24, 0xFF,
+ 0xDF, 0x0F, 0xF9, 0x1F, 0x09, 0x1D, 0x99, 0x0A, 0xD5, 0xFC, 0x49,
+ 0xFC, 0x23, 0x00, 0xDD, 0x00, 0x16, 0x00, 0xFC, 0xFF, 0xF7, 0xFF,
+ 0x63, 0x00, 0xF5, 0x00, 0x9E, 0xFE, 0x41, 0xFB, 0x46, 0x01, 0x69,
+ 0x13, 0x1D, 0x21, 0x63, 0x1A, 0x4B, 0x07, 0xE2, 0xFB, 0xFD, 0xFC,
+ 0x89, 0x00, 0xBA, 0x00, 0x04, 0x00, 0xFF, 0xFF, 0xFA, 0xFF, 0x8B,
+ 0x00, 0xD5, 0x00, 0xDE, 0xFD, 0x53, 0xFB, 0xD9, 0x03, 0xD0, 0x16,
+ 0x8A, 0x21, 0x51, 0x17, 0x47, 0x04, 0x5E, 0xFB, 0xC0, 0xFD, 0xCD,
+ 0x00, 0x92, 0x00, 0xFB, 0xFF, 0x00, 0x00, 0xFF, 0xFF, 0x02, 0x00,
+ 0xB4, 0x00, 0x96, 0x00, 0x1B, 0xFD, 0xC7, 0xFB, 0xCF, 0x06, 0xF0,
+ 0x19, 0x3A, 0x21, 0xF2, 0x13, 0xA4, 0x01, 0x3E, 0xFB, 0x81, 0xFE,
+ 0xF2, 0x00, 0x69, 0x00, 0xF7, 0xFF, 0xFD, 0xFF, 0x12, 0x00, 0xD9,
+ 0x00, 0x35, 0x00, 0x63, 0xFC, 0xA8, 0xFC, 0x13, 0x0A, 0xA9, 0x1C,
+ 0x32, 0x20, 0x6B, 0x10, 0x71, 0xFF, 0x71, 0xFB, 0x34, 0xFF, 0xFB,
+ 0x00, 0x44, 0x00, 0xF8, 0xFF, 0xFA, 0xFF, 0x2B, 0x00, 0xF3, 0x00,
+ 0xB3, 0xFF, 0xCA, 0xFB, 0x01, 0xFE, 0x8C, 0x0D, 0xDD, 0x1E, 0x7C,
+ 0x1E, 0xDE, 0x0C, 0xB5, 0xFD, 0xE4, 0xFB, 0xCE, 0xFF, 0xEF, 0x00,
+ 0x25, 0x00, 0xFA, 0xFF, 0xF7, 0xFF, 0x4A, 0x00, 0xFC, 0x00, 0x13,
+ 0xFF, 0x61, 0xFB, 0xD4, 0xFF, 0x1A, 0x11, 0x72, 0x20, 0x2D, 0x1C,
+ 0x6D, 0x09, 0x74, 0xFC, 0x85, 0xFC, 0x4A, 0x00, 0xD2, 0x00, 0x0F,
+ 0x00, 0xFD, 0xFF, 0xF7, 0xFF, 0x70, 0x00, 0xED, 0x00, 0x5D, 0xFE,
+ 0x3D, 0xFB, 0x1E, 0x02, 0x9C, 0x14, 0x58, 0x21, 0x5E, 0x19, 0x36,
+ 0x06, 0xA8, 0xFB, 0x40, 0xFD, 0xA4, 0x00, 0xAD, 0x00, 0x00, 0x00,
+ 0xFF, 0xFF, 0x00, 0x00, 0xFC, 0xFF, 0x9A, 0x00, 0xC3, 0x00, 0x9A,
+ 0xFD, 0x6F, 0xFB, 0xD5, 0x04, 0xEF, 0x17, 0x83, 0x21, 0x2D, 0x16,
+ 0x52, 0x03, 0x49, 0xFB, 0x04, 0xFE, 0xDD, 0x00, 0x83, 0x00, 0xF9,
+ 0xFF, 0xFF, 0xFF, 0x07, 0x00, 0xC2, 0x00, 0x78, 0x00, 0xD9, 0xFC,
+ 0x08, 0xFC, 0xE9, 0x07, 0xEF, 0x1A, 0xF3, 0x20, 0xBC, 0x12, 0xD4,
+ 0x00, 0x47, 0xFB, 0xC1, 0xFE, 0xF8, 0x00, 0x5B, 0x00, 0xF7, 0xFF,
+ 0xFC, 0xFF, 0x1A, 0x00, 0xE3, 0x00, 0x0B, 0x00, 0x2A, 0xFC, 0x12,
+ 0xFD, 0x42, 0x0B, 0x7D, 0x1D, 0xAD, 0x1F, 0x2F, 0x0F, 0xC9, 0xFE,
+ 0x92, 0xFB, 0x6C, 0xFF, 0xF9, 0x00, 0x38, 0x00, 0xF8, 0xFF, 0xF9,
+ 0xFF, 0x35, 0x00, 0xF8, 0x00, 0x7E, 0xFF, 0x9F, 0xFB, 0x95, 0xFE,
+ 0xC6, 0x0E, 0x7C, 0x1F, 0xC0, 0x1D, 0xA9, 0x0B, 0x38, 0xFD, 0x18,
+ 0xFC, 0xFD, 0xFF, 0xE6, 0x00, 0x1D, 0x00, 0xFB, 0xFF, 0xF7, 0xFF,
+ 0x57, 0x00, 0xFA, 0x00, 0xD6, 0xFE, 0x4C, 0xFB, 0x92, 0x00, 0x54,
+ 0x12, 0xD6, 0x20, 0x41, 0x1B, 0x49, 0x08, 0x20, 0xFC, 0xC3, 0xFC,
+ 0x6D, 0x00, 0xC6, 0x00, 0x09, 0x00, 0xFE, 0xFF, 0xF8, 0xFF, 0x7E,
+ 0x00, 0xE2, 0x00, 0x1A, 0xFE, 0x44, 0xFB, 0x03, 0x03, 0xCA, 0x15,
+ 0x7C, 0x21, 0x4C, 0x18, 0x2B, 0x05, 0x7B, 0xFB, 0x83, 0xFD, 0xBC,
+ 0x00, 0x9E, 0x00, 0xFD, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF,
+ 0xA8, 0x00, 0xAD, 0x00, 0x56, 0xFD, 0x98, 0xFB, 0xDC, 0x05, 0x04,
+ 0x19, 0x66, 0x21, 0x02, 0x15, 0x69, 0x02, 0x3E, 0xFB, 0x47, 0xFE,
+ 0xEA, 0x00, 0x75, 0x00, 0xF8, 0xFF, 0xFE, 0xFF, 0x0D, 0x00, 0xCE,
+ 0x00, 0x56, 0x00, 0x99, 0xFC, 0x56, 0xFC, 0x0B, 0x09, 0xE0, 0x1B,
+ 0x96, 0x20, 0x83, 0x11, 0x11, 0x00, 0x59, 0xFB, 0xFF, 0xFE, 0xFB,
+ 0x00, 0x4E, 0x00, 0xF7, 0xFF, 0xFB, 0xFF, 0x22, 0x00, 0xEC, 0x00,
+ 0xDE, 0xFF, 0xF5, 0xFB, 0x8A, 0xFD, 0x77, 0x0C, 0x3F, 0x1E, 0x14,
+ 0x1F, 0xF5, 0x0D, 0x31, 0xFE, 0xBB, 0xFB, 0xA2, 0xFF, 0xF5, 0x00,
+ 0x2E, 0x00, 0xF9, 0xFF, 0xF8, 0xFF, 0x40, 0x00, 0xFB, 0x00, 0x47,
+ 0xFF, 0x7B, 0xFB, 0x37, 0xFF, 0x02, 0x10, 0x07, 0x20, 0xF2, 0x1C,
+ 0x78, 0x0A, 0xCA, 0xFC, 0x50, 0xFC, 0x27, 0x00, 0xDC, 0x00, 0x15,
+ 0x00, 0xFC, 0xFF, 0xF7, 0xFF, 0x64, 0x00, 0xF5, 0x00, 0x97, 0xFE,
+ 0x40, 0xFB, 0x5D, 0x01, 0x8B, 0x13, 0x25, 0x21, 0x47, 0x1A, 0x2C,
+ 0x07, 0xDB, 0xFB, 0x05, 0xFD, 0x8C, 0x00, 0xB9, 0x00, 0x04, 0x00,
+ 0xFF, 0xFF, 0xFA, 0xFF, 0x8D, 0x00, 0xD3, 0x00, 0xD6, 0xFD, 0x56,
+ 0xFB, 0xF4, 0x03, 0xF0, 0x16, 0x8A, 0x21, 0x31, 0x17, 0x2B, 0x04,
+ 0x5B, 0xFB, 0xC7, 0xFD, 0xCF, 0x00, 0x90, 0x00, 0xFA, 0xFF, 0x00,
+ 0x00, 0xFF, 0xFF, 0x03, 0x00, 0xB6, 0x00, 0x92, 0x00, 0x13, 0xFD,
+ 0xCD, 0xFB, 0xEE, 0x06, 0x0D, 0x1A, 0x33, 0x21, 0xD0, 0x13, 0x8C,
+ 0x01, 0x3E, 0xFB, 0x88, 0xFE, 0xF3, 0x00, 0x67, 0x00, 0xF7, 0xFF,
+ 0x06, 0x00, 0x1D, 0x00, 0x03, 0xFF, 0xFE, 0x00, 0xA1, 0x02, 0xA6,
+ 0xF8, 0x56, 0x02, 0xA5, 0x28, 0xA5, 0x28, 0x56, 0x02, 0xA6, 0xF8,
+ 0xA1, 0x02, 0xFE, 0x00, 0x03, 0xFF, 0x1D, 0x00, 0x06, 0x00, 0x00,
+ 0x00, 0x21, 0x00, 0xA6, 0xFF, 0x3F, 0xFF, 0x0B, 0x03, 0x42, 0xFE,
+ 0x3E, 0xF8, 0x7F, 0x15, 0xAC, 0x30, 0x7F, 0x15, 0x3E, 0xF8, 0x42,
+ 0xFE, 0x0B, 0x03, 0x3F, 0xFF, 0xA6, 0xFF, 0x21, 0x00, 0x00, 0x00,
+ 0xFA, 0xFF, 0xCE, 0xFF, 0x14, 0x01, 0x00, 0xFD, 0x35, 0x06, 0xD5,
+ 0xF4, 0xDA, 0x15, 0x92, 0x40, 0xAE, 0xFE, 0xF3, 0xFC, 0x68, 0x03,
+ 0x86, 0xFD, 0x51, 0x01, 0x8B, 0xFF, 0x11, 0x00, 0x01, 0x00, 0xEC,
+ 0xFF, 0xF9, 0xFF, 0xC6, 0x00, 0x55, 0xFD, 0x35, 0x06, 0x90, 0xF3,
+ 0xE5, 0x1C, 0x6B, 0x3D, 0x71, 0xFA, 0x34, 0xFF, 0x46, 0x02, 0xFF,
+ 0xFD, 0x2D, 0x01, 0x90, 0xFF, 0x10, 0x00, 0x03, 0x00, 0xDB, 0xFF,
+ 0x2D, 0x00, 0x60, 0x00, 0xE1, 0xFD, 0xCE, 0x05, 0xED, 0xF2, 0xF3,
+ 0x23, 0x20, 0x39, 0x22, 0xF7, 0x44, 0x01, 0x1F, 0x01, 0x89, 0xFE,
+ 0xFB, 0x00, 0x9C, 0xFF, 0x0D, 0x00, 0x06, 0x00, 0xC9, 0xFF, 0x68,
+ 0x00, 0xE5, 0xFF, 0xA0, 0xFE, 0xFB, 0x04, 0x0C, 0xF3, 0xC5, 0x2A,
+ 0xD8, 0x33, 0xC9, 0xF4, 0x0B, 0x03, 0x05, 0x00, 0x1A, 0xFF, 0xC1,
+ 0x00, 0xAD, 0xFF, 0x0A, 0x00, 0x09, 0x00, 0xB5, 0xFF, 0xA5, 0x00,
+ 0x5C, 0xFF, 0x8C, 0xFF, 0xBF, 0x03, 0x06, 0xF4, 0x22, 0x31, 0xC8,
+ 0x2D, 0x63, 0xF3, 0x76, 0x04, 0x08, 0xFF, 0xA7, 0xFF, 0x84, 0x00,
+ 0xC0, 0xFF, 0x07, 0x00, 0x0C, 0x00, 0xA4, 0xFF, 0xE1, 0x00, 0xCB,
+ 0xFE, 0x9B, 0x00, 0x21, 0x02, 0xEE, 0xF5, 0xCD, 0x36, 0x24, 0x27,
+ 0xE1, 0xF2, 0x7A, 0x05, 0x33, 0xFE, 0x2A, 0x00, 0x47, 0x00, 0xD3,
+ 0xFF, 0x04, 0x00, 0x0F, 0x00, 0x95, 0xFF, 0x17, 0x01, 0x3D, 0xFE,
+ 0xBD, 0x01, 0x30, 0x00, 0xCC, 0xF8, 0x92, 0x3B, 0x2A, 0x20, 0x2E,
+ 0xF3, 0x12, 0x06, 0x8F, 0xFD, 0x9A, 0x00, 0x10, 0x00, 0xE5, 0xFF,
+ 0x02, 0x00, 0x10, 0x00, 0x8C, 0xFF, 0x42, 0x01, 0xBB, 0xFD, 0xE4,
+ 0x02, 0x01, 0xFE, 0x9C, 0xFC, 0x45, 0x3F, 0x16, 0x19, 0x2D, 0xF4,
+ 0x41, 0x06, 0x21, 0xFD, 0xF3, 0x00, 0xE0, 0xFF, 0xF4, 0xFF, 0x01,
+ 0x00, 0x10, 0x00, 0x8B, 0xFF, 0x5D, 0x01, 0x4F, 0xFD, 0xFB, 0x03,
+ 0xB2, 0xFB, 0x53, 0x01, 0xC2, 0x41, 0x24, 0x12, 0xBA, 0xF5, 0x0F,
+ 0x06, 0xE9, 0xFC, 0x33, 0x01, 0xBB, 0xFF, 0x00, 0x00, 0x00, 0x00,
+ 0x0D, 0x00, 0x93, 0xFF, 0x63, 0x01, 0x04, 0xFD, 0xEF, 0x04, 0x62,
+ 0xF9, 0xD7, 0x06, 0xF2, 0x42, 0x8D, 0x0B, 0xB0, 0xF7, 0x87, 0x05,
+ 0xE6, 0xFC, 0x58, 0x01, 0xA0, 0xFF, 0x09, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x07, 0x00, 0xA5, 0xFF, 0x52, 0x01, 0xE2, 0xFC, 0xAD, 0x05,
+ 0x35, 0xF7, 0x08, 0x0D, 0xCB, 0x42, 0x81, 0x05, 0xE8, 0xF9, 0xBB,
+ 0x04, 0x12, 0xFD, 0x64, 0x01, 0x90, 0xFF, 0x0E, 0x00, 0x00, 0x00,
+ 0xFE, 0xFF, 0xC2, 0xFF, 0x27, 0x01, 0xF1, 0xFC, 0x22, 0x06, 0x54,
+ 0xF5, 0xB8, 0x13, 0x4A, 0x41, 0x29, 0x00, 0x3C, 0xFC, 0xBD, 0x03,
+ 0x66, 0xFD, 0x58, 0x01, 0x8A, 0xFF, 0x11, 0x00, 0x01, 0x00, 0xF1,
+ 0xFF, 0xEB, 0xFF, 0xE1, 0x00, 0x35, 0xFD, 0x40, 0x06, 0xE4, 0xF3,
+ 0xB7, 0x1A, 0x85, 0x3E, 0xA6, 0xFB, 0x86, 0xFE, 0xA0, 0x02, 0xD7,
+ 0xFD, 0x39, 0x01, 0x8E, 0xFF, 0x10, 0x00, 0x03, 0x00, 0xE1, 0xFF,
+ 0x1C, 0x00, 0x82, 0x00, 0xB0, 0xFD, 0xF9, 0x05, 0x0C, 0xF3, 0xCB,
+ 0x21, 0x8F, 0x3A, 0x0D, 0xF8, 0xA9, 0x00, 0x79, 0x01, 0x5D, 0xFE,
+ 0x0B, 0x01, 0x98, 0xFF, 0x0E, 0x00, 0x05, 0x00, 0xCE, 0xFF, 0x55,
+ 0x00, 0x0D, 0x00, 0x60, 0xFE, 0x48, 0x05, 0xEC, 0xF2, 0xB6, 0x28,
+ 0x91, 0x35, 0x68, 0xF5, 0x88, 0x02, 0x5A, 0x00, 0xED, 0xFE, 0xD4,
+ 0x00, 0xA8, 0xFF, 0x0B, 0x00, 0x08, 0x00, 0xBB, 0xFF, 0x92, 0x00,
+ 0x87, 0xFF, 0x3F, 0xFF, 0x2B, 0x04, 0xA1, 0xF3, 0x3D, 0x2F, 0xB8,
+ 0x2F, 0xB8, 0xF3, 0x11, 0x04, 0x52, 0xFF, 0x7C, 0xFF, 0x97, 0x00,
+ 0xBA, 0xFF, 0x08, 0x00, 0x0B, 0x00, 0xA9, 0xFF, 0xCF, 0x00, 0xF8,
+ 0xFE, 0x44, 0x00, 0xAA, 0x02, 0x3E, 0xF5, 0x24, 0x35, 0x3B, 0x29,
+ 0xF2, 0xF2, 0x35, 0x05, 0x70, 0xFE, 0x03, 0x00, 0x5A, 0x00, 0xCD,
+ 0xFF, 0x05, 0x00, 0x0E, 0x00, 0x99, 0xFF, 0x07, 0x01, 0x68, 0xFE,
+ 0x63, 0x01, 0xD0, 0x00, 0xD0, 0xF7, 0x35, 0x3A, 0x55, 0x22, 0x02,
+ 0xF3, 0xEF, 0x05, 0xBC, 0xFD, 0x7A, 0x00, 0x20, 0x00, 0xDF, 0xFF,
+ 0x03, 0x00, 0x10, 0x00, 0x8E, 0xFF, 0x36, 0x01, 0xE1, 0xFD, 0x8A,
+ 0x02, 0xB2, 0xFE, 0x56, 0xFB, 0x40, 0x3E, 0x42, 0x1B, 0xCE, 0xF3,
+ 0x3E, 0x06, 0x3D, 0xFD, 0xDB, 0x00, 0xEE, 0xFF, 0xF0, 0xFF, 0x01,
+ 0x00, 0x11, 0x00, 0x8A, 0xFF, 0x57, 0x01, 0x6D, 0xFD, 0xA8, 0x03,
+ 0x69, 0xFC, 0xC8, 0xFF, 0x20, 0x41, 0x40, 0x14, 0x33, 0xF5, 0x28,
+ 0x06, 0xF5, 0xFC, 0x22, 0x01, 0xC5, 0xFF, 0xFD, 0xFF, 0x00, 0x00,
+ 0x0F, 0x00, 0x8F, 0xFF, 0x64, 0x01, 0x17, 0xFD, 0xA9, 0x04, 0x16,
+ 0xFA, 0x10, 0x05, 0xB8, 0x42, 0x87, 0x0D, 0x0D, 0xF7, 0xB9, 0x05,
+ 0xE2, 0xFC, 0x50, 0x01, 0xA7, 0xFF, 0x07, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x0A, 0x00, 0x9E, 0xFF, 0x5A, 0x01, 0xE8, 0xFC, 0x7A, 0x05,
+ 0xDA, 0xF7, 0x10, 0x0B, 0xFB, 0x42, 0x4B, 0x07, 0x35, 0xF9, 0x00,
+ 0x05, 0x00, 0xFD, 0x63, 0x01, 0x94, 0xFF, 0x0D, 0x00, 0x00, 0x00,
+ 0x01, 0x00, 0xB8, 0xFF, 0x37, 0x01, 0xE7, 0xFC, 0x07, 0x06, 0xDE,
+ 0xF5, 0x9F, 0x11, 0xE4, 0x41, 0xB8, 0x01, 0x84, 0xFB, 0x0F, 0x04,
+ 0x48, 0xFD, 0x5E, 0x01, 0x8B, 0xFF, 0x10, 0x00, 0x01, 0x00, 0xF5,
+ 0xFF, 0xDD, 0xFF, 0xF9, 0x00, 0x1B, 0xFD, 0x41, 0x06, 0x47, 0xF4,
+ 0x8B, 0x18, 0x81, 0x3F, 0xF1, 0xFC, 0xD5, 0xFD, 0xFA, 0x02, 0xB2,
+ 0xFD, 0x45, 0x01, 0x8C, 0xFF, 0x11, 0x00, 0x02, 0x00, 0xE6, 0xFF,
+ 0x0C, 0x00, 0xA2, 0x00, 0x85, 0xFD, 0x1A, 0x06, 0x3C, 0xF3, 0x9F,
+ 0x1F, 0xE6, 0x3B, 0x0E, 0xF9, 0x07, 0x00, 0xD4, 0x01, 0x33, 0xFE,
+ 0x1B, 0x01, 0x94, 0xFF, 0x0F, 0x00, 0x04, 0x00, 0xD4, 0xFF, 0x43,
+ 0x00, 0x33, 0x00, 0x25, 0xFE, 0x89, 0x05, 0xE0, 0xF2, 0x9C, 0x26,
+ 0x33, 0x37, 0x1E, 0xF6, 0xFD, 0x01, 0xB0, 0x00, 0xC0, 0xFE, 0xE6,
+ 0x00, 0xA2, 0xFF, 0x0C, 0x00, 0x07, 0x00, 0xC1, 0xFF, 0x7F, 0x00,
+ 0xB2, 0xFF, 0xF6, 0xFE, 0x8E, 0x04, 0x51, 0xF3, 0x49, 0x2D, 0x98,
+ 0x31, 0x23, 0xF4, 0xA2, 0x03, 0xA0, 0xFF, 0x51, 0xFF, 0xAA, 0x00,
+ 0xB4, 0xFF, 0x09, 0x00, 0x0A, 0x00, 0xAE, 0xFF, 0xBD, 0x00, 0x25,
+ 0xFF, 0xF1, 0xFF, 0x2B, 0x03, 0xA5, 0xF4, 0x68, 0x33, 0x48, 0x2B,
+ 0x17, 0xF3, 0xE7, 0x04, 0xB1, 0xFE, 0xDB, 0xFF, 0x6C, 0x00, 0xC7,
+ 0xFF, 0x06, 0x00, 0x0D, 0x00, 0x9E, 0xFF, 0xF7, 0x00, 0x94, 0xFE,
+ 0x09, 0x01, 0x6A, 0x01, 0xEB, 0xF6, 0xC1, 0x38, 0x7D, 0x24, 0xE8,
+ 0xF2, 0xC1, 0x05, 0xEE, 0xFD, 0x57, 0x00, 0x31, 0x00, 0xDA, 0xFF,
+ 0x03, 0x00, 0x10, 0x00, 0x91, 0xFF, 0x29, 0x01, 0x09, 0xFE, 0x2F,
+ 0x02, 0x5F, 0xFF, 0x27, 0xFA, 0x20, 0x3D, 0x70, 0x1D, 0x7D, 0xF3,
+ 0x31, 0x06, 0x5E, 0xFD, 0xBF, 0x00, 0xFD, 0xFF, 0xEB, 0xFF, 0x02,
+ 0x00, 0x11, 0x00, 0x8B, 0xFF, 0x4E, 0x01, 0x8E, 0xFD, 0x52, 0x03,
+ 0x20, 0xFD, 0x52, 0xFE, 0x60, 0x40, 0x63, 0x16, 0xB7, 0xF4, 0x39,
+ 0x06, 0x05, 0xFD, 0x0F, 0x01, 0xD1, 0xFF, 0xF9, 0xFF, 0x00, 0x00,
+ 0x10, 0x00, 0x8D, 0xFF, 0x62, 0x01, 0x2E, 0xFD, 0x5E, 0x04, 0xCC,
+ 0xFA, 0x5B, 0x03, 0x5E, 0x42, 0x8E, 0x0F, 0x71, 0xF6, 0xE4, 0x05,
+ 0xE2, 0xFC, 0x45, 0x01, 0xAF, 0xFF, 0x04, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x0B, 0x00, 0x99, 0xFF, 0x60, 0x01, 0xF2, 0xFC, 0x40, 0x05,
+ 0x85, 0xF8, 0x26, 0x09, 0x0C, 0x43, 0x26, 0x09, 0x85, 0xF8, 0x40,
+ 0x05, 0xF2, 0xFC, 0x60, 0x01, 0x99, 0xFF, 0x0B, 0x00, 0x00, 0x00,
+ 0x04, 0x00, 0xAF, 0xFF, 0x45, 0x01, 0xE2, 0xFC, 0xE4, 0x05, 0x71,
+ 0xF6, 0x8E, 0x0F, 0x5E, 0x42, 0x5B, 0x03, 0xCC, 0xFA, 0x5E, 0x04,
+ 0x2E, 0xFD, 0x62, 0x01, 0x8D, 0xFF, 0x10, 0x00, 0x00, 0x00, 0xF9,
+ 0xFF, 0xD1, 0xFF, 0x0F, 0x01, 0x05, 0xFD, 0x39, 0x06, 0xB7, 0xF4,
+ 0x63, 0x16, 0x60, 0x40, 0x52, 0xFE, 0x20, 0xFD, 0x52, 0x03, 0x8E,
+ 0xFD, 0x4E, 0x01, 0x8B, 0xFF, 0x11, 0x00, 0x02, 0x00, 0xEB, 0xFF,
+ 0xFD, 0xFF, 0xBF, 0x00, 0x5E, 0xFD, 0x31, 0x06, 0x7D, 0xF3, 0x70,
+ 0x1D, 0x20, 0x3D, 0x27, 0xFA, 0x5F, 0xFF, 0x2F, 0x02, 0x09, 0xFE,
+ 0x29, 0x01, 0x91, 0xFF, 0x10, 0x00, 0x03, 0x00, 0xDA, 0xFF, 0x31,
+ 0x00, 0x57, 0x00, 0xEE, 0xFD, 0xC1, 0x05, 0xE8, 0xF2, 0x7D, 0x24,
+ 0xC1, 0x38, 0xEB, 0xF6, 0x6A, 0x01, 0x09, 0x01, 0x94, 0xFE, 0xF7,
+ 0x00, 0x9E, 0xFF, 0x0D, 0x00, 0x06, 0x00, 0xC7, 0xFF, 0x6C, 0x00,
+ 0xDB, 0xFF, 0xB1, 0xFE, 0xE7, 0x04, 0x17, 0xF3, 0x48, 0x2B, 0x68,
+ 0x33, 0xA5, 0xF4, 0x2B, 0x03, 0xF1, 0xFF, 0x25, 0xFF, 0xBD, 0x00,
+ 0xAE, 0xFF, 0x0A, 0x00, 0x09, 0x00, 0xB4, 0xFF, 0xAA, 0x00, 0x51,
+ 0xFF, 0xA0, 0xFF, 0xA2, 0x03, 0x23, 0xF4, 0x98, 0x31, 0x49, 0x2D,
+ 0x51, 0xF3, 0x8E, 0x04, 0xF6, 0xFE, 0xB2, 0xFF, 0x7F, 0x00, 0xC1,
+ 0xFF, 0x07, 0x00, 0x0C, 0x00, 0xA2, 0xFF, 0xE6, 0x00, 0xC0, 0xFE,
+ 0xB0, 0x00, 0xFD, 0x01, 0x1E, 0xF6, 0x33, 0x37, 0x9C, 0x26, 0xE0,
+ 0xF2, 0x89, 0x05, 0x25, 0xFE, 0x33, 0x00, 0x43, 0x00, 0xD4, 0xFF,
+ 0x04, 0x00, 0x0F, 0x00, 0x94, 0xFF, 0x1B, 0x01, 0x33, 0xFE, 0xD4,
+ 0x01, 0x07, 0x00, 0x0E, 0xF9, 0xE6, 0x3B, 0x9F, 0x1F, 0x3C, 0xF3,
+ 0x1A, 0x06, 0x85, 0xFD, 0xA2, 0x00, 0x0C, 0x00, 0xE6, 0xFF, 0x02,
+ 0x00, 0x11, 0x00, 0x8C, 0xFF, 0x45, 0x01, 0xB2, 0xFD, 0xFA, 0x02,
+ 0xD5, 0xFD, 0xF1, 0xFC, 0x81, 0x3F, 0x8B, 0x18, 0x47, 0xF4, 0x41,
+ 0x06, 0x1B, 0xFD, 0xF9, 0x00, 0xDD, 0xFF, 0xF5, 0xFF, 0x01, 0x00,
+ 0x10, 0x00, 0x8B, 0xFF, 0x5E, 0x01, 0x48, 0xFD, 0x0F, 0x04, 0x84,
+ 0xFB, 0xB8, 0x01, 0xE4, 0x41, 0x9F, 0x11, 0xDE, 0xF5, 0x07, 0x06,
+ 0xE7, 0xFC, 0x37, 0x01, 0xB8, 0xFF, 0x01, 0x00, 0x00, 0x00, 0x0D,
+ 0x00, 0x94, 0xFF, 0x63, 0x01, 0x00, 0xFD, 0x00, 0x05, 0x35, 0xF9,
+ 0x4B, 0x07, 0xFB, 0x42, 0x10, 0x0B, 0xDA, 0xF7, 0x7A, 0x05, 0xE8,
+ 0xFC, 0x5A, 0x01, 0x9E, 0xFF, 0x0A, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x07, 0x00, 0xA7, 0xFF, 0x50, 0x01, 0xE2, 0xFC, 0xB9, 0x05, 0x0D,
+ 0xF7, 0x87, 0x0D, 0xB8, 0x42, 0x10, 0x05, 0x16, 0xFA, 0xA9, 0x04,
+ 0x17, 0xFD, 0x64, 0x01, 0x8F, 0xFF, 0x0F, 0x00, 0x00, 0x00, 0xFD,
+ 0xFF, 0xC5, 0xFF, 0x22, 0x01, 0xF5, 0xFC, 0x28, 0x06, 0x33, 0xF5,
+ 0x40, 0x14, 0x20, 0x41, 0xC8, 0xFF, 0x69, 0xFC, 0xA8, 0x03, 0x6D,
+ 0xFD, 0x57, 0x01, 0x8A, 0xFF, 0x11, 0x00, 0x01, 0x00, 0xF0, 0xFF,
+ 0xEE, 0xFF, 0xDB, 0x00, 0x3D, 0xFD, 0x3E, 0x06, 0xCE, 0xF3, 0x42,
+ 0x1B, 0x40, 0x3E, 0x56, 0xFB, 0xB2, 0xFE, 0x8A, 0x02, 0xE1, 0xFD,
+ 0x36, 0x01, 0x8E, 0xFF, 0x10, 0x00, 0x03, 0x00, 0xDF, 0xFF, 0x20,
+ 0x00, 0x7A, 0x00, 0xBC, 0xFD, 0xEF, 0x05, 0x02, 0xF3, 0x55, 0x22,
+ 0x35, 0x3A, 0xD0, 0xF7, 0xD0, 0x00, 0x63, 0x01, 0x68, 0xFE, 0x07,
+ 0x01, 0x99, 0xFF, 0x0E, 0x00, 0x05, 0x00, 0xCD, 0xFF, 0x5A, 0x00,
+ 0x03, 0x00, 0x70, 0xFE, 0x35, 0x05, 0xF2, 0xF2, 0x3B, 0x29, 0x24,
+ 0x35, 0x3E, 0xF5, 0xAA, 0x02, 0x44, 0x00, 0xF8, 0xFE, 0xCF, 0x00,
+ 0xA9, 0xFF, 0x0B, 0x00, 0x08, 0x00, 0xBA, 0xFF, 0x97, 0x00, 0x7C,
+ 0xFF, 0x52, 0xFF, 0x11, 0x04, 0xB8, 0xF3, 0xB8, 0x2F, 0x3D, 0x2F,
+ 0xA1, 0xF3, 0x2B, 0x04, 0x3F, 0xFF, 0x87, 0xFF, 0x92, 0x00, 0xBB,
+ 0xFF, 0x08, 0x00, 0x0B, 0x00, 0xA8, 0xFF, 0xD4, 0x00, 0xED, 0xFE,
+ 0x5A, 0x00, 0x88, 0x02, 0x68, 0xF5, 0x91, 0x35, 0xB6, 0x28, 0xEC,
+ 0xF2, 0x48, 0x05, 0x60, 0xFE, 0x0D, 0x00, 0x55, 0x00, 0xCE, 0xFF,
+ 0x05, 0x00, 0x0E, 0x00, 0x98, 0xFF, 0x0B, 0x01, 0x5D, 0xFE, 0x79,
+ 0x01, 0xA9, 0x00, 0x0D, 0xF8, 0x8F, 0x3A, 0xCB, 0x21, 0x0C, 0xF3,
+ 0xF9, 0x05, 0xB0, 0xFD, 0x82, 0x00, 0x1C, 0x00, 0xE1, 0xFF, 0x03,
+ 0x00, 0x10, 0x00, 0x8E, 0xFF, 0x39, 0x01, 0xD7, 0xFD, 0xA0, 0x02,
+ 0x86, 0xFE, 0xA6, 0xFB, 0x85, 0x3E, 0xB7, 0x1A, 0xE4, 0xF3, 0x40,
+ 0x06, 0x35, 0xFD, 0xE1, 0x00, 0xEB, 0xFF, 0xF1, 0xFF, 0x01, 0x00,
+ 0x11, 0x00, 0x8A, 0xFF, 0x58, 0x01, 0x66, 0xFD, 0xBD, 0x03, 0x3C,
+ 0xFC, 0x29, 0x00, 0x4A, 0x41, 0xB8, 0x13, 0x54, 0xF5, 0x22, 0x06,
+ 0xF1, 0xFC, 0x27, 0x01, 0xC2, 0xFF, 0xFE, 0xFF, 0x00, 0x00, 0x0E,
+ 0x00, 0x90, 0xFF, 0x64, 0x01, 0x12, 0xFD, 0xBB, 0x04, 0xE8, 0xF9,
+ 0x81, 0x05, 0xCB, 0x42, 0x08, 0x0D, 0x35, 0xF7, 0xAD, 0x05, 0xE2,
+ 0xFC, 0x52, 0x01, 0xA5, 0xFF, 0x07, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x09, 0x00, 0xA0, 0xFF, 0x58, 0x01, 0xE6, 0xFC, 0x87, 0x05, 0xB0,
+ 0xF7, 0x8D, 0x0B, 0xF2, 0x42, 0xD7, 0x06, 0x62, 0xF9, 0xEF, 0x04,
+ 0x04, 0xFD, 0x63, 0x01, 0x93, 0xFF, 0x0D, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0xBB, 0xFF, 0x33, 0x01, 0xE9, 0xFC, 0x0F, 0x06, 0xBA, 0xF5,
+ 0x24, 0x12, 0xC2, 0x41, 0x53, 0x01, 0xB2, 0xFB, 0xFB, 0x03, 0x4F,
+ 0xFD, 0x5D, 0x01, 0x8B, 0xFF, 0x10, 0x00, 0x01, 0x00, 0xF4, 0xFF,
+ 0xE0, 0xFF, 0xF3, 0x00, 0x21, 0xFD, 0x41, 0x06, 0x2D, 0xF4, 0x16,
+ 0x19, 0x45, 0x3F, 0x9C, 0xFC, 0x01, 0xFE, 0xE4, 0x02, 0xBB, 0xFD,
+ 0x42, 0x01, 0x8C, 0xFF, 0x10, 0x00, 0x02, 0x00, 0xE5, 0xFF, 0x10,
+ 0x00, 0x9A, 0x00, 0x8F, 0xFD, 0x12, 0x06, 0x2E, 0xF3, 0x2A, 0x20,
+ 0x92, 0x3B, 0xCC, 0xF8, 0x30, 0x00, 0xBD, 0x01, 0x3D, 0xFE, 0x17,
+ 0x01, 0x95, 0xFF, 0x0F, 0x00, 0x04, 0x00, 0xD3, 0xFF, 0x47, 0x00,
+ 0x2A, 0x00, 0x33, 0xFE, 0x7A, 0x05, 0xE1, 0xF2, 0x24, 0x27, 0xCD,
+ 0x36, 0xEE, 0xF5, 0x21, 0x02, 0x9B, 0x00, 0xCB, 0xFE, 0xE1, 0x00,
+ 0xA4, 0xFF, 0x0C, 0x00, 0x07, 0x00, 0xC0, 0xFF, 0x84, 0x00, 0xA7,
+ 0xFF, 0x08, 0xFF, 0x76, 0x04, 0x63, 0xF3, 0xC8, 0x2D, 0x22, 0x31,
+ 0x06, 0xF4, 0xBF, 0x03, 0x8C, 0xFF, 0x5C, 0xFF, 0xA5, 0x00, 0xB5,
+ 0xFF, 0x09, 0x00, 0x0A, 0x00, 0xAD, 0xFF, 0xC1, 0x00, 0x1A, 0xFF,
+ 0x05, 0x00, 0x0B, 0x03, 0xC9, 0xF4, 0xD8, 0x33, 0xC5, 0x2A, 0x0C,
+ 0xF3, 0xFB, 0x04, 0xA0, 0xFE, 0xE5, 0xFF, 0x68, 0x00, 0xC9, 0xFF,
+ 0x06, 0x00, 0x0D, 0x00, 0x9C, 0xFF, 0xFB, 0x00, 0x89, 0xFE, 0x1F,
+ 0x01, 0x44, 0x01, 0x22, 0xF7, 0x20, 0x39, 0xF3, 0x23, 0xED, 0xF2,
+ 0xCE, 0x05, 0xE1, 0xFD, 0x60, 0x00, 0x2D, 0x00, 0xDB, 0xFF, 0x03,
+ 0x00, 0x10, 0x00, 0x90, 0xFF, 0x2D, 0x01, 0xFF, 0xFD, 0x46, 0x02,
+ 0x34, 0xFF, 0x71, 0xFA, 0x6B, 0x3D, 0xE5, 0x1C, 0x90, 0xF3, 0x35,
+ 0x06, 0x55, 0xFD, 0xC6, 0x00, 0xF9, 0xFF, 0xEC, 0xFF, 0x01, 0x00,
+ 0x11, 0x00, 0x8B, 0xFF, 0x51, 0x01, 0x86, 0xFD, 0x68, 0x03, 0xF3,
+ 0xFC, 0xAE, 0xFE, 0x92, 0x40, 0xDA, 0x15, 0xD5, 0xF4, 0x35, 0x06,
+ 0x00, 0xFD, 0x14, 0x01, 0xCE, 0xFF, 0xFA, 0xFF, 0x00, 0x00, 0x0F,
+ 0x00, 0x8D, 0xFF, 0x63, 0x01, 0x28, 0xFD, 0x71, 0x04, 0x9E, 0xFA,
+ 0xC7, 0x03, 0x79, 0x42, 0x0B, 0x0F, 0x97, 0xF6, 0xDA, 0x05, 0xE2,
+ 0xFC, 0x48, 0x01, 0xAD, 0xFF, 0x05, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x0B, 0x00, 0x9A, 0xFF, 0x5F, 0x01, 0xEF, 0xFC, 0x4F, 0x05, 0x5A,
+ 0xF8, 0x9F, 0x09, 0x0A, 0x43, 0xAE, 0x08, 0xB1, 0xF8, 0x30, 0x05,
+ 0xF5, 0xFC, 0x61, 0x01, 0x97, 0xFF, 0x0C, 0x00, 0x00, 0x00, 0x03,
+ 0x00, 0xB1, 0xFF, 0x41, 0x01, 0xE3, 0xFC, 0xED, 0x05, 0x4C, 0xF6,
+ 0x11, 0x10, 0x42, 0x42, 0xF1, 0x02, 0xFA, 0xFA, 0x4B, 0x04, 0x34,
+ 0xFD, 0x61, 0x01, 0x8C, 0xFF, 0x10, 0x00, 0x01, 0x00, 0xF8, 0xFF,
+ 0xD4, 0xFF, 0x0A, 0x01, 0x0A, 0xFD, 0x3C, 0x06, 0x9A, 0xF4, 0xED,
+ 0x16, 0x2A, 0x40, 0xF8, 0xFD, 0x4D, 0xFD, 0x3C, 0x03, 0x97, 0xFD,
+ 0x4C, 0x01, 0x8B, 0xFF, 0x11, 0x00, 0x02, 0x00, 0xEA, 0xFF, 0x00,
+ 0x00, 0xB8, 0x00, 0x67, 0xFD, 0x2C, 0x06, 0x6B, 0xF3, 0xFC, 0x1D,
+ 0xD3, 0x3C, 0xDF, 0xF9, 0x89, 0xFF, 0x18, 0x02, 0x13, 0xFE, 0x26,
+ 0x01, 0x92, 0xFF, 0x0F, 0x00, 0x04, 0x00, 0xD9, 0xFF, 0x36, 0x00,
+ 0x4E, 0x00, 0xFB, 0xFD, 0xB4, 0x05, 0xE4, 0xF2, 0x04, 0x25, 0x5F,
+ 0x38, 0xB6, 0xF6, 0x90, 0x01, 0xF3, 0x00, 0x9F, 0xFE, 0xF3, 0x00,
+ 0x9F, 0xFF, 0x0D, 0x00, 0x06, 0x00, 0xC6, 0xFF, 0x71, 0x00, 0xD1,
+ 0xFF, 0xC2, 0xFE, 0xD1, 0x04, 0x23, 0xF3, 0xC9, 0x2B, 0xF5, 0x32,
+ 0x83, 0xF4, 0x49, 0x03, 0xDC, 0xFF, 0x30, 0xFF, 0xB8, 0x00, 0xB0,
+ 0xFF, 0x0A, 0x00, 0x09, 0x00, 0xB3, 0xFF, 0xAE, 0x00, 0x46, 0xFF,
+ 0xB4, 0xFF, 0x85, 0x03, 0x42, 0xF4, 0x0E, 0x32, 0xCA, 0x2C, 0x41,
+ 0xF3, 0xA5, 0x04, 0xE4, 0xFE, 0xBC, 0xFF, 0x7A, 0x00, 0xC3, 0xFF,
+ 0x07, 0x00, 0x0D, 0x00, 0xA1, 0xFF, 0xEA, 0x00, 0xB5, 0xFE, 0xC6,
+ 0x00, 0xD9, 0x01, 0x4F, 0xF6, 0x99, 0x37, 0x16, 0x26, 0xE0, 0xF2,
+ 0x98, 0x05, 0x16, 0xFE, 0x3C, 0x00, 0x3F, 0x00, 0xD6, 0xFF, 0x04,
+ 0x00, 0x0F, 0x00, 0x93, 0xFF, 0x1F, 0x01, 0x28, 0xFE, 0xEB, 0x01,
+ 0xDD, 0xFF, 0x52, 0xF9, 0x36, 0x3C, 0x13, 0x1F, 0x4B, 0xF3, 0x20,
+ 0x06, 0x7B, 0xFD, 0xA9, 0x00, 0x08, 0x00, 0xE7, 0xFF, 0x02, 0x00,
+ 0x11, 0x00, 0x8C, 0xFF, 0x47, 0x01, 0xA9, 0xFD, 0x10, 0x03, 0xA8,
+ 0xFD, 0x47, 0xFD, 0xBB, 0x3F, 0x01, 0x18, 0x62, 0xF4, 0x40, 0x06,
+ 0x15, 0xFD, 0xFF, 0x00, 0xDA, 0xFF, 0xF6, 0xFF, 0x01, 0x00, 0x10,
+ 0x00, 0x8B, 0xFF, 0x5F, 0x01, 0x41, 0xFD, 0x23, 0x04, 0x56, 0xFB,
+ 0x1F, 0x02, 0x06, 0x42, 0x19, 0x11, 0x02, 0xF6, 0xFF, 0x05, 0xE5,
+ 0xFC, 0x3B, 0x01, 0xB6, 0xFF, 0x02, 0x00, 0x00, 0x00, 0x0D, 0x00,
+ 0x95, 0xFF, 0x62, 0x01, 0xFC, 0xFC, 0x10, 0x05, 0x09, 0xF9, 0xC1,
+ 0x07, 0x03, 0x43, 0x94, 0x0A, 0x05, 0xF8, 0x6C, 0x05, 0xEA, 0xFC,
+ 0x5C, 0x01, 0x9D, 0xFF, 0x0A, 0x00, 0x00, 0x00, 0x00, 0x00, 0x06,
+ 0x00, 0xA9, 0xFF, 0x4D, 0x01, 0xE1, 0xFC, 0xC4, 0x05, 0xE6, 0xF6,
+ 0x08, 0x0E, 0xA5, 0x42, 0xA1, 0x04, 0x43, 0xFA, 0x97, 0x04, 0x1D,
+ 0xFD, 0x64, 0x01, 0x8F, 0xFF, 0x0F, 0x00, 0x00, 0x00, 0xFC, 0xFF,
+ 0xC8, 0xFF, 0x1E, 0x01, 0xF8, 0xFC, 0x2D, 0x06, 0x13, 0xF5, 0xC8,
+ 0x14, 0xF2, 0x40, 0x69, 0xFF, 0x97, 0xFC, 0x92, 0x03, 0x75, 0xFD,
+ 0x55, 0x01, 0x8A, 0xFF, 0x11, 0x00, 0x01, 0x00, 0xEF, 0xFF, 0xF2,
+ 0xFF, 0xD4, 0x00, 0x45, 0xFD, 0x3B, 0x06, 0xB8, 0xF3, 0xCE, 0x1B,
+ 0xFB, 0x3D, 0x08, 0xFB, 0xDE, 0xFE, 0x73, 0x02, 0xEB, 0xFD, 0x33,
+ 0x01, 0x8F, 0xFF, 0x10, 0x00, 0x03, 0x00, 0xDE, 0xFF, 0x25, 0x00,
+ 0x71, 0x00, 0xC8, 0xFD, 0xE5, 0x05, 0xFA, 0xF2, 0xDF, 0x22, 0xDB,
+ 0x39, 0x94, 0xF7, 0xF7, 0x00, 0x4C, 0x01, 0x73, 0xFE, 0x03, 0x01,
+ 0x9A, 0xFF, 0x0E, 0x00, 0x05, 0x00, 0xCC, 0xFF, 0x5E, 0x00, 0xF9,
+ 0xFF, 0x80, 0xFE, 0x23, 0x05, 0xF9, 0xF2, 0xC0, 0x29, 0xB8, 0x34,
+ 0x16, 0xF5, 0xCB, 0x02, 0x2F, 0x00, 0x03, 0xFF, 0xCA, 0x00, 0xAA,
+ 0xFF, 0x0B, 0x00, 0x08, 0x00, 0xB8, 0xFF, 0x9B, 0x00, 0x72, 0xFF,
+ 0x65, 0xFF, 0xF6, 0x03, 0xD1, 0xF3, 0x31, 0x30, 0xC1, 0x2E, 0x8B,
+ 0xF3, 0x45, 0x04, 0x2D, 0xFF, 0x92, 0xFF, 0x8D, 0x00, 0xBD, 0xFF,
+ 0x08, 0x00, 0x0C, 0x00, 0xA6, 0xFF, 0xD8, 0x00, 0xE2, 0xFE, 0x6F,
+ 0x00, 0x66, 0x02, 0x93, 0xF5, 0xFB, 0x35, 0x31, 0x28, 0xE7, 0xF2,
+ 0x59, 0x05, 0x51, 0xFE, 0x17, 0x00, 0x50, 0x00, 0xD0, 0xFF, 0x05,
+ 0x00, 0x0E, 0x00, 0x97, 0xFF, 0x0F, 0x01, 0x53, 0xFE, 0x90, 0x01,
+ 0x81, 0x00, 0x4B, 0xF8, 0xE6, 0x3A, 0x3F, 0x21, 0x16, 0xF3, 0x02,
+ 0x06, 0xA5, 0xFD, 0x8A, 0x00, 0x18, 0x00, 0xE2, 0xFF, 0x02, 0x00,
+ 0x10, 0x00, 0x8D, 0xFF, 0x3C, 0x01, 0xCE, 0xFD, 0xB7, 0x02, 0x5A,
+ 0xFE, 0xF7, 0xFB, 0xC6, 0x3E, 0x2C, 0x1A, 0xFC, 0xF3, 0x41, 0x06,
+ 0x2E, 0xFD, 0xE7, 0x00, 0xE7, 0xFF, 0xF2, 0xFF, 0x01, 0x00, 0x10,
+ 0x00, 0x8B, 0xFF, 0x5A, 0x01, 0x5E, 0xFD, 0xD2, 0x03, 0x0E, 0xFC,
+ 0x8B, 0x00, 0x75, 0x41, 0x32, 0x13, 0x75, 0xF5, 0x1C, 0x06, 0xEE,
+ 0xFC, 0x2B, 0x01, 0xC0, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x0E, 0x00,
+ 0x91, 0xFF, 0x64, 0x01, 0x0D, 0xFD, 0xCD, 0x04, 0xBB, 0xF9, 0xF2,
+ 0x05, 0xD9, 0x42, 0x88, 0x0C, 0x5E, 0xF7, 0xA1, 0x05, 0xE3, 0xFC,
+ 0x54, 0x01, 0xA3, 0xFF, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x09,
+ 0x00, 0xA2, 0xFF, 0x56, 0x01, 0xE5, 0xFC, 0x94, 0x05, 0x87, 0xF7,
+ 0x0A, 0x0C, 0xE6, 0x42, 0x64, 0x06, 0x8E, 0xF9, 0xDE, 0x04, 0x09,
+ 0xFD, 0x64, 0x01, 0x92, 0xFF, 0x0E, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0xBD, 0xFF, 0x2F, 0x01, 0xEC, 0xFC, 0x16, 0x06, 0x98, 0xF5, 0xAB,
+ 0x12, 0x9C, 0x41, 0xEE, 0x00, 0xE0, 0xFB, 0xE6, 0x03, 0x57, 0xFD,
+ 0x5B, 0x01, 0x8B, 0xFF, 0x10, 0x00, 0x01, 0x00, 0xF3, 0xFF, 0xE4,
+ 0xFF, 0xED, 0x00, 0x27, 0xFD, 0x41, 0x06, 0x14, 0xF4, 0xA1, 0x19,
+ 0x06, 0x3F, 0x49, 0xFC, 0x2E, 0xFE, 0xCD, 0x02, 0xC4, 0xFD, 0x3F,
+ 0x01, 0x8D, 0xFF, 0x10, 0x00, 0x02, 0x00, 0xE3, 0xFF, 0x14, 0x00,
+ 0x92, 0x00, 0x9A, 0xFD, 0x0A, 0x06, 0x22, 0xF3, 0xB4, 0x20, 0x3C,
+ 0x3B, 0x8B, 0xF8, 0x58, 0x00, 0xA7, 0x01, 0x48, 0xFE, 0x13, 0x01,
+ 0x96, 0xFF, 0x0F, 0x00, 0x04, 0x00, 0xD1, 0xFF, 0x4C, 0x00, 0x20,
+ 0x00, 0x42, 0xFE, 0x6A, 0x05, 0xE3, 0xF2, 0xAB, 0x27, 0x66, 0x36,
+ 0xC0, 0xF5, 0x44, 0x02, 0x85, 0x00, 0xD7, 0xFE, 0xDD, 0x00, 0xA5,
+ 0xFF, 0x0C, 0x00, 0x07, 0x00, 0xBE, 0xFF, 0x89, 0x00, 0x9D, 0xFF,
+ 0x1A, 0xFF, 0x5E, 0x04, 0x76, 0xF3, 0x45, 0x2E, 0xAA, 0x30, 0xEB,
+ 0xF3, 0xDB, 0x03, 0x79, 0xFF, 0x67, 0xFF, 0xA0, 0x00, 0xB7, 0xFF,
+ 0x09, 0x00, 0x0B, 0x00, 0xAC, 0xFF, 0xC6, 0x00, 0x0E, 0xFF, 0x1A,
+ 0x00, 0xEB, 0x02, 0xEF, 0xF4, 0x49, 0x34, 0x43, 0x2A, 0x02, 0xF3,
+ 0x0F, 0x05, 0x90, 0xFE, 0xEF, 0xFF, 0x63, 0x00, 0xCA, 0xFF, 0x06,
+ 0x00, 0x0E, 0x00, 0x9B, 0xFF, 0xFF, 0x00, 0x7E, 0xFE, 0x36, 0x01,
+ 0x1E, 0x01, 0x5B, 0xF7, 0x7E, 0x39, 0x69, 0x23, 0xF3, 0xF2, 0xD9,
+ 0x05, 0xD4, 0xFD, 0x69, 0x00, 0x29, 0x00, 0xDD, 0xFF, 0x03, 0x00,
+ 0x10, 0x00, 0x90, 0xFF, 0x30, 0x01, 0xF5, 0xFD, 0x5C, 0x02, 0x09,
+ 0xFF, 0xBC, 0xFA, 0xB5, 0x3D, 0x5A, 0x1C, 0xA3, 0xF3, 0x38, 0x06,
+ 0x4D, 0xFD, 0xCD, 0x00, 0xF5, 0xFF, 0xED, 0xFF, 0x01, 0x00, 0x11,
+ 0x00, 0x8B, 0xFF, 0x53, 0x01, 0x7E, 0xFD, 0x7D, 0x03, 0xC5, 0xFC,
+ 0x0B, 0xFF, 0xC3, 0x40, 0x51, 0x15, 0xF4, 0xF4, 0x31, 0x06, 0xFC,
+ 0xFC, 0x19, 0x01, 0xCB, 0xFF, 0xFB, 0xFF, 0x00, 0x00, 0x0F, 0x00,
+ 0x8E, 0xFF, 0x63, 0x01, 0x22, 0xFD, 0x84, 0x04, 0x71, 0xFA, 0x34,
+ 0x04, 0x90, 0x42, 0x89, 0x0E, 0xBE, 0xF6, 0xCF, 0x05, 0xE1, 0xFC,
+ 0x4A, 0x01, 0xAB, 0xFF, 0x06, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0B,
+ 0x00, 0x9B, 0xFF, 0x5D, 0x01, 0xEC, 0xFC, 0x5D, 0x05, 0x2F, 0xF8,
+ 0x19, 0x0A, 0x07, 0x43, 0x37, 0x08, 0xDD, 0xF8, 0x21, 0x05, 0xF8,
+ 0xFC, 0x62, 0x01, 0x96, 0xFF, 0x0C, 0x00, 0x00, 0x00, 0x03, 0x00,
+ 0xB4, 0xFF, 0x3E, 0x01, 0xE4, 0xFC, 0xF6, 0x05, 0x26, 0xF6, 0x95,
+ 0x10, 0x26, 0x42, 0x87, 0x02, 0x28, 0xFB, 0x37, 0x04, 0x3B, 0xFD,
+ 0x60, 0x01, 0x8C, 0xFF, 0x10, 0x00, 0x01, 0x00, 0xF7, 0xFF, 0xD7,
+ 0xFF, 0x04, 0x01, 0x0F, 0xFD, 0x3E, 0x06, 0x7D, 0xF4, 0x76, 0x17,
+ 0xF4, 0x3F, 0x9F, 0xFD, 0x7B, 0xFD, 0x26, 0x03, 0xA0, 0xFD, 0x4A,
+ 0x01, 0x8B, 0xFF, 0x11, 0x00, 0x02, 0x00, 0xE9, 0xFF, 0x04, 0x00,
+ 0xB1, 0x00, 0x71, 0xFD, 0x26, 0x06, 0x5A, 0xF3, 0x88, 0x1E, 0x87,
+ 0x3C, 0x98, 0xF9, 0xB3, 0xFF, 0x02, 0x02, 0x1E, 0xFE, 0x22, 0x01,
+ 0x93, 0xFF, 0x0F, 0x00, 0x04, 0x00, 0xD7, 0xFF, 0x3A, 0x00, 0x45,
+ 0x00, 0x09, 0xFE, 0xA7, 0x05, 0xE1, 0xF2, 0x8D, 0x25, 0xFD, 0x37,
+ 0x82, 0xF6, 0xB5, 0x01, 0xDC, 0x00, 0xAA, 0xFE, 0xEE, 0x00, 0xA0,
+ 0xFF, 0x0D, 0x00, 0x06, 0x00, 0xC4, 0xFF, 0x76, 0x00, 0xC7, 0xFF,
+ 0xD3, 0xFE, 0xBC, 0x04, 0x31, 0xF3, 0x4A, 0x2C, 0x83, 0x32, 0x61,
+ 0xF4, 0x68, 0x03, 0xC8, 0xFF, 0x3B, 0xFF, 0xB3, 0x00, 0xB1, 0xFF,
+ 0x0A, 0x00, 0x0A, 0x00, 0xB1, 0xFF, 0xB3, 0x00, 0x3B, 0xFF, 0xC8,
+ 0xFF, 0x68, 0x03, 0x61, 0xF4, 0x83, 0x32, 0x4A, 0x2C, 0x31, 0xF3,
+ 0xBC, 0x04, 0xD3, 0xFE, 0xC7, 0xFF, 0x76, 0x00, 0xC4, 0xFF, 0x06,
+ 0x00, 0x0D, 0x00, 0xA0, 0xFF, 0xEE, 0x00, 0xAA, 0xFE, 0xDC, 0x00,
+ 0xB5, 0x01, 0x82, 0xF6, 0xFD, 0x37, 0x8D, 0x25, 0xE1, 0xF2, 0xA7,
+ 0x05, 0x09, 0xFE, 0x45, 0x00, 0x3A, 0x00, 0xD7, 0xFF, 0x04, 0x00,
+ 0x0F, 0x00, 0x93, 0xFF, 0x22, 0x01, 0x1E, 0xFE, 0x02, 0x02, 0xB3,
+ 0xFF, 0x98, 0xF9, 0x87, 0x3C, 0x88, 0x1E, 0x5A, 0xF3, 0x26, 0x06,
+ 0x71, 0xFD, 0xB1, 0x00, 0x04, 0x00, 0xE9, 0xFF, 0x02, 0x00, 0x11,
+ 0x00, 0x8B, 0xFF, 0x4A, 0x01, 0xA0, 0xFD, 0x26, 0x03, 0x7B, 0xFD,
+ 0x9F, 0xFD, 0xF4, 0x3F, 0x76, 0x17, 0x7D, 0xF4, 0x3E, 0x06, 0x0F,
+ 0xFD, 0x04, 0x01, 0xD7, 0xFF, 0xF7, 0xFF, 0x01, 0x00, 0x10, 0x00,
+ 0x8C, 0xFF, 0x60, 0x01, 0x3B, 0xFD, 0x37, 0x04, 0x28, 0xFB, 0x87,
+ 0x02, 0x26, 0x42, 0x95, 0x10, 0x26, 0xF6, 0xF6, 0x05, 0xE4, 0xFC,
+ 0x3E, 0x01, 0xB4, 0xFF, 0x03, 0x00, 0x00, 0x00, 0x0C, 0x00, 0x96,
+ 0xFF, 0x62, 0x01, 0xF8, 0xFC, 0x21, 0x05, 0xDD, 0xF8, 0x37, 0x08,
+ 0x07, 0x43, 0x19, 0x0A, 0x2F, 0xF8, 0x5D, 0x05, 0xEC, 0xFC, 0x5D,
+ 0x01, 0x9B, 0xFF, 0x0B, 0x00, 0x00, 0x00, 0x00, 0x00, 0x06, 0x00,
+ 0xAB, 0xFF, 0x4A, 0x01, 0xE1, 0xFC, 0xCF, 0x05, 0xBE, 0xF6, 0x89,
+ 0x0E, 0x90, 0x42, 0x34, 0x04, 0x71, 0xFA, 0x84, 0x04, 0x22, 0xFD,
+ 0x63, 0x01, 0x8E, 0xFF, 0x0F, 0x00, 0x00, 0x00, 0xFB, 0xFF, 0xCB,
+ 0xFF, 0x19, 0x01, 0xFC, 0xFC, 0x31, 0x06, 0xF4, 0xF4, 0x51, 0x15,
+ 0xC3, 0x40, 0x0B, 0xFF, 0xC5, 0xFC, 0x7D, 0x03, 0x7E, 0xFD, 0x53,
+ 0x01, 0x8B, 0xFF, 0x11, 0x00, 0x01, 0x00, 0xED, 0xFF, 0xF5, 0xFF,
+ 0xCD, 0x00, 0x4D, 0xFD, 0x38, 0x06, 0xA3, 0xF3, 0x5A, 0x1C, 0xB5,
+ 0x3D, 0xBC, 0xFA, 0x09, 0xFF, 0x5C, 0x02, 0xF5, 0xFD, 0x30, 0x01,
+ 0x90, 0xFF, 0x10, 0x00, 0x03, 0x00, 0xDD, 0xFF, 0x29, 0x00, 0x69,
+ 0x00, 0xD4, 0xFD, 0xD9, 0x05, 0xF3, 0xF2, 0x69, 0x23, 0x7E, 0x39,
+ 0x5B, 0xF7, 0x1E, 0x01, 0x36, 0x01, 0x7E, 0xFE, 0xFF, 0x00, 0x9B,
+ 0xFF, 0x0E, 0x00, 0x06, 0x00, 0xCA, 0xFF, 0x63, 0x00, 0xEF, 0xFF,
+ 0x90, 0xFE, 0x0F, 0x05, 0x02, 0xF3, 0x43, 0x2A, 0x49, 0x34, 0xEF,
+ 0xF4, 0xEB, 0x02, 0x1A, 0x00, 0x0E, 0xFF, 0xC6, 0x00, 0xAC, 0xFF,
+ 0x0B, 0x00, 0x09, 0x00, 0xB7, 0xFF, 0xA0, 0x00, 0x67, 0xFF, 0x79,
+ 0xFF, 0xDB, 0x03, 0xEB, 0xF3, 0xAA, 0x30, 0x45, 0x2E, 0x76, 0xF3,
+ 0x5E, 0x04, 0x1A, 0xFF, 0x9D, 0xFF, 0x89, 0x00, 0xBE, 0xFF, 0x07,
+ 0x00, 0x0C, 0x00, 0xA5, 0xFF, 0xDD, 0x00, 0xD7, 0xFE, 0x85, 0x00,
+ 0x44, 0x02, 0xC0, 0xF5, 0x66, 0x36, 0xAB, 0x27, 0xE3, 0xF2, 0x6A,
+ 0x05, 0x42, 0xFE, 0x20, 0x00, 0x4C, 0x00, 0xD1, 0xFF, 0x04, 0x00,
+ 0x0F, 0x00, 0x96, 0xFF, 0x13, 0x01, 0x48, 0xFE, 0xA7, 0x01, 0x58,
+ 0x00, 0x8B, 0xF8, 0x3C, 0x3B, 0xB4, 0x20, 0x22, 0xF3, 0x0A, 0x06,
+ 0x9A, 0xFD, 0x92, 0x00, 0x14, 0x00, 0xE3, 0xFF, 0x02, 0x00, 0x10,
+ 0x00, 0x8D, 0xFF, 0x3F, 0x01, 0xC4, 0xFD, 0xCD, 0x02, 0x2E, 0xFE,
+ 0x49, 0xFC, 0x06, 0x3F, 0xA1, 0x19, 0x14, 0xF4, 0x41, 0x06, 0x27,
+ 0xFD, 0xED, 0x00, 0xE4, 0xFF, 0xF3, 0xFF, 0x01, 0x00, 0x10, 0x00,
+ 0x8B, 0xFF, 0x5B, 0x01, 0x57, 0xFD, 0xE6, 0x03, 0xE0, 0xFB, 0xEE,
+ 0x00, 0x9C, 0x41, 0xAB, 0x12, 0x98, 0xF5, 0x16, 0x06, 0xEC, 0xFC,
+ 0x2F, 0x01, 0xBD, 0xFF, 0x00, 0x00, 0x00, 0x00, 0x0E, 0x00, 0x92,
+ 0xFF, 0x64, 0x01, 0x09, 0xFD, 0xDE, 0x04, 0x8E, 0xF9, 0x64, 0x06,
+ 0xE6, 0x42, 0x0A, 0x0C, 0x87, 0xF7, 0x94, 0x05, 0xE5, 0xFC, 0x56,
+ 0x01, 0xA2, 0xFF, 0x09, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, 0x00,
+ 0xA3, 0xFF, 0x54, 0x01, 0xE3, 0xFC, 0xA1, 0x05, 0x5E, 0xF7, 0x88,
+ 0x0C, 0xD9, 0x42, 0xF2, 0x05, 0xBB, 0xF9, 0xCD, 0x04, 0x0D, 0xFD,
+ 0x64, 0x01, 0x91, 0xFF, 0x0E, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0xC0,
+ 0xFF, 0x2B, 0x01, 0xEE, 0xFC, 0x1C, 0x06, 0x75, 0xF5, 0x32, 0x13,
+ 0x75, 0x41, 0x8B, 0x00, 0x0E, 0xFC, 0xD2, 0x03, 0x5E, 0xFD, 0x5A,
+ 0x01, 0x8B, 0xFF, 0x10, 0x00, 0x01, 0x00, 0xF2, 0xFF, 0xE7, 0xFF,
+ 0xE7, 0x00, 0x2E, 0xFD, 0x41, 0x06, 0xFC, 0xF3, 0x2C, 0x1A, 0xC6,
+ 0x3E, 0xF7, 0xFB, 0x5A, 0xFE, 0xB7, 0x02, 0xCE, 0xFD, 0x3C, 0x01,
+ 0x8D, 0xFF, 0x10, 0x00, 0x02, 0x00, 0xE2, 0xFF, 0x18, 0x00, 0x8A,
+ 0x00, 0xA5, 0xFD, 0x02, 0x06, 0x16, 0xF3, 0x3F, 0x21, 0xE6, 0x3A,
+ 0x4B, 0xF8, 0x81, 0x00, 0x90, 0x01, 0x53, 0xFE, 0x0F, 0x01, 0x97,
+ 0xFF, 0x0E, 0x00, 0x05, 0x00, 0xD0, 0xFF, 0x50, 0x00, 0x17, 0x00,
+ 0x51, 0xFE, 0x59, 0x05, 0xE7, 0xF2, 0x31, 0x28, 0xFB, 0x35, 0x93,
+ 0xF5, 0x66, 0x02, 0x6F, 0x00, 0xE2, 0xFE, 0xD8, 0x00, 0xA6, 0xFF,
+ 0x0C, 0x00, 0x08, 0x00, 0xBD, 0xFF, 0x8D, 0x00, 0x92, 0xFF, 0x2D,
+ 0xFF, 0x45, 0x04, 0x8B, 0xF3, 0xC1, 0x2E, 0x31, 0x30, 0xD1, 0xF3,
+ 0xF6, 0x03, 0x65, 0xFF, 0x72, 0xFF, 0x9B, 0x00, 0xB8, 0xFF, 0x08,
+ 0x00, 0x0B, 0x00, 0xAA, 0xFF, 0xCA, 0x00, 0x03, 0xFF, 0x2F, 0x00,
+ 0xCB, 0x02, 0x16, 0xF5, 0xB8, 0x34, 0xC0, 0x29, 0xF9, 0xF2, 0x23,
+ 0x05, 0x80, 0xFE, 0xF9, 0xFF, 0x5E, 0x00, 0xCC, 0xFF, 0x05, 0x00,
+ 0x0E, 0x00, 0x9A, 0xFF, 0x03, 0x01, 0x73, 0xFE, 0x4C, 0x01, 0xF7,
+ 0x00, 0x94, 0xF7, 0xDB, 0x39, 0xDF, 0x22, 0xFA, 0xF2, 0xE5, 0x05,
+ 0xC8, 0xFD, 0x71, 0x00, 0x25, 0x00, 0xDE, 0xFF, 0x03, 0x00, 0x10,
+ 0x00, 0x8F, 0xFF, 0x33, 0x01, 0xEB, 0xFD, 0x73, 0x02, 0xDE, 0xFE,
+ 0x08, 0xFB, 0xFB, 0x3D, 0xCE, 0x1B, 0xB8, 0xF3, 0x3B, 0x06, 0x45,
+ 0xFD, 0xD4, 0x00, 0xF2, 0xFF, 0xEF, 0xFF, 0x01, 0x00, 0x11, 0x00,
+ 0x8A, 0xFF, 0x55, 0x01, 0x75, 0xFD, 0x92, 0x03, 0x97, 0xFC, 0x69,
+ 0xFF, 0xF2, 0x40, 0xC8, 0x14, 0x13, 0xF5, 0x2D, 0x06, 0xF8, 0xFC,
+ 0x1E, 0x01, 0xC8, 0xFF, 0xFC, 0xFF, 0x00, 0x00, 0x0F, 0x00, 0x8F,
+ 0xFF, 0x64, 0x01, 0x1D, 0xFD, 0x97, 0x04, 0x43, 0xFA, 0xA1, 0x04,
+ 0xA5, 0x42, 0x08, 0x0E, 0xE6, 0xF6, 0xC4, 0x05, 0xE1, 0xFC, 0x4D,
+ 0x01, 0xA9, 0xFF, 0x06, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0A, 0x00,
+ 0x9D, 0xFF, 0x5C, 0x01, 0xEA, 0xFC, 0x6C, 0x05, 0x05, 0xF8, 0x94,
+ 0x0A, 0x03, 0x43, 0xC1, 0x07, 0x09, 0xF9, 0x10, 0x05, 0xFC, 0xFC,
+ 0x62, 0x01, 0x95, 0xFF, 0x0D, 0x00, 0x00, 0x00, 0x02, 0x00, 0xB6,
+ 0xFF, 0x3B, 0x01, 0xE5, 0xFC, 0xFF, 0x05, 0x02, 0xF6, 0x19, 0x11,
+ 0x06, 0x42, 0x1F, 0x02, 0x56, 0xFB, 0x23, 0x04, 0x41, 0xFD, 0x5F,
+ 0x01, 0x8B, 0xFF, 0x10, 0x00, 0x01, 0x00, 0xF6, 0xFF, 0xDA, 0xFF,
+ 0xFF, 0x00, 0x15, 0xFD, 0x40, 0x06, 0x62, 0xF4, 0x01, 0x18, 0xBB,
+ 0x3F, 0x47, 0xFD, 0xA8, 0xFD, 0x10, 0x03, 0xA9, 0xFD, 0x47, 0x01,
+ 0x8C, 0xFF, 0x11, 0x00, 0x02, 0x00, 0xE7, 0xFF, 0x08, 0x00, 0xA9,
+ 0x00, 0x7B, 0xFD, 0x20, 0x06, 0x4B, 0xF3, 0x13, 0x1F, 0x36, 0x3C,
+ 0x52, 0xF9, 0xDD, 0xFF, 0xEB, 0x01, 0x28, 0xFE, 0x1F, 0x01, 0x93,
+ 0xFF, 0x0F, 0x00, 0x04, 0x00, 0xD6, 0xFF, 0x3F, 0x00, 0x3C, 0x00,
+ 0x16, 0xFE, 0x98, 0x05, 0xE0, 0xF2, 0x16, 0x26, 0x99, 0x37, 0x4F,
+ 0xF6, 0xD9, 0x01, 0xC6, 0x00, 0xB5, 0xFE, 0xEA, 0x00, 0xA1, 0xFF,
+ 0x0D, 0x00, 0x07, 0x00, 0xC3, 0xFF, 0x7A, 0x00, 0xBC, 0xFF, 0xE4,
+ 0xFE, 0xA5, 0x04, 0x41, 0xF3, 0xCA, 0x2C, 0x0E, 0x32, 0x42, 0xF4,
+ 0x85, 0x03, 0xB4, 0xFF, 0x46, 0xFF, 0xAE, 0x00, 0xB3, 0xFF, 0x09,
+ 0x00, 0x0A, 0x00, 0xB0, 0xFF, 0xB8, 0x00, 0x30, 0xFF, 0xDC, 0xFF,
+ 0x49, 0x03, 0x83, 0xF4, 0xF5, 0x32, 0xC9, 0x2B, 0x23, 0xF3, 0xD1,
+ 0x04, 0xC2, 0xFE, 0xD1, 0xFF, 0x71, 0x00, 0xC6, 0xFF, 0x06, 0x00,
+ 0x0D, 0x00, 0x9F, 0xFF, 0xF3, 0x00, 0x9F, 0xFE, 0xF3, 0x00, 0x90,
+ 0x01, 0xB6, 0xF6, 0x5F, 0x38, 0x04, 0x25, 0xE4, 0xF2, 0xB4, 0x05,
+ 0xFB, 0xFD, 0x4E, 0x00, 0x36, 0x00, 0xD9, 0xFF, 0x04, 0x00, 0x0F,
+ 0x00, 0x92, 0xFF, 0x26, 0x01, 0x13, 0xFE, 0x18, 0x02, 0x89, 0xFF,
+ 0xDF, 0xF9, 0xD3, 0x3C, 0xFC, 0x1D, 0x6B, 0xF3, 0x2C, 0x06, 0x67,
+ 0xFD, 0xB8, 0x00, 0x00, 0x00, 0xEA, 0xFF, 0x02, 0x00, 0x11, 0x00,
+ 0x8B, 0xFF, 0x4C, 0x01, 0x97, 0xFD, 0x3C, 0x03, 0x4D, 0xFD, 0xF8,
+ 0xFD, 0x2A, 0x40, 0xED, 0x16, 0x9A, 0xF4, 0x3C, 0x06, 0x0A, 0xFD,
+ 0x0A, 0x01, 0xD4, 0xFF, 0xF8, 0xFF, 0x01, 0x00, 0x10, 0x00, 0x8C,
+ 0xFF, 0x61, 0x01, 0x34, 0xFD, 0x4B, 0x04, 0xFA, 0xFA, 0xF1, 0x02,
+ 0x42, 0x42, 0x11, 0x10, 0x4C, 0xF6, 0xED, 0x05, 0xE3, 0xFC, 0x41,
+ 0x01, 0xB1, 0xFF, 0x03, 0x00, 0x00, 0x00, 0x0C, 0x00, 0x97, 0xFF,
+ 0x61, 0x01, 0xF5, 0xFC, 0x30, 0x05, 0xB1, 0xF8, 0xAE, 0x08, 0x0A,
+ 0x43, 0x9F, 0x09, 0x5A, 0xF8, 0x4F, 0x05, 0xEF, 0xFC, 0x5F, 0x01,
+ 0x9A, 0xFF, 0x0B, 0x00, 0x00, 0x00, 0x00, 0x00, 0x05, 0x00, 0xAD,
+ 0xFF, 0x48, 0x01, 0xE2, 0xFC, 0xDA, 0x05, 0x97, 0xF6, 0x0B, 0x0F,
+ 0x79, 0x42, 0xC7, 0x03, 0x9E, 0xFA, 0x71, 0x04, 0x28, 0xFD, 0x63,
+ 0x01, 0x8D, 0xFF, 0x0F, 0x00
+};
+
+static u16
+coefficient_sizes[8 * 2] = {
+ /* Playback */
+ 0x00C0, 0x5000, 0x0060, 0x2800, 0x0040, 0x0060, 0x1400, 0x0000,
+ /* capture */
+ 0x0020, 0x1260, 0x0020, 0x1260, 0x0000, 0x0040, 0x1260, 0x0000,
+};
+
diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile
new file mode 100644
index 0000000..4ba07d4
--- /dev/null
+++ b/sound/pci/oxygen/Makefile
@@ -0,0 +1,9 @@
+snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o
+snd-hifier-objs := hifier.o
+snd-oxygen-objs := oxygen.o
+snd-virtuoso-objs := virtuoso.o
+
+obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o
+obj-$(CONFIG_SND_HIFIER) += snd-hifier.o
+obj-$(CONFIG_SND_OXYGEN) += snd-oxygen.o
+obj-$(CONFIG_SND_VIRTUOSO) += snd-virtuoso.o
diff --git a/sound/pci/oxygen/ak4396.h b/sound/pci/oxygen/ak4396.h
new file mode 100644
index 0000000..551c1cf
--- /dev/null
+++ b/sound/pci/oxygen/ak4396.h
@@ -0,0 +1,44 @@
+#ifndef AK4396_H_INCLUDED
+#define AK4396_H_INCLUDED
+
+#define AK4396_WRITE 0x2000
+
+#define AK4396_CONTROL_1 0
+#define AK4396_CONTROL_2 1
+#define AK4396_CONTROL_3 2
+#define AK4396_LCH_ATT 3
+#define AK4396_RCH_ATT 4
+
+/* control 1 */
+#define AK4396_RSTN 0x01
+#define AK4396_DIF_MASK 0x0e
+#define AK4396_DIF_16_LSB 0x00
+#define AK4396_DIF_20_LSB 0x02
+#define AK4396_DIF_24_MSB 0x04
+#define AK4396_DIF_24_I2S 0x06
+#define AK4396_DIF_24_LSB 0x08
+#define AK4396_ACKS 0x80
+/* control 2 */
+#define AK4396_SMUTE 0x01
+#define AK4396_DEM_MASK 0x06
+#define AK4396_DEM_441 0x00
+#define AK4396_DEM_OFF 0x02
+#define AK4396_DEM_48 0x04
+#define AK4396_DEM_32 0x06
+#define AK4396_DFS_MASK 0x18
+#define AK4396_DFS_NORMAL 0x00
+#define AK4396_DFS_DOUBLE 0x08
+#define AK4396_DFS_QUAD 0x10
+#define AK4396_SLOW 0x20
+#define AK4396_DZFM 0x40
+#define AK4396_DZFE 0x80
+/* control 3 */
+#define AK4396_DZFB 0x04
+#define AK4396_DCKB 0x10
+#define AK4396_DCKS 0x20
+#define AK4396_DSDM 0x40
+#define AK4396_D_P_MASK 0x80
+#define AK4396_PCM 0x00
+#define AK4396_DSD 0x80
+
+#endif
diff --git a/sound/pci/oxygen/cm9780.h b/sound/pci/oxygen/cm9780.h
new file mode 100644
index 0000000..1445967
--- /dev/null
+++ b/sound/pci/oxygen/cm9780.h
@@ -0,0 +1,63 @@
+#ifndef CM9780_H_INCLUDED
+#define CM9780_H_INCLUDED
+
+#define CM9780_JACK 0x62
+#define CM9780_MIXER 0x64
+#define CM9780_GPIO_SETUP 0x70
+#define CM9780_GPIO_STATUS 0x72
+
+/* jack control */
+#define CM9780_RSOE 0x0001
+#define CM9780_CBOE 0x0002
+#define CM9780_SSOE 0x0004
+#define CM9780_FROE 0x0008
+#define CM9780_HP2FMICOE 0x0010
+#define CM9780_CB2MICOE 0x0020
+#define CM9780_FMIC2LI 0x0040
+#define CM9780_FMIC2MIC 0x0080
+#define CM9780_HP2LI 0x0100
+#define CM9780_HP2MIC 0x0200
+#define CM9780_MIC2LI 0x0400
+#define CM9780_MIC2MIC 0x0800
+#define CM9780_LI2LI 0x1000
+#define CM9780_LI2MIC 0x2000
+#define CM9780_LO2LI 0x4000
+#define CM9780_LO2MIC 0x8000
+
+/* mixer control */
+#define CM9780_BSTSEL 0x0001
+#define CM9780_STRO_MIC 0x0002
+#define CM9780_SPDI_FREX 0x0004
+#define CM9780_SPDI_SSEX 0x0008
+#define CM9780_SPDI_CBEX 0x0010
+#define CM9780_SPDI_RSEX 0x0020
+#define CM9780_MIX2FR 0x0040
+#define CM9780_MIX2SS 0x0080
+#define CM9780_MIX2CB 0x0100
+#define CM9780_MIX2RS 0x0200
+#define CM9780_MIX2FR_EX 0x0400
+#define CM9780_MIX2SS_EX 0x0800
+#define CM9780_MIX2CB_EX 0x1000
+#define CM9780_MIX2RS_EX 0x2000
+#define CM9780_P47_IO 0x4000
+#define CM9780_PCBSW 0x8000
+
+/* GPIO setup */
+#define CM9780_GPI0EN 0x0001
+#define CM9780_GPI1EN 0x0002
+#define CM9780_SENSE_P 0x0004
+#define CM9780_LOCK_P 0x0008
+#define CM9780_GPIO0P 0x0010
+#define CM9780_GPIO1P 0x0020
+#define CM9780_GPIO0IO 0x0100
+#define CM9780_GPIO1IO 0x0200
+
+/* GPIO status */
+#define CM9780_GPO0 0x0001
+#define CM9780_GPO1 0x0002
+#define CM9780_GPIO0S 0x0010
+#define CM9780_GPIO1S 0x0020
+#define CM9780_GPII0S 0x0100
+#define CM9780_GPII1S 0x0200
+
+#endif
diff --git a/sound/pci/oxygen/cs4362a.h b/sound/pci/oxygen/cs4362a.h
new file mode 100644
index 0000000..6a4fedf
--- /dev/null
+++ b/sound/pci/oxygen/cs4362a.h
@@ -0,0 +1,69 @@
+/* register 01h */
+#define CS4362A_PDN 0x01
+#define CS4362A_DAC1_DIS 0x02
+#define CS4362A_DAC2_DIS 0x04
+#define CS4362A_DAC3_DIS 0x08
+#define CS4362A_MCLKDIV 0x20
+#define CS4362A_FREEZE 0x40
+#define CS4362A_CPEN 0x80
+/* register 02h */
+#define CS4362A_DIF_MASK 0x70
+#define CS4362A_DIF_LJUST 0x00
+#define CS4362A_DIF_I2S 0x10
+#define CS4362A_DIF_RJUST_16 0x20
+#define CS4362A_DIF_RJUST_24 0x30
+#define CS4362A_DIF_RJUST_20 0x40
+#define CS4362A_DIF_RJUST_18 0x50
+/* register 03h */
+#define CS4362A_MUTEC_MASK 0x03
+#define CS4362A_MUTEC_6 0x00
+#define CS4362A_MUTEC_1 0x01
+#define CS4362A_MUTEC_3 0x03
+#define CS4362A_AMUTE 0x04
+#define CS4362A_MUTEC_POL 0x08
+#define CS4362A_RMP_UP 0x10
+#define CS4362A_SNGLVOL 0x20
+#define CS4362A_ZERO_CROSS 0x40
+#define CS4362A_SOFT_RAMP 0x80
+/* register 04h */
+#define CS4362A_RMP_DN 0x01
+#define CS4362A_DEM_MASK 0x06
+#define CS4362A_DEM_NONE 0x00
+#define CS4362A_DEM_44100 0x02
+#define CS4362A_DEM_48000 0x04
+#define CS4362A_DEM_32000 0x06
+#define CS4362A_FILT_SEL 0x10
+/* register 05h */
+#define CS4362A_INV_A1 0x01
+#define CS4362A_INV_B1 0x02
+#define CS4362A_INV_A2 0x04
+#define CS4362A_INV_B2 0x08
+#define CS4362A_INV_A3 0x10
+#define CS4362A_INV_B3 0x20
+/* register 06h */
+#define CS4362A_FM_MASK 0x03
+#define CS4362A_FM_SINGLE 0x00
+#define CS4362A_FM_DOUBLE 0x01
+#define CS4362A_FM_QUAD 0x02
+#define CS4362A_FM_DSD 0x03
+#define CS4362A_ATAPI_MASK 0x7c
+#define CS4362A_ATAPI_B_MUTE 0x00
+#define CS4362A_ATAPI_B_R 0x04
+#define CS4362A_ATAPI_B_L 0x08
+#define CS4362A_ATAPI_B_LR 0x0c
+#define CS4362A_ATAPI_A_MUTE 0x00
+#define CS4362A_ATAPI_A_R 0x10
+#define CS4362A_ATAPI_A_L 0x20
+#define CS4362A_ATAPI_A_LR 0x30
+#define CS4362A_ATAPI_MIX_LR_VOL 0x40
+#define CS4362A_A_EQ_B 0x80
+/* register 07h */
+#define CS4362A_VOL_MASK 0x7f
+#define CS4362A_MUTE 0x80
+/* register 08h: like 07h */
+/* registers 09h..0Bh: like 06h..08h */
+/* registers 0Ch..0Eh: like 06h..08h */
+/* register 12h */
+#define CS4362A_REV_MASK 0x07
+#define CS4362A_PART_MASK 0xf8
+#define CS4362A_PART_CS4362A 0x50
diff --git a/sound/pci/oxygen/cs4398.h b/sound/pci/oxygen/cs4398.h
new file mode 100644
index 0000000..5faf5ef
--- /dev/null
+++ b/sound/pci/oxygen/cs4398.h
@@ -0,0 +1,69 @@
+/* register 1 */
+#define CS4398_REV_MASK 0x07
+#define CS4398_PART_MASK 0xf8
+#define CS4398_PART_CS4398 0x70
+/* register 2 */
+#define CS4398_FM_MASK 0x03
+#define CS4398_FM_SINGLE 0x00
+#define CS4398_FM_DOUBLE 0x01
+#define CS4398_FM_QUAD 0x02
+#define CS4398_FM_DSD 0x03
+#define CS4398_DEM_MASK 0x0c
+#define CS4398_DEM_NONE 0x00
+#define CS4398_DEM_44100 0x04
+#define CS4398_DEM_48000 0x08
+#define CS4398_DEM_32000 0x0c
+#define CS4398_DIF_MASK 0x70
+#define CS4398_DIF_LJUST 0x00
+#define CS4398_DIF_I2S 0x10
+#define CS4398_DIF_RJUST_16 0x20
+#define CS4398_DIF_RJUST_24 0x30
+#define CS4398_DIF_RJUST_20 0x40
+#define CS4398_DIF_RJUST_18 0x50
+#define CS4398_DSD_SRC 0x80
+/* register 3 */
+#define CS4398_ATAPI_MASK 0x1f
+#define CS4398_ATAPI_B_MUTE 0x00
+#define CS4398_ATAPI_B_R 0x01
+#define CS4398_ATAPI_B_L 0x02
+#define CS4398_ATAPI_B_LR 0x03
+#define CS4398_ATAPI_A_MUTE 0x00
+#define CS4398_ATAPI_A_R 0x04
+#define CS4398_ATAPI_A_L 0x08
+#define CS4398_ATAPI_A_LR 0x0c
+#define CS4398_ATAPI_MIX_LR_VOL 0x10
+#define CS4398_INVERT_B 0x20
+#define CS4398_INVERT_A 0x40
+#define CS4398_VOL_B_EQ_A 0x80
+/* register 4 */
+#define CS4398_MUTEP_MASK 0x03
+#define CS4398_MUTEP_AUTO 0x00
+#define CS4398_MUTEP_LOW 0x02
+#define CS4398_MUTEP_HIGH 0x03
+#define CS4398_MUTE_B 0x08
+#define CS4398_MUTE_A 0x10
+#define CS4398_MUTEC_A_EQ_B 0x20
+#define CS4398_DAMUTE 0x40
+#define CS4398_PAMUTE 0x80
+/* register 5 */
+#define CS4398_VOL_A_MASK 0xff
+/* register 6 */
+#define CS4398_VOL_B_MASK 0xff
+/* register 7 */
+#define CS4398_DIR_DSD 0x01
+#define CS4398_FILT_SEL 0x04
+#define CS4398_RMP_DN 0x10
+#define CS4398_RMP_UP 0x20
+#define CS4398_ZERO_CROSS 0x40
+#define CS4398_SOFT_RAMP 0x80
+/* register 8 */
+#define CS4398_MCLKDIV3 0x08
+#define CS4398_MCLKDIV2 0x10
+#define CS4398_FREEZE 0x20
+#define CS4398_CPEN 0x40
+#define CS4398_PDN 0x80
+/* register 9 */
+#define CS4398_DSD_PM_EN 0x01
+#define CS4398_DSD_PM_MODE 0x02
+#define CS4398_INVALID_DSD 0x04
+#define CS4398_STATIC_DSD 0x08
diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
new file mode 100644
index 0000000..1ab833f
--- /dev/null
+++ b/sound/pci/oxygen/hifier.c
@@ -0,0 +1,216 @@
+/*
+ * C-Media CMI8788 driver for the MediaTek/TempoTec HiFier Fantasia
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/pci.h>
+#include <sound/control.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include "oxygen.h"
+#include "ak4396.h"
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("TempoTec HiFier driver");
+MODULE_LICENSE("GPL v2");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "card index");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "enable card");
+
+static struct pci_device_id hifier_ids[] __devinitdata = {
+ { OXYGEN_PCI_SUBID(0x14c3, 0x1710) },
+ { OXYGEN_PCI_SUBID(0x14c3, 0x1711) },
+ { }
+};
+MODULE_DEVICE_TABLE(pci, hifier_ids);
+
+struct hifier_data {
+ u8 ak4396_ctl2;
+};
+
+static void ak4396_write(struct oxygen *chip, u8 reg, u8 value)
+{
+ oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER |
+ OXYGEN_SPI_DATA_LENGTH_2 |
+ OXYGEN_SPI_CLOCK_160 |
+ (0 << OXYGEN_SPI_CODEC_SHIFT) |
+ OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
+ AK4396_WRITE | (reg << 8) | value);
+}
+
+static void update_ak4396_volume(struct oxygen *chip)
+{
+ ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]);
+ ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]);
+}
+
+static void hifier_registers_init(struct oxygen *chip)
+{
+ struct hifier_data *data = chip->model_data;
+
+ ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
+ ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2);
+ ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM);
+ update_ak4396_volume(chip);
+}
+
+static void hifier_init(struct oxygen *chip)
+{
+ struct hifier_data *data = chip->model_data;
+
+ data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+ hifier_registers_init(chip);
+
+ snd_component_add(chip->card, "AK4396");
+ snd_component_add(chip->card, "CS5340");
+}
+
+static void hifier_cleanup(struct oxygen *chip)
+{
+}
+
+static void hifier_resume(struct oxygen *chip)
+{
+ hifier_registers_init(chip);
+}
+
+static void set_ak4396_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ struct hifier_data *data = chip->model_data;
+ u8 value;
+
+ value = data->ak4396_ctl2 & ~AK4396_DFS_MASK;
+ if (params_rate(params) <= 54000)
+ value |= AK4396_DFS_NORMAL;
+ else if (params_rate(params) <= 108000)
+ value |= AK4396_DFS_DOUBLE;
+ else
+ value |= AK4396_DFS_QUAD;
+ data->ak4396_ctl2 = value;
+
+ msleep(1); /* wait for the new MCLK to become stable */
+
+ ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB);
+ ak4396_write(chip, AK4396_CONTROL_2, value);
+ ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
+}
+
+static void update_ak4396_mute(struct oxygen *chip)
+{
+ struct hifier_data *data = chip->model_data;
+ u8 value;
+
+ value = data->ak4396_ctl2 & ~AK4396_SMUTE;
+ if (chip->dac_mute)
+ value |= AK4396_SMUTE;
+ data->ak4396_ctl2 = value;
+ ak4396_write(chip, AK4396_CONTROL_2, value);
+}
+
+static void set_cs5340_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+}
+
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
+
+static int hifier_control_filter(struct snd_kcontrol_new *template)
+{
+ if (!strcmp(template->name, "Stereo Upmixing"))
+ return 1; /* stereo only - we don't need upmixing */
+ return 0;
+}
+
+static const struct oxygen_model model_hifier = {
+ .shortname = "C-Media CMI8787",
+ .longname = "C-Media Oxygen HD Audio",
+ .chip = "CMI8788",
+ .owner = THIS_MODULE,
+ .init = hifier_init,
+ .control_filter = hifier_control_filter,
+ .cleanup = hifier_cleanup,
+ .resume = hifier_resume,
+ .set_dac_params = set_ak4396_params,
+ .set_adc_params = set_cs5340_params,
+ .update_dac_volume = update_ak4396_volume,
+ .update_dac_mute = update_ak4396_mute,
+ .dac_tlv = ak4396_db_scale,
+ .model_data_size = sizeof(struct hifier_data),
+ .device_config = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_1,
+ .dac_channels = 2,
+ .dac_volume_min = 0,
+ .dac_volume_max = 255,
+ .function_flags = OXYGEN_FUNCTION_SPI,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+};
+
+static int __devinit hifier_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ ++dev;
+ return -ENOENT;
+ }
+ err = oxygen_pci_probe(pci, index[dev], id[dev], &model_hifier, 0);
+ if (err >= 0)
+ ++dev;
+ return err;
+}
+
+static struct pci_driver hifier_driver = {
+ .name = "CMI8787HiFier",
+ .id_table = hifier_ids,
+ .probe = hifier_probe,
+ .remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+ .suspend = oxygen_pci_suspend,
+ .resume = oxygen_pci_resume,
+#endif
+};
+
+static int __init alsa_card_hifier_init(void)
+{
+ return pci_register_driver(&hifier_driver);
+}
+
+static void __exit alsa_card_hifier_exit(void)
+{
+ pci_unregister_driver(&hifier_driver);
+}
+
+module_init(alsa_card_hifier_init)
+module_exit(alsa_card_hifier_exit)
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
new file mode 100644
index 0000000..b60f621
--- /dev/null
+++ b/sound/pci/oxygen/oxygen.c
@@ -0,0 +1,381 @@
+/*
+ * C-Media CMI8788 driver for C-Media's reference design and for the X-Meridian
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/*
+ * SPI 0 -> 1st AK4396 (front)
+ * SPI 1 -> 2nd AK4396 (surround)
+ * SPI 2 -> 3rd AK4396 (center/LFE)
+ * SPI 3 -> WM8785
+ * SPI 4 -> 4th AK4396 (back)
+ *
+ * GPIO 0 -> DFS0 of AK5385
+ * GPIO 1 -> DFS1 of AK5385
+ */
+
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <linux/pci.h>
+#include <sound/ac97_codec.h>
+#include <sound/control.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include "oxygen.h"
+#include "ak4396.h"
+#include "wm8785.h"
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("C-Media CMI8788 driver");
+MODULE_LICENSE("GPL v2");
+MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "card index");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "enable card");
+
+enum {
+ MODEL_CMEDIA_REF, /* C-Media's reference design */
+ MODEL_MERIDIAN, /* AuzenTech X-Meridian */
+};
+
+static struct pci_device_id oxygen_ids[] __devinitdata = {
+ { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x13f6, 0x0010), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN },
+ { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF },
+ { }
+};
+MODULE_DEVICE_TABLE(pci, oxygen_ids);
+
+
+#define GPIO_AK5385_DFS_MASK 0x0003
+#define GPIO_AK5385_DFS_NORMAL 0x0000
+#define GPIO_AK5385_DFS_DOUBLE 0x0001
+#define GPIO_AK5385_DFS_QUAD 0x0002
+
+struct generic_data {
+ u8 ak4396_ctl2;
+ u16 saved_wm8785_registers[2];
+};
+
+static void ak4396_write(struct oxygen *chip, unsigned int codec,
+ u8 reg, u8 value)
+{
+ /* maps ALSA channel pair number to SPI output */
+ static const u8 codec_spi_map[4] = {
+ 0, 1, 2, 4
+ };
+ oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER |
+ OXYGEN_SPI_DATA_LENGTH_2 |
+ OXYGEN_SPI_CLOCK_160 |
+ (codec_spi_map[codec] << OXYGEN_SPI_CODEC_SHIFT) |
+ OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
+ AK4396_WRITE | (reg << 8) | value);
+}
+
+static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value)
+{
+ struct generic_data *data = chip->model_data;
+
+ oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER |
+ OXYGEN_SPI_DATA_LENGTH_2 |
+ OXYGEN_SPI_CLOCK_160 |
+ (3 << OXYGEN_SPI_CODEC_SHIFT) |
+ OXYGEN_SPI_CEN_LATCH_CLOCK_LO,
+ (reg << 9) | value);
+ if (reg < ARRAY_SIZE(data->saved_wm8785_registers))
+ data->saved_wm8785_registers[reg] = value;
+}
+
+static void update_ak4396_volume(struct oxygen *chip)
+{
+ unsigned int i;
+
+ for (i = 0; i < 4; ++i) {
+ ak4396_write(chip, i,
+ AK4396_LCH_ATT, chip->dac_volume[i * 2]);
+ ak4396_write(chip, i,
+ AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]);
+ }
+}
+
+static void ak4396_registers_init(struct oxygen *chip)
+{
+ struct generic_data *data = chip->model_data;
+ unsigned int i;
+
+ for (i = 0; i < 4; ++i) {
+ ak4396_write(chip, i,
+ AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
+ ak4396_write(chip, i,
+ AK4396_CONTROL_2, data->ak4396_ctl2);
+ ak4396_write(chip, i,
+ AK4396_CONTROL_3, AK4396_PCM);
+ }
+ update_ak4396_volume(chip);
+}
+
+static void ak4396_init(struct oxygen *chip)
+{
+ struct generic_data *data = chip->model_data;
+
+ data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL;
+ ak4396_registers_init(chip);
+ snd_component_add(chip->card, "AK4396");
+}
+
+static void ak5385_init(struct oxygen *chip)
+{
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_AK5385_DFS_MASK);
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_AK5385_DFS_MASK);
+ snd_component_add(chip->card, "AK5385");
+}
+
+static void wm8785_registers_init(struct oxygen *chip)
+{
+ struct generic_data *data = chip->model_data;
+
+ wm8785_write(chip, WM8785_R7, 0);
+ wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]);
+ wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]);
+}
+
+static void wm8785_init(struct oxygen *chip)
+{
+ struct generic_data *data = chip->model_data;
+
+ data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE |
+ WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST;
+ data->saved_wm8785_registers[1] = WM8785_WL_24;
+ wm8785_registers_init(chip);
+ snd_component_add(chip->card, "WM8785");
+}
+
+static void generic_init(struct oxygen *chip)
+{
+ ak4396_init(chip);
+ wm8785_init(chip);
+}
+
+static void meridian_init(struct oxygen *chip)
+{
+ ak4396_init(chip);
+ ak5385_init(chip);
+}
+
+static void generic_cleanup(struct oxygen *chip)
+{
+}
+
+static void generic_resume(struct oxygen *chip)
+{
+ ak4396_registers_init(chip);
+ wm8785_registers_init(chip);
+}
+
+static void meridian_resume(struct oxygen *chip)
+{
+ ak4396_registers_init(chip);
+}
+
+static void set_ak4396_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ struct generic_data *data = chip->model_data;
+ unsigned int i;
+ u8 value;
+
+ value = data->ak4396_ctl2 & ~AK4396_DFS_MASK;
+ if (params_rate(params) <= 54000)
+ value |= AK4396_DFS_NORMAL;
+ else if (params_rate(params) <= 108000)
+ value |= AK4396_DFS_DOUBLE;
+ else
+ value |= AK4396_DFS_QUAD;
+ data->ak4396_ctl2 = value;
+
+ msleep(1); /* wait for the new MCLK to become stable */
+
+ for (i = 0; i < 4; ++i) {
+ ak4396_write(chip, i,
+ AK4396_CONTROL_1, AK4396_DIF_24_MSB);
+ ak4396_write(chip, i,
+ AK4396_CONTROL_2, value);
+ ak4396_write(chip, i,
+ AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN);
+ }
+}
+
+static void update_ak4396_mute(struct oxygen *chip)
+{
+ struct generic_data *data = chip->model_data;
+ unsigned int i;
+ u8 value;
+
+ value = data->ak4396_ctl2 & ~AK4396_SMUTE;
+ if (chip->dac_mute)
+ value |= AK4396_SMUTE;
+ data->ak4396_ctl2 = value;
+ for (i = 0; i < 4; ++i)
+ ak4396_write(chip, i, AK4396_CONTROL_2, value);
+}
+
+static void set_wm8785_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned int value;
+
+ wm8785_write(chip, WM8785_R7, 0);
+
+ value = WM8785_MCR_SLAVE | WM8785_FORMAT_LJUST;
+ if (params_rate(params) <= 48000)
+ value |= WM8785_OSR_SINGLE;
+ else if (params_rate(params) <= 96000)
+ value |= WM8785_OSR_DOUBLE;
+ else
+ value |= WM8785_OSR_QUAD;
+ wm8785_write(chip, WM8785_R0, value);
+
+ if (snd_pcm_format_width(params_format(params)) <= 16)
+ value = WM8785_WL_16;
+ else
+ value = WM8785_WL_24;
+ wm8785_write(chip, WM8785_R1, value);
+}
+
+static void set_ak5385_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned int value;
+
+ if (params_rate(params) <= 54000)
+ value = GPIO_AK5385_DFS_NORMAL;
+ else if (params_rate(params) <= 108000)
+ value = GPIO_AK5385_DFS_DOUBLE;
+ else
+ value = GPIO_AK5385_DFS_QUAD;
+ oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
+ value, GPIO_AK5385_DFS_MASK);
+}
+
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
+
+static int generic_probe(struct oxygen *chip, unsigned long driver_data)
+{
+ if (driver_data == MODEL_MERIDIAN) {
+ chip->model.init = meridian_init;
+ chip->model.resume = meridian_resume;
+ chip->model.set_adc_params = set_ak5385_params;
+ chip->model.device_config = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_1_FROM_SPDIF;
+ chip->model.misc_flags = OXYGEN_MISC_MIDI;
+ chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT;
+ }
+ return 0;
+}
+
+static const struct oxygen_model model_generic = {
+ .shortname = "C-Media CMI8788",
+ .longname = "C-Media Oxygen HD Audio",
+ .chip = "CMI8788",
+ .owner = THIS_MODULE,
+ .probe = generic_probe,
+ .init = generic_init,
+ .cleanup = generic_cleanup,
+ .resume = generic_resume,
+ .set_dac_params = set_ak4396_params,
+ .set_adc_params = set_wm8785_params,
+ .update_dac_volume = update_ak4396_volume,
+ .update_dac_mute = update_ak4396_mute,
+ .dac_tlv = ak4396_db_scale,
+ .model_data_size = sizeof(struct generic_data),
+ .device_config = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ PLAYBACK_2_TO_AC97_1 |
+ CAPTURE_0_FROM_I2S_1 |
+ CAPTURE_1_FROM_SPDIF |
+ CAPTURE_2_FROM_AC97_1,
+ .dac_channels = 8,
+ .dac_volume_min = 0,
+ .dac_volume_max = 255,
+ .function_flags = OXYGEN_FUNCTION_SPI |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+};
+
+static int __devinit generic_oxygen_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ ++dev;
+ return -ENOENT;
+ }
+ err = oxygen_pci_probe(pci, index[dev], id[dev],
+ &model_generic, pci_id->driver_data);
+ if (err >= 0)
+ ++dev;
+ return err;
+}
+
+static struct pci_driver oxygen_driver = {
+ .name = "CMI8788",
+ .id_table = oxygen_ids,
+ .probe = generic_oxygen_probe,
+ .remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+ .suspend = oxygen_pci_suspend,
+ .resume = oxygen_pci_resume,
+#endif
+};
+
+static int __init alsa_card_oxygen_init(void)
+{
+ return pci_register_driver(&oxygen_driver);
+}
+
+static void __exit alsa_card_oxygen_exit(void)
+{
+ pci_unregister_driver(&oxygen_driver);
+}
+
+module_init(alsa_card_oxygen_init)
+module_exit(alsa_card_oxygen_exit)
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
new file mode 100644
index 0000000..19107c6
--- /dev/null
+++ b/sound/pci/oxygen/oxygen.h
@@ -0,0 +1,232 @@
+#ifndef OXYGEN_H_INCLUDED
+#define OXYGEN_H_INCLUDED
+
+#include <linux/mutex.h>
+#include <linux/spinlock.h>
+#include <linux/wait.h>
+#include <linux/workqueue.h>
+#include "oxygen_regs.h"
+
+/* 1 << PCM_x == OXYGEN_CHANNEL_x */
+#define PCM_A 0
+#define PCM_B 1
+#define PCM_C 2
+#define PCM_SPDIF 3
+#define PCM_MULTICH 4
+#define PCM_AC97 5
+#define PCM_COUNT 6
+
+#define OXYGEN_IO_SIZE 0x100
+
+/* model-specific configuration of outputs/inputs */
+#define PLAYBACK_0_TO_I2S 0x0001
+ /* PLAYBACK_0_TO_AC97_0 not implemented */
+#define PLAYBACK_1_TO_SPDIF 0x0004
+#define PLAYBACK_2_TO_AC97_1 0x0008
+#define CAPTURE_0_FROM_I2S_1 0x0010
+#define CAPTURE_0_FROM_I2S_2 0x0020
+ /* CAPTURE_0_FROM_AC97_0 not implemented */
+#define CAPTURE_1_FROM_SPDIF 0x0080
+#define CAPTURE_2_FROM_I2S_2 0x0100
+#define CAPTURE_2_FROM_AC97_1 0x0200
+ /* CAPTURE_3_FROM_I2S_3 not implemented */
+#define MIDI_OUTPUT 0x0800
+#define MIDI_INPUT 0x1000
+
+enum {
+ CONTROL_SPDIF_PCM,
+ CONTROL_SPDIF_INPUT_BITS,
+ CONTROL_MIC_CAPTURE_SWITCH,
+ CONTROL_LINE_CAPTURE_SWITCH,
+ CONTROL_CD_CAPTURE_SWITCH,
+ CONTROL_AUX_CAPTURE_SWITCH,
+ CONTROL_COUNT
+};
+
+#define OXYGEN_PCI_SUBID(sv, sd) \
+ .vendor = PCI_VENDOR_ID_CMEDIA, \
+ .device = 0x8788, \
+ .subvendor = sv, \
+ .subdevice = sd
+
+struct pci_dev;
+struct snd_card;
+struct snd_pcm_substream;
+struct snd_pcm_hardware;
+struct snd_pcm_hw_params;
+struct snd_kcontrol_new;
+struct snd_rawmidi;
+struct oxygen;
+
+struct oxygen_model {
+ const char *shortname;
+ const char *longname;
+ const char *chip;
+ struct module *owner;
+ int (*probe)(struct oxygen *chip, unsigned long driver_data);
+ void (*init)(struct oxygen *chip);
+ int (*control_filter)(struct snd_kcontrol_new *template);
+ int (*mixer_init)(struct oxygen *chip);
+ void (*cleanup)(struct oxygen *chip);
+ void (*suspend)(struct oxygen *chip);
+ void (*resume)(struct oxygen *chip);
+ void (*pcm_hardware_filter)(unsigned int channel,
+ struct snd_pcm_hardware *hardware);
+ void (*set_dac_params)(struct oxygen *chip,
+ struct snd_pcm_hw_params *params);
+ void (*set_adc_params)(struct oxygen *chip,
+ struct snd_pcm_hw_params *params);
+ void (*update_dac_volume)(struct oxygen *chip);
+ void (*update_dac_mute)(struct oxygen *chip);
+ void (*gpio_changed)(struct oxygen *chip);
+ void (*uart_input)(struct oxygen *chip);
+ void (*ac97_switch)(struct oxygen *chip,
+ unsigned int reg, unsigned int mute);
+ const unsigned int *dac_tlv;
+ size_t model_data_size;
+ unsigned int device_config;
+ u8 dac_channels;
+ u8 dac_volume_min;
+ u8 dac_volume_max;
+ u8 misc_flags;
+ u8 function_flags;
+ u16 dac_i2s_format;
+ u16 adc_i2s_format;
+};
+
+struct oxygen {
+ unsigned long addr;
+ spinlock_t reg_lock;
+ struct mutex mutex;
+ struct snd_card *card;
+ struct pci_dev *pci;
+ struct snd_rawmidi *midi;
+ int irq;
+ void *model_data;
+ unsigned int interrupt_mask;
+ u8 dac_volume[8];
+ u8 dac_mute;
+ u8 pcm_active;
+ u8 pcm_running;
+ u8 dac_routing;
+ u8 spdif_playback_enable;
+ u8 revision;
+ u8 has_ac97_0;
+ u8 has_ac97_1;
+ u32 spdif_bits;
+ u32 spdif_pcm_bits;
+ struct snd_pcm_substream *streams[PCM_COUNT];
+ struct snd_kcontrol *controls[CONTROL_COUNT];
+ struct work_struct spdif_input_bits_work;
+ struct work_struct gpio_work;
+ wait_queue_head_t ac97_waitqueue;
+ union {
+ u8 _8[OXYGEN_IO_SIZE];
+ __le16 _16[OXYGEN_IO_SIZE / 2];
+ __le32 _32[OXYGEN_IO_SIZE / 4];
+ } saved_registers;
+ u16 saved_ac97_registers[2][0x40];
+ unsigned int uart_input_count;
+ u8 uart_input[32];
+ struct oxygen_model model;
+};
+
+/* oxygen_lib.c */
+
+int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
+ const struct oxygen_model *model,
+ unsigned long driver_data);
+void oxygen_pci_remove(struct pci_dev *pci);
+#ifdef CONFIG_PM
+int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
+int oxygen_pci_resume(struct pci_dev *pci);
+#endif
+
+/* oxygen_mixer.c */
+
+int oxygen_mixer_init(struct oxygen *chip);
+void oxygen_update_dac_routing(struct oxygen *chip);
+void oxygen_update_spdif_source(struct oxygen *chip);
+
+/* oxygen_pcm.c */
+
+int oxygen_pcm_init(struct oxygen *chip);
+
+/* oxygen_io.c */
+
+u8 oxygen_read8(struct oxygen *chip, unsigned int reg);
+u16 oxygen_read16(struct oxygen *chip, unsigned int reg);
+u32 oxygen_read32(struct oxygen *chip, unsigned int reg);
+void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value);
+void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value);
+void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value);
+void oxygen_write8_masked(struct oxygen *chip, unsigned int reg,
+ u8 value, u8 mask);
+void oxygen_write16_masked(struct oxygen *chip, unsigned int reg,
+ u16 value, u16 mask);
+void oxygen_write32_masked(struct oxygen *chip, unsigned int reg,
+ u32 value, u32 mask);
+
+u16 oxygen_read_ac97(struct oxygen *chip, unsigned int codec,
+ unsigned int index);
+void oxygen_write_ac97(struct oxygen *chip, unsigned int codec,
+ unsigned int index, u16 data);
+void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec,
+ unsigned int index, u16 data, u16 mask);
+
+void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data);
+void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data);
+
+void oxygen_reset_uart(struct oxygen *chip);
+void oxygen_write_uart(struct oxygen *chip, u8 data);
+
+static inline void oxygen_set_bits8(struct oxygen *chip,
+ unsigned int reg, u8 value)
+{
+ oxygen_write8_masked(chip, reg, value, value);
+}
+
+static inline void oxygen_set_bits16(struct oxygen *chip,
+ unsigned int reg, u16 value)
+{
+ oxygen_write16_masked(chip, reg, value, value);
+}
+
+static inline void oxygen_set_bits32(struct oxygen *chip,
+ unsigned int reg, u32 value)
+{
+ oxygen_write32_masked(chip, reg, value, value);
+}
+
+static inline void oxygen_clear_bits8(struct oxygen *chip,
+ unsigned int reg, u8 value)
+{
+ oxygen_write8_masked(chip, reg, 0, value);
+}
+
+static inline void oxygen_clear_bits16(struct oxygen *chip,
+ unsigned int reg, u16 value)
+{
+ oxygen_write16_masked(chip, reg, 0, value);
+}
+
+static inline void oxygen_clear_bits32(struct oxygen *chip,
+ unsigned int reg, u32 value)
+{
+ oxygen_write32_masked(chip, reg, 0, value);
+}
+
+static inline void oxygen_ac97_set_bits(struct oxygen *chip, unsigned int codec,
+ unsigned int index, u16 value)
+{
+ oxygen_write_ac97_masked(chip, codec, index, value, value);
+}
+
+static inline void oxygen_ac97_clear_bits(struct oxygen *chip,
+ unsigned int codec,
+ unsigned int index, u16 value)
+{
+ oxygen_write_ac97_masked(chip, codec, index, 0, value);
+}
+
+#endif
diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c
new file mode 100644
index 0000000..3126c4b
--- /dev/null
+++ b/sound/pci/oxygen/oxygen_io.c
@@ -0,0 +1,256 @@
+/*
+ * C-Media CMI8788 driver - helper functions
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/sched.h>
+#include <sound/core.h>
+#include <sound/mpu401.h>
+#include <asm/io.h>
+#include "oxygen.h"
+
+u8 oxygen_read8(struct oxygen *chip, unsigned int reg)
+{
+ return inb(chip->addr + reg);
+}
+EXPORT_SYMBOL(oxygen_read8);
+
+u16 oxygen_read16(struct oxygen *chip, unsigned int reg)
+{
+ return inw(chip->addr + reg);
+}
+EXPORT_SYMBOL(oxygen_read16);
+
+u32 oxygen_read32(struct oxygen *chip, unsigned int reg)
+{
+ return inl(chip->addr + reg);
+}
+EXPORT_SYMBOL(oxygen_read32);
+
+void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value)
+{
+ outb(value, chip->addr + reg);
+ chip->saved_registers._8[reg] = value;
+}
+EXPORT_SYMBOL(oxygen_write8);
+
+void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value)
+{
+ outw(value, chip->addr + reg);
+ chip->saved_registers._16[reg / 2] = cpu_to_le16(value);
+}
+EXPORT_SYMBOL(oxygen_write16);
+
+void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value)
+{
+ outl(value, chip->addr + reg);
+ chip->saved_registers._32[reg / 4] = cpu_to_le32(value);
+}
+EXPORT_SYMBOL(oxygen_write32);
+
+void oxygen_write8_masked(struct oxygen *chip, unsigned int reg,
+ u8 value, u8 mask)
+{
+ u8 tmp = inb(chip->addr + reg);
+ tmp &= ~mask;
+ tmp |= value & mask;
+ outb(tmp, chip->addr + reg);
+ chip->saved_registers._8[reg] = tmp;
+}
+EXPORT_SYMBOL(oxygen_write8_masked);
+
+void oxygen_write16_masked(struct oxygen *chip, unsigned int reg,
+ u16 value, u16 mask)
+{
+ u16 tmp = inw(chip->addr + reg);
+ tmp &= ~mask;
+ tmp |= value & mask;
+ outw(tmp, chip->addr + reg);
+ chip->saved_registers._16[reg / 2] = cpu_to_le16(tmp);
+}
+EXPORT_SYMBOL(oxygen_write16_masked);
+
+void oxygen_write32_masked(struct oxygen *chip, unsigned int reg,
+ u32 value, u32 mask)
+{
+ u32 tmp = inl(chip->addr + reg);
+ tmp &= ~mask;
+ tmp |= value & mask;
+ outl(tmp, chip->addr + reg);
+ chip->saved_registers._32[reg / 4] = cpu_to_le32(tmp);
+}
+EXPORT_SYMBOL(oxygen_write32_masked);
+
+static int oxygen_ac97_wait(struct oxygen *chip, unsigned int mask)
+{
+ u8 status = 0;
+
+ /*
+ * Reading the status register also clears the bits, so we have to save
+ * the read bits in status.
+ */
+ wait_event_timeout(chip->ac97_waitqueue,
+ ({ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS);
+ status & mask; }),
+ msecs_to_jiffies(1) + 1);
+ /*
+ * Check even after a timeout because this function should not require
+ * the AC'97 interrupt to be enabled.
+ */
+ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS);
+ return status & mask ? 0 : -EIO;
+}
+
+/*
+ * About 10% of AC'97 register reads or writes fail to complete, but even those
+ * where the controller indicates completion aren't guaranteed to have actually
+ * happened.
+ *
+ * It's hard to assign blame to either the controller or the codec because both
+ * were made by C-Media ...
+ */
+
+void oxygen_write_ac97(struct oxygen *chip, unsigned int codec,
+ unsigned int index, u16 data)
+{
+ unsigned int count, succeeded;
+ u32 reg;
+
+ reg = data;
+ reg |= index << OXYGEN_AC97_REG_ADDR_SHIFT;
+ reg |= OXYGEN_AC97_REG_DIR_WRITE;
+ reg |= codec << OXYGEN_AC97_REG_CODEC_SHIFT;
+ succeeded = 0;
+ for (count = 5; count > 0; --count) {
+ udelay(5);
+ oxygen_write32(chip, OXYGEN_AC97_REGS, reg);
+ /* require two "completed" writes, just to be sure */
+ if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_WRITE_DONE) >= 0 &&
+ ++succeeded >= 2) {
+ chip->saved_ac97_registers[codec][index / 2] = data;
+ return;
+ }
+ }
+ snd_printk(KERN_ERR "AC'97 write timeout\n");
+}
+EXPORT_SYMBOL(oxygen_write_ac97);
+
+u16 oxygen_read_ac97(struct oxygen *chip, unsigned int codec,
+ unsigned int index)
+{
+ unsigned int count;
+ unsigned int last_read = UINT_MAX;
+ u32 reg;
+
+ reg = index << OXYGEN_AC97_REG_ADDR_SHIFT;
+ reg |= OXYGEN_AC97_REG_DIR_READ;
+ reg |= codec << OXYGEN_AC97_REG_CODEC_SHIFT;
+ for (count = 5; count > 0; --count) {
+ udelay(5);
+ oxygen_write32(chip, OXYGEN_AC97_REGS, reg);
+ udelay(10);
+ if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_READ_DONE) >= 0) {
+ u16 value = oxygen_read16(chip, OXYGEN_AC97_REGS);
+ /* we require two consecutive reads of the same value */
+ if (value == last_read)
+ return value;
+ last_read = value;
+ /*
+ * Invert the register value bits to make sure that two
+ * consecutive unsuccessful reads do not return the same
+ * value.
+ */
+ reg ^= 0xffff;
+ }
+ }
+ snd_printk(KERN_ERR "AC'97 read timeout on codec %u\n", codec);
+ return 0;
+}
+EXPORT_SYMBOL(oxygen_read_ac97);
+
+void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec,
+ unsigned int index, u16 data, u16 mask)
+{
+ u16 value = oxygen_read_ac97(chip, codec, index);
+ value &= ~mask;
+ value |= data & mask;
+ oxygen_write_ac97(chip, codec, index, value);
+}
+EXPORT_SYMBOL(oxygen_write_ac97_masked);
+
+void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data)
+{
+ unsigned int count;
+
+ /* should not need more than 7.68 us (24 * 320 ns) */
+ count = 10;
+ while ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & OXYGEN_SPI_BUSY)
+ && count > 0) {
+ udelay(1);
+ --count;
+ }
+
+ oxygen_write8(chip, OXYGEN_SPI_DATA1, data);
+ oxygen_write8(chip, OXYGEN_SPI_DATA2, data >> 8);
+ if (control & OXYGEN_SPI_DATA_LENGTH_3)
+ oxygen_write8(chip, OXYGEN_SPI_DATA3, data >> 16);
+ oxygen_write8(chip, OXYGEN_SPI_CONTROL, control);
+}
+EXPORT_SYMBOL(oxygen_write_spi);
+
+void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data)
+{
+ unsigned long timeout;
+
+ /* should not need more than about 300 us */
+ timeout = jiffies + msecs_to_jiffies(1);
+ do {
+ if (!(oxygen_read16(chip, OXYGEN_2WIRE_BUS_STATUS)
+ & OXYGEN_2WIRE_BUSY))
+ break;
+ udelay(1);
+ cond_resched();
+ } while (time_after_eq(timeout, jiffies));
+
+ oxygen_write8(chip, OXYGEN_2WIRE_MAP, map);
+ oxygen_write8(chip, OXYGEN_2WIRE_DATA, data);
+ oxygen_write8(chip, OXYGEN_2WIRE_CONTROL,
+ device | OXYGEN_2WIRE_DIR_WRITE);
+}
+EXPORT_SYMBOL(oxygen_write_i2c);
+
+static void _write_uart(struct oxygen *chip, unsigned int port, u8 data)
+{
+ if (oxygen_read8(chip, OXYGEN_MPU401 + 1) & MPU401_TX_FULL)
+ msleep(1);
+ oxygen_write8(chip, OXYGEN_MPU401 + port, data);
+}
+
+void oxygen_reset_uart(struct oxygen *chip)
+{
+ _write_uart(chip, 1, MPU401_RESET);
+ msleep(1); /* wait for ACK */
+ _write_uart(chip, 1, MPU401_ENTER_UART);
+}
+EXPORT_SYMBOL(oxygen_reset_uart);
+
+void oxygen_write_uart(struct oxygen *chip, u8 data)
+{
+ _write_uart(chip, 0, data);
+}
+EXPORT_SYMBOL(oxygen_write_uart);
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
new file mode 100644
index 0000000..84f481d
--- /dev/null
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -0,0 +1,675 @@
+/*
+ * C-Media CMI8788 driver - main driver module
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/mutex.h>
+#include <linux/pci.h>
+#include <sound/ac97_codec.h>
+#include <sound/asoundef.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/mpu401.h>
+#include <sound/pcm.h>
+#include "oxygen.h"
+#include "cm9780.h"
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("C-Media CMI8788 helper library");
+MODULE_LICENSE("GPL v2");
+
+
+static inline int oxygen_uart_input_ready(struct oxygen *chip)
+{
+ return !(oxygen_read8(chip, OXYGEN_MPU401 + 1) & MPU401_RX_EMPTY);
+}
+
+static void oxygen_read_uart(struct oxygen *chip)
+{
+ if (unlikely(!oxygen_uart_input_ready(chip))) {
+ /* no data, but read it anyway to clear the interrupt */
+ oxygen_read8(chip, OXYGEN_MPU401);
+ return;
+ }
+ do {
+ u8 data = oxygen_read8(chip, OXYGEN_MPU401);
+ if (data == MPU401_ACK)
+ continue;
+ if (chip->uart_input_count >= ARRAY_SIZE(chip->uart_input))
+ chip->uart_input_count = 0;
+ chip->uart_input[chip->uart_input_count++] = data;
+ } while (oxygen_uart_input_ready(chip));
+ if (chip->model.uart_input)
+ chip->model.uart_input(chip);
+}
+
+static irqreturn_t oxygen_interrupt(int dummy, void *dev_id)
+{
+ struct oxygen *chip = dev_id;
+ unsigned int status, clear, elapsed_streams, i;
+
+ status = oxygen_read16(chip, OXYGEN_INTERRUPT_STATUS);
+ if (!status)
+ return IRQ_NONE;
+
+ spin_lock(&chip->reg_lock);
+
+ clear = status & (OXYGEN_CHANNEL_A |
+ OXYGEN_CHANNEL_B |
+ OXYGEN_CHANNEL_C |
+ OXYGEN_CHANNEL_SPDIF |
+ OXYGEN_CHANNEL_MULTICH |
+ OXYGEN_CHANNEL_AC97 |
+ OXYGEN_INT_SPDIF_IN_DETECT |
+ OXYGEN_INT_GPIO |
+ OXYGEN_INT_AC97);
+ if (clear) {
+ if (clear & OXYGEN_INT_SPDIF_IN_DETECT)
+ chip->interrupt_mask &= ~OXYGEN_INT_SPDIF_IN_DETECT;
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK,
+ chip->interrupt_mask & ~clear);
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK,
+ chip->interrupt_mask);
+ }
+
+ elapsed_streams = status & chip->pcm_running;
+
+ spin_unlock(&chip->reg_lock);
+
+ for (i = 0; i < PCM_COUNT; ++i)
+ if ((elapsed_streams & (1 << i)) && chip->streams[i])
+ snd_pcm_period_elapsed(chip->streams[i]);
+
+ if (status & OXYGEN_INT_SPDIF_IN_DETECT) {
+ spin_lock(&chip->reg_lock);
+ i = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL);
+ if (i & (OXYGEN_SPDIF_SENSE_INT | OXYGEN_SPDIF_LOCK_INT |
+ OXYGEN_SPDIF_RATE_INT)) {
+ /* write the interrupt bit(s) to clear */
+ oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, i);
+ schedule_work(&chip->spdif_input_bits_work);
+ }
+ spin_unlock(&chip->reg_lock);
+ }
+
+ if (status & OXYGEN_INT_GPIO)
+ schedule_work(&chip->gpio_work);
+
+ if (status & OXYGEN_INT_MIDI) {
+ if (chip->midi)
+ snd_mpu401_uart_interrupt(0, chip->midi->private_data);
+ else
+ oxygen_read_uart(chip);
+ }
+
+ if (status & OXYGEN_INT_AC97)
+ wake_up(&chip->ac97_waitqueue);
+
+ return IRQ_HANDLED;
+}
+
+static void oxygen_spdif_input_bits_changed(struct work_struct *work)
+{
+ struct oxygen *chip = container_of(work, struct oxygen,
+ spdif_input_bits_work);
+ u32 reg;
+
+ /*
+ * This function gets called when there is new activity on the SPDIF
+ * input, or when we lose lock on the input signal, or when the rate
+ * changes.
+ */
+ msleep(1);
+ spin_lock_irq(&chip->reg_lock);
+ reg = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL);
+ if ((reg & (OXYGEN_SPDIF_SENSE_STATUS |
+ OXYGEN_SPDIF_LOCK_STATUS))
+ == OXYGEN_SPDIF_SENSE_STATUS) {
+ /*
+ * If we detect activity on the SPDIF input but cannot lock to
+ * a signal, the clock bit is likely to be wrong.
+ */
+ reg ^= OXYGEN_SPDIF_IN_CLOCK_MASK;
+ oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, reg);
+ spin_unlock_irq(&chip->reg_lock);
+ msleep(1);
+ spin_lock_irq(&chip->reg_lock);
+ reg = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL);
+ if ((reg & (OXYGEN_SPDIF_SENSE_STATUS |
+ OXYGEN_SPDIF_LOCK_STATUS))
+ == OXYGEN_SPDIF_SENSE_STATUS) {
+ /* nothing detected with either clock; give up */
+ if ((reg & OXYGEN_SPDIF_IN_CLOCK_MASK)
+ == OXYGEN_SPDIF_IN_CLOCK_192) {
+ /*
+ * Reset clock to <= 96 kHz because this is
+ * more likely to be received next time.
+ */
+ reg &= ~OXYGEN_SPDIF_IN_CLOCK_MASK;
+ reg |= OXYGEN_SPDIF_IN_CLOCK_96;
+ oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, reg);
+ }
+ }
+ }
+ spin_unlock_irq(&chip->reg_lock);
+
+ if (chip->controls[CONTROL_SPDIF_INPUT_BITS]) {
+ spin_lock_irq(&chip->reg_lock);
+ chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT;
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK,
+ chip->interrupt_mask);
+ spin_unlock_irq(&chip->reg_lock);
+
+ /*
+ * We don't actually know that any channel status bits have
+ * changed, but let's send a notification just to be sure.
+ */
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->controls[CONTROL_SPDIF_INPUT_BITS]->id);
+ }
+}
+
+static void oxygen_gpio_changed(struct work_struct *work)
+{
+ struct oxygen *chip = container_of(work, struct oxygen, gpio_work);
+
+ if (chip->model.gpio_changed)
+ chip->model.gpio_changed(chip);
+}
+
+#ifdef CONFIG_PROC_FS
+static void oxygen_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct oxygen *chip = entry->private_data;
+ int i, j;
+
+ snd_iprintf(buffer, "CMI8788\n\n");
+ for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) {
+ snd_iprintf(buffer, "%02x:", i);
+ for (j = 0; j < 0x10; ++j)
+ snd_iprintf(buffer, " %02x", oxygen_read8(chip, i + j));
+ snd_iprintf(buffer, "\n");
+ }
+ if (mutex_lock_interruptible(&chip->mutex) < 0)
+ return;
+ if (chip->has_ac97_0) {
+ snd_iprintf(buffer, "\nAC97\n");
+ for (i = 0; i < 0x80; i += 0x10) {
+ snd_iprintf(buffer, "%02x:", i);
+ for (j = 0; j < 0x10; j += 2)
+ snd_iprintf(buffer, " %04x",
+ oxygen_read_ac97(chip, 0, i + j));
+ snd_iprintf(buffer, "\n");
+ }
+ }
+ if (chip->has_ac97_1) {
+ snd_iprintf(buffer, "\nAC97 2\n");
+ for (i = 0; i < 0x80; i += 0x10) {
+ snd_iprintf(buffer, "%02x:", i);
+ for (j = 0; j < 0x10; j += 2)
+ snd_iprintf(buffer, " %04x",
+ oxygen_read_ac97(chip, 1, i + j));
+ snd_iprintf(buffer, "\n");
+ }
+ }
+ mutex_unlock(&chip->mutex);
+}
+
+static void oxygen_proc_init(struct oxygen *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(chip->card, "cmi8788", &entry))
+ snd_info_set_text_ops(entry, chip, oxygen_proc_read);
+}
+#else
+#define oxygen_proc_init(chip)
+#endif
+
+static void oxygen_init(struct oxygen *chip)
+{
+ unsigned int i;
+
+ chip->dac_routing = 1;
+ for (i = 0; i < 8; ++i)
+ chip->dac_volume[i] = chip->model.dac_volume_min;
+ chip->dac_mute = 1;
+ chip->spdif_playback_enable = 1;
+ chip->spdif_bits = OXYGEN_SPDIF_C | OXYGEN_SPDIF_ORIGINAL |
+ (IEC958_AES1_CON_PCM_CODER << OXYGEN_SPDIF_CATEGORY_SHIFT);
+ chip->spdif_pcm_bits = chip->spdif_bits;
+
+ if (oxygen_read8(chip, OXYGEN_REVISION) & OXYGEN_REVISION_2)
+ chip->revision = 2;
+ else
+ chip->revision = 1;
+
+ if (chip->revision == 1)
+ oxygen_set_bits8(chip, OXYGEN_MISC,
+ OXYGEN_MISC_PCI_MEM_W_1_CLOCK);
+
+ i = oxygen_read16(chip, OXYGEN_AC97_CONTROL);
+ chip->has_ac97_0 = (i & OXYGEN_AC97_CODEC_0) != 0;
+ chip->has_ac97_1 = (i & OXYGEN_AC97_CODEC_1) != 0;
+
+ oxygen_write8_masked(chip, OXYGEN_FUNCTION,
+ OXYGEN_FUNCTION_RESET_CODEC |
+ chip->model.function_flags,
+ OXYGEN_FUNCTION_RESET_CODEC |
+ OXYGEN_FUNCTION_2WIRE_SPI_MASK |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5);
+ oxygen_write8(chip, OXYGEN_DMA_STATUS, 0);
+ oxygen_write8(chip, OXYGEN_DMA_PAUSE, 0);
+ oxygen_write8(chip, OXYGEN_PLAY_CHANNELS,
+ OXYGEN_PLAY_CHANNELS_2 |
+ OXYGEN_DMA_A_BURST_8 |
+ OXYGEN_DMA_MULTICH_BURST_8);
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+ oxygen_write8_masked(chip, OXYGEN_MISC,
+ chip->model.misc_flags,
+ OXYGEN_MISC_WRITE_PCI_SUBID |
+ OXYGEN_MISC_REC_C_FROM_SPDIF |
+ OXYGEN_MISC_REC_B_FROM_AC97 |
+ OXYGEN_MISC_REC_A_FROM_MULTICH |
+ OXYGEN_MISC_MIDI);
+ oxygen_write8(chip, OXYGEN_REC_FORMAT,
+ (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_A_SHIFT) |
+ (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_B_SHIFT) |
+ (OXYGEN_FORMAT_16 << OXYGEN_REC_FORMAT_C_SHIFT));
+ oxygen_write8(chip, OXYGEN_PLAY_FORMAT,
+ (OXYGEN_FORMAT_16 << OXYGEN_SPDIF_FORMAT_SHIFT) |
+ (OXYGEN_FORMAT_16 << OXYGEN_MULTICH_FORMAT_SHIFT));
+ oxygen_write8(chip, OXYGEN_REC_CHANNELS, OXYGEN_REC_CHANNELS_2_2_2);
+ oxygen_write16(chip, OXYGEN_I2S_MULTICH_FORMAT,
+ OXYGEN_RATE_48000 | chip->model.dac_i2s_format |
+ OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 |
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
+ if (chip->model.device_config & CAPTURE_0_FROM_I2S_1)
+ oxygen_write16(chip, OXYGEN_I2S_A_FORMAT,
+ OXYGEN_RATE_48000 | chip->model.adc_i2s_format |
+ OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 |
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
+ else
+ oxygen_write16(chip, OXYGEN_I2S_A_FORMAT,
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK);
+ if (chip->model.device_config & (CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_2_FROM_I2S_2))
+ oxygen_write16(chip, OXYGEN_I2S_B_FORMAT,
+ OXYGEN_RATE_48000 | chip->model.adc_i2s_format |
+ OXYGEN_I2S_MCLK_256 | OXYGEN_I2S_BITS_16 |
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64);
+ else
+ oxygen_write16(chip, OXYGEN_I2S_B_FORMAT,
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK);
+ oxygen_write16(chip, OXYGEN_I2S_C_FORMAT,
+ OXYGEN_I2S_MASTER | OXYGEN_I2S_MUTE_MCLK);
+ oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_OUT_ENABLE |
+ OXYGEN_SPDIF_LOOPBACK);
+ if (chip->model.device_config & CAPTURE_1_FROM_SPDIF)
+ oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_SENSE_MASK |
+ OXYGEN_SPDIF_LOCK_MASK |
+ OXYGEN_SPDIF_RATE_MASK |
+ OXYGEN_SPDIF_LOCK_PAR |
+ OXYGEN_SPDIF_IN_CLOCK_96,
+ OXYGEN_SPDIF_SENSE_MASK |
+ OXYGEN_SPDIF_LOCK_MASK |
+ OXYGEN_SPDIF_RATE_MASK |
+ OXYGEN_SPDIF_SENSE_PAR |
+ OXYGEN_SPDIF_LOCK_PAR |
+ OXYGEN_SPDIF_IN_CLOCK_MASK);
+ else
+ oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_SENSE_MASK |
+ OXYGEN_SPDIF_LOCK_MASK |
+ OXYGEN_SPDIF_RATE_MASK);
+ oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits);
+ oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+ OXYGEN_2WIRE_LENGTH_8 |
+ OXYGEN_2WIRE_INTERRUPT_MASK |
+ OXYGEN_2WIRE_SPEED_STANDARD);
+ oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK);
+ oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0);
+ oxygen_write16(chip, OXYGEN_GPIO_INTERRUPT_MASK, 0);
+ oxygen_write16(chip, OXYGEN_PLAY_ROUTING,
+ OXYGEN_PLAY_MULTICH_I2S_DAC |
+ OXYGEN_PLAY_SPDIF_SPDIF |
+ (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) |
+ (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
+ (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
+ (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT));
+ oxygen_write8(chip, OXYGEN_REC_ROUTING,
+ OXYGEN_REC_A_ROUTE_I2S_ADC_1 |
+ OXYGEN_REC_B_ROUTE_I2S_ADC_2 |
+ OXYGEN_REC_C_ROUTE_SPDIF);
+ oxygen_write8(chip, OXYGEN_ADC_MONITOR, 0);
+ oxygen_write8(chip, OXYGEN_A_MONITOR_ROUTING,
+ (0 << OXYGEN_A_MONITOR_ROUTE_0_SHIFT) |
+ (1 << OXYGEN_A_MONITOR_ROUTE_1_SHIFT) |
+ (2 << OXYGEN_A_MONITOR_ROUTE_2_SHIFT) |
+ (3 << OXYGEN_A_MONITOR_ROUTE_3_SHIFT));
+
+ if (chip->has_ac97_0 | chip->has_ac97_1)
+ oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK,
+ OXYGEN_AC97_INT_READ_DONE |
+ OXYGEN_AC97_INT_WRITE_DONE);
+ else
+ oxygen_write8(chip, OXYGEN_AC97_INTERRUPT_MASK, 0);
+ oxygen_write32(chip, OXYGEN_AC97_OUT_CONFIG, 0);
+ oxygen_write32(chip, OXYGEN_AC97_IN_CONFIG, 0);
+ if (!(chip->has_ac97_0 | chip->has_ac97_1))
+ oxygen_set_bits16(chip, OXYGEN_AC97_CONTROL,
+ OXYGEN_AC97_CLOCK_DISABLE);
+ if (!chip->has_ac97_0) {
+ oxygen_set_bits16(chip, OXYGEN_AC97_CONTROL,
+ OXYGEN_AC97_NO_CODEC_0);
+ } else {
+ oxygen_write_ac97(chip, 0, AC97_RESET, 0);
+ msleep(1);
+ oxygen_ac97_set_bits(chip, 0, CM9780_GPIO_SETUP,
+ CM9780_GPIO0IO | CM9780_GPIO1IO);
+ oxygen_ac97_set_bits(chip, 0, CM9780_MIXER,
+ CM9780_BSTSEL | CM9780_STRO_MIC |
+ CM9780_MIX2FR | CM9780_PCBSW);
+ oxygen_ac97_set_bits(chip, 0, CM9780_JACK,
+ CM9780_RSOE | CM9780_CBOE |
+ CM9780_SSOE | CM9780_FROE |
+ CM9780_MIC2MIC | CM9780_LI2LI);
+ oxygen_write_ac97(chip, 0, AC97_MASTER, 0x0000);
+ oxygen_write_ac97(chip, 0, AC97_PC_BEEP, 0x8000);
+ oxygen_write_ac97(chip, 0, AC97_MIC, 0x8808);
+ oxygen_write_ac97(chip, 0, AC97_LINE, 0x0808);
+ oxygen_write_ac97(chip, 0, AC97_CD, 0x8808);
+ oxygen_write_ac97(chip, 0, AC97_VIDEO, 0x8808);
+ oxygen_write_ac97(chip, 0, AC97_AUX, 0x8808);
+ oxygen_write_ac97(chip, 0, AC97_REC_GAIN, 0x8000);
+ oxygen_write_ac97(chip, 0, AC97_CENTER_LFE_MASTER, 0x8080);
+ oxygen_write_ac97(chip, 0, AC97_SURROUND_MASTER, 0x8080);
+ oxygen_ac97_clear_bits(chip, 0, CM9780_GPIO_STATUS,
+ CM9780_GPO0);
+ /* power down unused ADCs and DACs */
+ oxygen_ac97_set_bits(chip, 0, AC97_POWERDOWN,
+ AC97_PD_PR0 | AC97_PD_PR1);
+ oxygen_ac97_set_bits(chip, 0, AC97_EXTENDED_STATUS,
+ AC97_EA_PRI | AC97_EA_PRJ | AC97_EA_PRK);
+ }
+ if (chip->has_ac97_1) {
+ oxygen_set_bits32(chip, OXYGEN_AC97_OUT_CONFIG,
+ OXYGEN_AC97_CODEC1_SLOT3 |
+ OXYGEN_AC97_CODEC1_SLOT4);
+ oxygen_write_ac97(chip, 1, AC97_RESET, 0);
+ msleep(1);
+ oxygen_write_ac97(chip, 1, AC97_MASTER, 0x0000);
+ oxygen_write_ac97(chip, 1, AC97_HEADPHONE, 0x8000);
+ oxygen_write_ac97(chip, 1, AC97_PC_BEEP, 0x8000);
+ oxygen_write_ac97(chip, 1, AC97_MIC, 0x8808);
+ oxygen_write_ac97(chip, 1, AC97_LINE, 0x8808);
+ oxygen_write_ac97(chip, 1, AC97_CD, 0x8808);
+ oxygen_write_ac97(chip, 1, AC97_VIDEO, 0x8808);
+ oxygen_write_ac97(chip, 1, AC97_AUX, 0x8808);
+ oxygen_write_ac97(chip, 1, AC97_PCM, 0x0808);
+ oxygen_write_ac97(chip, 1, AC97_REC_SEL, 0x0000);
+ oxygen_write_ac97(chip, 1, AC97_REC_GAIN, 0x0000);
+ oxygen_ac97_set_bits(chip, 1, 0x6a, 0x0040);
+ }
+}
+
+static void oxygen_card_free(struct snd_card *card)
+{
+ struct oxygen *chip = card->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->interrupt_mask = 0;
+ chip->pcm_running = 0;
+ oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+ spin_unlock_irq(&chip->reg_lock);
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ flush_scheduled_work();
+ chip->model.cleanup(chip);
+ mutex_destroy(&chip->mutex);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+}
+
+int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
+ const struct oxygen_model *model,
+ unsigned long driver_data)
+{
+ struct snd_card *card;
+ struct oxygen *chip;
+ int err;
+
+ card = snd_card_new(index, id, model->owner,
+ sizeof *chip + model->model_data_size);
+ if (!card)
+ return -ENOMEM;
+
+ chip = card->private_data;
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->model = *model;
+ chip->model_data = chip + 1;
+ spin_lock_init(&chip->reg_lock);
+ mutex_init(&chip->mutex);
+ INIT_WORK(&chip->spdif_input_bits_work,
+ oxygen_spdif_input_bits_changed);
+ INIT_WORK(&chip->gpio_work, oxygen_gpio_changed);
+ init_waitqueue_head(&chip->ac97_waitqueue);
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ goto err_card;
+
+ err = pci_request_regions(pci, model->chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot reserve PCI resources\n");
+ goto err_pci_enable;
+ }
+
+ if (!(pci_resource_flags(pci, 0) & IORESOURCE_IO) ||
+ pci_resource_len(pci, 0) < OXYGEN_IO_SIZE) {
+ snd_printk(KERN_ERR "invalid PCI I/O range\n");
+ err = -ENXIO;
+ goto err_pci_regions;
+ }
+ chip->addr = pci_resource_start(pci, 0);
+
+ pci_set_master(pci);
+ snd_card_set_dev(card, &pci->dev);
+ card->private_free = oxygen_card_free;
+
+ if (chip->model.probe) {
+ err = chip->model.probe(chip, driver_data);
+ if (err < 0)
+ goto err_card;
+ }
+ oxygen_init(chip);
+ chip->model.init(chip);
+
+ err = request_irq(pci->irq, oxygen_interrupt, IRQF_SHARED,
+ chip->model.chip, chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot grab interrupt %d\n", pci->irq);
+ goto err_card;
+ }
+ chip->irq = pci->irq;
+
+ strcpy(card->driver, chip->model.chip);
+ strcpy(card->shortname, chip->model.shortname);
+ sprintf(card->longname, "%s (rev %u) at %#lx, irq %i",
+ chip->model.longname, chip->revision, chip->addr, chip->irq);
+ strcpy(card->mixername, chip->model.chip);
+ snd_component_add(card, chip->model.chip);
+
+ err = oxygen_pcm_init(chip);
+ if (err < 0)
+ goto err_card;
+
+ err = oxygen_mixer_init(chip);
+ if (err < 0)
+ goto err_card;
+
+ if (chip->model.device_config & (MIDI_OUTPUT | MIDI_INPUT)) {
+ unsigned int info_flags = MPU401_INFO_INTEGRATED;
+ if (chip->model.device_config & MIDI_OUTPUT)
+ info_flags |= MPU401_INFO_OUTPUT;
+ if (chip->model.device_config & MIDI_INPUT)
+ info_flags |= MPU401_INFO_INPUT;
+ err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
+ chip->addr + OXYGEN_MPU401,
+ info_flags, 0, 0,
+ &chip->midi);
+ if (err < 0)
+ goto err_card;
+ }
+
+ oxygen_proc_init(chip);
+
+ spin_lock_irq(&chip->reg_lock);
+ if (chip->model.device_config & CAPTURE_1_FROM_SPDIF)
+ chip->interrupt_mask |= OXYGEN_INT_SPDIF_IN_DETECT;
+ if (chip->has_ac97_0 | chip->has_ac97_1)
+ chip->interrupt_mask |= OXYGEN_INT_AC97;
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
+ spin_unlock_irq(&chip->reg_lock);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto err_card;
+
+ pci_set_drvdata(pci, card);
+ return 0;
+
+err_pci_regions:
+ pci_release_regions(pci);
+err_pci_enable:
+ pci_disable_device(pci);
+err_card:
+ snd_card_free(card);
+ return err;
+}
+EXPORT_SYMBOL(oxygen_pci_probe);
+
+void oxygen_pci_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+EXPORT_SYMBOL(oxygen_pci_remove);
+
+#ifdef CONFIG_PM
+int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct oxygen *chip = card->private_data;
+ unsigned int i, saved_interrupt_mask;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ for (i = 0; i < PCM_COUNT; ++i)
+ if (chip->streams[i])
+ snd_pcm_suspend(chip->streams[i]);
+
+ if (chip->model.suspend)
+ chip->model.suspend(chip);
+
+ spin_lock_irq(&chip->reg_lock);
+ saved_interrupt_mask = chip->interrupt_mask;
+ chip->interrupt_mask = 0;
+ oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+ spin_unlock_irq(&chip->reg_lock);
+
+ synchronize_irq(chip->irq);
+ flush_scheduled_work();
+ chip->interrupt_mask = saved_interrupt_mask;
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+EXPORT_SYMBOL(oxygen_pci_suspend);
+
+static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = {
+ 0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff,
+ 0x00300000, 0x00000fe4, 0x0ff7001f, 0x00000000
+};
+static const u32 ac97_registers_to_restore[2][0x40 / 32] = {
+ { 0x18284fa2, 0x03060000 },
+ { 0x00007fa6, 0x00200000 }
+};
+
+static inline int is_bit_set(const u32 *bitmap, unsigned int bit)
+{
+ return bitmap[bit / 32] & (1 << (bit & 31));
+}
+
+static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec)
+{
+ unsigned int i;
+
+ oxygen_write_ac97(chip, codec, AC97_RESET, 0);
+ msleep(1);
+ for (i = 1; i < 0x40; ++i)
+ if (is_bit_set(ac97_registers_to_restore[codec], i))
+ oxygen_write_ac97(chip, codec, i * 2,
+ chip->saved_ac97_registers[codec][i]);
+}
+
+int oxygen_pci_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct oxygen *chip = card->private_data;
+ unsigned int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ snd_printk(KERN_ERR "cannot reenable device");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ oxygen_write16(chip, OXYGEN_DMA_STATUS, 0);
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0);
+ for (i = 0; i < OXYGEN_IO_SIZE; ++i)
+ if (is_bit_set(registers_to_restore, i))
+ oxygen_write8(chip, i, chip->saved_registers._8[i]);
+ if (chip->has_ac97_0)
+ oxygen_restore_ac97(chip, 0);
+ if (chip->has_ac97_1)
+ oxygen_restore_ac97(chip, 1);
+
+ if (chip->model.resume)
+ chip->model.resume(chip);
+
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+EXPORT_SYMBOL(oxygen_pci_resume);
+#endif /* CONFIG_PM */
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
new file mode 100644
index 0000000..304da16
--- /dev/null
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -0,0 +1,1004 @@
+/*
+ * C-Media CMI8788 driver - mixer code
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/mutex.h>
+#include <sound/ac97_codec.h>
+#include <sound/asoundef.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include "oxygen.h"
+#include "cm9780.h"
+
+static int dac_volume_info(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_info *info)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ info->count = chip->model.dac_channels;
+ info->value.integer.min = chip->model.dac_volume_min;
+ info->value.integer.max = chip->model.dac_volume_max;
+ return 0;
+}
+
+static int dac_volume_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int i;
+
+ mutex_lock(&chip->mutex);
+ for (i = 0; i < chip->model.dac_channels; ++i)
+ value->value.integer.value[i] = chip->dac_volume[i];
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static int dac_volume_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int i;
+ int changed;
+
+ changed = 0;
+ mutex_lock(&chip->mutex);
+ for (i = 0; i < chip->model.dac_channels; ++i)
+ if (value->value.integer.value[i] != chip->dac_volume[i]) {
+ chip->dac_volume[i] = value->value.integer.value[i];
+ changed = 1;
+ }
+ if (changed)
+ chip->model.update_dac_volume(chip);
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+static int dac_mute_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ mutex_lock(&chip->mutex);
+ value->value.integer.value[0] = !chip->dac_mute;
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static int dac_mute_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ int changed;
+
+ mutex_lock(&chip->mutex);
+ changed = !value->value.integer.value[0] != chip->dac_mute;
+ if (changed) {
+ chip->dac_mute = !value->value.integer.value[0];
+ chip->model.update_dac_mute(chip);
+ }
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info)
+{
+ static const char *const names[3] = {
+ "Front", "Front+Surround", "Front+Surround+Back"
+ };
+ struct oxygen *chip = ctl->private_data;
+ unsigned int count = 2 + (chip->model.dac_channels == 8);
+
+ info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ info->count = 1;
+ info->value.enumerated.items = count;
+ if (info->value.enumerated.item >= count)
+ info->value.enumerated.item = count - 1;
+ strcpy(info->value.enumerated.name, names[info->value.enumerated.item]);
+ return 0;
+}
+
+static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ mutex_lock(&chip->mutex);
+ value->value.enumerated.item[0] = chip->dac_routing;
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+void oxygen_update_dac_routing(struct oxygen *chip)
+{
+ /* DAC 0: front, DAC 1: surround, DAC 2: center/LFE, DAC 3: back */
+ static const unsigned int reg_values[3] = {
+ /* stereo -> front */
+ (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) |
+ (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
+ (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
+ (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT),
+ /* stereo -> front+surround */
+ (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) |
+ (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
+ (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
+ (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT),
+ /* stereo -> front+surround+back */
+ (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) |
+ (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
+ (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
+ (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT),
+ };
+ u8 channels;
+ unsigned int reg_value;
+
+ channels = oxygen_read8(chip, OXYGEN_PLAY_CHANNELS) &
+ OXYGEN_PLAY_CHANNELS_MASK;
+ if (channels == OXYGEN_PLAY_CHANNELS_2)
+ reg_value = reg_values[chip->dac_routing];
+ else if (channels == OXYGEN_PLAY_CHANNELS_8)
+ /* in 7.1 mode, "rear" channels go to the "back" jack */
+ reg_value = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) |
+ (3 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
+ (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
+ (1 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+ else
+ reg_value = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) |
+ (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
+ (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
+ (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+ oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
+ OXYGEN_PLAY_DAC0_SOURCE_MASK |
+ OXYGEN_PLAY_DAC1_SOURCE_MASK |
+ OXYGEN_PLAY_DAC2_SOURCE_MASK |
+ OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
+static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int count = 2 + (chip->model.dac_channels == 8);
+ int changed;
+
+ mutex_lock(&chip->mutex);
+ changed = value->value.enumerated.item[0] != chip->dac_routing;
+ if (changed) {
+ chip->dac_routing = min(value->value.enumerated.item[0],
+ count - 1);
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_update_dac_routing(chip);
+ spin_unlock_irq(&chip->reg_lock);
+ }
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+static int spdif_switch_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ mutex_lock(&chip->mutex);
+ value->value.integer.value[0] = chip->spdif_playback_enable;
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static unsigned int oxygen_spdif_rate(unsigned int oxygen_rate)
+{
+ switch (oxygen_rate) {
+ case OXYGEN_RATE_32000:
+ return IEC958_AES3_CON_FS_32000 << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ case OXYGEN_RATE_44100:
+ return IEC958_AES3_CON_FS_44100 << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ default: /* OXYGEN_RATE_48000 */
+ return IEC958_AES3_CON_FS_48000 << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ case OXYGEN_RATE_64000:
+ return 0xb << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ case OXYGEN_RATE_88200:
+ return IEC958_AES3_CON_FS_88200 << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ case OXYGEN_RATE_96000:
+ return IEC958_AES3_CON_FS_96000 << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ case OXYGEN_RATE_176400:
+ return IEC958_AES3_CON_FS_176400 << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ case OXYGEN_RATE_192000:
+ return IEC958_AES3_CON_FS_192000 << OXYGEN_SPDIF_CS_RATE_SHIFT;
+ }
+}
+
+void oxygen_update_spdif_source(struct oxygen *chip)
+{
+ u32 old_control, new_control;
+ u16 old_routing, new_routing;
+ unsigned int oxygen_rate;
+
+ old_control = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL);
+ old_routing = oxygen_read16(chip, OXYGEN_PLAY_ROUTING);
+ if (chip->pcm_active & (1 << PCM_SPDIF)) {
+ new_control = old_control | OXYGEN_SPDIF_OUT_ENABLE;
+ new_routing = (old_routing & ~OXYGEN_PLAY_SPDIF_MASK)
+ | OXYGEN_PLAY_SPDIF_SPDIF;
+ oxygen_rate = (old_control >> OXYGEN_SPDIF_OUT_RATE_SHIFT)
+ & OXYGEN_I2S_RATE_MASK;
+ /* S/PDIF rate was already set by the caller */
+ } else if ((chip->pcm_active & (1 << PCM_MULTICH)) &&
+ chip->spdif_playback_enable) {
+ new_routing = (old_routing & ~OXYGEN_PLAY_SPDIF_MASK)
+ | OXYGEN_PLAY_SPDIF_MULTICH_01;
+ oxygen_rate = oxygen_read16(chip, OXYGEN_I2S_MULTICH_FORMAT)
+ & OXYGEN_I2S_RATE_MASK;
+ new_control = (old_control & ~OXYGEN_SPDIF_OUT_RATE_MASK) |
+ (oxygen_rate << OXYGEN_SPDIF_OUT_RATE_SHIFT) |
+ OXYGEN_SPDIF_OUT_ENABLE;
+ } else {
+ new_control = old_control & ~OXYGEN_SPDIF_OUT_ENABLE;
+ new_routing = old_routing;
+ oxygen_rate = OXYGEN_RATE_44100;
+ }
+ if (old_routing != new_routing) {
+ oxygen_write32(chip, OXYGEN_SPDIF_CONTROL,
+ new_control & ~OXYGEN_SPDIF_OUT_ENABLE);
+ oxygen_write16(chip, OXYGEN_PLAY_ROUTING, new_routing);
+ }
+ if (new_control & OXYGEN_SPDIF_OUT_ENABLE)
+ oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS,
+ oxygen_spdif_rate(oxygen_rate) |
+ ((chip->pcm_active & (1 << PCM_SPDIF)) ?
+ chip->spdif_pcm_bits : chip->spdif_bits));
+ oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, new_control);
+}
+
+static int spdif_switch_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ int changed;
+
+ mutex_lock(&chip->mutex);
+ changed = value->value.integer.value[0] != chip->spdif_playback_enable;
+ if (changed) {
+ chip->spdif_playback_enable = !!value->value.integer.value[0];
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_update_spdif_source(chip);
+ spin_unlock_irq(&chip->reg_lock);
+ }
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+static int spdif_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info)
+{
+ info->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ info->count = 1;
+ return 0;
+}
+
+static void oxygen_to_iec958(u32 bits, struct snd_ctl_elem_value *value)
+{
+ value->value.iec958.status[0] =
+ bits & (OXYGEN_SPDIF_NONAUDIO | OXYGEN_SPDIF_C |
+ OXYGEN_SPDIF_PREEMPHASIS);
+ value->value.iec958.status[1] = /* category and original */
+ bits >> OXYGEN_SPDIF_CATEGORY_SHIFT;
+}
+
+static u32 iec958_to_oxygen(struct snd_ctl_elem_value *value)
+{
+ u32 bits;
+
+ bits = value->value.iec958.status[0] &
+ (OXYGEN_SPDIF_NONAUDIO | OXYGEN_SPDIF_C |
+ OXYGEN_SPDIF_PREEMPHASIS);
+ bits |= value->value.iec958.status[1] << OXYGEN_SPDIF_CATEGORY_SHIFT;
+ if (bits & OXYGEN_SPDIF_NONAUDIO)
+ bits |= OXYGEN_SPDIF_V;
+ return bits;
+}
+
+static inline void write_spdif_bits(struct oxygen *chip, u32 bits)
+{
+ oxygen_write32_masked(chip, OXYGEN_SPDIF_OUTPUT_BITS, bits,
+ OXYGEN_SPDIF_NONAUDIO |
+ OXYGEN_SPDIF_C |
+ OXYGEN_SPDIF_PREEMPHASIS |
+ OXYGEN_SPDIF_CATEGORY_MASK |
+ OXYGEN_SPDIF_ORIGINAL |
+ OXYGEN_SPDIF_V);
+}
+
+static int spdif_default_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ mutex_lock(&chip->mutex);
+ oxygen_to_iec958(chip->spdif_bits, value);
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static int spdif_default_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u32 new_bits;
+ int changed;
+
+ new_bits = iec958_to_oxygen(value);
+ mutex_lock(&chip->mutex);
+ changed = new_bits != chip->spdif_bits;
+ if (changed) {
+ chip->spdif_bits = new_bits;
+ if (!(chip->pcm_active & (1 << PCM_SPDIF)))
+ write_spdif_bits(chip, new_bits);
+ }
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+static int spdif_mask_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ value->value.iec958.status[0] = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS;
+ value->value.iec958.status[1] =
+ IEC958_AES1_CON_CATEGORY | IEC958_AES1_CON_ORIGINAL;
+ return 0;
+}
+
+static int spdif_pcm_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ mutex_lock(&chip->mutex);
+ oxygen_to_iec958(chip->spdif_pcm_bits, value);
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static int spdif_pcm_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u32 new_bits;
+ int changed;
+
+ new_bits = iec958_to_oxygen(value);
+ mutex_lock(&chip->mutex);
+ changed = new_bits != chip->spdif_pcm_bits;
+ if (changed) {
+ chip->spdif_pcm_bits = new_bits;
+ if (chip->pcm_active & (1 << PCM_SPDIF))
+ write_spdif_bits(chip, new_bits);
+ }
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+static int spdif_input_mask_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ value->value.iec958.status[0] = 0xff;
+ value->value.iec958.status[1] = 0xff;
+ value->value.iec958.status[2] = 0xff;
+ value->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static int spdif_input_default_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u32 bits;
+
+ bits = oxygen_read32(chip, OXYGEN_SPDIF_INPUT_BITS);
+ value->value.iec958.status[0] = bits;
+ value->value.iec958.status[1] = bits >> 8;
+ value->value.iec958.status[2] = bits >> 16;
+ value->value.iec958.status[3] = bits >> 24;
+ return 0;
+}
+
+static int spdif_loopback_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+
+ value->value.integer.value[0] =
+ !!(oxygen_read32(chip, OXYGEN_SPDIF_CONTROL)
+ & OXYGEN_SPDIF_LOOPBACK);
+ return 0;
+}
+
+static int spdif_loopback_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u32 oldreg, newreg;
+ int changed;
+
+ spin_lock_irq(&chip->reg_lock);
+ oldreg = oxygen_read32(chip, OXYGEN_SPDIF_CONTROL);
+ if (value->value.integer.value[0])
+ newreg = oldreg | OXYGEN_SPDIF_LOOPBACK;
+ else
+ newreg = oldreg & ~OXYGEN_SPDIF_LOOPBACK;
+ changed = newreg != oldreg;
+ if (changed)
+ oxygen_write32(chip, OXYGEN_SPDIF_CONTROL, newreg);
+ spin_unlock_irq(&chip->reg_lock);
+ return changed;
+}
+
+static int monitor_volume_info(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_info *info)
+{
+ info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ info->count = 1;
+ info->value.integer.min = 0;
+ info->value.integer.max = 1;
+ return 0;
+}
+
+static int monitor_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u8 bit = ctl->private_value;
+ int invert = ctl->private_value & (1 << 8);
+
+ value->value.integer.value[0] =
+ !!invert ^ !!(oxygen_read8(chip, OXYGEN_ADC_MONITOR) & bit);
+ return 0;
+}
+
+static int monitor_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u8 bit = ctl->private_value;
+ int invert = ctl->private_value & (1 << 8);
+ u8 oldreg, newreg;
+ int changed;
+
+ spin_lock_irq(&chip->reg_lock);
+ oldreg = oxygen_read8(chip, OXYGEN_ADC_MONITOR);
+ if ((!!value->value.integer.value[0] ^ !!invert) != 0)
+ newreg = oldreg | bit;
+ else
+ newreg = oldreg & ~bit;
+ changed = newreg != oldreg;
+ if (changed)
+ oxygen_write8(chip, OXYGEN_ADC_MONITOR, newreg);
+ spin_unlock_irq(&chip->reg_lock);
+ return changed;
+}
+
+static int ac97_switch_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int codec = (ctl->private_value >> 24) & 1;
+ unsigned int index = ctl->private_value & 0xff;
+ unsigned int bitnr = (ctl->private_value >> 8) & 0xff;
+ int invert = ctl->private_value & (1 << 16);
+ u16 reg;
+
+ mutex_lock(&chip->mutex);
+ reg = oxygen_read_ac97(chip, codec, index);
+ mutex_unlock(&chip->mutex);
+ if (!(reg & (1 << bitnr)) ^ !invert)
+ value->value.integer.value[0] = 1;
+ else
+ value->value.integer.value[0] = 0;
+ return 0;
+}
+
+static void mute_ac97_ctl(struct oxygen *chip, unsigned int control)
+{
+ unsigned int priv_idx;
+ u16 value;
+
+ if (!chip->controls[control])
+ return;
+ priv_idx = chip->controls[control]->private_value & 0xff;
+ value = oxygen_read_ac97(chip, 0, priv_idx);
+ if (!(value & 0x8000)) {
+ oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000);
+ if (chip->model.ac97_switch)
+ chip->model.ac97_switch(chip, priv_idx, 0x8000);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->controls[control]->id);
+ }
+}
+
+static int ac97_switch_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int codec = (ctl->private_value >> 24) & 1;
+ unsigned int index = ctl->private_value & 0xff;
+ unsigned int bitnr = (ctl->private_value >> 8) & 0xff;
+ int invert = ctl->private_value & (1 << 16);
+ u16 oldreg, newreg;
+ int change;
+
+ mutex_lock(&chip->mutex);
+ oldreg = oxygen_read_ac97(chip, codec, index);
+ newreg = oldreg;
+ if (!value->value.integer.value[0] ^ !invert)
+ newreg |= 1 << bitnr;
+ else
+ newreg &= ~(1 << bitnr);
+ change = newreg != oldreg;
+ if (change) {
+ oxygen_write_ac97(chip, codec, index, newreg);
+ if (codec == 0 && chip->model.ac97_switch)
+ chip->model.ac97_switch(chip, index, newreg & 0x8000);
+ if (index == AC97_LINE) {
+ oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS,
+ newreg & 0x8000 ?
+ CM9780_GPO0 : 0, CM9780_GPO0);
+ if (!(newreg & 0x8000)) {
+ mute_ac97_ctl(chip, CONTROL_MIC_CAPTURE_SWITCH);
+ mute_ac97_ctl(chip, CONTROL_CD_CAPTURE_SWITCH);
+ mute_ac97_ctl(chip, CONTROL_AUX_CAPTURE_SWITCH);
+ }
+ } else if ((index == AC97_MIC || index == AC97_CD ||
+ index == AC97_VIDEO || index == AC97_AUX) &&
+ bitnr == 15 && !(newreg & 0x8000)) {
+ mute_ac97_ctl(chip, CONTROL_LINE_CAPTURE_SWITCH);
+ oxygen_write_ac97_masked(chip, 0, CM9780_GPIO_STATUS,
+ CM9780_GPO0, CM9780_GPO0);
+ }
+ }
+ mutex_unlock(&chip->mutex);
+ return change;
+}
+
+static int ac97_volume_info(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_info *info)
+{
+ info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ info->count = 2;
+ info->value.integer.min = 0;
+ info->value.integer.max = 0x1f;
+ return 0;
+}
+
+static int ac97_volume_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int codec = (ctl->private_value >> 24) & 1;
+ unsigned int index = ctl->private_value & 0xff;
+ u16 reg;
+
+ mutex_lock(&chip->mutex);
+ reg = oxygen_read_ac97(chip, codec, index);
+ mutex_unlock(&chip->mutex);
+ value->value.integer.value[0] = 31 - (reg & 0x1f);
+ value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f);
+ return 0;
+}
+
+static int ac97_volume_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int codec = (ctl->private_value >> 24) & 1;
+ unsigned int index = ctl->private_value & 0xff;
+ u16 oldreg, newreg;
+ int change;
+
+ mutex_lock(&chip->mutex);
+ oldreg = oxygen_read_ac97(chip, codec, index);
+ newreg = oldreg;
+ newreg = (newreg & ~0x1f) |
+ (31 - (value->value.integer.value[0] & 0x1f));
+ newreg = (newreg & ~0x1f00) |
+ ((31 - (value->value.integer.value[0] & 0x1f)) << 8);
+ change = newreg != oldreg;
+ if (change)
+ oxygen_write_ac97(chip, codec, index, newreg);
+ mutex_unlock(&chip->mutex);
+ return change;
+}
+
+static int ac97_fp_rec_volume_info(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_info *info)
+{
+ info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ info->count = 2;
+ info->value.integer.min = 0;
+ info->value.integer.max = 7;
+ return 0;
+}
+
+static int ac97_fp_rec_volume_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u16 reg;
+
+ mutex_lock(&chip->mutex);
+ reg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN);
+ mutex_unlock(&chip->mutex);
+ value->value.integer.value[0] = reg & 7;
+ value->value.integer.value[1] = (reg >> 8) & 7;
+ return 0;
+}
+
+static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u16 oldreg, newreg;
+ int change;
+
+ mutex_lock(&chip->mutex);
+ oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN);
+ newreg = oldreg & ~0x0707;
+ newreg = newreg | (value->value.integer.value[0] & 7);
+ newreg = newreg | ((value->value.integer.value[0] & 7) << 8);
+ change = newreg != oldreg;
+ if (change)
+ oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg);
+ mutex_unlock(&chip->mutex);
+ return change;
+}
+
+#define AC97_SWITCH(xname, codec, index, bitnr, invert) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .info = snd_ctl_boolean_mono_info, \
+ .get = ac97_switch_get, \
+ .put = ac97_switch_put, \
+ .private_value = ((codec) << 24) | ((invert) << 16) | \
+ ((bitnr) << 8) | (index), \
+ }
+#define AC97_VOLUME(xname, codec, index) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = ac97_volume_info, \
+ .get = ac97_volume_get, \
+ .put = ac97_volume_put, \
+ .tlv = { .p = ac97_db_scale, }, \
+ .private_value = ((codec) << 24) | (index), \
+ }
+
+static DECLARE_TLV_DB_SCALE(monitor_db_scale, -1000, 1000, 0);
+static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0);
+static DECLARE_TLV_DB_SCALE(ac97_rec_db_scale, 0, 150, 0);
+
+static const struct snd_kcontrol_new controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = dac_volume_info,
+ .get = dac_volume_get,
+ .put = dac_volume_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = dac_mute_get,
+ .put = dac_mute_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Stereo Upmixing",
+ .info = upmix_info,
+ .get = upmix_get,
+ .put = upmix_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
+ .info = snd_ctl_boolean_mono_info,
+ .get = spdif_switch_get,
+ .put = spdif_switch_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .device = 1,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .info = spdif_info,
+ .get = spdif_default_get,
+ .put = spdif_default_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .device = 1,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = spdif_info,
+ .get = spdif_mask_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .device = 1,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM),
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .info = spdif_info,
+ .get = spdif_pcm_get,
+ .put = spdif_pcm_put,
+ },
+};
+
+static const struct snd_kcontrol_new spdif_input_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .device = 1,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = spdif_info,
+ .get = spdif_input_mask_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .device = 1,
+ .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT),
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = spdif_info,
+ .get = spdif_input_default_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("Loopback ", NONE, SWITCH),
+ .info = snd_ctl_boolean_mono_info,
+ .get = spdif_loopback_get,
+ .put = spdif_loopback_put,
+ },
+};
+
+static const struct {
+ unsigned int pcm_dev;
+ struct snd_kcontrol_new controls[2];
+} monitor_controls[] = {
+ {
+ .pcm_dev = CAPTURE_0_FROM_I2S_1,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_A,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_A_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+ {
+ .pcm_dev = CAPTURE_0_FROM_I2S_2,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+ {
+ .pcm_dev = CAPTURE_2_FROM_I2S_2,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Switch",
+ .index = 1,
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Input Monitor Volume",
+ .index = 1,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_B_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+ {
+ .pcm_dev = CAPTURE_1_FROM_SPDIF,
+ .controls = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Input Monitor Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_C,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Input Monitor Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = monitor_volume_info,
+ .get = monitor_get,
+ .put = monitor_put,
+ .private_value = OXYGEN_ADC_MONITOR_C_HALF_VOL
+ | (1 << 8),
+ .tlv = { .p = monitor_db_scale, },
+ },
+ },
+ },
+};
+
+static const struct snd_kcontrol_new ac97_controls[] = {
+ AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC),
+ AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1),
+ AC97_SWITCH("Mic Boost (+20dB)", 0, AC97_MIC, 6, 0),
+ AC97_SWITCH("Line Capture Switch", 0, AC97_LINE, 15, 1),
+ AC97_VOLUME("CD Capture Volume", 0, AC97_CD),
+ AC97_SWITCH("CD Capture Switch", 0, AC97_CD, 15, 1),
+ AC97_VOLUME("Aux Capture Volume", 0, AC97_AUX),
+ AC97_SWITCH("Aux Capture Switch", 0, AC97_AUX, 15, 1),
+};
+
+static const struct snd_kcontrol_new ac97_fp_controls[] = {
+ AC97_VOLUME("Front Panel Playback Volume", 1, AC97_HEADPHONE),
+ AC97_SWITCH("Front Panel Playback Switch", 1, AC97_HEADPHONE, 15, 1),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Front Panel Capture Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = ac97_fp_rec_volume_info,
+ .get = ac97_fp_rec_volume_get,
+ .put = ac97_fp_rec_volume_put,
+ .tlv = { .p = ac97_rec_db_scale, },
+ },
+ AC97_SWITCH("Front Panel Capture Switch", 1, AC97_REC_GAIN, 15, 1),
+};
+
+static void oxygen_any_ctl_free(struct snd_kcontrol *ctl)
+{
+ struct oxygen *chip = ctl->private_data;
+ unsigned int i;
+
+ /* I'm too lazy to write a function for each control :-) */
+ for (i = 0; i < ARRAY_SIZE(chip->controls); ++i)
+ chip->controls[i] = NULL;
+}
+
+static int add_controls(struct oxygen *chip,
+ const struct snd_kcontrol_new controls[],
+ unsigned int count)
+{
+ static const char *const known_ctl_names[CONTROL_COUNT] = {
+ [CONTROL_SPDIF_PCM] =
+ SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM),
+ [CONTROL_SPDIF_INPUT_BITS] =
+ SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT),
+ [CONTROL_MIC_CAPTURE_SWITCH] = "Mic Capture Switch",
+ [CONTROL_LINE_CAPTURE_SWITCH] = "Line Capture Switch",
+ [CONTROL_CD_CAPTURE_SWITCH] = "CD Capture Switch",
+ [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch",
+ };
+ unsigned int i, j;
+ struct snd_kcontrol_new template;
+ struct snd_kcontrol *ctl;
+ int err;
+
+ for (i = 0; i < count; ++i) {
+ template = controls[i];
+ if (chip->model.control_filter) {
+ err = chip->model.control_filter(&template);
+ if (err < 0)
+ return err;
+ if (err == 1)
+ continue;
+ }
+ if (!strcmp(template.name, "Master Playback Volume") &&
+ chip->model.dac_tlv) {
+ template.tlv.p = chip->model.dac_tlv;
+ template.access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ }
+ ctl = snd_ctl_new1(&template, chip);
+ if (!ctl)
+ return -ENOMEM;
+ err = snd_ctl_add(chip->card, ctl);
+ if (err < 0)
+ return err;
+ for (j = 0; j < CONTROL_COUNT; ++j)
+ if (!strcmp(ctl->id.name, known_ctl_names[j])) {
+ chip->controls[j] = ctl;
+ ctl->private_free = oxygen_any_ctl_free;
+ }
+ }
+ return 0;
+}
+
+int oxygen_mixer_init(struct oxygen *chip)
+{
+ unsigned int i;
+ int err;
+
+ err = add_controls(chip, controls, ARRAY_SIZE(controls));
+ if (err < 0)
+ return err;
+ if (chip->model.device_config & CAPTURE_1_FROM_SPDIF) {
+ err = add_controls(chip, spdif_input_controls,
+ ARRAY_SIZE(spdif_input_controls));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < ARRAY_SIZE(monitor_controls); ++i) {
+ if (!(chip->model.device_config & monitor_controls[i].pcm_dev))
+ continue;
+ err = add_controls(chip, monitor_controls[i].controls,
+ ARRAY_SIZE(monitor_controls[i].controls));
+ if (err < 0)
+ return err;
+ }
+ if (chip->has_ac97_0) {
+ err = add_controls(chip, ac97_controls,
+ ARRAY_SIZE(ac97_controls));
+ if (err < 0)
+ return err;
+ }
+ if (chip->has_ac97_1) {
+ err = add_controls(chip, ac97_fp_controls,
+ ARRAY_SIZE(ac97_fp_controls));
+ if (err < 0)
+ return err;
+ }
+ return chip->model.mixer_init ? chip->model.mixer_init(chip) : 0;
+}
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
new file mode 100644
index 0000000..c262049
--- /dev/null
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -0,0 +1,746 @@
+/*
+ * C-Media CMI8788 driver - PCM code
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/pci.h>
+#include <sound/control.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "oxygen.h"
+
+/* most DMA channels have a 16-bit counter for 32-bit words */
+#define BUFFER_BYTES_MAX ((1 << 16) * 4)
+/* the multichannel DMA channel has a 24-bit counter */
+#define BUFFER_BYTES_MAX_MULTICH ((1 << 24) * 4)
+
+#define PERIOD_BYTES_MIN 64
+
+#define DEFAULT_BUFFER_BYTES (BUFFER_BYTES_MAX / 2)
+#define DEFAULT_BUFFER_BYTES_MULTICH (1024 * 1024)
+
+static const struct snd_pcm_hardware oxygen_stereo_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000,
+ .rate_min = 32000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = BUFFER_BYTES_MAX,
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = BUFFER_BYTES_MAX / 2,
+ .periods_min = 2,
+ .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN,
+};
+static const struct snd_pcm_hardware oxygen_multichannel_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000,
+ .rate_min = 32000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 8,
+ .buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH,
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2,
+ .periods_min = 2,
+ .periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN,
+};
+static const struct snd_pcm_hardware oxygen_ac97_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = BUFFER_BYTES_MAX,
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = BUFFER_BYTES_MAX / 2,
+ .periods_min = 2,
+ .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN,
+};
+
+static const struct snd_pcm_hardware *const oxygen_hardware[PCM_COUNT] = {
+ [PCM_A] = &oxygen_stereo_hardware,
+ [PCM_B] = &oxygen_stereo_hardware,
+ [PCM_C] = &oxygen_stereo_hardware,
+ [PCM_SPDIF] = &oxygen_stereo_hardware,
+ [PCM_MULTICH] = &oxygen_multichannel_hardware,
+ [PCM_AC97] = &oxygen_ac97_hardware,
+};
+
+static inline unsigned int
+oxygen_substream_channel(struct snd_pcm_substream *substream)
+{
+ return (unsigned int)(uintptr_t)substream->runtime->private_data;
+}
+
+static int oxygen_open(struct snd_pcm_substream *substream,
+ unsigned int channel)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ runtime->private_data = (void *)(uintptr_t)channel;
+ if (channel == PCM_B && chip->has_ac97_1 &&
+ (chip->model.device_config & CAPTURE_2_FROM_AC97_1))
+ runtime->hw = oxygen_ac97_hardware;
+ else
+ runtime->hw = *oxygen_hardware[channel];
+ switch (channel) {
+ case PCM_C:
+ runtime->hw.rates &= ~(SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_64000);
+ runtime->hw.rate_min = 44100;
+ break;
+ case PCM_MULTICH:
+ runtime->hw.channels_max = chip->model.dac_channels;
+ break;
+ }
+ if (chip->model.pcm_hardware_filter)
+ chip->model.pcm_hardware_filter(channel, &runtime->hw);
+ err = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (err < 0)
+ return err;
+ if (runtime->hw.formats & SNDRV_PCM_FMTBIT_S32_LE) {
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+ }
+ if (runtime->hw.channels_max > 2) {
+ err = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ 2);
+ if (err < 0)
+ return err;
+ }
+ if (channel == PCM_MULTICH) {
+ err = snd_pcm_hw_constraint_minmax
+ (runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 0, 8192000);
+ if (err < 0)
+ return err;
+ }
+ snd_pcm_set_sync(substream);
+ chip->streams[channel] = substream;
+
+ mutex_lock(&chip->mutex);
+ chip->pcm_active |= 1 << channel;
+ if (channel == PCM_SPDIF) {
+ chip->spdif_pcm_bits = chip->spdif_bits;
+ chip->controls[CONTROL_SPDIF_PCM]->vd[0].access &=
+ ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ &chip->controls[CONTROL_SPDIF_PCM]->id);
+ }
+ mutex_unlock(&chip->mutex);
+
+ return 0;
+}
+
+static int oxygen_rec_a_open(struct snd_pcm_substream *substream)
+{
+ return oxygen_open(substream, PCM_A);
+}
+
+static int oxygen_rec_b_open(struct snd_pcm_substream *substream)
+{
+ return oxygen_open(substream, PCM_B);
+}
+
+static int oxygen_rec_c_open(struct snd_pcm_substream *substream)
+{
+ return oxygen_open(substream, PCM_C);
+}
+
+static int oxygen_spdif_open(struct snd_pcm_substream *substream)
+{
+ return oxygen_open(substream, PCM_SPDIF);
+}
+
+static int oxygen_multich_open(struct snd_pcm_substream *substream)
+{
+ return oxygen_open(substream, PCM_MULTICH);
+}
+
+static int oxygen_ac97_open(struct snd_pcm_substream *substream)
+{
+ return oxygen_open(substream, PCM_AC97);
+}
+
+static int oxygen_close(struct snd_pcm_substream *substream)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ unsigned int channel = oxygen_substream_channel(substream);
+
+ mutex_lock(&chip->mutex);
+ chip->pcm_active &= ~(1 << channel);
+ if (channel == PCM_SPDIF) {
+ chip->controls[CONTROL_SPDIF_PCM]->vd[0].access |=
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ &chip->controls[CONTROL_SPDIF_PCM]->id);
+ }
+ if (channel == PCM_SPDIF || channel == PCM_MULTICH)
+ oxygen_update_spdif_source(chip);
+ mutex_unlock(&chip->mutex);
+
+ chip->streams[channel] = NULL;
+ return 0;
+}
+
+static unsigned int oxygen_format(struct snd_pcm_hw_params *hw_params)
+{
+ if (params_format(hw_params) == SNDRV_PCM_FORMAT_S32_LE)
+ return OXYGEN_FORMAT_24;
+ else
+ return OXYGEN_FORMAT_16;
+}
+
+static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params)
+{
+ switch (params_rate(hw_params)) {
+ case 32000:
+ return OXYGEN_RATE_32000;
+ case 44100:
+ return OXYGEN_RATE_44100;
+ default: /* 48000 */
+ return OXYGEN_RATE_48000;
+ case 64000:
+ return OXYGEN_RATE_64000;
+ case 88200:
+ return OXYGEN_RATE_88200;
+ case 96000:
+ return OXYGEN_RATE_96000;
+ case 176400:
+ return OXYGEN_RATE_176400;
+ case 192000:
+ return OXYGEN_RATE_192000;
+ }
+}
+
+static unsigned int oxygen_i2s_mclk(struct snd_pcm_hw_params *hw_params)
+{
+ if (params_rate(hw_params) <= 96000)
+ return OXYGEN_I2S_MCLK_256;
+ else
+ return OXYGEN_I2S_MCLK_128;
+}
+
+static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params)
+{
+ if (params_format(hw_params) == SNDRV_PCM_FORMAT_S32_LE)
+ return OXYGEN_I2S_BITS_24;
+ else
+ return OXYGEN_I2S_BITS_16;
+}
+
+static unsigned int oxygen_play_channels(struct snd_pcm_hw_params *hw_params)
+{
+ switch (params_channels(hw_params)) {
+ default: /* 2 */
+ return OXYGEN_PLAY_CHANNELS_2;
+ case 4:
+ return OXYGEN_PLAY_CHANNELS_4;
+ case 6:
+ return OXYGEN_PLAY_CHANNELS_6;
+ case 8:
+ return OXYGEN_PLAY_CHANNELS_8;
+ }
+}
+
+static const unsigned int channel_base_registers[PCM_COUNT] = {
+ [PCM_A] = OXYGEN_DMA_A_ADDRESS,
+ [PCM_B] = OXYGEN_DMA_B_ADDRESS,
+ [PCM_C] = OXYGEN_DMA_C_ADDRESS,
+ [PCM_SPDIF] = OXYGEN_DMA_SPDIF_ADDRESS,
+ [PCM_MULTICH] = OXYGEN_DMA_MULTICH_ADDRESS,
+ [PCM_AC97] = OXYGEN_DMA_AC97_ADDRESS,
+};
+
+static int oxygen_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ unsigned int channel = oxygen_substream_channel(substream);
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ oxygen_write32(chip, channel_base_registers[channel],
+ (u32)substream->runtime->dma_addr);
+ if (channel == PCM_MULTICH) {
+ oxygen_write32(chip, OXYGEN_DMA_MULTICH_COUNT,
+ params_buffer_bytes(hw_params) / 4 - 1);
+ oxygen_write32(chip, OXYGEN_DMA_MULTICH_TCOUNT,
+ params_period_bytes(hw_params) / 4 - 1);
+ } else {
+ oxygen_write16(chip, channel_base_registers[channel] + 4,
+ params_buffer_bytes(hw_params) / 4 - 1);
+ oxygen_write16(chip, channel_base_registers[channel] + 6,
+ params_period_bytes(hw_params) / 4 - 1);
+ }
+ return 0;
+}
+
+static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ err = oxygen_hw_params(substream, hw_params);
+ if (err < 0)
+ return err;
+
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_write8_masked(chip, OXYGEN_REC_FORMAT,
+ oxygen_format(hw_params) << OXYGEN_REC_FORMAT_A_SHIFT,
+ OXYGEN_REC_FORMAT_A_MASK);
+ oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT,
+ oxygen_rate(hw_params) |
+ oxygen_i2s_mclk(hw_params) |
+ chip->model.adc_i2s_format |
+ oxygen_i2s_bits(hw_params),
+ OXYGEN_I2S_RATE_MASK |
+ OXYGEN_I2S_FORMAT_MASK |
+ OXYGEN_I2S_MCLK_MASK |
+ OXYGEN_I2S_BITS_MASK);
+ spin_unlock_irq(&chip->reg_lock);
+
+ mutex_lock(&chip->mutex);
+ chip->model.set_adc_params(chip, hw_params);
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ int is_ac97;
+ int err;
+
+ err = oxygen_hw_params(substream, hw_params);
+ if (err < 0)
+ return err;
+
+ is_ac97 = chip->has_ac97_1 &&
+ (chip->model.device_config & CAPTURE_2_FROM_AC97_1);
+
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_write8_masked(chip, OXYGEN_REC_FORMAT,
+ oxygen_format(hw_params) << OXYGEN_REC_FORMAT_B_SHIFT,
+ OXYGEN_REC_FORMAT_B_MASK);
+ if (!is_ac97)
+ oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT,
+ oxygen_rate(hw_params) |
+ oxygen_i2s_mclk(hw_params) |
+ chip->model.adc_i2s_format |
+ oxygen_i2s_bits(hw_params),
+ OXYGEN_I2S_RATE_MASK |
+ OXYGEN_I2S_FORMAT_MASK |
+ OXYGEN_I2S_MCLK_MASK |
+ OXYGEN_I2S_BITS_MASK);
+ spin_unlock_irq(&chip->reg_lock);
+
+ if (!is_ac97) {
+ mutex_lock(&chip->mutex);
+ chip->model.set_adc_params(chip, hw_params);
+ mutex_unlock(&chip->mutex);
+ }
+ return 0;
+}
+
+static int oxygen_rec_c_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ err = oxygen_hw_params(substream, hw_params);
+ if (err < 0)
+ return err;
+
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_write8_masked(chip, OXYGEN_REC_FORMAT,
+ oxygen_format(hw_params) << OXYGEN_REC_FORMAT_C_SHIFT,
+ OXYGEN_REC_FORMAT_C_MASK);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int oxygen_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ err = oxygen_hw_params(substream, hw_params);
+ if (err < 0)
+ return err;
+
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_OUT_ENABLE);
+ oxygen_write8_masked(chip, OXYGEN_PLAY_FORMAT,
+ oxygen_format(hw_params) << OXYGEN_SPDIF_FORMAT_SHIFT,
+ OXYGEN_SPDIF_FORMAT_MASK);
+ oxygen_write32_masked(chip, OXYGEN_SPDIF_CONTROL,
+ oxygen_rate(hw_params) << OXYGEN_SPDIF_OUT_RATE_SHIFT,
+ OXYGEN_SPDIF_OUT_RATE_MASK);
+ oxygen_update_spdif_source(chip);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int oxygen_multich_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ int err;
+
+ err = oxygen_hw_params(substream, hw_params);
+ if (err < 0)
+ return err;
+
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_write8_masked(chip, OXYGEN_PLAY_CHANNELS,
+ oxygen_play_channels(hw_params),
+ OXYGEN_PLAY_CHANNELS_MASK);
+ oxygen_write8_masked(chip, OXYGEN_PLAY_FORMAT,
+ oxygen_format(hw_params) << OXYGEN_MULTICH_FORMAT_SHIFT,
+ OXYGEN_MULTICH_FORMAT_MASK);
+ oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT,
+ oxygen_rate(hw_params) |
+ chip->model.dac_i2s_format |
+ oxygen_i2s_bits(hw_params),
+ OXYGEN_I2S_RATE_MASK |
+ OXYGEN_I2S_FORMAT_MASK |
+ OXYGEN_I2S_BITS_MASK);
+ oxygen_update_dac_routing(chip);
+ oxygen_update_spdif_source(chip);
+ spin_unlock_irq(&chip->reg_lock);
+
+ mutex_lock(&chip->mutex);
+ chip->model.set_dac_params(chip, hw_params);
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static int oxygen_hw_free(struct snd_pcm_substream *substream)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ unsigned int channel = oxygen_substream_channel(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->interrupt_mask &= ~(1 << channel);
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
+ spin_unlock_irq(&chip->reg_lock);
+
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int oxygen_spdif_hw_free(struct snd_pcm_substream *substream)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL,
+ OXYGEN_SPDIF_OUT_ENABLE);
+ spin_unlock_irq(&chip->reg_lock);
+ return oxygen_hw_free(substream);
+}
+
+static int oxygen_prepare(struct snd_pcm_substream *substream)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ unsigned int channel = oxygen_substream_channel(substream);
+ unsigned int channel_mask = 1 << channel;
+
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_set_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask);
+ oxygen_clear_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask);
+
+ chip->interrupt_mask |= channel_mask;
+ oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+ unsigned int mask = 0;
+ int pausing;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ pausing = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pausing = 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) == chip) {
+ mask |= 1 << oxygen_substream_channel(s);
+ snd_pcm_trigger_done(s, substream);
+ }
+ }
+
+ spin_lock(&chip->reg_lock);
+ if (!pausing) {
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ chip->pcm_running |= mask;
+ else
+ chip->pcm_running &= ~mask;
+ oxygen_write8(chip, OXYGEN_DMA_STATUS, chip->pcm_running);
+ } else {
+ if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH)
+ oxygen_set_bits8(chip, OXYGEN_DMA_PAUSE, mask);
+ else
+ oxygen_clear_bits8(chip, OXYGEN_DMA_PAUSE, mask);
+ }
+ spin_unlock(&chip->reg_lock);
+ return 0;
+}
+
+static snd_pcm_uframes_t oxygen_pointer(struct snd_pcm_substream *substream)
+{
+ struct oxygen *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int channel = oxygen_substream_channel(substream);
+ u32 curr_addr;
+
+ /* no spinlock, this read should be atomic */
+ curr_addr = oxygen_read32(chip, channel_base_registers[channel]);
+ return bytes_to_frames(runtime, curr_addr - (u32)runtime->dma_addr);
+}
+
+static struct snd_pcm_ops oxygen_rec_a_ops = {
+ .open = oxygen_rec_a_open,
+ .close = oxygen_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = oxygen_rec_a_hw_params,
+ .hw_free = oxygen_hw_free,
+ .prepare = oxygen_prepare,
+ .trigger = oxygen_trigger,
+ .pointer = oxygen_pointer,
+};
+
+static struct snd_pcm_ops oxygen_rec_b_ops = {
+ .open = oxygen_rec_b_open,
+ .close = oxygen_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = oxygen_rec_b_hw_params,
+ .hw_free = oxygen_hw_free,
+ .prepare = oxygen_prepare,
+ .trigger = oxygen_trigger,
+ .pointer = oxygen_pointer,
+};
+
+static struct snd_pcm_ops oxygen_rec_c_ops = {
+ .open = oxygen_rec_c_open,
+ .close = oxygen_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = oxygen_rec_c_hw_params,
+ .hw_free = oxygen_hw_free,
+ .prepare = oxygen_prepare,
+ .trigger = oxygen_trigger,
+ .pointer = oxygen_pointer,
+};
+
+static struct snd_pcm_ops oxygen_spdif_ops = {
+ .open = oxygen_spdif_open,
+ .close = oxygen_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = oxygen_spdif_hw_params,
+ .hw_free = oxygen_spdif_hw_free,
+ .prepare = oxygen_prepare,
+ .trigger = oxygen_trigger,
+ .pointer = oxygen_pointer,
+};
+
+static struct snd_pcm_ops oxygen_multich_ops = {
+ .open = oxygen_multich_open,
+ .close = oxygen_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = oxygen_multich_hw_params,
+ .hw_free = oxygen_hw_free,
+ .prepare = oxygen_prepare,
+ .trigger = oxygen_trigger,
+ .pointer = oxygen_pointer,
+};
+
+static struct snd_pcm_ops oxygen_ac97_ops = {
+ .open = oxygen_ac97_open,
+ .close = oxygen_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = oxygen_hw_params,
+ .hw_free = oxygen_hw_free,
+ .prepare = oxygen_prepare,
+ .trigger = oxygen_trigger,
+ .pointer = oxygen_pointer,
+};
+
+static void oxygen_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+int oxygen_pcm_init(struct oxygen *chip)
+{
+ struct snd_pcm *pcm;
+ int outs, ins;
+ int err;
+
+ outs = !!(chip->model.device_config & PLAYBACK_0_TO_I2S);
+ ins = !!(chip->model.device_config & (CAPTURE_0_FROM_I2S_1 |
+ CAPTURE_0_FROM_I2S_2));
+ if (outs | ins) {
+ err = snd_pcm_new(chip->card, "Multichannel",
+ 0, outs, ins, &pcm);
+ if (err < 0)
+ return err;
+ if (outs)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &oxygen_multich_ops);
+ if (chip->model.device_config & CAPTURE_0_FROM_I2S_1)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &oxygen_rec_a_ops);
+ else if (chip->model.device_config & CAPTURE_0_FROM_I2S_2)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &oxygen_rec_b_ops);
+ pcm->private_data = chip;
+ pcm->private_free = oxygen_pcm_free;
+ strcpy(pcm->name, "Multichannel");
+ if (outs)
+ snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ DEFAULT_BUFFER_BYTES_MULTICH,
+ BUFFER_BYTES_MAX_MULTICH);
+ if (ins)
+ snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ DEFAULT_BUFFER_BYTES,
+ BUFFER_BYTES_MAX);
+ }
+
+ outs = !!(chip->model.device_config & PLAYBACK_1_TO_SPDIF);
+ ins = !!(chip->model.device_config & CAPTURE_1_FROM_SPDIF);
+ if (outs | ins) {
+ err = snd_pcm_new(chip->card, "Digital", 1, outs, ins, &pcm);
+ if (err < 0)
+ return err;
+ if (outs)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &oxygen_spdif_ops);
+ if (ins)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &oxygen_rec_c_ops);
+ pcm->private_data = chip;
+ pcm->private_free = oxygen_pcm_free;
+ strcpy(pcm->name, "Digital");
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ DEFAULT_BUFFER_BYTES,
+ BUFFER_BYTES_MAX);
+ }
+
+ if (chip->has_ac97_1) {
+ outs = !!(chip->model.device_config & PLAYBACK_2_TO_AC97_1);
+ ins = !!(chip->model.device_config & CAPTURE_2_FROM_AC97_1);
+ } else {
+ outs = 0;
+ ins = !!(chip->model.device_config & CAPTURE_2_FROM_I2S_2);
+ }
+ if (outs | ins) {
+ err = snd_pcm_new(chip->card, outs ? "AC97" : "Analog2",
+ 2, outs, ins, &pcm);
+ if (err < 0)
+ return err;
+ if (outs) {
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &oxygen_ac97_ops);
+ oxygen_write8_masked(chip, OXYGEN_REC_ROUTING,
+ OXYGEN_REC_B_ROUTE_AC97_1,
+ OXYGEN_REC_B_ROUTE_MASK);
+ }
+ if (ins)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &oxygen_rec_b_ops);
+ pcm->private_data = chip;
+ pcm->private_free = oxygen_pcm_free;
+ strcpy(pcm->name, outs ? "Front Panel" : "Analog 2");
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ DEFAULT_BUFFER_BYTES,
+ BUFFER_BYTES_MAX);
+ }
+ return 0;
+}
diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h
new file mode 100644
index 0000000..72de159
--- /dev/null
+++ b/sound/pci/oxygen/oxygen_regs.h
@@ -0,0 +1,453 @@
+#ifndef OXYGEN_REGS_H_INCLUDED
+#define OXYGEN_REGS_H_INCLUDED
+
+/* recording channel A */
+#define OXYGEN_DMA_A_ADDRESS 0x00 /* 32-bit base address */
+#define OXYGEN_DMA_A_COUNT 0x04 /* buffer counter (dwords) */
+#define OXYGEN_DMA_A_TCOUNT 0x06 /* interrupt counter (dwords) */
+
+/* recording channel B */
+#define OXYGEN_DMA_B_ADDRESS 0x08
+#define OXYGEN_DMA_B_COUNT 0x0c
+#define OXYGEN_DMA_B_TCOUNT 0x0e
+
+/* recording channel C */
+#define OXYGEN_DMA_C_ADDRESS 0x10
+#define OXYGEN_DMA_C_COUNT 0x14
+#define OXYGEN_DMA_C_TCOUNT 0x16
+
+/* SPDIF playback channel */
+#define OXYGEN_DMA_SPDIF_ADDRESS 0x18
+#define OXYGEN_DMA_SPDIF_COUNT 0x1c
+#define OXYGEN_DMA_SPDIF_TCOUNT 0x1e
+
+/* multichannel playback channel */
+#define OXYGEN_DMA_MULTICH_ADDRESS 0x20
+#define OXYGEN_DMA_MULTICH_COUNT 0x24 /* 24 bits */
+#define OXYGEN_DMA_MULTICH_TCOUNT 0x28 /* 24 bits */
+
+/* AC'97 (front panel) playback channel */
+#define OXYGEN_DMA_AC97_ADDRESS 0x30
+#define OXYGEN_DMA_AC97_COUNT 0x34
+#define OXYGEN_DMA_AC97_TCOUNT 0x36
+
+/* all registers 0x00..0x36 return current position on read */
+
+#define OXYGEN_DMA_STATUS 0x40 /* 1 = running, 0 = stop */
+#define OXYGEN_CHANNEL_A 0x01
+#define OXYGEN_CHANNEL_B 0x02
+#define OXYGEN_CHANNEL_C 0x04
+#define OXYGEN_CHANNEL_SPDIF 0x08
+#define OXYGEN_CHANNEL_MULTICH 0x10
+#define OXYGEN_CHANNEL_AC97 0x20
+
+#define OXYGEN_DMA_PAUSE 0x41 /* 1 = pause */
+/* OXYGEN_CHANNEL_* */
+
+#define OXYGEN_DMA_RESET 0x42
+/* OXYGEN_CHANNEL_* */
+
+#define OXYGEN_PLAY_CHANNELS 0x43
+#define OXYGEN_PLAY_CHANNELS_MASK 0x03
+#define OXYGEN_PLAY_CHANNELS_2 0x00
+#define OXYGEN_PLAY_CHANNELS_4 0x01
+#define OXYGEN_PLAY_CHANNELS_6 0x02
+#define OXYGEN_PLAY_CHANNELS_8 0x03
+#define OXYGEN_DMA_A_BURST_MASK 0x04
+#define OXYGEN_DMA_A_BURST_8 0x00 /* dwords */
+#define OXYGEN_DMA_A_BURST_16 0x04
+#define OXYGEN_DMA_MULTICH_BURST_MASK 0x08
+#define OXYGEN_DMA_MULTICH_BURST_8 0x00
+#define OXYGEN_DMA_MULTICH_BURST_16 0x08
+
+#define OXYGEN_INTERRUPT_MASK 0x44
+/* OXYGEN_CHANNEL_* */
+#define OXYGEN_INT_SPDIF_IN_DETECT 0x0100
+#define OXYGEN_INT_MCU 0x0200
+#define OXYGEN_INT_2WIRE 0x0400
+#define OXYGEN_INT_GPIO 0x0800
+#define OXYGEN_INT_MCB 0x2000
+#define OXYGEN_INT_AC97 0x4000
+
+#define OXYGEN_INTERRUPT_STATUS 0x46
+/* OXYGEN_CHANNEL_* amd OXYGEN_INT_* */
+#define OXYGEN_INT_MIDI 0x1000
+
+#define OXYGEN_MISC 0x48
+#define OXYGEN_MISC_WRITE_PCI_SUBID 0x01
+#define OXYGEN_MISC_LATENCY_3F 0x02
+#define OXYGEN_MISC_REC_C_FROM_SPDIF 0x04
+#define OXYGEN_MISC_REC_B_FROM_AC97 0x08
+#define OXYGEN_MISC_REC_A_FROM_MULTICH 0x10
+#define OXYGEN_MISC_PCI_MEM_W_1_CLOCK 0x20
+#define OXYGEN_MISC_MIDI 0x40
+#define OXYGEN_MISC_CRYSTAL_MASK 0x80
+#define OXYGEN_MISC_CRYSTAL_24576 0x00
+#define OXYGEN_MISC_CRYSTAL_27 0x80 /* MHz */
+
+#define OXYGEN_REC_FORMAT 0x4a
+#define OXYGEN_REC_FORMAT_A_MASK 0x03
+#define OXYGEN_REC_FORMAT_A_SHIFT 0
+#define OXYGEN_REC_FORMAT_B_MASK 0x0c
+#define OXYGEN_REC_FORMAT_B_SHIFT 2
+#define OXYGEN_REC_FORMAT_C_MASK 0x30
+#define OXYGEN_REC_FORMAT_C_SHIFT 4
+#define OXYGEN_FORMAT_16 0x00
+#define OXYGEN_FORMAT_24 0x01
+#define OXYGEN_FORMAT_32 0x02
+
+#define OXYGEN_PLAY_FORMAT 0x4b
+#define OXYGEN_SPDIF_FORMAT_MASK 0x03
+#define OXYGEN_SPDIF_FORMAT_SHIFT 0
+#define OXYGEN_MULTICH_FORMAT_MASK 0x0c
+#define OXYGEN_MULTICH_FORMAT_SHIFT 2
+/* OXYGEN_FORMAT_* */
+
+#define OXYGEN_REC_CHANNELS 0x4c
+#define OXYGEN_REC_CHANNELS_MASK 0x07
+#define OXYGEN_REC_CHANNELS_2_2_2 0x00 /* DMA A, B, C */
+#define OXYGEN_REC_CHANNELS_4_2_2 0x01
+#define OXYGEN_REC_CHANNELS_6_0_2 0x02
+#define OXYGEN_REC_CHANNELS_6_2_0 0x03
+#define OXYGEN_REC_CHANNELS_8_0_0 0x04
+
+#define OXYGEN_FUNCTION 0x50
+#define OXYGEN_FUNCTION_CLOCK_MASK 0x01
+#define OXYGEN_FUNCTION_CLOCK_PLL 0x00
+#define OXYGEN_FUNCTION_CLOCK_CRYSTAL 0x01
+#define OXYGEN_FUNCTION_RESET_CODEC 0x02
+#define OXYGEN_FUNCTION_RESET_POL 0x04
+#define OXYGEN_FUNCTION_PWDN 0x08
+#define OXYGEN_FUNCTION_PWDN_EN 0x10
+#define OXYGEN_FUNCTION_PWDN_POL 0x20
+#define OXYGEN_FUNCTION_2WIRE_SPI_MASK 0x40
+#define OXYGEN_FUNCTION_SPI 0x00
+#define OXYGEN_FUNCTION_2WIRE 0x40
+#define OXYGEN_FUNCTION_ENABLE_SPI_4_5 0x80 /* 0 = EEPROM */
+
+#define OXYGEN_I2S_MULTICH_FORMAT 0x60
+#define OXYGEN_I2S_RATE_MASK 0x0007 /* LRCK */
+#define OXYGEN_RATE_32000 0x0000
+#define OXYGEN_RATE_44100 0x0001
+#define OXYGEN_RATE_48000 0x0002
+#define OXYGEN_RATE_64000 0x0003
+#define OXYGEN_RATE_88200 0x0004
+#define OXYGEN_RATE_96000 0x0005
+#define OXYGEN_RATE_176400 0x0006
+#define OXYGEN_RATE_192000 0x0007
+#define OXYGEN_I2S_FORMAT_MASK 0x0008
+#define OXYGEN_I2S_FORMAT_I2S 0x0000
+#define OXYGEN_I2S_FORMAT_LJUST 0x0008
+#define OXYGEN_I2S_MCLK_MASK 0x0030 /* MCLK/LRCK */
+#define OXYGEN_I2S_MCLK_128 0x0000
+#define OXYGEN_I2S_MCLK_256 0x0010
+#define OXYGEN_I2S_MCLK_512 0x0020
+#define OXYGEN_I2S_BITS_MASK 0x00c0
+#define OXYGEN_I2S_BITS_16 0x0000
+#define OXYGEN_I2S_BITS_20 0x0040
+#define OXYGEN_I2S_BITS_24 0x0080
+#define OXYGEN_I2S_BITS_32 0x00c0
+#define OXYGEN_I2S_MASTER 0x0100
+#define OXYGEN_I2S_BCLK_MASK 0x0600 /* BCLK/LRCK */
+#define OXYGEN_I2S_BCLK_64 0x0000
+#define OXYGEN_I2S_BCLK_128 0x0200
+#define OXYGEN_I2S_BCLK_256 0x0400
+#define OXYGEN_I2S_MUTE_MCLK 0x0800
+
+#define OXYGEN_I2S_A_FORMAT 0x62
+#define OXYGEN_I2S_B_FORMAT 0x64
+#define OXYGEN_I2S_C_FORMAT 0x66
+/* like OXYGEN_I2S_MULTICH_FORMAT */
+
+#define OXYGEN_SPDIF_CONTROL 0x70
+#define OXYGEN_SPDIF_OUT_ENABLE 0x00000002
+#define OXYGEN_SPDIF_LOOPBACK 0x00000004 /* in to out */
+#define OXYGEN_SPDIF_SENSE_MASK 0x00000008
+#define OXYGEN_SPDIF_LOCK_MASK 0x00000010
+#define OXYGEN_SPDIF_RATE_MASK 0x00000020
+#define OXYGEN_SPDIF_SPDVALID 0x00000040
+#define OXYGEN_SPDIF_SENSE_PAR 0x00000200
+#define OXYGEN_SPDIF_LOCK_PAR 0x00000400
+#define OXYGEN_SPDIF_SENSE_STATUS 0x00000800
+#define OXYGEN_SPDIF_LOCK_STATUS 0x00001000
+#define OXYGEN_SPDIF_SENSE_INT 0x00002000 /* r/wc */
+#define OXYGEN_SPDIF_LOCK_INT 0x00004000 /* r/wc */
+#define OXYGEN_SPDIF_RATE_INT 0x00008000 /* r/wc */
+#define OXYGEN_SPDIF_IN_CLOCK_MASK 0x00010000
+#define OXYGEN_SPDIF_IN_CLOCK_96 0x00000000 /* <= 96 kHz */
+#define OXYGEN_SPDIF_IN_CLOCK_192 0x00010000 /* > 96 kHz */
+#define OXYGEN_SPDIF_OUT_RATE_MASK 0x07000000
+#define OXYGEN_SPDIF_OUT_RATE_SHIFT 24
+/* OXYGEN_RATE_* << OXYGEN_SPDIF_OUT_RATE_SHIFT */
+
+#define OXYGEN_SPDIF_OUTPUT_BITS 0x74
+#define OXYGEN_SPDIF_NONAUDIO 0x00000002
+#define OXYGEN_SPDIF_C 0x00000004
+#define OXYGEN_SPDIF_PREEMPHASIS 0x00000008
+#define OXYGEN_SPDIF_CATEGORY_MASK 0x000007f0
+#define OXYGEN_SPDIF_CATEGORY_SHIFT 4
+#define OXYGEN_SPDIF_ORIGINAL 0x00000800
+#define OXYGEN_SPDIF_CS_RATE_MASK 0x0000f000
+#define OXYGEN_SPDIF_CS_RATE_SHIFT 12
+#define OXYGEN_SPDIF_V 0x00010000 /* 0 = valid */
+
+#define OXYGEN_SPDIF_INPUT_BITS 0x78
+/* 32 bits, IEC958_AES_* */
+
+#define OXYGEN_EEPROM_CONTROL 0x80
+#define OXYGEN_EEPROM_ADDRESS_MASK 0x7f
+#define OXYGEN_EEPROM_DIR_MASK 0x80
+#define OXYGEN_EEPROM_DIR_READ 0x00
+#define OXYGEN_EEPROM_DIR_WRITE 0x80
+
+#define OXYGEN_EEPROM_STATUS 0x81
+#define OXYGEN_EEPROM_VALID 0x40
+#define OXYGEN_EEPROM_BUSY 0x80
+
+#define OXYGEN_EEPROM_DATA 0x82 /* 16 bits */
+
+#define OXYGEN_2WIRE_CONTROL 0x90
+#define OXYGEN_2WIRE_DIR_MASK 0x01
+#define OXYGEN_2WIRE_DIR_WRITE 0x00
+#define OXYGEN_2WIRE_DIR_READ 0x01
+#define OXYGEN_2WIRE_ADDRESS_MASK 0xfe /* slave device address */
+#define OXYGEN_2WIRE_ADDRESS_SHIFT 1
+
+#define OXYGEN_2WIRE_MAP 0x91 /* address, 8 bits */
+#define OXYGEN_2WIRE_DATA 0x92 /* data, 16 bits */
+
+#define OXYGEN_2WIRE_BUS_STATUS 0x94
+#define OXYGEN_2WIRE_BUSY 0x0001
+#define OXYGEN_2WIRE_LENGTH_MASK 0x0002
+#define OXYGEN_2WIRE_LENGTH_8 0x0000
+#define OXYGEN_2WIRE_LENGTH_16 0x0002
+#define OXYGEN_2WIRE_MANUAL_READ 0x0004 /* 0 = auto read */
+#define OXYGEN_2WIRE_WRITE_MAP_ONLY 0x0008
+#define OXYGEN_2WIRE_SLAVE_AD_MASK 0x0030 /* AD0, AD1 */
+#define OXYGEN_2WIRE_INTERRUPT_MASK 0x0040 /* 0 = int. if not responding */
+#define OXYGEN_2WIRE_SLAVE_NO_RESPONSE 0x0080
+#define OXYGEN_2WIRE_SPEED_MASK 0x0100
+#define OXYGEN_2WIRE_SPEED_STANDARD 0x0000
+#define OXYGEN_2WIRE_SPEED_FAST 0x0100
+#define OXYGEN_2WIRE_CLOCK_SYNC 0x0200
+#define OXYGEN_2WIRE_BUS_RESET 0x0400
+
+#define OXYGEN_SPI_CONTROL 0x98
+#define OXYGEN_SPI_BUSY 0x01 /* read */
+#define OXYGEN_SPI_TRIGGER 0x01 /* write */
+#define OXYGEN_SPI_DATA_LENGTH_MASK 0x02
+#define OXYGEN_SPI_DATA_LENGTH_2 0x00
+#define OXYGEN_SPI_DATA_LENGTH_3 0x02
+#define OXYGEN_SPI_CLOCK_MASK 0xc0
+#define OXYGEN_SPI_CLOCK_160 0x00 /* ns */
+#define OXYGEN_SPI_CLOCK_320 0x40
+#define OXYGEN_SPI_CLOCK_640 0x80
+#define OXYGEN_SPI_CLOCK_1280 0xc0
+#define OXYGEN_SPI_CODEC_MASK 0x70 /* 0..5 */
+#define OXYGEN_SPI_CODEC_SHIFT 4
+#define OXYGEN_SPI_CEN_MASK 0x80
+#define OXYGEN_SPI_CEN_LATCH_CLOCK_LO 0x00
+#define OXYGEN_SPI_CEN_LATCH_CLOCK_HI 0x80
+
+#define OXYGEN_SPI_DATA1 0x99
+#define OXYGEN_SPI_DATA2 0x9a
+#define OXYGEN_SPI_DATA3 0x9b
+
+#define OXYGEN_MPU401 0xa0
+
+#define OXYGEN_MPU401_CONTROL 0xa2
+#define OXYGEN_MPU401_LOOPBACK 0x01 /* TXD to RXD */
+
+#define OXYGEN_GPI_DATA 0xa4
+/* bits 0..5 = pin XGPI0..XGPI5 */
+
+#define OXYGEN_GPI_INTERRUPT_MASK 0xa5
+/* bits 0..5, 1 = enable */
+
+#define OXYGEN_GPIO_DATA 0xa6
+/* bits 0..9 */
+
+#define OXYGEN_GPIO_CONTROL 0xa8
+/* bits 0..9, 0 = input, 1 = output */
+#define OXYGEN_GPIO1_XSLAVE_RDY 0x8000
+
+#define OXYGEN_GPIO_INTERRUPT_MASK 0xaa
+/* bits 0..9, 1 = enable */
+
+#define OXYGEN_DEVICE_SENSE 0xac
+#define OXYGEN_HEAD_PHONE_DETECT 0x01
+#define OXYGEN_HEAD_PHONE_MASK 0x06
+#define OXYGEN_HEAD_PHONE_PASSIVE_SPK 0x00
+#define OXYGEN_HEAD_PHONE_HP 0x02
+#define OXYGEN_HEAD_PHONE_ACTIVE_SPK 0x04
+
+#define OXYGEN_MCU_2WIRE_DATA 0xb0
+
+#define OXYGEN_MCU_2WIRE_MAP 0xb2
+
+#define OXYGEN_MCU_2WIRE_STATUS 0xb3
+#define OXYGEN_MCU_2WIRE_BUSY 0x01
+#define OXYGEN_MCU_2WIRE_LENGTH_MASK 0x06
+#define OXYGEN_MCU_2WIRE_LENGTH_1 0x00
+#define OXYGEN_MCU_2WIRE_LENGTH_2 0x02
+#define OXYGEN_MCU_2WIRE_LENGTH_3 0x04
+#define OXYGEN_MCU_2WIRE_WRITE 0x08 /* r/wc */
+#define OXYGEN_MCU_2WIRE_READ 0x10 /* r/wc */
+#define OXYGEN_MCU_2WIRE_DRV_XACT_FAIL 0x20 /* r/wc */
+#define OXYGEN_MCU_2WIRE_RESET 0x40
+
+#define OXYGEN_MCU_2WIRE_CONTROL 0xb4
+#define OXYGEN_MCU_2WIRE_DRV_ACK 0x01
+#define OXYGEN_MCU_2WIRE_DRV_XACT 0x02
+#define OXYGEN_MCU_2WIRE_INT_MASK 0x04
+#define OXYGEN_MCU_2WIRE_SYNC_MASK 0x08
+#define OXYGEN_MCU_2WIRE_SYNC_RDY_PIN 0x00
+#define OXYGEN_MCU_2WIRE_SYNC_DATA 0x08
+#define OXYGEN_MCU_2WIRE_ADDRESS_MASK 0x30
+#define OXYGEN_MCU_2WIRE_ADDRESS_10 0x00
+#define OXYGEN_MCU_2WIRE_ADDRESS_12 0x10
+#define OXYGEN_MCU_2WIRE_ADDRESS_14 0x20
+#define OXYGEN_MCU_2WIRE_ADDRESS_16 0x30
+#define OXYGEN_MCU_2WIRE_INT_POL 0x40
+#define OXYGEN_MCU_2WIRE_SYNC_ENABLE 0x80
+
+#define OXYGEN_PLAY_ROUTING 0xc0
+#define OXYGEN_PLAY_MUTE01 0x0001
+#define OXYGEN_PLAY_MUTE23 0x0002
+#define OXYGEN_PLAY_MUTE45 0x0004
+#define OXYGEN_PLAY_MUTE67 0x0008
+#define OXYGEN_PLAY_MULTICH_MASK 0x0010
+#define OXYGEN_PLAY_MULTICH_I2S_DAC 0x0000
+#define OXYGEN_PLAY_MULTICH_AC97 0x0010
+#define OXYGEN_PLAY_SPDIF_MASK 0x00e0
+#define OXYGEN_PLAY_SPDIF_SPDIF 0x0000
+#define OXYGEN_PLAY_SPDIF_MULTICH_01 0x0020
+#define OXYGEN_PLAY_SPDIF_MULTICH_23 0x0040
+#define OXYGEN_PLAY_SPDIF_MULTICH_45 0x0060
+#define OXYGEN_PLAY_SPDIF_MULTICH_67 0x0080
+#define OXYGEN_PLAY_SPDIF_REC_A 0x00a0
+#define OXYGEN_PLAY_SPDIF_REC_B 0x00c0
+#define OXYGEN_PLAY_SPDIF_I2S_ADC_3 0x00e0
+#define OXYGEN_PLAY_DAC0_SOURCE_MASK 0x0300
+#define OXYGEN_PLAY_DAC0_SOURCE_SHIFT 8
+#define OXYGEN_PLAY_DAC1_SOURCE_MASK 0x0c00
+#define OXYGEN_PLAY_DAC1_SOURCE_SHIFT 10
+#define OXYGEN_PLAY_DAC2_SOURCE_MASK 0x3000
+#define OXYGEN_PLAY_DAC2_SOURCE_SHIFT 12
+#define OXYGEN_PLAY_DAC3_SOURCE_MASK 0xc000
+#define OXYGEN_PLAY_DAC3_SOURCE_SHIFT 14
+
+#define OXYGEN_REC_ROUTING 0xc2
+#define OXYGEN_MUTE_I2S_ADC_1 0x01
+#define OXYGEN_MUTE_I2S_ADC_2 0x02
+#define OXYGEN_MUTE_I2S_ADC_3 0x04
+#define OXYGEN_REC_A_ROUTE_MASK 0x08
+#define OXYGEN_REC_A_ROUTE_I2S_ADC_1 0x00
+#define OXYGEN_REC_A_ROUTE_AC97_0 0x08
+#define OXYGEN_REC_B_ROUTE_MASK 0x10
+#define OXYGEN_REC_B_ROUTE_I2S_ADC_2 0x00
+#define OXYGEN_REC_B_ROUTE_AC97_1 0x10
+#define OXYGEN_REC_C_ROUTE_MASK 0x20
+#define OXYGEN_REC_C_ROUTE_SPDIF 0x00
+#define OXYGEN_REC_C_ROUTE_I2S_ADC_3 0x20
+
+#define OXYGEN_ADC_MONITOR 0xc3
+#define OXYGEN_ADC_MONITOR_A 0x01
+#define OXYGEN_ADC_MONITOR_A_HALF_VOL 0x02
+#define OXYGEN_ADC_MONITOR_B 0x04
+#define OXYGEN_ADC_MONITOR_B_HALF_VOL 0x08
+#define OXYGEN_ADC_MONITOR_C 0x10
+#define OXYGEN_ADC_MONITOR_C_HALF_VOL 0x20
+
+#define OXYGEN_A_MONITOR_ROUTING 0xc4
+#define OXYGEN_A_MONITOR_ROUTE_0_MASK 0x03
+#define OXYGEN_A_MONITOR_ROUTE_0_SHIFT 0
+#define OXYGEN_A_MONITOR_ROUTE_1_MASK 0x0c
+#define OXYGEN_A_MONITOR_ROUTE_1_SHIFT 2
+#define OXYGEN_A_MONITOR_ROUTE_2_MASK 0x30
+#define OXYGEN_A_MONITOR_ROUTE_2_SHIFT 4
+#define OXYGEN_A_MONITOR_ROUTE_3_MASK 0xc0
+#define OXYGEN_A_MONITOR_ROUTE_3_SHIFT 6
+
+#define OXYGEN_AC97_CONTROL 0xd0
+#define OXYGEN_AC97_COLD_RESET 0x0001
+#define OXYGEN_AC97_SUSPENDED 0x0002 /* read */
+#define OXYGEN_AC97_RESUME 0x0002 /* write */
+#define OXYGEN_AC97_CLOCK_DISABLE 0x0004
+#define OXYGEN_AC97_NO_CODEC_0 0x0008
+#define OXYGEN_AC97_CODEC_0 0x0010
+#define OXYGEN_AC97_CODEC_1 0x0020
+
+#define OXYGEN_AC97_INTERRUPT_MASK 0xd2
+#define OXYGEN_AC97_INT_READ_DONE 0x01
+#define OXYGEN_AC97_INT_WRITE_DONE 0x02
+#define OXYGEN_AC97_INT_CODEC_0 0x10
+#define OXYGEN_AC97_INT_CODEC_1 0x20
+
+#define OXYGEN_AC97_INTERRUPT_STATUS 0xd3
+/* OXYGEN_AC97_INT_* */
+
+#define OXYGEN_AC97_OUT_CONFIG 0xd4
+#define OXYGEN_AC97_CODEC1_SLOT3 0x00000001
+#define OXYGEN_AC97_CODEC1_SLOT3_VSR 0x00000002
+#define OXYGEN_AC97_CODEC1_SLOT4 0x00000010
+#define OXYGEN_AC97_CODEC1_SLOT4_VSR 0x00000020
+#define OXYGEN_AC97_CODEC0_FRONTL 0x00000100
+#define OXYGEN_AC97_CODEC0_FRONTR 0x00000200
+#define OXYGEN_AC97_CODEC0_SIDEL 0x00000400
+#define OXYGEN_AC97_CODEC0_SIDER 0x00000800
+#define OXYGEN_AC97_CODEC0_CENTER 0x00001000
+#define OXYGEN_AC97_CODEC0_BASE 0x00002000
+#define OXYGEN_AC97_CODEC0_REARL 0x00004000
+#define OXYGEN_AC97_CODEC0_REARR 0x00008000
+
+#define OXYGEN_AC97_IN_CONFIG 0xd8
+#define OXYGEN_AC97_CODEC1_LINEL 0x00000001
+#define OXYGEN_AC97_CODEC1_LINEL_VSR 0x00000002
+#define OXYGEN_AC97_CODEC1_LINEL_16 0x00000000
+#define OXYGEN_AC97_CODEC1_LINEL_18 0x00000004
+#define OXYGEN_AC97_CODEC1_LINEL_20 0x00000008
+#define OXYGEN_AC97_CODEC1_LINER 0x00000010
+#define OXYGEN_AC97_CODEC1_LINER_VSR 0x00000020
+#define OXYGEN_AC97_CODEC1_LINER_16 0x00000000
+#define OXYGEN_AC97_CODEC1_LINER_18 0x00000040
+#define OXYGEN_AC97_CODEC1_LINER_20 0x00000080
+#define OXYGEN_AC97_CODEC0_LINEL 0x00000100
+#define OXYGEN_AC97_CODEC0_LINER 0x00000200
+
+#define OXYGEN_AC97_REGS 0xdc
+#define OXYGEN_AC97_REG_DATA_MASK 0x0000ffff
+#define OXYGEN_AC97_REG_ADDR_MASK 0x007f0000
+#define OXYGEN_AC97_REG_ADDR_SHIFT 16
+#define OXYGEN_AC97_REG_DIR_MASK 0x00800000
+#define OXYGEN_AC97_REG_DIR_WRITE 0x00000000
+#define OXYGEN_AC97_REG_DIR_READ 0x00800000
+#define OXYGEN_AC97_REG_CODEC_MASK 0x01000000
+#define OXYGEN_AC97_REG_CODEC_SHIFT 24
+
+#define OXYGEN_TEST 0xe0
+#define OXYGEN_TEST_RAM_SUCCEEDED 0x01
+#define OXYGEN_TEST_PLAYBACK_RAM 0x02
+#define OXYGEN_TEST_RECORD_RAM 0x04
+#define OXYGEN_TEST_PLL 0x08
+#define OXYGEN_TEST_2WIRE_LOOPBACK 0x10
+
+#define OXYGEN_DMA_FLUSH 0xe1
+/* OXYGEN_CHANNEL_* */
+
+#define OXYGEN_CODEC_VERSION 0xe4
+#define OXYGEN_XCID_MASK 0x07
+
+#define OXYGEN_REVISION 0xe6
+#define OXYGEN_REVISION_XPKGID_MASK 0x0007
+#define OXYGEN_REVISION_MASK 0xfff8
+#define OXYGEN_REVISION_2 0x0008 /* bit flag */
+#define OXYGEN_REVISION_8787 0x0014 /* 8 bits */
+
+#define OXYGEN_OFFSIN_48K 0xe8
+#define OXYGEN_OFFSBASE_48K 0xe9
+#define OXYGEN_OFFSBASE_MASK 0x0fff
+#define OXYGEN_OFFSIN_44K 0xec
+#define OXYGEN_OFFSBASE_44K 0xed
+
+#endif
diff --git a/sound/pci/oxygen/pcm1796.h b/sound/pci/oxygen/pcm1796.h
new file mode 100644
index 0000000..698bf46
--- /dev/null
+++ b/sound/pci/oxygen/pcm1796.h
@@ -0,0 +1,58 @@
+#ifndef PCM1796_H_INCLUDED
+#define PCM1796_H_INCLUDED
+
+/* register 16 */
+#define PCM1796_ATL_MASK 0xff
+/* register 17 */
+#define PCM1796_ATR_MASK 0xff
+/* register 18 */
+#define PCM1796_MUTE 0x01
+#define PCM1796_DME 0x02
+#define PCM1796_DMF_MASK 0x0c
+#define PCM1796_DMF_DISABLED 0x00
+#define PCM1796_DMF_48 0x04
+#define PCM1796_DMF_441 0x08
+#define PCM1796_DMF_32 0x0c
+#define PCM1796_FMT_MASK 0x70
+#define PCM1796_FMT_16_RJUST 0x00
+#define PCM1796_FMT_20_RJUST 0x10
+#define PCM1796_FMT_24_RJUST 0x20
+#define PCM1796_FMT_24_LJUST 0x30
+#define PCM1796_FMT_16_I2S 0x40
+#define PCM1796_FMT_24_I2S 0x50
+#define PCM1796_ATLD 0x80
+/* register 19 */
+#define PCM1796_INZD 0x01
+#define PCM1796_FLT_MASK 0x02
+#define PCM1796_FLT_SHARP 0x00
+#define PCM1796_FLT_SLOW 0x02
+#define PCM1796_DFMS 0x04
+#define PCM1796_OPE 0x10
+#define PCM1796_ATS_MASK 0x60
+#define PCM1796_ATS_1 0x00
+#define PCM1796_ATS_2 0x20
+#define PCM1796_ATS_4 0x40
+#define PCM1796_ATS_8 0x60
+#define PCM1796_REV 0x80
+/* register 20 */
+#define PCM1796_OS_MASK 0x03
+#define PCM1796_OS_64 0x00
+#define PCM1796_OS_32 0x01
+#define PCM1796_OS_128 0x02
+#define PCM1796_CHSL_MASK 0x04
+#define PCM1796_CHSL_LEFT 0x00
+#define PCM1796_CHSL_RIGHT 0x04
+#define PCM1796_MONO 0x08
+#define PCM1796_DFTH 0x10
+#define PCM1796_DSD 0x20
+#define PCM1796_SRST 0x40
+/* register 21 */
+#define PCM1796_PCMZ 0x01
+#define PCM1796_DZ_MASK 0x06
+/* register 22 */
+#define PCM1796_ZFGL 0x01
+#define PCM1796_ZFGR 0x02
+/* register 23 */
+#define PCM1796_ID_MASK 0x1f
+
+#endif
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
new file mode 100644
index 0000000..d74cc25
--- /dev/null
+++ b/sound/pci/oxygen/virtuoso.c
@@ -0,0 +1,958 @@
+/*
+ * C-Media CMI8788 driver for Asus Xonar cards
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License, version 2.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this driver; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/*
+ * Xonar D2/D2X
+ * ------------
+ *
+ * CMI8788:
+ *
+ * SPI 0 -> 1st PCM1796 (front)
+ * SPI 1 -> 2nd PCM1796 (surround)
+ * SPI 2 -> 3rd PCM1796 (center/LFE)
+ * SPI 4 -> 4th PCM1796 (back)
+ *
+ * GPIO 2 -> M0 of CS5381
+ * GPIO 3 -> M1 of CS5381
+ * GPIO 5 <- external power present (D2X only)
+ * GPIO 7 -> ALT
+ * GPIO 8 -> enable output to speakers
+ */
+
+/*
+ * Xonar D1/DX
+ * -----------
+ *
+ * CMI8788:
+ *
+ * I²C <-> CS4398 (front)
+ * <-> CS4362A (surround, center/LFE, back)
+ *
+ * GPI 0 <- external power present (DX only)
+ *
+ * GPIO 0 -> enable output to speakers
+ * GPIO 1 -> enable front panel I/O
+ * GPIO 2 -> M0 of CS5361
+ * GPIO 3 -> M1 of CS5361
+ * GPIO 8 -> route input jack to line-in (0) or mic-in (1)
+ *
+ * CS4398:
+ *
+ * AD0 <- 1
+ * AD1 <- 1
+ *
+ * CS4362A:
+ *
+ * AD0 <- 0
+ */
+
+/*
+ * Xonar HDAV1.3 (Deluxe)
+ * ----------------------
+ *
+ * CMI8788:
+ *
+ * I²C <-> PCM1796 (front)
+ *
+ * GPI 0 <- external power present
+ *
+ * GPIO 0 -> enable output to speakers
+ * GPIO 2 -> M0 of CS5381
+ * GPIO 3 -> M1 of CS5381
+ * GPIO 8 -> route input jack to line-in (0) or mic-in (1)
+ *
+ * TXD -> HDMI controller
+ * RXD <- HDMI controller
+ *
+ * PCM1796 front: AD1,0 <- 0,0
+ *
+ * no daughterboard
+ * ----------------
+ *
+ * GPIO 4 <- 1
+ *
+ * H6 daughterboard
+ * ----------------
+ *
+ * GPIO 4 <- 0
+ * GPIO 5 <- 0
+ *
+ * I²C <-> PCM1796 (surround)
+ * <-> PCM1796 (center/LFE)
+ * <-> PCM1796 (back)
+ *
+ * PCM1796 surround: AD1,0 <- 0,1
+ * PCM1796 center/LFE: AD1,0 <- 1,0
+ * PCM1796 back: AD1,0 <- 1,1
+ *
+ * unknown daughterboard
+ * ---------------------
+ *
+ * GPIO 4 <- 0
+ * GPIO 5 <- 1
+ *
+ * I²C <-> CS4362A (surround, center/LFE, back)
+ *
+ * CS4362A: AD0 <- 0
+ */
+
+#include <linux/pci.h>
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <sound/ac97_codec.h>
+#include <sound/asoundef.h>
+#include <sound/control.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include "oxygen.h"
+#include "cm9780.h"
+#include "pcm1796.h"
+#include "cs4398.h"
+#include "cs4362a.h"
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("Asus AVx00 driver");
+MODULE_LICENSE("GPL v2");
+MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "card index");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "enable card");
+
+enum {
+ MODEL_D2,
+ MODEL_D2X,
+ MODEL_D1,
+ MODEL_DX,
+ MODEL_HDAV, /* without daughterboard */
+ MODEL_HDAV_H6, /* with H6 daughterboard */
+};
+
+static struct pci_device_id xonar_ids[] __devinitdata = {
+ { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 },
+ { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX },
+ { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
+ { OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV },
+ { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 },
+ { }
+};
+MODULE_DEVICE_TABLE(pci, xonar_ids);
+
+
+#define GPIO_CS53x1_M_MASK 0x000c
+#define GPIO_CS53x1_M_SINGLE 0x0000
+#define GPIO_CS53x1_M_DOUBLE 0x0004
+#define GPIO_CS53x1_M_QUAD 0x0008
+
+#define GPIO_D2X_EXT_POWER 0x0020
+#define GPIO_D2_ALT 0x0080
+#define GPIO_D2_OUTPUT_ENABLE 0x0100
+
+#define GPI_DX_EXT_POWER 0x01
+#define GPIO_DX_OUTPUT_ENABLE 0x0001
+#define GPIO_DX_FRONT_PANEL 0x0002
+#define GPIO_DX_INPUT_ROUTE 0x0100
+
+#define GPIO_HDAV_DB_MASK 0x0030
+#define GPIO_HDAV_DB_H6 0x0000
+#define GPIO_HDAV_DB_XX 0x0020
+
+#define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ADx=i, /W=0 */
+#define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */
+#define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */
+
+struct xonar_data {
+ unsigned int model;
+ unsigned int anti_pop_delay;
+ unsigned int dacs;
+ u16 output_enable_bit;
+ u8 ext_power_reg;
+ u8 ext_power_int_reg;
+ u8 ext_power_bit;
+ u8 has_power;
+ u8 pcm1796_oversampling;
+ u8 cs4398_fm;
+ u8 cs4362a_fm;
+ u8 hdmi_params[5];
+};
+
+static void xonar_gpio_changed(struct oxygen *chip);
+
+static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
+ u8 reg, u8 value)
+{
+ /* maps ALSA channel pair number to SPI output */
+ static const u8 codec_map[4] = {
+ 0, 1, 2, 4
+ };
+ oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER |
+ OXYGEN_SPI_DATA_LENGTH_2 |
+ OXYGEN_SPI_CLOCK_160 |
+ (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) |
+ OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
+ (reg << 8) | value);
+}
+
+static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec,
+ u8 reg, u8 value)
+{
+ oxygen_write_i2c(chip, I2C_DEVICE_PCM1796(codec), reg, value);
+}
+
+static void pcm1796_write(struct oxygen *chip, unsigned int codec,
+ u8 reg, u8 value)
+{
+ if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) ==
+ OXYGEN_FUNCTION_SPI)
+ pcm1796_write_spi(chip, codec, reg, value);
+ else
+ pcm1796_write_i2c(chip, codec, reg, value);
+}
+
+static void cs4398_write(struct oxygen *chip, u8 reg, u8 value)
+{
+ oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value);
+}
+
+static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value)
+{
+ oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value);
+}
+
+static void hdmi_write_command(struct oxygen *chip, u8 command,
+ unsigned int count, const u8 *params)
+{
+ unsigned int i;
+ u8 checksum;
+
+ oxygen_write_uart(chip, 0xfb);
+ oxygen_write_uart(chip, 0xef);
+ oxygen_write_uart(chip, command);
+ oxygen_write_uart(chip, count);
+ for (i = 0; i < count; ++i)
+ oxygen_write_uart(chip, params[i]);
+ checksum = 0xfb + 0xef + command + count;
+ for (i = 0; i < count; ++i)
+ checksum += params[i];
+ oxygen_write_uart(chip, checksum);
+}
+
+static void xonar_enable_output(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ msleep(data->anti_pop_delay);
+ oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+}
+
+static void xonar_common_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ if (data->ext_power_reg) {
+ oxygen_set_bits8(chip, data->ext_power_int_reg,
+ data->ext_power_bit);
+ chip->interrupt_mask |= OXYGEN_INT_GPIO;
+ chip->model.gpio_changed = xonar_gpio_changed;
+ data->has_power = !!(oxygen_read8(chip, data->ext_power_reg)
+ & data->ext_power_bit);
+ }
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
+ GPIO_CS53x1_M_MASK | data->output_enable_bit);
+ oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
+ GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK);
+ oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC);
+ xonar_enable_output(chip);
+}
+
+static void update_pcm1796_volume(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+ unsigned int i;
+
+ for (i = 0; i < data->dacs; ++i) {
+ pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]);
+ pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]);
+ }
+}
+
+static void update_pcm1796_mute(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+ unsigned int i;
+ u8 value;
+
+ value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD;
+ if (chip->dac_mute)
+ value |= PCM1796_MUTE;
+ for (i = 0; i < data->dacs; ++i)
+ pcm1796_write(chip, i, 18, value);
+}
+
+static void pcm1796_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+ unsigned int i;
+
+ for (i = 0; i < data->dacs; ++i) {
+ pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1);
+ pcm1796_write(chip, i, 20, data->pcm1796_oversampling);
+ pcm1796_write(chip, i, 21, 0);
+ }
+ update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */
+ update_pcm1796_volume(chip);
+}
+
+static void xonar_d2_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ data->anti_pop_delay = 300;
+ data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
+ data->pcm1796_oversampling = PCM1796_OS_64;
+ if (data->model == MODEL_D2X) {
+ data->ext_power_reg = OXYGEN_GPIO_DATA;
+ data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK;
+ data->ext_power_bit = GPIO_D2X_EXT_POWER;
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
+ GPIO_D2X_EXT_POWER);
+ }
+
+ pcm1796_init(chip);
+
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT);
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT);
+
+ xonar_common_init(chip);
+
+ snd_component_add(chip->card, "PCM1796");
+ snd_component_add(chip->card, "CS5381");
+}
+
+static void update_cs4362a_volumes(struct oxygen *chip)
+{
+ u8 mute;
+
+ mute = chip->dac_mute ? CS4362A_MUTE : 0;
+ cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute);
+ cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute);
+ cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute);
+ cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute);
+ cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute);
+ cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute);
+}
+
+static void update_cs43xx_volume(struct oxygen *chip)
+{
+ cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2);
+ cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2);
+ update_cs4362a_volumes(chip);
+}
+
+static void update_cs43xx_mute(struct oxygen *chip)
+{
+ u8 reg;
+
+ reg = CS4398_MUTEP_LOW | CS4398_PAMUTE;
+ if (chip->dac_mute)
+ reg |= CS4398_MUTE_B | CS4398_MUTE_A;
+ cs4398_write(chip, 4, reg);
+ update_cs4362a_volumes(chip);
+}
+
+static void cs43xx_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ /* set CPEN (control port mode) and power down */
+ cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN);
+ cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
+ /* configure */
+ cs4398_write(chip, 2, data->cs4398_fm);
+ cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L);
+ cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP |
+ CS4398_ZERO_CROSS | CS4398_SOFT_RAMP);
+ cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST);
+ cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE |
+ CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP);
+ cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE);
+ cs4362a_write(chip, 0x05, 0);
+ cs4362a_write(chip, 0x06, data->cs4362a_fm);
+ cs4362a_write(chip, 0x09, data->cs4362a_fm);
+ cs4362a_write(chip, 0x0c, data->cs4362a_fm);
+ update_cs43xx_volume(chip);
+ update_cs43xx_mute(chip);
+ /* clear power down */
+ cs4398_write(chip, 8, CS4398_CPEN);
+ cs4362a_write(chip, 0x01, CS4362A_CPEN);
+}
+
+static void xonar_d1_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ data->anti_pop_delay = 800;
+ data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
+ data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST;
+ data->cs4362a_fm = CS4362A_FM_SINGLE |
+ CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
+ if (data->model == MODEL_DX) {
+ data->ext_power_reg = OXYGEN_GPI_DATA;
+ data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+ data->ext_power_bit = GPI_DX_EXT_POWER;
+ }
+
+ oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+ OXYGEN_2WIRE_LENGTH_8 |
+ OXYGEN_2WIRE_INTERRUPT_MASK |
+ OXYGEN_2WIRE_SPEED_FAST);
+
+ cs43xx_init(chip);
+
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
+ GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA,
+ GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE);
+
+ xonar_common_init(chip);
+
+ snd_component_add(chip->card, "CS4398");
+ snd_component_add(chip->card, "CS4362A");
+ snd_component_add(chip->card, "CS5361");
+}
+
+static void xonar_hdav_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+ u8 param;
+
+ oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+ OXYGEN_2WIRE_LENGTH_8 |
+ OXYGEN_2WIRE_INTERRUPT_MASK |
+ OXYGEN_2WIRE_SPEED_FAST);
+
+ data->anti_pop_delay = 100;
+ data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
+ data->ext_power_reg = OXYGEN_GPI_DATA;
+ data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+ data->ext_power_bit = GPI_DX_EXT_POWER;
+ data->pcm1796_oversampling = PCM1796_OS_64;
+
+ pcm1796_init(chip);
+
+ oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DX_INPUT_ROUTE);
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DX_INPUT_ROUTE);
+
+ oxygen_reset_uart(chip);
+ param = 0;
+ hdmi_write_command(chip, 0x61, 1, &param);
+ param = 1;
+ hdmi_write_command(chip, 0x74, 1, &param);
+ data->hdmi_params[1] = IEC958_AES3_CON_FS_48000;
+ data->hdmi_params[4] = 1;
+ hdmi_write_command(chip, 0x54, 5, data->hdmi_params);
+
+ xonar_common_init(chip);
+
+ snd_component_add(chip->card, "PCM1796");
+ snd_component_add(chip->card, "CS5381");
+}
+
+static void xonar_disable_output(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit);
+}
+
+static void xonar_d2_cleanup(struct oxygen *chip)
+{
+ xonar_disable_output(chip);
+}
+
+static void xonar_d1_cleanup(struct oxygen *chip)
+{
+ xonar_disable_output(chip);
+ cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN);
+ oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
+}
+
+static void xonar_hdav_cleanup(struct oxygen *chip)
+{
+ u8 param = 0;
+
+ hdmi_write_command(chip, 0x74, 1, &param);
+ xonar_disable_output(chip);
+}
+
+static void xonar_d2_suspend(struct oxygen *chip)
+{
+ xonar_d2_cleanup(chip);
+}
+
+static void xonar_d1_suspend(struct oxygen *chip)
+{
+ xonar_d1_cleanup(chip);
+}
+
+static void xonar_hdav_suspend(struct oxygen *chip)
+{
+ xonar_hdav_cleanup(chip);
+ msleep(2);
+}
+
+static void xonar_d2_resume(struct oxygen *chip)
+{
+ pcm1796_init(chip);
+ xonar_enable_output(chip);
+}
+
+static void xonar_d1_resume(struct oxygen *chip)
+{
+ cs43xx_init(chip);
+ xonar_enable_output(chip);
+}
+
+static void xonar_hdav_resume(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+ u8 param;
+
+ oxygen_reset_uart(chip);
+ param = 0;
+ hdmi_write_command(chip, 0x61, 1, &param);
+ param = 1;
+ hdmi_write_command(chip, 0x74, 1, &param);
+ hdmi_write_command(chip, 0x54, 5, data->hdmi_params);
+ pcm1796_init(chip);
+ xonar_enable_output(chip);
+}
+
+static void xonar_hdav_pcm_hardware_filter(unsigned int channel,
+ struct snd_pcm_hardware *hardware)
+{
+ if (channel == PCM_MULTICH) {
+ hardware->rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000;
+ hardware->rate_min = 44100;
+ }
+}
+
+static void set_pcm1796_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ struct xonar_data *data = chip->model_data;
+ unsigned int i;
+
+ data->pcm1796_oversampling =
+ params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64;
+ for (i = 0; i < data->dacs; ++i)
+ pcm1796_write(chip, i, 20, data->pcm1796_oversampling);
+}
+
+static void set_cs53x1_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ unsigned int value;
+
+ if (params_rate(params) <= 54000)
+ value = GPIO_CS53x1_M_SINGLE;
+ else if (params_rate(params) <= 108000)
+ value = GPIO_CS53x1_M_DOUBLE;
+ else
+ value = GPIO_CS53x1_M_QUAD;
+ oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
+ value, GPIO_CS53x1_M_MASK);
+}
+
+static void set_cs43xx_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ struct xonar_data *data = chip->model_data;
+
+ data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST;
+ data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
+ if (params_rate(params) <= 50000) {
+ data->cs4398_fm |= CS4398_FM_SINGLE;
+ data->cs4362a_fm |= CS4362A_FM_SINGLE;
+ } else if (params_rate(params) <= 100000) {
+ data->cs4398_fm |= CS4398_FM_DOUBLE;
+ data->cs4362a_fm |= CS4362A_FM_DOUBLE;
+ } else {
+ data->cs4398_fm |= CS4398_FM_QUAD;
+ data->cs4362a_fm |= CS4362A_FM_QUAD;
+ }
+ cs4398_write(chip, 2, data->cs4398_fm);
+ cs4362a_write(chip, 0x06, data->cs4362a_fm);
+ cs4362a_write(chip, 0x09, data->cs4362a_fm);
+ cs4362a_write(chip, 0x0c, data->cs4362a_fm);
+}
+
+static void set_hdmi_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ struct xonar_data *data = chip->model_data;
+
+ data->hdmi_params[0] = 0; /* 1 = non-audio */
+ switch (params_rate(params)) {
+ case 44100:
+ data->hdmi_params[1] = IEC958_AES3_CON_FS_44100;
+ break;
+ case 48000:
+ data->hdmi_params[1] = IEC958_AES3_CON_FS_48000;
+ break;
+ default: /* 96000 */
+ data->hdmi_params[1] = IEC958_AES3_CON_FS_96000;
+ break;
+ case 192000:
+ data->hdmi_params[1] = IEC958_AES3_CON_FS_192000;
+ break;
+ }
+ data->hdmi_params[2] = params_channels(params) / 2 - 1;
+ if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE)
+ data->hdmi_params[3] = 0;
+ else
+ data->hdmi_params[3] = 0xc0;
+ data->hdmi_params[4] = 1; /* ? */
+ hdmi_write_command(chip, 0x54, 5, data->hdmi_params);
+}
+
+static void set_hdav_params(struct oxygen *chip,
+ struct snd_pcm_hw_params *params)
+{
+ set_pcm1796_params(chip, params);
+ set_hdmi_params(chip, params);
+}
+
+static void xonar_gpio_changed(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+ u8 has_power;
+
+ has_power = !!(oxygen_read8(chip, data->ext_power_reg)
+ & data->ext_power_bit);
+ if (has_power != data->has_power) {
+ data->has_power = has_power;
+ if (has_power) {
+ snd_printk(KERN_NOTICE "power restored\n");
+ } else {
+ snd_printk(KERN_CRIT
+ "Hey! Don't unplug the power cable!\n");
+ /* TODO: stop PCMs */
+ }
+ }
+}
+
+static void xonar_hdav_uart_input(struct oxygen *chip)
+{
+ if (chip->uart_input_count >= 2 &&
+ chip->uart_input[chip->uart_input_count - 2] == 'O' &&
+ chip->uart_input[chip->uart_input_count - 1] == 'K') {
+ printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:");
+ print_hex_dump_bytes("", DUMP_PREFIX_OFFSET,
+ chip->uart_input, chip->uart_input_count);
+ chip->uart_input_count = 0;
+ }
+}
+
+static int gpio_bit_switch_get(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u16 bit = ctl->private_value;
+
+ value->value.integer.value[0] =
+ !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit);
+ return 0;
+}
+
+static int gpio_bit_switch_put(struct snd_kcontrol *ctl,
+ struct snd_ctl_elem_value *value)
+{
+ struct oxygen *chip = ctl->private_data;
+ u16 bit = ctl->private_value;
+ u16 old_bits, new_bits;
+ int changed;
+
+ spin_lock_irq(&chip->reg_lock);
+ old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA);
+ if (value->value.integer.value[0])
+ new_bits = old_bits | bit;
+ else
+ new_bits = old_bits & ~bit;
+ changed = new_bits != old_bits;
+ if (changed)
+ oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits);
+ spin_unlock_irq(&chip->reg_lock);
+ return changed;
+}
+
+static const struct snd_kcontrol_new alt_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Loopback Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = gpio_bit_switch_get,
+ .put = gpio_bit_switch_put,
+ .private_value = GPIO_D2_ALT,
+};
+
+static const struct snd_kcontrol_new front_panel_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Front Panel Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = gpio_bit_switch_get,
+ .put = gpio_bit_switch_put,
+ .private_value = GPIO_DX_FRONT_PANEL,
+};
+
+static void xonar_line_mic_ac97_switch(struct oxygen *chip,
+ unsigned int reg, unsigned int mute)
+{
+ if (reg == AC97_LINE) {
+ spin_lock_irq(&chip->reg_lock);
+ oxygen_write16_masked(chip, OXYGEN_GPIO_DATA,
+ mute ? GPIO_DX_INPUT_ROUTE : 0,
+ GPIO_DX_INPUT_ROUTE);
+ spin_unlock_irq(&chip->reg_lock);
+ }
+}
+
+static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0);
+static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
+
+static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
+{
+ if (!strncmp(template->name, "CD Capture ", 11))
+ /* CD in is actually connected to the video in pin */
+ template->private_value ^= AC97_CD ^ AC97_VIDEO;
+ return 0;
+}
+
+static int xonar_d1_control_filter(struct snd_kcontrol_new *template)
+{
+ if (!strncmp(template->name, "CD Capture ", 11))
+ return 1; /* no CD input */
+ return 0;
+}
+
+static int xonar_d2_mixer_init(struct oxygen *chip)
+{
+ return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip));
+}
+
+static int xonar_d1_mixer_init(struct oxygen *chip)
+{
+ return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip));
+}
+
+static int xonar_model_probe(struct oxygen *chip, unsigned long driver_data)
+{
+ static const char *const names[] = {
+ [MODEL_D1] = "Xonar D1",
+ [MODEL_DX] = "Xonar DX",
+ [MODEL_D2] = "Xonar D2",
+ [MODEL_D2X] = "Xonar D2X",
+ [MODEL_HDAV] = "Xonar HDAV1.3",
+ [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6",
+ };
+ static const u8 dacs[] = {
+ [MODEL_D1] = 2,
+ [MODEL_DX] = 2,
+ [MODEL_D2] = 4,
+ [MODEL_D2X] = 4,
+ [MODEL_HDAV] = 1,
+ [MODEL_HDAV_H6] = 4,
+ };
+ struct xonar_data *data = chip->model_data;
+
+ data->model = driver_data;
+ if (data->model == MODEL_HDAV) {
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
+ GPIO_HDAV_DB_MASK);
+ switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) &
+ GPIO_HDAV_DB_MASK) {
+ case GPIO_HDAV_DB_H6:
+ data->model = MODEL_HDAV_H6;
+ break;
+ case GPIO_HDAV_DB_XX:
+ snd_printk(KERN_ERR "unknown daughterboard\n");
+ return -ENODEV;
+ }
+ }
+
+ data->dacs = dacs[data->model];
+ chip->model.shortname = names[data->model];
+ return 0;
+}
+
+static const struct oxygen_model model_xonar_d2 = {
+ .longname = "Asus Virtuoso 200",
+ .chip = "AV200",
+ .owner = THIS_MODULE,
+ .probe = xonar_model_probe,
+ .init = xonar_d2_init,
+ .control_filter = xonar_d2_control_filter,
+ .mixer_init = xonar_d2_mixer_init,
+ .cleanup = xonar_d2_cleanup,
+ .suspend = xonar_d2_suspend,
+ .resume = xonar_d2_resume,
+ .set_dac_params = set_pcm1796_params,
+ .set_adc_params = set_cs53x1_params,
+ .update_dac_volume = update_pcm1796_volume,
+ .update_dac_mute = update_pcm1796_mute,
+ .dac_tlv = pcm1796_db_scale,
+ .model_data_size = sizeof(struct xonar_data),
+ .device_config = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_1_FROM_SPDIF |
+ MIDI_OUTPUT |
+ MIDI_INPUT,
+ .dac_channels = 8,
+ .dac_volume_min = 0x0f,
+ .dac_volume_max = 0xff,
+ .misc_flags = OXYGEN_MISC_MIDI,
+ .function_flags = OXYGEN_FUNCTION_SPI |
+ OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+};
+
+static const struct oxygen_model model_xonar_d1 = {
+ .longname = "Asus Virtuoso 100",
+ .chip = "AV200",
+ .owner = THIS_MODULE,
+ .probe = xonar_model_probe,
+ .init = xonar_d1_init,
+ .control_filter = xonar_d1_control_filter,
+ .mixer_init = xonar_d1_mixer_init,
+ .cleanup = xonar_d1_cleanup,
+ .suspend = xonar_d1_suspend,
+ .resume = xonar_d1_resume,
+ .set_dac_params = set_cs43xx_params,
+ .set_adc_params = set_cs53x1_params,
+ .update_dac_volume = update_cs43xx_volume,
+ .update_dac_mute = update_cs43xx_mute,
+ .ac97_switch = xonar_line_mic_ac97_switch,
+ .dac_tlv = cs4362a_db_scale,
+ .model_data_size = sizeof(struct xonar_data),
+ .device_config = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2,
+ .dac_channels = 8,
+ .dac_volume_min = 0,
+ .dac_volume_max = 127,
+ .function_flags = OXYGEN_FUNCTION_2WIRE,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+};
+
+static const struct oxygen_model model_xonar_hdav = {
+ .longname = "Asus Virtuoso 200",
+ .chip = "AV200",
+ .owner = THIS_MODULE,
+ .probe = xonar_model_probe,
+ .init = xonar_hdav_init,
+ .cleanup = xonar_hdav_cleanup,
+ .suspend = xonar_hdav_suspend,
+ .resume = xonar_hdav_resume,
+ .pcm_hardware_filter = xonar_hdav_pcm_hardware_filter,
+ .set_dac_params = set_hdav_params,
+ .set_adc_params = set_cs53x1_params,
+ .update_dac_volume = update_pcm1796_volume,
+ .update_dac_mute = update_pcm1796_mute,
+ .uart_input = xonar_hdav_uart_input,
+ .ac97_switch = xonar_line_mic_ac97_switch,
+ .dac_tlv = pcm1796_db_scale,
+ .model_data_size = sizeof(struct xonar_data),
+ .device_config = PLAYBACK_0_TO_I2S |
+ PLAYBACK_1_TO_SPDIF |
+ CAPTURE_0_FROM_I2S_2,
+ .dac_channels = 8,
+ .dac_volume_min = 0x0f,
+ .dac_volume_max = 0xff,
+ .misc_flags = OXYGEN_MISC_MIDI,
+ .function_flags = OXYGEN_FUNCTION_2WIRE,
+ .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+ .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+};
+
+static int __devinit xonar_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static const struct oxygen_model *const models[] = {
+ [MODEL_D1] = &model_xonar_d1,
+ [MODEL_DX] = &model_xonar_d1,
+ [MODEL_D2] = &model_xonar_d2,
+ [MODEL_D2X] = &model_xonar_d2,
+ [MODEL_HDAV] = &model_xonar_hdav,
+ };
+ static int dev;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ ++dev;
+ return -ENOENT;
+ }
+ BUG_ON(pci_id->driver_data >= ARRAY_SIZE(models));
+ err = oxygen_pci_probe(pci, index[dev], id[dev],
+ models[pci_id->driver_data],
+ pci_id->driver_data);
+ if (err >= 0)
+ ++dev;
+ return err;
+}
+
+static struct pci_driver xonar_driver = {
+ .name = "AV200",
+ .id_table = xonar_ids,
+ .probe = xonar_probe,
+ .remove = __devexit_p(oxygen_pci_remove),
+#ifdef CONFIG_PM
+ .suspend = oxygen_pci_suspend,
+ .resume = oxygen_pci_resume,
+#endif
+};
+
+static int __init alsa_card_xonar_init(void)
+{
+ return pci_register_driver(&xonar_driver);
+}
+
+static void __exit alsa_card_xonar_exit(void)
+{
+ pci_unregister_driver(&xonar_driver);
+}
+
+module_init(alsa_card_xonar_init)
+module_exit(alsa_card_xonar_exit)
diff --git a/sound/pci/oxygen/wm8785.h b/sound/pci/oxygen/wm8785.h
new file mode 100644
index 0000000..8c23e31
--- /dev/null
+++ b/sound/pci/oxygen/wm8785.h
@@ -0,0 +1,45 @@
+#ifndef WM8785_H_INCLUDED
+#define WM8785_H_INCLUDED
+
+#define WM8785_R0 0
+#define WM8785_R1 1
+#define WM8785_R2 2
+#define WM8785_R7 7
+
+/* R0 */
+#define WM8785_MCR_MASK 0x007
+#define WM8785_MCR_SLAVE 0x000
+#define WM8785_MCR_MASTER_128 0x001
+#define WM8785_MCR_MASTER_192 0x002
+#define WM8785_MCR_MASTER_256 0x003
+#define WM8785_MCR_MASTER_384 0x004
+#define WM8785_MCR_MASTER_512 0x005
+#define WM8785_MCR_MASTER_768 0x006
+#define WM8785_OSR_MASK 0x018
+#define WM8785_OSR_SINGLE 0x000
+#define WM8785_OSR_DOUBLE 0x008
+#define WM8785_OSR_QUAD 0x010
+#define WM8785_FORMAT_MASK 0x060
+#define WM8785_FORMAT_RJUST 0x000
+#define WM8785_FORMAT_LJUST 0x020
+#define WM8785_FORMAT_I2S 0x040
+#define WM8785_FORMAT_DSP 0x060
+/* R1 */
+#define WM8785_WL_MASK 0x003
+#define WM8785_WL_16 0x000
+#define WM8785_WL_20 0x001
+#define WM8785_WL_24 0x002
+#define WM8785_WL_32 0x003
+#define WM8785_LRP 0x004
+#define WM8785_BCLKINV 0x008
+#define WM8785_LRSWAP 0x010
+#define WM8785_DEVNO_MASK 0x0e0
+/* R2 */
+#define WM8785_HPFR 0x001
+#define WM8785_HPFL 0x002
+#define WM8785_SDODIS 0x004
+#define WM8785_PWRDNR 0x008
+#define WM8785_PWRDNL 0x010
+#define WM8785_TDM_MASK 0x1c0
+
+#endif
diff --git a/sound/pci/pcxhr/Makefile b/sound/pci/pcxhr/Makefile
new file mode 100644
index 0000000..10473c0
--- /dev/null
+++ b/sound/pci/pcxhr/Makefile
@@ -0,0 +1,2 @@
+snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o
+obj-$(CONFIG_SND_PCXHR) += snd-pcxhr.o
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
new file mode 100644
index 0000000..73de6e9
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -0,0 +1,1373 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * main file with alsa callbacks
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <linux/delay.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "pcxhr.h"
+#include "pcxhr_mixer.h"
+#include "pcxhr_hwdep.h"
+#include "pcxhr_core.h"
+
+#define DRIVER_NAME "pcxhr"
+
+MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>");
+MODULE_DESCRIPTION("Digigram " DRIVER_NAME " " PCXHR_DRIVER_VERSION_STRING);
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int mono[SNDRV_CARDS]; /* capture in mono only */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Digigram " DRIVER_NAME " soundcard");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Digigram " DRIVER_NAME " soundcard");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Digigram " DRIVER_NAME " soundcard");
+module_param_array(mono, bool, NULL, 0444);
+MODULE_PARM_DESC(mono, "Mono capture mode (default is stereo)");
+
+enum {
+ PCI_ID_VX882HR,
+ PCI_ID_PCX882HR,
+ PCI_ID_VX881HR,
+ PCI_ID_PCX881HR,
+ PCI_ID_PCX1222HR,
+ PCI_ID_PCX1221HR,
+ PCI_ID_LAST
+};
+
+static struct pci_device_id pcxhr_ids[] = {
+ { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, /* VX882HR */
+ { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, /* PCX882HR */
+ { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, /* VX881HR */
+ { 0x10b5, 0x9656, 0x1369, 0xb301, 0, 0, PCI_ID_PCX881HR, }, /* PCX881HR */
+ { 0x10b5, 0x9656, 0x1369, 0xb501, 0, 0, PCI_ID_PCX1222HR, }, /* PCX1222HR */
+ { 0x10b5, 0x9656, 0x1369, 0xb701, 0, 0, PCI_ID_PCX1221HR, }, /* PCX1221HR */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, pcxhr_ids);
+
+struct board_parameters {
+ char* board_name;
+ short playback_chips;
+ short capture_chips;
+ short firmware_num;
+};
+static struct board_parameters pcxhr_board_params[] = {
+[PCI_ID_VX882HR] = { "VX882HR", 4, 4, 41, },
+[PCI_ID_PCX882HR] = { "PCX882HR", 4, 4, 41, },
+[PCI_ID_VX881HR] = { "VX881HR", 4, 4, 41, },
+[PCI_ID_PCX881HR] = { "PCX881HR", 4, 4, 41, },
+[PCI_ID_PCX1222HR] = { "PCX1222HR", 6, 1, 42, },
+[PCI_ID_PCX1221HR] = { "PCX1221HR", 6, 1, 42, },
+};
+
+
+static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg,
+ unsigned int* realfreq)
+{
+ unsigned int reg;
+
+ if (freq < 6900 || freq > 110250)
+ return -EINVAL;
+ reg = (28224000 * 10) / freq;
+ reg = (reg + 5) / 10;
+ if (reg < 0x200)
+ *pllreg = reg + 0x800;
+ else if (reg < 0x400)
+ *pllreg = reg & 0x1ff;
+ else if (reg < 0x800) {
+ *pllreg = ((reg >> 1) & 0x1ff) + 0x200;
+ reg &= ~1;
+ } else {
+ *pllreg = ((reg >> 2) & 0x1ff) + 0x400;
+ reg &= ~3;
+ }
+ if (realfreq)
+ *realfreq = ((28224000 * 10) / reg + 5) / 10;
+ return 0;
+}
+
+
+#define PCXHR_FREQ_REG_MASK 0x1f
+#define PCXHR_FREQ_QUARTZ_48000 0x00
+#define PCXHR_FREQ_QUARTZ_24000 0x01
+#define PCXHR_FREQ_QUARTZ_12000 0x09
+#define PCXHR_FREQ_QUARTZ_32000 0x08
+#define PCXHR_FREQ_QUARTZ_16000 0x04
+#define PCXHR_FREQ_QUARTZ_8000 0x0c
+#define PCXHR_FREQ_QUARTZ_44100 0x02
+#define PCXHR_FREQ_QUARTZ_22050 0x0a
+#define PCXHR_FREQ_QUARTZ_11025 0x06
+#define PCXHR_FREQ_PLL 0x05
+#define PCXHR_FREQ_QUARTZ_192000 0x10
+#define PCXHR_FREQ_QUARTZ_96000 0x18
+#define PCXHR_FREQ_QUARTZ_176400 0x14
+#define PCXHR_FREQ_QUARTZ_88200 0x1c
+#define PCXHR_FREQ_QUARTZ_128000 0x12
+#define PCXHR_FREQ_QUARTZ_64000 0x1a
+
+#define PCXHR_FREQ_WORD_CLOCK 0x0f
+#define PCXHR_FREQ_SYNC_AES 0x0e
+#define PCXHR_FREQ_AES_1 0x07
+#define PCXHR_FREQ_AES_2 0x0b
+#define PCXHR_FREQ_AES_3 0x03
+#define PCXHR_FREQ_AES_4 0x0d
+
+#define PCXHR_MODIFY_CLOCK_S_BIT 0x04
+
+#define PCXHR_IRQ_TIMER_FREQ 92000
+#define PCXHR_IRQ_TIMER_PERIOD 48
+
+static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate,
+ unsigned int *reg, unsigned int *freq)
+{
+ unsigned int val, realfreq, pllreg;
+ struct pcxhr_rmh rmh;
+ int err;
+
+ realfreq = rate;
+ switch (mgr->use_clock_type) {
+ case PCXHR_CLOCK_TYPE_INTERNAL : /* clock by quartz or pll */
+ switch (rate) {
+ case 48000 : val = PCXHR_FREQ_QUARTZ_48000; break;
+ case 24000 : val = PCXHR_FREQ_QUARTZ_24000; break;
+ case 12000 : val = PCXHR_FREQ_QUARTZ_12000; break;
+ case 32000 : val = PCXHR_FREQ_QUARTZ_32000; break;
+ case 16000 : val = PCXHR_FREQ_QUARTZ_16000; break;
+ case 8000 : val = PCXHR_FREQ_QUARTZ_8000; break;
+ case 44100 : val = PCXHR_FREQ_QUARTZ_44100; break;
+ case 22050 : val = PCXHR_FREQ_QUARTZ_22050; break;
+ case 11025 : val = PCXHR_FREQ_QUARTZ_11025; break;
+ case 192000 : val = PCXHR_FREQ_QUARTZ_192000; break;
+ case 96000 : val = PCXHR_FREQ_QUARTZ_96000; break;
+ case 176400 : val = PCXHR_FREQ_QUARTZ_176400; break;
+ case 88200 : val = PCXHR_FREQ_QUARTZ_88200; break;
+ case 128000 : val = PCXHR_FREQ_QUARTZ_128000; break;
+ case 64000 : val = PCXHR_FREQ_QUARTZ_64000; break;
+ default :
+ val = PCXHR_FREQ_PLL;
+ /* get the value for the pll register */
+ err = pcxhr_pll_freq_register(rate, &pllreg, &realfreq);
+ if (err)
+ return err;
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
+ rmh.cmd[0] |= IO_NUM_REG_GENCLK;
+ rmh.cmd[1] = pllreg & MASK_DSP_WORD;
+ rmh.cmd[2] = pllreg >> 24;
+ rmh.cmd_len = 3;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err < 0) {
+ snd_printk(KERN_ERR
+ "error CMD_ACCESS_IO_WRITE for PLL register : %x!\n",
+ err );
+ return err;
+ }
+ }
+ break;
+ case PCXHR_CLOCK_TYPE_WORD_CLOCK : val = PCXHR_FREQ_WORD_CLOCK; break;
+ case PCXHR_CLOCK_TYPE_AES_SYNC : val = PCXHR_FREQ_SYNC_AES; break;
+ case PCXHR_CLOCK_TYPE_AES_1 : val = PCXHR_FREQ_AES_1; break;
+ case PCXHR_CLOCK_TYPE_AES_2 : val = PCXHR_FREQ_AES_2; break;
+ case PCXHR_CLOCK_TYPE_AES_3 : val = PCXHR_FREQ_AES_3; break;
+ case PCXHR_CLOCK_TYPE_AES_4 : val = PCXHR_FREQ_AES_4; break;
+ default : return -EINVAL;
+ }
+ *reg = val;
+ *freq = realfreq;
+ return 0;
+}
+
+
+int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate)
+{
+ unsigned int val, realfreq, speed;
+ struct pcxhr_rmh rmh;
+ int err, changed;
+
+ if (rate == 0)
+ return 0; /* nothing to do */
+
+ err = pcxhr_get_clock_reg(mgr, rate, &val, &realfreq);
+ if (err)
+ return err;
+
+ /* codec speed modes */
+ if (rate < 55000)
+ speed = 0; /* single speed */
+ else if (rate < 100000)
+ speed = 1; /* dual speed */
+ else
+ speed = 2; /* quad speed */
+ if (mgr->codec_speed != speed) {
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* mute outputs */
+ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set speed ratio */
+ rmh.cmd[0] |= IO_NUM_SPEED_RATIO;
+ rmh.cmd[1] = speed;
+ rmh.cmd_len = 2;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+ }
+ /* set the new frequency */
+ snd_printdd("clock register : set %x\n", val);
+ err = pcxhr_write_io_num_reg_cont(mgr, PCXHR_FREQ_REG_MASK, val, &changed);
+ if (err)
+ return err;
+ mgr->sample_rate_real = realfreq;
+ mgr->cur_clock_type = mgr->use_clock_type;
+
+ /* unmute after codec speed modes */
+ if (mgr->codec_speed != speed) {
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* unmute outputs */
+ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+ mgr->codec_speed = speed; /* save new codec speed */
+ }
+
+ if (changed) {
+ pcxhr_init_rmh(&rmh, CMD_MODIFY_CLOCK);
+ rmh.cmd[0] |= PCXHR_MODIFY_CLOCK_S_BIT; /* resync fifos */
+ if (rate < PCXHR_IRQ_TIMER_FREQ)
+ rmh.cmd[1] = PCXHR_IRQ_TIMER_PERIOD;
+ else
+ rmh.cmd[1] = PCXHR_IRQ_TIMER_PERIOD * 2;
+ rmh.cmd[2] = rate;
+ rmh.cmd_len = 3;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+ }
+ snd_printdd("pcxhr_set_clock to %dHz (realfreq=%d)\n", rate, realfreq);
+ return 0;
+}
+
+
+int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type,
+ int *sample_rate)
+{
+ struct pcxhr_rmh rmh;
+ unsigned char reg;
+ int err, rate;
+
+ switch (clock_type) {
+ case PCXHR_CLOCK_TYPE_WORD_CLOCK : reg = REG_STATUS_WORD_CLOCK; break;
+ case PCXHR_CLOCK_TYPE_AES_SYNC : reg = REG_STATUS_AES_SYNC; break;
+ case PCXHR_CLOCK_TYPE_AES_1 : reg = REG_STATUS_AES_1; break;
+ case PCXHR_CLOCK_TYPE_AES_2 : reg = REG_STATUS_AES_2; break;
+ case PCXHR_CLOCK_TYPE_AES_3 : reg = REG_STATUS_AES_3; break;
+ case PCXHR_CLOCK_TYPE_AES_4 : reg = REG_STATUS_AES_4; break;
+ default : return -EINVAL;
+ }
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);
+ rmh.cmd_len = 2;
+ rmh.cmd[0] |= IO_NUM_REG_STATUS;
+ if (mgr->last_reg_stat != reg) {
+ rmh.cmd[1] = reg;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+ udelay(100); /* wait minimum 2 sample_frames at 32kHz ! */
+ mgr->last_reg_stat = reg;
+ }
+ rmh.cmd[1] = REG_STATUS_CURRENT;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+ switch (rmh.stat[1] & 0x0f) {
+ case REG_STATUS_SYNC_32000 : rate = 32000; break;
+ case REG_STATUS_SYNC_44100 : rate = 44100; break;
+ case REG_STATUS_SYNC_48000 : rate = 48000; break;
+ case REG_STATUS_SYNC_64000 : rate = 64000; break;
+ case REG_STATUS_SYNC_88200 : rate = 88200; break;
+ case REG_STATUS_SYNC_96000 : rate = 96000; break;
+ case REG_STATUS_SYNC_128000 : rate = 128000; break;
+ case REG_STATUS_SYNC_176400 : rate = 176400; break;
+ case REG_STATUS_SYNC_192000 : rate = 192000; break;
+ default: rate = 0;
+ }
+ snd_printdd("External clock is at %d Hz\n", rate);
+ *sample_rate = rate;
+ return 0;
+}
+
+
+/*
+ * start or stop playback/capture substream
+ */
+static int pcxhr_set_stream_state(struct pcxhr_stream *stream)
+{
+ int err;
+ struct snd_pcxhr *chip;
+ struct pcxhr_rmh rmh;
+ int stream_mask, start;
+
+ if (stream->status == PCXHR_STREAM_STATUS_SCHEDULE_RUN)
+ start = 1;
+ else {
+ if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) {
+ snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state CANNOT be stopped\n");
+ return -EINVAL;
+ }
+ start = 0;
+ }
+ if (!stream->substream)
+ return -EINVAL;
+
+ stream->timer_abs_periods = 0;
+ stream->timer_period_frag = 0; /* reset theoretical stream pos */
+ stream->timer_buf_periods = 0;
+ stream->timer_is_synced = 0;
+
+ stream_mask = stream->pipe->is_capture ? 1 : 1<<stream->substream->number;
+
+ pcxhr_init_rmh(&rmh, start ? CMD_START_STREAM : CMD_STOP_STREAM);
+ pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture,
+ stream->pipe->first_audio, 0, stream_mask);
+
+ chip = snd_pcm_substream_chip(stream->substream);
+
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err)
+ snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state err=%x;\n", err);
+ stream->status = start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED;
+ return err;
+}
+
+#define HEADER_FMT_BASE_LIN 0xfed00000
+#define HEADER_FMT_BASE_FLOAT 0xfad00000
+#define HEADER_FMT_INTEL 0x00008000
+#define HEADER_FMT_24BITS 0x00004000
+#define HEADER_FMT_16BITS 0x00002000
+#define HEADER_FMT_UPTO11 0x00000200
+#define HEADER_FMT_UPTO32 0x00000100
+#define HEADER_FMT_MONO 0x00000080
+
+static int pcxhr_set_format(struct pcxhr_stream *stream)
+{
+ int err, is_capture, sample_rate, stream_num;
+ struct snd_pcxhr *chip;
+ struct pcxhr_rmh rmh;
+ unsigned int header;
+
+ switch (stream->format) {
+ case SNDRV_PCM_FORMAT_U8:
+ header = HEADER_FMT_BASE_LIN;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS | HEADER_FMT_INTEL;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS | HEADER_FMT_INTEL;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3BE:
+ header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS;
+ break;
+ case SNDRV_PCM_FORMAT_FLOAT_LE:
+ header = HEADER_FMT_BASE_FLOAT | HEADER_FMT_INTEL;
+ break;
+ default:
+ snd_printk(KERN_ERR "error pcxhr_set_format() : unknown format\n");
+ return -EINVAL;
+ }
+ chip = snd_pcm_substream_chip(stream->substream);
+
+ sample_rate = chip->mgr->sample_rate;
+ if (sample_rate <= 32000 && sample_rate !=0) {
+ if (sample_rate <= 11025)
+ header |= HEADER_FMT_UPTO11;
+ else
+ header |= HEADER_FMT_UPTO32;
+ }
+ if (stream->channels == 1)
+ header |= HEADER_FMT_MONO;
+
+ is_capture = stream->pipe->is_capture;
+ stream_num = is_capture ? 0 : stream->substream->number;
+
+ pcxhr_init_rmh(&rmh, is_capture ? CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT);
+ pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0);
+ if (is_capture)
+ rmh.cmd[0] |= 1<<12;
+ rmh.cmd[1] = 0;
+ rmh.cmd[2] = header >> 8;
+ rmh.cmd[3] = (header & 0xff) << 16;
+ rmh.cmd_len = 4;
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err)
+ snd_printk(KERN_ERR "ERROR pcxhr_set_format err=%x;\n", err);
+ return err;
+}
+
+static int pcxhr_update_r_buffer(struct pcxhr_stream *stream)
+{
+ int err, is_capture, stream_num;
+ struct pcxhr_rmh rmh;
+ struct snd_pcm_substream *subs = stream->substream;
+ struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
+
+ is_capture = (subs->stream == SNDRV_PCM_STREAM_CAPTURE);
+ stream_num = is_capture ? 0 : subs->number;
+
+ snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n",
+ is_capture ? 'c' : 'p',
+ chip->chip_idx, (void *)(long)subs->runtime->dma_addr,
+ subs->runtime->dma_bytes, subs->number);
+
+ pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS);
+ pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0);
+
+ /* max buffer size is 2 MByte */
+ snd_BUG_ON(subs->runtime->dma_bytes >= 0x200000);
+ rmh.cmd[1] = subs->runtime->dma_bytes * 8; /* size in bits */
+ rmh.cmd[2] = subs->runtime->dma_addr >> 24; /* most significant byte */
+ rmh.cmd[2] |= 1<<19; /* this is a circular buffer */
+ rmh.cmd[3] = subs->runtime->dma_addr & MASK_DSP_WORD; /* least 3 significant bytes */
+ rmh.cmd_len = 4;
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err)
+ snd_printk(KERN_ERR "ERROR CMD_UPDATE_R_BUFFERS err=%x;\n", err);
+ return err;
+}
+
+
+#if 0
+static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, snd_pcm_uframes_t *sample_count)
+{
+ struct pcxhr_rmh rmh;
+ int err;
+ pcxhr_t *chip = snd_pcm_substream_chip(stream->substream);
+ pcxhr_init_rmh(&rmh, CMD_PIPE_SAMPLE_COUNT);
+ pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture, 0, 0,
+ 1<<stream->pipe->first_audio);
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err == 0) {
+ *sample_count = ((snd_pcm_uframes_t)rmh.stat[0]) << 24;
+ *sample_count += (snd_pcm_uframes_t)rmh.stat[1];
+ }
+ snd_printdd("PIPE_SAMPLE_COUNT = %lx\n", *sample_count);
+ return err;
+}
+#endif
+
+static inline int pcxhr_stream_scheduled_get_pipe(struct pcxhr_stream *stream,
+ struct pcxhr_pipe **pipe)
+{
+ if (stream->status == PCXHR_STREAM_STATUS_SCHEDULE_RUN) {
+ *pipe = stream->pipe;
+ return 1;
+ }
+ return 0;
+}
+
+static void pcxhr_trigger_tasklet(unsigned long arg)
+{
+ unsigned long flags;
+ int i, j, err;
+ struct pcxhr_pipe *pipe;
+ struct snd_pcxhr *chip;
+ struct pcxhr_mgr *mgr = (struct pcxhr_mgr*)(arg);
+ int capture_mask = 0;
+ int playback_mask = 0;
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ struct timeval my_tv1, my_tv2;
+ do_gettimeofday(&my_tv1);
+#endif
+ mutex_lock(&mgr->setup_mutex);
+
+ /* check the pipes concerned and build pipe_array */
+ for (i = 0; i < mgr->num_cards; i++) {
+ chip = mgr->chip[i];
+ for (j = 0; j < chip->nb_streams_capt; j++) {
+ if (pcxhr_stream_scheduled_get_pipe(&chip->capture_stream[j], &pipe))
+ capture_mask |= (1 << pipe->first_audio);
+ }
+ for (j = 0; j < chip->nb_streams_play; j++) {
+ if (pcxhr_stream_scheduled_get_pipe(&chip->playback_stream[j], &pipe)) {
+ playback_mask |= (1 << pipe->first_audio);
+ break; /* add only once, as all playback streams of
+ * one chip use the same pipe
+ */
+ }
+ }
+ }
+ if (capture_mask == 0 && playback_mask == 0) {
+ mutex_unlock(&mgr->setup_mutex);
+ snd_printk(KERN_ERR "pcxhr_trigger_tasklet : no pipes\n");
+ return;
+ }
+
+ snd_printdd("pcxhr_trigger_tasklet : playback_mask=%x capture_mask=%x\n",
+ playback_mask, capture_mask);
+
+ /* synchronous stop of all the pipes concerned */
+ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0);
+ if (err) {
+ mutex_unlock(&mgr->setup_mutex);
+ snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error stop pipes (P%x C%x)\n",
+ playback_mask, capture_mask);
+ return;
+ }
+
+ /* unfortunately the dsp lost format and buffer info with the stop pipe */
+ for (i = 0; i < mgr->num_cards; i++) {
+ struct pcxhr_stream *stream;
+ chip = mgr->chip[i];
+ for (j = 0; j < chip->nb_streams_capt; j++) {
+ stream = &chip->capture_stream[j];
+ if (pcxhr_stream_scheduled_get_pipe(stream, &pipe)) {
+ err = pcxhr_set_format(stream);
+ err = pcxhr_update_r_buffer(stream);
+ }
+ }
+ for (j = 0; j < chip->nb_streams_play; j++) {
+ stream = &chip->playback_stream[j];
+ if (pcxhr_stream_scheduled_get_pipe(stream, &pipe)) {
+ err = pcxhr_set_format(stream);
+ err = pcxhr_update_r_buffer(stream);
+ }
+ }
+ }
+ /* start all the streams */
+ for (i = 0; i < mgr->num_cards; i++) {
+ struct pcxhr_stream *stream;
+ chip = mgr->chip[i];
+ for (j = 0; j < chip->nb_streams_capt; j++) {
+ stream = &chip->capture_stream[j];
+ if (pcxhr_stream_scheduled_get_pipe(stream, &pipe))
+ err = pcxhr_set_stream_state(stream);
+ }
+ for (j = 0; j < chip->nb_streams_play; j++) {
+ stream = &chip->playback_stream[j];
+ if (pcxhr_stream_scheduled_get_pipe(stream, &pipe))
+ err = pcxhr_set_stream_state(stream);
+ }
+ }
+
+ /* synchronous start of all the pipes concerned */
+ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1);
+ if (err) {
+ mutex_unlock(&mgr->setup_mutex);
+ snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error start pipes (P%x C%x)\n",
+ playback_mask, capture_mask);
+ return;
+ }
+
+ /* put the streams into the running state now (increment pointer by interrupt) */
+ spin_lock_irqsave(&mgr->lock, flags);
+ for ( i =0; i < mgr->num_cards; i++) {
+ struct pcxhr_stream *stream;
+ chip = mgr->chip[i];
+ for(j = 0; j < chip->nb_streams_capt; j++) {
+ stream = &chip->capture_stream[j];
+ if(stream->status == PCXHR_STREAM_STATUS_STARTED)
+ stream->status = PCXHR_STREAM_STATUS_RUNNING;
+ }
+ for (j = 0; j < chip->nb_streams_play; j++) {
+ stream = &chip->playback_stream[j];
+ if (stream->status == PCXHR_STREAM_STATUS_STARTED) {
+ /* playback will already have advanced ! */
+ stream->timer_period_frag += PCXHR_GRANULARITY;
+ stream->status = PCXHR_STREAM_STATUS_RUNNING;
+ }
+ }
+ }
+ spin_unlock_irqrestore(&mgr->lock, flags);
+
+ mutex_unlock(&mgr->setup_mutex);
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ do_gettimeofday(&my_tv2);
+ snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n",
+ (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
+#endif
+}
+
+
+/*
+ * trigger callback
+ */
+static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd)
+{
+ struct pcxhr_stream *stream;
+ struct snd_pcm_substream *s;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_printdd("SNDRV_PCM_TRIGGER_START\n");
+ if (snd_pcm_stream_linked(subs)) {
+ struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
+ snd_pcm_group_for_each_entry(s, subs) {
+ if (snd_pcm_substream_chip(s) != chip)
+ continue;
+ stream = s->runtime->private_data;
+ stream->status =
+ PCXHR_STREAM_STATUS_SCHEDULE_RUN;
+ snd_pcm_trigger_done(s, subs);
+ }
+ tasklet_hi_schedule(&chip->mgr->trigger_taskq);
+ } else {
+ stream = subs->runtime->private_data;
+ snd_printdd("Only one Substream %c %d\n",
+ stream->pipe->is_capture ? 'C' : 'P',
+ stream->pipe->first_audio);
+ if (pcxhr_set_format(stream))
+ return -EINVAL;
+ if (pcxhr_update_r_buffer(stream))
+ return -EINVAL;
+
+ stream->status = PCXHR_STREAM_STATUS_SCHEDULE_RUN;
+ if (pcxhr_set_stream_state(stream))
+ return -EINVAL;
+ stream->status = PCXHR_STREAM_STATUS_RUNNING;
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_printdd("SNDRV_PCM_TRIGGER_STOP\n");
+ snd_pcm_group_for_each_entry(s, subs) {
+ stream = s->runtime->private_data;
+ stream->status = PCXHR_STREAM_STATUS_SCHEDULE_STOP;
+ if (pcxhr_set_stream_state(stream))
+ return -EINVAL;
+ snd_pcm_trigger_done(s, subs);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /* TODO */
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+static int pcxhr_hardware_timer(struct pcxhr_mgr *mgr, int start)
+{
+ struct pcxhr_rmh rmh;
+ int err;
+
+ pcxhr_init_rmh(&rmh, CMD_SET_TIMER_INTERRUPT);
+ if (start) {
+ mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; /* last dsp time invalid */
+ rmh.cmd[0] |= PCXHR_GRANULARITY;
+ }
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err < 0)
+ snd_printk(KERN_ERR "error pcxhr_hardware_timer err(%x)\n", err);
+ return err;
+}
+
+/*
+ * prepare callback for all pcms
+ */
+static int pcxhr_prepare(struct snd_pcm_substream *subs)
+{
+ struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
+ struct pcxhr_mgr *mgr = chip->mgr;
+ /*
+ struct pcxhr_stream *stream = (pcxhr_stream_t*)subs->runtime->private_data;
+ */
+ int err = 0;
+
+ snd_printdd("pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n",
+ subs->runtime->period_size, subs->runtime->periods,
+ subs->runtime->buffer_size);
+
+ /*
+ if(subs->runtime->period_size <= PCXHR_GRANULARITY) {
+ snd_printk(KERN_ERR "pcxhr_prepare : error period_size too small (%x)\n",
+ (unsigned int)subs->runtime->period_size);
+ return -EINVAL;
+ }
+ */
+
+ mutex_lock(&mgr->setup_mutex);
+
+ do {
+ /* if the stream was stopped before, format and buffer were reset */
+ /*
+ if(stream->status == PCXHR_STREAM_STATUS_STOPPED) {
+ err = pcxhr_set_format(stream);
+ if(err) break;
+ err = pcxhr_update_r_buffer(stream);
+ if(err) break;
+ }
+ */
+
+ /* only the first stream can choose the sample rate */
+ /* the further opened streams will be limited to its frequency (see open) */
+ /* set the clock only once (first stream) */
+ if (mgr->sample_rate != subs->runtime->rate) {
+ err = pcxhr_set_clock(mgr, subs->runtime->rate);
+ if (err)
+ break;
+ if (mgr->sample_rate == 0)
+ /* start the DSP-timer */
+ err = pcxhr_hardware_timer(mgr, 1);
+ mgr->sample_rate = subs->runtime->rate;
+ }
+ } while(0); /* do only once (so we can use break instead of goto) */
+
+ mutex_unlock(&mgr->setup_mutex);
+
+ return err;
+}
+
+
+/*
+ * HW_PARAMS callback for all pcms
+ */
+static int pcxhr_hw_params(struct snd_pcm_substream *subs,
+ struct snd_pcm_hw_params *hw)
+{
+ struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
+ struct pcxhr_mgr *mgr = chip->mgr;
+ struct pcxhr_stream *stream = subs->runtime->private_data;
+ snd_pcm_format_t format;
+ int err;
+ int channels;
+
+ /* set up channels */
+ channels = params_channels(hw);
+
+ /* set up format for the stream */
+ format = params_format(hw);
+
+ mutex_lock(&mgr->setup_mutex);
+
+ stream->channels = channels;
+ stream->format = format;
+
+ /* set the format to the board */
+ /*
+ err = pcxhr_set_format(stream);
+ if(err) {
+ mutex_unlock(&mgr->setup_mutex);
+ return err;
+ }
+ */
+ /* allocate buffer */
+ err = snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw));
+
+ /*
+ if (err > 0) {
+ err = pcxhr_update_r_buffer(stream);
+ }
+ */
+ mutex_unlock(&mgr->setup_mutex);
+
+ return err;
+}
+
+static int pcxhr_hw_free(struct snd_pcm_substream *subs)
+{
+ snd_pcm_lib_free_pages(subs);
+ return 0;
+}
+
+
+/*
+ * CONFIGURATION SPACE for all pcms, mono pcm must update channels_max
+ */
+static struct snd_pcm_hardware pcxhr_caps =
+{
+ .info = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ 0 /*SNDRV_PCM_INFO_PAUSE*/),
+ .formats = ( SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
+ SNDRV_PCM_FMTBIT_FLOAT_LE ),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (32*1024),
+ /* 1 byte == 1 frame U8 mono (PCXHR_GRANULARITY is frames!) */
+ .period_bytes_min = (2*PCXHR_GRANULARITY),
+ .period_bytes_max = (16*1024),
+ .periods_min = 2,
+ .periods_max = (32*1024/PCXHR_GRANULARITY),
+};
+
+
+static int pcxhr_open(struct snd_pcm_substream *subs)
+{
+ struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
+ struct pcxhr_mgr *mgr = chip->mgr;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct pcxhr_stream *stream;
+
+ mutex_lock(&mgr->setup_mutex);
+
+ /* copy the struct snd_pcm_hardware struct */
+ runtime->hw = pcxhr_caps;
+
+ if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) {
+ snd_printdd("pcxhr_open playback chip%d subs%d\n",
+ chip->chip_idx, subs->number);
+ stream = &chip->playback_stream[subs->number];
+ } else {
+ snd_printdd("pcxhr_open capture chip%d subs%d\n",
+ chip->chip_idx, subs->number);
+ if (mgr->mono_capture)
+ runtime->hw.channels_max = 1;
+ else
+ runtime->hw.channels_min = 2;
+ stream = &chip->capture_stream[subs->number];
+ }
+ if (stream->status != PCXHR_STREAM_STATUS_FREE){
+ /* streams in use */
+ snd_printk(KERN_ERR "pcxhr_open chip%d subs%d in use\n",
+ chip->chip_idx, subs->number);
+ mutex_unlock(&mgr->setup_mutex);
+ return -EBUSY;
+ }
+
+ /* if a sample rate is already used or fixed by external clock,
+ * the stream cannot change
+ */
+ if (mgr->sample_rate)
+ runtime->hw.rate_min = runtime->hw.rate_max = mgr->sample_rate;
+ else {
+ if (mgr->use_clock_type != PCXHR_CLOCK_TYPE_INTERNAL) {
+ int external_rate;
+ if (pcxhr_get_external_clock(mgr, mgr->use_clock_type,
+ &external_rate) ||
+ external_rate == 0) {
+ /* cannot detect the external clock rate */
+ mutex_unlock(&mgr->setup_mutex);
+ return -EBUSY;
+ }
+ runtime->hw.rate_min = runtime->hw.rate_max = external_rate;
+ }
+ }
+
+ stream->status = PCXHR_STREAM_STATUS_OPEN;
+ stream->substream = subs;
+ stream->channels = 0; /* not configured yet */
+
+ runtime->private_data = stream;
+
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4);
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
+
+ snd_pcm_set_sync(subs);
+
+ mgr->ref_count_rate++;
+
+ mutex_unlock(&mgr->setup_mutex);
+ return 0;
+}
+
+
+static int pcxhr_close(struct snd_pcm_substream *subs)
+{
+ struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
+ struct pcxhr_mgr *mgr = chip->mgr;
+ struct pcxhr_stream *stream = subs->runtime->private_data;
+
+ mutex_lock(&mgr->setup_mutex);
+
+ snd_printdd("pcxhr_close chip%d subs%d\n", chip->chip_idx, subs->number);
+
+ /* sample rate released */
+ if (--mgr->ref_count_rate == 0) {
+ mgr->sample_rate = 0; /* the sample rate is no more locked */
+ pcxhr_hardware_timer(mgr, 0); /* stop the DSP-timer */
+ }
+
+ stream->status = PCXHR_STREAM_STATUS_FREE;
+ stream->substream = NULL;
+
+ mutex_unlock(&mgr->setup_mutex);
+
+ return 0;
+}
+
+
+static snd_pcm_uframes_t pcxhr_stream_pointer(struct snd_pcm_substream *subs)
+{
+ unsigned long flags;
+ u_int32_t timer_period_frag;
+ int timer_buf_periods;
+ struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct pcxhr_stream *stream = runtime->private_data;
+
+ spin_lock_irqsave(&chip->mgr->lock, flags);
+
+ /* get the period fragment and the nb of periods in the buffer */
+ timer_period_frag = stream->timer_period_frag;
+ timer_buf_periods = stream->timer_buf_periods;
+
+ spin_unlock_irqrestore(&chip->mgr->lock, flags);
+
+ return (snd_pcm_uframes_t)((timer_buf_periods * runtime->period_size) +
+ timer_period_frag);
+}
+
+
+static struct snd_pcm_ops pcxhr_ops = {
+ .open = pcxhr_open,
+ .close = pcxhr_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = pcxhr_prepare,
+ .hw_params = pcxhr_hw_params,
+ .hw_free = pcxhr_hw_free,
+ .trigger = pcxhr_trigger,
+ .pointer = pcxhr_stream_pointer,
+};
+
+/*
+ */
+int pcxhr_create_pcm(struct snd_pcxhr *chip)
+{
+ int err;
+ struct snd_pcm *pcm;
+ char name[32];
+
+ sprintf(name, "pcxhr %d", chip->chip_idx);
+ if ((err = snd_pcm_new(chip->card, name, 0,
+ chip->nb_streams_play,
+ chip->nb_streams_capt, &pcm)) < 0) {
+ snd_printk(KERN_ERR "cannot create pcm %s\n", name);
+ return err;
+ }
+ pcm->private_data = chip;
+
+ if (chip->nb_streams_play)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcxhr_ops);
+ if (chip->nb_streams_capt)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcxhr_ops);
+
+ pcm->info_flags = 0;
+ strcpy(pcm->name, name);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->mgr->pci),
+ 32*1024, 32*1024);
+ chip->pcm = pcm;
+ return 0;
+}
+
+static int pcxhr_chip_free(struct snd_pcxhr *chip)
+{
+ kfree(chip);
+ return 0;
+}
+
+static int pcxhr_chip_dev_free(struct snd_device *device)
+{
+ struct snd_pcxhr *chip = device->device_data;
+ return pcxhr_chip_free(chip);
+}
+
+
+/*
+ */
+static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, int idx)
+{
+ int err;
+ struct snd_pcxhr *chip;
+ static struct snd_device_ops ops = {
+ .dev_free = pcxhr_chip_dev_free,
+ };
+
+ mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (! chip) {
+ snd_printk(KERN_ERR "cannot allocate chip\n");
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->chip_idx = idx;
+ chip->mgr = mgr;
+
+ if (idx < mgr->playback_chips)
+ /* stereo or mono streams */
+ chip->nb_streams_play = PCXHR_PLAYBACK_STREAMS;
+
+ if (idx < mgr->capture_chips) {
+ if (mgr->mono_capture)
+ chip->nb_streams_capt = 2; /* 2 mono streams (left+right) */
+ else
+ chip->nb_streams_capt = 1; /* or 1 stereo stream */
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ pcxhr_chip_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &mgr->pci->dev);
+
+ return 0;
+}
+
+/* proc interface */
+static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_pcxhr *chip = entry->private_data;
+ struct pcxhr_mgr *mgr = chip->mgr;
+
+ snd_iprintf(buffer, "\n%s\n", mgr->longname);
+
+ /* stats available when embedded DSP is running */
+ if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) {
+ struct pcxhr_rmh rmh;
+ short ver_maj = (mgr->dsp_version >> 16) & 0xff;
+ short ver_min = (mgr->dsp_version >> 8) & 0xff;
+ short ver_build = mgr->dsp_version & 0xff;
+ snd_iprintf(buffer, "module version %s\n", PCXHR_DRIVER_VERSION_STRING);
+ snd_iprintf(buffer, "dsp version %d.%d.%d\n", ver_maj, ver_min, ver_build);
+ if (mgr->board_has_analog)
+ snd_iprintf(buffer, "analog io available\n");
+ else
+ snd_iprintf(buffer, "digital only board\n");
+
+ /* calc cpu load of the dsp */
+ pcxhr_init_rmh(&rmh, CMD_GET_DSP_RESOURCES);
+ if( ! pcxhr_send_msg(mgr, &rmh) ) {
+ int cur = rmh.stat[0];
+ int ref = rmh.stat[1];
+ if (ref > 0) {
+ if (mgr->sample_rate_real != 0 &&
+ mgr->sample_rate_real != 48000) {
+ ref = (ref * 48000) / mgr->sample_rate_real;
+ if (mgr->sample_rate_real >= PCXHR_IRQ_TIMER_FREQ)
+ ref *= 2;
+ }
+ cur = 100 - (100 * cur) / ref;
+ snd_iprintf(buffer, "cpu load %d%%\n", cur);
+ snd_iprintf(buffer, "buffer pool %d/%d kWords\n",
+ rmh.stat[2], rmh.stat[3]);
+ }
+ }
+ snd_iprintf(buffer, "dma granularity : %d\n", PCXHR_GRANULARITY);
+ snd_iprintf(buffer, "dsp time errors : %d\n", mgr->dsp_time_err);
+ snd_iprintf(buffer, "dsp async pipe xrun errors : %d\n",
+ mgr->async_err_pipe_xrun);
+ snd_iprintf(buffer, "dsp async stream xrun errors : %d\n",
+ mgr->async_err_stream_xrun);
+ snd_iprintf(buffer, "dsp async last other error : %x\n",
+ mgr->async_err_other_last);
+ /* debug zone dsp */
+ rmh.cmd[0] = 0x4200 + PCXHR_SIZE_MAX_STATUS;
+ rmh.cmd_len = 1;
+ rmh.stat_len = PCXHR_SIZE_MAX_STATUS;
+ rmh.dsp_stat = 0;
+ rmh.cmd_idx = CMD_LAST_INDEX;
+ if( ! pcxhr_send_msg(mgr, &rmh) ) {
+ int i;
+ for (i = 0; i < rmh.stat_len; i++)
+ snd_iprintf(buffer, "debug[%02d] = %06x\n", i, rmh.stat[i]);
+ }
+ } else
+ snd_iprintf(buffer, "no firmware loaded\n");
+ snd_iprintf(buffer, "\n");
+}
+static void pcxhr_proc_sync(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_pcxhr *chip = entry->private_data;
+ struct pcxhr_mgr *mgr = chip->mgr;
+ static char *texts[7] = {
+ "Internal", "Word", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4"
+ };
+
+ snd_iprintf(buffer, "\n%s\n", mgr->longname);
+ snd_iprintf(buffer, "Current Sample Clock\t: %s\n", texts[mgr->cur_clock_type]);
+ snd_iprintf(buffer, "Current Sample Rate\t= %d\n", mgr->sample_rate_real);
+
+ /* commands available when embedded DSP is running */
+ if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) {
+ int i, err, sample_rate;
+ for (i = PCXHR_CLOCK_TYPE_WORD_CLOCK; i< (3 + mgr->capture_chips); i++) {
+ err = pcxhr_get_external_clock(mgr, i, &sample_rate);
+ if (err)
+ break;
+ snd_iprintf(buffer, "%s Clock\t\t= %d\n", texts[i], sample_rate);
+ }
+ } else
+ snd_iprintf(buffer, "no firmware loaded\n");
+ snd_iprintf(buffer, "\n");
+}
+
+static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "info", &entry))
+ snd_info_set_text_ops(entry, chip, pcxhr_proc_info);
+ if (! snd_card_proc_new(chip->card, "sync", &entry))
+ snd_info_set_text_ops(entry, chip, pcxhr_proc_sync);
+}
+/* end of proc interface */
+
+/*
+ * release all the cards assigned to a manager instance
+ */
+static int pcxhr_free(struct pcxhr_mgr *mgr)
+{
+ unsigned int i;
+
+ for (i = 0; i < mgr->num_cards; i++) {
+ if (mgr->chip[i])
+ snd_card_free(mgr->chip[i]->card);
+ }
+
+ /* reset board if some firmware was loaded */
+ if(mgr->dsp_loaded) {
+ pcxhr_reset_board(mgr);
+ snd_printdd("reset pcxhr !\n");
+ }
+
+ /* release irq */
+ if (mgr->irq >= 0)
+ free_irq(mgr->irq, mgr);
+
+ pci_release_regions(mgr->pci);
+
+ /* free hostport purgebuffer */
+ if (mgr->hostport.area) {
+ snd_dma_free_pages(&mgr->hostport);
+ mgr->hostport.area = NULL;
+ }
+
+ kfree(mgr->prmh);
+
+ pci_disable_device(mgr->pci);
+ kfree(mgr);
+ return 0;
+}
+
+/*
+ * probe function - creates the card manager
+ */
+static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct pcxhr_mgr *mgr;
+ unsigned int i;
+ int err;
+ size_t size;
+ char *card_name;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (! enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ pci_set_master(pci);
+
+ /* check if we can restrict PCI DMA transfers to 32 bits */
+ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ /* alloc card manager */
+ mgr = kzalloc(sizeof(*mgr), GFP_KERNEL);
+ if (! mgr) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) {
+ kfree(mgr);
+ pci_disable_device(pci);
+ return -ENODEV;
+ }
+ card_name = pcxhr_board_params[pci_id->driver_data].board_name;
+ mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips;
+ mgr->capture_chips = pcxhr_board_params[pci_id->driver_data].capture_chips;
+ mgr->firmware_num = pcxhr_board_params[pci_id->driver_data].firmware_num;
+ mgr->mono_capture = mono[dev];
+
+ /* resource assignment */
+ if ((err = pci_request_regions(pci, card_name)) < 0) {
+ kfree(mgr);
+ pci_disable_device(pci);
+ return err;
+ }
+ for (i = 0; i < 3; i++)
+ mgr->port[i] = pci_resource_start(pci, i);
+
+ mgr->pci = pci;
+ mgr->irq = -1;
+
+ if (request_irq(pci->irq, pcxhr_interrupt, IRQF_SHARED,
+ card_name, mgr)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ pcxhr_free(mgr);
+ return -EBUSY;
+ }
+ mgr->irq = pci->irq;
+
+ sprintf(mgr->shortname, "Digigram %s", card_name);
+ sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, 0x%lx irq %i", mgr->shortname,
+ mgr->port[0], mgr->port[1], mgr->port[2], mgr->irq);
+
+ /* ISR spinlock */
+ spin_lock_init(&mgr->lock);
+ spin_lock_init(&mgr->msg_lock);
+
+ /* init setup mutex*/
+ mutex_init(&mgr->setup_mutex);
+
+ /* init taslket */
+ tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, (unsigned long) mgr);
+ tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet, (unsigned long) mgr);
+ mgr->prmh = kmalloc(sizeof(*mgr->prmh) +
+ sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS),
+ GFP_KERNEL);
+ if (! mgr->prmh) {
+ pcxhr_free(mgr);
+ return -ENOMEM;
+ }
+
+ for (i=0; i < PCXHR_MAX_CARDS; i++) {
+ struct snd_card *card;
+ char tmpid[16];
+ int idx;
+
+ if (i >= max(mgr->playback_chips, mgr->capture_chips))
+ break;
+ mgr->num_cards++;
+
+ if (index[dev] < 0)
+ idx = index[dev];
+ else
+ idx = index[dev] + i;
+
+ snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : card_name, i);
+ card = snd_card_new(idx, tmpid, THIS_MODULE, 0);
+
+ if (! card) {
+ snd_printk(KERN_ERR "cannot allocate the card %d\n", i);
+ pcxhr_free(mgr);
+ return -ENOMEM;
+ }
+
+ strcpy(card->driver, DRIVER_NAME);
+ sprintf(card->shortname, "%s [PCM #%d]", mgr->shortname, i);
+ sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i);
+
+ if ((err = pcxhr_create(mgr, card, i)) < 0) {
+ pcxhr_free(mgr);
+ return err;
+ }
+
+ if (i == 0)
+ /* init proc interface only for chip0 */
+ pcxhr_proc_init(mgr->chip[i]);
+
+ if ((err = snd_card_register(card)) < 0) {
+ pcxhr_free(mgr);
+ return err;
+ }
+ }
+
+ /* create hostport purgebuffer */
+ size = PAGE_ALIGN(sizeof(struct pcxhr_hostport));
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, &mgr->hostport) < 0) {
+ pcxhr_free(mgr);
+ return -ENOMEM;
+ }
+ /* init purgebuffer */
+ memset(mgr->hostport.area, 0, size);
+
+ /* create a DSP loader */
+ err = pcxhr_setup_firmware(mgr);
+ if (err < 0) {
+ pcxhr_free(mgr);
+ return err;
+ }
+
+ pci_set_drvdata(pci, mgr);
+ dev++;
+ return 0;
+}
+
+static void __devexit pcxhr_remove(struct pci_dev *pci)
+{
+ pcxhr_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Digigram pcxhr",
+ .id_table = pcxhr_ids,
+ .probe = pcxhr_probe,
+ .remove = __devexit_p(pcxhr_remove),
+};
+
+static int __init pcxhr_module_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit pcxhr_module_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(pcxhr_module_init)
+module_exit(pcxhr_module_exit)
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
new file mode 100644
index 0000000..6520647
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -0,0 +1,189 @@
+/*
+ * Driver for Digigram pcxhr soundcards
+ *
+ * main header file
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_PCXHR_H
+#define __SOUND_PCXHR_H
+
+#include <linux/interrupt.h>
+#include <linux/mutex.h>
+#include <sound/pcm.h>
+
+#define PCXHR_DRIVER_VERSION 0x000804 /* 0.8.4 */
+#define PCXHR_DRIVER_VERSION_STRING "0.8.4" /* 0.8.4 */
+
+
+#define PCXHR_MAX_CARDS 6
+#define PCXHR_PLAYBACK_STREAMS 4
+
+#define PCXHR_GRANULARITY 96 /* transfer granularity (should be min 96 and multiple of 48) */
+#define PCXHR_GRANULARITY_MIN 96 /* transfer granularity of pipes and the dsp time (MBOX4) */
+
+struct snd_pcxhr;
+struct pcxhr_mgr;
+
+struct pcxhr_stream;
+struct pcxhr_pipe;
+
+enum pcxhr_clock_type {
+ PCXHR_CLOCK_TYPE_INTERNAL = 0,
+ PCXHR_CLOCK_TYPE_WORD_CLOCK,
+ PCXHR_CLOCK_TYPE_AES_SYNC,
+ PCXHR_CLOCK_TYPE_AES_1,
+ PCXHR_CLOCK_TYPE_AES_2,
+ PCXHR_CLOCK_TYPE_AES_3,
+ PCXHR_CLOCK_TYPE_AES_4,
+};
+
+struct pcxhr_mgr {
+ unsigned int num_cards;
+ struct snd_pcxhr *chip[PCXHR_MAX_CARDS];
+
+ struct pci_dev *pci;
+
+ int irq;
+
+ /* card access with 1 mem bar and 2 io bar's */
+ unsigned long port[3];
+
+ /* share the name */
+ char shortname[32]; /* short name of this soundcard */
+ char longname[96]; /* name of this soundcard */
+
+ /* message tasklet */
+ struct tasklet_struct msg_taskq;
+ struct pcxhr_rmh *prmh;
+ /* trigger tasklet */
+ struct tasklet_struct trigger_taskq;
+
+ spinlock_t lock; /* interrupt spinlock */
+ spinlock_t msg_lock; /* message spinlock */
+
+ struct mutex setup_mutex; /* mutex used in hw_params, open and close */
+ struct mutex mixer_mutex; /* mutex for mixer */
+
+ /* hardware interface */
+ unsigned int dsp_loaded; /* bit flags of loaded dsp indices */
+ unsigned int dsp_version; /* read from embedded once firmware is loaded */
+ int board_has_analog; /* if 0 the board is digital only */
+ int mono_capture; /* if 1 the board does mono capture */
+ int playback_chips; /* 4 or 6 */
+ int capture_chips; /* 4 or 1 */
+ int firmware_num; /* 41 or 42 */
+
+ struct snd_dma_buffer hostport;
+
+ enum pcxhr_clock_type use_clock_type; /* clock type selected by mixer */
+ enum pcxhr_clock_type cur_clock_type; /* current clock type synced */
+ int sample_rate;
+ int ref_count_rate;
+ int timer_toggle; /* timer interrupt toggles between the two values 0x200 and 0x300 */
+ int dsp_time_last; /* the last dsp time (read by interrupt) */
+ int dsp_time_err; /* dsp time errors */
+ unsigned int src_it_dsp; /* dsp interrupt source */
+ unsigned int io_num_reg_cont; /* backup of IO_NUM_REG_CONT */
+ unsigned int codec_speed; /* speed mode of the codecs */
+ unsigned int sample_rate_real; /* current real sample rate */
+ int last_reg_stat;
+ int async_err_stream_xrun;
+ int async_err_pipe_xrun;
+ int async_err_other_last;
+};
+
+
+enum pcxhr_stream_status {
+ PCXHR_STREAM_STATUS_FREE,
+ PCXHR_STREAM_STATUS_OPEN,
+ PCXHR_STREAM_STATUS_SCHEDULE_RUN,
+ PCXHR_STREAM_STATUS_STARTED,
+ PCXHR_STREAM_STATUS_RUNNING,
+ PCXHR_STREAM_STATUS_SCHEDULE_STOP,
+ PCXHR_STREAM_STATUS_STOPPED,
+ PCXHR_STREAM_STATUS_PAUSED
+};
+
+struct pcxhr_stream {
+ struct snd_pcm_substream *substream;
+ snd_pcm_format_t format;
+ struct pcxhr_pipe *pipe;
+
+ enum pcxhr_stream_status status; /* free, open, running, draining, pause */
+
+ u_int64_t timer_abs_periods; /* timer: samples elapsed since TRIGGER_START (multiple of period_size) */
+ u_int32_t timer_period_frag; /* timer: samples elapsed since last call to snd_pcm_period_elapsed (0..period_size) */
+ u_int32_t timer_buf_periods; /* nb of periods in the buffer that have already elapsed */
+ int timer_is_synced; /* if(0) : timer needs to be resynced with real hardware pointer */
+
+ int channels;
+};
+
+
+enum pcxhr_pipe_status {
+ PCXHR_PIPE_UNDEFINED,
+ PCXHR_PIPE_DEFINED
+};
+
+struct pcxhr_pipe {
+ enum pcxhr_pipe_status status;
+ int is_capture; /* this is a capture pipe */
+ int first_audio; /* first audio num */
+};
+
+
+struct snd_pcxhr {
+ struct snd_card *card;
+ struct pcxhr_mgr *mgr;
+ int chip_idx; /* zero based */
+
+ struct snd_pcm *pcm; /* PCM */
+
+ struct pcxhr_pipe playback_pipe; /* 1 stereo pipe only */
+ struct pcxhr_pipe capture_pipe[2]; /* 1 stereo pipe or 2 mono pipes */
+
+ struct pcxhr_stream playback_stream[PCXHR_PLAYBACK_STREAMS];
+ struct pcxhr_stream capture_stream[2]; /* 1 stereo stream or 2 mono streams */
+ int nb_streams_play;
+ int nb_streams_capt;
+
+ int analog_playback_active[2]; /* Mixer : Master Playback active (!mute) */
+ int analog_playback_volume[2]; /* Mixer : Master Playback Volume */
+ int analog_capture_volume[2]; /* Mixer : Master Capture Volume */
+ int digital_playback_active[PCXHR_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Active [streams][stereo]*/
+ int digital_playback_volume[PCXHR_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Volume [streams][stereo]*/
+ int digital_capture_volume[2]; /* Mixer : Digital Capture Volume [stereo] */
+ int monitoring_active[2]; /* Mixer : Monitoring Active */
+ int monitoring_volume[2]; /* Mixer : Monitoring Volume */
+ int audio_capture_source; /* Mixer : Audio Capture Source */
+ unsigned char aes_bits[5]; /* Mixer : IEC958_AES bits */
+};
+
+struct pcxhr_hostport
+{
+ char purgebuffer[6];
+ char reserved[2];
+};
+
+/* exported */
+int pcxhr_create_pcm(struct snd_pcxhr *chip);
+int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate);
+int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, int *sample_rate);
+
+#endif /* __SOUND_PCXHR_H */
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
new file mode 100644
index 0000000..7143259
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -0,0 +1,1224 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * low level interface with interrupt and message handling implementation
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/firmware.h>
+#include <linux/interrupt.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include "pcxhr.h"
+#include "pcxhr_mixer.h"
+#include "pcxhr_hwdep.h"
+#include "pcxhr_core.h"
+
+
+/* registers used on the PLX (port 1) */
+#define PCXHR_PLX_OFFSET_MIN 0x40
+#define PCXHR_PLX_MBOX0 0x40
+#define PCXHR_PLX_MBOX1 0x44
+#define PCXHR_PLX_MBOX2 0x48
+#define PCXHR_PLX_MBOX3 0x4C
+#define PCXHR_PLX_MBOX4 0x50
+#define PCXHR_PLX_MBOX5 0x54
+#define PCXHR_PLX_MBOX6 0x58
+#define PCXHR_PLX_MBOX7 0x5C
+#define PCXHR_PLX_L2PCIDB 0x64
+#define PCXHR_PLX_IRQCS 0x68
+#define PCXHR_PLX_CHIPSC 0x6C
+
+/* registers used on the DSP (port 2) */
+#define PCXHR_DSP_ICR 0x00
+#define PCXHR_DSP_CVR 0x04
+#define PCXHR_DSP_ISR 0x08
+#define PCXHR_DSP_IVR 0x0C
+#define PCXHR_DSP_RXH 0x14
+#define PCXHR_DSP_TXH 0x14
+#define PCXHR_DSP_RXM 0x18
+#define PCXHR_DSP_TXM 0x18
+#define PCXHR_DSP_RXL 0x1C
+#define PCXHR_DSP_TXL 0x1C
+#define PCXHR_DSP_RESET 0x20
+#define PCXHR_DSP_OFFSET_MAX 0x20
+
+/* access to the card */
+#define PCXHR_PLX 1
+#define PCXHR_DSP 2
+
+#if (PCXHR_DSP_OFFSET_MAX > PCXHR_PLX_OFFSET_MIN)
+#undef PCXHR_REG_TO_PORT(x)
+#else
+#define PCXHR_REG_TO_PORT(x) ((x)>PCXHR_DSP_OFFSET_MAX ? PCXHR_PLX : PCXHR_DSP)
+#endif
+#define PCXHR_INPB(mgr,x) inb((mgr)->port[PCXHR_REG_TO_PORT(x)] + (x))
+#define PCXHR_INPL(mgr,x) inl((mgr)->port[PCXHR_REG_TO_PORT(x)] + (x))
+#define PCXHR_OUTPB(mgr,x,data) outb((data), (mgr)->port[PCXHR_REG_TO_PORT(x)] + (x))
+#define PCXHR_OUTPL(mgr,x,data) outl((data), (mgr)->port[PCXHR_REG_TO_PORT(x)] + (x))
+/* attention : access the PCXHR_DSP_* registers with inb and outb only ! */
+
+/* params used with PCXHR_PLX_MBOX0 */
+#define PCXHR_MBOX0_HF5 (1 << 0)
+#define PCXHR_MBOX0_HF4 (1 << 1)
+#define PCXHR_MBOX0_BOOT_HERE (1 << 23)
+/* params used with PCXHR_PLX_IRQCS */
+#define PCXHR_IRQCS_ENABLE_PCIIRQ (1 << 8)
+#define PCXHR_IRQCS_ENABLE_PCIDB (1 << 9)
+#define PCXHR_IRQCS_ACTIVE_PCIDB (1 << 13)
+/* params used with PCXHR_PLX_CHIPSC */
+#define PCXHR_CHIPSC_INIT_VALUE 0x100D767E
+#define PCXHR_CHIPSC_RESET_XILINX (1 << 16)
+#define PCXHR_CHIPSC_GPI_USERI (1 << 17)
+#define PCXHR_CHIPSC_DATA_CLK (1 << 24)
+#define PCXHR_CHIPSC_DATA_IN (1 << 26)
+
+/* params used with PCXHR_DSP_ICR */
+#define PCXHR_ICR_HI08_RREQ 0x01
+#define PCXHR_ICR_HI08_TREQ 0x02
+#define PCXHR_ICR_HI08_HDRQ 0x04
+#define PCXHR_ICR_HI08_HF0 0x08
+#define PCXHR_ICR_HI08_HF1 0x10
+#define PCXHR_ICR_HI08_HLEND 0x20
+#define PCXHR_ICR_HI08_INIT 0x80
+/* params used with PCXHR_DSP_CVR */
+#define PCXHR_CVR_HI08_HC 0x80
+/* params used with PCXHR_DSP_ISR */
+#define PCXHR_ISR_HI08_RXDF 0x01
+#define PCXHR_ISR_HI08_TXDE 0x02
+#define PCXHR_ISR_HI08_TRDY 0x04
+#define PCXHR_ISR_HI08_ERR 0x08
+#define PCXHR_ISR_HI08_CHK 0x10
+#define PCXHR_ISR_HI08_HREQ 0x80
+
+
+/* constants used for delay in msec */
+#define PCXHR_WAIT_DEFAULT 2
+#define PCXHR_WAIT_IT 25
+#define PCXHR_WAIT_IT_EXTRA 65
+
+/*
+ * pcxhr_check_reg_bit - wait for the specified bit is set/reset on a register
+ * @reg: register to check
+ * @mask: bit mask
+ * @bit: resultant bit to be checked
+ * @time: time-out of loop in msec
+ *
+ * returns zero if a bit matches, or a negative error code.
+ */
+static int pcxhr_check_reg_bit(struct pcxhr_mgr *mgr, unsigned int reg,
+ unsigned char mask, unsigned char bit, int time,
+ unsigned char* read)
+{
+ int i = 0;
+ unsigned long end_time = jiffies + (time * HZ + 999) / 1000;
+ do {
+ *read = PCXHR_INPB(mgr, reg);
+ if ((*read & mask) == bit) {
+ if (i > 100)
+ snd_printdd("ATTENTION! check_reg(%x) loopcount=%d\n",
+ reg, i);
+ return 0;
+ }
+ i++;
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "pcxhr_check_reg_bit: timeout, reg=%x, mask=0x%x, val=0x%x\n",
+ reg, mask, *read);
+ return -EIO;
+}
+
+/* constants used with pcxhr_check_reg_bit() */
+#define PCXHR_TIMEOUT_DSP 200
+
+
+#define PCXHR_MASK_EXTRA_INFO 0x0000FE
+#define PCXHR_MASK_IT_HF0 0x000100
+#define PCXHR_MASK_IT_HF1 0x000200
+#define PCXHR_MASK_IT_NO_HF0_HF1 0x000400
+#define PCXHR_MASK_IT_MANAGE_HF5 0x000800
+#define PCXHR_MASK_IT_WAIT 0x010000
+#define PCXHR_MASK_IT_WAIT_EXTRA 0x020000
+
+#define PCXHR_IT_SEND_BYTE_XILINX (0x0000003C | PCXHR_MASK_IT_HF0)
+#define PCXHR_IT_TEST_XILINX (0x0000003C | PCXHR_MASK_IT_HF1 | \
+ PCXHR_MASK_IT_MANAGE_HF5)
+#define PCXHR_IT_DOWNLOAD_BOOT (0x0000000C | PCXHR_MASK_IT_HF1 | \
+ PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT)
+#define PCXHR_IT_RESET_BOARD_FUNC (0x0000000C | PCXHR_MASK_IT_HF0 | \
+ PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT_EXTRA)
+#define PCXHR_IT_DOWNLOAD_DSP (0x0000000C | \
+ PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT)
+#define PCXHR_IT_DEBUG (0x0000005A | PCXHR_MASK_IT_NO_HF0_HF1)
+#define PCXHR_IT_RESET_SEMAPHORE (0x0000005C | PCXHR_MASK_IT_NO_HF0_HF1)
+#define PCXHR_IT_MESSAGE (0x00000074 | PCXHR_MASK_IT_NO_HF0_HF1)
+#define PCXHR_IT_RESET_CHK (0x00000076 | PCXHR_MASK_IT_NO_HF0_HF1)
+#define PCXHR_IT_UPDATE_RBUFFER (0x00000078 | PCXHR_MASK_IT_NO_HF0_HF1)
+
+static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atomic)
+{
+ int err;
+ unsigned char reg;
+
+ if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) {
+ /* clear hf5 bit */
+ PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX0,
+ PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & ~PCXHR_MBOX0_HF5);
+ }
+ if ((itdsp & PCXHR_MASK_IT_NO_HF0_HF1) == 0) {
+ reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ;
+ if (itdsp & PCXHR_MASK_IT_HF0)
+ reg |= PCXHR_ICR_HI08_HF0;
+ if (itdsp & PCXHR_MASK_IT_HF1)
+ reg |= PCXHR_ICR_HI08_HF1;
+ PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg);
+ }
+ reg = (unsigned char)(((itdsp & PCXHR_MASK_EXTRA_INFO) >> 1) | PCXHR_CVR_HI08_HC);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_CVR, reg);
+ if (itdsp & PCXHR_MASK_IT_WAIT) {
+ if (atomic)
+ mdelay(PCXHR_WAIT_IT);
+ else
+ msleep(PCXHR_WAIT_IT);
+ }
+ if (itdsp & PCXHR_MASK_IT_WAIT_EXTRA) {
+ if (atomic)
+ mdelay(PCXHR_WAIT_IT_EXTRA);
+ else
+ msleep(PCXHR_WAIT_IT);
+ }
+ /* wait for CVR_HI08_HC == 0 */
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_CVR, PCXHR_CVR_HI08_HC, 0,
+ PCXHR_TIMEOUT_DSP, &reg);
+ if (err) {
+ snd_printk(KERN_ERR "pcxhr_send_it_dsp : TIMEOUT CVR\n");
+ return err;
+ }
+ if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) {
+ /* wait for hf5 bit */
+ err = pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0, PCXHR_MBOX0_HF5,
+ PCXHR_MBOX0_HF5, PCXHR_TIMEOUT_DSP, &reg);
+ if (err) {
+ snd_printk(KERN_ERR "pcxhr_send_it_dsp : TIMEOUT HF5\n");
+ return err;
+ }
+ }
+ return 0; /* retry not handled here */
+}
+
+void pcxhr_reset_xilinx_com(struct pcxhr_mgr *mgr)
+{
+ /* reset second xilinx */
+ PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC,
+ PCXHR_CHIPSC_INIT_VALUE & ~PCXHR_CHIPSC_RESET_XILINX);
+}
+
+static void pcxhr_enable_irq(struct pcxhr_mgr *mgr, int enable)
+{
+ unsigned int reg = PCXHR_INPL(mgr, PCXHR_PLX_IRQCS);
+ /* enable/disable interrupts */
+ if (enable)
+ reg |= (PCXHR_IRQCS_ENABLE_PCIIRQ | PCXHR_IRQCS_ENABLE_PCIDB);
+ else
+ reg &= ~(PCXHR_IRQCS_ENABLE_PCIIRQ | PCXHR_IRQCS_ENABLE_PCIDB);
+ PCXHR_OUTPL(mgr, PCXHR_PLX_IRQCS, reg);
+}
+
+void pcxhr_reset_dsp(struct pcxhr_mgr *mgr)
+{
+ /* disable interrupts */
+ pcxhr_enable_irq(mgr, 0);
+
+ /* let's reset the DSP */
+ PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, 0);
+ msleep( PCXHR_WAIT_DEFAULT ); /* wait 2 msec */
+ PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, 3);
+ msleep( PCXHR_WAIT_DEFAULT ); /* wait 2 msec */
+
+ /* reset mailbox */
+ PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX0, 0);
+}
+
+void pcxhr_enable_dsp(struct pcxhr_mgr *mgr)
+{
+ /* enable interrupts */
+ pcxhr_enable_irq(mgr, 1);
+}
+
+/*
+ * load the xilinx image
+ */
+int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilinx, int second)
+{
+ unsigned int i;
+ unsigned int chipsc;
+ unsigned char data;
+ unsigned char mask;
+ const unsigned char *image;
+
+ /* test first xilinx */
+ chipsc = PCXHR_INPL(mgr, PCXHR_PLX_CHIPSC);
+ /* REV01 cards do not support the PCXHR_CHIPSC_GPI_USERI bit anymore */
+ /* this bit will always be 1; no possibility to test presence of first xilinx */
+ if(second) {
+ if ((chipsc & PCXHR_CHIPSC_GPI_USERI) == 0) {
+ snd_printk(KERN_ERR "error loading first xilinx\n");
+ return -EINVAL;
+ }
+ /* activate second xilinx */
+ chipsc |= PCXHR_CHIPSC_RESET_XILINX;
+ PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc);
+ msleep( PCXHR_WAIT_DEFAULT ); /* wait 2 msec */
+ }
+ image = xilinx->data;
+ for (i = 0; i < xilinx->size; i++, image++) {
+ data = *image;
+ mask = 0x80;
+ while (mask) {
+ chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | PCXHR_CHIPSC_DATA_IN);
+ if (data & mask)
+ chipsc |= PCXHR_CHIPSC_DATA_IN;
+ PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc);
+ chipsc |= PCXHR_CHIPSC_DATA_CLK;
+ PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc);
+ mask >>= 1;
+ }
+ /* don't take too much time in this loop... */
+ cond_resched();
+ }
+ chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | PCXHR_CHIPSC_DATA_IN);
+ PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc);
+ /* wait 2 msec (time to boot the xilinx before any access) */
+ msleep( PCXHR_WAIT_DEFAULT );
+ return 0;
+}
+
+/*
+ * send an executable file to the DSP
+ */
+static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp)
+{
+ int err;
+ unsigned int i;
+ unsigned int len;
+ const unsigned char *data;
+ unsigned char dummy;
+ /* check the length of boot image */
+ if (dsp->size <= 0)
+ return -EINVAL;
+ if (dsp->size % 3)
+ return -EINVAL;
+ if (snd_BUG_ON(!dsp->data))
+ return -EINVAL;
+ /* transfert data buffer from PC to DSP */
+ for (i = 0; i < dsp->size; i += 3) {
+ data = dsp->data + i;
+ if (i == 0) {
+ /* test data header consistency */
+ len = (unsigned int)((data[0]<<16) + (data[1]<<8) + data[2]);
+ if (len && dsp->size != (len + 2) * 3)
+ return -EINVAL;
+ }
+ /* wait DSP ready for new transfer */
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY,
+ PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, &dummy);
+ if (err) {
+ snd_printk(KERN_ERR "dsp loading error at position %d\n", i);
+ return err;
+ }
+ /* send host data */
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXH, data[0]);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXM, data[1]);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXL, data[2]);
+
+ /* don't take too much time in this loop... */
+ cond_resched();
+ }
+ /* give some time to boot the DSP */
+ msleep(PCXHR_WAIT_DEFAULT);
+ return 0;
+}
+
+/*
+ * load the eeprom image
+ */
+int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eeprom)
+{
+ int err;
+ unsigned char reg;
+
+ /* init value of the ICR register */
+ reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ;
+ if (PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & PCXHR_MBOX0_BOOT_HERE) {
+ /* no need to load the eeprom binary, but init the HI08 interface */
+ PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg | PCXHR_ICR_HI08_INIT);
+ msleep(PCXHR_WAIT_DEFAULT);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg);
+ msleep(PCXHR_WAIT_DEFAULT);
+ snd_printdd("no need to load eeprom boot\n");
+ return 0;
+ }
+ PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg);
+
+ err = pcxhr_download_dsp(mgr, eeprom);
+ if (err)
+ return err;
+ /* wait for chk bit */
+ return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK,
+ PCXHR_ISR_HI08_CHK, PCXHR_TIMEOUT_DSP, &reg);
+}
+
+/*
+ * load the boot image
+ */
+int pcxhr_load_boot_binary(struct pcxhr_mgr *mgr, const struct firmware *boot)
+{
+ int err;
+ unsigned int physaddr = mgr->hostport.addr;
+ unsigned char dummy;
+
+ /* send the hostport address to the DSP (only the upper 24 bit !) */
+ if (snd_BUG_ON(physaddr & 0xff))
+ return -EINVAL;
+ PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX1, (physaddr >> 8));
+
+ err = pcxhr_send_it_dsp(mgr, PCXHR_IT_DOWNLOAD_BOOT, 0);
+ if (err)
+ return err;
+ /* clear hf5 bit */
+ PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX0,
+ PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & ~PCXHR_MBOX0_HF5);
+
+ err = pcxhr_download_dsp(mgr, boot);
+ if (err)
+ return err;
+ /* wait for hf5 bit */
+ return pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0, PCXHR_MBOX0_HF5,
+ PCXHR_MBOX0_HF5, PCXHR_TIMEOUT_DSP, &dummy);
+}
+
+/*
+ * load the final dsp image
+ */
+int pcxhr_load_dsp_binary(struct pcxhr_mgr *mgr, const struct firmware *dsp)
+{
+ int err;
+ unsigned char dummy;
+ err = pcxhr_send_it_dsp(mgr, PCXHR_IT_RESET_BOARD_FUNC, 0);
+ if (err)
+ return err;
+ err = pcxhr_send_it_dsp(mgr, PCXHR_IT_DOWNLOAD_DSP, 0);
+ if (err)
+ return err;
+ err = pcxhr_download_dsp(mgr, dsp);
+ if (err)
+ return err;
+ /* wait for chk bit */
+ return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK,
+ PCXHR_ISR_HI08_CHK, PCXHR_TIMEOUT_DSP, &dummy);
+}
+
+
+struct pcxhr_cmd_info {
+ u32 opcode; /* command word */
+ u16 st_length; /* status length */
+ u16 st_type; /* status type (RMH_SSIZE_XXX) */
+};
+
+/* RMH status type */
+enum {
+ RMH_SSIZE_FIXED = 0, /* status size fix (st_length = 0..x) */
+ RMH_SSIZE_ARG = 1, /* status size given in the LSB byte (used with st_length = 1) */
+ RMH_SSIZE_MASK = 2, /* status size given in bitmask (used with st_length = 1) */
+};
+
+/*
+ * Array of DSP commands
+ */
+static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = {
+[CMD_VERSION] = { 0x010000, 1, RMH_SSIZE_FIXED },
+[CMD_SUPPORTED] = { 0x020000, 4, RMH_SSIZE_FIXED },
+[CMD_TEST_IT] = { 0x040000, 1, RMH_SSIZE_FIXED },
+[CMD_SEND_IRQA] = { 0x070001, 0, RMH_SSIZE_FIXED },
+[CMD_ACCESS_IO_WRITE] = { 0x090000, 1, RMH_SSIZE_ARG },
+[CMD_ACCESS_IO_READ] = { 0x094000, 1, RMH_SSIZE_ARG },
+[CMD_ASYNC] = { 0x0a0000, 1, RMH_SSIZE_ARG },
+[CMD_MODIFY_CLOCK] = { 0x0d0000, 0, RMH_SSIZE_FIXED },
+[CMD_RESYNC_AUDIO_INPUTS] = { 0x0e0000, 0, RMH_SSIZE_FIXED },
+[CMD_GET_DSP_RESOURCES] = { 0x100000, 4, RMH_SSIZE_FIXED },
+[CMD_SET_TIMER_INTERRUPT] = { 0x110000, 0, RMH_SSIZE_FIXED },
+[CMD_RES_PIPE] = { 0x400000, 0, RMH_SSIZE_FIXED },
+[CMD_FREE_PIPE] = { 0x410000, 0, RMH_SSIZE_FIXED },
+[CMD_CONF_PIPE] = { 0x422101, 0, RMH_SSIZE_FIXED },
+[CMD_STOP_PIPE] = { 0x470004, 0, RMH_SSIZE_FIXED },
+[CMD_PIPE_SAMPLE_COUNT] = { 0x49a000, 2, RMH_SSIZE_FIXED },
+[CMD_CAN_START_PIPE] = { 0x4b0000, 1, RMH_SSIZE_FIXED },
+[CMD_START_STREAM] = { 0x802000, 0, RMH_SSIZE_FIXED },
+[CMD_STREAM_OUT_LEVEL_ADJUST] = { 0x822000, 0, RMH_SSIZE_FIXED },
+[CMD_STOP_STREAM] = { 0x832000, 0, RMH_SSIZE_FIXED },
+[CMD_UPDATE_R_BUFFERS] = { 0x840000, 0, RMH_SSIZE_FIXED },
+[CMD_FORMAT_STREAM_OUT] = { 0x860000, 0, RMH_SSIZE_FIXED },
+[CMD_FORMAT_STREAM_IN] = { 0x870000, 0, RMH_SSIZE_FIXED },
+[CMD_STREAM_SAMPLE_COUNT] = { 0x902000, 2, RMH_SSIZE_FIXED }, /* stat_len = nb_streams * 2 */
+[CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED },
+};
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+static char* cmd_names[] = {
+[CMD_VERSION] = "CMD_VERSION",
+[CMD_SUPPORTED] = "CMD_SUPPORTED",
+[CMD_TEST_IT] = "CMD_TEST_IT",
+[CMD_SEND_IRQA] = "CMD_SEND_IRQA",
+[CMD_ACCESS_IO_WRITE] = "CMD_ACCESS_IO_WRITE",
+[CMD_ACCESS_IO_READ] = "CMD_ACCESS_IO_READ",
+[CMD_ASYNC] = "CMD_ASYNC",
+[CMD_MODIFY_CLOCK] = "CMD_MODIFY_CLOCK",
+[CMD_RESYNC_AUDIO_INPUTS] = "CMD_RESYNC_AUDIO_INPUTS",
+[CMD_GET_DSP_RESOURCES] = "CMD_GET_DSP_RESOURCES",
+[CMD_SET_TIMER_INTERRUPT] = "CMD_SET_TIMER_INTERRUPT",
+[CMD_RES_PIPE] = "CMD_RES_PIPE",
+[CMD_FREE_PIPE] = "CMD_FREE_PIPE",
+[CMD_CONF_PIPE] = "CMD_CONF_PIPE",
+[CMD_STOP_PIPE] = "CMD_STOP_PIPE",
+[CMD_PIPE_SAMPLE_COUNT] = "CMD_PIPE_SAMPLE_COUNT",
+[CMD_CAN_START_PIPE] = "CMD_CAN_START_PIPE",
+[CMD_START_STREAM] = "CMD_START_STREAM",
+[CMD_STREAM_OUT_LEVEL_ADJUST] = "CMD_STREAM_OUT_LEVEL_ADJUST",
+[CMD_STOP_STREAM] = "CMD_STOP_STREAM",
+[CMD_UPDATE_R_BUFFERS] = "CMD_UPDATE_R_BUFFERS",
+[CMD_FORMAT_STREAM_OUT] = "CMD_FORMAT_STREAM_OUT",
+[CMD_FORMAT_STREAM_IN] = "CMD_FORMAT_STREAM_IN",
+[CMD_STREAM_SAMPLE_COUNT] = "CMD_STREAM_SAMPLE_COUNT",
+[CMD_AUDIO_LEVEL_ADJUST] = "CMD_AUDIO_LEVEL_ADJUST",
+};
+#endif
+
+
+static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
+{
+ int err;
+ int i;
+ u32 data;
+ u32 size_mask;
+ unsigned char reg;
+ int max_stat_len;
+
+ if (rmh->stat_len < PCXHR_SIZE_MAX_STATUS)
+ max_stat_len = PCXHR_SIZE_MAX_STATUS;
+ else max_stat_len = rmh->stat_len;
+
+ for (i = 0; i < rmh->stat_len; i++) {
+ /* wait for receiver full */
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF,
+ PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, &reg);
+ if (err) {
+ snd_printk(KERN_ERR "ERROR RMH stat: ISR:RXDF=1 (ISR = %x; i=%d )\n",
+ reg, i);
+ return err;
+ }
+ /* read data */
+ data = PCXHR_INPB(mgr, PCXHR_DSP_TXH) << 16;
+ data |= PCXHR_INPB(mgr, PCXHR_DSP_TXM) << 8;
+ data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL);
+
+ /* need to update rmh->stat_len on the fly ?? */
+ if (i==0) {
+ if (rmh->dsp_stat != RMH_SSIZE_FIXED) {
+ if (rmh->dsp_stat == RMH_SSIZE_ARG) {
+ rmh->stat_len = (u16)(data & 0x0000ff) + 1;
+ data &= 0xffff00;
+ } else {
+ /* rmh->dsp_stat == RMH_SSIZE_MASK */
+ rmh->stat_len = 1;
+ size_mask = data;
+ while (size_mask) {
+ if (size_mask & 1)
+ rmh->stat_len++;
+ size_mask >>= 1;
+ }
+ }
+ }
+ }
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ if (rmh->cmd_idx < CMD_LAST_INDEX)
+ snd_printdd(" stat[%d]=%x\n", i, data);
+#endif
+ if (i < max_stat_len)
+ rmh->stat[i] = data;
+ }
+ if (rmh->stat_len > max_stat_len) {
+ snd_printdd("PCXHR : rmh->stat_len=%x too big\n", rmh->stat_len);
+ rmh->stat_len = max_stat_len;
+ }
+ return 0;
+}
+
+static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
+{
+ int err;
+ int i;
+ u32 data;
+ unsigned char reg;
+
+ if (snd_BUG_ON(rmh->cmd_len >= PCXHR_SIZE_MAX_CMD))
+ return -EINVAL;
+ err = pcxhr_send_it_dsp(mgr, PCXHR_IT_MESSAGE, 1);
+ if (err) {
+ snd_printk(KERN_ERR "pcxhr_send_message : ED_DSP_CRASHED\n");
+ return err;
+ }
+ /* wait for chk bit */
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK,
+ PCXHR_ISR_HI08_CHK, PCXHR_TIMEOUT_DSP, &reg);
+ if (err)
+ return err;
+ /* reset irq chk */
+ err = pcxhr_send_it_dsp(mgr, PCXHR_IT_RESET_CHK, 1);
+ if (err)
+ return err;
+ /* wait for chk bit == 0*/
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK, 0,
+ PCXHR_TIMEOUT_DSP, &reg);
+ if (err)
+ return err;
+
+ data = rmh->cmd[0];
+
+ if (rmh->cmd_len > 1)
+ data |= 0x008000; /* MASK_MORE_THAN_1_WORD_COMMAND */
+ else
+ data &= 0xff7fff; /* MASK_1_WORD_COMMAND */
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ if (rmh->cmd_idx < CMD_LAST_INDEX)
+ snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]);
+#endif
+
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY,
+ PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, &reg);
+ if (err)
+ return err;
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXH, (data>>16)&0xFF);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXM, (data>>8)&0xFF);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXL, (data&0xFF));
+
+ if (rmh->cmd_len > 1) {
+ /* send length */
+ data = rmh->cmd_len - 1;
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY,
+ PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, &reg);
+ if (err)
+ return err;
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXH, (data>>16)&0xFF);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXM, (data>>8)&0xFF);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXL, (data&0xFF));
+
+ for (i=1; i < rmh->cmd_len; i++) {
+ /* send other words */
+ data = rmh->cmd[i];
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ if (rmh->cmd_idx < CMD_LAST_INDEX)
+ snd_printdd(" cmd[%d]=%x\n", i, data);
+#endif
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR,
+ PCXHR_ISR_HI08_TRDY,
+ PCXHR_ISR_HI08_TRDY,
+ PCXHR_TIMEOUT_DSP, &reg);
+ if (err)
+ return err;
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXH, (data>>16)&0xFF);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXM, (data>>8)&0xFF);
+ PCXHR_OUTPB(mgr, PCXHR_DSP_TXL, (data&0xFF));
+ }
+ }
+ /* wait for chk bit */
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK,
+ PCXHR_ISR_HI08_CHK, PCXHR_TIMEOUT_DSP, &reg);
+ if (err)
+ return err;
+ /* test status ISR */
+ if (reg & PCXHR_ISR_HI08_ERR) {
+ /* ERROR, wait for receiver full */
+ err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF,
+ PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, &reg);
+ if (err) {
+ snd_printk(KERN_ERR "ERROR RMH: ISR:RXDF=1 (ISR = %x)\n", reg);
+ return err;
+ }
+ /* read error code */
+ data = PCXHR_INPB(mgr, PCXHR_DSP_TXH) << 16;
+ data |= PCXHR_INPB(mgr, PCXHR_DSP_TXM) << 8;
+ data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL);
+ snd_printk(KERN_ERR "ERROR RMH(%d): 0x%x\n", rmh->cmd_idx, data);
+ err = -EINVAL;
+ } else {
+ /* read the response data */
+ err = pcxhr_read_rmh_status(mgr, rmh);
+ }
+ /* reset semaphore */
+ if (pcxhr_send_it_dsp(mgr, PCXHR_IT_RESET_SEMAPHORE, 1) < 0)
+ return -EIO;
+ return err;
+}
+
+
+/**
+ * pcxhr_init_rmh - initialize the RMH instance
+ * @rmh: the rmh pointer to be initialized
+ * @cmd: the rmh command to be set
+ */
+void pcxhr_init_rmh(struct pcxhr_rmh *rmh, int cmd)
+{
+ if (snd_BUG_ON(cmd >= CMD_LAST_INDEX))
+ return;
+ rmh->cmd[0] = pcxhr_dsp_cmds[cmd].opcode;
+ rmh->cmd_len = 1;
+ rmh->stat_len = pcxhr_dsp_cmds[cmd].st_length;
+ rmh->dsp_stat = pcxhr_dsp_cmds[cmd].st_type;
+ rmh->cmd_idx = cmd;
+}
+
+
+void pcxhr_set_pipe_cmd_params(struct pcxhr_rmh *rmh, int capture,
+ unsigned int param1, unsigned int param2,
+ unsigned int param3)
+{
+ snd_BUG_ON(param1 > MASK_FIRST_FIELD);
+ if (capture)
+ rmh->cmd[0] |= 0x800; /* COMMAND_RECORD_MASK */
+ if (param1)
+ rmh->cmd[0] |= (param1 << FIELD_SIZE);
+ if (param2) {
+ snd_BUG_ON(param2 > MASK_FIRST_FIELD);
+ rmh->cmd[0] |= param2;
+ }
+ if(param3) {
+ snd_BUG_ON(param3 > MASK_DSP_WORD);
+ rmh->cmd[1] = param3;
+ rmh->cmd_len = 2;
+ }
+}
+
+/*
+ * pcxhr_send_msg - send a DSP message with spinlock
+ * @rmh: the rmh record to send and receive
+ *
+ * returns 0 if successful, or a negative error code.
+ */
+int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh)
+{
+ unsigned long flags;
+ int err;
+ spin_lock_irqsave(&mgr->msg_lock, flags);
+ err = pcxhr_send_msg_nolock(mgr, rmh);
+ spin_unlock_irqrestore(&mgr->msg_lock, flags);
+ return err;
+}
+
+static inline int pcxhr_pipes_running(struct pcxhr_mgr *mgr)
+{
+ int start_mask = PCXHR_INPL(mgr, PCXHR_PLX_MBOX2);
+ /* least segnificant 12 bits are the pipe states for the playback audios */
+ /* next 12 bits are the pipe states for the capture audios
+ * (PCXHR_PIPE_STATE_CAPTURE_OFFSET)
+ */
+ start_mask &= 0xffffff;
+ snd_printdd("CMD_PIPE_STATE MBOX2=0x%06x\n", start_mask);
+ return start_mask;
+}
+
+#define PCXHR_PIPE_STATE_CAPTURE_OFFSET 12
+#define MAX_WAIT_FOR_DSP 20
+
+static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, int audio_mask, int *retry)
+{
+ struct pcxhr_rmh rmh;
+ int err;
+ int audio = 0;
+
+ *retry = 0;
+ while (audio_mask) {
+ if (audio_mask & 1) {
+ pcxhr_init_rmh(&rmh, CMD_CAN_START_PIPE);
+ if (audio < PCXHR_PIPE_STATE_CAPTURE_OFFSET) {
+ /* can start playback pipe */
+ pcxhr_set_pipe_cmd_params(&rmh, 0, audio, 0, 0);
+ } else {
+ /* can start capture pipe */
+ pcxhr_set_pipe_cmd_params(&rmh, 1, audio -
+ PCXHR_PIPE_STATE_CAPTURE_OFFSET,
+ 0, 0);
+ }
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err) {
+ snd_printk(KERN_ERR
+ "error pipe start (CMD_CAN_START_PIPE) err=%x!\n",
+ err);
+ return err;
+ }
+ /* if the pipe couldn't be prepaired for start, retry it later */
+ if (rmh.stat[0] == 0)
+ *retry |= (1<<audio);
+ }
+ audio_mask>>=1;
+ audio++;
+ }
+ return 0;
+}
+
+static int pcxhr_stop_pipes(struct pcxhr_mgr *mgr, int audio_mask)
+{
+ struct pcxhr_rmh rmh;
+ int err;
+ int audio = 0;
+
+ while (audio_mask) {
+ if (audio_mask & 1) {
+ pcxhr_init_rmh(&rmh, CMD_STOP_PIPE);
+ if (audio < PCXHR_PIPE_STATE_CAPTURE_OFFSET) {
+ /* stop playback pipe */
+ pcxhr_set_pipe_cmd_params(&rmh, 0, audio, 0, 0);
+ } else {
+ /* stop capture pipe */
+ pcxhr_set_pipe_cmd_params(&rmh, 1, audio -
+ PCXHR_PIPE_STATE_CAPTURE_OFFSET,
+ 0, 0);
+ }
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err) {
+ snd_printk(KERN_ERR
+ "error pipe stop (CMD_STOP_PIPE) err=%x!\n",
+ err);
+ return err;
+ }
+ }
+ audio_mask>>=1;
+ audio++;
+ }
+ return 0;
+}
+
+static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask)
+{
+ struct pcxhr_rmh rmh;
+ int err;
+ int audio = 0;
+
+ while (audio_mask) {
+ if (audio_mask & 1) {
+ pcxhr_init_rmh(&rmh, CMD_CONF_PIPE);
+ if (audio < PCXHR_PIPE_STATE_CAPTURE_OFFSET)
+ pcxhr_set_pipe_cmd_params(&rmh, 0, 0, 0, 1 << audio);
+ else
+ pcxhr_set_pipe_cmd_params(&rmh, 1, 0, 0,
+ 1 << (audio - PCXHR_PIPE_STATE_CAPTURE_OFFSET));
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err) {
+ snd_printk(KERN_ERR
+ "error pipe start (CMD_CONF_PIPE) err=%x!\n",
+ err);
+ return err;
+ }
+ }
+ audio_mask>>=1;
+ audio++;
+ }
+ /* now fire the interrupt on the card */
+ pcxhr_init_rmh(&rmh, CMD_SEND_IRQA);
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err) {
+ snd_printk(KERN_ERR "error pipe start (CMD_SEND_IRQA) err=%x!\n", err );
+ return err;
+ }
+ return 0;
+}
+
+
+
+int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_mask, int start)
+{
+ int state, i, err;
+ int audio_mask;
+
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ struct timeval my_tv1, my_tv2;
+ do_gettimeofday(&my_tv1);
+#endif
+ audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET));
+ /* current pipe state (playback + record) */
+ state = pcxhr_pipes_running(mgr);
+ snd_printdd("pcxhr_set_pipe_state %s (mask %x current %x)\n",
+ start ? "START" : "STOP", audio_mask, state);
+ if (start) {
+ audio_mask &= ~state; /* start only pipes that are not yet started */
+ state = audio_mask;
+ for (i = 0; i < MAX_WAIT_FOR_DSP; i++) {
+ err = pcxhr_prepair_pipe_start(mgr, state, &state);
+ if (err)
+ return err;
+ if (state == 0)
+ break; /* success, all pipes prepaired for start */
+ mdelay(1); /* otherwise wait 1 millisecond and retry */
+ }
+ } else {
+ audio_mask &= state; /* stop only pipes that are started */
+ }
+ if (audio_mask == 0)
+ return 0;
+
+ err = pcxhr_toggle_pipes(mgr, audio_mask);
+ if (err)
+ return err;
+
+ i = 0;
+ while (1) {
+ state = pcxhr_pipes_running(mgr);
+ /* have all pipes the new state ? */
+ if ((state & audio_mask) == (start ? audio_mask : 0))
+ break;
+ if (++i >= MAX_WAIT_FOR_DSP * 100) {
+ snd_printk(KERN_ERR "error pipe start/stop (ED_NO_RESPONSE_AT_IRQA)\n");
+ return -EBUSY;
+ }
+ udelay(10); /* wait 10 microseconds */
+ }
+ if (!start) {
+ err = pcxhr_stop_pipes(mgr, audio_mask);
+ if (err)
+ return err;
+ }
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ do_gettimeofday(&my_tv2);
+ snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n",
+ (long)(my_tv2.tv_usec - my_tv1.tv_usec), err);
+#endif
+ return 0;
+}
+
+int pcxhr_write_io_num_reg_cont(struct pcxhr_mgr *mgr, unsigned int mask,
+ unsigned int value, int *changed)
+{
+ struct pcxhr_rmh rmh;
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&mgr->msg_lock, flags);
+ if ((mgr->io_num_reg_cont & mask) == value) {
+ snd_printdd("IO_NUM_REG_CONT mask %x already is set to %x\n", mask, value);
+ if (changed)
+ *changed = 0;
+ spin_unlock_irqrestore(&mgr->msg_lock, flags);
+ return 0; /* already programmed */
+ }
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
+ rmh.cmd[0] |= IO_NUM_REG_CONT;
+ rmh.cmd[1] = mask;
+ rmh.cmd[2] = value;
+ rmh.cmd_len = 3;
+ err = pcxhr_send_msg_nolock(mgr, &rmh);
+ if (err == 0) {
+ mgr->io_num_reg_cont &= ~mask;
+ mgr->io_num_reg_cont |= value;
+ if (changed)
+ *changed = 1;
+ }
+ spin_unlock_irqrestore(&mgr->msg_lock, flags);
+ return err;
+}
+
+#define PCXHR_IRQ_TIMER 0x000300
+#define PCXHR_IRQ_FREQ_CHANGE 0x000800
+#define PCXHR_IRQ_TIME_CODE 0x001000
+#define PCXHR_IRQ_NOTIFY 0x002000
+#define PCXHR_IRQ_ASYNC 0x008000
+#define PCXHR_IRQ_MASK 0x00bb00
+#define PCXHR_FATAL_DSP_ERR 0xff0000
+
+enum pcxhr_async_err_src {
+ PCXHR_ERR_PIPE,
+ PCXHR_ERR_STREAM,
+ PCXHR_ERR_AUDIO
+};
+
+static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err,
+ enum pcxhr_async_err_src err_src, int pipe,
+ int is_capture)
+{
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ static char* err_src_name[] = {
+ [PCXHR_ERR_PIPE] = "Pipe",
+ [PCXHR_ERR_STREAM] = "Stream",
+ [PCXHR_ERR_AUDIO] = "Audio"
+ };
+#endif
+ if (err & 0xfff)
+ err &= 0xfff;
+ else
+ err = ((err >> 12) & 0xfff);
+ if (!err)
+ return 0;
+ snd_printdd("CMD_ASYNC : Error %s %s Pipe %d err=%x\n", err_src_name[err_src],
+ is_capture ? "Record" : "Play", pipe, err);
+ if (err == 0xe01)
+ mgr->async_err_stream_xrun++;
+ else if (err == 0xe10)
+ mgr->async_err_pipe_xrun++;
+ else
+ mgr->async_err_other_last = (int)err;
+ return 1;
+}
+
+
+void pcxhr_msg_tasklet(unsigned long arg)
+{
+ struct pcxhr_mgr *mgr = (struct pcxhr_mgr *)(arg);
+ struct pcxhr_rmh *prmh = mgr->prmh;
+ int err;
+ int i, j;
+
+ if (mgr->src_it_dsp & PCXHR_IRQ_FREQ_CHANGE)
+ snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occured\n");
+ if (mgr->src_it_dsp & PCXHR_IRQ_TIME_CODE)
+ snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n");
+ if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY)
+ snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n");
+ if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) {
+ snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n");
+
+ pcxhr_init_rmh(prmh, CMD_ASYNC);
+ prmh->cmd[0] |= 1; /* add SEL_ASYNC_EVENTS */
+ /* this is the only one extra long response command */
+ prmh->stat_len = PCXHR_SIZE_MAX_LONG_STATUS;
+ err = pcxhr_send_msg(mgr, prmh);
+ if (err)
+ snd_printk(KERN_ERR "ERROR pcxhr_msg_tasklet=%x;\n", err);
+ i = 1;
+ while (i < prmh->stat_len) {
+ int nb_audio = (prmh->stat[i] >> FIELD_SIZE) & MASK_FIRST_FIELD;
+ int nb_stream = (prmh->stat[i] >> (2*FIELD_SIZE)) & MASK_FIRST_FIELD;
+ int pipe = prmh->stat[i] & MASK_FIRST_FIELD;
+ int is_capture = prmh->stat[i] & 0x400000;
+ u32 err2;
+
+ if (prmh->stat[i] & 0x800000) { /* if BIT_END */
+ snd_printdd("TASKLET : End%sPipe %d\n",
+ is_capture ? "Record" : "Play", pipe);
+ }
+ i++;
+ err2 = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1];
+ if (err2)
+ pcxhr_handle_async_err(mgr, err2,
+ PCXHR_ERR_PIPE,
+ pipe, is_capture);
+ i += 2;
+ for (j = 0; j < nb_stream; j++) {
+ err2 = prmh->stat[i] ?
+ prmh->stat[i] : prmh->stat[i+1];
+ if (err2)
+ pcxhr_handle_async_err(mgr, err2,
+ PCXHR_ERR_STREAM,
+ pipe,
+ is_capture);
+ i += 2;
+ }
+ for (j = 0; j < nb_audio; j++) {
+ err2 = prmh->stat[i] ?
+ prmh->stat[i] : prmh->stat[i+1];
+ if (err2)
+ pcxhr_handle_async_err(mgr, err2,
+ PCXHR_ERR_AUDIO,
+ pipe,
+ is_capture);
+ i += 2;
+ }
+ }
+ }
+}
+
+static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr,
+ struct pcxhr_stream *stream)
+{
+ u_int64_t hw_sample_count;
+ struct pcxhr_rmh rmh;
+ int err, stream_mask;
+
+ stream_mask = stream->pipe->is_capture ? 1 : 1<<stream->substream->number;
+
+ /* get sample count for one stream */
+ pcxhr_init_rmh(&rmh, CMD_STREAM_SAMPLE_COUNT);
+ pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture,
+ stream->pipe->first_audio, 0, stream_mask);
+ /* rmh.stat_len = 2; */ /* 2 resp data for each stream of the pipe */
+
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return 0;
+
+ hw_sample_count = ((u_int64_t)rmh.stat[0]) << 24;
+ hw_sample_count += (u_int64_t)rmh.stat[1];
+
+ snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n",
+ stream->pipe->is_capture ? 'C':'P', stream->substream->number,
+ (long unsigned int)hw_sample_count,
+ (long unsigned int)(stream->timer_abs_periods +
+ stream->timer_period_frag + PCXHR_GRANULARITY));
+
+ return hw_sample_count;
+}
+
+static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr,
+ struct pcxhr_stream *stream, int samples_to_add)
+{
+ if (stream->substream && (stream->status == PCXHR_STREAM_STATUS_RUNNING)) {
+ u_int64_t new_sample_count;
+ int elapsed = 0;
+ int hardware_read = 0;
+ struct snd_pcm_runtime *runtime = stream->substream->runtime;
+
+ if (samples_to_add < 0) {
+ stream->timer_is_synced = 0;
+ /* add default if no hardware_read possible */
+ samples_to_add = PCXHR_GRANULARITY;
+ }
+
+ if (!stream->timer_is_synced) {
+ if (stream->timer_abs_periods != 0 ||
+ stream->timer_period_frag + PCXHR_GRANULARITY >=
+ runtime->period_size) {
+ new_sample_count = pcxhr_stream_read_position(mgr, stream);
+ hardware_read = 1;
+ if (new_sample_count >= PCXHR_GRANULARITY_MIN) {
+ /* sub security offset because of jitter and
+ * finer granularity of dsp time (MBOX4)
+ */
+ new_sample_count -= PCXHR_GRANULARITY_MIN;
+ stream->timer_is_synced = 1;
+ }
+ }
+ }
+ if (!hardware_read) {
+ /* if we didn't try to sync the position, increment it
+ * by PCXHR_GRANULARITY every timer interrupt
+ */
+ new_sample_count = stream->timer_abs_periods +
+ stream->timer_period_frag + samples_to_add;
+ }
+ while (1) {
+ u_int64_t new_elapse_pos = stream->timer_abs_periods +
+ runtime->period_size;
+ if (new_elapse_pos > new_sample_count)
+ break;
+ elapsed = 1;
+ stream->timer_buf_periods++;
+ if (stream->timer_buf_periods >= runtime->periods)
+ stream->timer_buf_periods = 0;
+ stream->timer_abs_periods = new_elapse_pos;
+ }
+ if (new_sample_count >= stream->timer_abs_periods)
+ stream->timer_period_frag = (u_int32_t)(new_sample_count -
+ stream->timer_abs_periods);
+ else
+ snd_printk(KERN_ERR "ERROR new_sample_count too small ??? %lx\n",
+ (long unsigned int)new_sample_count);
+
+ if (elapsed) {
+ spin_unlock(&mgr->lock);
+ snd_pcm_period_elapsed(stream->substream);
+ spin_lock(&mgr->lock);
+ }
+ }
+}
+
+
+irqreturn_t pcxhr_interrupt(int irq, void *dev_id)
+{
+ struct pcxhr_mgr *mgr = dev_id;
+ unsigned int reg;
+ int i, j;
+ struct snd_pcxhr *chip;
+
+ spin_lock(&mgr->lock);
+
+ reg = PCXHR_INPL(mgr, PCXHR_PLX_IRQCS);
+ if (! (reg & PCXHR_IRQCS_ACTIVE_PCIDB)) {
+ spin_unlock(&mgr->lock);
+ return IRQ_NONE; /* this device did not cause the interrupt */
+ }
+
+ /* clear interrupt */
+ reg = PCXHR_INPL(mgr, PCXHR_PLX_L2PCIDB);
+ PCXHR_OUTPL(mgr, PCXHR_PLX_L2PCIDB, reg);
+
+ /* timer irq occured */
+ if (reg & PCXHR_IRQ_TIMER) {
+ int timer_toggle = reg & PCXHR_IRQ_TIMER;
+ /* is a 24 bit counter */
+ int dsp_time_new = PCXHR_INPL(mgr, PCXHR_PLX_MBOX4) & PCXHR_DSP_TIME_MASK;
+ int dsp_time_diff = dsp_time_new - mgr->dsp_time_last;
+
+ if (dsp_time_diff < 0 && mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID) {
+ snd_printdd("ERROR DSP TIME old(%d) new(%d) -> "
+ "resynchronize all streams\n",
+ mgr->dsp_time_last, dsp_time_new);
+ mgr->dsp_time_err++;
+ }
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ if (dsp_time_diff == 0)
+ snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new);
+ else if (dsp_time_diff >= (2*PCXHR_GRANULARITY))
+ snd_printdd("ERROR DSP TIME TOO BIG old(%d) add(%d)\n",
+ mgr->dsp_time_last, dsp_time_new - mgr->dsp_time_last);
+#endif
+ mgr->dsp_time_last = dsp_time_new;
+
+ if (timer_toggle == mgr->timer_toggle)
+ snd_printdd("ERROR TIMER TOGGLE\n");
+ mgr->timer_toggle = timer_toggle;
+
+ reg &= ~PCXHR_IRQ_TIMER;
+ for (i = 0; i < mgr->num_cards; i++) {
+ chip = mgr->chip[i];
+ for (j = 0; j < chip->nb_streams_capt; j++)
+ pcxhr_update_timer_pos(mgr, &chip->capture_stream[j],
+ dsp_time_diff);
+ }
+ for (i = 0; i < mgr->num_cards; i++) {
+ chip = mgr->chip[i];
+ for (j = 0; j < chip->nb_streams_play; j++)
+ pcxhr_update_timer_pos(mgr, &chip->playback_stream[j],
+ dsp_time_diff);
+ }
+ }
+ /* other irq's handled in the tasklet */
+ if (reg & PCXHR_IRQ_MASK) {
+
+ /* as we didn't request any notifications, some kind of xrun error
+ * will probably occured
+ */
+ /* better resynchronize all streams next interrupt : */
+ mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID;
+
+ mgr->src_it_dsp = reg;
+ tasklet_hi_schedule(&mgr->msg_taskq);
+ }
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ if (reg & PCXHR_FATAL_DSP_ERR)
+ snd_printdd("FATAL DSP ERROR : %x\n", reg);
+#endif
+ spin_unlock(&mgr->lock);
+ return IRQ_HANDLED; /* this device caused the interrupt */
+}
diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h
new file mode 100644
index 0000000..d9a4ab6
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -0,0 +1,200 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * low level interface with interrupt ans message handling
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_PCXHR_CORE_H
+#define __SOUND_PCXHR_CORE_H
+
+struct firmware;
+struct pcxhr_mgr;
+
+/* init and firmware download commands */
+void pcxhr_reset_xilinx_com(struct pcxhr_mgr *mgr);
+void pcxhr_reset_dsp(struct pcxhr_mgr *mgr);
+void pcxhr_enable_dsp(struct pcxhr_mgr *mgr);
+int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilinx, int second);
+int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eeprom);
+int pcxhr_load_boot_binary(struct pcxhr_mgr *mgr, const struct firmware *boot);
+int pcxhr_load_dsp_binary(struct pcxhr_mgr *mgr, const struct firmware *dsp);
+
+/* DSP time available on MailBox4 register : 24 bit time samples() */
+#define PCXHR_DSP_TIME_MASK 0x00ffffff
+#define PCXHR_DSP_TIME_INVALID 0x10000000
+
+
+#define PCXHR_SIZE_MAX_CMD 8
+#define PCXHR_SIZE_MAX_STATUS 16
+#define PCXHR_SIZE_MAX_LONG_STATUS 256
+
+struct pcxhr_rmh {
+ u16 cmd_len; /* length of the command to send (WORDs) */
+ u16 stat_len; /* length of the status received (WORDs) */
+ u16 dsp_stat; /* status type, RMP_SSIZE_XXX */
+ u16 cmd_idx; /* index of the command */
+ u32 cmd[PCXHR_SIZE_MAX_CMD];
+ u32 stat[PCXHR_SIZE_MAX_STATUS];
+};
+
+enum {
+ CMD_VERSION, /* cmd_len = 2 stat_len = 1 */
+ CMD_SUPPORTED, /* cmd_len = 1 stat_len = 4 */
+ CMD_TEST_IT, /* cmd_len = 1 stat_len = 1 */
+ CMD_SEND_IRQA, /* cmd_len = 1 stat_len = 0 */
+ CMD_ACCESS_IO_WRITE, /* cmd_len >= 1 stat_len >= 1 */
+ CMD_ACCESS_IO_READ, /* cmd_len >= 1 stat_len >= 1 */
+ CMD_ASYNC, /* cmd_len = 1 stat_len = 1 */
+ CMD_MODIFY_CLOCK, /* cmd_len = 3 stat_len = 0 */
+ CMD_RESYNC_AUDIO_INPUTS, /* cmd_len = 1 stat_len = 0 */
+ CMD_GET_DSP_RESOURCES, /* cmd_len = 1 stat_len = 4 */
+ CMD_SET_TIMER_INTERRUPT, /* cmd_len = 1 stat_len = 0 */
+ CMD_RES_PIPE, /* cmd_len = 2 stat_len = 0 */
+ CMD_FREE_PIPE, /* cmd_len = 1 stat_len = 0 */
+ CMD_CONF_PIPE, /* cmd_len = 2 stat_len = 0 */
+ CMD_STOP_PIPE, /* cmd_len = 1 stat_len = 0 */
+ CMD_PIPE_SAMPLE_COUNT, /* cmd_len = 2 stat_len = 2 */
+ CMD_CAN_START_PIPE, /* cmd_len >= 1 stat_len = 1 */
+ CMD_START_STREAM, /* cmd_len = 2 stat_len = 0 */
+ CMD_STREAM_OUT_LEVEL_ADJUST, /* cmd_len >= 1 stat_len = 0 */
+ CMD_STOP_STREAM, /* cmd_len = 2 stat_len = 0 */
+ CMD_UPDATE_R_BUFFERS, /* cmd_len = 4 stat_len = 0 */
+ CMD_FORMAT_STREAM_OUT, /* cmd_len >= 2 stat_len = 0 */
+ CMD_FORMAT_STREAM_IN, /* cmd_len >= 4 stat_len = 0 */
+ CMD_STREAM_SAMPLE_COUNT, /* cmd_len = 2 stat_len = (2 * nb_stream) */
+ CMD_AUDIO_LEVEL_ADJUST, /* cmd_len = 3 stat_len = 0 */
+ CMD_LAST_INDEX
+};
+
+#define MASK_DSP_WORD 0x00ffffff
+#define MASK_ALL_STREAM 0x00ffffff
+#define MASK_DSP_WORD_LEVEL 0x000001ff
+#define MASK_FIRST_FIELD 0x0000001f
+#define FIELD_SIZE 5
+
+/*
+ init the rmh struct; by default cmd_len is set to 1
+ */
+void pcxhr_init_rmh(struct pcxhr_rmh *rmh, int cmd);
+
+void pcxhr_set_pipe_cmd_params(struct pcxhr_rmh* rmh, int capture, unsigned int param1,
+ unsigned int param2, unsigned int param3);
+
+/*
+ send the rmh
+ */
+int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh);
+
+
+/* values used for CMD_ACCESS_IO_WRITE and CMD_ACCESS_IO_READ */
+#define IO_NUM_REG_CONT 0
+#define IO_NUM_REG_GENCLK 1
+#define IO_NUM_REG_MUTE_OUT 2
+#define IO_NUM_SPEED_RATIO 4
+#define IO_NUM_REG_STATUS 5
+#define IO_NUM_REG_CUER 10
+#define IO_NUM_UER_CHIP_REG 11
+#define IO_NUM_REG_OUT_ANA_LEVEL 20
+#define IO_NUM_REG_IN_ANA_LEVEL 21
+
+
+#define REG_CONT_UNMUTE_INPUTS 0x020000
+
+/* parameters used with register IO_NUM_REG_STATUS */
+#define REG_STATUS_OPTIONS 0
+#define REG_STATUS_AES_SYNC 8
+#define REG_STATUS_AES_1 9
+#define REG_STATUS_AES_2 10
+#define REG_STATUS_AES_3 11
+#define REG_STATUS_AES_4 12
+#define REG_STATUS_WORD_CLOCK 13
+#define REG_STATUS_INTER_SYNC 14
+#define REG_STATUS_CURRENT 0x80
+/* results */
+#define REG_STATUS_OPT_NO_VIDEO_SIGNAL 0x01
+#define REG_STATUS_OPT_DAUGHTER_MASK 0x1c
+#define REG_STATUS_OPT_ANALOG_BOARD 0x00
+#define REG_STATUS_OPT_NO_DAUGHTER 0x1c
+#define REG_STATUS_OPT_COMPANION_MASK 0xe0
+#define REG_STATUS_OPT_NO_COMPANION 0xe0
+#define REG_STATUS_SYNC_32000 0x00
+#define REG_STATUS_SYNC_44100 0x01
+#define REG_STATUS_SYNC_48000 0x02
+#define REG_STATUS_SYNC_64000 0x03
+#define REG_STATUS_SYNC_88200 0x04
+#define REG_STATUS_SYNC_96000 0x05
+#define REG_STATUS_SYNC_128000 0x06
+#define REG_STATUS_SYNC_176400 0x07
+#define REG_STATUS_SYNC_192000 0x08
+
+int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_mask, int start);
+
+int pcxhr_write_io_num_reg_cont(struct pcxhr_mgr *mgr, unsigned int mask,
+ unsigned int value, int *changed);
+
+/* codec parameters */
+#define CS8416_RUN 0x200401
+#define CS8416_FORMAT_DETECT 0x200b00
+#define CS8416_CSB0 0x201900
+#define CS8416_CSB1 0x201a00
+#define CS8416_CSB2 0x201b00
+#define CS8416_CSB3 0x201c00
+#define CS8416_CSB4 0x201d00
+#define CS8416_VERSION 0x207f00
+
+#define CS8420_DATA_FLOW_CTL 0x200301
+#define CS8420_CLOCK_SRC_CTL 0x200401
+#define CS8420_RECEIVER_ERRORS 0x201000
+#define CS8420_SRC_RATIO 0x201e00
+#define CS8420_CSB0 0x202000
+#define CS8420_CSB1 0x202100
+#define CS8420_CSB2 0x202200
+#define CS8420_CSB3 0x202300
+#define CS8420_CSB4 0x202400
+#define CS8420_VERSION 0x207f00
+
+#define CS4271_MODE_CTL_1 0x200101
+#define CS4271_DAC_CTL 0x200201
+#define CS4271_VOLMIX 0x200301
+#define CS4271_VOLMUTE_LEFT 0x200401
+#define CS4271_VOLMUTE_RIGHT 0x200501
+#define CS4271_ADC_CTL 0x200601
+#define CS4271_MODE_CTL_2 0x200701
+
+#define CHIP_SIG_AND_MAP_SPI 0xff7f00
+
+/* codec selection */
+#define CS4271_01_CS 0x160018
+#define CS4271_23_CS 0x160019
+#define CS4271_45_CS 0x16001a
+#define CS4271_67_CS 0x16001b
+#define CS4271_89_CS 0x16001c
+#define CS4271_AB_CS 0x16001d
+#define CS8420_01_CS 0x080090
+#define CS8420_23_CS 0x080092
+#define CS8420_45_CS 0x080094
+#define CS8420_67_CS 0x080096
+#define CS8416_01_CS 0x080098
+
+
+/* interrupt handling */
+irqreturn_t pcxhr_interrupt(int irq, void *dev_id);
+void pcxhr_msg_tasklet(unsigned long arg);
+
+#endif /* __SOUND_PCXHR_CORE_H */
diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c
new file mode 100644
index 0000000..96640d9
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_hwdep.c
@@ -0,0 +1,446 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * hwdep device manager
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/interrupt.h>
+#include <linux/vmalloc.h>
+#include <linux/firmware.h>
+#include <linux/pci.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/hwdep.h>
+#include "pcxhr.h"
+#include "pcxhr_mixer.h"
+#include "pcxhr_hwdep.h"
+#include "pcxhr_core.h"
+
+
+#if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE)
+#if !defined(CONFIG_USE_PCXHRLOADER) && !defined(CONFIG_SND_PCXHR) /* built-in kernel */
+#define SND_PCXHR_FW_LOADER /* use the standard firmware loader */
+#endif
+#endif
+
+
+/*
+ * get basic information and init pcxhr card
+ */
+
+static int pcxhr_init_board(struct pcxhr_mgr *mgr)
+{
+ int err;
+ struct pcxhr_rmh rmh;
+ int card_streams;
+
+ /* calc the number of all streams used */
+ if (mgr->mono_capture)
+ card_streams = mgr->capture_chips * 2;
+ else
+ card_streams = mgr->capture_chips;
+ card_streams += mgr->playback_chips * PCXHR_PLAYBACK_STREAMS;
+
+ /* enable interrupts */
+ pcxhr_enable_dsp(mgr);
+
+ pcxhr_init_rmh(&rmh, CMD_SUPPORTED);
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+ /* test 8 or 12 phys out */
+ if ((rmh.stat[0] & MASK_FIRST_FIELD) != mgr->playback_chips * 2)
+ return -EINVAL;
+ /* test 8 or 2 phys in */
+ if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) !=
+ mgr->capture_chips * 2)
+ return -EINVAL;
+ /* test max nb substream per board */
+ if ((rmh.stat[1] & 0x5F) < card_streams)
+ return -EINVAL;
+ /* test max nb substream per pipe */
+ if (((rmh.stat[1] >> 7) & 0x5F) < PCXHR_PLAYBACK_STREAMS)
+ return -EINVAL;
+
+ pcxhr_init_rmh(&rmh, CMD_VERSION);
+ /* firmware num for DSP */
+ rmh.cmd[0] |= mgr->firmware_num;
+ /* transfer granularity in samples (should be multiple of 48) */
+ rmh.cmd[1] = (1<<23) + PCXHR_GRANULARITY;
+ rmh.cmd_len = 2;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+ snd_printdd("PCXHR DSP version is %d.%d.%d\n",
+ (rmh.stat[0]>>16)&0xff, (rmh.stat[0]>>8)&0xff, rmh.stat[0]&0xff);
+ mgr->dsp_version = rmh.stat[0];
+
+ /* get options */
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);
+ rmh.cmd[0] |= IO_NUM_REG_STATUS;
+ rmh.cmd[1] = REG_STATUS_OPTIONS;
+ rmh.cmd_len = 2;
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err)
+ return err;
+
+ if ((rmh.stat[1] & REG_STATUS_OPT_DAUGHTER_MASK) == REG_STATUS_OPT_ANALOG_BOARD)
+ mgr->board_has_analog = 1; /* analog addon board available */
+ else
+ /* analog addon board not available -> no support for instance */
+ return -EINVAL;
+
+ /* unmute inputs */
+ err = pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS,
+ REG_CONT_UNMUTE_INPUTS, NULL);
+ if (err)
+ return err;
+ /* unmute outputs */
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* a write to IO_NUM_REG_MUTE_OUT mutes! */
+ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
+ err = pcxhr_send_msg(mgr, &rmh);
+ return err;
+}
+
+void pcxhr_reset_board(struct pcxhr_mgr *mgr)
+{
+ struct pcxhr_rmh rmh;
+
+ if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) {
+ /* mute outputs */
+ /* a read to IO_NUM_REG_MUTE_OUT register unmutes! */
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
+ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT;
+ pcxhr_send_msg(mgr, &rmh);
+ /* mute inputs */
+ pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, 0, NULL);
+ }
+ /* reset pcxhr dsp */
+ if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_DSP_EPRM_INDEX))
+ pcxhr_reset_dsp(mgr);
+ /* reset second xilinx */
+ if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_XLX_COM_INDEX))
+ pcxhr_reset_xilinx_com(mgr);
+ return;
+}
+
+
+/*
+ * allocate a playback/capture pipe (pcmp0/pcmc0)
+ */
+static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe,
+ int is_capture, int pin)
+{
+ int stream_count, audio_count;
+ int err;
+ struct pcxhr_rmh rmh;
+
+ if (is_capture) {
+ stream_count = 1;
+ if (mgr->mono_capture)
+ audio_count = 1;
+ else
+ audio_count = 2;
+ } else {
+ stream_count = PCXHR_PLAYBACK_STREAMS;
+ audio_count = 2; /* always stereo */
+ }
+ snd_printdd("snd_add_ref_pipe pin(%d) pcm%c0\n", pin, is_capture ? 'c' : 'p');
+ pipe->is_capture = is_capture;
+ pipe->first_audio = pin;
+ /* define pipe (P_PCM_ONLY_MASK (0x020000) is not necessary) */
+ pcxhr_init_rmh(&rmh, CMD_RES_PIPE);
+ pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, audio_count, stream_count);
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err < 0) {
+ snd_printk(KERN_ERR "error pipe allocation (CMD_RES_PIPE) err=%x!\n", err );
+ return err;
+ }
+ pipe->status = PCXHR_PIPE_DEFINED;
+
+ return 0;
+}
+
+/*
+ * free playback/capture pipe (pcmp0/pcmc0)
+ */
+#if 0
+static int pcxhr_dsp_free_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe)
+{
+ struct pcxhr_rmh rmh;
+ int capture_mask = 0;
+ int playback_mask = 0;
+ int err = 0;
+
+ if (pipe->is_capture)
+ capture_mask = (1 << pipe->first_audio);
+ else
+ playback_mask = (1 << pipe->first_audio);
+
+ /* stop one pipe */
+ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0);
+ if (err < 0)
+ snd_printk(KERN_ERR "error stopping pipe!\n");
+ /* release the pipe */
+ pcxhr_init_rmh(&rmh, CMD_FREE_PIPE);
+ pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, 0, 0);
+ err = pcxhr_send_msg(mgr, &rmh);
+ if (err < 0)
+ snd_printk(KERN_ERR "error pipe release (CMD_FREE_PIPE) err(%x)\n", err);
+ pipe->status = PCXHR_PIPE_UNDEFINED;
+ return err;
+}
+#endif
+
+
+static int pcxhr_config_pipes(struct pcxhr_mgr *mgr)
+{
+ int err, i, j;
+ struct snd_pcxhr *chip;
+ struct pcxhr_pipe *pipe;
+
+ /* allocate the pipes on the dsp */
+ for (i = 0; i < mgr->num_cards; i++) {
+ chip = mgr->chip[i];
+ if (chip->nb_streams_play) {
+ pipe = &chip->playback_pipe;
+ err = pcxhr_dsp_allocate_pipe( mgr, pipe, 0, i*2);
+ if (err)
+ return err;
+ for(j = 0; j < chip->nb_streams_play; j++)
+ chip->playback_stream[j].pipe = pipe;
+ }
+ for (j = 0; j < chip->nb_streams_capt; j++) {
+ pipe = &chip->capture_pipe[j];
+ err = pcxhr_dsp_allocate_pipe(mgr, pipe, 1, i*2 + j);
+ if (err)
+ return err;
+ chip->capture_stream[j].pipe = pipe;
+ }
+ }
+ return 0;
+}
+
+static int pcxhr_start_pipes(struct pcxhr_mgr *mgr)
+{
+ int i, j;
+ struct snd_pcxhr *chip;
+ int playback_mask = 0;
+ int capture_mask = 0;
+
+ /* start all the pipes on the dsp */
+ for (i = 0; i < mgr->num_cards; i++) {
+ chip = mgr->chip[i];
+ if (chip->nb_streams_play)
+ playback_mask |= (1 << chip->playback_pipe.first_audio);
+ for (j = 0; j < chip->nb_streams_capt; j++)
+ capture_mask |= (1 << chip->capture_pipe[j].first_audio);
+ }
+ return pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1);
+}
+
+
+static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, const struct firmware *dsp)
+{
+ int err, card_index;
+
+ snd_printdd("loading dsp [%d] size = %Zd\n", index, dsp->size);
+
+ switch (index) {
+ case PCXHR_FIRMWARE_XLX_INT_INDEX:
+ pcxhr_reset_xilinx_com(mgr);
+ return pcxhr_load_xilinx_binary(mgr, dsp, 0);
+
+ case PCXHR_FIRMWARE_XLX_COM_INDEX:
+ pcxhr_reset_xilinx_com(mgr);
+ return pcxhr_load_xilinx_binary(mgr, dsp, 1);
+
+ case PCXHR_FIRMWARE_DSP_EPRM_INDEX:
+ pcxhr_reset_dsp(mgr);
+ return pcxhr_load_eeprom_binary(mgr, dsp);
+
+ case PCXHR_FIRMWARE_DSP_BOOT_INDEX:
+ return pcxhr_load_boot_binary(mgr, dsp);
+
+ case PCXHR_FIRMWARE_DSP_MAIN_INDEX:
+ err = pcxhr_load_dsp_binary(mgr, dsp);
+ if (err)
+ return err;
+ break; /* continue with first init */
+ default:
+ snd_printk(KERN_ERR "wrong file index\n");
+ return -EFAULT;
+ } /* end of switch file index*/
+
+ /* first communication with embedded */
+ err = pcxhr_init_board(mgr);
+ if (err < 0) {
+ snd_printk(KERN_ERR "pcxhr could not be set up\n");
+ return err;
+ }
+ err = pcxhr_config_pipes(mgr);
+ if (err < 0) {
+ snd_printk(KERN_ERR "pcxhr pipes could not be set up\n");
+ return err;
+ }
+ /* create devices and mixer in accordance with HW options*/
+ for (card_index = 0; card_index < mgr->num_cards; card_index++) {
+ struct snd_pcxhr *chip = mgr->chip[card_index];
+
+ if ((err = pcxhr_create_pcm(chip)) < 0)
+ return err;
+
+ if (card_index == 0) {
+ if ((err = pcxhr_create_mixer(chip->mgr)) < 0)
+ return err;
+ }
+ if ((err = snd_card_register(chip->card)) < 0)
+ return err;
+ }
+ err = pcxhr_start_pipes(mgr);
+ if (err < 0) {
+ snd_printk(KERN_ERR "pcxhr pipes could not be started\n");
+ return err;
+ }
+ snd_printdd("pcxhr firmware downloaded and successfully set up\n");
+
+ return 0;
+}
+
+/*
+ * fw loader entry
+ */
+#ifdef SND_PCXHR_FW_LOADER
+
+int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
+{
+ static char *fw_files[5] = {
+ "xi_1_882.dat",
+ "xc_1_882.dat",
+ "e321_512.e56",
+ "b321_512.b56",
+ "d321_512.d56"
+ };
+ char path[32];
+
+ const struct firmware *fw_entry;
+ int i, err;
+
+ for (i = 0; i < ARRAY_SIZE(fw_files); i++) {
+ sprintf(path, "pcxhr/%s", fw_files[i]);
+ if (request_firmware(&fw_entry, path, &mgr->pci->dev)) {
+ snd_printk(KERN_ERR "pcxhr: can't load firmware %s\n", path);
+ return -ENOENT;
+ }
+ /* fake hwdep dsp record */
+ err = pcxhr_dsp_load(mgr, i, fw_entry);
+ release_firmware(fw_entry);
+ if (err < 0)
+ return err;
+ mgr->dsp_loaded |= 1 << i;
+ }
+ return 0;
+}
+
+MODULE_FIRMWARE("pcxhr/xi_1_882.dat");
+MODULE_FIRMWARE("pcxhr/xc_1_882.dat");
+MODULE_FIRMWARE("pcxhr/e321_512.e56");
+MODULE_FIRMWARE("pcxhr/b321_512.b56");
+MODULE_FIRMWARE("pcxhr/d321_512.d56");
+
+#else /* old style firmware loading */
+
+/* pcxhr hwdep interface id string */
+#define PCXHR_HWDEP_ID "pcxhr loader"
+
+
+static int pcxhr_hwdep_dsp_status(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_status *info)
+{
+ strcpy(info->id, "pcxhr");
+ info->num_dsps = PCXHR_FIRMWARE_FILES_MAX_INDEX;
+
+ if (hw->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))
+ info->chip_ready = 1;
+
+ info->version = PCXHR_DRIVER_VERSION;
+ return 0;
+}
+
+static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_image *dsp)
+{
+ struct pcxhr_mgr *mgr = hw->private_data;
+ int err;
+ struct firmware fw;
+
+ fw.size = dsp->length;
+ fw.data = vmalloc(fw.size);
+ if (! fw.data) {
+ snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image (%lu bytes)\n",
+ (unsigned long)fw.size);
+ return -ENOMEM;
+ }
+ if (copy_from_user((void *)fw.data, dsp->image, dsp->length)) {
+ vfree(fw.data);
+ return -EFAULT;
+ }
+ err = pcxhr_dsp_load(mgr, dsp->index, &fw);
+ vfree(fw.data);
+ if (err < 0)
+ return err;
+ mgr->dsp_loaded |= 1 << dsp->index;
+ return 0;
+}
+
+static int pcxhr_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+static int pcxhr_hwdep_release(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
+{
+ int err;
+ struct snd_hwdep *hw;
+
+ /* only create hwdep interface for first cardX (see "index" module parameter)*/
+ if ((err = snd_hwdep_new(mgr->chip[0]->card, PCXHR_HWDEP_ID, 0, &hw)) < 0)
+ return err;
+
+ hw->iface = SNDRV_HWDEP_IFACE_PCXHR;
+ hw->private_data = mgr;
+ hw->ops.open = pcxhr_hwdep_open;
+ hw->ops.release = pcxhr_hwdep_release;
+ hw->ops.dsp_status = pcxhr_hwdep_dsp_status;
+ hw->ops.dsp_load = pcxhr_hwdep_dsp_load;
+ hw->exclusive = 1;
+ mgr->dsp_loaded = 0;
+ sprintf(hw->name, PCXHR_HWDEP_ID);
+
+ if ((err = snd_card_register(mgr->chip[0]->card)) < 0)
+ return err;
+ return 0;
+}
+
+#endif /* SND_PCXHR_FW_LOADER */
diff --git a/sound/pci/pcxhr/pcxhr_hwdep.h b/sound/pci/pcxhr/pcxhr_hwdep.h
new file mode 100644
index 0000000..f561909
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_hwdep.h
@@ -0,0 +1,40 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * definitions and makros for basic card access
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_PCXHR_HWDEP_H
+#define __SOUND_PCXHR_HWDEP_H
+
+
+/* firmware status codes */
+#define PCXHR_FIRMWARE_XLX_INT_INDEX 0
+#define PCXHR_FIRMWARE_XLX_COM_INDEX 1
+#define PCXHR_FIRMWARE_DSP_EPRM_INDEX 2
+#define PCXHR_FIRMWARE_DSP_BOOT_INDEX 3
+#define PCXHR_FIRMWARE_DSP_MAIN_INDEX 4
+#define PCXHR_FIRMWARE_FILES_MAX_INDEX 5
+
+
+/* exported */
+int pcxhr_setup_firmware(struct pcxhr_mgr *mgr);
+void pcxhr_reset_board(struct pcxhr_mgr *mgr);
+
+#endif /* __SOUND_PCXHR_HWDEP_H */
diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
new file mode 100644
index 0000000..aabc7bc
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -0,0 +1,1058 @@
+#define __NO_VERSION__
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * mixer callbacks
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/time.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include "pcxhr.h"
+#include "pcxhr_hwdep.h"
+#include "pcxhr_core.h"
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/asoundef.h>
+#include "pcxhr_mixer.h"
+
+
+#define PCXHR_ANALOG_CAPTURE_LEVEL_MIN 0 /* -96.0 dB */
+#define PCXHR_ANALOG_CAPTURE_LEVEL_MAX 255 /* +31.5 dB */
+#define PCXHR_ANALOG_CAPTURE_ZERO_LEVEL 224 /* +16.0 dB ( +31.5 dB - fix level +15.5 dB ) */
+
+#define PCXHR_ANALOG_PLAYBACK_LEVEL_MIN 0 /* -128.0 dB */
+#define PCXHR_ANALOG_PLAYBACK_LEVEL_MAX 128 /* 0.0 dB */
+#define PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL 104 /* -24.0 dB ( 0.0 dB - fix level +24.0 dB ) */
+
+static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 3150);
+static const DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -10400, 100, 2400);
+
+static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel)
+{
+ int err, vol;
+ struct pcxhr_rmh rmh;
+
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
+ if (is_capture) {
+ rmh.cmd[0] |= IO_NUM_REG_IN_ANA_LEVEL;
+ rmh.cmd[2] = chip->analog_capture_volume[channel];
+ } else {
+ rmh.cmd[0] |= IO_NUM_REG_OUT_ANA_LEVEL;
+ if (chip->analog_playback_active[channel])
+ vol = chip->analog_playback_volume[channel];
+ else
+ vol = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN;
+ rmh.cmd[2] = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX - vol; /* playback analog levels are inversed */
+ }
+ rmh.cmd[1] = 1 << ((2 * chip->chip_idx) + channel); /* audio mask */
+ rmh.cmd_len = 3;
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err < 0) {
+ snd_printk(KERN_DEBUG "error update_analog_audio_level card(%d) "
+ "is_capture(%d) err(%x)\n", chip->chip_idx, is_capture, err);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * analog level control
+ */
+static int pcxhr_analog_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ if (kcontrol->private_value == 0) { /* playback */
+ uinfo->value.integer.min = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN; /* -128 dB */
+ uinfo->value.integer.max = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX; /* 0 dB */
+ } else { /* capture */
+ uinfo->value.integer.min = PCXHR_ANALOG_CAPTURE_LEVEL_MIN; /* -96 dB */
+ uinfo->value.integer.max = PCXHR_ANALOG_CAPTURE_LEVEL_MAX; /* 31.5 dB */
+ }
+ return 0;
+}
+
+static int pcxhr_analog_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if (kcontrol->private_value == 0) { /* playback */
+ ucontrol->value.integer.value[0] = chip->analog_playback_volume[0];
+ ucontrol->value.integer.value[1] = chip->analog_playback_volume[1];
+ } else { /* capture */
+ ucontrol->value.integer.value[0] = chip->analog_capture_volume[0];
+ ucontrol->value.integer.value[1] = chip->analog_capture_volume[1];
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int is_capture, i;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ is_capture = (kcontrol->private_value != 0);
+ for (i = 0; i < 2; i++) {
+ int new_volume = ucontrol->value.integer.value[i];
+ int *stored_volume = is_capture ?
+ &chip->analog_capture_volume[i] :
+ &chip->analog_playback_volume[i];
+ if (is_capture) {
+ if (new_volume < PCXHR_ANALOG_CAPTURE_LEVEL_MIN ||
+ new_volume > PCXHR_ANALOG_CAPTURE_LEVEL_MAX)
+ continue;
+ } else {
+ if (new_volume < PCXHR_ANALOG_PLAYBACK_LEVEL_MIN ||
+ new_volume > PCXHR_ANALOG_PLAYBACK_LEVEL_MAX)
+ continue;
+ }
+ if (*stored_volume != new_volume) {
+ *stored_volume = new_volume;
+ changed = 1;
+ pcxhr_update_analog_audio_level(chip, is_capture, i);
+ }
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new pcxhr_control_analog_level = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ /* name will be filled later */
+ .info = pcxhr_analog_vol_info,
+ .get = pcxhr_analog_vol_get,
+ .put = pcxhr_analog_vol_put,
+ /* tlv will be filled later */
+};
+
+/* shared */
+#define pcxhr_sw_info snd_ctl_boolean_stereo_info
+
+static int pcxhr_audio_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->analog_playback_active[0];
+ ucontrol->value.integer.value[1] = chip->analog_playback_active[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int i, changed = 0;
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for(i = 0; i < 2; i++) {
+ if (chip->analog_playback_active[i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->analog_playback_active[i] =
+ !!ucontrol->value.integer.value[i];
+ changed = 1;
+ /* update playback levels */
+ pcxhr_update_analog_audio_level(chip, 0, i);
+ }
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new pcxhr_control_output_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = pcxhr_sw_info, /* shared */
+ .get = pcxhr_audio_sw_get,
+ .put = pcxhr_audio_sw_put
+};
+
+
+#define PCXHR_DIGITAL_LEVEL_MIN 0x000 /* -110 dB */
+#define PCXHR_DIGITAL_LEVEL_MAX 0x1ff /* +18 dB */
+#define PCXHR_DIGITAL_ZERO_LEVEL 0x1b7 /* 0 dB */
+
+static const DECLARE_TLV_DB_SCALE(db_scale_digital, -10975, 25, 1800);
+
+#define MORE_THAN_ONE_STREAM_LEVEL 0x000001
+#define VALID_STREAM_PAN_LEVEL_MASK 0x800000
+#define VALID_STREAM_LEVEL_MASK 0x400000
+#define VALID_STREAM_LEVEL_1_MASK 0x200000
+#define VALID_STREAM_LEVEL_2_MASK 0x100000
+
+static int pcxhr_update_playback_stream_level(struct snd_pcxhr* chip, int idx)
+{
+ int err;
+ struct pcxhr_rmh rmh;
+ struct pcxhr_pipe *pipe = &chip->playback_pipe;
+ int left, right;
+
+ if (chip->digital_playback_active[idx][0])
+ left = chip->digital_playback_volume[idx][0];
+ else
+ left = PCXHR_DIGITAL_LEVEL_MIN;
+ if (chip->digital_playback_active[idx][1])
+ right = chip->digital_playback_volume[idx][1];
+ else
+ right = PCXHR_DIGITAL_LEVEL_MIN;
+
+ pcxhr_init_rmh(&rmh, CMD_STREAM_OUT_LEVEL_ADJUST);
+ /* add pipe and stream mask */
+ pcxhr_set_pipe_cmd_params(&rmh, 0, pipe->first_audio, 0, 1<<idx);
+ /* volume left->left / right->right panoramic level */
+ rmh.cmd[0] |= MORE_THAN_ONE_STREAM_LEVEL;
+ rmh.cmd[2] = VALID_STREAM_PAN_LEVEL_MASK | VALID_STREAM_LEVEL_1_MASK;
+ rmh.cmd[2] |= (left << 10);
+ rmh.cmd[3] = VALID_STREAM_PAN_LEVEL_MASK | VALID_STREAM_LEVEL_2_MASK;
+ rmh.cmd[3] |= right;
+ rmh.cmd_len = 4;
+
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err < 0) {
+ snd_printk(KERN_DEBUG "error update_playback_stream_level "
+ "card(%d) err(%x)\n", chip->chip_idx, err);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+#define AUDIO_IO_HAS_MUTE_LEVEL 0x400000
+#define AUDIO_IO_HAS_MUTE_MONITOR_1 0x200000
+#define VALID_AUDIO_IO_DIGITAL_LEVEL 0x000001
+#define VALID_AUDIO_IO_MONITOR_LEVEL 0x000002
+#define VALID_AUDIO_IO_MUTE_LEVEL 0x000004
+#define VALID_AUDIO_IO_MUTE_MONITOR_1 0x000008
+
+static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, int channel)
+{
+ int err;
+ struct pcxhr_rmh rmh;
+ struct pcxhr_pipe *pipe;
+
+ if (capture)
+ pipe = &chip->capture_pipe[0];
+ else
+ pipe = &chip->playback_pipe;
+
+ pcxhr_init_rmh(&rmh, CMD_AUDIO_LEVEL_ADJUST);
+ /* add channel mask */
+ pcxhr_set_pipe_cmd_params(&rmh, capture, 0, 0, 1 << (channel + pipe->first_audio));
+ /* TODO : if mask (3 << pipe->first_audio) is used, left and right channel
+ * will be programmed to the same params
+ */
+ if (capture) {
+ rmh.cmd[0] |= VALID_AUDIO_IO_DIGITAL_LEVEL;
+ /* VALID_AUDIO_IO_MUTE_LEVEL not yet handled (capture pipe level) */
+ rmh.cmd[2] = chip->digital_capture_volume[channel];
+ } else {
+ rmh.cmd[0] |= VALID_AUDIO_IO_MONITOR_LEVEL | VALID_AUDIO_IO_MUTE_MONITOR_1;
+ /* VALID_AUDIO_IO_DIGITAL_LEVEL and VALID_AUDIO_IO_MUTE_LEVEL not yet
+ * handled (playback pipe level)
+ */
+ rmh.cmd[2] = chip->monitoring_volume[channel] << 10;
+ if (chip->monitoring_active[channel] == 0)
+ rmh.cmd[2] |= AUDIO_IO_HAS_MUTE_MONITOR_1;
+ }
+ rmh.cmd_len = 3;
+
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if(err<0) {
+ snd_printk(KERN_DEBUG "error update_audio_level card(%d) err(%x)\n",
+ chip->chip_idx, err);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/* shared */
+static int pcxhr_digital_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = PCXHR_DIGITAL_LEVEL_MIN; /* -109.5 dB */
+ uinfo->value.integer.max = PCXHR_DIGITAL_LEVEL_MAX; /* 18.0 dB */
+ return 0;
+}
+
+
+static int pcxhr_pcm_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+ int *stored_volume;
+ int is_capture = kcontrol->private_value;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if (is_capture)
+ stored_volume = chip->digital_capture_volume; /* digital capture */
+ else
+ stored_volume = chip->digital_playback_volume[idx]; /* digital playback */
+ ucontrol->value.integer.value[0] = stored_volume[0];
+ ucontrol->value.integer.value[1] = stored_volume[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+ int changed = 0;
+ int is_capture = kcontrol->private_value;
+ int *stored_volume;
+ int i;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if (is_capture) /* digital capture */
+ stored_volume = chip->digital_capture_volume;
+ else /* digital playback */
+ stored_volume = chip->digital_playback_volume[idx];
+ for (i = 0; i < 2; i++) {
+ int vol = ucontrol->value.integer.value[i];
+ if (vol < PCXHR_DIGITAL_LEVEL_MIN ||
+ vol > PCXHR_DIGITAL_LEVEL_MAX)
+ continue;
+ if (stored_volume[i] != vol) {
+ stored_volume[i] = vol;
+ changed = 1;
+ if (is_capture) /* update capture volume */
+ pcxhr_update_audio_pipe_level(chip, 1, i);
+ }
+ }
+ if (!is_capture && changed) /* update playback volume */
+ pcxhr_update_playback_stream_level(chip, idx);
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_pcxhr_pcm_vol =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ /* name will be filled later */
+ /* count will be filled later */
+ .info = pcxhr_digital_vol_info, /* shared */
+ .get = pcxhr_pcm_vol_get,
+ .put = pcxhr_pcm_vol_put,
+ .tlv = { .p = db_scale_digital },
+};
+
+
+static int pcxhr_pcm_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->digital_playback_active[idx][0];
+ ucontrol->value.integer.value[1] = chip->digital_playback_active[idx][1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */
+ int i, j;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ j = idx;
+ for (i = 0; i < 2; i++) {
+ if (chip->digital_playback_active[j][i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->digital_playback_active[j][i] =
+ !!ucontrol->value.integer.value[i];
+ changed = 1;
+ }
+ }
+ if (changed)
+ pcxhr_update_playback_stream_level(chip, idx);
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new pcxhr_control_pcm_switch = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .count = PCXHR_PLAYBACK_STREAMS,
+ .info = pcxhr_sw_info, /* shared */
+ .get = pcxhr_pcm_sw_get,
+ .put = pcxhr_pcm_sw_put
+};
+
+
+/*
+ * monitoring level control
+ */
+
+static int pcxhr_monitor_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->monitoring_volume[0];
+ ucontrol->value.integer.value[1] = chip->monitoring_volume[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int i;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for (i = 0; i < 2; i++) {
+ if (chip->monitoring_volume[i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->monitoring_volume[i] =
+ !!ucontrol->value.integer.value[i];
+ if(chip->monitoring_active[i])
+ /* update monitoring volume and mute */
+ /* do only when monitoring is unmuted */
+ pcxhr_update_audio_pipe_level(chip, 0, i);
+ changed = 1;
+ }
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new pcxhr_control_monitor_vol = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Monitoring Volume",
+ .info = pcxhr_digital_vol_info, /* shared */
+ .get = pcxhr_monitor_vol_get,
+ .put = pcxhr_monitor_vol_put,
+ .tlv = { .p = db_scale_digital },
+};
+
+/*
+ * monitoring switch control
+ */
+
+static int pcxhr_monitor_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ mutex_lock(&chip->mgr->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->monitoring_active[0];
+ ucontrol->value.integer.value[1] = chip->monitoring_active[1];
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int i;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for (i = 0; i < 2; i++) {
+ if (chip->monitoring_active[i] !=
+ ucontrol->value.integer.value[i]) {
+ chip->monitoring_active[i] =
+ !!ucontrol->value.integer.value[i];
+ changed |= (1<<i); /* mask 0x01 and 0x02 */
+ }
+ }
+ if (changed & 0x01)
+ /* update left monitoring volume and mute */
+ pcxhr_update_audio_pipe_level(chip, 0, 0);
+ if (changed & 0x02)
+ /* update right monitoring volume and mute */
+ pcxhr_update_audio_pipe_level(chip, 0, 1);
+
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return (changed != 0);
+}
+
+static struct snd_kcontrol_new pcxhr_control_monitor_sw = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitoring Switch",
+ .info = pcxhr_sw_info, /* shared */
+ .get = pcxhr_monitor_sw_get,
+ .put = pcxhr_monitor_sw_put
+};
+
+
+
+/*
+ * audio source select
+ */
+#define PCXHR_SOURCE_AUDIO01_UER 0x000100
+#define PCXHR_SOURCE_AUDIO01_SYNC 0x000200
+#define PCXHR_SOURCE_AUDIO23_UER 0x000400
+#define PCXHR_SOURCE_AUDIO45_UER 0x001000
+#define PCXHR_SOURCE_AUDIO67_UER 0x040000
+
+static int pcxhr_set_audio_source(struct snd_pcxhr* chip)
+{
+ struct pcxhr_rmh rmh;
+ unsigned int mask, reg;
+ unsigned int codec;
+ int err, use_src, changed;
+
+ switch (chip->chip_idx) {
+ case 0 : mask = PCXHR_SOURCE_AUDIO01_UER; codec = CS8420_01_CS; break;
+ case 1 : mask = PCXHR_SOURCE_AUDIO23_UER; codec = CS8420_23_CS; break;
+ case 2 : mask = PCXHR_SOURCE_AUDIO45_UER; codec = CS8420_45_CS; break;
+ case 3 : mask = PCXHR_SOURCE_AUDIO67_UER; codec = CS8420_67_CS; break;
+ default: return -EINVAL;
+ }
+ reg = 0; /* audio source from analog plug */
+ use_src = 0; /* do not activate codec SRC */
+
+ if (chip->audio_capture_source != 0) {
+ reg = mask; /* audio source from digital plug */
+ if (chip->audio_capture_source == 2)
+ use_src = 1;
+ }
+ /* set the input source */
+ pcxhr_write_io_num_reg_cont(chip->mgr, mask, reg, &changed);
+ /* resync them (otherwise channel inversion possible) */
+ if (changed) {
+ pcxhr_init_rmh(&rmh, CMD_RESYNC_AUDIO_INPUTS);
+ rmh.cmd[0] |= (1 << chip->chip_idx);
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err)
+ return err;
+ }
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set codec SRC on off */
+ rmh.cmd_len = 3;
+ rmh.cmd[0] |= IO_NUM_UER_CHIP_REG;
+ rmh.cmd[1] = codec;
+ rmh.cmd[2] = (CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x54);
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if(err)
+ return err;
+ rmh.cmd[2] = (CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x49);
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ return err;
+}
+
+static int pcxhr_audio_src_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {"Analog", "Digital", "Digi+SRC"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int pcxhr_audio_src_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = chip->audio_capture_source;
+ return 0;
+}
+
+static int pcxhr_audio_src_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int ret = 0;
+
+ if (ucontrol->value.enumerated.item[0] >= 3)
+ return -EINVAL;
+ mutex_lock(&chip->mgr->mixer_mutex);
+ if (chip->audio_capture_source != ucontrol->value.enumerated.item[0]) {
+ chip->audio_capture_source = ucontrol->value.enumerated.item[0];
+ pcxhr_set_audio_source(chip);
+ ret = 1;
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return ret;
+}
+
+static struct snd_kcontrol_new pcxhr_control_audio_src = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = pcxhr_audio_src_info,
+ .get = pcxhr_audio_src_get,
+ .put = pcxhr_audio_src_put,
+};
+
+
+/*
+ * clock type selection
+ * enum pcxhr_clock_type {
+ * PCXHR_CLOCK_TYPE_INTERNAL = 0,
+ * PCXHR_CLOCK_TYPE_WORD_CLOCK,
+ * PCXHR_CLOCK_TYPE_AES_SYNC,
+ * PCXHR_CLOCK_TYPE_AES_1,
+ * PCXHR_CLOCK_TYPE_AES_2,
+ * PCXHR_CLOCK_TYPE_AES_3,
+ * PCXHR_CLOCK_TYPE_AES_4,
+ * };
+ */
+
+static int pcxhr_clock_type_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[7] = {
+ "Internal", "WordClock", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4"
+ };
+ struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol);
+ int clock_items = 3 + mgr->capture_chips;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = clock_items;
+ if (uinfo->value.enumerated.item >= clock_items)
+ uinfo->value.enumerated.item = clock_items-1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int pcxhr_clock_type_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = mgr->use_clock_type;
+ return 0;
+}
+
+static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol);
+ unsigned int clock_items = 3 + mgr->capture_chips;
+ int rate, ret = 0;
+
+ if (ucontrol->value.enumerated.item[0] >= clock_items)
+ return -EINVAL;
+ mutex_lock(&mgr->mixer_mutex);
+ if (mgr->use_clock_type != ucontrol->value.enumerated.item[0]) {
+ mutex_lock(&mgr->setup_mutex);
+ mgr->use_clock_type = ucontrol->value.enumerated.item[0];
+ if (mgr->use_clock_type)
+ pcxhr_get_external_clock(mgr, mgr->use_clock_type, &rate);
+ else
+ rate = mgr->sample_rate;
+ if (rate) {
+ pcxhr_set_clock(mgr, rate);
+ if (mgr->sample_rate)
+ mgr->sample_rate = rate;
+ }
+ mutex_unlock(&mgr->setup_mutex);
+ ret = 1; /* return 1 even if the set was not done. ok ? */
+ }
+ mutex_unlock(&mgr->mixer_mutex);
+ return ret;
+}
+
+static struct snd_kcontrol_new pcxhr_control_clock_type = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Clock Mode",
+ .info = pcxhr_clock_type_info,
+ .get = pcxhr_clock_type_get,
+ .put = pcxhr_clock_type_put,
+};
+
+/*
+ * clock rate control
+ * specific control that scans the sample rates on the external plugs
+ */
+static int pcxhr_clock_rate_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3 + mgr->capture_chips;
+ uinfo->value.integer.min = 0; /* clock not present */
+ uinfo->value.integer.max = 192000; /* max sample rate 192 kHz */
+ return 0;
+}
+
+static int pcxhr_clock_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol);
+ int i, err, rate;
+
+ mutex_lock(&mgr->mixer_mutex);
+ for(i = 0; i < 3 + mgr->capture_chips; i++) {
+ if (i == PCXHR_CLOCK_TYPE_INTERNAL)
+ rate = mgr->sample_rate_real;
+ else {
+ err = pcxhr_get_external_clock(mgr, i, &rate);
+ if (err)
+ break;
+ }
+ ucontrol->value.integer.value[i] = rate;
+ }
+ mutex_unlock(&mgr->mixer_mutex);
+ return 0;
+}
+
+static struct snd_kcontrol_new pcxhr_control_clock_rate = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "Clock Rates",
+ .info = pcxhr_clock_rate_info,
+ .get = pcxhr_clock_rate_get,
+};
+
+/*
+ * IEC958 status bits
+ */
+static int pcxhr_iec958_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char* aes_bits)
+{
+ int i, err;
+ unsigned char temp;
+ struct pcxhr_rmh rmh;
+
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ);
+ rmh.cmd[0] |= IO_NUM_UER_CHIP_REG;
+ switch (chip->chip_idx) {
+ case 0: rmh.cmd[1] = CS8420_01_CS; break; /* use CS8416_01_CS for AES SYNC plug */
+ case 1: rmh.cmd[1] = CS8420_23_CS; break;
+ case 2: rmh.cmd[1] = CS8420_45_CS; break;
+ case 3: rmh.cmd[1] = CS8420_67_CS; break;
+ default: return -EINVAL;
+ }
+ switch (aes_idx) {
+ case 0: rmh.cmd[2] = CS8420_CSB0; break; /* use CS8416_CSBx for AES SYNC plug */
+ case 1: rmh.cmd[2] = CS8420_CSB1; break;
+ case 2: rmh.cmd[2] = CS8420_CSB2; break;
+ case 3: rmh.cmd[2] = CS8420_CSB3; break;
+ case 4: rmh.cmd[2] = CS8420_CSB4; break;
+ default: return -EINVAL;
+ }
+ rmh.cmd[1] &= 0x0fffff; /* size and code the chip id for the fpga */
+ rmh.cmd[2] &= CHIP_SIG_AND_MAP_SPI; /* chip signature + map for spi read */
+ rmh.cmd_len = 3;
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err)
+ return err;
+ temp = 0;
+ for (i = 0; i < 8; i++) {
+ /* attention : reversed bit order (not with CS8416_01_CS) */
+ temp <<= 1;
+ if (rmh.stat[1] & (1 << i))
+ temp |= 1;
+ }
+ snd_printdd("read iec958 AES %d byte %d = 0x%x\n", chip->chip_idx, aes_idx, temp);
+ *aes_bits = temp;
+ return 0;
+}
+
+static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ unsigned char aes_bits;
+ int i, err;
+
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for(i = 0; i < 5; i++) {
+ if (kcontrol->private_value == 0) /* playback */
+ aes_bits = chip->aes_bits[i];
+ else { /* capture */
+ err = pcxhr_iec958_capture_byte(chip, i, &aes_bits);
+ if (err)
+ break;
+ }
+ ucontrol->value.iec958.status[i] = aes_bits;
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return 0;
+}
+
+static int pcxhr_iec958_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int i;
+ for (i = 0; i < 5; i++)
+ ucontrol->value.iec958.status[i] = 0xff;
+ return 0;
+}
+
+static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char aes_bits)
+{
+ int i, err, cmd;
+ unsigned char new_bits = aes_bits;
+ unsigned char old_bits = chip->aes_bits[aes_idx];
+ struct pcxhr_rmh rmh;
+
+ for (i = 0; i < 8; i++) {
+ if ((old_bits & 0x01) != (new_bits & 0x01)) {
+ cmd = chip->chip_idx & 0x03; /* chip index 0..3 */
+ if(chip->chip_idx > 3)
+ /* new bit used if chip_idx>3 (PCX1222HR) */
+ cmd |= 1 << 22;
+ cmd |= ((aes_idx << 3) + i) << 2; /* add bit offset */
+ cmd |= (new_bits & 0x01) << 23; /* add bit value */
+ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE);
+ rmh.cmd[0] |= IO_NUM_REG_CUER;
+ rmh.cmd[1] = cmd;
+ rmh.cmd_len = 2;
+ snd_printdd("write iec958 AES %d byte %d bit %d (cmd %x)\n",
+ chip->chip_idx, aes_idx, i, cmd);
+ err = pcxhr_send_msg(chip->mgr, &rmh);
+ if (err)
+ return err;
+ }
+ old_bits >>= 1;
+ new_bits >>= 1;
+ }
+ chip->aes_bits[aes_idx] = aes_bits;
+ return 0;
+}
+
+static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol);
+ int i, changed = 0;
+
+ /* playback */
+ mutex_lock(&chip->mgr->mixer_mutex);
+ for (i = 0; i < 5; i++) {
+ if (ucontrol->value.iec958.status[i] != chip->aes_bits[i]) {
+ pcxhr_iec958_update_byte(chip, i, ucontrol->value.iec958.status[i]);
+ changed = 1;
+ }
+ }
+ mutex_unlock(&chip->mgr->mixer_mutex);
+ return changed;
+}
+
+static struct snd_kcontrol_new pcxhr_control_playback_iec958_mask = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .info = pcxhr_iec958_info,
+ .get = pcxhr_iec958_mask_get
+};
+static struct snd_kcontrol_new pcxhr_control_playback_iec958 = {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = pcxhr_iec958_info,
+ .get = pcxhr_iec958_get,
+ .put = pcxhr_iec958_put,
+ .private_value = 0 /* playback */
+};
+
+static struct snd_kcontrol_new pcxhr_control_capture_iec958_mask = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,MASK),
+ .info = pcxhr_iec958_info,
+ .get = pcxhr_iec958_mask_get
+};
+static struct snd_kcontrol_new pcxhr_control_capture_iec958 = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT),
+ .info = pcxhr_iec958_info,
+ .get = pcxhr_iec958_get,
+ .private_value = 1 /* capture */
+};
+
+static void pcxhr_init_audio_levels(struct snd_pcxhr *chip)
+{
+ int i;
+
+ for (i = 0; i < 2; i++) {
+ if (chip->nb_streams_play) {
+ int j;
+ /* at boot time the digital volumes are unmuted 0dB */
+ for (j = 0; j < PCXHR_PLAYBACK_STREAMS; j++) {
+ chip->digital_playback_active[j][i] = 1;
+ chip->digital_playback_volume[j][i] = PCXHR_DIGITAL_ZERO_LEVEL;
+ }
+ /* after boot, only two bits are set on the uer interface */
+ chip->aes_bits[0] = IEC958_AES0_PROFESSIONAL | IEC958_AES0_PRO_FS_48000;
+/* only for test purpose, remove later */
+#ifdef CONFIG_SND_DEBUG
+ /* analog volumes for playback (is LEVEL_MIN after boot) */
+ chip->analog_playback_active[i] = 1;
+ chip->analog_playback_volume[i] = PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL;
+ pcxhr_update_analog_audio_level(chip, 0, i);
+#endif
+/* test end */
+ }
+ if (chip->nb_streams_capt) {
+ /* at boot time the digital volumes are unmuted 0dB */
+ chip->digital_capture_volume[i] = PCXHR_DIGITAL_ZERO_LEVEL;
+/* only for test purpose, remove later */
+#ifdef CONFIG_SND_DEBUG
+ /* analog volumes for playback (is LEVEL_MIN after boot) */
+ chip->analog_capture_volume[i] = PCXHR_ANALOG_CAPTURE_ZERO_LEVEL;
+ pcxhr_update_analog_audio_level(chip, 1, i);
+#endif
+/* test end */
+ }
+ }
+
+ return;
+}
+
+
+int pcxhr_create_mixer(struct pcxhr_mgr *mgr)
+{
+ struct snd_pcxhr *chip;
+ int err, i;
+
+ mutex_init(&mgr->mixer_mutex); /* can be in another place */
+
+ for (i = 0; i < mgr->num_cards; i++) {
+ struct snd_kcontrol_new temp;
+ chip = mgr->chip[i];
+
+ if (chip->nb_streams_play) {
+ /* analog output level control */
+ temp = pcxhr_control_analog_level;
+ temp.name = "Master Playback Volume";
+ temp.private_value = 0; /* playback */
+ temp.tlv.p = db_scale_analog_playback;
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ /* output mute controls */
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_output_switch,
+ chip))) < 0)
+ return err;
+
+ temp = snd_pcxhr_pcm_vol;
+ temp.name = "PCM Playback Volume";
+ temp.count = PCXHR_PLAYBACK_STREAMS;
+ temp.private_value = 0; /* playback */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_pcm_switch,
+ chip))) < 0)
+ return err;
+
+ /* IEC958 controls */
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_playback_iec958_mask,
+ chip))) < 0)
+ return err;
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_playback_iec958,
+ chip))) < 0)
+ return err;
+ }
+ if (chip->nb_streams_capt) {
+ /* analog input level control only on first two chips !*/
+ temp = pcxhr_control_analog_level;
+ temp.name = "Master Capture Volume";
+ temp.private_value = 1; /* capture */
+ temp.tlv.p = db_scale_analog_capture;
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+
+ temp = snd_pcxhr_pcm_vol;
+ temp.name = "PCM Capture Volume";
+ temp.count = 1;
+ temp.private_value = 1; /* capture */
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0)
+ return err;
+ /* Audio source */
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_audio_src,
+ chip))) < 0)
+ return err;
+ /* IEC958 controls */
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_capture_iec958_mask,
+ chip))) < 0)
+ return err;
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_capture_iec958,
+ chip))) < 0)
+ return err;
+ }
+ /* monitoring only if playback and capture device available */
+ if (chip->nb_streams_capt > 0 && chip->nb_streams_play > 0) {
+ /* monitoring */
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_monitor_vol,
+ chip))) < 0)
+ return err;
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_monitor_sw,
+ chip))) < 0)
+ return err;
+ }
+
+ if (i == 0) {
+ /* clock mode only one control per pcxhr */
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_clock_type,
+ mgr))) < 0)
+ return err;
+ /* non standard control used to scan the external clock presence/frequencies */
+ if ((err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&pcxhr_control_clock_rate,
+ mgr))) < 0)
+ return err;
+ }
+
+ /* init values for the mixer data */
+ pcxhr_init_audio_levels(chip);
+ }
+
+ return 0;
+}
diff --git a/sound/pci/pcxhr/pcxhr_mixer.h b/sound/pci/pcxhr/pcxhr_mixer.h
new file mode 100644
index 0000000..4348d0e
--- /dev/null
+++ b/sound/pci/pcxhr/pcxhr_mixer.h
@@ -0,0 +1,29 @@
+/*
+ * Driver for Digigram pcxhr compatible soundcards
+ *
+ * include file for mixer
+ *
+ * Copyright (c) 2004 by Digigram <alsa@digigram.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_PCXHR_MIXER_H
+#define __SOUND_PCXHR_MIXER_H
+
+/* exported */
+int pcxhr_create_mixer(struct pcxhr_mgr *mgr);
+
+#endif /* __SOUND_PCXHR_MIXER_H */
diff --git a/sound/pci/riptide/Makefile b/sound/pci/riptide/Makefile
new file mode 100644
index 0000000..dcd2e64
--- /dev/null
+++ b/sound/pci/riptide/Makefile
@@ -0,0 +1,3 @@
+snd-riptide-objs := riptide.o
+
+obj-$(CONFIG_SND_RIPTIDE) += snd-riptide.o
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
new file mode 100644
index 0000000..e9f0706
--- /dev/null
+++ b/sound/pci/riptide/riptide.c
@@ -0,0 +1,2239 @@
+/*
+ * Driver for the Conexant Riptide Soundchip
+ *
+ * Copyright (c) 2004 Peter Gruber <nokos@gmx.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+/*
+ History:
+ - 02/15/2004 first release
+
+ This Driver is based on the OSS Driver version from Linuxant (riptide-0.6lnxtbeta03111100)
+ credits from the original files:
+
+ MODULE NAME: cnxt_rt.h
+ AUTHOR: K. Lazarev (Transcribed by KNL)
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Created 02/1/2000 KNL
+
+ MODULE NAME: int_mdl.c
+ AUTHOR: Konstantin Lazarev (Transcribed by KNL)
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Created 10/01/99 KNL
+
+ MODULE NAME: riptide.h
+ AUTHOR: O. Druzhinin (Transcribed by OLD)
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Created 10/16/97 OLD
+
+ MODULE NAME: Rp_Cmdif.cpp
+ AUTHOR: O. Druzhinin (Transcribed by OLD)
+ K. Lazarev (Transcribed by KNL)
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Adopted from NT4 driver 6/22/99 OLD
+ Ported to Linux 9/01/99 KNL
+
+ MODULE NAME: rt_hw.c
+ AUTHOR: O. Druzhinin (Transcribed by OLD)
+ C. Lazarev (Transcribed by CNL)
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Created 11/18/97 OLD
+ Hardware functions for RipTide 11/24/97 CNL
+ (ES1) are coded
+ Hardware functions for RipTide 12/24/97 CNL
+ (A0) are coded
+ Hardware functions for RipTide 03/20/98 CNL
+ (A1) are coded
+ Boot loader is included 05/07/98 CNL
+ Redesigned for WDM 07/27/98 CNL
+ Redesigned for Linux 09/01/99 CNL
+
+ MODULE NAME: rt_hw.h
+ AUTHOR: C. Lazarev (Transcribed by CNL)
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Created 11/18/97 CNL
+
+ MODULE NAME: rt_mdl.c
+ AUTHOR: Konstantin Lazarev (Transcribed by KNL)
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Created 10/01/99 KNL
+
+ MODULE NAME: mixer.h
+ AUTHOR: K. Kenney
+ HISTORY: Major Revision Date By
+ ----------------------------- -------- -----
+ Created from MS W95 Sample 11/28/95 KRS
+ RipTide 10/15/97 KRS
+ Adopted for Windows NT driver 01/20/98 CNL
+*/
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/gameport.h>
+#include <linux/device.h>
+#include <linux/firmware.h>
+#include <asm/io.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/ac97_codec.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK 1
+#endif
+
+MODULE_AUTHOR("Peter Gruber <nokos@gmx.net>");
+MODULE_DESCRIPTION("riptide");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Conexant,Riptide}}");
+MODULE_FIRMWARE("riptide.hex");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE;
+
+#ifdef SUPPORT_JOYSTICK
+static int joystick_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x200 };
+#endif
+static int mpu_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x330 };
+static int opl3_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x388 };
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Riptide soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Riptide soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Riptide soundcard.");
+#ifdef SUPPORT_JOYSTICK
+module_param_array(joystick_port, int, NULL, 0444);
+MODULE_PARM_DESC(joystick_port, "Joystick port # for Riptide soundcard.");
+#endif
+module_param_array(mpu_port, int, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU401 port # for Riptide driver.");
+module_param_array(opl3_port, int, NULL, 0444);
+MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver.");
+
+/*
+ */
+
+#define MPU401_HW_RIPTIDE MPU401_HW_MPU401
+#define OPL3_HW_RIPTIDE OPL3_HW_OPL3
+
+#define PCI_EXT_CapId 0x40
+#define PCI_EXT_NextCapPrt 0x41
+#define PCI_EXT_PWMC 0x42
+#define PCI_EXT_PWSCR 0x44
+#define PCI_EXT_Data00 0x46
+#define PCI_EXT_PMSCR_BSE 0x47
+#define PCI_EXT_SB_Base 0x48
+#define PCI_EXT_FM_Base 0x4a
+#define PCI_EXT_MPU_Base 0x4C
+#define PCI_EXT_Game_Base 0x4E
+#define PCI_EXT_Legacy_Mask 0x50
+#define PCI_EXT_AsicRev 0x52
+#define PCI_EXT_Reserved3 0x53
+
+#define LEGACY_ENABLE_ALL 0x8000 /* legacy device options */
+#define LEGACY_ENABLE_SB 0x4000
+#define LEGACY_ENABLE_FM 0x2000
+#define LEGACY_ENABLE_MPU_INT 0x1000
+#define LEGACY_ENABLE_MPU 0x0800
+#define LEGACY_ENABLE_GAMEPORT 0x0400
+
+#define MAX_WRITE_RETRY 10 /* cmd interface limits */
+#define MAX_ERROR_COUNT 10
+#define CMDIF_TIMEOUT 500000
+#define RESET_TRIES 5
+
+#define READ_PORT_ULONG(p) inl((unsigned long)&(p))
+#define WRITE_PORT_ULONG(p,x) outl(x,(unsigned long)&(p))
+
+#define READ_AUDIO_CONTROL(p) READ_PORT_ULONG(p->audio_control)
+#define WRITE_AUDIO_CONTROL(p,x) WRITE_PORT_ULONG(p->audio_control,x)
+#define UMASK_AUDIO_CONTROL(p,x) WRITE_PORT_ULONG(p->audio_control,READ_PORT_ULONG(p->audio_control)|x)
+#define MASK_AUDIO_CONTROL(p,x) WRITE_PORT_ULONG(p->audio_control,READ_PORT_ULONG(p->audio_control)&x)
+#define READ_AUDIO_STATUS(p) READ_PORT_ULONG(p->audio_status)
+
+#define SET_GRESET(p) UMASK_AUDIO_CONTROL(p,0x0001) /* global reset switch */
+#define UNSET_GRESET(p) MASK_AUDIO_CONTROL(p,~0x0001)
+#define SET_AIE(p) UMASK_AUDIO_CONTROL(p,0x0004) /* interrupt enable */
+#define UNSET_AIE(p) MASK_AUDIO_CONTROL(p,~0x0004)
+#define SET_AIACK(p) UMASK_AUDIO_CONTROL(p,0x0008) /* interrupt acknowledge */
+#define UNSET_AIACKT(p) MASKAUDIO_CONTROL(p,~0x0008)
+#define SET_ECMDAE(p) UMASK_AUDIO_CONTROL(p,0x0010)
+#define UNSET_ECMDAE(p) MASK_AUDIO_CONTROL(p,~0x0010)
+#define SET_ECMDBE(p) UMASK_AUDIO_CONTROL(p,0x0020)
+#define UNSET_ECMDBE(p) MASK_AUDIO_CONTROL(p,~0x0020)
+#define SET_EDATAF(p) UMASK_AUDIO_CONTROL(p,0x0040)
+#define UNSET_EDATAF(p) MASK_AUDIO_CONTROL(p,~0x0040)
+#define SET_EDATBF(p) UMASK_AUDIO_CONTROL(p,0x0080)
+#define UNSET_EDATBF(p) MASK_AUDIO_CONTROL(p,~0x0080)
+#define SET_ESBIRQON(p) UMASK_AUDIO_CONTROL(p,0x0100)
+#define UNSET_ESBIRQON(p) MASK_AUDIO_CONTROL(p,~0x0100)
+#define SET_EMPUIRQ(p) UMASK_AUDIO_CONTROL(p,0x0200)
+#define UNSET_EMPUIRQ(p) MASK_AUDIO_CONTROL(p,~0x0200)
+#define IS_CMDE(a) (READ_PORT_ULONG(a->stat)&0x1) /* cmd empty */
+#define IS_DATF(a) (READ_PORT_ULONG(a->stat)&0x2) /* data filled */
+#define IS_READY(p) (READ_AUDIO_STATUS(p)&0x0001)
+#define IS_DLREADY(p) (READ_AUDIO_STATUS(p)&0x0002)
+#define IS_DLERR(p) (READ_AUDIO_STATUS(p)&0x0004)
+#define IS_GERR(p) (READ_AUDIO_STATUS(p)&0x0008) /* error ! */
+#define IS_CMDAEIRQ(p) (READ_AUDIO_STATUS(p)&0x0010)
+#define IS_CMDBEIRQ(p) (READ_AUDIO_STATUS(p)&0x0020)
+#define IS_DATAFIRQ(p) (READ_AUDIO_STATUS(p)&0x0040)
+#define IS_DATBFIRQ(p) (READ_AUDIO_STATUS(p)&0x0080)
+#define IS_EOBIRQ(p) (READ_AUDIO_STATUS(p)&0x0100) /* interrupt status */
+#define IS_EOSIRQ(p) (READ_AUDIO_STATUS(p)&0x0200)
+#define IS_EOCIRQ(p) (READ_AUDIO_STATUS(p)&0x0400)
+#define IS_UNSLIRQ(p) (READ_AUDIO_STATUS(p)&0x0800)
+#define IS_SBIRQ(p) (READ_AUDIO_STATUS(p)&0x1000)
+#define IS_MPUIRQ(p) (READ_AUDIO_STATUS(p)&0x2000)
+
+#define RESP 0x00000001 /* command flags */
+#define PARM 0x00000002
+#define CMDA 0x00000004
+#define CMDB 0x00000008
+#define NILL 0x00000000
+
+#define LONG0(a) ((u32)a) /* shifts and masks */
+#define BYTE0(a) (LONG0(a)&0xff)
+#define BYTE1(a) (BYTE0(a)<<8)
+#define BYTE2(a) (BYTE0(a)<<16)
+#define BYTE3(a) (BYTE0(a)<<24)
+#define WORD0(a) (LONG0(a)&0xffff)
+#define WORD1(a) (WORD0(a)<<8)
+#define WORD2(a) (WORD0(a)<<16)
+#define TRINIB0(a) (LONG0(a)&0xffffff)
+#define TRINIB1(a) (TRINIB0(a)<<8)
+
+#define RET(a) ((union cmdret *)(a))
+
+#define SEND_GETV(p,b) sendcmd(p,RESP,GETV,0,RET(b)) /* get version */
+#define SEND_GETC(p,b,c) sendcmd(p,PARM|RESP,GETC,c,RET(b))
+#define SEND_GUNS(p,b) sendcmd(p,RESP,GUNS,0,RET(b))
+#define SEND_SCID(p,b) sendcmd(p,RESP,SCID,0,RET(b))
+#define SEND_RMEM(p,b,c,d) sendcmd(p,PARM|RESP,RMEM|BYTE1(b),LONG0(c),RET(d)) /* memory access for firmware write */
+#define SEND_SMEM(p,b,c) sendcmd(p,PARM,SMEM|BYTE1(b),LONG0(c),RET(0)) /* memory access for firmware write */
+#define SEND_WMEM(p,b,c) sendcmd(p,PARM,WMEM|BYTE1(b),LONG0(c),RET(0)) /* memory access for firmware write */
+#define SEND_SDTM(p,b,c) sendcmd(p,PARM|RESP,SDTM|TRINIB1(b),0,RET(c)) /* memory access for firmware write */
+#define SEND_GOTO(p,b) sendcmd(p,PARM,GOTO,LONG0(b),RET(0)) /* memory access for firmware write */
+#define SEND_SETDPLL(p) sendcmd(p,0,ARM_SETDPLL,0,RET(0))
+#define SEND_SSTR(p,b,c) sendcmd(p,PARM,SSTR|BYTE3(b),LONG0(c),RET(0)) /* start stream */
+#define SEND_PSTR(p,b) sendcmd(p,PARM,PSTR,BYTE3(b),RET(0)) /* pause stream */
+#define SEND_KSTR(p,b) sendcmd(p,PARM,KSTR,BYTE3(b),RET(0)) /* stop stream */
+#define SEND_KDMA(p) sendcmd(p,0,KDMA,0,RET(0)) /* stop all dma */
+#define SEND_GPOS(p,b,c,d) sendcmd(p,PARM|RESP,GPOS,BYTE3(c)|BYTE2(b),RET(d)) /* get position in dma */
+#define SEND_SETF(p,b,c,d,e,f,g) sendcmd(p,PARM,SETF|WORD1(b)|BYTE3(c),d|BYTE1(e)|BYTE2(f)|BYTE3(g),RET(0)) /* set sample format at mixer */
+#define SEND_GSTS(p,b,c,d) sendcmd(p,PARM|RESP,GSTS,BYTE3(c)|BYTE2(b),RET(d))
+#define SEND_NGPOS(p,b,c,d) sendcmd(p,PARM|RESP,NGPOS,BYTE3(c)|BYTE2(b),RET(d))
+#define SEND_PSEL(p,b,c) sendcmd(p,PARM,PSEL,BYTE2(b)|BYTE3(c),RET(0)) /* activate lbus path */
+#define SEND_PCLR(p,b,c) sendcmd(p,PARM,PCLR,BYTE2(b)|BYTE3(c),RET(0)) /* deactivate lbus path */
+#define SEND_PLST(p,b) sendcmd(p,PARM,PLST,BYTE3(b),RET(0))
+#define SEND_RSSV(p,b,c,d) sendcmd(p,PARM|RESP,RSSV,BYTE2(b)|BYTE3(c),RET(d))
+#define SEND_LSEL(p,b,c,d,e,f,g,h) sendcmd(p,PARM,LSEL|BYTE1(b)|BYTE2(c)|BYTE3(d),BYTE0(e)|BYTE1(f)|BYTE2(g)|BYTE3(h),RET(0)) /* select paths for internal connections */
+#define SEND_SSRC(p,b,c,d,e) sendcmd(p,PARM,SSRC|BYTE1(b)|WORD2(c),WORD0(d)|WORD2(e),RET(0)) /* configure source */
+#define SEND_SLST(p,b) sendcmd(p,PARM,SLST,BYTE3(b),RET(0))
+#define SEND_RSRC(p,b,c) sendcmd(p,RESP,RSRC|BYTE1(b),0,RET(c)) /* read source config */
+#define SEND_SSRB(p,b,c) sendcmd(p,PARM,SSRB|BYTE1(b),WORD2(c),RET(0))
+#define SEND_SDGV(p,b,c,d,e) sendcmd(p,PARM,SDGV|BYTE2(b)|BYTE3(c),WORD0(d)|WORD2(e),RET(0)) /* set digital mixer */
+#define SEND_RDGV(p,b,c,d) sendcmd(p,PARM|RESP,RDGV|BYTE2(b)|BYTE3(c),0,RET(d)) /* read digital mixer */
+#define SEND_DLST(p,b) sendcmd(p,PARM,DLST,BYTE3(b),RET(0))
+#define SEND_SACR(p,b,c) sendcmd(p,PARM,SACR,WORD0(b)|WORD2(c),RET(0)) /* set AC97 register */
+#define SEND_RACR(p,b,c) sendcmd(p,PARM|RESP,RACR,WORD2(b),RET(c)) /* get AC97 register */
+#define SEND_ALST(p,b) sendcmd(p,PARM,ALST,BYTE3(b),RET(0))
+#define SEND_TXAC(p,b,c,d,e,f) sendcmd(p,PARM,TXAC|BYTE1(b)|WORD2(c),WORD0(d)|BYTE2(e)|BYTE3(f),RET(0))
+#define SEND_RXAC(p,b,c,d) sendcmd(p,PARM|RESP,RXAC,BYTE2(b)|BYTE3(c),RET(d))
+#define SEND_SI2S(p,b) sendcmd(p,PARM,SI2S,WORD2(b),RET(0))
+
+#define EOB_STATUS 0x80000000 /* status flags : block boundary */
+#define EOS_STATUS 0x40000000 /* : stoppped */
+#define EOC_STATUS 0x20000000 /* : stream end */
+#define ERR_STATUS 0x10000000
+#define EMPTY_STATUS 0x08000000
+
+#define IEOB_ENABLE 0x1 /* enable interrupts for status notification above */
+#define IEOS_ENABLE 0x2
+#define IEOC_ENABLE 0x4
+#define RDONCE 0x8
+#define DESC_MAX_MASK 0xff
+
+#define ST_PLAY 0x1 /* stream states */
+#define ST_STOP 0x2
+#define ST_PAUSE 0x4
+
+#define I2S_INTDEC 3 /* config for I2S link */
+#define I2S_MERGER 0
+#define I2S_SPLITTER 0
+#define I2S_MIXER 7
+#define I2S_RATE 44100
+
+#define MODEM_INTDEC 4 /* config for modem link */
+#define MODEM_MERGER 3
+#define MODEM_SPLITTER 0
+#define MODEM_MIXER 11
+
+#define FM_INTDEC 3 /* config for FM/OPL3 link */
+#define FM_MERGER 0
+#define FM_SPLITTER 0
+#define FM_MIXER 9
+
+#define SPLIT_PATH 0x80 /* path splitting flag */
+
+enum FIRMWARE {
+ DATA_REC = 0, EXT_END_OF_FILE, EXT_SEG_ADDR_REC, EXT_GOTO_CMD_REC,
+ EXT_LIN_ADDR_REC,
+};
+
+enum CMDS {
+ GETV = 0x00, GETC, GUNS, SCID, RMEM =
+ 0x10, SMEM, WMEM, SDTM, GOTO, SSTR =
+ 0x20, PSTR, KSTR, KDMA, GPOS, SETF, GSTS, NGPOS, PSEL =
+ 0x30, PCLR, PLST, RSSV, LSEL, SSRC = 0x40, SLST, RSRC, SSRB, SDGV =
+ 0x50, RDGV, DLST, SACR = 0x60, RACR, ALST, TXAC, RXAC, SI2S =
+ 0x70, ARM_SETDPLL = 0x72,
+};
+
+enum E1SOURCE {
+ ARM2LBUS_FIFO0 = 0, ARM2LBUS_FIFO1, ARM2LBUS_FIFO2, ARM2LBUS_FIFO3,
+ ARM2LBUS_FIFO4, ARM2LBUS_FIFO5, ARM2LBUS_FIFO6, ARM2LBUS_FIFO7,
+ ARM2LBUS_FIFO8, ARM2LBUS_FIFO9, ARM2LBUS_FIFO10, ARM2LBUS_FIFO11,
+ ARM2LBUS_FIFO12, ARM2LBUS_FIFO13, ARM2LBUS_FIFO14, ARM2LBUS_FIFO15,
+ INTER0_OUT, INTER1_OUT, INTER2_OUT, INTER3_OUT, INTER4_OUT,
+ INTERM0_OUT, INTERM1_OUT, INTERM2_OUT, INTERM3_OUT, INTERM4_OUT,
+ INTERM5_OUT, INTERM6_OUT, DECIMM0_OUT, DECIMM1_OUT, DECIMM2_OUT,
+ DECIMM3_OUT, DECIM0_OUT, SR3_4_OUT, OPL3_SAMPLE, ASRC0, ASRC1,
+ ACLNK2PADC, ACLNK2MODEM0RX, ACLNK2MIC, ACLNK2MODEM1RX, ACLNK2HNDMIC,
+ DIGITAL_MIXER_OUT0, GAINFUNC0_OUT, GAINFUNC1_OUT, GAINFUNC2_OUT,
+ GAINFUNC3_OUT, GAINFUNC4_OUT, SOFTMODEMTX, SPLITTER0_OUTL,
+ SPLITTER0_OUTR, SPLITTER1_OUTL, SPLITTER1_OUTR, SPLITTER2_OUTL,
+ SPLITTER2_OUTR, SPLITTER3_OUTL, SPLITTER3_OUTR, MERGER0_OUT,
+ MERGER1_OUT, MERGER2_OUT, MERGER3_OUT, ARM2LBUS_FIFO_DIRECT, NO_OUT
+};
+
+enum E2SINK {
+ LBUS2ARM_FIFO0 = 0, LBUS2ARM_FIFO1, LBUS2ARM_FIFO2, LBUS2ARM_FIFO3,
+ LBUS2ARM_FIFO4, LBUS2ARM_FIFO5, LBUS2ARM_FIFO6, LBUS2ARM_FIFO7,
+ INTER0_IN, INTER1_IN, INTER2_IN, INTER3_IN, INTER4_IN, INTERM0_IN,
+ INTERM1_IN, INTERM2_IN, INTERM3_IN, INTERM4_IN, INTERM5_IN, INTERM6_IN,
+ DECIMM0_IN, DECIMM1_IN, DECIMM2_IN, DECIMM3_IN, DECIM0_IN, SR3_4_IN,
+ PDAC2ACLNK, MODEM0TX2ACLNK, MODEM1TX2ACLNK, HNDSPK2ACLNK,
+ DIGITAL_MIXER_IN0, DIGITAL_MIXER_IN1, DIGITAL_MIXER_IN2,
+ DIGITAL_MIXER_IN3, DIGITAL_MIXER_IN4, DIGITAL_MIXER_IN5,
+ DIGITAL_MIXER_IN6, DIGITAL_MIXER_IN7, DIGITAL_MIXER_IN8,
+ DIGITAL_MIXER_IN9, DIGITAL_MIXER_IN10, DIGITAL_MIXER_IN11,
+ GAINFUNC0_IN, GAINFUNC1_IN, GAINFUNC2_IN, GAINFUNC3_IN, GAINFUNC4_IN,
+ SOFTMODEMRX, SPLITTER0_IN, SPLITTER1_IN, SPLITTER2_IN, SPLITTER3_IN,
+ MERGER0_INL, MERGER0_INR, MERGER1_INL, MERGER1_INR, MERGER2_INL,
+ MERGER2_INR, MERGER3_INL, MERGER3_INR, E2SINK_MAX
+};
+
+enum LBUS_SINK {
+ LS_SRC_INTERPOLATOR = 0, LS_SRC_INTERPOLATORM, LS_SRC_DECIMATOR,
+ LS_SRC_DECIMATORM, LS_MIXER_IN, LS_MIXER_GAIN_FUNCTION,
+ LS_SRC_SPLITTER, LS_SRC_MERGER, LS_NONE1, LS_NONE2,
+};
+
+enum RT_CHANNEL_IDS {
+ M0TX = 0, M1TX, TAMTX, HSSPKR, PDAC, DSNDTX0, DSNDTX1, DSNDTX2,
+ DSNDTX3, DSNDTX4, DSNDTX5, DSNDTX6, DSNDTX7, WVSTRTX, COP3DTX, SPARE,
+ M0RX, HSMIC, M1RX, CLEANRX, MICADC, PADC, COPRX1, COPRX2,
+ CHANNEL_ID_COUNTER
+};
+
+enum { SB_CMD = 0, MODEM_CMD, I2S_CMD0, I2S_CMD1, FM_CMD, MAX_CMD };
+
+struct lbuspath {
+ unsigned char *noconv;
+ unsigned char *stereo;
+ unsigned char *mono;
+};
+
+struct cmdport {
+ u32 data1; /* cmd,param */
+ u32 data2; /* param */
+ u32 stat; /* status */
+ u32 pad[5];
+};
+
+struct riptideport {
+ u32 audio_control; /* status registers */
+ u32 audio_status;
+ u32 pad[2];
+ struct cmdport port[2]; /* command ports */
+};
+
+struct cmdif {
+ struct riptideport *hwport;
+ spinlock_t lock;
+ unsigned int cmdcnt; /* cmd statistics */
+ unsigned int cmdtime;
+ unsigned int cmdtimemax;
+ unsigned int cmdtimemin;
+ unsigned int errcnt;
+ int is_reset;
+};
+
+struct riptide_firmware {
+ u16 ASIC;
+ u16 CODEC;
+ u16 AUXDSP;
+ u16 PROG;
+};
+
+union cmdret {
+ u8 retbytes[8];
+ u16 retwords[4];
+ u32 retlongs[2];
+};
+
+union firmware_version {
+ union cmdret ret;
+ struct riptide_firmware firmware;
+};
+
+#define get_pcmhwdev(substream) (struct pcmhw *)(substream->runtime->private_data)
+
+#define PLAYBACK_SUBSTREAMS 3
+struct snd_riptide {
+ struct snd_card *card;
+ struct pci_dev *pci;
+ const struct firmware *fw_entry;
+
+ struct cmdif *cif;
+
+ struct snd_pcm *pcm;
+ struct snd_pcm *pcm_i2s;
+ struct snd_rawmidi *rmidi;
+ struct snd_opl3 *opl3;
+ struct snd_ac97 *ac97;
+ struct snd_ac97_bus *ac97_bus;
+
+ struct snd_pcm_substream *playback_substream[PLAYBACK_SUBSTREAMS];
+ struct snd_pcm_substream *capture_substream;
+
+ int openstreams;
+
+ int irq;
+ unsigned long port;
+ unsigned short mpuaddr;
+ unsigned short opladdr;
+#ifdef SUPPORT_JOYSTICK
+ unsigned short gameaddr;
+#endif
+ struct resource *res_port;
+
+ unsigned short device_id;
+
+ union firmware_version firmware;
+
+ spinlock_t lock;
+ struct tasklet_struct riptide_tq;
+ struct snd_info_entry *proc_entry;
+
+ unsigned long received_irqs;
+ unsigned long handled_irqs;
+#ifdef CONFIG_PM
+ int in_suspend;
+#endif
+};
+
+struct sgd { /* scatter gather desriptor */
+ u32 dwNextLink;
+ u32 dwSegPtrPhys;
+ u32 dwSegLen;
+ u32 dwStat_Ctl;
+};
+
+struct pcmhw { /* pcm descriptor */
+ struct lbuspath paths;
+ unsigned char *lbuspath;
+ unsigned char source;
+ unsigned char intdec[2];
+ unsigned char mixer;
+ unsigned char id;
+ unsigned char state;
+ unsigned int rate;
+ unsigned int channels;
+ snd_pcm_format_t format;
+ struct snd_dma_buffer sgdlist;
+ struct sgd *sgdbuf;
+ unsigned int size;
+ unsigned int pages;
+ unsigned int oldpos;
+ unsigned int pointer;
+};
+
+#define CMDRET_ZERO (union cmdret){{(u32)0, (u32) 0}}
+
+static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
+ union cmdret *ret);
+static int getsourcesink(struct cmdif *cif, unsigned char source,
+ unsigned char sink, unsigned char *a,
+ unsigned char *b);
+static int snd_riptide_initialize(struct snd_riptide *chip);
+static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip);
+
+/*
+ */
+
+static struct pci_device_id snd_riptide_ids[] = {
+ {
+ .vendor = 0x127a,.device = 0x4310,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {
+ .vendor = 0x127a,.device = 0x4320,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {
+ .vendor = 0x127a,.device = 0x4330,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {
+ .vendor = 0x127a,.device = 0x4340,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {0,},
+};
+
+#ifdef SUPPORT_JOYSTICK
+static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = {
+ {
+ .vendor = 0x127a,.device = 0x4312,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {
+ .vendor = 0x127a,.device = 0x4322,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {.vendor = 0x127a,.device = 0x4332,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {.vendor = 0x127a,.device = 0x4342,
+ .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
+ },
+ {0,},
+};
+#endif
+
+MODULE_DEVICE_TABLE(pci, snd_riptide_ids);
+
+/*
+ */
+
+static unsigned char lbusin2out[E2SINK_MAX + 1][2] = {
+ {NO_OUT, LS_NONE1}, {NO_OUT, LS_NONE2}, {NO_OUT, LS_NONE1}, {NO_OUT,
+ LS_NONE2},
+ {NO_OUT, LS_NONE1}, {NO_OUT, LS_NONE2}, {NO_OUT, LS_NONE1}, {NO_OUT,
+ LS_NONE2},
+ {INTER0_OUT, LS_SRC_INTERPOLATOR}, {INTER1_OUT, LS_SRC_INTERPOLATOR},
+ {INTER2_OUT, LS_SRC_INTERPOLATOR}, {INTER3_OUT, LS_SRC_INTERPOLATOR},
+ {INTER4_OUT, LS_SRC_INTERPOLATOR}, {INTERM0_OUT, LS_SRC_INTERPOLATORM},
+ {INTERM1_OUT, LS_SRC_INTERPOLATORM}, {INTERM2_OUT,
+ LS_SRC_INTERPOLATORM},
+ {INTERM3_OUT, LS_SRC_INTERPOLATORM}, {INTERM4_OUT,
+ LS_SRC_INTERPOLATORM},
+ {INTERM5_OUT, LS_SRC_INTERPOLATORM}, {INTERM6_OUT,
+ LS_SRC_INTERPOLATORM},
+ {DECIMM0_OUT, LS_SRC_DECIMATORM}, {DECIMM1_OUT, LS_SRC_DECIMATORM},
+ {DECIMM2_OUT, LS_SRC_DECIMATORM}, {DECIMM3_OUT, LS_SRC_DECIMATORM},
+ {DECIM0_OUT, LS_SRC_DECIMATOR}, {SR3_4_OUT, LS_NONE1}, {NO_OUT,
+ LS_NONE2},
+ {NO_OUT, LS_NONE1}, {NO_OUT, LS_NONE2}, {NO_OUT, LS_NONE1},
+ {DIGITAL_MIXER_OUT0, LS_MIXER_IN}, {DIGITAL_MIXER_OUT0, LS_MIXER_IN},
+ {DIGITAL_MIXER_OUT0, LS_MIXER_IN}, {DIGITAL_MIXER_OUT0, LS_MIXER_IN},
+ {DIGITAL_MIXER_OUT0, LS_MIXER_IN}, {DIGITAL_MIXER_OUT0, LS_MIXER_IN},
+ {DIGITAL_MIXER_OUT0, LS_MIXER_IN}, {DIGITAL_MIXER_OUT0, LS_MIXER_IN},
+ {DIGITAL_MIXER_OUT0, LS_MIXER_IN}, {DIGITAL_MIXER_OUT0, LS_MIXER_IN},
+ {DIGITAL_MIXER_OUT0, LS_MIXER_IN}, {DIGITAL_MIXER_OUT0, LS_MIXER_IN},
+ {GAINFUNC0_OUT, LS_MIXER_GAIN_FUNCTION}, {GAINFUNC1_OUT,
+ LS_MIXER_GAIN_FUNCTION},
+ {GAINFUNC2_OUT, LS_MIXER_GAIN_FUNCTION}, {GAINFUNC3_OUT,
+ LS_MIXER_GAIN_FUNCTION},
+ {GAINFUNC4_OUT, LS_MIXER_GAIN_FUNCTION}, {SOFTMODEMTX, LS_NONE1},
+ {SPLITTER0_OUTL, LS_SRC_SPLITTER}, {SPLITTER1_OUTL, LS_SRC_SPLITTER},
+ {SPLITTER2_OUTL, LS_SRC_SPLITTER}, {SPLITTER3_OUTL, LS_SRC_SPLITTER},
+ {MERGER0_OUT, LS_SRC_MERGER}, {MERGER0_OUT, LS_SRC_MERGER},
+ {MERGER1_OUT, LS_SRC_MERGER},
+ {MERGER1_OUT, LS_SRC_MERGER}, {MERGER2_OUT, LS_SRC_MERGER},
+ {MERGER2_OUT, LS_SRC_MERGER},
+ {MERGER3_OUT, LS_SRC_MERGER}, {MERGER3_OUT, LS_SRC_MERGER}, {NO_OUT,
+ LS_NONE2},
+};
+
+static unsigned char lbus_play_opl3[] = {
+ DIGITAL_MIXER_IN0 + FM_MIXER, 0xff
+};
+static unsigned char lbus_play_modem[] = {
+ DIGITAL_MIXER_IN0 + MODEM_MIXER, 0xff
+};
+static unsigned char lbus_play_i2s[] = {
+ INTER0_IN + I2S_INTDEC, DIGITAL_MIXER_IN0 + I2S_MIXER, 0xff
+};
+static unsigned char lbus_play_out[] = {
+ PDAC2ACLNK, 0xff
+};
+static unsigned char lbus_play_outhp[] = {
+ HNDSPK2ACLNK, 0xff
+};
+static unsigned char lbus_play_noconv1[] = {
+ DIGITAL_MIXER_IN0, 0xff
+};
+static unsigned char lbus_play_stereo1[] = {
+ INTER0_IN, DIGITAL_MIXER_IN0, 0xff
+};
+static unsigned char lbus_play_mono1[] = {
+ INTERM0_IN, DIGITAL_MIXER_IN0, 0xff
+};
+static unsigned char lbus_play_noconv2[] = {
+ DIGITAL_MIXER_IN1, 0xff
+};
+static unsigned char lbus_play_stereo2[] = {
+ INTER1_IN, DIGITAL_MIXER_IN1, 0xff
+};
+static unsigned char lbus_play_mono2[] = {
+ INTERM1_IN, DIGITAL_MIXER_IN1, 0xff
+};
+static unsigned char lbus_play_noconv3[] = {
+ DIGITAL_MIXER_IN2, 0xff
+};
+static unsigned char lbus_play_stereo3[] = {
+ INTER2_IN, DIGITAL_MIXER_IN2, 0xff
+};
+static unsigned char lbus_play_mono3[] = {
+ INTERM2_IN, DIGITAL_MIXER_IN2, 0xff
+};
+static unsigned char lbus_rec_noconv1[] = {
+ LBUS2ARM_FIFO5, 0xff
+};
+static unsigned char lbus_rec_stereo1[] = {
+ DECIM0_IN, LBUS2ARM_FIFO5, 0xff
+};
+static unsigned char lbus_rec_mono1[] = {
+ DECIMM3_IN, LBUS2ARM_FIFO5, 0xff
+};
+
+static unsigned char play_ids[] = { 4, 1, 2, };
+static unsigned char play_sources[] = {
+ ARM2LBUS_FIFO4, ARM2LBUS_FIFO1, ARM2LBUS_FIFO2,
+};
+static struct lbuspath lbus_play_paths[] = {
+ {
+ .noconv = lbus_play_noconv1,
+ .stereo = lbus_play_stereo1,
+ .mono = lbus_play_mono1,
+ },
+ {
+ .noconv = lbus_play_noconv2,
+ .stereo = lbus_play_stereo2,
+ .mono = lbus_play_mono2,
+ },
+ {
+ .noconv = lbus_play_noconv3,
+ .stereo = lbus_play_stereo3,
+ .mono = lbus_play_mono3,
+ },
+};
+static struct lbuspath lbus_rec_path = {
+ .noconv = lbus_rec_noconv1,
+ .stereo = lbus_rec_stereo1,
+ .mono = lbus_rec_mono1,
+};
+
+#define FIRMWARE_VERSIONS 1
+static union firmware_version firmware_versions[] = {
+ {
+ .firmware = {
+ .ASIC = 3,
+ .CODEC = 2,
+ .AUXDSP = 3,
+ .PROG = 773,
+ },
+ },
+};
+
+static u32 atoh(const unsigned char *in, unsigned int len)
+{
+ u32 sum = 0;
+ unsigned int mult = 1;
+ unsigned char c;
+
+ while (len) {
+ c = in[len - 1];
+ if ((c >= '0') && (c <= '9'))
+ sum += mult * (c - '0');
+ else if ((c >= 'A') && (c <= 'F'))
+ sum += mult * (c - ('A' - 10));
+ else if ((c >= 'a') && (c <= 'f'))
+ sum += mult * (c - ('a' - 10));
+ mult *= 16;
+ --len;
+ }
+ return sum;
+}
+
+static int senddata(struct cmdif *cif, const unsigned char *in, u32 offset)
+{
+ u32 addr;
+ u32 data;
+ u32 i;
+ const unsigned char *p;
+
+ i = atoh(&in[1], 2);
+ addr = offset + atoh(&in[3], 4);
+ if (SEND_SMEM(cif, 0, addr) != 0)
+ return -EACCES;
+ p = in + 9;
+ while (i) {
+ data = atoh(p, 8);
+ if (SEND_WMEM(cif, 2,
+ ((data & 0x0f0f0f0f) << 4) | ((data & 0xf0f0f0f0)
+ >> 4)))
+ return -EACCES;
+ i -= 4;
+ p += 8;
+ }
+ return 0;
+}
+
+static int loadfirmware(struct cmdif *cif, const unsigned char *img,
+ unsigned int size)
+{
+ const unsigned char *in;
+ u32 laddr, saddr, t, val;
+ int err = 0;
+
+ laddr = saddr = 0;
+ while (size > 0 && err == 0) {
+ in = img;
+ if (in[0] == ':') {
+ t = atoh(&in[7], 2);
+ switch (t) {
+ case DATA_REC:
+ err = senddata(cif, in, laddr + saddr);
+ break;
+ case EXT_SEG_ADDR_REC:
+ saddr = atoh(&in[9], 4) << 4;
+ break;
+ case EXT_LIN_ADDR_REC:
+ laddr = atoh(&in[9], 4) << 16;
+ break;
+ case EXT_GOTO_CMD_REC:
+ val = atoh(&in[9], 8);
+ if (SEND_GOTO(cif, val) != 0)
+ err = -EACCES;
+ break;
+ case EXT_END_OF_FILE:
+ size = 0;
+ break;
+ default:
+ break;
+ }
+ while (size > 0) {
+ size--;
+ if (*img++ == '\n')
+ break;
+ }
+ }
+ }
+ snd_printdd("load firmware return %d\n", err);
+ return err;
+}
+
+static void
+alloclbuspath(struct cmdif *cif, unsigned char source,
+ unsigned char *path, unsigned char *mixer, unsigned char *s)
+{
+ while (*path != 0xff) {
+ unsigned char sink, type;
+
+ sink = *path & (~SPLIT_PATH);
+ if (sink != E2SINK_MAX) {
+ snd_printdd("alloc path 0x%x->0x%x\n", source, sink);
+ SEND_PSEL(cif, source, sink);
+ source = lbusin2out[sink][0];
+ type = lbusin2out[sink][1];
+ if (type == LS_MIXER_IN) {
+ if (mixer)
+ *mixer = sink - DIGITAL_MIXER_IN0;
+ }
+ if (type == LS_SRC_DECIMATORM ||
+ type == LS_SRC_DECIMATOR ||
+ type == LS_SRC_INTERPOLATORM ||
+ type == LS_SRC_INTERPOLATOR) {
+ if (s) {
+ if (s[0] != 0xff)
+ s[1] = sink;
+ else
+ s[0] = sink;
+ }
+ }
+ }
+ if (*path++ & SPLIT_PATH) {
+ unsigned char *npath = path;
+
+ while (*npath != 0xff)
+ npath++;
+ alloclbuspath(cif, source + 1, ++npath, mixer, s);
+ }
+ }
+}
+
+static void
+freelbuspath(struct cmdif *cif, unsigned char source, unsigned char *path)
+{
+ while (*path != 0xff) {
+ unsigned char sink;
+
+ sink = *path & (~SPLIT_PATH);
+ if (sink != E2SINK_MAX) {
+ snd_printdd("free path 0x%x->0x%x\n", source, sink);
+ SEND_PCLR(cif, source, sink);
+ source = lbusin2out[sink][0];
+ }
+ if (*path++ & SPLIT_PATH) {
+ unsigned char *npath = path;
+
+ while (*npath != 0xff)
+ npath++;
+ freelbuspath(cif, source + 1, ++npath);
+ }
+ }
+}
+
+static int writearm(struct cmdif *cif, u32 addr, u32 data, u32 mask)
+{
+ union cmdret rptr = CMDRET_ZERO;
+ unsigned int i = MAX_WRITE_RETRY;
+ int flag = 1;
+
+ SEND_RMEM(cif, 0x02, addr, &rptr);
+ rptr.retlongs[0] &= (~mask);
+
+ while (--i) {
+ SEND_SMEM(cif, 0x01, addr);
+ SEND_WMEM(cif, 0x02, (rptr.retlongs[0] | data));
+ SEND_RMEM(cif, 0x02, addr, &rptr);
+ if ((rptr.retlongs[0] & data) == data) {
+ flag = 0;
+ break;
+ } else
+ rptr.retlongs[0] &= ~mask;
+ }
+ snd_printdd("send arm 0x%x 0x%x 0x%x return %d\n", addr, data, mask,
+ flag);
+ return flag;
+}
+
+static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm,
+ union cmdret *ret)
+{
+ int i, j;
+ int err;
+ unsigned int time = 0;
+ unsigned long irqflags;
+ struct riptideport *hwport;
+ struct cmdport *cmdport = NULL;
+
+ if (snd_BUG_ON(!cif))
+ return -EINVAL;
+
+ hwport = cif->hwport;
+ if (cif->errcnt > MAX_ERROR_COUNT) {
+ if (cif->is_reset) {
+ snd_printk(KERN_ERR
+ "Riptide: Too many failed cmds, reinitializing\n");
+ if (riptide_reset(cif, NULL) == 0) {
+ cif->errcnt = 0;
+ return -EIO;
+ }
+ }
+ snd_printk(KERN_ERR "Riptide: Initialization failed.\n");
+ return -EINVAL;
+ }
+ if (ret) {
+ ret->retlongs[0] = 0;
+ ret->retlongs[1] = 0;
+ }
+ i = 0;
+ spin_lock_irqsave(&cif->lock, irqflags);
+ while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport))
+ udelay(10);
+ if (i >= CMDIF_TIMEOUT) {
+ err = -EBUSY;
+ goto errout;
+ }
+
+ err = 0;
+ for (j = 0, time = 0; time < CMDIF_TIMEOUT; j++, time += 2) {
+ cmdport = &(hwport->port[j % 2]);
+ if (IS_DATF(cmdport)) { /* free pending data */
+ READ_PORT_ULONG(cmdport->data1);
+ READ_PORT_ULONG(cmdport->data2);
+ }
+ if (IS_CMDE(cmdport)) {
+ if (flags & PARM) /* put data */
+ WRITE_PORT_ULONG(cmdport->data2, parm);
+ WRITE_PORT_ULONG(cmdport->data1, cmd); /* write cmd */
+ if ((flags & RESP) && ret) {
+ while (!IS_DATF(cmdport) &&
+ time++ < CMDIF_TIMEOUT)
+ udelay(10);
+ if (time < CMDIF_TIMEOUT) { /* read response */
+ ret->retlongs[0] =
+ READ_PORT_ULONG(cmdport->data1);
+ ret->retlongs[1] =
+ READ_PORT_ULONG(cmdport->data2);
+ } else {
+ err = -ENOSYS;
+ goto errout;
+ }
+ }
+ break;
+ }
+ udelay(20);
+ }
+ if (time == CMDIF_TIMEOUT) {
+ err = -ENODATA;
+ goto errout;
+ }
+ spin_unlock_irqrestore(&cif->lock, irqflags);
+
+ cif->cmdcnt++; /* update command statistics */
+ cif->cmdtime += time;
+ if (time > cif->cmdtimemax)
+ cif->cmdtimemax = time;
+ if (time < cif->cmdtimemin)
+ cif->cmdtimemin = time;
+ if ((cif->cmdcnt) % 1000 == 0)
+ snd_printdd
+ ("send cmd %d time: %d mintime: %d maxtime %d err: %d\n",
+ cif->cmdcnt, cif->cmdtime, cif->cmdtimemin,
+ cif->cmdtimemax, cif->errcnt);
+ return 0;
+
+ errout:
+ cif->errcnt++;
+ spin_unlock_irqrestore(&cif->lock, irqflags);
+ snd_printdd
+ ("send cmd %d hw: 0x%x flag: 0x%x cmd: 0x%x parm: 0x%x ret: 0x%x 0x%x CMDE: %d DATF: %d failed %d\n",
+ cif->cmdcnt, (int)((void *)&(cmdport->stat) - (void *)hwport),
+ flags, cmd, parm, ret ? ret->retlongs[0] : 0,
+ ret ? ret->retlongs[1] : 0, IS_CMDE(cmdport), IS_DATF(cmdport),
+ err);
+ return err;
+}
+
+static int
+setmixer(struct cmdif *cif, short num, unsigned short rval, unsigned short lval)
+{
+ union cmdret rptr = CMDRET_ZERO;
+ int i = 0;
+
+ snd_printdd("sent mixer %d: 0x%d 0x%d\n", num, rval, lval);
+ do {
+ SEND_SDGV(cif, num, num, rval, lval);
+ SEND_RDGV(cif, num, num, &rptr);
+ if (rptr.retwords[0] == lval && rptr.retwords[1] == rval)
+ return 0;
+ } while (i++ < MAX_WRITE_RETRY);
+ snd_printdd("sent mixer failed\n");
+ return -EIO;
+}
+
+static int getpaths(struct cmdif *cif, unsigned char *o)
+{
+ unsigned char src[E2SINK_MAX];
+ unsigned char sink[E2SINK_MAX];
+ int i, j = 0;
+
+ for (i = 0; i < E2SINK_MAX; i++) {
+ getsourcesink(cif, i, i, &src[i], &sink[i]);
+ if (sink[i] < E2SINK_MAX) {
+ o[j++] = sink[i];
+ o[j++] = i;
+ }
+ }
+ return j;
+}
+
+static int
+getsourcesink(struct cmdif *cif, unsigned char source, unsigned char sink,
+ unsigned char *a, unsigned char *b)
+{
+ union cmdret rptr = CMDRET_ZERO;
+
+ if (SEND_RSSV(cif, source, sink, &rptr) &&
+ SEND_RSSV(cif, source, sink, &rptr))
+ return -EIO;
+ *a = rptr.retbytes[0];
+ *b = rptr.retbytes[1];
+ snd_printdd("getsourcesink 0x%x 0x%x\n", *a, *b);
+ return 0;
+}
+
+static int
+getsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int *rate)
+{
+ unsigned char *s;
+ unsigned int p[2] = { 0, 0 };
+ int i;
+ union cmdret rptr = CMDRET_ZERO;
+
+ s = intdec;
+ for (i = 0; i < 2; i++) {
+ if (*s != 0xff) {
+ if (SEND_RSRC(cif, *s, &rptr) &&
+ SEND_RSRC(cif, *s, &rptr))
+ return -EIO;
+ p[i] += rptr.retwords[1];
+ p[i] *= rptr.retwords[2];
+ p[i] += rptr.retwords[3];
+ p[i] /= 65536;
+ }
+ s++;
+ }
+ if (p[0]) {
+ if (p[1] != p[0])
+ snd_printdd("rates differ %d %d\n", p[0], p[1]);
+ *rate = (unsigned int)p[0];
+ } else
+ *rate = (unsigned int)p[1];
+ snd_printdd("getsampleformat %d %d %d\n", intdec[0], intdec[1], *rate);
+ return 0;
+}
+
+static int
+setsampleformat(struct cmdif *cif,
+ unsigned char mixer, unsigned char id,
+ unsigned char channels, unsigned char format)
+{
+ unsigned char w, ch, sig, order;
+
+ snd_printdd
+ ("setsampleformat mixer: %d id: %d channels: %d format: %d\n",
+ mixer, id, channels, format);
+ ch = channels == 1;
+ w = snd_pcm_format_width(format) == 8;
+ sig = snd_pcm_format_unsigned(format) != 0;
+ order = snd_pcm_format_big_endian(format) != 0;
+
+ if (SEND_SETF(cif, mixer, w, ch, order, sig, id) &&
+ SEND_SETF(cif, mixer, w, ch, order, sig, id)) {
+ snd_printdd("setsampleformat failed\n");
+ return -EIO;
+ }
+ return 0;
+}
+
+static int
+setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate)
+{
+ u32 D, M, N;
+ union cmdret rptr = CMDRET_ZERO;
+ int i;
+
+ snd_printdd("setsamplerate intdec: %d,%d rate: %d\n", intdec[0],
+ intdec[1], rate);
+ D = 48000;
+ M = ((rate == 48000) ? 47999 : rate) * 65536;
+ N = M % D;
+ M /= D;
+ for (i = 0; i < 2; i++) {
+ if (*intdec != 0xff) {
+ do {
+ SEND_SSRC(cif, *intdec, D, M, N);
+ SEND_RSRC(cif, *intdec, &rptr);
+ } while (rptr.retwords[1] != D &&
+ rptr.retwords[2] != M &&
+ rptr.retwords[3] != N &&
+ i++ < MAX_WRITE_RETRY);
+ if (i == MAX_WRITE_RETRY) {
+ snd_printdd("sent samplerate %d: %d failed\n",
+ *intdec, rate);
+ return -EIO;
+ }
+ }
+ intdec++;
+ }
+ return 0;
+}
+
+static int
+getmixer(struct cmdif *cif, short num, unsigned short *rval,
+ unsigned short *lval)
+{
+ union cmdret rptr = CMDRET_ZERO;
+
+ if (SEND_RDGV(cif, num, num, &rptr) && SEND_RDGV(cif, num, num, &rptr))
+ return -EIO;
+ *rval = rptr.retwords[0];
+ *lval = rptr.retwords[1];
+ snd_printdd("got mixer %d: 0x%d 0x%d\n", num, *rval, *lval);
+ return 0;
+}
+
+static void riptide_handleirq(unsigned long dev_id)
+{
+ struct snd_riptide *chip = (void *)dev_id;
+ struct cmdif *cif = chip->cif;
+ struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1];
+ struct snd_pcm_runtime *runtime;
+ struct pcmhw *data = NULL;
+ unsigned int pos, period_bytes;
+ struct sgd *c;
+ int i, j;
+ unsigned int flag;
+
+ if (!cif)
+ return;
+
+ for (i = 0; i < PLAYBACK_SUBSTREAMS; i++)
+ substream[i] = chip->playback_substream[i];
+ substream[i] = chip->capture_substream;
+ for (i = 0; i < PLAYBACK_SUBSTREAMS + 1; i++) {
+ if (substream[i] &&
+ (runtime = substream[i]->runtime) &&
+ (data = runtime->private_data) && data->state != ST_STOP) {
+ pos = 0;
+ for (j = 0; j < data->pages; j++) {
+ c = &data->sgdbuf[j];
+ flag = le32_to_cpu(c->dwStat_Ctl);
+ if (flag & EOB_STATUS)
+ pos += le32_to_cpu(c->dwSegLen);
+ if (flag & EOC_STATUS)
+ pos += le32_to_cpu(c->dwSegLen);
+ if ((flag & EOS_STATUS)
+ && (data->state == ST_PLAY)) {
+ data->state = ST_STOP;
+ snd_printk(KERN_ERR
+ "Riptide: DMA stopped unexpectedly\n");
+ }
+ c->dwStat_Ctl =
+ cpu_to_le32(flag &
+ ~(EOS_STATUS | EOB_STATUS |
+ EOC_STATUS));
+ }
+ data->pointer += pos;
+ pos += data->oldpos;
+ if (data->state != ST_STOP) {
+ period_bytes =
+ frames_to_bytes(runtime,
+ runtime->period_size);
+ snd_printdd
+ ("interrupt 0x%x after 0x%lx of 0x%lx frames in period\n",
+ READ_AUDIO_STATUS(cif->hwport),
+ bytes_to_frames(runtime, pos),
+ runtime->period_size);
+ j = 0;
+ if (pos >= period_bytes) {
+ j++;
+ while (pos >= period_bytes)
+ pos -= period_bytes;
+ }
+ data->oldpos = pos;
+ if (j > 0)
+ snd_pcm_period_elapsed(substream[i]);
+ }
+ }
+ }
+}
+
+#ifdef CONFIG_PM
+static int riptide_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_riptide *chip = card->private_data;
+
+ chip->in_suspend = 1;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_ac97_suspend(chip->ac97);
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int riptide_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_riptide *chip = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "riptide: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+ snd_riptide_initialize(chip);
+ snd_ac97_resume(chip->ac97);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ chip->in_suspend = 0;
+ return 0;
+}
+#endif
+
+static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip)
+{
+ int timeout, tries;
+ union cmdret rptr = CMDRET_ZERO;
+ union firmware_version firmware;
+ int i, j, err, has_firmware;
+
+ if (!cif)
+ return -EINVAL;
+
+ cif->cmdcnt = 0;
+ cif->cmdtime = 0;
+ cif->cmdtimemax = 0;
+ cif->cmdtimemin = 0xffffffff;
+ cif->errcnt = 0;
+ cif->is_reset = 0;
+
+ tries = RESET_TRIES;
+ has_firmware = 0;
+ while (has_firmware == 0 && tries-- > 0) {
+ for (i = 0; i < 2; i++) {
+ WRITE_PORT_ULONG(cif->hwport->port[i].data1, 0);
+ WRITE_PORT_ULONG(cif->hwport->port[i].data2, 0);
+ }
+ SET_GRESET(cif->hwport);
+ udelay(100);
+ UNSET_GRESET(cif->hwport);
+ udelay(100);
+
+ for (timeout = 100000; --timeout; udelay(10)) {
+ if (IS_READY(cif->hwport) && !IS_GERR(cif->hwport))
+ break;
+ }
+ if (timeout == 0) {
+ snd_printk(KERN_ERR
+ "Riptide: device not ready, audio status: 0x%x ready: %d gerr: %d\n",
+ READ_AUDIO_STATUS(cif->hwport),
+ IS_READY(cif->hwport), IS_GERR(cif->hwport));
+ return -EIO;
+ } else {
+ snd_printdd
+ ("Riptide: audio status: 0x%x ready: %d gerr: %d\n",
+ READ_AUDIO_STATUS(cif->hwport),
+ IS_READY(cif->hwport), IS_GERR(cif->hwport));
+ }
+
+ SEND_GETV(cif, &rptr);
+ for (i = 0; i < 4; i++)
+ firmware.ret.retwords[i] = rptr.retwords[i];
+
+ snd_printdd
+ ("Firmware version: ASIC: %d CODEC %d AUXDSP %d PROG %d\n",
+ firmware.firmware.ASIC, firmware.firmware.CODEC,
+ firmware.firmware.AUXDSP, firmware.firmware.PROG);
+
+ for (j = 0; j < FIRMWARE_VERSIONS; j++) {
+ has_firmware = 1;
+ for (i = 0; i < 4; i++) {
+ if (firmware_versions[j].ret.retwords[i] !=
+ firmware.ret.retwords[i])
+ has_firmware = 0;
+ }
+ if (has_firmware)
+ break;
+ }
+
+ if (chip != NULL && has_firmware == 0) {
+ snd_printdd("Writing Firmware\n");
+ if (!chip->fw_entry) {
+ if ((err =
+ request_firmware(&chip->fw_entry,
+ "riptide.hex",
+ &chip->pci->dev)) != 0) {
+ snd_printk(KERN_ERR
+ "Riptide: Firmware not available %d\n",
+ err);
+ return -EIO;
+ }
+ }
+ err = loadfirmware(cif, chip->fw_entry->data,
+ chip->fw_entry->size);
+ if (err)
+ snd_printk(KERN_ERR
+ "Riptide: Could not load firmware %d\n",
+ err);
+ }
+ }
+
+ SEND_SACR(cif, 0, AC97_RESET);
+ SEND_RACR(cif, AC97_RESET, &rptr);
+ snd_printdd("AC97: 0x%x 0x%x\n", rptr.retlongs[0], rptr.retlongs[1]);
+
+ SEND_PLST(cif, 0);
+ SEND_SLST(cif, 0);
+ SEND_DLST(cif, 0);
+ SEND_ALST(cif, 0);
+ SEND_KDMA(cif);
+
+ writearm(cif, 0x301F8, 1, 1);
+ writearm(cif, 0x301F4, 1, 1);
+
+ SEND_LSEL(cif, MODEM_CMD, 0, 0, MODEM_INTDEC, MODEM_MERGER,
+ MODEM_SPLITTER, MODEM_MIXER);
+ setmixer(cif, MODEM_MIXER, 0x7fff, 0x7fff);
+ alloclbuspath(cif, ARM2LBUS_FIFO13, lbus_play_modem, NULL, NULL);
+
+ SEND_LSEL(cif, FM_CMD, 0, 0, FM_INTDEC, FM_MERGER, FM_SPLITTER,
+ FM_MIXER);
+ setmixer(cif, FM_MIXER, 0x7fff, 0x7fff);
+ writearm(cif, 0x30648 + FM_MIXER * 4, 0x01, 0x00000005);
+ writearm(cif, 0x301A8, 0x02, 0x00000002);
+ writearm(cif, 0x30264, 0x08, 0xffffffff);
+ alloclbuspath(cif, OPL3_SAMPLE, lbus_play_opl3, NULL, NULL);
+
+ SEND_SSRC(cif, I2S_INTDEC, 48000,
+ ((u32) I2S_RATE * 65536) / 48000,
+ ((u32) I2S_RATE * 65536) % 48000);
+ SEND_LSEL(cif, I2S_CMD0, 0, 0, I2S_INTDEC, I2S_MERGER, I2S_SPLITTER,
+ I2S_MIXER);
+ SEND_SI2S(cif, 1);
+ alloclbuspath(cif, ARM2LBUS_FIFO0, lbus_play_i2s, NULL, NULL);
+ alloclbuspath(cif, DIGITAL_MIXER_OUT0, lbus_play_out, NULL, NULL);
+ alloclbuspath(cif, DIGITAL_MIXER_OUT0, lbus_play_outhp, NULL, NULL);
+
+ SET_AIACK(cif->hwport);
+ SET_AIE(cif->hwport);
+ SET_AIACK(cif->hwport);
+ cif->is_reset = 1;
+ if (chip) {
+ for (i = 0; i < 4; i++)
+ chip->firmware.ret.retwords[i] =
+ firmware.ret.retwords[i];
+ }
+
+ return 0;
+}
+
+static struct snd_pcm_hardware snd_riptide_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats =
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8
+ | SNDRV_PCM_FMTBIT_U16_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5500,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (64 * 1024),
+ .period_bytes_min = PAGE_SIZE >> 1,
+ .period_bytes_max = PAGE_SIZE << 8,
+ .periods_min = 2,
+ .periods_max = 64,
+ .fifo_size = 0,
+};
+static struct snd_pcm_hardware snd_riptide_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats =
+ SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8
+ | SNDRV_PCM_FMTBIT_U16_LE,
+ .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5500,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (64 * 1024),
+ .period_bytes_min = PAGE_SIZE >> 1,
+ .period_bytes_max = PAGE_SIZE << 3,
+ .periods_min = 2,
+ .periods_max = 64,
+ .fifo_size = 0,
+};
+
+static snd_pcm_uframes_t snd_riptide_pointer(struct snd_pcm_substream
+ *substream)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcmhw *data = get_pcmhwdev(substream);
+ struct cmdif *cif = chip->cif;
+ union cmdret rptr = CMDRET_ZERO;
+ snd_pcm_uframes_t ret;
+
+ SEND_GPOS(cif, 0, data->id, &rptr);
+ if (data->size && runtime->period_size) {
+ snd_printdd
+ ("pointer stream %d position 0x%x(0x%x in buffer) bytes 0x%lx(0x%lx in period) frames\n",
+ data->id, rptr.retlongs[1], rptr.retlongs[1] % data->size,
+ bytes_to_frames(runtime, rptr.retlongs[1]),
+ bytes_to_frames(runtime,
+ rptr.retlongs[1]) % runtime->period_size);
+ if (rptr.retlongs[1] > data->pointer)
+ ret =
+ bytes_to_frames(runtime,
+ rptr.retlongs[1] % data->size);
+ else
+ ret =
+ bytes_to_frames(runtime,
+ data->pointer % data->size);
+ } else {
+ snd_printdd("stream not started or strange parms (%d %ld)\n",
+ data->size, runtime->period_size);
+ ret = bytes_to_frames(runtime, 0);
+ }
+ return ret;
+}
+
+static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int i, j;
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct pcmhw *data = get_pcmhwdev(substream);
+ struct cmdif *cif = chip->cif;
+ union cmdret rptr = CMDRET_ZERO;
+
+ spin_lock(&chip->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (!(data->state & ST_PLAY)) {
+ SEND_SSTR(cif, data->id, data->sgdlist.addr);
+ SET_AIE(cif->hwport);
+ data->state = ST_PLAY;
+ if (data->mixer != 0xff)
+ setmixer(cif, data->mixer, 0x7fff, 0x7fff);
+ chip->openstreams++;
+ data->oldpos = 0;
+ data->pointer = 0;
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (data->mixer != 0xff)
+ setmixer(cif, data->mixer, 0, 0);
+ setmixer(cif, data->mixer, 0, 0);
+ SEND_KSTR(cif, data->id);
+ data->state = ST_STOP;
+ chip->openstreams--;
+ j = 0;
+ do {
+ i = rptr.retlongs[1];
+ SEND_GPOS(cif, 0, data->id, &rptr);
+ udelay(1);
+ } while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY);
+ if (j >= MAX_WRITE_RETRY)
+ snd_printk(KERN_ERR "Riptide: Could not stop stream!");
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (!(data->state & ST_PAUSE)) {
+ SEND_PSTR(cif, data->id);
+ data->state |= ST_PAUSE;
+ chip->openstreams--;
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (data->state & ST_PAUSE) {
+ SEND_SSTR(cif, data->id, data->sgdlist.addr);
+ data->state &= ~ST_PAUSE;
+ chip->openstreams++;
+ }
+ break;
+ default:
+ spin_unlock(&chip->lock);
+ return -EINVAL;
+ }
+ spin_unlock(&chip->lock);
+ return 0;
+}
+
+static int snd_riptide_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcmhw *data = get_pcmhwdev(substream);
+ struct cmdif *cif = chip->cif;
+ unsigned char *lbuspath = NULL;
+ unsigned int rate, channels;
+ int err = 0;
+ snd_pcm_format_t format;
+
+ if (snd_BUG_ON(!cif || !data))
+ return -EINVAL;
+
+ snd_printdd("prepare id %d ch: %d f:0x%x r:%d\n", data->id,
+ runtime->channels, runtime->format, runtime->rate);
+
+ spin_lock_irq(&chip->lock);
+ channels = runtime->channels;
+ format = runtime->format;
+ rate = runtime->rate;
+ switch (channels) {
+ case 1:
+ if (rate == 48000 && format == SNDRV_PCM_FORMAT_S16_LE)
+ lbuspath = data->paths.noconv;
+ else
+ lbuspath = data->paths.mono;
+ break;
+ case 2:
+ if (rate == 48000 && format == SNDRV_PCM_FORMAT_S16_LE)
+ lbuspath = data->paths.noconv;
+ else
+ lbuspath = data->paths.stereo;
+ break;
+ }
+ snd_printdd("use sgdlist at 0x%p\n",
+ data->sgdlist.area);
+ if (data->sgdlist.area) {
+ unsigned int i, j, size, pages, f, pt, period;
+ struct sgd *c, *p = NULL;
+
+ size = frames_to_bytes(runtime, runtime->buffer_size);
+ period = frames_to_bytes(runtime, runtime->period_size);
+ f = PAGE_SIZE;
+ while ((size + (f >> 1) - 1) <= (f << 7) && (f << 1) > period)
+ f = f >> 1;
+ pages = (size + f - 1) / f;
+ data->size = size;
+ data->pages = pages;
+ snd_printdd
+ ("create sgd size: 0x%x pages %d of size 0x%x for period 0x%x\n",
+ size, pages, f, period);
+ pt = 0;
+ j = 0;
+ for (i = 0; i < pages; i++) {
+ unsigned int ofs, addr;
+ c = &data->sgdbuf[i];
+ if (p)
+ p->dwNextLink = cpu_to_le32(data->sgdlist.addr +
+ (i *
+ sizeof(struct
+ sgd)));
+ c->dwNextLink = cpu_to_le32(data->sgdlist.addr);
+ ofs = j << PAGE_SHIFT;
+ addr = snd_pcm_sgbuf_get_addr(substream, ofs) + pt;
+ c->dwSegPtrPhys = cpu_to_le32(addr);
+ pt = (pt + f) % PAGE_SIZE;
+ if (pt == 0)
+ j++;
+ c->dwSegLen = cpu_to_le32(f);
+ c->dwStat_Ctl =
+ cpu_to_le32(IEOB_ENABLE | IEOS_ENABLE |
+ IEOC_ENABLE);
+ p = c;
+ size -= f;
+ }
+ data->sgdbuf[i].dwSegLen = cpu_to_le32(size);
+ }
+ if (lbuspath && lbuspath != data->lbuspath) {
+ if (data->lbuspath)
+ freelbuspath(cif, data->source, data->lbuspath);
+ alloclbuspath(cif, data->source, lbuspath,
+ &data->mixer, data->intdec);
+ data->lbuspath = lbuspath;
+ data->rate = 0;
+ }
+ if (data->rate != rate || data->format != format ||
+ data->channels != channels) {
+ data->rate = rate;
+ data->format = format;
+ data->channels = channels;
+ if (setsampleformat
+ (cif, data->mixer, data->id, channels, format)
+ || setsamplerate(cif, data->intdec, rate))
+ err = -EIO;
+ }
+ spin_unlock_irq(&chip->lock);
+ return err;
+}
+
+static int
+snd_riptide_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct pcmhw *data = get_pcmhwdev(substream);
+ struct snd_dma_buffer *sgdlist = &data->sgdlist;
+ int err;
+
+ snd_printdd("hw params id %d (sgdlist: 0x%p 0x%lx %d)\n", data->id,
+ sgdlist->area, (unsigned long)sgdlist->addr,
+ (int)sgdlist->bytes);
+ if (sgdlist->area)
+ snd_dma_free_pages(sgdlist);
+ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ sizeof(struct sgd) * (DESC_MAX_MASK + 1),
+ sgdlist)) < 0) {
+ snd_printk(KERN_ERR "Riptide: failed to alloc %d dma bytes\n",
+ (int)sizeof(struct sgd) * (DESC_MAX_MASK + 1));
+ return err;
+ }
+ data->sgdbuf = (struct sgd *)sgdlist->area;
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int snd_riptide_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct pcmhw *data = get_pcmhwdev(substream);
+ struct cmdif *cif = chip->cif;
+
+ if (cif && data) {
+ if (data->lbuspath)
+ freelbuspath(cif, data->source, data->lbuspath);
+ data->lbuspath = NULL;
+ data->source = 0xff;
+ data->intdec[0] = 0xff;
+ data->intdec[1] = 0xff;
+
+ if (data->sgdlist.area) {
+ snd_dma_free_pages(&data->sgdlist);
+ data->sgdlist.area = NULL;
+ }
+ }
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_riptide_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcmhw *data;
+ int sub_num = substream->number;
+
+ chip->playback_substream[sub_num] = substream;
+ runtime->hw = snd_riptide_playback;
+ data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL);
+ data->paths = lbus_play_paths[sub_num];
+ data->id = play_ids[sub_num];
+ data->source = play_sources[sub_num];
+ data->intdec[0] = 0xff;
+ data->intdec[1] = 0xff;
+ data->state = ST_STOP;
+ runtime->private_data = data;
+ return snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+}
+
+static int snd_riptide_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcmhw *data;
+
+ chip->capture_substream = substream;
+ runtime->hw = snd_riptide_capture;
+ data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL);
+ data->paths = lbus_rec_path;
+ data->id = PADC;
+ data->source = ACLNK2PADC;
+ data->intdec[0] = 0xff;
+ data->intdec[1] = 0xff;
+ data->state = ST_STOP;
+ runtime->private_data = data;
+ return snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+}
+
+static int snd_riptide_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct pcmhw *data = get_pcmhwdev(substream);
+ int sub_num = substream->number;
+
+ substream->runtime->private_data = NULL;
+ chip->playback_substream[sub_num] = NULL;
+ kfree(data);
+ return 0;
+}
+
+static int snd_riptide_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_riptide *chip = snd_pcm_substream_chip(substream);
+ struct pcmhw *data = get_pcmhwdev(substream);
+
+ substream->runtime->private_data = NULL;
+ chip->capture_substream = NULL;
+ kfree(data);
+ return 0;
+}
+
+static struct snd_pcm_ops snd_riptide_playback_ops = {
+ .open = snd_riptide_playback_open,
+ .close = snd_riptide_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_riptide_hw_params,
+ .hw_free = snd_riptide_hw_free,
+ .prepare = snd_riptide_prepare,
+ .page = snd_pcm_sgbuf_ops_page,
+ .trigger = snd_riptide_trigger,
+ .pointer = snd_riptide_pointer,
+};
+static struct snd_pcm_ops snd_riptide_capture_ops = {
+ .open = snd_riptide_capture_open,
+ .close = snd_riptide_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_riptide_hw_params,
+ .hw_free = snd_riptide_hw_free,
+ .prepare = snd_riptide_prepare,
+ .page = snd_pcm_sgbuf_ops_page,
+ .trigger = snd_riptide_trigger,
+ .pointer = snd_riptide_pointer,
+};
+
+static int __devinit
+snd_riptide_pcm(struct snd_riptide *chip, int device, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err =
+ snd_pcm_new(chip->card, "RIPTIDE", device, PLAYBACK_SUBSTREAMS, 1,
+ &pcm)) < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_riptide_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_riptide_capture_ops);
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "RIPTIDE");
+ chip->pcm = pcm;
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 64 * 1024, 128 * 1024);
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static irqreturn_t
+snd_riptide_interrupt(int irq, void *dev_id)
+{
+ struct snd_riptide *chip = dev_id;
+ struct cmdif *cif = chip->cif;
+
+ if (cif) {
+ chip->received_irqs++;
+ if (IS_EOBIRQ(cif->hwport) || IS_EOSIRQ(cif->hwport) ||
+ IS_EOCIRQ(cif->hwport)) {
+ chip->handled_irqs++;
+ tasklet_hi_schedule(&chip->riptide_tq);
+ }
+ if (chip->rmidi && IS_MPUIRQ(cif->hwport)) {
+ chip->handled_irqs++;
+ snd_mpu401_uart_interrupt(irq,
+ chip->rmidi->private_data);
+ }
+ SET_AIACK(cif->hwport);
+ }
+ return IRQ_HANDLED;
+}
+
+static void
+snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct snd_riptide *chip = ac97->private_data;
+ struct cmdif *cif = chip->cif;
+ union cmdret rptr = CMDRET_ZERO;
+ int i = 0;
+
+ if (snd_BUG_ON(!cif))
+ return;
+
+ snd_printdd("Write AC97 reg 0x%x 0x%x\n", reg, val);
+ do {
+ SEND_SACR(cif, val, reg);
+ SEND_RACR(cif, reg, &rptr);
+ } while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY);
+ if (i == MAX_WRITE_RETRY)
+ snd_printdd("Write AC97 reg failed\n");
+}
+
+static unsigned short snd_riptide_codec_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct snd_riptide *chip = ac97->private_data;
+ struct cmdif *cif = chip->cif;
+ union cmdret rptr = CMDRET_ZERO;
+
+ if (snd_BUG_ON(!cif))
+ return 0;
+
+ if (SEND_RACR(cif, reg, &rptr) != 0)
+ SEND_RACR(cif, reg, &rptr);
+ snd_printdd("Read AC97 reg 0x%x got 0x%x\n", reg, rptr.retwords[1]);
+ return rptr.retwords[1];
+}
+
+static int snd_riptide_initialize(struct snd_riptide *chip)
+{
+ struct cmdif *cif;
+ unsigned int device_id;
+ int err;
+
+ if (snd_BUG_ON(!chip))
+ return -EINVAL;
+
+ cif = chip->cif;
+ if (!cif) {
+ if ((cif = kzalloc(sizeof(struct cmdif), GFP_KERNEL)) == NULL)
+ return -ENOMEM;
+ cif->hwport = (struct riptideport *)chip->port;
+ spin_lock_init(&cif->lock);
+ chip->cif = cif;
+ }
+ cif->is_reset = 0;
+ if ((err = riptide_reset(cif, chip)) != 0)
+ return err;
+ device_id = chip->device_id;
+ switch (device_id) {
+ case 0x4310:
+ case 0x4320:
+ case 0x4330:
+ snd_printdd("Modem enable?\n");
+ SEND_SETDPLL(cif);
+ break;
+ }
+ snd_printdd("Enabling MPU IRQs\n");
+ if (chip->rmidi)
+ SET_EMPUIRQ(cif->hwport);
+ return err;
+}
+
+static int snd_riptide_free(struct snd_riptide *chip)
+{
+ struct cmdif *cif;
+
+ if (!chip)
+ return 0;
+
+ if ((cif = chip->cif)) {
+ SET_GRESET(cif->hwport);
+ udelay(100);
+ UNSET_GRESET(cif->hwport);
+ kfree(chip->cif);
+ }
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ if (chip->fw_entry)
+ release_firmware(chip->fw_entry);
+ release_and_free_resource(chip->res_port);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_riptide_dev_free(struct snd_device *device)
+{
+ struct snd_riptide *chip = device->device_data;
+
+ return snd_riptide_free(chip);
+}
+
+static int __devinit
+snd_riptide_create(struct snd_card *card, struct pci_dev *pci,
+ struct snd_riptide **rchip)
+{
+ struct snd_riptide *chip;
+ struct riptideport *hwport;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_riptide_dev_free,
+ };
+
+ *rchip = NULL;
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ if (!(chip = kzalloc(sizeof(struct snd_riptide), GFP_KERNEL)))
+ return -ENOMEM;
+
+ spin_lock_init(&chip->lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->openstreams = 0;
+ chip->port = pci_resource_start(pci, 0);
+ chip->received_irqs = 0;
+ chip->handled_irqs = 0;
+ chip->cif = NULL;
+ tasklet_init(&chip->riptide_tq, riptide_handleirq, (unsigned long)chip);
+
+ if ((chip->res_port =
+ request_region(chip->port, 64, "RIPTIDE")) == NULL) {
+ snd_printk(KERN_ERR
+ "Riptide: unable to grab region 0x%lx-0x%lx\n",
+ chip->port, chip->port + 64 - 1);
+ snd_riptide_free(chip);
+ return -EBUSY;
+ }
+ hwport = (struct riptideport *)chip->port;
+ UNSET_AIE(hwport);
+
+ if (request_irq(pci->irq, snd_riptide_interrupt, IRQF_SHARED,
+ "RIPTIDE", chip)) {
+ snd_printk(KERN_ERR "Riptide: unable to grab IRQ %d\n",
+ pci->irq);
+ snd_riptide_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ chip->device_id = pci->device;
+ pci_set_master(pci);
+ if ((err = snd_riptide_initialize(chip)) < 0) {
+ snd_riptide_free(chip);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_riptide_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+}
+
+static void
+snd_riptide_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_riptide *chip = entry->private_data;
+ struct pcmhw *data;
+ int i;
+ struct cmdif *cif = NULL;
+ unsigned char p[256];
+ unsigned short rval = 0, lval = 0;
+ unsigned int rate;
+
+ if (!chip)
+ return;
+
+ snd_iprintf(buffer, "%s\n\n", chip->card->longname);
+ snd_iprintf(buffer, "Device ID: 0x%x\nReceived IRQs: (%ld)%ld\nPorts:",
+ chip->device_id, chip->handled_irqs, chip->received_irqs);
+ for (i = 0; i < 64; i += 4)
+ snd_iprintf(buffer, "%c%02x: %08x",
+ (i % 16) ? ' ' : '\n', i, inl(chip->port + i));
+ if ((cif = chip->cif)) {
+ snd_iprintf(buffer,
+ "\nVersion: ASIC: %d CODEC: %d AUXDSP: %d PROG: %d",
+ chip->firmware.firmware.ASIC,
+ chip->firmware.firmware.CODEC,
+ chip->firmware.firmware.AUXDSP,
+ chip->firmware.firmware.PROG);
+ snd_iprintf(buffer, "\nDigital mixer:");
+ for (i = 0; i < 12; i++) {
+ getmixer(cif, i, &rval, &lval);
+ snd_iprintf(buffer, "\n %d: %d %d", i, rval, lval);
+ }
+ snd_iprintf(buffer,
+ "\nARM Commands num: %d failed: %d time: %d max: %d min: %d",
+ cif->cmdcnt, cif->errcnt,
+ cif->cmdtime, cif->cmdtimemax, cif->cmdtimemin);
+ }
+ snd_iprintf(buffer, "\nOpen streams %d:\n", chip->openstreams);
+ for (i = 0; i < PLAYBACK_SUBSTREAMS; i++) {
+ if (chip->playback_substream[i]
+ && chip->playback_substream[i]->runtime
+ && (data =
+ chip->playback_substream[i]->runtime->private_data)) {
+ snd_iprintf(buffer,
+ "stream: %d mixer: %d source: %d (%d,%d)\n",
+ data->id, data->mixer, data->source,
+ data->intdec[0], data->intdec[1]);
+ if (!(getsamplerate(cif, data->intdec, &rate)))
+ snd_iprintf(buffer, "rate: %d\n", rate);
+ }
+ }
+ if (chip->capture_substream
+ && chip->capture_substream->runtime
+ && (data = chip->capture_substream->runtime->private_data)) {
+ snd_iprintf(buffer,
+ "stream: %d mixer: %d source: %d (%d,%d)\n",
+ data->id, data->mixer,
+ data->source, data->intdec[0], data->intdec[1]);
+ if (!(getsamplerate(cif, data->intdec, &rate)))
+ snd_iprintf(buffer, "rate: %d\n", rate);
+ }
+ snd_iprintf(buffer, "Paths:\n");
+ i = getpaths(cif, p);
+ while (i--) {
+ snd_iprintf(buffer, "%x->%x ", p[i - 1], p[i]);
+ i--;
+ }
+ snd_iprintf(buffer, "\n");
+}
+
+static void __devinit snd_riptide_proc_init(struct snd_riptide *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(chip->card, "riptide", &entry))
+ snd_info_set_text_ops(entry, chip, snd_riptide_proc_read);
+}
+
+static int __devinit snd_riptide_mixer(struct snd_riptide *chip)
+{
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+ int err = 0;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_riptide_codec_write,
+ .read = snd_riptide_codec_read,
+ };
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.scaps = AC97_SCAP_SKIP_MODEM;
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &pbus)) < 0)
+ return err;
+
+ chip->ac97_bus = pbus;
+ ac97.pci = chip->pci;
+ if ((err = snd_ac97_mixer(pbus, &ac97, &chip->ac97)) < 0)
+ return err;
+ return err;
+}
+
+#ifdef SUPPORT_JOYSTICK
+static int have_joystick;
+static struct pci_dev *riptide_gameport_pci;
+static struct gameport *riptide_gameport;
+
+static int __devinit
+snd_riptide_joystick_probe(struct pci_dev *pci, const struct pci_device_id *id)
+{
+ static int dev;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ if (joystick_port[dev]) {
+ riptide_gameport = gameport_allocate_port();
+ if (riptide_gameport) {
+ if (!request_region
+ (joystick_port[dev], 8, "Riptide gameport")) {
+ snd_printk(KERN_WARNING
+ "Riptide: cannot grab gameport 0x%x\n",
+ joystick_port[dev]);
+ gameport_free_port(riptide_gameport);
+ riptide_gameport = NULL;
+ } else {
+ riptide_gameport_pci = pci;
+ riptide_gameport->io = joystick_port[dev];
+ gameport_register_port(riptide_gameport);
+ }
+ }
+ }
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_riptide_joystick_remove(struct pci_dev *pci)
+{
+ if (riptide_gameport) {
+ if (riptide_gameport_pci == pci) {
+ release_region(riptide_gameport->io, 8);
+ riptide_gameport_pci = NULL;
+ gameport_unregister_port(riptide_gameport);
+ riptide_gameport = NULL;
+ }
+ }
+}
+#endif
+
+static int __devinit
+snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_riptide *chip;
+ unsigned short addr;
+ int err = 0;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ if ((err = snd_riptide_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = chip;
+ if ((err = snd_riptide_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_riptide_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_write_config_word(chip->pci, PCI_EXT_Legacy_Mask, LEGACY_ENABLE_ALL
+ | (opl3_port[dev] ? LEGACY_ENABLE_FM : 0)
+#ifdef SUPPORT_JOYSTICK
+ | (joystick_port[dev] ? LEGACY_ENABLE_GAMEPORT :
+ 0)
+#endif
+ | (mpu_port[dev]
+ ? (LEGACY_ENABLE_MPU_INT | LEGACY_ENABLE_MPU) :
+ 0)
+ | ((chip->irq << 4) & 0xF0));
+ if ((addr = mpu_port[dev]) != 0) {
+ pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, addr);
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE,
+ addr, 0, chip->irq, 0,
+ &chip->rmidi)) < 0)
+ snd_printk(KERN_WARNING
+ "Riptide: Can't Allocate MPU at 0x%x\n",
+ addr);
+ else
+ chip->mpuaddr = addr;
+ }
+ if ((addr = opl3_port[dev]) != 0) {
+ pci_write_config_word(chip->pci, PCI_EXT_FM_Base, addr);
+ if ((err = snd_opl3_create(card, addr, addr + 2,
+ OPL3_HW_RIPTIDE, 0,
+ &chip->opl3)) < 0)
+ snd_printk(KERN_WARNING
+ "Riptide: Can't Allocate OPL3 at 0x%x\n",
+ addr);
+ else {
+ chip->opladdr = addr;
+ if ((err =
+ snd_opl3_hwdep_new(chip->opl3, 0, 1, NULL)) < 0)
+ snd_printk(KERN_WARNING
+ "Riptide: Can't Allocate OPL3-HWDEP\n");
+ }
+ }
+#ifdef SUPPORT_JOYSTICK
+ if ((addr = joystick_port[dev]) != 0) {
+ pci_write_config_word(chip->pci, PCI_EXT_Game_Base, addr);
+ chip->gameaddr = addr;
+ }
+#endif
+
+ strcpy(card->driver, "RIPTIDE");
+ strcpy(card->shortname, "Riptide");
+#ifdef SUPPORT_JOYSTICK
+ snprintf(card->longname, sizeof(card->longname),
+ "%s at 0x%lx, irq %i mpu 0x%x opl3 0x%x gameport 0x%x",
+ card->shortname, chip->port, chip->irq, chip->mpuaddr,
+ chip->opladdr, chip->gameaddr);
+#else
+ snprintf(card->longname, sizeof(card->longname),
+ "%s at 0x%lx, irq %i mpu 0x%x opl3 0x%x",
+ card->shortname, chip->port, chip->irq, chip->mpuaddr,
+ chip->opladdr);
+#endif
+ snd_riptide_proc_init(chip);
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_card_riptide_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "RIPTIDE",
+ .id_table = snd_riptide_ids,
+ .probe = snd_card_riptide_probe,
+ .remove = __devexit_p(snd_card_riptide_remove),
+#ifdef CONFIG_PM
+ .suspend = riptide_suspend,
+ .resume = riptide_resume,
+#endif
+};
+
+#ifdef SUPPORT_JOYSTICK
+static struct pci_driver joystick_driver = {
+ .name = "Riptide Joystick",
+ .id_table = snd_riptide_joystick_ids,
+ .probe = snd_riptide_joystick_probe,
+ .remove = __devexit_p(snd_riptide_joystick_remove),
+};
+#endif
+
+static int __init alsa_card_riptide_init(void)
+{
+ int err;
+ if ((err = pci_register_driver(&driver)) < 0)
+ return err;
+#if defined(SUPPORT_JOYSTICK)
+ if (pci_register_driver(&joystick_driver) < 0) {
+ have_joystick = 0;
+ snd_printk(KERN_INFO "no joystick found\n");
+ } else
+ have_joystick = 1;
+#endif
+ return 0;
+}
+
+static void __exit alsa_card_riptide_exit(void)
+{
+ pci_unregister_driver(&driver);
+#if defined(SUPPORT_JOYSTICK)
+ if (have_joystick)
+ pci_unregister_driver(&joystick_driver);
+#endif
+}
+
+module_init(alsa_card_riptide_init);
+module_exit(alsa_card_riptide_exit);
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
new file mode 100644
index 0000000..e7ef3a1
--- /dev/null
+++ b/sound/pci/rme32.c
@@ -0,0 +1,2007 @@
+/*
+ * ALSA driver for RME Digi32, Digi32/8 and Digi32 PRO audio interfaces
+ *
+ * Copyright (c) 2002-2004 Martin Langer <martin-langer@gmx.de>,
+ * Pilo Chambert <pilo.c@wanadoo.fr>
+ *
+ * Thanks to : Anders Torger <torger@ludd.luth.se>,
+ * Henk Hesselink <henk@anda.nl>
+ * for writing the digi96-driver
+ * and RME for all informations.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ *
+ * ****************************************************************************
+ *
+ * Note #1 "Sek'd models" ................................... martin 2002-12-07
+ *
+ * Identical soundcards by Sek'd were labeled:
+ * RME Digi 32 = Sek'd Prodif 32
+ * RME Digi 32 Pro = Sek'd Prodif 96
+ * RME Digi 32/8 = Sek'd Prodif Gold
+ *
+ * ****************************************************************************
+ *
+ * Note #2 "full duplex mode" ............................... martin 2002-12-07
+ *
+ * Full duplex doesn't work. All cards (32, 32/8, 32Pro) are working identical
+ * in this mode. Rec data and play data are using the same buffer therefore. At
+ * first you have got the playing bits in the buffer and then (after playing
+ * them) they were overwitten by the captured sound of the CS8412/14. Both
+ * modes (play/record) are running harmonically hand in hand in the same buffer
+ * and you have only one start bit plus one interrupt bit to control this
+ * paired action.
+ * This is opposite to the latter rme96 where playing and capturing is totally
+ * separated and so their full duplex mode is supported by alsa (using two
+ * start bits and two interrupts for two different buffers).
+ * But due to the wrong sequence of playing and capturing ALSA shows no solved
+ * full duplex support for the rme32 at the moment. That's bad, but I'm not
+ * able to solve it. Are you motivated enough to solve this problem now? Your
+ * patch would be welcome!
+ *
+ * ****************************************************************************
+ *
+ * "The story after the long seeking" -- tiwai
+ *
+ * Ok, the situation regarding the full duplex is now improved a bit.
+ * In the fullduplex mode (given by the module parameter), the hardware buffer
+ * is split to halves for read and write directions at the DMA pointer.
+ * That is, the half above the current DMA pointer is used for write, and
+ * the half below is used for read. To mangle this strange behavior, an
+ * software intermediate buffer is introduced. This is, of course, not good
+ * from the viewpoint of the data transfer efficiency. However, this allows
+ * you to use arbitrary buffer sizes, instead of the fixed I/O buffer size.
+ *
+ * ****************************************************************************
+ */
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm-indirect.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int fullduplex[SNDRV_CARDS]; // = {[0 ... (SNDRV_CARDS - 1)] = 1};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for RME Digi32 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for RME Digi32 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable RME Digi32 soundcard.");
+module_param_array(fullduplex, bool, NULL, 0444);
+MODULE_PARM_DESC(fullduplex, "Support full-duplex mode.");
+MODULE_AUTHOR("Martin Langer <martin-langer@gmx.de>, Pilo Chambert <pilo.c@wanadoo.fr>");
+MODULE_DESCRIPTION("RME Digi32, Digi32/8, Digi32 PRO");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}");
+
+/* Defines for RME Digi32 series */
+#define RME32_SPDIF_NCHANNELS 2
+
+/* Playback and capture buffer size */
+#define RME32_BUFFER_SIZE 0x20000
+
+/* IO area size */
+#define RME32_IO_SIZE 0x30000
+
+/* IO area offsets */
+#define RME32_IO_DATA_BUFFER 0x0
+#define RME32_IO_CONTROL_REGISTER 0x20000
+#define RME32_IO_GET_POS 0x20000
+#define RME32_IO_CONFIRM_ACTION_IRQ 0x20004
+#define RME32_IO_RESET_POS 0x20100
+
+/* Write control register bits */
+#define RME32_WCR_START (1 << 0) /* startbit */
+#define RME32_WCR_MONO (1 << 1) /* 0=stereo, 1=mono
+ Setting the whole card to mono
+ doesn't seem to be very useful.
+ A software-solution can handle
+ full-duplex with one direction in
+ stereo and the other way in mono.
+ So, the hardware should work all
+ the time in stereo! */
+#define RME32_WCR_MODE24 (1 << 2) /* 0=16bit, 1=32bit */
+#define RME32_WCR_SEL (1 << 3) /* 0=input on output, 1=normal playback/capture */
+#define RME32_WCR_FREQ_0 (1 << 4) /* frequency (play) */
+#define RME32_WCR_FREQ_1 (1 << 5)
+#define RME32_WCR_INP_0 (1 << 6) /* input switch */
+#define RME32_WCR_INP_1 (1 << 7)
+#define RME32_WCR_RESET (1 << 8) /* Reset address */
+#define RME32_WCR_MUTE (1 << 9) /* digital mute for output */
+#define RME32_WCR_PRO (1 << 10) /* 1=professional, 0=consumer */
+#define RME32_WCR_DS_BM (1 << 11) /* 1=DoubleSpeed (only PRO-Version); 1=BlockMode (only Adat-Version) */
+#define RME32_WCR_ADAT (1 << 12) /* Adat Mode (only Adat-Version) */
+#define RME32_WCR_AUTOSYNC (1 << 13) /* AutoSync */
+#define RME32_WCR_PD (1 << 14) /* DAC Reset (only PRO-Version) */
+#define RME32_WCR_EMP (1 << 15) /* 1=Emphasis on (only PRO-Version) */
+
+#define RME32_WCR_BITPOS_FREQ_0 4
+#define RME32_WCR_BITPOS_FREQ_1 5
+#define RME32_WCR_BITPOS_INP_0 6
+#define RME32_WCR_BITPOS_INP_1 7
+
+/* Read control register bits */
+#define RME32_RCR_AUDIO_ADDR_MASK 0x1ffff
+#define RME32_RCR_LOCK (1 << 23) /* 1=locked, 0=not locked */
+#define RME32_RCR_ERF (1 << 26) /* 1=Error, 0=no Error */
+#define RME32_RCR_FREQ_0 (1 << 27) /* CS841x frequency (record) */
+#define RME32_RCR_FREQ_1 (1 << 28)
+#define RME32_RCR_FREQ_2 (1 << 29)
+#define RME32_RCR_KMODE (1 << 30) /* card mode: 1=PLL, 0=quartz */
+#define RME32_RCR_IRQ (1 << 31) /* interrupt */
+
+#define RME32_RCR_BITPOS_F0 27
+#define RME32_RCR_BITPOS_F1 28
+#define RME32_RCR_BITPOS_F2 29
+
+/* Input types */
+#define RME32_INPUT_OPTICAL 0
+#define RME32_INPUT_COAXIAL 1
+#define RME32_INPUT_INTERNAL 2
+#define RME32_INPUT_XLR 3
+
+/* Clock modes */
+#define RME32_CLOCKMODE_SLAVE 0
+#define RME32_CLOCKMODE_MASTER_32 1
+#define RME32_CLOCKMODE_MASTER_44 2
+#define RME32_CLOCKMODE_MASTER_48 3
+
+/* Block sizes in bytes */
+#define RME32_BLOCK_SIZE 8192
+
+/* Software intermediate buffer (max) size */
+#define RME32_MID_BUFFER_SIZE (1024*1024)
+
+/* Hardware revisions */
+#define RME32_32_REVISION 192
+#define RME32_328_REVISION_OLD 100
+#define RME32_328_REVISION_NEW 101
+#define RME32_PRO_REVISION_WITH_8412 192
+#define RME32_PRO_REVISION_WITH_8414 150
+
+
+struct rme32 {
+ spinlock_t lock;
+ int irq;
+ unsigned long port;
+ void __iomem *iobase;
+
+ u32 wcreg; /* cached write control register value */
+ u32 wcreg_spdif; /* S/PDIF setup */
+ u32 wcreg_spdif_stream; /* S/PDIF setup (temporary) */
+ u32 rcreg; /* cached read control register value */
+
+ u8 rev; /* card revision number */
+
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ int playback_frlog; /* log2 of framesize */
+ int capture_frlog;
+
+ size_t playback_periodsize; /* in bytes, zero if not used */
+ size_t capture_periodsize; /* in bytes, zero if not used */
+
+ unsigned int fullduplex_mode;
+ int running;
+
+ struct snd_pcm_indirect playback_pcm;
+ struct snd_pcm_indirect capture_pcm;
+
+ struct snd_card *card;
+ struct snd_pcm *spdif_pcm;
+ struct snd_pcm *adat_pcm;
+ struct pci_dev *pci;
+ struct snd_kcontrol *spdif_ctl;
+};
+
+static struct pci_device_id snd_rme32_ids[] = {
+ {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,},
+ {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,},
+ {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,},
+ {0,}
+};
+
+MODULE_DEVICE_TABLE(pci, snd_rme32_ids);
+
+#define RME32_ISWORKING(rme32) ((rme32)->wcreg & RME32_WCR_START)
+#define RME32_PRO_WITH_8414(rme32) ((rme32)->pci->device == PCI_DEVICE_ID_RME_DIGI32_PRO && (rme32)->rev == RME32_PRO_REVISION_WITH_8414)
+
+static int snd_rme32_playback_prepare(struct snd_pcm_substream *substream);
+
+static int snd_rme32_capture_prepare(struct snd_pcm_substream *substream);
+
+static int snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
+
+static void snd_rme32_proc_init(struct rme32 * rme32);
+
+static int snd_rme32_create_switches(struct snd_card *card, struct rme32 * rme32);
+
+static inline unsigned int snd_rme32_pcm_byteptr(struct rme32 * rme32)
+{
+ return (readl(rme32->iobase + RME32_IO_GET_POS)
+ & RME32_RCR_AUDIO_ADDR_MASK);
+}
+
+/* silence callback for halfduplex mode */
+static int snd_rme32_playback_silence(struct snd_pcm_substream *substream, int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ count <<= rme32->playback_frlog;
+ pos <<= rme32->playback_frlog;
+ memset_io(rme32->iobase + RME32_IO_DATA_BUFFER + pos, 0, count);
+ return 0;
+}
+
+/* copy callback for halfduplex mode */
+static int snd_rme32_playback_copy(struct snd_pcm_substream *substream, int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *src, snd_pcm_uframes_t count)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ count <<= rme32->playback_frlog;
+ pos <<= rme32->playback_frlog;
+ if (copy_from_user_toio(rme32->iobase + RME32_IO_DATA_BUFFER + pos,
+ src, count))
+ return -EFAULT;
+ return 0;
+}
+
+/* copy callback for halfduplex mode */
+static int snd_rme32_capture_copy(struct snd_pcm_substream *substream, int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *dst, snd_pcm_uframes_t count)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ count <<= rme32->capture_frlog;
+ pos <<= rme32->capture_frlog;
+ if (copy_to_user_fromio(dst,
+ rme32->iobase + RME32_IO_DATA_BUFFER + pos,
+ count))
+ return -EFAULT;
+ return 0;
+}
+
+/*
+ * SPDIF I/O capabilities (half-duplex mode)
+ */
+static struct snd_pcm_hardware snd_rme32_spdif_info = {
+ .info = (SNDRV_PCM_INFO_MMAP_IOMEM |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 32000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = RME32_BUFFER_SIZE,
+ .period_bytes_min = RME32_BLOCK_SIZE,
+ .period_bytes_max = RME32_BLOCK_SIZE,
+ .periods_min = RME32_BUFFER_SIZE / RME32_BLOCK_SIZE,
+ .periods_max = RME32_BUFFER_SIZE / RME32_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+/*
+ * ADAT I/O capabilities (half-duplex mode)
+ */
+static struct snd_pcm_hardware snd_rme32_adat_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP_IOMEM |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats= SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 8,
+ .channels_max = 8,
+ .buffer_bytes_max = RME32_BUFFER_SIZE,
+ .period_bytes_min = RME32_BLOCK_SIZE,
+ .period_bytes_max = RME32_BLOCK_SIZE,
+ .periods_min = RME32_BUFFER_SIZE / RME32_BLOCK_SIZE,
+ .periods_max = RME32_BUFFER_SIZE / RME32_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+/*
+ * SPDIF I/O capabilities (full-duplex mode)
+ */
+static struct snd_pcm_hardware snd_rme32_spdif_fd_info = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 32000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = RME32_MID_BUFFER_SIZE,
+ .period_bytes_min = RME32_BLOCK_SIZE,
+ .period_bytes_max = RME32_BLOCK_SIZE,
+ .periods_min = 2,
+ .periods_max = RME32_MID_BUFFER_SIZE / RME32_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+/*
+ * ADAT I/O capabilities (full-duplex mode)
+ */
+static struct snd_pcm_hardware snd_rme32_adat_fd_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats= SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 8,
+ .channels_max = 8,
+ .buffer_bytes_max = RME32_MID_BUFFER_SIZE,
+ .period_bytes_min = RME32_BLOCK_SIZE,
+ .period_bytes_max = RME32_BLOCK_SIZE,
+ .periods_min = 2,
+ .periods_max = RME32_MID_BUFFER_SIZE / RME32_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+static void snd_rme32_reset_dac(struct rme32 *rme32)
+{
+ writel(rme32->wcreg | RME32_WCR_PD,
+ rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+}
+
+static int snd_rme32_playback_getrate(struct rme32 * rme32)
+{
+ int rate;
+
+ rate = ((rme32->wcreg >> RME32_WCR_BITPOS_FREQ_0) & 1) +
+ (((rme32->wcreg >> RME32_WCR_BITPOS_FREQ_1) & 1) << 1);
+ switch (rate) {
+ case 1:
+ rate = 32000;
+ break;
+ case 2:
+ rate = 44100;
+ break;
+ case 3:
+ rate = 48000;
+ break;
+ default:
+ return -1;
+ }
+ return (rme32->wcreg & RME32_WCR_DS_BM) ? rate << 1 : rate;
+}
+
+static int snd_rme32_capture_getrate(struct rme32 * rme32, int *is_adat)
+{
+ int n;
+
+ *is_adat = 0;
+ if (rme32->rcreg & RME32_RCR_LOCK) {
+ /* ADAT rate */
+ *is_adat = 1;
+ }
+ if (rme32->rcreg & RME32_RCR_ERF) {
+ return -1;
+ }
+
+ /* S/PDIF rate */
+ n = ((rme32->rcreg >> RME32_RCR_BITPOS_F0) & 1) +
+ (((rme32->rcreg >> RME32_RCR_BITPOS_F1) & 1) << 1) +
+ (((rme32->rcreg >> RME32_RCR_BITPOS_F2) & 1) << 2);
+
+ if (RME32_PRO_WITH_8414(rme32))
+ switch (n) { /* supporting the CS8414 */
+ case 0:
+ case 1:
+ case 2:
+ return -1;
+ case 3:
+ return 96000;
+ case 4:
+ return 88200;
+ case 5:
+ return 48000;
+ case 6:
+ return 44100;
+ case 7:
+ return 32000;
+ default:
+ return -1;
+ break;
+ }
+ else
+ switch (n) { /* supporting the CS8412 */
+ case 0:
+ return -1;
+ case 1:
+ return 48000;
+ case 2:
+ return 44100;
+ case 3:
+ return 32000;
+ case 4:
+ return 48000;
+ case 5:
+ return 44100;
+ case 6:
+ return 44056;
+ case 7:
+ return 32000;
+ default:
+ break;
+ }
+ return -1;
+}
+
+static int snd_rme32_playback_setrate(struct rme32 * rme32, int rate)
+{
+ int ds;
+
+ ds = rme32->wcreg & RME32_WCR_DS_BM;
+ switch (rate) {
+ case 32000:
+ rme32->wcreg &= ~RME32_WCR_DS_BM;
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) &
+ ~RME32_WCR_FREQ_1;
+ break;
+ case 44100:
+ rme32->wcreg &= ~RME32_WCR_DS_BM;
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_1) &
+ ~RME32_WCR_FREQ_0;
+ break;
+ case 48000:
+ rme32->wcreg &= ~RME32_WCR_DS_BM;
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) |
+ RME32_WCR_FREQ_1;
+ break;
+ case 64000:
+ if (rme32->pci->device != PCI_DEVICE_ID_RME_DIGI32_PRO)
+ return -EINVAL;
+ rme32->wcreg |= RME32_WCR_DS_BM;
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) &
+ ~RME32_WCR_FREQ_1;
+ break;
+ case 88200:
+ if (rme32->pci->device != PCI_DEVICE_ID_RME_DIGI32_PRO)
+ return -EINVAL;
+ rme32->wcreg |= RME32_WCR_DS_BM;
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_1) &
+ ~RME32_WCR_FREQ_0;
+ break;
+ case 96000:
+ if (rme32->pci->device != PCI_DEVICE_ID_RME_DIGI32_PRO)
+ return -EINVAL;
+ rme32->wcreg |= RME32_WCR_DS_BM;
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) |
+ RME32_WCR_FREQ_1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if ((!ds && rme32->wcreg & RME32_WCR_DS_BM) ||
+ (ds && !(rme32->wcreg & RME32_WCR_DS_BM)))
+ {
+ /* change to/from double-speed: reset the DAC (if available) */
+ snd_rme32_reset_dac(rme32);
+ } else {
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ }
+ return 0;
+}
+
+static int snd_rme32_setclockmode(struct rme32 * rme32, int mode)
+{
+ switch (mode) {
+ case RME32_CLOCKMODE_SLAVE:
+ /* AutoSync */
+ rme32->wcreg = (rme32->wcreg & ~RME32_WCR_FREQ_0) &
+ ~RME32_WCR_FREQ_1;
+ break;
+ case RME32_CLOCKMODE_MASTER_32:
+ /* Internal 32.0kHz */
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) &
+ ~RME32_WCR_FREQ_1;
+ break;
+ case RME32_CLOCKMODE_MASTER_44:
+ /* Internal 44.1kHz */
+ rme32->wcreg = (rme32->wcreg & ~RME32_WCR_FREQ_0) |
+ RME32_WCR_FREQ_1;
+ break;
+ case RME32_CLOCKMODE_MASTER_48:
+ /* Internal 48.0kHz */
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_FREQ_0) |
+ RME32_WCR_FREQ_1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static int snd_rme32_getclockmode(struct rme32 * rme32)
+{
+ return ((rme32->wcreg >> RME32_WCR_BITPOS_FREQ_0) & 1) +
+ (((rme32->wcreg >> RME32_WCR_BITPOS_FREQ_1) & 1) << 1);
+}
+
+static int snd_rme32_setinputtype(struct rme32 * rme32, int type)
+{
+ switch (type) {
+ case RME32_INPUT_OPTICAL:
+ rme32->wcreg = (rme32->wcreg & ~RME32_WCR_INP_0) &
+ ~RME32_WCR_INP_1;
+ break;
+ case RME32_INPUT_COAXIAL:
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_INP_0) &
+ ~RME32_WCR_INP_1;
+ break;
+ case RME32_INPUT_INTERNAL:
+ rme32->wcreg = (rme32->wcreg & ~RME32_WCR_INP_0) |
+ RME32_WCR_INP_1;
+ break;
+ case RME32_INPUT_XLR:
+ rme32->wcreg = (rme32->wcreg | RME32_WCR_INP_0) |
+ RME32_WCR_INP_1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static int snd_rme32_getinputtype(struct rme32 * rme32)
+{
+ return ((rme32->wcreg >> RME32_WCR_BITPOS_INP_0) & 1) +
+ (((rme32->wcreg >> RME32_WCR_BITPOS_INP_1) & 1) << 1);
+}
+
+static void
+snd_rme32_setframelog(struct rme32 * rme32, int n_channels, int is_playback)
+{
+ int frlog;
+
+ if (n_channels == 2) {
+ frlog = 1;
+ } else {
+ /* assume 8 channels */
+ frlog = 3;
+ }
+ if (is_playback) {
+ frlog += (rme32->wcreg & RME32_WCR_MODE24) ? 2 : 1;
+ rme32->playback_frlog = frlog;
+ } else {
+ frlog += (rme32->wcreg & RME32_WCR_MODE24) ? 2 : 1;
+ rme32->capture_frlog = frlog;
+ }
+}
+
+static int snd_rme32_setformat(struct rme32 * rme32, int format)
+{
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ rme32->wcreg &= ~RME32_WCR_MODE24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ rme32->wcreg |= RME32_WCR_MODE24;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static int
+snd_rme32_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ int err, rate, dummy;
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (rme32->fullduplex_mode) {
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (err < 0)
+ return err;
+ } else {
+ runtime->dma_area = (void __force *)(rme32->iobase +
+ RME32_IO_DATA_BUFFER);
+ runtime->dma_addr = rme32->port + RME32_IO_DATA_BUFFER;
+ runtime->dma_bytes = RME32_BUFFER_SIZE;
+ }
+
+ spin_lock_irq(&rme32->lock);
+ if ((rme32->rcreg & RME32_RCR_KMODE) &&
+ (rate = snd_rme32_capture_getrate(rme32, &dummy)) > 0) {
+ /* AutoSync */
+ if ((int)params_rate(params) != rate) {
+ spin_unlock_irq(&rme32->lock);
+ return -EIO;
+ }
+ } else if ((err = snd_rme32_playback_setrate(rme32, params_rate(params))) < 0) {
+ spin_unlock_irq(&rme32->lock);
+ return err;
+ }
+ if ((err = snd_rme32_setformat(rme32, params_format(params))) < 0) {
+ spin_unlock_irq(&rme32->lock);
+ return err;
+ }
+
+ snd_rme32_setframelog(rme32, params_channels(params), 1);
+ if (rme32->capture_periodsize != 0) {
+ if (params_period_size(params) << rme32->playback_frlog != rme32->capture_periodsize) {
+ spin_unlock_irq(&rme32->lock);
+ return -EBUSY;
+ }
+ }
+ rme32->playback_periodsize = params_period_size(params) << rme32->playback_frlog;
+ /* S/PDIF setup */
+ if ((rme32->wcreg & RME32_WCR_ADAT) == 0) {
+ rme32->wcreg &= ~(RME32_WCR_PRO | RME32_WCR_EMP);
+ rme32->wcreg |= rme32->wcreg_spdif_stream;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ }
+ spin_unlock_irq(&rme32->lock);
+
+ return 0;
+}
+
+static int
+snd_rme32_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ int err, isadat, rate;
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (rme32->fullduplex_mode) {
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (err < 0)
+ return err;
+ } else {
+ runtime->dma_area = (void __force *)rme32->iobase +
+ RME32_IO_DATA_BUFFER;
+ runtime->dma_addr = rme32->port + RME32_IO_DATA_BUFFER;
+ runtime->dma_bytes = RME32_BUFFER_SIZE;
+ }
+
+ spin_lock_irq(&rme32->lock);
+ /* enable AutoSync for record-preparing */
+ rme32->wcreg |= RME32_WCR_AUTOSYNC;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+
+ if ((err = snd_rme32_setformat(rme32, params_format(params))) < 0) {
+ spin_unlock_irq(&rme32->lock);
+ return err;
+ }
+ if ((err = snd_rme32_playback_setrate(rme32, params_rate(params))) < 0) {
+ spin_unlock_irq(&rme32->lock);
+ return err;
+ }
+ if ((rate = snd_rme32_capture_getrate(rme32, &isadat)) > 0) {
+ if ((int)params_rate(params) != rate) {
+ spin_unlock_irq(&rme32->lock);
+ return -EIO;
+ }
+ if ((isadat && runtime->hw.channels_min == 2) ||
+ (!isadat && runtime->hw.channels_min == 8)) {
+ spin_unlock_irq(&rme32->lock);
+ return -EIO;
+ }
+ }
+ /* AutoSync off for recording */
+ rme32->wcreg &= ~RME32_WCR_AUTOSYNC;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+
+ snd_rme32_setframelog(rme32, params_channels(params), 0);
+ if (rme32->playback_periodsize != 0) {
+ if (params_period_size(params) << rme32->capture_frlog !=
+ rme32->playback_periodsize) {
+ spin_unlock_irq(&rme32->lock);
+ return -EBUSY;
+ }
+ }
+ rme32->capture_periodsize =
+ params_period_size(params) << rme32->capture_frlog;
+ spin_unlock_irq(&rme32->lock);
+
+ return 0;
+}
+
+static int snd_rme32_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ if (! rme32->fullduplex_mode)
+ return 0;
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static void snd_rme32_pcm_start(struct rme32 * rme32, int from_pause)
+{
+ if (!from_pause) {
+ writel(0, rme32->iobase + RME32_IO_RESET_POS);
+ }
+
+ rme32->wcreg |= RME32_WCR_START;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+}
+
+static void snd_rme32_pcm_stop(struct rme32 * rme32, int to_pause)
+{
+ /*
+ * Check if there is an unconfirmed IRQ, if so confirm it, or else
+ * the hardware will not stop generating interrupts
+ */
+ rme32->rcreg = readl(rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ if (rme32->rcreg & RME32_RCR_IRQ) {
+ writel(0, rme32->iobase + RME32_IO_CONFIRM_ACTION_IRQ);
+ }
+ rme32->wcreg &= ~RME32_WCR_START;
+ if (rme32->wcreg & RME32_WCR_SEL)
+ rme32->wcreg |= RME32_WCR_MUTE;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ if (! to_pause)
+ writel(0, rme32->iobase + RME32_IO_RESET_POS);
+}
+
+static irqreturn_t snd_rme32_interrupt(int irq, void *dev_id)
+{
+ struct rme32 *rme32 = (struct rme32 *) dev_id;
+
+ rme32->rcreg = readl(rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ if (!(rme32->rcreg & RME32_RCR_IRQ)) {
+ return IRQ_NONE;
+ } else {
+ if (rme32->capture_substream) {
+ snd_pcm_period_elapsed(rme32->capture_substream);
+ }
+ if (rme32->playback_substream) {
+ snd_pcm_period_elapsed(rme32->playback_substream);
+ }
+ writel(0, rme32->iobase + RME32_IO_CONFIRM_ACTION_IRQ);
+ }
+ return IRQ_HANDLED;
+}
+
+static unsigned int period_bytes[] = { RME32_BLOCK_SIZE };
+
+
+static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = {
+ .count = ARRAY_SIZE(period_bytes),
+ .list = period_bytes,
+ .mask = 0
+};
+
+static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_runtime *runtime)
+{
+ if (! rme32->fullduplex_mode) {
+ snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ RME32_BUFFER_SIZE, RME32_BUFFER_SIZE);
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ &hw_constraints_period_bytes);
+ }
+}
+
+static int snd_rme32_playback_spdif_open(struct snd_pcm_substream *substream)
+{
+ int rate, dummy;
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_pcm_set_sync(substream);
+
+ spin_lock_irq(&rme32->lock);
+ if (rme32->playback_substream != NULL) {
+ spin_unlock_irq(&rme32->lock);
+ return -EBUSY;
+ }
+ rme32->wcreg &= ~RME32_WCR_ADAT;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ rme32->playback_substream = substream;
+ spin_unlock_irq(&rme32->lock);
+
+ if (rme32->fullduplex_mode)
+ runtime->hw = snd_rme32_spdif_fd_info;
+ else
+ runtime->hw = snd_rme32_spdif_info;
+ if (rme32->pci->device == PCI_DEVICE_ID_RME_DIGI32_PRO) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
+ if ((rme32->rcreg & RME32_RCR_KMODE) &&
+ (rate = snd_rme32_capture_getrate(rme32, &dummy)) > 0) {
+ /* AutoSync */
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+
+ snd_rme32_set_buffer_constraint(rme32, runtime);
+
+ rme32->wcreg_spdif_stream = rme32->wcreg_spdif;
+ rme32->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(rme32->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &rme32->spdif_ctl->id);
+ return 0;
+}
+
+static int snd_rme32_capture_spdif_open(struct snd_pcm_substream *substream)
+{
+ int isadat, rate;
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_pcm_set_sync(substream);
+
+ spin_lock_irq(&rme32->lock);
+ if (rme32->capture_substream != NULL) {
+ spin_unlock_irq(&rme32->lock);
+ return -EBUSY;
+ }
+ rme32->capture_substream = substream;
+ spin_unlock_irq(&rme32->lock);
+
+ if (rme32->fullduplex_mode)
+ runtime->hw = snd_rme32_spdif_fd_info;
+ else
+ runtime->hw = snd_rme32_spdif_info;
+ if (RME32_PRO_WITH_8414(rme32)) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
+ if ((rate = snd_rme32_capture_getrate(rme32, &isadat)) > 0) {
+ if (isadat) {
+ return -EIO;
+ }
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+
+ snd_rme32_set_buffer_constraint(rme32, runtime);
+
+ return 0;
+}
+
+static int
+snd_rme32_playback_adat_open(struct snd_pcm_substream *substream)
+{
+ int rate, dummy;
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_pcm_set_sync(substream);
+
+ spin_lock_irq(&rme32->lock);
+ if (rme32->playback_substream != NULL) {
+ spin_unlock_irq(&rme32->lock);
+ return -EBUSY;
+ }
+ rme32->wcreg |= RME32_WCR_ADAT;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ rme32->playback_substream = substream;
+ spin_unlock_irq(&rme32->lock);
+
+ if (rme32->fullduplex_mode)
+ runtime->hw = snd_rme32_adat_fd_info;
+ else
+ runtime->hw = snd_rme32_adat_info;
+ if ((rme32->rcreg & RME32_RCR_KMODE) &&
+ (rate = snd_rme32_capture_getrate(rme32, &dummy)) > 0) {
+ /* AutoSync */
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+
+ snd_rme32_set_buffer_constraint(rme32, runtime);
+ return 0;
+}
+
+static int
+snd_rme32_capture_adat_open(struct snd_pcm_substream *substream)
+{
+ int isadat, rate;
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (rme32->fullduplex_mode)
+ runtime->hw = snd_rme32_adat_fd_info;
+ else
+ runtime->hw = snd_rme32_adat_info;
+ if ((rate = snd_rme32_capture_getrate(rme32, &isadat)) > 0) {
+ if (!isadat) {
+ return -EIO;
+ }
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+
+ snd_pcm_set_sync(substream);
+
+ spin_lock_irq(&rme32->lock);
+ if (rme32->capture_substream != NULL) {
+ spin_unlock_irq(&rme32->lock);
+ return -EBUSY;
+ }
+ rme32->capture_substream = substream;
+ spin_unlock_irq(&rme32->lock);
+
+ snd_rme32_set_buffer_constraint(rme32, runtime);
+ return 0;
+}
+
+static int snd_rme32_playback_close(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ int spdif = 0;
+
+ spin_lock_irq(&rme32->lock);
+ rme32->playback_substream = NULL;
+ rme32->playback_periodsize = 0;
+ spdif = (rme32->wcreg & RME32_WCR_ADAT) == 0;
+ spin_unlock_irq(&rme32->lock);
+ if (spdif) {
+ rme32->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(rme32->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ &rme32->spdif_ctl->id);
+ }
+ return 0;
+}
+
+static int snd_rme32_capture_close(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme32->lock);
+ rme32->capture_substream = NULL;
+ rme32->capture_periodsize = 0;
+ spin_unlock(&rme32->lock);
+ return 0;
+}
+
+static int snd_rme32_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme32->lock);
+ if (rme32->fullduplex_mode) {
+ memset(&rme32->playback_pcm, 0, sizeof(rme32->playback_pcm));
+ rme32->playback_pcm.hw_buffer_size = RME32_BUFFER_SIZE;
+ rme32->playback_pcm.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ } else {
+ writel(0, rme32->iobase + RME32_IO_RESET_POS);
+ }
+ if (rme32->wcreg & RME32_WCR_SEL)
+ rme32->wcreg &= ~RME32_WCR_MUTE;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ spin_unlock_irq(&rme32->lock);
+ return 0;
+}
+
+static int snd_rme32_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme32->lock);
+ if (rme32->fullduplex_mode) {
+ memset(&rme32->capture_pcm, 0, sizeof(rme32->capture_pcm));
+ rme32->capture_pcm.hw_buffer_size = RME32_BUFFER_SIZE;
+ rme32->capture_pcm.hw_queue_size = RME32_BUFFER_SIZE / 2;
+ rme32->capture_pcm.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ } else {
+ writel(0, rme32->iobase + RME32_IO_RESET_POS);
+ }
+ spin_unlock_irq(&rme32->lock);
+ return 0;
+}
+
+static int
+snd_rme32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+
+ spin_lock(&rme32->lock);
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s != rme32->playback_substream &&
+ s != rme32->capture_substream)
+ continue;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ rme32->running |= (1 << s->stream);
+ if (rme32->fullduplex_mode) {
+ /* remember the current DMA position */
+ if (s == rme32->playback_substream) {
+ rme32->playback_pcm.hw_io =
+ rme32->playback_pcm.hw_data = snd_rme32_pcm_byteptr(rme32);
+ } else {
+ rme32->capture_pcm.hw_io =
+ rme32->capture_pcm.hw_data = snd_rme32_pcm_byteptr(rme32);
+ }
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ rme32->running &= ~(1 << s->stream);
+ break;
+ }
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ /* prefill playback buffer */
+ if (cmd == SNDRV_PCM_TRIGGER_START && rme32->fullduplex_mode) {
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == rme32->playback_substream) {
+ s->ops->ack(s);
+ break;
+ }
+ }
+ }
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (rme32->running && ! RME32_ISWORKING(rme32))
+ snd_rme32_pcm_start(rme32, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (! rme32->running && RME32_ISWORKING(rme32))
+ snd_rme32_pcm_stop(rme32, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (rme32->running && RME32_ISWORKING(rme32))
+ snd_rme32_pcm_stop(rme32, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (rme32->running && ! RME32_ISWORKING(rme32))
+ snd_rme32_pcm_start(rme32, 1);
+ break;
+ }
+ spin_unlock(&rme32->lock);
+ return 0;
+}
+
+/* pointer callback for halfduplex mode */
+static snd_pcm_uframes_t
+snd_rme32_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ return snd_rme32_pcm_byteptr(rme32) >> rme32->playback_frlog;
+}
+
+static snd_pcm_uframes_t
+snd_rme32_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ return snd_rme32_pcm_byteptr(rme32) >> rme32->capture_frlog;
+}
+
+
+/* ack and pointer callbacks for fullduplex mode */
+static void snd_rme32_pb_trans_copy(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ memcpy_toio(rme32->iobase + RME32_IO_DATA_BUFFER + rec->hw_data,
+ substream->runtime->dma_area + rec->sw_data, bytes);
+}
+
+static int snd_rme32_playback_fd_ack(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_indirect *rec, *cprec;
+
+ rec = &rme32->playback_pcm;
+ cprec = &rme32->capture_pcm;
+ spin_lock(&rme32->lock);
+ rec->hw_queue_size = RME32_BUFFER_SIZE;
+ if (rme32->running & (1 << SNDRV_PCM_STREAM_CAPTURE))
+ rec->hw_queue_size -= cprec->hw_ready;
+ spin_unlock(&rme32->lock);
+ snd_pcm_indirect_playback_transfer(substream, rec,
+ snd_rme32_pb_trans_copy);
+ return 0;
+}
+
+static void snd_rme32_cp_trans_copy(struct snd_pcm_substream *substream,
+ struct snd_pcm_indirect *rec, size_t bytes)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ memcpy_fromio(substream->runtime->dma_area + rec->sw_data,
+ rme32->iobase + RME32_IO_DATA_BUFFER + rec->hw_data,
+ bytes);
+}
+
+static int snd_rme32_capture_fd_ack(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ snd_pcm_indirect_capture_transfer(substream, &rme32->capture_pcm,
+ snd_rme32_cp_trans_copy);
+ return 0;
+}
+
+static snd_pcm_uframes_t
+snd_rme32_playback_fd_pointer(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ return snd_pcm_indirect_playback_pointer(substream, &rme32->playback_pcm,
+ snd_rme32_pcm_byteptr(rme32));
+}
+
+static snd_pcm_uframes_t
+snd_rme32_capture_fd_pointer(struct snd_pcm_substream *substream)
+{
+ struct rme32 *rme32 = snd_pcm_substream_chip(substream);
+ return snd_pcm_indirect_capture_pointer(substream, &rme32->capture_pcm,
+ snd_rme32_pcm_byteptr(rme32));
+}
+
+/* for halfduplex mode */
+static struct snd_pcm_ops snd_rme32_playback_spdif_ops = {
+ .open = snd_rme32_playback_spdif_open,
+ .close = snd_rme32_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_playback_hw_params,
+ .hw_free = snd_rme32_pcm_hw_free,
+ .prepare = snd_rme32_playback_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_playback_pointer,
+ .copy = snd_rme32_playback_copy,
+ .silence = snd_rme32_playback_silence,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static struct snd_pcm_ops snd_rme32_capture_spdif_ops = {
+ .open = snd_rme32_capture_spdif_open,
+ .close = snd_rme32_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_capture_hw_params,
+ .hw_free = snd_rme32_pcm_hw_free,
+ .prepare = snd_rme32_capture_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_capture_pointer,
+ .copy = snd_rme32_capture_copy,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static struct snd_pcm_ops snd_rme32_playback_adat_ops = {
+ .open = snd_rme32_playback_adat_open,
+ .close = snd_rme32_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_playback_hw_params,
+ .prepare = snd_rme32_playback_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_playback_pointer,
+ .copy = snd_rme32_playback_copy,
+ .silence = snd_rme32_playback_silence,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static struct snd_pcm_ops snd_rme32_capture_adat_ops = {
+ .open = snd_rme32_capture_adat_open,
+ .close = snd_rme32_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_capture_hw_params,
+ .prepare = snd_rme32_capture_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_capture_pointer,
+ .copy = snd_rme32_capture_copy,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+/* for fullduplex mode */
+static struct snd_pcm_ops snd_rme32_playback_spdif_fd_ops = {
+ .open = snd_rme32_playback_spdif_open,
+ .close = snd_rme32_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_playback_hw_params,
+ .hw_free = snd_rme32_pcm_hw_free,
+ .prepare = snd_rme32_playback_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_playback_fd_pointer,
+ .ack = snd_rme32_playback_fd_ack,
+};
+
+static struct snd_pcm_ops snd_rme32_capture_spdif_fd_ops = {
+ .open = snd_rme32_capture_spdif_open,
+ .close = snd_rme32_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_capture_hw_params,
+ .hw_free = snd_rme32_pcm_hw_free,
+ .prepare = snd_rme32_capture_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_capture_fd_pointer,
+ .ack = snd_rme32_capture_fd_ack,
+};
+
+static struct snd_pcm_ops snd_rme32_playback_adat_fd_ops = {
+ .open = snd_rme32_playback_adat_open,
+ .close = snd_rme32_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_playback_hw_params,
+ .prepare = snd_rme32_playback_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_playback_fd_pointer,
+ .ack = snd_rme32_playback_fd_ack,
+};
+
+static struct snd_pcm_ops snd_rme32_capture_adat_fd_ops = {
+ .open = snd_rme32_capture_adat_open,
+ .close = snd_rme32_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme32_capture_hw_params,
+ .prepare = snd_rme32_capture_prepare,
+ .trigger = snd_rme32_pcm_trigger,
+ .pointer = snd_rme32_capture_fd_pointer,
+ .ack = snd_rme32_capture_fd_ack,
+};
+
+static void snd_rme32_free(void *private_data)
+{
+ struct rme32 *rme32 = (struct rme32 *) private_data;
+
+ if (rme32 == NULL) {
+ return;
+ }
+ if (rme32->irq >= 0) {
+ snd_rme32_pcm_stop(rme32, 0);
+ free_irq(rme32->irq, (void *) rme32);
+ rme32->irq = -1;
+ }
+ if (rme32->iobase) {
+ iounmap(rme32->iobase);
+ rme32->iobase = NULL;
+ }
+ if (rme32->port) {
+ pci_release_regions(rme32->pci);
+ rme32->port = 0;
+ }
+ pci_disable_device(rme32->pci);
+}
+
+static void snd_rme32_free_spdif_pcm(struct snd_pcm *pcm)
+{
+ struct rme32 *rme32 = (struct rme32 *) pcm->private_data;
+ rme32->spdif_pcm = NULL;
+}
+
+static void
+snd_rme32_free_adat_pcm(struct snd_pcm *pcm)
+{
+ struct rme32 *rme32 = (struct rme32 *) pcm->private_data;
+ rme32->adat_pcm = NULL;
+}
+
+static int __devinit snd_rme32_create(struct rme32 * rme32)
+{
+ struct pci_dev *pci = rme32->pci;
+ int err;
+
+ rme32->irq = -1;
+ spin_lock_init(&rme32->lock);
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ if ((err = pci_request_regions(pci, "RME32")) < 0)
+ return err;
+ rme32->port = pci_resource_start(rme32->pci, 0);
+
+ rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE);
+ if (!rme32->iobase) {
+ snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n",
+ rme32->port, rme32->port + RME32_IO_SIZE - 1);
+ return -ENOMEM;
+ }
+
+ if (request_irq(pci->irq, snd_rme32_interrupt, IRQF_SHARED,
+ "RME32", rme32)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ return -EBUSY;
+ }
+ rme32->irq = pci->irq;
+
+ /* read the card's revision number */
+ pci_read_config_byte(pci, 8, &rme32->rev);
+
+ /* set up ALSA pcm device for S/PDIF */
+ if ((err = snd_pcm_new(rme32->card, "Digi32 IEC958", 0, 1, 1, &rme32->spdif_pcm)) < 0) {
+ return err;
+ }
+ rme32->spdif_pcm->private_data = rme32;
+ rme32->spdif_pcm->private_free = snd_rme32_free_spdif_pcm;
+ strcpy(rme32->spdif_pcm->name, "Digi32 IEC958");
+ if (rme32->fullduplex_mode) {
+ snd_pcm_set_ops(rme32->spdif_pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_rme32_playback_spdif_fd_ops);
+ snd_pcm_set_ops(rme32->spdif_pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_rme32_capture_spdif_fd_ops);
+ snd_pcm_lib_preallocate_pages_for_all(rme32->spdif_pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 0, RME32_MID_BUFFER_SIZE);
+ rme32->spdif_pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+ } else {
+ snd_pcm_set_ops(rme32->spdif_pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_rme32_playback_spdif_ops);
+ snd_pcm_set_ops(rme32->spdif_pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_rme32_capture_spdif_ops);
+ rme32->spdif_pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
+ }
+
+ /* set up ALSA pcm device for ADAT */
+ if ((pci->device == PCI_DEVICE_ID_RME_DIGI32) ||
+ (pci->device == PCI_DEVICE_ID_RME_DIGI32_PRO)) {
+ /* ADAT is not available on DIGI32 and DIGI32 Pro */
+ rme32->adat_pcm = NULL;
+ }
+ else {
+ if ((err = snd_pcm_new(rme32->card, "Digi32 ADAT", 1,
+ 1, 1, &rme32->adat_pcm)) < 0)
+ {
+ return err;
+ }
+ rme32->adat_pcm->private_data = rme32;
+ rme32->adat_pcm->private_free = snd_rme32_free_adat_pcm;
+ strcpy(rme32->adat_pcm->name, "Digi32 ADAT");
+ if (rme32->fullduplex_mode) {
+ snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_rme32_playback_adat_fd_ops);
+ snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_rme32_capture_adat_fd_ops);
+ snd_pcm_lib_preallocate_pages_for_all(rme32->adat_pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 0, RME32_MID_BUFFER_SIZE);
+ rme32->adat_pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+ } else {
+ snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_rme32_playback_adat_ops);
+ snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_rme32_capture_adat_ops);
+ rme32->adat_pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
+ }
+ }
+
+
+ rme32->playback_periodsize = 0;
+ rme32->capture_periodsize = 0;
+
+ /* make sure playback/capture is stopped, if by some reason active */
+ snd_rme32_pcm_stop(rme32, 0);
+
+ /* reset DAC */
+ snd_rme32_reset_dac(rme32);
+
+ /* reset buffer pointer */
+ writel(0, rme32->iobase + RME32_IO_RESET_POS);
+
+ /* set default values in registers */
+ rme32->wcreg = RME32_WCR_SEL | /* normal playback */
+ RME32_WCR_INP_0 | /* input select */
+ RME32_WCR_MUTE; /* muting on */
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+
+
+ /* init switch interface */
+ if ((err = snd_rme32_create_switches(rme32->card, rme32)) < 0) {
+ return err;
+ }
+
+ /* init proc interface */
+ snd_rme32_proc_init(rme32);
+
+ rme32->capture_substream = NULL;
+ rme32->playback_substream = NULL;
+
+ return 0;
+}
+
+/*
+ * proc interface
+ */
+
+static void
+snd_rme32_proc_read(struct snd_info_entry * entry, struct snd_info_buffer *buffer)
+{
+ int n;
+ struct rme32 *rme32 = (struct rme32 *) entry->private_data;
+
+ rme32->rcreg = readl(rme32->iobase + RME32_IO_CONTROL_REGISTER);
+
+ snd_iprintf(buffer, rme32->card->longname);
+ snd_iprintf(buffer, " (index #%d)\n", rme32->card->number + 1);
+
+ snd_iprintf(buffer, "\nGeneral settings\n");
+ if (rme32->fullduplex_mode)
+ snd_iprintf(buffer, " Full-duplex mode\n");
+ else
+ snd_iprintf(buffer, " Half-duplex mode\n");
+ if (RME32_PRO_WITH_8414(rme32)) {
+ snd_iprintf(buffer, " receiver: CS8414\n");
+ } else {
+ snd_iprintf(buffer, " receiver: CS8412\n");
+ }
+ if (rme32->wcreg & RME32_WCR_MODE24) {
+ snd_iprintf(buffer, " format: 24 bit");
+ } else {
+ snd_iprintf(buffer, " format: 16 bit");
+ }
+ if (rme32->wcreg & RME32_WCR_MONO) {
+ snd_iprintf(buffer, ", Mono\n");
+ } else {
+ snd_iprintf(buffer, ", Stereo\n");
+ }
+
+ snd_iprintf(buffer, "\nInput settings\n");
+ switch (snd_rme32_getinputtype(rme32)) {
+ case RME32_INPUT_OPTICAL:
+ snd_iprintf(buffer, " input: optical");
+ break;
+ case RME32_INPUT_COAXIAL:
+ snd_iprintf(buffer, " input: coaxial");
+ break;
+ case RME32_INPUT_INTERNAL:
+ snd_iprintf(buffer, " input: internal");
+ break;
+ case RME32_INPUT_XLR:
+ snd_iprintf(buffer, " input: XLR");
+ break;
+ }
+ if (snd_rme32_capture_getrate(rme32, &n) < 0) {
+ snd_iprintf(buffer, "\n sample rate: no valid signal\n");
+ } else {
+ if (n) {
+ snd_iprintf(buffer, " (8 channels)\n");
+ } else {
+ snd_iprintf(buffer, " (2 channels)\n");
+ }
+ snd_iprintf(buffer, " sample rate: %d Hz\n",
+ snd_rme32_capture_getrate(rme32, &n));
+ }
+
+ snd_iprintf(buffer, "\nOutput settings\n");
+ if (rme32->wcreg & RME32_WCR_SEL) {
+ snd_iprintf(buffer, " output signal: normal playback");
+ } else {
+ snd_iprintf(buffer, " output signal: same as input");
+ }
+ if (rme32->wcreg & RME32_WCR_MUTE) {
+ snd_iprintf(buffer, " (muted)\n");
+ } else {
+ snd_iprintf(buffer, "\n");
+ }
+
+ /* master output frequency */
+ if (!
+ ((!(rme32->wcreg & RME32_WCR_FREQ_0))
+ && (!(rme32->wcreg & RME32_WCR_FREQ_1)))) {
+ snd_iprintf(buffer, " sample rate: %d Hz\n",
+ snd_rme32_playback_getrate(rme32));
+ }
+ if (rme32->rcreg & RME32_RCR_KMODE) {
+ snd_iprintf(buffer, " sample clock source: AutoSync\n");
+ } else {
+ snd_iprintf(buffer, " sample clock source: Internal\n");
+ }
+ if (rme32->wcreg & RME32_WCR_PRO) {
+ snd_iprintf(buffer, " format: AES/EBU (professional)\n");
+ } else {
+ snd_iprintf(buffer, " format: IEC958 (consumer)\n");
+ }
+ if (rme32->wcreg & RME32_WCR_EMP) {
+ snd_iprintf(buffer, " emphasis: on\n");
+ } else {
+ snd_iprintf(buffer, " emphasis: off\n");
+ }
+}
+
+static void __devinit snd_rme32_proc_init(struct rme32 * rme32)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(rme32->card, "rme32", &entry))
+ snd_info_set_text_ops(entry, rme32, snd_rme32_proc_read);
+}
+
+/*
+ * control interface
+ */
+
+#define snd_rme32_info_loopback_control snd_ctl_boolean_mono_info
+
+static int
+snd_rme32_get_loopback_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme32->lock);
+ ucontrol->value.integer.value[0] =
+ rme32->wcreg & RME32_WCR_SEL ? 0 : 1;
+ spin_unlock_irq(&rme32->lock);
+ return 0;
+}
+static int
+snd_rme32_put_loopback_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ucontrol->value.integer.value[0] ? 0 : RME32_WCR_SEL;
+ spin_lock_irq(&rme32->lock);
+ val = (rme32->wcreg & ~RME32_WCR_SEL) | val;
+ change = val != rme32->wcreg;
+ if (ucontrol->value.integer.value[0])
+ val &= ~RME32_WCR_MUTE;
+ else
+ val |= RME32_WCR_MUTE;
+ rme32->wcreg = val;
+ writel(val, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ spin_unlock_irq(&rme32->lock);
+ return change;
+}
+
+static int
+snd_rme32_info_inputtype_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+ static char *texts[4] = { "Optical", "Coaxial", "Internal", "XLR" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ switch (rme32->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI32:
+ case PCI_DEVICE_ID_RME_DIGI32_8:
+ uinfo->value.enumerated.items = 3;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI32_PRO:
+ uinfo->value.enumerated.items = 4;
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ if (uinfo->value.enumerated.item >
+ uinfo->value.enumerated.items - 1) {
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ }
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+static int
+snd_rme32_get_inputtype_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+ unsigned int items = 3;
+
+ spin_lock_irq(&rme32->lock);
+ ucontrol->value.enumerated.item[0] = snd_rme32_getinputtype(rme32);
+
+ switch (rme32->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI32:
+ case PCI_DEVICE_ID_RME_DIGI32_8:
+ items = 3;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI32_PRO:
+ items = 4;
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ if (ucontrol->value.enumerated.item[0] >= items) {
+ ucontrol->value.enumerated.item[0] = items - 1;
+ }
+
+ spin_unlock_irq(&rme32->lock);
+ return 0;
+}
+static int
+snd_rme32_put_inputtype_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change, items = 3;
+
+ switch (rme32->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI32:
+ case PCI_DEVICE_ID_RME_DIGI32_8:
+ items = 3;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI32_PRO:
+ items = 4;
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ val = ucontrol->value.enumerated.item[0] % items;
+
+ spin_lock_irq(&rme32->lock);
+ change = val != (unsigned int)snd_rme32_getinputtype(rme32);
+ snd_rme32_setinputtype(rme32, val);
+ spin_unlock_irq(&rme32->lock);
+ return change;
+}
+
+static int
+snd_rme32_info_clockmode_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = { "AutoSync",
+ "Internal 32.0kHz",
+ "Internal 44.1kHz",
+ "Internal 48.0kHz" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3) {
+ uinfo->value.enumerated.item = 3;
+ }
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+static int
+snd_rme32_get_clockmode_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme32->lock);
+ ucontrol->value.enumerated.item[0] = snd_rme32_getclockmode(rme32);
+ spin_unlock_irq(&rme32->lock);
+ return 0;
+}
+static int
+snd_rme32_put_clockmode_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ucontrol->value.enumerated.item[0] % 3;
+ spin_lock_irq(&rme32->lock);
+ change = val != (unsigned int)snd_rme32_getclockmode(rme32);
+ snd_rme32_setclockmode(rme32, val);
+ spin_unlock_irq(&rme32->lock);
+ return change;
+}
+
+static u32 snd_rme32_convert_from_aes(struct snd_aes_iec958 * aes)
+{
+ u32 val = 0;
+ val |= (aes->status[0] & IEC958_AES0_PROFESSIONAL) ? RME32_WCR_PRO : 0;
+ if (val & RME32_WCR_PRO)
+ val |= (aes->status[0] & IEC958_AES0_PRO_EMPHASIS_5015) ? RME32_WCR_EMP : 0;
+ else
+ val |= (aes->status[0] & IEC958_AES0_CON_EMPHASIS_5015) ? RME32_WCR_EMP : 0;
+ return val;
+}
+
+static void snd_rme32_convert_to_aes(struct snd_aes_iec958 * aes, u32 val)
+{
+ aes->status[0] = ((val & RME32_WCR_PRO) ? IEC958_AES0_PROFESSIONAL : 0);
+ if (val & RME32_WCR_PRO)
+ aes->status[0] |= (val & RME32_WCR_EMP) ? IEC958_AES0_PRO_EMPHASIS_5015 : 0;
+ else
+ aes->status[0] |= (val & RME32_WCR_EMP) ? IEC958_AES0_CON_EMPHASIS_5015 : 0;
+}
+
+static int snd_rme32_control_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme32_control_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+
+ snd_rme32_convert_to_aes(&ucontrol->value.iec958,
+ rme32->wcreg_spdif);
+ return 0;
+}
+
+static int snd_rme32_control_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_rme32_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&rme32->lock);
+ change = val != rme32->wcreg_spdif;
+ rme32->wcreg_spdif = val;
+ spin_unlock_irq(&rme32->lock);
+ return change;
+}
+
+static int snd_rme32_control_spdif_stream_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme32_control_spdif_stream_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *
+ ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+
+ snd_rme32_convert_to_aes(&ucontrol->value.iec958,
+ rme32->wcreg_spdif_stream);
+ return 0;
+}
+
+static int snd_rme32_control_spdif_stream_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *
+ ucontrol)
+{
+ struct rme32 *rme32 = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_rme32_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&rme32->lock);
+ change = val != rme32->wcreg_spdif_stream;
+ rme32->wcreg_spdif_stream = val;
+ rme32->wcreg &= ~(RME32_WCR_PRO | RME32_WCR_EMP);
+ rme32->wcreg |= val;
+ writel(rme32->wcreg, rme32->iobase + RME32_IO_CONTROL_REGISTER);
+ spin_unlock_irq(&rme32->lock);
+ return change;
+}
+
+static int snd_rme32_control_spdif_mask_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme32_control_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *
+ ucontrol)
+{
+ ucontrol->value.iec958.status[0] = kcontrol->private_value;
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_rme32_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .info = snd_rme32_control_spdif_info,
+ .get = snd_rme32_control_spdif_get,
+ .put = snd_rme32_control_spdif_put
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM),
+ .info = snd_rme32_control_spdif_stream_info,
+ .get = snd_rme32_control_spdif_stream_get,
+ .put = snd_rme32_control_spdif_stream_put
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK),
+ .info = snd_rme32_control_spdif_mask_info,
+ .get = snd_rme32_control_spdif_mask_get,
+ .private_value = IEC958_AES0_PROFESSIONAL | IEC958_AES0_CON_EMPHASIS
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK),
+ .info = snd_rme32_control_spdif_mask_info,
+ .get = snd_rme32_control_spdif_mask_get,
+ .private_value = IEC958_AES0_PROFESSIONAL | IEC958_AES0_PRO_EMPHASIS
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Connector",
+ .info = snd_rme32_info_inputtype_control,
+ .get = snd_rme32_get_inputtype_control,
+ .put = snd_rme32_put_inputtype_control
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Loopback Input",
+ .info = snd_rme32_info_loopback_control,
+ .get = snd_rme32_get_loopback_control,
+ .put = snd_rme32_put_loopback_control
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Sample Clock Source",
+ .info = snd_rme32_info_clockmode_control,
+ .get = snd_rme32_get_clockmode_control,
+ .put = snd_rme32_put_clockmode_control
+ }
+};
+
+static int snd_rme32_create_switches(struct snd_card *card, struct rme32 * rme32)
+{
+ int idx, err;
+ struct snd_kcontrol *kctl;
+
+ for (idx = 0; idx < (int)ARRAY_SIZE(snd_rme32_controls); idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_rme32_controls[idx], rme32))) < 0)
+ return err;
+ if (idx == 1) /* IEC958 (S/PDIF) Stream */
+ rme32->spdif_ctl = kctl;
+ }
+
+ return 0;
+}
+
+/*
+ * Card initialisation
+ */
+
+static void snd_rme32_card_free(struct snd_card *card)
+{
+ snd_rme32_free(card->private_data);
+}
+
+static int __devinit
+snd_rme32_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct rme32 *rme32;
+ struct snd_card *card;
+ int err;
+
+ if (dev >= SNDRV_CARDS) {
+ return -ENODEV;
+ }
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct rme32))) == NULL)
+ return -ENOMEM;
+ card->private_free = snd_rme32_card_free;
+ rme32 = (struct rme32 *) card->private_data;
+ rme32->card = card;
+ rme32->pci = pci;
+ snd_card_set_dev(card, &pci->dev);
+ if (fullduplex[dev])
+ rme32->fullduplex_mode = 1;
+ if ((err = snd_rme32_create(rme32)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "Digi32");
+ switch (rme32->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI32:
+ strcpy(card->shortname, "RME Digi32");
+ break;
+ case PCI_DEVICE_ID_RME_DIGI32_8:
+ strcpy(card->shortname, "RME Digi32/8");
+ break;
+ case PCI_DEVICE_ID_RME_DIGI32_PRO:
+ strcpy(card->shortname, "RME Digi32 PRO");
+ break;
+ }
+ sprintf(card->longname, "%s (Rev. %d) at 0x%lx, irq %d",
+ card->shortname, rme32->rev, rme32->port, rme32->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_rme32_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "RME Digi32",
+ .id_table = snd_rme32_ids,
+ .probe = snd_rme32_probe,
+ .remove = __devexit_p(snd_rme32_remove),
+};
+
+static int __init alsa_card_rme32_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_rme32_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_rme32_init)
+module_exit(alsa_card_rme32_exit)
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
new file mode 100644
index 0000000..3fdd488
--- /dev/null
+++ b/sound/pci/rme96.c
@@ -0,0 +1,2420 @@
+/*
+ * ALSA driver for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST audio
+ * interfaces
+ *
+ * Copyright (c) 2000, 2001 Anders Torger <torger@ludd.luth.se>
+ *
+ * Thanks to Henk Hesselink <henk@anda.nl> for the analog volume control
+ * code.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+
+/* note, two last pcis should be equal, it is not a bug */
+
+MODULE_AUTHOR("Anders Torger <torger@ludd.luth.se>");
+MODULE_DESCRIPTION("RME Digi96, Digi96/8, Digi96/8 PRO, Digi96/8 PST, "
+ "Digi96/8 PAD");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{RME,Digi96},"
+ "{RME,Digi96/8},"
+ "{RME,Digi96/8 PRO},"
+ "{RME,Digi96/8 PST},"
+ "{RME,Digi96/8 PAD}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for RME Digi96 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for RME Digi96 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
+
+/*
+ * Defines for RME Digi96 series, from internal RME reference documents
+ * dated 12.01.00
+ */
+
+#define RME96_SPDIF_NCHANNELS 2
+
+/* Playback and capture buffer size */
+#define RME96_BUFFER_SIZE 0x10000
+
+/* IO area size */
+#define RME96_IO_SIZE 0x60000
+
+/* IO area offsets */
+#define RME96_IO_PLAY_BUFFER 0x0
+#define RME96_IO_REC_BUFFER 0x10000
+#define RME96_IO_CONTROL_REGISTER 0x20000
+#define RME96_IO_ADDITIONAL_REG 0x20004
+#define RME96_IO_CONFIRM_PLAY_IRQ 0x20008
+#define RME96_IO_CONFIRM_REC_IRQ 0x2000C
+#define RME96_IO_SET_PLAY_POS 0x40000
+#define RME96_IO_RESET_PLAY_POS 0x4FFFC
+#define RME96_IO_SET_REC_POS 0x50000
+#define RME96_IO_RESET_REC_POS 0x5FFFC
+#define RME96_IO_GET_PLAY_POS 0x20000
+#define RME96_IO_GET_REC_POS 0x30000
+
+/* Write control register bits */
+#define RME96_WCR_START (1 << 0)
+#define RME96_WCR_START_2 (1 << 1)
+#define RME96_WCR_GAIN_0 (1 << 2)
+#define RME96_WCR_GAIN_1 (1 << 3)
+#define RME96_WCR_MODE24 (1 << 4)
+#define RME96_WCR_MODE24_2 (1 << 5)
+#define RME96_WCR_BM (1 << 6)
+#define RME96_WCR_BM_2 (1 << 7)
+#define RME96_WCR_ADAT (1 << 8)
+#define RME96_WCR_FREQ_0 (1 << 9)
+#define RME96_WCR_FREQ_1 (1 << 10)
+#define RME96_WCR_DS (1 << 11)
+#define RME96_WCR_PRO (1 << 12)
+#define RME96_WCR_EMP (1 << 13)
+#define RME96_WCR_SEL (1 << 14)
+#define RME96_WCR_MASTER (1 << 15)
+#define RME96_WCR_PD (1 << 16)
+#define RME96_WCR_INP_0 (1 << 17)
+#define RME96_WCR_INP_1 (1 << 18)
+#define RME96_WCR_THRU_0 (1 << 19)
+#define RME96_WCR_THRU_1 (1 << 20)
+#define RME96_WCR_THRU_2 (1 << 21)
+#define RME96_WCR_THRU_3 (1 << 22)
+#define RME96_WCR_THRU_4 (1 << 23)
+#define RME96_WCR_THRU_5 (1 << 24)
+#define RME96_WCR_THRU_6 (1 << 25)
+#define RME96_WCR_THRU_7 (1 << 26)
+#define RME96_WCR_DOLBY (1 << 27)
+#define RME96_WCR_MONITOR_0 (1 << 28)
+#define RME96_WCR_MONITOR_1 (1 << 29)
+#define RME96_WCR_ISEL (1 << 30)
+#define RME96_WCR_IDIS (1 << 31)
+
+#define RME96_WCR_BITPOS_GAIN_0 2
+#define RME96_WCR_BITPOS_GAIN_1 3
+#define RME96_WCR_BITPOS_FREQ_0 9
+#define RME96_WCR_BITPOS_FREQ_1 10
+#define RME96_WCR_BITPOS_INP_0 17
+#define RME96_WCR_BITPOS_INP_1 18
+#define RME96_WCR_BITPOS_MONITOR_0 28
+#define RME96_WCR_BITPOS_MONITOR_1 29
+
+/* Read control register bits */
+#define RME96_RCR_AUDIO_ADDR_MASK 0xFFFF
+#define RME96_RCR_IRQ_2 (1 << 16)
+#define RME96_RCR_T_OUT (1 << 17)
+#define RME96_RCR_DEV_ID_0 (1 << 21)
+#define RME96_RCR_DEV_ID_1 (1 << 22)
+#define RME96_RCR_LOCK (1 << 23)
+#define RME96_RCR_VERF (1 << 26)
+#define RME96_RCR_F0 (1 << 27)
+#define RME96_RCR_F1 (1 << 28)
+#define RME96_RCR_F2 (1 << 29)
+#define RME96_RCR_AUTOSYNC (1 << 30)
+#define RME96_RCR_IRQ (1 << 31)
+
+#define RME96_RCR_BITPOS_F0 27
+#define RME96_RCR_BITPOS_F1 28
+#define RME96_RCR_BITPOS_F2 29
+
+/* Additonal register bits */
+#define RME96_AR_WSEL (1 << 0)
+#define RME96_AR_ANALOG (1 << 1)
+#define RME96_AR_FREQPAD_0 (1 << 2)
+#define RME96_AR_FREQPAD_1 (1 << 3)
+#define RME96_AR_FREQPAD_2 (1 << 4)
+#define RME96_AR_PD2 (1 << 5)
+#define RME96_AR_DAC_EN (1 << 6)
+#define RME96_AR_CLATCH (1 << 7)
+#define RME96_AR_CCLK (1 << 8)
+#define RME96_AR_CDATA (1 << 9)
+
+#define RME96_AR_BITPOS_F0 2
+#define RME96_AR_BITPOS_F1 3
+#define RME96_AR_BITPOS_F2 4
+
+/* Monitor tracks */
+#define RME96_MONITOR_TRACKS_1_2 0
+#define RME96_MONITOR_TRACKS_3_4 1
+#define RME96_MONITOR_TRACKS_5_6 2
+#define RME96_MONITOR_TRACKS_7_8 3
+
+/* Attenuation */
+#define RME96_ATTENUATION_0 0
+#define RME96_ATTENUATION_6 1
+#define RME96_ATTENUATION_12 2
+#define RME96_ATTENUATION_18 3
+
+/* Input types */
+#define RME96_INPUT_OPTICAL 0
+#define RME96_INPUT_COAXIAL 1
+#define RME96_INPUT_INTERNAL 2
+#define RME96_INPUT_XLR 3
+#define RME96_INPUT_ANALOG 4
+
+/* Clock modes */
+#define RME96_CLOCKMODE_SLAVE 0
+#define RME96_CLOCKMODE_MASTER 1
+#define RME96_CLOCKMODE_WORDCLOCK 2
+
+/* Block sizes in bytes */
+#define RME96_SMALL_BLOCK_SIZE 2048
+#define RME96_LARGE_BLOCK_SIZE 8192
+
+/* Volume control */
+#define RME96_AD1852_VOL_BITS 14
+#define RME96_AD1855_VOL_BITS 10
+
+
+struct rme96 {
+ spinlock_t lock;
+ int irq;
+ unsigned long port;
+ void __iomem *iobase;
+
+ u32 wcreg; /* cached write control register value */
+ u32 wcreg_spdif; /* S/PDIF setup */
+ u32 wcreg_spdif_stream; /* S/PDIF setup (temporary) */
+ u32 rcreg; /* cached read control register value */
+ u32 areg; /* cached additional register value */
+ u16 vol[2]; /* cached volume of analog output */
+
+ u8 rev; /* card revision number */
+
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ int playback_frlog; /* log2 of framesize */
+ int capture_frlog;
+
+ size_t playback_periodsize; /* in bytes, zero if not used */
+ size_t capture_periodsize; /* in bytes, zero if not used */
+
+ struct snd_card *card;
+ struct snd_pcm *spdif_pcm;
+ struct snd_pcm *adat_pcm;
+ struct pci_dev *pci;
+ struct snd_kcontrol *spdif_ctl;
+};
+
+static struct pci_device_id snd_rme96_ids[] = {
+ { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+ { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+ { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+ { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_rme96_ids);
+
+#define RME96_ISPLAYING(rme96) ((rme96)->wcreg & RME96_WCR_START)
+#define RME96_ISRECORDING(rme96) ((rme96)->wcreg & RME96_WCR_START_2)
+#define RME96_HAS_ANALOG_IN(rme96) ((rme96)->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST)
+#define RME96_HAS_ANALOG_OUT(rme96) ((rme96)->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PRO || \
+ (rme96)->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST)
+#define RME96_DAC_IS_1852(rme96) (RME96_HAS_ANALOG_OUT(rme96) && (rme96)->rev >= 4)
+#define RME96_DAC_IS_1855(rme96) (((rme96)->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST && (rme96)->rev < 4) || \
+ ((rme96)->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PRO && (rme96)->rev == 2))
+#define RME96_185X_MAX_OUT(rme96) ((1 << (RME96_DAC_IS_1852(rme96) ? RME96_AD1852_VOL_BITS : RME96_AD1855_VOL_BITS)) - 1)
+
+static int
+snd_rme96_playback_prepare(struct snd_pcm_substream *substream);
+
+static int
+snd_rme96_capture_prepare(struct snd_pcm_substream *substream);
+
+static int
+snd_rme96_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd);
+
+static int
+snd_rme96_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd);
+
+static snd_pcm_uframes_t
+snd_rme96_playback_pointer(struct snd_pcm_substream *substream);
+
+static snd_pcm_uframes_t
+snd_rme96_capture_pointer(struct snd_pcm_substream *substream);
+
+static void __devinit
+snd_rme96_proc_init(struct rme96 *rme96);
+
+static int
+snd_rme96_create_switches(struct snd_card *card,
+ struct rme96 *rme96);
+
+static int
+snd_rme96_getinputtype(struct rme96 *rme96);
+
+static inline unsigned int
+snd_rme96_playback_ptr(struct rme96 *rme96)
+{
+ return (readl(rme96->iobase + RME96_IO_GET_PLAY_POS)
+ & RME96_RCR_AUDIO_ADDR_MASK) >> rme96->playback_frlog;
+}
+
+static inline unsigned int
+snd_rme96_capture_ptr(struct rme96 *rme96)
+{
+ return (readl(rme96->iobase + RME96_IO_GET_REC_POS)
+ & RME96_RCR_AUDIO_ADDR_MASK) >> rme96->capture_frlog;
+}
+
+static int
+snd_rme96_playback_silence(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ count <<= rme96->playback_frlog;
+ pos <<= rme96->playback_frlog;
+ memset_io(rme96->iobase + RME96_IO_PLAY_BUFFER + pos,
+ 0, count);
+ return 0;
+}
+
+static int
+snd_rme96_playback_copy(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *src,
+ snd_pcm_uframes_t count)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ count <<= rme96->playback_frlog;
+ pos <<= rme96->playback_frlog;
+ copy_from_user_toio(rme96->iobase + RME96_IO_PLAY_BUFFER + pos, src,
+ count);
+ return 0;
+}
+
+static int
+snd_rme96_capture_copy(struct snd_pcm_substream *substream,
+ int channel, /* not used (interleaved data) */
+ snd_pcm_uframes_t pos,
+ void __user *dst,
+ snd_pcm_uframes_t count)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ count <<= rme96->capture_frlog;
+ pos <<= rme96->capture_frlog;
+ copy_to_user_fromio(dst, rme96->iobase + RME96_IO_REC_BUFFER + pos,
+ count);
+ return 0;
+}
+
+/*
+ * Digital output capabilities (S/PDIF)
+ */
+static struct snd_pcm_hardware snd_rme96_playback_spdif_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP_IOMEM |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = RME96_BUFFER_SIZE,
+ .period_bytes_min = RME96_SMALL_BLOCK_SIZE,
+ .period_bytes_max = RME96_LARGE_BLOCK_SIZE,
+ .periods_min = RME96_BUFFER_SIZE / RME96_LARGE_BLOCK_SIZE,
+ .periods_max = RME96_BUFFER_SIZE / RME96_SMALL_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+/*
+ * Digital input capabilities (S/PDIF)
+ */
+static struct snd_pcm_hardware snd_rme96_capture_spdif_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP_IOMEM |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = RME96_BUFFER_SIZE,
+ .period_bytes_min = RME96_SMALL_BLOCK_SIZE,
+ .period_bytes_max = RME96_LARGE_BLOCK_SIZE,
+ .periods_min = RME96_BUFFER_SIZE / RME96_LARGE_BLOCK_SIZE,
+ .periods_max = RME96_BUFFER_SIZE / RME96_SMALL_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+/*
+ * Digital output capabilities (ADAT)
+ */
+static struct snd_pcm_hardware snd_rme96_playback_adat_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP_IOMEM |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 8,
+ .channels_max = 8,
+ .buffer_bytes_max = RME96_BUFFER_SIZE,
+ .period_bytes_min = RME96_SMALL_BLOCK_SIZE,
+ .period_bytes_max = RME96_LARGE_BLOCK_SIZE,
+ .periods_min = RME96_BUFFER_SIZE / RME96_LARGE_BLOCK_SIZE,
+ .periods_max = RME96_BUFFER_SIZE / RME96_SMALL_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+/*
+ * Digital input capabilities (ADAT)
+ */
+static struct snd_pcm_hardware snd_rme96_capture_adat_info =
+{
+ .info = (SNDRV_PCM_INFO_MMAP_IOMEM |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 8,
+ .channels_max = 8,
+ .buffer_bytes_max = RME96_BUFFER_SIZE,
+ .period_bytes_min = RME96_SMALL_BLOCK_SIZE,
+ .period_bytes_max = RME96_LARGE_BLOCK_SIZE,
+ .periods_min = RME96_BUFFER_SIZE / RME96_LARGE_BLOCK_SIZE,
+ .periods_max = RME96_BUFFER_SIZE / RME96_SMALL_BLOCK_SIZE,
+ .fifo_size = 0,
+};
+
+/*
+ * The CDATA, CCLK and CLATCH bits can be used to write to the SPI interface
+ * of the AD1852 or AD1852 D/A converter on the board. CDATA must be set up
+ * on the falling edge of CCLK and be stable on the rising edge. The rising
+ * edge of CLATCH after the last data bit clocks in the whole data word.
+ * A fast processor could probably drive the SPI interface faster than the
+ * DAC can handle (3MHz for the 1855, unknown for the 1852). The udelay(1)
+ * limits the data rate to 500KHz and only causes a delay of 33 microsecs.
+ *
+ * NOTE: increased delay from 1 to 10, since there where problems setting
+ * the volume.
+ */
+static void
+snd_rme96_write_SPI(struct rme96 *rme96, u16 val)
+{
+ int i;
+
+ for (i = 0; i < 16; i++) {
+ if (val & 0x8000) {
+ rme96->areg |= RME96_AR_CDATA;
+ } else {
+ rme96->areg &= ~RME96_AR_CDATA;
+ }
+ rme96->areg &= ~(RME96_AR_CCLK | RME96_AR_CLATCH);
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ udelay(10);
+ rme96->areg |= RME96_AR_CCLK;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ udelay(10);
+ val <<= 1;
+ }
+ rme96->areg &= ~(RME96_AR_CCLK | RME96_AR_CDATA);
+ rme96->areg |= RME96_AR_CLATCH;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ udelay(10);
+ rme96->areg &= ~RME96_AR_CLATCH;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+}
+
+static void
+snd_rme96_apply_dac_volume(struct rme96 *rme96)
+{
+ if (RME96_DAC_IS_1852(rme96)) {
+ snd_rme96_write_SPI(rme96, (rme96->vol[0] << 2) | 0x0);
+ snd_rme96_write_SPI(rme96, (rme96->vol[1] << 2) | 0x2);
+ } else if (RME96_DAC_IS_1855(rme96)) {
+ snd_rme96_write_SPI(rme96, (rme96->vol[0] & 0x3FF) | 0x000);
+ snd_rme96_write_SPI(rme96, (rme96->vol[1] & 0x3FF) | 0x400);
+ }
+}
+
+static void
+snd_rme96_reset_dac(struct rme96 *rme96)
+{
+ writel(rme96->wcreg | RME96_WCR_PD,
+ rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+}
+
+static int
+snd_rme96_getmontracks(struct rme96 *rme96)
+{
+ return ((rme96->wcreg >> RME96_WCR_BITPOS_MONITOR_0) & 1) +
+ (((rme96->wcreg >> RME96_WCR_BITPOS_MONITOR_1) & 1) << 1);
+}
+
+static int
+snd_rme96_setmontracks(struct rme96 *rme96,
+ int montracks)
+{
+ if (montracks & 1) {
+ rme96->wcreg |= RME96_WCR_MONITOR_0;
+ } else {
+ rme96->wcreg &= ~RME96_WCR_MONITOR_0;
+ }
+ if (montracks & 2) {
+ rme96->wcreg |= RME96_WCR_MONITOR_1;
+ } else {
+ rme96->wcreg &= ~RME96_WCR_MONITOR_1;
+ }
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static int
+snd_rme96_getattenuation(struct rme96 *rme96)
+{
+ return ((rme96->wcreg >> RME96_WCR_BITPOS_GAIN_0) & 1) +
+ (((rme96->wcreg >> RME96_WCR_BITPOS_GAIN_1) & 1) << 1);
+}
+
+static int
+snd_rme96_setattenuation(struct rme96 *rme96,
+ int attenuation)
+{
+ switch (attenuation) {
+ case 0:
+ rme96->wcreg = (rme96->wcreg & ~RME96_WCR_GAIN_0) &
+ ~RME96_WCR_GAIN_1;
+ break;
+ case 1:
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_GAIN_0) &
+ ~RME96_WCR_GAIN_1;
+ break;
+ case 2:
+ rme96->wcreg = (rme96->wcreg & ~RME96_WCR_GAIN_0) |
+ RME96_WCR_GAIN_1;
+ break;
+ case 3:
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_GAIN_0) |
+ RME96_WCR_GAIN_1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static int
+snd_rme96_capture_getrate(struct rme96 *rme96,
+ int *is_adat)
+{
+ int n, rate;
+
+ *is_adat = 0;
+ if (rme96->areg & RME96_AR_ANALOG) {
+ /* Analog input, overrides S/PDIF setting */
+ n = ((rme96->areg >> RME96_AR_BITPOS_F0) & 1) +
+ (((rme96->areg >> RME96_AR_BITPOS_F1) & 1) << 1);
+ switch (n) {
+ case 1:
+ rate = 32000;
+ break;
+ case 2:
+ rate = 44100;
+ break;
+ case 3:
+ rate = 48000;
+ break;
+ default:
+ return -1;
+ }
+ return (rme96->areg & RME96_AR_BITPOS_F2) ? rate << 1 : rate;
+ }
+
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ if (rme96->rcreg & RME96_RCR_LOCK) {
+ /* ADAT rate */
+ *is_adat = 1;
+ if (rme96->rcreg & RME96_RCR_T_OUT) {
+ return 48000;
+ }
+ return 44100;
+ }
+
+ if (rme96->rcreg & RME96_RCR_VERF) {
+ return -1;
+ }
+
+ /* S/PDIF rate */
+ n = ((rme96->rcreg >> RME96_RCR_BITPOS_F0) & 1) +
+ (((rme96->rcreg >> RME96_RCR_BITPOS_F1) & 1) << 1) +
+ (((rme96->rcreg >> RME96_RCR_BITPOS_F2) & 1) << 2);
+
+ switch (n) {
+ case 0:
+ if (rme96->rcreg & RME96_RCR_T_OUT) {
+ return 64000;
+ }
+ return -1;
+ case 3: return 96000;
+ case 4: return 88200;
+ case 5: return 48000;
+ case 6: return 44100;
+ case 7: return 32000;
+ default:
+ break;
+ }
+ return -1;
+}
+
+static int
+snd_rme96_playback_getrate(struct rme96 *rme96)
+{
+ int rate, dummy;
+
+ if (!(rme96->wcreg & RME96_WCR_MASTER) &&
+ snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
+ (rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0)
+ {
+ /* slave clock */
+ return rate;
+ }
+ rate = ((rme96->wcreg >> RME96_WCR_BITPOS_FREQ_0) & 1) +
+ (((rme96->wcreg >> RME96_WCR_BITPOS_FREQ_1) & 1) << 1);
+ switch (rate) {
+ case 1:
+ rate = 32000;
+ break;
+ case 2:
+ rate = 44100;
+ break;
+ case 3:
+ rate = 48000;
+ break;
+ default:
+ return -1;
+ }
+ return (rme96->wcreg & RME96_WCR_DS) ? rate << 1 : rate;
+}
+
+static int
+snd_rme96_playback_setrate(struct rme96 *rme96,
+ int rate)
+{
+ int ds;
+
+ ds = rme96->wcreg & RME96_WCR_DS;
+ switch (rate) {
+ case 32000:
+ rme96->wcreg &= ~RME96_WCR_DS;
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_FREQ_0) &
+ ~RME96_WCR_FREQ_1;
+ break;
+ case 44100:
+ rme96->wcreg &= ~RME96_WCR_DS;
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_FREQ_1) &
+ ~RME96_WCR_FREQ_0;
+ break;
+ case 48000:
+ rme96->wcreg &= ~RME96_WCR_DS;
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_FREQ_0) |
+ RME96_WCR_FREQ_1;
+ break;
+ case 64000:
+ rme96->wcreg |= RME96_WCR_DS;
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_FREQ_0) &
+ ~RME96_WCR_FREQ_1;
+ break;
+ case 88200:
+ rme96->wcreg |= RME96_WCR_DS;
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_FREQ_1) &
+ ~RME96_WCR_FREQ_0;
+ break;
+ case 96000:
+ rme96->wcreg |= RME96_WCR_DS;
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_FREQ_0) |
+ RME96_WCR_FREQ_1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ if ((!ds && rme96->wcreg & RME96_WCR_DS) ||
+ (ds && !(rme96->wcreg & RME96_WCR_DS)))
+ {
+ /* change to/from double-speed: reset the DAC (if available) */
+ snd_rme96_reset_dac(rme96);
+ } else {
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ }
+ return 0;
+}
+
+static int
+snd_rme96_capture_analog_setrate(struct rme96 *rme96,
+ int rate)
+{
+ switch (rate) {
+ case 32000:
+ rme96->areg = ((rme96->areg | RME96_AR_FREQPAD_0) &
+ ~RME96_AR_FREQPAD_1) & ~RME96_AR_FREQPAD_2;
+ break;
+ case 44100:
+ rme96->areg = ((rme96->areg & ~RME96_AR_FREQPAD_0) |
+ RME96_AR_FREQPAD_1) & ~RME96_AR_FREQPAD_2;
+ break;
+ case 48000:
+ rme96->areg = ((rme96->areg | RME96_AR_FREQPAD_0) |
+ RME96_AR_FREQPAD_1) & ~RME96_AR_FREQPAD_2;
+ break;
+ case 64000:
+ if (rme96->rev < 4) {
+ return -EINVAL;
+ }
+ rme96->areg = ((rme96->areg | RME96_AR_FREQPAD_0) &
+ ~RME96_AR_FREQPAD_1) | RME96_AR_FREQPAD_2;
+ break;
+ case 88200:
+ if (rme96->rev < 4) {
+ return -EINVAL;
+ }
+ rme96->areg = ((rme96->areg & ~RME96_AR_FREQPAD_0) |
+ RME96_AR_FREQPAD_1) | RME96_AR_FREQPAD_2;
+ break;
+ case 96000:
+ rme96->areg = ((rme96->areg | RME96_AR_FREQPAD_0) |
+ RME96_AR_FREQPAD_1) | RME96_AR_FREQPAD_2;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ return 0;
+}
+
+static int
+snd_rme96_setclockmode(struct rme96 *rme96,
+ int mode)
+{
+ switch (mode) {
+ case RME96_CLOCKMODE_SLAVE:
+ /* AutoSync */
+ rme96->wcreg &= ~RME96_WCR_MASTER;
+ rme96->areg &= ~RME96_AR_WSEL;
+ break;
+ case RME96_CLOCKMODE_MASTER:
+ /* Internal */
+ rme96->wcreg |= RME96_WCR_MASTER;
+ rme96->areg &= ~RME96_AR_WSEL;
+ break;
+ case RME96_CLOCKMODE_WORDCLOCK:
+ /* Word clock is a master mode */
+ rme96->wcreg |= RME96_WCR_MASTER;
+ rme96->areg |= RME96_AR_WSEL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ return 0;
+}
+
+static int
+snd_rme96_getclockmode(struct rme96 *rme96)
+{
+ if (rme96->areg & RME96_AR_WSEL) {
+ return RME96_CLOCKMODE_WORDCLOCK;
+ }
+ return (rme96->wcreg & RME96_WCR_MASTER) ? RME96_CLOCKMODE_MASTER :
+ RME96_CLOCKMODE_SLAVE;
+}
+
+static int
+snd_rme96_setinputtype(struct rme96 *rme96,
+ int type)
+{
+ int n;
+
+ switch (type) {
+ case RME96_INPUT_OPTICAL:
+ rme96->wcreg = (rme96->wcreg & ~RME96_WCR_INP_0) &
+ ~RME96_WCR_INP_1;
+ break;
+ case RME96_INPUT_COAXIAL:
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_INP_0) &
+ ~RME96_WCR_INP_1;
+ break;
+ case RME96_INPUT_INTERNAL:
+ rme96->wcreg = (rme96->wcreg & ~RME96_WCR_INP_0) |
+ RME96_WCR_INP_1;
+ break;
+ case RME96_INPUT_XLR:
+ if ((rme96->pci->device != PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST &&
+ rme96->pci->device != PCI_DEVICE_ID_RME_DIGI96_8_PRO) ||
+ (rme96->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST &&
+ rme96->rev > 4))
+ {
+ /* Only Digi96/8 PRO and Digi96/8 PAD supports XLR */
+ return -EINVAL;
+ }
+ rme96->wcreg = (rme96->wcreg | RME96_WCR_INP_0) |
+ RME96_WCR_INP_1;
+ break;
+ case RME96_INPUT_ANALOG:
+ if (!RME96_HAS_ANALOG_IN(rme96)) {
+ return -EINVAL;
+ }
+ rme96->areg |= RME96_AR_ANALOG;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ if (rme96->rev < 4) {
+ /*
+ * Revision less than 004 does not support 64 and
+ * 88.2 kHz
+ */
+ if (snd_rme96_capture_getrate(rme96, &n) == 88200) {
+ snd_rme96_capture_analog_setrate(rme96, 44100);
+ }
+ if (snd_rme96_capture_getrate(rme96, &n) == 64000) {
+ snd_rme96_capture_analog_setrate(rme96, 32000);
+ }
+ }
+ return 0;
+ default:
+ return -EINVAL;
+ }
+ if (type != RME96_INPUT_ANALOG && RME96_HAS_ANALOG_IN(rme96)) {
+ rme96->areg &= ~RME96_AR_ANALOG;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ }
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static int
+snd_rme96_getinputtype(struct rme96 *rme96)
+{
+ if (rme96->areg & RME96_AR_ANALOG) {
+ return RME96_INPUT_ANALOG;
+ }
+ return ((rme96->wcreg >> RME96_WCR_BITPOS_INP_0) & 1) +
+ (((rme96->wcreg >> RME96_WCR_BITPOS_INP_1) & 1) << 1);
+}
+
+static void
+snd_rme96_setframelog(struct rme96 *rme96,
+ int n_channels,
+ int is_playback)
+{
+ int frlog;
+
+ if (n_channels == 2) {
+ frlog = 1;
+ } else {
+ /* assume 8 channels */
+ frlog = 3;
+ }
+ if (is_playback) {
+ frlog += (rme96->wcreg & RME96_WCR_MODE24) ? 2 : 1;
+ rme96->playback_frlog = frlog;
+ } else {
+ frlog += (rme96->wcreg & RME96_WCR_MODE24_2) ? 2 : 1;
+ rme96->capture_frlog = frlog;
+ }
+}
+
+static int
+snd_rme96_playback_setformat(struct rme96 *rme96,
+ int format)
+{
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ rme96->wcreg &= ~RME96_WCR_MODE24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ rme96->wcreg |= RME96_WCR_MODE24;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static int
+snd_rme96_capture_setformat(struct rme96 *rme96,
+ int format)
+{
+ switch (format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ rme96->wcreg &= ~RME96_WCR_MODE24_2;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ rme96->wcreg |= RME96_WCR_MODE24_2;
+ break;
+ default:
+ return -EINVAL;
+ }
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ return 0;
+}
+
+static void
+snd_rme96_set_period_properties(struct rme96 *rme96,
+ size_t period_bytes)
+{
+ switch (period_bytes) {
+ case RME96_LARGE_BLOCK_SIZE:
+ rme96->wcreg &= ~RME96_WCR_ISEL;
+ break;
+ case RME96_SMALL_BLOCK_SIZE:
+ rme96->wcreg |= RME96_WCR_ISEL;
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ rme96->wcreg &= ~RME96_WCR_IDIS;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+}
+
+static int
+snd_rme96_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err, rate, dummy;
+
+ runtime->dma_area = (void __force *)(rme96->iobase +
+ RME96_IO_PLAY_BUFFER);
+ runtime->dma_addr = rme96->port + RME96_IO_PLAY_BUFFER;
+ runtime->dma_bytes = RME96_BUFFER_SIZE;
+
+ spin_lock_irq(&rme96->lock);
+ if (!(rme96->wcreg & RME96_WCR_MASTER) &&
+ snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
+ (rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0)
+ {
+ /* slave clock */
+ if ((int)params_rate(params) != rate) {
+ spin_unlock_irq(&rme96->lock);
+ return -EIO;
+ }
+ } else if ((err = snd_rme96_playback_setrate(rme96, params_rate(params))) < 0) {
+ spin_unlock_irq(&rme96->lock);
+ return err;
+ }
+ if ((err = snd_rme96_playback_setformat(rme96, params_format(params))) < 0) {
+ spin_unlock_irq(&rme96->lock);
+ return err;
+ }
+ snd_rme96_setframelog(rme96, params_channels(params), 1);
+ if (rme96->capture_periodsize != 0) {
+ if (params_period_size(params) << rme96->playback_frlog !=
+ rme96->capture_periodsize)
+ {
+ spin_unlock_irq(&rme96->lock);
+ return -EBUSY;
+ }
+ }
+ rme96->playback_periodsize =
+ params_period_size(params) << rme96->playback_frlog;
+ snd_rme96_set_period_properties(rme96, rme96->playback_periodsize);
+ /* S/PDIF setup */
+ if ((rme96->wcreg & RME96_WCR_ADAT) == 0) {
+ rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP);
+ writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ }
+ spin_unlock_irq(&rme96->lock);
+
+ return 0;
+}
+
+static int
+snd_rme96_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err, isadat, rate;
+
+ runtime->dma_area = (void __force *)(rme96->iobase +
+ RME96_IO_REC_BUFFER);
+ runtime->dma_addr = rme96->port + RME96_IO_REC_BUFFER;
+ runtime->dma_bytes = RME96_BUFFER_SIZE;
+
+ spin_lock_irq(&rme96->lock);
+ if ((err = snd_rme96_capture_setformat(rme96, params_format(params))) < 0) {
+ spin_unlock_irq(&rme96->lock);
+ return err;
+ }
+ if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
+ if ((err = snd_rme96_capture_analog_setrate(rme96,
+ params_rate(params))) < 0)
+ {
+ spin_unlock_irq(&rme96->lock);
+ return err;
+ }
+ } else if ((rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0) {
+ if ((int)params_rate(params) != rate) {
+ spin_unlock_irq(&rme96->lock);
+ return -EIO;
+ }
+ if ((isadat && runtime->hw.channels_min == 2) ||
+ (!isadat && runtime->hw.channels_min == 8))
+ {
+ spin_unlock_irq(&rme96->lock);
+ return -EIO;
+ }
+ }
+ snd_rme96_setframelog(rme96, params_channels(params), 0);
+ if (rme96->playback_periodsize != 0) {
+ if (params_period_size(params) << rme96->capture_frlog !=
+ rme96->playback_periodsize)
+ {
+ spin_unlock_irq(&rme96->lock);
+ return -EBUSY;
+ }
+ }
+ rme96->capture_periodsize =
+ params_period_size(params) << rme96->capture_frlog;
+ snd_rme96_set_period_properties(rme96, rme96->capture_periodsize);
+ spin_unlock_irq(&rme96->lock);
+
+ return 0;
+}
+
+static void
+snd_rme96_playback_start(struct rme96 *rme96,
+ int from_pause)
+{
+ if (!from_pause) {
+ writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
+ }
+
+ rme96->wcreg |= RME96_WCR_START;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+}
+
+static void
+snd_rme96_capture_start(struct rme96 *rme96,
+ int from_pause)
+{
+ if (!from_pause) {
+ writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
+ }
+
+ rme96->wcreg |= RME96_WCR_START_2;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+}
+
+static void
+snd_rme96_playback_stop(struct rme96 *rme96)
+{
+ /*
+ * Check if there is an unconfirmed IRQ, if so confirm it, or else
+ * the hardware will not stop generating interrupts
+ */
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ if (rme96->rcreg & RME96_RCR_IRQ) {
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
+ }
+ rme96->wcreg &= ~RME96_WCR_START;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+}
+
+static void
+snd_rme96_capture_stop(struct rme96 *rme96)
+{
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ if (rme96->rcreg & RME96_RCR_IRQ_2) {
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+ }
+ rme96->wcreg &= ~RME96_WCR_START_2;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+}
+
+static irqreturn_t
+snd_rme96_interrupt(int irq,
+ void *dev_id)
+{
+ struct rme96 *rme96 = (struct rme96 *)dev_id;
+
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ /* fastpath out, to ease interrupt sharing */
+ if (!((rme96->rcreg & RME96_RCR_IRQ) ||
+ (rme96->rcreg & RME96_RCR_IRQ_2)))
+ {
+ return IRQ_NONE;
+ }
+
+ if (rme96->rcreg & RME96_RCR_IRQ) {
+ /* playback */
+ snd_pcm_period_elapsed(rme96->playback_substream);
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
+ }
+ if (rme96->rcreg & RME96_RCR_IRQ_2) {
+ /* capture */
+ snd_pcm_period_elapsed(rme96->capture_substream);
+ writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+ }
+ return IRQ_HANDLED;
+}
+
+static unsigned int period_bytes[] = { RME96_SMALL_BLOCK_SIZE, RME96_LARGE_BLOCK_SIZE };
+
+static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = {
+ .count = ARRAY_SIZE(period_bytes),
+ .list = period_bytes,
+ .mask = 0
+};
+
+static void
+rme96_set_buffer_size_constraint(struct rme96 *rme96,
+ struct snd_pcm_runtime *runtime)
+{
+ unsigned int size;
+
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
+ if ((size = rme96->playback_periodsize) != 0 ||
+ (size = rme96->capture_periodsize) != 0)
+ snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ size, size);
+ else
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ &hw_constraints_period_bytes);
+}
+
+static int
+snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
+{
+ int rate, dummy;
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irq(&rme96->lock);
+ if (rme96->playback_substream != NULL) {
+ spin_unlock_irq(&rme96->lock);
+ return -EBUSY;
+ }
+ rme96->wcreg &= ~RME96_WCR_ADAT;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ rme96->playback_substream = substream;
+ spin_unlock_irq(&rme96->lock);
+
+ runtime->hw = snd_rme96_playback_spdif_info;
+ if (!(rme96->wcreg & RME96_WCR_MASTER) &&
+ snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
+ (rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0)
+ {
+ /* slave clock */
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+ rme96_set_buffer_size_constraint(rme96, runtime);
+
+ rme96->wcreg_spdif_stream = rme96->wcreg_spdif;
+ rme96->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(rme96->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &rme96->spdif_ctl->id);
+ return 0;
+}
+
+static int
+snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
+{
+ int isadat, rate;
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_rme96_capture_spdif_info;
+ if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
+ (rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
+ {
+ if (isadat) {
+ return -EIO;
+ }
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+
+ spin_lock_irq(&rme96->lock);
+ if (rme96->capture_substream != NULL) {
+ spin_unlock_irq(&rme96->lock);
+ return -EBUSY;
+ }
+ rme96->capture_substream = substream;
+ spin_unlock_irq(&rme96->lock);
+
+ rme96_set_buffer_size_constraint(rme96, runtime);
+ return 0;
+}
+
+static int
+snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
+{
+ int rate, dummy;
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irq(&rme96->lock);
+ if (rme96->playback_substream != NULL) {
+ spin_unlock_irq(&rme96->lock);
+ return -EBUSY;
+ }
+ rme96->wcreg |= RME96_WCR_ADAT;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ rme96->playback_substream = substream;
+ spin_unlock_irq(&rme96->lock);
+
+ runtime->hw = snd_rme96_playback_adat_info;
+ if (!(rme96->wcreg & RME96_WCR_MASTER) &&
+ snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
+ (rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0)
+ {
+ /* slave clock */
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+ rme96_set_buffer_size_constraint(rme96, runtime);
+ return 0;
+}
+
+static int
+snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
+{
+ int isadat, rate;
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_rme96_capture_adat_info;
+ if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
+ /* makes no sense to use analog input. Note that analog
+ expension cards AEB4/8-I are RME96_INPUT_INTERNAL */
+ return -EIO;
+ }
+ if ((rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0) {
+ if (!isadat) {
+ return -EIO;
+ }
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+
+ spin_lock_irq(&rme96->lock);
+ if (rme96->capture_substream != NULL) {
+ spin_unlock_irq(&rme96->lock);
+ return -EBUSY;
+ }
+ rme96->capture_substream = substream;
+ spin_unlock_irq(&rme96->lock);
+
+ rme96_set_buffer_size_constraint(rme96, runtime);
+ return 0;
+}
+
+static int
+snd_rme96_playback_close(struct snd_pcm_substream *substream)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ int spdif = 0;
+
+ spin_lock_irq(&rme96->lock);
+ if (RME96_ISPLAYING(rme96)) {
+ snd_rme96_playback_stop(rme96);
+ }
+ rme96->playback_substream = NULL;
+ rme96->playback_periodsize = 0;
+ spdif = (rme96->wcreg & RME96_WCR_ADAT) == 0;
+ spin_unlock_irq(&rme96->lock);
+ if (spdif) {
+ rme96->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(rme96->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &rme96->spdif_ctl->id);
+ }
+ return 0;
+}
+
+static int
+snd_rme96_capture_close(struct snd_pcm_substream *substream)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme96->lock);
+ if (RME96_ISRECORDING(rme96)) {
+ snd_rme96_capture_stop(rme96);
+ }
+ rme96->capture_substream = NULL;
+ rme96->capture_periodsize = 0;
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+
+static int
+snd_rme96_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme96->lock);
+ if (RME96_ISPLAYING(rme96)) {
+ snd_rme96_playback_stop(rme96);
+ }
+ writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+
+static int
+snd_rme96_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme96->lock);
+ if (RME96_ISRECORDING(rme96)) {
+ snd_rme96_capture_stop(rme96);
+ }
+ writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+
+static int
+snd_rme96_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (!RME96_ISPLAYING(rme96)) {
+ if (substream != rme96->playback_substream) {
+ return -EBUSY;
+ }
+ snd_rme96_playback_start(rme96, 0);
+ }
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (RME96_ISPLAYING(rme96)) {
+ if (substream != rme96->playback_substream) {
+ return -EBUSY;
+ }
+ snd_rme96_playback_stop(rme96);
+ }
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (RME96_ISPLAYING(rme96)) {
+ snd_rme96_playback_stop(rme96);
+ }
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!RME96_ISPLAYING(rme96)) {
+ snd_rme96_playback_start(rme96, 1);
+ }
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int
+snd_rme96_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (!RME96_ISRECORDING(rme96)) {
+ if (substream != rme96->capture_substream) {
+ return -EBUSY;
+ }
+ snd_rme96_capture_start(rme96, 0);
+ }
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (RME96_ISRECORDING(rme96)) {
+ if (substream != rme96->capture_substream) {
+ return -EBUSY;
+ }
+ snd_rme96_capture_stop(rme96);
+ }
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (RME96_ISRECORDING(rme96)) {
+ snd_rme96_capture_stop(rme96);
+ }
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!RME96_ISRECORDING(rme96)) {
+ snd_rme96_capture_start(rme96, 1);
+ }
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+snd_rme96_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ return snd_rme96_playback_ptr(rme96);
+}
+
+static snd_pcm_uframes_t
+snd_rme96_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+ return snd_rme96_capture_ptr(rme96);
+}
+
+static struct snd_pcm_ops snd_rme96_playback_spdif_ops = {
+ .open = snd_rme96_playback_spdif_open,
+ .close = snd_rme96_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme96_playback_hw_params,
+ .prepare = snd_rme96_playback_prepare,
+ .trigger = snd_rme96_playback_trigger,
+ .pointer = snd_rme96_playback_pointer,
+ .copy = snd_rme96_playback_copy,
+ .silence = snd_rme96_playback_silence,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static struct snd_pcm_ops snd_rme96_capture_spdif_ops = {
+ .open = snd_rme96_capture_spdif_open,
+ .close = snd_rme96_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme96_capture_hw_params,
+ .prepare = snd_rme96_capture_prepare,
+ .trigger = snd_rme96_capture_trigger,
+ .pointer = snd_rme96_capture_pointer,
+ .copy = snd_rme96_capture_copy,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static struct snd_pcm_ops snd_rme96_playback_adat_ops = {
+ .open = snd_rme96_playback_adat_open,
+ .close = snd_rme96_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme96_playback_hw_params,
+ .prepare = snd_rme96_playback_prepare,
+ .trigger = snd_rme96_playback_trigger,
+ .pointer = snd_rme96_playback_pointer,
+ .copy = snd_rme96_playback_copy,
+ .silence = snd_rme96_playback_silence,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static struct snd_pcm_ops snd_rme96_capture_adat_ops = {
+ .open = snd_rme96_capture_adat_open,
+ .close = snd_rme96_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_rme96_capture_hw_params,
+ .prepare = snd_rme96_capture_prepare,
+ .trigger = snd_rme96_capture_trigger,
+ .pointer = snd_rme96_capture_pointer,
+ .copy = snd_rme96_capture_copy,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static void
+snd_rme96_free(void *private_data)
+{
+ struct rme96 *rme96 = (struct rme96 *)private_data;
+
+ if (rme96 == NULL) {
+ return;
+ }
+ if (rme96->irq >= 0) {
+ snd_rme96_playback_stop(rme96);
+ snd_rme96_capture_stop(rme96);
+ rme96->areg &= ~RME96_AR_DAC_EN;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ free_irq(rme96->irq, (void *)rme96);
+ rme96->irq = -1;
+ }
+ if (rme96->iobase) {
+ iounmap(rme96->iobase);
+ rme96->iobase = NULL;
+ }
+ if (rme96->port) {
+ pci_release_regions(rme96->pci);
+ rme96->port = 0;
+ }
+ pci_disable_device(rme96->pci);
+}
+
+static void
+snd_rme96_free_spdif_pcm(struct snd_pcm *pcm)
+{
+ struct rme96 *rme96 = (struct rme96 *) pcm->private_data;
+ rme96->spdif_pcm = NULL;
+}
+
+static void
+snd_rme96_free_adat_pcm(struct snd_pcm *pcm)
+{
+ struct rme96 *rme96 = (struct rme96 *) pcm->private_data;
+ rme96->adat_pcm = NULL;
+}
+
+static int __devinit
+snd_rme96_create(struct rme96 *rme96)
+{
+ struct pci_dev *pci = rme96->pci;
+ int err;
+
+ rme96->irq = -1;
+ spin_lock_init(&rme96->lock);
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ if ((err = pci_request_regions(pci, "RME96")) < 0)
+ return err;
+ rme96->port = pci_resource_start(rme96->pci, 0);
+
+ rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE);
+ if (!rme96->iobase) {
+ snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1);
+ return -ENOMEM;
+ }
+
+ if (request_irq(pci->irq, snd_rme96_interrupt, IRQF_SHARED,
+ "RME96", rme96)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ return -EBUSY;
+ }
+ rme96->irq = pci->irq;
+
+ /* read the card's revision number */
+ pci_read_config_byte(pci, 8, &rme96->rev);
+
+ /* set up ALSA pcm device for S/PDIF */
+ if ((err = snd_pcm_new(rme96->card, "Digi96 IEC958", 0,
+ 1, 1, &rme96->spdif_pcm)) < 0)
+ {
+ return err;
+ }
+ rme96->spdif_pcm->private_data = rme96;
+ rme96->spdif_pcm->private_free = snd_rme96_free_spdif_pcm;
+ strcpy(rme96->spdif_pcm->name, "Digi96 IEC958");
+ snd_pcm_set_ops(rme96->spdif_pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_rme96_playback_spdif_ops);
+ snd_pcm_set_ops(rme96->spdif_pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme96_capture_spdif_ops);
+
+ rme96->spdif_pcm->info_flags = 0;
+
+ /* set up ALSA pcm device for ADAT */
+ if (pci->device == PCI_DEVICE_ID_RME_DIGI96) {
+ /* ADAT is not available on the base model */
+ rme96->adat_pcm = NULL;
+ } else {
+ if ((err = snd_pcm_new(rme96->card, "Digi96 ADAT", 1,
+ 1, 1, &rme96->adat_pcm)) < 0)
+ {
+ return err;
+ }
+ rme96->adat_pcm->private_data = rme96;
+ rme96->adat_pcm->private_free = snd_rme96_free_adat_pcm;
+ strcpy(rme96->adat_pcm->name, "Digi96 ADAT");
+ snd_pcm_set_ops(rme96->adat_pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_rme96_playback_adat_ops);
+ snd_pcm_set_ops(rme96->adat_pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme96_capture_adat_ops);
+
+ rme96->adat_pcm->info_flags = 0;
+ }
+
+ rme96->playback_periodsize = 0;
+ rme96->capture_periodsize = 0;
+
+ /* make sure playback/capture is stopped, if by some reason active */
+ snd_rme96_playback_stop(rme96);
+ snd_rme96_capture_stop(rme96);
+
+ /* set default values in registers */
+ rme96->wcreg =
+ RME96_WCR_FREQ_1 | /* set 44.1 kHz playback */
+ RME96_WCR_SEL | /* normal playback */
+ RME96_WCR_MASTER | /* set to master clock mode */
+ RME96_WCR_INP_0; /* set coaxial input */
+
+ rme96->areg = RME96_AR_FREQPAD_1; /* set 44.1 kHz analog capture */
+
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+ /* reset the ADC */
+ writel(rme96->areg | RME96_AR_PD2,
+ rme96->iobase + RME96_IO_ADDITIONAL_REG);
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+ /* reset and enable the DAC (order is important). */
+ snd_rme96_reset_dac(rme96);
+ rme96->areg |= RME96_AR_DAC_EN;
+ writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+ /* reset playback and record buffer pointers */
+ writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
+ writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
+
+ /* reset volume */
+ rme96->vol[0] = rme96->vol[1] = 0;
+ if (RME96_HAS_ANALOG_OUT(rme96)) {
+ snd_rme96_apply_dac_volume(rme96);
+ }
+
+ /* init switch interface */
+ if ((err = snd_rme96_create_switches(rme96->card, rme96)) < 0) {
+ return err;
+ }
+
+ /* init proc interface */
+ snd_rme96_proc_init(rme96);
+
+ return 0;
+}
+
+/*
+ * proc interface
+ */
+
+static void
+snd_rme96_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ int n;
+ struct rme96 *rme96 = (struct rme96 *)entry->private_data;
+
+ rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+
+ snd_iprintf(buffer, rme96->card->longname);
+ snd_iprintf(buffer, " (index #%d)\n", rme96->card->number + 1);
+
+ snd_iprintf(buffer, "\nGeneral settings\n");
+ if (rme96->wcreg & RME96_WCR_IDIS) {
+ snd_iprintf(buffer, " period size: N/A (interrupts "
+ "disabled)\n");
+ } else if (rme96->wcreg & RME96_WCR_ISEL) {
+ snd_iprintf(buffer, " period size: 2048 bytes\n");
+ } else {
+ snd_iprintf(buffer, " period size: 8192 bytes\n");
+ }
+ snd_iprintf(buffer, "\nInput settings\n");
+ switch (snd_rme96_getinputtype(rme96)) {
+ case RME96_INPUT_OPTICAL:
+ snd_iprintf(buffer, " input: optical");
+ break;
+ case RME96_INPUT_COAXIAL:
+ snd_iprintf(buffer, " input: coaxial");
+ break;
+ case RME96_INPUT_INTERNAL:
+ snd_iprintf(buffer, " input: internal");
+ break;
+ case RME96_INPUT_XLR:
+ snd_iprintf(buffer, " input: XLR");
+ break;
+ case RME96_INPUT_ANALOG:
+ snd_iprintf(buffer, " input: analog");
+ break;
+ }
+ if (snd_rme96_capture_getrate(rme96, &n) < 0) {
+ snd_iprintf(buffer, "\n sample rate: no valid signal\n");
+ } else {
+ if (n) {
+ snd_iprintf(buffer, " (8 channels)\n");
+ } else {
+ snd_iprintf(buffer, " (2 channels)\n");
+ }
+ snd_iprintf(buffer, " sample rate: %d Hz\n",
+ snd_rme96_capture_getrate(rme96, &n));
+ }
+ if (rme96->wcreg & RME96_WCR_MODE24_2) {
+ snd_iprintf(buffer, " sample format: 24 bit\n");
+ } else {
+ snd_iprintf(buffer, " sample format: 16 bit\n");
+ }
+
+ snd_iprintf(buffer, "\nOutput settings\n");
+ if (rme96->wcreg & RME96_WCR_SEL) {
+ snd_iprintf(buffer, " output signal: normal playback\n");
+ } else {
+ snd_iprintf(buffer, " output signal: same as input\n");
+ }
+ snd_iprintf(buffer, " sample rate: %d Hz\n",
+ snd_rme96_playback_getrate(rme96));
+ if (rme96->wcreg & RME96_WCR_MODE24) {
+ snd_iprintf(buffer, " sample format: 24 bit\n");
+ } else {
+ snd_iprintf(buffer, " sample format: 16 bit\n");
+ }
+ if (rme96->areg & RME96_AR_WSEL) {
+ snd_iprintf(buffer, " sample clock source: word clock\n");
+ } else if (rme96->wcreg & RME96_WCR_MASTER) {
+ snd_iprintf(buffer, " sample clock source: internal\n");
+ } else if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
+ snd_iprintf(buffer, " sample clock source: autosync (internal anyway due to analog input setting)\n");
+ } else if (snd_rme96_capture_getrate(rme96, &n) < 0) {
+ snd_iprintf(buffer, " sample clock source: autosync (internal anyway due to no valid signal)\n");
+ } else {
+ snd_iprintf(buffer, " sample clock source: autosync\n");
+ }
+ if (rme96->wcreg & RME96_WCR_PRO) {
+ snd_iprintf(buffer, " format: AES/EBU (professional)\n");
+ } else {
+ snd_iprintf(buffer, " format: IEC958 (consumer)\n");
+ }
+ if (rme96->wcreg & RME96_WCR_EMP) {
+ snd_iprintf(buffer, " emphasis: on\n");
+ } else {
+ snd_iprintf(buffer, " emphasis: off\n");
+ }
+ if (rme96->wcreg & RME96_WCR_DOLBY) {
+ snd_iprintf(buffer, " non-audio (dolby): on\n");
+ } else {
+ snd_iprintf(buffer, " non-audio (dolby): off\n");
+ }
+ if (RME96_HAS_ANALOG_IN(rme96)) {
+ snd_iprintf(buffer, "\nAnalog output settings\n");
+ switch (snd_rme96_getmontracks(rme96)) {
+ case RME96_MONITOR_TRACKS_1_2:
+ snd_iprintf(buffer, " monitored ADAT tracks: 1+2\n");
+ break;
+ case RME96_MONITOR_TRACKS_3_4:
+ snd_iprintf(buffer, " monitored ADAT tracks: 3+4\n");
+ break;
+ case RME96_MONITOR_TRACKS_5_6:
+ snd_iprintf(buffer, " monitored ADAT tracks: 5+6\n");
+ break;
+ case RME96_MONITOR_TRACKS_7_8:
+ snd_iprintf(buffer, " monitored ADAT tracks: 7+8\n");
+ break;
+ }
+ switch (snd_rme96_getattenuation(rme96)) {
+ case RME96_ATTENUATION_0:
+ snd_iprintf(buffer, " attenuation: 0 dB\n");
+ break;
+ case RME96_ATTENUATION_6:
+ snd_iprintf(buffer, " attenuation: -6 dB\n");
+ break;
+ case RME96_ATTENUATION_12:
+ snd_iprintf(buffer, " attenuation: -12 dB\n");
+ break;
+ case RME96_ATTENUATION_18:
+ snd_iprintf(buffer, " attenuation: -18 dB\n");
+ break;
+ }
+ snd_iprintf(buffer, " volume left: %u\n", rme96->vol[0]);
+ snd_iprintf(buffer, " volume right: %u\n", rme96->vol[1]);
+ }
+}
+
+static void __devinit
+snd_rme96_proc_init(struct rme96 *rme96)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(rme96->card, "rme96", &entry))
+ snd_info_set_text_ops(entry, rme96, snd_rme96_proc_read);
+}
+
+/*
+ * control interface
+ */
+
+#define snd_rme96_info_loopback_control snd_ctl_boolean_mono_info
+
+static int
+snd_rme96_get_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme96->lock);
+ ucontrol->value.integer.value[0] = rme96->wcreg & RME96_WCR_SEL ? 0 : 1;
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+static int
+snd_rme96_put_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ucontrol->value.integer.value[0] ? 0 : RME96_WCR_SEL;
+ spin_lock_irq(&rme96->lock);
+ val = (rme96->wcreg & ~RME96_WCR_SEL) | val;
+ change = val != rme96->wcreg;
+ rme96->wcreg = val;
+ writel(val, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ spin_unlock_irq(&rme96->lock);
+ return change;
+}
+
+static int
+snd_rme96_info_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *_texts[5] = { "Optical", "Coaxial", "Internal", "XLR", "Analog" };
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ char *texts[5] = { _texts[0], _texts[1], _texts[2], _texts[3], _texts[4] };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ switch (rme96->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI96:
+ case PCI_DEVICE_ID_RME_DIGI96_8:
+ uinfo->value.enumerated.items = 3;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PRO:
+ uinfo->value.enumerated.items = 4;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST:
+ if (rme96->rev > 4) {
+ /* PST */
+ uinfo->value.enumerated.items = 4;
+ texts[3] = _texts[4]; /* Analog instead of XLR */
+ } else {
+ /* PAD */
+ uinfo->value.enumerated.items = 5;
+ }
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ if (uinfo->value.enumerated.item > uinfo->value.enumerated.items - 1) {
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ }
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+static int
+snd_rme96_get_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ unsigned int items = 3;
+
+ spin_lock_irq(&rme96->lock);
+ ucontrol->value.enumerated.item[0] = snd_rme96_getinputtype(rme96);
+
+ switch (rme96->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI96:
+ case PCI_DEVICE_ID_RME_DIGI96_8:
+ items = 3;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PRO:
+ items = 4;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST:
+ if (rme96->rev > 4) {
+ /* for handling PST case, (INPUT_ANALOG is moved to INPUT_XLR */
+ if (ucontrol->value.enumerated.item[0] == RME96_INPUT_ANALOG) {
+ ucontrol->value.enumerated.item[0] = RME96_INPUT_XLR;
+ }
+ items = 4;
+ } else {
+ items = 5;
+ }
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ if (ucontrol->value.enumerated.item[0] >= items) {
+ ucontrol->value.enumerated.item[0] = items - 1;
+ }
+
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+static int
+snd_rme96_put_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change, items = 3;
+
+ switch (rme96->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI96:
+ case PCI_DEVICE_ID_RME_DIGI96_8:
+ items = 3;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PRO:
+ items = 4;
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST:
+ if (rme96->rev > 4) {
+ items = 4;
+ } else {
+ items = 5;
+ }
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ val = ucontrol->value.enumerated.item[0] % items;
+
+ /* special case for PST */
+ if (rme96->pci->device == PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST && rme96->rev > 4) {
+ if (val == RME96_INPUT_XLR) {
+ val = RME96_INPUT_ANALOG;
+ }
+ }
+
+ spin_lock_irq(&rme96->lock);
+ change = (int)val != snd_rme96_getinputtype(rme96);
+ snd_rme96_setinputtype(rme96, val);
+ spin_unlock_irq(&rme96->lock);
+ return change;
+}
+
+static int
+snd_rme96_info_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = { "AutoSync", "Internal", "Word" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2) {
+ uinfo->value.enumerated.item = 2;
+ }
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+static int
+snd_rme96_get_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme96->lock);
+ ucontrol->value.enumerated.item[0] = snd_rme96_getclockmode(rme96);
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+static int
+snd_rme96_put_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ucontrol->value.enumerated.item[0] % 3;
+ spin_lock_irq(&rme96->lock);
+ change = (int)val != snd_rme96_getclockmode(rme96);
+ snd_rme96_setclockmode(rme96, val);
+ spin_unlock_irq(&rme96->lock);
+ return change;
+}
+
+static int
+snd_rme96_info_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = { "0 dB", "-6 dB", "-12 dB", "-18 dB" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3) {
+ uinfo->value.enumerated.item = 3;
+ }
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+static int
+snd_rme96_get_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme96->lock);
+ ucontrol->value.enumerated.item[0] = snd_rme96_getattenuation(rme96);
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+static int
+snd_rme96_put_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ucontrol->value.enumerated.item[0] % 4;
+ spin_lock_irq(&rme96->lock);
+
+ change = (int)val != snd_rme96_getattenuation(rme96);
+ snd_rme96_setattenuation(rme96, val);
+ spin_unlock_irq(&rme96->lock);
+ return change;
+}
+
+static int
+snd_rme96_info_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = { "1+2", "3+4", "5+6", "7+8" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3) {
+ uinfo->value.enumerated.item = 3;
+ }
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+static int
+snd_rme96_get_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme96->lock);
+ ucontrol->value.enumerated.item[0] = snd_rme96_getmontracks(rme96);
+ spin_unlock_irq(&rme96->lock);
+ return 0;
+}
+static int
+snd_rme96_put_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ucontrol->value.enumerated.item[0] % 4;
+ spin_lock_irq(&rme96->lock);
+ change = (int)val != snd_rme96_getmontracks(rme96);
+ snd_rme96_setmontracks(rme96, val);
+ spin_unlock_irq(&rme96->lock);
+ return change;
+}
+
+static u32 snd_rme96_convert_from_aes(struct snd_aes_iec958 *aes)
+{
+ u32 val = 0;
+ val |= (aes->status[0] & IEC958_AES0_PROFESSIONAL) ? RME96_WCR_PRO : 0;
+ val |= (aes->status[0] & IEC958_AES0_NONAUDIO) ? RME96_WCR_DOLBY : 0;
+ if (val & RME96_WCR_PRO)
+ val |= (aes->status[0] & IEC958_AES0_PRO_EMPHASIS_5015) ? RME96_WCR_EMP : 0;
+ else
+ val |= (aes->status[0] & IEC958_AES0_CON_EMPHASIS_5015) ? RME96_WCR_EMP : 0;
+ return val;
+}
+
+static void snd_rme96_convert_to_aes(struct snd_aes_iec958 *aes, u32 val)
+{
+ aes->status[0] = ((val & RME96_WCR_PRO) ? IEC958_AES0_PROFESSIONAL : 0) |
+ ((val & RME96_WCR_DOLBY) ? IEC958_AES0_NONAUDIO : 0);
+ if (val & RME96_WCR_PRO)
+ aes->status[0] |= (val & RME96_WCR_EMP) ? IEC958_AES0_PRO_EMPHASIS_5015 : 0;
+ else
+ aes->status[0] |= (val & RME96_WCR_EMP) ? IEC958_AES0_CON_EMPHASIS_5015 : 0;
+}
+
+static int snd_rme96_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme96_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ snd_rme96_convert_to_aes(&ucontrol->value.iec958, rme96->wcreg_spdif);
+ return 0;
+}
+
+static int snd_rme96_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_rme96_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&rme96->lock);
+ change = val != rme96->wcreg_spdif;
+ rme96->wcreg_spdif = val;
+ spin_unlock_irq(&rme96->lock);
+ return change;
+}
+
+static int snd_rme96_control_spdif_stream_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme96_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ snd_rme96_convert_to_aes(&ucontrol->value.iec958, rme96->wcreg_spdif_stream);
+ return 0;
+}
+
+static int snd_rme96_control_spdif_stream_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_rme96_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&rme96->lock);
+ change = val != rme96->wcreg_spdif_stream;
+ rme96->wcreg_spdif_stream = val;
+ rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP);
+ rme96->wcreg |= val;
+ writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
+ spin_unlock_irq(&rme96->lock);
+ return change;
+}
+
+static int snd_rme96_control_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme96_control_spdif_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = kcontrol->private_value;
+ return 0;
+}
+
+static int
+snd_rme96_dac_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = RME96_185X_MAX_OUT(rme96);
+ return 0;
+}
+
+static int
+snd_rme96_dac_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *u)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme96->lock);
+ u->value.integer.value[0] = rme96->vol[0];
+ u->value.integer.value[1] = rme96->vol[1];
+ spin_unlock_irq(&rme96->lock);
+
+ return 0;
+}
+
+static int
+snd_rme96_dac_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *u)
+{
+ struct rme96 *rme96 = snd_kcontrol_chip(kcontrol);
+ int change = 0;
+ unsigned int vol, maxvol;
+
+
+ if (!RME96_HAS_ANALOG_OUT(rme96))
+ return -EINVAL;
+ maxvol = RME96_185X_MAX_OUT(rme96);
+ spin_lock_irq(&rme96->lock);
+ vol = u->value.integer.value[0];
+ if (vol != rme96->vol[0] && vol <= maxvol) {
+ rme96->vol[0] = vol;
+ change = 1;
+ }
+ vol = u->value.integer.value[1];
+ if (vol != rme96->vol[1] && vol <= maxvol) {
+ rme96->vol[1] = vol;
+ change = 1;
+ }
+ if (change)
+ snd_rme96_apply_dac_volume(rme96);
+ spin_unlock_irq(&rme96->lock);
+
+ return change;
+}
+
+static struct snd_kcontrol_new snd_rme96_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_rme96_control_spdif_info,
+ .get = snd_rme96_control_spdif_get,
+ .put = snd_rme96_control_spdif_put
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_rme96_control_spdif_stream_info,
+ .get = snd_rme96_control_spdif_stream_get,
+ .put = snd_rme96_control_spdif_stream_put
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_rme96_control_spdif_mask_info,
+ .get = snd_rme96_control_spdif_mask_get,
+ .private_value = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_CON_EMPHASIS
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+ .info = snd_rme96_control_spdif_mask_info,
+ .get = snd_rme96_control_spdif_mask_get,
+ .private_value = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_PRO_EMPHASIS
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Connector",
+ .info = snd_rme96_info_inputtype_control,
+ .get = snd_rme96_get_inputtype_control,
+ .put = snd_rme96_put_inputtype_control
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Loopback Input",
+ .info = snd_rme96_info_loopback_control,
+ .get = snd_rme96_get_loopback_control,
+ .put = snd_rme96_put_loopback_control
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Sample Clock Source",
+ .info = snd_rme96_info_clockmode_control,
+ .get = snd_rme96_get_clockmode_control,
+ .put = snd_rme96_put_clockmode_control
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Tracks",
+ .info = snd_rme96_info_montracks_control,
+ .get = snd_rme96_get_montracks_control,
+ .put = snd_rme96_put_montracks_control
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Attenuation",
+ .info = snd_rme96_info_attenuation_control,
+ .get = snd_rme96_get_attenuation_control,
+ .put = snd_rme96_put_attenuation_control
+},
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Playback Volume",
+ .info = snd_rme96_dac_volume_info,
+ .get = snd_rme96_dac_volume_get,
+ .put = snd_rme96_dac_volume_put
+}
+};
+
+static int
+snd_rme96_create_switches(struct snd_card *card,
+ struct rme96 *rme96)
+{
+ int idx, err;
+ struct snd_kcontrol *kctl;
+
+ for (idx = 0; idx < 7; idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_rme96_controls[idx], rme96))) < 0)
+ return err;
+ if (idx == 1) /* IEC958 (S/PDIF) Stream */
+ rme96->spdif_ctl = kctl;
+ }
+
+ if (RME96_HAS_ANALOG_OUT(rme96)) {
+ for (idx = 7; idx < 10; idx++)
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_rme96_controls[idx], rme96))) < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/*
+ * Card initialisation
+ */
+
+static void snd_rme96_card_free(struct snd_card *card)
+{
+ snd_rme96_free(card->private_data);
+}
+
+static int __devinit
+snd_rme96_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct rme96 *rme96;
+ struct snd_card *card;
+ int err;
+ u8 val;
+
+ if (dev >= SNDRV_CARDS) {
+ return -ENODEV;
+ }
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+ if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct rme96))) == NULL)
+ return -ENOMEM;
+ card->private_free = snd_rme96_card_free;
+ rme96 = (struct rme96 *)card->private_data;
+ rme96->card = card;
+ rme96->pci = pci;
+ snd_card_set_dev(card, &pci->dev);
+ if ((err = snd_rme96_create(rme96)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "Digi96");
+ switch (rme96->pci->device) {
+ case PCI_DEVICE_ID_RME_DIGI96:
+ strcpy(card->shortname, "RME Digi96");
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8:
+ strcpy(card->shortname, "RME Digi96/8");
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PRO:
+ strcpy(card->shortname, "RME Digi96/8 PRO");
+ break;
+ case PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST:
+ pci_read_config_byte(rme96->pci, 8, &val);
+ if (val < 5) {
+ strcpy(card->shortname, "RME Digi96/8 PAD");
+ } else {
+ strcpy(card->shortname, "RME Digi96/8 PST");
+ }
+ break;
+ }
+ sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname,
+ rme96->port, rme96->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_rme96_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "RME Digi96",
+ .id_table = snd_rme96_ids,
+ .probe = snd_rme96_probe,
+ .remove = __devexit_p(snd_rme96_remove),
+};
+
+static int __init alsa_card_rme96_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_rme96_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_rme96_init)
+module_exit(alsa_card_rme96_exit)
diff --git a/sound/pci/rme9652/Makefile b/sound/pci/rme9652/Makefile
new file mode 100644
index 0000000..dcba560
--- /dev/null
+++ b/sound/pci/rme9652/Makefile
@@ -0,0 +1,13 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-rme9652-objs := rme9652.o
+snd-hdsp-objs := hdsp.o
+snd-hdspm-objs := hdspm.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_RME9652) += snd-rme9652.o
+obj-$(CONFIG_SND_HDSP) += snd-hdsp.o
+obj-$(CONFIG_SND_HDSPM) +=snd-hdspm.o
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
new file mode 100644
index 0000000..736246f
--- /dev/null
+++ b/sound/pci/rme9652/hdsp.c
@@ -0,0 +1,5212 @@
+/*
+ * ALSA driver for RME Hammerfall DSP audio interface(s)
+ *
+ * Copyright (c) 2002 Paul Davis
+ * Marcus Andersson
+ * Thomas Charbonnel
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/firmware.h>
+#include <linux/moduleparam.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+#include <sound/rawmidi.h>
+#include <sound/hwdep.h>
+#include <sound/initval.h>
+#include <sound/hdsp.h>
+
+#include <asm/byteorder.h>
+#include <asm/current.h>
+#include <asm/io.h>
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for RME Hammerfall DSP interface.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for RME Hammerfall DSP interface.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable/disable specific Hammerfall DSP soundcards.");
+MODULE_AUTHOR("Paul Davis <paul@linuxaudiosystems.com>, Marcus Andersson, Thomas Charbonnel <thomas@undata.org>");
+MODULE_DESCRIPTION("RME Hammerfall DSP");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP},"
+ "{RME HDSP-9652},"
+ "{RME HDSP-9632}}");
+#ifdef HDSP_FW_LOADER
+MODULE_FIRMWARE("multiface_firmware.bin");
+MODULE_FIRMWARE("multiface_firmware_rev11.bin");
+MODULE_FIRMWARE("digiface_firmware.bin");
+MODULE_FIRMWARE("digiface_firmware_rev11.bin");
+#endif
+
+#define HDSP_MAX_CHANNELS 26
+#define HDSP_MAX_DS_CHANNELS 14
+#define HDSP_MAX_QS_CHANNELS 8
+#define DIGIFACE_SS_CHANNELS 26
+#define DIGIFACE_DS_CHANNELS 14
+#define MULTIFACE_SS_CHANNELS 18
+#define MULTIFACE_DS_CHANNELS 14
+#define H9652_SS_CHANNELS 26
+#define H9652_DS_CHANNELS 14
+/* This does not include possible Analog Extension Boards
+ AEBs are detected at card initialization
+*/
+#define H9632_SS_CHANNELS 12
+#define H9632_DS_CHANNELS 8
+#define H9632_QS_CHANNELS 4
+
+/* Write registers. These are defined as byte-offsets from the iobase value.
+ */
+#define HDSP_resetPointer 0
+#define HDSP_freqReg 0
+#define HDSP_outputBufferAddress 32
+#define HDSP_inputBufferAddress 36
+#define HDSP_controlRegister 64
+#define HDSP_interruptConfirmation 96
+#define HDSP_outputEnable 128
+#define HDSP_control2Reg 256
+#define HDSP_midiDataOut0 352
+#define HDSP_midiDataOut1 356
+#define HDSP_fifoData 368
+#define HDSP_inputEnable 384
+
+/* Read registers. These are defined as byte-offsets from the iobase value
+ */
+
+#define HDSP_statusRegister 0
+#define HDSP_timecode 128
+#define HDSP_status2Register 192
+#define HDSP_midiDataIn0 360
+#define HDSP_midiDataIn1 364
+#define HDSP_midiStatusOut0 384
+#define HDSP_midiStatusOut1 388
+#define HDSP_midiStatusIn0 392
+#define HDSP_midiStatusIn1 396
+#define HDSP_fifoStatus 400
+
+/* the meters are regular i/o-mapped registers, but offset
+ considerably from the rest. the peak registers are reset
+ when read; the least-significant 4 bits are full-scale counters;
+ the actual peak value is in the most-significant 24 bits.
+*/
+
+#define HDSP_playbackPeakLevel 4096 /* 26 * 32 bit values */
+#define HDSP_inputPeakLevel 4224 /* 26 * 32 bit values */
+#define HDSP_outputPeakLevel 4352 /* (26+2) * 32 bit values */
+#define HDSP_playbackRmsLevel 4612 /* 26 * 64 bit values */
+#define HDSP_inputRmsLevel 4868 /* 26 * 64 bit values */
+
+
+/* This is for H9652 cards
+ Peak values are read downward from the base
+ Rms values are read upward
+ There are rms values for the outputs too
+ 26*3 values are read in ss mode
+ 14*3 in ds mode, with no gap between values
+*/
+#define HDSP_9652_peakBase 7164
+#define HDSP_9652_rmsBase 4096
+
+/* c.f. the hdsp_9632_meters_t struct */
+#define HDSP_9632_metersBase 4096
+
+#define HDSP_IO_EXTENT 7168
+
+/* control2 register bits */
+
+#define HDSP_TMS 0x01
+#define HDSP_TCK 0x02
+#define HDSP_TDI 0x04
+#define HDSP_JTAG 0x08
+#define HDSP_PWDN 0x10
+#define HDSP_PROGRAM 0x020
+#define HDSP_CONFIG_MODE_0 0x040
+#define HDSP_CONFIG_MODE_1 0x080
+#define HDSP_VERSION_BIT 0x100
+#define HDSP_BIGENDIAN_MODE 0x200
+#define HDSP_RD_MULTIPLE 0x400
+#define HDSP_9652_ENABLE_MIXER 0x800
+#define HDSP_TDO 0x10000000
+
+#define HDSP_S_PROGRAM (HDSP_PROGRAM|HDSP_CONFIG_MODE_0)
+#define HDSP_S_LOAD (HDSP_PROGRAM|HDSP_CONFIG_MODE_1)
+
+/* Control Register bits */
+
+#define HDSP_Start (1<<0) /* start engine */
+#define HDSP_Latency0 (1<<1) /* buffer size = 2^n where n is defined by Latency{2,1,0} */
+#define HDSP_Latency1 (1<<2) /* [ see above ] */
+#define HDSP_Latency2 (1<<3) /* [ see above ] */
+#define HDSP_ClockModeMaster (1<<4) /* 1=Master, 0=Slave/Autosync */
+#define HDSP_AudioInterruptEnable (1<<5) /* what do you think ? */
+#define HDSP_Frequency0 (1<<6) /* 0=44.1kHz/88.2kHz/176.4kHz 1=48kHz/96kHz/192kHz */
+#define HDSP_Frequency1 (1<<7) /* 0=32kHz/64kHz/128kHz */
+#define HDSP_DoubleSpeed (1<<8) /* 0=normal speed, 1=double speed */
+#define HDSP_SPDIFProfessional (1<<9) /* 0=consumer, 1=professional */
+#define HDSP_SPDIFEmphasis (1<<10) /* 0=none, 1=on */
+#define HDSP_SPDIFNonAudio (1<<11) /* 0=off, 1=on */
+#define HDSP_SPDIFOpticalOut (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */
+#define HDSP_SyncRef2 (1<<13)
+#define HDSP_SPDIFInputSelect0 (1<<14)
+#define HDSP_SPDIFInputSelect1 (1<<15)
+#define HDSP_SyncRef0 (1<<16)
+#define HDSP_SyncRef1 (1<<17)
+#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */
+#define HDSP_XLRBreakoutCable (1<<20) /* For H9632 cards */
+#define HDSP_Midi0InterruptEnable (1<<22)
+#define HDSP_Midi1InterruptEnable (1<<23)
+#define HDSP_LineOut (1<<24)
+#define HDSP_ADGain0 (1<<25) /* From here : H9632 specific */
+#define HDSP_ADGain1 (1<<26)
+#define HDSP_DAGain0 (1<<27)
+#define HDSP_DAGain1 (1<<28)
+#define HDSP_PhoneGain0 (1<<29)
+#define HDSP_PhoneGain1 (1<<30)
+#define HDSP_QuadSpeed (1<<31)
+
+#define HDSP_ADGainMask (HDSP_ADGain0|HDSP_ADGain1)
+#define HDSP_ADGainMinus10dBV HDSP_ADGainMask
+#define HDSP_ADGainPlus4dBu (HDSP_ADGain0)
+#define HDSP_ADGainLowGain 0
+
+#define HDSP_DAGainMask (HDSP_DAGain0|HDSP_DAGain1)
+#define HDSP_DAGainHighGain HDSP_DAGainMask
+#define HDSP_DAGainPlus4dBu (HDSP_DAGain0)
+#define HDSP_DAGainMinus10dBV 0
+
+#define HDSP_PhoneGainMask (HDSP_PhoneGain0|HDSP_PhoneGain1)
+#define HDSP_PhoneGain0dB HDSP_PhoneGainMask
+#define HDSP_PhoneGainMinus6dB (HDSP_PhoneGain0)
+#define HDSP_PhoneGainMinus12dB 0
+
+#define HDSP_LatencyMask (HDSP_Latency0|HDSP_Latency1|HDSP_Latency2)
+#define HDSP_FrequencyMask (HDSP_Frequency0|HDSP_Frequency1|HDSP_DoubleSpeed|HDSP_QuadSpeed)
+
+#define HDSP_SPDIFInputMask (HDSP_SPDIFInputSelect0|HDSP_SPDIFInputSelect1)
+#define HDSP_SPDIFInputADAT1 0
+#define HDSP_SPDIFInputCoaxial (HDSP_SPDIFInputSelect0)
+#define HDSP_SPDIFInputCdrom (HDSP_SPDIFInputSelect1)
+#define HDSP_SPDIFInputAES (HDSP_SPDIFInputSelect0|HDSP_SPDIFInputSelect1)
+
+#define HDSP_SyncRefMask (HDSP_SyncRef0|HDSP_SyncRef1|HDSP_SyncRef2)
+#define HDSP_SyncRef_ADAT1 0
+#define HDSP_SyncRef_ADAT2 (HDSP_SyncRef0)
+#define HDSP_SyncRef_ADAT3 (HDSP_SyncRef1)
+#define HDSP_SyncRef_SPDIF (HDSP_SyncRef0|HDSP_SyncRef1)
+#define HDSP_SyncRef_WORD (HDSP_SyncRef2)
+#define HDSP_SyncRef_ADAT_SYNC (HDSP_SyncRef0|HDSP_SyncRef2)
+
+/* Sample Clock Sources */
+
+#define HDSP_CLOCK_SOURCE_AUTOSYNC 0
+#define HDSP_CLOCK_SOURCE_INTERNAL_32KHZ 1
+#define HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ 2
+#define HDSP_CLOCK_SOURCE_INTERNAL_48KHZ 3
+#define HDSP_CLOCK_SOURCE_INTERNAL_64KHZ 4
+#define HDSP_CLOCK_SOURCE_INTERNAL_88_2KHZ 5
+#define HDSP_CLOCK_SOURCE_INTERNAL_96KHZ 6
+#define HDSP_CLOCK_SOURCE_INTERNAL_128KHZ 7
+#define HDSP_CLOCK_SOURCE_INTERNAL_176_4KHZ 8
+#define HDSP_CLOCK_SOURCE_INTERNAL_192KHZ 9
+
+/* Preferred sync reference choices - used by "pref_sync_ref" control switch */
+
+#define HDSP_SYNC_FROM_WORD 0
+#define HDSP_SYNC_FROM_SPDIF 1
+#define HDSP_SYNC_FROM_ADAT1 2
+#define HDSP_SYNC_FROM_ADAT_SYNC 3
+#define HDSP_SYNC_FROM_ADAT2 4
+#define HDSP_SYNC_FROM_ADAT3 5
+
+/* SyncCheck status */
+
+#define HDSP_SYNC_CHECK_NO_LOCK 0
+#define HDSP_SYNC_CHECK_LOCK 1
+#define HDSP_SYNC_CHECK_SYNC 2
+
+/* AutoSync references - used by "autosync_ref" control switch */
+
+#define HDSP_AUTOSYNC_FROM_WORD 0
+#define HDSP_AUTOSYNC_FROM_ADAT_SYNC 1
+#define HDSP_AUTOSYNC_FROM_SPDIF 2
+#define HDSP_AUTOSYNC_FROM_NONE 3
+#define HDSP_AUTOSYNC_FROM_ADAT1 4
+#define HDSP_AUTOSYNC_FROM_ADAT2 5
+#define HDSP_AUTOSYNC_FROM_ADAT3 6
+
+/* Possible sources of S/PDIF input */
+
+#define HDSP_SPDIFIN_OPTICAL 0 /* optical (ADAT1) */
+#define HDSP_SPDIFIN_COAXIAL 1 /* coaxial (RCA) */
+#define HDSP_SPDIFIN_INTERNAL 2 /* internal (CDROM) */
+#define HDSP_SPDIFIN_AES 3 /* xlr for H9632 (AES)*/
+
+#define HDSP_Frequency32KHz HDSP_Frequency0
+#define HDSP_Frequency44_1KHz HDSP_Frequency1
+#define HDSP_Frequency48KHz (HDSP_Frequency1|HDSP_Frequency0)
+#define HDSP_Frequency64KHz (HDSP_DoubleSpeed|HDSP_Frequency0)
+#define HDSP_Frequency88_2KHz (HDSP_DoubleSpeed|HDSP_Frequency1)
+#define HDSP_Frequency96KHz (HDSP_DoubleSpeed|HDSP_Frequency1|HDSP_Frequency0)
+/* For H9632 cards */
+#define HDSP_Frequency128KHz (HDSP_QuadSpeed|HDSP_DoubleSpeed|HDSP_Frequency0)
+#define HDSP_Frequency176_4KHz (HDSP_QuadSpeed|HDSP_DoubleSpeed|HDSP_Frequency1)
+#define HDSP_Frequency192KHz (HDSP_QuadSpeed|HDSP_DoubleSpeed|HDSP_Frequency1|HDSP_Frequency0)
+/* RME says n = 104857600000000, but in the windows MADI driver, I see:
+ return 104857600000000 / rate; // 100 MHz
+ return 110100480000000 / rate; // 105 MHz
+*/
+#define DDS_NUMERATOR 104857600000000ULL; /* = 2^20 * 10^8 */
+
+#define hdsp_encode_latency(x) (((x)<<1) & HDSP_LatencyMask)
+#define hdsp_decode_latency(x) (((x) & HDSP_LatencyMask)>>1)
+
+#define hdsp_encode_spdif_in(x) (((x)&0x3)<<14)
+#define hdsp_decode_spdif_in(x) (((x)>>14)&0x3)
+
+/* Status Register bits */
+
+#define HDSP_audioIRQPending (1<<0)
+#define HDSP_Lock2 (1<<1) /* this is for Digiface and H9652 */
+#define HDSP_spdifFrequency3 HDSP_Lock2 /* this is for H9632 only */
+#define HDSP_Lock1 (1<<2)
+#define HDSP_Lock0 (1<<3)
+#define HDSP_SPDIFSync (1<<4)
+#define HDSP_TimecodeLock (1<<5)
+#define HDSP_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
+#define HDSP_Sync2 (1<<16)
+#define HDSP_Sync1 (1<<17)
+#define HDSP_Sync0 (1<<18)
+#define HDSP_DoubleSpeedStatus (1<<19)
+#define HDSP_ConfigError (1<<20)
+#define HDSP_DllError (1<<21)
+#define HDSP_spdifFrequency0 (1<<22)
+#define HDSP_spdifFrequency1 (1<<23)
+#define HDSP_spdifFrequency2 (1<<24)
+#define HDSP_SPDIFErrorFlag (1<<25)
+#define HDSP_BufferID (1<<26)
+#define HDSP_TimecodeSync (1<<27)
+#define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */
+#define HDSP_AEBI (1<<29) /* 0 = present, 1 = absent */
+#define HDSP_midi0IRQPending (1<<30)
+#define HDSP_midi1IRQPending (1<<31)
+
+#define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2)
+#define HDSP_spdifFrequencyMask_9632 (HDSP_spdifFrequency0|\
+ HDSP_spdifFrequency1|\
+ HDSP_spdifFrequency2|\
+ HDSP_spdifFrequency3)
+
+#define HDSP_spdifFrequency32KHz (HDSP_spdifFrequency0)
+#define HDSP_spdifFrequency44_1KHz (HDSP_spdifFrequency1)
+#define HDSP_spdifFrequency48KHz (HDSP_spdifFrequency0|HDSP_spdifFrequency1)
+
+#define HDSP_spdifFrequency64KHz (HDSP_spdifFrequency2)
+#define HDSP_spdifFrequency88_2KHz (HDSP_spdifFrequency0|HDSP_spdifFrequency2)
+#define HDSP_spdifFrequency96KHz (HDSP_spdifFrequency2|HDSP_spdifFrequency1)
+
+/* This is for H9632 cards */
+#define HDSP_spdifFrequency128KHz (HDSP_spdifFrequency0|\
+ HDSP_spdifFrequency1|\
+ HDSP_spdifFrequency2)
+#define HDSP_spdifFrequency176_4KHz HDSP_spdifFrequency3
+#define HDSP_spdifFrequency192KHz (HDSP_spdifFrequency3|HDSP_spdifFrequency0)
+
+/* Status2 Register bits */
+
+#define HDSP_version0 (1<<0)
+#define HDSP_version1 (1<<1)
+#define HDSP_version2 (1<<2)
+#define HDSP_wc_lock (1<<3)
+#define HDSP_wc_sync (1<<4)
+#define HDSP_inp_freq0 (1<<5)
+#define HDSP_inp_freq1 (1<<6)
+#define HDSP_inp_freq2 (1<<7)
+#define HDSP_SelSyncRef0 (1<<8)
+#define HDSP_SelSyncRef1 (1<<9)
+#define HDSP_SelSyncRef2 (1<<10)
+
+#define HDSP_wc_valid (HDSP_wc_lock|HDSP_wc_sync)
+
+#define HDSP_systemFrequencyMask (HDSP_inp_freq0|HDSP_inp_freq1|HDSP_inp_freq2)
+#define HDSP_systemFrequency32 (HDSP_inp_freq0)
+#define HDSP_systemFrequency44_1 (HDSP_inp_freq1)
+#define HDSP_systemFrequency48 (HDSP_inp_freq0|HDSP_inp_freq1)
+#define HDSP_systemFrequency64 (HDSP_inp_freq2)
+#define HDSP_systemFrequency88_2 (HDSP_inp_freq0|HDSP_inp_freq2)
+#define HDSP_systemFrequency96 (HDSP_inp_freq1|HDSP_inp_freq2)
+/* FIXME : more values for 9632 cards ? */
+
+#define HDSP_SelSyncRefMask (HDSP_SelSyncRef0|HDSP_SelSyncRef1|HDSP_SelSyncRef2)
+#define HDSP_SelSyncRef_ADAT1 0
+#define HDSP_SelSyncRef_ADAT2 (HDSP_SelSyncRef0)
+#define HDSP_SelSyncRef_ADAT3 (HDSP_SelSyncRef1)
+#define HDSP_SelSyncRef_SPDIF (HDSP_SelSyncRef0|HDSP_SelSyncRef1)
+#define HDSP_SelSyncRef_WORD (HDSP_SelSyncRef2)
+#define HDSP_SelSyncRef_ADAT_SYNC (HDSP_SelSyncRef0|HDSP_SelSyncRef2)
+
+/* Card state flags */
+
+#define HDSP_InitializationComplete (1<<0)
+#define HDSP_FirmwareLoaded (1<<1)
+#define HDSP_FirmwareCached (1<<2)
+
+/* FIFO wait times, defined in terms of 1/10ths of msecs */
+
+#define HDSP_LONG_WAIT 5000
+#define HDSP_SHORT_WAIT 30
+
+#define UNITY_GAIN 32768
+#define MINUS_INFINITY_GAIN 0
+
+/* the size of a substream (1 mono data stream) */
+
+#define HDSP_CHANNEL_BUFFER_SAMPLES (16*1024)
+#define HDSP_CHANNEL_BUFFER_BYTES (4*HDSP_CHANNEL_BUFFER_SAMPLES)
+
+/* the size of the area we need to allocate for DMA transfers. the
+ size is the same regardless of the number of channels - the
+ Multiface still uses the same memory area.
+
+ Note that we allocate 1 more channel than is apparently needed
+ because the h/w seems to write 1 byte beyond the end of the last
+ page. Sigh.
+*/
+
+#define HDSP_DMA_AREA_BYTES ((HDSP_MAX_CHANNELS+1) * HDSP_CHANNEL_BUFFER_BYTES)
+#define HDSP_DMA_AREA_KILOBYTES (HDSP_DMA_AREA_BYTES/1024)
+
+/* use hotplug firmeare loader? */
+#if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE)
+#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP)
+#define HDSP_FW_LOADER
+#endif
+#endif
+
+struct hdsp_9632_meters {
+ u32 input_peak[16];
+ u32 playback_peak[16];
+ u32 output_peak[16];
+ u32 xxx_peak[16];
+ u32 padding[64];
+ u32 input_rms_low[16];
+ u32 playback_rms_low[16];
+ u32 output_rms_low[16];
+ u32 xxx_rms_low[16];
+ u32 input_rms_high[16];
+ u32 playback_rms_high[16];
+ u32 output_rms_high[16];
+ u32 xxx_rms_high[16];
+};
+
+struct hdsp_midi {
+ struct hdsp *hdsp;
+ int id;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *input;
+ struct snd_rawmidi_substream *output;
+ char istimer; /* timer in use */
+ struct timer_list timer;
+ spinlock_t lock;
+ int pending;
+};
+
+struct hdsp {
+ spinlock_t lock;
+ struct snd_pcm_substream *capture_substream;
+ struct snd_pcm_substream *playback_substream;
+ struct hdsp_midi midi[2];
+ struct tasklet_struct midi_tasklet;
+ int use_midi_tasklet;
+ int precise_ptr;
+ u32 control_register; /* cached value */
+ u32 control2_register; /* cached value */
+ u32 creg_spdif;
+ u32 creg_spdif_stream;
+ int clock_source_locked;
+ char *card_name; /* digiface/multiface */
+ enum HDSP_IO_Type io_type; /* ditto, but for code use */
+ unsigned short firmware_rev;
+ unsigned short state; /* stores state bits */
+ u32 firmware_cache[24413]; /* this helps recover from accidental iobox power failure */
+ size_t period_bytes; /* guess what this is */
+ unsigned char max_channels;
+ unsigned char qs_in_channels; /* quad speed mode for H9632 */
+ unsigned char ds_in_channels;
+ unsigned char ss_in_channels; /* different for multiface/digiface */
+ unsigned char qs_out_channels;
+ unsigned char ds_out_channels;
+ unsigned char ss_out_channels;
+
+ struct snd_dma_buffer capture_dma_buf;
+ struct snd_dma_buffer playback_dma_buf;
+ unsigned char *capture_buffer; /* suitably aligned address */
+ unsigned char *playback_buffer; /* suitably aligned address */
+
+ pid_t capture_pid;
+ pid_t playback_pid;
+ int running;
+ int system_sample_rate;
+ char *channel_map;
+ int dev;
+ int irq;
+ unsigned long port;
+ void __iomem *iobase;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_hwdep *hwdep;
+ struct pci_dev *pci;
+ struct snd_kcontrol *spdif_ctl;
+ unsigned short mixer_matrix[HDSP_MATRIX_MIXER_SIZE];
+ unsigned int dds_value; /* last value written to freq register */
+};
+
+/* These tables map the ALSA channels 1..N to the channels that we
+ need to use in order to find the relevant channel buffer. RME
+ refer to this kind of mapping as between "the ADAT channel and
+ the DMA channel." We index it using the logical audio channel,
+ and the value is the DMA channel (i.e. channel buffer number)
+ where the data for that channel can be read/written from/to.
+*/
+
+static char channel_map_df_ss[HDSP_MAX_CHANNELS] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17,
+ 18, 19, 20, 21, 22, 23, 24, 25
+};
+
+static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */
+ /* Analog */
+ 0, 1, 2, 3, 4, 5, 6, 7,
+ /* ADAT 2 */
+ 16, 17, 18, 19, 20, 21, 22, 23,
+ /* SPDIF */
+ 24, 25,
+ -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+static char channel_map_ds[HDSP_MAX_CHANNELS] = {
+ /* ADAT channels are remapped */
+ 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23,
+ /* channels 12 and 13 are S/PDIF */
+ 24, 25,
+ /* others don't exist */
+ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = {
+ /* ADAT channels */
+ 0, 1, 2, 3, 4, 5, 6, 7,
+ /* SPDIF */
+ 8, 9,
+ /* Analog */
+ 10, 11,
+ /* AO4S-192 and AI4S-192 extension boards */
+ 12, 13, 14, 15,
+ /* others don't exist */
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1
+};
+
+static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = {
+ /* ADAT */
+ 1, 3, 5, 7,
+ /* SPDIF */
+ 8, 9,
+ /* Analog */
+ 10, 11,
+ /* AO4S-192 and AI4S-192 extension boards */
+ 12, 13, 14, 15,
+ /* others don't exist */
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1
+};
+
+static char channel_map_H9632_qs[HDSP_MAX_CHANNELS] = {
+ /* ADAT is disabled in this mode */
+ /* SPDIF */
+ 8, 9,
+ /* Analog */
+ 10, 11,
+ /* AO4S-192 and AI4S-192 extension boards */
+ 12, 13, 14, 15,
+ /* others don't exist */
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1
+};
+
+static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer *dmab, size_t size)
+{
+ dmab->dev.type = SNDRV_DMA_TYPE_DEV;
+ dmab->dev.dev = snd_dma_pci_data(pci);
+ if (snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
+ if (dmab->bytes >= size)
+ return 0;
+ }
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, dmab) < 0)
+ return -ENOMEM;
+ return 0;
+}
+
+static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci)
+{
+ if (dmab->area) {
+ dmab->dev.dev = NULL; /* make it anonymous */
+ snd_dma_reserve_buf(dmab, snd_dma_pci_buf_id(pci));
+ }
+}
+
+
+static struct pci_device_id snd_hdsp_ids[] = {
+ {
+ .vendor = PCI_VENDOR_ID_XILINX,
+ .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP,
+ .subvendor = PCI_ANY_ID,
+ .subdevice = PCI_ANY_ID,
+ }, /* RME Hammerfall-DSP */
+ { 0, },
+};
+
+MODULE_DEVICE_TABLE(pci, snd_hdsp_ids);
+
+/* prototypes */
+static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp);
+static int snd_hdsp_create_pcm(struct snd_card *card, struct hdsp *hdsp);
+static int snd_hdsp_enable_io (struct hdsp *hdsp);
+static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp);
+static void snd_hdsp_initialize_channels (struct hdsp *hdsp);
+static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout);
+static int hdsp_autosync_ref(struct hdsp *hdsp);
+static int snd_hdsp_set_defaults(struct hdsp *hdsp);
+static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp);
+
+static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out)
+{
+ switch (hdsp->io_type) {
+ case Multiface:
+ case Digiface:
+ default:
+ if (hdsp->firmware_rev == 0xa)
+ return (64 * out) + (32 + (in));
+ else
+ return (52 * out) + (26 + (in));
+ case H9632:
+ return (32 * out) + (16 + (in));
+ case H9652:
+ return (52 * out) + (26 + (in));
+ }
+}
+
+static int hdsp_input_to_output_key (struct hdsp *hdsp, int in, int out)
+{
+ switch (hdsp->io_type) {
+ case Multiface:
+ case Digiface:
+ default:
+ if (hdsp->firmware_rev == 0xa)
+ return (64 * out) + in;
+ else
+ return (52 * out) + in;
+ case H9632:
+ return (32 * out) + in;
+ case H9652:
+ return (52 * out) + in;
+ }
+}
+
+static void hdsp_write(struct hdsp *hdsp, int reg, int val)
+{
+ writel(val, hdsp->iobase + reg);
+}
+
+static unsigned int hdsp_read(struct hdsp *hdsp, int reg)
+{
+ return readl (hdsp->iobase + reg);
+}
+
+static int hdsp_check_for_iobox (struct hdsp *hdsp)
+{
+
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0;
+ if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) {
+ snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n");
+ hdsp->state &= ~HDSP_FirmwareLoaded;
+ return -EIO;
+ }
+ return 0;
+
+}
+
+static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
+
+ int i;
+ unsigned long flags;
+
+ if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
+
+ snd_printk ("Hammerfall-DSP: loading firmware\n");
+
+ hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM);
+ hdsp_write (hdsp, HDSP_fifoData, 0);
+
+ if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) {
+ snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n");
+ return -EIO;
+ }
+
+ hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD);
+
+ for (i = 0; i < 24413; ++i) {
+ hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]);
+ if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) {
+ snd_printk ("Hammerfall-DSP: timeout during firmware loading\n");
+ return -EIO;
+ }
+ }
+
+ ssleep(3);
+
+ if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) {
+ snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n");
+ return -EIO;
+ }
+
+#ifdef SNDRV_BIG_ENDIAN
+ hdsp->control2_register = HDSP_BIGENDIAN_MODE;
+#else
+ hdsp->control2_register = 0;
+#endif
+ hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register);
+ snd_printk ("Hammerfall-DSP: finished firmware loading\n");
+
+ }
+ if (hdsp->state & HDSP_InitializationComplete) {
+ snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n");
+ spin_lock_irqsave(&hdsp->lock, flags);
+ snd_hdsp_set_defaults(hdsp);
+ spin_unlock_irqrestore(&hdsp->lock, flags);
+ }
+
+ hdsp->state |= HDSP_FirmwareLoaded;
+
+ return 0;
+}
+
+static int hdsp_get_iobox_version (struct hdsp *hdsp)
+{
+ if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
+
+ hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM);
+ hdsp_write (hdsp, HDSP_fifoData, 0);
+ if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0)
+ return -EIO;
+
+ hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD);
+ hdsp_write (hdsp, HDSP_fifoData, 0);
+
+ if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT)) {
+ hdsp->io_type = Multiface;
+ hdsp_write (hdsp, HDSP_control2Reg, HDSP_VERSION_BIT);
+ hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD);
+ hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT);
+ } else {
+ hdsp->io_type = Digiface;
+ }
+ } else {
+ /* firmware was already loaded, get iobox type */
+ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1)
+ hdsp->io_type = Multiface;
+ else
+ hdsp->io_type = Digiface;
+ }
+ return 0;
+}
+
+
+#ifdef HDSP_FW_LOADER
+static int hdsp_request_fw_loader(struct hdsp *hdsp);
+#endif
+
+static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand)
+{
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
+ return 0;
+ if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
+ hdsp->state &= ~HDSP_FirmwareLoaded;
+ if (! load_on_demand)
+ return -EIO;
+ snd_printk(KERN_ERR "Hammerfall-DSP: firmware not present.\n");
+ /* try to load firmware */
+ if (! (hdsp->state & HDSP_FirmwareCached)) {
+#ifdef HDSP_FW_LOADER
+ if (! hdsp_request_fw_loader(hdsp))
+ return 0;
+#endif
+ snd_printk(KERN_ERR
+ "Hammerfall-DSP: No firmware loaded nor "
+ "cached, please upload firmware.\n");
+ return -EIO;
+ }
+ if (snd_hdsp_load_firmware_from_cache(hdsp) != 0) {
+ snd_printk(KERN_ERR
+ "Hammerfall-DSP: Firmware loading from "
+ "cache failed, please upload manually.\n");
+ return -EIO;
+ }
+ }
+ return 0;
+}
+
+
+static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout)
+{
+ int i;
+
+ /* the fifoStatus registers reports on how many words
+ are available in the command FIFO.
+ */
+
+ for (i = 0; i < timeout; i++) {
+
+ if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count)
+ return 0;
+
+ /* not very friendly, but we only do this during a firmware
+ load and changing the mixer, so we just put up with it.
+ */
+
+ udelay (100);
+ }
+
+ snd_printk ("Hammerfall-DSP: wait for FIFO status <= %d failed after %d iterations\n",
+ count, timeout);
+ return -1;
+}
+
+static int hdsp_read_gain (struct hdsp *hdsp, unsigned int addr)
+{
+ if (addr >= HDSP_MATRIX_MIXER_SIZE)
+ return 0;
+
+ return hdsp->mixer_matrix[addr];
+}
+
+static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short data)
+{
+ unsigned int ad;
+
+ if (addr >= HDSP_MATRIX_MIXER_SIZE)
+ return -1;
+
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632) {
+
+ /* from martin bjornsen:
+
+ "You can only write dwords to the
+ mixer memory which contain two
+ mixer values in the low and high
+ word. So if you want to change
+ value 0 you have to read value 1
+ from the cache and write both to
+ the first dword in the mixer
+ memory."
+ */
+
+ if (hdsp->io_type == H9632 && addr >= 512)
+ return 0;
+
+ if (hdsp->io_type == H9652 && addr >= 1352)
+ return 0;
+
+ hdsp->mixer_matrix[addr] = data;
+
+
+ /* `addr' addresses a 16-bit wide address, but
+ the address space accessed via hdsp_write
+ uses byte offsets. put another way, addr
+ varies from 0 to 1351, but to access the
+ corresponding memory location, we need
+ to access 0 to 2703 ...
+ */
+ ad = addr/2;
+
+ hdsp_write (hdsp, 4096 + (ad*4),
+ (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) +
+ hdsp->mixer_matrix[addr&0x7fe]);
+
+ return 0;
+
+ } else {
+
+ ad = (addr << 16) + data;
+
+ if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT))
+ return -1;
+
+ hdsp_write (hdsp, HDSP_fifoData, ad);
+ hdsp->mixer_matrix[addr] = data;
+
+ }
+
+ return 0;
+}
+
+static int snd_hdsp_use_is_exclusive(struct hdsp *hdsp)
+{
+ unsigned long flags;
+ int ret = 1;
+
+ spin_lock_irqsave(&hdsp->lock, flags);
+ if ((hdsp->playback_pid != hdsp->capture_pid) &&
+ (hdsp->playback_pid >= 0) && (hdsp->capture_pid >= 0))
+ ret = 0;
+ spin_unlock_irqrestore(&hdsp->lock, flags);
+ return ret;
+}
+
+static int hdsp_spdif_sample_rate(struct hdsp *hdsp)
+{
+ unsigned int status = hdsp_read(hdsp, HDSP_statusRegister);
+ unsigned int rate_bits = (status & HDSP_spdifFrequencyMask);
+
+ /* For the 9632, the mask is different */
+ if (hdsp->io_type == H9632)
+ rate_bits = (status & HDSP_spdifFrequencyMask_9632);
+
+ if (status & HDSP_SPDIFErrorFlag)
+ return 0;
+
+ switch (rate_bits) {
+ case HDSP_spdifFrequency32KHz: return 32000;
+ case HDSP_spdifFrequency44_1KHz: return 44100;
+ case HDSP_spdifFrequency48KHz: return 48000;
+ case HDSP_spdifFrequency64KHz: return 64000;
+ case HDSP_spdifFrequency88_2KHz: return 88200;
+ case HDSP_spdifFrequency96KHz: return 96000;
+ case HDSP_spdifFrequency128KHz:
+ if (hdsp->io_type == H9632) return 128000;
+ break;
+ case HDSP_spdifFrequency176_4KHz:
+ if (hdsp->io_type == H9632) return 176400;
+ break;
+ case HDSP_spdifFrequency192KHz:
+ if (hdsp->io_type == H9632) return 192000;
+ break;
+ default:
+ break;
+ }
+ snd_printk ("Hammerfall-DSP: unknown spdif frequency status; bits = 0x%x, status = 0x%x\n", rate_bits, status);
+ return 0;
+}
+
+static int hdsp_external_sample_rate(struct hdsp *hdsp)
+{
+ unsigned int status2 = hdsp_read(hdsp, HDSP_status2Register);
+ unsigned int rate_bits = status2 & HDSP_systemFrequencyMask;
+
+ /* For the 9632 card, there seems to be no bit for indicating external
+ * sample rate greater than 96kHz. The card reports the corresponding
+ * single speed. So the best means seems to get spdif rate when
+ * autosync reference is spdif */
+ if (hdsp->io_type == H9632 &&
+ hdsp_autosync_ref(hdsp) == HDSP_AUTOSYNC_FROM_SPDIF)
+ return hdsp_spdif_sample_rate(hdsp);
+
+ switch (rate_bits) {
+ case HDSP_systemFrequency32: return 32000;
+ case HDSP_systemFrequency44_1: return 44100;
+ case HDSP_systemFrequency48: return 48000;
+ case HDSP_systemFrequency64: return 64000;
+ case HDSP_systemFrequency88_2: return 88200;
+ case HDSP_systemFrequency96: return 96000;
+ default:
+ return 0;
+ }
+}
+
+static void hdsp_compute_period_size(struct hdsp *hdsp)
+{
+ hdsp->period_bytes = 1 << ((hdsp_decode_latency(hdsp->control_register) + 8));
+}
+
+static snd_pcm_uframes_t hdsp_hw_pointer(struct hdsp *hdsp)
+{
+ int position;
+
+ position = hdsp_read(hdsp, HDSP_statusRegister);
+
+ if (!hdsp->precise_ptr)
+ return (position & HDSP_BufferID) ? (hdsp->period_bytes / 4) : 0;
+
+ position &= HDSP_BufferPositionMask;
+ position /= 4;
+ position &= (hdsp->period_bytes/2) - 1;
+ return position;
+}
+
+static void hdsp_reset_hw_pointer(struct hdsp *hdsp)
+{
+ hdsp_write (hdsp, HDSP_resetPointer, 0);
+ if (hdsp->io_type == H9632 && hdsp->firmware_rev >= 152)
+ /* HDSP_resetPointer = HDSP_freqReg, which is strange and
+ * requires (?) to write again DDS value after a reset pointer
+ * (at least, it works like this) */
+ hdsp_write (hdsp, HDSP_freqReg, hdsp->dds_value);
+}
+
+static void hdsp_start_audio(struct hdsp *s)
+{
+ s->control_register |= (HDSP_AudioInterruptEnable | HDSP_Start);
+ hdsp_write(s, HDSP_controlRegister, s->control_register);
+}
+
+static void hdsp_stop_audio(struct hdsp *s)
+{
+ s->control_register &= ~(HDSP_Start | HDSP_AudioInterruptEnable);
+ hdsp_write(s, HDSP_controlRegister, s->control_register);
+}
+
+static void hdsp_silence_playback(struct hdsp *hdsp)
+{
+ memset(hdsp->playback_buffer, 0, HDSP_DMA_AREA_BYTES);
+}
+
+static int hdsp_set_interrupt_interval(struct hdsp *s, unsigned int frames)
+{
+ int n;
+
+ spin_lock_irq(&s->lock);
+
+ frames >>= 7;
+ n = 0;
+ while (frames) {
+ n++;
+ frames >>= 1;
+ }
+
+ s->control_register &= ~HDSP_LatencyMask;
+ s->control_register |= hdsp_encode_latency(n);
+
+ hdsp_write(s, HDSP_controlRegister, s->control_register);
+
+ hdsp_compute_period_size(s);
+
+ spin_unlock_irq(&s->lock);
+
+ return 0;
+}
+
+static void hdsp_set_dds_value(struct hdsp *hdsp, int rate)
+{
+ u64 n;
+ u32 r;
+
+ if (rate >= 112000)
+ rate /= 4;
+ else if (rate >= 56000)
+ rate /= 2;
+
+ n = DDS_NUMERATOR;
+ div64_32(&n, rate, &r);
+ /* n should be less than 2^32 for being written to FREQ register */
+ snd_BUG_ON(n >> 32);
+ /* HDSP_freqReg and HDSP_resetPointer are the same, so keep the DDS
+ value to write it after a reset */
+ hdsp->dds_value = n;
+ hdsp_write(hdsp, HDSP_freqReg, hdsp->dds_value);
+}
+
+static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally)
+{
+ int reject_if_open = 0;
+ int current_rate;
+ int rate_bits;
+
+ /* ASSUMPTION: hdsp->lock is either held, or
+ there is no need for it (e.g. during module
+ initialization).
+ */
+
+ if (!(hdsp->control_register & HDSP_ClockModeMaster)) {
+ if (called_internally) {
+ /* request from ctl or card initialization */
+ snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n");
+ return -1;
+ } else {
+ /* hw_param request while in AutoSync mode */
+ int external_freq = hdsp_external_sample_rate(hdsp);
+ int spdif_freq = hdsp_spdif_sample_rate(hdsp);
+
+ if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1))
+ snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n");
+ else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1))
+ snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n");
+ else if (rate != external_freq) {
+ snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n");
+ return -1;
+ }
+ }
+ }
+
+ current_rate = hdsp->system_sample_rate;
+
+ /* Changing from a "single speed" to a "double speed" rate is
+ not allowed if any substreams are open. This is because
+ such a change causes a shift in the location of
+ the DMA buffers and a reduction in the number of available
+ buffers.
+
+ Note that a similar but essentially insoluble problem
+ exists for externally-driven rate changes. All we can do
+ is to flag rate changes in the read/write routines. */
+
+ if (rate > 96000 && hdsp->io_type != H9632)
+ return -EINVAL;
+
+ switch (rate) {
+ case 32000:
+ if (current_rate > 48000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency32KHz;
+ break;
+ case 44100:
+ if (current_rate > 48000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency44_1KHz;
+ break;
+ case 48000:
+ if (current_rate > 48000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency48KHz;
+ break;
+ case 64000:
+ if (current_rate <= 48000 || current_rate > 96000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency64KHz;
+ break;
+ case 88200:
+ if (current_rate <= 48000 || current_rate > 96000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency88_2KHz;
+ break;
+ case 96000:
+ if (current_rate <= 48000 || current_rate > 96000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency96KHz;
+ break;
+ case 128000:
+ if (current_rate < 128000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency128KHz;
+ break;
+ case 176400:
+ if (current_rate < 128000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency176_4KHz;
+ break;
+ case 192000:
+ if (current_rate < 128000)
+ reject_if_open = 1;
+ rate_bits = HDSP_Frequency192KHz;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (reject_if_open && (hdsp->capture_pid >= 0 || hdsp->playback_pid >= 0)) {
+ snd_printk ("Hammerfall-DSP: cannot change speed mode (capture PID = %d, playback PID = %d)\n",
+ hdsp->capture_pid,
+ hdsp->playback_pid);
+ return -EBUSY;
+ }
+
+ hdsp->control_register &= ~HDSP_FrequencyMask;
+ hdsp->control_register |= rate_bits;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+
+ /* For HDSP9632 rev 152, need to set DDS value in FREQ register */
+ if (hdsp->io_type == H9632 && hdsp->firmware_rev >= 152)
+ hdsp_set_dds_value(hdsp, rate);
+
+ if (rate >= 128000) {
+ hdsp->channel_map = channel_map_H9632_qs;
+ } else if (rate > 48000) {
+ if (hdsp->io_type == H9632)
+ hdsp->channel_map = channel_map_H9632_ds;
+ else
+ hdsp->channel_map = channel_map_ds;
+ } else {
+ switch (hdsp->io_type) {
+ case Multiface:
+ hdsp->channel_map = channel_map_mf_ss;
+ break;
+ case Digiface:
+ case H9652:
+ hdsp->channel_map = channel_map_df_ss;
+ break;
+ case H9632:
+ hdsp->channel_map = channel_map_H9632_ss;
+ break;
+ default:
+ /* should never happen */
+ break;
+ }
+ }
+
+ hdsp->system_sample_rate = rate;
+
+ return 0;
+}
+
+/*----------------------------------------------------------------------------
+ MIDI
+ ----------------------------------------------------------------------------*/
+
+static unsigned char snd_hdsp_midi_read_byte (struct hdsp *hdsp, int id)
+{
+ /* the hardware already does the relevant bit-mask with 0xff */
+ if (id)
+ return hdsp_read(hdsp, HDSP_midiDataIn1);
+ else
+ return hdsp_read(hdsp, HDSP_midiDataIn0);
+}
+
+static void snd_hdsp_midi_write_byte (struct hdsp *hdsp, int id, int val)
+{
+ /* the hardware already does the relevant bit-mask with 0xff */
+ if (id)
+ hdsp_write(hdsp, HDSP_midiDataOut1, val);
+ else
+ hdsp_write(hdsp, HDSP_midiDataOut0, val);
+}
+
+static int snd_hdsp_midi_input_available (struct hdsp *hdsp, int id)
+{
+ if (id)
+ return (hdsp_read(hdsp, HDSP_midiStatusIn1) & 0xff);
+ else
+ return (hdsp_read(hdsp, HDSP_midiStatusIn0) & 0xff);
+}
+
+static int snd_hdsp_midi_output_possible (struct hdsp *hdsp, int id)
+{
+ int fifo_bytes_used;
+
+ if (id)
+ fifo_bytes_used = hdsp_read(hdsp, HDSP_midiStatusOut1) & 0xff;
+ else
+ fifo_bytes_used = hdsp_read(hdsp, HDSP_midiStatusOut0) & 0xff;
+
+ if (fifo_bytes_used < 128)
+ return 128 - fifo_bytes_used;
+ else
+ return 0;
+}
+
+static void snd_hdsp_flush_midi_input (struct hdsp *hdsp, int id)
+{
+ while (snd_hdsp_midi_input_available (hdsp, id))
+ snd_hdsp_midi_read_byte (hdsp, id);
+}
+
+static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi)
+{
+ unsigned long flags;
+ int n_pending;
+ int to_write;
+ int i;
+ unsigned char buf[128];
+
+ /* Output is not interrupt driven */
+
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if (hmidi->output) {
+ if (!snd_rawmidi_transmit_empty (hmidi->output)) {
+ if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) {
+ if (n_pending > (int)sizeof (buf))
+ n_pending = sizeof (buf);
+
+ if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) {
+ for (i = 0; i < to_write; ++i)
+ snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]);
+ }
+ }
+ }
+ }
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ return 0;
+}
+
+static int snd_hdsp_midi_input_read (struct hdsp_midi *hmidi)
+{
+ unsigned char buf[128]; /* this buffer is designed to match the MIDI input FIFO size */
+ unsigned long flags;
+ int n_pending;
+ int i;
+
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if ((n_pending = snd_hdsp_midi_input_available (hmidi->hdsp, hmidi->id)) > 0) {
+ if (hmidi->input) {
+ if (n_pending > (int)sizeof (buf))
+ n_pending = sizeof (buf);
+ for (i = 0; i < n_pending; ++i)
+ buf[i] = snd_hdsp_midi_read_byte (hmidi->hdsp, hmidi->id);
+ if (n_pending)
+ snd_rawmidi_receive (hmidi->input, buf, n_pending);
+ } else {
+ /* flush the MIDI input FIFO */
+ while (--n_pending)
+ snd_hdsp_midi_read_byte (hmidi->hdsp, hmidi->id);
+ }
+ }
+ hmidi->pending = 0;
+ if (hmidi->id)
+ hmidi->hdsp->control_register |= HDSP_Midi1InterruptEnable;
+ else
+ hmidi->hdsp->control_register |= HDSP_Midi0InterruptEnable;
+ hdsp_write(hmidi->hdsp, HDSP_controlRegister, hmidi->hdsp->control_register);
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ return snd_hdsp_midi_output_write (hmidi);
+}
+
+static void snd_hdsp_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct hdsp *hdsp;
+ struct hdsp_midi *hmidi;
+ unsigned long flags;
+ u32 ie;
+
+ hmidi = (struct hdsp_midi *) substream->rmidi->private_data;
+ hdsp = hmidi->hdsp;
+ ie = hmidi->id ? HDSP_Midi1InterruptEnable : HDSP_Midi0InterruptEnable;
+ spin_lock_irqsave (&hdsp->lock, flags);
+ if (up) {
+ if (!(hdsp->control_register & ie)) {
+ snd_hdsp_flush_midi_input (hdsp, hmidi->id);
+ hdsp->control_register |= ie;
+ }
+ } else {
+ hdsp->control_register &= ~ie;
+ tasklet_kill(&hdsp->midi_tasklet);
+ }
+
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ spin_unlock_irqrestore (&hdsp->lock, flags);
+}
+
+static void snd_hdsp_midi_output_timer(unsigned long data)
+{
+ struct hdsp_midi *hmidi = (struct hdsp_midi *) data;
+ unsigned long flags;
+
+ snd_hdsp_midi_output_write(hmidi);
+ spin_lock_irqsave (&hmidi->lock, flags);
+
+ /* this does not bump hmidi->istimer, because the
+ kernel automatically removed the timer when it
+ expired, and we are now adding it back, thus
+ leaving istimer wherever it was set before.
+ */
+
+ if (hmidi->istimer) {
+ hmidi->timer.expires = 1 + jiffies;
+ add_timer(&hmidi->timer);
+ }
+
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+}
+
+static void snd_hdsp_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct hdsp_midi *hmidi;
+ unsigned long flags;
+
+ hmidi = (struct hdsp_midi *) substream->rmidi->private_data;
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if (up) {
+ if (!hmidi->istimer) {
+ init_timer(&hmidi->timer);
+ hmidi->timer.function = snd_hdsp_midi_output_timer;
+ hmidi->timer.data = (unsigned long) hmidi;
+ hmidi->timer.expires = 1 + jiffies;
+ add_timer(&hmidi->timer);
+ hmidi->istimer++;
+ }
+ } else {
+ if (hmidi->istimer && --hmidi->istimer <= 0)
+ del_timer (&hmidi->timer);
+ }
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ if (up)
+ snd_hdsp_midi_output_write(hmidi);
+}
+
+static int snd_hdsp_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct hdsp_midi *hmidi;
+
+ hmidi = (struct hdsp_midi *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ snd_hdsp_flush_midi_input (hmidi->hdsp, hmidi->id);
+ hmidi->input = substream;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdsp_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct hdsp_midi *hmidi;
+
+ hmidi = (struct hdsp_midi *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->output = substream;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdsp_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct hdsp_midi *hmidi;
+
+ snd_hdsp_midi_input_trigger (substream, 0);
+
+ hmidi = (struct hdsp_midi *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->input = NULL;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdsp_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct hdsp_midi *hmidi;
+
+ snd_hdsp_midi_output_trigger (substream, 0);
+
+ hmidi = (struct hdsp_midi *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->output = NULL;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static struct snd_rawmidi_ops snd_hdsp_midi_output =
+{
+ .open = snd_hdsp_midi_output_open,
+ .close = snd_hdsp_midi_output_close,
+ .trigger = snd_hdsp_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_hdsp_midi_input =
+{
+ .open = snd_hdsp_midi_input_open,
+ .close = snd_hdsp_midi_input_close,
+ .trigger = snd_hdsp_midi_input_trigger,
+};
+
+static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int id)
+{
+ char buf[32];
+
+ hdsp->midi[id].id = id;
+ hdsp->midi[id].rmidi = NULL;
+ hdsp->midi[id].input = NULL;
+ hdsp->midi[id].output = NULL;
+ hdsp->midi[id].hdsp = hdsp;
+ hdsp->midi[id].istimer = 0;
+ hdsp->midi[id].pending = 0;
+ spin_lock_init (&hdsp->midi[id].lock);
+
+ sprintf (buf, "%s MIDI %d", card->shortname, id+1);
+ if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0)
+ return -1;
+
+ sprintf (hdsp->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1);
+ hdsp->midi[id].rmidi->private_data = &hdsp->midi[id];
+
+ snd_rawmidi_set_ops (hdsp->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_hdsp_midi_output);
+ snd_rawmidi_set_ops (hdsp->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_hdsp_midi_input);
+
+ hdsp->midi[id].rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return 0;
+}
+
+/*-----------------------------------------------------------------------------
+ Control Interface
+ ----------------------------------------------------------------------------*/
+
+static u32 snd_hdsp_convert_from_aes(struct snd_aes_iec958 *aes)
+{
+ u32 val = 0;
+ val |= (aes->status[0] & IEC958_AES0_PROFESSIONAL) ? HDSP_SPDIFProfessional : 0;
+ val |= (aes->status[0] & IEC958_AES0_NONAUDIO) ? HDSP_SPDIFNonAudio : 0;
+ if (val & HDSP_SPDIFProfessional)
+ val |= (aes->status[0] & IEC958_AES0_PRO_EMPHASIS_5015) ? HDSP_SPDIFEmphasis : 0;
+ else
+ val |= (aes->status[0] & IEC958_AES0_CON_EMPHASIS_5015) ? HDSP_SPDIFEmphasis : 0;
+ return val;
+}
+
+static void snd_hdsp_convert_to_aes(struct snd_aes_iec958 *aes, u32 val)
+{
+ aes->status[0] = ((val & HDSP_SPDIFProfessional) ? IEC958_AES0_PROFESSIONAL : 0) |
+ ((val & HDSP_SPDIFNonAudio) ? IEC958_AES0_NONAUDIO : 0);
+ if (val & HDSP_SPDIFProfessional)
+ aes->status[0] |= (val & HDSP_SPDIFEmphasis) ? IEC958_AES0_PRO_EMPHASIS_5015 : 0;
+ else
+ aes->status[0] |= (val & HDSP_SPDIFEmphasis) ? IEC958_AES0_CON_EMPHASIS_5015 : 0;
+}
+
+static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif);
+ return 0;
+}
+
+static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&hdsp->lock);
+ change = val != hdsp->creg_spdif;
+ hdsp->creg_spdif = val;
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream);
+ return 0;
+}
+
+static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&hdsp->lock);
+ change = val != hdsp->creg_spdif_stream;
+ hdsp->creg_spdif_stream = val;
+ hdsp->control_register &= ~(HDSP_SPDIFProfessional | HDSP_SPDIFNonAudio | HDSP_SPDIFEmphasis);
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register |= val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+static int snd_hdsp_control_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_hdsp_control_spdif_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = kcontrol->private_value;
+ return 0;
+}
+
+#define HDSP_SPDIF_IN(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_spdif_in, \
+ .get = snd_hdsp_get_spdif_in, \
+ .put = snd_hdsp_put_spdif_in }
+
+static unsigned int hdsp_spdif_in(struct hdsp *hdsp)
+{
+ return hdsp_decode_spdif_in(hdsp->control_register & HDSP_SPDIFInputMask);
+}
+
+static int hdsp_set_spdif_input(struct hdsp *hdsp, int in)
+{
+ hdsp->control_register &= ~HDSP_SPDIFInputMask;
+ hdsp->control_register |= hdsp_encode_spdif_in(in);
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {"Optical", "Coaxial", "Internal", "AES"};
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = ((hdsp->io_type == H9632) ? 4 : 3);
+ if (uinfo->value.enumerated.item > ((hdsp->io_type == H9632) ? 3 : 2))
+ uinfo->value.enumerated.item = ((hdsp->io_type == H9632) ? 3 : 2);
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3);
+ spin_lock_irq(&hdsp->lock);
+ change = val != hdsp_spdif_in(hdsp);
+ if (change)
+ hdsp_set_spdif_input(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_SPDIF_OUT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_hdsp_info_spdif_bits, \
+ .get = snd_hdsp_get_spdif_out, .put = snd_hdsp_put_spdif_out }
+
+static int hdsp_spdif_out(struct hdsp *hdsp)
+{
+ return (hdsp->control_register & HDSP_SPDIFOpticalOut) ? 1 : 0;
+}
+
+static int hdsp_set_spdif_output(struct hdsp *hdsp, int out)
+{
+ if (out)
+ hdsp->control_register |= HDSP_SPDIFOpticalOut;
+ else
+ hdsp->control_register &= ~HDSP_SPDIFOpticalOut;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+#define snd_hdsp_info_spdif_bits snd_ctl_boolean_mono_info
+
+static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_spdif_out(hdsp);
+ hdsp_set_spdif_output(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_SPDIF_PROFESSIONAL(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_hdsp_info_spdif_bits, \
+ .get = snd_hdsp_get_spdif_professional, .put = snd_hdsp_put_spdif_professional }
+
+static int hdsp_spdif_professional(struct hdsp *hdsp)
+{
+ return (hdsp->control_register & HDSP_SPDIFProfessional) ? 1 : 0;
+}
+
+static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val)
+{
+ if (val)
+ hdsp->control_register |= HDSP_SPDIFProfessional;
+ else
+ hdsp->control_register &= ~HDSP_SPDIFProfessional;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_spdif_professional(hdsp);
+ hdsp_set_spdif_professional(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_SPDIF_EMPHASIS(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_hdsp_info_spdif_bits, \
+ .get = snd_hdsp_get_spdif_emphasis, .put = snd_hdsp_put_spdif_emphasis }
+
+static int hdsp_spdif_emphasis(struct hdsp *hdsp)
+{
+ return (hdsp->control_register & HDSP_SPDIFEmphasis) ? 1 : 0;
+}
+
+static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val)
+{
+ if (val)
+ hdsp->control_register |= HDSP_SPDIFEmphasis;
+ else
+ hdsp->control_register &= ~HDSP_SPDIFEmphasis;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_spdif_emphasis(hdsp);
+ hdsp_set_spdif_emphasis(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_SPDIF_NON_AUDIO(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_hdsp_info_spdif_bits, \
+ .get = snd_hdsp_get_spdif_nonaudio, .put = snd_hdsp_put_spdif_nonaudio }
+
+static int hdsp_spdif_nonaudio(struct hdsp *hdsp)
+{
+ return (hdsp->control_register & HDSP_SPDIFNonAudio) ? 1 : 0;
+}
+
+static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val)
+{
+ if (val)
+ hdsp->control_register |= HDSP_SPDIFNonAudio;
+ else
+ hdsp->control_register &= ~HDSP_SPDIFNonAudio;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_spdif_nonaudio(hdsp);
+ hdsp_set_spdif_nonaudio(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_SPDIF_SAMPLE_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdsp_info_spdif_sample_rate, \
+ .get = snd_hdsp_get_spdif_sample_rate \
+}
+
+static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ switch (hdsp_spdif_sample_rate(hdsp)) {
+ case 32000:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ case 44100:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ case 48000:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case 64000:
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case 88200:
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ case 96000:
+ ucontrol->value.enumerated.item[0] = 5;
+ break;
+ case 128000:
+ ucontrol->value.enumerated.item[0] = 7;
+ break;
+ case 176400:
+ ucontrol->value.enumerated.item[0] = 8;
+ break;
+ case 192000:
+ ucontrol->value.enumerated.item[0] = 9;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 6;
+ }
+ return 0;
+}
+
+#define HDSP_SYSTEM_SAMPLE_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdsp_info_system_sample_rate, \
+ .get = snd_hdsp_get_system_sample_rate \
+}
+
+static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate;
+ return 0;
+}
+
+#define HDSP_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdsp_info_autosync_sample_rate, \
+ .get = snd_hdsp_get_autosync_sample_rate \
+}
+
+static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ switch (hdsp_external_sample_rate(hdsp)) {
+ case 32000:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ case 44100:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ case 48000:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case 64000:
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case 88200:
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ case 96000:
+ ucontrol->value.enumerated.item[0] = 5;
+ break;
+ case 128000:
+ ucontrol->value.enumerated.item[0] = 7;
+ break;
+ case 176400:
+ ucontrol->value.enumerated.item[0] = 8;
+ break;
+ case 192000:
+ ucontrol->value.enumerated.item[0] = 9;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 6;
+ }
+ return 0;
+}
+
+#define HDSP_SYSTEM_CLOCK_MODE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdsp_info_system_clock_mode, \
+ .get = snd_hdsp_get_system_clock_mode \
+}
+
+static int hdsp_system_clock_mode(struct hdsp *hdsp)
+{
+ if (hdsp->control_register & HDSP_ClockModeMaster)
+ return 0;
+ else if (hdsp_external_sample_rate(hdsp) != hdsp->system_sample_rate)
+ return 0;
+ return 1;
+}
+
+static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"Master", "Slave" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp);
+ return 0;
+}
+
+#define HDSP_CLOCK_SOURCE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_clock_source, \
+ .get = snd_hdsp_get_clock_source, \
+ .put = snd_hdsp_put_clock_source \
+}
+
+static int hdsp_clock_source(struct hdsp *hdsp)
+{
+ if (hdsp->control_register & HDSP_ClockModeMaster) {
+ switch (hdsp->system_sample_rate) {
+ case 32000:
+ return 1;
+ case 44100:
+ return 2;
+ case 48000:
+ return 3;
+ case 64000:
+ return 4;
+ case 88200:
+ return 5;
+ case 96000:
+ return 6;
+ case 128000:
+ return 7;
+ case 176400:
+ return 8;
+ case 192000:
+ return 9;
+ default:
+ return 3;
+ }
+ } else {
+ return 0;
+ }
+}
+
+static int hdsp_set_clock_source(struct hdsp *hdsp, int mode)
+{
+ int rate;
+ switch (mode) {
+ case HDSP_CLOCK_SOURCE_AUTOSYNC:
+ if (hdsp_external_sample_rate(hdsp) != 0) {
+ if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) {
+ hdsp->control_register &= ~HDSP_ClockModeMaster;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+ }
+ }
+ return -1;
+ case HDSP_CLOCK_SOURCE_INTERNAL_32KHZ:
+ rate = 32000;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ:
+ rate = 44100;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ:
+ rate = 48000;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_64KHZ:
+ rate = 64000;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_88_2KHZ:
+ rate = 88200;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_96KHZ:
+ rate = 96000;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_128KHZ:
+ rate = 128000;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_176_4KHZ:
+ rate = 176400;
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ:
+ rate = 192000;
+ break;
+ default:
+ rate = 48000;
+ }
+ hdsp->control_register |= HDSP_ClockModeMaster;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ hdsp_set_rate(hdsp, rate, 1);
+ return 0;
+}
+
+static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" };
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ if (hdsp->io_type == H9632)
+ uinfo->value.enumerated.items = 10;
+ else
+ uinfo->value.enumerated.items = 7;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0];
+ if (val < 0) val = 0;
+ if (hdsp->io_type == H9632) {
+ if (val > 9)
+ val = 9;
+ } else {
+ if (val > 6)
+ val = 6;
+ }
+ spin_lock_irq(&hdsp->lock);
+ if (val != hdsp_clock_source(hdsp))
+ change = (hdsp_set_clock_source(hdsp, val) == 0) ? 1 : 0;
+ else
+ change = 0;
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define snd_hdsp_info_clock_source_lock snd_ctl_boolean_mono_info
+
+static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = hdsp->clock_source_locked;
+ return 0;
+}
+
+static int snd_hdsp_put_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+
+ change = (int)ucontrol->value.integer.value[0] != hdsp->clock_source_locked;
+ if (change)
+ hdsp->clock_source_locked = !!ucontrol->value.integer.value[0];
+ return change;
+}
+
+#define HDSP_DA_GAIN(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_da_gain, \
+ .get = snd_hdsp_get_da_gain, \
+ .put = snd_hdsp_put_da_gain \
+}
+
+static int hdsp_da_gain(struct hdsp *hdsp)
+{
+ switch (hdsp->control_register & HDSP_DAGainMask) {
+ case HDSP_DAGainHighGain:
+ return 0;
+ case HDSP_DAGainPlus4dBu:
+ return 1;
+ case HDSP_DAGainMinus10dBV:
+ return 2;
+ default:
+ return 1;
+ }
+}
+
+static int hdsp_set_da_gain(struct hdsp *hdsp, int mode)
+{
+ hdsp->control_register &= ~HDSP_DAGainMask;
+ switch (mode) {
+ case 0:
+ hdsp->control_register |= HDSP_DAGainHighGain;
+ break;
+ case 1:
+ hdsp->control_register |= HDSP_DAGainPlus4dBu;
+ break;
+ case 2:
+ hdsp->control_register |= HDSP_DAGainMinus10dBV;
+ break;
+ default:
+ return -1;
+
+ }
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0];
+ if (val < 0) val = 0;
+ if (val > 2) val = 2;
+ spin_lock_irq(&hdsp->lock);
+ if (val != hdsp_da_gain(hdsp))
+ change = (hdsp_set_da_gain(hdsp, val) == 0) ? 1 : 0;
+ else
+ change = 0;
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_AD_GAIN(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_ad_gain, \
+ .get = snd_hdsp_get_ad_gain, \
+ .put = snd_hdsp_put_ad_gain \
+}
+
+static int hdsp_ad_gain(struct hdsp *hdsp)
+{
+ switch (hdsp->control_register & HDSP_ADGainMask) {
+ case HDSP_ADGainMinus10dBV:
+ return 0;
+ case HDSP_ADGainPlus4dBu:
+ return 1;
+ case HDSP_ADGainLowGain:
+ return 2;
+ default:
+ return 1;
+ }
+}
+
+static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode)
+{
+ hdsp->control_register &= ~HDSP_ADGainMask;
+ switch (mode) {
+ case 0:
+ hdsp->control_register |= HDSP_ADGainMinus10dBV;
+ break;
+ case 1:
+ hdsp->control_register |= HDSP_ADGainPlus4dBu;
+ break;
+ case 2:
+ hdsp->control_register |= HDSP_ADGainLowGain;
+ break;
+ default:
+ return -1;
+
+ }
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0];
+ if (val < 0) val = 0;
+ if (val > 2) val = 2;
+ spin_lock_irq(&hdsp->lock);
+ if (val != hdsp_ad_gain(hdsp))
+ change = (hdsp_set_ad_gain(hdsp, val) == 0) ? 1 : 0;
+ else
+ change = 0;
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_PHONE_GAIN(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_phone_gain, \
+ .get = snd_hdsp_get_phone_gain, \
+ .put = snd_hdsp_put_phone_gain \
+}
+
+static int hdsp_phone_gain(struct hdsp *hdsp)
+{
+ switch (hdsp->control_register & HDSP_PhoneGainMask) {
+ case HDSP_PhoneGain0dB:
+ return 0;
+ case HDSP_PhoneGainMinus6dB:
+ return 1;
+ case HDSP_PhoneGainMinus12dB:
+ return 2;
+ default:
+ return 0;
+ }
+}
+
+static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode)
+{
+ hdsp->control_register &= ~HDSP_PhoneGainMask;
+ switch (mode) {
+ case 0:
+ hdsp->control_register |= HDSP_PhoneGain0dB;
+ break;
+ case 1:
+ hdsp->control_register |= HDSP_PhoneGainMinus6dB;
+ break;
+ case 2:
+ hdsp->control_register |= HDSP_PhoneGainMinus12dB;
+ break;
+ default:
+ return -1;
+
+ }
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"0 dB", "-6 dB", "-12 dB"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0];
+ if (val < 0) val = 0;
+ if (val > 2) val = 2;
+ spin_lock_irq(&hdsp->lock);
+ if (val != hdsp_phone_gain(hdsp))
+ change = (hdsp_set_phone_gain(hdsp, val) == 0) ? 1 : 0;
+ else
+ change = 0;
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_XLR_BREAKOUT_CABLE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_xlr_breakout_cable, \
+ .get = snd_hdsp_get_xlr_breakout_cable, \
+ .put = snd_hdsp_put_xlr_breakout_cable \
+}
+
+static int hdsp_xlr_breakout_cable(struct hdsp *hdsp)
+{
+ if (hdsp->control_register & HDSP_XLRBreakoutCable)
+ return 1;
+ return 0;
+}
+
+static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode)
+{
+ if (mode)
+ hdsp->control_register |= HDSP_XLRBreakoutCable;
+ else
+ hdsp->control_register &= ~HDSP_XLRBreakoutCable;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+#define snd_hdsp_info_xlr_breakout_cable snd_ctl_boolean_mono_info
+
+static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_xlr_breakout_cable(hdsp);
+ hdsp_set_xlr_breakout_cable(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+/* (De)activates old RME Analog Extension Board
+ These are connected to the internal ADAT connector
+ Switching this on desactivates external ADAT
+*/
+#define HDSP_AEB(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_aeb, \
+ .get = snd_hdsp_get_aeb, \
+ .put = snd_hdsp_put_aeb \
+}
+
+static int hdsp_aeb(struct hdsp *hdsp)
+{
+ if (hdsp->control_register & HDSP_AnalogExtensionBoard)
+ return 1;
+ return 0;
+}
+
+static int hdsp_set_aeb(struct hdsp *hdsp, int mode)
+{
+ if (mode)
+ hdsp->control_register |= HDSP_AnalogExtensionBoard;
+ else
+ hdsp->control_register &= ~HDSP_AnalogExtensionBoard;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+#define snd_hdsp_info_aeb snd_ctl_boolean_mono_info
+
+static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_aeb(hdsp);
+ hdsp_set_aeb(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_PREF_SYNC_REF(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_pref_sync_ref, \
+ .get = snd_hdsp_get_pref_sync_ref, \
+ .put = snd_hdsp_put_pref_sync_ref \
+}
+
+static int hdsp_pref_sync_ref(struct hdsp *hdsp)
+{
+ /* Notice that this looks at the requested sync source,
+ not the one actually in use.
+ */
+
+ switch (hdsp->control_register & HDSP_SyncRefMask) {
+ case HDSP_SyncRef_ADAT1:
+ return HDSP_SYNC_FROM_ADAT1;
+ case HDSP_SyncRef_ADAT2:
+ return HDSP_SYNC_FROM_ADAT2;
+ case HDSP_SyncRef_ADAT3:
+ return HDSP_SYNC_FROM_ADAT3;
+ case HDSP_SyncRef_SPDIF:
+ return HDSP_SYNC_FROM_SPDIF;
+ case HDSP_SyncRef_WORD:
+ return HDSP_SYNC_FROM_WORD;
+ case HDSP_SyncRef_ADAT_SYNC:
+ return HDSP_SYNC_FROM_ADAT_SYNC;
+ default:
+ return HDSP_SYNC_FROM_WORD;
+ }
+ return 0;
+}
+
+static int hdsp_set_pref_sync_ref(struct hdsp *hdsp, int pref)
+{
+ hdsp->control_register &= ~HDSP_SyncRefMask;
+ switch (pref) {
+ case HDSP_SYNC_FROM_ADAT1:
+ hdsp->control_register &= ~HDSP_SyncRefMask; /* clear SyncRef bits */
+ break;
+ case HDSP_SYNC_FROM_ADAT2:
+ hdsp->control_register |= HDSP_SyncRef_ADAT2;
+ break;
+ case HDSP_SYNC_FROM_ADAT3:
+ hdsp->control_register |= HDSP_SyncRef_ADAT3;
+ break;
+ case HDSP_SYNC_FROM_SPDIF:
+ hdsp->control_register |= HDSP_SyncRef_SPDIF;
+ break;
+ case HDSP_SYNC_FROM_WORD:
+ hdsp->control_register |= HDSP_SyncRef_WORD;
+ break;
+ case HDSP_SYNC_FROM_ADAT_SYNC:
+ hdsp->control_register |= HDSP_SyncRef_ADAT_SYNC;
+ break;
+ default:
+ return -1;
+ }
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" };
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+
+ switch (hdsp->io_type) {
+ case Digiface:
+ case H9652:
+ uinfo->value.enumerated.items = 6;
+ break;
+ case Multiface:
+ uinfo->value.enumerated.items = 4;
+ break;
+ case H9632:
+ uinfo->value.enumerated.items = 3;
+ break;
+ default:
+ uinfo->value.enumerated.items = 0;
+ break;
+ }
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change, max;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+
+ switch (hdsp->io_type) {
+ case Digiface:
+ case H9652:
+ max = 6;
+ break;
+ case Multiface:
+ max = 4;
+ break;
+ case H9632:
+ max = 3;
+ break;
+ default:
+ return -EIO;
+ }
+
+ val = ucontrol->value.enumerated.item[0] % max;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_pref_sync_ref(hdsp);
+ hdsp_set_pref_sync_ref(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_AUTOSYNC_REF(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdsp_info_autosync_ref, \
+ .get = snd_hdsp_get_autosync_ref, \
+}
+
+static int hdsp_autosync_ref(struct hdsp *hdsp)
+{
+ /* This looks at the autosync selected sync reference */
+ unsigned int status2 = hdsp_read(hdsp, HDSP_status2Register);
+
+ switch (status2 & HDSP_SelSyncRefMask) {
+ case HDSP_SelSyncRef_WORD:
+ return HDSP_AUTOSYNC_FROM_WORD;
+ case HDSP_SelSyncRef_ADAT_SYNC:
+ return HDSP_AUTOSYNC_FROM_ADAT_SYNC;
+ case HDSP_SelSyncRef_SPDIF:
+ return HDSP_AUTOSYNC_FROM_SPDIF;
+ case HDSP_SelSyncRefMask:
+ return HDSP_AUTOSYNC_FROM_NONE;
+ case HDSP_SelSyncRef_ADAT1:
+ return HDSP_AUTOSYNC_FROM_ADAT1;
+ case HDSP_SelSyncRef_ADAT2:
+ return HDSP_AUTOSYNC_FROM_ADAT2;
+ case HDSP_SelSyncRef_ADAT3:
+ return HDSP_AUTOSYNC_FROM_ADAT3;
+ default:
+ return HDSP_AUTOSYNC_FROM_WORD;
+ }
+ return 0;
+}
+
+static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 7;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp);
+ return 0;
+}
+
+#define HDSP_LINE_OUT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_line_out, \
+ .get = snd_hdsp_get_line_out, \
+ .put = snd_hdsp_put_line_out \
+}
+
+static int hdsp_line_out(struct hdsp *hdsp)
+{
+ return (hdsp->control_register & HDSP_LineOut) ? 1 : 0;
+}
+
+static int hdsp_set_line_output(struct hdsp *hdsp, int out)
+{
+ if (out)
+ hdsp->control_register |= HDSP_LineOut;
+ else
+ hdsp->control_register &= ~HDSP_LineOut;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ return 0;
+}
+
+#define snd_hdsp_info_line_out snd_ctl_boolean_mono_info
+
+static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdsp->lock);
+ ucontrol->value.integer.value[0] = hdsp_line_out(hdsp);
+ spin_unlock_irq(&hdsp->lock);
+ return 0;
+}
+
+static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp_line_out(hdsp);
+ hdsp_set_line_output(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_PRECISE_POINTER(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_precise_pointer, \
+ .get = snd_hdsp_get_precise_pointer, \
+ .put = snd_hdsp_put_precise_pointer \
+}
+
+static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise)
+{
+ if (precise)
+ hdsp->precise_ptr = 1;
+ else
+ hdsp->precise_ptr = 0;
+ return 0;
+}
+
+#define snd_hdsp_info_precise_pointer snd_ctl_boolean_mono_info
+
+static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdsp->lock);
+ ucontrol->value.integer.value[0] = hdsp->precise_ptr;
+ spin_unlock_irq(&hdsp->lock);
+ return 0;
+}
+
+static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp->precise_ptr;
+ hdsp_set_precise_pointer(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_USE_MIDI_TASKLET(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_use_midi_tasklet, \
+ .get = snd_hdsp_get_use_midi_tasklet, \
+ .put = snd_hdsp_put_use_midi_tasklet \
+}
+
+static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet)
+{
+ if (use_tasklet)
+ hdsp->use_midi_tasklet = 1;
+ else
+ hdsp->use_midi_tasklet = 0;
+ return 0;
+}
+
+#define snd_hdsp_info_use_midi_tasklet snd_ctl_boolean_mono_info
+
+static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdsp->lock);
+ ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet;
+ spin_unlock_irq(&hdsp->lock);
+ return 0;
+}
+
+static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdsp->lock);
+ change = (int)val != hdsp->use_midi_tasklet;
+ hdsp_set_use_midi_tasklet(hdsp, val);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_MIXER(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .device = 0, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdsp_info_mixer, \
+ .get = snd_hdsp_get_mixer, \
+ .put = snd_hdsp_put_mixer \
+}
+
+static int snd_hdsp_info_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 65536;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int source;
+ int destination;
+ int addr;
+
+ source = ucontrol->value.integer.value[0];
+ destination = ucontrol->value.integer.value[1];
+
+ if (source >= hdsp->max_channels)
+ addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination);
+ else
+ addr = hdsp_input_to_output_key(hdsp,source, destination);
+
+ spin_lock_irq(&hdsp->lock);
+ ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr);
+ spin_unlock_irq(&hdsp->lock);
+ return 0;
+}
+
+static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int source;
+ int destination;
+ int gain;
+ int addr;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+
+ source = ucontrol->value.integer.value[0];
+ destination = ucontrol->value.integer.value[1];
+
+ if (source >= hdsp->max_channels)
+ addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels, destination);
+ else
+ addr = hdsp_input_to_output_key(hdsp,source, destination);
+
+ gain = ucontrol->value.integer.value[2];
+
+ spin_lock_irq(&hdsp->lock);
+ change = gain != hdsp_read_gain(hdsp, addr);
+ if (change)
+ hdsp_write_gain(hdsp, addr, gain);
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+#define HDSP_WC_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdsp_info_sync_check, \
+ .get = snd_hdsp_get_wc_sync_check \
+}
+
+static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {"No Lock", "Lock", "Sync" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int hdsp_wc_sync_check(struct hdsp *hdsp)
+{
+ int status2 = hdsp_read(hdsp, HDSP_status2Register);
+ if (status2 & HDSP_wc_lock) {
+ if (status2 & HDSP_wc_sync)
+ return 2;
+ else
+ return 1;
+ } else
+ return 0;
+ return 0;
+}
+
+static int snd_hdsp_get_wc_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_wc_sync_check(hdsp);
+ return 0;
+}
+
+#define HDSP_SPDIF_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdsp_info_sync_check, \
+ .get = snd_hdsp_get_spdif_sync_check \
+}
+
+static int hdsp_spdif_sync_check(struct hdsp *hdsp)
+{
+ int status = hdsp_read(hdsp, HDSP_statusRegister);
+ if (status & HDSP_SPDIFErrorFlag)
+ return 0;
+ else {
+ if (status & HDSP_SPDIFSync)
+ return 2;
+ else
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_hdsp_get_spdif_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_spdif_sync_check(hdsp);
+ return 0;
+}
+
+#define HDSP_ADATSYNC_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdsp_info_sync_check, \
+ .get = snd_hdsp_get_adatsync_sync_check \
+}
+
+static int hdsp_adatsync_sync_check(struct hdsp *hdsp)
+{
+ int status = hdsp_read(hdsp, HDSP_statusRegister);
+ if (status & HDSP_TimecodeLock) {
+ if (status & HDSP_TimecodeSync)
+ return 2;
+ else
+ return 1;
+ } else
+ return 0;
+}
+
+static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_adatsync_sync_check(hdsp);
+ return 0;
+}
+
+#define HDSP_ADAT_SYNC_CHECK \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdsp_info_sync_check, \
+ .get = snd_hdsp_get_adat_sync_check \
+}
+
+static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx)
+{
+ int status = hdsp_read(hdsp, HDSP_statusRegister);
+
+ if (status & (HDSP_Lock0>>idx)) {
+ if (status & (HDSP_Sync0>>idx))
+ return 2;
+ else
+ return 1;
+ } else
+ return 0;
+}
+
+static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ int offset;
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ offset = ucontrol->id.index - 1;
+ snd_BUG_ON(offset < 0);
+
+ switch (hdsp->io_type) {
+ case Digiface:
+ case H9652:
+ if (offset >= 3)
+ return -EINVAL;
+ break;
+ case Multiface:
+ case H9632:
+ if (offset >= 1)
+ return -EINVAL;
+ break;
+ default:
+ return -EIO;
+ }
+
+ ucontrol->value.enumerated.item[0] = hdsp_adat_sync_check(hdsp, offset);
+ return 0;
+}
+
+#define HDSP_DDS_OFFSET(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdsp_info_dds_offset, \
+ .get = snd_hdsp_get_dds_offset, \
+ .put = snd_hdsp_put_dds_offset \
+}
+
+static int hdsp_dds_offset(struct hdsp *hdsp)
+{
+ u64 n;
+ u32 r;
+ unsigned int dds_value = hdsp->dds_value;
+ int system_sample_rate = hdsp->system_sample_rate;
+
+ if (!dds_value)
+ return 0;
+
+ n = DDS_NUMERATOR;
+ /*
+ * dds_value = n / rate
+ * rate = n / dds_value
+ */
+ div64_32(&n, dds_value, &r);
+ if (system_sample_rate >= 112000)
+ n *= 4;
+ else if (system_sample_rate >= 56000)
+ n *= 2;
+ return ((int)n) - system_sample_rate;
+}
+
+static int hdsp_set_dds_offset(struct hdsp *hdsp, int offset_hz)
+{
+ int rate = hdsp->system_sample_rate + offset_hz;
+ hdsp_set_dds_value(hdsp, rate);
+ return 0;
+}
+
+static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = -5000;
+ uinfo->value.integer.max = 5000;
+ return 0;
+}
+
+static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp);
+ return 0;
+}
+
+static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdsp_use_is_exclusive(hdsp))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0];
+ spin_lock_irq(&hdsp->lock);
+ if (val != hdsp_dds_offset(hdsp))
+ change = (hdsp_set_dds_offset(hdsp, val) == 0) ? 1 : 0;
+ else
+ change = 0;
+ spin_unlock_irq(&hdsp->lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_hdsp_9632_controls[] = {
+HDSP_DA_GAIN("DA Gain", 0),
+HDSP_AD_GAIN("AD Gain", 0),
+HDSP_PHONE_GAIN("Phones Gain", 0),
+HDSP_XLR_BREAKOUT_CABLE("XLR Breakout Cable", 0),
+HDSP_DDS_OFFSET("DDS Sample Rate Offset", 0)
+};
+
+static struct snd_kcontrol_new snd_hdsp_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_hdsp_control_spdif_info,
+ .get = snd_hdsp_control_spdif_get,
+ .put = snd_hdsp_control_spdif_put,
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_hdsp_control_spdif_stream_info,
+ .get = snd_hdsp_control_spdif_stream_get,
+ .put = snd_hdsp_control_spdif_stream_put,
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_hdsp_control_spdif_mask_info,
+ .get = snd_hdsp_control_spdif_mask_get,
+ .private_value = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_CON_EMPHASIS,
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+ .info = snd_hdsp_control_spdif_mask_info,
+ .get = snd_hdsp_control_spdif_mask_get,
+ .private_value = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_PRO_EMPHASIS,
+},
+HDSP_MIXER("Mixer", 0),
+HDSP_SPDIF_IN("IEC958 Input Connector", 0),
+HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0),
+HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0),
+HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0),
+HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0),
+/* 'Sample Clock Source' complies with the alsa control naming scheme */
+HDSP_CLOCK_SOURCE("Sample Clock Source", 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Sample Clock Source Locking",
+ .info = snd_hdsp_info_clock_source_lock,
+ .get = snd_hdsp_get_clock_source_lock,
+ .put = snd_hdsp_put_clock_source_lock,
+},
+HDSP_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
+HDSP_PREF_SYNC_REF("Preferred Sync Reference", 0),
+HDSP_AUTOSYNC_REF("AutoSync Reference", 0),
+HDSP_SPDIF_SAMPLE_RATE("SPDIF Sample Rate", 0),
+HDSP_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
+/* 'External Rate' complies with the alsa control naming scheme */
+HDSP_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+HDSP_WC_SYNC_CHECK("Word Clock Lock Status", 0),
+HDSP_SPDIF_SYNC_CHECK("SPDIF Lock Status", 0),
+HDSP_ADATSYNC_SYNC_CHECK("ADAT Sync Lock Status", 0),
+HDSP_LINE_OUT("Line Out", 0),
+HDSP_PRECISE_POINTER("Precise Pointer", 0),
+HDSP_USE_MIDI_TASKLET("Use Midi Tasklet", 0),
+};
+
+static struct snd_kcontrol_new snd_hdsp_96xx_aeb = HDSP_AEB("Analog Extension Board", 0);
+static struct snd_kcontrol_new snd_hdsp_adat_sync_check = HDSP_ADAT_SYNC_CHECK;
+
+static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp)
+{
+ unsigned int idx;
+ int err;
+ struct snd_kcontrol *kctl;
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_controls); idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_controls[idx], hdsp))) < 0)
+ return err;
+ if (idx == 1) /* IEC958 (S/PDIF) Stream */
+ hdsp->spdif_ctl = kctl;
+ }
+
+ /* ADAT SyncCheck status */
+ snd_hdsp_adat_sync_check.name = "ADAT Lock Status";
+ snd_hdsp_adat_sync_check.index = 1;
+ if ((err = snd_ctl_add (card, kctl = snd_ctl_new1(&snd_hdsp_adat_sync_check, hdsp))))
+ return err;
+ if (hdsp->io_type == Digiface || hdsp->io_type == H9652) {
+ for (idx = 1; idx < 3; ++idx) {
+ snd_hdsp_adat_sync_check.index = idx+1;
+ if ((err = snd_ctl_add (card, kctl = snd_ctl_new1(&snd_hdsp_adat_sync_check, hdsp))))
+ return err;
+ }
+ }
+
+ /* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */
+ if (hdsp->io_type == H9632) {
+ for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_9632_controls[idx], hdsp))) < 0)
+ return err;
+ }
+ }
+
+ /* AEB control for H96xx card */
+ if (hdsp->io_type == H9632 || hdsp->io_type == H9652) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_96xx_aeb, hdsp))) < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/*------------------------------------------------------------
+ /proc interface
+ ------------------------------------------------------------*/
+
+static void
+snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct hdsp *hdsp = (struct hdsp *) entry->private_data;
+ unsigned int status;
+ unsigned int status2;
+ char *pref_sync_ref;
+ char *autosync_ref;
+ char *system_clock_mode;
+ char *clock_source;
+ int x;
+
+ if (hdsp_check_for_iobox (hdsp)) {
+ snd_iprintf(buffer, "No I/O box connected.\nPlease connect one and upload firmware.\n");
+ return;
+ }
+
+ if (hdsp_check_for_firmware(hdsp, 0)) {
+ if (hdsp->state & HDSP_FirmwareCached) {
+ if (snd_hdsp_load_firmware_from_cache(hdsp) != 0) {
+ snd_iprintf(buffer, "Firmware loading from cache failed, please upload manually.\n");
+ return;
+ }
+ } else {
+ int err = -EINVAL;
+#ifdef HDSP_FW_LOADER
+ err = hdsp_request_fw_loader(hdsp);
+#endif
+ if (err < 0) {
+ snd_iprintf(buffer,
+ "No firmware loaded nor cached, "
+ "please upload firmware.\n");
+ return;
+ }
+ }
+ }
+
+ status = hdsp_read(hdsp, HDSP_statusRegister);
+ status2 = hdsp_read(hdsp, HDSP_status2Register);
+
+ snd_iprintf(buffer, "%s (Card #%d)\n", hdsp->card_name, hdsp->card->number + 1);
+ snd_iprintf(buffer, "Buffers: capture %p playback %p\n",
+ hdsp->capture_buffer, hdsp->playback_buffer);
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ hdsp->irq, hdsp->port, (unsigned long)hdsp->iobase);
+ snd_iprintf(buffer, "Control register: 0x%x\n", hdsp->control_register);
+ snd_iprintf(buffer, "Control2 register: 0x%x\n", hdsp->control2_register);
+ snd_iprintf(buffer, "Status register: 0x%x\n", status);
+ snd_iprintf(buffer, "Status2 register: 0x%x\n", status2);
+ snd_iprintf(buffer, "FIFO status: %d\n", hdsp_read(hdsp, HDSP_fifoStatus) & 0xff);
+ snd_iprintf(buffer, "MIDI1 Output status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusOut0));
+ snd_iprintf(buffer, "MIDI1 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn0));
+ snd_iprintf(buffer, "MIDI2 Output status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusOut1));
+ snd_iprintf(buffer, "MIDI2 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn1));
+ snd_iprintf(buffer, "Use Midi Tasklet: %s\n", hdsp->use_midi_tasklet ? "on" : "off");
+
+ snd_iprintf(buffer, "\n");
+
+ x = 1 << (6 + hdsp_decode_latency(hdsp->control_register & HDSP_LatencyMask));
+
+ snd_iprintf(buffer, "Buffer Size (Latency): %d samples (2 periods of %lu bytes)\n", x, (unsigned long) hdsp->period_bytes);
+ snd_iprintf(buffer, "Hardware pointer (frames): %ld\n", hdsp_hw_pointer(hdsp));
+ snd_iprintf(buffer, "Precise pointer: %s\n", hdsp->precise_ptr ? "on" : "off");
+ snd_iprintf(buffer, "Line out: %s\n", (hdsp->control_register & HDSP_LineOut) ? "on" : "off");
+
+ snd_iprintf(buffer, "Firmware version: %d\n", (status2&HDSP_version0)|(status2&HDSP_version1)<<1|(status2&HDSP_version2)<<2);
+
+ snd_iprintf(buffer, "\n");
+
+
+ switch (hdsp_clock_source(hdsp)) {
+ case HDSP_CLOCK_SOURCE_AUTOSYNC:
+ clock_source = "AutoSync";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_32KHZ:
+ clock_source = "Internal 32 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ:
+ clock_source = "Internal 44.1 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ:
+ clock_source = "Internal 48 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_64KHZ:
+ clock_source = "Internal 64 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_88_2KHZ:
+ clock_source = "Internal 88.2 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_96KHZ:
+ clock_source = "Internal 96 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_128KHZ:
+ clock_source = "Internal 128 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_176_4KHZ:
+ clock_source = "Internal 176.4 kHz";
+ break;
+ case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ:
+ clock_source = "Internal 192 kHz";
+ break;
+ default:
+ clock_source = "Error";
+ }
+ snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source);
+
+ if (hdsp_system_clock_mode(hdsp))
+ system_clock_mode = "Slave";
+ else
+ system_clock_mode = "Master";
+
+ switch (hdsp_pref_sync_ref (hdsp)) {
+ case HDSP_SYNC_FROM_WORD:
+ pref_sync_ref = "Word Clock";
+ break;
+ case HDSP_SYNC_FROM_ADAT_SYNC:
+ pref_sync_ref = "ADAT Sync";
+ break;
+ case HDSP_SYNC_FROM_SPDIF:
+ pref_sync_ref = "SPDIF";
+ break;
+ case HDSP_SYNC_FROM_ADAT1:
+ pref_sync_ref = "ADAT1";
+ break;
+ case HDSP_SYNC_FROM_ADAT2:
+ pref_sync_ref = "ADAT2";
+ break;
+ case HDSP_SYNC_FROM_ADAT3:
+ pref_sync_ref = "ADAT3";
+ break;
+ default:
+ pref_sync_ref = "Word Clock";
+ break;
+ }
+ snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref);
+
+ switch (hdsp_autosync_ref (hdsp)) {
+ case HDSP_AUTOSYNC_FROM_WORD:
+ autosync_ref = "Word Clock";
+ break;
+ case HDSP_AUTOSYNC_FROM_ADAT_SYNC:
+ autosync_ref = "ADAT Sync";
+ break;
+ case HDSP_AUTOSYNC_FROM_SPDIF:
+ autosync_ref = "SPDIF";
+ break;
+ case HDSP_AUTOSYNC_FROM_NONE:
+ autosync_ref = "None";
+ break;
+ case HDSP_AUTOSYNC_FROM_ADAT1:
+ autosync_ref = "ADAT1";
+ break;
+ case HDSP_AUTOSYNC_FROM_ADAT2:
+ autosync_ref = "ADAT2";
+ break;
+ case HDSP_AUTOSYNC_FROM_ADAT3:
+ autosync_ref = "ADAT3";
+ break;
+ default:
+ autosync_ref = "---";
+ break;
+ }
+ snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref);
+
+ snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp));
+
+ snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode);
+
+ snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate);
+ snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No");
+
+ snd_iprintf(buffer, "\n");
+
+ switch (hdsp_spdif_in(hdsp)) {
+ case HDSP_SPDIFIN_OPTICAL:
+ snd_iprintf(buffer, "IEC958 input: Optical\n");
+ break;
+ case HDSP_SPDIFIN_COAXIAL:
+ snd_iprintf(buffer, "IEC958 input: Coaxial\n");
+ break;
+ case HDSP_SPDIFIN_INTERNAL:
+ snd_iprintf(buffer, "IEC958 input: Internal\n");
+ break;
+ case HDSP_SPDIFIN_AES:
+ snd_iprintf(buffer, "IEC958 input: AES\n");
+ break;
+ default:
+ snd_iprintf(buffer, "IEC958 input: ???\n");
+ break;
+ }
+
+ if (hdsp->control_register & HDSP_SPDIFOpticalOut)
+ snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n");
+ else
+ snd_iprintf(buffer, "IEC958 output: Coaxial only\n");
+
+ if (hdsp->control_register & HDSP_SPDIFProfessional)
+ snd_iprintf(buffer, "IEC958 quality: Professional\n");
+ else
+ snd_iprintf(buffer, "IEC958 quality: Consumer\n");
+
+ if (hdsp->control_register & HDSP_SPDIFEmphasis)
+ snd_iprintf(buffer, "IEC958 emphasis: on\n");
+ else
+ snd_iprintf(buffer, "IEC958 emphasis: off\n");
+
+ if (hdsp->control_register & HDSP_SPDIFNonAudio)
+ snd_iprintf(buffer, "IEC958 NonAudio: on\n");
+ else
+ snd_iprintf(buffer, "IEC958 NonAudio: off\n");
+ if ((x = hdsp_spdif_sample_rate (hdsp)) != 0)
+ snd_iprintf (buffer, "IEC958 sample rate: %d\n", x);
+ else
+ snd_iprintf (buffer, "IEC958 sample rate: Error flag set\n");
+
+ snd_iprintf(buffer, "\n");
+
+ /* Sync Check */
+ x = status & HDSP_Sync0;
+ if (status & HDSP_Lock0)
+ snd_iprintf(buffer, "ADAT1: %s\n", x ? "Sync" : "Lock");
+ else
+ snd_iprintf(buffer, "ADAT1: No Lock\n");
+
+ switch (hdsp->io_type) {
+ case Digiface:
+ case H9652:
+ x = status & HDSP_Sync1;
+ if (status & HDSP_Lock1)
+ snd_iprintf(buffer, "ADAT2: %s\n", x ? "Sync" : "Lock");
+ else
+ snd_iprintf(buffer, "ADAT2: No Lock\n");
+ x = status & HDSP_Sync2;
+ if (status & HDSP_Lock2)
+ snd_iprintf(buffer, "ADAT3: %s\n", x ? "Sync" : "Lock");
+ else
+ snd_iprintf(buffer, "ADAT3: No Lock\n");
+ break;
+ default:
+ /* relax */
+ break;
+ }
+
+ x = status & HDSP_SPDIFSync;
+ if (status & HDSP_SPDIFErrorFlag)
+ snd_iprintf (buffer, "SPDIF: No Lock\n");
+ else
+ snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock");
+
+ x = status2 & HDSP_wc_sync;
+ if (status2 & HDSP_wc_lock)
+ snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock");
+ else
+ snd_iprintf (buffer, "Word Clock: No Lock\n");
+
+ x = status & HDSP_TimecodeSync;
+ if (status & HDSP_TimecodeLock)
+ snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock");
+ else
+ snd_iprintf(buffer, "ADAT Sync: No Lock\n");
+
+ snd_iprintf(buffer, "\n");
+
+ /* Informations about H9632 specific controls */
+ if (hdsp->io_type == H9632) {
+ char *tmp;
+
+ switch (hdsp_ad_gain(hdsp)) {
+ case 0:
+ tmp = "-10 dBV";
+ break;
+ case 1:
+ tmp = "+4 dBu";
+ break;
+ default:
+ tmp = "Lo Gain";
+ break;
+ }
+ snd_iprintf(buffer, "AD Gain : %s\n", tmp);
+
+ switch (hdsp_da_gain(hdsp)) {
+ case 0:
+ tmp = "Hi Gain";
+ break;
+ case 1:
+ tmp = "+4 dBu";
+ break;
+ default:
+ tmp = "-10 dBV";
+ break;
+ }
+ snd_iprintf(buffer, "DA Gain : %s\n", tmp);
+
+ switch (hdsp_phone_gain(hdsp)) {
+ case 0:
+ tmp = "0 dB";
+ break;
+ case 1:
+ tmp = "-6 dB";
+ break;
+ default:
+ tmp = "-12 dB";
+ break;
+ }
+ snd_iprintf(buffer, "Phones Gain : %s\n", tmp);
+
+ snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no");
+
+ if (hdsp->control_register & HDSP_AnalogExtensionBoard)
+ snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n");
+ else
+ snd_iprintf(buffer, "AEB : off (ADAT1 external)\n");
+ snd_iprintf(buffer, "\n");
+ }
+
+}
+
+static void snd_hdsp_proc_init(struct hdsp *hdsp)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(hdsp->card, "hdsp", &entry))
+ snd_info_set_text_ops(entry, hdsp, snd_hdsp_proc_read);
+}
+
+static void snd_hdsp_free_buffers(struct hdsp *hdsp)
+{
+ snd_hammerfall_free_buffer(&hdsp->capture_dma_buf, hdsp->pci);
+ snd_hammerfall_free_buffer(&hdsp->playback_dma_buf, hdsp->pci);
+}
+
+static int __devinit snd_hdsp_initialize_memory(struct hdsp *hdsp)
+{
+ unsigned long pb_bus, cb_bus;
+
+ if (snd_hammerfall_get_buffer(hdsp->pci, &hdsp->capture_dma_buf, HDSP_DMA_AREA_BYTES) < 0 ||
+ snd_hammerfall_get_buffer(hdsp->pci, &hdsp->playback_dma_buf, HDSP_DMA_AREA_BYTES) < 0) {
+ if (hdsp->capture_dma_buf.area)
+ snd_dma_free_pages(&hdsp->capture_dma_buf);
+ printk(KERN_ERR "%s: no buffers available\n", hdsp->card_name);
+ return -ENOMEM;
+ }
+
+ /* Align to bus-space 64K boundary */
+
+ cb_bus = ALIGN(hdsp->capture_dma_buf.addr, 0x10000ul);
+ pb_bus = ALIGN(hdsp->playback_dma_buf.addr, 0x10000ul);
+
+ /* Tell the card where it is */
+
+ hdsp_write(hdsp, HDSP_inputBufferAddress, cb_bus);
+ hdsp_write(hdsp, HDSP_outputBufferAddress, pb_bus);
+
+ hdsp->capture_buffer = hdsp->capture_dma_buf.area + (cb_bus - hdsp->capture_dma_buf.addr);
+ hdsp->playback_buffer = hdsp->playback_dma_buf.area + (pb_bus - hdsp->playback_dma_buf.addr);
+
+ return 0;
+}
+
+static int snd_hdsp_set_defaults(struct hdsp *hdsp)
+{
+ unsigned int i;
+
+ /* ASSUMPTION: hdsp->lock is either held, or
+ there is no need to hold it (e.g. during module
+ initialization).
+ */
+
+ /* set defaults:
+
+ SPDIF Input via Coax
+ Master clock mode
+ maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer,
+ which implies 2 4096 sample, 32Kbyte periods).
+ Enable line out.
+ */
+
+ hdsp->control_register = HDSP_ClockModeMaster |
+ HDSP_SPDIFInputCoaxial |
+ hdsp_encode_latency(7) |
+ HDSP_LineOut;
+
+
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+
+#ifdef SNDRV_BIG_ENDIAN
+ hdsp->control2_register = HDSP_BIGENDIAN_MODE;
+#else
+ hdsp->control2_register = 0;
+#endif
+ if (hdsp->io_type == H9652)
+ snd_hdsp_9652_enable_mixer (hdsp);
+ else
+ hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register);
+
+ hdsp_reset_hw_pointer(hdsp);
+ hdsp_compute_period_size(hdsp);
+
+ /* silence everything */
+
+ for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i)
+ hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN;
+
+ for (i = 0; i < ((hdsp->io_type == H9652 || hdsp->io_type == H9632) ? 1352 : HDSP_MATRIX_MIXER_SIZE); ++i) {
+ if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN))
+ return -EIO;
+ }
+
+ /* H9632 specific defaults */
+ if (hdsp->io_type == H9632) {
+ hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB);
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ }
+
+ /* set a default rate so that the channel map is set up.
+ */
+
+ hdsp_set_rate(hdsp, 48000, 1);
+
+ return 0;
+}
+
+static void hdsp_midi_tasklet(unsigned long arg)
+{
+ struct hdsp *hdsp = (struct hdsp *)arg;
+
+ if (hdsp->midi[0].pending)
+ snd_hdsp_midi_input_read (&hdsp->midi[0]);
+ if (hdsp->midi[1].pending)
+ snd_hdsp_midi_input_read (&hdsp->midi[1]);
+}
+
+static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
+{
+ struct hdsp *hdsp = (struct hdsp *) dev_id;
+ unsigned int status;
+ int audio;
+ int midi0;
+ int midi1;
+ unsigned int midi0status;
+ unsigned int midi1status;
+ int schedule = 0;
+
+ status = hdsp_read(hdsp, HDSP_statusRegister);
+
+ audio = status & HDSP_audioIRQPending;
+ midi0 = status & HDSP_midi0IRQPending;
+ midi1 = status & HDSP_midi1IRQPending;
+
+ if (!audio && !midi0 && !midi1)
+ return IRQ_NONE;
+
+ hdsp_write(hdsp, HDSP_interruptConfirmation, 0);
+
+ midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff;
+ midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff;
+
+ if (audio) {
+ if (hdsp->capture_substream)
+ snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
+
+ if (hdsp->playback_substream)
+ snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream);
+ }
+
+ if (midi0 && midi0status) {
+ if (hdsp->use_midi_tasklet) {
+ /* we disable interrupts for this input until processing is done */
+ hdsp->control_register &= ~HDSP_Midi0InterruptEnable;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ hdsp->midi[0].pending = 1;
+ schedule = 1;
+ } else {
+ snd_hdsp_midi_input_read (&hdsp->midi[0]);
+ }
+ }
+ if (hdsp->io_type != Multiface && hdsp->io_type != H9632 && midi1 && midi1status) {
+ if (hdsp->use_midi_tasklet) {
+ /* we disable interrupts for this input until processing is done */
+ hdsp->control_register &= ~HDSP_Midi1InterruptEnable;
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
+ hdsp->midi[1].pending = 1;
+ schedule = 1;
+ } else {
+ snd_hdsp_midi_input_read (&hdsp->midi[1]);
+ }
+ }
+ if (hdsp->use_midi_tasklet && schedule)
+ tasklet_hi_schedule(&hdsp->midi_tasklet);
+ return IRQ_HANDLED;
+}
+
+static snd_pcm_uframes_t snd_hdsp_hw_pointer(struct snd_pcm_substream *substream)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ return hdsp_hw_pointer(hdsp);
+}
+
+static char *hdsp_channel_buffer_location(struct hdsp *hdsp,
+ int stream,
+ int channel)
+
+{
+ int mapped_channel;
+
+ if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels))
+ return NULL;
+
+ if ((mapped_channel = hdsp->channel_map[channel]) < 0)
+ return NULL;
+
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
+ return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES);
+ else
+ return hdsp->playback_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES);
+}
+
+static int snd_hdsp_playback_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ if (snd_BUG_ON(pos + count > HDSP_CHANNEL_BUFFER_BYTES / 4))
+ return -EINVAL;
+
+ channel_buf = hdsp_channel_buffer_location (hdsp, substream->pstr->stream, channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ if (copy_from_user(channel_buf + pos * 4, src, count * 4))
+ return -EFAULT;
+ return count;
+}
+
+static int snd_hdsp_capture_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, void __user *dst, snd_pcm_uframes_t count)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ if (snd_BUG_ON(pos + count > HDSP_CHANNEL_BUFFER_BYTES / 4))
+ return -EINVAL;
+
+ channel_buf = hdsp_channel_buffer_location (hdsp, substream->pstr->stream, channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ if (copy_to_user(dst, channel_buf + pos * 4, count * 4))
+ return -EFAULT;
+ return count;
+}
+
+static int snd_hdsp_hw_silence(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, snd_pcm_uframes_t count)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ channel_buf = hdsp_channel_buffer_location (hdsp, substream->pstr->stream, channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ memset(channel_buf + pos * 4, 0, count * 4);
+ return count;
+}
+
+static int snd_hdsp_reset(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *other;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = hdsp->capture_substream;
+ else
+ other = hdsp->playback_substream;
+ if (hdsp->running)
+ runtime->status->hw_ptr = hdsp_hw_pointer(hdsp);
+ else
+ runtime->status->hw_ptr = 0;
+ if (other) {
+ struct snd_pcm_substream *s;
+ struct snd_pcm_runtime *oruntime = other->runtime;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == other) {
+ oruntime->status->hw_ptr = runtime->status->hw_ptr;
+ break;
+ }
+ }
+ }
+ return 0;
+}
+
+static int snd_hdsp_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ int err;
+ pid_t this_pid;
+ pid_t other_pid;
+
+ if (hdsp_check_for_iobox (hdsp))
+ return -EIO;
+
+ if (hdsp_check_for_firmware(hdsp, 1))
+ return -EIO;
+
+ spin_lock_irq(&hdsp->lock);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ hdsp->control_register &= ~(HDSP_SPDIFProfessional | HDSP_SPDIFNonAudio | HDSP_SPDIFEmphasis);
+ hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register |= hdsp->creg_spdif_stream);
+ this_pid = hdsp->playback_pid;
+ other_pid = hdsp->capture_pid;
+ } else {
+ this_pid = hdsp->capture_pid;
+ other_pid = hdsp->playback_pid;
+ }
+
+ if ((other_pid > 0) && (this_pid != other_pid)) {
+
+ /* The other stream is open, and not by the same
+ task as this one. Make sure that the parameters
+ that matter are the same.
+ */
+
+ if (params_rate(params) != hdsp->system_sample_rate) {
+ spin_unlock_irq(&hdsp->lock);
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE);
+ return -EBUSY;
+ }
+
+ if (params_period_size(params) != hdsp->period_bytes / 4) {
+ spin_unlock_irq(&hdsp->lock);
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return -EBUSY;
+ }
+
+ /* We're fine. */
+
+ spin_unlock_irq(&hdsp->lock);
+ return 0;
+
+ } else {
+ spin_unlock_irq(&hdsp->lock);
+ }
+
+ /* how to make sure that the rate matches an externally-set one ?
+ */
+
+ spin_lock_irq(&hdsp->lock);
+ if (! hdsp->clock_source_locked) {
+ if ((err = hdsp_set_rate(hdsp, params_rate(params), 0)) < 0) {
+ spin_unlock_irq(&hdsp->lock);
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE);
+ return err;
+ }
+ }
+ spin_unlock_irq(&hdsp->lock);
+
+ if ((err = hdsp_set_interrupt_interval(hdsp, params_period_size(params))) < 0) {
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return err;
+ }
+
+ return 0;
+}
+
+static int snd_hdsp_channel_info(struct snd_pcm_substream *substream,
+ struct snd_pcm_channel_info *info)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ int mapped_channel;
+
+ if (snd_BUG_ON(info->channel >= hdsp->max_channels))
+ return -EINVAL;
+
+ if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
+ return -EINVAL;
+
+ info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES;
+ info->first = 0;
+ info->step = 32;
+ return 0;
+}
+
+static int snd_hdsp_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL1_RESET:
+ return snd_hdsp_reset(substream);
+ case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
+ return snd_hdsp_channel_info(substream, arg);
+ default:
+ break;
+ }
+
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *other;
+ int running;
+
+ if (hdsp_check_for_iobox (hdsp))
+ return -EIO;
+
+ if (hdsp_check_for_firmware(hdsp, 0)) /* no auto-loading in trigger */
+ return -EIO;
+
+ spin_lock(&hdsp->lock);
+ running = hdsp->running;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ running |= 1 << substream->stream;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ running &= ~(1 << substream->stream);
+ break;
+ default:
+ snd_BUG();
+ spin_unlock(&hdsp->lock);
+ return -EINVAL;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = hdsp->capture_substream;
+ else
+ other = hdsp->playback_substream;
+
+ if (other) {
+ struct snd_pcm_substream *s;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == other) {
+ snd_pcm_trigger_done(s, substream);
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ running |= 1 << s->stream;
+ else
+ running &= ~(1 << s->stream);
+ goto _ok;
+ }
+ }
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ if (!(running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) &&
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ hdsp_silence_playback(hdsp);
+ } else {
+ if (running &&
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ hdsp_silence_playback(hdsp);
+ }
+ } else {
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ hdsp_silence_playback(hdsp);
+ }
+ _ok:
+ snd_pcm_trigger_done(substream, substream);
+ if (!hdsp->running && running)
+ hdsp_start_audio(hdsp);
+ else if (hdsp->running && !running)
+ hdsp_stop_audio(hdsp);
+ hdsp->running = running;
+ spin_unlock(&hdsp->lock);
+
+ return 0;
+}
+
+static int snd_hdsp_prepare(struct snd_pcm_substream *substream)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ int result = 0;
+
+ if (hdsp_check_for_iobox (hdsp))
+ return -EIO;
+
+ if (hdsp_check_for_firmware(hdsp, 1))
+ return -EIO;
+
+ spin_lock_irq(&hdsp->lock);
+ if (!hdsp->running)
+ hdsp_reset_hw_pointer(hdsp);
+ spin_unlock_irq(&hdsp->lock);
+ return result;
+}
+
+static struct snd_pcm_hardware snd_hdsp_playback_subinfo =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_DOUBLE),
+#ifdef SNDRV_BIG_ENDIAN
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+#else
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+#endif
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 14,
+ .channels_max = HDSP_MAX_CHANNELS,
+ .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4) * 10,
+ .period_bytes_max = (8192 * 4) * HDSP_MAX_CHANNELS,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0
+};
+
+static struct snd_pcm_hardware snd_hdsp_capture_subinfo =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START),
+#ifdef SNDRV_BIG_ENDIAN
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+#else
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+#endif
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 14,
+ .channels_max = HDSP_MAX_CHANNELS,
+ .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4) * 10,
+ .period_bytes_max = (8192 * 4) * HDSP_MAX_CHANNELS,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0
+};
+
+static unsigned int hdsp_period_sizes[] = { 64, 128, 256, 512, 1024, 2048, 4096, 8192 };
+
+static struct snd_pcm_hw_constraint_list hdsp_hw_constraints_period_sizes = {
+ .count = ARRAY_SIZE(hdsp_period_sizes),
+ .list = hdsp_period_sizes,
+ .mask = 0
+};
+
+static unsigned int hdsp_9632_sample_rates[] = { 32000, 44100, 48000, 64000, 88200, 96000, 128000, 176400, 192000 };
+
+static struct snd_pcm_hw_constraint_list hdsp_hw_constraints_9632_sample_rates = {
+ .count = ARRAY_SIZE(hdsp_9632_sample_rates),
+ .list = hdsp_9632_sample_rates,
+ .mask = 0
+};
+
+static int snd_hdsp_hw_rule_in_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct hdsp *hdsp = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ if (hdsp->io_type == H9632) {
+ unsigned int list[3];
+ list[0] = hdsp->qs_in_channels;
+ list[1] = hdsp->ds_in_channels;
+ list[2] = hdsp->ss_in_channels;
+ return snd_interval_list(c, 3, list, 0);
+ } else {
+ unsigned int list[2];
+ list[0] = hdsp->ds_in_channels;
+ list[1] = hdsp->ss_in_channels;
+ return snd_interval_list(c, 2, list, 0);
+ }
+}
+
+static int snd_hdsp_hw_rule_out_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ unsigned int list[3];
+ struct hdsp *hdsp = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ if (hdsp->io_type == H9632) {
+ list[0] = hdsp->qs_out_channels;
+ list[1] = hdsp->ds_out_channels;
+ list[2] = hdsp->ss_out_channels;
+ return snd_interval_list(c, 3, list, 0);
+ } else {
+ list[0] = hdsp->ds_out_channels;
+ list[1] = hdsp->ss_out_channels;
+ }
+ return snd_interval_list(c, 2, list, 0);
+}
+
+static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct hdsp *hdsp = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (r->min > 96000 && hdsp->io_type == H9632) {
+ struct snd_interval t = {
+ .min = hdsp->qs_in_channels,
+ .max = hdsp->qs_in_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ } else if (r->min > 48000 && r->max <= 96000) {
+ struct snd_interval t = {
+ .min = hdsp->ds_in_channels,
+ .max = hdsp->ds_in_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ } else if (r->max < 64000) {
+ struct snd_interval t = {
+ .min = hdsp->ss_in_channels,
+ .max = hdsp->ss_in_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ }
+ return 0;
+}
+
+static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct hdsp *hdsp = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (r->min > 96000 && hdsp->io_type == H9632) {
+ struct snd_interval t = {
+ .min = hdsp->qs_out_channels,
+ .max = hdsp->qs_out_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ } else if (r->min > 48000 && r->max <= 96000) {
+ struct snd_interval t = {
+ .min = hdsp->ds_out_channels,
+ .max = hdsp->ds_out_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ } else if (r->max < 64000) {
+ struct snd_interval t = {
+ .min = hdsp->ss_out_channels,
+ .max = hdsp->ss_out_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ }
+ return 0;
+}
+
+static int snd_hdsp_hw_rule_rate_out_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct hdsp *hdsp = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (c->min >= hdsp->ss_out_channels) {
+ struct snd_interval t = {
+ .min = 32000,
+ .max = 48000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ } else if (c->max <= hdsp->qs_out_channels && hdsp->io_type == H9632) {
+ struct snd_interval t = {
+ .min = 128000,
+ .max = 192000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ } else if (c->max <= hdsp->ds_out_channels) {
+ struct snd_interval t = {
+ .min = 64000,
+ .max = 96000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ }
+ return 0;
+}
+
+static int snd_hdsp_hw_rule_rate_in_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct hdsp *hdsp = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (c->min >= hdsp->ss_in_channels) {
+ struct snd_interval t = {
+ .min = 32000,
+ .max = 48000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ } else if (c->max <= hdsp->qs_in_channels && hdsp->io_type == H9632) {
+ struct snd_interval t = {
+ .min = 128000,
+ .max = 192000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ } else if (c->max <= hdsp->ds_in_channels) {
+ struct snd_interval t = {
+ .min = 64000,
+ .max = 96000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ }
+ return 0;
+}
+
+static int snd_hdsp_playback_open(struct snd_pcm_substream *substream)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (hdsp_check_for_iobox (hdsp))
+ return -EIO;
+
+ if (hdsp_check_for_firmware(hdsp, 1))
+ return -EIO;
+
+ spin_lock_irq(&hdsp->lock);
+
+ snd_pcm_set_sync(substream);
+
+ runtime->hw = snd_hdsp_playback_subinfo;
+ runtime->dma_area = hdsp->playback_buffer;
+ runtime->dma_bytes = HDSP_DMA_AREA_BYTES;
+
+ hdsp->playback_pid = current->pid;
+ hdsp->playback_substream = substream;
+
+ spin_unlock_irq(&hdsp->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes);
+ if (hdsp->clock_source_locked) {
+ runtime->hw.rate_min = runtime->hw.rate_max = hdsp->system_sample_rate;
+ } else if (hdsp->io_type == H9632) {
+ runtime->hw.rate_max = 192000;
+ runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates);
+ }
+ if (hdsp->io_type == H9632) {
+ runtime->hw.channels_min = hdsp->qs_out_channels;
+ runtime->hw.channels_max = hdsp->ss_out_channels;
+ }
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdsp_hw_rule_out_channels, hdsp,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdsp_hw_rule_out_channels_rate, hdsp,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_hdsp_hw_rule_rate_out_channels, hdsp,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+
+ hdsp->creg_spdif_stream = hdsp->creg_spdif;
+ hdsp->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id);
+ return 0;
+}
+
+static int snd_hdsp_playback_release(struct snd_pcm_substream *substream)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&hdsp->lock);
+
+ hdsp->playback_pid = -1;
+ hdsp->playback_substream = NULL;
+
+ spin_unlock_irq(&hdsp->lock);
+
+ hdsp->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(hdsp->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &hdsp->spdif_ctl->id);
+ return 0;
+}
+
+
+static int snd_hdsp_capture_open(struct snd_pcm_substream *substream)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (hdsp_check_for_iobox (hdsp))
+ return -EIO;
+
+ if (hdsp_check_for_firmware(hdsp, 1))
+ return -EIO;
+
+ spin_lock_irq(&hdsp->lock);
+
+ snd_pcm_set_sync(substream);
+
+ runtime->hw = snd_hdsp_capture_subinfo;
+ runtime->dma_area = hdsp->capture_buffer;
+ runtime->dma_bytes = HDSP_DMA_AREA_BYTES;
+
+ hdsp->capture_pid = current->pid;
+ hdsp->capture_substream = substream;
+
+ spin_unlock_irq(&hdsp->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hdsp_hw_constraints_period_sizes);
+ if (hdsp->io_type == H9632) {
+ runtime->hw.channels_min = hdsp->qs_in_channels;
+ runtime->hw.channels_max = hdsp->ss_in_channels;
+ runtime->hw.rate_max = 192000;
+ runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates);
+ }
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdsp_hw_rule_in_channels, hdsp,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdsp_hw_rule_in_channels_rate, hdsp,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_hdsp_hw_rule_rate_in_channels, hdsp,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ return 0;
+}
+
+static int snd_hdsp_capture_release(struct snd_pcm_substream *substream)
+{
+ struct hdsp *hdsp = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&hdsp->lock);
+
+ hdsp->capture_pid = -1;
+ hdsp->capture_substream = NULL;
+
+ spin_unlock_irq(&hdsp->lock);
+ return 0;
+}
+
+static int snd_hdsp_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file)
+{
+ /* we have nothing to initialize but the call is required */
+ return 0;
+}
+
+
+/* helper functions for copying meter values */
+static inline int copy_u32_le(void __user *dest, void __iomem *src)
+{
+ u32 val = readl(src);
+ return copy_to_user(dest, &val, 4);
+}
+
+static inline int copy_u64_le(void __user *dest, void __iomem *src_low, void __iomem *src_high)
+{
+ u32 rms_low, rms_high;
+ u64 rms;
+ rms_low = readl(src_low);
+ rms_high = readl(src_high);
+ rms = ((u64)rms_high << 32) | rms_low;
+ return copy_to_user(dest, &rms, 8);
+}
+
+static inline int copy_u48_le(void __user *dest, void __iomem *src_low, void __iomem *src_high)
+{
+ u32 rms_low, rms_high;
+ u64 rms;
+ rms_low = readl(src_low) & 0xffffff00;
+ rms_high = readl(src_high) & 0xffffff00;
+ rms = ((u64)rms_high << 32) | rms_low;
+ return copy_to_user(dest, &rms, 8);
+}
+
+static int hdsp_9652_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rms)
+{
+ int doublespeed = 0;
+ int i, j, channels, ofs;
+
+ if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DoubleSpeedStatus)
+ doublespeed = 1;
+ channels = doublespeed ? 14 : 26;
+ for (i = 0, j = 0; i < 26; ++i) {
+ if (doublespeed && (i & 4))
+ continue;
+ ofs = HDSP_9652_peakBase - j * 4;
+ if (copy_u32_le(&peak_rms->input_peaks[i], hdsp->iobase + ofs))
+ return -EFAULT;
+ ofs -= channels * 4;
+ if (copy_u32_le(&peak_rms->playback_peaks[i], hdsp->iobase + ofs))
+ return -EFAULT;
+ ofs -= channels * 4;
+ if (copy_u32_le(&peak_rms->output_peaks[i], hdsp->iobase + ofs))
+ return -EFAULT;
+ ofs = HDSP_9652_rmsBase + j * 8;
+ if (copy_u48_le(&peak_rms->input_rms[i], hdsp->iobase + ofs,
+ hdsp->iobase + ofs + 4))
+ return -EFAULT;
+ ofs += channels * 8;
+ if (copy_u48_le(&peak_rms->playback_rms[i], hdsp->iobase + ofs,
+ hdsp->iobase + ofs + 4))
+ return -EFAULT;
+ ofs += channels * 8;
+ if (copy_u48_le(&peak_rms->output_rms[i], hdsp->iobase + ofs,
+ hdsp->iobase + ofs + 4))
+ return -EFAULT;
+ j++;
+ }
+ return 0;
+}
+
+static int hdsp_9632_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rms)
+{
+ int i, j;
+ struct hdsp_9632_meters __iomem *m;
+ int doublespeed = 0;
+
+ if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DoubleSpeedStatus)
+ doublespeed = 1;
+ m = (struct hdsp_9632_meters __iomem *)(hdsp->iobase+HDSP_9632_metersBase);
+ for (i = 0, j = 0; i < 16; ++i, ++j) {
+ if (copy_u32_le(&peak_rms->input_peaks[i], &m->input_peak[j]))
+ return -EFAULT;
+ if (copy_u32_le(&peak_rms->playback_peaks[i], &m->playback_peak[j]))
+ return -EFAULT;
+ if (copy_u32_le(&peak_rms->output_peaks[i], &m->output_peak[j]))
+ return -EFAULT;
+ if (copy_u64_le(&peak_rms->input_rms[i], &m->input_rms_low[j],
+ &m->input_rms_high[j]))
+ return -EFAULT;
+ if (copy_u64_le(&peak_rms->playback_rms[i], &m->playback_rms_low[j],
+ &m->playback_rms_high[j]))
+ return -EFAULT;
+ if (copy_u64_le(&peak_rms->output_rms[i], &m->output_rms_low[j],
+ &m->output_rms_high[j]))
+ return -EFAULT;
+ if (doublespeed && i == 3) i += 4;
+ }
+ return 0;
+}
+
+static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rms)
+{
+ int i;
+
+ for (i = 0; i < 26; i++) {
+ if (copy_u32_le(&peak_rms->playback_peaks[i],
+ hdsp->iobase + HDSP_playbackPeakLevel + i * 4))
+ return -EFAULT;
+ if (copy_u32_le(&peak_rms->input_peaks[i],
+ hdsp->iobase + HDSP_inputPeakLevel + i * 4))
+ return -EFAULT;
+ }
+ for (i = 0; i < 28; i++) {
+ if (copy_u32_le(&peak_rms->output_peaks[i],
+ hdsp->iobase + HDSP_outputPeakLevel + i * 4))
+ return -EFAULT;
+ }
+ for (i = 0; i < 26; ++i) {
+ if (copy_u64_le(&peak_rms->playback_rms[i],
+ hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4,
+ hdsp->iobase + HDSP_playbackRmsLevel + i * 8))
+ return -EFAULT;
+ if (copy_u64_le(&peak_rms->input_rms[i],
+ hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4,
+ hdsp->iobase + HDSP_inputRmsLevel + i * 8))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct hdsp *hdsp = (struct hdsp *)hw->private_data;
+ void __user *argp = (void __user *)arg;
+ int err;
+
+ switch (cmd) {
+ case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: {
+ struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg;
+
+ err = hdsp_check_for_iobox(hdsp);
+ if (err < 0)
+ return err;
+
+ err = hdsp_check_for_firmware(hdsp, 1);
+ if (err < 0)
+ return err;
+
+ if (!(hdsp->state & HDSP_FirmwareLoaded)) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n");
+ return -EINVAL;
+ }
+
+ switch (hdsp->io_type) {
+ case H9652:
+ return hdsp_9652_get_peak(hdsp, peak_rms);
+ case H9632:
+ return hdsp_9632_get_peak(hdsp, peak_rms);
+ default:
+ return hdsp_get_peak(hdsp, peak_rms);
+ }
+ }
+ case SNDRV_HDSP_IOCTL_GET_CONFIG_INFO: {
+ struct hdsp_config_info info;
+ unsigned long flags;
+ int i;
+
+ err = hdsp_check_for_iobox(hdsp);
+ if (err < 0)
+ return err;
+
+ err = hdsp_check_for_firmware(hdsp, 1);
+ if (err < 0)
+ return err;
+
+ spin_lock_irqsave(&hdsp->lock, flags);
+ info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp);
+ info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp);
+ if (hdsp->io_type != H9632)
+ info.adatsync_sync_check = (unsigned char)hdsp_adatsync_sync_check(hdsp);
+ info.spdif_sync_check = (unsigned char)hdsp_spdif_sync_check(hdsp);
+ for (i = 0; i < ((hdsp->io_type != Multiface && hdsp->io_type != H9632) ? 3 : 1); ++i)
+ info.adat_sync_check[i] = (unsigned char)hdsp_adat_sync_check(hdsp, i);
+ info.spdif_in = (unsigned char)hdsp_spdif_in(hdsp);
+ info.spdif_out = (unsigned char)hdsp_spdif_out(hdsp);
+ info.spdif_professional = (unsigned char)hdsp_spdif_professional(hdsp);
+ info.spdif_emphasis = (unsigned char)hdsp_spdif_emphasis(hdsp);
+ info.spdif_nonaudio = (unsigned char)hdsp_spdif_nonaudio(hdsp);
+ info.spdif_sample_rate = hdsp_spdif_sample_rate(hdsp);
+ info.system_sample_rate = hdsp->system_sample_rate;
+ info.autosync_sample_rate = hdsp_external_sample_rate(hdsp);
+ info.system_clock_mode = (unsigned char)hdsp_system_clock_mode(hdsp);
+ info.clock_source = (unsigned char)hdsp_clock_source(hdsp);
+ info.autosync_ref = (unsigned char)hdsp_autosync_ref(hdsp);
+ info.line_out = (unsigned char)hdsp_line_out(hdsp);
+ if (hdsp->io_type == H9632) {
+ info.da_gain = (unsigned char)hdsp_da_gain(hdsp);
+ info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp);
+ info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp);
+ info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp);
+
+ }
+ if (hdsp->io_type == H9632 || hdsp->io_type == H9652)
+ info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp);
+ spin_unlock_irqrestore(&hdsp->lock, flags);
+ if (copy_to_user(argp, &info, sizeof(info)))
+ return -EFAULT;
+ break;
+ }
+ case SNDRV_HDSP_IOCTL_GET_9632_AEB: {
+ struct hdsp_9632_aeb h9632_aeb;
+
+ if (hdsp->io_type != H9632) return -EINVAL;
+ h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS;
+ h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS;
+ if (copy_to_user(argp, &h9632_aeb, sizeof(h9632_aeb)))
+ return -EFAULT;
+ break;
+ }
+ case SNDRV_HDSP_IOCTL_GET_VERSION: {
+ struct hdsp_version hdsp_version;
+ int err;
+
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL;
+ if (hdsp->io_type == Undefined) {
+ if ((err = hdsp_get_iobox_version(hdsp)) < 0)
+ return err;
+ }
+ hdsp_version.io_type = hdsp->io_type;
+ hdsp_version.firmware_rev = hdsp->firmware_rev;
+ if ((err = copy_to_user(argp, &hdsp_version, sizeof(hdsp_version))))
+ return -EFAULT;
+ break;
+ }
+ case SNDRV_HDSP_IOCTL_UPLOAD_FIRMWARE: {
+ struct hdsp_firmware __user *firmware;
+ u32 __user *firmware_data;
+ int err;
+
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL;
+ /* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */
+ if (hdsp->io_type == Undefined) return -EINVAL;
+
+ if (hdsp->state & (HDSP_FirmwareCached | HDSP_FirmwareLoaded))
+ return -EBUSY;
+
+ snd_printk(KERN_INFO "Hammerfall-DSP: initializing firmware upload\n");
+ firmware = (struct hdsp_firmware __user *)argp;
+
+ if (get_user(firmware_data, &firmware->firmware_data))
+ return -EFAULT;
+
+ if (hdsp_check_for_iobox (hdsp))
+ return -EIO;
+
+ if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0)
+ return -EFAULT;
+
+ hdsp->state |= HDSP_FirmwareCached;
+
+ if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0)
+ return err;
+
+ if (!(hdsp->state & HDSP_InitializationComplete)) {
+ if ((err = snd_hdsp_enable_io(hdsp)) < 0)
+ return err;
+
+ snd_hdsp_initialize_channels(hdsp);
+ snd_hdsp_initialize_midi_flush(hdsp);
+
+ if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n");
+ return err;
+ }
+ }
+ break;
+ }
+ case SNDRV_HDSP_IOCTL_GET_MIXER: {
+ struct hdsp_mixer __user *mixer = (struct hdsp_mixer __user *)argp;
+ if (copy_to_user(mixer->matrix, hdsp->mixer_matrix, sizeof(unsigned short)*HDSP_MATRIX_MIXER_SIZE))
+ return -EFAULT;
+ break;
+ }
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static struct snd_pcm_ops snd_hdsp_playback_ops = {
+ .open = snd_hdsp_playback_open,
+ .close = snd_hdsp_playback_release,
+ .ioctl = snd_hdsp_ioctl,
+ .hw_params = snd_hdsp_hw_params,
+ .prepare = snd_hdsp_prepare,
+ .trigger = snd_hdsp_trigger,
+ .pointer = snd_hdsp_hw_pointer,
+ .copy = snd_hdsp_playback_copy,
+ .silence = snd_hdsp_hw_silence,
+};
+
+static struct snd_pcm_ops snd_hdsp_capture_ops = {
+ .open = snd_hdsp_capture_open,
+ .close = snd_hdsp_capture_release,
+ .ioctl = snd_hdsp_ioctl,
+ .hw_params = snd_hdsp_hw_params,
+ .prepare = snd_hdsp_prepare,
+ .trigger = snd_hdsp_trigger,
+ .pointer = snd_hdsp_hw_pointer,
+ .copy = snd_hdsp_capture_copy,
+};
+
+static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
+{
+ struct snd_hwdep *hw;
+ int err;
+
+ if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0)
+ return err;
+
+ hdsp->hwdep = hw;
+ hw->private_data = hdsp;
+ strcpy(hw->name, "HDSP hwdep interface");
+
+ hw->ops.open = snd_hdsp_hwdep_dummy_op;
+ hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
+ hw->ops.release = snd_hdsp_hwdep_dummy_op;
+
+ return 0;
+}
+
+static int snd_hdsp_create_pcm(struct snd_card *card, struct hdsp *hdsp)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(card, hdsp->card_name, 0, 1, 1, &pcm)) < 0)
+ return err;
+
+ hdsp->pcm = pcm;
+ pcm->private_data = hdsp;
+ strcpy(pcm->name, hdsp->card_name);
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_hdsp_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_hdsp_capture_ops);
+
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+
+ return 0;
+}
+
+static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp)
+{
+ hdsp->control2_register |= HDSP_9652_ENABLE_MIXER;
+ hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register);
+}
+
+static int snd_hdsp_enable_io (struct hdsp *hdsp)
+{
+ int i;
+
+ if (hdsp_fifo_wait (hdsp, 0, 100)) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n");
+ return -EIO;
+ }
+
+ for (i = 0; i < hdsp->max_channels; ++i) {
+ hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1);
+ hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1);
+ }
+
+ return 0;
+}
+
+static void snd_hdsp_initialize_channels(struct hdsp *hdsp)
+{
+ int status, aebi_channels, aebo_channels;
+
+ switch (hdsp->io_type) {
+ case Digiface:
+ hdsp->card_name = "RME Hammerfall DSP + Digiface";
+ hdsp->ss_in_channels = hdsp->ss_out_channels = DIGIFACE_SS_CHANNELS;
+ hdsp->ds_in_channels = hdsp->ds_out_channels = DIGIFACE_DS_CHANNELS;
+ break;
+
+ case H9652:
+ hdsp->card_name = "RME Hammerfall HDSP 9652";
+ hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS;
+ hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS;
+ break;
+
+ case H9632:
+ status = hdsp_read(hdsp, HDSP_statusRegister);
+ /* HDSP_AEBx bits are low when AEB are connected */
+ aebi_channels = (status & HDSP_AEBI) ? 0 : 4;
+ aebo_channels = (status & HDSP_AEBO) ? 0 : 4;
+ hdsp->card_name = "RME Hammerfall HDSP 9632";
+ hdsp->ss_in_channels = H9632_SS_CHANNELS+aebi_channels;
+ hdsp->ds_in_channels = H9632_DS_CHANNELS+aebi_channels;
+ hdsp->qs_in_channels = H9632_QS_CHANNELS+aebi_channels;
+ hdsp->ss_out_channels = H9632_SS_CHANNELS+aebo_channels;
+ hdsp->ds_out_channels = H9632_DS_CHANNELS+aebo_channels;
+ hdsp->qs_out_channels = H9632_QS_CHANNELS+aebo_channels;
+ break;
+
+ case Multiface:
+ hdsp->card_name = "RME Hammerfall DSP + Multiface";
+ hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS;
+ hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS;
+ break;
+
+ default:
+ /* should never get here */
+ break;
+ }
+}
+
+static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp)
+{
+ snd_hdsp_flush_midi_input (hdsp, 0);
+ snd_hdsp_flush_midi_input (hdsp, 1);
+}
+
+static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp)
+{
+ int err;
+
+ if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n");
+ return err;
+ }
+
+
+ if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n");
+ return err;
+ }
+
+ if (hdsp->io_type == Digiface || hdsp->io_type == H9652) {
+ if ((err = snd_hdsp_create_midi(card, hdsp, 1)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: Error creating second midi interface\n");
+ return err;
+ }
+ }
+
+ if ((err = snd_hdsp_create_controls(card, hdsp)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: Error creating ctl interface\n");
+ return err;
+ }
+
+ snd_hdsp_proc_init(hdsp);
+
+ hdsp->system_sample_rate = -1;
+ hdsp->playback_pid = -1;
+ hdsp->capture_pid = -1;
+ hdsp->capture_substream = NULL;
+ hdsp->playback_substream = NULL;
+
+ if ((err = snd_hdsp_set_defaults(hdsp)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n");
+ return err;
+ }
+
+ if (!(hdsp->state & HDSP_InitializationComplete)) {
+ strcpy(card->shortname, "Hammerfall DSP");
+ sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name,
+ hdsp->port, hdsp->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n");
+ return err;
+ }
+ hdsp->state |= HDSP_InitializationComplete;
+ }
+
+ return 0;
+}
+
+#ifdef HDSP_FW_LOADER
+/* load firmware via hotplug fw loader */
+static int hdsp_request_fw_loader(struct hdsp *hdsp)
+{
+ const char *fwfile;
+ const struct firmware *fw;
+ int err;
+
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
+ return 0;
+ if (hdsp->io_type == Undefined) {
+ if ((err = hdsp_get_iobox_version(hdsp)) < 0)
+ return err;
+ if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
+ return 0;
+ }
+
+ /* caution: max length of firmware filename is 30! */
+ switch (hdsp->io_type) {
+ case Multiface:
+ if (hdsp->firmware_rev == 0xa)
+ fwfile = "multiface_firmware.bin";
+ else
+ fwfile = "multiface_firmware_rev11.bin";
+ break;
+ case Digiface:
+ if (hdsp->firmware_rev == 0xa)
+ fwfile = "digiface_firmware.bin";
+ else
+ fwfile = "digiface_firmware_rev11.bin";
+ break;
+ default:
+ snd_printk(KERN_ERR "Hammerfall-DSP: invalid io_type %d\n", hdsp->io_type);
+ return -EINVAL;
+ }
+
+ if (request_firmware(&fw, fwfile, &hdsp->pci->dev)) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: cannot load firmware %s\n", fwfile);
+ return -ENOENT;
+ }
+ if (fw->size < sizeof(hdsp->firmware_cache)) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: too short firmware size %d (expected %d)\n",
+ (int)fw->size, (int)sizeof(hdsp->firmware_cache));
+ release_firmware(fw);
+ return -EINVAL;
+ }
+
+ memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache));
+
+ release_firmware(fw);
+
+ hdsp->state |= HDSP_FirmwareCached;
+
+ if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0)
+ return err;
+
+ if (!(hdsp->state & HDSP_InitializationComplete)) {
+ if ((err = snd_hdsp_enable_io(hdsp)) < 0)
+ return err;
+
+ if ((err = snd_hdsp_create_hwdep(hdsp->card, hdsp)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: error creating hwdep device\n");
+ return err;
+ }
+ snd_hdsp_initialize_channels(hdsp);
+ snd_hdsp_initialize_midi_flush(hdsp);
+ if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n");
+ return err;
+ }
+ }
+ return 0;
+}
+#endif
+
+static int __devinit snd_hdsp_create(struct snd_card *card,
+ struct hdsp *hdsp)
+{
+ struct pci_dev *pci = hdsp->pci;
+ int err;
+ int is_9652 = 0;
+ int is_9632 = 0;
+
+ hdsp->irq = -1;
+ hdsp->state = 0;
+ hdsp->midi[0].rmidi = NULL;
+ hdsp->midi[1].rmidi = NULL;
+ hdsp->midi[0].input = NULL;
+ hdsp->midi[1].input = NULL;
+ hdsp->midi[0].output = NULL;
+ hdsp->midi[1].output = NULL;
+ hdsp->midi[0].pending = 0;
+ hdsp->midi[1].pending = 0;
+ spin_lock_init(&hdsp->midi[0].lock);
+ spin_lock_init(&hdsp->midi[1].lock);
+ hdsp->iobase = NULL;
+ hdsp->control_register = 0;
+ hdsp->control2_register = 0;
+ hdsp->io_type = Undefined;
+ hdsp->max_channels = 26;
+
+ hdsp->card = card;
+
+ spin_lock_init(&hdsp->lock);
+
+ tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp);
+
+ pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev);
+ hdsp->firmware_rev &= 0xff;
+
+ /* From Martin Bjoernsen :
+ "It is important that the card's latency timer register in
+ the PCI configuration space is set to a value much larger
+ than 0 by the computer's BIOS or the driver.
+ The windows driver always sets this 8 bit register [...]
+ to its maximum 255 to avoid problems with some computers."
+ */
+ pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF);
+
+ strcpy(card->driver, "H-DSP");
+ strcpy(card->mixername, "Xilinx FPGA");
+
+ if (hdsp->firmware_rev < 0xa)
+ return -ENODEV;
+ else if (hdsp->firmware_rev < 0x64)
+ hdsp->card_name = "RME Hammerfall DSP";
+ else if (hdsp->firmware_rev < 0x96) {
+ hdsp->card_name = "RME HDSP 9652";
+ is_9652 = 1;
+ } else {
+ hdsp->card_name = "RME HDSP 9632";
+ hdsp->max_channels = 16;
+ is_9632 = 1;
+ }
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ pci_set_master(hdsp->pci);
+
+ if ((err = pci_request_regions(pci, "hdsp")) < 0)
+ return err;
+ hdsp->port = pci_resource_start(pci, 0);
+ if ((hdsp->iobase = ioremap_nocache(hdsp->port, HDSP_IO_EXTENT)) == NULL) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: unable to remap region 0x%lx-0x%lx\n", hdsp->port, hdsp->port + HDSP_IO_EXTENT - 1);
+ return -EBUSY;
+ }
+
+ if (request_irq(pci->irq, snd_hdsp_interrupt, IRQF_SHARED,
+ "hdsp", hdsp)) {
+ snd_printk(KERN_ERR "Hammerfall-DSP: unable to use IRQ %d\n", pci->irq);
+ return -EBUSY;
+ }
+
+ hdsp->irq = pci->irq;
+ hdsp->precise_ptr = 0;
+ hdsp->use_midi_tasklet = 1;
+ hdsp->dds_value = 0;
+
+ if ((err = snd_hdsp_initialize_memory(hdsp)) < 0)
+ return err;
+
+ if (!is_9652 && !is_9632) {
+ /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
+ ssleep(2);
+
+ err = hdsp_check_for_iobox(hdsp);
+ if (err < 0)
+ return err;
+
+ if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
+#ifdef HDSP_FW_LOADER
+ if ((err = hdsp_request_fw_loader(hdsp)) < 0)
+ /* we don't fail as this can happen
+ if userspace is not ready for
+ firmware upload
+ */
+ snd_printk(KERN_ERR "Hammerfall-DSP: couldn't get firmware from userspace. try using hdsploader\n");
+ else
+ /* init is complete, we return */
+ return 0;
+#endif
+ /* we defer initialization */
+ snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n");
+ if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
+ return err;
+ return 0;
+ } else {
+ snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n");
+ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1)
+ hdsp->io_type = Multiface;
+ else
+ hdsp->io_type = Digiface;
+ }
+ }
+
+ if ((err = snd_hdsp_enable_io(hdsp)) != 0)
+ return err;
+
+ if (is_9652)
+ hdsp->io_type = H9652;
+
+ if (is_9632)
+ hdsp->io_type = H9632;
+
+ if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
+ return err;
+
+ snd_hdsp_initialize_channels(hdsp);
+ snd_hdsp_initialize_midi_flush(hdsp);
+
+ hdsp->state |= HDSP_FirmwareLoaded;
+
+ if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0)
+ return err;
+
+ return 0;
+}
+
+static int snd_hdsp_free(struct hdsp *hdsp)
+{
+ if (hdsp->port) {
+ /* stop the audio, and cancel all interrupts */
+ tasklet_kill(&hdsp->midi_tasklet);
+ hdsp->control_register &= ~(HDSP_Start|HDSP_AudioInterruptEnable|HDSP_Midi0InterruptEnable|HDSP_Midi1InterruptEnable);
+ hdsp_write (hdsp, HDSP_controlRegister, hdsp->control_register);
+ }
+
+ if (hdsp->irq >= 0)
+ free_irq(hdsp->irq, (void *)hdsp);
+
+ snd_hdsp_free_buffers(hdsp);
+
+ if (hdsp->iobase)
+ iounmap(hdsp->iobase);
+
+ if (hdsp->port)
+ pci_release_regions(hdsp->pci);
+
+ pci_disable_device(hdsp->pci);
+ return 0;
+}
+
+static void snd_hdsp_card_free(struct snd_card *card)
+{
+ struct hdsp *hdsp = (struct hdsp *) card->private_data;
+
+ if (hdsp)
+ snd_hdsp_free(hdsp);
+}
+
+static int __devinit snd_hdsp_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct hdsp *hdsp;
+ struct snd_card *card;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ if (!(card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct hdsp))))
+ return -ENOMEM;
+
+ hdsp = (struct hdsp *) card->private_data;
+ card->private_free = snd_hdsp_card_free;
+ hdsp->dev = dev;
+ hdsp->pci = pci;
+ snd_card_set_dev(card, &pci->dev);
+
+ if ((err = snd_hdsp_create(card, hdsp)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->shortname, "Hammerfall DSP");
+ sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name,
+ hdsp->port, hdsp->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_hdsp_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "RME Hammerfall DSP",
+ .id_table = snd_hdsp_ids,
+ .probe = snd_hdsp_probe,
+ .remove = __devexit_p(snd_hdsp_remove),
+};
+
+static int __init alsa_card_hdsp_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_hdsp_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_hdsp_init)
+module_exit(alsa_card_hdsp_exit)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
new file mode 100644
index 0000000..98762f9
--- /dev/null
+++ b/sound/pci/rme9652/hdspm.c
@@ -0,0 +1,4566 @@
+/*
+ * ALSA driver for RME Hammerfall DSP MADI audio interface(s)
+ *
+ * Copyright (c) 2003 Winfried Ritsch (IEM)
+ * code based on hdsp.c Paul Davis
+ * Marcus Andersson
+ * Thomas Charbonnel
+ * Modified 2006-06-01 for AES32 support by Remy Bruno
+ * <remy.bruno@trinnov.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <asm/io.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+#include <sound/rawmidi.h>
+#include <sound/hwdep.h>
+#include <sound/initval.h>
+
+#include <sound/hdspm.h>
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */
+
+/* Disable precise pointer at start */
+static int precise_ptr[SNDRV_CARDS];
+
+/* Send all playback to line outs */
+static int line_outs_monitor[SNDRV_CARDS];
+
+/* Enable Analog Outs on Channel 63/64 by default */
+static int enable_monitor[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for RME HDSPM interface.");
+
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for RME HDSPM interface.");
+
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable/disable specific HDSPM soundcards.");
+
+module_param_array(precise_ptr, bool, NULL, 0444);
+MODULE_PARM_DESC(precise_ptr, "Enable or disable precise pointer.");
+
+module_param_array(line_outs_monitor, bool, NULL, 0444);
+MODULE_PARM_DESC(line_outs_monitor,
+ "Send playback streams to analog outs by default.");
+
+module_param_array(enable_monitor, bool, NULL, 0444);
+MODULE_PARM_DESC(enable_monitor,
+ "Enable Analog Out on Channel 63/64 by default.");
+
+MODULE_AUTHOR
+ ("Winfried Ritsch <ritsch_AT_iem.at>, "
+ "Paul Davis <paul@linuxaudiosystems.com>, "
+ "Marcus Andersson, Thomas Charbonnel <thomas@undata.org>, "
+ "Remy Bruno <remy.bruno@trinnov.com>");
+MODULE_DESCRIPTION("RME HDSPM");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
+
+/* --- Write registers. ---
+ These are defined as byte-offsets from the iobase value. */
+
+#define HDSPM_controlRegister 64
+#define HDSPM_interruptConfirmation 96
+#define HDSPM_control2Reg 256 /* not in specs ???????? */
+#define HDSPM_freqReg 256 /* for AES32 */
+#define HDSPM_midiDataOut0 352 /* just believe in old code */
+#define HDSPM_midiDataOut1 356
+#define HDSPM_eeprom_wr 384 /* for AES32 */
+
+/* DMA enable for 64 channels, only Bit 0 is relevant */
+#define HDSPM_outputEnableBase 512 /* 512-767 input DMA */
+#define HDSPM_inputEnableBase 768 /* 768-1023 output DMA */
+
+/* 16 page addresses for each of the 64 channels DMA buffer in and out
+ (each 64k=16*4k) Buffer must be 4k aligned (which is default i386 ????) */
+#define HDSPM_pageAddressBufferOut 8192
+#define HDSPM_pageAddressBufferIn (HDSPM_pageAddressBufferOut+64*16*4)
+
+#define HDSPM_MADI_mixerBase 32768 /* 32768-65535 for 2x64x64 Fader */
+
+#define HDSPM_MATRIX_MIXER_SIZE 8192 /* = 2*64*64 * 4 Byte => 32kB */
+
+/* --- Read registers. ---
+ These are defined as byte-offsets from the iobase value */
+#define HDSPM_statusRegister 0
+/*#define HDSPM_statusRegister2 96 */
+/* after RME Windows driver sources, status2 is 4-byte word # 48 = word at
+ * offset 192, for AES32 *and* MADI
+ * => need to check that offset 192 is working on MADI */
+#define HDSPM_statusRegister2 192
+#define HDSPM_timecodeRegister 128
+
+#define HDSPM_midiDataIn0 360
+#define HDSPM_midiDataIn1 364
+
+/* status is data bytes in MIDI-FIFO (0-128) */
+#define HDSPM_midiStatusOut0 384
+#define HDSPM_midiStatusOut1 388
+#define HDSPM_midiStatusIn0 392
+#define HDSPM_midiStatusIn1 396
+
+
+/* the meters are regular i/o-mapped registers, but offset
+ considerably from the rest. the peak registers are reset
+ when read; the least-significant 4 bits are full-scale counters;
+ the actual peak value is in the most-significant 24 bits.
+*/
+#define HDSPM_MADI_peakrmsbase 4096 /* 4096-8191 2x64x32Bit Meters */
+
+/* --- Control Register bits --------- */
+#define HDSPM_Start (1<<0) /* start engine */
+
+#define HDSPM_Latency0 (1<<1) /* buffer size = 2^n */
+#define HDSPM_Latency1 (1<<2) /* where n is defined */
+#define HDSPM_Latency2 (1<<3) /* by Latency{2,1,0} */
+
+#define HDSPM_ClockModeMaster (1<<4) /* 1=Master, 0=Slave/Autosync */
+
+#define HDSPM_AudioInterruptEnable (1<<5) /* what do you think ? */
+
+#define HDSPM_Frequency0 (1<<6) /* 0=44.1kHz/88.2kHz 1=48kHz/96kHz */
+#define HDSPM_Frequency1 (1<<7) /* 0=32kHz/64kHz */
+#define HDSPM_DoubleSpeed (1<<8) /* 0=normal speed, 1=double speed */
+#define HDSPM_QuadSpeed (1<<31) /* quad speed bit */
+
+#define HDSPM_Professional (1<<9) /* Professional */ /* AES32 ONLY */
+#define HDSPM_TX_64ch (1<<10) /* Output 64channel MODE=1,
+ 56channelMODE=0 */ /* MADI ONLY*/
+#define HDSPM_Emphasis (1<<10) /* Emphasis */ /* AES32 ONLY */
+
+#define HDSPM_AutoInp (1<<11) /* Auto Input (takeover) == Safe Mode,
+ 0=off, 1=on */ /* MADI ONLY */
+#define HDSPM_Dolby (1<<11) /* Dolby = "NonAudio" ?? */ /* AES32 ONLY */
+
+#define HDSPM_InputSelect0 (1<<14) /* Input select 0= optical, 1=coax
+ * -- MADI ONLY
+ */
+#define HDSPM_InputSelect1 (1<<15) /* should be 0 */
+
+#define HDSPM_SyncRef0 (1<<16) /* 0=WOrd, 1=MADI */
+#define HDSPM_SyncRef1 (1<<17) /* for AES32: SyncRefN codes the AES # */
+#define HDSPM_SyncRef2 (1<<13)
+#define HDSPM_SyncRef3 (1<<25)
+
+#define HDSPM_SMUX (1<<18) /* Frame ??? */ /* MADI ONY */
+#define HDSPM_clr_tms (1<<19) /* clear track marker, do not use
+ AES additional bits in
+ lower 5 Audiodatabits ??? */
+#define HDSPM_taxi_reset (1<<20) /* ??? */ /* MADI ONLY ? */
+#define HDSPM_WCK48 (1<<20) /* Frame ??? = HDSPM_SMUX */ /* AES32 ONLY */
+
+#define HDSPM_Midi0InterruptEnable (1<<22)
+#define HDSPM_Midi1InterruptEnable (1<<23)
+
+#define HDSPM_LineOut (1<<24) /* Analog Out on channel 63/64 on=1, mute=0 */
+
+#define HDSPM_DS_DoubleWire (1<<26) /* AES32 ONLY */
+#define HDSPM_QS_DoubleWire (1<<27) /* AES32 ONLY */
+#define HDSPM_QS_QuadWire (1<<28) /* AES32 ONLY */
+
+#define HDSPM_wclk_sel (1<<30)
+
+/* --- bit helper defines */
+#define HDSPM_LatencyMask (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
+#define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1|\
+ HDSPM_DoubleSpeed|HDSPM_QuadSpeed)
+#define HDSPM_InputMask (HDSPM_InputSelect0|HDSPM_InputSelect1)
+#define HDSPM_InputOptical 0
+#define HDSPM_InputCoaxial (HDSPM_InputSelect0)
+#define HDSPM_SyncRefMask (HDSPM_SyncRef0|HDSPM_SyncRef1|\
+ HDSPM_SyncRef2|HDSPM_SyncRef3)
+#define HDSPM_SyncRef_Word 0
+#define HDSPM_SyncRef_MADI (HDSPM_SyncRef0)
+
+#define HDSPM_SYNC_FROM_WORD 0 /* Preferred sync reference */
+#define HDSPM_SYNC_FROM_MADI 1 /* choices - used by "pref_sync_ref" */
+
+#define HDSPM_Frequency32KHz HDSPM_Frequency0
+#define HDSPM_Frequency44_1KHz HDSPM_Frequency1
+#define HDSPM_Frequency48KHz (HDSPM_Frequency1|HDSPM_Frequency0)
+#define HDSPM_Frequency64KHz (HDSPM_DoubleSpeed|HDSPM_Frequency0)
+#define HDSPM_Frequency88_2KHz (HDSPM_DoubleSpeed|HDSPM_Frequency1)
+#define HDSPM_Frequency96KHz (HDSPM_DoubleSpeed|HDSPM_Frequency1|\
+ HDSPM_Frequency0)
+#define HDSPM_Frequency128KHz (HDSPM_QuadSpeed|HDSPM_Frequency0)
+#define HDSPM_Frequency176_4KHz (HDSPM_QuadSpeed|HDSPM_Frequency1)
+#define HDSPM_Frequency192KHz (HDSPM_QuadSpeed|HDSPM_Frequency1|\
+ HDSPM_Frequency0)
+
+/* --- for internal discrimination */
+#define HDSPM_CLOCK_SOURCE_AUTOSYNC 0 /* Sample Clock Sources */
+#define HDSPM_CLOCK_SOURCE_INTERNAL_32KHZ 1
+#define HDSPM_CLOCK_SOURCE_INTERNAL_44_1KHZ 2
+#define HDSPM_CLOCK_SOURCE_INTERNAL_48KHZ 3
+#define HDSPM_CLOCK_SOURCE_INTERNAL_64KHZ 4
+#define HDSPM_CLOCK_SOURCE_INTERNAL_88_2KHZ 5
+#define HDSPM_CLOCK_SOURCE_INTERNAL_96KHZ 6
+#define HDSPM_CLOCK_SOURCE_INTERNAL_128KHZ 7
+#define HDSPM_CLOCK_SOURCE_INTERNAL_176_4KHZ 8
+#define HDSPM_CLOCK_SOURCE_INTERNAL_192KHZ 9
+
+/* Synccheck Status */
+#define HDSPM_SYNC_CHECK_NO_LOCK 0
+#define HDSPM_SYNC_CHECK_LOCK 1
+#define HDSPM_SYNC_CHECK_SYNC 2
+
+/* AutoSync References - used by "autosync_ref" control switch */
+#define HDSPM_AUTOSYNC_FROM_WORD 0
+#define HDSPM_AUTOSYNC_FROM_MADI 1
+#define HDSPM_AUTOSYNC_FROM_NONE 2
+
+/* Possible sources of MADI input */
+#define HDSPM_OPTICAL 0 /* optical */
+#define HDSPM_COAXIAL 1 /* BNC */
+
+#define hdspm_encode_latency(x) (((x)<<1) & HDSPM_LatencyMask)
+#define hdspm_decode_latency(x) (((x) & HDSPM_LatencyMask)>>1)
+
+#define hdspm_encode_in(x) (((x)&0x3)<<14)
+#define hdspm_decode_in(x) (((x)>>14)&0x3)
+
+/* --- control2 register bits --- */
+#define HDSPM_TMS (1<<0)
+#define HDSPM_TCK (1<<1)
+#define HDSPM_TDI (1<<2)
+#define HDSPM_JTAG (1<<3)
+#define HDSPM_PWDN (1<<4)
+#define HDSPM_PROGRAM (1<<5)
+#define HDSPM_CONFIG_MODE_0 (1<<6)
+#define HDSPM_CONFIG_MODE_1 (1<<7)
+/*#define HDSPM_VERSION_BIT (1<<8) not defined any more*/
+#define HDSPM_BIGENDIAN_MODE (1<<9)
+#define HDSPM_RD_MULTIPLE (1<<10)
+
+/* --- Status Register bits --- */ /* MADI ONLY */ /* Bits defined here and
+ that do not conflict with specific bits for AES32 seem to be valid also
+ for the AES32
+ */
+#define HDSPM_audioIRQPending (1<<0) /* IRQ is high and pending */
+#define HDSPM_RX_64ch (1<<1) /* Input 64chan. MODE=1, 56chn MODE=0 */
+#define HDSPM_AB_int (1<<2) /* InputChannel Opt=0, Coax=1
+ * (like inp0)
+ */
+#define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */
+
+#define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
+ /* since 64byte accurate last 6 bits
+ are not used */
+
+#define HDSPM_madiSync (1<<18) /* MADI is in sync */
+#define HDSPM_DoubleSpeedStatus (1<<19) /* (input) card in double speed */
+
+#define HDSPM_madiFreq0 (1<<22) /* system freq 0=error */
+#define HDSPM_madiFreq1 (1<<23) /* 1=32, 2=44.1 3=48 */
+#define HDSPM_madiFreq2 (1<<24) /* 4=64, 5=88.2 6=96 */
+#define HDSPM_madiFreq3 (1<<25) /* 7=128, 8=176.4 9=192 */
+
+#define HDSPM_BufferID (1<<26) /* (Double)Buffer ID toggles with
+ * Interrupt
+ */
+#define HDSPM_midi0IRQPending (1<<30) /* MIDI IRQ is pending */
+#define HDSPM_midi1IRQPending (1<<31) /* and aktiv */
+
+/* --- status bit helpers */
+#define HDSPM_madiFreqMask (HDSPM_madiFreq0|HDSPM_madiFreq1|\
+ HDSPM_madiFreq2|HDSPM_madiFreq3)
+#define HDSPM_madiFreq32 (HDSPM_madiFreq0)
+#define HDSPM_madiFreq44_1 (HDSPM_madiFreq1)
+#define HDSPM_madiFreq48 (HDSPM_madiFreq0|HDSPM_madiFreq1)
+#define HDSPM_madiFreq64 (HDSPM_madiFreq2)
+#define HDSPM_madiFreq88_2 (HDSPM_madiFreq0|HDSPM_madiFreq2)
+#define HDSPM_madiFreq96 (HDSPM_madiFreq1|HDSPM_madiFreq2)
+#define HDSPM_madiFreq128 (HDSPM_madiFreq0|HDSPM_madiFreq1|HDSPM_madiFreq2)
+#define HDSPM_madiFreq176_4 (HDSPM_madiFreq3)
+#define HDSPM_madiFreq192 (HDSPM_madiFreq3|HDSPM_madiFreq0)
+
+/* Status2 Register bits */ /* MADI ONLY */
+
+#define HDSPM_version0 (1<<0) /* not realy defined but I guess */
+#define HDSPM_version1 (1<<1) /* in former cards it was ??? */
+#define HDSPM_version2 (1<<2)
+
+#define HDSPM_wcLock (1<<3) /* Wordclock is detected and locked */
+#define HDSPM_wcSync (1<<4) /* Wordclock is in sync with systemclock */
+
+#define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */
+#define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */
+#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */
+/* missing Bit for 111=128, 1000=176.4, 1001=192 */
+
+#define HDSPM_SelSyncRef0 (1<<8) /* Sync Source in slave mode */
+#define HDSPM_SelSyncRef1 (1<<9) /* 000=word, 001=MADI, */
+#define HDSPM_SelSyncRef2 (1<<10) /* 111=no valid signal */
+
+#define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync)
+
+#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2)
+#define HDSPM_wcFreq32 (HDSPM_wc_freq0)
+#define HDSPM_wcFreq44_1 (HDSPM_wc_freq1)
+#define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1)
+#define HDSPM_wcFreq64 (HDSPM_wc_freq2)
+#define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2)
+#define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2)
+
+
+#define HDSPM_SelSyncRefMask (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|\
+ HDSPM_SelSyncRef2)
+#define HDSPM_SelSyncRef_WORD 0
+#define HDSPM_SelSyncRef_MADI (HDSPM_SelSyncRef0)
+#define HDSPM_SelSyncRef_NVALID (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|\
+ HDSPM_SelSyncRef2)
+
+/*
+ For AES32, bits for status, status2 and timecode are different
+*/
+/* status */
+#define HDSPM_AES32_wcLock 0x0200000
+#define HDSPM_AES32_wcFreq_bit 22
+/* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function
+ HDSPM_bit2freq */
+#define HDSPM_AES32_syncref_bit 16
+/* (status >> HDSPM_AES32_syncref_bit) & 0xF gives sync source */
+
+#define HDSPM_AES32_AUTOSYNC_FROM_WORD 0
+#define HDSPM_AES32_AUTOSYNC_FROM_AES1 1
+#define HDSPM_AES32_AUTOSYNC_FROM_AES2 2
+#define HDSPM_AES32_AUTOSYNC_FROM_AES3 3
+#define HDSPM_AES32_AUTOSYNC_FROM_AES4 4
+#define HDSPM_AES32_AUTOSYNC_FROM_AES5 5
+#define HDSPM_AES32_AUTOSYNC_FROM_AES6 6
+#define HDSPM_AES32_AUTOSYNC_FROM_AES7 7
+#define HDSPM_AES32_AUTOSYNC_FROM_AES8 8
+#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9
+
+/* status2 */
+/* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */
+#define HDSPM_LockAES 0x80
+#define HDSPM_LockAES1 0x80
+#define HDSPM_LockAES2 0x40
+#define HDSPM_LockAES3 0x20
+#define HDSPM_LockAES4 0x10
+#define HDSPM_LockAES5 0x8
+#define HDSPM_LockAES6 0x4
+#define HDSPM_LockAES7 0x2
+#define HDSPM_LockAES8 0x1
+/*
+ Timecode
+ After windows driver sources, bits 4*i to 4*i+3 give the input frequency on
+ AES i+1
+ bits 3210
+ 0001 32kHz
+ 0010 44.1kHz
+ 0011 48kHz
+ 0100 64kHz
+ 0101 88.2kHz
+ 0110 96kHz
+ 0111 128kHz
+ 1000 176.4kHz
+ 1001 192kHz
+ NB: Timecode register doesn't seem to work on AES32 card revision 230
+*/
+
+/* Mixer Values */
+#define UNITY_GAIN 32768 /* = 65536/2 */
+#define MINUS_INFINITY_GAIN 0
+
+/* Number of channels for different Speed Modes */
+#define MADI_SS_CHANNELS 64
+#define MADI_DS_CHANNELS 32
+#define MADI_QS_CHANNELS 16
+
+/* the size of a substream (1 mono data stream) */
+#define HDSPM_CHANNEL_BUFFER_SAMPLES (16*1024)
+#define HDSPM_CHANNEL_BUFFER_BYTES (4*HDSPM_CHANNEL_BUFFER_SAMPLES)
+
+/* the size of the area we need to allocate for DMA transfers. the
+ size is the same regardless of the number of channels, and
+ also the latency to use.
+ for one direction !!!
+*/
+#define HDSPM_DMA_AREA_BYTES (HDSPM_MAX_CHANNELS * HDSPM_CHANNEL_BUFFER_BYTES)
+#define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
+
+/* revisions >= 230 indicate AES32 card */
+#define HDSPM_AESREVISION 230
+
+/* speed factor modes */
+#define HDSPM_SPEED_SINGLE 0
+#define HDSPM_SPEED_DOUBLE 1
+#define HDSPM_SPEED_QUAD 2
+/* names for speed modes */
+static char *hdspm_speed_names[] = { "single", "double", "quad" };
+
+struct hdspm_midi {
+ struct hdspm *hdspm;
+ int id;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *input;
+ struct snd_rawmidi_substream *output;
+ char istimer; /* timer in use */
+ struct timer_list timer;
+ spinlock_t lock;
+ int pending;
+};
+
+struct hdspm {
+ spinlock_t lock;
+ /* only one playback and/or capture stream */
+ struct snd_pcm_substream *capture_substream;
+ struct snd_pcm_substream *playback_substream;
+
+ char *card_name; /* for procinfo */
+ unsigned short firmware_rev; /* dont know if relevant (yes if AES32)*/
+
+ unsigned char is_aes32; /* indicates if card is AES32 */
+
+ int precise_ptr; /* use precise pointers, to be tested */
+ int monitor_outs; /* set up monitoring outs init flag */
+
+ u32 control_register; /* cached value */
+ u32 control2_register; /* cached value */
+
+ struct hdspm_midi midi[2];
+ struct tasklet_struct midi_tasklet;
+
+ size_t period_bytes;
+ unsigned char ss_channels; /* channels of card in single speed */
+ unsigned char ds_channels; /* Double Speed */
+ unsigned char qs_channels; /* Quad Speed */
+
+ unsigned char *playback_buffer; /* suitably aligned address */
+ unsigned char *capture_buffer; /* suitably aligned address */
+
+ pid_t capture_pid; /* process id which uses capture */
+ pid_t playback_pid; /* process id which uses capture */
+ int running; /* running status */
+
+ int last_external_sample_rate; /* samplerate mystic ... */
+ int last_internal_sample_rate;
+ int system_sample_rate;
+
+ char *channel_map; /* channel map for DS and Quadspeed */
+
+ int dev; /* Hardware vars... */
+ int irq;
+ unsigned long port;
+ void __iomem *iobase;
+
+ int irq_count; /* for debug */
+
+ struct snd_card *card; /* one card */
+ struct snd_pcm *pcm; /* has one pcm */
+ struct snd_hwdep *hwdep; /* and a hwdep for additional ioctl */
+ struct pci_dev *pci; /* and an pci info */
+
+ /* Mixer vars */
+ /* fast alsa mixer */
+ struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS];
+ /* but input to much, so not used */
+ struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS];
+ /* full mixer accessable over mixer ioctl or hwdep-device */
+ struct hdspm_mixer *mixer;
+
+};
+
+/* These tables map the ALSA channels 1..N to the channels that we
+ need to use in order to find the relevant channel buffer. RME
+ refer to this kind of mapping as between "the ADAT channel and
+ the DMA channel." We index it using the logical audio channel,
+ and the value is the DMA channel (i.e. channel buffer number)
+ where the data for that channel can be read/written from/to.
+*/
+
+static char channel_map_madi_ss[HDSPM_MAX_CHANNELS] = {
+ 0, 1, 2, 3, 4, 5, 6, 7,
+ 8, 9, 10, 11, 12, 13, 14, 15,
+ 16, 17, 18, 19, 20, 21, 22, 23,
+ 24, 25, 26, 27, 28, 29, 30, 31,
+ 32, 33, 34, 35, 36, 37, 38, 39,
+ 40, 41, 42, 43, 44, 45, 46, 47,
+ 48, 49, 50, 51, 52, 53, 54, 55,
+ 56, 57, 58, 59, 60, 61, 62, 63
+};
+
+
+static struct pci_device_id snd_hdspm_ids[] __devinitdata = {
+ {
+ .vendor = PCI_VENDOR_ID_XILINX,
+ .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI,
+ .subvendor = PCI_ANY_ID,
+ .subdevice = PCI_ANY_ID,
+ .class = 0,
+ .class_mask = 0,
+ .driver_data = 0},
+ {0,}
+};
+
+MODULE_DEVICE_TABLE(pci, snd_hdspm_ids);
+
+/* prototypes */
+static int __devinit snd_hdspm_create_alsa_devices(struct snd_card *card,
+ struct hdspm * hdspm);
+static int __devinit snd_hdspm_create_pcm(struct snd_card *card,
+ struct hdspm * hdspm);
+
+static inline void snd_hdspm_initialize_midi_flush(struct hdspm * hdspm);
+static int hdspm_update_simple_mixer_controls(struct hdspm * hdspm);
+static int hdspm_autosync_ref(struct hdspm * hdspm);
+static int snd_hdspm_set_defaults(struct hdspm * hdspm);
+static void hdspm_set_sgbuf(struct hdspm * hdspm,
+ struct snd_pcm_substream *substream,
+ unsigned int reg, int channels);
+
+static inline int HDSPM_bit2freq(int n)
+{
+ static const int bit2freq_tab[] = {
+ 0, 32000, 44100, 48000, 64000, 88200,
+ 96000, 128000, 176400, 192000 };
+ if (n < 1 || n > 9)
+ return 0;
+ return bit2freq_tab[n];
+}
+
+/* Write/read to/from HDSPM with Adresses in Bytes
+ not words but only 32Bit writes are allowed */
+
+static inline void hdspm_write(struct hdspm * hdspm, unsigned int reg,
+ unsigned int val)
+{
+ writel(val, hdspm->iobase + reg);
+}
+
+static inline unsigned int hdspm_read(struct hdspm * hdspm, unsigned int reg)
+{
+ return readl(hdspm->iobase + reg);
+}
+
+/* for each output channel (chan) I have an Input (in) and Playback (pb) Fader
+ mixer is write only on hardware so we have to cache him for read
+ each fader is a u32, but uses only the first 16 bit */
+
+static inline int hdspm_read_in_gain(struct hdspm * hdspm, unsigned int chan,
+ unsigned int in)
+{
+ if (chan >= HDSPM_MIXER_CHANNELS || in >= HDSPM_MIXER_CHANNELS)
+ return 0;
+
+ return hdspm->mixer->ch[chan].in[in];
+}
+
+static inline int hdspm_read_pb_gain(struct hdspm * hdspm, unsigned int chan,
+ unsigned int pb)
+{
+ if (chan >= HDSPM_MIXER_CHANNELS || pb >= HDSPM_MIXER_CHANNELS)
+ return 0;
+ return hdspm->mixer->ch[chan].pb[pb];
+}
+
+static int hdspm_write_in_gain(struct hdspm *hdspm, unsigned int chan,
+ unsigned int in, unsigned short data)
+{
+ if (chan >= HDSPM_MIXER_CHANNELS || in >= HDSPM_MIXER_CHANNELS)
+ return -1;
+
+ hdspm_write(hdspm,
+ HDSPM_MADI_mixerBase +
+ ((in + 128 * chan) * sizeof(u32)),
+ (hdspm->mixer->ch[chan].in[in] = data & 0xFFFF));
+ return 0;
+}
+
+static int hdspm_write_pb_gain(struct hdspm *hdspm, unsigned int chan,
+ unsigned int pb, unsigned short data)
+{
+ if (chan >= HDSPM_MIXER_CHANNELS || pb >= HDSPM_MIXER_CHANNELS)
+ return -1;
+
+ hdspm_write(hdspm,
+ HDSPM_MADI_mixerBase +
+ ((64 + pb + 128 * chan) * sizeof(u32)),
+ (hdspm->mixer->ch[chan].pb[pb] = data & 0xFFFF));
+ return 0;
+}
+
+
+/* enable DMA for specific channels, now available for DSP-MADI */
+static inline void snd_hdspm_enable_in(struct hdspm * hdspm, int i, int v)
+{
+ hdspm_write(hdspm, HDSPM_inputEnableBase + (4 * i), v);
+}
+
+static inline void snd_hdspm_enable_out(struct hdspm * hdspm, int i, int v)
+{
+ hdspm_write(hdspm, HDSPM_outputEnableBase + (4 * i), v);
+}
+
+/* check if same process is writing and reading */
+static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm)
+{
+ unsigned long flags;
+ int ret = 1;
+
+ spin_lock_irqsave(&hdspm->lock, flags);
+ if ((hdspm->playback_pid != hdspm->capture_pid) &&
+ (hdspm->playback_pid >= 0) && (hdspm->capture_pid >= 0)) {
+ ret = 0;
+ }
+ spin_unlock_irqrestore(&hdspm->lock, flags);
+ return ret;
+}
+
+/* check for external sample rate */
+static int hdspm_external_sample_rate(struct hdspm *hdspm)
+{
+ if (hdspm->is_aes32) {
+ unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ unsigned int timecode =
+ hdspm_read(hdspm, HDSPM_timecodeRegister);
+
+ int syncref = hdspm_autosync_ref(hdspm);
+
+ if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD &&
+ status & HDSPM_AES32_wcLock)
+ return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit)
+ & 0xF);
+ if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 &&
+ syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 &&
+ status2 & (HDSPM_LockAES >>
+ (syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)))
+ return HDSPM_bit2freq((timecode >>
+ (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF);
+ return 0;
+ } else {
+ unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ unsigned int rate_bits;
+ int rate = 0;
+
+ /* if wordclock has synced freq and wordclock is valid */
+ if ((status2 & HDSPM_wcLock) != 0 &&
+ (status & HDSPM_SelSyncRef0) == 0) {
+
+ rate_bits = status2 & HDSPM_wcFreqMask;
+
+ switch (rate_bits) {
+ case HDSPM_wcFreq32:
+ rate = 32000;
+ break;
+ case HDSPM_wcFreq44_1:
+ rate = 44100;
+ break;
+ case HDSPM_wcFreq48:
+ rate = 48000;
+ break;
+ case HDSPM_wcFreq64:
+ rate = 64000;
+ break;
+ case HDSPM_wcFreq88_2:
+ rate = 88200;
+ break;
+ case HDSPM_wcFreq96:
+ rate = 96000;
+ break;
+ /* Quadspeed Bit missing ???? */
+ default:
+ rate = 0;
+ break;
+ }
+ }
+
+ /* if rate detected and Syncref is Word than have it,
+ * word has priority to MADI
+ */
+ if (rate != 0 &&
+ (status2 & HDSPM_SelSyncRefMask) == HDSPM_SelSyncRef_WORD)
+ return rate;
+
+ /* maby a madi input (which is taken if sel sync is madi) */
+ if (status & HDSPM_madiLock) {
+ rate_bits = status & HDSPM_madiFreqMask;
+
+ switch (rate_bits) {
+ case HDSPM_madiFreq32:
+ rate = 32000;
+ break;
+ case HDSPM_madiFreq44_1:
+ rate = 44100;
+ break;
+ case HDSPM_madiFreq48:
+ rate = 48000;
+ break;
+ case HDSPM_madiFreq64:
+ rate = 64000;
+ break;
+ case HDSPM_madiFreq88_2:
+ rate = 88200;
+ break;
+ case HDSPM_madiFreq96:
+ rate = 96000;
+ break;
+ case HDSPM_madiFreq128:
+ rate = 128000;
+ break;
+ case HDSPM_madiFreq176_4:
+ rate = 176400;
+ break;
+ case HDSPM_madiFreq192:
+ rate = 192000;
+ break;
+ default:
+ rate = 0;
+ break;
+ }
+ }
+ return rate;
+ }
+}
+
+/* Latency function */
+static inline void hdspm_compute_period_size(struct hdspm * hdspm)
+{
+ hdspm->period_bytes =
+ 1 << ((hdspm_decode_latency(hdspm->control_register) + 8));
+}
+
+static snd_pcm_uframes_t hdspm_hw_pointer(struct hdspm * hdspm)
+{
+ int position;
+
+ position = hdspm_read(hdspm, HDSPM_statusRegister);
+
+ if (!hdspm->precise_ptr)
+ return (position & HDSPM_BufferID) ?
+ (hdspm->period_bytes / 4) : 0;
+
+ /* hwpointer comes in bytes and is 64Bytes accurate (by docu since
+ PCI Burst)
+ i have experimented that it is at most 64 Byte to much for playing
+ so substraction of 64 byte should be ok for ALSA, but use it only
+ for application where you know what you do since if you come to
+ near with record pointer it can be a disaster */
+
+ position &= HDSPM_BufferPositionMask;
+ position = ((position - 64) % (2 * hdspm->period_bytes)) / 4;
+
+ return position;
+}
+
+
+static inline void hdspm_start_audio(struct hdspm * s)
+{
+ s->control_register |= (HDSPM_AudioInterruptEnable | HDSPM_Start);
+ hdspm_write(s, HDSPM_controlRegister, s->control_register);
+}
+
+static inline void hdspm_stop_audio(struct hdspm * s)
+{
+ s->control_register &= ~(HDSPM_Start | HDSPM_AudioInterruptEnable);
+ hdspm_write(s, HDSPM_controlRegister, s->control_register);
+}
+
+/* should I silence all or only opened ones ? doit all for first even is 4MB*/
+static void hdspm_silence_playback(struct hdspm *hdspm)
+{
+ int i;
+ int n = hdspm->period_bytes;
+ void *buf = hdspm->playback_buffer;
+
+ if (buf == NULL)
+ return;
+
+ for (i = 0; i < HDSPM_MAX_CHANNELS; i++) {
+ memset(buf, 0, n);
+ buf += HDSPM_CHANNEL_BUFFER_BYTES;
+ }
+}
+
+static int hdspm_set_interrupt_interval(struct hdspm * s, unsigned int frames)
+{
+ int n;
+
+ spin_lock_irq(&s->lock);
+
+ frames >>= 7;
+ n = 0;
+ while (frames) {
+ n++;
+ frames >>= 1;
+ }
+ s->control_register &= ~HDSPM_LatencyMask;
+ s->control_register |= hdspm_encode_latency(n);
+
+ hdspm_write(s, HDSPM_controlRegister, s->control_register);
+
+ hdspm_compute_period_size(s);
+
+ spin_unlock_irq(&s->lock);
+
+ return 0;
+}
+
+static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
+{
+ u64 n;
+ u32 r;
+
+ if (rate >= 112000)
+ rate /= 4;
+ else if (rate >= 56000)
+ rate /= 2;
+
+ /* RME says n = 104857600000000, but in the windows MADI driver, I see:
+// return 104857600000000 / rate; // 100 MHz
+ return 110100480000000 / rate; // 105 MHz
+ */
+ /* n = 104857600000000ULL; */ /* = 2^20 * 10^8 */
+ n = 110100480000000ULL; /* Value checked for AES32 and MADI */
+ div64_32(&n, rate, &r);
+ /* n should be less than 2^32 for being written to FREQ register */
+ snd_BUG_ON(n >> 32);
+ hdspm_write(hdspm, HDSPM_freqReg, (u32)n);
+}
+
+/* dummy set rate lets see what happens */
+static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally)
+{
+ int current_rate;
+ int rate_bits;
+ int not_set = 0;
+ int current_speed, target_speed;
+
+ /* ASSUMPTION: hdspm->lock is either set, or there is no need for
+ it (e.g. during module initialization).
+ */
+
+ if (!(hdspm->control_register & HDSPM_ClockModeMaster)) {
+
+ /* SLAVE --- */
+ if (called_internally) {
+
+ /* request from ctl or card initialization
+ just make a warning an remember setting
+ for future master mode switching */
+
+ snd_printk(KERN_WARNING "HDSPM: "
+ "Warning: device is not running "
+ "as a clock master.\n");
+ not_set = 1;
+ } else {
+
+ /* hw_param request while in AutoSync mode */
+ int external_freq =
+ hdspm_external_sample_rate(hdspm);
+
+ if (hdspm_autosync_ref(hdspm) ==
+ HDSPM_AUTOSYNC_FROM_NONE) {
+
+ snd_printk(KERN_WARNING "HDSPM: "
+ "Detected no Externel Sync \n");
+ not_set = 1;
+
+ } else if (rate != external_freq) {
+
+ snd_printk(KERN_WARNING "HDSPM: "
+ "Warning: No AutoSync source for "
+ "requested rate\n");
+ not_set = 1;
+ }
+ }
+ }
+
+ current_rate = hdspm->system_sample_rate;
+
+ /* Changing between Singe, Double and Quad speed is not
+ allowed if any substreams are open. This is because such a change
+ causes a shift in the location of the DMA buffers and a reduction
+ in the number of available buffers.
+
+ Note that a similar but essentially insoluble problem exists for
+ externally-driven rate changes. All we can do is to flag rate
+ changes in the read/write routines.
+ */
+
+ if (current_rate <= 48000)
+ current_speed = HDSPM_SPEED_SINGLE;
+ else if (current_rate <= 96000)
+ current_speed = HDSPM_SPEED_DOUBLE;
+ else
+ current_speed = HDSPM_SPEED_QUAD;
+
+ if (rate <= 48000)
+ target_speed = HDSPM_SPEED_SINGLE;
+ else if (rate <= 96000)
+ target_speed = HDSPM_SPEED_DOUBLE;
+ else
+ target_speed = HDSPM_SPEED_QUAD;
+
+ switch (rate) {
+ case 32000:
+ rate_bits = HDSPM_Frequency32KHz;
+ break;
+ case 44100:
+ rate_bits = HDSPM_Frequency44_1KHz;
+ break;
+ case 48000:
+ rate_bits = HDSPM_Frequency48KHz;
+ break;
+ case 64000:
+ rate_bits = HDSPM_Frequency64KHz;
+ break;
+ case 88200:
+ rate_bits = HDSPM_Frequency88_2KHz;
+ break;
+ case 96000:
+ rate_bits = HDSPM_Frequency96KHz;
+ break;
+ case 128000:
+ rate_bits = HDSPM_Frequency128KHz;
+ break;
+ case 176400:
+ rate_bits = HDSPM_Frequency176_4KHz;
+ break;
+ case 192000:
+ rate_bits = HDSPM_Frequency192KHz;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (current_speed != target_speed
+ && (hdspm->capture_pid >= 0 || hdspm->playback_pid >= 0)) {
+ snd_printk
+ (KERN_ERR "HDSPM: "
+ "cannot change from %s speed to %s speed mode "
+ "(capture PID = %d, playback PID = %d)\n",
+ hdspm_speed_names[current_speed],
+ hdspm_speed_names[target_speed],
+ hdspm->capture_pid, hdspm->playback_pid);
+ return -EBUSY;
+ }
+
+ hdspm->control_register &= ~HDSPM_FrequencyMask;
+ hdspm->control_register |= rate_bits;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ /* For AES32, need to set DDS value in FREQ register
+ For MADI, also apparently */
+ hdspm_set_dds_value(hdspm, rate);
+
+ if (hdspm->is_aes32 && rate != current_rate)
+ hdspm_write(hdspm, HDSPM_eeprom_wr, 0);
+
+ /* For AES32 and for MADI (at least rev 204), channel_map needs to
+ * always be channel_map_madi_ss, whatever the sample rate */
+ hdspm->channel_map = channel_map_madi_ss;
+
+ hdspm->system_sample_rate = rate;
+
+ if (not_set != 0)
+ return -1;
+
+ return 0;
+}
+
+/* mainly for init to 0 on load */
+static void all_in_all_mixer(struct hdspm * hdspm, int sgain)
+{
+ int i, j;
+ unsigned int gain;
+
+ if (sgain > UNITY_GAIN)
+ gain = UNITY_GAIN;
+ else if (sgain < 0)
+ gain = 0;
+ else
+ gain = sgain;
+
+ for (i = 0; i < HDSPM_MIXER_CHANNELS; i++)
+ for (j = 0; j < HDSPM_MIXER_CHANNELS; j++) {
+ hdspm_write_in_gain(hdspm, i, j, gain);
+ hdspm_write_pb_gain(hdspm, i, j, gain);
+ }
+}
+
+/*----------------------------------------------------------------------------
+ MIDI
+ ----------------------------------------------------------------------------*/
+
+static inline unsigned char snd_hdspm_midi_read_byte (struct hdspm *hdspm,
+ int id)
+{
+ /* the hardware already does the relevant bit-mask with 0xff */
+ if (id)
+ return hdspm_read(hdspm, HDSPM_midiDataIn1);
+ else
+ return hdspm_read(hdspm, HDSPM_midiDataIn0);
+}
+
+static inline void snd_hdspm_midi_write_byte (struct hdspm *hdspm, int id,
+ int val)
+{
+ /* the hardware already does the relevant bit-mask with 0xff */
+ if (id)
+ hdspm_write(hdspm, HDSPM_midiDataOut1, val);
+ else
+ hdspm_write(hdspm, HDSPM_midiDataOut0, val);
+}
+
+static inline int snd_hdspm_midi_input_available (struct hdspm *hdspm, int id)
+{
+ if (id)
+ return (hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xff);
+ else
+ return (hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xff);
+}
+
+static inline int snd_hdspm_midi_output_possible (struct hdspm *hdspm, int id)
+{
+ int fifo_bytes_used;
+
+ if (id)
+ fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut1);
+ else
+ fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut0);
+ fifo_bytes_used &= 0xff;
+
+ if (fifo_bytes_used < 128)
+ return 128 - fifo_bytes_used;
+ else
+ return 0;
+}
+
+static void snd_hdspm_flush_midi_input(struct hdspm *hdspm, int id)
+{
+ while (snd_hdspm_midi_input_available (hdspm, id))
+ snd_hdspm_midi_read_byte (hdspm, id);
+}
+
+static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi)
+{
+ unsigned long flags;
+ int n_pending;
+ int to_write;
+ int i;
+ unsigned char buf[128];
+
+ /* Output is not interrupt driven */
+
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if (hmidi->output &&
+ !snd_rawmidi_transmit_empty (hmidi->output)) {
+ n_pending = snd_hdspm_midi_output_possible (hmidi->hdspm,
+ hmidi->id);
+ if (n_pending > 0) {
+ if (n_pending > (int)sizeof (buf))
+ n_pending = sizeof (buf);
+
+ to_write = snd_rawmidi_transmit (hmidi->output, buf,
+ n_pending);
+ if (to_write > 0) {
+ for (i = 0; i < to_write; ++i)
+ snd_hdspm_midi_write_byte (hmidi->hdspm,
+ hmidi->id,
+ buf[i]);
+ }
+ }
+ }
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ return 0;
+}
+
+static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi)
+{
+ unsigned char buf[128]; /* this buffer is designed to match the MIDI
+ * input FIFO size
+ */
+ unsigned long flags;
+ int n_pending;
+ int i;
+
+ spin_lock_irqsave (&hmidi->lock, flags);
+ n_pending = snd_hdspm_midi_input_available (hmidi->hdspm, hmidi->id);
+ if (n_pending > 0) {
+ if (hmidi->input) {
+ if (n_pending > (int)sizeof (buf))
+ n_pending = sizeof (buf);
+ for (i = 0; i < n_pending; ++i)
+ buf[i] = snd_hdspm_midi_read_byte (hmidi->hdspm,
+ hmidi->id);
+ if (n_pending)
+ snd_rawmidi_receive (hmidi->input, buf,
+ n_pending);
+ } else {
+ /* flush the MIDI input FIFO */
+ while (n_pending--)
+ snd_hdspm_midi_read_byte (hmidi->hdspm,
+ hmidi->id);
+ }
+ }
+ hmidi->pending = 0;
+ if (hmidi->id)
+ hmidi->hdspm->control_register |= HDSPM_Midi1InterruptEnable;
+ else
+ hmidi->hdspm->control_register |= HDSPM_Midi0InterruptEnable;
+ hdspm_write(hmidi->hdspm, HDSPM_controlRegister,
+ hmidi->hdspm->control_register);
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ return snd_hdspm_midi_output_write (hmidi);
+}
+
+static void
+snd_hdspm_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct hdspm *hdspm;
+ struct hdspm_midi *hmidi;
+ unsigned long flags;
+ u32 ie;
+
+ hmidi = substream->rmidi->private_data;
+ hdspm = hmidi->hdspm;
+ ie = hmidi->id ?
+ HDSPM_Midi1InterruptEnable : HDSPM_Midi0InterruptEnable;
+ spin_lock_irqsave (&hdspm->lock, flags);
+ if (up) {
+ if (!(hdspm->control_register & ie)) {
+ snd_hdspm_flush_midi_input (hdspm, hmidi->id);
+ hdspm->control_register |= ie;
+ }
+ } else {
+ hdspm->control_register &= ~ie;
+ }
+
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ spin_unlock_irqrestore (&hdspm->lock, flags);
+}
+
+static void snd_hdspm_midi_output_timer(unsigned long data)
+{
+ struct hdspm_midi *hmidi = (struct hdspm_midi *) data;
+ unsigned long flags;
+
+ snd_hdspm_midi_output_write(hmidi);
+ spin_lock_irqsave (&hmidi->lock, flags);
+
+ /* this does not bump hmidi->istimer, because the
+ kernel automatically removed the timer when it
+ expired, and we are now adding it back, thus
+ leaving istimer wherever it was set before.
+ */
+
+ if (hmidi->istimer) {
+ hmidi->timer.expires = 1 + jiffies;
+ add_timer(&hmidi->timer);
+ }
+
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+}
+
+static void
+snd_hdspm_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct hdspm_midi *hmidi;
+ unsigned long flags;
+
+ hmidi = substream->rmidi->private_data;
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if (up) {
+ if (!hmidi->istimer) {
+ init_timer(&hmidi->timer);
+ hmidi->timer.function = snd_hdspm_midi_output_timer;
+ hmidi->timer.data = (unsigned long) hmidi;
+ hmidi->timer.expires = 1 + jiffies;
+ add_timer(&hmidi->timer);
+ hmidi->istimer++;
+ }
+ } else {
+ if (hmidi->istimer && --hmidi->istimer <= 0)
+ del_timer (&hmidi->timer);
+ }
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ if (up)
+ snd_hdspm_midi_output_write(hmidi);
+}
+
+static int snd_hdspm_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ struct hdspm_midi *hmidi;
+
+ hmidi = substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ snd_hdspm_flush_midi_input (hmidi->hdspm, hmidi->id);
+ hmidi->input = substream;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct hdspm_midi *hmidi;
+
+ hmidi = substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->output = substream;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ struct hdspm_midi *hmidi;
+
+ snd_hdspm_midi_input_trigger (substream, 0);
+
+ hmidi = substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->input = NULL;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct hdspm_midi *hmidi;
+
+ snd_hdspm_midi_output_trigger (substream, 0);
+
+ hmidi = substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->output = NULL;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static struct snd_rawmidi_ops snd_hdspm_midi_output =
+{
+ .open = snd_hdspm_midi_output_open,
+ .close = snd_hdspm_midi_output_close,
+ .trigger = snd_hdspm_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_hdspm_midi_input =
+{
+ .open = snd_hdspm_midi_input_open,
+ .close = snd_hdspm_midi_input_close,
+ .trigger = snd_hdspm_midi_input_trigger,
+};
+
+static int __devinit snd_hdspm_create_midi (struct snd_card *card,
+ struct hdspm *hdspm, int id)
+{
+ int err;
+ char buf[32];
+
+ hdspm->midi[id].id = id;
+ hdspm->midi[id].hdspm = hdspm;
+ spin_lock_init (&hdspm->midi[id].lock);
+
+ sprintf (buf, "%s MIDI %d", card->shortname, id+1);
+ err = snd_rawmidi_new (card, buf, id, 1, 1, &hdspm->midi[id].rmidi);
+ if (err < 0)
+ return err;
+
+ sprintf (hdspm->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1);
+ hdspm->midi[id].rmidi->private_data = &hdspm->midi[id];
+
+ snd_rawmidi_set_ops(hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_hdspm_midi_output);
+ snd_rawmidi_set_ops(hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_hdspm_midi_input);
+
+ hdspm->midi[id].rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return 0;
+}
+
+
+static void hdspm_midi_tasklet(unsigned long arg)
+{
+ struct hdspm *hdspm = (struct hdspm *)arg;
+
+ if (hdspm->midi[0].pending)
+ snd_hdspm_midi_input_read (&hdspm->midi[0]);
+ if (hdspm->midi[1].pending)
+ snd_hdspm_midi_input_read (&hdspm->midi[1]);
+}
+
+
+/*-----------------------------------------------------------------------------
+ Status Interface
+ ----------------------------------------------------------------------------*/
+
+/* get the system sample rate which is set */
+
+#define HDSPM_SYSTEM_SAMPLE_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_system_sample_rate, \
+ .get = snd_hdspm_get_system_sample_rate \
+}
+
+static int snd_hdspm_info_system_sample_rate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_hdspm_get_system_sample_rate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *
+ ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm->system_sample_rate;
+ return 0;
+}
+
+#define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_autosync_sample_rate, \
+ .get = snd_hdspm_get_autosync_sample_rate \
+}
+
+static int snd_hdspm_info_autosync_sample_rate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "32000", "44100", "48000",
+ "64000", "88200", "96000",
+ "128000", "176400", "192000",
+ "None"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 10;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *
+ ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ switch (hdspm_external_sample_rate(hdspm)) {
+ case 32000:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ case 44100:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ case 48000:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case 64000:
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case 88200:
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ case 96000:
+ ucontrol->value.enumerated.item[0] = 5;
+ break;
+ case 128000:
+ ucontrol->value.enumerated.item[0] = 6;
+ break;
+ case 176400:
+ ucontrol->value.enumerated.item[0] = 7;
+ break;
+ case 192000:
+ ucontrol->value.enumerated.item[0] = 8;
+ break;
+
+ default:
+ ucontrol->value.enumerated.item[0] = 9;
+ }
+ return 0;
+}
+
+#define HDSPM_SYSTEM_CLOCK_MODE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_system_clock_mode, \
+ .get = snd_hdspm_get_system_clock_mode, \
+}
+
+
+
+static int hdspm_system_clock_mode(struct hdspm * hdspm)
+{
+ /* Always reflect the hardware info, rme is never wrong !!!! */
+
+ if (hdspm->control_register & HDSPM_ClockModeMaster)
+ return 0;
+ return 1;
+}
+
+static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "Master", "Slave" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_hdspm_get_system_clock_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] =
+ hdspm_system_clock_mode(hdspm);
+ return 0;
+}
+
+#define HDSPM_CLOCK_SOURCE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_clock_source, \
+ .get = snd_hdspm_get_clock_source, \
+ .put = snd_hdspm_put_clock_source \
+}
+
+static int hdspm_clock_source(struct hdspm * hdspm)
+{
+ if (hdspm->control_register & HDSPM_ClockModeMaster) {
+ switch (hdspm->system_sample_rate) {
+ case 32000:
+ return 1;
+ case 44100:
+ return 2;
+ case 48000:
+ return 3;
+ case 64000:
+ return 4;
+ case 88200:
+ return 5;
+ case 96000:
+ return 6;
+ case 128000:
+ return 7;
+ case 176400:
+ return 8;
+ case 192000:
+ return 9;
+ default:
+ return 3;
+ }
+ } else {
+ return 0;
+ }
+}
+
+static int hdspm_set_clock_source(struct hdspm * hdspm, int mode)
+{
+ int rate;
+ switch (mode) {
+
+ case HDSPM_CLOCK_SOURCE_AUTOSYNC:
+ if (hdspm_external_sample_rate(hdspm) != 0) {
+ hdspm->control_register &= ~HDSPM_ClockModeMaster;
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ return 0;
+ }
+ return -1;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_32KHZ:
+ rate = 32000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_44_1KHZ:
+ rate = 44100;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_48KHZ:
+ rate = 48000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_64KHZ:
+ rate = 64000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_88_2KHZ:
+ rate = 88200;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_96KHZ:
+ rate = 96000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_128KHZ:
+ rate = 128000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_176_4KHZ:
+ rate = 176400;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_192KHZ:
+ rate = 192000;
+ break;
+
+ default:
+ rate = 44100;
+ }
+ hdspm->control_register |= HDSPM_ClockModeMaster;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ hdspm_set_rate(hdspm, rate, 1);
+ return 0;
+}
+
+static int snd_hdspm_info_clock_source(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "AutoSync",
+ "Internal 32.0 kHz", "Internal 44.1 kHz",
+ "Internal 48.0 kHz",
+ "Internal 64.0 kHz", "Internal 88.2 kHz",
+ "Internal 96.0 kHz",
+ "Internal 128.0 kHz", "Internal 176.4 kHz",
+ "Internal 192.0 kHz"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 10;
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int snd_hdspm_get_clock_source(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_clock_source(hdspm);
+ return 0;
+}
+
+static int snd_hdspm_put_clock_source(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0];
+ if (val < 0)
+ val = 0;
+ if (val > 9)
+ val = 9;
+ spin_lock_irq(&hdspm->lock);
+ if (val != hdspm_clock_source(hdspm))
+ change = (hdspm_set_clock_source(hdspm, val) == 0) ? 1 : 0;
+ else
+ change = 0;
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_PREF_SYNC_REF(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_pref_sync_ref, \
+ .get = snd_hdspm_get_pref_sync_ref, \
+ .put = snd_hdspm_put_pref_sync_ref \
+}
+
+static int hdspm_pref_sync_ref(struct hdspm * hdspm)
+{
+ /* Notice that this looks at the requested sync source,
+ not the one actually in use.
+ */
+ if (hdspm->is_aes32) {
+ switch (hdspm->control_register & HDSPM_SyncRefMask) {
+ /* number gives AES index, except for 0 which
+ corresponds to WordClock */
+ case 0: return 0;
+ case HDSPM_SyncRef0: return 1;
+ case HDSPM_SyncRef1: return 2;
+ case HDSPM_SyncRef1+HDSPM_SyncRef0: return 3;
+ case HDSPM_SyncRef2: return 4;
+ case HDSPM_SyncRef2+HDSPM_SyncRef0: return 5;
+ case HDSPM_SyncRef2+HDSPM_SyncRef1: return 6;
+ case HDSPM_SyncRef2+HDSPM_SyncRef1+HDSPM_SyncRef0: return 7;
+ case HDSPM_SyncRef3: return 8;
+ }
+ } else {
+ switch (hdspm->control_register & HDSPM_SyncRefMask) {
+ case HDSPM_SyncRef_Word:
+ return HDSPM_SYNC_FROM_WORD;
+ case HDSPM_SyncRef_MADI:
+ return HDSPM_SYNC_FROM_MADI;
+ }
+ }
+
+ return HDSPM_SYNC_FROM_WORD;
+}
+
+static int hdspm_set_pref_sync_ref(struct hdspm * hdspm, int pref)
+{
+ hdspm->control_register &= ~HDSPM_SyncRefMask;
+
+ if (hdspm->is_aes32) {
+ switch (pref) {
+ case 0:
+ hdspm->control_register |= 0;
+ break;
+ case 1:
+ hdspm->control_register |= HDSPM_SyncRef0;
+ break;
+ case 2:
+ hdspm->control_register |= HDSPM_SyncRef1;
+ break;
+ case 3:
+ hdspm->control_register |= HDSPM_SyncRef1+HDSPM_SyncRef0;
+ break;
+ case 4:
+ hdspm->control_register |= HDSPM_SyncRef2;
+ break;
+ case 5:
+ hdspm->control_register |= HDSPM_SyncRef2+HDSPM_SyncRef0;
+ break;
+ case 6:
+ hdspm->control_register |= HDSPM_SyncRef2+HDSPM_SyncRef1;
+ break;
+ case 7:
+ hdspm->control_register |=
+ HDSPM_SyncRef2+HDSPM_SyncRef1+HDSPM_SyncRef0;
+ break;
+ case 8:
+ hdspm->control_register |= HDSPM_SyncRef3;
+ break;
+ default:
+ return -1;
+ }
+ } else {
+ switch (pref) {
+ case HDSPM_SYNC_FROM_MADI:
+ hdspm->control_register |= HDSPM_SyncRef_MADI;
+ break;
+ case HDSPM_SYNC_FROM_WORD:
+ hdspm->control_register |= HDSPM_SyncRef_Word;
+ break;
+ default:
+ return -1;
+ }
+ }
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ return 0;
+}
+
+static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ if (hdspm->is_aes32) {
+ static char *texts[] = { "Word", "AES1", "AES2", "AES3",
+ "AES4", "AES5", "AES6", "AES7", "AES8" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+
+ uinfo->value.enumerated.items = 9;
+
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ } else {
+ static char *texts[] = { "Word", "MADI" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+
+ uinfo->value.enumerated.items = 2;
+
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_pref_sync_ref(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_pref_sync_ref(hdspm);
+ return 0;
+}
+
+static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change, max;
+ unsigned int val;
+
+ max = hdspm->is_aes32 ? 9 : 2;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+
+ val = ucontrol->value.enumerated.item[0] % max;
+
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_pref_sync_ref(hdspm);
+ hdspm_set_pref_sync_ref(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_AUTOSYNC_REF(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_autosync_ref, \
+ .get = snd_hdspm_get_autosync_ref, \
+}
+
+static int hdspm_autosync_ref(struct hdspm * hdspm)
+{
+ if (hdspm->is_aes32) {
+ unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) &
+ 0xF;
+ if (syncref == 0)
+ return HDSPM_AES32_AUTOSYNC_FROM_WORD;
+ if (syncref <= 8)
+ return syncref;
+ return HDSPM_AES32_AUTOSYNC_FROM_NONE;
+ } else {
+ /* This looks at the autosync selected sync reference */
+ unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+
+ switch (status2 & HDSPM_SelSyncRefMask) {
+ case HDSPM_SelSyncRef_WORD:
+ return HDSPM_AUTOSYNC_FROM_WORD;
+ case HDSPM_SelSyncRef_MADI:
+ return HDSPM_AUTOSYNC_FROM_MADI;
+ case HDSPM_SelSyncRef_NVALID:
+ return HDSPM_AUTOSYNC_FROM_NONE;
+ default:
+ return 0;
+ }
+
+ return 0;
+ }
+}
+
+static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ if (hdspm->is_aes32) {
+ static char *texts[] = { "WordClock", "AES1", "AES2", "AES3",
+ "AES4", "AES5", "AES6", "AES7", "AES8", "None"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 10;
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ } else {
+ static char *texts[] = { "WordClock", "MADI", "None" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_autosync_ref(hdspm);
+ return 0;
+}
+
+#define HDSPM_LINE_OUT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_line_out, \
+ .get = snd_hdspm_get_line_out, \
+ .put = snd_hdspm_put_line_out \
+}
+
+static int hdspm_line_out(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_LineOut) ? 1 : 0;
+}
+
+
+static int hdspm_set_line_output(struct hdspm * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_LineOut;
+ else
+ hdspm->control_register &= ~HDSPM_LineOut;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+#define snd_hdspm_info_line_out snd_ctl_boolean_mono_info
+
+static int snd_hdspm_get_line_out(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_line_out(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_line_out(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_line_out(hdspm);
+ hdspm_set_line_output(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_TX_64(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_tx_64, \
+ .get = snd_hdspm_get_tx_64, \
+ .put = snd_hdspm_put_tx_64 \
+}
+
+static int hdspm_tx_64(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_TX_64ch) ? 1 : 0;
+}
+
+static int hdspm_set_tx_64(struct hdspm * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_TX_64ch;
+ else
+ hdspm->control_register &= ~HDSPM_TX_64ch;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+#define snd_hdspm_info_tx_64 snd_ctl_boolean_mono_info
+
+static int snd_hdspm_get_tx_64(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_tx_64(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_tx_64(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_tx_64(hdspm);
+ hdspm_set_tx_64(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_C_TMS(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_c_tms, \
+ .get = snd_hdspm_get_c_tms, \
+ .put = snd_hdspm_put_c_tms \
+}
+
+static int hdspm_c_tms(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_clr_tms) ? 1 : 0;
+}
+
+static int hdspm_set_c_tms(struct hdspm * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_clr_tms;
+ else
+ hdspm->control_register &= ~HDSPM_clr_tms;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+#define snd_hdspm_info_c_tms snd_ctl_boolean_mono_info
+
+static int snd_hdspm_get_c_tms(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_c_tms(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_c_tms(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_c_tms(hdspm);
+ hdspm_set_c_tms(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_SAFE_MODE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_safe_mode, \
+ .get = snd_hdspm_get_safe_mode, \
+ .put = snd_hdspm_put_safe_mode \
+}
+
+static int hdspm_safe_mode(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_AutoInp) ? 1 : 0;
+}
+
+static int hdspm_set_safe_mode(struct hdspm * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_AutoInp;
+ else
+ hdspm->control_register &= ~HDSPM_AutoInp;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+#define snd_hdspm_info_safe_mode snd_ctl_boolean_mono_info
+
+static int snd_hdspm_get_safe_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_safe_mode(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_safe_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_safe_mode(hdspm);
+ hdspm_set_safe_mode(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_EMPHASIS(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_emphasis, \
+ .get = snd_hdspm_get_emphasis, \
+ .put = snd_hdspm_put_emphasis \
+}
+
+static int hdspm_emphasis(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_Emphasis) ? 1 : 0;
+}
+
+static int hdspm_set_emphasis(struct hdspm * hdspm, int emp)
+{
+ if (emp)
+ hdspm->control_register |= HDSPM_Emphasis;
+ else
+ hdspm->control_register &= ~HDSPM_Emphasis;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+#define snd_hdspm_info_emphasis snd_ctl_boolean_mono_info
+
+static int snd_hdspm_get_emphasis(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_emphasis(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_emphasis(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_emphasis(hdspm);
+ hdspm_set_emphasis(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_DOLBY(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_dolby, \
+ .get = snd_hdspm_get_dolby, \
+ .put = snd_hdspm_put_dolby \
+}
+
+static int hdspm_dolby(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_Dolby) ? 1 : 0;
+}
+
+static int hdspm_set_dolby(struct hdspm * hdspm, int dol)
+{
+ if (dol)
+ hdspm->control_register |= HDSPM_Dolby;
+ else
+ hdspm->control_register &= ~HDSPM_Dolby;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+#define snd_hdspm_info_dolby snd_ctl_boolean_mono_info
+
+static int snd_hdspm_get_dolby(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_dolby(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_dolby(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_dolby(hdspm);
+ hdspm_set_dolby(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_PROFESSIONAL(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_professional, \
+ .get = snd_hdspm_get_professional, \
+ .put = snd_hdspm_put_professional \
+}
+
+static int hdspm_professional(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_Professional) ? 1 : 0;
+}
+
+static int hdspm_set_professional(struct hdspm * hdspm, int dol)
+{
+ if (dol)
+ hdspm->control_register |= HDSPM_Professional;
+ else
+ hdspm->control_register &= ~HDSPM_Professional;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+#define snd_hdspm_info_professional snd_ctl_boolean_mono_info
+
+static int snd_hdspm_get_professional(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_professional(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_professional(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_professional(hdspm);
+ hdspm_set_professional(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_INPUT_SELECT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_input_select, \
+ .get = snd_hdspm_get_input_select, \
+ .put = snd_hdspm_put_input_select \
+}
+
+static int hdspm_input_select(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_InputSelect0) ? 1 : 0;
+}
+
+static int hdspm_set_input_select(struct hdspm * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_InputSelect0;
+ else
+ hdspm->control_register &= ~HDSPM_InputSelect0;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "optical", "coaxial" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int snd_hdspm_get_input_select(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_input_select(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_input_select(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_input_select(hdspm);
+ hdspm_set_input_select(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_DS_WIRE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_ds_wire, \
+ .get = snd_hdspm_get_ds_wire, \
+ .put = snd_hdspm_put_ds_wire \
+}
+
+static int hdspm_ds_wire(struct hdspm * hdspm)
+{
+ return (hdspm->control_register & HDSPM_DS_DoubleWire) ? 1 : 0;
+}
+
+static int hdspm_set_ds_wire(struct hdspm * hdspm, int ds)
+{
+ if (ds)
+ hdspm->control_register |= HDSPM_DS_DoubleWire;
+ else
+ hdspm->control_register &= ~HDSPM_DS_DoubleWire;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "Single", "Double" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int snd_hdspm_get_ds_wire(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_ds_wire(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_ds_wire(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_ds_wire(hdspm);
+ hdspm_set_ds_wire(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_QS_WIRE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_qs_wire, \
+ .get = snd_hdspm_get_qs_wire, \
+ .put = snd_hdspm_put_qs_wire \
+}
+
+static int hdspm_qs_wire(struct hdspm * hdspm)
+{
+ if (hdspm->control_register & HDSPM_QS_DoubleWire)
+ return 1;
+ if (hdspm->control_register & HDSPM_QS_QuadWire)
+ return 2;
+ return 0;
+}
+
+static int hdspm_set_qs_wire(struct hdspm * hdspm, int mode)
+{
+ hdspm->control_register &= ~(HDSPM_QS_DoubleWire | HDSPM_QS_QuadWire);
+ switch (mode) {
+ case 0:
+ break;
+ case 1:
+ hdspm->control_register |= HDSPM_QS_DoubleWire;
+ break;
+ case 2:
+ hdspm->control_register |= HDSPM_QS_QuadWire;
+ break;
+ }
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "Single", "Double", "Quad" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int snd_hdspm_get_qs_wire(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_qs_wire(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0];
+ if (val < 0)
+ val = 0;
+ if (val > 2)
+ val = 2;
+ spin_lock_irq(&hdspm->lock);
+ change = val != hdspm_qs_wire(hdspm);
+ hdspm_set_qs_wire(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+/* Simple Mixer
+ deprecated since to much faders ???
+ MIXER interface says output (source, destination, value)
+ where source > MAX_channels are playback channels
+ on MADICARD
+ - playback mixer matrix: [channelout+64] [output] [value]
+ - input(thru) mixer matrix: [channelin] [output] [value]
+ (better do 2 kontrols for seperation ?)
+*/
+
+#define HDSPM_MIXER(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .device = 0, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_mixer, \
+ .get = snd_hdspm_get_mixer, \
+ .put = snd_hdspm_put_mixer \
+}
+
+static int snd_hdspm_info_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 65535;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+static int snd_hdspm_get_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int source;
+ int destination;
+
+ source = ucontrol->value.integer.value[0];
+ if (source < 0)
+ source = 0;
+ else if (source >= 2 * HDSPM_MAX_CHANNELS)
+ source = 2 * HDSPM_MAX_CHANNELS - 1;
+
+ destination = ucontrol->value.integer.value[1];
+ if (destination < 0)
+ destination = 0;
+ else if (destination >= HDSPM_MAX_CHANNELS)
+ destination = HDSPM_MAX_CHANNELS - 1;
+
+ spin_lock_irq(&hdspm->lock);
+ if (source >= HDSPM_MAX_CHANNELS)
+ ucontrol->value.integer.value[2] =
+ hdspm_read_pb_gain(hdspm, destination,
+ source - HDSPM_MAX_CHANNELS);
+ else
+ ucontrol->value.integer.value[2] =
+ hdspm_read_in_gain(hdspm, destination, source);
+
+ spin_unlock_irq(&hdspm->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_put_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ int source;
+ int destination;
+ int gain;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+
+ source = ucontrol->value.integer.value[0];
+ destination = ucontrol->value.integer.value[1];
+
+ if (source < 0 || source >= 2 * HDSPM_MAX_CHANNELS)
+ return -1;
+ if (destination < 0 || destination >= HDSPM_MAX_CHANNELS)
+ return -1;
+
+ gain = ucontrol->value.integer.value[2];
+
+ spin_lock_irq(&hdspm->lock);
+
+ if (source >= HDSPM_MAX_CHANNELS)
+ change = gain != hdspm_read_pb_gain(hdspm, destination,
+ source -
+ HDSPM_MAX_CHANNELS);
+ else
+ change = gain != hdspm_read_in_gain(hdspm, destination,
+ source);
+
+ if (change) {
+ if (source >= HDSPM_MAX_CHANNELS)
+ hdspm_write_pb_gain(hdspm, destination,
+ source - HDSPM_MAX_CHANNELS,
+ gain);
+ else
+ hdspm_write_in_gain(hdspm, destination, source,
+ gain);
+ }
+ spin_unlock_irq(&hdspm->lock);
+
+ return change;
+}
+
+/* The simple mixer control(s) provide gain control for the
+ basic 1:1 mappings of playback streams to output
+ streams.
+*/
+
+#define HDSPM_PLAYBACK_MIXER \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | \
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_playback_mixer, \
+ .get = snd_hdspm_get_playback_mixer, \
+ .put = snd_hdspm_put_playback_mixer \
+}
+
+static int snd_hdspm_info_playback_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 65536;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+static int snd_hdspm_get_playback_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int channel;
+ int mapped_channel;
+
+ channel = ucontrol->id.index - 1;
+
+ if (snd_BUG_ON(channel < 0 || channel >= HDSPM_MAX_CHANNELS))
+ return -EINVAL;
+
+ mapped_channel = hdspm->channel_map[channel];
+ if (mapped_channel < 0)
+ return -EINVAL;
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] =
+ hdspm_read_pb_gain(hdspm, mapped_channel, mapped_channel);
+ spin_unlock_irq(&hdspm->lock);
+
+ /*
+ snd_printdd("get pb mixer index %d, channel %d, mapped_channel %d, "
+ "value %d\n",
+ ucontrol->id.index, channel, mapped_channel,
+ ucontrol->value.integer.value[0]);
+ */
+ return 0;
+}
+
+static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ int channel;
+ int mapped_channel;
+ int gain;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+
+ channel = ucontrol->id.index - 1;
+
+ if (snd_BUG_ON(channel < 0 || channel >= HDSPM_MAX_CHANNELS))
+ return -EINVAL;
+
+ mapped_channel = hdspm->channel_map[channel];
+ if (mapped_channel < 0)
+ return -EINVAL;
+
+ gain = ucontrol->value.integer.value[0];
+
+ spin_lock_irq(&hdspm->lock);
+ change =
+ gain != hdspm_read_pb_gain(hdspm, mapped_channel,
+ mapped_channel);
+ if (change)
+ hdspm_write_pb_gain(hdspm, mapped_channel, mapped_channel,
+ gain);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_WC_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_sync_check, \
+ .get = snd_hdspm_get_wc_sync_check \
+}
+
+static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "No Lock", "Lock", "Sync" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int hdspm_wc_sync_check(struct hdspm * hdspm)
+{
+ if (hdspm->is_aes32) {
+ int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ if (status & HDSPM_AES32_wcLock) {
+ /* I don't know how to differenciate sync from lock.
+ Doing as if sync for now */
+ return 2;
+ }
+ return 0;
+ } else {
+ int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ if (status2 & HDSPM_wcLock) {
+ if (status2 & HDSPM_wcSync)
+ return 2;
+ else
+ return 1;
+ }
+ return 0;
+ }
+}
+
+static int snd_hdspm_get_wc_sync_check(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_wc_sync_check(hdspm);
+ return 0;
+}
+
+
+#define HDSPM_MADI_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_sync_check, \
+ .get = snd_hdspm_get_madisync_sync_check \
+}
+
+static int hdspm_madisync_sync_check(struct hdspm * hdspm)
+{
+ int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ if (status & HDSPM_madiLock) {
+ if (status & HDSPM_madiSync)
+ return 2;
+ else
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_madisync_sync_check(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *
+ ucontrol)
+{
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] =
+ hdspm_madisync_sync_check(hdspm);
+ return 0;
+}
+
+
+#define HDSPM_AES_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_sync_check, \
+ .get = snd_hdspm_get_aes_sync_check \
+}
+
+static int hdspm_aes_sync_check(struct hdspm * hdspm, int idx)
+{
+ int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ if (status2 & (HDSPM_LockAES >> idx)) {
+ /* I don't know how to differenciate sync from lock.
+ Doing as if sync for now */
+ return 2;
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_aes_sync_check(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int offset;
+ struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+
+ offset = ucontrol->id.index - 1;
+ if (offset < 0 || offset >= 8)
+ return -EINVAL;
+
+ ucontrol->value.enumerated.item[0] =
+ hdspm_aes_sync_check(hdspm, offset);
+ return 0;
+}
+
+
+static struct snd_kcontrol_new snd_hdspm_controls_madi[] = {
+
+ HDSPM_MIXER("Mixer", 0),
+/* 'Sample Clock Source' complies with the alsa control naming scheme */
+ HDSPM_CLOCK_SOURCE("Sample Clock Source", 0),
+
+ HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
+ HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
+ HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
+ HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
+/* 'External Rate' complies with the alsa control naming scheme */
+ HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+ HDSPM_WC_SYNC_CHECK("Word Clock Lock Status", 0),
+ HDSPM_MADI_SYNC_CHECK("MADI Sync Lock Status", 0),
+ HDSPM_LINE_OUT("Line Out", 0),
+ HDSPM_TX_64("TX 64 channels mode", 0),
+ HDSPM_C_TMS("Clear Track Marker", 0),
+ HDSPM_SAFE_MODE("Safe Mode", 0),
+ HDSPM_INPUT_SELECT("Input Select", 0),
+};
+
+static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
+
+ HDSPM_MIXER("Mixer", 0),
+/* 'Sample Clock Source' complies with the alsa control naming scheme */
+ HDSPM_CLOCK_SOURCE("Sample Clock Source", 0),
+
+ HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
+ HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
+ HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
+ HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
+/* 'External Rate' complies with the alsa control naming scheme */
+ HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+ HDSPM_WC_SYNC_CHECK("Word Clock Lock Status", 0),
+/* HDSPM_AES_SYNC_CHECK("AES Lock Status", 0),*/ /* created in snd_hdspm_create_controls() */
+ HDSPM_LINE_OUT("Line Out", 0),
+ HDSPM_EMPHASIS("Emphasis", 0),
+ HDSPM_DOLBY("Non Audio", 0),
+ HDSPM_PROFESSIONAL("Professional", 0),
+ HDSPM_C_TMS("Clear Track Marker", 0),
+ HDSPM_DS_WIRE("Double Speed Wire Mode", 0),
+ HDSPM_QS_WIRE("Quad Speed Wire Mode", 0),
+};
+
+static struct snd_kcontrol_new snd_hdspm_playback_mixer = HDSPM_PLAYBACK_MIXER;
+
+
+static int hdspm_update_simple_mixer_controls(struct hdspm * hdspm)
+{
+ int i;
+
+ for (i = hdspm->ds_channels; i < hdspm->ss_channels; ++i) {
+ if (hdspm->system_sample_rate > 48000) {
+ hdspm->playback_mixer_ctls[i]->vd[0].access =
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE |
+ SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+ } else {
+ hdspm->playback_mixer_ctls[i]->vd[0].access =
+ SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+ }
+ snd_ctl_notify(hdspm->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ &hdspm->playback_mixer_ctls[i]->id);
+ }
+
+ return 0;
+}
+
+
+static int snd_hdspm_create_controls(struct snd_card *card, struct hdspm * hdspm)
+{
+ unsigned int idx, limit;
+ int err;
+ struct snd_kcontrol *kctl;
+
+ /* add control list first */
+ if (hdspm->is_aes32) {
+ struct snd_kcontrol_new aes_sync_ctl =
+ HDSPM_AES_SYNC_CHECK("AES Lock Status", 0);
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_hdspm_controls_aes32);
+ idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_hdspm_controls_aes32[idx],
+ hdspm));
+ if (err < 0)
+ return err;
+ }
+ for (idx = 1; idx <= 8; idx++) {
+ aes_sync_ctl.index = idx;
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&aes_sync_ctl, hdspm));
+ if (err < 0)
+ return err;
+ }
+ } else {
+ for (idx = 0; idx < ARRAY_SIZE(snd_hdspm_controls_madi);
+ idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_hdspm_controls_madi[idx],
+ hdspm));
+ if (err < 0)
+ return err;
+ }
+ }
+
+ /* Channel playback mixer as default control
+ Note: the whole matrix would be 128*HDSPM_MIXER_CHANNELS Faders,
+ thats too * big for any alsamixer they are accesible via special
+ IOCTL on hwdep and the mixer 2dimensional mixer control
+ */
+
+ snd_hdspm_playback_mixer.name = "Chn";
+ limit = HDSPM_MAX_CHANNELS;
+
+ /* The index values are one greater than the channel ID so that
+ * alsamixer will display them correctly. We want to use the index
+ * for fast lookup of the relevant channel, but if we use it at all,
+ * most ALSA software does the wrong thing with it ...
+ */
+
+ for (idx = 0; idx < limit; ++idx) {
+ snd_hdspm_playback_mixer.index = idx + 1;
+ kctl = snd_ctl_new1(&snd_hdspm_playback_mixer, hdspm);
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
+ return err;
+ hdspm->playback_mixer_ctls[idx] = kctl;
+ }
+
+ return 0;
+}
+
+/*------------------------------------------------------------
+ /proc interface
+ ------------------------------------------------------------*/
+
+static void
+snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
+ struct snd_info_buffer *buffer)
+{
+ struct hdspm *hdspm = entry->private_data;
+ unsigned int status;
+ unsigned int status2;
+ char *pref_sync_ref;
+ char *autosync_ref;
+ char *system_clock_mode;
+ char *clock_source;
+ char *insel;
+ char *syncref;
+ int x, x2;
+
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+
+ snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
+ hdspm->card_name, hdspm->card->number + 1,
+ hdspm->firmware_rev,
+ (status2 & HDSPM_version0) |
+ (status2 & HDSPM_version1) | (status2 &
+ HDSPM_version2));
+
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+ snd_iprintf(buffer, "--- System ---\n");
+
+ snd_iprintf(buffer,
+ "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+ status & HDSPM_audioIRQPending,
+ (status & HDSPM_midi0IRQPending) ? 1 : 0,
+ (status & HDSPM_midi1IRQPending) ? 1 : 0,
+ hdspm->irq_count);
+ snd_iprintf(buffer,
+ "HW pointer: id = %d, rawptr = %d (%d->%d) "
+ "estimated= %ld (bytes)\n",
+ ((status & HDSPM_BufferID) ? 1 : 0),
+ (status & HDSPM_BufferPositionMask),
+ (status & HDSPM_BufferPositionMask) %
+ (2 * (int)hdspm->period_bytes),
+ ((status & HDSPM_BufferPositionMask) - 64) %
+ (2 * (int)hdspm->period_bytes),
+ (long) hdspm_hw_pointer(hdspm) * 4);
+
+ snd_iprintf(buffer,
+ "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+ hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+ snd_iprintf(buffer,
+ "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
+ "status2=0x%x\n",
+ hdspm->control_register, hdspm->control2_register,
+ status, status2);
+
+ snd_iprintf(buffer, "--- Settings ---\n");
+
+ x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
+ HDSPM_LatencyMask));
+
+ snd_iprintf(buffer,
+ "Size (Latency): %d samples (2 periods of %lu bytes)\n",
+ x, (unsigned long) hdspm->period_bytes);
+
+ snd_iprintf(buffer, "Line out: %s, Precise Pointer: %s\n",
+ (hdspm->control_register & HDSPM_LineOut) ? "on " : "off",
+ (hdspm->precise_ptr) ? "on" : "off");
+
+ switch (hdspm->control_register & HDSPM_InputMask) {
+ case HDSPM_InputOptical:
+ insel = "Optical";
+ break;
+ case HDSPM_InputCoaxial:
+ insel = "Coaxial";
+ break;
+ default:
+ insel = "Unkown";
+ }
+
+ switch (hdspm->control_register & HDSPM_SyncRefMask) {
+ case HDSPM_SyncRef_Word:
+ syncref = "WordClock";
+ break;
+ case HDSPM_SyncRef_MADI:
+ syncref = "MADI";
+ break;
+ default:
+ syncref = "Unkown";
+ }
+ snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel,
+ syncref);
+
+ snd_iprintf(buffer,
+ "ClearTrackMarker = %s, Transmit in %s Channel Mode, "
+ "Auto Input %s\n",
+ (hdspm->
+ control_register & HDSPM_clr_tms) ? "on" : "off",
+ (hdspm->
+ control_register & HDSPM_TX_64ch) ? "64" : "56",
+ (hdspm->
+ control_register & HDSPM_AutoInp) ? "on" : "off");
+
+ switch (hdspm_clock_source(hdspm)) {
+ case HDSPM_CLOCK_SOURCE_AUTOSYNC:
+ clock_source = "AutoSync";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_32KHZ:
+ clock_source = "Internal 32 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_44_1KHZ:
+ clock_source = "Internal 44.1 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_48KHZ:
+ clock_source = "Internal 48 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_64KHZ:
+ clock_source = "Internal 64 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_88_2KHZ:
+ clock_source = "Internal 88.2 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_96KHZ:
+ clock_source = "Internal 96 kHz";
+ break;
+ default:
+ clock_source = "Error";
+ }
+ snd_iprintf(buffer, "Sample Clock Source: %s\n", clock_source);
+ if (!(hdspm->control_register & HDSPM_ClockModeMaster))
+ system_clock_mode = "Slave";
+ else
+ system_clock_mode = "Master";
+ snd_iprintf(buffer, "System Clock Mode: %s\n", system_clock_mode);
+
+ switch (hdspm_pref_sync_ref(hdspm)) {
+ case HDSPM_SYNC_FROM_WORD:
+ pref_sync_ref = "Word Clock";
+ break;
+ case HDSPM_SYNC_FROM_MADI:
+ pref_sync_ref = "MADI Sync";
+ break;
+ default:
+ pref_sync_ref = "XXXX Clock";
+ break;
+ }
+ snd_iprintf(buffer, "Preferred Sync Reference: %s\n",
+ pref_sync_ref);
+
+ snd_iprintf(buffer, "System Clock Frequency: %d\n",
+ hdspm->system_sample_rate);
+
+
+ snd_iprintf(buffer, "--- Status:\n");
+
+ x = status & HDSPM_madiSync;
+ x2 = status2 & HDSPM_wcSync;
+
+ snd_iprintf(buffer, "Inputs MADI=%s, WordClock=%s\n",
+ (status & HDSPM_madiLock) ? (x ? "Sync" : "Lock") :
+ "NoLock",
+ (status2 & HDSPM_wcLock) ? (x2 ? "Sync" : "Lock") :
+ "NoLock");
+
+ switch (hdspm_autosync_ref(hdspm)) {
+ case HDSPM_AUTOSYNC_FROM_WORD:
+ autosync_ref = "Word Clock";
+ break;
+ case HDSPM_AUTOSYNC_FROM_MADI:
+ autosync_ref = "MADI Sync";
+ break;
+ case HDSPM_AUTOSYNC_FROM_NONE:
+ autosync_ref = "Input not valid";
+ break;
+ default:
+ autosync_ref = "---";
+ break;
+ }
+ snd_iprintf(buffer,
+ "AutoSync: Reference= %s, Freq=%d (MADI = %d, Word = %d)\n",
+ autosync_ref, hdspm_external_sample_rate(hdspm),
+ (status & HDSPM_madiFreqMask) >> 22,
+ (status2 & HDSPM_wcFreqMask) >> 5);
+
+ snd_iprintf(buffer, "Input: %s, Mode=%s\n",
+ (status & HDSPM_AB_int) ? "Coax" : "Optical",
+ (status & HDSPM_RX_64ch) ? "64 channels" :
+ "56 channels");
+
+ snd_iprintf(buffer, "\n");
+}
+
+static void
+snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
+ struct snd_info_buffer *buffer)
+{
+ struct hdspm *hdspm = entry->private_data;
+ unsigned int status;
+ unsigned int status2;
+ unsigned int timecode;
+ int pref_syncref;
+ char *autosync_ref;
+ char *system_clock_mode;
+ char *clock_source;
+ int x;
+
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
+
+ snd_iprintf(buffer, "%s (Card #%d) Rev.%x\n",
+ hdspm->card_name, hdspm->card->number + 1,
+ hdspm->firmware_rev);
+
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+ snd_iprintf(buffer, "--- System ---\n");
+
+ snd_iprintf(buffer,
+ "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+ status & HDSPM_audioIRQPending,
+ (status & HDSPM_midi0IRQPending) ? 1 : 0,
+ (status & HDSPM_midi1IRQPending) ? 1 : 0,
+ hdspm->irq_count);
+ snd_iprintf(buffer,
+ "HW pointer: id = %d, rawptr = %d (%d->%d) "
+ "estimated= %ld (bytes)\n",
+ ((status & HDSPM_BufferID) ? 1 : 0),
+ (status & HDSPM_BufferPositionMask),
+ (status & HDSPM_BufferPositionMask) %
+ (2 * (int)hdspm->period_bytes),
+ ((status & HDSPM_BufferPositionMask) - 64) %
+ (2 * (int)hdspm->period_bytes),
+ (long) hdspm_hw_pointer(hdspm) * 4);
+
+ snd_iprintf(buffer,
+ "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+ hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+ snd_iprintf(buffer,
+ "Register: ctrl1=0x%x, status1=0x%x, status2=0x%x, "
+ "timecode=0x%x\n",
+ hdspm->control_register,
+ status, status2, timecode);
+
+ snd_iprintf(buffer, "--- Settings ---\n");
+
+ x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
+ HDSPM_LatencyMask));
+
+ snd_iprintf(buffer,
+ "Size (Latency): %d samples (2 periods of %lu bytes)\n",
+ x, (unsigned long) hdspm->period_bytes);
+
+ snd_iprintf(buffer, "Line out: %s, Precise Pointer: %s\n",
+ (hdspm->
+ control_register & HDSPM_LineOut) ? "on " : "off",
+ (hdspm->precise_ptr) ? "on" : "off");
+
+ snd_iprintf(buffer,
+ "ClearTrackMarker %s, Emphasis %s, Dolby %s\n",
+ (hdspm->
+ control_register & HDSPM_clr_tms) ? "on" : "off",
+ (hdspm->
+ control_register & HDSPM_Emphasis) ? "on" : "off",
+ (hdspm->
+ control_register & HDSPM_Dolby) ? "on" : "off");
+
+ switch (hdspm_clock_source(hdspm)) {
+ case HDSPM_CLOCK_SOURCE_AUTOSYNC:
+ clock_source = "AutoSync";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_32KHZ:
+ clock_source = "Internal 32 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_44_1KHZ:
+ clock_source = "Internal 44.1 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_48KHZ:
+ clock_source = "Internal 48 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_64KHZ:
+ clock_source = "Internal 64 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_88_2KHZ:
+ clock_source = "Internal 88.2 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_96KHZ:
+ clock_source = "Internal 96 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_128KHZ:
+ clock_source = "Internal 128 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_176_4KHZ:
+ clock_source = "Internal 176.4 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_192KHZ:
+ clock_source = "Internal 192 kHz";
+ break;
+ default:
+ clock_source = "Error";
+ }
+ snd_iprintf(buffer, "Sample Clock Source: %s\n", clock_source);
+ if (!(hdspm->control_register & HDSPM_ClockModeMaster))
+ system_clock_mode = "Slave";
+ else
+ system_clock_mode = "Master";
+ snd_iprintf(buffer, "System Clock Mode: %s\n", system_clock_mode);
+
+ pref_syncref = hdspm_pref_sync_ref(hdspm);
+ if (pref_syncref == 0)
+ snd_iprintf(buffer, "Preferred Sync Reference: Word Clock\n");
+ else
+ snd_iprintf(buffer, "Preferred Sync Reference: AES%d\n",
+ pref_syncref);
+
+ snd_iprintf(buffer, "System Clock Frequency: %d\n",
+ hdspm->system_sample_rate);
+
+ snd_iprintf(buffer, "Double speed: %s\n",
+ hdspm->control_register & HDSPM_DS_DoubleWire?
+ "Double wire" : "Single wire");
+ snd_iprintf(buffer, "Quad speed: %s\n",
+ hdspm->control_register & HDSPM_QS_DoubleWire?
+ "Double wire" :
+ hdspm->control_register & HDSPM_QS_QuadWire?
+ "Quad wire" : "Single wire");
+
+ snd_iprintf(buffer, "--- Status:\n");
+
+ snd_iprintf(buffer, "Word: %s Frequency: %d\n",
+ (status & HDSPM_AES32_wcLock)? "Sync " : "No Lock",
+ HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF));
+
+ for (x = 0; x < 8; x++) {
+ snd_iprintf(buffer, "AES%d: %s Frequency: %d\n",
+ x+1,
+ (status2 & (HDSPM_LockAES >> x)) ?
+ "Sync ": "No Lock",
+ HDSPM_bit2freq((timecode >> (4*x)) & 0xF));
+ }
+
+ switch (hdspm_autosync_ref(hdspm)) {
+ case HDSPM_AES32_AUTOSYNC_FROM_NONE: autosync_ref="None"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_WORD: autosync_ref="Word Clock"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES1: autosync_ref="AES1"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES2: autosync_ref="AES2"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES3: autosync_ref="AES3"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES4: autosync_ref="AES4"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES5: autosync_ref="AES5"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES6: autosync_ref="AES6"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES7: autosync_ref="AES7"; break;
+ case HDSPM_AES32_AUTOSYNC_FROM_AES8: autosync_ref="AES8"; break;
+ default: autosync_ref = "---"; break;
+ }
+ snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref);
+
+ snd_iprintf(buffer, "\n");
+}
+
+#ifdef CONFIG_SND_DEBUG
+static void
+snd_hdspm_proc_read_debug(struct snd_info_entry * entry,
+ struct snd_info_buffer *buffer)
+{
+ struct hdspm *hdspm = entry->private_data;
+
+ int j,i;
+
+ for (i = 0; i < 256 /* 1024*64 */; i += j) {
+ snd_iprintf(buffer, "0x%08X: ", i);
+ for (j = 0; j < 16; j += 4)
+ snd_iprintf(buffer, "%08X ", hdspm_read(hdspm, i + j));
+ snd_iprintf(buffer, "\n");
+ }
+}
+#endif
+
+
+
+static void __devinit snd_hdspm_proc_init(struct hdspm * hdspm)
+{
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(hdspm->card, "hdspm", &entry))
+ snd_info_set_text_ops(entry, hdspm,
+ hdspm->is_aes32 ?
+ snd_hdspm_proc_read_aes32 :
+ snd_hdspm_proc_read_madi);
+#ifdef CONFIG_SND_DEBUG
+ /* debug file to read all hdspm registers */
+ if (!snd_card_proc_new(hdspm->card, "debug", &entry))
+ snd_info_set_text_ops(entry, hdspm,
+ snd_hdspm_proc_read_debug);
+#endif
+}
+
+/*------------------------------------------------------------
+ hdspm intitialize
+ ------------------------------------------------------------*/
+
+static int snd_hdspm_set_defaults(struct hdspm * hdspm)
+{
+ unsigned int i;
+
+ /* ASSUMPTION: hdspm->lock is either held, or there is no need to
+ hold it (e.g. during module initialization).
+ */
+
+ /* set defaults: */
+
+ if (hdspm->is_aes32)
+ hdspm->control_register =
+ HDSPM_ClockModeMaster | /* Master Cloack Mode on */
+ hdspm_encode_latency(7) | /* latency maximum =
+ * 8192 samples
+ */
+ HDSPM_SyncRef0 | /* AES1 is syncclock */
+ HDSPM_LineOut | /* Analog output in */
+ HDSPM_Professional; /* Professional mode */
+ else
+ hdspm->control_register =
+ HDSPM_ClockModeMaster | /* Master Cloack Mode on */
+ hdspm_encode_latency(7) | /* latency maximum =
+ * 8192 samples
+ */
+ HDSPM_InputCoaxial | /* Input Coax not Optical */
+ HDSPM_SyncRef_MADI | /* Madi is syncclock */
+ HDSPM_LineOut | /* Analog output in */
+ HDSPM_TX_64ch | /* transmit in 64ch mode */
+ HDSPM_AutoInp; /* AutoInput chossing (takeover) */
+
+ /* ! HDSPM_Frequency0|HDSPM_Frequency1 = 44.1khz */
+ /* ! HDSPM_DoubleSpeed HDSPM_QuadSpeed = normal speed */
+ /* ! HDSPM_clr_tms = do not clear bits in track marks */
+
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ if (!hdspm->is_aes32) {
+ /* No control2 register for AES32 */
+#ifdef SNDRV_BIG_ENDIAN
+ hdspm->control2_register = HDSPM_BIGENDIAN_MODE;
+#else
+ hdspm->control2_register = 0;
+#endif
+
+ hdspm_write(hdspm, HDSPM_control2Reg, hdspm->control2_register);
+ }
+ hdspm_compute_period_size(hdspm);
+
+ /* silence everything */
+
+ all_in_all_mixer(hdspm, 0 * UNITY_GAIN);
+
+ if (line_outs_monitor[hdspm->dev]) {
+
+ snd_printk(KERN_INFO "HDSPM: "
+ "sending all playback streams to line outs.\n");
+
+ for (i = 0; i < HDSPM_MIXER_CHANNELS; i++) {
+ if (hdspm_write_pb_gain(hdspm, i, i, UNITY_GAIN))
+ return -EIO;
+ }
+ }
+
+ /* set a default rate so that the channel map is set up. */
+ hdspm->channel_map = channel_map_madi_ss;
+ hdspm_set_rate(hdspm, 44100, 1);
+
+ return 0;
+}
+
+
+/*------------------------------------------------------------
+ interrupt
+ ------------------------------------------------------------*/
+
+static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
+{
+ struct hdspm *hdspm = (struct hdspm *) dev_id;
+ unsigned int status;
+ int audio;
+ int midi0;
+ int midi1;
+ unsigned int midi0status;
+ unsigned int midi1status;
+ int schedule = 0;
+
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+
+ audio = status & HDSPM_audioIRQPending;
+ midi0 = status & HDSPM_midi0IRQPending;
+ midi1 = status & HDSPM_midi1IRQPending;
+
+ if (!audio && !midi0 && !midi1)
+ return IRQ_NONE;
+
+ hdspm_write(hdspm, HDSPM_interruptConfirmation, 0);
+ hdspm->irq_count++;
+
+ midi0status = hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xff;
+ midi1status = hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xff;
+
+ if (audio) {
+
+ if (hdspm->capture_substream)
+ snd_pcm_period_elapsed(hdspm->capture_substream);
+
+ if (hdspm->playback_substream)
+ snd_pcm_period_elapsed(hdspm->playback_substream);
+ }
+
+ if (midi0 && midi0status) {
+ /* we disable interrupts for this input until processing
+ * is done
+ */
+ hdspm->control_register &= ~HDSPM_Midi0InterruptEnable;
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ hdspm->midi[0].pending = 1;
+ schedule = 1;
+ }
+ if (midi1 && midi1status) {
+ /* we disable interrupts for this input until processing
+ * is done
+ */
+ hdspm->control_register &= ~HDSPM_Midi1InterruptEnable;
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ hdspm->midi[1].pending = 1;
+ schedule = 1;
+ }
+ if (schedule)
+ tasklet_hi_schedule(&hdspm->midi_tasklet);
+ return IRQ_HANDLED;
+}
+
+/*------------------------------------------------------------
+ pcm interface
+ ------------------------------------------------------------*/
+
+
+static snd_pcm_uframes_t snd_hdspm_hw_pointer(struct snd_pcm_substream *
+ substream)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ return hdspm_hw_pointer(hdspm);
+}
+
+static char *hdspm_channel_buffer_location(struct hdspm * hdspm,
+ int stream, int channel)
+{
+ int mapped_channel;
+
+ if (snd_BUG_ON(channel < 0 || channel >= HDSPM_MAX_CHANNELS))
+ return NULL;
+
+ mapped_channel = hdspm->channel_map[channel];
+ if (mapped_channel < 0)
+ return NULL;
+
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
+ return hdspm->capture_buffer +
+ mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
+ else
+ return hdspm->playback_buffer +
+ mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
+}
+
+
+/* dont know why need it ??? */
+static int snd_hdspm_playback_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t pos,
+ void __user *src, snd_pcm_uframes_t count)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ if (snd_BUG_ON(pos + count > HDSPM_CHANNEL_BUFFER_BYTES / 4))
+ return -EINVAL;
+
+ channel_buf =
+ hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
+ channel);
+
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+
+ return copy_from_user(channel_buf + pos * 4, src, count * 4);
+}
+
+static int snd_hdspm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t pos,
+ void __user *dst, snd_pcm_uframes_t count)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ if (snd_BUG_ON(pos + count > HDSPM_CHANNEL_BUFFER_BYTES / 4))
+ return -EINVAL;
+
+ channel_buf =
+ hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
+ channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ return copy_to_user(dst, channel_buf + pos * 4, count * 4);
+}
+
+static int snd_hdspm_hw_silence(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ channel_buf =
+ hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
+ channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ memset(channel_buf + pos * 4, 0, count * 4);
+ return 0;
+}
+
+static int snd_hdspm_reset(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *other;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = hdspm->capture_substream;
+ else
+ other = hdspm->playback_substream;
+
+ if (hdspm->running)
+ runtime->status->hw_ptr = hdspm_hw_pointer(hdspm);
+ else
+ runtime->status->hw_ptr = 0;
+ if (other) {
+ struct snd_pcm_substream *s;
+ struct snd_pcm_runtime *oruntime = other->runtime;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == other) {
+ oruntime->status->hw_ptr =
+ runtime->status->hw_ptr;
+ break;
+ }
+ }
+ }
+ return 0;
+}
+
+static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ int err;
+ int i;
+ pid_t this_pid;
+ pid_t other_pid;
+
+ spin_lock_irq(&hdspm->lock);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ this_pid = hdspm->playback_pid;
+ other_pid = hdspm->capture_pid;
+ } else {
+ this_pid = hdspm->capture_pid;
+ other_pid = hdspm->playback_pid;
+ }
+
+ if (other_pid > 0 && this_pid != other_pid) {
+
+ /* The other stream is open, and not by the same
+ task as this one. Make sure that the parameters
+ that matter are the same.
+ */
+
+ if (params_rate(params) != hdspm->system_sample_rate) {
+ spin_unlock_irq(&hdspm->lock);
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ return -EBUSY;
+ }
+
+ if (params_period_size(params) != hdspm->period_bytes / 4) {
+ spin_unlock_irq(&hdspm->lock);
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return -EBUSY;
+ }
+
+ }
+ /* We're fine. */
+ spin_unlock_irq(&hdspm->lock);
+
+ /* how to make sure that the rate matches an externally-set one ? */
+
+ spin_lock_irq(&hdspm->lock);
+ err = hdspm_set_rate(hdspm, params_rate(params), 0);
+ if (err < 0) {
+ spin_unlock_irq(&hdspm->lock);
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ return err;
+ }
+ spin_unlock_irq(&hdspm->lock);
+
+ err = hdspm_set_interrupt_interval(hdspm,
+ params_period_size(params));
+ if (err < 0) {
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return err;
+ }
+
+ /* Memory allocation, takashi's method, dont know if we should
+ * spinlock
+ */
+ /* malloc all buffer even if not enabled to get sure */
+ /* Update for MADI rev 204: we need to allocate for all channels,
+ * otherwise it doesn't work at 96kHz */
+ err =
+ snd_pcm_lib_malloc_pages(substream, HDSPM_DMA_AREA_BYTES);
+ if (err < 0)
+ return err;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ hdspm_set_sgbuf(hdspm, substream, HDSPM_pageAddressBufferOut,
+ params_channels(params));
+
+ for (i = 0; i < params_channels(params); ++i)
+ snd_hdspm_enable_out(hdspm, i, 1);
+
+ hdspm->playback_buffer =
+ (unsigned char *) substream->runtime->dma_area;
+ snd_printdd("Allocated sample buffer for playback at %p\n",
+ hdspm->playback_buffer);
+ } else {
+ hdspm_set_sgbuf(hdspm, substream, HDSPM_pageAddressBufferIn,
+ params_channels(params));
+
+ for (i = 0; i < params_channels(params); ++i)
+ snd_hdspm_enable_in(hdspm, i, 1);
+
+ hdspm->capture_buffer =
+ (unsigned char *) substream->runtime->dma_area;
+ snd_printdd("Allocated sample buffer for capture at %p\n",
+ hdspm->capture_buffer);
+ }
+ /*
+ snd_printdd("Allocated sample buffer for %s at 0x%08X\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ "playback" : "capture",
+ snd_pcm_sgbuf_get_addr(substream, 0));
+ */
+ /*
+ snd_printdd("set_hwparams: %s %d Hz, %d channels, bs = %d\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ "playback" : "capture",
+ params_rate(params), params_channels(params),
+ params_buffer_size(params));
+ */
+ return 0;
+}
+
+static int snd_hdspm_hw_free(struct snd_pcm_substream *substream)
+{
+ int i;
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ /* params_channels(params) should be enough,
+ but to get sure in case of error */
+ for (i = 0; i < HDSPM_MAX_CHANNELS; ++i)
+ snd_hdspm_enable_out(hdspm, i, 0);
+
+ hdspm->playback_buffer = NULL;
+ } else {
+ for (i = 0; i < HDSPM_MAX_CHANNELS; ++i)
+ snd_hdspm_enable_in(hdspm, i, 0);
+
+ hdspm->capture_buffer = NULL;
+
+ }
+
+ snd_pcm_lib_free_pages(substream);
+
+ return 0;
+}
+
+static int snd_hdspm_channel_info(struct snd_pcm_substream *substream,
+ struct snd_pcm_channel_info * info)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ int mapped_channel;
+
+ if (snd_BUG_ON(info->channel >= HDSPM_MAX_CHANNELS))
+ return -EINVAL;
+
+ mapped_channel = hdspm->channel_map[info->channel];
+ if (mapped_channel < 0)
+ return -EINVAL;
+
+ info->offset = mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
+ info->first = 0;
+ info->step = 32;
+ return 0;
+}
+
+static int snd_hdspm_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL1_RESET:
+ return snd_hdspm_reset(substream);
+
+ case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
+ {
+ struct snd_pcm_channel_info *info = arg;
+ return snd_hdspm_channel_info(substream, info);
+ }
+ default:
+ break;
+ }
+
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static int snd_hdspm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *other;
+ int running;
+
+ spin_lock(&hdspm->lock);
+ running = hdspm->running;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ running |= 1 << substream->stream;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ running &= ~(1 << substream->stream);
+ break;
+ default:
+ snd_BUG();
+ spin_unlock(&hdspm->lock);
+ return -EINVAL;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = hdspm->capture_substream;
+ else
+ other = hdspm->playback_substream;
+
+ if (other) {
+ struct snd_pcm_substream *s;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == other) {
+ snd_pcm_trigger_done(s, substream);
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ running |= 1 << s->stream;
+ else
+ running &= ~(1 << s->stream);
+ goto _ok;
+ }
+ }
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ if (!(running & (1 << SNDRV_PCM_STREAM_PLAYBACK))
+ && substream->stream ==
+ SNDRV_PCM_STREAM_CAPTURE)
+ hdspm_silence_playback(hdspm);
+ } else {
+ if (running &&
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ hdspm_silence_playback(hdspm);
+ }
+ } else {
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ hdspm_silence_playback(hdspm);
+ }
+ _ok:
+ snd_pcm_trigger_done(substream, substream);
+ if (!hdspm->running && running)
+ hdspm_start_audio(hdspm);
+ else if (hdspm->running && !running)
+ hdspm_stop_audio(hdspm);
+ hdspm->running = running;
+ spin_unlock(&hdspm->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_prepare(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static unsigned int period_sizes[] =
+ { 64, 128, 256, 512, 1024, 2048, 4096, 8192 };
+
+static struct snd_pcm_hardware snd_hdspm_playback_subinfo = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_DOUBLE),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000 ),
+ .rate_min = 32000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = HDSPM_MAX_CHANNELS,
+ .buffer_bytes_max =
+ HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4),
+ .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0
+};
+
+static struct snd_pcm_hardware snd_hdspm_capture_subinfo = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000),
+ .rate_min = 32000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = HDSPM_MAX_CHANNELS,
+ .buffer_bytes_max =
+ HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4),
+ .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes = {
+ .count = ARRAY_SIZE(period_sizes),
+ .list = period_sizes,
+ .mask = 0
+};
+
+
+static int snd_hdspm_hw_rule_channels_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule * rule)
+{
+ struct hdspm *hdspm = rule->private;
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+
+ if (r->min > 48000 && r->max <= 96000) {
+ struct snd_interval t = {
+ .min = hdspm->ds_channels,
+ .max = hdspm->ds_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ } else if (r->max < 64000) {
+ struct snd_interval t = {
+ .min = hdspm->ss_channels,
+ .max = hdspm->ss_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ }
+ return 0;
+}
+
+static int snd_hdspm_hw_rule_rate_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule * rule)
+{
+ struct hdspm *hdspm = rule->private;
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+
+ if (c->min >= hdspm->ss_channels) {
+ struct snd_interval t = {
+ .min = 32000,
+ .max = 48000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ } else if (c->max <= hdspm->ds_channels) {
+ struct snd_interval t = {
+ .min = 64000,
+ .max = 96000,
+ .integer = 1,
+ };
+
+ return snd_interval_refine(r, &t);
+ }
+ return 0;
+}
+
+static int snd_hdspm_hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ unsigned int list[3];
+ struct hdspm *hdspm = rule->private;
+ struct snd_interval *c = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ if (hdspm->is_aes32) {
+ list[0] = hdspm->qs_channels;
+ list[1] = hdspm->ds_channels;
+ list[2] = hdspm->ss_channels;
+ return snd_interval_list(c, 3, list, 0);
+ } else {
+ list[0] = hdspm->ds_channels;
+ list[1] = hdspm->ss_channels;
+ return snd_interval_list(c, 2, list, 0);
+ }
+}
+
+
+static unsigned int hdspm_aes32_sample_rates[] = {
+ 32000, 44100, 48000, 64000, 88200, 96000, 128000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list
+hdspm_hw_constraints_aes32_sample_rates = {
+ .count = ARRAY_SIZE(hdspm_aes32_sample_rates),
+ .list = hdspm_aes32_sample_rates,
+ .mask = 0
+};
+
+static int snd_hdspm_playback_open(struct snd_pcm_substream *substream)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irq(&hdspm->lock);
+
+ snd_pcm_set_sync(substream);
+
+ runtime->hw = snd_hdspm_playback_subinfo;
+
+ if (hdspm->capture_substream == NULL)
+ hdspm_stop_audio(hdspm);
+
+ hdspm->playback_pid = current->pid;
+ hdspm->playback_substream = substream;
+
+ spin_unlock_irq(&hdspm->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes);
+
+ if (hdspm->is_aes32) {
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hdspm_hw_constraints_aes32_sample_rates);
+ } else {
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdspm_hw_rule_channels, hdspm,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdspm_hw_rule_channels_rate, hdspm,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_hdspm_hw_rule_rate_channels, hdspm,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ }
+ return 0;
+}
+
+static int snd_hdspm_playback_release(struct snd_pcm_substream *substream)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&hdspm->lock);
+
+ hdspm->playback_pid = -1;
+ hdspm->playback_substream = NULL;
+
+ spin_unlock_irq(&hdspm->lock);
+
+ return 0;
+}
+
+
+static int snd_hdspm_capture_open(struct snd_pcm_substream *substream)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irq(&hdspm->lock);
+ snd_pcm_set_sync(substream);
+ runtime->hw = snd_hdspm_capture_subinfo;
+
+ if (hdspm->playback_substream == NULL)
+ hdspm_stop_audio(hdspm);
+
+ hdspm->capture_pid = current->pid;
+ hdspm->capture_substream = substream;
+
+ spin_unlock_irq(&hdspm->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes);
+ if (hdspm->is_aes32) {
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hdspm_hw_constraints_aes32_sample_rates);
+ } else {
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdspm_hw_rule_channels, hdspm,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdspm_hw_rule_channels_rate, hdspm,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_hdspm_hw_rule_rate_channels, hdspm,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ }
+ return 0;
+}
+
+static int snd_hdspm_capture_release(struct snd_pcm_substream *substream)
+{
+ struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&hdspm->lock);
+
+ hdspm->capture_pid = -1;
+ hdspm->capture_substream = NULL;
+
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep * hw, struct file *file)
+{
+ /* we have nothing to initialize but the call is required */
+ return 0;
+}
+
+
+static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct hdspm *hdspm = hw->private_data;
+ struct hdspm_mixer_ioctl mixer;
+ struct hdspm_config_info info;
+ struct hdspm_version hdspm_version;
+ struct hdspm_peak_rms_ioctl rms;
+
+ switch (cmd) {
+
+ case SNDRV_HDSPM_IOCTL_GET_PEAK_RMS:
+ if (copy_from_user(&rms, (void __user *)arg, sizeof(rms)))
+ return -EFAULT;
+ /* maybe there is a chance to memorymap in future
+ * so dont touch just copy
+ */
+ if(copy_to_user_fromio((void __user *)rms.peak,
+ hdspm->iobase+HDSPM_MADI_peakrmsbase,
+ sizeof(struct hdspm_peak_rms)) != 0 )
+ return -EFAULT;
+
+ break;
+
+
+ case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO:
+
+ spin_lock_irq(&hdspm->lock);
+ info.pref_sync_ref = hdspm_pref_sync_ref(hdspm);
+ info.wordclock_sync_check = hdspm_wc_sync_check(hdspm);
+
+ info.system_sample_rate = hdspm->system_sample_rate;
+ info.autosync_sample_rate =
+ hdspm_external_sample_rate(hdspm);
+ info.system_clock_mode = hdspm_system_clock_mode(hdspm);
+ info.clock_source = hdspm_clock_source(hdspm);
+ info.autosync_ref = hdspm_autosync_ref(hdspm);
+ info.line_out = hdspm_line_out(hdspm);
+ info.passthru = 0;
+ spin_unlock_irq(&hdspm->lock);
+ if (copy_to_user((void __user *) arg, &info, sizeof(info)))
+ return -EFAULT;
+ break;
+
+ case SNDRV_HDSPM_IOCTL_GET_VERSION:
+ hdspm_version.firmware_rev = hdspm->firmware_rev;
+ if (copy_to_user((void __user *) arg, &hdspm_version,
+ sizeof(hdspm_version)))
+ return -EFAULT;
+ break;
+
+ case SNDRV_HDSPM_IOCTL_GET_MIXER:
+ if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer)))
+ return -EFAULT;
+ if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer,
+ sizeof(struct hdspm_mixer)))
+ return -EFAULT;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static struct snd_pcm_ops snd_hdspm_playback_ops = {
+ .open = snd_hdspm_playback_open,
+ .close = snd_hdspm_playback_release,
+ .ioctl = snd_hdspm_ioctl,
+ .hw_params = snd_hdspm_hw_params,
+ .hw_free = snd_hdspm_hw_free,
+ .prepare = snd_hdspm_prepare,
+ .trigger = snd_hdspm_trigger,
+ .pointer = snd_hdspm_hw_pointer,
+ .copy = snd_hdspm_playback_copy,
+ .silence = snd_hdspm_hw_silence,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static struct snd_pcm_ops snd_hdspm_capture_ops = {
+ .open = snd_hdspm_capture_open,
+ .close = snd_hdspm_capture_release,
+ .ioctl = snd_hdspm_ioctl,
+ .hw_params = snd_hdspm_hw_params,
+ .hw_free = snd_hdspm_hw_free,
+ .prepare = snd_hdspm_prepare,
+ .trigger = snd_hdspm_trigger,
+ .pointer = snd_hdspm_hw_pointer,
+ .copy = snd_hdspm_capture_copy,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
+ struct hdspm * hdspm)
+{
+ struct snd_hwdep *hw;
+ int err;
+
+ err = snd_hwdep_new(card, "HDSPM hwdep", 0, &hw);
+ if (err < 0)
+ return err;
+
+ hdspm->hwdep = hw;
+ hw->private_data = hdspm;
+ strcpy(hw->name, "HDSPM hwdep interface");
+
+ hw->ops.open = snd_hdspm_hwdep_dummy_op;
+ hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
+ hw->ops.release = snd_hdspm_hwdep_dummy_op;
+
+ return 0;
+}
+
+
+/*------------------------------------------------------------
+ memory interface
+ ------------------------------------------------------------*/
+static int __devinit snd_hdspm_preallocate_memory(struct hdspm * hdspm)
+{
+ int err;
+ struct snd_pcm *pcm;
+ size_t wanted;
+
+ pcm = hdspm->pcm;
+
+ wanted = HDSPM_DMA_AREA_BYTES;
+
+ err =
+ snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(hdspm->pci),
+ wanted,
+ wanted);
+ if (err < 0) {
+ snd_printdd("Could not preallocate %zd Bytes\n", wanted);
+
+ return err;
+ } else
+ snd_printdd(" Preallocated %zd Bytes\n", wanted);
+
+ return 0;
+}
+
+static void hdspm_set_sgbuf(struct hdspm * hdspm,
+ struct snd_pcm_substream *substream,
+ unsigned int reg, int channels)
+{
+ int i;
+ for (i = 0; i < (channels * 16); i++)
+ hdspm_write(hdspm, reg + 4 * i,
+ snd_pcm_sgbuf_get_addr(substream, 4096 * i));
+}
+
+/* ------------- ALSA Devices ---------------------------- */
+static int __devinit snd_hdspm_create_pcm(struct snd_card *card,
+ struct hdspm * hdspm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(card, hdspm->card_name, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ hdspm->pcm = pcm;
+ pcm->private_data = hdspm;
+ strcpy(pcm->name, hdspm->card_name);
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_hdspm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_hdspm_capture_ops);
+
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+
+ err = snd_hdspm_preallocate_memory(hdspm);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static inline void snd_hdspm_initialize_midi_flush(struct hdspm * hdspm)
+{
+ snd_hdspm_flush_midi_input(hdspm, 0);
+ snd_hdspm_flush_midi_input(hdspm, 1);
+}
+
+static int __devinit snd_hdspm_create_alsa_devices(struct snd_card *card,
+ struct hdspm * hdspm)
+{
+ int err;
+
+ snd_printdd("Create card...\n");
+ err = snd_hdspm_create_pcm(card, hdspm);
+ if (err < 0)
+ return err;
+
+ err = snd_hdspm_create_midi(card, hdspm, 0);
+ if (err < 0)
+ return err;
+
+ err = snd_hdspm_create_midi(card, hdspm, 1);
+ if (err < 0)
+ return err;
+
+ err = snd_hdspm_create_controls(card, hdspm);
+ if (err < 0)
+ return err;
+
+ err = snd_hdspm_create_hwdep(card, hdspm);
+ if (err < 0)
+ return err;
+
+ snd_printdd("proc init...\n");
+ snd_hdspm_proc_init(hdspm);
+
+ hdspm->system_sample_rate = -1;
+ hdspm->last_external_sample_rate = -1;
+ hdspm->last_internal_sample_rate = -1;
+ hdspm->playback_pid = -1;
+ hdspm->capture_pid = -1;
+ hdspm->capture_substream = NULL;
+ hdspm->playback_substream = NULL;
+
+ snd_printdd("Set defaults...\n");
+ err = snd_hdspm_set_defaults(hdspm);
+ if (err < 0)
+ return err;
+
+ snd_printdd("Update mixer controls...\n");
+ hdspm_update_simple_mixer_controls(hdspm);
+
+ snd_printdd("Initializeing complete ???\n");
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_printk(KERN_ERR "HDSPM: error registering card\n");
+ return err;
+ }
+
+ snd_printdd("... yes now\n");
+
+ return 0;
+}
+
+static int __devinit snd_hdspm_create(struct snd_card *card,
+ struct hdspm *hdspm,
+ int precise_ptr, int enable_monitor)
+{
+ struct pci_dev *pci = hdspm->pci;
+ int err;
+ unsigned long io_extent;
+
+ hdspm->irq = -1;
+
+ spin_lock_init(&hdspm->midi[0].lock);
+ spin_lock_init(&hdspm->midi[1].lock);
+
+ hdspm->card = card;
+
+ spin_lock_init(&hdspm->lock);
+
+ tasklet_init(&hdspm->midi_tasklet,
+ hdspm_midi_tasklet, (unsigned long) hdspm);
+
+ pci_read_config_word(hdspm->pci,
+ PCI_CLASS_REVISION, &hdspm->firmware_rev);
+
+ hdspm->is_aes32 = (hdspm->firmware_rev >= HDSPM_AESREVISION);
+
+ strcpy(card->mixername, "Xilinx FPGA");
+ if (hdspm->is_aes32) {
+ strcpy(card->driver, "HDSPAES32");
+ hdspm->card_name = "RME HDSPM AES32";
+ } else {
+ strcpy(card->driver, "HDSPM");
+ hdspm->card_name = "RME HDSPM MADI";
+ }
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ pci_set_master(hdspm->pci);
+
+ err = pci_request_regions(pci, "hdspm");
+ if (err < 0)
+ return err;
+
+ hdspm->port = pci_resource_start(pci, 0);
+ io_extent = pci_resource_len(pci, 0);
+
+ snd_printdd("grabbed memory region 0x%lx-0x%lx\n",
+ hdspm->port, hdspm->port + io_extent - 1);
+
+
+ hdspm->iobase = ioremap_nocache(hdspm->port, io_extent);
+ if (!hdspm->iobase) {
+ snd_printk(KERN_ERR "HDSPM: "
+ "unable to remap region 0x%lx-0x%lx\n",
+ hdspm->port, hdspm->port + io_extent - 1);
+ return -EBUSY;
+ }
+ snd_printdd("remapped region (0x%lx) 0x%lx-0x%lx\n",
+ (unsigned long)hdspm->iobase, hdspm->port,
+ hdspm->port + io_extent - 1);
+
+ if (request_irq(pci->irq, snd_hdspm_interrupt,
+ IRQF_SHARED, "hdspm", hdspm)) {
+ snd_printk(KERN_ERR "HDSPM: unable to use IRQ %d\n", pci->irq);
+ return -EBUSY;
+ }
+
+ snd_printdd("use IRQ %d\n", pci->irq);
+
+ hdspm->irq = pci->irq;
+ hdspm->precise_ptr = precise_ptr;
+
+ hdspm->monitor_outs = enable_monitor;
+
+ snd_printdd("kmalloc Mixer memory of %zd Bytes\n",
+ sizeof(struct hdspm_mixer));
+ hdspm->mixer = kzalloc(sizeof(struct hdspm_mixer), GFP_KERNEL);
+ if (!hdspm->mixer) {
+ snd_printk(KERN_ERR "HDSPM: "
+ "unable to kmalloc Mixer memory of %d Bytes\n",
+ (int)sizeof(struct hdspm_mixer));
+ return err;
+ }
+
+ hdspm->ss_channels = MADI_SS_CHANNELS;
+ hdspm->ds_channels = MADI_DS_CHANNELS;
+ hdspm->qs_channels = MADI_QS_CHANNELS;
+
+ snd_printdd("create alsa devices.\n");
+ err = snd_hdspm_create_alsa_devices(card, hdspm);
+ if (err < 0)
+ return err;
+
+ snd_hdspm_initialize_midi_flush(hdspm);
+
+ return 0;
+}
+
+static int snd_hdspm_free(struct hdspm * hdspm)
+{
+
+ if (hdspm->port) {
+
+ /* stop th audio, and cancel all interrupts */
+ hdspm->control_register &=
+ ~(HDSPM_Start | HDSPM_AudioInterruptEnable |
+ HDSPM_Midi0InterruptEnable | HDSPM_Midi1InterruptEnable);
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ }
+
+ if (hdspm->irq >= 0)
+ free_irq(hdspm->irq, (void *) hdspm);
+
+ kfree(hdspm->mixer);
+
+ if (hdspm->iobase)
+ iounmap(hdspm->iobase);
+
+ if (hdspm->port)
+ pci_release_regions(hdspm->pci);
+
+ pci_disable_device(hdspm->pci);
+ return 0;
+}
+
+static void snd_hdspm_card_free(struct snd_card *card)
+{
+ struct hdspm *hdspm = card->private_data;
+
+ if (hdspm)
+ snd_hdspm_free(hdspm);
+}
+
+static int __devinit snd_hdspm_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct hdspm *hdspm;
+ struct snd_card *card;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev],
+ THIS_MODULE, sizeof(struct hdspm));
+ if (!card)
+ return -ENOMEM;
+
+ hdspm = card->private_data;
+ card->private_free = snd_hdspm_card_free;
+ hdspm->dev = dev;
+ hdspm->pci = pci;
+
+ snd_card_set_dev(card, &pci->dev);
+
+ err = snd_hdspm_create(card, hdspm, precise_ptr[dev],
+ enable_monitor[dev]);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->shortname, "HDSPM MADI");
+ sprintf(card->longname, "%s at 0x%lx, irq %d", hdspm->card_name,
+ hdspm->port, hdspm->irq);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_hdspm_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "RME Hammerfall DSP MADI",
+ .id_table = snd_hdspm_ids,
+ .probe = snd_hdspm_probe,
+ .remove = __devexit_p(snd_hdspm_remove),
+};
+
+
+static int __init alsa_card_hdspm_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_hdspm_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_hdspm_init)
+module_exit(alsa_card_hdspm_exit)
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
new file mode 100644
index 0000000..2570907
--- /dev/null
+++ b/sound/pci/rme9652/rme9652.c
@@ -0,0 +1,2653 @@
+/*
+ * ALSA driver for RME Digi9652 audio interfaces
+ *
+ * Copyright (c) 1999 IEM - Winfried Ritsch
+ * Copyright (c) 1999-2001 Paul Davis
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+
+#include <asm/current.h>
+#include <asm/io.h>
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int precise_ptr[SNDRV_CARDS]; /* Enable precise pointer */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for RME Digi9652 (Hammerfall) soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for RME Digi9652 (Hammerfall) soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable/disable specific RME96{52,36} soundcards.");
+module_param_array(precise_ptr, bool, NULL, 0444);
+MODULE_PARM_DESC(precise_ptr, "Enable precise pointer (doesn't work reliably).");
+MODULE_AUTHOR("Paul Davis <pbd@op.net>, Winfried Ritsch");
+MODULE_DESCRIPTION("RME Digi9652/Digi9636");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{RME,Hammerfall},"
+ "{RME,Hammerfall-Light}}");
+
+/* The Hammerfall has two sets of 24 ADAT + 2 S/PDIF channels, one for
+ capture, one for playback. Both the ADAT and S/PDIF channels appear
+ to the host CPU in the same block of memory. There is no functional
+ difference between them in terms of access.
+
+ The Hammerfall Light is identical to the Hammerfall, except that it
+ has 2 sets 18 channels (16 ADAT + 2 S/PDIF) for capture and playback.
+*/
+
+#define RME9652_NCHANNELS 26
+#define RME9636_NCHANNELS 18
+
+/* Preferred sync source choices - used by "sync_pref" control switch */
+
+#define RME9652_SYNC_FROM_SPDIF 0
+#define RME9652_SYNC_FROM_ADAT1 1
+#define RME9652_SYNC_FROM_ADAT2 2
+#define RME9652_SYNC_FROM_ADAT3 3
+
+/* Possible sources of S/PDIF input */
+
+#define RME9652_SPDIFIN_OPTICAL 0 /* optical (ADAT1) */
+#define RME9652_SPDIFIN_COAXIAL 1 /* coaxial (RCA) */
+#define RME9652_SPDIFIN_INTERN 2 /* internal (CDROM) */
+
+/* ------------- Status-Register bits --------------------- */
+
+#define RME9652_IRQ (1<<0) /* IRQ is High if not reset by irq_clear */
+#define RME9652_lock_2 (1<<1) /* ADAT 3-PLL: 1=locked, 0=unlocked */
+#define RME9652_lock_1 (1<<2) /* ADAT 2-PLL: 1=locked, 0=unlocked */
+#define RME9652_lock_0 (1<<3) /* ADAT 1-PLL: 1=locked, 0=unlocked */
+#define RME9652_fs48 (1<<4) /* sample rate is 0=44.1/88.2,1=48/96 Khz */
+#define RME9652_wsel_rd (1<<5) /* if Word-Clock is used and valid then 1 */
+ /* bits 6-15 encode h/w buffer pointer position */
+#define RME9652_sync_2 (1<<16) /* if ADAT-IN 3 in sync to system clock */
+#define RME9652_sync_1 (1<<17) /* if ADAT-IN 2 in sync to system clock */
+#define RME9652_sync_0 (1<<18) /* if ADAT-IN 1 in sync to system clock */
+#define RME9652_DS_rd (1<<19) /* 1=Double Speed Mode, 0=Normal Speed */
+#define RME9652_tc_busy (1<<20) /* 1=time-code copy in progress (960ms) */
+#define RME9652_tc_out (1<<21) /* time-code out bit */
+#define RME9652_F_0 (1<<22) /* 000=64kHz, 100=88.2kHz, 011=96kHz */
+#define RME9652_F_1 (1<<23) /* 111=32kHz, 110=44.1kHz, 101=48kHz, */
+#define RME9652_F_2 (1<<24) /* external Crystal Chip if ERF=1 */
+#define RME9652_ERF (1<<25) /* Error-Flag of SDPIF Receiver (1=No Lock) */
+#define RME9652_buffer_id (1<<26) /* toggles by each interrupt on rec/play */
+#define RME9652_tc_valid (1<<27) /* 1 = a signal is detected on time-code input */
+#define RME9652_SPDIF_READ (1<<28) /* byte available from Rev 1.5+ S/PDIF interface */
+
+#define RME9652_sync (RME9652_sync_0|RME9652_sync_1|RME9652_sync_2)
+#define RME9652_lock (RME9652_lock_0|RME9652_lock_1|RME9652_lock_2)
+#define RME9652_F (RME9652_F_0|RME9652_F_1|RME9652_F_2)
+#define rme9652_decode_spdif_rate(x) ((x)>>22)
+
+/* Bit 6..15 : h/w buffer pointer */
+
+#define RME9652_buf_pos 0x000FFC0
+
+/* Bits 31,30,29 are bits 5,4,3 of h/w pointer position on later
+ Rev G EEPROMS and Rev 1.5 cards or later.
+*/
+
+#define RME9652_REV15_buf_pos(x) ((((x)&0xE0000000)>>26)|((x)&RME9652_buf_pos))
+
+/* amount of io space we remap for register access. i'm not sure we
+ even need this much, but 1K is nice round number :)
+*/
+
+#define RME9652_IO_EXTENT 1024
+
+#define RME9652_init_buffer 0
+#define RME9652_play_buffer 32 /* holds ptr to 26x64kBit host RAM */
+#define RME9652_rec_buffer 36 /* holds ptr to 26x64kBit host RAM */
+#define RME9652_control_register 64
+#define RME9652_irq_clear 96
+#define RME9652_time_code 100 /* useful if used with alesis adat */
+#define RME9652_thru_base 128 /* 132...228 Thru for 26 channels */
+
+/* Read-only registers */
+
+/* Writing to any of the register locations writes to the status
+ register. We'll use the first location as our point of access.
+*/
+
+#define RME9652_status_register 0
+
+/* --------- Control-Register Bits ---------------- */
+
+
+#define RME9652_start_bit (1<<0) /* start record/play */
+ /* bits 1-3 encode buffersize/latency */
+#define RME9652_Master (1<<4) /* Clock Mode Master=1,Slave/Auto=0 */
+#define RME9652_IE (1<<5) /* Interrupt Enable */
+#define RME9652_freq (1<<6) /* samplerate 0=44.1/88.2, 1=48/96 kHz */
+#define RME9652_freq1 (1<<7) /* if 0, 32kHz, else always 1 */
+#define RME9652_DS (1<<8) /* Doule Speed 0=44.1/48, 1=88.2/96 Khz */
+#define RME9652_PRO (1<<9) /* S/PDIF out: 0=consumer, 1=professional */
+#define RME9652_EMP (1<<10) /* Emphasis 0=None, 1=ON */
+#define RME9652_Dolby (1<<11) /* Non-audio bit 1=set, 0=unset */
+#define RME9652_opt_out (1<<12) /* Use 1st optical OUT as SPDIF: 1=yes,0=no */
+#define RME9652_wsel (1<<13) /* use Wordclock as sync (overwrites master) */
+#define RME9652_inp_0 (1<<14) /* SPDIF-IN: 00=optical (ADAT1), */
+#define RME9652_inp_1 (1<<15) /* 01=koaxial (Cinch), 10=Internal CDROM */
+#define RME9652_SyncPref_ADAT2 (1<<16)
+#define RME9652_SyncPref_ADAT3 (1<<17)
+#define RME9652_SPDIF_RESET (1<<18) /* Rev 1.5+: h/w S/PDIF receiver */
+#define RME9652_SPDIF_SELECT (1<<19)
+#define RME9652_SPDIF_CLOCK (1<<20)
+#define RME9652_SPDIF_WRITE (1<<21)
+#define RME9652_ADAT1_INTERNAL (1<<22) /* Rev 1.5+: if set, internal CD connector carries ADAT */
+
+/* buffersize = 512Bytes * 2^n, where n is made from Bit2 ... Bit0 */
+
+#define RME9652_latency 0x0e
+#define rme9652_encode_latency(x) (((x)&0x7)<<1)
+#define rme9652_decode_latency(x) (((x)>>1)&0x7)
+#define rme9652_running_double_speed(s) ((s)->control_register & RME9652_DS)
+#define RME9652_inp (RME9652_inp_0|RME9652_inp_1)
+#define rme9652_encode_spdif_in(x) (((x)&0x3)<<14)
+#define rme9652_decode_spdif_in(x) (((x)>>14)&0x3)
+
+#define RME9652_SyncPref_Mask (RME9652_SyncPref_ADAT2|RME9652_SyncPref_ADAT3)
+#define RME9652_SyncPref_ADAT1 0
+#define RME9652_SyncPref_SPDIF (RME9652_SyncPref_ADAT2|RME9652_SyncPref_ADAT3)
+
+/* the size of a substream (1 mono data stream) */
+
+#define RME9652_CHANNEL_BUFFER_SAMPLES (16*1024)
+#define RME9652_CHANNEL_BUFFER_BYTES (4*RME9652_CHANNEL_BUFFER_SAMPLES)
+
+/* the size of the area we need to allocate for DMA transfers. the
+ size is the same regardless of the number of channels - the
+ 9636 still uses the same memory area.
+
+ Note that we allocate 1 more channel than is apparently needed
+ because the h/w seems to write 1 byte beyond the end of the last
+ page. Sigh.
+*/
+
+#define RME9652_DMA_AREA_BYTES ((RME9652_NCHANNELS+1) * RME9652_CHANNEL_BUFFER_BYTES)
+#define RME9652_DMA_AREA_KILOBYTES (RME9652_DMA_AREA_BYTES/1024)
+
+struct snd_rme9652 {
+ int dev;
+
+ spinlock_t lock;
+ int irq;
+ unsigned long port;
+ void __iomem *iobase;
+
+ int precise_ptr;
+
+ u32 control_register; /* cached value */
+ u32 thru_bits; /* thru 1=on, 0=off channel 1=Bit1... channel 26= Bit26 */
+
+ u32 creg_spdif;
+ u32 creg_spdif_stream;
+
+ char *card_name; /* hammerfall or hammerfall light names */
+
+ size_t hw_offsetmask; /* &-with status register to get real hw_offset */
+ size_t prev_hw_offset; /* previous hw offset */
+ size_t max_jitter; /* maximum jitter in frames for
+ hw pointer */
+ size_t period_bytes; /* guess what this is */
+
+ unsigned char ds_channels;
+ unsigned char ss_channels; /* different for hammerfall/hammerfall-light */
+
+ struct snd_dma_buffer playback_dma_buf;
+ struct snd_dma_buffer capture_dma_buf;
+
+ unsigned char *capture_buffer; /* suitably aligned address */
+ unsigned char *playback_buffer; /* suitably aligned address */
+
+ pid_t capture_pid;
+ pid_t playback_pid;
+
+ struct snd_pcm_substream *capture_substream;
+ struct snd_pcm_substream *playback_substream;
+ int running;
+
+ int passthru; /* non-zero if doing pass-thru */
+ int hw_rev; /* h/w rev * 10 (i.e. 1.5 has hw_rev = 15) */
+
+ int last_spdif_sample_rate; /* so that we can catch externally ... */
+ int last_adat_sample_rate; /* ... induced rate changes */
+
+ char *channel_map;
+
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct pci_dev *pci;
+ struct snd_kcontrol *spdif_ctl;
+
+};
+
+/* These tables map the ALSA channels 1..N to the channels that we
+ need to use in order to find the relevant channel buffer. RME
+ refer to this kind of mapping as between "the ADAT channel and
+ the DMA channel." We index it using the logical audio channel,
+ and the value is the DMA channel (i.e. channel buffer number)
+ where the data for that channel can be read/written from/to.
+*/
+
+static char channel_map_9652_ss[26] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17,
+ 18, 19, 20, 21, 22, 23, 24, 25
+};
+
+static char channel_map_9636_ss[26] = {
+ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15,
+ /* channels 16 and 17 are S/PDIF */
+ 24, 25,
+ /* channels 18-25 don't exist */
+ -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+static char channel_map_9652_ds[26] = {
+ /* ADAT channels are remapped */
+ 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23,
+ /* channels 12 and 13 are S/PDIF */
+ 24, 25,
+ /* others don't exist */
+ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+static char channel_map_9636_ds[26] = {
+ /* ADAT channels are remapped */
+ 1, 3, 5, 7, 9, 11, 13, 15,
+ /* channels 8 and 9 are S/PDIF */
+ 24, 25
+ /* others don't exist */
+ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer *dmab, size_t size)
+{
+ dmab->dev.type = SNDRV_DMA_TYPE_DEV;
+ dmab->dev.dev = snd_dma_pci_data(pci);
+ if (snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
+ if (dmab->bytes >= size)
+ return 0;
+ }
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, dmab) < 0)
+ return -ENOMEM;
+ return 0;
+}
+
+static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci)
+{
+ if (dmab->area) {
+ dmab->dev.dev = NULL; /* make it anonymous */
+ snd_dma_reserve_buf(dmab, snd_dma_pci_buf_id(pci));
+ }
+}
+
+
+static struct pci_device_id snd_rme9652_ids[] = {
+ {
+ .vendor = 0x10ee,
+ .device = 0x3fc4,
+ .subvendor = PCI_ANY_ID,
+ .subdevice = PCI_ANY_ID,
+ }, /* RME Digi9652 */
+ { 0, },
+};
+
+MODULE_DEVICE_TABLE(pci, snd_rme9652_ids);
+
+static inline void rme9652_write(struct snd_rme9652 *rme9652, int reg, int val)
+{
+ writel(val, rme9652->iobase + reg);
+}
+
+static inline unsigned int rme9652_read(struct snd_rme9652 *rme9652, int reg)
+{
+ return readl(rme9652->iobase + reg);
+}
+
+static inline int snd_rme9652_use_is_exclusive(struct snd_rme9652 *rme9652)
+{
+ unsigned long flags;
+ int ret = 1;
+
+ spin_lock_irqsave(&rme9652->lock, flags);
+ if ((rme9652->playback_pid != rme9652->capture_pid) &&
+ (rme9652->playback_pid >= 0) && (rme9652->capture_pid >= 0)) {
+ ret = 0;
+ }
+ spin_unlock_irqrestore(&rme9652->lock, flags);
+ return ret;
+}
+
+static inline int rme9652_adat_sample_rate(struct snd_rme9652 *rme9652)
+{
+ if (rme9652_running_double_speed(rme9652)) {
+ return (rme9652_read(rme9652, RME9652_status_register) &
+ RME9652_fs48) ? 96000 : 88200;
+ } else {
+ return (rme9652_read(rme9652, RME9652_status_register) &
+ RME9652_fs48) ? 48000 : 44100;
+ }
+}
+
+static inline void rme9652_compute_period_size(struct snd_rme9652 *rme9652)
+{
+ unsigned int i;
+
+ i = rme9652->control_register & RME9652_latency;
+ rme9652->period_bytes = 1 << ((rme9652_decode_latency(i) + 8));
+ rme9652->hw_offsetmask =
+ (rme9652->period_bytes * 2 - 1) & RME9652_buf_pos;
+ rme9652->max_jitter = 80;
+}
+
+static snd_pcm_uframes_t rme9652_hw_pointer(struct snd_rme9652 *rme9652)
+{
+ int status;
+ unsigned int offset, frag;
+ snd_pcm_uframes_t period_size = rme9652->period_bytes / 4;
+ snd_pcm_sframes_t delta;
+
+ status = rme9652_read(rme9652, RME9652_status_register);
+ if (!rme9652->precise_ptr)
+ return (status & RME9652_buffer_id) ? period_size : 0;
+ offset = status & RME9652_buf_pos;
+
+ /* The hardware may give a backward movement for up to 80 frames
+ Martin Kirst <martin.kirst@freenet.de> knows the details.
+ */
+
+ delta = rme9652->prev_hw_offset - offset;
+ delta &= 0xffff;
+ if (delta <= (snd_pcm_sframes_t)rme9652->max_jitter * 4)
+ offset = rme9652->prev_hw_offset;
+ else
+ rme9652->prev_hw_offset = offset;
+ offset &= rme9652->hw_offsetmask;
+ offset /= 4;
+ frag = status & RME9652_buffer_id;
+
+ if (offset < period_size) {
+ if (offset > rme9652->max_jitter) {
+ if (frag)
+ printk(KERN_ERR "Unexpected hw_pointer position (bufid == 0): status: %x offset: %d\n", status, offset);
+ } else if (!frag)
+ return 0;
+ offset -= rme9652->max_jitter;
+ if ((int)offset < 0)
+ offset += period_size * 2;
+ } else {
+ if (offset > period_size + rme9652->max_jitter) {
+ if (!frag)
+ printk(KERN_ERR "Unexpected hw_pointer position (bufid == 1): status: %x offset: %d\n", status, offset);
+ } else if (frag)
+ return period_size;
+ offset -= rme9652->max_jitter;
+ }
+
+ return offset;
+}
+
+static inline void rme9652_reset_hw_pointer(struct snd_rme9652 *rme9652)
+{
+ int i;
+
+ /* reset the FIFO pointer to zero. We do this by writing to 8
+ registers, each of which is a 32bit wide register, and set
+ them all to zero. Note that s->iobase is a pointer to
+ int32, not pointer to char.
+ */
+
+ for (i = 0; i < 8; i++) {
+ rme9652_write(rme9652, i * 4, 0);
+ udelay(10);
+ }
+ rme9652->prev_hw_offset = 0;
+}
+
+static inline void rme9652_start(struct snd_rme9652 *s)
+{
+ s->control_register |= (RME9652_IE | RME9652_start_bit);
+ rme9652_write(s, RME9652_control_register, s->control_register);
+}
+
+static inline void rme9652_stop(struct snd_rme9652 *s)
+{
+ s->control_register &= ~(RME9652_start_bit | RME9652_IE);
+ rme9652_write(s, RME9652_control_register, s->control_register);
+}
+
+static int rme9652_set_interrupt_interval(struct snd_rme9652 *s,
+ unsigned int frames)
+{
+ int restart = 0;
+ int n;
+
+ spin_lock_irq(&s->lock);
+
+ if ((restart = s->running)) {
+ rme9652_stop(s);
+ }
+
+ frames >>= 7;
+ n = 0;
+ while (frames) {
+ n++;
+ frames >>= 1;
+ }
+
+ s->control_register &= ~RME9652_latency;
+ s->control_register |= rme9652_encode_latency(n);
+
+ rme9652_write(s, RME9652_control_register, s->control_register);
+
+ rme9652_compute_period_size(s);
+
+ if (restart)
+ rme9652_start(s);
+
+ spin_unlock_irq(&s->lock);
+
+ return 0;
+}
+
+static int rme9652_set_rate(struct snd_rme9652 *rme9652, int rate)
+{
+ int restart;
+ int reject_if_open = 0;
+ int xrate;
+
+ if (!snd_rme9652_use_is_exclusive (rme9652)) {
+ return -EBUSY;
+ }
+
+ /* Changing from a "single speed" to a "double speed" rate is
+ not allowed if any substreams are open. This is because
+ such a change causes a shift in the location of
+ the DMA buffers and a reduction in the number of available
+ buffers.
+
+ Note that a similar but essentially insoluble problem
+ exists for externally-driven rate changes. All we can do
+ is to flag rate changes in the read/write routines.
+ */
+
+ spin_lock_irq(&rme9652->lock);
+ xrate = rme9652_adat_sample_rate(rme9652);
+
+ switch (rate) {
+ case 44100:
+ if (xrate > 48000) {
+ reject_if_open = 1;
+ }
+ rate = 0;
+ break;
+ case 48000:
+ if (xrate > 48000) {
+ reject_if_open = 1;
+ }
+ rate = RME9652_freq;
+ break;
+ case 88200:
+ if (xrate < 48000) {
+ reject_if_open = 1;
+ }
+ rate = RME9652_DS;
+ break;
+ case 96000:
+ if (xrate < 48000) {
+ reject_if_open = 1;
+ }
+ rate = RME9652_DS | RME9652_freq;
+ break;
+ default:
+ spin_unlock_irq(&rme9652->lock);
+ return -EINVAL;
+ }
+
+ if (reject_if_open && (rme9652->capture_pid >= 0 || rme9652->playback_pid >= 0)) {
+ spin_unlock_irq(&rme9652->lock);
+ return -EBUSY;
+ }
+
+ if ((restart = rme9652->running)) {
+ rme9652_stop(rme9652);
+ }
+ rme9652->control_register &= ~(RME9652_freq | RME9652_DS);
+ rme9652->control_register |= rate;
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+
+ if (restart) {
+ rme9652_start(rme9652);
+ }
+
+ if (rate & RME9652_DS) {
+ if (rme9652->ss_channels == RME9652_NCHANNELS) {
+ rme9652->channel_map = channel_map_9652_ds;
+ } else {
+ rme9652->channel_map = channel_map_9636_ds;
+ }
+ } else {
+ if (rme9652->ss_channels == RME9652_NCHANNELS) {
+ rme9652->channel_map = channel_map_9652_ss;
+ } else {
+ rme9652->channel_map = channel_map_9636_ss;
+ }
+ }
+
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static void rme9652_set_thru(struct snd_rme9652 *rme9652, int channel, int enable)
+{
+ int i;
+
+ rme9652->passthru = 0;
+
+ if (channel < 0) {
+
+ /* set thru for all channels */
+
+ if (enable) {
+ for (i = 0; i < RME9652_NCHANNELS; i++) {
+ rme9652->thru_bits |= (1 << i);
+ rme9652_write(rme9652, RME9652_thru_base + i * 4, 1);
+ }
+ } else {
+ for (i = 0; i < RME9652_NCHANNELS; i++) {
+ rme9652->thru_bits &= ~(1 << i);
+ rme9652_write(rme9652, RME9652_thru_base + i * 4, 0);
+ }
+ }
+
+ } else {
+ int mapped_channel;
+
+ mapped_channel = rme9652->channel_map[channel];
+
+ if (enable) {
+ rme9652->thru_bits |= (1 << mapped_channel);
+ } else {
+ rme9652->thru_bits &= ~(1 << mapped_channel);
+ }
+
+ rme9652_write(rme9652,
+ RME9652_thru_base + mapped_channel * 4,
+ enable ? 1 : 0);
+ }
+}
+
+static int rme9652_set_passthru(struct snd_rme9652 *rme9652, int onoff)
+{
+ if (onoff) {
+ rme9652_set_thru(rme9652, -1, 1);
+
+ /* we don't want interrupts, so do a
+ custom version of rme9652_start().
+ */
+
+ rme9652->control_register =
+ RME9652_inp_0 |
+ rme9652_encode_latency(7) |
+ RME9652_start_bit;
+
+ rme9652_reset_hw_pointer(rme9652);
+
+ rme9652_write(rme9652, RME9652_control_register,
+ rme9652->control_register);
+ rme9652->passthru = 1;
+ } else {
+ rme9652_set_thru(rme9652, -1, 0);
+ rme9652_stop(rme9652);
+ rme9652->passthru = 0;
+ }
+
+ return 0;
+}
+
+static void rme9652_spdif_set_bit (struct snd_rme9652 *rme9652, int mask, int onoff)
+{
+ if (onoff)
+ rme9652->control_register |= mask;
+ else
+ rme9652->control_register &= ~mask;
+
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+}
+
+static void rme9652_spdif_write_byte (struct snd_rme9652 *rme9652, const int val)
+{
+ long mask;
+ long i;
+
+ for (i = 0, mask = 0x80; i < 8; i++, mask >>= 1) {
+ if (val & mask)
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_WRITE, 1);
+ else
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_WRITE, 0);
+
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_CLOCK, 1);
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_CLOCK, 0);
+ }
+}
+
+static int rme9652_spdif_read_byte (struct snd_rme9652 *rme9652)
+{
+ long mask;
+ long val;
+ long i;
+
+ val = 0;
+
+ for (i = 0, mask = 0x80; i < 8; i++, mask >>= 1) {
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_CLOCK, 1);
+ if (rme9652_read (rme9652, RME9652_status_register) & RME9652_SPDIF_READ)
+ val |= mask;
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_CLOCK, 0);
+ }
+
+ return val;
+}
+
+static void rme9652_write_spdif_codec (struct snd_rme9652 *rme9652, const int address, const int data)
+{
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_SELECT, 1);
+ rme9652_spdif_write_byte (rme9652, 0x20);
+ rme9652_spdif_write_byte (rme9652, address);
+ rme9652_spdif_write_byte (rme9652, data);
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_SELECT, 0);
+}
+
+
+static int rme9652_spdif_read_codec (struct snd_rme9652 *rme9652, const int address)
+{
+ int ret;
+
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_SELECT, 1);
+ rme9652_spdif_write_byte (rme9652, 0x20);
+ rme9652_spdif_write_byte (rme9652, address);
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_SELECT, 0);
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_SELECT, 1);
+
+ rme9652_spdif_write_byte (rme9652, 0x21);
+ ret = rme9652_spdif_read_byte (rme9652);
+ rme9652_spdif_set_bit (rme9652, RME9652_SPDIF_SELECT, 0);
+
+ return ret;
+}
+
+static void rme9652_initialize_spdif_receiver (struct snd_rme9652 *rme9652)
+{
+ /* XXX what unsets this ? */
+
+ rme9652->control_register |= RME9652_SPDIF_RESET;
+
+ rme9652_write_spdif_codec (rme9652, 4, 0x40);
+ rme9652_write_spdif_codec (rme9652, 17, 0x13);
+ rme9652_write_spdif_codec (rme9652, 6, 0x02);
+}
+
+static inline int rme9652_spdif_sample_rate(struct snd_rme9652 *s)
+{
+ unsigned int rate_bits;
+
+ if (rme9652_read(s, RME9652_status_register) & RME9652_ERF) {
+ return -1; /* error condition */
+ }
+
+ if (s->hw_rev == 15) {
+
+ int x, y, ret;
+
+ x = rme9652_spdif_read_codec (s, 30);
+
+ if (x != 0)
+ y = 48000 * 64 / x;
+ else
+ y = 0;
+
+ if (y > 30400 && y < 33600) ret = 32000;
+ else if (y > 41900 && y < 46000) ret = 44100;
+ else if (y > 46000 && y < 50400) ret = 48000;
+ else if (y > 60800 && y < 67200) ret = 64000;
+ else if (y > 83700 && y < 92000) ret = 88200;
+ else if (y > 92000 && y < 100000) ret = 96000;
+ else ret = 0;
+ return ret;
+ }
+
+ rate_bits = rme9652_read(s, RME9652_status_register) & RME9652_F;
+
+ switch (rme9652_decode_spdif_rate(rate_bits)) {
+ case 0x7:
+ return 32000;
+ break;
+
+ case 0x6:
+ return 44100;
+ break;
+
+ case 0x5:
+ return 48000;
+ break;
+
+ case 0x4:
+ return 88200;
+ break;
+
+ case 0x3:
+ return 96000;
+ break;
+
+ case 0x0:
+ return 64000;
+ break;
+
+ default:
+ snd_printk(KERN_ERR "%s: unknown S/PDIF input rate (bits = 0x%x)\n",
+ s->card_name, rate_bits);
+ return 0;
+ break;
+ }
+}
+
+/*-----------------------------------------------------------------------------
+ Control Interface
+ ----------------------------------------------------------------------------*/
+
+static u32 snd_rme9652_convert_from_aes(struct snd_aes_iec958 *aes)
+{
+ u32 val = 0;
+ val |= (aes->status[0] & IEC958_AES0_PROFESSIONAL) ? RME9652_PRO : 0;
+ val |= (aes->status[0] & IEC958_AES0_NONAUDIO) ? RME9652_Dolby : 0;
+ if (val & RME9652_PRO)
+ val |= (aes->status[0] & IEC958_AES0_PRO_EMPHASIS_5015) ? RME9652_EMP : 0;
+ else
+ val |= (aes->status[0] & IEC958_AES0_CON_EMPHASIS_5015) ? RME9652_EMP : 0;
+ return val;
+}
+
+static void snd_rme9652_convert_to_aes(struct snd_aes_iec958 *aes, u32 val)
+{
+ aes->status[0] = ((val & RME9652_PRO) ? IEC958_AES0_PROFESSIONAL : 0) |
+ ((val & RME9652_Dolby) ? IEC958_AES0_NONAUDIO : 0);
+ if (val & RME9652_PRO)
+ aes->status[0] |= (val & RME9652_EMP) ? IEC958_AES0_PRO_EMPHASIS_5015 : 0;
+ else
+ aes->status[0] |= (val & RME9652_EMP) ? IEC958_AES0_CON_EMPHASIS_5015 : 0;
+}
+
+static int snd_rme9652_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme9652_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ snd_rme9652_convert_to_aes(&ucontrol->value.iec958, rme9652->creg_spdif);
+ return 0;
+}
+
+static int snd_rme9652_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_rme9652_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&rme9652->lock);
+ change = val != rme9652->creg_spdif;
+ rme9652->creg_spdif = val;
+ spin_unlock_irq(&rme9652->lock);
+ return change;
+}
+
+static int snd_rme9652_control_spdif_stream_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme9652_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ snd_rme9652_convert_to_aes(&ucontrol->value.iec958, rme9652->creg_spdif_stream);
+ return 0;
+}
+
+static int snd_rme9652_control_spdif_stream_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ u32 val;
+
+ val = snd_rme9652_convert_from_aes(&ucontrol->value.iec958);
+ spin_lock_irq(&rme9652->lock);
+ change = val != rme9652->creg_spdif_stream;
+ rme9652->creg_spdif_stream = val;
+ rme9652->control_register &= ~(RME9652_PRO | RME9652_Dolby | RME9652_EMP);
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register |= val);
+ spin_unlock_irq(&rme9652->lock);
+ return change;
+}
+
+static int snd_rme9652_control_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_rme9652_control_spdif_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = kcontrol->private_value;
+ return 0;
+}
+
+#define RME9652_ADAT1_IN(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_rme9652_info_adat1_in, \
+ .get = snd_rme9652_get_adat1_in, \
+ .put = snd_rme9652_put_adat1_in }
+
+static unsigned int rme9652_adat1_in(struct snd_rme9652 *rme9652)
+{
+ if (rme9652->control_register & RME9652_ADAT1_INTERNAL)
+ return 1;
+ return 0;
+}
+
+static int rme9652_set_adat1_input(struct snd_rme9652 *rme9652, int internal)
+{
+ int restart = 0;
+
+ if (internal) {
+ rme9652->control_register |= RME9652_ADAT1_INTERNAL;
+ } else {
+ rme9652->control_register &= ~RME9652_ADAT1_INTERNAL;
+ }
+
+ /* XXX do we actually need to stop the card when we do this ? */
+
+ if ((restart = rme9652->running)) {
+ rme9652_stop(rme9652);
+ }
+
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+
+ if (restart) {
+ rme9652_start(rme9652);
+ }
+
+ return 0;
+}
+
+static int snd_rme9652_info_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = {"ADAT1", "Internal"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_rme9652_get_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme9652->lock);
+ ucontrol->value.enumerated.item[0] = rme9652_adat1_in(rme9652);
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static int snd_rme9652_put_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_rme9652_use_is_exclusive(rme9652))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0] % 2;
+ spin_lock_irq(&rme9652->lock);
+ change = val != rme9652_adat1_in(rme9652);
+ if (change)
+ rme9652_set_adat1_input(rme9652, val);
+ spin_unlock_irq(&rme9652->lock);
+ return change;
+}
+
+#define RME9652_SPDIF_IN(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_rme9652_info_spdif_in, \
+ .get = snd_rme9652_get_spdif_in, .put = snd_rme9652_put_spdif_in }
+
+static unsigned int rme9652_spdif_in(struct snd_rme9652 *rme9652)
+{
+ return rme9652_decode_spdif_in(rme9652->control_register &
+ RME9652_inp);
+}
+
+static int rme9652_set_spdif_input(struct snd_rme9652 *rme9652, int in)
+{
+ int restart = 0;
+
+ rme9652->control_register &= ~RME9652_inp;
+ rme9652->control_register |= rme9652_encode_spdif_in(in);
+
+ if ((restart = rme9652->running)) {
+ rme9652_stop(rme9652);
+ }
+
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+
+ if (restart) {
+ rme9652_start(rme9652);
+ }
+
+ return 0;
+}
+
+static int snd_rme9652_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {"ADAT1", "Coaxial", "Internal"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_rme9652_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme9652->lock);
+ ucontrol->value.enumerated.item[0] = rme9652_spdif_in(rme9652);
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static int snd_rme9652_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_rme9652_use_is_exclusive(rme9652))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0] % 3;
+ spin_lock_irq(&rme9652->lock);
+ change = val != rme9652_spdif_in(rme9652);
+ if (change)
+ rme9652_set_spdif_input(rme9652, val);
+ spin_unlock_irq(&rme9652->lock);
+ return change;
+}
+
+#define RME9652_SPDIF_OUT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_rme9652_info_spdif_out, \
+ .get = snd_rme9652_get_spdif_out, .put = snd_rme9652_put_spdif_out }
+
+static int rme9652_spdif_out(struct snd_rme9652 *rme9652)
+{
+ return (rme9652->control_register & RME9652_opt_out) ? 1 : 0;
+}
+
+static int rme9652_set_spdif_output(struct snd_rme9652 *rme9652, int out)
+{
+ int restart = 0;
+
+ if (out) {
+ rme9652->control_register |= RME9652_opt_out;
+ } else {
+ rme9652->control_register &= ~RME9652_opt_out;
+ }
+
+ if ((restart = rme9652->running)) {
+ rme9652_stop(rme9652);
+ }
+
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+
+ if (restart) {
+ rme9652_start(rme9652);
+ }
+
+ return 0;
+}
+
+#define snd_rme9652_info_spdif_out snd_ctl_boolean_mono_info
+
+static int snd_rme9652_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme9652->lock);
+ ucontrol->value.integer.value[0] = rme9652_spdif_out(rme9652);
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static int snd_rme9652_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_rme9652_use_is_exclusive(rme9652))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&rme9652->lock);
+ change = (int)val != rme9652_spdif_out(rme9652);
+ rme9652_set_spdif_output(rme9652, val);
+ spin_unlock_irq(&rme9652->lock);
+ return change;
+}
+
+#define RME9652_SYNC_MODE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_rme9652_info_sync_mode, \
+ .get = snd_rme9652_get_sync_mode, .put = snd_rme9652_put_sync_mode }
+
+static int rme9652_sync_mode(struct snd_rme9652 *rme9652)
+{
+ if (rme9652->control_register & RME9652_wsel) {
+ return 2;
+ } else if (rme9652->control_register & RME9652_Master) {
+ return 1;
+ } else {
+ return 0;
+ }
+}
+
+static int rme9652_set_sync_mode(struct snd_rme9652 *rme9652, int mode)
+{
+ int restart = 0;
+
+ switch (mode) {
+ case 0:
+ rme9652->control_register &=
+ ~(RME9652_Master | RME9652_wsel);
+ break;
+ case 1:
+ rme9652->control_register =
+ (rme9652->control_register & ~RME9652_wsel) | RME9652_Master;
+ break;
+ case 2:
+ rme9652->control_register |=
+ (RME9652_Master | RME9652_wsel);
+ break;
+ }
+
+ if ((restart = rme9652->running)) {
+ rme9652_stop(rme9652);
+ }
+
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+
+ if (restart) {
+ rme9652_start(rme9652);
+ }
+
+ return 0;
+}
+
+static int snd_rme9652_info_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[3] = {"AutoSync", "Master", "Word Clock"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_rme9652_get_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme9652->lock);
+ ucontrol->value.enumerated.item[0] = rme9652_sync_mode(rme9652);
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static int snd_rme9652_put_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ val = ucontrol->value.enumerated.item[0] % 3;
+ spin_lock_irq(&rme9652->lock);
+ change = (int)val != rme9652_sync_mode(rme9652);
+ rme9652_set_sync_mode(rme9652, val);
+ spin_unlock_irq(&rme9652->lock);
+ return change;
+}
+
+#define RME9652_SYNC_PREF(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_rme9652_info_sync_pref, \
+ .get = snd_rme9652_get_sync_pref, .put = snd_rme9652_put_sync_pref }
+
+static int rme9652_sync_pref(struct snd_rme9652 *rme9652)
+{
+ switch (rme9652->control_register & RME9652_SyncPref_Mask) {
+ case RME9652_SyncPref_ADAT1:
+ return RME9652_SYNC_FROM_ADAT1;
+ case RME9652_SyncPref_ADAT2:
+ return RME9652_SYNC_FROM_ADAT2;
+ case RME9652_SyncPref_ADAT3:
+ return RME9652_SYNC_FROM_ADAT3;
+ case RME9652_SyncPref_SPDIF:
+ return RME9652_SYNC_FROM_SPDIF;
+ }
+ /* Not reachable */
+ return 0;
+}
+
+static int rme9652_set_sync_pref(struct snd_rme9652 *rme9652, int pref)
+{
+ int restart;
+
+ rme9652->control_register &= ~RME9652_SyncPref_Mask;
+ switch (pref) {
+ case RME9652_SYNC_FROM_ADAT1:
+ rme9652->control_register |= RME9652_SyncPref_ADAT1;
+ break;
+ case RME9652_SYNC_FROM_ADAT2:
+ rme9652->control_register |= RME9652_SyncPref_ADAT2;
+ break;
+ case RME9652_SYNC_FROM_ADAT3:
+ rme9652->control_register |= RME9652_SyncPref_ADAT3;
+ break;
+ case RME9652_SYNC_FROM_SPDIF:
+ rme9652->control_register |= RME9652_SyncPref_SPDIF;
+ break;
+ }
+
+ if ((restart = rme9652->running)) {
+ rme9652_stop(rme9652);
+ }
+
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+
+ if (restart) {
+ rme9652_start(rme9652);
+ }
+
+ return 0;
+}
+
+static int snd_rme9652_info_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {"IEC958 In", "ADAT1 In", "ADAT2 In", "ADAT3 In"};
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = rme9652->ss_channels == RME9652_NCHANNELS ? 4 : 3;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_rme9652_get_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme9652->lock);
+ ucontrol->value.enumerated.item[0] = rme9652_sync_pref(rme9652);
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static int snd_rme9652_put_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change, max;
+ unsigned int val;
+
+ if (!snd_rme9652_use_is_exclusive(rme9652))
+ return -EBUSY;
+ max = rme9652->ss_channels == RME9652_NCHANNELS ? 4 : 3;
+ val = ucontrol->value.enumerated.item[0] % max;
+ spin_lock_irq(&rme9652->lock);
+ change = (int)val != rme9652_sync_pref(rme9652);
+ rme9652_set_sync_pref(rme9652, val);
+ spin_unlock_irq(&rme9652->lock);
+ return change;
+}
+
+static int snd_rme9652_info_thru(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = rme9652->ss_channels;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_rme9652_get_thru(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ unsigned int k;
+ u32 thru_bits = rme9652->thru_bits;
+
+ for (k = 0; k < rme9652->ss_channels; ++k) {
+ ucontrol->value.integer.value[k] = !!(thru_bits & (1 << k));
+ }
+ return 0;
+}
+
+static int snd_rme9652_put_thru(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int chn;
+ u32 thru_bits = 0;
+
+ if (!snd_rme9652_use_is_exclusive(rme9652))
+ return -EBUSY;
+
+ for (chn = 0; chn < rme9652->ss_channels; ++chn) {
+ if (ucontrol->value.integer.value[chn])
+ thru_bits |= 1 << chn;
+ }
+
+ spin_lock_irq(&rme9652->lock);
+ change = thru_bits ^ rme9652->thru_bits;
+ if (change) {
+ for (chn = 0; chn < rme9652->ss_channels; ++chn) {
+ if (!(change & (1 << chn)))
+ continue;
+ rme9652_set_thru(rme9652,chn,thru_bits&(1<<chn));
+ }
+ }
+ spin_unlock_irq(&rme9652->lock);
+ return !!change;
+}
+
+#define RME9652_PASSTHRU(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_rme9652_info_passthru, \
+ .put = snd_rme9652_put_passthru, \
+ .get = snd_rme9652_get_passthru }
+
+#define snd_rme9652_info_passthru snd_ctl_boolean_mono_info
+
+static int snd_rme9652_get_passthru(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme9652->lock);
+ ucontrol->value.integer.value[0] = rme9652->passthru;
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static int snd_rme9652_put_passthru(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+ int err = 0;
+
+ if (!snd_rme9652_use_is_exclusive(rme9652))
+ return -EBUSY;
+
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&rme9652->lock);
+ change = (ucontrol->value.integer.value[0] != rme9652->passthru);
+ if (change)
+ err = rme9652_set_passthru(rme9652, val);
+ spin_unlock_irq(&rme9652->lock);
+ return err ? err : change;
+}
+
+/* Read-only switches */
+
+#define RME9652_SPDIF_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_rme9652_info_spdif_rate, \
+ .get = snd_rme9652_get_spdif_rate }
+
+static int snd_rme9652_info_spdif_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 96000;
+ return 0;
+}
+
+static int snd_rme9652_get_spdif_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&rme9652->lock);
+ ucontrol->value.integer.value[0] = rme9652_spdif_sample_rate(rme9652);
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+#define RME9652_ADAT_SYNC(xname, xindex, xidx) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_rme9652_info_adat_sync, \
+ .get = snd_rme9652_get_adat_sync, .private_value = xidx }
+
+static int snd_rme9652_info_adat_sync(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {"No Lock", "Lock", "No Lock Sync", "Lock Sync"};
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_rme9652_get_adat_sync(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+ unsigned int mask1, mask2, val;
+
+ switch (kcontrol->private_value) {
+ case 0: mask1 = RME9652_lock_0; mask2 = RME9652_sync_0; break;
+ case 1: mask1 = RME9652_lock_1; mask2 = RME9652_sync_1; break;
+ case 2: mask1 = RME9652_lock_2; mask2 = RME9652_sync_2; break;
+ default: return -EINVAL;
+ }
+ val = rme9652_read(rme9652, RME9652_status_register);
+ ucontrol->value.enumerated.item[0] = (val & mask1) ? 1 : 0;
+ ucontrol->value.enumerated.item[0] |= (val & mask2) ? 2 : 0;
+ return 0;
+}
+
+#define RME9652_TC_VALID(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_rme9652_info_tc_valid, \
+ .get = snd_rme9652_get_tc_valid }
+
+#define snd_rme9652_info_tc_valid snd_ctl_boolean_mono_info
+
+static int snd_rme9652_get_tc_valid(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] =
+ (rme9652_read(rme9652, RME9652_status_register) & RME9652_tc_valid) ? 1 : 0;
+ return 0;
+}
+
+#ifdef ALSA_HAS_STANDARD_WAY_OF_RETURNING_TIMECODE
+
+/* FIXME: this routine needs a port to the new control API --jk */
+
+static int snd_rme9652_get_tc_value(void *private_data,
+ snd_kswitch_t *kswitch,
+ snd_switch_t *uswitch)
+{
+ struct snd_rme9652 *s = (struct snd_rme9652 *) private_data;
+ u32 value;
+ int i;
+
+ uswitch->type = SNDRV_SW_TYPE_DWORD;
+
+ if ((rme9652_read(s, RME9652_status_register) &
+ RME9652_tc_valid) == 0) {
+ uswitch->value.data32[0] = 0;
+ return 0;
+ }
+
+ /* timecode request */
+
+ rme9652_write(s, RME9652_time_code, 0);
+
+ /* XXX bug alert: loop-based timing !!!! */
+
+ for (i = 0; i < 50; i++) {
+ if (!(rme9652_read(s, i * 4) & RME9652_tc_busy))
+ break;
+ }
+
+ if (!(rme9652_read(s, i * 4) & RME9652_tc_busy)) {
+ return -EIO;
+ }
+
+ value = 0;
+
+ for (i = 0; i < 32; i++) {
+ value >>= 1;
+
+ if (rme9652_read(s, i * 4) & RME9652_tc_out)
+ value |= 0x80000000;
+ }
+
+ if (value > 2 * 60 * 48000) {
+ value -= 2 * 60 * 48000;
+ } else {
+ value = 0;
+ }
+
+ uswitch->value.data32[0] = value;
+
+ return 0;
+}
+
+#endif /* ALSA_HAS_STANDARD_WAY_OF_RETURNING_TIMECODE */
+
+static struct snd_kcontrol_new snd_rme9652_controls[] = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_rme9652_control_spdif_info,
+ .get = snd_rme9652_control_spdif_get,
+ .put = snd_rme9652_control_spdif_put,
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_rme9652_control_spdif_stream_info,
+ .get = snd_rme9652_control_spdif_stream_get,
+ .put = snd_rme9652_control_spdif_stream_put,
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_rme9652_control_spdif_mask_info,
+ .get = snd_rme9652_control_spdif_mask_get,
+ .private_value = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_CON_EMPHASIS,
+},
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+ .info = snd_rme9652_control_spdif_mask_info,
+ .get = snd_rme9652_control_spdif_mask_get,
+ .private_value = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_PROFESSIONAL |
+ IEC958_AES0_PRO_EMPHASIS,
+},
+RME9652_SPDIF_IN("IEC958 Input Connector", 0),
+RME9652_SPDIF_OUT("IEC958 Output also on ADAT1", 0),
+RME9652_SYNC_MODE("Sync Mode", 0),
+RME9652_SYNC_PREF("Preferred Sync Source", 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channels Thru",
+ .index = 0,
+ .info = snd_rme9652_info_thru,
+ .get = snd_rme9652_get_thru,
+ .put = snd_rme9652_put_thru,
+},
+RME9652_SPDIF_RATE("IEC958 Sample Rate", 0),
+RME9652_ADAT_SYNC("ADAT1 Sync Check", 0, 0),
+RME9652_ADAT_SYNC("ADAT2 Sync Check", 0, 1),
+RME9652_TC_VALID("Timecode Valid", 0),
+RME9652_PASSTHRU("Passthru", 0)
+};
+
+static struct snd_kcontrol_new snd_rme9652_adat3_check =
+RME9652_ADAT_SYNC("ADAT3 Sync Check", 0, 2);
+
+static struct snd_kcontrol_new snd_rme9652_adat1_input =
+RME9652_ADAT1_IN("ADAT1 Input Source", 0);
+
+static int snd_rme9652_create_controls(struct snd_card *card, struct snd_rme9652 *rme9652)
+{
+ unsigned int idx;
+ int err;
+ struct snd_kcontrol *kctl;
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_rme9652_controls); idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_rme9652_controls[idx], rme9652))) < 0)
+ return err;
+ if (idx == 1) /* IEC958 (S/PDIF) Stream */
+ rme9652->spdif_ctl = kctl;
+ }
+
+ if (rme9652->ss_channels == RME9652_NCHANNELS)
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_rme9652_adat3_check, rme9652))) < 0)
+ return err;
+
+ if (rme9652->hw_rev >= 15)
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_rme9652_adat1_input, rme9652))) < 0)
+ return err;
+
+ return 0;
+}
+
+/*------------------------------------------------------------
+ /proc interface
+ ------------------------------------------------------------*/
+
+static void
+snd_rme9652_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_rme9652 *rme9652 = (struct snd_rme9652 *) entry->private_data;
+ u32 thru_bits = rme9652->thru_bits;
+ int show_auto_sync_source = 0;
+ int i;
+ unsigned int status;
+ int x;
+
+ status = rme9652_read(rme9652, RME9652_status_register);
+
+ snd_iprintf(buffer, "%s (Card #%d)\n", rme9652->card_name, rme9652->card->number + 1);
+ snd_iprintf(buffer, "Buffers: capture %p playback %p\n",
+ rme9652->capture_buffer, rme9652->playback_buffer);
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ rme9652->irq, rme9652->port, (unsigned long)rme9652->iobase);
+ snd_iprintf(buffer, "Control register: %x\n", rme9652->control_register);
+
+ snd_iprintf(buffer, "\n");
+
+ x = 1 << (6 + rme9652_decode_latency(rme9652->control_register &
+ RME9652_latency));
+
+ snd_iprintf(buffer, "Latency: %d samples (2 periods of %lu bytes)\n",
+ x, (unsigned long) rme9652->period_bytes);
+ snd_iprintf(buffer, "Hardware pointer (frames): %ld\n",
+ rme9652_hw_pointer(rme9652));
+ snd_iprintf(buffer, "Passthru: %s\n",
+ rme9652->passthru ? "yes" : "no");
+
+ if ((rme9652->control_register & (RME9652_Master | RME9652_wsel)) == 0) {
+ snd_iprintf(buffer, "Clock mode: autosync\n");
+ show_auto_sync_source = 1;
+ } else if (rme9652->control_register & RME9652_wsel) {
+ if (status & RME9652_wsel_rd) {
+ snd_iprintf(buffer, "Clock mode: word clock\n");
+ } else {
+ snd_iprintf(buffer, "Clock mode: word clock (no signal)\n");
+ }
+ } else {
+ snd_iprintf(buffer, "Clock mode: master\n");
+ }
+
+ if (show_auto_sync_source) {
+ switch (rme9652->control_register & RME9652_SyncPref_Mask) {
+ case RME9652_SyncPref_ADAT1:
+ snd_iprintf(buffer, "Pref. sync source: ADAT1\n");
+ break;
+ case RME9652_SyncPref_ADAT2:
+ snd_iprintf(buffer, "Pref. sync source: ADAT2\n");
+ break;
+ case RME9652_SyncPref_ADAT3:
+ snd_iprintf(buffer, "Pref. sync source: ADAT3\n");
+ break;
+ case RME9652_SyncPref_SPDIF:
+ snd_iprintf(buffer, "Pref. sync source: IEC958\n");
+ break;
+ default:
+ snd_iprintf(buffer, "Pref. sync source: ???\n");
+ }
+ }
+
+ if (rme9652->hw_rev >= 15)
+ snd_iprintf(buffer, "\nADAT1 Input source: %s\n",
+ (rme9652->control_register & RME9652_ADAT1_INTERNAL) ?
+ "Internal" : "ADAT1 optical");
+
+ snd_iprintf(buffer, "\n");
+
+ switch (rme9652_decode_spdif_in(rme9652->control_register &
+ RME9652_inp)) {
+ case RME9652_SPDIFIN_OPTICAL:
+ snd_iprintf(buffer, "IEC958 input: ADAT1\n");
+ break;
+ case RME9652_SPDIFIN_COAXIAL:
+ snd_iprintf(buffer, "IEC958 input: Coaxial\n");
+ break;
+ case RME9652_SPDIFIN_INTERN:
+ snd_iprintf(buffer, "IEC958 input: Internal\n");
+ break;
+ default:
+ snd_iprintf(buffer, "IEC958 input: ???\n");
+ break;
+ }
+
+ if (rme9652->control_register & RME9652_opt_out) {
+ snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n");
+ } else {
+ snd_iprintf(buffer, "IEC958 output: Coaxial only\n");
+ }
+
+ if (rme9652->control_register & RME9652_PRO) {
+ snd_iprintf(buffer, "IEC958 quality: Professional\n");
+ } else {
+ snd_iprintf(buffer, "IEC958 quality: Consumer\n");
+ }
+
+ if (rme9652->control_register & RME9652_EMP) {
+ snd_iprintf(buffer, "IEC958 emphasis: on\n");
+ } else {
+ snd_iprintf(buffer, "IEC958 emphasis: off\n");
+ }
+
+ if (rme9652->control_register & RME9652_Dolby) {
+ snd_iprintf(buffer, "IEC958 Dolby: on\n");
+ } else {
+ snd_iprintf(buffer, "IEC958 Dolby: off\n");
+ }
+
+ i = rme9652_spdif_sample_rate(rme9652);
+
+ if (i < 0) {
+ snd_iprintf(buffer,
+ "IEC958 sample rate: error flag set\n");
+ } else if (i == 0) {
+ snd_iprintf(buffer, "IEC958 sample rate: undetermined\n");
+ } else {
+ snd_iprintf(buffer, "IEC958 sample rate: %d\n", i);
+ }
+
+ snd_iprintf(buffer, "\n");
+
+ snd_iprintf(buffer, "ADAT Sample rate: %dHz\n",
+ rme9652_adat_sample_rate(rme9652));
+
+ /* Sync Check */
+
+ x = status & RME9652_sync_0;
+ if (status & RME9652_lock_0) {
+ snd_iprintf(buffer, "ADAT1: %s\n", x ? "Sync" : "Lock");
+ } else {
+ snd_iprintf(buffer, "ADAT1: No Lock\n");
+ }
+
+ x = status & RME9652_sync_1;
+ if (status & RME9652_lock_1) {
+ snd_iprintf(buffer, "ADAT2: %s\n", x ? "Sync" : "Lock");
+ } else {
+ snd_iprintf(buffer, "ADAT2: No Lock\n");
+ }
+
+ x = status & RME9652_sync_2;
+ if (status & RME9652_lock_2) {
+ snd_iprintf(buffer, "ADAT3: %s\n", x ? "Sync" : "Lock");
+ } else {
+ snd_iprintf(buffer, "ADAT3: No Lock\n");
+ }
+
+ snd_iprintf(buffer, "\n");
+
+ snd_iprintf(buffer, "Timecode signal: %s\n",
+ (status & RME9652_tc_valid) ? "yes" : "no");
+
+ /* thru modes */
+
+ snd_iprintf(buffer, "Punch Status:\n\n");
+
+ for (i = 0; i < rme9652->ss_channels; i++) {
+ if (thru_bits & (1 << i)) {
+ snd_iprintf(buffer, "%2d: on ", i + 1);
+ } else {
+ snd_iprintf(buffer, "%2d: off ", i + 1);
+ }
+
+ if (((i + 1) % 8) == 0) {
+ snd_iprintf(buffer, "\n");
+ }
+ }
+
+ snd_iprintf(buffer, "\n");
+}
+
+static void __devinit snd_rme9652_proc_init(struct snd_rme9652 *rme9652)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(rme9652->card, "rme9652", &entry))
+ snd_info_set_text_ops(entry, rme9652, snd_rme9652_proc_read);
+}
+
+static void snd_rme9652_free_buffers(struct snd_rme9652 *rme9652)
+{
+ snd_hammerfall_free_buffer(&rme9652->capture_dma_buf, rme9652->pci);
+ snd_hammerfall_free_buffer(&rme9652->playback_dma_buf, rme9652->pci);
+}
+
+static int snd_rme9652_free(struct snd_rme9652 *rme9652)
+{
+ if (rme9652->irq >= 0)
+ rme9652_stop(rme9652);
+ snd_rme9652_free_buffers(rme9652);
+
+ if (rme9652->irq >= 0)
+ free_irq(rme9652->irq, (void *)rme9652);
+ if (rme9652->iobase)
+ iounmap(rme9652->iobase);
+ if (rme9652->port)
+ pci_release_regions(rme9652->pci);
+
+ pci_disable_device(rme9652->pci);
+ return 0;
+}
+
+static int __devinit snd_rme9652_initialize_memory(struct snd_rme9652 *rme9652)
+{
+ unsigned long pb_bus, cb_bus;
+
+ if (snd_hammerfall_get_buffer(rme9652->pci, &rme9652->capture_dma_buf, RME9652_DMA_AREA_BYTES) < 0 ||
+ snd_hammerfall_get_buffer(rme9652->pci, &rme9652->playback_dma_buf, RME9652_DMA_AREA_BYTES) < 0) {
+ if (rme9652->capture_dma_buf.area)
+ snd_dma_free_pages(&rme9652->capture_dma_buf);
+ printk(KERN_ERR "%s: no buffers available\n", rme9652->card_name);
+ return -ENOMEM;
+ }
+
+ /* Align to bus-space 64K boundary */
+
+ cb_bus = ALIGN(rme9652->capture_dma_buf.addr, 0x10000ul);
+ pb_bus = ALIGN(rme9652->playback_dma_buf.addr, 0x10000ul);
+
+ /* Tell the card where it is */
+
+ rme9652_write(rme9652, RME9652_rec_buffer, cb_bus);
+ rme9652_write(rme9652, RME9652_play_buffer, pb_bus);
+
+ rme9652->capture_buffer = rme9652->capture_dma_buf.area + (cb_bus - rme9652->capture_dma_buf.addr);
+ rme9652->playback_buffer = rme9652->playback_dma_buf.area + (pb_bus - rme9652->playback_dma_buf.addr);
+
+ return 0;
+}
+
+static void snd_rme9652_set_defaults(struct snd_rme9652 *rme9652)
+{
+ unsigned int k;
+
+ /* ASSUMPTION: rme9652->lock is either held, or
+ there is no need to hold it (e.g. during module
+ initialization).
+ */
+
+ /* set defaults:
+
+ SPDIF Input via Coax
+ autosync clock mode
+ maximum latency (7 = 8192 samples, 64Kbyte buffer,
+ which implies 2 4096 sample, 32Kbyte periods).
+
+ if rev 1.5, initialize the S/PDIF receiver.
+
+ */
+
+ rme9652->control_register =
+ RME9652_inp_0 | rme9652_encode_latency(7);
+
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register);
+
+ rme9652_reset_hw_pointer(rme9652);
+ rme9652_compute_period_size(rme9652);
+
+ /* default: thru off for all channels */
+
+ for (k = 0; k < RME9652_NCHANNELS; ++k)
+ rme9652_write(rme9652, RME9652_thru_base + k * 4, 0);
+
+ rme9652->thru_bits = 0;
+ rme9652->passthru = 0;
+
+ /* set a default rate so that the channel map is set up */
+
+ rme9652_set_rate(rme9652, 48000);
+}
+
+static irqreturn_t snd_rme9652_interrupt(int irq, void *dev_id)
+{
+ struct snd_rme9652 *rme9652 = (struct snd_rme9652 *) dev_id;
+
+ if (!(rme9652_read(rme9652, RME9652_status_register) & RME9652_IRQ)) {
+ return IRQ_NONE;
+ }
+
+ rme9652_write(rme9652, RME9652_irq_clear, 0);
+
+ if (rme9652->capture_substream) {
+ snd_pcm_period_elapsed(rme9652->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
+ }
+
+ if (rme9652->playback_substream) {
+ snd_pcm_period_elapsed(rme9652->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream);
+ }
+ return IRQ_HANDLED;
+}
+
+static snd_pcm_uframes_t snd_rme9652_hw_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ return rme9652_hw_pointer(rme9652);
+}
+
+static char *rme9652_channel_buffer_location(struct snd_rme9652 *rme9652,
+ int stream,
+ int channel)
+
+{
+ int mapped_channel;
+
+ if (snd_BUG_ON(channel < 0 || channel >= RME9652_NCHANNELS))
+ return NULL;
+
+ if ((mapped_channel = rme9652->channel_map[channel]) < 0) {
+ return NULL;
+ }
+
+ if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ return rme9652->capture_buffer +
+ (mapped_channel * RME9652_CHANNEL_BUFFER_BYTES);
+ } else {
+ return rme9652->playback_buffer +
+ (mapped_channel * RME9652_CHANNEL_BUFFER_BYTES);
+ }
+}
+
+static int snd_rme9652_playback_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ if (snd_BUG_ON(pos + count > RME9652_CHANNEL_BUFFER_BYTES / 4))
+ return -EINVAL;
+
+ channel_buf = rme9652_channel_buffer_location (rme9652,
+ substream->pstr->stream,
+ channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ if (copy_from_user(channel_buf + pos * 4, src, count * 4))
+ return -EFAULT;
+ return count;
+}
+
+static int snd_rme9652_capture_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, void __user *dst, snd_pcm_uframes_t count)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ if (snd_BUG_ON(pos + count > RME9652_CHANNEL_BUFFER_BYTES / 4))
+ return -EINVAL;
+
+ channel_buf = rme9652_channel_buffer_location (rme9652,
+ substream->pstr->stream,
+ channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ if (copy_to_user(dst, channel_buf + pos * 4, count * 4))
+ return -EFAULT;
+ return count;
+}
+
+static int snd_rme9652_hw_silence(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, snd_pcm_uframes_t count)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ channel_buf = rme9652_channel_buffer_location (rme9652,
+ substream->pstr->stream,
+ channel);
+ if (snd_BUG_ON(!channel_buf))
+ return -EIO;
+ memset(channel_buf + pos * 4, 0, count * 4);
+ return count;
+}
+
+static int snd_rme9652_reset(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *other;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = rme9652->capture_substream;
+ else
+ other = rme9652->playback_substream;
+ if (rme9652->running)
+ runtime->status->hw_ptr = rme9652_hw_pointer(rme9652);
+ else
+ runtime->status->hw_ptr = 0;
+ if (other) {
+ struct snd_pcm_substream *s;
+ struct snd_pcm_runtime *oruntime = other->runtime;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == other) {
+ oruntime->status->hw_ptr = runtime->status->hw_ptr;
+ break;
+ }
+ }
+ }
+ return 0;
+}
+
+static int snd_rme9652_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ int err;
+ pid_t this_pid;
+ pid_t other_pid;
+
+ spin_lock_irq(&rme9652->lock);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ rme9652->control_register &= ~(RME9652_PRO | RME9652_Dolby | RME9652_EMP);
+ rme9652_write(rme9652, RME9652_control_register, rme9652->control_register |= rme9652->creg_spdif_stream);
+ this_pid = rme9652->playback_pid;
+ other_pid = rme9652->capture_pid;
+ } else {
+ this_pid = rme9652->capture_pid;
+ other_pid = rme9652->playback_pid;
+ }
+
+ if ((other_pid > 0) && (this_pid != other_pid)) {
+
+ /* The other stream is open, and not by the same
+ task as this one. Make sure that the parameters
+ that matter are the same.
+ */
+
+ if ((int)params_rate(params) !=
+ rme9652_adat_sample_rate(rme9652)) {
+ spin_unlock_irq(&rme9652->lock);
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE);
+ return -EBUSY;
+ }
+
+ if (params_period_size(params) != rme9652->period_bytes / 4) {
+ spin_unlock_irq(&rme9652->lock);
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return -EBUSY;
+ }
+
+ /* We're fine. */
+
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+
+ } else {
+ spin_unlock_irq(&rme9652->lock);
+ }
+
+ /* how to make sure that the rate matches an externally-set one ?
+ */
+
+ if ((err = rme9652_set_rate(rme9652, params_rate(params))) < 0) {
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_RATE);
+ return err;
+ }
+
+ if ((err = rme9652_set_interrupt_interval(rme9652, params_period_size(params))) < 0) {
+ _snd_pcm_hw_param_setempty(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return err;
+ }
+
+ return 0;
+}
+
+static int snd_rme9652_channel_info(struct snd_pcm_substream *substream,
+ struct snd_pcm_channel_info *info)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ int chn;
+
+ if (snd_BUG_ON(info->channel >= RME9652_NCHANNELS))
+ return -EINVAL;
+
+ if ((chn = rme9652->channel_map[info->channel]) < 0) {
+ return -EINVAL;
+ }
+
+ info->offset = chn * RME9652_CHANNEL_BUFFER_BYTES;
+ info->first = 0;
+ info->step = 32;
+ return 0;
+}
+
+static int snd_rme9652_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL1_RESET:
+ {
+ return snd_rme9652_reset(substream);
+ }
+ case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
+ {
+ struct snd_pcm_channel_info *info = arg;
+ return snd_rme9652_channel_info(substream, info);
+ }
+ default:
+ break;
+ }
+
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static void rme9652_silence_playback(struct snd_rme9652 *rme9652)
+{
+ memset(rme9652->playback_buffer, 0, RME9652_DMA_AREA_BYTES);
+}
+
+static int snd_rme9652_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *other;
+ int running;
+ spin_lock(&rme9652->lock);
+ running = rme9652->running;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ running |= 1 << substream->stream;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ running &= ~(1 << substream->stream);
+ break;
+ default:
+ snd_BUG();
+ spin_unlock(&rme9652->lock);
+ return -EINVAL;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = rme9652->capture_substream;
+ else
+ other = rme9652->playback_substream;
+
+ if (other) {
+ struct snd_pcm_substream *s;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == other) {
+ snd_pcm_trigger_done(s, substream);
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ running |= 1 << s->stream;
+ else
+ running &= ~(1 << s->stream);
+ goto _ok;
+ }
+ }
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ if (!(running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) &&
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ rme9652_silence_playback(rme9652);
+ } else {
+ if (running &&
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ rme9652_silence_playback(rme9652);
+ }
+ } else {
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ rme9652_silence_playback(rme9652);
+ }
+ _ok:
+ snd_pcm_trigger_done(substream, substream);
+ if (!rme9652->running && running)
+ rme9652_start(rme9652);
+ else if (rme9652->running && !running)
+ rme9652_stop(rme9652);
+ rme9652->running = running;
+ spin_unlock(&rme9652->lock);
+
+ return 0;
+}
+
+static int snd_rme9652_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ unsigned long flags;
+ int result = 0;
+
+ spin_lock_irqsave(&rme9652->lock, flags);
+ if (!rme9652->running)
+ rme9652_reset_hw_pointer(rme9652);
+ spin_unlock_irqrestore(&rme9652->lock, flags);
+ return result;
+}
+
+static struct snd_pcm_hardware snd_rme9652_playback_subinfo =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_DOUBLE),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 26,
+ .buffer_bytes_max = RME9652_CHANNEL_BUFFER_BYTES * 26,
+ .period_bytes_min = (64 * 4) * 10,
+ .period_bytes_max = (8192 * 4) * 26,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_rme9652_capture_subinfo =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 26,
+ .buffer_bytes_max = RME9652_CHANNEL_BUFFER_BYTES *26,
+ .period_bytes_min = (64 * 4) * 10,
+ .period_bytes_max = (8192 * 4) * 26,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static unsigned int period_sizes[] = { 64, 128, 256, 512, 1024, 2048, 4096, 8192 };
+
+static struct snd_pcm_hw_constraint_list hw_constraints_period_sizes = {
+ .count = ARRAY_SIZE(period_sizes),
+ .list = period_sizes,
+ .mask = 0
+};
+
+static int snd_rme9652_hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_rme9652 *rme9652 = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ unsigned int list[2] = { rme9652->ds_channels, rme9652->ss_channels };
+ return snd_interval_list(c, 2, list, 0);
+}
+
+static int snd_rme9652_hw_rule_channels_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_rme9652 *rme9652 = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (r->min > 48000) {
+ struct snd_interval t = {
+ .min = rme9652->ds_channels,
+ .max = rme9652->ds_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ } else if (r->max < 88200) {
+ struct snd_interval t = {
+ .min = rme9652->ss_channels,
+ .max = rme9652->ss_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ }
+ return 0;
+}
+
+static int snd_rme9652_hw_rule_rate_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_rme9652 *rme9652 = rule->private;
+ struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ if (c->min >= rme9652->ss_channels) {
+ struct snd_interval t = {
+ .min = 44100,
+ .max = 48000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ } else if (c->max <= rme9652->ds_channels) {
+ struct snd_interval t = {
+ .min = 88200,
+ .max = 96000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ }
+ return 0;
+}
+
+static int snd_rme9652_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irq(&rme9652->lock);
+
+ snd_pcm_set_sync(substream);
+
+ runtime->hw = snd_rme9652_playback_subinfo;
+ runtime->dma_area = rme9652->playback_buffer;
+ runtime->dma_bytes = RME9652_DMA_AREA_BYTES;
+
+ if (rme9652->capture_substream == NULL) {
+ rme9652_stop(rme9652);
+ rme9652_set_thru(rme9652, -1, 0);
+ }
+
+ rme9652->playback_pid = current->pid;
+ rme9652->playback_substream = substream;
+
+ spin_unlock_irq(&rme9652->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_period_sizes);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_rme9652_hw_rule_channels, rme9652,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_rme9652_hw_rule_channels_rate, rme9652,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_rme9652_hw_rule_rate_channels, rme9652,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+
+ rme9652->creg_spdif_stream = rme9652->creg_spdif;
+ rme9652->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(rme9652->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &rme9652->spdif_ctl->id);
+ return 0;
+}
+
+static int snd_rme9652_playback_release(struct snd_pcm_substream *substream)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme9652->lock);
+
+ rme9652->playback_pid = -1;
+ rme9652->playback_substream = NULL;
+
+ spin_unlock_irq(&rme9652->lock);
+
+ rme9652->spdif_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(rme9652->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &rme9652->spdif_ctl->id);
+ return 0;
+}
+
+
+static int snd_rme9652_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ spin_lock_irq(&rme9652->lock);
+
+ snd_pcm_set_sync(substream);
+
+ runtime->hw = snd_rme9652_capture_subinfo;
+ runtime->dma_area = rme9652->capture_buffer;
+ runtime->dma_bytes = RME9652_DMA_AREA_BYTES;
+
+ if (rme9652->playback_substream == NULL) {
+ rme9652_stop(rme9652);
+ rme9652_set_thru(rme9652, -1, 0);
+ }
+
+ rme9652->capture_pid = current->pid;
+ rme9652->capture_substream = substream;
+
+ spin_unlock_irq(&rme9652->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_period_sizes);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_rme9652_hw_rule_channels, rme9652,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_rme9652_hw_rule_channels_rate, rme9652,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_rme9652_hw_rule_rate_channels, rme9652,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ return 0;
+}
+
+static int snd_rme9652_capture_release(struct snd_pcm_substream *substream)
+{
+ struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&rme9652->lock);
+
+ rme9652->capture_pid = -1;
+ rme9652->capture_substream = NULL;
+
+ spin_unlock_irq(&rme9652->lock);
+ return 0;
+}
+
+static struct snd_pcm_ops snd_rme9652_playback_ops = {
+ .open = snd_rme9652_playback_open,
+ .close = snd_rme9652_playback_release,
+ .ioctl = snd_rme9652_ioctl,
+ .hw_params = snd_rme9652_hw_params,
+ .prepare = snd_rme9652_prepare,
+ .trigger = snd_rme9652_trigger,
+ .pointer = snd_rme9652_hw_pointer,
+ .copy = snd_rme9652_playback_copy,
+ .silence = snd_rme9652_hw_silence,
+};
+
+static struct snd_pcm_ops snd_rme9652_capture_ops = {
+ .open = snd_rme9652_capture_open,
+ .close = snd_rme9652_capture_release,
+ .ioctl = snd_rme9652_ioctl,
+ .hw_params = snd_rme9652_hw_params,
+ .prepare = snd_rme9652_prepare,
+ .trigger = snd_rme9652_trigger,
+ .pointer = snd_rme9652_hw_pointer,
+ .copy = snd_rme9652_capture_copy,
+};
+
+static int __devinit snd_rme9652_create_pcm(struct snd_card *card,
+ struct snd_rme9652 *rme9652)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(card,
+ rme9652->card_name,
+ 0, 1, 1, &pcm)) < 0) {
+ return err;
+ }
+
+ rme9652->pcm = pcm;
+ pcm->private_data = rme9652;
+ strcpy(pcm->name, rme9652->card_name);
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_rme9652_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme9652_capture_ops);
+
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+
+ return 0;
+}
+
+static int __devinit snd_rme9652_create(struct snd_card *card,
+ struct snd_rme9652 *rme9652,
+ int precise_ptr)
+{
+ struct pci_dev *pci = rme9652->pci;
+ int err;
+ int status;
+ unsigned short rev;
+
+ rme9652->irq = -1;
+ rme9652->card = card;
+
+ pci_read_config_word(rme9652->pci, PCI_CLASS_REVISION, &rev);
+
+ switch (rev & 0xff) {
+ case 3:
+ case 4:
+ case 8:
+ case 9:
+ break;
+
+ default:
+ /* who knows? */
+ return -ENODEV;
+ }
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ spin_lock_init(&rme9652->lock);
+
+ if ((err = pci_request_regions(pci, "rme9652")) < 0)
+ return err;
+ rme9652->port = pci_resource_start(pci, 0);
+ rme9652->iobase = ioremap_nocache(rme9652->port, RME9652_IO_EXTENT);
+ if (rme9652->iobase == NULL) {
+ snd_printk(KERN_ERR "unable to remap region 0x%lx-0x%lx\n", rme9652->port, rme9652->port + RME9652_IO_EXTENT - 1);
+ return -EBUSY;
+ }
+
+ if (request_irq(pci->irq, snd_rme9652_interrupt, IRQF_SHARED,
+ "rme9652", rme9652)) {
+ snd_printk(KERN_ERR "unable to request IRQ %d\n", pci->irq);
+ return -EBUSY;
+ }
+ rme9652->irq = pci->irq;
+ rme9652->precise_ptr = precise_ptr;
+
+ /* Determine the h/w rev level of the card. This seems like
+ a particularly kludgy way to encode it, but its what RME
+ chose to do, so we follow them ...
+ */
+
+ status = rme9652_read(rme9652, RME9652_status_register);
+ if (rme9652_decode_spdif_rate(status&RME9652_F) == 1) {
+ rme9652->hw_rev = 15;
+ } else {
+ rme9652->hw_rev = 11;
+ }
+
+ /* Differentiate between the standard Hammerfall, and the
+ "Light", which does not have the expansion board. This
+ method comes from information received from Mathhias
+ Clausen at RME. Display the EEPROM and h/w revID where
+ relevant.
+ */
+
+ switch (rev) {
+ case 8: /* original eprom */
+ strcpy(card->driver, "RME9636");
+ if (rme9652->hw_rev == 15) {
+ rme9652->card_name = "RME Digi9636 (Rev 1.5)";
+ } else {
+ rme9652->card_name = "RME Digi9636";
+ }
+ rme9652->ss_channels = RME9636_NCHANNELS;
+ break;
+ case 9: /* W36_G EPROM */
+ strcpy(card->driver, "RME9636");
+ rme9652->card_name = "RME Digi9636 (Rev G)";
+ rme9652->ss_channels = RME9636_NCHANNELS;
+ break;
+ case 4: /* W52_G EPROM */
+ strcpy(card->driver, "RME9652");
+ rme9652->card_name = "RME Digi9652 (Rev G)";
+ rme9652->ss_channels = RME9652_NCHANNELS;
+ break;
+ case 3: /* original eprom */
+ strcpy(card->driver, "RME9652");
+ if (rme9652->hw_rev == 15) {
+ rme9652->card_name = "RME Digi9652 (Rev 1.5)";
+ } else {
+ rme9652->card_name = "RME Digi9652";
+ }
+ rme9652->ss_channels = RME9652_NCHANNELS;
+ break;
+ }
+
+ rme9652->ds_channels = (rme9652->ss_channels - 2) / 2 + 2;
+
+ pci_set_master(rme9652->pci);
+
+ if ((err = snd_rme9652_initialize_memory(rme9652)) < 0) {
+ return err;
+ }
+
+ if ((err = snd_rme9652_create_pcm(card, rme9652)) < 0) {
+ return err;
+ }
+
+ if ((err = snd_rme9652_create_controls(card, rme9652)) < 0) {
+ return err;
+ }
+
+ snd_rme9652_proc_init(rme9652);
+
+ rme9652->last_spdif_sample_rate = -1;
+ rme9652->last_adat_sample_rate = -1;
+ rme9652->playback_pid = -1;
+ rme9652->capture_pid = -1;
+ rme9652->capture_substream = NULL;
+ rme9652->playback_substream = NULL;
+
+ snd_rme9652_set_defaults(rme9652);
+
+ if (rme9652->hw_rev == 15) {
+ rme9652_initialize_spdif_receiver (rme9652);
+ }
+
+ return 0;
+}
+
+static void snd_rme9652_card_free(struct snd_card *card)
+{
+ struct snd_rme9652 *rme9652 = (struct snd_rme9652 *) card->private_data;
+
+ if (rme9652)
+ snd_rme9652_free(rme9652);
+}
+
+static int __devinit snd_rme9652_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_rme9652 *rme9652;
+ struct snd_card *card;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_rme9652));
+
+ if (!card)
+ return -ENOMEM;
+
+ rme9652 = (struct snd_rme9652 *) card->private_data;
+ card->private_free = snd_rme9652_card_free;
+ rme9652->dev = dev;
+ rme9652->pci = pci;
+ snd_card_set_dev(card, &pci->dev);
+
+ if ((err = snd_rme9652_create(card, rme9652, precise_ptr[dev])) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->shortname, rme9652->card_name);
+
+ sprintf(card->longname, "%s at 0x%lx, irq %d",
+ card->shortname, rme9652->port, rme9652->irq);
+
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_rme9652_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "RME Digi9652 (Hammerfall)",
+ .id_table = snd_rme9652_ids,
+ .probe = snd_rme9652_probe,
+ .remove = __devexit_p(snd_rme9652_remove),
+};
+
+static int __init alsa_card_hammerfall_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_hammerfall_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_hammerfall_init)
+module_exit(alsa_card_hammerfall_exit)
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
new file mode 100644
index 0000000..df2007e
--- /dev/null
+++ b/sound/pci/sis7019.c
@@ -0,0 +1,1459 @@
+/*
+ * Driver for SiS7019 Audio Accelerator
+ *
+ * Copyright (C) 2004-2007, David Dillow
+ * Written by David Dillow <dave@thedillows.org>
+ * Inspired by the Trident 4D-WaveDX/NX driver.
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, version 2.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/time.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include "sis7019.h"
+
+MODULE_AUTHOR("David Dillow <dave@thedillows.org>");
+MODULE_DESCRIPTION("SiS7019");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int enable = 1;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator.");
+module_param(enable, bool, 0444);
+MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator.");
+
+static struct pci_device_id snd_sis7019_ids[] = {
+ { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_sis7019_ids);
+
+/* There are three timing modes for the voices.
+ *
+ * For both playback and capture, when the buffer is one or two periods long,
+ * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt
+ * to let us know when the periods have ended.
+ *
+ * When performing playback with more than two periods per buffer, we set
+ * the "Stop Sample Offset" and tell the hardware to interrupt us when we
+ * reach it. We then update the offset and continue on until we are
+ * interrupted for the next period.
+ *
+ * Capture channels do not have a SSO, so we allocate a playback channel to
+ * use as a timer for the capture periods. We use the SSO on the playback
+ * channel to clock out virtual periods, and adjust the virtual period length
+ * to maintain synchronization. This algorithm came from the Trident driver.
+ *
+ * FIXME: It'd be nice to make use of some of the synth features in the
+ * hardware, but a woeful lack of documentation is a significant roadblock.
+ */
+struct voice {
+ u16 flags;
+#define VOICE_IN_USE 1
+#define VOICE_CAPTURE 2
+#define VOICE_SSO_TIMING 4
+#define VOICE_SYNC_TIMING 8
+ u16 sync_cso;
+ u16 period_size;
+ u16 buffer_size;
+ u16 sync_period_size;
+ u16 sync_buffer_size;
+ u32 sso;
+ u32 vperiod;
+ struct snd_pcm_substream *substream;
+ struct voice *timing;
+ void __iomem *ctrl_base;
+ void __iomem *wave_base;
+ void __iomem *sync_base;
+ int num;
+};
+
+/* We need four pages to store our wave parameters during a suspend. If
+ * we're not doing power management, we still need to allocate a page
+ * for the silence buffer.
+ */
+#ifdef CONFIG_PM
+#define SIS_SUSPEND_PAGES 4
+#else
+#define SIS_SUSPEND_PAGES 1
+#endif
+
+struct sis7019 {
+ unsigned long ioport;
+ void __iomem *ioaddr;
+ int irq;
+ int codecs_present;
+
+ struct pci_dev *pci;
+ struct snd_pcm *pcm;
+ struct snd_card *card;
+ struct snd_ac97 *ac97[3];
+
+ /* Protect against more than one thread hitting the AC97
+ * registers (in a more polite manner than pounding the hardware
+ * semaphore)
+ */
+ struct mutex ac97_mutex;
+
+ /* voice_lock protects allocation/freeing of the voice descriptions
+ */
+ spinlock_t voice_lock;
+
+ struct voice voices[64];
+ struct voice capture_voice;
+
+ /* Allocate pages to store the internal wave state during
+ * suspends. When we're operating, this can be used as a silence
+ * buffer for a timing channel.
+ */
+ void *suspend_state[SIS_SUSPEND_PAGES];
+
+ int silence_users;
+ dma_addr_t silence_dma_addr;
+};
+
+#define SIS_PRIMARY_CODEC_PRESENT 0x0001
+#define SIS_SECONDARY_CODEC_PRESENT 0x0002
+#define SIS_TERTIARY_CODEC_PRESENT 0x0004
+
+/* The HW offset parameters (Loop End, Stop Sample, End Sample) have a
+ * documented range of 8-0xfff8 samples. Given that they are 0-based,
+ * that places our period/buffer range at 9-0xfff9 samples. That makes the
+ * max buffer size 0xfff9 samples * 2 channels * 2 bytes per sample, and
+ * max samples / min samples gives us the max periods in a buffer.
+ *
+ * We'll add a constraint upon open that limits the period and buffer sample
+ * size to values that are legal for the hardware.
+ */
+static struct snd_pcm_hardware sis_playback_hw_info = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (0xfff9 * 4),
+ .period_bytes_min = 9,
+ .period_bytes_max = (0xfff9 * 4),
+ .periods_min = 1,
+ .periods_max = (0xfff9 / 9),
+};
+
+static struct snd_pcm_hardware sis_capture_hw_info = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (0xfff9 * 4),
+ .period_bytes_min = 9,
+ .period_bytes_max = (0xfff9 * 4),
+ .periods_min = 1,
+ .periods_max = (0xfff9 / 9),
+};
+
+static void sis_update_sso(struct voice *voice, u16 period)
+{
+ void __iomem *base = voice->ctrl_base;
+
+ voice->sso += period;
+ if (voice->sso >= voice->buffer_size)
+ voice->sso -= voice->buffer_size;
+
+ /* Enforce the documented hardware minimum offset */
+ if (voice->sso < 8)
+ voice->sso = 8;
+
+ /* The SSO is in the upper 16 bits of the register. */
+ writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2);
+}
+
+static void sis_update_voice(struct voice *voice)
+{
+ if (voice->flags & VOICE_SSO_TIMING) {
+ sis_update_sso(voice, voice->period_size);
+ } else if (voice->flags & VOICE_SYNC_TIMING) {
+ int sync;
+
+ /* If we've not hit the end of the virtual period, update
+ * our records and keep going.
+ */
+ if (voice->vperiod > voice->period_size) {
+ voice->vperiod -= voice->period_size;
+ if (voice->vperiod < voice->period_size)
+ sis_update_sso(voice, voice->vperiod);
+ else
+ sis_update_sso(voice, voice->period_size);
+ return;
+ }
+
+ /* Calculate our relative offset between the target and
+ * the actual CSO value. Since we're operating in a loop,
+ * if the value is more than half way around, we can
+ * consider ourselves wrapped.
+ */
+ sync = voice->sync_cso;
+ sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO);
+ if (sync > (voice->sync_buffer_size / 2))
+ sync -= voice->sync_buffer_size;
+
+ /* If sync is positive, then we interrupted too early, and
+ * we'll need to come back in a few samples and try again.
+ * There's a minimum wait, as it takes some time for the DMA
+ * engine to startup, etc...
+ */
+ if (sync > 0) {
+ if (sync < 16)
+ sync = 16;
+ sis_update_sso(voice, sync);
+ return;
+ }
+
+ /* Ok, we interrupted right on time, or (hopefully) just
+ * a bit late. We'll adjst our next waiting period based
+ * on how close we got.
+ *
+ * We need to stay just behind the actual channel to ensure
+ * it really is past a period when we get our interrupt --
+ * otherwise we'll fall into the early code above and have
+ * a minimum wait time, which makes us quite late here,
+ * eating into the user's time to refresh the buffer, esp.
+ * if using small periods.
+ *
+ * If we're less than 9 samples behind, we're on target.
+ */
+ if (sync > -9)
+ voice->vperiod = voice->sync_period_size + 1;
+ else
+ voice->vperiod = voice->sync_period_size - 4;
+
+ if (voice->vperiod < voice->buffer_size) {
+ sis_update_sso(voice, voice->vperiod);
+ voice->vperiod = 0;
+ } else
+ sis_update_sso(voice, voice->period_size);
+
+ sync = voice->sync_cso + voice->sync_period_size;
+ if (sync >= voice->sync_buffer_size)
+ sync -= voice->sync_buffer_size;
+ voice->sync_cso = sync;
+ }
+
+ snd_pcm_period_elapsed(voice->substream);
+}
+
+static void sis_voice_irq(u32 status, struct voice *voice)
+{
+ int bit;
+
+ while (status) {
+ bit = __ffs(status);
+ status >>= bit + 1;
+ voice += bit;
+ sis_update_voice(voice);
+ voice++;
+ }
+}
+
+static irqreturn_t sis_interrupt(int irq, void *dev)
+{
+ struct sis7019 *sis = dev;
+ unsigned long io = sis->ioport;
+ struct voice *voice;
+ u32 intr, status;
+
+ /* We only use the DMA interrupts, and we don't enable any other
+ * source of interrupts. But, it is possible to see an interupt
+ * status that didn't actually interrupt us, so eliminate anything
+ * we're not expecting to avoid falsely claiming an IRQ, and an
+ * ensuing endless loop.
+ */
+ intr = inl(io + SIS_GISR);
+ intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS |
+ SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS;
+ if (!intr)
+ return IRQ_NONE;
+
+ do {
+ status = inl(io + SIS_PISR_A);
+ if (status) {
+ sis_voice_irq(status, sis->voices);
+ outl(status, io + SIS_PISR_A);
+ }
+
+ status = inl(io + SIS_PISR_B);
+ if (status) {
+ sis_voice_irq(status, &sis->voices[32]);
+ outl(status, io + SIS_PISR_B);
+ }
+
+ status = inl(io + SIS_RISR);
+ if (status) {
+ voice = &sis->capture_voice;
+ if (!voice->timing)
+ snd_pcm_period_elapsed(voice->substream);
+
+ outl(status, io + SIS_RISR);
+ }
+
+ outl(intr, io + SIS_GISR);
+ intr = inl(io + SIS_GISR);
+ intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS |
+ SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS;
+ } while (intr);
+
+ return IRQ_HANDLED;
+}
+
+static u32 sis_rate_to_delta(unsigned int rate)
+{
+ u32 delta;
+
+ /* This was copied from the trident driver, but it seems its gotten
+ * around a bit... nevertheless, it works well.
+ *
+ * We special case 44100 and 8000 since rounding with the equation
+ * does not give us an accurate enough value. For 11025 and 22050
+ * the equation gives us the best answer. All other frequencies will
+ * also use the equation. JDW
+ */
+ if (rate == 44100)
+ delta = 0xeb3;
+ else if (rate == 8000)
+ delta = 0x2ab;
+ else if (rate == 48000)
+ delta = 0x1000;
+ else
+ delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff;
+ return delta;
+}
+
+static void __sis_map_silence(struct sis7019 *sis)
+{
+ /* Helper function: must hold sis->voice_lock on entry */
+ if (!sis->silence_users)
+ sis->silence_dma_addr = pci_map_single(sis->pci,
+ sis->suspend_state[0],
+ 4096, PCI_DMA_TODEVICE);
+ sis->silence_users++;
+}
+
+static void __sis_unmap_silence(struct sis7019 *sis)
+{
+ /* Helper function: must hold sis->voice_lock on entry */
+ sis->silence_users--;
+ if (!sis->silence_users)
+ pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096,
+ PCI_DMA_TODEVICE);
+}
+
+static void sis_free_voice(struct sis7019 *sis, struct voice *voice)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sis->voice_lock, flags);
+ if (voice->timing) {
+ __sis_unmap_silence(sis);
+ voice->timing->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING |
+ VOICE_SYNC_TIMING);
+ voice->timing = NULL;
+ }
+ voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING);
+ spin_unlock_irqrestore(&sis->voice_lock, flags);
+}
+
+static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis)
+{
+ /* Must hold the voice_lock on entry */
+ struct voice *voice;
+ int i;
+
+ for (i = 0; i < 64; i++) {
+ voice = &sis->voices[i];
+ if (voice->flags & VOICE_IN_USE)
+ continue;
+ voice->flags |= VOICE_IN_USE;
+ goto found_one;
+ }
+ voice = NULL;
+
+found_one:
+ return voice;
+}
+
+static struct voice *sis_alloc_playback_voice(struct sis7019 *sis)
+{
+ struct voice *voice;
+ unsigned long flags;
+
+ spin_lock_irqsave(&sis->voice_lock, flags);
+ voice = __sis_alloc_playback_voice(sis);
+ spin_unlock_irqrestore(&sis->voice_lock, flags);
+
+ return voice;
+}
+
+static int sis_alloc_timing_voice(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct sis7019 *sis = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *voice = runtime->private_data;
+ unsigned int period_size, buffer_size;
+ unsigned long flags;
+ int needed;
+
+ /* If there are one or two periods per buffer, we don't need a
+ * timing voice, as we can use the capture channel's interrupts
+ * to clock out the periods.
+ */
+ period_size = params_period_size(hw_params);
+ buffer_size = params_buffer_size(hw_params);
+ needed = (period_size != buffer_size &&
+ period_size != (buffer_size / 2));
+
+ if (needed && !voice->timing) {
+ spin_lock_irqsave(&sis->voice_lock, flags);
+ voice->timing = __sis_alloc_playback_voice(sis);
+ if (voice->timing)
+ __sis_map_silence(sis);
+ spin_unlock_irqrestore(&sis->voice_lock, flags);
+ if (!voice->timing)
+ return -ENOMEM;
+ voice->timing->substream = substream;
+ } else if (!needed && voice->timing) {
+ sis_free_voice(sis, voice);
+ voice->timing = NULL;
+ }
+
+ return 0;
+}
+
+static int sis_playback_open(struct snd_pcm_substream *substream)
+{
+ struct sis7019 *sis = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *voice;
+
+ voice = sis_alloc_playback_voice(sis);
+ if (!voice)
+ return -EAGAIN;
+
+ voice->substream = substream;
+ runtime->private_data = voice;
+ runtime->hw = sis_playback_hw_info;
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ 9, 0xfff9);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ 9, 0xfff9);
+ snd_pcm_set_sync(substream);
+ return 0;
+}
+
+static int sis_substream_close(struct snd_pcm_substream *substream)
+{
+ struct sis7019 *sis = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *voice = runtime->private_data;
+
+ sis_free_voice(sis, voice);
+ return 0;
+}
+
+static int sis_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int sis_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *voice = runtime->private_data;
+ void __iomem *ctrl_base = voice->ctrl_base;
+ void __iomem *wave_base = voice->wave_base;
+ u32 format, dma_addr, control, sso_eso, delta, reg;
+ u16 leo;
+
+ /* We rely on the PCM core to ensure that the parameters for this
+ * substream do not change on us while we're programming the HW.
+ */
+ format = 0;
+ if (snd_pcm_format_width(runtime->format) == 8)
+ format |= SIS_PLAY_DMA_FORMAT_8BIT;
+ if (!snd_pcm_format_signed(runtime->format))
+ format |= SIS_PLAY_DMA_FORMAT_UNSIGNED;
+ if (runtime->channels == 1)
+ format |= SIS_PLAY_DMA_FORMAT_MONO;
+
+ /* The baseline setup is for a single period per buffer, and
+ * we add bells and whistles as needed from there.
+ */
+ dma_addr = runtime->dma_addr;
+ leo = runtime->buffer_size - 1;
+ control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO;
+ sso_eso = leo;
+
+ if (runtime->period_size == (runtime->buffer_size / 2)) {
+ control |= SIS_PLAY_DMA_INTR_AT_MLP;
+ } else if (runtime->period_size != runtime->buffer_size) {
+ voice->flags |= VOICE_SSO_TIMING;
+ voice->sso = runtime->period_size - 1;
+ voice->period_size = runtime->period_size;
+ voice->buffer_size = runtime->buffer_size;
+
+ control &= ~SIS_PLAY_DMA_INTR_AT_LEO;
+ control |= SIS_PLAY_DMA_INTR_AT_SSO;
+ sso_eso |= (runtime->period_size - 1) << 16;
+ }
+
+ delta = sis_rate_to_delta(runtime->rate);
+
+ /* Ok, we're ready to go, set up the channel.
+ */
+ writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO);
+ writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE);
+ writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL);
+ writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO);
+
+ for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4)
+ writel(0, wave_base + reg);
+
+ writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL);
+ writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION);
+ writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE |
+ SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE |
+ SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE,
+ wave_base + SIS_WAVE_CHANNEL_CONTROL);
+
+ /* Force PCI writes to post. */
+ readl(ctrl_base);
+
+ return 0;
+}
+
+static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct sis7019 *sis = snd_pcm_substream_chip(substream);
+ unsigned long io = sis->ioport;
+ struct snd_pcm_substream *s;
+ struct voice *voice;
+ void *chip;
+ int starting;
+ u32 record = 0;
+ u32 play[2] = { 0, 0 };
+
+ /* No locks needed, as the PCM core will hold the locks on the
+ * substreams, and the HW will only start/stop the indicated voices
+ * without changing the state of the others.
+ */
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ starting = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ starting = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ /* Make sure it is for us... */
+ chip = snd_pcm_substream_chip(s);
+ if (chip != sis)
+ continue;
+
+ voice = s->runtime->private_data;
+ if (voice->flags & VOICE_CAPTURE) {
+ record |= 1 << voice->num;
+ voice = voice->timing;
+ }
+
+ /* voice could be NULL if this a recording stream, and it
+ * doesn't have an external timing channel.
+ */
+ if (voice)
+ play[voice->num / 32] |= 1 << (voice->num & 0x1f);
+
+ snd_pcm_trigger_done(s, substream);
+ }
+
+ if (starting) {
+ if (record)
+ outl(record, io + SIS_RECORD_START_REG);
+ if (play[0])
+ outl(play[0], io + SIS_PLAY_START_A_REG);
+ if (play[1])
+ outl(play[1], io + SIS_PLAY_START_B_REG);
+ } else {
+ if (record)
+ outl(record, io + SIS_RECORD_STOP_REG);
+ if (play[0])
+ outl(play[0], io + SIS_PLAY_STOP_A_REG);
+ if (play[1])
+ outl(play[1], io + SIS_PLAY_STOP_B_REG);
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *voice = runtime->private_data;
+ u32 cso;
+
+ cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO);
+ cso &= 0xffff;
+ return cso;
+}
+
+static int sis_capture_open(struct snd_pcm_substream *substream)
+{
+ struct sis7019 *sis = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *voice = &sis->capture_voice;
+ unsigned long flags;
+
+ /* FIXME: The driver only supports recording from one channel
+ * at the moment, but it could support more.
+ */
+ spin_lock_irqsave(&sis->voice_lock, flags);
+ if (voice->flags & VOICE_IN_USE)
+ voice = NULL;
+ else
+ voice->flags |= VOICE_IN_USE;
+ spin_unlock_irqrestore(&sis->voice_lock, flags);
+
+ if (!voice)
+ return -EAGAIN;
+
+ voice->substream = substream;
+ runtime->private_data = voice;
+ runtime->hw = sis_capture_hw_info;
+ runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC];
+ snd_pcm_limit_hw_rates(runtime);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ 9, 0xfff9);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ 9, 0xfff9);
+ snd_pcm_set_sync(substream);
+ return 0;
+}
+
+static int sis_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct sis7019 *sis = snd_pcm_substream_chip(substream);
+ int rc;
+
+ rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE,
+ params_rate(hw_params));
+ if (rc)
+ goto out;
+
+ rc = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (rc < 0)
+ goto out;
+
+ rc = sis_alloc_timing_voice(substream, hw_params);
+
+out:
+ return rc;
+}
+
+static void sis_prepare_timing_voice(struct voice *voice,
+ struct snd_pcm_substream *substream)
+{
+ struct sis7019 *sis = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *timing = voice->timing;
+ void __iomem *play_base = timing->ctrl_base;
+ void __iomem *wave_base = timing->wave_base;
+ u16 buffer_size, period_size;
+ u32 format, control, sso_eso, delta;
+ u32 vperiod, sso, reg;
+
+ /* Set our initial buffer and period as large as we can given a
+ * single page of silence.
+ */
+ buffer_size = 4096 / runtime->channels;
+ buffer_size /= snd_pcm_format_size(runtime->format, 1);
+ period_size = buffer_size;
+
+ /* Initially, we want to interrupt just a bit behind the end of
+ * the period we're clocking out. 10 samples seems to give a good
+ * delay.
+ *
+ * We want to spread our interrupts throughout the virtual period,
+ * so that we don't end up with two interrupts back to back at the
+ * end -- this helps minimize the effects of any jitter. Adjust our
+ * clocking period size so that the last period is at least a fourth
+ * of a full period.
+ *
+ * This is all moot if we don't need to use virtual periods.
+ */
+ vperiod = runtime->period_size + 10;
+ if (vperiod > period_size) {
+ u16 tail = vperiod % period_size;
+ u16 quarter_period = period_size / 4;
+
+ if (tail && tail < quarter_period) {
+ u16 loops = vperiod / period_size;
+
+ tail = quarter_period - tail;
+ tail += loops - 1;
+ tail /= loops;
+ period_size -= tail;
+ }
+
+ sso = period_size - 1;
+ } else {
+ /* The initial period will fit inside the buffer, so we
+ * don't need to use virtual periods -- disable them.
+ */
+ period_size = runtime->period_size;
+ sso = vperiod - 1;
+ vperiod = 0;
+ }
+
+ /* The interrupt handler implements the timing syncronization, so
+ * setup its state.
+ */
+ timing->flags |= VOICE_SYNC_TIMING;
+ timing->sync_base = voice->ctrl_base;
+ timing->sync_cso = runtime->period_size - 1;
+ timing->sync_period_size = runtime->period_size;
+ timing->sync_buffer_size = runtime->buffer_size;
+ timing->period_size = period_size;
+ timing->buffer_size = buffer_size;
+ timing->sso = sso;
+ timing->vperiod = vperiod;
+
+ /* Using unsigned samples with the all-zero silence buffer
+ * forces the output to the lower rail, killing playback.
+ * So ignore unsigned vs signed -- it doesn't change the timing.
+ */
+ format = 0;
+ if (snd_pcm_format_width(runtime->format) == 8)
+ format = SIS_CAPTURE_DMA_FORMAT_8BIT;
+ if (runtime->channels == 1)
+ format |= SIS_CAPTURE_DMA_FORMAT_MONO;
+
+ control = timing->buffer_size - 1;
+ control |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO;
+ sso_eso = timing->buffer_size - 1;
+ sso_eso |= timing->sso << 16;
+
+ delta = sis_rate_to_delta(runtime->rate);
+
+ /* We've done the math, now configure the channel.
+ */
+ writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO);
+ writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE);
+ writel(control, play_base + SIS_PLAY_DMA_CONTROL);
+ writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO);
+
+ for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4)
+ writel(0, wave_base + reg);
+
+ writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL);
+ writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION);
+ writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE |
+ SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE |
+ SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE,
+ wave_base + SIS_WAVE_CHANNEL_CONTROL);
+}
+
+static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voice *voice = runtime->private_data;
+ void __iomem *rec_base = voice->ctrl_base;
+ u32 format, dma_addr, control;
+ u16 leo;
+
+ /* We rely on the PCM core to ensure that the parameters for this
+ * substream do not change on us while we're programming the HW.
+ */
+ format = 0;
+ if (snd_pcm_format_width(runtime->format) == 8)
+ format = SIS_CAPTURE_DMA_FORMAT_8BIT;
+ if (!snd_pcm_format_signed(runtime->format))
+ format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED;
+ if (runtime->channels == 1)
+ format |= SIS_CAPTURE_DMA_FORMAT_MONO;
+
+ dma_addr = runtime->dma_addr;
+ leo = runtime->buffer_size - 1;
+ control = leo | SIS_CAPTURE_DMA_LOOP;
+
+ /* If we've got more than two periods per buffer, then we have
+ * use a timing voice to clock out the periods. Otherwise, we can
+ * use the capture channel's interrupts.
+ */
+ if (voice->timing) {
+ sis_prepare_timing_voice(voice, substream);
+ } else {
+ control |= SIS_CAPTURE_DMA_INTR_AT_LEO;
+ if (runtime->period_size != runtime->buffer_size)
+ control |= SIS_CAPTURE_DMA_INTR_AT_MLP;
+ }
+
+ writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO);
+ writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE);
+ writel(control, rec_base + SIS_CAPTURE_DMA_CONTROL);
+
+ /* Force the writes to post. */
+ readl(rec_base);
+
+ return 0;
+}
+
+static struct snd_pcm_ops sis_playback_ops = {
+ .open = sis_playback_open,
+ .close = sis_substream_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = sis_playback_hw_params,
+ .hw_free = sis_hw_free,
+ .prepare = sis_pcm_playback_prepare,
+ .trigger = sis_pcm_trigger,
+ .pointer = sis_pcm_pointer,
+};
+
+static struct snd_pcm_ops sis_capture_ops = {
+ .open = sis_capture_open,
+ .close = sis_substream_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = sis_capture_hw_params,
+ .hw_free = sis_hw_free,
+ .prepare = sis_pcm_capture_prepare,
+ .trigger = sis_pcm_trigger,
+ .pointer = sis_pcm_pointer,
+};
+
+static int __devinit sis_pcm_create(struct sis7019 *sis)
+{
+ struct snd_pcm *pcm;
+ int rc;
+
+ /* We have 64 voices, and the driver currently records from
+ * only one channel, though that could change in the future.
+ */
+ rc = snd_pcm_new(sis->card, "SiS7019", 0, 64, 1, &pcm);
+ if (rc)
+ return rc;
+
+ pcm->private_data = sis;
+ strcpy(pcm->name, "SiS7019");
+ sis->pcm = pcm;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops);
+
+ /* Try to preallocate some memory, but it's not the end of the
+ * world if this fails.
+ */
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(sis->pci), 64*1024, 128*1024);
+
+ return 0;
+}
+
+static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd)
+{
+ unsigned long io = sis->ioport;
+ unsigned short val = 0xffff;
+ u16 status;
+ u16 rdy;
+ int count;
+ static const u16 codec_ready[3] = {
+ SIS_AC97_STATUS_CODEC_READY,
+ SIS_AC97_STATUS_CODEC2_READY,
+ SIS_AC97_STATUS_CODEC3_READY,
+ };
+
+ rdy = codec_ready[codec];
+
+
+ /* Get the AC97 semaphore -- software first, so we don't spin
+ * pounding out IO reads on the hardware semaphore...
+ */
+ mutex_lock(&sis->ac97_mutex);
+
+ count = 0xffff;
+ while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count)
+ udelay(1);
+
+ if (!count)
+ goto timeout;
+
+ /* ... and wait for any outstanding commands to complete ...
+ */
+ count = 0xffff;
+ do {
+ status = inw(io + SIS_AC97_STATUS);
+ if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY))
+ break;
+
+ udelay(1);
+ } while (--count);
+
+ if (!count)
+ goto timeout_sema;
+
+ /* ... before sending our command and waiting for it to finish ...
+ */
+ outl(cmd, io + SIS_AC97_CMD);
+ udelay(10);
+
+ count = 0xffff;
+ while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)
+ udelay(1);
+
+ /* ... and reading the results (if any).
+ */
+ val = inl(io + SIS_AC97_CMD) >> 16;
+
+timeout_sema:
+ outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
+timeout:
+ mutex_unlock(&sis->ac97_mutex);
+
+ if (!count) {
+ printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n",
+ codec, cmd);
+ }
+
+ return val;
+}
+
+static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ static const u32 cmd[3] = {
+ SIS_AC97_CMD_CODEC_WRITE,
+ SIS_AC97_CMD_CODEC2_WRITE,
+ SIS_AC97_CMD_CODEC3_WRITE,
+ };
+ sis_ac97_rw(ac97->private_data, ac97->num,
+ (val << 16) | (reg << 8) | cmd[ac97->num]);
+}
+
+static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ static const u32 cmd[3] = {
+ SIS_AC97_CMD_CODEC_READ,
+ SIS_AC97_CMD_CODEC2_READ,
+ SIS_AC97_CMD_CODEC3_READ,
+ };
+ return sis_ac97_rw(ac97->private_data, ac97->num,
+ (reg << 8) | cmd[ac97->num]);
+}
+
+static int __devinit sis_mixer_create(struct sis7019 *sis)
+{
+ struct snd_ac97_bus *bus;
+ struct snd_ac97_template ac97;
+ static struct snd_ac97_bus_ops ops = {
+ .write = sis_ac97_write,
+ .read = sis_ac97_read,
+ };
+ int rc;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = sis;
+
+ rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus);
+ if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT)
+ rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]);
+ ac97.num = 1;
+ if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT))
+ rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]);
+ ac97.num = 2;
+ if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT))
+ rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]);
+
+ /* If we return an error here, then snd_card_free() should
+ * free up any ac97 codecs that got created, as well as the bus.
+ */
+ return rc;
+}
+
+static void sis_free_suspend(struct sis7019 *sis)
+{
+ int i;
+
+ for (i = 0; i < SIS_SUSPEND_PAGES; i++)
+ kfree(sis->suspend_state[i]);
+}
+
+static int sis_chip_free(struct sis7019 *sis)
+{
+ /* Reset the chip, and disable all interrputs.
+ */
+ outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR);
+ udelay(10);
+ outl(0, sis->ioport + SIS_GCR);
+ outl(0, sis->ioport + SIS_GIER);
+
+ /* Now, free everything we allocated.
+ */
+ if (sis->irq >= 0)
+ free_irq(sis->irq, sis);
+
+ if (sis->ioaddr)
+ iounmap(sis->ioaddr);
+
+ pci_release_regions(sis->pci);
+ pci_disable_device(sis->pci);
+
+ sis_free_suspend(sis);
+ return 0;
+}
+
+static int sis_dev_free(struct snd_device *dev)
+{
+ struct sis7019 *sis = dev->device_data;
+ return sis_chip_free(sis);
+}
+
+static int sis_chip_init(struct sis7019 *sis)
+{
+ unsigned long io = sis->ioport;
+ void __iomem *ioaddr = sis->ioaddr;
+ u16 status;
+ int count;
+ int i;
+
+ /* Reset the audio controller
+ */
+ outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR);
+ udelay(10);
+ outl(0, io + SIS_GCR);
+
+ /* Get the AC-link semaphore, and reset the codecs
+ */
+ count = 0xffff;
+ while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count)
+ udelay(1);
+
+ if (!count)
+ return -EIO;
+
+ outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD);
+ udelay(10);
+
+ count = 0xffff;
+ while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)
+ udelay(1);
+
+ /* Now that we've finished the reset, find out what's attached.
+ */
+ status = inl(io + SIS_AC97_STATUS);
+ if (status & SIS_AC97_STATUS_CODEC_READY)
+ sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT;
+ if (status & SIS_AC97_STATUS_CODEC2_READY)
+ sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT;
+ if (status & SIS_AC97_STATUS_CODEC3_READY)
+ sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT;
+
+ /* All done, let go of the semaphore, and check for errors
+ */
+ outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA);
+ if (!sis->codecs_present || !count)
+ return -EIO;
+
+ /* Let the hardware know that the audio driver is alive,
+ * and enable PCM slots on the AC-link for L/R playback (3 & 4) and
+ * record channels. We're going to want to use Variable Rate Audio
+ * for recording, to avoid needlessly resampling from 48kHZ.
+ */
+ outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF);
+ outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE |
+ SIS_AC97_CONF_PCM_CAP_MIC_ENABLE |
+ SIS_AC97_CONF_PCM_CAP_LR_ENABLE |
+ SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF);
+
+ /* All AC97 PCM slots should be sourced from sub-mixer 0.
+ */
+ outl(0, io + SIS_AC97_PSR);
+
+ /* There is only one valid DMA setup for a PCI environment.
+ */
+ outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR);
+
+ /* Reset the syncronization groups for all of the channels
+ * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc.
+ * we'll need to change how we handle these. Until then, we just
+ * assign sub-mixer 0 to all playback channels, and avoid any
+ * attenuation on the audio.
+ */
+ outl(0, io + SIS_PLAY_SYNC_GROUP_A);
+ outl(0, io + SIS_PLAY_SYNC_GROUP_B);
+ outl(0, io + SIS_PLAY_SYNC_GROUP_C);
+ outl(0, io + SIS_PLAY_SYNC_GROUP_D);
+ outl(0, io + SIS_MIXER_SYNC_GROUP);
+
+ for (i = 0; i < 64; i++) {
+ writel(i, SIS_MIXER_START_ADDR(ioaddr, i));
+ writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN |
+ SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i));
+ }
+
+ /* Don't attenuate any audio set for the wave amplifier.
+ *
+ * FIXME: Maximum attenuation is set for the music amp, which will
+ * need to change if we start using the synth engine.
+ */
+ outl(0xffff0000, io + SIS_WEVCR);
+
+ /* Ensure that the wave engine is in normal operating mode.
+ */
+ outl(0, io + SIS_WECCR);
+
+ /* Go ahead and enable the DMA interrupts. They won't go live
+ * until we start a channel.
+ */
+ outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE |
+ SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int sis_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct sis7019 *sis = card->private_data;
+ void __iomem *ioaddr = sis->ioaddr;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(sis->pcm);
+ if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT)
+ snd_ac97_suspend(sis->ac97[0]);
+ if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)
+ snd_ac97_suspend(sis->ac97[1]);
+ if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)
+ snd_ac97_suspend(sis->ac97[2]);
+
+ /* snd_pcm_suspend_all() stopped all channels, so we're quiescent.
+ */
+ if (sis->irq >= 0) {
+ free_irq(sis->irq, sis);
+ sis->irq = -1;
+ }
+
+ /* Save the internal state away
+ */
+ for (i = 0; i < 4; i++) {
+ memcpy_fromio(sis->suspend_state[i], ioaddr, 4096);
+ ioaddr += 4096;
+ }
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int sis_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct sis7019 *sis = card->private_data;
+ void __iomem *ioaddr = sis->ioaddr;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "sis7019: unable to re-enable device\n");
+ goto error;
+ }
+
+ if (sis_chip_init(sis)) {
+ printk(KERN_ERR "sis7019: unable to re-init controller\n");
+ goto error;
+ }
+
+ if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+ card->shortname, sis)) {
+ printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq);
+ goto error;
+ }
+
+ /* Restore saved state, then clear out the page we use for the
+ * silence buffer.
+ */
+ for (i = 0; i < 4; i++) {
+ memcpy_toio(ioaddr, sis->suspend_state[i], 4096);
+ ioaddr += 4096;
+ }
+
+ memset(sis->suspend_state[0], 0, 4096);
+
+ sis->irq = pci->irq;
+ pci_set_master(pci);
+
+ if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT)
+ snd_ac97_resume(sis->ac97[0]);
+ if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)
+ snd_ac97_resume(sis->ac97[1]);
+ if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)
+ snd_ac97_resume(sis->ac97[2]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+
+error:
+ snd_card_disconnect(card);
+ return -EIO;
+}
+#endif /* CONFIG_PM */
+
+static int sis_alloc_suspend(struct sis7019 *sis)
+{
+ int i;
+
+ /* We need 16K to store the internal wave engine state during a
+ * suspend, but we don't need it to be contiguous, so play nice
+ * with the memory system. We'll also use this area for a silence
+ * buffer.
+ */
+ for (i = 0; i < SIS_SUSPEND_PAGES; i++) {
+ sis->suspend_state[i] = kmalloc(4096, GFP_KERNEL);
+ if (!sis->suspend_state[i])
+ return -ENOMEM;
+ }
+ memset(sis->suspend_state[0], 0, 4096);
+
+ return 0;
+}
+
+static int __devinit sis_chip_create(struct snd_card *card,
+ struct pci_dev *pci)
+{
+ struct sis7019 *sis = card->private_data;
+ struct voice *voice;
+ static struct snd_device_ops ops = {
+ .dev_free = sis_dev_free,
+ };
+ int rc;
+ int i;
+
+ rc = pci_enable_device(pci);
+ if (rc)
+ goto error_out;
+
+ if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) {
+ printk(KERN_ERR "sis7019: architecture does not support "
+ "30-bit PCI busmaster DMA");
+ goto error_out_enabled;
+ }
+
+ memset(sis, 0, sizeof(*sis));
+ mutex_init(&sis->ac97_mutex);
+ spin_lock_init(&sis->voice_lock);
+ sis->card = card;
+ sis->pci = pci;
+ sis->irq = -1;
+ sis->ioport = pci_resource_start(pci, 0);
+
+ rc = pci_request_regions(pci, "SiS7019");
+ if (rc) {
+ printk(KERN_ERR "sis7019: unable request regions\n");
+ goto error_out_enabled;
+ }
+
+ rc = -EIO;
+ sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000);
+ if (!sis->ioaddr) {
+ printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n");
+ goto error_out_cleanup;
+ }
+
+ rc = sis_alloc_suspend(sis);
+ if (rc < 0) {
+ printk(KERN_ERR "sis7019: unable to allocate state storage\n");
+ goto error_out_cleanup;
+ }
+
+ rc = sis_chip_init(sis);
+ if (rc)
+ goto error_out_cleanup;
+
+ if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+ card->shortname, sis)) {
+ printk(KERN_ERR "unable to allocate irq %d\n", sis->irq);
+ goto error_out_cleanup;
+ }
+
+ sis->irq = pci->irq;
+ pci_set_master(pci);
+
+ for (i = 0; i < 64; i++) {
+ voice = &sis->voices[i];
+ voice->num = i;
+ voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i);
+ voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i);
+ }
+
+ voice = &sis->capture_voice;
+ voice->flags = VOICE_CAPTURE;
+ voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN;
+ voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num);
+
+ rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops);
+ if (rc)
+ goto error_out_cleanup;
+
+ snd_card_set_dev(card, &pci->dev);
+
+ return 0;
+
+error_out_cleanup:
+ sis_chip_free(sis);
+
+error_out_enabled:
+ pci_disable_device(pci);
+
+error_out:
+ return rc;
+}
+
+static int __devinit snd_sis7019_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct sis7019 *sis;
+ int rc;
+
+ rc = -ENOENT;
+ if (!enable)
+ goto error_out;
+
+ rc = -ENOMEM;
+ card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis));
+ if (!card)
+ goto error_out;
+
+ strcpy(card->driver, "SiS7019");
+ strcpy(card->shortname, "SiS7019");
+ rc = sis_chip_create(card, pci);
+ if (rc)
+ goto card_error_out;
+
+ sis = card->private_data;
+
+ rc = sis_mixer_create(sis);
+ if (rc)
+ goto card_error_out;
+
+ rc = sis_pcm_create(sis);
+ if (rc)
+ goto card_error_out;
+
+ snprintf(card->longname, sizeof(card->longname),
+ "%s Audio Accelerator with %s at 0x%lx, irq %d",
+ card->shortname, snd_ac97_get_short_name(sis->ac97[0]),
+ sis->ioport, sis->irq);
+
+ rc = snd_card_register(card);
+ if (rc)
+ goto card_error_out;
+
+ pci_set_drvdata(pci, card);
+ return 0;
+
+card_error_out:
+ snd_card_free(card);
+
+error_out:
+ return rc;
+}
+
+static void __devexit snd_sis7019_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver sis7019_driver = {
+ .name = "SiS7019",
+ .id_table = snd_sis7019_ids,
+ .probe = snd_sis7019_probe,
+ .remove = __devexit_p(snd_sis7019_remove),
+
+#ifdef CONFIG_PM
+ .suspend = sis_suspend,
+ .resume = sis_resume,
+#endif
+};
+
+static int __init sis7019_init(void)
+{
+ return pci_register_driver(&sis7019_driver);
+}
+
+static void __exit sis7019_exit(void)
+{
+ pci_unregister_driver(&sis7019_driver);
+}
+
+module_init(sis7019_init);
+module_exit(sis7019_exit);
diff --git a/sound/pci/sis7019.h b/sound/pci/sis7019.h
new file mode 100644
index 0000000..013b673
--- /dev/null
+++ b/sound/pci/sis7019.h
@@ -0,0 +1,342 @@
+#ifndef __sis7019_h__
+#define __sis7019_h__
+
+/*
+ * Definitions for SiS7019 Audio Accelerator
+ *
+ * Copyright (C) 2004-2007, David Dillow
+ * Written by David Dillow <dave@thedillows.org>
+ * Inspired by the Trident 4D-WaveDX/NX driver.
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, version 2.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+/* General Control Register */
+#define SIS_GCR 0x00
+#define SIS_GCR_MACRO_POWER_DOWN 0x80000000
+#define SIS_GCR_MODEM_ENABLE 0x00010000
+#define SIS_GCR_SOFTWARE_RESET 0x00000001
+
+/* General Interrupt Enable Register */
+#define SIS_GIER 0x04
+#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000
+#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000
+#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000
+#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000
+#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000
+#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010
+#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008
+#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004
+#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002
+#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001
+
+/* General Interrupt Status Register */
+#define SIS_GISR 0x08
+#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000
+#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000
+#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000
+#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000
+#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000
+#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010
+#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008
+#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004
+#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002
+#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001
+
+/* DMA Control Register */
+#define SIS_DMA_CSR 0x10
+#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d
+#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200
+#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100
+#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010
+#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008
+#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004
+#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002
+#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001
+
+/* Playback Channel Start Registers */
+#define SIS_PLAY_START_A_REG 0x14
+#define SIS_PLAY_START_B_REG 0x18
+
+/* Playback Channel Stop Registers */
+#define SIS_PLAY_STOP_A_REG 0x1c
+#define SIS_PLAY_STOP_B_REG 0x20
+
+/* Recording Channel Start Register */
+#define SIS_RECORD_START_REG 0x24
+
+/* Recording Channel Stop Register */
+#define SIS_RECORD_STOP_REG 0x28
+
+/* Playback Interrupt Status Registers */
+#define SIS_PISR_A 0x2c
+#define SIS_PISR_B 0x30
+
+/* Recording Interrupt Status Register */
+#define SIS_RISR 0x34
+
+/* AC97 AC-link Playback Source Register */
+#define SIS_AC97_PSR 0x40
+#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000
+#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000
+#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000
+#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000
+#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00
+#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0
+#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f
+
+/* AC97 AC-link Command Register */
+#define SIS_AC97_CMD 0x50
+#define SIS_AC97_CMD_DATA_MASK 0xffff0000
+#define SIS_AC97_CMD_REG_MASK 0x0000ff00
+#define SIS_AC97_CMD_CODEC3_READ 0x0000000d
+#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c
+#define SIS_AC97_CMD_CODEC2_READ 0x0000000b
+#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a
+#define SIS_AC97_CMD_CODEC_READ 0x00000009
+#define SIS_AC97_CMD_CODEC_WRITE 0x00000008
+#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005
+#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004
+#define SIS_AC97_CMD_DONE 0x00000000
+
+/* AC97 AC-link Semaphore Register */
+#define SIS_AC97_SEMA 0x54
+#define SIS_AC97_SEMA_BUSY 0x00000001
+#define SIS_AC97_SEMA_RELEASE 0x00000000
+
+/* AC97 AC-link Status Register */
+#define SIS_AC97_STATUS 0x58
+#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000
+#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000
+#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000
+#define SIS_AC97_STATUS_CODEC3_READY 0x00000400
+#define SIS_AC97_STATUS_CODEC2_READY 0x00000200
+#define SIS_AC97_STATUS_CODEC_READY 0x00000100
+#define SIS_AC97_STATUS_WARM_RESET 0x00000080
+#define SIS_AC97_STATUS_COLD_RESET 0x00000040
+#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020
+#define SIS_AC97_STATUS_NORMAL 0x00000010
+#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004
+#define SIS_AC97_STATUS_SEMAPHORE 0x00000002
+#define SIS_AC97_STATUS_BUSY 0x00000001
+
+/* AC97 AC-link Audio Configuration Register */
+#define SIS_AC97_CONF 0x5c
+#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000
+#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000
+#define SIS_AC97_CONF_PR6_ENABLE 0x20000000
+#define SIS_AC97_CONF_PR5_ENABLE 0x10000000
+#define SIS_AC97_CONF_PR4_ENABLE 0x08000000
+#define SIS_AC97_CONF_PR3_ENABLE 0x04000000
+#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000
+#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000
+#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000
+#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000
+#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000
+#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000
+#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000
+#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000
+#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000
+#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200
+#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100
+#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080
+#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040
+#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020
+#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010
+#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008
+#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004
+#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002
+#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001
+
+/* Playback Channel Sync Group registers */
+#define SIS_PLAY_SYNC_GROUP_A 0x80
+#define SIS_PLAY_SYNC_GROUP_B 0x84
+#define SIS_PLAY_SYNC_GROUP_C 0x88
+#define SIS_PLAY_SYNC_GROUP_D 0x8c
+#define SIS_MIXER_SYNC_GROUP 0x90
+
+/* Wave Engine Config and Control Register */
+#define SIS_WECCR 0xa0
+#define SIS_WECCR_TESTMODE_MASK 0x00300000
+#define SIS_WECCR_TESTMODE_NORMAL 0x00000000
+#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000
+#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000
+#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000
+#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000
+#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000
+#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000
+#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000
+#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000
+#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000
+#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00
+#define SIS_WECCR_COMMAND_MASK 0x00000007
+#define SIS_WECCR_COMMAND_NONE 0x00000000
+#define SIS_WECCR_COMMAND_DONE 0x00000000
+#define SIS_WECCR_COMMAND_PAUSE 0x00000001
+#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002
+#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003
+#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004
+
+/* Wave Engine Volume Control Register */
+#define SIS_WEVCR 0xa4
+#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000
+#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000
+#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00
+#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff
+
+/* Wave Engine Interrupt Status Registers */
+#define SIS_WEISR_A 0xa8
+#define SIS_WEISR_B 0xac
+
+
+/* Playback DMA parameters (paramter RAM) */
+#define SIS_PLAY_DMA_OFFSET 0x0000
+#define SIS_PLAY_DMA_SIZE 0x10
+#define SIS_PLAY_DMA_ADDR(addr, num) \
+ ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET)
+
+#define SIS_PLAY_DMA_FORMAT_CSO 0x00
+#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000
+#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000
+#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000
+#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff
+#define SIS_PLAY_DMA_BASE 0x04
+#define SIS_PLAY_DMA_CONTROL 0x08
+#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000
+#define SIS_PLAY_DMA_RELEASE 0x02000000
+#define SIS_PLAY_DMA_LOOP 0x01000000
+#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000
+#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000
+#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000
+#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000
+#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff
+#define SIS_PLAY_DMA_SSO_ESO 0x0c
+#define SIS_PLAY_DMA_SSO_MASK 0xffff0000
+#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff
+
+/* Capture DMA parameters (paramter RAM) */
+#define SIS_CAPTURE_DMA_OFFSET 0x0800
+#define SIS_CAPTURE_DMA_SIZE 0x10
+#define SIS_CAPTURE_DMA_ADDR(addr, num) \
+ ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET)
+
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14
+#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15
+#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16
+#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17
+#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18
+#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19
+#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20
+
+#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00
+#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000
+#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000
+#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000
+#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000
+#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000
+#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000
+#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000
+#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff
+#define SIS_CAPTURE_DMA_BASE 0x04
+#define SIS_CAPTURE_DMA_CONTROL 0x08
+#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000
+#define SIS_CAPTURE_DMA_RELEASE 0x02000000
+#define SIS_CAPTURE_DMA_LOOP 0x01000000
+#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000
+#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000
+#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff
+#define SIS_CAPTURE_DMA_RESERVED 0x0c
+
+
+/* Mixer routing list start pointer (parameter RAM) */
+#define SIS_MIXER_START_OFFSET 0x1000
+#define SIS_MIXER_START_SIZE 0x04
+#define SIS_MIXER_START_ADDR(addr, num) \
+ ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET)
+
+#define SIS_MIXER_START_MASK 0x0000007f
+
+/* Mixer routing table (parameter RAM) */
+#define SIS_MIXER_OFFSET 0x1400
+#define SIS_MIXER_SIZE 0x04
+#define SIS_MIXER_ADDR(addr, num) \
+ ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET)
+
+#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000
+#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000
+#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0x00ff0000
+#define SIS_MIXER_LEFT_NO_ATTEN 0x00ff0000
+#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00
+#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000
+#define SIS_MIXER_DEST_MASK 0x0000007f
+#define SIS_MIXER_DEST_0 0x00000020
+#define SIS_MIXER_DEST_1 0x00000021
+#define SIS_MIXER_DEST_2 0x00000022
+#define SIS_MIXER_DEST_3 0x00000023
+#define SIS_MIXER_DEST_4 0x00000024
+#define SIS_MIXER_DEST_5 0x00000025
+#define SIS_MIXER_DEST_6 0x00000026
+#define SIS_MIXER_DEST_7 0x00000027
+#define SIS_MIXER_DEST_8 0x00000028
+#define SIS_MIXER_DEST_9 0x00000029
+#define SIS_MIXER_DEST_10 0x0000002a
+#define SIS_MIXER_DEST_11 0x0000002b
+#define SIS_MIXER_DEST_12 0x0000002c
+#define SIS_MIXER_DEST_13 0x0000002d
+#define SIS_MIXER_DEST_14 0x0000002e
+#define SIS_MIXER_DEST_15 0x0000002f
+
+/* Wave Engine Control Parameters (parameter RAM) */
+#define SIS_WAVE_OFFSET 0x2000
+#define SIS_WAVE_SIZE 0x40
+#define SIS_WAVE_ADDR(addr, num) \
+ ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET)
+
+#define SIS_WAVE_GENERAL 0x00
+#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000
+#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000
+#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000
+#define SIS_WAVE_GENERAL_ARTICULATION 0x04
+#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000
+#define SIS_WAVE_ARTICULATION 0x08
+#define SIS_WAVE_TIMER 0x0c
+#define SIS_WAVE_GENERATOR 0x10
+#define SIS_WAVE_CHANNEL_CONTROL 0x14
+#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000
+#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000
+#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000
+#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000
+#define SIS_WAVE_LFO_EG_CONTROL 0x18
+#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c
+#define SIS_WAVE_LFO_EG_CONTROL_3 0x20
+#define SIS_WAVE_LFO_EG_CONTROL_4 0x24
+
+#endif /* __sis7019_h__ */
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
new file mode 100644
index 0000000..cd408b8
--- /dev/null
+++ b/sound/pci/sonicvibes.c
@@ -0,0 +1,1515 @@
+/*
+ * Driver for S3 SonicVibes soundcard
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ * BUGS:
+ * It looks like 86c617 rev 3 doesn't supports DDMA buffers above 16MB?
+ * Driver sometimes hangs... Nobody knows why at this moment...
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("S3 SonicVibes PCI");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}");
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK 1
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int reverb[SNDRV_CARDS];
+static int mge[SNDRV_CARDS];
+static unsigned int dmaio = 0x7a00; /* DDMA i/o address */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for S3 SonicVibes soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for S3 SonicVibes soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable S3 SonicVibes soundcard.");
+module_param_array(reverb, bool, NULL, 0444);
+MODULE_PARM_DESC(reverb, "Enable reverb (SRAM is present) for S3 SonicVibes soundcard.");
+module_param_array(mge, bool, NULL, 0444);
+MODULE_PARM_DESC(mge, "MIC Gain Enable for S3 SonicVibes soundcard.");
+module_param(dmaio, uint, 0444);
+MODULE_PARM_DESC(dmaio, "DDMA i/o base address for S3 SonicVibes soundcard.");
+
+/*
+ * Enhanced port direct registers
+ */
+
+#define SV_REG(sonic, x) ((sonic)->enh_port + SV_REG_##x)
+
+#define SV_REG_CONTROL 0x00 /* R/W: CODEC/Mixer control register */
+#define SV_ENHANCED 0x01 /* audio mode select - enhanced mode */
+#define SV_TEST 0x02 /* test bit */
+#define SV_REVERB 0x04 /* reverb enable */
+#define SV_WAVETABLE 0x08 /* wavetable active / FM active if not set */
+#define SV_INTA 0x20 /* INTA driving - should be always 1 */
+#define SV_RESET 0x80 /* reset chip */
+#define SV_REG_IRQMASK 0x01 /* R/W: CODEC/Mixer interrupt mask register */
+#define SV_DMAA_MASK 0x01 /* mask DMA-A interrupt */
+#define SV_DMAC_MASK 0x04 /* mask DMA-C interrupt */
+#define SV_SPEC_MASK 0x08 /* special interrupt mask - should be always masked */
+#define SV_UD_MASK 0x40 /* Up/Down button interrupt mask */
+#define SV_MIDI_MASK 0x80 /* mask MIDI interrupt */
+#define SV_REG_STATUS 0x02 /* R/O: CODEC/Mixer status register */
+#define SV_DMAA_IRQ 0x01 /* DMA-A interrupt */
+#define SV_DMAC_IRQ 0x04 /* DMA-C interrupt */
+#define SV_SPEC_IRQ 0x08 /* special interrupt */
+#define SV_UD_IRQ 0x40 /* Up/Down interrupt */
+#define SV_MIDI_IRQ 0x80 /* MIDI interrupt */
+#define SV_REG_INDEX 0x04 /* R/W: CODEC/Mixer index address register */
+#define SV_MCE 0x40 /* mode change enable */
+#define SV_TRD 0x80 /* DMA transfer request disabled */
+#define SV_REG_DATA 0x05 /* R/W: CODEC/Mixer index data register */
+
+/*
+ * Enhanced port indirect registers
+ */
+
+#define SV_IREG_LEFT_ADC 0x00 /* Left ADC Input Control */
+#define SV_IREG_RIGHT_ADC 0x01 /* Right ADC Input Control */
+#define SV_IREG_LEFT_AUX1 0x02 /* Left AUX1 Input Control */
+#define SV_IREG_RIGHT_AUX1 0x03 /* Right AUX1 Input Control */
+#define SV_IREG_LEFT_CD 0x04 /* Left CD Input Control */
+#define SV_IREG_RIGHT_CD 0x05 /* Right CD Input Control */
+#define SV_IREG_LEFT_LINE 0x06 /* Left Line Input Control */
+#define SV_IREG_RIGHT_LINE 0x07 /* Right Line Input Control */
+#define SV_IREG_MIC 0x08 /* MIC Input Control */
+#define SV_IREG_GAME_PORT 0x09 /* Game Port Control */
+#define SV_IREG_LEFT_SYNTH 0x0a /* Left Synth Input Control */
+#define SV_IREG_RIGHT_SYNTH 0x0b /* Right Synth Input Control */
+#define SV_IREG_LEFT_AUX2 0x0c /* Left AUX2 Input Control */
+#define SV_IREG_RIGHT_AUX2 0x0d /* Right AUX2 Input Control */
+#define SV_IREG_LEFT_ANALOG 0x0e /* Left Analog Mixer Output Control */
+#define SV_IREG_RIGHT_ANALOG 0x0f /* Right Analog Mixer Output Control */
+#define SV_IREG_LEFT_PCM 0x10 /* Left PCM Input Control */
+#define SV_IREG_RIGHT_PCM 0x11 /* Right PCM Input Control */
+#define SV_IREG_DMA_DATA_FMT 0x12 /* DMA Data Format */
+#define SV_IREG_PC_ENABLE 0x13 /* Playback/Capture Enable Register */
+#define SV_IREG_UD_BUTTON 0x14 /* Up/Down Button Register */
+#define SV_IREG_REVISION 0x15 /* Revision */
+#define SV_IREG_ADC_OUTPUT_CTRL 0x16 /* ADC Output Control */
+#define SV_IREG_DMA_A_UPPER 0x18 /* DMA A Upper Base Count */
+#define SV_IREG_DMA_A_LOWER 0x19 /* DMA A Lower Base Count */
+#define SV_IREG_DMA_C_UPPER 0x1c /* DMA C Upper Base Count */
+#define SV_IREG_DMA_C_LOWER 0x1d /* DMA C Lower Base Count */
+#define SV_IREG_PCM_RATE_LOW 0x1e /* PCM Sampling Rate Low Byte */
+#define SV_IREG_PCM_RATE_HIGH 0x1f /* PCM Sampling Rate High Byte */
+#define SV_IREG_SYNTH_RATE_LOW 0x20 /* Synthesizer Sampling Rate Low Byte */
+#define SV_IREG_SYNTH_RATE_HIGH 0x21 /* Synthesizer Sampling Rate High Byte */
+#define SV_IREG_ADC_CLOCK 0x22 /* ADC Clock Source Selection */
+#define SV_IREG_ADC_ALT_RATE 0x23 /* ADC Alternative Sampling Rate Selection */
+#define SV_IREG_ADC_PLL_M 0x24 /* ADC PLL M Register */
+#define SV_IREG_ADC_PLL_N 0x25 /* ADC PLL N Register */
+#define SV_IREG_SYNTH_PLL_M 0x26 /* Synthesizer PLL M Register */
+#define SV_IREG_SYNTH_PLL_N 0x27 /* Synthesizer PLL N Register */
+#define SV_IREG_MPU401 0x2a /* MPU-401 UART Operation */
+#define SV_IREG_DRIVE_CTRL 0x2b /* Drive Control */
+#define SV_IREG_SRS_SPACE 0x2c /* SRS Space Control */
+#define SV_IREG_SRS_CENTER 0x2d /* SRS Center Control */
+#define SV_IREG_WAVE_SOURCE 0x2e /* Wavetable Sample Source Select */
+#define SV_IREG_ANALOG_POWER 0x30 /* Analog Power Down Control */
+#define SV_IREG_DIGITAL_POWER 0x31 /* Digital Power Down Control */
+
+#define SV_IREG_ADC_PLL SV_IREG_ADC_PLL_M
+#define SV_IREG_SYNTH_PLL SV_IREG_SYNTH_PLL_M
+
+/*
+ * DMA registers
+ */
+
+#define SV_DMA_ADDR0 0x00
+#define SV_DMA_ADDR1 0x01
+#define SV_DMA_ADDR2 0x02
+#define SV_DMA_ADDR3 0x03
+#define SV_DMA_COUNT0 0x04
+#define SV_DMA_COUNT1 0x05
+#define SV_DMA_COUNT2 0x06
+#define SV_DMA_MODE 0x0b
+#define SV_DMA_RESET 0x0d
+#define SV_DMA_MASK 0x0f
+
+/*
+ * Record sources
+ */
+
+#define SV_RECSRC_RESERVED (0x00<<5)
+#define SV_RECSRC_CD (0x01<<5)
+#define SV_RECSRC_DAC (0x02<<5)
+#define SV_RECSRC_AUX2 (0x03<<5)
+#define SV_RECSRC_LINE (0x04<<5)
+#define SV_RECSRC_AUX1 (0x05<<5)
+#define SV_RECSRC_MIC (0x06<<5)
+#define SV_RECSRC_OUT (0x07<<5)
+
+/*
+ * constants
+ */
+
+#define SV_FULLRATE 48000
+#define SV_REFFREQUENCY 24576000
+#define SV_ADCMULT 512
+
+#define SV_MODE_PLAY 1
+#define SV_MODE_CAPTURE 2
+
+/*
+
+ */
+
+struct sonicvibes {
+ unsigned long dma1size;
+ unsigned long dma2size;
+ int irq;
+
+ unsigned long sb_port;
+ unsigned long enh_port;
+ unsigned long synth_port;
+ unsigned long midi_port;
+ unsigned long game_port;
+ unsigned int dmaa_port;
+ struct resource *res_dmaa;
+ unsigned int dmac_port;
+ struct resource *res_dmac;
+
+ unsigned char enable;
+ unsigned char irqmask;
+ unsigned char revision;
+ unsigned char format;
+ unsigned char srs_space;
+ unsigned char srs_center;
+ unsigned char mpu_switch;
+ unsigned char wave_source;
+
+ unsigned int mode;
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+ struct snd_rawmidi *rmidi;
+ struct snd_hwdep *fmsynth; /* S3FM */
+
+ spinlock_t reg_lock;
+
+ unsigned int p_dma_size;
+ unsigned int c_dma_size;
+
+ struct snd_kcontrol *master_mute;
+ struct snd_kcontrol *master_volume;
+
+#ifdef SUPPORT_JOYSTICK
+ struct gameport *gameport;
+#endif
+};
+
+static struct pci_device_id snd_sonic_ids[] = {
+ { 0x5333, 0xca00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_sonic_ids);
+
+static struct snd_ratden sonicvibes_adc_clock = {
+ .num_min = 4000 * 65536,
+ .num_max = 48000UL * 65536,
+ .num_step = 1,
+ .den = 65536,
+};
+static struct snd_pcm_hw_constraint_ratdens snd_sonicvibes_hw_constraints_adc_clock = {
+ .nrats = 1,
+ .rats = &sonicvibes_adc_clock,
+};
+
+/*
+ * common I/O routines
+ */
+
+static inline void snd_sonicvibes_setdmaa(struct sonicvibes * sonic,
+ unsigned int addr,
+ unsigned int count)
+{
+ count--;
+ outl(addr, sonic->dmaa_port + SV_DMA_ADDR0);
+ outl(count, sonic->dmaa_port + SV_DMA_COUNT0);
+ outb(0x18, sonic->dmaa_port + SV_DMA_MODE);
+#if 0
+ printk("program dmaa: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmaa_port + SV_DMA_ADDR0));
+#endif
+}
+
+static inline void snd_sonicvibes_setdmac(struct sonicvibes * sonic,
+ unsigned int addr,
+ unsigned int count)
+{
+ /* note: dmac is working in word mode!!! */
+ count >>= 1;
+ count--;
+ outl(addr, sonic->dmac_port + SV_DMA_ADDR0);
+ outl(count, sonic->dmac_port + SV_DMA_COUNT0);
+ outb(0x14, sonic->dmac_port + SV_DMA_MODE);
+#if 0
+ printk("program dmac: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmac_port + SV_DMA_ADDR0));
+#endif
+}
+
+static inline unsigned int snd_sonicvibes_getdmaa(struct sonicvibes * sonic)
+{
+ return (inl(sonic->dmaa_port + SV_DMA_COUNT0) & 0xffffff) + 1;
+}
+
+static inline unsigned int snd_sonicvibes_getdmac(struct sonicvibes * sonic)
+{
+ /* note: dmac is working in word mode!!! */
+ return ((inl(sonic->dmac_port + SV_DMA_COUNT0) & 0xffffff) + 1) << 1;
+}
+
+static void snd_sonicvibes_out1(struct sonicvibes * sonic,
+ unsigned char reg,
+ unsigned char value)
+{
+ outb(reg, SV_REG(sonic, INDEX));
+ udelay(10);
+ outb(value, SV_REG(sonic, DATA));
+ udelay(10);
+}
+
+static void snd_sonicvibes_out(struct sonicvibes * sonic,
+ unsigned char reg,
+ unsigned char value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sonic->reg_lock, flags);
+ outb(reg, SV_REG(sonic, INDEX));
+ udelay(10);
+ outb(value, SV_REG(sonic, DATA));
+ udelay(10);
+ spin_unlock_irqrestore(&sonic->reg_lock, flags);
+}
+
+static unsigned char snd_sonicvibes_in1(struct sonicvibes * sonic, unsigned char reg)
+{
+ unsigned char value;
+
+ outb(reg, SV_REG(sonic, INDEX));
+ udelay(10);
+ value = inb(SV_REG(sonic, DATA));
+ udelay(10);
+ return value;
+}
+
+static unsigned char snd_sonicvibes_in(struct sonicvibes * sonic, unsigned char reg)
+{
+ unsigned long flags;
+ unsigned char value;
+
+ spin_lock_irqsave(&sonic->reg_lock, flags);
+ outb(reg, SV_REG(sonic, INDEX));
+ udelay(10);
+ value = inb(SV_REG(sonic, DATA));
+ udelay(10);
+ spin_unlock_irqrestore(&sonic->reg_lock, flags);
+ return value;
+}
+
+#if 0
+static void snd_sonicvibes_debug(struct sonicvibes * sonic)
+{
+ printk("SV REGS: INDEX = 0x%02x ", inb(SV_REG(sonic, INDEX)));
+ printk(" STATUS = 0x%02x\n", inb(SV_REG(sonic, STATUS)));
+ printk(" 0x00: left input = 0x%02x ", snd_sonicvibes_in(sonic, 0x00));
+ printk(" 0x20: synth rate low = 0x%02x\n", snd_sonicvibes_in(sonic, 0x20));
+ printk(" 0x01: right input = 0x%02x ", snd_sonicvibes_in(sonic, 0x01));
+ printk(" 0x21: synth rate high = 0x%02x\n", snd_sonicvibes_in(sonic, 0x21));
+ printk(" 0x02: left AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x02));
+ printk(" 0x22: ADC clock = 0x%02x\n", snd_sonicvibes_in(sonic, 0x22));
+ printk(" 0x03: right AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x03));
+ printk(" 0x23: ADC alt rate = 0x%02x\n", snd_sonicvibes_in(sonic, 0x23));
+ printk(" 0x04: left CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x04));
+ printk(" 0x24: ADC pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x24));
+ printk(" 0x05: right CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x05));
+ printk(" 0x25: ADC pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x25));
+ printk(" 0x06: left line = 0x%02x ", snd_sonicvibes_in(sonic, 0x06));
+ printk(" 0x26: Synth pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x26));
+ printk(" 0x07: right line = 0x%02x ", snd_sonicvibes_in(sonic, 0x07));
+ printk(" 0x27: Synth pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x27));
+ printk(" 0x08: MIC = 0x%02x ", snd_sonicvibes_in(sonic, 0x08));
+ printk(" 0x28: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x28));
+ printk(" 0x09: Game port = 0x%02x ", snd_sonicvibes_in(sonic, 0x09));
+ printk(" 0x29: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x29));
+ printk(" 0x0a: left synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0a));
+ printk(" 0x2a: MPU401 = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2a));
+ printk(" 0x0b: right synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0b));
+ printk(" 0x2b: drive ctrl = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2b));
+ printk(" 0x0c: left AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0c));
+ printk(" 0x2c: SRS space = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2c));
+ printk(" 0x0d: right AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0d));
+ printk(" 0x2d: SRS center = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2d));
+ printk(" 0x0e: left analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0e));
+ printk(" 0x2e: wave source = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2e));
+ printk(" 0x0f: right analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0f));
+ printk(" 0x2f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2f));
+ printk(" 0x10: left PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x10));
+ printk(" 0x30: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x30));
+ printk(" 0x11: right PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x11));
+ printk(" 0x31: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x31));
+ printk(" 0x12: DMA data format = 0x%02x ", snd_sonicvibes_in(sonic, 0x12));
+ printk(" 0x32: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x32));
+ printk(" 0x13: P/C enable = 0x%02x ", snd_sonicvibes_in(sonic, 0x13));
+ printk(" 0x33: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x33));
+ printk(" 0x14: U/D button = 0x%02x ", snd_sonicvibes_in(sonic, 0x14));
+ printk(" 0x34: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x34));
+ printk(" 0x15: revision = 0x%02x ", snd_sonicvibes_in(sonic, 0x15));
+ printk(" 0x35: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x35));
+ printk(" 0x16: ADC output ctrl = 0x%02x ", snd_sonicvibes_in(sonic, 0x16));
+ printk(" 0x36: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x36));
+ printk(" 0x17: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x17));
+ printk(" 0x37: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x37));
+ printk(" 0x18: DMA A upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x18));
+ printk(" 0x38: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x38));
+ printk(" 0x19: DMA A lower cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x19));
+ printk(" 0x39: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x39));
+ printk(" 0x1a: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1a));
+ printk(" 0x3a: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3a));
+ printk(" 0x1b: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1b));
+ printk(" 0x3b: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3b));
+ printk(" 0x1c: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1c));
+ printk(" 0x3c: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3c));
+ printk(" 0x1d: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1d));
+ printk(" 0x3d: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3d));
+ printk(" 0x1e: PCM rate low = 0x%02x ", snd_sonicvibes_in(sonic, 0x1e));
+ printk(" 0x3e: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3e));
+ printk(" 0x1f: PCM rate high = 0x%02x ", snd_sonicvibes_in(sonic, 0x1f));
+ printk(" 0x3f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3f));
+}
+
+#endif
+
+static void snd_sonicvibes_setfmt(struct sonicvibes * sonic,
+ unsigned char mask,
+ unsigned char value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sonic->reg_lock, flags);
+ outb(SV_MCE | SV_IREG_DMA_DATA_FMT, SV_REG(sonic, INDEX));
+ if (mask) {
+ sonic->format = inb(SV_REG(sonic, DATA));
+ udelay(10);
+ }
+ sonic->format = (sonic->format & mask) | value;
+ outb(sonic->format, SV_REG(sonic, DATA));
+ udelay(10);
+ outb(0, SV_REG(sonic, INDEX));
+ udelay(10);
+ spin_unlock_irqrestore(&sonic->reg_lock, flags);
+}
+
+static void snd_sonicvibes_pll(unsigned int rate,
+ unsigned int *res_r,
+ unsigned int *res_m,
+ unsigned int *res_n)
+{
+ unsigned int r, m = 0, n = 0;
+ unsigned int xm, xn, xr, xd, metric = ~0U;
+
+ if (rate < 625000 / SV_ADCMULT)
+ rate = 625000 / SV_ADCMULT;
+ if (rate > 150000000 / SV_ADCMULT)
+ rate = 150000000 / SV_ADCMULT;
+ /* slight violation of specs, needed for continuous sampling rates */
+ for (r = 0; rate < 75000000 / SV_ADCMULT; r += 0x20, rate <<= 1);
+ for (xn = 3; xn < 33; xn++) /* 35 */
+ for (xm = 3; xm < 257; xm++) {
+ xr = ((SV_REFFREQUENCY / SV_ADCMULT) * xm) / xn;
+ if (xr >= rate)
+ xd = xr - rate;
+ else
+ xd = rate - xr;
+ if (xd < metric) {
+ metric = xd;
+ m = xm - 2;
+ n = xn - 2;
+ }
+ }
+ *res_r = r;
+ *res_m = m;
+ *res_n = n;
+#if 0
+ printk("metric = %i, xm = %i, xn = %i\n", metric, xm, xn);
+ printk("pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n);
+#endif
+}
+
+static void snd_sonicvibes_setpll(struct sonicvibes * sonic,
+ unsigned char reg,
+ unsigned int rate)
+{
+ unsigned long flags;
+ unsigned int r, m, n;
+
+ snd_sonicvibes_pll(rate, &r, &m, &n);
+ if (sonic != NULL) {
+ spin_lock_irqsave(&sonic->reg_lock, flags);
+ snd_sonicvibes_out1(sonic, reg, m);
+ snd_sonicvibes_out1(sonic, reg + 1, r | n);
+ spin_unlock_irqrestore(&sonic->reg_lock, flags);
+ }
+}
+
+static void snd_sonicvibes_set_adc_rate(struct sonicvibes * sonic, unsigned int rate)
+{
+ unsigned long flags;
+ unsigned int div;
+ unsigned char clock;
+
+ div = 48000 / rate;
+ if (div > 8)
+ div = 8;
+ if ((48000 / div) == rate) { /* use the alternate clock */
+ clock = 0x10;
+ } else { /* use the PLL source */
+ clock = 0x00;
+ snd_sonicvibes_setpll(sonic, SV_IREG_ADC_PLL, rate);
+ }
+ spin_lock_irqsave(&sonic->reg_lock, flags);
+ snd_sonicvibes_out1(sonic, SV_IREG_ADC_ALT_RATE, (div - 1) << 4);
+ snd_sonicvibes_out1(sonic, SV_IREG_ADC_CLOCK, clock);
+ spin_unlock_irqrestore(&sonic->reg_lock, flags);
+}
+
+static int snd_sonicvibes_hw_constraint_dac_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ unsigned int rate, div, r, m, n;
+
+ if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min ==
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max) {
+ rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min;
+ div = 48000 / rate;
+ if (div > 8)
+ div = 8;
+ if ((48000 / div) == rate) {
+ params->rate_num = rate;
+ params->rate_den = 1;
+ } else {
+ snd_sonicvibes_pll(rate, &r, &m, &n);
+ snd_BUG_ON(SV_REFFREQUENCY % 16);
+ snd_BUG_ON(SV_ADCMULT % 512);
+ params->rate_num = (SV_REFFREQUENCY/16) * (n+2) * r;
+ params->rate_den = (SV_ADCMULT/512) * (m+2);
+ }
+ }
+ return 0;
+}
+
+static void snd_sonicvibes_set_dac_rate(struct sonicvibes * sonic, unsigned int rate)
+{
+ unsigned int div;
+ unsigned long flags;
+
+ div = (rate * 65536 + SV_FULLRATE / 2) / SV_FULLRATE;
+ if (div > 65535)
+ div = 65535;
+ spin_lock_irqsave(&sonic->reg_lock, flags);
+ snd_sonicvibes_out1(sonic, SV_IREG_PCM_RATE_HIGH, div >> 8);
+ snd_sonicvibes_out1(sonic, SV_IREG_PCM_RATE_LOW, div);
+ spin_unlock_irqrestore(&sonic->reg_lock, flags);
+}
+
+static int snd_sonicvibes_trigger(struct sonicvibes * sonic, int what, int cmd)
+{
+ int result = 0;
+
+ spin_lock(&sonic->reg_lock);
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ if (!(sonic->enable & what)) {
+ sonic->enable |= what;
+ snd_sonicvibes_out1(sonic, SV_IREG_PC_ENABLE, sonic->enable);
+ }
+ } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ if (sonic->enable & what) {
+ sonic->enable &= ~what;
+ snd_sonicvibes_out1(sonic, SV_IREG_PC_ENABLE, sonic->enable);
+ }
+ } else {
+ result = -EINVAL;
+ }
+ spin_unlock(&sonic->reg_lock);
+ return result;
+}
+
+static irqreturn_t snd_sonicvibes_interrupt(int irq, void *dev_id)
+{
+ struct sonicvibes *sonic = dev_id;
+ unsigned char status;
+
+ status = inb(SV_REG(sonic, STATUS));
+ if (!(status & (SV_DMAA_IRQ | SV_DMAC_IRQ | SV_MIDI_IRQ)))
+ return IRQ_NONE;
+ if (status == 0xff) { /* failure */
+ outb(sonic->irqmask = ~0, SV_REG(sonic, IRQMASK));
+ snd_printk(KERN_ERR "IRQ failure - interrupts disabled!!\n");
+ return IRQ_HANDLED;
+ }
+ if (sonic->pcm) {
+ if (status & SV_DMAA_IRQ)
+ snd_pcm_period_elapsed(sonic->playback_substream);
+ if (status & SV_DMAC_IRQ)
+ snd_pcm_period_elapsed(sonic->capture_substream);
+ }
+ if (sonic->rmidi) {
+ if (status & SV_MIDI_IRQ)
+ snd_mpu401_uart_interrupt(irq, sonic->rmidi->private_data);
+ }
+ if (status & SV_UD_IRQ) {
+ unsigned char udreg;
+ int vol, oleft, oright, mleft, mright;
+
+ spin_lock(&sonic->reg_lock);
+ udreg = snd_sonicvibes_in1(sonic, SV_IREG_UD_BUTTON);
+ vol = udreg & 0x3f;
+ if (!(udreg & 0x40))
+ vol = -vol;
+ oleft = mleft = snd_sonicvibes_in1(sonic, SV_IREG_LEFT_ANALOG);
+ oright = mright = snd_sonicvibes_in1(sonic, SV_IREG_RIGHT_ANALOG);
+ oleft &= 0x1f;
+ oright &= 0x1f;
+ oleft += vol;
+ if (oleft < 0)
+ oleft = 0;
+ if (oleft > 0x1f)
+ oleft = 0x1f;
+ oright += vol;
+ if (oright < 0)
+ oright = 0;
+ if (oright > 0x1f)
+ oright = 0x1f;
+ if (udreg & 0x80) {
+ mleft ^= 0x80;
+ mright ^= 0x80;
+ }
+ oleft |= mleft & 0x80;
+ oright |= mright & 0x80;
+ snd_sonicvibes_out1(sonic, SV_IREG_LEFT_ANALOG, oleft);
+ snd_sonicvibes_out1(sonic, SV_IREG_RIGHT_ANALOG, oright);
+ spin_unlock(&sonic->reg_lock);
+ snd_ctl_notify(sonic->card, SNDRV_CTL_EVENT_MASK_VALUE, &sonic->master_mute->id);
+ snd_ctl_notify(sonic->card, SNDRV_CTL_EVENT_MASK_VALUE, &sonic->master_volume->id);
+ }
+ return IRQ_HANDLED;
+}
+
+/*
+ * PCM part
+ */
+
+static int snd_sonicvibes_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ return snd_sonicvibes_trigger(sonic, 1, cmd);
+}
+
+static int snd_sonicvibes_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ return snd_sonicvibes_trigger(sonic, 2, cmd);
+}
+
+static int snd_sonicvibes_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_sonicvibes_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_sonicvibes_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned char fmt = 0;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ sonic->p_dma_size = size;
+ count--;
+ if (runtime->channels > 1)
+ fmt |= 1;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ fmt |= 2;
+ snd_sonicvibes_setfmt(sonic, ~3, fmt);
+ snd_sonicvibes_set_dac_rate(sonic, runtime->rate);
+ spin_lock_irq(&sonic->reg_lock);
+ snd_sonicvibes_setdmaa(sonic, runtime->dma_addr, size);
+ snd_sonicvibes_out1(sonic, SV_IREG_DMA_A_UPPER, count >> 8);
+ snd_sonicvibes_out1(sonic, SV_IREG_DMA_A_LOWER, count);
+ spin_unlock_irq(&sonic->reg_lock);
+ return 0;
+}
+
+static int snd_sonicvibes_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned char fmt = 0;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned int count = snd_pcm_lib_period_bytes(substream);
+
+ sonic->c_dma_size = size;
+ count >>= 1;
+ count--;
+ if (runtime->channels > 1)
+ fmt |= 0x10;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ fmt |= 0x20;
+ snd_sonicvibes_setfmt(sonic, ~0x30, fmt);
+ snd_sonicvibes_set_adc_rate(sonic, runtime->rate);
+ spin_lock_irq(&sonic->reg_lock);
+ snd_sonicvibes_setdmac(sonic, runtime->dma_addr, size);
+ snd_sonicvibes_out1(sonic, SV_IREG_DMA_C_UPPER, count >> 8);
+ snd_sonicvibes_out1(sonic, SV_IREG_DMA_C_LOWER, count);
+ spin_unlock_irq(&sonic->reg_lock);
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_sonicvibes_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(sonic->enable & 1))
+ return 0;
+ ptr = sonic->p_dma_size - snd_sonicvibes_getdmaa(sonic);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_sonicvibes_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ size_t ptr;
+ if (!(sonic->enable & 2))
+ return 0;
+ ptr = sonic->c_dma_size - snd_sonicvibes_getdmac(sonic);
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static struct snd_pcm_hardware snd_sonicvibes_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 32,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_sonicvibes_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 32,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_sonicvibes_playback_open(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ sonic->mode |= SV_MODE_PLAY;
+ sonic->playback_substream = substream;
+ runtime->hw = snd_sonicvibes_playback;
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, snd_sonicvibes_hw_constraint_dac_rate, NULL, SNDRV_PCM_HW_PARAM_RATE, -1);
+ return 0;
+}
+
+static int snd_sonicvibes_capture_open(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ sonic->mode |= SV_MODE_CAPTURE;
+ sonic->capture_substream = substream;
+ runtime->hw = snd_sonicvibes_capture;
+ snd_pcm_hw_constraint_ratdens(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &snd_sonicvibes_hw_constraints_adc_clock);
+ return 0;
+}
+
+static int snd_sonicvibes_playback_close(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+
+ sonic->playback_substream = NULL;
+ sonic->mode &= ~SV_MODE_PLAY;
+ return 0;
+}
+
+static int snd_sonicvibes_capture_close(struct snd_pcm_substream *substream)
+{
+ struct sonicvibes *sonic = snd_pcm_substream_chip(substream);
+
+ sonic->capture_substream = NULL;
+ sonic->mode &= ~SV_MODE_CAPTURE;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_sonicvibes_playback_ops = {
+ .open = snd_sonicvibes_playback_open,
+ .close = snd_sonicvibes_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sonicvibes_hw_params,
+ .hw_free = snd_sonicvibes_hw_free,
+ .prepare = snd_sonicvibes_playback_prepare,
+ .trigger = snd_sonicvibes_playback_trigger,
+ .pointer = snd_sonicvibes_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_sonicvibes_capture_ops = {
+ .open = snd_sonicvibes_capture_open,
+ .close = snd_sonicvibes_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sonicvibes_hw_params,
+ .hw_free = snd_sonicvibes_hw_free,
+ .prepare = snd_sonicvibes_capture_prepare,
+ .trigger = snd_sonicvibes_capture_trigger,
+ .pointer = snd_sonicvibes_capture_pointer,
+};
+
+static int __devinit snd_sonicvibes_pcm(struct sonicvibes * sonic, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(sonic->card, "s3_86c617", device, 1, 1, &pcm)) < 0)
+ return err;
+ if (snd_BUG_ON(!pcm))
+ return -EINVAL;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sonicvibes_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sonicvibes_capture_ops);
+
+ pcm->private_data = sonic;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "S3 SonicVibes");
+ sonic->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(sonic->pci), 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/*
+ * Mixer part
+ */
+
+#define SONICVIBES_MUX(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_sonicvibes_info_mux, \
+ .get = snd_sonicvibes_get_mux, .put = snd_sonicvibes_put_mux }
+
+static int snd_sonicvibes_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[7] = {
+ "CD", "PCM", "Aux1", "Line", "Aux0", "Mic", "Mix"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 2;
+ uinfo->value.enumerated.items = 7;
+ if (uinfo->value.enumerated.item >= 7)
+ uinfo->value.enumerated.item = 6;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_sonicvibes_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&sonic->reg_lock);
+ ucontrol->value.enumerated.item[0] = ((snd_sonicvibes_in1(sonic, SV_IREG_LEFT_ADC) & SV_RECSRC_OUT) >> 5) - 1;
+ ucontrol->value.enumerated.item[1] = ((snd_sonicvibes_in1(sonic, SV_IREG_RIGHT_ADC) & SV_RECSRC_OUT) >> 5) - 1;
+ spin_unlock_irq(&sonic->reg_lock);
+ return 0;
+}
+
+static int snd_sonicvibes_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol);
+ unsigned short left, right, oval1, oval2;
+ int change;
+
+ if (ucontrol->value.enumerated.item[0] >= 7 ||
+ ucontrol->value.enumerated.item[1] >= 7)
+ return -EINVAL;
+ left = (ucontrol->value.enumerated.item[0] + 1) << 5;
+ right = (ucontrol->value.enumerated.item[1] + 1) << 5;
+ spin_lock_irq(&sonic->reg_lock);
+ oval1 = snd_sonicvibes_in1(sonic, SV_IREG_LEFT_ADC);
+ oval2 = snd_sonicvibes_in1(sonic, SV_IREG_RIGHT_ADC);
+ left = (oval1 & ~SV_RECSRC_OUT) | left;
+ right = (oval2 & ~SV_RECSRC_OUT) | right;
+ change = left != oval1 || right != oval2;
+ snd_sonicvibes_out1(sonic, SV_IREG_LEFT_ADC, left);
+ snd_sonicvibes_out1(sonic, SV_IREG_RIGHT_ADC, right);
+ spin_unlock_irq(&sonic->reg_lock);
+ return change;
+}
+
+#define SONICVIBES_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_sonicvibes_info_single, \
+ .get = snd_sonicvibes_get_single, .put = snd_sonicvibes_put_single, \
+ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) }
+
+static int snd_sonicvibes_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_sonicvibes_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irq(&sonic->reg_lock);
+ ucontrol->value.integer.value[0] = (snd_sonicvibes_in1(sonic, reg)>> shift) & mask;
+ spin_unlock_irq(&sonic->reg_lock);
+ if (invert)
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_sonicvibes_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short val, oval;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+ spin_lock_irq(&sonic->reg_lock);
+ oval = snd_sonicvibes_in1(sonic, reg);
+ val = (oval & ~(mask << shift)) | val;
+ change = val != oval;
+ snd_sonicvibes_out1(sonic, reg, val);
+ spin_unlock_irq(&sonic->reg_lock);
+ return change;
+}
+
+#define SONICVIBES_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_sonicvibes_info_double, \
+ .get = snd_sonicvibes_get_double, .put = snd_sonicvibes_put_double, \
+ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) }
+
+static int snd_sonicvibes_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_sonicvibes_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+
+ spin_lock_irq(&sonic->reg_lock);
+ ucontrol->value.integer.value[0] = (snd_sonicvibes_in1(sonic, left_reg) >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (snd_sonicvibes_in1(sonic, right_reg) >> shift_right) & mask;
+ spin_unlock_irq(&sonic->reg_lock);
+ if (invert) {
+ ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
+ }
+ return 0;
+}
+
+static int snd_sonicvibes_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned short val1, val2, oval1, oval2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irq(&sonic->reg_lock);
+ oval1 = snd_sonicvibes_in1(sonic, left_reg);
+ oval2 = snd_sonicvibes_in1(sonic, right_reg);
+ val1 = (oval1 & ~(mask << shift_left)) | val1;
+ val2 = (oval2 & ~(mask << shift_right)) | val2;
+ change = val1 != oval1 || val2 != oval2;
+ snd_sonicvibes_out1(sonic, left_reg, val1);
+ snd_sonicvibes_out1(sonic, right_reg, val2);
+ spin_unlock_irq(&sonic->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_sonicvibes_controls[] __devinitdata = {
+SONICVIBES_DOUBLE("Capture Volume", 0, SV_IREG_LEFT_ADC, SV_IREG_RIGHT_ADC, 0, 0, 15, 0),
+SONICVIBES_DOUBLE("Aux Playback Switch", 0, SV_IREG_LEFT_AUX1, SV_IREG_RIGHT_AUX1, 7, 7, 1, 1),
+SONICVIBES_DOUBLE("Aux Playback Volume", 0, SV_IREG_LEFT_AUX1, SV_IREG_RIGHT_AUX1, 0, 0, 31, 1),
+SONICVIBES_DOUBLE("CD Playback Switch", 0, SV_IREG_LEFT_CD, SV_IREG_RIGHT_CD, 7, 7, 1, 1),
+SONICVIBES_DOUBLE("CD Playback Volume", 0, SV_IREG_LEFT_CD, SV_IREG_RIGHT_CD, 0, 0, 31, 1),
+SONICVIBES_DOUBLE("Line Playback Switch", 0, SV_IREG_LEFT_LINE, SV_IREG_RIGHT_LINE, 7, 7, 1, 1),
+SONICVIBES_DOUBLE("Line Playback Volume", 0, SV_IREG_LEFT_LINE, SV_IREG_RIGHT_LINE, 0, 0, 31, 1),
+SONICVIBES_SINGLE("Mic Playback Switch", 0, SV_IREG_MIC, 7, 1, 1),
+SONICVIBES_SINGLE("Mic Playback Volume", 0, SV_IREG_MIC, 0, 15, 1),
+SONICVIBES_SINGLE("Mic Boost", 0, SV_IREG_LEFT_ADC, 4, 1, 0),
+SONICVIBES_DOUBLE("Synth Playback Switch", 0, SV_IREG_LEFT_SYNTH, SV_IREG_RIGHT_SYNTH, 7, 7, 1, 1),
+SONICVIBES_DOUBLE("Synth Playback Volume", 0, SV_IREG_LEFT_SYNTH, SV_IREG_RIGHT_SYNTH, 0, 0, 31, 1),
+SONICVIBES_DOUBLE("Aux Playback Switch", 1, SV_IREG_LEFT_AUX2, SV_IREG_RIGHT_AUX2, 7, 7, 1, 1),
+SONICVIBES_DOUBLE("Aux Playback Volume", 1, SV_IREG_LEFT_AUX2, SV_IREG_RIGHT_AUX2, 0, 0, 31, 1),
+SONICVIBES_DOUBLE("Master Playback Switch", 0, SV_IREG_LEFT_ANALOG, SV_IREG_RIGHT_ANALOG, 7, 7, 1, 1),
+SONICVIBES_DOUBLE("Master Playback Volume", 0, SV_IREG_LEFT_ANALOG, SV_IREG_RIGHT_ANALOG, 0, 0, 31, 1),
+SONICVIBES_DOUBLE("PCM Playback Switch", 0, SV_IREG_LEFT_PCM, SV_IREG_RIGHT_PCM, 7, 7, 1, 1),
+SONICVIBES_DOUBLE("PCM Playback Volume", 0, SV_IREG_LEFT_PCM, SV_IREG_RIGHT_PCM, 0, 0, 63, 1),
+SONICVIBES_SINGLE("Loopback Capture Switch", 0, SV_IREG_ADC_OUTPUT_CTRL, 0, 1, 0),
+SONICVIBES_SINGLE("Loopback Capture Volume", 0, SV_IREG_ADC_OUTPUT_CTRL, 2, 63, 1),
+SONICVIBES_MUX("Capture Source", 0)
+};
+
+static void snd_sonicvibes_master_free(struct snd_kcontrol *kcontrol)
+{
+ struct sonicvibes *sonic = snd_kcontrol_chip(kcontrol);
+ sonic->master_mute = NULL;
+ sonic->master_volume = NULL;
+}
+
+static int __devinit snd_sonicvibes_mixer(struct sonicvibes * sonic)
+{
+ struct snd_card *card;
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+ int err;
+
+ if (snd_BUG_ON(!sonic || !sonic->card))
+ return -EINVAL;
+ card = sonic->card;
+ strcpy(card->mixername, "S3 SonicVibes");
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_sonicvibes_controls); idx++) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_sonicvibes_controls[idx], sonic))) < 0)
+ return err;
+ switch (idx) {
+ case 0:
+ case 1: kctl->private_free = snd_sonicvibes_master_free; break;
+ }
+ }
+ return 0;
+}
+
+/*
+
+ */
+
+static void snd_sonicvibes_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct sonicvibes *sonic = entry->private_data;
+ unsigned char tmp;
+
+ tmp = sonic->srs_space & 0x0f;
+ snd_iprintf(buffer, "SRS 3D : %s\n",
+ sonic->srs_space & 0x80 ? "off" : "on");
+ snd_iprintf(buffer, "SRS Space : %s\n",
+ tmp == 0x00 ? "100%" :
+ tmp == 0x01 ? "75%" :
+ tmp == 0x02 ? "50%" :
+ tmp == 0x03 ? "25%" : "0%");
+ tmp = sonic->srs_center & 0x0f;
+ snd_iprintf(buffer, "SRS Center : %s\n",
+ tmp == 0x00 ? "100%" :
+ tmp == 0x01 ? "75%" :
+ tmp == 0x02 ? "50%" :
+ tmp == 0x03 ? "25%" : "0%");
+ tmp = sonic->wave_source & 0x03;
+ snd_iprintf(buffer, "WaveTable Source : %s\n",
+ tmp == 0x00 ? "on-board ROM" :
+ tmp == 0x01 ? "PCI bus" : "on-board ROM + PCI bus");
+ tmp = sonic->mpu_switch;
+ snd_iprintf(buffer, "Onboard synth : %s\n", tmp & 0x01 ? "on" : "off");
+ snd_iprintf(buffer, "Ext. Rx to synth : %s\n", tmp & 0x02 ? "on" : "off");
+ snd_iprintf(buffer, "MIDI to ext. Tx : %s\n", tmp & 0x04 ? "on" : "off");
+}
+
+static void __devinit snd_sonicvibes_proc_init(struct sonicvibes * sonic)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(sonic->card, "sonicvibes", &entry))
+ snd_info_set_text_ops(entry, sonic, snd_sonicvibes_proc_read);
+}
+
+/*
+
+ */
+
+#ifdef SUPPORT_JOYSTICK
+static struct snd_kcontrol_new snd_sonicvibes_game_control __devinitdata =
+SONICVIBES_SINGLE("Joystick Speed", 0, SV_IREG_GAME_PORT, 1, 15, 0);
+
+static int __devinit snd_sonicvibes_create_gameport(struct sonicvibes *sonic)
+{
+ struct gameport *gp;
+
+ sonic->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "sonicvibes: cannot allocate memory for gameport\n");
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "SonicVibes Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(sonic->pci));
+ gameport_set_dev_parent(gp, &sonic->pci->dev);
+ gp->io = sonic->game_port;
+
+ gameport_register_port(gp);
+
+ snd_ctl_add(sonic->card, snd_ctl_new1(&snd_sonicvibes_game_control, sonic));
+
+ return 0;
+}
+
+static void snd_sonicvibes_free_gameport(struct sonicvibes *sonic)
+{
+ if (sonic->gameport) {
+ gameport_unregister_port(sonic->gameport);
+ sonic->gameport = NULL;
+ }
+}
+#else
+static inline int snd_sonicvibes_create_gameport(struct sonicvibes *sonic) { return -ENOSYS; }
+static inline void snd_sonicvibes_free_gameport(struct sonicvibes *sonic) { }
+#endif
+
+static int snd_sonicvibes_free(struct sonicvibes *sonic)
+{
+ snd_sonicvibes_free_gameport(sonic);
+ pci_write_config_dword(sonic->pci, 0x40, sonic->dmaa_port);
+ pci_write_config_dword(sonic->pci, 0x48, sonic->dmac_port);
+ if (sonic->irq >= 0)
+ free_irq(sonic->irq, sonic);
+ release_and_free_resource(sonic->res_dmaa);
+ release_and_free_resource(sonic->res_dmac);
+ pci_release_regions(sonic->pci);
+ pci_disable_device(sonic->pci);
+ kfree(sonic);
+ return 0;
+}
+
+static int snd_sonicvibes_dev_free(struct snd_device *device)
+{
+ struct sonicvibes *sonic = device->device_data;
+ return snd_sonicvibes_free(sonic);
+}
+
+static int __devinit snd_sonicvibes_create(struct snd_card *card,
+ struct pci_dev *pci,
+ int reverb,
+ int mge,
+ struct sonicvibes ** rsonic)
+{
+ struct sonicvibes *sonic;
+ unsigned int dmaa, dmac;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_sonicvibes_dev_free,
+ };
+
+ *rsonic = NULL;
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ /* check, if we can restrict PCI DMA transfers to 24 bits */
+ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ sonic = kzalloc(sizeof(*sonic), GFP_KERNEL);
+ if (sonic == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ spin_lock_init(&sonic->reg_lock);
+ sonic->card = card;
+ sonic->pci = pci;
+ sonic->irq = -1;
+
+ if ((err = pci_request_regions(pci, "S3 SonicVibes")) < 0) {
+ kfree(sonic);
+ pci_disable_device(pci);
+ return err;
+ }
+
+ sonic->sb_port = pci_resource_start(pci, 0);
+ sonic->enh_port = pci_resource_start(pci, 1);
+ sonic->synth_port = pci_resource_start(pci, 2);
+ sonic->midi_port = pci_resource_start(pci, 3);
+ sonic->game_port = pci_resource_start(pci, 4);
+
+ if (request_irq(pci->irq, snd_sonicvibes_interrupt, IRQF_SHARED,
+ "S3 SonicVibes", sonic)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_sonicvibes_free(sonic);
+ return -EBUSY;
+ }
+ sonic->irq = pci->irq;
+
+ pci_read_config_dword(pci, 0x40, &dmaa);
+ pci_read_config_dword(pci, 0x48, &dmac);
+ dmaio &= ~0x0f;
+ dmaa &= ~0x0f;
+ dmac &= ~0x0f;
+ if (!dmaa) {
+ dmaa = dmaio;
+ dmaio += 0x10;
+ snd_printk(KERN_INFO "BIOS did not allocate DDMA channel A i/o, allocated at 0x%x\n", dmaa);
+ }
+ if (!dmac) {
+ dmac = dmaio;
+ dmaio += 0x10;
+ snd_printk(KERN_INFO "BIOS did not allocate DDMA channel C i/o, allocated at 0x%x\n", dmac);
+ }
+ pci_write_config_dword(pci, 0x40, dmaa);
+ pci_write_config_dword(pci, 0x48, dmac);
+
+ if ((sonic->res_dmaa = request_region(dmaa, 0x10, "S3 SonicVibes DDMA-A")) == NULL) {
+ snd_sonicvibes_free(sonic);
+ snd_printk(KERN_ERR "unable to grab DDMA-A port at 0x%x-0x%x\n", dmaa, dmaa + 0x10 - 1);
+ return -EBUSY;
+ }
+ if ((sonic->res_dmac = request_region(dmac, 0x10, "S3 SonicVibes DDMA-C")) == NULL) {
+ snd_sonicvibes_free(sonic);
+ snd_printk(KERN_ERR "unable to grab DDMA-C port at 0x%x-0x%x\n", dmac, dmac + 0x10 - 1);
+ return -EBUSY;
+ }
+
+ pci_read_config_dword(pci, 0x40, &sonic->dmaa_port);
+ pci_read_config_dword(pci, 0x48, &sonic->dmac_port);
+ sonic->dmaa_port &= ~0x0f;
+ sonic->dmac_port &= ~0x0f;
+ pci_write_config_dword(pci, 0x40, sonic->dmaa_port | 9); /* enable + enhanced */
+ pci_write_config_dword(pci, 0x48, sonic->dmac_port | 9); /* enable */
+ /* ok.. initialize S3 SonicVibes chip */
+ outb(SV_RESET, SV_REG(sonic, CONTROL)); /* reset chip */
+ udelay(100);
+ outb(0, SV_REG(sonic, CONTROL)); /* release reset */
+ udelay(100);
+ outb(SV_ENHANCED | SV_INTA | (reverb ? SV_REVERB : 0), SV_REG(sonic, CONTROL));
+ inb(SV_REG(sonic, STATUS)); /* clear IRQs */
+#if 1
+ snd_sonicvibes_out(sonic, SV_IREG_DRIVE_CTRL, 0); /* drive current 16mA */
+#else
+ snd_sonicvibes_out(sonic, SV_IREG_DRIVE_CTRL, 0x40); /* drive current 8mA */
+#endif
+ snd_sonicvibes_out(sonic, SV_IREG_PC_ENABLE, sonic->enable = 0); /* disable playback & capture */
+ outb(sonic->irqmask = ~(SV_DMAA_MASK | SV_DMAC_MASK | SV_UD_MASK), SV_REG(sonic, IRQMASK));
+ inb(SV_REG(sonic, STATUS)); /* clear IRQs */
+ snd_sonicvibes_out(sonic, SV_IREG_ADC_CLOCK, 0); /* use PLL as clock source */
+ snd_sonicvibes_out(sonic, SV_IREG_ANALOG_POWER, 0); /* power up analog parts */
+ snd_sonicvibes_out(sonic, SV_IREG_DIGITAL_POWER, 0); /* power up digital parts */
+ snd_sonicvibes_setpll(sonic, SV_IREG_ADC_PLL, 8000);
+ snd_sonicvibes_out(sonic, SV_IREG_SRS_SPACE, sonic->srs_space = 0x80); /* SRS space off */
+ snd_sonicvibes_out(sonic, SV_IREG_SRS_CENTER, sonic->srs_center = 0x00);/* SRS center off */
+ snd_sonicvibes_out(sonic, SV_IREG_MPU401, sonic->mpu_switch = 0x05); /* MPU-401 switch */
+ snd_sonicvibes_out(sonic, SV_IREG_WAVE_SOURCE, sonic->wave_source = 0x00); /* onboard ROM */
+ snd_sonicvibes_out(sonic, SV_IREG_PCM_RATE_LOW, (8000 * 65536 / SV_FULLRATE) & 0xff);
+ snd_sonicvibes_out(sonic, SV_IREG_PCM_RATE_HIGH, ((8000 * 65536 / SV_FULLRATE) >> 8) & 0xff);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_ADC, mge ? 0xd0 : 0xc0);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_ADC, 0xc0);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_AUX1, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_AUX1, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_CD, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_CD, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_LINE, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_LINE, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_MIC, 0x8f);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_SYNTH, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_SYNTH, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_AUX2, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_AUX2, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_ANALOG, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_ANALOG, 0x9f);
+ snd_sonicvibes_out(sonic, SV_IREG_LEFT_PCM, 0xbf);
+ snd_sonicvibes_out(sonic, SV_IREG_RIGHT_PCM, 0xbf);
+ snd_sonicvibes_out(sonic, SV_IREG_ADC_OUTPUT_CTRL, 0xfc);
+#if 0
+ snd_sonicvibes_debug(sonic);
+#endif
+ sonic->revision = snd_sonicvibes_in(sonic, SV_IREG_REVISION);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sonic, &ops)) < 0) {
+ snd_sonicvibes_free(sonic);
+ return err;
+ }
+
+ snd_sonicvibes_proc_init(sonic);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rsonic = sonic;
+ return 0;
+}
+
+/*
+ * MIDI section
+ */
+
+static struct snd_kcontrol_new snd_sonicvibes_midi_controls[] __devinitdata = {
+SONICVIBES_SINGLE("SonicVibes Wave Source RAM", 0, SV_IREG_WAVE_SOURCE, 0, 1, 0),
+SONICVIBES_SINGLE("SonicVibes Wave Source RAM+ROM", 0, SV_IREG_WAVE_SOURCE, 1, 1, 0),
+SONICVIBES_SINGLE("SonicVibes Onboard Synth", 0, SV_IREG_MPU401, 0, 1, 0),
+SONICVIBES_SINGLE("SonicVibes External Rx to Synth", 0, SV_IREG_MPU401, 1, 1, 0),
+SONICVIBES_SINGLE("SonicVibes External Tx", 0, SV_IREG_MPU401, 2, 1, 0)
+};
+
+static int snd_sonicvibes_midi_input_open(struct snd_mpu401 * mpu)
+{
+ struct sonicvibes *sonic = mpu->private_data;
+ outb(sonic->irqmask &= ~SV_MIDI_MASK, SV_REG(sonic, IRQMASK));
+ return 0;
+}
+
+static void snd_sonicvibes_midi_input_close(struct snd_mpu401 * mpu)
+{
+ struct sonicvibes *sonic = mpu->private_data;
+ outb(sonic->irqmask |= SV_MIDI_MASK, SV_REG(sonic, IRQMASK));
+}
+
+static int __devinit snd_sonicvibes_midi(struct sonicvibes * sonic,
+ struct snd_rawmidi *rmidi)
+{
+ struct snd_mpu401 * mpu = rmidi->private_data;
+ struct snd_card *card = sonic->card;
+ struct snd_rawmidi_str *dir;
+ unsigned int idx;
+ int err;
+
+ mpu->private_data = sonic;
+ mpu->open_input = snd_sonicvibes_midi_input_open;
+ mpu->close_input = snd_sonicvibes_midi_input_close;
+ dir = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+ for (idx = 0; idx < ARRAY_SIZE(snd_sonicvibes_midi_controls); idx++)
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_sonicvibes_midi_controls[idx], sonic))) < 0)
+ return err;
+ return 0;
+}
+
+static int __devinit snd_sonic_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct sonicvibes *sonic;
+ struct snd_rawmidi *midi_uart;
+ struct snd_opl3 *opl3;
+ int idx, err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+ for (idx = 0; idx < 5; idx++) {
+ if (pci_resource_start(pci, idx) == 0 ||
+ !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) {
+ snd_card_free(card);
+ return -ENODEV;
+ }
+ }
+ if ((err = snd_sonicvibes_create(card, pci,
+ reverb[dev] ? 1 : 0,
+ mge[dev] ? 1 : 0,
+ &sonic)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "SonicVibes");
+ strcpy(card->shortname, "S3 SonicVibes");
+ sprintf(card->longname, "%s rev %i at 0x%llx, irq %i",
+ card->shortname,
+ sonic->revision,
+ (unsigned long long)pci_resource_start(pci, 1),
+ sonic->irq);
+
+ if ((err = snd_sonicvibes_pcm(sonic, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_sonicvibes_mixer(sonic)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES,
+ sonic->midi_port, MPU401_INFO_INTEGRATED,
+ sonic->irq, 0,
+ &midi_uart)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ snd_sonicvibes_midi(sonic, midi_uart);
+ if ((err = snd_opl3_create(card, sonic->synth_port,
+ sonic->synth_port + 2,
+ OPL3_HW_OPL3_SV, 1, &opl3)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_sonicvibes_create_gameport(sonic);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_sonic_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "S3 SonicVibes",
+ .id_table = snd_sonic_ids,
+ .probe = snd_sonic_probe,
+ .remove = __devexit_p(snd_sonic_remove),
+};
+
+static int __init alsa_card_sonicvibes_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_sonicvibes_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_sonicvibes_init)
+module_exit(alsa_card_sonicvibes_exit)
diff --git a/sound/pci/trident/Makefile b/sound/pci/trident/Makefile
new file mode 100644
index 0000000..88676b5
--- /dev/null
+++ b/sound/pci/trident/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-trident-objs := trident.o trident_main.o trident_memory.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_TRIDENT) += snd-trident.o
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
new file mode 100644
index 0000000..d94b16f
--- /dev/null
+++ b/sound/pci/trident/trident.c
@@ -0,0 +1,196 @@
+/*
+ * Driver for Trident 4DWave DX/NX & SiS SI7018 Audio PCI soundcard
+ *
+ * Driver was originated by Trident <audio@tridentmicro.com>
+ * Fri Feb 19 15:55:28 MST 1999
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/time.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/trident.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, <audio@tridentmicro.com>");
+MODULE_DESCRIPTION("Trident 4D-WaveDX/NX & SiS SI7018");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Trident,4DWave DX},"
+ "{Trident,4DWave NX},"
+ "{SiS,SI7018 PCI Audio},"
+ "{Best Union,Miss Melody 4DWave PCI},"
+ "{HIS,4DWave PCI},"
+ "{Warpspeed,ONSpeed 4DWave PCI},"
+ "{Aztech Systems,PCI 64-Q3D},"
+ "{Addonics,SV 750},"
+ "{CHIC,True Sound 4Dwave},"
+ "{Shark,Predator4D-PCI},"
+ "{Jaton,SonicWave 4D},"
+ "{Hoontech,SoundTrack Digital 4DWave NX}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 32};
+static int wavetable_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8192};
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Trident 4DWave PCI soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Trident 4DWave PCI soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Trident 4DWave PCI soundcard.");
+module_param_array(pcm_channels, int, NULL, 0444);
+MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM.");
+module_param_array(wavetable_size, int, NULL, 0444);
+MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth.");
+
+static struct pci_device_id snd_trident_ids[] = {
+ {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX),
+ PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0},
+ {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX),
+ 0, 0, 0},
+ {PCI_DEVICE(PCI_VENDOR_ID_SI, PCI_DEVICE_ID_SI_7018), 0, 0, 0},
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_trident_ids);
+
+static int __devinit snd_trident_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_trident *trident;
+ const char *str;
+ int err, pcm_dev = 0;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_trident_create(card, pci,
+ pcm_channels[dev],
+ ((pci->vendor << 16) | pci->device) == TRIDENT_DEVICE_ID_SI7018 ? 1 : 2,
+ wavetable_size[dev],
+ &trident)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = trident;
+
+ switch (trident->device) {
+ case TRIDENT_DEVICE_ID_DX:
+ str = "TRID4DWAVEDX";
+ break;
+ case TRIDENT_DEVICE_ID_NX:
+ str = "TRID4DWAVENX";
+ break;
+ case TRIDENT_DEVICE_ID_SI7018:
+ str = "SI7018";
+ break;
+ default:
+ str = "Unknown";
+ }
+ strcpy(card->driver, str);
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ strcpy(card->shortname, "SiS ");
+ } else {
+ strcpy(card->shortname, "Trident ");
+ }
+ strcat(card->shortname, card->driver);
+ sprintf(card->longname, "%s PCI Audio at 0x%lx, irq %d",
+ card->shortname, trident->port, trident->irq);
+
+ if ((err = snd_trident_pcm(trident, pcm_dev++, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ switch (trident->device) {
+ case TRIDENT_DEVICE_ID_DX:
+ case TRIDENT_DEVICE_ID_NX:
+ if ((err = snd_trident_foldback_pcm(trident, pcm_dev++, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ break;
+ }
+ if (trident->device == TRIDENT_DEVICE_ID_NX || trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ if ((err = snd_trident_spdif_pcm(trident, pcm_dev++, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018 &&
+ (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE,
+ trident->midi_port,
+ MPU401_INFO_INTEGRATED,
+ trident->irq, 0, &trident->rmidi)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_trident_create_gameport(trident);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_trident_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Trident4DWaveAudio",
+ .id_table = snd_trident_ids,
+ .probe = snd_trident_probe,
+ .remove = __devexit_p(snd_trident_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_trident_suspend,
+ .resume = snd_trident_resume,
+#endif
+};
+
+static int __init alsa_card_trident_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_trident_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_trident_init)
+module_exit(alsa_card_trident_exit)
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
new file mode 100644
index 0000000..c612b43
--- /dev/null
+++ b/sound/pci/trident/trident_main.c
@@ -0,0 +1,3976 @@
+/*
+ * Maintained by Jaroslav Kysela <perex@perex.cz>
+ * Originated by audio@tridentmicro.com
+ * Fri Feb 19 15:55:28 MST 1999
+ * Routines for control of Trident 4DWave (DX and NX) chip
+ *
+ * BUGS:
+ *
+ * TODO:
+ * ---
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ * SiS7018 S/PDIF support by Thomas Winischhofer <thomas@winischhofer.net>
+ */
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/gameport.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/trident.h>
+#include <sound/asoundef.h>
+
+#include <asm/io.h>
+
+static int snd_trident_pcm_mixer_build(struct snd_trident *trident,
+ struct snd_trident_voice * voice,
+ struct snd_pcm_substream *substream);
+static int snd_trident_pcm_mixer_free(struct snd_trident *trident,
+ struct snd_trident_voice * voice,
+ struct snd_pcm_substream *substream);
+static irqreturn_t snd_trident_interrupt(int irq, void *dev_id);
+static int snd_trident_sis_reset(struct snd_trident *trident);
+
+static void snd_trident_clear_voices(struct snd_trident * trident,
+ unsigned short v_min, unsigned short v_max);
+static int snd_trident_free(struct snd_trident *trident);
+
+/*
+ * common I/O routines
+ */
+
+
+#if 0
+static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice)
+{
+ unsigned int val, tmp;
+
+ printk("Trident voice %i:\n", voice);
+ outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR));
+ val = inl(TRID_REG(trident, CH_LBA));
+ printk("LBA: 0x%x\n", val);
+ val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC));
+ printk("GVSel: %i\n", val >> 31);
+ printk("Pan: 0x%x\n", (val >> 24) & 0x7f);
+ printk("Vol: 0x%x\n", (val >> 16) & 0xff);
+ printk("CTRL: 0x%x\n", (val >> 12) & 0x0f);
+ printk("EC: 0x%x\n", val & 0x0fff);
+ if (trident->device != TRIDENT_DEVICE_ID_NX) {
+ val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS));
+ printk("CSO: 0x%x\n", val >> 16);
+ printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff);
+ printk("FMS: 0x%x\n", val & 0x0f);
+ val = inl(TRID_REG(trident, CH_DX_ESO_DELTA));
+ printk("ESO: 0x%x\n", val >> 16);
+ printk("Delta: 0x%x\n", val & 0xffff);
+ val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL));
+ } else { // TRIDENT_DEVICE_ID_NX
+ val = inl(TRID_REG(trident, CH_NX_DELTA_CSO));
+ tmp = (val >> 24) & 0xff;
+ printk("CSO: 0x%x\n", val & 0x00ffffff);
+ val = inl(TRID_REG(trident, CH_NX_DELTA_ESO));
+ tmp |= (val >> 16) & 0xff00;
+ printk("Delta: 0x%x\n", tmp);
+ printk("ESO: 0x%x\n", val & 0x00ffffff);
+ val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL));
+ printk("Alpha: 0x%x\n", val >> 20);
+ printk("FMS: 0x%x\n", (val >> 16) & 0x0f);
+ }
+ printk("FMC: 0x%x\n", (val >> 14) & 3);
+ printk("RVol: 0x%x\n", (val >> 7) & 0x7f);
+ printk("CVol: 0x%x\n", val & 0x7f);
+}
+#endif
+
+/*---------------------------------------------------------------------------
+ unsigned short snd_trident_codec_read(struct snd_ac97 *ac97, unsigned short reg)
+
+ Description: This routine will do all of the reading from the external
+ CODEC (AC97).
+
+ Parameters: ac97 - ac97 codec structure
+ reg - CODEC register index, from AC97 Hal.
+
+ returns: 16 bit value read from the AC97.
+
+ ---------------------------------------------------------------------------*/
+static unsigned short snd_trident_codec_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ unsigned int data = 0, treg;
+ unsigned short count = 0xffff;
+ unsigned long flags;
+ struct snd_trident *trident = ac97->private_data;
+
+ spin_lock_irqsave(&trident->reg_lock, flags);
+ if (trident->device == TRIDENT_DEVICE_ID_DX) {
+ data = (DX_AC97_BUSY_READ | (reg & 0x000000ff));
+ outl(data, TRID_REG(trident, DX_ACR1_AC97_R));
+ do {
+ data = inl(TRID_REG(trident, DX_ACR1_AC97_R));
+ if ((data & DX_AC97_BUSY_READ) == 0)
+ break;
+ } while (--count);
+ } else if (trident->device == TRIDENT_DEVICE_ID_NX) {
+ data = (NX_AC97_BUSY_READ | (reg & 0x000000ff));
+ treg = ac97->num == 0 ? NX_ACR2_AC97_R_PRIMARY : NX_ACR3_AC97_R_SECONDARY;
+ outl(data, TRID_REG(trident, treg));
+ do {
+ data = inl(TRID_REG(trident, treg));
+ if ((data & 0x00000C00) == 0)
+ break;
+ } while (--count);
+ } else if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ data = SI_AC97_BUSY_READ | SI_AC97_AUDIO_BUSY | (reg & 0x000000ff);
+ if (ac97->num == 1)
+ data |= SI_AC97_SECONDARY;
+ outl(data, TRID_REG(trident, SI_AC97_READ));
+ do {
+ data = inl(TRID_REG(trident, SI_AC97_READ));
+ if ((data & (SI_AC97_BUSY_READ)) == 0)
+ break;
+ } while (--count);
+ }
+
+ if (count == 0 && !trident->ac97_detect) {
+ snd_printk(KERN_ERR "ac97 codec read TIMEOUT [0x%x/0x%x]!!!\n",
+ reg, data);
+ data = 0;
+ }
+
+ spin_unlock_irqrestore(&trident->reg_lock, flags);
+ return ((unsigned short) (data >> 16));
+}
+
+/*---------------------------------------------------------------------------
+ void snd_trident_codec_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short wdata)
+
+ Description: This routine will do all of the writing to the external
+ CODEC (AC97).
+
+ Parameters: ac97 - ac97 codec structure
+ reg - CODEC register index, from AC97 Hal.
+ data - Lower 16 bits are the data to write to CODEC.
+
+ returns: TRUE if everything went ok, else FALSE.
+
+ ---------------------------------------------------------------------------*/
+static void snd_trident_codec_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short wdata)
+{
+ unsigned int address, data;
+ unsigned short count = 0xffff;
+ unsigned long flags;
+ struct snd_trident *trident = ac97->private_data;
+
+ data = ((unsigned long) wdata) << 16;
+
+ spin_lock_irqsave(&trident->reg_lock, flags);
+ if (trident->device == TRIDENT_DEVICE_ID_DX) {
+ address = DX_ACR0_AC97_W;
+
+ /* read AC-97 write register status */
+ do {
+ if ((inw(TRID_REG(trident, address)) & DX_AC97_BUSY_WRITE) == 0)
+ break;
+ } while (--count);
+
+ data |= (DX_AC97_BUSY_WRITE | (reg & 0x000000ff));
+ } else if (trident->device == TRIDENT_DEVICE_ID_NX) {
+ address = NX_ACR1_AC97_W;
+
+ /* read AC-97 write register status */
+ do {
+ if ((inw(TRID_REG(trident, address)) & NX_AC97_BUSY_WRITE) == 0)
+ break;
+ } while (--count);
+
+ data |= (NX_AC97_BUSY_WRITE | (ac97->num << 8) | (reg & 0x000000ff));
+ } else if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ address = SI_AC97_WRITE;
+
+ /* read AC-97 write register status */
+ do {
+ if ((inw(TRID_REG(trident, address)) & (SI_AC97_BUSY_WRITE)) == 0)
+ break;
+ } while (--count);
+
+ data |= SI_AC97_BUSY_WRITE | SI_AC97_AUDIO_BUSY | (reg & 0x000000ff);
+ if (ac97->num == 1)
+ data |= SI_AC97_SECONDARY;
+ } else {
+ address = 0; /* keep GCC happy */
+ count = 0; /* return */
+ }
+
+ if (count == 0) {
+ spin_unlock_irqrestore(&trident->reg_lock, flags);
+ return;
+ }
+ outl(data, TRID_REG(trident, address));
+ spin_unlock_irqrestore(&trident->reg_lock, flags);
+}
+
+/*---------------------------------------------------------------------------
+ void snd_trident_enable_eso(struct snd_trident *trident)
+
+ Description: This routine will enable end of loop interrupts.
+ End of loop interrupts will occur when a running
+ channel reaches ESO.
+ Also enables middle of loop interrupts.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_enable_eso(struct snd_trident * trident)
+{
+ unsigned int val;
+
+ val = inl(TRID_REG(trident, T4D_LFO_GC_CIR));
+ val |= ENDLP_IE;
+ val |= MIDLP_IE;
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018)
+ val |= BANK_B_EN;
+ outl(val, TRID_REG(trident, T4D_LFO_GC_CIR));
+}
+
+/*---------------------------------------------------------------------------
+ void snd_trident_disable_eso(struct snd_trident *trident)
+
+ Description: This routine will disable end of loop interrupts.
+ End of loop interrupts will occur when a running
+ channel reaches ESO.
+ Also disables middle of loop interrupts.
+
+ Parameters:
+ trident - pointer to target device class for 4DWave.
+
+ returns: TRUE if everything went ok, else FALSE.
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_disable_eso(struct snd_trident * trident)
+{
+ unsigned int tmp;
+
+ tmp = inl(TRID_REG(trident, T4D_LFO_GC_CIR));
+ tmp &= ~ENDLP_IE;
+ tmp &= ~MIDLP_IE;
+ outl(tmp, TRID_REG(trident, T4D_LFO_GC_CIR));
+}
+
+/*---------------------------------------------------------------------------
+ void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice)
+
+ Description: Start a voice, any channel 0 thru 63.
+ This routine automatically handles the fact that there are
+ more than 32 channels available.
+
+ Parameters : voice - Voice number 0 thru n.
+ trident - pointer to target device class for 4DWave.
+
+ Return Value: None.
+
+ ---------------------------------------------------------------------------*/
+
+void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice)
+{
+ unsigned int mask = 1 << (voice & 0x1f);
+ unsigned int reg = (voice & 0x20) ? T4D_START_B : T4D_START_A;
+
+ outl(mask, TRID_REG(trident, reg));
+}
+
+EXPORT_SYMBOL(snd_trident_start_voice);
+
+/*---------------------------------------------------------------------------
+ void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice)
+
+ Description: Stop a voice, any channel 0 thru 63.
+ This routine automatically handles the fact that there are
+ more than 32 channels available.
+
+ Parameters : voice - Voice number 0 thru n.
+ trident - pointer to target device class for 4DWave.
+
+ Return Value: None.
+
+ ---------------------------------------------------------------------------*/
+
+void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice)
+{
+ unsigned int mask = 1 << (voice & 0x1f);
+ unsigned int reg = (voice & 0x20) ? T4D_STOP_B : T4D_STOP_A;
+
+ outl(mask, TRID_REG(trident, reg));
+}
+
+EXPORT_SYMBOL(snd_trident_stop_voice);
+
+/*---------------------------------------------------------------------------
+ int snd_trident_allocate_pcm_channel(struct snd_trident *trident)
+
+ Description: Allocate hardware channel in Bank B (32-63).
+
+ Parameters : trident - pointer to target device class for 4DWave.
+
+ Return Value: hardware channel - 32-63 or -1 when no channel is available
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_allocate_pcm_channel(struct snd_trident * trident)
+{
+ int idx;
+
+ if (trident->ChanPCMcnt >= trident->ChanPCM)
+ return -1;
+ for (idx = 31; idx >= 0; idx--) {
+ if (!(trident->ChanMap[T4D_BANK_B] & (1 << idx))) {
+ trident->ChanMap[T4D_BANK_B] |= 1 << idx;
+ trident->ChanPCMcnt++;
+ return idx + 32;
+ }
+ }
+ return -1;
+}
+
+/*---------------------------------------------------------------------------
+ void snd_trident_free_pcm_channel(int channel)
+
+ Description: Free hardware channel in Bank B (32-63)
+
+ Parameters : trident - pointer to target device class for 4DWave.
+ channel - hardware channel number 0-63
+
+ Return Value: none
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_free_pcm_channel(struct snd_trident *trident, int channel)
+{
+ if (channel < 32 || channel > 63)
+ return;
+ channel &= 0x1f;
+ if (trident->ChanMap[T4D_BANK_B] & (1 << channel)) {
+ trident->ChanMap[T4D_BANK_B] &= ~(1 << channel);
+ trident->ChanPCMcnt--;
+ }
+}
+
+/*---------------------------------------------------------------------------
+ unsigned int snd_trident_allocate_synth_channel(void)
+
+ Description: Allocate hardware channel in Bank A (0-31).
+
+ Parameters : trident - pointer to target device class for 4DWave.
+
+ Return Value: hardware channel - 0-31 or -1 when no channel is available
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_allocate_synth_channel(struct snd_trident * trident)
+{
+ int idx;
+
+ for (idx = 31; idx >= 0; idx--) {
+ if (!(trident->ChanMap[T4D_BANK_A] & (1 << idx))) {
+ trident->ChanMap[T4D_BANK_A] |= 1 << idx;
+ trident->synth.ChanSynthCount++;
+ return idx;
+ }
+ }
+ return -1;
+}
+
+/*---------------------------------------------------------------------------
+ void snd_trident_free_synth_channel( int channel )
+
+ Description: Free hardware channel in Bank B (0-31).
+
+ Parameters : trident - pointer to target device class for 4DWave.
+ channel - hardware channel number 0-63
+
+ Return Value: none
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_free_synth_channel(struct snd_trident *trident, int channel)
+{
+ if (channel < 0 || channel > 31)
+ return;
+ channel &= 0x1f;
+ if (trident->ChanMap[T4D_BANK_A] & (1 << channel)) {
+ trident->ChanMap[T4D_BANK_A] &= ~(1 << channel);
+ trident->synth.ChanSynthCount--;
+ }
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_write_voice_regs
+
+ Description: This routine will complete and write the 5 hardware channel
+ registers to hardware.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ voice - synthesizer voice structure
+ Each register field.
+
+ ---------------------------------------------------------------------------*/
+
+void snd_trident_write_voice_regs(struct snd_trident * trident,
+ struct snd_trident_voice * voice)
+{
+ unsigned int FmcRvolCvol;
+ unsigned int regs[5];
+
+ regs[1] = voice->LBA;
+ regs[4] = (voice->GVSel << 31) |
+ ((voice->Pan & 0x0000007f) << 24) |
+ ((voice->CTRL & 0x0000000f) << 12);
+ FmcRvolCvol = ((voice->FMC & 3) << 14) |
+ ((voice->RVol & 0x7f) << 7) |
+ (voice->CVol & 0x7f);
+
+ switch (trident->device) {
+ case TRIDENT_DEVICE_ID_SI7018:
+ regs[4] |= voice->number > 31 ?
+ (voice->Vol & 0x000003ff) :
+ ((voice->Vol & 0x00003fc) << (16-2)) |
+ (voice->EC & 0x00000fff);
+ regs[0] = (voice->CSO << 16) | ((voice->Alpha & 0x00000fff) << 4) |
+ (voice->FMS & 0x0000000f);
+ regs[2] = (voice->ESO << 16) | (voice->Delta & 0x0ffff);
+ regs[3] = (voice->Attribute << 16) | FmcRvolCvol;
+ break;
+ case TRIDENT_DEVICE_ID_DX:
+ regs[4] |= ((voice->Vol & 0x000003fc) << (16-2)) |
+ (voice->EC & 0x00000fff);
+ regs[0] = (voice->CSO << 16) | ((voice->Alpha & 0x00000fff) << 4) |
+ (voice->FMS & 0x0000000f);
+ regs[2] = (voice->ESO << 16) | (voice->Delta & 0x0ffff);
+ regs[3] = FmcRvolCvol;
+ break;
+ case TRIDENT_DEVICE_ID_NX:
+ regs[4] |= ((voice->Vol & 0x000003fc) << (16-2)) |
+ (voice->EC & 0x00000fff);
+ regs[0] = (voice->Delta << 24) | (voice->CSO & 0x00ffffff);
+ regs[2] = ((voice->Delta << 16) & 0xff000000) |
+ (voice->ESO & 0x00ffffff);
+ regs[3] = (voice->Alpha << 20) |
+ ((voice->FMS & 0x0000000f) << 16) | FmcRvolCvol;
+ break;
+ default:
+ snd_BUG();
+ return;
+ }
+
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+ outl(regs[0], TRID_REG(trident, CH_START + 0));
+ outl(regs[1], TRID_REG(trident, CH_START + 4));
+ outl(regs[2], TRID_REG(trident, CH_START + 8));
+ outl(regs[3], TRID_REG(trident, CH_START + 12));
+ outl(regs[4], TRID_REG(trident, CH_START + 16));
+
+#if 0
+ printk("written %i channel:\n", voice->number);
+ printk(" regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0)));
+ printk(" regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4)));
+ printk(" regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8)));
+ printk(" regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12)));
+ printk(" regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16)));
+#endif
+}
+
+EXPORT_SYMBOL(snd_trident_write_voice_regs);
+
+/*---------------------------------------------------------------------------
+ snd_trident_write_cso_reg
+
+ Description: This routine will write the new CSO offset
+ register to hardware.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ voice - synthesizer voice structure
+ CSO - new CSO value
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_write_cso_reg(struct snd_trident * trident,
+ struct snd_trident_voice * voice,
+ unsigned int CSO)
+{
+ voice->CSO = CSO;
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+ if (trident->device != TRIDENT_DEVICE_ID_NX) {
+ outw(voice->CSO, TRID_REG(trident, CH_DX_CSO_ALPHA_FMS) + 2);
+ } else {
+ outl((voice->Delta << 24) |
+ (voice->CSO & 0x00ffffff), TRID_REG(trident, CH_NX_DELTA_CSO));
+ }
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_write_eso_reg
+
+ Description: This routine will write the new ESO offset
+ register to hardware.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ voice - synthesizer voice structure
+ ESO - new ESO value
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_write_eso_reg(struct snd_trident * trident,
+ struct snd_trident_voice * voice,
+ unsigned int ESO)
+{
+ voice->ESO = ESO;
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+ if (trident->device != TRIDENT_DEVICE_ID_NX) {
+ outw(voice->ESO, TRID_REG(trident, CH_DX_ESO_DELTA) + 2);
+ } else {
+ outl(((voice->Delta << 16) & 0xff000000) | (voice->ESO & 0x00ffffff),
+ TRID_REG(trident, CH_NX_DELTA_ESO));
+ }
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_write_vol_reg
+
+ Description: This routine will write the new voice volume
+ register to hardware.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ voice - synthesizer voice structure
+ Vol - new voice volume
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_write_vol_reg(struct snd_trident * trident,
+ struct snd_trident_voice * voice,
+ unsigned int Vol)
+{
+ voice->Vol = Vol;
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+ switch (trident->device) {
+ case TRIDENT_DEVICE_ID_DX:
+ case TRIDENT_DEVICE_ID_NX:
+ outb(voice->Vol >> 2, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 2));
+ break;
+ case TRIDENT_DEVICE_ID_SI7018:
+ // printk("voice->Vol = 0x%x\n", voice->Vol);
+ outw((voice->CTRL << 12) | voice->Vol,
+ TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC));
+ break;
+ }
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_write_pan_reg
+
+ Description: This routine will write the new voice pan
+ register to hardware.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ voice - synthesizer voice structure
+ Pan - new pan value
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_write_pan_reg(struct snd_trident * trident,
+ struct snd_trident_voice * voice,
+ unsigned int Pan)
+{
+ voice->Pan = Pan;
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+ outb(((voice->GVSel & 0x01) << 7) | (voice->Pan & 0x7f),
+ TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 3));
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_write_rvol_reg
+
+ Description: This routine will write the new reverb volume
+ register to hardware.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ voice - synthesizer voice structure
+ RVol - new reverb volume
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_write_rvol_reg(struct snd_trident * trident,
+ struct snd_trident_voice * voice,
+ unsigned int RVol)
+{
+ voice->RVol = RVol;
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+ outw(((voice->FMC & 0x0003) << 14) | ((voice->RVol & 0x007f) << 7) |
+ (voice->CVol & 0x007f),
+ TRID_REG(trident, trident->device == TRIDENT_DEVICE_ID_NX ?
+ CH_NX_ALPHA_FMS_FMC_RVOL_CVOL : CH_DX_FMC_RVOL_CVOL));
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_write_cvol_reg
+
+ Description: This routine will write the new chorus volume
+ register to hardware.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ voice - synthesizer voice structure
+ CVol - new chorus volume
+
+ ---------------------------------------------------------------------------*/
+
+static void snd_trident_write_cvol_reg(struct snd_trident * trident,
+ struct snd_trident_voice * voice,
+ unsigned int CVol)
+{
+ voice->CVol = CVol;
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+ outw(((voice->FMC & 0x0003) << 14) | ((voice->RVol & 0x007f) << 7) |
+ (voice->CVol & 0x007f),
+ TRID_REG(trident, trident->device == TRIDENT_DEVICE_ID_NX ?
+ CH_NX_ALPHA_FMS_FMC_RVOL_CVOL : CH_DX_FMC_RVOL_CVOL));
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_convert_rate
+
+ Description: This routine converts rate in HZ to hardware delta value.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ rate - Real or Virtual channel number.
+
+ Returns: Delta value.
+
+ ---------------------------------------------------------------------------*/
+static unsigned int snd_trident_convert_rate(unsigned int rate)
+{
+ unsigned int delta;
+
+ // We special case 44100 and 8000 since rounding with the equation
+ // does not give us an accurate enough value. For 11025 and 22050
+ // the equation gives us the best answer. All other frequencies will
+ // also use the equation. JDW
+ if (rate == 44100)
+ delta = 0xeb3;
+ else if (rate == 8000)
+ delta = 0x2ab;
+ else if (rate == 48000)
+ delta = 0x1000;
+ else
+ delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff;
+ return delta;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_convert_adc_rate
+
+ Description: This routine converts rate in HZ to hardware delta value.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ rate - Real or Virtual channel number.
+
+ Returns: Delta value.
+
+ ---------------------------------------------------------------------------*/
+static unsigned int snd_trident_convert_adc_rate(unsigned int rate)
+{
+ unsigned int delta;
+
+ // We special case 44100 and 8000 since rounding with the equation
+ // does not give us an accurate enough value. For 11025 and 22050
+ // the equation gives us the best answer. All other frequencies will
+ // also use the equation. JDW
+ if (rate == 44100)
+ delta = 0x116a;
+ else if (rate == 8000)
+ delta = 0x6000;
+ else if (rate == 48000)
+ delta = 0x1000;
+ else
+ delta = ((48000 << 12) / rate) & 0x0000ffff;
+ return delta;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_spurious_threshold
+
+ Description: This routine converts rate in HZ to spurious threshold.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ rate - Real or Virtual channel number.
+
+ Returns: Delta value.
+
+ ---------------------------------------------------------------------------*/
+static unsigned int snd_trident_spurious_threshold(unsigned int rate,
+ unsigned int period_size)
+{
+ unsigned int res = (rate * period_size) / 48000;
+ if (res < 64)
+ res = res / 2;
+ else
+ res -= 32;
+ return res;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_control_mode
+
+ Description: This routine returns a control mode for a PCM channel.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+ substream - PCM substream
+
+ Returns: Control value.
+
+ ---------------------------------------------------------------------------*/
+static unsigned int snd_trident_control_mode(struct snd_pcm_substream *substream)
+{
+ unsigned int CTRL;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ /* set ctrl mode
+ CTRL default: 8-bit (unsigned) mono, loop mode enabled
+ */
+ CTRL = 0x00000001;
+ if (snd_pcm_format_width(runtime->format) == 16)
+ CTRL |= 0x00000008; // 16-bit data
+ if (snd_pcm_format_signed(runtime->format))
+ CTRL |= 0x00000002; // signed data
+ if (runtime->channels > 1)
+ CTRL |= 0x00000004; // stereo data
+ return CTRL;
+}
+
+/*
+ * PCM part
+ */
+
+/*---------------------------------------------------------------------------
+ snd_trident_ioctl
+
+ Description: Device I/O control handler for playback/capture parameters.
+
+ Parameters: substream - PCM substream class
+ cmd - what ioctl message to process
+ arg - additional message infoarg
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd,
+ void *arg)
+{
+ /* FIXME: it seems that with small periods the behaviour of
+ trident hardware is unpredictable and interrupt generator
+ is broken */
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_allocate_pcm_mem
+
+ Description: Allocate PCM ring buffer for given substream
+
+ Parameters: substream - PCM substream class
+ hw_params - hardware parameters
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_allocate_pcm_mem(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ if (trident->tlb.entries) {
+ if (err > 0) { /* change */
+ if (voice->memblk)
+ snd_trident_free_pages(trident, voice->memblk);
+ voice->memblk = snd_trident_alloc_pages(trident, substream);
+ if (voice->memblk == NULL)
+ return -ENOMEM;
+ }
+ }
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_allocate_evoice
+
+ Description: Allocate extra voice as interrupt generator
+
+ Parameters: substream - PCM substream class
+ hw_params - hardware parameters
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_allocate_evoice(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident_voice *evoice = voice->extra;
+
+ /* voice management */
+
+ if (params_buffer_size(hw_params) / 2 != params_period_size(hw_params)) {
+ if (evoice == NULL) {
+ evoice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_PCM, 0, 0);
+ if (evoice == NULL)
+ return -ENOMEM;
+ voice->extra = evoice;
+ evoice->substream = substream;
+ }
+ } else {
+ if (evoice != NULL) {
+ snd_trident_free_voice(trident, evoice);
+ voice->extra = evoice = NULL;
+ }
+ }
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_hw_params
+
+ Description: Set the hardware parameters for the playback device.
+
+ Parameters: substream - PCM substream class
+ hw_params - hardware parameters
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ int err;
+
+ err = snd_trident_allocate_pcm_mem(substream, hw_params);
+ if (err >= 0)
+ err = snd_trident_allocate_evoice(substream, hw_params);
+ return err;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_playback_hw_free
+
+ Description: Release the hardware resources for the playback device.
+
+ Parameters: substream - PCM substream class
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident_voice *evoice = voice ? voice->extra : NULL;
+
+ if (trident->tlb.entries) {
+ if (voice && voice->memblk) {
+ snd_trident_free_pages(trident, voice->memblk);
+ voice->memblk = NULL;
+ }
+ }
+ snd_pcm_lib_free_pages(substream);
+ if (evoice != NULL) {
+ snd_trident_free_voice(trident, evoice);
+ voice->extra = NULL;
+ }
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_playback_prepare
+
+ Description: Prepare playback device for playback.
+
+ Parameters: substream - PCM substream class
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident_voice *evoice = voice->extra;
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[substream->number];
+
+ spin_lock_irq(&trident->reg_lock);
+
+ /* set delta (rate) value */
+ voice->Delta = snd_trident_convert_rate(runtime->rate);
+ voice->spurious_threshold = snd_trident_spurious_threshold(runtime->rate, runtime->period_size);
+
+ /* set Loop Begin Address */
+ if (voice->memblk)
+ voice->LBA = voice->memblk->offset;
+ else
+ voice->LBA = runtime->dma_addr;
+
+ voice->CSO = 0;
+ voice->ESO = runtime->buffer_size - 1; /* in samples */
+ voice->CTRL = snd_trident_control_mode(substream);
+ voice->FMC = 3;
+ voice->GVSel = 1;
+ voice->EC = 0;
+ voice->Alpha = 0;
+ voice->FMS = 0;
+ voice->Vol = mix->vol;
+ voice->RVol = mix->rvol;
+ voice->CVol = mix->cvol;
+ voice->Pan = mix->pan;
+ voice->Attribute = 0;
+#if 0
+ voice->Attribute = (1<<(30-16))|(2<<(26-16))|
+ (0<<(24-16))|(0x1f<<(19-16));
+#else
+ voice->Attribute = 0;
+#endif
+
+ snd_trident_write_voice_regs(trident, voice);
+
+ if (evoice != NULL) {
+ evoice->Delta = voice->Delta;
+ evoice->spurious_threshold = voice->spurious_threshold;
+ evoice->LBA = voice->LBA;
+ evoice->CSO = 0;
+ evoice->ESO = (runtime->period_size * 2) + 4 - 1; /* in samples */
+ evoice->CTRL = voice->CTRL;
+ evoice->FMC = 3;
+ evoice->GVSel = trident->device == TRIDENT_DEVICE_ID_SI7018 ? 0 : 1;
+ evoice->EC = 0;
+ evoice->Alpha = 0;
+ evoice->FMS = 0;
+ evoice->Vol = 0x3ff; /* mute */
+ evoice->RVol = evoice->CVol = 0x7f; /* mute */
+ evoice->Pan = 0x7f; /* mute */
+#if 0
+ evoice->Attribute = (1<<(30-16))|(2<<(26-16))|
+ (0<<(24-16))|(0x1f<<(19-16));
+#else
+ evoice->Attribute = 0;
+#endif
+ snd_trident_write_voice_regs(trident, evoice);
+ evoice->isync2 = 1;
+ evoice->isync_mark = runtime->period_size;
+ evoice->ESO = (runtime->period_size * 2) - 1;
+ }
+
+ spin_unlock_irq(&trident->reg_lock);
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_capture_hw_params
+
+ Description: Set the hardware parameters for the capture device.
+
+ Parameters: substream - PCM substream class
+ hw_params - hardware parameters
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_trident_allocate_pcm_mem(substream, hw_params);
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_capture_prepare
+
+ Description: Prepare capture device for playback.
+
+ Parameters: substream - PCM substream class
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ unsigned int val, ESO_bytes;
+
+ spin_lock_irq(&trident->reg_lock);
+
+ // Initilize the channel and set channel Mode
+ outb(0, TRID_REG(trident, LEGACY_DMAR15));
+
+ // Set DMA channel operation mode register
+ outb(0x54, TRID_REG(trident, LEGACY_DMAR11));
+
+ // Set channel buffer Address, DMAR0 expects contiguous PCI memory area
+ voice->LBA = runtime->dma_addr;
+ outl(voice->LBA, TRID_REG(trident, LEGACY_DMAR0));
+ if (voice->memblk)
+ voice->LBA = voice->memblk->offset;
+
+ // set ESO
+ ESO_bytes = snd_pcm_lib_buffer_bytes(substream) - 1;
+ outb((ESO_bytes & 0x00ff0000) >> 16, TRID_REG(trident, LEGACY_DMAR6));
+ outw((ESO_bytes & 0x0000ffff), TRID_REG(trident, LEGACY_DMAR4));
+ ESO_bytes++;
+
+ // Set channel sample rate, 4.12 format
+ val = (((unsigned int) 48000L << 12) + (runtime->rate/2)) / runtime->rate;
+ outw(val, TRID_REG(trident, T4D_SBDELTA_DELTA_R));
+
+ // Set channel interrupt blk length
+ if (snd_pcm_format_width(runtime->format) == 16) {
+ val = (unsigned short) ((ESO_bytes >> 1) - 1);
+ } else {
+ val = (unsigned short) (ESO_bytes - 1);
+ }
+
+ outl((val << 16) | val, TRID_REG(trident, T4D_SBBL_SBCL));
+
+ // Right now, set format and start to run captureing,
+ // continuous run loop enable.
+ trident->bDMAStart = 0x19; // 0001 1001b
+
+ if (snd_pcm_format_width(runtime->format) == 16)
+ trident->bDMAStart |= 0x80;
+ if (snd_pcm_format_signed(runtime->format))
+ trident->bDMAStart |= 0x20;
+ if (runtime->channels > 1)
+ trident->bDMAStart |= 0x40;
+
+ // Prepare capture intr channel
+
+ voice->Delta = snd_trident_convert_rate(runtime->rate);
+ voice->spurious_threshold = snd_trident_spurious_threshold(runtime->rate, runtime->period_size);
+ voice->isync = 1;
+ voice->isync_mark = runtime->period_size;
+ voice->isync_max = runtime->buffer_size;
+
+ // Set voice parameters
+ voice->CSO = 0;
+ voice->ESO = voice->isync_ESO = (runtime->period_size * 2) + 6 - 1;
+ voice->CTRL = snd_trident_control_mode(substream);
+ voice->FMC = 3;
+ voice->RVol = 0x7f;
+ voice->CVol = 0x7f;
+ voice->GVSel = 1;
+ voice->Pan = 0x7f; /* mute */
+ voice->Vol = 0x3ff; /* mute */
+ voice->EC = 0;
+ voice->Alpha = 0;
+ voice->FMS = 0;
+ voice->Attribute = 0;
+
+ snd_trident_write_voice_regs(trident, voice);
+
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_si7018_capture_hw_params
+
+ Description: Set the hardware parameters for the capture device.
+
+ Parameters: substream - PCM substream class
+ hw_params - hardware parameters
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_si7018_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+
+ return snd_trident_allocate_evoice(substream, hw_params);
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_si7018_capture_hw_free
+
+ Description: Release the hardware resources for the capture device.
+
+ Parameters: substream - PCM substream class
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_si7018_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident_voice *evoice = voice ? voice->extra : NULL;
+
+ snd_pcm_lib_free_pages(substream);
+ if (evoice != NULL) {
+ snd_trident_free_voice(trident, evoice);
+ voice->extra = NULL;
+ }
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_si7018_capture_prepare
+
+ Description: Prepare capture device for playback.
+
+ Parameters: substream - PCM substream class
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_si7018_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident_voice *evoice = voice->extra;
+
+ spin_lock_irq(&trident->reg_lock);
+
+ voice->LBA = runtime->dma_addr;
+ voice->Delta = snd_trident_convert_adc_rate(runtime->rate);
+ voice->spurious_threshold = snd_trident_spurious_threshold(runtime->rate, runtime->period_size);
+
+ // Set voice parameters
+ voice->CSO = 0;
+ voice->ESO = runtime->buffer_size - 1; /* in samples */
+ voice->CTRL = snd_trident_control_mode(substream);
+ voice->FMC = 0;
+ voice->RVol = 0;
+ voice->CVol = 0;
+ voice->GVSel = 1;
+ voice->Pan = T4D_DEFAULT_PCM_PAN;
+ voice->Vol = 0;
+ voice->EC = 0;
+ voice->Alpha = 0;
+ voice->FMS = 0;
+
+ voice->Attribute = (2 << (30-16)) |
+ (2 << (26-16)) |
+ (2 << (24-16)) |
+ (1 << (23-16));
+
+ snd_trident_write_voice_regs(trident, voice);
+
+ if (evoice != NULL) {
+ evoice->Delta = snd_trident_convert_rate(runtime->rate);
+ evoice->spurious_threshold = voice->spurious_threshold;
+ evoice->LBA = voice->LBA;
+ evoice->CSO = 0;
+ evoice->ESO = (runtime->period_size * 2) + 20 - 1; /* in samples, 20 means correction */
+ evoice->CTRL = voice->CTRL;
+ evoice->FMC = 3;
+ evoice->GVSel = 0;
+ evoice->EC = 0;
+ evoice->Alpha = 0;
+ evoice->FMS = 0;
+ evoice->Vol = 0x3ff; /* mute */
+ evoice->RVol = evoice->CVol = 0x7f; /* mute */
+ evoice->Pan = 0x7f; /* mute */
+ evoice->Attribute = 0;
+ snd_trident_write_voice_regs(trident, evoice);
+ evoice->isync2 = 1;
+ evoice->isync_mark = runtime->period_size;
+ evoice->ESO = (runtime->period_size * 2) - 1;
+ }
+
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_foldback_prepare
+
+ Description: Prepare foldback capture device for playback.
+
+ Parameters: substream - PCM substream class
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_foldback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident_voice *evoice = voice->extra;
+
+ spin_lock_irq(&trident->reg_lock);
+
+ /* Set channel buffer Address */
+ if (voice->memblk)
+ voice->LBA = voice->memblk->offset;
+ else
+ voice->LBA = runtime->dma_addr;
+
+ /* set target ESO for channel */
+ voice->ESO = runtime->buffer_size - 1; /* in samples */
+
+ /* set sample rate */
+ voice->Delta = 0x1000;
+ voice->spurious_threshold = snd_trident_spurious_threshold(48000, runtime->period_size);
+
+ voice->CSO = 0;
+ voice->CTRL = snd_trident_control_mode(substream);
+ voice->FMC = 3;
+ voice->RVol = 0x7f;
+ voice->CVol = 0x7f;
+ voice->GVSel = 1;
+ voice->Pan = 0x7f; /* mute */
+ voice->Vol = 0x3ff; /* mute */
+ voice->EC = 0;
+ voice->Alpha = 0;
+ voice->FMS = 0;
+ voice->Attribute = 0;
+
+ /* set up capture channel */
+ outb(((voice->number & 0x3f) | 0x80), TRID_REG(trident, T4D_RCI + voice->foldback_chan));
+
+ snd_trident_write_voice_regs(trident, voice);
+
+ if (evoice != NULL) {
+ evoice->Delta = voice->Delta;
+ evoice->spurious_threshold = voice->spurious_threshold;
+ evoice->LBA = voice->LBA;
+ evoice->CSO = 0;
+ evoice->ESO = (runtime->period_size * 2) + 4 - 1; /* in samples */
+ evoice->CTRL = voice->CTRL;
+ evoice->FMC = 3;
+ evoice->GVSel = trident->device == TRIDENT_DEVICE_ID_SI7018 ? 0 : 1;
+ evoice->EC = 0;
+ evoice->Alpha = 0;
+ evoice->FMS = 0;
+ evoice->Vol = 0x3ff; /* mute */
+ evoice->RVol = evoice->CVol = 0x7f; /* mute */
+ evoice->Pan = 0x7f; /* mute */
+ evoice->Attribute = 0;
+ snd_trident_write_voice_regs(trident, evoice);
+ evoice->isync2 = 1;
+ evoice->isync_mark = runtime->period_size;
+ evoice->ESO = (runtime->period_size * 2) - 1;
+ }
+
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_hw_params
+
+ Description: Set the hardware parameters for the spdif device.
+
+ Parameters: substream - PCM substream class
+ hw_params - hardware parameters
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_spdif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ unsigned int old_bits = 0, change = 0;
+ int err;
+
+ err = snd_trident_allocate_pcm_mem(substream, hw_params);
+ if (err < 0)
+ return err;
+
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ err = snd_trident_allocate_evoice(substream, hw_params);
+ if (err < 0)
+ return err;
+ }
+
+ /* prepare SPDIF channel */
+ spin_lock_irq(&trident->reg_lock);
+ old_bits = trident->spdif_pcm_bits;
+ if (old_bits & IEC958_AES0_PROFESSIONAL)
+ trident->spdif_pcm_bits &= ~IEC958_AES0_PRO_FS;
+ else
+ trident->spdif_pcm_bits &= ~(IEC958_AES3_CON_FS << 24);
+ if (params_rate(hw_params) >= 48000) {
+ trident->spdif_pcm_ctrl = 0x3c; // 48000 Hz
+ trident->spdif_pcm_bits |=
+ trident->spdif_bits & IEC958_AES0_PROFESSIONAL ?
+ IEC958_AES0_PRO_FS_48000 :
+ (IEC958_AES3_CON_FS_48000 << 24);
+ }
+ else if (params_rate(hw_params) >= 44100) {
+ trident->spdif_pcm_ctrl = 0x3e; // 44100 Hz
+ trident->spdif_pcm_bits |=
+ trident->spdif_bits & IEC958_AES0_PROFESSIONAL ?
+ IEC958_AES0_PRO_FS_44100 :
+ (IEC958_AES3_CON_FS_44100 << 24);
+ }
+ else {
+ trident->spdif_pcm_ctrl = 0x3d; // 32000 Hz
+ trident->spdif_pcm_bits |=
+ trident->spdif_bits & IEC958_AES0_PROFESSIONAL ?
+ IEC958_AES0_PRO_FS_32000 :
+ (IEC958_AES3_CON_FS_32000 << 24);
+ }
+ change = old_bits != trident->spdif_pcm_bits;
+ spin_unlock_irq(&trident->reg_lock);
+
+ if (change)
+ snd_ctl_notify(trident->card, SNDRV_CTL_EVENT_MASK_VALUE, &trident->spdif_pcm_ctl->id);
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_prepare
+
+ Description: Prepare SPDIF device for playback.
+
+ Parameters: substream - PCM substream class
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_spdif_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident_voice *evoice = voice->extra;
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[substream->number];
+ unsigned int RESO, LBAO;
+ unsigned int temp;
+
+ spin_lock_irq(&trident->reg_lock);
+
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+
+ /* set delta (rate) value */
+ voice->Delta = snd_trident_convert_rate(runtime->rate);
+ voice->spurious_threshold = snd_trident_spurious_threshold(runtime->rate, runtime->period_size);
+
+ /* set Loop Back Address */
+ LBAO = runtime->dma_addr;
+ if (voice->memblk)
+ voice->LBA = voice->memblk->offset;
+ else
+ voice->LBA = LBAO;
+
+ voice->isync = 1;
+ voice->isync3 = 1;
+ voice->isync_mark = runtime->period_size;
+ voice->isync_max = runtime->buffer_size;
+
+ /* set target ESO for channel */
+ RESO = runtime->buffer_size - 1;
+ voice->ESO = voice->isync_ESO = (runtime->period_size * 2) + 6 - 1;
+
+ /* set ctrl mode */
+ voice->CTRL = snd_trident_control_mode(substream);
+
+ voice->FMC = 3;
+ voice->RVol = 0x7f;
+ voice->CVol = 0x7f;
+ voice->GVSel = 1;
+ voice->Pan = 0x7f;
+ voice->Vol = 0x3ff;
+ voice->EC = 0;
+ voice->CSO = 0;
+ voice->Alpha = 0;
+ voice->FMS = 0;
+ voice->Attribute = 0;
+
+ /* prepare surrogate IRQ channel */
+ snd_trident_write_voice_regs(trident, voice);
+
+ outw((RESO & 0xffff), TRID_REG(trident, NX_SPESO));
+ outb((RESO >> 16), TRID_REG(trident, NX_SPESO + 2));
+ outl((LBAO & 0xfffffffc), TRID_REG(trident, NX_SPLBA));
+ outw((voice->CSO & 0xffff), TRID_REG(trident, NX_SPCTRL_SPCSO));
+ outb((voice->CSO >> 16), TRID_REG(trident, NX_SPCTRL_SPCSO + 2));
+
+ /* set SPDIF setting */
+ outb(trident->spdif_pcm_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
+ outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS));
+
+ } else { /* SiS */
+
+ /* set delta (rate) value */
+ voice->Delta = 0x800;
+ voice->spurious_threshold = snd_trident_spurious_threshold(48000, runtime->period_size);
+
+ /* set Loop Begin Address */
+ if (voice->memblk)
+ voice->LBA = voice->memblk->offset;
+ else
+ voice->LBA = runtime->dma_addr;
+
+ voice->CSO = 0;
+ voice->ESO = runtime->buffer_size - 1; /* in samples */
+ voice->CTRL = snd_trident_control_mode(substream);
+ voice->FMC = 3;
+ voice->GVSel = 1;
+ voice->EC = 0;
+ voice->Alpha = 0;
+ voice->FMS = 0;
+ voice->Vol = mix->vol;
+ voice->RVol = mix->rvol;
+ voice->CVol = mix->cvol;
+ voice->Pan = mix->pan;
+ voice->Attribute = (1<<(30-16))|(7<<(26-16))|
+ (0<<(24-16))|(0<<(19-16));
+
+ snd_trident_write_voice_regs(trident, voice);
+
+ if (evoice != NULL) {
+ evoice->Delta = voice->Delta;
+ evoice->spurious_threshold = voice->spurious_threshold;
+ evoice->LBA = voice->LBA;
+ evoice->CSO = 0;
+ evoice->ESO = (runtime->period_size * 2) + 4 - 1; /* in samples */
+ evoice->CTRL = voice->CTRL;
+ evoice->FMC = 3;
+ evoice->GVSel = trident->device == TRIDENT_DEVICE_ID_SI7018 ? 0 : 1;
+ evoice->EC = 0;
+ evoice->Alpha = 0;
+ evoice->FMS = 0;
+ evoice->Vol = 0x3ff; /* mute */
+ evoice->RVol = evoice->CVol = 0x7f; /* mute */
+ evoice->Pan = 0x7f; /* mute */
+ evoice->Attribute = 0;
+ snd_trident_write_voice_regs(trident, evoice);
+ evoice->isync2 = 1;
+ evoice->isync_mark = runtime->period_size;
+ evoice->ESO = (runtime->period_size * 2) - 1;
+ }
+
+ outl(trident->spdif_pcm_bits, TRID_REG(trident, SI_SPDIF_CS));
+ temp = inl(TRID_REG(trident, T4D_LFO_GC_CIR));
+ temp &= ~(1<<19);
+ outl(temp, TRID_REG(trident, T4D_LFO_GC_CIR));
+ temp = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ temp |= SPDIF_EN;
+ outl(temp, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ }
+
+ spin_unlock_irq(&trident->reg_lock);
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_trigger
+
+ Description: Start/stop devices
+
+ Parameters: substream - PCM substream class
+ cmd - trigger command (STOP, GO)
+
+ Returns: Error status
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_substream *s;
+ unsigned int what, whati, capture_flag, spdif_flag;
+ struct snd_trident_voice *voice, *evoice;
+ unsigned int val, go;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ go = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ go = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+ what = whati = capture_flag = spdif_flag = 0;
+ spin_lock(&trident->reg_lock);
+ val = inl(TRID_REG(trident, T4D_STIMER)) & 0x00ffffff;
+ snd_pcm_group_for_each_entry(s, substream) {
+ if ((struct snd_trident *) snd_pcm_substream_chip(s) == trident) {
+ voice = s->runtime->private_data;
+ evoice = voice->extra;
+ what |= 1 << (voice->number & 0x1f);
+ if (evoice == NULL) {
+ whati |= 1 << (voice->number & 0x1f);
+ } else {
+ what |= 1 << (evoice->number & 0x1f);
+ whati |= 1 << (evoice->number & 0x1f);
+ if (go)
+ evoice->stimer = val;
+ }
+ if (go) {
+ voice->running = 1;
+ voice->stimer = val;
+ } else {
+ voice->running = 0;
+ }
+ snd_pcm_trigger_done(s, substream);
+ if (voice->capture)
+ capture_flag = 1;
+ if (voice->spdif)
+ spdif_flag = 1;
+ }
+ }
+ if (spdif_flag) {
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+ outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS));
+ val = trident->spdif_pcm_ctrl;
+ if (!go)
+ val &= ~(0x28);
+ outb(val, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
+ } else {
+ outl(trident->spdif_pcm_bits, TRID_REG(trident, SI_SPDIF_CS));
+ val = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) | SPDIF_EN;
+ outl(val, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ }
+ }
+ if (!go)
+ outl(what, TRID_REG(trident, T4D_STOP_B));
+ val = inl(TRID_REG(trident, T4D_AINTEN_B));
+ if (go) {
+ val |= whati;
+ } else {
+ val &= ~whati;
+ }
+ outl(val, TRID_REG(trident, T4D_AINTEN_B));
+ if (go) {
+ outl(what, TRID_REG(trident, T4D_START_B));
+
+ if (capture_flag && trident->device != TRIDENT_DEVICE_ID_SI7018)
+ outb(trident->bDMAStart, TRID_REG(trident, T4D_SBCTRL_SBE2R_SBDD));
+ } else {
+ if (capture_flag && trident->device != TRIDENT_DEVICE_ID_SI7018)
+ outb(0x00, TRID_REG(trident, T4D_SBCTRL_SBE2R_SBDD));
+ }
+ spin_unlock(&trident->reg_lock);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_playback_pointer
+
+ Description: This routine return the playback position
+
+ Parameters: substream - PCM substream class
+
+ Returns: position of buffer
+
+ ---------------------------------------------------------------------------*/
+
+static snd_pcm_uframes_t snd_trident_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ unsigned int cso;
+
+ if (!voice->running)
+ return 0;
+
+ spin_lock(&trident->reg_lock);
+
+ outb(voice->number, TRID_REG(trident, T4D_LFO_GC_CIR));
+
+ if (trident->device != TRIDENT_DEVICE_ID_NX) {
+ cso = inw(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS + 2));
+ } else { // ID_4DWAVE_NX
+ cso = (unsigned int) inl(TRID_REG(trident, CH_NX_DELTA_CSO)) & 0x00ffffff;
+ }
+
+ spin_unlock(&trident->reg_lock);
+
+ if (cso >= runtime->buffer_size)
+ cso = 0;
+
+ return cso;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_capture_pointer
+
+ Description: This routine return the capture position
+
+ Parameters: pcm1 - PCM device class
+
+ Returns: position of buffer
+
+ ---------------------------------------------------------------------------*/
+
+static snd_pcm_uframes_t snd_trident_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ unsigned int result;
+
+ if (!voice->running)
+ return 0;
+
+ result = inw(TRID_REG(trident, T4D_SBBL_SBCL));
+ if (runtime->channels > 1)
+ result >>= 1;
+ if (result > 0)
+ result = runtime->buffer_size - result;
+
+ return result;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_pointer
+
+ Description: This routine return the SPDIF playback position
+
+ Parameters: substream - PCM substream class
+
+ Returns: position of buffer
+
+ ---------------------------------------------------------------------------*/
+
+static snd_pcm_uframes_t snd_trident_spdif_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+ unsigned int result;
+
+ if (!voice->running)
+ return 0;
+
+ result = inl(TRID_REG(trident, NX_SPCTRL_SPCSO)) & 0x00ffffff;
+
+ return result;
+}
+
+/*
+ * Playback support device description
+ */
+
+static struct snd_pcm_hardware snd_trident_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (256*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (256*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+ * Capture support device description
+ */
+
+static struct snd_pcm_hardware snd_trident_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */),
+ .formats = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 4000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+ * Foldback capture support device description
+ */
+
+static struct snd_pcm_hardware snd_trident_foldback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/*
+ * SPDIF playback support device description
+ */
+
+static struct snd_pcm_hardware snd_trident_spdif =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 32000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_trident_spdif_7018 =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_PAUSE /* | SNDRV_PCM_INFO_RESUME */),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static void snd_trident_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ struct snd_trident_voice *voice = runtime->private_data;
+ struct snd_trident *trident;
+
+ if (voice) {
+ trident = voice->trident;
+ snd_trident_free_voice(trident, voice);
+ }
+}
+
+static int snd_trident_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice;
+
+ voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_PCM, 0, 0);
+ if (voice == NULL)
+ return -EAGAIN;
+ snd_trident_pcm_mixer_build(trident, voice, substream);
+ voice->substream = substream;
+ runtime->private_data = voice;
+ runtime->private_free = snd_trident_pcm_free_substream;
+ runtime->hw = snd_trident_playback;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 64*1024);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_playback_close
+
+ Description: This routine will close the 4DWave playback device. For now
+ we will simply free the dma transfer buffer.
+
+ Parameters: substream - PCM substream class
+
+ ---------------------------------------------------------------------------*/
+static int snd_trident_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_trident_voice *voice = runtime->private_data;
+
+ snd_trident_pcm_mixer_free(trident, voice, substream);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_open
+
+ Description: This routine will open the 4DWave SPDIF device.
+
+ Parameters: substream - PCM substream class
+
+ Returns: status - success or failure flag
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_trident_voice *voice;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_PCM, 0, 0);
+ if (voice == NULL)
+ return -EAGAIN;
+ voice->spdif = 1;
+ voice->substream = substream;
+ spin_lock_irq(&trident->reg_lock);
+ trident->spdif_pcm_bits = trident->spdif_bits;
+ spin_unlock_irq(&trident->reg_lock);
+
+ runtime->private_data = voice;
+ runtime->private_free = snd_trident_pcm_free_substream;
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ runtime->hw = snd_trident_spdif;
+ } else {
+ runtime->hw = snd_trident_spdif_7018;
+ }
+
+ trident->spdif_pcm_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(trident->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &trident->spdif_pcm_ctl->id);
+
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 64*1024);
+ return 0;
+}
+
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_close
+
+ Description: This routine will close the 4DWave SPDIF device.
+
+ Parameters: substream - PCM substream class
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ unsigned int temp;
+
+ spin_lock_irq(&trident->reg_lock);
+ // restore default SPDIF setting
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+ outb(trident->spdif_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
+ outl(trident->spdif_bits, TRID_REG(trident, NX_SPCSTATUS));
+ } else {
+ outl(trident->spdif_bits, TRID_REG(trident, SI_SPDIF_CS));
+ temp = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ if (trident->spdif_ctrl) {
+ temp |= SPDIF_EN;
+ } else {
+ temp &= ~SPDIF_EN;
+ }
+ outl(temp, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ }
+ spin_unlock_irq(&trident->reg_lock);
+ trident->spdif_pcm_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(trident->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &trident->spdif_pcm_ctl->id);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_capture_open
+
+ Description: This routine will open the 4DWave capture device.
+
+ Parameters: substream - PCM substream class
+
+ Returns: status - success or failure flag
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_trident_voice *voice;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_PCM, 0, 0);
+ if (voice == NULL)
+ return -EAGAIN;
+ voice->capture = 1;
+ voice->substream = substream;
+ runtime->private_data = voice;
+ runtime->private_free = snd_trident_pcm_free_substream;
+ runtime->hw = snd_trident_capture;
+ snd_pcm_set_sync(substream);
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 64*1024);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_capture_close
+
+ Description: This routine will close the 4DWave capture device. For now
+ we will simply free the dma transfer buffer.
+
+ Parameters: substream - PCM substream class
+
+ ---------------------------------------------------------------------------*/
+static int snd_trident_capture_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_foldback_open
+
+ Description: This routine will open the 4DWave foldback capture device.
+
+ Parameters: substream - PCM substream class
+
+ Returns: status - success or failure flag
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_foldback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_trident_voice *voice;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_PCM, 0, 0);
+ if (voice == NULL)
+ return -EAGAIN;
+ voice->foldback_chan = substream->number;
+ voice->substream = substream;
+ runtime->private_data = voice;
+ runtime->private_free = snd_trident_pcm_free_substream;
+ runtime->hw = snd_trident_foldback;
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 64*1024);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_foldback_close
+
+ Description: This routine will close the 4DWave foldback capture device.
+ For now we will simply free the dma transfer buffer.
+
+ Parameters: substream - PCM substream class
+
+ ---------------------------------------------------------------------------*/
+static int snd_trident_foldback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_trident *trident = snd_pcm_substream_chip(substream);
+ struct snd_trident_voice *voice;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ voice = runtime->private_data;
+
+ /* stop capture channel */
+ spin_lock_irq(&trident->reg_lock);
+ outb(0x00, TRID_REG(trident, T4D_RCI + voice->foldback_chan));
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ PCM operations
+ ---------------------------------------------------------------------------*/
+
+static struct snd_pcm_ops snd_trident_playback_ops = {
+ .open = snd_trident_playback_open,
+ .close = snd_trident_playback_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_hw_params,
+ .hw_free = snd_trident_hw_free,
+ .prepare = snd_trident_playback_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_trident_nx_playback_ops = {
+ .open = snd_trident_playback_open,
+ .close = snd_trident_playback_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_hw_params,
+ .hw_free = snd_trident_hw_free,
+ .prepare = snd_trident_playback_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_playback_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static struct snd_pcm_ops snd_trident_capture_ops = {
+ .open = snd_trident_capture_open,
+ .close = snd_trident_capture_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_capture_hw_params,
+ .hw_free = snd_trident_hw_free,
+ .prepare = snd_trident_capture_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_capture_pointer,
+};
+
+static struct snd_pcm_ops snd_trident_si7018_capture_ops = {
+ .open = snd_trident_capture_open,
+ .close = snd_trident_capture_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_si7018_capture_hw_params,
+ .hw_free = snd_trident_si7018_capture_hw_free,
+ .prepare = snd_trident_si7018_capture_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_trident_foldback_ops = {
+ .open = snd_trident_foldback_open,
+ .close = snd_trident_foldback_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_hw_params,
+ .hw_free = snd_trident_hw_free,
+ .prepare = snd_trident_foldback_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_trident_nx_foldback_ops = {
+ .open = snd_trident_foldback_open,
+ .close = snd_trident_foldback_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_hw_params,
+ .hw_free = snd_trident_hw_free,
+ .prepare = snd_trident_foldback_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_playback_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static struct snd_pcm_ops snd_trident_spdif_ops = {
+ .open = snd_trident_spdif_open,
+ .close = snd_trident_spdif_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_spdif_hw_params,
+ .hw_free = snd_trident_hw_free,
+ .prepare = snd_trident_spdif_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_spdif_pointer,
+};
+
+static struct snd_pcm_ops snd_trident_spdif_7018_ops = {
+ .open = snd_trident_spdif_open,
+ .close = snd_trident_spdif_close,
+ .ioctl = snd_trident_ioctl,
+ .hw_params = snd_trident_spdif_hw_params,
+ .hw_free = snd_trident_hw_free,
+ .prepare = snd_trident_spdif_prepare,
+ .trigger = snd_trident_trigger,
+ .pointer = snd_trident_playback_pointer,
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_pcm
+
+ Description: This routine registers the 4DWave device for PCM support.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+
+ Returns: None
+
+ ---------------------------------------------------------------------------*/
+
+int __devinit snd_trident_pcm(struct snd_trident * trident,
+ int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(trident->card, "trident_dx_nx", device, trident->ChanPCM, 1, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = trident;
+
+ if (trident->tlb.entries) {
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_trident_nx_playback_ops);
+ } else {
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_trident_playback_ops);
+ }
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ trident->device != TRIDENT_DEVICE_ID_SI7018 ?
+ &snd_trident_capture_ops :
+ &snd_trident_si7018_capture_ops);
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strcpy(pcm->name, "Trident 4DWave");
+ trident->pcm = pcm;
+
+ if (trident->tlb.entries) {
+ struct snd_pcm_substream *substream;
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next)
+ snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(trident->pci),
+ 64*1024, 128*1024);
+ snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
+ SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci),
+ 64*1024, 128*1024);
+ } else {
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(trident->pci), 64*1024, 128*1024);
+ }
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_foldback_pcm
+
+ Description: This routine registers the 4DWave device for foldback PCM support.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+
+ Returns: None
+
+ ---------------------------------------------------------------------------*/
+
+int __devinit snd_trident_foldback_pcm(struct snd_trident * trident,
+ int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *foldback;
+ int err;
+ int num_chan = 3;
+ struct snd_pcm_substream *substream;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if (trident->device == TRIDENT_DEVICE_ID_NX)
+ num_chan = 4;
+ if ((err = snd_pcm_new(trident->card, "trident_dx_nx", device, 0, num_chan, &foldback)) < 0)
+ return err;
+
+ foldback->private_data = trident;
+ if (trident->tlb.entries)
+ snd_pcm_set_ops(foldback, SNDRV_PCM_STREAM_CAPTURE, &snd_trident_nx_foldback_ops);
+ else
+ snd_pcm_set_ops(foldback, SNDRV_PCM_STREAM_CAPTURE, &snd_trident_foldback_ops);
+ foldback->info_flags = 0;
+ strcpy(foldback->name, "Trident 4DWave");
+ substream = foldback->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ strcpy(substream->name, "Front Mixer");
+ substream = substream->next;
+ strcpy(substream->name, "Reverb Mixer");
+ substream = substream->next;
+ strcpy(substream->name, "Chorus Mixer");
+ if (num_chan == 4) {
+ substream = substream->next;
+ strcpy(substream->name, "Second AC'97 ADC");
+ }
+ trident->foldback = foldback;
+
+ if (trident->tlb.entries)
+ snd_pcm_lib_preallocate_pages_for_all(foldback, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(trident->pci), 0, 128*1024);
+ else
+ snd_pcm_lib_preallocate_pages_for_all(foldback, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(trident->pci), 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = foldback;
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif
+
+ Description: This routine registers the 4DWave-NX device for SPDIF support.
+
+ Parameters: trident - pointer to target device class for 4DWave-NX.
+
+ Returns: None
+
+ ---------------------------------------------------------------------------*/
+
+int __devinit snd_trident_spdif_pcm(struct snd_trident * trident,
+ int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *spdif;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(trident->card, "trident_dx_nx IEC958", device, 1, 0, &spdif)) < 0)
+ return err;
+
+ spdif->private_data = trident;
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+ snd_pcm_set_ops(spdif, SNDRV_PCM_STREAM_PLAYBACK, &snd_trident_spdif_ops);
+ } else {
+ snd_pcm_set_ops(spdif, SNDRV_PCM_STREAM_PLAYBACK, &snd_trident_spdif_7018_ops);
+ }
+ spdif->info_flags = 0;
+ strcpy(spdif->name, "Trident 4DWave IEC958");
+ trident->spdif = spdif;
+
+ snd_pcm_lib_preallocate_pages_for_all(spdif, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci), 64*1024, 128*1024);
+
+ if (rpcm)
+ *rpcm = spdif;
+ return 0;
+}
+
+/*
+ * Mixer part
+ */
+
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_control
+
+ Description: enable/disable S/PDIF out from ac97 mixer
+ ---------------------------------------------------------------------------*/
+
+#define snd_trident_spdif_control_info snd_ctl_boolean_mono_info
+
+static int snd_trident_spdif_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+
+ spin_lock_irq(&trident->reg_lock);
+ val = trident->spdif_ctrl;
+ ucontrol->value.integer.value[0] = val == kcontrol->private_value;
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+static int snd_trident_spdif_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int change;
+
+ val = ucontrol->value.integer.value[0] ? (unsigned char) kcontrol->private_value : 0x00;
+ spin_lock_irq(&trident->reg_lock);
+ /* S/PDIF C Channel bits 0-31 : 48khz, SCMS disabled */
+ change = trident->spdif_ctrl != val;
+ trident->spdif_ctrl = val;
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+ if ((inb(TRID_REG(trident, NX_SPCTRL_SPCSO + 3)) & 0x10) == 0) {
+ outl(trident->spdif_bits, TRID_REG(trident, NX_SPCSTATUS));
+ outb(trident->spdif_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
+ }
+ } else {
+ if (trident->spdif == NULL) {
+ unsigned int temp;
+ outl(trident->spdif_bits, TRID_REG(trident, SI_SPDIF_CS));
+ temp = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) & ~SPDIF_EN;
+ if (val)
+ temp |= SPDIF_EN;
+ outl(temp, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ }
+ }
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_spdif_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH),
+ .info = snd_trident_spdif_control_info,
+ .get = snd_trident_spdif_control_get,
+ .put = snd_trident_spdif_control_put,
+ .private_value = 0x28,
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_default
+
+ Description: put/get the S/PDIF default settings
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_spdif_default_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_trident_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&trident->reg_lock);
+ ucontrol->value.iec958.status[0] = (trident->spdif_bits >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (trident->spdif_bits >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (trident->spdif_bits >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (trident->spdif_bits >> 24) & 0xff;
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+static int snd_trident_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+ spin_lock_irq(&trident->reg_lock);
+ change = trident->spdif_bits != val;
+ trident->spdif_bits = val;
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+ if ((inb(TRID_REG(trident, NX_SPCTRL_SPCSO + 3)) & 0x10) == 0)
+ outl(trident->spdif_bits, TRID_REG(trident, NX_SPCSTATUS));
+ } else {
+ if (trident->spdif == NULL)
+ outl(trident->spdif_bits, TRID_REG(trident, SI_SPDIF_CS));
+ }
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_spdif_default __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_trident_spdif_default_info,
+ .get = snd_trident_spdif_default_get,
+ .put = snd_trident_spdif_default_put
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_mask
+
+ Description: put/get the S/PDIF mask
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_spdif_mask_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_trident_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_trident_spdif_mask __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .info = snd_trident_spdif_mask_info,
+ .get = snd_trident_spdif_mask_get,
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_spdif_stream
+
+ Description: put/get the S/PDIF stream settings
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_spdif_stream_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_trident_spdif_stream_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&trident->reg_lock);
+ ucontrol->value.iec958.status[0] = (trident->spdif_pcm_bits >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (trident->spdif_pcm_bits >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (trident->spdif_pcm_bits >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (trident->spdif_pcm_bits >> 24) & 0xff;
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+static int snd_trident_spdif_stream_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+ spin_lock_irq(&trident->reg_lock);
+ change = trident->spdif_pcm_bits != val;
+ trident->spdif_pcm_bits = val;
+ if (trident->spdif != NULL) {
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+ outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS));
+ } else {
+ outl(trident->spdif_bits, TRID_REG(trident, SI_SPDIF_CS));
+ }
+ }
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_spdif_stream __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_trident_spdif_stream_info,
+ .get = snd_trident_spdif_stream_get,
+ .put = snd_trident_spdif_stream_put
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_ac97_control
+
+ Description: enable/disable rear path for ac97
+ ---------------------------------------------------------------------------*/
+
+#define snd_trident_ac97_control_info snd_ctl_boolean_mono_info
+
+static int snd_trident_ac97_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+
+ spin_lock_irq(&trident->reg_lock);
+ val = trident->ac97_ctrl = inl(TRID_REG(trident, NX_ACR0_AC97_COM_STAT));
+ ucontrol->value.integer.value[0] = (val & (1 << kcontrol->private_value)) ? 1 : 0;
+ spin_unlock_irq(&trident->reg_lock);
+ return 0;
+}
+
+static int snd_trident_ac97_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned char val;
+ int change = 0;
+
+ spin_lock_irq(&trident->reg_lock);
+ val = trident->ac97_ctrl = inl(TRID_REG(trident, NX_ACR0_AC97_COM_STAT));
+ val &= ~(1 << kcontrol->private_value);
+ if (ucontrol->value.integer.value[0])
+ val |= 1 << kcontrol->private_value;
+ change = val != trident->ac97_ctrl;
+ trident->ac97_ctrl = val;
+ outl(trident->ac97_ctrl = val, TRID_REG(trident, NX_ACR0_AC97_COM_STAT));
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_ac97_rear_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Rear Path",
+ .info = snd_trident_ac97_control_info,
+ .get = snd_trident_ac97_control_get,
+ .put = snd_trident_ac97_control_put,
+ .private_value = 4,
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_vol_control
+
+ Description: wave & music volume control
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_vol_control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_trident_vol_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+
+ val = trident->musicvol_wavevol;
+ ucontrol->value.integer.value[0] = 255 - ((val >> kcontrol->private_value) & 0xff);
+ ucontrol->value.integer.value[1] = 255 - ((val >> (kcontrol->private_value + 8)) & 0xff);
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_gvol, -6375, 25, 0);
+
+static int snd_trident_vol_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+
+ spin_lock_irq(&trident->reg_lock);
+ val = trident->musicvol_wavevol;
+ val &= ~(0xffff << kcontrol->private_value);
+ val |= ((255 - (ucontrol->value.integer.value[0] & 0xff)) |
+ ((255 - (ucontrol->value.integer.value[1] & 0xff)) << 8)) << kcontrol->private_value;
+ change = val != trident->musicvol_wavevol;
+ outl(trident->musicvol_wavevol = val, TRID_REG(trident, T4D_MUSICVOL_WAVEVOL));
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_vol_music_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Music Playback Volume",
+ .info = snd_trident_vol_control_info,
+ .get = snd_trident_vol_control_get,
+ .put = snd_trident_vol_control_put,
+ .private_value = 16,
+ .tlv = { .p = db_scale_gvol },
+};
+
+static struct snd_kcontrol_new snd_trident_vol_wave_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Wave Playback Volume",
+ .info = snd_trident_vol_control_info,
+ .get = snd_trident_vol_control_get,
+ .put = snd_trident_vol_control_put,
+ .private_value = 0,
+ .tlv = { .p = db_scale_gvol },
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_pcm_vol_control
+
+ Description: PCM front volume control
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_pcm_vol_control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018)
+ uinfo->value.integer.max = 1023;
+ return 0;
+}
+
+static int snd_trident_pcm_vol_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ ucontrol->value.integer.value[0] = 1023 - mix->vol;
+ } else {
+ ucontrol->value.integer.value[0] = 255 - (mix->vol>>2);
+ }
+ return 0;
+}
+
+static int snd_trident_pcm_vol_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+ unsigned int val;
+ int change = 0;
+
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ val = 1023 - (ucontrol->value.integer.value[0] & 1023);
+ } else {
+ val = (255 - (ucontrol->value.integer.value[0] & 255)) << 2;
+ }
+ spin_lock_irq(&trident->reg_lock);
+ change = val != mix->vol;
+ mix->vol = val;
+ if (mix->voice != NULL)
+ snd_trident_write_vol_reg(trident, mix->voice, val);
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_pcm_vol_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Front Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .count = 32,
+ .info = snd_trident_pcm_vol_control_info,
+ .get = snd_trident_pcm_vol_control_get,
+ .put = snd_trident_pcm_vol_control_put,
+ /* FIXME: no tlv yet */
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_pcm_pan_control
+
+ Description: PCM front pan control
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_pcm_pan_control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 127;
+ return 0;
+}
+
+static int snd_trident_pcm_pan_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+
+ ucontrol->value.integer.value[0] = mix->pan;
+ if (ucontrol->value.integer.value[0] & 0x40) {
+ ucontrol->value.integer.value[0] = (0x3f - (ucontrol->value.integer.value[0] & 0x3f));
+ } else {
+ ucontrol->value.integer.value[0] |= 0x40;
+ }
+ return 0;
+}
+
+static int snd_trident_pcm_pan_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+ unsigned char val;
+ int change = 0;
+
+ if (ucontrol->value.integer.value[0] & 0x40)
+ val = ucontrol->value.integer.value[0] & 0x3f;
+ else
+ val = (0x3f - (ucontrol->value.integer.value[0] & 0x3f)) | 0x40;
+ spin_lock_irq(&trident->reg_lock);
+ change = val != mix->pan;
+ mix->pan = val;
+ if (mix->voice != NULL)
+ snd_trident_write_pan_reg(trident, mix->voice, val);
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_pcm_pan_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Pan Playback Control",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .count = 32,
+ .info = snd_trident_pcm_pan_control_info,
+ .get = snd_trident_pcm_pan_control_get,
+ .put = snd_trident_pcm_pan_control_put,
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_pcm_rvol_control
+
+ Description: PCM reverb volume control
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_pcm_rvol_control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 127;
+ return 0;
+}
+
+static int snd_trident_pcm_rvol_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+
+ ucontrol->value.integer.value[0] = 127 - mix->rvol;
+ return 0;
+}
+
+static int snd_trident_pcm_rvol_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+ unsigned short val;
+ int change = 0;
+
+ val = 0x7f - (ucontrol->value.integer.value[0] & 0x7f);
+ spin_lock_irq(&trident->reg_lock);
+ change = val != mix->rvol;
+ mix->rvol = val;
+ if (mix->voice != NULL)
+ snd_trident_write_rvol_reg(trident, mix->voice, val);
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_crvol, -3175, 25, 1);
+
+static struct snd_kcontrol_new snd_trident_pcm_rvol_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Reverb Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .count = 32,
+ .info = snd_trident_pcm_rvol_control_info,
+ .get = snd_trident_pcm_rvol_control_get,
+ .put = snd_trident_pcm_rvol_control_put,
+ .tlv = { .p = db_scale_crvol },
+};
+
+/*---------------------------------------------------------------------------
+ snd_trident_pcm_cvol_control
+
+ Description: PCM chorus volume control
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_pcm_cvol_control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 127;
+ return 0;
+}
+
+static int snd_trident_pcm_cvol_control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+
+ ucontrol->value.integer.value[0] = 127 - mix->cvol;
+ return 0;
+}
+
+static int snd_trident_pcm_cvol_control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_trident *trident = snd_kcontrol_chip(kcontrol);
+ struct snd_trident_pcm_mixer *mix = &trident->pcm_mixer[snd_ctl_get_ioffnum(kcontrol, &ucontrol->id)];
+ unsigned short val;
+ int change = 0;
+
+ val = 0x7f - (ucontrol->value.integer.value[0] & 0x7f);
+ spin_lock_irq(&trident->reg_lock);
+ change = val != mix->cvol;
+ mix->cvol = val;
+ if (mix->voice != NULL)
+ snd_trident_write_cvol_reg(trident, mix->voice, val);
+ spin_unlock_irq(&trident->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_trident_pcm_cvol_control __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Chorus Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .count = 32,
+ .info = snd_trident_pcm_cvol_control_info,
+ .get = snd_trident_pcm_cvol_control_get,
+ .put = snd_trident_pcm_cvol_control_put,
+ .tlv = { .p = db_scale_crvol },
+};
+
+static void snd_trident_notify_pcm_change1(struct snd_card *card,
+ struct snd_kcontrol *kctl,
+ int num, int activate)
+{
+ struct snd_ctl_elem_id id;
+
+ if (! kctl)
+ return;
+ if (activate)
+ kctl->vd[num].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ else
+ kctl->vd[num].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ snd_ctl_build_ioff(&id, kctl, num));
+}
+
+static void snd_trident_notify_pcm_change(struct snd_trident *trident,
+ struct snd_trident_pcm_mixer *tmix,
+ int num, int activate)
+{
+ snd_trident_notify_pcm_change1(trident->card, trident->ctl_vol, num, activate);
+ snd_trident_notify_pcm_change1(trident->card, trident->ctl_pan, num, activate);
+ snd_trident_notify_pcm_change1(trident->card, trident->ctl_rvol, num, activate);
+ snd_trident_notify_pcm_change1(trident->card, trident->ctl_cvol, num, activate);
+}
+
+static int snd_trident_pcm_mixer_build(struct snd_trident *trident,
+ struct snd_trident_voice *voice,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_trident_pcm_mixer *tmix;
+
+ if (snd_BUG_ON(!trident || !voice || !substream))
+ return -EINVAL;
+ tmix = &trident->pcm_mixer[substream->number];
+ tmix->voice = voice;
+ tmix->vol = T4D_DEFAULT_PCM_VOL;
+ tmix->pan = T4D_DEFAULT_PCM_PAN;
+ tmix->rvol = T4D_DEFAULT_PCM_RVOL;
+ tmix->cvol = T4D_DEFAULT_PCM_CVOL;
+ snd_trident_notify_pcm_change(trident, tmix, substream->number, 1);
+ return 0;
+}
+
+static int snd_trident_pcm_mixer_free(struct snd_trident *trident, struct snd_trident_voice *voice, struct snd_pcm_substream *substream)
+{
+ struct snd_trident_pcm_mixer *tmix;
+
+ if (snd_BUG_ON(!trident || !substream))
+ return -EINVAL;
+ tmix = &trident->pcm_mixer[substream->number];
+ tmix->voice = NULL;
+ snd_trident_notify_pcm_change(trident, tmix, substream->number, 0);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_mixer
+
+ Description: This routine registers the 4DWave device for mixer support.
+
+ Parameters: trident - pointer to target device class for 4DWave.
+
+ Returns: None
+
+ ---------------------------------------------------------------------------*/
+
+static int __devinit snd_trident_mixer(struct snd_trident * trident, int pcm_spdif_device)
+{
+ struct snd_ac97_template _ac97;
+ struct snd_card *card = trident->card;
+ struct snd_kcontrol *kctl;
+ struct snd_ctl_elem_value *uctl;
+ int idx, err, retries = 2;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_trident_codec_write,
+ .read = snd_trident_codec_read,
+ };
+
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (!uctl)
+ return -ENOMEM;
+
+ if ((err = snd_ac97_bus(trident->card, 0, &ops, NULL, &trident->ac97_bus)) < 0)
+ goto __out;
+
+ memset(&_ac97, 0, sizeof(_ac97));
+ _ac97.private_data = trident;
+ trident->ac97_detect = 1;
+
+ __again:
+ if ((err = snd_ac97_mixer(trident->ac97_bus, &_ac97, &trident->ac97)) < 0) {
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ if ((err = snd_trident_sis_reset(trident)) < 0)
+ goto __out;
+ if (retries-- > 0)
+ goto __again;
+ err = -EIO;
+ }
+ goto __out;
+ }
+
+ /* secondary codec? */
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018 &&
+ (inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) & SI_AC97_PRIMARY_READY) != 0) {
+ _ac97.num = 1;
+ err = snd_ac97_mixer(trident->ac97_bus, &_ac97, &trident->ac97_sec);
+ if (err < 0)
+ snd_printk(KERN_ERR "SI7018: the secondary codec - invalid access\n");
+#if 0 // only for my testing purpose --jk
+ {
+ struct snd_ac97 *mc97;
+ err = snd_ac97_modem(trident->card, &_ac97, &mc97);
+ if (err < 0)
+ snd_printk(KERN_ERR "snd_ac97_modem returned error %i\n", err);
+ }
+#endif
+ }
+
+ trident->ac97_detect = 0;
+
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_trident_vol_wave_control, trident))) < 0)
+ goto __out;
+ kctl->put(kctl, uctl);
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_trident_vol_music_control, trident))) < 0)
+ goto __out;
+ kctl->put(kctl, uctl);
+ outl(trident->musicvol_wavevol = 0x00000000, TRID_REG(trident, T4D_MUSICVOL_WAVEVOL));
+ } else {
+ outl(trident->musicvol_wavevol = 0xffff0000, TRID_REG(trident, T4D_MUSICVOL_WAVEVOL));
+ }
+
+ for (idx = 0; idx < 32; idx++) {
+ struct snd_trident_pcm_mixer *tmix;
+
+ tmix = &trident->pcm_mixer[idx];
+ tmix->voice = NULL;
+ }
+ if ((trident->ctl_vol = snd_ctl_new1(&snd_trident_pcm_vol_control, trident)) == NULL)
+ goto __nomem;
+ if ((err = snd_ctl_add(card, trident->ctl_vol)))
+ goto __out;
+
+ if ((trident->ctl_pan = snd_ctl_new1(&snd_trident_pcm_pan_control, trident)) == NULL)
+ goto __nomem;
+ if ((err = snd_ctl_add(card, trident->ctl_pan)))
+ goto __out;
+
+ if ((trident->ctl_rvol = snd_ctl_new1(&snd_trident_pcm_rvol_control, trident)) == NULL)
+ goto __nomem;
+ if ((err = snd_ctl_add(card, trident->ctl_rvol)))
+ goto __out;
+
+ if ((trident->ctl_cvol = snd_ctl_new1(&snd_trident_pcm_cvol_control, trident)) == NULL)
+ goto __nomem;
+ if ((err = snd_ctl_add(card, trident->ctl_cvol)))
+ goto __out;
+
+ if (trident->device == TRIDENT_DEVICE_ID_NX) {
+ if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_trident_ac97_rear_control, trident))) < 0)
+ goto __out;
+ kctl->put(kctl, uctl);
+ }
+ if (trident->device == TRIDENT_DEVICE_ID_NX || trident->device == TRIDENT_DEVICE_ID_SI7018) {
+
+ kctl = snd_ctl_new1(&snd_trident_spdif_control, trident);
+ if (kctl == NULL) {
+ err = -ENOMEM;
+ goto __out;
+ }
+ if (trident->ac97->ext_id & AC97_EI_SPDIF)
+ kctl->id.index++;
+ if (trident->ac97_sec && (trident->ac97_sec->ext_id & AC97_EI_SPDIF))
+ kctl->id.index++;
+ idx = kctl->id.index;
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ goto __out;
+ kctl->put(kctl, uctl);
+
+ kctl = snd_ctl_new1(&snd_trident_spdif_default, trident);
+ if (kctl == NULL) {
+ err = -ENOMEM;
+ goto __out;
+ }
+ kctl->id.index = idx;
+ kctl->id.device = pcm_spdif_device;
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ goto __out;
+
+ kctl = snd_ctl_new1(&snd_trident_spdif_mask, trident);
+ if (kctl == NULL) {
+ err = -ENOMEM;
+ goto __out;
+ }
+ kctl->id.index = idx;
+ kctl->id.device = pcm_spdif_device;
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ goto __out;
+
+ kctl = snd_ctl_new1(&snd_trident_spdif_stream, trident);
+ if (kctl == NULL) {
+ err = -ENOMEM;
+ goto __out;
+ }
+ kctl->id.index = idx;
+ kctl->id.device = pcm_spdif_device;
+ if ((err = snd_ctl_add(card, kctl)) < 0)
+ goto __out;
+ trident->spdif_pcm_ctl = kctl;
+ }
+
+ err = 0;
+ goto __out;
+
+ __nomem:
+ err = -ENOMEM;
+
+ __out:
+ kfree(uctl);
+
+ return err;
+}
+
+/*
+ * gameport interface
+ */
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+
+static unsigned char snd_trident_gameport_read(struct gameport *gameport)
+{
+ struct snd_trident *chip = gameport_get_port_data(gameport);
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+ return inb(TRID_REG(chip, GAMEPORT_LEGACY));
+}
+
+static void snd_trident_gameport_trigger(struct gameport *gameport)
+{
+ struct snd_trident *chip = gameport_get_port_data(gameport);
+
+ if (snd_BUG_ON(!chip))
+ return;
+ outb(0xff, TRID_REG(chip, GAMEPORT_LEGACY));
+}
+
+static int snd_trident_gameport_cooked_read(struct gameport *gameport, int *axes, int *buttons)
+{
+ struct snd_trident *chip = gameport_get_port_data(gameport);
+ int i;
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+
+ *buttons = (~inb(TRID_REG(chip, GAMEPORT_LEGACY)) >> 4) & 0xf;
+
+ for (i = 0; i < 4; i++) {
+ axes[i] = inw(TRID_REG(chip, GAMEPORT_AXES + i * 2));
+ if (axes[i] == 0xffff) axes[i] = -1;
+ }
+
+ return 0;
+}
+
+static int snd_trident_gameport_open(struct gameport *gameport, int mode)
+{
+ struct snd_trident *chip = gameport_get_port_data(gameport);
+
+ if (snd_BUG_ON(!chip))
+ return 0;
+
+ switch (mode) {
+ case GAMEPORT_MODE_COOKED:
+ outb(GAMEPORT_MODE_ADC, TRID_REG(chip, GAMEPORT_GCR));
+ msleep(20);
+ return 0;
+ case GAMEPORT_MODE_RAW:
+ outb(0, TRID_REG(chip, GAMEPORT_GCR));
+ return 0;
+ default:
+ return -1;
+ }
+}
+
+int __devinit snd_trident_create_gameport(struct snd_trident *chip)
+{
+ struct gameport *gp;
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "trident: cannot allocate memory for gameport\n");
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "Trident 4DWave");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+
+ gameport_set_port_data(gp, chip);
+ gp->fuzz = 64;
+ gp->read = snd_trident_gameport_read;
+ gp->trigger = snd_trident_gameport_trigger;
+ gp->cooked_read = snd_trident_gameport_cooked_read;
+ gp->open = snd_trident_gameport_open;
+
+ gameport_register_port(gp);
+
+ return 0;
+}
+
+static inline void snd_trident_free_gameport(struct snd_trident *chip)
+{
+ if (chip->gameport) {
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+ }
+}
+#else
+int __devinit snd_trident_create_gameport(struct snd_trident *chip) { return -ENOSYS; }
+static inline void snd_trident_free_gameport(struct snd_trident *chip) { }
+#endif /* CONFIG_GAMEPORT */
+
+/*
+ * delay for 1 tick
+ */
+static inline void do_delay(struct snd_trident *chip)
+{
+ schedule_timeout_uninterruptible(1);
+}
+
+/*
+ * SiS reset routine
+ */
+
+static int snd_trident_sis_reset(struct snd_trident *trident)
+{
+ unsigned long end_time;
+ unsigned int i;
+ int r;
+
+ r = trident->in_suspend ? 0 : 2; /* count of retries */
+ __si7018_retry:
+ pci_write_config_byte(trident->pci, 0x46, 0x04); /* SOFTWARE RESET */
+ udelay(100);
+ pci_write_config_byte(trident->pci, 0x46, 0x00);
+ udelay(100);
+ /* disable AC97 GPIO interrupt */
+ outb(0x00, TRID_REG(trident, SI_AC97_GPIO));
+ /* initialize serial interface, force cold reset */
+ i = PCMOUT|SURROUT|CENTEROUT|LFEOUT|SECONDARY_ID|COLD_RESET;
+ outl(i, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ udelay(1000);
+ /* remove cold reset */
+ i &= ~COLD_RESET;
+ outl(i, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ udelay(2000);
+ /* wait, until the codec is ready */
+ end_time = (jiffies + (HZ * 3) / 4) + 1;
+ do {
+ if ((inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) & SI_AC97_PRIMARY_READY) != 0)
+ goto __si7018_ok;
+ do_delay(trident);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "AC'97 codec ready error [0x%x]\n", inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)));
+ if (r-- > 0) {
+ end_time = jiffies + HZ;
+ do {
+ do_delay(trident);
+ } while (time_after_eq(end_time, jiffies));
+ goto __si7018_retry;
+ }
+ __si7018_ok:
+ /* wait for the second codec */
+ do {
+ if ((inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) & SI_AC97_SECONDARY_READY) != 0)
+ break;
+ do_delay(trident);
+ } while (time_after_eq(end_time, jiffies));
+ /* enable 64 channel mode */
+ outl(BANK_B_EN, TRID_REG(trident, T4D_LFO_GC_CIR));
+ return 0;
+}
+
+/*
+ * /proc interface
+ */
+
+static void snd_trident_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_trident *trident = entry->private_data;
+ char *s;
+
+ switch (trident->device) {
+ case TRIDENT_DEVICE_ID_SI7018:
+ s = "SiS 7018 Audio";
+ break;
+ case TRIDENT_DEVICE_ID_DX:
+ s = "Trident 4DWave PCI DX";
+ break;
+ case TRIDENT_DEVICE_ID_NX:
+ s = "Trident 4DWave PCI NX";
+ break;
+ default:
+ s = "???";
+ }
+ snd_iprintf(buffer, "%s\n\n", s);
+ snd_iprintf(buffer, "Spurious IRQs : %d\n", trident->spurious_irq_count);
+ snd_iprintf(buffer, "Spurious IRQ dlta: %d\n", trident->spurious_irq_max_delta);
+ if (trident->device == TRIDENT_DEVICE_ID_NX || trident->device == TRIDENT_DEVICE_ID_SI7018)
+ snd_iprintf(buffer, "IEC958 Mixer Out : %s\n", trident->spdif_ctrl == 0x28 ? "on" : "off");
+ if (trident->device == TRIDENT_DEVICE_ID_NX) {
+ snd_iprintf(buffer, "Rear Speakers : %s\n", trident->ac97_ctrl & 0x00000010 ? "on" : "off");
+ if (trident->tlb.entries) {
+ snd_iprintf(buffer,"\nVirtual Memory\n");
+ snd_iprintf(buffer, "Memory Maximum : %d\n", trident->tlb.memhdr->size);
+ snd_iprintf(buffer, "Memory Used : %d\n", trident->tlb.memhdr->used);
+ snd_iprintf(buffer, "Memory Free : %d\n", snd_util_mem_avail(trident->tlb.memhdr));
+ }
+ }
+}
+
+static void __devinit snd_trident_proc_init(struct snd_trident * trident)
+{
+ struct snd_info_entry *entry;
+ const char *s = "trident";
+
+ if (trident->device == TRIDENT_DEVICE_ID_SI7018)
+ s = "sis7018";
+ if (! snd_card_proc_new(trident->card, s, &entry))
+ snd_info_set_text_ops(entry, trident, snd_trident_proc_read);
+}
+
+static int snd_trident_dev_free(struct snd_device *device)
+{
+ struct snd_trident *trident = device->device_data;
+ return snd_trident_free(trident);
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_tlb_alloc
+
+ Description: Allocate and set up the TLB page table on 4D NX.
+ Each entry has 4 bytes (physical PCI address).
+
+ Parameters: trident - pointer to target device class for 4DWave.
+
+ Returns: 0 or negative error code
+
+ ---------------------------------------------------------------------------*/
+
+static int __devinit snd_trident_tlb_alloc(struct snd_trident *trident)
+{
+ int i;
+
+ /* TLB array must be aligned to 16kB !!! so we allocate
+ 32kB region and correct offset when necessary */
+
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci),
+ 2 * SNDRV_TRIDENT_MAX_PAGES * 4, &trident->tlb.buffer) < 0) {
+ snd_printk(KERN_ERR "trident: unable to allocate TLB buffer\n");
+ return -ENOMEM;
+ }
+ trident->tlb.entries = (unsigned int*)ALIGN((unsigned long)trident->tlb.buffer.area, SNDRV_TRIDENT_MAX_PAGES * 4);
+ trident->tlb.entries_dmaaddr = ALIGN(trident->tlb.buffer.addr, SNDRV_TRIDENT_MAX_PAGES * 4);
+ /* allocate shadow TLB page table (virtual addresses) */
+ trident->tlb.shadow_entries = vmalloc(SNDRV_TRIDENT_MAX_PAGES*sizeof(unsigned long));
+ if (trident->tlb.shadow_entries == NULL) {
+ snd_printk(KERN_ERR "trident: unable to allocate shadow TLB entries\n");
+ return -ENOMEM;
+ }
+ /* allocate and setup silent page and initialise TLB entries */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci),
+ SNDRV_TRIDENT_PAGE_SIZE, &trident->tlb.silent_page) < 0) {
+ snd_printk(KERN_ERR "trident: unable to allocate silent page\n");
+ return -ENOMEM;
+ }
+ memset(trident->tlb.silent_page.area, 0, SNDRV_TRIDENT_PAGE_SIZE);
+ for (i = 0; i < SNDRV_TRIDENT_MAX_PAGES; i++) {
+ trident->tlb.entries[i] = cpu_to_le32(trident->tlb.silent_page.addr & ~(SNDRV_TRIDENT_PAGE_SIZE-1));
+ trident->tlb.shadow_entries[i] = (unsigned long)trident->tlb.silent_page.area;
+ }
+
+ /* use emu memory block manager code to manage tlb page allocation */
+ trident->tlb.memhdr = snd_util_memhdr_new(SNDRV_TRIDENT_PAGE_SIZE * SNDRV_TRIDENT_MAX_PAGES);
+ if (trident->tlb.memhdr == NULL)
+ return -ENOMEM;
+
+ trident->tlb.memhdr->block_extra_size = sizeof(struct snd_trident_memblk_arg);
+ return 0;
+}
+
+/*
+ * initialize 4D DX chip
+ */
+
+static void snd_trident_stop_all_voices(struct snd_trident *trident)
+{
+ outl(0xffffffff, TRID_REG(trident, T4D_STOP_A));
+ outl(0xffffffff, TRID_REG(trident, T4D_STOP_B));
+ outl(0, TRID_REG(trident, T4D_AINTEN_A));
+ outl(0, TRID_REG(trident, T4D_AINTEN_B));
+}
+
+static int snd_trident_4d_dx_init(struct snd_trident *trident)
+{
+ struct pci_dev *pci = trident->pci;
+ unsigned long end_time;
+
+ /* reset the legacy configuration and whole audio/wavetable block */
+ pci_write_config_dword(pci, 0x40, 0); /* DDMA */
+ pci_write_config_byte(pci, 0x44, 0); /* ports */
+ pci_write_config_byte(pci, 0x45, 0); /* Legacy DMA */
+ pci_write_config_byte(pci, 0x46, 4); /* reset */
+ udelay(100);
+ pci_write_config_byte(pci, 0x46, 0); /* release reset */
+ udelay(100);
+
+ /* warm reset of the AC'97 codec */
+ outl(0x00000001, TRID_REG(trident, DX_ACR2_AC97_COM_STAT));
+ udelay(100);
+ outl(0x00000000, TRID_REG(trident, DX_ACR2_AC97_COM_STAT));
+ /* DAC on, disable SB IRQ and try to force ADC valid signal */
+ trident->ac97_ctrl = 0x0000004a;
+ outl(trident->ac97_ctrl, TRID_REG(trident, DX_ACR2_AC97_COM_STAT));
+ /* wait, until the codec is ready */
+ end_time = (jiffies + (HZ * 3) / 4) + 1;
+ do {
+ if ((inl(TRID_REG(trident, DX_ACR2_AC97_COM_STAT)) & 0x0010) != 0)
+ goto __dx_ok;
+ do_delay(trident);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "AC'97 codec ready error\n");
+ return -EIO;
+
+ __dx_ok:
+ snd_trident_stop_all_voices(trident);
+
+ return 0;
+}
+
+/*
+ * initialize 4D NX chip
+ */
+static int snd_trident_4d_nx_init(struct snd_trident *trident)
+{
+ struct pci_dev *pci = trident->pci;
+ unsigned long end_time;
+
+ /* reset the legacy configuration and whole audio/wavetable block */
+ pci_write_config_dword(pci, 0x40, 0); /* DDMA */
+ pci_write_config_byte(pci, 0x44, 0); /* ports */
+ pci_write_config_byte(pci, 0x45, 0); /* Legacy DMA */
+
+ pci_write_config_byte(pci, 0x46, 1); /* reset */
+ udelay(100);
+ pci_write_config_byte(pci, 0x46, 0); /* release reset */
+ udelay(100);
+
+ /* warm reset of the AC'97 codec */
+ outl(0x00000001, TRID_REG(trident, NX_ACR0_AC97_COM_STAT));
+ udelay(100);
+ outl(0x00000000, TRID_REG(trident, NX_ACR0_AC97_COM_STAT));
+ /* wait, until the codec is ready */
+ end_time = (jiffies + (HZ * 3) / 4) + 1;
+ do {
+ if ((inl(TRID_REG(trident, NX_ACR0_AC97_COM_STAT)) & 0x0008) != 0)
+ goto __nx_ok;
+ do_delay(trident);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "AC'97 codec ready error [0x%x]\n", inl(TRID_REG(trident, NX_ACR0_AC97_COM_STAT)));
+ return -EIO;
+
+ __nx_ok:
+ /* DAC on */
+ trident->ac97_ctrl = 0x00000002;
+ outl(trident->ac97_ctrl, TRID_REG(trident, NX_ACR0_AC97_COM_STAT));
+ /* disable SB IRQ */
+ outl(NX_SB_IRQ_DISABLE, TRID_REG(trident, T4D_MISCINT));
+
+ snd_trident_stop_all_voices(trident);
+
+ if (trident->tlb.entries != NULL) {
+ unsigned int i;
+ /* enable virtual addressing via TLB */
+ i = trident->tlb.entries_dmaaddr;
+ i |= 0x00000001;
+ outl(i, TRID_REG(trident, NX_TLBC));
+ } else {
+ outl(0, TRID_REG(trident, NX_TLBC));
+ }
+ /* initialize S/PDIF */
+ outl(trident->spdif_bits, TRID_REG(trident, NX_SPCSTATUS));
+ outb(trident->spdif_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
+
+ return 0;
+}
+
+/*
+ * initialize sis7018 chip
+ */
+static int snd_trident_sis_init(struct snd_trident *trident)
+{
+ int err;
+
+ if ((err = snd_trident_sis_reset(trident)) < 0)
+ return err;
+
+ snd_trident_stop_all_voices(trident);
+
+ /* initialize S/PDIF */
+ outl(trident->spdif_bits, TRID_REG(trident, SI_SPDIF_CS));
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_create
+
+ Description: This routine will create the device specific class for
+ the 4DWave card. It will also perform basic initialization.
+
+ Parameters: card - which card to create
+ pci - interface to PCI bus resource info
+ dma1ptr - playback dma buffer
+ dma2ptr - capture dma buffer
+ irqptr - interrupt resource info
+
+ Returns: 4DWave device class private data
+
+ ---------------------------------------------------------------------------*/
+
+int __devinit snd_trident_create(struct snd_card *card,
+ struct pci_dev *pci,
+ int pcm_streams,
+ int pcm_spdif_device,
+ int max_wavetable_size,
+ struct snd_trident ** rtrident)
+{
+ struct snd_trident *trident;
+ int i, err;
+ struct snd_trident_voice *voice;
+ struct snd_trident_pcm_mixer *tmix;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_trident_dev_free,
+ };
+
+ *rtrident = NULL;
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ /* check, if we can restrict PCI DMA transfers to 30 bits */
+ if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_30BIT_MASK) < 0) {
+ snd_printk(KERN_ERR "architecture does not support 30bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ trident = kzalloc(sizeof(*trident), GFP_KERNEL);
+ if (trident == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ trident->device = (pci->vendor << 16) | pci->device;
+ trident->card = card;
+ trident->pci = pci;
+ spin_lock_init(&trident->reg_lock);
+ spin_lock_init(&trident->event_lock);
+ spin_lock_init(&trident->voice_alloc);
+ if (pcm_streams < 1)
+ pcm_streams = 1;
+ if (pcm_streams > 32)
+ pcm_streams = 32;
+ trident->ChanPCM = pcm_streams;
+ if (max_wavetable_size < 0 )
+ max_wavetable_size = 0;
+ trident->synth.max_size = max_wavetable_size * 1024;
+ trident->irq = -1;
+
+ trident->midi_port = TRID_REG(trident, T4D_MPU401_BASE);
+ pci_set_master(pci);
+
+ if ((err = pci_request_regions(pci, "Trident Audio")) < 0) {
+ kfree(trident);
+ pci_disable_device(pci);
+ return err;
+ }
+ trident->port = pci_resource_start(pci, 0);
+
+ if (request_irq(pci->irq, snd_trident_interrupt, IRQF_SHARED,
+ "Trident Audio", trident)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_trident_free(trident);
+ return -EBUSY;
+ }
+ trident->irq = pci->irq;
+
+ /* allocate 16k-aligned TLB for NX cards */
+ trident->tlb.entries = NULL;
+ trident->tlb.buffer.area = NULL;
+ if (trident->device == TRIDENT_DEVICE_ID_NX) {
+ if ((err = snd_trident_tlb_alloc(trident)) < 0) {
+ snd_trident_free(trident);
+ return err;
+ }
+ }
+
+ trident->spdif_bits = trident->spdif_pcm_bits = SNDRV_PCM_DEFAULT_CON_SPDIF;
+
+ /* initialize chip */
+ switch (trident->device) {
+ case TRIDENT_DEVICE_ID_DX:
+ err = snd_trident_4d_dx_init(trident);
+ break;
+ case TRIDENT_DEVICE_ID_NX:
+ err = snd_trident_4d_nx_init(trident);
+ break;
+ case TRIDENT_DEVICE_ID_SI7018:
+ err = snd_trident_sis_init(trident);
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ if (err < 0) {
+ snd_trident_free(trident);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, trident, &ops)) < 0) {
+ snd_trident_free(trident);
+ return err;
+ }
+
+ if ((err = snd_trident_mixer(trident, pcm_spdif_device)) < 0)
+ return err;
+
+ /* initialise synth voices */
+ for (i = 0; i < 64; i++) {
+ voice = &trident->synth.voices[i];
+ voice->number = i;
+ voice->trident = trident;
+ }
+ /* initialize pcm mixer entries */
+ for (i = 0; i < 32; i++) {
+ tmix = &trident->pcm_mixer[i];
+ tmix->vol = T4D_DEFAULT_PCM_VOL;
+ tmix->pan = T4D_DEFAULT_PCM_PAN;
+ tmix->rvol = T4D_DEFAULT_PCM_RVOL;
+ tmix->cvol = T4D_DEFAULT_PCM_CVOL;
+ }
+
+ snd_trident_enable_eso(trident);
+
+ snd_trident_proc_init(trident);
+ snd_card_set_dev(card, &pci->dev);
+ *rtrident = trident;
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_free
+
+ Description: This routine will free the device specific class for
+ the 4DWave card.
+
+ Parameters: trident - device specific private data for 4DWave card
+
+ Returns: None.
+
+ ---------------------------------------------------------------------------*/
+
+static int snd_trident_free(struct snd_trident *trident)
+{
+ snd_trident_free_gameport(trident);
+ snd_trident_disable_eso(trident);
+ // Disable S/PDIF out
+ if (trident->device == TRIDENT_DEVICE_ID_NX)
+ outb(0x00, TRID_REG(trident, NX_SPCTRL_SPCSO + 3));
+ else if (trident->device == TRIDENT_DEVICE_ID_SI7018) {
+ outl(0, TRID_REG(trident, SI_SERIAL_INTF_CTRL));
+ }
+ if (trident->irq >= 0)
+ free_irq(trident->irq, trident);
+ if (trident->tlb.buffer.area) {
+ outl(0, TRID_REG(trident, NX_TLBC));
+ if (trident->tlb.memhdr)
+ snd_util_memhdr_free(trident->tlb.memhdr);
+ if (trident->tlb.silent_page.area)
+ snd_dma_free_pages(&trident->tlb.silent_page);
+ vfree(trident->tlb.shadow_entries);
+ snd_dma_free_pages(&trident->tlb.buffer);
+ }
+ pci_release_regions(trident->pci);
+ pci_disable_device(trident->pci);
+ kfree(trident);
+ return 0;
+}
+
+/*---------------------------------------------------------------------------
+ snd_trident_interrupt
+
+ Description: ISR for Trident 4DWave device
+
+ Parameters: trident - device specific private data for 4DWave card
+
+ Problems: It seems that Trident chips generates interrupts more than
+ one time in special cases. The spurious interrupts are
+ detected via sample timer (T4D_STIMER) and computing
+ corresponding delta value. The limits are detected with
+ the method try & fail so it is possible that it won't
+ work on all computers. [jaroslav]
+
+ Returns: None.
+
+ ---------------------------------------------------------------------------*/
+
+static irqreturn_t snd_trident_interrupt(int irq, void *dev_id)
+{
+ struct snd_trident *trident = dev_id;
+ unsigned int audio_int, chn_int, stimer, channel, mask, tmp;
+ int delta;
+ struct snd_trident_voice *voice;
+
+ audio_int = inl(TRID_REG(trident, T4D_MISCINT));
+ if ((audio_int & (ADDRESS_IRQ|MPU401_IRQ)) == 0)
+ return IRQ_NONE;
+ if (audio_int & ADDRESS_IRQ) {
+ // get interrupt status for all channels
+ spin_lock(&trident->reg_lock);
+ stimer = inl(TRID_REG(trident, T4D_STIMER)) & 0x00ffffff;
+ chn_int = inl(TRID_REG(trident, T4D_AINT_A));
+ if (chn_int == 0)
+ goto __skip1;
+ outl(chn_int, TRID_REG(trident, T4D_AINT_A)); /* ack */
+ __skip1:
+ chn_int = inl(TRID_REG(trident, T4D_AINT_B));
+ if (chn_int == 0)
+ goto __skip2;
+ for (channel = 63; channel >= 32; channel--) {
+ mask = 1 << (channel&0x1f);
+ if ((chn_int & mask) == 0)
+ continue;
+ voice = &trident->synth.voices[channel];
+ if (!voice->pcm || voice->substream == NULL) {
+ outl(mask, TRID_REG(trident, T4D_STOP_B));
+ continue;
+ }
+ delta = (int)stimer - (int)voice->stimer;
+ if (delta < 0)
+ delta = -delta;
+ if ((unsigned int)delta < voice->spurious_threshold) {
+ /* do some statistics here */
+ trident->spurious_irq_count++;
+ if (trident->spurious_irq_max_delta < (unsigned int)delta)
+ trident->spurious_irq_max_delta = delta;
+ continue;
+ }
+ voice->stimer = stimer;
+ if (voice->isync) {
+ if (!voice->isync3) {
+ tmp = inw(TRID_REG(trident, T4D_SBBL_SBCL));
+ if (trident->bDMAStart & 0x40)
+ tmp >>= 1;
+ if (tmp > 0)
+ tmp = voice->isync_max - tmp;
+ } else {
+ tmp = inl(TRID_REG(trident, NX_SPCTRL_SPCSO)) & 0x00ffffff;
+ }
+ if (tmp < voice->isync_mark) {
+ if (tmp > 0x10)
+ tmp = voice->isync_ESO - 7;
+ else
+ tmp = voice->isync_ESO + 2;
+ /* update ESO for IRQ voice to preserve sync */
+ snd_trident_stop_voice(trident, voice->number);
+ snd_trident_write_eso_reg(trident, voice, tmp);
+ snd_trident_start_voice(trident, voice->number);
+ }
+ } else if (voice->isync2) {
+ voice->isync2 = 0;
+ /* write original ESO and update CSO for IRQ voice to preserve sync */
+ snd_trident_stop_voice(trident, voice->number);
+ snd_trident_write_cso_reg(trident, voice, voice->isync_mark);
+ snd_trident_write_eso_reg(trident, voice, voice->ESO);
+ snd_trident_start_voice(trident, voice->number);
+ }
+#if 0
+ if (voice->extra) {
+ /* update CSO for extra voice to preserve sync */
+ snd_trident_stop_voice(trident, voice->extra->number);
+ snd_trident_write_cso_reg(trident, voice->extra, 0);
+ snd_trident_start_voice(trident, voice->extra->number);
+ }
+#endif
+ spin_unlock(&trident->reg_lock);
+ snd_pcm_period_elapsed(voice->substream);
+ spin_lock(&trident->reg_lock);
+ }
+ outl(chn_int, TRID_REG(trident, T4D_AINT_B)); /* ack */
+ __skip2:
+ spin_unlock(&trident->reg_lock);
+ }
+ if (audio_int & MPU401_IRQ) {
+ if (trident->rmidi) {
+ snd_mpu401_uart_interrupt(irq, trident->rmidi->private_data);
+ } else {
+ inb(TRID_REG(trident, T4D_MPUR0));
+ }
+ }
+ // outl((ST_TARGET_REACHED | MIXER_OVERFLOW | MIXER_UNDERFLOW), TRID_REG(trident, T4D_MISCINT));
+ return IRQ_HANDLED;
+}
+
+struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, int type, int client, int port)
+{
+ struct snd_trident_voice *pvoice;
+ unsigned long flags;
+ int idx;
+
+ spin_lock_irqsave(&trident->voice_alloc, flags);
+ if (type == SNDRV_TRIDENT_VOICE_TYPE_PCM) {
+ idx = snd_trident_allocate_pcm_channel(trident);
+ if(idx < 0) {
+ spin_unlock_irqrestore(&trident->voice_alloc, flags);
+ return NULL;
+ }
+ pvoice = &trident->synth.voices[idx];
+ pvoice->use = 1;
+ pvoice->pcm = 1;
+ pvoice->capture = 0;
+ pvoice->spdif = 0;
+ pvoice->memblk = NULL;
+ pvoice->substream = NULL;
+ spin_unlock_irqrestore(&trident->voice_alloc, flags);
+ return pvoice;
+ }
+ if (type == SNDRV_TRIDENT_VOICE_TYPE_SYNTH) {
+ idx = snd_trident_allocate_synth_channel(trident);
+ if(idx < 0) {
+ spin_unlock_irqrestore(&trident->voice_alloc, flags);
+ return NULL;
+ }
+ pvoice = &trident->synth.voices[idx];
+ pvoice->use = 1;
+ pvoice->synth = 1;
+ pvoice->client = client;
+ pvoice->port = port;
+ pvoice->memblk = NULL;
+ spin_unlock_irqrestore(&trident->voice_alloc, flags);
+ return pvoice;
+ }
+ if (type == SNDRV_TRIDENT_VOICE_TYPE_MIDI) {
+ }
+ spin_unlock_irqrestore(&trident->voice_alloc, flags);
+ return NULL;
+}
+
+EXPORT_SYMBOL(snd_trident_alloc_voice);
+
+void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voice *voice)
+{
+ unsigned long flags;
+ void (*private_free)(struct snd_trident_voice *);
+ void *private_data;
+
+ if (voice == NULL || !voice->use)
+ return;
+ snd_trident_clear_voices(trident, voice->number, voice->number);
+ spin_lock_irqsave(&trident->voice_alloc, flags);
+ private_free = voice->private_free;
+ private_data = voice->private_data;
+ voice->private_free = NULL;
+ voice->private_data = NULL;
+ if (voice->pcm)
+ snd_trident_free_pcm_channel(trident, voice->number);
+ if (voice->synth)
+ snd_trident_free_synth_channel(trident, voice->number);
+ voice->use = voice->pcm = voice->synth = voice->midi = 0;
+ voice->capture = voice->spdif = 0;
+ voice->sample_ops = NULL;
+ voice->substream = NULL;
+ voice->extra = NULL;
+ spin_unlock_irqrestore(&trident->voice_alloc, flags);
+ if (private_free)
+ private_free(voice);
+}
+
+EXPORT_SYMBOL(snd_trident_free_voice);
+
+static void snd_trident_clear_voices(struct snd_trident * trident, unsigned short v_min, unsigned short v_max)
+{
+ unsigned int i, val, mask[2] = { 0, 0 };
+
+ if (snd_BUG_ON(v_min > 63 || v_max > 63))
+ return;
+ for (i = v_min; i <= v_max; i++)
+ mask[i >> 5] |= 1 << (i & 0x1f);
+ if (mask[0]) {
+ outl(mask[0], TRID_REG(trident, T4D_STOP_A));
+ val = inl(TRID_REG(trident, T4D_AINTEN_A));
+ outl(val & ~mask[0], TRID_REG(trident, T4D_AINTEN_A));
+ }
+ if (mask[1]) {
+ outl(mask[1], TRID_REG(trident, T4D_STOP_B));
+ val = inl(TRID_REG(trident, T4D_AINTEN_B));
+ outl(val & ~mask[1], TRID_REG(trident, T4D_AINTEN_B));
+ }
+}
+
+#ifdef CONFIG_PM
+int snd_trident_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_trident *trident = card->private_data;
+
+ trident->in_suspend = 1;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(trident->pcm);
+ snd_pcm_suspend_all(trident->foldback);
+ snd_pcm_suspend_all(trident->spdif);
+
+ snd_ac97_suspend(trident->ac97);
+ snd_ac97_suspend(trident->ac97_sec);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+int snd_trident_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_trident *trident = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "trident: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ switch (trident->device) {
+ case TRIDENT_DEVICE_ID_DX:
+ snd_trident_4d_dx_init(trident);
+ break;
+ case TRIDENT_DEVICE_ID_NX:
+ snd_trident_4d_nx_init(trident);
+ break;
+ case TRIDENT_DEVICE_ID_SI7018:
+ snd_trident_sis_init(trident);
+ break;
+ }
+
+ snd_ac97_resume(trident->ac97);
+ snd_ac97_resume(trident->ac97_sec);
+
+ /* restore some registers */
+ outl(trident->musicvol_wavevol, TRID_REG(trident, T4D_MUSICVOL_WAVEVOL));
+
+ snd_trident_enable_eso(trident);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ trident->in_suspend = 0;
+ return 0;
+}
+#endif /* CONFIG_PM */
diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c
new file mode 100644
index 0000000..f9779e2
--- /dev/null
+++ b/sound/pci/trident/trident_memory.c
@@ -0,0 +1,315 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ * Copyright (c) by Scott McNab <sdm@fractalgraphics.com.au>
+ *
+ * Trident 4DWave-NX memory page allocation (TLB area)
+ * Trident chip can handle only 16MByte of the memory at the same time.
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <asm/io.h>
+#include <linux/pci.h>
+#include <linux/time.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/trident.h>
+
+/* page arguments of these two macros are Trident page (4096 bytes), not like
+ * aligned pages in others
+ */
+#define __set_tlb_bus(trident,page,ptr,addr) \
+ do { (trident)->tlb.entries[page] = cpu_to_le32((addr) & ~(SNDRV_TRIDENT_PAGE_SIZE-1)); \
+ (trident)->tlb.shadow_entries[page] = (ptr); } while (0)
+#define __tlb_to_ptr(trident,page) \
+ (void*)((trident)->tlb.shadow_entries[page])
+#define __tlb_to_addr(trident,page) \
+ (dma_addr_t)le32_to_cpu((trident->tlb.entries[page]) & ~(SNDRV_TRIDENT_PAGE_SIZE - 1))
+
+#if PAGE_SIZE == 4096
+/* page size == SNDRV_TRIDENT_PAGE_SIZE */
+#define ALIGN_PAGE_SIZE PAGE_SIZE /* minimum page size for allocation */
+#define MAX_ALIGN_PAGES SNDRV_TRIDENT_MAX_PAGES /* maxmium aligned pages */
+/* fill TLB entrie(s) corresponding to page with ptr */
+#define set_tlb_bus(trident,page,ptr,addr) __set_tlb_bus(trident,page,ptr,addr)
+/* fill TLB entrie(s) corresponding to page with silence pointer */
+#define set_silent_tlb(trident,page) __set_tlb_bus(trident, page, (unsigned long)trident->tlb.silent_page.area, trident->tlb.silent_page.addr)
+/* get aligned page from offset address */
+#define get_aligned_page(offset) ((offset) >> 12)
+/* get offset address from aligned page */
+#define aligned_page_offset(page) ((page) << 12)
+/* get buffer address from aligned page */
+#define page_to_ptr(trident,page) __tlb_to_ptr(trident, page)
+/* get PCI physical address from aligned page */
+#define page_to_addr(trident,page) __tlb_to_addr(trident, page)
+
+#elif PAGE_SIZE == 8192
+/* page size == SNDRV_TRIDENT_PAGE_SIZE x 2*/
+#define ALIGN_PAGE_SIZE PAGE_SIZE
+#define MAX_ALIGN_PAGES (SNDRV_TRIDENT_MAX_PAGES / 2)
+#define get_aligned_page(offset) ((offset) >> 13)
+#define aligned_page_offset(page) ((page) << 13)
+#define page_to_ptr(trident,page) __tlb_to_ptr(trident, (page) << 1)
+#define page_to_addr(trident,page) __tlb_to_addr(trident, (page) << 1)
+
+/* fill TLB entries -- we need to fill two entries */
+static inline void set_tlb_bus(struct snd_trident *trident, int page,
+ unsigned long ptr, dma_addr_t addr)
+{
+ page <<= 1;
+ __set_tlb_bus(trident, page, ptr, addr);
+ __set_tlb_bus(trident, page+1, ptr + SNDRV_TRIDENT_PAGE_SIZE, addr + SNDRV_TRIDENT_PAGE_SIZE);
+}
+static inline void set_silent_tlb(struct snd_trident *trident, int page)
+{
+ page <<= 1;
+ __set_tlb_bus(trident, page, (unsigned long)trident->tlb.silent_page.area, trident->tlb.silent_page.addr);
+ __set_tlb_bus(trident, page+1, (unsigned long)trident->tlb.silent_page.area, trident->tlb.silent_page.addr);
+}
+
+#else
+/* arbitrary size */
+#define UNIT_PAGES (PAGE_SIZE / SNDRV_TRIDENT_PAGE_SIZE)
+#define ALIGN_PAGE_SIZE (SNDRV_TRIDENT_PAGE_SIZE * UNIT_PAGES)
+#define MAX_ALIGN_PAGES (SNDRV_TRIDENT_MAX_PAGES / UNIT_PAGES)
+/* Note: if alignment doesn't match to the maximum size, the last few blocks
+ * become unusable. To use such blocks, you'll need to check the validity
+ * of accessing page in set_tlb_bus and set_silent_tlb. search_empty()
+ * should also check it, too.
+ */
+#define get_aligned_page(offset) ((offset) / ALIGN_PAGE_SIZE)
+#define aligned_page_offset(page) ((page) * ALIGN_PAGE_SIZE)
+#define page_to_ptr(trident,page) __tlb_to_ptr(trident, (page) * UNIT_PAGES)
+#define page_to_addr(trident,page) __tlb_to_addr(trident, (page) * UNIT_PAGES)
+
+/* fill TLB entries -- UNIT_PAGES entries must be filled */
+static inline void set_tlb_bus(struct snd_trident *trident, int page,
+ unsigned long ptr, dma_addr_t addr)
+{
+ int i;
+ page *= UNIT_PAGES;
+ for (i = 0; i < UNIT_PAGES; i++, page++) {
+ __set_tlb_bus(trident, page, ptr, addr);
+ ptr += SNDRV_TRIDENT_PAGE_SIZE;
+ addr += SNDRV_TRIDENT_PAGE_SIZE;
+ }
+}
+static inline void set_silent_tlb(struct snd_trident *trident, int page)
+{
+ int i;
+ page *= UNIT_PAGES;
+ for (i = 0; i < UNIT_PAGES; i++, page++)
+ __set_tlb_bus(trident, page, (unsigned long)trident->tlb.silent_page.area, trident->tlb.silent_page.addr);
+}
+
+#endif /* PAGE_SIZE */
+
+/* calculate buffer pointer from offset address */
+static inline void *offset_ptr(struct snd_trident *trident, int offset)
+{
+ char *ptr;
+ ptr = page_to_ptr(trident, get_aligned_page(offset));
+ ptr += offset % ALIGN_PAGE_SIZE;
+ return (void*)ptr;
+}
+
+/* first and last (aligned) pages of memory block */
+#define firstpg(blk) (((struct snd_trident_memblk_arg *)snd_util_memblk_argptr(blk))->first_page)
+#define lastpg(blk) (((struct snd_trident_memblk_arg *)snd_util_memblk_argptr(blk))->last_page)
+
+/*
+ * search empty pages which may contain given size
+ */
+static struct snd_util_memblk *
+search_empty(struct snd_util_memhdr *hdr, int size)
+{
+ struct snd_util_memblk *blk, *prev;
+ int page, psize;
+ struct list_head *p;
+
+ psize = get_aligned_page(size + ALIGN_PAGE_SIZE -1);
+ prev = NULL;
+ page = 0;
+ list_for_each(p, &hdr->block) {
+ blk = list_entry(p, struct snd_util_memblk, list);
+ if (page + psize <= firstpg(blk))
+ goto __found_pages;
+ page = lastpg(blk) + 1;
+ }
+ if (page + psize > MAX_ALIGN_PAGES)
+ return NULL;
+
+__found_pages:
+ /* create a new memory block */
+ blk = __snd_util_memblk_new(hdr, psize * ALIGN_PAGE_SIZE, p->prev);
+ if (blk == NULL)
+ return NULL;
+ blk->offset = aligned_page_offset(page); /* set aligned offset */
+ firstpg(blk) = page;
+ lastpg(blk) = page + psize - 1;
+ return blk;
+}
+
+
+/*
+ * check if the given pointer is valid for pages
+ */
+static int is_valid_page(unsigned long ptr)
+{
+ if (ptr & ~0x3fffffffUL) {
+ snd_printk(KERN_ERR "max memory size is 1GB!!\n");
+ return 0;
+ }
+ if (ptr & (SNDRV_TRIDENT_PAGE_SIZE-1)) {
+ snd_printk(KERN_ERR "page is not aligned\n");
+ return 0;
+ }
+ return 1;
+}
+
+/*
+ * page allocation for DMA (Scatter-Gather version)
+ */
+static struct snd_util_memblk *
+snd_trident_alloc_sg_pages(struct snd_trident *trident,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_util_memhdr *hdr;
+ struct snd_util_memblk *blk;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int idx, page;
+
+ if (snd_BUG_ON(runtime->dma_bytes <= 0 ||
+ runtime->dma_bytes > SNDRV_TRIDENT_MAX_PAGES *
+ SNDRV_TRIDENT_PAGE_SIZE))
+ return NULL;
+ hdr = trident->tlb.memhdr;
+ if (snd_BUG_ON(!hdr))
+ return NULL;
+
+
+
+ mutex_lock(&hdr->block_mutex);
+ blk = search_empty(hdr, runtime->dma_bytes);
+ if (blk == NULL) {
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+
+ /* set TLB entries */
+ idx = 0;
+ for (page = firstpg(blk); page <= lastpg(blk); page++, idx++) {
+ unsigned long ofs = idx << PAGE_SHIFT;
+ dma_addr_t addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+ unsigned long ptr = (unsigned long)
+ snd_pcm_sgbuf_get_ptr(substream, ofs);
+ if (! is_valid_page(addr)) {
+ __snd_util_mem_free(hdr, blk);
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+ set_tlb_bus(trident, page, ptr, addr);
+ }
+ mutex_unlock(&hdr->block_mutex);
+ return blk;
+}
+
+/*
+ * page allocation for DMA (contiguous version)
+ */
+static struct snd_util_memblk *
+snd_trident_alloc_cont_pages(struct snd_trident *trident,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_util_memhdr *hdr;
+ struct snd_util_memblk *blk;
+ int page;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ dma_addr_t addr;
+ unsigned long ptr;
+
+ if (snd_BUG_ON(runtime->dma_bytes <= 0 ||
+ runtime->dma_bytes > SNDRV_TRIDENT_MAX_PAGES *
+ SNDRV_TRIDENT_PAGE_SIZE))
+ return NULL;
+ hdr = trident->tlb.memhdr;
+ if (snd_BUG_ON(!hdr))
+ return NULL;
+
+ mutex_lock(&hdr->block_mutex);
+ blk = search_empty(hdr, runtime->dma_bytes);
+ if (blk == NULL) {
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+
+ /* set TLB entries */
+ addr = runtime->dma_addr;
+ ptr = (unsigned long)runtime->dma_area;
+ for (page = firstpg(blk); page <= lastpg(blk); page++,
+ ptr += SNDRV_TRIDENT_PAGE_SIZE, addr += SNDRV_TRIDENT_PAGE_SIZE) {
+ if (! is_valid_page(addr)) {
+ __snd_util_mem_free(hdr, blk);
+ mutex_unlock(&hdr->block_mutex);
+ return NULL;
+ }
+ set_tlb_bus(trident, page, ptr, addr);
+ }
+ mutex_unlock(&hdr->block_mutex);
+ return blk;
+}
+
+/*
+ * page allocation for DMA
+ */
+struct snd_util_memblk *
+snd_trident_alloc_pages(struct snd_trident *trident,
+ struct snd_pcm_substream *substream)
+{
+ if (snd_BUG_ON(!trident || !substream))
+ return NULL;
+ if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_SG)
+ return snd_trident_alloc_sg_pages(trident, substream);
+ else
+ return snd_trident_alloc_cont_pages(trident, substream);
+}
+
+
+/*
+ * release DMA buffer from page table
+ */
+int snd_trident_free_pages(struct snd_trident *trident,
+ struct snd_util_memblk *blk)
+{
+ struct snd_util_memhdr *hdr;
+ int page;
+
+ if (snd_BUG_ON(!trident || !blk))
+ return -EINVAL;
+
+ hdr = trident->tlb.memhdr;
+ mutex_lock(&hdr->block_mutex);
+ /* reset TLB entries */
+ for (page = firstpg(blk); page <= lastpg(blk); page++)
+ set_silent_tlb(trident, page);
+ /* free memory block */
+ __snd_util_mem_free(hdr, blk);
+ mutex_unlock(&hdr->block_mutex);
+ return 0;
+}
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
new file mode 100644
index 0000000..1aafe95
--- /dev/null
+++ b/sound/pci/via82xx.c
@@ -0,0 +1,2563 @@
+/*
+ * ALSA driver for VIA VT82xx (South Bridge)
+ *
+ * VT82C686A/B/C, VT8233A/C, VT8235
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ * Tjeerd.Mulder <Tjeerd.Mulder@fujitsu-siemens.com>
+ * 2002 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * Changes:
+ *
+ * Dec. 19, 2002 Takashi Iwai <tiwai@suse.de>
+ * - use the DSX channels for the first pcm playback.
+ * (on VIA8233, 8233C and 8235 only)
+ * this will allow you play simultaneously up to 4 streams.
+ * multi-channel playback is assigned to the second device
+ * on these chips.
+ * - support the secondary capture (on VIA8233/C,8235)
+ * - SPDIF support
+ * the DSX3 channel can be used for SPDIF output.
+ * on VIA8233A, this channel is assigned to the second pcm
+ * playback.
+ * the card config of alsa-lib will assign the correct
+ * device for applications.
+ * - clean up the code, separate low-level initialization
+ * routines for each chipset.
+ *
+ * Sep. 26, 2005 Karsten Wiese <annabellesgarden@yahoo.de>
+ * - Optimize position calculation for the 823x chips.
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/gameport.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+#include <sound/ac97_codec.h>
+#include <sound/mpu401.h>
+#include <sound/initval.h>
+
+#if 0
+#define POINTER_DEBUG
+#endif
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("VIA VT82xx audio");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C,pci},{VIA,VT8233A/C,8235}}");
+
+#if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE))
+#define SUPPORT_JOYSTICK 1
+#endif
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static long mpu_port;
+#ifdef SUPPORT_JOYSTICK
+static int joystick;
+#endif
+static int ac97_clock = 48000;
+static char *ac97_quirk;
+static int dxs_support;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for VIA 82xx bridge.");
+module_param(mpu_port, long, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 port. (VT82C686x only)");
+#ifdef SUPPORT_JOYSTICK
+module_param(joystick, bool, 0444);
+MODULE_PARM_DESC(joystick, "Enable joystick. (VT82C686x only)");
+#endif
+module_param(ac97_clock, int, 0444);
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz).");
+module_param(ac97_quirk, charp, 0444);
+MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
+module_param(dxs_support, int, 0444);
+MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
+
+/* revision numbers for via686 */
+#define VIA_REV_686_A 0x10
+#define VIA_REV_686_B 0x11
+#define VIA_REV_686_C 0x12
+#define VIA_REV_686_D 0x13
+#define VIA_REV_686_E 0x14
+#define VIA_REV_686_H 0x20
+
+/* revision numbers for via8233 */
+#define VIA_REV_PRE_8233 0x10 /* not in market */
+#define VIA_REV_8233C 0x20 /* 2 rec, 4 pb, 1 multi-pb */
+#define VIA_REV_8233 0x30 /* 2 rec, 4 pb, 1 multi-pb, spdif */
+#define VIA_REV_8233A 0x40 /* 1 rec, 1 multi-pb, spdf */
+#define VIA_REV_8235 0x50 /* 2 rec, 4 pb, 1 multi-pb, spdif */
+#define VIA_REV_8237 0x60
+#define VIA_REV_8251 0x70
+
+/*
+ * Direct registers
+ */
+
+#define VIAREG(via, x) ((via)->port + VIA_REG_##x)
+#define VIADEV_REG(viadev, x) ((viadev)->port + VIA_REG_##x)
+
+/* common offsets */
+#define VIA_REG_OFFSET_STATUS 0x00 /* byte - channel status */
+#define VIA_REG_STAT_ACTIVE 0x80 /* RO */
+#define VIA8233_SHADOW_STAT_ACTIVE 0x08 /* RO */
+#define VIA_REG_STAT_PAUSED 0x40 /* RO */
+#define VIA_REG_STAT_TRIGGER_QUEUED 0x08 /* RO */
+#define VIA_REG_STAT_STOPPED 0x04 /* RWC */
+#define VIA_REG_STAT_EOL 0x02 /* RWC */
+#define VIA_REG_STAT_FLAG 0x01 /* RWC */
+#define VIA_REG_OFFSET_CONTROL 0x01 /* byte - channel control */
+#define VIA_REG_CTRL_START 0x80 /* WO */
+#define VIA_REG_CTRL_TERMINATE 0x40 /* WO */
+#define VIA_REG_CTRL_AUTOSTART 0x20
+#define VIA_REG_CTRL_PAUSE 0x08 /* RW */
+#define VIA_REG_CTRL_INT_STOP 0x04
+#define VIA_REG_CTRL_INT_EOL 0x02
+#define VIA_REG_CTRL_INT_FLAG 0x01
+#define VIA_REG_CTRL_RESET 0x01 /* RW - probably reset? undocumented */
+#define VIA_REG_CTRL_INT (VIA_REG_CTRL_INT_FLAG | VIA_REG_CTRL_INT_EOL | VIA_REG_CTRL_AUTOSTART)
+#define VIA_REG_OFFSET_TYPE 0x02 /* byte - channel type (686 only) */
+#define VIA_REG_TYPE_AUTOSTART 0x80 /* RW - autostart at EOL */
+#define VIA_REG_TYPE_16BIT 0x20 /* RW */
+#define VIA_REG_TYPE_STEREO 0x10 /* RW */
+#define VIA_REG_TYPE_INT_LLINE 0x00
+#define VIA_REG_TYPE_INT_LSAMPLE 0x04
+#define VIA_REG_TYPE_INT_LESSONE 0x08
+#define VIA_REG_TYPE_INT_MASK 0x0c
+#define VIA_REG_TYPE_INT_EOL 0x02
+#define VIA_REG_TYPE_INT_FLAG 0x01
+#define VIA_REG_OFFSET_TABLE_PTR 0x04 /* dword - channel table pointer */
+#define VIA_REG_OFFSET_CURR_PTR 0x04 /* dword - channel current pointer */
+#define VIA_REG_OFFSET_STOP_IDX 0x08 /* dword - stop index, channel type, sample rate */
+#define VIA8233_REG_TYPE_16BIT 0x00200000 /* RW */
+#define VIA8233_REG_TYPE_STEREO 0x00100000 /* RW */
+#define VIA_REG_OFFSET_CURR_COUNT 0x0c /* dword - channel current count (24 bit) */
+#define VIA_REG_OFFSET_CURR_INDEX 0x0f /* byte - channel current index (for via8233 only) */
+
+#define DEFINE_VIA_REGSET(name,val) \
+enum {\
+ VIA_REG_##name##_STATUS = (val),\
+ VIA_REG_##name##_CONTROL = (val) + 0x01,\
+ VIA_REG_##name##_TYPE = (val) + 0x02,\
+ VIA_REG_##name##_TABLE_PTR = (val) + 0x04,\
+ VIA_REG_##name##_CURR_PTR = (val) + 0x04,\
+ VIA_REG_##name##_STOP_IDX = (val) + 0x08,\
+ VIA_REG_##name##_CURR_COUNT = (val) + 0x0c,\
+}
+
+/* playback block */
+DEFINE_VIA_REGSET(PLAYBACK, 0x00);
+DEFINE_VIA_REGSET(CAPTURE, 0x10);
+DEFINE_VIA_REGSET(FM, 0x20);
+
+/* AC'97 */
+#define VIA_REG_AC97 0x80 /* dword */
+#define VIA_REG_AC97_CODEC_ID_MASK (3<<30)
+#define VIA_REG_AC97_CODEC_ID_SHIFT 30
+#define VIA_REG_AC97_CODEC_ID_PRIMARY 0x00
+#define VIA_REG_AC97_CODEC_ID_SECONDARY 0x01
+#define VIA_REG_AC97_SECONDARY_VALID (1<<27)
+#define VIA_REG_AC97_PRIMARY_VALID (1<<25)
+#define VIA_REG_AC97_BUSY (1<<24)
+#define VIA_REG_AC97_READ (1<<23)
+#define VIA_REG_AC97_CMD_SHIFT 16
+#define VIA_REG_AC97_CMD_MASK 0x7e
+#define VIA_REG_AC97_DATA_SHIFT 0
+#define VIA_REG_AC97_DATA_MASK 0xffff
+
+#define VIA_REG_SGD_SHADOW 0x84 /* dword */
+/* via686 */
+#define VIA_REG_SGD_STAT_PB_FLAG (1<<0)
+#define VIA_REG_SGD_STAT_CP_FLAG (1<<1)
+#define VIA_REG_SGD_STAT_FM_FLAG (1<<2)
+#define VIA_REG_SGD_STAT_PB_EOL (1<<4)
+#define VIA_REG_SGD_STAT_CP_EOL (1<<5)
+#define VIA_REG_SGD_STAT_FM_EOL (1<<6)
+#define VIA_REG_SGD_STAT_PB_STOP (1<<8)
+#define VIA_REG_SGD_STAT_CP_STOP (1<<9)
+#define VIA_REG_SGD_STAT_FM_STOP (1<<10)
+#define VIA_REG_SGD_STAT_PB_ACTIVE (1<<12)
+#define VIA_REG_SGD_STAT_CP_ACTIVE (1<<13)
+#define VIA_REG_SGD_STAT_FM_ACTIVE (1<<14)
+/* via8233 */
+#define VIA8233_REG_SGD_STAT_FLAG (1<<0)
+#define VIA8233_REG_SGD_STAT_EOL (1<<1)
+#define VIA8233_REG_SGD_STAT_STOP (1<<2)
+#define VIA8233_REG_SGD_STAT_ACTIVE (1<<3)
+#define VIA8233_INTR_MASK(chan) ((VIA8233_REG_SGD_STAT_FLAG|VIA8233_REG_SGD_STAT_EOL) << ((chan) * 4))
+#define VIA8233_REG_SGD_CHAN_SDX 0
+#define VIA8233_REG_SGD_CHAN_MULTI 4
+#define VIA8233_REG_SGD_CHAN_REC 6
+#define VIA8233_REG_SGD_CHAN_REC1 7
+
+#define VIA_REG_GPI_STATUS 0x88
+#define VIA_REG_GPI_INTR 0x8c
+
+/* multi-channel and capture registers for via8233 */
+DEFINE_VIA_REGSET(MULTPLAY, 0x40);
+DEFINE_VIA_REGSET(CAPTURE_8233, 0x60);
+
+/* via8233-specific registers */
+#define VIA_REG_OFS_PLAYBACK_VOLUME_L 0x02 /* byte */
+#define VIA_REG_OFS_PLAYBACK_VOLUME_R 0x03 /* byte */
+#define VIA_REG_OFS_MULTPLAY_FORMAT 0x02 /* byte - format and channels */
+#define VIA_REG_MULTPLAY_FMT_8BIT 0x00
+#define VIA_REG_MULTPLAY_FMT_16BIT 0x80
+#define VIA_REG_MULTPLAY_FMT_CH_MASK 0x70 /* # channels << 4 (valid = 1,2,4,6) */
+#define VIA_REG_OFS_CAPTURE_FIFO 0x02 /* byte - bit 6 = fifo enable */
+#define VIA_REG_CAPTURE_FIFO_ENABLE 0x40
+
+#define VIA_DXS_MAX_VOLUME 31 /* max. volume (attenuation) of reg 0x32/33 */
+
+#define VIA_REG_CAPTURE_CHANNEL 0x63 /* byte - input select */
+#define VIA_REG_CAPTURE_CHANNEL_MIC 0x4
+#define VIA_REG_CAPTURE_CHANNEL_LINE 0
+#define VIA_REG_CAPTURE_SELECT_CODEC 0x03 /* recording source codec (0 = primary) */
+
+#define VIA_TBL_BIT_FLAG 0x40000000
+#define VIA_TBL_BIT_EOL 0x80000000
+
+/* pci space */
+#define VIA_ACLINK_STAT 0x40
+#define VIA_ACLINK_C11_READY 0x20
+#define VIA_ACLINK_C10_READY 0x10
+#define VIA_ACLINK_C01_READY 0x04 /* secondary codec ready */
+#define VIA_ACLINK_LOWPOWER 0x02 /* low-power state */
+#define VIA_ACLINK_C00_READY 0x01 /* primary codec ready */
+#define VIA_ACLINK_CTRL 0x41
+#define VIA_ACLINK_CTRL_ENABLE 0x80 /* 0: disable, 1: enable */
+#define VIA_ACLINK_CTRL_RESET 0x40 /* 0: assert, 1: de-assert */
+#define VIA_ACLINK_CTRL_SYNC 0x20 /* 0: release SYNC, 1: force SYNC hi */
+#define VIA_ACLINK_CTRL_SDO 0x10 /* 0: release SDO, 1: force SDO hi */
+#define VIA_ACLINK_CTRL_VRA 0x08 /* 0: disable VRA, 1: enable VRA */
+#define VIA_ACLINK_CTRL_PCM 0x04 /* 0: disable PCM, 1: enable PCM */
+#define VIA_ACLINK_CTRL_FM 0x02 /* via686 only */
+#define VIA_ACLINK_CTRL_SB 0x01 /* via686 only */
+#define VIA_ACLINK_CTRL_INIT (VIA_ACLINK_CTRL_ENABLE|\
+ VIA_ACLINK_CTRL_RESET|\
+ VIA_ACLINK_CTRL_PCM|\
+ VIA_ACLINK_CTRL_VRA)
+#define VIA_FUNC_ENABLE 0x42
+#define VIA_FUNC_MIDI_PNP 0x80 /* FIXME: it's 0x40 in the datasheet! */
+#define VIA_FUNC_MIDI_IRQMASK 0x40 /* FIXME: not documented! */
+#define VIA_FUNC_RX2C_WRITE 0x20
+#define VIA_FUNC_SB_FIFO_EMPTY 0x10
+#define VIA_FUNC_ENABLE_GAME 0x08
+#define VIA_FUNC_ENABLE_FM 0x04
+#define VIA_FUNC_ENABLE_MIDI 0x02
+#define VIA_FUNC_ENABLE_SB 0x01
+#define VIA_PNP_CONTROL 0x43
+#define VIA_FM_NMI_CTRL 0x48
+#define VIA8233_VOLCHG_CTRL 0x48
+#define VIA8233_SPDIF_CTRL 0x49
+#define VIA8233_SPDIF_DX3 0x08
+#define VIA8233_SPDIF_SLOT_MASK 0x03
+#define VIA8233_SPDIF_SLOT_1011 0x00
+#define VIA8233_SPDIF_SLOT_34 0x01
+#define VIA8233_SPDIF_SLOT_78 0x02
+#define VIA8233_SPDIF_SLOT_69 0x03
+
+/*
+ */
+
+#define VIA_DXS_AUTO 0
+#define VIA_DXS_ENABLE 1
+#define VIA_DXS_DISABLE 2
+#define VIA_DXS_48K 3
+#define VIA_DXS_NO_VRA 4
+#define VIA_DXS_SRC 5
+
+
+/*
+ * pcm stream
+ */
+
+struct snd_via_sg_table {
+ unsigned int offset;
+ unsigned int size;
+} ;
+
+#define VIA_TABLE_SIZE 255
+#define VIA_MAX_BUFSIZE (1<<24)
+
+struct viadev {
+ unsigned int reg_offset;
+ unsigned long port;
+ int direction; /* playback = 0, capture = 1 */
+ struct snd_pcm_substream *substream;
+ int running;
+ unsigned int tbl_entries; /* # descriptors */
+ struct snd_dma_buffer table;
+ struct snd_via_sg_table *idx_table;
+ /* for recovery from the unexpected pointer */
+ unsigned int lastpos;
+ unsigned int fragsize;
+ unsigned int bufsize;
+ unsigned int bufsize2;
+ int hwptr_done; /* processed frame position in the buffer */
+ int in_interrupt;
+ int shadow_shift;
+};
+
+
+enum { TYPE_CARD_VIA686 = 1, TYPE_CARD_VIA8233 };
+enum { TYPE_VIA686, TYPE_VIA8233, TYPE_VIA8233A };
+
+#define VIA_MAX_DEVS 7 /* 4 playback, 1 multi, 2 capture */
+
+struct via_rate_lock {
+ spinlock_t lock;
+ int rate;
+ int used;
+};
+
+struct via82xx {
+ int irq;
+
+ unsigned long port;
+ struct resource *mpu_res;
+ int chip_type;
+ unsigned char revision;
+
+ unsigned char old_legacy;
+ unsigned char old_legacy_cfg;
+#ifdef CONFIG_PM
+ unsigned char legacy_saved;
+ unsigned char legacy_cfg_saved;
+ unsigned char spdif_ctrl_saved;
+ unsigned char capture_src_saved[2];
+ unsigned int mpu_port_saved;
+#endif
+
+ unsigned char playback_volume[4][2]; /* for VIA8233/C/8235; default = 0 */
+ unsigned char playback_volume_c[2]; /* for VIA8233/C/8235; default = 0 */
+
+ unsigned int intr_mask; /* SGD_SHADOW mask to check interrupts */
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+
+ unsigned int num_devs;
+ unsigned int playback_devno, multi_devno, capture_devno;
+ struct viadev devs[VIA_MAX_DEVS];
+ struct via_rate_lock rates[2]; /* playback and capture */
+ unsigned int dxs_fixed: 1; /* DXS channel accepts only 48kHz */
+ unsigned int no_vra: 1; /* no need to set VRA on DXS channels */
+ unsigned int dxs_src: 1; /* use full SRC capabilities of DXS */
+ unsigned int spdif_on: 1; /* only spdif rates work to external DACs */
+
+ struct snd_pcm *pcms[2];
+ struct snd_rawmidi *rmidi;
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97;
+ unsigned int ac97_clock;
+ unsigned int ac97_secondary; /* secondary AC'97 codec is present */
+
+ spinlock_t reg_lock;
+ struct snd_info_entry *proc_entry;
+
+#ifdef SUPPORT_JOYSTICK
+ struct gameport *gameport;
+#endif
+};
+
+static struct pci_device_id snd_via82xx_ids[] = {
+ /* 0x1106, 0x3058 */
+ { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_82C686_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA686, }, /* 686A */
+ /* 0x1106, 0x3059 */
+ { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_8233_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA8233, }, /* VT8233 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_via82xx_ids);
+
+/*
+ */
+
+/*
+ * allocate and initialize the descriptor buffers
+ * periods = number of periods
+ * fragsize = period size in bytes
+ */
+static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substream,
+ struct pci_dev *pci,
+ unsigned int periods, unsigned int fragsize)
+{
+ unsigned int i, idx, ofs, rest;
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+
+ if (dev->table.area == NULL) {
+ /* the start of each lists must be aligned to 8 bytes,
+ * but the kernel pages are much bigger, so we don't care
+ */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ PAGE_ALIGN(VIA_TABLE_SIZE * 2 * 8),
+ &dev->table) < 0)
+ return -ENOMEM;
+ }
+ if (! dev->idx_table) {
+ dev->idx_table = kmalloc(sizeof(*dev->idx_table) * VIA_TABLE_SIZE, GFP_KERNEL);
+ if (! dev->idx_table)
+ return -ENOMEM;
+ }
+
+ /* fill the entries */
+ idx = 0;
+ ofs = 0;
+ for (i = 0; i < periods; i++) {
+ rest = fragsize;
+ /* fill descriptors for a period.
+ * a period can be split to several descriptors if it's
+ * over page boundary.
+ */
+ do {
+ unsigned int r;
+ unsigned int flag;
+ unsigned int addr;
+
+ if (idx >= VIA_TABLE_SIZE) {
+ snd_printk(KERN_ERR "via82xx: too much table size!\n");
+ return -EINVAL;
+ }
+ addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+ ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+ r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest);
+ rest -= r;
+ if (! rest) {
+ if (i == periods - 1)
+ flag = VIA_TBL_BIT_EOL; /* buffer boundary */
+ else
+ flag = VIA_TBL_BIT_FLAG; /* period boundary */
+ } else
+ flag = 0; /* period continues to the next */
+ // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest);
+ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+ dev->idx_table[idx].offset = ofs;
+ dev->idx_table[idx].size = r;
+ ofs += r;
+ idx++;
+ } while (rest > 0);
+ }
+ dev->tbl_entries = idx;
+ dev->bufsize = periods * fragsize;
+ dev->bufsize2 = dev->bufsize / 2;
+ dev->fragsize = fragsize;
+ return 0;
+}
+
+
+static int clean_via_table(struct viadev *dev, struct snd_pcm_substream *substream,
+ struct pci_dev *pci)
+{
+ if (dev->table.area) {
+ snd_dma_free_pages(&dev->table);
+ dev->table.area = NULL;
+ }
+ kfree(dev->idx_table);
+ dev->idx_table = NULL;
+ return 0;
+}
+
+/*
+ * Basic I/O
+ */
+
+static inline unsigned int snd_via82xx_codec_xread(struct via82xx *chip)
+{
+ return inl(VIAREG(chip, AC97));
+}
+
+static inline void snd_via82xx_codec_xwrite(struct via82xx *chip, unsigned int val)
+{
+ outl(val, VIAREG(chip, AC97));
+}
+
+static int snd_via82xx_codec_ready(struct via82xx *chip, int secondary)
+{
+ unsigned int timeout = 1000; /* 1ms */
+ unsigned int val;
+
+ while (timeout-- > 0) {
+ udelay(1);
+ if (!((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY))
+ return val & 0xffff;
+ }
+ snd_printk(KERN_ERR "codec_ready: codec %i is not ready [0x%x]\n",
+ secondary, snd_via82xx_codec_xread(chip));
+ return -EIO;
+}
+
+static int snd_via82xx_codec_valid(struct via82xx *chip, int secondary)
+{
+ unsigned int timeout = 1000; /* 1ms */
+ unsigned int val, val1;
+ unsigned int stat = !secondary ? VIA_REG_AC97_PRIMARY_VALID :
+ VIA_REG_AC97_SECONDARY_VALID;
+
+ while (timeout-- > 0) {
+ val = snd_via82xx_codec_xread(chip);
+ val1 = val & (VIA_REG_AC97_BUSY | stat);
+ if (val1 == stat)
+ return val & 0xffff;
+ udelay(1);
+ }
+ return -EIO;
+}
+
+static void snd_via82xx_codec_wait(struct snd_ac97 *ac97)
+{
+ struct via82xx *chip = ac97->private_data;
+ int err;
+ err = snd_via82xx_codec_ready(chip, ac97->num);
+ /* here we need to wait fairly for long time.. */
+ msleep(500);
+}
+
+static void snd_via82xx_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct via82xx *chip = ac97->private_data;
+ unsigned int xval;
+
+ xval = !ac97->num ? VIA_REG_AC97_CODEC_ID_PRIMARY : VIA_REG_AC97_CODEC_ID_SECONDARY;
+ xval <<= VIA_REG_AC97_CODEC_ID_SHIFT;
+ xval |= reg << VIA_REG_AC97_CMD_SHIFT;
+ xval |= val << VIA_REG_AC97_DATA_SHIFT;
+ snd_via82xx_codec_xwrite(chip, xval);
+ snd_via82xx_codec_ready(chip, ac97->num);
+}
+
+static unsigned short snd_via82xx_codec_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct via82xx *chip = ac97->private_data;
+ unsigned int xval, val = 0xffff;
+ int again = 0;
+
+ xval = ac97->num << VIA_REG_AC97_CODEC_ID_SHIFT;
+ xval |= ac97->num ? VIA_REG_AC97_SECONDARY_VALID : VIA_REG_AC97_PRIMARY_VALID;
+ xval |= VIA_REG_AC97_READ;
+ xval |= (reg & 0x7f) << VIA_REG_AC97_CMD_SHIFT;
+ while (1) {
+ if (again++ > 3) {
+ snd_printk(KERN_ERR "codec_read: codec %i is not valid [0x%x]\n",
+ ac97->num, snd_via82xx_codec_xread(chip));
+ return 0xffff;
+ }
+ snd_via82xx_codec_xwrite(chip, xval);
+ udelay (20);
+ if (snd_via82xx_codec_valid(chip, ac97->num) >= 0) {
+ udelay(25);
+ val = snd_via82xx_codec_xread(chip);
+ break;
+ }
+ }
+ return val & 0xffff;
+}
+
+static void snd_via82xx_channel_reset(struct via82xx *chip, struct viadev *viadev)
+{
+ outb(VIA_REG_CTRL_PAUSE | VIA_REG_CTRL_TERMINATE | VIA_REG_CTRL_RESET,
+ VIADEV_REG(viadev, OFFSET_CONTROL));
+ inb(VIADEV_REG(viadev, OFFSET_CONTROL));
+ udelay(50);
+ /* disable interrupts */
+ outb(0x00, VIADEV_REG(viadev, OFFSET_CONTROL));
+ /* clear interrupts */
+ outb(0x03, VIADEV_REG(viadev, OFFSET_STATUS));
+ outb(0x00, VIADEV_REG(viadev, OFFSET_TYPE)); /* for via686 */
+ // outl(0, VIADEV_REG(viadev, OFFSET_CURR_PTR));
+ viadev->lastpos = 0;
+ viadev->hwptr_done = 0;
+}
+
+
+/*
+ * Interrupt handler
+ * Used for 686 and 8233A
+ */
+static irqreturn_t snd_via686_interrupt(int irq, void *dev_id)
+{
+ struct via82xx *chip = dev_id;
+ unsigned int status;
+ unsigned int i;
+
+ status = inl(VIAREG(chip, SGD_SHADOW));
+ if (! (status & chip->intr_mask)) {
+ if (chip->rmidi)
+ /* check mpu401 interrupt */
+ return snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data);
+ return IRQ_NONE;
+ }
+
+ /* check status for each stream */
+ spin_lock(&chip->reg_lock);
+ for (i = 0; i < chip->num_devs; i++) {
+ struct viadev *viadev = &chip->devs[i];
+ unsigned char c_status = inb(VIADEV_REG(viadev, OFFSET_STATUS));
+ if (! (c_status & (VIA_REG_STAT_EOL|VIA_REG_STAT_FLAG|VIA_REG_STAT_STOPPED)))
+ continue;
+ if (viadev->substream && viadev->running) {
+ /*
+ * Update hwptr_done based on 'period elapsed'
+ * interrupts. We'll use it, when the chip returns 0
+ * for OFFSET_CURR_COUNT.
+ */
+ if (c_status & VIA_REG_STAT_EOL)
+ viadev->hwptr_done = 0;
+ else
+ viadev->hwptr_done += viadev->fragsize;
+ viadev->in_interrupt = c_status;
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(viadev->substream);
+ spin_lock(&chip->reg_lock);
+ viadev->in_interrupt = 0;
+ }
+ outb(c_status, VIADEV_REG(viadev, OFFSET_STATUS)); /* ack */
+ }
+ spin_unlock(&chip->reg_lock);
+ return IRQ_HANDLED;
+}
+
+/*
+ * Interrupt handler
+ */
+static irqreturn_t snd_via8233_interrupt(int irq, void *dev_id)
+{
+ struct via82xx *chip = dev_id;
+ unsigned int status;
+ unsigned int i;
+ int irqreturn = 0;
+
+ /* check status for each stream */
+ spin_lock(&chip->reg_lock);
+ status = inl(VIAREG(chip, SGD_SHADOW));
+
+ for (i = 0; i < chip->num_devs; i++) {
+ struct viadev *viadev = &chip->devs[i];
+ struct snd_pcm_substream *substream;
+ unsigned char c_status, shadow_status;
+
+ shadow_status = (status >> viadev->shadow_shift) &
+ (VIA8233_SHADOW_STAT_ACTIVE|VIA_REG_STAT_EOL|
+ VIA_REG_STAT_FLAG);
+ c_status = shadow_status & (VIA_REG_STAT_EOL|VIA_REG_STAT_FLAG);
+ if (!c_status)
+ continue;
+
+ substream = viadev->substream;
+ if (substream && viadev->running) {
+ /*
+ * Update hwptr_done based on 'period elapsed'
+ * interrupts. We'll use it, when the chip returns 0
+ * for OFFSET_CURR_COUNT.
+ */
+ if (c_status & VIA_REG_STAT_EOL)
+ viadev->hwptr_done = 0;
+ else
+ viadev->hwptr_done += viadev->fragsize;
+ viadev->in_interrupt = c_status;
+ if (shadow_status & VIA8233_SHADOW_STAT_ACTIVE)
+ viadev->in_interrupt |= VIA_REG_STAT_ACTIVE;
+ spin_unlock(&chip->reg_lock);
+
+ snd_pcm_period_elapsed(substream);
+
+ spin_lock(&chip->reg_lock);
+ viadev->in_interrupt = 0;
+ }
+ outb(c_status, VIADEV_REG(viadev, OFFSET_STATUS)); /* ack */
+ irqreturn = 1;
+ }
+ spin_unlock(&chip->reg_lock);
+ return IRQ_RETVAL(irqreturn);
+}
+
+/*
+ * PCM callbacks
+ */
+
+/*
+ * trigger callback
+ */
+static int snd_via82xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ unsigned char val;
+
+ if (chip->chip_type != TYPE_VIA686)
+ val = VIA_REG_CTRL_INT;
+ else
+ val = 0;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ val |= VIA_REG_CTRL_START;
+ viadev->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ val = VIA_REG_CTRL_TERMINATE;
+ viadev->running = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ val |= VIA_REG_CTRL_PAUSE;
+ viadev->running = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ viadev->running = 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ outb(val, VIADEV_REG(viadev, OFFSET_CONTROL));
+ if (cmd == SNDRV_PCM_TRIGGER_STOP)
+ snd_via82xx_channel_reset(chip, viadev);
+ return 0;
+}
+
+
+/*
+ * pointer callbacks
+ */
+
+/*
+ * calculate the linear position at the given sg-buffer index and the rest count
+ */
+
+#define check_invalid_pos(viadev,pos) \
+ ((pos) < viadev->lastpos && ((pos) >= viadev->bufsize2 ||\
+ viadev->lastpos < viadev->bufsize2))
+
+static inline unsigned int calc_linear_pos(struct viadev *viadev, unsigned int idx,
+ unsigned int count)
+{
+ unsigned int size, base, res;
+
+ size = viadev->idx_table[idx].size;
+ base = viadev->idx_table[idx].offset;
+ res = base + size - count;
+ if (res >= viadev->bufsize)
+ res -= viadev->bufsize;
+
+ /* check the validity of the calculated position */
+ if (size < count) {
+ snd_printd(KERN_ERR "invalid via82xx_cur_ptr (size = %d, count = %d)\n",
+ (int)size, (int)count);
+ res = viadev->lastpos;
+ } else {
+ if (! count) {
+ /* Some mobos report count = 0 on the DMA boundary,
+ * i.e. count = size indeed.
+ * Let's check whether this step is above the expected size.
+ */
+ int delta = res - viadev->lastpos;
+ if (delta < 0)
+ delta += viadev->bufsize;
+ if ((unsigned int)delta > viadev->fragsize)
+ res = base;
+ }
+ if (check_invalid_pos(viadev, res)) {
+#ifdef POINTER_DEBUG
+ printk(KERN_DEBUG "fail: idx = %i/%i, lastpos = 0x%x, "
+ "bufsize2 = 0x%x, offsize = 0x%x, size = 0x%x, "
+ "count = 0x%x\n", idx, viadev->tbl_entries,
+ viadev->lastpos, viadev->bufsize2,
+ viadev->idx_table[idx].offset,
+ viadev->idx_table[idx].size, count);
+#endif
+ /* count register returns full size when end of buffer is reached */
+ res = base + size;
+ if (check_invalid_pos(viadev, res)) {
+ snd_printd(KERN_ERR "invalid via82xx_cur_ptr (2), "
+ "using last valid pointer\n");
+ res = viadev->lastpos;
+ }
+ }
+ }
+ return res;
+}
+
+/*
+ * get the current pointer on via686
+ */
+static snd_pcm_uframes_t snd_via686_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ unsigned int idx, ptr, count, res;
+
+ if (snd_BUG_ON(!viadev->tbl_entries))
+ return 0;
+ if (!(inb(VIADEV_REG(viadev, OFFSET_STATUS)) & VIA_REG_STAT_ACTIVE))
+ return 0;
+
+ spin_lock(&chip->reg_lock);
+ count = inl(VIADEV_REG(viadev, OFFSET_CURR_COUNT)) & 0xffffff;
+ /* The via686a does not have the current index register,
+ * so we need to calculate the index from CURR_PTR.
+ */
+ ptr = inl(VIADEV_REG(viadev, OFFSET_CURR_PTR));
+ if (ptr <= (unsigned int)viadev->table.addr)
+ idx = 0;
+ else /* CURR_PTR holds the address + 8 */
+ idx = ((ptr - (unsigned int)viadev->table.addr) / 8 - 1) % viadev->tbl_entries;
+ res = calc_linear_pos(viadev, idx, count);
+ viadev->lastpos = res; /* remember the last position */
+ spin_unlock(&chip->reg_lock);
+
+ return bytes_to_frames(substream->runtime, res);
+}
+
+/*
+ * get the current pointer on via823x
+ */
+static snd_pcm_uframes_t snd_via8233_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ unsigned int idx, count, res;
+ int status;
+
+ if (snd_BUG_ON(!viadev->tbl_entries))
+ return 0;
+
+ spin_lock(&chip->reg_lock);
+ count = inl(VIADEV_REG(viadev, OFFSET_CURR_COUNT));
+ status = viadev->in_interrupt;
+ if (!status)
+ status = inb(VIADEV_REG(viadev, OFFSET_STATUS));
+
+ /* An apparent bug in the 8251 is worked around by sending a
+ * REG_CTRL_START. */
+ if (chip->revision == VIA_REV_8251 && (status & VIA_REG_STAT_EOL))
+ snd_via82xx_pcm_trigger(substream, SNDRV_PCM_TRIGGER_START);
+
+ if (!(status & VIA_REG_STAT_ACTIVE)) {
+ res = 0;
+ goto unlock;
+ }
+ if (count & 0xffffff) {
+ idx = count >> 24;
+ if (idx >= viadev->tbl_entries) {
+#ifdef POINTER_DEBUG
+ printk(KERN_DEBUG "fail: invalid idx = %i/%i\n", idx,
+ viadev->tbl_entries);
+#endif
+ res = viadev->lastpos;
+ } else {
+ count &= 0xffffff;
+ res = calc_linear_pos(viadev, idx, count);
+ }
+ } else {
+ res = viadev->hwptr_done;
+ if (!viadev->in_interrupt) {
+ if (status & VIA_REG_STAT_EOL) {
+ res = 0;
+ } else
+ if (status & VIA_REG_STAT_FLAG) {
+ res += viadev->fragsize;
+ }
+ }
+ }
+unlock:
+ viadev->lastpos = res;
+ spin_unlock(&chip->reg_lock);
+
+ return bytes_to_frames(substream->runtime, res);
+}
+
+
+/*
+ * hw_params callback:
+ * allocate the buffer and build up the buffer description table
+ */
+static int snd_via82xx_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ err = build_via_table(viadev, substream, chip->pci,
+ params_periods(hw_params),
+ params_period_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+/*
+ * hw_free callback:
+ * clean up the buffer description table and release the buffer
+ */
+static int snd_via82xx_hw_free(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+
+ clean_via_table(viadev, substream, chip->pci);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+
+/*
+ * set up the table pointer
+ */
+static void snd_via82xx_set_table_ptr(struct via82xx *chip, struct viadev *viadev)
+{
+ snd_via82xx_codec_ready(chip, 0);
+ outl((u32)viadev->table.addr, VIADEV_REG(viadev, OFFSET_TABLE_PTR));
+ udelay(20);
+ snd_via82xx_codec_ready(chip, 0);
+}
+
+/*
+ * prepare callback for playback and capture on via686
+ */
+static void via686_setup_format(struct via82xx *chip, struct viadev *viadev,
+ struct snd_pcm_runtime *runtime)
+{
+ snd_via82xx_channel_reset(chip, viadev);
+ /* this must be set after channel_reset */
+ snd_via82xx_set_table_ptr(chip, viadev);
+ outb(VIA_REG_TYPE_AUTOSTART |
+ (runtime->format == SNDRV_PCM_FORMAT_S16_LE ? VIA_REG_TYPE_16BIT : 0) |
+ (runtime->channels > 1 ? VIA_REG_TYPE_STEREO : 0) |
+ ((viadev->reg_offset & 0x10) == 0 ? VIA_REG_TYPE_INT_LSAMPLE : 0) |
+ VIA_REG_TYPE_INT_EOL |
+ VIA_REG_TYPE_INT_FLAG, VIADEV_REG(viadev, OFFSET_TYPE));
+}
+
+static int snd_via686_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate);
+ snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate);
+ via686_setup_format(chip, viadev, runtime);
+ return 0;
+}
+
+static int snd_via686_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ snd_ac97_set_rate(chip->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate);
+ via686_setup_format(chip, viadev, runtime);
+ return 0;
+}
+
+/*
+ * lock the current rate
+ */
+static int via_lock_rate(struct via_rate_lock *rec, int rate)
+{
+ int changed = 0;
+
+ spin_lock_irq(&rec->lock);
+ if (rec->rate != rate) {
+ if (rec->rate && rec->used > 1) /* already set */
+ changed = -EINVAL;
+ else {
+ rec->rate = rate;
+ changed = 1;
+ }
+ }
+ spin_unlock_irq(&rec->lock);
+ return changed;
+}
+
+/*
+ * prepare callback for DSX playback on via823x
+ */
+static int snd_via8233_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ac97_rate = chip->dxs_src ? 48000 : runtime->rate;
+ int rate_changed;
+ u32 rbits;
+
+ if ((rate_changed = via_lock_rate(&chip->rates[0], ac97_rate)) < 0)
+ return rate_changed;
+ if (rate_changed)
+ snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE,
+ chip->no_vra ? 48000 : runtime->rate);
+ if (chip->spdif_on && viadev->reg_offset == 0x30)
+ snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate);
+
+ if (runtime->rate == 48000)
+ rbits = 0xfffff;
+ else
+ rbits = (0x100000 / 48000) * runtime->rate +
+ ((0x100000 % 48000) * runtime->rate) / 48000;
+ snd_BUG_ON(rbits & ~0xfffff);
+ snd_via82xx_channel_reset(chip, viadev);
+ snd_via82xx_set_table_ptr(chip, viadev);
+ outb(chip->playback_volume[viadev->reg_offset / 0x10][0],
+ VIADEV_REG(viadev, OFS_PLAYBACK_VOLUME_L));
+ outb(chip->playback_volume[viadev->reg_offset / 0x10][1],
+ VIADEV_REG(viadev, OFS_PLAYBACK_VOLUME_R));
+ outl((runtime->format == SNDRV_PCM_FORMAT_S16_LE ? VIA8233_REG_TYPE_16BIT : 0) | /* format */
+ (runtime->channels > 1 ? VIA8233_REG_TYPE_STEREO : 0) | /* stereo */
+ rbits | /* rate */
+ 0xff000000, /* STOP index is never reached */
+ VIADEV_REG(viadev, OFFSET_STOP_IDX));
+ udelay(20);
+ snd_via82xx_codec_ready(chip, 0);
+ return 0;
+}
+
+/*
+ * prepare callback for multi-channel playback on via823x
+ */
+static int snd_via8233_multi_prepare(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int slots;
+ int fmt;
+
+ if (via_lock_rate(&chip->rates[0], runtime->rate) < 0)
+ return -EINVAL;
+ snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate);
+ snd_ac97_set_rate(chip->ac97, AC97_PCM_SURR_DAC_RATE, runtime->rate);
+ snd_ac97_set_rate(chip->ac97, AC97_PCM_LFE_DAC_RATE, runtime->rate);
+ snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate);
+ snd_via82xx_channel_reset(chip, viadev);
+ snd_via82xx_set_table_ptr(chip, viadev);
+
+ fmt = (runtime->format == SNDRV_PCM_FORMAT_S16_LE) ?
+ VIA_REG_MULTPLAY_FMT_16BIT : VIA_REG_MULTPLAY_FMT_8BIT;
+ fmt |= runtime->channels << 4;
+ outb(fmt, VIADEV_REG(viadev, OFS_MULTPLAY_FORMAT));
+#if 0
+ if (chip->revision == VIA_REV_8233A)
+ slots = 0;
+ else
+#endif
+ {
+ /* set sample number to slot 3, 4, 7, 8, 6, 9 (for VIA8233/C,8235) */
+ /* corresponding to FL, FR, RL, RR, C, LFE ?? */
+ switch (runtime->channels) {
+ case 1: slots = (1<<0) | (1<<4); break;
+ case 2: slots = (1<<0) | (2<<4); break;
+ case 3: slots = (1<<0) | (2<<4) | (5<<8); break;
+ case 4: slots = (1<<0) | (2<<4) | (3<<8) | (4<<12); break;
+ case 5: slots = (1<<0) | (2<<4) | (3<<8) | (4<<12) | (5<<16); break;
+ case 6: slots = (1<<0) | (2<<4) | (3<<8) | (4<<12) | (5<<16) | (6<<20); break;
+ default: slots = 0; break;
+ }
+ }
+ /* STOP index is never reached */
+ outl(0xff000000 | slots, VIADEV_REG(viadev, OFFSET_STOP_IDX));
+ udelay(20);
+ snd_via82xx_codec_ready(chip, 0);
+ return 0;
+}
+
+/*
+ * prepare callback for capture on via823x
+ */
+static int snd_via8233_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (via_lock_rate(&chip->rates[1], runtime->rate) < 0)
+ return -EINVAL;
+ snd_ac97_set_rate(chip->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate);
+ snd_via82xx_channel_reset(chip, viadev);
+ snd_via82xx_set_table_ptr(chip, viadev);
+ outb(VIA_REG_CAPTURE_FIFO_ENABLE, VIADEV_REG(viadev, OFS_CAPTURE_FIFO));
+ outl((runtime->format == SNDRV_PCM_FORMAT_S16_LE ? VIA8233_REG_TYPE_16BIT : 0) |
+ (runtime->channels > 1 ? VIA8233_REG_TYPE_STEREO : 0) |
+ 0xff000000, /* STOP index is never reached */
+ VIADEV_REG(viadev, OFFSET_STOP_IDX));
+ udelay(20);
+ snd_via82xx_codec_ready(chip, 0);
+ return 0;
+}
+
+
+/*
+ * pcm hardware definition, identical for both playback and capture
+ */
+static struct snd_pcm_hardware snd_via82xx_hw =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ /* SNDRV_PCM_INFO_RESUME | */
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = VIA_MAX_BUFSIZE,
+ .period_bytes_min = 32,
+ .period_bytes_max = VIA_MAX_BUFSIZE / 2,
+ .periods_min = 2,
+ .periods_max = VIA_TABLE_SIZE / 2,
+ .fifo_size = 0,
+};
+
+
+/*
+ * open callback skeleton
+ */
+static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ struct via_rate_lock *ratep;
+
+ runtime->hw = snd_via82xx_hw;
+
+ /* set the hw rate condition */
+ ratep = &chip->rates[viadev->direction];
+ spin_lock_irq(&ratep->lock);
+ ratep->used++;
+ if (chip->spdif_on && viadev->reg_offset == 0x30) {
+ /* DXS#3 and spdif is on */
+ runtime->hw.rates = chip->ac97->rates[AC97_RATES_SPDIF];
+ snd_pcm_limit_hw_rates(runtime);
+ } else if (chip->dxs_fixed && viadev->reg_offset < 0x40) {
+ /* fixed DXS playback rate */
+ runtime->hw.rates = SNDRV_PCM_RATE_48000;
+ runtime->hw.rate_min = runtime->hw.rate_max = 48000;
+ } else if (chip->dxs_src && viadev->reg_offset < 0x40) {
+ /* use full SRC capabilities of DXS */
+ runtime->hw.rates = (SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_48000);
+ runtime->hw.rate_min = 8000;
+ runtime->hw.rate_max = 48000;
+ } else if (! ratep->rate) {
+ int idx = viadev->direction ? AC97_RATES_ADC : AC97_RATES_FRONT_DAC;
+ runtime->hw.rates = chip->ac97->rates[idx];
+ snd_pcm_limit_hw_rates(runtime);
+ } else {
+ /* a fixed rate */
+ runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+ runtime->hw.rate_max = runtime->hw.rate_min = ratep->rate;
+ }
+ spin_unlock_irq(&ratep->lock);
+
+ /* we may remove following constaint when we modify table entries
+ in interrupt */
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ runtime->private_data = viadev;
+ viadev->substream = substream;
+
+ return 0;
+}
+
+
+/*
+ * open callback for playback on via686 and via823x DSX
+ */
+static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number];
+ int err;
+
+ if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * open callback for playback on via823x multi-channel
+ */
+static int snd_via8233_multi_open(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = &chip->devs[chip->multi_devno];
+ int err;
+ /* channels constraint for VIA8233A
+ * 3 and 5 channels are not supported
+ */
+ static unsigned int channels[] = {
+ 1, 2, 4, 6
+ };
+ static struct snd_pcm_hw_constraint_list hw_constraints_channels = {
+ .count = ARRAY_SIZE(channels),
+ .list = channels,
+ .mask = 0,
+ };
+
+ if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0)
+ return err;
+ substream->runtime->hw.channels_max = 6;
+ if (chip->revision == VIA_REV_8233A)
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ &hw_constraints_channels);
+ return 0;
+}
+
+/*
+ * open callback for capture on via686 and via823x
+ */
+static int snd_via82xx_capture_open(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = &chip->devs[chip->capture_devno + substream->pcm->device];
+
+ return snd_via82xx_pcm_open(chip, viadev, substream);
+}
+
+/*
+ * close callback
+ */
+static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ struct via_rate_lock *ratep;
+
+ /* release the rate lock */
+ ratep = &chip->rates[viadev->direction];
+ spin_lock_irq(&ratep->lock);
+ ratep->used--;
+ if (! ratep->used)
+ ratep->rate = 0;
+ spin_unlock_irq(&ratep->lock);
+ if (! ratep->rate) {
+ if (! viadev->direction) {
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_FRONT_DAC_RATE, 0);
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_SURR_DAC_RATE, 0);
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_LFE_DAC_RATE, 0);
+ } else
+ snd_ac97_update_power(chip->ac97,
+ AC97_PCM_LR_ADC_RATE, 0);
+ }
+ viadev->substream = NULL;
+ return 0;
+}
+
+
+/* via686 playback callbacks */
+static struct snd_pcm_ops snd_via686_playback_ops = {
+ .open = snd_via82xx_playback_open,
+ .close = snd_via82xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_via82xx_hw_params,
+ .hw_free = snd_via82xx_hw_free,
+ .prepare = snd_via686_playback_prepare,
+ .trigger = snd_via82xx_pcm_trigger,
+ .pointer = snd_via686_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/* via686 capture callbacks */
+static struct snd_pcm_ops snd_via686_capture_ops = {
+ .open = snd_via82xx_capture_open,
+ .close = snd_via82xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_via82xx_hw_params,
+ .hw_free = snd_via82xx_hw_free,
+ .prepare = snd_via686_capture_prepare,
+ .trigger = snd_via82xx_pcm_trigger,
+ .pointer = snd_via686_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/* via823x DSX playback callbacks */
+static struct snd_pcm_ops snd_via8233_playback_ops = {
+ .open = snd_via82xx_playback_open,
+ .close = snd_via82xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_via82xx_hw_params,
+ .hw_free = snd_via82xx_hw_free,
+ .prepare = snd_via8233_playback_prepare,
+ .trigger = snd_via82xx_pcm_trigger,
+ .pointer = snd_via8233_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/* via823x multi-channel playback callbacks */
+static struct snd_pcm_ops snd_via8233_multi_ops = {
+ .open = snd_via8233_multi_open,
+ .close = snd_via82xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_via82xx_hw_params,
+ .hw_free = snd_via82xx_hw_free,
+ .prepare = snd_via8233_multi_prepare,
+ .trigger = snd_via82xx_pcm_trigger,
+ .pointer = snd_via8233_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/* via823x capture callbacks */
+static struct snd_pcm_ops snd_via8233_capture_ops = {
+ .open = snd_via82xx_capture_open,
+ .close = snd_via82xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_via82xx_hw_params,
+ .hw_free = snd_via82xx_hw_free,
+ .prepare = snd_via8233_capture_prepare,
+ .trigger = snd_via82xx_pcm_trigger,
+ .pointer = snd_via8233_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+
+static void init_viadev(struct via82xx *chip, int idx, unsigned int reg_offset,
+ int shadow_pos, int direction)
+{
+ chip->devs[idx].reg_offset = reg_offset;
+ chip->devs[idx].shadow_shift = shadow_pos * 4;
+ chip->devs[idx].direction = direction;
+ chip->devs[idx].port = chip->port + reg_offset;
+}
+
+/*
+ * create pcm instances for VIA8233, 8233C and 8235 (not 8233A)
+ */
+static int __devinit snd_via8233_pcm_new(struct via82xx *chip)
+{
+ struct snd_pcm *pcm;
+ int i, err;
+
+ chip->playback_devno = 0; /* x 4 */
+ chip->multi_devno = 4; /* x 1 */
+ chip->capture_devno = 5; /* x 2 */
+ chip->num_devs = 7;
+ chip->intr_mask = 0x33033333; /* FLAG|EOL for rec0-1, mc, sdx0-3 */
+
+ /* PCM #0: 4 DSX playbacks and 1 capture */
+ err = snd_pcm_new(chip->card, chip->card->shortname, 0, 4, 1, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_via8233_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_via8233_capture_ops);
+ pcm->private_data = chip;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcms[0] = pcm;
+ /* set up playbacks */
+ for (i = 0; i < 4; i++)
+ init_viadev(chip, i, 0x10 * i, i, 0);
+ /* capture */
+ init_viadev(chip, chip->capture_devno, VIA_REG_CAPTURE_8233_STATUS, 6, 1);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, VIA_MAX_BUFSIZE);
+
+ /* PCM #1: multi-channel playback and 2nd capture */
+ err = snd_pcm_new(chip->card, chip->card->shortname, 1, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_via8233_multi_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_via8233_capture_ops);
+ pcm->private_data = chip;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcms[1] = pcm;
+ /* set up playback */
+ init_viadev(chip, chip->multi_devno, VIA_REG_MULTPLAY_STATUS, 4, 0);
+ /* set up capture */
+ init_viadev(chip, chip->capture_devno + 1, VIA_REG_CAPTURE_8233_STATUS + 0x10, 7, 1);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, VIA_MAX_BUFSIZE);
+ return 0;
+}
+
+/*
+ * create pcm instances for VIA8233A
+ */
+static int __devinit snd_via8233a_pcm_new(struct via82xx *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ chip->multi_devno = 0;
+ chip->playback_devno = 1;
+ chip->capture_devno = 2;
+ chip->num_devs = 3;
+ chip->intr_mask = 0x03033000; /* FLAG|EOL for rec0, mc, sdx3 */
+
+ /* PCM #0: multi-channel playback and capture */
+ err = snd_pcm_new(chip->card, chip->card->shortname, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_via8233_multi_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_via8233_capture_ops);
+ pcm->private_data = chip;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcms[0] = pcm;
+ /* set up playback */
+ init_viadev(chip, chip->multi_devno, VIA_REG_MULTPLAY_STATUS, 4, 0);
+ /* capture */
+ init_viadev(chip, chip->capture_devno, VIA_REG_CAPTURE_8233_STATUS, 6, 1);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, VIA_MAX_BUFSIZE);
+
+ /* SPDIF supported? */
+ if (! ac97_can_spdif(chip->ac97))
+ return 0;
+
+ /* PCM #1: DXS3 playback (for spdif) */
+ err = snd_pcm_new(chip->card, chip->card->shortname, 1, 1, 0, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_via8233_playback_ops);
+ pcm->private_data = chip;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcms[1] = pcm;
+ /* set up playback */
+ init_viadev(chip, chip->playback_devno, 0x30, 3, 0);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, VIA_MAX_BUFSIZE);
+ return 0;
+}
+
+/*
+ * create a pcm instance for via686a/b
+ */
+static int __devinit snd_via686_pcm_new(struct via82xx *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ chip->playback_devno = 0;
+ chip->capture_devno = 1;
+ chip->num_devs = 2;
+ chip->intr_mask = 0x77; /* FLAG | EOL for PB, CP, FM */
+
+ err = snd_pcm_new(chip->card, chip->card->shortname, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_via686_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_via686_capture_ops);
+ pcm->private_data = chip;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcms[0] = pcm;
+ init_viadev(chip, 0, VIA_REG_PLAYBACK_STATUS, 0, 0);
+ init_viadev(chip, 1, VIA_REG_CAPTURE_STATUS, 0, 1);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, VIA_MAX_BUFSIZE);
+ return 0;
+}
+
+
+/*
+ * Mixer part
+ */
+
+static int snd_via8233_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ /* formerly they were "Line" and "Mic", but it looks like that they
+ * have nothing to do with the actual physical connections...
+ */
+ static char *texts[2] = {
+ "Input1", "Input2"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item >= 2)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_via8233_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long port = chip->port + (kcontrol->id.index ? (VIA_REG_CAPTURE_CHANNEL + 0x10) : VIA_REG_CAPTURE_CHANNEL);
+ ucontrol->value.enumerated.item[0] = inb(port) & VIA_REG_CAPTURE_CHANNEL_MIC ? 1 : 0;
+ return 0;
+}
+
+static int snd_via8233_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long port = chip->port + (kcontrol->id.index ? (VIA_REG_CAPTURE_CHANNEL + 0x10) : VIA_REG_CAPTURE_CHANNEL);
+ u8 val, oval;
+
+ spin_lock_irq(&chip->reg_lock);
+ oval = inb(port);
+ val = oval & ~VIA_REG_CAPTURE_CHANNEL_MIC;
+ if (ucontrol->value.enumerated.item[0])
+ val |= VIA_REG_CAPTURE_CHANNEL_MIC;
+ if (val != oval)
+ outb(val, port);
+ spin_unlock_irq(&chip->reg_lock);
+ return val != oval;
+}
+
+static struct snd_kcontrol_new snd_via8233_capture_source __devinitdata = {
+ .name = "Input Source Select",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_via8233_capture_source_info,
+ .get = snd_via8233_capture_source_get,
+ .put = snd_via8233_capture_source_put,
+};
+
+#define snd_via8233_dxs3_spdif_info snd_ctl_boolean_mono_info
+
+static int snd_via8233_dxs3_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ u8 val;
+
+ pci_read_config_byte(chip->pci, VIA8233_SPDIF_CTRL, &val);
+ ucontrol->value.integer.value[0] = (val & VIA8233_SPDIF_DX3) ? 1 : 0;
+ return 0;
+}
+
+static int snd_via8233_dxs3_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ u8 val, oval;
+
+ pci_read_config_byte(chip->pci, VIA8233_SPDIF_CTRL, &oval);
+ val = oval & ~VIA8233_SPDIF_DX3;
+ if (ucontrol->value.integer.value[0])
+ val |= VIA8233_SPDIF_DX3;
+ /* save the spdif flag for rate filtering */
+ chip->spdif_on = ucontrol->value.integer.value[0] ? 1 : 0;
+ if (val != oval) {
+ pci_write_config_byte(chip->pci, VIA8233_SPDIF_CTRL, val);
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_via8233_dxs3_spdif_control __devinitdata = {
+ .name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH),
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_via8233_dxs3_spdif_info,
+ .get = snd_via8233_dxs3_spdif_get,
+ .put = snd_via8233_dxs3_spdif_put,
+};
+
+static int snd_via8233_dxs_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = VIA_DXS_MAX_VOLUME;
+ return 0;
+}
+
+static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+
+ ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0];
+ ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1];
+ return 0;
+}
+
+static int snd_via8233_pcmdxs_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume_c[0];
+ ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume_c[1];
+ return 0;
+}
+
+static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+ unsigned long port = chip->port + 0x10 * idx;
+ unsigned char val;
+ int i, change = 0;
+
+ for (i = 0; i < 2; i++) {
+ val = ucontrol->value.integer.value[i];
+ if (val > VIA_DXS_MAX_VOLUME)
+ val = VIA_DXS_MAX_VOLUME;
+ val = VIA_DXS_MAX_VOLUME - val;
+ change |= val != chip->playback_volume[idx][i];
+ if (change) {
+ chip->playback_volume[idx][i] = val;
+ outb(val, port + VIA_REG_OFS_PLAYBACK_VOLUME_L + i);
+ }
+ }
+ return change;
+}
+
+static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct via82xx *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int idx;
+ unsigned char val;
+ int i, change = 0;
+
+ for (i = 0; i < 2; i++) {
+ val = ucontrol->value.integer.value[i];
+ if (val > VIA_DXS_MAX_VOLUME)
+ val = VIA_DXS_MAX_VOLUME;
+ val = VIA_DXS_MAX_VOLUME - val;
+ if (val != chip->playback_volume_c[i]) {
+ change = 1;
+ chip->playback_volume_c[i] = val;
+ for (idx = 0; idx < 4; idx++) {
+ unsigned long port = chip->port + 0x10 * idx;
+ chip->playback_volume[idx][i] = val;
+ outb(val, port + VIA_REG_OFS_PLAYBACK_VOLUME_L + i);
+ }
+ }
+ }
+ return change;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1);
+
+static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = {
+ .name = "PCM Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .info = snd_via8233_dxs_volume_info,
+ .get = snd_via8233_pcmdxs_volume_get,
+ .put = snd_via8233_pcmdxs_volume_put,
+ .tlv = { .p = db_scale_dxs }
+};
+
+static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
+ .name = "VIA DXS Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .count = 4,
+ .info = snd_via8233_dxs_volume_info,
+ .get = snd_via8233_dxs_volume_get,
+ .put = snd_via8233_dxs_volume_put,
+ .tlv = { .p = db_scale_dxs }
+};
+
+/*
+ */
+
+static void snd_via82xx_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct via82xx *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void snd_via82xx_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct via82xx *chip = ac97->private_data;
+ chip->ac97 = NULL;
+}
+
+static struct ac97_quirk ac97_quirks[] = {
+ {
+ .subvendor = 0x1106,
+ .subdevice = 0x4161,
+ .codec_id = 0x56494161, /* VT1612A */
+ .name = "Soltek SL-75DRV5",
+ .type = AC97_TUNE_NONE
+ },
+ { /* FIXME: which codec? */
+ .subvendor = 0x1106,
+ .subdevice = 0x4161,
+ .name = "ASRock K7VT2",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1019,
+ .subdevice = 0x0a81,
+ .name = "ECS K7VTA3",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1019,
+ .subdevice = 0x0a85,
+ .name = "ECS L7VMM2",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1019,
+ .subdevice = 0x1841,
+ .name = "ECS K7VTA3",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x1849,
+ .subdevice = 0x3059,
+ .name = "ASRock K7VM2",
+ .type = AC97_TUNE_HP_ONLY /* VT1616 */
+ },
+ {
+ .subvendor = 0x14cd,
+ .subdevice = 0x7002,
+ .name = "Unknown",
+ .type = AC97_TUNE_ALC_JACK
+ },
+ {
+ .subvendor = 0x1071,
+ .subdevice = 0x8590,
+ .name = "Mitac Mobo",
+ .type = AC97_TUNE_ALC_JACK
+ },
+ {
+ .subvendor = 0x161f,
+ .subdevice = 0x202b,
+ .name = "Arima Notebook",
+ .type = AC97_TUNE_HP_ONLY,
+ },
+ {
+ .subvendor = 0x161f,
+ .subdevice = 0x2032,
+ .name = "Targa Traveller 811",
+ .type = AC97_TUNE_HP_ONLY,
+ },
+ {
+ .subvendor = 0x161f,
+ .subdevice = 0x2032,
+ .name = "m680x",
+ .type = AC97_TUNE_HP_ONLY, /* http://launchpad.net/bugs/38546 */
+ },
+ {
+ .subvendor = 0x1297,
+ .subdevice = 0xa232,
+ .name = "Shuttle AK32VN",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ { } /* terminator */
+};
+
+static int __devinit snd_via82xx_mixer_new(struct via82xx *chip, const char *quirk_override)
+{
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_via82xx_codec_write,
+ .read = snd_via82xx_codec_read,
+ .wait = snd_via82xx_codec_wait,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &chip->ac97_bus)) < 0)
+ return err;
+ chip->ac97_bus->private_free = snd_via82xx_mixer_free_ac97_bus;
+ chip->ac97_bus->clock = chip->ac97_clock;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_via82xx_mixer_free_ac97;
+ ac97.pci = chip->pci;
+ ac97.scaps = AC97_SCAP_SKIP_MODEM | AC97_SCAP_POWER_SAVE;
+ if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97)) < 0)
+ return err;
+
+ snd_ac97_tune_hardware(chip->ac97, ac97_quirks, quirk_override);
+
+ if (chip->chip_type != TYPE_VIA686) {
+ /* use slot 10/11 */
+ snd_ac97_update_bits(chip->ac97, AC97_EXTENDED_STATUS, 0x03 << 4, 0x03 << 4);
+ }
+
+ return 0;
+}
+
+#ifdef SUPPORT_JOYSTICK
+#define JOYSTICK_ADDR 0x200
+static int __devinit snd_via686_create_gameport(struct via82xx *chip, unsigned char *legacy)
+{
+ struct gameport *gp;
+ struct resource *r;
+
+ if (!joystick)
+ return -ENODEV;
+
+ r = request_region(JOYSTICK_ADDR, 8, "VIA686 gameport");
+ if (!r) {
+ printk(KERN_WARNING "via82xx: cannot reserve joystick port 0x%#x\n",
+ JOYSTICK_ADDR);
+ return -EBUSY;
+ }
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "via82xx: cannot allocate memory for gameport\n");
+ release_and_free_resource(r);
+ return -ENOMEM;
+ }
+
+ gameport_set_name(gp, "VIA686 Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+ gp->io = JOYSTICK_ADDR;
+ gameport_set_port_data(gp, r);
+
+ /* Enable legacy joystick port */
+ *legacy |= VIA_FUNC_ENABLE_GAME;
+ pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE, *legacy);
+
+ gameport_register_port(chip->gameport);
+
+ return 0;
+}
+
+static void snd_via686_free_gameport(struct via82xx *chip)
+{
+ if (chip->gameport) {
+ struct resource *r = gameport_get_port_data(chip->gameport);
+
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+ release_and_free_resource(r);
+ }
+}
+#else
+static inline int snd_via686_create_gameport(struct via82xx *chip, unsigned char *legacy)
+{
+ return -ENOSYS;
+}
+static inline void snd_via686_free_gameport(struct via82xx *chip) { }
+#endif
+
+
+/*
+ *
+ */
+
+static int __devinit snd_via8233_init_misc(struct via82xx *chip)
+{
+ int i, err, caps;
+ unsigned char val;
+
+ caps = chip->chip_type == TYPE_VIA8233A ? 1 : 2;
+ for (i = 0; i < caps; i++) {
+ snd_via8233_capture_source.index = i;
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_capture_source, chip));
+ if (err < 0)
+ return err;
+ }
+ if (ac97_can_spdif(chip->ac97)) {
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs3_spdif_control, chip));
+ if (err < 0)
+ return err;
+ }
+ if (chip->chip_type != TYPE_VIA8233A) {
+ /* when no h/w PCM volume control is found, use DXS volume control
+ * as the PCM vol control
+ */
+ struct snd_ctl_elem_id sid;
+ memset(&sid, 0, sizeof(sid));
+ strcpy(sid.name, "PCM Playback Volume");
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ if (! snd_ctl_find_id(chip->card, &sid)) {
+ snd_printd(KERN_INFO "Using DXS as PCM Playback\n");
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_pcmdxs_volume_control, chip));
+ if (err < 0)
+ return err;
+ }
+ else /* Using DXS when PCM emulation is enabled is really weird */
+ {
+ /* Standalone DXS controls */
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip));
+ if (err < 0)
+ return err;
+ }
+ }
+ /* select spdif data slot 10/11 */
+ pci_read_config_byte(chip->pci, VIA8233_SPDIF_CTRL, &val);
+ val = (val & ~VIA8233_SPDIF_SLOT_MASK) | VIA8233_SPDIF_SLOT_1011;
+ val &= ~VIA8233_SPDIF_DX3; /* SPDIF off as default */
+ pci_write_config_byte(chip->pci, VIA8233_SPDIF_CTRL, val);
+
+ return 0;
+}
+
+static int __devinit snd_via686_init_misc(struct via82xx *chip)
+{
+ unsigned char legacy, legacy_cfg;
+ int rev_h = 0;
+
+ legacy = chip->old_legacy;
+ legacy_cfg = chip->old_legacy_cfg;
+ legacy |= VIA_FUNC_MIDI_IRQMASK; /* FIXME: correct? (disable MIDI) */
+ legacy &= ~VIA_FUNC_ENABLE_GAME; /* disable joystick */
+ if (chip->revision >= VIA_REV_686_H) {
+ rev_h = 1;
+ if (mpu_port >= 0x200) { /* force MIDI */
+ mpu_port &= 0xfffc;
+ pci_write_config_dword(chip->pci, 0x18, mpu_port | 0x01);
+#ifdef CONFIG_PM
+ chip->mpu_port_saved = mpu_port;
+#endif
+ } else {
+ mpu_port = pci_resource_start(chip->pci, 2);
+ }
+ } else {
+ switch (mpu_port) { /* force MIDI */
+ case 0x300:
+ case 0x310:
+ case 0x320:
+ case 0x330:
+ legacy_cfg &= ~(3 << 2);
+ legacy_cfg |= (mpu_port & 0x0030) >> 2;
+ break;
+ default: /* no, use BIOS settings */
+ if (legacy & VIA_FUNC_ENABLE_MIDI)
+ mpu_port = 0x300 + ((legacy_cfg & 0x000c) << 2);
+ break;
+ }
+ }
+ if (mpu_port >= 0x200 &&
+ (chip->mpu_res = request_region(mpu_port, 2, "VIA82xx MPU401"))
+ != NULL) {
+ if (rev_h)
+ legacy |= VIA_FUNC_MIDI_PNP; /* enable PCI I/O 2 */
+ legacy |= VIA_FUNC_ENABLE_MIDI;
+ } else {
+ if (rev_h)
+ legacy &= ~VIA_FUNC_MIDI_PNP; /* disable PCI I/O 2 */
+ legacy &= ~VIA_FUNC_ENABLE_MIDI;
+ mpu_port = 0;
+ }
+
+ pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE, legacy);
+ pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg);
+ if (chip->mpu_res) {
+ if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A,
+ mpu_port, MPU401_INFO_INTEGRATED,
+ chip->irq, 0, &chip->rmidi) < 0) {
+ printk(KERN_WARNING "unable to initialize MPU-401"
+ " at 0x%lx, skipping\n", mpu_port);
+ legacy &= ~VIA_FUNC_ENABLE_MIDI;
+ } else {
+ legacy &= ~VIA_FUNC_MIDI_IRQMASK; /* enable MIDI interrupt */
+ }
+ pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE, legacy);
+ }
+
+ snd_via686_create_gameport(chip, &legacy);
+
+#ifdef CONFIG_PM
+ chip->legacy_saved = legacy;
+ chip->legacy_cfg_saved = legacy_cfg;
+#endif
+
+ return 0;
+}
+
+
+/*
+ * proc interface
+ */
+static void snd_via82xx_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct via82xx *chip = entry->private_data;
+ int i;
+
+ snd_iprintf(buffer, "%s\n\n", chip->card->longname);
+ for (i = 0; i < 0xa0; i += 4) {
+ snd_iprintf(buffer, "%02x: %08x\n", i, inl(chip->port + i));
+ }
+}
+
+static void __devinit snd_via82xx_proc_init(struct via82xx *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "via82xx", &entry))
+ snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read);
+}
+
+/*
+ *
+ */
+
+static int snd_via82xx_chip_init(struct via82xx *chip)
+{
+ unsigned int val;
+ unsigned long end_time;
+ unsigned char pval;
+
+#if 0 /* broken on K7M? */
+ if (chip->chip_type == TYPE_VIA686)
+ /* disable all legacy ports */
+ pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE, 0);
+#endif
+ pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
+ if (! (pval & VIA_ACLINK_C00_READY)) { /* codec not ready? */
+ /* deassert ACLink reset, force SYNC */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL,
+ VIA_ACLINK_CTRL_ENABLE |
+ VIA_ACLINK_CTRL_RESET |
+ VIA_ACLINK_CTRL_SYNC);
+ udelay(100);
+#if 1 /* FIXME: should we do full reset here for all chip models? */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, 0x00);
+ udelay(100);
+#else
+ /* deassert ACLink reset, force SYNC (warm AC'97 reset) */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL,
+ VIA_ACLINK_CTRL_RESET|VIA_ACLINK_CTRL_SYNC);
+ udelay(2);
+#endif
+ /* ACLink on, deassert ACLink reset, VSR, SGD data out */
+ /* note - FM data out has trouble with non VRA codecs !! */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_INIT);
+ udelay(100);
+ }
+
+ /* Make sure VRA is enabled, in case we didn't do a
+ * complete codec reset, above */
+ pci_read_config_byte(chip->pci, VIA_ACLINK_CTRL, &pval);
+ if ((pval & VIA_ACLINK_CTRL_INIT) != VIA_ACLINK_CTRL_INIT) {
+ /* ACLink on, deassert ACLink reset, VSR, SGD data out */
+ /* note - FM data out has trouble with non VRA codecs !! */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_INIT);
+ udelay(100);
+ }
+
+ /* wait until codec ready */
+ end_time = jiffies + msecs_to_jiffies(750);
+ do {
+ pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
+ if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+
+ if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)
+ snd_printk(KERN_ERR "AC'97 codec is not ready [0x%x]\n", val);
+
+#if 0 /* FIXME: we don't support the second codec yet so skip the detection now.. */
+ snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
+ VIA_REG_AC97_SECONDARY_VALID |
+ (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
+ end_time = jiffies + msecs_to_jiffies(750);
+ snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
+ VIA_REG_AC97_SECONDARY_VALID |
+ (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
+ do {
+ if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_SECONDARY_VALID) {
+ chip->ac97_secondary = 1;
+ goto __ac97_ok2;
+ }
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+ /* This is ok, the most of motherboards have only one codec */
+
+ __ac97_ok2:
+#endif
+
+ if (chip->chip_type == TYPE_VIA686) {
+ /* route FM trap to IRQ, disable FM trap */
+ pci_write_config_byte(chip->pci, VIA_FM_NMI_CTRL, 0);
+ /* disable all GPI interrupts */
+ outl(0, VIAREG(chip, GPI_INTR));
+ }
+
+ if (chip->chip_type != TYPE_VIA686) {
+ /* Workaround for Award BIOS bug:
+ * DXS channels don't work properly with VRA if MC97 is disabled.
+ */
+ struct pci_dev *pci;
+ pci = pci_get_device(0x1106, 0x3068, NULL); /* MC97 */
+ if (pci) {
+ unsigned char data;
+ pci_read_config_byte(pci, 0x44, &data);
+ pci_write_config_byte(pci, 0x44, data | 0x40);
+ pci_dev_put(pci);
+ }
+ }
+
+ if (chip->chip_type != TYPE_VIA8233A) {
+ int i, idx;
+ for (idx = 0; idx < 4; idx++) {
+ unsigned long port = chip->port + 0x10 * idx;
+ for (i = 0; i < 2; i++) {
+ chip->playback_volume[idx][i]=chip->playback_volume_c[i];
+ outb(chip->playback_volume_c[i],
+ port + VIA_REG_OFS_PLAYBACK_VOLUME_L + i);
+ }
+ }
+ }
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct via82xx *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < 2; i++)
+ snd_pcm_suspend_all(chip->pcms[i]);
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+ synchronize_irq(chip->irq);
+ snd_ac97_suspend(chip->ac97);
+
+ /* save misc values */
+ if (chip->chip_type != TYPE_VIA686) {
+ pci_read_config_byte(chip->pci, VIA8233_SPDIF_CTRL, &chip->spdif_ctrl_saved);
+ chip->capture_src_saved[0] = inb(chip->port + VIA_REG_CAPTURE_CHANNEL);
+ chip->capture_src_saved[1] = inb(chip->port + VIA_REG_CAPTURE_CHANNEL + 0x10);
+ }
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_via82xx_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct via82xx *chip = card->private_data;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "via82xx: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_via82xx_chip_init(chip);
+
+ if (chip->chip_type == TYPE_VIA686) {
+ if (chip->mpu_port_saved)
+ pci_write_config_dword(chip->pci, 0x18, chip->mpu_port_saved | 0x01);
+ pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE, chip->legacy_saved);
+ pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, chip->legacy_cfg_saved);
+ } else {
+ pci_write_config_byte(chip->pci, VIA8233_SPDIF_CTRL, chip->spdif_ctrl_saved);
+ outb(chip->capture_src_saved[0], chip->port + VIA_REG_CAPTURE_CHANNEL);
+ outb(chip->capture_src_saved[1], chip->port + VIA_REG_CAPTURE_CHANNEL + 0x10);
+ }
+
+ snd_ac97_resume(chip->ac97);
+
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static int snd_via82xx_free(struct via82xx *chip)
+{
+ unsigned int i;
+
+ if (chip->irq < 0)
+ goto __end_hw;
+ /* disable interrupts */
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ __end_hw:
+ release_and_free_resource(chip->mpu_res);
+ pci_release_regions(chip->pci);
+
+ if (chip->chip_type == TYPE_VIA686) {
+ snd_via686_free_gameport(chip);
+ pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE, chip->old_legacy);
+ pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, chip->old_legacy_cfg);
+ }
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_via82xx_dev_free(struct snd_device *device)
+{
+ struct via82xx *chip = device->device_data;
+ return snd_via82xx_free(chip);
+}
+
+static int __devinit snd_via82xx_create(struct snd_card *card,
+ struct pci_dev *pci,
+ int chip_type,
+ int revision,
+ unsigned int ac97_clock,
+ struct via82xx ** r_via)
+{
+ struct via82xx *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_via82xx_dev_free,
+ };
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ if ((chip = kzalloc(sizeof(*chip), GFP_KERNEL)) == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->chip_type = chip_type;
+ chip->revision = revision;
+
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->rates[0].lock);
+ spin_lock_init(&chip->rates[1].lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ pci_read_config_byte(pci, VIA_FUNC_ENABLE, &chip->old_legacy);
+ pci_read_config_byte(pci, VIA_PNP_CONTROL, &chip->old_legacy_cfg);
+ pci_write_config_byte(chip->pci, VIA_FUNC_ENABLE,
+ chip->old_legacy & ~(VIA_FUNC_ENABLE_SB|VIA_FUNC_ENABLE_FM));
+
+ if ((err = pci_request_regions(pci, card->driver)) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->port = pci_resource_start(pci, 0);
+ if (request_irq(pci->irq,
+ chip_type == TYPE_VIA8233 ?
+ snd_via8233_interrupt : snd_via686_interrupt,
+ IRQF_SHARED,
+ card->driver, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_via82xx_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ if (ac97_clock >= 8000 && ac97_clock <= 48000)
+ chip->ac97_clock = ac97_clock;
+ synchronize_irq(chip->irq);
+
+ if ((err = snd_via82xx_chip_init(chip)) < 0) {
+ snd_via82xx_free(chip);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_via82xx_free(chip);
+ return err;
+ }
+
+ /* The 8233 ac97 controller does not implement the master bit
+ * in the pci command register. IMHO this is a violation of the PCI spec.
+ * We call pci_set_master here because it does not hurt. */
+ pci_set_master(pci);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_via = chip;
+ return 0;
+}
+
+struct via823x_info {
+ int revision;
+ char *name;
+ int type;
+};
+static struct via823x_info via823x_cards[] __devinitdata = {
+ { VIA_REV_PRE_8233, "VIA 8233-Pre", TYPE_VIA8233 },
+ { VIA_REV_8233C, "VIA 8233C", TYPE_VIA8233 },
+ { VIA_REV_8233, "VIA 8233", TYPE_VIA8233 },
+ { VIA_REV_8233A, "VIA 8233A", TYPE_VIA8233A },
+ { VIA_REV_8235, "VIA 8235", TYPE_VIA8233 },
+ { VIA_REV_8237, "VIA 8237", TYPE_VIA8233 },
+ { VIA_REV_8251, "VIA 8251", TYPE_VIA8233 },
+};
+
+/*
+ * auto detection of DXS channel supports.
+ */
+
+static struct snd_pci_quirk dxs_whitelist[] __devinitdata = {
+ SND_PCI_QUIRK(0x1005, 0x4710, "Avance Logic Mobo", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K),
+ SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x1019, 0x0a85, "ECS L7VMM2", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x1019, 0, "ESC K8", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1019, 0xaa01, "ESC K8T890-A", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1025, 0x0033, "Acer Inspire 1353LM", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x1025, 0x0046, "Acer Aspire 1524 WLMi", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1043, 0, "ASUS A7/A8", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x1071, 0, "Diverse Notebook", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x10cf, 0x118e, "FSC Laptop", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1106, 0, "ASRock", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1297, 0xa231, "Shuttle AK31v2", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte GA-7VAXP", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1462, 0x3800, "MSI KT266", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1462, 0x7120, "MSI KT4V", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1462, 0x7142, "MSI K8MM-V", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1462, 0, "MSI Mobo", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x147b, 0x1401, "ABIT KD7(-RAID)", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x147b, 0x1411, "ABIT VA-20", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x147b, 0x1413, "ABIT KV8 Pro", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x147b, 0x1415, "ABIT AV8", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x14ff, 0x0403, "Twinhead mobo", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x14ff, 0x0408, "Twinhead laptop", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1558, 0x4701, "Clevo D470", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1584, 0x8120, "Diverse Laptop", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1584, 0x8123, "Targa/Uniwill", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x161f, 0x202b, "Amira Notebook", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x161f, 0x2032, "m680x machines", VIA_DXS_48K),
+ SND_PCI_QUIRK(0x1631, 0xe004, "PB EasyNote 3174", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1695, 0x3005, "EPoX EP-8K9A", VIA_DXS_ENABLE),
+ SND_PCI_QUIRK(0x1695, 0, "EPoX mobo", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x16f3, 0, "Jetway K8", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1734, 0, "FSC Laptop", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1849, 0x3059, "ASRock K7VM2", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x1849, 0, "ASRock mobo", VIA_DXS_SRC),
+ SND_PCI_QUIRK(0x1919, 0x200a, "Soltek SL-K8", VIA_DXS_NO_VRA),
+ SND_PCI_QUIRK(0x4005, 0x4710, "MSI K7T266", VIA_DXS_SRC),
+ { } /* terminator */
+};
+
+static int __devinit check_dxs_list(struct pci_dev *pci, int revision)
+{
+ const struct snd_pci_quirk *w;
+
+ w = snd_pci_quirk_lookup(pci, dxs_whitelist);
+ if (w) {
+ snd_printdd(KERN_INFO "via82xx: DXS white list for %s found\n",
+ w->name);
+ return w->value;
+ }
+
+ /* for newer revision, default to DXS_SRC */
+ if (revision >= VIA_REV_8235)
+ return VIA_DXS_SRC;
+
+ /*
+ * not detected, try 48k rate only to be sure.
+ */
+ printk(KERN_INFO "via82xx: Assuming DXS channels with 48k fixed sample rate.\n");
+ printk(KERN_INFO " Please try dxs_support=5 option\n");
+ printk(KERN_INFO " and report if it works on your machine.\n");
+ printk(KERN_INFO " For more details, read ALSA-Configuration.txt.\n");
+ return VIA_DXS_48K;
+};
+
+static int __devinit snd_via82xx_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct via82xx *chip;
+ int chip_type = 0, card_type;
+ unsigned int i;
+ int err;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ card_type = pci_id->driver_data;
+ switch (card_type) {
+ case TYPE_CARD_VIA686:
+ strcpy(card->driver, "VIA686A");
+ sprintf(card->shortname, "VIA 82C686A/B rev%x", pci->revision);
+ chip_type = TYPE_VIA686;
+ break;
+ case TYPE_CARD_VIA8233:
+ chip_type = TYPE_VIA8233;
+ sprintf(card->shortname, "VIA 823x rev%x", pci->revision);
+ for (i = 0; i < ARRAY_SIZE(via823x_cards); i++) {
+ if (pci->revision == via823x_cards[i].revision) {
+ chip_type = via823x_cards[i].type;
+ strcpy(card->shortname, via823x_cards[i].name);
+ break;
+ }
+ }
+ if (chip_type != TYPE_VIA8233A) {
+ if (dxs_support == VIA_DXS_AUTO)
+ dxs_support = check_dxs_list(pci, pci->revision);
+ /* force to use VIA8233 or 8233A model according to
+ * dxs_support module option
+ */
+ if (dxs_support == VIA_DXS_DISABLE)
+ chip_type = TYPE_VIA8233A;
+ else
+ chip_type = TYPE_VIA8233;
+ }
+ if (chip_type == TYPE_VIA8233A)
+ strcpy(card->driver, "VIA8233A");
+ else if (pci->revision >= VIA_REV_8237)
+ strcpy(card->driver, "VIA8237"); /* no slog assignment */
+ else
+ strcpy(card->driver, "VIA8233");
+ break;
+ default:
+ snd_printk(KERN_ERR "invalid card type %d\n", card_type);
+ err = -EINVAL;
+ goto __error;
+ }
+
+ if ((err = snd_via82xx_create(card, pci, chip_type, pci->revision,
+ ac97_clock, &chip)) < 0)
+ goto __error;
+ card->private_data = chip;
+ if ((err = snd_via82xx_mixer_new(chip, ac97_quirk)) < 0)
+ goto __error;
+
+ if (chip_type == TYPE_VIA686) {
+ if ((err = snd_via686_pcm_new(chip)) < 0 ||
+ (err = snd_via686_init_misc(chip)) < 0)
+ goto __error;
+ } else {
+ if (chip_type == TYPE_VIA8233A) {
+ if ((err = snd_via8233a_pcm_new(chip)) < 0)
+ goto __error;
+ // chip->dxs_fixed = 1; /* FIXME: use 48k for DXS #3? */
+ } else {
+ if ((err = snd_via8233_pcm_new(chip)) < 0)
+ goto __error;
+ if (dxs_support == VIA_DXS_48K)
+ chip->dxs_fixed = 1;
+ else if (dxs_support == VIA_DXS_NO_VRA)
+ chip->no_vra = 1;
+ else if (dxs_support == VIA_DXS_SRC) {
+ chip->no_vra = 1;
+ chip->dxs_src = 1;
+ }
+ }
+ if ((err = snd_via8233_init_misc(chip)) < 0)
+ goto __error;
+ }
+
+ /* disable interrupts */
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+
+ snprintf(card->longname, sizeof(card->longname),
+ "%s with %s at %#lx, irq %d", card->shortname,
+ snd_ac97_get_short_name(chip->ac97), chip->port, chip->irq);
+
+ snd_via82xx_proc_init(chip);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ return 0;
+
+ __error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_via82xx_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "VIA 82xx Audio",
+ .id_table = snd_via82xx_ids,
+ .probe = snd_via82xx_probe,
+ .remove = __devexit_p(snd_via82xx_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_via82xx_suspend,
+ .resume = snd_via82xx_resume,
+#endif
+};
+
+static int __init alsa_card_via82xx_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_via82xx_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_via82xx_init)
+module_exit(alsa_card_via82xx_exit)
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
new file mode 100644
index 0000000..5bd79d2
--- /dev/null
+++ b/sound/pci/via82xx_modem.c
@@ -0,0 +1,1245 @@
+/*
+ * ALSA modem driver for VIA VT82xx (South Bridge)
+ *
+ * VT82C686A/B/C, VT8233A/C, VT8235
+ *
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
+ * Tjeerd.Mulder <Tjeerd.Mulder@fujitsu-siemens.com>
+ * 2002 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/*
+ * Changes:
+ *
+ * Sep. 2, 2004 Sasha Khapyorsky <sashak@alsa-project.org>
+ * Modified from original audio driver 'via82xx.c' to support AC97
+ * modems.
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/info.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+
+#if 0
+#define POINTER_DEBUG
+#endif
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("VIA VT82xx modem");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{VIA,VT82C686A/B/C modem,pci}}");
+
+static int index = -2; /* Exclude the first card */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int ac97_clock = 48000;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for VIA 82xx bridge.");
+module_param(ac97_clock, int, 0444);
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (default 48000Hz).");
+
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
+
+/*
+ * Direct registers
+ */
+
+#define VIAREG(via, x) ((via)->port + VIA_REG_##x)
+#define VIADEV_REG(viadev, x) ((viadev)->port + VIA_REG_##x)
+
+/* common offsets */
+#define VIA_REG_OFFSET_STATUS 0x00 /* byte - channel status */
+#define VIA_REG_STAT_ACTIVE 0x80 /* RO */
+#define VIA_REG_STAT_PAUSED 0x40 /* RO */
+#define VIA_REG_STAT_TRIGGER_QUEUED 0x08 /* RO */
+#define VIA_REG_STAT_STOPPED 0x04 /* RWC */
+#define VIA_REG_STAT_EOL 0x02 /* RWC */
+#define VIA_REG_STAT_FLAG 0x01 /* RWC */
+#define VIA_REG_OFFSET_CONTROL 0x01 /* byte - channel control */
+#define VIA_REG_CTRL_START 0x80 /* WO */
+#define VIA_REG_CTRL_TERMINATE 0x40 /* WO */
+#define VIA_REG_CTRL_AUTOSTART 0x20
+#define VIA_REG_CTRL_PAUSE 0x08 /* RW */
+#define VIA_REG_CTRL_INT_STOP 0x04
+#define VIA_REG_CTRL_INT_EOL 0x02
+#define VIA_REG_CTRL_INT_FLAG 0x01
+#define VIA_REG_CTRL_RESET 0x01 /* RW - probably reset? undocumented */
+#define VIA_REG_CTRL_INT (VIA_REG_CTRL_INT_FLAG | VIA_REG_CTRL_INT_EOL | VIA_REG_CTRL_AUTOSTART)
+#define VIA_REG_OFFSET_TYPE 0x02 /* byte - channel type (686 only) */
+#define VIA_REG_TYPE_AUTOSTART 0x80 /* RW - autostart at EOL */
+#define VIA_REG_TYPE_16BIT 0x20 /* RW */
+#define VIA_REG_TYPE_STEREO 0x10 /* RW */
+#define VIA_REG_TYPE_INT_LLINE 0x00
+#define VIA_REG_TYPE_INT_LSAMPLE 0x04
+#define VIA_REG_TYPE_INT_LESSONE 0x08
+#define VIA_REG_TYPE_INT_MASK 0x0c
+#define VIA_REG_TYPE_INT_EOL 0x02
+#define VIA_REG_TYPE_INT_FLAG 0x01
+#define VIA_REG_OFFSET_TABLE_PTR 0x04 /* dword - channel table pointer */
+#define VIA_REG_OFFSET_CURR_PTR 0x04 /* dword - channel current pointer */
+#define VIA_REG_OFFSET_STOP_IDX 0x08 /* dword - stop index, channel type, sample rate */
+#define VIA_REG_OFFSET_CURR_COUNT 0x0c /* dword - channel current count (24 bit) */
+#define VIA_REG_OFFSET_CURR_INDEX 0x0f /* byte - channel current index (for via8233 only) */
+
+#define DEFINE_VIA_REGSET(name,val) \
+enum {\
+ VIA_REG_##name##_STATUS = (val),\
+ VIA_REG_##name##_CONTROL = (val) + 0x01,\
+ VIA_REG_##name##_TYPE = (val) + 0x02,\
+ VIA_REG_##name##_TABLE_PTR = (val) + 0x04,\
+ VIA_REG_##name##_CURR_PTR = (val) + 0x04,\
+ VIA_REG_##name##_STOP_IDX = (val) + 0x08,\
+ VIA_REG_##name##_CURR_COUNT = (val) + 0x0c,\
+}
+
+/* modem block */
+DEFINE_VIA_REGSET(MO, 0x40);
+DEFINE_VIA_REGSET(MI, 0x50);
+
+/* AC'97 */
+#define VIA_REG_AC97 0x80 /* dword */
+#define VIA_REG_AC97_CODEC_ID_MASK (3<<30)
+#define VIA_REG_AC97_CODEC_ID_SHIFT 30
+#define VIA_REG_AC97_CODEC_ID_PRIMARY 0x00
+#define VIA_REG_AC97_CODEC_ID_SECONDARY 0x01
+#define VIA_REG_AC97_SECONDARY_VALID (1<<27)
+#define VIA_REG_AC97_PRIMARY_VALID (1<<25)
+#define VIA_REG_AC97_BUSY (1<<24)
+#define VIA_REG_AC97_READ (1<<23)
+#define VIA_REG_AC97_CMD_SHIFT 16
+#define VIA_REG_AC97_CMD_MASK 0x7e
+#define VIA_REG_AC97_DATA_SHIFT 0
+#define VIA_REG_AC97_DATA_MASK 0xffff
+
+#define VIA_REG_SGD_SHADOW 0x84 /* dword */
+#define VIA_REG_SGD_STAT_PB_FLAG (1<<0)
+#define VIA_REG_SGD_STAT_CP_FLAG (1<<1)
+#define VIA_REG_SGD_STAT_FM_FLAG (1<<2)
+#define VIA_REG_SGD_STAT_PB_EOL (1<<4)
+#define VIA_REG_SGD_STAT_CP_EOL (1<<5)
+#define VIA_REG_SGD_STAT_FM_EOL (1<<6)
+#define VIA_REG_SGD_STAT_PB_STOP (1<<8)
+#define VIA_REG_SGD_STAT_CP_STOP (1<<9)
+#define VIA_REG_SGD_STAT_FM_STOP (1<<10)
+#define VIA_REG_SGD_STAT_PB_ACTIVE (1<<12)
+#define VIA_REG_SGD_STAT_CP_ACTIVE (1<<13)
+#define VIA_REG_SGD_STAT_FM_ACTIVE (1<<14)
+#define VIA_REG_SGD_STAT_MR_FLAG (1<<16)
+#define VIA_REG_SGD_STAT_MW_FLAG (1<<17)
+#define VIA_REG_SGD_STAT_MR_EOL (1<<20)
+#define VIA_REG_SGD_STAT_MW_EOL (1<<21)
+#define VIA_REG_SGD_STAT_MR_STOP (1<<24)
+#define VIA_REG_SGD_STAT_MW_STOP (1<<25)
+#define VIA_REG_SGD_STAT_MR_ACTIVE (1<<28)
+#define VIA_REG_SGD_STAT_MW_ACTIVE (1<<29)
+
+#define VIA_REG_GPI_STATUS 0x88
+#define VIA_REG_GPI_INTR 0x8c
+
+#define VIA_TBL_BIT_FLAG 0x40000000
+#define VIA_TBL_BIT_EOL 0x80000000
+
+/* pci space */
+#define VIA_ACLINK_STAT 0x40
+#define VIA_ACLINK_C11_READY 0x20
+#define VIA_ACLINK_C10_READY 0x10
+#define VIA_ACLINK_C01_READY 0x04 /* secondary codec ready */
+#define VIA_ACLINK_LOWPOWER 0x02 /* low-power state */
+#define VIA_ACLINK_C00_READY 0x01 /* primary codec ready */
+#define VIA_ACLINK_CTRL 0x41
+#define VIA_ACLINK_CTRL_ENABLE 0x80 /* 0: disable, 1: enable */
+#define VIA_ACLINK_CTRL_RESET 0x40 /* 0: assert, 1: de-assert */
+#define VIA_ACLINK_CTRL_SYNC 0x20 /* 0: release SYNC, 1: force SYNC hi */
+#define VIA_ACLINK_CTRL_SDO 0x10 /* 0: release SDO, 1: force SDO hi */
+#define VIA_ACLINK_CTRL_VRA 0x08 /* 0: disable VRA, 1: enable VRA */
+#define VIA_ACLINK_CTRL_PCM 0x04 /* 0: disable PCM, 1: enable PCM */
+#define VIA_ACLINK_CTRL_FM 0x02 /* via686 only */
+#define VIA_ACLINK_CTRL_SB 0x01 /* via686 only */
+#define VIA_ACLINK_CTRL_INIT (VIA_ACLINK_CTRL_ENABLE|\
+ VIA_ACLINK_CTRL_RESET|\
+ VIA_ACLINK_CTRL_PCM)
+#define VIA_FUNC_ENABLE 0x42
+#define VIA_FUNC_MIDI_PNP 0x80 /* FIXME: it's 0x40 in the datasheet! */
+#define VIA_FUNC_MIDI_IRQMASK 0x40 /* FIXME: not documented! */
+#define VIA_FUNC_RX2C_WRITE 0x20
+#define VIA_FUNC_SB_FIFO_EMPTY 0x10
+#define VIA_FUNC_ENABLE_GAME 0x08
+#define VIA_FUNC_ENABLE_FM 0x04
+#define VIA_FUNC_ENABLE_MIDI 0x02
+#define VIA_FUNC_ENABLE_SB 0x01
+#define VIA_PNP_CONTROL 0x43
+#define VIA_MC97_CTRL 0x44
+#define VIA_MC97_CTRL_ENABLE 0x80
+#define VIA_MC97_CTRL_SECONDARY 0x40
+#define VIA_MC97_CTRL_INIT (VIA_MC97_CTRL_ENABLE|\
+ VIA_MC97_CTRL_SECONDARY)
+
+
+/*
+ * pcm stream
+ */
+
+struct snd_via_sg_table {
+ unsigned int offset;
+ unsigned int size;
+} ;
+
+#define VIA_TABLE_SIZE 255
+
+struct viadev {
+ unsigned int reg_offset;
+ unsigned long port;
+ int direction; /* playback = 0, capture = 1 */
+ struct snd_pcm_substream *substream;
+ int running;
+ unsigned int tbl_entries; /* # descriptors */
+ struct snd_dma_buffer table;
+ struct snd_via_sg_table *idx_table;
+ /* for recovery from the unexpected pointer */
+ unsigned int lastpos;
+ unsigned int bufsize;
+ unsigned int bufsize2;
+};
+
+enum { TYPE_CARD_VIA82XX_MODEM = 1 };
+
+#define VIA_MAX_MODEM_DEVS 2
+
+struct via82xx_modem {
+ int irq;
+
+ unsigned long port;
+
+ unsigned int intr_mask; /* SGD_SHADOW mask to check interrupts */
+
+ struct pci_dev *pci;
+ struct snd_card *card;
+
+ unsigned int num_devs;
+ unsigned int playback_devno, capture_devno;
+ struct viadev devs[VIA_MAX_MODEM_DEVS];
+
+ struct snd_pcm *pcms[2];
+
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97 *ac97;
+ unsigned int ac97_clock;
+ unsigned int ac97_secondary; /* secondary AC'97 codec is present */
+
+ spinlock_t reg_lock;
+ struct snd_info_entry *proc_entry;
+};
+
+static struct pci_device_id snd_via82xx_modem_ids[] = {
+ { 0x1106, 0x3068, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA82XX_MODEM, },
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_via82xx_modem_ids);
+
+/*
+ */
+
+/*
+ * allocate and initialize the descriptor buffers
+ * periods = number of periods
+ * fragsize = period size in bytes
+ */
+static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substream,
+ struct pci_dev *pci,
+ unsigned int periods, unsigned int fragsize)
+{
+ unsigned int i, idx, ofs, rest;
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+
+ if (dev->table.area == NULL) {
+ /* the start of each lists must be aligned to 8 bytes,
+ * but the kernel pages are much bigger, so we don't care
+ */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ PAGE_ALIGN(VIA_TABLE_SIZE * 2 * 8),
+ &dev->table) < 0)
+ return -ENOMEM;
+ }
+ if (! dev->idx_table) {
+ dev->idx_table = kmalloc(sizeof(*dev->idx_table) * VIA_TABLE_SIZE, GFP_KERNEL);
+ if (! dev->idx_table)
+ return -ENOMEM;
+ }
+
+ /* fill the entries */
+ idx = 0;
+ ofs = 0;
+ for (i = 0; i < periods; i++) {
+ rest = fragsize;
+ /* fill descriptors for a period.
+ * a period can be split to several descriptors if it's
+ * over page boundary.
+ */
+ do {
+ unsigned int r;
+ unsigned int flag;
+ unsigned int addr;
+
+ if (idx >= VIA_TABLE_SIZE) {
+ snd_printk(KERN_ERR "via82xx: too much table size!\n");
+ return -EINVAL;
+ }
+ addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+ ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr);
+ r = PAGE_SIZE - (ofs % PAGE_SIZE);
+ if (rest < r)
+ r = rest;
+ rest -= r;
+ if (! rest) {
+ if (i == periods - 1)
+ flag = VIA_TBL_BIT_EOL; /* buffer boundary */
+ else
+ flag = VIA_TBL_BIT_FLAG; /* period boundary */
+ } else
+ flag = 0; /* period continues to the next */
+ // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest);
+ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
+ dev->idx_table[idx].offset = ofs;
+ dev->idx_table[idx].size = r;
+ ofs += r;
+ idx++;
+ } while (rest > 0);
+ }
+ dev->tbl_entries = idx;
+ dev->bufsize = periods * fragsize;
+ dev->bufsize2 = dev->bufsize / 2;
+ return 0;
+}
+
+
+static int clean_via_table(struct viadev *dev, struct snd_pcm_substream *substream,
+ struct pci_dev *pci)
+{
+ if (dev->table.area) {
+ snd_dma_free_pages(&dev->table);
+ dev->table.area = NULL;
+ }
+ kfree(dev->idx_table);
+ dev->idx_table = NULL;
+ return 0;
+}
+
+/*
+ * Basic I/O
+ */
+
+static inline unsigned int snd_via82xx_codec_xread(struct via82xx_modem *chip)
+{
+ return inl(VIAREG(chip, AC97));
+}
+
+static inline void snd_via82xx_codec_xwrite(struct via82xx_modem *chip, unsigned int val)
+{
+ outl(val, VIAREG(chip, AC97));
+}
+
+static int snd_via82xx_codec_ready(struct via82xx_modem *chip, int secondary)
+{
+ unsigned int timeout = 1000; /* 1ms */
+ unsigned int val;
+
+ while (timeout-- > 0) {
+ udelay(1);
+ if (!((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY))
+ return val & 0xffff;
+ }
+ snd_printk(KERN_ERR "codec_ready: codec %i is not ready [0x%x]\n",
+ secondary, snd_via82xx_codec_xread(chip));
+ return -EIO;
+}
+
+static int snd_via82xx_codec_valid(struct via82xx_modem *chip, int secondary)
+{
+ unsigned int timeout = 1000; /* 1ms */
+ unsigned int val, val1;
+ unsigned int stat = !secondary ? VIA_REG_AC97_PRIMARY_VALID :
+ VIA_REG_AC97_SECONDARY_VALID;
+
+ while (timeout-- > 0) {
+ val = snd_via82xx_codec_xread(chip);
+ val1 = val & (VIA_REG_AC97_BUSY | stat);
+ if (val1 == stat)
+ return val & 0xffff;
+ udelay(1);
+ }
+ return -EIO;
+}
+
+static void snd_via82xx_codec_wait(struct snd_ac97 *ac97)
+{
+ struct via82xx_modem *chip = ac97->private_data;
+ int err;
+ err = snd_via82xx_codec_ready(chip, ac97->num);
+ /* here we need to wait fairly for long time.. */
+ msleep(500);
+}
+
+static void snd_via82xx_codec_write(struct snd_ac97 *ac97,
+ unsigned short reg,
+ unsigned short val)
+{
+ struct via82xx_modem *chip = ac97->private_data;
+ unsigned int xval;
+ if(reg == AC97_GPIO_STATUS) {
+ outl(val, VIAREG(chip, GPI_STATUS));
+ return;
+ }
+ xval = !ac97->num ? VIA_REG_AC97_CODEC_ID_PRIMARY : VIA_REG_AC97_CODEC_ID_SECONDARY;
+ xval <<= VIA_REG_AC97_CODEC_ID_SHIFT;
+ xval |= reg << VIA_REG_AC97_CMD_SHIFT;
+ xval |= val << VIA_REG_AC97_DATA_SHIFT;
+ snd_via82xx_codec_xwrite(chip, xval);
+ snd_via82xx_codec_ready(chip, ac97->num);
+}
+
+static unsigned short snd_via82xx_codec_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ struct via82xx_modem *chip = ac97->private_data;
+ unsigned int xval, val = 0xffff;
+ int again = 0;
+
+ xval = ac97->num << VIA_REG_AC97_CODEC_ID_SHIFT;
+ xval |= ac97->num ? VIA_REG_AC97_SECONDARY_VALID : VIA_REG_AC97_PRIMARY_VALID;
+ xval |= VIA_REG_AC97_READ;
+ xval |= (reg & 0x7f) << VIA_REG_AC97_CMD_SHIFT;
+ while (1) {
+ if (again++ > 3) {
+ snd_printk(KERN_ERR "codec_read: codec %i is not valid [0x%x]\n",
+ ac97->num, snd_via82xx_codec_xread(chip));
+ return 0xffff;
+ }
+ snd_via82xx_codec_xwrite(chip, xval);
+ udelay (20);
+ if (snd_via82xx_codec_valid(chip, ac97->num) >= 0) {
+ udelay(25);
+ val = snd_via82xx_codec_xread(chip);
+ break;
+ }
+ }
+ return val & 0xffff;
+}
+
+static void snd_via82xx_channel_reset(struct via82xx_modem *chip, struct viadev *viadev)
+{
+ outb(VIA_REG_CTRL_PAUSE | VIA_REG_CTRL_TERMINATE | VIA_REG_CTRL_RESET,
+ VIADEV_REG(viadev, OFFSET_CONTROL));
+ inb(VIADEV_REG(viadev, OFFSET_CONTROL));
+ udelay(50);
+ /* disable interrupts */
+ outb(0x00, VIADEV_REG(viadev, OFFSET_CONTROL));
+ /* clear interrupts */
+ outb(0x03, VIADEV_REG(viadev, OFFSET_STATUS));
+ outb(0x00, VIADEV_REG(viadev, OFFSET_TYPE)); /* for via686 */
+ // outl(0, VIADEV_REG(viadev, OFFSET_CURR_PTR));
+ viadev->lastpos = 0;
+}
+
+
+/*
+ * Interrupt handler
+ */
+
+static irqreturn_t snd_via82xx_interrupt(int irq, void *dev_id)
+{
+ struct via82xx_modem *chip = dev_id;
+ unsigned int status;
+ unsigned int i;
+
+ status = inl(VIAREG(chip, SGD_SHADOW));
+ if (! (status & chip->intr_mask)) {
+ return IRQ_NONE;
+ }
+// _skip_sgd:
+
+ /* check status for each stream */
+ spin_lock(&chip->reg_lock);
+ for (i = 0; i < chip->num_devs; i++) {
+ struct viadev *viadev = &chip->devs[i];
+ unsigned char c_status = inb(VIADEV_REG(viadev, OFFSET_STATUS));
+ c_status &= (VIA_REG_STAT_EOL|VIA_REG_STAT_FLAG|VIA_REG_STAT_STOPPED);
+ if (! c_status)
+ continue;
+ if (viadev->substream && viadev->running) {
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(viadev->substream);
+ spin_lock(&chip->reg_lock);
+ }
+ outb(c_status, VIADEV_REG(viadev, OFFSET_STATUS)); /* ack */
+ }
+ spin_unlock(&chip->reg_lock);
+ return IRQ_HANDLED;
+}
+
+/*
+ * PCM callbacks
+ */
+
+/*
+ * trigger callback
+ */
+static int snd_via82xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ unsigned char val = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ val |= VIA_REG_CTRL_START;
+ viadev->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ val = VIA_REG_CTRL_TERMINATE;
+ viadev->running = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ val |= VIA_REG_CTRL_PAUSE;
+ viadev->running = 0;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ viadev->running = 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+ outb(val, VIADEV_REG(viadev, OFFSET_CONTROL));
+ if (cmd == SNDRV_PCM_TRIGGER_STOP)
+ snd_via82xx_channel_reset(chip, viadev);
+ return 0;
+}
+
+/*
+ * pointer callbacks
+ */
+
+/*
+ * calculate the linear position at the given sg-buffer index and the rest count
+ */
+
+#define check_invalid_pos(viadev,pos) \
+ ((pos) < viadev->lastpos && ((pos) >= viadev->bufsize2 ||\
+ viadev->lastpos < viadev->bufsize2))
+
+static inline unsigned int calc_linear_pos(struct viadev *viadev, unsigned int idx,
+ unsigned int count)
+{
+ unsigned int size, res;
+
+ size = viadev->idx_table[idx].size;
+ res = viadev->idx_table[idx].offset + size - count;
+
+ /* check the validity of the calculated position */
+ if (size < count) {
+ snd_printd(KERN_ERR "invalid via82xx_cur_ptr (size = %d, count = %d)\n",
+ (int)size, (int)count);
+ res = viadev->lastpos;
+ } else if (check_invalid_pos(viadev, res)) {
+#ifdef POINTER_DEBUG
+ printk(KERN_DEBUG "fail: idx = %i/%i, lastpos = 0x%x, "
+ "bufsize2 = 0x%x, offsize = 0x%x, size = 0x%x, "
+ "count = 0x%x\n", idx, viadev->tbl_entries, viadev->lastpos,
+ viadev->bufsize2, viadev->idx_table[idx].offset,
+ viadev->idx_table[idx].size, count);
+#endif
+ if (count && size < count) {
+ snd_printd(KERN_ERR "invalid via82xx_cur_ptr, "
+ "using last valid pointer\n");
+ res = viadev->lastpos;
+ } else {
+ if (! count)
+ /* bogus count 0 on the DMA boundary? */
+ res = viadev->idx_table[idx].offset;
+ else
+ /* count register returns full size
+ * when end of buffer is reached
+ */
+ res = viadev->idx_table[idx].offset + size;
+ if (check_invalid_pos(viadev, res)) {
+ snd_printd(KERN_ERR "invalid via82xx_cur_ptr (2), "
+ "using last valid pointer\n");
+ res = viadev->lastpos;
+ }
+ }
+ }
+ viadev->lastpos = res; /* remember the last position */
+ if (res >= viadev->bufsize)
+ res -= viadev->bufsize;
+ return res;
+}
+
+/*
+ * get the current pointer on via686
+ */
+static snd_pcm_uframes_t snd_via686_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ unsigned int idx, ptr, count, res;
+
+ if (snd_BUG_ON(!viadev->tbl_entries))
+ return 0;
+ if (!(inb(VIADEV_REG(viadev, OFFSET_STATUS)) & VIA_REG_STAT_ACTIVE))
+ return 0;
+
+ spin_lock(&chip->reg_lock);
+ count = inl(VIADEV_REG(viadev, OFFSET_CURR_COUNT)) & 0xffffff;
+ /* The via686a does not have the current index register,
+ * so we need to calculate the index from CURR_PTR.
+ */
+ ptr = inl(VIADEV_REG(viadev, OFFSET_CURR_PTR));
+ if (ptr <= (unsigned int)viadev->table.addr)
+ idx = 0;
+ else /* CURR_PTR holds the address + 8 */
+ idx = ((ptr - (unsigned int)viadev->table.addr) / 8 - 1) %
+ viadev->tbl_entries;
+ res = calc_linear_pos(viadev, idx, count);
+ spin_unlock(&chip->reg_lock);
+
+ return bytes_to_frames(substream->runtime, res);
+}
+
+/*
+ * hw_params callback:
+ * allocate the buffer and build up the buffer description table
+ */
+static int snd_via82xx_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ err = build_via_table(viadev, substream, chip->pci,
+ params_periods(hw_params),
+ params_period_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ snd_ac97_write(chip->ac97, AC97_LINE1_RATE, params_rate(hw_params));
+ snd_ac97_write(chip->ac97, AC97_LINE1_LEVEL, 0);
+
+ return 0;
+}
+
+/*
+ * hw_free callback:
+ * clean up the buffer description table and release the buffer
+ */
+static int snd_via82xx_hw_free(struct snd_pcm_substream *substream)
+{
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+
+ clean_via_table(viadev, substream, chip->pci);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+
+/*
+ * set up the table pointer
+ */
+static void snd_via82xx_set_table_ptr(struct via82xx_modem *chip, struct viadev *viadev)
+{
+ snd_via82xx_codec_ready(chip, chip->ac97_secondary);
+ outl((u32)viadev->table.addr, VIADEV_REG(viadev, OFFSET_TABLE_PTR));
+ udelay(20);
+ snd_via82xx_codec_ready(chip, chip->ac97_secondary);
+}
+
+/*
+ * prepare callback for playback and capture
+ */
+static int snd_via82xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+
+ snd_via82xx_channel_reset(chip, viadev);
+ /* this must be set after channel_reset */
+ snd_via82xx_set_table_ptr(chip, viadev);
+ outb(VIA_REG_TYPE_AUTOSTART|VIA_REG_TYPE_INT_EOL|VIA_REG_TYPE_INT_FLAG,
+ VIADEV_REG(viadev, OFFSET_TYPE));
+ return 0;
+}
+
+/*
+ * pcm hardware definition, identical for both playback and capture
+ */
+static struct snd_pcm_hardware snd_via82xx_hw =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ /* SNDRV_PCM_INFO_RESUME | */
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 128 * 1024,
+ .periods_min = 2,
+ .periods_max = VIA_TABLE_SIZE / 2,
+ .fifo_size = 0,
+};
+
+
+/*
+ * open callback skeleton
+ */
+static int snd_via82xx_modem_pcm_open(struct via82xx_modem *chip, struct viadev *viadev,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+ static unsigned int rates[] = { 8000, 9600, 12000, 16000 };
+ static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+
+ runtime->hw = snd_via82xx_hw;
+
+ if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates)) < 0)
+ return err;
+
+ /* we may remove following constaint when we modify table entries
+ in interrupt */
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ runtime->private_data = viadev;
+ viadev->substream = substream;
+
+ return 0;
+}
+
+
+/*
+ * open callback for playback
+ */
+static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
+{
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number];
+
+ return snd_via82xx_modem_pcm_open(chip, viadev, substream);
+}
+
+/*
+ * open callback for capture
+ */
+static int snd_via82xx_capture_open(struct snd_pcm_substream *substream)
+{
+ struct via82xx_modem *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = &chip->devs[chip->capture_devno + substream->pcm->device];
+
+ return snd_via82xx_modem_pcm_open(chip, viadev, substream);
+}
+
+/*
+ * close callback
+ */
+static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct viadev *viadev = substream->runtime->private_data;
+
+ viadev->substream = NULL;
+ return 0;
+}
+
+
+/* via686 playback callbacks */
+static struct snd_pcm_ops snd_via686_playback_ops = {
+ .open = snd_via82xx_playback_open,
+ .close = snd_via82xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_via82xx_hw_params,
+ .hw_free = snd_via82xx_hw_free,
+ .prepare = snd_via82xx_pcm_prepare,
+ .trigger = snd_via82xx_pcm_trigger,
+ .pointer = snd_via686_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/* via686 capture callbacks */
+static struct snd_pcm_ops snd_via686_capture_ops = {
+ .open = snd_via82xx_capture_open,
+ .close = snd_via82xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_via82xx_hw_params,
+ .hw_free = snd_via82xx_hw_free,
+ .prepare = snd_via82xx_pcm_prepare,
+ .trigger = snd_via82xx_pcm_trigger,
+ .pointer = snd_via686_pcm_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+
+static void init_viadev(struct via82xx_modem *chip, int idx, unsigned int reg_offset,
+ int direction)
+{
+ chip->devs[idx].reg_offset = reg_offset;
+ chip->devs[idx].direction = direction;
+ chip->devs[idx].port = chip->port + reg_offset;
+}
+
+/*
+ * create a pcm instance for via686a/b
+ */
+static int __devinit snd_via686_pcm_new(struct via82xx_modem *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ chip->playback_devno = 0;
+ chip->capture_devno = 1;
+ chip->num_devs = 2;
+ chip->intr_mask = 0x330000; /* FLAGS | EOL for MR, MW */
+
+ err = snd_pcm_new(chip->card, chip->card->shortname, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_via686_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_via686_capture_ops);
+ pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
+ pcm->private_data = chip;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcms[0] = pcm;
+ init_viadev(chip, 0, VIA_REG_MO_STATUS, 0);
+ init_viadev(chip, 1, VIA_REG_MI_STATUS, 1);
+
+ if ((err = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 128*1024)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * Mixer part
+ */
+
+
+static void snd_via82xx_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct via82xx_modem *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void snd_via82xx_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct via82xx_modem *chip = ac97->private_data;
+ chip->ac97 = NULL;
+}
+
+
+static int __devinit snd_via82xx_mixer_new(struct via82xx_modem *chip)
+{
+ struct snd_ac97_template ac97;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_via82xx_codec_write,
+ .read = snd_via82xx_codec_read,
+ .wait = snd_via82xx_codec_wait,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &chip->ac97_bus)) < 0)
+ return err;
+ chip->ac97_bus->private_free = snd_via82xx_mixer_free_ac97_bus;
+ chip->ac97_bus->clock = chip->ac97_clock;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_via82xx_mixer_free_ac97;
+ ac97.pci = chip->pci;
+ ac97.scaps = AC97_SCAP_SKIP_AUDIO | AC97_SCAP_POWER_SAVE;
+ ac97.num = chip->ac97_secondary;
+
+ if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * proc interface
+ */
+static void snd_via82xx_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct via82xx_modem *chip = entry->private_data;
+ int i;
+
+ snd_iprintf(buffer, "%s\n\n", chip->card->longname);
+ for (i = 0; i < 0xa0; i += 4) {
+ snd_iprintf(buffer, "%02x: %08x\n", i, inl(chip->port + i));
+ }
+}
+
+static void __devinit snd_via82xx_proc_init(struct via82xx_modem *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "via82xx", &entry))
+ snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read);
+}
+
+/*
+ *
+ */
+
+static int snd_via82xx_chip_init(struct via82xx_modem *chip)
+{
+ unsigned int val;
+ unsigned long end_time;
+ unsigned char pval;
+
+ pci_read_config_byte(chip->pci, VIA_MC97_CTRL, &pval);
+ if((pval & VIA_MC97_CTRL_INIT) != VIA_MC97_CTRL_INIT) {
+ pci_write_config_byte(chip->pci, 0x44, pval|VIA_MC97_CTRL_INIT);
+ udelay(100);
+ }
+
+ pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
+ if (! (pval & VIA_ACLINK_C00_READY)) { /* codec not ready? */
+ /* deassert ACLink reset, force SYNC */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL,
+ VIA_ACLINK_CTRL_ENABLE |
+ VIA_ACLINK_CTRL_RESET |
+ VIA_ACLINK_CTRL_SYNC);
+ udelay(100);
+#if 1 /* FIXME: should we do full reset here for all chip models? */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, 0x00);
+ udelay(100);
+#else
+ /* deassert ACLink reset, force SYNC (warm AC'97 reset) */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL,
+ VIA_ACLINK_CTRL_RESET|VIA_ACLINK_CTRL_SYNC);
+ udelay(2);
+#endif
+ /* ACLink on, deassert ACLink reset, VSR, SGD data out */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_INIT);
+ udelay(100);
+ }
+
+ pci_read_config_byte(chip->pci, VIA_ACLINK_CTRL, &pval);
+ if ((pval & VIA_ACLINK_CTRL_INIT) != VIA_ACLINK_CTRL_INIT) {
+ /* ACLink on, deassert ACLink reset, VSR, SGD data out */
+ pci_write_config_byte(chip->pci, VIA_ACLINK_CTRL, VIA_ACLINK_CTRL_INIT);
+ udelay(100);
+ }
+
+ /* wait until codec ready */
+ end_time = jiffies + msecs_to_jiffies(750);
+ do {
+ pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval);
+ if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+
+ if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY)
+ snd_printk(KERN_ERR "AC'97 codec is not ready [0x%x]\n", val);
+
+ snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
+ VIA_REG_AC97_SECONDARY_VALID |
+ (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
+ end_time = jiffies + msecs_to_jiffies(750);
+ snd_via82xx_codec_xwrite(chip, VIA_REG_AC97_READ |
+ VIA_REG_AC97_SECONDARY_VALID |
+ (VIA_REG_AC97_CODEC_ID_SECONDARY << VIA_REG_AC97_CODEC_ID_SHIFT));
+ do {
+ if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_SECONDARY_VALID) {
+ chip->ac97_secondary = 1;
+ goto __ac97_ok2;
+ }
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+ /* This is ok, the most of motherboards have only one codec */
+
+ __ac97_ok2:
+
+ /* route FM trap to IRQ, disable FM trap */
+ // pci_write_config_byte(chip->pci, VIA_FM_NMI_CTRL, 0);
+ /* disable all GPI interrupts */
+ outl(0, VIAREG(chip, GPI_INTR));
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct via82xx_modem *chip = card->private_data;
+ int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ for (i = 0; i < 2; i++)
+ snd_pcm_suspend_all(chip->pcms[i]);
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+ synchronize_irq(chip->irq);
+ snd_ac97_suspend(chip->ac97);
+
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+static int snd_via82xx_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct via82xx_modem *chip = card->private_data;
+ int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "via82xx-modem: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+
+ snd_via82xx_chip_init(chip);
+
+ snd_ac97_resume(chip->ac97);
+
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static int snd_via82xx_free(struct via82xx_modem *chip)
+{
+ unsigned int i;
+
+ if (chip->irq < 0)
+ goto __end_hw;
+ /* disable interrupts */
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+
+ __end_hw:
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_via82xx_dev_free(struct snd_device *device)
+{
+ struct via82xx_modem *chip = device->device_data;
+ return snd_via82xx_free(chip);
+}
+
+static int __devinit snd_via82xx_create(struct snd_card *card,
+ struct pci_dev *pci,
+ int chip_type,
+ int revision,
+ unsigned int ac97_clock,
+ struct via82xx_modem ** r_via)
+{
+ struct via82xx_modem *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_via82xx_dev_free,
+ };
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ if ((chip = kzalloc(sizeof(*chip), GFP_KERNEL)) == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ spin_lock_init(&chip->reg_lock);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ if ((err = pci_request_regions(pci, card->driver)) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->port = pci_resource_start(pci, 0);
+ if (request_irq(pci->irq, snd_via82xx_interrupt, IRQF_SHARED,
+ card->driver, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_via82xx_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+ if (ac97_clock >= 8000 && ac97_clock <= 48000)
+ chip->ac97_clock = ac97_clock;
+ synchronize_irq(chip->irq);
+
+ if ((err = snd_via82xx_chip_init(chip)) < 0) {
+ snd_via82xx_free(chip);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_via82xx_free(chip);
+ return err;
+ }
+
+ /* The 8233 ac97 controller does not implement the master bit
+ * in the pci command register. IMHO this is a violation of the PCI spec.
+ * We call pci_set_master here because it does not hurt. */
+ pci_set_master(pci);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *r_via = chip;
+ return 0;
+}
+
+
+static int __devinit snd_via82xx_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ struct snd_card *card;
+ struct via82xx_modem *chip;
+ int chip_type = 0, card_type;
+ unsigned int i;
+ int err;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ card_type = pci_id->driver_data;
+ switch (card_type) {
+ case TYPE_CARD_VIA82XX_MODEM:
+ strcpy(card->driver, "VIA82XX-MODEM");
+ sprintf(card->shortname, "VIA 82XX modem");
+ break;
+ default:
+ snd_printk(KERN_ERR "invalid card type %d\n", card_type);
+ err = -EINVAL;
+ goto __error;
+ }
+
+ if ((err = snd_via82xx_create(card, pci, chip_type, pci->revision,
+ ac97_clock, &chip)) < 0)
+ goto __error;
+ card->private_data = chip;
+ if ((err = snd_via82xx_mixer_new(chip)) < 0)
+ goto __error;
+
+ if ((err = snd_via686_pcm_new(chip)) < 0 )
+ goto __error;
+
+ /* disable interrupts */
+ for (i = 0; i < chip->num_devs; i++)
+ snd_via82xx_channel_reset(chip, &chip->devs[i]);
+
+ sprintf(card->longname, "%s at 0x%lx, irq %d",
+ card->shortname, chip->port, chip->irq);
+
+ snd_via82xx_proc_init(chip);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ return 0;
+
+ __error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit snd_via82xx_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "VIA 82xx Modem",
+ .id_table = snd_via82xx_modem_ids,
+ .probe = snd_via82xx_probe,
+ .remove = __devexit_p(snd_via82xx_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_via82xx_suspend,
+ .resume = snd_via82xx_resume,
+#endif
+};
+
+static int __init alsa_card_via82xx_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_via82xx_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_via82xx_init)
+module_exit(alsa_card_via82xx_exit)
diff --git a/sound/pci/vx222/Makefile b/sound/pci/vx222/Makefile
new file mode 100644
index 0000000..a4d08d4
--- /dev/null
+++ b/sound/pci/vx222/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-vx222-objs := vx222.o vx222_ops.o
+
+obj-$(CONFIG_SND_VX222) += snd-vx222.o
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
new file mode 100644
index 0000000..acc352f
--- /dev/null
+++ b/sound/pci/vx222/vx222.c
@@ -0,0 +1,314 @@
+/*
+ * Driver for Digigram VX222 V2/Mic PCI soundcards
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include "vx222.h"
+
+#define CARD_NAME "VX222"
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("Digigram VX222 V2/Mic");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Digigram," CARD_NAME "}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int mic[SNDRV_CARDS]; /* microphone */
+static int ibl[SNDRV_CARDS]; /* microphone */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Digigram " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Digigram " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard.");
+module_param_array(mic, bool, NULL, 0444);
+MODULE_PARM_DESC(mic, "Enable Microphone.");
+module_param_array(ibl, int, NULL, 0444);
+MODULE_PARM_DESC(ibl, "Capture IBL size.");
+
+/*
+ */
+
+enum {
+ VX_PCI_VX222_OLD,
+ VX_PCI_VX222_NEW
+};
+
+static struct pci_device_id snd_vx222_ids[] = {
+ { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */
+ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_vx222_ids);
+
+
+/*
+ */
+
+static const DECLARE_TLV_DB_SCALE(db_scale_old_vol, -11350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_akm, -7350, 50, 0);
+
+static struct snd_vx_hardware vx222_old_hw = {
+
+ .name = "VX222/Old",
+ .type = VX_TYPE_BOARD,
+ /* hw specs */
+ .num_codecs = 1,
+ .num_ins = 1,
+ .num_outs = 1,
+ .output_level_max = VX_ANALOG_OUT_LEVEL_MAX,
+ .output_level_db_scale = db_scale_old_vol,
+};
+
+static struct snd_vx_hardware vx222_v2_hw = {
+
+ .name = "VX222/v2",
+ .type = VX_TYPE_V2,
+ /* hw specs */
+ .num_codecs = 1,
+ .num_ins = 1,
+ .num_outs = 1,
+ .output_level_max = VX2_AKM_LEVEL_MAX,
+ .output_level_db_scale = db_scale_akm,
+};
+
+static struct snd_vx_hardware vx222_mic_hw = {
+
+ .name = "VX222/Mic",
+ .type = VX_TYPE_MIC,
+ /* hw specs */
+ .num_codecs = 1,
+ .num_ins = 1,
+ .num_outs = 1,
+ .output_level_max = VX2_AKM_LEVEL_MAX,
+ .output_level_db_scale = db_scale_akm,
+};
+
+
+/*
+ */
+static int snd_vx222_free(struct vx_core *chip)
+{
+ struct snd_vx222 *vx = (struct snd_vx222 *)chip;
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void*)chip);
+ if (vx->port[0])
+ pci_release_regions(vx->pci);
+ pci_disable_device(vx->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_vx222_dev_free(struct snd_device *device)
+{
+ struct vx_core *chip = device->device_data;
+ return snd_vx222_free(chip);
+}
+
+
+static int __devinit snd_vx222_create(struct snd_card *card, struct pci_dev *pci,
+ struct snd_vx_hardware *hw,
+ struct snd_vx222 **rchip)
+{
+ struct vx_core *chip;
+ struct snd_vx222 *vx;
+ int i, err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_vx222_dev_free,
+ };
+ struct snd_vx_ops *vx_ops;
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ pci_set_master(pci);
+
+ vx_ops = hw->type == VX_TYPE_BOARD ? &vx222_old_ops : &vx222_ops;
+ chip = snd_vx_create(card, hw, vx_ops,
+ sizeof(struct snd_vx222) - sizeof(struct vx_core));
+ if (! chip) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ vx = (struct snd_vx222 *)chip;
+ vx->pci = pci;
+
+ if ((err = pci_request_regions(pci, CARD_NAME)) < 0) {
+ snd_vx222_free(chip);
+ return err;
+ }
+ for (i = 0; i < 2; i++)
+ vx->port[i] = pci_resource_start(pci, i + 1);
+
+ if (request_irq(pci->irq, snd_vx_irq_handler, IRQF_SHARED,
+ CARD_NAME, chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_vx222_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_vx222_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = vx;
+ return 0;
+}
+
+
+static int __devinit snd_vx222_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct snd_vx_hardware *hw;
+ struct snd_vx222 *vx;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ switch ((int)pci_id->driver_data) {
+ case VX_PCI_VX222_OLD:
+ hw = &vx222_old_hw;
+ break;
+ case VX_PCI_VX222_NEW:
+ default:
+ if (mic[dev])
+ hw = &vx222_mic_hw;
+ else
+ hw = &vx222_v2_hw;
+ break;
+ }
+ if ((err = snd_vx222_create(card, pci, hw, &vx)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ card->private_data = vx;
+ vx->core.ibl.size = ibl[dev];
+
+ sprintf(card->longname, "%s at 0x%lx & 0x%lx, irq %i",
+ card->shortname, vx->port[0], vx->port[1], vx->core.irq);
+ snd_printdd("%s at 0x%lx & 0x%lx, irq %i\n",
+ card->shortname, vx->port[0], vx->port[1], vx->core.irq);
+
+#ifdef SND_VX_FW_LOADER
+ vx->core.dev = &pci->dev;
+#endif
+
+ if ((err = snd_vx_setup_firmware(&vx->core)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_vx222_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+#ifdef CONFIG_PM
+static int snd_vx222_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_vx222 *vx = card->private_data;
+ int err;
+
+ err = snd_vx_suspend(&vx->core, state);
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return err;
+}
+
+static int snd_vx222_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_vx222 *vx = card->private_data;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "vx222: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+ return snd_vx_resume(&vx->core);
+}
+#endif
+
+static struct pci_driver driver = {
+ .name = "Digigram VX222",
+ .id_table = snd_vx222_ids,
+ .probe = snd_vx222_probe,
+ .remove = __devexit_p(snd_vx222_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_vx222_suspend,
+ .resume = snd_vx222_resume,
+#endif
+};
+
+static int __init alsa_card_vx222_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_vx222_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_vx222_init)
+module_exit(alsa_card_vx222_exit)
diff --git a/sound/pci/vx222/vx222.h b/sound/pci/vx222/vx222.h
new file mode 100644
index 0000000..2f0d78f
--- /dev/null
+++ b/sound/pci/vx222/vx222.h
@@ -0,0 +1,114 @@
+/*
+ * Driver for Digigram VX222 PCI soundcards
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __VX222_H
+#define __VX222_H
+
+#include <sound/vx_core.h>
+
+struct snd_vx222 {
+
+ struct vx_core core;
+
+ /* h/w config; for PLX and for DSP */
+ struct pci_dev *pci;
+ unsigned long port[2];
+
+ unsigned int regCDSP; /* current CDSP register */
+ unsigned int regCFG; /* current CFG register */
+ unsigned int regSELMIC; /* current SELMIC reg. (for VX222 Mic) */
+
+ int input_level[2]; /* input level for vx222 mic */
+ int mic_level; /* mic level for vx222 mic */
+};
+
+/* we use a lookup table with 148 values, see vx_mixer.c */
+#define VX2_AKM_LEVEL_MAX 0x93
+
+extern struct snd_vx_ops vx222_ops;
+extern struct snd_vx_ops vx222_old_ops;
+
+/* Offset of registers with base equal to portDSP. */
+#define VX_RESET_DMA_REGISTER_OFFSET 0x00000008
+
+/* Constants used to access the INTCSR register. */
+#define VX_INTCSR_VALUE 0x00000001
+#define VX_PCI_INTERRUPT_MASK 0x00000040
+
+/* Constants used to access the CDSP register (0x20). */
+#define VX_CDSP_TEST1_MASK 0x00000080
+#define VX_CDSP_TOR1_MASK 0x00000040
+#define VX_CDSP_TOR2_MASK 0x00000020
+#define VX_CDSP_RESERVED0_0_MASK 0x00000010
+#define VX_CDSP_CODEC_RESET_MASK 0x00000008
+#define VX_CDSP_VALID_IRQ_MASK 0x00000004
+#define VX_CDSP_TEST0_MASK 0x00000002
+#define VX_CDSP_DSP_RESET_MASK 0x00000001
+
+#define VX_CDSP_GPIO_OUT_MASK 0x00000060
+#define VX_GPIO_OUT_BIT_OFFSET 5 // transform output to bit 0 and 1
+
+/* Constants used to access the CFG register (0x24). */
+#define VX_CFG_SYNCDSP_MASK 0x00000080
+#define VX_CFG_RESERVED0_0_MASK 0x00000040
+#define VX_CFG_RESERVED1_0_MASK 0x00000020
+#define VX_CFG_RESERVED2_0_MASK 0x00000010
+#define VX_CFG_DATAIN_SEL_MASK 0x00000008 // 0 (ana), 1 (UER)
+#define VX_CFG_RESERVED3_0_MASK 0x00000004
+#define VX_CFG_RESERVED4_0_MASK 0x00000002
+#define VX_CFG_CLOCKIN_SEL_MASK 0x00000001 // 0 (internal), 1 (AES/EBU)
+
+/* Constants used to access the STATUS register (0x30). */
+#define VX_STATUS_DATA_XICOR_MASK 0x00000080
+#define VX_STATUS_VAL_TEST1_MASK 0x00000040
+#define VX_STATUS_VAL_TEST0_MASK 0x00000020
+#define VX_STATUS_RESERVED0_MASK 0x00000010
+#define VX_STATUS_VAL_TOR1_MASK 0x00000008
+#define VX_STATUS_VAL_TOR0_MASK 0x00000004
+#define VX_STATUS_LEVEL_IN_MASK 0x00000002 // 6 dBu (0), 22 dBu (1)
+#define VX_STATUS_MEMIRQ_MASK 0x00000001
+
+#define VX_STATUS_GPIO_IN_MASK 0x0000000C
+#define VX_GPIO_IN_BIT_OFFSET 0 // leave input as bit 2 and 3
+
+/* Constants used to access the MICRO INPUT SELECT register (0x40). */
+#define MICRO_SELECT_INPUT_NORM 0x00
+#define MICRO_SELECT_INPUT_MUTE 0x01
+#define MICRO_SELECT_INPUT_LIMIT 0x02
+#define MICRO_SELECT_INPUT_MASK 0x03
+
+#define MICRO_SELECT_PREAMPLI_G_0 0x00
+#define MICRO_SELECT_PREAMPLI_G_1 0x04
+#define MICRO_SELECT_PREAMPLI_G_2 0x08
+#define MICRO_SELECT_PREAMPLI_G_3 0x0C
+#define MICRO_SELECT_PREAMPLI_MASK 0x0C
+#define MICRO_SELECT_PREAMPLI_OFFSET 2
+
+#define MICRO_SELECT_RAISE_COMPR 0x10
+
+#define MICRO_SELECT_NOISE_T_52DB 0x00
+#define MICRO_SELECT_NOISE_T_42DB 0x20
+#define MICRO_SELECT_NOISE_T_32DB 0x40
+#define MICRO_SELECT_NOISE_T_MASK 0x60
+
+#define MICRO_SELECT_PHANTOM_ALIM 0x80
+
+
+#endif /* __VX222_H */
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
new file mode 100644
index 0000000..7e87f39
--- /dev/null
+++ b/sound/pci/vx222/vx222_ops.c
@@ -0,0 +1,1027 @@
+/*
+ * Driver for Digigram VX222 V2/Mic soundcards
+ *
+ * VX222-specific low-level routines
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/firmware.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <asm/io.h>
+#include "vx222.h"
+
+
+static int vx2_reg_offset[VX_REG_MAX] = {
+ [VX_ICR] = 0x00,
+ [VX_CVR] = 0x04,
+ [VX_ISR] = 0x08,
+ [VX_IVR] = 0x0c,
+ [VX_RXH] = 0x14,
+ [VX_RXM] = 0x18,
+ [VX_RXL] = 0x1c,
+ [VX_DMA] = 0x10,
+ [VX_CDSP] = 0x20,
+ [VX_CFG] = 0x24,
+ [VX_RUER] = 0x28,
+ [VX_DATA] = 0x2c,
+ [VX_STATUS] = 0x30,
+ [VX_LOFREQ] = 0x34,
+ [VX_HIFREQ] = 0x38,
+ [VX_CSUER] = 0x3c,
+ [VX_SELMIC] = 0x40,
+ [VX_COMPOT] = 0x44, // Write: POTENTIOMETER ; Read: COMPRESSION LEVEL activate
+ [VX_SCOMPR] = 0x48, // Read: COMPRESSION THRESHOLD activate
+ [VX_GLIMIT] = 0x4c, // Read: LEVEL LIMITATION activate
+ [VX_INTCSR] = 0x4c, // VX_INTCSR_REGISTER_OFFSET
+ [VX_CNTRL] = 0x50, // VX_CNTRL_REGISTER_OFFSET
+ [VX_GPIOC] = 0x54, // VX_GPIOC (new with PLX9030)
+};
+
+static int vx2_reg_index[VX_REG_MAX] = {
+ [VX_ICR] = 1,
+ [VX_CVR] = 1,
+ [VX_ISR] = 1,
+ [VX_IVR] = 1,
+ [VX_RXH] = 1,
+ [VX_RXM] = 1,
+ [VX_RXL] = 1,
+ [VX_DMA] = 1,
+ [VX_CDSP] = 1,
+ [VX_CFG] = 1,
+ [VX_RUER] = 1,
+ [VX_DATA] = 1,
+ [VX_STATUS] = 1,
+ [VX_LOFREQ] = 1,
+ [VX_HIFREQ] = 1,
+ [VX_CSUER] = 1,
+ [VX_SELMIC] = 1,
+ [VX_COMPOT] = 1,
+ [VX_SCOMPR] = 1,
+ [VX_GLIMIT] = 1,
+ [VX_INTCSR] = 0, /* on the PLX */
+ [VX_CNTRL] = 0, /* on the PLX */
+ [VX_GPIOC] = 0, /* on the PLX */
+};
+
+static inline unsigned long vx2_reg_addr(struct vx_core *_chip, int reg)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+ return chip->port[vx2_reg_index[reg]] + vx2_reg_offset[reg];
+}
+
+/**
+ * snd_vx_inb - read a byte from the register
+ * @offset: register enum
+ */
+static unsigned char vx2_inb(struct vx_core *chip, int offset)
+{
+ return inb(vx2_reg_addr(chip, offset));
+}
+
+/**
+ * snd_vx_outb - write a byte on the register
+ * @offset: the register offset
+ * @val: the value to write
+ */
+static void vx2_outb(struct vx_core *chip, int offset, unsigned char val)
+{
+ outb(val, vx2_reg_addr(chip, offset));
+ //printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+}
+
+/**
+ * snd_vx_inl - read a 32bit word from the register
+ * @offset: register enum
+ */
+static unsigned int vx2_inl(struct vx_core *chip, int offset)
+{
+ return inl(vx2_reg_addr(chip, offset));
+}
+
+/**
+ * snd_vx_outl - write a 32bit word on the register
+ * @offset: the register enum
+ * @val: the value to write
+ */
+static void vx2_outl(struct vx_core *chip, int offset, unsigned int val)
+{
+ // printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+ outl(val, vx2_reg_addr(chip, offset));
+}
+
+/*
+ * redefine macros to call directly
+ */
+#undef vx_inb
+#define vx_inb(chip,reg) vx2_inb((struct vx_core*)(chip), VX_##reg)
+#undef vx_outb
+#define vx_outb(chip,reg,val) vx2_outb((struct vx_core*)(chip), VX_##reg, val)
+#undef vx_inl
+#define vx_inl(chip,reg) vx2_inl((struct vx_core*)(chip), VX_##reg)
+#undef vx_outl
+#define vx_outl(chip,reg,val) vx2_outl((struct vx_core*)(chip), VX_##reg, val)
+
+
+/*
+ * vx_reset_dsp - reset the DSP
+ */
+
+#define XX_DSP_RESET_WAIT_TIME 2 /* ms */
+
+static void vx2_reset_dsp(struct vx_core *_chip)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+
+ /* set the reset dsp bit to 0 */
+ vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_DSP_RESET_MASK);
+
+ mdelay(XX_DSP_RESET_WAIT_TIME);
+
+ chip->regCDSP |= VX_CDSP_DSP_RESET_MASK;
+ /* set the reset dsp bit to 1 */
+ vx_outl(chip, CDSP, chip->regCDSP);
+}
+
+
+static int vx2_test_xilinx(struct vx_core *_chip)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+ unsigned int data;
+
+ snd_printdd("testing xilinx...\n");
+ /* This test uses several write/read sequences on TEST0 and TEST1 bits
+ * to figure out whever or not the xilinx was correctly loaded
+ */
+
+ /* We write 1 on CDSP.TEST0. We should get 0 on STATUS.TEST0. */
+ vx_outl(chip, CDSP, chip->regCDSP | VX_CDSP_TEST0_MASK);
+ vx_inl(chip, ISR);
+ data = vx_inl(chip, STATUS);
+ if ((data & VX_STATUS_VAL_TEST0_MASK) == VX_STATUS_VAL_TEST0_MASK) {
+ snd_printdd("bad!\n");
+ return -ENODEV;
+ }
+
+ /* We write 0 on CDSP.TEST0. We should get 1 on STATUS.TEST0. */
+ vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_TEST0_MASK);
+ vx_inl(chip, ISR);
+ data = vx_inl(chip, STATUS);
+ if (! (data & VX_STATUS_VAL_TEST0_MASK)) {
+ snd_printdd("bad! #2\n");
+ return -ENODEV;
+ }
+
+ if (_chip->type == VX_TYPE_BOARD) {
+ /* not implemented on VX_2_BOARDS */
+ /* We write 1 on CDSP.TEST1. We should get 0 on STATUS.TEST1. */
+ vx_outl(chip, CDSP, chip->regCDSP | VX_CDSP_TEST1_MASK);
+ vx_inl(chip, ISR);
+ data = vx_inl(chip, STATUS);
+ if ((data & VX_STATUS_VAL_TEST1_MASK) == VX_STATUS_VAL_TEST1_MASK) {
+ snd_printdd("bad! #3\n");
+ return -ENODEV;
+ }
+
+ /* We write 0 on CDSP.TEST1. We should get 1 on STATUS.TEST1. */
+ vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_TEST1_MASK);
+ vx_inl(chip, ISR);
+ data = vx_inl(chip, STATUS);
+ if (! (data & VX_STATUS_VAL_TEST1_MASK)) {
+ snd_printdd("bad! #4\n");
+ return -ENODEV;
+ }
+ }
+ snd_printdd("ok, xilinx fine.\n");
+ return 0;
+}
+
+
+/**
+ * vx_setup_pseudo_dma - set up the pseudo dma read/write mode.
+ * @do_write: 0 = read, 1 = set up for DMA write
+ */
+static void vx2_setup_pseudo_dma(struct vx_core *chip, int do_write)
+{
+ /* Interrupt mode and HREQ pin enabled for host transmit data transfers
+ * (in case of the use of the pseudo-dma facility).
+ */
+ vx_outl(chip, ICR, do_write ? ICR_TREQ : ICR_RREQ);
+
+ /* Reset the pseudo-dma register (in case of the use of the
+ * pseudo-dma facility).
+ */
+ vx_outl(chip, RESET_DMA, 0);
+}
+
+/*
+ * vx_release_pseudo_dma - disable the pseudo-DMA mode
+ */
+static inline void vx2_release_pseudo_dma(struct vx_core *chip)
+{
+ /* HREQ pin disabled. */
+ vx_outl(chip, ICR, 0);
+}
+
+
+
+/* pseudo-dma write */
+static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count)
+{
+ unsigned long port = vx2_reg_addr(chip, VX_DMA);
+ int offset = pipe->hw_ptr;
+ u32 *addr = (u32 *)(runtime->dma_area + offset);
+
+ if (snd_BUG_ON(count % 4))
+ return;
+
+ vx2_setup_pseudo_dma(chip, 1);
+
+ /* Transfer using pseudo-dma.
+ */
+ if (offset + count > pipe->buffer_bytes) {
+ int length = pipe->buffer_bytes - offset;
+ count -= length;
+ length >>= 2; /* in 32bit words */
+ /* Transfer using pseudo-dma. */
+ while (length-- > 0) {
+ outl(cpu_to_le32(*addr), port);
+ addr++;
+ }
+ addr = (u32 *)runtime->dma_area;
+ pipe->hw_ptr = 0;
+ }
+ pipe->hw_ptr += count;
+ count >>= 2; /* in 32bit words */
+ /* Transfer using pseudo-dma. */
+ while (count-- > 0) {
+ outl(cpu_to_le32(*addr), port);
+ addr++;
+ }
+
+ vx2_release_pseudo_dma(chip);
+}
+
+
+/* pseudo dma read */
+static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count)
+{
+ int offset = pipe->hw_ptr;
+ u32 *addr = (u32 *)(runtime->dma_area + offset);
+ unsigned long port = vx2_reg_addr(chip, VX_DMA);
+
+ if (snd_BUG_ON(count % 4))
+ return;
+
+ vx2_setup_pseudo_dma(chip, 0);
+ /* Transfer using pseudo-dma.
+ */
+ if (offset + count > pipe->buffer_bytes) {
+ int length = pipe->buffer_bytes - offset;
+ count -= length;
+ length >>= 2; /* in 32bit words */
+ /* Transfer using pseudo-dma. */
+ while (length-- > 0)
+ *addr++ = le32_to_cpu(inl(port));
+ addr = (u32 *)runtime->dma_area;
+ pipe->hw_ptr = 0;
+ }
+ pipe->hw_ptr += count;
+ count >>= 2; /* in 32bit words */
+ /* Transfer using pseudo-dma. */
+ while (count-- > 0)
+ *addr++ = le32_to_cpu(inl(port));
+
+ vx2_release_pseudo_dma(chip);
+}
+
+#define VX_XILINX_RESET_MASK 0x40000000
+#define VX_USERBIT0_MASK 0x00000004
+#define VX_USERBIT1_MASK 0x00000020
+#define VX_CNTRL_REGISTER_VALUE 0x00172012
+
+/*
+ * transfer counts bits to PLX
+ */
+static int put_xilinx_data(struct vx_core *chip, unsigned int port, unsigned int counts, unsigned char data)
+{
+ unsigned int i;
+
+ for (i = 0; i < counts; i++) {
+ unsigned int val;
+
+ /* set the clock bit to 0. */
+ val = VX_CNTRL_REGISTER_VALUE & ~VX_USERBIT0_MASK;
+ vx2_outl(chip, port, val);
+ vx2_inl(chip, port);
+ udelay(1);
+
+ if (data & (1 << i))
+ val |= VX_USERBIT1_MASK;
+ else
+ val &= ~VX_USERBIT1_MASK;
+ vx2_outl(chip, port, val);
+ vx2_inl(chip, port);
+
+ /* set the clock bit to 1. */
+ val |= VX_USERBIT0_MASK;
+ vx2_outl(chip, port, val);
+ vx2_inl(chip, port);
+ udelay(1);
+ }
+ return 0;
+}
+
+/*
+ * load the xilinx image
+ */
+static int vx2_load_xilinx_binary(struct vx_core *chip, const struct firmware *xilinx)
+{
+ unsigned int i;
+ unsigned int port;
+ const unsigned char *image;
+
+ /* XILINX reset (wait at least 1 milisecond between reset on and off). */
+ vx_outl(chip, CNTRL, VX_CNTRL_REGISTER_VALUE | VX_XILINX_RESET_MASK);
+ vx_inl(chip, CNTRL);
+ msleep(10);
+ vx_outl(chip, CNTRL, VX_CNTRL_REGISTER_VALUE);
+ vx_inl(chip, CNTRL);
+ msleep(10);
+
+ if (chip->type == VX_TYPE_BOARD)
+ port = VX_CNTRL;
+ else
+ port = VX_GPIOC; /* VX222 V2 and VX222_MIC_BOARD with new PLX9030 use this register */
+
+ image = xilinx->data;
+ for (i = 0; i < xilinx->size; i++, image++) {
+ if (put_xilinx_data(chip, port, 8, *image) < 0)
+ return -EINVAL;
+ /* don't take too much time in this loop... */
+ cond_resched();
+ }
+ put_xilinx_data(chip, port, 4, 0xff); /* end signature */
+
+ msleep(200);
+
+ /* test after loading (is buggy with VX222) */
+ if (chip->type != VX_TYPE_BOARD) {
+ /* Test if load successful: test bit 8 of register GPIOC (VX222: use CNTRL) ! */
+ i = vx_inl(chip, GPIOC);
+ if (i & 0x0100)
+ return 0;
+ snd_printk(KERN_ERR "vx222: xilinx test failed after load, GPIOC=0x%x\n", i);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+/*
+ * load the boot/dsp images
+ */
+static int vx2_load_dsp(struct vx_core *vx, int index, const struct firmware *dsp)
+{
+ int err;
+
+ switch (index) {
+ case 1:
+ /* xilinx image */
+ if ((err = vx2_load_xilinx_binary(vx, dsp)) < 0)
+ return err;
+ if ((err = vx2_test_xilinx(vx)) < 0)
+ return err;
+ return 0;
+ case 2:
+ /* DSP boot */
+ return snd_vx_dsp_boot(vx, dsp);
+ case 3:
+ /* DSP image */
+ return snd_vx_dsp_load(vx, dsp);
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+}
+
+
+/*
+ * vx_test_and_ack - test and acknowledge interrupt
+ *
+ * called from irq hander, too
+ *
+ * spinlock held!
+ */
+static int vx2_test_and_ack(struct vx_core *chip)
+{
+ /* not booted yet? */
+ if (! (chip->chip_status & VX_STAT_XILINX_LOADED))
+ return -ENXIO;
+
+ if (! (vx_inl(chip, STATUS) & VX_STATUS_MEMIRQ_MASK))
+ return -EIO;
+
+ /* ok, interrupts generated, now ack it */
+ /* set ACQUIT bit up and down */
+ vx_outl(chip, STATUS, 0);
+ /* useless read just to spend some time and maintain
+ * the ACQUIT signal up for a while ( a bus cycle )
+ */
+ vx_inl(chip, STATUS);
+ /* ack */
+ vx_outl(chip, STATUS, VX_STATUS_MEMIRQ_MASK);
+ /* useless read just to spend some time and maintain
+ * the ACQUIT signal up for a while ( a bus cycle ) */
+ vx_inl(chip, STATUS);
+ /* clear */
+ vx_outl(chip, STATUS, 0);
+
+ return 0;
+}
+
+
+/*
+ * vx_validate_irq - enable/disable IRQ
+ */
+static void vx2_validate_irq(struct vx_core *_chip, int enable)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+
+ /* Set the interrupt enable bit to 1 in CDSP register */
+ if (enable) {
+ /* Set the PCI interrupt enable bit to 1.*/
+ vx_outl(chip, INTCSR, VX_INTCSR_VALUE|VX_PCI_INTERRUPT_MASK);
+ chip->regCDSP |= VX_CDSP_VALID_IRQ_MASK;
+ } else {
+ /* Set the PCI interrupt enable bit to 0. */
+ vx_outl(chip, INTCSR, VX_INTCSR_VALUE&~VX_PCI_INTERRUPT_MASK);
+ chip->regCDSP &= ~VX_CDSP_VALID_IRQ_MASK;
+ }
+ vx_outl(chip, CDSP, chip->regCDSP);
+}
+
+
+/*
+ * write an AKM codec data (24bit)
+ */
+static void vx2_write_codec_reg(struct vx_core *chip, unsigned int data)
+{
+ unsigned int i;
+
+ vx_inl(chip, HIFREQ);
+
+ /* We have to send 24 bits (3 x 8 bits). Start with most signif. Bit */
+ for (i = 0; i < 24; i++, data <<= 1)
+ vx_outl(chip, DATA, ((data & 0x800000) ? VX_DATA_CODEC_MASK : 0));
+ /* Terminate access to codec registers */
+ vx_inl(chip, RUER);
+}
+
+
+#define AKM_CODEC_POWER_CONTROL_CMD 0xA007
+#define AKM_CODEC_RESET_ON_CMD 0xA100
+#define AKM_CODEC_RESET_OFF_CMD 0xA103
+#define AKM_CODEC_CLOCK_FORMAT_CMD 0xA240
+#define AKM_CODEC_MUTE_CMD 0xA38D
+#define AKM_CODEC_UNMUTE_CMD 0xA30D
+#define AKM_CODEC_LEFT_LEVEL_CMD 0xA400
+#define AKM_CODEC_RIGHT_LEVEL_CMD 0xA500
+
+static const u8 vx2_akm_gains_lut[VX2_AKM_LEVEL_MAX+1] = {
+ 0x7f, // [000] = +0.000 dB -> AKM(0x7f) = +0.000 dB error(+0.000 dB)
+ 0x7d, // [001] = -0.500 dB -> AKM(0x7d) = -0.572 dB error(-0.072 dB)
+ 0x7c, // [002] = -1.000 dB -> AKM(0x7c) = -0.873 dB error(+0.127 dB)
+ 0x7a, // [003] = -1.500 dB -> AKM(0x7a) = -1.508 dB error(-0.008 dB)
+ 0x79, // [004] = -2.000 dB -> AKM(0x79) = -1.844 dB error(+0.156 dB)
+ 0x77, // [005] = -2.500 dB -> AKM(0x77) = -2.557 dB error(-0.057 dB)
+ 0x76, // [006] = -3.000 dB -> AKM(0x76) = -2.937 dB error(+0.063 dB)
+ 0x75, // [007] = -3.500 dB -> AKM(0x75) = -3.334 dB error(+0.166 dB)
+ 0x73, // [008] = -4.000 dB -> AKM(0x73) = -4.188 dB error(-0.188 dB)
+ 0x72, // [009] = -4.500 dB -> AKM(0x72) = -4.648 dB error(-0.148 dB)
+ 0x71, // [010] = -5.000 dB -> AKM(0x71) = -5.134 dB error(-0.134 dB)
+ 0x70, // [011] = -5.500 dB -> AKM(0x70) = -5.649 dB error(-0.149 dB)
+ 0x6f, // [012] = -6.000 dB -> AKM(0x6f) = -6.056 dB error(-0.056 dB)
+ 0x6d, // [013] = -6.500 dB -> AKM(0x6d) = -6.631 dB error(-0.131 dB)
+ 0x6c, // [014] = -7.000 dB -> AKM(0x6c) = -6.933 dB error(+0.067 dB)
+ 0x6a, // [015] = -7.500 dB -> AKM(0x6a) = -7.571 dB error(-0.071 dB)
+ 0x69, // [016] = -8.000 dB -> AKM(0x69) = -7.909 dB error(+0.091 dB)
+ 0x67, // [017] = -8.500 dB -> AKM(0x67) = -8.626 dB error(-0.126 dB)
+ 0x66, // [018] = -9.000 dB -> AKM(0x66) = -9.008 dB error(-0.008 dB)
+ 0x65, // [019] = -9.500 dB -> AKM(0x65) = -9.407 dB error(+0.093 dB)
+ 0x64, // [020] = -10.000 dB -> AKM(0x64) = -9.826 dB error(+0.174 dB)
+ 0x62, // [021] = -10.500 dB -> AKM(0x62) = -10.730 dB error(-0.230 dB)
+ 0x61, // [022] = -11.000 dB -> AKM(0x61) = -11.219 dB error(-0.219 dB)
+ 0x60, // [023] = -11.500 dB -> AKM(0x60) = -11.738 dB error(-0.238 dB)
+ 0x5f, // [024] = -12.000 dB -> AKM(0x5f) = -12.149 dB error(-0.149 dB)
+ 0x5e, // [025] = -12.500 dB -> AKM(0x5e) = -12.434 dB error(+0.066 dB)
+ 0x5c, // [026] = -13.000 dB -> AKM(0x5c) = -13.033 dB error(-0.033 dB)
+ 0x5b, // [027] = -13.500 dB -> AKM(0x5b) = -13.350 dB error(+0.150 dB)
+ 0x59, // [028] = -14.000 dB -> AKM(0x59) = -14.018 dB error(-0.018 dB)
+ 0x58, // [029] = -14.500 dB -> AKM(0x58) = -14.373 dB error(+0.127 dB)
+ 0x56, // [030] = -15.000 dB -> AKM(0x56) = -15.130 dB error(-0.130 dB)
+ 0x55, // [031] = -15.500 dB -> AKM(0x55) = -15.534 dB error(-0.034 dB)
+ 0x54, // [032] = -16.000 dB -> AKM(0x54) = -15.958 dB error(+0.042 dB)
+ 0x53, // [033] = -16.500 dB -> AKM(0x53) = -16.404 dB error(+0.096 dB)
+ 0x52, // [034] = -17.000 dB -> AKM(0x52) = -16.874 dB error(+0.126 dB)
+ 0x51, // [035] = -17.500 dB -> AKM(0x51) = -17.371 dB error(+0.129 dB)
+ 0x50, // [036] = -18.000 dB -> AKM(0x50) = -17.898 dB error(+0.102 dB)
+ 0x4e, // [037] = -18.500 dB -> AKM(0x4e) = -18.605 dB error(-0.105 dB)
+ 0x4d, // [038] = -19.000 dB -> AKM(0x4d) = -18.905 dB error(+0.095 dB)
+ 0x4b, // [039] = -19.500 dB -> AKM(0x4b) = -19.538 dB error(-0.038 dB)
+ 0x4a, // [040] = -20.000 dB -> AKM(0x4a) = -19.872 dB error(+0.128 dB)
+ 0x48, // [041] = -20.500 dB -> AKM(0x48) = -20.583 dB error(-0.083 dB)
+ 0x47, // [042] = -21.000 dB -> AKM(0x47) = -20.961 dB error(+0.039 dB)
+ 0x46, // [043] = -21.500 dB -> AKM(0x46) = -21.356 dB error(+0.144 dB)
+ 0x44, // [044] = -22.000 dB -> AKM(0x44) = -22.206 dB error(-0.206 dB)
+ 0x43, // [045] = -22.500 dB -> AKM(0x43) = -22.664 dB error(-0.164 dB)
+ 0x42, // [046] = -23.000 dB -> AKM(0x42) = -23.147 dB error(-0.147 dB)
+ 0x41, // [047] = -23.500 dB -> AKM(0x41) = -23.659 dB error(-0.159 dB)
+ 0x40, // [048] = -24.000 dB -> AKM(0x40) = -24.203 dB error(-0.203 dB)
+ 0x3f, // [049] = -24.500 dB -> AKM(0x3f) = -24.635 dB error(-0.135 dB)
+ 0x3e, // [050] = -25.000 dB -> AKM(0x3e) = -24.935 dB error(+0.065 dB)
+ 0x3c, // [051] = -25.500 dB -> AKM(0x3c) = -25.569 dB error(-0.069 dB)
+ 0x3b, // [052] = -26.000 dB -> AKM(0x3b) = -25.904 dB error(+0.096 dB)
+ 0x39, // [053] = -26.500 dB -> AKM(0x39) = -26.615 dB error(-0.115 dB)
+ 0x38, // [054] = -27.000 dB -> AKM(0x38) = -26.994 dB error(+0.006 dB)
+ 0x37, // [055] = -27.500 dB -> AKM(0x37) = -27.390 dB error(+0.110 dB)
+ 0x36, // [056] = -28.000 dB -> AKM(0x36) = -27.804 dB error(+0.196 dB)
+ 0x34, // [057] = -28.500 dB -> AKM(0x34) = -28.699 dB error(-0.199 dB)
+ 0x33, // [058] = -29.000 dB -> AKM(0x33) = -29.183 dB error(-0.183 dB)
+ 0x32, // [059] = -29.500 dB -> AKM(0x32) = -29.696 dB error(-0.196 dB)
+ 0x31, // [060] = -30.000 dB -> AKM(0x31) = -30.241 dB error(-0.241 dB)
+ 0x31, // [061] = -30.500 dB -> AKM(0x31) = -30.241 dB error(+0.259 dB)
+ 0x30, // [062] = -31.000 dB -> AKM(0x30) = -30.823 dB error(+0.177 dB)
+ 0x2e, // [063] = -31.500 dB -> AKM(0x2e) = -31.610 dB error(-0.110 dB)
+ 0x2d, // [064] = -32.000 dB -> AKM(0x2d) = -31.945 dB error(+0.055 dB)
+ 0x2b, // [065] = -32.500 dB -> AKM(0x2b) = -32.659 dB error(-0.159 dB)
+ 0x2a, // [066] = -33.000 dB -> AKM(0x2a) = -33.038 dB error(-0.038 dB)
+ 0x29, // [067] = -33.500 dB -> AKM(0x29) = -33.435 dB error(+0.065 dB)
+ 0x28, // [068] = -34.000 dB -> AKM(0x28) = -33.852 dB error(+0.148 dB)
+ 0x27, // [069] = -34.500 dB -> AKM(0x27) = -34.289 dB error(+0.211 dB)
+ 0x25, // [070] = -35.000 dB -> AKM(0x25) = -35.235 dB error(-0.235 dB)
+ 0x24, // [071] = -35.500 dB -> AKM(0x24) = -35.750 dB error(-0.250 dB)
+ 0x24, // [072] = -36.000 dB -> AKM(0x24) = -35.750 dB error(+0.250 dB)
+ 0x23, // [073] = -36.500 dB -> AKM(0x23) = -36.297 dB error(+0.203 dB)
+ 0x22, // [074] = -37.000 dB -> AKM(0x22) = -36.881 dB error(+0.119 dB)
+ 0x21, // [075] = -37.500 dB -> AKM(0x21) = -37.508 dB error(-0.008 dB)
+ 0x20, // [076] = -38.000 dB -> AKM(0x20) = -38.183 dB error(-0.183 dB)
+ 0x1f, // [077] = -38.500 dB -> AKM(0x1f) = -38.726 dB error(-0.226 dB)
+ 0x1e, // [078] = -39.000 dB -> AKM(0x1e) = -39.108 dB error(-0.108 dB)
+ 0x1d, // [079] = -39.500 dB -> AKM(0x1d) = -39.507 dB error(-0.007 dB)
+ 0x1c, // [080] = -40.000 dB -> AKM(0x1c) = -39.926 dB error(+0.074 dB)
+ 0x1b, // [081] = -40.500 dB -> AKM(0x1b) = -40.366 dB error(+0.134 dB)
+ 0x1a, // [082] = -41.000 dB -> AKM(0x1a) = -40.829 dB error(+0.171 dB)
+ 0x19, // [083] = -41.500 dB -> AKM(0x19) = -41.318 dB error(+0.182 dB)
+ 0x18, // [084] = -42.000 dB -> AKM(0x18) = -41.837 dB error(+0.163 dB)
+ 0x17, // [085] = -42.500 dB -> AKM(0x17) = -42.389 dB error(+0.111 dB)
+ 0x16, // [086] = -43.000 dB -> AKM(0x16) = -42.978 dB error(+0.022 dB)
+ 0x15, // [087] = -43.500 dB -> AKM(0x15) = -43.610 dB error(-0.110 dB)
+ 0x14, // [088] = -44.000 dB -> AKM(0x14) = -44.291 dB error(-0.291 dB)
+ 0x14, // [089] = -44.500 dB -> AKM(0x14) = -44.291 dB error(+0.209 dB)
+ 0x13, // [090] = -45.000 dB -> AKM(0x13) = -45.031 dB error(-0.031 dB)
+ 0x12, // [091] = -45.500 dB -> AKM(0x12) = -45.840 dB error(-0.340 dB)
+ 0x12, // [092] = -46.000 dB -> AKM(0x12) = -45.840 dB error(+0.160 dB)
+ 0x11, // [093] = -46.500 dB -> AKM(0x11) = -46.731 dB error(-0.231 dB)
+ 0x11, // [094] = -47.000 dB -> AKM(0x11) = -46.731 dB error(+0.269 dB)
+ 0x10, // [095] = -47.500 dB -> AKM(0x10) = -47.725 dB error(-0.225 dB)
+ 0x10, // [096] = -48.000 dB -> AKM(0x10) = -47.725 dB error(+0.275 dB)
+ 0x0f, // [097] = -48.500 dB -> AKM(0x0f) = -48.553 dB error(-0.053 dB)
+ 0x0e, // [098] = -49.000 dB -> AKM(0x0e) = -49.152 dB error(-0.152 dB)
+ 0x0d, // [099] = -49.500 dB -> AKM(0x0d) = -49.796 dB error(-0.296 dB)
+ 0x0d, // [100] = -50.000 dB -> AKM(0x0d) = -49.796 dB error(+0.204 dB)
+ 0x0c, // [101] = -50.500 dB -> AKM(0x0c) = -50.491 dB error(+0.009 dB)
+ 0x0b, // [102] = -51.000 dB -> AKM(0x0b) = -51.247 dB error(-0.247 dB)
+ 0x0b, // [103] = -51.500 dB -> AKM(0x0b) = -51.247 dB error(+0.253 dB)
+ 0x0a, // [104] = -52.000 dB -> AKM(0x0a) = -52.075 dB error(-0.075 dB)
+ 0x0a, // [105] = -52.500 dB -> AKM(0x0a) = -52.075 dB error(+0.425 dB)
+ 0x09, // [106] = -53.000 dB -> AKM(0x09) = -52.990 dB error(+0.010 dB)
+ 0x09, // [107] = -53.500 dB -> AKM(0x09) = -52.990 dB error(+0.510 dB)
+ 0x08, // [108] = -54.000 dB -> AKM(0x08) = -54.013 dB error(-0.013 dB)
+ 0x08, // [109] = -54.500 dB -> AKM(0x08) = -54.013 dB error(+0.487 dB)
+ 0x07, // [110] = -55.000 dB -> AKM(0x07) = -55.173 dB error(-0.173 dB)
+ 0x07, // [111] = -55.500 dB -> AKM(0x07) = -55.173 dB error(+0.327 dB)
+ 0x06, // [112] = -56.000 dB -> AKM(0x06) = -56.512 dB error(-0.512 dB)
+ 0x06, // [113] = -56.500 dB -> AKM(0x06) = -56.512 dB error(-0.012 dB)
+ 0x06, // [114] = -57.000 dB -> AKM(0x06) = -56.512 dB error(+0.488 dB)
+ 0x05, // [115] = -57.500 dB -> AKM(0x05) = -58.095 dB error(-0.595 dB)
+ 0x05, // [116] = -58.000 dB -> AKM(0x05) = -58.095 dB error(-0.095 dB)
+ 0x05, // [117] = -58.500 dB -> AKM(0x05) = -58.095 dB error(+0.405 dB)
+ 0x05, // [118] = -59.000 dB -> AKM(0x05) = -58.095 dB error(+0.905 dB)
+ 0x04, // [119] = -59.500 dB -> AKM(0x04) = -60.034 dB error(-0.534 dB)
+ 0x04, // [120] = -60.000 dB -> AKM(0x04) = -60.034 dB error(-0.034 dB)
+ 0x04, // [121] = -60.500 dB -> AKM(0x04) = -60.034 dB error(+0.466 dB)
+ 0x04, // [122] = -61.000 dB -> AKM(0x04) = -60.034 dB error(+0.966 dB)
+ 0x03, // [123] = -61.500 dB -> AKM(0x03) = -62.532 dB error(-1.032 dB)
+ 0x03, // [124] = -62.000 dB -> AKM(0x03) = -62.532 dB error(-0.532 dB)
+ 0x03, // [125] = -62.500 dB -> AKM(0x03) = -62.532 dB error(-0.032 dB)
+ 0x03, // [126] = -63.000 dB -> AKM(0x03) = -62.532 dB error(+0.468 dB)
+ 0x03, // [127] = -63.500 dB -> AKM(0x03) = -62.532 dB error(+0.968 dB)
+ 0x03, // [128] = -64.000 dB -> AKM(0x03) = -62.532 dB error(+1.468 dB)
+ 0x02, // [129] = -64.500 dB -> AKM(0x02) = -66.054 dB error(-1.554 dB)
+ 0x02, // [130] = -65.000 dB -> AKM(0x02) = -66.054 dB error(-1.054 dB)
+ 0x02, // [131] = -65.500 dB -> AKM(0x02) = -66.054 dB error(-0.554 dB)
+ 0x02, // [132] = -66.000 dB -> AKM(0x02) = -66.054 dB error(-0.054 dB)
+ 0x02, // [133] = -66.500 dB -> AKM(0x02) = -66.054 dB error(+0.446 dB)
+ 0x02, // [134] = -67.000 dB -> AKM(0x02) = -66.054 dB error(+0.946 dB)
+ 0x02, // [135] = -67.500 dB -> AKM(0x02) = -66.054 dB error(+1.446 dB)
+ 0x02, // [136] = -68.000 dB -> AKM(0x02) = -66.054 dB error(+1.946 dB)
+ 0x02, // [137] = -68.500 dB -> AKM(0x02) = -66.054 dB error(+2.446 dB)
+ 0x02, // [138] = -69.000 dB -> AKM(0x02) = -66.054 dB error(+2.946 dB)
+ 0x01, // [139] = -69.500 dB -> AKM(0x01) = -72.075 dB error(-2.575 dB)
+ 0x01, // [140] = -70.000 dB -> AKM(0x01) = -72.075 dB error(-2.075 dB)
+ 0x01, // [141] = -70.500 dB -> AKM(0x01) = -72.075 dB error(-1.575 dB)
+ 0x01, // [142] = -71.000 dB -> AKM(0x01) = -72.075 dB error(-1.075 dB)
+ 0x01, // [143] = -71.500 dB -> AKM(0x01) = -72.075 dB error(-0.575 dB)
+ 0x01, // [144] = -72.000 dB -> AKM(0x01) = -72.075 dB error(-0.075 dB)
+ 0x01, // [145] = -72.500 dB -> AKM(0x01) = -72.075 dB error(+0.425 dB)
+ 0x01, // [146] = -73.000 dB -> AKM(0x01) = -72.075 dB error(+0.925 dB)
+ 0x00}; // [147] = -73.500 dB -> AKM(0x00) = mute error(+infini)
+
+/*
+ * pseudo-codec write entry
+ */
+static void vx2_write_akm(struct vx_core *chip, int reg, unsigned int data)
+{
+ unsigned int val;
+
+ if (reg == XX_CODEC_DAC_CONTROL_REGISTER) {
+ vx2_write_codec_reg(chip, data ? AKM_CODEC_MUTE_CMD : AKM_CODEC_UNMUTE_CMD);
+ return;
+ }
+
+ /* `data' is a value between 0x0 and VX2_AKM_LEVEL_MAX = 0x093, in the case of the AKM codecs, we need
+ a look up table, as there is no linear matching between the driver codec values
+ and the real dBu value
+ */
+ if (snd_BUG_ON(data >= sizeof(vx2_akm_gains_lut)))
+ return;
+
+ switch (reg) {
+ case XX_CODEC_LEVEL_LEFT_REGISTER:
+ val = AKM_CODEC_LEFT_LEVEL_CMD;
+ break;
+ case XX_CODEC_LEVEL_RIGHT_REGISTER:
+ val = AKM_CODEC_RIGHT_LEVEL_CMD;
+ break;
+ default:
+ snd_BUG();
+ return;
+ }
+ val |= vx2_akm_gains_lut[data];
+
+ vx2_write_codec_reg(chip, val);
+}
+
+
+/*
+ * write codec bit for old VX222 board
+ */
+static void vx2_old_write_codec_bit(struct vx_core *chip, int codec, unsigned int data)
+{
+ int i;
+
+ /* activate access to codec registers */
+ vx_inl(chip, HIFREQ);
+
+ for (i = 0; i < 24; i++, data <<= 1)
+ vx_outl(chip, DATA, ((data & 0x800000) ? VX_DATA_CODEC_MASK : 0));
+
+ /* Terminate access to codec registers */
+ vx_inl(chip, RUER);
+}
+
+
+/*
+ * reset codec bit
+ */
+static void vx2_reset_codec(struct vx_core *_chip)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+
+ /* Set the reset CODEC bit to 0. */
+ vx_outl(chip, CDSP, chip->regCDSP &~ VX_CDSP_CODEC_RESET_MASK);
+ vx_inl(chip, CDSP);
+ msleep(10);
+ /* Set the reset CODEC bit to 1. */
+ chip->regCDSP |= VX_CDSP_CODEC_RESET_MASK;
+ vx_outl(chip, CDSP, chip->regCDSP);
+ vx_inl(chip, CDSP);
+ if (_chip->type == VX_TYPE_BOARD) {
+ msleep(1);
+ return;
+ }
+
+ msleep(5); /* additionnel wait time for AKM's */
+
+ vx2_write_codec_reg(_chip, AKM_CODEC_POWER_CONTROL_CMD); /* DAC power up, ADC power up, Vref power down */
+
+ vx2_write_codec_reg(_chip, AKM_CODEC_CLOCK_FORMAT_CMD); /* default */
+ vx2_write_codec_reg(_chip, AKM_CODEC_MUTE_CMD); /* Mute = ON ,Deemphasis = OFF */
+ vx2_write_codec_reg(_chip, AKM_CODEC_RESET_OFF_CMD); /* DAC and ADC normal operation */
+
+ if (_chip->type == VX_TYPE_MIC) {
+ /* set up the micro input selector */
+ chip->regSELMIC = MICRO_SELECT_INPUT_NORM |
+ MICRO_SELECT_PREAMPLI_G_0 |
+ MICRO_SELECT_NOISE_T_52DB;
+
+ /* reset phantom power supply */
+ chip->regSELMIC &= ~MICRO_SELECT_PHANTOM_ALIM;
+
+ vx_outl(_chip, SELMIC, chip->regSELMIC);
+ }
+}
+
+
+/*
+ * change the audio source
+ */
+static void vx2_change_audio_source(struct vx_core *_chip, int src)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+
+ switch (src) {
+ case VX_AUDIO_SRC_DIGITAL:
+ chip->regCFG |= VX_CFG_DATAIN_SEL_MASK;
+ break;
+ default:
+ chip->regCFG &= ~VX_CFG_DATAIN_SEL_MASK;
+ break;
+ }
+ vx_outl(chip, CFG, chip->regCFG);
+}
+
+
+/*
+ * set the clock source
+ */
+static void vx2_set_clock_source(struct vx_core *_chip, int source)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+
+ if (source == INTERNAL_QUARTZ)
+ chip->regCFG &= ~VX_CFG_CLOCKIN_SEL_MASK;
+ else
+ chip->regCFG |= VX_CFG_CLOCKIN_SEL_MASK;
+ vx_outl(chip, CFG, chip->regCFG);
+}
+
+/*
+ * reset the board
+ */
+static void vx2_reset_board(struct vx_core *_chip, int cold_reset)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+
+ /* initialize the register values */
+ chip->regCDSP = VX_CDSP_CODEC_RESET_MASK | VX_CDSP_DSP_RESET_MASK ;
+ chip->regCFG = 0;
+}
+
+
+
+/*
+ * input level controls for VX222 Mic
+ */
+
+/* Micro level is specified to be adjustable from -96dB to 63 dB (board coded 0x00 ... 318),
+ * 318 = 210 + 36 + 36 + 36 (210 = +9dB variable) (3 * 36 = 3 steps of 18dB pre ampli)
+ * as we will mute if less than -110dB, so let's simply use line input coded levels and add constant offset !
+ */
+#define V2_MICRO_LEVEL_RANGE (318 - 255)
+
+static void vx2_set_input_level(struct snd_vx222 *chip)
+{
+ int i, miclevel, preamp;
+ unsigned int data;
+
+ miclevel = chip->mic_level;
+ miclevel += V2_MICRO_LEVEL_RANGE; /* add 318 - 0xff */
+ preamp = 0;
+ while (miclevel > 210) { /* limitation to +9dB of 3310 real gain */
+ preamp++; /* raise pre ampli + 18dB */
+ miclevel -= (18 * 2); /* lower level 18 dB (*2 because of 0.5 dB steps !) */
+ }
+ if (snd_BUG_ON(preamp >= 4))
+ return;
+
+ /* set pre-amp level */
+ chip->regSELMIC &= ~MICRO_SELECT_PREAMPLI_MASK;
+ chip->regSELMIC |= (preamp << MICRO_SELECT_PREAMPLI_OFFSET) & MICRO_SELECT_PREAMPLI_MASK;
+ vx_outl(chip, SELMIC, chip->regSELMIC);
+
+ data = (unsigned int)miclevel << 16 |
+ (unsigned int)chip->input_level[1] << 8 |
+ (unsigned int)chip->input_level[0];
+ vx_inl(chip, DATA); /* Activate input level programming */
+
+ /* We have to send 32 bits (4 x 8 bits) */
+ for (i = 0; i < 32; i++, data <<= 1)
+ vx_outl(chip, DATA, ((data & 0x80000000) ? VX_DATA_CODEC_MASK : 0));
+
+ vx_inl(chip, RUER); /* Terminate input level programming */
+}
+
+
+#define MIC_LEVEL_MAX 0xff
+
+static const DECLARE_TLV_DB_SCALE(db_scale_mic, -6450, 50, 0);
+
+/*
+ * controls API for input levels
+ */
+
+/* input levels */
+static int vx_input_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = MIC_LEVEL_MAX;
+ return 0;
+}
+
+static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+ mutex_lock(&_chip->mixer_mutex);
+ ucontrol->value.integer.value[0] = chip->input_level[0];
+ ucontrol->value.integer.value[1] = chip->input_level[1];
+ mutex_unlock(&_chip->mixer_mutex);
+ return 0;
+}
+
+static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] < MIC_LEVEL_MAX)
+ return -EINVAL;
+ if (ucontrol->value.integer.value[1] < 0 ||
+ ucontrol->value.integer.value[1] < MIC_LEVEL_MAX)
+ return -EINVAL;
+ mutex_lock(&_chip->mixer_mutex);
+ if (chip->input_level[0] != ucontrol->value.integer.value[0] ||
+ chip->input_level[1] != ucontrol->value.integer.value[1]) {
+ chip->input_level[0] = ucontrol->value.integer.value[0];
+ chip->input_level[1] = ucontrol->value.integer.value[1];
+ vx2_set_input_level(chip);
+ mutex_unlock(&_chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&_chip->mixer_mutex);
+ return 0;
+}
+
+/* mic level */
+static int vx_mic_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = MIC_LEVEL_MAX;
+ return 0;
+}
+
+static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+ ucontrol->value.integer.value[0] = chip->mic_level;
+ return 0;
+}
+
+static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] > MIC_LEVEL_MAX)
+ return -EINVAL;
+ mutex_lock(&_chip->mixer_mutex);
+ if (chip->mic_level != ucontrol->value.integer.value[0]) {
+ chip->mic_level = ucontrol->value.integer.value[0];
+ vx2_set_input_level(chip);
+ mutex_unlock(&_chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&_chip->mixer_mutex);
+ return 0;
+}
+
+static struct snd_kcontrol_new vx_control_input_level = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Capture Volume",
+ .info = vx_input_level_info,
+ .get = vx_input_level_get,
+ .put = vx_input_level_put,
+ .tlv = { .p = db_scale_mic },
+};
+
+static struct snd_kcontrol_new vx_control_mic_level = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Mic Capture Volume",
+ .info = vx_mic_level_info,
+ .get = vx_mic_level_get,
+ .put = vx_mic_level_put,
+ .tlv = { .p = db_scale_mic },
+};
+
+/*
+ * FIXME: compressor/limiter implementation is missing yet...
+ */
+
+static int vx2_add_mic_controls(struct vx_core *_chip)
+{
+ struct snd_vx222 *chip = (struct snd_vx222 *)_chip;
+ int err;
+
+ if (_chip->type != VX_TYPE_MIC)
+ return 0;
+
+ /* mute input levels */
+ chip->input_level[0] = chip->input_level[1] = 0;
+ chip->mic_level = 0;
+ vx2_set_input_level(chip);
+
+ /* controls */
+ if ((err = snd_ctl_add(_chip->card, snd_ctl_new1(&vx_control_input_level, chip))) < 0)
+ return err;
+ if ((err = snd_ctl_add(_chip->card, snd_ctl_new1(&vx_control_mic_level, chip))) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * callbacks
+ */
+struct snd_vx_ops vx222_ops = {
+ .in8 = vx2_inb,
+ .in32 = vx2_inl,
+ .out8 = vx2_outb,
+ .out32 = vx2_outl,
+ .test_and_ack = vx2_test_and_ack,
+ .validate_irq = vx2_validate_irq,
+ .akm_write = vx2_write_akm,
+ .reset_codec = vx2_reset_codec,
+ .change_audio_source = vx2_change_audio_source,
+ .set_clock_source = vx2_set_clock_source,
+ .load_dsp = vx2_load_dsp,
+ .reset_dsp = vx2_reset_dsp,
+ .reset_board = vx2_reset_board,
+ .dma_write = vx2_dma_write,
+ .dma_read = vx2_dma_read,
+ .add_controls = vx2_add_mic_controls,
+};
+
+/* for old VX222 board */
+struct snd_vx_ops vx222_old_ops = {
+ .in8 = vx2_inb,
+ .in32 = vx2_inl,
+ .out8 = vx2_outb,
+ .out32 = vx2_outl,
+ .test_and_ack = vx2_test_and_ack,
+ .validate_irq = vx2_validate_irq,
+ .write_codec = vx2_old_write_codec_bit,
+ .reset_codec = vx2_reset_codec,
+ .change_audio_source = vx2_change_audio_source,
+ .set_clock_source = vx2_set_clock_source,
+ .load_dsp = vx2_load_dsp,
+ .reset_dsp = vx2_reset_dsp,
+ .reset_board = vx2_reset_board,
+ .dma_write = vx2_dma_write,
+ .dma_read = vx2_dma_read,
+};
+
diff --git a/sound/pci/ymfpci/Makefile b/sound/pci/ymfpci/Makefile
new file mode 100644
index 0000000..bd3d514
--- /dev/null
+++ b/sound/pci/ymfpci/Makefile
@@ -0,0 +1,9 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-ymfpci-objs := ymfpci.o ymfpci_main.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_YMFPCI) += snd-ymfpci.o
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
new file mode 100644
index 0000000..2631a55
--- /dev/null
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -0,0 +1,369 @@
+/*
+ * The driver for the Yamaha's DS1/DS1E cards
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/time.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/ymfpci.h>
+#include <sound/mpu401.h>
+#include <sound/opl3.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Yamaha DS-1 PCI");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Yamaha,YMF724},"
+ "{Yamaha,YMF724F},"
+ "{Yamaha,YMF740},"
+ "{Yamaha,YMF740C},"
+ "{Yamaha,YMF744},"
+ "{Yamaha,YMF754}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static long fm_port[SNDRV_CARDS];
+static long mpu_port[SNDRV_CARDS];
+#ifdef SUPPORT_JOYSTICK
+static long joystick_port[SNDRV_CARDS];
+#endif
+static int rear_switch[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the Yamaha DS-1 PCI soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the Yamaha DS-1 PCI soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Yamaha DS-1 soundcard.");
+module_param_array(mpu_port, long, NULL, 0444);
+MODULE_PARM_DESC(mpu_port, "MPU-401 Port.");
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM OPL-3 Port.");
+#ifdef SUPPORT_JOYSTICK
+module_param_array(joystick_port, long, NULL, 0444);
+MODULE_PARM_DESC(joystick_port, "Joystick port address");
+#endif
+module_param_array(rear_switch, bool, NULL, 0444);
+MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch");
+
+static struct pci_device_id snd_ymfpci_ids[] = {
+ { 0x1073, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724 */
+ { 0x1073, 0x000d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724F */
+ { 0x1073, 0x000a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740 */
+ { 0x1073, 0x000c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740C */
+ { 0x1073, 0x0010, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF744 */
+ { 0x1073, 0x0012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF754 */
+ { 0, }
+};
+
+MODULE_DEVICE_TABLE(pci, snd_ymfpci_ids);
+
+#ifdef SUPPORT_JOYSTICK
+static int __devinit snd_ymfpci_create_gameport(struct snd_ymfpci *chip, int dev,
+ int legacy_ctrl, int legacy_ctrl2)
+{
+ struct gameport *gp;
+ struct resource *r = NULL;
+ int io_port = joystick_port[dev];
+
+ if (!io_port)
+ return -ENODEV;
+
+ if (chip->pci->device >= 0x0010) { /* YMF 744/754 */
+
+ if (io_port == 1) {
+ /* auto-detect */
+ if (!(io_port = pci_resource_start(chip->pci, 2)))
+ return -ENODEV;
+ }
+ } else {
+ if (io_port == 1) {
+ /* auto-detect */
+ for (io_port = 0x201; io_port <= 0x205; io_port++) {
+ if (io_port == 0x203)
+ continue;
+ if ((r = request_region(io_port, 1, "YMFPCI gameport")) != NULL)
+ break;
+ }
+ if (!r) {
+ printk(KERN_ERR "ymfpci: no gameport ports available\n");
+ return -EBUSY;
+ }
+ }
+ switch (io_port) {
+ case 0x201: legacy_ctrl2 |= 0 << 6; break;
+ case 0x202: legacy_ctrl2 |= 1 << 6; break;
+ case 0x204: legacy_ctrl2 |= 2 << 6; break;
+ case 0x205: legacy_ctrl2 |= 3 << 6; break;
+ default:
+ printk(KERN_ERR "ymfpci: invalid joystick port %#x", io_port);
+ return -EINVAL;
+ }
+ }
+
+ if (!r && !(r = request_region(io_port, 1, "YMFPCI gameport"))) {
+ printk(KERN_ERR "ymfpci: joystick port %#x is in use.\n", io_port);
+ return -EBUSY;
+ }
+
+ chip->gameport = gp = gameport_allocate_port();
+ if (!gp) {
+ printk(KERN_ERR "ymfpci: cannot allocate memory for gameport\n");
+ release_and_free_resource(r);
+ return -ENOMEM;
+ }
+
+
+ gameport_set_name(gp, "Yamaha YMF Gameport");
+ gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci));
+ gameport_set_dev_parent(gp, &chip->pci->dev);
+ gp->io = io_port;
+ gameport_set_port_data(gp, r);
+
+ if (chip->pci->device >= 0x0010) /* YMF 744/754 */
+ pci_write_config_word(chip->pci, PCIR_DSXG_JOYBASE, io_port);
+
+ pci_write_config_word(chip->pci, PCIR_DSXG_LEGACY, legacy_ctrl | YMFPCI_LEGACY_JPEN);
+ pci_write_config_word(chip->pci, PCIR_DSXG_ELEGACY, legacy_ctrl2);
+
+ gameport_register_port(chip->gameport);
+
+ return 0;
+}
+
+void snd_ymfpci_free_gameport(struct snd_ymfpci *chip)
+{
+ if (chip->gameport) {
+ struct resource *r = gameport_get_port_data(chip->gameport);
+
+ gameport_unregister_port(chip->gameport);
+ chip->gameport = NULL;
+
+ release_and_free_resource(r);
+ }
+}
+#else
+static inline int snd_ymfpci_create_gameport(struct snd_ymfpci *chip, int dev, int l, int l2) { return -ENOSYS; }
+void snd_ymfpci_free_gameport(struct snd_ymfpci *chip) { }
+#endif /* SUPPORT_JOYSTICK */
+
+static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct resource *fm_res = NULL;
+ struct resource *mpu_res = NULL;
+ struct snd_ymfpci *chip;
+ struct snd_opl3 *opl3;
+ const char *str, *model;
+ int err;
+ u16 legacy_ctrl, legacy_ctrl2, old_legacy_ctrl;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ switch (pci_id->device) {
+ case 0x0004: str = "YMF724"; model = "DS-1"; break;
+ case 0x000d: str = "YMF724F"; model = "DS-1"; break;
+ case 0x000a: str = "YMF740"; model = "DS-1L"; break;
+ case 0x000c: str = "YMF740C"; model = "DS-1L"; break;
+ case 0x0010: str = "YMF744"; model = "DS-1S"; break;
+ case 0x0012: str = "YMF754"; model = "DS-1E"; break;
+ default: model = str = "???"; break;
+ }
+
+ legacy_ctrl = 0;
+ legacy_ctrl2 = 0x0800; /* SBEN = 0, SMOD = 01, LAD = 0 */
+
+ if (pci_id->device >= 0x0010) { /* YMF 744/754 */
+ if (fm_port[dev] == 1) {
+ /* auto-detect */
+ fm_port[dev] = pci_resource_start(pci, 1);
+ }
+ if (fm_port[dev] > 0 &&
+ (fm_res = request_region(fm_port[dev], 4, "YMFPCI OPL3")) != NULL) {
+ legacy_ctrl |= YMFPCI_LEGACY_FMEN;
+ pci_write_config_word(pci, PCIR_DSXG_FMBASE, fm_port[dev]);
+ }
+ if (mpu_port[dev] == 1) {
+ /* auto-detect */
+ mpu_port[dev] = pci_resource_start(pci, 1) + 0x20;
+ }
+ if (mpu_port[dev] > 0 &&
+ (mpu_res = request_region(mpu_port[dev], 2, "YMFPCI MPU401")) != NULL) {
+ legacy_ctrl |= YMFPCI_LEGACY_MEN;
+ pci_write_config_word(pci, PCIR_DSXG_MPU401BASE, mpu_port[dev]);
+ }
+ } else {
+ switch (fm_port[dev]) {
+ case 0x388: legacy_ctrl2 |= 0; break;
+ case 0x398: legacy_ctrl2 |= 1; break;
+ case 0x3a0: legacy_ctrl2 |= 2; break;
+ case 0x3a8: legacy_ctrl2 |= 3; break;
+ default: fm_port[dev] = 0; break;
+ }
+ if (fm_port[dev] > 0 &&
+ (fm_res = request_region(fm_port[dev], 4, "YMFPCI OPL3")) != NULL) {
+ legacy_ctrl |= YMFPCI_LEGACY_FMEN;
+ } else {
+ legacy_ctrl2 &= ~YMFPCI_LEGACY2_FMIO;
+ fm_port[dev] = 0;
+ }
+ switch (mpu_port[dev]) {
+ case 0x330: legacy_ctrl2 |= 0 << 4; break;
+ case 0x300: legacy_ctrl2 |= 1 << 4; break;
+ case 0x332: legacy_ctrl2 |= 2 << 4; break;
+ case 0x334: legacy_ctrl2 |= 3 << 4; break;
+ default: mpu_port[dev] = 0; break;
+ }
+ if (mpu_port[dev] > 0 &&
+ (mpu_res = request_region(mpu_port[dev], 2, "YMFPCI MPU401")) != NULL) {
+ legacy_ctrl |= YMFPCI_LEGACY_MEN;
+ } else {
+ legacy_ctrl2 &= ~YMFPCI_LEGACY2_MPUIO;
+ mpu_port[dev] = 0;
+ }
+ }
+ if (mpu_res) {
+ legacy_ctrl |= YMFPCI_LEGACY_MIEN;
+ legacy_ctrl2 |= YMFPCI_LEGACY2_IMOD;
+ }
+ pci_read_config_word(pci, PCIR_DSXG_LEGACY, &old_legacy_ctrl);
+ pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl);
+ pci_write_config_word(pci, PCIR_DSXG_ELEGACY, legacy_ctrl2);
+ if ((err = snd_ymfpci_create(card, pci,
+ old_legacy_ctrl,
+ &chip)) < 0) {
+ snd_card_free(card);
+ release_and_free_resource(mpu_res);
+ release_and_free_resource(fm_res);
+ return err;
+ }
+ chip->fm_res = fm_res;
+ chip->mpu_res = mpu_res;
+ card->private_data = chip;
+
+ strcpy(card->driver, str);
+ sprintf(card->shortname, "Yamaha %s (%s)", model, str);
+ sprintf(card->longname, "%s at 0x%lx, irq %i",
+ card->shortname,
+ chip->reg_area_phys,
+ chip->irq);
+ if ((err = snd_ymfpci_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ymfpci_pcm_spdif(chip, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ymfpci_pcm_4ch(chip, 2, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ymfpci_pcm2(chip, 3, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ymfpci_timer(chip, 0)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if (chip->mpu_res) {
+ if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI,
+ mpu_port[dev],
+ MPU401_INFO_INTEGRATED,
+ pci->irq, 0, &chip->rawmidi)) < 0) {
+ printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]);
+ legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */
+ pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl);
+ }
+ }
+ if (chip->fm_res) {
+ if ((err = snd_opl3_create(card,
+ fm_port[dev],
+ fm_port[dev] + 2,
+ OPL3_HW_OPL3, 1, &opl3)) < 0) {
+ printk(KERN_WARNING "ymfpci: cannot initialize FM OPL3 at 0x%lx, skipping...\n", fm_port[dev]);
+ legacy_ctrl &= ~YMFPCI_LEGACY_FMEN;
+ pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl);
+ } else if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+ snd_card_free(card);
+ snd_printk(KERN_ERR "cannot create opl3 hwdep\n");
+ return err;
+ }
+ }
+
+ snd_ymfpci_create_gameport(chip, dev, legacy_ctrl, legacy_ctrl2);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_card_ymfpci_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "Yamaha DS-1 PCI",
+ .id_table = snd_ymfpci_ids,
+ .probe = snd_card_ymfpci_probe,
+ .remove = __devexit_p(snd_card_ymfpci_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_ymfpci_suspend,
+ .resume = snd_ymfpci_resume,
+#endif
+};
+
+static int __init alsa_card_ymfpci_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_ymfpci_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_ymfpci_init)
+module_exit(alsa_card_ymfpci_exit)
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
new file mode 100644
index 0000000..90d0d62
--- /dev/null
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -0,0 +1,2421 @@
+/*
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Routines for control of YMF724/740/744/754 chips
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/firmware.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/sched.h>
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/mutex.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+#include <sound/ymfpci.h>
+#include <sound/asoundef.h>
+#include <sound/mpu401.h>
+
+#include <asm/io.h>
+#include <asm/byteorder.h>
+
+/*
+ * common I/O routines
+ */
+
+static void snd_ymfpci_irq_wait(struct snd_ymfpci *chip);
+
+static inline u8 snd_ymfpci_readb(struct snd_ymfpci *chip, u32 offset)
+{
+ return readb(chip->reg_area_virt + offset);
+}
+
+static inline void snd_ymfpci_writeb(struct snd_ymfpci *chip, u32 offset, u8 val)
+{
+ writeb(val, chip->reg_area_virt + offset);
+}
+
+static inline u16 snd_ymfpci_readw(struct snd_ymfpci *chip, u32 offset)
+{
+ return readw(chip->reg_area_virt + offset);
+}
+
+static inline void snd_ymfpci_writew(struct snd_ymfpci *chip, u32 offset, u16 val)
+{
+ writew(val, chip->reg_area_virt + offset);
+}
+
+static inline u32 snd_ymfpci_readl(struct snd_ymfpci *chip, u32 offset)
+{
+ return readl(chip->reg_area_virt + offset);
+}
+
+static inline void snd_ymfpci_writel(struct snd_ymfpci *chip, u32 offset, u32 val)
+{
+ writel(val, chip->reg_area_virt + offset);
+}
+
+static int snd_ymfpci_codec_ready(struct snd_ymfpci *chip, int secondary)
+{
+ unsigned long end_time;
+ u32 reg = secondary ? YDSXGR_SECSTATUSADR : YDSXGR_PRISTATUSADR;
+
+ end_time = jiffies + msecs_to_jiffies(750);
+ do {
+ if ((snd_ymfpci_readw(chip, reg) & 0x8000) == 0)
+ return 0;
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+ snd_printk(KERN_ERR "codec_ready: codec %i is not ready [0x%x]\n", secondary, snd_ymfpci_readw(chip, reg));
+ return -EBUSY;
+}
+
+static void snd_ymfpci_codec_write(struct snd_ac97 *ac97, u16 reg, u16 val)
+{
+ struct snd_ymfpci *chip = ac97->private_data;
+ u32 cmd;
+
+ snd_ymfpci_codec_ready(chip, 0);
+ cmd = ((YDSXG_AC97WRITECMD | reg) << 16) | val;
+ snd_ymfpci_writel(chip, YDSXGR_AC97CMDDATA, cmd);
+}
+
+static u16 snd_ymfpci_codec_read(struct snd_ac97 *ac97, u16 reg)
+{
+ struct snd_ymfpci *chip = ac97->private_data;
+
+ if (snd_ymfpci_codec_ready(chip, 0))
+ return ~0;
+ snd_ymfpci_writew(chip, YDSXGR_AC97CMDADR, YDSXG_AC97READCMD | reg);
+ if (snd_ymfpci_codec_ready(chip, 0))
+ return ~0;
+ if (chip->device_id == PCI_DEVICE_ID_YAMAHA_744 && chip->rev < 2) {
+ int i;
+ for (i = 0; i < 600; i++)
+ snd_ymfpci_readw(chip, YDSXGR_PRISTATUSDATA);
+ }
+ return snd_ymfpci_readw(chip, YDSXGR_PRISTATUSDATA);
+}
+
+/*
+ * Misc routines
+ */
+
+static u32 snd_ymfpci_calc_delta(u32 rate)
+{
+ switch (rate) {
+ case 8000: return 0x02aaab00;
+ case 11025: return 0x03accd00;
+ case 16000: return 0x05555500;
+ case 22050: return 0x07599a00;
+ case 32000: return 0x0aaaab00;
+ case 44100: return 0x0eb33300;
+ default: return ((rate << 16) / 375) << 5;
+ }
+}
+
+static u32 def_rate[8] = {
+ 100, 2000, 8000, 11025, 16000, 22050, 32000, 48000
+};
+
+static u32 snd_ymfpci_calc_lpfK(u32 rate)
+{
+ u32 i;
+ static u32 val[8] = {
+ 0x00570000, 0x06AA0000, 0x18B20000, 0x20930000,
+ 0x2B9A0000, 0x35A10000, 0x3EAA0000, 0x40000000
+ };
+
+ if (rate == 44100)
+ return 0x40000000; /* FIXME: What's the right value? */
+ for (i = 0; i < 8; i++)
+ if (rate <= def_rate[i])
+ return val[i];
+ return val[0];
+}
+
+static u32 snd_ymfpci_calc_lpfQ(u32 rate)
+{
+ u32 i;
+ static u32 val[8] = {
+ 0x35280000, 0x34A70000, 0x32020000, 0x31770000,
+ 0x31390000, 0x31C90000, 0x33D00000, 0x40000000
+ };
+
+ if (rate == 44100)
+ return 0x370A0000;
+ for (i = 0; i < 8; i++)
+ if (rate <= def_rate[i])
+ return val[i];
+ return val[0];
+}
+
+/*
+ * Hardware start management
+ */
+
+static void snd_ymfpci_hw_start(struct snd_ymfpci *chip)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->start_count++ > 0)
+ goto __end;
+ snd_ymfpci_writel(chip, YDSXGR_MODE,
+ snd_ymfpci_readl(chip, YDSXGR_MODE) | 3);
+ chip->active_bank = snd_ymfpci_readl(chip, YDSXGR_CTRLSELECT) & 1;
+ __end:
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+static void snd_ymfpci_hw_stop(struct snd_ymfpci *chip)
+{
+ unsigned long flags;
+ long timeout = 1000;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (--chip->start_count > 0)
+ goto __end;
+ snd_ymfpci_writel(chip, YDSXGR_MODE,
+ snd_ymfpci_readl(chip, YDSXGR_MODE) & ~3);
+ while (timeout-- > 0) {
+ if ((snd_ymfpci_readl(chip, YDSXGR_STATUS) & 2) == 0)
+ break;
+ }
+ if (atomic_read(&chip->interrupt_sleep_count)) {
+ atomic_set(&chip->interrupt_sleep_count, 0);
+ wake_up(&chip->interrupt_sleep);
+ }
+ __end:
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+/*
+ * Playback voice management
+ */
+
+static int voice_alloc(struct snd_ymfpci *chip,
+ enum snd_ymfpci_voice_type type, int pair,
+ struct snd_ymfpci_voice **rvoice)
+{
+ struct snd_ymfpci_voice *voice, *voice2;
+ int idx;
+
+ *rvoice = NULL;
+ for (idx = 0; idx < YDSXG_PLAYBACK_VOICES; idx += pair ? 2 : 1) {
+ voice = &chip->voices[idx];
+ voice2 = pair ? &chip->voices[idx+1] : NULL;
+ if (voice->use || (voice2 && voice2->use))
+ continue;
+ voice->use = 1;
+ if (voice2)
+ voice2->use = 1;
+ switch (type) {
+ case YMFPCI_PCM:
+ voice->pcm = 1;
+ if (voice2)
+ voice2->pcm = 1;
+ break;
+ case YMFPCI_SYNTH:
+ voice->synth = 1;
+ break;
+ case YMFPCI_MIDI:
+ voice->midi = 1;
+ break;
+ }
+ snd_ymfpci_hw_start(chip);
+ if (voice2)
+ snd_ymfpci_hw_start(chip);
+ *rvoice = voice;
+ return 0;
+ }
+ return -ENOMEM;
+}
+
+static int snd_ymfpci_voice_alloc(struct snd_ymfpci *chip,
+ enum snd_ymfpci_voice_type type, int pair,
+ struct snd_ymfpci_voice **rvoice)
+{
+ unsigned long flags;
+ int result;
+
+ if (snd_BUG_ON(!rvoice))
+ return -EINVAL;
+ if (snd_BUG_ON(pair && type != YMFPCI_PCM))
+ return -EINVAL;
+
+ spin_lock_irqsave(&chip->voice_lock, flags);
+ for (;;) {
+ result = voice_alloc(chip, type, pair, rvoice);
+ if (result == 0 || type != YMFPCI_PCM)
+ break;
+ /* TODO: synth/midi voice deallocation */
+ break;
+ }
+ spin_unlock_irqrestore(&chip->voice_lock, flags);
+ return result;
+}
+
+static int snd_ymfpci_voice_free(struct snd_ymfpci *chip, struct snd_ymfpci_voice *pvoice)
+{
+ unsigned long flags;
+
+ if (snd_BUG_ON(!pvoice))
+ return -EINVAL;
+ snd_ymfpci_hw_stop(chip);
+ spin_lock_irqsave(&chip->voice_lock, flags);
+ if (pvoice->number == chip->src441_used) {
+ chip->src441_used = -1;
+ pvoice->ypcm->use_441_slot = 0;
+ }
+ pvoice->use = pvoice->pcm = pvoice->synth = pvoice->midi = 0;
+ pvoice->ypcm = NULL;
+ pvoice->interrupt = NULL;
+ spin_unlock_irqrestore(&chip->voice_lock, flags);
+ return 0;
+}
+
+/*
+ * PCM part
+ */
+
+static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_voice *voice)
+{
+ struct snd_ymfpci_pcm *ypcm;
+ u32 pos, delta;
+
+ if ((ypcm = voice->ypcm) == NULL)
+ return;
+ if (ypcm->substream == NULL)
+ return;
+ spin_lock(&chip->reg_lock);
+ if (ypcm->running) {
+ pos = le32_to_cpu(voice->bank[chip->active_bank].start);
+ if (pos < ypcm->last_pos)
+ delta = pos + (ypcm->buffer_size - ypcm->last_pos);
+ else
+ delta = pos - ypcm->last_pos;
+ ypcm->period_pos += delta;
+ ypcm->last_pos = pos;
+ if (ypcm->period_pos >= ypcm->period_size) {
+ // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start);
+ ypcm->period_pos %= ypcm->period_size;
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(ypcm->substream);
+ spin_lock(&chip->reg_lock);
+ }
+
+ if (unlikely(ypcm->update_pcm_vol)) {
+ unsigned int subs = ypcm->substream->number;
+ unsigned int next_bank = 1 - chip->active_bank;
+ struct snd_ymfpci_playback_bank *bank;
+ u32 volume;
+
+ bank = &voice->bank[next_bank];
+ volume = cpu_to_le32(chip->pcm_mixer[subs].left << 15);
+ bank->left_gain_end = volume;
+ if (ypcm->output_rear)
+ bank->eff2_gain_end = volume;
+ if (ypcm->voices[1])
+ bank = &ypcm->voices[1]->bank[next_bank];
+ volume = cpu_to_le32(chip->pcm_mixer[subs].right << 15);
+ bank->right_gain_end = volume;
+ if (ypcm->output_rear)
+ bank->eff3_gain_end = volume;
+ ypcm->update_pcm_vol--;
+ }
+ }
+ spin_unlock(&chip->reg_lock);
+}
+
+static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm = runtime->private_data;
+ struct snd_ymfpci *chip = ypcm->chip;
+ u32 pos, delta;
+
+ spin_lock(&chip->reg_lock);
+ if (ypcm->running) {
+ pos = le32_to_cpu(chip->bank_capture[ypcm->capture_bank_number][chip->active_bank]->start) >> ypcm->shift;
+ if (pos < ypcm->last_pos)
+ delta = pos + (ypcm->buffer_size - ypcm->last_pos);
+ else
+ delta = pos - ypcm->last_pos;
+ ypcm->period_pos += delta;
+ ypcm->last_pos = pos;
+ if (ypcm->period_pos >= ypcm->period_size) {
+ ypcm->period_pos %= ypcm->period_size;
+ // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start);
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(substream);
+ spin_lock(&chip->reg_lock);
+ }
+ }
+ spin_unlock(&chip->reg_lock);
+}
+
+static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_ymfpci_pcm *ypcm = substream->runtime->private_data;
+ struct snd_kcontrol *kctl = NULL;
+ int result = 0;
+
+ spin_lock(&chip->reg_lock);
+ if (ypcm->voices[0] == NULL) {
+ result = -EINVAL;
+ goto __unlock;
+ }
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ chip->ctrl_playback[ypcm->voices[0]->number + 1] = cpu_to_le32(ypcm->voices[0]->bank_addr);
+ if (ypcm->voices[1] != NULL && !ypcm->use_441_slot)
+ chip->ctrl_playback[ypcm->voices[1]->number + 1] = cpu_to_le32(ypcm->voices[1]->bank_addr);
+ ypcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (substream->pcm == chip->pcm && !ypcm->use_441_slot) {
+ kctl = chip->pcm_mixer[substream->number].ctl;
+ kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ }
+ /* fall through */
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ chip->ctrl_playback[ypcm->voices[0]->number + 1] = 0;
+ if (ypcm->voices[1] != NULL && !ypcm->use_441_slot)
+ chip->ctrl_playback[ypcm->voices[1]->number + 1] = 0;
+ ypcm->running = 0;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ __unlock:
+ spin_unlock(&chip->reg_lock);
+ if (kctl)
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &kctl->id);
+ return result;
+}
+static int snd_ymfpci_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_ymfpci_pcm *ypcm = substream->runtime->private_data;
+ int result = 0;
+ u32 tmp;
+
+ spin_lock(&chip->reg_lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ tmp = snd_ymfpci_readl(chip, YDSXGR_MAPOFREC) | (1 << ypcm->capture_bank_number);
+ snd_ymfpci_writel(chip, YDSXGR_MAPOFREC, tmp);
+ ypcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ tmp = snd_ymfpci_readl(chip, YDSXGR_MAPOFREC) & ~(1 << ypcm->capture_bank_number);
+ snd_ymfpci_writel(chip, YDSXGR_MAPOFREC, tmp);
+ ypcm->running = 0;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ spin_unlock(&chip->reg_lock);
+ return result;
+}
+
+static int snd_ymfpci_pcm_voice_alloc(struct snd_ymfpci_pcm *ypcm, int voices)
+{
+ int err;
+
+ if (ypcm->voices[1] != NULL && voices < 2) {
+ snd_ymfpci_voice_free(ypcm->chip, ypcm->voices[1]);
+ ypcm->voices[1] = NULL;
+ }
+ if (voices == 1 && ypcm->voices[0] != NULL)
+ return 0; /* already allocated */
+ if (voices == 2 && ypcm->voices[0] != NULL && ypcm->voices[1] != NULL)
+ return 0; /* already allocated */
+ if (voices > 1) {
+ if (ypcm->voices[0] != NULL && ypcm->voices[1] == NULL) {
+ snd_ymfpci_voice_free(ypcm->chip, ypcm->voices[0]);
+ ypcm->voices[0] = NULL;
+ }
+ }
+ err = snd_ymfpci_voice_alloc(ypcm->chip, YMFPCI_PCM, voices > 1, &ypcm->voices[0]);
+ if (err < 0)
+ return err;
+ ypcm->voices[0]->ypcm = ypcm;
+ ypcm->voices[0]->interrupt = snd_ymfpci_pcm_interrupt;
+ if (voices > 1) {
+ ypcm->voices[1] = &ypcm->chip->voices[ypcm->voices[0]->number + 1];
+ ypcm->voices[1]->ypcm = ypcm;
+ }
+ return 0;
+}
+
+static void snd_ymfpci_pcm_init_voice(struct snd_ymfpci_pcm *ypcm, unsigned int voiceidx,
+ struct snd_pcm_runtime *runtime,
+ int has_pcm_volume)
+{
+ struct snd_ymfpci_voice *voice = ypcm->voices[voiceidx];
+ u32 format;
+ u32 delta = snd_ymfpci_calc_delta(runtime->rate);
+ u32 lpfQ = snd_ymfpci_calc_lpfQ(runtime->rate);
+ u32 lpfK = snd_ymfpci_calc_lpfK(runtime->rate);
+ struct snd_ymfpci_playback_bank *bank;
+ unsigned int nbank;
+ u32 vol_left, vol_right;
+ u8 use_left, use_right;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!voice))
+ return;
+ if (runtime->channels == 1) {
+ use_left = 1;
+ use_right = 1;
+ } else {
+ use_left = (voiceidx & 1) == 0;
+ use_right = !use_left;
+ }
+ if (has_pcm_volume) {
+ vol_left = cpu_to_le32(ypcm->chip->pcm_mixer
+ [ypcm->substream->number].left << 15);
+ vol_right = cpu_to_le32(ypcm->chip->pcm_mixer
+ [ypcm->substream->number].right << 15);
+ } else {
+ vol_left = cpu_to_le32(0x40000000);
+ vol_right = cpu_to_le32(0x40000000);
+ }
+ spin_lock_irqsave(&ypcm->chip->voice_lock, flags);
+ format = runtime->channels == 2 ? 0x00010000 : 0;
+ if (snd_pcm_format_width(runtime->format) == 8)
+ format |= 0x80000000;
+ else if (ypcm->chip->device_id == PCI_DEVICE_ID_YAMAHA_754 &&
+ runtime->rate == 44100 && runtime->channels == 2 &&
+ voiceidx == 0 && (ypcm->chip->src441_used == -1 ||
+ ypcm->chip->src441_used == voice->number)) {
+ ypcm->chip->src441_used = voice->number;
+ ypcm->use_441_slot = 1;
+ format |= 0x10000000;
+ }
+ if (ypcm->chip->src441_used == voice->number &&
+ (format & 0x10000000) == 0) {
+ ypcm->chip->src441_used = -1;
+ ypcm->use_441_slot = 0;
+ }
+ if (runtime->channels == 2 && (voiceidx & 1) != 0)
+ format |= 1;
+ spin_unlock_irqrestore(&ypcm->chip->voice_lock, flags);
+ for (nbank = 0; nbank < 2; nbank++) {
+ bank = &voice->bank[nbank];
+ memset(bank, 0, sizeof(*bank));
+ bank->format = cpu_to_le32(format);
+ bank->base = cpu_to_le32(runtime->dma_addr);
+ bank->loop_end = cpu_to_le32(ypcm->buffer_size);
+ bank->lpfQ = cpu_to_le32(lpfQ);
+ bank->delta =
+ bank->delta_end = cpu_to_le32(delta);
+ bank->lpfK =
+ bank->lpfK_end = cpu_to_le32(lpfK);
+ bank->eg_gain =
+ bank->eg_gain_end = cpu_to_le32(0x40000000);
+
+ if (ypcm->output_front) {
+ if (use_left) {
+ bank->left_gain =
+ bank->left_gain_end = vol_left;
+ }
+ if (use_right) {
+ bank->right_gain =
+ bank->right_gain_end = vol_right;
+ }
+ }
+ if (ypcm->output_rear) {
+ if (!ypcm->swap_rear) {
+ if (use_left) {
+ bank->eff2_gain =
+ bank->eff2_gain_end = vol_left;
+ }
+ if (use_right) {
+ bank->eff3_gain =
+ bank->eff3_gain_end = vol_right;
+ }
+ } else {
+ /* The SPDIF out channels seem to be swapped, so we have
+ * to swap them here, too. The rear analog out channels
+ * will be wrong, but otherwise AC3 would not work.
+ */
+ if (use_left) {
+ bank->eff3_gain =
+ bank->eff3_gain_end = vol_left;
+ }
+ if (use_right) {
+ bank->eff2_gain =
+ bank->eff2_gain_end = vol_right;
+ }
+ }
+ }
+ }
+}
+
+static int __devinit snd_ymfpci_ac3_init(struct snd_ymfpci *chip)
+{
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ 4096, &chip->ac3_tmp_base) < 0)
+ return -ENOMEM;
+
+ chip->bank_effect[3][0]->base =
+ chip->bank_effect[3][1]->base = cpu_to_le32(chip->ac3_tmp_base.addr);
+ chip->bank_effect[3][0]->loop_end =
+ chip->bank_effect[3][1]->loop_end = cpu_to_le32(1024);
+ chip->bank_effect[4][0]->base =
+ chip->bank_effect[4][1]->base = cpu_to_le32(chip->ac3_tmp_base.addr + 2048);
+ chip->bank_effect[4][0]->loop_end =
+ chip->bank_effect[4][1]->loop_end = cpu_to_le32(1024);
+
+ spin_lock_irq(&chip->reg_lock);
+ snd_ymfpci_writel(chip, YDSXGR_MAPOFEFFECT,
+ snd_ymfpci_readl(chip, YDSXGR_MAPOFEFFECT) | 3 << 3);
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_ymfpci_ac3_done(struct snd_ymfpci *chip)
+{
+ spin_lock_irq(&chip->reg_lock);
+ snd_ymfpci_writel(chip, YDSXGR_MAPOFEFFECT,
+ snd_ymfpci_readl(chip, YDSXGR_MAPOFEFFECT) & ~(3 << 3));
+ spin_unlock_irq(&chip->reg_lock);
+ // snd_ymfpci_irq_wait(chip);
+ if (chip->ac3_tmp_base.area) {
+ snd_dma_free_pages(&chip->ac3_tmp_base);
+ chip->ac3_tmp_base.area = NULL;
+ }
+ return 0;
+}
+
+static int snd_ymfpci_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm = runtime->private_data;
+ int err;
+
+ if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
+ return err;
+ if ((err = snd_ymfpci_pcm_voice_alloc(ypcm, params_channels(hw_params))) < 0)
+ return err;
+ return 0;
+}
+
+static int snd_ymfpci_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm;
+
+ if (runtime->private_data == NULL)
+ return 0;
+ ypcm = runtime->private_data;
+
+ /* wait, until the PCI operations are not finished */
+ snd_ymfpci_irq_wait(chip);
+ snd_pcm_lib_free_pages(substream);
+ if (ypcm->voices[1]) {
+ snd_ymfpci_voice_free(chip, ypcm->voices[1]);
+ ypcm->voices[1] = NULL;
+ }
+ if (ypcm->voices[0]) {
+ snd_ymfpci_voice_free(chip, ypcm->voices[0]);
+ ypcm->voices[0] = NULL;
+ }
+ return 0;
+}
+
+static int snd_ymfpci_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm = runtime->private_data;
+ struct snd_kcontrol *kctl;
+ unsigned int nvoice;
+
+ ypcm->period_size = runtime->period_size;
+ ypcm->buffer_size = runtime->buffer_size;
+ ypcm->period_pos = 0;
+ ypcm->last_pos = 0;
+ for (nvoice = 0; nvoice < runtime->channels; nvoice++)
+ snd_ymfpci_pcm_init_voice(ypcm, nvoice, runtime,
+ substream->pcm == chip->pcm);
+
+ if (substream->pcm == chip->pcm && !ypcm->use_441_slot) {
+ kctl = chip->pcm_mixer[substream->number].ctl;
+ kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &kctl->id);
+ }
+ return 0;
+}
+
+static int snd_ymfpci_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_ymfpci_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+
+ /* wait, until the PCI operations are not finished */
+ snd_ymfpci_irq_wait(chip);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_ymfpci_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm = runtime->private_data;
+ struct snd_ymfpci_capture_bank * bank;
+ int nbank;
+ u32 rate, format;
+
+ ypcm->period_size = runtime->period_size;
+ ypcm->buffer_size = runtime->buffer_size;
+ ypcm->period_pos = 0;
+ ypcm->last_pos = 0;
+ ypcm->shift = 0;
+ rate = ((48000 * 4096) / runtime->rate) - 1;
+ format = 0;
+ if (runtime->channels == 2) {
+ format |= 2;
+ ypcm->shift++;
+ }
+ if (snd_pcm_format_width(runtime->format) == 8)
+ format |= 1;
+ else
+ ypcm->shift++;
+ switch (ypcm->capture_bank_number) {
+ case 0:
+ snd_ymfpci_writel(chip, YDSXGR_RECFORMAT, format);
+ snd_ymfpci_writel(chip, YDSXGR_RECSLOTSR, rate);
+ break;
+ case 1:
+ snd_ymfpci_writel(chip, YDSXGR_ADCFORMAT, format);
+ snd_ymfpci_writel(chip, YDSXGR_ADCSLOTSR, rate);
+ break;
+ }
+ for (nbank = 0; nbank < 2; nbank++) {
+ bank = chip->bank_capture[ypcm->capture_bank_number][nbank];
+ bank->base = cpu_to_le32(runtime->dma_addr);
+ bank->loop_end = cpu_to_le32(ypcm->buffer_size << ypcm->shift);
+ bank->start = 0;
+ bank->num_of_loops = 0;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_ymfpci_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm = runtime->private_data;
+ struct snd_ymfpci_voice *voice = ypcm->voices[0];
+
+ if (!(ypcm->running && voice))
+ return 0;
+ return le32_to_cpu(voice->bank[chip->active_bank].start);
+}
+
+static snd_pcm_uframes_t snd_ymfpci_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm = runtime->private_data;
+
+ if (!ypcm->running)
+ return 0;
+ return le32_to_cpu(chip->bank_capture[ypcm->capture_bank_number][chip->active_bank]->start) >> ypcm->shift;
+}
+
+static void snd_ymfpci_irq_wait(struct snd_ymfpci *chip)
+{
+ wait_queue_t wait;
+ int loops = 4;
+
+ while (loops-- > 0) {
+ if ((snd_ymfpci_readl(chip, YDSXGR_MODE) & 3) == 0)
+ continue;
+ init_waitqueue_entry(&wait, current);
+ add_wait_queue(&chip->interrupt_sleep, &wait);
+ atomic_inc(&chip->interrupt_sleep_count);
+ schedule_timeout_uninterruptible(msecs_to_jiffies(50));
+ remove_wait_queue(&chip->interrupt_sleep, &wait);
+ }
+}
+
+static irqreturn_t snd_ymfpci_interrupt(int irq, void *dev_id)
+{
+ struct snd_ymfpci *chip = dev_id;
+ u32 status, nvoice, mode;
+ struct snd_ymfpci_voice *voice;
+
+ status = snd_ymfpci_readl(chip, YDSXGR_STATUS);
+ if (status & 0x80000000) {
+ chip->active_bank = snd_ymfpci_readl(chip, YDSXGR_CTRLSELECT) & 1;
+ spin_lock(&chip->voice_lock);
+ for (nvoice = 0; nvoice < YDSXG_PLAYBACK_VOICES; nvoice++) {
+ voice = &chip->voices[nvoice];
+ if (voice->interrupt)
+ voice->interrupt(chip, voice);
+ }
+ for (nvoice = 0; nvoice < YDSXG_CAPTURE_VOICES; nvoice++) {
+ if (chip->capture_substream[nvoice])
+ snd_ymfpci_pcm_capture_interrupt(chip->capture_substream[nvoice]);
+ }
+#if 0
+ for (nvoice = 0; nvoice < YDSXG_EFFECT_VOICES; nvoice++) {
+ if (chip->effect_substream[nvoice])
+ snd_ymfpci_pcm_effect_interrupt(chip->effect_substream[nvoice]);
+ }
+#endif
+ spin_unlock(&chip->voice_lock);
+ spin_lock(&chip->reg_lock);
+ snd_ymfpci_writel(chip, YDSXGR_STATUS, 0x80000000);
+ mode = snd_ymfpci_readl(chip, YDSXGR_MODE) | 2;
+ snd_ymfpci_writel(chip, YDSXGR_MODE, mode);
+ spin_unlock(&chip->reg_lock);
+
+ if (atomic_read(&chip->interrupt_sleep_count)) {
+ atomic_set(&chip->interrupt_sleep_count, 0);
+ wake_up(&chip->interrupt_sleep);
+ }
+ }
+
+ status = snd_ymfpci_readw(chip, YDSXGR_INTFLAG);
+ if (status & 1) {
+ if (chip->timer)
+ snd_timer_interrupt(chip->timer, chip->timer->sticks);
+ }
+ snd_ymfpci_writew(chip, YDSXGR_INTFLAG, status);
+
+ if (chip->rawmidi)
+ snd_mpu401_uart_interrupt(irq, chip->rawmidi->private_data);
+ return IRQ_HANDLED;
+}
+
+static struct snd_pcm_hardware snd_ymfpci_playback =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 256 * 1024, /* FIXME: enough? */
+ .period_bytes_min = 64,
+ .period_bytes_max = 256 * 1024, /* FIXME: enough? */
+ .periods_min = 3,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware snd_ymfpci_capture =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 256 * 1024, /* FIXME: enough? */
+ .period_bytes_min = 64,
+ .period_bytes_max = 256 * 1024, /* FIXME: enough? */
+ .periods_min = 3,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static void snd_ymfpci_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ kfree(runtime->private_data);
+}
+
+static int snd_ymfpci_playback_open_1(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm;
+
+ ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL);
+ if (ypcm == NULL)
+ return -ENOMEM;
+ ypcm->chip = chip;
+ ypcm->type = PLAYBACK_VOICE;
+ ypcm->substream = substream;
+ runtime->hw = snd_ymfpci_playback;
+ runtime->private_data = ypcm;
+ runtime->private_free = snd_ymfpci_pcm_free_substream;
+ /* FIXME? True value is 256/48 = 5.33333 ms */
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX);
+ return 0;
+}
+
+/* call with spinlock held */
+static void ymfpci_open_extension(struct snd_ymfpci *chip)
+{
+ if (! chip->rear_opened) {
+ if (! chip->spdif_opened) /* set AC3 */
+ snd_ymfpci_writel(chip, YDSXGR_MODE,
+ snd_ymfpci_readl(chip, YDSXGR_MODE) | (1 << 30));
+ /* enable second codec (4CHEN) */
+ snd_ymfpci_writew(chip, YDSXGR_SECCONFIG,
+ (snd_ymfpci_readw(chip, YDSXGR_SECCONFIG) & ~0x0330) | 0x0010);
+ }
+}
+
+/* call with spinlock held */
+static void ymfpci_close_extension(struct snd_ymfpci *chip)
+{
+ if (! chip->rear_opened) {
+ if (! chip->spdif_opened)
+ snd_ymfpci_writel(chip, YDSXGR_MODE,
+ snd_ymfpci_readl(chip, YDSXGR_MODE) & ~(1 << 30));
+ snd_ymfpci_writew(chip, YDSXGR_SECCONFIG,
+ (snd_ymfpci_readw(chip, YDSXGR_SECCONFIG) & ~0x0330) & ~0x0010);
+ }
+}
+
+static int snd_ymfpci_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm;
+ int err;
+
+ if ((err = snd_ymfpci_playback_open_1(substream)) < 0)
+ return err;
+ ypcm = runtime->private_data;
+ ypcm->output_front = 1;
+ ypcm->output_rear = chip->mode_dup4ch ? 1 : 0;
+ ypcm->swap_rear = 0;
+ spin_lock_irq(&chip->reg_lock);
+ if (ypcm->output_rear) {
+ ymfpci_open_extension(chip);
+ chip->rear_opened++;
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_ymfpci_playback_spdif_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm;
+ int err;
+
+ if ((err = snd_ymfpci_playback_open_1(substream)) < 0)
+ return err;
+ ypcm = runtime->private_data;
+ ypcm->output_front = 0;
+ ypcm->output_rear = 1;
+ ypcm->swap_rear = 1;
+ spin_lock_irq(&chip->reg_lock);
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTCTRL,
+ snd_ymfpci_readw(chip, YDSXGR_SPDIFOUTCTRL) | 2);
+ ymfpci_open_extension(chip);
+ chip->spdif_pcm_bits = chip->spdif_bits;
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTSTATUS, chip->spdif_pcm_bits);
+ chip->spdif_opened++;
+ spin_unlock_irq(&chip->reg_lock);
+
+ chip->spdif_pcm_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &chip->spdif_pcm_ctl->id);
+ return 0;
+}
+
+static int snd_ymfpci_playback_4ch_open(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm;
+ int err;
+
+ if ((err = snd_ymfpci_playback_open_1(substream)) < 0)
+ return err;
+ ypcm = runtime->private_data;
+ ypcm->output_front = 0;
+ ypcm->output_rear = 1;
+ ypcm->swap_rear = 0;
+ spin_lock_irq(&chip->reg_lock);
+ ymfpci_open_extension(chip);
+ chip->rear_opened++;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_ymfpci_capture_open(struct snd_pcm_substream *substream,
+ u32 capture_bank_number)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm;
+
+ ypcm = kzalloc(sizeof(*ypcm), GFP_KERNEL);
+ if (ypcm == NULL)
+ return -ENOMEM;
+ ypcm->chip = chip;
+ ypcm->type = capture_bank_number + CAPTURE_REC;
+ ypcm->substream = substream;
+ ypcm->capture_bank_number = capture_bank_number;
+ chip->capture_substream[capture_bank_number] = substream;
+ runtime->hw = snd_ymfpci_capture;
+ /* FIXME? True value is 256/48 = 5.33333 ms */
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 5333, UINT_MAX);
+ runtime->private_data = ypcm;
+ runtime->private_free = snd_ymfpci_pcm_free_substream;
+ snd_ymfpci_hw_start(chip);
+ return 0;
+}
+
+static int snd_ymfpci_capture_rec_open(struct snd_pcm_substream *substream)
+{
+ return snd_ymfpci_capture_open(substream, 0);
+}
+
+static int snd_ymfpci_capture_ac97_open(struct snd_pcm_substream *substream)
+{
+ return snd_ymfpci_capture_open(substream, 1);
+}
+
+static int snd_ymfpci_playback_close_1(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+static int snd_ymfpci_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_ymfpci_pcm *ypcm = substream->runtime->private_data;
+
+ spin_lock_irq(&chip->reg_lock);
+ if (ypcm->output_rear && chip->rear_opened > 0) {
+ chip->rear_opened--;
+ ymfpci_close_extension(chip);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return snd_ymfpci_playback_close_1(substream);
+}
+
+static int snd_ymfpci_playback_spdif_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ chip->spdif_opened = 0;
+ ymfpci_close_extension(chip);
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTCTRL,
+ snd_ymfpci_readw(chip, YDSXGR_SPDIFOUTCTRL) & ~2);
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTSTATUS, chip->spdif_bits);
+ spin_unlock_irq(&chip->reg_lock);
+ chip->spdif_pcm_ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &chip->spdif_pcm_ctl->id);
+ return snd_ymfpci_playback_close_1(substream);
+}
+
+static int snd_ymfpci_playback_4ch_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&chip->reg_lock);
+ if (chip->rear_opened > 0) {
+ chip->rear_opened--;
+ ymfpci_close_extension(chip);
+ }
+ spin_unlock_irq(&chip->reg_lock);
+ return snd_ymfpci_playback_close_1(substream);
+}
+
+static int snd_ymfpci_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_ymfpci *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ymfpci_pcm *ypcm = runtime->private_data;
+
+ if (ypcm != NULL) {
+ chip->capture_substream[ypcm->capture_bank_number] = NULL;
+ snd_ymfpci_hw_stop(chip);
+ }
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ymfpci_playback_ops = {
+ .open = snd_ymfpci_playback_open,
+ .close = snd_ymfpci_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ymfpci_playback_hw_params,
+ .hw_free = snd_ymfpci_playback_hw_free,
+ .prepare = snd_ymfpci_playback_prepare,
+ .trigger = snd_ymfpci_playback_trigger,
+ .pointer = snd_ymfpci_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_ymfpci_capture_rec_ops = {
+ .open = snd_ymfpci_capture_rec_open,
+ .close = snd_ymfpci_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ymfpci_capture_hw_params,
+ .hw_free = snd_ymfpci_capture_hw_free,
+ .prepare = snd_ymfpci_capture_prepare,
+ .trigger = snd_ymfpci_capture_trigger,
+ .pointer = snd_ymfpci_capture_pointer,
+};
+
+int __devinit snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(chip->card, "YMFPCI", device, 32, 1, &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ymfpci_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ymfpci_capture_rec_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "YMFPCI");
+ chip->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ymfpci_capture_ac97_ops = {
+ .open = snd_ymfpci_capture_ac97_open,
+ .close = snd_ymfpci_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ymfpci_capture_hw_params,
+ .hw_free = snd_ymfpci_capture_hw_free,
+ .prepare = snd_ymfpci_capture_prepare,
+ .trigger = snd_ymfpci_capture_trigger,
+ .pointer = snd_ymfpci_capture_pointer,
+};
+
+int __devinit snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(chip->card, "YMFPCI - PCM2", device, 0, 1, &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ymfpci_capture_ac97_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ sprintf(pcm->name, "YMFPCI - %s",
+ chip->device_id == PCI_DEVICE_ID_YAMAHA_754 ? "Direct Recording" : "AC'97");
+ chip->pcm2 = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ymfpci_playback_spdif_ops = {
+ .open = snd_ymfpci_playback_spdif_open,
+ .close = snd_ymfpci_playback_spdif_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ymfpci_playback_hw_params,
+ .hw_free = snd_ymfpci_playback_hw_free,
+ .prepare = snd_ymfpci_playback_prepare,
+ .trigger = snd_ymfpci_playback_trigger,
+ .pointer = snd_ymfpci_playback_pointer,
+};
+
+int __devinit snd_ymfpci_pcm_spdif(struct snd_ymfpci *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(chip->card, "YMFPCI - IEC958", device, 1, 0, &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ymfpci_playback_spdif_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "YMFPCI - IEC958");
+ chip->pcm_spdif = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static struct snd_pcm_ops snd_ymfpci_playback_4ch_ops = {
+ .open = snd_ymfpci_playback_4ch_open,
+ .close = snd_ymfpci_playback_4ch_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ymfpci_playback_hw_params,
+ .hw_free = snd_ymfpci_playback_hw_free,
+ .prepare = snd_ymfpci_playback_prepare,
+ .trigger = snd_ymfpci_playback_trigger,
+ .pointer = snd_ymfpci_playback_pointer,
+};
+
+int __devinit snd_ymfpci_pcm_4ch(struct snd_ymfpci *chip, int device, struct snd_pcm ** rpcm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(chip->card, "YMFPCI - Rear", device, 1, 0, &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ymfpci_playback_4ch_ops);
+
+ /* global setup */
+ pcm->info_flags = 0;
+ strcpy(pcm->name, "YMFPCI - Rear PCM");
+ chip->pcm_4ch = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
+
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
+}
+
+static int snd_ymfpci_spdif_default_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ymfpci_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&chip->reg_lock);
+ ucontrol->value.iec958.status[0] = (chip->spdif_bits >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (chip->spdif_bits >> 8) & 0xff;
+ ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS_48000;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_ymfpci_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ((ucontrol->value.iec958.status[0] & 0x3e) << 0) |
+ (ucontrol->value.iec958.status[1] << 8);
+ spin_lock_irq(&chip->reg_lock);
+ change = chip->spdif_bits != val;
+ chip->spdif_bits = val;
+ if ((snd_ymfpci_readw(chip, YDSXGR_SPDIFOUTCTRL) & 1) && chip->pcm_spdif == NULL)
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTSTATUS, chip->spdif_bits);
+ spin_unlock_irq(&chip->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ymfpci_spdif_default __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_ymfpci_spdif_default_info,
+ .get = snd_ymfpci_spdif_default_get,
+ .put = snd_ymfpci_spdif_default_put
+};
+
+static int snd_ymfpci_spdif_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ymfpci_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&chip->reg_lock);
+ ucontrol->value.iec958.status[0] = 0x3e;
+ ucontrol->value.iec958.status[1] = 0xff;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ymfpci_spdif_mask __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_ymfpci_spdif_mask_info,
+ .get = snd_ymfpci_spdif_mask_get,
+};
+
+static int snd_ymfpci_spdif_stream_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ymfpci_spdif_stream_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&chip->reg_lock);
+ ucontrol->value.iec958.status[0] = (chip->spdif_pcm_bits >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (chip->spdif_pcm_bits >> 8) & 0xff;
+ ucontrol->value.iec958.status[3] = IEC958_AES3_CON_FS_48000;
+ spin_unlock_irq(&chip->reg_lock);
+ return 0;
+}
+
+static int snd_ymfpci_spdif_stream_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change;
+
+ val = ((ucontrol->value.iec958.status[0] & 0x3e) << 0) |
+ (ucontrol->value.iec958.status[1] << 8);
+ spin_lock_irq(&chip->reg_lock);
+ change = chip->spdif_pcm_bits != val;
+ chip->spdif_pcm_bits = val;
+ if ((snd_ymfpci_readw(chip, YDSXGR_SPDIFOUTCTRL) & 2))
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTSTATUS, chip->spdif_pcm_bits);
+ spin_unlock_irq(&chip->reg_lock);
+ return change;
+}
+
+static struct snd_kcontrol_new snd_ymfpci_spdif_stream __devinitdata =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+ .info = snd_ymfpci_spdif_stream_info,
+ .get = snd_ymfpci_spdif_stream_get,
+ .put = snd_ymfpci_spdif_stream_put
+};
+
+static int snd_ymfpci_drec_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info)
+{
+ static char *texts[3] = {"AC'97", "IEC958", "ZV Port"};
+
+ info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ info->count = 1;
+ info->value.enumerated.items = 3;
+ if (info->value.enumerated.item > 2)
+ info->value.enumerated.item = 2;
+ strcpy(info->value.enumerated.name, texts[info->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ymfpci_drec_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ u16 reg;
+
+ spin_lock_irq(&chip->reg_lock);
+ reg = snd_ymfpci_readw(chip, YDSXGR_GLOBALCTRL);
+ spin_unlock_irq(&chip->reg_lock);
+ if (!(reg & 0x100))
+ value->value.enumerated.item[0] = 0;
+ else
+ value->value.enumerated.item[0] = 1 + ((reg & 0x200) != 0);
+ return 0;
+}
+
+static int snd_ymfpci_drec_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *value)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ u16 reg, old_reg;
+
+ spin_lock_irq(&chip->reg_lock);
+ old_reg = snd_ymfpci_readw(chip, YDSXGR_GLOBALCTRL);
+ if (value->value.enumerated.item[0] == 0)
+ reg = old_reg & ~0x100;
+ else
+ reg = (old_reg & ~0x300) | 0x100 | ((value->value.enumerated.item[0] == 2) << 9);
+ snd_ymfpci_writew(chip, YDSXGR_GLOBALCTRL, reg);
+ spin_unlock_irq(&chip->reg_lock);
+ return reg != old_reg;
+}
+
+static struct snd_kcontrol_new snd_ymfpci_drec_source __devinitdata = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Direct Recording Source",
+ .info = snd_ymfpci_drec_source_info,
+ .get = snd_ymfpci_drec_source_get,
+ .put = snd_ymfpci_drec_source_put
+};
+
+/*
+ * Mixer controls
+ */
+
+#define YMFPCI_SINGLE(xname, xindex, reg, shift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .info = snd_ymfpci_info_single, \
+ .get = snd_ymfpci_get_single, .put = snd_ymfpci_put_single, \
+ .private_value = ((reg) | ((shift) << 16)) }
+
+#define snd_ymfpci_info_single snd_ctl_boolean_mono_info
+
+static int snd_ymfpci_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xffff;
+ unsigned int shift = (kcontrol->private_value >> 16) & 0xff;
+ unsigned int mask = 1;
+
+ switch (reg) {
+ case YDSXGR_SPDIFOUTCTRL: break;
+ case YDSXGR_SPDIFINCTRL: break;
+ default: return -EINVAL;
+ }
+ ucontrol->value.integer.value[0] =
+ (snd_ymfpci_readl(chip, reg) >> shift) & mask;
+ return 0;
+}
+
+static int snd_ymfpci_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xffff;
+ unsigned int shift = (kcontrol->private_value >> 16) & 0xff;
+ unsigned int mask = 1;
+ int change;
+ unsigned int val, oval;
+
+ switch (reg) {
+ case YDSXGR_SPDIFOUTCTRL: break;
+ case YDSXGR_SPDIFINCTRL: break;
+ default: return -EINVAL;
+ }
+ val = (ucontrol->value.integer.value[0] & mask);
+ val <<= shift;
+ spin_lock_irq(&chip->reg_lock);
+ oval = snd_ymfpci_readl(chip, reg);
+ val = (oval & ~(mask << shift)) | val;
+ change = val != oval;
+ snd_ymfpci_writel(chip, reg, val);
+ spin_unlock_irq(&chip->reg_lock);
+ return change;
+}
+
+static const DECLARE_TLV_DB_LINEAR(db_scale_native, TLV_DB_GAIN_MUTE, 0);
+
+#define YMFPCI_DOUBLE(xname, xindex, reg) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \
+ .info = snd_ymfpci_info_double, \
+ .get = snd_ymfpci_get_double, .put = snd_ymfpci_put_double, \
+ .private_value = reg, \
+ .tlv = { .p = db_scale_native } }
+
+static int snd_ymfpci_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int reg = kcontrol->private_value;
+
+ if (reg < 0x80 || reg >= 0xc0)
+ return -EINVAL;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 16383;
+ return 0;
+}
+
+static int snd_ymfpci_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value;
+ unsigned int shift_left = 0, shift_right = 16, mask = 16383;
+ unsigned int val;
+
+ if (reg < 0x80 || reg >= 0xc0)
+ return -EINVAL;
+ spin_lock_irq(&chip->reg_lock);
+ val = snd_ymfpci_readl(chip, reg);
+ spin_unlock_irq(&chip->reg_lock);
+ ucontrol->value.integer.value[0] = (val >> shift_left) & mask;
+ ucontrol->value.integer.value[1] = (val >> shift_right) & mask;
+ return 0;
+}
+
+static int snd_ymfpci_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value;
+ unsigned int shift_left = 0, shift_right = 16, mask = 16383;
+ int change;
+ unsigned int val1, val2, oval;
+
+ if (reg < 0x80 || reg >= 0xc0)
+ return -EINVAL;
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+ spin_lock_irq(&chip->reg_lock);
+ oval = snd_ymfpci_readl(chip, reg);
+ val1 = (oval & ~((mask << shift_left) | (mask << shift_right))) | val1 | val2;
+ change = val1 != oval;
+ snd_ymfpci_writel(chip, reg, val1);
+ spin_unlock_irq(&chip->reg_lock);
+ return change;
+}
+
+static int snd_ymfpci_put_nativedacvol(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = YDSXGR_NATIVEDACOUTVOL;
+ unsigned int reg2 = YDSXGR_BUF441OUTVOL;
+ int change;
+ unsigned int value, oval;
+
+ value = ucontrol->value.integer.value[0] & 0x3fff;
+ value |= (ucontrol->value.integer.value[1] & 0x3fff) << 16;
+ spin_lock_irq(&chip->reg_lock);
+ oval = snd_ymfpci_readl(chip, reg);
+ change = value != oval;
+ snd_ymfpci_writel(chip, reg, value);
+ snd_ymfpci_writel(chip, reg2, value);
+ spin_unlock_irq(&chip->reg_lock);
+ return change;
+}
+
+/*
+ * 4ch duplication
+ */
+#define snd_ymfpci_info_dup4ch snd_ctl_boolean_mono_info
+
+static int snd_ymfpci_get_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = chip->mode_dup4ch;
+ return 0;
+}
+
+static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ int change;
+ change = (ucontrol->value.integer.value[0] != chip->mode_dup4ch);
+ if (change)
+ chip->mode_dup4ch = !!ucontrol->value.integer.value[0];
+ return change;
+}
+
+
+static struct snd_kcontrol_new snd_ymfpci_controls[] __devinitdata = {
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Wave Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_ymfpci_info_double,
+ .get = snd_ymfpci_get_double,
+ .put = snd_ymfpci_put_nativedacvol,
+ .private_value = YDSXGR_NATIVEDACOUTVOL,
+ .tlv = { .p = db_scale_native },
+},
+YMFPCI_DOUBLE("Wave Capture Volume", 0, YDSXGR_NATIVEDACLOOPVOL),
+YMFPCI_DOUBLE("Digital Capture Volume", 0, YDSXGR_NATIVEDACINVOL),
+YMFPCI_DOUBLE("Digital Capture Volume", 1, YDSXGR_NATIVEADCINVOL),
+YMFPCI_DOUBLE("ADC Playback Volume", 0, YDSXGR_PRIADCOUTVOL),
+YMFPCI_DOUBLE("ADC Capture Volume", 0, YDSXGR_PRIADCLOOPVOL),
+YMFPCI_DOUBLE("ADC Playback Volume", 1, YDSXGR_SECADCOUTVOL),
+YMFPCI_DOUBLE("ADC Capture Volume", 1, YDSXGR_SECADCLOOPVOL),
+YMFPCI_DOUBLE("FM Legacy Volume", 0, YDSXGR_LEGACYOUTVOL),
+YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ", PLAYBACK,VOLUME), 0, YDSXGR_ZVOUTVOL),
+YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("", CAPTURE,VOLUME), 0, YDSXGR_ZVLOOPVOL),
+YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ",PLAYBACK,VOLUME), 1, YDSXGR_SPDIFOUTVOL),
+YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL),
+YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0),
+YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0),
+YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "4ch Duplication",
+ .info = snd_ymfpci_info_dup4ch,
+ .get = snd_ymfpci_get_dup4ch,
+ .put = snd_ymfpci_put_dup4ch,
+},
+};
+
+
+/*
+ * GPIO
+ */
+
+static int snd_ymfpci_get_gpio_out(struct snd_ymfpci *chip, int pin)
+{
+ u16 reg, mode;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ reg = snd_ymfpci_readw(chip, YDSXGR_GPIOFUNCENABLE);
+ reg &= ~(1 << (pin + 8));
+ reg |= (1 << pin);
+ snd_ymfpci_writew(chip, YDSXGR_GPIOFUNCENABLE, reg);
+ /* set the level mode for input line */
+ mode = snd_ymfpci_readw(chip, YDSXGR_GPIOTYPECONFIG);
+ mode &= ~(3 << (pin * 2));
+ snd_ymfpci_writew(chip, YDSXGR_GPIOTYPECONFIG, mode);
+ snd_ymfpci_writew(chip, YDSXGR_GPIOFUNCENABLE, reg | (1 << (pin + 8)));
+ mode = snd_ymfpci_readw(chip, YDSXGR_GPIOINSTATUS);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return (mode >> pin) & 1;
+}
+
+static int snd_ymfpci_set_gpio_out(struct snd_ymfpci *chip, int pin, int enable)
+{
+ u16 reg;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ reg = snd_ymfpci_readw(chip, YDSXGR_GPIOFUNCENABLE);
+ reg &= ~(1 << pin);
+ reg &= ~(1 << (pin + 8));
+ snd_ymfpci_writew(chip, YDSXGR_GPIOFUNCENABLE, reg);
+ snd_ymfpci_writew(chip, YDSXGR_GPIOOUTCTRL, enable << pin);
+ snd_ymfpci_writew(chip, YDSXGR_GPIOFUNCENABLE, reg | (1 << (pin + 8)));
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ return 0;
+}
+
+#define snd_ymfpci_gpio_sw_info snd_ctl_boolean_mono_info
+
+static int snd_ymfpci_gpio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ int pin = (int)kcontrol->private_value;
+ ucontrol->value.integer.value[0] = snd_ymfpci_get_gpio_out(chip, pin);
+ return 0;
+}
+
+static int snd_ymfpci_gpio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ int pin = (int)kcontrol->private_value;
+
+ if (snd_ymfpci_get_gpio_out(chip, pin) != ucontrol->value.integer.value[0]) {
+ snd_ymfpci_set_gpio_out(chip, pin, !!ucontrol->value.integer.value[0]);
+ ucontrol->value.integer.value[0] = snd_ymfpci_get_gpio_out(chip, pin);
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ymfpci_rear_shared __devinitdata = {
+ .name = "Shared Rear/Line-In Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_ymfpci_gpio_sw_info,
+ .get = snd_ymfpci_gpio_sw_get,
+ .put = snd_ymfpci_gpio_sw_put,
+ .private_value = 2,
+};
+
+/*
+ * PCM voice volume
+ */
+
+static int snd_ymfpci_pcm_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0x8000;
+ return 0;
+}
+
+static int snd_ymfpci_pcm_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int subs = kcontrol->id.subdevice;
+
+ ucontrol->value.integer.value[0] = chip->pcm_mixer[subs].left;
+ ucontrol->value.integer.value[1] = chip->pcm_mixer[subs].right;
+ return 0;
+}
+
+static int snd_ymfpci_pcm_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ymfpci *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int subs = kcontrol->id.subdevice;
+ struct snd_pcm_substream *substream;
+ unsigned long flags;
+
+ if (ucontrol->value.integer.value[0] != chip->pcm_mixer[subs].left ||
+ ucontrol->value.integer.value[1] != chip->pcm_mixer[subs].right) {
+ chip->pcm_mixer[subs].left = ucontrol->value.integer.value[0];
+ chip->pcm_mixer[subs].right = ucontrol->value.integer.value[1];
+ if (chip->pcm_mixer[subs].left > 0x8000)
+ chip->pcm_mixer[subs].left = 0x8000;
+ if (chip->pcm_mixer[subs].right > 0x8000)
+ chip->pcm_mixer[subs].right = 0x8000;
+
+ substream = (struct snd_pcm_substream *)kcontrol->private_value;
+ spin_lock_irqsave(&chip->voice_lock, flags);
+ if (substream->runtime && substream->runtime->private_data) {
+ struct snd_ymfpci_pcm *ypcm = substream->runtime->private_data;
+ if (!ypcm->use_441_slot)
+ ypcm->update_pcm_vol = 2;
+ }
+ spin_unlock_irqrestore(&chip->voice_lock, flags);
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_ymfpci_pcm_volume __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "PCM Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .info = snd_ymfpci_pcm_vol_info,
+ .get = snd_ymfpci_pcm_vol_get,
+ .put = snd_ymfpci_pcm_vol_put,
+};
+
+
+/*
+ * Mixer routines
+ */
+
+static void snd_ymfpci_mixer_free_ac97_bus(struct snd_ac97_bus *bus)
+{
+ struct snd_ymfpci *chip = bus->private_data;
+ chip->ac97_bus = NULL;
+}
+
+static void snd_ymfpci_mixer_free_ac97(struct snd_ac97 *ac97)
+{
+ struct snd_ymfpci *chip = ac97->private_data;
+ chip->ac97 = NULL;
+}
+
+int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch)
+{
+ struct snd_ac97_template ac97;
+ struct snd_kcontrol *kctl;
+ struct snd_pcm_substream *substream;
+ unsigned int idx;
+ int err;
+ static struct snd_ac97_bus_ops ops = {
+ .write = snd_ymfpci_codec_write,
+ .read = snd_ymfpci_codec_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &chip->ac97_bus)) < 0)
+ return err;
+ chip->ac97_bus->private_free = snd_ymfpci_mixer_free_ac97_bus;
+ chip->ac97_bus->no_vra = 1; /* YMFPCI doesn't need VRA */
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ ac97.private_free = snd_ymfpci_mixer_free_ac97;
+ if ((err = snd_ac97_mixer(chip->ac97_bus, &ac97, &chip->ac97)) < 0)
+ return err;
+
+ /* to be sure */
+ snd_ac97_update_bits(chip->ac97, AC97_EXTENDED_STATUS,
+ AC97_EA_VRA|AC97_EA_VRM, 0);
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_ymfpci_controls); idx++) {
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_controls[idx], chip))) < 0)
+ return err;
+ }
+
+ /* add S/PDIF control */
+ if (snd_BUG_ON(!chip->pcm_spdif))
+ return -ENXIO;
+ if ((err = snd_ctl_add(chip->card, kctl = snd_ctl_new1(&snd_ymfpci_spdif_default, chip))) < 0)
+ return err;
+ kctl->id.device = chip->pcm_spdif->device;
+ if ((err = snd_ctl_add(chip->card, kctl = snd_ctl_new1(&snd_ymfpci_spdif_mask, chip))) < 0)
+ return err;
+ kctl->id.device = chip->pcm_spdif->device;
+ if ((err = snd_ctl_add(chip->card, kctl = snd_ctl_new1(&snd_ymfpci_spdif_stream, chip))) < 0)
+ return err;
+ kctl->id.device = chip->pcm_spdif->device;
+ chip->spdif_pcm_ctl = kctl;
+
+ /* direct recording source */
+ if (chip->device_id == PCI_DEVICE_ID_YAMAHA_754 &&
+ (err = snd_ctl_add(chip->card, kctl = snd_ctl_new1(&snd_ymfpci_drec_source, chip))) < 0)
+ return err;
+
+ /*
+ * shared rear/line-in
+ */
+ if (rear_switch) {
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_rear_shared, chip))) < 0)
+ return err;
+ }
+
+ /* per-voice volume */
+ substream = chip->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ for (idx = 0; idx < 32; ++idx) {
+ kctl = snd_ctl_new1(&snd_ymfpci_pcm_volume, chip);
+ if (!kctl)
+ return -ENOMEM;
+ kctl->id.device = chip->pcm->device;
+ kctl->id.subdevice = idx;
+ kctl->private_value = (unsigned long)substream;
+ if ((err = snd_ctl_add(chip->card, kctl)) < 0)
+ return err;
+ chip->pcm_mixer[idx].left = 0x8000;
+ chip->pcm_mixer[idx].right = 0x8000;
+ chip->pcm_mixer[idx].ctl = kctl;
+ substream = substream->next;
+ }
+
+ return 0;
+}
+
+
+/*
+ * timer
+ */
+
+static int snd_ymfpci_timer_start(struct snd_timer *timer)
+{
+ struct snd_ymfpci *chip;
+ unsigned long flags;
+ unsigned int count;
+
+ chip = snd_timer_chip(timer);
+ count = (timer->sticks << 1) - 1;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_ymfpci_writew(chip, YDSXGR_TIMERCOUNT, count);
+ snd_ymfpci_writeb(chip, YDSXGR_TIMERCTRL, 0x03);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_ymfpci_timer_stop(struct snd_timer *timer)
+{
+ struct snd_ymfpci *chip;
+ unsigned long flags;
+
+ chip = snd_timer_chip(timer);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_ymfpci_writeb(chip, YDSXGR_TIMERCTRL, 0x00);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+static int snd_ymfpci_timer_precise_resolution(struct snd_timer *timer,
+ unsigned long *num, unsigned long *den)
+{
+ *num = 1;
+ *den = 48000;
+ return 0;
+}
+
+static struct snd_timer_hardware snd_ymfpci_timer_hw = {
+ .flags = SNDRV_TIMER_HW_AUTO,
+ .resolution = 20833, /* 1/fs = 20.8333...us */
+ .ticks = 0x8000,
+ .start = snd_ymfpci_timer_start,
+ .stop = snd_ymfpci_timer_stop,
+ .precise_resolution = snd_ymfpci_timer_precise_resolution,
+};
+
+int __devinit snd_ymfpci_timer(struct snd_ymfpci *chip, int device)
+{
+ struct snd_timer *timer = NULL;
+ struct snd_timer_id tid;
+ int err;
+
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = chip->card->number;
+ tid.device = device;
+ tid.subdevice = 0;
+ if ((err = snd_timer_new(chip->card, "YMFPCI", &tid, &timer)) >= 0) {
+ strcpy(timer->name, "YMFPCI timer");
+ timer->private_data = chip;
+ timer->hw = snd_ymfpci_timer_hw;
+ }
+ chip->timer = timer;
+ return err;
+}
+
+
+/*
+ * proc interface
+ */
+
+static void snd_ymfpci_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_ymfpci *chip = entry->private_data;
+ int i;
+
+ snd_iprintf(buffer, "YMFPCI\n\n");
+ for (i = 0; i <= YDSXGR_WORKBASE; i += 4)
+ snd_iprintf(buffer, "%04x: %04x\n", i, snd_ymfpci_readl(chip, i));
+}
+
+static int __devinit snd_ymfpci_proc_init(struct snd_card *card, struct snd_ymfpci *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(card, "ymfpci", &entry))
+ snd_info_set_text_ops(entry, chip, snd_ymfpci_proc_read);
+ return 0;
+}
+
+/*
+ * initialization routines
+ */
+
+static void snd_ymfpci_aclink_reset(struct pci_dev * pci)
+{
+ u8 cmd;
+
+ pci_read_config_byte(pci, PCIR_DSXG_CTRL, &cmd);
+#if 0 // force to reset
+ if (cmd & 0x03) {
+#endif
+ pci_write_config_byte(pci, PCIR_DSXG_CTRL, cmd & 0xfc);
+ pci_write_config_byte(pci, PCIR_DSXG_CTRL, cmd | 0x03);
+ pci_write_config_byte(pci, PCIR_DSXG_CTRL, cmd & 0xfc);
+ pci_write_config_word(pci, PCIR_DSXG_PWRCTRL1, 0);
+ pci_write_config_word(pci, PCIR_DSXG_PWRCTRL2, 0);
+#if 0
+ }
+#endif
+}
+
+static void snd_ymfpci_enable_dsp(struct snd_ymfpci *chip)
+{
+ snd_ymfpci_writel(chip, YDSXGR_CONFIG, 0x00000001);
+}
+
+static void snd_ymfpci_disable_dsp(struct snd_ymfpci *chip)
+{
+ u32 val;
+ int timeout = 1000;
+
+ val = snd_ymfpci_readl(chip, YDSXGR_CONFIG);
+ if (val)
+ snd_ymfpci_writel(chip, YDSXGR_CONFIG, 0x00000000);
+ while (timeout-- > 0) {
+ val = snd_ymfpci_readl(chip, YDSXGR_STATUS);
+ if ((val & 0x00000002) == 0)
+ break;
+ }
+}
+
+static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip)
+{
+ int err, is_1e;
+ const char *name;
+
+ err = request_firmware(&chip->dsp_microcode, "yamaha/ds1_dsp.fw",
+ &chip->pci->dev);
+ if (err >= 0) {
+ if (chip->dsp_microcode->size != YDSXG_DSPLENGTH) {
+ snd_printk(KERN_ERR "DSP microcode has wrong size\n");
+ err = -EINVAL;
+ }
+ }
+ if (err < 0)
+ return err;
+ is_1e = chip->device_id == PCI_DEVICE_ID_YAMAHA_724F ||
+ chip->device_id == PCI_DEVICE_ID_YAMAHA_740C ||
+ chip->device_id == PCI_DEVICE_ID_YAMAHA_744 ||
+ chip->device_id == PCI_DEVICE_ID_YAMAHA_754;
+ name = is_1e ? "yamaha/ds1e_ctrl.fw" : "yamaha/ds1_ctrl.fw";
+ err = request_firmware(&chip->controller_microcode, name,
+ &chip->pci->dev);
+ if (err >= 0) {
+ if (chip->controller_microcode->size != YDSXG_CTRLLENGTH) {
+ snd_printk(KERN_ERR "controller microcode"
+ " has wrong size\n");
+ err = -EINVAL;
+ }
+ }
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+MODULE_FIRMWARE("yamaha/ds1_dsp.fw");
+MODULE_FIRMWARE("yamaha/ds1_ctrl.fw");
+MODULE_FIRMWARE("yamaha/ds1e_ctrl.fw");
+
+static void snd_ymfpci_download_image(struct snd_ymfpci *chip)
+{
+ int i;
+ u16 ctrl;
+ const __le32 *inst;
+
+ snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0x00000000);
+ snd_ymfpci_disable_dsp(chip);
+ snd_ymfpci_writel(chip, YDSXGR_MODE, 0x00010000);
+ snd_ymfpci_writel(chip, YDSXGR_MODE, 0x00000000);
+ snd_ymfpci_writel(chip, YDSXGR_MAPOFREC, 0x00000000);
+ snd_ymfpci_writel(chip, YDSXGR_MAPOFEFFECT, 0x00000000);
+ snd_ymfpci_writel(chip, YDSXGR_PLAYCTRLBASE, 0x00000000);
+ snd_ymfpci_writel(chip, YDSXGR_RECCTRLBASE, 0x00000000);
+ snd_ymfpci_writel(chip, YDSXGR_EFFCTRLBASE, 0x00000000);
+ ctrl = snd_ymfpci_readw(chip, YDSXGR_GLOBALCTRL);
+ snd_ymfpci_writew(chip, YDSXGR_GLOBALCTRL, ctrl & ~0x0007);
+
+ /* setup DSP instruction code */
+ inst = (const __le32 *)chip->dsp_microcode->data;
+ for (i = 0; i < YDSXG_DSPLENGTH / 4; i++)
+ snd_ymfpci_writel(chip, YDSXGR_DSPINSTRAM + (i << 2),
+ le32_to_cpu(inst[i]));
+
+ /* setup control instruction code */
+ inst = (const __le32 *)chip->controller_microcode->data;
+ for (i = 0; i < YDSXG_CTRLLENGTH / 4; i++)
+ snd_ymfpci_writel(chip, YDSXGR_CTRLINSTRAM + (i << 2),
+ le32_to_cpu(inst[i]));
+
+ snd_ymfpci_enable_dsp(chip);
+}
+
+static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip)
+{
+ long size, playback_ctrl_size;
+ int voice, bank, reg;
+ u8 *ptr;
+ dma_addr_t ptr_addr;
+
+ playback_ctrl_size = 4 + 4 * YDSXG_PLAYBACK_VOICES;
+ chip->bank_size_playback = snd_ymfpci_readl(chip, YDSXGR_PLAYCTRLSIZE) << 2;
+ chip->bank_size_capture = snd_ymfpci_readl(chip, YDSXGR_RECCTRLSIZE) << 2;
+ chip->bank_size_effect = snd_ymfpci_readl(chip, YDSXGR_EFFCTRLSIZE) << 2;
+ chip->work_size = YDSXG_DEFAULT_WORK_SIZE;
+
+ size = ALIGN(playback_ctrl_size, 0x100) +
+ ALIGN(chip->bank_size_playback * 2 * YDSXG_PLAYBACK_VOICES, 0x100) +
+ ALIGN(chip->bank_size_capture * 2 * YDSXG_CAPTURE_VOICES, 0x100) +
+ ALIGN(chip->bank_size_effect * 2 * YDSXG_EFFECT_VOICES, 0x100) +
+ chip->work_size;
+ /* work_ptr must be aligned to 256 bytes, but it's already
+ covered with the kernel page allocation mechanism */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ size, &chip->work_ptr) < 0)
+ return -ENOMEM;
+ ptr = chip->work_ptr.area;
+ ptr_addr = chip->work_ptr.addr;
+ memset(ptr, 0, size); /* for sure */
+
+ chip->bank_base_playback = ptr;
+ chip->bank_base_playback_addr = ptr_addr;
+ chip->ctrl_playback = (u32 *)ptr;
+ chip->ctrl_playback[0] = cpu_to_le32(YDSXG_PLAYBACK_VOICES);
+ ptr += ALIGN(playback_ctrl_size, 0x100);
+ ptr_addr += ALIGN(playback_ctrl_size, 0x100);
+ for (voice = 0; voice < YDSXG_PLAYBACK_VOICES; voice++) {
+ chip->voices[voice].number = voice;
+ chip->voices[voice].bank = (struct snd_ymfpci_playback_bank *)ptr;
+ chip->voices[voice].bank_addr = ptr_addr;
+ for (bank = 0; bank < 2; bank++) {
+ chip->bank_playback[voice][bank] = (struct snd_ymfpci_playback_bank *)ptr;
+ ptr += chip->bank_size_playback;
+ ptr_addr += chip->bank_size_playback;
+ }
+ }
+ ptr = (char *)ALIGN((unsigned long)ptr, 0x100);
+ ptr_addr = ALIGN(ptr_addr, 0x100);
+ chip->bank_base_capture = ptr;
+ chip->bank_base_capture_addr = ptr_addr;
+ for (voice = 0; voice < YDSXG_CAPTURE_VOICES; voice++)
+ for (bank = 0; bank < 2; bank++) {
+ chip->bank_capture[voice][bank] = (struct snd_ymfpci_capture_bank *)ptr;
+ ptr += chip->bank_size_capture;
+ ptr_addr += chip->bank_size_capture;
+ }
+ ptr = (char *)ALIGN((unsigned long)ptr, 0x100);
+ ptr_addr = ALIGN(ptr_addr, 0x100);
+ chip->bank_base_effect = ptr;
+ chip->bank_base_effect_addr = ptr_addr;
+ for (voice = 0; voice < YDSXG_EFFECT_VOICES; voice++)
+ for (bank = 0; bank < 2; bank++) {
+ chip->bank_effect[voice][bank] = (struct snd_ymfpci_effect_bank *)ptr;
+ ptr += chip->bank_size_effect;
+ ptr_addr += chip->bank_size_effect;
+ }
+ ptr = (char *)ALIGN((unsigned long)ptr, 0x100);
+ ptr_addr = ALIGN(ptr_addr, 0x100);
+ chip->work_base = ptr;
+ chip->work_base_addr = ptr_addr;
+
+ snd_BUG_ON(ptr + chip->work_size !=
+ chip->work_ptr.area + chip->work_ptr.bytes);
+
+ snd_ymfpci_writel(chip, YDSXGR_PLAYCTRLBASE, chip->bank_base_playback_addr);
+ snd_ymfpci_writel(chip, YDSXGR_RECCTRLBASE, chip->bank_base_capture_addr);
+ snd_ymfpci_writel(chip, YDSXGR_EFFCTRLBASE, chip->bank_base_effect_addr);
+ snd_ymfpci_writel(chip, YDSXGR_WORKBASE, chip->work_base_addr);
+ snd_ymfpci_writel(chip, YDSXGR_WORKSIZE, chip->work_size >> 2);
+
+ /* S/PDIF output initialization */
+ chip->spdif_bits = chip->spdif_pcm_bits = SNDRV_PCM_DEFAULT_CON_SPDIF & 0xffff;
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTCTRL, 0);
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFOUTSTATUS, chip->spdif_bits);
+
+ /* S/PDIF input initialization */
+ snd_ymfpci_writew(chip, YDSXGR_SPDIFINCTRL, 0);
+
+ /* digital mixer setup */
+ for (reg = 0x80; reg < 0xc0; reg += 4)
+ snd_ymfpci_writel(chip, reg, 0);
+ snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0x3fff3fff);
+ snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0x3fff3fff);
+ snd_ymfpci_writel(chip, YDSXGR_ZVOUTVOL, 0x3fff3fff);
+ snd_ymfpci_writel(chip, YDSXGR_SPDIFOUTVOL, 0x3fff3fff);
+ snd_ymfpci_writel(chip, YDSXGR_NATIVEADCINVOL, 0x3fff3fff);
+ snd_ymfpci_writel(chip, YDSXGR_NATIVEDACINVOL, 0x3fff3fff);
+ snd_ymfpci_writel(chip, YDSXGR_PRIADCLOOPVOL, 0x3fff3fff);
+ snd_ymfpci_writel(chip, YDSXGR_LEGACYOUTVOL, 0x3fff3fff);
+
+ return 0;
+}
+
+static int snd_ymfpci_free(struct snd_ymfpci *chip)
+{
+ u16 ctrl;
+
+ if (snd_BUG_ON(!chip))
+ return -EINVAL;
+
+ if (chip->res_reg_area) { /* don't touch busy hardware */
+ snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0);
+ snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0);
+ snd_ymfpci_writel(chip, YDSXGR_LEGACYOUTVOL, 0);
+ snd_ymfpci_writel(chip, YDSXGR_STATUS, ~0);
+ snd_ymfpci_disable_dsp(chip);
+ snd_ymfpci_writel(chip, YDSXGR_PLAYCTRLBASE, 0);
+ snd_ymfpci_writel(chip, YDSXGR_RECCTRLBASE, 0);
+ snd_ymfpci_writel(chip, YDSXGR_EFFCTRLBASE, 0);
+ snd_ymfpci_writel(chip, YDSXGR_WORKBASE, 0);
+ snd_ymfpci_writel(chip, YDSXGR_WORKSIZE, 0);
+ ctrl = snd_ymfpci_readw(chip, YDSXGR_GLOBALCTRL);
+ snd_ymfpci_writew(chip, YDSXGR_GLOBALCTRL, ctrl & ~0x0007);
+ }
+
+ snd_ymfpci_ac3_done(chip);
+
+ /* Set PCI device to D3 state */
+#if 0
+ /* FIXME: temporarily disabled, otherwise we cannot fire up
+ * the chip again unless reboot. ACPI bug?
+ */
+ pci_set_power_state(chip->pci, 3);
+#endif
+
+#ifdef CONFIG_PM
+ vfree(chip->saved_regs);
+#endif
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+ release_and_free_resource(chip->mpu_res);
+ release_and_free_resource(chip->fm_res);
+ snd_ymfpci_free_gameport(chip);
+ if (chip->reg_area_virt)
+ iounmap(chip->reg_area_virt);
+ if (chip->work_ptr.area)
+ snd_dma_free_pages(&chip->work_ptr);
+
+ release_and_free_resource(chip->res_reg_area);
+
+ pci_write_config_word(chip->pci, 0x40, chip->old_legacy_ctrl);
+
+ pci_disable_device(chip->pci);
+ release_firmware(chip->dsp_microcode);
+ release_firmware(chip->controller_microcode);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_ymfpci_dev_free(struct snd_device *device)
+{
+ struct snd_ymfpci *chip = device->device_data;
+ return snd_ymfpci_free(chip);
+}
+
+#ifdef CONFIG_PM
+static int saved_regs_index[] = {
+ /* spdif */
+ YDSXGR_SPDIFOUTCTRL,
+ YDSXGR_SPDIFOUTSTATUS,
+ YDSXGR_SPDIFINCTRL,
+ /* volumes */
+ YDSXGR_PRIADCLOOPVOL,
+ YDSXGR_NATIVEDACINVOL,
+ YDSXGR_NATIVEDACOUTVOL,
+ YDSXGR_BUF441OUTVOL,
+ YDSXGR_NATIVEADCINVOL,
+ YDSXGR_SPDIFLOOPVOL,
+ YDSXGR_SPDIFOUTVOL,
+ YDSXGR_ZVOUTVOL,
+ YDSXGR_LEGACYOUTVOL,
+ /* address bases */
+ YDSXGR_PLAYCTRLBASE,
+ YDSXGR_RECCTRLBASE,
+ YDSXGR_EFFCTRLBASE,
+ YDSXGR_WORKBASE,
+ /* capture set up */
+ YDSXGR_MAPOFREC,
+ YDSXGR_RECFORMAT,
+ YDSXGR_RECSLOTSR,
+ YDSXGR_ADCFORMAT,
+ YDSXGR_ADCSLOTSR,
+};
+#define YDSXGR_NUM_SAVED_REGS ARRAY_SIZE(saved_regs_index)
+
+int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_ymfpci *chip = card->private_data;
+ unsigned int i;
+
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ snd_pcm_suspend_all(chip->pcm2);
+ snd_pcm_suspend_all(chip->pcm_spdif);
+ snd_pcm_suspend_all(chip->pcm_4ch);
+ snd_ac97_suspend(chip->ac97);
+ for (i = 0; i < YDSXGR_NUM_SAVED_REGS; i++)
+ chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]);
+ chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE);
+ snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0);
+ snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0);
+ snd_ymfpci_disable_dsp(chip);
+ pci_disable_device(pci);
+ pci_save_state(pci);
+ pci_set_power_state(pci, pci_choose_state(pci, state));
+ return 0;
+}
+
+int snd_ymfpci_resume(struct pci_dev *pci)
+{
+ struct snd_card *card = pci_get_drvdata(pci);
+ struct snd_ymfpci *chip = card->private_data;
+ unsigned int i;
+
+ pci_set_power_state(pci, PCI_D0);
+ pci_restore_state(pci);
+ if (pci_enable_device(pci) < 0) {
+ printk(KERN_ERR "ymfpci: pci_enable_device failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
+ pci_set_master(pci);
+ snd_ymfpci_aclink_reset(pci);
+ snd_ymfpci_codec_ready(chip, 0);
+ snd_ymfpci_download_image(chip);
+ udelay(100);
+
+ for (i = 0; i < YDSXGR_NUM_SAVED_REGS; i++)
+ snd_ymfpci_writel(chip, saved_regs_index[i], chip->saved_regs[i]);
+
+ snd_ac97_resume(chip->ac97);
+
+ /* start hw again */
+ if (chip->start_count > 0) {
+ spin_lock_irq(&chip->reg_lock);
+ snd_ymfpci_writel(chip, YDSXGR_MODE, chip->saved_ydsxgr_mode);
+ chip->active_bank = snd_ymfpci_readl(chip, YDSXGR_CTRLSELECT);
+ spin_unlock_irq(&chip->reg_lock);
+ }
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+int __devinit snd_ymfpci_create(struct snd_card *card,
+ struct pci_dev * pci,
+ unsigned short old_legacy_ctrl,
+ struct snd_ymfpci ** rchip)
+{
+ struct snd_ymfpci *chip;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_ymfpci_dev_free,
+ };
+
+ *rchip = NULL;
+
+ /* enable PCI device */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+ chip->old_legacy_ctrl = old_legacy_ctrl;
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->voice_lock);
+ init_waitqueue_head(&chip->interrupt_sleep);
+ atomic_set(&chip->interrupt_sleep_count, 0);
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+ chip->device_id = pci->device;
+ chip->rev = pci->revision;
+ chip->reg_area_phys = pci_resource_start(pci, 0);
+ chip->reg_area_virt = ioremap_nocache(chip->reg_area_phys, 0x8000);
+ pci_set_master(pci);
+ chip->src441_used = -1;
+
+ if ((chip->res_reg_area = request_mem_region(chip->reg_area_phys, 0x8000, "YMFPCI")) == NULL) {
+ snd_printk(KERN_ERR "unable to grab memory region 0x%lx-0x%lx\n", chip->reg_area_phys, chip->reg_area_phys + 0x8000 - 1);
+ snd_ymfpci_free(chip);
+ return -EBUSY;
+ }
+ if (request_irq(pci->irq, snd_ymfpci_interrupt, IRQF_SHARED,
+ "YMFPCI", chip)) {
+ snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+ snd_ymfpci_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ snd_ymfpci_aclink_reset(pci);
+ if (snd_ymfpci_codec_ready(chip, 0) < 0) {
+ snd_ymfpci_free(chip);
+ return -EIO;
+ }
+
+ err = snd_ymfpci_request_firmware(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "firmware request failed: %d\n", err);
+ snd_ymfpci_free(chip);
+ return err;
+ }
+ snd_ymfpci_download_image(chip);
+
+ udelay(100); /* seems we need a delay after downloading image.. */
+
+ if (snd_ymfpci_memalloc(chip) < 0) {
+ snd_ymfpci_free(chip);
+ return -EIO;
+ }
+
+ if ((err = snd_ymfpci_ac3_init(chip)) < 0) {
+ snd_ymfpci_free(chip);
+ return err;
+ }
+
+#ifdef CONFIG_PM
+ chip->saved_regs = vmalloc(YDSXGR_NUM_SAVED_REGS * sizeof(u32));
+ if (chip->saved_regs == NULL) {
+ snd_ymfpci_free(chip);
+ return -ENOMEM;
+ }
+#endif
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_ymfpci_free(chip);
+ return err;
+ }
+
+ snd_ymfpci_proc_init(card, chip);
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+}
diff --git a/sound/pcmcia/Kconfig b/sound/pcmcia/Kconfig
new file mode 100644
index 0000000..7fbb190
--- /dev/null
+++ b/sound/pcmcia/Kconfig
@@ -0,0 +1,33 @@
+# ALSA PCMCIA drivers
+
+menuconfig SND_PCMCIA
+ bool "PCMCIA sound devices"
+ depends on PCMCIA
+ default y
+ help
+ Support for sound devices connected via the PCMCIA bus.
+
+if SND_PCMCIA && PCMCIA
+
+config SND_VXPOCKET
+ tristate "Digigram VXpocket"
+ select SND_VX_LIB
+ help
+ Say Y here to include support for Digigram VXpocket and
+ VXpocket 440 soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-vxpocket.
+
+config SND_PDAUDIOCF
+ tristate "Sound Core PDAudioCF"
+ select SND_PCM
+ help
+ Say Y here to include support for Sound Core PDAudioCF
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-pdaudiocf.
+
+endif # SND_PCMCIA
+
diff --git a/sound/pcmcia/Makefile b/sound/pcmcia/Makefile
new file mode 100644
index 0000000..beef2e3
--- /dev/null
+++ b/sound/pcmcia/Makefile
@@ -0,0 +1,6 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+obj-$(CONFIG_SND) += vx/ pdaudiocf/
diff --git a/sound/pcmcia/pdaudiocf/Makefile b/sound/pcmcia/pdaudiocf/Makefile
new file mode 100644
index 0000000..e892d72
--- /dev/null
+++ b/sound/pcmcia/pdaudiocf/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2004 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-pdaudiocf-objs := pdaudiocf.o pdaudiocf_core.o pdaudiocf_irq.o pdaudiocf_pcm.o
+
+obj-$(CONFIG_SND_PDAUDIOCF) += snd-pdaudiocf.o
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
new file mode 100644
index 0000000..819aaaa
--- /dev/null
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -0,0 +1,314 @@
+/*
+ * Driver for Sound Core PDAudioCF soundcard
+ *
+ * Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/core.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <pcmcia/ciscode.h>
+#include <pcmcia/cisreg.h>
+#include "pdaudiocf.h"
+#include <sound/initval.h>
+#include <linux/init.h>
+
+/*
+ */
+
+#define CARD_NAME "PDAudio-CF"
+
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
+MODULE_DESCRIPTION("Sound Core " CARD_NAME);
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Sound Core," CARD_NAME "}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
+
+/*
+ */
+
+static struct snd_card *card_list[SNDRV_CARDS];
+
+/*
+ * prototypes
+ */
+static int pdacf_config(struct pcmcia_device *link);
+static void snd_pdacf_detach(struct pcmcia_device *p_dev);
+
+static void pdacf_release(struct pcmcia_device *link)
+{
+ pcmcia_disable_device(link);
+}
+
+/*
+ * destructor
+ */
+static int snd_pdacf_free(struct snd_pdacf *pdacf)
+{
+ struct pcmcia_device *link = pdacf->p_dev;
+
+ pdacf_release(link);
+
+ card_list[pdacf->index] = NULL;
+ pdacf->card = NULL;
+
+ kfree(pdacf);
+ return 0;
+}
+
+static int snd_pdacf_dev_free(struct snd_device *device)
+{
+ struct snd_pdacf *chip = device->device_data;
+ return snd_pdacf_free(chip);
+}
+
+/*
+ * snd_pdacf_attach - attach callback for cs
+ */
+static int snd_pdacf_probe(struct pcmcia_device *link)
+{
+ int i;
+ struct snd_pdacf *pdacf;
+ struct snd_card *card;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_pdacf_dev_free,
+ };
+
+ snd_printdd(KERN_DEBUG "pdacf_attach called\n");
+ /* find an empty slot from the card list */
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ if (! card_list[i])
+ break;
+ }
+ if (i >= SNDRV_CARDS) {
+ snd_printk(KERN_ERR "pdacf: too many cards found\n");
+ return -EINVAL;
+ }
+ if (! enable[i])
+ return -ENODEV; /* disabled explicitly */
+
+ /* ok, create a card instance */
+ card = snd_card_new(index[i], id[i], THIS_MODULE, 0);
+ if (card == NULL) {
+ snd_printk(KERN_ERR "pdacf: cannot create a card instance\n");
+ return -ENOMEM;
+ }
+
+ pdacf = snd_pdacf_create(card);
+ if (! pdacf)
+ return -EIO;
+
+ if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, pdacf, &ops) < 0) {
+ kfree(pdacf);
+ snd_card_free(card);
+ return -ENODEV;
+ }
+
+ snd_card_set_dev(card, &handle_to_dev(link));
+
+ pdacf->index = i;
+ card_list[i] = card;
+
+ pdacf->p_dev = link;
+ link->priv = pdacf;
+
+ link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO;
+ link->io.NumPorts1 = 16;
+
+ link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT | IRQ_FORCED_PULSE;
+ // link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED;
+
+ link->irq.IRQInfo1 = 0 /* | IRQ_LEVEL_ID */;
+ link->irq.Handler = pdacf_interrupt;
+ link->irq.Instance = pdacf;
+ link->conf.Attributes = CONF_ENABLE_IRQ;
+ link->conf.IntType = INT_MEMORY_AND_IO;
+ link->conf.ConfigIndex = 1;
+ link->conf.Present = PRESENT_OPTION;
+
+ return pdacf_config(link);
+}
+
+
+/**
+ * snd_pdacf_assign_resources - initialize the hardware and card instance.
+ * @port: i/o port for the card
+ * @irq: irq number for the card
+ *
+ * this function assigns the specified port and irq, boot the card,
+ * create pcm and control instances, and initialize the rest hardware.
+ *
+ * returns 0 if successful, or a negative error code.
+ */
+static int snd_pdacf_assign_resources(struct snd_pdacf *pdacf, int port, int irq)
+{
+ int err;
+ struct snd_card *card = pdacf->card;
+
+ snd_printdd(KERN_DEBUG "pdacf assign resources: port = 0x%x, irq = %d\n", port, irq);
+ pdacf->port = port;
+ pdacf->irq = irq;
+ pdacf->chip_status |= PDAUDIOCF_STAT_IS_CONFIGURED;
+
+ err = snd_pdacf_ak4117_create(pdacf);
+ if (err < 0)
+ return err;
+
+ strcpy(card->driver, "PDAudio-CF");
+ sprintf(card->shortname, "Core Sound %s", card->driver);
+ sprintf(card->longname, "%s at 0x%x, irq %i",
+ card->shortname, port, irq);
+
+ err = snd_pdacf_pcm_new(pdacf);
+ if (err < 0)
+ return err;
+
+ if ((err = snd_card_register(card)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * snd_pdacf_detach - detach callback for cs
+ */
+static void snd_pdacf_detach(struct pcmcia_device *link)
+{
+ struct snd_pdacf *chip = link->priv;
+
+ snd_printdd(KERN_DEBUG "pdacf_detach called\n");
+
+ if (chip->chip_status & PDAUDIOCF_STAT_IS_CONFIGURED)
+ snd_pdacf_powerdown(chip);
+ chip->chip_status |= PDAUDIOCF_STAT_IS_STALE; /* to be sure */
+ snd_card_disconnect(chip->card);
+ snd_card_free_when_closed(chip->card);
+}
+
+/*
+ * configuration callback
+ */
+
+#define CS_CHECK(fn, ret) \
+do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0)
+
+static int pdacf_config(struct pcmcia_device *link)
+{
+ struct snd_pdacf *pdacf = link->priv;
+ int last_fn, last_ret;
+
+ snd_printdd(KERN_DEBUG "pdacf_config called\n");
+ link->conf.ConfigIndex = 0x5;
+
+ CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io));
+ CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq));
+ CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf));
+
+ if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0)
+ goto failed;
+
+ link->dev_node = &pdacf->node;
+ return 0;
+
+cs_failed:
+ cs_error(link, last_fn, last_ret);
+failed:
+ pcmcia_disable_device(link);
+ return -ENODEV;
+}
+
+#ifdef CONFIG_PM
+
+static int pdacf_suspend(struct pcmcia_device *link)
+{
+ struct snd_pdacf *chip = link->priv;
+
+ snd_printdd(KERN_DEBUG "SUSPEND\n");
+ if (chip) {
+ snd_printdd(KERN_DEBUG "snd_pdacf_suspend calling\n");
+ snd_pdacf_suspend(chip, PMSG_SUSPEND);
+ }
+
+ return 0;
+}
+
+static int pdacf_resume(struct pcmcia_device *link)
+{
+ struct snd_pdacf *chip = link->priv;
+
+ snd_printdd(KERN_DEBUG "RESUME\n");
+ if (pcmcia_dev_present(link)) {
+ if (chip) {
+ snd_printdd(KERN_DEBUG "calling snd_pdacf_resume\n");
+ snd_pdacf_resume(chip);
+ }
+ }
+ snd_printdd(KERN_DEBUG "resume done!\n");
+
+ return 0;
+}
+
+#endif
+
+/*
+ * Module entry points
+ */
+static struct pcmcia_device_id snd_pdacf_ids[] = {
+ /* this is too general PCMCIA_DEVICE_MANF_CARD(0x015d, 0x4c45), */
+ PCMCIA_DEVICE_PROD_ID12("Core Sound","PDAudio-CF",0x396d19d2,0x71717b49),
+ PCMCIA_DEVICE_NULL
+};
+MODULE_DEVICE_TABLE(pcmcia, snd_pdacf_ids);
+
+static struct pcmcia_driver pdacf_cs_driver = {
+ .owner = THIS_MODULE,
+ .drv = {
+ .name = "snd-pdaudiocf",
+ },
+ .probe = snd_pdacf_probe,
+ .remove = snd_pdacf_detach,
+ .id_table = snd_pdacf_ids,
+#ifdef CONFIG_PM
+ .suspend = pdacf_suspend,
+ .resume = pdacf_resume,
+#endif
+
+};
+
+static int __init init_pdacf(void)
+{
+ return pcmcia_register_driver(&pdacf_cs_driver);
+}
+
+static void __exit exit_pdacf(void)
+{
+ pcmcia_unregister_driver(&pdacf_cs_driver);
+}
+
+module_init(init_pdacf);
+module_exit(exit_pdacf);
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
new file mode 100644
index 0000000..b060183
--- /dev/null
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -0,0 +1,145 @@
+/*
+ * Driver for Sound Cors PDAudioCF soundcard
+ *
+ * Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __PDAUDIOCF_H
+#define __PDAUDIOCF_H
+
+#include <sound/pcm.h>
+#include <asm/io.h>
+#include <linux/interrupt.h>
+#include <pcmcia/cs_types.h>
+#include <pcmcia/cs.h>
+#include <pcmcia/cistpl.h>
+#include <pcmcia/ds.h>
+
+#include <sound/ak4117.h>
+
+/* PDAUDIOCF registers */
+#define PDAUDIOCF_REG_MD 0x00 /* music data, R/O */
+#define PDAUDIOCF_REG_WDP 0x02 /* write data pointer / 2, R/O */
+#define PDAUDIOCF_REG_RDP 0x04 /* read data pointer / 2, R/O */
+#define PDAUDIOCF_REG_TCR 0x06 /* test control register W/O */
+#define PDAUDIOCF_REG_SCR 0x08 /* status and control, R/W (see bit description) */
+#define PDAUDIOCF_REG_ISR 0x0a /* interrupt status, R/O */
+#define PDAUDIOCF_REG_IER 0x0c /* interrupt enable, R/W */
+#define PDAUDIOCF_REG_AK_IFR 0x0e /* AK interface register, R/W */
+
+/* PDAUDIOCF_REG_TCR */
+#define PDAUDIOCF_ELIMAKMBIT (1<<0) /* simulate AKM music data */
+#define PDAUDIOCF_TESTDATASEL (1<<1) /* test data selection, 0 = 0x55, 1 = pseudo-random */
+
+/* PDAUDIOCF_REG_SCR */
+#define PDAUDIOCF_AK_SBP (1<<0) /* serial port busy flag */
+#define PDAUDIOCF_RST (1<<2) /* FPGA, AKM + SRAM buffer reset */
+#define PDAUDIOCF_PDN (1<<3) /* power down bit */
+#define PDAUDIOCF_CLKDIV0 (1<<4) /* choose 24.576Mhz clock divided by 1,2,3 or 4 */
+#define PDAUDIOCF_CLKDIV1 (1<<5)
+#define PDAUDIOCF_RECORD (1<<6) /* start capturing to SRAM */
+#define PDAUDIOCF_AK_SDD (1<<7) /* music data detected */
+#define PDAUDIOCF_RED_LED_OFF (1<<8) /* red LED off override */
+#define PDAUDIOCF_BLUE_LED_OFF (1<<9) /* blue LED off override */
+#define PDAUDIOCF_DATAFMT0 (1<<10) /* data format bits: 00 = 16-bit, 01 = 18-bit */
+#define PDAUDIOCF_DATAFMT1 (1<<11) /* 10 = 20-bit, 11 = 24-bit, all right justified */
+#define PDAUDIOCF_FPGAREV(x) ((x>>12)&0x0f) /* FPGA revision */
+
+/* PDAUDIOCF_REG_ISR */
+#define PDAUDIOCF_IRQLVL (1<<0) /* Buffer level IRQ */
+#define PDAUDIOCF_IRQOVR (1<<1) /* Overrun IRQ */
+#define PDAUDIOCF_IRQAKM (1<<2) /* AKM IRQ */
+
+/* PDAUDIOCF_REG_IER */
+#define PDAUDIOCF_IRQLVLEN0 (1<<0) /* fill threshold levels; 00 = none, 01 = 1/8th of buffer */
+#define PDAUDIOCF_IRQLVLEN1 (1<<1) /* 10 = 1/4th of buffer, 11 = 1/2th of buffer */
+#define PDAUDIOCF_IRQOVREN (1<<2) /* enable overrun IRQ */
+#define PDAUDIOCF_IRQAKMEN (1<<3) /* enable AKM IRQ */
+#define PDAUDIOCF_BLUEDUTY0 (1<<8) /* blue LED duty cycle; 00 = 100%, 01 = 50% */
+#define PDAUDIOCF_BLUEDUTY1 (1<<9) /* 02 = 25%, 11 = 12% */
+#define PDAUDIOCF_REDDUTY0 (1<<10) /* red LED duty cycle; 00 = 100%, 01 = 50% */
+#define PDAUDIOCF_REDDUTY1 (1<<11) /* 02 = 25%, 11 = 12% */
+#define PDAUDIOCF_BLUESDD (1<<12) /* blue LED against SDD bit */
+#define PDAUDIOCF_BLUEMODULATE (1<<13) /* save power when 100% duty cycle selected */
+#define PDAUDIOCF_REDMODULATE (1<<14) /* save power when 100% duty cycle selected */
+#define PDAUDIOCF_HALFRATE (1<<15) /* slow both LED blinks by half (also spdif detect rate) */
+
+/* chip status */
+#define PDAUDIOCF_STAT_IS_STALE (1<<0)
+#define PDAUDIOCF_STAT_IS_CONFIGURED (1<<1)
+#define PDAUDIOCF_STAT_IS_SUSPENDED (1<<2)
+
+struct snd_pdacf {
+ struct snd_card *card;
+ int index;
+
+ unsigned long port;
+ int irq;
+
+ spinlock_t reg_lock;
+ unsigned short regmap[8];
+ unsigned short suspend_reg_scr;
+ struct tasklet_struct tq;
+
+ spinlock_t ak4117_lock;
+ struct ak4117 *ak4117;
+
+ unsigned int chip_status;
+
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *pcm_substream;
+ unsigned int pcm_running: 1;
+ unsigned int pcm_channels;
+ unsigned int pcm_swab;
+ unsigned int pcm_little;
+ unsigned int pcm_frame;
+ unsigned int pcm_sample;
+ unsigned int pcm_xor;
+ unsigned int pcm_size;
+ unsigned int pcm_period;
+ unsigned int pcm_tdone;
+ unsigned int pcm_hwptr;
+ void *pcm_area;
+
+ /* pcmcia stuff */
+ struct pcmcia_device *p_dev;
+ dev_node_t node;
+};
+
+static inline void pdacf_reg_write(struct snd_pdacf *chip, unsigned char reg, unsigned short val)
+{
+ outw(chip->regmap[reg>>1] = val, chip->port + reg);
+}
+
+static inline unsigned short pdacf_reg_read(struct snd_pdacf *chip, unsigned char reg)
+{
+ return inw(chip->port + reg);
+}
+
+struct snd_pdacf *snd_pdacf_create(struct snd_card *card);
+int snd_pdacf_ak4117_create(struct snd_pdacf *pdacf);
+void snd_pdacf_powerdown(struct snd_pdacf *chip);
+#ifdef CONFIG_PM
+int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state);
+int snd_pdacf_resume(struct snd_pdacf *chip);
+#endif
+int snd_pdacf_pcm_new(struct snd_pdacf *chip);
+irqreturn_t pdacf_interrupt(int irq, void *dev);
+void pdacf_tasklet(unsigned long private_data);
+void pdacf_reinit(struct snd_pdacf *chip, int resume);
+
+#endif /* __PDAUDIOCF_H */
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
new file mode 100644
index 0000000..dfa40b0
--- /dev/null
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
@@ -0,0 +1,290 @@
+/*
+ * Driver for Sound Core PDAudioCF soundcard
+ *
+ * Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include "pdaudiocf.h"
+#include <sound/initval.h>
+
+/*
+ *
+ */
+static unsigned char pdacf_ak4117_read(void *private_data, unsigned char reg)
+{
+ struct snd_pdacf *chip = private_data;
+ unsigned long timeout;
+ unsigned long flags;
+ unsigned char res;
+
+ spin_lock_irqsave(&chip->ak4117_lock, flags);
+ timeout = 1000;
+ while (pdacf_reg_read(chip, PDAUDIOCF_REG_SCR) & PDAUDIOCF_AK_SBP) {
+ udelay(5);
+ if (--timeout == 0) {
+ spin_unlock_irqrestore(&chip->ak4117_lock, flags);
+ snd_printk(KERN_ERR "AK4117 ready timeout (read)\n");
+ return 0;
+ }
+ }
+ pdacf_reg_write(chip, PDAUDIOCF_REG_AK_IFR, (u16)reg << 8);
+ timeout = 1000;
+ while (pdacf_reg_read(chip, PDAUDIOCF_REG_SCR) & PDAUDIOCF_AK_SBP) {
+ udelay(5);
+ if (--timeout == 0) {
+ spin_unlock_irqrestore(&chip->ak4117_lock, flags);
+ snd_printk(KERN_ERR "AK4117 read timeout (read2)\n");
+ return 0;
+ }
+ }
+ res = (unsigned char)pdacf_reg_read(chip, PDAUDIOCF_REG_AK_IFR);
+ spin_unlock_irqrestore(&chip->ak4117_lock, flags);
+ return res;
+}
+
+static void pdacf_ak4117_write(void *private_data, unsigned char reg, unsigned char val)
+{
+ struct snd_pdacf *chip = private_data;
+ unsigned long timeout;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->ak4117_lock, flags);
+ timeout = 1000;
+ while (inw(chip->port + PDAUDIOCF_REG_SCR) & PDAUDIOCF_AK_SBP) {
+ udelay(5);
+ if (--timeout == 0) {
+ spin_unlock_irqrestore(&chip->ak4117_lock, flags);
+ snd_printk(KERN_ERR "AK4117 ready timeout (write)\n");
+ return;
+ }
+ }
+ outw((u16)reg << 8 | val | (1<<13), chip->port + PDAUDIOCF_REG_AK_IFR);
+ spin_unlock_irqrestore(&chip->ak4117_lock, flags);
+}
+
+#if 0
+void pdacf_dump(struct snd_pdacf *chip)
+{
+ printk("PDAUDIOCF DUMP (0x%lx):\n", chip->port);
+ printk("WPD : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_WDP));
+ printk("RDP : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_RDP));
+ printk("TCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_TCR));
+ printk("SCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_SCR));
+ printk("ISR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_ISR));
+ printk("IER : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_IER));
+ printk("AK_IFR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_AK_IFR));
+}
+#endif
+
+static int pdacf_reset(struct snd_pdacf *chip, int powerdown)
+{
+ u16 val;
+
+ val = pdacf_reg_read(chip, PDAUDIOCF_REG_SCR);
+ val |= PDAUDIOCF_PDN;
+ val &= ~PDAUDIOCF_RECORD; /* for sure */
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, val);
+ udelay(5);
+ val |= PDAUDIOCF_RST;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, val);
+ udelay(200);
+ val &= ~PDAUDIOCF_RST;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, val);
+ udelay(5);
+ if (!powerdown) {
+ val &= ~PDAUDIOCF_PDN;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, val);
+ udelay(200);
+ }
+ return 0;
+}
+
+void pdacf_reinit(struct snd_pdacf *chip, int resume)
+{
+ pdacf_reset(chip, 0);
+ if (resume)
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, chip->suspend_reg_scr);
+ snd_ak4117_reinit(chip->ak4117);
+ pdacf_reg_write(chip, PDAUDIOCF_REG_TCR, chip->regmap[PDAUDIOCF_REG_TCR>>1]);
+ pdacf_reg_write(chip, PDAUDIOCF_REG_IER, chip->regmap[PDAUDIOCF_REG_IER>>1]);
+}
+
+static void pdacf_proc_read(struct snd_info_entry * entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_pdacf *chip = entry->private_data;
+ u16 tmp;
+
+ snd_iprintf(buffer, "PDAudioCF\n\n");
+ tmp = pdacf_reg_read(chip, PDAUDIOCF_REG_SCR);
+ snd_iprintf(buffer, "FPGA revision : 0x%x\n", PDAUDIOCF_FPGAREV(tmp));
+
+}
+
+static void pdacf_proc_init(struct snd_pdacf *chip)
+{
+ struct snd_info_entry *entry;
+
+ if (! snd_card_proc_new(chip->card, "pdaudiocf", &entry))
+ snd_info_set_text_ops(entry, chip, pdacf_proc_read);
+}
+
+struct snd_pdacf *snd_pdacf_create(struct snd_card *card)
+{
+ struct snd_pdacf *chip;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return NULL;
+ chip->card = card;
+ spin_lock_init(&chip->reg_lock);
+ spin_lock_init(&chip->ak4117_lock);
+ tasklet_init(&chip->tq, pdacf_tasklet, (unsigned long)chip);
+ card->private_data = chip;
+
+ pdacf_proc_init(chip);
+ return chip;
+}
+
+static void snd_pdacf_ak4117_change(struct ak4117 *ak4117, unsigned char c0, unsigned char c1)
+{
+ struct snd_pdacf *chip = ak4117->change_callback_private;
+ unsigned long flags;
+ u16 val;
+
+ if (!(c0 & AK4117_UNLCK))
+ return;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = chip->regmap[PDAUDIOCF_REG_SCR>>1];
+ if (ak4117->rcs0 & AK4117_UNLCK)
+ val |= PDAUDIOCF_BLUE_LED_OFF;
+ else
+ val &= ~PDAUDIOCF_BLUE_LED_OFF;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+}
+
+int snd_pdacf_ak4117_create(struct snd_pdacf *chip)
+{
+ int err;
+ u16 val;
+ /* design note: if we unmask PLL unlock, parity, valid, audio or auto bit interrupts */
+ /* from AK4117 then INT1 pin from AK4117 will be high all time, because PCMCIA interrupts are */
+ /* egde based and FPGA does logical OR for all interrupt sources, we cannot use these */
+ /* high-rate sources */
+ static unsigned char pgm[5] = {
+ AK4117_XTL_24_576M | AK4117_EXCT, /* AK4117_REG_PWRDN */
+ AK4117_CM_PLL_XTAL | AK4117_PKCS_128fs | AK4117_XCKS_128fs, /* AK4117_REQ_CLOCK */
+ AK4117_EFH_1024LRCLK | AK4117_DIF_24R | AK4117_IPS, /* AK4117_REG_IO */
+ 0xff, /* AK4117_REG_INT0_MASK */
+ AK4117_MAUTO | AK4117_MAUD | AK4117_MULK | AK4117_MPAR | AK4117_MV, /* AK4117_REG_INT1_MASK */
+ };
+
+ err = pdacf_reset(chip, 0);
+ if (err < 0)
+ return err;
+ err = snd_ak4117_create(chip->card, pdacf_ak4117_read, pdacf_ak4117_write, pgm, chip, &chip->ak4117);
+ if (err < 0)
+ return err;
+
+ val = pdacf_reg_read(chip, PDAUDIOCF_REG_TCR);
+#if 1 /* normal operation */
+ val &= ~(PDAUDIOCF_ELIMAKMBIT|PDAUDIOCF_TESTDATASEL);
+#else /* debug */
+ val |= PDAUDIOCF_ELIMAKMBIT;
+ val &= ~PDAUDIOCF_TESTDATASEL;
+#endif
+ pdacf_reg_write(chip, PDAUDIOCF_REG_TCR, val);
+
+ /* setup the FPGA to match AK4117 setup */
+ val = pdacf_reg_read(chip, PDAUDIOCF_REG_SCR);
+ val &= ~(PDAUDIOCF_CLKDIV0 | PDAUDIOCF_CLKDIV1); /* use 24.576Mhz clock */
+ val &= ~(PDAUDIOCF_RED_LED_OFF|PDAUDIOCF_BLUE_LED_OFF);
+ val |= PDAUDIOCF_DATAFMT0 | PDAUDIOCF_DATAFMT1; /* 24-bit data */
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, val);
+
+ /* setup LEDs and IRQ */
+ val = pdacf_reg_read(chip, PDAUDIOCF_REG_IER);
+ val &= ~(PDAUDIOCF_IRQLVLEN0 | PDAUDIOCF_IRQLVLEN1);
+ val &= ~(PDAUDIOCF_BLUEDUTY0 | PDAUDIOCF_REDDUTY0 | PDAUDIOCF_REDDUTY1);
+ val |= PDAUDIOCF_BLUEDUTY1 | PDAUDIOCF_HALFRATE;
+ val |= PDAUDIOCF_IRQOVREN | PDAUDIOCF_IRQAKMEN;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_IER, val);
+
+ chip->ak4117->change_callback_private = chip;
+ chip->ak4117->change_callback = snd_pdacf_ak4117_change;
+
+ /* update LED status */
+ snd_pdacf_ak4117_change(chip->ak4117, AK4117_UNLCK, 0);
+
+ return 0;
+}
+
+void snd_pdacf_powerdown(struct snd_pdacf *chip)
+{
+ u16 val;
+
+ val = pdacf_reg_read(chip, PDAUDIOCF_REG_SCR);
+ chip->suspend_reg_scr = val;
+ val |= PDAUDIOCF_RED_LED_OFF | PDAUDIOCF_BLUE_LED_OFF;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, val);
+ /* disable interrupts, but use direct write to preserve old register value in chip->regmap */
+ val = inw(chip->port + PDAUDIOCF_REG_IER);
+ val &= ~(PDAUDIOCF_IRQOVREN|PDAUDIOCF_IRQAKMEN|PDAUDIOCF_IRQLVLEN0|PDAUDIOCF_IRQLVLEN1);
+ outw(val, chip->port + PDAUDIOCF_REG_IER);
+ pdacf_reset(chip, 1);
+}
+
+#ifdef CONFIG_PM
+
+int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state)
+{
+ u16 val;
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(chip->pcm);
+ /* disable interrupts, but use direct write to preserve old register value in chip->regmap */
+ val = inw(chip->port + PDAUDIOCF_REG_IER);
+ val &= ~(PDAUDIOCF_IRQOVREN|PDAUDIOCF_IRQAKMEN|PDAUDIOCF_IRQLVLEN0|PDAUDIOCF_IRQLVLEN1);
+ outw(val, chip->port + PDAUDIOCF_REG_IER);
+ chip->chip_status |= PDAUDIOCF_STAT_IS_SUSPENDED; /* ignore interrupts from now */
+ snd_pdacf_powerdown(chip);
+ return 0;
+}
+
+static inline int check_signal(struct snd_pdacf *chip)
+{
+ return (chip->ak4117->rcs0 & AK4117_UNLCK) == 0;
+}
+
+int snd_pdacf_resume(struct snd_pdacf *chip)
+{
+ int timeout = 40;
+
+ pdacf_reinit(chip, 1);
+ /* wait for AK4117's PLL */
+ while (timeout-- > 0 &&
+ (snd_ak4117_external_rate(chip->ak4117) <= 0 || !check_signal(chip)))
+ mdelay(1);
+ chip->chip_status &= ~PDAUDIOCF_STAT_IS_SUSPENDED;
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
+ return 0;
+}
+#endif
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
new file mode 100644
index 0000000..fa4b113
--- /dev/null
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
@@ -0,0 +1,325 @@
+/*
+ * Driver for Sound Core PDAudioCF soundcard
+ *
+ * Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/core.h>
+#include "pdaudiocf.h"
+#include <sound/initval.h>
+#include <asm/irq_regs.h>
+
+/*
+ *
+ */
+irqreturn_t pdacf_interrupt(int irq, void *dev)
+{
+ struct snd_pdacf *chip = dev;
+ unsigned short stat;
+
+ if ((chip->chip_status & (PDAUDIOCF_STAT_IS_STALE|
+ PDAUDIOCF_STAT_IS_CONFIGURED|
+ PDAUDIOCF_STAT_IS_SUSPENDED)) != PDAUDIOCF_STAT_IS_CONFIGURED)
+ return IRQ_HANDLED; /* IRQ_NONE here? */
+
+ stat = inw(chip->port + PDAUDIOCF_REG_ISR);
+ if (stat & (PDAUDIOCF_IRQLVL|PDAUDIOCF_IRQOVR)) {
+ if (stat & PDAUDIOCF_IRQOVR) /* should never happen */
+ snd_printk(KERN_ERR "PDAUDIOCF SRAM buffer overrun detected!\n");
+ if (chip->pcm_substream)
+ tasklet_hi_schedule(&chip->tq);
+ if (!(stat & PDAUDIOCF_IRQAKM))
+ stat |= PDAUDIOCF_IRQAKM; /* check rate */
+ }
+ if (get_irq_regs() != NULL)
+ snd_ak4117_check_rate_and_errors(chip->ak4117, 0);
+ return IRQ_HANDLED;
+}
+
+static inline void pdacf_transfer_mono16(u16 *dst, u16 xor, unsigned int size, unsigned long rdp_port)
+{
+ while (size-- > 0) {
+ *dst++ = inw(rdp_port) ^ xor;
+ inw(rdp_port);
+ }
+}
+
+static inline void pdacf_transfer_mono32(u32 *dst, u32 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ inw(rdp_port);
+ *dst++ = ((((u32)val2 & 0xff) << 24) | ((u32)val1 << 8)) ^ xor;
+ }
+}
+
+static inline void pdacf_transfer_stereo16(u16 *dst, u16 xor, unsigned int size, unsigned long rdp_port)
+{
+ while (size-- > 0) {
+ *dst++ = inw(rdp_port) ^ xor;
+ *dst++ = inw(rdp_port) ^ xor;
+ }
+}
+
+static inline void pdacf_transfer_stereo32(u32 *dst, u32 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2, val3;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ val3 = inw(rdp_port);
+ *dst++ = ((((u32)val2 & 0xff) << 24) | ((u32)val1 << 8)) ^ xor;
+ *dst++ = (((u32)val3 << 16) | (val2 & 0xff00)) ^ xor;
+ }
+}
+
+static inline void pdacf_transfer_mono16sw(u16 *dst, u16 xor, unsigned int size, unsigned long rdp_port)
+{
+ while (size-- > 0) {
+ *dst++ = swab16(inw(rdp_port) ^ xor);
+ inw(rdp_port);
+ }
+}
+
+static inline void pdacf_transfer_mono32sw(u32 *dst, u32 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ inw(rdp_port);
+ *dst++ = swab32((((val2 & 0xff) << 24) | ((u32)val1 << 8)) ^ xor);
+ }
+}
+
+static inline void pdacf_transfer_stereo16sw(u16 *dst, u16 xor, unsigned int size, unsigned long rdp_port)
+{
+ while (size-- > 0) {
+ *dst++ = swab16(inw(rdp_port) ^ xor);
+ *dst++ = swab16(inw(rdp_port) ^ xor);
+ }
+}
+
+static inline void pdacf_transfer_stereo32sw(u32 *dst, u32 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2, val3;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ val3 = inw(rdp_port);
+ *dst++ = swab32((((val2 & 0xff) << 24) | ((u32)val1 << 8)) ^ xor);
+ *dst++ = swab32((((u32)val3 << 16) | (val2 & 0xff00)) ^ xor);
+ }
+}
+
+static inline void pdacf_transfer_mono24le(u8 *dst, u16 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2;
+ register u32 xval1;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ inw(rdp_port);
+ xval1 = (((val2 & 0xff) << 8) | (val1 << 16)) ^ xor;
+ *dst++ = (u8)(xval1 >> 8);
+ *dst++ = (u8)(xval1 >> 16);
+ *dst++ = (u8)(xval1 >> 24);
+ }
+}
+
+static inline void pdacf_transfer_mono24be(u8 *dst, u16 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2;
+ register u32 xval1;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ inw(rdp_port);
+ xval1 = (((val2 & 0xff) << 8) | (val1 << 16)) ^ xor;
+ *dst++ = (u8)(xval1 >> 24);
+ *dst++ = (u8)(xval1 >> 16);
+ *dst++ = (u8)(xval1 >> 8);
+ }
+}
+
+static inline void pdacf_transfer_stereo24le(u8 *dst, u32 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2, val3;
+ register u32 xval1, xval2;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ val3 = inw(rdp_port);
+ xval1 = ((((u32)val2 & 0xff) << 24) | ((u32)val1 << 8)) ^ xor;
+ xval2 = (((u32)val3 << 16) | (val2 & 0xff00)) ^ xor;
+ *dst++ = (u8)(xval1 >> 8);
+ *dst++ = (u8)(xval1 >> 16);
+ *dst++ = (u8)(xval1 >> 24);
+ *dst++ = (u8)(xval2 >> 8);
+ *dst++ = (u8)(xval2 >> 16);
+ *dst++ = (u8)(xval2 >> 24);
+ }
+}
+
+static inline void pdacf_transfer_stereo24be(u8 *dst, u32 xor, unsigned int size, unsigned long rdp_port)
+{
+ register u16 val1, val2, val3;
+ register u32 xval1, xval2;
+
+ while (size-- > 0) {
+ val1 = inw(rdp_port);
+ val2 = inw(rdp_port);
+ val3 = inw(rdp_port);
+ xval1 = ((((u32)val2 & 0xff) << 24) | ((u32)val1 << 8)) ^ xor;
+ xval2 = (((u32)val3 << 16) | (val2 & 0xff00)) ^ xor;
+ *dst++ = (u8)(xval1 >> 24);
+ *dst++ = (u8)(xval1 >> 16);
+ *dst++ = (u8)(xval1 >> 8);
+ *dst++ = (u8)(xval2 >> 24);
+ *dst++ = (u8)(xval2 >> 16);
+ *dst++ = (u8)(xval2 >> 8);
+ }
+}
+
+static void pdacf_transfer(struct snd_pdacf *chip, unsigned int size, unsigned int off)
+{
+ unsigned long rdp_port = chip->port + PDAUDIOCF_REG_MD;
+ unsigned int xor = chip->pcm_xor;
+
+ if (chip->pcm_sample == 3) {
+ if (chip->pcm_little) {
+ if (chip->pcm_channels == 1) {
+ pdacf_transfer_mono24le((char *)chip->pcm_area + (off * 3), xor, size, rdp_port);
+ } else {
+ pdacf_transfer_stereo24le((char *)chip->pcm_area + (off * 6), xor, size, rdp_port);
+ }
+ } else {
+ if (chip->pcm_channels == 1) {
+ pdacf_transfer_mono24be((char *)chip->pcm_area + (off * 3), xor, size, rdp_port);
+ } else {
+ pdacf_transfer_stereo24be((char *)chip->pcm_area + (off * 6), xor, size, rdp_port);
+ }
+ }
+ return;
+ }
+ if (chip->pcm_swab == 0) {
+ if (chip->pcm_channels == 1) {
+ if (chip->pcm_frame == 2) {
+ pdacf_transfer_mono16((u16 *)chip->pcm_area + off, xor, size, rdp_port);
+ } else {
+ pdacf_transfer_mono32((u32 *)chip->pcm_area + off, xor, size, rdp_port);
+ }
+ } else {
+ if (chip->pcm_frame == 2) {
+ pdacf_transfer_stereo16((u16 *)chip->pcm_area + (off * 2), xor, size, rdp_port);
+ } else {
+ pdacf_transfer_stereo32((u32 *)chip->pcm_area + (off * 2), xor, size, rdp_port);
+ }
+ }
+ } else {
+ if (chip->pcm_channels == 1) {
+ if (chip->pcm_frame == 2) {
+ pdacf_transfer_mono16sw((u16 *)chip->pcm_area + off, xor, size, rdp_port);
+ } else {
+ pdacf_transfer_mono32sw((u32 *)chip->pcm_area + off, xor, size, rdp_port);
+ }
+ } else {
+ if (chip->pcm_frame == 2) {
+ pdacf_transfer_stereo16sw((u16 *)chip->pcm_area + (off * 2), xor, size, rdp_port);
+ } else {
+ pdacf_transfer_stereo32sw((u32 *)chip->pcm_area + (off * 2), xor, size, rdp_port);
+ }
+ }
+ }
+}
+
+void pdacf_tasklet(unsigned long private_data)
+{
+ struct snd_pdacf *chip = (struct snd_pdacf *) private_data;
+ int size, off, cont, rdp, wdp;
+
+ if ((chip->chip_status & (PDAUDIOCF_STAT_IS_STALE|PDAUDIOCF_STAT_IS_CONFIGURED)) != PDAUDIOCF_STAT_IS_CONFIGURED)
+ return;
+
+ if (chip->pcm_substream == NULL || chip->pcm_substream->runtime == NULL || !snd_pcm_running(chip->pcm_substream))
+ return;
+
+ rdp = inw(chip->port + PDAUDIOCF_REG_RDP);
+ wdp = inw(chip->port + PDAUDIOCF_REG_WDP);
+ // printk("TASKLET: rdp = %x, wdp = %x\n", rdp, wdp);
+ size = wdp - rdp;
+ if (size < 0)
+ size += 0x10000;
+ if (size == 0)
+ size = 0x10000;
+ size /= chip->pcm_frame;
+ if (size > 64)
+ size -= 32;
+
+#if 0
+ chip->pcm_hwptr += size;
+ chip->pcm_hwptr %= chip->pcm_size;
+ chip->pcm_tdone += size;
+ if (chip->pcm_frame == 2) {
+ unsigned long rdp_port = chip->port + PDAUDIOCF_REG_MD;
+ while (size-- > 0) {
+ inw(rdp_port);
+ inw(rdp_port);
+ }
+ } else {
+ unsigned long rdp_port = chip->port + PDAUDIOCF_REG_MD;
+ while (size-- > 0) {
+ inw(rdp_port);
+ inw(rdp_port);
+ inw(rdp_port);
+ }
+ }
+#else
+ off = chip->pcm_hwptr + chip->pcm_tdone;
+ off %= chip->pcm_size;
+ chip->pcm_tdone += size;
+ while (size > 0) {
+ cont = chip->pcm_size - off;
+ if (cont > size)
+ cont = size;
+ pdacf_transfer(chip, cont, off);
+ off += cont;
+ off %= chip->pcm_size;
+ size -= cont;
+ }
+#endif
+ spin_lock(&chip->reg_lock);
+ while (chip->pcm_tdone >= chip->pcm_period) {
+ chip->pcm_hwptr += chip->pcm_period;
+ chip->pcm_hwptr %= chip->pcm_size;
+ chip->pcm_tdone -= chip->pcm_period;
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(chip->pcm_substream);
+ spin_lock(&chip->reg_lock);
+ }
+ spin_unlock(&chip->reg_lock);
+ // printk("TASKLET: end\n");
+}
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
new file mode 100644
index 0000000..01066c9
--- /dev/null
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -0,0 +1,349 @@
+/*
+ * Driver for Sound Core PDAudioCF soundcards
+ *
+ * PCM part
+ *
+ * Copyright (c) 2003 by Jaroslav Kysela <perex@perex.cz>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/slab.h>
+#include <linux/vmalloc.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/asoundef.h>
+#include "pdaudiocf.h"
+
+
+/*
+ * we use a vmalloc'ed (sg-)buffer
+ */
+
+/* get the physical page pointer on the given offset */
+static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, unsigned long offset)
+{
+ void *pageptr = subs->runtime->dma_area + offset;
+ return vmalloc_to_page(pageptr);
+}
+
+/*
+ * hw_params callback
+ * NOTE: this may be called not only once per pcm open!
+ */
+static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ if (runtime->dma_area) {
+ if (runtime->dma_bytes >= size)
+ return 0; /* already enough large */
+ vfree(runtime->dma_area);
+ }
+ runtime->dma_area = vmalloc_32(size);
+ if (! runtime->dma_area)
+ return -ENOMEM;
+ runtime->dma_bytes = size;
+ return 0;
+}
+
+/*
+ * hw_free callback
+ * NOTE: this may be called not only once per pcm open!
+ */
+static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ vfree(runtime->dma_area);
+ runtime->dma_area = NULL;
+ return 0;
+}
+
+/*
+ * clear the SRAM contents
+ */
+static int pdacf_pcm_clear_sram(struct snd_pdacf *chip)
+{
+ int max_loop = 64 * 1024;
+
+ while (inw(chip->port + PDAUDIOCF_REG_RDP) != inw(chip->port + PDAUDIOCF_REG_WDP)) {
+ if (max_loop-- < 0)
+ return -EIO;
+ inw(chip->port + PDAUDIOCF_REG_MD);
+ }
+ return 0;
+}
+
+/*
+ * pdacf_pcm_trigger - trigger callback for capture
+ */
+static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
+{
+ struct snd_pdacf *chip = snd_pcm_substream_chip(subs);
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ int inc, ret = 0, rate;
+ unsigned short mask, val, tmp;
+
+ if (chip->chip_status & PDAUDIOCF_STAT_IS_STALE)
+ return -EBUSY;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ chip->pcm_hwptr = 0;
+ chip->pcm_tdone = 0;
+ /* fall thru */
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ mask = 0;
+ val = PDAUDIOCF_RECORD;
+ inc = 1;
+ rate = snd_ak4117_check_rate_and_errors(chip->ak4117, AK4117_CHECK_NO_STAT|AK4117_CHECK_NO_RATE);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ mask = PDAUDIOCF_RECORD;
+ val = 0;
+ inc = -1;
+ rate = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+ spin_lock(&chip->reg_lock);
+ chip->pcm_running += inc;
+ tmp = pdacf_reg_read(chip, PDAUDIOCF_REG_SCR);
+ if (chip->pcm_running) {
+ if ((chip->ak4117->rcs0 & AK4117_UNLCK) || runtime->rate != rate) {
+ chip->pcm_running -= inc;
+ ret = -EIO;
+ goto __end;
+ }
+ }
+ tmp &= ~mask;
+ tmp |= val;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, tmp);
+ __end:
+ spin_unlock(&chip->reg_lock);
+ snd_ak4117_check_rate_and_errors(chip->ak4117, AK4117_CHECK_NO_RATE);
+ return ret;
+}
+
+/*
+ * pdacf_pcm_hw_params - hw_params callback for playback and capture
+ */
+static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params));
+}
+
+/*
+ * pdacf_pcm_hw_free - hw_free callback for playback and capture
+ */
+static int pdacf_pcm_hw_free(struct snd_pcm_substream *subs)
+{
+ return snd_pcm_free_vmalloc_buffer(subs);
+}
+
+/*
+ * pdacf_pcm_prepare - prepare callback for playback and capture
+ */
+static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
+{
+ struct snd_pdacf *chip = snd_pcm_substream_chip(subs);
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ u16 val, nval, aval;
+
+ if (chip->chip_status & PDAUDIOCF_STAT_IS_STALE)
+ return -EBUSY;
+
+ chip->pcm_channels = runtime->channels;
+
+ chip->pcm_little = snd_pcm_format_little_endian(runtime->format) > 0;
+#ifdef SNDRV_LITTLE_ENDIAN
+ chip->pcm_swab = snd_pcm_format_big_endian(runtime->format) > 0;
+#else
+ chip->pcm_swab = chip->pcm_little;
+#endif
+
+ if (snd_pcm_format_unsigned(runtime->format))
+ chip->pcm_xor = 0x80008000;
+
+ if (pdacf_pcm_clear_sram(chip) < 0)
+ return -EIO;
+
+ val = nval = pdacf_reg_read(chip, PDAUDIOCF_REG_SCR);
+ nval &= ~(PDAUDIOCF_DATAFMT0|PDAUDIOCF_DATAFMT1);
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_S16_BE:
+ break;
+ default: /* 24-bit */
+ nval |= PDAUDIOCF_DATAFMT0 | PDAUDIOCF_DATAFMT1;
+ break;
+ }
+ aval = 0;
+ chip->pcm_sample = 4;
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_S16_BE:
+ aval = AK4117_DIF_16R;
+ chip->pcm_frame = 2;
+ chip->pcm_sample = 2;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ case SNDRV_PCM_FORMAT_S24_3BE:
+ chip->pcm_sample = 3;
+ /* fall through */
+ default: /* 24-bit */
+ aval = AK4117_DIF_24R;
+ chip->pcm_frame = 3;
+ chip->pcm_xor &= 0xffff0000;
+ break;
+ }
+
+ if (val != nval) {
+ snd_ak4117_reg_write(chip->ak4117, AK4117_REG_IO, AK4117_DIF2|AK4117_DIF1|AK4117_DIF0, aval);
+ pdacf_reg_write(chip, PDAUDIOCF_REG_SCR, nval);
+ }
+
+ val = pdacf_reg_read(chip, PDAUDIOCF_REG_IER);
+ val &= ~(PDAUDIOCF_IRQLVLEN1);
+ val |= PDAUDIOCF_IRQLVLEN0;
+ pdacf_reg_write(chip, PDAUDIOCF_REG_IER, val);
+
+ chip->pcm_size = runtime->buffer_size;
+ chip->pcm_period = runtime->period_size;
+ chip->pcm_area = runtime->dma_area;
+
+ return 0;
+}
+
+
+/*
+ * capture hw information
+ */
+
+static struct snd_pcm_hardware pdacf_pcm_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
+ .rates = SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000,
+ .rate_min = 32000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (512*1024),
+ .period_bytes_min = 8*1024,
+ .period_bytes_max = (64*1024),
+ .periods_min = 2,
+ .periods_max = 128,
+ .fifo_size = 0,
+};
+
+
+/*
+ * pdacf_pcm_capture_open - open callback for capture
+ */
+static int pdacf_pcm_capture_open(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ struct snd_pdacf *chip = snd_pcm_substream_chip(subs);
+
+ if (chip->chip_status & PDAUDIOCF_STAT_IS_STALE)
+ return -EBUSY;
+
+ runtime->hw = pdacf_pcm_capture_hw;
+ runtime->private_data = chip;
+ chip->pcm_substream = subs;
+
+ return 0;
+}
+
+/*
+ * pdacf_pcm_capture_close - close callback for capture
+ */
+static int pdacf_pcm_capture_close(struct snd_pcm_substream *subs)
+{
+ struct snd_pdacf *chip = snd_pcm_substream_chip(subs);
+
+ if (!chip)
+ return -EINVAL;
+ pdacf_reinit(chip, 0);
+ chip->pcm_substream = NULL;
+ return 0;
+}
+
+
+/*
+ * pdacf_pcm_capture_pointer - pointer callback for capture
+ */
+static snd_pcm_uframes_t pdacf_pcm_capture_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_pdacf *chip = snd_pcm_substream_chip(subs);
+ return chip->pcm_hwptr;
+}
+
+/*
+ * operators for PCM capture
+ */
+static struct snd_pcm_ops pdacf_pcm_capture_ops = {
+ .open = pdacf_pcm_capture_open,
+ .close = pdacf_pcm_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pdacf_pcm_hw_params,
+ .hw_free = pdacf_pcm_hw_free,
+ .prepare = pdacf_pcm_prepare,
+ .trigger = pdacf_pcm_trigger,
+ .pointer = pdacf_pcm_capture_pointer,
+ .page = snd_pcm_get_vmalloc_page,
+};
+
+
+/*
+ * snd_pdacf_pcm_new - create and initialize a pcm
+ */
+int snd_pdacf_pcm_new(struct snd_pdacf *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(chip->card, "PDAudioCF", 0, 0, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pdacf_pcm_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcm = pcm;
+
+ err = snd_ak4117_build(chip->ak4117, pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
diff --git a/sound/pcmcia/vx/Makefile b/sound/pcmcia/vx/Makefile
new file mode 100644
index 0000000..2bb42ea
--- /dev/null
+++ b/sound/pcmcia/vx/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-vxpocket-objs := vxpocket.o vxp_ops.o vxp_mixer.o
+
+obj-$(CONFIG_SND_VXPOCKET) += snd-vxpocket.o
diff --git a/sound/pcmcia/vx/vxp_mixer.c b/sound/pcmcia/vx/vxp_mixer.c
new file mode 100644
index 0000000..a4a6642
--- /dev/null
+++ b/sound/pcmcia/vx/vxp_mixer.c
@@ -0,0 +1,151 @@
+/*
+ * Driver for Digigram VXpocket soundcards
+ *
+ * VX-pocket mixer
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include "vxpocket.h"
+
+#define MIC_LEVEL_MIN 0
+#define MIC_LEVEL_MAX 8
+
+/*
+ * mic level control (for VXPocket)
+ */
+static int vx_mic_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = MIC_LEVEL_MAX;
+ return 0;
+}
+
+static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+ ucontrol->value.integer.value[0] = chip->mic_level;
+ return 0;
+}
+
+static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+ unsigned int val = ucontrol->value.integer.value[0];
+
+ if (val > MIC_LEVEL_MAX)
+ return -EINVAL;
+ mutex_lock(&_chip->mixer_mutex);
+ if (chip->mic_level != ucontrol->value.integer.value[0]) {
+ vx_set_mic_level(_chip, ucontrol->value.integer.value[0]);
+ chip->mic_level = ucontrol->value.integer.value[0];
+ mutex_unlock(&_chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&_chip->mixer_mutex);
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(db_scale_mic, -21, 3, 0);
+
+static struct snd_kcontrol_new vx_control_mic_level = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ),
+ .name = "Mic Capture Volume",
+ .info = vx_mic_level_info,
+ .get = vx_mic_level_get,
+ .put = vx_mic_level_put,
+ .tlv = { .p = db_scale_mic },
+};
+
+/*
+ * mic boost level control (for VXP440)
+ */
+#define vx_mic_boost_info snd_ctl_boolean_mono_info
+
+static int vx_mic_boost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+ ucontrol->value.integer.value[0] = chip->mic_level;
+ return 0;
+}
+
+static int vx_mic_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct vx_core *_chip = snd_kcontrol_chip(kcontrol);
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+ int val = !!ucontrol->value.integer.value[0];
+ mutex_lock(&_chip->mixer_mutex);
+ if (chip->mic_level != val) {
+ vx_set_mic_boost(_chip, val);
+ chip->mic_level = val;
+ mutex_unlock(&_chip->mixer_mutex);
+ return 1;
+ }
+ mutex_unlock(&_chip->mixer_mutex);
+ return 0;
+}
+
+static struct snd_kcontrol_new vx_control_mic_boost = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Boost",
+ .info = vx_mic_boost_info,
+ .get = vx_mic_boost_get,
+ .put = vx_mic_boost_put,
+};
+
+
+int vxp_add_mic_controls(struct vx_core *_chip)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+ int err;
+
+ /* mute input levels */
+ chip->mic_level = 0;
+ switch (_chip->type) {
+ case VX_TYPE_VXPOCKET:
+ vx_set_mic_level(_chip, 0);
+ break;
+ case VX_TYPE_VXP440:
+ vx_set_mic_boost(_chip, 0);
+ break;
+ }
+
+ /* mic level */
+ switch (_chip->type) {
+ case VX_TYPE_VXPOCKET:
+ if ((err = snd_ctl_add(_chip->card, snd_ctl_new1(&vx_control_mic_level, chip))) < 0)
+ return err;
+ break;
+ case VX_TYPE_VXP440:
+ if ((err = snd_ctl_add(_chip->card, snd_ctl_new1(&vx_control_mic_boost, chip))) < 0)
+ return err;
+ break;
+ }
+
+ return 0;
+}
+
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
new file mode 100644
index 0000000..989e04a
--- /dev/null
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -0,0 +1,614 @@
+/*
+ * Driver for Digigram VXpocket soundcards
+ *
+ * lowlevel routines for VXpocket soundcards
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <asm/io.h>
+#include "vxpocket.h"
+
+
+static int vxp_reg_offset[VX_REG_MAX] = {
+ [VX_ICR] = 0x00, // ICR
+ [VX_CVR] = 0x01, // CVR
+ [VX_ISR] = 0x02, // ISR
+ [VX_IVR] = 0x03, // IVR
+ [VX_RXH] = 0x05, // RXH
+ [VX_RXM] = 0x06, // RXM
+ [VX_RXL] = 0x07, // RXL
+ [VX_DMA] = 0x04, // DMA
+ [VX_CDSP] = 0x08, // CDSP
+ [VX_LOFREQ] = 0x09, // LFREQ
+ [VX_HIFREQ] = 0x0a, // HFREQ
+ [VX_DATA] = 0x0b, // DATA
+ [VX_MICRO] = 0x0c, // MICRO
+ [VX_DIALOG] = 0x0d, // DIALOG
+ [VX_CSUER] = 0x0e, // CSUER
+ [VX_RUER] = 0x0f, // RUER
+};
+
+
+static inline unsigned long vxp_reg_addr(struct vx_core *_chip, int reg)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+ return chip->port + vxp_reg_offset[reg];
+}
+
+/*
+ * snd_vx_inb - read a byte from the register
+ * @offset: register offset
+ */
+static unsigned char vxp_inb(struct vx_core *chip, int offset)
+{
+ return inb(vxp_reg_addr(chip, offset));
+}
+
+/*
+ * snd_vx_outb - write a byte on the register
+ * @offset: the register offset
+ * @val: the value to write
+ */
+static void vxp_outb(struct vx_core *chip, int offset, unsigned char val)
+{
+ outb(val, vxp_reg_addr(chip, offset));
+}
+
+/*
+ * redefine macros to call directly
+ */
+#undef vx_inb
+#define vx_inb(chip,reg) vxp_inb((struct vx_core *)(chip), VX_##reg)
+#undef vx_outb
+#define vx_outb(chip,reg,val) vxp_outb((struct vx_core *)(chip), VX_##reg,val)
+
+
+/*
+ * vx_check_magic - check the magic word on xilinx
+ *
+ * returns zero if a magic word is detected, or a negative error code.
+ */
+static int vx_check_magic(struct vx_core *chip)
+{
+ unsigned long end_time = jiffies + HZ / 5;
+ int c;
+ do {
+ c = vx_inb(chip, CDSP);
+ if (c == CDSP_MAGIC)
+ return 0;
+ msleep(10);
+ } while (time_after_eq(end_time, jiffies));
+ snd_printk(KERN_ERR "cannot find xilinx magic word (%x)\n", c);
+ return -EIO;
+}
+
+
+/*
+ * vx_reset_dsp - reset the DSP
+ */
+
+#define XX_DSP_RESET_WAIT_TIME 2 /* ms */
+
+static void vxp_reset_dsp(struct vx_core *_chip)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ /* set the reset dsp bit to 1 */
+ vx_outb(chip, CDSP, chip->regCDSP | VXP_CDSP_DSP_RESET_MASK);
+ vx_inb(chip, CDSP);
+ mdelay(XX_DSP_RESET_WAIT_TIME);
+ /* reset the bit */
+ chip->regCDSP &= ~VXP_CDSP_DSP_RESET_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+ vx_inb(chip, CDSP);
+ mdelay(XX_DSP_RESET_WAIT_TIME);
+}
+
+/*
+ * reset codec bit
+ */
+static void vxp_reset_codec(struct vx_core *_chip)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ /* Set the reset CODEC bit to 1. */
+ vx_outb(chip, CDSP, chip->regCDSP | VXP_CDSP_CODEC_RESET_MASK);
+ vx_inb(chip, CDSP);
+ msleep(10);
+ /* Set the reset CODEC bit to 0. */
+ chip->regCDSP &= ~VXP_CDSP_CODEC_RESET_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+ vx_inb(chip, CDSP);
+ msleep(1);
+}
+
+/*
+ * vx_load_xilinx_binary - load the xilinx binary image
+ * the binary image is the binary array converted from the bitstream file.
+ */
+static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware *fw)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+ unsigned int i;
+ int c;
+ int regCSUER, regRUER;
+ const unsigned char *image;
+ unsigned char data;
+
+ /* Switch to programmation mode */
+ chip->regDIALOG |= VXP_DLG_XILINX_REPROG_MASK;
+ vx_outb(chip, DIALOG, chip->regDIALOG);
+
+ /* Save register CSUER and RUER */
+ regCSUER = vx_inb(chip, CSUER);
+ regRUER = vx_inb(chip, RUER);
+
+ /* reset HF0 and HF1 */
+ vx_outb(chip, ICR, 0);
+
+ /* Wait for answer HF2 equal to 1 */
+ snd_printdd(KERN_DEBUG "check ISR_HF2\n");
+ if (vx_check_isr(_chip, ISR_HF2, ISR_HF2, 20) < 0)
+ goto _error;
+
+ /* set HF1 for loading xilinx binary */
+ vx_outb(chip, ICR, ICR_HF1);
+ image = fw->data;
+ for (i = 0; i < fw->size; i++, image++) {
+ data = *image;
+ if (vx_wait_isr_bit(_chip, ISR_TX_EMPTY) < 0)
+ goto _error;
+ vx_outb(chip, TXL, data);
+ /* wait for reading */
+ if (vx_wait_for_rx_full(_chip) < 0)
+ goto _error;
+ c = vx_inb(chip, RXL);
+ if (c != (int)data)
+ snd_printk(KERN_ERR "vxpocket: load xilinx mismatch at %d: 0x%x != 0x%x\n", i, c, (int)data);
+ }
+
+ /* reset HF1 */
+ vx_outb(chip, ICR, 0);
+
+ /* wait for HF3 */
+ if (vx_check_isr(_chip, ISR_HF3, ISR_HF3, 20) < 0)
+ goto _error;
+
+ /* read the number of bytes received */
+ if (vx_wait_for_rx_full(_chip) < 0)
+ goto _error;
+
+ c = (int)vx_inb(chip, RXH) << 16;
+ c |= (int)vx_inb(chip, RXM) << 8;
+ c |= vx_inb(chip, RXL);
+
+ snd_printdd(KERN_DEBUG "xilinx: dsp size received 0x%x, orig 0x%Zx\n", c, fw->size);
+
+ vx_outb(chip, ICR, ICR_HF0);
+
+ /* TEMPO 250ms : wait until Xilinx is downloaded */
+ msleep(300);
+
+ /* test magical word */
+ if (vx_check_magic(_chip) < 0)
+ goto _error;
+
+ /* Restore register 0x0E and 0x0F (thus replacing COR and FCSR) */
+ vx_outb(chip, CSUER, regCSUER);
+ vx_outb(chip, RUER, regRUER);
+
+ /* Reset the Xilinx's signal enabling IO access */
+ chip->regDIALOG |= VXP_DLG_XILINX_REPROG_MASK;
+ vx_outb(chip, DIALOG, chip->regDIALOG);
+ vx_inb(chip, DIALOG);
+ msleep(10);
+ chip->regDIALOG &= ~VXP_DLG_XILINX_REPROG_MASK;
+ vx_outb(chip, DIALOG, chip->regDIALOG);
+ vx_inb(chip, DIALOG);
+
+ /* Reset of the Codec */
+ vxp_reset_codec(_chip);
+ vx_reset_dsp(_chip);
+
+ return 0;
+
+ _error:
+ vx_outb(chip, CSUER, regCSUER);
+ vx_outb(chip, RUER, regRUER);
+ chip->regDIALOG &= ~VXP_DLG_XILINX_REPROG_MASK;
+ vx_outb(chip, DIALOG, chip->regDIALOG);
+ return -EIO;
+}
+
+
+/*
+ * vxp_load_dsp - load_dsp callback
+ */
+static int vxp_load_dsp(struct vx_core *vx, int index, const struct firmware *fw)
+{
+ int err;
+
+ switch (index) {
+ case 0:
+ /* xilinx boot */
+ if ((err = vx_check_magic(vx)) < 0)
+ return err;
+ if ((err = snd_vx_load_boot_image(vx, fw)) < 0)
+ return err;
+ return 0;
+ case 1:
+ /* xilinx image */
+ return vxp_load_xilinx_binary(vx, fw);
+ case 2:
+ /* DSP boot */
+ return snd_vx_dsp_boot(vx, fw);
+ case 3:
+ /* DSP image */
+ return snd_vx_dsp_load(vx, fw);
+ default:
+ snd_BUG();
+ return -EINVAL;
+ }
+}
+
+
+/*
+ * vx_test_and_ack - test and acknowledge interrupt
+ *
+ * called from irq hander, too
+ *
+ * spinlock held!
+ */
+static int vxp_test_and_ack(struct vx_core *_chip)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ /* not booted yet? */
+ if (! (_chip->chip_status & VX_STAT_XILINX_LOADED))
+ return -ENXIO;
+
+ if (! (vx_inb(chip, DIALOG) & VXP_DLG_MEMIRQ_MASK))
+ return -EIO;
+
+ /* ok, interrupts generated, now ack it */
+ /* set ACQUIT bit up and down */
+ vx_outb(chip, DIALOG, chip->regDIALOG | VXP_DLG_ACK_MEMIRQ_MASK);
+ /* useless read just to spend some time and maintain
+ * the ACQUIT signal up for a while ( a bus cycle )
+ */
+ vx_inb(chip, DIALOG);
+ vx_outb(chip, DIALOG, chip->regDIALOG & ~VXP_DLG_ACK_MEMIRQ_MASK);
+
+ return 0;
+}
+
+
+/*
+ * vx_validate_irq - enable/disable IRQ
+ */
+static void vxp_validate_irq(struct vx_core *_chip, int enable)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ /* Set the interrupt enable bit to 1 in CDSP register */
+ if (enable)
+ chip->regCDSP |= VXP_CDSP_VALID_IRQ_MASK;
+ else
+ chip->regCDSP &= ~VXP_CDSP_VALID_IRQ_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+}
+
+/*
+ * vx_setup_pseudo_dma - set up the pseudo dma read/write mode.
+ * @do_write: 0 = read, 1 = set up for DMA write
+ */
+static void vx_setup_pseudo_dma(struct vx_core *_chip, int do_write)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ /* Interrupt mode and HREQ pin enabled for host transmit / receive data transfers */
+ vx_outb(chip, ICR, do_write ? ICR_TREQ : ICR_RREQ);
+ /* Reset the pseudo-dma register */
+ vx_inb(chip, ISR);
+ vx_outb(chip, ISR, 0);
+
+ /* Select DMA in read/write transfer mode and in 16-bit accesses */
+ chip->regDIALOG |= VXP_DLG_DMA16_SEL_MASK;
+ chip->regDIALOG |= do_write ? VXP_DLG_DMAWRITE_SEL_MASK : VXP_DLG_DMAREAD_SEL_MASK;
+ vx_outb(chip, DIALOG, chip->regDIALOG);
+
+}
+
+/*
+ * vx_release_pseudo_dma - disable the pseudo-DMA mode
+ */
+static void vx_release_pseudo_dma(struct vx_core *_chip)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ /* Disable DMA and 16-bit accesses */
+ chip->regDIALOG &= ~(VXP_DLG_DMAWRITE_SEL_MASK|
+ VXP_DLG_DMAREAD_SEL_MASK|
+ VXP_DLG_DMA16_SEL_MASK);
+ vx_outb(chip, DIALOG, chip->regDIALOG);
+ /* HREQ pin disabled. */
+ vx_outb(chip, ICR, 0);
+}
+
+/*
+ * vx_pseudo_dma_write - write bulk data on pseudo-DMA mode
+ * @count: data length to transfer in bytes
+ *
+ * data size must be aligned to 6 bytes to ensure the 24bit alignment on DSP.
+ * NB: call with a certain lock!
+ */
+static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count)
+{
+ long port = vxp_reg_addr(chip, VX_DMA);
+ int offset = pipe->hw_ptr;
+ unsigned short *addr = (unsigned short *)(runtime->dma_area + offset);
+
+ vx_setup_pseudo_dma(chip, 1);
+ if (offset + count > pipe->buffer_bytes) {
+ int length = pipe->buffer_bytes - offset;
+ count -= length;
+ length >>= 1; /* in 16bit words */
+ /* Transfer using pseudo-dma. */
+ while (length-- > 0) {
+ outw(cpu_to_le16(*addr), port);
+ addr++;
+ }
+ addr = (unsigned short *)runtime->dma_area;
+ pipe->hw_ptr = 0;
+ }
+ pipe->hw_ptr += count;
+ count >>= 1; /* in 16bit words */
+ /* Transfer using pseudo-dma. */
+ while (count-- > 0) {
+ outw(cpu_to_le16(*addr), port);
+ addr++;
+ }
+ vx_release_pseudo_dma(chip);
+}
+
+
+/*
+ * vx_pseudo_dma_read - read bulk data on pseudo DMA mode
+ * @offset: buffer offset in bytes
+ * @count: data length to transfer in bytes
+ *
+ * the read length must be aligned to 6 bytes, as well as write.
+ * NB: call with a certain lock!
+ */
+static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
+ struct vx_pipe *pipe, int count)
+{
+ struct snd_vxpocket *pchip = (struct snd_vxpocket *)chip;
+ long port = vxp_reg_addr(chip, VX_DMA);
+ int offset = pipe->hw_ptr;
+ unsigned short *addr = (unsigned short *)(runtime->dma_area + offset);
+
+ if (snd_BUG_ON(count % 2))
+ return;
+ vx_setup_pseudo_dma(chip, 0);
+ if (offset + count > pipe->buffer_bytes) {
+ int length = pipe->buffer_bytes - offset;
+ count -= length;
+ length >>= 1; /* in 16bit words */
+ /* Transfer using pseudo-dma. */
+ while (length-- > 0)
+ *addr++ = le16_to_cpu(inw(port));
+ addr = (unsigned short *)runtime->dma_area;
+ pipe->hw_ptr = 0;
+ }
+ pipe->hw_ptr += count;
+ count >>= 1; /* in 16bit words */
+ /* Transfer using pseudo-dma. */
+ while (count-- > 1)
+ *addr++ = le16_to_cpu(inw(port));
+ /* Disable DMA */
+ pchip->regDIALOG &= ~VXP_DLG_DMAREAD_SEL_MASK;
+ vx_outb(chip, DIALOG, pchip->regDIALOG);
+ /* Read the last word (16 bits) */
+ *addr = le16_to_cpu(inw(port));
+ /* Disable 16-bit accesses */
+ pchip->regDIALOG &= ~VXP_DLG_DMA16_SEL_MASK;
+ vx_outb(chip, DIALOG, pchip->regDIALOG);
+ /* HREQ pin disabled. */
+ vx_outb(chip, ICR, 0);
+}
+
+
+/*
+ * write a codec data (24bit)
+ */
+static void vxp_write_codec_reg(struct vx_core *chip, int codec, unsigned int data)
+{
+ int i;
+
+ /* Activate access to the corresponding codec register */
+ if (! codec)
+ vx_inb(chip, LOFREQ);
+ else
+ vx_inb(chip, CODEC2);
+
+ /* We have to send 24 bits (3 x 8 bits). Start with most signif. Bit */
+ for (i = 0; i < 24; i++, data <<= 1)
+ vx_outb(chip, DATA, ((data & 0x800000) ? VX_DATA_CODEC_MASK : 0));
+
+ /* Terminate access to codec registers */
+ vx_inb(chip, HIFREQ);
+}
+
+
+/*
+ * vx_set_mic_boost - set mic boost level (on vxp440 only)
+ * @boost: 0 = 20dB, 1 = +38dB
+ */
+void vx_set_mic_boost(struct vx_core *chip, int boost)
+{
+ struct snd_vxpocket *pchip = (struct snd_vxpocket *)chip;
+ unsigned long flags;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ if (pchip->regCDSP & P24_CDSP_MICS_SEL_MASK) {
+ if (boost) {
+ /* boost: 38 dB */
+ pchip->regCDSP &= ~P24_CDSP_MIC20_SEL_MASK;
+ pchip->regCDSP |= P24_CDSP_MIC38_SEL_MASK;
+ } else {
+ /* minimum value: 20 dB */
+ pchip->regCDSP |= P24_CDSP_MIC20_SEL_MASK;
+ pchip->regCDSP &= ~P24_CDSP_MIC38_SEL_MASK;
+ }
+ vx_outb(chip, CDSP, pchip->regCDSP);
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+/*
+ * remap the linear value (0-8) to the actual value (0-15)
+ */
+static int vx_compute_mic_level(int level)
+{
+ switch (level) {
+ case 5: level = 6 ; break;
+ case 6: level = 8 ; break;
+ case 7: level = 11; break;
+ case 8: level = 15; break;
+ default: break ;
+ }
+ return level;
+}
+
+/*
+ * vx_set_mic_level - set mic level (on vxpocket only)
+ * @level: the mic level = 0 - 8 (max)
+ */
+void vx_set_mic_level(struct vx_core *chip, int level)
+{
+ struct snd_vxpocket *pchip = (struct snd_vxpocket *)chip;
+ unsigned long flags;
+
+ if (chip->chip_status & VX_STAT_IS_STALE)
+ return;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ if (pchip->regCDSP & VXP_CDSP_MIC_SEL_MASK) {
+ level = vx_compute_mic_level(level);
+ vx_outb(chip, MICRO, level);
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+/*
+ * change the input audio source
+ */
+static void vxp_change_audio_source(struct vx_core *_chip, int src)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ switch (src) {
+ case VX_AUDIO_SRC_DIGITAL:
+ chip->regCDSP |= VXP_CDSP_DATAIN_SEL_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+ break;
+ case VX_AUDIO_SRC_LINE:
+ chip->regCDSP &= ~VXP_CDSP_DATAIN_SEL_MASK;
+ if (_chip->type == VX_TYPE_VXP440)
+ chip->regCDSP &= ~P24_CDSP_MICS_SEL_MASK;
+ else
+ chip->regCDSP &= ~VXP_CDSP_MIC_SEL_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+ break;
+ case VX_AUDIO_SRC_MIC:
+ chip->regCDSP &= ~VXP_CDSP_DATAIN_SEL_MASK;
+ /* reset mic levels */
+ if (_chip->type == VX_TYPE_VXP440) {
+ chip->regCDSP &= ~P24_CDSP_MICS_SEL_MASK;
+ if (chip->mic_level)
+ chip->regCDSP |= P24_CDSP_MIC38_SEL_MASK;
+ else
+ chip->regCDSP |= P24_CDSP_MIC20_SEL_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+ } else {
+ chip->regCDSP |= VXP_CDSP_MIC_SEL_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+ vx_outb(chip, MICRO, vx_compute_mic_level(chip->mic_level));
+ }
+ break;
+ }
+}
+
+/*
+ * change the clock source
+ * source = INTERNAL_QUARTZ or UER_SYNC
+ */
+static void vxp_set_clock_source(struct vx_core *_chip, int source)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ if (source == INTERNAL_QUARTZ)
+ chip->regCDSP &= ~VXP_CDSP_CLOCKIN_SEL_MASK;
+ else
+ chip->regCDSP |= VXP_CDSP_CLOCKIN_SEL_MASK;
+ vx_outb(chip, CDSP, chip->regCDSP);
+}
+
+
+/*
+ * reset the board
+ */
+static void vxp_reset_board(struct vx_core *_chip, int cold_reset)
+{
+ struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip;
+
+ chip->regCDSP = 0;
+ chip->regDIALOG = 0;
+}
+
+
+/*
+ * callbacks
+ */
+/* exported */
+struct snd_vx_ops snd_vxpocket_ops = {
+ .in8 = vxp_inb,
+ .out8 = vxp_outb,
+ .test_and_ack = vxp_test_and_ack,
+ .validate_irq = vxp_validate_irq,
+ .write_codec = vxp_write_codec_reg,
+ .reset_codec = vxp_reset_codec,
+ .change_audio_source = vxp_change_audio_source,
+ .set_clock_source = vxp_set_clock_source,
+ .load_dsp = vxp_load_dsp,
+ .add_controls = vxp_add_mic_controls,
+ .reset_dsp = vxp_reset_dsp,
+ .reset_board = vxp_reset_board,
+ .dma_write = vxp_dma_write,
+ .dma_read = vxp_dma_read,
+};
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
new file mode 100644
index 0000000..706602a
--- /dev/null
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -0,0 +1,384 @@
+/*
+ * Driver for Digigram VXpocket V2/440 soundcards
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <linux/init.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include "vxpocket.h"
+#include <pcmcia/ciscode.h>
+#include <pcmcia/cisreg.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+/*
+ */
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("Digigram VXPocket");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Digigram,VXPocket},{Digigram,VXPocket440}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */
+static int ibl[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for VXPocket soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for VXPocket soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable VXPocket soundcard.");
+module_param_array(ibl, int, NULL, 0444);
+MODULE_PARM_DESC(ibl, "Capture IBL size for VXPocket soundcard.");
+
+
+/*
+ */
+
+static unsigned int card_alloc;
+
+
+/*
+ */
+static void vxpocket_release(struct pcmcia_device *link)
+{
+ pcmcia_disable_device(link);
+}
+
+/*
+ * destructor, called from snd_card_free_when_closed()
+ */
+static int snd_vxpocket_dev_free(struct snd_device *device)
+{
+ struct vx_core *chip = device->device_data;
+
+ snd_vx_free_firmware(chip);
+ kfree(chip);
+ return 0;
+}
+
+
+/*
+ * Hardware information
+ */
+
+/* VX-pocket V2
+ *
+ * 1 DSP, 1 sync UER
+ * 1 programmable clock (NIY)
+ * 1 stereo analog input (line/micro)
+ * 1 stereo analog output
+ * Only output levels can be modified
+ */
+
+static const DECLARE_TLV_DB_SCALE(db_scale_old_vol, -11350, 50, 0);
+
+static struct snd_vx_hardware vxpocket_hw = {
+ .name = "VXPocket",
+ .type = VX_TYPE_VXPOCKET,
+
+ /* hardware specs */
+ .num_codecs = 1,
+ .num_ins = 1,
+ .num_outs = 1,
+ .output_level_max = VX_ANALOG_OUT_LEVEL_MAX,
+ .output_level_db_scale = db_scale_old_vol,
+};
+
+/* VX-pocket 440
+ *
+ * 1 DSP, 1 sync UER, 1 sync World Clock (NIY)
+ * SMPTE (NIY)
+ * 2 stereo analog input (line/micro)
+ * 2 stereo analog output
+ * Only output levels can be modified
+ * UER, but only for the first two inputs and outputs.
+ */
+
+static struct snd_vx_hardware vxp440_hw = {
+ .name = "VXPocket440",
+ .type = VX_TYPE_VXP440,
+
+ /* hardware specs */
+ .num_codecs = 2,
+ .num_ins = 2,
+ .num_outs = 2,
+ .output_level_max = VX_ANALOG_OUT_LEVEL_MAX,
+ .output_level_db_scale = db_scale_old_vol,
+};
+
+
+/*
+ * create vxpocket instance
+ */
+static struct snd_vxpocket *snd_vxpocket_new(struct snd_card *card, int ibl,
+ struct pcmcia_device *link)
+{
+ struct vx_core *chip;
+ struct snd_vxpocket *vxp;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_vxpocket_dev_free,
+ };
+
+ chip = snd_vx_create(card, &vxpocket_hw, &snd_vxpocket_ops,
+ sizeof(struct snd_vxpocket) - sizeof(struct vx_core));
+ if (! chip)
+ return NULL;
+
+ if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops) < 0) {
+ kfree(chip);
+ return NULL;
+ }
+ chip->ibl.size = ibl;
+
+ vxp = (struct snd_vxpocket *)chip;
+
+ vxp->p_dev = link;
+ link->priv = chip;
+
+ link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO;
+ link->io.NumPorts1 = 16;
+
+ link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT;
+
+ link->irq.IRQInfo1 = IRQ_LEVEL_ID;
+ link->irq.Handler = &snd_vx_irq_handler;
+ link->irq.Instance = chip;
+
+ link->conf.Attributes = CONF_ENABLE_IRQ;
+ link->conf.IntType = INT_MEMORY_AND_IO;
+ link->conf.ConfigIndex = 1;
+ link->conf.Present = PRESENT_OPTION;
+
+ return vxp;
+}
+
+
+/**
+ * snd_vxpocket_assign_resources - initialize the hardware and card instance.
+ * @port: i/o port for the card
+ * @irq: irq number for the card
+ *
+ * this function assigns the specified port and irq, boot the card,
+ * create pcm and control instances, and initialize the rest hardware.
+ *
+ * returns 0 if successful, or a negative error code.
+ */
+static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq)
+{
+ int err;
+ struct snd_card *card = chip->card;
+ struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip;
+
+ snd_printdd(KERN_DEBUG "vxpocket assign resources: port = 0x%x, irq = %d\n", port, irq);
+ vxp->port = port;
+
+ sprintf(card->shortname, "Digigram %s", card->driver);
+ sprintf(card->longname, "%s at 0x%x, irq %i",
+ card->shortname, port, irq);
+
+ chip->irq = irq;
+
+ if ((err = snd_vx_setup_firmware(chip)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * configuration callback
+ */
+
+#define CS_CHECK(fn, ret) \
+do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0)
+
+static int vxpocket_config(struct pcmcia_device *link)
+{
+ struct vx_core *chip = link->priv;
+ struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip;
+ int last_fn, last_ret;
+
+ snd_printdd(KERN_DEBUG "vxpocket_config called\n");
+
+ /* redefine hardware record according to the VERSION1 string */
+ if (!strcmp(link->prod_id[1], "VX-POCKET")) {
+ snd_printdd("VX-pocket is detected\n");
+ } else {
+ snd_printdd("VX-pocket 440 is detected\n");
+ /* overwrite the hardware information */
+ chip->hw = &vxp440_hw;
+ chip->type = vxp440_hw.type;
+ strcpy(chip->card->driver, vxp440_hw.name);
+ }
+
+ CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io));
+ CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq));
+ CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf));
+
+ chip->dev = &handle_to_dev(link);
+ snd_card_set_dev(chip->card, chip->dev);
+
+ if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq.AssignedIRQ) < 0)
+ goto failed;
+
+ link->dev_node = &vxp->node;
+ return 0;
+
+cs_failed:
+ cs_error(link, last_fn, last_ret);
+failed:
+ pcmcia_disable_device(link);
+ return -ENODEV;
+}
+
+#ifdef CONFIG_PM
+
+static int vxp_suspend(struct pcmcia_device *link)
+{
+ struct vx_core *chip = link->priv;
+
+ snd_printdd(KERN_DEBUG "SUSPEND\n");
+ if (chip) {
+ snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n");
+ snd_vx_suspend(chip, PMSG_SUSPEND);
+ }
+
+ return 0;
+}
+
+static int vxp_resume(struct pcmcia_device *link)
+{
+ struct vx_core *chip = link->priv;
+
+ snd_printdd(KERN_DEBUG "RESUME\n");
+ if (pcmcia_dev_present(link)) {
+ //struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip;
+ if (chip) {
+ snd_printdd(KERN_DEBUG "calling snd_vx_resume\n");
+ snd_vx_resume(chip);
+ }
+ }
+ snd_printdd(KERN_DEBUG "resume done!\n");
+
+ return 0;
+}
+
+#endif
+
+
+/*
+ */
+static int vxpocket_probe(struct pcmcia_device *p_dev)
+{
+ struct snd_card *card;
+ struct snd_vxpocket *vxp;
+ int i;
+
+ /* find an empty slot from the card list */
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ if (!(card_alloc & (1 << i)))
+ break;
+ }
+ if (i >= SNDRV_CARDS) {
+ snd_printk(KERN_ERR "vxpocket: too many cards found\n");
+ return -EINVAL;
+ }
+ if (! enable[i])
+ return -ENODEV; /* disabled explicitly */
+
+ /* ok, create a card instance */
+ card = snd_card_new(index[i], id[i], THIS_MODULE, 0);
+ if (card == NULL) {
+ snd_printk(KERN_ERR "vxpocket: cannot create a card instance\n");
+ return -ENOMEM;
+ }
+
+ vxp = snd_vxpocket_new(card, ibl[i], p_dev);
+ if (! vxp) {
+ snd_card_free(card);
+ return -ENODEV;
+ }
+ card->private_data = vxp;
+
+ vxp->index = i;
+ card_alloc |= 1 << i;
+
+ vxp->p_dev = p_dev;
+
+ return vxpocket_config(p_dev);
+}
+
+static void vxpocket_detach(struct pcmcia_device *link)
+{
+ struct snd_vxpocket *vxp;
+ struct vx_core *chip;
+
+ if (! link)
+ return;
+
+ vxp = link->priv;
+ chip = (struct vx_core *)vxp;
+ card_alloc &= ~(1 << vxp->index);
+
+ chip->chip_status |= VX_STAT_IS_STALE; /* to be sure */
+ snd_card_disconnect(chip->card);
+ vxpocket_release(link);
+ snd_card_free_when_closed(chip->card);
+}
+
+/*
+ * Module entry points
+ */
+
+static struct pcmcia_device_id vxp_ids[] = {
+ PCMCIA_DEVICE_MANF_CARD(0x01f1, 0x0100),
+ PCMCIA_DEVICE_NULL
+};
+MODULE_DEVICE_TABLE(pcmcia, vxp_ids);
+
+static struct pcmcia_driver vxp_cs_driver = {
+ .owner = THIS_MODULE,
+ .drv = {
+ .name = "snd-vxpocket",
+ },
+ .probe = vxpocket_probe,
+ .remove = vxpocket_detach,
+ .id_table = vxp_ids,
+#ifdef CONFIG_PM
+ .suspend = vxp_suspend,
+ .resume = vxp_resume,
+#endif
+};
+
+static int __init init_vxpocket(void)
+{
+ return pcmcia_register_driver(&vxp_cs_driver);
+}
+
+static void __exit exit_vxpocket(void)
+{
+ pcmcia_unregister_driver(&vxp_cs_driver);
+}
+
+module_init(init_vxpocket);
+module_exit(exit_vxpocket);
diff --git a/sound/pcmcia/vx/vxpocket.h b/sound/pcmcia/vx/vxpocket.h
new file mode 100644
index 0000000..27ea002
--- /dev/null
+++ b/sound/pcmcia/vx/vxpocket.h
@@ -0,0 +1,93 @@
+/*
+ * Driver for Digigram VXpocket soundcards
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __VXPOCKET_H
+#define __VXPOCKET_H
+
+#include <sound/vx_core.h>
+
+#include <pcmcia/cs_types.h>
+#include <pcmcia/cs.h>
+#include <pcmcia/cistpl.h>
+#include <pcmcia/ds.h>
+
+struct snd_vxpocket {
+
+ struct vx_core core;
+
+ unsigned long port;
+
+ int mic_level; /* analog mic level (or boost) */
+
+ unsigned int regCDSP; /* current CDSP register */
+ unsigned int regDIALOG; /* current DIALOG register */
+
+ int index; /* card index */
+
+ /* pcmcia stuff */
+ struct pcmcia_device *p_dev;
+ dev_node_t node;
+};
+
+extern struct snd_vx_ops snd_vxpocket_ops;
+
+void vx_set_mic_boost(struct vx_core *chip, int boost);
+void vx_set_mic_level(struct vx_core *chip, int level);
+
+int vxp_add_mic_controls(struct vx_core *chip);
+
+/* Constants used to access the CDSP register (0x08). */
+#define CDSP_MAGIC 0xA7 /* magic value (for read) */
+/* for write */
+#define VXP_CDSP_CLOCKIN_SEL_MASK 0x80 /* 0 (internal), 1 (AES/EBU) */
+#define VXP_CDSP_DATAIN_SEL_MASK 0x40 /* 0 (analog), 1 (UER) */
+#define VXP_CDSP_SMPTE_SEL_MASK 0x20
+#define VXP_CDSP_RESERVED_MASK 0x10
+#define VXP_CDSP_MIC_SEL_MASK 0x08
+#define VXP_CDSP_VALID_IRQ_MASK 0x04
+#define VXP_CDSP_CODEC_RESET_MASK 0x02
+#define VXP_CDSP_DSP_RESET_MASK 0x01
+/* VXPOCKET 240/440 */
+#define P24_CDSP_MICS_SEL_MASK 0x18
+#define P24_CDSP_MIC20_SEL_MASK 0x10
+#define P24_CDSP_MIC38_SEL_MASK 0x08
+
+/* Constants used to access the MEMIRQ register (0x0C). */
+#define P44_MEMIRQ_MASTER_SLAVE_SEL_MASK 0x08
+#define P44_MEMIRQ_SYNCED_ALONE_SEL_MASK 0x04
+#define P44_MEMIRQ_WCLK_OUT_IN_SEL_MASK 0x02 /* Not used */
+#define P44_MEMIRQ_WCLK_UER_SEL_MASK 0x01 /* Not used */
+
+/* Micro levels (0x0C) */
+
+/* Constants used to access the DIALOG register (0x0D). */
+#define VXP_DLG_XILINX_REPROG_MASK 0x80 /* W */
+#define VXP_DLG_DATA_XICOR_MASK 0x80 /* R */
+#define VXP_DLG_RESERVED4_0_MASK 0x40
+#define VXP_DLG_RESERVED2_0_MASK 0x20
+#define VXP_DLG_RESERVED1_0_MASK 0x10
+#define VXP_DLG_DMAWRITE_SEL_MASK 0x08 /* W */
+#define VXP_DLG_DMAREAD_SEL_MASK 0x04 /* W */
+#define VXP_DLG_MEMIRQ_MASK 0x02 /* R */
+#define VXP_DLG_DMA16_SEL_MASK 0x02 /* W */
+#define VXP_DLG_ACK_MEMIRQ_MASK 0x01 /* R/W */
+
+
+#endif /* __VXPOCKET_H */
diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig
new file mode 100644
index 0000000..777de2b
--- /dev/null
+++ b/sound/ppc/Kconfig
@@ -0,0 +1,51 @@
+# ALSA PowerMac drivers
+
+menuconfig SND_PPC
+ bool "PowerPC sound devices"
+ depends on PPC64 || PPC32
+ default y
+ help
+ Support for sound devices specific to PowerPC architectures.
+
+if SND_PPC
+
+config SND_POWERMAC
+ tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)"
+ depends on I2C && INPUT && PPC_PMAC
+ select SND_PCM
+ help
+ Say Y here to include support for the integrated sound device.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-powermac.
+
+config SND_POWERMAC_AUTO_DRC
+ bool "Toggle DRC automatically at headphone/line plug-in"
+ depends on SND_POWERMAC
+ default y
+ help
+ Say Y here to enable the automatic toggle of DRC (dynamic
+ range compression) on Tumbler/Snapper.
+ If this feature is enabled, DRC is turned off when the
+ headphone/line jack is plugged, and turned on when unplugged.
+
+ Note that you can turn on/off DRC manually even without this
+ option.
+
+config SND_PS3
+ tristate "PS3 Audio support"
+ depends on PS3_PS3AV
+ select SND_PCM
+ default m
+ help
+ Say Y here to include support for audio on the PS3
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd_ps3.
+
+config SND_PS3_DEFAULT_START_DELAY
+ int "Startup delay time in ms"
+ depends on SND_PS3
+ default "2000"
+
+endif # SND_PPC
diff --git a/sound/ppc/Makefile b/sound/ppc/Makefile
new file mode 100644
index 0000000..679c45a
--- /dev/null
+++ b/sound/ppc/Makefile
@@ -0,0 +1,10 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o
+obj-$(CONFIG_SND_PS3) += snd_ps3.o
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
new file mode 100644
index 0000000..7bd33e6
--- /dev/null
+++ b/sound/ppc/awacs.c
@@ -0,0 +1,1078 @@
+/*
+ * PMac AWACS lowlevel functions
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ * code based on dmasound.c.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <asm/io.h>
+#include <asm/nvram.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "pmac.h"
+
+
+#ifdef CONFIG_ADB_CUDA
+#define PMAC_AMP_AVAIL
+#endif
+
+#ifdef PMAC_AMP_AVAIL
+struct awacs_amp {
+ unsigned char amp_master;
+ unsigned char amp_vol[2][2];
+ unsigned char amp_tone[2];
+};
+
+#define CHECK_CUDA_AMP() (sys_ctrler == SYS_CTRLER_CUDA)
+
+#endif /* PMAC_AMP_AVAIL */
+
+
+static void snd_pmac_screamer_wait(struct snd_pmac *chip)
+{
+ long timeout = 2000;
+ while (!(in_le32(&chip->awacs->codec_stat) & MASK_VALID)) {
+ mdelay(1);
+ if (! --timeout) {
+ snd_printd("snd_pmac_screamer_wait timeout\n");
+ break;
+ }
+ }
+}
+
+/*
+ * write AWACS register
+ */
+static void
+snd_pmac_awacs_write(struct snd_pmac *chip, int val)
+{
+ long timeout = 5000000;
+
+ if (chip->model == PMAC_SCREAMER)
+ snd_pmac_screamer_wait(chip);
+ out_le32(&chip->awacs->codec_ctrl, val | (chip->subframe << 22));
+ while (in_le32(&chip->awacs->codec_ctrl) & MASK_NEWECMD) {
+ if (! --timeout) {
+ snd_printd("snd_pmac_awacs_write timeout\n");
+ break;
+ }
+ }
+}
+
+static void
+snd_pmac_awacs_write_reg(struct snd_pmac *chip, int reg, int val)
+{
+ snd_pmac_awacs_write(chip, val | (reg << 12));
+ chip->awacs_reg[reg] = val;
+}
+
+static void
+snd_pmac_awacs_write_noreg(struct snd_pmac *chip, int reg, int val)
+{
+ snd_pmac_awacs_write(chip, val | (reg << 12));
+}
+
+#ifdef CONFIG_PM
+/* Recalibrate chip */
+static void screamer_recalibrate(struct snd_pmac *chip)
+{
+ if (chip->model != PMAC_SCREAMER)
+ return;
+
+ /* Sorry for the horrible delays... I hope to get that improved
+ * by making the whole PM process asynchronous in a future version
+ */
+ snd_pmac_awacs_write_noreg(chip, 1, chip->awacs_reg[1]);
+ if (chip->manufacturer == 0x1)
+ /* delay for broken crystal part */
+ msleep(750);
+ snd_pmac_awacs_write_noreg(chip, 1,
+ chip->awacs_reg[1] | MASK_RECALIBRATE |
+ MASK_CMUTE | MASK_AMUTE);
+ snd_pmac_awacs_write_noreg(chip, 1, chip->awacs_reg[1]);
+ snd_pmac_awacs_write_noreg(chip, 6, chip->awacs_reg[6]);
+}
+
+#else
+#define screamer_recalibrate(chip) /* NOP */
+#endif
+
+
+/*
+ * additional callback to set the pcm format
+ */
+static void snd_pmac_awacs_set_format(struct snd_pmac *chip)
+{
+ chip->awacs_reg[1] &= ~MASK_SAMPLERATE;
+ chip->awacs_reg[1] |= chip->rate_index << 3;
+ snd_pmac_awacs_write_reg(chip, 1, chip->awacs_reg[1]);
+}
+
+
+/*
+ * AWACS volume callbacks
+ */
+/*
+ * volumes: 0-15 stereo
+ */
+static int snd_pmac_awacs_info_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 15;
+ return 0;
+}
+
+static int snd_pmac_awacs_get_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int lshift = (kcontrol->private_value >> 8) & 0xff;
+ int inverted = (kcontrol->private_value >> 16) & 1;
+ unsigned long flags;
+ int vol[2];
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ vol[0] = (chip->awacs_reg[reg] >> lshift) & 0xf;
+ vol[1] = chip->awacs_reg[reg] & 0xf;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (inverted) {
+ vol[0] = 0x0f - vol[0];
+ vol[1] = 0x0f - vol[1];
+ }
+ ucontrol->value.integer.value[0] = vol[0];
+ ucontrol->value.integer.value[1] = vol[1];
+ return 0;
+}
+
+static int snd_pmac_awacs_put_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int lshift = (kcontrol->private_value >> 8) & 0xff;
+ int inverted = (kcontrol->private_value >> 16) & 1;
+ int val, oldval;
+ unsigned long flags;
+ unsigned int vol[2];
+
+ vol[0] = ucontrol->value.integer.value[0];
+ vol[1] = ucontrol->value.integer.value[1];
+ if (vol[0] > 0x0f || vol[1] > 0x0f)
+ return -EINVAL;
+ if (inverted) {
+ vol[0] = 0x0f - vol[0];
+ vol[1] = 0x0f - vol[1];
+ }
+ vol[0] &= 0x0f;
+ vol[1] &= 0x0f;
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ oldval = chip->awacs_reg[reg];
+ val = oldval & ~(0xf | (0xf << lshift));
+ val |= vol[0] << lshift;
+ val |= vol[1];
+ if (oldval != val)
+ snd_pmac_awacs_write_reg(chip, reg, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return oldval != reg;
+}
+
+
+#define AWACS_VOLUME(xname, xreg, xshift, xinverted) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = snd_pmac_awacs_info_volume, \
+ .get = snd_pmac_awacs_get_volume, \
+ .put = snd_pmac_awacs_put_volume, \
+ .private_value = (xreg) | ((xshift) << 8) | ((xinverted) << 16) }
+
+/*
+ * mute master/ogain for AWACS: mono
+ */
+static int snd_pmac_awacs_get_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int invert = (kcontrol->private_value >> 16) & 1;
+ int val;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = (chip->awacs_reg[reg] >> shift) & 1;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (invert)
+ val = 1 - val;
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+}
+
+static int snd_pmac_awacs_put_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int invert = (kcontrol->private_value >> 16) & 1;
+ int mask = 1 << shift;
+ int val, changed;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val = chip->awacs_reg[reg] & ~mask;
+ if (ucontrol->value.integer.value[0] != invert)
+ val |= mask;
+ changed = chip->awacs_reg[reg] != val;
+ if (changed)
+ snd_pmac_awacs_write_reg(chip, reg, val);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return changed;
+}
+
+#define AWACS_SWITCH(xname, xreg, xshift, xinvert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = snd_pmac_boolean_mono_info, \
+ .get = snd_pmac_awacs_get_switch, \
+ .put = snd_pmac_awacs_put_switch, \
+ .private_value = (xreg) | ((xshift) << 8) | ((xinvert) << 16) }
+
+
+#ifdef PMAC_AMP_AVAIL
+/*
+ * controls for perch/whisper extension cards, e.g. G3 desktop
+ *
+ * TDA7433 connected via i2c address 0x45 (= 0x8a),
+ * accessed through cuda
+ */
+static void awacs_set_cuda(int reg, int val)
+{
+ struct adb_request req;
+ cuda_request(&req, NULL, 5, CUDA_PACKET, CUDA_GET_SET_IIC, 0x8a,
+ reg, val);
+ while (! req.complete)
+ cuda_poll();
+}
+
+/*
+ * level = 0 - 14, 7 = 0 dB
+ */
+static void awacs_amp_set_tone(struct awacs_amp *amp, int bass, int treble)
+{
+ amp->amp_tone[0] = bass;
+ amp->amp_tone[1] = treble;
+ if (bass > 7)
+ bass = (14 - bass) + 8;
+ if (treble > 7)
+ treble = (14 - treble) + 8;
+ awacs_set_cuda(2, (bass << 4) | treble);
+}
+
+/*
+ * vol = 0 - 31 (attenuation), 32 = mute bit, stereo
+ */
+static int awacs_amp_set_vol(struct awacs_amp *amp, int index,
+ int lvol, int rvol, int do_check)
+{
+ if (do_check && amp->amp_vol[index][0] == lvol &&
+ amp->amp_vol[index][1] == rvol)
+ return 0;
+ awacs_set_cuda(3 + index, lvol);
+ awacs_set_cuda(5 + index, rvol);
+ amp->amp_vol[index][0] = lvol;
+ amp->amp_vol[index][1] = rvol;
+ return 1;
+}
+
+/*
+ * 0 = -79 dB, 79 = 0 dB, 99 = +20 dB
+ */
+static void awacs_amp_set_master(struct awacs_amp *amp, int vol)
+{
+ amp->amp_master = vol;
+ if (vol <= 79)
+ vol = 32 + (79 - vol);
+ else
+ vol = 32 - (vol - 79);
+ awacs_set_cuda(1, vol);
+}
+
+static void awacs_amp_free(struct snd_pmac *chip)
+{
+ struct awacs_amp *amp = chip->mixer_data;
+ if (!amp)
+ return;
+ kfree(amp);
+ chip->mixer_data = NULL;
+ chip->mixer_free = NULL;
+}
+
+
+/*
+ * mixer controls
+ */
+static int snd_pmac_awacs_info_volume_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 31;
+ return 0;
+}
+
+static int snd_pmac_awacs_get_volume_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ struct awacs_amp *amp = chip->mixer_data;
+
+ ucontrol->value.integer.value[0] = 31 - (amp->amp_vol[index][0] & 31);
+ ucontrol->value.integer.value[1] = 31 - (amp->amp_vol[index][1] & 31);
+ return 0;
+}
+
+static int snd_pmac_awacs_put_volume_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ int vol[2];
+ struct awacs_amp *amp = chip->mixer_data;
+
+ vol[0] = (31 - (ucontrol->value.integer.value[0] & 31))
+ | (amp->amp_vol[index][0] & 32);
+ vol[1] = (31 - (ucontrol->value.integer.value[1] & 31))
+ | (amp->amp_vol[index][1] & 32);
+ return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1);
+}
+
+static int snd_pmac_awacs_get_switch_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ struct awacs_amp *amp = chip->mixer_data;
+
+ ucontrol->value.integer.value[0] = (amp->amp_vol[index][0] & 32)
+ ? 0 : 1;
+ ucontrol->value.integer.value[1] = (amp->amp_vol[index][1] & 32)
+ ? 0 : 1;
+ return 0;
+}
+
+static int snd_pmac_awacs_put_switch_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ int vol[2];
+ struct awacs_amp *amp = chip->mixer_data;
+
+ vol[0] = (ucontrol->value.integer.value[0] ? 0 : 32)
+ | (amp->amp_vol[index][0] & 31);
+ vol[1] = (ucontrol->value.integer.value[1] ? 0 : 32)
+ | (amp->amp_vol[index][1] & 31);
+ return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1);
+}
+
+static int snd_pmac_awacs_info_tone_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 14;
+ return 0;
+}
+
+static int snd_pmac_awacs_get_tone_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ struct awacs_amp *amp = chip->mixer_data;
+
+ ucontrol->value.integer.value[0] = amp->amp_tone[index];
+ return 0;
+}
+
+static int snd_pmac_awacs_put_tone_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ struct awacs_amp *amp = chip->mixer_data;
+ unsigned int val;
+
+ val = ucontrol->value.integer.value[0];
+ if (val > 14)
+ return -EINVAL;
+ if (val != amp->amp_tone[index]) {
+ amp->amp_tone[index] = val;
+ awacs_amp_set_tone(amp, amp->amp_tone[0], amp->amp_tone[1]);
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_pmac_awacs_info_master_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 99;
+ return 0;
+}
+
+static int snd_pmac_awacs_get_master_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct awacs_amp *amp = chip->mixer_data;
+
+ ucontrol->value.integer.value[0] = amp->amp_master;
+ return 0;
+}
+
+static int snd_pmac_awacs_put_master_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct awacs_amp *amp = chip->mixer_data;
+ unsigned int val;
+
+ val = ucontrol->value.integer.value[0];
+ if (val > 99)
+ return -EINVAL;
+ if (val != amp->amp_master) {
+ amp->amp_master = val;
+ awacs_amp_set_master(amp, amp->amp_master);
+ return 1;
+ }
+ return 0;
+}
+
+#define AMP_CH_SPK 0
+#define AMP_CH_HD 1
+
+static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __initdata = {
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PC Speaker Playback Volume",
+ .info = snd_pmac_awacs_info_volume_amp,
+ .get = snd_pmac_awacs_get_volume_amp,
+ .put = snd_pmac_awacs_put_volume_amp,
+ .private_value = AMP_CH_SPK,
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Playback Volume",
+ .info = snd_pmac_awacs_info_volume_amp,
+ .get = snd_pmac_awacs_get_volume_amp,
+ .put = snd_pmac_awacs_put_volume_amp,
+ .private_value = AMP_CH_HD,
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Tone Control - Bass",
+ .info = snd_pmac_awacs_info_tone_amp,
+ .get = snd_pmac_awacs_get_tone_amp,
+ .put = snd_pmac_awacs_put_tone_amp,
+ .private_value = 0,
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Tone Control - Treble",
+ .info = snd_pmac_awacs_info_tone_amp,
+ .get = snd_pmac_awacs_get_tone_amp,
+ .put = snd_pmac_awacs_put_tone_amp,
+ .private_value = 1,
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Amp Master Playback Volume",
+ .info = snd_pmac_awacs_info_master_amp,
+ .get = snd_pmac_awacs_get_master_amp,
+ .put = snd_pmac_awacs_put_master_amp,
+ },
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __initdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Playback Switch",
+ .info = snd_pmac_boolean_stereo_info,
+ .get = snd_pmac_awacs_get_switch_amp,
+ .put = snd_pmac_awacs_put_switch_amp,
+ .private_value = AMP_CH_HD,
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __initdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PC Speaker Playback Switch",
+ .info = snd_pmac_boolean_stereo_info,
+ .get = snd_pmac_awacs_get_switch_amp,
+ .put = snd_pmac_awacs_put_switch_amp,
+ .private_value = AMP_CH_SPK,
+};
+
+#endif /* PMAC_AMP_AVAIL */
+
+
+/*
+ * mic boost for screamer
+ */
+static int snd_pmac_screamer_mic_boost_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 3;
+ return 0;
+}
+
+static int snd_pmac_screamer_mic_boost_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int val = 0;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->awacs_reg[6] & MASK_MIC_BOOST)
+ val |= 2;
+ if (chip->awacs_reg[0] & MASK_GAINLINE)
+ val |= 1;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+}
+
+static int snd_pmac_screamer_mic_boost_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int val0, val6;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ val0 = chip->awacs_reg[0] & ~MASK_GAINLINE;
+ val6 = chip->awacs_reg[6] & ~MASK_MIC_BOOST;
+ if (ucontrol->value.integer.value[0] & 1)
+ val0 |= MASK_GAINLINE;
+ if (ucontrol->value.integer.value[0] & 2)
+ val6 |= MASK_MIC_BOOST;
+ if (val0 != chip->awacs_reg[0]) {
+ snd_pmac_awacs_write_reg(chip, 0, val0);
+ changed = 1;
+ }
+ if (val6 != chip->awacs_reg[6]) {
+ snd_pmac_awacs_write_reg(chip, 6, val6);
+ changed = 1;
+ }
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return changed;
+}
+
+/*
+ * lists of mixer elements
+ */
+static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __initdata = {
+ AWACS_SWITCH("Master Capture Switch", 1, SHIFT_LOOPTHRU, 0),
+ AWACS_VOLUME("Master Capture Volume", 0, 4, 0),
+/* AWACS_SWITCH("Unknown Playback Switch", 6, SHIFT_PAROUT0, 0), */
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = {
+ AWACS_VOLUME("Master Playback Volume", 2, 6, 1),
+ AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1),
+ AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
+ AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = {
+ AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
+ AWACS_VOLUME("Master Playback Volume", 5, 6, 1),
+ AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_g4agp[] __initdata = {
+ AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
+ AWACS_VOLUME("Master Playback Volume", 5, 6, 1),
+ AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
+ AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = {
+ AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
+ AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
+ AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = {
+ AWACS_VOLUME("Master Playback Volume", 2, 6, 1),
+ AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
+};
+
+/* FIXME: is this correct order?
+ * screamer (powerbook G3 pismo) seems to have different bits...
+ */
+static struct snd_kcontrol_new snd_pmac_awacs_mixers2[] __initdata = {
+ AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_LINE, 0),
+ AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_MIC, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __initdata = {
+ AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
+ AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_LINE, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata =
+AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1);
+
+static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata =
+AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1);
+
+static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = {
+ AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __initdata = {
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic Boost Capture Volume",
+ .info = snd_pmac_screamer_mic_boost_info,
+ .get = snd_pmac_screamer_mic_boost_get,
+ .put = snd_pmac_screamer_mic_boost_put,
+ },
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __initdata =
+{
+ AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __initdata =
+{
+ AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
+ AWACS_SWITCH("CD Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __initdata =
+{
+ AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
+ AWACS_SWITCH("Mic Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0),
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __initdata = {
+ AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1),
+};
+
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __initdata =
+AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1);
+
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __initdata =
+AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1);
+
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __initdata =
+AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0);
+
+
+/*
+ * add new mixer elements to the card
+ */
+static int build_mixers(struct snd_pmac *chip, int nums,
+ struct snd_kcontrol_new *mixers)
+{
+ int i, err;
+
+ for (i = 0; i < nums; i++) {
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&mixers[i], chip));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+
+/*
+ * restore all registers
+ */
+static void awacs_restore_all_regs(struct snd_pmac *chip)
+{
+ snd_pmac_awacs_write_noreg(chip, 0, chip->awacs_reg[0]);
+ snd_pmac_awacs_write_noreg(chip, 1, chip->awacs_reg[1]);
+ snd_pmac_awacs_write_noreg(chip, 2, chip->awacs_reg[2]);
+ snd_pmac_awacs_write_noreg(chip, 4, chip->awacs_reg[4]);
+ if (chip->model == PMAC_SCREAMER) {
+ snd_pmac_awacs_write_noreg(chip, 5, chip->awacs_reg[5]);
+ snd_pmac_awacs_write_noreg(chip, 6, chip->awacs_reg[6]);
+ snd_pmac_awacs_write_noreg(chip, 7, chip->awacs_reg[7]);
+ }
+}
+
+#ifdef CONFIG_PM
+static void snd_pmac_awacs_suspend(struct snd_pmac *chip)
+{
+ snd_pmac_awacs_write_noreg(chip, 1, (chip->awacs_reg[1]
+ | MASK_AMUTE | MASK_CMUTE));
+}
+
+static void snd_pmac_awacs_resume(struct snd_pmac *chip)
+{
+ if (machine_is_compatible("PowerBook3,1")
+ || machine_is_compatible("PowerBook3,2")) {
+ msleep(100);
+ snd_pmac_awacs_write_reg(chip, 1,
+ chip->awacs_reg[1] & ~MASK_PAROUT);
+ msleep(300);
+ }
+
+ awacs_restore_all_regs(chip);
+ if (chip->model == PMAC_SCREAMER) {
+ /* reset power bits in reg 6 */
+ mdelay(5);
+ snd_pmac_awacs_write_noreg(chip, 6, chip->awacs_reg[6]);
+ }
+ screamer_recalibrate(chip);
+#ifdef PMAC_AMP_AVAIL
+ if (chip->mixer_data) {
+ struct awacs_amp *amp = chip->mixer_data;
+ awacs_amp_set_vol(amp, 0,
+ amp->amp_vol[0][0], amp->amp_vol[0][1], 0);
+ awacs_amp_set_vol(amp, 1,
+ amp->amp_vol[1][0], amp->amp_vol[1][1], 0);
+ awacs_amp_set_tone(amp, amp->amp_tone[0], amp->amp_tone[1]);
+ awacs_amp_set_master(amp, amp->amp_master);
+ }
+#endif
+}
+#endif /* CONFIG_PM */
+
+#define IS_PM7500 (machine_is_compatible("AAPL,7500"))
+#define IS_BEIGE (machine_is_compatible("AAPL,Gossamer"))
+#define IS_IMAC1 (machine_is_compatible("PowerMac2,1"))
+#define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \
+ || machine_is_compatible("PowerMac4,1"))
+#define IS_G4AGP (machine_is_compatible("PowerMac3,1"))
+
+static int imac1, imac2;
+
+#ifdef PMAC_SUPPORT_AUTOMUTE
+/*
+ * auto-mute stuffs
+ */
+static int snd_pmac_awacs_detect_headphone(struct snd_pmac *chip)
+{
+ return (in_le32(&chip->awacs->codec_stat) & chip->hp_stat_mask) ? 1 : 0;
+}
+
+#ifdef PMAC_AMP_AVAIL
+static int toggle_amp_mute(struct awacs_amp *amp, int index, int mute)
+{
+ int vol[2];
+ vol[0] = amp->amp_vol[index][0] & 31;
+ vol[1] = amp->amp_vol[index][1] & 31;
+ if (mute) {
+ vol[0] |= 32;
+ vol[1] |= 32;
+ }
+ return awacs_amp_set_vol(amp, index, vol[0], vol[1], 1);
+}
+#endif
+
+static void snd_pmac_awacs_update_automute(struct snd_pmac *chip, int do_notify)
+{
+ if (chip->auto_mute) {
+#ifdef PMAC_AMP_AVAIL
+ if (chip->mixer_data) {
+ struct awacs_amp *amp = chip->mixer_data;
+ int changed;
+ if (snd_pmac_awacs_detect_headphone(chip)) {
+ changed = toggle_amp_mute(amp, AMP_CH_HD, 0);
+ changed |= toggle_amp_mute(amp, AMP_CH_SPK, 1);
+ } else {
+ changed = toggle_amp_mute(amp, AMP_CH_HD, 1);
+ changed |= toggle_amp_mute(amp, AMP_CH_SPK, 0);
+ }
+ if (do_notify && ! changed)
+ return;
+ } else
+#endif
+ {
+ int reg = chip->awacs_reg[1]
+ | (MASK_HDMUTE | MASK_SPKMUTE);
+ if (imac1) {
+ reg &= ~MASK_SPKMUTE;
+ reg |= MASK_PAROUT1;
+ } else if (imac2) {
+ reg &= ~MASK_SPKMUTE;
+ reg &= ~MASK_PAROUT1;
+ }
+ if (snd_pmac_awacs_detect_headphone(chip))
+ reg &= ~MASK_HDMUTE;
+ else if (imac1)
+ reg &= ~MASK_PAROUT1;
+ else if (imac2)
+ reg |= MASK_PAROUT1;
+ else
+ reg &= ~MASK_SPKMUTE;
+ if (do_notify && reg == chip->awacs_reg[1])
+ return;
+ snd_pmac_awacs_write_reg(chip, 1, reg);
+ }
+ if (do_notify) {
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_sw_ctl->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->speaker_sw_ctl->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->hp_detect_ctl->id);
+ }
+ }
+}
+#endif /* PMAC_SUPPORT_AUTOMUTE */
+
+
+/*
+ * initialize chip
+ */
+int __init
+snd_pmac_awacs_init(struct snd_pmac *chip)
+{
+ int pm7500 = IS_PM7500;
+ int beige = IS_BEIGE;
+ int g4agp = IS_G4AGP;
+ int imac;
+ int err, vol;
+
+ imac1 = IS_IMAC1;
+ imac2 = IS_IMAC2;
+ imac = imac1 || imac2;
+ /* looks like MASK_GAINLINE triggers something, so we set here
+ * as start-up
+ */
+ chip->awacs_reg[0] = MASK_MUX_CD | 0xff | MASK_GAINLINE;
+ chip->awacs_reg[1] = MASK_CMUTE | MASK_AMUTE;
+ /* FIXME: Only machines with external SRS module need MASK_PAROUT */
+ if (chip->has_iic || chip->device_id == 0x5 ||
+ /* chip->_device_id == 0x8 || */
+ chip->device_id == 0xb)
+ chip->awacs_reg[1] |= MASK_PAROUT;
+ /* get default volume from nvram */
+ // vol = (~nvram_read_byte(0x1308) & 7) << 1;
+ // vol = ((pmac_xpram_read( 8 ) & 7 ) << 1 );
+ vol = 0x0f; /* no, on alsa, muted as default */
+ vol = vol + (vol << 6);
+ chip->awacs_reg[2] = vol;
+ chip->awacs_reg[4] = vol;
+ if (chip->model == PMAC_SCREAMER) {
+ /* FIXME: screamer has loopthru vol control */
+ chip->awacs_reg[5] = vol;
+ /* FIXME: maybe should be vol << 3 for PCMCIA speaker */
+ chip->awacs_reg[6] = MASK_MIC_BOOST;
+ chip->awacs_reg[7] = 0;
+ }
+
+ awacs_restore_all_regs(chip);
+ chip->manufacturer = (in_le32(&chip->awacs->codec_stat) >> 8) & 0xf;
+ screamer_recalibrate(chip);
+
+ chip->revision = (in_le32(&chip->awacs->codec_stat) >> 12) & 0xf;
+#ifdef PMAC_AMP_AVAIL
+ if (chip->revision == 3 && chip->has_iic && CHECK_CUDA_AMP()) {
+ struct awacs_amp *amp = kzalloc(sizeof(*amp), GFP_KERNEL);
+ if (! amp)
+ return -ENOMEM;
+ chip->mixer_data = amp;
+ chip->mixer_free = awacs_amp_free;
+ /* mute and zero vol */
+ awacs_amp_set_vol(amp, 0, 63, 63, 0);
+ awacs_amp_set_vol(amp, 1, 63, 63, 0);
+ awacs_amp_set_tone(amp, 7, 7); /* 0 dB */
+ awacs_amp_set_master(amp, 79); /* 0 dB */
+ }
+#endif /* PMAC_AMP_AVAIL */
+
+ if (chip->hp_stat_mask == 0) {
+ /* set headphone-jack detection bit */
+ switch (chip->model) {
+ case PMAC_AWACS:
+ chip->hp_stat_mask = pm7500 ? MASK_HDPCONN
+ : MASK_LOCONN;
+ break;
+ case PMAC_SCREAMER:
+ switch (chip->device_id) {
+ case 0x08:
+ case 0x0B:
+ chip->hp_stat_mask = imac
+ ? MASK_LOCONN_IMAC |
+ MASK_HDPLCONN_IMAC |
+ MASK_HDPRCONN_IMAC
+ : MASK_HDPCONN;
+ break;
+ case 0x00:
+ case 0x05:
+ chip->hp_stat_mask = MASK_LOCONN;
+ break;
+ default:
+ chip->hp_stat_mask = MASK_HDPCONN;
+ break;
+ }
+ break;
+ default:
+ snd_BUG();
+ break;
+ }
+ }
+
+ /*
+ * build mixers
+ */
+ strcpy(chip->card->mixername, "PowerMac AWACS");
+
+ err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers),
+ snd_pmac_awacs_mixers);
+ if (err < 0)
+ return err;
+ if (beige || g4agp)
+ ;
+ else if (chip->model == PMAC_SCREAMER)
+ err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2),
+ snd_pmac_screamer_mixers2);
+ else if (!pm7500)
+ err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers2),
+ snd_pmac_awacs_mixers2);
+ if (err < 0)
+ return err;
+ if (pm7500)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500),
+ snd_pmac_awacs_mixers_pmac7500);
+ else if (beige)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_screamer_mixers_beige),
+ snd_pmac_screamer_mixers_beige);
+ else if (imac)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_screamer_mixers_imac),
+ snd_pmac_screamer_mixers_imac);
+ else if (g4agp)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_screamer_mixers_g4agp),
+ snd_pmac_screamer_mixers_g4agp);
+ else
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_awacs_mixers_pmac),
+ snd_pmac_awacs_mixers_pmac);
+ if (err < 0)
+ return err;
+ chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp)
+ ? &snd_pmac_awacs_master_sw_imac
+ : &snd_pmac_awacs_master_sw, chip);
+ err = snd_ctl_add(chip->card, chip->master_sw_ctl);
+ if (err < 0)
+ return err;
+#ifdef PMAC_AMP_AVAIL
+ if (chip->mixer_data) {
+ /* use amplifier. the signal is connected from route A
+ * to the amp. the amp has its headphone and speaker
+ * volumes and mute switches, so we use them instead of
+ * screamer registers.
+ * in this case, it seems the route C is not used.
+ */
+ err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_amp_vol),
+ snd_pmac_awacs_amp_vol);
+ if (err < 0)
+ return err;
+ /* overwrite */
+ chip->master_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_hp_sw,
+ chip);
+ err = snd_ctl_add(chip->card, chip->master_sw_ctl);
+ if (err < 0)
+ return err;
+ chip->speaker_sw_ctl = snd_ctl_new1(&snd_pmac_awacs_amp_spk_sw,
+ chip);
+ err = snd_ctl_add(chip->card, chip->speaker_sw_ctl);
+ if (err < 0)
+ return err;
+ } else
+#endif /* PMAC_AMP_AVAIL */
+ {
+ /* route A = headphone, route C = speaker */
+ err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol),
+ snd_pmac_awacs_speaker_vol);
+ if (err < 0)
+ return err;
+ chip->speaker_sw_ctl = snd_ctl_new1(imac1
+ ? &snd_pmac_awacs_speaker_sw_imac1
+ : imac2
+ ? &snd_pmac_awacs_speaker_sw_imac2
+ : &snd_pmac_awacs_speaker_sw, chip);
+ err = snd_ctl_add(chip->card, chip->speaker_sw_ctl);
+ if (err < 0)
+ return err;
+ }
+
+ if (beige || g4agp)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige),
+ snd_pmac_screamer_mic_boost_beige);
+ else if (imac)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_screamer_mic_boost_imac),
+ snd_pmac_screamer_mic_boost_imac);
+ else if (chip->model == PMAC_SCREAMER)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_screamer_mic_boost),
+ snd_pmac_screamer_mic_boost);
+ else if (pm7500)
+ err = build_mixers(chip,
+ ARRAY_SIZE(snd_pmac_awacs_mic_boost_pmac7500),
+ snd_pmac_awacs_mic_boost_pmac7500);
+ else
+ err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mic_boost),
+ snd_pmac_awacs_mic_boost);
+ if (err < 0)
+ return err;
+
+ /*
+ * set lowlevel callbacks
+ */
+ chip->set_format = snd_pmac_awacs_set_format;
+#ifdef CONFIG_PM
+ chip->suspend = snd_pmac_awacs_suspend;
+ chip->resume = snd_pmac_awacs_resume;
+#endif
+#ifdef PMAC_SUPPORT_AUTOMUTE
+ err = snd_pmac_add_automute(chip);
+ if (err < 0)
+ return err;
+ chip->detect_headphone = snd_pmac_awacs_detect_headphone;
+ chip->update_automute = snd_pmac_awacs_update_automute;
+ snd_pmac_awacs_update_automute(chip, 0); /* update the status only */
+#endif
+ if (chip->model == PMAC_SCREAMER) {
+ snd_pmac_awacs_write_noreg(chip, 6, chip->awacs_reg[6]);
+ snd_pmac_awacs_write_noreg(chip, 0, chip->awacs_reg[0]);
+ }
+
+ return 0;
+}
diff --git a/sound/ppc/awacs.h b/sound/ppc/awacs.h
new file mode 100644
index 0000000..c33e6a5
--- /dev/null
+++ b/sound/ppc/awacs.h
@@ -0,0 +1,205 @@
+/*
+ * Driver for PowerMac AWACS onboard soundchips
+ * Copyright (c) 2001 by Takashi Iwai <tiwai@suse.de>
+ * based on dmasound.c.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#ifndef __AWACS_H
+#define __AWACS_H
+
+/*******************************/
+/* AWACs Audio Register Layout */
+/*******************************/
+
+struct awacs_regs {
+ unsigned control; /* Audio control register */
+ unsigned pad0[3];
+ unsigned codec_ctrl; /* Codec control register */
+ unsigned pad1[3];
+ unsigned codec_stat; /* Codec status register */
+ unsigned pad2[3];
+ unsigned clip_count; /* Clipping count register */
+ unsigned pad3[3];
+ unsigned byteswap; /* Data is little-endian if 1 */
+};
+
+/*******************/
+/* Audio Bit Masks */
+/*******************/
+
+/* Audio Control Reg Bit Masks */
+/* ----- ------- --- --- ----- */
+#define MASK_ISFSEL (0xf) /* Input SubFrame Select */
+#define MASK_OSFSEL (0xf << 4) /* Output SubFrame Select */
+#define MASK_RATE (0x7 << 8) /* Sound Rate */
+#define MASK_CNTLERR (0x1 << 11) /* Error */
+#define MASK_PORTCHG (0x1 << 12) /* Port Change */
+#define MASK_IEE (0x1 << 13) /* Enable Interrupt on Error */
+#define MASK_IEPC (0x1 << 14) /* Enable Interrupt on Port Change */
+#define MASK_SSFSEL (0x3 << 15) /* Status SubFrame Select */
+
+/* Audio Codec Control Reg Bit Masks */
+/* ----- ----- ------- --- --- ----- */
+#define MASK_NEWECMD (0x1 << 24) /* Lock: don't write to reg when 1 */
+#define MASK_EMODESEL (0x3 << 22) /* Send info out on which frame? */
+#define MASK_EXMODEADDR (0x3ff << 12) /* Extended Mode Address -- 10 bits */
+#define MASK_EXMODEDATA (0xfff) /* Extended Mode Data -- 12 bits */
+
+/* Audio Codec Control Address Values / Masks */
+/* ----- ----- ------- ------- ------ - ----- */
+#define MASK_ADDR0 (0x0 << 12) /* Expanded Data Mode Address 0 */
+#define MASK_ADDR_MUX MASK_ADDR0 /* Mux Control */
+#define MASK_ADDR_GAIN MASK_ADDR0
+
+#define MASK_ADDR1 (0x1 << 12) /* Expanded Data Mode Address 1 */
+#define MASK_ADDR_MUTE MASK_ADDR1
+#define MASK_ADDR_RATE MASK_ADDR1
+
+#define MASK_ADDR2 (0x2 << 12) /* Expanded Data Mode Address 2 */
+#define MASK_ADDR_VOLA MASK_ADDR2 /* Volume Control A -- Headphones */
+#define MASK_ADDR_VOLHD MASK_ADDR2
+
+#define MASK_ADDR4 (0x4 << 12) /* Expanded Data Mode Address 4 */
+#define MASK_ADDR_VOLC MASK_ADDR4 /* Volume Control C -- Speaker */
+#define MASK_ADDR_VOLSPK MASK_ADDR4
+
+/* additional registers of screamer */
+#define MASK_ADDR5 (0x5 << 12) /* Expanded Data Mode Address 5 */
+#define MASK_ADDR6 (0x6 << 12) /* Expanded Data Mode Address 6 */
+#define MASK_ADDR7 (0x7 << 12) /* Expanded Data Mode Address 7 */
+
+/* Address 0 Bit Masks & Macros */
+/* ------- - --- ----- - ------ */
+#define MASK_GAINRIGHT (0xf) /* Gain Right Mask */
+#define MASK_GAINLEFT (0xf << 4) /* Gain Left Mask */
+#define MASK_GAINLINE (0x1 << 8) /* Disable Mic preamp */
+#define MASK_GAINMIC (0x0 << 8) /* Enable Mic preamp */
+#define MASK_MUX_CD (0x1 << 9) /* Select CD in MUX */
+#define MASK_MUX_MIC (0x1 << 10) /* Select Mic in MUX */
+#define MASK_MUX_AUDIN (0x1 << 11) /* Select Audio In in MUX */
+#define MASK_MUX_LINE MASK_MUX_AUDIN
+#define SHIFT_GAINLINE 8
+#define SHIFT_MUX_CD 9
+#define SHIFT_MUX_MIC 10
+#define SHIFT_MUX_LINE 11
+
+#define GAINRIGHT(x) ((x) & MASK_GAINRIGHT)
+#define GAINLEFT(x) (((x) << 4) & MASK_GAINLEFT)
+
+/* Address 1 Bit Masks */
+/* ------- - --- ----- */
+#define MASK_ADDR1RES1 (0x3) /* Reserved */
+#define MASK_RECALIBRATE (0x1 << 2) /* Recalibrate */
+#define MASK_SAMPLERATE (0x7 << 3) /* Sample Rate: */
+#define MASK_LOOPTHRU (0x1 << 6) /* Loopthrough Enable */
+#define SHIFT_LOOPTHRU 6
+#define MASK_CMUTE (0x1 << 7) /* Output C (Speaker) Mute when 1 */
+#define MASK_SPKMUTE MASK_CMUTE
+#define SHIFT_SPKMUTE 7
+#define MASK_ADDR1RES2 (0x1 << 8) /* Reserved */
+#define MASK_AMUTE (0x1 << 9) /* Output A (Headphone) Mute when 1 */
+#define MASK_HDMUTE MASK_AMUTE
+#define SHIFT_HDMUTE 9
+#define MASK_PAROUT (0x3 << 10) /* Parallel Out (???) */
+#define MASK_PAROUT0 (0x1 << 10) /* Parallel Out (???) */
+#define MASK_PAROUT1 (0x1 << 11) /* Parallel Out (enable speaker) */
+#define SHIFT_PAROUT 10
+#define SHIFT_PAROUT0 10
+#define SHIFT_PAROUT1 11
+
+#define SAMPLERATE_48000 (0x0 << 3) /* 48 or 44.1 kHz */
+#define SAMPLERATE_32000 (0x1 << 3) /* 32 or 29.4 kHz */
+#define SAMPLERATE_24000 (0x2 << 3) /* 24 or 22.05 kHz */
+#define SAMPLERATE_19200 (0x3 << 3) /* 19.2 or 17.64 kHz */
+#define SAMPLERATE_16000 (0x4 << 3) /* 16 or 14.7 kHz */
+#define SAMPLERATE_12000 (0x5 << 3) /* 12 or 11.025 kHz */
+#define SAMPLERATE_9600 (0x6 << 3) /* 9.6 or 8.82 kHz */
+#define SAMPLERATE_8000 (0x7 << 3) /* 8 or 7.35 kHz */
+
+/* Address 2 & 4 Bit Masks & Macros */
+/* ------- - - - --- ----- - ------ */
+#define MASK_OUTVOLRIGHT (0xf) /* Output Right Volume */
+#define MASK_ADDR2RES1 (0x2 << 4) /* Reserved */
+#define MASK_ADDR4RES1 MASK_ADDR2RES1
+#define MASK_OUTVOLLEFT (0xf << 6) /* Output Left Volume */
+#define MASK_ADDR2RES2 (0x2 << 10) /* Reserved */
+#define MASK_ADDR4RES2 MASK_ADDR2RES2
+
+#define VOLRIGHT(x) (((~(x)) & MASK_OUTVOLRIGHT))
+#define VOLLEFT(x) (((~(x)) << 6) & MASK_OUTVOLLEFT)
+
+/* address 6 */
+#define MASK_MIC_BOOST (0x4) /* screamer mic boost */
+#define SHIFT_MIC_BOOST 2
+
+/* Audio Codec Status Reg Bit Masks */
+/* ----- ----- ------ --- --- ----- */
+#define MASK_EXTEND (0x1 << 23) /* Extend */
+#define MASK_VALID (0x1 << 22) /* Valid Data? */
+#define MASK_OFLEFT (0x1 << 21) /* Overflow Left */
+#define MASK_OFRIGHT (0x1 << 20) /* Overflow Right */
+#define MASK_ERRCODE (0xf << 16) /* Error Code */
+#define MASK_REVISION (0xf << 12) /* Revision Number */
+#define MASK_MFGID (0xf << 8) /* Mfg. ID */
+#define MASK_CODSTATRES (0xf << 4) /* bits 4 - 7 reserved */
+#define MASK_INSENSE (0xf) /* port sense bits: */
+#define MASK_HDPCONN 8 /* headphone plugged in */
+#define MASK_LOCONN 4 /* line-out plugged in */
+#define MASK_LICONN 2 /* line-in plugged in */
+#define MASK_MICCONN 1 /* microphone plugged in */
+#define MASK_LICONN_IMAC 8 /* line-in plugged in */
+#define MASK_HDPRCONN_IMAC 4 /* headphone right plugged in */
+#define MASK_HDPLCONN_IMAC 2 /* headphone left plugged in */
+#define MASK_LOCONN_IMAC 1 /* line-out plugged in */
+
+/* Clipping Count Reg Bit Masks */
+/* -------- ----- --- --- ----- */
+#define MASK_CLIPLEFT (0xff << 7) /* Clipping Count, Left Channel */
+#define MASK_CLIPRIGHT (0xff) /* Clipping Count, Right Channel */
+
+/* DBDMA ChannelStatus Bit Masks */
+/* ----- ------------- --- ----- */
+#define MASK_CSERR (0x1 << 7) /* Error */
+#define MASK_EOI (0x1 << 6) /* End of Input --
+ only for Input Channel */
+#define MASK_CSUNUSED (0x1f << 1) /* bits 1-5 not used */
+#define MASK_WAIT (0x1) /* Wait */
+
+/* Various Rates */
+/* ------- ----- */
+#define RATE_48000 (0x0 << 8) /* 48 kHz */
+#define RATE_44100 (0x0 << 8) /* 44.1 kHz */
+#define RATE_32000 (0x1 << 8) /* 32 kHz */
+#define RATE_29400 (0x1 << 8) /* 29.4 kHz */
+#define RATE_24000 (0x2 << 8) /* 24 kHz */
+#define RATE_22050 (0x2 << 8) /* 22.05 kHz */
+#define RATE_19200 (0x3 << 8) /* 19.2 kHz */
+#define RATE_17640 (0x3 << 8) /* 17.64 kHz */
+#define RATE_16000 (0x4 << 8) /* 16 kHz */
+#define RATE_14700 (0x4 << 8) /* 14.7 kHz */
+#define RATE_12000 (0x5 << 8) /* 12 kHz */
+#define RATE_11025 (0x5 << 8) /* 11.025 kHz */
+#define RATE_9600 (0x6 << 8) /* 9.6 kHz */
+#define RATE_8820 (0x6 << 8) /* 8.82 kHz */
+#define RATE_8000 (0x7 << 8) /* 8 kHz */
+#define RATE_7350 (0x7 << 8) /* 7.35 kHz */
+
+#define RATE_LOW 1 /* HIGH = 48kHz, etc; LOW = 44.1kHz, etc. */
+
+
+#endif /* __AWACS_H */
diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c
new file mode 100644
index 0000000..89f5c32
--- /dev/null
+++ b/sound/ppc/beep.c
@@ -0,0 +1,285 @@
+/*
+ * Beep using pcm
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <asm/io.h>
+#include <asm/irq.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/input.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include "pmac.h"
+
+struct pmac_beep {
+ int running; /* boolean */
+ int volume; /* mixer volume: 0-100 */
+ int volume_play; /* currently playing volume */
+ int hz;
+ int nsamples;
+ short *buf; /* allocated wave buffer */
+ dma_addr_t addr; /* physical address of buffer */
+ struct input_dev *dev;
+};
+
+/*
+ * stop beep if running
+ */
+void snd_pmac_beep_stop(struct snd_pmac *chip)
+{
+ struct pmac_beep *beep = chip->beep;
+ if (beep && beep->running) {
+ beep->running = 0;
+ snd_pmac_beep_dma_stop(chip);
+ }
+}
+
+/*
+ * Stuff for outputting a beep. The values range from -327 to +327
+ * so we can multiply by an amplitude in the range 0..100 to get a
+ * signed short value to put in the output buffer.
+ */
+static short beep_wform[256] = {
+ 0, 40, 79, 117, 153, 187, 218, 245,
+ 269, 288, 304, 316, 323, 327, 327, 324,
+ 318, 310, 299, 288, 275, 262, 249, 236,
+ 224, 213, 204, 196, 190, 186, 183, 182,
+ 182, 183, 186, 189, 192, 196, 200, 203,
+ 206, 208, 209, 209, 209, 207, 204, 201,
+ 197, 193, 188, 183, 179, 174, 170, 166,
+ 163, 161, 160, 159, 159, 160, 161, 162,
+ 164, 166, 168, 169, 171, 171, 171, 170,
+ 169, 167, 163, 159, 155, 150, 144, 139,
+ 133, 128, 122, 117, 113, 110, 107, 105,
+ 103, 103, 103, 103, 104, 104, 105, 105,
+ 105, 103, 101, 97, 92, 86, 78, 68,
+ 58, 45, 32, 18, 3, -11, -26, -41,
+ -55, -68, -79, -88, -95, -100, -102, -102,
+ -99, -93, -85, -75, -62, -48, -33, -16,
+ 0, 16, 33, 48, 62, 75, 85, 93,
+ 99, 102, 102, 100, 95, 88, 79, 68,
+ 55, 41, 26, 11, -3, -18, -32, -45,
+ -58, -68, -78, -86, -92, -97, -101, -103,
+ -105, -105, -105, -104, -104, -103, -103, -103,
+ -103, -105, -107, -110, -113, -117, -122, -128,
+ -133, -139, -144, -150, -155, -159, -163, -167,
+ -169, -170, -171, -171, -171, -169, -168, -166,
+ -164, -162, -161, -160, -159, -159, -160, -161,
+ -163, -166, -170, -174, -179, -183, -188, -193,
+ -197, -201, -204, -207, -209, -209, -209, -208,
+ -206, -203, -200, -196, -192, -189, -186, -183,
+ -182, -182, -183, -186, -190, -196, -204, -213,
+ -224, -236, -249, -262, -275, -288, -299, -310,
+ -318, -324, -327, -327, -323, -316, -304, -288,
+ -269, -245, -218, -187, -153, -117, -79, -40,
+};
+
+#define BEEP_SRATE 22050 /* 22050 Hz sample rate */
+#define BEEP_BUFLEN 512
+#define BEEP_VOLUME 15 /* 0 - 100 */
+
+static int snd_pmac_beep_event(struct input_dev *dev, unsigned int type,
+ unsigned int code, int hz)
+{
+ struct snd_pmac *chip;
+ struct pmac_beep *beep;
+ unsigned long flags;
+ int beep_speed = 0;
+ int srate;
+ int period, ncycles, nsamples;
+ int i, j, f;
+ short *p;
+
+ if (type != EV_SND)
+ return -1;
+
+ switch (code) {
+ case SND_BELL: if (hz) hz = 1000;
+ case SND_TONE: break;
+ default: return -1;
+ }
+
+ chip = input_get_drvdata(dev);
+ if (! chip || (beep = chip->beep) == NULL)
+ return -1;
+
+ if (! hz) {
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (beep->running)
+ snd_pmac_beep_stop(chip);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+ }
+
+ beep_speed = snd_pmac_rate_index(chip, &chip->playback, BEEP_SRATE);
+ srate = chip->freq_table[beep_speed];
+
+ if (hz <= srate / BEEP_BUFLEN || hz > srate / 2)
+ hz = 1000;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ if (chip->playback.running || chip->capture.running || beep->running) {
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+ }
+ beep->running = 1;
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ if (hz == beep->hz && beep->volume == beep->volume_play) {
+ nsamples = beep->nsamples;
+ } else {
+ period = srate * 256 / hz; /* fixed point */
+ ncycles = BEEP_BUFLEN * 256 / period;
+ nsamples = (period * ncycles) >> 8;
+ f = ncycles * 65536 / nsamples;
+ j = 0;
+ p = beep->buf;
+ for (i = 0; i < nsamples; ++i, p += 2) {
+ p[0] = p[1] = beep_wform[j >> 8] * beep->volume;
+ j = (j + f) & 0xffff;
+ }
+ beep->hz = hz;
+ beep->volume_play = beep->volume;
+ beep->nsamples = nsamples;
+ }
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_pmac_beep_dma_start(chip, beep->nsamples * 4, beep->addr, beep_speed);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ return 0;
+}
+
+/*
+ * beep volume mixer
+ */
+
+static int snd_pmac_info_beep(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int snd_pmac_get_beep(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ if (snd_BUG_ON(!chip->beep))
+ return -ENXIO;
+ ucontrol->value.integer.value[0] = chip->beep->volume;
+ return 0;
+}
+
+static int snd_pmac_put_beep(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int oval, nval;
+ if (snd_BUG_ON(!chip->beep))
+ return -ENXIO;
+ oval = chip->beep->volume;
+ nval = ucontrol->value.integer.value[0];
+ if (nval > 100)
+ return -EINVAL;
+ chip->beep->volume = nval;
+ return oval != chip->beep->volume;
+}
+
+static struct snd_kcontrol_new snd_pmac_beep_mixer = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Beep Playback Volume",
+ .info = snd_pmac_info_beep,
+ .get = snd_pmac_get_beep,
+ .put = snd_pmac_put_beep,
+};
+
+/* Initialize beep stuff */
+int __init snd_pmac_attach_beep(struct snd_pmac *chip)
+{
+ struct pmac_beep *beep;
+ struct input_dev *input_dev;
+ struct snd_kcontrol *beep_ctl;
+ void *dmabuf;
+ int err = -ENOMEM;
+
+ beep = kzalloc(sizeof(*beep), GFP_KERNEL);
+ if (! beep)
+ return -ENOMEM;
+ dmabuf = dma_alloc_coherent(&chip->pdev->dev, BEEP_BUFLEN * 4,
+ &beep->addr, GFP_KERNEL);
+ input_dev = input_allocate_device();
+ if (! dmabuf || ! input_dev)
+ goto fail1;
+
+ /* FIXME: set more better values */
+ input_dev->name = "PowerMac Beep";
+ input_dev->phys = "powermac/beep";
+ input_dev->id.bustype = BUS_ADB;
+ input_dev->id.vendor = 0x001f;
+ input_dev->id.product = 0x0001;
+ input_dev->id.version = 0x0100;
+
+ input_dev->evbit[0] = BIT_MASK(EV_SND);
+ input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE);
+ input_dev->event = snd_pmac_beep_event;
+ input_dev->dev.parent = &chip->pdev->dev;
+ input_set_drvdata(input_dev, chip);
+
+ beep->dev = input_dev;
+ beep->buf = dmabuf;
+ beep->volume = BEEP_VOLUME;
+ beep->running = 0;
+
+ beep_ctl = snd_ctl_new1(&snd_pmac_beep_mixer, chip);
+ err = snd_ctl_add(chip->card, beep_ctl);
+ if (err < 0)
+ goto fail1;
+
+ chip->beep = beep;
+
+ err = input_register_device(beep->dev);
+ if (err)
+ goto fail2;
+
+ return 0;
+
+ fail2: snd_ctl_remove(chip->card, beep_ctl);
+ fail1: input_free_device(input_dev);
+ if (dmabuf)
+ dma_free_coherent(&chip->pdev->dev, BEEP_BUFLEN * 4,
+ dmabuf, beep->addr);
+ kfree(beep);
+ return err;
+}
+
+void snd_pmac_detach_beep(struct snd_pmac *chip)
+{
+ if (chip->beep) {
+ input_unregister_device(chip->beep->dev);
+ dma_free_coherent(&chip->pdev->dev, BEEP_BUFLEN * 4,
+ chip->beep->buf, chip->beep->addr);
+ kfree(chip->beep);
+ chip->beep = NULL;
+ }
+}
diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c
new file mode 100644
index 0000000..f860d39
--- /dev/null
+++ b/sound/ppc/burgundy.c
@@ -0,0 +1,733 @@
+/*
+ * PMac Burgundy lowlevel functions
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ * code based on dmasound.c.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <asm/io.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include "pmac.h"
+#include "burgundy.h"
+
+
+/* Waits for busy flag to clear */
+static inline void
+snd_pmac_burgundy_busy_wait(struct snd_pmac *chip)
+{
+ int timeout = 50;
+ while ((in_le32(&chip->awacs->codec_ctrl) & MASK_NEWECMD) && timeout--)
+ udelay(1);
+ if (! timeout)
+ printk(KERN_DEBUG "burgundy_busy_wait: timeout\n");
+}
+
+static inline void
+snd_pmac_burgundy_extend_wait(struct snd_pmac *chip)
+{
+ int timeout;
+ timeout = 50;
+ while (!(in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--)
+ udelay(1);
+ if (! timeout)
+ printk(KERN_DEBUG "burgundy_extend_wait: timeout #1\n");
+ timeout = 50;
+ while ((in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--)
+ udelay(1);
+ if (! timeout)
+ printk(KERN_DEBUG "burgundy_extend_wait: timeout #2\n");
+}
+
+static void
+snd_pmac_burgundy_wcw(struct snd_pmac *chip, unsigned addr, unsigned val)
+{
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x200c00 + (val & 0xff));
+ snd_pmac_burgundy_busy_wait(chip);
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x200d00 +((val>>8) & 0xff));
+ snd_pmac_burgundy_busy_wait(chip);
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x200e00 +((val>>16) & 0xff));
+ snd_pmac_burgundy_busy_wait(chip);
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x200f00 +((val>>24) & 0xff));
+ snd_pmac_burgundy_busy_wait(chip);
+}
+
+static unsigned
+snd_pmac_burgundy_rcw(struct snd_pmac *chip, unsigned addr)
+{
+ unsigned val = 0;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x100000);
+ snd_pmac_burgundy_busy_wait(chip);
+ snd_pmac_burgundy_extend_wait(chip);
+ val += (in_le32(&chip->awacs->codec_stat) >> 4) & 0xff;
+
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x100100);
+ snd_pmac_burgundy_busy_wait(chip);
+ snd_pmac_burgundy_extend_wait(chip);
+ val += ((in_le32(&chip->awacs->codec_stat)>>4) & 0xff) <<8;
+
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x100200);
+ snd_pmac_burgundy_busy_wait(chip);
+ snd_pmac_burgundy_extend_wait(chip);
+ val += ((in_le32(&chip->awacs->codec_stat)>>4) & 0xff) <<16;
+
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x100300);
+ snd_pmac_burgundy_busy_wait(chip);
+ snd_pmac_burgundy_extend_wait(chip);
+ val += ((in_le32(&chip->awacs->codec_stat)>>4) & 0xff) <<24;
+
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ return val;
+}
+
+static void
+snd_pmac_burgundy_wcb(struct snd_pmac *chip, unsigned int addr,
+ unsigned int val)
+{
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x300000 + (val & 0xff));
+ snd_pmac_burgundy_busy_wait(chip);
+}
+
+static unsigned
+snd_pmac_burgundy_rcb(struct snd_pmac *chip, unsigned int addr)
+{
+ unsigned val = 0;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->reg_lock, flags);
+
+ out_le32(&chip->awacs->codec_ctrl, addr + 0x100000);
+ snd_pmac_burgundy_busy_wait(chip);
+ snd_pmac_burgundy_extend_wait(chip);
+ val += (in_le32(&chip->awacs->codec_stat) >> 4) & 0xff;
+
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+ return val;
+}
+
+#define BASE2ADDR(base) ((base) << 12)
+#define ADDR2BASE(addr) ((addr) >> 12)
+
+/*
+ * Burgundy volume: 0 - 100, stereo, word reg
+ */
+static void
+snd_pmac_burgundy_write_volume(struct snd_pmac *chip, unsigned int address,
+ long *volume, int shift)
+{
+ int hardvolume, lvolume, rvolume;
+
+ if (volume[0] < 0 || volume[0] > 100 ||
+ volume[1] < 0 || volume[1] > 100)
+ return; /* -EINVAL */
+ lvolume = volume[0] ? volume[0] + BURGUNDY_VOLUME_OFFSET : 0;
+ rvolume = volume[1] ? volume[1] + BURGUNDY_VOLUME_OFFSET : 0;
+
+ hardvolume = lvolume + (rvolume << shift);
+ if (shift == 8)
+ hardvolume |= hardvolume << 16;
+
+ snd_pmac_burgundy_wcw(chip, address, hardvolume);
+}
+
+static void
+snd_pmac_burgundy_read_volume(struct snd_pmac *chip, unsigned int address,
+ long *volume, int shift)
+{
+ int wvolume;
+
+ wvolume = snd_pmac_burgundy_rcw(chip, address);
+
+ volume[0] = wvolume & 0xff;
+ if (volume[0] >= BURGUNDY_VOLUME_OFFSET)
+ volume[0] -= BURGUNDY_VOLUME_OFFSET;
+ else
+ volume[0] = 0;
+ volume[1] = (wvolume >> shift) & 0xff;
+ if (volume[1] >= BURGUNDY_VOLUME_OFFSET)
+ volume[1] -= BURGUNDY_VOLUME_OFFSET;
+ else
+ volume[1] = 0;
+}
+
+static int snd_pmac_burgundy_info_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int snd_pmac_burgundy_get_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff);
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ snd_pmac_burgundy_read_volume(chip, addr,
+ ucontrol->value.integer.value, shift);
+ return 0;
+}
+
+static int snd_pmac_burgundy_put_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff);
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ long nvoices[2];
+
+ snd_pmac_burgundy_write_volume(chip, addr,
+ ucontrol->value.integer.value, shift);
+ snd_pmac_burgundy_read_volume(chip, addr, nvoices, shift);
+ return (nvoices[0] != ucontrol->value.integer.value[0] ||
+ nvoices[1] != ucontrol->value.integer.value[1]);
+}
+
+#define BURGUNDY_VOLUME_W(xname, xindex, addr, shift) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\
+ .info = snd_pmac_burgundy_info_volume,\
+ .get = snd_pmac_burgundy_get_volume,\
+ .put = snd_pmac_burgundy_put_volume,\
+ .private_value = ((ADDR2BASE(addr) & 0xff) | ((shift) << 8)) }
+
+/*
+ * Burgundy volume: 0 - 100, stereo, 2-byte reg
+ */
+static void
+snd_pmac_burgundy_write_volume_2b(struct snd_pmac *chip, unsigned int address,
+ long *volume, int off)
+{
+ int lvolume, rvolume;
+
+ off |= off << 2;
+ lvolume = volume[0] ? volume[0] + BURGUNDY_VOLUME_OFFSET : 0;
+ rvolume = volume[1] ? volume[1] + BURGUNDY_VOLUME_OFFSET : 0;
+
+ snd_pmac_burgundy_wcb(chip, address + off, lvolume);
+ snd_pmac_burgundy_wcb(chip, address + off + 0x500, rvolume);
+}
+
+static void
+snd_pmac_burgundy_read_volume_2b(struct snd_pmac *chip, unsigned int address,
+ long *volume, int off)
+{
+ volume[0] = snd_pmac_burgundy_rcb(chip, address + off);
+ if (volume[0] >= BURGUNDY_VOLUME_OFFSET)
+ volume[0] -= BURGUNDY_VOLUME_OFFSET;
+ else
+ volume[0] = 0;
+ volume[1] = snd_pmac_burgundy_rcb(chip, address + off + 0x100);
+ if (volume[1] >= BURGUNDY_VOLUME_OFFSET)
+ volume[1] -= BURGUNDY_VOLUME_OFFSET;
+ else
+ volume[1] = 0;
+}
+
+static int snd_pmac_burgundy_info_volume_2b(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int snd_pmac_burgundy_get_volume_2b(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff);
+ int off = kcontrol->private_value & 0x300;
+ snd_pmac_burgundy_read_volume_2b(chip, addr,
+ ucontrol->value.integer.value, off);
+ return 0;
+}
+
+static int snd_pmac_burgundy_put_volume_2b(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff);
+ int off = kcontrol->private_value & 0x300;
+ long nvoices[2];
+
+ snd_pmac_burgundy_write_volume_2b(chip, addr,
+ ucontrol->value.integer.value, off);
+ snd_pmac_burgundy_read_volume_2b(chip, addr, nvoices, off);
+ return (nvoices[0] != ucontrol->value.integer.value[0] ||
+ nvoices[1] != ucontrol->value.integer.value[1]);
+}
+
+#define BURGUNDY_VOLUME_2B(xname, xindex, addr, off) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\
+ .info = snd_pmac_burgundy_info_volume_2b,\
+ .get = snd_pmac_burgundy_get_volume_2b,\
+ .put = snd_pmac_burgundy_put_volume_2b,\
+ .private_value = ((ADDR2BASE(addr) & 0xff) | ((off) << 8)) }
+
+/*
+ * Burgundy gain/attenuation: 0 - 15, mono/stereo, byte reg
+ */
+static int snd_pmac_burgundy_info_gain(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = stereo + 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 15;
+ return 0;
+}
+
+static int snd_pmac_burgundy_get_gain(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff);
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ int atten = (kcontrol->private_value >> 25) & 1;
+ int oval;
+
+ oval = snd_pmac_burgundy_rcb(chip, addr);
+ if (atten)
+ oval = ~oval & 0xff;
+ ucontrol->value.integer.value[0] = oval & 0xf;
+ if (stereo)
+ ucontrol->value.integer.value[1] = (oval >> 4) & 0xf;
+ return 0;
+}
+
+static int snd_pmac_burgundy_put_gain(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR(kcontrol->private_value & 0xff);
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ int atten = (kcontrol->private_value >> 25) & 1;
+ int oval, val;
+
+ oval = snd_pmac_burgundy_rcb(chip, addr);
+ if (atten)
+ oval = ~oval & 0xff;
+ val = ucontrol->value.integer.value[0];
+ if (stereo)
+ val |= ucontrol->value.integer.value[1] << 4;
+ else
+ val |= ucontrol->value.integer.value[0] << 4;
+ if (atten)
+ val = ~val & 0xff;
+ snd_pmac_burgundy_wcb(chip, addr, val);
+ return val != oval;
+}
+
+#define BURGUNDY_VOLUME_B(xname, xindex, addr, stereo, atten) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\
+ .info = snd_pmac_burgundy_info_gain,\
+ .get = snd_pmac_burgundy_get_gain,\
+ .put = snd_pmac_burgundy_put_gain,\
+ .private_value = (ADDR2BASE(addr) | ((stereo) << 24) | ((atten) << 25)) }
+
+/*
+ * Burgundy switch: 0/1, mono/stereo, word reg
+ */
+static int snd_pmac_burgundy_info_switch_w(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = stereo + 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_pmac_burgundy_get_switch_w(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff);
+ int lmask = 1 << (kcontrol->private_value & 0xff);
+ int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff);
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ int val = snd_pmac_burgundy_rcw(chip, addr);
+ ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0;
+ if (stereo)
+ ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0;
+ return 0;
+}
+
+static int snd_pmac_burgundy_put_switch_w(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff);
+ int lmask = 1 << (kcontrol->private_value & 0xff);
+ int rmask = 1 << ((kcontrol->private_value >> 8) & 0xff);
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ int val, oval;
+ oval = snd_pmac_burgundy_rcw(chip, addr);
+ val = oval & ~(lmask | (stereo ? rmask : 0));
+ if (ucontrol->value.integer.value[0])
+ val |= lmask;
+ if (stereo && ucontrol->value.integer.value[1])
+ val |= rmask;
+ snd_pmac_burgundy_wcw(chip, addr, val);
+ return val != oval;
+}
+
+#define BURGUNDY_SWITCH_W(xname, xindex, addr, lbit, rbit, stereo) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\
+ .info = snd_pmac_burgundy_info_switch_w,\
+ .get = snd_pmac_burgundy_get_switch_w,\
+ .put = snd_pmac_burgundy_put_switch_w,\
+ .private_value = ((lbit) | ((rbit) << 8)\
+ | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) }
+
+/*
+ * Burgundy switch: 0/1, mono/stereo, byte reg, bit mask
+ */
+static int snd_pmac_burgundy_info_switch_b(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = stereo + 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_pmac_burgundy_get_switch_b(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff);
+ int lmask = kcontrol->private_value & 0xff;
+ int rmask = (kcontrol->private_value >> 8) & 0xff;
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ int val = snd_pmac_burgundy_rcb(chip, addr);
+ ucontrol->value.integer.value[0] = (val & lmask) ? 1 : 0;
+ if (stereo)
+ ucontrol->value.integer.value[1] = (val & rmask) ? 1 : 0;
+ return 0;
+}
+
+static int snd_pmac_burgundy_put_switch_b(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int addr = BASE2ADDR((kcontrol->private_value >> 16) & 0xff);
+ int lmask = kcontrol->private_value & 0xff;
+ int rmask = (kcontrol->private_value >> 8) & 0xff;
+ int stereo = (kcontrol->private_value >> 24) & 1;
+ int val, oval;
+ oval = snd_pmac_burgundy_rcb(chip, addr);
+ val = oval & ~(lmask | rmask);
+ if (ucontrol->value.integer.value[0])
+ val |= lmask;
+ if (stereo && ucontrol->value.integer.value[1])
+ val |= rmask;
+ snd_pmac_burgundy_wcb(chip, addr, val);
+ return val != oval;
+}
+
+#define BURGUNDY_SWITCH_B(xname, xindex, addr, lmask, rmask, stereo) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\
+ .info = snd_pmac_burgundy_info_switch_b,\
+ .get = snd_pmac_burgundy_get_switch_b,\
+ .put = snd_pmac_burgundy_put_switch_b,\
+ .private_value = ((lmask) | ((rmask) << 8)\
+ | (ADDR2BASE(addr) << 16) | ((stereo) << 24)) }
+
+/*
+ * Burgundy mixers
+ */
+static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __initdata = {
+ BURGUNDY_VOLUME_W("Master Playback Volume", 0,
+ MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8),
+ BURGUNDY_VOLUME_W("CD Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_VOLCD, 16),
+ BURGUNDY_VOLUME_2B("Input Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_VOLMIX01, 2),
+ BURGUNDY_VOLUME_2B("Mixer Playback Volume", 0,
+ MASK_ADDR_BURGUNDY_VOLMIX23, 0),
+ BURGUNDY_VOLUME_B("CD Gain Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_GAINCD, 1, 0),
+ BURGUNDY_SWITCH_W("Master Capture Switch", 0,
+ MASK_ADDR_BURGUNDY_OUTPUTENABLES, 24, 0, 0),
+ BURGUNDY_SWITCH_W("CD Capture Switch", 0,
+ MASK_ADDR_BURGUNDY_CAPTURESELECTS, 0, 16, 1),
+ BURGUNDY_SWITCH_W("CD Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 0, 16, 1),
+/* BURGUNDY_SWITCH_W("Loop Capture Switch", 0,
+ * MASK_ADDR_BURGUNDY_CAPTURESELECTS, 8, 24, 1),
+ * BURGUNDY_SWITCH_B("Mixer out Capture Switch", 0,
+ * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x02, 0, 0),
+ * BURGUNDY_SWITCH_B("Mixer Capture Switch", 0,
+ * MASK_ADDR_BURGUNDY_HOSTIFAD, 0x01, 0, 0),
+ * BURGUNDY_SWITCH_B("PCM out Capture Switch", 0,
+ * MASK_ADDR_BURGUNDY_HOSTIFEH, 0x02, 0, 0),
+ */ BURGUNDY_SWITCH_B("PCM Capture Switch", 0,
+ MASK_ADDR_BURGUNDY_HOSTIFEH, 0x01, 0, 0)
+};
+static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __initdata = {
+ BURGUNDY_VOLUME_W("Line in Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_VOLLINE, 16),
+ BURGUNDY_VOLUME_W("Mic Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_VOLMIC, 16),
+ BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_GAINLINE, 1, 0),
+ BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_GAINMIC, 1, 0),
+ BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0,
+ MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1),
+ BURGUNDY_VOLUME_B("Line out Playback Volume", 0,
+ MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1),
+ BURGUNDY_VOLUME_B("Headphone Playback Volume", 0,
+ MASK_ADDR_BURGUNDY_ATTENHP, 1, 1),
+ BURGUNDY_SWITCH_W("Line in Capture Switch", 0,
+ MASK_ADDR_BURGUNDY_CAPTURESELECTS, 1, 17, 1),
+ BURGUNDY_SWITCH_W("Mic Capture Switch", 0,
+ MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1),
+ BURGUNDY_SWITCH_W("Line in Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 1, 17, 1),
+ BURGUNDY_SWITCH_W("Mic Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1),
+ BURGUNDY_SWITCH_B("Mic Boost Capture Switch", 0,
+ MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1)
+};
+static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __initdata = {
+ BURGUNDY_VOLUME_W("Line in Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_VOLMIC, 16),
+ BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0,
+ MASK_ADDR_BURGUNDY_GAINMIC, 1, 0),
+ BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0,
+ MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1),
+ BURGUNDY_VOLUME_B("Line out Playback Volume", 0,
+ MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1),
+ BURGUNDY_SWITCH_W("Line in Capture Switch", 0,
+ MASK_ADDR_BURGUNDY_CAPTURESELECTS, 2, 18, 1),
+ BURGUNDY_SWITCH_W("Line in Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_OUTPUTSELECTS, 2, 18, 1),
+/* BURGUNDY_SWITCH_B("Line in Boost Capture Switch", 0,
+ * MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) */
+};
+static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __initdata =
+BURGUNDY_SWITCH_B("Master Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ BURGUNDY_OUTPUT_LEFT | BURGUNDY_LINEOUT_LEFT | BURGUNDY_HP_LEFT,
+ BURGUNDY_OUTPUT_RIGHT | BURGUNDY_LINEOUT_RIGHT | BURGUNDY_HP_RIGHT, 1);
+static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __initdata =
+BURGUNDY_SWITCH_B("Master Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ BURGUNDY_OUTPUT_INTERN
+ | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
+static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __initdata =
+BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
+static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __initdata =
+BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ BURGUNDY_OUTPUT_INTERN, 0, 0);
+static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __initdata =
+BURGUNDY_SWITCH_B("Line out Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ BURGUNDY_LINEOUT_LEFT, BURGUNDY_LINEOUT_RIGHT, 1);
+static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __initdata =
+BURGUNDY_SWITCH_B("Line out Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
+static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __initdata =
+BURGUNDY_SWITCH_B("Headphone Playback Switch", 0,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ BURGUNDY_HP_LEFT, BURGUNDY_HP_RIGHT, 1);
+
+
+#ifdef PMAC_SUPPORT_AUTOMUTE
+/*
+ * auto-mute stuffs
+ */
+static int snd_pmac_burgundy_detect_headphone(struct snd_pmac *chip)
+{
+ return (in_le32(&chip->awacs->codec_stat) & chip->hp_stat_mask) ? 1 : 0;
+}
+
+static void snd_pmac_burgundy_update_automute(struct snd_pmac *chip, int do_notify)
+{
+ if (chip->auto_mute) {
+ int imac = machine_is_compatible("iMac");
+ int reg, oreg;
+ reg = oreg = snd_pmac_burgundy_rcb(chip,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES);
+ reg &= imac ? ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT
+ | BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT)
+ : ~(BURGUNDY_OUTPUT_LEFT | BURGUNDY_OUTPUT_RIGHT
+ | BURGUNDY_OUTPUT_INTERN);
+ if (snd_pmac_burgundy_detect_headphone(chip))
+ reg |= imac ? (BURGUNDY_HP_LEFT | BURGUNDY_HP_RIGHT)
+ : (BURGUNDY_OUTPUT_LEFT
+ | BURGUNDY_OUTPUT_RIGHT);
+ else
+ reg |= imac ? (BURGUNDY_OUTPUT_LEFT
+ | BURGUNDY_OUTPUT_RIGHT)
+ : (BURGUNDY_OUTPUT_INTERN);
+ if (do_notify && reg == oreg)
+ return;
+ snd_pmac_burgundy_wcb(chip,
+ MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, reg);
+ if (do_notify) {
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_sw_ctl->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->speaker_sw_ctl->id);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->hp_detect_ctl->id);
+ }
+ }
+}
+#endif /* PMAC_SUPPORT_AUTOMUTE */
+
+
+/*
+ * initialize burgundy
+ */
+int __init snd_pmac_burgundy_init(struct snd_pmac *chip)
+{
+ int imac = machine_is_compatible("iMac");
+ int i, err;
+
+ /* Checks to see the chip is alive and kicking */
+ if ((in_le32(&chip->awacs->codec_ctrl) & MASK_ERRCODE) == 0xf0000) {
+ printk(KERN_WARNING "pmac burgundy: disabled by MacOS :-(\n");
+ return 1;
+ }
+
+ snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_OUTPUTENABLES,
+ DEF_BURGUNDY_OUTPUTENABLES);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
+ DEF_BURGUNDY_MORE_OUTPUTENABLES);
+ snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_OUTPUTSELECTS,
+ DEF_BURGUNDY_OUTPUTSELECTS);
+
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL21,
+ DEF_BURGUNDY_INPSEL21);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_INPSEL3,
+ imac ? DEF_BURGUNDY_INPSEL3_IMAC
+ : DEF_BURGUNDY_INPSEL3_PMAC);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINCD,
+ DEF_BURGUNDY_GAINCD);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINLINE,
+ DEF_BURGUNDY_GAINLINE);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINMIC,
+ DEF_BURGUNDY_GAINMIC);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_GAINMODEM,
+ DEF_BURGUNDY_GAINMODEM);
+
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_ATTENSPEAKER,
+ DEF_BURGUNDY_ATTENSPEAKER);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_ATTENLINEOUT,
+ DEF_BURGUNDY_ATTENLINEOUT);
+ snd_pmac_burgundy_wcb(chip, MASK_ADDR_BURGUNDY_ATTENHP,
+ DEF_BURGUNDY_ATTENHP);
+
+ snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_MASTER_VOLUME,
+ DEF_BURGUNDY_MASTER_VOLUME);
+ snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_VOLCD,
+ DEF_BURGUNDY_VOLCD);
+ snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_VOLLINE,
+ DEF_BURGUNDY_VOLLINE);
+ snd_pmac_burgundy_wcw(chip, MASK_ADDR_BURGUNDY_VOLMIC,
+ DEF_BURGUNDY_VOLMIC);
+
+ if (chip->hp_stat_mask == 0) {
+ /* set headphone-jack detection bit */
+ if (imac)
+ chip->hp_stat_mask = BURGUNDY_HPDETECT_IMAC_UPPER
+ | BURGUNDY_HPDETECT_IMAC_LOWER
+ | BURGUNDY_HPDETECT_IMAC_SIDE;
+ else
+ chip->hp_stat_mask = BURGUNDY_HPDETECT_PMAC_BACK;
+ }
+ /*
+ * build burgundy mixers
+ */
+ strcpy(chip->card->mixername, "PowerMac Burgundy");
+
+ for (i = 0; i < ARRAY_SIZE(snd_pmac_burgundy_mixers); i++) {
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&snd_pmac_burgundy_mixers[i], chip));
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < (imac ? ARRAY_SIZE(snd_pmac_burgundy_mixers_imac)
+ : ARRAY_SIZE(snd_pmac_burgundy_mixers_pmac)); i++) {
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(imac ? &snd_pmac_burgundy_mixers_imac[i]
+ : &snd_pmac_burgundy_mixers_pmac[i], chip));
+ if (err < 0)
+ return err;
+ }
+ chip->master_sw_ctl = snd_ctl_new1(imac
+ ? &snd_pmac_burgundy_master_sw_imac
+ : &snd_pmac_burgundy_master_sw_pmac, chip);
+ err = snd_ctl_add(chip->card, chip->master_sw_ctl);
+ if (err < 0)
+ return err;
+ chip->master_sw_ctl = snd_ctl_new1(imac
+ ? &snd_pmac_burgundy_line_sw_imac
+ : &snd_pmac_burgundy_line_sw_pmac, chip);
+ err = snd_ctl_add(chip->card, chip->master_sw_ctl);
+ if (err < 0)
+ return err;
+ if (imac) {
+ chip->master_sw_ctl = snd_ctl_new1(
+ &snd_pmac_burgundy_hp_sw_imac, chip);
+ err = snd_ctl_add(chip->card, chip->master_sw_ctl);
+ if (err < 0)
+ return err;
+ }
+ chip->speaker_sw_ctl = snd_ctl_new1(imac
+ ? &snd_pmac_burgundy_speaker_sw_imac
+ : &snd_pmac_burgundy_speaker_sw_pmac, chip);
+ err = snd_ctl_add(chip->card, chip->speaker_sw_ctl);
+ if (err < 0)
+ return err;
+#ifdef PMAC_SUPPORT_AUTOMUTE
+ err = snd_pmac_add_automute(chip);
+ if (err < 0)
+ return err;
+
+ chip->detect_headphone = snd_pmac_burgundy_detect_headphone;
+ chip->update_automute = snd_pmac_burgundy_update_automute;
+ snd_pmac_burgundy_update_automute(chip, 0); /* update the status only */
+#endif
+
+ return 0;
+}
diff --git a/sound/ppc/burgundy.h b/sound/ppc/burgundy.h
new file mode 100644
index 0000000..7a7f9cf
--- /dev/null
+++ b/sound/ppc/burgundy.h
@@ -0,0 +1,114 @@
+/*
+ * Driver for PowerMac Burgundy onboard soundchips
+ * Copyright (c) 2001 by Takashi Iwai <tiwai@suse.de>
+ * based on dmasound.c.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#ifndef __BURGUNDY_H
+#define __BURGUNDY_H
+
+#define MASK_ADDR_BURGUNDY_INPBOOST (0x10 << 12)
+#define MASK_ADDR_BURGUNDY_INPSEL21 (0x11 << 12)
+#define MASK_ADDR_BURGUNDY_INPSEL3 (0x12 << 12)
+
+#define MASK_ADDR_BURGUNDY_GAINCH1 (0x13 << 12)
+#define MASK_ADDR_BURGUNDY_GAINCH2 (0x14 << 12)
+#define MASK_ADDR_BURGUNDY_GAINCH3 (0x15 << 12)
+#define MASK_ADDR_BURGUNDY_GAINCH4 (0x16 << 12)
+
+#define MASK_ADDR_BURGUNDY_VOLCH1 (0x20 << 12)
+#define MASK_ADDR_BURGUNDY_VOLCH2 (0x21 << 12)
+#define MASK_ADDR_BURGUNDY_VOLCH3 (0x22 << 12)
+#define MASK_ADDR_BURGUNDY_VOLCH4 (0x23 << 12)
+
+#define MASK_ADDR_BURGUNDY_CAPTURESELECTS (0x2A << 12)
+#define MASK_ADDR_BURGUNDY_OUTPUTSELECTS (0x2B << 12)
+#define MASK_ADDR_BURGUNDY_VOLMIX01 (0x2D << 12)
+#define MASK_ADDR_BURGUNDY_VOLMIX23 (0x2E << 12)
+#define MASK_ADDR_BURGUNDY_OUTPUTENABLES (0x2F << 12)
+
+#define MASK_ADDR_BURGUNDY_MASTER_VOLUME (0x30 << 12)
+
+#define MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES (0x60 << 12)
+
+#define MASK_ADDR_BURGUNDY_ATTENSPEAKER (0x62 << 12)
+#define MASK_ADDR_BURGUNDY_ATTENLINEOUT (0x63 << 12)
+#define MASK_ADDR_BURGUNDY_ATTENHP (0x64 << 12)
+#define MASK_ADDR_BURGUNDY_ATTENMONO (0x65 << 12)
+
+#define MASK_ADDR_BURGUNDY_HOSTIFAD (0x78 << 12)
+#define MASK_ADDR_BURGUNDY_HOSTIFEH (0x79 << 12)
+
+#define MASK_ADDR_BURGUNDY_VOLCD (MASK_ADDR_BURGUNDY_VOLCH1)
+#define MASK_ADDR_BURGUNDY_VOLLINE (MASK_ADDR_BURGUNDY_VOLCH2)
+#define MASK_ADDR_BURGUNDY_VOLMIC (MASK_ADDR_BURGUNDY_VOLCH3)
+#define MASK_ADDR_BURGUNDY_VOLMODEM (MASK_ADDR_BURGUNDY_VOLCH4)
+
+#define MASK_ADDR_BURGUNDY_GAINCD (MASK_ADDR_BURGUNDY_GAINCH1)
+#define MASK_ADDR_BURGUNDY_GAINLINE (MASK_ADDR_BURGUNDY_GAINCH2)
+#define MASK_ADDR_BURGUNDY_GAINMIC (MASK_ADDR_BURGUNDY_GAINCH3)
+#define MASK_ADDR_BURGUNDY_GAINMODEM (MASK_ADDR_BURGUNDY_VOLCH4)
+
+
+/* These are all default values for the burgundy */
+#define DEF_BURGUNDY_INPSEL21 (0xAA)
+#define DEF_BURGUNDY_INPSEL3_IMAC (0x0A)
+#define DEF_BURGUNDY_INPSEL3_PMAC (0x05)
+
+#define DEF_BURGUNDY_GAINCD (0x33)
+#define DEF_BURGUNDY_GAINLINE (0x44)
+#define DEF_BURGUNDY_GAINMIC (0x44)
+#define DEF_BURGUNDY_GAINMODEM (0x06)
+
+/* Remember: lowest volume here is 0x9B (155) */
+#define DEF_BURGUNDY_VOLCD (0xCCCCCCCC)
+#define DEF_BURGUNDY_VOLLINE (0x00000000)
+#define DEF_BURGUNDY_VOLMIC (0x00000000)
+#define DEF_BURGUNDY_VOLMODEM (0xCCCCCCCC)
+
+#define DEF_BURGUNDY_OUTPUTSELECTS (0x010F010F)
+#define DEF_BURGUNDY_OUTPUTENABLES (0x0100000A)
+
+/* #define DEF_BURGUNDY_MASTER_VOLUME (0xFFFFFFFF) */ /* too loud */
+#define DEF_BURGUNDY_MASTER_VOLUME (0xDDDDDDDD)
+
+#define DEF_BURGUNDY_MORE_OUTPUTENABLES (0x7E)
+
+#define DEF_BURGUNDY_ATTENSPEAKER (0x44)
+#define DEF_BURGUNDY_ATTENLINEOUT (0xCC)
+#define DEF_BURGUNDY_ATTENHP (0xCC)
+
+/* MORE_OUTPUTENABLES bits */
+#define BURGUNDY_OUTPUT_LEFT 0x02
+#define BURGUNDY_OUTPUT_RIGHT 0x04
+#define BURGUNDY_LINEOUT_LEFT 0x08
+#define BURGUNDY_LINEOUT_RIGHT 0x10
+#define BURGUNDY_HP_LEFT 0x20
+#define BURGUNDY_HP_RIGHT 0x40
+#define BURGUNDY_OUTPUT_INTERN 0x80
+
+/* Headphone detection bits */
+#define BURGUNDY_HPDETECT_PMAC_BACK 0x04
+#define BURGUNDY_HPDETECT_IMAC_SIDE 0x04
+#define BURGUNDY_HPDETECT_IMAC_UPPER 0x08
+#define BURGUNDY_HPDETECT_IMAC_LOWER 0x01
+
+/* Volume offset */
+#define BURGUNDY_VOLUME_OFFSET 155
+
+#endif /* __BURGUNDY_H */
diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c
new file mode 100644
index 0000000..8a5b290
--- /dev/null
+++ b/sound/ppc/daca.c
@@ -0,0 +1,282 @@
+/*
+ * PMac DACA lowlevel functions
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/kmod.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "pmac.h"
+
+/* i2c address */
+#define DACA_I2C_ADDR 0x4d
+
+/* registers */
+#define DACA_REG_SR 0x01
+#define DACA_REG_AVOL 0x02
+#define DACA_REG_GCFG 0x03
+
+/* maximum volume value */
+#define DACA_VOL_MAX 0x38
+
+
+struct pmac_daca {
+ struct pmac_keywest i2c;
+ int left_vol, right_vol;
+ unsigned int deemphasis : 1;
+ unsigned int amp_on : 1;
+};
+
+
+/*
+ * initialize / detect DACA
+ */
+static int daca_init_client(struct pmac_keywest *i2c)
+{
+ unsigned short wdata = 0x00;
+ /* SR: no swap, 1bit delay, 32-48kHz */
+ /* GCFG: power amp inverted, DAC on */
+ if (i2c_smbus_write_byte_data(i2c->client, DACA_REG_SR, 0x08) < 0 ||
+ i2c_smbus_write_byte_data(i2c->client, DACA_REG_GCFG, 0x05) < 0)
+ return -EINVAL;
+ return i2c_smbus_write_block_data(i2c->client, DACA_REG_AVOL,
+ 2, (unsigned char*)&wdata);
+}
+
+/*
+ * update volume
+ */
+static int daca_set_volume(struct pmac_daca *mix)
+{
+ unsigned char data[2];
+
+ if (! mix->i2c.client)
+ return -ENODEV;
+
+ if (mix->left_vol > DACA_VOL_MAX)
+ data[0] = DACA_VOL_MAX;
+ else
+ data[0] = mix->left_vol;
+ if (mix->right_vol > DACA_VOL_MAX)
+ data[1] = DACA_VOL_MAX;
+ else
+ data[1] = mix->right_vol;
+ data[1] |= mix->deemphasis ? 0x40 : 0;
+ if (i2c_smbus_write_block_data(mix->i2c.client, DACA_REG_AVOL,
+ 2, data) < 0) {
+ snd_printk("failed to set volume \n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/* deemphasis switch */
+#define daca_info_deemphasis snd_ctl_boolean_mono_info
+
+static int daca_get_deemphasis(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_daca *mix;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ ucontrol->value.integer.value[0] = mix->deemphasis ? 1 : 0;
+ return 0;
+}
+
+static int daca_put_deemphasis(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_daca *mix;
+ int change;
+
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ change = mix->deemphasis != ucontrol->value.integer.value[0];
+ if (change) {
+ mix->deemphasis = !!ucontrol->value.integer.value[0];
+ daca_set_volume(mix);
+ }
+ return change;
+}
+
+/* output volume */
+static int daca_info_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = DACA_VOL_MAX;
+ return 0;
+}
+
+static int daca_get_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_daca *mix;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ ucontrol->value.integer.value[0] = mix->left_vol;
+ ucontrol->value.integer.value[1] = mix->right_vol;
+ return 0;
+}
+
+static int daca_put_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_daca *mix;
+ unsigned int vol[2];
+ int change;
+
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ vol[0] = ucontrol->value.integer.value[0];
+ vol[1] = ucontrol->value.integer.value[1];
+ if (vol[0] > DACA_VOL_MAX || vol[1] > DACA_VOL_MAX)
+ return -EINVAL;
+ change = mix->left_vol != vol[0] ||
+ mix->right_vol != vol[1];
+ if (change) {
+ mix->left_vol = vol[0];
+ mix->right_vol = vol[1];
+ daca_set_volume(mix);
+ }
+ return change;
+}
+
+/* amplifier switch */
+#define daca_info_amp daca_info_deemphasis
+
+static int daca_get_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_daca *mix;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ ucontrol->value.integer.value[0] = mix->amp_on ? 1 : 0;
+ return 0;
+}
+
+static int daca_put_amp(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_daca *mix;
+ int change;
+
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ change = mix->amp_on != ucontrol->value.integer.value[0];
+ if (change) {
+ mix->amp_on = !!ucontrol->value.integer.value[0];
+ i2c_smbus_write_byte_data(mix->i2c.client, DACA_REG_GCFG,
+ mix->amp_on ? 0x05 : 0x04);
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new daca_mixers[] = {
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Deemphasis Switch",
+ .info = daca_info_deemphasis,
+ .get = daca_get_deemphasis,
+ .put = daca_put_deemphasis
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .info = daca_info_volume,
+ .get = daca_get_volume,
+ .put = daca_put_volume
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Power Amplifier Switch",
+ .info = daca_info_amp,
+ .get = daca_get_amp,
+ .put = daca_put_amp
+ },
+};
+
+
+#ifdef CONFIG_PM
+static void daca_resume(struct snd_pmac *chip)
+{
+ struct pmac_daca *mix = chip->mixer_data;
+ i2c_smbus_write_byte_data(mix->i2c.client, DACA_REG_SR, 0x08);
+ i2c_smbus_write_byte_data(mix->i2c.client, DACA_REG_GCFG,
+ mix->amp_on ? 0x05 : 0x04);
+ daca_set_volume(mix);
+}
+#endif /* CONFIG_PM */
+
+
+static void daca_cleanup(struct snd_pmac *chip)
+{
+ struct pmac_daca *mix = chip->mixer_data;
+ if (! mix)
+ return;
+ snd_pmac_keywest_cleanup(&mix->i2c);
+ kfree(mix);
+ chip->mixer_data = NULL;
+}
+
+/* exported */
+int __init snd_pmac_daca_init(struct snd_pmac *chip)
+{
+ int i, err;
+ struct pmac_daca *mix;
+
+ request_module("i2c-powermac");
+
+ mix = kzalloc(sizeof(*mix), GFP_KERNEL);
+ if (! mix)
+ return -ENOMEM;
+ chip->mixer_data = mix;
+ chip->mixer_free = daca_cleanup;
+ mix->amp_on = 1; /* default on */
+
+ mix->i2c.addr = DACA_I2C_ADDR;
+ mix->i2c.init_client = daca_init_client;
+ mix->i2c.name = "DACA";
+ if ((err = snd_pmac_keywest_init(&mix->i2c)) < 0)
+ return err;
+
+ /*
+ * build mixers
+ */
+ strcpy(chip->card->mixername, "PowerMac DACA");
+
+ for (i = 0; i < ARRAY_SIZE(daca_mixers); i++) {
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&daca_mixers[i], chip))) < 0)
+ return err;
+ }
+
+#ifdef CONFIG_PM
+ chip->resume = daca_resume;
+#endif
+
+ return 0;
+}
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
new file mode 100644
index 0000000..6ff99ed
--- /dev/null
+++ b/sound/ppc/keywest.c
@@ -0,0 +1,141 @@
+/*
+ * common keywest i2c layer
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include "pmac.h"
+
+/*
+ * we have to keep a static variable here since i2c attach_adapter
+ * callback cannot pass a private data.
+ */
+static struct pmac_keywest *keywest_ctx;
+
+
+static int keywest_attach_adapter(struct i2c_adapter *adapter);
+static int keywest_detach_client(struct i2c_client *client);
+
+struct i2c_driver keywest_driver = {
+ .driver = {
+ .name = "PMac Keywest Audio",
+ },
+ .attach_adapter = &keywest_attach_adapter,
+ .detach_client = &keywest_detach_client,
+};
+
+
+#ifndef i2c_device_name
+#define i2c_device_name(x) ((x)->name)
+#endif
+
+static int keywest_attach_adapter(struct i2c_adapter *adapter)
+{
+ int err;
+ struct i2c_client *new_client;
+
+ if (! keywest_ctx)
+ return -EINVAL;
+
+ if (strncmp(i2c_device_name(adapter), "mac-io", 6))
+ return 0; /* ignored */
+
+ new_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
+ if (! new_client)
+ return -ENOMEM;
+
+ new_client->addr = keywest_ctx->addr;
+ i2c_set_clientdata(new_client, keywest_ctx);
+ new_client->adapter = adapter;
+ new_client->driver = &keywest_driver;
+ new_client->flags = 0;
+
+ strcpy(i2c_device_name(new_client), keywest_ctx->name);
+ keywest_ctx->client = new_client;
+
+ /* Tell the i2c layer a new client has arrived */
+ if (i2c_attach_client(new_client)) {
+ snd_printk(KERN_ERR "tumbler: cannot attach i2c client\n");
+ err = -ENODEV;
+ goto __err;
+ }
+
+ return 0;
+
+ __err:
+ kfree(new_client);
+ keywest_ctx->client = NULL;
+ return err;
+}
+
+static int keywest_detach_client(struct i2c_client *client)
+{
+ if (! keywest_ctx)
+ return 0;
+ if (client == keywest_ctx->client)
+ keywest_ctx->client = NULL;
+
+ i2c_detach_client(client);
+ kfree(client);
+ return 0;
+}
+
+/* exported */
+void snd_pmac_keywest_cleanup(struct pmac_keywest *i2c)
+{
+ if (keywest_ctx && keywest_ctx == i2c) {
+ i2c_del_driver(&keywest_driver);
+ keywest_ctx = NULL;
+ }
+}
+
+int __init snd_pmac_tumbler_post_init(void)
+{
+ int err;
+
+ if (!keywest_ctx || !keywest_ctx->client)
+ return -ENXIO;
+
+ if ((err = keywest_ctx->init_client(keywest_ctx)) < 0) {
+ snd_printk(KERN_ERR "tumbler: %i :cannot initialize the MCS\n", err);
+ return err;
+ }
+ return 0;
+}
+
+/* exported */
+int __init snd_pmac_keywest_init(struct pmac_keywest *i2c)
+{
+ int err;
+
+ if (keywest_ctx)
+ return -EBUSY;
+
+ keywest_ctx = i2c;
+
+ if ((err = i2c_add_driver(&keywest_driver))) {
+ snd_printk(KERN_ERR "cannot register keywest i2c driver\n");
+ return err;
+ }
+ return 0;
+}
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
new file mode 100644
index 0000000..a38c0c7
--- /dev/null
+++ b/sound/ppc/pmac.c
@@ -0,0 +1,1412 @@
+/*
+ * PMac DBDMA lowlevel functions
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ * code based on dmasound.c.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <asm/io.h>
+#include <asm/irq.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include "pmac.h"
+#include <sound/pcm_params.h>
+#include <asm/pmac_feature.h>
+#include <asm/pci-bridge.h>
+
+
+/* fixed frequency table for awacs, screamer, burgundy, DACA (44100 max) */
+static int awacs_freqs[8] = {
+ 44100, 29400, 22050, 17640, 14700, 11025, 8820, 7350
+};
+/* fixed frequency table for tumbler */
+static int tumbler_freqs[1] = {
+ 44100
+};
+
+
+/*
+ * we will allocate a single 'emergency' dbdma cmd block to use if the
+ * tx status comes up "DEAD". This happens on some PowerComputing Pmac
+ * clones, either owing to a bug in dbdma or some interaction between
+ * IDE and sound. However, this measure would deal with DEAD status if
+ * it appeared elsewhere.
+ */
+static struct pmac_dbdma emergency_dbdma;
+static int emergency_in_use;
+
+
+/*
+ * allocate DBDMA command arrays
+ */
+static int snd_pmac_dbdma_alloc(struct snd_pmac *chip, struct pmac_dbdma *rec, int size)
+{
+ unsigned int rsize = sizeof(struct dbdma_cmd) * (size + 1);
+
+ rec->space = dma_alloc_coherent(&chip->pdev->dev, rsize,
+ &rec->dma_base, GFP_KERNEL);
+ if (rec->space == NULL)
+ return -ENOMEM;
+ rec->size = size;
+ memset(rec->space, 0, rsize);
+ rec->cmds = (void __iomem *)DBDMA_ALIGN(rec->space);
+ rec->addr = rec->dma_base + (unsigned long)((char *)rec->cmds - (char *)rec->space);
+
+ return 0;
+}
+
+static void snd_pmac_dbdma_free(struct snd_pmac *chip, struct pmac_dbdma *rec)
+{
+ if (rec->space) {
+ unsigned int rsize = sizeof(struct dbdma_cmd) * (rec->size + 1);
+
+ dma_free_coherent(&chip->pdev->dev, rsize, rec->space, rec->dma_base);
+ }
+}
+
+
+/*
+ * pcm stuff
+ */
+
+/*
+ * look up frequency table
+ */
+
+unsigned int snd_pmac_rate_index(struct snd_pmac *chip, struct pmac_stream *rec, unsigned int rate)
+{
+ int i, ok, found;
+
+ ok = rec->cur_freqs;
+ if (rate > chip->freq_table[0])
+ return 0;
+ found = 0;
+ for (i = 0; i < chip->num_freqs; i++, ok >>= 1) {
+ if (! (ok & 1)) continue;
+ found = i;
+ if (rate >= chip->freq_table[i])
+ break;
+ }
+ return found;
+}
+
+/*
+ * check whether another stream is active
+ */
+static inline int another_stream(int stream)
+{
+ return (stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+}
+
+/*
+ * allocate buffers
+ */
+static int snd_pmac_pcm_hw_params(struct snd_pcm_substream *subs,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw_params));
+}
+
+/*
+ * release buffers
+ */
+static int snd_pmac_pcm_hw_free(struct snd_pcm_substream *subs)
+{
+ snd_pcm_lib_free_pages(subs);
+ return 0;
+}
+
+/*
+ * get a stream of the opposite direction
+ */
+static struct pmac_stream *snd_pmac_get_stream(struct snd_pmac *chip, int stream)
+{
+ switch (stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ return &chip->playback;
+ case SNDRV_PCM_STREAM_CAPTURE:
+ return &chip->capture;
+ default:
+ snd_BUG();
+ return NULL;
+ }
+}
+
+/*
+ * wait while run status is on
+ */
+static inline void
+snd_pmac_wait_ack(struct pmac_stream *rec)
+{
+ int timeout = 50000;
+ while ((in_le32(&rec->dma->status) & RUN) && timeout-- > 0)
+ udelay(1);
+}
+
+/*
+ * set the format and rate to the chip.
+ * call the lowlevel function if defined (e.g. for AWACS).
+ */
+static void snd_pmac_pcm_set_format(struct snd_pmac *chip)
+{
+ /* set up frequency and format */
+ out_le32(&chip->awacs->control, chip->control_mask | (chip->rate_index << 8));
+ out_le32(&chip->awacs->byteswap, chip->format == SNDRV_PCM_FORMAT_S16_LE ? 1 : 0);
+ if (chip->set_format)
+ chip->set_format(chip);
+}
+
+/*
+ * stop the DMA transfer
+ */
+static inline void snd_pmac_dma_stop(struct pmac_stream *rec)
+{
+ out_le32(&rec->dma->control, (RUN|WAKE|FLUSH|PAUSE) << 16);
+ snd_pmac_wait_ack(rec);
+}
+
+/*
+ * set the command pointer address
+ */
+static inline void snd_pmac_dma_set_command(struct pmac_stream *rec, struct pmac_dbdma *cmd)
+{
+ out_le32(&rec->dma->cmdptr, cmd->addr);
+}
+
+/*
+ * start the DMA
+ */
+static inline void snd_pmac_dma_run(struct pmac_stream *rec, int status)
+{
+ out_le32(&rec->dma->control, status | (status << 16));
+}
+
+
+/*
+ * prepare playback/capture stream
+ */
+static int snd_pmac_pcm_prepare(struct snd_pmac *chip, struct pmac_stream *rec, struct snd_pcm_substream *subs)
+{
+ int i;
+ volatile struct dbdma_cmd __iomem *cp;
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ int rate_index;
+ long offset;
+ struct pmac_stream *astr;
+
+ rec->dma_size = snd_pcm_lib_buffer_bytes(subs);
+ rec->period_size = snd_pcm_lib_period_bytes(subs);
+ rec->nperiods = rec->dma_size / rec->period_size;
+ rec->cur_period = 0;
+ rate_index = snd_pmac_rate_index(chip, rec, runtime->rate);
+
+ /* set up constraints */
+ astr = snd_pmac_get_stream(chip, another_stream(rec->stream));
+ if (! astr)
+ return -EINVAL;
+ astr->cur_freqs = 1 << rate_index;
+ astr->cur_formats = 1 << runtime->format;
+ chip->rate_index = rate_index;
+ chip->format = runtime->format;
+
+ /* We really want to execute a DMA stop command, after the AWACS
+ * is initialized.
+ * For reasons I don't understand, it stops the hissing noise
+ * common to many PowerBook G3 systems and random noise otherwise
+ * captured on iBook2's about every third time. -ReneR
+ */
+ spin_lock_irq(&chip->reg_lock);
+ snd_pmac_dma_stop(rec);
+ st_le16(&chip->extra_dma.cmds->command, DBDMA_STOP);
+ snd_pmac_dma_set_command(rec, &chip->extra_dma);
+ snd_pmac_dma_run(rec, RUN);
+ spin_unlock_irq(&chip->reg_lock);
+ mdelay(5);
+ spin_lock_irq(&chip->reg_lock);
+ /* continuous DMA memory type doesn't provide the physical address,
+ * so we need to resolve the address here...
+ */
+ offset = runtime->dma_addr;
+ for (i = 0, cp = rec->cmd.cmds; i < rec->nperiods; i++, cp++) {
+ st_le32(&cp->phy_addr, offset);
+ st_le16(&cp->req_count, rec->period_size);
+ /*st_le16(&cp->res_count, 0);*/
+ st_le16(&cp->xfer_status, 0);
+ offset += rec->period_size;
+ }
+ /* make loop */
+ st_le16(&cp->command, DBDMA_NOP + BR_ALWAYS);
+ st_le32(&cp->cmd_dep, rec->cmd.addr);
+
+ snd_pmac_dma_stop(rec);
+ snd_pmac_dma_set_command(rec, &rec->cmd);
+ spin_unlock_irq(&chip->reg_lock);
+
+ return 0;
+}
+
+
+/*
+ * PCM trigger/stop
+ */
+static int snd_pmac_pcm_trigger(struct snd_pmac *chip, struct pmac_stream *rec,
+ struct snd_pcm_substream *subs, int cmd)
+{
+ volatile struct dbdma_cmd __iomem *cp;
+ int i, command;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (rec->running)
+ return -EBUSY;
+ command = (subs->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ OUTPUT_MORE : INPUT_MORE) + INTR_ALWAYS;
+ spin_lock(&chip->reg_lock);
+ snd_pmac_beep_stop(chip);
+ snd_pmac_pcm_set_format(chip);
+ for (i = 0, cp = rec->cmd.cmds; i < rec->nperiods; i++, cp++)
+ out_le16(&cp->command, command);
+ snd_pmac_dma_set_command(rec, &rec->cmd);
+ (void)in_le32(&rec->dma->status);
+ snd_pmac_dma_run(rec, RUN|WAKE);
+ rec->running = 1;
+ spin_unlock(&chip->reg_lock);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ spin_lock(&chip->reg_lock);
+ rec->running = 0;
+ /*printk("stopped!!\n");*/
+ snd_pmac_dma_stop(rec);
+ for (i = 0, cp = rec->cmd.cmds; i < rec->nperiods; i++, cp++)
+ out_le16(&cp->command, DBDMA_STOP);
+ spin_unlock(&chip->reg_lock);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * return the current pointer
+ */
+inline
+static snd_pcm_uframes_t snd_pmac_pcm_pointer(struct snd_pmac *chip,
+ struct pmac_stream *rec,
+ struct snd_pcm_substream *subs)
+{
+ int count = 0;
+
+#if 1 /* hmm.. how can we get the current dma pointer?? */
+ int stat;
+ volatile struct dbdma_cmd __iomem *cp = &rec->cmd.cmds[rec->cur_period];
+ stat = ld_le16(&cp->xfer_status);
+ if (stat & (ACTIVE|DEAD)) {
+ count = in_le16(&cp->res_count);
+ if (count)
+ count = rec->period_size - count;
+ }
+#endif
+ count += rec->cur_period * rec->period_size;
+ /*printk("pointer=%d\n", count);*/
+ return bytes_to_frames(subs->runtime, count);
+}
+
+/*
+ * playback
+ */
+
+static int snd_pmac_playback_prepare(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+ return snd_pmac_pcm_prepare(chip, &chip->playback, subs);
+}
+
+static int snd_pmac_playback_trigger(struct snd_pcm_substream *subs,
+ int cmd)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+ return snd_pmac_pcm_trigger(chip, &chip->playback, subs, cmd);
+}
+
+static snd_pcm_uframes_t snd_pmac_playback_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+ return snd_pmac_pcm_pointer(chip, &chip->playback, subs);
+}
+
+
+/*
+ * capture
+ */
+
+static int snd_pmac_capture_prepare(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+ return snd_pmac_pcm_prepare(chip, &chip->capture, subs);
+}
+
+static int snd_pmac_capture_trigger(struct snd_pcm_substream *subs,
+ int cmd)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+ return snd_pmac_pcm_trigger(chip, &chip->capture, subs, cmd);
+}
+
+static snd_pcm_uframes_t snd_pmac_capture_pointer(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+ return snd_pmac_pcm_pointer(chip, &chip->capture, subs);
+}
+
+
+/*
+ * Handle DEAD DMA transfers:
+ * if the TX status comes up "DEAD" - reported on some Power Computing machines
+ * we need to re-start the dbdma - but from a different physical start address
+ * and with a different transfer length. It would get very messy to do this
+ * with the normal dbdma_cmd blocks - we would have to re-write the buffer start
+ * addresses each time. So, we will keep a single dbdma_cmd block which can be
+ * fiddled with.
+ * When DEAD status is first reported the content of the faulted dbdma block is
+ * copied into the emergency buffer and we note that the buffer is in use.
+ * we then bump the start physical address by the amount that was successfully
+ * output before it died.
+ * On any subsequent DEAD result we just do the bump-ups (we know that we are
+ * already using the emergency dbdma_cmd).
+ * CHECK: this just tries to "do it". It is possible that we should abandon
+ * xfers when the number of residual bytes gets below a certain value - I can
+ * see that this might cause a loop-forever if a too small transfer causes
+ * DEAD status. However this is a TODO for now - we'll see what gets reported.
+ * When we get a successful transfer result with the emergency buffer we just
+ * pretend that it completed using the original dmdma_cmd and carry on. The
+ * 'next_cmd' field will already point back to the original loop of blocks.
+ */
+static inline void snd_pmac_pcm_dead_xfer(struct pmac_stream *rec,
+ volatile struct dbdma_cmd __iomem *cp)
+{
+ unsigned short req, res ;
+ unsigned int phy ;
+
+ /* printk(KERN_WARNING "snd-powermac: DMA died - patching it up!\n"); */
+
+ /* to clear DEAD status we must first clear RUN
+ set it to quiescent to be on the safe side */
+ (void)in_le32(&rec->dma->status);
+ out_le32(&rec->dma->control, (RUN|PAUSE|FLUSH|WAKE) << 16);
+
+ if (!emergency_in_use) { /* new problem */
+ memcpy((void *)emergency_dbdma.cmds, (void *)cp,
+ sizeof(struct dbdma_cmd));
+ emergency_in_use = 1;
+ st_le16(&cp->xfer_status, 0);
+ st_le16(&cp->req_count, rec->period_size);
+ cp = emergency_dbdma.cmds;
+ }
+
+ /* now bump the values to reflect the amount
+ we haven't yet shifted */
+ req = ld_le16(&cp->req_count);
+ res = ld_le16(&cp->res_count);
+ phy = ld_le32(&cp->phy_addr);
+ phy += (req - res);
+ st_le16(&cp->req_count, res);
+ st_le16(&cp->res_count, 0);
+ st_le16(&cp->xfer_status, 0);
+ st_le32(&cp->phy_addr, phy);
+
+ st_le32(&cp->cmd_dep, rec->cmd.addr
+ + sizeof(struct dbdma_cmd)*((rec->cur_period+1)%rec->nperiods));
+
+ st_le16(&cp->command, OUTPUT_MORE | BR_ALWAYS | INTR_ALWAYS);
+
+ /* point at our patched up command block */
+ out_le32(&rec->dma->cmdptr, emergency_dbdma.addr);
+
+ /* we must re-start the controller */
+ (void)in_le32(&rec->dma->status);
+ /* should complete clearing the DEAD status */
+ out_le32(&rec->dma->control, ((RUN|WAKE) << 16) + (RUN|WAKE));
+}
+
+/*
+ * update playback/capture pointer from interrupts
+ */
+static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec)
+{
+ volatile struct dbdma_cmd __iomem *cp;
+ int c;
+ int stat;
+
+ spin_lock(&chip->reg_lock);
+ if (rec->running) {
+ for (c = 0; c < rec->nperiods; c++) { /* at most all fragments */
+
+ if (emergency_in_use) /* already using DEAD xfer? */
+ cp = emergency_dbdma.cmds;
+ else
+ cp = &rec->cmd.cmds[rec->cur_period];
+
+ stat = ld_le16(&cp->xfer_status);
+
+ if (stat & DEAD) {
+ snd_pmac_pcm_dead_xfer(rec, cp);
+ break; /* this block is still going */
+ }
+
+ if (emergency_in_use)
+ emergency_in_use = 0 ; /* done that */
+
+ if (! (stat & ACTIVE))
+ break;
+
+ /*printk("update frag %d\n", rec->cur_period);*/
+ st_le16(&cp->xfer_status, 0);
+ st_le16(&cp->req_count, rec->period_size);
+ /*st_le16(&cp->res_count, 0);*/
+ rec->cur_period++;
+ if (rec->cur_period >= rec->nperiods) {
+ rec->cur_period = 0;
+ }
+
+ spin_unlock(&chip->reg_lock);
+ snd_pcm_period_elapsed(rec->substream);
+ spin_lock(&chip->reg_lock);
+ }
+ }
+ spin_unlock(&chip->reg_lock);
+}
+
+
+/*
+ * hw info
+ */
+
+static struct snd_pcm_hardware snd_pmac_playback =
+{
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_44100,
+ .rate_min = 7350,
+ .rate_max = 44100,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 131072,
+ .period_bytes_min = 256,
+ .period_bytes_max = 16384,
+ .periods_min = 3,
+ .periods_max = PMAC_MAX_FRAGS,
+};
+
+static struct snd_pcm_hardware snd_pmac_capture =
+{
+ .info = (SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_44100,
+ .rate_min = 7350,
+ .rate_max = 44100,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 131072,
+ .period_bytes_min = 256,
+ .period_bytes_max = 16384,
+ .periods_min = 3,
+ .periods_max = PMAC_MAX_FRAGS,
+};
+
+
+#if 0 // NYI
+static int snd_pmac_hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pmac *chip = rule->private;
+ struct pmac_stream *rec = snd_pmac_get_stream(chip, rule->deps[0]);
+ int i, freq_table[8], num_freqs;
+
+ if (! rec)
+ return -EINVAL;
+ num_freqs = 0;
+ for (i = chip->num_freqs - 1; i >= 0; i--) {
+ if (rec->cur_freqs & (1 << i))
+ freq_table[num_freqs++] = chip->freq_table[i];
+ }
+
+ return snd_interval_list(hw_param_interval(params, rule->var),
+ num_freqs, freq_table, 0);
+}
+
+static int snd_pmac_hw_rule_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pmac *chip = rule->private;
+ struct pmac_stream *rec = snd_pmac_get_stream(chip, rule->deps[0]);
+
+ if (! rec)
+ return -EINVAL;
+ return snd_mask_refine_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT),
+ rec->cur_formats);
+}
+#endif // NYI
+
+static int snd_pmac_pcm_open(struct snd_pmac *chip, struct pmac_stream *rec,
+ struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ int i;
+
+ /* look up frequency table and fill bit mask */
+ runtime->hw.rates = 0;
+ for (i = 0; i < chip->num_freqs; i++)
+ if (chip->freqs_ok & (1 << i))
+ runtime->hw.rates |=
+ snd_pcm_rate_to_rate_bit(chip->freq_table[i]);
+
+ /* check for minimum and maximum rates */
+ for (i = 0; i < chip->num_freqs; i++) {
+ if (chip->freqs_ok & (1 << i)) {
+ runtime->hw.rate_max = chip->freq_table[i];
+ break;
+ }
+ }
+ for (i = chip->num_freqs - 1; i >= 0; i--) {
+ if (chip->freqs_ok & (1 << i)) {
+ runtime->hw.rate_min = chip->freq_table[i];
+ break;
+ }
+ }
+ runtime->hw.formats = chip->formats_ok;
+ if (chip->can_capture) {
+ if (! chip->can_duplex)
+ runtime->hw.info |= SNDRV_PCM_INFO_HALF_DUPLEX;
+ runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX;
+ }
+ runtime->private_data = rec;
+ rec->substream = subs;
+
+#if 0 /* FIXME: still under development.. */
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_pmac_hw_rule_rate, chip, rec->stream, -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ snd_pmac_hw_rule_format, chip, rec->stream, -1);
+#endif
+
+ runtime->hw.periods_max = rec->cmd.size - 1;
+
+ /* constraints to fix choppy sound */
+ snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ return 0;
+}
+
+static int snd_pmac_pcm_close(struct snd_pmac *chip, struct pmac_stream *rec,
+ struct snd_pcm_substream *subs)
+{
+ struct pmac_stream *astr;
+
+ snd_pmac_dma_stop(rec);
+
+ astr = snd_pmac_get_stream(chip, another_stream(rec->stream));
+ if (! astr)
+ return -EINVAL;
+
+ /* reset constraints */
+ astr->cur_freqs = chip->freqs_ok;
+ astr->cur_formats = chip->formats_ok;
+
+ return 0;
+}
+
+static int snd_pmac_playback_open(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+
+ subs->runtime->hw = snd_pmac_playback;
+ return snd_pmac_pcm_open(chip, &chip->playback, subs);
+}
+
+static int snd_pmac_capture_open(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+
+ subs->runtime->hw = snd_pmac_capture;
+ return snd_pmac_pcm_open(chip, &chip->capture, subs);
+}
+
+static int snd_pmac_playback_close(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+
+ return snd_pmac_pcm_close(chip, &chip->playback, subs);
+}
+
+static int snd_pmac_capture_close(struct snd_pcm_substream *subs)
+{
+ struct snd_pmac *chip = snd_pcm_substream_chip(subs);
+
+ return snd_pmac_pcm_close(chip, &chip->capture, subs);
+}
+
+/*
+ */
+
+static struct snd_pcm_ops snd_pmac_playback_ops = {
+ .open = snd_pmac_playback_open,
+ .close = snd_pmac_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_pmac_pcm_hw_params,
+ .hw_free = snd_pmac_pcm_hw_free,
+ .prepare = snd_pmac_playback_prepare,
+ .trigger = snd_pmac_playback_trigger,
+ .pointer = snd_pmac_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_pmac_capture_ops = {
+ .open = snd_pmac_capture_open,
+ .close = snd_pmac_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_pmac_pcm_hw_params,
+ .hw_free = snd_pmac_pcm_hw_free,
+ .prepare = snd_pmac_capture_prepare,
+ .trigger = snd_pmac_capture_trigger,
+ .pointer = snd_pmac_capture_pointer,
+};
+
+int __init snd_pmac_pcm_new(struct snd_pmac *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+ int num_captures = 1;
+
+ if (! chip->can_capture)
+ num_captures = 0;
+ err = snd_pcm_new(chip->card, chip->card->driver, 0, 1, num_captures, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_pmac_playback_ops);
+ if (chip->can_capture)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_pmac_capture_ops);
+
+ pcm->private_data = chip;
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+ strcpy(pcm->name, chip->card->shortname);
+ chip->pcm = pcm;
+
+ chip->formats_ok = SNDRV_PCM_FMTBIT_S16_BE;
+ if (chip->can_byte_swap)
+ chip->formats_ok |= SNDRV_PCM_FMTBIT_S16_LE;
+
+ chip->playback.cur_formats = chip->formats_ok;
+ chip->capture.cur_formats = chip->formats_ok;
+ chip->playback.cur_freqs = chip->freqs_ok;
+ chip->capture.cur_freqs = chip->freqs_ok;
+
+ /* preallocate 64k buffer */
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ &chip->pdev->dev,
+ 64 * 1024, 64 * 1024);
+
+ return 0;
+}
+
+
+static void snd_pmac_dbdma_reset(struct snd_pmac *chip)
+{
+ out_le32(&chip->playback.dma->control, (RUN|PAUSE|FLUSH|WAKE|DEAD) << 16);
+ snd_pmac_wait_ack(&chip->playback);
+ out_le32(&chip->capture.dma->control, (RUN|PAUSE|FLUSH|WAKE|DEAD) << 16);
+ snd_pmac_wait_ack(&chip->capture);
+}
+
+
+/*
+ * handling beep
+ */
+void snd_pmac_beep_dma_start(struct snd_pmac *chip, int bytes, unsigned long addr, int speed)
+{
+ struct pmac_stream *rec = &chip->playback;
+
+ snd_pmac_dma_stop(rec);
+ st_le16(&chip->extra_dma.cmds->req_count, bytes);
+ st_le16(&chip->extra_dma.cmds->xfer_status, 0);
+ st_le32(&chip->extra_dma.cmds->cmd_dep, chip->extra_dma.addr);
+ st_le32(&chip->extra_dma.cmds->phy_addr, addr);
+ st_le16(&chip->extra_dma.cmds->command, OUTPUT_MORE + BR_ALWAYS);
+ out_le32(&chip->awacs->control,
+ (in_le32(&chip->awacs->control) & ~0x1f00)
+ | (speed << 8));
+ out_le32(&chip->awacs->byteswap, 0);
+ snd_pmac_dma_set_command(rec, &chip->extra_dma);
+ snd_pmac_dma_run(rec, RUN);
+}
+
+void snd_pmac_beep_dma_stop(struct snd_pmac *chip)
+{
+ snd_pmac_dma_stop(&chip->playback);
+ st_le16(&chip->extra_dma.cmds->command, DBDMA_STOP);
+ snd_pmac_pcm_set_format(chip); /* reset format */
+}
+
+
+/*
+ * interrupt handlers
+ */
+static irqreturn_t
+snd_pmac_tx_intr(int irq, void *devid)
+{
+ struct snd_pmac *chip = devid;
+ snd_pmac_pcm_update(chip, &chip->playback);
+ return IRQ_HANDLED;
+}
+
+
+static irqreturn_t
+snd_pmac_rx_intr(int irq, void *devid)
+{
+ struct snd_pmac *chip = devid;
+ snd_pmac_pcm_update(chip, &chip->capture);
+ return IRQ_HANDLED;
+}
+
+
+static irqreturn_t
+snd_pmac_ctrl_intr(int irq, void *devid)
+{
+ struct snd_pmac *chip = devid;
+ int ctrl = in_le32(&chip->awacs->control);
+
+ /*printk("pmac: control interrupt.. 0x%x\n", ctrl);*/
+ if (ctrl & MASK_PORTCHG) {
+ /* do something when headphone is plugged/unplugged? */
+ if (chip->update_automute)
+ chip->update_automute(chip, 1);
+ }
+ if (ctrl & MASK_CNTLERR) {
+ int err = (in_le32(&chip->awacs->codec_stat) & MASK_ERRCODE) >> 16;
+ if (err && chip->model <= PMAC_SCREAMER)
+ snd_printk(KERN_DEBUG "error %x\n", err);
+ }
+ /* Writing 1s to the CNTLERR and PORTCHG bits clears them... */
+ out_le32(&chip->awacs->control, ctrl);
+ return IRQ_HANDLED;
+}
+
+
+/*
+ * a wrapper to feature call for compatibility
+ */
+static void snd_pmac_sound_feature(struct snd_pmac *chip, int enable)
+{
+ if (ppc_md.feature_call)
+ ppc_md.feature_call(PMAC_FTR_SOUND_CHIP_ENABLE, chip->node, 0, enable);
+}
+
+/*
+ * release resources
+ */
+
+static int snd_pmac_free(struct snd_pmac *chip)
+{
+ /* stop sounds */
+ if (chip->initialized) {
+ snd_pmac_dbdma_reset(chip);
+ /* disable interrupts from awacs interface */
+ out_le32(&chip->awacs->control, in_le32(&chip->awacs->control) & 0xfff);
+ }
+
+ if (chip->node)
+ snd_pmac_sound_feature(chip, 0);
+
+ /* clean up mixer if any */
+ if (chip->mixer_free)
+ chip->mixer_free(chip);
+
+ snd_pmac_detach_beep(chip);
+
+ /* release resources */
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void*)chip);
+ if (chip->tx_irq >= 0)
+ free_irq(chip->tx_irq, (void*)chip);
+ if (chip->rx_irq >= 0)
+ free_irq(chip->rx_irq, (void*)chip);
+ snd_pmac_dbdma_free(chip, &chip->playback.cmd);
+ snd_pmac_dbdma_free(chip, &chip->capture.cmd);
+ snd_pmac_dbdma_free(chip, &chip->extra_dma);
+ snd_pmac_dbdma_free(chip, &emergency_dbdma);
+ if (chip->macio_base)
+ iounmap(chip->macio_base);
+ if (chip->latch_base)
+ iounmap(chip->latch_base);
+ if (chip->awacs)
+ iounmap(chip->awacs);
+ if (chip->playback.dma)
+ iounmap(chip->playback.dma);
+ if (chip->capture.dma)
+ iounmap(chip->capture.dma);
+
+ if (chip->node) {
+ int i;
+ for (i = 0; i < 3; i++) {
+ if (chip->requested & (1 << i))
+ release_mem_region(chip->rsrc[i].start,
+ chip->rsrc[i].end -
+ chip->rsrc[i].start + 1);
+ }
+ }
+
+ if (chip->pdev)
+ pci_dev_put(chip->pdev);
+ of_node_put(chip->node);
+ kfree(chip);
+ return 0;
+}
+
+
+/*
+ * free the device
+ */
+static int snd_pmac_dev_free(struct snd_device *device)
+{
+ struct snd_pmac *chip = device->device_data;
+ return snd_pmac_free(chip);
+}
+
+
+/*
+ * check the machine support byteswap (little-endian)
+ */
+
+static void __init detect_byte_swap(struct snd_pmac *chip)
+{
+ struct device_node *mio;
+
+ /* if seems that Keylargo can't byte-swap */
+ for (mio = chip->node->parent; mio; mio = mio->parent) {
+ if (strcmp(mio->name, "mac-io") == 0) {
+ if (of_device_is_compatible(mio, "Keylargo"))
+ chip->can_byte_swap = 0;
+ break;
+ }
+ }
+
+ /* it seems the Pismo & iBook can't byte-swap in hardware. */
+ if (machine_is_compatible("PowerBook3,1") ||
+ machine_is_compatible("PowerBook2,1"))
+ chip->can_byte_swap = 0 ;
+
+ if (machine_is_compatible("PowerBook2,1"))
+ chip->can_duplex = 0;
+}
+
+
+/*
+ * detect a sound chip
+ */
+static int __init snd_pmac_detect(struct snd_pmac *chip)
+{
+ struct device_node *sound;
+ struct device_node *dn;
+ const unsigned int *prop;
+ unsigned int l;
+ struct macio_chip* macio;
+
+ if (!machine_is(powermac))
+ return -ENODEV;
+
+ chip->subframe = 0;
+ chip->revision = 0;
+ chip->freqs_ok = 0xff; /* all ok */
+ chip->model = PMAC_AWACS;
+ chip->can_byte_swap = 1;
+ chip->can_duplex = 1;
+ chip->can_capture = 1;
+ chip->num_freqs = ARRAY_SIZE(awacs_freqs);
+ chip->freq_table = awacs_freqs;
+ chip->pdev = NULL;
+
+ chip->control_mask = MASK_IEPC | MASK_IEE | 0x11; /* default */
+
+ /* check machine type */
+ if (machine_is_compatible("AAPL,3400/2400")
+ || machine_is_compatible("AAPL,3500"))
+ chip->is_pbook_3400 = 1;
+ else if (machine_is_compatible("PowerBook1,1")
+ || machine_is_compatible("AAPL,PowerBook1998"))
+ chip->is_pbook_G3 = 1;
+ chip->node = of_find_node_by_name(NULL, "awacs");
+ sound = of_node_get(chip->node);
+
+ /*
+ * powermac G3 models have a node called "davbus"
+ * with a child called "sound".
+ */
+ if (!chip->node)
+ chip->node = of_find_node_by_name(NULL, "davbus");
+ /*
+ * if we didn't find a davbus device, try 'i2s-a' since
+ * this seems to be what iBooks have
+ */
+ if (! chip->node) {
+ chip->node = of_find_node_by_name(NULL, "i2s-a");
+ if (chip->node && chip->node->parent &&
+ chip->node->parent->parent) {
+ if (of_device_is_compatible(chip->node->parent->parent,
+ "K2-Keylargo"))
+ chip->is_k2 = 1;
+ }
+ }
+ if (! chip->node)
+ return -ENODEV;
+
+ if (!sound) {
+ sound = of_find_node_by_name(NULL, "sound");
+ while (sound && sound->parent != chip->node)
+ sound = of_find_node_by_name(sound, "sound");
+ }
+ if (! sound) {
+ of_node_put(chip->node);
+ chip->node = NULL;
+ return -ENODEV;
+ }
+ prop = of_get_property(sound, "sub-frame", NULL);
+ if (prop && *prop < 16)
+ chip->subframe = *prop;
+ prop = of_get_property(sound, "layout-id", NULL);
+ if (prop) {
+ /* partly deprecate snd-powermac, for those machines
+ * that have a layout-id property for now */
+ printk(KERN_INFO "snd-powermac no longer handles any "
+ "machines with a layout-id property "
+ "in the device-tree, use snd-aoa.\n");
+ of_node_put(sound);
+ of_node_put(chip->node);
+ chip->node = NULL;
+ return -ENODEV;
+ }
+ /* This should be verified on older screamers */
+ if (of_device_is_compatible(sound, "screamer")) {
+ chip->model = PMAC_SCREAMER;
+ // chip->can_byte_swap = 0; /* FIXME: check this */
+ }
+ if (of_device_is_compatible(sound, "burgundy")) {
+ chip->model = PMAC_BURGUNDY;
+ chip->control_mask = MASK_IEPC | 0x11; /* disable IEE */
+ }
+ if (of_device_is_compatible(sound, "daca")) {
+ chip->model = PMAC_DACA;
+ chip->can_capture = 0; /* no capture */
+ chip->can_duplex = 0;
+ // chip->can_byte_swap = 0; /* FIXME: check this */
+ chip->control_mask = MASK_IEPC | 0x11; /* disable IEE */
+ }
+ if (of_device_is_compatible(sound, "tumbler")) {
+ chip->model = PMAC_TUMBLER;
+ chip->can_capture = 0; /* no capture */
+ chip->can_duplex = 0;
+ // chip->can_byte_swap = 0; /* FIXME: check this */
+ chip->num_freqs = ARRAY_SIZE(tumbler_freqs);
+ chip->freq_table = tumbler_freqs;
+ chip->control_mask = MASK_IEPC | 0x11; /* disable IEE */
+ }
+ if (of_device_is_compatible(sound, "snapper")) {
+ chip->model = PMAC_SNAPPER;
+ // chip->can_byte_swap = 0; /* FIXME: check this */
+ chip->num_freqs = ARRAY_SIZE(tumbler_freqs);
+ chip->freq_table = tumbler_freqs;
+ chip->control_mask = MASK_IEPC | 0x11; /* disable IEE */
+ }
+ prop = of_get_property(sound, "device-id", NULL);
+ if (prop)
+ chip->device_id = *prop;
+ dn = of_find_node_by_name(NULL, "perch");
+ chip->has_iic = (dn != NULL);
+ of_node_put(dn);
+
+ /* We need the PCI device for DMA allocations, let's use a crude method
+ * for now ...
+ */
+ macio = macio_find(chip->node, macio_unknown);
+ if (macio == NULL)
+ printk(KERN_WARNING "snd-powermac: can't locate macio !\n");
+ else {
+ struct pci_dev *pdev = NULL;
+
+ for_each_pci_dev(pdev) {
+ struct device_node *np = pci_device_to_OF_node(pdev);
+ if (np && np == macio->of_node) {
+ chip->pdev = pdev;
+ break;
+ }
+ }
+ }
+ if (chip->pdev == NULL)
+ printk(KERN_WARNING "snd-powermac: can't locate macio PCI"
+ " device !\n");
+
+ detect_byte_swap(chip);
+
+ /* look for a property saying what sample rates
+ are available */
+ prop = of_get_property(sound, "sample-rates", &l);
+ if (! prop)
+ prop = of_get_property(sound, "output-frame-rates", &l);
+ if (prop) {
+ int i;
+ chip->freqs_ok = 0;
+ for (l /= sizeof(int); l > 0; --l) {
+ unsigned int r = *prop++;
+ /* Apple 'Fixed' format */
+ if (r >= 0x10000)
+ r >>= 16;
+ for (i = 0; i < chip->num_freqs; ++i) {
+ if (r == chip->freq_table[i]) {
+ chip->freqs_ok |= (1 << i);
+ break;
+ }
+ }
+ }
+ } else {
+ /* assume only 44.1khz */
+ chip->freqs_ok = 1;
+ }
+
+ of_node_put(sound);
+ return 0;
+}
+
+#ifdef PMAC_SUPPORT_AUTOMUTE
+/*
+ * auto-mute
+ */
+static int pmac_auto_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = chip->auto_mute;
+ return 0;
+}
+
+static int pmac_auto_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ if (ucontrol->value.integer.value[0] != chip->auto_mute) {
+ chip->auto_mute = !!ucontrol->value.integer.value[0];
+ if (chip->update_automute)
+ chip->update_automute(chip, 1);
+ return 1;
+ }
+ return 0;
+}
+
+static int pmac_hp_detect_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ if (chip->detect_headphone)
+ ucontrol->value.integer.value[0] = chip->detect_headphone(chip);
+ else
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static struct snd_kcontrol_new auto_mute_controls[] __initdata = {
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Auto Mute Switch",
+ .info = snd_pmac_boolean_mono_info,
+ .get = pmac_auto_mute_get,
+ .put = pmac_auto_mute_put,
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Detection",
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .info = snd_pmac_boolean_mono_info,
+ .get = pmac_hp_detect_get,
+ },
+};
+
+int __init snd_pmac_add_automute(struct snd_pmac *chip)
+{
+ int err;
+ chip->auto_mute = 1;
+ err = snd_ctl_add(chip->card, snd_ctl_new1(&auto_mute_controls[0], chip));
+ if (err < 0) {
+ printk(KERN_ERR "snd-powermac: Failed to add automute control\n");
+ return err;
+ }
+ chip->hp_detect_ctl = snd_ctl_new1(&auto_mute_controls[1], chip);
+ return snd_ctl_add(chip->card, chip->hp_detect_ctl);
+}
+#endif /* PMAC_SUPPORT_AUTOMUTE */
+
+/*
+ * create and detect a pmac chip record
+ */
+int __init snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
+{
+ struct snd_pmac *chip;
+ struct device_node *np;
+ int i, err;
+ unsigned int irq;
+ unsigned long ctrl_addr, txdma_addr, rxdma_addr;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_pmac_dev_free,
+ };
+
+ *chip_return = NULL;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+ chip->card = card;
+
+ spin_lock_init(&chip->reg_lock);
+ chip->irq = chip->tx_irq = chip->rx_irq = -1;
+
+ chip->playback.stream = SNDRV_PCM_STREAM_PLAYBACK;
+ chip->capture.stream = SNDRV_PCM_STREAM_CAPTURE;
+
+ if ((err = snd_pmac_detect(chip)) < 0)
+ goto __error;
+
+ if (snd_pmac_dbdma_alloc(chip, &chip->playback.cmd, PMAC_MAX_FRAGS + 1) < 0 ||
+ snd_pmac_dbdma_alloc(chip, &chip->capture.cmd, PMAC_MAX_FRAGS + 1) < 0 ||
+ snd_pmac_dbdma_alloc(chip, &chip->extra_dma, 2) < 0 ||
+ snd_pmac_dbdma_alloc(chip, &emergency_dbdma, 2) < 0) {
+ err = -ENOMEM;
+ goto __error;
+ }
+
+ np = chip->node;
+ chip->requested = 0;
+ if (chip->is_k2) {
+ static char *rnames[] = {
+ "Sound Control", "Sound DMA" };
+ for (i = 0; i < 2; i ++) {
+ if (of_address_to_resource(np->parent, i,
+ &chip->rsrc[i])) {
+ printk(KERN_ERR "snd: can't translate rsrc "
+ " %d (%s)\n", i, rnames[i]);
+ err = -ENODEV;
+ goto __error;
+ }
+ if (request_mem_region(chip->rsrc[i].start,
+ chip->rsrc[i].end -
+ chip->rsrc[i].start + 1,
+ rnames[i]) == NULL) {
+ printk(KERN_ERR "snd: can't request rsrc "
+ " %d (%s: 0x%016llx:%016llx)\n",
+ i, rnames[i],
+ (unsigned long long)chip->rsrc[i].start,
+ (unsigned long long)chip->rsrc[i].end);
+ err = -ENODEV;
+ goto __error;
+ }
+ chip->requested |= (1 << i);
+ }
+ ctrl_addr = chip->rsrc[0].start;
+ txdma_addr = chip->rsrc[1].start;
+ rxdma_addr = txdma_addr + 0x100;
+ } else {
+ static char *rnames[] = {
+ "Sound Control", "Sound Tx DMA", "Sound Rx DMA" };
+ for (i = 0; i < 3; i ++) {
+ if (of_address_to_resource(np, i,
+ &chip->rsrc[i])) {
+ printk(KERN_ERR "snd: can't translate rsrc "
+ " %d (%s)\n", i, rnames[i]);
+ err = -ENODEV;
+ goto __error;
+ }
+ if (request_mem_region(chip->rsrc[i].start,
+ chip->rsrc[i].end -
+ chip->rsrc[i].start + 1,
+ rnames[i]) == NULL) {
+ printk(KERN_ERR "snd: can't request rsrc "
+ " %d (%s: 0x%016llx:%016llx)\n",
+ i, rnames[i],
+ (unsigned long long)chip->rsrc[i].start,
+ (unsigned long long)chip->rsrc[i].end);
+ err = -ENODEV;
+ goto __error;
+ }
+ chip->requested |= (1 << i);
+ }
+ ctrl_addr = chip->rsrc[0].start;
+ txdma_addr = chip->rsrc[1].start;
+ rxdma_addr = chip->rsrc[2].start;
+ }
+
+ chip->awacs = ioremap(ctrl_addr, 0x1000);
+ chip->playback.dma = ioremap(txdma_addr, 0x100);
+ chip->capture.dma = ioremap(rxdma_addr, 0x100);
+ if (chip->model <= PMAC_BURGUNDY) {
+ irq = irq_of_parse_and_map(np, 0);
+ if (request_irq(irq, snd_pmac_ctrl_intr, 0,
+ "PMac", (void*)chip)) {
+ snd_printk(KERN_ERR "pmac: unable to grab IRQ %d\n",
+ irq);
+ err = -EBUSY;
+ goto __error;
+ }
+ chip->irq = irq;
+ }
+ irq = irq_of_parse_and_map(np, 1);
+ if (request_irq(irq, snd_pmac_tx_intr, 0, "PMac Output", (void*)chip)){
+ snd_printk(KERN_ERR "pmac: unable to grab IRQ %d\n", irq);
+ err = -EBUSY;
+ goto __error;
+ }
+ chip->tx_irq = irq;
+ irq = irq_of_parse_and_map(np, 2);
+ if (request_irq(irq, snd_pmac_rx_intr, 0, "PMac Input", (void*)chip)) {
+ snd_printk(KERN_ERR "pmac: unable to grab IRQ %d\n", irq);
+ err = -EBUSY;
+ goto __error;
+ }
+ chip->rx_irq = irq;
+
+ snd_pmac_sound_feature(chip, 1);
+
+ /* reset & enable interrupts */
+ if (chip->model <= PMAC_BURGUNDY)
+ out_le32(&chip->awacs->control, chip->control_mask);
+
+ /* Powerbooks have odd ways of enabling inputs such as
+ an expansion-bay CD or sound from an internal modem
+ or a PC-card modem. */
+ if (chip->is_pbook_3400) {
+ /* Enable CD and PC-card sound inputs. */
+ /* This is done by reading from address
+ * f301a000, + 0x10 to enable the expansion-bay
+ * CD sound input, + 0x80 to enable the PC-card
+ * sound input. The 0x100 enables the SCSI bus
+ * terminator power.
+ */
+ chip->latch_base = ioremap (0xf301a000, 0x1000);
+ in_8(chip->latch_base + 0x190);
+ } else if (chip->is_pbook_G3) {
+ struct device_node* mio;
+ for (mio = chip->node->parent; mio; mio = mio->parent) {
+ if (strcmp(mio->name, "mac-io") == 0) {
+ struct resource r;
+ if (of_address_to_resource(mio, 0, &r) == 0)
+ chip->macio_base =
+ ioremap(r.start, 0x40);
+ break;
+ }
+ }
+ /* Enable CD sound input. */
+ /* The relevant bits for writing to this byte are 0x8f.
+ * I haven't found out what the 0x80 bit does.
+ * For the 0xf bits, writing 3 or 7 enables the CD
+ * input, any other value disables it. Values
+ * 1, 3, 5, 7 enable the microphone. Values 0, 2,
+ * 4, 6, 8 - f enable the input from the modem.
+ */
+ if (chip->macio_base)
+ out_8(chip->macio_base + 0x37, 3);
+ }
+
+ /* Reset dbdma channels */
+ snd_pmac_dbdma_reset(chip);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0)
+ goto __error;
+
+ *chip_return = chip;
+ return 0;
+
+ __error:
+ snd_pmac_free(chip);
+ return err;
+}
+
+
+/*
+ * sleep notify for powerbook
+ */
+
+#ifdef CONFIG_PM
+
+/*
+ * Save state when going to sleep, restore it afterwards.
+ */
+
+void snd_pmac_suspend(struct snd_pmac *chip)
+{
+ unsigned long flags;
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ if (chip->suspend)
+ chip->suspend(chip);
+ snd_pcm_suspend_all(chip->pcm);
+ spin_lock_irqsave(&chip->reg_lock, flags);
+ snd_pmac_beep_stop(chip);
+ spin_unlock_irqrestore(&chip->reg_lock, flags);
+ if (chip->irq >= 0)
+ disable_irq(chip->irq);
+ if (chip->tx_irq >= 0)
+ disable_irq(chip->tx_irq);
+ if (chip->rx_irq >= 0)
+ disable_irq(chip->rx_irq);
+ snd_pmac_sound_feature(chip, 0);
+}
+
+void snd_pmac_resume(struct snd_pmac *chip)
+{
+ snd_pmac_sound_feature(chip, 1);
+ if (chip->resume)
+ chip->resume(chip);
+ /* enable CD sound input */
+ if (chip->macio_base && chip->is_pbook_G3)
+ out_8(chip->macio_base + 0x37, 3);
+ else if (chip->is_pbook_3400)
+ in_8(chip->latch_base + 0x190);
+
+ snd_pmac_pcm_set_format(chip);
+
+ if (chip->irq >= 0)
+ enable_irq(chip->irq);
+ if (chip->tx_irq >= 0)
+ enable_irq(chip->tx_irq);
+ if (chip->rx_irq >= 0)
+ enable_irq(chip->rx_irq);
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
+}
+
+#endif /* CONFIG_PM */
+
diff --git a/sound/ppc/pmac.h b/sound/ppc/pmac.h
new file mode 100644
index 0000000..25c512c
--- /dev/null
+++ b/sound/ppc/pmac.h
@@ -0,0 +1,210 @@
+/*
+ * Driver for PowerMac onboard soundchips
+ * Copyright (c) 2001 by Takashi Iwai <tiwai@suse.de>
+ * based on dmasound.c.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#ifndef __PMAC_H
+#define __PMAC_H
+
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include "awacs.h"
+
+#include <linux/adb.h>
+#ifdef CONFIG_ADB_CUDA
+#include <linux/cuda.h>
+#endif
+#ifdef CONFIG_ADB_PMU
+#include <linux/pmu.h>
+#endif
+#include <linux/nvram.h>
+#include <linux/tty.h>
+#include <linux/vt_kern.h>
+#include <asm/dbdma.h>
+#include <asm/prom.h>
+#include <asm/machdep.h>
+
+/* maximum number of fragments */
+#define PMAC_MAX_FRAGS 32
+
+
+#define PMAC_SUPPORT_AUTOMUTE
+
+/*
+ * DBDMA space
+ */
+struct pmac_dbdma {
+ dma_addr_t dma_base;
+ dma_addr_t addr;
+ struct dbdma_cmd __iomem *cmds;
+ void *space;
+ int size;
+};
+
+/*
+ * playback/capture stream
+ */
+struct pmac_stream {
+ int running; /* boolean */
+
+ int stream; /* PLAYBACK/CAPTURE */
+
+ int dma_size; /* in bytes */
+ int period_size; /* in bytes */
+ int buffer_size; /* in kbytes */
+ int nperiods, cur_period;
+
+ struct pmac_dbdma cmd;
+ volatile struct dbdma_regs __iomem *dma;
+
+ struct snd_pcm_substream *substream;
+
+ unsigned int cur_freqs; /* currently available frequencies */
+ unsigned int cur_formats; /* currently available formats */
+};
+
+
+/*
+ */
+
+enum snd_pmac_model {
+ PMAC_AWACS, PMAC_SCREAMER, PMAC_BURGUNDY, PMAC_DACA, PMAC_TUMBLER,
+ PMAC_SNAPPER
+};
+
+struct snd_pmac {
+ struct snd_card *card;
+
+ /* h/w info */
+ struct device_node *node;
+ struct pci_dev *pdev;
+ unsigned int revision;
+ unsigned int manufacturer;
+ unsigned int subframe;
+ unsigned int device_id;
+ enum snd_pmac_model model;
+
+ unsigned int has_iic : 1;
+ unsigned int is_pbook_3400 : 1;
+ unsigned int is_pbook_G3 : 1;
+ unsigned int is_k2 : 1;
+
+ unsigned int can_byte_swap : 1;
+ unsigned int can_duplex : 1;
+ unsigned int can_capture : 1;
+
+ unsigned int auto_mute : 1;
+ unsigned int initialized : 1;
+ unsigned int feature_is_set : 1;
+
+ unsigned int requested;
+ struct resource rsrc[3];
+
+ int num_freqs;
+ int *freq_table;
+ unsigned int freqs_ok; /* bit flags */
+ unsigned int formats_ok; /* pcm hwinfo */
+ int active;
+ int rate_index;
+ int format; /* current format */
+
+ spinlock_t reg_lock;
+ volatile struct awacs_regs __iomem *awacs;
+ int awacs_reg[8]; /* register cache */
+ unsigned int hp_stat_mask;
+
+ unsigned char __iomem *latch_base;
+ unsigned char __iomem *macio_base;
+
+ struct pmac_stream playback;
+ struct pmac_stream capture;
+
+ struct pmac_dbdma extra_dma;
+
+ int irq, tx_irq, rx_irq;
+
+ struct snd_pcm *pcm;
+
+ struct pmac_beep *beep;
+
+ unsigned int control_mask; /* control mask */
+
+ /* mixer stuffs */
+ void *mixer_data;
+ void (*mixer_free)(struct snd_pmac *);
+ struct snd_kcontrol *master_sw_ctl;
+ struct snd_kcontrol *speaker_sw_ctl;
+ struct snd_kcontrol *drc_sw_ctl; /* only used for tumbler -ReneR */
+ struct snd_kcontrol *hp_detect_ctl;
+ struct snd_kcontrol *lineout_sw_ctl;
+
+ /* lowlevel callbacks */
+ void (*set_format)(struct snd_pmac *chip);
+ void (*update_automute)(struct snd_pmac *chip, int do_notify);
+ int (*detect_headphone)(struct snd_pmac *chip);
+#ifdef CONFIG_PM
+ void (*suspend)(struct snd_pmac *chip);
+ void (*resume)(struct snd_pmac *chip);
+#endif
+
+};
+
+
+/* exported functions */
+int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return);
+int snd_pmac_pcm_new(struct snd_pmac *chip);
+int snd_pmac_attach_beep(struct snd_pmac *chip);
+void snd_pmac_detach_beep(struct snd_pmac *chip);
+void snd_pmac_beep_stop(struct snd_pmac *chip);
+unsigned int snd_pmac_rate_index(struct snd_pmac *chip, struct pmac_stream *rec, unsigned int rate);
+
+void snd_pmac_beep_dma_start(struct snd_pmac *chip, int bytes, unsigned long addr, int speed);
+void snd_pmac_beep_dma_stop(struct snd_pmac *chip);
+
+#ifdef CONFIG_PM
+void snd_pmac_suspend(struct snd_pmac *chip);
+void snd_pmac_resume(struct snd_pmac *chip);
+#endif
+
+/* initialize mixer */
+int snd_pmac_awacs_init(struct snd_pmac *chip);
+int snd_pmac_burgundy_init(struct snd_pmac *chip);
+int snd_pmac_daca_init(struct snd_pmac *chip);
+int snd_pmac_tumbler_init(struct snd_pmac *chip);
+int snd_pmac_tumbler_post_init(void);
+
+/* i2c functions */
+struct pmac_keywest {
+ int addr;
+ struct i2c_client *client;
+ int id;
+ int (*init_client)(struct pmac_keywest *i2c);
+ char *name;
+};
+
+int snd_pmac_keywest_init(struct pmac_keywest *i2c);
+void snd_pmac_keywest_cleanup(struct pmac_keywest *i2c);
+
+/* misc */
+#define snd_pmac_boolean_stereo_info snd_ctl_boolean_stereo_info
+#define snd_pmac_boolean_mono_info snd_ctl_boolean_mono_info
+
+int snd_pmac_add_automute(struct snd_pmac *chip);
+
+#endif /* __PMAC_H */
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
new file mode 100644
index 0000000..c936225
--- /dev/null
+++ b/sound/ppc/powermac.c
@@ -0,0 +1,195 @@
+/*
+ * Driver for PowerMac AWACS
+ * Copyright (c) 2001 by Takashi Iwai <tiwai@suse.de>
+ * based on dmasound.c.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/platform_device.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include "pmac.h"
+#include "awacs.h"
+#include "burgundy.h"
+
+#define CHIP_NAME "PMac"
+
+MODULE_DESCRIPTION("PowerMac");
+MODULE_SUPPORTED_DEVICE("{{Apple,PowerMac}}");
+MODULE_LICENSE("GPL");
+
+static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
+static int enable_beep = 1;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for " CHIP_NAME " soundchip.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for " CHIP_NAME " soundchip.");
+module_param(enable_beep, bool, 0444);
+MODULE_PARM_DESC(enable_beep, "Enable beep using PCM.");
+
+static struct platform_device *device;
+
+
+/*
+ */
+
+static int __init snd_pmac_probe(struct platform_device *devptr)
+{
+ struct snd_card *card;
+ struct snd_pmac *chip;
+ char *name_ext;
+ int err;
+
+ card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_pmac_new(card, &chip)) < 0)
+ goto __error;
+ card->private_data = chip;
+
+ switch (chip->model) {
+ case PMAC_BURGUNDY:
+ strcpy(card->driver, "PMac Burgundy");
+ strcpy(card->shortname, "PowerMac Burgundy");
+ sprintf(card->longname, "%s (Dev %d) Sub-frame %d",
+ card->shortname, chip->device_id, chip->subframe);
+ if ((err = snd_pmac_burgundy_init(chip)) < 0)
+ goto __error;
+ break;
+ case PMAC_DACA:
+ strcpy(card->driver, "PMac DACA");
+ strcpy(card->shortname, "PowerMac DACA");
+ sprintf(card->longname, "%s (Dev %d) Sub-frame %d",
+ card->shortname, chip->device_id, chip->subframe);
+ if ((err = snd_pmac_daca_init(chip)) < 0)
+ goto __error;
+ break;
+ case PMAC_TUMBLER:
+ case PMAC_SNAPPER:
+ name_ext = chip->model == PMAC_TUMBLER ? "Tumbler" : "Snapper";
+ sprintf(card->driver, "PMac %s", name_ext);
+ sprintf(card->shortname, "PowerMac %s", name_ext);
+ sprintf(card->longname, "%s (Dev %d) Sub-frame %d",
+ card->shortname, chip->device_id, chip->subframe);
+ if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0)
+ goto __error;
+ break;
+ case PMAC_AWACS:
+ case PMAC_SCREAMER:
+ name_ext = chip->model == PMAC_SCREAMER ? "Screamer" : "AWACS";
+ sprintf(card->driver, "PMac %s", name_ext);
+ sprintf(card->shortname, "PowerMac %s", name_ext);
+ if (chip->is_pbook_3400)
+ name_ext = " [PB3400]";
+ else if (chip->is_pbook_G3)
+ name_ext = " [PBG3]";
+ else
+ name_ext = "";
+ sprintf(card->longname, "%s%s Rev %d",
+ card->shortname, name_ext, chip->revision);
+ if ((err = snd_pmac_awacs_init(chip)) < 0)
+ goto __error;
+ break;
+ default:
+ snd_printk("unsupported hardware %d\n", chip->model);
+ err = -EINVAL;
+ goto __error;
+ }
+
+ if ((err = snd_pmac_pcm_new(chip)) < 0)
+ goto __error;
+
+ chip->initialized = 1;
+ if (enable_beep)
+ snd_pmac_attach_beep(chip);
+
+ snd_card_set_dev(card, &devptr->dev);
+
+ if ((err = snd_card_register(card)) < 0)
+ goto __error;
+
+ platform_set_drvdata(devptr, card);
+ return 0;
+
+__error:
+ snd_card_free(card);
+ return err;
+}
+
+
+static int __devexit snd_pmac_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_pmac_driver_suspend(struct platform_device *devptr, pm_message_t state)
+{
+ struct snd_card *card = platform_get_drvdata(devptr);
+ snd_pmac_suspend(card->private_data);
+ return 0;
+}
+
+static int snd_pmac_driver_resume(struct platform_device *devptr)
+{
+ struct snd_card *card = platform_get_drvdata(devptr);
+ snd_pmac_resume(card->private_data);
+ return 0;
+}
+#endif
+
+#define SND_PMAC_DRIVER "snd_powermac"
+
+static struct platform_driver snd_pmac_driver = {
+ .probe = snd_pmac_probe,
+ .remove = __devexit_p(snd_pmac_remove),
+#ifdef CONFIG_PM
+ .suspend = snd_pmac_driver_suspend,
+ .resume = snd_pmac_driver_resume,
+#endif
+ .driver = {
+ .name = SND_PMAC_DRIVER
+ },
+};
+
+static int __init alsa_card_pmac_init(void)
+{
+ int err;
+
+ if ((err = platform_driver_register(&snd_pmac_driver)) < 0)
+ return err;
+ device = platform_device_register_simple(SND_PMAC_DRIVER, -1, NULL, 0);
+ return 0;
+
+}
+
+static void __exit alsa_card_pmac_exit(void)
+{
+ if (!IS_ERR(device))
+ platform_device_unregister(device);
+ platform_driver_unregister(&snd_pmac_driver);
+}
+
+module_init(alsa_card_pmac_init)
+module_exit(alsa_card_pmac_exit)
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
new file mode 100644
index 0000000..8f9e385
--- /dev/null
+++ b/sound/ppc/snd_ps3.c
@@ -0,0 +1,1208 @@
+/*
+ * Audio support for PS3
+ * Copyright (C) 2007 Sony Computer Entertainment Inc.
+ * All rights reserved.
+ * Copyright 2006, 2007 Sony Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; version 2 of the Licence.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/interrupt.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/asound.h>
+#include <sound/memalloc.h>
+#include <sound/pcm_params.h>
+#include <sound/control.h>
+#include <linux/dmapool.h>
+#include <linux/dma-mapping.h>
+#include <asm/firmware.h>
+#include <asm/dma.h>
+#include <asm/lv1call.h>
+#include <asm/ps3.h>
+#include <asm/ps3av.h>
+
+#include "snd_ps3_reg.h"
+#include "snd_ps3.h"
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("PS3 sound driver");
+MODULE_AUTHOR("Sony Computer Entertainment Inc.");
+
+/* module entries */
+static int __init snd_ps3_init(void);
+static void __exit snd_ps3_exit(void);
+
+/* ALSA snd driver ops */
+static int snd_ps3_pcm_open(struct snd_pcm_substream *substream);
+static int snd_ps3_pcm_close(struct snd_pcm_substream *substream);
+static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream);
+static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd);
+static snd_pcm_uframes_t snd_ps3_pcm_pointer(struct snd_pcm_substream
+ *substream);
+static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params);
+static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream);
+
+
+/* ps3_system_bus_driver entries */
+static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev);
+static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev);
+
+/* address setup */
+static int snd_ps3_map_mmio(void);
+static void snd_ps3_unmap_mmio(void);
+static int snd_ps3_allocate_irq(void);
+static void snd_ps3_free_irq(void);
+static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start);
+
+/* interrupt handler */
+static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id);
+
+
+/* set sampling rate/format */
+static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream);
+/* take effect parameter change */
+static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card);
+/* initialize avsetting and take it effect */
+static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card);
+/* setup dma */
+static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
+ enum snd_ps3_dma_filltype filltype);
+static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card);
+
+static dma_addr_t v_to_bus(struct snd_ps3_card_info *, void *vaddr, int ch);
+
+
+module_init(snd_ps3_init);
+module_exit(snd_ps3_exit);
+
+/*
+ * global
+ */
+static struct snd_ps3_card_info the_card;
+
+static int snd_ps3_start_delay = CONFIG_SND_PS3_DEFAULT_START_DELAY;
+
+module_param_named(start_delay, snd_ps3_start_delay, uint, 0644);
+MODULE_PARM_DESC(start_delay, "time to insert silent data in milisec");
+
+static int index = SNDRV_DEFAULT_IDX1;
+static char *id = SNDRV_DEFAULT_STR1;
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for PS3 soundchip.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for PS3 soundchip.");
+
+
+/*
+ * PS3 audio register access
+ */
+static inline u32 read_reg(unsigned int reg)
+{
+ return in_be32(the_card.mapped_mmio_vaddr + reg);
+}
+static inline void write_reg(unsigned int reg, u32 val)
+{
+ out_be32(the_card.mapped_mmio_vaddr + reg, val);
+}
+static inline void update_reg(unsigned int reg, u32 or_val)
+{
+ u32 newval = read_reg(reg) | or_val;
+ write_reg(reg, newval);
+}
+static inline void update_mask_reg(unsigned int reg, u32 mask, u32 or_val)
+{
+ u32 newval = (read_reg(reg) & mask) | or_val;
+ write_reg(reg, newval);
+}
+
+/*
+ * ALSA defs
+ */
+static const struct snd_pcm_hardware snd_ps3_pcm_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE),
+ .rates = (SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 44100,
+ .rate_max = 96000,
+
+ .channels_min = 2, /* stereo only */
+ .channels_max = 2,
+
+ .buffer_bytes_max = PS3_AUDIO_FIFO_SIZE * 64,
+
+ /* interrupt by four stages */
+ .period_bytes_min = PS3_AUDIO_FIFO_STAGE_SIZE * 4,
+ .period_bytes_max = PS3_AUDIO_FIFO_STAGE_SIZE * 4,
+
+ .periods_min = 16,
+ .periods_max = 32, /* buffer_size_max/ period_bytes_max */
+
+ .fifo_size = PS3_AUDIO_FIFO_SIZE
+};
+
+static struct snd_pcm_ops snd_ps3_pcm_spdif_ops =
+{
+ .open = snd_ps3_pcm_open,
+ .close = snd_ps3_pcm_close,
+ .prepare = snd_ps3_pcm_prepare,
+ .ioctl = snd_pcm_lib_ioctl,
+ .trigger = snd_ps3_pcm_trigger,
+ .pointer = snd_ps3_pcm_pointer,
+ .hw_params = snd_ps3_pcm_hw_params,
+ .hw_free = snd_ps3_pcm_hw_free
+};
+
+static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card,
+ int count, int force_stop)
+{
+ int dma_ch, done, retries, stop_forced = 0;
+ uint32_t status;
+
+ for (dma_ch = 0; dma_ch < 8; dma_ch ++) {
+ retries = count;
+ do {
+ status = read_reg(PS3_AUDIO_KICK(dma_ch)) &
+ PS3_AUDIO_KICK_STATUS_MASK;
+ switch (status) {
+ case PS3_AUDIO_KICK_STATUS_DONE:
+ case PS3_AUDIO_KICK_STATUS_NOTIFY:
+ case PS3_AUDIO_KICK_STATUS_CLEAR:
+ case PS3_AUDIO_KICK_STATUS_ERROR:
+ done = 1;
+ break;
+ default:
+ done = 0;
+ udelay(10);
+ }
+ } while (!done && --retries);
+ if (!retries && force_stop) {
+ pr_info("%s: DMA ch %d is not stopped.",
+ __func__, dma_ch);
+ /* last resort. force to stop dma.
+ * NOTE: this cause DMA done interrupts
+ */
+ update_reg(PS3_AUDIO_CONFIG, PS3_AUDIO_CONFIG_CLEAR);
+ stop_forced = 1;
+ }
+ }
+ return stop_forced;
+}
+
+/*
+ * wait for all dma is done.
+ * NOTE: caller should reset card->running before call.
+ * If not, the interrupt handler will re-start DMA,
+ * then DMA is never stopped.
+ */
+static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card)
+{
+ int stop_forced;
+ /*
+ * wait for the last dma is done
+ */
+
+ /*
+ * expected maximum DMA done time is 5.7ms + something (DMA itself).
+ * 5.7ms is from 16bit/sample 2ch 44.1Khz; the time next
+ * DMA kick event would occur.
+ */
+ stop_forced = snd_ps3_verify_dma_stop(card, 700, 1);
+
+ /*
+ * clear outstanding interrupts.
+ */
+ update_reg(PS3_AUDIO_INTR_0, 0);
+ update_reg(PS3_AUDIO_AX_IS, 0);
+
+ /*
+ *revert CLEAR bit since it will not reset automatically after DMA stop
+ */
+ if (stop_forced)
+ update_mask_reg(PS3_AUDIO_CONFIG, ~PS3_AUDIO_CONFIG_CLEAR, 0);
+ /* ensure the hardware sees changes */
+ wmb();
+}
+
+static void snd_ps3_kick_dma(struct snd_ps3_card_info *card)
+{
+
+ update_reg(PS3_AUDIO_KICK(0), PS3_AUDIO_KICK_REQUEST);
+ /* ensure the hardware sees the change */
+ wmb();
+}
+
+/*
+ * convert virtual addr to ioif bus addr.
+ */
+static dma_addr_t v_to_bus(struct snd_ps3_card_info *card,
+ void * paddr,
+ int ch)
+{
+ return card->dma_start_bus_addr[ch] +
+ (paddr - card->dma_start_vaddr[ch]);
+};
+
+
+/*
+ * increment ring buffer pointer.
+ * NOTE: caller must hold write spinlock
+ */
+static void snd_ps3_bump_buffer(struct snd_ps3_card_info *card,
+ enum snd_ps3_ch ch, size_t byte_count,
+ int stage)
+{
+ if (!stage)
+ card->dma_last_transfer_vaddr[ch] =
+ card->dma_next_transfer_vaddr[ch];
+ card->dma_next_transfer_vaddr[ch] += byte_count;
+ if ((card->dma_start_vaddr[ch] + (card->dma_buffer_size / 2)) <=
+ card->dma_next_transfer_vaddr[ch]) {
+ card->dma_next_transfer_vaddr[ch] = card->dma_start_vaddr[ch];
+ }
+}
+/*
+ * setup dmac to send data to audio and attenuate samples on the ring buffer
+ */
+static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
+ enum snd_ps3_dma_filltype filltype)
+{
+ /* this dmac does not support over 4G */
+ uint32_t dma_addr;
+ int fill_stages, dma_ch, stage;
+ enum snd_ps3_ch ch;
+ uint32_t ch0_kick_event = 0; /* initialize to mute gcc */
+ void *start_vaddr;
+ unsigned long irqsave;
+ int silent = 0;
+
+ switch (filltype) {
+ case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL:
+ silent = 1;
+ /* intentionally fall thru */
+ case SND_PS3_DMA_FILLTYPE_FIRSTFILL:
+ ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS;
+ break;
+
+ case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING:
+ silent = 1;
+ /* intentionally fall thru */
+ case SND_PS3_DMA_FILLTYPE_RUNNING:
+ ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY;
+ break;
+ }
+
+ snd_ps3_verify_dma_stop(card, 700, 0);
+ fill_stages = 4;
+ spin_lock_irqsave(&card->dma_lock, irqsave);
+ for (ch = 0; ch < 2; ch++) {
+ start_vaddr = card->dma_next_transfer_vaddr[0];
+ for (stage = 0; stage < fill_stages; stage ++) {
+ dma_ch = stage * 2 + ch;
+ if (silent)
+ dma_addr = card->null_buffer_start_dma_addr;
+ else
+ dma_addr =
+ v_to_bus(card,
+ card->dma_next_transfer_vaddr[ch],
+ ch);
+
+ write_reg(PS3_AUDIO_SOURCE(dma_ch),
+ (PS3_AUDIO_SOURCE_TARGET_SYSTEM_MEMORY |
+ dma_addr));
+
+ /* dst: fixed to 3wire#0 */
+ if (ch == 0)
+ write_reg(PS3_AUDIO_DEST(dma_ch),
+ (PS3_AUDIO_DEST_TARGET_AUDIOFIFO |
+ PS3_AUDIO_AO_3W_LDATA(0)));
+ else
+ write_reg(PS3_AUDIO_DEST(dma_ch),
+ (PS3_AUDIO_DEST_TARGET_AUDIOFIFO |
+ PS3_AUDIO_AO_3W_RDATA(0)));
+
+ /* count always 1 DMA block (1/2 stage = 128 bytes) */
+ write_reg(PS3_AUDIO_DMASIZE(dma_ch), 0);
+ /* bump pointer if needed */
+ if (!silent)
+ snd_ps3_bump_buffer(card, ch,
+ PS3_AUDIO_DMAC_BLOCK_SIZE,
+ stage);
+
+ /* kick event */
+ if (dma_ch == 0)
+ write_reg(PS3_AUDIO_KICK(dma_ch),
+ ch0_kick_event);
+ else
+ write_reg(PS3_AUDIO_KICK(dma_ch),
+ PS3_AUDIO_KICK_EVENT_AUDIO_DMA(dma_ch
+ - 1) |
+ PS3_AUDIO_KICK_REQUEST);
+ }
+ }
+ /* ensure the hardware sees the change */
+ wmb();
+ spin_unlock_irqrestore(&card->dma_lock, irqsave);
+
+ return 0;
+}
+
+/*
+ * audio mute on/off
+ * mute_on : 0 output enabled
+ * 1 mute
+ */
+static int snd_ps3_mute(int mute_on)
+{
+ return ps3av_audio_mute(mute_on);
+}
+
+/*
+ * PCM operators
+ */
+static int snd_ps3_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
+ int pcm_index;
+
+ pcm_index = substream->pcm->device;
+ /* to retrieve substream/runtime in interrupt handler */
+ card->substream = substream;
+
+ runtime->hw = snd_ps3_pcm_hw;
+
+ card->start_delay = snd_ps3_start_delay;
+
+ /* mute off */
+ snd_ps3_mute(0); /* this function sleep */
+
+ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ PS3_AUDIO_FIFO_STAGE_SIZE * 4 * 2);
+ return 0;
+};
+
+static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ size_t size;
+
+ /* alloc transport buffer */
+ size = params_buffer_bytes(hw_params);
+ snd_pcm_lib_malloc_pages(substream, size);
+ return 0;
+};
+
+static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream,
+ unsigned int delay_ms)
+{
+ int ret;
+ int rate ;
+
+ rate = substream->runtime->rate;
+ ret = snd_pcm_format_size(substream->runtime->format,
+ rate * delay_ms / 1000)
+ * substream->runtime->channels;
+
+ pr_debug(KERN_ERR "%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n",
+ __func__,
+ delay_ms,
+ rate,
+ snd_pcm_format_size(substream->runtime->format, rate),
+ rate * delay_ms / 1000,
+ ret);
+
+ return ret;
+};
+
+static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
+ unsigned long irqsave;
+
+ if (!snd_ps3_set_avsetting(substream)) {
+ /* some parameter changed */
+ write_reg(PS3_AUDIO_AX_IE,
+ PS3_AUDIO_AX_IE_ASOBEIE(0) |
+ PS3_AUDIO_AX_IE_ASOBUIE(0));
+ /*
+ * let SPDIF device re-lock with SPDIF signal,
+ * start with some silence
+ */
+ card->silent = snd_ps3_delay_to_bytes(substream,
+ card->start_delay) /
+ (PS3_AUDIO_FIFO_STAGE_SIZE * 4); /* every 4 times */
+ }
+
+ /* restart ring buffer pointer */
+ spin_lock_irqsave(&card->dma_lock, irqsave);
+ {
+ card->dma_buffer_size = runtime->dma_bytes;
+
+ card->dma_last_transfer_vaddr[SND_PS3_CH_L] =
+ card->dma_next_transfer_vaddr[SND_PS3_CH_L] =
+ card->dma_start_vaddr[SND_PS3_CH_L] =
+ runtime->dma_area;
+ card->dma_start_bus_addr[SND_PS3_CH_L] = runtime->dma_addr;
+
+ card->dma_last_transfer_vaddr[SND_PS3_CH_R] =
+ card->dma_next_transfer_vaddr[SND_PS3_CH_R] =
+ card->dma_start_vaddr[SND_PS3_CH_R] =
+ runtime->dma_area + (runtime->dma_bytes / 2);
+ card->dma_start_bus_addr[SND_PS3_CH_R] =
+ runtime->dma_addr + (runtime->dma_bytes / 2);
+
+ pr_debug("%s: vaddr=%p bus=%#lx\n", __func__,
+ card->dma_start_vaddr[SND_PS3_CH_L],
+ card->dma_start_bus_addr[SND_PS3_CH_L]);
+
+ }
+ spin_unlock_irqrestore(&card->dma_lock, irqsave);
+
+ /* ensure the hardware sees the change */
+ mb();
+
+ return 0;
+};
+
+static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* clear outstanding interrupts */
+ update_reg(PS3_AUDIO_AX_IS, 0);
+
+ spin_lock(&card->dma_lock);
+ {
+ card->running = 1;
+ }
+ spin_unlock(&card->dma_lock);
+
+ snd_ps3_program_dma(card,
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
+ snd_ps3_kick_dma(card);
+ while (read_reg(PS3_AUDIO_KICK(7)) &
+ PS3_AUDIO_KICK_STATUS_MASK) {
+ udelay(1);
+ }
+ snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_RUNNING);
+ snd_ps3_kick_dma(card);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ spin_lock(&card->dma_lock);
+ {
+ card->running = 0;
+ }
+ spin_unlock(&card->dma_lock);
+ snd_ps3_wait_for_dma_stop(card);
+ break;
+ default:
+ break;
+
+ }
+
+ return ret;
+};
+
+/*
+ * report current pointer
+ */
+static snd_pcm_uframes_t snd_ps3_pcm_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
+ size_t bytes;
+ snd_pcm_uframes_t ret;
+
+ spin_lock(&card->dma_lock);
+ {
+ bytes = (size_t)(card->dma_last_transfer_vaddr[SND_PS3_CH_L] -
+ card->dma_start_vaddr[SND_PS3_CH_L]);
+ }
+ spin_unlock(&card->dma_lock);
+
+ ret = bytes_to_frames(substream->runtime, bytes * 2);
+
+ return ret;
+};
+
+static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ int ret;
+ ret = snd_pcm_lib_free_pages(substream);
+ return ret;
+};
+
+static int snd_ps3_pcm_close(struct snd_pcm_substream *substream)
+{
+ /* mute on */
+ snd_ps3_mute(1);
+ return 0;
+};
+
+static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card)
+{
+ /*
+ * avsetting driver seems to never change the followings
+ * so, init them here once
+ */
+
+ /* no dma interrupt needed */
+ write_reg(PS3_AUDIO_INTR_EN_0, 0);
+
+ /* use every 4 buffer empty interrupt */
+ update_mask_reg(PS3_AUDIO_AX_IC,
+ PS3_AUDIO_AX_IC_AASOIMD_MASK,
+ PS3_AUDIO_AX_IC_AASOIMD_EVERY4);
+
+ /* enable 3wire clocks */
+ update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
+ ~(PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED |
+ PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED),
+ 0);
+ update_reg(PS3_AUDIO_AO_3WMCTRL,
+ PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT);
+}
+
+/*
+ * av setting
+ * NOTE: calling this function may generate audio interrupt.
+ */
+static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card)
+{
+ int ret, retries, i;
+ pr_debug("%s: start\n", __func__);
+
+ ret = ps3av_set_audio_mode(card->avs.avs_audio_ch,
+ card->avs.avs_audio_rate,
+ card->avs.avs_audio_width,
+ card->avs.avs_audio_format,
+ card->avs.avs_audio_source);
+ /*
+ * Reset the following unwanted settings:
+ */
+
+ /* disable all 3wire buffers */
+ update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
+ ~(PS3_AUDIO_AO_3WMCTRL_ASOEN(0) |
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(1) |
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(2) |
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(3)),
+ 0);
+ wmb(); /* ensure the hardware sees the change */
+ /* wait for actually stopped */
+ retries = 1000;
+ while ((read_reg(PS3_AUDIO_AO_3WMCTRL) &
+ (PS3_AUDIO_AO_3WMCTRL_ASORUN(0) |
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(1) |
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(2) |
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(3))) &&
+ --retries) {
+ udelay(1);
+ }
+
+ /* reset buffer pointer */
+ for (i = 0; i < 4; i++) {
+ update_reg(PS3_AUDIO_AO_3WCTRL(i),
+ PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET);
+ udelay(10);
+ }
+ wmb(); /* ensure the hardware actually start resetting */
+
+ /* enable 3wire#0 buffer */
+ update_reg(PS3_AUDIO_AO_3WMCTRL, PS3_AUDIO_AO_3WMCTRL_ASOEN(0));
+
+
+ /* In 24bit mode,ALSA inserts a zero byte at first byte of per sample */
+ update_mask_reg(PS3_AUDIO_AO_3WCTRL(0),
+ ~PS3_AUDIO_AO_3WCTRL_ASODF,
+ PS3_AUDIO_AO_3WCTRL_ASODF_LSB);
+ update_mask_reg(PS3_AUDIO_AO_SPDCTRL(0),
+ ~PS3_AUDIO_AO_SPDCTRL_SPODF,
+ PS3_AUDIO_AO_SPDCTRL_SPODF_LSB);
+ /* ensure all the setting above is written back to register */
+ wmb();
+ /* avsetting driver altered AX_IE, caller must reset it if you want */
+ pr_debug("%s: end\n", __func__);
+ return ret;
+}
+
+static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card)
+{
+ int ret;
+ pr_debug("%s: start\n", __func__);
+ card->avs.avs_audio_ch = PS3AV_CMD_AUDIO_NUM_OF_CH_2;
+ card->avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
+ card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
+ card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM;
+ card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL;
+ memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8);
+
+ ret = snd_ps3_change_avsetting(card);
+
+ snd_ps3_audio_fixup(card);
+
+ /* to start to generate SPDIF signal, fill data */
+ snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
+ snd_ps3_kick_dma(card);
+ pr_debug("%s: end\n", __func__);
+ return ret;
+}
+
+/*
+ * set sampling rate according to the substream
+ */
+static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream)
+{
+ struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
+ struct snd_ps3_avsetting_info avs;
+ int ret;
+
+ avs = card->avs;
+
+ pr_debug("%s: called freq=%d width=%d\n", __func__,
+ substream->runtime->rate,
+ snd_pcm_format_width(substream->runtime->format));
+
+ pr_debug("%s: before freq=%d width=%d\n", __func__,
+ card->avs.avs_audio_rate, card->avs.avs_audio_width);
+
+ /* sample rate */
+ switch (substream->runtime->rate) {
+ case 44100:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K;
+ break;
+ case 48000:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
+ break;
+ case 88200:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K;
+ break;
+ case 96000:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K;
+ break;
+ default:
+ pr_info("%s: invalid rate %d\n", __func__,
+ substream->runtime->rate);
+ return 1;
+ }
+
+ /* width */
+ switch (snd_pcm_format_width(substream->runtime->format)) {
+ case 16:
+ avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
+ break;
+ case 24:
+ avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24;
+ break;
+ default:
+ pr_info("%s: invalid width %d\n", __func__,
+ snd_pcm_format_width(substream->runtime->format));
+ return 1;
+ }
+
+ memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8);
+
+ if (memcmp(&card->avs, &avs, sizeof(avs))) {
+ pr_debug("%s: after freq=%d width=%d\n", __func__,
+ card->avs.avs_audio_rate, card->avs.avs_audio_width);
+
+ card->avs = avs;
+ snd_ps3_change_avsetting(card);
+ ret = 0;
+ } else
+ ret = 1;
+
+ /* check CS non-audio bit and mute accordingly */
+ if (avs.avs_cs_info[0] & 0x02)
+ ps3av_audio_mute_analog(1); /* mute if non-audio */
+ else
+ ps3av_audio_mute_analog(0);
+
+ return ret;
+}
+
+/*
+ * SPDIF status bits controls
+ */
+static int snd_ps3_spdif_mask_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+/* FIXME: ps3av_set_audio_mode() assumes only consumer mode */
+static int snd_ps3_spdif_cmask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ memset(ucontrol->value.iec958.status, 0xff, 8);
+ return 0;
+}
+
+static int snd_ps3_spdif_pmask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return 0;
+}
+
+static int snd_ps3_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ memcpy(ucontrol->value.iec958.status, ps3av_mode_cs_info, 8);
+ return 0;
+}
+
+static int snd_ps3_spdif_default_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (memcmp(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8)) {
+ memcpy(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8);
+ return 1;
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new spdif_ctls[] = {
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .info = snd_ps3_spdif_mask_info,
+ .get = snd_ps3_spdif_cmask_get,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+ .info = snd_ps3_spdif_mask_info,
+ .get = snd_ps3_spdif_pmask_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .info = snd_ps3_spdif_mask_info,
+ .get = snd_ps3_spdif_default_get,
+ .put = snd_ps3_spdif_default_put,
+ },
+};
+
+
+static int snd_ps3_map_mmio(void)
+{
+ the_card.mapped_mmio_vaddr =
+ ioremap(the_card.ps3_dev->m_region->bus_addr,
+ the_card.ps3_dev->m_region->len);
+
+ if (!the_card.mapped_mmio_vaddr) {
+ pr_info("%s: ioremap 0 failed p=%#lx l=%#lx \n",
+ __func__, the_card.ps3_dev->m_region->lpar_addr,
+ the_card.ps3_dev->m_region->len);
+ return -ENXIO;
+ }
+
+ return 0;
+};
+
+static void snd_ps3_unmap_mmio(void)
+{
+ iounmap(the_card.mapped_mmio_vaddr);
+ the_card.mapped_mmio_vaddr = NULL;
+}
+
+static int snd_ps3_allocate_irq(void)
+{
+ int ret;
+ u64 lpar_addr, lpar_size;
+ u64 __iomem *mapped;
+
+ /* FIXME: move this to device_init (H/W probe) */
+
+ /* get irq outlet */
+ ret = lv1_gpu_device_map(1, &lpar_addr, &lpar_size);
+ if (ret) {
+ pr_info("%s: device map 1 failed %d\n", __func__,
+ ret);
+ return -ENXIO;
+ }
+
+ mapped = ioremap(lpar_addr, lpar_size);
+ if (!mapped) {
+ pr_info("%s: ioremap 1 failed \n", __func__);
+ return -ENXIO;
+ }
+
+ the_card.audio_irq_outlet = in_be64(mapped);
+
+ iounmap(mapped);
+ ret = lv1_gpu_device_unmap(1);
+ if (ret)
+ pr_info("%s: unmap 1 failed\n", __func__);
+
+ /* irq */
+ ret = ps3_irq_plug_setup(PS3_BINDING_CPU_ANY,
+ the_card.audio_irq_outlet,
+ &the_card.irq_no);
+ if (ret) {
+ pr_info("%s:ps3_alloc_irq failed (%d)\n", __func__, ret);
+ return ret;
+ }
+
+ ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED,
+ SND_PS3_DRIVER_NAME, &the_card);
+ if (ret) {
+ pr_info("%s: request_irq failed (%d)\n", __func__, ret);
+ goto cleanup_irq;
+ }
+
+ return 0;
+
+ cleanup_irq:
+ ps3_irq_plug_destroy(the_card.irq_no);
+ return ret;
+};
+
+static void snd_ps3_free_irq(void)
+{
+ free_irq(the_card.irq_no, &the_card);
+ ps3_irq_plug_destroy(the_card.irq_no);
+}
+
+static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start)
+{
+ uint64_t val;
+ int ret;
+
+ val = (ioaddr_start & (0x0fUL << 32)) >> (32 - 20) |
+ (0x03UL << 24) |
+ (0x0fUL << 12) |
+ (PS3_AUDIO_IOID);
+
+ ret = lv1_gpu_attribute(0x100, 0x007, val, 0, 0);
+ if (ret)
+ pr_info("%s: gpu_attribute failed %d\n", __func__,
+ ret);
+}
+
+static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
+{
+ int i, ret;
+ u64 lpar_addr, lpar_size;
+
+ BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1));
+ BUG_ON(dev->match_id != PS3_MATCH_ID_SOUND);
+
+ the_card.ps3_dev = dev;
+
+ ret = ps3_open_hv_device(dev);
+
+ if (ret)
+ return -ENXIO;
+
+ /* setup MMIO */
+ ret = lv1_gpu_device_map(2, &lpar_addr, &lpar_size);
+ if (ret) {
+ pr_info("%s: device map 2 failed %d\n", __func__, ret);
+ goto clean_open;
+ }
+ ps3_mmio_region_init(dev, dev->m_region, lpar_addr, lpar_size,
+ PAGE_SHIFT);
+
+ ret = snd_ps3_map_mmio();
+ if (ret)
+ goto clean_dev_map;
+
+ /* setup DMA area */
+ ps3_dma_region_init(dev, dev->d_region,
+ PAGE_SHIFT, /* use system page size */
+ 0, /* dma type; not used */
+ NULL,
+ _ALIGN_UP(SND_PS3_DMA_REGION_SIZE, PAGE_SIZE));
+ dev->d_region->ioid = PS3_AUDIO_IOID;
+
+ ret = ps3_dma_region_create(dev->d_region);
+ if (ret) {
+ pr_info("%s: region_create\n", __func__);
+ goto clean_mmio;
+ }
+
+ snd_ps3_audio_set_base_addr(dev->d_region->bus_addr);
+
+ /* CONFIG_SND_PS3_DEFAULT_START_DELAY */
+ the_card.start_delay = snd_ps3_start_delay;
+
+ /* irq */
+ if (snd_ps3_allocate_irq()) {
+ ret = -ENXIO;
+ goto clean_dma_region;
+ }
+
+ /* create card instance */
+ the_card.card = snd_card_new(index, id, THIS_MODULE, 0);
+ if (!the_card.card) {
+ ret = -ENXIO;
+ goto clean_irq;
+ }
+
+ strcpy(the_card.card->driver, "PS3");
+ strcpy(the_card.card->shortname, "PS3");
+ strcpy(the_card.card->longname, "PS3 sound");
+
+ /* create control elements */
+ for (i = 0; i < ARRAY_SIZE(spdif_ctls); i++) {
+ ret = snd_ctl_add(the_card.card,
+ snd_ctl_new1(&spdif_ctls[i], &the_card));
+ if (ret < 0)
+ goto clean_card;
+ }
+
+ /* create PCM devices instance */
+ /* NOTE:this driver works assuming pcm:substream = 1:1 */
+ ret = snd_pcm_new(the_card.card,
+ "SPDIF",
+ 0, /* instance index, will be stored pcm.device*/
+ 1, /* output substream */
+ 0, /* input substream */
+ &(the_card.pcm));
+ if (ret)
+ goto clean_card;
+
+ the_card.pcm->private_data = &the_card;
+ strcpy(the_card.pcm->name, "SPDIF");
+
+ /* set pcm ops */
+ snd_pcm_set_ops(the_card.pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_ps3_pcm_spdif_ops);
+
+ the_card.pcm->info_flags = SNDRV_PCM_INFO_NONINTERLEAVED;
+ /* pre-alloc PCM DMA buffer*/
+ ret = snd_pcm_lib_preallocate_pages_for_all(the_card.pcm,
+ SNDRV_DMA_TYPE_DEV,
+ &dev->core,
+ SND_PS3_PCM_PREALLOC_SIZE,
+ SND_PS3_PCM_PREALLOC_SIZE);
+ if (ret < 0) {
+ pr_info("%s: prealloc failed\n", __func__);
+ goto clean_card;
+ }
+
+ /*
+ * allocate null buffer
+ * its size should be lager than PS3_AUDIO_FIFO_STAGE_SIZE * 2
+ * PAGE_SIZE is enogh
+ */
+ if (!(the_card.null_buffer_start_vaddr =
+ dma_alloc_coherent(&the_card.ps3_dev->core,
+ PAGE_SIZE,
+ &the_card.null_buffer_start_dma_addr,
+ GFP_KERNEL))) {
+ pr_info("%s: nullbuffer alloc failed\n", __func__);
+ goto clean_preallocate;
+ }
+ pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__,
+ the_card.null_buffer_start_vaddr,
+ the_card.null_buffer_start_dma_addr);
+ /* set default sample rate/word width */
+ snd_ps3_init_avsetting(&the_card);
+
+ /* register the card */
+ snd_card_set_dev(the_card.card, &dev->core);
+ ret = snd_card_register(the_card.card);
+ if (ret < 0)
+ goto clean_dma_map;
+
+ pr_info("%s started. start_delay=%dms\n",
+ the_card.card->longname, the_card.start_delay);
+ return 0;
+
+clean_dma_map:
+ dma_free_coherent(&the_card.ps3_dev->core,
+ PAGE_SIZE,
+ the_card.null_buffer_start_vaddr,
+ the_card.null_buffer_start_dma_addr);
+clean_preallocate:
+ snd_pcm_lib_preallocate_free_for_all(the_card.pcm);
+clean_card:
+ snd_card_free(the_card.card);
+clean_irq:
+ snd_ps3_free_irq();
+clean_dma_region:
+ ps3_dma_region_free(dev->d_region);
+clean_mmio:
+ snd_ps3_unmap_mmio();
+clean_dev_map:
+ lv1_gpu_device_unmap(2);
+clean_open:
+ ps3_close_hv_device(dev);
+ /*
+ * there is no destructor function to pcm.
+ * midlayer automatically releases if the card removed
+ */
+ return ret;
+}; /* snd_ps3_probe */
+
+/* called when module removal */
+static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev)
+{
+ int ret;
+ pr_info("%s:start id=%d\n", __func__, dev->match_id);
+ if (dev->match_id != PS3_MATCH_ID_SOUND)
+ return -ENXIO;
+
+ /*
+ * ctl and preallocate buffer will be freed in
+ * snd_card_free
+ */
+ ret = snd_card_free(the_card.card);
+ if (ret)
+ pr_info("%s: ctl freecard=%d\n", __func__, ret);
+
+ dma_free_coherent(&dev->core,
+ PAGE_SIZE,
+ the_card.null_buffer_start_vaddr,
+ the_card.null_buffer_start_dma_addr);
+
+ ps3_dma_region_free(dev->d_region);
+
+ snd_ps3_free_irq();
+ snd_ps3_unmap_mmio();
+
+ lv1_gpu_device_unmap(2);
+ ps3_close_hv_device(dev);
+ pr_info("%s:end id=%d\n", __func__, dev->match_id);
+ return 0;
+} /* snd_ps3_remove */
+
+static struct ps3_system_bus_driver snd_ps3_bus_driver_info = {
+ .match_id = PS3_MATCH_ID_SOUND,
+ .probe = snd_ps3_driver_probe,
+ .remove = snd_ps3_driver_remove,
+ .shutdown = snd_ps3_driver_remove,
+ .core = {
+ .name = SND_PS3_DRIVER_NAME,
+ .owner = THIS_MODULE,
+ },
+};
+
+
+/*
+ * Interrupt handler
+ */
+static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id)
+{
+
+ uint32_t port_intr;
+ int underflow_occured = 0;
+ struct snd_ps3_card_info *card = dev_id;
+
+ if (!card->running) {
+ update_reg(PS3_AUDIO_AX_IS, 0);
+ update_reg(PS3_AUDIO_INTR_0, 0);
+ return IRQ_HANDLED;
+ }
+
+ port_intr = read_reg(PS3_AUDIO_AX_IS);
+ /*
+ *serial buffer empty detected (every 4 times),
+ *program next dma and kick it
+ */
+ if (port_intr & PS3_AUDIO_AX_IE_ASOBEIE(0)) {
+ write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBEIE(0));
+ if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
+ write_reg(PS3_AUDIO_AX_IS, port_intr);
+ underflow_occured = 1;
+ }
+ if (card->silent) {
+ /* we are still in silent time */
+ snd_ps3_program_dma(card,
+ (underflow_occured) ?
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL :
+ SND_PS3_DMA_FILLTYPE_SILENT_RUNNING);
+ snd_ps3_kick_dma(card);
+ card->silent --;
+ } else {
+ snd_ps3_program_dma(card,
+ (underflow_occured) ?
+ SND_PS3_DMA_FILLTYPE_FIRSTFILL :
+ SND_PS3_DMA_FILLTYPE_RUNNING);
+ snd_ps3_kick_dma(card);
+ snd_pcm_period_elapsed(card->substream);
+ }
+ } else if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
+ write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBUIE(0));
+ /*
+ * serial out underflow, but buffer empty not detected.
+ * in this case, fill fifo with 0 to recover. After
+ * filling dummy data, serial automatically start to
+ * consume them and then will generate normal buffer
+ * empty interrupts.
+ * If both buffer underflow and buffer empty are occured,
+ * it is better to do nomal data transfer than empty one
+ */
+ snd_ps3_program_dma(card,
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
+ snd_ps3_kick_dma(card);
+ snd_ps3_program_dma(card,
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
+ snd_ps3_kick_dma(card);
+ }
+ /* clear interrupt cause */
+ return IRQ_HANDLED;
+};
+
+/*
+ * module/subsystem initialize/terminate
+ */
+static int __init snd_ps3_init(void)
+{
+ int ret;
+
+ if (!firmware_has_feature(FW_FEATURE_PS3_LV1))
+ return -ENXIO;
+
+ memset(&the_card, 0, sizeof(the_card));
+ spin_lock_init(&the_card.dma_lock);
+
+ /* register systembus DRIVER, this calls our probe() func */
+ ret = ps3_system_bus_driver_register(&snd_ps3_bus_driver_info);
+
+ return ret;
+}
+
+static void __exit snd_ps3_exit(void)
+{
+ ps3_system_bus_driver_unregister(&snd_ps3_bus_driver_info);
+}
+
+MODULE_ALIAS(PS3_MODULE_ALIAS_SOUND);
diff --git a/sound/ppc/snd_ps3.h b/sound/ppc/snd_ps3.h
new file mode 100644
index 0000000..326fb29
--- /dev/null
+++ b/sound/ppc/snd_ps3.h
@@ -0,0 +1,136 @@
+/*
+ * Audio support for PS3
+ * Copyright (C) 2007 Sony Computer Entertainment Inc.
+ * All rights reserved.
+ * Copyright 2006, 2007 Sony Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; version 2 of the Licence.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#if !defined(_SND_PS3_H_)
+#define _SND_PS3_H_
+
+#include <linux/irqreturn.h>
+
+#define SND_PS3_DRIVER_NAME "snd_ps3"
+
+enum snd_ps3_out_channel {
+ SND_PS3_OUT_SPDIF_0,
+ SND_PS3_OUT_SPDIF_1,
+ SND_PS3_OUT_SERIAL_0,
+ SND_PS3_OUT_DEVS
+};
+
+enum snd_ps3_dma_filltype {
+ SND_PS3_DMA_FILLTYPE_FIRSTFILL,
+ SND_PS3_DMA_FILLTYPE_RUNNING,
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL,
+ SND_PS3_DMA_FILLTYPE_SILENT_RUNNING
+};
+
+enum snd_ps3_ch {
+ SND_PS3_CH_L = 0,
+ SND_PS3_CH_R = 1,
+ SND_PS3_CH_MAX = 2
+};
+
+struct snd_ps3_avsetting_info {
+ uint32_t avs_audio_ch; /* fixed */
+ uint32_t avs_audio_rate;
+ uint32_t avs_audio_width;
+ uint32_t avs_audio_format; /* fixed */
+ uint32_t avs_audio_source; /* fixed */
+ unsigned char avs_cs_info[8];
+};
+/*
+ * PS3 audio 'card' instance
+ * there should be only ONE hardware.
+ */
+struct snd_ps3_card_info {
+ struct ps3_system_bus_device *ps3_dev;
+ struct snd_card *card;
+
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+
+ /* hvc info */
+ u64 audio_lpar_addr;
+ u64 audio_lpar_size;
+
+ /* registers */
+ void __iomem *mapped_mmio_vaddr;
+
+ /* irq */
+ u64 audio_irq_outlet;
+ unsigned int irq_no;
+
+ /* remember avsetting */
+ struct snd_ps3_avsetting_info avs;
+
+ /* dma buffer management */
+ spinlock_t dma_lock;
+ /* dma_lock start */
+ void * dma_start_vaddr[2]; /* 0 for L, 1 for R */
+ dma_addr_t dma_start_bus_addr[2];
+ size_t dma_buffer_size;
+ void * dma_last_transfer_vaddr[2];
+ void * dma_next_transfer_vaddr[2];
+ int silent;
+ /* dma_lock end */
+
+ int running;
+
+ /* null buffer */
+ void *null_buffer_start_vaddr;
+ dma_addr_t null_buffer_start_dma_addr;
+
+ /* start delay */
+ unsigned int start_delay;
+
+};
+
+
+/* PS3 audio DMAC block size in bytes */
+#define PS3_AUDIO_DMAC_BLOCK_SIZE (128)
+/* one stage (stereo) of audio FIFO in bytes */
+#define PS3_AUDIO_FIFO_STAGE_SIZE (256)
+/* how many stages the fifo have */
+#define PS3_AUDIO_FIFO_STAGE_COUNT (8)
+/* fifo size 128 bytes * 8 stages * stereo (2ch) */
+#define PS3_AUDIO_FIFO_SIZE \
+ (PS3_AUDIO_FIFO_STAGE_SIZE * PS3_AUDIO_FIFO_STAGE_COUNT)
+
+/* PS3 audio DMAC max block count in one dma shot = 128 (0x80) blocks*/
+#define PS3_AUDIO_DMAC_MAX_BLOCKS (PS3_AUDIO_DMASIZE_BLOCKS_MASK + 1)
+
+#define PS3_AUDIO_NORMAL_DMA_START_CH (0)
+#define PS3_AUDIO_NORMAL_DMA_COUNT (8)
+#define PS3_AUDIO_NULL_DMA_START_CH \
+ (PS3_AUDIO_NORMAL_DMA_START_CH + PS3_AUDIO_NORMAL_DMA_COUNT)
+#define PS3_AUDIO_NULL_DMA_COUNT (2)
+
+#define SND_PS3_MAX_VOL (0x0F)
+#define SND_PS3_MIN_VOL (0x00)
+#define SND_PS3_MIN_ATT SND_PS3_MIN_VOL
+#define SND_PS3_MAX_ATT SND_PS3_MAX_VOL
+
+#define SND_PS3_PCM_PREALLOC_SIZE \
+ (PS3_AUDIO_DMAC_BLOCK_SIZE * PS3_AUDIO_DMAC_MAX_BLOCKS * 4)
+
+#define SND_PS3_DMA_REGION_SIZE \
+ (SND_PS3_PCM_PREALLOC_SIZE + PAGE_SIZE)
+
+#define PS3_AUDIO_IOID (1UL)
+
+#endif /* _SND_PS3_H_ */
diff --git a/sound/ppc/snd_ps3_reg.h b/sound/ppc/snd_ps3_reg.h
new file mode 100644
index 0000000..03fdee4
--- /dev/null
+++ b/sound/ppc/snd_ps3_reg.h
@@ -0,0 +1,891 @@
+/*
+ * Audio support for PS3
+ * Copyright (C) 2007 Sony Computer Entertainment Inc.
+ * Copyright 2006, 2007 Sony Corporation
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/*
+ * interrupt / configure registers
+ */
+
+#define PS3_AUDIO_INTR_0 (0x00000100)
+#define PS3_AUDIO_INTR_EN_0 (0x00000140)
+#define PS3_AUDIO_CONFIG (0x00000200)
+
+/*
+ * DMAC registers
+ * n:0..9
+ */
+#define PS3_AUDIO_DMAC_REGBASE(x) (0x0000210 + 0x20 * (x))
+
+#define PS3_AUDIO_KICK(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x00)
+#define PS3_AUDIO_SOURCE(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x04)
+#define PS3_AUDIO_DEST(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x08)
+#define PS3_AUDIO_DMASIZE(n) (PS3_AUDIO_DMAC_REGBASE(n) + 0x0C)
+
+/*
+ * mute control
+ */
+#define PS3_AUDIO_AX_MCTRL (0x00004000)
+#define PS3_AUDIO_AX_ISBP (0x00004004)
+#define PS3_AUDIO_AX_AOBP (0x00004008)
+#define PS3_AUDIO_AX_IC (0x00004010)
+#define PS3_AUDIO_AX_IE (0x00004014)
+#define PS3_AUDIO_AX_IS (0x00004018)
+
+/*
+ * three wire serial
+ * n:0..3
+ */
+#define PS3_AUDIO_AO_MCTRL (0x00006000)
+#define PS3_AUDIO_AO_3WMCTRL (0x00006004)
+
+#define PS3_AUDIO_AO_3WCTRL(n) (0x00006200 + 0x200 * (n))
+
+/*
+ * S/PDIF
+ * n:0..1
+ * x:0..11
+ * y:0..5
+ */
+#define PS3_AUDIO_AO_SPD_REGBASE(n) (0x00007200 + 0x200 * (n))
+
+#define PS3_AUDIO_AO_SPDCTRL(n) \
+ (PS3_AUDIO_AO_SPD_REGBASE(n) + 0x00)
+#define PS3_AUDIO_AO_SPDUB(n, x) \
+ (PS3_AUDIO_AO_SPD_REGBASE(n) + 0x04 + 0x04 * (x))
+#define PS3_AUDIO_AO_SPDCS(n, y) \
+ (PS3_AUDIO_AO_SPD_REGBASE(n) + 0x34 + 0x04 * (y))
+
+
+/*
+ PS3_AUDIO_INTR_0 register tells an interrupt handler which audio
+ DMA channel triggered the interrupt. The interrupt status for a channel
+ can be cleared by writing a '1' to the corresponding bit. A new interrupt
+ cannot be generated until the previous interrupt has been cleared.
+
+ Note that the status reported by PS3_AUDIO_INTR_0 is independent of the
+ value of PS3_AUDIO_INTR_EN_0.
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0 0 0 0 0 0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C| INTR_0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+#define PS3_AUDIO_INTR_0_CHAN(n) (1 << ((n) * 2))
+#define PS3_AUDIO_INTR_0_CHAN9 PS3_AUDIO_INTR_0_CHAN(9)
+#define PS3_AUDIO_INTR_0_CHAN8 PS3_AUDIO_INTR_0_CHAN(8)
+#define PS3_AUDIO_INTR_0_CHAN7 PS3_AUDIO_INTR_0_CHAN(7)
+#define PS3_AUDIO_INTR_0_CHAN6 PS3_AUDIO_INTR_0_CHAN(6)
+#define PS3_AUDIO_INTR_0_CHAN5 PS3_AUDIO_INTR_0_CHAN(5)
+#define PS3_AUDIO_INTR_0_CHAN4 PS3_AUDIO_INTR_0_CHAN(4)
+#define PS3_AUDIO_INTR_0_CHAN3 PS3_AUDIO_INTR_0_CHAN(3)
+#define PS3_AUDIO_INTR_0_CHAN2 PS3_AUDIO_INTR_0_CHAN(2)
+#define PS3_AUDIO_INTR_0_CHAN1 PS3_AUDIO_INTR_0_CHAN(1)
+#define PS3_AUDIO_INTR_0_CHAN0 PS3_AUDIO_INTR_0_CHAN(0)
+
+/*
+ The PS3_AUDIO_INTR_EN_0 register specifies which DMA channels can generate
+ an interrupt to the PU. Each bit of PS3_AUDIO_INTR_EN_0 is ANDed with the
+ corresponding bit in PS3_AUDIO_INTR_0. The resulting bits are OR'd together
+ to generate the Audio interrupt.
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0 0 0 0 0 0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C|0|C| INTR_EN_0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+
+ Bit assignments are same as PS3_AUDIO_INTR_0
+*/
+
+/*
+ PS3_AUDIO_CONFIG
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0 0 0 0 C|0 0 0 0 0 0 0 0| CONFIG
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+
+*/
+
+/* The CLEAR field cancels all pending transfers, and stops any running DMA
+ transfers. Any interrupts associated with the canceled transfers
+ will occur as if the transfer had finished.
+ Since this bit is designed to recover from DMA related issues
+ which are caused by unpredictable situations, it is prefered to wait
+ for normal DMA transfer end without using this bit.
+*/
+#define PS3_AUDIO_CONFIG_CLEAR (1 << 8) /* RWIVF */
+
+/*
+ PS3_AUDIO_AX_MCTRL: Audio Port Mute Control Register
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|A|A|A|0 0 0 0 0 0 0|S|S|A|A|A|A| AX_MCTRL
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+
+/* 3 Wire Audio Serial Output Channel Mutes (0..3) */
+#define PS3_AUDIO_AX_MCTRL_ASOMT(n) (1 << (3 - (n))) /* RWIVF */
+#define PS3_AUDIO_AX_MCTRL_ASO3MT (1 << 0) /* RWIVF */
+#define PS3_AUDIO_AX_MCTRL_ASO2MT (1 << 1) /* RWIVF */
+#define PS3_AUDIO_AX_MCTRL_ASO1MT (1 << 2) /* RWIVF */
+#define PS3_AUDIO_AX_MCTRL_ASO0MT (1 << 3) /* RWIVF */
+
+/* S/PDIF mutes (0,1)*/
+#define PS3_AUDIO_AX_MCTRL_SPOMT(n) (1 << (5 - (n))) /* RWIVF */
+#define PS3_AUDIO_AX_MCTRL_SPO1MT (1 << 4) /* RWIVF */
+#define PS3_AUDIO_AX_MCTRL_SPO0MT (1 << 5) /* RWIVF */
+
+/* All 3 Wire Serial Outputs Mute */
+#define PS3_AUDIO_AX_MCTRL_AASOMT (1 << 13) /* RWIVF */
+
+/* All S/PDIF Mute */
+#define PS3_AUDIO_AX_MCTRL_ASPOMT (1 << 14) /* RWIVF */
+
+/* All Audio Outputs Mute */
+#define PS3_AUDIO_AX_MCTRL_AAOMT (1 << 15) /* RWIVF */
+
+/*
+ S/PDIF Outputs Buffer Read/Write Pointer Register
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0|0|SPO0B|0|SPO1B|0 0 0 0 0 0 0 0|0|SPO0B|0|SPO1B| AX_ISBP
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+
+*/
+/*
+ S/PDIF Output Channel Read Buffer Numbers
+ Buffer number is value of field.
+ Indicates current read access buffer ID from Audio Data
+ Transfer controller of S/PDIF Output
+*/
+
+#define PS3_AUDIO_AX_ISBP_SPOBRN_MASK(n) (0x7 << 4 * (1 - (n))) /* R-IUF */
+#define PS3_AUDIO_AX_ISBP_SPO1BRN_MASK (0x7 << 0) /* R-IUF */
+#define PS3_AUDIO_AX_ISBP_SPO0BRN_MASK (0x7 << 4) /* R-IUF */
+
+/*
+S/PDIF Output Channel Buffer Write Numbers
+Indicates current write access buffer ID from bus master.
+*/
+#define PS3_AUDIO_AX_ISBP_SPOBWN_MASK(n) (0x7 << 4 * (5 - (n))) /* R-IUF */
+#define PS3_AUDIO_AX_ISBP_SPO1BWN_MASK (0x7 << 16) /* R-IUF */
+#define PS3_AUDIO_AX_ISBP_SPO0BWN_MASK (0x7 << 20) /* R-IUF */
+
+/*
+ 3 Wire Audio Serial Outputs Buffer Read/Write
+ Pointer Register
+ Buffer number is value of field
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0|ASO0B|0|ASO1B|0|ASO2B|0|ASO3B|0|ASO0B|0|ASO1B|0|ASO2B|0|ASO3B| AX_AOBP
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+
+/*
+3 Wire Audio Serial Output Channel Buffer Read Numbers
+Indicates current read access buffer Id from Audio Data Transfer
+Controller of 3 Wire Audio Serial Output Channels
+*/
+#define PS3_AUDIO_AX_AOBP_ASOBRN_MASK(n) (0x7 << 4 * (3 - (n))) /* R-IUF */
+
+#define PS3_AUDIO_AX_AOBP_ASO3BRN_MASK (0x7 << 0) /* R-IUF */
+#define PS3_AUDIO_AX_AOBP_ASO2BRN_MASK (0x7 << 4) /* R-IUF */
+#define PS3_AUDIO_AX_AOBP_ASO1BRN_MASK (0x7 << 8) /* R-IUF */
+#define PS3_AUDIO_AX_AOBP_ASO0BRN_MASK (0x7 << 12) /* R-IUF */
+
+/*
+3 Wire Audio Serial Output Channel Buffer Write Numbers
+Indicates current write access buffer ID from bus master.
+*/
+#define PS3_AUDIO_AX_AOBP_ASOBWN_MASK(n) (0x7 << 4 * (7 - (n))) /* R-IUF */
+
+#define PS3_AUDIO_AX_AOBP_ASO3BWN_MASK (0x7 << 16) /* R-IUF */
+#define PS3_AUDIO_AX_AOBP_ASO2BWN_MASK (0x7 << 20) /* R-IUF */
+#define PS3_AUDIO_AX_AOBP_ASO1BWN_MASK (0x7 << 24) /* R-IUF */
+#define PS3_AUDIO_AX_AOBP_ASO0BWN_MASK (0x7 << 28) /* R-IUF */
+
+
+
+/*
+Audio Port Interrupt Condition Register
+For the fields in this register, the following values apply:
+0 = Interrupt is generated every interrupt event.
+1 = Interrupt is generated every 2 interrupt events.
+2 = Interrupt is generated every 4 interrupt events.
+3 = Reserved
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0|0 0|SPO|0 0|SPO|0 0|AAS|0 0 0 0 0 0 0 0 0 0 0 0| AX_IC
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+/*
+All 3-Wire Audio Serial Outputs Interrupt Mode
+Configures the Interrupt and Signal Notification
+condition of all 3-wire Audio Serial Outputs.
+*/
+#define PS3_AUDIO_AX_IC_AASOIMD_MASK (0x3 << 12) /* RWIVF */
+#define PS3_AUDIO_AX_IC_AASOIMD_EVERY1 (0x0 << 12) /* RWI-V */
+#define PS3_AUDIO_AX_IC_AASOIMD_EVERY2 (0x1 << 12) /* RW--V */
+#define PS3_AUDIO_AX_IC_AASOIMD_EVERY4 (0x2 << 12) /* RW--V */
+
+/*
+S/PDIF Output Channel Interrupt Modes
+Configures the Interrupt and signal Notification
+conditions of S/PDIF output channels.
+*/
+#define PS3_AUDIO_AX_IC_SPO1IMD_MASK (0x3 << 16) /* RWIVF */
+#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY1 (0x0 << 16) /* RWI-V */
+#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY2 (0x1 << 16) /* RW--V */
+#define PS3_AUDIO_AX_IC_SPO1IMD_EVERY4 (0x2 << 16) /* RW--V */
+
+#define PS3_AUDIO_AX_IC_SPO0IMD_MASK (0x3 << 20) /* RWIVF */
+#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY1 (0x0 << 20) /* RWI-V */
+#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY2 (0x1 << 20) /* RW--V */
+#define PS3_AUDIO_AX_IC_SPO0IMD_EVERY4 (0x2 << 20) /* RW--V */
+
+/*
+Audio Port interrupt Enable Register
+Configures whether to enable or disable each Interrupt Generation.
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0|S|S|0 0|A|A|A|A|0 0 0 0|S|S|0 0|S|S|0 0|A|A|A|A| AX_IE
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+
+*/
+
+/*
+3 Wire Audio Serial Output Channel Buffer Underflow
+Interrupt Enables
+Select enable/disable of Buffer Underflow Interrupts for
+3-Wire Audio Serial Output Channels
+DISABLED=Interrupt generation disabled.
+*/
+#define PS3_AUDIO_AX_IE_ASOBUIE(n) (1 << (3 - (n))) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO3BUIE (1 << 0) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO2BUIE (1 << 1) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO1BUIE (1 << 2) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO0BUIE (1 << 3) /* RWIVF */
+
+/* S/PDIF Output Channel Buffer Underflow Interrupt Enables */
+
+#define PS3_AUDIO_AX_IE_SPOBUIE(n) (1 << (7 - (n))) /* RWIVF */
+#define PS3_AUDIO_AX_IE_SPO1BUIE (1 << 6) /* RWIVF */
+#define PS3_AUDIO_AX_IE_SPO0BUIE (1 << 7) /* RWIVF */
+
+/* S/PDIF Output Channel One Block Transfer Completion Interrupt Enables */
+
+#define PS3_AUDIO_AX_IE_SPOBTCIE(n) (1 << (11 - (n))) /* RWIVF */
+#define PS3_AUDIO_AX_IE_SPO1BTCIE (1 << 10) /* RWIVF */
+#define PS3_AUDIO_AX_IE_SPO0BTCIE (1 << 11) /* RWIVF */
+
+/* 3-Wire Audio Serial Output Channel Buffer Empty Interrupt Enables */
+
+#define PS3_AUDIO_AX_IE_ASOBEIE(n) (1 << (19 - (n))) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO3BEIE (1 << 16) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO2BEIE (1 << 17) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO1BEIE (1 << 18) /* RWIVF */
+#define PS3_AUDIO_AX_IE_ASO0BEIE (1 << 19) /* RWIVF */
+
+/* S/PDIF Output Channel Buffer Empty Interrupt Enables */
+
+#define PS3_AUDIO_AX_IE_SPOBEIE(n) (1 << (23 - (n))) /* RWIVF */
+#define PS3_AUDIO_AX_IE_SPO1BEIE (1 << 22) /* RWIVF */
+#define PS3_AUDIO_AX_IE_SPO0BEIE (1 << 23) /* RWIVF */
+
+/*
+Audio Port Interrupt Status Register
+Indicates Interrupt status, which interrupt has occured, and can clear
+each interrupt in this register.
+Writing 1b to a field containing 1b clears field and de-asserts interrupt.
+Writing 0b to a field has no effect.
+Field vaules are the following:
+0 - Interrupt hasn't occured.
+1 - Interrupt has occured.
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0|S|S|0 0|A|A|A|A|0 0 0 0|S|S|0 0|S|S|0 0|A|A|A|A| AX_IS
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+
+ Bit assignment are same as AX_IE
+*/
+
+/*
+Audio Output Master Control Register
+Configures Master Clock and other master Audio Output Settings
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0|SCKSE|0|SCKSE| MR0 | MR1 |MCL|MCL|0 0 0 0|0 0 0 0 0 0 0 0| AO_MCTRL
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+
+/*
+MCLK Output Control
+Controls mclko[1] output.
+0 - Disable output (fixed at High)
+1 - Output clock produced by clock selected
+with scksel1 by mr1
+2 - Reserved
+3 - Reserved
+*/
+
+#define PS3_AUDIO_AO_MCTRL_MCLKC1_MASK (0x3 << 12) /* RWIVF */
+#define PS3_AUDIO_AO_MCTRL_MCLKC1_DISABLED (0x0 << 12) /* RWI-V */
+#define PS3_AUDIO_AO_MCTRL_MCLKC1_ENABLED (0x1 << 12) /* RW--V */
+#define PS3_AUDIO_AO_MCTRL_MCLKC1_RESVD2 (0x2 << 12) /* RW--V */
+#define PS3_AUDIO_AO_MCTRL_MCLKC1_RESVD3 (0x3 << 12) /* RW--V */
+
+/*
+MCLK Output Control
+Controls mclko[0] output.
+0 - Disable output (fixed at High)
+1 - Output clock produced by clock selected
+with SCKSEL0 by MR0
+2 - Reserved
+3 - Reserved
+*/
+#define PS3_AUDIO_AO_MCTRL_MCLKC0_MASK (0x3 << 14) /* RWIVF */
+#define PS3_AUDIO_AO_MCTRL_MCLKC0_DISABLED (0x0 << 14) /* RWI-V */
+#define PS3_AUDIO_AO_MCTRL_MCLKC0_ENABLED (0x1 << 14) /* RW--V */
+#define PS3_AUDIO_AO_MCTRL_MCLKC0_RESVD2 (0x2 << 14) /* RW--V */
+#define PS3_AUDIO_AO_MCTRL_MCLKC0_RESVD3 (0x3 << 14) /* RW--V */
+/*
+Master Clock Rate 1
+Sets the divide ration of Master Clock1 (clock output from
+mclko[1] for the input clock selected by scksel1.
+*/
+#define PS3_AUDIO_AO_MCTRL_MR1_MASK (0xf << 16)
+#define PS3_AUDIO_AO_MCTRL_MR1_DEFAULT (0x0 << 16) /* RWI-V */
+/*
+Master Clock Rate 0
+Sets the divide ratio of Master Clock0 (clock output from
+mclko[0] for the input clock selected by scksel0).
+*/
+#define PS3_AUDIO_AO_MCTRL_MR0_MASK (0xf << 20) /* RWIVF */
+#define PS3_AUDIO_AO_MCTRL_MR0_DEFAULT (0x0 << 20) /* RWI-V */
+/*
+System Clock Select 0/1
+Selects the system clock to be used as Master Clock 0/1
+Input the system clock that is appropriate for the sampling
+rate.
+*/
+#define PS3_AUDIO_AO_MCTRL_SCKSEL1_MASK (0x7 << 24) /* RWIVF */
+#define PS3_AUDIO_AO_MCTRL_SCKSEL1_DEFAULT (0x2 << 24) /* RWI-V */
+
+#define PS3_AUDIO_AO_MCTRL_SCKSEL0_MASK (0x7 << 28) /* RWIVF */
+#define PS3_AUDIO_AO_MCTRL_SCKSEL0_DEFAULT (0x2 << 28) /* RWI-V */
+
+
+/*
+3-Wire Audio Output Master Control Register
+Configures clock, 3-Wire Audio Serial Output Enable, and
+other 3-Wire Audio Serial Output Master Settings
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |A|A|A|A|0 0 0|A| ASOSR |0 0 0 0|A|A|A|A|A|A|0|1|0 0 0 0 0 0 0 0| AO_3WMCTRL
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+
+
+/*
+LRCKO Polarity
+0 - Reserved
+1 - default
+*/
+#define PS3_AUDIO_AO_3WMCTRL_ASOPLRCK (1 << 8) /* RWIVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT (1 << 8) /* RW--V */
+
+/* LRCK Output Disable */
+
+#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD (1 << 10) /* RWIVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_ENABLED (0 << 10) /* RW--V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED (1 << 10) /* RWI-V */
+
+/* Bit Clock Output Disable */
+
+#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD (1 << 11) /* RWIVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_ENABLED (0 << 11) /* RW--V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED (1 << 11) /* RWI-V */
+
+/*
+3-Wire Audio Serial Output Channel 0-3 Operational
+Status. Each bit becomes 1 after each 3-Wire Audio
+Serial Output Channel N is in action by setting 1 to
+asoen.
+Each bit becomes 0 after each 3-Wire Audio Serial Output
+Channel N is out of action by setting 0 to asoen.
+*/
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN(n) (1 << (15 - (n))) /* R-IVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(n) (0 << (15 - (n))) /* R-I-V */
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(n) (1 << (15 - (n))) /* R---V */
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN0 \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(0)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN0_STOPPED \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(0)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN0_RUNNING \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(0)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN1 \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(1)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN1_STOPPED \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(1)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN1_RUNNING \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(1)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN2 \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(2)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN2_STOPPED \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(2)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN2_RUNNING \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(2)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN3 \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(3)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN3_STOPPED \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_STOPPED(3)
+#define PS3_AUDIO_AO_3WMCTRL_ASORUN3_RUNNING \
+ PS3_AUDIO_AO_3WMCTRL_ASORUN_RUNNING(3)
+
+/*
+Sampling Rate
+Specifies the divide ratio of the bit clock (clock output
+from bclko) used by the 3-wire Audio Output Clock, whcih
+is applied to the master clock selected by mcksel.
+Data output is synchronized with this clock.
+*/
+#define PS3_AUDIO_AO_3WMCTRL_ASOSR_MASK (0xf << 20) /* RWIVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV2 (0x1 << 20) /* RWI-V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV4 (0x2 << 20) /* RW--V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV8 (0x4 << 20) /* RW--V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOSR_DIV12 (0x6 << 20) /* RW--V */
+
+/*
+Master Clock Select
+0 - Master Clock 0
+1 - Master Clock 1
+*/
+#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL (1 << 24) /* RWIVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL_CLK0 (0 << 24) /* RWI-V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOMCKSEL_CLK1 (1 << 24) /* RW--V */
+
+/*
+Enables and disables 4ch 3-Wire Audio Serial Output
+operation. Each Bit from 0 to 3 corresponds to an
+output channel, which means that each output channel
+can be enabled or disabled individually. When
+multiple channels are enabled at the same time, output
+operations are performed in synchronization.
+Bit 0 - Output Channel 0 (SDOUT[0])
+Bit 1 - Output Channel 1 (SDOUT[1])
+Bit 2 - Output Channel 2 (SDOUT[2])
+Bit 3 - Output Channel 3 (SDOUT[3])
+*/
+#define PS3_AUDIO_AO_3WMCTRL_ASOEN(n) (1 << (31 - (n))) /* RWIVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(n) (0 << (31 - (n))) /* RWI-V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(n) (1 << (31 - (n))) /* RW--V */
+
+#define PS3_AUDIO_AO_3WMCTRL_ASOEN0 \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(0) /* RWIVF */
+#define PS3_AUDIO_AO_3WMCTRL_ASOEN0_DISABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(0) /* RWI-V */
+#define PS3_AUDIO_AO_3WMCTRL_ASOEN0_ENABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(0) /* RW--V */
+#define PS3_AUDIO_A1_3WMCTRL_ASOEN0 \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(1) /* RWIVF */
+#define PS3_AUDIO_A1_3WMCTRL_ASOEN0_DISABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(1) /* RWI-V */
+#define PS3_AUDIO_A1_3WMCTRL_ASOEN0_ENABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(1) /* RW--V */
+#define PS3_AUDIO_A2_3WMCTRL_ASOEN0 \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(2) /* RWIVF */
+#define PS3_AUDIO_A2_3WMCTRL_ASOEN0_DISABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(2) /* RWI-V */
+#define PS3_AUDIO_A2_3WMCTRL_ASOEN0_ENABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(2) /* RW--V */
+#define PS3_AUDIO_A3_3WMCTRL_ASOEN0 \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(3) /* RWIVF */
+#define PS3_AUDIO_A3_3WMCTRL_ASOEN0_DISABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_DISABLED(3) /* RWI-V */
+#define PS3_AUDIO_A3_3WMCTRL_ASOEN0_ENABLED \
+ PS3_AUDIO_AO_3WMCTRL_ASOEN_ENABLED(3) /* RW--V */
+
+/*
+3-Wire Audio Serial output Channel 0-3 Control Register
+Configures settings for 3-Wire Serial Audio Output Channel 0-3
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|A|0 0 0 0|A|0|ASO|0 0 0|0|0|0|0|0| AO_3WCTRL
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+
+*/
+/*
+Data Bit Mode
+Specifies the number of data bits
+0 - 16 bits
+1 - reserved
+2 - 20 bits
+3 - 24 bits
+*/
+#define PS3_AUDIO_AO_3WCTRL_ASODB_MASK (0x3 << 8) /* RWIVF */
+#define PS3_AUDIO_AO_3WCTRL_ASODB_16BIT (0x0 << 8) /* RWI-V */
+#define PS3_AUDIO_AO_3WCTRL_ASODB_RESVD (0x1 << 8) /* RWI-V */
+#define PS3_AUDIO_AO_3WCTRL_ASODB_20BIT (0x2 << 8) /* RW--V */
+#define PS3_AUDIO_AO_3WCTRL_ASODB_24BIT (0x3 << 8) /* RW--V */
+/*
+Data Format Mode
+Specifies the data format where (LSB side or MSB) the data(in 20 bit
+or 24 bit resolution mode) is put in a 32 bit field.
+0 - Data put on LSB side
+1 - Data put on MSB side
+*/
+#define PS3_AUDIO_AO_3WCTRL_ASODF (1 << 11) /* RWIVF */
+#define PS3_AUDIO_AO_3WCTRL_ASODF_LSB (0 << 11) /* RWI-V */
+#define PS3_AUDIO_AO_3WCTRL_ASODF_MSB (1 << 11) /* RW--V */
+/*
+Buffer Reset
+Performs buffer reset. Writing 1 to this bit initializes the
+corresponding 3-Wire Audio Output buffers(both L and R).
+*/
+#define PS3_AUDIO_AO_3WCTRL_ASOBRST (1 << 16) /* CWIVF */
+#define PS3_AUDIO_AO_3WCTRL_ASOBRST_IDLE (0 << 16) /* -WI-V */
+#define PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET (1 << 16) /* -W--T */
+
+/*
+S/PDIF Audio Output Channel 0/1 Control Register
+Configures settings for S/PDIF Audio Output Channel 0/1.
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |S|0 0 0|S|0 0|S| SPOSR |0 0|SPO|0 0 0 0|S|0|SPO|0 0 0 0 0 0 0|S| AO_SPDCTRL
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+/*
+Buffer reset. Writing 1 to this bit initializes the
+corresponding S/PDIF output buffer pointer.
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPOBRST (1 << 0) /* CWIVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPOBRST_IDLE (0 << 0) /* -WI-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPOBRST_RESET (1 << 0) /* -W--T */
+
+/*
+Data Bit Mode
+Specifies number of data bits
+0 - 16 bits
+1 - Reserved
+2 - 20 bits
+3 - 24 bits
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPODB_MASK (0x3 << 8) /* RWIVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPODB_16BIT (0x0 << 8) /* RWI-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPODB_RESVD (0x1 << 8) /* RW--V */
+#define PS3_AUDIO_AO_SPDCTRL_SPODB_20BIT (0x2 << 8) /* RW--V */
+#define PS3_AUDIO_AO_SPDCTRL_SPODB_24BIT (0x3 << 8) /* RW--V */
+/*
+Data format Mode
+Specifies the data format, where (LSB side or MSB)
+the data(in 20 or 24 bit resolution) is put in the
+32 bit field.
+0 - LSB Side
+1 - MSB Side
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPODF (1 << 11) /* RWIVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPODF_LSB (0 << 11) /* RWI-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPODF_MSB (1 << 11) /* RW--V */
+/*
+Source Select
+Specifies the source of the S/PDIF output. When 0, output
+operation is controlled by 3wen[0] of AO_3WMCTRL register.
+The SR must have the same setting as the a0_3wmctrl reg.
+0 - 3-Wire Audio OUT Ch0 Buffer
+1 - S/PDIF buffer
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPOSS_MASK (0x3 << 16) /* RWIVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPOSS_3WEN (0x0 << 16) /* RWI-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPOSS_SPDIF (0x1 << 16) /* RW--V */
+/*
+Sampling Rate
+Specifies the divide ratio of the bit clock (clock output
+from bclko) used by the S/PDIF Output Clock, which
+is applied to the master clock selected by mcksel.
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPOSR (0xf << 20) /* RWIVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV2 (0x1 << 20) /* RWI-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV4 (0x2 << 20) /* RW--V */
+#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV8 (0x4 << 20) /* RW--V */
+#define PS3_AUDIO_AO_SPDCTRL_SPOSR_DIV12 (0x6 << 20) /* RW--V */
+/*
+Master Clock Select
+0 - Master Clock 0
+1 - Master Clock 1
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL (1 << 24) /* RWIVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL_CLK0 (0 << 24) /* RWI-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPOMCKSEL_CLK1 (1 << 24) /* RW--V */
+
+/*
+S/PDIF Output Channel Operational Status
+This bit becomes 1 after S/PDIF Output Channel is in
+action by setting 1 to spoen. This bit becomes 0
+after S/PDIF Output Channel is out of action by setting
+0 to spoen.
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPORUN (1 << 27) /* R-IVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPORUN_STOPPED (0 << 27) /* R-I-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPORUN_RUNNING (1 << 27) /* R---V */
+
+/*
+S/PDIF Audio Output Channel Output Enable
+Enables and disables output operation. This bit is used
+only when sposs = 1
+*/
+#define PS3_AUDIO_AO_SPDCTRL_SPOEN (1 << 31) /* RWIVF */
+#define PS3_AUDIO_AO_SPDCTRL_SPOEN_DISABLED (0 << 31) /* RWI-V */
+#define PS3_AUDIO_AO_SPDCTRL_SPOEN_ENABLED (1 << 31) /* RW--V */
+
+/*
+S/PDIF Audio Output Channel Channel Status
+Setting Registers.
+Configures channel status bit settings for each block
+(192 bits).
+Output is performed from the MSB(AO_SPDCS0 register bit 31).
+The same value is added for subframes within the same frame.
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ | SPOCS | AO_SPDCS
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+
+S/PDIF Audio Output Channel User Bit Setting
+Configures user bit settings for each block (384 bits).
+Output is performed from the MSB(ao_spdub0 register bit 31).
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ | SPOUB | AO_SPDUB
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+/*****************************************************************************
+ *
+ * DMAC register
+ *
+ *****************************************************************************/
+/*
+The PS3_AUDIO_KICK register is used to initiate a DMA transfer and monitor
+its status
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0|STATU|0 0 0| EVENT |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|R| KICK
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+/*
+The REQUEST field is written to ACTIVE to initiate a DMA request when EVENT
+occurs.
+It will return to the DONE state when the request is completed.
+The registers for a DMA channel should only be written if REQUEST is IDLE.
+*/
+
+#define PS3_AUDIO_KICK_REQUEST (1 << 0) /* RWIVF */
+#define PS3_AUDIO_KICK_REQUEST_IDLE (0 << 0) /* RWI-V */
+#define PS3_AUDIO_KICK_REQUEST_ACTIVE (1 << 0) /* -W--T */
+
+/*
+ *The EVENT field is used to set the event in which
+ *the DMA request becomes active.
+ */
+#define PS3_AUDIO_KICK_EVENT_MASK (0x1f << 16) /* RWIVF */
+#define PS3_AUDIO_KICK_EVENT_ALWAYS (0x00 << 16) /* RWI-V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY (0x01 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT0_UNDERFLOW (0x02 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT1_EMPTY (0x03 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT1_UNDERFLOW (0x04 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT2_EMPTY (0x05 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT2_UNDERFLOW (0x06 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT3_EMPTY (0x07 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SERIALOUT3_UNDERFLOW (0x08 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SPDIF0_BLOCKTRANSFERCOMPLETE \
+ (0x09 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SPDIF0_UNDERFLOW (0x0A << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SPDIF0_EMPTY (0x0B << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SPDIF1_BLOCKTRANSFERCOMPLETE \
+ (0x0C << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SPDIF1_UNDERFLOW (0x0D << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_SPDIF1_EMPTY (0x0E << 16) /* RW--V */
+
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA(n) \
+ ((0x13 + (n)) << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA0 (0x13 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA1 (0x14 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA2 (0x15 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA3 (0x16 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA4 (0x17 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA5 (0x18 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA6 (0x19 << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA7 (0x1A << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA8 (0x1B << 16) /* RW--V */
+#define PS3_AUDIO_KICK_EVENT_AUDIO_DMA9 (0x1C << 16) /* RW--V */
+
+/*
+The STATUS field can be used to monitor the progress of a DMA request.
+DONE indicates the previous request has completed.
+EVENT indicates that the DMA engine is waiting for the EVENT to occur.
+PENDING indicates that the DMA engine has not started processing this
+request, but the EVENT has occured.
+DMA indicates that the data transfer is in progress.
+NOTIFY indicates that the notifier signalling end of transfer is being written.
+CLEAR indicated that the previous transfer was cleared.
+ERROR indicates the previous transfer requested an unsupported
+source/destination combination.
+*/
+
+#define PS3_AUDIO_KICK_STATUS_MASK (0x7 << 24) /* R-IVF */
+#define PS3_AUDIO_KICK_STATUS_DONE (0x0 << 24) /* R-I-V */
+#define PS3_AUDIO_KICK_STATUS_EVENT (0x1 << 24) /* R---V */
+#define PS3_AUDIO_KICK_STATUS_PENDING (0x2 << 24) /* R---V */
+#define PS3_AUDIO_KICK_STATUS_DMA (0x3 << 24) /* R---V */
+#define PS3_AUDIO_KICK_STATUS_NOTIFY (0x4 << 24) /* R---V */
+#define PS3_AUDIO_KICK_STATUS_CLEAR (0x5 << 24) /* R---V */
+#define PS3_AUDIO_KICK_STATUS_ERROR (0x6 << 24) /* R---V */
+
+/*
+The PS3_AUDIO_SOURCE register specifies the source address for transfers.
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ | START |0 0 0 0 0|TAR| SOURCE
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+
+/*
+The Audio DMA engine uses 128-byte transfers, thus the address must be aligned
+to a 128 byte boundary. The low seven bits are assumed to be 0.
+*/
+
+#define PS3_AUDIO_SOURCE_START_MASK (0x01FFFFFF << 7) /* RWIUF */
+
+/*
+The TARGET field specifies the memory space containing the source address.
+*/
+
+#define PS3_AUDIO_SOURCE_TARGET_MASK (3 << 0) /* RWIVF */
+#define PS3_AUDIO_SOURCE_TARGET_SYSTEM_MEMORY (2 << 0) /* RW--V */
+
+/*
+The PS3_AUDIO_DEST register specifies the destination address for transfers.
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ | START |0 0 0 0 0|TAR| DEST
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+
+/*
+The Audio DMA engine uses 128-byte transfers, thus the address must be aligned
+to a 128 byte boundary. The low seven bits are assumed to be 0.
+*/
+
+#define PS3_AUDIO_DEST_START_MASK (0x01FFFFFF << 7) /* RWIUF */
+
+/*
+The TARGET field specifies the memory space containing the destination address
+AUDIOFIFO = Audio WriteData FIFO,
+*/
+
+#define PS3_AUDIO_DEST_TARGET_MASK (3 << 0) /* RWIVF */
+#define PS3_AUDIO_DEST_TARGET_AUDIOFIFO (1 << 0) /* RW--V */
+
+/*
+PS3_AUDIO_DMASIZE specifies the number of 128-byte blocks + 1 to transfer.
+So a value of 0 means 128-bytes will get transfered.
+
+
+ 31 24 23 16 15 8 7 0
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+ |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0| BLOCKS | DMASIZE
+ +-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-|-+-+-+-+-+-+-+-+
+*/
+
+
+#define PS3_AUDIO_DMASIZE_BLOCKS_MASK (0x7f << 0) /* RWIUF */
+
+/*
+ * source/destination address for internal fifos
+ */
+#define PS3_AUDIO_AO_3W_LDATA(n) (0x1000 + (0x100 * (n)))
+#define PS3_AUDIO_AO_3W_RDATA(n) (0x1080 + (0x100 * (n)))
+
+#define PS3_AUDIO_AO_SPD_DATA(n) (0x2000 + (0x400 * (n)))
+
+
+/*
+ * field attiribute
+ *
+ * Read
+ * ' ' = Other Information
+ * '-' = Field is part of a write-only register
+ * 'C' = Value read is always the same, constant value line follows (C)
+ * 'R' = Value is read
+ *
+ * Write
+ * ' ' = Other Information
+ * '-' = Must not be written (D), value ignored when written (R,A,F)
+ * 'W' = Can be written
+ *
+ * Internal State
+ * ' ' = Other Information
+ * '-' = No internal state
+ * 'X' = Internal state, initial value is unknown
+ * 'I' = Internal state, initial value is known and follows (I)
+ *
+ * Declaration/Size
+ * ' ' = Other Information
+ * '-' = Does Not Apply
+ * 'V' = Type is void
+ * 'U' = Type is unsigned integer
+ * 'S' = Type is signed integer
+ * 'F' = Type is IEEE floating point
+ * '1' = Byte size (008)
+ * '2' = Short size (016)
+ * '3' = Three byte size (024)
+ * '4' = Word size (032)
+ * '8' = Double size (064)
+ *
+ * Define Indicator
+ * ' ' = Other Information
+ * 'D' = Device
+ * 'M' = Memory
+ * 'R' = Register
+ * 'A' = Array of Registers
+ * 'F' = Field
+ * 'V' = Value
+ * 'T' = Task
+ */
+
diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c
new file mode 100644
index 0000000..f746e15
--- /dev/null
+++ b/sound/ppc/tumbler.c
@@ -0,0 +1,1483 @@
+/*
+ * PMac Tumbler/Snapper lowlevel functions
+ *
+ * Copyright (c) by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * Rene Rebe <rene.rebe@gmx.net>:
+ * * update from shadow registers on wakeup and headphone plug
+ * * automatically toggle DRC on headphone plug
+ *
+ */
+
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <linux/kmod.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <sound/core.h>
+#include <asm/io.h>
+#include <asm/irq.h>
+#include <asm/machdep.h>
+#include <asm/pmac_feature.h>
+#include "pmac.h"
+#include "tumbler_volume.h"
+
+#undef DEBUG
+
+#ifdef DEBUG
+#define DBG(fmt...) printk(fmt)
+#else
+#define DBG(fmt...)
+#endif
+
+/* i2c address for tumbler */
+#define TAS_I2C_ADDR 0x34
+
+/* registers */
+#define TAS_REG_MCS 0x01 /* main control */
+#define TAS_REG_DRC 0x02
+#define TAS_REG_VOL 0x04
+#define TAS_REG_TREBLE 0x05
+#define TAS_REG_BASS 0x06
+#define TAS_REG_INPUT1 0x07
+#define TAS_REG_INPUT2 0x08
+
+/* tas3001c */
+#define TAS_REG_PCM TAS_REG_INPUT1
+
+/* tas3004 */
+#define TAS_REG_LMIX TAS_REG_INPUT1
+#define TAS_REG_RMIX TAS_REG_INPUT2
+#define TAS_REG_MCS2 0x43 /* main control 2 */
+#define TAS_REG_ACS 0x40 /* analog control */
+
+/* mono volumes for tas3001c/tas3004 */
+enum {
+ VOL_IDX_PCM_MONO, /* tas3001c only */
+ VOL_IDX_BASS, VOL_IDX_TREBLE,
+ VOL_IDX_LAST_MONO
+};
+
+/* stereo volumes for tas3004 */
+enum {
+ VOL_IDX_PCM, VOL_IDX_PCM2, VOL_IDX_ADC,
+ VOL_IDX_LAST_MIX
+};
+
+struct pmac_gpio {
+ unsigned int addr;
+ u8 active_val;
+ u8 inactive_val;
+ u8 active_state;
+};
+
+struct pmac_tumbler {
+ struct pmac_keywest i2c;
+ struct pmac_gpio audio_reset;
+ struct pmac_gpio amp_mute;
+ struct pmac_gpio line_mute;
+ struct pmac_gpio line_detect;
+ struct pmac_gpio hp_mute;
+ struct pmac_gpio hp_detect;
+ int headphone_irq;
+ int lineout_irq;
+ unsigned int save_master_vol[2];
+ unsigned int master_vol[2];
+ unsigned int save_master_switch[2];
+ unsigned int master_switch[2];
+ unsigned int mono_vol[VOL_IDX_LAST_MONO];
+ unsigned int mix_vol[VOL_IDX_LAST_MIX][2]; /* stereo volumes for tas3004 */
+ int drc_range;
+ int drc_enable;
+ int capture_source;
+ int anded_reset;
+ int auto_mute_notify;
+ int reset_on_sleep;
+ u8 acs;
+};
+
+
+/*
+ */
+
+static int send_init_client(struct pmac_keywest *i2c, unsigned int *regs)
+{
+ while (*regs > 0) {
+ int err, count = 10;
+ do {
+ err = i2c_smbus_write_byte_data(i2c->client,
+ regs[0], regs[1]);
+ if (err >= 0)
+ break;
+ DBG("(W) i2c error %d\n", err);
+ mdelay(10);
+ } while (count--);
+ if (err < 0)
+ return -ENXIO;
+ regs += 2;
+ }
+ return 0;
+}
+
+
+static int tumbler_init_client(struct pmac_keywest *i2c)
+{
+ static unsigned int regs[] = {
+ /* normal operation, SCLK=64fps, i2s output, i2s input, 16bit width */
+ TAS_REG_MCS, (1<<6)|(2<<4)|(2<<2)|0,
+ 0, /* terminator */
+ };
+ DBG("(I) tumbler init client\n");
+ return send_init_client(i2c, regs);
+}
+
+static int snapper_init_client(struct pmac_keywest *i2c)
+{
+ static unsigned int regs[] = {
+ /* normal operation, SCLK=64fps, i2s output, 16bit width */
+ TAS_REG_MCS, (1<<6)|(2<<4)|0,
+ /* normal operation, all-pass mode */
+ TAS_REG_MCS2, (1<<1),
+ /* normal output, no deemphasis, A input, power-up, line-in */
+ TAS_REG_ACS, 0,
+ 0, /* terminator */
+ };
+ DBG("(I) snapper init client\n");
+ return send_init_client(i2c, regs);
+}
+
+/*
+ * gpio access
+ */
+#define do_gpio_write(gp, val) \
+ pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, (gp)->addr, val)
+#define do_gpio_read(gp) \
+ pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, (gp)->addr, 0)
+#define tumbler_gpio_free(gp) /* NOP */
+
+static void write_audio_gpio(struct pmac_gpio *gp, int active)
+{
+ if (! gp->addr)
+ return;
+ active = active ? gp->active_val : gp->inactive_val;
+ do_gpio_write(gp, active);
+ DBG("(I) gpio %x write %d\n", gp->addr, active);
+}
+
+static int check_audio_gpio(struct pmac_gpio *gp)
+{
+ int ret;
+
+ if (! gp->addr)
+ return 0;
+
+ ret = do_gpio_read(gp);
+
+ return (ret & 0x1) == (gp->active_val & 0x1);
+}
+
+static int read_audio_gpio(struct pmac_gpio *gp)
+{
+ int ret;
+ if (! gp->addr)
+ return 0;
+ ret = do_gpio_read(gp);
+ ret = (ret & 0x02) !=0;
+ return ret == gp->active_state;
+}
+
+/*
+ * update master volume
+ */
+static int tumbler_set_master_volume(struct pmac_tumbler *mix)
+{
+ unsigned char block[6];
+ unsigned int left_vol, right_vol;
+
+ if (! mix->i2c.client)
+ return -ENODEV;
+
+ if (! mix->master_switch[0])
+ left_vol = 0;
+ else {
+ left_vol = mix->master_vol[0];
+ if (left_vol >= ARRAY_SIZE(master_volume_table))
+ left_vol = ARRAY_SIZE(master_volume_table) - 1;
+ left_vol = master_volume_table[left_vol];
+ }
+ if (! mix->master_switch[1])
+ right_vol = 0;
+ else {
+ right_vol = mix->master_vol[1];
+ if (right_vol >= ARRAY_SIZE(master_volume_table))
+ right_vol = ARRAY_SIZE(master_volume_table) - 1;
+ right_vol = master_volume_table[right_vol];
+ }
+
+ block[0] = (left_vol >> 16) & 0xff;
+ block[1] = (left_vol >> 8) & 0xff;
+ block[2] = (left_vol >> 0) & 0xff;
+
+ block[3] = (right_vol >> 16) & 0xff;
+ block[4] = (right_vol >> 8) & 0xff;
+ block[5] = (right_vol >> 0) & 0xff;
+
+ if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_VOL, 6,
+ block) < 0) {
+ snd_printk("failed to set volume \n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/* output volume */
+static int tumbler_info_master_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = ARRAY_SIZE(master_volume_table) - 1;
+ return 0;
+}
+
+static int tumbler_get_master_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix = chip->mixer_data;
+
+ ucontrol->value.integer.value[0] = mix->master_vol[0];
+ ucontrol->value.integer.value[1] = mix->master_vol[1];
+ return 0;
+}
+
+static int tumbler_put_master_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix = chip->mixer_data;
+ unsigned int vol[2];
+ int change;
+
+ vol[0] = ucontrol->value.integer.value[0];
+ vol[1] = ucontrol->value.integer.value[1];
+ if (vol[0] >= ARRAY_SIZE(master_volume_table) ||
+ vol[1] >= ARRAY_SIZE(master_volume_table))
+ return -EINVAL;
+ change = mix->master_vol[0] != vol[0] ||
+ mix->master_vol[1] != vol[1];
+ if (change) {
+ mix->master_vol[0] = vol[0];
+ mix->master_vol[1] = vol[1];
+ tumbler_set_master_volume(mix);
+ }
+ return change;
+}
+
+/* output switch */
+static int tumbler_get_master_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix = chip->mixer_data;
+
+ ucontrol->value.integer.value[0] = mix->master_switch[0];
+ ucontrol->value.integer.value[1] = mix->master_switch[1];
+ return 0;
+}
+
+static int tumbler_put_master_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix = chip->mixer_data;
+ int change;
+
+ change = mix->master_switch[0] != ucontrol->value.integer.value[0] ||
+ mix->master_switch[1] != ucontrol->value.integer.value[1];
+ if (change) {
+ mix->master_switch[0] = !!ucontrol->value.integer.value[0];
+ mix->master_switch[1] = !!ucontrol->value.integer.value[1];
+ tumbler_set_master_volume(mix);
+ }
+ return change;
+}
+
+
+/*
+ * TAS3001c dynamic range compression
+ */
+
+#define TAS3001_DRC_MAX 0x5f
+
+static int tumbler_set_drc(struct pmac_tumbler *mix)
+{
+ unsigned char val[2];
+
+ if (! mix->i2c.client)
+ return -ENODEV;
+
+ if (mix->drc_enable) {
+ val[0] = 0xc1; /* enable, 3:1 compression */
+ if (mix->drc_range > TAS3001_DRC_MAX)
+ val[1] = 0xf0;
+ else if (mix->drc_range < 0)
+ val[1] = 0x91;
+ else
+ val[1] = mix->drc_range + 0x91;
+ } else {
+ val[0] = 0;
+ val[1] = 0;
+ }
+
+ if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC,
+ 2, val) < 0) {
+ snd_printk("failed to set DRC\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * TAS3004
+ */
+
+#define TAS3004_DRC_MAX 0xef
+
+static int snapper_set_drc(struct pmac_tumbler *mix)
+{
+ unsigned char val[6];
+
+ if (! mix->i2c.client)
+ return -ENODEV;
+
+ if (mix->drc_enable)
+ val[0] = 0x50; /* 3:1 above threshold */
+ else
+ val[0] = 0x51; /* disabled */
+ val[1] = 0x02; /* 1:1 below threshold */
+ if (mix->drc_range > 0xef)
+ val[2] = 0xef;
+ else if (mix->drc_range < 0)
+ val[2] = 0x00;
+ else
+ val[2] = mix->drc_range;
+ val[3] = 0xb0;
+ val[4] = 0x60;
+ val[5] = 0xa0;
+
+ if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC,
+ 6, val) < 0) {
+ snd_printk("failed to set DRC\n");
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int tumbler_info_drc_value(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max =
+ chip->model == PMAC_TUMBLER ? TAS3001_DRC_MAX : TAS3004_DRC_MAX;
+ return 0;
+}
+
+static int tumbler_get_drc_value(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ ucontrol->value.integer.value[0] = mix->drc_range;
+ return 0;
+}
+
+static int tumbler_put_drc_value(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ unsigned int val;
+ int change;
+
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ val = ucontrol->value.integer.value[0];
+ if (chip->model == PMAC_TUMBLER) {
+ if (val > TAS3001_DRC_MAX)
+ return -EINVAL;
+ } else {
+ if (val > TAS3004_DRC_MAX)
+ return -EINVAL;
+ }
+ change = mix->drc_range != val;
+ if (change) {
+ mix->drc_range = val;
+ if (chip->model == PMAC_TUMBLER)
+ tumbler_set_drc(mix);
+ else
+ snapper_set_drc(mix);
+ }
+ return change;
+}
+
+static int tumbler_get_drc_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ ucontrol->value.integer.value[0] = mix->drc_enable;
+ return 0;
+}
+
+static int tumbler_put_drc_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ int change;
+
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ change = mix->drc_enable != ucontrol->value.integer.value[0];
+ if (change) {
+ mix->drc_enable = !!ucontrol->value.integer.value[0];
+ if (chip->model == PMAC_TUMBLER)
+ tumbler_set_drc(mix);
+ else
+ snapper_set_drc(mix);
+ }
+ return change;
+}
+
+
+/*
+ * mono volumes
+ */
+
+struct tumbler_mono_vol {
+ int index;
+ int reg;
+ int bytes;
+ unsigned int max;
+ unsigned int *table;
+};
+
+static int tumbler_set_mono_volume(struct pmac_tumbler *mix,
+ struct tumbler_mono_vol *info)
+{
+ unsigned char block[4];
+ unsigned int vol;
+ int i;
+
+ if (! mix->i2c.client)
+ return -ENODEV;
+
+ vol = mix->mono_vol[info->index];
+ if (vol >= info->max)
+ vol = info->max - 1;
+ vol = info->table[vol];
+ for (i = 0; i < info->bytes; i++)
+ block[i] = (vol >> ((info->bytes - i - 1) * 8)) & 0xff;
+ if (i2c_smbus_write_i2c_block_data(mix->i2c.client, info->reg,
+ info->bytes, block) < 0) {
+ snd_printk("failed to set mono volume %d\n", info->index);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int tumbler_info_mono(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct tumbler_mono_vol *info = (struct tumbler_mono_vol *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = info->max - 1;
+ return 0;
+}
+
+static int tumbler_get_mono(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tumbler_mono_vol *info = (struct tumbler_mono_vol *)kcontrol->private_value;
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ ucontrol->value.integer.value[0] = mix->mono_vol[info->index];
+ return 0;
+}
+
+static int tumbler_put_mono(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct tumbler_mono_vol *info = (struct tumbler_mono_vol *)kcontrol->private_value;
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ unsigned int vol;
+ int change;
+
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ vol = ucontrol->value.integer.value[0];
+ if (vol >= info->max)
+ return -EINVAL;
+ change = mix->mono_vol[info->index] != vol;
+ if (change) {
+ mix->mono_vol[info->index] = vol;
+ tumbler_set_mono_volume(mix, info);
+ }
+ return change;
+}
+
+/* TAS3001c mono volumes */
+static struct tumbler_mono_vol tumbler_pcm_vol_info = {
+ .index = VOL_IDX_PCM_MONO,
+ .reg = TAS_REG_PCM,
+ .bytes = 3,
+ .max = ARRAY_SIZE(mixer_volume_table),
+ .table = mixer_volume_table,
+};
+
+static struct tumbler_mono_vol tumbler_bass_vol_info = {
+ .index = VOL_IDX_BASS,
+ .reg = TAS_REG_BASS,
+ .bytes = 1,
+ .max = ARRAY_SIZE(bass_volume_table),
+ .table = bass_volume_table,
+};
+
+static struct tumbler_mono_vol tumbler_treble_vol_info = {
+ .index = VOL_IDX_TREBLE,
+ .reg = TAS_REG_TREBLE,
+ .bytes = 1,
+ .max = ARRAY_SIZE(treble_volume_table),
+ .table = treble_volume_table,
+};
+
+/* TAS3004 mono volumes */
+static struct tumbler_mono_vol snapper_bass_vol_info = {
+ .index = VOL_IDX_BASS,
+ .reg = TAS_REG_BASS,
+ .bytes = 1,
+ .max = ARRAY_SIZE(snapper_bass_volume_table),
+ .table = snapper_bass_volume_table,
+};
+
+static struct tumbler_mono_vol snapper_treble_vol_info = {
+ .index = VOL_IDX_TREBLE,
+ .reg = TAS_REG_TREBLE,
+ .bytes = 1,
+ .max = ARRAY_SIZE(snapper_treble_volume_table),
+ .table = snapper_treble_volume_table,
+};
+
+
+#define DEFINE_MONO(xname,type) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,\
+ .name = xname, \
+ .info = tumbler_info_mono, \
+ .get = tumbler_get_mono, \
+ .put = tumbler_put_mono, \
+ .private_value = (unsigned long)(&tumbler_##type##_vol_info), \
+}
+
+#define DEFINE_SNAPPER_MONO(xname,type) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,\
+ .name = xname, \
+ .info = tumbler_info_mono, \
+ .get = tumbler_get_mono, \
+ .put = tumbler_put_mono, \
+ .private_value = (unsigned long)(&snapper_##type##_vol_info), \
+}
+
+
+/*
+ * snapper mixer volumes
+ */
+
+static int snapper_set_mix_vol1(struct pmac_tumbler *mix, int idx, int ch, int reg)
+{
+ int i, j, vol;
+ unsigned char block[9];
+
+ vol = mix->mix_vol[idx][ch];
+ if (vol >= ARRAY_SIZE(mixer_volume_table)) {
+ vol = ARRAY_SIZE(mixer_volume_table) - 1;
+ mix->mix_vol[idx][ch] = vol;
+ }
+
+ for (i = 0; i < 3; i++) {
+ vol = mix->mix_vol[i][ch];
+ vol = mixer_volume_table[vol];
+ for (j = 0; j < 3; j++)
+ block[i * 3 + j] = (vol >> ((2 - j) * 8)) & 0xff;
+ }
+ if (i2c_smbus_write_i2c_block_data(mix->i2c.client, reg,
+ 9, block) < 0) {
+ snd_printk("failed to set mono volume %d\n", reg);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int snapper_set_mix_vol(struct pmac_tumbler *mix, int idx)
+{
+ if (! mix->i2c.client)
+ return -ENODEV;
+ if (snapper_set_mix_vol1(mix, idx, 0, TAS_REG_LMIX) < 0 ||
+ snapper_set_mix_vol1(mix, idx, 1, TAS_REG_RMIX) < 0)
+ return -EINVAL;
+ return 0;
+}
+
+static int snapper_info_mix(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = ARRAY_SIZE(mixer_volume_table) - 1;
+ return 0;
+}
+
+static int snapper_get_mix(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int idx = (int)kcontrol->private_value;
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ ucontrol->value.integer.value[0] = mix->mix_vol[idx][0];
+ ucontrol->value.integer.value[1] = mix->mix_vol[idx][1];
+ return 0;
+}
+
+static int snapper_put_mix(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int idx = (int)kcontrol->private_value;
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ unsigned int vol[2];
+ int change;
+
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ vol[0] = ucontrol->value.integer.value[0];
+ vol[1] = ucontrol->value.integer.value[1];
+ if (vol[0] >= ARRAY_SIZE(mixer_volume_table) ||
+ vol[1] >= ARRAY_SIZE(mixer_volume_table))
+ return -EINVAL;
+ change = mix->mix_vol[idx][0] != vol[0] ||
+ mix->mix_vol[idx][1] != vol[1];
+ if (change) {
+ mix->mix_vol[idx][0] = vol[0];
+ mix->mix_vol[idx][1] = vol[1];
+ snapper_set_mix_vol(mix, idx);
+ }
+ return change;
+}
+
+
+/*
+ * mute switches. FIXME: Turn that into software mute when both outputs are muted
+ * to avoid codec reset on ibook M7
+ */
+
+enum { TUMBLER_MUTE_HP, TUMBLER_MUTE_AMP, TUMBLER_MUTE_LINE };
+
+static int tumbler_get_mute_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ struct pmac_gpio *gp;
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ switch(kcontrol->private_value) {
+ case TUMBLER_MUTE_HP:
+ gp = &mix->hp_mute; break;
+ case TUMBLER_MUTE_AMP:
+ gp = &mix->amp_mute; break;
+ case TUMBLER_MUTE_LINE:
+ gp = &mix->line_mute; break;
+ default:
+ gp = NULL;
+ }
+ if (gp == NULL)
+ return -EINVAL;
+ ucontrol->value.integer.value[0] = !check_audio_gpio(gp);
+ return 0;
+}
+
+static int tumbler_put_mute_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix;
+ struct pmac_gpio *gp;
+ int val;
+#ifdef PMAC_SUPPORT_AUTOMUTE
+ if (chip->update_automute && chip->auto_mute)
+ return 0; /* don't touch in the auto-mute mode */
+#endif
+ if (! (mix = chip->mixer_data))
+ return -ENODEV;
+ switch(kcontrol->private_value) {
+ case TUMBLER_MUTE_HP:
+ gp = &mix->hp_mute; break;
+ case TUMBLER_MUTE_AMP:
+ gp = &mix->amp_mute; break;
+ case TUMBLER_MUTE_LINE:
+ gp = &mix->line_mute; break;
+ default:
+ gp = NULL;
+ }
+ if (gp == NULL)
+ return -EINVAL;
+ val = ! check_audio_gpio(gp);
+ if (val != ucontrol->value.integer.value[0]) {
+ write_audio_gpio(gp, ! ucontrol->value.integer.value[0]);
+ return 1;
+ }
+ return 0;
+}
+
+static int snapper_set_capture_source(struct pmac_tumbler *mix)
+{
+ if (! mix->i2c.client)
+ return -ENODEV;
+ if (mix->capture_source)
+ mix->acs = mix->acs |= 2;
+ else
+ mix->acs &= ~2;
+ return i2c_smbus_write_byte_data(mix->i2c.client, TAS_REG_ACS, mix->acs);
+}
+
+static int snapper_info_capture_source(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2] = {
+ "Line", "Mic"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snapper_get_capture_source(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix = chip->mixer_data;
+
+ ucontrol->value.enumerated.item[0] = mix->capture_source;
+ return 0;
+}
+
+static int snapper_put_capture_source(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pmac *chip = snd_kcontrol_chip(kcontrol);
+ struct pmac_tumbler *mix = chip->mixer_data;
+ int change;
+
+ change = ucontrol->value.enumerated.item[0] != mix->capture_source;
+ if (change) {
+ mix->capture_source = !!ucontrol->value.enumerated.item[0];
+ snapper_set_capture_source(mix);
+ }
+ return change;
+}
+
+#define DEFINE_SNAPPER_MIX(xname,idx,ofs) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,\
+ .name = xname, \
+ .info = snapper_info_mix, \
+ .get = snapper_get_mix, \
+ .put = snapper_put_mix, \
+ .index = idx,\
+ .private_value = ofs, \
+}
+
+
+/*
+ */
+static struct snd_kcontrol_new tumbler_mixers[] __initdata = {
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .info = tumbler_info_master_volume,
+ .get = tumbler_get_master_volume,
+ .put = tumbler_put_master_volume
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_pmac_boolean_stereo_info,
+ .get = tumbler_get_master_switch,
+ .put = tumbler_put_master_switch
+ },
+ DEFINE_MONO("Tone Control - Bass", bass),
+ DEFINE_MONO("Tone Control - Treble", treble),
+ DEFINE_MONO("PCM Playback Volume", pcm),
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DRC Range",
+ .info = tumbler_info_drc_value,
+ .get = tumbler_get_drc_value,
+ .put = tumbler_put_drc_value
+ },
+};
+
+static struct snd_kcontrol_new snapper_mixers[] __initdata = {
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .info = tumbler_info_master_volume,
+ .get = tumbler_get_master_volume,
+ .put = tumbler_put_master_volume
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_pmac_boolean_stereo_info,
+ .get = tumbler_get_master_switch,
+ .put = tumbler_put_master_switch
+ },
+ DEFINE_SNAPPER_MIX("PCM Playback Volume", 0, VOL_IDX_PCM),
+ DEFINE_SNAPPER_MIX("PCM Playback Volume", 1, VOL_IDX_PCM2),
+ DEFINE_SNAPPER_MIX("Monitor Mix Volume", 0, VOL_IDX_ADC),
+ DEFINE_SNAPPER_MONO("Tone Control - Bass", bass),
+ DEFINE_SNAPPER_MONO("Tone Control - Treble", treble),
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DRC Range",
+ .info = tumbler_info_drc_value,
+ .get = tumbler_get_drc_value,
+ .put = tumbler_put_drc_value
+ },
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source", /* FIXME: "Capture Source" doesn't work properly */
+ .info = snapper_info_capture_source,
+ .get = snapper_get_capture_source,
+ .put = snapper_put_capture_source
+ },
+};
+
+static struct snd_kcontrol_new tumbler_hp_sw __initdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Headphone Playback Switch",
+ .info = snd_pmac_boolean_mono_info,
+ .get = tumbler_get_mute_switch,
+ .put = tumbler_put_mute_switch,
+ .private_value = TUMBLER_MUTE_HP,
+};
+static struct snd_kcontrol_new tumbler_speaker_sw __initdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PC Speaker Playback Switch",
+ .info = snd_pmac_boolean_mono_info,
+ .get = tumbler_get_mute_switch,
+ .put = tumbler_put_mute_switch,
+ .private_value = TUMBLER_MUTE_AMP,
+};
+static struct snd_kcontrol_new tumbler_lineout_sw __initdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Out Playback Switch",
+ .info = snd_pmac_boolean_mono_info,
+ .get = tumbler_get_mute_switch,
+ .put = tumbler_put_mute_switch,
+ .private_value = TUMBLER_MUTE_LINE,
+};
+static struct snd_kcontrol_new tumbler_drc_sw __initdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DRC Switch",
+ .info = snd_pmac_boolean_mono_info,
+ .get = tumbler_get_drc_switch,
+ .put = tumbler_put_drc_switch
+};
+
+
+#ifdef PMAC_SUPPORT_AUTOMUTE
+/*
+ * auto-mute stuffs
+ */
+static int tumbler_detect_headphone(struct snd_pmac *chip)
+{
+ struct pmac_tumbler *mix = chip->mixer_data;
+ int detect = 0;
+
+ if (mix->hp_detect.addr)
+ detect |= read_audio_gpio(&mix->hp_detect);
+ return detect;
+}
+
+static int tumbler_detect_lineout(struct snd_pmac *chip)
+{
+ struct pmac_tumbler *mix = chip->mixer_data;
+ int detect = 0;
+
+ if (mix->line_detect.addr)
+ detect |= read_audio_gpio(&mix->line_detect);
+ return detect;
+}
+
+static void check_mute(struct snd_pmac *chip, struct pmac_gpio *gp, int val, int do_notify,
+ struct snd_kcontrol *sw)
+{
+ if (check_audio_gpio(gp) != val) {
+ write_audio_gpio(gp, val);
+ if (do_notify)
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &sw->id);
+ }
+}
+
+static struct work_struct device_change;
+static struct snd_pmac *device_change_chip;
+
+static void device_change_handler(struct work_struct *work)
+{
+ struct snd_pmac *chip = device_change_chip;
+ struct pmac_tumbler *mix;
+ int headphone, lineout;
+
+ if (!chip)
+ return;
+
+ mix = chip->mixer_data;
+ if (snd_BUG_ON(!mix))
+ return;
+
+ headphone = tumbler_detect_headphone(chip);
+ lineout = tumbler_detect_lineout(chip);
+
+ DBG("headphone: %d, lineout: %d\n", headphone, lineout);
+
+ if (headphone || lineout) {
+ /* unmute headphone/lineout & mute speaker */
+ if (headphone)
+ check_mute(chip, &mix->hp_mute, 0, mix->auto_mute_notify,
+ chip->master_sw_ctl);
+ if (lineout && mix->line_mute.addr != 0)
+ check_mute(chip, &mix->line_mute, 0, mix->auto_mute_notify,
+ chip->lineout_sw_ctl);
+ if (mix->anded_reset)
+ msleep(10);
+ check_mute(chip, &mix->amp_mute, 1, mix->auto_mute_notify,
+ chip->speaker_sw_ctl);
+ } else {
+ /* unmute speaker, mute others */
+ check_mute(chip, &mix->amp_mute, 0, mix->auto_mute_notify,
+ chip->speaker_sw_ctl);
+ if (mix->anded_reset)
+ msleep(10);
+ check_mute(chip, &mix->hp_mute, 1, mix->auto_mute_notify,
+ chip->master_sw_ctl);
+ if (mix->line_mute.addr != 0)
+ check_mute(chip, &mix->line_mute, 1, mix->auto_mute_notify,
+ chip->lineout_sw_ctl);
+ }
+ if (mix->auto_mute_notify)
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->hp_detect_ctl->id);
+
+#ifdef CONFIG_SND_POWERMAC_AUTO_DRC
+ mix->drc_enable = ! (headphone || lineout);
+ if (mix->auto_mute_notify)
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->drc_sw_ctl->id);
+ if (chip->model == PMAC_TUMBLER)
+ tumbler_set_drc(mix);
+ else
+ snapper_set_drc(mix);
+#endif
+
+ /* reset the master volume so the correct amplification is applied */
+ tumbler_set_master_volume(mix);
+}
+
+static void tumbler_update_automute(struct snd_pmac *chip, int do_notify)
+{
+ if (chip->auto_mute) {
+ struct pmac_tumbler *mix;
+ mix = chip->mixer_data;
+ if (snd_BUG_ON(!mix))
+ return;
+ mix->auto_mute_notify = do_notify;
+ schedule_work(&device_change);
+ }
+}
+#endif /* PMAC_SUPPORT_AUTOMUTE */
+
+
+/* interrupt - headphone plug changed */
+static irqreturn_t headphone_intr(int irq, void *devid)
+{
+ struct snd_pmac *chip = devid;
+ if (chip->update_automute && chip->initialized) {
+ chip->update_automute(chip, 1);
+ return IRQ_HANDLED;
+ }
+ return IRQ_NONE;
+}
+
+/* look for audio-gpio device */
+static struct device_node *find_audio_device(const char *name)
+{
+ struct device_node *gpiop;
+ struct device_node *np;
+
+ gpiop = of_find_node_by_name(NULL, "gpio");
+ if (! gpiop)
+ return NULL;
+
+ for (np = of_get_next_child(gpiop, NULL); np;
+ np = of_get_next_child(gpiop, np)) {
+ const char *property = of_get_property(np, "audio-gpio", NULL);
+ if (property && strcmp(property, name) == 0)
+ break;
+ }
+ of_node_put(gpiop);
+ return np;
+}
+
+/* look for audio-gpio device */
+static struct device_node *find_compatible_audio_device(const char *name)
+{
+ struct device_node *gpiop;
+ struct device_node *np;
+
+ gpiop = of_find_node_by_name(NULL, "gpio");
+ if (!gpiop)
+ return NULL;
+
+ for (np = of_get_next_child(gpiop, NULL); np;
+ np = of_get_next_child(gpiop, np)) {
+ if (of_device_is_compatible(np, name))
+ break;
+ }
+ of_node_put(gpiop);
+ return np;
+}
+
+/* find an audio device and get its address */
+static long tumbler_find_device(const char *device, const char *platform,
+ struct pmac_gpio *gp, int is_compatible)
+{
+ struct device_node *node;
+ const u32 *base;
+ u32 addr;
+ long ret;
+
+ if (is_compatible)
+ node = find_compatible_audio_device(device);
+ else
+ node = find_audio_device(device);
+ if (! node) {
+ DBG("(W) cannot find audio device %s !\n", device);
+ snd_printdd("cannot find device %s\n", device);
+ return -ENODEV;
+ }
+
+ base = of_get_property(node, "AAPL,address", NULL);
+ if (! base) {
+ base = of_get_property(node, "reg", NULL);
+ if (!base) {
+ DBG("(E) cannot find address for device %s !\n", device);
+ snd_printd("cannot find address for device %s\n", device);
+ of_node_put(node);
+ return -ENODEV;
+ }
+ addr = *base;
+ if (addr < 0x50)
+ addr += 0x50;
+ } else
+ addr = *base;
+
+ gp->addr = addr & 0x0000ffff;
+ /* Try to find the active state, default to 0 ! */
+ base = of_get_property(node, "audio-gpio-active-state", NULL);
+ if (base) {
+ gp->active_state = *base;
+ gp->active_val = (*base) ? 0x5 : 0x4;
+ gp->inactive_val = (*base) ? 0x4 : 0x5;
+ } else {
+ const u32 *prop = NULL;
+ gp->active_state = 0;
+ gp->active_val = 0x4;
+ gp->inactive_val = 0x5;
+ /* Here are some crude hacks to extract the GPIO polarity and
+ * open collector informations out of the do-platform script
+ * as we don't yet have an interpreter for these things
+ */
+ if (platform)
+ prop = of_get_property(node, platform, NULL);
+ if (prop) {
+ if (prop[3] == 0x9 && prop[4] == 0x9) {
+ gp->active_val = 0xd;
+ gp->inactive_val = 0xc;
+ }
+ if (prop[3] == 0x1 && prop[4] == 0x1) {
+ gp->active_val = 0x5;
+ gp->inactive_val = 0x4;
+ }
+ }
+ }
+
+ DBG("(I) GPIO device %s found, offset: %x, active state: %d !\n",
+ device, gp->addr, gp->active_state);
+
+ ret = irq_of_parse_and_map(node, 0);
+ of_node_put(node);
+ return ret;
+}
+
+/* reset audio */
+static void tumbler_reset_audio(struct snd_pmac *chip)
+{
+ struct pmac_tumbler *mix = chip->mixer_data;
+
+ if (mix->anded_reset) {
+ DBG("(I) codec anded reset !\n");
+ write_audio_gpio(&mix->hp_mute, 0);
+ write_audio_gpio(&mix->amp_mute, 0);
+ msleep(200);
+ write_audio_gpio(&mix->hp_mute, 1);
+ write_audio_gpio(&mix->amp_mute, 1);
+ msleep(100);
+ write_audio_gpio(&mix->hp_mute, 0);
+ write_audio_gpio(&mix->amp_mute, 0);
+ msleep(100);
+ } else {
+ DBG("(I) codec normal reset !\n");
+
+ write_audio_gpio(&mix->audio_reset, 0);
+ msleep(200);
+ write_audio_gpio(&mix->audio_reset, 1);
+ msleep(100);
+ write_audio_gpio(&mix->audio_reset, 0);
+ msleep(100);
+ }
+}
+
+#ifdef CONFIG_PM
+/* suspend mixer */
+static void tumbler_suspend(struct snd_pmac *chip)
+{
+ struct pmac_tumbler *mix = chip->mixer_data;
+
+ if (mix->headphone_irq >= 0)
+ disable_irq(mix->headphone_irq);
+ if (mix->lineout_irq >= 0)
+ disable_irq(mix->lineout_irq);
+ mix->save_master_switch[0] = mix->master_switch[0];
+ mix->save_master_switch[1] = mix->master_switch[1];
+ mix->save_master_vol[0] = mix->master_vol[0];
+ mix->save_master_vol[1] = mix->master_vol[1];
+ mix->master_switch[0] = mix->master_switch[1] = 0;
+ tumbler_set_master_volume(mix);
+ if (!mix->anded_reset) {
+ write_audio_gpio(&mix->amp_mute, 1);
+ write_audio_gpio(&mix->hp_mute, 1);
+ }
+ if (chip->model == PMAC_SNAPPER) {
+ mix->acs |= 1;
+ i2c_smbus_write_byte_data(mix->i2c.client, TAS_REG_ACS, mix->acs);
+ }
+ if (mix->anded_reset) {
+ write_audio_gpio(&mix->amp_mute, 1);
+ write_audio_gpio(&mix->hp_mute, 1);
+ } else
+ write_audio_gpio(&mix->audio_reset, 1);
+}
+
+/* resume mixer */
+static void tumbler_resume(struct snd_pmac *chip)
+{
+ struct pmac_tumbler *mix = chip->mixer_data;
+
+ mix->acs &= ~1;
+ mix->master_switch[0] = mix->save_master_switch[0];
+ mix->master_switch[1] = mix->save_master_switch[1];
+ mix->master_vol[0] = mix->save_master_vol[0];
+ mix->master_vol[1] = mix->save_master_vol[1];
+ tumbler_reset_audio(chip);
+ if (mix->i2c.client && mix->i2c.init_client) {
+ if (mix->i2c.init_client(&mix->i2c) < 0)
+ printk(KERN_ERR "tumbler_init_client error\n");
+ } else
+ printk(KERN_ERR "tumbler: i2c is not initialized\n");
+ if (chip->model == PMAC_TUMBLER) {
+ tumbler_set_mono_volume(mix, &tumbler_pcm_vol_info);
+ tumbler_set_mono_volume(mix, &tumbler_bass_vol_info);
+ tumbler_set_mono_volume(mix, &tumbler_treble_vol_info);
+ tumbler_set_drc(mix);
+ } else {
+ snapper_set_mix_vol(mix, VOL_IDX_PCM);
+ snapper_set_mix_vol(mix, VOL_IDX_PCM2);
+ snapper_set_mix_vol(mix, VOL_IDX_ADC);
+ tumbler_set_mono_volume(mix, &snapper_bass_vol_info);
+ tumbler_set_mono_volume(mix, &snapper_treble_vol_info);
+ snapper_set_drc(mix);
+ snapper_set_capture_source(mix);
+ }
+ tumbler_set_master_volume(mix);
+ if (chip->update_automute)
+ chip->update_automute(chip, 0);
+ if (mix->headphone_irq >= 0) {
+ unsigned char val;
+
+ enable_irq(mix->headphone_irq);
+ /* activate headphone status interrupts */
+ val = do_gpio_read(&mix->hp_detect);
+ do_gpio_write(&mix->hp_detect, val | 0x80);
+ }
+ if (mix->lineout_irq >= 0)
+ enable_irq(mix->lineout_irq);
+}
+#endif
+
+/* initialize tumbler */
+static int __init tumbler_init(struct snd_pmac *chip)
+{
+ int irq;
+ struct pmac_tumbler *mix = chip->mixer_data;
+
+ if (tumbler_find_device("audio-hw-reset",
+ "platform-do-hw-reset",
+ &mix->audio_reset, 0) < 0)
+ tumbler_find_device("hw-reset",
+ "platform-do-hw-reset",
+ &mix->audio_reset, 1);
+ if (tumbler_find_device("amp-mute",
+ "platform-do-amp-mute",
+ &mix->amp_mute, 0) < 0)
+ tumbler_find_device("amp-mute",
+ "platform-do-amp-mute",
+ &mix->amp_mute, 1);
+ if (tumbler_find_device("headphone-mute",
+ "platform-do-headphone-mute",
+ &mix->hp_mute, 0) < 0)
+ tumbler_find_device("headphone-mute",
+ "platform-do-headphone-mute",
+ &mix->hp_mute, 1);
+ if (tumbler_find_device("line-output-mute",
+ "platform-do-lineout-mute",
+ &mix->line_mute, 0) < 0)
+ tumbler_find_device("line-output-mute",
+ "platform-do-lineout-mute",
+ &mix->line_mute, 1);
+ irq = tumbler_find_device("headphone-detect",
+ NULL, &mix->hp_detect, 0);
+ if (irq <= NO_IRQ)
+ irq = tumbler_find_device("headphone-detect",
+ NULL, &mix->hp_detect, 1);
+ if (irq <= NO_IRQ)
+ irq = tumbler_find_device("keywest-gpio15",
+ NULL, &mix->hp_detect, 1);
+ mix->headphone_irq = irq;
+ irq = tumbler_find_device("line-output-detect",
+ NULL, &mix->line_detect, 0);
+ if (irq <= NO_IRQ)
+ irq = tumbler_find_device("line-output-detect",
+ NULL, &mix->line_detect, 1);
+ mix->lineout_irq = irq;
+
+ tumbler_reset_audio(chip);
+
+ return 0;
+}
+
+static void tumbler_cleanup(struct snd_pmac *chip)
+{
+ struct pmac_tumbler *mix = chip->mixer_data;
+ if (! mix)
+ return;
+
+ if (mix->headphone_irq >= 0)
+ free_irq(mix->headphone_irq, chip);
+ if (mix->lineout_irq >= 0)
+ free_irq(mix->lineout_irq, chip);
+ tumbler_gpio_free(&mix->audio_reset);
+ tumbler_gpio_free(&mix->amp_mute);
+ tumbler_gpio_free(&mix->hp_mute);
+ tumbler_gpio_free(&mix->hp_detect);
+ snd_pmac_keywest_cleanup(&mix->i2c);
+ kfree(mix);
+ chip->mixer_data = NULL;
+}
+
+/* exported */
+int __init snd_pmac_tumbler_init(struct snd_pmac *chip)
+{
+ int i, err;
+ struct pmac_tumbler *mix;
+ const u32 *paddr;
+ struct device_node *tas_node, *np;
+ char *chipname;
+
+ request_module("i2c-powermac");
+
+ mix = kzalloc(sizeof(*mix), GFP_KERNEL);
+ if (! mix)
+ return -ENOMEM;
+ mix->headphone_irq = -1;
+
+ chip->mixer_data = mix;
+ chip->mixer_free = tumbler_cleanup;
+ mix->anded_reset = 0;
+ mix->reset_on_sleep = 1;
+
+ for (np = chip->node->child; np; np = np->sibling) {
+ if (!strcmp(np->name, "sound")) {
+ if (of_get_property(np, "has-anded-reset", NULL))
+ mix->anded_reset = 1;
+ if (of_get_property(np, "layout-id", NULL))
+ mix->reset_on_sleep = 0;
+ break;
+ }
+ }
+ if ((err = tumbler_init(chip)) < 0)
+ return err;
+
+ /* set up TAS */
+ tas_node = of_find_node_by_name(NULL, "deq");
+ if (tas_node == NULL)
+ tas_node = of_find_node_by_name(NULL, "codec");
+ if (tas_node == NULL)
+ return -ENODEV;
+
+ paddr = of_get_property(tas_node, "i2c-address", NULL);
+ if (paddr == NULL)
+ paddr = of_get_property(tas_node, "reg", NULL);
+ if (paddr)
+ mix->i2c.addr = (*paddr) >> 1;
+ else
+ mix->i2c.addr = TAS_I2C_ADDR;
+ of_node_put(tas_node);
+
+ DBG("(I) TAS i2c address is: %x\n", mix->i2c.addr);
+
+ if (chip->model == PMAC_TUMBLER) {
+ mix->i2c.init_client = tumbler_init_client;
+ mix->i2c.name = "TAS3001c";
+ chipname = "Tumbler";
+ } else {
+ mix->i2c.init_client = snapper_init_client;
+ mix->i2c.name = "TAS3004";
+ chipname = "Snapper";
+ }
+
+ if ((err = snd_pmac_keywest_init(&mix->i2c)) < 0)
+ return err;
+
+ /*
+ * build mixers
+ */
+ sprintf(chip->card->mixername, "PowerMac %s", chipname);
+
+ if (chip->model == PMAC_TUMBLER) {
+ for (i = 0; i < ARRAY_SIZE(tumbler_mixers); i++) {
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&tumbler_mixers[i], chip))) < 0)
+ return err;
+ }
+ } else {
+ for (i = 0; i < ARRAY_SIZE(snapper_mixers); i++) {
+ if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snapper_mixers[i], chip))) < 0)
+ return err;
+ }
+ }
+ chip->master_sw_ctl = snd_ctl_new1(&tumbler_hp_sw, chip);
+ if ((err = snd_ctl_add(chip->card, chip->master_sw_ctl)) < 0)
+ return err;
+ chip->speaker_sw_ctl = snd_ctl_new1(&tumbler_speaker_sw, chip);
+ if ((err = snd_ctl_add(chip->card, chip->speaker_sw_ctl)) < 0)
+ return err;
+ if (mix->line_mute.addr != 0) {
+ chip->lineout_sw_ctl = snd_ctl_new1(&tumbler_lineout_sw, chip);
+ if ((err = snd_ctl_add(chip->card, chip->lineout_sw_ctl)) < 0)
+ return err;
+ }
+ chip->drc_sw_ctl = snd_ctl_new1(&tumbler_drc_sw, chip);
+ if ((err = snd_ctl_add(chip->card, chip->drc_sw_ctl)) < 0)
+ return err;
+
+ /* set initial DRC range to 60% */
+ if (chip->model == PMAC_TUMBLER)
+ mix->drc_range = (TAS3001_DRC_MAX * 6) / 10;
+ else
+ mix->drc_range = (TAS3004_DRC_MAX * 6) / 10;
+ mix->drc_enable = 1; /* will be changed later if AUTO_DRC is set */
+ if (chip->model == PMAC_TUMBLER)
+ tumbler_set_drc(mix);
+ else
+ snapper_set_drc(mix);
+
+#ifdef CONFIG_PM
+ chip->suspend = tumbler_suspend;
+ chip->resume = tumbler_resume;
+#endif
+
+ INIT_WORK(&device_change, device_change_handler);
+ device_change_chip = chip;
+
+#ifdef PMAC_SUPPORT_AUTOMUTE
+ if ((mix->headphone_irq >=0 || mix->lineout_irq >= 0)
+ && (err = snd_pmac_add_automute(chip)) < 0)
+ return err;
+ chip->detect_headphone = tumbler_detect_headphone;
+ chip->update_automute = tumbler_update_automute;
+ tumbler_update_automute(chip, 0); /* update the status only */
+
+ /* activate headphone status interrupts */
+ if (mix->headphone_irq >= 0) {
+ unsigned char val;
+ if ((err = request_irq(mix->headphone_irq, headphone_intr, 0,
+ "Sound Headphone Detection", chip)) < 0)
+ return 0;
+ /* activate headphone status interrupts */
+ val = do_gpio_read(&mix->hp_detect);
+ do_gpio_write(&mix->hp_detect, val | 0x80);
+ }
+ if (mix->lineout_irq >= 0) {
+ unsigned char val;
+ if ((err = request_irq(mix->lineout_irq, headphone_intr, 0,
+ "Sound Lineout Detection", chip)) < 0)
+ return 0;
+ /* activate headphone status interrupts */
+ val = do_gpio_read(&mix->line_detect);
+ do_gpio_write(&mix->line_detect, val | 0x80);
+ }
+#endif
+
+ return 0;
+}
diff --git a/sound/ppc/tumbler_volume.h b/sound/ppc/tumbler_volume.h
new file mode 100644
index 0000000..ef8d85d
--- /dev/null
+++ b/sound/ppc/tumbler_volume.h
@@ -0,0 +1,250 @@
+/* volume tables, taken from TAS3001c data manual */
+/* volume gain values */
+/* 0 = -70 dB, 175 = 18.0 dB in 0.5 dB step */
+static unsigned int master_volume_table[] = {
+ 0x00000015, 0x00000016, 0x00000017,
+ 0x00000019, 0x0000001a, 0x0000001c,
+ 0x0000001d, 0x0000001f, 0x00000021,
+ 0x00000023, 0x00000025, 0x00000027,
+ 0x00000029, 0x0000002c, 0x0000002e,
+ 0x00000031, 0x00000034, 0x00000037,
+ 0x0000003a, 0x0000003e, 0x00000042,
+ 0x00000045, 0x0000004a, 0x0000004e,
+ 0x00000053, 0x00000057, 0x0000005d,
+ 0x00000062, 0x00000068, 0x0000006e,
+ 0x00000075, 0x0000007b, 0x00000083,
+ 0x0000008b, 0x00000093, 0x0000009b,
+ 0x000000a5, 0x000000ae, 0x000000b9,
+ 0x000000c4, 0x000000cf, 0x000000dc,
+ 0x000000e9, 0x000000f6, 0x00000105,
+ 0x00000114, 0x00000125, 0x00000136,
+ 0x00000148, 0x0000015c, 0x00000171,
+ 0x00000186, 0x0000019e, 0x000001b6,
+ 0x000001d0, 0x000001eb, 0x00000209,
+ 0x00000227, 0x00000248, 0x0000026b,
+ 0x0000028f, 0x000002b6, 0x000002df,
+ 0x0000030b, 0x00000339, 0x0000036a,
+ 0x0000039e, 0x000003d5, 0x0000040f,
+ 0x0000044c, 0x0000048d, 0x000004d2,
+ 0x0000051c, 0x00000569, 0x000005bb,
+ 0x00000612, 0x0000066e, 0x000006d0,
+ 0x00000737, 0x000007a5, 0x00000818,
+ 0x00000893, 0x00000915, 0x0000099f,
+ 0x00000a31, 0x00000acc, 0x00000b6f,
+ 0x00000c1d, 0x00000cd5, 0x00000d97,
+ 0x00000e65, 0x00000f40, 0x00001027,
+ 0x0000111c, 0x00001220, 0x00001333,
+ 0x00001456, 0x0000158a, 0x000016d1,
+ 0x0000182b, 0x0000199a, 0x00001b1e,
+ 0x00001cb9, 0x00001e6d, 0x0000203a,
+ 0x00002223, 0x00002429, 0x0000264e,
+ 0x00002893, 0x00002afa, 0x00002d86,
+ 0x00003039, 0x00003314, 0x0000361b,
+ 0x00003950, 0x00003cb5, 0x0000404e,
+ 0x0000441d, 0x00004827, 0x00004c6d,
+ 0x000050f4, 0x000055c0, 0x00005ad5,
+ 0x00006037, 0x000065ea, 0x00006bf4,
+ 0x0000725a, 0x00007920, 0x0000804e,
+ 0x000087e8, 0x00008ff6, 0x0000987d,
+ 0x0000a186, 0x0000ab19, 0x0000b53c,
+ 0x0000bff9, 0x0000cb59, 0x0000d766,
+ 0x0000e429, 0x0000f1ae, 0x00010000,
+ 0x00010f2b, 0x00011f3d, 0x00013042,
+ 0x00014249, 0x00015562, 0x0001699c,
+ 0x00017f09, 0x000195bc, 0x0001adc6,
+ 0x0001c73d, 0x0001e237, 0x0001feca,
+ 0x00021d0e, 0x00023d1d, 0x00025f12,
+ 0x0002830b, 0x0002a925, 0x0002d182,
+ 0x0002fc42, 0x0003298b, 0x00035983,
+ 0x00038c53, 0x0003c225, 0x0003fb28,
+ 0x0004378b, 0x00047783, 0x0004bb44,
+ 0x0005030a, 0x00054f10, 0x00059f98,
+ 0x0005f4e5, 0x00064f40, 0x0006aef6,
+ 0x00071457, 0x00077fbb, 0x0007f17b,
+};
+
+/* treble table for TAS3001c */
+/* 0 = -18 dB, 72 = 18 dB in 0.5 dB step */
+static unsigned int treble_volume_table[] = {
+ 0x96, 0x95, 0x94,
+ 0x93, 0x92, 0x91,
+ 0x90, 0x8f, 0x8e,
+ 0x8d, 0x8c, 0x8b,
+ 0x8a, 0x89, 0x88,
+ 0x87, 0x86, 0x85,
+ 0x84, 0x83, 0x82,
+ 0x81, 0x80, 0x7f,
+ 0x7e, 0x7d, 0x7c,
+ 0x7b, 0x7a, 0x79,
+ 0x78, 0x77, 0x76,
+ 0x75, 0x74, 0x73,
+ 0x72, 0x71, 0x70,
+ 0x6e, 0x6d, 0x6c,
+ 0x6b, 0x69, 0x68,
+ 0x66, 0x65, 0x63,
+ 0x62, 0x60, 0x5e,
+ 0x5c, 0x5a, 0x57,
+ 0x55, 0x52, 0x4f,
+ 0x4c, 0x49, 0x45,
+ 0x42, 0x3e, 0x3a,
+ 0x36, 0x32, 0x2d,
+ 0x28, 0x22, 0x1c,
+ 0x16, 0x10, 0x09,
+ 0x01,
+};
+
+/* bass table for TAS3001c */
+/* 0 = -18 dB, 72 = 18 dB in 0.5 dB step */
+static unsigned int bass_volume_table[] = {
+ 0x86, 0x82, 0x7f,
+ 0x7d, 0x7a, 0x78,
+ 0x76, 0x74, 0x72,
+ 0x70, 0x6e, 0x6d,
+ 0x6b, 0x69, 0x66,
+ 0x64, 0x61, 0x5f,
+ 0x5d, 0x5c, 0x5a,
+ 0x59, 0x58, 0x56,
+ 0x55, 0x54, 0x53,
+ 0x51, 0x4f, 0x4d,
+ 0x4b, 0x49, 0x46,
+ 0x44, 0x42, 0x40,
+ 0x3e, 0x3c, 0x3b,
+ 0x39, 0x38, 0x36,
+ 0x35, 0x33, 0x31,
+ 0x30, 0x2e, 0x2c,
+ 0x2b, 0x29, 0x28,
+ 0x26, 0x25, 0x23,
+ 0x21, 0x1f, 0x1c,
+ 0x19, 0x18, 0x17,
+ 0x16, 0x14, 0x13,
+ 0x12, 0x10, 0x0f,
+ 0x0d, 0x0b, 0x0a,
+ 0x08, 0x06, 0x03,
+ 0x01,
+};
+
+/* mixer (pcm) volume table */
+/* 0 = -70 dB, 175 = 18.0 dB in 0.5 dB step */
+static unsigned int mixer_volume_table[] = {
+ 0x00014b, 0x00015f, 0x000174,
+ 0x00018a, 0x0001a1, 0x0001ba,
+ 0x0001d4, 0x0001f0, 0x00020d,
+ 0x00022c, 0x00024d, 0x000270,
+ 0x000295, 0x0002bc, 0x0002e6,
+ 0x000312, 0x000340, 0x000372,
+ 0x0003a6, 0x0003dd, 0x000418,
+ 0x000456, 0x000498, 0x0004de,
+ 0x000528, 0x000576, 0x0005c9,
+ 0x000620, 0x00067d, 0x0006e0,
+ 0x000748, 0x0007b7, 0x00082c,
+ 0x0008a8, 0x00092b, 0x0009b6,
+ 0x000a49, 0x000ae5, 0x000b8b,
+ 0x000c3a, 0x000cf3, 0x000db8,
+ 0x000e88, 0x000f64, 0x00104e,
+ 0x001145, 0x00124b, 0x001361,
+ 0x001487, 0x0015be, 0x001708,
+ 0x001865, 0x0019d8, 0x001b60,
+ 0x001cff, 0x001eb7, 0x002089,
+ 0x002276, 0x002481, 0x0026ab,
+ 0x0028f5, 0x002b63, 0x002df5,
+ 0x0030ae, 0x003390, 0x00369e,
+ 0x0039db, 0x003d49, 0x0040ea,
+ 0x0044c3, 0x0048d6, 0x004d27,
+ 0x0051b9, 0x005691, 0x005bb2,
+ 0x006121, 0x0066e3, 0x006cfb,
+ 0x007370, 0x007a48, 0x008186,
+ 0x008933, 0x009154, 0x0099f1,
+ 0x00a310, 0x00acba, 0x00b6f6,
+ 0x00c1cd, 0x00cd49, 0x00d973,
+ 0x00e655, 0x00f3fb, 0x010270,
+ 0x0111c0, 0x0121f9, 0x013328,
+ 0x01455b, 0x0158a2, 0x016d0e,
+ 0x0182af, 0x019999, 0x01b1de,
+ 0x01cb94, 0x01e6cf, 0x0203a7,
+ 0x022235, 0x024293, 0x0264db,
+ 0x02892c, 0x02afa3, 0x02d862,
+ 0x03038a, 0x033142, 0x0361af,
+ 0x0394fa, 0x03cb50, 0x0404de,
+ 0x0441d5, 0x048268, 0x04c6d0,
+ 0x050f44, 0x055c04, 0x05ad50,
+ 0x06036e, 0x065ea5, 0x06bf44,
+ 0x07259d, 0x079207, 0x0804dc,
+ 0x087e80, 0x08ff59, 0x0987d5,
+ 0x0a1866, 0x0ab189, 0x0b53be,
+ 0x0bff91, 0x0cb591, 0x0d765a,
+ 0x0e4290, 0x0f1adf, 0x100000,
+ 0x10f2b4, 0x11f3c9, 0x13041a,
+ 0x14248e, 0x15561a, 0x1699c0,
+ 0x17f094, 0x195bb8, 0x1adc61,
+ 0x1c73d5, 0x1e236d, 0x1fec98,
+ 0x21d0d9, 0x23d1cd, 0x25f125,
+ 0x2830af, 0x2a9254, 0x2d1818,
+ 0x2fc420, 0x3298b0, 0x35982f,
+ 0x38c528, 0x3c224c, 0x3fb278,
+ 0x437880, 0x477828, 0x4bb446,
+ 0x5030a1, 0x54f106, 0x59f980,
+ 0x5f4e52, 0x64f403, 0x6aef5d,
+ 0x714575, 0x77fbaa, 0x7f17af,
+};
+
+
+/* treble table for TAS3004 */
+/* 0 = -18 dB, 72 = 18 dB in 0.5 dB step */
+static unsigned int snapper_treble_volume_table[] = {
+ 0x96, 0x95, 0x94,
+ 0x93, 0x92, 0x91,
+ 0x90, 0x8f, 0x8e,
+ 0x8d, 0x8c, 0x8b,
+ 0x8a, 0x89, 0x88,
+ 0x87, 0x86, 0x85,
+ 0x84, 0x83, 0x82,
+ 0x81, 0x80, 0x7f,
+ 0x7e, 0x7d, 0x7c,
+ 0x7b, 0x7a, 0x79,
+ 0x78, 0x77, 0x76,
+ 0x75, 0x74, 0x73,
+ 0x72, 0x71, 0x70,
+ 0x6f, 0x6d, 0x6c,
+ 0x6b, 0x69, 0x68,
+ 0x67, 0x65, 0x63,
+ 0x62, 0x60, 0x5d,
+ 0x5b, 0x59, 0x56,
+ 0x53, 0x51, 0x4d,
+ 0x4a, 0x47, 0x43,
+ 0x3f, 0x3b, 0x36,
+ 0x31, 0x2c, 0x26,
+ 0x20, 0x1a, 0x13,
+ 0x08, 0x04, 0x01,
+ 0x01,
+};
+
+/* bass table for TAS3004 */
+/* 0 = -18 dB, 72 = 18 dB in 0.5 dB step */
+static unsigned int snapper_bass_volume_table[] = {
+ 0x96, 0x95, 0x94,
+ 0x93, 0x92, 0x91,
+ 0x90, 0x8f, 0x8e,
+ 0x8d, 0x8c, 0x8b,
+ 0x8a, 0x89, 0x88,
+ 0x87, 0x86, 0x85,
+ 0x84, 0x83, 0x82,
+ 0x81, 0x80, 0x7f,
+ 0x7e, 0x7d, 0x7c,
+ 0x7b, 0x7a, 0x79,
+ 0x78, 0x77, 0x76,
+ 0x75, 0x74, 0x73,
+ 0x72, 0x71, 0x6f,
+ 0x6e, 0x6d, 0x6b,
+ 0x6a, 0x69, 0x67,
+ 0x66, 0x65, 0x63,
+ 0x62, 0x61, 0x5f,
+ 0x5d, 0x5b, 0x58,
+ 0x55, 0x52, 0x4f,
+ 0x4c, 0x49, 0x46,
+ 0x43, 0x3f, 0x3b,
+ 0x37, 0x33, 0x2e,
+ 0x29, 0x24, 0x1e,
+ 0x18, 0x11, 0x0a,
+ 0x01,
+};
+
diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig
new file mode 100644
index 0000000..cfc1439
--- /dev/null
+++ b/sound/sh/Kconfig
@@ -0,0 +1,22 @@
+# ALSA SH drivers
+
+menuconfig SND_SUPERH
+ bool "SUPERH sound devices"
+ depends on SUPERH
+ default y
+ help
+ Support for sound devices specific to SUPERH architectures.
+ Drivers that are implemented on ASoC can be found in
+ "ALSA for SoC audio support" section.
+
+if SND_SUPERH
+
+config SND_AICA
+ tristate "Dreamcast Yamaha AICA sound"
+ depends on SH_DREAMCAST
+ select SND_PCM
+ help
+ ALSA Sound driver for the SEGA Dreamcast console.
+
+endif # SND_SUPERH
+
diff --git a/sound/sh/Makefile b/sound/sh/Makefile
new file mode 100644
index 0000000..8fdcb6e
--- /dev/null
+++ b/sound/sh/Makefile
@@ -0,0 +1,8 @@
+#
+# Makefile for ALSA
+#
+
+snd-aica-objs := aica.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AICA) += snd-aica.o
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
new file mode 100644
index 0000000..7c920f3
--- /dev/null
+++ b/sound/sh/aica.c
@@ -0,0 +1,688 @@
+/*
+* This code is licenced under
+* the General Public Licence
+* version 2
+*
+* Copyright Adrian McMenamin 2005, 2006, 2007
+* <adrian@mcmen.demon.co.uk>
+* Requires firmware (BSD licenced) available from:
+* http://linuxdc.cvs.sourceforge.net/linuxdc/linux-sh-dc/sound/oss/aica/firmware/
+* or the maintainer
+*
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of version 2 of the GNU General Public License as published by
+* the Free Software Foundation.
+*
+* This program is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+* GNU General Public License for more details.
+*
+* You should have received a copy of the GNU General Public License
+* along with this program; if not, write to the Free Software
+* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*
+*/
+
+#include <linux/init.h>
+#include <linux/jiffies.h>
+#include <linux/slab.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/firmware.h>
+#include <linux/timer.h>
+#include <linux/delay.h>
+#include <linux/workqueue.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <asm/io.h>
+#include <asm/dma.h>
+#include <mach/sysasic.h>
+#include "aica.h"
+
+MODULE_AUTHOR("Adrian McMenamin <adrian@mcmen.demon.co.uk>");
+MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}");
+
+/* module parameters */
+#define CARD_NAME "AICA"
+static int index = -1;
+static char *id;
+static int enable = 1;
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+module_param(enable, bool, 0644);
+MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
+
+/* Use workqueue */
+static struct workqueue_struct *aica_queue;
+
+/* Simple platform device */
+static struct platform_device *pd;
+static struct resource aica_memory_space[2] = {
+ {
+ .name = "AICA ARM CONTROL",
+ .start = ARM_RESET_REGISTER,
+ .flags = IORESOURCE_MEM,
+ .end = ARM_RESET_REGISTER + 3,
+ },
+ {
+ .name = "AICA Sound RAM",
+ .start = SPU_MEMORY_BASE,
+ .flags = IORESOURCE_MEM,
+ .end = SPU_MEMORY_BASE + 0x200000 - 1,
+ },
+};
+
+/* SPU specific functions */
+/* spu_write_wait - wait for G2-SH FIFO to clear */
+static void spu_write_wait(void)
+{
+ int time_count;
+ time_count = 0;
+ while (1) {
+ if (!(readl(G2_FIFO) & 0x11))
+ break;
+ /* To ensure hardware failure doesn't wedge kernel */
+ time_count++;
+ if (time_count > 0x10000) {
+ snd_printk
+ ("WARNING: G2 FIFO appears to be blocked.\n");
+ break;
+ }
+ }
+}
+
+/* spu_memset - write to memory in SPU address space */
+static void spu_memset(u32 toi, u32 what, int length)
+{
+ int i;
+ unsigned long flags;
+ if (snd_BUG_ON(length % 4))
+ return;
+ for (i = 0; i < length; i++) {
+ if (!(i % 8))
+ spu_write_wait();
+ local_irq_save(flags);
+ writel(what, toi + SPU_MEMORY_BASE);
+ local_irq_restore(flags);
+ toi++;
+ }
+}
+
+/* spu_memload - write to SPU address space */
+static void spu_memload(u32 toi, void *from, int length)
+{
+ unsigned long flags;
+ u32 *froml = from;
+ u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi);
+ int i;
+ u32 val;
+ length = DIV_ROUND_UP(length, 4);
+ spu_write_wait();
+ for (i = 0; i < length; i++) {
+ if (!(i % 8))
+ spu_write_wait();
+ val = *froml;
+ local_irq_save(flags);
+ writel(val, to);
+ local_irq_restore(flags);
+ froml++;
+ to++;
+ }
+}
+
+/* spu_disable - set spu registers to stop sound output */
+static void spu_disable(void)
+{
+ int i;
+ unsigned long flags;
+ u32 regval;
+ spu_write_wait();
+ regval = readl(ARM_RESET_REGISTER);
+ regval |= 1;
+ spu_write_wait();
+ local_irq_save(flags);
+ writel(regval, ARM_RESET_REGISTER);
+ local_irq_restore(flags);
+ for (i = 0; i < 64; i++) {
+ spu_write_wait();
+ regval = readl(SPU_REGISTER_BASE + (i * 0x80));
+ regval = (regval & ~0x4000) | 0x8000;
+ spu_write_wait();
+ local_irq_save(flags);
+ writel(regval, SPU_REGISTER_BASE + (i * 0x80));
+ local_irq_restore(flags);
+ }
+}
+
+/* spu_enable - set spu registers to enable sound output */
+static void spu_enable(void)
+{
+ unsigned long flags;
+ u32 regval = readl(ARM_RESET_REGISTER);
+ regval &= ~1;
+ spu_write_wait();
+ local_irq_save(flags);
+ writel(regval, ARM_RESET_REGISTER);
+ local_irq_restore(flags);
+}
+
+/*
+ * Halt the sound processor, clear the memory,
+ * load some default ARM7 code, and then restart ARM7
+*/
+static void spu_reset(void)
+{
+ unsigned long flags;
+ spu_disable();
+ spu_memset(0, 0, 0x200000 / 4);
+ /* Put ARM7 in endless loop */
+ local_irq_save(flags);
+ ctrl_outl(0xea000002, SPU_MEMORY_BASE);
+ local_irq_restore(flags);
+ spu_enable();
+}
+
+/* aica_chn_start - write to spu to start playback */
+static void aica_chn_start(void)
+{
+ unsigned long flags;
+ spu_write_wait();
+ local_irq_save(flags);
+ writel(AICA_CMD_KICK | AICA_CMD_START, (u32 *) AICA_CONTROL_POINT);
+ local_irq_restore(flags);
+}
+
+/* aica_chn_halt - write to spu to halt playback */
+static void aica_chn_halt(void)
+{
+ unsigned long flags;
+ spu_write_wait();
+ local_irq_save(flags);
+ writel(AICA_CMD_KICK | AICA_CMD_STOP, (u32 *) AICA_CONTROL_POINT);
+ local_irq_restore(flags);
+}
+
+/* ALSA code below */
+static struct snd_pcm_hardware snd_pcm_aica_playback_hw = {
+ .info = (SNDRV_PCM_INFO_NONINTERLEAVED),
+ .formats =
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_IMA_ADPCM),
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = AICA_BUFFER_SIZE,
+ .period_bytes_min = AICA_PERIOD_SIZE,
+ .period_bytes_max = AICA_PERIOD_SIZE,
+ .periods_min = AICA_PERIOD_NUMBER,
+ .periods_max = AICA_PERIOD_NUMBER,
+};
+
+static int aica_dma_transfer(int channels, int buffer_size,
+ struct snd_pcm_substream *substream)
+{
+ int q, err, period_offset;
+ struct snd_card_aica *dreamcastcard;
+ struct snd_pcm_runtime *runtime;
+ unsigned long flags;
+ err = 0;
+ dreamcastcard = substream->pcm->private_data;
+ period_offset = dreamcastcard->clicks;
+ period_offset %= (AICA_PERIOD_NUMBER / channels);
+ runtime = substream->runtime;
+ for (q = 0; q < channels; q++) {
+ local_irq_save(flags);
+ err = dma_xfer(AICA_DMA_CHANNEL,
+ (unsigned long) (runtime->dma_area +
+ (AICA_BUFFER_SIZE * q) /
+ channels +
+ AICA_PERIOD_SIZE *
+ period_offset),
+ AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET +
+ AICA_PERIOD_SIZE * period_offset,
+ buffer_size / channels, AICA_DMA_MODE);
+ if (unlikely(err < 0)) {
+ local_irq_restore(flags);
+ break;
+ }
+ dma_wait_for_completion(AICA_DMA_CHANNEL);
+ local_irq_restore(flags);
+ }
+ return err;
+}
+
+static void startup_aica(struct snd_card_aica *dreamcastcard)
+{
+ spu_memload(AICA_CHANNEL0_CONTROL_OFFSET,
+ dreamcastcard->channel, sizeof(struct aica_channel));
+ aica_chn_start();
+}
+
+static void run_spu_dma(struct work_struct *work)
+{
+ int buffer_size;
+ struct snd_pcm_runtime *runtime;
+ struct snd_card_aica *dreamcastcard;
+ dreamcastcard =
+ container_of(work, struct snd_card_aica, spu_dma_work);
+ runtime = dreamcastcard->substream->runtime;
+ if (unlikely(dreamcastcard->dma_check == 0)) {
+ buffer_size =
+ frames_to_bytes(runtime, runtime->buffer_size);
+ if (runtime->channels > 1)
+ dreamcastcard->channel->flags |= 0x01;
+ aica_dma_transfer(runtime->channels, buffer_size,
+ dreamcastcard->substream);
+ startup_aica(dreamcastcard);
+ dreamcastcard->clicks =
+ buffer_size / (AICA_PERIOD_SIZE * runtime->channels);
+ return;
+ } else {
+ aica_dma_transfer(runtime->channels,
+ AICA_PERIOD_SIZE * runtime->channels,
+ dreamcastcard->substream);
+ snd_pcm_period_elapsed(dreamcastcard->substream);
+ dreamcastcard->clicks++;
+ if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER))
+ dreamcastcard->clicks %= AICA_PERIOD_NUMBER;
+ mod_timer(&dreamcastcard->timer, jiffies + 1);
+ }
+}
+
+static void aica_period_elapsed(unsigned long timer_var)
+{
+ /*timer function - so cannot sleep */
+ int play_period;
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm_substream *substream;
+ struct snd_card_aica *dreamcastcard;
+ substream = (struct snd_pcm_substream *) timer_var;
+ runtime = substream->runtime;
+ dreamcastcard = substream->pcm->private_data;
+ /* Have we played out an additional period? */
+ play_period =
+ frames_to_bytes(runtime,
+ readl
+ (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) /
+ AICA_PERIOD_SIZE;
+ if (play_period == dreamcastcard->current_period) {
+ /* reschedule the timer */
+ mod_timer(&(dreamcastcard->timer), jiffies + 1);
+ return;
+ }
+ if (runtime->channels > 1)
+ dreamcastcard->current_period = play_period;
+ if (unlikely(dreamcastcard->dma_check == 0))
+ dreamcastcard->dma_check = 1;
+ queue_work(aica_queue, &(dreamcastcard->spu_dma_work));
+}
+
+static void spu_begin_dma(struct snd_pcm_substream *substream)
+{
+ struct snd_card_aica *dreamcastcard;
+ struct snd_pcm_runtime *runtime;
+ runtime = substream->runtime;
+ dreamcastcard = substream->pcm->private_data;
+ /*get the queue to do the work */
+ queue_work(aica_queue, &(dreamcastcard->spu_dma_work));
+ /* Timer may already be running */
+ if (unlikely(dreamcastcard->timer.data)) {
+ mod_timer(&dreamcastcard->timer, jiffies + 4);
+ return;
+ }
+ init_timer(&(dreamcastcard->timer));
+ dreamcastcard->timer.data = (unsigned long) substream;
+ dreamcastcard->timer.function = aica_period_elapsed;
+ dreamcastcard->timer.expires = jiffies + 4;
+ add_timer(&(dreamcastcard->timer));
+}
+
+static int snd_aicapcm_pcm_open(struct snd_pcm_substream
+ *substream)
+{
+ struct snd_pcm_runtime *runtime;
+ struct aica_channel *channel;
+ struct snd_card_aica *dreamcastcard;
+ if (!enable)
+ return -ENOENT;
+ dreamcastcard = substream->pcm->private_data;
+ channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL);
+ if (!channel)
+ return -ENOMEM;
+ /* set defaults for channel */
+ channel->sfmt = SM_8BIT;
+ channel->cmd = AICA_CMD_START;
+ channel->vol = dreamcastcard->master_volume;
+ channel->pan = 0x80;
+ channel->pos = 0;
+ channel->flags = 0; /* default to mono */
+ dreamcastcard->channel = channel;
+ runtime = substream->runtime;
+ runtime->hw = snd_pcm_aica_playback_hw;
+ spu_enable();
+ dreamcastcard->clicks = 0;
+ dreamcastcard->current_period = 0;
+ dreamcastcard->dma_check = 0;
+ return 0;
+}
+
+static int snd_aicapcm_pcm_close(struct snd_pcm_substream
+ *substream)
+{
+ struct snd_card_aica *dreamcastcard = substream->pcm->private_data;
+ flush_workqueue(aica_queue);
+ if (dreamcastcard->timer.data)
+ del_timer(&dreamcastcard->timer);
+ kfree(dreamcastcard->channel);
+ spu_disable();
+ return 0;
+}
+
+static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream
+ *substream)
+{
+ /* Free the DMA buffer */
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream
+ *substream, struct snd_pcm_hw_params
+ *hw_params)
+{
+ /* Allocate a DMA buffer using ALSA built-ins */
+ return
+ snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream
+ *substream)
+{
+ struct snd_card_aica *dreamcastcard = substream->pcm->private_data;
+ if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE)
+ dreamcastcard->channel->sfmt = SM_16BIT;
+ dreamcastcard->channel->freq = substream->runtime->rate;
+ dreamcastcard->substream = substream;
+ return 0;
+}
+
+static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream
+ *substream, int cmd)
+{
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ spu_begin_dma(substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ aica_chn_halt();
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream
+ *substream)
+{
+ return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER);
+}
+
+static struct snd_pcm_ops snd_aicapcm_playback_ops = {
+ .open = snd_aicapcm_pcm_open,
+ .close = snd_aicapcm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_aicapcm_pcm_hw_params,
+ .hw_free = snd_aicapcm_pcm_hw_free,
+ .prepare = snd_aicapcm_pcm_prepare,
+ .trigger = snd_aicapcm_pcm_trigger,
+ .pointer = snd_aicapcm_pcm_pointer,
+};
+
+/* TO DO: set up to handle more than one pcm instance */
+static int __init snd_aicapcmchip(struct snd_card_aica
+ *dreamcastcard, int pcm_index)
+{
+ struct snd_pcm *pcm;
+ int err;
+ /* AICA has no capture ability */
+ err =
+ snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0,
+ &pcm);
+ if (unlikely(err < 0))
+ return err;
+ pcm->private_data = dreamcastcard;
+ strcpy(pcm->name, "AICA PCM");
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_aicapcm_playback_ops);
+ /* Allocate the DMA buffers */
+ err =
+ snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data
+ (GFP_KERNEL),
+ AICA_BUFFER_SIZE,
+ AICA_BUFFER_SIZE);
+ return err;
+}
+
+/* Mixer controls */
+#define aica_pcmswitch_info snd_ctl_boolean_mono_info
+
+static int aica_pcmswitch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 1; /* TO DO: Fix me */
+ return 0;
+}
+
+static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (ucontrol->value.integer.value[0] == 1)
+ return 0; /* TO DO: Fix me */
+ else
+ aica_chn_halt();
+ return 0;
+}
+
+static int aica_pcmvolume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xFF;
+ return 0;
+}
+
+static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_card_aica *dreamcastcard;
+ dreamcastcard = kcontrol->private_data;
+ if (unlikely(!dreamcastcard->channel))
+ return -ETXTBSY; /* we've not yet been set up */
+ ucontrol->value.integer.value[0] = dreamcastcard->channel->vol;
+ return 0;
+}
+
+static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_card_aica *dreamcastcard;
+ unsigned int vol;
+ dreamcastcard = kcontrol->private_data;
+ if (unlikely(!dreamcastcard->channel))
+ return -ETXTBSY;
+ vol = ucontrol->value.integer.value[0];
+ if (vol > 0xff)
+ return -EINVAL;
+ if (unlikely(dreamcastcard->channel->vol == vol))
+ return 0;
+ dreamcastcard->channel->vol = ucontrol->value.integer.value[0];
+ dreamcastcard->master_volume = ucontrol->value.integer.value[0];
+ spu_memload(AICA_CHANNEL0_CONTROL_OFFSET,
+ dreamcastcard->channel, sizeof(struct aica_channel));
+ return 1;
+}
+
+static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .index = 0,
+ .info = aica_pcmswitch_info,
+ .get = aica_pcmswitch_get,
+ .put = aica_pcmswitch_put
+};
+
+static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Volume",
+ .index = 0,
+ .info = aica_pcmvolume_info,
+ .get = aica_pcmvolume_get,
+ .put = aica_pcmvolume_put
+};
+
+static int load_aica_firmware(void)
+{
+ int err;
+ const struct firmware *fw_entry;
+ spu_reset();
+ err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev);
+ if (unlikely(err))
+ return err;
+ /* write firware into memory */
+ spu_disable();
+ spu_memload(0, fw_entry->data, fw_entry->size);
+ spu_enable();
+ release_firmware(fw_entry);
+ return err;
+}
+
+static int __devinit add_aicamixer_controls(struct snd_card_aica
+ *dreamcastcard)
+{
+ int err;
+ err = snd_ctl_add
+ (dreamcastcard->card,
+ snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard));
+ if (unlikely(err < 0))
+ return err;
+ err = snd_ctl_add
+ (dreamcastcard->card,
+ snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard));
+ if (unlikely(err < 0))
+ return err;
+ return 0;
+}
+
+static int __devexit snd_aica_remove(struct platform_device *devptr)
+{
+ struct snd_card_aica *dreamcastcard;
+ dreamcastcard = platform_get_drvdata(devptr);
+ if (unlikely(!dreamcastcard))
+ return -ENODEV;
+ snd_card_free(dreamcastcard->card);
+ kfree(dreamcastcard);
+ platform_set_drvdata(devptr, NULL);
+ return 0;
+}
+
+static int __devinit snd_aica_probe(struct platform_device *devptr)
+{
+ int err;
+ struct snd_card_aica *dreamcastcard;
+ dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL);
+ if (unlikely(!dreamcastcard))
+ return -ENOMEM;
+ dreamcastcard->card =
+ snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0);
+ if (unlikely(!dreamcastcard->card)) {
+ kfree(dreamcastcard);
+ return -ENODEV;
+ }
+ strcpy(dreamcastcard->card->driver, "snd_aica");
+ strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER);
+ strcpy(dreamcastcard->card->longname,
+ "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast");
+ /* Prepare to use the queue */
+ INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma);
+ /* Load the PCM 'chip' */
+ err = snd_aicapcmchip(dreamcastcard, 0);
+ if (unlikely(err < 0))
+ goto freedreamcast;
+ snd_card_set_dev(dreamcastcard->card, &devptr->dev);
+ dreamcastcard->timer.data = 0;
+ dreamcastcard->channel = NULL;
+ /* Add basic controls */
+ err = add_aicamixer_controls(dreamcastcard);
+ if (unlikely(err < 0))
+ goto freedreamcast;
+ /* Register the card with ALSA subsystem */
+ err = snd_card_register(dreamcastcard->card);
+ if (unlikely(err < 0))
+ goto freedreamcast;
+ platform_set_drvdata(devptr, dreamcastcard);
+ aica_queue = create_workqueue(CARD_NAME);
+ if (unlikely(!aica_queue))
+ goto freedreamcast;
+ snd_printk
+ ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n");
+ return 0;
+ freedreamcast:
+ snd_card_free(dreamcastcard->card);
+ kfree(dreamcastcard);
+ return err;
+}
+
+static struct platform_driver snd_aica_driver = {
+ .probe = snd_aica_probe,
+ .remove = __devexit_p(snd_aica_remove),
+ .driver = {
+ .name = SND_AICA_DRIVER},
+};
+
+static int __init aica_init(void)
+{
+ int err;
+ err = platform_driver_register(&snd_aica_driver);
+ if (unlikely(err < 0))
+ return err;
+ pd = platform_device_register_simple(SND_AICA_DRIVER, -1,
+ aica_memory_space, 2);
+ if (IS_ERR(pd)) {
+ platform_driver_unregister(&snd_aica_driver);
+ return PTR_ERR(pd);
+ }
+ /* Load the firmware */
+ return load_aica_firmware();
+}
+
+static void __exit aica_exit(void)
+{
+ /* Destroy the aica kernel thread *
+ * being extra cautious to check if it exists*/
+ if (likely(aica_queue))
+ destroy_workqueue(aica_queue);
+ platform_device_unregister(pd);
+ platform_driver_unregister(&snd_aica_driver);
+ /* Kill any sound still playing and reset ARM7 to safe state */
+ spu_reset();
+}
+
+module_init(aica_init);
+module_exit(aica_exit);
diff --git a/sound/sh/aica.h b/sound/sh/aica.h
new file mode 100644
index 0000000..d098baa
--- /dev/null
+++ b/sound/sh/aica.h
@@ -0,0 +1,81 @@
+/* aica.h
+ * Header file for ALSA driver for
+ * Sega Dreamcast Yamaha AICA sound
+ * Copyright Adrian McMenamin
+ * <adrian@mcmen.demon.co.uk>
+ * 2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as published by
+ * the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/* SPU memory and register constants etc */
+#define G2_FIFO 0xa05f688c
+#define SPU_MEMORY_BASE 0xA0800000
+#define ARM_RESET_REGISTER 0xA0702C00
+#define SPU_REGISTER_BASE 0xA0700000
+
+/* AICA channels stuff */
+#define AICA_CONTROL_POINT 0xA0810000
+#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008
+#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004
+
+/* Command values */
+#define AICA_CMD_KICK 0x80000000
+#define AICA_CMD_NONE 0
+#define AICA_CMD_START 1
+#define AICA_CMD_STOP 2
+#define AICA_CMD_VOL 3
+
+/* Sound modes */
+#define SM_8BIT 1
+#define SM_16BIT 0
+#define SM_ADPCM 2
+
+/* Buffer and period size */
+#define AICA_BUFFER_SIZE 0x8000
+#define AICA_PERIOD_SIZE 0x800
+#define AICA_PERIOD_NUMBER 16
+
+#define AICA_CHANNEL0_OFFSET 0x11000
+#define AICA_CHANNEL1_OFFSET 0x21000
+#define CHANNEL_OFFSET 0x10000
+
+#define AICA_DMA_CHANNEL 5
+#define AICA_DMA_MODE 5
+
+#define SND_AICA_DRIVER "AICA"
+
+struct aica_channel {
+ uint32_t cmd; /* Command ID */
+ uint32_t pos; /* Sample position */
+ uint32_t length; /* Sample length */
+ uint32_t freq; /* Frequency */
+ uint32_t vol; /* Volume 0-255 */
+ uint32_t pan; /* Pan 0-255 */
+ uint32_t sfmt; /* Sound format */
+ uint32_t flags; /* Bit flags */
+};
+
+struct snd_card_aica {
+ struct work_struct spu_dma_work;
+ struct snd_card *card;
+ struct aica_channel *channel;
+ struct snd_pcm_substream *substream;
+ int clicks;
+ int current_period;
+ struct timer_list timer;
+ int master_volume;
+ int dma_check;
+};
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
new file mode 100644
index 0000000..4dfda66
--- /dev/null
+++ b/sound/soc/Kconfig
@@ -0,0 +1,41 @@
+#
+# SoC audio configuration
+#
+
+menuconfig SND_SOC
+ tristate "ALSA for SoC audio support"
+ select SND_PCM
+ select AC97_BUS if SND_SOC_AC97_BUS
+ ---help---
+
+ If you want ASoC support, you should say Y here and also to the
+ specific driver for your SoC platform below.
+
+ ASoC provides power efficient ALSA support for embedded battery powered
+ SoC based systems like PDA's, Phones and Personal Media Players.
+
+ This ASoC audio support can also be built as a module. If so, the module
+ will be called snd-soc-core.
+
+if SND_SOC
+
+config SND_SOC_AC97_BUS
+ bool
+
+# All the supported Soc's
+source "sound/soc/at32/Kconfig"
+source "sound/soc/at91/Kconfig"
+source "sound/soc/au1x/Kconfig"
+source "sound/soc/pxa/Kconfig"
+source "sound/soc/s3c24xx/Kconfig"
+source "sound/soc/sh/Kconfig"
+source "sound/soc/fsl/Kconfig"
+source "sound/soc/davinci/Kconfig"
+source "sound/soc/omap/Kconfig"
+source "sound/soc/blackfin/Kconfig"
+
+# Supported codecs
+source "sound/soc/codecs/Kconfig"
+
+endif # SND_SOC
+
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
new file mode 100644
index 0000000..d849349
--- /dev/null
+++ b/sound/soc/Makefile
@@ -0,0 +1,5 @@
+snd-soc-core-objs := soc-core.o soc-dapm.o
+
+obj-$(CONFIG_SND_SOC) += snd-soc-core.o
+obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
+obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/
diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig
new file mode 100644
index 0000000..b0765e8
--- /dev/null
+++ b/sound/soc/at32/Kconfig
@@ -0,0 +1,34 @@
+config SND_AT32_SOC
+ tristate "SoC Audio for the Atmel AT32 System-on-a-Chip"
+ depends on AVR32 && SND_SOC
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the AT32 SSC interface. You will also need to
+ to select the audio interfaces to support below.
+
+
+config SND_AT32_SOC_SSC
+ tristate
+
+
+
+config SND_AT32_SOC_PLAYPAQ
+ tristate "SoC Audio support for PlayPaq with WM8510"
+ depends on SND_AT32_SOC && BOARD_PLAYPAQ
+ select SND_AT32_SOC_SSC
+ select SND_SOC_WM8510
+ help
+ Say Y or M here if you want to add support for SoC audio
+ on the LRS PlayPaq.
+
+
+
+config SND_AT32_SOC_PLAYPAQ_SLAVE
+ bool "Run CODEC on PlayPaq in slave mode"
+ depends on SND_AT32_SOC_PLAYPAQ
+ default n
+ help
+ Say Y if you want to run with the AT32 SSC generating the BCLK
+ and FRAME signals on the PlayPaq. Unless you want to play
+ with the AT32 as the SSC master, you probably want to say N here,
+ as this will give you better sound quality.
diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile
new file mode 100644
index 0000000..c03e55e
--- /dev/null
+++ b/sound/soc/at32/Makefile
@@ -0,0 +1,11 @@
+# AT32 Platform Support
+snd-soc-at32-objs := at32-pcm.o
+snd-soc-at32-ssc-objs := at32-ssc.o
+
+obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o
+obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o
+
+# AT32 Machine Support
+snd-soc-playpaq-objs := playpaq_wm8510.o
+
+obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c
new file mode 100644
index 0000000..c83584f
--- /dev/null
+++ b/sound/soc/at32/at32-pcm.c
@@ -0,0 +1,492 @@
+/* sound/soc/at32/at32-pcm.c
+ * ASoC PCM interface for Atmel AT32 SoC
+ *
+ * Copyright (C) 2008 Long Range Systems
+ * Geoffrey Wossum <gwossum@acm.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Note that this is basically a port of the sound/soc/at91-pcm.c to
+ * the AVR32 kernel. Thanks to Frank Mandarino for that code.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/atmel_pdc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "at32-pcm.h"
+
+
+
+/*--------------------------------------------------------------------------*\
+ * Hardware definition
+\*--------------------------------------------------------------------------*/
+/* TODO: These values were taken from the AT91 platform driver, check
+ * them against real values for AT32
+ */
+static const struct snd_pcm_hardware at32_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE),
+
+ .formats = SNDRV_PCM_FMTBIT_S16,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */
+ .periods_min = 2,
+ .periods_max = 1024,
+ .buffer_bytes_max = 32 * 1024,
+};
+
+
+
+/*--------------------------------------------------------------------------*\
+ * Data types
+\*--------------------------------------------------------------------------*/
+struct at32_runtime_data {
+ struct at32_pcm_dma_params *params;
+ dma_addr_t dma_buffer; /* physical address of DMA buffer */
+ dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */
+ size_t period_size;
+
+ dma_addr_t period_ptr; /* physical address of next period */
+ int periods; /* period index of period_ptr */
+
+ /* Save PDC registers (for power management) */
+ u32 pdc_xpr_save;
+ u32 pdc_xcr_save;
+ u32 pdc_xnpr_save;
+ u32 pdc_xncr_save;
+};
+
+
+
+/*--------------------------------------------------------------------------*\
+ * Helper functions
+\*--------------------------------------------------------------------------*/
+static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *dmabuf = &substream->dma_buffer;
+ size_t size = at32_pcm_hardware.buffer_bytes_max;
+
+ dmabuf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dmabuf->dev.dev = pcm->card->dev;
+ dmabuf->private_data = NULL;
+ dmabuf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &dmabuf->addr, GFP_KERNEL);
+ pr_debug("at32_pcm: preallocate_dma_buffer: "
+ "area=%p, addr=%p, size=%ld\n",
+ (void *)dmabuf->area, (void *)dmabuf->addr, size);
+
+ if (!dmabuf->area)
+ return -ENOMEM;
+
+ dmabuf->bytes = size;
+ return 0;
+}
+
+
+
+/*--------------------------------------------------------------------------*\
+ * ISR
+\*--------------------------------------------------------------------------*/
+static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *rtd = substream->runtime;
+ struct at32_runtime_data *prtd = rtd->private_data;
+ struct at32_pcm_dma_params *params = prtd->params;
+ static int count;
+
+ count++;
+ if (ssc_sr & params->mask->ssc_endbuf) {
+ pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ "underrun" : "overrun", params->name, ssc_sr, count);
+
+ /* re-start the PDC */
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end)
+ prtd->period_ptr = prtd->dma_buffer;
+
+
+ ssc_writex(params->ssc->regs, params->pdc->xpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_enable);
+ }
+
+
+ if (ssc_sr & params->mask->ssc_endx) {
+ /* Load the PDC next pointer and counter registers */
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end)
+ prtd->period_ptr = prtd->dma_buffer;
+ ssc_writex(params->ssc->regs, params->pdc->xnpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+ }
+
+
+ snd_pcm_period_elapsed(substream);
+}
+
+
+
+/*--------------------------------------------------------------------------*\
+ * PCM operations
+\*--------------------------------------------------------------------------*/
+static int at32_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at32_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* this may get called several times by oss emulation
+ * with different params
+ */
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ prtd->params = rtd->dai->cpu_dai->dma_data;
+ prtd->params->dma_intr_handler = at32_pcm_dma_irq;
+
+ prtd->dma_buffer = runtime->dma_addr;
+ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
+ prtd->period_size = params_period_bytes(params);
+
+ pr_debug("hw_params: DMA for %s initialized "
+ "(dma_bytes=%ld, period_size=%ld)\n",
+ prtd->params->name, runtime->dma_bytes, prtd->period_size);
+
+ return 0;
+}
+
+
+
+static int at32_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct at32_runtime_data *prtd = substream->runtime->private_data;
+ struct at32_pcm_dma_params *params = prtd->params;
+
+ if (params != NULL) {
+ ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+ params->mask->pdc_disable);
+ prtd->params->dma_intr_handler = NULL;
+ }
+
+ return 0;
+}
+
+
+
+static int at32_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct at32_runtime_data *prtd = substream->runtime->private_data;
+ struct at32_pcm_dma_params *params = prtd->params;
+
+ ssc_writex(params->ssc->regs, SSC_IDR,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
+
+ return 0;
+}
+
+
+static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *rtd = substream->runtime;
+ struct at32_runtime_data *prtd = rtd->private_data;
+ struct at32_pcm_dma_params *params = prtd->params;
+ int ret = 0;
+
+ pr_debug("at32_pcm_trigger: buffer_size = %ld, "
+ "dma_area = %p, dma_bytes = %ld\n",
+ rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->period_ptr = prtd->dma_buffer;
+
+ ssc_writex(params->ssc->regs, params->pdc->xpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ prtd->period_ptr += prtd->period_size;
+ ssc_writex(params->ssc->regs, params->pdc->xnpr,
+ prtd->period_ptr);
+ ssc_writex(params->ssc->regs, params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ pr_debug("trigger: period_ptr=%lx, xpr=%x, "
+ "xcr=%d, xnpr=%x, xncr=%d\n",
+ (unsigned long)prtd->period_ptr,
+ ssc_readx(params->ssc->regs, params->pdc->xpr),
+ ssc_readx(params->ssc->regs, params->pdc->xcr),
+ ssc_readx(params->ssc->regs, params->pdc->xnpr),
+ ssc_readx(params->ssc->regs, params->pdc->xncr));
+
+ ssc_writex(params->ssc->regs, SSC_IER,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+ ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+ params->mask->pdc_enable);
+
+ pr_debug("sr=%x, imr=%x\n",
+ ssc_readx(params->ssc->regs, SSC_SR),
+ ssc_readx(params->ssc->regs, SSC_IER));
+ break; /* SNDRV_PCM_TRIGGER_START */
+
+
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
+ break;
+
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_enable);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+
+
+static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at32_runtime_data *prtd = runtime->private_data;
+ struct at32_pcm_dma_params *params = prtd->params;
+ dma_addr_t ptr;
+ snd_pcm_uframes_t x;
+
+ ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr);
+ x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
+
+ if (x == runtime->buffer_size)
+ x = 0;
+
+ return x;
+}
+
+
+
+static int at32_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at32_runtime_data *prtd;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware);
+
+ /* ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ runtime->private_data = prtd;
+
+
+out:
+ return ret;
+}
+
+
+
+static int at32_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct at32_runtime_data *prtd = substream->runtime->private_data;
+
+ kfree(prtd);
+ return 0;
+}
+
+
+static int at32_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ return remap_pfn_range(vma, vma->vm_start,
+ substream->dma_buffer.addr >> PAGE_SHIFT,
+ vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+
+
+static struct snd_pcm_ops at32_pcm_ops = {
+ .open = at32_pcm_open,
+ .close = at32_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = at32_pcm_hw_params,
+ .hw_free = at32_pcm_hw_free,
+ .prepare = at32_pcm_prepare,
+ .trigger = at32_pcm_trigger,
+ .pointer = at32_pcm_pointer,
+ .mmap = at32_pcm_mmap,
+};
+
+
+
+/*--------------------------------------------------------------------------*\
+ * ASoC platform driver
+\*--------------------------------------------------------------------------*/
+static u64 at32_pcm_dmamask = 0xffffffff;
+
+static int at32_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &at32_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = at32_pcm_preallocate_dma_buffer(
+ pcm, SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n");
+ ret = at32_pcm_preallocate_dma_buffer(
+ pcm, SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+
+
+out:
+ return ret;
+}
+
+
+
+static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (substream == NULL)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+
+
+#ifdef CONFIG_PM
+static int at32_pcm_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct at32_runtime_data *prtd;
+ struct at32_pcm_dma_params *params;
+
+ if (runtime == NULL)
+ return 0;
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* Disable the PDC and save the PDC registers */
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+ params->mask->pdc_disable);
+
+ prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
+ prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
+ prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr);
+ prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr);
+
+ return 0;
+}
+
+
+
+static int at32_pcm_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct at32_runtime_data *prtd;
+ struct at32_pcm_dma_params *params;
+
+ if (runtime == NULL)
+ return 0;
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* Restore the PDC registers and enable the PDC */
+ ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save);
+ ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save);
+ ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
+ ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
+
+ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable);
+ return 0;
+}
+#else /* CONFIG_PM */
+# define at32_pcm_suspend NULL
+# define at32_pcm_resume NULL
+#endif /* CONFIG_PM */
+
+
+
+struct snd_soc_platform at32_soc_platform = {
+ .name = "at32-audio",
+ .pcm_ops = &at32_pcm_ops,
+ .pcm_new = at32_pcm_new,
+ .pcm_free = at32_pcm_free_dma_buffers,
+ .suspend = at32_pcm_suspend,
+ .resume = at32_pcm_resume,
+};
+EXPORT_SYMBOL_GPL(at32_soc_platform);
+
+
+
+MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
+MODULE_DESCRIPTION("Atmel AT32 PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h
new file mode 100644
index 0000000..2a52430
--- /dev/null
+++ b/sound/soc/at32/at32-pcm.h
@@ -0,0 +1,79 @@
+/* sound/soc/at32/at32-pcm.h
+ * ASoC PCM interface for Atmel AT32 SoC
+ *
+ * Copyright (C) 2008 Long Range Systems
+ * Geoffrey Wossum <gwossum@acm.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __SOUND_SOC_AT32_AT32_PCM_H
+#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__
+
+#include <linux/atmel-ssc.h>
+
+
+/*
+ * Registers and status bits that are required by the PCM driver
+ * TODO: Is ptcr really used?
+ */
+struct at32_pdc_regs {
+ u32 xpr; /* PDC RX/TX pointer */
+ u32 xcr; /* PDC RX/TX counter */
+ u32 xnpr; /* PDC next RX/TX pointer */
+ u32 xncr; /* PDC next RX/TX counter */
+ u32 ptcr; /* PDC transfer control */
+};
+
+
+
+/*
+ * SSC mask info
+ */
+struct at32_ssc_mask {
+ u32 ssc_enable; /* SSC RX/TX enable */
+ u32 ssc_disable; /* SSC RX/TX disable */
+ u32 ssc_endx; /* SSC ENDTX or ENDRX */
+ u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */
+ u32 pdc_enable; /* PDC RX/TX enable */
+ u32 pdc_disable; /* PDC RX/TX disable */
+};
+
+
+
+/*
+ * This structure, shared between the PCM driver and the interface,
+ * contains all information required by the PCM driver to perform the
+ * PDC DMA operation. All fields except dma_intr_handler() are initialized
+ * by the interface. The dms_intr_handler() pointer is set by the PCM
+ * driver and called by the interface SSC interrupt handler if it is
+ * non-NULL.
+ */
+struct at32_pcm_dma_params {
+ char *name; /* stream identifier */
+ int pdc_xfer_size; /* PDC counter increment in bytes */
+ struct ssc_device *ssc; /* SSC device for stream */
+ struct at32_pdc_regs *pdc; /* PDC register info */
+ struct at32_ssc_mask *mask; /* SSC mask info */
+ struct snd_pcm_substream *substream;
+ void (*dma_intr_handler) (u32, struct snd_pcm_substream *);
+};
+
+
+
+/*
+ * The AT32 ASoC platform driver
+ */
+extern struct snd_soc_platform at32_soc_platform;
+
+
+
+/*
+ * SSC register access (since ssc_writel() / ssc_readl() require literal name)
+ */
+#define ssc_readx(base, reg) (__raw_readl((base) + (reg)))
+#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg))
+
+#endif /* __SOUND_SOC_AT32_AT32_PCM_H */
diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c
new file mode 100644
index 0000000..4ef6492
--- /dev/null
+++ b/sound/soc/at32/at32-ssc.c
@@ -0,0 +1,849 @@
+/* sound/soc/at32/at32-ssc.c
+ * ASoC platform driver for AT32 using SSC as DAI
+ *
+ * Copyright (C) 2008 Long Range Systems
+ * Geoffrey Wossum <gwossum@acm.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Note that this is basically a port of the sound/soc/at91-ssc.c to
+ * the AVR32 kernel. Thanks to Frank Mandarino for that code.
+ */
+
+/* #define DEBUG */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+#include <linux/atmel_pdc.h>
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "at32-pcm.h"
+#include "at32-ssc.h"
+
+
+
+/*-------------------------------------------------------------------------*\
+ * Constants
+\*-------------------------------------------------------------------------*/
+#define NUM_SSC_DEVICES 3
+
+/*
+ * SSC direction masks
+ */
+#define SSC_DIR_MASK_UNUSED 0
+#define SSC_DIR_MASK_PLAYBACK 1
+#define SSC_DIR_MASK_CAPTURE 2
+
+/*
+ * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These
+ * are expected to be used with SSC_BF
+ */
+/* START bit field values */
+#define SSC_START_CONTINUOUS 0
+#define SSC_START_TX_RX 1
+#define SSC_START_LOW_RF 2
+#define SSC_START_HIGH_RF 3
+#define SSC_START_FALLING_RF 4
+#define SSC_START_RISING_RF 5
+#define SSC_START_LEVEL_RF 6
+#define SSC_START_EDGE_RF 7
+#define SSS_START_COMPARE_0 8
+
+/* CKI bit field values */
+#define SSC_CKI_FALLING 0
+#define SSC_CKI_RISING 1
+
+/* CKO bit field values */
+#define SSC_CKO_NONE 0
+#define SSC_CKO_CONTINUOUS 1
+#define SSC_CKO_TRANSFER 2
+
+/* CKS bit field values */
+#define SSC_CKS_DIV 0
+#define SSC_CKS_CLOCK 1
+#define SSC_CKS_PIN 2
+
+/* FSEDGE bit field values */
+#define SSC_FSEDGE_POSITIVE 0
+#define SSC_FSEDGE_NEGATIVE 1
+
+/* FSOS bit field values */
+#define SSC_FSOS_NONE 0
+#define SSC_FSOS_NEGATIVE 1
+#define SSC_FSOS_POSITIVE 2
+#define SSC_FSOS_LOW 3
+#define SSC_FSOS_HIGH 4
+#define SSC_FSOS_TOGGLE 5
+
+#define START_DELAY 1
+
+
+
+/*-------------------------------------------------------------------------*\
+ * Module data
+\*-------------------------------------------------------------------------*/
+/*
+ * SSC PDC registered required by the PCM DMA engine
+ */
+static struct at32_pdc_regs pdc_tx_reg = {
+ .xpr = SSC_PDC_TPR,
+ .xcr = SSC_PDC_TCR,
+ .xnpr = SSC_PDC_TNPR,
+ .xncr = SSC_PDC_TNCR,
+};
+
+
+
+static struct at32_pdc_regs pdc_rx_reg = {
+ .xpr = SSC_PDC_RPR,
+ .xcr = SSC_PDC_RCR,
+ .xnpr = SSC_PDC_RNPR,
+ .xncr = SSC_PDC_RNCR,
+};
+
+
+
+/*
+ * SSC and PDC status bits for transmit and receive
+ */
+static struct at32_ssc_mask ssc_tx_mask = {
+ .ssc_enable = SSC_BIT(CR_TXEN),
+ .ssc_disable = SSC_BIT(CR_TXDIS),
+ .ssc_endx = SSC_BIT(SR_ENDTX),
+ .ssc_endbuf = SSC_BIT(SR_TXBUFE),
+ .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN),
+ .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS),
+};
+
+
+
+static struct at32_ssc_mask ssc_rx_mask = {
+ .ssc_enable = SSC_BIT(CR_RXEN),
+ .ssc_disable = SSC_BIT(CR_RXDIS),
+ .ssc_endx = SSC_BIT(SR_ENDRX),
+ .ssc_endbuf = SSC_BIT(SR_RXBUFF),
+ .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN),
+ .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS),
+};
+
+
+
+/*
+ * DMA parameters for each SSC
+ */
+static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
+ {
+ {
+ .name = "SSC0 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC0 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ },
+ },
+ {
+ {
+ .name = "SSC1 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC1 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ },
+ },
+ {
+ {
+ .name = "SSC2 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC2 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ },
+ },
+};
+
+
+
+static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = {
+ {
+ .name = "ssc0",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
+ .dir_mask = SSC_DIR_MASK_UNUSED,
+ .initialized = 0,
+ },
+ {
+ .name = "ssc1",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
+ .dir_mask = SSC_DIR_MASK_UNUSED,
+ .initialized = 0,
+ },
+ {
+ .name = "ssc2",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
+ .dir_mask = SSC_DIR_MASK_UNUSED,
+ .initialized = 0,
+ },
+};
+
+
+
+
+/*-------------------------------------------------------------------------*\
+ * ISR
+\*-------------------------------------------------------------------------*/
+/*
+ * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt
+ * handler in the PCM driver.
+ */
+static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id)
+{
+ struct at32_ssc_info *ssc_p = dev_id;
+ struct at32_pcm_dma_params *dma_params;
+ u32 ssc_sr;
+ u32 ssc_substream_mask;
+ int i;
+
+ ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) &
+ ssc_readl(ssc_p->ssc->regs, IMR));
+
+ /*
+ * Loop through substreams attached to this SSC. If a DMA-related
+ * interrupt occured on that substream, call the DMA interrupt
+ * handler function, if one has been registered in the dma_param
+ * structure by the PCM driver.
+ */
+ for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
+ dma_params = ssc_p->dma_params[i];
+
+ if ((dma_params != NULL) &&
+ (dma_params->dma_intr_handler != NULL)) {
+ ssc_substream_mask = (dma_params->mask->ssc_endx |
+ dma_params->mask->ssc_endbuf);
+ if (ssc_sr & ssc_substream_mask) {
+ dma_params->dma_intr_handler(ssc_sr,
+ dma_params->
+ substream);
+ }
+ }
+ }
+
+
+ return IRQ_HANDLED;
+}
+
+/*-------------------------------------------------------------------------*\
+ * DAI functions
+\*-------------------------------------------------------------------------*/
+/*
+ * Startup. Only that one substream allowed in each direction.
+ */
+static int at32_ssc_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ int dir_mask;
+
+ dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE);
+
+ spin_lock_irq(&ssc_p->lock);
+ if (ssc_p->dir_mask & dir_mask) {
+ spin_unlock_irq(&ssc_p->lock);
+ return -EBUSY;
+ }
+ ssc_p->dir_mask |= dir_mask;
+ spin_unlock_irq(&ssc_p->lock);
+
+ return 0;
+}
+
+
+
+/*
+ * Shutdown. Clear DMA parameters and shutdown the SSC if there
+ * are no other substreams open.
+ */
+static void at32_ssc_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct at32_pcm_dma_params *dma_params;
+ int dir_mask;
+
+ dma_params = ssc_p->dma_params[substream->stream];
+
+ if (dma_params != NULL) {
+ ssc_writel(dma_params->ssc->regs, CR,
+ dma_params->mask->ssc_disable);
+ pr_debug("%s disabled SSC_SR=0x%08x\n",
+ (substream->stream ? "receiver" : "transmit"),
+ ssc_readl(ssc_p->ssc->regs, SR));
+
+ dma_params->ssc = NULL;
+ dma_params->substream = NULL;
+ ssc_p->dma_params[substream->stream] = NULL;
+ }
+
+
+ dir_mask = 1 << substream->stream;
+ spin_lock_irq(&ssc_p->lock);
+ ssc_p->dir_mask &= ~dir_mask;
+ if (!ssc_p->dir_mask) {
+ /* Shutdown the SSC clock */
+ pr_debug("at32-ssc: Stopping user %d clock\n",
+ ssc_p->ssc->user);
+ clk_disable(ssc_p->ssc->clk);
+
+ if (ssc_p->initialized) {
+ free_irq(ssc_p->ssc->irq, ssc_p);
+ ssc_p->initialized = 0;
+ }
+
+ /* Reset the SSC */
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+
+ /* clear the SSC dividers */
+ ssc_p->cmr_div = 0;
+ ssc_p->tcmr_period = 0;
+ ssc_p->rcmr_period = 0;
+ }
+ spin_unlock_irq(&ssc_p->lock);
+}
+
+
+
+/*
+ * Set the SSC system clock rate
+ */
+static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ /* TODO: What the heck do I do here? */
+ return 0;
+}
+
+
+
+/*
+ * Record DAI format for use by hw_params()
+ */
+static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ ssc_p->daifmt = fmt;
+ return 0;
+}
+
+
+
+/*
+ * Record SSC clock dividers for use in hw_params()
+ */
+static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ switch (div_id) {
+ case AT32_SSC_CMR_DIV:
+ /*
+ * The same master clock divider is used for both
+ * transmit and receive, so if a value has already
+ * been set, it must match this value
+ */
+ if (ssc_p->cmr_div == 0)
+ ssc_p->cmr_div = div;
+ else if (div != ssc_p->cmr_div)
+ return -EBUSY;
+ break;
+
+ case AT32_SSC_TCMR_PERIOD:
+ ssc_p->tcmr_period = div;
+ break;
+
+ case AT32_SSC_RCMR_PERIOD:
+ ssc_p->rcmr_period = div;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+
+/*
+ * Configure the SSC
+ */
+static int at32_ssc_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int id = rtd->dai->cpu_dai->id;
+ struct at32_ssc_info *ssc_p = &ssc_info[id];
+ struct at32_pcm_dma_params *dma_params;
+ int channels, bits;
+ u32 tfmr, rfmr, tcmr, rcmr;
+ int start_event;
+ int ret;
+
+
+ /*
+ * Currently, there is only one set of dma_params for each direction.
+ * If more are added, this code will have to be changed to select
+ * the proper set
+ */
+ dma_params = &ssc_dma_params[id][substream->stream];
+ dma_params->ssc = ssc_p->ssc;
+ dma_params->substream = substream;
+
+ ssc_p->dma_params[substream->stream] = dma_params;
+
+
+ /*
+ * The cpu_dai->dma_data field is only used to communicate the
+ * appropriate DMA parameters to the PCM driver's hw_params()
+ * function. It should not be used for other purposes as it
+ * is common to all substreams.
+ */
+ rtd->dai->cpu_dai->dma_data = dma_params;
+
+ channels = params_channels(params);
+
+
+ /*
+ * Determine sample size in bits and the PDC increment
+ */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ bits = 8;
+ dma_params->pdc_xfer_size = 1;
+ break;
+
+ case SNDRV_PCM_FORMAT_S16:
+ bits = 16;
+ dma_params->pdc_xfer_size = 2;
+ break;
+
+ case SNDRV_PCM_FORMAT_S24:
+ bits = 24;
+ dma_params->pdc_xfer_size = 4;
+ break;
+
+ case SNDRV_PCM_FORMAT_S32:
+ bits = 32;
+ dma_params->pdc_xfer_size = 4;
+ break;
+
+ default:
+ pr_warning("at32-ssc: Unsupported PCM format %d",
+ params_format(params));
+ return -EINVAL;
+ }
+ pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n",
+ bits, dma_params->pdc_xfer_size, channels);
+
+
+ /*
+ * The SSC only supports up to 16-bit samples in I2S format, due
+ * to the size of the Frame Mode Register FSLEN field.
+ */
+ if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S)
+ if (bits > 16) {
+ pr_warning("at32-ssc: "
+ "sample size %d is too large for I2S\n",
+ bits);
+ return -EINVAL;
+ }
+
+
+ /*
+ * Compute the SSC register settings
+ */
+ switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+ SND_SOC_DAIFMT_MASTER_MASK)) {
+ case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+ /*
+ * I2S format, SSC provides BCLK and LRS clocks.
+ *
+ * The SSC transmit and receive clocks are generated from the
+ * MCK divider, and the BCLK signal is output on the SSC TK line
+ */
+ pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n");
+ rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
+ SSC_BF(RCMR_STTDLY, START_DELAY) |
+ SSC_BF(RCMR_START, SSC_START_FALLING_RF) |
+ SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
+ SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
+ SSC_BF(RCMR_CKS, SSC_CKS_DIV));
+
+ rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
+ SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) |
+ SSC_BF(RFMR_FSLEN, bits - 1) |
+ SSC_BF(RFMR_DATNB, channels - 1) |
+ SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
+
+ tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
+ SSC_BF(TCMR_STTDLY, START_DELAY) |
+ SSC_BF(TCMR_START, SSC_START_FALLING_RF) |
+ SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
+ SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
+ SSC_BF(TCMR_CKS, SSC_CKS_DIV));
+
+ tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
+ SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) |
+ SSC_BF(TFMR_FSLEN, bits - 1) |
+ SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) |
+ SSC_BF(TFMR_DATLEN, bits - 1));
+ break;
+
+
+ case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
+ /*
+ * I2S format, CODEC supplies BCLK and LRC clock.
+ *
+ * The SSC transmit clock is obtained from the BCLK signal
+ * on the TK line, and the SSC receive clock is generated from
+ * the transmit clock.
+ *
+ * For single channel data, one sample is transferred on the
+ * falling edge of the LRC clock. For two channel data, one
+ * sample is transferred on both edges of the LRC clock.
+ */
+ pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n");
+ start_event = ((channels == 1) ?
+ SSC_START_FALLING_RF : SSC_START_EDGE_RF);
+
+ rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) |
+ SSC_BF(RCMR_START, start_event) |
+ SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
+ SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
+ SSC_BF(RCMR_CKS, SSC_CKS_CLOCK));
+
+ rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
+ SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) |
+ SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
+
+ tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) |
+ SSC_BF(TCMR_START, start_event) |
+ SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
+ SSC_BF(TCMR_CKO, SSC_CKO_NONE) |
+ SSC_BF(TCMR_CKS, SSC_CKS_PIN));
+
+ tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
+ SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) |
+ SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
+ break;
+
+
+ case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+ /*
+ * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
+ *
+ * The SSC transmit and receive clocks are generated from the
+ * MCK divider, and the BCLK signal is output on the SSC TK line
+ */
+ pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n");
+ rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
+ SSC_BF(RCMR_STTDLY, 1) |
+ SSC_BF(RCMR_START, SSC_START_RISING_RF) |
+ SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
+ SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
+ SSC_BF(RCMR_CKS, SSC_CKS_DIV));
+
+ rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
+ SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) |
+ SSC_BF(RFMR_DATNB, channels - 1) |
+ SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
+
+ tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
+ SSC_BF(TCMR_STTDLY, 1) |
+ SSC_BF(TCMR_START, SSC_START_RISING_RF) |
+ SSC_BF(TCMR_CKI, SSC_CKI_RISING) |
+ SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
+ SSC_BF(TCMR_CKS, SSC_CKS_DIV));
+
+ tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
+ SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) |
+ SSC_BF(TFMR_DATNB, channels - 1) |
+ SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
+ break;
+
+
+ case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+ default:
+ pr_warning("at32-ssc: unsupported DAI format 0x%x\n",
+ ssc_p->daifmt);
+ return -EINVAL;
+ break;
+ }
+ pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
+ rcmr, rfmr, tcmr, tfmr);
+
+
+ if (!ssc_p->initialized) {
+ /* enable peripheral clock */
+ pr_debug("at32-ssc: Starting clock\n");
+ clk_enable(ssc_p->ssc->clk);
+
+ /* Reset the SSC and its PDC registers */
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+
+ ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
+
+ ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
+ ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
+
+ ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0,
+ ssc_p->name, ssc_p);
+ if (ret < 0) {
+ pr_warning("at32-ssc: request irq failed (%d)\n", ret);
+ pr_debug("at32-ssc: Stopping clock\n");
+ clk_disable(ssc_p->ssc->clk);
+ return ret;
+ }
+
+ ssc_p->initialized = 1;
+ }
+
+ /* Set SSC clock mode register */
+ ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div);
+
+ /* set receive clock mode and format */
+ ssc_writel(ssc_p->ssc->regs, RCMR, rcmr);
+ ssc_writel(ssc_p->ssc->regs, RFMR, rfmr);
+
+ /* set transmit clock mode and format */
+ ssc_writel(ssc_p->ssc->regs, TCMR, tcmr);
+ ssc_writel(ssc_p->ssc->regs, TFMR, tfmr);
+
+ pr_debug("at32-ssc: SSC initialized\n");
+ return 0;
+}
+
+
+
+static int at32_ssc_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct at32_pcm_dma_params *dma_params;
+
+ dma_params = ssc_p->dma_params[substream->stream];
+
+ ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable);
+
+ return 0;
+}
+
+
+
+#ifdef CONFIG_PM
+static int at32_ssc_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct at32_ssc_info *ssc_p;
+
+ if (!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ /* Save the status register before disabling transmit and receive */
+ ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR);
+ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS));
+
+ /* Save the current interrupt mask, then disable unmasked interrupts */
+ ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR);
+ ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr);
+
+ ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR);
+ ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR);
+ ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR);
+ ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR);
+ ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR);
+
+ return 0;
+}
+
+
+
+static int at32_ssc_resume(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct at32_ssc_info *ssc_p;
+ u32 cr;
+
+ if (!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ /* restore SSC register settings */
+ ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr);
+ ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr);
+ ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr);
+ ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr);
+ ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr);
+
+ /* re-enable interrupts */
+ ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
+
+ /* Re-enable recieve and transmit as appropriate */
+ cr = 0;
+ cr |=
+ (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
+ cr |=
+ (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0;
+ ssc_writel(ssc_p->ssc->regs, CR, cr);
+
+ return 0;
+}
+#else /* CONFIG_PM */
+# define at32_ssc_suspend NULL
+# define at32_ssc_resume NULL
+#endif /* CONFIG_PM */
+
+
+#define AT32_SSC_RATES \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+
+#define AT32_SSC_FORMATS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \
+ SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32)
+
+
+struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = {
+ {
+ .name = "at32-ssc0",
+ .id = 0,
+ .type = SND_SOC_DAI_PCM,
+ .suspend = at32_ssc_suspend,
+ .resume = at32_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT32_SSC_RATES,
+ .formats = AT32_SSC_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT32_SSC_RATES,
+ .formats = AT32_SSC_FORMATS,
+ },
+ .ops = {
+ .startup = at32_ssc_startup,
+ .shutdown = at32_ssc_shutdown,
+ .prepare = at32_ssc_prepare,
+ .hw_params = at32_ssc_hw_params,
+ },
+ .dai_ops = {
+ .set_sysclk = at32_ssc_set_dai_sysclk,
+ .set_fmt = at32_ssc_set_dai_fmt,
+ .set_clkdiv = at32_ssc_set_dai_clkdiv,
+ },
+ .private_data = &ssc_info[0],
+ },
+ {
+ .name = "at32-ssc1",
+ .id = 1,
+ .type = SND_SOC_DAI_PCM,
+ .suspend = at32_ssc_suspend,
+ .resume = at32_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT32_SSC_RATES,
+ .formats = AT32_SSC_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT32_SSC_RATES,
+ .formats = AT32_SSC_FORMATS,
+ },
+ .ops = {
+ .startup = at32_ssc_startup,
+ .shutdown = at32_ssc_shutdown,
+ .prepare = at32_ssc_prepare,
+ .hw_params = at32_ssc_hw_params,
+ },
+ .dai_ops = {
+ .set_sysclk = at32_ssc_set_dai_sysclk,
+ .set_fmt = at32_ssc_set_dai_fmt,
+ .set_clkdiv = at32_ssc_set_dai_clkdiv,
+ },
+ .private_data = &ssc_info[1],
+ },
+ {
+ .name = "at32-ssc2",
+ .id = 2,
+ .type = SND_SOC_DAI_PCM,
+ .suspend = at32_ssc_suspend,
+ .resume = at32_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT32_SSC_RATES,
+ .formats = AT32_SSC_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT32_SSC_RATES,
+ .formats = AT32_SSC_FORMATS,
+ },
+ .ops = {
+ .startup = at32_ssc_startup,
+ .shutdown = at32_ssc_shutdown,
+ .prepare = at32_ssc_prepare,
+ .hw_params = at32_ssc_hw_params,
+ },
+ .dai_ops = {
+ .set_sysclk = at32_ssc_set_dai_sysclk,
+ .set_fmt = at32_ssc_set_dai_fmt,
+ .set_clkdiv = at32_ssc_set_dai_clkdiv,
+ },
+ .private_data = &ssc_info[2],
+ },
+};
+EXPORT_SYMBOL_GPL(at32_ssc_dai);
+
+
+MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
+MODULE_DESCRIPTION("AT32 SSC ASoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h
new file mode 100644
index 0000000..3c052db
--- /dev/null
+++ b/sound/soc/at32/at32-ssc.h
@@ -0,0 +1,59 @@
+/* sound/soc/at32/at32-ssc.h
+ * ASoC SSC interface for Atmel AT32 SoC
+ *
+ * Copyright (C) 2008 Long Range Systems
+ * Geoffrey Wossum <gwossum@acm.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __SOUND_SOC_AT32_AT32_SSC_H
+#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__
+
+#include <linux/types.h>
+#include <linux/atmel-ssc.h>
+
+#include "at32-pcm.h"
+
+
+
+struct at32_ssc_state {
+ u32 ssc_cmr;
+ u32 ssc_rcmr;
+ u32 ssc_rfmr;
+ u32 ssc_tcmr;
+ u32 ssc_tfmr;
+ u32 ssc_sr;
+ u32 ssc_imr;
+};
+
+
+
+struct at32_ssc_info {
+ char *name;
+ struct ssc_device *ssc;
+ spinlock_t lock; /* lock for dir_mask */
+ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
+ unsigned short initialized; /* true if SSC has been initialized */
+ unsigned short daifmt;
+ unsigned short cmr_div;
+ unsigned short tcmr_period;
+ unsigned short rcmr_period;
+ struct at32_pcm_dma_params *dma_params[2];
+ struct at32_ssc_state ssc_state;
+};
+
+
+/* SSC divider ids */
+#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */
+#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
+#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
+
+
+extern struct snd_soc_dai at32_ssc_dai[];
+
+
+
+#endif /* __SOUND_SOC_AT32_AT32_SSC_H */
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
new file mode 100644
index 0000000..b1966e4
--- /dev/null
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -0,0 +1,513 @@
+/* sound/soc/at32/playpaq_wm8510.c
+ * ASoC machine driver for PlayPaq using WM8510 codec
+ *
+ * Copyright (C) 2008 Long Range Systems
+ * Geoffrey Wossum <gwossum@acm.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c
+ *
+ * NOTE: If you don't have the AT32 enhanced portmux configured (which
+ * isn't currently in the mainline or Atmel patched kernel), you will
+ * need to set the MCLK pin (PA30) to peripheral A in your board initialization
+ * code. Something like:
+ * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0);
+ *
+ */
+
+/* #define DEBUG */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/errno.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/at32ap700x.h>
+#include <mach/portmux.h>
+
+#include "../codecs/wm8510.h"
+#include "at32-pcm.h"
+#include "at32-ssc.h"
+
+
+/*-------------------------------------------------------------------------*\
+ * constants
+\*-------------------------------------------------------------------------*/
+#define MCLK_PIN GPIO_PIN_PA(30)
+#define MCLK_PERIPH GPIO_PERIPH_A
+
+
+/*-------------------------------------------------------------------------*\
+ * data types
+\*-------------------------------------------------------------------------*/
+/* SSC clocking data */
+struct ssc_clock_data {
+ /* CMR div */
+ unsigned int cmr_div;
+
+ /* Frame period (as needed by xCMR.PERIOD) */
+ unsigned int period;
+
+ /* The SSC clock rate these settings where calculated for */
+ unsigned long ssc_rate;
+};
+
+
+/*-------------------------------------------------------------------------*\
+ * module data
+\*-------------------------------------------------------------------------*/
+static struct clk *_gclk0;
+static struct clk *_pll0;
+
+#define CODEC_CLK (_gclk0)
+
+
+/*-------------------------------------------------------------------------*\
+ * Sound SOC operations
+\*-------------------------------------------------------------------------*/
+#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct at32_ssc_info *ssc_p = cpu_dai->private_data;
+ struct ssc_device *ssc = ssc_p->ssc;
+ struct ssc_clock_data cd;
+ unsigned int rate, width_bits, channels;
+ unsigned int bitrate, ssc_div;
+ unsigned actual_rate;
+
+
+ /*
+ * Figure out required bitrate
+ */
+ rate = params_rate(params);
+ channels = params_channels(params);
+ width_bits = snd_pcm_format_physical_width(params_format(params));
+ bitrate = rate * width_bits * channels;
+
+
+ /*
+ * Figure out required SSC divider and period for required bitrate
+ */
+ cd.ssc_rate = clk_get_rate(ssc->clk);
+ ssc_div = cd.ssc_rate / bitrate;
+ cd.cmr_div = ssc_div / 2;
+ if (ssc_div & 1) {
+ /* round cmr_div up */
+ cd.cmr_div++;
+ }
+ cd.period = width_bits - 1;
+
+
+ /*
+ * Find actual rate, compare to requested rate
+ */
+ actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
+ pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
+ rate, actual_rate);
+
+
+ return cd;
+}
+#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
+
+
+
+static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct at32_ssc_info *ssc_p = cpu_dai->private_data;
+ struct ssc_device *ssc = ssc_p->ssc;
+ unsigned int pll_out = 0, bclk = 0, mclk_div = 0;
+ int ret;
+
+
+ /* Due to difficulties with getting the correct clocks from the AT32's
+ * PLL0, we're going to let the CODEC be in charge of all the clocks
+ */
+#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+ const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+#else
+ struct ssc_clock_data cd;
+ const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS);
+#endif
+
+ if (ssc == NULL) {
+ pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n");
+ return -EINVAL;
+ }
+
+
+ /*
+ * Figure out PLL and BCLK dividers for WM8510
+ */
+ switch (params_rate(params)) {
+ case 48000:
+ pll_out = 12288000;
+ mclk_div = WM8510_MCLKDIV_1;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
+ case 44100:
+ pll_out = 11289600;
+ mclk_div = WM8510_MCLKDIV_1;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
+ case 22050:
+ pll_out = 11289600;
+ mclk_div = WM8510_MCLKDIV_2;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
+ case 16000:
+ pll_out = 12288000;
+ mclk_div = WM8510_MCLKDIV_3;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
+ case 11025:
+ pll_out = 11289600;
+ mclk_div = WM8510_MCLKDIV_4;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
+ case 8000:
+ pll_out = 12288000;
+ mclk_div = WM8510_MCLKDIV_6;
+ bclk = WM8510_BCLKDIV_8;
+ break;
+
+ default:
+ pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+
+
+ /*
+ * set CPU and CODEC DAI configuration
+ */
+ ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+ if (ret < 0) {
+ pr_warning("playpaq_wm8510: "
+ "Failed to set CODEC DAI format (%d)\n",
+ ret);
+ return ret;
+ }
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+ if (ret < 0) {
+ pr_warning("playpaq_wm8510: "
+ "Failed to set CPU DAI format (%d)\n",
+ ret);
+ return ret;
+ }
+
+
+ /*
+ * Set CPU clock configuration
+ */
+#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+ cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai);
+ pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n",
+ cd.cmr_div, cd.period);
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div);
+ if (ret < 0) {
+ pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n",
+ ret);
+ return ret;
+ }
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD,
+ cd.period);
+ if (ret < 0) {
+ pr_warning("playpaq_wm8510: "
+ "Failed to set CPU transmit period (%d)\n",
+ ret);
+ return ret;
+ }
+#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
+
+
+ /*
+ * Set CODEC clock configuration
+ */
+ pr_debug("playpaq_wm8510: "
+ "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n",
+ clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div);
+
+
+#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk);
+ if (ret < 0) {
+ pr_warning
+ ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n",
+ ret);
+ return ret;
+ }
+#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
+
+
+ ret = snd_soc_dai_set_pll(codec_dai, 0,
+ clk_get_rate(CODEC_CLK), pll_out);
+ if (ret < 0) {
+ pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
+ ret);
+ return ret;
+ }
+
+
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div);
+ if (ret < 0) {
+ pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n",
+ ret);
+ return ret;
+ }
+
+
+ return 0;
+}
+
+
+
+static struct snd_soc_ops playpaq_wm8510_ops = {
+ .hw_params = playpaq_wm8510_hw_params,
+};
+
+
+
+static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* speaker connected to SPKOUT */
+ {"Ext Spk", NULL, "SPKOUTP"},
+ {"Ext Spk", NULL, "SPKOUTN"},
+
+ {"Mic Bias", NULL, "Int Mic"},
+ {"MICN", NULL, "Mic Bias"},
+ {"MICP", NULL, "Mic Bias"},
+};
+
+
+
+static int playpaq_wm8510_init(struct snd_soc_codec *codec)
+{
+ int i;
+
+ /*
+ * Add DAPM widgets
+ */
+ for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]);
+
+
+
+ /*
+ * Setup audio path interconnects
+ */
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+
+
+ /* always connected pins */
+ snd_soc_dapm_enable_pin(codec, "Int Mic");
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ snd_soc_dapm_sync(codec);
+
+
+
+ /* Make CSB show PLL rate */
+ snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV,
+ WM8510_OPCLKDIV_1 | 4);
+
+ return 0;
+}
+
+
+
+static struct snd_soc_dai_link playpaq_wm8510_dai = {
+ .name = "WM8510",
+ .stream_name = "WM8510 PCM",
+ .cpu_dai = &at32_ssc_dai[0],
+ .codec_dai = &wm8510_dai,
+ .init = playpaq_wm8510_init,
+ .ops = &playpaq_wm8510_ops,
+};
+
+
+
+static struct snd_soc_machine snd_soc_machine_playpaq = {
+ .name = "LRS_PlayPaq_WM8510",
+ .dai_link = &playpaq_wm8510_dai,
+ .num_links = 1,
+};
+
+
+
+static struct wm8510_setup_data playpaq_wm8510_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1a,
+};
+
+
+
+static struct snd_soc_device playpaq_wm8510_snd_devdata = {
+ .machine = &snd_soc_machine_playpaq,
+ .platform = &at32_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8510,
+ .codec_data = &playpaq_wm8510_setup,
+};
+
+static struct platform_device *playpaq_snd_device;
+
+
+static int __init playpaq_asoc_init(void)
+{
+ int ret = 0;
+ struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data;
+ struct ssc_device *ssc = NULL;
+
+
+ /*
+ * Request SSC device
+ */
+ ssc = ssc_request(0);
+ if (IS_ERR(ssc)) {
+ ret = PTR_ERR(ssc);
+ goto err_ssc;
+ }
+ ssc_p->ssc = ssc;
+
+
+ /*
+ * Configure MCLK for WM8510
+ */
+ _gclk0 = clk_get(NULL, "gclk0");
+ if (IS_ERR(_gclk0)) {
+ _gclk0 = NULL;
+ goto err_gclk0;
+ }
+ _pll0 = clk_get(NULL, "pll0");
+ if (IS_ERR(_pll0)) {
+ _pll0 = NULL;
+ goto err_pll0;
+ }
+ if (clk_set_parent(_gclk0, _pll0)) {
+ pr_warning("snd-soc-playpaq: "
+ "Failed to set PLL0 as parent for DAC clock\n");
+ goto err_set_clk;
+ }
+ clk_set_rate(CODEC_CLK, 12000000);
+ clk_enable(CODEC_CLK);
+
+#if defined CONFIG_AT32_ENHANCED_PORTMUX
+ at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0);
+#endif
+
+
+ /*
+ * Create and register platform device
+ */
+ playpaq_snd_device = platform_device_alloc("soc-audio", 0);
+ if (playpaq_snd_device == NULL) {
+ ret = -ENOMEM;
+ goto err_device_alloc;
+ }
+
+ platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata);
+ playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev;
+
+ ret = platform_device_add(playpaq_snd_device);
+ if (ret) {
+ pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n",
+ ret);
+ goto err_device_add;
+ }
+
+ return 0;
+
+
+err_device_add:
+ if (playpaq_snd_device != NULL) {
+ platform_device_put(playpaq_snd_device);
+ playpaq_snd_device = NULL;
+ }
+err_device_alloc:
+err_set_clk:
+ if (_pll0 != NULL) {
+ clk_put(_pll0);
+ _pll0 = NULL;
+ }
+err_pll0:
+ if (_gclk0 != NULL) {
+ clk_put(_gclk0);
+ _gclk0 = NULL;
+ }
+err_gclk0:
+ ssc_free(ssc);
+err_ssc:
+ return ret;
+}
+
+
+static void __exit playpaq_asoc_exit(void)
+{
+ struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data;
+ struct ssc_device *ssc;
+
+ if (ssc_p != NULL) {
+ ssc = ssc_p->ssc;
+ if (ssc != NULL)
+ ssc_free(ssc);
+ ssc_p->ssc = NULL;
+ }
+
+ if (_gclk0 != NULL) {
+ clk_put(_gclk0);
+ _gclk0 = NULL;
+ }
+ if (_pll0 != NULL) {
+ clk_put(_pll0);
+ _pll0 = NULL;
+ }
+
+#if defined CONFIG_AT32_ENHANCED_PORTMUX
+ at32_free_pin(MCLK_PIN);
+#endif
+
+ platform_device_unregister(playpaq_snd_device);
+ playpaq_snd_device = NULL;
+}
+
+module_init(playpaq_asoc_init);
+module_exit(playpaq_asoc_exit);
+
+MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
+MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
new file mode 100644
index 0000000..85a8832
--- /dev/null
+++ b/sound/soc/at91/Kconfig
@@ -0,0 +1,10 @@
+config SND_AT91_SOC
+ tristate "SoC Audio for the Atmel AT91 System-on-Chip"
+ depends on ARCH_AT91
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the AT91 SSC interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_AT91_SOC_SSC
+ tristate
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
new file mode 100644
index 0000000..b817f11
--- /dev/null
+++ b/sound/soc/at91/Makefile
@@ -0,0 +1,6 @@
+# AT91 Platform Support
+snd-soc-at91-objs := at91-pcm.o
+snd-soc-at91-ssc-objs := at91-ssc.o
+
+obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
+obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
new file mode 100644
index 0000000..7ab48bd
--- /dev/null
+++ b/sound/soc/at91/at91-pcm.c
@@ -0,0 +1,434 @@
+/*
+ * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ * Created: Mar 3, 2006
+ *
+ * Based on pxa2xx-pcm.c by:
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/atmel_pdc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+#include <mach/at91_ssc.h>
+
+#include "at91-pcm.h"
+
+#if 0
+#define DBG(x...) printk(KERN_INFO "at91-pcm: " x)
+#else
+#define DBG(x...)
+#endif
+
+static const struct snd_pcm_hardware at91_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8192,
+ .periods_min = 2,
+ .periods_max = 1024,
+ .buffer_bytes_max = 32 * 1024,
+};
+
+struct at91_runtime_data {
+ struct at91_pcm_dma_params *params;
+ dma_addr_t dma_buffer; /* physical address of dma buffer */
+ dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */
+ size_t period_size;
+ dma_addr_t period_ptr; /* physical address of next period */
+ u32 pdc_xpr_save; /* PDC register save */
+ u32 pdc_xcr_save;
+ u32 pdc_xnpr_save;
+ u32 pdc_xncr_save;
+};
+
+static void at91_pcm_dma_irq(u32 ssc_sr,
+ struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+ static int count = 0;
+
+ count++;
+
+ if (ssc_sr & params->mask->ssc_endbuf) {
+
+ printk(KERN_WARNING
+ "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+ ? "underrun" : "overrun",
+ params->name, ssc_sr, count);
+
+ /* re-start the PDC */
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
+
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end) {
+ prtd->period_ptr = prtd->dma_buffer;
+ }
+
+ at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
+ }
+
+ if (ssc_sr & params->mask->ssc_endx) {
+
+ /* Load the PDC next pointer and counter registers */
+ prtd->period_ptr += prtd->period_size;
+ if (prtd->period_ptr >= prtd->dma_buffer_end) {
+ prtd->period_ptr = prtd->dma_buffer;
+ }
+ at91_ssc_write(params->ssc_base + params->pdc->xnpr,
+ prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+ }
+
+ snd_pcm_period_elapsed(substream);
+}
+
+static int at91_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at91_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* this may get called several times by oss emulation
+ * with different params */
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ prtd->params = rtd->dai->cpu_dai->dma_data;
+ prtd->params->dma_intr_handler = at91_pcm_dma_irq;
+
+ prtd->dma_buffer = runtime->dma_addr;
+ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
+ prtd->period_size = params_period_bytes(params);
+
+ DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n",
+ prtd->params->name, runtime->dma_bytes, prtd->period_size);
+ return 0;
+}
+
+static int at91_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+
+ if (params != NULL) {
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
+ prtd->params->dma_intr_handler = NULL;
+ }
+
+ return 0;
+}
+
+static int at91_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+
+ at91_ssc_write(params->ssc_base + AT91_SSC_IDR,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
+ return 0;
+}
+
+static int at91_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->period_ptr = prtd->dma_buffer;
+
+ at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xcr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ prtd->period_ptr += prtd->period_size;
+ at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
+ at91_ssc_write(params->ssc_base + params->pdc->xncr,
+ prtd->period_size / params->pdc_xfer_size);
+
+ DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n",
+ (unsigned long) prtd->period_ptr,
+ at91_ssc_read(params->ssc_base + params->pdc->xpr),
+ at91_ssc_read(params->ssc_base + params->pdc->xcr),
+ at91_ssc_read(params->ssc_base + params->pdc->xnpr),
+ at91_ssc_read(params->ssc_base + params->pdc->xncr));
+
+ at91_ssc_write(params->ssc_base + AT91_SSC_IER,
+ params->mask->ssc_endx | params->mask->ssc_endbuf);
+
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR,
+ params->mask->pdc_enable);
+
+ DBG("sr=%lx imr=%lx\n",
+ at91_ssc_read(params->ssc_base + AT91_SSC_SR),
+ at91_ssc_read(params->ssc_base + AT91_SSC_IMR));
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t at91_pcm_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at91_runtime_data *prtd = runtime->private_data;
+ struct at91_pcm_dma_params *params = prtd->params;
+ dma_addr_t ptr;
+ snd_pcm_uframes_t x;
+
+ ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr);
+ x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
+
+ if (x == runtime->buffer_size)
+ x = 0;
+ return x;
+}
+
+static int at91_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct at91_runtime_data *prtd;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware);
+
+ /* ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ runtime->private_data = prtd;
+
+ out:
+ return ret;
+}
+
+static int at91_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct at91_runtime_data *prtd = substream->runtime->private_data;
+
+ kfree(prtd);
+ return 0;
+}
+
+static int at91_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+struct snd_pcm_ops at91_pcm_ops = {
+ .open = at91_pcm_open,
+ .close = at91_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = at91_pcm_hw_params,
+ .hw_free = at91_pcm_hw_free,
+ .prepare = at91_pcm_prepare,
+ .trigger = at91_pcm_trigger,
+ .pointer = at91_pcm_pointer,
+ .mmap = at91_pcm_mmap,
+};
+
+static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = at91_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+
+ DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+ (void *) buf->area,
+ (void *) buf->addr,
+ size);
+
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static u64 at91_pcm_dmamask = 0xffffffff;
+
+static int at91_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &at91_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = at91_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = at91_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+#ifdef CONFIG_PM
+static int at91_pcm_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct at91_runtime_data *prtd;
+ struct at91_pcm_dma_params *params;
+
+ if (!runtime)
+ return 0;
+
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* disable the PDC and save the PDC registers */
+
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
+
+ prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr);
+ prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr);
+ prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr);
+ prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr);
+
+ return 0;
+}
+
+static int at91_pcm_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = dai->runtime;
+ struct at91_runtime_data *prtd;
+ struct at91_pcm_dma_params *params;
+
+ if (!runtime)
+ return 0;
+
+ prtd = runtime->private_data;
+ params = prtd->params;
+
+ /* restore the PDC registers and enable the PDC */
+ at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save);
+ at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save);
+ at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save);
+ at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save);
+
+ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable);
+ return 0;
+}
+#else
+#define at91_pcm_suspend NULL
+#define at91_pcm_resume NULL
+#endif
+
+struct snd_soc_platform at91_soc_platform = {
+ .name = "at91-audio",
+ .pcm_ops = &at91_pcm_ops,
+ .pcm_new = at91_pcm_new,
+ .pcm_free = at91_pcm_free_dma_buffers,
+ .suspend = at91_pcm_suspend,
+ .resume = at91_pcm_resume,
+};
+
+EXPORT_SYMBOL_GPL(at91_soc_platform);
+
+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
+MODULE_DESCRIPTION("Atmel AT91 PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
new file mode 100644
index 0000000..e5aada2
--- /dev/null
+++ b/sound/soc/at91/at91-pcm.h
@@ -0,0 +1,72 @@
+/*
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ * Created: Mar 3, 2006
+ *
+ * Based on pxa2xx-pcm.h by:
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AT91_PCM_H
+#define _AT91_PCM_H
+
+#include <mach/hardware.h>
+
+struct at91_ssc_periph {
+ void __iomem *base;
+ u32 pid;
+};
+
+/*
+ * Registers and status bits that are required by the PCM driver.
+ */
+struct at91_pdc_regs {
+ unsigned int xpr; /* PDC recv/trans pointer */
+ unsigned int xcr; /* PDC recv/trans counter */
+ unsigned int xnpr; /* PDC next recv/trans pointer */
+ unsigned int xncr; /* PDC next recv/trans counter */
+ unsigned int ptcr; /* PDC transfer control */
+};
+
+struct at91_ssc_mask {
+ u32 ssc_enable; /* SSC recv/trans enable */
+ u32 ssc_disable; /* SSC recv/trans disable */
+ u32 ssc_endx; /* SSC ENDTX or ENDRX */
+ u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */
+ u32 pdc_enable; /* PDC recv/trans enable */
+ u32 pdc_disable; /* PDC recv/trans disable */
+};
+
+/*
+ * This structure, shared between the PCM driver and the interface,
+ * contains all information required by the PCM driver to perform the
+ * PDC DMA operation. All fields except dma_intr_handler() are initialized
+ * by the interface. The dms_intr_handler() pointer is set by the PCM
+ * driver and called by the interface SSC interrupt handler if it is
+ * non-NULL.
+ */
+struct at91_pcm_dma_params {
+ char *name; /* stream identifier */
+ int pdc_xfer_size; /* PDC counter increment in bytes */
+ void __iomem *ssc_base; /* SSC base address */
+ struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */
+ struct at91_ssc_mask *mask;/* SSC & PDC status bits */
+ struct snd_pcm_substream *substream;
+ void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
+};
+
+extern struct snd_soc_platform at91_soc_platform;
+
+#define at91_ssc_read(a) ((unsigned long) __raw_readl(a))
+#define at91_ssc_write(a,v) __raw_writel((v),(a))
+
+#endif /* _AT91_PCM_H */
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
new file mode 100644
index 0000000..1b61cc4
--- /dev/null
+++ b/sound/soc/at91/at91-ssc.c
@@ -0,0 +1,791 @@
+/*
+ * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/atmel_pdc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+#include <mach/at91_pmc.h>
+#include <mach/at91_ssc.h>
+
+#include "at91-pcm.h"
+#include "at91-ssc.h"
+
+#if 0
+#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x)
+#else
+#define DBG(x...)
+#endif
+
+#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
+#define NUM_SSC_DEVICES 1
+#else
+#define NUM_SSC_DEVICES 3
+#endif
+
+
+/*
+ * SSC PDC registers required by the PCM DMA engine.
+ */
+static struct at91_pdc_regs pdc_tx_reg = {
+ .xpr = ATMEL_PDC_TPR,
+ .xcr = ATMEL_PDC_TCR,
+ .xnpr = ATMEL_PDC_TNPR,
+ .xncr = ATMEL_PDC_TNCR,
+};
+
+static struct at91_pdc_regs pdc_rx_reg = {
+ .xpr = ATMEL_PDC_RPR,
+ .xcr = ATMEL_PDC_RCR,
+ .xnpr = ATMEL_PDC_RNPR,
+ .xncr = ATMEL_PDC_RNCR,
+};
+
+/*
+ * SSC & PDC status bits for transmit and receive.
+ */
+static struct at91_ssc_mask ssc_tx_mask = {
+ .ssc_enable = AT91_SSC_TXEN,
+ .ssc_disable = AT91_SSC_TXDIS,
+ .ssc_endx = AT91_SSC_ENDTX,
+ .ssc_endbuf = AT91_SSC_TXBUFE,
+ .pdc_enable = ATMEL_PDC_TXTEN,
+ .pdc_disable = ATMEL_PDC_TXTDIS,
+};
+
+static struct at91_ssc_mask ssc_rx_mask = {
+ .ssc_enable = AT91_SSC_RXEN,
+ .ssc_disable = AT91_SSC_RXDIS,
+ .ssc_endx = AT91_SSC_ENDRX,
+ .ssc_endbuf = AT91_SSC_RXBUFF,
+ .pdc_enable = ATMEL_PDC_RXTEN,
+ .pdc_disable = ATMEL_PDC_RXTDIS,
+};
+
+
+/*
+ * DMA parameters.
+ */
+static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
+ {{
+ .name = "SSC0 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC0 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ }},
+#if NUM_SSC_DEVICES == 3
+ {{
+ .name = "SSC1 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC1 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ }},
+ {{
+ .name = "SSC2 PCM out",
+ .pdc = &pdc_tx_reg,
+ .mask = &ssc_tx_mask,
+ },
+ {
+ .name = "SSC2 PCM in",
+ .pdc = &pdc_rx_reg,
+ .mask = &ssc_rx_mask,
+ }},
+#endif
+};
+
+struct at91_ssc_state {
+ u32 ssc_cmr;
+ u32 ssc_rcmr;
+ u32 ssc_rfmr;
+ u32 ssc_tcmr;
+ u32 ssc_tfmr;
+ u32 ssc_sr;
+ u32 ssc_imr;
+};
+
+static struct at91_ssc_info {
+ char *name;
+ struct at91_ssc_periph ssc;
+ spinlock_t lock; /* lock for dir_mask */
+ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */
+ unsigned short initialized; /* 1=SSC has been initialized */
+ unsigned short daifmt;
+ unsigned short cmr_div;
+ unsigned short tcmr_period;
+ unsigned short rcmr_period;
+ struct at91_pcm_dma_params *dma_params[2];
+ struct at91_ssc_state ssc_state;
+
+} ssc_info[NUM_SSC_DEVICES] = {
+ {
+ .name = "ssc0",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
+ .dir_mask = 0,
+ .initialized = 0,
+ },
+#if NUM_SSC_DEVICES == 3
+ {
+ .name = "ssc1",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
+ .dir_mask = 0,
+ .initialized = 0,
+ },
+ {
+ .name = "ssc2",
+ .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
+ .dir_mask = 0,
+ .initialized = 0,
+ },
+#endif
+};
+
+static unsigned int at91_ssc_sysclk;
+
+/*
+ * SSC interrupt handler. Passes PDC interrupts to the DMA
+ * interrupt handler in the PCM driver.
+ */
+static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id)
+{
+ struct at91_ssc_info *ssc_p = dev_id;
+ struct at91_pcm_dma_params *dma_params;
+ u32 ssc_sr;
+ int i;
+
+ ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)
+ & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
+
+ /*
+ * Loop through the substreams attached to this SSC. If
+ * a DMA-related interrupt occurred on that substream, call
+ * the DMA interrupt handler function, if one has been
+ * registered in the dma_params structure by the PCM driver.
+ */
+ for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
+ dma_params = ssc_p->dma_params[i];
+
+ if (dma_params != NULL && dma_params->dma_intr_handler != NULL &&
+ (ssc_sr &
+ (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf)))
+
+ dma_params->dma_intr_handler(ssc_sr, dma_params->substream);
+ }
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * Startup. Only that one substream allowed in each direction.
+ */
+static int at91_ssc_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ int dir_mask;
+
+ DBG("ssc_startup: SSC_SR=0x%08lx\n",
+ at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
+ dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2;
+
+ spin_lock_irq(&ssc_p->lock);
+ if (ssc_p->dir_mask & dir_mask) {
+ spin_unlock_irq(&ssc_p->lock);
+ return -EBUSY;
+ }
+ ssc_p->dir_mask |= dir_mask;
+ spin_unlock_irq(&ssc_p->lock);
+
+ return 0;
+}
+
+/*
+ * Shutdown. Clear DMA parameters and shutdown the SSC if there
+ * are no other substreams open.
+ */
+static void at91_ssc_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct at91_pcm_dma_params *dma_params;
+ int dir, dir_mask;
+
+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ dma_params = ssc_p->dma_params[dir];
+
+ if (dma_params != NULL) {
+ at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
+ dma_params->mask->ssc_disable);
+ DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"),
+ at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
+
+ dma_params->ssc_base = NULL;
+ dma_params->substream = NULL;
+ ssc_p->dma_params[dir] = NULL;
+ }
+
+ dir_mask = 1 << dir;
+
+ spin_lock_irq(&ssc_p->lock);
+ ssc_p->dir_mask &= ~dir_mask;
+ if (!ssc_p->dir_mask) {
+ /* Shutdown the SSC clock. */
+ DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
+ at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
+
+ if (ssc_p->initialized) {
+ free_irq(ssc_p->ssc.pid, ssc_p);
+ ssc_p->initialized = 0;
+ }
+
+ /* Reset the SSC */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
+
+ /* Clear the SSC dividers */
+ ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
+ }
+ spin_unlock_irq(&ssc_p->lock);
+}
+
+/*
+ * Record the SSC system clock rate.
+ */
+static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ /*
+ * The only clock supplied to the SSC is the AT91 master clock,
+ * which is only used if the SSC is generating BCLK and/or
+ * LRC clocks.
+ */
+ switch (clk_id) {
+ case AT91_SYSCLK_MCK:
+ at91_ssc_sysclk = freq;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Record the DAI format for use in hw_params().
+ */
+static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ ssc_p->daifmt = fmt;
+ return 0;
+}
+
+/*
+ * Record SSC clock dividers for use in hw_params().
+ */
+static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+ switch (div_id) {
+ case AT91SSC_CMR_DIV:
+ /*
+ * The same master clock divider is used for both
+ * transmit and receive, so if a value has already
+ * been set, it must match this value.
+ */
+ if (ssc_p->cmr_div == 0)
+ ssc_p->cmr_div = div;
+ else
+ if (div != ssc_p->cmr_div)
+ return -EBUSY;
+ break;
+
+ case AT91SSC_TCMR_PERIOD:
+ ssc_p->tcmr_period = div;
+ break;
+
+ case AT91SSC_RCMR_PERIOD:
+ ssc_p->rcmr_period = div;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Configure the SSC.
+ */
+static int at91_ssc_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int id = rtd->dai->cpu_dai->id;
+ struct at91_ssc_info *ssc_p = &ssc_info[id];
+ struct at91_pcm_dma_params *dma_params;
+ int dir, channels, bits;
+ u32 tfmr, rfmr, tcmr, rcmr;
+ int start_event;
+ int ret;
+
+ /*
+ * Currently, there is only one set of dma params for
+ * each direction. If more are added, this code will
+ * have to be changed to select the proper set.
+ */
+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ dma_params = &ssc_dma_params[id][dir];
+ dma_params->ssc_base = ssc_p->ssc.base;
+ dma_params->substream = substream;
+
+ ssc_p->dma_params[dir] = dma_params;
+
+ /*
+ * The cpu_dai->dma_data field is only used to communicate the
+ * appropriate DMA parameters to the pcm driver hw_params()
+ * function. It should not be used for other purposes
+ * as it is common to all substreams.
+ */
+ rtd->dai->cpu_dai->dma_data = dma_params;
+
+ channels = params_channels(params);
+
+ /*
+ * Determine sample size in bits and the PDC increment.
+ */
+ switch(params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ bits = 8;
+ dma_params->pdc_xfer_size = 1;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bits = 16;
+ dma_params->pdc_xfer_size = 2;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bits = 24;
+ dma_params->pdc_xfer_size = 4;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bits = 32;
+ dma_params->pdc_xfer_size = 4;
+ break;
+ default:
+ printk(KERN_WARNING "at91-ssc: unsupported PCM format\n");
+ return -EINVAL;
+ }
+
+ /*
+ * The SSC only supports up to 16-bit samples in I2S format, due
+ * to the size of the Frame Mode Register FSLEN field.
+ */
+ if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
+ && bits > 16) {
+ printk(KERN_WARNING
+ "at91-ssc: sample size %d is too large for I2S\n", bits);
+ return -EINVAL;
+ }
+
+ /*
+ * Compute SSC register settings.
+ */
+ switch (ssc_p->daifmt
+ & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
+
+ case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+ /*
+ * I2S format, SSC provides BCLK and LRC clocks.
+ *
+ * The SSC transmit and receive clocks are generated from the
+ * MCK divider, and the BCLK signal is output on the SSC TK line.
+ */
+ rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START)
+ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
+
+ rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS)
+ | (((bits - 1) << 16) & AT91_SSC_FSLEN)
+ | (((channels - 1) << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_LOOP)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+
+ tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START)
+ | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
+
+ tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( 0 << 23) & AT91_SSC_FSDEN)
+ | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS)
+ | (((bits - 1) << 16) & AT91_SSC_FSLEN)
+ | (((channels - 1) << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_DATDEF)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+ break;
+
+ case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
+ /*
+ * I2S format, CODEC supplies BCLK and LRC clocks.
+ *
+ * The SSC transmit clock is obtained from the BCLK signal on
+ * on the TK line, and the SSC receive clock is generated from the
+ * transmit clock.
+ *
+ * For single channel data, one sample is transferred on the falling
+ * edge of the LRC clock. For two channel data, one sample is
+ * transferred on both edges of the LRC clock.
+ */
+ start_event = channels == 1
+ ? AT91_SSC_START_FALLING_RF
+ : AT91_SSC_START_EDGE_RF;
+
+ rcmr = (( 0 << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( start_event ) & AT91_SSC_START)
+ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS);
+
+ rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS)
+ | (( 0 << 16) & AT91_SSC_FSLEN)
+ | (( 0 << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_LOOP)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+
+ tcmr = (( 0 << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( start_event ) & AT91_SSC_START)
+ | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS);
+
+ tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( 0 << 23) & AT91_SSC_FSDEN)
+ | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS)
+ | (( 0 << 16) & AT91_SSC_FSLEN)
+ | (( 0 << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_DATDEF)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+ /*
+ * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks.
+ *
+ * The SSC transmit and receive clocks are generated from the
+ * MCK divider, and the BCLK signal is output on the SSC TK line.
+ */
+ rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START)
+ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
+
+ rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS)
+ | (( 0 << 16) & AT91_SSC_FSLEN)
+ | (((channels - 1) << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_LOOP)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+
+ tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD)
+ | (( 1 << 16) & AT91_SSC_STTDLY)
+ | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START)
+ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI)
+ | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO)
+ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS);
+
+ tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE)
+ | (( 0 << 23) & AT91_SSC_FSDEN)
+ | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS)
+ | (( 0 << 16) & AT91_SSC_FSLEN)
+ | (((channels - 1) << 8) & AT91_SSC_DATNB)
+ | (( 1 << 7) & AT91_SSC_MSBF)
+ | (( 0 << 5) & AT91_SSC_DATDEF)
+ | (((bits - 1) << 0) & AT91_SSC_DATALEN);
+
+
+
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+ default:
+ printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n",
+ ssc_p->daifmt);
+ return -EINVAL;
+ break;
+ }
+ DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr);
+
+ if (!ssc_p->initialized) {
+
+ /* Enable PMC peripheral clock for this SSC */
+ DBG("Starting pid %d clock\n", ssc_p->ssc.pid);
+ at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
+
+ /* Reset the SSC and its PDC registers */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
+
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0);
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0);
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0);
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0);
+ at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0);
+
+ if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt,
+ 0, ssc_p->name, ssc_p)) < 0) {
+ printk(KERN_WARNING "at91-ssc: request_irq failure\n");
+
+ DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
+ at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
+ return ret;
+ }
+
+ ssc_p->initialized = 1;
+ }
+
+ /* set SSC clock mode register */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div);
+
+ /* set receive clock mode and format */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr);
+
+ /* set transmit clock mode and format */
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr);
+
+ DBG("hw_params: SSC initialized\n");
+ return 0;
+}
+
+
+static int at91_ssc_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+ struct at91_pcm_dma_params *dma_params;
+ int dir;
+
+ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+ dma_params = ssc_p->dma_params[dir];
+
+ at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
+ dma_params->mask->ssc_enable);
+
+ DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit",
+ at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR));
+ return 0;
+}
+
+
+#ifdef CONFIG_PM
+static int at91_ssc_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct at91_ssc_info *ssc_p;
+
+ if(!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ /* Save the status register before disabling transmit and receive. */
+ ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
+ AT91_SSC_TXDIS | AT91_SSC_RXDIS);
+
+ /* Save the current interrupt mask, then disable unmasked interrupts. */
+ ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr);
+
+ ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR);
+ ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR);
+ ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR);
+ ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR);
+ ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR);
+
+ return 0;
+}
+
+static int at91_ssc_resume(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct at91_ssc_info *ssc_p;
+
+ if(!cpu_dai->active)
+ return 0;
+
+ ssc_p = &ssc_info[cpu_dai->id];
+
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr);
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr);
+
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr);
+
+ at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
+ ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) |
+ ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0));
+
+ return 0;
+}
+
+#else
+#define at91_ssc_suspend NULL
+#define at91_ssc_resume NULL
+#endif
+
+#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = {
+ { .name = "at91-ssc0",
+ .id = 0,
+ .type = SND_SOC_DAI_PCM,
+ .suspend = at91_ssc_suspend,
+ .resume = at91_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_SSC_RATES,
+ .formats = AT91_SSC_FORMATS,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_SSC_RATES,
+ .formats = AT91_SSC_FORMATS,},
+ .ops = {
+ .startup = at91_ssc_startup,
+ .shutdown = at91_ssc_shutdown,
+ .prepare = at91_ssc_prepare,
+ .hw_params = at91_ssc_hw_params,},
+ .dai_ops = {
+ .set_sysclk = at91_ssc_set_dai_sysclk,
+ .set_fmt = at91_ssc_set_dai_fmt,
+ .set_clkdiv = at91_ssc_set_dai_clkdiv,},
+ .private_data = &ssc_info[0].ssc,
+ },
+#if NUM_SSC_DEVICES == 3
+ { .name = "at91-ssc1",
+ .id = 1,
+ .type = SND_SOC_DAI_PCM,
+ .suspend = at91_ssc_suspend,
+ .resume = at91_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_SSC_RATES,
+ .formats = AT91_SSC_FORMATS,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_SSC_RATES,
+ .formats = AT91_SSC_FORMATS,},
+ .ops = {
+ .startup = at91_ssc_startup,
+ .shutdown = at91_ssc_shutdown,
+ .prepare = at91_ssc_prepare,
+ .hw_params = at91_ssc_hw_params,},
+ .dai_ops = {
+ .set_sysclk = at91_ssc_set_dai_sysclk,
+ .set_fmt = at91_ssc_set_dai_fmt,
+ .set_clkdiv = at91_ssc_set_dai_clkdiv,},
+ .private_data = &ssc_info[1].ssc,
+ },
+ { .name = "at91-ssc2",
+ .id = 2,
+ .type = SND_SOC_DAI_PCM,
+ .suspend = at91_ssc_suspend,
+ .resume = at91_ssc_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_SSC_RATES,
+ .formats = AT91_SSC_FORMATS,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AT91_SSC_RATES,
+ .formats = AT91_SSC_FORMATS,},
+ .ops = {
+ .startup = at91_ssc_startup,
+ .shutdown = at91_ssc_shutdown,
+ .prepare = at91_ssc_prepare,
+ .hw_params = at91_ssc_hw_params,},
+ .dai_ops = {
+ .set_sysclk = at91_ssc_set_dai_sysclk,
+ .set_fmt = at91_ssc_set_dai_fmt,
+ .set_clkdiv = at91_ssc_set_dai_clkdiv,},
+ .private_data = &ssc_info[2].ssc,
+ },
+#endif
+};
+
+EXPORT_SYMBOL_GPL(at91_ssc_dai);
+
+/* Module information */
+MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com");
+MODULE_DESCRIPTION("AT91 SSC ASoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h
new file mode 100644
index 0000000..6b7bf38
--- /dev/null
+++ b/sound/soc/at91/at91-ssc.h
@@ -0,0 +1,27 @@
+/*
+ * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ * Endrelia Technologies Inc.
+ * Created: Jan 9, 2007
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AT91_SSC_H
+#define _AT91_SSC_H
+
+/* SSC system clock ids */
+#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */
+
+/* SSC divider ids */
+#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */
+#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
+#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
+
+extern struct snd_soc_dai at91_ssc_dai[];
+
+#endif /* _AT91_SSC_H */
+
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 0000000..410a893
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,32 @@
+##
+## Au1200/Au1550 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+ tristate "SoC Audio for Au1200/Au1250/Au1550"
+ depends on SOC_AU1200 || SOC_AU1550
+ help
+ This option enables support for the Programmable Serial
+ Controllers in AC97 and I2S mode, and the Descriptor-Based DMA
+ Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC.
+
+config SND_SOC_AU1XPSC_I2S
+ tristate
+
+config SND_SOC_AU1XPSC_AC97
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_SAMPLE_PSC_AC97
+ tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
+ depends on SND_SOC_AU1XPSC
+ select SND_SOC_AU1XPSC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ This is a sample AC97 sound machine for use in Au12x0/Au1550
+ based systems which have audio on PSC1 (e.g. Db1200 demoboard).
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 0000000..6c6950b
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -0,0 +1,13 @@
+# Au1200/Au1550 PSC audio
+snd-soc-au1xpsc-dbdma-objs := dbdma2.o
+snd-soc-au1xpsc-i2s-objs := psc-i2s.o
+snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+
+obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
+obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
+obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
+
+# Boards
+snd-soc-sample-ac97-objs := sample-ac97.o
+
+obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 0000000..1466d93
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -0,0 +1,421 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * DMA glue for Au1x-PSC audio.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ * of a PSC. Multiple independent audio devices are impossible
+ * with ASoC v1.
+ */
+
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/*#define PCM_DEBUG*/
+
+#define MSG(x...) printk(KERN_INFO "au1xpsc_pcm: " x)
+#ifdef PCM_DEBUG
+#define DBG MSG
+#else
+#define DBG(x...) do {} while (0)
+#endif
+
+struct au1xpsc_audio_dmadata {
+ /* DDMA control data */
+ unsigned int ddma_id; /* DDMA direction ID for this PSC */
+ u32 ddma_chan; /* DDMA context */
+
+ /* PCM context (for irq handlers) */
+ struct snd_pcm_substream *substream;
+ unsigned long curr_period; /* current segment DDMA is working on */
+ unsigned long q_period; /* queue period(s) */
+ unsigned long dma_area; /* address of queued DMA area */
+ unsigned long dma_area_s; /* start address of DMA area */
+ unsigned long pos; /* current byte position being played */
+ unsigned long periods; /* number of SG segments in total */
+ unsigned long period_bytes; /* size in bytes of one SG segment */
+
+ /* runtime data */
+ int msbits;
+};
+
+/* instance data. There can be only one, MacLeod!!!! */
+static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
+
+/*
+ * These settings are somewhat okay, at least on my machine audio plays
+ * almost skip-free. Especially the 64kB buffer seems to help a LOT.
+ */
+#define AU1XPSC_PERIOD_MIN_BYTES 1024
+#define AU1XPSC_BUFFER_MIN_BYTES 65536
+
+#define AU1XPSC_PCM_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | \
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE | \
+ 0)
+
+/* PCM hardware DMA capabilities - platform specific */
+static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = AU1XPSC_PCM_FMTS,
+ .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
+ .period_bytes_max = 4096 * 1024 - 1,
+ .periods_min = 2,
+ .periods_max = 4096, /* 2 to as-much-as-you-like */
+ .buffer_bytes_max = 4096 * 1024 - 1,
+ .fifo_size = 16, /* fifo entries of AC97/I2S PSC */
+};
+
+static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
+{
+ au1xxx_dbdma_put_source_flags(cd->ddma_chan,
+ (void *)phys_to_virt(cd->dma_area),
+ cd->period_bytes, DDMA_FLAGS_IE);
+
+ /* update next-to-queue period */
+ ++cd->q_period;
+ cd->dma_area += cd->period_bytes;
+ if (cd->q_period >= cd->periods) {
+ cd->q_period = 0;
+ cd->dma_area = cd->dma_area_s;
+ }
+}
+
+static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
+{
+ au1xxx_dbdma_put_dest_flags(cd->ddma_chan,
+ (void *)phys_to_virt(cd->dma_area),
+ cd->period_bytes, DDMA_FLAGS_IE);
+
+ /* update next-to-queue period */
+ ++cd->q_period;
+ cd->dma_area += cd->period_bytes;
+ if (cd->q_period >= cd->periods) {
+ cd->q_period = 0;
+ cd->dma_area = cd->dma_area_s;
+ }
+}
+
+static void au1x_pcm_dmatx_cb(int irq, void *dev_id)
+{
+ struct au1xpsc_audio_dmadata *cd = dev_id;
+
+ cd->pos += cd->period_bytes;
+ if (++cd->curr_period >= cd->periods) {
+ cd->pos = 0;
+ cd->curr_period = 0;
+ }
+ snd_pcm_period_elapsed(cd->substream);
+ au1x_pcm_queue_tx(cd);
+}
+
+static void au1x_pcm_dmarx_cb(int irq, void *dev_id)
+{
+ struct au1xpsc_audio_dmadata *cd = dev_id;
+
+ cd->pos += cd->period_bytes;
+ if (++cd->curr_period >= cd->periods) {
+ cd->pos = 0;
+ cd->curr_period = 0;
+ }
+ snd_pcm_period_elapsed(cd->substream);
+ au1x_pcm_queue_rx(cd);
+}
+
+static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd)
+{
+ if (pcd->ddma_chan) {
+ au1xxx_dbdma_stop(pcd->ddma_chan);
+ au1xxx_dbdma_reset(pcd->ddma_chan);
+ au1xxx_dbdma_chan_free(pcd->ddma_chan);
+ pcd->ddma_chan = 0;
+ pcd->msbits = 0;
+ }
+}
+
+/* in case of missing DMA ring or changed TX-source / RX-dest bit widths,
+ * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according
+ * to ALSA-supplied sample depth. This is due to limitations in the dbdma api
+ * (cannot adjust source/dest widths of already allocated descriptor ring).
+ */
+static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
+ int stype, int msbits)
+{
+ /* DMA only in 8/16/32 bit widths */
+ if (msbits == 24)
+ msbits = 32;
+
+ /* check current config: correct bits and descriptors allocated? */
+ if ((pcd->ddma_chan) && (msbits == pcd->msbits))
+ goto out; /* all ok! */
+
+ au1x_pcm_dbdma_free(pcd);
+
+ if (stype == PCM_RX)
+ pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
+ DSCR_CMD0_ALWAYS,
+ au1x_pcm_dmarx_cb, (void *)pcd);
+ else
+ pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS,
+ pcd->ddma_id,
+ au1x_pcm_dmatx_cb, (void *)pcd);
+
+ if (!pcd->ddma_chan)
+ return -ENOMEM;;
+
+ au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits);
+ au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2);
+
+ pcd->msbits = msbits;
+
+ au1xxx_dbdma_stop(pcd->ddma_chan);
+ au1xxx_dbdma_reset(pcd->ddma_chan);
+
+out:
+ return 0;
+}
+
+static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct au1xpsc_audio_dmadata *pcd;
+ int stype, ret;
+
+ ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (ret < 0)
+ goto out;
+
+ stype = SUBSTREAM_TYPE(substream);
+ pcd = au1xpsc_audio_pcmdma[stype];
+
+ DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
+ "runtime->min_align %d\n",
+ (unsigned long)runtime->dma_area,
+ (unsigned long)runtime->dma_addr, runtime->dma_bytes,
+ runtime->min_align);
+
+ DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits,
+ params_periods(params), params_period_bytes(params), stype);
+
+ ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits);
+ if (ret) {
+ MSG("DDMA channel (re)alloc failed!\n");
+ goto out;
+ }
+
+ pcd->substream = substream;
+ pcd->period_bytes = params_period_bytes(params);
+ pcd->periods = params_periods(params);
+ pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr;
+ pcd->q_period = 0;
+ pcd->curr_period = 0;
+ pcd->pos = 0;
+
+ ret = 0;
+out:
+ return ret;
+}
+
+static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct au1xpsc_audio_dmadata *pcd =
+ au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)];
+
+ au1xxx_dbdma_reset(pcd->ddma_chan);
+
+ if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+ au1x_pcm_queue_rx(pcd);
+ au1x_pcm_queue_rx(pcd);
+ } else {
+ au1x_pcm_queue_tx(pcd);
+ au1x_pcm_queue_tx(pcd);
+ }
+
+ return 0;
+}
+
+static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ au1xxx_dbdma_start(c);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ au1xxx_dbdma_stop(c);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t
+au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ return bytes_to_frames(substream->runtime,
+ au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos);
+}
+
+static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
+{
+ snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
+ return 0;
+}
+
+static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
+{
+ au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]);
+ return 0;
+}
+
+struct snd_pcm_ops au1xpsc_pcm_ops = {
+ .open = au1xpsc_pcm_open,
+ .close = au1xpsc_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = au1xpsc_pcm_hw_params,
+ .hw_free = au1xpsc_pcm_hw_free,
+ .prepare = au1xpsc_pcm_prepare,
+ .trigger = au1xpsc_pcm_trigger,
+ .pointer = au1xpsc_pcm_pointer,
+};
+
+static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int au1xpsc_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
+
+ return 0;
+}
+
+static int au1xpsc_pcm_probe(struct platform_device *pdev)
+{
+ struct resource *r;
+ int ret;
+
+ if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX])
+ return -EBUSY;
+
+ /* TX DMA */
+ au1xpsc_audio_pcmdma[PCM_TX]
+ = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+ if (!au1xpsc_audio_pcmdma[PCM_TX])
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out1;
+ }
+ (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start;
+
+ /* RX DMA */
+ au1xpsc_audio_pcmdma[PCM_RX]
+ = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+ if (!au1xpsc_audio_pcmdma[PCM_RX])
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r) {
+ ret = -ENODEV;
+ goto out2;
+ }
+ (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start;
+
+ return 0;
+
+out2:
+ kfree(au1xpsc_audio_pcmdma[PCM_RX]);
+ au1xpsc_audio_pcmdma[PCM_RX] = NULL;
+out1:
+ kfree(au1xpsc_audio_pcmdma[PCM_TX]);
+ au1xpsc_audio_pcmdma[PCM_TX] = NULL;
+ return ret;
+}
+
+static int au1xpsc_pcm_remove(struct platform_device *pdev)
+{
+ int i;
+
+ for (i = 0; i < 2; i++) {
+ if (au1xpsc_audio_pcmdma[i]) {
+ au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]);
+ kfree(au1xpsc_audio_pcmdma[i]);
+ au1xpsc_audio_pcmdma[i] = NULL;
+ }
+ }
+
+ return 0;
+}
+
+/* au1xpsc audio platform */
+struct snd_soc_platform au1xpsc_soc_platform = {
+ .name = "au1xpsc-pcm-dbdma",
+ .probe = au1xpsc_pcm_probe,
+ .remove = au1xpsc_pcm_remove,
+ .pcm_ops = &au1xpsc_pcm_ops,
+ .pcm_new = au1xpsc_pcm_new,
+ .pcm_free = au1xpsc_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);
+
+static int __init au1xpsc_audio_dbdma_init(void)
+{
+ au1xpsc_audio_pcmdma[PCM_TX] = NULL;
+ au1xpsc_audio_pcmdma[PCM_RX] = NULL;
+ return 0;
+}
+
+static void __exit au1xpsc_audio_dbdma_exit(void)
+{
+}
+
+module_init(au1xpsc_audio_dbdma_init);
+module_exit(au1xpsc_audio_dbdma_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 0000000..57facba
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -0,0 +1,387 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Au1xxx-PSC AC97 glue.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ * of a PSC. Multiple independent audio devices are impossible
+ * with ASoC v1.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+#define AC97_DIR \
+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AC97_RATES \
+ SNDRV_PCM_RATE_8000_48000
+
+#define AC97_FMTS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
+
+#define AC97PCR_START(stype) \
+ ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+#define AC97PCR_STOP(stype) \
+ ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+#define AC97PCR_CLRFIFO(stype) \
+ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+
+/* instance data. There can be only one, MacLeod!!!! */
+static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
+
+/* AC97 controller reads codec register */
+static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ /* FIXME */
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+ unsigned short data, tmo;
+
+ au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata));
+ au_sync();
+
+ tmo = 1000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
+ udelay(2);
+
+ if (!tmo)
+ data = 0xffff;
+ else
+ data = au_readl(AC97_CDC(pscdata)) & 0xffff;
+
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ au_sync();
+
+ return data;
+}
+
+/* AC97 controller writes to codec register */
+static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ /* FIXME */
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+ unsigned int tmo;
+
+ au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata));
+ au_sync();
+ tmo = 1000;
+ while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
+ au_sync();
+
+ au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ au_sync();
+}
+
+/* AC97 controller asserts a warm reset */
+static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ /* FIXME */
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+
+ au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata));
+ au_sync();
+ msleep(10);
+ au_writel(0, AC97_RST(pscdata));
+ au_sync();
+}
+
+static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ /* FIXME */
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+ int i;
+
+ /* disable PSC during cold reset */
+ au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+ au_sync();
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata));
+ au_sync();
+
+ /* issue cold reset */
+ au_writel(PSC_AC97RST_RST, AC97_RST(pscdata));
+ au_sync();
+ msleep(500);
+ au_writel(0, AC97_RST(pscdata));
+ au_sync();
+
+ /* enable PSC */
+ au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+ au_sync();
+
+ /* wait for PSC to indicate it's ready */
+ i = 100000;
+ while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
+ au_sync();
+
+ if (i == 0) {
+ printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n");
+ return;
+ }
+
+ /* enable the ac97 function */
+ au_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ au_sync();
+
+ /* wait for AC97 core to become ready */
+ i = 100000;
+ while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
+ au_sync();
+ if (i == 0)
+ printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = au1xpsc_ac97_read,
+ .write = au1xpsc_ac97_write,
+ .reset = au1xpsc_ac97_cold_reset,
+ .warm_reset = au1xpsc_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ /* FIXME */
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+ unsigned long r, stat;
+ int chans, stype = SUBSTREAM_TYPE(substream);
+
+ chans = params_channels(params);
+
+ r = au_readl(AC97_CFG(pscdata));
+ stat = au_readl(AC97_STAT(pscdata));
+
+ /* already active? */
+ if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) {
+ /* reject parameters not currently set up */
+ if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) ||
+ (pscdata->rate != params_rate(params)))
+ return -EINVAL;
+ } else {
+ /* disable AC97 device controller first */
+ au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ au_sync();
+
+ /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
+ r &= ~PSC_AC97CFG_LEN_MASK;
+ r |= PSC_AC97CFG_SET_LEN(params->msbits);
+
+ /* channels: enable slots for front L/R channel */
+ if (stype == PCM_TX) {
+ r &= ~PSC_AC97CFG_TXSLOT_MASK;
+ r |= PSC_AC97CFG_TXSLOT_ENA(3);
+ r |= PSC_AC97CFG_TXSLOT_ENA(4);
+ } else {
+ r &= ~PSC_AC97CFG_RXSLOT_MASK;
+ r |= PSC_AC97CFG_RXSLOT_ENA(3);
+ r |= PSC_AC97CFG_RXSLOT_ENA(4);
+ }
+
+ /* finally enable the AC97 controller again */
+ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ au_sync();
+
+ pscdata->cfg = r;
+ pscdata->rate = params_rate(params);
+ }
+
+ return 0;
+}
+
+static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ /* FIXME */
+ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+ int ret, stype = SUBSTREAM_TYPE(substream);
+
+ ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ au_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
+ au_sync();
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
+ au_sync();
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static int au1xpsc_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ int ret;
+ struct resource *r;
+ unsigned long sel;
+
+ if (au1xpsc_ac97_workdata)
+ return -EBUSY;
+
+ au1xpsc_ac97_workdata =
+ kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
+ if (!au1xpsc_ac97_workdata)
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ au1xpsc_ac97_workdata->ioarea =
+ request_mem_region(r->start, r->end - r->start + 1,
+ "au1xpsc_ac97");
+ if (!au1xpsc_ac97_workdata->ioarea)
+ goto out0;
+
+ au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff);
+ if (!au1xpsc_ac97_workdata->mmio)
+ goto out1;
+
+ /* configuration: max dma trigger threshold, enable ac97 */
+ au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
+ PSC_AC97CFG_TT_FIFO8 |
+ PSC_AC97CFG_DE_ENABLE;
+
+ /* preserve PSC clock source set up by platform (dev.platform_data
+ * is already occupied by soc layer)
+ */
+ sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK;
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+ au_sync();
+ au_writel(0, PSC_SEL(au1xpsc_ac97_workdata));
+ au_sync();
+ au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata));
+ au_sync();
+ /* next up: cold reset. Dont check for PSC-ready now since
+ * there may not be any codec clock yet.
+ */
+
+ return 0;
+
+out1:
+ release_resource(au1xpsc_ac97_workdata->ioarea);
+ kfree(au1xpsc_ac97_workdata->ioarea);
+out0:
+ kfree(au1xpsc_ac97_workdata);
+ au1xpsc_ac97_workdata = NULL;
+ return ret;
+}
+
+static void au1xpsc_ac97_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ /* disable PSC completely */
+ au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+ au_sync();
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+ au_sync();
+
+ iounmap(au1xpsc_ac97_workdata->mmio);
+ release_resource(au1xpsc_ac97_workdata->ioarea);
+ kfree(au1xpsc_ac97_workdata->ioarea);
+ kfree(au1xpsc_ac97_workdata);
+ au1xpsc_ac97_workdata = NULL;
+}
+
+static int au1xpsc_ac97_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ /* save interesting registers and disable PSC */
+ au1xpsc_ac97_workdata->pm[0] =
+ au_readl(PSC_SEL(au1xpsc_ac97_workdata));
+
+ au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+ au_sync();
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+ au_sync();
+
+ return 0;
+}
+
+static int au1xpsc_ac97_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ /* restore PSC clock config */
+ au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
+ PSC_SEL(au1xpsc_ac97_workdata));
+ au_sync();
+
+ /* after this point the ac97 core will cold-reset the codec.
+ * During cold-reset the PSC is reinitialized and the last
+ * configuration set up in hw_params() is restored.
+ */
+ return 0;
+}
+
+struct snd_soc_dai au1xpsc_ac97_dai = {
+ .name = "au1xpsc_ac97",
+ .type = SND_SOC_DAI_AC97,
+ .probe = au1xpsc_ac97_probe,
+ .remove = au1xpsc_ac97_remove,
+ .suspend = au1xpsc_ac97_suspend,
+ .resume = au1xpsc_ac97_resume,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = {
+ .trigger = au1xpsc_ac97_trigger,
+ .hw_params = au1xpsc_ac97_hw_params,
+ },
+};
+EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
+
+static int __init au1xpsc_ac97_init(void)
+{
+ au1xpsc_ac97_workdata = NULL;
+ return 0;
+}
+
+static void __exit au1xpsc_ac97_exit(void)
+{
+}
+
+module_init(au1xpsc_ac97_init);
+module_exit(au1xpsc_ac97_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 0000000..9384702
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -0,0 +1,414 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Au1xxx-PSC I2S glue.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ * of a PSC. Multiple independent audio devices are impossible
+ * with ASoC v1.
+ * NOTE: so far only PSC slave mode (bit- and frameclock) is supported.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/* supported I2S DAI hardware formats */
+#define AU1XPSC_I2S_DAIFMT \
+ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \
+ SND_SOC_DAIFMT_NB_NF)
+
+/* supported I2S direction */
+#define AU1XPSC_I2S_DIR \
+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AU1XPSC_I2S_RATES \
+ SNDRV_PCM_RATE_8000_192000
+
+#define AU1XPSC_I2S_FMTS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+#define I2SSTAT_BUSY(stype) \
+ ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+#define I2SPCR_START(stype) \
+ ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+#define I2SPCR_STOP(stype) \
+ ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+#define I2SPCR_CLRFIFO(stype) \
+ ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+
+
+/* instance data. There can be only one, MacLeod!!!! */
+static struct au1xpsc_audio_data *au1xpsc_i2s_workdata;
+
+static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
+ unsigned long ct;
+ int ret;
+
+ ret = -EINVAL;
+
+ ct = pscdata->cfg;
+
+ ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ); /* left-justified */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ct |= PSC_I2SCFG_XM; /* enable I2S mode */
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ ct |= PSC_I2SCFG_MLJ; /* LSB (right-) justified */
+ break;
+ default:
+ goto out;
+ }
+
+ ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI); /* IB-IF */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ ct |= PSC_I2SCFG_BI;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ ct |= PSC_I2SCFG_WI;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ goto out;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */
+ ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */
+ ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */
+ break;
+ default:
+ goto out;
+ }
+
+ pscdata->cfg = ct;
+ ret = 0;
+out:
+ return ret;
+}
+
+static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
+
+ int cfgbits;
+ unsigned long stat;
+
+ /* check if the PSC is already streaming data */
+ stat = au_readl(I2S_STAT(pscdata));
+ if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) {
+ /* reject parameters not currently set up in hardware */
+ cfgbits = au_readl(I2S_CFG(pscdata));
+ if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) ||
+ (params_rate(params) != pscdata->rate))
+ return -EINVAL;
+ } else {
+ /* set sample bitdepth */
+ pscdata->cfg &= ~(0x1f << 4);
+ pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits);
+ /* remember current rate for other stream */
+ pscdata->rate = params_rate(params);
+ }
+ return 0;
+}
+
+/* Configure PSC late: on my devel systems the codec is I2S master and
+ * supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit. ASoC
+ * uses aggressive PM and switches the codec off when it is not in use
+ * which also means the PSC unit doesn't get any clocks and is therefore
+ * dead. That's why this chunk here gets called from the trigger callback
+ * because I can be reasonably certain the codec is driving the clocks.
+ */
+static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata)
+{
+ unsigned long tmo;
+
+ /* bring PSC out of sleep, and configure I2S unit */
+ au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+ au_sync();
+
+ tmo = 1000000;
+ while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo)
+ tmo--;
+
+ if (!tmo)
+ goto psc_err;
+
+ au_writel(0, I2S_CFG(pscdata));
+ au_sync();
+ au_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata));
+ au_sync();
+
+ /* wait for I2S controller to become ready */
+ tmo = 1000000;
+ while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo)
+ tmo--;
+
+ if (tmo)
+ return 0;
+
+psc_err:
+ au_writel(0, I2S_CFG(pscdata));
+ au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+ au_sync();
+ return -ETIMEDOUT;
+}
+
+static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype)
+{
+ unsigned long tmo, stat;
+ int ret;
+
+ ret = 0;
+
+ /* if both TX and RX are idle, configure the PSC */
+ stat = au_readl(I2S_STAT(pscdata));
+ if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
+ ret = au1xpsc_i2s_configure(pscdata);
+ if (ret)
+ goto out;
+ }
+
+ au_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata));
+ au_sync();
+ au_writel(I2SPCR_START(stype), I2S_PCR(pscdata));
+ au_sync();
+
+ /* wait for start confirmation */
+ tmo = 1000000;
+ while (!(au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+ tmo--;
+
+ if (!tmo) {
+ au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+ au_sync();
+ ret = -ETIMEDOUT;
+ }
+out:
+ return ret;
+}
+
+static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
+{
+ unsigned long tmo, stat;
+
+ au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+ au_sync();
+
+ /* wait for stop confirmation */
+ tmo = 1000000;
+ while ((au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+ tmo--;
+
+ /* if both TX and RX are idle, disable PSC */
+ stat = au_readl(I2S_STAT(pscdata));
+ if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
+ au_writel(0, I2S_CFG(pscdata));
+ au_sync();
+ au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+ au_sync();
+ }
+ return 0;
+}
+
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
+ int ret, stype = SUBSTREAM_TYPE(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ ret = au1xpsc_i2s_start(pscdata, stype);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ ret = au1xpsc_i2s_stop(pscdata, stype);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static int au1xpsc_i2s_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct resource *r;
+ unsigned long sel;
+ int ret;
+
+ if (au1xpsc_i2s_workdata)
+ return -EBUSY;
+
+ au1xpsc_i2s_workdata =
+ kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
+ if (!au1xpsc_i2s_workdata)
+ return -ENOMEM;
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ ret = -ENODEV;
+ goto out0;
+ }
+
+ ret = -EBUSY;
+ au1xpsc_i2s_workdata->ioarea =
+ request_mem_region(r->start, r->end - r->start + 1,
+ "au1xpsc_i2s");
+ if (!au1xpsc_i2s_workdata->ioarea)
+ goto out0;
+
+ au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff);
+ if (!au1xpsc_i2s_workdata->mmio)
+ goto out1;
+
+ /* preserve PSC clock source set up by platform (dev.platform_data
+ * is already occupied by soc layer)
+ */
+ sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK;
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+ au_sync();
+ au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata));
+ au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+ au_sync();
+
+ /* preconfigure: set max rx/tx fifo depths */
+ au1xpsc_i2s_workdata->cfg |=
+ PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8;
+
+ /* don't wait for I2S core to become ready now; clocks may not
+ * be running yet; depending on clock input for PSC a wait might
+ * time out.
+ */
+
+ return 0;
+
+out1:
+ release_resource(au1xpsc_i2s_workdata->ioarea);
+ kfree(au1xpsc_i2s_workdata->ioarea);
+out0:
+ kfree(au1xpsc_i2s_workdata);
+ au1xpsc_i2s_workdata = NULL;
+ return ret;
+}
+
+static void au1xpsc_i2s_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+ au_sync();
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+ au_sync();
+
+ iounmap(au1xpsc_i2s_workdata->mmio);
+ release_resource(au1xpsc_i2s_workdata->ioarea);
+ kfree(au1xpsc_i2s_workdata->ioarea);
+ kfree(au1xpsc_i2s_workdata);
+ au1xpsc_i2s_workdata = NULL;
+}
+
+static int au1xpsc_i2s_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ /* save interesting register and disable PSC */
+ au1xpsc_i2s_workdata->pm[0] =
+ au_readl(PSC_SEL(au1xpsc_i2s_workdata));
+
+ au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+ au_sync();
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+ au_sync();
+
+ return 0;
+}
+
+static int au1xpsc_i2s_resume(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ /* select I2S mode and PSC clock */
+ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+ au_sync();
+ au_writel(0, PSC_SEL(au1xpsc_i2s_workdata));
+ au_sync();
+ au_writel(au1xpsc_i2s_workdata->pm[0],
+ PSC_SEL(au1xpsc_i2s_workdata));
+ au_sync();
+
+ return 0;
+}
+
+struct snd_soc_dai au1xpsc_i2s_dai = {
+ .name = "au1xpsc_i2s",
+ .type = SND_SOC_DAI_I2S,
+ .probe = au1xpsc_i2s_probe,
+ .remove = au1xpsc_i2s_remove,
+ .suspend = au1xpsc_i2s_suspend,
+ .resume = au1xpsc_i2s_resume,
+ .playback = {
+ .rates = AU1XPSC_I2S_RATES,
+ .formats = AU1XPSC_I2S_FMTS,
+ .channels_min = 2,
+ .channels_max = 8, /* 2 without external help */
+ },
+ .capture = {
+ .rates = AU1XPSC_I2S_RATES,
+ .formats = AU1XPSC_I2S_FMTS,
+ .channels_min = 2,
+ .channels_max = 8, /* 2 without external help */
+ },
+ .ops = {
+ .trigger = au1xpsc_i2s_trigger,
+ .hw_params = au1xpsc_i2s_hw_params,
+ },
+ .dai_ops = {
+ .set_fmt = au1xpsc_i2s_set_fmt,
+ },
+};
+EXPORT_SYMBOL(au1xpsc_i2s_dai);
+
+static int __init au1xpsc_i2s_init(void)
+{
+ au1xpsc_i2s_workdata = NULL;
+ return 0;
+}
+
+static void __exit au1xpsc_i2s_exit(void)
+{
+}
+
+module_init(au1xpsc_i2s_init);
+module_exit(au1xpsc_i2s_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 0000000..8fdb1a0
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -0,0 +1,53 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ * of a PSC. Multiple independent audio devices are impossible
+ * with ASoC v1.
+ */
+
+#ifndef _AU1X_PCM_H
+#define _AU1X_PCM_H
+
+extern struct snd_soc_dai au1xpsc_ac97_dai;
+extern struct snd_soc_dai au1xpsc_i2s_dai;
+extern struct snd_soc_platform au1xpsc_soc_platform;
+extern struct snd_ac97_bus_ops soc_ac97_ops;
+
+struct au1xpsc_audio_data {
+ void __iomem *mmio;
+
+ unsigned long cfg;
+ unsigned long rate;
+
+ unsigned long pm[2];
+ struct resource *ioarea;
+};
+
+#define PCM_TX 0
+#define PCM_RX 1
+
+#define SUBSTREAM_TYPE(substream) \
+ ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
+
+/* easy access macros */
+#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
+#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
+#define I2S_STAT(x) ((unsigned long)((x)->mmio) + PSC_I2SSTAT_OFFSET)
+#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG_OFFSET)
+#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR_OFFSET)
+#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG_OFFSET)
+#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC_OFFSET)
+#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT_OFFSET)
+#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR_OFFSET)
+#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST_OFFSET)
+#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT_OFFSET)
+
+#endif
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 0000000..f75ae7f
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * Sample Au12x0/Au1550 PSC AC97 sound machine.
+ *
+ * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms outlined in the file COPYING at the root of this
+ * source archive.
+ *
+ * This is a very generic AC97 sound machine driver for boards which
+ * have (AC97) audio at PSC1 (e.g. DB1200 demoboards).
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+
+#include "../codecs/ac97.h"
+#include "psc.h"
+
+static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */
+ .codec_dai = &ac97_dai, /* see codecs/ac97.c */
+ .init = au1xpsc_sample_ac97_init,
+ .ops = NULL,
+};
+
+static struct snd_soc_machine au1xpsc_sample_ac97_machine = {
+ .name = "Au1xxx PSC AC97 Audio",
+ .dai_link = &au1xpsc_sample_ac97_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
+ .machine = &au1xpsc_sample_ac97_machine,
+ .platform = &au1xpsc_soc_platform, /* see dbdma2.c */
+ .codec_dev = &soc_codec_dev_ac97,
+};
+
+static struct resource au1xpsc_psc1_res[] = {
+ [0] = {
+ .start = CPHYSADDR(PSC1_BASE_ADDR),
+ .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff,
+ .flags = IORESOURCE_MEM,
+ },
+ [1] = {
+#ifdef CONFIG_SOC_AU1200
+ .start = AU1200_PSC1_INT,
+ .end = AU1200_PSC1_INT,
+#elif defined(CONFIG_SOC_AU1550)
+ .start = AU1550_PSC1_INT,
+ .end = AU1550_PSC1_INT,
+#endif
+ .flags = IORESOURCE_IRQ,
+ },
+ [2] = {
+ .start = DSCR_CMD0_PSC1_TX,
+ .end = DSCR_CMD0_PSC1_TX,
+ .flags = IORESOURCE_DMA,
+ },
+ [3] = {
+ .start = DSCR_CMD0_PSC1_RX,
+ .end = DSCR_CMD0_PSC1_RX,
+ .flags = IORESOURCE_DMA,
+ },
+};
+
+static struct platform_device *au1xpsc_sample_ac97_dev;
+
+static int __init au1xpsc_sample_ac97_load(void)
+{
+ int ret;
+
+#ifdef CONFIG_SOC_AU1200
+ unsigned long io;
+
+ /* modify sys_pinfunc for AC97 on PSC1 */
+ io = au_readl(SYS_PINFUNC);
+ io |= SYS_PINFUNC_P1C;
+ io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B);
+ au_writel(io, SYS_PINFUNC);
+ au_sync();
+#endif
+
+ ret = -ENOMEM;
+
+ /* setup PSC clock source for AC97 part: external clock provided
+ * by codec. The psc-ac97.c driver depends on this setting!
+ */
+ au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET);
+ au_sync();
+
+ au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1);
+ if (!au1xpsc_sample_ac97_dev)
+ goto out;
+
+ au1xpsc_sample_ac97_dev->resource =
+ kmemdup(au1xpsc_psc1_res, sizeof(struct resource) *
+ ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL);
+ au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res);
+ au1xpsc_sample_ac97_dev->id = 1;
+
+ platform_set_drvdata(au1xpsc_sample_ac97_dev,
+ &au1xpsc_sample_ac97_devdata);
+ au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev;
+ ret = platform_device_add(au1xpsc_sample_ac97_dev);
+
+ if (ret) {
+ platform_device_put(au1xpsc_sample_ac97_dev);
+ au1xpsc_sample_ac97_dev = NULL;
+ }
+
+out:
+ return ret;
+}
+
+static void __exit au1xpsc_sample_ac97_exit(void)
+{
+ platform_device_unregister(au1xpsc_sample_ac97_dev);
+}
+
+module_init(au1xpsc_sample_ac97_load);
+module_exit(au1xpsc_sample_ac97_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
new file mode 100644
index 0000000..dc00620
--- /dev/null
+++ b/sound/soc/blackfin/Kconfig
@@ -0,0 +1,101 @@
+config SND_BF5XX_I2S
+ tristate "SoC I2S Audio for the ADI BF5xx chip"
+ depends on BLACKFIN && SND_SOC
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the Blackfin SPORT (synchronous serial ports) interface in I2S
+ mode (supports single stereo In/Out).
+ You will also need to select the audio interfaces to support below.
+
+config SND_BF5XX_SOC_SSM2602
+ tristate "SoC SSM2602 Audio support for BF52x ezkit"
+ depends on SND_BF5XX_I2S
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_SSM2602
+ select I2C
+ select I2C_BLACKFIN_TWI
+ help
+ Say Y if you want to add support for SoC audio on BF527-EZKIT.
+
+config SND_BF5XX_SOC_AD73311
+ tristate "SoC AD73311 Audio support for Blackfin"
+ depends on SND_BF5XX_I2S
+ select SND_BF5XX_SOC_I2S
+ select SND_SOC_AD73311
+ help
+ Say Y if you want to add support for AD73311 codec on Blackfin.
+
+config SND_BFIN_AD73311_SE
+ int "PF pin for AD73311L Chip Select"
+ depends on SND_BF5XX_SOC_AD73311
+ default 4
+ help
+ Enter the GPIO used to control AD73311's SE pin. Acceptable
+ values are 0 to 7
+
+config SND_BF5XX_AC97
+ tristate "SoC AC97 Audio for the ADI BF5xx chip"
+ depends on BLACKFIN && SND_SOC
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the Blackfin SPORT (synchronous serial ports) interface in slot 16
+ mode (pseudo AC97 interface).
+ You will also need to select the audio interfaces to support below.
+
+ Note:
+ AC97 codecs which do not implment the slot-16 mode will not function
+ properly with this driver. This driver is known to work with the
+ Analog Devices line of AC97 codecs.
+
+config SND_MMAP_SUPPORT
+ bool "Enable MMAP Support"
+ depends on SND_BF5XX_AC97
+ default y
+ help
+ Say y if you want AC97 driver to support mmap mode.
+ We introduce an intermediate buffer to simulate mmap.
+
+config SND_BF5XX_SOC_SPORT
+ tristate
+
+config SND_BF5XX_SOC_I2S
+ tristate
+ select SND_BF5XX_SOC_SPORT
+
+config SND_BF5XX_SOC_AC97
+ tristate
+ select AC97_BUS
+ select SND_SOC_AC97_BUS
+ select SND_BF5XX_SOC_SPORT
+
+config SND_BF5XX_SOC_AD1980
+ tristate "SoC AD1980/1 Audio support for BF5xx"
+ depends on SND_BF5XX_AC97
+ select SND_BF5XX_SOC_AC97
+ select SND_SOC_AD1980
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
+config SND_BF5XX_SPORT_NUM
+ int "Set a SPORT for Sound chip"
+ depends on (SND_BF5XX_I2S || SND_BF5XX_AC97)
+ range 0 3 if BF54x
+ range 0 1 if (BF53x || BF561)
+ default 0
+ help
+ Set the correct SPORT for sound chip.
+
+config SND_BF5XX_HAVE_COLD_RESET
+ bool "BOARD has COLD Reset GPIO"
+ depends on SND_BF5XX_AC97
+ default y if BFIN548_EZKIT
+ default n if !BFIN548_EZKIT
+
+config SND_BF5XX_RESET_GPIO_NUM
+ int "Set a GPIO for cold reset"
+ depends on SND_BF5XX_HAVE_COLD_RESET
+ range 0 159
+ default 19 if BFIN548_EZKIT
+ default 5 if BFIN537_STAMP
+ help
+ Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
new file mode 100644
index 0000000..97bb37a
--- /dev/null
+++ b/sound/soc/blackfin/Makefile
@@ -0,0 +1,21 @@
+# Blackfin Platform Support
+snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o
+snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o
+snd-soc-bf5xx-sport-objs := bf5xx-sport.o
+snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o
+snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o
+
+obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o
+obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o
+obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o
+obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o
+obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
+
+# Blackfin Machine Support
+snd-ad1980-objs := bf5xx-ad1980.o
+snd-ssm2602-objs := bf5xx-ssm2602.o
+snd-ad73311-objs := bf5xx-ad73311.o
+
+obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
+obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
+obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
new file mode 100644
index 0000000..25e50d2
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -0,0 +1,457 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-ac97-pcm.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: DMA Driver for AC97 sound chip
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+
+#include "bf5xx-ac97-pcm.h"
+#include "bf5xx-ac97.h"
+#include "bf5xx-sport.h"
+
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ bf5xx_pcm_to_ac97(
+ (struct ac97_frame *)sport->tx_dma_buf + sport->tx_pos,
+ (__u32 *)runtime->dma_area + sport->tx_pos, count);
+ sport->tx_pos += runtime->period_size;
+ if (sport->tx_pos >= runtime->buffer_size)
+ sport->tx_pos %= runtime->buffer_size;
+ sport->tx_delay_pos = sport->tx_pos;
+ } else {
+ bf5xx_ac97_to_pcm(
+ (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
+ (__u32 *)runtime->dma_area + sport->rx_pos, count);
+ sport->rx_pos += runtime->period_size;
+ if (sport->rx_pos >= runtime->buffer_size)
+ sport->rx_pos %= runtime->buffer_size;
+ }
+}
+#endif
+
+static void bf5xx_dma_irq(void *data)
+{
+ struct snd_pcm_substream *pcm = data;
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ struct sport_device *sport = runtime->private_data;
+ bf5xx_mmap_copy(pcm, runtime->period_size);
+ if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (sport->once == 0) {
+ snd_pcm_period_elapsed(pcm);
+ bf5xx_mmap_copy(pcm, runtime->period_size);
+ sport->once = 1;
+ }
+ }
+#endif
+ snd_pcm_period_elapsed(pcm);
+}
+
+/* The memory size for pure pcm data is 128*1024 = 0x20000 bytes.
+ * The total rx/tx buffer is for ac97 frame to hold all pcm data
+ * is 0x20000 * sizeof(struct ac97_frame) / 4.
+ */
+#ifdef CONFIG_SND_MMAP_SUPPORT
+static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+#else
+static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+#endif
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 0x10000,
+ .periods_min = 1,
+ .periods_max = PAGE_SIZE/32,
+ .buffer_bytes_max = 0x20000, /* 128 kbytes */
+ .fifo_size = 16,
+};
+
+static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ size_t size = bf5xx_pcm_hardware.buffer_bytes_max
+ * sizeof(struct ac97_frame) / 4;
+
+ snd_pcm_lib_malloc_pages(substream, size);
+
+ return 0;
+}
+
+static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ memset(runtime->dma_area, 0, runtime->buffer_size);
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+
+ /* An intermediate buffer is introduced for implementing mmap for
+ * SPORT working in TMD mode(include AC97).
+ */
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
+ runtime->period_size * sizeof(struct ac97_frame));
+ } else {
+ sport_set_rx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods,
+ runtime->period_size * sizeof(struct ac97_frame));
+ }
+#else
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_tx_dma(sport, runtime->dma_area, runtime->periods,
+ runtime->period_size * sizeof(struct ac97_frame));
+ } else {
+ sport_set_rx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_rx_dma(sport, runtime->dma_area, runtime->periods,
+ runtime->period_size * sizeof(struct ac97_frame));
+ }
+#endif
+ return 0;
+}
+
+static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ int ret = 0;
+
+ pr_debug("%s enter\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ bf5xx_mmap_copy(substream, runtime->period_size);
+ snd_pcm_period_elapsed(substream);
+ sport->tx_delay_pos = 0;
+ sport_tx_start(sport);
+ }
+ else
+ sport_rx_start(sport);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ sport->tx_pos = 0;
+#endif
+ sport_tx_stop(sport);
+ } else {
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ sport->rx_pos = 0;
+#endif
+ sport_rx_stop(sport);
+ }
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ unsigned int curr;
+
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ curr = sport->tx_delay_pos;
+ else
+ curr = sport->rx_pos;
+#else
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ curr = sport_curr_offset_tx(sport) / sizeof(struct ac97_frame);
+ else
+ curr = sport_curr_offset_rx(sport) / sizeof(struct ac97_frame);
+
+#endif
+ return curr;
+}
+
+static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+
+ pr_debug("%s enter\n", __func__);
+ snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ if (sport_handle != NULL)
+ runtime->private_data = sport_handle;
+ else {
+ pr_err("sport_handle is NULL\n");
+ return -1;
+ }
+ return 0;
+
+ out:
+ return ret;
+}
+
+static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+
+ pr_debug("%s enter\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ sport->once = 0;
+ memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
+ } else
+ memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
+
+ return 0;
+}
+
+#ifdef CONFIG_SND_MMAP_SUPPORT
+static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ size_t size = vma->vm_end - vma->vm_start;
+ vma->vm_start = (unsigned long)runtime->dma_area;
+ vma->vm_end = vma->vm_start + size;
+ vma->vm_flags |= VM_SHARED;
+ return 0 ;
+}
+#else
+static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos,
+ void __user *buf, snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ pr_debug("%s copy pos:0x%lx count:0x%lx\n",
+ substream->stream ? "Capture" : "Playback", pos, count);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ bf5xx_pcm_to_ac97(
+ (struct ac97_frame *)runtime->dma_area + pos,
+ buf, count);
+ else
+ bf5xx_ac97_to_pcm(
+ (struct ac97_frame *)runtime->dma_area + pos,
+ buf, count);
+ return 0;
+}
+#endif
+
+struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
+ .open = bf5xx_pcm_open,
+ .close = bf5xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = bf5xx_pcm_hw_params,
+ .hw_free = bf5xx_pcm_hw_free,
+ .prepare = bf5xx_pcm_prepare,
+ .trigger = bf5xx_pcm_trigger,
+ .pointer = bf5xx_pcm_pointer,
+#ifdef CONFIG_SND_MMAP_SUPPORT
+ .mmap = bf5xx_pcm_mmap,
+#else
+ .copy = bf5xx_pcm_copy,
+#endif
+};
+
+static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = bf5xx_pcm_hardware.buffer_bytes_max
+ * sizeof(struct ac97_frame) / 4;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area) {
+ pr_err("Failed to allocate dma memory\n");
+ pr_err("Please increase uncached DMA memory region\n");
+ return -ENOMEM;
+ }
+ buf->bytes = size;
+
+ pr_debug("%s, area:%p, size:0x%08lx\n", __func__,
+ buf->area, buf->bytes);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sport_handle->tx_buf = buf->area;
+ else
+ sport_handle->rx_buf = buf->area;
+
+/*
+ * Need to allocate local buffer when enable
+ * MMAP for SPORT working in TMD mode (include AC97).
+ */
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (!sport_handle->tx_dma_buf) {
+ sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \
+ size, &sport_handle->tx_dma_phy, GFP_KERNEL);
+ if (!sport_handle->tx_dma_buf) {
+ pr_err("Failed to allocate memory for tx dma \
+ buf - Please increase uncached DMA \
+ memory region\n");
+ return -ENOMEM;
+ } else
+ memset(sport_handle->tx_dma_buf, 0, size);
+ } else
+ memset(sport_handle->tx_dma_buf, 0, size);
+ } else {
+ if (!sport_handle->rx_dma_buf) {
+ sport_handle->rx_dma_buf = dma_alloc_coherent(NULL, \
+ size, &sport_handle->rx_dma_phy, GFP_KERNEL);
+ if (!sport_handle->rx_dma_buf) {
+ pr_err("Failed to allocate memory for rx dma \
+ buf - Please increase uncached DMA \
+ memory region\n");
+ return -ENOMEM;
+ } else
+ memset(sport_handle->rx_dma_buf, 0, size);
+ } else
+ memset(sport_handle->rx_dma_buf, 0, size);
+ }
+#endif
+ return 0;
+}
+
+static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ size_t size = bf5xx_pcm_hardware.buffer_bytes_max *
+ sizeof(struct ac97_frame) / 4;
+#endif
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ dma_free_coherent(NULL, buf->bytes, buf->area, 0);
+ buf->area = NULL;
+#if defined(CONFIG_SND_MMAP_SUPPORT)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (sport_handle->tx_dma_buf)
+ dma_free_coherent(NULL, size, \
+ sport_handle->tx_dma_buf, 0);
+ sport_handle->tx_dma_buf = NULL;
+ } else {
+
+ if (sport_handle->rx_dma_buf)
+ dma_free_coherent(NULL, size, \
+ sport_handle->rx_dma_buf, 0);
+ sport_handle->rx_dma_buf = NULL;
+ }
+#endif
+ }
+ if (sport_handle)
+ sport_done(sport_handle);
+}
+
+static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK;
+
+int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ pr_debug("%s enter\n", __func__);
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &bf5xx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+ if (dai->playback.channels_min) {
+ ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+struct snd_soc_platform bf5xx_ac97_soc_platform = {
+ .name = "bf5xx-audio",
+ .pcm_ops = &bf5xx_pcm_ac97_ops,
+ .pcm_new = bf5xx_pcm_ac97_new,
+ .pcm_free = bf5xx_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform);
+
+MODULE_AUTHOR("Cliff Cai");
+MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.h b/sound/soc/blackfin/bf5xx-ac97-pcm.h
new file mode 100644
index 0000000..350125a
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.h
@@ -0,0 +1,29 @@
+/*
+ * linux/sound/arm/bf5xx-ac97-pcm.h -- ALSA PCM interface for the Blackfin
+ *
+ * Copyright 2007 Analog Device Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _BF5XX_AC97_PCM_H
+#define _BF5XX_AC97_PCM_H
+
+struct bf5xx_pcm_dma_params {
+ char *name; /* stream identifier */
+};
+
+struct bf5xx_gpio {
+ u32 sys;
+ u32 rx;
+ u32 tx;
+ u32 clk;
+ u32 frm;
+};
+
+/* platform data */
+extern struct snd_soc_platform bf5xx_ac97_soc_platform;
+
+#endif
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
new file mode 100644
index 0000000..5e5aafb
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -0,0 +1,406 @@
+/*
+ * bf5xx-ac97.c -- AC97 support for the ADI blackfin chip.
+ *
+ * Author: Roy Huang
+ * Created: 11th. June 2007
+ * Copyright: Analog Device Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/irq.h>
+#include <asm/portmux.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+
+#include "bf5xx-sport.h"
+#include "bf5xx-ac97.h"
+
+#if defined(CONFIG_BF54x)
+#define PIN_REQ_SPORT_0 {P_SPORT0_TFS, P_SPORT0_DTPRI, P_SPORT0_TSCLK, \
+ P_SPORT0_RFS, P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}
+
+#define PIN_REQ_SPORT_1 {P_SPORT1_TFS, P_SPORT1_DTPRI, P_SPORT1_TSCLK, \
+ P_SPORT1_RFS, P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}
+
+#define PIN_REQ_SPORT_2 {P_SPORT2_TFS, P_SPORT2_DTPRI, P_SPORT2_TSCLK, \
+ P_SPORT2_RFS, P_SPORT2_DRPRI, P_SPORT2_RSCLK, 0}
+
+#define PIN_REQ_SPORT_3 {P_SPORT3_TFS, P_SPORT3_DTPRI, P_SPORT3_TSCLK, \
+ P_SPORT3_RFS, P_SPORT3_DRPRI, P_SPORT3_RSCLK, 0}
+#else
+#define PIN_REQ_SPORT_0 {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, \
+ P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}
+
+#define PIN_REQ_SPORT_1 {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, \
+ P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}
+#endif
+
+static int *cmd_count;
+static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
+
+#if defined(CONFIG_BF54x)
+static struct sport_param sport_params[4] = {
+ {
+ .dma_rx_chan = CH_SPORT0_RX,
+ .dma_tx_chan = CH_SPORT0_TX,
+ .err_irq = IRQ_SPORT0_ERR,
+ .regs = (struct sport_register *)SPORT0_TCR1,
+ },
+ {
+ .dma_rx_chan = CH_SPORT1_RX,
+ .dma_tx_chan = CH_SPORT1_TX,
+ .err_irq = IRQ_SPORT1_ERR,
+ .regs = (struct sport_register *)SPORT1_TCR1,
+ },
+ {
+ .dma_rx_chan = CH_SPORT2_RX,
+ .dma_tx_chan = CH_SPORT2_TX,
+ .err_irq = IRQ_SPORT2_ERR,
+ .regs = (struct sport_register *)SPORT2_TCR1,
+ },
+ {
+ .dma_rx_chan = CH_SPORT3_RX,
+ .dma_tx_chan = CH_SPORT3_TX,
+ .err_irq = IRQ_SPORT3_ERR,
+ .regs = (struct sport_register *)SPORT3_TCR1,
+ }
+};
+#else
+static struct sport_param sport_params[2] = {
+ {
+ .dma_rx_chan = CH_SPORT0_RX,
+ .dma_tx_chan = CH_SPORT0_TX,
+ .err_irq = IRQ_SPORT0_ERROR,
+ .regs = (struct sport_register *)SPORT0_TCR1,
+ },
+ {
+ .dma_rx_chan = CH_SPORT1_RX,
+ .dma_tx_chan = CH_SPORT1_TX,
+ .err_irq = IRQ_SPORT1_ERROR,
+ .regs = (struct sport_register *)SPORT1_TCR1,
+ }
+};
+#endif
+
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
+ size_t count)
+{
+ while (count--) {
+ dst->ac97_tag = TAG_VALID | TAG_PCM;
+ (dst++)->ac97_pcm = *src++;
+ }
+}
+EXPORT_SYMBOL(bf5xx_pcm_to_ac97);
+
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+ size_t count)
+{
+ while (count--)
+ *(dst++) = (src++)->ac97_pcm;
+}
+EXPORT_SYMBOL(bf5xx_ac97_to_pcm);
+
+static unsigned int sport_tx_curr_frag(struct sport_device *sport)
+{
+ return sport->tx_curr_frag = sport_curr_offset_tx(sport) / \
+ sport->tx_fragsize;
+}
+
+static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data)
+{
+ struct sport_device *sport = sport_handle;
+ int nextfrag = sport_tx_curr_frag(sport);
+ struct ac97_frame *nextwrite;
+
+ sport_incfrag(sport, &nextfrag, 1);
+
+ nextwrite = (struct ac97_frame *)(sport->tx_buf + \
+ nextfrag * sport->tx_fragsize);
+ pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n",
+ sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]);
+ nextwrite[cmd_count[nextfrag]].ac97_tag |= TAG_CMD;
+ nextwrite[cmd_count[nextfrag]].ac97_addr = addr;
+ nextwrite[cmd_count[nextfrag]].ac97_data = data;
+ ++cmd_count[nextfrag];
+ pr_debug("ac97_sport: Inserting %02x/%04x into fragment %d\n",
+ addr >> 8, data, nextfrag);
+}
+
+static unsigned short bf5xx_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct ac97_frame out_frame[2], in_frame[2];
+
+ pr_debug("%s enter 0x%x\n", __func__, reg);
+
+ /* When dma descriptor is enabled, the register should not be read */
+ if (sport_handle->tx_run || sport_handle->rx_run) {
+ pr_err("Could you send a mail to cliff.cai@analog.com "
+ "to report this?\n");
+ return -EFAULT;
+ }
+
+ memset(&out_frame, 0, 2 * sizeof(struct ac97_frame));
+ memset(&in_frame, 0, 2 * sizeof(struct ac97_frame));
+ out_frame[0].ac97_tag = TAG_VALID | TAG_CMD;
+ out_frame[0].ac97_addr = ((reg << 8) | 0x8000);
+ sport_send_and_recv(sport_handle, (unsigned char *)&out_frame,
+ (unsigned char *)&in_frame,
+ 2 * sizeof(struct ac97_frame));
+ return in_frame[1].ac97_data;
+}
+
+void bf5xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ pr_debug("%s enter 0x%x:0x%04x\n", __func__, reg, val);
+
+ if (sport_handle->tx_run) {
+ enqueue_cmd(ac97, (reg << 8), val); /* write */
+ enqueue_cmd(ac97, (reg << 8) | 0x8000, 0); /* read back */
+ } else {
+ struct ac97_frame frame;
+ memset(&frame, 0, sizeof(struct ac97_frame));
+ frame.ac97_tag = TAG_VALID | TAG_CMD;
+ frame.ac97_addr = (reg << 8);
+ frame.ac97_data = val;
+ sport_send_and_recv(sport_handle, (unsigned char *)&frame, \
+ NULL, sizeof(struct ac97_frame));
+ }
+}
+
+static void bf5xx_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+#if defined(CONFIG_BF54x) || defined(CONFIG_BF561) || \
+ (defined(BF537_FAMILY) && (CONFIG_SND_BF5XX_SPORT_NUM == 1))
+
+#define CONCAT(a, b, c) a ## b ## c
+#define BFIN_SPORT_RFS(x) CONCAT(P_SPORT, x, _RFS)
+
+ u16 per = BFIN_SPORT_RFS(CONFIG_SND_BF5XX_SPORT_NUM);
+ u16 gpio = P_IDENT(BFIN_SPORT_RFS(CONFIG_SND_BF5XX_SPORT_NUM));
+
+ pr_debug("%s enter\n", __func__);
+
+ peripheral_free(per);
+ gpio_request(gpio, "bf5xx-ac97");
+ gpio_direction_output(gpio, 1);
+ udelay(2);
+ gpio_set_value(gpio, 0);
+ udelay(1);
+ gpio_free(gpio);
+ peripheral_request(per, "soc-audio");
+#else
+ pr_info("%s: Not implemented\n", __func__);
+#endif
+}
+
+static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ pr_debug("%s enter\n", __func__);
+
+ /* It is specified for bf548-ezkit */
+ gpio_set_value(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 0);
+ /* Keep reset pin low for 1 ms */
+ mdelay(1);
+ gpio_set_value(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1);
+ /* Wait for bit clock recover */
+ mdelay(1);
+#else
+ pr_info("%s: Not implemented\n", __func__);
+#endif
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = bf5xx_ac97_read,
+ .write = bf5xx_ac97_write,
+ .warm_reset = bf5xx_ac97_warm_reset,
+ .reset = bf5xx_ac97_cold_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+#ifdef CONFIG_PM
+static int bf5xx_ac97_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct sport_device *sport =
+ (struct sport_device *)dai->private_data;
+
+ pr_debug("%s : sport %d\n", __func__, dai->id);
+ if (!dai->active)
+ return 0;
+ if (dai->capture.active)
+ sport_rx_stop(sport);
+ if (dai->playback.active)
+ sport_tx_stop(sport);
+ return 0;
+}
+
+static int bf5xx_ac97_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ int ret;
+ struct sport_device *sport =
+ (struct sport_device *)dai->private_data;
+
+ pr_debug("%s : sport %d\n", __func__, dai->id);
+ if (!dai->active)
+ return 0;
+
+ ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1));
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1));
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ if (dai->capture.active)
+ sport_rx_start(sport);
+ if (dai->playback.active)
+ sport_tx_start(sport);
+ return 0;
+}
+
+#else
+#define bf5xx_ac97_suspend NULL
+#define bf5xx_ac97_resume NULL
+#endif
+
+static int bf5xx_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ int ret;
+#if defined(CONFIG_BF54x)
+ u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1,
+ PIN_REQ_SPORT_2, PIN_REQ_SPORT_3};
+#else
+ u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1};
+#endif
+ cmd_count = (int *)get_zeroed_page(GFP_KERNEL);
+ if (cmd_count == NULL)
+ return -ENOMEM;
+
+ if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
+ pr_err("Requesting Peripherals failed\n");
+ return -EFAULT;
+ }
+
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ /* Request PB3 as reset pin */
+ if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) {
+ pr_err("Failed to request GPIO_%d for reset\n",
+ CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+ peripheral_free_list(&sport_req[sport_num][0]);
+ return -1;
+ }
+ gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1);
+#endif
+ sport_handle = sport_init(&sport_params[sport_num], 2, \
+ sizeof(struct ac97_frame), NULL);
+ if (!sport_handle) {
+ peripheral_free_list(&sport_req[sport_num][0]);
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+ return -ENODEV;
+ }
+ /*SPORT works in TDM mode to simulate AC97 transfers*/
+ ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ kfree(sport_handle);
+ peripheral_free_list(&sport_req[sport_num][0]);
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+ return -EBUSY;
+ }
+
+ ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1));
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ kfree(sport_handle);
+ peripheral_free_list(&sport_req[sport_num][0]);
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+ return -EBUSY;
+ }
+
+ ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1));
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ kfree(sport_handle);
+ peripheral_free_list(&sport_req[sport_num][0]);
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+ return -EBUSY;
+ }
+ return 0;
+}
+
+static void bf5xx_ac97_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ free_page((unsigned long)cmd_count);
+ cmd_count = NULL;
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+ gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+}
+
+struct snd_soc_dai bfin_ac97_dai = {
+ .name = "bf5xx-ac97",
+ .id = 0,
+ .type = SND_SOC_DAI_AC97,
+ .probe = bf5xx_ac97_probe,
+ .remove = bf5xx_ac97_remove,
+ .suspend = bf5xx_ac97_suspend,
+ .resume = bf5xx_ac97_resume,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+};
+EXPORT_SYMBOL_GPL(bfin_ac97_dai);
+
+MODULE_AUTHOR("Roy Huang");
+MODULE_DESCRIPTION("AC97 driver for ADI Blackfin");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
new file mode 100644
index 0000000..3f77cc5
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -0,0 +1,36 @@
+/*
+ * linux/sound/arm/bf5xx-ac97.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _BF5XX_AC97_H
+#define _BF5XX_AC97_H
+
+extern struct snd_ac97_bus_ops bf5xx_ac97_ops;
+extern struct snd_ac97 *ac97;
+/* Frame format in memory, only support stereo currently */
+struct ac97_frame {
+ u16 ac97_tag; /* slot 0 */
+ u16 ac97_addr; /* slot 1 */
+ u16 ac97_data; /* slot 2 */
+ u32 ac97_pcm; /* slot 3 and 4: left and right pcm data */
+} __attribute__ ((packed));
+
+#define TAG_VALID 0x8000
+#define TAG_CMD 0x6000
+#define TAG_PCM_LEFT 0x1000
+#define TAG_PCM_RIGHT 0x0800
+#define TAG_PCM (TAG_PCM_LEFT | TAG_PCM_RIGHT)
+
+extern struct snd_soc_dai bfin_ac97_dai;
+
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
+ size_t count);
+
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+ size_t count);
+
+#endif
diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c
new file mode 100644
index 0000000..124425d
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ad1980.c
@@ -0,0 +1,113 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-ad1980.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: Board driver for AD1980/1 audio codec
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <asm/dma.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <linux/gpio.h>
+#include <asm/portmux.h>
+
+#include "../codecs/ad1980.h"
+#include "bf5xx-sport.h"
+#include "bf5xx-ac97-pcm.h"
+#include "bf5xx-ac97.h"
+
+static struct snd_soc_machine bf5xx_board;
+
+static int bf5xx_board_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ pr_debug("%s enter\n", __func__);
+ cpu_dai->private_data = sport_handle;
+ return 0;
+}
+
+static struct snd_soc_ops bf5xx_board_ops = {
+ .startup = bf5xx_board_startup,
+};
+
+static struct snd_soc_dai_link bf5xx_board_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &bfin_ac97_dai,
+ .codec_dai = &ad1980_dai,
+ .ops = &bf5xx_board_ops,
+};
+
+static struct snd_soc_machine bf5xx_board = {
+ .name = "bf5xx-board",
+ .dai_link = &bf5xx_board_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_board_snd_devdata = {
+ .machine = &bf5xx_board,
+ .platform = &bf5xx_ac97_soc_platform,
+ .codec_dev = &soc_codec_dev_ad1980,
+};
+
+static struct platform_device *bf5xx_board_snd_device;
+
+static int __init bf5xx_board_init(void)
+{
+ int ret;
+
+ bf5xx_board_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bf5xx_board_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(bf5xx_board_snd_device, &bf5xx_board_snd_devdata);
+ bf5xx_board_snd_devdata.dev = &bf5xx_board_snd_device->dev;
+ ret = platform_device_add(bf5xx_board_snd_device);
+
+ if (ret)
+ platform_device_put(bf5xx_board_snd_device);
+
+ return ret;
+}
+
+static void __exit bf5xx_board_exit(void)
+{
+ platform_device_unregister(bf5xx_board_snd_device);
+}
+
+module_init(bf5xx_board_init);
+module_exit(bf5xx_board_exit);
+
+/* Module information */
+MODULE_AUTHOR("Cliff Cai");
+MODULE_DESCRIPTION("ALSA SoC AD1980/1 BF5xx board");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
new file mode 100644
index 0000000..622c9b9
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -0,0 +1,240 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-ad73311.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Thur Sep 25 2008
+ * Description: Board driver for ad73311 sound chip
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+
+#include <asm/blackfin.h>
+#include <asm/cacheflush.h>
+#include <asm/irq.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "../codecs/ad73311.h"
+#include "bf5xx-sport.h"
+#include "bf5xx-i2s-pcm.h"
+#include "bf5xx-i2s.h"
+
+#if CONFIG_SND_BF5XX_SPORT_NUM == 0
+#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1
+#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1
+#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2
+#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16
+#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT
+#else
+#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1
+#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1
+#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2
+#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16
+#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT
+#endif
+
+#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
+
+static struct snd_soc_machine bf5xx_ad73311;
+
+static int snd_ad73311_startup(void)
+{
+ pr_debug("%s enter\n", __func__);
+
+ /* Pull up SE pin on AD73311L */
+ gpio_set_value(GPIO_SE, 1);
+ return 0;
+}
+
+static int snd_ad73311_configure(void)
+{
+ unsigned short ctrl_regs[6];
+ unsigned short status = 0;
+ int count = 0;
+
+ /* DMCLK = MCLK = 16.384 MHz
+ * SCLK = DMCLK/8 = 2.048 MHz
+ * Sample Rate = DMCLK/2048 = 8 KHz
+ */
+ ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \
+ REGB_SCDIV(0) | REGB_DIRATE(0);
+ ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \
+ REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ;
+ ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \
+ REGD_IGS(2);
+ ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f);
+ ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ;
+ ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA;
+
+ local_irq_disable();
+ snd_ad73311_startup();
+ udelay(1);
+
+ bfin_write_SPORT_TCR1(TFSR);
+ bfin_write_SPORT_TCR2(0xF);
+ SSYNC();
+
+ /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to
+ * FIFO before enable SPORT to transfer the data
+ */
+ for (count = 0; count < 6; count++)
+ bfin_write_SPORT_TX16(ctrl_regs[count]);
+ SSYNC();
+ bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN);
+ SSYNC();
+
+ /* When TUVF is set, the data is already send out */
+ while (!(status & TUVF) && count++ < 10000) {
+ udelay(1);
+ status = bfin_read_SPORT_STAT();
+ SSYNC();
+ }
+ bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN);
+ SSYNC();
+ local_irq_enable();
+
+ if (count == 10000) {
+ printk(KERN_ERR "ad73311: failed to configure codec\n");
+ return -1;
+ }
+ return 0;
+}
+
+static int bf5xx_probe(struct platform_device *pdev)
+{
+ int err;
+ if (gpio_request(GPIO_SE, "AD73311_SE")) {
+ printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE);
+ return -EBUSY;
+ }
+
+ gpio_direction_output(GPIO_SE, 0);
+
+ err = snd_ad73311_configure();
+ if (err < 0)
+ return -EFAULT;
+
+ return 0;
+}
+
+static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ pr_debug("%s enter\n", __func__);
+ cpu_dai->private_data = sport_handle;
+ return 0;
+}
+
+static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+
+ pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
+ params_format(params));
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+
+static struct snd_soc_ops bf5xx_ad73311_ops = {
+ .startup = bf5xx_ad73311_startup,
+ .hw_params = bf5xx_ad73311_hw_params,
+};
+
+static struct snd_soc_dai_link bf5xx_ad73311_dai = {
+ .name = "ad73311",
+ .stream_name = "AD73311",
+ .cpu_dai = &bf5xx_i2s_dai,
+ .codec_dai = &ad73311_dai,
+ .ops = &bf5xx_ad73311_ops,
+};
+
+static struct snd_soc_machine bf5xx_ad73311 = {
+ .name = "bf5xx_ad73311",
+ .probe = bf5xx_probe,
+ .dai_link = &bf5xx_ad73311_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
+ .machine = &bf5xx_ad73311,
+ .platform = &bf5xx_i2s_soc_platform,
+ .codec_dev = &soc_codec_dev_ad73311,
+};
+
+static struct platform_device *bf52x_ad73311_snd_device;
+
+static int __init bf5xx_ad73311_init(void)
+{
+ int ret;
+
+ pr_debug("%s enter\n", __func__);
+ bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bf52x_ad73311_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata);
+ bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev;
+ ret = platform_device_add(bf52x_ad73311_snd_device);
+
+ if (ret)
+ platform_device_put(bf52x_ad73311_snd_device);
+
+ return ret;
+}
+
+static void __exit bf5xx_ad73311_exit(void)
+{
+ pr_debug("%s enter\n", __func__);
+ platform_device_unregister(bf52x_ad73311_snd_device);
+}
+
+module_init(bf5xx_ad73311_init);
+module_exit(bf5xx_ad73311_exit);
+
+/* Module information */
+MODULE_AUTHOR("Cliff Cai");
+MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
new file mode 100644
index 0000000..61fccf9
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -0,0 +1,288 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-i2s-pcm.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: DMA driver for i2s codec
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+
+#include "bf5xx-i2s-pcm.h"
+#include "bf5xx-i2s.h"
+#include "bf5xx-sport.h"
+
+static void bf5xx_dma_irq(void *data)
+{
+ struct snd_pcm_substream *pcm = data;
+ snd_pcm_period_elapsed(pcm);
+}
+
+static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 0x10000,
+ .periods_min = 1,
+ .periods_max = PAGE_SIZE/32,
+ .buffer_bytes_max = 0x20000, /* 128 kbytes */
+ .fifo_size = 16,
+};
+
+static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ size_t size = bf5xx_pcm_hardware.buffer_bytes_max;
+ snd_pcm_lib_malloc_pages(substream, size);
+
+ return 0;
+}
+
+static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+
+ return 0;
+}
+
+static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ int period_bytes = frames_to_bytes(runtime, runtime->period_size);
+
+ pr_debug("%s enter\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_tx_dma(sport, runtime->dma_area,
+ runtime->periods, period_bytes);
+ } else {
+ sport_set_rx_callback(sport, bf5xx_dma_irq, substream);
+ sport_config_rx_dma(sport, runtime->dma_area,
+ runtime->periods, period_bytes);
+ }
+
+ return 0;
+}
+
+static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ int ret = 0;
+
+ pr_debug("%s enter\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sport_tx_start(sport);
+ else
+ sport_rx_start(sport);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sport_tx_stop(sport);
+ else
+ sport_rx_stop(sport);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sport_device *sport = runtime->private_data;
+ unsigned int diff;
+ snd_pcm_uframes_t frames;
+ pr_debug("%s enter\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ diff = sport_curr_offset_tx(sport);
+ frames = bytes_to_frames(substream->runtime, diff);
+ } else {
+ diff = sport_curr_offset_rx(sport);
+ frames = bytes_to_frames(substream->runtime, diff);
+ }
+ return frames;
+}
+
+static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret;
+
+ pr_debug("%s enter\n", __func__);
+ snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime, \
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ if (sport_handle != NULL)
+ runtime->private_data = sport_handle;
+ else {
+ pr_err("sport_handle is NULL\n");
+ return -1;
+ }
+ return 0;
+
+ out:
+ return ret;
+}
+
+static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ size_t size = vma->vm_end - vma->vm_start;
+ vma->vm_start = (unsigned long)runtime->dma_area;
+ vma->vm_end = vma->vm_start + size;
+ vma->vm_flags |= VM_SHARED;
+
+ return 0 ;
+}
+
+struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
+ .open = bf5xx_pcm_open,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = bf5xx_pcm_hw_params,
+ .hw_free = bf5xx_pcm_hw_free,
+ .prepare = bf5xx_pcm_prepare,
+ .trigger = bf5xx_pcm_trigger,
+ .pointer = bf5xx_pcm_pointer,
+ .mmap = bf5xx_pcm_mmap,
+};
+
+static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = bf5xx_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area) {
+ pr_err("Failed to allocate dma memory \
+ Please increase uncached DMA memory region\n");
+ return -ENOMEM;
+ }
+ buf->bytes = size;
+
+ pr_debug("%s, area:%p, size:0x%08lx\n", __func__,
+ buf->area, buf->bytes);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sport_handle->tx_buf = buf->area;
+ else
+ sport_handle->rx_buf = buf->area;
+
+ return 0;
+}
+
+static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ dma_free_coherent(NULL, buf->bytes, buf->area, 0);
+ buf->area = NULL;
+ }
+ if (sport_handle)
+ sport_done(sport_handle);
+}
+
+static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK;
+
+int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ pr_debug("%s enter\n", __func__);
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &bf5xx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+ if (dai->playback.channels_min) {
+ ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = bf5xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+struct snd_soc_platform bf5xx_i2s_soc_platform = {
+ .name = "bf5xx-audio",
+ .pcm_ops = &bf5xx_pcm_i2s_ops,
+ .pcm_new = bf5xx_pcm_i2s_new,
+ .pcm_free = bf5xx_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
+
+MODULE_AUTHOR("Cliff Cai");
+MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h
new file mode 100644
index 0000000..4d4609a
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.h
@@ -0,0 +1,29 @@
+/*
+ * linux/sound/arm/bf5xx-i2s-pcm.h -- ALSA PCM interface for the Blackfin
+ *
+ * Copyright 2007 Analog Device Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _BF5XX_I2S_PCM_H
+#define _BF5XX_I2S_PCM_H
+
+struct bf5xx_pcm_dma_params {
+ char *name; /* stream identifier */
+};
+
+struct bf5xx_gpio {
+ u32 sys;
+ u32 rx;
+ u32 tx;
+ u32 clk;
+ u32 frm;
+};
+
+/* platform data */
+extern struct snd_soc_platform bf5xx_i2s_soc_platform;
+
+#endif
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
new file mode 100644
index 0000000..e020c16
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -0,0 +1,321 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-i2s.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: Blackfin I2S CPU DAI driver
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/irq.h>
+#include <asm/portmux.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+
+#include "bf5xx-sport.h"
+#include "bf5xx-i2s.h"
+
+struct bf5xx_i2s_port {
+ u16 tcr1;
+ u16 rcr1;
+ u16 tcr2;
+ u16 rcr2;
+ int counter;
+};
+
+static struct bf5xx_i2s_port bf5xx_i2s;
+static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
+
+static struct sport_param sport_params[2] = {
+ {
+ .dma_rx_chan = CH_SPORT0_RX,
+ .dma_tx_chan = CH_SPORT0_TX,
+ .err_irq = IRQ_SPORT0_ERROR,
+ .regs = (struct sport_register *)SPORT0_TCR1,
+ },
+ {
+ .dma_rx_chan = CH_SPORT1_RX,
+ .dma_tx_chan = CH_SPORT1_TX,
+ .err_irq = IRQ_SPORT1_ERROR,
+ .regs = (struct sport_register *)SPORT1_TCR1,
+ }
+};
+
+/*
+ * Setting the TFS pin selector for SPORT 0 based on whether the selected
+ * port id F or G. If the port is F then no conflict should exist for the
+ * TFS. When Port G is selected and EMAC then there is a conflict between
+ * the PHY interrupt line and TFS. Current settings prevent the conflict
+ * by ignoring the TFS pin when Port G is selected. This allows both
+ * ssm2602 using Port G and EMAC concurrently.
+ */
+#ifdef CONFIG_BF527_SPORT0_PORTF
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
+#else
+#define LOCAL_SPORT0_TFS (0)
+#endif
+
+static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
+ P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0},
+ {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI,
+ P_SPORT1_RSCLK, P_SPORT1_TFS, 0} };
+
+static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ int ret = 0;
+
+ /* interface format:support I2S,slave mode */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ bf5xx_i2s.tcr1 |= TFSR | TCKFE;
+ bf5xx_i2s.rcr1 |= RFSR | RCKFE;
+ bf5xx_i2s.tcr2 |= TSFSE;
+ bf5xx_i2s.rcr2 |= RSFSE;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ bf5xx_i2s.tcr1 |= TFSR;
+ bf5xx_i2s.rcr1 |= RFSR;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ret = -EINVAL;
+ break;
+ default:
+ printk(KERN_ERR "%s: Unknown DAI format type\n", __func__);
+ ret = -EINVAL;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ case SND_SOC_DAIFMT_CBS_CFM:
+ ret = -EINVAL;
+ break;
+ default:
+ printk(KERN_ERR "%s: Unknown DAI master type\n", __func__);
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s enter\n", __func__);
+
+ /*this counter is used for counting how many pcm streams are opened*/
+ bf5xx_i2s.counter++;
+ return 0;
+}
+
+static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ int ret = 0;
+
+ bf5xx_i2s.tcr2 &= ~0x1f;
+ bf5xx_i2s.rcr2 &= ~0x1f;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bf5xx_i2s.tcr2 |= 15;
+ bf5xx_i2s.rcr2 |= 15;
+ sport_handle->wdsize = 2;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bf5xx_i2s.tcr2 |= 23;
+ bf5xx_i2s.rcr2 |= 23;
+ sport_handle->wdsize = 3;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bf5xx_i2s.tcr2 |= 31;
+ bf5xx_i2s.rcr2 |= 31;
+ sport_handle->wdsize = 4;
+ break;
+ }
+
+ if (bf5xx_i2s.counter == 1) {
+ /*
+ * TX and RX are not independent,they are enabled at the
+ * same time, even if only one side is running. So, we
+ * need to configure both of them at the time when the first
+ * stream is opened.
+ *
+ * CPU DAI:slave mode.
+ */
+ ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
+ bf5xx_i2s.rcr2, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
+ bf5xx_i2s.tcr2, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+ }
+
+ return 0;
+}
+
+static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s enter\n", __func__);
+ bf5xx_i2s.counter--;
+}
+
+static int bf5xx_i2s_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ pr_debug("%s enter\n", __func__);
+ if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
+ pr_err("Requesting Peripherals failed\n");
+ return -EFAULT;
+ }
+
+ /* request DMA for SPORT */
+ sport_handle = sport_init(&sport_params[sport_num], 4, \
+ 2 * sizeof(u32), NULL);
+ if (!sport_handle) {
+ peripheral_free_list(&sport_req[sport_num][0]);
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+static void bf5xx_i2s_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ pr_debug("%s enter\n", __func__);
+ peripheral_free_list(&sport_req[sport_num][0]);
+}
+
+#ifdef CONFIG_PM
+static int bf5xx_i2s_suspend(struct platform_device *dev,
+ struct snd_soc_dai *dai)
+{
+ struct sport_device *sport =
+ (struct sport_device *)dai->private_data;
+
+ pr_debug("%s : sport %d\n", __func__, dai->id);
+ if (!dai->active)
+ return 0;
+ if (dai->capture.active)
+ sport_rx_stop(sport);
+ if (dai->playback.active)
+ sport_tx_stop(sport);
+ return 0;
+}
+
+static int bf5xx_i2s_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ int ret;
+ struct sport_device *sport =
+ (struct sport_device *)dai->private_data;
+
+ pr_debug("%s : sport %d\n", __func__, dai->id);
+ if (!dai->active)
+ return 0;
+
+ ret = sport_config_rx(sport_handle, RFSR | RCKFE, RSFSE|0x1f, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ ret = sport_config_tx(sport_handle, TFSR | TCKFE, TSFSE|0x1f, 0, 0);
+ if (ret) {
+ pr_err("SPORT is busy!\n");
+ return -EBUSY;
+ }
+
+ if (dai->capture.active)
+ sport_rx_start(sport);
+ if (dai->playback.active)
+ sport_tx_start(sport);
+ return 0;
+}
+
+#else
+#define bf5xx_i2s_suspend NULL
+#define bf5xx_i2s_resume NULL
+#endif
+
+#define BF5XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_96000)
+
+#define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai bf5xx_i2s_dai = {
+ .name = "bf5xx-i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .probe = bf5xx_i2s_probe,
+ .remove = bf5xx_i2s_remove,
+ .suspend = bf5xx_i2s_suspend,
+ .resume = bf5xx_i2s_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = BF5XX_I2S_RATES,
+ .formats = BF5XX_I2S_FORMATS,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = BF5XX_I2S_RATES,
+ .formats = BF5XX_I2S_FORMATS,},
+ .ops = {
+ .startup = bf5xx_i2s_startup,
+ .shutdown = bf5xx_i2s_shutdown,
+ .hw_params = bf5xx_i2s_hw_params,},
+ .dai_ops = {
+ .set_fmt = bf5xx_i2s_set_dai_fmt,
+ },
+};
+EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
+
+/* Module information */
+MODULE_AUTHOR("Cliff Cai");
+MODULE_DESCRIPTION("I2S driver for ADI Blackfin");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h
new file mode 100644
index 0000000..7107d1a
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-i2s.h
@@ -0,0 +1,14 @@
+/*
+ * linux/sound/arm/bf5xx-i2s.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _BF5XX_I2S_H
+#define _BF5XX_I2S_H
+
+extern struct snd_soc_dai bf5xx_i2s_dai;
+
+#endif
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
new file mode 100644
index 0000000..3b99e48
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -0,0 +1,1032 @@
+/*
+ * File: bf5xx_sport.c
+ * Based on:
+ * Author: Roy Huang <roy.huang@analog.com>
+ *
+ * Created: Tue Sep 21 10:52:42 CEST 2004
+ * Description:
+ * Blackfin SPORT Driver
+ *
+ * Copyright 2004-2007 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/kernel.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/dma-mapping.h>
+#include <linux/gpio.h>
+#include <linux/bug.h>
+#include <asm/portmux.h>
+#include <asm/dma.h>
+#include <asm/blackfin.h>
+#include <asm/cacheflush.h>
+
+#include "bf5xx-sport.h"
+/* delay between frame sync pulse and first data bit in multichannel mode */
+#define FRAME_DELAY (1<<12)
+
+struct sport_device *sport_handle;
+EXPORT_SYMBOL(sport_handle);
+/* note: multichannel is in units of 8 channels,
+ * tdm_count is # channels NOT / 8 ! */
+int sport_set_multichannel(struct sport_device *sport,
+ int tdm_count, u32 mask, int packed)
+{
+ pr_debug("%s tdm_count=%d mask:0x%08x packed=%d\n", __func__,
+ tdm_count, mask, packed);
+
+ if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN))
+ return -EBUSY;
+
+ if (tdm_count & 0x7)
+ return -EINVAL;
+
+ if (tdm_count > 32)
+ return -EINVAL; /* Only support less than 32 channels now */
+
+ if (tdm_count) {
+ sport->regs->mcmc1 = ((tdm_count>>3)-1) << 12;
+ sport->regs->mcmc2 = FRAME_DELAY | MCMEN | \
+ (packed ? (MCDTXPE|MCDRXPE) : 0);
+
+ sport->regs->mtcs0 = mask;
+ sport->regs->mrcs0 = mask;
+ sport->regs->mtcs1 = 0;
+ sport->regs->mrcs1 = 0;
+ sport->regs->mtcs2 = 0;
+ sport->regs->mrcs2 = 0;
+ sport->regs->mtcs3 = 0;
+ sport->regs->mrcs3 = 0;
+ } else {
+ sport->regs->mcmc1 = 0;
+ sport->regs->mcmc2 = 0;
+
+ sport->regs->mtcs0 = 0;
+ sport->regs->mrcs0 = 0;
+ }
+
+ sport->regs->mtcs1 = 0; sport->regs->mtcs2 = 0; sport->regs->mtcs3 = 0;
+ sport->regs->mrcs1 = 0; sport->regs->mrcs2 = 0; sport->regs->mrcs3 = 0;
+
+ SSYNC();
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_set_multichannel);
+
+int sport_config_rx(struct sport_device *sport, unsigned int rcr1,
+ unsigned int rcr2, unsigned int clkdiv, unsigned int fsdiv)
+{
+ if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN))
+ return -EBUSY;
+
+ sport->regs->rcr1 = rcr1;
+ sport->regs->rcr2 = rcr2;
+ sport->regs->rclkdiv = clkdiv;
+ sport->regs->rfsdiv = fsdiv;
+
+ SSYNC();
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_config_rx);
+
+int sport_config_tx(struct sport_device *sport, unsigned int tcr1,
+ unsigned int tcr2, unsigned int clkdiv, unsigned int fsdiv)
+{
+ if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN))
+ return -EBUSY;
+
+ sport->regs->tcr1 = tcr1;
+ sport->regs->tcr2 = tcr2;
+ sport->regs->tclkdiv = clkdiv;
+ sport->regs->tfsdiv = fsdiv;
+
+ SSYNC();
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_config_tx);
+
+static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
+ size_t fragsize, unsigned int cfg,
+ unsigned int x_count, unsigned int ycount, size_t wdsize)
+{
+
+ int i;
+
+ for (i = 0; i < fragcount; ++i) {
+ desc[i].next_desc_addr = (unsigned long)&(desc[i + 1]);
+ desc[i].start_addr = (unsigned long)buf + i*fragsize;
+ desc[i].cfg = cfg;
+ desc[i].x_count = x_count;
+ desc[i].x_modify = wdsize;
+ desc[i].y_count = ycount;
+ desc[i].y_modify = wdsize;
+ }
+
+ /* make circular */
+ desc[fragcount-1].next_desc_addr = (unsigned long)desc;
+
+ pr_debug("setup desc: desc0=%p, next0=%lx, desc1=%p,"
+ "next1=%lx\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n",
+ &(desc[0]), desc[0].next_desc_addr,
+ &(desc[1]), desc[1].next_desc_addr,
+ desc[0].x_count, desc[0].y_count,
+ desc[0].start_addr, desc[0].cfg);
+}
+
+static int sport_start(struct sport_device *sport)
+{
+ enable_dma(sport->dma_rx_chan);
+ enable_dma(sport->dma_tx_chan);
+ sport->regs->rcr1 |= RSPEN;
+ sport->regs->tcr1 |= TSPEN;
+ SSYNC();
+
+ return 0;
+}
+
+static int sport_stop(struct sport_device *sport)
+{
+ sport->regs->tcr1 &= ~TSPEN;
+ sport->regs->rcr1 &= ~RSPEN;
+ SSYNC();
+
+ disable_dma(sport->dma_rx_chan);
+ disable_dma(sport->dma_tx_chan);
+ return 0;
+}
+
+static inline int sport_hook_rx_dummy(struct sport_device *sport)
+{
+ struct dmasg *desc, temp_desc;
+ unsigned long flags;
+
+ BUG_ON(sport->dummy_rx_desc == NULL);
+ BUG_ON(sport->curr_rx_desc == sport->dummy_rx_desc);
+
+ /* Maybe the dummy buffer descriptor ring is damaged */
+ sport->dummy_rx_desc->next_desc_addr = \
+ (unsigned long)(sport->dummy_rx_desc+1);
+
+ local_irq_save(flags);
+ desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_rx_chan);
+ /* Copy the descriptor which will be damaged to backup */
+ temp_desc = *desc;
+ desc->x_count = 0xa;
+ desc->y_count = 0;
+ desc->next_desc_addr = (unsigned long)(sport->dummy_rx_desc);
+ local_irq_restore(flags);
+ /* Waiting for dummy buffer descriptor is already hooked*/
+ while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) -
+ sizeof(struct dmasg)) !=
+ (unsigned long)sport->dummy_rx_desc)
+ ;
+ sport->curr_rx_desc = sport->dummy_rx_desc;
+ /* Restore the damaged descriptor */
+ *desc = temp_desc;
+
+ return 0;
+}
+
+static inline int sport_rx_dma_start(struct sport_device *sport, int dummy)
+{
+ if (dummy) {
+ sport->dummy_rx_desc->next_desc_addr = \
+ (unsigned long) sport->dummy_rx_desc;
+ sport->curr_rx_desc = sport->dummy_rx_desc;
+ } else
+ sport->curr_rx_desc = sport->dma_rx_desc;
+
+ set_dma_next_desc_addr(sport->dma_rx_chan, \
+ (unsigned long)(sport->curr_rx_desc));
+ set_dma_x_count(sport->dma_rx_chan, 0);
+ set_dma_x_modify(sport->dma_rx_chan, 0);
+ set_dma_config(sport->dma_rx_chan, (DMAFLOW_LARGE | NDSIZE_9 | \
+ WDSIZE_32 | WNR));
+ set_dma_curr_addr(sport->dma_rx_chan, sport->curr_rx_desc->start_addr);
+ SSYNC();
+
+ return 0;
+}
+
+static inline int sport_tx_dma_start(struct sport_device *sport, int dummy)
+{
+ if (dummy) {
+ sport->dummy_tx_desc->next_desc_addr = \
+ (unsigned long) sport->dummy_tx_desc;
+ sport->curr_tx_desc = sport->dummy_tx_desc;
+ } else
+ sport->curr_tx_desc = sport->dma_tx_desc;
+
+ set_dma_next_desc_addr(sport->dma_tx_chan, \
+ (unsigned long)(sport->curr_tx_desc));
+ set_dma_x_count(sport->dma_tx_chan, 0);
+ set_dma_x_modify(sport->dma_tx_chan, 0);
+ set_dma_config(sport->dma_tx_chan,
+ (DMAFLOW_LARGE | NDSIZE_9 | WDSIZE_32));
+ set_dma_curr_addr(sport->dma_tx_chan, sport->curr_tx_desc->start_addr);
+ SSYNC();
+
+ return 0;
+}
+
+int sport_rx_start(struct sport_device *sport)
+{
+ unsigned long flags;
+ pr_debug("%s enter\n", __func__);
+ if (sport->rx_run)
+ return -EBUSY;
+ if (sport->tx_run) {
+ /* tx is running, rx is not running */
+ BUG_ON(sport->dma_rx_desc == NULL);
+ BUG_ON(sport->curr_rx_desc != sport->dummy_rx_desc);
+ local_irq_save(flags);
+ while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) -
+ sizeof(struct dmasg)) !=
+ (unsigned long)sport->dummy_rx_desc)
+ ;
+ sport->dummy_rx_desc->next_desc_addr =
+ (unsigned long)(sport->dma_rx_desc);
+ local_irq_restore(flags);
+ sport->curr_rx_desc = sport->dma_rx_desc;
+ } else {
+ sport_tx_dma_start(sport, 1);
+ sport_rx_dma_start(sport, 0);
+ sport_start(sport);
+ }
+
+ sport->rx_run = 1;
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_rx_start);
+
+int sport_rx_stop(struct sport_device *sport)
+{
+ pr_debug("%s enter\n", __func__);
+
+ if (!sport->rx_run)
+ return 0;
+ if (sport->tx_run) {
+ /* TX dma is still running, hook the dummy buffer */
+ sport_hook_rx_dummy(sport);
+ } else {
+ /* Both rx and tx dma will be stopped */
+ sport_stop(sport);
+ sport->curr_rx_desc = NULL;
+ sport->curr_tx_desc = NULL;
+ }
+
+ sport->rx_run = 0;
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_rx_stop);
+
+static inline int sport_hook_tx_dummy(struct sport_device *sport)
+{
+ struct dmasg *desc, temp_desc;
+ unsigned long flags;
+
+ BUG_ON(sport->dummy_tx_desc == NULL);
+ BUG_ON(sport->curr_tx_desc == sport->dummy_tx_desc);
+
+ sport->dummy_tx_desc->next_desc_addr = \
+ (unsigned long)(sport->dummy_tx_desc+1);
+
+ /* Shorten the time on last normal descriptor */
+ local_irq_save(flags);
+ desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_tx_chan);
+ /* Store the descriptor which will be damaged */
+ temp_desc = *desc;
+ desc->x_count = 0xa;
+ desc->y_count = 0;
+ desc->next_desc_addr = (unsigned long)(sport->dummy_tx_desc);
+ local_irq_restore(flags);
+ /* Waiting for dummy buffer descriptor is already hooked*/
+ while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - \
+ sizeof(struct dmasg)) != \
+ (unsigned long)sport->dummy_tx_desc)
+ ;
+ sport->curr_tx_desc = sport->dummy_tx_desc;
+ /* Restore the damaged descriptor */
+ *desc = temp_desc;
+
+ return 0;
+}
+
+int sport_tx_start(struct sport_device *sport)
+{
+ unsigned flags;
+ pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__,
+ sport->tx_run, sport->rx_run);
+ if (sport->tx_run)
+ return -EBUSY;
+ if (sport->rx_run) {
+ BUG_ON(sport->dma_tx_desc == NULL);
+ BUG_ON(sport->curr_tx_desc != sport->dummy_tx_desc);
+ /* Hook the normal buffer descriptor */
+ local_irq_save(flags);
+ while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) -
+ sizeof(struct dmasg)) !=
+ (unsigned long)sport->dummy_tx_desc)
+ ;
+ sport->dummy_tx_desc->next_desc_addr =
+ (unsigned long)(sport->dma_tx_desc);
+ local_irq_restore(flags);
+ sport->curr_tx_desc = sport->dma_tx_desc;
+ } else {
+
+ sport_tx_dma_start(sport, 0);
+ /* Let rx dma run the dummy buffer */
+ sport_rx_dma_start(sport, 1);
+ sport_start(sport);
+ }
+ sport->tx_run = 1;
+ return 0;
+}
+EXPORT_SYMBOL(sport_tx_start);
+
+int sport_tx_stop(struct sport_device *sport)
+{
+ if (!sport->tx_run)
+ return 0;
+ if (sport->rx_run) {
+ /* RX is still running, hook the dummy buffer */
+ sport_hook_tx_dummy(sport);
+ } else {
+ /* Both rx and tx dma stopped */
+ sport_stop(sport);
+ sport->curr_rx_desc = NULL;
+ sport->curr_tx_desc = NULL;
+ }
+
+ sport->tx_run = 0;
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_tx_stop);
+
+static inline int compute_wdsize(size_t wdsize)
+{
+ switch (wdsize) {
+ case 1:
+ return WDSIZE_8;
+ case 2:
+ return WDSIZE_16;
+ case 4:
+ default:
+ return WDSIZE_32;
+ }
+}
+
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+ int fragcount, size_t fragsize)
+{
+ unsigned int x_count;
+ unsigned int y_count;
+ unsigned int cfg;
+ dma_addr_t addr;
+
+ pr_debug("%s buf:%p, frag:%d, fragsize:0x%lx\n", __func__, \
+ buf, fragcount, fragsize);
+
+ x_count = fragsize / sport->wdsize;
+ y_count = 0;
+
+ /* for fragments larger than 64k words we use 2d dma,
+ * denote fragecount as two numbers' mutliply and both of them
+ * are less than 64k.*/
+ if (x_count >= 0x10000) {
+ int i, count = x_count;
+
+ for (i = 16; i > 0; i--) {
+ x_count = 1 << i;
+ if ((count & (x_count - 1)) == 0) {
+ y_count = count >> i;
+ if (y_count < 0x10000)
+ break;
+ }
+ }
+ if (i == 0)
+ return -EINVAL;
+ }
+ pr_debug("%s(x_count:0x%x, y_count:0x%x)\n", __func__,
+ x_count, y_count);
+
+ if (sport->dma_rx_desc)
+ dma_free_coherent(NULL, sport->rx_desc_bytes,
+ sport->dma_rx_desc, 0);
+
+ /* Allocate a new descritor ring as current one. */
+ sport->dma_rx_desc = dma_alloc_coherent(NULL, \
+ fragcount * sizeof(struct dmasg), &addr, 0);
+ sport->rx_desc_bytes = fragcount * sizeof(struct dmasg);
+
+ if (!sport->dma_rx_desc) {
+ pr_err("Failed to allocate memory for rx desc\n");
+ return -ENOMEM;
+ }
+
+ sport->rx_buf = buf;
+ sport->rx_fragsize = fragsize;
+ sport->rx_frags = fragcount;
+
+ cfg = 0x7000 | DI_EN | compute_wdsize(sport->wdsize) | WNR | \
+ (DESC_ELEMENT_COUNT << 8); /* large descriptor mode */
+
+ if (y_count != 0)
+ cfg |= DMA2D;
+
+ setup_desc(sport->dma_rx_desc, buf, fragcount, fragsize,
+ cfg|DMAEN, x_count, y_count, sport->wdsize);
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_config_rx_dma);
+
+int sport_config_tx_dma(struct sport_device *sport, void *buf, \
+ int fragcount, size_t fragsize)
+{
+ unsigned int x_count;
+ unsigned int y_count;
+ unsigned int cfg;
+ dma_addr_t addr;
+
+ pr_debug("%s buf:%p, fragcount:%d, fragsize:0x%lx\n",
+ __func__, buf, fragcount, fragsize);
+
+ x_count = fragsize/sport->wdsize;
+ y_count = 0;
+
+ /* for fragments larger than 64k words we use 2d dma,
+ * denote fragecount as two numbers' mutliply and both of them
+ * are less than 64k.*/
+ if (x_count >= 0x10000) {
+ int i, count = x_count;
+
+ for (i = 16; i > 0; i--) {
+ x_count = 1 << i;
+ if ((count & (x_count - 1)) == 0) {
+ y_count = count >> i;
+ if (y_count < 0x10000)
+ break;
+ }
+ }
+ if (i == 0)
+ return -EINVAL;
+ }
+ pr_debug("%s x_count:0x%x, y_count:0x%x\n", __func__,
+ x_count, y_count);
+
+
+ if (sport->dma_tx_desc) {
+ dma_free_coherent(NULL, sport->tx_desc_bytes, \
+ sport->dma_tx_desc, 0);
+ }
+
+ sport->dma_tx_desc = dma_alloc_coherent(NULL, \
+ fragcount * sizeof(struct dmasg), &addr, 0);
+ sport->tx_desc_bytes = fragcount * sizeof(struct dmasg);
+ if (!sport->dma_tx_desc) {
+ pr_err("Failed to allocate memory for tx desc\n");
+ return -ENOMEM;
+ }
+
+ sport->tx_buf = buf;
+ sport->tx_fragsize = fragsize;
+ sport->tx_frags = fragcount;
+ cfg = 0x7000 | DI_EN | compute_wdsize(sport->wdsize) | \
+ (DESC_ELEMENT_COUNT << 8); /* large descriptor mode */
+
+ if (y_count != 0)
+ cfg |= DMA2D;
+
+ setup_desc(sport->dma_tx_desc, buf, fragcount, fragsize,
+ cfg|DMAEN, x_count, y_count, sport->wdsize);
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_config_tx_dma);
+
+/* setup dummy dma descriptor ring, which don't generate interrupts,
+ * the x_modify is set to 0 */
+static int sport_config_rx_dummy(struct sport_device *sport)
+{
+ struct dmasg *desc;
+ unsigned config;
+
+ pr_debug("%s entered\n", __func__);
+#if L1_DATA_A_LENGTH != 0
+ desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc));
+#else
+ {
+ dma_addr_t addr;
+ desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0);
+ }
+#endif
+ if (desc == NULL) {
+ pr_err("Failed to allocate memory for dummy rx desc\n");
+ return -ENOMEM;
+ }
+ memset(desc, 0, 2 * sizeof(*desc));
+ sport->dummy_rx_desc = desc;
+ desc->start_addr = (unsigned long)sport->dummy_buf;
+ config = DMAFLOW_LARGE | NDSIZE_9 | compute_wdsize(sport->wdsize)
+ | WNR | DMAEN;
+ desc->cfg = config;
+ desc->x_count = sport->dummy_count/sport->wdsize;
+ desc->x_modify = sport->wdsize;
+ desc->y_count = 0;
+ desc->y_modify = 0;
+ memcpy(desc+1, desc, sizeof(*desc));
+ desc->next_desc_addr = (unsigned long)(desc+1);
+ desc[1].next_desc_addr = (unsigned long)desc;
+ return 0;
+}
+
+static int sport_config_tx_dummy(struct sport_device *sport)
+{
+ struct dmasg *desc;
+ unsigned int config;
+
+ pr_debug("%s entered\n", __func__);
+
+#if L1_DATA_A_LENGTH != 0
+ desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc));
+#else
+ {
+ dma_addr_t addr;
+ desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0);
+ }
+#endif
+ if (!desc) {
+ pr_err("Failed to allocate memory for dummy tx desc\n");
+ return -ENOMEM;
+ }
+ memset(desc, 0, 2 * sizeof(*desc));
+ sport->dummy_tx_desc = desc;
+ desc->start_addr = (unsigned long)sport->dummy_buf + \
+ sport->dummy_count;
+ config = DMAFLOW_LARGE | NDSIZE_9 |
+ compute_wdsize(sport->wdsize) | DMAEN;
+ desc->cfg = config;
+ desc->x_count = sport->dummy_count/sport->wdsize;
+ desc->x_modify = sport->wdsize;
+ desc->y_count = 0;
+ desc->y_modify = 0;
+ memcpy(desc+1, desc, sizeof(*desc));
+ desc->next_desc_addr = (unsigned long)(desc+1);
+ desc[1].next_desc_addr = (unsigned long)desc;
+ return 0;
+}
+
+unsigned long sport_curr_offset_rx(struct sport_device *sport)
+{
+ unsigned long curr = get_dma_curr_addr(sport->dma_rx_chan);
+
+ return (unsigned char *)curr - sport->rx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_rx);
+
+unsigned long sport_curr_offset_tx(struct sport_device *sport)
+{
+ unsigned long curr = get_dma_curr_addr(sport->dma_tx_chan);
+
+ return (unsigned char *)curr - sport->tx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_tx);
+
+void sport_incfrag(struct sport_device *sport, int *frag, int tx)
+{
+ ++(*frag);
+ if (tx == 1 && *frag == sport->tx_frags)
+ *frag = 0;
+
+ if (tx == 0 && *frag == sport->rx_frags)
+ *frag = 0;
+}
+EXPORT_SYMBOL(sport_incfrag);
+
+void sport_decfrag(struct sport_device *sport, int *frag, int tx)
+{
+ --(*frag);
+ if (tx == 1 && *frag == 0)
+ *frag = sport->tx_frags;
+
+ if (tx == 0 && *frag == 0)
+ *frag = sport->rx_frags;
+}
+EXPORT_SYMBOL(sport_decfrag);
+
+static int sport_check_status(struct sport_device *sport,
+ unsigned int *sport_stat,
+ unsigned int *rx_stat,
+ unsigned int *tx_stat)
+{
+ int status = 0;
+
+ if (sport_stat) {
+ SSYNC();
+ status = sport->regs->stat;
+ if (status & (TOVF|TUVF|ROVF|RUVF))
+ sport->regs->stat = (status & (TOVF|TUVF|ROVF|RUVF));
+ SSYNC();
+ *sport_stat = status;
+ }
+
+ if (rx_stat) {
+ SSYNC();
+ status = get_dma_curr_irqstat(sport->dma_rx_chan);
+ if (status & (DMA_DONE|DMA_ERR))
+ clear_dma_irqstat(sport->dma_rx_chan);
+ SSYNC();
+ *rx_stat = status;
+ }
+
+ if (tx_stat) {
+ SSYNC();
+ status = get_dma_curr_irqstat(sport->dma_tx_chan);
+ if (status & (DMA_DONE|DMA_ERR))
+ clear_dma_irqstat(sport->dma_tx_chan);
+ SSYNC();
+ *tx_stat = status;
+ }
+
+ return 0;
+}
+
+int sport_dump_stat(struct sport_device *sport, char *buf, size_t len)
+{
+ int ret;
+
+ ret = snprintf(buf, len,
+ "sts: 0x%04x\n"
+ "rx dma %d sts: 0x%04x tx dma %d sts: 0x%04x\n",
+ sport->regs->stat,
+ sport->dma_rx_chan,
+ get_dma_curr_irqstat(sport->dma_rx_chan),
+ sport->dma_tx_chan,
+ get_dma_curr_irqstat(sport->dma_tx_chan));
+ buf += ret;
+ len -= ret;
+
+ ret += snprintf(buf, len,
+ "curr_rx_desc:0x%p, curr_tx_desc:0x%p\n"
+ "dma_rx_desc:0x%p, dma_tx_desc:0x%p\n"
+ "dummy_rx_desc:0x%p, dummy_tx_desc:0x%p\n",
+ sport->curr_rx_desc, sport->curr_tx_desc,
+ sport->dma_rx_desc, sport->dma_tx_desc,
+ sport->dummy_rx_desc, sport->dummy_tx_desc);
+
+ return ret;
+}
+
+static irqreturn_t rx_handler(int irq, void *dev_id)
+{
+ unsigned int rx_stat;
+ struct sport_device *sport = dev_id;
+
+ pr_debug("%s enter\n", __func__);
+ sport_check_status(sport, NULL, &rx_stat, NULL);
+ if (!(rx_stat & DMA_DONE))
+ pr_err("rx dma is already stopped\n");
+
+ if (sport->rx_callback) {
+ sport->rx_callback(sport->rx_data);
+ return IRQ_HANDLED;
+ }
+
+ return IRQ_NONE;
+}
+
+static irqreturn_t tx_handler(int irq, void *dev_id)
+{
+ unsigned int tx_stat;
+ struct sport_device *sport = dev_id;
+ pr_debug("%s enter\n", __func__);
+ sport_check_status(sport, NULL, NULL, &tx_stat);
+ if (!(tx_stat & DMA_DONE)) {
+ pr_err("tx dma is already stopped\n");
+ return IRQ_HANDLED;
+ }
+ if (sport->tx_callback) {
+ sport->tx_callback(sport->tx_data);
+ return IRQ_HANDLED;
+ }
+
+ return IRQ_NONE;
+}
+
+static irqreturn_t err_handler(int irq, void *dev_id)
+{
+ unsigned int status = 0;
+ struct sport_device *sport = dev_id;
+
+ pr_debug("%s\n", __func__);
+ if (sport_check_status(sport, &status, NULL, NULL)) {
+ pr_err("error checking status ??");
+ return IRQ_NONE;
+ }
+
+ if (status & (TOVF|TUVF|ROVF|RUVF)) {
+ pr_info("sport status error:%s%s%s%s\n",
+ status & TOVF ? " TOVF" : "",
+ status & TUVF ? " TUVF" : "",
+ status & ROVF ? " ROVF" : "",
+ status & RUVF ? " RUVF" : "");
+ if (status & TOVF || status & TUVF) {
+ disable_dma(sport->dma_tx_chan);
+ if (sport->tx_run)
+ sport_tx_dma_start(sport, 0);
+ else
+ sport_tx_dma_start(sport, 1);
+ enable_dma(sport->dma_tx_chan);
+ } else {
+ disable_dma(sport->dma_rx_chan);
+ if (sport->rx_run)
+ sport_rx_dma_start(sport, 0);
+ else
+ sport_rx_dma_start(sport, 1);
+ enable_dma(sport->dma_rx_chan);
+ }
+ }
+ status = sport->regs->stat;
+ if (status & (TOVF|TUVF|ROVF|RUVF))
+ sport->regs->stat = (status & (TOVF|TUVF|ROVF|RUVF));
+ SSYNC();
+
+ if (sport->err_callback)
+ sport->err_callback(sport->err_data);
+
+ return IRQ_HANDLED;
+}
+
+int sport_set_rx_callback(struct sport_device *sport,
+ void (*rx_callback)(void *), void *rx_data)
+{
+ BUG_ON(rx_callback == NULL);
+ sport->rx_callback = rx_callback;
+ sport->rx_data = rx_data;
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_set_rx_callback);
+
+int sport_set_tx_callback(struct sport_device *sport,
+ void (*tx_callback)(void *), void *tx_data)
+{
+ BUG_ON(tx_callback == NULL);
+ sport->tx_callback = tx_callback;
+ sport->tx_data = tx_data;
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_set_tx_callback);
+
+int sport_set_err_callback(struct sport_device *sport,
+ void (*err_callback)(void *), void *err_data)
+{
+ BUG_ON(err_callback == NULL);
+ sport->err_callback = err_callback;
+ sport->err_data = err_data;
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_set_err_callback);
+
+struct sport_device *sport_init(struct sport_param *param, unsigned wdsize,
+ unsigned dummy_count, void *private_data)
+{
+ int ret;
+ struct sport_device *sport;
+ pr_debug("%s enter\n", __func__);
+ BUG_ON(param == NULL);
+ BUG_ON(wdsize == 0 || dummy_count == 0);
+ sport = kmalloc(sizeof(struct sport_device), GFP_KERNEL);
+ if (!sport) {
+ pr_err("Failed to allocate for sport device\n");
+ return NULL;
+ }
+
+ memset(sport, 0, sizeof(struct sport_device));
+ sport->dma_rx_chan = param->dma_rx_chan;
+ sport->dma_tx_chan = param->dma_tx_chan;
+ sport->err_irq = param->err_irq;
+ sport->regs = param->regs;
+ sport->private_data = private_data;
+
+ if (request_dma(sport->dma_rx_chan, "SPORT RX Data") == -EBUSY) {
+ pr_err("Failed to request RX dma %d\n", \
+ sport->dma_rx_chan);
+ goto __init_err1;
+ }
+ if (set_dma_callback(sport->dma_rx_chan, rx_handler, sport) != 0) {
+ pr_err("Failed to request RX irq %d\n", \
+ sport->dma_rx_chan);
+ goto __init_err2;
+ }
+
+ if (request_dma(sport->dma_tx_chan, "SPORT TX Data") == -EBUSY) {
+ pr_err("Failed to request TX dma %d\n", \
+ sport->dma_tx_chan);
+ goto __init_err2;
+ }
+
+ if (set_dma_callback(sport->dma_tx_chan, tx_handler, sport) != 0) {
+ pr_err("Failed to request TX irq %d\n", \
+ sport->dma_tx_chan);
+ goto __init_err3;
+ }
+
+ if (request_irq(sport->err_irq, err_handler, IRQF_SHARED, "SPORT err",
+ sport) < 0) {
+ pr_err("Failed to request err irq:%d\n", \
+ sport->err_irq);
+ goto __init_err3;
+ }
+
+ pr_err("dma rx:%d tx:%d, err irq:%d, regs:%p\n",
+ sport->dma_rx_chan, sport->dma_tx_chan,
+ sport->err_irq, sport->regs);
+
+ sport->wdsize = wdsize;
+ sport->dummy_count = dummy_count;
+
+#if L1_DATA_A_LENGTH != 0
+ sport->dummy_buf = l1_data_sram_alloc(dummy_count * 2);
+#else
+ sport->dummy_buf = kmalloc(dummy_count * 2, GFP_KERNEL);
+#endif
+ if (sport->dummy_buf == NULL) {
+ pr_err("Failed to allocate dummy buffer\n");
+ goto __error;
+ }
+
+ memset(sport->dummy_buf, 0, dummy_count * 2);
+ ret = sport_config_rx_dummy(sport);
+ if (ret) {
+ pr_err("Failed to config rx dummy ring\n");
+ goto __error;
+ }
+ ret = sport_config_tx_dummy(sport);
+ if (ret) {
+ pr_err("Failed to config tx dummy ring\n");
+ goto __error;
+ }
+
+ return sport;
+__error:
+ free_irq(sport->err_irq, sport);
+__init_err3:
+ free_dma(sport->dma_tx_chan);
+__init_err2:
+ free_dma(sport->dma_rx_chan);
+__init_err1:
+ kfree(sport);
+ return NULL;
+}
+EXPORT_SYMBOL(sport_init);
+
+void sport_done(struct sport_device *sport)
+{
+ if (sport == NULL)
+ return;
+
+ sport_stop(sport);
+ if (sport->dma_rx_desc)
+ dma_free_coherent(NULL, sport->rx_desc_bytes,
+ sport->dma_rx_desc, 0);
+ if (sport->dma_tx_desc)
+ dma_free_coherent(NULL, sport->tx_desc_bytes,
+ sport->dma_tx_desc, 0);
+
+#if L1_DATA_A_LENGTH != 0
+ l1_data_sram_free(sport->dummy_rx_desc);
+ l1_data_sram_free(sport->dummy_tx_desc);
+ l1_data_sram_free(sport->dummy_buf);
+#else
+ dma_free_coherent(NULL, 2*sizeof(struct dmasg),
+ sport->dummy_rx_desc, 0);
+ dma_free_coherent(NULL, 2*sizeof(struct dmasg),
+ sport->dummy_tx_desc, 0);
+ kfree(sport->dummy_buf);
+#endif
+ free_dma(sport->dma_rx_chan);
+ free_dma(sport->dma_tx_chan);
+ free_irq(sport->err_irq, sport);
+
+ kfree(sport);
+ sport = NULL;
+}
+EXPORT_SYMBOL(sport_done);
+/*
+* It is only used to send several bytes when dma is not enabled
+ * sport controller is configured but not enabled.
+ * Multichannel cannot works with pio mode */
+/* Used by ac97 to write and read codec register */
+int sport_send_and_recv(struct sport_device *sport, u8 *out_data, \
+ u8 *in_data, int len)
+{
+ unsigned short dma_config;
+ unsigned short status;
+ unsigned long flags;
+ unsigned long wait = 0;
+
+ pr_debug("%s enter, out_data:%p, in_data:%p len:%d\n", \
+ __func__, out_data, in_data, len);
+ pr_debug("tcr1:0x%04x, tcr2:0x%04x, tclkdiv:0x%04x, tfsdiv:0x%04x\n"
+ "mcmc1:0x%04x, mcmc2:0x%04x\n",
+ sport->regs->tcr1, sport->regs->tcr2,
+ sport->regs->tclkdiv, sport->regs->tfsdiv,
+ sport->regs->mcmc1, sport->regs->mcmc2);
+ flush_dcache_range((unsigned)out_data, (unsigned)(out_data + len));
+
+ /* Enable tx dma */
+ dma_config = (RESTART | WDSIZE_16 | DI_EN);
+ set_dma_start_addr(sport->dma_tx_chan, (unsigned long)out_data);
+ set_dma_x_count(sport->dma_tx_chan, len/2);
+ set_dma_x_modify(sport->dma_tx_chan, 2);
+ set_dma_config(sport->dma_tx_chan, dma_config);
+ enable_dma(sport->dma_tx_chan);
+
+ if (in_data != NULL) {
+ invalidate_dcache_range((unsigned)in_data, \
+ (unsigned)(in_data + len));
+ /* Enable rx dma */
+ dma_config = (RESTART | WDSIZE_16 | WNR | DI_EN);
+ set_dma_start_addr(sport->dma_rx_chan, (unsigned long)in_data);
+ set_dma_x_count(sport->dma_rx_chan, len/2);
+ set_dma_x_modify(sport->dma_rx_chan, 2);
+ set_dma_config(sport->dma_rx_chan, dma_config);
+ enable_dma(sport->dma_rx_chan);
+ }
+
+ local_irq_save(flags);
+ sport->regs->tcr1 |= TSPEN;
+ sport->regs->rcr1 |= RSPEN;
+ SSYNC();
+
+ status = get_dma_curr_irqstat(sport->dma_tx_chan);
+ while (status & DMA_RUN) {
+ udelay(1);
+ status = get_dma_curr_irqstat(sport->dma_tx_chan);
+ pr_debug("DMA status:0x%04x\n", status);
+ if (wait++ > 100)
+ goto __over;
+ }
+ status = sport->regs->stat;
+ wait = 0;
+
+ while (!(status & TXHRE)) {
+ pr_debug("sport status:0x%04x\n", status);
+ udelay(1);
+ status = *(unsigned short *)&sport->regs->stat;
+ if (wait++ > 1000)
+ goto __over;
+ }
+ /* Wait for the last byte sent out */
+ udelay(20);
+ pr_debug("sport status:0x%04x\n", status);
+
+__over:
+ sport->regs->tcr1 &= ~TSPEN;
+ sport->regs->rcr1 &= ~RSPEN;
+ SSYNC();
+ disable_dma(sport->dma_tx_chan);
+ /* Clear the status */
+ clear_dma_irqstat(sport->dma_tx_chan);
+ if (in_data != NULL) {
+ disable_dma(sport->dma_rx_chan);
+ clear_dma_irqstat(sport->dma_rx_chan);
+ }
+ SSYNC();
+ local_irq_restore(flags);
+
+ return 0;
+}
+EXPORT_SYMBOL(sport_send_and_recv);
+
+MODULE_AUTHOR("Roy Huang");
+MODULE_DESCRIPTION("SPORT driver for ADI Blackfin");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h
new file mode 100644
index 0000000..fcadcc0
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-sport.h
@@ -0,0 +1,194 @@
+/*
+ * File: bf5xx_ac97_sport.h
+ * Based on:
+ * Author: Roy Huang <roy.huang@analog.com>
+ *
+ * Created:
+ * Description:
+ *
+ * Copyright 2004-2007 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+
+#ifndef __BF5XX_SPORT_H__
+#define __BF5XX_SPORT_H__
+
+#include <linux/types.h>
+#include <linux/wait.h>
+#include <linux/workqueue.h>
+#include <asm/dma.h>
+
+struct sport_register {
+ u16 tcr1; u16 reserved0;
+ u16 tcr2; u16 reserved1;
+ u16 tclkdiv; u16 reserved2;
+ u16 tfsdiv; u16 reserved3;
+ u32 tx;
+ u32 reserved_l0;
+ u32 rx;
+ u32 reserved_l1;
+ u16 rcr1; u16 reserved4;
+ u16 rcr2; u16 reserved5;
+ u16 rclkdiv; u16 reserved6;
+ u16 rfsdiv; u16 reserved7;
+ u16 stat; u16 reserved8;
+ u16 chnl; u16 reserved9;
+ u16 mcmc1; u16 reserved10;
+ u16 mcmc2; u16 reserved11;
+ u32 mtcs0;
+ u32 mtcs1;
+ u32 mtcs2;
+ u32 mtcs3;
+ u32 mrcs0;
+ u32 mrcs1;
+ u32 mrcs2;
+ u32 mrcs3;
+};
+
+#define DESC_ELEMENT_COUNT 9
+
+struct sport_device {
+ int dma_rx_chan;
+ int dma_tx_chan;
+ int err_irq;
+ struct sport_register *regs;
+
+ unsigned char *rx_buf;
+ unsigned char *tx_buf;
+ unsigned int rx_fragsize;
+ unsigned int tx_fragsize;
+ unsigned int rx_frags;
+ unsigned int tx_frags;
+ unsigned int wdsize;
+
+ /* for dummy dma transfer */
+ void *dummy_buf;
+ unsigned int dummy_count;
+
+ /* DMA descriptor ring head of current audio stream*/
+ struct dmasg *dma_rx_desc;
+ struct dmasg *dma_tx_desc;
+ unsigned int rx_desc_bytes;
+ unsigned int tx_desc_bytes;
+
+ unsigned int rx_run:1; /* rx is running */
+ unsigned int tx_run:1; /* tx is running */
+
+ struct dmasg *dummy_rx_desc;
+ struct dmasg *dummy_tx_desc;
+
+ struct dmasg *curr_rx_desc;
+ struct dmasg *curr_tx_desc;
+
+ int rx_curr_frag;
+ int tx_curr_frag;
+
+ unsigned int rcr1;
+ unsigned int rcr2;
+ int rx_tdm_count;
+
+ unsigned int tcr1;
+ unsigned int tcr2;
+ int tx_tdm_count;
+
+ void (*rx_callback)(void *data);
+ void *rx_data;
+ void (*tx_callback)(void *data);
+ void *tx_data;
+ void (*err_callback)(void *data);
+ void *err_data;
+ unsigned char *tx_dma_buf;
+ unsigned char *rx_dma_buf;
+#ifdef CONFIG_SND_MMAP_SUPPORT
+ dma_addr_t tx_dma_phy;
+ dma_addr_t rx_dma_phy;
+ int tx_pos;/*pcm sample count*/
+ int rx_pos;
+ unsigned int tx_buffer_size;
+ unsigned int rx_buffer_size;
+ int tx_delay_pos;
+ int once;
+#endif
+ void *private_data;
+};
+
+extern struct sport_device *sport_handle;
+
+struct sport_param {
+ int dma_rx_chan;
+ int dma_tx_chan;
+ int err_irq;
+ struct sport_register *regs;
+};
+
+struct sport_device *sport_init(struct sport_param *param, unsigned wdsize,
+ unsigned dummy_count, void *private_data);
+
+void sport_done(struct sport_device *sport);
+
+/* first use these ...*/
+
+/* note: multichannel is in units of 8 channels, tdm_count is number of channels
+ * NOT / 8 ! all channels are enabled by default */
+int sport_set_multichannel(struct sport_device *sport, int tdm_count,
+ u32 mask, int packed);
+
+int sport_config_rx(struct sport_device *sport,
+ unsigned int rcr1, unsigned int rcr2,
+ unsigned int clkdiv, unsigned int fsdiv);
+
+int sport_config_tx(struct sport_device *sport,
+ unsigned int tcr1, unsigned int tcr2,
+ unsigned int clkdiv, unsigned int fsdiv);
+
+/* ... then these: */
+
+/* buffer size (in bytes) == fragcount * fragsize_bytes */
+
+/* this is not a very general api, it sets the dma to 2d autobuffer mode */
+
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+ int fragcount, size_t fragsize_bytes);
+
+int sport_config_tx_dma(struct sport_device *sport, void *buf,
+ int fragcount, size_t fragsize_bytes);
+
+int sport_tx_start(struct sport_device *sport);
+int sport_tx_stop(struct sport_device *sport);
+int sport_rx_start(struct sport_device *sport);
+int sport_rx_stop(struct sport_device *sport);
+
+/* for use in interrupt handler */
+unsigned long sport_curr_offset_rx(struct sport_device *sport);
+unsigned long sport_curr_offset_tx(struct sport_device *sport);
+
+void sport_incfrag(struct sport_device *sport, int *frag, int tx);
+void sport_decfrag(struct sport_device *sport, int *frag, int tx);
+
+int sport_set_rx_callback(struct sport_device *sport,
+ void (*rx_callback)(void *), void *rx_data);
+int sport_set_tx_callback(struct sport_device *sport,
+ void (*tx_callback)(void *), void *tx_data);
+int sport_set_err_callback(struct sport_device *sport,
+ void (*err_callback)(void *), void *err_data);
+
+int sport_send_and_recv(struct sport_device *sport, u8 *out_data, \
+ u8 *in_data, int len);
+#endif /* BF53X_SPORT_H */
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
new file mode 100644
index 0000000..e15f67f
--- /dev/null
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -0,0 +1,186 @@
+/*
+ * File: sound/soc/blackfin/bf5xx-ssm2602.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: board driver for SSM2602 sound chip
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm_params.h>
+
+#include <asm/dma.h>
+#include <asm/portmux.h>
+#include <linux/gpio.h>
+#include "../codecs/ssm2602.h"
+#include "bf5xx-sport.h"
+#include "bf5xx-i2s-pcm.h"
+#include "bf5xx-i2s.h"
+
+static struct snd_soc_machine bf5xx_ssm2602;
+
+static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ pr_debug("%s enter\n", __func__);
+ cpu_dai->private_data = sport_handle;
+ return 0;
+}
+
+static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
+ params_format(params));
+ /*
+ * If you are using a crystal source which frequency is not 12MHz
+ * then modify the below case statement with frequency of the crystal.
+ *
+ * If you are using the SPORT to generate clocking then this is
+ * where to do it.
+ */
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 12000000;
+ break;
+ }
+
+ /*
+ * CODEC is master for BCLK and LRC in this configuration.
+ */
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops bf5xx_ssm2602_ops = {
+ .startup = bf5xx_ssm2602_startup,
+ .hw_params = bf5xx_ssm2602_hw_params,
+};
+
+static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
+ .name = "ssm2602",
+ .stream_name = "SSM2602",
+ .cpu_dai = &bf5xx_i2s_dai,
+ .codec_dai = &ssm2602_dai,
+ .ops = &bf5xx_ssm2602_ops,
+};
+
+/*
+ * SSM2602 2 wire address is determined by CSB
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+
+static struct ssm2602_setup_data bf5xx_ssm2602_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1b,
+};
+
+static struct snd_soc_machine bf5xx_ssm2602 = {
+ .name = "bf5xx_ssm2602",
+ .dai_link = &bf5xx_ssm2602_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
+ .machine = &bf5xx_ssm2602,
+ .platform = &bf5xx_i2s_soc_platform,
+ .codec_dev = &soc_codec_dev_ssm2602,
+ .codec_data = &bf5xx_ssm2602_setup,
+};
+
+static struct platform_device *bf52x_ssm2602_snd_device;
+
+static int __init bf5xx_ssm2602_init(void)
+{
+ int ret;
+
+ pr_debug("%s enter\n", __func__);
+ bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!bf52x_ssm2602_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(bf52x_ssm2602_snd_device,
+ &bf5xx_ssm2602_snd_devdata);
+ bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev;
+ ret = platform_device_add(bf52x_ssm2602_snd_device);
+
+ if (ret)
+ platform_device_put(bf52x_ssm2602_snd_device);
+
+ return ret;
+}
+
+static void __exit bf5xx_ssm2602_exit(void)
+{
+ pr_debug("%s enter\n", __func__);
+ platform_device_unregister(bf52x_ssm2602_snd_device);
+}
+
+module_init(bf5xx_ssm2602_init);
+module_exit(bf5xx_ssm2602_exit);
+
+/* Module information */
+MODULE_AUTHOR("Cliff Cai");
+MODULE_DESCRIPTION("ALSA SoC SSM2602 BF527-EZKIT");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
new file mode 100644
index 0000000..38a0e3b
--- /dev/null
+++ b/sound/soc/codecs/Kconfig
@@ -0,0 +1,112 @@
+config SND_SOC_ALL_CODECS
+ tristate "Build all ASoC CODEC drivers"
+ depends on I2C
+ select SPI
+ select SPI_MASTER
+ select SND_SOC_AD73311
+ select SND_SOC_AK4535
+ select SND_SOC_CS4270
+ select SND_SOC_SSM2602
+ select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC26
+ select SND_SOC_TLV320AIC3X
+ select SND_SOC_UDA1380
+ select SND_SOC_WM8510
+ select SND_SOC_WM8580
+ select SND_SOC_WM8731
+ select SND_SOC_WM8750
+ select SND_SOC_WM8753
+ select SND_SOC_WM8900
+ select SND_SOC_WM8903
+ select SND_SOC_WM8971
+ select SND_SOC_WM8990
+ help
+ Normally ASoC codec drivers are only built if a machine driver which
+ uses them is also built since they are only usable with a machine
+ driver. Selecting this option will allow these drivers to be built
+ without an explicit machine driver for test and development purposes.
+
+ If unsure select "N".
+
+
+config SND_SOC_AC97_CODEC
+ tristate
+ select SND_AC97_CODEC
+
+config SND_SOC_AD1980
+ tristate
+
+config SND_SOC_AD73311
+ tristate
+
+config SND_SOC_AK4535
+ tristate
+
+# Cirrus Logic CS4270 Codec
+config SND_SOC_CS4270
+ tristate
+
+# Cirrus Logic CS4270 Codec Hardware Mute Support
+# Select if you have external muting circuitry attached to your CS4270.
+config SND_SOC_CS4270_HWMUTE
+ bool
+ depends on SND_SOC_CS4270
+
+# Cirrus Logic CS4270 Codec VD = 3.3V Errata
+# Select if you are affected by the errata where the part will not function
+# if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will
+# not select any sample rates that require MCLK to be divided by 1.5.
+config SND_SOC_CS4270_VD33_ERRATA
+ bool
+ depends on SND_SOC_CS4270
+
+config SND_SOC_SSM2602
+ tristate
+
+config SND_SOC_TLV320AIC23
+ tristate
+ depends on I2C
+
+config SND_SOC_TLV320AIC26
+ tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
+ depends on SPI
+
+config SND_SOC_TLV320AIC3X
+ tristate
+ depends on I2C
+
+config SND_SOC_UDA1380
+ tristate
+
+config SND_SOC_WM8510
+ tristate
+
+config SND_SOC_WM8580
+ tristate
+
+config SND_SOC_WM8731
+ tristate
+
+config SND_SOC_WM8750
+ tristate
+
+config SND_SOC_WM8753
+ tristate
+
+config SND_SOC_WM8900
+ tristate
+
+config SND_SOC_WM8903
+ tristate
+
+config SND_SOC_WM8971
+ tristate
+
+config SND_SOC_WM8990
+ tristate
+
+config SND_SOC_WM9712
+ tristate
+
+config SND_SOC_WM9713
+ tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
new file mode 100644
index 0000000..90f0a58
--- /dev/null
+++ b/sound/soc/codecs/Makefile
@@ -0,0 +1,43 @@
+snd-soc-ac97-objs := ac97.o
+snd-soc-ad1980-objs := ad1980.o
+snd-soc-ad73311-objs := ad73311.o
+snd-soc-ak4535-objs := ak4535.o
+snd-soc-cs4270-objs := cs4270.o
+snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-tlv320aic23-objs := tlv320aic23.o
+snd-soc-tlv320aic26-objs := tlv320aic26.o
+snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8510-objs := wm8510.o
+snd-soc-wm8580-objs := wm8580.o
+snd-soc-wm8731-objs := wm8731.o
+snd-soc-wm8750-objs := wm8750.o
+snd-soc-wm8753-objs := wm8753.o
+snd-soc-wm8900-objs := wm8900.o
+snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8971-objs := wm8971.o
+snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9712-objs := wm9712.o
+snd-soc-wm9713-objs := wm9713.o
+
+obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
+obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
+obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
+obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
+obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o
+obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
+obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
+obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
+obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
+obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
+obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
+obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
new file mode 100644
index 0000000..bd1ebdc
--- /dev/null
+++ b/sound/soc/codecs/ac97.c
@@ -0,0 +1,179 @@
+/*
+ * ac97.c -- ALSA Soc AC97 codec support
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Generic AC97 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include "ac97.h"
+
+#define AC97_VERSION "0.6"
+
+static int ac97_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
+ return snd_ac97_set_rate(codec->ac97, reg, runtime->rate);
+}
+
+#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai ac97_dai = {
+ .name = "AC97 HiFi",
+ .type = SND_SOC_DAI_AC97,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = STD_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = STD_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_prepare,},
+};
+EXPORT_SYMBOL_GPL(ac97_dai);
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ return soc_ac97_ops.read(codec->ac97, reg);
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ return 0;
+}
+
+static int ac97_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97_template ac97_template;
+ int ret = 0;
+
+ printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (!socdev->codec)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->name = "AC97";
+ codec->owner = THIS_MODULE;
+ codec->dai = &ac97_dai;
+ codec->num_dai = 1;
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto err;
+
+ /* add codec as bus device for standard ac97 */
+ ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus);
+ if (ret < 0)
+ goto bus_err;
+
+ memset(&ac97_template, 0, sizeof(struct snd_ac97_template));
+ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97);
+ if (ret < 0)
+ goto bus_err;
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0)
+ goto bus_err;
+ return 0;
+
+bus_err:
+ snd_soc_free_pcms(socdev);
+
+err:
+ kfree(socdev->codec->reg_cache);
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int ac97_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (!codec)
+ return 0;
+
+ snd_soc_free_pcms(socdev);
+ kfree(socdev->codec->reg_cache);
+ kfree(socdev->codec);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_ac97_suspend(socdev->codec->ac97);
+
+ return 0;
+}
+
+static int ac97_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_ac97_resume(socdev->codec->ac97);
+
+ return 0;
+}
+#else
+#define ac97_soc_suspend NULL
+#define ac97_soc_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_ac97 = {
+ .probe = ac97_soc_probe,
+ .remove = ac97_soc_remove,
+ .suspend = ac97_soc_suspend,
+ .resume = ac97_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
+
+MODULE_DESCRIPTION("Soc Generic AC97 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h
new file mode 100644
index 0000000..281aa42
--- /dev/null
+++ b/sound/soc/codecs/ac97.h
@@ -0,0 +1,19 @@
+/*
+ * linux/sound/codecs/ac97.h -- ALSA SoC Layer
+ *
+ * Author: Liam Girdwood
+ * Created: Dec 1st 2005
+ * Copyright: Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_AC97_H
+#define __LINUX_SND_SOC_AC97_H
+
+extern struct snd_soc_codec_device soc_codec_dev_ac97;
+extern struct snd_soc_dai ac97_dai;
+
+#endif
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
new file mode 100644
index 0000000..1397b8e
--- /dev/null
+++ b/sound/soc/codecs/ad1980.c
@@ -0,0 +1,308 @@
+/*
+ * ad1980.c -- ALSA Soc AD1980 codec support
+ *
+ * Copyright: Analog Device Inc.
+ * Author: Roy Huang <roy.huang@analog.com>
+ * Cliff Cai <cliff.cai@analog.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "ad1980.h"
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val);
+
+/*
+ * AD1980 register cache
+ */
+static const u16 ad1980_reg[] = {
+ 0x0090, 0x8000, 0x8000, 0x8000, /* 0 - 6 */
+ 0x0000, 0x0000, 0x8008, 0x8008, /* 8 - e */
+ 0x8808, 0x8808, 0x0000, 0x8808, /* 10 - 16 */
+ 0x8808, 0x0000, 0x8000, 0x0000, /* 18 - 1e */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 20 - 26 */
+ 0x03c7, 0x0000, 0xbb80, 0xbb80, /* 28 - 2e */
+ 0xbb80, 0xbb80, 0x0000, 0x8080, /* 30 - 36 */
+ 0x8080, 0x2000, 0x0000, 0x0000, /* 38 - 3e */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+ 0x8080, 0x0000, 0x0000, 0x0000, /* 60 - 66 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */
+ 0x0000, 0x0000, 0x1001, 0x0000, /* 70 - 76 */
+ 0x0000, 0x0000, 0x4144, 0x5370 /* 78 - 7e */
+};
+
+static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
+ "Stereo Mix", "Mono Mix", "Phone"};
+
+static const struct soc_enum ad1980_cap_src =
+ SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel);
+
+static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = {
+SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
+
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
+
+SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
+SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
+
+SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 31, 0),
+SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1),
+
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
+SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+SOC_SINGLE("Phone Capture Volume", AC97_PHONE, 0, 31, 1),
+SOC_SINGLE("Phone Capture Switch", AC97_PHONE, 15, 1, 1),
+
+SOC_SINGLE("Mic Volume", AC97_MIC, 0, 31, 1),
+SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+
+SOC_SINGLE("Stereo Mic Switch", AC97_AD_MISC, 6, 1, 0),
+SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
+
+SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
+
+SOC_ENUM("Capture Source", ad1980_cap_src),
+
+SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
+};
+
+/* add non dapm controls */
+static int ad1980_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) {
+ err = snd_ctl_add(codec->card, snd_soc_cnew(
+ &ad1980_snd_ac97_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ switch (reg) {
+ case AC97_RESET:
+ case AC97_INT_PAGING:
+ case AC97_POWERDOWN:
+ case AC97_EXTENDED_STATUS:
+ case AC97_VENDOR_ID1:
+ case AC97_VENDOR_ID2:
+ return soc_ac97_ops.read(codec->ac97, reg);
+ default:
+ reg = reg >> 1;
+
+ if (reg >= (ARRAY_SIZE(ad1980_reg)))
+ return -EINVAL;
+
+ return cache[reg];
+ }
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ reg = reg >> 1;
+ if (reg < (ARRAY_SIZE(ad1980_reg)))
+ cache[reg] = val;
+
+ return 0;
+}
+
+struct snd_soc_dai ad1980_dai = {
+ .name = "AC97",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+};
+EXPORT_SYMBOL_GPL(ad1980_dai);
+
+static int ad1980_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ u16 retry_cnt = 0;
+
+retry:
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (ac97_read(codec, AC97_RESET) == 0x0090)
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ /* Set bit 16slot in register 74h, then every slot will has only 16
+ * bits. This command is sent out in 20bit mode, in which case the
+ * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/
+ ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900);
+
+ if (ac97_read(codec, AC97_RESET) != 0x0090)
+ goto err;
+ return 0;
+
+err:
+ while (retry_cnt++ < 10)
+ goto retry;
+
+ printk(KERN_ERR "AD1980 AC97 reset failed\n");
+ return -EIO;
+}
+
+static int ad1980_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+ u16 vendor_id2;
+
+ printk(KERN_INFO "AD1980 SoC Audio Codec\n");
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache =
+ kzalloc(sizeof(u16) * ARRAY_SIZE(ad1980_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ memcpy(codec->reg_cache, ad1980_reg, sizeof(u16) * \
+ ARRAY_SIZE(ad1980_reg));
+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ad1980_reg);
+ codec->reg_cache_step = 2;
+ codec->name = "AD1980";
+ codec->owner = THIS_MODULE;
+ codec->dai = &ad1980_dai;
+ codec->num_dai = 1;
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
+ goto codec_err;
+ }
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+
+ ret = ad1980_reset(codec, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "AC97 link error\n");
+ goto reset_err;
+ }
+
+ /* Read out vendor ID to make sure it is ad1980 */
+ if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144)
+ goto reset_err;
+
+ vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2);
+
+ if (vendor_id2 != 0x5370) {
+ if (vendor_id2 != 0x5374)
+ goto reset_err;
+ else
+ printk(KERN_WARNING "ad1980: "
+ "Found AD1981 - only 2/2 IN/OUT Channels "
+ "supported\n");
+ }
+
+ ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */
+ ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */
+ ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */
+
+ ad1980_add_controls(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "ad1980: failed to register card\n");
+ goto reset_err;
+ }
+
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+
+codec_err:
+ kfree(codec->reg_cache);
+
+cache_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int ad1980_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_dapm_free(socdev);
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad1980 = {
+ .probe = ad1980_soc_probe,
+ .remove = ad1980_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad1980);
+
+MODULE_DESCRIPTION("ASoC ad1980 driver");
+MODULE_AUTHOR("Roy Huang, Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h
new file mode 100644
index 0000000..db6c850
--- /dev/null
+++ b/sound/soc/codecs/ad1980.h
@@ -0,0 +1,23 @@
+/*
+ * ad1980.h -- ad1980 Soc Audio driver
+ */
+
+#ifndef _AD1980_H
+#define _AD1980_H
+/* Bit definition of Power-Down Control/Status Register */
+#define ADC 0x0001
+#define DAC 0x0002
+#define ANL 0x0004
+#define REF 0x0008
+#define PR0 0x0100
+#define PR1 0x0200
+#define PR2 0x0400
+#define PR3 0x0800
+#define PR4 0x1000
+#define PR5 0x2000
+#define PR6 0x4000
+
+extern struct snd_soc_dai ad1980_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad1980;
+
+#endif
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
new file mode 100644
index 0000000..37af860
--- /dev/null
+++ b/sound/soc/codecs/ad73311.c
@@ -0,0 +1,107 @@
+/*
+ * ad73311.c -- ALSA Soc AD73311 codec support
+ *
+ * Copyright: Analog Device Inc.
+ * Author: Cliff Cai <cliff.cai@analog.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Revision history
+ * 25th Sep 2008 Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "ad73311.h"
+
+struct snd_soc_dai ad73311_dai = {
+ .name = "AD73311",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+};
+EXPORT_SYMBOL_GPL(ad73311_dai);
+
+static int ad73311_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+ mutex_init(&codec->mutex);
+ codec->name = "AD73311";
+ codec->owner = THIS_MODULE;
+ codec->dai = &ad73311_dai;
+ codec->num_dai = 1;
+ socdev->codec = codec;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "ad73311: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "ad73311: failed to register card\n");
+ goto register_err;
+ }
+
+ return ret;
+
+register_err:
+ snd_soc_free_pcms(socdev);
+pcm_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int ad73311_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+ snd_soc_free_pcms(socdev);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad73311 = {
+ .probe = ad73311_soc_probe,
+ .remove = ad73311_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
+
+MODULE_DESCRIPTION("ASoC ad73311 driver");
+MODULE_AUTHOR("Cliff Cai ");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
new file mode 100644
index 0000000..507ce0c
--- /dev/null
+++ b/sound/soc/codecs/ad73311.h
@@ -0,0 +1,90 @@
+/*
+ * File: sound/soc/codec/ad73311.h
+ * Based on:
+ * Author: Cliff Cai <cliff.cai@analog.com>
+ *
+ * Created: Thur Sep 25, 2008
+ * Description: definitions for AD73311 registers
+ *
+ *
+ * Modified:
+ * Copyright 2006 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef __AD73311_H__
+#define __AD73311_H__
+
+#define AD_CONTROL 0x8000
+#define AD_DATA 0x0000
+#define AD_READ 0x4000
+#define AD_WRITE 0x0000
+
+/* Control register A */
+#define CTRL_REG_A (0 << 8)
+
+#define REGA_MODE_PRO 0x00
+#define REGA_MODE_DATA 0x01
+#define REGA_MODE_MIXED 0x03
+#define REGA_DLB 0x04
+#define REGA_SLB 0x08
+#define REGA_DEVC(x) ((x & 0x7) << 4)
+#define REGA_RESET 0x80
+
+/* Control register B */
+#define CTRL_REG_B (1 << 8)
+
+#define REGB_DIRATE(x) (x & 0x3)
+#define REGB_SCDIV(x) ((x & 0x3) << 2)
+#define REGB_MCDIV(x) ((x & 0x7) << 4)
+#define REGB_CEE (1 << 7)
+
+/* Control register C */
+#define CTRL_REG_C (2 << 8)
+
+#define REGC_PUDEV (1 << 0)
+#define REGC_PUADC (1 << 3)
+#define REGC_PUDAC (1 << 4)
+#define REGC_PUREF (1 << 5)
+#define REGC_REFUSE (1 << 6)
+
+/* Control register D */
+#define CTRL_REG_D (3 << 8)
+
+#define REGD_IGS(x) (x & 0x7)
+#define REGD_RMOD (1 << 3)
+#define REGD_OGS(x) ((x & 0x7) << 4)
+#define REGD_MUTE (x << 7)
+
+/* Control register E */
+#define CTRL_REG_E (4 << 8)
+
+#define REGE_DA(x) (x & 0x1f)
+#define REGE_IBYP (1 << 5)
+
+/* Control register F */
+#define CTRL_REG_F (5 << 8)
+
+#define REGF_SEEN (1 << 5)
+#define REGF_INV (1 << 6)
+#define REGF_ALB (1 << 7)
+
+extern struct snd_soc_dai ad73311_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad73311;
+#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
new file mode 100644
index 0000000..2a89b58
--- /dev/null
+++ b/sound/soc/codecs/ak4535.c
@@ -0,0 +1,694 @@
+/*
+ * ak4535.c -- AK4535 ALSA Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "ak4535.h"
+
+#define AK4535_VERSION "0.3"
+
+struct snd_soc_codec_device soc_codec_dev_ak4535;
+
+/* codec private data */
+struct ak4535_priv {
+ unsigned int sysclk;
+};
+
+/*
+ * ak4535 register cache
+ */
+static const u16 ak4535_reg[AK4535_CACHEREGNUM] = {
+ 0x0000, 0x0080, 0x0000, 0x0003,
+ 0x0002, 0x0000, 0x0011, 0x0001,
+ 0x0000, 0x0040, 0x0036, 0x0010,
+ 0x0000, 0x0000, 0x0057, 0x0000,
+};
+
+/*
+ * read ak4535 register cache
+ */
+static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4535_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+static inline unsigned int ak4535_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 data;
+ data = reg;
+
+ if (codec->hw_write(codec->control_data, &data, 1) != 1)
+ return -EIO;
+
+ if (codec->hw_read(codec->control_data, &data, 1) != 1)
+ return -EIO;
+
+ return data;
+};
+
+/*
+ * write ak4535 register cache
+ */
+static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= AK4535_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the AK4535 register space
+ */
+static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D8 AK4535 register offset
+ * D7...D0 register data
+ */
+ data[0] = reg & 0xff;
+ data[1] = value & 0xff;
+
+ ak4535_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static int ak4535_sync(struct snd_soc_codec *codec)
+{
+ u16 *cache = codec->reg_cache;
+ int i, r = 0;
+
+ for (i = 0; i < AK4535_CACHEREGNUM; i++)
+ r |= ak4535_write(codec, i, cache[i]);
+
+ return r;
+};
+
+static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
+static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"};
+static const char *ak4535_hp_out[] = {"Stereo", "Mono"};
+static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"};
+static const char *ak4535_mic_select[] = {"Internal", "External"};
+
+static const struct soc_enum ak4535_enum[] = {
+ SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain),
+ SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out),
+ SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out),
+ SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp),
+ SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select),
+};
+
+static const struct snd_kcontrol_new ak4535_snd_controls[] = {
+ SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0),
+ SOC_ENUM("Mono 1 Output", ak4535_enum[1]),
+ SOC_ENUM("Mono 1 Gain", ak4535_enum[0]),
+ SOC_ENUM("Headphone Output", ak4535_enum[2]),
+ SOC_ENUM("Playback Deemphasis", ak4535_enum[3]),
+ SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0),
+ SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0),
+ SOC_ENUM("Mic Select", ak4535_enum[4]),
+ SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0),
+ SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0),
+ SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0),
+ SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0),
+ SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0),
+ SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0),
+ SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0),
+ SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1),
+ SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1),
+ SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0),
+ SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
+};
+
+/* add non dapm controls */
+static int ak4535_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Mono 1 Mixer */
+static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
+ SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0),
+};
+
+/* Stereo Mixer */
+static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0),
+ SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0),
+};
+
+/* Input Mixer */
+static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0),
+ SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new ak4535_input_mux_control =
+ SOC_DAPM_ENUM("Input Select", ak4535_enum[4]);
+
+/* HP L switch */
+static const struct snd_kcontrol_new ak4535_hpl_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1);
+
+/* HP R switch */
+static const struct snd_kcontrol_new ak4535_hpr_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1);
+
+/* mono 2 switch */
+static const struct snd_kcontrol_new ak4535_mono2_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0);
+
+/* Line out switch */
+static const struct snd_kcontrol_new ak4535_line_control =
+ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0);
+
+/* ak4535 dapm widgets */
+static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4535_stereo_mixer_controls[0],
+ ARRAY_SIZE(ak4535_stereo_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4535_mono1_mixer_controls[0],
+ ARRAY_SIZE(ak4535_mono1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4535_input_mixer_controls[0],
+ ARRAY_SIZE(ak4535_input_mixer_controls)),
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &ak4535_input_mux_control),
+ SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0),
+ SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_mono2_control),
+ /* speaker powersave bit */
+ SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_line_control),
+ SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_hpl_control),
+ SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0,
+ &ak4535_hpr_control),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+ SND_SOC_DAPM_OUTPUT("SPP"),
+ SND_SOC_DAPM_OUTPUT("SPN"),
+ SND_SOC_DAPM_OUTPUT("MOUT1"),
+ SND_SOC_DAPM_OUTPUT("MOUT2"),
+ SND_SOC_DAPM_OUTPUT("MICOUT"),
+ SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0),
+ SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0),
+ SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0),
+ SND_SOC_DAPM_INPUT("MICIN"),
+ SND_SOC_DAPM_INPUT("MICEXT"),
+ SND_SOC_DAPM_INPUT("AUX"),
+ SND_SOC_DAPM_INPUT("MIN"),
+ SND_SOC_DAPM_INPUT("AIN"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /*stereo mixer */
+ {"Stereo Mixer", "Playback Switch", "DAC"},
+ {"Stereo Mixer", "Mic Sidetone Switch", "Mic"},
+ {"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
+
+ /* mono1 mixer */
+ {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"},
+ {"Mono1 Mixer", "Mono Playback Switch", "DAC"},
+
+ /* Mic */
+ {"Mic", NULL, "AIN"},
+ {"Input Mux", "Internal", "Mic Int Bias"},
+ {"Input Mux", "External", "Mic Ext Bias"},
+ {"Mic Int Bias", NULL, "MICIN"},
+ {"Mic Ext Bias", NULL, "MICEXT"},
+ {"MICOUT", NULL, "Input Mux"},
+
+ /* line out */
+ {"LOUT", NULL, "Line Out Enable"},
+ {"ROUT", NULL, "Line Out Enable"},
+ {"Line Out Enable", "Switch", "Line Out"},
+ {"Line Out", NULL, "Stereo Mixer"},
+
+ /* mono1 out */
+ {"MOUT1", NULL, "Mono Out"},
+ {"Mono Out", NULL, "Mono1 Mixer"},
+
+ /* left HP */
+ {"HPL", NULL, "Left HP Enable"},
+ {"Left HP Enable", "Switch", "HP L Amp"},
+ {"HP L Amp", NULL, "Stereo Mixer"},
+
+ /* right HP */
+ {"HPR", NULL, "Right HP Enable"},
+ {"Right HP Enable", "Switch", "HP R Amp"},
+ {"HP R Amp", NULL, "Stereo Mixer"},
+
+ /* speaker */
+ {"SPP", NULL, "Speaker Enable"},
+ {"SPN", NULL, "Speaker Enable"},
+ {"Speaker Enable", "Switch", "Spk Amp"},
+ {"Spk Amp", NULL, "MIN"},
+
+ /* mono 2 */
+ {"MOUT2", NULL, "Mono 2 Enable"},
+ {"Mono 2 Enable", "Switch", "Stereo Mixer"},
+
+ /* Aux In */
+ {"Aux In", NULL, "AUX"},
+
+ /* ADC */
+ {"ADC", NULL, "Input Mixer"},
+ {"Input Mixer", "Mic Capture Switch", "Mic"},
+ {"Input Mixer", "Aux Capture Switch", "Aux In"},
+};
+
+static int ak4535_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets,
+ ARRAY_SIZE(ak4535_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ak4535_priv *ak4535 = codec->private_data;
+
+ ak4535->sysclk = freq;
+ return 0;
+}
+
+static int ak4535_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct ak4535_priv *ak4535 = codec->private_data;
+ u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
+ int rate = params_rate(params), fs = 256;
+
+ if (rate)
+ fs = ak4535->sysclk / rate;
+
+ /* set fs */
+ switch (fs) {
+ case 1024:
+ mode2 |= (0x2 << 5);
+ break;
+ case 512:
+ mode2 |= (0x1 << 5);
+ break;
+ case 256:
+ break;
+ }
+
+ /* set rate */
+ ak4535_write(codec, AK4535_MODE2, mode2);
+ return 0;
+}
+
+static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 mode1 = 0;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ mode1 = 0x0002;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ mode1 = 0x0001;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* use 32 fs for BCLK to save power */
+ mode1 |= 0x4;
+
+ ak4535_write(codec, AK4535_MODE1, mode1);
+ return 0;
+}
+
+static int ak4535_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
+ if (!mute)
+ ak4535_write(codec, AK4535_DAC, mute_reg);
+ else
+ ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
+ return 0;
+}
+
+static int ak4535_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 i;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ ak4535_mute(codec->dai, 0);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ ak4535_mute(codec->dai, 1);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ i = ak4535_read_reg_cache(codec, AK4535_PM1);
+ ak4535_write(codec, AK4535_PM1, i | 0x80);
+ i = ak4535_read_reg_cache(codec, AK4535_PM2);
+ ak4535_write(codec, AK4535_PM2, i & (~0x80));
+ break;
+ case SND_SOC_BIAS_OFF:
+ i = ak4535_read_reg_cache(codec, AK4535_PM1);
+ ak4535_write(codec, AK4535_PM1, i & (~0x80));
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai ak4535_dai = {
+ .name = "AK4535",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4535_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4535_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = ak4535_hw_params,
+ },
+ .dai_ops = {
+ .set_fmt = ak4535_set_dai_fmt,
+ .digital_mute = ak4535_mute,
+ .set_sysclk = ak4535_set_dai_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(ak4535_dai);
+
+static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int ak4535_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ ak4535_sync(codec);
+ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ ak4535_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the AK4535 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int ak4535_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "AK4535";
+ codec->owner = THIS_MODULE;
+ codec->read = ak4535_read_reg_cache;
+ codec->write = ak4535_write;
+ codec->set_bias_level = ak4535_set_bias_level;
+ codec->dai = &ak4535_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(ak4535_reg);
+ codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4535: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ak4535_add_controls(codec);
+ ak4535_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "ak4535: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+
+ return ret;
+}
+
+static struct snd_soc_device *ak4535_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static int ak4535_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = ak4535_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = ak4535_init(socdev);
+ if (ret < 0)
+ printk(KERN_ERR "failed to initialise AK4535\n");
+
+ return ret;
+}
+
+static int ak4535_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id ak4535_i2c_id[] = {
+ { "ak4535", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id);
+
+static struct i2c_driver ak4535_i2c_driver = {
+ .driver = {
+ .name = "AK4535 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4535_i2c_probe,
+ .remove = ak4535_i2c_remove,
+ .id_table = ak4535_i2c_id,
+};
+
+static int ak4535_add_i2c_device(struct platform_device *pdev,
+ const struct ak4535_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&ak4535_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "ak4535", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&ak4535_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+static int ak4535_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct ak4535_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct ak4535_priv *ak4535;
+ int ret;
+
+ printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL);
+ if (ak4535 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = ak4535;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ak4535_socdev = socdev;
+ ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ codec->hw_read = (hw_read_t)i2c_master_recv;
+ ret = ak4535_add_i2c_device(pdev, setup);
+ }
+#endif
+
+ if (ret != 0) {
+ kfree(codec->private_data);
+ kfree(codec);
+ }
+ return ret;
+}
+
+/* power down chip */
+static int ak4535_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&ak4535_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ak4535 = {
+ .probe = ak4535_probe,
+ .remove = ak4535_remove,
+ .suspend = ak4535_suspend,
+ .resume = ak4535_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
+
+MODULE_DESCRIPTION("Soc AK4535 driver");
+MODULE_AUTHOR("Richard Purdie");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h
new file mode 100644
index 0000000..c7a5870
--- /dev/null
+++ b/sound/soc/codecs/ak4535.h
@@ -0,0 +1,47 @@
+/*
+ * ak4535.h -- AK4535 Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AK4535_H
+#define _AK4535_H
+
+/* AK4535 register space */
+
+#define AK4535_PM1 0x0
+#define AK4535_PM2 0x1
+#define AK4535_SIG1 0x2
+#define AK4535_SIG2 0x3
+#define AK4535_MODE1 0x4
+#define AK4535_MODE2 0x5
+#define AK4535_DAC 0x6
+#define AK4535_MIC 0x7
+#define AK4535_TIMER 0x8
+#define AK4535_ALC1 0x9
+#define AK4535_ALC2 0xa
+#define AK4535_PGA 0xb
+#define AK4535_LATT 0xc
+#define AK4535_RATT 0xd
+#define AK4535_VOL 0xe
+#define AK4535_STATUS 0xf
+
+#define AK4535_CACHEREGNUM 0x10
+
+struct ak4535_setup_data {
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai ak4535_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4535;
+
+#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
new file mode 100644
index 0000000..0bbd945
--- /dev/null
+++ b/sound/soc/codecs/cs4270.c
@@ -0,0 +1,765 @@
+/*
+ * CS4270 ALSA SoC (ASoC) codec driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under
+ * the terms of the GNU General Public License version 2. This program
+ * is licensed "as is" without any warranty of any kind, whether express
+ * or implied.
+ *
+ * This is an ASoC device driver for the Cirrus Logic CS4270 codec.
+ *
+ * Current features/limitations:
+ *
+ * 1) Software mode is supported. Stand-alone mode is automatically
+ * selected if I2C is disabled or if a CS4270 is not found on the I2C
+ * bus. However, stand-alone mode is only partially implemented because
+ * there is no mechanism yet for this driver and the machine driver to
+ * communicate the values of the M0, M1, MCLK1, and MCLK2 pins.
+ * 2) Only I2C is supported, not SPI
+ * 3) Only Master mode is supported, not Slave.
+ * 4) The machine driver's 'startup' function must call
+ * cs4270_set_dai_sysclk() with the value of MCLK.
+ * 5) Only I2S and left-justified modes are supported
+ * 6) Power management is not supported
+ * 7) The only supported control is volume and hardware mute (if enabled)
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <linux/i2c.h>
+
+#include "cs4270.h"
+
+/* If I2C is defined, then we support software mode. However, if we're
+ not compiled as module but I2C is, then we can't use I2C calls. */
+#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
+#define USE_I2C
+#endif
+
+/* Private data for the CS4270 */
+struct cs4270_private {
+ unsigned int mclk; /* Input frequency of the MCLK pin */
+ unsigned int mode; /* The mode (I2S or left-justified) */
+};
+
+/*
+ * The codec isn't really big-endian or little-endian, since the I2S
+ * interface requires data to be sent serially with the MSbit first.
+ * However, to support BE and LE I2S devices, we specify both here. That
+ * way, ALSA will always match the bit patterns.
+ */
+#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
+
+#ifdef USE_I2C
+
+/* CS4270 registers addresses */
+#define CS4270_CHIPID 0x01 /* Chip ID */
+#define CS4270_PWRCTL 0x02 /* Power Control */
+#define CS4270_MODE 0x03 /* Mode Control */
+#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */
+#define CS4270_TRANS 0x05 /* Transition Control */
+#define CS4270_MUTE 0x06 /* Mute Control */
+#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */
+#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */
+
+#define CS4270_FIRSTREG 0x01
+#define CS4270_LASTREG 0x08
+#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1)
+
+/* Bit masks for the CS4270 registers */
+#define CS4270_CHIPID_ID 0xF0
+#define CS4270_CHIPID_REV 0x0F
+#define CS4270_PWRCTL_FREEZE 0x80
+#define CS4270_PWRCTL_PDN_ADC 0x20
+#define CS4270_PWRCTL_PDN_DAC 0x02
+#define CS4270_PWRCTL_PDN 0x01
+#define CS4270_MODE_SPEED_MASK 0x30
+#define CS4270_MODE_1X 0x00
+#define CS4270_MODE_2X 0x10
+#define CS4270_MODE_4X 0x20
+#define CS4270_MODE_SLAVE 0x30
+#define CS4270_MODE_DIV_MASK 0x0E
+#define CS4270_MODE_DIV1 0x00
+#define CS4270_MODE_DIV15 0x02
+#define CS4270_MODE_DIV2 0x04
+#define CS4270_MODE_DIV3 0x06
+#define CS4270_MODE_DIV4 0x08
+#define CS4270_MODE_POPGUARD 0x01
+#define CS4270_FORMAT_FREEZE_A 0x80
+#define CS4270_FORMAT_FREEZE_B 0x40
+#define CS4270_FORMAT_LOOPBACK 0x20
+#define CS4270_FORMAT_DAC_MASK 0x18
+#define CS4270_FORMAT_DAC_LJ 0x00
+#define CS4270_FORMAT_DAC_I2S 0x08
+#define CS4270_FORMAT_DAC_RJ16 0x18
+#define CS4270_FORMAT_DAC_RJ24 0x10
+#define CS4270_FORMAT_ADC_MASK 0x01
+#define CS4270_FORMAT_ADC_LJ 0x00
+#define CS4270_FORMAT_ADC_I2S 0x01
+#define CS4270_TRANS_ONE_VOL 0x80
+#define CS4270_TRANS_SOFT 0x40
+#define CS4270_TRANS_ZERO 0x20
+#define CS4270_TRANS_INV_ADC_A 0x08
+#define CS4270_TRANS_INV_ADC_B 0x10
+#define CS4270_TRANS_INV_DAC_A 0x02
+#define CS4270_TRANS_INV_DAC_B 0x04
+#define CS4270_TRANS_DEEMPH 0x01
+#define CS4270_MUTE_AUTO 0x20
+#define CS4270_MUTE_ADC_A 0x08
+#define CS4270_MUTE_ADC_B 0x10
+#define CS4270_MUTE_POLARITY 0x04
+#define CS4270_MUTE_DAC_A 0x01
+#define CS4270_MUTE_DAC_B 0x02
+
+/*
+ * Clock Ratio Selection for Master Mode with I2C enabled
+ *
+ * The data for this chart is taken from Table 5 of the CS4270 reference
+ * manual.
+ *
+ * This table is used to determine how to program the Mode Control register.
+ * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling
+ * rates the CS4270 currently supports.
+ *
+ * Each element in this array corresponds to the ratios in mclk_ratios[].
+ * These two arrays need to be in sync.
+ *
+ * 'speed_mode' is the corresponding bit pattern to be written to the
+ * MODE bits of the Mode Control Register
+ *
+ * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of
+ * the Mode Control Register.
+ *
+ * In situations where a single ratio is represented by multiple speed
+ * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick
+ * double-speed instead of quad-speed. However, the CS4270 errata states
+ * that Divide-By-1.5 can cause failures, so we avoid that mode where
+ * possible.
+ *
+ * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not
+ * work if VD = 3.3V. If this effects you, select the
+ * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will
+ * never select any sample rates that require divide-by-1.5.
+ */
+static struct {
+ unsigned int ratio;
+ u8 speed_mode;
+ u8 mclk;
+} cs4270_mode_ratios[] = {
+ {64, CS4270_MODE_4X, CS4270_MODE_DIV1},
+#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA
+ {96, CS4270_MODE_4X, CS4270_MODE_DIV15},
+#endif
+ {128, CS4270_MODE_2X, CS4270_MODE_DIV1},
+ {192, CS4270_MODE_4X, CS4270_MODE_DIV3},
+ {256, CS4270_MODE_1X, CS4270_MODE_DIV1},
+ {384, CS4270_MODE_2X, CS4270_MODE_DIV3},
+ {512, CS4270_MODE_1X, CS4270_MODE_DIV2},
+ {768, CS4270_MODE_1X, CS4270_MODE_DIV3},
+ {1024, CS4270_MODE_1X, CS4270_MODE_DIV4}
+};
+
+/* The number of MCLK/LRCK ratios supported by the CS4270 */
+#define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios)
+
+/*
+ * Determine the CS4270 samples rates.
+ *
+ * 'freq' is the input frequency to MCLK. The other parameters are ignored.
+ *
+ * The value of MCLK is used to determine which sample rates are supported
+ * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine
+ * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024.
+ *
+ * This function calculates the nine ratios and determines which ones match
+ * a standard sample rate. If there's a match, then it is added to the list
+ * of support sample rates.
+ *
+ * This function must be called by the machine driver's 'startup' function,
+ * otherwise the list of supported sample rates will not be available in
+ * time for ALSA.
+ *
+ * Note that in stand-alone mode, the sample rate is determined by input
+ * pins M0, M1, MDIV1, and MDIV2. Also in stand-alone mode, divide-by-3
+ * is not a programmable option. However, divide-by-3 is not an available
+ * option in stand-alone mode. This cases two problems: a ratio of 768 is
+ * not available (it requires divide-by-3) and B) ratios 192 and 384 can
+ * only be selected with divide-by-1.5, but there is an errate that make
+ * this selection difficult.
+ *
+ * In addition, there is no mechanism for communicating with the machine
+ * driver what the input settings can be. This would need to be implemented
+ * for stand-alone mode to work.
+ */
+static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs4270_private *cs4270 = codec->private_data;
+ unsigned int rates = 0;
+ unsigned int rate_min = -1;
+ unsigned int rate_max = 0;
+ unsigned int i;
+
+ cs4270->mclk = freq;
+
+ for (i = 0; i < NUM_MCLK_RATIOS; i++) {
+ unsigned int rate = freq / cs4270_mode_ratios[i].ratio;
+ rates |= snd_pcm_rate_to_rate_bit(rate);
+ if (rate < rate_min)
+ rate_min = rate;
+ if (rate > rate_max)
+ rate_max = rate;
+ }
+ /* FIXME: soc should support a rate list */
+ rates &= ~SNDRV_PCM_RATE_KNOT;
+
+ if (!rates) {
+ printk(KERN_ERR "cs4270: could not find a valid sample rate\n");
+ return -EINVAL;
+ }
+
+ codec_dai->playback.rates = rates;
+ codec_dai->playback.rate_min = rate_min;
+ codec_dai->playback.rate_max = rate_max;
+
+ codec_dai->capture.rates = rates;
+ codec_dai->capture.rate_min = rate_min;
+ codec_dai->capture.rate_max = rate_max;
+
+ return 0;
+}
+
+/*
+ * Configure the codec for the selected audio format
+ *
+ * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the
+ * codec accordingly.
+ *
+ * Currently, this function only supports SND_SOC_DAIFMT_I2S and
+ * SND_SOC_DAIFMT_LEFT_J. The CS4270 codec also supports right-justified
+ * data for playback only, but ASoC currently does not support different
+ * formats for playback vs. record.
+ */
+static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int format)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cs4270_private *cs4270 = codec->private_data;
+ int ret = 0;
+
+ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_LEFT_J:
+ cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
+ break;
+ default:
+ printk(KERN_ERR "cs4270: invalid DAI format\n");
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+/*
+ * A list of addresses on which this CS4270 could use. I2C addresses are
+ * 7 bits. For the CS4270, the upper four bits are always 1001, and the
+ * lower three bits are determined via the AD2, AD1, and AD0 pins
+ * (respectively).
+ */
+static const unsigned short normal_i2c[] = {
+ 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END
+};
+I2C_CLIENT_INSMOD;
+
+/*
+ * Pre-fill the CS4270 register cache.
+ *
+ * We use the auto-increment feature of the CS4270 to read all registers in
+ * one shot.
+ */
+static int cs4270_fill_cache(struct snd_soc_codec *codec)
+{
+ u8 *cache = codec->reg_cache;
+ struct i2c_client *i2c_client = codec->control_data;
+ s32 length;
+
+ length = i2c_smbus_read_i2c_block_data(i2c_client,
+ CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
+
+ if (length != CS4270_NUMREGS) {
+ printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n",
+ i2c_client->addr);
+ return -EIO;
+ }
+
+ return 0;
+}
+
+/*
+ * Read from the CS4270 register cache.
+ *
+ * This CS4270 registers are cached to avoid excessive I2C I/O operations.
+ * After the initial read to pre-fill the cache, the CS4270 never updates
+ * the register values, so we won't have a cache coherncy problem.
+ */
+static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+
+ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
+ return -EIO;
+
+ return cache[reg - CS4270_FIRSTREG];
+}
+
+/*
+ * Write to a CS4270 register via the I2C bus.
+ *
+ * This function writes the given value to the given CS4270 register, and
+ * also updates the register cache.
+ *
+ * Note that we don't use the hw_write function pointer of snd_soc_codec.
+ * That's because it's too clunky: the hw_write_t prototype does not match
+ * i2c_smbus_write_byte_data(), and it's just another layer of overhead.
+ */
+static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 *cache = codec->reg_cache;
+
+ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
+ return -EIO;
+
+ /* Only perform an I2C operation if the new value is different */
+ if (cache[reg - CS4270_FIRSTREG] != value) {
+ struct i2c_client *client = codec->control_data;
+ if (i2c_smbus_write_byte_data(client, reg, value)) {
+ printk(KERN_ERR "cs4270: I2C write failed\n");
+ return -EIO;
+ }
+
+ /* We've written to the hardware, so update the cache */
+ cache[reg - CS4270_FIRSTREG] = value;
+ }
+
+ return 0;
+}
+
+/*
+ * Program the CS4270 with the given hardware parameters.
+ *
+ * The .dai_ops functions are used to provide board-specific data, like
+ * input frequencies, to this driver. This function takes that information,
+ * combines it with the hardware parameters provided, and programs the
+ * hardware accordingly.
+ */
+static int cs4270_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct cs4270_private *cs4270 = codec->private_data;
+ int ret;
+ unsigned int i;
+ unsigned int rate;
+ unsigned int ratio;
+ int reg;
+
+ /* Figure out which MCLK/LRCK ratio to use */
+
+ rate = params_rate(params); /* Sampling rate, in Hz */
+ ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */
+
+ for (i = 0; i < NUM_MCLK_RATIOS; i++) {
+ if (cs4270_mode_ratios[i].ratio == ratio)
+ break;
+ }
+
+ if (i == NUM_MCLK_RATIOS) {
+ /* We did not find a matching ratio */
+ printk(KERN_ERR "cs4270: could not find matching ratio\n");
+ return -EINVAL;
+ }
+
+ /* Freeze and power-down the codec */
+
+ ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE |
+ CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC |
+ CS4270_PWRCTL_PDN);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: I2C write failed\n");
+ return ret;
+ }
+
+ /* Program the mode control register */
+
+ reg = snd_soc_read(codec, CS4270_MODE);
+ reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK);
+ reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk;
+
+ ret = snd_soc_write(codec, CS4270_MODE, reg);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: I2C write failed\n");
+ return ret;
+ }
+
+ /* Program the format register */
+
+ reg = snd_soc_read(codec, CS4270_FORMAT);
+ reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK);
+
+ switch (cs4270->mode) {
+ case SND_SOC_DAIFMT_I2S:
+ reg |= CS4270_FORMAT_DAC_I2S | CS4270_FORMAT_ADC_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ;
+ break;
+ default:
+ printk(KERN_ERR "cs4270: unknown format\n");
+ return -EINVAL;
+ }
+
+ ret = snd_soc_write(codec, CS4270_FORMAT, reg);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: I2C write failed\n");
+ return ret;
+ }
+
+ /* Disable auto-mute. This feature appears to be buggy, because in
+ some situations, auto-mute will not deactivate when it should. */
+
+ reg = snd_soc_read(codec, CS4270_MUTE);
+ reg &= ~CS4270_MUTE_AUTO;
+ ret = snd_soc_write(codec, CS4270_MUTE, reg);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: I2C write failed\n");
+ return ret;
+ }
+
+ /* Thaw and power-up the codec */
+
+ ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: I2C write failed\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
+
+/*
+ * Set the CS4270 external mute
+ *
+ * This function toggles the mute bits in the MUTE register. The CS4270's
+ * mute capability is intended for external muting circuitry, so if the
+ * board does not have the MUTEA or MUTEB pins connected to such circuitry,
+ * then this function will do nothing.
+ */
+static int cs4270_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int reg6;
+
+ reg6 = snd_soc_read(codec, CS4270_MUTE);
+
+ if (mute)
+ reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
+ CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
+ else
+ reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
+ CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+
+ return snd_soc_write(codec, CS4270_MUTE, reg6);
+}
+
+#endif
+
+static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *);
+
+/* A list of non-DAPM controls that the CS4270 supports */
+static const struct snd_kcontrol_new cs4270_snd_controls[] = {
+ SOC_DOUBLE_R("Master Playback Volume",
+ CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1)
+};
+
+static const struct i2c_device_id cs4270_id[] = {
+ {"cs4270", 0},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, cs4270_id);
+
+static struct i2c_driver cs4270_i2c_driver = {
+ .driver = {
+ .name = "CS4270 I2C",
+ .owner = THIS_MODULE,
+ },
+ .id_table = cs4270_id,
+ .probe = cs4270_i2c_probe,
+};
+
+/*
+ * Global variable to store socdev for i2c probe function.
+ *
+ * If struct i2c_driver had a private_data field, we wouldn't need to use
+ * cs4270_socdec. This is the only way to pass the socdev structure to
+ * cs4270_i2c_probe().
+ *
+ * The real solution to cs4270_socdev is to create a mechanism
+ * that maps I2C addresses to snd_soc_device structures. Perhaps the
+ * creation of the snd_soc_device object should be moved out of
+ * cs4270_probe() and into cs4270_i2c_probe(), but that would make this
+ * driver dependent on I2C. The CS4270 supports "stand-alone" mode, whereby
+ * the chip is *not* connected to the I2C bus, but is instead configured via
+ * input pins.
+ */
+static struct snd_soc_device *cs4270_socdev;
+
+/*
+ * Initialize the I2C interface of the CS4270
+ *
+ * This function is called for whenever the I2C subsystem finds a device
+ * at a particular address.
+ *
+ * Note: snd_soc_new_pcms() must be called before this function can be called,
+ * because of snd_ctl_add().
+ */
+static int cs4270_i2c_probe(struct i2c_client *i2c_client,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = cs4270_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ int ret = 0;
+
+ /* Probing all possible addresses has one drawback: if there are
+ multiple CS4270s on the bus, then you cannot specify which
+ socdev is matched with which CS4270. For now, we just reject
+ this I2C device if the socdev already has one attached. */
+ if (codec->control_data)
+ return -ENODEV;
+
+ /* Note: codec_dai->codec is NULL here */
+
+ codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL);
+ if (!codec->reg_cache) {
+ printk(KERN_ERR "cs4270: could not allocate register cache\n");
+ ret = -ENOMEM;
+ goto error;
+ }
+
+ /* Verify that we have a CS4270 */
+
+ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: failed to read I2C\n");
+ goto error;
+ }
+ /* The top four bits of the chip ID should be 1100. */
+ if ((ret & 0xF0) != 0xC0) {
+ /* The device at this address is not a CS4270 codec */
+ ret = -ENODEV;
+ goto error;
+ }
+
+ printk(KERN_INFO "cs4270: found device at I2C address %X\n",
+ i2c_client->addr);
+ printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF);
+
+ codec->control_data = i2c_client;
+ codec->read = cs4270_read_reg_cache;
+ codec->write = cs4270_i2c_write;
+ codec->reg_cache_size = CS4270_NUMREGS;
+
+ /* The I2C interface is set up, so pre-fill our register cache */
+
+ ret = cs4270_fill_cache(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: failed to fill register cache\n");
+ goto error;
+ }
+
+ /* Add the non-DAPM controls */
+
+ for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) {
+ struct snd_kcontrol *kctrl =
+ snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
+
+ ret = snd_ctl_add(codec->card, kctrl);
+ if (ret < 0)
+ goto error;
+ }
+
+ i2c_set_clientdata(i2c_client, codec);
+
+ return 0;
+
+error:
+ codec->control_data = NULL;
+
+ kfree(codec->reg_cache);
+ codec->reg_cache = NULL;
+ codec->reg_cache_size = 0;
+
+ return ret;
+}
+
+#endif /* USE_I2C*/
+
+struct snd_soc_dai cs4270_dai = {
+ .name = "CS4270",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = 0,
+ .formats = CS4270_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = 0,
+ .formats = CS4270_FORMATS,
+ },
+};
+EXPORT_SYMBOL_GPL(cs4270_dai);
+
+/*
+ * ASoC probe function
+ *
+ * This function is called when the machine driver calls
+ * platform_device_add().
+ */
+static int cs4270_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "CS4270 ALSA SoC Codec\n");
+
+ /* Allocate enough space for the snd_soc_codec structure
+ and our private data together. */
+ codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) +
+ sizeof(struct cs4270_private), GFP_KERNEL);
+ if (!codec) {
+ printk(KERN_ERR "cs4270: Could not allocate codec structure\n");
+ return -ENOMEM;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->name = "CS4270";
+ codec->owner = THIS_MODULE;
+ codec->dai = &cs4270_dai;
+ codec->num_dai = 1;
+ codec->private_data = (void *) codec +
+ ALIGN(sizeof(struct snd_soc_codec), 4);
+
+ socdev->codec = codec;
+
+ /* Register PCMs */
+
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: failed to create PCMs\n");
+ goto error_free_codec;
+ }
+
+#ifdef USE_I2C
+ cs4270_socdev = socdev;
+
+ ret = i2c_add_driver(&cs4270_i2c_driver);
+ if (ret) {
+ printk(KERN_ERR "cs4270: failed to attach driver");
+ goto error_free_pcms;
+ }
+
+ /* Did we find a CS4270 on the I2C bus? */
+ if (codec->control_data) {
+ /* Initialize codec ops */
+ cs4270_dai.ops.hw_params = cs4270_hw_params;
+ cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk;
+ cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt;
+#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
+ cs4270_dai.dai_ops.digital_mute = cs4270_mute;
+#endif
+ } else
+ printk(KERN_INFO "cs4270: no I2C device found, "
+ "using stand-alone mode\n");
+#else
+ printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
+#endif
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "cs4270: failed to register card\n");
+ goto error_del_driver;
+ }
+
+ return 0;
+
+error_del_driver:
+#ifdef USE_I2C
+ i2c_del_driver(&cs4270_i2c_driver);
+
+error_free_pcms:
+#endif
+ snd_soc_free_pcms(socdev);
+
+error_free_codec:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+
+ return ret;
+}
+
+static int cs4270_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+
+#ifdef USE_I2C
+ i2c_del_driver(&cs4270_i2c_driver);
+#endif
+
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+
+ return 0;
+}
+
+/*
+ * ASoC codec device structure
+ *
+ * Assign this variable to the codec_dev field of the machine driver's
+ * snd_soc_device structure.
+ */
+struct snd_soc_codec_device soc_codec_device_cs4270 = {
+ .probe = cs4270_probe,
+ .remove = cs4270_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h
new file mode 100644
index 0000000..adc6cd9
--- /dev/null
+++ b/sound/soc/codecs/cs4270.h
@@ -0,0 +1,28 @@
+/*
+ * Cirrus Logic CS4270 ALSA SoC Codec Driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under
+ * the terms of the GNU General Public License version 2. This program
+ * is licensed "as is" without any warranty of any kind, whether express
+ * or implied.
+ */
+
+#ifndef _CS4270_H
+#define _CS4270_H
+
+/*
+ * The ASoC codec DAI structure for the CS4270. Assign this structure to
+ * the .codec_dai field of your machine driver's snd_soc_dai_link structure.
+ */
+extern struct snd_soc_dai cs4270_dai;
+
+/*
+ * The ASoC codec device structure for the CS4270. Assign this structure
+ * to the .codec_dev field of your machine driver's snd_soc_device
+ * structure.
+ */
+extern struct snd_soc_codec_device soc_codec_device_cs4270;
+
+#endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
new file mode 100644
index 0000000..44ef0da
--- /dev/null
+++ b/sound/soc/codecs/ssm2602.c
@@ -0,0 +1,775 @@
+/*
+ * File: sound/soc/codecs/ssm2602.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: Driver for ssm2602 sound chip
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "ssm2602.h"
+
+#define SSM2602_VERSION "0.1"
+
+struct snd_soc_codec_device soc_codec_dev_ssm2602;
+
+/* codec private data */
+struct ssm2602_priv {
+ unsigned int sysclk;
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+/*
+ * ssm2602 register cache
+ * We can't read the ssm2602 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
+ 0x0017, 0x0017, 0x0079, 0x0079,
+ 0x0000, 0x0000, 0x0000, 0x000a,
+ 0x0000, 0x0000
+};
+
+/*
+ * read ssm2602 register cache
+ */
+static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == SSM2602_RESET)
+ return 0;
+ if (reg >= SSM2602_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write ssm2602 register cache
+ */
+static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= SSM2602_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the ssm2602 register space
+ */
+static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 ssm2602 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ ssm2602_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define ssm2602_reset(c) ssm2602_write(c, SSM2602_RESET, 0)
+
+/*Appending several "None"s just for OSS mixer use*/
+static const char *ssm2602_input_select[] = {
+ "Line", "Mic", "None", "None", "None",
+ "None", "None", "None",
+};
+
+static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum ssm2602_enum[] = {
+ SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select),
+ SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph),
+};
+
+static const struct snd_kcontrol_new ssm2602_snd_controls[] = {
+
+SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V,
+ 0, 127, 0),
+SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V,
+ 7, 1, 0),
+
+SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0),
+SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
+
+SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
+SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
+
+SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1),
+
+SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1),
+SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0),
+
+SOC_ENUM("Capture Source", ssm2602_enum[0]),
+
+SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
+};
+
+/* add non dapm controls */
+static int ssm2602_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Output Mixer */
+static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new ssm2602_input_mux_controls =
+SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]);
+
+static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1,
+ &ssm2602_output_mixer_controls[0],
+ ARRAY_SIZE(ssm2602_output_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
+SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls),
+SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1),
+SND_SOC_DAPM_INPUT("MICIN"),
+SND_SOC_DAPM_INPUT("RLINEIN"),
+SND_SOC_DAPM_INPUT("LLINEIN"),
+};
+
+static const struct snd_soc_dapm_route audio_conn[] = {
+ /* output mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+ /* outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+
+ /* input mux */
+ {"Input Mux", "Line", "Line Input"},
+ {"Input Mux", "Mic", "Mic Bias"},
+ {"ADC", NULL, "Input Mux"},
+
+ /* inputs */
+ {"Line Input", NULL, "LLINEIN"},
+ {"Line Input", NULL, "RLINEIN"},
+ {"Mic Bias", NULL, "MICIN"},
+};
+
+static int ssm2602_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets,
+ ARRAY_SIZE(ssm2602_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:4;
+ u8 bosr:1;
+ u8 usb:1;
+};
+
+/* codec mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0, 0x0},
+ {18432000, 48000, 384, 0x0, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x0, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0x6, 0x0, 0x0},
+ {18432000, 32000, 576, 0x6, 0x1, 0x0},
+ {12000000, 32000, 375, 0x6, 0x0, 0x1},
+
+ /* 8k */
+ {12288000, 8000, 1536, 0x3, 0x0, 0x0},
+ {18432000, 8000, 2304, 0x3, 0x1, 0x0},
+ {11289600, 8000, 1408, 0xb, 0x0, 0x0},
+ {16934400, 8000, 2112, 0xb, 0x1, 0x0},
+ {12000000, 8000, 1500, 0x3, 0x0, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0x7, 0x0, 0x0},
+ {18432000, 96000, 192, 0x7, 0x1, 0x0},
+ {12000000, 96000, 125, 0x7, 0x0, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x8, 0x0, 0x0},
+ {16934400, 44100, 384, 0x8, 0x1, 0x0},
+ {12000000, 44100, 272, 0x8, 0x1, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0xf, 0x0, 0x0},
+ {16934400, 88200, 192, 0xf, 0x1, 0x0},
+ {12000000, 88200, 136, 0xf, 0x1, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return i;
+}
+
+static int ssm2602_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ u16 srate;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
+ u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
+ int i = get_coeff(ssm2602->sysclk, params_rate(params));
+
+ /*no match is found*/
+ if (i == ARRAY_SIZE(coeff_div))
+ return -EINVAL;
+
+ srate = (coeff_div[i].sr << 2) |
+ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+ ssm2602_write(codec, SSM2602_SRATE, srate);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x000c;
+ break;
+ }
+ ssm2602_write(codec, SSM2602_IFACE, iface);
+ ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
+ return 0;
+}
+
+static int ssm2602_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
+ struct snd_pcm_runtime *master_runtime;
+
+ /* The DAI has shared clocks so if we already have a playback or
+ * capture going then constrain this substream to match it.
+ */
+ if (ssm2602->master_substream) {
+ master_runtime = ssm2602->master_substream->runtime;
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ ssm2602->slave_substream = substream;
+ } else
+ ssm2602->master_substream = substream;
+
+ return 0;
+}
+
+static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ /* set active */
+ ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
+
+ return 0;
+}
+
+static void ssm2602_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ /* deactivate */
+ if (!codec->active)
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+}
+
+static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE;
+ if (mute)
+ ssm2602_write(codec, SSM2602_APDIGI,
+ mute_reg | APDIGI_ENABLE_DAC_MUTE);
+ else
+ ssm2602_write(codec, SSM2602_APDIGI, mute_reg);
+ return 0;
+}
+
+static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ssm2602_priv *ssm2602 = codec->private_data;
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ ssm2602->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ ssm2602_write(codec, SSM2602_IFACE, iface);
+ return 0;
+}
+
+static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* vref/mid, osc on, dac unmute */
+ ssm2602_write(codec, SSM2602_PWR, reg);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* everything off except vref/vmid, */
+ ssm2602_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+ ssm2602_write(codec, SSM2602_PWR, 0xffff);
+ break;
+
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+struct snd_soc_dai ssm2602_dai = {
+ .name = "SSM2602",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SSM2602_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SSM2602_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,},
+ .ops = {
+ .startup = ssm2602_startup,
+ .prepare = ssm2602_pcm_prepare,
+ .hw_params = ssm2602_hw_params,
+ .shutdown = ssm2602_shutdown,
+ },
+ .dai_ops = {
+ .digital_mute = ssm2602_mute,
+ .set_sysclk = ssm2602_set_dai_sysclk,
+ .set_fmt = ssm2602_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(ssm2602_dai);
+
+static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int ssm2602_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(ssm2602_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ ssm2602_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the ssm2602 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int ssm2602_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "SSM2602";
+ codec->owner = THIS_MODULE;
+ codec->read = ssm2602_read_reg_cache;
+ codec->write = ssm2602_write;
+ codec->set_bias_level = ssm2602_set_bias_level;
+ codec->dai = &ssm2602_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = sizeof(ssm2602_reg);
+ codec->reg_cache = kmemdup(ssm2602_reg, sizeof(ssm2602_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ ssm2602_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ pr_err("ssm2602: failed to create pcms\n");
+ goto pcm_err;
+ }
+ /*power on device*/
+ ssm2602_write(codec, SSM2602_ACTIVE, 0);
+ /* set the update bits */
+ reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL);
+ ssm2602_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH);
+ reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL);
+ ssm2602_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH);
+ reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V);
+ ssm2602_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH);
+ reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V);
+ ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH);
+ /*select Line in as default input*/
+ ssm2602_write(codec, SSM2602_APANA,
+ APANA_ENABLE_MIC_BOOST2 | APANA_SELECT_DAC |
+ APANA_ENABLE_MIC_BOOST);
+ ssm2602_write(codec, SSM2602_PWR, 0);
+
+ ssm2602_add_controls(codec);
+ ssm2602_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ pr_err("ssm2602: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *ssm2602_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * ssm2602 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static int ssm2602_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = ssm2602_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = ssm2602_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise SSM2602\n");
+
+ return ret;
+}
+
+static int ssm2602_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id ssm2602_i2c_id[] = {
+ { "ssm2602", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
+/* corgi i2c codec control layer */
+static struct i2c_driver ssm2602_i2c_driver = {
+ .driver = {
+ .name = "SSM2602 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = ssm2602_i2c_probe,
+ .remove = ssm2602_i2c_remove,
+ .id_table = ssm2602_i2c_id,
+};
+
+static int ssm2602_add_i2c_device(struct platform_device *pdev,
+ const struct ssm2602_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&ssm2602_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "ssm2602", I2C_NAME_SIZE);
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+ return 0;
+err_driver:
+ i2c_del_driver(&ssm2602_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+static int ssm2602_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct ssm2602_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct ssm2602_priv *ssm2602;
+ int ret = 0;
+
+ pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL);
+ if (ssm2602 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = ssm2602;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ssm2602_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = ssm2602_add_i2c_device(pdev, setup);
+ }
+#else
+ /* other interfaces */
+#endif
+ return ret;
+}
+
+/* remove everything here */
+static int ssm2602_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&ssm2602_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
+ .probe = ssm2602_probe,
+ .remove = ssm2602_remove,
+ .suspend = ssm2602_suspend,
+ .resume = ssm2602_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
+
+MODULE_DESCRIPTION("ASoC ssm2602 driver");
+MODULE_AUTHOR("Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h
new file mode 100644
index 0000000..f344e6d
--- /dev/null
+++ b/sound/soc/codecs/ssm2602.h
@@ -0,0 +1,130 @@
+/*
+ * File: sound/soc/codecs/ssm2602.h
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef _SSM2602_H
+#define _SSM2602_H
+
+/* SSM2602 Codec Register definitions */
+
+#define SSM2602_LINVOL 0x00
+#define SSM2602_RINVOL 0x01
+#define SSM2602_LOUT1V 0x02
+#define SSM2602_ROUT1V 0x03
+#define SSM2602_APANA 0x04
+#define SSM2602_APDIGI 0x05
+#define SSM2602_PWR 0x06
+#define SSM2602_IFACE 0x07
+#define SSM2602_SRATE 0x08
+#define SSM2602_ACTIVE 0x09
+#define SSM2602_RESET 0x0f
+
+/*SSM2602 Codec Register Field definitions
+ *(Mask value to extract the corresponding Register field)
+ */
+
+/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/
+#define LINVOL_LIN_VOL 0x01F /* Left Channel PGA Volume control */
+#define LINVOL_LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */
+#define LINVOL_LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */
+
+/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/
+#define RINVOL_RIN_VOL 0x01F /* Right Channel PGA Volume control */
+#define RINVOL_RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */
+#define RINVOL_RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */
+
+/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/
+#define LOUT1V_LHP_VOL 0x07F /* Left Channel Headphone volume control */
+#define LOUT1V_ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */
+#define LOUT1V_LRHP_BOTH 0x100 /* Left Channel Headphone volume update */
+
+/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/
+#define ROUT1V_RHP_VOL 0x07F /* Right Channel Headphone volume control */
+#define ROUT1V_ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */
+#define ROUT1V_RLHP_BOTH 0x100 /* Right Channel Headphone volume update */
+
+/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/
+#define APANA_ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */
+#define APANA_ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */
+#define APANA_ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */
+#define APANA_ENABLE_BYPASS 0x008 /* Line input bypass to line output */
+#define APANA_SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */
+#define APANA_ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */
+#define APANA_SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */
+#define APANA_ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */
+
+/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/
+#define APDIGI_ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */
+#define APDIGI_DE_EMPHASIS 0x006 /* De-Emphasis Control */
+#define APDIGI_ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */
+#define APDIGI_STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */
+
+/*Power Down Control (SSM2602_REG_POWER)
+ *(1=Enable PowerDown, 0=Disable PowerDown)
+ */
+#define PWR_LINE_IN_PDN 0x001 /* Line Input Power Down */
+#define PWR_MIC_PDN 0x002 /* Microphone Input & Bias Power Down */
+#define PWR_ADC_PDN 0x004 /* ADC Power Down */
+#define PWR_DAC_PDN 0x008 /* DAC Power Down */
+#define PWR_OUT_PDN 0x010 /* Outputs Power Down */
+#define PWR_OSC_PDN 0x020 /* Oscillator Power Down */
+#define PWR_CLK_OUT_PDN 0x040 /* CLKOUT Power Down */
+#define PWR_POWER_OFF 0x080 /* POWEROFF Mode */
+
+/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/
+#define IFACE_IFACE_FORMAT 0x003 /* Digital Audio input format control */
+#define IFACE_AUDIO_DATA_LEN 0x00C /* Audio Data word length control */
+#define IFACE_DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */
+#define IFACE_DAC_LR_SWAP 0x020 /* Swap DAC data control */
+#define IFACE_ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */
+#define IFACE_BCLK_INVERT 0x080 /* Bit Clock Inversion control */
+
+/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/
+#define SRATE_ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */
+#define SRATE_BOS_RATE 0x002 /* Base Over-Sampling rate */
+#define SRATE_SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */
+#define SRATE_CORECLK_DIV2 0x040 /* Core Clock divider select */
+#define SRATE_CLKOUT_DIV2 0x080 /* Clock Out divider select */
+
+/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/
+#define ACTIVE_ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */
+
+/*********************************************************************/
+
+#define SSM2602_CACHEREGNUM 10
+
+#define SSM2602_SYSCLK 0
+#define SSM2602_DAI 0
+
+struct ssm2602_setup_data {
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai ssm2602_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ssm2602;
+
+#endif
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
new file mode 100644
index 0000000..44308da
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -0,0 +1,714 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Notes:
+ * The AIC23 is a driver for a low power stereo audio
+ * codec tlv320aic23
+ *
+ * The machine layer should disable unsupported inputs/outputs by
+ * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+
+#include "tlv320aic23.h"
+
+#define AIC23_VERSION "0.1"
+
+struct tlv320aic23_srate_reg_info {
+ u32 sample_rate;
+ u8 control; /* SR3, SR2, SR1, SR0 and BOSR */
+ u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
+};
+
+/*
+ * AIC23 register cache
+ */
+static const u16 tlv320aic23_reg[] = {
+ 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */
+ 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */
+ 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */
+};
+
+/*
+ * read tlv320aic23 register cache
+ */
+static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
+ *codec, unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write tlv320aic23 register cache
+ */
+static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, u16 value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the tlv320aic23 register space
+ */
+static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+
+ u8 data[2];
+
+ /* TLV320AIC23 has 7 bit address and 9 bits of data
+ * so we need to switch one data bit into reg and rest
+ * of data into val
+ */
+
+ if ((reg < 0 || reg > 9) && (reg != 15)) {
+ printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+ return -1;
+ }
+
+ data[0] = (reg << 1) | (value >> 8 & 0x01);
+ data[1] = value & 0xff;
+
+ tlv320aic23_write_reg_cache(codec, reg, value);
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+
+ printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+ value, reg);
+
+ return -EIO;
+}
+
+static const char *rec_src_text[] = { "Line", "Mic" };
+static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum rec_src_enum =
+ SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Input Select", rec_src_enum);
+
+static const struct soc_enum tlv320aic23_rec_src =
+ SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+static const struct soc_enum tlv320aic23_deemph =
+ SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
+static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
+
+static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 val, reg;
+
+ val = (ucontrol->value.integer.value[0] & 0x07);
+
+ /* linear conversion to userspace
+ * 000 = -6db
+ * 001 = -9db
+ * 010 = -12db
+ * 011 = -18db (Min)
+ * 100 = 0db (Max)
+ */
+ val = (val >= 4) ? 4 : (3 - val);
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
+ tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
+
+ return 0;
+}
+
+static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ u16 val;
+
+ val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
+ val = val >> 6;
+ val = (val >= 4) ? 4 : (3 - val);
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+
+}
+
+#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
+ .put = snd_soc_tlv320aic23_put_volsw, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
+ TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
+ SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
+ SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
+ TLV320AIC23_RINVOL, 7, 1, 0),
+ SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
+ TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
+ SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
+ SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
+ SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
+ 6, 4, 0, sidetone_vol_tlv),
+ SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
+};
+
+/* add non dapm controls */
+static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
+{
+
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&tlv320aic23_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+
+}
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
+ SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
+ SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+ &tlv320aic23_rec_src_mux_controls),
+ SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
+ &tlv320aic23_output_mixer_controls[0],
+ ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
+ SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("LHPOUT"),
+ SND_SOC_DAPM_OUTPUT("RHPOUT"),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+
+ SND_SOC_DAPM_INPUT("LLINEIN"),
+ SND_SOC_DAPM_INPUT("RLINEIN"),
+
+ SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* Output Mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
+
+ /* Outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+
+ /* Inputs */
+ {"Line Input", "NULL", "LLINEIN"},
+ {"Line Input", "NULL", "RLINEIN"},
+
+ {"Mic Input", "NULL", "MICIN"},
+
+ /* input mux */
+ {"Capture Source", "Line", "Line Input"},
+ {"Capture Source", "Mic", "Mic Input"},
+ {"ADC", NULL, "Capture Source"},
+
+};
+
+/* tlv320aic23 related */
+static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
+ {4000, 0x06, 1}, /* 4000 */
+ {8000, 0x06, 0}, /* 8000 */
+ {16000, 0x0C, 1}, /* 16000 */
+ {22050, 0x11, 1}, /* 22050 */
+ {24000, 0x00, 1}, /* 24000 */
+ {32000, 0x0C, 0}, /* 32000 */
+ {44100, 0x11, 0}, /* 44100 */
+ {48000, 0x00, 0}, /* 48000 */
+ {88200, 0x1F, 0}, /* 88200 */
+ {96000, 0x0E, 0}, /* 96000 */
+};
+
+static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+ /* set up audio path interconnects */
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 iface_reg, data;
+ u8 count = 0;
+
+ iface_reg =
+ tlv320aic23_read_reg_cache(codec,
+ TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
+
+ /* Search for the right sample rate */
+ /* Verify what happens if the rate is not supported
+ * now it goes to 96Khz */
+ while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
+ (count < ARRAY_SIZE(srate_reg_info))) {
+ count++;
+ }
+
+ data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
+ (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
+ TLV320AIC23_USB_CLK_ON;
+
+ tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface_reg |= (0x01 << 2);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface_reg |= (0x02 << 2);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface_reg |= (0x03 << 2);
+ break;
+ }
+ tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+ return 0;
+}
+
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* set active */
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+
+ return 0;
+}
+
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* deactivate */
+ if (!codec->active) {
+ udelay(50);
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ }
+}
+
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 reg;
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
+ if (mute)
+ reg |= TLV320AIC23_DACM_MUTE;
+
+ else
+ reg &= ~TLV320AIC23_DACM_MUTE;
+
+ tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
+
+ return 0;
+}
+
+static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface_reg;
+
+ iface_reg =
+ tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface_reg |= TLV320AIC23_MS_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface_reg |= TLV320AIC23_FOR_I2S;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface_reg |= TLV320AIC23_FOR_DSP;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface_reg |= TLV320AIC23_FOR_LJUST;
+ break;
+ default:
+ return -EINVAL;
+
+ }
+
+ tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+ return 0;
+}
+
+static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ switch (freq) {
+ case 12000000:
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* vref/mid, osc on, dac unmute */
+ tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* everything off except vref/vmid, */
+ tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai tlv320aic23_dai = {
+ .name = "tlv320aic23",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .ops = {
+ .prepare = tlv320aic23_pcm_prepare,
+ .hw_params = tlv320aic23_hw_params,
+ .shutdown = tlv320aic23_shutdown,
+ },
+ .dai_ops = {
+ .digital_mute = tlv320aic23_mute,
+ .set_fmt = tlv320aic23_set_dai_fmt,
+ .set_sysclk = tlv320aic23_set_dai_sysclk,
+ }
+};
+EXPORT_SYMBOL_GPL(tlv320aic23_dai);
+
+static int tlv320aic23_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int tlv320aic23_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u16 reg;
+
+ /* Sync reg_cache with the hardware */
+ for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+ u16 val = tlv320aic23_read_reg_cache(codec, reg);
+ tlv320aic23_write(codec, reg, val);
+ }
+
+ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
+
+ return 0;
+}
+
+/*
+ * initialise the AIC23 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int tlv320aic23_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+ u16 reg;
+
+ codec->name = "tlv320aic23";
+ codec->owner = THIS_MODULE;
+ codec->read = tlv320aic23_read_reg_cache;
+ codec->write = tlv320aic23_write;
+ codec->set_bias_level = tlv320aic23_set_bias_level;
+ codec->dai = &tlv320aic23_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
+ codec->reg_cache =
+ kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* Reset codec */
+ tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+
+ /* Unmute input */
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
+ tlv320aic23_write(codec, TLV320AIC23_LINVOL,
+ (reg & (~TLV320AIC23_LIM_MUTED)) |
+ (TLV320AIC23_LRS_ENABLED));
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
+ tlv320aic23_write(codec, TLV320AIC23_RINVOL,
+ (reg & (~TLV320AIC23_LIM_MUTED)) |
+ TLV320AIC23_LRS_ENABLED);
+
+ reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
+ tlv320aic23_write(codec, TLV320AIC23_ANLG,
+ (reg) & (~TLV320AIC23_BYPASS_ON) &
+ (~TLV320AIC23_MICM_MUTED));
+
+ /* Default output volume */
+ tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
+ TLV320AIC23_DEFAULT_OUT_VOL &
+ TLV320AIC23_OUT_VOL_MASK);
+ tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
+ TLV320AIC23_DEFAULT_OUT_VOL &
+ TLV320AIC23_OUT_VOL_MASK);
+
+ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
+
+ tlv320aic23_add_controls(codec);
+ tlv320aic23_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "tlv320aic23: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+static struct snd_soc_device *tlv320aic23_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * If the i2c layer weren't so broken, we could pass this kind of data
+ * around
+ */
+static int tlv320aic23_codec_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *i2c_id)
+{
+ struct snd_soc_device *socdev = tlv320aic23_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+ return -EINVAL;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = tlv320aic23_init(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
+{
+ put_device(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id tlv320aic23_id[] = {
+ {"tlv320aic23", 0},
+ {}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
+
+static struct i2c_driver tlv320aic23_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic23",
+ },
+ .probe = tlv320aic23_codec_probe,
+ .remove = __exit_p(tlv320aic23_i2c_remove),
+ .id_table = tlv320aic23_id,
+};
+
+#endif
+
+static int tlv320aic23_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ tlv320aic23_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ codec->hw_write = (hw_write_t) i2c_master_send;
+ codec->hw_read = NULL;
+ ret = i2c_add_driver(&tlv320aic23_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+#endif
+ return ret;
+}
+
+static int tlv320aic23_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&tlv320aic23_i2c_driver);
+#endif
+ kfree(codec->reg_cache);
+ kfree(codec);
+
+ return 0;
+}
+struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
+ .probe = tlv320aic23_probe,
+ .remove = tlv320aic23_remove,
+ .suspend = tlv320aic23_suspend,
+ .resume = tlv320aic23_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
new file mode 100644
index 0000000..79d1faf
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -0,0 +1,122 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _TLV320AIC23_H
+#define _TLV320AIC23_H
+
+/* Codec TLV320AIC23 */
+#define TLV320AIC23_LINVOL 0x00
+#define TLV320AIC23_RINVOL 0x01
+#define TLV320AIC23_LCHNVOL 0x02
+#define TLV320AIC23_RCHNVOL 0x03
+#define TLV320AIC23_ANLG 0x04
+#define TLV320AIC23_DIGT 0x05
+#define TLV320AIC23_PWR 0x06
+#define TLV320AIC23_DIGT_FMT 0x07
+#define TLV320AIC23_SRATE 0x08
+#define TLV320AIC23_ACTIVE 0x09
+#define TLV320AIC23_RESET 0x0F
+
+/* Left (right) line input volume control register */
+#define TLV320AIC23_LRS_ENABLED 0x0100
+#define TLV320AIC23_LIM_MUTED 0x0080
+#define TLV320AIC23_LIV_DEFAULT 0x0017
+#define TLV320AIC23_LIV_MAX 0x001f
+#define TLV320AIC23_LIV_MIN 0x0000
+
+/* Left (right) channel headphone volume control register */
+#define TLV320AIC23_LZC_ON 0x0080
+#define TLV320AIC23_LHV_DEFAULT 0x0079
+#define TLV320AIC23_LHV_MAX 0x007f
+#define TLV320AIC23_LHV_MIN 0x0000
+
+/* Analog audio path control register */
+#define TLV320AIC23_STA_REG(x) ((x)<<6)
+#define TLV320AIC23_STE_ENABLED 0x0020
+#define TLV320AIC23_DAC_SELECTED 0x0010
+#define TLV320AIC23_BYPASS_ON 0x0008
+#define TLV320AIC23_INSEL_MIC 0x0004
+#define TLV320AIC23_MICM_MUTED 0x0002
+#define TLV320AIC23_MICB_20DB 0x0001
+
+/* Digital audio path control register */
+#define TLV320AIC23_DACM_MUTE 0x0008
+#define TLV320AIC23_DEEMP_32K 0x0002
+#define TLV320AIC23_DEEMP_44K 0x0004
+#define TLV320AIC23_DEEMP_48K 0x0006
+#define TLV320AIC23_ADCHP_ON 0x0001
+
+/* Power control down register */
+#define TLV320AIC23_DEVICE_PWR_OFF 0x0080
+#define TLV320AIC23_CLK_OFF 0x0040
+#define TLV320AIC23_OSC_OFF 0x0020
+#define TLV320AIC23_OUT_OFF 0x0010
+#define TLV320AIC23_DAC_OFF 0x0008
+#define TLV320AIC23_ADC_OFF 0x0004
+#define TLV320AIC23_MIC_OFF 0x0002
+#define TLV320AIC23_LINE_OFF 0x0001
+
+/* Digital audio interface register */
+#define TLV320AIC23_MS_MASTER 0x0040
+#define TLV320AIC23_LRSWAP_ON 0x0020
+#define TLV320AIC23_LRP_ON 0x0010
+#define TLV320AIC23_IWL_16 0x0000
+#define TLV320AIC23_IWL_20 0x0004
+#define TLV320AIC23_IWL_24 0x0008
+#define TLV320AIC23_IWL_32 0x000C
+#define TLV320AIC23_FOR_I2S 0x0002
+#define TLV320AIC23_FOR_DSP 0x0003
+#define TLV320AIC23_FOR_LJUST 0x0001
+
+/* Sample rate control register */
+#define TLV320AIC23_CLKOUT_HALF 0x0080
+#define TLV320AIC23_CLKIN_HALF 0x0040
+#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */
+#define TLV320AIC23_USB_CLK_ON 0x0001
+#define TLV320AIC23_SR_MASK 0xf
+#define TLV320AIC23_CLKOUT_SHIFT 7
+#define TLV320AIC23_CLKIN_SHIFT 6
+#define TLV320AIC23_SR_SHIFT 2
+#define TLV320AIC23_BOSR_SHIFT 1
+
+/* Digital interface register */
+#define TLV320AIC23_ACT_ON 0x0001
+
+/*
+ * AUDIO related MACROS
+ */
+
+#define TLV320AIC23_DEFAULT_OUT_VOL 0x70
+#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10
+
+#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN
+#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX
+#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \
+ TLV320AIC23_OUT_VOL_MIN)
+#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX
+
+#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN
+#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX
+#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \
+ TLV320AIC23_IN_VOL_MIN)
+#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX
+
+#define TLV320AIC23_SIDETONE_MASK 0x1c0
+#define TLV320AIC23_SIDETONE_0 0x100
+#define TLV320AIC23_SIDETONE_6 0x000
+#define TLV320AIC23_SIDETONE_9 0x040
+#define TLV320AIC23_SIDETONE_12 0x080
+#define TLV320AIC23_SIDETONE_18 0x0c0
+
+extern struct snd_soc_dai tlv320aic23_dai;
+extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
+
+#endif /* _TLV320AIC23_H */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
new file mode 100644
index 0000000..bed8a9e
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -0,0 +1,520 @@
+/*
+ * Texas Instruments TLV320AIC26 low power audio CODEC
+ * ALSA SoC CODEC driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/device.h>
+#include <linux/sysfs.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc-of-simple.h>
+#include <sound/initval.h>
+
+#include "tlv320aic26.h"
+
+MODULE_DESCRIPTION("ASoC TLV320AIC26 codec driver");
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_LICENSE("GPL");
+
+/* AIC26 driver private data */
+struct aic26 {
+ struct spi_device *spi;
+ struct snd_soc_codec codec;
+ u16 reg_cache[AIC26_NUM_REGS]; /* shadow registers */
+ int master;
+ int datfm;
+ int mclk;
+
+ /* Keyclick parameters */
+ int keyclick_amplitude;
+ int keyclick_freq;
+ int keyclick_len;
+};
+
+/* ---------------------------------------------------------------------
+ * Register access routines
+ */
+static unsigned int aic26_reg_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct aic26 *aic26 = codec->private_data;
+ u16 *cache = codec->reg_cache;
+ u16 cmd, value;
+ u8 buffer[2];
+ int rc;
+
+ if (reg >= AIC26_NUM_REGS) {
+ WARN_ON_ONCE(1);
+ return 0;
+ }
+
+ /* Do SPI transfer; first 16bits are command; remaining is
+ * register contents */
+ cmd = AIC26_READ_COMMAND_WORD(reg);
+ buffer[0] = (cmd >> 8) & 0xff;
+ buffer[1] = cmd & 0xff;
+ rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2);
+ if (rc) {
+ dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
+ return -EIO;
+ }
+ value = (buffer[0] << 8) | buffer[1];
+
+ /* Update the cache before returning with the value */
+ cache[reg] = value;
+ return value;
+}
+
+static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg >= AIC26_NUM_REGS) {
+ WARN_ON_ONCE(1);
+ return 0;
+ }
+
+ return cache[reg];
+}
+
+static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ struct aic26 *aic26 = codec->private_data;
+ u16 *cache = codec->reg_cache;
+ u16 cmd;
+ u8 buffer[4];
+ int rc;
+
+ if (reg >= AIC26_NUM_REGS) {
+ WARN_ON_ONCE(1);
+ return -EINVAL;
+ }
+
+ /* Do SPI transfer; first 16bits are command; remaining is data
+ * to write into register */
+ cmd = AIC26_WRITE_COMMAND_WORD(reg);
+ buffer[0] = (cmd >> 8) & 0xff;
+ buffer[1] = cmd & 0xff;
+ buffer[2] = value >> 8;
+ buffer[3] = value;
+ rc = spi_write(aic26->spi, buffer, 4);
+ if (rc) {
+ dev_err(&aic26->spi->dev, "AIC26 reg read error\n");
+ return -EIO;
+ }
+
+ /* update cache before returning */
+ cache[reg] = value;
+ return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * Digital Audio Interface Operations
+ */
+static int aic26_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct aic26 *aic26 = codec->private_data;
+ int fsref, divisor, wlen, pval, jval, dval, qval;
+ u16 reg;
+
+ dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
+ substream, params);
+ dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
+ params_format(params));
+
+ switch (params_rate(params)) {
+ case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
+ case 11025: fsref = 44100; divisor = AIC26_DIV_4; break;
+ case 12000: fsref = 48000; divisor = AIC26_DIV_4; break;
+ case 16000: fsref = 48000; divisor = AIC26_DIV_3; break;
+ case 22050: fsref = 44100; divisor = AIC26_DIV_2; break;
+ case 24000: fsref = 48000; divisor = AIC26_DIV_2; break;
+ case 32000: fsref = 48000; divisor = AIC26_DIV_1_5; break;
+ case 44100: fsref = 44100; divisor = AIC26_DIV_1; break;
+ case 48000: fsref = 48000; divisor = AIC26_DIV_1; break;
+ default:
+ dev_dbg(&aic26->spi->dev, "bad rate\n"); return -EINVAL;
+ }
+
+ /* select data word length */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
+ case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
+ case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
+ case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
+ default:
+ dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
+ }
+
+ /* Configure PLL */
+ pval = 1;
+ jval = (fsref == 44100) ? 7 : 8;
+ dval = (fsref == 44100) ? 5264 : 1920;
+ qval = 0;
+ reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
+ aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
+ reg = dval << 2;
+ aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg);
+
+ /* Audio Control 3 (master mode, fsref rate) */
+ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3);
+ reg &= ~0xf800;
+ if (aic26->master)
+ reg |= 0x0800;
+ if (fsref == 48000)
+ reg |= 0x2000;
+ aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg);
+
+ /* Audio Control 1 (FSref divisor) */
+ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1);
+ reg &= ~0x0fff;
+ reg |= wlen | aic26->datfm | (divisor << 3) | divisor;
+ aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg);
+
+ return 0;
+}
+
+/**
+ * aic26_mute - Mute control to reduce noise when changing audio format
+ */
+static int aic26_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct aic26 *aic26 = codec->private_data;
+ u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN);
+
+ dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
+ dai, mute);
+
+ if (mute)
+ reg |= 0x8080;
+ else
+ reg &= ~0x8080;
+ aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg);
+
+ return 0;
+}
+
+static int aic26_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic26 *aic26 = codec->private_data;
+
+ dev_dbg(&aic26->spi->dev, "aic26_set_sysclk(dai=%p, clk_id==%i,"
+ " freq=%i, dir=%i)\n",
+ codec_dai, clk_id, freq, dir);
+
+ /* MCLK needs to fall between 2MHz and 50 MHz */
+ if ((freq < 2000000) || (freq > 50000000))
+ return -EINVAL;
+
+ aic26->mclk = freq;
+ return 0;
+}
+
+static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic26 *aic26 = codec->private_data;
+
+ dev_dbg(&aic26->spi->dev, "aic26_set_fmt(dai=%p, fmt==%i)\n",
+ codec_dai, fmt);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM: aic26->master = 1; break;
+ case SND_SOC_DAIFMT_CBS_CFS: aic26->master = 0; break;
+ default:
+ dev_dbg(&aic26->spi->dev, "bad master\n"); return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S: aic26->datfm = AIC26_DATFM_I2S; break;
+ case SND_SOC_DAIFMT_DSP_A: aic26->datfm = AIC26_DATFM_DSP; break;
+ case SND_SOC_DAIFMT_RIGHT_J: aic26->datfm = AIC26_DATFM_RIGHTJ; break;
+ case SND_SOC_DAIFMT_LEFT_J: aic26->datfm = AIC26_DATFM_LEFTJ; break;
+ default:
+ dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
+ }
+
+ return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * Digital Audio Interface Definition
+ */
+#define AIC26_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
+#define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
+
+struct snd_soc_dai aic26_dai = {
+ .name = "tlv320aic26",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC26_RATES,
+ .formats = AIC26_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC26_RATES,
+ .formats = AIC26_FORMATS,
+ },
+ .ops = {
+ .hw_params = aic26_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = aic26_mute,
+ .set_sysclk = aic26_set_sysclk,
+ .set_fmt = aic26_set_fmt,
+ },
+};
+EXPORT_SYMBOL_GPL(aic26_dai);
+
+/* ---------------------------------------------------------------------
+ * ALSA controls
+ */
+static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
+static const struct soc_enum aic26_capture_src_enum =
+ SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12, 2, aic26_capture_src_text);
+
+static const struct snd_kcontrol_new aic26_snd_controls[] = {
+ /* Output */
+ SOC_DOUBLE("PCM Playback Volume", AIC26_REG_DAC_GAIN, 8, 0, 0x7f, 1),
+ SOC_DOUBLE("PCM Playback Switch", AIC26_REG_DAC_GAIN, 15, 7, 1, 1),
+ SOC_SINGLE("PCM Capture Volume", AIC26_REG_ADC_GAIN, 8, 0x7f, 0),
+ SOC_SINGLE("PCM Capture Mute", AIC26_REG_ADC_GAIN, 15, 1, 1),
+ SOC_SINGLE("Keyclick activate", AIC26_REG_AUDIO_CTRL2, 15, 0x1, 0),
+ SOC_SINGLE("Keyclick amplitude", AIC26_REG_AUDIO_CTRL2, 12, 0x7, 0),
+ SOC_SINGLE("Keyclick frequency", AIC26_REG_AUDIO_CTRL2, 8, 0x7, 0),
+ SOC_SINGLE("Keyclick period", AIC26_REG_AUDIO_CTRL2, 4, 0xf, 0),
+ SOC_ENUM("Capture Source", aic26_capture_src_enum),
+};
+
+/* ---------------------------------------------------------------------
+ * SoC CODEC portion of driver: probe and release routines
+ */
+static int aic26_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct snd_kcontrol *kcontrol;
+ struct aic26 *aic26;
+ int i, ret, err;
+
+ dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
+ dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
+ dev_dbg(&pdev->dev, "codec_data=%p\n", socdev->codec_data);
+
+ /* Fetch the relevant aic26 private data here (it's already been
+ * stored in the .codec pointer) */
+ aic26 = socdev->codec_data;
+ if (aic26 == NULL) {
+ dev_err(&pdev->dev, "aic26: missing codec pointer\n");
+ return -ENODEV;
+ }
+ codec = &aic26->codec;
+ socdev->codec = codec;
+
+ dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
+ &pdev->dev, socdev->dev);
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "aic26: failed to create pcms\n");
+ return -ENODEV;
+ }
+
+ /* register controls */
+ dev_dbg(&pdev->dev, "Registering controls\n");
+ for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
+ kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
+ err = snd_ctl_add(codec->card, kcontrol);
+ WARN_ON(err < 0);
+ }
+
+ /* CODEC is setup, we can register the card now */
+ dev_dbg(&pdev->dev, "Registering card\n");
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ dev_err(&pdev->dev, "aic26: failed to register card\n");
+ goto card_err;
+ }
+ return 0;
+
+ card_err:
+ snd_soc_free_pcms(socdev);
+ return ret;
+}
+
+static int aic26_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ snd_soc_free_pcms(socdev);
+ return 0;
+}
+
+struct snd_soc_codec_device aic26_soc_codec_dev = {
+ .probe = aic26_probe,
+ .remove = aic26_remove,
+};
+EXPORT_SYMBOL_GPL(aic26_soc_codec_dev);
+
+/* ---------------------------------------------------------------------
+ * SPI device portion of driver: sysfs files for debugging
+ */
+
+static ssize_t aic26_keyclick_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct aic26 *aic26 = dev_get_drvdata(dev);
+ int val, amp, freq, len;
+
+ val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+ amp = (val >> 12) & 0x7;
+ freq = (125 << ((val >> 8) & 0x7)) >> 1;
+ len = 2 * (1 + ((val >> 4) & 0xf));
+
+ return sprintf(buf, "amp=%x freq=%iHz len=%iclks\n", amp, freq, len);
+}
+
+/* Any write to the keyclick attribute will trigger the keyclick event */
+static ssize_t aic26_keyclick_set(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf, size_t count)
+{
+ struct aic26 *aic26 = dev_get_drvdata(dev);
+ int val;
+
+ val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2);
+ val |= 0x8000;
+ aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val);
+
+ return count;
+}
+
+static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
+
+/* ---------------------------------------------------------------------
+ * SPI device portion of driver: probe and release routines and SPI
+ * driver registration.
+ */
+static int aic26_spi_probe(struct spi_device *spi)
+{
+ struct aic26 *aic26;
+ int rc, i, reg;
+
+ dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
+
+ /* Allocate driver data */
+ aic26 = kzalloc(sizeof *aic26, GFP_KERNEL);
+ if (!aic26)
+ return -ENOMEM;
+
+ /* Initialize the driver data */
+ aic26->spi = spi;
+ dev_set_drvdata(&spi->dev, aic26);
+
+ /* Setup what we can in the codec structure so that the register
+ * access functions will work as expected. More will be filled
+ * out when it is probed by the SoC CODEC part of this driver */
+ aic26->codec.private_data = aic26;
+ aic26->codec.name = "aic26";
+ aic26->codec.owner = THIS_MODULE;
+ aic26->codec.dai = &aic26_dai;
+ aic26->codec.num_dai = 1;
+ aic26->codec.read = aic26_reg_read;
+ aic26->codec.write = aic26_reg_write;
+ aic26->master = 1;
+ mutex_init(&aic26->codec.mutex);
+ INIT_LIST_HEAD(&aic26->codec.dapm_widgets);
+ INIT_LIST_HEAD(&aic26->codec.dapm_paths);
+ aic26->codec.reg_cache_size = AIC26_NUM_REGS;
+ aic26->codec.reg_cache = aic26->reg_cache;
+
+ /* Reset the codec to power on defaults */
+ aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
+
+ /* Power up CODEC */
+ aic26_reg_write(&aic26->codec, AIC26_REG_POWER_CTRL, 0);
+
+ /* Audio Control 3 (master mode, fsref rate) */
+ reg = aic26_reg_read(&aic26->codec, AIC26_REG_AUDIO_CTRL3);
+ reg &= ~0xf800;
+ reg |= 0x0800; /* set master mode */
+ aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL3, reg);
+
+ /* Fill register cache */
+ for (i = 0; i < ARRAY_SIZE(aic26->reg_cache); i++)
+ aic26_reg_read(&aic26->codec, i);
+
+ /* Register the sysfs files for debugging */
+ /* Create SysFS files */
+ rc = device_create_file(&spi->dev, &dev_attr_keyclick);
+ if (rc)
+ dev_info(&spi->dev, "error creating sysfs files\n");
+
+#if defined(CONFIG_SND_SOC_OF_SIMPLE)
+ /* Tell the of_soc helper about this codec */
+ of_snd_soc_register_codec(&aic26_soc_codec_dev, aic26, &aic26_dai,
+ spi->dev.archdata.of_node);
+#endif
+
+ dev_dbg(&spi->dev, "SPI device initialized\n");
+ return 0;
+}
+
+static int aic26_spi_remove(struct spi_device *spi)
+{
+ struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
+
+ kfree(aic26);
+
+ return 0;
+}
+
+static struct spi_driver aic26_spi = {
+ .driver = {
+ .name = "tlv320aic26",
+ .owner = THIS_MODULE,
+ },
+ .probe = aic26_spi_probe,
+ .remove = aic26_spi_remove,
+};
+
+static int __init aic26_init(void)
+{
+ return spi_register_driver(&aic26_spi);
+}
+module_init(aic26_init);
+
+static void __exit aic26_exit(void)
+{
+ spi_unregister_driver(&aic26_spi);
+}
+module_exit(aic26_exit);
diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
new file mode 100644
index 0000000..786ba16
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic26.h
@@ -0,0 +1,96 @@
+/*
+ * Texas Instruments TLV320AIC26 low power audio CODEC
+ * register definitions
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ */
+
+#ifndef _TLV320AIC16_H_
+#define _TLV320AIC16_H_
+
+/* AIC26 Registers */
+#define AIC26_READ_COMMAND_WORD(addr) ((1 << 15) | (addr << 5))
+#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5))
+#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset)
+#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0)
+
+/* Page 0: Auxillary data registers */
+#define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05)
+#define AIC26_REG_BAT2 AIC26_PAGE_ADDR(0, 0x06)
+#define AIC26_REG_AUX AIC26_PAGE_ADDR(0, 0x07)
+#define AIC26_REG_TEMP1 AIC26_PAGE_ADDR(0, 0x09)
+#define AIC26_REG_TEMP2 AIC26_PAGE_ADDR(0, 0x0A)
+
+/* Page 1: Auxillary control registers */
+#define AIC26_REG_AUX_ADC AIC26_PAGE_ADDR(1, 0x00)
+#define AIC26_REG_STATUS AIC26_PAGE_ADDR(1, 0x01)
+#define AIC26_REG_REFERENCE AIC26_PAGE_ADDR(1, 0x03)
+#define AIC26_REG_RESET AIC26_PAGE_ADDR(1, 0x04)
+
+/* Page 2: Audio control registers */
+#define AIC26_REG_AUDIO_CTRL1 AIC26_PAGE_ADDR(2, 0x00)
+#define AIC26_REG_ADC_GAIN AIC26_PAGE_ADDR(2, 0x01)
+#define AIC26_REG_DAC_GAIN AIC26_PAGE_ADDR(2, 0x02)
+#define AIC26_REG_SIDETONE AIC26_PAGE_ADDR(2, 0x03)
+#define AIC26_REG_AUDIO_CTRL2 AIC26_PAGE_ADDR(2, 0x04)
+#define AIC26_REG_POWER_CTRL AIC26_PAGE_ADDR(2, 0x05)
+#define AIC26_REG_AUDIO_CTRL3 AIC26_PAGE_ADDR(2, 0x06)
+
+#define AIC26_REG_FILTER_COEFF_L_N0 AIC26_PAGE_ADDR(2, 0x07)
+#define AIC26_REG_FILTER_COEFF_L_N1 AIC26_PAGE_ADDR(2, 0x08)
+#define AIC26_REG_FILTER_COEFF_L_N2 AIC26_PAGE_ADDR(2, 0x09)
+#define AIC26_REG_FILTER_COEFF_L_N3 AIC26_PAGE_ADDR(2, 0x0A)
+#define AIC26_REG_FILTER_COEFF_L_N4 AIC26_PAGE_ADDR(2, 0x0B)
+#define AIC26_REG_FILTER_COEFF_L_N5 AIC26_PAGE_ADDR(2, 0x0C)
+#define AIC26_REG_FILTER_COEFF_L_D1 AIC26_PAGE_ADDR(2, 0x0D)
+#define AIC26_REG_FILTER_COEFF_L_D2 AIC26_PAGE_ADDR(2, 0x0E)
+#define AIC26_REG_FILTER_COEFF_L_D4 AIC26_PAGE_ADDR(2, 0x0F)
+#define AIC26_REG_FILTER_COEFF_L_D5 AIC26_PAGE_ADDR(2, 0x10)
+#define AIC26_REG_FILTER_COEFF_R_N0 AIC26_PAGE_ADDR(2, 0x11)
+#define AIC26_REG_FILTER_COEFF_R_N1 AIC26_PAGE_ADDR(2, 0x12)
+#define AIC26_REG_FILTER_COEFF_R_N2 AIC26_PAGE_ADDR(2, 0x13)
+#define AIC26_REG_FILTER_COEFF_R_N3 AIC26_PAGE_ADDR(2, 0x14)
+#define AIC26_REG_FILTER_COEFF_R_N4 AIC26_PAGE_ADDR(2, 0x15)
+#define AIC26_REG_FILTER_COEFF_R_N5 AIC26_PAGE_ADDR(2, 0x16)
+#define AIC26_REG_FILTER_COEFF_R_D1 AIC26_PAGE_ADDR(2, 0x17)
+#define AIC26_REG_FILTER_COEFF_R_D2 AIC26_PAGE_ADDR(2, 0x18)
+#define AIC26_REG_FILTER_COEFF_R_D4 AIC26_PAGE_ADDR(2, 0x19)
+#define AIC26_REG_FILTER_COEFF_R_D5 AIC26_PAGE_ADDR(2, 0x1A)
+
+#define AIC26_REG_PLL_PROG1 AIC26_PAGE_ADDR(2, 0x1B)
+#define AIC26_REG_PLL_PROG2 AIC26_PAGE_ADDR(2, 0x1C)
+#define AIC26_REG_AUDIO_CTRL4 AIC26_PAGE_ADDR(2, 0x1D)
+#define AIC26_REG_AUDIO_CTRL5 AIC26_PAGE_ADDR(2, 0x1E)
+
+/* fsref dividers; used in register 'Audio Control 1' */
+enum aic26_divisors {
+ AIC26_DIV_1 = 0,
+ AIC26_DIV_1_5 = 1,
+ AIC26_DIV_2 = 2,
+ AIC26_DIV_3 = 3,
+ AIC26_DIV_4 = 4,
+ AIC26_DIV_5 = 5,
+ AIC26_DIV_5_5 = 6,
+ AIC26_DIV_6 = 7,
+};
+
+/* Digital data format */
+enum aic26_datfm {
+ AIC26_DATFM_I2S = 0 << 8,
+ AIC26_DATFM_DSP = 1 << 8,
+ AIC26_DATFM_RIGHTJ = 2 << 8, /* right justified */
+ AIC26_DATFM_LEFTJ = 3 << 8, /* left justified */
+};
+
+/* Sample word length in bits; used in register 'Audio Control 1' */
+enum aic26_wlen {
+ AIC26_WLEN_16 = 0 << 10,
+ AIC26_WLEN_20 = 1 << 10,
+ AIC26_WLEN_24 = 2 << 10,
+ AIC26_WLEN_32 = 3 << 10,
+};
+
+extern struct snd_soc_dai aic26_dai;
+extern struct snd_soc_codec_device aic26_soc_codec_dev;
+
+#endif /* _TLV320AIC16_H_ */
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
new file mode 100644
index 0000000..cff276e
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -0,0 +1,1346 @@
+/*
+ * ALSA SoC TLV320AIC3X codec driver
+ *
+ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * Based on sound/soc/codecs/wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Notes:
+ * The AIC3X is a driver for a low power stereo audio
+ * codecs aic31, aic32, aic33.
+ *
+ * It supports full aic33 codec functionality.
+ * The compatibility with aic32, aic31 is as follows:
+ * aic32 | aic31
+ * ---------------------------------------
+ * MONO_LOUT -> N/A | MONO_LOUT -> N/A
+ * | IN1L -> LINE1L
+ * | IN1R -> LINE1R
+ * | IN2L -> LINE2L
+ * | IN2R -> LINE2R
+ * | MIC3L/R -> N/A
+ * truncated internal functionality in
+ * accordance with documentation
+ * ---------------------------------------
+ *
+ * Hence the machine layer should disable unsupported inputs/outputs by
+ * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "tlv320aic3x.h"
+
+#define AIC3X_VERSION "0.2"
+
+/* codec private data */
+struct aic3x_priv {
+ unsigned int sysclk;
+ int master;
+};
+
+/*
+ * AIC3X register cache
+ * We can't read the AIC3X register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = {
+ 0x00, 0x00, 0x00, 0x10, /* 0 */
+ 0x04, 0x00, 0x00, 0x00, /* 4 */
+ 0x00, 0x00, 0x00, 0x01, /* 8 */
+ 0x00, 0x00, 0x00, 0x80, /* 12 */
+ 0x80, 0xff, 0xff, 0x78, /* 16 */
+ 0x78, 0x78, 0x78, 0x78, /* 20 */
+ 0x78, 0x00, 0x00, 0xfe, /* 24 */
+ 0x00, 0x00, 0xfe, 0x00, /* 28 */
+ 0x18, 0x18, 0x00, 0x00, /* 32 */
+ 0x00, 0x00, 0x00, 0x00, /* 36 */
+ 0x00, 0x00, 0x00, 0x80, /* 40 */
+ 0x80, 0x00, 0x00, 0x00, /* 44 */
+ 0x00, 0x00, 0x00, 0x04, /* 48 */
+ 0x00, 0x00, 0x00, 0x00, /* 52 */
+ 0x00, 0x00, 0x04, 0x00, /* 56 */
+ 0x00, 0x00, 0x00, 0x00, /* 60 */
+ 0x00, 0x04, 0x00, 0x00, /* 64 */
+ 0x00, 0x00, 0x00, 0x00, /* 68 */
+ 0x04, 0x00, 0x00, 0x00, /* 72 */
+ 0x00, 0x00, 0x00, 0x00, /* 76 */
+ 0x00, 0x00, 0x00, 0x00, /* 80 */
+ 0x00, 0x00, 0x00, 0x00, /* 84 */
+ 0x00, 0x00, 0x00, 0x00, /* 88 */
+ 0x00, 0x00, 0x00, 0x00, /* 92 */
+ 0x00, 0x00, 0x00, 0x00, /* 96 */
+ 0x00, 0x00, 0x02, /* 100 */
+};
+
+/*
+ * read aic3x register cache
+ */
+static inline unsigned int aic3x_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u8 *cache = codec->reg_cache;
+ if (reg >= AIC3X_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write aic3x register cache
+ */
+static inline void aic3x_write_reg_cache(struct snd_soc_codec *codec,
+ u8 reg, u8 value)
+{
+ u8 *cache = codec->reg_cache;
+ if (reg >= AIC3X_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the aic3x register space
+ */
+static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D8 aic3x register offset
+ * D7...D0 register data
+ */
+ data[0] = reg & 0xff;
+ data[1] = value & 0xff;
+
+ aic3x_write_reg_cache(codec, data[0], data[1]);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+/*
+ * read from the aic3x register space
+ */
+static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
+ u8 *value)
+{
+ *value = reg & 0xff;
+ if (codec->hw_read(codec->control_data, value, 1) != 1)
+ return -EIO;
+
+ aic3x_write_reg_cache(codec, reg, *value);
+ return 0;
+}
+
+#define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw_aic3x, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) }
+
+/*
+ * All input lines are connected when !0xf and disconnected with 0xf bit field,
+ * so we have to use specific dapm_put call for input mixer
+ */
+static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0x0f;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0x01;
+ unsigned short val, val_mask;
+ int ret;
+ struct snd_soc_dapm_path *path;
+ int found = 0;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+
+ mask = 0xf;
+ if (val)
+ val = mask;
+
+ if (invert)
+ val = mask - val;
+ val_mask = mask << shift;
+ val = val << shift;
+
+ mutex_lock(&widget->codec->mutex);
+
+ if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) {
+ /* find dapm widget path assoc with kcontrol */
+ list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ if (path->kcontrol != kcontrol)
+ continue;
+
+ /* found, now check type */
+ found = 1;
+ if (val)
+ /* new connection */
+ path->connect = invert ? 0 : 1;
+ else
+ /* old connection must be powered down */
+ path->connect = invert ? 1 : 0;
+ break;
+ }
+
+ if (found)
+ snd_soc_dapm_sync(widget->codec);
+ }
+
+ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
+
+ mutex_unlock(&widget->codec->mutex);
+ return ret;
+}
+
+static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" };
+static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" };
+static const char *aic3x_left_hpcom_mux[] =
+ { "differential of HPLOUT", "constant VCM", "single-ended" };
+static const char *aic3x_right_hpcom_mux[] =
+ { "differential of HPROUT", "constant VCM", "single-ended",
+ "differential of HPLCOM", "external feedback" };
+static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
+static const char *aic3x_adc_hpf[] =
+ { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
+
+#define LDAC_ENUM 0
+#define RDAC_ENUM 1
+#define LHPCOM_ENUM 2
+#define RHPCOM_ENUM 3
+#define LINE1L_ENUM 4
+#define LINE1R_ENUM 5
+#define LINE2L_ENUM 6
+#define LINE2R_ENUM 7
+#define ADC_HPF_ENUM 8
+
+static const struct soc_enum aic3x_enum[] = {
+ SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
+ SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux),
+ SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux),
+ SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux),
+ SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+ SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+ SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+ SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+ SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
+};
+
+static const struct snd_kcontrol_new aic3x_snd_controls[] = {
+ /* Output */
+ SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1),
+
+ SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
+ DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
+ 0x01, 0),
+ SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
+ PGAR_2_RLOPM_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+ LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
+
+ SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
+ DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1),
+ SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
+ SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL,
+ PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL,
+ LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1),
+
+ SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL,
+ DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
+ 0x01, 0),
+ SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+ PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
+ LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
+
+ SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL,
+ DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
+ 0x01, 0),
+ SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
+ PGAR_2_HPRCOM_VOL, 0, 0x7f, 1),
+ SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
+ LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
+
+ /*
+ * Note: enable Automatic input Gain Controller with care. It can
+ * adjust PGA to max value when ADC is on and will never go back.
+ */
+ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
+
+ /* Input */
+ SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0),
+ SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
+
+ SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
+};
+
+/* add non dapm controls */
+static int aic3x_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&aic3x_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Left DAC Mux */
+static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
+
+/* Right DAC Mux */
+static const struct snd_kcontrol_new aic3x_right_dac_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]);
+
+/* Left HPCOM Mux */
+static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]);
+
+/* Right HPCOM Mux */
+static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]);
+
+/* Left DAC_L1 Mixer */
+static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0),
+};
+
+/* Right DAC_R1 Mixer */
+static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0),
+};
+
+/* Left PGA Mixer */
+static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = {
+ SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1),
+};
+
+/* Right PGA Mixer */
+static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
+ SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1),
+ SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1),
+};
+
+/* Left Line1 Mux */
+static const struct snd_kcontrol_new aic3x_left_line1_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_ENUM]);
+
+/* Right Line1 Mux */
+static const struct snd_kcontrol_new aic3x_right_line1_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_ENUM]);
+
+/* Left Line2 Mux */
+static const struct snd_kcontrol_new aic3x_left_line2_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]);
+
+/* Right Line2 Mux */
+static const struct snd_kcontrol_new aic3x_right_line2_mux_controls =
+SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]);
+
+/* Left PGA Bypass Mixer */
+static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+};
+
+/* Right PGA Bypass Mixer */
+static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
+};
+
+/* Left Line2 Bypass Mixer */
+static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
+};
+
+/* Right Line2 Bypass Mixer */
+static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0),
+ SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
+ /* Left DAC to Left Outputs */
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0),
+ SND_SOC_DAPM_MUX("Left DAC Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_dac_mux_controls),
+ SND_SOC_DAPM_MIXER("Left DAC_L1 Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_dac_mixer_controls[0],
+ ARRAY_SIZE(aic3x_left_dac_mixer_controls)),
+ SND_SOC_DAPM_MUX("Left HPCOM Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_hpcom_mux_controls),
+ SND_SOC_DAPM_PGA("Left Line Out", LLOPM_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left HP Out", HPLOUT_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left HP Com", HPLCOM_CTRL, 0, 0, NULL, 0),
+
+ /* Right DAC to Right Outputs */
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", DAC_PWR, 6, 0),
+ SND_SOC_DAPM_MUX("Right DAC Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_dac_mux_controls),
+ SND_SOC_DAPM_MIXER("Right DAC_R1 Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_dac_mixer_controls[0],
+ ARRAY_SIZE(aic3x_right_dac_mixer_controls)),
+ SND_SOC_DAPM_MUX("Right HPCOM Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_hpcom_mux_controls),
+ SND_SOC_DAPM_PGA("Right Line Out", RLOPM_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0),
+
+ /* Mono Output */
+ SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
+
+ /* Left Inputs to Left ADC */
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
+ SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_pga_mixer_controls[0],
+ ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
+ SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_line1_mux_controls),
+ SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_line2_mux_controls),
+
+ /* Right Inputs to Right ADC */
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
+ LINE1R_2_RADC_CTRL, 2, 0),
+ SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_pga_mixer_controls[0],
+ ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
+ SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_line1_mux_controls),
+ SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_line2_mux_controls),
+
+ /*
+ * Not a real mic bias widget but similar function. This is for dynamic
+ * control of GPIO1 digital mic modulator clock output function when
+ * using digital mic.
+ */
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk",
+ AIC3X_GPIO1_REG, 4, 0xf,
+ AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK,
+ AIC3X_GPIO1_FUNC_DISABLED),
+
+ /*
+ * Also similar function like mic bias. Selects digital mic with
+ * configurable oversampling rate instead of ADC converter.
+ */
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128",
+ AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64",
+ AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32",
+ AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0),
+
+ /* Mic Bias */
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V",
+ MICBIAS_CTRL, 6, 3, 1, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V",
+ MICBIAS_CTRL, 6, 3, 2, 0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD",
+ MICBIAS_CTRL, 6, 3, 3, 0),
+
+ /* Left PGA to Left Output bypass */
+ SND_SOC_DAPM_MIXER("Left PGA Bypass Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_pga_bp_mixer_controls[0],
+ ARRAY_SIZE(aic3x_left_pga_bp_mixer_controls)),
+
+ /* Right PGA to Right Output bypass */
+ SND_SOC_DAPM_MIXER("Right PGA Bypass Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_pga_bp_mixer_controls[0],
+ ARRAY_SIZE(aic3x_right_pga_bp_mixer_controls)),
+
+ /* Left Line2 to Left Output bypass */
+ SND_SOC_DAPM_MIXER("Left Line2 Bypass Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_left_line2_bp_mixer_controls[0],
+ ARRAY_SIZE(aic3x_left_line2_bp_mixer_controls)),
+
+ /* Right Line2 to Right Output bypass */
+ SND_SOC_DAPM_MIXER("Right Line2 Bypass Mixer", SND_SOC_NOPM, 0, 0,
+ &aic3x_right_line2_bp_mixer_controls[0],
+ ARRAY_SIZE(aic3x_right_line2_bp_mixer_controls)),
+
+ SND_SOC_DAPM_OUTPUT("LLOUT"),
+ SND_SOC_DAPM_OUTPUT("RLOUT"),
+ SND_SOC_DAPM_OUTPUT("MONO_LOUT"),
+ SND_SOC_DAPM_OUTPUT("HPLOUT"),
+ SND_SOC_DAPM_OUTPUT("HPROUT"),
+ SND_SOC_DAPM_OUTPUT("HPLCOM"),
+ SND_SOC_DAPM_OUTPUT("HPRCOM"),
+
+ SND_SOC_DAPM_INPUT("MIC3L"),
+ SND_SOC_DAPM_INPUT("MIC3R"),
+ SND_SOC_DAPM_INPUT("LINE1L"),
+ SND_SOC_DAPM_INPUT("LINE1R"),
+ SND_SOC_DAPM_INPUT("LINE2L"),
+ SND_SOC_DAPM_INPUT("LINE2R"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* Left Output */
+ {"Left DAC Mux", "DAC_L1", "Left DAC"},
+ {"Left DAC Mux", "DAC_L2", "Left DAC"},
+ {"Left DAC Mux", "DAC_L3", "Left DAC"},
+
+ {"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"},
+ {"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"},
+ {"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"},
+ {"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"},
+ {"Left Line Out", NULL, "Left DAC Mux"},
+ {"Left HP Out", NULL, "Left DAC Mux"},
+
+ {"Left HPCOM Mux", "differential of HPLOUT", "Left DAC_L1 Mixer"},
+ {"Left HPCOM Mux", "constant VCM", "Left DAC_L1 Mixer"},
+ {"Left HPCOM Mux", "single-ended", "Left DAC_L1 Mixer"},
+
+ {"Left Line Out", NULL, "Left DAC_L1 Mixer"},
+ {"Mono Out", NULL, "Left DAC_L1 Mixer"},
+ {"Left HP Out", NULL, "Left DAC_L1 Mixer"},
+ {"Left HP Com", NULL, "Left HPCOM Mux"},
+
+ {"LLOUT", NULL, "Left Line Out"},
+ {"LLOUT", NULL, "Left Line Out"},
+ {"HPLOUT", NULL, "Left HP Out"},
+ {"HPLCOM", NULL, "Left HP Com"},
+
+ /* Right Output */
+ {"Right DAC Mux", "DAC_R1", "Right DAC"},
+ {"Right DAC Mux", "DAC_R2", "Right DAC"},
+ {"Right DAC Mux", "DAC_R3", "Right DAC"},
+
+ {"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"},
+ {"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"},
+ {"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"},
+ {"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"},
+ {"Right Line Out", NULL, "Right DAC Mux"},
+ {"Right HP Out", NULL, "Right DAC Mux"},
+
+ {"Right HPCOM Mux", "differential of HPROUT", "Right DAC_R1 Mixer"},
+ {"Right HPCOM Mux", "constant VCM", "Right DAC_R1 Mixer"},
+ {"Right HPCOM Mux", "single-ended", "Right DAC_R1 Mixer"},
+ {"Right HPCOM Mux", "differential of HPLCOM", "Right DAC_R1 Mixer"},
+ {"Right HPCOM Mux", "external feedback", "Right DAC_R1 Mixer"},
+
+ {"Right Line Out", NULL, "Right DAC_R1 Mixer"},
+ {"Mono Out", NULL, "Right DAC_R1 Mixer"},
+ {"Right HP Out", NULL, "Right DAC_R1 Mixer"},
+ {"Right HP Com", NULL, "Right HPCOM Mux"},
+
+ {"RLOUT", NULL, "Right Line Out"},
+ {"RLOUT", NULL, "Right Line Out"},
+ {"HPROUT", NULL, "Right HP Out"},
+ {"HPRCOM", NULL, "Right HP Com"},
+
+ /* Mono Output */
+ {"MONO_LOUT", NULL, "Mono Out"},
+ {"MONO_LOUT", NULL, "Mono Out"},
+
+ /* Left Input */
+ {"Left Line1L Mux", "single-ended", "LINE1L"},
+ {"Left Line1L Mux", "differential", "LINE1L"},
+
+ {"Left Line2L Mux", "single-ended", "LINE2L"},
+ {"Left Line2L Mux", "differential", "LINE2L"},
+
+ {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
+ {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
+ {"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
+
+ {"Left ADC", NULL, "Left PGA Mixer"},
+ {"Left ADC", NULL, "GPIO1 dmic modclk"},
+
+ /* Right Input */
+ {"Right Line1R Mux", "single-ended", "LINE1R"},
+ {"Right Line1R Mux", "differential", "LINE1R"},
+
+ {"Right Line2R Mux", "single-ended", "LINE2R"},
+ {"Right Line2R Mux", "differential", "LINE2R"},
+
+ {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"},
+ {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"},
+ {"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
+
+ {"Right ADC", NULL, "Right PGA Mixer"},
+ {"Right ADC", NULL, "GPIO1 dmic modclk"},
+
+ /* Left PGA Bypass */
+ {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"},
+ {"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"},
+
+ {"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"},
+ {"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"},
+ {"Left HPCOM Mux", "single-ended", "Left PGA Bypass Mixer"},
+
+ {"Left Line Out", NULL, "Left PGA Bypass Mixer"},
+ {"Mono Out", NULL, "Left PGA Bypass Mixer"},
+ {"Left HP Out", NULL, "Left PGA Bypass Mixer"},
+
+ /* Right PGA Bypass */
+ {"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"},
+ {"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"},
+
+ {"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"},
+ {"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"},
+ {"Right HPCOM Mux", "single-ended", "Right PGA Bypass Mixer"},
+ {"Right HPCOM Mux", "differential of HPLCOM", "Right PGA Bypass Mixer"},
+ {"Right HPCOM Mux", "external feedback", "Right PGA Bypass Mixer"},
+
+ {"Right Line Out", NULL, "Right PGA Bypass Mixer"},
+ {"Mono Out", NULL, "Right PGA Bypass Mixer"},
+ {"Right HP Out", NULL, "Right PGA Bypass Mixer"},
+
+ /* Left Line2 Bypass */
+ {"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"},
+ {"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"},
+
+ {"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"},
+ {"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"},
+ {"Left HPCOM Mux", "single-ended", "Left Line2 Bypass Mixer"},
+
+ {"Left Line Out", NULL, "Left Line2 Bypass Mixer"},
+ {"Mono Out", NULL, "Left Line2 Bypass Mixer"},
+ {"Left HP Out", NULL, "Left Line2 Bypass Mixer"},
+
+ /* Right Line2 Bypass */
+ {"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"},
+ {"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"},
+
+ {"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"},
+ {"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"},
+ {"Right HPCOM Mux", "single-ended", "Right Line2 Bypass Mixer"},
+ {"Right HPCOM Mux", "differential of HPLCOM", "Right Line2 Bypass Mixer"},
+ {"Right HPCOM Mux", "external feedback", "Right Line2 Bypass Mixer"},
+
+ {"Right Line Out", NULL, "Right Line2 Bypass Mixer"},
+ {"Mono Out", NULL, "Right Line2 Bypass Mixer"},
+ {"Right HP Out", NULL, "Right Line2 Bypass Mixer"},
+
+ /*
+ * Logical path between digital mic enable and GPIO1 modulator clock
+ * output function
+ */
+ {"GPIO1 dmic modclk", NULL, "DMic Rate 128"},
+ {"GPIO1 dmic modclk", NULL, "DMic Rate 64"},
+ {"GPIO1 dmic modclk", NULL, "DMic Rate 32"},
+};
+
+static int aic3x_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+ ARRAY_SIZE(aic3x_dapm_widgets));
+
+ /* set up audio path interconnects */
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int aic3x_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct aic3x_priv *aic3x = codec->private_data;
+ int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
+ u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
+ u16 pll_d = 1;
+
+ /* select data word length */
+ data =
+ aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ data |= (0x01 << 4);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ data |= (0x02 << 4);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ data |= (0x03 << 4);
+ break;
+ }
+ aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data);
+
+ /* Fsref can be 44100 or 48000 */
+ fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000;
+
+ /* Try to find a value for Q which allows us to bypass the PLL and
+ * generate CODEC_CLK directly. */
+ for (pll_q = 2; pll_q < 18; pll_q++)
+ if (aic3x->sysclk / (128 * pll_q) == fsref) {
+ bypass_pll = 1;
+ break;
+ }
+
+ if (bypass_pll) {
+ pll_q &= 0xf;
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
+ aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
+ } else
+ aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+
+ /* Route Left DAC to left channel input and
+ * right DAC to right channel input */
+ data = (LDAC2LCH | RDAC2RCH);
+ data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000;
+ if (params_rate(params) >= 64000)
+ data |= DUAL_RATE_MODE;
+ aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
+
+ /* codec sample rate select */
+ data = (fsref * 20) / params_rate(params);
+ if (params_rate(params) < 64000)
+ data /= 2;
+ data /= 5;
+ data -= 2;
+ data |= (data << 4);
+ aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data);
+
+ if (bypass_pll)
+ return 0;
+
+ /* Use PLL
+ * find an apropriate setup for j, d, r and p by iterating over
+ * p and r - j and d are calculated for each fraction.
+ * Up to 128 values are probed, the closest one wins the game.
+ * The sysclk is divided by 1000 to prevent integer overflows.
+ */
+ codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000);
+
+ for (r = 1; r <= 16; r++)
+ for (p = 1; p <= 8; p++) {
+ int clk, tmp = (codec_clk * pll_r * 10) / pll_p;
+ u8 j = tmp / 10000;
+ u16 d = tmp % 10000;
+
+ if (j > 63)
+ continue;
+
+ if (d != 0 && aic3x->sysclk < 10000000)
+ continue;
+
+ /* This is actually 1000 * ((j + (d/10000)) * r) / p
+ * The term had to be converted to get rid of the
+ * division by 10000 */
+ clk = ((10000 * j * r) + (d * r)) / (10 * p);
+
+ /* check whether this values get closer than the best
+ * ones we had before */
+ if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) {
+ pll_j = j; pll_d = d; pll_r = r; pll_p = p;
+ last_clk = clk;
+ }
+
+ /* Early exit for exact matches */
+ if (clk == codec_clk)
+ break;
+ }
+
+ if (last_clk == 0) {
+ printk(KERN_ERR "%s(): unable to setup PLL\n", __func__);
+ return -EINVAL;
+ }
+
+ data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT));
+ aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT);
+ aic3x_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT);
+ aic3x_write(codec, AIC3X_PLL_PROGC_REG, (pll_d >> 6) << PLLD_MSB_SHIFT);
+ aic3x_write(codec, AIC3X_PLL_PROGD_REG,
+ (pll_d & 0x3F) << PLLD_LSB_SHIFT);
+
+ return 0;
+}
+
+static int aic3x_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON;
+ u8 rdac_reg = aic3x_read_reg_cache(codec, RDAC_VOL) & ~MUTE_ON;
+
+ if (mute) {
+ aic3x_write(codec, LDAC_VOL, ldac_reg | MUTE_ON);
+ aic3x_write(codec, RDAC_VOL, rdac_reg | MUTE_ON);
+ } else {
+ aic3x_write(codec, LDAC_VOL, ldac_reg);
+ aic3x_write(codec, RDAC_VOL, rdac_reg);
+ }
+
+ return 0;
+}
+
+static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic3x_priv *aic3x = codec->private_data;
+
+ aic3x->sysclk = freq;
+ return 0;
+}
+
+static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic3x_priv *aic3x = codec->private_data;
+ u8 iface_areg, iface_breg;
+
+ iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
+ iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aic3x->master = 1;
+ iface_areg |= BIT_CLK_MASTER | WORD_CLK_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aic3x->master = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /*
+ * match both interface format and signal polarities since they
+ * are fixed
+ */
+ switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+ SND_SOC_DAIFMT_INV_MASK)) {
+ case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
+ break;
+ case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
+ iface_breg |= (0x01 << 6);
+ break;
+ case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
+ iface_breg |= (0x02 << 6);
+ break;
+ case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
+ iface_breg |= (0x03 << 6);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
+ aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
+
+ return 0;
+}
+
+static int aic3x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct aic3x_priv *aic3x = codec->private_data;
+ u8 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* all power is driven by DAPM system */
+ if (aic3x->master) {
+ /* enable pll */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG,
+ reg | PLL_ENABLE);
+ }
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /*
+ * all power is driven by DAPM system,
+ * so output power is safe if bypass was set
+ */
+ if (aic3x->master) {
+ /* disable pll */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG,
+ reg & ~PLL_ENABLE);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* force all power off */
+ reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL);
+ aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON);
+ reg = aic3x_read_reg_cache(codec, LINE1R_2_RADC_CTRL);
+ aic3x_write(codec, LINE1R_2_RADC_CTRL, reg & ~RADC_PWR_ON);
+
+ reg = aic3x_read_reg_cache(codec, DAC_PWR);
+ aic3x_write(codec, DAC_PWR, reg & ~(LDAC_PWR_ON | RDAC_PWR_ON));
+
+ reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL);
+ aic3x_write(codec, HPLOUT_CTRL, reg & ~HPLOUT_PWR_ON);
+ reg = aic3x_read_reg_cache(codec, HPROUT_CTRL);
+ aic3x_write(codec, HPROUT_CTRL, reg & ~HPROUT_PWR_ON);
+
+ reg = aic3x_read_reg_cache(codec, HPLCOM_CTRL);
+ aic3x_write(codec, HPLCOM_CTRL, reg & ~HPLCOM_PWR_ON);
+ reg = aic3x_read_reg_cache(codec, HPRCOM_CTRL);
+ aic3x_write(codec, HPRCOM_CTRL, reg & ~HPRCOM_PWR_ON);
+
+ reg = aic3x_read_reg_cache(codec, MONOLOPM_CTRL);
+ aic3x_write(codec, MONOLOPM_CTRL, reg & ~MONOLOPM_PWR_ON);
+
+ reg = aic3x_read_reg_cache(codec, LLOPM_CTRL);
+ aic3x_write(codec, LLOPM_CTRL, reg & ~LLOPM_PWR_ON);
+ reg = aic3x_read_reg_cache(codec, RLOPM_CTRL);
+ aic3x_write(codec, RLOPM_CTRL, reg & ~RLOPM_PWR_ON);
+
+ if (aic3x->master) {
+ /* disable pll */
+ reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
+ aic3x_write(codec, AIC3X_PLL_PROGA_REG,
+ reg & ~PLL_ENABLE);
+ }
+ break;
+ }
+ codec->bias_level = level;
+
+ return 0;
+}
+
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state)
+{
+ u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+ u8 bit = gpio ? 3: 0;
+ u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit);
+ aic3x_write(codec, reg, val | (!!state << bit));
+}
+EXPORT_SYMBOL_GPL(aic3x_set_gpio);
+
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
+{
+ u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+ u8 val, bit = gpio ? 2: 1;
+
+ aic3x_read(codec, reg, &val);
+ return (val >> bit) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_get_gpio);
+
+int aic3x_headset_detected(struct snd_soc_codec *codec)
+{
+ u8 val;
+ aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
+ return (val >> 2) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_headset_detected);
+
+#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
+#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai aic3x_dai = {
+ .name = "tlv320aic3x",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AIC3X_RATES,
+ .formats = AIC3X_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AIC3X_RATES,
+ .formats = AIC3X_FORMATS,},
+ .ops = {
+ .hw_params = aic3x_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = aic3x_mute,
+ .set_sysclk = aic3x_set_dai_sysclk,
+ .set_fmt = aic3x_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(aic3x_dai);
+
+static int aic3x_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int aic3x_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u8 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++) {
+ data[0] = i;
+ data[1] = cache[i];
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ aic3x_set_bias_level(codec, codec->suspend_bias_level);
+
+ return 0;
+}
+
+/*
+ * initialise the AIC3X driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int aic3x_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct aic3x_setup_data *setup = socdev->codec_data;
+ int reg, ret = 0;
+
+ codec->name = "tlv320aic3x";
+ codec->owner = THIS_MODULE;
+ codec->read = aic3x_read_reg_cache;
+ codec->write = aic3x_write;
+ codec->set_bias_level = aic3x_set_bias_level;
+ codec->dai = &aic3x_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(aic3x_reg);
+ codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT);
+ aic3x_write(codec, AIC3X_RESET, SOFT_RESET);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "aic3x: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* DAC default volume and mute */
+ aic3x_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON);
+ aic3x_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON);
+
+ /* DAC to HP default volume and route to Output mixer */
+ aic3x_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON);
+ aic3x_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON);
+ aic3x_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON);
+ aic3x_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON);
+ /* DAC to Line Out default volume and route to Output mixer */
+ aic3x_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
+ aic3x_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
+ /* DAC to Mono Line Out default volume and route to Output mixer */
+ aic3x_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
+ aic3x_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
+
+ /* unmute all outputs */
+ reg = aic3x_read_reg_cache(codec, LLOPM_CTRL);
+ aic3x_write(codec, LLOPM_CTRL, reg | UNMUTE);
+ reg = aic3x_read_reg_cache(codec, RLOPM_CTRL);
+ aic3x_write(codec, RLOPM_CTRL, reg | UNMUTE);
+ reg = aic3x_read_reg_cache(codec, MONOLOPM_CTRL);
+ aic3x_write(codec, MONOLOPM_CTRL, reg | UNMUTE);
+ reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL);
+ aic3x_write(codec, HPLOUT_CTRL, reg | UNMUTE);
+ reg = aic3x_read_reg_cache(codec, HPROUT_CTRL);
+ aic3x_write(codec, HPROUT_CTRL, reg | UNMUTE);
+ reg = aic3x_read_reg_cache(codec, HPLCOM_CTRL);
+ aic3x_write(codec, HPLCOM_CTRL, reg | UNMUTE);
+ reg = aic3x_read_reg_cache(codec, HPRCOM_CTRL);
+ aic3x_write(codec, HPRCOM_CTRL, reg | UNMUTE);
+
+ /* ADC default volume and unmute */
+ aic3x_write(codec, LADC_VOL, DEFAULT_GAIN);
+ aic3x_write(codec, RADC_VOL, DEFAULT_GAIN);
+ /* By default route Line1 to ADC PGA mixer */
+ aic3x_write(codec, LINE1L_2_LADC_CTRL, 0x0);
+ aic3x_write(codec, LINE1R_2_RADC_CTRL, 0x0);
+
+ /* PGA to HP Bypass default volume, disconnect from Output Mixer */
+ aic3x_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL);
+ aic3x_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL);
+ aic3x_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL);
+ aic3x_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL);
+ /* PGA to Line Out default volume, disconnect from Output Mixer */
+ aic3x_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
+ aic3x_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
+ /* PGA to Mono Line Out default volume, disconnect from Output Mixer */
+ aic3x_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL);
+ aic3x_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL);
+
+ /* Line2 to HP Bypass default volume, disconnect from Output Mixer */
+ aic3x_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
+ aic3x_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
+ aic3x_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
+ aic3x_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
+ /* Line2 Line Out default volume, disconnect from Output Mixer */
+ aic3x_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
+ aic3x_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+ /* Line2 to Mono Out default volume, disconnect from Output Mixer */
+ aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
+ aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
+
+ /* off, with power on */
+ aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* setup GPIO functions */
+ aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
+ aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
+
+ aic3x_add_controls(codec);
+ aic3x_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "aic3x: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *aic3x_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * AIC3X 2 wire address can be up to 4 devices with device addresses
+ * 0x18, 0x19, 0x1A, 0x1B
+ */
+
+/*
+ * If the i2c layer weren't so broken, we could pass this kind of data
+ * around
+ */
+static int aic3x_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = aic3x_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = aic3x_init(socdev);
+ if (ret < 0)
+ printk(KERN_ERR "aic3x: failed to initialise AIC3X\n");
+ return ret;
+}
+
+static int aic3x_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id aic3x_i2c_id[] = {
+ { "tlv320aic3x", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
+
+/* machine i2c codec control layer */
+static struct i2c_driver aic3x_i2c_driver = {
+ .driver = {
+ .name = "aic3x I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = aic3x_i2c_probe,
+ .remove = aic3x_i2c_remove,
+ .id_table = aic3x_i2c_id,
+};
+
+static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len)
+{
+ value[0] = i2c_smbus_read_byte_data(client, value[0]);
+ return (len == 1);
+}
+
+static int aic3x_add_i2c_device(struct platform_device *pdev,
+ const struct aic3x_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&aic3x_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "tlv320aic3x", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&aic3x_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+static int aic3x_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct aic3x_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct aic3x_priv *aic3x;
+ int ret = 0;
+
+ printk(KERN_INFO "AIC3X Audio Codec %s\n", AIC3X_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
+ if (aic3x == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = aic3x;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ aic3x_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t) i2c_master_send;
+ codec->hw_read = (hw_read_t) aic3x_i2c_read;
+ ret = aic3x_add_i2c_device(pdev, setup);
+ }
+#else
+ /* Add other interfaces here */
+#endif
+
+ if (ret != 0) {
+ kfree(codec->private_data);
+ kfree(codec);
+ }
+ return ret;
+}
+
+static int aic3x_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* power down chip */
+ if (codec->control_data)
+ aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&aic3x_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_aic3x = {
+ .probe = aic3x_probe,
+ .remove = aic3x_remove,
+ .suspend = aic3x_suspend,
+ .resume = aic3x_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
new file mode 100644
index 0000000..00a195a
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -0,0 +1,235 @@
+/*
+ * ALSA SoC TLV320AIC3X codec driver
+ *
+ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AIC3X_H
+#define _AIC3X_H
+
+/* AIC3X register space */
+#define AIC3X_CACHEREGNUM 103
+
+/* Page select register */
+#define AIC3X_PAGE_SELECT 0
+/* Software reset register */
+#define AIC3X_RESET 1
+/* Codec Sample rate select register */
+#define AIC3X_SAMPLE_RATE_SEL_REG 2
+/* PLL progrramming register A */
+#define AIC3X_PLL_PROGA_REG 3
+/* PLL progrramming register B */
+#define AIC3X_PLL_PROGB_REG 4
+/* PLL progrramming register C */
+#define AIC3X_PLL_PROGC_REG 5
+/* PLL progrramming register D */
+#define AIC3X_PLL_PROGD_REG 6
+/* Codec datapath setup register */
+#define AIC3X_CODEC_DATAPATH_REG 7
+/* Audio serial data interface control register A */
+#define AIC3X_ASD_INTF_CTRLA 8
+/* Audio serial data interface control register B */
+#define AIC3X_ASD_INTF_CTRLB 9
+/* Audio overflow status and PLL R value programming register */
+#define AIC3X_OVRF_STATUS_AND_PLLR_REG 11
+/* Audio codec digital filter control register */
+#define AIC3X_CODEC_DFILT_CTRL 12
+
+/* ADC PGA Gain control registers */
+#define LADC_VOL 15
+#define RADC_VOL 16
+/* MIC3 control registers */
+#define MIC3LR_2_LADC_CTRL 17
+#define MIC3LR_2_RADC_CTRL 18
+/* Line1 Input control registers */
+#define LINE1L_2_LADC_CTRL 19
+#define LINE1R_2_RADC_CTRL 22
+/* Line2 Input control registers */
+#define LINE2L_2_LADC_CTRL 20
+#define LINE2R_2_RADC_CTRL 23
+/* MICBIAS Control Register */
+#define MICBIAS_CTRL 25
+
+/* AGC Control Registers A, B, C */
+#define LAGC_CTRL_A 26
+#define LAGC_CTRL_B 27
+#define LAGC_CTRL_C 28
+#define RAGC_CTRL_A 29
+#define RAGC_CTRL_B 30
+#define RAGC_CTRL_C 31
+
+/* DAC Power and Left High Power Output control registers */
+#define DAC_PWR 37
+#define HPLCOM_CFG 37
+/* Right High Power Output control registers */
+#define HPRCOM_CFG 38
+/* DAC Output Switching control registers */
+#define DAC_LINE_MUX 41
+/* High Power Output Driver Pop Reduction registers */
+#define HPOUT_POP_REDUCTION 42
+/* DAC Digital control registers */
+#define LDAC_VOL 43
+#define RDAC_VOL 44
+/* High Power Output control registers */
+#define LINE2L_2_HPLOUT_VOL 45
+#define LINE2R_2_HPROUT_VOL 62
+#define PGAL_2_HPLOUT_VOL 46
+#define PGAR_2_HPROUT_VOL 63
+#define DACL1_2_HPLOUT_VOL 47
+#define DACR1_2_HPROUT_VOL 64
+#define HPLOUT_CTRL 51
+#define HPROUT_CTRL 65
+/* High Power COM control registers */
+#define LINE2L_2_HPLCOM_VOL 52
+#define LINE2R_2_HPRCOM_VOL 69
+#define PGAL_2_HPLCOM_VOL 53
+#define PGAR_2_HPRCOM_VOL 70
+#define DACL1_2_HPLCOM_VOL 54
+#define DACR1_2_HPRCOM_VOL 71
+#define HPLCOM_CTRL 58
+#define HPRCOM_CTRL 72
+/* Mono Line Output Plus/Minus control registers */
+#define LINE2L_2_MONOLOPM_VOL 73
+#define LINE2R_2_MONOLOPM_VOL 76
+#define PGAL_2_MONOLOPM_VOL 74
+#define PGAR_2_MONOLOPM_VOL 77
+#define DACL1_2_MONOLOPM_VOL 75
+#define DACR1_2_MONOLOPM_VOL 78
+#define MONOLOPM_CTRL 79
+/* Line Output Plus/Minus control registers */
+#define LINE2L_2_LLOPM_VOL 80
+#define LINE2R_2_RLOPM_VOL 90
+#define PGAL_2_LLOPM_VOL 81
+#define PGAR_2_RLOPM_VOL 91
+#define DACL1_2_LLOPM_VOL 82
+#define DACR1_2_RLOPM_VOL 92
+#define LLOPM_CTRL 86
+#define RLOPM_CTRL 93
+/* GPIO/IRQ registers */
+#define AIC3X_STICKY_IRQ_FLAGS_REG 96
+#define AIC3X_RT_IRQ_FLAGS_REG 97
+#define AIC3X_GPIO1_REG 98
+#define AIC3X_GPIO2_REG 99
+#define AIC3X_GPIOA_REG 100
+#define AIC3X_GPIOB_REG 101
+/* Clock generation control register */
+#define AIC3X_CLKGEN_CTRL_REG 102
+
+/* Page select register bits */
+#define PAGE0_SELECT 0
+#define PAGE1_SELECT 1
+
+/* Audio serial data interface control register A bits */
+#define BIT_CLK_MASTER 0x80
+#define WORD_CLK_MASTER 0x40
+
+/* Codec Datapath setup register 7 */
+#define FSREF_44100 (1 << 7)
+#define FSREF_48000 (0 << 7)
+#define DUAL_RATE_MODE ((1 << 5) | (1 << 6))
+#define LDAC2LCH (0x1 << 3)
+#define RDAC2RCH (0x1 << 1)
+
+/* PLL registers bitfields */
+#define PLLP_SHIFT 0
+#define PLLQ_SHIFT 3
+#define PLLR_SHIFT 0
+#define PLLJ_SHIFT 2
+#define PLLD_MSB_SHIFT 0
+#define PLLD_LSB_SHIFT 2
+
+/* Clock generation register bits */
+#define CODEC_CLKIN_PLLDIV 0
+#define CODEC_CLKIN_CLKDIV 1
+#define PLL_CLKIN_SHIFT 4
+#define MCLK_SOURCE 0x0
+#define PLL_CLKDIV_SHIFT 0
+
+/* Software reset register bits */
+#define SOFT_RESET 0x80
+
+/* PLL progrramming register A bits */
+#define PLL_ENABLE 0x80
+
+/* Route bits */
+#define ROUTE_ON 0x80
+
+/* Mute bits */
+#define UNMUTE 0x08
+#define MUTE_ON 0x80
+
+/* Power bits */
+#define LADC_PWR_ON 0x04
+#define RADC_PWR_ON 0x04
+#define LDAC_PWR_ON 0x80
+#define RDAC_PWR_ON 0x40
+#define HPLOUT_PWR_ON 0x01
+#define HPROUT_PWR_ON 0x01
+#define HPLCOM_PWR_ON 0x01
+#define HPRCOM_PWR_ON 0x01
+#define MONOLOPM_PWR_ON 0x01
+#define LLOPM_PWR_ON 0x01
+#define RLOPM_PWR_ON 0x01
+
+#define INVERT_VOL(val) (0x7f - val)
+
+/* Default output volume (inverted) */
+#define DEFAULT_VOL INVERT_VOL(0x50)
+/* Default input volume */
+#define DEFAULT_GAIN 0x20
+
+/* GPIO API */
+enum {
+ AIC3X_GPIO1_FUNC_DISABLED = 0,
+ AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX = 2,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4,
+ AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5,
+ AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6,
+ AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7,
+ AIC3X_GPIO1_FUNC_INPUT = 8,
+ AIC3X_GPIO1_FUNC_OUTPUT = 9,
+ AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10,
+ AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11,
+ AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12,
+ AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13,
+ AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14,
+ AIC3X_GPIO1_FUNC_ALL_IRQ = 16
+};
+
+enum {
+ AIC3X_GPIO2_FUNC_DISABLED = 0,
+ AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2,
+ AIC3X_GPIO2_FUNC_INPUT = 3,
+ AIC3X_GPIO2_FUNC_OUTPUT = 4,
+ AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5,
+ AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8,
+ AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9,
+ AIC3X_GPIO2_FUNC_ALL_IRQ = 10,
+ AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11,
+ AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12,
+ AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13,
+ AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14,
+ AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15
+};
+
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+int aic3x_headset_detected(struct snd_soc_codec *codec);
+
+struct aic3x_setup_data {
+ int i2c_bus;
+ unsigned short i2c_address;
+ unsigned int gpio_func[2];
+};
+
+extern struct snd_soc_dai aic3x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_aic3x;
+
+#endif /* _AIC3X_H */
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
new file mode 100644
index 0000000..a69ee72
--- /dev/null
+++ b/sound/soc/codecs/uda1380.c
@@ -0,0 +1,849 @@
+/*
+ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com>
+ * Improved support for DAPM and audio routing/mixing capabilities,
+ * added TLV support.
+ *
+ * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC
+ * codec model.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ * Copyright 2005 Openedhand Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/types.h>
+#include <linux/string.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/ioctl.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "uda1380.h"
+
+#define UDA1380_VERSION "0.6"
+
+/*
+ * uda1380 register cache
+ */
+static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
+ 0x0502, 0x0000, 0x0000, 0x3f3f,
+ 0x0202, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0xff00, 0x0000, 0x4800,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x8000, 0x0002, 0x0000,
+};
+
+/*
+ * read uda1380 register cache
+ */
+static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == UDA1380_RESET)
+ return 0;
+ if (reg >= UDA1380_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write uda1380 register cache
+ */
+static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= UDA1380_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the UDA1380 register space
+ */
+static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+
+ /* data is
+ * data[0] is register offset
+ * data[1] is MS byte
+ * data[2] is LS byte
+ */
+ data[0] = reg;
+ data[1] = (value & 0xff00) >> 8;
+ data[2] = value & 0x00ff;
+
+ uda1380_write_reg_cache(codec, reg, value);
+
+ /* the interpolator & decimator regs must only be written when the
+ * codec DAI is active.
+ */
+ if (!codec->active && (reg >= UDA1380_MVOL))
+ return 0;
+ pr_debug("uda1380: hw write %x val %x\n", reg, value);
+ if (codec->hw_write(codec->control_data, data, 3) == 3) {
+ unsigned int val;
+ i2c_master_send(codec->control_data, data, 1);
+ i2c_master_recv(codec->control_data, data, 2);
+ val = (data[0]<<8) | data[1];
+ if (val != value) {
+ pr_debug("uda1380: READ BACK VAL %x\n",
+ (data[0]<<8) | data[1]);
+ return -EIO;
+ }
+ return 0;
+ } else
+ return -EIO;
+}
+
+#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0)
+
+/* declarations of ALSA reg_elem_REAL controls */
+static const char *uda1380_deemp[] = {
+ "None",
+ "32kHz",
+ "44.1kHz",
+ "48kHz",
+ "96kHz",
+};
+static const char *uda1380_input_sel[] = {
+ "Line",
+ "Mic + Line R",
+ "Line L",
+ "Mic",
+};
+static const char *uda1380_output_sel[] = {
+ "DAC",
+ "Analog Mixer",
+};
+static const char *uda1380_spf_mode[] = {
+ "Flat",
+ "Minimum1",
+ "Minimum2",
+ "Maximum"
+};
+static const char *uda1380_capture_sel[] = {
+ "ADC",
+ "Digital Mixer"
+};
+static const char *uda1380_sel_ns[] = {
+ "3rd-order",
+ "5th-order"
+};
+static const char *uda1380_mix_control[] = {
+ "off",
+ "PCM only",
+ "before sound processing",
+ "after sound processing"
+};
+static const char *uda1380_sdet_setting[] = {
+ "3200",
+ "4800",
+ "9600",
+ "19200"
+};
+static const char *uda1380_os_setting[] = {
+ "single-speed",
+ "double-speed (no mixing)",
+ "quad-speed (no mixing)"
+};
+
+static const struct soc_enum uda1380_deemp_enum[] = {
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
+};
+static const struct soc_enum uda1380_input_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
+static const struct soc_enum uda1380_output_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */
+static const struct soc_enum uda1380_spf_enum =
+ SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */
+static const struct soc_enum uda1380_capture_sel_enum =
+ SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */
+static const struct soc_enum uda1380_sel_ns_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */
+static const struct soc_enum uda1380_mix_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */
+static const struct soc_enum uda1380_sdet_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */
+static const struct soc_enum uda1380_os_enum =
+ SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */
+
+/*
+ * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
+ */
+static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1);
+
+/*
+ * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored),
+ * from -66 dB in 0.5 dB steps (2 dB steps, really) and
+ * from -52 dB in 0.25 dB steps
+ */
+static const unsigned int mvol_tlv[] = {
+ TLV_DB_RANGE_HEAD(3),
+ 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1),
+ 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0),
+ 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0),
+};
+
+/*
+ * from -72 dB in 1.5 dB steps (6 dB steps really),
+ * from -66 dB in 0.75 dB steps (3 dB steps really),
+ * from -60 dB in 0.5 dB steps (2 dB steps really) and
+ * from -46 dB in 0.25 dB steps
+ */
+static const unsigned int vc_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1),
+ 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0),
+ 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0),
+ 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0),
+};
+
+/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */
+static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0);
+
+/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts
+ * off at 18 dB max) */
+static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0);
+
+/* from -63 to 24 dB in 0.5 dB steps (-128...48) */
+static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1);
+
+/* from 0 to 24 dB in 3 dB steps */
+static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
+
+/* from 0 to 30 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0);
+
+static const struct snd_kcontrol_new uda1380_snd_controls[] = {
+ SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */
+ SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv), /* MVCL, MVCR */
+ SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv), /* VC2 */
+ SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv), /* VC1 */
+ SOC_ENUM("Sound Processing Filter", uda1380_spf_enum), /* M */
+ SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), /* TRL, TRR */
+ SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv), /* BBL, BBR */
+/**/ SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1), /* MTM */
+ SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1), /* MT2 from decimation filter */
+ SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]), /* DE2 */
+ SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1), /* MT1, from digital data input */
+ SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]), /* DE1 */
+ SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */
+ SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */
+ SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */
+ SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */
+ SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */
+ SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */
+ SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */
+ SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv), /* ML_DEC, MR_DEC */
+/**/ SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1), /* MT_ADC */
+ SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */
+ SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0), /* ADCPOL_INV */
+ SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv), /* VGA_CTRL */
+ SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0), /* SKIP_DCFIL (before decimator) */
+ SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0), /* EN_DCFIL (at output of decimator) */
+ SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), /* TODO: enum, see table 62 */
+ SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), /* AGC_LEVEL */
+ /* -5.5, -8, -11.5, -14 dBFS */
+ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
+};
+
+/* add non dapm controls */
+static int uda1380_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Input mux */
+static const struct snd_kcontrol_new uda1380_input_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
+
+/* Output mux */
+static const struct snd_kcontrol_new uda1380_output_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_output_sel_enum);
+
+/* Capture mux */
+static const struct snd_kcontrol_new uda1380_capture_mux_control =
+ SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum);
+
+
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_input_mux_control),
+ SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_output_mux_control),
+ SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0,
+ &uda1380_capture_mux_control),
+ SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
+ SND_SOC_DAPM_INPUT("VINM"),
+ SND_SOC_DAPM_INPUT("VINL"),
+ SND_SOC_DAPM_INPUT("VINR"),
+ SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("VOUTLHP"),
+ SND_SOC_DAPM_OUTPUT("VOUTRHP"),
+ SND_SOC_DAPM_OUTPUT("VOUTL"),
+ SND_SOC_DAPM_OUTPUT("VOUTR"),
+ SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
+ SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* output mux */
+ {"HeadPhone Driver", NULL, "Output Mux"},
+ {"VOUTR", NULL, "Output Mux"},
+ {"VOUTL", NULL, "Output Mux"},
+
+ {"Analog Mixer", NULL, "VINR"},
+ {"Analog Mixer", NULL, "VINL"},
+ {"Analog Mixer", NULL, "DAC"},
+
+ {"Output Mux", "DAC", "DAC"},
+ {"Output Mux", "Analog Mixer", "Analog Mixer"},
+
+ /* {"DAC", "Digital Mixer", "I2S" } */
+
+ /* headphone driver */
+ {"VOUTLHP", NULL, "HeadPhone Driver"},
+ {"VOUTRHP", NULL, "HeadPhone Driver"},
+
+ /* input mux */
+ {"Left ADC", NULL, "Input Mux"},
+ {"Input Mux", "Mic", "Mic LNA"},
+ {"Input Mux", "Mic + Line R", "Mic LNA"},
+ {"Input Mux", "Line L", "Left PGA"},
+ {"Input Mux", "Line", "Left PGA"},
+
+ /* right input */
+ {"Right ADC", "Mic + Line R", "Right PGA"},
+ {"Right ADC", "Line", "Right PGA"},
+
+ /* inputs */
+ {"Mic LNA", NULL, "VINM"},
+ {"Left PGA", NULL, "VINL"},
+ {"Right PGA", NULL, "VINR"},
+};
+
+static int uda1380_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int iface;
+
+ /* set up DAI based upon fmt */
+ iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+ iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
+
+ /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= R01_SFORI_I2S | R01_SFORO_I2S;
+ break;
+ case SND_SOC_DAIFMT_LSB:
+ iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
+ break;
+ case SND_SOC_DAIFMT_MSB:
+ iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+ }
+
+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+ iface |= R01_SIM;
+
+ uda1380_write(codec, UDA1380_IFACE, iface);
+
+ return 0;
+}
+
+/*
+ * Flush reg cache
+ * We can only write the interpolator and decimator registers
+ * when the DAI is being clocked by the CPU DAI. It's up to the
+ * machine and cpu DAI driver to do this before we are called.
+ */
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, reg_start, reg_end, clk;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ reg_start = UDA1380_MVOL;
+ reg_end = UDA1380_MIXER;
+ } else {
+ reg_start = UDA1380_DEC;
+ reg_end = UDA1380_AGC;
+ }
+
+ /* FIXME disable DAC_CLK */
+ clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+ uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
+
+ for (reg = reg_start; reg <= reg_end; reg++) {
+ pr_debug("uda1380: flush reg %x val %x:", reg,
+ uda1380_read_reg_cache(codec, reg));
+ uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+ }
+
+ /* FIXME enable DAC_CLK */
+ uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
+
+ return 0;
+}
+
+static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+ /* set WSPLL power and divider if running from this clock */
+ if (clk & R00_DAC_CLK) {
+ int rate = params_rate(params);
+ u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+ clk &= ~0x3; /* clear SEL_LOOP_DIV */
+ switch (rate) {
+ case 6250 ... 12500:
+ clk |= 0x0;
+ break;
+ case 12501 ... 25000:
+ clk |= 0x1;
+ break;
+ case 25001 ... 50000:
+ clk |= 0x2;
+ break;
+ case 50001 ... 100000:
+ clk |= 0x3;
+ break;
+ }
+ uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ clk |= R00_EN_DAC | R00_EN_INT;
+ else
+ clk |= R00_EN_ADC | R00_EN_DEC;
+
+ uda1380_write(codec, UDA1380_CLK, clk);
+ return 0;
+}
+
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+ /* shut down WSPLL power if running from this clock */
+ if (clk & R00_DAC_CLK) {
+ u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+ uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ clk &= ~(R00_EN_DAC | R00_EN_INT);
+ else
+ clk &= ~(R00_EN_ADC | R00_EN_DEC);
+
+ uda1380_write(codec, UDA1380_CLK, clk);
+}
+
+static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
+
+ /* FIXME: mute(codec,0) is called when the magician clock is already
+ * set to WSPLL, but for some unknown reason writing to interpolator
+ * registers works only when clocked by SYSCLK */
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+ uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
+ if (mute)
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
+ else
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg);
+ uda1380_write(codec, UDA1380_CLK, clk);
+ return 0;
+}
+
+static int uda1380_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
+ break;
+ case SND_SOC_BIAS_OFF:
+ uda1380_write(codec, UDA1380_PM, 0x0);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai uda1380_dai[] = {
+{
+ .name = "UDA1380",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .digital_mute = uda1380_mute,
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+{ /* playback only - dual interface */
+ .name = "UDA1380",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .digital_mute = uda1380_mute,
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+{ /* capture only - dual interface*/
+ .name = "UDA1380",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = UDA1380_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .hw_params = uda1380_pcm_hw_params,
+ .shutdown = uda1380_pcm_shutdown,
+ .prepare = uda1380_pcm_prepare,
+ },
+ .dai_ops = {
+ .set_fmt = uda1380_set_dai_fmt,
+ },
+},
+};
+EXPORT_SYMBOL_GPL(uda1380_dai);
+
+static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int uda1380_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ uda1380_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the UDA1380 driver
+ * register mixer and dsp interfaces with the kernel
+ */
+static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "UDA1380";
+ codec->owner = THIS_MODULE;
+ codec->read = uda1380_read_reg_cache;
+ codec->write = uda1380_write;
+ codec->set_bias_level = uda1380_set_bias_level;
+ codec->dai = uda1380_dai;
+ codec->num_dai = ARRAY_SIZE(uda1380_dai);
+ codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+ codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
+ codec->reg_cache_step = 1;
+ uda1380_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ pr_err("uda1380: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ /* set clock input */
+ switch (dac_clk) {
+ case UDA1380_DAC_CLK_SYSCLK:
+ uda1380_write(codec, UDA1380_CLK, 0);
+ break;
+ case UDA1380_DAC_CLK_WSPLL:
+ uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
+ break;
+ }
+
+ /* uda1380 init */
+ uda1380_add_controls(codec);
+ uda1380_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ pr_err("uda1380: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *uda1380_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static int uda1380_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = uda1380_socdev;
+ struct uda1380_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = uda1380_init(socdev, setup->dac_clk);
+ if (ret < 0)
+ pr_err("uda1380: failed to initialise UDA1380\n");
+
+ return ret;
+}
+
+static int uda1380_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id uda1380_i2c_id[] = {
+ { "uda1380", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id);
+
+static struct i2c_driver uda1380_i2c_driver = {
+ .driver = {
+ .name = "UDA1380 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = uda1380_i2c_probe,
+ .remove = uda1380_i2c_remove,
+ .id_table = uda1380_i2c_id,
+};
+
+static int uda1380_add_i2c_device(struct platform_device *pdev,
+ const struct uda1380_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&uda1380_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "uda1380", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&uda1380_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+static int uda1380_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct uda1380_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret;
+
+ pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ uda1380_socdev = socdev;
+ ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = uda1380_add_i2c_device(pdev, setup);
+ }
+#endif
+
+ if (ret != 0)
+ kfree(codec);
+ return ret;
+}
+
+/* power down chip */
+static int uda1380_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&uda1380_i2c_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_uda1380 = {
+ .probe = uda1380_probe,
+ .remove = uda1380_remove,
+ .suspend = uda1380_suspend,
+ .resume = uda1380_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
+
+MODULE_AUTHOR("Giorgio Padrin");
+MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h
new file mode 100644
index 0000000..c55c17a
--- /dev/null
+++ b/sound/soc/codecs/uda1380.h
@@ -0,0 +1,90 @@
+/*
+ * Audio support for Philips UDA1380
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ */
+
+#ifndef _UDA1380_H
+#define _UDA1380_H
+
+#define UDA1380_CLK 0x00
+#define UDA1380_IFACE 0x01
+#define UDA1380_PM 0x02
+#define UDA1380_AMIX 0x03
+#define UDA1380_HP 0x04
+#define UDA1380_MVOL 0x10
+#define UDA1380_MIXVOL 0x11
+#define UDA1380_MODE 0x12
+#define UDA1380_DEEMP 0x13
+#define UDA1380_MIXER 0x14
+#define UDA1380_INTSTAT 0x18
+#define UDA1380_DEC 0x20
+#define UDA1380_PGA 0x21
+#define UDA1380_ADC 0x22
+#define UDA1380_AGC 0x23
+#define UDA1380_DECSTAT 0x28
+#define UDA1380_RESET 0x7f
+
+#define UDA1380_CACHEREGNUM 0x24
+
+/* Register flags */
+#define R00_EN_ADC 0x0800
+#define R00_EN_DEC 0x0400
+#define R00_EN_DAC 0x0200
+#define R00_EN_INT 0x0100
+#define R00_DAC_CLK 0x0010
+#define R01_SFORI_I2S 0x0000
+#define R01_SFORI_LSB16 0x0100
+#define R01_SFORI_LSB18 0x0200
+#define R01_SFORI_LSB20 0x0300
+#define R01_SFORI_MSB 0x0500
+#define R01_SFORI_MASK 0x0700
+#define R01_SFORO_I2S 0x0000
+#define R01_SFORO_LSB16 0x0001
+#define R01_SFORO_LSB18 0x0002
+#define R01_SFORO_LSB20 0x0003
+#define R01_SFORO_LSB24 0x0004
+#define R01_SFORO_MSB 0x0005
+#define R01_SFORO_MASK 0x0007
+#define R01_SEL_SOURCE 0x0040
+#define R01_SIM 0x0010
+#define R02_PON_PLL 0x8000
+#define R02_PON_HP 0x2000
+#define R02_PON_DAC 0x0400
+#define R02_PON_BIAS 0x0100
+#define R02_EN_AVC 0x0080
+#define R02_PON_AVC 0x0040
+#define R02_PON_LNA 0x0010
+#define R02_PON_PGAL 0x0008
+#define R02_PON_ADCL 0x0004
+#define R02_PON_PGAR 0x0002
+#define R02_PON_ADCR 0x0001
+#define R13_MTM 0x4000
+#define R14_SILENCE 0x0080
+#define R14_SDET_ON 0x0040
+#define R21_MT_ADC 0x8000
+#define R22_SEL_LNA 0x0008
+#define R22_SEL_MIC 0x0004
+#define R22_SKIP_DCFIL 0x0002
+#define R23_AGC_EN 0x0001
+
+struct uda1380_setup_data {
+ int i2c_bus;
+ unsigned short i2c_address;
+ int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL 1
+};
+
+#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */
+#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */
+#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
+
+extern struct snd_soc_dai uda1380_dai[3];
+extern struct snd_soc_codec_device soc_codec_dev_uda1380;
+
+#endif /* _UDA1380_H */
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
new file mode 100644
index 0000000..d8ca2da
--- /dev/null
+++ b/sound/soc/codecs/wm8510.c
@@ -0,0 +1,895 @@
+/*
+ * wm8510.c -- WM8510 ALSA Soc Audio driver
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8510.h"
+
+#define WM8510_VERSION "0.6"
+
+struct snd_soc_codec_device soc_codec_dev_wm8510;
+
+/*
+ * wm8510 register cache
+ * We can't read the WM8510 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0050, 0x0000, 0x0140, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x00ff,
+ 0x0000, 0x0000, 0x0100, 0x00ff,
+ 0x0000, 0x0000, 0x012c, 0x002c,
+ 0x002c, 0x002c, 0x002c, 0x0000,
+ 0x0032, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0038, 0x000b, 0x0032, 0x0000,
+ 0x0008, 0x000c, 0x0093, 0x00e9,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0003, 0x0010, 0x0000, 0x0000,
+ 0x0000, 0x0002, 0x0001, 0x0000,
+ 0x0000, 0x0000, 0x0039, 0x0000,
+ 0x0001,
+};
+
+#define WM8510_POWER1_BIASEN 0x08
+#define WM8510_POWER1_BUFIOEN 0x10
+
+/*
+ * read wm8510 register cache
+ */
+static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == WM8510_RESET)
+ return 0;
+ if (reg >= WM8510_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8510 register cache
+ */
+static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= WM8510_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8510 register space
+ */
+static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8510 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8510_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0)
+
+static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8510_alc[] = { "ALC", "Limiter" };
+
+static const struct soc_enum wm8510_enum[] = {
+ SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */
+ SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */
+ SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp),
+ SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc),
+};
+
+static const struct snd_kcontrol_new wm8510_snd_controls[] = {
+
+SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0),
+
+SOC_ENUM("DAC Companding", wm8510_enum[1]),
+SOC_ENUM("ADC Companding", wm8510_enum[0]),
+
+SOC_ENUM("Playback De-emphasis", wm8510_enum[2]),
+SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0),
+
+SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0),
+
+SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0),
+SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0),
+SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0),
+
+SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0),
+
+SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0),
+SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0),
+SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0),
+
+SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0),
+SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0),
+
+SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0),
+SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0),
+SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0),
+
+SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0),
+SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0),
+SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0),
+
+SOC_ENUM("ALC Capture Mode", wm8510_enum[3]),
+SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0),
+
+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0),
+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0),
+
+SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0),
+SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0),
+
+SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0),
+SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1),
+SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0),
+SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0),
+
+SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0),
+SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1),
+};
+
+/* add non dapm controls */
+static int wm8510_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8510_snd_controls[i], codec,
+ NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Speaker Output Mixer */
+static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0),
+};
+
+/* Mono Output Mixer */
+static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8510_boost_controls[] = {
+SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 1),
+SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0),
+SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0),
+};
+
+static const struct snd_kcontrol_new wm8510_micpga_controls[] = {
+SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0),
+SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0),
+SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0,
+ &wm8510_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm8510_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0,
+ &wm8510_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8510_mono_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0),
+SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
+
+SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0,
+ &wm8510_micpga_controls[0],
+ ARRAY_SIZE(wm8510_micpga_controls)),
+SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0,
+ &wm8510_boost_controls[0],
+ ARRAY_SIZE(wm8510_boost_controls)),
+
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0),
+
+SND_SOC_DAPM_INPUT("MICN"),
+SND_SOC_DAPM_INPUT("MICP"),
+SND_SOC_DAPM_INPUT("AUX"),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Mono output mixer */
+ {"Mono Mixer", "PCM Playback Switch", "DAC"},
+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Speaker output mixer */
+ {"Speaker Mixer", "PCM Playback Switch", "DAC"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Outputs */
+ {"Mono Out", NULL, "Mono Mixer"},
+ {"MONOOUT", NULL, "Mono Out"},
+ {"SpkN Out", NULL, "Speaker Mixer"},
+ {"SpkP Out", NULL, "Speaker Mixer"},
+ {"SPKOUTN", NULL, "SpkN Out"},
+ {"SPKOUTP", NULL, "SpkP Out"},
+
+ /* Microphone PGA */
+ {"Mic PGA", "MICN Switch", "MICN"},
+ {"Mic PGA", "MICP Switch", "MICP"},
+ { "Mic PGA", "AUX Switch", "Aux Input" },
+
+ /* Boost Mixer */
+ {"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+ {"Boost Mixer", "Mic Volume", "MICP"},
+ {"Boost Mixer", "Aux Volume", "Aux Input"},
+
+ {"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8510_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets,
+ ARRAY_SIZE(wm8510_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct pll_ {
+ unsigned int pre_div:4; /* prescale - 1 */
+ unsigned int n:4;
+ unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+
+static void pll_factors(unsigned int target, unsigned int source)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div.pre_div = 1;
+ Ndiv = target / source;
+ } else
+ pll_div.pre_div = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8510 N value %d outwith recommended range!d\n",
+ Ndiv);
+
+ pll_div.n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div.k = K;
+}
+
+static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ if (freq_in == 0 || freq_out == 0) {
+ /* Clock CODEC directly from MCLK */
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
+ wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff);
+
+ /* Turn off PLL */
+ reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
+ wm8510_write(codec, WM8510_POWER1, reg & 0x1df);
+ return 0;
+ }
+
+ pll_factors(freq_out*8, freq_in);
+
+ wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
+ wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
+ wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff);
+ wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff);
+ reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
+ wm8510_write(codec, WM8510_POWER1, reg | 0x020);
+
+ /* Run CODEC from PLL instead of MCLK */
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
+ wm8510_write(codec, WM8510_CLOCK, reg | 0x100);
+
+ return 0;
+}
+
+/*
+ * Configure WM8510 clock dividers.
+ */
+static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8510_OPCLKDIV:
+ reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf;
+ wm8510_write(codec, WM8510_GPIO, reg | div);
+ break;
+ case WM8510_MCLKDIV:
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f;
+ wm8510_write(codec, WM8510_CLOCK, reg | div);
+ break;
+ case WM8510_ADCCLK:
+ reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7;
+ wm8510_write(codec, WM8510_ADC, reg | div);
+ break;
+ case WM8510_DACCLK:
+ reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7;
+ wm8510_write(codec, WM8510_DAC, reg | div);
+ break;
+ case WM8510_BCLKDIV:
+ reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3;
+ wm8510_write(codec, WM8510_CLOCK, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+ u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ clk |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0010;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0008;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x00018;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0180;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0100;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0080;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8510_write(codec, WM8510_IFACE, iface);
+ wm8510_write(codec, WM8510_CLOCK, clk);
+ return 0;
+}
+
+static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f;
+ u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0020;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0040;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x0060;
+ break;
+ }
+
+ /* filter coefficient */
+ switch (params_rate(params)) {
+ case SNDRV_PCM_RATE_8000:
+ adn |= 0x5 << 1;
+ break;
+ case SNDRV_PCM_RATE_11025:
+ adn |= 0x4 << 1;
+ break;
+ case SNDRV_PCM_RATE_16000:
+ adn |= 0x3 << 1;
+ break;
+ case SNDRV_PCM_RATE_22050:
+ adn |= 0x2 << 1;
+ break;
+ case SNDRV_PCM_RATE_32000:
+ adn |= 0x1 << 1;
+ break;
+ case SNDRV_PCM_RATE_44100:
+ case SNDRV_PCM_RATE_48000:
+ break;
+ }
+
+ wm8510_write(codec, WM8510_IFACE, iface);
+ wm8510_write(codec, WM8510_ADD, adn);
+ return 0;
+}
+
+static int wm8510_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
+
+ if (mute)
+ wm8510_write(codec, WM8510_DAC, mute_reg | 0x40);
+ else
+ wm8510_write(codec, WM8510_DAC, mute_reg);
+ return 0;
+}
+
+/* liam need to make this lower power with dapm */
+static int wm8510_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ power1 |= 0x1; /* VMID 50k */
+ wm8510_write(codec, WM8510_POWER1, power1);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
+
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Initial cap charge at VMID 5k */
+ wm8510_write(codec, WM8510_POWER1, power1 | 0x3);
+ mdelay(100);
+ }
+
+ power1 |= 0x2; /* VMID 500k */
+ wm8510_write(codec, WM8510_POWER1, power1);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ wm8510_write(codec, WM8510_POWER1, 0);
+ wm8510_write(codec, WM8510_POWER2, 0);
+ wm8510_write(codec, WM8510_POWER3, 0);
+ break;
+ }
+
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai wm8510_dai = {
+ .name = "WM8510 HiFi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8510_RATES,
+ .formats = WM8510_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8510_RATES,
+ .formats = WM8510_FORMATS,},
+ .ops = {
+ .hw_params = wm8510_pcm_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = wm8510_mute,
+ .set_fmt = wm8510_set_dai_fmt,
+ .set_clkdiv = wm8510_set_dai_clkdiv,
+ .set_pll = wm8510_set_dai_pll,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8510_dai);
+
+static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8510_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8510_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the WM8510 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8510_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "WM8510";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8510_read_reg_cache;
+ codec->write = wm8510_write;
+ codec->set_bias_level = wm8510_set_bias_level;
+ codec->dai = &wm8510_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8510_reg);
+ codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8510_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8510: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8510_add_controls(codec);
+ wm8510_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8510: failed to register card\n");
+ goto card_err;
+ }
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *wm8510_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8510 2 wire address is 0x1a
+ */
+
+static int wm8510_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8510_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8510_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8510\n");
+
+ return ret;
+}
+
+static int wm8510_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8510_i2c_id[] = {
+ { "wm8510", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id);
+
+static struct i2c_driver wm8510_i2c_driver = {
+ .driver = {
+ .name = "WM8510 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8510_i2c_probe,
+ .remove = wm8510_i2c_remove,
+ .id_table = wm8510_i2c_id,
+};
+
+static int wm8510_add_i2c_device(struct platform_device *pdev,
+ const struct wm8510_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8510_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8510", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8510_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8510_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8510_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8510_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8510\n");
+
+ return ret;
+}
+
+static int __devexit wm8510_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8510_spi_driver = {
+ .driver = {
+ .name = "wm8510",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8510_spi_probe,
+ .remove = __devexit_p(wm8510_spi_remove),
+};
+
+static int wm8510_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
+static int wm8510_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8510_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ pr_info("WM8510 Audio Codec %s", WM8510_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ wm8510_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8510_add_i2c_device(pdev, setup);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8510_spi_write;
+ ret = spi_register_driver(&wm8510_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
+#endif
+
+ if (ret != 0)
+ kfree(codec);
+ return ret;
+}
+
+/* power down chip */
+static int wm8510_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8510_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8510_spi_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8510 = {
+ .probe = wm8510_probe,
+ .remove = wm8510_remove,
+ .suspend = wm8510_suspend,
+ .resume = wm8510_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
+
+MODULE_DESCRIPTION("ASoC WM8510 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
new file mode 100644
index 0000000..bdefcf5
--- /dev/null
+++ b/sound/soc/codecs/wm8510.h
@@ -0,0 +1,105 @@
+/*
+ * wm8510.h -- WM8510 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8510_H
+#define _WM8510_H
+
+/* WM8510 register space */
+
+#define WM8510_RESET 0x0
+#define WM8510_POWER1 0x1
+#define WM8510_POWER2 0x2
+#define WM8510_POWER3 0x3
+#define WM8510_IFACE 0x4
+#define WM8510_COMP 0x5
+#define WM8510_CLOCK 0x6
+#define WM8510_ADD 0x7
+#define WM8510_GPIO 0x8
+#define WM8510_DAC 0xa
+#define WM8510_DACVOL 0xb
+#define WM8510_ADC 0xe
+#define WM8510_ADCVOL 0xf
+#define WM8510_EQ1 0x12
+#define WM8510_EQ2 0x13
+#define WM8510_EQ3 0x14
+#define WM8510_EQ4 0x15
+#define WM8510_EQ5 0x16
+#define WM8510_DACLIM1 0x18
+#define WM8510_DACLIM2 0x19
+#define WM8510_NOTCH1 0x1b
+#define WM8510_NOTCH2 0x1c
+#define WM8510_NOTCH3 0x1d
+#define WM8510_NOTCH4 0x1e
+#define WM8510_ALC1 0x20
+#define WM8510_ALC2 0x21
+#define WM8510_ALC3 0x22
+#define WM8510_NGATE 0x23
+#define WM8510_PLLN 0x24
+#define WM8510_PLLK1 0x25
+#define WM8510_PLLK2 0x26
+#define WM8510_PLLK3 0x27
+#define WM8510_ATTEN 0x28
+#define WM8510_INPUT 0x2c
+#define WM8510_INPPGA 0x2d
+#define WM8510_ADCBOOST 0x2f
+#define WM8510_OUTPUT 0x31
+#define WM8510_SPKMIX 0x32
+#define WM8510_SPKVOL 0x36
+#define WM8510_MONOMIX 0x38
+
+#define WM8510_CACHEREGNUM 57
+
+/* Clock divider Id's */
+#define WM8510_OPCLKDIV 0
+#define WM8510_MCLKDIV 1
+#define WM8510_ADCCLK 2
+#define WM8510_DACCLK 3
+#define WM8510_BCLKDIV 4
+
+/* DAC clock dividers */
+#define WM8510_DACCLK_F2 (1 << 3)
+#define WM8510_DACCLK_F4 (0 << 3)
+
+/* ADC clock dividers */
+#define WM8510_ADCCLK_F2 (1 << 3)
+#define WM8510_ADCCLK_F4 (0 << 3)
+
+/* PLL Out dividers */
+#define WM8510_OPCLKDIV_1 (0 << 4)
+#define WM8510_OPCLKDIV_2 (1 << 4)
+#define WM8510_OPCLKDIV_3 (2 << 4)
+#define WM8510_OPCLKDIV_4 (3 << 4)
+
+/* BCLK clock dividers */
+#define WM8510_BCLKDIV_1 (0 << 2)
+#define WM8510_BCLKDIV_2 (1 << 2)
+#define WM8510_BCLKDIV_4 (2 << 2)
+#define WM8510_BCLKDIV_8 (3 << 2)
+#define WM8510_BCLKDIV_16 (4 << 2)
+#define WM8510_BCLKDIV_32 (5 << 2)
+
+/* MCLK clock dividers */
+#define WM8510_MCLKDIV_1 (0 << 5)
+#define WM8510_MCLKDIV_1_5 (1 << 5)
+#define WM8510_MCLKDIV_2 (2 << 5)
+#define WM8510_MCLKDIV_3 (3 << 5)
+#define WM8510_MCLKDIV_4 (4 << 5)
+#define WM8510_MCLKDIV_6 (5 << 5)
+#define WM8510_MCLKDIV_8 (6 << 5)
+#define WM8510_MCLKDIV_12 (7 << 5)
+
+struct wm8510_setup_data {
+ int spi;
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8510_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8510;
+
+#endif
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
new file mode 100644
index 0000000..627ebfb
--- /dev/null
+++ b/sound/soc/codecs/wm8580.c
@@ -0,0 +1,1053 @@
+/*
+ * wm8580.c -- WM8580 ALSA Soc Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Notes:
+ * The WM8580 is a multichannel codec with S/PDIF support, featuring six
+ * DAC channels and two ADC channels.
+ *
+ * Currently only the primary audio interface is supported - S/PDIF and
+ * the secondary audio interfaces are not.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+
+#include "wm8580.h"
+
+#define WM8580_VERSION "0.1"
+
+struct pll_state {
+ unsigned int in;
+ unsigned int out;
+};
+
+/* codec private data */
+struct wm8580_priv {
+ struct pll_state a;
+ struct pll_state b;
+};
+
+/* WM8580 register space */
+#define WM8580_PLLA1 0x00
+#define WM8580_PLLA2 0x01
+#define WM8580_PLLA3 0x02
+#define WM8580_PLLA4 0x03
+#define WM8580_PLLB1 0x04
+#define WM8580_PLLB2 0x05
+#define WM8580_PLLB3 0x06
+#define WM8580_PLLB4 0x07
+#define WM8580_CLKSEL 0x08
+#define WM8580_PAIF1 0x09
+#define WM8580_PAIF2 0x0A
+#define WM8580_SAIF1 0x0B
+#define WM8580_PAIF3 0x0C
+#define WM8580_PAIF4 0x0D
+#define WM8580_SAIF2 0x0E
+#define WM8580_DAC_CONTROL1 0x0F
+#define WM8580_DAC_CONTROL2 0x10
+#define WM8580_DAC_CONTROL3 0x11
+#define WM8580_DAC_CONTROL4 0x12
+#define WM8580_DAC_CONTROL5 0x13
+#define WM8580_DIGITAL_ATTENUATION_DACL1 0x14
+#define WM8580_DIGITAL_ATTENUATION_DACR1 0x15
+#define WM8580_DIGITAL_ATTENUATION_DACL2 0x16
+#define WM8580_DIGITAL_ATTENUATION_DACR2 0x17
+#define WM8580_DIGITAL_ATTENUATION_DACL3 0x18
+#define WM8580_DIGITAL_ATTENUATION_DACR3 0x19
+#define WM8580_MASTER_DIGITAL_ATTENUATION 0x1C
+#define WM8580_ADC_CONTROL1 0x1D
+#define WM8580_SPDTXCHAN0 0x1E
+#define WM8580_SPDTXCHAN1 0x1F
+#define WM8580_SPDTXCHAN2 0x20
+#define WM8580_SPDTXCHAN3 0x21
+#define WM8580_SPDTXCHAN4 0x22
+#define WM8580_SPDTXCHAN5 0x23
+#define WM8580_SPDMODE 0x24
+#define WM8580_INTMASK 0x25
+#define WM8580_GPO1 0x26
+#define WM8580_GPO2 0x27
+#define WM8580_GPO3 0x28
+#define WM8580_GPO4 0x29
+#define WM8580_GPO5 0x2A
+#define WM8580_INTSTAT 0x2B
+#define WM8580_SPDRXCHAN1 0x2C
+#define WM8580_SPDRXCHAN2 0x2D
+#define WM8580_SPDRXCHAN3 0x2E
+#define WM8580_SPDRXCHAN4 0x2F
+#define WM8580_SPDRXCHAN5 0x30
+#define WM8580_SPDSTAT 0x31
+#define WM8580_PWRDN1 0x32
+#define WM8580_PWRDN2 0x33
+#define WM8580_READBACK 0x34
+#define WM8580_RESET 0x35
+
+/* PLLB4 (register 7h) */
+#define WM8580_PLLB4_MCLKOUTSRC_MASK 0x60
+#define WM8580_PLLB4_MCLKOUTSRC_PLLA 0x20
+#define WM8580_PLLB4_MCLKOUTSRC_PLLB 0x40
+#define WM8580_PLLB4_MCLKOUTSRC_OSC 0x60
+
+#define WM8580_PLLB4_CLKOUTSRC_MASK 0x180
+#define WM8580_PLLB4_CLKOUTSRC_PLLACLK 0x080
+#define WM8580_PLLB4_CLKOUTSRC_PLLBCLK 0x100
+#define WM8580_PLLB4_CLKOUTSRC_OSCCLK 0x180
+
+/* CLKSEL (register 8h) */
+#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03
+#define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01
+#define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02
+
+/* AIF control 1 (registers 9h-bh) */
+#define WM8580_AIF_RATE_MASK 0x7
+#define WM8580_AIF_RATE_128 0x0
+#define WM8580_AIF_RATE_192 0x1
+#define WM8580_AIF_RATE_256 0x2
+#define WM8580_AIF_RATE_384 0x3
+#define WM8580_AIF_RATE_512 0x4
+#define WM8580_AIF_RATE_768 0x5
+#define WM8580_AIF_RATE_1152 0x6
+
+#define WM8580_AIF_BCLKSEL_MASK 0x18
+#define WM8580_AIF_BCLKSEL_64 0x00
+#define WM8580_AIF_BCLKSEL_128 0x08
+#define WM8580_AIF_BCLKSEL_256 0x10
+#define WM8580_AIF_BCLKSEL_SYSCLK 0x18
+
+#define WM8580_AIF_MS 0x20
+
+#define WM8580_AIF_CLKSRC_MASK 0xc0
+#define WM8580_AIF_CLKSRC_PLLA 0x40
+#define WM8580_AIF_CLKSRC_PLLB 0x40
+#define WM8580_AIF_CLKSRC_MCLK 0xc0
+
+/* AIF control 2 (registers ch-eh) */
+#define WM8580_AIF_FMT_MASK 0x03
+#define WM8580_AIF_FMT_RIGHTJ 0x00
+#define WM8580_AIF_FMT_LEFTJ 0x01
+#define WM8580_AIF_FMT_I2S 0x02
+#define WM8580_AIF_FMT_DSP 0x03
+
+#define WM8580_AIF_LENGTH_MASK 0x0c
+#define WM8580_AIF_LENGTH_16 0x00
+#define WM8580_AIF_LENGTH_20 0x04
+#define WM8580_AIF_LENGTH_24 0x08
+#define WM8580_AIF_LENGTH_32 0x0c
+
+#define WM8580_AIF_LRP 0x10
+#define WM8580_AIF_BCP 0x20
+
+/* Powerdown Register 1 (register 32h) */
+#define WM8580_PWRDN1_PWDN 0x001
+#define WM8580_PWRDN1_ALLDACPD 0x040
+
+/* Powerdown Register 2 (register 33h) */
+#define WM8580_PWRDN2_OSSCPD 0x001
+#define WM8580_PWRDN2_PLLAPD 0x002
+#define WM8580_PWRDN2_PLLBPD 0x004
+#define WM8580_PWRDN2_SPDIFPD 0x008
+#define WM8580_PWRDN2_SPDIFTXD 0x010
+#define WM8580_PWRDN2_SPDIFRXD 0x020
+
+#define WM8580_DAC_CONTROL5_MUTEALL 0x10
+
+/*
+ * wm8580 register cache
+ * We can't read the WM8580 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8580_reg[] = {
+ 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/
+ 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/
+ 0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/
+ 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/
+ 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/
+ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/
+ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R27*/
+ 0x01f0, 0x0040, 0x0000, 0x0000, /*R31(0x1F)*/
+ 0x0000, 0x0000, 0x0031, 0x000b, /*R35*/
+ 0x0039, 0x0000, 0x0010, 0x0032, /*R39*/
+ 0x0054, 0x0076, 0x0098, 0x0000, /*R43(0x2B)*/
+ 0x0000, 0x0000, 0x0000, 0x0000, /*R47*/
+ 0x0000, 0x0000, 0x005e, 0x003e, /*R51(0x33)*/
+ 0x0000, 0x0000 /*R53*/
+};
+
+/*
+ * read wm8580 register cache
+ */
+static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+ return cache[reg];
+}
+
+/*
+ * write wm8580 register cache
+ */
+static inline void wm8580_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8580 register space
+ */
+static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+
+ /* Registers are 9 bits wide */
+ value &= 0x1ff;
+
+ switch (reg) {
+ case WM8580_RESET:
+ /* Uncached */
+ break;
+ default:
+ if (value == wm8580_read_reg_cache(codec, reg))
+ return 0;
+ }
+
+ /* data is
+ * D15..D9 WM8580 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8580_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static inline unsigned int wm8580_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ default:
+ return wm8580_read_reg_cache(codec, reg);
+ }
+}
+
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+
+static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int reg2 = (kcontrol->private_value >> 24) & 0xff;
+ int ret;
+ u16 val;
+
+ /* Clear the register cache so we write without VU set */
+ wm8580_write_reg_cache(codec, reg, 0);
+ wm8580_write_reg_cache(codec, reg2, 0);
+
+ ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+ if (ret < 0)
+ return ret;
+
+ /* Now write again with the volume update bit set */
+ val = wm8580_read_reg_cache(codec, reg);
+ wm8580_write(codec, reg, val | 0x0100);
+
+ val = wm8580_read_reg_cache(codec, reg2);
+ wm8580_write(codec, reg2, val | 0x0100);
+
+ return 0;
+}
+
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_2r, \
+ .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
+ .private_value = (reg_left) | ((shift) << 8) | \
+ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+
+static const struct snd_kcontrol_new wm8580_snd_controls[] = {
+SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
+ WM8580_DIGITAL_ATTENUATION_DACL1,
+ WM8580_DIGITAL_ATTENUATION_DACR1,
+ 0, 0xff, 0, dac_tlv),
+SOC_WM8580_OUT_DOUBLE_R_TLV("DAC2 Playback Volume",
+ WM8580_DIGITAL_ATTENUATION_DACL2,
+ WM8580_DIGITAL_ATTENUATION_DACR2,
+ 0, 0xff, 0, dac_tlv),
+SOC_WM8580_OUT_DOUBLE_R_TLV("DAC3 Playback Volume",
+ WM8580_DIGITAL_ATTENUATION_DACL3,
+ WM8580_DIGITAL_ATTENUATION_DACR3,
+ 0, 0xff, 0, dac_tlv),
+
+SOC_SINGLE("DAC1 Deemphasis Switch", WM8580_DAC_CONTROL3, 0, 1, 0),
+SOC_SINGLE("DAC2 Deemphasis Switch", WM8580_DAC_CONTROL3, 1, 1, 0),
+SOC_SINGLE("DAC3 Deemphasis Switch", WM8580_DAC_CONTROL3, 2, 1, 0),
+
+SOC_DOUBLE("DAC1 Invert Switch", WM8580_DAC_CONTROL4, 0, 1, 1, 0),
+SOC_DOUBLE("DAC2 Invert Switch", WM8580_DAC_CONTROL4, 2, 3, 1, 0),
+SOC_DOUBLE("DAC3 Invert Switch", WM8580_DAC_CONTROL4, 4, 5, 1, 0),
+
+SOC_SINGLE("DAC ZC Switch", WM8580_DAC_CONTROL5, 5, 1, 0),
+SOC_SINGLE("DAC1 Switch", WM8580_DAC_CONTROL5, 0, 1, 0),
+SOC_SINGLE("DAC2 Switch", WM8580_DAC_CONTROL5, 1, 1, 0),
+SOC_SINGLE("DAC3 Switch", WM8580_DAC_CONTROL5, 2, 1, 0),
+
+SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0),
+SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0),
+};
+
+/* Add non-DAPM controls */
+static int wm8580_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8580_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1),
+SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1),
+SND_SOC_DAPM_DAC("DAC3", "Playback", WM8580_PWRDN1, 4, 1),
+
+SND_SOC_DAPM_OUTPUT("VOUT1L"),
+SND_SOC_DAPM_OUTPUT("VOUT1R"),
+SND_SOC_DAPM_OUTPUT("VOUT2L"),
+SND_SOC_DAPM_OUTPUT("VOUT2R"),
+SND_SOC_DAPM_OUTPUT("VOUT3L"),
+SND_SOC_DAPM_OUTPUT("VOUT3R"),
+
+SND_SOC_DAPM_ADC("ADC", "Capture", WM8580_PWRDN1, 1, 1),
+
+SND_SOC_DAPM_INPUT("AINL"),
+SND_SOC_DAPM_INPUT("AINR"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ { "VOUT1L", NULL, "DAC1" },
+ { "VOUT1R", NULL, "DAC1" },
+
+ { "VOUT2L", NULL, "DAC2" },
+ { "VOUT2R", NULL, "DAC2" },
+
+ { "VOUT3L", NULL, "DAC3" },
+ { "VOUT3R", NULL, "DAC3" },
+
+ { "ADC", NULL, "AINL" },
+ { "ADC", NULL, "AINR" },
+};
+
+static int wm8580_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets,
+ ARRAY_SIZE(wm8580_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 prescale:1;
+ u32 postscale:1;
+ u32 freqmode:2;
+ u32 n:4;
+ u32 k:24;
+};
+
+/* The size in bits of the pll divide */
+#define FIXED_PLL_SIZE (1 << 22)
+
+/* PLL rate to output rate divisions */
+static struct {
+ unsigned int div;
+ unsigned int freqmode;
+ unsigned int postscale;
+} post_table[] = {
+ { 2, 0, 0 },
+ { 4, 0, 1 },
+ { 4, 1, 0 },
+ { 8, 1, 1 },
+ { 8, 2, 0 },
+ { 16, 2, 1 },
+ { 12, 3, 0 },
+ { 24, 3, 1 }
+};
+
+static int pll_factors(struct _pll_div *pll_div, unsigned int target,
+ unsigned int source)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod;
+ int i;
+
+ pr_debug("wm8580: PLL %dHz->%dHz\n", source, target);
+
+ /* Scale the output frequency up; the PLL should run in the
+ * region of 90-100MHz.
+ */
+ for (i = 0; i < ARRAY_SIZE(post_table); i++) {
+ if (target * post_table[i].div >= 90000000 &&
+ target * post_table[i].div <= 100000000) {
+ pll_div->freqmode = post_table[i].freqmode;
+ pll_div->postscale = post_table[i].postscale;
+ target *= post_table[i].div;
+ break;
+ }
+ }
+
+ if (i == ARRAY_SIZE(post_table)) {
+ printk(KERN_ERR "wm8580: Unable to scale output frequency "
+ "%u\n", target);
+ return -EINVAL;
+ }
+
+ Ndiv = target / source;
+
+ if (Ndiv < 5) {
+ source /= 2;
+ pll_div->prescale = 1;
+ Ndiv = target / source;
+ } else
+ pll_div->prescale = 0;
+
+ if ((Ndiv < 5) || (Ndiv > 13)) {
+ printk(KERN_ERR
+ "WM8580 N=%d outside supported range\n", Ndiv);
+ return -EINVAL;
+ }
+
+ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ pll_div->k = K;
+
+ pr_debug("PLL %x.%x prescale %d freqmode %d postscale %d\n",
+ pll_div->n, pll_div->k, pll_div->prescale, pll_div->freqmode,
+ pll_div->postscale);
+
+ return 0;
+}
+
+static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ int offset;
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8580_priv *wm8580 = codec->private_data;
+ struct pll_state *state;
+ struct _pll_div pll_div;
+ unsigned int reg;
+ unsigned int pwr_mask;
+ int ret;
+
+ /* GCC isn't able to work out the ifs below for initialising/using
+ * pll_div so suppress warnings.
+ */
+ memset(&pll_div, 0, sizeof(pll_div));
+
+ switch (pll_id) {
+ case WM8580_PLLA:
+ state = &wm8580->a;
+ offset = 0;
+ pwr_mask = WM8580_PWRDN2_PLLAPD;
+ break;
+ case WM8580_PLLB:
+ state = &wm8580->b;
+ offset = 4;
+ pwr_mask = WM8580_PWRDN2_PLLBPD;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ if (freq_in && freq_out) {
+ ret = pll_factors(&pll_div, freq_out, freq_in);
+ if (ret != 0)
+ return ret;
+ }
+
+ state->in = freq_in;
+ state->out = freq_out;
+
+ /* Always disable the PLL - it is not safe to leave it running
+ * while reprogramming it.
+ */
+ reg = wm8580_read(codec, WM8580_PWRDN2);
+ wm8580_write(codec, WM8580_PWRDN2, reg | pwr_mask);
+
+ if (!freq_in || !freq_out)
+ return 0;
+
+ wm8580_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff);
+ wm8580_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0xff);
+ wm8580_write(codec, WM8580_PLLA3 + offset,
+ (pll_div.k >> 18 & 0xf) | (pll_div.n << 4));
+
+ reg = wm8580_read(codec, WM8580_PLLA4 + offset);
+ reg &= ~0x3f;
+ reg |= pll_div.prescale | pll_div.postscale << 1 |
+ pll_div.freqmode << 4;
+
+ wm8580_write(codec, WM8580_PLLA4 + offset, reg);
+
+ /* All done, turn it on */
+ reg = wm8580_read(codec, WM8580_PWRDN2);
+ wm8580_write(codec, WM8580_PWRDN2, reg & ~pwr_mask);
+
+ return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai_link *dai = rtd->dai;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id);
+
+ paifb &= ~WM8580_AIF_LENGTH_MASK;
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ paifb |= WM8580_AIF_LENGTH_20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ paifb |= WM8580_AIF_LENGTH_24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ paifb |= WM8580_AIF_LENGTH_24;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb);
+ return 0;
+}
+
+static int wm8580_set_paif_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int aifa;
+ unsigned int aifb;
+ int can_invert_lrclk;
+
+ aifa = wm8580_read(codec, WM8580_PAIF1 + codec_dai->id);
+ aifb = wm8580_read(codec, WM8580_PAIF3 + codec_dai->id);
+
+ aifb &= ~(WM8580_AIF_FMT_MASK | WM8580_AIF_LRP | WM8580_AIF_BCP);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aifa &= ~WM8580_AIF_MS;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aifa |= WM8580_AIF_MS;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ can_invert_lrclk = 1;
+ aifb |= WM8580_AIF_FMT_I2S;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ can_invert_lrclk = 1;
+ aifb |= WM8580_AIF_FMT_RIGHTJ;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ can_invert_lrclk = 1;
+ aifb |= WM8580_AIF_FMT_LEFTJ;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ can_invert_lrclk = 0;
+ aifb |= WM8580_AIF_FMT_DSP;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ can_invert_lrclk = 0;
+ aifb |= WM8580_AIF_FMT_DSP;
+ aifb |= WM8580_AIF_LRP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+
+ case SND_SOC_DAIFMT_IB_IF:
+ if (!can_invert_lrclk)
+ return -EINVAL;
+ aifb |= WM8580_AIF_BCP;
+ aifb |= WM8580_AIF_LRP;
+ break;
+
+ case SND_SOC_DAIFMT_IB_NF:
+ aifb |= WM8580_AIF_BCP;
+ break;
+
+ case SND_SOC_DAIFMT_NB_IF:
+ if (!can_invert_lrclk)
+ return -EINVAL;
+ aifb |= WM8580_AIF_LRP;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ wm8580_write(codec, WM8580_PAIF1 + codec_dai->id, aifa);
+ wm8580_write(codec, WM8580_PAIF3 + codec_dai->id, aifb);
+
+ return 0;
+}
+
+static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int reg;
+
+ switch (div_id) {
+ case WM8580_MCLK:
+ reg = wm8580_read(codec, WM8580_PLLB4);
+ reg &= ~WM8580_PLLB4_MCLKOUTSRC_MASK;
+
+ switch (div) {
+ case WM8580_CLKSRC_MCLK:
+ /* Input */
+ break;
+
+ case WM8580_CLKSRC_PLLA:
+ reg |= WM8580_PLLB4_MCLKOUTSRC_PLLA;
+ break;
+ case WM8580_CLKSRC_PLLB:
+ reg |= WM8580_PLLB4_MCLKOUTSRC_PLLB;
+ break;
+
+ case WM8580_CLKSRC_OSC:
+ reg |= WM8580_PLLB4_MCLKOUTSRC_OSC;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ wm8580_write(codec, WM8580_PLLB4, reg);
+ break;
+
+ case WM8580_DAC_CLKSEL:
+ reg = wm8580_read(codec, WM8580_CLKSEL);
+ reg &= ~WM8580_CLKSEL_DAC_CLKSEL_MASK;
+
+ switch (div) {
+ case WM8580_CLKSRC_MCLK:
+ break;
+
+ case WM8580_CLKSRC_PLLA:
+ reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLA;
+ break;
+
+ case WM8580_CLKSRC_PLLB:
+ reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLB;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ wm8580_write(codec, WM8580_CLKSEL, reg);
+ break;
+
+ case WM8580_CLKOUTSRC:
+ reg = wm8580_read(codec, WM8580_PLLB4);
+ reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK;
+
+ switch (div) {
+ case WM8580_CLKSRC_NONE:
+ break;
+
+ case WM8580_CLKSRC_PLLA:
+ reg |= WM8580_PLLB4_CLKOUTSRC_PLLACLK;
+ break;
+
+ case WM8580_CLKSRC_PLLB:
+ reg |= WM8580_PLLB4_CLKOUTSRC_PLLBCLK;
+ break;
+
+ case WM8580_CLKSRC_OSC:
+ reg |= WM8580_PLLB4_CLKOUTSRC_OSCCLK;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ wm8580_write(codec, WM8580_PLLB4, reg);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int reg;
+
+ reg = wm8580_read(codec, WM8580_DAC_CONTROL5);
+
+ if (mute)
+ reg |= WM8580_DAC_CONTROL5_MUTEALL;
+ else
+ reg &= ~WM8580_DAC_CONTROL5_MUTEALL;
+
+ wm8580_write(codec, WM8580_DAC_CONTROL5, reg);
+
+ return 0;
+}
+
+static int wm8580_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ case SND_SOC_BIAS_STANDBY:
+ break;
+ case SND_SOC_BIAS_OFF:
+ reg = wm8580_read(codec, WM8580_PWRDN1);
+ wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai wm8580_dai[] = {
+ {
+ .name = "WM8580 PAIFRX",
+ .id = 0,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = WM8580_FORMATS,
+ },
+ .ops = {
+ .hw_params = wm8580_paif_hw_params,
+ },
+ .dai_ops = {
+ .set_fmt = wm8580_set_paif_dai_fmt,
+ .set_clkdiv = wm8580_set_dai_clkdiv,
+ .set_pll = wm8580_set_dai_pll,
+ .digital_mute = wm8580_digital_mute,
+ },
+ },
+ {
+ .name = "WM8580 PAIFTX",
+ .id = 1,
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = WM8580_FORMATS,
+ },
+ .ops = {
+ .hw_params = wm8580_paif_hw_params,
+ },
+ .dai_ops = {
+ .set_fmt = wm8580_set_paif_dai_fmt,
+ .set_clkdiv = wm8580_set_dai_clkdiv,
+ .set_pll = wm8580_set_dai_pll,
+ },
+ },
+};
+EXPORT_SYMBOL_GPL(wm8580_dai);
+
+/*
+ * initialise the WM8580 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8580_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ codec->name = "WM8580";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8580_read_reg_cache;
+ codec->write = wm8580_write;
+ codec->set_bias_level = wm8580_set_bias_level;
+ codec->dai = wm8580_dai;
+ codec->num_dai = ARRAY_SIZE(wm8580_dai);
+ codec->reg_cache_size = ARRAY_SIZE(wm8580_reg);
+ codec->reg_cache = kmemdup(wm8580_reg, sizeof(wm8580_reg),
+ GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ /* Get the codec into a known state */
+ wm8580_write(codec, WM8580_RESET, 0);
+
+ /* Power up and get individual control of the DACs */
+ wm8580_write(codec, WM8580_PWRDN1, wm8580_read(codec, WM8580_PWRDN1) &
+ ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD));
+
+ /* Make VMID high impedence */
+ wm8580_write(codec, WM8580_ADC_CONTROL1,
+ wm8580_read(codec, WM8580_ADC_CONTROL1) & ~0x100);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1,
+ SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8580: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ wm8580_add_controls(codec);
+ wm8580_add_widgets(codec);
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8580: failed to register card\n");
+ goto card_err;
+ }
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static struct snd_soc_device *wm8580_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8580 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8580_i2c_driver;
+static struct i2c_client client_template;
+
+static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = wm8580_socdev;
+ struct wm8580_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = wm8580_init(socdev);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "failed to initialise WM8580\n");
+ goto err;
+ }
+
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int wm8580_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int wm8580_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, wm8580_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8580_i2c_driver = {
+ .driver = {
+ .name = "WM8580 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .attach_adapter = wm8580_i2c_attach,
+ .detach_client = wm8580_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "WM8580",
+ .driver = &wm8580_i2c_driver,
+};
+#endif
+
+static int wm8580_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8580_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8580_priv *wm8580;
+ int ret = 0;
+
+ pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL);
+ if (wm8580 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8580;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8580_socdev = socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&wm8580_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+ /* Add other interfaces here */
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int wm8580_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8580_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8580 = {
+ .probe = wm8580_probe,
+ .remove = wm8580_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
+
+MODULE_DESCRIPTION("ASoC WM8580 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
new file mode 100644
index 0000000..589ddab
--- /dev/null
+++ b/sound/soc/codecs/wm8580.h
@@ -0,0 +1,42 @@
+/*
+ * wm8580.h -- audio driver for WM8580
+ *
+ * Copyright 2008 Samsung Electronics.
+ * Author: Ryu Euiyoul
+ * ryu.real@gmail.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _WM8580_H
+#define _WM8580_H
+
+#define WM8580_PLLA 1
+#define WM8580_PLLB 2
+
+#define WM8580_MCLK 1
+#define WM8580_DAC_CLKSEL 2
+#define WM8580_CLKOUTSRC 3
+
+#define WM8580_CLKSRC_MCLK 1
+#define WM8580_CLKSRC_PLLA 2
+#define WM8580_CLKSRC_PLLB 3
+#define WM8580_CLKSRC_OSC 4
+#define WM8580_CLKSRC_NONE 5
+
+struct wm8580_setup_data {
+ unsigned short i2c_address;
+};
+
+#define WM8580_DAI_PAIFRX 0
+#define WM8580_DAI_PAIFTX 1
+
+extern struct snd_soc_dai wm8580_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_wm8580;
+
+#endif
+
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
new file mode 100644
index 0000000..7f8a7e3
--- /dev/null
+++ b/sound/soc/codecs/wm8731.c
@@ -0,0 +1,797 @@
+/*
+ * wm8731.c -- WM8731 ALSA SoC Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8731.h"
+
+#define WM8731_VERSION "0.13"
+
+struct snd_soc_codec_device soc_codec_dev_wm8731;
+
+/* codec private data */
+struct wm8731_priv {
+ unsigned int sysclk;
+};
+
+/*
+ * wm8731 register cache
+ * We can't read the WM8731 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 wm8731_reg[WM8731_CACHEREGNUM] = {
+ 0x0097, 0x0097, 0x0079, 0x0079,
+ 0x000a, 0x0008, 0x009f, 0x000a,
+ 0x0000, 0x0000
+};
+
+/*
+ * read wm8731 register cache
+ */
+static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == WM8731_RESET)
+ return 0;
+ if (reg >= WM8731_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8731 register cache
+ */
+static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= WM8731_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8731 register space
+ */
+static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8731 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8731_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0)
+
+static const char *wm8731_input_select[] = {"Line In", "Mic"};
+static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum wm8731_enum[] = {
+ SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select),
+ SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph),
+};
+
+static const struct snd_kcontrol_new wm8731_snd_controls[] = {
+
+SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V,
+ 0, 127, 0),
+SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V,
+ 7, 1, 0),
+
+SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0),
+SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1),
+
+SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0),
+SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1),
+
+SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1),
+
+SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1),
+SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
+
+SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
+};
+
+/* add non dapm controls */
+static int wm8731_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8731_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/* Output Mixer */
+static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new wm8731_input_mux_controls =
+SOC_DAPM_ENUM("Input Select", wm8731_enum[0]);
+
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
+ &wm8731_output_mixer_controls[0],
+ ARRAY_SIZE(wm8731_output_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1),
+SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &wm8731_input_mux_controls),
+SND_SOC_DAPM_PGA("Line Input", WM8731_PWR, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8731_PWR, 1, 1),
+SND_SOC_DAPM_INPUT("MICIN"),
+SND_SOC_DAPM_INPUT("RLINEIN"),
+SND_SOC_DAPM_INPUT("LLINEIN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* output mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+ /* outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+
+ /* input mux */
+ {"Input Mux", "Line In", "Line Input"},
+ {"Input Mux", "Mic", "Mic Bias"},
+ {"ADC", NULL, "Input Mux"},
+
+ /* inputs */
+ {"Line Input", NULL, "LLINEIN"},
+ {"Line Input", NULL, "RLINEIN"},
+ {"Mic Bias", NULL, "MICIN"},
+};
+
+static int wm8731_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ ARRAY_SIZE(wm8731_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:4;
+ u8 bosr:1;
+ u8 usb:1;
+};
+
+/* codec mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0, 0x0},
+ {18432000, 48000, 384, 0x0, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x0, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0x6, 0x0, 0x0},
+ {18432000, 32000, 576, 0x6, 0x1, 0x0},
+ {12000000, 32000, 375, 0x6, 0x0, 0x1},
+
+ /* 8k */
+ {12288000, 8000, 1536, 0x3, 0x0, 0x0},
+ {18432000, 8000, 2304, 0x3, 0x1, 0x0},
+ {11289600, 8000, 1408, 0xb, 0x0, 0x0},
+ {16934400, 8000, 2112, 0xb, 0x1, 0x0},
+ {12000000, 8000, 1500, 0x3, 0x0, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0x7, 0x0, 0x0},
+ {18432000, 96000, 192, 0x7, 0x1, 0x0},
+ {12000000, 96000, 125, 0x7, 0x0, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x8, 0x0, 0x0},
+ {16934400, 44100, 384, 0x8, 0x1, 0x0},
+ {12000000, 44100, 272, 0x8, 0x1, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0xf, 0x0, 0x0},
+ {16934400, 88200, 192, 0xf, 0x1, 0x0},
+ {12000000, 88200, 136, 0xf, 0x1, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return 0;
+}
+
+static int wm8731_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8731_priv *wm8731 = codec->private_data;
+ u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
+ int i = get_coeff(wm8731->sysclk, params_rate(params));
+ u16 srate = (coeff_div[i].sr << 2) |
+ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
+ wm8731_write(codec, WM8731_SRATE, srate);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ }
+
+ wm8731_write(codec, WM8731_IFACE, iface);
+ return 0;
+}
+
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* set active */
+ wm8731_write(codec, WM8731_ACTIVE, 0x0001);
+
+ return 0;
+}
+
+static void wm8731_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* deactivate */
+ if (!codec->active) {
+ udelay(50);
+ wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ }
+}
+
+static int wm8731_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
+
+ if (mute)
+ wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8);
+ else
+ wm8731_write(codec, WM8731_APDIGI, mute_reg);
+ return 0;
+}
+
+static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8731_priv *wm8731 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8731->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+
+static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ wm8731_write(codec, WM8731_IFACE, iface);
+ return 0;
+}
+
+static int wm8731_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* vref/mid, osc on, dac unmute */
+ wm8731_write(codec, WM8731_PWR, reg);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* everything off except vref/vmid, */
+ wm8731_write(codec, WM8731_PWR, reg | 0x0040);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ wm8731_write(codec, WM8731_PWR, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8731_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+#define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8731_dai = {
+ .name = "WM8731",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .ops = {
+ .prepare = wm8731_pcm_prepare,
+ .hw_params = wm8731_hw_params,
+ .shutdown = wm8731_shutdown,
+ },
+ .dai_ops = {
+ .digital_mute = wm8731_mute,
+ .set_sysclk = wm8731_set_dai_sysclk,
+ .set_fmt = wm8731_set_dai_fmt,
+ }
+};
+EXPORT_SYMBOL_GPL(wm8731_dai);
+
+static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8731_write(codec, WM8731_ACTIVE, 0x0);
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8731_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8731_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+/*
+ * initialise the WM8731 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8731_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "WM8731";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8731_read_reg_cache;
+ codec->write = wm8731_write;
+ codec->set_bias_level = wm8731_set_bias_level;
+ codec->dai = &wm8731_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8731_reg);
+ codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8731_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8731: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* power on device */
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* set the update bits */
+ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
+ wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
+ reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
+ wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
+ reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
+ wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
+ reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
+ wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
+
+ wm8731_add_controls(codec);
+ wm8731_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8731: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *wm8731_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8731 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+
+static int wm8731_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8731_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8731_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8731\n");
+
+ return ret;
+}
+
+static int wm8731_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8731_i2c_id[] = {
+ { "wm8731", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
+
+static struct i2c_driver wm8731_i2c_driver = {
+ .driver = {
+ .name = "WM8731 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8731_i2c_probe,
+ .remove = wm8731_i2c_remove,
+ .id_table = wm8731_i2c_id,
+};
+
+static int wm8731_add_i2c_device(struct platform_device *pdev,
+ const struct wm8731_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8731_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8731_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8731_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8731_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8731_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8731\n");
+
+ return ret;
+}
+
+static int __devexit wm8731_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8731_spi_driver = {
+ .driver = {
+ .name = "wm8731",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8731_spi_probe,
+ .remove = __devexit_p(wm8731_spi_remove),
+};
+
+static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
+static int wm8731_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8731_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8731_priv *wm8731;
+ int ret = 0;
+
+ pr_info("WM8731 Audio Codec %s", WM8731_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+ if (wm8731 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8731;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ wm8731_socdev = socdev;
+ ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8731_add_i2c_device(pdev, setup);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8731_spi_write;
+ ret = spi_register_driver(&wm8731_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
+#endif
+
+ if (ret != 0) {
+ kfree(codec->private_data);
+ kfree(codec);
+ }
+ return ret;
+}
+
+/* power down chip */
+static int wm8731_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8731_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8731_spi_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8731 = {
+ .probe = wm8731_probe,
+ .remove = wm8731_remove,
+ .suspend = wm8731_suspend,
+ .resume = wm8731_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
+
+MODULE_DESCRIPTION("ASoC WM8731 driver");
+MODULE_AUTHOR("Richard Purdie");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
new file mode 100644
index 0000000..95190e9
--- /dev/null
+++ b/sound/soc/codecs/wm8731.h
@@ -0,0 +1,46 @@
+/*
+ * wm8731.h -- WM8731 Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8731_H
+#define _WM8731_H
+
+/* WM8731 register space */
+
+#define WM8731_LINVOL 0x00
+#define WM8731_RINVOL 0x01
+#define WM8731_LOUT1V 0x02
+#define WM8731_ROUT1V 0x03
+#define WM8731_APANA 0x04
+#define WM8731_APDIGI 0x05
+#define WM8731_PWR 0x06
+#define WM8731_IFACE 0x07
+#define WM8731_SRATE 0x08
+#define WM8731_ACTIVE 0x09
+#define WM8731_RESET 0x0f
+
+#define WM8731_CACHEREGNUM 10
+
+#define WM8731_SYSCLK 0
+#define WM8731_DAI 0
+
+struct wm8731_setup_data {
+ int spi;
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8731_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8731;
+
+#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
new file mode 100644
index 0000000..9b7296e
--- /dev/null
+++ b/sound/soc/codecs/wm8750.c
@@ -0,0 +1,1091 @@
+/*
+ * wm8750.c -- WM8750 ALSA SoC audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8750.h"
+
+#define WM8750_VERSION "0.12"
+
+/* codec private data */
+struct wm8750_priv {
+ unsigned int sysclk;
+};
+
+/*
+ * wm8750 register cache
+ * We can't read the WM8750 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8750_reg[] = {
+ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */
+ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */
+ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */
+ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */
+ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */
+ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */
+ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */
+ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */
+ 0x0079, 0x0079, 0x0079, /* 40 */
+};
+
+/*
+ * read wm8750 register cache
+ */
+static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8750_CACHE_REGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8750 register cache
+ */
+static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8750_CACHE_REGNUM)
+ return;
+ cache[reg] = value;
+}
+
+static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8753 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8750_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0)
+
+/*
+ * WM8750 Controls
+ */
+static const char *wm8750_bass[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm8750_bass_filter[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
+static const char *wm8750_treble[] = {"8kHz", "4kHz"};
+static const char *wm8750_3d_lc[] = {"200Hz", "500Hz"};
+static const char *wm8750_3d_uc[] = {"2.2kHz", "1.5kHz"};
+static const char *wm8750_3d_func[] = {"Capture", "Playback"};
+static const char *wm8750_alc_func[] = {"Off", "Right", "Left", "Stereo"};
+static const char *wm8750_ng_type[] = {"Constant PGA Gain",
+ "Mute ADC Output"};
+static const char *wm8750_line_mux[] = {"Line 1", "Line 2", "Line 3", "PGA",
+ "Differential"};
+static const char *wm8750_pga_sel[] = {"Line 1", "Line 2", "Line 3",
+ "Differential"};
+static const char *wm8750_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut",
+ "ROUT1"};
+static const char *wm8750_diff_sel[] = {"Line 1", "Line 2"};
+static const char *wm8750_adcpol[] = {"Normal", "L Invert", "R Invert",
+ "L + R Invert"};
+static const char *wm8750_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *wm8750_mono_mux[] = {"Stereo", "Mono (Left)",
+ "Mono (Right)", "Digital Mono"};
+
+static const struct soc_enum wm8750_enum[] = {
+SOC_ENUM_SINGLE(WM8750_BASS, 7, 2, wm8750_bass),
+SOC_ENUM_SINGLE(WM8750_BASS, 6, 2, wm8750_bass_filter),
+SOC_ENUM_SINGLE(WM8750_TREBLE, 6, 2, wm8750_treble),
+SOC_ENUM_SINGLE(WM8750_3D, 5, 2, wm8750_3d_lc),
+SOC_ENUM_SINGLE(WM8750_3D, 6, 2, wm8750_3d_uc),
+SOC_ENUM_SINGLE(WM8750_3D, 7, 2, wm8750_3d_func),
+SOC_ENUM_SINGLE(WM8750_ALC1, 7, 4, wm8750_alc_func),
+SOC_ENUM_SINGLE(WM8750_NGATE, 1, 2, wm8750_ng_type),
+SOC_ENUM_SINGLE(WM8750_LOUTM1, 0, 5, wm8750_line_mux),
+SOC_ENUM_SINGLE(WM8750_ROUTM1, 0, 5, wm8750_line_mux),
+SOC_ENUM_SINGLE(WM8750_LADCIN, 6, 4, wm8750_pga_sel), /* 10 */
+SOC_ENUM_SINGLE(WM8750_RADCIN, 6, 4, wm8750_pga_sel),
+SOC_ENUM_SINGLE(WM8750_ADCTL2, 7, 4, wm8750_out3),
+SOC_ENUM_SINGLE(WM8750_ADCIN, 8, 2, wm8750_diff_sel),
+SOC_ENUM_SINGLE(WM8750_ADCDAC, 5, 4, wm8750_adcpol),
+SOC_ENUM_SINGLE(WM8750_ADCDAC, 1, 4, wm8750_deemph),
+SOC_ENUM_SINGLE(WM8750_ADCIN, 6, 4, wm8750_mono_mux), /* 16 */
+
+};
+
+static const struct snd_kcontrol_new wm8750_snd_controls[] = {
+
+SOC_DOUBLE_R("Capture Volume", WM8750_LINVOL, WM8750_RINVOL, 0, 63, 0),
+SOC_DOUBLE_R("Capture ZC Switch", WM8750_LINVOL, WM8750_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8750_LINVOL, WM8750_RINVOL, 7, 1, 1),
+
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8750_LOUT1V,
+ WM8750_ROUT1V, 7, 1, 0),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8750_LOUT2V,
+ WM8750_ROUT2V, 7, 1, 0),
+
+SOC_ENUM("Playback De-emphasis", wm8750_enum[15]),
+
+SOC_ENUM("Capture Polarity", wm8750_enum[14]),
+SOC_SINGLE("Playback 6dB Attenuate", WM8750_ADCDAC, 7, 1, 0),
+SOC_SINGLE("Capture 6dB Attenuate", WM8750_ADCDAC, 8, 1, 0),
+
+SOC_DOUBLE_R("PCM Volume", WM8750_LDAC, WM8750_RDAC, 0, 255, 0),
+
+SOC_ENUM("Bass Boost", wm8750_enum[0]),
+SOC_ENUM("Bass Filter", wm8750_enum[1]),
+SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 1),
+SOC_ENUM("Treble Cut-off", wm8750_enum[2]),
+
+SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0),
+SOC_SINGLE("3D Volume", WM8750_3D, 1, 15, 0),
+SOC_ENUM("3D Lower Cut-off", wm8750_enum[3]),
+SOC_ENUM("3D Upper Cut-off", wm8750_enum[4]),
+SOC_ENUM("3D Mode", wm8750_enum[5]),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8750_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8750_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", wm8750_enum[6]),
+SOC_SINGLE("ALC Capture ZC Switch", WM8750_ALC2, 7, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8750_ALC2, 0, 15, 0),
+SOC_SINGLE("ALC Capture Decay Time", WM8750_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack Time", WM8750_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8750_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", wm8750_enum[4]),
+SOC_SINGLE("ALC Capture NG Switch", WM8750_NGATE, 0, 1, 0),
+
+SOC_SINGLE("Left ADC Capture Volume", WM8750_LADC, 0, 255, 0),
+SOC_SINGLE("Right ADC Capture Volume", WM8750_RADC, 0, 255, 0),
+
+SOC_SINGLE("ZC Timeout Switch", WM8750_ADCTL1, 0, 1, 0),
+SOC_SINGLE("Playback Invert Switch", WM8750_ADCTL1, 1, 1, 0),
+
+SOC_SINGLE("Right Speaker Playback Invert Switch", WM8750_ADCTL2, 4, 1, 0),
+
+/* Unimplemented */
+/* ADCDAC Bit 0 - ADCHPD */
+/* ADCDAC Bit 4 - HPOR */
+/* ADCTL1 Bit 2,3 - DATSEL */
+/* ADCTL1 Bit 4,5 - DMONOMIX */
+/* ADCTL1 Bit 6,7 - VSEL */
+/* ADCTL2 Bit 2 - LRCM */
+/* ADCTL2 Bit 3 - TRI */
+/* ADCTL3 Bit 5 - HPFLREN */
+/* ADCTL3 Bit 6 - VROI */
+/* ADCTL3 Bit 7,8 - ADCLRM */
+/* ADCIN Bit 4 - LDCM */
+/* ADCIN Bit 5 - RDCM */
+
+SOC_DOUBLE_R("Mic Boost", WM8750_LADCIN, WM8750_RADCIN, 4, 3, 0),
+
+SOC_DOUBLE_R("Bypass Left Playback Volume", WM8750_LOUTM1,
+ WM8750_LOUTM2, 4, 7, 1),
+SOC_DOUBLE_R("Bypass Right Playback Volume", WM8750_ROUTM1,
+ WM8750_ROUTM2, 4, 7, 1),
+SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8750_MOUTM1,
+ WM8750_MOUTM2, 4, 7, 1),
+
+SOC_SINGLE("Mono Playback ZC Switch", WM8750_MOUTV, 7, 1, 0),
+
+SOC_DOUBLE_R("Headphone Playback Volume", WM8750_LOUT1V, WM8750_ROUT1V,
+ 0, 127, 0),
+SOC_DOUBLE_R("Speaker Playback Volume", WM8750_LOUT2V, WM8750_ROUT2V,
+ 0, 127, 0),
+
+SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8750_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8750_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8750_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("Playback Switch", WM8750_LOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_LOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_LOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8750_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_ROUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_ROUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM8750_ROUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_ROUTM2, 7, 1, 0),
+};
+
+/* Mono Mixer */
+static const struct snd_kcontrol_new wm8750_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_MOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_MOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_MOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_MOUTM2, 7, 1, 0),
+};
+
+/* Left Line Mux */
+static const struct snd_kcontrol_new wm8750_left_line_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[8]);
+
+/* Right Line Mux */
+static const struct snd_kcontrol_new wm8750_right_line_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[9]);
+
+/* Left PGA Mux */
+static const struct snd_kcontrol_new wm8750_left_pga_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[10]);
+
+/* Right PGA Mux */
+static const struct snd_kcontrol_new wm8750_right_pga_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[11]);
+
+/* Out 3 Mux */
+static const struct snd_kcontrol_new wm8750_out3_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[12]);
+
+/* Differential Mux */
+static const struct snd_kcontrol_new wm8750_diffmux_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[13]);
+
+/* Mono ADC Mux */
+static const struct snd_kcontrol_new wm8750_monomux_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[16]);
+
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8750_left_mixer_controls[0],
+ ARRAY_SIZE(wm8750_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8750_right_mixer_controls[0],
+ ARRAY_SIZE(wm8750_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono Mixer", WM8750_PWR2, 2, 0,
+ &wm8750_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8750_mono_mixer_controls)),
+
+ SND_SOC_DAPM_PGA("Right Out 2", WM8750_PWR2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 2", WM8750_PWR2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Out 1", WM8750_PWR2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 1", WM8750_PWR2, 6, 0, NULL, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8750_PWR2, 7, 0),
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8750_PWR2, 8, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8750_PWR1, 1, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8750_PWR1, 2, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8750_PWR1, 3, 0),
+
+ SND_SOC_DAPM_MUX("Left PGA Mux", WM8750_PWR1, 5, 0,
+ &wm8750_left_pga_controls),
+ SND_SOC_DAPM_MUX("Right PGA Mux", WM8750_PWR1, 4, 0,
+ &wm8750_right_pga_controls),
+ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_left_line_controls),
+ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_right_line_controls),
+
+ SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8750_out3_controls),
+ SND_SOC_DAPM_PGA("Out 3", WM8750_PWR2, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono Out 1", WM8750_PWR2, 2, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_diffmux_controls),
+ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_monomux_controls),
+ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8750_monomux_controls),
+
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("MONO1"),
+ SND_SOC_DAPM_OUTPUT("OUT3"),
+ SND_SOC_DAPM_OUTPUT("VREF"),
+
+ SND_SOC_DAPM_INPUT("LINPUT1"),
+ SND_SOC_DAPM_INPUT("LINPUT2"),
+ SND_SOC_DAPM_INPUT("LINPUT3"),
+ SND_SOC_DAPM_INPUT("RINPUT1"),
+ SND_SOC_DAPM_INPUT("RINPUT2"),
+ SND_SOC_DAPM_INPUT("RINPUT3"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* left mixer */
+ {"Left Mixer", "Playback Switch", "Left DAC"},
+ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Left Mixer", "Right Playback Switch", "Right DAC"},
+ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* right mixer */
+ {"Right Mixer", "Left Playback Switch", "Left DAC"},
+ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Right Mixer", "Playback Switch", "Right DAC"},
+ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* left out 1 */
+ {"Left Out 1", NULL, "Left Mixer"},
+ {"LOUT1", NULL, "Left Out 1"},
+
+ /* left out 2 */
+ {"Left Out 2", NULL, "Left Mixer"},
+ {"LOUT2", NULL, "Left Out 2"},
+
+ /* right out 1 */
+ {"Right Out 1", NULL, "Right Mixer"},
+ {"ROUT1", NULL, "Right Out 1"},
+
+ /* right out 2 */
+ {"Right Out 2", NULL, "Right Mixer"},
+ {"ROUT2", NULL, "Right Out 2"},
+
+ /* mono mixer */
+ {"Mono Mixer", "Left Playback Switch", "Left DAC"},
+ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Mono Mixer", "Right Playback Switch", "Right DAC"},
+ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* mono out */
+ {"Mono Out 1", NULL, "Mono Mixer"},
+ {"MONO1", NULL, "Mono Out 1"},
+
+ /* out 3 */
+ {"Out3 Mux", "VREF", "VREF"},
+ {"Out3 Mux", "ROUT1 + Vol", "ROUT1"},
+ {"Out3 Mux", "ROUT1", "Right Mixer"},
+ {"Out3 Mux", "MonoOut", "MONO1"},
+ {"Out 3", NULL, "Out3 Mux"},
+ {"OUT3", NULL, "Out 3"},
+
+ /* Left Line Mux */
+ {"Left Line Mux", "Line 1", "LINPUT1"},
+ {"Left Line Mux", "Line 2", "LINPUT2"},
+ {"Left Line Mux", "Line 3", "LINPUT3"},
+ {"Left Line Mux", "PGA", "Left PGA Mux"},
+ {"Left Line Mux", "Differential", "Differential Mux"},
+
+ /* Right Line Mux */
+ {"Right Line Mux", "Line 1", "RINPUT1"},
+ {"Right Line Mux", "Line 2", "RINPUT2"},
+ {"Right Line Mux", "Line 3", "RINPUT3"},
+ {"Right Line Mux", "PGA", "Right PGA Mux"},
+ {"Right Line Mux", "Differential", "Differential Mux"},
+
+ /* Left PGA Mux */
+ {"Left PGA Mux", "Line 1", "LINPUT1"},
+ {"Left PGA Mux", "Line 2", "LINPUT2"},
+ {"Left PGA Mux", "Line 3", "LINPUT3"},
+ {"Left PGA Mux", "Differential", "Differential Mux"},
+
+ /* Right PGA Mux */
+ {"Right PGA Mux", "Line 1", "RINPUT1"},
+ {"Right PGA Mux", "Line 2", "RINPUT2"},
+ {"Right PGA Mux", "Line 3", "RINPUT3"},
+ {"Right PGA Mux", "Differential", "Differential Mux"},
+
+ /* Differential Mux */
+ {"Differential Mux", "Line 1", "LINPUT1"},
+ {"Differential Mux", "Line 1", "RINPUT1"},
+ {"Differential Mux", "Line 2", "LINPUT2"},
+ {"Differential Mux", "Line 2", "RINPUT2"},
+
+ /* Left ADC Mux */
+ {"Left ADC Mux", "Stereo", "Left PGA Mux"},
+ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"},
+ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"},
+
+ /* Right ADC Mux */
+ {"Right ADC Mux", "Stereo", "Right PGA Mux"},
+ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"},
+ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"},
+
+ /* ADC */
+ {"Left ADC", NULL, "Left ADC Mux"},
+ {"Right ADC", NULL, "Right ADC Mux"},
+};
+
+static int wm8750_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ ARRAY_SIZE(wm8750_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:5;
+ u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 8k */
+ {12288000, 8000, 1536, 0x6, 0x0},
+ {11289600, 8000, 1408, 0x16, 0x0},
+ {18432000, 8000, 2304, 0x7, 0x0},
+ {16934400, 8000, 2112, 0x17, 0x0},
+ {12000000, 8000, 1500, 0x6, 0x1},
+
+ /* 11.025k */
+ {11289600, 11025, 1024, 0x18, 0x0},
+ {16934400, 11025, 1536, 0x19, 0x0},
+ {12000000, 11025, 1088, 0x19, 0x1},
+
+ /* 16k */
+ {12288000, 16000, 768, 0xa, 0x0},
+ {18432000, 16000, 1152, 0xb, 0x0},
+ {12000000, 16000, 750, 0xa, 0x1},
+
+ /* 22.05k */
+ {11289600, 22050, 512, 0x1a, 0x0},
+ {16934400, 22050, 768, 0x1b, 0x0},
+ {12000000, 22050, 544, 0x1b, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0xc, 0x0},
+ {18432000, 32000, 576, 0xd, 0x0},
+ {12000000, 32000, 375, 0xa, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x10, 0x0},
+ {16934400, 44100, 384, 0x11, 0x0},
+ {12000000, 44100, 272, 0x11, 0x1},
+
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0},
+ {18432000, 48000, 384, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0x1e, 0x0},
+ {16934400, 88200, 192, 0x1f, 0x0},
+ {12000000, 88200, 136, 0x1f, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0xe, 0x0},
+ {18432000, 96000, 192, 0xf, 0x0},
+ {12000000, 96000, 125, 0xe, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+
+ printk(KERN_ERR "wm8750: could not get coeff for mclk %d @ rate %d\n",
+ mclk, rate);
+ return -EINVAL;
+}
+
+static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8750_priv *wm8750 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8750->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8750_write(codec, WM8750_IFACE, iface);
+ return 0;
+}
+
+static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8750_priv *wm8750 = codec->private_data;
+ u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
+ u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
+ int coeff = get_coeff(wm8750->sysclk, params_rate(params));
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x000c;
+ break;
+ }
+
+ /* set iface & srate */
+ wm8750_write(codec, WM8750_IFACE, iface);
+ if (coeff >= 0)
+ wm8750_write(codec, WM8750_SRATE, srate |
+ (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+ return 0;
+}
+
+static int wm8750_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
+
+ if (mute)
+ wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8);
+ else
+ wm8750_write(codec, WM8750_ADCDAC, mute_reg);
+ return 0;
+}
+
+static int wm8750_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* set vmid to 50k and unmute dac */
+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* set vmid to 5k for quick power up */
+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* mute dac and set vmid to 500k, enable VREF */
+ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
+ break;
+ case SND_SOC_BIAS_OFF:
+ wm8750_write(codec, WM8750_PWR1, 0x0001);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8750_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8750_dai = {
+ .name = "WM8750",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8750_RATES,
+ .formats = WM8750_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8750_RATES,
+ .formats = WM8750_FORMATS,},
+ .ops = {
+ .hw_params = wm8750_pcm_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = wm8750_mute,
+ .set_fmt = wm8750_set_dai_fmt,
+ .set_sysclk = wm8750_set_dai_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8750_dai);
+
+static void wm8750_work(struct work_struct *work)
+{
+ struct snd_soc_codec *codec =
+ container_of(work, struct snd_soc_codec, delayed_work.work);
+ wm8750_set_bias_level(codec, codec->bias_level);
+}
+
+static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8750_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) {
+ if (i == WM8750_RESET)
+ continue;
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* charge wm8750 caps */
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_ON;
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(1000));
+ }
+
+ return 0;
+}
+
+/*
+ * initialise the WM8750 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8750_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "WM8750";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8750_read_reg_cache;
+ codec->write = wm8750_write;
+ codec->set_bias_level = wm8750_set_bias_level;
+ codec->dai = &wm8750_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8750_reg);
+ codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8750_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8750: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* charge output caps */
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_STANDBY;
+ schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+
+ /* set the update bits */
+ reg = wm8750_read_reg_cache(codec, WM8750_LDAC);
+ wm8750_write(codec, WM8750_LDAC, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_RDAC);
+ wm8750_write(codec, WM8750_RDAC, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V);
+ wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V);
+ wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V);
+ wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V);
+ wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_LINVOL);
+ wm8750_write(codec, WM8750_LINVOL, reg | 0x0100);
+ reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
+ wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
+
+ wm8750_add_controls(codec);
+ wm8750_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8750: failed to register card\n");
+ goto card_err;
+ }
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static struct snd_soc_device *wm8750_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8750 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+
+static int wm8750_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8750_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8750_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8750\n");
+
+ return ret;
+}
+
+static int wm8750_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8750_i2c_id[] = {
+ { "wm8750", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id);
+
+static struct i2c_driver wm8750_i2c_driver = {
+ .driver = {
+ .name = "WM8750 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8750_i2c_probe,
+ .remove = wm8750_i2c_remove,
+ .id_table = wm8750_i2c_id,
+};
+
+static int wm8750_add_i2c_device(struct platform_device *pdev,
+ const struct wm8750_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8750_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8750", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8750_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8750_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8750_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8750_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8750\n");
+
+ return ret;
+}
+
+static int __devexit wm8750_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8750_spi_driver = {
+ .driver = {
+ .name = "wm8750",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8750_spi_probe,
+ .remove = __devexit_p(wm8750_spi_remove),
+};
+
+static int wm8750_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+#endif
+
+static int wm8750_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8750_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec;
+ struct wm8750_priv *wm8750;
+ int ret;
+
+ pr_info("WM8750 Audio Codec %s", WM8750_VERSION);
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL);
+ if (wm8750 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8750;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8750_socdev = socdev;
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work);
+
+ ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8750_add_i2c_device(pdev, setup);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8750_spi_write;
+ ret = spi_register_driver(&wm8750_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
+#endif
+
+ if (ret != 0) {
+ kfree(codec->private_data);
+ kfree(codec);
+ }
+ return ret;
+}
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(dwork);
+
+ /* if there was any work waiting then we run it now and
+ * wait for it's completion */
+ if (ret) {
+ schedule_delayed_work(dwork, 0);
+ flush_scheduled_work();
+ }
+ return ret;
+}
+
+/* power down chip */
+static int wm8750_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ run_delayed_work(&codec->delayed_work);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8750_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8750_spi_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8750 = {
+ .probe = wm8750_probe,
+ .remove = wm8750_remove,
+ .suspend = wm8750_suspend,
+ .resume = wm8750_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
+
+MODULE_DESCRIPTION("ASoC WM8750 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h
new file mode 100644
index 0000000..1dc100e
--- /dev/null
+++ b/sound/soc/codecs/wm8750.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _WM8750_H
+#define _WM8750_H
+
+/* WM8750 register space */
+
+#define WM8750_LINVOL 0x00
+#define WM8750_RINVOL 0x01
+#define WM8750_LOUT1V 0x02
+#define WM8750_ROUT1V 0x03
+#define WM8750_ADCDAC 0x05
+#define WM8750_IFACE 0x07
+#define WM8750_SRATE 0x08
+#define WM8750_LDAC 0x0a
+#define WM8750_RDAC 0x0b
+#define WM8750_BASS 0x0c
+#define WM8750_TREBLE 0x0d
+#define WM8750_RESET 0x0f
+#define WM8750_3D 0x10
+#define WM8750_ALC1 0x11
+#define WM8750_ALC2 0x12
+#define WM8750_ALC3 0x13
+#define WM8750_NGATE 0x14
+#define WM8750_LADC 0x15
+#define WM8750_RADC 0x16
+#define WM8750_ADCTL1 0x17
+#define WM8750_ADCTL2 0x18
+#define WM8750_PWR1 0x19
+#define WM8750_PWR2 0x1a
+#define WM8750_ADCTL3 0x1b
+#define WM8750_ADCIN 0x1f
+#define WM8750_LADCIN 0x20
+#define WM8750_RADCIN 0x21
+#define WM8750_LOUTM1 0x22
+#define WM8750_LOUTM2 0x23
+#define WM8750_ROUTM1 0x24
+#define WM8750_ROUTM2 0x25
+#define WM8750_MOUTM1 0x26
+#define WM8750_MOUTM2 0x27
+#define WM8750_LOUT2V 0x28
+#define WM8750_ROUT2V 0x29
+#define WM8750_MOUTV 0x2a
+
+#define WM8750_CACHE_REGNUM 0x2a
+
+#define WM8750_SYSCLK 0
+
+struct wm8750_setup_data {
+ int spi;
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8750_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8750;
+
+#endif
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
new file mode 100644
index 0000000..d426eaa
--- /dev/null
+++ b/sound/soc/codecs/wm8753.c
@@ -0,0 +1,1882 @@
+/*
+ * wm8753.c -- WM8753 ALSA Soc Audio driver
+ *
+ * Copyright 2003 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Notes:
+ * The WM8753 is a low power, high quality stereo codec with integrated PCM
+ * codec designed for portable digital telephony applications.
+ *
+ * Dual DAI:-
+ *
+ * This driver support 2 DAI PCM's. This makes the default PCM available for
+ * HiFi audio (e.g. MP3, ogg) playback/capture and the other PCM available for
+ * voice.
+ *
+ * Please note that the voice PCM can be connected directly to a Bluetooth
+ * codec or GSM modem and thus cannot be read or written to, although it is
+ * available to be configured with snd_hw_params(), etc and kcontrols in the
+ * normal alsa manner.
+ *
+ * Fast DAI switching:-
+ *
+ * The driver can now fast switch between the DAI configurations via a
+ * an alsa kcontrol. This allows the PCM to remain open.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include "wm8753.h"
+
+#define WM8753_VERSION "0.16"
+
+static int caps_charge = 2000;
+module_param(caps_charge, int, 0);
+MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
+
+static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
+ unsigned int mode);
+
+/* codec private data */
+struct wm8753_priv {
+ unsigned int sysclk;
+ unsigned int pcmclk;
+};
+
+/*
+ * wm8753 register cache
+ * We can't read the WM8753 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8753_reg[] = {
+ 0x0008, 0x0000, 0x000a, 0x000a,
+ 0x0033, 0x0000, 0x0007, 0x00ff,
+ 0x00ff, 0x000f, 0x000f, 0x007b,
+ 0x0000, 0x0032, 0x0000, 0x00c3,
+ 0x00c3, 0x00c0, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0055,
+ 0x0005, 0x0050, 0x0055, 0x0050,
+ 0x0055, 0x0050, 0x0055, 0x0079,
+ 0x0079, 0x0079, 0x0079, 0x0079,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0097, 0x0097, 0x0000, 0x0004,
+ 0x0000, 0x0083, 0x0024, 0x01ba,
+ 0x0000, 0x0083, 0x0024, 0x01ba,
+ 0x0000, 0x0000
+};
+
+/*
+ * read wm8753 register cache
+ */
+static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
+ return -1;
+ return cache[reg - 1];
+}
+
+/*
+ * write wm8753 register cache
+ */
+static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg < 1 || reg > 0x3f)
+ return;
+ cache[reg - 1] = value;
+}
+
+/*
+ * write to the WM8753 register space
+ */
+static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8753 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8753_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8753_reset(c) wm8753_write(c, WM8753_RESET, 0)
+
+/*
+ * WM8753 Controls
+ */
+static const char *wm8753_base[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm8753_base_filter[] =
+ {"130Hz @ 48kHz", "200Hz @ 48kHz", "100Hz @ 16kHz", "400Hz @ 48kHz",
+ "100Hz @ 8kHz", "200Hz @ 8kHz"};
+static const char *wm8753_treble[] = {"8kHz", "4kHz"};
+static const char *wm8753_alc_func[] = {"Off", "Right", "Left", "Stereo"};
+static const char *wm8753_ng_type[] = {"Constant PGA Gain", "Mute ADC Output"};
+static const char *wm8753_3d_func[] = {"Capture", "Playback"};
+static const char *wm8753_3d_uc[] = {"2.2kHz", "1.5kHz"};
+static const char *wm8753_3d_lc[] = {"200Hz", "500Hz"};
+static const char *wm8753_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"};
+static const char *wm8753_mono_mix[] = {"Stereo", "Left", "Right", "Mono"};
+static const char *wm8753_dac_phase[] = {"Non Inverted", "Inverted"};
+static const char *wm8753_line_mix[] = {"Line 1 + 2", "Line 1 - 2",
+ "Line 1", "Line 2"};
+static const char *wm8753_mono_mux[] = {"Line Mix", "Rx Mix"};
+static const char *wm8753_right_mux[] = {"Line 2", "Rx Mix"};
+static const char *wm8753_left_mux[] = {"Line 1", "Rx Mix"};
+static const char *wm8753_rxmsel[] = {"RXP - RXN", "RXP + RXN", "RXP", "RXN"};
+static const char *wm8753_sidetone_mux[] = {"Left PGA", "Mic 1", "Mic 2",
+ "Right PGA"};
+static const char *wm8753_mono2_src[] = {"Inverted Mono 1", "Left", "Right",
+ "Left + Right"};
+static const char *wm8753_out3[] = {"VREF", "ROUT2", "Left + Right"};
+static const char *wm8753_out4[] = {"VREF", "Capture ST", "LOUT2"};
+static const char *wm8753_radcsel[] = {"PGA", "Line or RXP-RXN", "Sidetone"};
+static const char *wm8753_ladcsel[] = {"PGA", "Line or RXP-RXN", "Line"};
+static const char *wm8753_mono_adc[] = {"Stereo", "Analogue Mix Left",
+ "Analogue Mix Right", "Digital Mono Mix"};
+static const char *wm8753_adc_hp[] = {"3.4Hz @ 48kHz", "82Hz @ 16k",
+ "82Hz @ 8kHz", "170Hz @ 8kHz"};
+static const char *wm8753_adc_filter[] = {"HiFi", "Voice"};
+static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"};
+static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"};
+static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC",
+ "Channel Swap"};
+static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"};
+
+static const struct soc_enum wm8753_enum[] = {
+SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base),
+SOC_ENUM_SINGLE(WM8753_BASS, 4, 6, wm8753_base_filter),
+SOC_ENUM_SINGLE(WM8753_TREBLE, 6, 2, wm8753_treble),
+SOC_ENUM_SINGLE(WM8753_ALC1, 7, 4, wm8753_alc_func),
+SOC_ENUM_SINGLE(WM8753_NGATE, 1, 2, wm8753_ng_type),
+SOC_ENUM_SINGLE(WM8753_3D, 7, 2, wm8753_3d_func),
+SOC_ENUM_SINGLE(WM8753_3D, 6, 2, wm8753_3d_uc),
+SOC_ENUM_SINGLE(WM8753_3D, 5, 2, wm8753_3d_lc),
+SOC_ENUM_SINGLE(WM8753_DAC, 1, 4, wm8753_deemp),
+SOC_ENUM_SINGLE(WM8753_DAC, 4, 4, wm8753_mono_mix),
+SOC_ENUM_SINGLE(WM8753_DAC, 6, 2, wm8753_dac_phase),
+SOC_ENUM_SINGLE(WM8753_INCTL1, 3, 4, wm8753_line_mix),
+SOC_ENUM_SINGLE(WM8753_INCTL1, 2, 2, wm8753_mono_mux),
+SOC_ENUM_SINGLE(WM8753_INCTL1, 1, 2, wm8753_right_mux),
+SOC_ENUM_SINGLE(WM8753_INCTL1, 0, 2, wm8753_left_mux),
+SOC_ENUM_SINGLE(WM8753_INCTL2, 6, 4, wm8753_rxmsel),
+SOC_ENUM_SINGLE(WM8753_INCTL2, 4, 4, wm8753_sidetone_mux),
+SOC_ENUM_SINGLE(WM8753_OUTCTL, 7, 4, wm8753_mono2_src),
+SOC_ENUM_SINGLE(WM8753_OUTCTL, 0, 3, wm8753_out3),
+SOC_ENUM_SINGLE(WM8753_ADCTL2, 7, 3, wm8753_out4),
+SOC_ENUM_SINGLE(WM8753_ADCIN, 2, 3, wm8753_radcsel),
+SOC_ENUM_SINGLE(WM8753_ADCIN, 0, 3, wm8753_ladcsel),
+SOC_ENUM_SINGLE(WM8753_ADCIN, 4, 4, wm8753_mono_adc),
+SOC_ENUM_SINGLE(WM8753_ADC, 2, 4, wm8753_adc_hp),
+SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter),
+SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel),
+SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode),
+SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel),
+SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase),
+};
+
+
+static int wm8753_get_dai(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL);
+
+ ucontrol->value.integer.value[0] = (mode & 0xc) >> 2;
+ return 0;
+}
+
+static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL);
+
+ if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0])
+ return 0;
+
+ mode &= 0xfff3;
+ mode |= (ucontrol->value.integer.value[0] << 2);
+
+ wm8753_write(codec, WM8753_IOCTL, mode);
+ wm8753_set_dai_mode(codec, ucontrol->value.integer.value[0]);
+ return 1;
+}
+
+static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const unsigned int out_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ /* 0000000 - 0101111 = "Analogue mute" */
+ 0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0),
+ 48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0),
+};
+static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0);
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
+
+static const struct snd_kcontrol_new wm8753_snd_controls[] = {
+SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0,
+ adc_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0,
+ 127, 0, out_tlv),
+
+SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7,
+ 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4,
+ 7, 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7,
+ 1, voice_mix_tlv),
+
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7,
+ 1, 0),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7,
+ 1, 0),
+
+SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv),
+SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1,
+ mix_tlv),
+SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1,
+ voice_mix_tlv),
+SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
+
+SOC_ENUM("Bass Boost", wm8753_enum[0]),
+SOC_ENUM("Bass Filter", wm8753_enum[1]),
+SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1),
+SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
+
+SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1,
+ rec_mix_tlv),
+SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1,
+ rec_mix_tlv),
+
+SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0,
+ pga_tlv),
+SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1),
+
+SOC_ENUM("Capture Filter Select", wm8753_enum[23]),
+SOC_ENUM("Capture Filter Cut-off", wm8753_enum[24]),
+SOC_SINGLE("Capture Filter Switch", WM8753_ADC, 0, 1, 1),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8753_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8753_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", wm8753_enum[3]),
+SOC_SINGLE("ALC Capture ZC Switch", WM8753_ALC2, 8, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8753_ALC2, 0, 15, 1),
+SOC_SINGLE("ALC Capture Decay Time", WM8753_ALC3, 4, 15, 1),
+SOC_SINGLE("ALC Capture Attack Time", WM8753_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8753_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", wm8753_enum[4]),
+SOC_SINGLE("ALC Capture NG Switch", WM8753_NGATE, 0, 1, 0),
+
+SOC_ENUM("3D Function", wm8753_enum[5]),
+SOC_ENUM("3D Upper Cut-off", wm8753_enum[6]),
+SOC_ENUM("3D Lower Cut-off", wm8753_enum[7]),
+SOC_SINGLE("3D Volume", WM8753_3D, 1, 15, 0),
+SOC_SINGLE("3D Switch", WM8753_3D, 0, 1, 0),
+
+SOC_SINGLE("Capture 6dB Attenuate", WM8753_ADCTL1, 2, 1, 0),
+SOC_SINGLE("Playback 6dB Attenuate", WM8753_ADCTL1, 1, 1, 0),
+
+SOC_ENUM("De-emphasis", wm8753_enum[8]),
+SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
+SOC_ENUM("Playback Phase", wm8753_enum[10]),
+
+SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv),
+SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv),
+
+SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
+
+SOC_ENUM("ADC Data Select", wm8753_enum[27]),
+SOC_ENUM("ROUT2 Phase", wm8753_enum[28]),
+};
+
+/* add non dapm controls */
+static int wm8753_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8753_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8753_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_LOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_LOUTM2, 7, 1, 0),
+SOC_DAPM_SINGLE("Left Playback Switch", WM8753_LOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_LOUTM1, 7, 1, 0),
+};
+
+/* Right mixer */
+static const struct snd_kcontrol_new wm8753_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_ROUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_ROUTM2, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8753_ROUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_ROUTM1, 7, 1, 0),
+};
+
+/* Mono mixer */
+static const struct snd_kcontrol_new wm8753_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8753_MOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8753_MOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_MOUTM2, 3, 1, 0),
+SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_MOUTM2, 7, 1, 0),
+SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_MOUTM1, 7, 1, 0),
+};
+
+/* Mono 2 Mux */
+static const struct snd_kcontrol_new wm8753_mono2_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[17]);
+
+/* Out 3 Mux */
+static const struct snd_kcontrol_new wm8753_out3_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[18]);
+
+/* Out 4 Mux */
+static const struct snd_kcontrol_new wm8753_out4_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[19]);
+
+/* ADC Mono Mix */
+static const struct snd_kcontrol_new wm8753_adc_mono_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[22]);
+
+/* Record mixer */
+static const struct snd_kcontrol_new wm8753_record_mixer_controls[] = {
+SOC_DAPM_SINGLE("Voice Capture Switch", WM8753_RECMIX2, 3, 1, 0),
+SOC_DAPM_SINGLE("Left Capture Switch", WM8753_RECMIX1, 3, 1, 0),
+SOC_DAPM_SINGLE("Right Capture Switch", WM8753_RECMIX1, 7, 1, 0),
+};
+
+/* Left ADC mux */
+static const struct snd_kcontrol_new wm8753_adc_left_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[21]);
+
+/* Right ADC mux */
+static const struct snd_kcontrol_new wm8753_adc_right_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[20]);
+
+/* MIC mux */
+static const struct snd_kcontrol_new wm8753_mic_mux_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[16]);
+
+/* ALC mixer */
+static const struct snd_kcontrol_new wm8753_alc_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Capture Switch", WM8753_INCTL2, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic2 Capture Switch", WM8753_INCTL2, 2, 1, 0),
+SOC_DAPM_SINGLE("Mic1 Capture Switch", WM8753_INCTL2, 1, 1, 0),
+SOC_DAPM_SINGLE("Rx Capture Switch", WM8753_INCTL2, 0, 1, 0),
+};
+
+/* Left Line mux */
+static const struct snd_kcontrol_new wm8753_line_left_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[14]);
+
+/* Right Line mux */
+static const struct snd_kcontrol_new wm8753_line_right_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[13]);
+
+/* Mono Line mux */
+static const struct snd_kcontrol_new wm8753_line_mono_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[12]);
+
+/* Line mux and mixer */
+static const struct snd_kcontrol_new wm8753_line_mux_mix_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[11]);
+
+/* Rx mux and mixer */
+static const struct snd_kcontrol_new wm8753_rx_mux_mix_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[15]);
+
+/* Mic Selector Mux */
+static const struct snd_kcontrol_new wm8753_mic_sel_mux_controls =
+SOC_DAPM_ENUM("Route", wm8753_enum[25]);
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8753_PWR1, 5, 0),
+SND_SOC_DAPM_MIXER("Left Mixer", WM8753_PWR4, 0, 0,
+ &wm8753_left_mixer_controls[0], ARRAY_SIZE(wm8753_left_mixer_controls)),
+SND_SOC_DAPM_PGA("Left Out 1", WM8753_PWR3, 8, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left Out 2", WM8753_PWR3, 6, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", WM8753_PWR1, 3, 0),
+SND_SOC_DAPM_OUTPUT("LOUT1"),
+SND_SOC_DAPM_OUTPUT("LOUT2"),
+SND_SOC_DAPM_MIXER("Right Mixer", WM8753_PWR4, 1, 0,
+ &wm8753_right_mixer_controls[0], ARRAY_SIZE(wm8753_right_mixer_controls)),
+SND_SOC_DAPM_PGA("Right Out 1", WM8753_PWR3, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Out 2", WM8753_PWR3, 5, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", WM8753_PWR1, 2, 0),
+SND_SOC_DAPM_OUTPUT("ROUT1"),
+SND_SOC_DAPM_OUTPUT("ROUT2"),
+SND_SOC_DAPM_MIXER("Mono Mixer", WM8753_PWR4, 2, 0,
+ &wm8753_mono_mixer_controls[0], ARRAY_SIZE(wm8753_mono_mixer_controls)),
+SND_SOC_DAPM_PGA("Mono Out 1", WM8753_PWR3, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Mono Out 2", WM8753_PWR3, 1, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0),
+SND_SOC_DAPM_OUTPUT("MONO1"),
+SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls),
+SND_SOC_DAPM_OUTPUT("MONO2"),
+SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0),
+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls),
+SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_MUX("Out4 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out4_controls),
+SND_SOC_DAPM_PGA("Out 4", WM8753_PWR3, 3, 0, NULL, 0),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_MIXER("Playback Mixer", WM8753_PWR4, 3, 0,
+ &wm8753_record_mixer_controls[0],
+ ARRAY_SIZE(wm8753_record_mixer_controls)),
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8753_PWR2, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8753_PWR2, 2, 0),
+SND_SOC_DAPM_MUX("Capture Left Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8753_adc_mono_controls),
+SND_SOC_DAPM_MUX("Capture Right Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8753_adc_mono_controls),
+SND_SOC_DAPM_MUX("Capture Left Mux", SND_SOC_NOPM, 0, 0,
+ &wm8753_adc_left_controls),
+SND_SOC_DAPM_MUX("Capture Right Mux", SND_SOC_NOPM, 0, 0,
+ &wm8753_adc_right_controls),
+SND_SOC_DAPM_MUX("Mic Sidetone Mux", SND_SOC_NOPM, 0, 0,
+ &wm8753_mic_mux_controls),
+SND_SOC_DAPM_PGA("Left Capture Volume", WM8753_PWR2, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Capture Volume", WM8753_PWR2, 4, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("ALC Mixer", WM8753_PWR2, 6, 0,
+ &wm8753_alc_mixer_controls[0], ARRAY_SIZE(wm8753_alc_mixer_controls)),
+SND_SOC_DAPM_MUX("Line Left Mux", SND_SOC_NOPM, 0, 0,
+ &wm8753_line_left_controls),
+SND_SOC_DAPM_MUX("Line Right Mux", SND_SOC_NOPM, 0, 0,
+ &wm8753_line_right_controls),
+SND_SOC_DAPM_MUX("Line Mono Mux", SND_SOC_NOPM, 0, 0,
+ &wm8753_line_mono_controls),
+SND_SOC_DAPM_MUX("Line Mixer", WM8753_PWR2, 0, 0,
+ &wm8753_line_mux_mix_controls),
+SND_SOC_DAPM_MUX("Rx Mixer", WM8753_PWR2, 1, 0,
+ &wm8753_rx_mux_mix_controls),
+SND_SOC_DAPM_PGA("Mic 1 Volume", WM8753_PWR2, 8, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Mic 2 Volume", WM8753_PWR2, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("Mic Selection Mux", SND_SOC_NOPM, 0, 0,
+ &wm8753_mic_sel_mux_controls),
+SND_SOC_DAPM_INPUT("LINE1"),
+SND_SOC_DAPM_INPUT("LINE2"),
+SND_SOC_DAPM_INPUT("RXP"),
+SND_SOC_DAPM_INPUT("RXN"),
+SND_SOC_DAPM_INPUT("ACIN"),
+SND_SOC_DAPM_OUTPUT("ACOP"),
+SND_SOC_DAPM_INPUT("MIC1N"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2N"),
+SND_SOC_DAPM_INPUT("MIC2"),
+SND_SOC_DAPM_VMID("VREF"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* left mixer */
+ {"Left Mixer", "Left Playback Switch", "Left DAC"},
+ {"Left Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Left Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
+ {"Left Mixer", "Bypass Playback Switch", "Line Left Mux"},
+
+ /* right mixer */
+ {"Right Mixer", "Right Playback Switch", "Right DAC"},
+ {"Right Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Right Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
+ {"Right Mixer", "Bypass Playback Switch", "Line Right Mux"},
+
+ /* mono mixer */
+ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Mono Mixer", "Left Playback Switch", "Left DAC"},
+ {"Mono Mixer", "Right Playback Switch", "Right DAC"},
+ {"Mono Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
+ {"Mono Mixer", "Bypass Playback Switch", "Line Mono Mux"},
+
+ /* left out */
+ {"Left Out 1", NULL, "Left Mixer"},
+ {"Left Out 2", NULL, "Left Mixer"},
+ {"LOUT1", NULL, "Left Out 1"},
+ {"LOUT2", NULL, "Left Out 2"},
+
+ /* right out */
+ {"Right Out 1", NULL, "Right Mixer"},
+ {"Right Out 2", NULL, "Right Mixer"},
+ {"ROUT1", NULL, "Right Out 1"},
+ {"ROUT2", NULL, "Right Out 2"},
+
+ /* mono 1 out */
+ {"Mono Out 1", NULL, "Mono Mixer"},
+ {"MONO1", NULL, "Mono Out 1"},
+
+ /* mono 2 out */
+ {"Mono 2 Mux", "Left + Right", "Out3 Left + Right"},
+ {"Mono 2 Mux", "Inverted Mono 1", "MONO1"},
+ {"Mono 2 Mux", "Left", "Left Mixer"},
+ {"Mono 2 Mux", "Right", "Right Mixer"},
+ {"Mono Out 2", NULL, "Mono 2 Mux"},
+ {"MONO2", NULL, "Mono Out 2"},
+
+ /* out 3 */
+ {"Out3 Left + Right", NULL, "Left Mixer"},
+ {"Out3 Left + Right", NULL, "Right Mixer"},
+ {"Out3 Mux", "VREF", "VREF"},
+ {"Out3 Mux", "Left + Right", "Out3 Left + Right"},
+ {"Out3 Mux", "ROUT2", "ROUT2"},
+ {"Out 3", NULL, "Out3 Mux"},
+ {"OUT3", NULL, "Out 3"},
+
+ /* out 4 */
+ {"Out4 Mux", "VREF", "VREF"},
+ {"Out4 Mux", "Capture ST", "Playback Mixer"},
+ {"Out4 Mux", "LOUT2", "LOUT2"},
+ {"Out 4", NULL, "Out4 Mux"},
+ {"OUT4", NULL, "Out 4"},
+
+ /* record mixer */
+ {"Playback Mixer", "Left Capture Switch", "Left Mixer"},
+ {"Playback Mixer", "Voice Capture Switch", "Mono Mixer"},
+ {"Playback Mixer", "Right Capture Switch", "Right Mixer"},
+
+ /* Mic/SideTone Mux */
+ {"Mic Sidetone Mux", "Left PGA", "Left Capture Volume"},
+ {"Mic Sidetone Mux", "Right PGA", "Right Capture Volume"},
+ {"Mic Sidetone Mux", "Mic 1", "Mic 1 Volume"},
+ {"Mic Sidetone Mux", "Mic 2", "Mic 2 Volume"},
+
+ /* Capture Left Mux */
+ {"Capture Left Mux", "PGA", "Left Capture Volume"},
+ {"Capture Left Mux", "Line or RXP-RXN", "Line Left Mux"},
+ {"Capture Left Mux", "Line", "LINE1"},
+
+ /* Capture Right Mux */
+ {"Capture Right Mux", "PGA", "Right Capture Volume"},
+ {"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"},
+ {"Capture Right Mux", "Sidetone", "Playback Mixer"},
+
+ /* Mono Capture mixer-mux */
+ {"Capture Right Mixer", "Stereo", "Capture Right Mux"},
+ {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
+ {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
+ {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
+ {"Capture Right Mixer", "Analogue Mix Right", "Capture Right Mux"},
+ {"Capture Left Mixer", "Digital Mono Mix", "Capture Left Mux"},
+ {"Capture Left Mixer", "Digital Mono Mix", "Capture Right Mux"},
+ {"Capture Right Mixer", "Digital Mono Mix", "Capture Left Mux"},
+ {"Capture Right Mixer", "Digital Mono Mix", "Capture Right Mux"},
+
+ /* ADC */
+ {"Left ADC", NULL, "Capture Left Mixer"},
+ {"Right ADC", NULL, "Capture Right Mixer"},
+
+ /* Left Capture Volume */
+ {"Left Capture Volume", NULL, "ACIN"},
+
+ /* Right Capture Volume */
+ {"Right Capture Volume", NULL, "Mic 2 Volume"},
+
+ /* ALC Mixer */
+ {"ALC Mixer", "Line Capture Switch", "Line Mixer"},
+ {"ALC Mixer", "Mic2 Capture Switch", "Mic 2 Volume"},
+ {"ALC Mixer", "Mic1 Capture Switch", "Mic 1 Volume"},
+ {"ALC Mixer", "Rx Capture Switch", "Rx Mixer"},
+
+ /* Line Left Mux */
+ {"Line Left Mux", "Line 1", "LINE1"},
+ {"Line Left Mux", "Rx Mix", "Rx Mixer"},
+
+ /* Line Right Mux */
+ {"Line Right Mux", "Line 2", "LINE2"},
+ {"Line Right Mux", "Rx Mix", "Rx Mixer"},
+
+ /* Line Mono Mux */
+ {"Line Mono Mux", "Line Mix", "Line Mixer"},
+ {"Line Mono Mux", "Rx Mix", "Rx Mixer"},
+
+ /* Line Mixer/Mux */
+ {"Line Mixer", "Line 1 + 2", "LINE1"},
+ {"Line Mixer", "Line 1 - 2", "LINE1"},
+ {"Line Mixer", "Line 1 + 2", "LINE2"},
+ {"Line Mixer", "Line 1 - 2", "LINE2"},
+ {"Line Mixer", "Line 1", "LINE1"},
+ {"Line Mixer", "Line 2", "LINE2"},
+
+ /* Rx Mixer/Mux */
+ {"Rx Mixer", "RXP - RXN", "RXP"},
+ {"Rx Mixer", "RXP + RXN", "RXP"},
+ {"Rx Mixer", "RXP - RXN", "RXN"},
+ {"Rx Mixer", "RXP + RXN", "RXN"},
+ {"Rx Mixer", "RXP", "RXP"},
+ {"Rx Mixer", "RXN", "RXN"},
+
+ /* Mic 1 Volume */
+ {"Mic 1 Volume", NULL, "MIC1N"},
+ {"Mic 1 Volume", NULL, "Mic Selection Mux"},
+
+ /* Mic 2 Volume */
+ {"Mic 2 Volume", NULL, "MIC2N"},
+ {"Mic 2 Volume", NULL, "MIC2"},
+
+ /* Mic Selector Mux */
+ {"Mic Selection Mux", "Mic 1", "MIC1"},
+ {"Mic Selection Mux", "Mic 2", "MIC2N"},
+ {"Mic Selection Mux", "Mic 3", "MIC2"},
+
+ /* ACOP */
+ {"ACOP", NULL, "ALC Mixer"},
+};
+
+static int wm8753_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+ ARRAY_SIZE(wm8753_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 div2:1;
+ u32 n:4;
+ u32 k:24;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 22) * 10)
+
+static void pll_factors(struct _pll_div *pll_div, unsigned int target,
+ unsigned int source)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div->div2 = 1;
+ Ndiv = target / source;
+ } else
+ pll_div->div2 = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "wm8753: unsupported N = %d\n", Ndiv);
+
+ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div->k = K;
+}
+
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ u16 reg, enable;
+ int offset;
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ if (pll_id < WM8753_PLL1 || pll_id > WM8753_PLL2)
+ return -ENODEV;
+
+ if (pll_id == WM8753_PLL1) {
+ offset = 0;
+ enable = 0x10;
+ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xffef;
+ } else {
+ offset = 4;
+ enable = 0x8;
+ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfff7;
+ }
+
+ if (!freq_in || !freq_out) {
+ /* disable PLL */
+ wm8753_write(codec, WM8753_PLL1CTL1 + offset, 0x0026);
+ wm8753_write(codec, WM8753_CLOCK, reg);
+ return 0;
+ } else {
+ u16 value = 0;
+ struct _pll_div pll_div;
+
+ pll_factors(&pll_div, freq_out * 8, freq_in);
+
+ /* set up N and K PLL divisor ratios */
+ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */
+ value = (pll_div.n << 5) + ((pll_div.k & 0x3c0000) >> 18);
+ wm8753_write(codec, WM8753_PLL1CTL2 + offset, value);
+
+ /* bits 8:0 = PLL_K[17:9] */
+ value = (pll_div.k & 0x03fe00) >> 9;
+ wm8753_write(codec, WM8753_PLL1CTL3 + offset, value);
+
+ /* bits 8:0 = PLL_K[8:0] */
+ value = pll_div.k & 0x0001ff;
+ wm8753_write(codec, WM8753_PLL1CTL4 + offset, value);
+
+ /* set PLL as input and enable */
+ wm8753_write(codec, WM8753_PLL1CTL1 + offset, 0x0027 |
+ (pll_div.div2 << 3));
+ wm8753_write(codec, WM8753_CLOCK, reg | enable);
+ }
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u8 sr:5;
+ u8 usb:1;
+};
+
+/* codec hifi mclk (after PLL) clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 8k */
+ {12288000, 8000, 0x6, 0x0},
+ {11289600, 8000, 0x16, 0x0},
+ {18432000, 8000, 0x7, 0x0},
+ {16934400, 8000, 0x17, 0x0},
+ {12000000, 8000, 0x6, 0x1},
+
+ /* 11.025k */
+ {11289600, 11025, 0x18, 0x0},
+ {16934400, 11025, 0x19, 0x0},
+ {12000000, 11025, 0x19, 0x1},
+
+ /* 16k */
+ {12288000, 16000, 0xa, 0x0},
+ {18432000, 16000, 0xb, 0x0},
+ {12000000, 16000, 0xa, 0x1},
+
+ /* 22.05k */
+ {11289600, 22050, 0x1a, 0x0},
+ {16934400, 22050, 0x1b, 0x0},
+ {12000000, 22050, 0x1b, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 0xc, 0x0},
+ {18432000, 32000, 0xd, 0x0},
+ {12000000, 32000, 0xa, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 0x10, 0x0},
+ {16934400, 44100, 0x11, 0x0},
+ {12000000, 44100, 0x11, 0x1},
+
+ /* 48k */
+ {12288000, 48000, 0x0, 0x0},
+ {18432000, 48000, 0x1, 0x0},
+ {12000000, 48000, 0x0, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 0x1e, 0x0},
+ {16934400, 88200, 0x1f, 0x0},
+ {12000000, 88200, 0x1f, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 0xe, 0x0},
+ {18432000, 96000, 0xf, 0x0},
+ {12000000, 96000, 0xe, 0x1},
+};
+
+static int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return -EINVAL;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8753_priv *wm8753 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ if (clk_id == WM8753_MCLK) {
+ wm8753->sysclk = freq;
+ return 0;
+ } else if (clk_id == WM8753_PCMCLK) {
+ wm8753->pcmclk = freq;
+ return 0;
+ }
+ break;
+ }
+ return -EINVAL;
+}
+
+/*
+ * Set's ADC and Voice DAC format.
+ */
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01ec;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ voice |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ voice |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ voice |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ voice |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8753_write(codec, WM8753_PCM, voice);
+ return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8753_priv *wm8753 = codec->private_data;
+ u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3;
+ u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ voice |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ voice |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ voice |= 0x000c;
+ break;
+ }
+
+ /* sample rate */
+ if (params_rate(params) * 384 == wm8753->pcmclk)
+ srate |= 0x80;
+ wm8753_write(codec, WM8753_SRATE1, srate);
+
+ wm8753_write(codec, WM8753_PCM, voice);
+ return 0;
+}
+
+/*
+ * Set's PCM dai fmt and BCLK.
+ */
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 voice, ioctl;
+
+ voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x011f;
+ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x015d;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ioctl |= 0x2;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ voice |= 0x0040;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /* frame inversion not valid for DSP modes */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ voice |= 0x0080;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ voice &= ~0x0010;
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ voice |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ voice |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ voice |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8753_write(codec, WM8753_PCM, voice);
+ wm8753_write(codec, WM8753_IOCTL, ioctl);
+ return 0;
+}
+
+static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8753_PCMDIV:
+ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0x003f;
+ wm8753_write(codec, WM8753_CLOCK, reg | div);
+ break;
+ case WM8753_BCLKDIV:
+ reg = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x01c7;
+ wm8753_write(codec, WM8753_SRATE2, reg | div);
+ break;
+ case WM8753_VXCLKDIV:
+ reg = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x003f;
+ wm8753_write(codec, WM8753_SRATE2, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * Set's HiFi DAC format.
+ */
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01e0;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ hifi |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ hifi |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ hifi |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ hifi |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8753_write(codec, WM8753_HIFI, hifi);
+ return 0;
+}
+
+/*
+ * Set's I2S DAI format.
+ */
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 ioctl, hifi;
+
+ hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x011f;
+ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x00ae;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ioctl |= 0x1;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ hifi |= 0x0040;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /* frame inversion not valid for DSP modes */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ hifi |= 0x0080;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ hifi &= ~0x0010;
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ hifi |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ hifi |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ hifi |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8753_write(codec, WM8753_HIFI, hifi);
+ wm8753_write(codec, WM8753_IOCTL, ioctl);
+ return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8753_priv *wm8753 = codec->private_data;
+ u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
+ u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3;
+ int coeff;
+
+ /* is digital filter coefficient valid ? */
+ coeff = get_coeff(wm8753->sysclk, params_rate(params));
+ if (coeff < 0) {
+ printk(KERN_ERR "wm8753 invalid MCLK or rate\n");
+ return coeff;
+ }
+ wm8753_write(codec, WM8753_SRATE1, srate | (coeff_div[coeff].sr << 1) |
+ coeff_div[coeff].usb);
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ hifi |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ hifi |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ hifi |= 0x000c;
+ break;
+ }
+
+ wm8753_write(codec, WM8753_HIFI, hifi);
+ return 0;
+}
+
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 clock;
+
+ /* set clk source as pcmclk */
+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
+ wm8753_write(codec, WM8753_CLOCK, clock);
+
+ if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
+ return -EINVAL;
+ return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
+}
+
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
+ return -EINVAL;
+ return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+}
+
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 clock;
+
+ /* set clk source as pcmclk */
+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
+ wm8753_write(codec, WM8753_CLOCK, clock);
+
+ if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
+ return -EINVAL;
+ return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+}
+
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 clock;
+
+ /* set clk source as mclk */
+ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb;
+ wm8753_write(codec, WM8753_CLOCK, clock | 0x4);
+
+ if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
+ return -EINVAL;
+ if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
+ return -EINVAL;
+ return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+}
+
+static int wm8753_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
+
+ /* the digital mute covers the HiFi and Voice DAC's on the WM8753.
+ * make sure we check if they are not both active when we mute */
+ if (mute && dai->id == 1) {
+ if (!wm8753_dai[WM8753_DAI_VOICE].playback.active ||
+ !wm8753_dai[WM8753_DAI_HIFI].playback.active)
+ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
+ } else {
+ if (mute)
+ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8);
+ else
+ wm8753_write(codec, WM8753_DAC, mute_reg);
+ }
+
+ return 0;
+}
+
+static int wm8753_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* set vmid to 50k and unmute dac */
+ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* set vmid to 5k for quick power up */
+ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* mute dac and set vmid to 500k, enable VREF */
+ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
+ break;
+ case SND_SOC_BIAS_OFF:
+ wm8753_write(codec, WM8753_PWR1, 0x0001);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8753_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define WM8753_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+/*
+ * The WM8753 supports upto 4 different and mutually exclusive DAI
+ * configurations. This gives 2 PCM's available for use, hifi and voice.
+ * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
+ * is connected between the wm8753 and a BT codec or GSM modem.
+ *
+ * 1. Voice over PCM DAI - HIFI DAC over HIFI DAI
+ * 2. Voice over HIFI DAI - HIFI disabled
+ * 3. Voice disabled - HIFI over HIFI
+ * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
+ */
+static const struct snd_soc_dai wm8753_all_dai[] = {
+/* DAI HiFi mode 1 */
+{ .name = "WM8753 HiFi",
+ .id = 1,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .capture = { /* dummy for fast DAI switching */
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .ops = {
+ .hw_params = wm8753_i2s_hw_params,},
+ .dai_ops = {
+ .digital_mute = wm8753_mute,
+ .set_fmt = wm8753_mode1h_set_dai_fmt,
+ .set_clkdiv = wm8753_set_dai_clkdiv,
+ .set_pll = wm8753_set_dai_pll,
+ .set_sysclk = wm8753_set_dai_sysclk,
+ },
+},
+/* DAI Voice mode 1 */
+{ .name = "WM8753 Voice",
+ .id = 1,
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .ops = {
+ .hw_params = wm8753_pcm_hw_params,},
+ .dai_ops = {
+ .digital_mute = wm8753_mute,
+ .set_fmt = wm8753_mode1v_set_dai_fmt,
+ .set_clkdiv = wm8753_set_dai_clkdiv,
+ .set_pll = wm8753_set_dai_pll,
+ .set_sysclk = wm8753_set_dai_sysclk,
+ },
+},
+/* DAI HiFi mode 2 - dummy */
+{ .name = "WM8753 HiFi",
+ .id = 2,
+},
+/* DAI Voice mode 2 */
+{ .name = "WM8753 Voice",
+ .id = 2,
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .ops = {
+ .hw_params = wm8753_pcm_hw_params,},
+ .dai_ops = {
+ .digital_mute = wm8753_mute,
+ .set_fmt = wm8753_mode2_set_dai_fmt,
+ .set_clkdiv = wm8753_set_dai_clkdiv,
+ .set_pll = wm8753_set_dai_pll,
+ .set_sysclk = wm8753_set_dai_sysclk,
+ },
+},
+/* DAI HiFi mode 3 */
+{ .name = "WM8753 HiFi",
+ .id = 3,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .ops = {
+ .hw_params = wm8753_i2s_hw_params,},
+ .dai_ops = {
+ .digital_mute = wm8753_mute,
+ .set_fmt = wm8753_mode3_4_set_dai_fmt,
+ .set_clkdiv = wm8753_set_dai_clkdiv,
+ .set_pll = wm8753_set_dai_pll,
+ .set_sysclk = wm8753_set_dai_sysclk,
+ },
+},
+/* DAI Voice mode 3 - dummy */
+{ .name = "WM8753 Voice",
+ .id = 3,
+},
+/* DAI HiFi mode 4 */
+{ .name = "WM8753 HiFi",
+ .id = 4,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8753_RATES,
+ .formats = WM8753_FORMATS,},
+ .ops = {
+ .hw_params = wm8753_i2s_hw_params,},
+ .dai_ops = {
+ .digital_mute = wm8753_mute,
+ .set_fmt = wm8753_mode3_4_set_dai_fmt,
+ .set_clkdiv = wm8753_set_dai_clkdiv,
+ .set_pll = wm8753_set_dai_pll,
+ .set_sysclk = wm8753_set_dai_sysclk,
+ },
+},
+/* DAI Voice mode 4 - dummy */
+{ .name = "WM8753 Voice",
+ .id = 4,
+},
+};
+
+struct snd_soc_dai wm8753_dai[2];
+EXPORT_SYMBOL_GPL(wm8753_dai);
+
+static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
+{
+ if (mode < 4) {
+ int playback_active, capture_active, codec_active, pop_wait;
+ void *private_data;
+
+ playback_active = wm8753_dai[0].playback.active;
+ capture_active = wm8753_dai[0].capture.active;
+ codec_active = wm8753_dai[0].active;
+ private_data = wm8753_dai[0].private_data;
+ pop_wait = wm8753_dai[0].pop_wait;
+ wm8753_dai[0] = wm8753_all_dai[mode << 1];
+ wm8753_dai[0].playback.active = playback_active;
+ wm8753_dai[0].capture.active = capture_active;
+ wm8753_dai[0].active = codec_active;
+ wm8753_dai[0].private_data = private_data;
+ wm8753_dai[0].pop_wait = pop_wait;
+
+ playback_active = wm8753_dai[1].playback.active;
+ capture_active = wm8753_dai[1].capture.active;
+ codec_active = wm8753_dai[1].active;
+ private_data = wm8753_dai[1].private_data;
+ pop_wait = wm8753_dai[1].pop_wait;
+ wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1];
+ wm8753_dai[1].playback.active = playback_active;
+ wm8753_dai[1].capture.active = capture_active;
+ wm8753_dai[1].active = codec_active;
+ wm8753_dai[1].private_data = private_data;
+ wm8753_dai[1].pop_wait = pop_wait;
+ }
+ wm8753_dai[0].codec = codec;
+ wm8753_dai[1].codec = codec;
+}
+
+static void wm8753_work(struct work_struct *work)
+{
+ struct snd_soc_codec *codec =
+ container_of(work, struct snd_soc_codec, delayed_work.work);
+ wm8753_set_bias_level(codec, codec->bias_level);
+}
+
+static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* we only need to suspend if we are a valid card */
+ if (!codec->card)
+ return 0;
+
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8753_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* we only need to resume if we are a valid card */
+ if (!codec->card)
+ return 0;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
+ if (i + 1 == WM8753_RESET)
+ continue;
+ data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* charge wm8753 caps */
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_ON;
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(caps_charge));
+ }
+
+ return 0;
+}
+
+/*
+ * initialise the WM8753 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8753_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "WM8753";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8753_read_reg_cache;
+ codec->write = wm8753_write;
+ codec->set_bias_level = wm8753_set_bias_level;
+ codec->dai = wm8753_dai;
+ codec->num_dai = 2;
+ codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
+ codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8753_set_dai_mode(codec, 0);
+
+ wm8753_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8753: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* charge output caps */
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+ codec->bias_level = SND_SOC_BIAS_STANDBY;
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(caps_charge));
+
+ /* set the update bits */
+ reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
+ wm8753_write(codec, WM8753_LDAC, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_RDAC);
+ wm8753_write(codec, WM8753_RDAC, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_LADC);
+ wm8753_write(codec, WM8753_LADC, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_RADC);
+ wm8753_write(codec, WM8753_RADC, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V);
+ wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V);
+ wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V);
+ wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V);
+ wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_LINVOL);
+ wm8753_write(codec, WM8753_LINVOL, reg | 0x0100);
+ reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
+ wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
+
+ wm8753_add_controls(codec);
+ wm8753_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8753: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static struct snd_soc_device *wm8753_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8753 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+
+static int wm8753_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8753_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8753_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8753\n");
+
+ return ret;
+}
+
+static int wm8753_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8753_i2c_id[] = {
+ { "wm8753", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
+
+static struct i2c_driver wm8753_i2c_driver = {
+ .driver = {
+ .name = "WM8753 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8753_i2c_probe,
+ .remove = wm8753_i2c_remove,
+ .id_table = wm8753_i2c_id,
+};
+
+static int wm8753_add_i2c_device(struct platform_device *pdev,
+ const struct wm8753_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8753_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8753", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8753_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8753_spi_probe(struct spi_device *spi)
+{
+ struct snd_soc_device *socdev = wm8753_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ codec->control_data = spi;
+
+ ret = wm8753_init(socdev);
+ if (ret < 0)
+ dev_err(&spi->dev, "failed to initialise WM8753\n");
+
+ return ret;
+}
+
+static int __devexit wm8753_spi_remove(struct spi_device *spi)
+{
+ return 0;
+}
+
+static struct spi_driver wm8753_spi_driver = {
+ .driver = {
+ .name = "wm8753",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8753_spi_probe,
+ .remove = __devexit_p(wm8753_spi_remove),
+};
+
+static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+#endif
+
+
+static int wm8753_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8753_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8753_priv *wm8753;
+ int ret = 0;
+
+ pr_info("WM8753 Audio Codec %s", WM8753_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
+ if (wm8753 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8753;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8753_socdev = socdev;
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8753_add_i2c_device(pdev, setup);
+ }
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ if (setup->spi) {
+ codec->hw_write = (hw_write_t)wm8753_spi_write;
+ ret = spi_register_driver(&wm8753_spi_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add spi driver");
+ }
+#endif
+
+ if (ret != 0) {
+ kfree(codec->private_data);
+ kfree(codec);
+ }
+ return ret;
+}
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(dwork);
+
+ /* if there was any work waiting then we run it now and
+ * wait for it's completion */
+ if (ret) {
+ schedule_delayed_work(dwork, 0);
+ flush_scheduled_work();
+ }
+ return ret;
+}
+
+/* power down chip */
+static int wm8753_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ run_delayed_work(&codec->delayed_work);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8753_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8753_spi_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8753 = {
+ .probe = wm8753_probe,
+ .remove = wm8753_remove,
+ .suspend = wm8753_suspend,
+ .resume = wm8753_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
+
+MODULE_DESCRIPTION("ASoC WM8753 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
new file mode 100644
index 0000000..f55704c
--- /dev/null
+++ b/sound/soc/codecs/wm8753.h
@@ -0,0 +1,127 @@
+/*
+ * wm8753.h -- audio driver for WM8753
+ *
+ * Copyright 2003 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _WM8753_H
+#define _WM8753_H
+
+/* WM8753 register space */
+
+#define WM8753_DAC 0x01
+#define WM8753_ADC 0x02
+#define WM8753_PCM 0x03
+#define WM8753_HIFI 0x04
+#define WM8753_IOCTL 0x05
+#define WM8753_SRATE1 0x06
+#define WM8753_SRATE2 0x07
+#define WM8753_LDAC 0x08
+#define WM8753_RDAC 0x09
+#define WM8753_BASS 0x0a
+#define WM8753_TREBLE 0x0b
+#define WM8753_ALC1 0x0c
+#define WM8753_ALC2 0x0d
+#define WM8753_ALC3 0x0e
+#define WM8753_NGATE 0x0f
+#define WM8753_LADC 0x10
+#define WM8753_RADC 0x11
+#define WM8753_ADCTL1 0x12
+#define WM8753_3D 0x13
+#define WM8753_PWR1 0x14
+#define WM8753_PWR2 0x15
+#define WM8753_PWR3 0x16
+#define WM8753_PWR4 0x17
+#define WM8753_ID 0x18
+#define WM8753_INTPOL 0x19
+#define WM8753_INTEN 0x1a
+#define WM8753_GPIO1 0x1b
+#define WM8753_GPIO2 0x1c
+#define WM8753_RESET 0x1f
+#define WM8753_RECMIX1 0x20
+#define WM8753_RECMIX2 0x21
+#define WM8753_LOUTM1 0x22
+#define WM8753_LOUTM2 0x23
+#define WM8753_ROUTM1 0x24
+#define WM8753_ROUTM2 0x25
+#define WM8753_MOUTM1 0x26
+#define WM8753_MOUTM2 0x27
+#define WM8753_LOUT1V 0x28
+#define WM8753_ROUT1V 0x29
+#define WM8753_LOUT2V 0x2a
+#define WM8753_ROUT2V 0x2b
+#define WM8753_MOUTV 0x2c
+#define WM8753_OUTCTL 0x2d
+#define WM8753_ADCIN 0x2e
+#define WM8753_INCTL1 0x2f
+#define WM8753_INCTL2 0x30
+#define WM8753_LINVOL 0x31
+#define WM8753_RINVOL 0x32
+#define WM8753_MICBIAS 0x33
+#define WM8753_CLOCK 0x34
+#define WM8753_PLL1CTL1 0x35
+#define WM8753_PLL1CTL2 0x36
+#define WM8753_PLL1CTL3 0x37
+#define WM8753_PLL1CTL4 0x38
+#define WM8753_PLL2CTL1 0x39
+#define WM8753_PLL2CTL2 0x3a
+#define WM8753_PLL2CTL3 0x3b
+#define WM8753_PLL2CTL4 0x3c
+#define WM8753_BIASCTL 0x3d
+#define WM8753_ADCTL2 0x3f
+
+struct wm8753_setup_data {
+ int spi;
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+#define WM8753_PLL1 0
+#define WM8753_PLL2 1
+
+/* clock inputs */
+#define WM8753_MCLK 0
+#define WM8753_PCMCLK 1
+
+/* clock divider id's */
+#define WM8753_PCMDIV 0
+#define WM8753_BCLKDIV 1
+#define WM8753_VXCLKDIV 2
+
+/* PCM clock dividers */
+#define WM8753_PCM_DIV_1 (0 << 6)
+#define WM8753_PCM_DIV_3 (2 << 6)
+#define WM8753_PCM_DIV_5_5 (3 << 6)
+#define WM8753_PCM_DIV_2 (4 << 6)
+#define WM8753_PCM_DIV_4 (5 << 6)
+#define WM8753_PCM_DIV_6 (6 << 6)
+#define WM8753_PCM_DIV_8 (7 << 6)
+
+/* BCLK clock dividers */
+#define WM8753_BCLK_DIV_1 (0 << 3)
+#define WM8753_BCLK_DIV_2 (1 << 3)
+#define WM8753_BCLK_DIV_4 (2 << 3)
+#define WM8753_BCLK_DIV_8 (3 << 3)
+#define WM8753_BCLK_DIV_16 (4 << 3)
+
+/* VXCLK clock dividers */
+#define WM8753_VXCLK_DIV_1 (0 << 6)
+#define WM8753_VXCLK_DIV_2 (1 << 6)
+#define WM8753_VXCLK_DIV_4 (2 << 6)
+#define WM8753_VXCLK_DIV_8 (3 << 6)
+#define WM8753_VXCLK_DIV_16 (4 << 6)
+
+#define WM8753_DAI_HIFI 0
+#define WM8753_DAI_VOICE 1
+
+extern struct snd_soc_dai wm8753_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm8753;
+
+#endif
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
new file mode 100644
index 0000000..3b326c9
--- /dev/null
+++ b/sound/soc/codecs/wm8900.c
@@ -0,0 +1,1541 @@
+/*
+ * wm8900.c -- WM8900 ALSA Soc Audio driver
+ *
+ * Copyright 2007, 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * TODO:
+ * - Tristating.
+ * - TDM.
+ * - Jack detect.
+ * - FLL source configuration, currently only MCLK is supported.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8900.h"
+
+/* WM8900 register space */
+#define WM8900_REG_RESET 0x0
+#define WM8900_REG_ID 0x0
+#define WM8900_REG_POWER1 0x1
+#define WM8900_REG_POWER2 0x2
+#define WM8900_REG_POWER3 0x3
+#define WM8900_REG_AUDIO1 0x4
+#define WM8900_REG_AUDIO2 0x5
+#define WM8900_REG_CLOCKING1 0x6
+#define WM8900_REG_CLOCKING2 0x7
+#define WM8900_REG_AUDIO3 0x8
+#define WM8900_REG_AUDIO4 0x9
+#define WM8900_REG_DACCTRL 0xa
+#define WM8900_REG_LDAC_DV 0xb
+#define WM8900_REG_RDAC_DV 0xc
+#define WM8900_REG_SIDETONE 0xd
+#define WM8900_REG_ADCCTRL 0xe
+#define WM8900_REG_LADC_DV 0xf
+#define WM8900_REG_RADC_DV 0x10
+#define WM8900_REG_GPIO 0x12
+#define WM8900_REG_INCTL 0x15
+#define WM8900_REG_LINVOL 0x16
+#define WM8900_REG_RINVOL 0x17
+#define WM8900_REG_INBOOSTMIX1 0x18
+#define WM8900_REG_INBOOSTMIX2 0x19
+#define WM8900_REG_ADCPATH 0x1a
+#define WM8900_REG_AUXBOOST 0x1b
+#define WM8900_REG_ADDCTL 0x1e
+#define WM8900_REG_FLLCTL1 0x24
+#define WM8900_REG_FLLCTL2 0x25
+#define WM8900_REG_FLLCTL3 0x26
+#define WM8900_REG_FLLCTL4 0x27
+#define WM8900_REG_FLLCTL5 0x28
+#define WM8900_REG_FLLCTL6 0x29
+#define WM8900_REG_LOUTMIXCTL1 0x2c
+#define WM8900_REG_ROUTMIXCTL1 0x2d
+#define WM8900_REG_BYPASS1 0x2e
+#define WM8900_REG_BYPASS2 0x2f
+#define WM8900_REG_AUXOUT_CTL 0x30
+#define WM8900_REG_LOUT1CTL 0x33
+#define WM8900_REG_ROUT1CTL 0x34
+#define WM8900_REG_LOUT2CTL 0x35
+#define WM8900_REG_ROUT2CTL 0x36
+#define WM8900_REG_HPCTL1 0x3a
+#define WM8900_REG_OUTBIASCTL 0x73
+
+#define WM8900_MAXREG 0x80
+
+#define WM8900_REG_ADDCTL_OUT1_DIS 0x80
+#define WM8900_REG_ADDCTL_OUT2_DIS 0x40
+#define WM8900_REG_ADDCTL_VMID_DIS 0x20
+#define WM8900_REG_ADDCTL_BIAS_SRC 0x10
+#define WM8900_REG_ADDCTL_VMID_SOFTST 0x04
+#define WM8900_REG_ADDCTL_TEMP_SD 0x02
+
+#define WM8900_REG_GPIO_TEMP_ENA 0x2
+
+#define WM8900_REG_POWER1_STARTUP_BIAS_ENA 0x0100
+#define WM8900_REG_POWER1_BIAS_ENA 0x0008
+#define WM8900_REG_POWER1_VMID_BUF_ENA 0x0004
+#define WM8900_REG_POWER1_FLL_ENA 0x0040
+
+#define WM8900_REG_POWER2_SYSCLK_ENA 0x8000
+#define WM8900_REG_POWER2_ADCL_ENA 0x0002
+#define WM8900_REG_POWER2_ADCR_ENA 0x0001
+
+#define WM8900_REG_POWER3_DACL_ENA 0x0002
+#define WM8900_REG_POWER3_DACR_ENA 0x0001
+
+#define WM8900_REG_AUDIO1_AIF_FMT_MASK 0x0018
+#define WM8900_REG_AUDIO1_LRCLK_INV 0x0080
+#define WM8900_REG_AUDIO1_BCLK_INV 0x0100
+
+#define WM8900_REG_CLOCKING1_BCLK_DIR 0x1
+#define WM8900_REG_CLOCKING1_MCLK_SRC 0x100
+#define WM8900_REG_CLOCKING1_BCLK_MASK (~0x01e)
+#define WM8900_REG_CLOCKING1_OPCLK_MASK (~0x7000)
+
+#define WM8900_REG_CLOCKING2_ADC_CLKDIV 0xe0
+#define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c
+
+#define WM8900_REG_DACCTRL_MUTE 0x004
+#define WM8900_REG_DACCTRL_AIF_LRCLKRATE 0x400
+
+#define WM8900_REG_AUDIO3_ADCLRC_DIR 0x0800
+
+#define WM8900_REG_AUDIO4_DACLRC_DIR 0x0800
+
+#define WM8900_REG_FLLCTL1_OSC_ENA 0x100
+
+#define WM8900_REG_FLLCTL6_FLL_SLOW_LOCK_REF 0x100
+
+#define WM8900_REG_HPCTL1_HP_IPSTAGE_ENA 0x80
+#define WM8900_REG_HPCTL1_HP_OPSTAGE_ENA 0x40
+#define WM8900_REG_HPCTL1_HP_CLAMP_IP 0x20
+#define WM8900_REG_HPCTL1_HP_CLAMP_OP 0x10
+#define WM8900_REG_HPCTL1_HP_SHORT 0x08
+#define WM8900_REG_HPCTL1_HP_SHORT2 0x04
+
+#define WM8900_LRC_MASK 0xfc00
+
+struct snd_soc_codec_device soc_codec_dev_wm8900;
+
+struct wm8900_priv {
+ u32 fll_in; /* FLL input frequency */
+ u32 fll_out; /* FLL output frequency */
+};
+
+/*
+ * wm8900 register cache. We can't read the entire register space and we
+ * have slow control buses so we cache the registers.
+ */
+static const u16 wm8900_reg_defaults[WM8900_MAXREG] = {
+ 0x8900, 0x0000,
+ 0xc000, 0x0000,
+ 0x4050, 0x4000,
+ 0x0008, 0x0000,
+ 0x0040, 0x0040,
+ 0x1004, 0x00c0,
+ 0x00c0, 0x0000,
+ 0x0100, 0x00c0,
+ 0x00c0, 0x0000,
+ 0xb001, 0x0000,
+ 0x0000, 0x0044,
+ 0x004c, 0x004c,
+ 0x0044, 0x0044,
+ 0x0000, 0x0044,
+ 0x0000, 0x0000,
+ 0x0002, 0x0000,
+ 0x0000, 0x0000,
+ 0x0000, 0x0000,
+ 0x0008, 0x0000,
+ 0x0000, 0x0008,
+ 0x0097, 0x0100,
+ 0x0000, 0x0000,
+ 0x0050, 0x0050,
+ 0x0055, 0x0055,
+ 0x0055, 0x0000,
+ 0x0000, 0x0079,
+ 0x0079, 0x0079,
+ 0x0079, 0x0000,
+ /* Remaining registers all zero */
+};
+
+/*
+ * read wm8900 register cache
+ */
+static inline unsigned int wm8900_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ BUG_ON(reg >= WM8900_MAXREG);
+
+ if (reg == WM8900_REG_ID)
+ return 0;
+
+ return cache[reg];
+}
+
+/*
+ * write wm8900 register cache
+ */
+static inline void wm8900_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+
+ BUG_ON(reg >= WM8900_MAXREG);
+
+ cache[reg] = value;
+}
+
+/*
+ * write to the WM8900 register space
+ */
+static int wm8900_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+
+ if (value == wm8900_read_reg_cache(codec, reg))
+ return 0;
+
+ /* data is
+ * D15..D9 WM8900 register offset
+ * D8...D0 register data
+ */
+ data[0] = reg;
+ data[1] = value >> 8;
+ data[2] = value & 0x00ff;
+
+ wm8900_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 3) == 3)
+ return 0;
+ else
+ return -EIO;
+}
+
+/*
+ * Read from the wm8900.
+ */
+static unsigned int wm8900_chip_read(struct snd_soc_codec *codec, u8 reg)
+{
+ struct i2c_msg xfer[2];
+ u16 data;
+ int ret;
+ struct i2c_client *client = codec->control_data;
+
+ BUG_ON(reg != WM8900_REG_ID && reg != WM8900_REG_POWER1);
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 1;
+ xfer[0].buf = &reg;
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 2;
+ xfer[1].buf = (u8 *)&data;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret != 2) {
+ printk(KERN_CRIT "i2c_transfer returned %d\n", ret);
+ return 0;
+ }
+
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+
+/*
+ * Read from the WM8900 register space. Most registers can't be read
+ * and are therefore supplied from cache.
+ */
+static unsigned int wm8900_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ switch (reg) {
+ case WM8900_REG_ID:
+ return wm8900_chip_read(codec, reg);
+ default:
+ return wm8900_read_reg_cache(codec, reg);
+ }
+}
+
+static void wm8900_reset(struct snd_soc_codec *codec)
+{
+ wm8900_write(codec, WM8900_REG_RESET, 0);
+
+ memcpy(codec->reg_cache, wm8900_reg_defaults,
+ sizeof(codec->reg_cache));
+}
+
+static int wm8900_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 hpctl1 = wm8900_read(codec, WM8900_REG_HPCTL1);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Clamp headphone outputs */
+ hpctl1 = WM8900_REG_HPCTL1_HP_CLAMP_IP |
+ WM8900_REG_HPCTL1_HP_CLAMP_OP;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
+ /* Enable the input stage */
+ hpctl1 &= ~WM8900_REG_HPCTL1_HP_CLAMP_IP;
+ hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT |
+ WM8900_REG_HPCTL1_HP_SHORT2 |
+ WM8900_REG_HPCTL1_HP_IPSTAGE_ENA;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+
+ msleep(400);
+
+ /* Enable the output stage */
+ hpctl1 &= ~WM8900_REG_HPCTL1_HP_CLAMP_OP;
+ hpctl1 |= WM8900_REG_HPCTL1_HP_OPSTAGE_ENA;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+
+ /* Remove the shorts */
+ hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT2;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ /* Short the output */
+ hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+
+ /* Disable the output stage */
+ hpctl1 &= ~WM8900_REG_HPCTL1_HP_OPSTAGE_ENA;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+
+ /* Clamp the outputs and power down input */
+ hpctl1 |= WM8900_REG_HPCTL1_HP_CLAMP_IP |
+ WM8900_REG_HPCTL1_HP_CLAMP_OP;
+ hpctl1 &= ~WM8900_REG_HPCTL1_HP_IPSTAGE_ENA;
+ wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1);
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ /* Disable everything */
+ wm8900_write(codec, WM8900_REG_HPCTL1, 0);
+ break;
+
+ default:
+ BUG();
+ }
+
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -5700, 100, 0);
+
+static const DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 0);
+
+static const DECLARE_TLV_DB_SCALE(in_boost_tlv, -1200, 600, 0);
+
+static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1200, 100, 0);
+
+static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0);
+
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1);
+
+static const DECLARE_TLV_DB_SCALE(adc_svol_tlv, -3600, 300, 0);
+
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1);
+
+static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" };
+
+static const struct soc_enum mic_bias_level =
+SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt);
+
+static const char *dac_mute_rate_txt[] = { "Fast", "Slow" };
+
+static const struct soc_enum dac_mute_rate =
+SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt);
+
+static const char *dac_deemphasis_txt[] = {
+ "Disabled", "32kHz", "44.1kHz", "48kHz"
+};
+
+static const struct soc_enum dac_deemphasis =
+SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt);
+
+static const char *adc_hpf_cut_txt[] = {
+ "Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"
+};
+
+static const struct soc_enum adc_hpf_cut =
+SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt);
+
+static const char *lr_txt[] = {
+ "Left", "Right"
+};
+
+static const struct soc_enum aifl_src =
+SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt);
+
+static const struct soc_enum aifr_src =
+SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt);
+
+static const struct soc_enum dacl_src =
+SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt);
+
+static const struct soc_enum dacr_src =
+SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt);
+
+static const char *sidetone_txt[] = {
+ "Disabled", "Left ADC", "Right ADC"
+};
+
+static const struct soc_enum dacl_sidetone =
+SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt);
+
+static const struct soc_enum dacr_sidetone =
+SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt);
+
+static const struct snd_kcontrol_new wm8900_snd_controls[] = {
+SOC_ENUM("Mic Bias Level", mic_bias_level),
+
+SOC_SINGLE_TLV("Left Input PGA Volume", WM8900_REG_LINVOL, 0, 31, 0,
+ in_pga_tlv),
+SOC_SINGLE("Left Input PGA Switch", WM8900_REG_LINVOL, 6, 1, 1),
+SOC_SINGLE("Left Input PGA ZC Switch", WM8900_REG_LINVOL, 7, 1, 0),
+
+SOC_SINGLE_TLV("Right Input PGA Volume", WM8900_REG_RINVOL, 0, 31, 0,
+ in_pga_tlv),
+SOC_SINGLE("Right Input PGA Switch", WM8900_REG_RINVOL, 6, 1, 1),
+SOC_SINGLE("Right Input PGA ZC Switch", WM8900_REG_RINVOL, 7, 1, 0),
+
+SOC_SINGLE("DAC Soft Mute Switch", WM8900_REG_DACCTRL, 6, 1, 1),
+SOC_ENUM("DAC Mute Rate", dac_mute_rate),
+SOC_SINGLE("DAC Mono Switch", WM8900_REG_DACCTRL, 9, 1, 0),
+SOC_ENUM("DAC Deemphasis", dac_deemphasis),
+SOC_SINGLE("DAC Sloping Stopband Filter Switch", WM8900_REG_DACCTRL, 8, 1, 0),
+SOC_SINGLE("DAC Sigma-Delta Modulator Clock Switch", WM8900_REG_DACCTRL,
+ 12, 1, 0),
+
+SOC_SINGLE("ADC HPF Switch", WM8900_REG_ADCCTRL, 8, 1, 0),
+SOC_ENUM("ADC HPF Cut-Off", adc_hpf_cut),
+SOC_DOUBLE("ADC Invert Switch", WM8900_REG_ADCCTRL, 1, 0, 1, 0),
+SOC_SINGLE_TLV("Left ADC Sidetone Volume", WM8900_REG_SIDETONE, 9, 12, 0,
+ adc_svol_tlv),
+SOC_SINGLE_TLV("Right ADC Sidetone Volume", WM8900_REG_SIDETONE, 5, 12, 0,
+ adc_svol_tlv),
+SOC_ENUM("Left Digital Audio Source", aifl_src),
+SOC_ENUM("Right Digital Audio Source", aifr_src),
+
+SOC_SINGLE_TLV("DAC Input Boost Volume", WM8900_REG_AUDIO2, 10, 4, 0,
+ dac_boost_tlv),
+SOC_ENUM("Left DAC Source", dacl_src),
+SOC_ENUM("Right DAC Source", dacr_src),
+SOC_ENUM("Left DAC Sidetone", dacl_sidetone),
+SOC_ENUM("Right DAC Sidetone", dacr_sidetone),
+SOC_DOUBLE("DAC Invert Switch", WM8900_REG_DACCTRL, 1, 0, 1, 0),
+
+SOC_DOUBLE_R_TLV("Digital Playback Volume",
+ WM8900_REG_LDAC_DV, WM8900_REG_RDAC_DV,
+ 1, 96, 0, dac_tlv),
+SOC_DOUBLE_R_TLV("Digital Capture Volume",
+ WM8900_REG_LADC_DV, WM8900_REG_RADC_DV, 1, 119, 0, adc_tlv),
+
+SOC_SINGLE_TLV("LINPUT3 Bypass Volume", WM8900_REG_LOUTMIXCTL1, 4, 7, 0,
+ out_mix_tlv),
+SOC_SINGLE_TLV("RINPUT3 Bypass Volume", WM8900_REG_ROUTMIXCTL1, 4, 7, 0,
+ out_mix_tlv),
+SOC_SINGLE_TLV("Left AUX Bypass Volume", WM8900_REG_AUXOUT_CTL, 4, 7, 0,
+ out_mix_tlv),
+SOC_SINGLE_TLV("Right AUX Bypass Volume", WM8900_REG_AUXOUT_CTL, 0, 7, 0,
+ out_mix_tlv),
+
+SOC_SINGLE_TLV("LeftIn to RightOut Mixer Volume", WM8900_REG_BYPASS1, 0, 7, 0,
+ out_mix_tlv),
+SOC_SINGLE_TLV("LeftIn to LeftOut Mixer Volume", WM8900_REG_BYPASS1, 4, 7, 0,
+ out_mix_tlv),
+SOC_SINGLE_TLV("RightIn to LeftOut Mixer Volume", WM8900_REG_BYPASS2, 0, 7, 0,
+ out_mix_tlv),
+SOC_SINGLE_TLV("RightIn to RightOut Mixer Volume", WM8900_REG_BYPASS2, 4, 7, 0,
+ out_mix_tlv),
+
+SOC_SINGLE_TLV("IN2L Boost Volume", WM8900_REG_INBOOSTMIX1, 0, 3, 0,
+ in_boost_tlv),
+SOC_SINGLE_TLV("IN3L Boost Volume", WM8900_REG_INBOOSTMIX1, 4, 3, 0,
+ in_boost_tlv),
+SOC_SINGLE_TLV("IN2R Boost Volume", WM8900_REG_INBOOSTMIX2, 0, 3, 0,
+ in_boost_tlv),
+SOC_SINGLE_TLV("IN3R Boost Volume", WM8900_REG_INBOOSTMIX2, 4, 3, 0,
+ in_boost_tlv),
+SOC_SINGLE_TLV("Left AUX Boost Volume", WM8900_REG_AUXBOOST, 4, 3, 0,
+ in_boost_tlv),
+SOC_SINGLE_TLV("Right AUX Boost Volume", WM8900_REG_AUXBOOST, 0, 3, 0,
+ in_boost_tlv),
+
+SOC_DOUBLE_R_TLV("LINEOUT1 Volume", WM8900_REG_LOUT1CTL, WM8900_REG_ROUT1CTL,
+ 0, 63, 0, out_pga_tlv),
+SOC_DOUBLE_R("LINEOUT1 Switch", WM8900_REG_LOUT1CTL, WM8900_REG_ROUT1CTL,
+ 6, 1, 1),
+SOC_DOUBLE_R("LINEOUT1 ZC Switch", WM8900_REG_LOUT1CTL, WM8900_REG_ROUT1CTL,
+ 7, 1, 0),
+
+SOC_DOUBLE_R_TLV("LINEOUT2 Volume",
+ WM8900_REG_LOUT2CTL, WM8900_REG_ROUT2CTL,
+ 0, 63, 0, out_pga_tlv),
+SOC_DOUBLE_R("LINEOUT2 Switch",
+ WM8900_REG_LOUT2CTL, WM8900_REG_ROUT2CTL, 6, 1, 1),
+SOC_DOUBLE_R("LINEOUT2 ZC Switch",
+ WM8900_REG_LOUT2CTL, WM8900_REG_ROUT2CTL, 7, 1, 0),
+SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1,
+ 0, 1, 1),
+
+};
+
+/* add non dapm controls */
+static int wm8900_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8900_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new wm8900_dapm_loutput2_control =
+SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0);
+
+static const struct snd_kcontrol_new wm8900_dapm_routput2_control =
+SOC_DAPM_SINGLE("LINEOUT2R Switch", WM8900_REG_POWER3, 5, 1, 0);
+
+static const struct snd_kcontrol_new wm8900_loutmix_controls[] = {
+SOC_DAPM_SINGLE("LINPUT3 Bypass Switch", WM8900_REG_LOUTMIXCTL1, 7, 1, 0),
+SOC_DAPM_SINGLE("AUX Bypass Switch", WM8900_REG_AUXOUT_CTL, 7, 1, 0),
+SOC_DAPM_SINGLE("Left Input Mixer Switch", WM8900_REG_BYPASS1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Input Mixer Switch", WM8900_REG_BYPASS2, 3, 1, 0),
+SOC_DAPM_SINGLE("DACL Switch", WM8900_REG_LOUTMIXCTL1, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8900_routmix_controls[] = {
+SOC_DAPM_SINGLE("RINPUT3 Bypass Switch", WM8900_REG_ROUTMIXCTL1, 7, 1, 0),
+SOC_DAPM_SINGLE("AUX Bypass Switch", WM8900_REG_AUXOUT_CTL, 3, 1, 0),
+SOC_DAPM_SINGLE("Left Input Mixer Switch", WM8900_REG_BYPASS1, 3, 1, 0),
+SOC_DAPM_SINGLE("Right Input Mixer Switch", WM8900_REG_BYPASS2, 7, 1, 0),
+SOC_DAPM_SINGLE("DACR Switch", WM8900_REG_ROUTMIXCTL1, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8900_linmix_controls[] = {
+SOC_DAPM_SINGLE("LINPUT2 Switch", WM8900_REG_INBOOSTMIX1, 2, 1, 1),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8900_REG_INBOOSTMIX1, 6, 1, 1),
+SOC_DAPM_SINGLE("AUX Switch", WM8900_REG_AUXBOOST, 6, 1, 1),
+SOC_DAPM_SINGLE("Input PGA Switch", WM8900_REG_ADCPATH, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8900_rinmix_controls[] = {
+SOC_DAPM_SINGLE("RINPUT2 Switch", WM8900_REG_INBOOSTMIX2, 2, 1, 1),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INBOOSTMIX2, 6, 1, 1),
+SOC_DAPM_SINGLE("AUX Switch", WM8900_REG_AUXBOOST, 2, 1, 1),
+SOC_DAPM_SINGLE("Input PGA Switch", WM8900_REG_ADCPATH, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8900_linpga_controls[] = {
+SOC_DAPM_SINGLE("LINPUT1 Switch", WM8900_REG_INCTL, 6, 1, 0),
+SOC_DAPM_SINGLE("LINPUT2 Switch", WM8900_REG_INCTL, 5, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8900_REG_INCTL, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8900_rinpga_controls[] = {
+SOC_DAPM_SINGLE("RINPUT1 Switch", WM8900_REG_INCTL, 2, 1, 0),
+SOC_DAPM_SINGLE("RINPUT2 Switch", WM8900_REG_INCTL, 1, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0),
+};
+
+static const char *wm9700_lp_mux[] = { "Disabled", "Enabled" };
+
+static const struct soc_enum wm8900_lineout2_lp_mux =
+SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm9700_lp_mux);
+
+static const struct snd_kcontrol_new wm8900_lineout2_lp =
+SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux);
+
+static const struct snd_soc_dapm_widget wm8900_dapm_widgets[] = {
+
+/* Externally visible pins */
+SND_SOC_DAPM_OUTPUT("LINEOUT1L"),
+SND_SOC_DAPM_OUTPUT("LINEOUT1R"),
+SND_SOC_DAPM_OUTPUT("LINEOUT2L"),
+SND_SOC_DAPM_OUTPUT("LINEOUT2R"),
+SND_SOC_DAPM_OUTPUT("HP_L"),
+SND_SOC_DAPM_OUTPUT("HP_R"),
+
+SND_SOC_DAPM_INPUT("RINPUT1"),
+SND_SOC_DAPM_INPUT("LINPUT1"),
+SND_SOC_DAPM_INPUT("RINPUT2"),
+SND_SOC_DAPM_INPUT("LINPUT2"),
+SND_SOC_DAPM_INPUT("RINPUT3"),
+SND_SOC_DAPM_INPUT("LINPUT3"),
+SND_SOC_DAPM_INPUT("AUX"),
+
+SND_SOC_DAPM_VMID("VMID"),
+
+/* Input */
+SND_SOC_DAPM_MIXER("Left Input PGA", WM8900_REG_POWER2, 3, 0,
+ wm8900_linpga_controls,
+ ARRAY_SIZE(wm8900_linpga_controls)),
+SND_SOC_DAPM_MIXER("Right Input PGA", WM8900_REG_POWER2, 2, 0,
+ wm8900_rinpga_controls,
+ ARRAY_SIZE(wm8900_rinpga_controls)),
+
+SND_SOC_DAPM_MIXER("Left Input Mixer", WM8900_REG_POWER2, 5, 0,
+ wm8900_linmix_controls,
+ ARRAY_SIZE(wm8900_linmix_controls)),
+SND_SOC_DAPM_MIXER("Right Input Mixer", WM8900_REG_POWER2, 4, 0,
+ wm8900_rinmix_controls,
+ ARRAY_SIZE(wm8900_rinmix_controls)),
+
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8900_REG_POWER1, 4, 0),
+
+SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8900_REG_POWER2, 1, 0),
+SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8900_REG_POWER2, 0, 0),
+
+/* Output */
+SND_SOC_DAPM_DAC("DACL", "Left HiFi Playback", WM8900_REG_POWER3, 1, 0),
+SND_SOC_DAPM_DAC("DACR", "Right HiFi Playback", WM8900_REG_POWER3, 0, 0),
+
+SND_SOC_DAPM_PGA_E("Headphone Amplifier", WM8900_REG_POWER3, 7, 0, NULL, 0,
+ wm8900_hp_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_PGA("LINEOUT1L PGA", WM8900_REG_POWER2, 8, 0, NULL, 0),
+SND_SOC_DAPM_PGA("LINEOUT1R PGA", WM8900_REG_POWER2, 7, 0, NULL, 0),
+
+SND_SOC_DAPM_MUX("LINEOUT2 LP", SND_SOC_NOPM, 0, 0, &wm8900_lineout2_lp),
+SND_SOC_DAPM_PGA("LINEOUT2L PGA", WM8900_REG_POWER3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("LINEOUT2R PGA", WM8900_REG_POWER3, 5, 0, NULL, 0),
+
+SND_SOC_DAPM_MIXER("Left Output Mixer", WM8900_REG_POWER3, 3, 0,
+ wm8900_loutmix_controls,
+ ARRAY_SIZE(wm8900_loutmix_controls)),
+SND_SOC_DAPM_MIXER("Right Output Mixer", WM8900_REG_POWER3, 2, 0,
+ wm8900_routmix_controls,
+ ARRAY_SIZE(wm8900_routmix_controls)),
+};
+
+/* Target, Path, Source */
+static const struct snd_soc_dapm_route audio_map[] = {
+/* Inputs */
+{"Left Input PGA", "LINPUT1 Switch", "LINPUT1"},
+{"Left Input PGA", "LINPUT2 Switch", "LINPUT2"},
+{"Left Input PGA", "LINPUT3 Switch", "LINPUT3"},
+
+{"Right Input PGA", "RINPUT1 Switch", "RINPUT1"},
+{"Right Input PGA", "RINPUT2 Switch", "RINPUT2"},
+{"Right Input PGA", "RINPUT3 Switch", "RINPUT3"},
+
+{"Left Input Mixer", "LINPUT2 Switch", "LINPUT2"},
+{"Left Input Mixer", "LINPUT3 Switch", "LINPUT3"},
+{"Left Input Mixer", "AUX Switch", "AUX"},
+{"Left Input Mixer", "Input PGA Switch", "Left Input PGA"},
+
+{"Right Input Mixer", "RINPUT2 Switch", "RINPUT2"},
+{"Right Input Mixer", "RINPUT3 Switch", "RINPUT3"},
+{"Right Input Mixer", "AUX Switch", "AUX"},
+{"Right Input Mixer", "Input PGA Switch", "Right Input PGA"},
+
+{"ADCL", NULL, "Left Input Mixer"},
+{"ADCR", NULL, "Right Input Mixer"},
+
+/* Outputs */
+{"LINEOUT1L", NULL, "LINEOUT1L PGA"},
+{"LINEOUT1L PGA", NULL, "Left Output Mixer"},
+{"LINEOUT1R", NULL, "LINEOUT1R PGA"},
+{"LINEOUT1R PGA", NULL, "Right Output Mixer"},
+
+{"LINEOUT2L PGA", NULL, "Left Output Mixer"},
+{"LINEOUT2 LP", "Disabled", "LINEOUT2L PGA"},
+{"LINEOUT2 LP", "Enabled", "Left Output Mixer"},
+{"LINEOUT2L", NULL, "LINEOUT2 LP"},
+
+{"LINEOUT2R PGA", NULL, "Right Output Mixer"},
+{"LINEOUT2 LP", "Disabled", "LINEOUT2R PGA"},
+{"LINEOUT2 LP", "Enabled", "Right Output Mixer"},
+{"LINEOUT2R", NULL, "LINEOUT2 LP"},
+
+{"Left Output Mixer", "LINPUT3 Bypass Switch", "LINPUT3"},
+{"Left Output Mixer", "AUX Bypass Switch", "AUX"},
+{"Left Output Mixer", "Left Input Mixer Switch", "Left Input Mixer"},
+{"Left Output Mixer", "Right Input Mixer Switch", "Right Input Mixer"},
+{"Left Output Mixer", "DACL Switch", "DACL"},
+
+{"Right Output Mixer", "RINPUT3 Bypass Switch", "RINPUT3"},
+{"Right Output Mixer", "AUX Bypass Switch", "AUX"},
+{"Right Output Mixer", "Left Input Mixer Switch", "Left Input Mixer"},
+{"Right Output Mixer", "Right Input Mixer Switch", "Right Input Mixer"},
+{"Right Output Mixer", "DACR Switch", "DACR"},
+
+/* Note that the headphone output stage needs to be connected
+ * externally to LINEOUT2 via DC blocking capacitors. Other
+ * configurations are not supported.
+ *
+ * Note also that left and right headphone paths are treated as a
+ * mono path.
+ */
+{"Headphone Amplifier", NULL, "LINEOUT2 LP"},
+{"Headphone Amplifier", NULL, "LINEOUT2 LP"},
+{"HP_L", NULL, "Headphone Amplifier"},
+{"HP_R", NULL, "Headphone Amplifier"},
+};
+
+static int wm8900_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8900_dapm_widgets,
+ ARRAY_SIZE(wm8900_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+
+ return 0;
+}
+
+static int wm8900_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 reg;
+
+ reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ reg |= 0x20;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ reg |= 0x40;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ reg |= 0x60;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8900_write(codec, WM8900_REG_AUDIO1, reg);
+
+ return 0;
+}
+
+/* FLL divisors */
+struct _fll_div {
+ u16 fll_ratio;
+ u16 fllclk_div;
+ u16 fll_slow_lock_ref;
+ u16 n;
+ u16 k;
+};
+
+/* The size in bits of the FLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
+ unsigned int Fout)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod, target;
+ unsigned int div;
+
+ BUG_ON(!Fout);
+
+ /* The FLL must run at 90-100MHz which is then scaled down to
+ * the output value by FLLCLK_DIV. */
+ target = Fout;
+ div = 1;
+ while (target < 90000000) {
+ div *= 2;
+ target *= 2;
+ }
+
+ if (target > 100000000)
+ printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d"
+ " Fout=%d\n", target, Fref, Fout);
+ if (div > 32) {
+ printk(KERN_ERR "wm8900: Invalid FLL division rate %u, "
+ "Fref=%d, Fout=%d, target=%d\n",
+ div, Fref, Fout, target);
+ return -EINVAL;
+ }
+
+ fll_div->fllclk_div = div >> 2;
+
+ if (Fref < 48000)
+ fll_div->fll_slow_lock_ref = 1;
+ else
+ fll_div->fll_slow_lock_ref = 0;
+
+ Ndiv = target / Fref;
+
+ if (Fref < 1000000)
+ fll_div->fll_ratio = 8;
+ else
+ fll_div->fll_ratio = 1;
+
+ fll_div->n = Ndiv / fll_div->fll_ratio;
+ Nmod = (target / fll_div->fll_ratio) % Fref;
+
+ /* Calculate fractional part - scale up so we can round. */
+ Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, Fref);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ fll_div->k = K / 10;
+
+ BUG_ON(target != Fout * (fll_div->fllclk_div << 2));
+ BUG_ON(!K && target != Fref * fll_div->fll_ratio * fll_div->n);
+
+ return 0;
+}
+
+static int wm8900_set_fll(struct snd_soc_codec *codec,
+ int fll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct wm8900_priv *wm8900 = codec->private_data;
+ struct _fll_div fll_div;
+ unsigned int reg;
+
+ if (wm8900->fll_in == freq_in && wm8900->fll_out == freq_out)
+ return 0;
+
+ /* The digital side should be disabled during any change. */
+ reg = wm8900_read(codec, WM8900_REG_POWER1);
+ wm8900_write(codec, WM8900_REG_POWER1,
+ reg & (~WM8900_REG_POWER1_FLL_ENA));
+
+ /* Disable the FLL? */
+ if (!freq_in || !freq_out) {
+ reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
+ wm8900_write(codec, WM8900_REG_CLOCKING1,
+ reg & (~WM8900_REG_CLOCKING1_MCLK_SRC));
+
+ reg = wm8900_read(codec, WM8900_REG_FLLCTL1);
+ wm8900_write(codec, WM8900_REG_FLLCTL1,
+ reg & (~WM8900_REG_FLLCTL1_OSC_ENA));
+
+ wm8900->fll_in = freq_in;
+ wm8900->fll_out = freq_out;
+
+ return 0;
+ }
+
+ if (fll_factors(&fll_div, freq_in, freq_out) != 0)
+ goto reenable;
+
+ wm8900->fll_in = freq_in;
+ wm8900->fll_out = freq_out;
+
+ /* The osclilator *MUST* be enabled before we enable the
+ * digital circuit. */
+ wm8900_write(codec, WM8900_REG_FLLCTL1,
+ fll_div.fll_ratio | WM8900_REG_FLLCTL1_OSC_ENA);
+
+ wm8900_write(codec, WM8900_REG_FLLCTL4, fll_div.n >> 5);
+ wm8900_write(codec, WM8900_REG_FLLCTL5,
+ (fll_div.fllclk_div << 6) | (fll_div.n & 0x1f));
+
+ if (fll_div.k) {
+ wm8900_write(codec, WM8900_REG_FLLCTL2,
+ (fll_div.k >> 8) | 0x100);
+ wm8900_write(codec, WM8900_REG_FLLCTL3, fll_div.k & 0xff);
+ } else
+ wm8900_write(codec, WM8900_REG_FLLCTL2, 0);
+
+ if (fll_div.fll_slow_lock_ref)
+ wm8900_write(codec, WM8900_REG_FLLCTL6,
+ WM8900_REG_FLLCTL6_FLL_SLOW_LOCK_REF);
+ else
+ wm8900_write(codec, WM8900_REG_FLLCTL6, 0);
+
+ reg = wm8900_read(codec, WM8900_REG_POWER1);
+ wm8900_write(codec, WM8900_REG_POWER1,
+ reg | WM8900_REG_POWER1_FLL_ENA);
+
+reenable:
+ reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
+ wm8900_write(codec, WM8900_REG_CLOCKING1,
+ reg | WM8900_REG_CLOCKING1_MCLK_SRC);
+
+ return 0;
+}
+
+static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out);
+}
+
+static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int reg;
+
+ switch (div_id) {
+ case WM8900_BCLK_DIV:
+ reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
+ wm8900_write(codec, WM8900_REG_CLOCKING1,
+ div | (reg & WM8900_REG_CLOCKING1_BCLK_MASK));
+ break;
+ case WM8900_OPCLK_DIV:
+ reg = wm8900_read(codec, WM8900_REG_CLOCKING1);
+ wm8900_write(codec, WM8900_REG_CLOCKING1,
+ div | (reg & WM8900_REG_CLOCKING1_OPCLK_MASK));
+ break;
+ case WM8900_DAC_LRCLK:
+ reg = wm8900_read(codec, WM8900_REG_AUDIO4);
+ wm8900_write(codec, WM8900_REG_AUDIO4,
+ div | (reg & WM8900_LRC_MASK));
+ break;
+ case WM8900_ADC_LRCLK:
+ reg = wm8900_read(codec, WM8900_REG_AUDIO3);
+ wm8900_write(codec, WM8900_REG_AUDIO3,
+ div | (reg & WM8900_LRC_MASK));
+ break;
+ case WM8900_DAC_CLKDIV:
+ reg = wm8900_read(codec, WM8900_REG_CLOCKING2);
+ wm8900_write(codec, WM8900_REG_CLOCKING2,
+ div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV));
+ break;
+ case WM8900_ADC_CLKDIV:
+ reg = wm8900_read(codec, WM8900_REG_CLOCKING2);
+ wm8900_write(codec, WM8900_REG_CLOCKING2,
+ div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV));
+ break;
+ case WM8900_LRCLK_MODE:
+ reg = wm8900_read(codec, WM8900_REG_DACCTRL);
+ wm8900_write(codec, WM8900_REG_DACCTRL,
+ div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE));
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+static int wm8900_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int clocking1, aif1, aif3, aif4;
+
+ clocking1 = wm8900_read(codec, WM8900_REG_CLOCKING1);
+ aif1 = wm8900_read(codec, WM8900_REG_AUDIO1);
+ aif3 = wm8900_read(codec, WM8900_REG_AUDIO3);
+ aif4 = wm8900_read(codec, WM8900_REG_AUDIO4);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ clocking1 &= ~WM8900_REG_CLOCKING1_BCLK_DIR;
+ aif3 &= ~WM8900_REG_AUDIO3_ADCLRC_DIR;
+ aif4 &= ~WM8900_REG_AUDIO4_DACLRC_DIR;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ clocking1 &= ~WM8900_REG_CLOCKING1_BCLK_DIR;
+ aif3 |= WM8900_REG_AUDIO3_ADCLRC_DIR;
+ aif4 |= WM8900_REG_AUDIO4_DACLRC_DIR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ clocking1 |= WM8900_REG_CLOCKING1_BCLK_DIR;
+ aif3 |= WM8900_REG_AUDIO3_ADCLRC_DIR;
+ aif4 |= WM8900_REG_AUDIO4_DACLRC_DIR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ clocking1 |= WM8900_REG_CLOCKING1_BCLK_DIR;
+ aif3 &= ~WM8900_REG_AUDIO3_ADCLRC_DIR;
+ aif4 &= ~WM8900_REG_AUDIO4_DACLRC_DIR;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ aif1 |= WM8900_REG_AUDIO1_AIF_FMT_MASK;
+ aif1 &= ~WM8900_REG_AUDIO1_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ aif1 |= WM8900_REG_AUDIO1_AIF_FMT_MASK;
+ aif1 |= WM8900_REG_AUDIO1_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ aif1 &= ~WM8900_REG_AUDIO1_AIF_FMT_MASK;
+ aif1 |= 0x10;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ aif1 &= ~WM8900_REG_AUDIO1_AIF_FMT_MASK;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ aif1 &= ~WM8900_REG_AUDIO1_AIF_FMT_MASK;
+ aif1 |= 0x8;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /* frame inversion not valid for DSP modes */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ aif1 &= ~WM8900_REG_AUDIO1_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif1 |= WM8900_REG_AUDIO1_BCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ aif1 &= ~WM8900_REG_AUDIO1_BCLK_INV;
+ aif1 &= ~WM8900_REG_AUDIO1_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ aif1 |= WM8900_REG_AUDIO1_BCLK_INV;
+ aif1 |= WM8900_REG_AUDIO1_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif1 |= WM8900_REG_AUDIO1_BCLK_INV;
+ aif1 &= ~WM8900_REG_AUDIO1_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ aif1 &= ~WM8900_REG_AUDIO1_BCLK_INV;
+ aif1 |= WM8900_REG_AUDIO1_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8900_write(codec, WM8900_REG_CLOCKING1, clocking1);
+ wm8900_write(codec, WM8900_REG_AUDIO1, aif1);
+ wm8900_write(codec, WM8900_REG_AUDIO3, aif3);
+ wm8900_write(codec, WM8900_REG_AUDIO4, aif4);
+
+ return 0;
+}
+
+static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ reg = wm8900_read(codec, WM8900_REG_DACCTRL);
+
+ if (mute)
+ reg |= WM8900_REG_DACCTRL_MUTE;
+ else
+ reg &= ~WM8900_REG_DACCTRL_MUTE;
+
+ wm8900_write(codec, WM8900_REG_DACCTRL, reg);
+
+ return 0;
+}
+
+#define WM8900_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+#define WM8900_PCM_FORMATS \
+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
+ SNDRV_PCM_FORMAT_S24_LE)
+
+struct snd_soc_dai wm8900_dai = {
+ .name = "WM8900 HiFi",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8900_RATES,
+ .formats = WM8900_PCM_FORMATS,
+ },
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8900_RATES,
+ .formats = WM8900_PCM_FORMATS,
+ },
+ .ops = {
+ .hw_params = wm8900_hw_params,
+ },
+ .dai_ops = {
+ .set_clkdiv = wm8900_set_dai_clkdiv,
+ .set_pll = wm8900_set_dai_pll,
+ .set_fmt = wm8900_set_dai_fmt,
+ .digital_mute = wm8900_digital_mute,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8900_dai);
+
+static int wm8900_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* Enable thermal shutdown */
+ reg = wm8900_read(codec, WM8900_REG_GPIO);
+ wm8900_write(codec, WM8900_REG_GPIO,
+ reg | WM8900_REG_GPIO_TEMP_ENA);
+ reg = wm8900_read(codec, WM8900_REG_ADDCTL);
+ wm8900_write(codec, WM8900_REG_ADDCTL,
+ reg | WM8900_REG_ADDCTL_TEMP_SD);
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ /* Charge capacitors if initial power up */
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* STARTUP_BIAS_ENA on */
+ wm8900_write(codec, WM8900_REG_POWER1,
+ WM8900_REG_POWER1_STARTUP_BIAS_ENA);
+
+ /* Startup bias mode */
+ wm8900_write(codec, WM8900_REG_ADDCTL,
+ WM8900_REG_ADDCTL_BIAS_SRC |
+ WM8900_REG_ADDCTL_VMID_SOFTST);
+
+ /* VMID 2x50k */
+ wm8900_write(codec, WM8900_REG_POWER1,
+ WM8900_REG_POWER1_STARTUP_BIAS_ENA | 0x1);
+
+ /* Allow capacitors to charge */
+ schedule_timeout_interruptible(msecs_to_jiffies(400));
+
+ /* Enable bias */
+ wm8900_write(codec, WM8900_REG_POWER1,
+ WM8900_REG_POWER1_STARTUP_BIAS_ENA |
+ WM8900_REG_POWER1_BIAS_ENA | 0x1);
+
+ wm8900_write(codec, WM8900_REG_ADDCTL, 0);
+
+ wm8900_write(codec, WM8900_REG_POWER1,
+ WM8900_REG_POWER1_BIAS_ENA | 0x1);
+ }
+
+ reg = wm8900_read(codec, WM8900_REG_POWER1);
+ wm8900_write(codec, WM8900_REG_POWER1,
+ (reg & WM8900_REG_POWER1_FLL_ENA) |
+ WM8900_REG_POWER1_BIAS_ENA | 0x1);
+ wm8900_write(codec, WM8900_REG_POWER2,
+ WM8900_REG_POWER2_SYSCLK_ENA);
+ wm8900_write(codec, WM8900_REG_POWER3, 0);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* Startup bias enable */
+ reg = wm8900_read(codec, WM8900_REG_POWER1);
+ wm8900_write(codec, WM8900_REG_POWER1,
+ reg & WM8900_REG_POWER1_STARTUP_BIAS_ENA);
+ wm8900_write(codec, WM8900_REG_ADDCTL,
+ WM8900_REG_ADDCTL_BIAS_SRC |
+ WM8900_REG_ADDCTL_VMID_SOFTST);
+
+ /* Discharge caps */
+ wm8900_write(codec, WM8900_REG_POWER1,
+ WM8900_REG_POWER1_STARTUP_BIAS_ENA);
+ schedule_timeout_interruptible(msecs_to_jiffies(500));
+
+ /* Remove clamp */
+ wm8900_write(codec, WM8900_REG_HPCTL1, 0);
+
+ /* Power down */
+ wm8900_write(codec, WM8900_REG_ADDCTL, 0);
+ wm8900_write(codec, WM8900_REG_POWER1, 0);
+ wm8900_write(codec, WM8900_REG_POWER2, 0);
+ wm8900_write(codec, WM8900_REG_POWER3, 0);
+
+ /* Need to let things settle before stopping the clock
+ * to ensure that restart works, see "Stopping the
+ * master clock" in the datasheet. */
+ schedule_timeout_interruptible(msecs_to_jiffies(1));
+ wm8900_write(codec, WM8900_REG_POWER2,
+ WM8900_REG_POWER2_SYSCLK_ENA);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static int wm8900_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8900_priv *wm8900 = codec->private_data;
+ int fll_out = wm8900->fll_out;
+ int fll_in = wm8900->fll_in;
+ int ret;
+
+ /* Stop the FLL in an orderly fashion */
+ ret = wm8900_set_fll(codec, 0, 0, 0);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "Failed to stop FLL\n");
+ return ret;
+ }
+
+ wm8900->fll_out = fll_out;
+ wm8900->fll_in = fll_in;
+
+ wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm8900_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8900_priv *wm8900 = codec->private_data;
+ u16 *cache;
+ int i, ret;
+
+ cache = kmemdup(codec->reg_cache, sizeof(wm8900_reg_defaults),
+ GFP_KERNEL);
+
+ wm8900_reset(codec);
+ wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Restart the FLL? */
+ if (wm8900->fll_out) {
+ int fll_out = wm8900->fll_out;
+ int fll_in = wm8900->fll_in;
+
+ wm8900->fll_in = 0;
+ wm8900->fll_out = 0;
+
+ ret = wm8900_set_fll(codec, 0, fll_in, fll_out);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "Failed to restart FLL\n");
+ return ret;
+ }
+ }
+
+ if (cache) {
+ for (i = 0; i < WM8900_MAXREG; i++)
+ wm8900_write(codec, i, cache[i]);
+ kfree(cache);
+ } else
+ dev_err(&pdev->dev, "Unable to allocate register cache\n");
+
+ return 0;
+}
+
+/*
+ * initialise the WM8900 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8900_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+ unsigned int reg;
+ struct i2c_client *i2c_client = socdev->codec->control_data;
+
+ codec->name = "WM8900";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8900_read;
+ codec->write = wm8900_write;
+ codec->dai = &wm8900_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8900_MAXREG;
+ codec->reg_cache = kmemdup(wm8900_reg_defaults,
+ sizeof(wm8900_reg_defaults), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ reg = wm8900_read(codec, WM8900_REG_ID);
+ if (reg != 0x8900) {
+ dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n",
+ reg);
+ return -ENODEV;
+ }
+
+ codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
+ if (codec->private_data == NULL) {
+ ret = -ENOMEM;
+ goto priv_err;
+ }
+
+ /* Read back from the chip */
+ reg = wm8900_chip_read(codec, WM8900_REG_POWER1);
+ reg = (reg >> 12) & 0xf;
+ dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg);
+
+ wm8900_reset(codec);
+
+ /* Latch the volume update bits */
+ wm8900_write(codec, WM8900_REG_LINVOL,
+ wm8900_read(codec, WM8900_REG_LINVOL) | 0x100);
+ wm8900_write(codec, WM8900_REG_RINVOL,
+ wm8900_read(codec, WM8900_REG_RINVOL) | 0x100);
+ wm8900_write(codec, WM8900_REG_LOUT1CTL,
+ wm8900_read(codec, WM8900_REG_LOUT1CTL) | 0x100);
+ wm8900_write(codec, WM8900_REG_ROUT1CTL,
+ wm8900_read(codec, WM8900_REG_ROUT1CTL) | 0x100);
+ wm8900_write(codec, WM8900_REG_LOUT2CTL,
+ wm8900_read(codec, WM8900_REG_LOUT2CTL) | 0x100);
+ wm8900_write(codec, WM8900_REG_ROUT2CTL,
+ wm8900_read(codec, WM8900_REG_ROUT2CTL) | 0x100);
+ wm8900_write(codec, WM8900_REG_LDAC_DV,
+ wm8900_read(codec, WM8900_REG_LDAC_DV) | 0x100);
+ wm8900_write(codec, WM8900_REG_RDAC_DV,
+ wm8900_read(codec, WM8900_REG_RDAC_DV) | 0x100);
+ wm8900_write(codec, WM8900_REG_LADC_DV,
+ wm8900_read(codec, WM8900_REG_LADC_DV) | 0x100);
+ wm8900_write(codec, WM8900_REG_RADC_DV,
+ wm8900_read(codec, WM8900_REG_RADC_DV) | 0x100);
+
+ /* Set the DAC and mixer output bias */
+ wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81);
+
+ /* Register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "Failed to register new PCMs\n");
+ goto pcm_err;
+ }
+
+ /* Turn the chip on */
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ wm8900_add_controls(codec);
+ wm8900_add_widgets(codec);
+
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ dev_err(&i2c_client->dev, "Failed to register card\n");
+ goto card_err;
+ }
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+priv_err:
+ kfree(codec->private_data);
+ return ret;
+}
+
+static struct snd_soc_device *wm8900_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8900_i2c_driver;
+static struct i2c_client client_template;
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+ struct snd_soc_device *socdev = wm8900_socdev;
+ struct wm8900_setup_data *setup = socdev->codec_data;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c;
+ int ret;
+
+ if (addr != setup->i2c_address)
+ return -ENODEV;
+
+ dev_err(&adap->dev, "Probe on %x\n", addr);
+
+ client_template.adapter = adap;
+ client_template.addr = addr;
+
+ i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+ if (i2c == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
+ dev_err(&adap->dev,
+ "failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = wm8900_init(socdev);
+ if (ret < 0) {
+ dev_err(&adap->dev, "failed to initialise WM8900\n");
+ goto err;
+ }
+ return ret;
+
+err:
+ kfree(codec);
+ kfree(i2c);
+ return ret;
+}
+
+static int wm8900_i2c_detach(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ i2c_detach_client(client);
+ kfree(codec->reg_cache);
+ kfree(client);
+ return 0;
+}
+
+static int wm8900_i2c_attach(struct i2c_adapter *adap)
+{
+ return i2c_probe(adap, &addr_data, wm8900_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8900_i2c_driver = {
+ .driver = {
+ .name = "WM8900 I2C codec",
+ .owner = THIS_MODULE,
+ },
+ .attach_adapter = wm8900_i2c_attach,
+ .detach_client = wm8900_i2c_detach,
+ .command = NULL,
+};
+
+static struct i2c_client client_template = {
+ .name = "WM8900",
+ .driver = &wm8900_i2c_driver,
+};
+#endif
+
+static int wm8900_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8900_setup_data *setup;
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ dev_info(&pdev->dev, "WM8900 Audio Codec\n");
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ socdev->codec = codec;
+
+ codec->set_bias_level = wm8900_set_bias_level;
+
+ wm8900_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&wm8900_i2c_driver);
+ if (ret != 0)
+ printk(KERN_ERR "can't add i2c driver");
+ }
+#else
+#error Non-I2C interfaces not yet supported
+#endif
+ return ret;
+}
+
+/* power down chip */
+static int wm8900_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8900_i2c_driver);
+#endif
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8900 = {
+ .probe = wm8900_probe,
+ .remove = wm8900_remove,
+ .suspend = wm8900_suspend,
+ .resume = wm8900_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
+
+MODULE_DESCRIPTION("ASoC WM8900 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h
new file mode 100644
index 0000000..ba450d9
--- /dev/null
+++ b/sound/soc/codecs/wm8900.h
@@ -0,0 +1,64 @@
+/*
+ * wm8900.h -- WM890 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8900_H
+#define _WM8900_H
+
+#define WM8900_FLL 1
+
+#define WM8900_BCLK_DIV 1
+#define WM8900_ADC_CLKDIV 2
+#define WM8900_DAC_CLKDIV 3
+#define WM8900_ADC_LRCLK 4
+#define WM8900_DAC_LRCLK 5
+#define WM8900_OPCLK_DIV 6
+#define WM8900_LRCLK_MODE 7
+
+#define WM8900_BCLK_DIV_1 0x00
+#define WM8900_BCLK_DIV_1_5 0x02
+#define WM8900_BCLK_DIV_2 0x04
+#define WM8900_BCLK_DIV_3 0x06
+#define WM8900_BCLK_DIV_4 0x08
+#define WM8900_BCLK_DIV_5_5 0x0a
+#define WM8900_BCLK_DIV_6 0x0c
+#define WM8900_BCLK_DIV_8 0x0e
+#define WM8900_BCLK_DIV_11 0x10
+#define WM8900_BCLK_DIV_12 0x12
+#define WM8900_BCLK_DIV_16 0x14
+#define WM8900_BCLK_DIV_22 0x16
+#define WM8900_BCLK_DIV_24 0x18
+#define WM8900_BCLK_DIV_32 0x1a
+#define WM8900_BCLK_DIV_44 0x1c
+#define WM8900_BCLK_DIV_48 0x1e
+
+#define WM8900_ADC_CLKDIV_1 0x00
+#define WM8900_ADC_CLKDIV_1_5 0x20
+#define WM8900_ADC_CLKDIV_2 0x40
+#define WM8900_ADC_CLKDIV_3 0x60
+#define WM8900_ADC_CLKDIV_4 0x80
+#define WM8900_ADC_CLKDIV_5_5 0xa0
+#define WM8900_ADC_CLKDIV_6 0xc0
+
+#define WM8900_DAC_CLKDIV_1 0x00
+#define WM8900_DAC_CLKDIV_1_5 0x04
+#define WM8900_DAC_CLKDIV_2 0x08
+#define WM8900_DAC_CLKDIV_3 0x0c
+#define WM8900_DAC_CLKDIV_4 0x10
+#define WM8900_DAC_CLKDIV_5_5 0x14
+#define WM8900_DAC_CLKDIV_6 0x18
+
+#define WM8900_
+
+struct wm8900_setup_data {
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8900_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8900;
+
+#endif
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
new file mode 100644
index 0000000..ce40d78
--- /dev/null
+++ b/sound/soc/codecs/wm8903.c
@@ -0,0 +1,1813 @@
+/*
+ * wm8903.c -- WM8903 ALSA SoC Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * TODO:
+ * - TDM mode configuration.
+ * - Mic detect.
+ * - Digital microphone support.
+ * - Interrupt support (mic detect and sequencer).
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8903.h"
+
+struct wm8903_priv {
+ int sysclk;
+
+ /* Reference counts */
+ int charge_pump_users;
+ int class_w_users;
+ int playback_active;
+ int capture_active;
+
+ struct snd_pcm_substream *master_substream;
+ struct snd_pcm_substream *slave_substream;
+};
+
+/* Register defaults at reset */
+static u16 wm8903_reg_defaults[] = {
+ 0x8903, /* R0 - SW Reset and ID */
+ 0x0000, /* R1 - Revision Number */
+ 0x0000, /* R2 */
+ 0x0000, /* R3 */
+ 0x0018, /* R4 - Bias Control 0 */
+ 0x0000, /* R5 - VMID Control 0 */
+ 0x0000, /* R6 - Mic Bias Control 0 */
+ 0x0000, /* R7 */
+ 0x0001, /* R8 - Analogue DAC 0 */
+ 0x0000, /* R9 */
+ 0x0001, /* R10 - Analogue ADC 0 */
+ 0x0000, /* R11 */
+ 0x0000, /* R12 - Power Management 0 */
+ 0x0000, /* R13 - Power Management 1 */
+ 0x0000, /* R14 - Power Management 2 */
+ 0x0000, /* R15 - Power Management 3 */
+ 0x0000, /* R16 - Power Management 4 */
+ 0x0000, /* R17 - Power Management 5 */
+ 0x0000, /* R18 - Power Management 6 */
+ 0x0000, /* R19 */
+ 0x0400, /* R20 - Clock Rates 0 */
+ 0x0D07, /* R21 - Clock Rates 1 */
+ 0x0000, /* R22 - Clock Rates 2 */
+ 0x0000, /* R23 */
+ 0x0050, /* R24 - Audio Interface 0 */
+ 0x0242, /* R25 - Audio Interface 1 */
+ 0x0008, /* R26 - Audio Interface 2 */
+ 0x0022, /* R27 - Audio Interface 3 */
+ 0x0000, /* R28 */
+ 0x0000, /* R29 */
+ 0x00C0, /* R30 - DAC Digital Volume Left */
+ 0x00C0, /* R31 - DAC Digital Volume Right */
+ 0x0000, /* R32 - DAC Digital 0 */
+ 0x0000, /* R33 - DAC Digital 1 */
+ 0x0000, /* R34 */
+ 0x0000, /* R35 */
+ 0x00C0, /* R36 - ADC Digital Volume Left */
+ 0x00C0, /* R37 - ADC Digital Volume Right */
+ 0x0000, /* R38 - ADC Digital 0 */
+ 0x0073, /* R39 - Digital Microphone 0 */
+ 0x09BF, /* R40 - DRC 0 */
+ 0x3241, /* R41 - DRC 1 */
+ 0x0020, /* R42 - DRC 2 */
+ 0x0000, /* R43 - DRC 3 */
+ 0x0085, /* R44 - Analogue Left Input 0 */
+ 0x0085, /* R45 - Analogue Right Input 0 */
+ 0x0044, /* R46 - Analogue Left Input 1 */
+ 0x0044, /* R47 - Analogue Right Input 1 */
+ 0x0000, /* R48 */
+ 0x0000, /* R49 */
+ 0x0008, /* R50 - Analogue Left Mix 0 */
+ 0x0004, /* R51 - Analogue Right Mix 0 */
+ 0x0000, /* R52 - Analogue Spk Mix Left 0 */
+ 0x0000, /* R53 - Analogue Spk Mix Left 1 */
+ 0x0000, /* R54 - Analogue Spk Mix Right 0 */
+ 0x0000, /* R55 - Analogue Spk Mix Right 1 */
+ 0x0000, /* R56 */
+ 0x002D, /* R57 - Analogue OUT1 Left */
+ 0x002D, /* R58 - Analogue OUT1 Right */
+ 0x0039, /* R59 - Analogue OUT2 Left */
+ 0x0039, /* R60 - Analogue OUT2 Right */
+ 0x0100, /* R61 */
+ 0x0139, /* R62 - Analogue OUT3 Left */
+ 0x0139, /* R63 - Analogue OUT3 Right */
+ 0x0000, /* R64 */
+ 0x0000, /* R65 - Analogue SPK Output Control 0 */
+ 0x0000, /* R66 */
+ 0x0010, /* R67 - DC Servo 0 */
+ 0x0100, /* R68 */
+ 0x00A4, /* R69 - DC Servo 2 */
+ 0x0807, /* R70 */
+ 0x0000, /* R71 */
+ 0x0000, /* R72 */
+ 0x0000, /* R73 */
+ 0x0000, /* R74 */
+ 0x0000, /* R75 */
+ 0x0000, /* R76 */
+ 0x0000, /* R77 */
+ 0x0000, /* R78 */
+ 0x000E, /* R79 */
+ 0x0000, /* R80 */
+ 0x0000, /* R81 */
+ 0x0000, /* R82 */
+ 0x0000, /* R83 */
+ 0x0000, /* R84 */
+ 0x0000, /* R85 */
+ 0x0000, /* R86 */
+ 0x0006, /* R87 */
+ 0x0000, /* R88 */
+ 0x0000, /* R89 */
+ 0x0000, /* R90 - Analogue HP 0 */
+ 0x0060, /* R91 */
+ 0x0000, /* R92 */
+ 0x0000, /* R93 */
+ 0x0000, /* R94 - Analogue Lineout 0 */
+ 0x0060, /* R95 */
+ 0x0000, /* R96 */
+ 0x0000, /* R97 */
+ 0x0000, /* R98 - Charge Pump 0 */
+ 0x1F25, /* R99 */
+ 0x2B19, /* R100 */
+ 0x01C0, /* R101 */
+ 0x01EF, /* R102 */
+ 0x2B00, /* R103 */
+ 0x0000, /* R104 - Class W 0 */
+ 0x01C0, /* R105 */
+ 0x1C10, /* R106 */
+ 0x0000, /* R107 */
+ 0x0000, /* R108 - Write Sequencer 0 */
+ 0x0000, /* R109 - Write Sequencer 1 */
+ 0x0000, /* R110 - Write Sequencer 2 */
+ 0x0000, /* R111 - Write Sequencer 3 */
+ 0x0000, /* R112 - Write Sequencer 4 */
+ 0x0000, /* R113 */
+ 0x0000, /* R114 - Control Interface */
+ 0x0000, /* R115 */
+ 0x00A8, /* R116 - GPIO Control 1 */
+ 0x00A8, /* R117 - GPIO Control 2 */
+ 0x00A8, /* R118 - GPIO Control 3 */
+ 0x0220, /* R119 - GPIO Control 4 */
+ 0x01A0, /* R120 - GPIO Control 5 */
+ 0x0000, /* R121 - Interrupt Status 1 */
+ 0xFFFF, /* R122 - Interrupt Status 1 Mask */
+ 0x0000, /* R123 - Interrupt Polarity 1 */
+ 0x0000, /* R124 */
+ 0x0003, /* R125 */
+ 0x0000, /* R126 - Interrupt Control */
+ 0x0000, /* R127 */
+ 0x0005, /* R128 */
+ 0x0000, /* R129 - Control Interface Test 1 */
+ 0x0000, /* R130 */
+ 0x0000, /* R131 */
+ 0x0000, /* R132 */
+ 0x0000, /* R133 */
+ 0x0000, /* R134 */
+ 0x03FF, /* R135 */
+ 0x0007, /* R136 */
+ 0x0040, /* R137 */
+ 0x0000, /* R138 */
+ 0x0000, /* R139 */
+ 0x0000, /* R140 */
+ 0x0000, /* R141 */
+ 0x0000, /* R142 */
+ 0x0000, /* R143 */
+ 0x0000, /* R144 */
+ 0x0000, /* R145 */
+ 0x0000, /* R146 */
+ 0x0000, /* R147 */
+ 0x4000, /* R148 */
+ 0x6810, /* R149 - Charge Pump Test 1 */
+ 0x0004, /* R150 */
+ 0x0000, /* R151 */
+ 0x0000, /* R152 */
+ 0x0000, /* R153 */
+ 0x0000, /* R154 */
+ 0x0000, /* R155 */
+ 0x0000, /* R156 */
+ 0x0000, /* R157 */
+ 0x0000, /* R158 */
+ 0x0000, /* R159 */
+ 0x0000, /* R160 */
+ 0x0000, /* R161 */
+ 0x0000, /* R162 */
+ 0x0000, /* R163 */
+ 0x0028, /* R164 - Clock Rate Test 4 */
+ 0x0004, /* R165 */
+ 0x0000, /* R166 */
+ 0x0060, /* R167 */
+ 0x0000, /* R168 */
+ 0x0000, /* R169 */
+ 0x0000, /* R170 */
+ 0x0000, /* R171 */
+ 0x0000, /* R172 - Analogue Output Bias 0 */
+};
+
+static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ BUG_ON(reg >= ARRAY_SIZE(wm8903_reg_defaults));
+
+ return cache[reg];
+}
+
+static unsigned int wm8903_hw_read(struct snd_soc_codec *codec, u8 reg)
+{
+ struct i2c_msg xfer[2];
+ u16 data;
+ int ret;
+ struct i2c_client *client = codec->control_data;
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 1;
+ xfer[0].buf = &reg;
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 2;
+ xfer[1].buf = (u8 *)&data;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret != 2) {
+ pr_err("i2c_transfer returned %d\n", ret);
+ return 0;
+ }
+
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+
+static unsigned int wm8903_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ case WM8903_SW_RESET_AND_ID:
+ case WM8903_REVISION_NUMBER:
+ case WM8903_INTERRUPT_STATUS_1:
+ case WM8903_WRITE_SEQUENCER_4:
+ return wm8903_hw_read(codec, reg);
+
+ default:
+ return wm8903_read_reg_cache(codec, reg);
+ }
+}
+
+static void wm8903_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+
+ BUG_ON(reg >= ARRAY_SIZE(wm8903_reg_defaults));
+
+ switch (reg) {
+ case WM8903_SW_RESET_AND_ID:
+ case WM8903_REVISION_NUMBER:
+ break;
+
+ default:
+ cache[reg] = value;
+ break;
+ }
+}
+
+static int wm8903_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+
+ wm8903_write_reg_cache(codec, reg, value);
+
+ /* Data format is 1 byte of address followed by 2 bytes of data */
+ data[0] = reg;
+ data[1] = (value >> 8) & 0xff;
+ data[2] = value & 0xff;
+
+ if (codec->hw_write(codec->control_data, data, 3) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
+{
+ u16 reg[5];
+ struct i2c_client *i2c = codec->control_data;
+
+ BUG_ON(start > 48);
+
+ /* Enable the sequencer */
+ reg[0] = wm8903_read(codec, WM8903_WRITE_SEQUENCER_0);
+ reg[0] |= WM8903_WSEQ_ENA;
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
+
+ dev_dbg(&i2c->dev, "Starting sequence at %d\n", start);
+
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_3,
+ start | WM8903_WSEQ_START);
+
+ /* Wait for it to complete. If we have the interrupt wired up then
+ * we could block waiting for an interrupt, though polling may still
+ * be desirable for diagnostic purposes.
+ */
+ do {
+ msleep(10);
+
+ reg[4] = wm8903_read(codec, WM8903_WRITE_SEQUENCER_4);
+ } while (reg[4] & WM8903_WSEQ_BUSY);
+
+ dev_dbg(&i2c->dev, "Sequence complete\n");
+
+ /* Disable the sequencer again */
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_0,
+ reg[0] & ~WM8903_WSEQ_ENA);
+
+ return 0;
+}
+
+static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
+{
+ int i;
+
+ /* There really ought to be something better we can do here :/ */
+ for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
+ cache[i] = wm8903_hw_read(codec, i);
+}
+
+static void wm8903_reset(struct snd_soc_codec *codec)
+{
+ wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0);
+}
+
+#define WM8903_OUTPUT_SHORT 0x8
+#define WM8903_OUTPUT_OUT 0x4
+#define WM8903_OUTPUT_INT 0x2
+#define WM8903_OUTPUT_IN 0x1
+
+/*
+ * Event for headphone and line out amplifier power changes. Special
+ * power up/down sequences are required in order to maximise pop/click
+ * performance.
+ */
+static int wm8903_output_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8903_priv *wm8903 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
+ u16 val;
+ u16 reg;
+ int shift;
+ u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0);
+
+ switch (w->reg) {
+ case WM8903_POWER_MANAGEMENT_2:
+ reg = WM8903_ANALOGUE_HP_0;
+ break;
+ case WM8903_POWER_MANAGEMENT_3:
+ reg = WM8903_ANALOGUE_LINEOUT_0;
+ break;
+ default:
+ BUG();
+ }
+
+ switch (w->shift) {
+ case 0:
+ shift = 0;
+ break;
+ case 1:
+ shift = 4;
+ break;
+ default:
+ BUG();
+ }
+
+ if (event & SND_SOC_DAPM_PRE_PMU) {
+ val = wm8903_read(codec, reg);
+
+ /* Short the output */
+ val &= ~(WM8903_OUTPUT_SHORT << shift);
+ wm8903_write(codec, reg, val);
+
+ wm8903->charge_pump_users++;
+
+ dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
+ wm8903->charge_pump_users);
+
+ if (wm8903->charge_pump_users == 1) {
+ dev_dbg(&i2c->dev, "Enabling charge pump\n");
+ wm8903_write(codec, WM8903_CHARGE_PUMP_0,
+ cp_reg | WM8903_CP_ENA);
+ mdelay(4);
+ }
+ }
+
+ if (event & SND_SOC_DAPM_POST_PMU) {
+ val = wm8903_read(codec, reg);
+
+ val |= (WM8903_OUTPUT_IN << shift);
+ wm8903_write(codec, reg, val);
+
+ val |= (WM8903_OUTPUT_INT << shift);
+ wm8903_write(codec, reg, val);
+
+ /* Turn on the output ENA_OUTP */
+ val |= (WM8903_OUTPUT_OUT << shift);
+ wm8903_write(codec, reg, val);
+
+ /* Remove the short */
+ val |= (WM8903_OUTPUT_SHORT << shift);
+ wm8903_write(codec, reg, val);
+ }
+
+ if (event & SND_SOC_DAPM_PRE_PMD) {
+ val = wm8903_read(codec, reg);
+
+ /* Short the output */
+ val &= ~(WM8903_OUTPUT_SHORT << shift);
+ wm8903_write(codec, reg, val);
+
+ /* Then disable the intermediate and output stages */
+ val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
+ WM8903_OUTPUT_IN) << shift);
+ wm8903_write(codec, reg, val);
+ }
+
+ if (event & SND_SOC_DAPM_POST_PMD) {
+ wm8903->charge_pump_users--;
+
+ dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
+ wm8903->charge_pump_users);
+
+ if (wm8903->charge_pump_users == 0) {
+ dev_dbg(&i2c->dev, "Disabling charge pump\n");
+ wm8903_write(codec, WM8903_CHARGE_PUMP_0,
+ cp_reg & ~WM8903_CP_ENA);
+ }
+ }
+
+ return 0;
+}
+
+/*
+ * When used with DAC outputs only the WM8903 charge pump supports
+ * operation in class W mode, providing very low power consumption
+ * when used with digital sources. Enable and disable this mode
+ * automatically depending on the mixer configuration.
+ *
+ * All the relevant controls are simple switches.
+ */
+static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_codec *codec = widget->codec;
+ struct wm8903_priv *wm8903 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
+ u16 reg;
+ int ret;
+
+ reg = wm8903_read(codec, WM8903_CLASS_W_0);
+
+ /* Turn it off if we're about to enable bypass */
+ if (ucontrol->value.integer.value[0]) {
+ if (wm8903->class_w_users == 0) {
+ dev_dbg(&i2c->dev, "Disabling Class W\n");
+ wm8903_write(codec, WM8903_CLASS_W_0, reg &
+ ~(WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V));
+ }
+ wm8903->class_w_users++;
+ }
+
+ /* Implement the change */
+ ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
+
+ /* If we've just disabled the last bypass path turn Class W on */
+ if (!ucontrol->value.integer.value[0]) {
+ if (wm8903->class_w_users == 1) {
+ dev_dbg(&i2c->dev, "Enabling Class W\n");
+ wm8903_write(codec, WM8903_CLASS_W_0, reg |
+ WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V);
+ }
+ wm8903->class_w_users--;
+ }
+
+ dev_dbg(&i2c->dev, "Bypass use count now %d\n",
+ wm8903->class_w_users);
+
+ return ret;
+}
+
+#define SOC_DAPM_SINGLE_W(xname, reg, shift, max, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_dapm_get_volsw, .put = wm8903_class_w_put, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+/* ALSA can only do steps of .01dB */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
+
+static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
+
+static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_tlv_amp, -2250, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_tlv_min, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(drc_tlv_max, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(drc_tlv_startup, -300, 50, 0);
+
+static const char *drc_slope_text[] = {
+ "1", "1/2", "1/4", "1/8", "1/16", "0"
+};
+
+static const struct soc_enum drc_slope_r0 =
+ SOC_ENUM_SINGLE(WM8903_DRC_2, 3, 6, drc_slope_text);
+
+static const struct soc_enum drc_slope_r1 =
+ SOC_ENUM_SINGLE(WM8903_DRC_2, 0, 6, drc_slope_text);
+
+static const char *drc_attack_text[] = {
+ "instantaneous",
+ "363us", "762us", "1.45ms", "2.9ms", "5.8ms", "11.6ms", "23.2ms",
+ "46.4ms", "92.8ms", "185.6ms"
+};
+
+static const struct soc_enum drc_attack =
+ SOC_ENUM_SINGLE(WM8903_DRC_1, 12, 11, drc_attack_text);
+
+static const char *drc_decay_text[] = {
+ "186ms", "372ms", "743ms", "1.49s", "2.97s", "5.94s", "11.89s",
+ "23.87s", "47.56s"
+};
+
+static const struct soc_enum drc_decay =
+ SOC_ENUM_SINGLE(WM8903_DRC_1, 8, 9, drc_decay_text);
+
+static const char *drc_ff_delay_text[] = {
+ "5 samples", "9 samples"
+};
+
+static const struct soc_enum drc_ff_delay =
+ SOC_ENUM_SINGLE(WM8903_DRC_0, 5, 2, drc_ff_delay_text);
+
+static const char *drc_qr_decay_text[] = {
+ "0.725ms", "1.45ms", "5.8ms"
+};
+
+static const struct soc_enum drc_qr_decay =
+ SOC_ENUM_SINGLE(WM8903_DRC_1, 4, 3, drc_qr_decay_text);
+
+static const char *drc_smoothing_text[] = {
+ "Low", "Medium", "High"
+};
+
+static const struct soc_enum drc_smoothing =
+ SOC_ENUM_SINGLE(WM8903_DRC_0, 11, 3, drc_smoothing_text);
+
+static const char *soft_mute_text[] = {
+ "Fast (fs/2)", "Slow (fs/32)"
+};
+
+static const struct soc_enum soft_mute =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 10, 2, soft_mute_text);
+
+static const char *mute_mode_text[] = {
+ "Hard", "Soft"
+};
+
+static const struct soc_enum mute_mode =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 9, 2, mute_mode_text);
+
+static const char *dac_deemphasis_text[] = {
+ "Disabled", "32kHz", "44.1kHz", "48kHz"
+};
+
+static const struct soc_enum dac_deemphasis =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 1, 4, dac_deemphasis_text);
+
+static const char *companding_text[] = {
+ "ulaw", "alaw"
+};
+
+static const struct soc_enum dac_companding =
+ SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 0, 2, companding_text);
+
+static const struct soc_enum adc_companding =
+ SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 2, 2, companding_text);
+
+static const char *input_mode_text[] = {
+ "Single-Ended", "Differential Line", "Differential Mic"
+};
+
+static const struct soc_enum linput_mode_enum =
+ SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text);
+
+static const struct soc_enum rinput_mode_enum =
+ SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text);
+
+static const char *linput_mux_text[] = {
+ "IN1L", "IN2L", "IN3L"
+};
+
+static const struct soc_enum linput_enum =
+ SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 2, 3, linput_mux_text);
+
+static const struct soc_enum linput_inv_enum =
+ SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 4, 3, linput_mux_text);
+
+static const char *rinput_mux_text[] = {
+ "IN1R", "IN2R", "IN3R"
+};
+
+static const struct soc_enum rinput_enum =
+ SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 2, 3, rinput_mux_text);
+
+static const struct soc_enum rinput_inv_enum =
+ SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
+
+
+static const struct snd_kcontrol_new wm8903_snd_controls[] = {
+
+/* Input PGAs - No TLV since the scale depends on PGA mode */
+SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0,
+ 7, 1, 1),
+SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0,
+ 0, 31, 0),
+SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1,
+ 6, 1, 0),
+
+SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0,
+ 7, 1, 1),
+SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0,
+ 0, 31, 0),
+SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
+ 6, 1, 0),
+
+/* ADCs */
+SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0),
+SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0),
+SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1),
+SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1,
+ drc_tlv_thresh),
+SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp),
+SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min),
+SOC_SINGLE_TLV("DRC Maximum Gain Volume", WM8903_DRC_1, 0, 3, 0, drc_tlv_max),
+SOC_ENUM("DRC Attack Rate", drc_attack),
+SOC_ENUM("DRC Decay Rate", drc_decay),
+SOC_ENUM("DRC FF Delay", drc_ff_delay),
+SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0),
+SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0),
+SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
+SOC_ENUM("DRC QR Decay Rate", drc_qr_decay),
+SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0),
+SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0),
+SOC_ENUM("DRC Smoothing Threashold", drc_smoothing),
+SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
+
+SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
+ WM8903_ADC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv),
+SOC_ENUM("ADC Companding Mode", adc_companding),
+SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
+
+/* DAC */
+SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT,
+ WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
+SOC_ENUM("DAC Soft Mute Rate", soft_mute),
+SOC_ENUM("DAC Mute Mode", mute_mode),
+SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0),
+SOC_ENUM("DAC De-emphasis", dac_deemphasis),
+SOC_SINGLE("DAC Sloping Stopband Filter Switch",
+ WM8903_DAC_DIGITAL_1, 11, 1, 0),
+SOC_ENUM("DAC Companding Mode", dac_companding),
+SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0),
+
+/* Headphones */
+SOC_DOUBLE_R("Headphone Switch",
+ WM8903_ANALOGUE_OUT1_LEFT, WM8903_ANALOGUE_OUT1_RIGHT,
+ 8, 1, 1),
+SOC_DOUBLE_R("Headphone ZC Switch",
+ WM8903_ANALOGUE_OUT1_LEFT, WM8903_ANALOGUE_OUT1_RIGHT,
+ 6, 1, 0),
+SOC_DOUBLE_R_TLV("Headphone Volume",
+ WM8903_ANALOGUE_OUT1_LEFT, WM8903_ANALOGUE_OUT1_RIGHT,
+ 0, 63, 0, out_tlv),
+
+/* Line out */
+SOC_DOUBLE_R("Line Out Switch",
+ WM8903_ANALOGUE_OUT2_LEFT, WM8903_ANALOGUE_OUT2_RIGHT,
+ 8, 1, 1),
+SOC_DOUBLE_R("Line Out ZC Switch",
+ WM8903_ANALOGUE_OUT2_LEFT, WM8903_ANALOGUE_OUT2_RIGHT,
+ 6, 1, 0),
+SOC_DOUBLE_R_TLV("Line Out Volume",
+ WM8903_ANALOGUE_OUT2_LEFT, WM8903_ANALOGUE_OUT2_RIGHT,
+ 0, 63, 0, out_tlv),
+
+/* Speaker */
+SOC_DOUBLE_R("Speaker Switch",
+ WM8903_ANALOGUE_OUT3_LEFT, WM8903_ANALOGUE_OUT3_RIGHT, 8, 1, 1),
+SOC_DOUBLE_R("Speaker ZC Switch",
+ WM8903_ANALOGUE_OUT3_LEFT, WM8903_ANALOGUE_OUT3_RIGHT, 6, 1, 0),
+SOC_DOUBLE_R_TLV("Speaker Volume",
+ WM8903_ANALOGUE_OUT3_LEFT, WM8903_ANALOGUE_OUT3_RIGHT,
+ 0, 63, 0, out_tlv),
+};
+
+static int wm8903_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8903_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new linput_mode_mux =
+ SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum);
+
+static const struct snd_kcontrol_new rinput_mode_mux =
+ SOC_DAPM_ENUM("Right Input Mode Mux", rinput_mode_enum);
+
+static const struct snd_kcontrol_new linput_mux =
+ SOC_DAPM_ENUM("Left Input Mux", linput_enum);
+
+static const struct snd_kcontrol_new linput_inv_mux =
+ SOC_DAPM_ENUM("Left Inverting Input Mux", linput_inv_enum);
+
+static const struct snd_kcontrol_new rinput_mux =
+ SOC_DAPM_ENUM("Right Input Mux", rinput_enum);
+
+static const struct snd_kcontrol_new rinput_inv_mux =
+ SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum);
+
+static const struct snd_kcontrol_new left_output_mixer[] = {
+SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
+SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
+SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_output_mixer[] = {
+SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 3, 1, 0),
+SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 2, 1, 0),
+SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_speaker_mixer[] = {
+SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 3, 1, 0),
+SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 2, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 1, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0,
+ 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_speaker_mixer[] = {
+SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 3, 1, 0),
+SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 2, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
+ 1, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
+ 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("LINEOUTL"),
+SND_SOC_DAPM_OUTPUT("LINEOUTR"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8903_MIC_BIAS_CONTROL_0, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Input Mux", SND_SOC_NOPM, 0, 0, &linput_mux),
+SND_SOC_DAPM_MUX("Left Input Inverting Mux", SND_SOC_NOPM, 0, 0,
+ &linput_inv_mux),
+SND_SOC_DAPM_MUX("Left Input Mode Mux", SND_SOC_NOPM, 0, 0, &linput_mode_mux),
+
+SND_SOC_DAPM_MUX("Right Input Mux", SND_SOC_NOPM, 0, 0, &rinput_mux),
+SND_SOC_DAPM_MUX("Right Input Inverting Mux", SND_SOC_NOPM, 0, 0,
+ &rinput_inv_mux),
+SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux),
+
+SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0),
+
+SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
+SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
+
+SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
+SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
+
+SND_SOC_DAPM_MIXER("Left Output Mixer", WM8903_POWER_MANAGEMENT_1, 1, 0,
+ left_output_mixer, ARRAY_SIZE(left_output_mixer)),
+SND_SOC_DAPM_MIXER("Right Output Mixer", WM8903_POWER_MANAGEMENT_1, 0, 0,
+ right_output_mixer, ARRAY_SIZE(right_output_mixer)),
+
+SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
+ left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
+SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
+ right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
+
+SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
+ 1, 0, NULL, 0, wm8903_output_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
+ 0, 0, NULL, 0, wm8903_output_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
+ NULL, 0, wm8903_output_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
+ NULL, 0, wm8903_output_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0,
+ NULL, 0),
+
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+ { "Left Input Mux", "IN1L", "IN1L" },
+ { "Left Input Mux", "IN2L", "IN2L" },
+ { "Left Input Mux", "IN3L", "IN3L" },
+
+ { "Left Input Inverting Mux", "IN1L", "IN1L" },
+ { "Left Input Inverting Mux", "IN2L", "IN2L" },
+ { "Left Input Inverting Mux", "IN3L", "IN3L" },
+
+ { "Right Input Mux", "IN1R", "IN1R" },
+ { "Right Input Mux", "IN2R", "IN2R" },
+ { "Right Input Mux", "IN3R", "IN3R" },
+
+ { "Right Input Inverting Mux", "IN1R", "IN1R" },
+ { "Right Input Inverting Mux", "IN2R", "IN2R" },
+ { "Right Input Inverting Mux", "IN3R", "IN3R" },
+
+ { "Left Input Mode Mux", "Single-Ended", "Left Input Inverting Mux" },
+ { "Left Input Mode Mux", "Differential Line",
+ "Left Input Mux" },
+ { "Left Input Mode Mux", "Differential Line",
+ "Left Input Inverting Mux" },
+ { "Left Input Mode Mux", "Differential Mic",
+ "Left Input Mux" },
+ { "Left Input Mode Mux", "Differential Mic",
+ "Left Input Inverting Mux" },
+
+ { "Right Input Mode Mux", "Single-Ended",
+ "Right Input Inverting Mux" },
+ { "Right Input Mode Mux", "Differential Line",
+ "Right Input Mux" },
+ { "Right Input Mode Mux", "Differential Line",
+ "Right Input Inverting Mux" },
+ { "Right Input Mode Mux", "Differential Mic",
+ "Right Input Mux" },
+ { "Right Input Mode Mux", "Differential Mic",
+ "Right Input Inverting Mux" },
+
+ { "Left Input PGA", NULL, "Left Input Mode Mux" },
+ { "Right Input PGA", NULL, "Right Input Mode Mux" },
+
+ { "ADCL", NULL, "Left Input PGA" },
+ { "ADCR", NULL, "Right Input PGA" },
+
+ { "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" },
+ { "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" },
+ { "Left Output Mixer", "DACL Switch", "DACL" },
+ { "Left Output Mixer", "DACR Switch", "DACR" },
+
+ { "Right Output Mixer", "Left Bypass Switch", "Left Input PGA" },
+ { "Right Output Mixer", "Right Bypass Switch", "Right Input PGA" },
+ { "Right Output Mixer", "DACL Switch", "DACL" },
+ { "Right Output Mixer", "DACR Switch", "DACR" },
+
+ { "Left Speaker Mixer", "Left Bypass Switch", "Left Input PGA" },
+ { "Left Speaker Mixer", "Right Bypass Switch", "Right Input PGA" },
+ { "Left Speaker Mixer", "DACL Switch", "DACL" },
+ { "Left Speaker Mixer", "DACR Switch", "DACR" },
+
+ { "Right Speaker Mixer", "Left Bypass Switch", "Left Input PGA" },
+ { "Right Speaker Mixer", "Right Bypass Switch", "Right Input PGA" },
+ { "Right Speaker Mixer", "DACL Switch", "DACL" },
+ { "Right Speaker Mixer", "DACR Switch", "DACR" },
+
+ { "Left Line Output PGA", NULL, "Left Output Mixer" },
+ { "Right Line Output PGA", NULL, "Right Output Mixer" },
+
+ { "Left Headphone Output PGA", NULL, "Left Output Mixer" },
+ { "Right Headphone Output PGA", NULL, "Right Output Mixer" },
+
+ { "Left Speaker PGA", NULL, "Left Speaker Mixer" },
+ { "Right Speaker PGA", NULL, "Right Speaker Mixer" },
+
+ { "HPOUTL", NULL, "Left Headphone Output PGA" },
+ { "HPOUTR", NULL, "Right Headphone Output PGA" },
+
+ { "LINEOUTL", NULL, "Left Line Output PGA" },
+ { "LINEOUTR", NULL, "Right Line Output PGA" },
+
+ { "LOP", NULL, "Left Speaker PGA" },
+ { "LON", NULL, "Left Speaker PGA" },
+
+ { "ROP", NULL, "Right Speaker PGA" },
+ { "RON", NULL, "Right Speaker PGA" },
+};
+
+static int wm8903_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8903_dapm_widgets,
+ ARRAY_SIZE(wm8903_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(codec);
+
+ return 0;
+}
+
+static int wm8903_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct i2c_client *i2c = codec->control_data;
+ u16 reg, reg2;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ reg = wm8903_read(codec, WM8903_VMID_CONTROL_0);
+ reg &= ~(WM8903_VMID_RES_MASK);
+ reg |= WM8903_VMID_RES_50K;
+ wm8903_write(codec, WM8903_VMID_CONTROL_0, reg);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ wm8903_run_sequence(codec, 0);
+ wm8903_sync_reg_cache(codec, codec->reg_cache);
+
+ /* Enable low impedence charge pump output */
+ reg = wm8903_read(codec,
+ WM8903_CONTROL_INTERFACE_TEST_1);
+ wm8903_write(codec, WM8903_CONTROL_INTERFACE_TEST_1,
+ reg | WM8903_TEST_KEY);
+ reg2 = wm8903_read(codec, WM8903_CHARGE_PUMP_TEST_1);
+ wm8903_write(codec, WM8903_CHARGE_PUMP_TEST_1,
+ reg2 | WM8903_CP_SW_KELVIN_MODE_MASK);
+ wm8903_write(codec, WM8903_CONTROL_INTERFACE_TEST_1,
+ reg);
+
+ /* By default no bypass paths are enabled so
+ * enable Class W support.
+ */
+ dev_dbg(&i2c->dev, "Enabling Class W\n");
+ wm8903_write(codec, WM8903_CLASS_W_0, reg |
+ WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V);
+ }
+
+ reg = wm8903_read(codec, WM8903_VMID_CONTROL_0);
+ reg &= ~(WM8903_VMID_RES_MASK);
+ reg |= WM8903_VMID_RES_250K;
+ wm8903_write(codec, WM8903_VMID_CONTROL_0, reg);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ wm8903_run_sequence(codec, 32);
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+static int wm8903_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8903_priv *wm8903 = codec->private_data;
+
+ wm8903->sysclk = freq;
+
+ return 0;
+}
+
+static int wm8903_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 aif1 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_1);
+
+ aif1 &= ~(WM8903_LRCLK_DIR | WM8903_BCLK_DIR | WM8903_AIF_FMT_MASK |
+ WM8903_AIF_LRCLK_INV | WM8903_AIF_BCLK_INV);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ aif1 |= WM8903_LRCLK_DIR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif1 |= WM8903_LRCLK_DIR | WM8903_BCLK_DIR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ aif1 |= WM8903_BCLK_DIR;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ aif1 |= 0x3;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ aif1 |= 0x3 | WM8903_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ aif1 |= 0x2;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ aif1 |= 0x1;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* Clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /* frame inversion not valid for DSP modes */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif1 |= WM8903_AIF_BCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ aif1 |= WM8903_AIF_BCLK_INV | WM8903_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif1 |= WM8903_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ aif1 |= WM8903_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
+
+ return 0;
+}
+
+static int wm8903_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ reg = wm8903_read(codec, WM8903_DAC_DIGITAL_1);
+
+ if (mute)
+ reg |= WM8903_DAC_MUTE;
+ else
+ reg &= ~WM8903_DAC_MUTE;
+
+ wm8903_write(codec, WM8903_DAC_DIGITAL_1, reg);
+
+ return 0;
+}
+
+/* Lookup table for CLK_SYS/fs ratio. 256fs or more is recommended
+ * for optimal performance so we list the lower rates first and match
+ * on the last match we find. */
+static struct {
+ int div;
+ int rate;
+ int mode;
+ int mclk_div;
+} clk_sys_ratios[] = {
+ { 64, 0x0, 0x0, 1 },
+ { 68, 0x0, 0x1, 1 },
+ { 125, 0x0, 0x2, 1 },
+ { 128, 0x1, 0x0, 1 },
+ { 136, 0x1, 0x1, 1 },
+ { 192, 0x2, 0x0, 1 },
+ { 204, 0x2, 0x1, 1 },
+
+ { 64, 0x0, 0x0, 2 },
+ { 68, 0x0, 0x1, 2 },
+ { 125, 0x0, 0x2, 2 },
+ { 128, 0x1, 0x0, 2 },
+ { 136, 0x1, 0x1, 2 },
+ { 192, 0x2, 0x0, 2 },
+ { 204, 0x2, 0x1, 2 },
+
+ { 250, 0x2, 0x2, 1 },
+ { 256, 0x3, 0x0, 1 },
+ { 272, 0x3, 0x1, 1 },
+ { 384, 0x4, 0x0, 1 },
+ { 408, 0x4, 0x1, 1 },
+ { 375, 0x4, 0x2, 1 },
+ { 512, 0x5, 0x0, 1 },
+ { 544, 0x5, 0x1, 1 },
+ { 500, 0x5, 0x2, 1 },
+ { 768, 0x6, 0x0, 1 },
+ { 816, 0x6, 0x1, 1 },
+ { 750, 0x6, 0x2, 1 },
+ { 1024, 0x7, 0x0, 1 },
+ { 1088, 0x7, 0x1, 1 },
+ { 1000, 0x7, 0x2, 1 },
+ { 1408, 0x8, 0x0, 1 },
+ { 1496, 0x8, 0x1, 1 },
+ { 1536, 0x9, 0x0, 1 },
+ { 1632, 0x9, 0x1, 1 },
+ { 1500, 0x9, 0x2, 1 },
+
+ { 250, 0x2, 0x2, 2 },
+ { 256, 0x3, 0x0, 2 },
+ { 272, 0x3, 0x1, 2 },
+ { 384, 0x4, 0x0, 2 },
+ { 408, 0x4, 0x1, 2 },
+ { 375, 0x4, 0x2, 2 },
+ { 512, 0x5, 0x0, 2 },
+ { 544, 0x5, 0x1, 2 },
+ { 500, 0x5, 0x2, 2 },
+ { 768, 0x6, 0x0, 2 },
+ { 816, 0x6, 0x1, 2 },
+ { 750, 0x6, 0x2, 2 },
+ { 1024, 0x7, 0x0, 2 },
+ { 1088, 0x7, 0x1, 2 },
+ { 1000, 0x7, 0x2, 2 },
+ { 1408, 0x8, 0x0, 2 },
+ { 1496, 0x8, 0x1, 2 },
+ { 1536, 0x9, 0x0, 2 },
+ { 1632, 0x9, 0x1, 2 },
+ { 1500, 0x9, 0x2, 2 },
+};
+
+/* CLK_SYS/BCLK ratios - multiplied by 10 due to .5s */
+static struct {
+ int ratio;
+ int div;
+} bclk_divs[] = {
+ { 10, 0 },
+ { 15, 1 },
+ { 20, 2 },
+ { 30, 3 },
+ { 40, 4 },
+ { 50, 5 },
+ { 55, 6 },
+ { 60, 7 },
+ { 80, 8 },
+ { 100, 9 },
+ { 110, 10 },
+ { 120, 11 },
+ { 160, 12 },
+ { 200, 13 },
+ { 220, 14 },
+ { 240, 15 },
+ { 250, 16 },
+ { 300, 17 },
+ { 320, 18 },
+ { 440, 19 },
+ { 480, 20 },
+};
+
+/* Sample rates for DSP */
+static struct {
+ int rate;
+ int value;
+} sample_rates[] = {
+ { 8000, 0 },
+ { 11025, 1 },
+ { 12000, 2 },
+ { 16000, 3 },
+ { 22050, 4 },
+ { 24000, 5 },
+ { 32000, 6 },
+ { 44100, 7 },
+ { 48000, 8 },
+ { 88200, 9 },
+ { 96000, 10 },
+ { 0, 0 },
+};
+
+static int wm8903_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8903_priv *wm8903 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
+ struct snd_pcm_runtime *master_runtime;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ wm8903->playback_active++;
+ else
+ wm8903->capture_active++;
+
+ /* The DAI has shared clocks so if we already have a playback or
+ * capture going then constrain this substream to match it.
+ */
+ if (wm8903->master_substream) {
+ master_runtime = wm8903->master_substream->runtime;
+
+ dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
+ master_runtime->sample_bits,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
+
+ wm8903->slave_substream = substream;
+ } else
+ wm8903->master_substream = substream;
+
+ return 0;
+}
+
+static void wm8903_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8903_priv *wm8903 = codec->private_data;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ wm8903->playback_active--;
+ else
+ wm8903->capture_active--;
+
+ if (wm8903->master_substream == substream)
+ wm8903->master_substream = wm8903->slave_substream;
+
+ wm8903->slave_substream = NULL;
+}
+
+static int wm8903_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8903_priv *wm8903 = codec->private_data;
+ struct i2c_client *i2c = codec->control_data;
+ int fs = params_rate(params);
+ int bclk;
+ int bclk_div;
+ int i;
+ int dsp_config;
+ int clk_config;
+ int best_val;
+ int cur_val;
+ int clk_sys;
+
+ u16 aif1 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_1);
+ u16 aif2 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_2);
+ u16 aif3 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_3);
+ u16 clock0 = wm8903_read(codec, WM8903_CLOCK_RATES_0);
+ u16 clock1 = wm8903_read(codec, WM8903_CLOCK_RATES_1);
+
+ if (substream == wm8903->slave_substream) {
+ dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
+ return 0;
+ }
+
+ /* Configure sample rate logic for DSP - choose nearest rate */
+ dsp_config = 0;
+ best_val = abs(sample_rates[dsp_config].rate - fs);
+ for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+ cur_val = abs(sample_rates[i].rate - fs);
+ if (cur_val <= best_val) {
+ dsp_config = i;
+ best_val = cur_val;
+ }
+ }
+
+ /* Constraints should stop us hitting this but let's make sure */
+ if (wm8903->capture_active)
+ switch (sample_rates[dsp_config].rate) {
+ case 88200:
+ case 96000:
+ dev_err(&i2c->dev, "%dHz unsupported by ADC\n",
+ fs);
+ return -EINVAL;
+
+ default:
+ break;
+ }
+
+ dev_dbg(&i2c->dev, "DSP fs = %dHz\n", sample_rates[dsp_config].rate);
+ clock1 &= ~WM8903_SAMPLE_RATE_MASK;
+ clock1 |= sample_rates[dsp_config].value;
+
+ aif1 &= ~WM8903_AIF_WL_MASK;
+ bclk = 2 * fs;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bclk *= 16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ bclk *= 20;
+ aif1 |= 0x4;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bclk *= 24;
+ aif1 |= 0x8;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bclk *= 32;
+ aif1 |= 0xc;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ dev_dbg(&i2c->dev, "MCLK = %dHz, target sample rate = %dHz\n",
+ wm8903->sysclk, fs);
+
+ /* We may not have an MCLK which allows us to generate exactly
+ * the clock we want, particularly with USB derived inputs, so
+ * approximate.
+ */
+ clk_config = 0;
+ best_val = abs((wm8903->sysclk /
+ (clk_sys_ratios[0].mclk_div *
+ clk_sys_ratios[0].div)) - fs);
+ for (i = 1; i < ARRAY_SIZE(clk_sys_ratios); i++) {
+ cur_val = abs((wm8903->sysclk /
+ (clk_sys_ratios[i].mclk_div *
+ clk_sys_ratios[i].div)) - fs);
+
+ if (cur_val <= best_val) {
+ clk_config = i;
+ best_val = cur_val;
+ }
+ }
+
+ if (clk_sys_ratios[clk_config].mclk_div == 2) {
+ clock0 |= WM8903_MCLKDIV2;
+ clk_sys = wm8903->sysclk / 2;
+ } else {
+ clock0 &= ~WM8903_MCLKDIV2;
+ clk_sys = wm8903->sysclk;
+ }
+
+ clock1 &= ~(WM8903_CLK_SYS_RATE_MASK |
+ WM8903_CLK_SYS_MODE_MASK);
+ clock1 |= clk_sys_ratios[clk_config].rate << WM8903_CLK_SYS_RATE_SHIFT;
+ clock1 |= clk_sys_ratios[clk_config].mode << WM8903_CLK_SYS_MODE_SHIFT;
+
+ dev_dbg(&i2c->dev, "CLK_SYS_RATE=%x, CLK_SYS_MODE=%x div=%d\n",
+ clk_sys_ratios[clk_config].rate,
+ clk_sys_ratios[clk_config].mode,
+ clk_sys_ratios[clk_config].div);
+
+ dev_dbg(&i2c->dev, "Actual CLK_SYS = %dHz\n", clk_sys);
+
+ /* We may not get quite the right frequency if using
+ * approximate clocks so look for the closest match that is
+ * higher than the target (we need to ensure that there enough
+ * BCLKs to clock out the samples).
+ */
+ bclk_div = 0;
+ best_val = ((clk_sys * 10) / bclk_divs[0].ratio) - bclk;
+ i = 1;
+ while (i < ARRAY_SIZE(bclk_divs)) {
+ cur_val = ((clk_sys * 10) / bclk_divs[i].ratio) - bclk;
+ if (cur_val < 0) /* BCLK table is sorted */
+ break;
+ bclk_div = i;
+ best_val = cur_val;
+ i++;
+ }
+
+ aif2 &= ~WM8903_BCLK_DIV_MASK;
+ aif3 &= ~WM8903_LRCLK_RATE_MASK;
+
+ dev_dbg(&i2c->dev, "BCLK ratio %d for %dHz - actual BCLK = %dHz\n",
+ bclk_divs[bclk_div].ratio / 10, bclk,
+ (clk_sys * 10) / bclk_divs[bclk_div].ratio);
+
+ aif2 |= bclk_divs[bclk_div].div;
+ aif3 |= bclk / fs;
+
+ wm8903_write(codec, WM8903_CLOCK_RATES_0, clock0);
+ wm8903_write(codec, WM8903_CLOCK_RATES_1, clock1);
+ wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
+ wm8903_write(codec, WM8903_AUDIO_INTERFACE_2, aif2);
+ wm8903_write(codec, WM8903_AUDIO_INTERFACE_3, aif3);
+
+ return 0;
+}
+
+#define WM8903_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000)
+
+#define WM8903_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define WM8903_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8903_dai = {
+ .name = "WM8903",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8903_PLAYBACK_RATES,
+ .formats = WM8903_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = WM8903_CAPTURE_RATES,
+ .formats = WM8903_FORMATS,
+ },
+ .ops = {
+ .startup = wm8903_startup,
+ .shutdown = wm8903_shutdown,
+ .hw_params = wm8903_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = wm8903_digital_mute,
+ .set_fmt = wm8903_set_dai_fmt,
+ .set_sysclk = wm8903_set_dai_sysclk
+ }
+};
+EXPORT_SYMBOL_GPL(wm8903_dai);
+
+static int wm8903_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm8903_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c = codec->control_data;
+ int i;
+ u16 *reg_cache = codec->reg_cache;
+ u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+ GFP_KERNEL);
+
+ /* Bring the codec back up to standby first to minimise pop/clicks */
+ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8903_set_bias_level(codec, codec->suspend_bias_level);
+
+ /* Sync back everything else */
+ if (tmp_cache) {
+ for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
+ if (tmp_cache[i] != reg_cache[i])
+ wm8903_write(codec, i, tmp_cache[i]);
+ } else {
+ dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
+ }
+
+ return 0;
+}
+
+/*
+ * initialise the WM8903 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8903_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct i2c_client *i2c = codec->control_data;
+ int ret = 0;
+ u16 val;
+
+ val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
+ if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
+ dev_err(&i2c->dev,
+ "Device with ID register %x is not a WM8903\n", val);
+ return -ENODEV;
+ }
+
+ codec->name = "WM8903";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8903_read;
+ codec->write = wm8903_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8903_set_bias_level;
+ codec->dai = &wm8903_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults);
+ codec->reg_cache = kmemdup(wm8903_reg_defaults,
+ sizeof(wm8903_reg_defaults),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ dev_err(&i2c->dev, "Failed to allocate register cache\n");
+ return -ENOMEM;
+ }
+
+ val = wm8903_read(codec, WM8903_REVISION_NUMBER);
+ dev_info(&i2c->dev, "WM8903 revision %d\n",
+ val & WM8903_CHIP_REV_MASK);
+
+ wm8903_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* SYSCLK is required for pretty much anything */
+ wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA);
+
+ /* power on device */
+ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Latch volume update bits */
+ val = wm8903_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT);
+ val |= WM8903_ADCVU;
+ wm8903_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val);
+ wm8903_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val);
+
+ val = wm8903_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT);
+ val |= WM8903_DACVU;
+ wm8903_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val);
+ wm8903_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val);
+
+ val = wm8903_read(codec, WM8903_ANALOGUE_OUT1_LEFT);
+ val |= WM8903_HPOUTVU;
+ wm8903_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val);
+ wm8903_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val);
+
+ val = wm8903_read(codec, WM8903_ANALOGUE_OUT2_LEFT);
+ val |= WM8903_LINEOUTVU;
+ wm8903_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val);
+ wm8903_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val);
+
+ val = wm8903_read(codec, WM8903_ANALOGUE_OUT3_LEFT);
+ val |= WM8903_SPKVU;
+ wm8903_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val);
+ wm8903_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
+
+ /* Enable DAC soft mute by default */
+ val = wm8903_read(codec, WM8903_DAC_DIGITAL_1);
+ val |= WM8903_DAC_MUTEMODE;
+ wm8903_write(codec, WM8903_DAC_DIGITAL_1, val);
+
+ wm8903_add_controls(codec);
+ wm8903_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "wm8903: failed to register card\n");
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+static struct snd_soc_device *wm8903_socdev;
+
+static int wm8903_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8903_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8903_init(socdev);
+ if (ret < 0)
+ dev_err(&i2c->dev, "Device initialisation failed\n");
+
+ return ret;
+}
+
+static int wm8903_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+/* i2c codec control layer */
+static const struct i2c_device_id wm8903_i2c_id[] = {
+ { "wm8903", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id);
+
+static struct i2c_driver wm8903_i2c_driver = {
+ .driver = {
+ .name = "WM8903",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8903_i2c_probe,
+ .remove = wm8903_i2c_remove,
+ .id_table = wm8903_i2c_id,
+};
+
+static int wm8903_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8903_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8903_priv *wm8903;
+ struct i2c_board_info board_info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *i2c_client;
+ int ret = 0;
+
+ setup = socdev->codec_data;
+
+ if (!setup->i2c_address) {
+ dev_err(&pdev->dev, "No codec address provided\n");
+ return -ENODEV;
+ }
+
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
+ if (wm8903 == NULL) {
+ ret = -ENOMEM;
+ goto err_codec;
+ }
+
+ codec->private_data = wm8903;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ wm8903_socdev = socdev;
+
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = i2c_add_driver(&wm8903_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ goto err_priv;
+ } else {
+ memset(&board_info, 0, sizeof(board_info));
+ strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE);
+ board_info.addr = setup->i2c_address;
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "Can't get I2C bus %d\n",
+ setup->i2c_bus);
+ ret = -ENODEV;
+ goto err_adapter;
+ }
+
+ i2c_client = i2c_new_device(adapter, &board_info);
+ i2c_put_adapter(adapter);
+ if (i2c_client == NULL) {
+ dev_err(&pdev->dev,
+ "I2C driver registration failed\n");
+ ret = -ENODEV;
+ goto err_adapter;
+ }
+ }
+
+ return ret;
+
+err_adapter:
+ i2c_del_driver(&wm8903_i2c_driver);
+err_priv:
+ kfree(codec->private_data);
+err_codec:
+ kfree(codec);
+ return ret;
+}
+
+/* power down chip */
+static int wm8903_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+ i2c_unregister_device(socdev->codec->control_data);
+ i2c_del_driver(&wm8903_i2c_driver);
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8903 = {
+ .probe = wm8903_probe,
+ .remove = wm8903_remove,
+ .suspend = wm8903_suspend,
+ .resume = wm8903_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903);
+
+MODULE_DESCRIPTION("ASoC WM8903 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
new file mode 100644
index 0000000..cec622f
--- /dev/null
+++ b/sound/soc/codecs/wm8903.h
@@ -0,0 +1,1463 @@
+/*
+ * wm8903.h - WM8903 audio codec interface
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef _WM8903_H
+#define _WM8903_H
+
+#include <linux/i2c.h>
+
+extern struct snd_soc_dai wm8903_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8903;
+
+struct wm8903_setup_data {
+ int i2c_bus;
+ int i2c_address;
+};
+
+#define WM8903_MCLK_DIV_2 1
+#define WM8903_CLK_SYS 2
+#define WM8903_BCLK 3
+#define WM8903_LRCLK 4
+
+/*
+ * Register values.
+ */
+#define WM8903_SW_RESET_AND_ID 0x00
+#define WM8903_REVISION_NUMBER 0x01
+#define WM8903_BIAS_CONTROL_0 0x04
+#define WM8903_VMID_CONTROL_0 0x05
+#define WM8903_MIC_BIAS_CONTROL_0 0x06
+#define WM8903_ANALOGUE_DAC_0 0x08
+#define WM8903_ANALOGUE_ADC_0 0x0A
+#define WM8903_POWER_MANAGEMENT_0 0x0C
+#define WM8903_POWER_MANAGEMENT_1 0x0D
+#define WM8903_POWER_MANAGEMENT_2 0x0E
+#define WM8903_POWER_MANAGEMENT_3 0x0F
+#define WM8903_POWER_MANAGEMENT_4 0x10
+#define WM8903_POWER_MANAGEMENT_5 0x11
+#define WM8903_POWER_MANAGEMENT_6 0x12
+#define WM8903_CLOCK_RATES_0 0x14
+#define WM8903_CLOCK_RATES_1 0x15
+#define WM8903_CLOCK_RATES_2 0x16
+#define WM8903_AUDIO_INTERFACE_0 0x18
+#define WM8903_AUDIO_INTERFACE_1 0x19
+#define WM8903_AUDIO_INTERFACE_2 0x1A
+#define WM8903_AUDIO_INTERFACE_3 0x1B
+#define WM8903_DAC_DIGITAL_VOLUME_LEFT 0x1E
+#define WM8903_DAC_DIGITAL_VOLUME_RIGHT 0x1F
+#define WM8903_DAC_DIGITAL_0 0x20
+#define WM8903_DAC_DIGITAL_1 0x21
+#define WM8903_ADC_DIGITAL_VOLUME_LEFT 0x24
+#define WM8903_ADC_DIGITAL_VOLUME_RIGHT 0x25
+#define WM8903_ADC_DIGITAL_0 0x26
+#define WM8903_DIGITAL_MICROPHONE_0 0x27
+#define WM8903_DRC_0 0x28
+#define WM8903_DRC_1 0x29
+#define WM8903_DRC_2 0x2A
+#define WM8903_DRC_3 0x2B
+#define WM8903_ANALOGUE_LEFT_INPUT_0 0x2C
+#define WM8903_ANALOGUE_RIGHT_INPUT_0 0x2D
+#define WM8903_ANALOGUE_LEFT_INPUT_1 0x2E
+#define WM8903_ANALOGUE_RIGHT_INPUT_1 0x2F
+#define WM8903_ANALOGUE_LEFT_MIX_0 0x32
+#define WM8903_ANALOGUE_RIGHT_MIX_0 0x33
+#define WM8903_ANALOGUE_SPK_MIX_LEFT_0 0x34
+#define WM8903_ANALOGUE_SPK_MIX_LEFT_1 0x35
+#define WM8903_ANALOGUE_SPK_MIX_RIGHT_0 0x36
+#define WM8903_ANALOGUE_SPK_MIX_RIGHT_1 0x37
+#define WM8903_ANALOGUE_OUT1_LEFT 0x39
+#define WM8903_ANALOGUE_OUT1_RIGHT 0x3A
+#define WM8903_ANALOGUE_OUT2_LEFT 0x3B
+#define WM8903_ANALOGUE_OUT2_RIGHT 0x3C
+#define WM8903_ANALOGUE_OUT3_LEFT 0x3E
+#define WM8903_ANALOGUE_OUT3_RIGHT 0x3F
+#define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0 0x41
+#define WM8903_DC_SERVO_0 0x43
+#define WM8903_DC_SERVO_2 0x45
+#define WM8903_ANALOGUE_HP_0 0x5A
+#define WM8903_ANALOGUE_LINEOUT_0 0x5E
+#define WM8903_CHARGE_PUMP_0 0x62
+#define WM8903_CLASS_W_0 0x68
+#define WM8903_WRITE_SEQUENCER_0 0x6C
+#define WM8903_WRITE_SEQUENCER_1 0x6D
+#define WM8903_WRITE_SEQUENCER_2 0x6E
+#define WM8903_WRITE_SEQUENCER_3 0x6F
+#define WM8903_WRITE_SEQUENCER_4 0x70
+#define WM8903_CONTROL_INTERFACE 0x72
+#define WM8903_GPIO_CONTROL_1 0x74
+#define WM8903_GPIO_CONTROL_2 0x75
+#define WM8903_GPIO_CONTROL_3 0x76
+#define WM8903_GPIO_CONTROL_4 0x77
+#define WM8903_GPIO_CONTROL_5 0x78
+#define WM8903_INTERRUPT_STATUS_1 0x79
+#define WM8903_INTERRUPT_STATUS_1_MASK 0x7A
+#define WM8903_INTERRUPT_POLARITY_1 0x7B
+#define WM8903_INTERRUPT_CONTROL 0x7E
+#define WM8903_CONTROL_INTERFACE_TEST_1 0x81
+#define WM8903_CHARGE_PUMP_TEST_1 0x95
+#define WM8903_CLOCK_RATE_TEST_4 0xA4
+#define WM8903_ANALOGUE_OUTPUT_BIAS_0 0xAC
+
+#define WM8903_REGISTER_COUNT 75
+#define WM8903_MAX_REGISTER 0xAC
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - SW Reset and ID
+ */
+#define WM8903_SW_RESET_DEV_ID1_MASK 0xFFFF /* SW_RESET_DEV_ID1 - [15:0] */
+#define WM8903_SW_RESET_DEV_ID1_SHIFT 0 /* SW_RESET_DEV_ID1 - [15:0] */
+#define WM8903_SW_RESET_DEV_ID1_WIDTH 16 /* SW_RESET_DEV_ID1 - [15:0] */
+
+/*
+ * R1 (0x01) - Revision Number
+ */
+#define WM8903_CHIP_REV_MASK 0x000F /* CHIP_REV - [3:0] */
+#define WM8903_CHIP_REV_SHIFT 0 /* CHIP_REV - [3:0] */
+#define WM8903_CHIP_REV_WIDTH 4 /* CHIP_REV - [3:0] */
+
+/*
+ * R4 (0x04) - Bias Control 0
+ */
+#define WM8903_POBCTRL 0x0010 /* POBCTRL */
+#define WM8903_POBCTRL_MASK 0x0010 /* POBCTRL */
+#define WM8903_POBCTRL_SHIFT 4 /* POBCTRL */
+#define WM8903_POBCTRL_WIDTH 1 /* POBCTRL */
+#define WM8903_ISEL_MASK 0x000C /* ISEL - [3:2] */
+#define WM8903_ISEL_SHIFT 2 /* ISEL - [3:2] */
+#define WM8903_ISEL_WIDTH 2 /* ISEL - [3:2] */
+#define WM8903_STARTUP_BIAS_ENA 0x0002 /* STARTUP_BIAS_ENA */
+#define WM8903_STARTUP_BIAS_ENA_MASK 0x0002 /* STARTUP_BIAS_ENA */
+#define WM8903_STARTUP_BIAS_ENA_SHIFT 1 /* STARTUP_BIAS_ENA */
+#define WM8903_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */
+#define WM8903_BIAS_ENA 0x0001 /* BIAS_ENA */
+#define WM8903_BIAS_ENA_MASK 0x0001 /* BIAS_ENA */
+#define WM8903_BIAS_ENA_SHIFT 0 /* BIAS_ENA */
+#define WM8903_BIAS_ENA_WIDTH 1 /* BIAS_ENA */
+
+/*
+ * R5 (0x05) - VMID Control 0
+ */
+#define WM8903_VMID_TIE_ENA 0x0080 /* VMID_TIE_ENA */
+#define WM8903_VMID_TIE_ENA_MASK 0x0080 /* VMID_TIE_ENA */
+#define WM8903_VMID_TIE_ENA_SHIFT 7 /* VMID_TIE_ENA */
+#define WM8903_VMID_TIE_ENA_WIDTH 1 /* VMID_TIE_ENA */
+#define WM8903_BUFIO_ENA 0x0040 /* BUFIO_ENA */
+#define WM8903_BUFIO_ENA_MASK 0x0040 /* BUFIO_ENA */
+#define WM8903_BUFIO_ENA_SHIFT 6 /* BUFIO_ENA */
+#define WM8903_BUFIO_ENA_WIDTH 1 /* BUFIO_ENA */
+#define WM8903_VMID_IO_ENA 0x0020 /* VMID_IO_ENA */
+#define WM8903_VMID_IO_ENA_MASK 0x0020 /* VMID_IO_ENA */
+#define WM8903_VMID_IO_ENA_SHIFT 5 /* VMID_IO_ENA */
+#define WM8903_VMID_IO_ENA_WIDTH 1 /* VMID_IO_ENA */
+#define WM8903_VMID_SOFT_MASK 0x0018 /* VMID_SOFT - [4:3] */
+#define WM8903_VMID_SOFT_SHIFT 3 /* VMID_SOFT - [4:3] */
+#define WM8903_VMID_SOFT_WIDTH 2 /* VMID_SOFT - [4:3] */
+#define WM8903_VMID_RES_MASK 0x0006 /* VMID_RES - [2:1] */
+#define WM8903_VMID_RES_SHIFT 1 /* VMID_RES - [2:1] */
+#define WM8903_VMID_RES_WIDTH 2 /* VMID_RES - [2:1] */
+#define WM8903_VMID_BUF_ENA 0x0001 /* VMID_BUF_ENA */
+#define WM8903_VMID_BUF_ENA_MASK 0x0001 /* VMID_BUF_ENA */
+#define WM8903_VMID_BUF_ENA_SHIFT 0 /* VMID_BUF_ENA */
+#define WM8903_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */
+
+#define WM8903_VMID_RES_50K 2
+#define WM8903_VMID_RES_250K 3
+#define WM8903_VMID_RES_5K 4
+
+/*
+ * R6 (0x06) - Mic Bias Control 0
+ */
+#define WM8903_MICDET_HYST_ENA 0x0080 /* MICDET_HYST_ENA */
+#define WM8903_MICDET_HYST_ENA_MASK 0x0080 /* MICDET_HYST_ENA */
+#define WM8903_MICDET_HYST_ENA_SHIFT 7 /* MICDET_HYST_ENA */
+#define WM8903_MICDET_HYST_ENA_WIDTH 1 /* MICDET_HYST_ENA */
+#define WM8903_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */
+#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */
+#define WM8903_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */
+#define WM8903_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */
+#define WM8903_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */
+#define WM8903_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */
+#define WM8903_MICDET_ENA 0x0002 /* MICDET_ENA */
+#define WM8903_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */
+#define WM8903_MICDET_ENA_SHIFT 1 /* MICDET_ENA */
+#define WM8903_MICDET_ENA_WIDTH 1 /* MICDET_ENA */
+#define WM8903_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */
+#define WM8903_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */
+
+/*
+ * R8 (0x08) - Analogue DAC 0
+ */
+#define WM8903_DACBIAS_SEL_MASK 0x0018 /* DACBIAS_SEL - [4:3] */
+#define WM8903_DACBIAS_SEL_SHIFT 3 /* DACBIAS_SEL - [4:3] */
+#define WM8903_DACBIAS_SEL_WIDTH 2 /* DACBIAS_SEL - [4:3] */
+#define WM8903_DACVMID_BIAS_SEL_MASK 0x0006 /* DACVMID_BIAS_SEL - [2:1] */
+#define WM8903_DACVMID_BIAS_SEL_SHIFT 1 /* DACVMID_BIAS_SEL - [2:1] */
+#define WM8903_DACVMID_BIAS_SEL_WIDTH 2 /* DACVMID_BIAS_SEL - [2:1] */
+
+/*
+ * R10 (0x0A) - Analogue ADC 0
+ */
+#define WM8903_ADC_OSR128 0x0001 /* ADC_OSR128 */
+#define WM8903_ADC_OSR128_MASK 0x0001 /* ADC_OSR128 */
+#define WM8903_ADC_OSR128_SHIFT 0 /* ADC_OSR128 */
+#define WM8903_ADC_OSR128_WIDTH 1 /* ADC_OSR128 */
+
+/*
+ * R12 (0x0C) - Power Management 0
+ */
+#define WM8903_INL_ENA 0x0002 /* INL_ENA */
+#define WM8903_INL_ENA_MASK 0x0002 /* INL_ENA */
+#define WM8903_INL_ENA_SHIFT 1 /* INL_ENA */
+#define WM8903_INL_ENA_WIDTH 1 /* INL_ENA */
+#define WM8903_INR_ENA 0x0001 /* INR_ENA */
+#define WM8903_INR_ENA_MASK 0x0001 /* INR_ENA */
+#define WM8903_INR_ENA_SHIFT 0 /* INR_ENA */
+#define WM8903_INR_ENA_WIDTH 1 /* INR_ENA */
+
+/*
+ * R13 (0x0D) - Power Management 1
+ */
+#define WM8903_MIXOUTL_ENA 0x0002 /* MIXOUTL_ENA */
+#define WM8903_MIXOUTL_ENA_MASK 0x0002 /* MIXOUTL_ENA */
+#define WM8903_MIXOUTL_ENA_SHIFT 1 /* MIXOUTL_ENA */
+#define WM8903_MIXOUTL_ENA_WIDTH 1 /* MIXOUTL_ENA */
+#define WM8903_MIXOUTR_ENA 0x0001 /* MIXOUTR_ENA */
+#define WM8903_MIXOUTR_ENA_MASK 0x0001 /* MIXOUTR_ENA */
+#define WM8903_MIXOUTR_ENA_SHIFT 0 /* MIXOUTR_ENA */
+#define WM8903_MIXOUTR_ENA_WIDTH 1 /* MIXOUTR_ENA */
+
+/*
+ * R14 (0x0E) - Power Management 2
+ */
+#define WM8903_HPL_PGA_ENA 0x0002 /* HPL_PGA_ENA */
+#define WM8903_HPL_PGA_ENA_MASK 0x0002 /* HPL_PGA_ENA */
+#define WM8903_HPL_PGA_ENA_SHIFT 1 /* HPL_PGA_ENA */
+#define WM8903_HPL_PGA_ENA_WIDTH 1 /* HPL_PGA_ENA */
+#define WM8903_HPR_PGA_ENA 0x0001 /* HPR_PGA_ENA */
+#define WM8903_HPR_PGA_ENA_MASK 0x0001 /* HPR_PGA_ENA */
+#define WM8903_HPR_PGA_ENA_SHIFT 0 /* HPR_PGA_ENA */
+#define WM8903_HPR_PGA_ENA_WIDTH 1 /* HPR_PGA_ENA */
+
+/*
+ * R15 (0x0F) - Power Management 3
+ */
+#define WM8903_LINEOUTL_PGA_ENA 0x0002 /* LINEOUTL_PGA_ENA */
+#define WM8903_LINEOUTL_PGA_ENA_MASK 0x0002 /* LINEOUTL_PGA_ENA */
+#define WM8903_LINEOUTL_PGA_ENA_SHIFT 1 /* LINEOUTL_PGA_ENA */
+#define WM8903_LINEOUTL_PGA_ENA_WIDTH 1 /* LINEOUTL_PGA_ENA */
+#define WM8903_LINEOUTR_PGA_ENA 0x0001 /* LINEOUTR_PGA_ENA */
+#define WM8903_LINEOUTR_PGA_ENA_MASK 0x0001 /* LINEOUTR_PGA_ENA */
+#define WM8903_LINEOUTR_PGA_ENA_SHIFT 0 /* LINEOUTR_PGA_ENA */
+#define WM8903_LINEOUTR_PGA_ENA_WIDTH 1 /* LINEOUTR_PGA_ENA */
+
+/*
+ * R16 (0x10) - Power Management 4
+ */
+#define WM8903_MIXSPKL_ENA 0x0002 /* MIXSPKL_ENA */
+#define WM8903_MIXSPKL_ENA_MASK 0x0002 /* MIXSPKL_ENA */
+#define WM8903_MIXSPKL_ENA_SHIFT 1 /* MIXSPKL_ENA */
+#define WM8903_MIXSPKL_ENA_WIDTH 1 /* MIXSPKL_ENA */
+#define WM8903_MIXSPKR_ENA 0x0001 /* MIXSPKR_ENA */
+#define WM8903_MIXSPKR_ENA_MASK 0x0001 /* MIXSPKR_ENA */
+#define WM8903_MIXSPKR_ENA_SHIFT 0 /* MIXSPKR_ENA */
+#define WM8903_MIXSPKR_ENA_WIDTH 1 /* MIXSPKR_ENA */
+
+/*
+ * R17 (0x11) - Power Management 5
+ */
+#define WM8903_SPKL_ENA 0x0002 /* SPKL_ENA */
+#define WM8903_SPKL_ENA_MASK 0x0002 /* SPKL_ENA */
+#define WM8903_SPKL_ENA_SHIFT 1 /* SPKL_ENA */
+#define WM8903_SPKL_ENA_WIDTH 1 /* SPKL_ENA */
+#define WM8903_SPKR_ENA 0x0001 /* SPKR_ENA */
+#define WM8903_SPKR_ENA_MASK 0x0001 /* SPKR_ENA */
+#define WM8903_SPKR_ENA_SHIFT 0 /* SPKR_ENA */
+#define WM8903_SPKR_ENA_WIDTH 1 /* SPKR_ENA */
+
+/*
+ * R18 (0x12) - Power Management 6
+ */
+#define WM8903_DACL_ENA 0x0008 /* DACL_ENA */
+#define WM8903_DACL_ENA_MASK 0x0008 /* DACL_ENA */
+#define WM8903_DACL_ENA_SHIFT 3 /* DACL_ENA */
+#define WM8903_DACL_ENA_WIDTH 1 /* DACL_ENA */
+#define WM8903_DACR_ENA 0x0004 /* DACR_ENA */
+#define WM8903_DACR_ENA_MASK 0x0004 /* DACR_ENA */
+#define WM8903_DACR_ENA_SHIFT 2 /* DACR_ENA */
+#define WM8903_DACR_ENA_WIDTH 1 /* DACR_ENA */
+#define WM8903_ADCL_ENA 0x0002 /* ADCL_ENA */
+#define WM8903_ADCL_ENA_MASK 0x0002 /* ADCL_ENA */
+#define WM8903_ADCL_ENA_SHIFT 1 /* ADCL_ENA */
+#define WM8903_ADCL_ENA_WIDTH 1 /* ADCL_ENA */
+#define WM8903_ADCR_ENA 0x0001 /* ADCR_ENA */
+#define WM8903_ADCR_ENA_MASK 0x0001 /* ADCR_ENA */
+#define WM8903_ADCR_ENA_SHIFT 0 /* ADCR_ENA */
+#define WM8903_ADCR_ENA_WIDTH 1 /* ADCR_ENA */
+
+/*
+ * R20 (0x14) - Clock Rates 0
+ */
+#define WM8903_MCLKDIV2 0x0001 /* MCLKDIV2 */
+#define WM8903_MCLKDIV2_MASK 0x0001 /* MCLKDIV2 */
+#define WM8903_MCLKDIV2_SHIFT 0 /* MCLKDIV2 */
+#define WM8903_MCLKDIV2_WIDTH 1 /* MCLKDIV2 */
+
+/*
+ * R21 (0x15) - Clock Rates 1
+ */
+#define WM8903_CLK_SYS_RATE_MASK 0x3C00 /* CLK_SYS_RATE - [13:10] */
+#define WM8903_CLK_SYS_RATE_SHIFT 10 /* CLK_SYS_RATE - [13:10] */
+#define WM8903_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [13:10] */
+#define WM8903_CLK_SYS_MODE_MASK 0x0300 /* CLK_SYS_MODE - [9:8] */
+#define WM8903_CLK_SYS_MODE_SHIFT 8 /* CLK_SYS_MODE - [9:8] */
+#define WM8903_CLK_SYS_MODE_WIDTH 2 /* CLK_SYS_MODE - [9:8] */
+#define WM8903_SAMPLE_RATE_MASK 0x000F /* SAMPLE_RATE - [3:0] */
+#define WM8903_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [3:0] */
+#define WM8903_SAMPLE_RATE_WIDTH 4 /* SAMPLE_RATE - [3:0] */
+
+/*
+ * R22 (0x16) - Clock Rates 2
+ */
+#define WM8903_CLK_SYS_ENA 0x0004 /* CLK_SYS_ENA */
+#define WM8903_CLK_SYS_ENA_MASK 0x0004 /* CLK_SYS_ENA */
+#define WM8903_CLK_SYS_ENA_SHIFT 2 /* CLK_SYS_ENA */
+#define WM8903_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */
+#define WM8903_CLK_DSP_ENA 0x0002 /* CLK_DSP_ENA */
+#define WM8903_CLK_DSP_ENA_MASK 0x0002 /* CLK_DSP_ENA */
+#define WM8903_CLK_DSP_ENA_SHIFT 1 /* CLK_DSP_ENA */
+#define WM8903_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */
+#define WM8903_TO_ENA 0x0001 /* TO_ENA */
+#define WM8903_TO_ENA_MASK 0x0001 /* TO_ENA */
+#define WM8903_TO_ENA_SHIFT 0 /* TO_ENA */
+#define WM8903_TO_ENA_WIDTH 1 /* TO_ENA */
+
+/*
+ * R24 (0x18) - Audio Interface 0
+ */
+#define WM8903_DACL_DATINV 0x1000 /* DACL_DATINV */
+#define WM8903_DACL_DATINV_MASK 0x1000 /* DACL_DATINV */
+#define WM8903_DACL_DATINV_SHIFT 12 /* DACL_DATINV */
+#define WM8903_DACL_DATINV_WIDTH 1 /* DACL_DATINV */
+#define WM8903_DACR_DATINV 0x0800 /* DACR_DATINV */
+#define WM8903_DACR_DATINV_MASK 0x0800 /* DACR_DATINV */
+#define WM8903_DACR_DATINV_SHIFT 11 /* DACR_DATINV */
+#define WM8903_DACR_DATINV_WIDTH 1 /* DACR_DATINV */
+#define WM8903_DAC_BOOST_MASK 0x0600 /* DAC_BOOST - [10:9] */
+#define WM8903_DAC_BOOST_SHIFT 9 /* DAC_BOOST - [10:9] */
+#define WM8903_DAC_BOOST_WIDTH 2 /* DAC_BOOST - [10:9] */
+#define WM8903_LOOPBACK 0x0100 /* LOOPBACK */
+#define WM8903_LOOPBACK_MASK 0x0100 /* LOOPBACK */
+#define WM8903_LOOPBACK_SHIFT 8 /* LOOPBACK */
+#define WM8903_LOOPBACK_WIDTH 1 /* LOOPBACK */
+#define WM8903_AIFADCL_SRC 0x0080 /* AIFADCL_SRC */
+#define WM8903_AIFADCL_SRC_MASK 0x0080 /* AIFADCL_SRC */
+#define WM8903_AIFADCL_SRC_SHIFT 7 /* AIFADCL_SRC */
+#define WM8903_AIFADCL_SRC_WIDTH 1 /* AIFADCL_SRC */
+#define WM8903_AIFADCR_SRC 0x0040 /* AIFADCR_SRC */
+#define WM8903_AIFADCR_SRC_MASK 0x0040 /* AIFADCR_SRC */
+#define WM8903_AIFADCR_SRC_SHIFT 6 /* AIFADCR_SRC */
+#define WM8903_AIFADCR_SRC_WIDTH 1 /* AIFADCR_SRC */
+#define WM8903_AIFDACL_SRC 0x0020 /* AIFDACL_SRC */
+#define WM8903_AIFDACL_SRC_MASK 0x0020 /* AIFDACL_SRC */
+#define WM8903_AIFDACL_SRC_SHIFT 5 /* AIFDACL_SRC */
+#define WM8903_AIFDACL_SRC_WIDTH 1 /* AIFDACL_SRC */
+#define WM8903_AIFDACR_SRC 0x0010 /* AIFDACR_SRC */
+#define WM8903_AIFDACR_SRC_MASK 0x0010 /* AIFDACR_SRC */
+#define WM8903_AIFDACR_SRC_SHIFT 4 /* AIFDACR_SRC */
+#define WM8903_AIFDACR_SRC_WIDTH 1 /* AIFDACR_SRC */
+#define WM8903_ADC_COMP 0x0008 /* ADC_COMP */
+#define WM8903_ADC_COMP_MASK 0x0008 /* ADC_COMP */
+#define WM8903_ADC_COMP_SHIFT 3 /* ADC_COMP */
+#define WM8903_ADC_COMP_WIDTH 1 /* ADC_COMP */
+#define WM8903_ADC_COMPMODE 0x0004 /* ADC_COMPMODE */
+#define WM8903_ADC_COMPMODE_MASK 0x0004 /* ADC_COMPMODE */
+#define WM8903_ADC_COMPMODE_SHIFT 2 /* ADC_COMPMODE */
+#define WM8903_ADC_COMPMODE_WIDTH 1 /* ADC_COMPMODE */
+#define WM8903_DAC_COMP 0x0002 /* DAC_COMP */
+#define WM8903_DAC_COMP_MASK 0x0002 /* DAC_COMP */
+#define WM8903_DAC_COMP_SHIFT 1 /* DAC_COMP */
+#define WM8903_DAC_COMP_WIDTH 1 /* DAC_COMP */
+#define WM8903_DAC_COMPMODE 0x0001 /* DAC_COMPMODE */
+#define WM8903_DAC_COMPMODE_MASK 0x0001 /* DAC_COMPMODE */
+#define WM8903_DAC_COMPMODE_SHIFT 0 /* DAC_COMPMODE */
+#define WM8903_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */
+
+/*
+ * R25 (0x19) - Audio Interface 1
+ */
+#define WM8903_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */
+#define WM8903_AIFDAC_TDM_MASK 0x2000 /* AIFDAC_TDM */
+#define WM8903_AIFDAC_TDM_SHIFT 13 /* AIFDAC_TDM */
+#define WM8903_AIFDAC_TDM_WIDTH 1 /* AIFDAC_TDM */
+#define WM8903_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */
+#define WM8903_AIFDAC_TDM_CHAN_MASK 0x1000 /* AIFDAC_TDM_CHAN */
+#define WM8903_AIFDAC_TDM_CHAN_SHIFT 12 /* AIFDAC_TDM_CHAN */
+#define WM8903_AIFDAC_TDM_CHAN_WIDTH 1 /* AIFDAC_TDM_CHAN */
+#define WM8903_AIFADC_TDM 0x0800 /* AIFADC_TDM */
+#define WM8903_AIFADC_TDM_MASK 0x0800 /* AIFADC_TDM */
+#define WM8903_AIFADC_TDM_SHIFT 11 /* AIFADC_TDM */
+#define WM8903_AIFADC_TDM_WIDTH 1 /* AIFADC_TDM */
+#define WM8903_AIFADC_TDM_CHAN 0x0400 /* AIFADC_TDM_CHAN */
+#define WM8903_AIFADC_TDM_CHAN_MASK 0x0400 /* AIFADC_TDM_CHAN */
+#define WM8903_AIFADC_TDM_CHAN_SHIFT 10 /* AIFADC_TDM_CHAN */
+#define WM8903_AIFADC_TDM_CHAN_WIDTH 1 /* AIFADC_TDM_CHAN */
+#define WM8903_LRCLK_DIR 0x0200 /* LRCLK_DIR */
+#define WM8903_LRCLK_DIR_MASK 0x0200 /* LRCLK_DIR */
+#define WM8903_LRCLK_DIR_SHIFT 9 /* LRCLK_DIR */
+#define WM8903_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */
+#define WM8903_AIF_BCLK_INV 0x0080 /* AIF_BCLK_INV */
+#define WM8903_AIF_BCLK_INV_MASK 0x0080 /* AIF_BCLK_INV */
+#define WM8903_AIF_BCLK_INV_SHIFT 7 /* AIF_BCLK_INV */
+#define WM8903_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */
+#define WM8903_BCLK_DIR 0x0040 /* BCLK_DIR */
+#define WM8903_BCLK_DIR_MASK 0x0040 /* BCLK_DIR */
+#define WM8903_BCLK_DIR_SHIFT 6 /* BCLK_DIR */
+#define WM8903_BCLK_DIR_WIDTH 1 /* BCLK_DIR */
+#define WM8903_AIF_LRCLK_INV 0x0010 /* AIF_LRCLK_INV */
+#define WM8903_AIF_LRCLK_INV_MASK 0x0010 /* AIF_LRCLK_INV */
+#define WM8903_AIF_LRCLK_INV_SHIFT 4 /* AIF_LRCLK_INV */
+#define WM8903_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */
+#define WM8903_AIF_WL_MASK 0x000C /* AIF_WL - [3:2] */
+#define WM8903_AIF_WL_SHIFT 2 /* AIF_WL - [3:2] */
+#define WM8903_AIF_WL_WIDTH 2 /* AIF_WL - [3:2] */
+#define WM8903_AIF_FMT_MASK 0x0003 /* AIF_FMT - [1:0] */
+#define WM8903_AIF_FMT_SHIFT 0 /* AIF_FMT - [1:0] */
+#define WM8903_AIF_FMT_WIDTH 2 /* AIF_FMT - [1:0] */
+
+/*
+ * R26 (0x1A) - Audio Interface 2
+ */
+#define WM8903_BCLK_DIV_MASK 0x001F /* BCLK_DIV - [4:0] */
+#define WM8903_BCLK_DIV_SHIFT 0 /* BCLK_DIV - [4:0] */
+#define WM8903_BCLK_DIV_WIDTH 5 /* BCLK_DIV - [4:0] */
+
+/*
+ * R27 (0x1B) - Audio Interface 3
+ */
+#define WM8903_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */
+#define WM8903_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */
+#define WM8903_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */
+
+/*
+ * R30 (0x1E) - DAC Digital Volume Left
+ */
+#define WM8903_DACVU 0x0100 /* DACVU */
+#define WM8903_DACVU_MASK 0x0100 /* DACVU */
+#define WM8903_DACVU_SHIFT 8 /* DACVU */
+#define WM8903_DACVU_WIDTH 1 /* DACVU */
+#define WM8903_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */
+#define WM8903_DACL_VOL_SHIFT 0 /* DACL_VOL - [7:0] */
+#define WM8903_DACL_VOL_WIDTH 8 /* DACL_VOL - [7:0] */
+
+/*
+ * R31 (0x1F) - DAC Digital Volume Right
+ */
+#define WM8903_DACVU 0x0100 /* DACVU */
+#define WM8903_DACVU_MASK 0x0100 /* DACVU */
+#define WM8903_DACVU_SHIFT 8 /* DACVU */
+#define WM8903_DACVU_WIDTH 1 /* DACVU */
+#define WM8903_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */
+#define WM8903_DACR_VOL_SHIFT 0 /* DACR_VOL - [7:0] */
+#define WM8903_DACR_VOL_WIDTH 8 /* DACR_VOL - [7:0] */
+
+/*
+ * R32 (0x20) - DAC Digital 0
+ */
+#define WM8903_ADCL_DAC_SVOL_MASK 0x0F00 /* ADCL_DAC_SVOL - [11:8] */
+#define WM8903_ADCL_DAC_SVOL_SHIFT 8 /* ADCL_DAC_SVOL - [11:8] */
+#define WM8903_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [11:8] */
+#define WM8903_ADCR_DAC_SVOL_MASK 0x00F0 /* ADCR_DAC_SVOL - [7:4] */
+#define WM8903_ADCR_DAC_SVOL_SHIFT 4 /* ADCR_DAC_SVOL - [7:4] */
+#define WM8903_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [7:4] */
+#define WM8903_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */
+#define WM8903_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */
+#define WM8903_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */
+#define WM8903_ADC_TO_DACR_MASK 0x0003 /* ADC_TO_DACR - [1:0] */
+#define WM8903_ADC_TO_DACR_SHIFT 0 /* ADC_TO_DACR - [1:0] */
+#define WM8903_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [1:0] */
+
+/*
+ * R33 (0x21) - DAC Digital 1
+ */
+#define WM8903_DAC_MONO 0x1000 /* DAC_MONO */
+#define WM8903_DAC_MONO_MASK 0x1000 /* DAC_MONO */
+#define WM8903_DAC_MONO_SHIFT 12 /* DAC_MONO */
+#define WM8903_DAC_MONO_WIDTH 1 /* DAC_MONO */
+#define WM8903_DAC_SB_FILT 0x0800 /* DAC_SB_FILT */
+#define WM8903_DAC_SB_FILT_MASK 0x0800 /* DAC_SB_FILT */
+#define WM8903_DAC_SB_FILT_SHIFT 11 /* DAC_SB_FILT */
+#define WM8903_DAC_SB_FILT_WIDTH 1 /* DAC_SB_FILT */
+#define WM8903_DAC_MUTERATE 0x0400 /* DAC_MUTERATE */
+#define WM8903_DAC_MUTERATE_MASK 0x0400 /* DAC_MUTERATE */
+#define WM8903_DAC_MUTERATE_SHIFT 10 /* DAC_MUTERATE */
+#define WM8903_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */
+#define WM8903_DAC_MUTEMODE 0x0200 /* DAC_MUTEMODE */
+#define WM8903_DAC_MUTEMODE_MASK 0x0200 /* DAC_MUTEMODE */
+#define WM8903_DAC_MUTEMODE_SHIFT 9 /* DAC_MUTEMODE */
+#define WM8903_DAC_MUTEMODE_WIDTH 1 /* DAC_MUTEMODE */
+#define WM8903_DAC_MUTE 0x0008 /* DAC_MUTE */
+#define WM8903_DAC_MUTE_MASK 0x0008 /* DAC_MUTE */
+#define WM8903_DAC_MUTE_SHIFT 3 /* DAC_MUTE */
+#define WM8903_DAC_MUTE_WIDTH 1 /* DAC_MUTE */
+#define WM8903_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */
+#define WM8903_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */
+#define WM8903_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */
+
+/*
+ * R36 (0x24) - ADC Digital Volume Left
+ */
+#define WM8903_ADCVU 0x0100 /* ADCVU */
+#define WM8903_ADCVU_MASK 0x0100 /* ADCVU */
+#define WM8903_ADCVU_SHIFT 8 /* ADCVU */
+#define WM8903_ADCVU_WIDTH 1 /* ADCVU */
+#define WM8903_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */
+#define WM8903_ADCL_VOL_SHIFT 0 /* ADCL_VOL - [7:0] */
+#define WM8903_ADCL_VOL_WIDTH 8 /* ADCL_VOL - [7:0] */
+
+/*
+ * R37 (0x25) - ADC Digital Volume Right
+ */
+#define WM8903_ADCVU 0x0100 /* ADCVU */
+#define WM8903_ADCVU_MASK 0x0100 /* ADCVU */
+#define WM8903_ADCVU_SHIFT 8 /* ADCVU */
+#define WM8903_ADCVU_WIDTH 1 /* ADCVU */
+#define WM8903_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */
+#define WM8903_ADCR_VOL_SHIFT 0 /* ADCR_VOL - [7:0] */
+#define WM8903_ADCR_VOL_WIDTH 8 /* ADCR_VOL - [7:0] */
+
+/*
+ * R38 (0x26) - ADC Digital 0
+ */
+#define WM8903_ADC_HPF_CUT_MASK 0x0060 /* ADC_HPF_CUT - [6:5] */
+#define WM8903_ADC_HPF_CUT_SHIFT 5 /* ADC_HPF_CUT - [6:5] */
+#define WM8903_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [6:5] */
+#define WM8903_ADC_HPF_ENA 0x0010 /* ADC_HPF_ENA */
+#define WM8903_ADC_HPF_ENA_MASK 0x0010 /* ADC_HPF_ENA */
+#define WM8903_ADC_HPF_ENA_SHIFT 4 /* ADC_HPF_ENA */
+#define WM8903_ADC_HPF_ENA_WIDTH 1 /* ADC_HPF_ENA */
+#define WM8903_ADCL_DATINV 0x0002 /* ADCL_DATINV */
+#define WM8903_ADCL_DATINV_MASK 0x0002 /* ADCL_DATINV */
+#define WM8903_ADCL_DATINV_SHIFT 1 /* ADCL_DATINV */
+#define WM8903_ADCL_DATINV_WIDTH 1 /* ADCL_DATINV */
+#define WM8903_ADCR_DATINV 0x0001 /* ADCR_DATINV */
+#define WM8903_ADCR_DATINV_MASK 0x0001 /* ADCR_DATINV */
+#define WM8903_ADCR_DATINV_SHIFT 0 /* ADCR_DATINV */
+#define WM8903_ADCR_DATINV_WIDTH 1 /* ADCR_DATINV */
+
+/*
+ * R39 (0x27) - Digital Microphone 0
+ */
+#define WM8903_DIGMIC_MODE_SEL 0x0100 /* DIGMIC_MODE_SEL */
+#define WM8903_DIGMIC_MODE_SEL_MASK 0x0100 /* DIGMIC_MODE_SEL */
+#define WM8903_DIGMIC_MODE_SEL_SHIFT 8 /* DIGMIC_MODE_SEL */
+#define WM8903_DIGMIC_MODE_SEL_WIDTH 1 /* DIGMIC_MODE_SEL */
+#define WM8903_DIGMIC_CLK_SEL_L_MASK 0x00C0 /* DIGMIC_CLK_SEL_L - [7:6] */
+#define WM8903_DIGMIC_CLK_SEL_L_SHIFT 6 /* DIGMIC_CLK_SEL_L - [7:6] */
+#define WM8903_DIGMIC_CLK_SEL_L_WIDTH 2 /* DIGMIC_CLK_SEL_L - [7:6] */
+#define WM8903_DIGMIC_CLK_SEL_R_MASK 0x0030 /* DIGMIC_CLK_SEL_R - [5:4] */
+#define WM8903_DIGMIC_CLK_SEL_R_SHIFT 4 /* DIGMIC_CLK_SEL_R - [5:4] */
+#define WM8903_DIGMIC_CLK_SEL_R_WIDTH 2 /* DIGMIC_CLK_SEL_R - [5:4] */
+#define WM8903_DIGMIC_CLK_SEL_RT_MASK 0x000C /* DIGMIC_CLK_SEL_RT - [3:2] */
+#define WM8903_DIGMIC_CLK_SEL_RT_SHIFT 2 /* DIGMIC_CLK_SEL_RT - [3:2] */
+#define WM8903_DIGMIC_CLK_SEL_RT_WIDTH 2 /* DIGMIC_CLK_SEL_RT - [3:2] */
+#define WM8903_DIGMIC_CLK_SEL_MASK 0x0003 /* DIGMIC_CLK_SEL - [1:0] */
+#define WM8903_DIGMIC_CLK_SEL_SHIFT 0 /* DIGMIC_CLK_SEL - [1:0] */
+#define WM8903_DIGMIC_CLK_SEL_WIDTH 2 /* DIGMIC_CLK_SEL - [1:0] */
+
+/*
+ * R40 (0x28) - DRC 0
+ */
+#define WM8903_DRC_ENA 0x8000 /* DRC_ENA */
+#define WM8903_DRC_ENA_MASK 0x8000 /* DRC_ENA */
+#define WM8903_DRC_ENA_SHIFT 15 /* DRC_ENA */
+#define WM8903_DRC_ENA_WIDTH 1 /* DRC_ENA */
+#define WM8903_DRC_THRESH_HYST_MASK 0x1800 /* DRC_THRESH_HYST - [12:11] */
+#define WM8903_DRC_THRESH_HYST_SHIFT 11 /* DRC_THRESH_HYST - [12:11] */
+#define WM8903_DRC_THRESH_HYST_WIDTH 2 /* DRC_THRESH_HYST - [12:11] */
+#define WM8903_DRC_STARTUP_GAIN_MASK 0x07C0 /* DRC_STARTUP_GAIN - [10:6] */
+#define WM8903_DRC_STARTUP_GAIN_SHIFT 6 /* DRC_STARTUP_GAIN - [10:6] */
+#define WM8903_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [10:6] */
+#define WM8903_DRC_FF_DELAY 0x0020 /* DRC_FF_DELAY */
+#define WM8903_DRC_FF_DELAY_MASK 0x0020 /* DRC_FF_DELAY */
+#define WM8903_DRC_FF_DELAY_SHIFT 5 /* DRC_FF_DELAY */
+#define WM8903_DRC_FF_DELAY_WIDTH 1 /* DRC_FF_DELAY */
+#define WM8903_DRC_SMOOTH_ENA 0x0008 /* DRC_SMOOTH_ENA */
+#define WM8903_DRC_SMOOTH_ENA_MASK 0x0008 /* DRC_SMOOTH_ENA */
+#define WM8903_DRC_SMOOTH_ENA_SHIFT 3 /* DRC_SMOOTH_ENA */
+#define WM8903_DRC_SMOOTH_ENA_WIDTH 1 /* DRC_SMOOTH_ENA */
+#define WM8903_DRC_QR_ENA 0x0004 /* DRC_QR_ENA */
+#define WM8903_DRC_QR_ENA_MASK 0x0004 /* DRC_QR_ENA */
+#define WM8903_DRC_QR_ENA_SHIFT 2 /* DRC_QR_ENA */
+#define WM8903_DRC_QR_ENA_WIDTH 1 /* DRC_QR_ENA */
+#define WM8903_DRC_ANTICLIP_ENA 0x0002 /* DRC_ANTICLIP_ENA */
+#define WM8903_DRC_ANTICLIP_ENA_MASK 0x0002 /* DRC_ANTICLIP_ENA */
+#define WM8903_DRC_ANTICLIP_ENA_SHIFT 1 /* DRC_ANTICLIP_ENA */
+#define WM8903_DRC_ANTICLIP_ENA_WIDTH 1 /* DRC_ANTICLIP_ENA */
+#define WM8903_DRC_HYST_ENA 0x0001 /* DRC_HYST_ENA */
+#define WM8903_DRC_HYST_ENA_MASK 0x0001 /* DRC_HYST_ENA */
+#define WM8903_DRC_HYST_ENA_SHIFT 0 /* DRC_HYST_ENA */
+#define WM8903_DRC_HYST_ENA_WIDTH 1 /* DRC_HYST_ENA */
+
+/*
+ * R41 (0x29) - DRC 1
+ */
+#define WM8903_DRC_ATTACK_RATE_MASK 0xF000 /* DRC_ATTACK_RATE - [15:12] */
+#define WM8903_DRC_ATTACK_RATE_SHIFT 12 /* DRC_ATTACK_RATE - [15:12] */
+#define WM8903_DRC_ATTACK_RATE_WIDTH 4 /* DRC_ATTACK_RATE - [15:12] */
+#define WM8903_DRC_DECAY_RATE_MASK 0x0F00 /* DRC_DECAY_RATE - [11:8] */
+#define WM8903_DRC_DECAY_RATE_SHIFT 8 /* DRC_DECAY_RATE - [11:8] */
+#define WM8903_DRC_DECAY_RATE_WIDTH 4 /* DRC_DECAY_RATE - [11:8] */
+#define WM8903_DRC_THRESH_QR_MASK 0x00C0 /* DRC_THRESH_QR - [7:6] */
+#define WM8903_DRC_THRESH_QR_SHIFT 6 /* DRC_THRESH_QR - [7:6] */
+#define WM8903_DRC_THRESH_QR_WIDTH 2 /* DRC_THRESH_QR - [7:6] */
+#define WM8903_DRC_RATE_QR_MASK 0x0030 /* DRC_RATE_QR - [5:4] */
+#define WM8903_DRC_RATE_QR_SHIFT 4 /* DRC_RATE_QR - [5:4] */
+#define WM8903_DRC_RATE_QR_WIDTH 2 /* DRC_RATE_QR - [5:4] */
+#define WM8903_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */
+#define WM8903_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */
+#define WM8903_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */
+#define WM8903_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */
+#define WM8903_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */
+#define WM8903_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */
+
+/*
+ * R42 (0x2A) - DRC 2
+ */
+#define WM8903_DRC_R0_SLOPE_COMP_MASK 0x0038 /* DRC_R0_SLOPE_COMP - [5:3] */
+#define WM8903_DRC_R0_SLOPE_COMP_SHIFT 3 /* DRC_R0_SLOPE_COMP - [5:3] */
+#define WM8903_DRC_R0_SLOPE_COMP_WIDTH 3 /* DRC_R0_SLOPE_COMP - [5:3] */
+#define WM8903_DRC_R1_SLOPE_COMP_MASK 0x0007 /* DRC_R1_SLOPE_COMP - [2:0] */
+#define WM8903_DRC_R1_SLOPE_COMP_SHIFT 0 /* DRC_R1_SLOPE_COMP - [2:0] */
+#define WM8903_DRC_R1_SLOPE_COMP_WIDTH 3 /* DRC_R1_SLOPE_COMP - [2:0] */
+
+/*
+ * R43 (0x2B) - DRC 3
+ */
+#define WM8903_DRC_THRESH_COMP_MASK 0x07E0 /* DRC_THRESH_COMP - [10:5] */
+#define WM8903_DRC_THRESH_COMP_SHIFT 5 /* DRC_THRESH_COMP - [10:5] */
+#define WM8903_DRC_THRESH_COMP_WIDTH 6 /* DRC_THRESH_COMP - [10:5] */
+#define WM8903_DRC_AMP_COMP_MASK 0x001F /* DRC_AMP_COMP - [4:0] */
+#define WM8903_DRC_AMP_COMP_SHIFT 0 /* DRC_AMP_COMP - [4:0] */
+#define WM8903_DRC_AMP_COMP_WIDTH 5 /* DRC_AMP_COMP - [4:0] */
+
+/*
+ * R44 (0x2C) - Analogue Left Input 0
+ */
+#define WM8903_LINMUTE 0x0080 /* LINMUTE */
+#define WM8903_LINMUTE_MASK 0x0080 /* LINMUTE */
+#define WM8903_LINMUTE_SHIFT 7 /* LINMUTE */
+#define WM8903_LINMUTE_WIDTH 1 /* LINMUTE */
+#define WM8903_LIN_VOL_MASK 0x001F /* LIN_VOL - [4:0] */
+#define WM8903_LIN_VOL_SHIFT 0 /* LIN_VOL - [4:0] */
+#define WM8903_LIN_VOL_WIDTH 5 /* LIN_VOL - [4:0] */
+
+/*
+ * R45 (0x2D) - Analogue Right Input 0
+ */
+#define WM8903_RINMUTE 0x0080 /* RINMUTE */
+#define WM8903_RINMUTE_MASK 0x0080 /* RINMUTE */
+#define WM8903_RINMUTE_SHIFT 7 /* RINMUTE */
+#define WM8903_RINMUTE_WIDTH 1 /* RINMUTE */
+#define WM8903_RIN_VOL_MASK 0x001F /* RIN_VOL - [4:0] */
+#define WM8903_RIN_VOL_SHIFT 0 /* RIN_VOL - [4:0] */
+#define WM8903_RIN_VOL_WIDTH 5 /* RIN_VOL - [4:0] */
+
+/*
+ * R46 (0x2E) - Analogue Left Input 1
+ */
+#define WM8903_INL_CM_ENA 0x0040 /* INL_CM_ENA */
+#define WM8903_INL_CM_ENA_MASK 0x0040 /* INL_CM_ENA */
+#define WM8903_INL_CM_ENA_SHIFT 6 /* INL_CM_ENA */
+#define WM8903_INL_CM_ENA_WIDTH 1 /* INL_CM_ENA */
+#define WM8903_L_IP_SEL_N_MASK 0x0030 /* L_IP_SEL_N - [5:4] */
+#define WM8903_L_IP_SEL_N_SHIFT 4 /* L_IP_SEL_N - [5:4] */
+#define WM8903_L_IP_SEL_N_WIDTH 2 /* L_IP_SEL_N - [5:4] */
+#define WM8903_L_IP_SEL_P_MASK 0x000C /* L_IP_SEL_P - [3:2] */
+#define WM8903_L_IP_SEL_P_SHIFT 2 /* L_IP_SEL_P - [3:2] */
+#define WM8903_L_IP_SEL_P_WIDTH 2 /* L_IP_SEL_P - [3:2] */
+#define WM8903_L_MODE_MASK 0x0003 /* L_MODE - [1:0] */
+#define WM8903_L_MODE_SHIFT 0 /* L_MODE - [1:0] */
+#define WM8903_L_MODE_WIDTH 2 /* L_MODE - [1:0] */
+
+/*
+ * R47 (0x2F) - Analogue Right Input 1
+ */
+#define WM8903_INR_CM_ENA 0x0040 /* INR_CM_ENA */
+#define WM8903_INR_CM_ENA_MASK 0x0040 /* INR_CM_ENA */
+#define WM8903_INR_CM_ENA_SHIFT 6 /* INR_CM_ENA */
+#define WM8903_INR_CM_ENA_WIDTH 1 /* INR_CM_ENA */
+#define WM8903_R_IP_SEL_N_MASK 0x0030 /* R_IP_SEL_N - [5:4] */
+#define WM8903_R_IP_SEL_N_SHIFT 4 /* R_IP_SEL_N - [5:4] */
+#define WM8903_R_IP_SEL_N_WIDTH 2 /* R_IP_SEL_N - [5:4] */
+#define WM8903_R_IP_SEL_P_MASK 0x000C /* R_IP_SEL_P - [3:2] */
+#define WM8903_R_IP_SEL_P_SHIFT 2 /* R_IP_SEL_P - [3:2] */
+#define WM8903_R_IP_SEL_P_WIDTH 2 /* R_IP_SEL_P - [3:2] */
+#define WM8903_R_MODE_MASK 0x0003 /* R_MODE - [1:0] */
+#define WM8903_R_MODE_SHIFT 0 /* R_MODE - [1:0] */
+#define WM8903_R_MODE_WIDTH 2 /* R_MODE - [1:0] */
+
+/*
+ * R50 (0x32) - Analogue Left Mix 0
+ */
+#define WM8903_DACL_TO_MIXOUTL 0x0008 /* DACL_TO_MIXOUTL */
+#define WM8903_DACL_TO_MIXOUTL_MASK 0x0008 /* DACL_TO_MIXOUTL */
+#define WM8903_DACL_TO_MIXOUTL_SHIFT 3 /* DACL_TO_MIXOUTL */
+#define WM8903_DACL_TO_MIXOUTL_WIDTH 1 /* DACL_TO_MIXOUTL */
+#define WM8903_DACR_TO_MIXOUTL 0x0004 /* DACR_TO_MIXOUTL */
+#define WM8903_DACR_TO_MIXOUTL_MASK 0x0004 /* DACR_TO_MIXOUTL */
+#define WM8903_DACR_TO_MIXOUTL_SHIFT 2 /* DACR_TO_MIXOUTL */
+#define WM8903_DACR_TO_MIXOUTL_WIDTH 1 /* DACR_TO_MIXOUTL */
+#define WM8903_BYPASSL_TO_MIXOUTL 0x0002 /* BYPASSL_TO_MIXOUTL */
+#define WM8903_BYPASSL_TO_MIXOUTL_MASK 0x0002 /* BYPASSL_TO_MIXOUTL */
+#define WM8903_BYPASSL_TO_MIXOUTL_SHIFT 1 /* BYPASSL_TO_MIXOUTL */
+#define WM8903_BYPASSL_TO_MIXOUTL_WIDTH 1 /* BYPASSL_TO_MIXOUTL */
+#define WM8903_BYPASSR_TO_MIXOUTL 0x0001 /* BYPASSR_TO_MIXOUTL */
+#define WM8903_BYPASSR_TO_MIXOUTL_MASK 0x0001 /* BYPASSR_TO_MIXOUTL */
+#define WM8903_BYPASSR_TO_MIXOUTL_SHIFT 0 /* BYPASSR_TO_MIXOUTL */
+#define WM8903_BYPASSR_TO_MIXOUTL_WIDTH 1 /* BYPASSR_TO_MIXOUTL */
+
+/*
+ * R51 (0x33) - Analogue Right Mix 0
+ */
+#define WM8903_DACL_TO_MIXOUTR 0x0008 /* DACL_TO_MIXOUTR */
+#define WM8903_DACL_TO_MIXOUTR_MASK 0x0008 /* DACL_TO_MIXOUTR */
+#define WM8903_DACL_TO_MIXOUTR_SHIFT 3 /* DACL_TO_MIXOUTR */
+#define WM8903_DACL_TO_MIXOUTR_WIDTH 1 /* DACL_TO_MIXOUTR */
+#define WM8903_DACR_TO_MIXOUTR 0x0004 /* DACR_TO_MIXOUTR */
+#define WM8903_DACR_TO_MIXOUTR_MASK 0x0004 /* DACR_TO_MIXOUTR */
+#define WM8903_DACR_TO_MIXOUTR_SHIFT 2 /* DACR_TO_MIXOUTR */
+#define WM8903_DACR_TO_MIXOUTR_WIDTH 1 /* DACR_TO_MIXOUTR */
+#define WM8903_BYPASSL_TO_MIXOUTR 0x0002 /* BYPASSL_TO_MIXOUTR */
+#define WM8903_BYPASSL_TO_MIXOUTR_MASK 0x0002 /* BYPASSL_TO_MIXOUTR */
+#define WM8903_BYPASSL_TO_MIXOUTR_SHIFT 1 /* BYPASSL_TO_MIXOUTR */
+#define WM8903_BYPASSL_TO_MIXOUTR_WIDTH 1 /* BYPASSL_TO_MIXOUTR */
+#define WM8903_BYPASSR_TO_MIXOUTR 0x0001 /* BYPASSR_TO_MIXOUTR */
+#define WM8903_BYPASSR_TO_MIXOUTR_MASK 0x0001 /* BYPASSR_TO_MIXOUTR */
+#define WM8903_BYPASSR_TO_MIXOUTR_SHIFT 0 /* BYPASSR_TO_MIXOUTR */
+#define WM8903_BYPASSR_TO_MIXOUTR_WIDTH 1 /* BYPASSR_TO_MIXOUTR */
+
+/*
+ * R52 (0x34) - Analogue Spk Mix Left 0
+ */
+#define WM8903_DACL_TO_MIXSPKL 0x0008 /* DACL_TO_MIXSPKL */
+#define WM8903_DACL_TO_MIXSPKL_MASK 0x0008 /* DACL_TO_MIXSPKL */
+#define WM8903_DACL_TO_MIXSPKL_SHIFT 3 /* DACL_TO_MIXSPKL */
+#define WM8903_DACL_TO_MIXSPKL_WIDTH 1 /* DACL_TO_MIXSPKL */
+#define WM8903_DACR_TO_MIXSPKL 0x0004 /* DACR_TO_MIXSPKL */
+#define WM8903_DACR_TO_MIXSPKL_MASK 0x0004 /* DACR_TO_MIXSPKL */
+#define WM8903_DACR_TO_MIXSPKL_SHIFT 2 /* DACR_TO_MIXSPKL */
+#define WM8903_DACR_TO_MIXSPKL_WIDTH 1 /* DACR_TO_MIXSPKL */
+#define WM8903_BYPASSL_TO_MIXSPKL 0x0002 /* BYPASSL_TO_MIXSPKL */
+#define WM8903_BYPASSL_TO_MIXSPKL_MASK 0x0002 /* BYPASSL_TO_MIXSPKL */
+#define WM8903_BYPASSL_TO_MIXSPKL_SHIFT 1 /* BYPASSL_TO_MIXSPKL */
+#define WM8903_BYPASSL_TO_MIXSPKL_WIDTH 1 /* BYPASSL_TO_MIXSPKL */
+#define WM8903_BYPASSR_TO_MIXSPKL 0x0001 /* BYPASSR_TO_MIXSPKL */
+#define WM8903_BYPASSR_TO_MIXSPKL_MASK 0x0001 /* BYPASSR_TO_MIXSPKL */
+#define WM8903_BYPASSR_TO_MIXSPKL_SHIFT 0 /* BYPASSR_TO_MIXSPKL */
+#define WM8903_BYPASSR_TO_MIXSPKL_WIDTH 1 /* BYPASSR_TO_MIXSPKL */
+
+/*
+ * R53 (0x35) - Analogue Spk Mix Left 1
+ */
+#define WM8903_DACL_MIXSPKL_VOL 0x0008 /* DACL_MIXSPKL_VOL */
+#define WM8903_DACL_MIXSPKL_VOL_MASK 0x0008 /* DACL_MIXSPKL_VOL */
+#define WM8903_DACL_MIXSPKL_VOL_SHIFT 3 /* DACL_MIXSPKL_VOL */
+#define WM8903_DACL_MIXSPKL_VOL_WIDTH 1 /* DACL_MIXSPKL_VOL */
+#define WM8903_DACR_MIXSPKL_VOL 0x0004 /* DACR_MIXSPKL_VOL */
+#define WM8903_DACR_MIXSPKL_VOL_MASK 0x0004 /* DACR_MIXSPKL_VOL */
+#define WM8903_DACR_MIXSPKL_VOL_SHIFT 2 /* DACR_MIXSPKL_VOL */
+#define WM8903_DACR_MIXSPKL_VOL_WIDTH 1 /* DACR_MIXSPKL_VOL */
+#define WM8903_BYPASSL_MIXSPKL_VOL 0x0002 /* BYPASSL_MIXSPKL_VOL */
+#define WM8903_BYPASSL_MIXSPKL_VOL_MASK 0x0002 /* BYPASSL_MIXSPKL_VOL */
+#define WM8903_BYPASSL_MIXSPKL_VOL_SHIFT 1 /* BYPASSL_MIXSPKL_VOL */
+#define WM8903_BYPASSL_MIXSPKL_VOL_WIDTH 1 /* BYPASSL_MIXSPKL_VOL */
+#define WM8903_BYPASSR_MIXSPKL_VOL 0x0001 /* BYPASSR_MIXSPKL_VOL */
+#define WM8903_BYPASSR_MIXSPKL_VOL_MASK 0x0001 /* BYPASSR_MIXSPKL_VOL */
+#define WM8903_BYPASSR_MIXSPKL_VOL_SHIFT 0 /* BYPASSR_MIXSPKL_VOL */
+#define WM8903_BYPASSR_MIXSPKL_VOL_WIDTH 1 /* BYPASSR_MIXSPKL_VOL */
+
+/*
+ * R54 (0x36) - Analogue Spk Mix Right 0
+ */
+#define WM8903_DACL_TO_MIXSPKR 0x0008 /* DACL_TO_MIXSPKR */
+#define WM8903_DACL_TO_MIXSPKR_MASK 0x0008 /* DACL_TO_MIXSPKR */
+#define WM8903_DACL_TO_MIXSPKR_SHIFT 3 /* DACL_TO_MIXSPKR */
+#define WM8903_DACL_TO_MIXSPKR_WIDTH 1 /* DACL_TO_MIXSPKR */
+#define WM8903_DACR_TO_MIXSPKR 0x0004 /* DACR_TO_MIXSPKR */
+#define WM8903_DACR_TO_MIXSPKR_MASK 0x0004 /* DACR_TO_MIXSPKR */
+#define WM8903_DACR_TO_MIXSPKR_SHIFT 2 /* DACR_TO_MIXSPKR */
+#define WM8903_DACR_TO_MIXSPKR_WIDTH 1 /* DACR_TO_MIXSPKR */
+#define WM8903_BYPASSL_TO_MIXSPKR 0x0002 /* BYPASSL_TO_MIXSPKR */
+#define WM8903_BYPASSL_TO_MIXSPKR_MASK 0x0002 /* BYPASSL_TO_MIXSPKR */
+#define WM8903_BYPASSL_TO_MIXSPKR_SHIFT 1 /* BYPASSL_TO_MIXSPKR */
+#define WM8903_BYPASSL_TO_MIXSPKR_WIDTH 1 /* BYPASSL_TO_MIXSPKR */
+#define WM8903_BYPASSR_TO_MIXSPKR 0x0001 /* BYPASSR_TO_MIXSPKR */
+#define WM8903_BYPASSR_TO_MIXSPKR_MASK 0x0001 /* BYPASSR_TO_MIXSPKR */
+#define WM8903_BYPASSR_TO_MIXSPKR_SHIFT 0 /* BYPASSR_TO_MIXSPKR */
+#define WM8903_BYPASSR_TO_MIXSPKR_WIDTH 1 /* BYPASSR_TO_MIXSPKR */
+
+/*
+ * R55 (0x37) - Analogue Spk Mix Right 1
+ */
+#define WM8903_DACL_MIXSPKR_VOL 0x0008 /* DACL_MIXSPKR_VOL */
+#define WM8903_DACL_MIXSPKR_VOL_MASK 0x0008 /* DACL_MIXSPKR_VOL */
+#define WM8903_DACL_MIXSPKR_VOL_SHIFT 3 /* DACL_MIXSPKR_VOL */
+#define WM8903_DACL_MIXSPKR_VOL_WIDTH 1 /* DACL_MIXSPKR_VOL */
+#define WM8903_DACR_MIXSPKR_VOL 0x0004 /* DACR_MIXSPKR_VOL */
+#define WM8903_DACR_MIXSPKR_VOL_MASK 0x0004 /* DACR_MIXSPKR_VOL */
+#define WM8903_DACR_MIXSPKR_VOL_SHIFT 2 /* DACR_MIXSPKR_VOL */
+#define WM8903_DACR_MIXSPKR_VOL_WIDTH 1 /* DACR_MIXSPKR_VOL */
+#define WM8903_BYPASSL_MIXSPKR_VOL 0x0002 /* BYPASSL_MIXSPKR_VOL */
+#define WM8903_BYPASSL_MIXSPKR_VOL_MASK 0x0002 /* BYPASSL_MIXSPKR_VOL */
+#define WM8903_BYPASSL_MIXSPKR_VOL_SHIFT 1 /* BYPASSL_MIXSPKR_VOL */
+#define WM8903_BYPASSL_MIXSPKR_VOL_WIDTH 1 /* BYPASSL_MIXSPKR_VOL */
+#define WM8903_BYPASSR_MIXSPKR_VOL 0x0001 /* BYPASSR_MIXSPKR_VOL */
+#define WM8903_BYPASSR_MIXSPKR_VOL_MASK 0x0001 /* BYPASSR_MIXSPKR_VOL */
+#define WM8903_BYPASSR_MIXSPKR_VOL_SHIFT 0 /* BYPASSR_MIXSPKR_VOL */
+#define WM8903_BYPASSR_MIXSPKR_VOL_WIDTH 1 /* BYPASSR_MIXSPKR_VOL */
+
+/*
+ * R57 (0x39) - Analogue OUT1 Left
+ */
+#define WM8903_HPL_MUTE 0x0100 /* HPL_MUTE */
+#define WM8903_HPL_MUTE_MASK 0x0100 /* HPL_MUTE */
+#define WM8903_HPL_MUTE_SHIFT 8 /* HPL_MUTE */
+#define WM8903_HPL_MUTE_WIDTH 1 /* HPL_MUTE */
+#define WM8903_HPOUTVU 0x0080 /* HPOUTVU */
+#define WM8903_HPOUTVU_MASK 0x0080 /* HPOUTVU */
+#define WM8903_HPOUTVU_SHIFT 7 /* HPOUTVU */
+#define WM8903_HPOUTVU_WIDTH 1 /* HPOUTVU */
+#define WM8903_HPOUTLZC 0x0040 /* HPOUTLZC */
+#define WM8903_HPOUTLZC_MASK 0x0040 /* HPOUTLZC */
+#define WM8903_HPOUTLZC_SHIFT 6 /* HPOUTLZC */
+#define WM8903_HPOUTLZC_WIDTH 1 /* HPOUTLZC */
+#define WM8903_HPOUTL_VOL_MASK 0x003F /* HPOUTL_VOL - [5:0] */
+#define WM8903_HPOUTL_VOL_SHIFT 0 /* HPOUTL_VOL - [5:0] */
+#define WM8903_HPOUTL_VOL_WIDTH 6 /* HPOUTL_VOL - [5:0] */
+
+/*
+ * R58 (0x3A) - Analogue OUT1 Right
+ */
+#define WM8903_HPR_MUTE 0x0100 /* HPR_MUTE */
+#define WM8903_HPR_MUTE_MASK 0x0100 /* HPR_MUTE */
+#define WM8903_HPR_MUTE_SHIFT 8 /* HPR_MUTE */
+#define WM8903_HPR_MUTE_WIDTH 1 /* HPR_MUTE */
+#define WM8903_HPOUTVU 0x0080 /* HPOUTVU */
+#define WM8903_HPOUTVU_MASK 0x0080 /* HPOUTVU */
+#define WM8903_HPOUTVU_SHIFT 7 /* HPOUTVU */
+#define WM8903_HPOUTVU_WIDTH 1 /* HPOUTVU */
+#define WM8903_HPOUTRZC 0x0040 /* HPOUTRZC */
+#define WM8903_HPOUTRZC_MASK 0x0040 /* HPOUTRZC */
+#define WM8903_HPOUTRZC_SHIFT 6 /* HPOUTRZC */
+#define WM8903_HPOUTRZC_WIDTH 1 /* HPOUTRZC */
+#define WM8903_HPOUTR_VOL_MASK 0x003F /* HPOUTR_VOL - [5:0] */
+#define WM8903_HPOUTR_VOL_SHIFT 0 /* HPOUTR_VOL - [5:0] */
+#define WM8903_HPOUTR_VOL_WIDTH 6 /* HPOUTR_VOL - [5:0] */
+
+/*
+ * R59 (0x3B) - Analogue OUT2 Left
+ */
+#define WM8903_LINEOUTL_MUTE 0x0100 /* LINEOUTL_MUTE */
+#define WM8903_LINEOUTL_MUTE_MASK 0x0100 /* LINEOUTL_MUTE */
+#define WM8903_LINEOUTL_MUTE_SHIFT 8 /* LINEOUTL_MUTE */
+#define WM8903_LINEOUTL_MUTE_WIDTH 1 /* LINEOUTL_MUTE */
+#define WM8903_LINEOUTVU 0x0080 /* LINEOUTVU */
+#define WM8903_LINEOUTVU_MASK 0x0080 /* LINEOUTVU */
+#define WM8903_LINEOUTVU_SHIFT 7 /* LINEOUTVU */
+#define WM8903_LINEOUTVU_WIDTH 1 /* LINEOUTVU */
+#define WM8903_LINEOUTLZC 0x0040 /* LINEOUTLZC */
+#define WM8903_LINEOUTLZC_MASK 0x0040 /* LINEOUTLZC */
+#define WM8903_LINEOUTLZC_SHIFT 6 /* LINEOUTLZC */
+#define WM8903_LINEOUTLZC_WIDTH 1 /* LINEOUTLZC */
+#define WM8903_LINEOUTL_VOL_MASK 0x003F /* LINEOUTL_VOL - [5:0] */
+#define WM8903_LINEOUTL_VOL_SHIFT 0 /* LINEOUTL_VOL - [5:0] */
+#define WM8903_LINEOUTL_VOL_WIDTH 6 /* LINEOUTL_VOL - [5:0] */
+
+/*
+ * R60 (0x3C) - Analogue OUT2 Right
+ */
+#define WM8903_LINEOUTR_MUTE 0x0100 /* LINEOUTR_MUTE */
+#define WM8903_LINEOUTR_MUTE_MASK 0x0100 /* LINEOUTR_MUTE */
+#define WM8903_LINEOUTR_MUTE_SHIFT 8 /* LINEOUTR_MUTE */
+#define WM8903_LINEOUTR_MUTE_WIDTH 1 /* LINEOUTR_MUTE */
+#define WM8903_LINEOUTVU 0x0080 /* LINEOUTVU */
+#define WM8903_LINEOUTVU_MASK 0x0080 /* LINEOUTVU */
+#define WM8903_LINEOUTVU_SHIFT 7 /* LINEOUTVU */
+#define WM8903_LINEOUTVU_WIDTH 1 /* LINEOUTVU */
+#define WM8903_LINEOUTRZC 0x0040 /* LINEOUTRZC */
+#define WM8903_LINEOUTRZC_MASK 0x0040 /* LINEOUTRZC */
+#define WM8903_LINEOUTRZC_SHIFT 6 /* LINEOUTRZC */
+#define WM8903_LINEOUTRZC_WIDTH 1 /* LINEOUTRZC */
+#define WM8903_LINEOUTR_VOL_MASK 0x003F /* LINEOUTR_VOL - [5:0] */
+#define WM8903_LINEOUTR_VOL_SHIFT 0 /* LINEOUTR_VOL - [5:0] */
+#define WM8903_LINEOUTR_VOL_WIDTH 6 /* LINEOUTR_VOL - [5:0] */
+
+/*
+ * R62 (0x3E) - Analogue OUT3 Left
+ */
+#define WM8903_SPKL_MUTE 0x0100 /* SPKL_MUTE */
+#define WM8903_SPKL_MUTE_MASK 0x0100 /* SPKL_MUTE */
+#define WM8903_SPKL_MUTE_SHIFT 8 /* SPKL_MUTE */
+#define WM8903_SPKL_MUTE_WIDTH 1 /* SPKL_MUTE */
+#define WM8903_SPKVU 0x0080 /* SPKVU */
+#define WM8903_SPKVU_MASK 0x0080 /* SPKVU */
+#define WM8903_SPKVU_SHIFT 7 /* SPKVU */
+#define WM8903_SPKVU_WIDTH 1 /* SPKVU */
+#define WM8903_SPKLZC 0x0040 /* SPKLZC */
+#define WM8903_SPKLZC_MASK 0x0040 /* SPKLZC */
+#define WM8903_SPKLZC_SHIFT 6 /* SPKLZC */
+#define WM8903_SPKLZC_WIDTH 1 /* SPKLZC */
+#define WM8903_SPKL_VOL_MASK 0x003F /* SPKL_VOL - [5:0] */
+#define WM8903_SPKL_VOL_SHIFT 0 /* SPKL_VOL - [5:0] */
+#define WM8903_SPKL_VOL_WIDTH 6 /* SPKL_VOL - [5:0] */
+
+/*
+ * R63 (0x3F) - Analogue OUT3 Right
+ */
+#define WM8903_SPKR_MUTE 0x0100 /* SPKR_MUTE */
+#define WM8903_SPKR_MUTE_MASK 0x0100 /* SPKR_MUTE */
+#define WM8903_SPKR_MUTE_SHIFT 8 /* SPKR_MUTE */
+#define WM8903_SPKR_MUTE_WIDTH 1 /* SPKR_MUTE */
+#define WM8903_SPKVU 0x0080 /* SPKVU */
+#define WM8903_SPKVU_MASK 0x0080 /* SPKVU */
+#define WM8903_SPKVU_SHIFT 7 /* SPKVU */
+#define WM8903_SPKVU_WIDTH 1 /* SPKVU */
+#define WM8903_SPKRZC 0x0040 /* SPKRZC */
+#define WM8903_SPKRZC_MASK 0x0040 /* SPKRZC */
+#define WM8903_SPKRZC_SHIFT 6 /* SPKRZC */
+#define WM8903_SPKRZC_WIDTH 1 /* SPKRZC */
+#define WM8903_SPKR_VOL_MASK 0x003F /* SPKR_VOL - [5:0] */
+#define WM8903_SPKR_VOL_SHIFT 0 /* SPKR_VOL - [5:0] */
+#define WM8903_SPKR_VOL_WIDTH 6 /* SPKR_VOL - [5:0] */
+
+/*
+ * R65 (0x41) - Analogue SPK Output Control 0
+ */
+#define WM8903_SPK_DISCHARGE 0x0002 /* SPK_DISCHARGE */
+#define WM8903_SPK_DISCHARGE_MASK 0x0002 /* SPK_DISCHARGE */
+#define WM8903_SPK_DISCHARGE_SHIFT 1 /* SPK_DISCHARGE */
+#define WM8903_SPK_DISCHARGE_WIDTH 1 /* SPK_DISCHARGE */
+#define WM8903_VROI 0x0001 /* VROI */
+#define WM8903_VROI_MASK 0x0001 /* VROI */
+#define WM8903_VROI_SHIFT 0 /* VROI */
+#define WM8903_VROI_WIDTH 1 /* VROI */
+
+/*
+ * R67 (0x43) - DC Servo 0
+ */
+#define WM8903_DCS_MASTER_ENA 0x0010 /* DCS_MASTER_ENA */
+#define WM8903_DCS_MASTER_ENA_MASK 0x0010 /* DCS_MASTER_ENA */
+#define WM8903_DCS_MASTER_ENA_SHIFT 4 /* DCS_MASTER_ENA */
+#define WM8903_DCS_MASTER_ENA_WIDTH 1 /* DCS_MASTER_ENA */
+#define WM8903_DCS_ENA_MASK 0x000F /* DCS_ENA - [3:0] */
+#define WM8903_DCS_ENA_SHIFT 0 /* DCS_ENA - [3:0] */
+#define WM8903_DCS_ENA_WIDTH 4 /* DCS_ENA - [3:0] */
+
+/*
+ * R69 (0x45) - DC Servo 2
+ */
+#define WM8903_DCS_MODE_MASK 0x0003 /* DCS_MODE - [1:0] */
+#define WM8903_DCS_MODE_SHIFT 0 /* DCS_MODE - [1:0] */
+#define WM8903_DCS_MODE_WIDTH 2 /* DCS_MODE - [1:0] */
+
+/*
+ * R90 (0x5A) - Analogue HP 0
+ */
+#define WM8903_HPL_RMV_SHORT 0x0080 /* HPL_RMV_SHORT */
+#define WM8903_HPL_RMV_SHORT_MASK 0x0080 /* HPL_RMV_SHORT */
+#define WM8903_HPL_RMV_SHORT_SHIFT 7 /* HPL_RMV_SHORT */
+#define WM8903_HPL_RMV_SHORT_WIDTH 1 /* HPL_RMV_SHORT */
+#define WM8903_HPL_ENA_OUTP 0x0040 /* HPL_ENA_OUTP */
+#define WM8903_HPL_ENA_OUTP_MASK 0x0040 /* HPL_ENA_OUTP */
+#define WM8903_HPL_ENA_OUTP_SHIFT 6 /* HPL_ENA_OUTP */
+#define WM8903_HPL_ENA_OUTP_WIDTH 1 /* HPL_ENA_OUTP */
+#define WM8903_HPL_ENA_DLY 0x0020 /* HPL_ENA_DLY */
+#define WM8903_HPL_ENA_DLY_MASK 0x0020 /* HPL_ENA_DLY */
+#define WM8903_HPL_ENA_DLY_SHIFT 5 /* HPL_ENA_DLY */
+#define WM8903_HPL_ENA_DLY_WIDTH 1 /* HPL_ENA_DLY */
+#define WM8903_HPL_ENA 0x0010 /* HPL_ENA */
+#define WM8903_HPL_ENA_MASK 0x0010 /* HPL_ENA */
+#define WM8903_HPL_ENA_SHIFT 4 /* HPL_ENA */
+#define WM8903_HPL_ENA_WIDTH 1 /* HPL_ENA */
+#define WM8903_HPR_RMV_SHORT 0x0008 /* HPR_RMV_SHORT */
+#define WM8903_HPR_RMV_SHORT_MASK 0x0008 /* HPR_RMV_SHORT */
+#define WM8903_HPR_RMV_SHORT_SHIFT 3 /* HPR_RMV_SHORT */
+#define WM8903_HPR_RMV_SHORT_WIDTH 1 /* HPR_RMV_SHORT */
+#define WM8903_HPR_ENA_OUTP 0x0004 /* HPR_ENA_OUTP */
+#define WM8903_HPR_ENA_OUTP_MASK 0x0004 /* HPR_ENA_OUTP */
+#define WM8903_HPR_ENA_OUTP_SHIFT 2 /* HPR_ENA_OUTP */
+#define WM8903_HPR_ENA_OUTP_WIDTH 1 /* HPR_ENA_OUTP */
+#define WM8903_HPR_ENA_DLY 0x0002 /* HPR_ENA_DLY */
+#define WM8903_HPR_ENA_DLY_MASK 0x0002 /* HPR_ENA_DLY */
+#define WM8903_HPR_ENA_DLY_SHIFT 1 /* HPR_ENA_DLY */
+#define WM8903_HPR_ENA_DLY_WIDTH 1 /* HPR_ENA_DLY */
+#define WM8903_HPR_ENA 0x0001 /* HPR_ENA */
+#define WM8903_HPR_ENA_MASK 0x0001 /* HPR_ENA */
+#define WM8903_HPR_ENA_SHIFT 0 /* HPR_ENA */
+#define WM8903_HPR_ENA_WIDTH 1 /* HPR_ENA */
+
+/*
+ * R94 (0x5E) - Analogue Lineout 0
+ */
+#define WM8903_LINEOUTL_RMV_SHORT 0x0080 /* LINEOUTL_RMV_SHORT */
+#define WM8903_LINEOUTL_RMV_SHORT_MASK 0x0080 /* LINEOUTL_RMV_SHORT */
+#define WM8903_LINEOUTL_RMV_SHORT_SHIFT 7 /* LINEOUTL_RMV_SHORT */
+#define WM8903_LINEOUTL_RMV_SHORT_WIDTH 1 /* LINEOUTL_RMV_SHORT */
+#define WM8903_LINEOUTL_ENA_OUTP 0x0040 /* LINEOUTL_ENA_OUTP */
+#define WM8903_LINEOUTL_ENA_OUTP_MASK 0x0040 /* LINEOUTL_ENA_OUTP */
+#define WM8903_LINEOUTL_ENA_OUTP_SHIFT 6 /* LINEOUTL_ENA_OUTP */
+#define WM8903_LINEOUTL_ENA_OUTP_WIDTH 1 /* LINEOUTL_ENA_OUTP */
+#define WM8903_LINEOUTL_ENA_DLY 0x0020 /* LINEOUTL_ENA_DLY */
+#define WM8903_LINEOUTL_ENA_DLY_MASK 0x0020 /* LINEOUTL_ENA_DLY */
+#define WM8903_LINEOUTL_ENA_DLY_SHIFT 5 /* LINEOUTL_ENA_DLY */
+#define WM8903_LINEOUTL_ENA_DLY_WIDTH 1 /* LINEOUTL_ENA_DLY */
+#define WM8903_LINEOUTL_ENA 0x0010 /* LINEOUTL_ENA */
+#define WM8903_LINEOUTL_ENA_MASK 0x0010 /* LINEOUTL_ENA */
+#define WM8903_LINEOUTL_ENA_SHIFT 4 /* LINEOUTL_ENA */
+#define WM8903_LINEOUTL_ENA_WIDTH 1 /* LINEOUTL_ENA */
+#define WM8903_LINEOUTR_RMV_SHORT 0x0008 /* LINEOUTR_RMV_SHORT */
+#define WM8903_LINEOUTR_RMV_SHORT_MASK 0x0008 /* LINEOUTR_RMV_SHORT */
+#define WM8903_LINEOUTR_RMV_SHORT_SHIFT 3 /* LINEOUTR_RMV_SHORT */
+#define WM8903_LINEOUTR_RMV_SHORT_WIDTH 1 /* LINEOUTR_RMV_SHORT */
+#define WM8903_LINEOUTR_ENA_OUTP 0x0004 /* LINEOUTR_ENA_OUTP */
+#define WM8903_LINEOUTR_ENA_OUTP_MASK 0x0004 /* LINEOUTR_ENA_OUTP */
+#define WM8903_LINEOUTR_ENA_OUTP_SHIFT 2 /* LINEOUTR_ENA_OUTP */
+#define WM8903_LINEOUTR_ENA_OUTP_WIDTH 1 /* LINEOUTR_ENA_OUTP */
+#define WM8903_LINEOUTR_ENA_DLY 0x0002 /* LINEOUTR_ENA_DLY */
+#define WM8903_LINEOUTR_ENA_DLY_MASK 0x0002 /* LINEOUTR_ENA_DLY */
+#define WM8903_LINEOUTR_ENA_DLY_SHIFT 1 /* LINEOUTR_ENA_DLY */
+#define WM8903_LINEOUTR_ENA_DLY_WIDTH 1 /* LINEOUTR_ENA_DLY */
+#define WM8903_LINEOUTR_ENA 0x0001 /* LINEOUTR_ENA */
+#define WM8903_LINEOUTR_ENA_MASK 0x0001 /* LINEOUTR_ENA */
+#define WM8903_LINEOUTR_ENA_SHIFT 0 /* LINEOUTR_ENA */
+#define WM8903_LINEOUTR_ENA_WIDTH 1 /* LINEOUTR_ENA */
+
+/*
+ * R98 (0x62) - Charge Pump 0
+ */
+#define WM8903_CP_ENA 0x0001 /* CP_ENA */
+#define WM8903_CP_ENA_MASK 0x0001 /* CP_ENA */
+#define WM8903_CP_ENA_SHIFT 0 /* CP_ENA */
+#define WM8903_CP_ENA_WIDTH 1 /* CP_ENA */
+
+/*
+ * R104 (0x68) - Class W 0
+ */
+#define WM8903_CP_DYN_FREQ 0x0002 /* CP_DYN_FREQ */
+#define WM8903_CP_DYN_FREQ_MASK 0x0002 /* CP_DYN_FREQ */
+#define WM8903_CP_DYN_FREQ_SHIFT 1 /* CP_DYN_FREQ */
+#define WM8903_CP_DYN_FREQ_WIDTH 1 /* CP_DYN_FREQ */
+#define WM8903_CP_DYN_V 0x0001 /* CP_DYN_V */
+#define WM8903_CP_DYN_V_MASK 0x0001 /* CP_DYN_V */
+#define WM8903_CP_DYN_V_SHIFT 0 /* CP_DYN_V */
+#define WM8903_CP_DYN_V_WIDTH 1 /* CP_DYN_V */
+
+/*
+ * R108 (0x6C) - Write Sequencer 0
+ */
+#define WM8903_WSEQ_ENA 0x0100 /* WSEQ_ENA */
+#define WM8903_WSEQ_ENA_MASK 0x0100 /* WSEQ_ENA */
+#define WM8903_WSEQ_ENA_SHIFT 8 /* WSEQ_ENA */
+#define WM8903_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */
+#define WM8903_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */
+#define WM8903_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */
+#define WM8903_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */
+
+/*
+ * R109 (0x6D) - Write Sequencer 1
+ */
+#define WM8903_WSEQ_DATA_WIDTH_MASK 0x7000 /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM8903_WSEQ_DATA_WIDTH_SHIFT 12 /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM8903_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [14:12] */
+#define WM8903_WSEQ_DATA_START_MASK 0x0F00 /* WSEQ_DATA_START - [11:8] */
+#define WM8903_WSEQ_DATA_START_SHIFT 8 /* WSEQ_DATA_START - [11:8] */
+#define WM8903_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [11:8] */
+#define WM8903_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */
+#define WM8903_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */
+#define WM8903_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */
+
+/*
+ * R110 (0x6E) - Write Sequencer 2
+ */
+#define WM8903_WSEQ_EOS 0x4000 /* WSEQ_EOS */
+#define WM8903_WSEQ_EOS_MASK 0x4000 /* WSEQ_EOS */
+#define WM8903_WSEQ_EOS_SHIFT 14 /* WSEQ_EOS */
+#define WM8903_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */
+#define WM8903_WSEQ_DELAY_MASK 0x0F00 /* WSEQ_DELAY - [11:8] */
+#define WM8903_WSEQ_DELAY_SHIFT 8 /* WSEQ_DELAY - [11:8] */
+#define WM8903_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [11:8] */
+#define WM8903_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */
+#define WM8903_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */
+#define WM8903_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */
+
+/*
+ * R111 (0x6F) - Write Sequencer 3
+ */
+#define WM8903_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */
+#define WM8903_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */
+#define WM8903_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */
+#define WM8903_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */
+#define WM8903_WSEQ_START 0x0100 /* WSEQ_START */
+#define WM8903_WSEQ_START_MASK 0x0100 /* WSEQ_START */
+#define WM8903_WSEQ_START_SHIFT 8 /* WSEQ_START */
+#define WM8903_WSEQ_START_WIDTH 1 /* WSEQ_START */
+#define WM8903_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */
+#define WM8903_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */
+#define WM8903_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */
+
+/*
+ * R112 (0x70) - Write Sequencer 4
+ */
+#define WM8903_WSEQ_CURRENT_INDEX_MASK 0x03F0 /* WSEQ_CURRENT_INDEX - [9:4] */
+#define WM8903_WSEQ_CURRENT_INDEX_SHIFT 4 /* WSEQ_CURRENT_INDEX - [9:4] */
+#define WM8903_WSEQ_CURRENT_INDEX_WIDTH 6 /* WSEQ_CURRENT_INDEX - [9:4] */
+#define WM8903_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */
+#define WM8903_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */
+#define WM8903_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */
+#define WM8903_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */
+
+/*
+ * R114 (0x72) - Control Interface
+ */
+#define WM8903_MASK_WRITE_ENA 0x0001 /* MASK_WRITE_ENA */
+#define WM8903_MASK_WRITE_ENA_MASK 0x0001 /* MASK_WRITE_ENA */
+#define WM8903_MASK_WRITE_ENA_SHIFT 0 /* MASK_WRITE_ENA */
+#define WM8903_MASK_WRITE_ENA_WIDTH 1 /* MASK_WRITE_ENA */
+
+/*
+ * R116 (0x74) - GPIO Control 1
+ */
+#define WM8903_GP1_FN_MASK 0x1F00 /* GP1_FN - [12:8] */
+#define WM8903_GP1_FN_SHIFT 8 /* GP1_FN - [12:8] */
+#define WM8903_GP1_FN_WIDTH 5 /* GP1_FN - [12:8] */
+#define WM8903_GP1_DIR 0x0080 /* GP1_DIR */
+#define WM8903_GP1_DIR_MASK 0x0080 /* GP1_DIR */
+#define WM8903_GP1_DIR_SHIFT 7 /* GP1_DIR */
+#define WM8903_GP1_DIR_WIDTH 1 /* GP1_DIR */
+#define WM8903_GP1_OP_CFG 0x0040 /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_MASK 0x0040 /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_SHIFT 6 /* GP1_OP_CFG */
+#define WM8903_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */
+#define WM8903_GP1_IP_CFG 0x0020 /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_MASK 0x0020 /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_SHIFT 5 /* GP1_IP_CFG */
+#define WM8903_GP1_IP_CFG_WIDTH 1 /* GP1_IP_CFG */
+#define WM8903_GP1_LVL 0x0010 /* GP1_LVL */
+#define WM8903_GP1_LVL_MASK 0x0010 /* GP1_LVL */
+#define WM8903_GP1_LVL_SHIFT 4 /* GP1_LVL */
+#define WM8903_GP1_LVL_WIDTH 1 /* GP1_LVL */
+#define WM8903_GP1_PD 0x0008 /* GP1_PD */
+#define WM8903_GP1_PD_MASK 0x0008 /* GP1_PD */
+#define WM8903_GP1_PD_SHIFT 3 /* GP1_PD */
+#define WM8903_GP1_PD_WIDTH 1 /* GP1_PD */
+#define WM8903_GP1_PU 0x0004 /* GP1_PU */
+#define WM8903_GP1_PU_MASK 0x0004 /* GP1_PU */
+#define WM8903_GP1_PU_SHIFT 2 /* GP1_PU */
+#define WM8903_GP1_PU_WIDTH 1 /* GP1_PU */
+#define WM8903_GP1_INTMODE 0x0002 /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_MASK 0x0002 /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_SHIFT 1 /* GP1_INTMODE */
+#define WM8903_GP1_INTMODE_WIDTH 1 /* GP1_INTMODE */
+#define WM8903_GP1_DB 0x0001 /* GP1_DB */
+#define WM8903_GP1_DB_MASK 0x0001 /* GP1_DB */
+#define WM8903_GP1_DB_SHIFT 0 /* GP1_DB */
+#define WM8903_GP1_DB_WIDTH 1 /* GP1_DB */
+
+/*
+ * R117 (0x75) - GPIO Control 2
+ */
+#define WM8903_GP2_FN_MASK 0x1F00 /* GP2_FN - [12:8] */
+#define WM8903_GP2_FN_SHIFT 8 /* GP2_FN - [12:8] */
+#define WM8903_GP2_FN_WIDTH 5 /* GP2_FN - [12:8] */
+#define WM8903_GP2_DIR 0x0080 /* GP2_DIR */
+#define WM8903_GP2_DIR_MASK 0x0080 /* GP2_DIR */
+#define WM8903_GP2_DIR_SHIFT 7 /* GP2_DIR */
+#define WM8903_GP2_DIR_WIDTH 1 /* GP2_DIR */
+#define WM8903_GP2_OP_CFG 0x0040 /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_MASK 0x0040 /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_SHIFT 6 /* GP2_OP_CFG */
+#define WM8903_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */
+#define WM8903_GP2_IP_CFG 0x0020 /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_MASK 0x0020 /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_SHIFT 5 /* GP2_IP_CFG */
+#define WM8903_GP2_IP_CFG_WIDTH 1 /* GP2_IP_CFG */
+#define WM8903_GP2_LVL 0x0010 /* GP2_LVL */
+#define WM8903_GP2_LVL_MASK 0x0010 /* GP2_LVL */
+#define WM8903_GP2_LVL_SHIFT 4 /* GP2_LVL */
+#define WM8903_GP2_LVL_WIDTH 1 /* GP2_LVL */
+#define WM8903_GP2_PD 0x0008 /* GP2_PD */
+#define WM8903_GP2_PD_MASK 0x0008 /* GP2_PD */
+#define WM8903_GP2_PD_SHIFT 3 /* GP2_PD */
+#define WM8903_GP2_PD_WIDTH 1 /* GP2_PD */
+#define WM8903_GP2_PU 0x0004 /* GP2_PU */
+#define WM8903_GP2_PU_MASK 0x0004 /* GP2_PU */
+#define WM8903_GP2_PU_SHIFT 2 /* GP2_PU */
+#define WM8903_GP2_PU_WIDTH 1 /* GP2_PU */
+#define WM8903_GP2_INTMODE 0x0002 /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_MASK 0x0002 /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_SHIFT 1 /* GP2_INTMODE */
+#define WM8903_GP2_INTMODE_WIDTH 1 /* GP2_INTMODE */
+#define WM8903_GP2_DB 0x0001 /* GP2_DB */
+#define WM8903_GP2_DB_MASK 0x0001 /* GP2_DB */
+#define WM8903_GP2_DB_SHIFT 0 /* GP2_DB */
+#define WM8903_GP2_DB_WIDTH 1 /* GP2_DB */
+
+/*
+ * R118 (0x76) - GPIO Control 3
+ */
+#define WM8903_GP3_FN_MASK 0x1F00 /* GP3_FN - [12:8] */
+#define WM8903_GP3_FN_SHIFT 8 /* GP3_FN - [12:8] */
+#define WM8903_GP3_FN_WIDTH 5 /* GP3_FN - [12:8] */
+#define WM8903_GP3_DIR 0x0080 /* GP3_DIR */
+#define WM8903_GP3_DIR_MASK 0x0080 /* GP3_DIR */
+#define WM8903_GP3_DIR_SHIFT 7 /* GP3_DIR */
+#define WM8903_GP3_DIR_WIDTH 1 /* GP3_DIR */
+#define WM8903_GP3_OP_CFG 0x0040 /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_MASK 0x0040 /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_SHIFT 6 /* GP3_OP_CFG */
+#define WM8903_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */
+#define WM8903_GP3_IP_CFG 0x0020 /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_MASK 0x0020 /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_SHIFT 5 /* GP3_IP_CFG */
+#define WM8903_GP3_IP_CFG_WIDTH 1 /* GP3_IP_CFG */
+#define WM8903_GP3_LVL 0x0010 /* GP3_LVL */
+#define WM8903_GP3_LVL_MASK 0x0010 /* GP3_LVL */
+#define WM8903_GP3_LVL_SHIFT 4 /* GP3_LVL */
+#define WM8903_GP3_LVL_WIDTH 1 /* GP3_LVL */
+#define WM8903_GP3_PD 0x0008 /* GP3_PD */
+#define WM8903_GP3_PD_MASK 0x0008 /* GP3_PD */
+#define WM8903_GP3_PD_SHIFT 3 /* GP3_PD */
+#define WM8903_GP3_PD_WIDTH 1 /* GP3_PD */
+#define WM8903_GP3_PU 0x0004 /* GP3_PU */
+#define WM8903_GP3_PU_MASK 0x0004 /* GP3_PU */
+#define WM8903_GP3_PU_SHIFT 2 /* GP3_PU */
+#define WM8903_GP3_PU_WIDTH 1 /* GP3_PU */
+#define WM8903_GP3_INTMODE 0x0002 /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_MASK 0x0002 /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_SHIFT 1 /* GP3_INTMODE */
+#define WM8903_GP3_INTMODE_WIDTH 1 /* GP3_INTMODE */
+#define WM8903_GP3_DB 0x0001 /* GP3_DB */
+#define WM8903_GP3_DB_MASK 0x0001 /* GP3_DB */
+#define WM8903_GP3_DB_SHIFT 0 /* GP3_DB */
+#define WM8903_GP3_DB_WIDTH 1 /* GP3_DB */
+
+/*
+ * R119 (0x77) - GPIO Control 4
+ */
+#define WM8903_GP4_FN_MASK 0x1F00 /* GP4_FN - [12:8] */
+#define WM8903_GP4_FN_SHIFT 8 /* GP4_FN - [12:8] */
+#define WM8903_GP4_FN_WIDTH 5 /* GP4_FN - [12:8] */
+#define WM8903_GP4_DIR 0x0080 /* GP4_DIR */
+#define WM8903_GP4_DIR_MASK 0x0080 /* GP4_DIR */
+#define WM8903_GP4_DIR_SHIFT 7 /* GP4_DIR */
+#define WM8903_GP4_DIR_WIDTH 1 /* GP4_DIR */
+#define WM8903_GP4_OP_CFG 0x0040 /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_MASK 0x0040 /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_SHIFT 6 /* GP4_OP_CFG */
+#define WM8903_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */
+#define WM8903_GP4_IP_CFG 0x0020 /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_MASK 0x0020 /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_SHIFT 5 /* GP4_IP_CFG */
+#define WM8903_GP4_IP_CFG_WIDTH 1 /* GP4_IP_CFG */
+#define WM8903_GP4_LVL 0x0010 /* GP4_LVL */
+#define WM8903_GP4_LVL_MASK 0x0010 /* GP4_LVL */
+#define WM8903_GP4_LVL_SHIFT 4 /* GP4_LVL */
+#define WM8903_GP4_LVL_WIDTH 1 /* GP4_LVL */
+#define WM8903_GP4_PD 0x0008 /* GP4_PD */
+#define WM8903_GP4_PD_MASK 0x0008 /* GP4_PD */
+#define WM8903_GP4_PD_SHIFT 3 /* GP4_PD */
+#define WM8903_GP4_PD_WIDTH 1 /* GP4_PD */
+#define WM8903_GP4_PU 0x0004 /* GP4_PU */
+#define WM8903_GP4_PU_MASK 0x0004 /* GP4_PU */
+#define WM8903_GP4_PU_SHIFT 2 /* GP4_PU */
+#define WM8903_GP4_PU_WIDTH 1 /* GP4_PU */
+#define WM8903_GP4_INTMODE 0x0002 /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_MASK 0x0002 /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_SHIFT 1 /* GP4_INTMODE */
+#define WM8903_GP4_INTMODE_WIDTH 1 /* GP4_INTMODE */
+#define WM8903_GP4_DB 0x0001 /* GP4_DB */
+#define WM8903_GP4_DB_MASK 0x0001 /* GP4_DB */
+#define WM8903_GP4_DB_SHIFT 0 /* GP4_DB */
+#define WM8903_GP4_DB_WIDTH 1 /* GP4_DB */
+
+/*
+ * R120 (0x78) - GPIO Control 5
+ */
+#define WM8903_GP5_FN_MASK 0x1F00 /* GP5_FN - [12:8] */
+#define WM8903_GP5_FN_SHIFT 8 /* GP5_FN - [12:8] */
+#define WM8903_GP5_FN_WIDTH 5 /* GP5_FN - [12:8] */
+#define WM8903_GP5_DIR 0x0080 /* GP5_DIR */
+#define WM8903_GP5_DIR_MASK 0x0080 /* GP5_DIR */
+#define WM8903_GP5_DIR_SHIFT 7 /* GP5_DIR */
+#define WM8903_GP5_DIR_WIDTH 1 /* GP5_DIR */
+#define WM8903_GP5_OP_CFG 0x0040 /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_MASK 0x0040 /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_SHIFT 6 /* GP5_OP_CFG */
+#define WM8903_GP5_OP_CFG_WIDTH 1 /* GP5_OP_CFG */
+#define WM8903_GP5_IP_CFG 0x0020 /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_MASK 0x0020 /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_SHIFT 5 /* GP5_IP_CFG */
+#define WM8903_GP5_IP_CFG_WIDTH 1 /* GP5_IP_CFG */
+#define WM8903_GP5_LVL 0x0010 /* GP5_LVL */
+#define WM8903_GP5_LVL_MASK 0x0010 /* GP5_LVL */
+#define WM8903_GP5_LVL_SHIFT 4 /* GP5_LVL */
+#define WM8903_GP5_LVL_WIDTH 1 /* GP5_LVL */
+#define WM8903_GP5_PD 0x0008 /* GP5_PD */
+#define WM8903_GP5_PD_MASK 0x0008 /* GP5_PD */
+#define WM8903_GP5_PD_SHIFT 3 /* GP5_PD */
+#define WM8903_GP5_PD_WIDTH 1 /* GP5_PD */
+#define WM8903_GP5_PU 0x0004 /* GP5_PU */
+#define WM8903_GP5_PU_MASK 0x0004 /* GP5_PU */
+#define WM8903_GP5_PU_SHIFT 2 /* GP5_PU */
+#define WM8903_GP5_PU_WIDTH 1 /* GP5_PU */
+#define WM8903_GP5_INTMODE 0x0002 /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_MASK 0x0002 /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_SHIFT 1 /* GP5_INTMODE */
+#define WM8903_GP5_INTMODE_WIDTH 1 /* GP5_INTMODE */
+#define WM8903_GP5_DB 0x0001 /* GP5_DB */
+#define WM8903_GP5_DB_MASK 0x0001 /* GP5_DB */
+#define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */
+#define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */
+
+/*
+ * R121 (0x79) - Interrupt Status 1
+ */
+#define WM8903_MICSHRT_EINT 0x8000 /* MICSHRT_EINT */
+#define WM8903_MICSHRT_EINT_MASK 0x8000 /* MICSHRT_EINT */
+#define WM8903_MICSHRT_EINT_SHIFT 15 /* MICSHRT_EINT */
+#define WM8903_MICSHRT_EINT_WIDTH 1 /* MICSHRT_EINT */
+#define WM8903_MICDET_EINT 0x4000 /* MICDET_EINT */
+#define WM8903_MICDET_EINT_MASK 0x4000 /* MICDET_EINT */
+#define WM8903_MICDET_EINT_SHIFT 14 /* MICDET_EINT */
+#define WM8903_MICDET_EINT_WIDTH 1 /* MICDET_EINT */
+#define WM8903_WSEQ_BUSY_EINT 0x2000 /* WSEQ_BUSY_EINT */
+#define WM8903_WSEQ_BUSY_EINT_MASK 0x2000 /* WSEQ_BUSY_EINT */
+#define WM8903_WSEQ_BUSY_EINT_SHIFT 13 /* WSEQ_BUSY_EINT */
+#define WM8903_WSEQ_BUSY_EINT_WIDTH 1 /* WSEQ_BUSY_EINT */
+#define WM8903_GP5_EINT 0x0010 /* GP5_EINT */
+#define WM8903_GP5_EINT_MASK 0x0010 /* GP5_EINT */
+#define WM8903_GP5_EINT_SHIFT 4 /* GP5_EINT */
+#define WM8903_GP5_EINT_WIDTH 1 /* GP5_EINT */
+#define WM8903_GP4_EINT 0x0008 /* GP4_EINT */
+#define WM8903_GP4_EINT_MASK 0x0008 /* GP4_EINT */
+#define WM8903_GP4_EINT_SHIFT 3 /* GP4_EINT */
+#define WM8903_GP4_EINT_WIDTH 1 /* GP4_EINT */
+#define WM8903_GP3_EINT 0x0004 /* GP3_EINT */
+#define WM8903_GP3_EINT_MASK 0x0004 /* GP3_EINT */
+#define WM8903_GP3_EINT_SHIFT 2 /* GP3_EINT */
+#define WM8903_GP3_EINT_WIDTH 1 /* GP3_EINT */
+#define WM8903_GP2_EINT 0x0002 /* GP2_EINT */
+#define WM8903_GP2_EINT_MASK 0x0002 /* GP2_EINT */
+#define WM8903_GP2_EINT_SHIFT 1 /* GP2_EINT */
+#define WM8903_GP2_EINT_WIDTH 1 /* GP2_EINT */
+#define WM8903_GP1_EINT 0x0001 /* GP1_EINT */
+#define WM8903_GP1_EINT_MASK 0x0001 /* GP1_EINT */
+#define WM8903_GP1_EINT_SHIFT 0 /* GP1_EINT */
+#define WM8903_GP1_EINT_WIDTH 1 /* GP1_EINT */
+
+/*
+ * R122 (0x7A) - Interrupt Status 1 Mask
+ */
+#define WM8903_IM_MICSHRT_EINT 0x8000 /* IM_MICSHRT_EINT */
+#define WM8903_IM_MICSHRT_EINT_MASK 0x8000 /* IM_MICSHRT_EINT */
+#define WM8903_IM_MICSHRT_EINT_SHIFT 15 /* IM_MICSHRT_EINT */
+#define WM8903_IM_MICSHRT_EINT_WIDTH 1 /* IM_MICSHRT_EINT */
+#define WM8903_IM_MICDET_EINT 0x4000 /* IM_MICDET_EINT */
+#define WM8903_IM_MICDET_EINT_MASK 0x4000 /* IM_MICDET_EINT */
+#define WM8903_IM_MICDET_EINT_SHIFT 14 /* IM_MICDET_EINT */
+#define WM8903_IM_MICDET_EINT_WIDTH 1 /* IM_MICDET_EINT */
+#define WM8903_IM_WSEQ_BUSY_EINT 0x2000 /* IM_WSEQ_BUSY_EINT */
+#define WM8903_IM_WSEQ_BUSY_EINT_MASK 0x2000 /* IM_WSEQ_BUSY_EINT */
+#define WM8903_IM_WSEQ_BUSY_EINT_SHIFT 13 /* IM_WSEQ_BUSY_EINT */
+#define WM8903_IM_WSEQ_BUSY_EINT_WIDTH 1 /* IM_WSEQ_BUSY_EINT */
+#define WM8903_IM_GP5_EINT 0x0010 /* IM_GP5_EINT */
+#define WM8903_IM_GP5_EINT_MASK 0x0010 /* IM_GP5_EINT */
+#define WM8903_IM_GP5_EINT_SHIFT 4 /* IM_GP5_EINT */
+#define WM8903_IM_GP5_EINT_WIDTH 1 /* IM_GP5_EINT */
+#define WM8903_IM_GP4_EINT 0x0008 /* IM_GP4_EINT */
+#define WM8903_IM_GP4_EINT_MASK 0x0008 /* IM_GP4_EINT */
+#define WM8903_IM_GP4_EINT_SHIFT 3 /* IM_GP4_EINT */
+#define WM8903_IM_GP4_EINT_WIDTH 1 /* IM_GP4_EINT */
+#define WM8903_IM_GP3_EINT 0x0004 /* IM_GP3_EINT */
+#define WM8903_IM_GP3_EINT_MASK 0x0004 /* IM_GP3_EINT */
+#define WM8903_IM_GP3_EINT_SHIFT 2 /* IM_GP3_EINT */
+#define WM8903_IM_GP3_EINT_WIDTH 1 /* IM_GP3_EINT */
+#define WM8903_IM_GP2_EINT 0x0002 /* IM_GP2_EINT */
+#define WM8903_IM_GP2_EINT_MASK 0x0002 /* IM_GP2_EINT */
+#define WM8903_IM_GP2_EINT_SHIFT 1 /* IM_GP2_EINT */
+#define WM8903_IM_GP2_EINT_WIDTH 1 /* IM_GP2_EINT */
+#define WM8903_IM_GP1_EINT 0x0001 /* IM_GP1_EINT */
+#define WM8903_IM_GP1_EINT_MASK 0x0001 /* IM_GP1_EINT */
+#define WM8903_IM_GP1_EINT_SHIFT 0 /* IM_GP1_EINT */
+#define WM8903_IM_GP1_EINT_WIDTH 1 /* IM_GP1_EINT */
+
+/*
+ * R123 (0x7B) - Interrupt Polarity 1
+ */
+#define WM8903_MICSHRT_INV 0x8000 /* MICSHRT_INV */
+#define WM8903_MICSHRT_INV_MASK 0x8000 /* MICSHRT_INV */
+#define WM8903_MICSHRT_INV_SHIFT 15 /* MICSHRT_INV */
+#define WM8903_MICSHRT_INV_WIDTH 1 /* MICSHRT_INV */
+#define WM8903_MICDET_INV 0x4000 /* MICDET_INV */
+#define WM8903_MICDET_INV_MASK 0x4000 /* MICDET_INV */
+#define WM8903_MICDET_INV_SHIFT 14 /* MICDET_INV */
+#define WM8903_MICDET_INV_WIDTH 1 /* MICDET_INV */
+
+/*
+ * R126 (0x7E) - Interrupt Control
+ */
+#define WM8903_IRQ_POL 0x0001 /* IRQ_POL */
+#define WM8903_IRQ_POL_MASK 0x0001 /* IRQ_POL */
+#define WM8903_IRQ_POL_SHIFT 0 /* IRQ_POL */
+#define WM8903_IRQ_POL_WIDTH 1 /* IRQ_POL */
+
+/*
+ * R129 (0x81) - Control Interface Test 1
+ */
+#define WM8903_USER_KEY 0x0002 /* USER_KEY */
+#define WM8903_USER_KEY_MASK 0x0002 /* USER_KEY */
+#define WM8903_USER_KEY_SHIFT 1 /* USER_KEY */
+#define WM8903_USER_KEY_WIDTH 1 /* USER_KEY */
+#define WM8903_TEST_KEY 0x0001 /* TEST_KEY */
+#define WM8903_TEST_KEY_MASK 0x0001 /* TEST_KEY */
+#define WM8903_TEST_KEY_SHIFT 0 /* TEST_KEY */
+#define WM8903_TEST_KEY_WIDTH 1 /* TEST_KEY */
+
+/*
+ * R149 (0x95) - Charge Pump Test 1
+ */
+#define WM8903_CP_SW_KELVIN_MODE_MASK 0x0006 /* CP_SW_KELVIN_MODE - [2:1] */
+#define WM8903_CP_SW_KELVIN_MODE_SHIFT 1 /* CP_SW_KELVIN_MODE - [2:1] */
+#define WM8903_CP_SW_KELVIN_MODE_WIDTH 2 /* CP_SW_KELVIN_MODE - [2:1] */
+
+/*
+ * R164 (0xA4) - Clock Rate Test 4
+ */
+#define WM8903_ADC_DIG_MIC 0x0200 /* ADC_DIG_MIC */
+#define WM8903_ADC_DIG_MIC_MASK 0x0200 /* ADC_DIG_MIC */
+#define WM8903_ADC_DIG_MIC_SHIFT 9 /* ADC_DIG_MIC */
+#define WM8903_ADC_DIG_MIC_WIDTH 1 /* ADC_DIG_MIC */
+
+/*
+ * R172 (0xAC) - Analogue Output Bias 0
+ */
+#define WM8903_PGA_BIAS_MASK 0x0070 /* PGA_BIAS - [6:4] */
+#define WM8903_PGA_BIAS_SHIFT 4 /* PGA_BIAS - [6:4] */
+#define WM8903_PGA_BIAS_WIDTH 3 /* PGA_BIAS - [6:4] */
+
+#endif
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
new file mode 100644
index 0000000..f41a578
--- /dev/null
+++ b/sound/soc/codecs/wm8971.c
@@ -0,0 +1,941 @@
+/*
+ * wm8971.c -- WM8971 ALSA SoC Audio driver
+ *
+ * Copyright 2005 Lab126, Inc.
+ *
+ * Author: Kenneth Kiraly <kiraly@lab126.com>
+ *
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8971.h"
+
+#define WM8971_VERSION "0.9"
+
+#define WM8971_REG_COUNT 43
+
+static struct workqueue_struct *wm8971_workq = NULL;
+
+/* codec private data */
+struct wm8971_priv {
+ unsigned int sysclk;
+};
+
+/*
+ * wm8971 register cache
+ * We can't read the WM8971 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8971_reg[] = {
+ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */
+ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */
+ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */
+ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */
+ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */
+ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */
+ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */
+ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */
+ 0x0079, 0x0079, 0x0079, /* 40 */
+};
+
+static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg < WM8971_REG_COUNT)
+ return cache[reg];
+
+ return -1;
+}
+
+static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg < WM8971_REG_COUNT)
+ cache[reg] = value;
+}
+
+static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8753 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8971_write_reg_cache (codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0)
+
+/* WM8971 Controls */
+static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" };
+static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz",
+ "200Hz @ 48kHz" };
+static const char *wm8971_treble[] = { "8kHz", "4kHz" };
+static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" };
+static const char *wm8971_ng_type[] = { "Constant PGA Gain",
+ "Mute ADC Output" };
+static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)",
+ "Mono (Right)", "Digital Mono"};
+static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" };
+static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA",
+ "Differential"};
+static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA",
+ "Differential"};
+static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"};
+static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"};
+static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert",
+ "L + R Invert"};
+
+static const struct soc_enum wm8971_enum[] = {
+ SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */
+ SOC_ENUM_SINGLE(WM8971_BASS, 6, 2, wm8971_bass_filter),
+ SOC_ENUM_SINGLE(WM8971_TREBLE, 6, 2, wm8971_treble),
+ SOC_ENUM_SINGLE(WM8971_ALC1, 7, 4, wm8971_alc_func),
+ SOC_ENUM_SINGLE(WM8971_NGATE, 1, 2, wm8971_ng_type), /* 4 */
+ SOC_ENUM_SINGLE(WM8971_ADCDAC, 1, 4, wm8971_deemp),
+ SOC_ENUM_SINGLE(WM8971_ADCTL1, 4, 4, wm8971_mono_mux),
+ SOC_ENUM_SINGLE(WM8971_ADCTL1, 1, 2, wm8971_dac_phase),
+ SOC_ENUM_SINGLE(WM8971_LOUTM1, 0, 5, wm8971_lline_mux), /* 8 */
+ SOC_ENUM_SINGLE(WM8971_ROUTM1, 0, 5, wm8971_rline_mux),
+ SOC_ENUM_SINGLE(WM8971_LADCIN, 6, 4, wm8971_lpga_sel),
+ SOC_ENUM_SINGLE(WM8971_RADCIN, 6, 4, wm8971_rpga_sel),
+ SOC_ENUM_SINGLE(WM8971_ADCDAC, 5, 4, wm8971_adcpol), /* 12 */
+ SOC_ENUM_SINGLE(WM8971_ADCIN, 6, 4, wm8971_mono_mux),
+};
+
+static const struct snd_kcontrol_new wm8971_snd_controls[] = {
+ SOC_DOUBLE_R("Capture Volume", WM8971_LINVOL, WM8971_RINVOL, 0, 63, 0),
+ SOC_DOUBLE_R("Capture ZC Switch", WM8971_LINVOL, WM8971_RINVOL,
+ 6, 1, 0),
+ SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1),
+
+ SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V,
+ WM8971_ROUT1V, 7, 1, 0),
+ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V,
+ WM8971_ROUT2V, 7, 1, 0),
+ SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0),
+
+ SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0),
+
+ SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1,
+ WM8971_LOUTM2, 4, 7, 1),
+ SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1,
+ WM8971_ROUTM2, 4, 7, 1),
+ SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1,
+ WM8971_MOUTM2, 4, 7, 1),
+
+ SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V,
+ WM8971_ROUT1V, 0, 127, 0),
+ SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V,
+ WM8971_ROUT2V, 0, 127, 0),
+
+ SOC_ENUM("Bass Boost", wm8971_enum[0]),
+ SOC_ENUM("Bass Filter", wm8971_enum[1]),
+ SOC_SINGLE("Bass Volume", WM8971_BASS, 0, 7, 1),
+
+ SOC_SINGLE("Treble Volume", WM8971_TREBLE, 0, 7, 0),
+ SOC_ENUM("Treble Cut-off", wm8971_enum[2]),
+
+ SOC_SINGLE("Capture Filter Switch", WM8971_ADCDAC, 0, 1, 1),
+
+ SOC_SINGLE("ALC Target Volume", WM8971_ALC1, 0, 7, 0),
+ SOC_SINGLE("ALC Max Volume", WM8971_ALC1, 4, 7, 0),
+
+ SOC_SINGLE("ALC Capture Target Volume", WM8971_ALC1, 0, 7, 0),
+ SOC_SINGLE("ALC Capture Max Volume", WM8971_ALC1, 4, 7, 0),
+ SOC_ENUM("ALC Capture Function", wm8971_enum[3]),
+ SOC_SINGLE("ALC Capture ZC Switch", WM8971_ALC2, 7, 1, 0),
+ SOC_SINGLE("ALC Capture Hold Time", WM8971_ALC2, 0, 15, 0),
+ SOC_SINGLE("ALC Capture Decay Time", WM8971_ALC3, 4, 15, 0),
+ SOC_SINGLE("ALC Capture Attack Time", WM8971_ALC3, 0, 15, 0),
+ SOC_SINGLE("ALC Capture NG Threshold", WM8971_NGATE, 3, 31, 0),
+ SOC_ENUM("ALC Capture NG Type", wm8971_enum[4]),
+ SOC_SINGLE("ALC Capture NG Switch", WM8971_NGATE, 0, 1, 0),
+
+ SOC_SINGLE("Capture 6dB Attenuate", WM8971_ADCDAC, 8, 1, 0),
+ SOC_SINGLE("Playback 6dB Attenuate", WM8971_ADCDAC, 7, 1, 0),
+
+ SOC_ENUM("Playback De-emphasis", wm8971_enum[5]),
+ SOC_ENUM("Playback Function", wm8971_enum[6]),
+ SOC_ENUM("Playback Phase", wm8971_enum[7]),
+
+ SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
+};
+
+/* add non-DAPM controls */
+static int wm8971_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8971_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8971_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("Playback Switch", WM8971_LOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_LOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8971_LOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8971_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8971_ROUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_ROUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM8971_ROUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_ROUTM2, 7, 1, 0),
+};
+
+/* Mono Mixer */
+static const struct snd_kcontrol_new wm8971_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8971_MOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_MOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8971_MOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_MOUTM2, 7, 1, 0),
+};
+
+/* Left Line Mux */
+static const struct snd_kcontrol_new wm8971_left_line_controls =
+SOC_DAPM_ENUM("Route", wm8971_enum[8]);
+
+/* Right Line Mux */
+static const struct snd_kcontrol_new wm8971_right_line_controls =
+SOC_DAPM_ENUM("Route", wm8971_enum[9]);
+
+/* Left PGA Mux */
+static const struct snd_kcontrol_new wm8971_left_pga_controls =
+SOC_DAPM_ENUM("Route", wm8971_enum[10]);
+
+/* Right PGA Mux */
+static const struct snd_kcontrol_new wm8971_right_pga_controls =
+SOC_DAPM_ENUM("Route", wm8971_enum[11]);
+
+/* Mono ADC Mux */
+static const struct snd_kcontrol_new wm8971_monomux_controls =
+SOC_DAPM_ENUM("Route", wm8971_enum[13]);
+
+static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8971_left_mixer_controls[0],
+ ARRAY_SIZE(wm8971_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8971_right_mixer_controls[0],
+ ARRAY_SIZE(wm8971_right_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0,
+ &wm8971_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8971_mono_mixer_controls)),
+
+ SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Out 1", WM8971_PWR2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 1", WM8971_PWR2, 6, 0, NULL, 0),
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8971_PWR2, 7, 0),
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8971_PWR2, 8, 0),
+ SND_SOC_DAPM_PGA("Mono Out 1", WM8971_PWR2, 2, 0, NULL, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8971_PWR1, 1, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8971_PWR1, 2, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0),
+
+ SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0,
+ &wm8971_left_pga_controls),
+ SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0,
+ &wm8971_right_pga_controls),
+ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8971_left_line_controls),
+ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8971_right_line_controls),
+
+ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8971_monomux_controls),
+ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8971_monomux_controls),
+
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("MONO"),
+
+ SND_SOC_DAPM_INPUT("LINPUT1"),
+ SND_SOC_DAPM_INPUT("RINPUT1"),
+ SND_SOC_DAPM_INPUT("MIC"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* left mixer */
+ {"Left Mixer", "Playback Switch", "Left DAC"},
+ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Left Mixer", "Right Playback Switch", "Right DAC"},
+ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* right mixer */
+ {"Right Mixer", "Left Playback Switch", "Left DAC"},
+ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Right Mixer", "Playback Switch", "Right DAC"},
+ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* left out 1 */
+ {"Left Out 1", NULL, "Left Mixer"},
+ {"LOUT1", NULL, "Left Out 1"},
+
+ /* left out 2 */
+ {"Left Out 2", NULL, "Left Mixer"},
+ {"LOUT2", NULL, "Left Out 2"},
+
+ /* right out 1 */
+ {"Right Out 1", NULL, "Right Mixer"},
+ {"ROUT1", NULL, "Right Out 1"},
+
+ /* right out 2 */
+ {"Right Out 2", NULL, "Right Mixer"},
+ {"ROUT2", NULL, "Right Out 2"},
+
+ /* mono mixer */
+ {"Mono Mixer", "Left Playback Switch", "Left DAC"},
+ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"},
+ {"Mono Mixer", "Right Playback Switch", "Right DAC"},
+ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+ /* mono out */
+ {"Mono Out", NULL, "Mono Mixer"},
+ {"MONO1", NULL, "Mono Out"},
+
+ /* Left Line Mux */
+ {"Left Line Mux", "Line", "LINPUT1"},
+ {"Left Line Mux", "PGA", "Left PGA Mux"},
+ {"Left Line Mux", "Differential", "Differential Mux"},
+
+ /* Right Line Mux */
+ {"Right Line Mux", "Line", "RINPUT1"},
+ {"Right Line Mux", "Mic", "MIC"},
+ {"Right Line Mux", "PGA", "Right PGA Mux"},
+ {"Right Line Mux", "Differential", "Differential Mux"},
+
+ /* Left PGA Mux */
+ {"Left PGA Mux", "Line", "LINPUT1"},
+ {"Left PGA Mux", "Differential", "Differential Mux"},
+
+ /* Right PGA Mux */
+ {"Right PGA Mux", "Line", "RINPUT1"},
+ {"Right PGA Mux", "Differential", "Differential Mux"},
+
+ /* Differential Mux */
+ {"Differential Mux", "Line", "LINPUT1"},
+ {"Differential Mux", "Line", "RINPUT1"},
+
+ /* Left ADC Mux */
+ {"Left ADC Mux", "Stereo", "Left PGA Mux"},
+ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"},
+ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"},
+
+ /* Right ADC Mux */
+ {"Right ADC Mux", "Stereo", "Right PGA Mux"},
+ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"},
+ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"},
+
+ /* ADC */
+ {"Left ADC", NULL, "Left ADC Mux"},
+ {"Right ADC", NULL, "Right ADC Mux"},
+};
+
+static int wm8971_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8971_dapm_widgets,
+ ARRAY_SIZE(wm8971_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+
+ return 0;
+}
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:5;
+ u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 8k */
+ {12288000, 8000, 1536, 0x6, 0x0},
+ {11289600, 8000, 1408, 0x16, 0x0},
+ {18432000, 8000, 2304, 0x7, 0x0},
+ {16934400, 8000, 2112, 0x17, 0x0},
+ {12000000, 8000, 1500, 0x6, 0x1},
+
+ /* 11.025k */
+ {11289600, 11025, 1024, 0x18, 0x0},
+ {16934400, 11025, 1536, 0x19, 0x0},
+ {12000000, 11025, 1088, 0x19, 0x1},
+
+ /* 16k */
+ {12288000, 16000, 768, 0xa, 0x0},
+ {18432000, 16000, 1152, 0xb, 0x0},
+ {12000000, 16000, 750, 0xa, 0x1},
+
+ /* 22.05k */
+ {11289600, 22050, 512, 0x1a, 0x0},
+ {16934400, 22050, 768, 0x1b, 0x0},
+ {12000000, 22050, 544, 0x1b, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0xc, 0x0},
+ {18432000, 32000, 576, 0xd, 0x0},
+ {12000000, 32000, 375, 0xa, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x10, 0x0},
+ {16934400, 44100, 384, 0x11, 0x0},
+ {12000000, 44100, 272, 0x11, 0x1},
+
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0},
+ {18432000, 48000, 384, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0x1e, 0x0},
+ {16934400, 88200, 192, 0x1f, 0x0},
+ {12000000, 88200, 136, 0x1f, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0xe, 0x0},
+ {18432000, 96000, 192, 0xf, 0x0},
+ {12000000, 96000, 125, 0xe, 0x1},
+};
+
+static int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+ return -EINVAL;
+}
+
+static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8971_priv *wm8971 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8971->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8971_write(codec, WM8971_IFACE, iface);
+ return 0;
+}
+
+static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8971_priv *wm8971 = codec->private_data;
+ u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3;
+ u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0;
+ int coeff = get_coeff(wm8971->sysclk, params_rate(params));
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x000c;
+ break;
+ }
+
+ /* set iface & srate */
+ wm8971_write(codec, WM8971_IFACE, iface);
+ if (coeff >= 0)
+ wm8971_write(codec, WM8971_SRATE, srate |
+ (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+ return 0;
+}
+
+static int wm8971_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7;
+
+ if (mute)
+ wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8);
+ else
+ wm8971_write(codec, WM8971_ADCDAC, mute_reg);
+ return 0;
+}
+
+static int wm8971_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* set vmid to 50k and unmute dac */
+ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* mute dac and set vmid to 500k, enable VREF */
+ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140);
+ break;
+ case SND_SOC_BIAS_OFF:
+ wm8971_write(codec, WM8971_PWR1, 0x0001);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8971_dai = {
+ .name = "WM8971",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8971_RATES,
+ .formats = WM8971_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8971_RATES,
+ .formats = WM8971_FORMATS,},
+ .ops = {
+ .hw_params = wm8971_pcm_hw_params,
+ },
+ .dai_ops = {
+ .digital_mute = wm8971_mute,
+ .set_fmt = wm8971_set_dai_fmt,
+ .set_sysclk = wm8971_set_dai_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8971_dai);
+
+static void wm8971_work(struct work_struct *work)
+{
+ struct snd_soc_codec *codec =
+ container_of(work, struct snd_soc_codec, delayed_work.work);
+ wm8971_set_bias_level(codec, codec->bias_level);
+}
+
+static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8971_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+ u16 reg;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) {
+ if (i + 1 == WM8971_RESET)
+ continue;
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* charge wm8971 caps */
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+ reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
+ wm8971_write(codec, WM8971_PWR1, reg | 0x01c0);
+ codec->bias_level = SND_SOC_BIAS_ON;
+ queue_delayed_work(wm8971_workq, &codec->delayed_work,
+ msecs_to_jiffies(1000));
+ }
+
+ return 0;
+}
+
+static int wm8971_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg, ret = 0;
+
+ codec->name = "WM8971";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8971_read_reg_cache;
+ codec->write = wm8971_write;
+ codec->set_bias_level = wm8971_set_bias_level;
+ codec->dai = &wm8971_dai;
+ codec->reg_cache_size = ARRAY_SIZE(wm8971_reg);
+ codec->num_dai = 1;
+ codec->reg_cache = kmemdup(wm8971_reg, sizeof(wm8971_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8971_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8971: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* charge output caps - set vmid to 5k for quick power up */
+ reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e;
+ wm8971_write(codec, WM8971_PWR1, reg | 0x01c0);
+ codec->bias_level = SND_SOC_BIAS_STANDBY;
+ queue_delayed_work(wm8971_workq, &codec->delayed_work,
+ msecs_to_jiffies(1000));
+
+ /* set the update bits */
+ reg = wm8971_read_reg_cache(codec, WM8971_LDAC);
+ wm8971_write(codec, WM8971_LDAC, reg | 0x0100);
+ reg = wm8971_read_reg_cache(codec, WM8971_RDAC);
+ wm8971_write(codec, WM8971_RDAC, reg | 0x0100);
+
+ reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V);
+ wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100);
+ reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V);
+ wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100);
+
+ reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V);
+ wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100);
+ reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V);
+ wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100);
+
+ reg = wm8971_read_reg_cache(codec, WM8971_LINVOL);
+ wm8971_write(codec, WM8971_LINVOL, reg | 0x0100);
+ reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
+ wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
+
+ wm8971_add_controls(codec);
+ wm8971_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8971: failed to register card\n");
+ goto card_err;
+ }
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static struct snd_soc_device *wm8971_socdev;
+
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+
+static int wm8971_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8971_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+
+ codec->control_data = i2c;
+
+ ret = wm8971_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8971\n");
+
+ return ret;
+}
+
+static int wm8971_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8971_i2c_id[] = {
+ { "wm8971", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id);
+
+static struct i2c_driver wm8971_i2c_driver = {
+ .driver = {
+ .name = "WM8971 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8971_i2c_probe,
+ .remove = wm8971_i2c_remove,
+ .id_table = wm8971_i2c_id,
+};
+
+static int wm8971_add_i2c_device(struct platform_device *pdev,
+ const struct wm8971_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8971_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8971", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8971_i2c_driver);
+ return -ENODEV;
+}
+
+#endif
+
+static int wm8971_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8971_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8971_priv *wm8971;
+ int ret = 0;
+
+ pr_info("WM8971 Audio Codec %s", WM8971_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8971 = kzalloc(sizeof(struct wm8971_priv), GFP_KERNEL);
+ if (wm8971 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8971;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8971_socdev = socdev;
+
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8971_work);
+ wm8971_workq = create_workqueue("wm8971");
+ if (wm8971_workq == NULL) {
+ kfree(codec->private_data);
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8971_add_i2c_device(pdev, setup);
+ }
+#endif
+ /* Add other interfaces here */
+
+ if (ret != 0) {
+ destroy_workqueue(wm8971_workq);
+ kfree(codec->private_data);
+ kfree(codec);
+ }
+
+ return ret;
+}
+
+/* power down chip */
+static int wm8971_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ if (wm8971_workq)
+ destroy_workqueue(wm8971_workq);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8971_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8971 = {
+ .probe = wm8971_probe,
+ .remove = wm8971_remove,
+ .suspend = wm8971_suspend,
+ .resume = wm8971_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
+
+MODULE_DESCRIPTION("ASoC WM8971 driver");
+MODULE_AUTHOR("Lab126");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8971.h b/sound/soc/codecs/wm8971.h
new file mode 100644
index 0000000..ef4f08f
--- /dev/null
+++ b/sound/soc/codecs/wm8971.h
@@ -0,0 +1,64 @@
+/*
+ * wm8971.h -- audio driver for WM8971
+ *
+ * Copyright 2005 Lab126, Inc.
+ *
+ * Author: Kenneth Kiraly <kiraly@lab126.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _WM8971_H
+#define _WM8971_H
+
+#define WM8971_LINVOL 0x00
+#define WM8971_RINVOL 0x01
+#define WM8971_LOUT1V 0x02
+#define WM8971_ROUT1V 0x03
+#define WM8971_ADCDAC 0x05
+#define WM8971_IFACE 0x07
+#define WM8971_SRATE 0x08
+#define WM8971_LDAC 0x0a
+#define WM8971_RDAC 0x0b
+#define WM8971_BASS 0x0c
+#define WM8971_TREBLE 0x0d
+#define WM8971_RESET 0x0f
+#define WM8971_ALC1 0x11
+#define WM8971_ALC2 0x12
+#define WM8971_ALC3 0x13
+#define WM8971_NGATE 0x14
+#define WM8971_LADC 0x15
+#define WM8971_RADC 0x16
+#define WM8971_ADCTL1 0x17
+#define WM8971_ADCTL2 0x18
+#define WM8971_PWR1 0x19
+#define WM8971_PWR2 0x1a
+#define WM8971_ADCTL3 0x1b
+#define WM8971_ADCIN 0x1f
+#define WM8971_LADCIN 0x20
+#define WM8971_RADCIN 0x21
+#define WM8971_LOUTM1 0x22
+#define WM8971_LOUTM2 0x23
+#define WM8971_ROUTM1 0x24
+#define WM8971_ROUTM2 0x25
+#define WM8971_MOUTM1 0x26
+#define WM8971_MOUTM2 0x27
+#define WM8971_LOUT2V 0x28
+#define WM8971_ROUT2V 0x29
+#define WM8971_MOUTV 0x2A
+
+#define WM8971_SYSCLK 0
+
+struct wm8971_setup_data {
+ int i2c_bus;
+ unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8971_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8971;
+
+#endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
new file mode 100644
index 0000000..572d22b
--- /dev/null
+++ b/sound/soc/codecs/wm8990.c
@@ -0,0 +1,1635 @@
+/*
+ * wm8990.c -- WM8990 ALSA Soc Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include "wm8990.h"
+
+#define WM8990_VERSION "0.2"
+
+/* codec private data */
+struct wm8990_priv {
+ unsigned int sysclk;
+ unsigned int pcmclk;
+};
+
+/*
+ * wm8990 register cache. Note that register 0 is not included in the
+ * cache.
+ */
+static const u16 wm8990_reg[] = {
+ 0x8990, /* R0 - Reset */
+ 0x0000, /* R1 - Power Management (1) */
+ 0x6000, /* R2 - Power Management (2) */
+ 0x0000, /* R3 - Power Management (3) */
+ 0x4050, /* R4 - Audio Interface (1) */
+ 0x4000, /* R5 - Audio Interface (2) */
+ 0x01C8, /* R6 - Clocking (1) */
+ 0x0000, /* R7 - Clocking (2) */
+ 0x0040, /* R8 - Audio Interface (3) */
+ 0x0040, /* R9 - Audio Interface (4) */
+ 0x0004, /* R10 - DAC CTRL */
+ 0x00C0, /* R11 - Left DAC Digital Volume */
+ 0x00C0, /* R12 - Right DAC Digital Volume */
+ 0x0000, /* R13 - Digital Side Tone */
+ 0x0100, /* R14 - ADC CTRL */
+ 0x00C0, /* R15 - Left ADC Digital Volume */
+ 0x00C0, /* R16 - Right ADC Digital Volume */
+ 0x0000, /* R17 */
+ 0x0000, /* R18 - GPIO CTRL 1 */
+ 0x1000, /* R19 - GPIO1 & GPIO2 */
+ 0x1010, /* R20 - GPIO3 & GPIO4 */
+ 0x1010, /* R21 - GPIO5 & GPIO6 */
+ 0x8000, /* R22 - GPIOCTRL 2 */
+ 0x0800, /* R23 - GPIO_POL */
+ 0x008B, /* R24 - Left Line Input 1&2 Volume */
+ 0x008B, /* R25 - Left Line Input 3&4 Volume */
+ 0x008B, /* R26 - Right Line Input 1&2 Volume */
+ 0x008B, /* R27 - Right Line Input 3&4 Volume */
+ 0x0000, /* R28 - Left Output Volume */
+ 0x0000, /* R29 - Right Output Volume */
+ 0x0066, /* R30 - Line Outputs Volume */
+ 0x0022, /* R31 - Out3/4 Volume */
+ 0x0079, /* R32 - Left OPGA Volume */
+ 0x0079, /* R33 - Right OPGA Volume */
+ 0x0003, /* R34 - Speaker Volume */
+ 0x0003, /* R35 - ClassD1 */
+ 0x0000, /* R36 */
+ 0x0100, /* R37 - ClassD3 */
+ 0x0079, /* R38 - ClassD4 */
+ 0x0000, /* R39 - Input Mixer1 */
+ 0x0000, /* R40 - Input Mixer2 */
+ 0x0000, /* R41 - Input Mixer3 */
+ 0x0000, /* R42 - Input Mixer4 */
+ 0x0000, /* R43 - Input Mixer5 */
+ 0x0000, /* R44 - Input Mixer6 */
+ 0x0000, /* R45 - Output Mixer1 */
+ 0x0000, /* R46 - Output Mixer2 */
+ 0x0000, /* R47 - Output Mixer3 */
+ 0x0000, /* R48 - Output Mixer4 */
+ 0x0000, /* R49 - Output Mixer5 */
+ 0x0000, /* R50 - Output Mixer6 */
+ 0x0180, /* R51 - Out3/4 Mixer */
+ 0x0000, /* R52 - Line Mixer1 */
+ 0x0000, /* R53 - Line Mixer2 */
+ 0x0000, /* R54 - Speaker Mixer */
+ 0x0000, /* R55 - Additional Control */
+ 0x0000, /* R56 - AntiPOP1 */
+ 0x0000, /* R57 - AntiPOP2 */
+ 0x0000, /* R58 - MICBIAS */
+ 0x0000, /* R59 */
+ 0x0008, /* R60 - PLL1 */
+ 0x0031, /* R61 - PLL2 */
+ 0x0026, /* R62 - PLL3 */
+};
+
+/*
+ * read wm8990 register cache
+ */
+static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+ return cache[reg];
+}
+
+/*
+ * write wm8990 register cache
+ */
+static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+
+ /* Reset register is uncached */
+ if (reg == 0)
+ return;
+
+ cache[reg] = value;
+}
+
+/*
+ * write to the wm8990 register space
+ */
+static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[3];
+
+ data[0] = reg & 0xFF;
+ data[1] = (value >> 8) & 0xFF;
+ data[2] = value & 0xFF;
+
+ wm8990_write_reg_cache(codec, reg, value);
+
+ if (codec->hw_write(codec->control_data, data, 3) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0)
+
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100);
+
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+
+static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int ret;
+ u16 val;
+
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+ if (ret < 0)
+ return ret;
+
+ /* now hit the volume update bits (always bit 8) */
+ val = wm8990_read_reg_cache(codec, reg);
+ return wm8990_write(codec, reg, val | 0x0100);
+}
+
+#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\
+ tlv_array) {\
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \
+ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+static const char *wm8990_digital_sidetone[] =
+ {"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8990_left_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACL_SHIFT,
+ WM8990_ADC_TO_DACL_MASK,
+ wm8990_digital_sidetone);
+
+static const struct soc_enum wm8990_right_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACR_SHIFT,
+ WM8990_ADC_TO_DACR_MASK,
+ wm8990_digital_sidetone);
+
+static const char *wm8990_adcmode[] =
+ {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8990_right_adcmode_enum =
+SOC_ENUM_SINGLE(WM8990_ADC_CTRL,
+ WM8990_ADC_HPF_CUT_SHIFT,
+ WM8990_ADC_HPF_CUT_MASK,
+ wm8990_adcmode);
+
+static const struct snd_kcontrol_new wm8990_snd_controls[] = {
+/* INMIXL */
+SOC_SINGLE("LIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L12MNBST_BIT, 1, 0),
+SOC_SINGLE("LIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L34MNBST_BIT, 1, 0),
+/* INMIXR */
+SOC_SINGLE("RIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R12MNBST_BIT, 1, 0),
+SOC_SINGLE("RIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R34MNBST_BIT, 1, 0),
+
+/* LOMIX */
+SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER3,
+ WM8990_LLI3LOVOL_SHIFT, WM8990_LLI3LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3,
+ WM8990_LR12LOVOL_SHIFT, WM8990_LR12LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3,
+ WM8990_LL12LOVOL_SHIFT, WM8990_LL12LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER5,
+ WM8990_LRI3LOVOL_SHIFT, WM8990_LRI3LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER5,
+ WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER5,
+ WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv),
+
+/* ROMIX */
+SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER4,
+ WM8990_RRI3ROVOL_SHIFT, WM8990_RRI3ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4,
+ WM8990_RL12ROVOL_SHIFT, WM8990_RL12ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4,
+ WM8990_RR12ROVOL_SHIFT, WM8990_RR12ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER6,
+ WM8990_RLI3ROVOL_SHIFT, WM8990_RLI3ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER6,
+ WM8990_RLBROVOL_SHIFT, WM8990_RLBROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER6,
+ WM8990_RRBROVOL_SHIFT, WM8990_RRBROVOL_MASK, 1, out_mix_tlv),
+
+/* LOUT */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8990_LEFT_OUTPUT_VOLUME,
+ WM8990_LOUTVOL_SHIFT, WM8990_LOUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOUT ZC", WM8990_LEFT_OUTPUT_VOLUME, WM8990_LOZC_BIT, 1, 0),
+
+/* ROUT */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8990_RIGHT_OUTPUT_VOLUME,
+ WM8990_ROUTVOL_SHIFT, WM8990_ROUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROUT ZC", WM8990_RIGHT_OUTPUT_VOLUME, WM8990_ROZC_BIT, 1, 0),
+
+/* LOPGA */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8990_LEFT_OPGA_VOLUME,
+ WM8990_LOPGAVOL_SHIFT, WM8990_LOPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOPGA ZC Switch", WM8990_LEFT_OPGA_VOLUME,
+ WM8990_LOPGAZC_BIT, 1, 0),
+
+/* ROPGA */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8990_RIGHT_OPGA_VOLUME,
+ WM8990_ROPGAVOL_SHIFT, WM8990_ROPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROPGA ZC Switch", WM8990_RIGHT_OPGA_VOLUME,
+ WM8990_ROPGAZC_BIT, 1, 0),
+
+SOC_SINGLE("LON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_LONMUTE_BIT, 1, 0),
+SOC_SINGLE("LOP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_LOPMUTE_BIT, 1, 0),
+SOC_SINGLE("LOP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_LOATTN_BIT, 1, 0),
+SOC_SINGLE("RON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_RONMUTE_BIT, 1, 0),
+SOC_SINGLE("ROP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_ROPMUTE_BIT, 1, 0),
+SOC_SINGLE("ROP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME,
+ WM8990_ROATTN_BIT, 1, 0),
+
+SOC_SINGLE("OUT3 Mute Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT3MUTE_BIT, 1, 0),
+SOC_SINGLE("OUT3 Attenuation Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT3ATTN_BIT, 1, 0),
+
+SOC_SINGLE("OUT4 Mute Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT4MUTE_BIT, 1, 0),
+SOC_SINGLE("OUT4 Attenuation Switch", WM8990_OUT3_4_VOLUME,
+ WM8990_OUT4ATTN_BIT, 1, 0),
+
+SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1,
+ WM8990_CDMODE_BIT, 1, 0),
+
+SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME,
+ WM8990_SPKATTN_SHIFT, WM8990_SPKATTN_MASK, 0),
+SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3,
+ WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0),
+SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3,
+ WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0),
+SOC_SINGLE_TLV("Speaker Volume", WM8990_CLASSD4,
+ WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("Speaker ZC Switch", WM8990_CLASSD4,
+ WM8990_SPKZC_SHIFT, WM8990_SPKZC_MASK, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+ WM8990_LEFT_DAC_DIGITAL_VOLUME,
+ WM8990_DACL_VOL_SHIFT,
+ WM8990_DACL_VOL_MASK,
+ 0,
+ out_dac_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+ WM8990_RIGHT_DAC_DIGITAL_VOLUME,
+ WM8990_DACR_VOL_SHIFT,
+ WM8990_DACR_VOL_MASK,
+ 0,
+ out_dac_tlv),
+
+SOC_ENUM("Left Digital Sidetone", wm8990_left_digital_sidetone_enum),
+SOC_ENUM("Right Digital Sidetone", wm8990_right_digital_sidetone_enum),
+
+SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADCL_DAC_SVOL_SHIFT, WM8990_ADCL_DAC_SVOL_MASK, 0,
+ out_sidetone_tlv),
+SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADCR_DAC_SVOL_SHIFT, WM8990_ADCR_DAC_SVOL_MASK, 0,
+ out_sidetone_tlv),
+
+SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8990_ADC_CTRL,
+ WM8990_ADC_HPF_ENA_BIT, 1, 0),
+
+SOC_ENUM("ADC HPF Mode", wm8990_right_adcmode_enum),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+ WM8990_LEFT_ADC_DIGITAL_VOLUME,
+ WM8990_ADCL_VOL_SHIFT,
+ WM8990_ADCL_VOL_MASK,
+ 0,
+ in_adc_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+ WM8990_RIGHT_ADC_DIGITAL_VOLUME,
+ WM8990_ADCR_VOL_SHIFT,
+ WM8990_ADCR_VOL_MASK,
+ 0,
+ in_adc_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+ WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8990_LIN12VOL_SHIFT,
+ WM8990_LIN12VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("LIN12 ZC Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8990_LI12ZC_BIT, 1, 0),
+
+SOC_SINGLE("LIN12 Mute Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+ WM8990_LI12MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+ WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8990_LIN34VOL_SHIFT,
+ WM8990_LIN34VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("LIN34 ZC Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8990_LI34ZC_BIT, 1, 0),
+
+SOC_SINGLE("LIN34 Mute Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+ WM8990_LI34MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+ WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8990_RIN12VOL_SHIFT,
+ WM8990_RIN12VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("RIN12 ZC Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8990_RI12ZC_BIT, 1, 0),
+
+SOC_SINGLE("RIN12 Mute Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+ WM8990_RI12MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+ WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8990_RIN34VOL_SHIFT,
+ WM8990_RIN34VOL_MASK,
+ 0,
+ in_pga_tlv),
+
+SOC_SINGLE("RIN34 ZC Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8990_RI34ZC_BIT, 1, 0),
+
+SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+ WM8990_RI34MUTE_BIT, 1, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8990_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8990_snd_controls[i], codec,
+ NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+static int inmixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ u16 reg, fakepower;
+
+ reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2);
+ fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS);
+
+ if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) |
+ (1 << WM8990_AINLMUX_PWR_BIT))) {
+ reg |= WM8990_AINL_ENA;
+ } else {
+ reg &= ~WM8990_AINL_ENA;
+ }
+
+ if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) |
+ (1 << WM8990_AINRMUX_PWR_BIT))) {
+ reg |= WM8990_AINR_ENA;
+ } else {
+ reg &= ~WM8990_AINL_ENA;
+ }
+ wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg);
+
+ return 0;
+}
+
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ u32 reg_shift = kcontrol->private_value & 0xfff;
+ int ret = 0;
+ u16 reg;
+
+ switch (reg_shift) {
+ case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) :
+ reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1);
+ if (reg & WM8990_LDLO) {
+ printk(KERN_WARNING
+ "Cannot set as Output Mixer 1 LDLO Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8):
+ reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2);
+ if (reg & WM8990_RDRO) {
+ printk(KERN_WARNING
+ "Cannot set as Output Mixer 2 RDRO Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8):
+ reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+ if (reg & WM8990_LDSPK) {
+ printk(KERN_WARNING
+ "Cannot set as Speaker Mixer LDSPK Set\n");
+ ret = -1;
+ }
+ break;
+ case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8):
+ reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+ if (reg & WM8990_RDSPK) {
+ printk(KERN_WARNING
+ "Cannot set as Speaker Mixer RDSPK Set\n");
+ ret = -1;
+ }
+ break;
+ }
+
+ return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+ TLV_DB_RANGE_HEAD(1),
+ 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8990_dapm_lin12_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN1 Switch", WM8990_INPUT_MIXER2, WM8990_LMN1_BIT, 1, 0),
+SOC_DAPM_SINGLE("LIN2 Switch", WM8990_INPUT_MIXER2, WM8990_LMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8990_dapm_lin34_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN3 Switch", WM8990_INPUT_MIXER2, WM8990_LMN3_BIT, 1, 0),
+SOC_DAPM_SINGLE("LIN4 Switch", WM8990_INPUT_MIXER2, WM8990_LMP4_BIT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8990_dapm_rin12_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN1 Switch", WM8990_INPUT_MIXER2, WM8990_RMN1_BIT, 1, 0),
+SOC_DAPM_SINGLE("RIN2 Switch", WM8990_INPUT_MIXER2, WM8990_RMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8990_dapm_rin34_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN3 Switch", WM8990_INPUT_MIXER2, WM8990_RMN3_BIT, 1, 0),
+SOC_DAPM_SINGLE("RIN4 Switch", WM8990_INPUT_MIXER2, WM8990_RMP4_BIT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8990_dapm_inmixl_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8990_INPUT_MIXER3,
+ WM8990_LDBVOL_SHIFT, WM8990_LDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8990_INPUT_MIXER5, WM8990_LI2BVOL_SHIFT,
+ 7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("LINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT,
+ 1, 0),
+SOC_DAPM_SINGLE("LINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT,
+ 1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8990_dapm_inmixr_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8990_INPUT_MIXER4,
+ WM8990_RDBVOL_SHIFT, WM8990_RDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8990_INPUT_MIXER6, WM8990_RI2BVOL_SHIFT,
+ 7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("RINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT,
+ 1, 0),
+SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT,
+ 1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8990_ainlmux[] =
+ {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8990_ainlmux_enum =
+SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT,
+ ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux);
+
+static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls =
+SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8990_ainrmux[] =
+ {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8990_ainrmux_enum =
+SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT,
+ ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux);
+
+static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls =
+SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8990_dapm_rxvoice_controls[] = {
+SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8990_INPUT_MIXER5, WM8990_LR4BVOL_SHIFT,
+ WM8990_LR4BVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8990_INPUT_MIXER6, WM8990_RL4BVOL_SHIFT,
+ WM8990_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lomix_controls[] = {
+SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LRBLO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LLBLO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LRI3LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LLI3LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LR12LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LL12LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8990_OUTPUT_MIXER1,
+ WM8990_LDLO_BIT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8990_dapm_romix_controls[] = {
+SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RLBRO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RRBRO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RLI3RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RRI3RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RL12RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RR12RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8990_OUTPUT_MIXER2,
+ WM8990_RDRO_BIT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lonmix_controls[] = {
+SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1,
+ WM8990_LLOPGALON_BIT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER1,
+ WM8990_LROPGALON_BIT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8990_LINE_MIXER1,
+ WM8990_LOPLON_BIT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lopmix_controls[] = {
+SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER1,
+ WM8990_LR12LOP_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER1,
+ WM8990_LL12LOP_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1,
+ WM8990_LLOPGALOP_BIT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8990_dapm_ronmix_controls[] = {
+SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2,
+ WM8990_RROPGARON_BIT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER2,
+ WM8990_RLOPGARON_BIT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8990_LINE_MIXER2,
+ WM8990_ROPRON_BIT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8990_dapm_ropmix_controls[] = {
+SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER2,
+ WM8990_RL12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER2,
+ WM8990_RR12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2,
+ WM8990_RROPGAROP_BIT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8990_dapm_out3mix_controls[] = {
+SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER,
+ WM8990_LI4O3_BIT, 1, 0),
+SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8990_OUT3_4_MIXER,
+ WM8990_LPGAO3_BIT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8990_dapm_out4mix_controls[] = {
+SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8990_OUT3_4_MIXER,
+ WM8990_RPGAO4_BIT, 1, 0),
+SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER,
+ WM8990_RI4O4_BIT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8990_dapm_spkmix_controls[] = {
+SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LI2SPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LB2SPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LOPGASPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8990_SPEAKER_MIXER,
+ WM8990_LDSPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8990_SPEAKER_MIXER,
+ WM8990_RDSPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8990_SPEAKER_MIXER,
+ WM8990_ROPGASPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_RL12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8990_SPEAKER_MIXER,
+ WM8990_RI2SPK_BIT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8990_dapm_widgets[] = {
+/* Input Side */
+/* Input Lines */
+SND_SOC_DAPM_INPUT("LIN1"),
+SND_SOC_DAPM_INPUT("LIN2"),
+SND_SOC_DAPM_INPUT("LIN3"),
+SND_SOC_DAPM_INPUT("LIN4/RXN"),
+SND_SOC_DAPM_INPUT("RIN3"),
+SND_SOC_DAPM_INPUT("RIN4/RXP"),
+SND_SOC_DAPM_INPUT("RIN1"),
+SND_SOC_DAPM_INPUT("RIN2"),
+SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+/* DACs */
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2,
+ WM8990_ADCL_ENA_BIT, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8990_POWER_MANAGEMENT_2,
+ WM8990_ADCR_ENA_BIT, 0),
+
+/* Input PGAs */
+SND_SOC_DAPM_MIXER("LIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN12_ENA_BIT,
+ 0, &wm8990_dapm_lin12_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lin12_pga_controls)),
+SND_SOC_DAPM_MIXER("LIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN34_ENA_BIT,
+ 0, &wm8990_dapm_lin34_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lin34_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN12_ENA_BIT,
+ 0, &wm8990_dapm_rin12_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_rin12_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT,
+ 0, &wm8990_dapm_rin34_pga_controls[0],
+ ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)),
+
+/* INMIXL */
+SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
+ &wm8990_dapm_inmixl_controls[0],
+ ARRAY_SIZE(wm8990_dapm_inmixl_controls),
+ inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINLMUX */
+SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+ &wm8990_dapm_ainlmux_controls, inmixer_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* INMIXR */
+SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
+ &wm8990_dapm_inmixr_controls[0],
+ ARRAY_SIZE(wm8990_dapm_inmixr_controls),
+ inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINRMUX */
+SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+ &wm8990_dapm_ainrmux_controls, inmixer_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* Output Side */
+/* DACs */
+SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8990_POWER_MANAGEMENT_3,
+ WM8990_DACL_ENA_BIT, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8990_POWER_MANAGEMENT_3,
+ WM8990_DACR_ENA_BIT, 0),
+
+/* LOMIX */
+SND_SOC_DAPM_MIXER_E("LOMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOMIX_ENA_BIT,
+ 0, &wm8990_dapm_lomix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lomix_controls),
+ outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LONMIX */
+SND_SOC_DAPM_MIXER("LONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LON_ENA_BIT, 0,
+ &wm8990_dapm_lonmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lonmix_controls)),
+
+/* LOPMIX */
+SND_SOC_DAPM_MIXER("LOPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOP_ENA_BIT, 0,
+ &wm8990_dapm_lopmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_lopmix_controls)),
+
+/* OUT3MIX */
+SND_SOC_DAPM_MIXER("OUT3MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT3_ENA_BIT, 0,
+ &wm8990_dapm_out3mix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_out3mix_controls)),
+
+/* SPKMIX */
+SND_SOC_DAPM_MIXER_E("SPKMIX", WM8990_POWER_MANAGEMENT_1, WM8990_SPK_ENA_BIT, 0,
+ &wm8990_dapm_spkmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_spkmix_controls), outmixer_event,
+ SND_SOC_DAPM_PRE_REG),
+
+/* OUT4MIX */
+SND_SOC_DAPM_MIXER("OUT4MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT4_ENA_BIT, 0,
+ &wm8990_dapm_out4mix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_out4mix_controls)),
+
+/* ROPMIX */
+SND_SOC_DAPM_MIXER("ROPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROP_ENA_BIT, 0,
+ &wm8990_dapm_ropmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_ropmix_controls)),
+
+/* RONMIX */
+SND_SOC_DAPM_MIXER("RONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_RON_ENA_BIT, 0,
+ &wm8990_dapm_ronmix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_ronmix_controls)),
+
+/* ROMIX */
+SND_SOC_DAPM_MIXER_E("ROMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROMIX_ENA_BIT,
+ 0, &wm8990_dapm_romix_controls[0],
+ ARRAY_SIZE(wm8990_dapm_romix_controls),
+ outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LOUT PGA */
+SND_SOC_DAPM_PGA("LOUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_LOUT_ENA_BIT, 0,
+ NULL, 0),
+
+/* ROUT PGA */
+SND_SOC_DAPM_PGA("ROUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_ROUT_ENA_BIT, 0,
+ NULL, 0),
+
+/* LOPGA */
+SND_SOC_DAPM_PGA("LOPGA", WM8990_POWER_MANAGEMENT_3, WM8990_LOPGA_ENA_BIT, 0,
+ NULL, 0),
+
+/* ROPGA */
+SND_SOC_DAPM_PGA("ROPGA", WM8990_POWER_MANAGEMENT_3, WM8990_ROPGA_ENA_BIT, 0,
+ NULL, 0),
+
+/* MICBIAS */
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8990_POWER_MANAGEMENT_1,
+ WM8990_MICBIAS_ENA_BIT, 0),
+
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Make DACs turn on when playing even if not mixed into any outputs */
+ {"Internal DAC Sink", NULL, "Left DAC"},
+ {"Internal DAC Sink", NULL, "Right DAC"},
+
+ /* Make ADCs turn on when recording even if not mixed from any inputs */
+ {"Left ADC", NULL, "Internal ADC Source"},
+ {"Right ADC", NULL, "Internal ADC Source"},
+
+ /* Input Side */
+ /* LIN12 PGA */
+ {"LIN12 PGA", "LIN1 Switch", "LIN1"},
+ {"LIN12 PGA", "LIN2 Switch", "LIN2"},
+ /* LIN34 PGA */
+ {"LIN34 PGA", "LIN3 Switch", "LIN3"},
+ {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+ /* INMIXL */
+ {"INMIXL", "Record Left Volume", "LOMIX"},
+ {"INMIXL", "LIN2 Volume", "LIN2"},
+ {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+ {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+ /* AILNMUX */
+ {"AILNMUX", "INMIXL Mix", "INMIXL"},
+ {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
+ {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
+ {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
+ {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* ADC */
+ {"Left ADC", NULL, "AILNMUX"},
+
+ /* RIN12 PGA */
+ {"RIN12 PGA", "RIN1 Switch", "RIN1"},
+ {"RIN12 PGA", "RIN2 Switch", "RIN2"},
+ /* RIN34 PGA */
+ {"RIN34 PGA", "RIN3 Switch", "RIN3"},
+ {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+ /* INMIXL */
+ {"INMIXR", "Record Right Volume", "ROMIX"},
+ {"INMIXR", "RIN2 Volume", "RIN2"},
+ {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+ {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+ /* AIRNMUX */
+ {"AIRNMUX", "INMIXR Mix", "INMIXR"},
+ {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
+ {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
+ {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
+ {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+ /* ADC */
+ {"Right ADC", NULL, "AIRNMUX"},
+
+ /* LOMIX */
+ {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+ {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+ {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+ {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+ {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"},
+ {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"},
+ {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+ /* ROMIX */
+ {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+ {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+ {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+ {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+ {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"},
+ {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"},
+ {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+ /* SPKMIX */
+ {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+ {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+ {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"},
+ {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"},
+ {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+ {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+ {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+ {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"},
+
+ /* LONMIX */
+ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+ {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+ {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+ /* LOPMIX */
+ {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+ {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+ {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+ /* OUT3MIX */
+ {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+ {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+ /* OUT4MIX */
+ {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+ {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"},
+
+ /* RONMIX */
+ {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+ {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+ {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+ /* ROPMIX */
+ {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+ {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+ {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+ /* Out Mixer PGAs */
+ {"LOPGA", NULL, "LOMIX"},
+ {"ROPGA", NULL, "ROMIX"},
+
+ {"LOUT PGA", NULL, "LOMIX"},
+ {"ROUT PGA", NULL, "ROMIX"},
+
+ /* Output Pins */
+ {"LON", NULL, "LONMIX"},
+ {"LOP", NULL, "LOPMIX"},
+ {"OUT", NULL, "OUT3MIX"},
+ {"LOUT", NULL, "LOUT PGA"},
+ {"SPKN", NULL, "SPKMIX"},
+ {"ROUT", NULL, "ROUT PGA"},
+ {"OUT4", NULL, "OUT4MIX"},
+ {"ROP", NULL, "ROPMIX"},
+ {"RON", NULL, "RONMIX"},
+};
+
+static int wm8990_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets,
+ ARRAY_SIZE(wm8990_dapm_widgets));
+
+ /* set up the WM8990 audio map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 div2;
+ u32 n;
+ u32 k;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 16) * 10)
+
+static void pll_factors(struct _pll_div *pll_div, unsigned int target,
+ unsigned int source)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div->div2 = 1;
+ Ndiv = target / source;
+ } else
+ pll_div->div2 = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8990 N value outwith recommended range! N = %d\n", Ndiv);
+
+ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div->k = K;
+}
+
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ u16 reg;
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct _pll_div pll_div;
+
+ if (freq_in && freq_out) {
+ pll_factors(&pll_div, freq_out * 4, freq_in);
+
+ /* Turn on PLL */
+ reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ reg |= WM8990_PLL_ENA;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+
+ /* sysclk comes from PLL */
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2);
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC);
+
+ /* set up N , fractional mode and pre-divisor if neccessary */
+ wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM |
+ (pll_div.div2?WM8990_PRESCALE:0));
+ wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8));
+ wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF));
+ } else {
+ /* Turn on PLL */
+ reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ reg &= ~WM8990_PLL_ENA;
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+ }
+ return 0;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8990_priv *wm8990 = codec->private_data;
+
+ wm8990->sysclk = freq;
+ return 0;
+}
+
+/*
+ * Set's ADC and Voice DAC format.
+ */
+static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 audio1, audio3;
+
+ audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
+ audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ audio3 &= ~WM8990_AIF_MSTR1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ audio3 |= WM8990_AIF_MSTR1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ audio1 &= ~WM8990_AIF_FMT_MASK;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ audio1 |= WM8990_AIF_TMF_I2S;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ audio1 |= WM8990_AIF_TMF_RIGHTJ;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ audio1 |= WM8990_AIF_TMF_LEFTJ;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ audio1 |= WM8990_AIF_TMF_DSP;
+ audio1 &= ~WM8990_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ audio1 |= WM8990_AIF_TMF_DSP | WM8990_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3);
+ return 0;
+}
+
+static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8990_MCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ ~WM8990_MCLK_DIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ break;
+ case WM8990_DACCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ ~WM8990_DAC_CLKDIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ break;
+ case WM8990_ADCCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+ ~WM8990_ADC_CLKDIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+ break;
+ case WM8990_BCLK_DIV:
+ reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) &
+ ~WM8990_BCLK_DIV_MASK;
+ wm8990_write(codec, WM8990_CLOCKING_1, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8990_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
+
+ audio1 &= ~WM8990_AIF_WL_MASK;
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ audio1 |= WM8990_AIF_WL_20BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ audio1 |= WM8990_AIF_WL_24BITS;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ audio1 |= WM8990_AIF_WL_32BITS;
+ break;
+ }
+
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+ return 0;
+}
+
+static int wm8990_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 val;
+
+ val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE;
+
+ if (mute)
+ wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+ else
+ wm8990_write(codec, WM8990_DAC_CTRL, val);
+
+ return 0;
+}
+
+static int wm8990_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 val;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Enable all output discharge bits */
+ wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+ WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
+ WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
+ WM8990_DIS_ROUT);
+
+ /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL |
+ WM8990_VMIDTOG);
+
+ /* Delay to allow output caps to discharge */
+ msleep(msecs_to_jiffies(300));
+
+ /* Disable VMIDTOG */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL);
+
+ /* disable all output discharge bits */
+ wm8990_write(codec, WM8990_ANTIPOP1, 0);
+
+ /* Enable outputs */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00);
+
+ msleep(msecs_to_jiffies(50));
+
+ /* Enable VMID at 2x50k */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02);
+
+ msleep(msecs_to_jiffies(100));
+
+ /* Enable VREF */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+
+ msleep(msecs_to_jiffies(600));
+
+ /* Enable BUFIOEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL |
+ WM8990_BUFIOEN);
+
+ /* Disable outputs */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3);
+
+ /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
+ } else {
+ /* ON -> standby */
+
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* Enable POBCTRL and SOFT_ST */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_POBCTRL | WM8990_BUFIOEN);
+
+ /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+ WM8990_BUFDCOPEN | WM8990_POBCTRL |
+ WM8990_BUFIOEN);
+
+ /* mute DAC */
+ val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL);
+ wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+
+ /* Enable any disabled outputs */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+
+ /* Disable VMID */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01);
+
+ msleep(msecs_to_jiffies(300));
+
+ /* Enable all output discharge bits */
+ wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+ WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
+ WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
+ WM8990_DIS_ROUT);
+
+ /* Disable VREF */
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0);
+
+ /* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+ wm8990_write(codec, WM8990_ANTIPOP2, 0x0);
+ break;
+ }
+
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8990_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define WM8990_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+/*
+ * The WM8990 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+struct snd_soc_dai wm8990_dai = {
+/* ADC/DAC on primary */
+ .name = "WM8990 ADC/DAC Primary",
+ .id = 1,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8990_RATES,
+ .formats = WM8990_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8990_RATES,
+ .formats = WM8990_FORMATS,},
+ .ops = {
+ .hw_params = wm8990_hw_params,},
+ .dai_ops = {
+ .digital_mute = wm8990_mute,
+ .set_fmt = wm8990_set_dai_fmt,
+ .set_clkdiv = wm8990_set_dai_clkdiv,
+ .set_pll = wm8990_set_dai_pll,
+ .set_sysclk = wm8990_set_dai_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(wm8990_dai);
+
+static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ /* we only need to suspend if we are a valid card */
+ if (!codec->card)
+ return 0;
+
+ wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8990_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* we only need to resume if we are a valid card */
+ if (!codec->card)
+ return 0;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) {
+ if (i + 1 == WM8990_RESET)
+ continue;
+ data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+/*
+ * initialise the WM8990 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8990_init(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 reg;
+ int ret = 0;
+
+ codec->name = "WM8990";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8990_read_reg_cache;
+ codec->write = wm8990_write;
+ codec->set_bias_level = wm8990_set_bias_level;
+ codec->dai = &wm8990_dai;
+ codec->num_dai = 2;
+ codec->reg_cache_size = ARRAY_SIZE(wm8990_reg);
+ codec->reg_cache = kmemdup(wm8990_reg, sizeof(wm8990_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+
+ wm8990_reset(codec);
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8990: failed to create pcms\n");
+ goto pcm_err;
+ }
+
+ /* charge output caps */
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4);
+ wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1);
+
+ reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) &
+ ~WM8990_GPIO1_SEL_MASK;
+ wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1);
+
+ reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+ wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA);
+
+ wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+ wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+ wm8990_add_controls(codec);
+ wm8990_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm8990: failed to register card\n");
+ goto card_err;
+ }
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ kfree(codec->reg_cache);
+ return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+ around */
+static struct snd_soc_device *wm8990_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM891 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x34
+ * high = 0x36
+ */
+
+static int wm8990_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct snd_soc_device *socdev = wm8990_socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret;
+
+ i2c_set_clientdata(i2c, codec);
+ codec->control_data = i2c;
+
+ ret = wm8990_init(socdev);
+ if (ret < 0)
+ pr_err("failed to initialise WM8990\n");
+
+ return ret;
+}
+
+static int wm8990_i2c_remove(struct i2c_client *client)
+{
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
+ kfree(codec->reg_cache);
+ return 0;
+}
+
+static const struct i2c_device_id wm8990_i2c_id[] = {
+ { "wm8990", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id);
+
+static struct i2c_driver wm8990_i2c_driver = {
+ .driver = {
+ .name = "WM8990 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8990_i2c_probe,
+ .remove = wm8990_i2c_remove,
+ .id_table = wm8990_i2c_id,
+};
+
+static int wm8990_add_i2c_device(struct platform_device *pdev,
+ const struct wm8990_setup_data *setup)
+{
+ struct i2c_board_info info;
+ struct i2c_adapter *adapter;
+ struct i2c_client *client;
+ int ret;
+
+ ret = i2c_add_driver(&wm8990_i2c_driver);
+ if (ret != 0) {
+ dev_err(&pdev->dev, "can't add i2c driver\n");
+ return ret;
+ }
+
+ memset(&info, 0, sizeof(struct i2c_board_info));
+ info.addr = setup->i2c_address;
+ strlcpy(info.type, "wm8990", I2C_NAME_SIZE);
+
+ adapter = i2c_get_adapter(setup->i2c_bus);
+ if (!adapter) {
+ dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+ setup->i2c_bus);
+ goto err_driver;
+ }
+
+ client = i2c_new_device(adapter, &info);
+ i2c_put_adapter(adapter);
+ if (!client) {
+ dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+ (unsigned int)info.addr);
+ goto err_driver;
+ }
+
+ return 0;
+
+err_driver:
+ i2c_del_driver(&wm8990_i2c_driver);
+ return -ENODEV;
+}
+#endif
+
+static int wm8990_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct wm8990_setup_data *setup;
+ struct snd_soc_codec *codec;
+ struct wm8990_priv *wm8990;
+ int ret;
+
+ pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION);
+
+ setup = socdev->codec_data;
+ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (codec == NULL)
+ return -ENOMEM;
+
+ wm8990 = kzalloc(sizeof(struct wm8990_priv), GFP_KERNEL);
+ if (wm8990 == NULL) {
+ kfree(codec);
+ return -ENOMEM;
+ }
+
+ codec->private_data = wm8990;
+ socdev->codec = codec;
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8990_socdev = socdev;
+
+ ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ ret = wm8990_add_i2c_device(pdev, setup);
+ }
+#endif
+
+ if (ret != 0) {
+ kfree(codec->private_data);
+ kfree(codec);
+ }
+ return ret;
+}
+
+/* power down chip */
+static int wm8990_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec->control_data)
+ wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_unregister_device(codec->control_data);
+ i2c_del_driver(&wm8990_i2c_driver);
+#endif
+ kfree(codec->private_data);
+ kfree(codec);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8990 = {
+ .probe = wm8990_probe,
+ .remove = wm8990_remove,
+ .suspend = wm8990_suspend,
+ .resume = wm8990_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
+
+MODULE_DESCRIPTION("ASoC WM8990 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
new file mode 100644
index 0000000..0e192f3
--- /dev/null
+++ b/sound/soc/codecs/wm8990.h
@@ -0,0 +1,843 @@
+/*
+ * wm8990.h -- audio driver for WM8990
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef __WM8990REGISTERDEFS_H__
+#define __WM8990REGISTERDEFS_H__
+
+/*
+ * Register values.
+ */
+#define WM8990_RESET 0x00
+#define WM8990_POWER_MANAGEMENT_1 0x01
+#define WM8990_POWER_MANAGEMENT_2 0x02
+#define WM8990_POWER_MANAGEMENT_3 0x03
+#define WM8990_AUDIO_INTERFACE_1 0x04
+#define WM8990_AUDIO_INTERFACE_2 0x05
+#define WM8990_CLOCKING_1 0x06
+#define WM8990_CLOCKING_2 0x07
+#define WM8990_AUDIO_INTERFACE_3 0x08
+#define WM8990_AUDIO_INTERFACE_4 0x09
+#define WM8990_DAC_CTRL 0x0A
+#define WM8990_LEFT_DAC_DIGITAL_VOLUME 0x0B
+#define WM8990_RIGHT_DAC_DIGITAL_VOLUME 0x0C
+#define WM8990_DIGITAL_SIDE_TONE 0x0D
+#define WM8990_ADC_CTRL 0x0E
+#define WM8990_LEFT_ADC_DIGITAL_VOLUME 0x0F
+#define WM8990_RIGHT_ADC_DIGITAL_VOLUME 0x10
+#define WM8990_GPIO_CTRL_1 0x12
+#define WM8990_GPIO1_GPIO2 0x13
+#define WM8990_GPIO3_GPIO4 0x14
+#define WM8990_GPIO5_GPIO6 0x15
+#define WM8990_GPIOCTRL_2 0x16
+#define WM8990_GPIO_POL 0x17
+#define WM8990_LEFT_LINE_INPUT_1_2_VOLUME 0x18
+#define WM8990_LEFT_LINE_INPUT_3_4_VOLUME 0x19
+#define WM8990_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A
+#define WM8990_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B
+#define WM8990_LEFT_OUTPUT_VOLUME 0x1C
+#define WM8990_RIGHT_OUTPUT_VOLUME 0x1D
+#define WM8990_LINE_OUTPUTS_VOLUME 0x1E
+#define WM8990_OUT3_4_VOLUME 0x1F
+#define WM8990_LEFT_OPGA_VOLUME 0x20
+#define WM8990_RIGHT_OPGA_VOLUME 0x21
+#define WM8990_SPEAKER_VOLUME 0x22
+#define WM8990_CLASSD1 0x23
+#define WM8990_CLASSD3 0x25
+#define WM8990_CLASSD4 0x26
+#define WM8990_INPUT_MIXER1 0x27
+#define WM8990_INPUT_MIXER2 0x28
+#define WM8990_INPUT_MIXER3 0x29
+#define WM8990_INPUT_MIXER4 0x2A
+#define WM8990_INPUT_MIXER5 0x2B
+#define WM8990_INPUT_MIXER6 0x2C
+#define WM8990_OUTPUT_MIXER1 0x2D
+#define WM8990_OUTPUT_MIXER2 0x2E
+#define WM8990_OUTPUT_MIXER3 0x2F
+#define WM8990_OUTPUT_MIXER4 0x30
+#define WM8990_OUTPUT_MIXER5 0x31
+#define WM8990_OUTPUT_MIXER6 0x32
+#define WM8990_OUT3_4_MIXER 0x33
+#define WM8990_LINE_MIXER1 0x34
+#define WM8990_LINE_MIXER2 0x35
+#define WM8990_SPEAKER_MIXER 0x36
+#define WM8990_ADDITIONAL_CONTROL 0x37
+#define WM8990_ANTIPOP1 0x38
+#define WM8990_ANTIPOP2 0x39
+#define WM8990_MICBIAS 0x3A
+#define WM8990_PLL1 0x3C
+#define WM8990_PLL2 0x3D
+#define WM8990_PLL3 0x3E
+#define WM8990_INTDRIVBITS 0x3F
+
+#define WM8990_REGISTER_COUNT 60
+#define WM8990_MAX_REGISTER 0x3F
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Reset
+ */
+#define WM8990_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID */
+
+/*
+ * R1 (0x01) - Power Management (1)
+ */
+#define WM8990_SPK_ENA 0x1000 /* SPK_ENA */
+#define WM8990_SPK_ENA_BIT 12
+#define WM8990_OUT3_ENA 0x0800 /* OUT3_ENA */
+#define WM8990_OUT3_ENA_BIT 11
+#define WM8990_OUT4_ENA 0x0400 /* OUT4_ENA */
+#define WM8990_OUT4_ENA_BIT 10
+#define WM8990_LOUT_ENA 0x0200 /* LOUT_ENA */
+#define WM8990_LOUT_ENA_BIT 9
+#define WM8990_ROUT_ENA 0x0100 /* ROUT_ENA */
+#define WM8990_ROUT_ENA_BIT 8
+#define WM8990_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */
+#define WM8990_MICBIAS_ENA_BIT 4
+#define WM8990_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */
+#define WM8990_VREF_ENA 0x0001 /* VREF_ENA */
+#define WM8990_VREF_ENA_BIT 0
+
+/*
+ * R2 (0x02) - Power Management (2)
+ */
+#define WM8990_PLL_ENA 0x8000 /* PLL_ENA */
+#define WM8990_PLL_ENA_BIT 15
+#define WM8990_TSHUT_ENA 0x4000 /* TSHUT_ENA */
+#define WM8990_TSHUT_ENA_BIT 14
+#define WM8990_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */
+#define WM8990_TSHUT_OPDIS_BIT 13
+#define WM8990_OPCLK_ENA 0x0800 /* OPCLK_ENA */
+#define WM8990_OPCLK_ENA_BIT 11
+#define WM8990_AINL_ENA 0x0200 /* AINL_ENA */
+#define WM8990_AINL_ENA_BIT 9
+#define WM8990_AINR_ENA 0x0100 /* AINR_ENA */
+#define WM8990_AINR_ENA_BIT 8
+#define WM8990_LIN34_ENA 0x0080 /* LIN34_ENA */
+#define WM8990_LIN34_ENA_BIT 7
+#define WM8990_LIN12_ENA 0x0040 /* LIN12_ENA */
+#define WM8990_LIN12_ENA_BIT 6
+#define WM8990_RIN34_ENA 0x0020 /* RIN34_ENA */
+#define WM8990_RIN34_ENA_BIT 5
+#define WM8990_RIN12_ENA 0x0010 /* RIN12_ENA */
+#define WM8990_RIN12_ENA_BIT 4
+#define WM8990_ADCL_ENA 0x0002 /* ADCL_ENA */
+#define WM8990_ADCL_ENA_BIT 1
+#define WM8990_ADCR_ENA 0x0001 /* ADCR_ENA */
+#define WM8990_ADCR_ENA_BIT 0
+
+/*
+ * R3 (0x03) - Power Management (3)
+ */
+#define WM8990_LON_ENA 0x2000 /* LON_ENA */
+#define WM8990_LON_ENA_BIT 13
+#define WM8990_LOP_ENA 0x1000 /* LOP_ENA */
+#define WM8990_LOP_ENA_BIT 12
+#define WM8990_RON_ENA 0x0800 /* RON_ENA */
+#define WM8990_RON_ENA_BIT 11
+#define WM8990_ROP_ENA 0x0400 /* ROP_ENA */
+#define WM8990_ROP_ENA_BIT 10
+#define WM8990_LOPGA_ENA 0x0080 /* LOPGA_ENA */
+#define WM8990_LOPGA_ENA_BIT 7
+#define WM8990_ROPGA_ENA 0x0040 /* ROPGA_ENA */
+#define WM8990_ROPGA_ENA_BIT 6
+#define WM8990_LOMIX_ENA 0x0020 /* LOMIX_ENA */
+#define WM8990_LOMIX_ENA_BIT 5
+#define WM8990_ROMIX_ENA 0x0010 /* ROMIX_ENA */
+#define WM8990_ROMIX_ENA_BIT 4
+#define WM8990_DACL_ENA 0x0002 /* DACL_ENA */
+#define WM8990_DACL_ENA_BIT 1
+#define WM8990_DACR_ENA 0x0001 /* DACR_ENA */
+#define WM8990_DACR_ENA_BIT 0
+
+/*
+ * R4 (0x04) - Audio Interface (1)
+ */
+#define WM8990_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */
+#define WM8990_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */
+#define WM8990_AIFADC_TDM 0x2000 /* AIFADC_TDM */
+#define WM8990_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */
+#define WM8990_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */
+#define WM8990_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */
+#define WM8990_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */
+#define WM8990_AIF_WL_16BITS (0 << 5)
+#define WM8990_AIF_WL_20BITS (1 << 5)
+#define WM8990_AIF_WL_24BITS (2 << 5)
+#define WM8990_AIF_WL_32BITS (3 << 5)
+#define WM8990_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */
+#define WM8990_AIF_TMF_RIGHTJ (0 << 3)
+#define WM8990_AIF_TMF_LEFTJ (1 << 3)
+#define WM8990_AIF_TMF_I2S (2 << 3)
+#define WM8990_AIF_TMF_DSP (3 << 3)
+
+/*
+ * R5 (0x05) - Audio Interface (2)
+ */
+#define WM8990_DACL_SRC 0x8000 /* DACL_SRC */
+#define WM8990_DACR_SRC 0x4000 /* DACR_SRC */
+#define WM8990_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */
+#define WM8990_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */
+#define WM8990_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST */
+#define WM8990_DAC_COMP 0x0010 /* DAC_COMP */
+#define WM8990_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */
+#define WM8990_ADC_COMP 0x0004 /* ADC_COMP */
+#define WM8990_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */
+#define WM8990_LOOPBACK 0x0001 /* LOOPBACK */
+
+/*
+ * R6 (0x06) - Clocking (1)
+ */
+#define WM8990_TOCLK_RATE 0x8000 /* TOCLK_RATE */
+#define WM8990_TOCLK_ENA 0x4000 /* TOCLK_ENA */
+#define WM8990_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */
+#define WM8990_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */
+#define WM8990_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */
+#define WM8990_BCLK_DIV_1 (0x0 << 1)
+#define WM8990_BCLK_DIV_1_5 (0x1 << 1)
+#define WM8990_BCLK_DIV_2 (0x2 << 1)
+#define WM8990_BCLK_DIV_3 (0x3 << 1)
+#define WM8990_BCLK_DIV_4 (0x4 << 1)
+#define WM8990_BCLK_DIV_5_5 (0x5 << 1)
+#define WM8990_BCLK_DIV_6 (0x6 << 1)
+#define WM8990_BCLK_DIV_8 (0x7 << 1)
+#define WM8990_BCLK_DIV_11 (0x8 << 1)
+#define WM8990_BCLK_DIV_12 (0x9 << 1)
+#define WM8990_BCLK_DIV_16 (0xA << 1)
+#define WM8990_BCLK_DIV_22 (0xB << 1)
+#define WM8990_BCLK_DIV_24 (0xC << 1)
+#define WM8990_BCLK_DIV_32 (0xD << 1)
+#define WM8990_BCLK_DIV_44 (0xE << 1)
+#define WM8990_BCLK_DIV_48 (0xF << 1)
+
+/*
+ * R7 (0x07) - Clocking (2)
+ */
+#define WM8990_MCLK_SRC 0x8000 /* MCLK_SRC */
+#define WM8990_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */
+#define WM8990_CLK_FORCE 0x2000 /* CLK_FORCE */
+#define WM8990_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */
+#define WM8990_MCLK_DIV_1 (0 << 11)
+#define WM8990_MCLK_DIV_2 (2 << 11)
+#define WM8990_MCLK_INV 0x0400 /* MCLK_INV */
+#define WM8990_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV */
+#define WM8990_ADC_CLKDIV_1 (0 << 5)
+#define WM8990_ADC_CLKDIV_1_5 (1 << 5)
+#define WM8990_ADC_CLKDIV_2 (2 << 5)
+#define WM8990_ADC_CLKDIV_3 (3 << 5)
+#define WM8990_ADC_CLKDIV_4 (4 << 5)
+#define WM8990_ADC_CLKDIV_5_5 (5 << 5)
+#define WM8990_ADC_CLKDIV_6 (6 << 5)
+#define WM8990_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */
+#define WM8990_DAC_CLKDIV_1 (0 << 2)
+#define WM8990_DAC_CLKDIV_1_5 (1 << 2)
+#define WM8990_DAC_CLKDIV_2 (2 << 2)
+#define WM8990_DAC_CLKDIV_3 (3 << 2)
+#define WM8990_DAC_CLKDIV_4 (4 << 2)
+#define WM8990_DAC_CLKDIV_5_5 (5 << 2)
+#define WM8990_DAC_CLKDIV_6 (6 << 2)
+
+/*
+ * R8 (0x08) - Audio Interface (3)
+ */
+#define WM8990_AIF_MSTR1 0x8000 /* AIF_MSTR1 */
+#define WM8990_AIF_MSTR2 0x4000 /* AIF_MSTR2 */
+#define WM8990_AIF_SEL 0x2000 /* AIF_SEL */
+#define WM8990_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */
+#define WM8990_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE */
+
+/*
+ * R9 (0x09) - Audio Interface (4)
+ */
+#define WM8990_ALRCGPIO1 0x8000 /* ALRCGPIO1 */
+#define WM8990_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */
+#define WM8990_AIF_TRIS 0x2000 /* AIF_TRIS */
+#define WM8990_DACLRC_DIR 0x0800 /* DACLRC_DIR */
+#define WM8990_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE */
+
+/*
+ * R10 (0x0A) - DAC CTRL
+ */
+#define WM8990_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */
+#define WM8990_DAC_MONO 0x0200 /* DAC_MONO */
+#define WM8990_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */
+#define WM8990_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */
+#define WM8990_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */
+#define WM8990_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */
+#define WM8990_DAC_MUTE 0x0004 /* DAC_MUTE */
+#define WM8990_DACL_DATINV 0x0002 /* DACL_DATINV */
+#define WM8990_DACR_DATINV 0x0001 /* DACR_DATINV */
+
+/*
+ * R11 (0x0B) - Left DAC Digital Volume
+ */
+#define WM8990_DAC_VU 0x0100 /* DAC_VU */
+#define WM8990_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */
+#define WM8990_DACL_VOL_SHIFT 0
+/*
+ * R12 (0x0C) - Right DAC Digital Volume
+ */
+#define WM8990_DAC_VU 0x0100 /* DAC_VU */
+#define WM8990_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */
+#define WM8990_DACR_VOL_SHIFT 0
+/*
+ * R13 (0x0D) - Digital Side Tone
+ */
+#define WM8990_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL */
+#define WM8990_ADCL_DAC_SVOL_SHIFT 9
+#define WM8990_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL */
+#define WM8990_ADCR_DAC_SVOL_SHIFT 5
+#define WM8990_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */
+#define WM8990_ADC_TO_DACL_SHIFT 2
+#define WM8990_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */
+#define WM8990_ADC_TO_DACR_SHIFT 0
+
+/*
+ * R14 (0x0E) - ADC CTRL
+ */
+#define WM8990_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */
+#define WM8990_ADC_HPF_ENA_BIT 8
+#define WM8990_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */
+#define WM8990_ADC_HPF_CUT_SHIFT 5
+#define WM8990_ADCL_DATINV 0x0002 /* ADCL_DATINV */
+#define WM8990_ADCL_DATINV_BIT 1
+#define WM8990_ADCR_DATINV 0x0001 /* ADCR_DATINV */
+#define WM8990_ADCR_DATINV_BIT 0
+
+/*
+ * R15 (0x0F) - Left ADC Digital Volume
+ */
+#define WM8990_ADC_VU 0x0100 /* ADC_VU */
+#define WM8990_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */
+#define WM8990_ADCL_VOL_SHIFT 0
+
+/*
+ * R16 (0x10) - Right ADC Digital Volume
+ */
+#define WM8990_ADC_VU 0x0100 /* ADC_VU */
+#define WM8990_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */
+#define WM8990_ADCR_VOL_SHIFT 0
+
+/*
+ * R18 (0x12) - GPIO CTRL 1
+ */
+#define WM8990_IRQ 0x1000 /* IRQ */
+#define WM8990_TEMPOK 0x0800 /* TEMPOK */
+#define WM8990_MICSHRT 0x0400 /* MICSHRT */
+#define WM8990_MICDET 0x0200 /* MICDET */
+#define WM8990_PLL_LCK 0x0100 /* PLL_LCK */
+#define WM8990_GPI8_STATUS 0x0080 /* GPI8_STATUS */
+#define WM8990_GPI7_STATUS 0x0040 /* GPI7_STATUS */
+#define WM8990_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */
+#define WM8990_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */
+#define WM8990_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */
+#define WM8990_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */
+#define WM8990_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */
+#define WM8990_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */
+
+/*
+ * R19 (0x13) - GPIO1 & GPIO2
+ */
+#define WM8990_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */
+#define WM8990_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */
+#define WM8990_GPIO2_PU 0x2000 /* GPIO2_PU */
+#define WM8990_GPIO2_PD 0x1000 /* GPIO2_PD */
+#define WM8990_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */
+#define WM8990_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */
+#define WM8990_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */
+#define WM8990_GPIO1_PU 0x0020 /* GPIO1_PU */
+#define WM8990_GPIO1_PD 0x0010 /* GPIO1_PD */
+#define WM8990_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */
+
+/*
+ * R20 (0x14) - GPIO3 & GPIO4
+ */
+#define WM8990_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */
+#define WM8990_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */
+#define WM8990_GPIO4_PU 0x2000 /* GPIO4_PU */
+#define WM8990_GPIO4_PD 0x1000 /* GPIO4_PD */
+#define WM8990_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */
+#define WM8990_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */
+#define WM8990_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */
+#define WM8990_GPIO3_PU 0x0020 /* GPIO3_PU */
+#define WM8990_GPIO3_PD 0x0010 /* GPIO3_PD */
+#define WM8990_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */
+
+/*
+ * R21 (0x15) - GPIO5 & GPIO6
+ */
+#define WM8990_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */
+#define WM8990_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */
+#define WM8990_GPIO6_PU 0x2000 /* GPIO6_PU */
+#define WM8990_GPIO6_PD 0x1000 /* GPIO6_PD */
+#define WM8990_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */
+#define WM8990_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */
+#define WM8990_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */
+#define WM8990_GPIO5_PU 0x0020 /* GPIO5_PU */
+#define WM8990_GPIO5_PD 0x0010 /* GPIO5_PD */
+#define WM8990_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */
+
+/*
+ * R22 (0x16) - GPIOCTRL 2
+ */
+#define WM8990_RD_3W_ENA 0x8000 /* RD_3W_ENA */
+#define WM8990_MODE_3W4W 0x4000 /* MODE_3W4W */
+#define WM8990_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */
+#define WM8990_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */
+#define WM8990_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */
+#define WM8990_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */
+#define WM8990_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */
+#define WM8990_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */
+#define WM8990_GPI8_ENA 0x0010 /* GPI8_ENA */
+#define WM8990_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */
+#define WM8990_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */
+#define WM8990_GPI7_ENA 0x0001 /* GPI7_ENA */
+
+/*
+ * R23 (0x17) - GPIO_POL
+ */
+#define WM8990_IRQ_INV 0x1000 /* IRQ_INV */
+#define WM8990_TEMPOK_POL 0x0800 /* TEMPOK_POL */
+#define WM8990_MICSHRT_POL 0x0400 /* MICSHRT_POL */
+#define WM8990_MICDET_POL 0x0200 /* MICDET_POL */
+#define WM8990_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */
+#define WM8990_GPI8_POL 0x0080 /* GPI8_POL */
+#define WM8990_GPI7_POL 0x0040 /* GPI7_POL */
+#define WM8990_GPIO6_POL 0x0020 /* GPIO6_POL */
+#define WM8990_GPIO5_POL 0x0010 /* GPIO5_POL */
+#define WM8990_GPIO4_POL 0x0008 /* GPIO4_POL */
+#define WM8990_GPIO3_POL 0x0004 /* GPIO3_POL */
+#define WM8990_GPIO2_POL 0x0002 /* GPIO2_POL */
+#define WM8990_GPIO1_POL 0x0001 /* GPIO1_POL */
+
+/*
+ * R24 (0x18) - Left Line Input 1&2 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_LI12MUTE 0x0080 /* LI12MUTE */
+#define WM8990_LI12MUTE_BIT 7
+#define WM8990_LI12ZC 0x0040 /* LI12ZC */
+#define WM8990_LI12ZC_BIT 6
+#define WM8990_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */
+#define WM8990_LIN12VOL_SHIFT 0
+/*
+ * R25 (0x19) - Left Line Input 3&4 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_LI34MUTE 0x0080 /* LI34MUTE */
+#define WM8990_LI34MUTE_BIT 7
+#define WM8990_LI34ZC 0x0040 /* LI34ZC */
+#define WM8990_LI34ZC_BIT 6
+#define WM8990_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */
+#define WM8990_LIN34VOL_SHIFT 0
+
+/*
+ * R26 (0x1A) - Right Line Input 1&2 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_RI12MUTE 0x0080 /* RI12MUTE */
+#define WM8990_RI12MUTE_BIT 7
+#define WM8990_RI12ZC 0x0040 /* RI12ZC */
+#define WM8990_RI12ZC_BIT 6
+#define WM8990_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */
+#define WM8990_RIN12VOL_SHIFT 0
+
+/*
+ * R27 (0x1B) - Right Line Input 3&4 Volume
+ */
+#define WM8990_IPVU 0x0100 /* IPVU */
+#define WM8990_RI34MUTE 0x0080 /* RI34MUTE */
+#define WM8990_RI34MUTE_BIT 7
+#define WM8990_RI34ZC 0x0040 /* RI34ZC */
+#define WM8990_RI34ZC_BIT 6
+#define WM8990_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */
+#define WM8990_RIN34VOL_SHIFT 0
+
+/*
+ * R28 (0x1C) - Left Output Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_LOZC 0x0080 /* LOZC */
+#define WM8990_LOZC_BIT 7
+#define WM8990_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */
+#define WM8990_LOUTVOL_SHIFT 0
+/*
+ * R29 (0x1D) - Right Output Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_ROZC 0x0080 /* ROZC */
+#define WM8990_ROZC_BIT 7
+#define WM8990_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */
+#define WM8990_ROUTVOL_SHIFT 0
+/*
+ * R30 (0x1E) - Line Outputs Volume
+ */
+#define WM8990_LONMUTE 0x0040 /* LONMUTE */
+#define WM8990_LONMUTE_BIT 6
+#define WM8990_LOPMUTE 0x0020 /* LOPMUTE */
+#define WM8990_LOPMUTE_BIT 5
+#define WM8990_LOATTN 0x0010 /* LOATTN */
+#define WM8990_LOATTN_BIT 4
+#define WM8990_RONMUTE 0x0004 /* RONMUTE */
+#define WM8990_RONMUTE_BIT 2
+#define WM8990_ROPMUTE 0x0002 /* ROPMUTE */
+#define WM8990_ROPMUTE_BIT 1
+#define WM8990_ROATTN 0x0001 /* ROATTN */
+#define WM8990_ROATTN_BIT 0
+
+/*
+ * R31 (0x1F) - Out3/4 Volume
+ */
+#define WM8990_OUT3MUTE 0x0020 /* OUT3MUTE */
+#define WM8990_OUT3MUTE_BIT 5
+#define WM8990_OUT3ATTN 0x0010 /* OUT3ATTN */
+#define WM8990_OUT3ATTN_BIT 4
+#define WM8990_OUT4MUTE 0x0002 /* OUT4MUTE */
+#define WM8990_OUT4MUTE_BIT 1
+#define WM8990_OUT4ATTN 0x0001 /* OUT4ATTN */
+#define WM8990_OUT4ATTN_BIT 0
+
+/*
+ * R32 (0x20) - Left OPGA Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_LOPGAZC 0x0080 /* LOPGAZC */
+#define WM8990_LOPGAZC_BIT 7
+#define WM8990_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */
+#define WM8990_LOPGAVOL_SHIFT 0
+
+/*
+ * R33 (0x21) - Right OPGA Volume
+ */
+#define WM8990_OPVU 0x0100 /* OPVU */
+#define WM8990_ROPGAZC 0x0080 /* ROPGAZC */
+#define WM8990_ROPGAZC_BIT 7
+#define WM8990_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */
+#define WM8990_ROPGAVOL_SHIFT 0
+/*
+ * R34 (0x22) - Speaker Volume
+ */
+#define WM8990_SPKATTN_MASK 0x0003 /* SPKATTN - [1:0] */
+#define WM8990_SPKATTN_SHIFT 0
+
+/*
+ * R35 (0x23) - ClassD1
+ */
+#define WM8990_CDMODE 0x0100 /* CDMODE */
+#define WM8990_CDMODE_BIT 8
+
+/*
+ * R37 (0x25) - ClassD3
+ */
+#define WM8990_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */
+#define WM8990_DCGAIN_SHIFT 3
+#define WM8990_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */
+#define WM8990_ACGAIN_SHIFT 0
+
+/*
+ * R38 (0x26) - ClassD4
+ */
+#define WM8990_SPKZC_MASK 0x0001 /* SPKZC */
+#define WM8990_SPKZC_SHIFT 7 /* SPKZC */
+#define WM8990_SPKVOL_MASK 0x007F /* SPKVOL - [6:0] */
+#define WM8990_SPKVOL_SHIFT 0 /* SPKVOL - [6:0] */
+
+/*
+ * R39 (0x27) - Input Mixer1
+ */
+#define WM8990_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */
+#define WM8990_AINLMODE_SHIFT 2
+#define WM8990_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */
+#define WM8990_AINRMODE_SHIFT 0
+
+/*
+ * R40 (0x28) - Input Mixer2
+ */
+#define WM8990_LMP4 0x0080 /* LMP4 */
+#define WM8990_LMP4_BIT 7 /* LMP4 */
+#define WM8990_LMN3 0x0040 /* LMN3 */
+#define WM8990_LMN3_BIT 6 /* LMN3 */
+#define WM8990_LMP2 0x0020 /* LMP2 */
+#define WM8990_LMP2_BIT 5 /* LMP2 */
+#define WM8990_LMN1 0x0010 /* LMN1 */
+#define WM8990_LMN1_BIT 4 /* LMN1 */
+#define WM8990_RMP4 0x0008 /* RMP4 */
+#define WM8990_RMP4_BIT 3 /* RMP4 */
+#define WM8990_RMN3 0x0004 /* RMN3 */
+#define WM8990_RMN3_BIT 2 /* RMN3 */
+#define WM8990_RMP2 0x0002 /* RMP2 */
+#define WM8990_RMP2_BIT 1 /* RMP2 */
+#define WM8990_RMN1 0x0001 /* RMN1 */
+#define WM8990_RMN1_BIT 0 /* RMN1 */
+
+/*
+ * R41 (0x29) - Input Mixer3
+ */
+#define WM8990_L34MNB 0x0100 /* L34MNB */
+#define WM8990_L34MNB_BIT 8
+#define WM8990_L34MNBST 0x0080 /* L34MNBST */
+#define WM8990_L34MNBST_BIT 7
+#define WM8990_L12MNB 0x0020 /* L12MNB */
+#define WM8990_L12MNB_BIT 5
+#define WM8990_L12MNBST 0x0010 /* L12MNBST */
+#define WM8990_L12MNBST_BIT 4
+#define WM8990_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */
+#define WM8990_LDBVOL_SHIFT 0
+
+/*
+ * R42 (0x2A) - Input Mixer4
+ */
+#define WM8990_R34MNB 0x0100 /* R34MNB */
+#define WM8990_R34MNB_BIT 8
+#define WM8990_R34MNBST 0x0080 /* R34MNBST */
+#define WM8990_R34MNBST_BIT 7
+#define WM8990_R12MNB 0x0020 /* R12MNB */
+#define WM8990_R12MNB_BIT 5
+#define WM8990_R12MNBST 0x0010 /* R12MNBST */
+#define WM8990_R12MNBST_BIT 4
+#define WM8990_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */
+#define WM8990_RDBVOL_SHIFT 0
+
+/*
+ * R43 (0x2B) - Input Mixer5
+ */
+#define WM8990_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */
+#define WM8990_LI2BVOL_SHIFT 6
+#define WM8990_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */
+#define WM8990_LR4BVOL_SHIFT 3
+#define WM8990_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */
+#define WM8990_LL4BVOL_SHIFT 0
+
+/*
+ * R44 (0x2C) - Input Mixer6
+ */
+#define WM8990_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */
+#define WM8990_RI2BVOL_SHIFT 6
+#define WM8990_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */
+#define WM8990_RL4BVOL_SHIFT 3
+#define WM8990_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */
+#define WM8990_RR4BVOL_SHIFT 0
+
+/*
+ * R45 (0x2D) - Output Mixer1
+ */
+#define WM8990_LRBLO 0x0080 /* LRBLO */
+#define WM8990_LRBLO_BIT 7
+#define WM8990_LLBLO 0x0040 /* LLBLO */
+#define WM8990_LLBLO_BIT 6
+#define WM8990_LRI3LO 0x0020 /* LRI3LO */
+#define WM8990_LRI3LO_BIT 5
+#define WM8990_LLI3LO 0x0010 /* LLI3LO */
+#define WM8990_LLI3LO_BIT 4
+#define WM8990_LR12LO 0x0008 /* LR12LO */
+#define WM8990_LR12LO_BIT 3
+#define WM8990_LL12LO 0x0004 /* LL12LO */
+#define WM8990_LL12LO_BIT 2
+#define WM8990_LDLO 0x0001 /* LDLO */
+#define WM8990_LDLO_BIT 0
+
+/*
+ * R46 (0x2E) - Output Mixer2
+ */
+#define WM8990_RLBRO 0x0080 /* RLBRO */
+#define WM8990_RLBRO_BIT 7
+#define WM8990_RRBRO 0x0040 /* RRBRO */
+#define WM8990_RRBRO_BIT 6
+#define WM8990_RLI3RO 0x0020 /* RLI3RO */
+#define WM8990_RLI3RO_BIT 5
+#define WM8990_RRI3RO 0x0010 /* RRI3RO */
+#define WM8990_RRI3RO_BIT 4
+#define WM8990_RL12RO 0x0008 /* RL12RO */
+#define WM8990_RL12RO_BIT 3
+#define WM8990_RR12RO 0x0004 /* RR12RO */
+#define WM8990_RR12RO_BIT 2
+#define WM8990_RDRO 0x0001 /* RDRO */
+#define WM8990_RDRO_BIT 0
+
+/*
+ * R47 (0x2F) - Output Mixer3
+ */
+#define WM8990_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */
+#define WM8990_LLI3LOVOL_SHIFT 6
+#define WM8990_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */
+#define WM8990_LR12LOVOL_SHIFT 3
+#define WM8990_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */
+#define WM8990_LL12LOVOL_SHIFT 0
+
+/*
+ * R48 (0x30) - Output Mixer4
+ */
+#define WM8990_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */
+#define WM8990_RRI3ROVOL_SHIFT 6
+#define WM8990_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */
+#define WM8990_RL12ROVOL_SHIFT 3
+#define WM8990_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */
+#define WM8990_RR12ROVOL_SHIFT 0
+
+/*
+ * R49 (0x31) - Output Mixer5
+ */
+#define WM8990_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */
+#define WM8990_LRI3LOVOL_SHIFT 6
+#define WM8990_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */
+#define WM8990_LRBLOVOL_SHIFT 3
+#define WM8990_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */
+#define WM8990_LLBLOVOL_SHIFT 0
+
+/*
+ * R50 (0x32) - Output Mixer6
+ */
+#define WM8990_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */
+#define WM8990_RLI3ROVOL_SHIFT 6
+#define WM8990_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */
+#define WM8990_RLBROVOL_SHIFT 3
+#define WM8990_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */
+#define WM8990_RRBROVOL_SHIFT 0
+
+/*
+ * R51 (0x33) - Out3/4 Mixer
+ */
+#define WM8990_VSEL_MASK 0x0180 /* VSEL - [8:7] */
+#define WM8990_LI4O3 0x0020 /* LI4O3 */
+#define WM8990_LI4O3_BIT 5
+#define WM8990_LPGAO3 0x0010 /* LPGAO3 */
+#define WM8990_LPGAO3_BIT 4
+#define WM8990_RI4O4 0x0002 /* RI4O4 */
+#define WM8990_RI4O4_BIT 1
+#define WM8990_RPGAO4 0x0001 /* RPGAO4 */
+#define WM8990_RPGAO4_BIT 0
+/*
+ * R52 (0x34) - Line Mixer1
+ */
+#define WM8990_LLOPGALON 0x0040 /* LLOPGALON */
+#define WM8990_LLOPGALON_BIT 6
+#define WM8990_LROPGALON 0x0020 /* LROPGALON */
+#define WM8990_LROPGALON_BIT 5
+#define WM8990_LOPLON 0x0010 /* LOPLON */
+#define WM8990_LOPLON_BIT 4
+#define WM8990_LR12LOP 0x0004 /* LR12LOP */
+#define WM8990_LR12LOP_BIT 2
+#define WM8990_LL12LOP 0x0002 /* LL12LOP */
+#define WM8990_LL12LOP_BIT 1
+#define WM8990_LLOPGALOP 0x0001 /* LLOPGALOP */
+#define WM8990_LLOPGALOP_BIT 0
+/*
+ * R53 (0x35) - Line Mixer2
+ */
+#define WM8990_RROPGARON 0x0040 /* RROPGARON */
+#define WM8990_RROPGARON_BIT 6
+#define WM8990_RLOPGARON 0x0020 /* RLOPGARON */
+#define WM8990_RLOPGARON_BIT 5
+#define WM8990_ROPRON 0x0010 /* ROPRON */
+#define WM8990_ROPRON_BIT 4
+#define WM8990_RL12ROP 0x0004 /* RL12ROP */
+#define WM8990_RL12ROP_BIT 2
+#define WM8990_RR12ROP 0x0002 /* RR12ROP */
+#define WM8990_RR12ROP_BIT 1
+#define WM8990_RROPGAROP 0x0001 /* RROPGAROP */
+#define WM8990_RROPGAROP_BIT 0
+
+/*
+ * R54 (0x36) - Speaker Mixer
+ */
+#define WM8990_LB2SPK 0x0080 /* LB2SPK */
+#define WM8990_LB2SPK_BIT 7
+#define WM8990_RB2SPK 0x0040 /* RB2SPK */
+#define WM8990_RB2SPK_BIT 6
+#define WM8990_LI2SPK 0x0020 /* LI2SPK */
+#define WM8990_LI2SPK_BIT 5
+#define WM8990_RI2SPK 0x0010 /* RI2SPK */
+#define WM8990_RI2SPK_BIT 4
+#define WM8990_LOPGASPK 0x0008 /* LOPGASPK */
+#define WM8990_LOPGASPK_BIT 3
+#define WM8990_ROPGASPK 0x0004 /* ROPGASPK */
+#define WM8990_ROPGASPK_BIT 2
+#define WM8990_LDSPK 0x0002 /* LDSPK */
+#define WM8990_LDSPK_BIT 1
+#define WM8990_RDSPK 0x0001 /* RDSPK */
+#define WM8990_RDSPK_BIT 0
+
+/*
+ * R55 (0x37) - Additional Control
+ */
+#define WM8990_VROI 0x0001 /* VROI */
+
+/*
+ * R56 (0x38) - AntiPOP1
+ */
+#define WM8990_DIS_LLINE 0x0020 /* DIS_LLINE */
+#define WM8990_DIS_RLINE 0x0010 /* DIS_RLINE */
+#define WM8990_DIS_OUT3 0x0008 /* DIS_OUT3 */
+#define WM8990_DIS_OUT4 0x0004 /* DIS_OUT4 */
+#define WM8990_DIS_LOUT 0x0002 /* DIS_LOUT */
+#define WM8990_DIS_ROUT 0x0001 /* DIS_ROUT */
+
+/*
+ * R57 (0x39) - AntiPOP2
+ */
+#define WM8990_SOFTST 0x0040 /* SOFTST */
+#define WM8990_BUFIOEN 0x0008 /* BUFIOEN */
+#define WM8990_BUFDCOPEN 0x0004 /* BUFDCOPEN */
+#define WM8990_POBCTRL 0x0002 /* POBCTRL */
+#define WM8990_VMIDTOG 0x0001 /* VMIDTOG */
+
+/*
+ * R58 (0x3A) - MICBIAS
+ */
+#define WM8990_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */
+#define WM8990_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */
+#define WM8990_MCD 0x0004 /* MCD */
+#define WM8990_MBSEL 0x0001 /* MBSEL */
+
+/*
+ * R60 (0x3C) - PLL1
+ */
+#define WM8990_SDM 0x0080 /* SDM */
+#define WM8990_PRESCALE 0x0040 /* PRESCALE */
+#define WM8990_PLLN_MASK 0x000F /* PLLN - [3:0] */
+
+/*
+ * R61 (0x3D) - PLL2
+ */
+#define WM8990_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */
+
+/*
+ * R62 (0x3E) - PLL3
+ */
+#define WM8990_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */
+
+/*
+ * R63 (0x3F) - Internal Driver Bits
+ */
+#define WM8990_INMIXL_PWR_BIT 0
+#define WM8990_AINLMUX_PWR_BIT 1
+#define WM8990_INMIXR_PWR_BIT 2
+#define WM8990_AINRMUX_PWR_BIT 3
+
+struct wm8990_setup_data {
+ unsigned i2c_bus;
+ unsigned short i2c_address;
+};
+
+#define WM8990_MCLK_DIV 0
+#define WM8990_DACCLK_DIV 1
+#define WM8990_ADCCLK_DIV 2
+#define WM8990_BCLK_DIV 3
+
+extern struct snd_soc_dai wm8990_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8990;
+
+#endif /* __WM8990REGISTERDEFS_H__ */
+/*------------------------------ END OF FILE ---------------------------------*/
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
new file mode 100644
index 0000000..ffb471e
--- /dev/null
+++ b/sound/soc/codecs/wm9712.c
@@ -0,0 +1,750 @@
+/*
+ * wm9712.c -- ALSA Soc WM9712 codec support
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include "wm9712.h"
+
+#define WM9712_VERSION "0.4"
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val);
+
+/*
+ * WM9712 register cache
+ */
+static const u16 wm9712_reg[] = {
+ 0x6174, 0x8000, 0x8000, 0x8000, /* 6 */
+ 0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */
+ 0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */
+ 0xe808, 0x3000, 0x8000, 0x0000, /* 1e */
+ 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+ 0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */
+ 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+ 0x0000, 0x2000, 0x0000, 0x0000, /* 3e */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 46 */
+ 0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */
+ 0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */
+ 0x0008, 0x0000, 0x0000, 0x0000, /* 5e */
+ 0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+ 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */
+ 0x0000, 0x0000 /* virtual hp mixers */
+};
+
+/* virtual HP mixers regs */
+#define HPL_MIXER 0x80
+#define HPR_MIXER 0x82
+
+static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
+static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right",
+ "Mono"};
+static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"};
+static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
+static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2",
+ "Stereo"};
+static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer",
+ "Line", "Headphone Mixer", "Phone Mixer", "Phone"};
+static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"};
+static const char *wm9712_diff_sel[] = {"Mic", "Line"};
+
+static const struct soc_enum wm9712_enum[] = {
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select),
+SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux),
+SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src),
+SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc),
+SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base),
+SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain),
+SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type),
+SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel),
+};
+
+static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
+SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
+
+SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
+SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0),
+SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0),
+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
+SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+SOC_ENUM("ALC Function", wm9712_enum[0]),
+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+SOC_ENUM("ALC NG Type", wm9712_enum[10]),
+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
+
+SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1),
+SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
+
+SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
+SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1),
+SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
+
+SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
+SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1),
+SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1),
+
+SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1),
+SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1),
+SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1),
+
+SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1),
+SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
+
+SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
+SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
+
+SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
+SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
+SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0),
+
+SOC_ENUM("Bass Control", wm9712_enum[5]),
+SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
+SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
+SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1),
+SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
+
+SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
+
+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
+};
+
+/* add non dapm controls */
+static int wm9712_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm9712_snd_ac97_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* We have to create a fake left and right HP mixers because
+ * the codec only has a single control that is shared by both channels.
+ * This makes it impossible to determine the audio path.
+ */
+static int mixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ u16 l, r, beep, line, phone, mic, pcm, aux;
+
+ l = ac97_read(w->codec, HPL_MIXER);
+ r = ac97_read(w->codec, HPR_MIXER);
+ beep = ac97_read(w->codec, AC97_PC_BEEP);
+ mic = ac97_read(w->codec, AC97_VIDEO);
+ phone = ac97_read(w->codec, AC97_PHONE);
+ line = ac97_read(w->codec, AC97_LINE);
+ pcm = ac97_read(w->codec, AC97_PCM);
+ aux = ac97_read(w->codec, AC97_CD);
+
+ if (l & 0x1 || r & 0x1)
+ ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
+
+ if (l & 0x2 || r & 0x2)
+ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+
+ if (l & 0x4 || r & 0x4)
+ ac97_write(w->codec, AC97_LINE, line & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_LINE, line | 0x8000);
+
+ if (l & 0x8 || r & 0x8)
+ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+
+ if (l & 0x10 || r & 0x10)
+ ac97_write(w->codec, AC97_CD, aux & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_CD, aux | 0x8000);
+
+ if (l & 0x20 || r & 0x20)
+ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+
+ return 0;
+}
+
+/* Left Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0),
+};
+
+/* Right Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
+ SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0),
+};
+
+/* Speaker Mixer */
+static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1),
+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1),
+ SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1),
+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1),
+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1),
+};
+
+/* Phone Mixer */
+static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = {
+ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1),
+ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1),
+ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1),
+ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1),
+ SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1),
+ SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1),
+};
+
+/* ALC headphone mux */
+static const struct snd_kcontrol_new wm9712_alc_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[1]);
+
+/* out 3 mux */
+static const struct snd_kcontrol_new wm9712_out3_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[2]);
+
+/* spk mux */
+static const struct snd_kcontrol_new wm9712_spk_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[3]);
+
+/* Capture to Phone mux */
+static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[4]);
+
+/* Capture left select */
+static const struct snd_kcontrol_new wm9712_capture_selectl_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[8]);
+
+/* Capture right select */
+static const struct snd_kcontrol_new wm9712_capture_selectr_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[9]);
+
+/* Mic select */
+static const struct snd_kcontrol_new wm9712_mic_src_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+
+/* diff select */
+static const struct snd_kcontrol_new wm9712_diff_sel_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[11]);
+
+static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = {
+SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_alc_mux_controls),
+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_out3_mux_controls),
+SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_spk_mux_controls),
+SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_phone_mux_controls),
+SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_selectl_controls),
+SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
+ &wm9712_capture_selectr_controls),
+SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
+ &wm9712_diff_sel_controls),
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
+ &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
+ &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
+ &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
+ &wm9712_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm9712_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1),
+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1),
+SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("LOUT2"),
+SND_SOC_DAPM_OUTPUT("ROUT2"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_INPUT("LINEINL"),
+SND_SOC_DAPM_INPUT("LINEINR"),
+SND_SOC_DAPM_INPUT("PHONE"),
+SND_SOC_DAPM_INPUT("PCBEEP"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* virtual mixer - mixes left & right channels for spk and mono */
+ {"AC97 Mixer", NULL, "Left DAC"},
+ {"AC97 Mixer", NULL, "Right DAC"},
+
+ /* Left HP mixer */
+ {"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Left HP Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
+ {"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
+ {"Left HP Mixer", NULL, "ALC Sidetone Mux"},
+
+ /* Right HP mixer */
+ {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Right HP Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
+ {"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
+ {"Right HP Mixer", NULL, "ALC Sidetone Mux"},
+
+ /* speaker mixer */
+ {"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Speaker Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
+
+ /* Phone mixer */
+ {"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+ {"Phone Mixer", "Line Bypass Switch", "Line PGA"},
+ {"Phone Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"},
+ {"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"},
+
+ /* inputs */
+ {"Line PGA", NULL, "LINEINL"},
+ {"Line PGA", NULL, "LINEINR"},
+ {"Phone PGA", NULL, "PHONE"},
+ {"Mic PGA", NULL, "MIC1"},
+ {"Mic PGA", NULL, "MIC2"},
+
+ /* left capture selector */
+ {"Left Capture Select", "Mic", "MIC1"},
+ {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
+ {"Left Capture Select", "Line", "LINEINL"},
+ {"Left Capture Select", "Headphone Mixer", "Left HP Mixer"},
+ {"Left Capture Select", "Phone Mixer", "Phone Mixer"},
+ {"Left Capture Select", "Phone", "PHONE"},
+
+ /* right capture selector */
+ {"Right Capture Select", "Mic", "MIC2"},
+ {"Right Capture Select", "Speaker Mixer", "Speaker Mixer"},
+ {"Right Capture Select", "Line", "LINEINR"},
+ {"Right Capture Select", "Headphone Mixer", "Right HP Mixer"},
+ {"Right Capture Select", "Phone Mixer", "Phone Mixer"},
+ {"Right Capture Select", "Phone", "PHONE"},
+
+ /* ALC Sidetone */
+ {"ALC Sidetone Mux", "Stereo", "Left Capture Select"},
+ {"ALC Sidetone Mux", "Stereo", "Right Capture Select"},
+ {"ALC Sidetone Mux", "Left", "Left Capture Select"},
+ {"ALC Sidetone Mux", "Right", "Right Capture Select"},
+
+ /* ADC's */
+ {"Left ADC", NULL, "Left Capture Select"},
+ {"Right ADC", NULL, "Right Capture Select"},
+
+ /* outputs */
+ {"MONOOUT", NULL, "Phone Mixer"},
+ {"HPOUTL", NULL, "Headphone PGA"},
+ {"Headphone PGA", NULL, "Left HP Mixer"},
+ {"HPOUTR", NULL, "Headphone PGA"},
+ {"Headphone PGA", NULL, "Right HP Mixer"},
+
+ /* mono mixer */
+ {"Mono Mixer", NULL, "Left HP Mixer"},
+ {"Mono Mixer", NULL, "Right HP Mixer"},
+
+ /* Out3 Mux */
+ {"Out3 Mux", "Left", "Left HP Mixer"},
+ {"Out3 Mux", "Mono", "Phone Mixer"},
+ {"Out3 Mux", "Left + Right", "Mono Mixer"},
+ {"Out 3 PGA", NULL, "Out3 Mux"},
+ {"OUT3", NULL, "Out 3 PGA"},
+
+ /* speaker Mux */
+ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
+ {"Speaker Mux", "Headphone Mix", "Mono Mixer"},
+ {"Speaker PGA", NULL, "Speaker Mux"},
+ {"LOUT2", NULL, "Speaker PGA"},
+ {"ROUT2", NULL, "Speaker PGA"},
+};
+
+static int wm9712_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
+ ARRAY_SIZE(wm9712_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
+ reg == AC97_REC_GAIN)
+ return soc_ac97_ops.read(codec->ac97, reg);
+ else {
+ reg = reg >> 1;
+
+ if (reg > (ARRAY_SIZE(wm9712_reg)))
+ return -EIO;
+
+ return cache[reg];
+ }
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ reg = reg >> 1;
+ if (reg <= (ARRAY_SIZE(wm9712_reg)))
+ cache[reg] = val;
+
+ return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg;
+ u16 vra;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 vra, xsle;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+ xsle = ac97_read(codec, AC97_PCI_SID);
+ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+
+ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
+}
+
+#define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai wm9712_dai[] = {
+{
+ .name = "AC97 HiFi",
+ .type = SND_SOC_DAI_AC97_BUS,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9712_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9712_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_prepare,},
+},
+{
+ .name = "AC97 Aux",
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM9712_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_aux_prepare,},
+}
+};
+EXPORT_SYMBOL_GPL(wm9712_dai);
+
+static int wm9712_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* disable everything including AC link */
+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+ ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (ac97_read(codec, 0) == wm9712_reg[0])
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (ac97_read(codec, 0) != wm9712_reg[0])
+ goto err;
+ return 0;
+
+err:
+ printk(KERN_ERR "WM9712 AC97 reset failed\n");
+ return -EIO;
+}
+
+static int wm9712_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm9712_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9712_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ if (ret == 0) {
+ /* Sync reg_cache with the hardware after cold reset */
+ for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) {
+ if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
+ (i > 0x58 && i != 0x5c))
+ continue;
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+ }
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return ret;
+}
+
+static int wm9712_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache = kmemdup(wm9712_reg, sizeof(wm9712_reg), GFP_KERNEL);
+
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ codec->reg_cache_size = sizeof(wm9712_reg);
+ codec->reg_cache_step = 2;
+
+ codec->name = "WM9712";
+ codec->owner = THIS_MODULE;
+ codec->dai = wm9712_dai;
+ codec->num_dai = ARRAY_SIZE(wm9712_dai);
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ codec->set_bias_level = wm9712_set_bias_level;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
+ goto codec_err;
+ }
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ ret = wm9712_reset(codec, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "AC97 link error\n");
+ goto reset_err;
+ }
+
+ /* set alc mux to none */
+ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
+
+ wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm9712_add_controls(codec);
+ wm9712_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
+ printk(KERN_ERR "wm9712: failed to register card\n");
+ goto reset_err;
+ }
+
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+
+codec_err:
+ kfree(codec->reg_cache);
+
+cache_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int wm9712_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_dapm_free(socdev);
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9712 = {
+ .probe = wm9712_soc_probe,
+ .remove = wm9712_soc_remove,
+ .suspend = wm9712_soc_suspend,
+ .resume = wm9712_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
+
+MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
new file mode 100644
index 0000000..d29e8a1
--- /dev/null
+++ b/sound/soc/codecs/wm9712.h
@@ -0,0 +1,14 @@
+/*
+ * wm9712.h -- WM9712 Soc Audio driver
+ */
+
+#ifndef _WM9712_H
+#define _WM9712_H
+
+#define WM9712_DAI_AC97_HIFI 0
+#define WM9712_DAI_AC97_AUX 1
+
+extern struct snd_soc_dai wm9712_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9712;
+
+#endif
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
new file mode 100644
index 0000000..945b32e
--- /dev/null
+++ b/sound/soc/codecs/wm9713.c
@@ -0,0 +1,1306 @@
+/*
+ * wm9713.c -- ALSA Soc WM9713 codec support
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Features:-
+ *
+ * o Support for AC97 Codec, Voice DAC and Aux DAC
+ * o Support for DAPM
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "wm9713.h"
+
+#define WM9713_VERSION "0.15"
+
+struct wm9713_priv {
+ u32 pll_in; /* PLL input frequency */
+ u32 pll_out; /* PLL output frequency */
+};
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val);
+
+/*
+ * WM9713 register cache
+ * Reg 0x3c bit 15 is used by touch driver.
+ */
+static const u16 wm9713_reg[] = {
+ 0x6174, 0x8080, 0x8080, 0x8080,
+ 0xc880, 0xe808, 0xe808, 0x0808,
+ 0x00da, 0x8000, 0xd600, 0xaaa0,
+ 0xaaa0, 0xaaa0, 0x0000, 0x0000,
+ 0x0f0f, 0x0040, 0x0000, 0x7f00,
+ 0x0405, 0x0410, 0xbb80, 0xbb80,
+ 0x0000, 0xbb80, 0x0000, 0x4523,
+ 0x0000, 0x2000, 0x7eff, 0xffff,
+ 0x0000, 0x0000, 0x0080, 0x0000,
+ 0x0000, 0x0000, 0xfffe, 0xffff,
+ 0x0000, 0x0000, 0x0000, 0xfffe,
+ 0x4000, 0x0000, 0x0000, 0x0000,
+ 0xb032, 0x3e00, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0006,
+ 0x0001, 0x0000, 0x574d, 0x4c13,
+ 0x0000, 0x0000, 0x0000
+};
+
+/* virtual HP mixers regs */
+#define HPL_MIXER 0x80
+#define HPR_MIXER 0x82
+#define MICB_MUX 0x82
+
+static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
+static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
+static const char *wm9713_rec_src[] =
+ {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker",
+ "Mono Out", "Zh"};
+static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv",
+ "Mono Vmid", "Inv Vmid"};
+static const char *wm9713_spk_pga[] =
+ {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid",
+ "Speaker Vmid", "Inv Vmid"};
+static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone",
+ "Headphone Vmid"};
+static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"};
+static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"};
+static const char *wm9713_dac_inv[] =
+ {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone",
+ "Headphone Mono", "NC", "Vmid"};
+static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"};
+static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"};
+static const char *wm9713_micb_select[] = {"MPB", "MPA"};
+
+static const struct soc_enum wm9713_enum[] = {
+SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/
+SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */
+SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */
+SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */
+SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */
+};
+
+static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = {
+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1),
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1),
+SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1),
+SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1),
+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+
+SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
+SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
+
+SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1),
+SOC_ENUM("Capture Volume Steps", wm9713_enum[5]),
+SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0),
+SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
+
+SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1),
+SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
+SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
+
+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+SOC_ENUM("ALC Function", wm9713_enum[6]),
+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0),
+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+SOC_ENUM("ALC NG Type", wm9713_enum[17]),
+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0),
+
+SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0),
+SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
+
+SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0),
+SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1),
+
+SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1),
+SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0),
+SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1),
+
+SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1),
+SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
+
+SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
+SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
+SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
+
+SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
+SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
+SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1),
+
+SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1),
+SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1),
+SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1),
+
+SOC_ENUM("Bass Control", wm9713_enum[16]),
+SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
+SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1),
+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0),
+SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1),
+SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1),
+
+SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0),
+SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
+SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
+};
+
+/* add non dapm controls */
+static int wm9713_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm9713_snd_ac97_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* We have to create a fake left and right HP mixers because
+ * the codec only has a single control that is shared by both channels.
+ * This makes it impossible to determine the audio path using the current
+ * register map, thus we add a new (virtual) register to help determine the
+ * audio route within the device.
+ */
+static int mixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ u16 l, r, beep, tone, phone, rec, pcm, aux;
+
+ l = ac97_read(w->codec, HPL_MIXER);
+ r = ac97_read(w->codec, HPR_MIXER);
+ beep = ac97_read(w->codec, AC97_PC_BEEP);
+ tone = ac97_read(w->codec, AC97_MASTER_TONE);
+ phone = ac97_read(w->codec, AC97_PHONE);
+ rec = ac97_read(w->codec, AC97_REC_SEL);
+ pcm = ac97_read(w->codec, AC97_PCM);
+ aux = ac97_read(w->codec, AC97_AUX);
+
+ if (event & SND_SOC_DAPM_PRE_REG)
+ return 0;
+ if ((l & 0x1) || (r & 0x1))
+ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+
+ if ((l & 0x2) || (r & 0x2))
+ ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000);
+
+ if ((l & 0x4) || (r & 0x4))
+ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+
+ if ((l & 0x8) || (r & 0x8))
+ ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000);
+
+ if ((l & 0x10) || (r & 0x10))
+ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+
+ if ((l & 0x20) || (r & 0x20))
+ ac97_write(w->codec, AC97_AUX, aux & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_AUX, aux | 0x8000);
+
+ return 0;
+}
+
+/* Left Headphone Mixers */
+static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0),
+SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
+};
+
+/* Right Headphone Mixers */
+static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0),
+SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0),
+};
+
+/* headphone capture mux */
+static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[1]);
+
+/* headphone mic mux */
+static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[0]);
+
+/* Speaker Mixer */
+static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1),
+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
+};
+
+/* Mono Mixer */
+static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1),
+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1),
+SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1),
+SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1),
+};
+
+/* mono mic mux */
+static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[2]);
+
+/* mono output mux */
+static const struct snd_kcontrol_new wm9713_mono_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[7]);
+
+/* speaker left output mux */
+static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[8]);
+
+/* speaker right output mux */
+static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[9]);
+
+/* headphone left output mux */
+static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[10]);
+
+/* headphone right output mux */
+static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[11]);
+
+/* Out3 mux */
+static const struct snd_kcontrol_new wm9713_out3_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[12]);
+
+/* Out4 mux */
+static const struct snd_kcontrol_new wm9713_out4_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[13]);
+
+/* DAC inv mux 1 */
+static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[14]);
+
+/* DAC inv mux 2 */
+static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[15]);
+
+/* Capture source left */
+static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[3]);
+
+/* Capture source right */
+static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[4]);
+
+/* mic source */
+static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[18]);
+
+/* mic source B virtual control */
+static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[19]);
+
+static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = {
+SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_rec_mux_controls),
+SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_mic_mux_controls),
+SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_mono_mic_mux_controls),
+SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_mono_mux_controls),
+SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_spkl_mux_controls),
+SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_spkr_mux_controls),
+SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hpl_out_mux_controls),
+SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hpr_out_mux_controls),
+SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_out3_mux_controls),
+SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_out4_mux_controls),
+SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0,
+ &wm9713_dac_inv1_mux_controls),
+SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0,
+ &wm9713_dac_inv2_mux_controls),
+SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_rec_srcl_mux_controls),
+SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_rec_srcr_mux_controls),
+SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_mic_sel_mux_controls),
+SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_micb_sel_mux_controls),
+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
+ &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
+ &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1,
+ &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1,
+ &wm9713_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm9713_speaker_mixer_controls)),
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1),
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
+SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
+SND_SOC_DAPM_ADC("Left HiFi ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Right HiFi ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Left Voice ADC", "Left Voice Capture", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Right Voice ADC", "Right Voice Capture", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1),
+SND_SOC_DAPM_OUTPUT("MONO"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_OUTPUT("SPKL"),
+SND_SOC_DAPM_OUTPUT("SPKR"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_INPUT("LINEL"),
+SND_SOC_DAPM_INPUT("LINER"),
+SND_SOC_DAPM_INPUT("MONOIN"),
+SND_SOC_DAPM_INPUT("PCBEEP"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2A"),
+SND_SOC_DAPM_INPUT("MIC2B"),
+SND_SOC_DAPM_VMID("VMID"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* left HP mixer */
+ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"},
+ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
+ {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"},
+ {"Left HP Mixer", NULL, "Capture Headphone Mux"},
+
+ /* right HP mixer */
+ {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"},
+ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
+ {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"},
+ {"Right HP Mixer", NULL, "Capture Headphone Mux"},
+
+ /* virtual mixer - mixes left & right channels for spk and mono */
+ {"AC97 Mixer", NULL, "Left DAC"},
+ {"AC97 Mixer", NULL, "Right DAC"},
+ {"Line Mixer", NULL, "Right Line In"},
+ {"Line Mixer", NULL, "Left Line In"},
+ {"HP Mixer", NULL, "Left HP Mixer"},
+ {"HP Mixer", NULL, "Right HP Mixer"},
+ {"Capture Mixer", NULL, "Left Capture Source"},
+ {"Capture Mixer", NULL, "Right Capture Source"},
+
+ /* speaker mixer */
+ {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"},
+ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"},
+
+ /* mono mixer */
+ {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Mono Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"},
+ {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"},
+ {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"},
+ {"Mono Mixer", NULL, "Capture Mono Mux"},
+
+ /* DAC inv mux 1 */
+ {"DAC Inv Mux 1", "Mono", "Mono Mixer"},
+ {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"},
+ {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"},
+ {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"},
+ {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"},
+
+ /* DAC inv mux 2 */
+ {"DAC Inv Mux 2", "Mono", "Mono Mixer"},
+ {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"},
+ {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"},
+ {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"},
+ {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"},
+
+ /* headphone left mux */
+ {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"},
+
+ /* headphone right mux */
+ {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"},
+
+ /* speaker left mux */
+ {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"},
+ {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"},
+ {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"},
+
+ /* speaker right mux */
+ {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"},
+ {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"},
+ {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"},
+
+ /* mono mux */
+ {"Mono Out Mux", "Mono", "Mono Mixer"},
+ {"Mono Out Mux", "Inv", "DAC Inv Mux 1"},
+
+ /* out 3 mux */
+ {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"},
+
+ /* out 4 mux */
+ {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"},
+
+ /* output pga */
+ {"HPL", NULL, "Left Headphone"},
+ {"Left Headphone", NULL, "Left Headphone Out Mux"},
+ {"HPR", NULL, "Right Headphone"},
+ {"Right Headphone", NULL, "Right Headphone Out Mux"},
+ {"OUT3", NULL, "Out 3"},
+ {"Out 3", NULL, "Out 3 Mux"},
+ {"OUT4", NULL, "Out 4"},
+ {"Out 4", NULL, "Out 4 Mux"},
+ {"SPKL", NULL, "Left Speaker"},
+ {"Left Speaker", NULL, "Left Speaker Out Mux"},
+ {"SPKR", NULL, "Right Speaker"},
+ {"Right Speaker", NULL, "Right Speaker Out Mux"},
+ {"MONO", NULL, "Mono Out"},
+ {"Mono Out", NULL, "Mono Out Mux"},
+
+ /* input pga */
+ {"Left Line In", NULL, "LINEL"},
+ {"Right Line In", NULL, "LINER"},
+ {"Mono In", NULL, "MONOIN"},
+ {"Mic A PGA", NULL, "Mic A Pre Amp"},
+ {"Mic B PGA", NULL, "Mic B Pre Amp"},
+
+ /* left capture select */
+ {"Left Capture Source", "Mic 1", "Mic A Pre Amp"},
+ {"Left Capture Source", "Mic 2", "Mic B Pre Amp"},
+ {"Left Capture Source", "Line", "LINEL"},
+ {"Left Capture Source", "Mono In", "MONOIN"},
+ {"Left Capture Source", "Headphone", "Left HP Mixer"},
+ {"Left Capture Source", "Speaker", "Speaker Mixer"},
+ {"Left Capture Source", "Mono Out", "Mono Mixer"},
+
+ /* right capture select */
+ {"Right Capture Source", "Mic 1", "Mic A Pre Amp"},
+ {"Right Capture Source", "Mic 2", "Mic B Pre Amp"},
+ {"Right Capture Source", "Line", "LINER"},
+ {"Right Capture Source", "Mono In", "MONOIN"},
+ {"Right Capture Source", "Headphone", "Right HP Mixer"},
+ {"Right Capture Source", "Speaker", "Speaker Mixer"},
+ {"Right Capture Source", "Mono Out", "Mono Mixer"},
+
+ /* left ADC */
+ {"Left ADC", NULL, "Left Capture Source"},
+ {"Left Voice ADC", NULL, "Left ADC"},
+ {"Left HiFi ADC", NULL, "Left ADC"},
+
+ /* right ADC */
+ {"Right ADC", NULL, "Right Capture Source"},
+ {"Right Voice ADC", NULL, "Right ADC"},
+ {"Right HiFi ADC", NULL, "Right ADC"},
+
+ /* mic */
+ {"Mic A Pre Amp", NULL, "Mic A Source"},
+ {"Mic A Source", "Mic 1", "MIC1"},
+ {"Mic A Source", "Mic 2 A", "MIC2A"},
+ {"Mic A Source", "Mic 2 B", "Mic B Source"},
+ {"Mic B Pre Amp", "MPB", "Mic B Source"},
+ {"Mic B Source", NULL, "MIC2B"},
+
+ /* headphone capture */
+ {"Capture Headphone Mux", "Stereo", "Capture Mixer"},
+ {"Capture Headphone Mux", "Left", "Left Capture Source"},
+ {"Capture Headphone Mux", "Right", "Right Capture Source"},
+
+ /* mono capture */
+ {"Capture Mono Mux", "Stereo", "Capture Mixer"},
+ {"Capture Mono Mux", "Left", "Left Capture Source"},
+ {"Capture Mono Mux", "Right", "Right Capture Source"},
+};
+
+static int wm9713_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+ ARRAY_SIZE(wm9713_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
+ reg == AC97_CD)
+ return soc_ac97_ops.read(codec->ac97, reg);
+ else {
+ reg = reg >> 1;
+
+ if (reg > (ARRAY_SIZE(wm9713_reg)))
+ return -EIO;
+
+ return cache[reg];
+ }
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg < 0x7c)
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ reg = reg >> 1;
+ if (reg <= (ARRAY_SIZE(wm9713_reg)))
+ cache[reg] = val;
+
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 divsel:1;
+ u32 divctl:1;
+ u32 lf:1;
+ u32 n:4;
+ u32 k:24;
+};
+
+/* The size in bits of the PLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 22) * 10)
+
+static void pll_factors(struct _pll_div *pll_div, unsigned int source)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod, target;
+
+ /* The the PLL output is always 98.304MHz. */
+ target = 98304000;
+
+ /* If the input frequency is over 14.4MHz then scale it down. */
+ if (source > 14400000) {
+ source >>= 1;
+ pll_div->divsel = 1;
+
+ if (source > 14400000) {
+ source >>= 1;
+ pll_div->divctl = 1;
+ } else
+ pll_div->divctl = 0;
+
+ } else {
+ pll_div->divsel = 0;
+ pll_div->divctl = 0;
+ }
+
+ /* Low frequency sources require an additional divide in the
+ * loop.
+ */
+ if (source < 8192000) {
+ pll_div->lf = 1;
+ target >>= 2;
+ } else
+ pll_div->lf = 0;
+
+ Ndiv = target / source;
+ if ((Ndiv < 5) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM9713 PLL N value %d out of recommended range!\n",
+ Ndiv);
+
+ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div->k = K;
+}
+
+/**
+ * Please note that changing the PLL input frequency may require
+ * resynchronisation with the AC97 controller.
+ */
+static int wm9713_set_pll(struct snd_soc_codec *codec,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct wm9713_priv *wm9713 = codec->private_data;
+ u16 reg, reg2;
+ struct _pll_div pll_div;
+
+ /* turn PLL off ? */
+ if (freq_in == 0 || freq_out == 0) {
+ /* disable PLL power and select ext source */
+ reg = ac97_read(codec, AC97_HANDSET_RATE);
+ ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
+ reg = ac97_read(codec, AC97_EXTENDED_MID);
+ ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
+ wm9713->pll_out = 0;
+ return 0;
+ }
+
+ pll_factors(&pll_div, freq_in);
+
+ if (pll_div.k == 0) {
+ reg = (pll_div.n << 12) | (pll_div.lf << 11) |
+ (pll_div.divsel << 9) | (pll_div.divctl << 8);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+ } else {
+ /* write the fractional k to the reg 0x46 pages */
+ reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) |
+ (pll_div.divsel << 9) | (pll_div.divctl << 8);
+
+ /* K [21:20] */
+ reg = reg2 | (0x5 << 4) | (pll_div.k >> 20);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [19:16] */
+ reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [15:12] */
+ reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [11:8] */
+ reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [7:4] */
+ reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+ }
+
+ /* turn PLL on and select as source */
+ reg = ac97_read(codec, AC97_EXTENDED_MID);
+ ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
+ reg = ac97_read(codec, AC97_HANDSET_RATE);
+ ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
+ wm9713->pll_out = freq_out;
+ wm9713->pll_in = freq_in;
+
+ /* wait 10ms AC97 link frames for the link to stabilise */
+ schedule_timeout_interruptible(msecs_to_jiffies(10));
+ return 0;
+}
+
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
+}
+
+/*
+ * Tristate the PCM DAI lines, tristate can be disabled by calling
+ * wm9713_set_dai_fmt()
+ */
+static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
+ int tristate)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff;
+
+ if (tristate)
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
+
+ return 0;
+}
+
+/*
+ * Configure WM9713 clock dividers.
+ * Voice DAC needs 256 FS
+ */
+static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM9713_PCMCLK_DIV:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_CLKA_MULT:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_CLKB_MULT:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_HIFI_DIV:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_PCMBCLK_DIV:
+ reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff;
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div);
+ break;
+ case WM9713_PCMCLK_PLL_DIV:
+ reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80;
+ ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div);
+ break;
+ case WM9713_HIFI_PLL_DIV:
+ reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80;
+ ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5;
+ u16 reg = 0x8000;
+
+ /* clock masters */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ reg |= 0x4000;
+ gpio |= 0x0010;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ reg |= 0x6000;
+ gpio |= 0x0018;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ reg |= 0x2000;
+ gpio |= 0x001a;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ gpio |= 0x0012;
+ break;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ reg |= 0x00c0;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ reg |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ reg |= 0x0040;
+ break;
+ }
+
+ /* DAI format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ reg |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ reg |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ reg |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ reg |= 0x0043;
+ break;
+ }
+
+ ac97_write(codec, AC97_GPIO_CFG, gpio);
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
+ return 0;
+}
+
+static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ reg |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ reg |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ reg |= 0x000c;
+ break;
+ }
+
+ /* enable PCM interface in master mode */
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
+ return 0;
+}
+
+static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 status;
+
+ /* Gracefully shut down the voice interface. */
+ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
+ ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
+ schedule_timeout_interruptible(msecs_to_jiffies(1));
+ ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
+ ac97_write(codec, AC97_EXTENDED_MID, status);
+}
+
+static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg;
+ u16 vra;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 vra, xsle;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+ xsle = ac97_read(codec, AC97_PCI_SID);
+ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+
+ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
+}
+
+#define WM9713_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_11025 | \
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define WM9713_PCM_FORMATS \
+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
+ SNDRV_PCM_FORMAT_S24_LE)
+
+struct snd_soc_dai wm9713_dai[] = {
+{
+ .name = "AC97 HiFi",
+ .type = SND_SOC_DAI_AC97_BUS,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9713_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9713_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_hifi_prepare,},
+ .dai_ops = {
+ .set_clkdiv = wm9713_set_dai_clkdiv,
+ .set_pll = wm9713_set_dai_pll,},
+ },
+ {
+ .name = "AC97 Aux",
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM9713_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_aux_prepare,},
+ .dai_ops = {
+ .set_clkdiv = wm9713_set_dai_clkdiv,
+ .set_pll = wm9713_set_dai_pll,},
+ },
+ {
+ .name = "WM9713 Voice",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM9713_PCM_RATES,
+ .formats = WM9713_PCM_FORMATS,},
+ .capture = {
+ .stream_name = "Voice Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9713_PCM_RATES,
+ .formats = WM9713_PCM_FORMATS,},
+ .ops = {
+ .hw_params = wm9713_pcm_hw_params,
+ .shutdown = wm9713_voiceshutdown,},
+ .dai_ops = {
+ .set_clkdiv = wm9713_set_dai_clkdiv,
+ .set_pll = wm9713_set_dai_pll,
+ .set_fmt = wm9713_set_dai_fmt,
+ .set_tristate = wm9713_set_dai_tristate,
+ },
+ },
+};
+EXPORT_SYMBOL_GPL(wm9713_dai);
+
+int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (ac97_read(codec, 0) == wm9713_reg[0])
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (ac97_read(codec, 0) != wm9713_reg[0])
+ return -EIO;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wm9713_reset);
+
+static int wm9713_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* enable thermal shutdown */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* enable master bias and vmid */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* disable everything including AC link */
+ ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+ ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static int wm9713_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 reg;
+
+ /* Disable everything except touchpanel - that will be handled
+ * by the touch driver and left disabled if touch is not in
+ * use. */
+ reg = ac97_read(codec, AC97_EXTENDED_MID);
+ ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff);
+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+ ac97_write(codec, AC97_POWERDOWN, 0x6f00);
+ ac97_write(codec, AC97_POWERDOWN, 0xffff);
+
+ return 0;
+}
+
+static int wm9713_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm9713_priv *wm9713 = codec->private_data;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9713_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* do we need to re-start the PLL ? */
+ if (wm9713->pll_out)
+ wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
+
+ /* only synchronise the codec if warm reset failed */
+ if (ret == 0) {
+ for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) {
+ if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID ||
+ i == AC97_EXTENDED_MSTATUS || i > 0x66)
+ continue;
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+ }
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return ret;
+}
+
+static int wm9713_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0, reg;
+
+ printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ codec->reg_cache_size = sizeof(wm9713_reg);
+ codec->reg_cache_step = 2;
+
+ codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL);
+ if (codec->private_data == NULL) {
+ ret = -ENOMEM;
+ goto priv_err;
+ }
+
+ codec->name = "WM9713";
+ codec->owner = THIS_MODULE;
+ codec->dai = wm9713_dai;
+ codec->num_dai = ARRAY_SIZE(wm9713_dai);
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ codec->set_bias_level = wm9713_set_bias_level;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0)
+ goto codec_err;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ /* do a cold reset for the controller and then try
+ * a warm reset followed by an optional cold reset for codec */
+ wm9713_reset(codec, 0);
+ ret = wm9713_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "AC97 link error\n");
+ goto reset_err;
+ }
+
+ wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* unmute the adc - move to kcontrol */
+ reg = ac97_read(codec, AC97_CD) & 0x7fff;
+ ac97_write(codec, AC97_CD, reg);
+
+ wm9713_add_controls(codec);
+ wm9713_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0)
+ goto reset_err;
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+
+codec_err:
+ kfree(codec->private_data);
+
+priv_err:
+ kfree(codec->reg_cache);
+
+cache_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int wm9713_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_dapm_free(socdev);
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->private_data);
+ kfree(codec->reg_cache);
+ kfree(codec->dai);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9713 = {
+ .probe = wm9713_soc_probe,
+ .remove = wm9713_soc_remove,
+ .suspend = wm9713_soc_suspend,
+ .resume = wm9713_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713);
+
+MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h
new file mode 100644
index 0000000..63b8d81
--- /dev/null
+++ b/sound/soc/codecs/wm9713.h
@@ -0,0 +1,53 @@
+/*
+ * wm9713.h -- WM9713 Soc Audio driver
+ */
+
+#ifndef _WM9713_H
+#define _WM9713_H
+
+/* clock inputs */
+#define WM9713_CLKA_PIN 0
+#define WM9713_CLKB_PIN 1
+
+/* clock divider ID's */
+#define WM9713_PCMCLK_DIV 0
+#define WM9713_CLKA_MULT 1
+#define WM9713_CLKB_MULT 2
+#define WM9713_HIFI_DIV 3
+#define WM9713_PCMBCLK_DIV 4
+#define WM9713_PCMCLK_PLL_DIV 5
+#define WM9713_HIFI_PLL_DIV 6
+
+/* Calculate the appropriate bit mask for the external PCM clock divider */
+#define WM9713_PCMDIV(x) ((x - 1) << 8)
+
+/* Calculate the appropriate bit mask for the external HiFi clock divider */
+#define WM9713_HIFIDIV(x) ((x - 1) << 12)
+
+/* MCLK clock mulitipliers */
+#define WM9713_CLKA_X1 (0 << 1)
+#define WM9713_CLKA_X2 (1 << 1)
+#define WM9713_CLKB_X1 (0 << 2)
+#define WM9713_CLKB_X2 (1 << 2)
+
+/* MCLK clock MUX */
+#define WM9713_CLK_MUX_A (0 << 0)
+#define WM9713_CLK_MUX_B (1 << 0)
+
+/* Voice DAI BCLK divider */
+#define WM9713_PCMBCLK_DIV_1 (0 << 9)
+#define WM9713_PCMBCLK_DIV_2 (1 << 9)
+#define WM9713_PCMBCLK_DIV_4 (2 << 9)
+#define WM9713_PCMBCLK_DIV_8 (3 << 9)
+#define WM9713_PCMBCLK_DIV_16 (4 << 9)
+
+#define WM9713_DAI_AC97_HIFI 0
+#define WM9713_DAI_AC97_AUX 1
+#define WM9713_DAI_PCM_VOICE 2
+
+extern struct snd_soc_codec_device soc_codec_dev_wm9713;
+extern struct snd_soc_dai wm9713_dai[3];
+
+int wm9713_reset(struct snd_soc_codec *codec, int try_warm);
+
+#endif
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
new file mode 100644
index 0000000..8f7e338
--- /dev/null
+++ b/sound/soc/davinci/Kconfig
@@ -0,0 +1,19 @@
+config SND_DAVINCI_SOC
+ tristate "SoC Audio for the TI DAVINCI chip"
+ depends on ARCH_DAVINCI
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the DAVINCI AC97 or I2S interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_DAVINCI_SOC_I2S
+ tristate
+
+config SND_DAVINCI_SOC_EVM
+ tristate "SoC Audio support for DaVinci EVM"
+ depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+ select SND_DAVINCI_SOC_I2S
+ select SND_SOC_TLV320AIC3X
+ help
+ Say Y if you want to add support for SoC audio on TI
+ DaVinci EVM platform.
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
new file mode 100644
index 0000000..ca772e5
--- /dev/null
+++ b/sound/soc/davinci/Makefile
@@ -0,0 +1,11 @@
+# DAVINCI Platform Support
+snd-soc-davinci-objs := davinci-pcm.o
+snd-soc-davinci-i2s-objs := davinci-i2s.o
+
+obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o
+obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
+
+# DAVINCI Machine Support
+snd-soc-evm-objs := davinci-evm.o
+
+obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
new file mode 100644
index 0000000..9e6062c
--- /dev/null
+++ b/sound/soc/davinci/davinci-evm.c
@@ -0,0 +1,202 @@
+/*
+ * ASoC driver for TI DAVINCI EVM platform
+ *
+ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/dma.h>
+#include <mach/hardware.h>
+
+#include "../codecs/tlv320aic3x.h"
+#include "davinci-pcm.h"
+#include "davinci-i2s.h"
+
+#define EVM_CODEC_CLOCK 22579200
+
+static int evm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
+ SND_SOC_DAIFMT_IB_NF);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops evm_ops = {
+ .hw_params = evm_hw_params,
+};
+
+/* davinci-evm machine dapm widgets */
+static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+/* davinci-evm machine audio_mapnections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Headphone connected to HPLOUT, HPROUT */
+ {"Headphone Jack", NULL, "HPLOUT"},
+ {"Headphone Jack", NULL, "HPROUT"},
+
+ /* Line Out connected to LLOUT, RLOUT */
+ {"Line Out", NULL, "LLOUT"},
+ {"Line Out", NULL, "RLOUT"},
+
+ /* Mic connected to (MIC3L | MIC3R) */
+ {"MIC3L", NULL, "Mic Bias 2V"},
+ {"MIC3R", NULL, "Mic Bias 2V"},
+ {"Mic Bias 2V", NULL, "Mic Jack"},
+
+ /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */
+ {"LINE1L", NULL, "Line In"},
+ {"LINE2L", NULL, "Line In"},
+ {"LINE1R", NULL, "Line In"},
+ {"LINE2R", NULL, "Line In"},
+};
+
+/* Logic for a aic3x as connected on a davinci-evm */
+static int evm_aic3x_init(struct snd_soc_codec *codec)
+{
+ /* Add davinci-evm specific widgets */
+ snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+ ARRAY_SIZE(aic3x_dapm_widgets));
+
+ /* Set up davinci-evm specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* not connected */
+ snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
+ snd_soc_dapm_disable_pin(codec, "HPLCOM");
+ snd_soc_dapm_disable_pin(codec, "HPRCOM");
+
+ /* always connected */
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Out");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+/* davinci-evm digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai = &davinci_i2s_dai,
+ .codec_dai = &aic3x_dai,
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
+/* davinci-evm audio machine driver */
+static struct snd_soc_machine snd_soc_machine_evm = {
+ .name = "DaVinci EVM",
+ .dai_link = &evm_dai,
+ .num_links = 1,
+};
+
+/* evm audio private data */
+static struct aic3x_setup_data evm_aic3x_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1b,
+};
+
+/* evm audio subsystem */
+static struct snd_soc_device evm_snd_devdata = {
+ .machine = &snd_soc_machine_evm,
+ .platform = &davinci_soc_platform,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &evm_aic3x_setup,
+};
+
+static struct resource evm_snd_resources[] = {
+ {
+ .start = DAVINCI_MCBSP_BASE,
+ .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+ .flags = IORESOURCE_MEM,
+ },
+};
+
+static struct evm_snd_platform_data evm_snd_data = {
+ .tx_dma_ch = DM644X_DMACH_MCBSP_TX,
+ .rx_dma_ch = DM644X_DMACH_MCBSP_RX,
+};
+
+static struct platform_device *evm_snd_device;
+
+static int __init evm_init(void)
+{
+ int ret;
+
+ evm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!evm_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
+ evm_snd_devdata.dev = &evm_snd_device->dev;
+ evm_snd_device->dev.platform_data = &evm_snd_data;
+
+ ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
+ ARRAY_SIZE(evm_snd_resources));
+ if (ret) {
+ platform_device_put(evm_snd_device);
+ return ret;
+ }
+
+ ret = platform_device_add(evm_snd_device);
+ if (ret)
+ platform_device_put(evm_snd_device);
+
+ return ret;
+}
+
+static void __exit evm_exit(void)
+{
+ platform_device_unregister(evm_snd_device);
+}
+
+module_init(evm_init);
+module_exit(evm_exit);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
new file mode 100644
index 0000000..abb5fed
--- /dev/null
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -0,0 +1,409 @@
+/*
+ * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "davinci-pcm.h"
+
+#define DAVINCI_MCBSP_DRR_REG 0x00
+#define DAVINCI_MCBSP_DXR_REG 0x04
+#define DAVINCI_MCBSP_SPCR_REG 0x08
+#define DAVINCI_MCBSP_RCR_REG 0x0c
+#define DAVINCI_MCBSP_XCR_REG 0x10
+#define DAVINCI_MCBSP_SRGR_REG 0x14
+#define DAVINCI_MCBSP_PCR_REG 0x24
+
+#define DAVINCI_MCBSP_SPCR_RRST (1 << 0)
+#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4)
+#define DAVINCI_MCBSP_SPCR_XRST (1 << 16)
+#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20)
+#define DAVINCI_MCBSP_SPCR_GRST (1 << 22)
+#define DAVINCI_MCBSP_SPCR_FRST (1 << 23)
+#define DAVINCI_MCBSP_SPCR_FREE (1 << 25)
+
+#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5)
+#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8)
+#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16)
+#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21)
+
+#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5)
+#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8)
+#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16)
+#define DAVINCI_MCBSP_XCR_XFIG (1 << 18)
+#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21)
+
+#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8)
+#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16)
+#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28)
+
+#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0)
+#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1)
+#define DAVINCI_MCBSP_PCR_FSRP (1 << 2)
+#define DAVINCI_MCBSP_PCR_FSXP (1 << 3)
+#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8)
+#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9)
+#define DAVINCI_MCBSP_PCR_FSRM (1 << 10)
+#define DAVINCI_MCBSP_PCR_FSXM (1 << 11)
+
+#define MOD_REG_BIT(val, mask, set) do { \
+ if (set) { \
+ val |= mask; \
+ } else { \
+ val &= ~mask; \
+ } \
+} while (0)
+
+enum {
+ DAVINCI_MCBSP_WORD_8 = 0,
+ DAVINCI_MCBSP_WORD_12,
+ DAVINCI_MCBSP_WORD_16,
+ DAVINCI_MCBSP_WORD_20,
+ DAVINCI_MCBSP_WORD_24,
+ DAVINCI_MCBSP_WORD_32,
+};
+
+static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
+ .name = "I2S PCM Stereo out",
+};
+
+static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
+ .name = "I2S PCM Stereo in",
+};
+
+struct davinci_mcbsp_dev {
+ void __iomem *base;
+ struct clk *clk;
+ struct davinci_pcm_dma_params *dma_params[2];
+};
+
+static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
+ int reg, u32 val)
+{
+ __raw_writel(val, dev->base + reg);
+}
+
+static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg)
+{
+ return __raw_readl(dev->base + reg);
+}
+
+static void davinci_mcbsp_start(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ u32 w;
+
+ /* Start the sample generator and enable transmitter/receiver */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
+ else
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+ /* Start frame sync */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+}
+
+static void davinci_mcbsp_stop(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ u32 w;
+
+ /* Reset transmitter/receiver and sample rate/frame sync generators */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST |
+ DAVINCI_MCBSP_SPCR_FRST, 0);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0);
+ else
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+}
+
+static int davinci_i2s_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+
+ cpu_dai->dma_data = dev->dma_params[substream->stream];
+
+ return 0;
+}
+
+static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
+ u32 w;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG,
+ DAVINCI_MCBSP_PCR_FSXM |
+ DAVINCI_MCBSP_PCR_FSRM |
+ DAVINCI_MCBSP_PCR_CLKXM |
+ DAVINCI_MCBSP_PCR_CLKRM);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG,
+ DAVINCI_MCBSP_SRGR_FSGM);
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_NF:
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
+ DAVINCI_MCBSP_PCR_CLKRP, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP |
+ DAVINCI_MCBSP_PCR_FSRP, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
+ DAVINCI_MCBSP_PCR_CLKRP |
+ DAVINCI_MCBSP_PCR_FSXP |
+ DAVINCI_MCBSP_PCR_FSRP, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ struct snd_interval *i = NULL;
+ int mcbsp_word_length;
+ u32 w;
+
+ /* general line settings */
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
+ DAVINCI_MCBSP_SPCR_RINTM(3) |
+ DAVINCI_MCBSP_SPCR_XINTM(3) |
+ DAVINCI_MCBSP_SPCR_FREE);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG,
+ DAVINCI_MCBSP_RCR_RFRLEN1(1) |
+ DAVINCI_MCBSP_RCR_RDATDLY(1));
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG,
+ DAVINCI_MCBSP_XCR_XFRLEN1(1) |
+ DAVINCI_MCBSP_XCR_XDATDLY(1) |
+ DAVINCI_MCBSP_XCR_XFIG);
+
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
+
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
+
+ /* Determine xfer data type */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ dma_params->data_type = 1;
+ mcbsp_word_length = DAVINCI_MCBSP_WORD_8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ dma_params->data_type = 2;
+ mcbsp_word_length = DAVINCI_MCBSP_WORD_16;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ dma_params->data_type = 4;
+ mcbsp_word_length = DAVINCI_MCBSP_WORD_32;
+ break;
+ default:
+ printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n");
+ return -EINVAL;
+ }
+
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
+
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+
+ return 0;
+}
+
+static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ davinci_mcbsp_start(substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ davinci_mcbsp_stop(substream);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static int davinci_i2s_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+ struct davinci_mcbsp_dev *dev;
+ struct resource *mem, *ioarea;
+ struct evm_snd_platform_data *pdata;
+ int ret;
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ return -ENODEV;
+ }
+
+ ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1,
+ pdev->name);
+ if (!ioarea) {
+ dev_err(&pdev->dev, "McBSP region already claimed\n");
+ return -EBUSY;
+ }
+
+ dev = kzalloc(sizeof(struct davinci_mcbsp_dev), GFP_KERNEL);
+ if (!dev) {
+ ret = -ENOMEM;
+ goto err_release_region;
+ }
+
+ cpu_dai->private_data = dev;
+
+ dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+ if (IS_ERR(dev->clk)) {
+ ret = -ENODEV;
+ goto err_free_mem;
+ }
+ clk_enable(dev->clk);
+
+ dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+ pdata = pdev->dev.platform_data;
+
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch;
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+ (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
+
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch;
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+ (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
+
+ return 0;
+
+err_free_mem:
+ kfree(dev);
+err_release_region:
+ release_mem_region(mem->start, (mem->end - mem->start) + 1);
+
+ return ret;
+}
+
+static void davinci_i2s_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+ struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
+ struct resource *mem;
+
+ clk_disable(dev->clk);
+ clk_put(dev->clk);
+ dev->clk = NULL;
+
+ kfree(dev);
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ release_mem_region(mem->start, (mem->end - mem->start) + 1);
+}
+
+#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
+
+struct snd_soc_dai davinci_i2s_dai = {
+ .name = "davinci-i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .probe = davinci_i2s_probe,
+ .remove = davinci_i2s_remove,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .startup = davinci_i2s_startup,
+ .trigger = davinci_i2s_trigger,
+ .hw_params = davinci_i2s_hw_params,},
+ .dai_ops = {
+ .set_fmt = davinci_i2s_set_dai_fmt,
+ },
+};
+EXPORT_SYMBOL_GPL(davinci_i2s_dai);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h
new file mode 100644
index 0000000..241648c
--- /dev/null
+++ b/sound/soc/davinci/davinci-i2s.h
@@ -0,0 +1,17 @@
+/*
+ * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _DAVINCI_I2S_H
+#define _DAVINCI_I2S_H
+
+extern struct snd_soc_dai davinci_i2s_dai;
+
+#endif
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
new file mode 100644
index 0000000..76feaa6
--- /dev/null
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -0,0 +1,389 @@
+/*
+ * ALSA PCM interface for the TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+
+#include "davinci-pcm.h"
+
+#define DAVINCI_PCM_DEBUG 0
+#if DAVINCI_PCM_DEBUG
+#define DPRINTK(x...) printk(KERN_DEBUG x)
+#else
+#define DPRINTK(x...)
+#endif
+
+static struct snd_pcm_hardware davinci_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+struct davinci_runtime_data {
+ spinlock_t lock;
+ int period; /* current DMA period */
+ int master_lch; /* Master DMA channel */
+ int slave_lch; /* Slave DMA channel */
+ struct davinci_pcm_dma_params *params; /* DMA params */
+};
+
+static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int lch = prtd->slave_lch;
+ unsigned int period_size;
+ unsigned int dma_offset;
+ dma_addr_t dma_pos;
+ dma_addr_t src, dst;
+ unsigned short src_bidx, dst_bidx;
+ unsigned int data_type;
+ unsigned int count;
+
+ period_size = snd_pcm_lib_period_bytes(substream);
+ dma_offset = prtd->period * period_size;
+ dma_pos = runtime->dma_addr + dma_offset;
+
+ DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x "
+ "period_size=%x\n", lch, dma_pos, period_size);
+
+ data_type = prtd->params->data_type;
+ count = period_size / data_type;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ src = dma_pos;
+ dst = prtd->params->dma_addr;
+ src_bidx = data_type;
+ dst_bidx = 0;
+ } else {
+ src = prtd->params->dma_addr;
+ dst = dma_pos;
+ src_bidx = 0;
+ dst_bidx = data_type;
+ }
+
+ davinci_set_dma_src_params(lch, src, INCR, W8BIT);
+ davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
+ davinci_set_dma_src_index(lch, src_bidx, 0);
+ davinci_set_dma_dest_index(lch, dst_bidx, 0);
+ davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+
+ prtd->period++;
+ if (unlikely(prtd->period >= runtime->periods))
+ prtd->period = 0;
+}
+
+static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+{
+ struct snd_pcm_substream *substream = data;
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+
+ DPRINTK("lch=%d, status=0x%x\n", lch, ch_status);
+
+ if (unlikely(ch_status != DMA_COMPLETE))
+ return;
+
+ if (snd_pcm_running(substream)) {
+ snd_pcm_period_elapsed(substream);
+
+ spin_lock(&prtd->lock);
+ davinci_pcm_enqueue_dma(substream);
+ spin_unlock(&prtd->lock);
+ }
+}
+
+static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
+ int tcc = TCC_ANY;
+ int ret;
+
+ if (!dma_data)
+ return -ENODEV;
+
+ prtd->params = dma_data;
+
+ /* Request master DMA channel */
+ ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+ davinci_pcm_dma_irq, substream,
+ &prtd->master_lch, &tcc, EVENTQ_0);
+ if (ret)
+ return ret;
+
+ /* Request slave DMA channel */
+ ret = davinci_request_dma(PARAM_ANY, "Link",
+ NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
+ if (ret) {
+ davinci_free_dma(prtd->master_lch);
+ return ret;
+ }
+
+ /* Link slave DMA channel in loopback */
+ davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+
+ return 0;
+}
+
+static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ spin_lock(&prtd->lock);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ davinci_start_dma(prtd->master_lch);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ davinci_stop_dma(prtd->master_lch);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ spin_unlock(&prtd->lock);
+
+ return ret;
+}
+
+static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct paramentry_descriptor temp;
+
+ prtd->period = 0;
+ davinci_pcm_enqueue_dma(substream);
+
+ /* Get slave channel dma params for master channel startup */
+ davinci_get_dma_params(prtd->slave_lch, &temp);
+ davinci_set_dma_params(prtd->master_lch, &temp);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+davinci_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct davinci_runtime_data *prtd = runtime->private_data;
+ unsigned int offset;
+ dma_addr_t count;
+ dma_addr_t src, dst;
+
+ spin_lock(&prtd->lock);
+
+ davinci_dma_getposition(prtd->master_lch, &src, &dst);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ count = src - runtime->dma_addr;
+ else
+ count = dst - runtime->dma_addr;;
+
+ spin_unlock(&prtd->lock);
+
+ offset = bytes_to_frames(runtime, count);
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+
+ return offset;
+}
+
+static int davinci_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct davinci_runtime_data *prtd;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
+
+ prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ spin_lock_init(&prtd->lock);
+
+ runtime->private_data = prtd;
+
+ ret = davinci_pcm_dma_request(substream);
+ if (ret) {
+ printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n");
+ kfree(prtd);
+ }
+
+ return ret;
+}
+
+static int davinci_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct davinci_runtime_data *prtd = runtime->private_data;
+
+ davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+
+ davinci_free_dma(prtd->slave_lch);
+ davinci_free_dma(prtd->master_lch);
+
+ kfree(prtd);
+
+ return 0;
+}
+
+static int davinci_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int davinci_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int davinci_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+struct snd_pcm_ops davinci_pcm_ops = {
+ .open = davinci_pcm_open,
+ .close = davinci_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = davinci_pcm_hw_params,
+ .hw_free = davinci_pcm_hw_free,
+ .prepare = davinci_pcm_prepare,
+ .trigger = davinci_pcm_trigger,
+ .pointer = davinci_pcm_pointer,
+ .mmap = davinci_pcm_mmap,
+};
+
+static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = davinci_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+
+ DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+ (void *) buf->area, (void *) buf->addr, size);
+
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void davinci_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static u64 davinci_pcm_dmamask = 0xffffffff;
+
+static int davinci_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+ int ret;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &davinci_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = davinci_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = davinci_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+struct snd_soc_platform davinci_soc_platform = {
+ .name = "davinci-audio",
+ .pcm_ops = &davinci_pcm_ops,
+ .pcm_new = davinci_pcm_new,
+ .pcm_free = davinci_pcm_free,
+};
+EXPORT_SYMBOL_GPL(davinci_soc_platform);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
new file mode 100644
index 0000000..62cb4eb
--- /dev/null
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -0,0 +1,29 @@
+/*
+ * ALSA PCM interface for the TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _DAVINCI_PCM_H
+#define _DAVINCI_PCM_H
+
+struct davinci_pcm_dma_params {
+ char *name; /* stream identifier */
+ int channel; /* sync dma channel ID */
+ dma_addr_t dma_addr; /* device physical address for DMA */
+ unsigned int data_type; /* xfer data type */
+};
+
+struct evm_snd_platform_data {
+ int tx_dma_ch;
+ int rx_dma_ch;
+};
+
+extern struct snd_soc_platform davinci_soc_platform;
+
+#endif
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
new file mode 100644
index 0000000..8d73edc
--- /dev/null
+++ b/sound/soc/fsl/Kconfig
@@ -0,0 +1,27 @@
+config SND_SOC_OF_SIMPLE
+ tristate
+
+config SND_SOC_MPC8610
+ bool "ALSA SoC support for the MPC8610 SOC"
+ depends on MPC8610_HPCD
+ default y if MPC8610
+ help
+ Say Y if you want to add support for codecs attached to the SSI
+ device on an MPC8610.
+
+config SND_SOC_MPC8610_HPCD
+ bool "ALSA SoC support for the Freescale MPC8610 HPCD board"
+ depends on SND_SOC_MPC8610
+ select SND_SOC_CS4270
+ select SND_SOC_CS4270_VD33_ERRATA
+ default y if MPC8610_HPCD
+ help
+ Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
+
+config SND_SOC_MPC5200_I2S
+ tristate "Freescale MPC5200 PSC in I2S mode driver"
+ depends on SND_SOC && PPC_MPC52xx && PPC_BESTCOMM
+ select SND_SOC_OF_SIMPLE
+ select PPC_BESTCOMM_GEN_BD
+ help
+ Say Y here to support the MPC5200 PSCs in I2S mode.
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
new file mode 100644
index 0000000..035da4a
--- /dev/null
+++ b/sound/soc/fsl/Makefile
@@ -0,0 +1,11 @@
+# Simple machine driver that extracts configuration from the OF device tree
+obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o
+
+# MPC8610 HPCD Machine Support
+obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
+
+# MPC8610 Platform Support
+obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+
+obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
new file mode 100644
index 0000000..d2d3da9
--- /dev/null
+++ b/sound/soc/fsl/fsl_dma.c
@@ -0,0 +1,858 @@
+/*
+ * Freescale DMA ALSA SoC PCM driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
+ * under the terms of the GNU General Public License version 2. This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
+ *
+ * This driver implements ASoC support for the Elo DMA controller, which is
+ * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms,
+ * the PCM driver is what handles the DMA buffer.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/io.h>
+
+#include "fsl_dma.h"
+
+/*
+ * The formats that the DMA controller supports, which is anything
+ * that is 8, 16, or 32 bits.
+ */
+#define FSLDMA_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_U16_BE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_U24_LE | \
+ SNDRV_PCM_FMTBIT_U24_BE | \
+ SNDRV_PCM_FMTBIT_S32_LE | \
+ SNDRV_PCM_FMTBIT_S32_BE | \
+ SNDRV_PCM_FMTBIT_U32_LE | \
+ SNDRV_PCM_FMTBIT_U32_BE)
+
+#define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
+ SNDRV_PCM_RATE_CONTINUOUS)
+
+/* DMA global data. This structure is used by fsl_dma_open() to determine
+ * which DMA channels to assign to a substream. Unfortunately, ASoC V1 does
+ * not allow the machine driver to provide this information to the PCM
+ * driver in advance, and there's no way to differentiate between the two
+ * DMA controllers. So for now, this driver only supports one SSI device
+ * using two DMA channels. We cannot support multiple DMA devices.
+ *
+ * ssi_stx_phys: bus address of SSI STX register
+ * ssi_srx_phys: bus address of SSI SRX register
+ * dma_channel: pointer to the DMA channel's registers
+ * irq: IRQ for this DMA channel
+ * assigned: set to 1 if that DMA channel is assigned to a substream
+ */
+static struct {
+ dma_addr_t ssi_stx_phys;
+ dma_addr_t ssi_srx_phys;
+ struct ccsr_dma_channel __iomem *dma_channel[2];
+ unsigned int irq[2];
+ unsigned int assigned[2];
+} dma_global_data;
+
+/*
+ * The number of DMA links to use. Two is the bare minimum, but if you
+ * have really small links you might need more.
+ */
+#define NUM_DMA_LINKS 2
+
+/** fsl_dma_private: p-substream DMA data
+ *
+ * Each substream has a 1-to-1 association with a DMA channel.
+ *
+ * The link[] array is first because it needs to be aligned on a 32-byte
+ * boundary, so putting it first will ensure alignment without padding the
+ * structure.
+ *
+ * @link[]: array of link descriptors
+ * @controller_id: which DMA controller (0, 1, ...)
+ * @channel_id: which DMA channel on the controller (0, 1, 2, ...)
+ * @dma_channel: pointer to the DMA channel's registers
+ * @irq: IRQ for this DMA channel
+ * @substream: pointer to the substream object, needed by the ISR
+ * @ssi_sxx_phys: bus address of the STX or SRX register to use
+ * @ld_buf_phys: physical address of the LD buffer
+ * @current_link: index into link[] of the link currently being processed
+ * @dma_buf_phys: physical address of the DMA buffer
+ * @dma_buf_next: physical address of the next period to process
+ * @dma_buf_end: physical address of the byte after the end of the DMA
+ * @buffer period_size: the size of a single period
+ * @num_periods: the number of periods in the DMA buffer
+ */
+struct fsl_dma_private {
+ struct fsl_dma_link_descriptor link[NUM_DMA_LINKS];
+ unsigned int controller_id;
+ unsigned int channel_id;
+ struct ccsr_dma_channel __iomem *dma_channel;
+ unsigned int irq;
+ struct snd_pcm_substream *substream;
+ dma_addr_t ssi_sxx_phys;
+ dma_addr_t ld_buf_phys;
+ unsigned int current_link;
+ dma_addr_t dma_buf_phys;
+ dma_addr_t dma_buf_next;
+ dma_addr_t dma_buf_end;
+ size_t period_size;
+ unsigned int num_periods;
+};
+
+/**
+ * fsl_dma_hardare: define characteristics of the PCM hardware.
+ *
+ * The PCM hardware is the Freescale DMA controller. This structure defines
+ * the capabilities of that hardware.
+ *
+ * Since the sampling rate and data format are not controlled by the DMA
+ * controller, we specify no limits for those values. The only exception is
+ * period_bytes_min, which is set to a reasonably low value to prevent the
+ * DMA controller from generating too many interrupts per second.
+ *
+ * Since each link descriptor has a 32-bit byte count field, we set
+ * period_bytes_max to the largest 32-bit number. We also have no maximum
+ * number of periods.
+ *
+ * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
+ * limitation in the SSI driver requires the sample rates for playback and
+ * capture to be the same.
+ */
+static const struct snd_pcm_hardware fsl_dma_hardware = {
+
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_JOINT_DUPLEX,
+ .formats = FSLDMA_PCM_FORMATS,
+ .rates = FSLDMA_PCM_RATES,
+ .rate_min = 5512,
+ .rate_max = 192000,
+ .period_bytes_min = 512, /* A reasonable limit */
+ .period_bytes_max = (u32) -1,
+ .periods_min = NUM_DMA_LINKS,
+ .periods_max = (unsigned int) -1,
+ .buffer_bytes_max = 128 * 1024, /* A reasonable limit */
+};
+
+/**
+ * fsl_dma_abort_stream: tell ALSA that the DMA transfer has aborted
+ *
+ * This function should be called by the ISR whenever the DMA controller
+ * halts data transfer.
+ */
+static void fsl_dma_abort_stream(struct snd_pcm_substream *substream)
+{
+ unsigned long flags;
+
+ snd_pcm_stream_lock_irqsave(substream, flags);
+
+ if (snd_pcm_running(substream))
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+
+ snd_pcm_stream_unlock_irqrestore(substream, flags);
+}
+
+/**
+ * fsl_dma_update_pointers - update LD pointers to point to the next period
+ *
+ * As each period is completed, this function changes the the link
+ * descriptor pointers for that period to point to the next period.
+ */
+static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private)
+{
+ struct fsl_dma_link_descriptor *link =
+ &dma_private->link[dma_private->current_link];
+
+ /* Update our link descriptors to point to the next period */
+ if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ link->source_addr =
+ cpu_to_be32(dma_private->dma_buf_next);
+ else
+ link->dest_addr =
+ cpu_to_be32(dma_private->dma_buf_next);
+
+ /* Update our variables for next time */
+ dma_private->dma_buf_next += dma_private->period_size;
+
+ if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
+ dma_private->dma_buf_next = dma_private->dma_buf_phys;
+
+ if (++dma_private->current_link >= NUM_DMA_LINKS)
+ dma_private->current_link = 0;
+}
+
+/**
+ * fsl_dma_isr: interrupt handler for the DMA controller
+ *
+ * @irq: IRQ of the DMA channel
+ * @dev_id: pointer to the dma_private structure for this DMA channel
+ */
+static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
+{
+ struct fsl_dma_private *dma_private = dev_id;
+ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
+ irqreturn_t ret = IRQ_NONE;
+ u32 sr, sr2 = 0;
+
+ /* We got an interrupt, so read the status register to see what we
+ were interrupted for.
+ */
+ sr = in_be32(&dma_channel->sr);
+
+ if (sr & CCSR_DMA_SR_TE) {
+ dev_err(dma_private->substream->pcm->card->dev,
+ "DMA transmit error (controller=%u channel=%u irq=%u\n",
+ dma_private->controller_id,
+ dma_private->channel_id, irq);
+ fsl_dma_abort_stream(dma_private->substream);
+ sr2 |= CCSR_DMA_SR_TE;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_CH)
+ ret = IRQ_HANDLED;
+
+ if (sr & CCSR_DMA_SR_PE) {
+ dev_err(dma_private->substream->pcm->card->dev,
+ "DMA%u programming error (channel=%u irq=%u)\n",
+ dma_private->controller_id,
+ dma_private->channel_id, irq);
+ fsl_dma_abort_stream(dma_private->substream);
+ sr2 |= CCSR_DMA_SR_PE;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_EOLNI) {
+ sr2 |= CCSR_DMA_SR_EOLNI;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_CB)
+ ret = IRQ_HANDLED;
+
+ if (sr & CCSR_DMA_SR_EOSI) {
+ struct snd_pcm_substream *substream = dma_private->substream;
+
+ /* Tell ALSA we completed a period. */
+ snd_pcm_period_elapsed(substream);
+
+ /*
+ * Update our link descriptors to point to the next period. We
+ * only need to do this if the number of periods is not equal to
+ * the number of links.
+ */
+ if (dma_private->num_periods != NUM_DMA_LINKS)
+ fsl_dma_update_pointers(dma_private);
+
+ sr2 |= CCSR_DMA_SR_EOSI;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sr & CCSR_DMA_SR_EOLSI) {
+ sr2 |= CCSR_DMA_SR_EOLSI;
+ ret = IRQ_HANDLED;
+ }
+
+ /* Clear the bits that we set */
+ if (sr2)
+ out_be32(&dma_channel->sr, sr2);
+
+ return ret;
+}
+
+/**
+ * fsl_dma_new: initialize this PCM driver.
+ *
+ * This function is called when the codec driver calls snd_soc_new_pcms(),
+ * once for each .dai_link in the machine driver's snd_soc_machine
+ * structure.
+ */
+static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ static u64 fsl_dma_dmamask = DMA_BIT_MASK(32);
+ int ret;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &fsl_dma_dmamask;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = fsl_dma_dmamask;
+
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev,
+ fsl_dma_hardware.buffer_bytes_max,
+ &pcm->streams[0].substream->dma_buffer);
+ if (ret) {
+ dev_err(card->dev,
+ "Can't allocate playback DMA buffer (size=%u)\n",
+ fsl_dma_hardware.buffer_bytes_max);
+ return -ENOMEM;
+ }
+
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev,
+ fsl_dma_hardware.buffer_bytes_max,
+ &pcm->streams[1].substream->dma_buffer);
+ if (ret) {
+ snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+ dev_err(card->dev,
+ "Can't allocate capture DMA buffer (size=%u)\n",
+ fsl_dma_hardware.buffer_bytes_max);
+ return -ENOMEM;
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_dma_open: open a new substream.
+ *
+ * Each substream has its own DMA buffer.
+ *
+ * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link
+ * descriptors that ping-pong from one period to the next. For example, if
+ * there are six periods and two link descriptors, this is how they look
+ * before playback starts:
+ *
+ * The last link descriptor
+ * ____________ points back to the first
+ * | |
+ * V |
+ * ___ ___ |
+ * | |->| |->|
+ * |___| |___|
+ * | |
+ * | |
+ * V V
+ * _________________________________________
+ * | | | | | | | The DMA buffer is
+ * | | | | | | | divided into 6 parts
+ * |______|______|______|______|______|______|
+ *
+ * and here's how they look after the first period is finished playing:
+ *
+ * ____________
+ * | |
+ * V |
+ * ___ ___ |
+ * | |->| |->|
+ * |___| |___|
+ * | |
+ * |______________
+ * | |
+ * V V
+ * _________________________________________
+ * | | | | | | |
+ * | | | | | | |
+ * |______|______|______|______|______|______|
+ *
+ * The first link descriptor now points to the third period. The DMA
+ * controller is currently playing the second period. When it finishes, it
+ * will jump back to the first descriptor and play the third period.
+ *
+ * There are four reasons we do this:
+ *
+ * 1. The only way to get the DMA controller to automatically restart the
+ * transfer when it gets to the end of the buffer is to use chaining
+ * mode. Basic direct mode doesn't offer that feature.
+ * 2. We need to receive an interrupt at the end of every period. The DMA
+ * controller can generate an interrupt at the end of every link transfer
+ * (aka segment). Making each period into a DMA segment will give us the
+ * interrupts we need.
+ * 3. By creating only two link descriptors, regardless of the number of
+ * periods, we do not need to reallocate the link descriptors if the
+ * number of periods changes.
+ * 4. All of the audio data is still stored in a single, contiguous DMA
+ * buffer, which is what ALSA expects. We're just dividing it into
+ * contiguous parts, and creating a link descriptor for each one.
+ */
+static int fsl_dma_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private;
+ struct ccsr_dma_channel __iomem *dma_channel;
+ dma_addr_t ld_buf_phys;
+ u64 temp_link; /* Pointer to next link descriptor */
+ u32 mr;
+ unsigned int channel;
+ int ret = 0;
+ unsigned int i;
+
+ /*
+ * Reject any DMA buffer whose size is not a multiple of the period
+ * size. We need to make sure that the DMA buffer can be evenly divided
+ * into periods.
+ */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ dev_err(substream->pcm->card->dev, "invalid buffer size\n");
+ return ret;
+ }
+
+ channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ if (dma_global_data.assigned[channel]) {
+ dev_err(substream->pcm->card->dev,
+ "DMA channel already assigned\n");
+ return -EBUSY;
+ }
+
+ dma_private = dma_alloc_coherent(substream->pcm->dev,
+ sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL);
+ if (!dma_private) {
+ dev_err(substream->pcm->card->dev,
+ "can't allocate DMA private data\n");
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys;
+ else
+ dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys;
+
+ dma_private->dma_channel = dma_global_data.dma_channel[channel];
+ dma_private->irq = dma_global_data.irq[channel];
+ dma_private->substream = substream;
+ dma_private->ld_buf_phys = ld_buf_phys;
+ dma_private->dma_buf_phys = substream->dma_buffer.addr;
+
+ /* We only support one DMA controller for now */
+ dma_private->controller_id = 0;
+ dma_private->channel_id = channel;
+
+ ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private);
+ if (ret) {
+ dev_err(substream->pcm->card->dev,
+ "can't register ISR for IRQ %u (ret=%i)\n",
+ dma_private->irq, ret);
+ dma_free_coherent(substream->pcm->dev,
+ sizeof(struct fsl_dma_private),
+ dma_private, dma_private->ld_buf_phys);
+ return ret;
+ }
+
+ dma_global_data.assigned[channel] = 1;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
+ runtime->private_data = dma_private;
+
+ /* Program the fixed DMA controller parameters */
+
+ dma_channel = dma_private->dma_channel;
+
+ temp_link = dma_private->ld_buf_phys +
+ sizeof(struct fsl_dma_link_descriptor);
+
+ for (i = 0; i < NUM_DMA_LINKS; i++) {
+ struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+ link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+ link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+ link->next = cpu_to_be64(temp_link);
+
+ temp_link += sizeof(struct fsl_dma_link_descriptor);
+ }
+ /* The last link descriptor points to the first */
+ dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
+
+ /* Tell the DMA controller where the first link descriptor is */
+ out_be32(&dma_channel->clndar,
+ CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
+ out_be32(&dma_channel->eclndar,
+ CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
+
+ /* The manual says the BCR must be clear before enabling EMP */
+ out_be32(&dma_channel->bcr, 0);
+
+ /*
+ * Program the mode register for interrupts, external master control,
+ * and source/destination hold. Also clear the Channel Abort bit.
+ */
+ mr = in_be32(&dma_channel->mr) &
+ ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
+
+ /*
+ * We want External Master Start and External Master Pause enabled,
+ * because the SSI is controlling the DMA controller. We want the DMA
+ * controller to be set up in advance, and then we signal only the SSI
+ * to start transferring.
+ *
+ * We want End-Of-Segment Interrupts enabled, because this will generate
+ * an interrupt at the end of each segment (each link descriptor
+ * represents one segment). Each DMA segment is the same thing as an
+ * ALSA period, so this is how we get an interrupt at the end of every
+ * period.
+ *
+ * We want Error Interrupt enabled, so that we can get an error if
+ * the DMA controller is mis-programmed somehow.
+ */
+ mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
+ CCSR_DMA_MR_EMS_EN;
+
+ /* For playback, we want the destination address to be held. For
+ capture, set the source address to be held. */
+ mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
+
+ out_be32(&dma_channel->mr, mr);
+
+ return 0;
+}
+
+/**
+ * fsl_dma_hw_params: continue initializing the DMA links
+ *
+ * This function obtains hardware parameters about the opened stream and
+ * programs the DMA controller accordingly.
+ *
+ * Note that due to a quirk of the SSI's STX register, the target address
+ * for the DMA operations depends on the sample size. So we don't program
+ * the dest_addr (for playback -- source_addr for capture) fields in the
+ * link descriptors here. We do that in fsl_dma_prepare()
+ */
+static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+
+ dma_addr_t temp_addr; /* Pointer to next period */
+
+ unsigned int i;
+
+ /* Get all the parameters we need */
+ size_t buffer_size = params_buffer_bytes(hw_params);
+ size_t period_size = params_period_bytes(hw_params);
+
+ /* Initialize our DMA tracking variables */
+ dma_private->period_size = period_size;
+ dma_private->num_periods = params_periods(hw_params);
+ dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
+ dma_private->dma_buf_next = dma_private->dma_buf_phys +
+ (NUM_DMA_LINKS * period_size);
+ if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
+ dma_private->dma_buf_next = dma_private->dma_buf_phys;
+
+ /*
+ * The actual address in STX0 (destination for playback, source for
+ * capture) is based on the sample size, but we don't know the sample
+ * size in this function, so we'll have to adjust that later. See
+ * comments in fsl_dma_prepare().
+ *
+ * The DMA controller does not have a cache, so the CPU does not
+ * need to tell it to flush its cache. However, the DMA
+ * controller does need to tell the CPU to flush its cache.
+ * That's what the SNOOP bit does.
+ *
+ * Also, even though the DMA controller supports 36-bit addressing, for
+ * simplicity we currently support only 32-bit addresses for the audio
+ * buffer itself.
+ */
+ temp_addr = substream->dma_buffer.addr;
+
+ for (i = 0; i < NUM_DMA_LINKS; i++) {
+ struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+ link->count = cpu_to_be32(period_size);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ link->source_addr = cpu_to_be32(temp_addr);
+ else
+ link->dest_addr = cpu_to_be32(temp_addr);
+
+ temp_addr += period_size;
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_dma_prepare - prepare the DMA registers for playback.
+ *
+ * This function is called after the specifics of the audio data are known,
+ * i.e. snd_pcm_runtime is initialized.
+ *
+ * In this function, we finish programming the registers of the DMA
+ * controller that are dependent on the sample size.
+ *
+ * One of the drawbacks with big-endian is that when copying integers of
+ * different sizes to a fixed-sized register, the address to which the
+ * integer must be copied is dependent on the size of the integer.
+ *
+ * For example, if P is the address of a 32-bit register, and X is a 32-bit
+ * integer, then X should be copied to address P. However, if X is a 16-bit
+ * integer, then it should be copied to P+2. If X is an 8-bit register,
+ * then it should be copied to P+3.
+ *
+ * So for playback of 8-bit samples, the DMA controller must transfer single
+ * bytes from the DMA buffer to the last byte of the STX0 register, i.e.
+ * offset by 3 bytes. For 16-bit samples, the offset is two bytes.
+ *
+ * For 24-bit samples, the offset is 1 byte. However, the DMA controller
+ * does not support 3-byte copies (the DAHTS register supports only 1, 2, 4,
+ * and 8 bytes at a time). So we do not support packed 24-bit samples.
+ * 24-bit data must be padded to 32 bits.
+ */
+static int fsl_dma_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
+ u32 mr;
+ unsigned int i;
+ dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */
+ unsigned int frame_size; /* Number of bytes per frame */
+
+ ssi_sxx_phys = dma_private->ssi_sxx_phys;
+
+ mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK |
+ CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK);
+
+ switch (runtime->sample_bits) {
+ case 8:
+ mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
+ ssi_sxx_phys += 3;
+ break;
+ case 16:
+ mr |= CCSR_DMA_MR_DAHTS_2 | CCSR_DMA_MR_SAHTS_2;
+ ssi_sxx_phys += 2;
+ break;
+ case 32:
+ mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4;
+ break;
+ default:
+ dev_err(substream->pcm->card->dev,
+ "unsupported sample size %u\n", runtime->sample_bits);
+ return -EINVAL;
+ }
+
+ frame_size = runtime->frame_bits / 8;
+ /*
+ * BWC should always be a multiple of the frame size. BWC determines
+ * how many bytes are sent/received before the DMA controller checks the
+ * SSI to see if it needs to stop. For playback, the transmit FIFO can
+ * hold three frames, so we want to send two frames at a time. For
+ * capture, the receive FIFO is triggered when it contains one frame, so
+ * we want to receive one frame at a time.
+ */
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mr |= CCSR_DMA_MR_BWC(2 * frame_size);
+ else
+ mr |= CCSR_DMA_MR_BWC(frame_size);
+
+ out_be32(&dma_channel->mr, mr);
+
+ /*
+ * Program the address of the DMA transfer to/from the SSI.
+ */
+ for (i = 0; i < NUM_DMA_LINKS; i++) {
+ struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ link->dest_addr = cpu_to_be32(ssi_sxx_phys);
+ else
+ link->source_addr = cpu_to_be32(ssi_sxx_phys);
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_dma_pointer: determine the current position of the DMA transfer
+ *
+ * This function is called by ALSA when ALSA wants to know where in the
+ * stream buffer the hardware currently is.
+ *
+ * For playback, the SAR register contains the physical address of the most
+ * recent DMA transfer. For capture, the value is in the DAR register.
+ *
+ * The base address of the buffer is stored in the source_addr field of the
+ * first link descriptor.
+ */
+static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
+ dma_addr_t position;
+ snd_pcm_uframes_t frames;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ position = in_be32(&dma_channel->sar);
+ else
+ position = in_be32(&dma_channel->dar);
+
+ frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
+
+ /*
+ * If the current address is just past the end of the buffer, wrap it
+ * around.
+ */
+ if (frames == runtime->buffer_size)
+ frames = 0;
+
+ return frames;
+}
+
+/**
+ * fsl_dma_hw_free: release resources allocated in fsl_dma_hw_params()
+ *
+ * Release the resources allocated in fsl_dma_hw_params() and de-program the
+ * registers.
+ *
+ * This function can be called multiple times.
+ */
+static int fsl_dma_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+
+ if (dma_private) {
+ struct ccsr_dma_channel __iomem *dma_channel;
+
+ dma_channel = dma_private->dma_channel;
+
+ /* Stop the DMA */
+ out_be32(&dma_channel->mr, CCSR_DMA_MR_CA);
+ out_be32(&dma_channel->mr, 0);
+
+ /* Reset all the other registers */
+ out_be32(&dma_channel->sr, -1);
+ out_be32(&dma_channel->clndar, 0);
+ out_be32(&dma_channel->eclndar, 0);
+ out_be32(&dma_channel->satr, 0);
+ out_be32(&dma_channel->sar, 0);
+ out_be32(&dma_channel->datr, 0);
+ out_be32(&dma_channel->dar, 0);
+ out_be32(&dma_channel->bcr, 0);
+ out_be32(&dma_channel->nlndar, 0);
+ out_be32(&dma_channel->enlndar, 0);
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_dma_close: close the stream.
+ */
+static int fsl_dma_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct fsl_dma_private *dma_private = runtime->private_data;
+ int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ if (dma_private) {
+ if (dma_private->irq)
+ free_irq(dma_private->irq, dma_private);
+
+ if (dma_private->ld_buf_phys) {
+ dma_unmap_single(substream->pcm->dev,
+ dma_private->ld_buf_phys,
+ sizeof(dma_private->link), DMA_TO_DEVICE);
+ }
+
+ /* Deallocate the fsl_dma_private structure */
+ dma_free_coherent(substream->pcm->dev,
+ sizeof(struct fsl_dma_private),
+ dma_private, dma_private->ld_buf_phys);
+ substream->runtime->private_data = NULL;
+ }
+
+ dma_global_data.assigned[dir] = 0;
+
+ return 0;
+}
+
+/*
+ * Remove this PCM driver.
+ */
+static void fsl_dma_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ substream = pcm->streams[i].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+static struct snd_pcm_ops fsl_dma_ops = {
+ .open = fsl_dma_open,
+ .close = fsl_dma_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = fsl_dma_hw_params,
+ .hw_free = fsl_dma_hw_free,
+ .prepare = fsl_dma_prepare,
+ .pointer = fsl_dma_pointer,
+};
+
+struct snd_soc_platform fsl_soc_platform = {
+ .name = "fsl-dma",
+ .pcm_ops = &fsl_dma_ops,
+ .pcm_new = fsl_dma_new,
+ .pcm_free = fsl_dma_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(fsl_soc_platform);
+
+/**
+ * fsl_dma_configure: store the DMA parameters from the fabric driver.
+ *
+ * This function is called by the ASoC fabric driver to give us the DMA and
+ * SSI channel information.
+ *
+ * Unfortunately, ASoC V1 does make it possible to determine the DMA/SSI
+ * data when a substream is created, so for now we need to store this data
+ * into a global variable. This means that we can only support one DMA
+ * controller, and hence only one SSI.
+ */
+int fsl_dma_configure(struct fsl_dma_info *dma_info)
+{
+ static int initialized;
+
+ /* We only support one DMA controller for now */
+ if (initialized)
+ return 0;
+
+ dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys;
+ dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys;
+ dma_global_data.dma_channel[0] = dma_info->dma_channel[0];
+ dma_global_data.dma_channel[1] = dma_info->dma_channel[1];
+ dma_global_data.irq[0] = dma_info->dma_irq[0];
+ dma_global_data.irq[1] = dma_info->dma_irq[1];
+ dma_global_data.assigned[0] = 0;
+ dma_global_data.assigned[1] = 0;
+
+ initialized = 1;
+ return 1;
+}
+EXPORT_SYMBOL_GPL(fsl_dma_configure);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h
new file mode 100644
index 0000000..385d4a4
--- /dev/null
+++ b/sound/soc/fsl/fsl_dma.h
@@ -0,0 +1,149 @@
+/*
+ * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _MPC8610_PCM_H
+#define _MPC8610_PCM_H
+
+struct ccsr_dma {
+ u8 res0[0x100];
+ struct ccsr_dma_channel {
+ __be32 mr; /* Mode register */
+ __be32 sr; /* Status register */
+ __be32 eclndar; /* Current link descriptor extended addr reg */
+ __be32 clndar; /* Current link descriptor address register */
+ __be32 satr; /* Source attributes register */
+ __be32 sar; /* Source address register */
+ __be32 datr; /* Destination attributes register */
+ __be32 dar; /* Destination address register */
+ __be32 bcr; /* Byte count register */
+ __be32 enlndar; /* Next link descriptor extended address reg */
+ __be32 nlndar; /* Next link descriptor address register */
+ u8 res1[4];
+ __be32 eclsdar; /* Current list descriptor extended addr reg */
+ __be32 clsdar; /* Current list descriptor address register */
+ __be32 enlsdar; /* Next list descriptor extended address reg */
+ __be32 nlsdar; /* Next list descriptor address register */
+ __be32 ssr; /* Source stride register */
+ __be32 dsr; /* Destination stride register */
+ u8 res2[0x38];
+ } channel[4];
+ __be32 dgsr;
+};
+
+#define CCSR_DMA_MR_BWC_DISABLED 0x0F000000
+#define CCSR_DMA_MR_BWC_SHIFT 24
+#define CCSR_DMA_MR_BWC_MASK 0x0F000000
+#define CCSR_DMA_MR_BWC(x) \
+ ((ilog2(x) << CCSR_DMA_MR_BWC_SHIFT) & CCSR_DMA_MR_BWC_MASK)
+#define CCSR_DMA_MR_EMP_EN 0x00200000
+#define CCSR_DMA_MR_EMS_EN 0x00040000
+#define CCSR_DMA_MR_DAHTS_MASK 0x00030000
+#define CCSR_DMA_MR_DAHTS_1 0x00000000
+#define CCSR_DMA_MR_DAHTS_2 0x00010000
+#define CCSR_DMA_MR_DAHTS_4 0x00020000
+#define CCSR_DMA_MR_DAHTS_8 0x00030000
+#define CCSR_DMA_MR_SAHTS_MASK 0x0000C000
+#define CCSR_DMA_MR_SAHTS_1 0x00000000
+#define CCSR_DMA_MR_SAHTS_2 0x00004000
+#define CCSR_DMA_MR_SAHTS_4 0x00008000
+#define CCSR_DMA_MR_SAHTS_8 0x0000C000
+#define CCSR_DMA_MR_DAHE 0x00002000
+#define CCSR_DMA_MR_SAHE 0x00001000
+#define CCSR_DMA_MR_SRW 0x00000400
+#define CCSR_DMA_MR_EOSIE 0x00000200
+#define CCSR_DMA_MR_EOLNIE 0x00000100
+#define CCSR_DMA_MR_EOLSIE 0x00000080
+#define CCSR_DMA_MR_EIE 0x00000040
+#define CCSR_DMA_MR_XFE 0x00000020
+#define CCSR_DMA_MR_CDSM_SWSM 0x00000010
+#define CCSR_DMA_MR_CA 0x00000008
+#define CCSR_DMA_MR_CTM 0x00000004
+#define CCSR_DMA_MR_CC 0x00000002
+#define CCSR_DMA_MR_CS 0x00000001
+
+#define CCSR_DMA_SR_TE 0x00000080
+#define CCSR_DMA_SR_CH 0x00000020
+#define CCSR_DMA_SR_PE 0x00000010
+#define CCSR_DMA_SR_EOLNI 0x00000008
+#define CCSR_DMA_SR_CB 0x00000004
+#define CCSR_DMA_SR_EOSI 0x00000002
+#define CCSR_DMA_SR_EOLSI 0x00000001
+
+/* ECLNDAR takes bits 32-36 of the CLNDAR register */
+static inline u32 CCSR_DMA_ECLNDAR_ADDR(u64 x)
+{
+ return (x >> 32) & 0xf;
+}
+
+#define CCSR_DMA_CLNDAR_ADDR(x) ((x) & 0xFFFFFFFE)
+#define CCSR_DMA_CLNDAR_EOSIE 0x00000008
+
+/* SATR and DATR, combined */
+#define CCSR_DMA_ATR_PBATMU 0x20000000
+#define CCSR_DMA_ATR_TFLOWLVL_0 0x00000000
+#define CCSR_DMA_ATR_TFLOWLVL_1 0x06000000
+#define CCSR_DMA_ATR_TFLOWLVL_2 0x08000000
+#define CCSR_DMA_ATR_TFLOWLVL_3 0x0C000000
+#define CCSR_DMA_ATR_PCIORDER 0x02000000
+#define CCSR_DMA_ATR_SME 0x01000000
+#define CCSR_DMA_ATR_NOSNOOP 0x00040000
+#define CCSR_DMA_ATR_SNOOP 0x00050000
+#define CCSR_DMA_ATR_ESAD_MASK 0x0000000F
+
+/**
+ * List Descriptor for extended chaining mode DMA operations.
+ *
+ * The CLSDAR register points to the first (in a linked-list) List
+ * Descriptor. Each object must be aligned on a 32-byte boundary. Each
+ * list descriptor points to a linked-list of link Descriptors.
+ */
+struct fsl_dma_list_descriptor {
+ __be64 next; /* Address of next list descriptor */
+ __be64 first_link; /* Address of first link descriptor */
+ __be32 source; /* Source stride */
+ __be32 dest; /* Destination stride */
+ u8 res[8]; /* Reserved */
+} __attribute__ ((aligned(32), packed));
+
+/**
+ * Link Descriptor for basic and extended chaining mode DMA operations.
+ *
+ * A Link Descriptor points to a single DMA buffer. Each link descriptor
+ * must be aligned on a 32-byte boundary.
+ */
+struct fsl_dma_link_descriptor {
+ __be32 source_attr; /* Programmed into SATR register */
+ __be32 source_addr; /* Programmed into SAR register */
+ __be32 dest_attr; /* Programmed into DATR register */
+ __be32 dest_addr; /* Programmed into DAR register */
+ __be64 next; /* Address of next link descriptor */
+ __be32 count; /* Byte count */
+ u8 res[4]; /* Reserved */
+} __attribute__ ((aligned(32), packed));
+
+/* DMA information needed to create a snd_soc_dai object
+ *
+ * ssi_stx_phys: bus address of SSI STX register to use
+ * ssi_srx_phys: bus address of SSI SRX register to use
+ * dma[0]: points to the DMA channel to use for playback
+ * dma[1]: points to the DMA channel to use for capture
+ * dma_irq[0]: IRQ of the DMA channel to use for playback
+ * dma_irq[1]: IRQ of the DMA channel to use for capture
+ */
+struct fsl_dma_info {
+ dma_addr_t ssi_stx_phys;
+ dma_addr_t ssi_srx_phys;
+ struct ccsr_dma_channel __iomem *dma_channel[2];
+ unsigned int dma_irq[2];
+};
+
+extern struct snd_soc_platform fsl_soc_platform;
+
+int fsl_dma_configure(struct fsl_dma_info *dma_info);
+
+#endif
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
new file mode 100644
index 0000000..157a789
--- /dev/null
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -0,0 +1,697 @@
+/*
+ * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
+ * under the terms of the GNU General Public License version 2. This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/immap_86xx.h>
+
+#include "fsl_ssi.h"
+
+/**
+ * FSLSSI_I2S_RATES: sample rates supported by the I2S
+ *
+ * This driver currently only supports the SSI running in I2S slave mode,
+ * which means the codec determines the sample rate. Therefore, we tell
+ * ALSA that we support all rates and let the codec driver decide what rates
+ * are really supported.
+ */
+#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
+ SNDRV_PCM_RATE_CONTINUOUS)
+
+/**
+ * FSLSSI_I2S_FORMATS: audio formats supported by the SSI
+ *
+ * This driver currently only supports the SSI running in I2S slave mode.
+ *
+ * The SSI has a limitation in that the samples must be in the same byte
+ * order as the host CPU. This is because when multiple bytes are written
+ * to the STX register, the bytes and bits must be written in the same
+ * order. The STX is a shift register, so all the bits need to be aligned
+ * (bit-endianness must match byte-endianness). Processors typically write
+ * the bits within a byte in the same order that the bytes of a word are
+ * written in. So if the host CPU is big-endian, then only big-endian
+ * samples will be written to STX properly.
+ */
+#ifdef __BIG_ENDIAN
+#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_S20_3BE | \
+ SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S24_BE)
+#else
+#define FSLSSI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+#endif
+
+/**
+ * fsl_ssi_private: per-SSI private data
+ *
+ * @name: short name for this device ("SSI0", "SSI1", etc)
+ * @ssi: pointer to the SSI's registers
+ * @ssi_phys: physical address of the SSI registers
+ * @irq: IRQ of this SSI
+ * @first_stream: pointer to the stream that was opened first
+ * @second_stream: pointer to second stream
+ * @dev: struct device pointer
+ * @playback: the number of playback streams opened
+ * @capture: the number of capture streams opened
+ * @cpu_dai: the CPU DAI for this device
+ * @dev_attr: the sysfs device attribute structure
+ * @stats: SSI statistics
+ */
+struct fsl_ssi_private {
+ char name[8];
+ struct ccsr_ssi __iomem *ssi;
+ dma_addr_t ssi_phys;
+ unsigned int irq;
+ struct snd_pcm_substream *first_stream;
+ struct snd_pcm_substream *second_stream;
+ struct device *dev;
+ unsigned int playback;
+ unsigned int capture;
+ struct snd_soc_dai cpu_dai;
+ struct device_attribute dev_attr;
+
+ struct {
+ unsigned int rfrc;
+ unsigned int tfrc;
+ unsigned int cmdau;
+ unsigned int cmddu;
+ unsigned int rxt;
+ unsigned int rdr1;
+ unsigned int rdr0;
+ unsigned int tde1;
+ unsigned int tde0;
+ unsigned int roe1;
+ unsigned int roe0;
+ unsigned int tue1;
+ unsigned int tue0;
+ unsigned int tfs;
+ unsigned int rfs;
+ unsigned int tls;
+ unsigned int rls;
+ unsigned int rff1;
+ unsigned int rff0;
+ unsigned int tfe1;
+ unsigned int tfe0;
+ } stats;
+};
+
+/**
+ * fsl_ssi_isr: SSI interrupt handler
+ *
+ * Although it's possible to use the interrupt handler to send and receive
+ * data to/from the SSI, we use the DMA instead. Programming is more
+ * complicated, but the performance is much better.
+ *
+ * This interrupt handler is used only to gather statistics.
+ *
+ * @irq: IRQ of the SSI device
+ * @dev_id: pointer to the ssi_private structure for this SSI device
+ */
+static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
+{
+ struct fsl_ssi_private *ssi_private = dev_id;
+ struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+ irqreturn_t ret = IRQ_NONE;
+ __be32 sisr;
+ __be32 sisr2 = 0;
+
+ /* We got an interrupt, so read the status register to see what we
+ were interrupted for. We mask it with the Interrupt Enable register
+ so that we only check for events that we're interested in.
+ */
+ sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier);
+
+ if (sisr & CCSR_SSI_SISR_RFRC) {
+ ssi_private->stats.rfrc++;
+ sisr2 |= CCSR_SSI_SISR_RFRC;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TFRC) {
+ ssi_private->stats.tfrc++;
+ sisr2 |= CCSR_SSI_SISR_TFRC;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_CMDAU) {
+ ssi_private->stats.cmdau++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_CMDDU) {
+ ssi_private->stats.cmddu++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_RXT) {
+ ssi_private->stats.rxt++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_RDR1) {
+ ssi_private->stats.rdr1++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_RDR0) {
+ ssi_private->stats.rdr0++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TDE1) {
+ ssi_private->stats.tde1++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TDE0) {
+ ssi_private->stats.tde0++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_ROE1) {
+ ssi_private->stats.roe1++;
+ sisr2 |= CCSR_SSI_SISR_ROE1;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_ROE0) {
+ ssi_private->stats.roe0++;
+ sisr2 |= CCSR_SSI_SISR_ROE0;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TUE1) {
+ ssi_private->stats.tue1++;
+ sisr2 |= CCSR_SSI_SISR_TUE1;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TUE0) {
+ ssi_private->stats.tue0++;
+ sisr2 |= CCSR_SSI_SISR_TUE0;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TFS) {
+ ssi_private->stats.tfs++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_RFS) {
+ ssi_private->stats.rfs++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TLS) {
+ ssi_private->stats.tls++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_RLS) {
+ ssi_private->stats.rls++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_RFF1) {
+ ssi_private->stats.rff1++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_RFF0) {
+ ssi_private->stats.rff0++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TFE1) {
+ ssi_private->stats.tfe1++;
+ ret = IRQ_HANDLED;
+ }
+
+ if (sisr & CCSR_SSI_SISR_TFE0) {
+ ssi_private->stats.tfe0++;
+ ret = IRQ_HANDLED;
+ }
+
+ /* Clear the bits that we set */
+ if (sisr2)
+ out_be32(&ssi->sisr, sisr2);
+
+ return ret;
+}
+
+/**
+ * fsl_ssi_startup: create a new substream
+ *
+ * This is the first function called when a stream is opened.
+ *
+ * If this is the first stream open, then grab the IRQ and program most of
+ * the SSI registers.
+ */
+static int fsl_ssi_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
+
+ /*
+ * If this is the first stream opened, then request the IRQ
+ * and initialize the SSI registers.
+ */
+ if (!ssi_private->playback && !ssi_private->capture) {
+ struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+ int ret;
+
+ ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0,
+ ssi_private->name, ssi_private);
+ if (ret < 0) {
+ dev_err(substream->pcm->card->dev,
+ "could not claim irq %u\n", ssi_private->irq);
+ return ret;
+ }
+
+ /*
+ * Section 16.5 of the MPC8610 reference manual says that the
+ * SSI needs to be disabled before updating the registers we set
+ * here.
+ */
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+
+ /*
+ * Program the SSI into I2S Slave Non-Network Synchronous mode.
+ * Also enable the transmit and receive FIFO.
+ *
+ * FIXME: Little-endian samples require a different shift dir
+ */
+ clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK,
+ CCSR_SSI_SCR_TFR_CLK_DIS |
+ CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN);
+
+ out_be32(&ssi->stcr,
+ CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
+ CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TEFS |
+ CCSR_SSI_STCR_TSCKP);
+
+ out_be32(&ssi->srcr,
+ CCSR_SSI_SRCR_RXBIT0 | CCSR_SSI_SRCR_RFEN0 |
+ CCSR_SSI_SRCR_RFSI | CCSR_SSI_SRCR_REFS |
+ CCSR_SSI_SRCR_RSCKP);
+
+ /*
+ * The DC and PM bits are only used if the SSI is the clock
+ * master.
+ */
+
+ /* 4. Enable the interrupts and DMA requests */
+ out_be32(&ssi->sier,
+ CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE |
+ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN |
+ CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN |
+ CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE |
+ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN);
+
+ /*
+ * Set the watermark for transmit FIFI 0 and receive FIFO 0. We
+ * don't use FIFO 1. Since the SSI only supports stereo, the
+ * watermark should never be an odd number.
+ */
+ out_be32(&ssi->sfcsr,
+ CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2));
+
+ /*
+ * We keep the SSI disabled because if we enable it, then the
+ * DMA controller will start. It's not supposed to start until
+ * the SCR.TE (or SCR.RE) bit is set, but it does anyway. The
+ * DMA controller will transfer one "BWC" of data (i.e. the
+ * amount of data that the MR.BWC bits are set to). The reason
+ * this is bad is because at this point, the PCM driver has not
+ * finished initializing the DMA controller.
+ */
+ }
+
+ if (!ssi_private->first_stream)
+ ssi_private->first_stream = substream;
+ else {
+ /* This is the second stream open, so we need to impose sample
+ * rate and maybe sample size constraints. Note that this can
+ * cause a race condition if the second stream is opened before
+ * the first stream is fully initialized.
+ *
+ * We provide some protection by checking to make sure the first
+ * stream is initialized, but it's not perfect. ALSA sometimes
+ * re-initializes the driver with a different sample rate or
+ * size. If the second stream is opened before the first stream
+ * has received its final parameters, then the second stream may
+ * be constrained to the wrong sample rate or size.
+ *
+ * FIXME: This code does not handle opening and closing streams
+ * repeatedly. If you open two streams and then close the first
+ * one, you may not be able to open another stream until you
+ * close the second one as well.
+ */
+ struct snd_pcm_runtime *first_runtime =
+ ssi_private->first_stream->runtime;
+
+ if (!first_runtime->rate || !first_runtime->sample_bits) {
+ dev_err(substream->pcm->card->dev,
+ "set sample rate and size in %s stream first\n",
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+ ? "capture" : "playback");
+ return -EAGAIN;
+ }
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ first_runtime->rate, first_runtime->rate);
+
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ first_runtime->sample_bits,
+ first_runtime->sample_bits);
+
+ ssi_private->second_stream = substream;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ssi_private->playback++;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ssi_private->capture++;
+
+ return 0;
+}
+
+/**
+ * fsl_ssi_prepare: prepare the SSI.
+ *
+ * Most of the SSI registers have been programmed in the startup function,
+ * but the word length must be programmed here. Unfortunately, programming
+ * the SxCCR.WL bits requires the SSI to be temporarily disabled. This can
+ * cause a problem with supporting simultaneous playback and capture. If
+ * the SSI is already playing a stream, then that stream may be temporarily
+ * stopped when you start capture.
+ *
+ * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
+ * clock master.
+ */
+static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
+
+ struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+
+ if (substream == ssi_private->first_stream) {
+ u32 wl;
+
+ /* The SSI should always be disabled at this points (SSIEN=0) */
+ wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
+
+ /* In synchronous mode, the SSI uses STCCR for capture */
+ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_ssi_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ *
+ * The DMA channel is in external master start and pause mode, which
+ * means the SSI completely controls the flow of data.
+ */
+static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
+ struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ setbits32(&ssi->scr,
+ CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
+ } else {
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+ setbits32(&ssi->scr,
+ CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
+
+ /*
+ * I think we need this delay to allow time for the SSI
+ * to put data into its FIFO. Without it, ALSA starts
+ * to complain about overruns.
+ */
+ mdelay(1);
+ }
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+ else
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/**
+ * fsl_ssi_shutdown: shutdown the SSI
+ *
+ * Shutdown the SSI if there are no other substreams open.
+ */
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ssi_private->playback--;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ssi_private->capture--;
+
+ if (ssi_private->first_stream == substream)
+ ssi_private->first_stream = ssi_private->second_stream;
+
+ ssi_private->second_stream = NULL;
+
+ /*
+ * If this is the last active substream, disable the SSI and release
+ * the IRQ.
+ */
+ if (!ssi_private->playback && !ssi_private->capture) {
+ struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+
+ clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+
+ free_irq(ssi_private->irq, ssi_private);
+ }
+}
+
+/**
+ * fsl_ssi_set_sysclk: set the clock frequency and direction
+ *
+ * This function is called by the machine driver to tell us what the clock
+ * frequency and direction are.
+ *
+ * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
+ * and we don't care about the frequency. Return an error if the direction
+ * is not SND_SOC_CLOCK_IN.
+ *
+ * @clk_id: reserved, should be zero
+ * @freq: the frequency of the given clock ID, currently ignored
+ * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
+ */
+static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+
+ return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
+}
+
+/**
+ * fsl_ssi_set_fmt: set the serial format.
+ *
+ * This function is called by the machine driver to tell us what serial
+ * format to use.
+ *
+ * Currently, we only support I2S mode. Return an error if the format is
+ * not SND_SOC_DAIFMT_I2S.
+ *
+ * @format: one of SND_SOC_DAIFMT_xxx
+ */
+static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
+{
+ return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
+}
+
+/**
+ * fsl_ssi_dai_template: template CPU DAI for the SSI
+ */
+static struct snd_soc_dai fsl_ssi_dai_template = {
+ .playback = {
+ /* The SSI does not support monaural audio. */
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = FSLSSI_I2S_RATES,
+ .formats = FSLSSI_I2S_FORMATS,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = FSLSSI_I2S_RATES,
+ .formats = FSLSSI_I2S_FORMATS,
+ },
+ .ops = {
+ .startup = fsl_ssi_startup,
+ .prepare = fsl_ssi_prepare,
+ .shutdown = fsl_ssi_shutdown,
+ .trigger = fsl_ssi_trigger,
+ },
+ .dai_ops = {
+ .set_sysclk = fsl_ssi_set_sysclk,
+ .set_fmt = fsl_ssi_set_fmt,
+ },
+};
+
+/**
+ * fsl_sysfs_ssi_show: display SSI statistics
+ *
+ * Display the statistics for the current SSI device.
+ */
+static ssize_t fsl_sysfs_ssi_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct fsl_ssi_private *ssi_private =
+ container_of(attr, struct fsl_ssi_private, dev_attr);
+ ssize_t length;
+
+ length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc);
+ length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc);
+ length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau);
+ length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu);
+ length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt);
+ length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1);
+ length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0);
+ length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1);
+ length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0);
+ length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1);
+ length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0);
+ length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1);
+ length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0);
+ length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs);
+ length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs);
+ length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls);
+ length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls);
+ length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1);
+ length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0);
+ length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1);
+ length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0);
+
+ return length;
+}
+
+/**
+ * fsl_ssi_create_dai: create a snd_soc_dai structure
+ *
+ * This function is called by the machine driver to create a snd_soc_dai
+ * structure. The function creates an ssi_private object, which contains
+ * the snd_soc_dai. It also creates the sysfs statistics device.
+ */
+struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
+{
+ struct snd_soc_dai *fsl_ssi_dai;
+ struct fsl_ssi_private *ssi_private;
+ int ret = 0;
+ struct device_attribute *dev_attr;
+
+ ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL);
+ if (!ssi_private) {
+ dev_err(ssi_info->dev, "could not allocate DAI object\n");
+ return NULL;
+ }
+ memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template,
+ sizeof(struct snd_soc_dai));
+
+ fsl_ssi_dai = &ssi_private->cpu_dai;
+ dev_attr = &ssi_private->dev_attr;
+
+ sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id);
+ ssi_private->ssi = ssi_info->ssi;
+ ssi_private->ssi_phys = ssi_info->ssi_phys;
+ ssi_private->irq = ssi_info->irq;
+ ssi_private->dev = ssi_info->dev;
+
+ ssi_private->dev->driver_data = fsl_ssi_dai;
+
+ /* Initialize the the device_attribute structure */
+ dev_attr->attr.name = "ssi-stats";
+ dev_attr->attr.mode = S_IRUGO;
+ dev_attr->show = fsl_sysfs_ssi_show;
+
+ ret = device_create_file(ssi_private->dev, dev_attr);
+ if (ret) {
+ dev_err(ssi_info->dev, "could not create sysfs %s file\n",
+ ssi_private->dev_attr.attr.name);
+ kfree(fsl_ssi_dai);
+ return NULL;
+ }
+
+ fsl_ssi_dai->private_data = ssi_private;
+ fsl_ssi_dai->name = ssi_private->name;
+ fsl_ssi_dai->id = ssi_info->id;
+
+ return fsl_ssi_dai;
+}
+EXPORT_SYMBOL_GPL(fsl_ssi_create_dai);
+
+/**
+ * fsl_ssi_destroy_dai: destroy the snd_soc_dai object
+ *
+ * This function undoes the operations of fsl_ssi_create_dai()
+ */
+void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
+{
+ struct fsl_ssi_private *ssi_private =
+ container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai);
+
+ device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
+
+ kfree(ssi_private);
+}
+EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
new file mode 100644
index 0000000..83b44d7
--- /dev/null
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -0,0 +1,224 @@
+/*
+ * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 SoC
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
+ * under the terms of the GNU General Public License version 2. This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
+ */
+
+#ifndef _MPC8610_I2S_H
+#define _MPC8610_I2S_H
+
+/* SSI Register Map */
+struct ccsr_ssi {
+ __be32 stx0; /* 0x.0000 - SSI Transmit Data Register 0 */
+ __be32 stx1; /* 0x.0004 - SSI Transmit Data Register 1 */
+ __be32 srx0; /* 0x.0008 - SSI Receive Data Register 0 */
+ __be32 srx1; /* 0x.000C - SSI Receive Data Register 1 */
+ __be32 scr; /* 0x.0010 - SSI Control Register */
+ __be32 sisr; /* 0x.0014 - SSI Interrupt Status Register Mixed */
+ __be32 sier; /* 0x.0018 - SSI Interrupt Enable Register */
+ __be32 stcr; /* 0x.001C - SSI Transmit Configuration Register */
+ __be32 srcr; /* 0x.0020 - SSI Receive Configuration Register */
+ __be32 stccr; /* 0x.0024 - SSI Transmit Clock Control Register */
+ __be32 srccr; /* 0x.0028 - SSI Receive Clock Control Register */
+ __be32 sfcsr; /* 0x.002C - SSI FIFO Control/Status Register */
+ __be32 str; /* 0x.0030 - SSI Test Register */
+ __be32 sor; /* 0x.0034 - SSI Option Register */
+ __be32 sacnt; /* 0x.0038 - SSI AC97 Control Register */
+ __be32 sacadd; /* 0x.003C - SSI AC97 Command Address Register */
+ __be32 sacdat; /* 0x.0040 - SSI AC97 Command Data Register */
+ __be32 satag; /* 0x.0044 - SSI AC97 Tag Register */
+ __be32 stmsk; /* 0x.0048 - SSI Transmit Time Slot Mask Register */
+ __be32 srmsk; /* 0x.004C - SSI Receive Time Slot Mask Register */
+ __be32 saccst; /* 0x.0050 - SSI AC97 Channel Status Register */
+ __be32 saccen; /* 0x.0054 - SSI AC97 Channel Enable Register */
+ __be32 saccdis; /* 0x.0058 - SSI AC97 Channel Disable Register */
+};
+
+#define CCSR_SSI_SCR_RFR_CLK_DIS 0x00000800
+#define CCSR_SSI_SCR_TFR_CLK_DIS 0x00000400
+#define CCSR_SSI_SCR_TCH_EN 0x00000100
+#define CCSR_SSI_SCR_SYS_CLK_EN 0x00000080
+#define CCSR_SSI_SCR_I2S_MODE_MASK 0x00000060
+#define CCSR_SSI_SCR_I2S_MODE_NORMAL 0x00000000
+#define CCSR_SSI_SCR_I2S_MODE_MASTER 0x00000020
+#define CCSR_SSI_SCR_I2S_MODE_SLAVE 0x00000040
+#define CCSR_SSI_SCR_SYN 0x00000010
+#define CCSR_SSI_SCR_NET 0x00000008
+#define CCSR_SSI_SCR_RE 0x00000004
+#define CCSR_SSI_SCR_TE 0x00000002
+#define CCSR_SSI_SCR_SSIEN 0x00000001
+
+#define CCSR_SSI_SISR_RFRC 0x01000000
+#define CCSR_SSI_SISR_TFRC 0x00800000
+#define CCSR_SSI_SISR_CMDAU 0x00040000
+#define CCSR_SSI_SISR_CMDDU 0x00020000
+#define CCSR_SSI_SISR_RXT 0x00010000
+#define CCSR_SSI_SISR_RDR1 0x00008000
+#define CCSR_SSI_SISR_RDR0 0x00004000
+#define CCSR_SSI_SISR_TDE1 0x00002000
+#define CCSR_SSI_SISR_TDE0 0x00001000
+#define CCSR_SSI_SISR_ROE1 0x00000800
+#define CCSR_SSI_SISR_ROE0 0x00000400
+#define CCSR_SSI_SISR_TUE1 0x00000200
+#define CCSR_SSI_SISR_TUE0 0x00000100
+#define CCSR_SSI_SISR_TFS 0x00000080
+#define CCSR_SSI_SISR_RFS 0x00000040
+#define CCSR_SSI_SISR_TLS 0x00000020
+#define CCSR_SSI_SISR_RLS 0x00000010
+#define CCSR_SSI_SISR_RFF1 0x00000008
+#define CCSR_SSI_SISR_RFF0 0x00000004
+#define CCSR_SSI_SISR_TFE1 0x00000002
+#define CCSR_SSI_SISR_TFE0 0x00000001
+
+#define CCSR_SSI_SIER_RFRC_EN 0x01000000
+#define CCSR_SSI_SIER_TFRC_EN 0x00800000
+#define CCSR_SSI_SIER_RDMAE 0x00400000
+#define CCSR_SSI_SIER_RIE 0x00200000
+#define CCSR_SSI_SIER_TDMAE 0x00100000
+#define CCSR_SSI_SIER_TIE 0x00080000
+#define CCSR_SSI_SIER_CMDAU_EN 0x00040000
+#define CCSR_SSI_SIER_CMDDU_EN 0x00020000
+#define CCSR_SSI_SIER_RXT_EN 0x00010000
+#define CCSR_SSI_SIER_RDR1_EN 0x00008000
+#define CCSR_SSI_SIER_RDR0_EN 0x00004000
+#define CCSR_SSI_SIER_TDE1_EN 0x00002000
+#define CCSR_SSI_SIER_TDE0_EN 0x00001000
+#define CCSR_SSI_SIER_ROE1_EN 0x00000800
+#define CCSR_SSI_SIER_ROE0_EN 0x00000400
+#define CCSR_SSI_SIER_TUE1_EN 0x00000200
+#define CCSR_SSI_SIER_TUE0_EN 0x00000100
+#define CCSR_SSI_SIER_TFS_EN 0x00000080
+#define CCSR_SSI_SIER_RFS_EN 0x00000040
+#define CCSR_SSI_SIER_TLS_EN 0x00000020
+#define CCSR_SSI_SIER_RLS_EN 0x00000010
+#define CCSR_SSI_SIER_RFF1_EN 0x00000008
+#define CCSR_SSI_SIER_RFF0_EN 0x00000004
+#define CCSR_SSI_SIER_TFE1_EN 0x00000002
+#define CCSR_SSI_SIER_TFE0_EN 0x00000001
+
+#define CCSR_SSI_STCR_TXBIT0 0x00000200
+#define CCSR_SSI_STCR_TFEN1 0x00000100
+#define CCSR_SSI_STCR_TFEN0 0x00000080
+#define CCSR_SSI_STCR_TFDIR 0x00000040
+#define CCSR_SSI_STCR_TXDIR 0x00000020
+#define CCSR_SSI_STCR_TSHFD 0x00000010
+#define CCSR_SSI_STCR_TSCKP 0x00000008
+#define CCSR_SSI_STCR_TFSI 0x00000004
+#define CCSR_SSI_STCR_TFSL 0x00000002
+#define CCSR_SSI_STCR_TEFS 0x00000001
+
+#define CCSR_SSI_SRCR_RXEXT 0x00000400
+#define CCSR_SSI_SRCR_RXBIT0 0x00000200
+#define CCSR_SSI_SRCR_RFEN1 0x00000100
+#define CCSR_SSI_SRCR_RFEN0 0x00000080
+#define CCSR_SSI_SRCR_RFDIR 0x00000040
+#define CCSR_SSI_SRCR_RXDIR 0x00000020
+#define CCSR_SSI_SRCR_RSHFD 0x00000010
+#define CCSR_SSI_SRCR_RSCKP 0x00000008
+#define CCSR_SSI_SRCR_RFSI 0x00000004
+#define CCSR_SSI_SRCR_RFSL 0x00000002
+#define CCSR_SSI_SRCR_REFS 0x00000001
+
+/* STCCR and SRCCR */
+#define CCSR_SSI_SxCCR_DIV2 0x00040000
+#define CCSR_SSI_SxCCR_PSR 0x00020000
+#define CCSR_SSI_SxCCR_WL_SHIFT 13
+#define CCSR_SSI_SxCCR_WL_MASK 0x0001E000
+#define CCSR_SSI_SxCCR_WL(x) \
+ (((((x) / 2) - 1) << CCSR_SSI_SxCCR_WL_SHIFT) & CCSR_SSI_SxCCR_WL_MASK)
+#define CCSR_SSI_SxCCR_DC_SHIFT 8
+#define CCSR_SSI_SxCCR_DC_MASK 0x00001F00
+#define CCSR_SSI_SxCCR_DC(x) \
+ ((((x) - 1) << CCSR_SSI_SxCCR_DC_SHIFT) & CCSR_SSI_SxCCR_DC_MASK)
+#define CCSR_SSI_SxCCR_PM_SHIFT 0
+#define CCSR_SSI_SxCCR_PM_MASK 0x000000FF
+#define CCSR_SSI_SxCCR_PM(x) \
+ ((((x) - 1) << CCSR_SSI_SxCCR_PM_SHIFT) & CCSR_SSI_SxCCR_PM_MASK)
+
+/*
+ * The xFCNT bits are read-only, and the xFWM bits are read/write. Use the
+ * CCSR_SSI_SFCSR_xFCNTy() macros to read the FIFO counters, and use the
+ * CCSR_SSI_SFCSR_xFWMy() macros to set the watermarks.
+ */
+#define CCSR_SSI_SFCSR_RFCNT1_SHIFT 28
+#define CCSR_SSI_SFCSR_RFCNT1_MASK 0xF0000000
+#define CCSR_SSI_SFCSR_RFCNT1(x) \
+ (((x) & CCSR_SSI_SFCSR_RFCNT1_MASK) >> CCSR_SSI_SFCSR_RFCNT1_SHIFT)
+#define CCSR_SSI_SFCSR_TFCNT1_SHIFT 24
+#define CCSR_SSI_SFCSR_TFCNT1_MASK 0x0F000000
+#define CCSR_SSI_SFCSR_TFCNT1(x) \
+ (((x) & CCSR_SSI_SFCSR_TFCNT1_MASK) >> CCSR_SSI_SFCSR_TFCNT1_SHIFT)
+#define CCSR_SSI_SFCSR_RFWM1_SHIFT 20
+#define CCSR_SSI_SFCSR_RFWM1_MASK 0x00F00000
+#define CCSR_SSI_SFCSR_RFWM1(x) \
+ (((x) << CCSR_SSI_SFCSR_RFWM1_SHIFT) & CCSR_SSI_SFCSR_RFWM1_MASK)
+#define CCSR_SSI_SFCSR_TFWM1_SHIFT 16
+#define CCSR_SSI_SFCSR_TFWM1_MASK 0x000F0000
+#define CCSR_SSI_SFCSR_TFWM1(x) \
+ (((x) << CCSR_SSI_SFCSR_TFWM1_SHIFT) & CCSR_SSI_SFCSR_TFWM1_MASK)
+#define CCSR_SSI_SFCSR_RFCNT0_SHIFT 12
+#define CCSR_SSI_SFCSR_RFCNT0_MASK 0x0000F000
+#define CCSR_SSI_SFCSR_RFCNT0(x) \
+ (((x) & CCSR_SSI_SFCSR_RFCNT0_MASK) >> CCSR_SSI_SFCSR_RFCNT0_SHIFT)
+#define CCSR_SSI_SFCSR_TFCNT0_SHIFT 8
+#define CCSR_SSI_SFCSR_TFCNT0_MASK 0x00000F00
+#define CCSR_SSI_SFCSR_TFCNT0(x) \
+ (((x) & CCSR_SSI_SFCSR_TFCNT0_MASK) >> CCSR_SSI_SFCSR_TFCNT0_SHIFT)
+#define CCSR_SSI_SFCSR_RFWM0_SHIFT 4
+#define CCSR_SSI_SFCSR_RFWM0_MASK 0x000000F0
+#define CCSR_SSI_SFCSR_RFWM0(x) \
+ (((x) << CCSR_SSI_SFCSR_RFWM0_SHIFT) & CCSR_SSI_SFCSR_RFWM0_MASK)
+#define CCSR_SSI_SFCSR_TFWM0_SHIFT 0
+#define CCSR_SSI_SFCSR_TFWM0_MASK 0x0000000F
+#define CCSR_SSI_SFCSR_TFWM0(x) \
+ (((x) << CCSR_SSI_SFCSR_TFWM0_SHIFT) & CCSR_SSI_SFCSR_TFWM0_MASK)
+
+#define CCSR_SSI_STR_TEST 0x00008000
+#define CCSR_SSI_STR_RCK2TCK 0x00004000
+#define CCSR_SSI_STR_RFS2TFS 0x00002000
+#define CCSR_SSI_STR_RXSTATE(x) (((x) >> 8) & 0x1F)
+#define CCSR_SSI_STR_TXD2RXD 0x00000080
+#define CCSR_SSI_STR_TCK2RCK 0x00000040
+#define CCSR_SSI_STR_TFS2RFS 0x00000020
+#define CCSR_SSI_STR_TXSTATE(x) ((x) & 0x1F)
+
+#define CCSR_SSI_SOR_CLKOFF 0x00000040
+#define CCSR_SSI_SOR_RX_CLR 0x00000020
+#define CCSR_SSI_SOR_TX_CLR 0x00000010
+#define CCSR_SSI_SOR_INIT 0x00000008
+#define CCSR_SSI_SOR_WAIT_SHIFT 1
+#define CCSR_SSI_SOR_WAIT_MASK 0x00000006
+#define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT)
+#define CCSR_SSI_SOR_SYNRST 0x00000001
+
+/* Instantiation data for an SSI interface
+ *
+ * This structure contains all the information that the the SSI driver needs
+ * to instantiate an SSI interface with ALSA. The machine driver should
+ * create this structure, fill it in, call fsl_ssi_create_dai(), and then
+ * delete the structure.
+ *
+ * id: which SSI this is (0, 1, etc. )
+ * ssi: pointer to the SSI's registers
+ * ssi_phys: physical address of the SSI registers
+ * irq: IRQ of this SSI
+ * dev: struct device, used to create the sysfs statistics file
+*/
+struct fsl_ssi_info {
+ unsigned int id;
+ struct ccsr_ssi __iomem *ssi;
+ dma_addr_t ssi_phys;
+ unsigned int irq;
+ struct device *dev;
+};
+
+struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
+void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai);
+
+#endif
+
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
new file mode 100644
index 0000000..94a02ea
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -0,0 +1,886 @@
+/*
+ * Freescale MPC5200 PSC in I2S mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include <sysdev/bestcomm/bestcomm.h>
+#include <sysdev/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
+MODULE_LICENSE("GPL");
+
+/**
+ * PSC_I2S_RATES: sample rates supported by the I2S
+ *
+ * This driver currently only supports the PSC running in I2S slave mode,
+ * which means the codec determines the sample rate. Therefore, we tell
+ * ALSA that we support all rates and let the codec driver decide what rates
+ * are really supported.
+ */
+#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \
+ SNDRV_PCM_RATE_CONTINUOUS)
+
+/**
+ * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
+ */
+#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
+ SNDRV_PCM_FMTBIT_S32_BE)
+
+/**
+ * psc_i2s_stream - Data specific to a single stream (playback or capture)
+ * @active: flag indicating if the stream is active
+ * @psc_i2s: pointer back to parent psc_i2s data structure
+ * @bcom_task: bestcomm task structure
+ * @irq: irq number for bestcomm task
+ * @period_start: physical address of start of DMA region
+ * @period_end: physical address of end of DMA region
+ * @period_next_pt: physical address of next DMA buffer to enqueue
+ * @period_bytes: size of DMA period in bytes
+ */
+struct psc_i2s_stream {
+ int active;
+ struct psc_i2s *psc_i2s;
+ struct bcom_task *bcom_task;
+ int irq;
+ struct snd_pcm_substream *stream;
+ dma_addr_t period_start;
+ dma_addr_t period_end;
+ dma_addr_t period_next_pt;
+ dma_addr_t period_current_pt;
+ int period_bytes;
+};
+
+/**
+ * psc_i2s - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @dai: the CPU DAI for this device
+ * @sicr: Base value used in serial interface control register; mode is ORed
+ * with this value.
+ * @playback: Playback stream context data
+ * @capture: Capture stream context data
+ */
+struct psc_i2s {
+ char name[32];
+ struct mpc52xx_psc __iomem *psc_regs;
+ struct mpc52xx_psc_fifo __iomem *fifo_regs;
+ unsigned int irq;
+ struct device *dev;
+ struct snd_soc_dai dai;
+ spinlock_t lock;
+ u32 sicr;
+
+ /* per-stream data */
+ struct psc_i2s_stream playback;
+ struct psc_i2s_stream capture;
+
+ /* Statistics */
+ struct {
+ int overrun_count;
+ int underrun_count;
+ } stats;
+};
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
+{
+ struct psc_i2s *psc_i2s = _psc_i2s;
+ struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+ u16 isr;
+
+ isr = in_be16(&regs->mpc52xx_psc_isr);
+
+ /* Playback underrun error */
+ if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+ psc_i2s->stats.underrun_count++;
+
+ /* Capture overrun error */
+ if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+ psc_i2s->stats.overrun_count++;
+
+ out_8(&regs->command, 4 << 4); /* reset the error status */
+
+ return IRQ_HANDLED;
+}
+
+/**
+ * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue. Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
+{
+ struct bcom_bd *bd;
+
+ /* Prepare and enqueue the next buffer descriptor */
+ bd = bcom_prepare_next_buffer(s->bcom_task);
+ bd->status = s->period_bytes;
+ bd->data[0] = s->period_next_pt;
+ bcom_submit_next_buffer(s->bcom_task, NULL);
+
+ /* Update for next period */
+ s->period_next_pt += s->period_bytes;
+ if (s->period_next_pt >= s->period_end)
+ s->period_next_pt = s->period_start;
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
+{
+ struct psc_i2s_stream *s = _psc_i2s_stream;
+
+ /* For each finished period, dequeue the completed period buffer
+ * and enqueue a new one in it's place. */
+ while (bcom_buffer_done(s->bcom_task)) {
+ bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+ s->period_current_pt += s->period_bytes;
+ if (s->period_current_pt >= s->period_end)
+ s->period_current_pt = s->period_start;
+ psc_i2s_bcom_enqueue_next_buffer(s);
+ bcom_enable(s->bcom_task);
+ }
+
+ /* If the stream is active, then also inform the PCM middle layer
+ * of the period finished event. */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ return IRQ_HANDLED;
+}
+
+/**
+ * psc_i2s_startup: create a new substream
+ *
+ * This is the first function called when a stream is opened.
+ *
+ * If this is the first stream open, then grab the IRQ and program most of
+ * the PSC registers.
+ */
+static int psc_i2s_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ int rc;
+
+ dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
+
+ if (!psc_i2s->playback.active &&
+ !psc_i2s->capture.active) {
+ /* Setup the IRQs */
+ rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
+ "psc-i2s-status", psc_i2s);
+ rc |= request_irq(psc_i2s->capture.irq,
+ &psc_i2s_bcom_irq, IRQF_SHARED,
+ "psc-i2s-capture", &psc_i2s->capture);
+ rc |= request_irq(psc_i2s->playback.irq,
+ &psc_i2s_bcom_irq, IRQF_SHARED,
+ "psc-i2s-playback", &psc_i2s->playback);
+ if (rc) {
+ free_irq(psc_i2s->irq, psc_i2s);
+ free_irq(psc_i2s->capture.irq,
+ &psc_i2s->capture);
+ free_irq(psc_i2s->playback.irq,
+ &psc_i2s->playback);
+ return -ENODEV;
+ }
+ }
+
+ return 0;
+}
+
+static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ u32 mode;
+
+ dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ " periods=%i buffer_size=%i buffer_bytes=%i\n",
+ __func__, substream, params_period_size(params),
+ params_period_bytes(params), params_periods(params),
+ params_buffer_size(params), params_buffer_bytes(params));
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_BE:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_BE:
+ mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
+ break;
+ default:
+ dev_dbg(psc_i2s->dev, "invalid format\n");
+ return -EINVAL;
+ }
+ out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+/**
+ * psc_i2s_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct psc_i2s_stream *s;
+ struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
+ u16 imr;
+ u8 psc_cmd;
+ unsigned long flags;
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_i2s->capture;
+ else
+ s = &psc_i2s->playback;
+
+ dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
+ " stream_id=%i\n",
+ substream, cmd, substream->pstr->stream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ s->period_bytes = frames_to_bytes(runtime,
+ runtime->period_size);
+ s->period_start = virt_to_phys(runtime->dma_area);
+ s->period_end = s->period_start +
+ (s->period_bytes * runtime->periods);
+ s->period_next_pt = s->period_start;
+ s->period_current_pt = s->period_start;
+ s->active = 1;
+
+ /* First; reset everything */
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ out_8(&regs->command, MPC52xx_PSC_RST_RX);
+ out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+ } else {
+ out_8(&regs->command, MPC52xx_PSC_RST_TX);
+ out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+ }
+
+ /* Next, fill up the bestcomm bd queue and enable DMA.
+ * This will begin filling the PSC's fifo. */
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ bcom_gen_bd_rx_reset(s->bcom_task);
+ else
+ bcom_gen_bd_tx_reset(s->bcom_task);
+ while (!bcom_queue_full(s->bcom_task))
+ psc_i2s_bcom_enqueue_next_buffer(s);
+ bcom_enable(s->bcom_task);
+
+ /* Due to errata in the i2s mode; need to line up enabling
+ * the transmitter with a transition on the frame sync
+ * line */
+
+ spin_lock_irqsave(&psc_i2s->lock, flags);
+ /* first make sure it is low */
+ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
+ ;
+ /* then wait for the transition to high */
+ while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
+ ;
+ /* Finally, enable the PSC.
+ * Receiver must always be enabled; even when we only want
+ * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */
+ psc_cmd = MPC52xx_PSC_RX_ENABLE;
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ psc_cmd |= MPC52xx_PSC_TX_ENABLE;
+ out_8(&regs->command, psc_cmd);
+ spin_unlock_irqrestore(&psc_i2s->lock, flags);
+
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ /* Turn off the PSC */
+ s->active = 0;
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (!psc_i2s->playback.active) {
+ out_8(&regs->command, 2 << 4); /* reset rx */
+ out_8(&regs->command, 3 << 4); /* reset tx */
+ out_8(&regs->command, 4 << 4); /* reset err */
+ }
+ } else {
+ out_8(&regs->command, 3 << 4); /* reset tx */
+ out_8(&regs->command, 4 << 4); /* reset err */
+ if (!psc_i2s->capture.active)
+ out_8(&regs->command, 2 << 4); /* reset rx */
+ }
+
+ bcom_disable(s->bcom_task);
+ while (!bcom_queue_empty(s->bcom_task))
+ bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+ break;
+
+ default:
+ dev_dbg(psc_i2s->dev, "invalid command\n");
+ return -EINVAL;
+ }
+
+ /* Update interrupt enable settings */
+ imr = 0;
+ if (psc_i2s->playback.active)
+ imr |= MPC52xx_PSC_IMR_TXEMP;
+ if (psc_i2s->capture.active)
+ imr |= MPC52xx_PSC_IMR_ORERR;
+ out_be16(&regs->isr_imr.imr, imr);
+
+ return 0;
+}
+
+/**
+ * psc_i2s_shutdown: shutdown the data transfer on a stream
+ *
+ * Shutdown the PSC if there are no other substreams open.
+ */
+static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+
+ dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
+
+ /*
+ * If this is the last active substream, disable the PSC and release
+ * the IRQ.
+ */
+ if (!psc_i2s->playback.active &&
+ !psc_i2s->capture.active) {
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
+ out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
+ out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
+ out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
+ out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+
+ /* Release irqs */
+ free_irq(psc_i2s->irq, psc_i2s);
+ free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
+ free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
+ }
+}
+
+/**
+ * psc_i2s_set_sysclk: set the clock frequency and direction
+ *
+ * This function is called by the machine driver to tell us what the clock
+ * frequency and direction are.
+ *
+ * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN),
+ * and we don't care about the frequency. Return an error if the direction
+ * is not SND_SOC_CLOCK_IN.
+ *
+ * @clk_id: reserved, should be zero
+ * @freq: the frequency of the given clock ID, currently ignored
+ * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
+ */
+static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct psc_i2s *psc_i2s = cpu_dai->private_data;
+ dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+ cpu_dai, dir);
+ return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
+}
+
+/**
+ * psc_i2s_set_fmt: set the serial format.
+ *
+ * This function is called by the machine driver to tell us what serial
+ * format to use.
+ *
+ * This driver only supports I2S mode. Return an error if the format is
+ * not SND_SOC_DAIFMT_I2S.
+ *
+ * @format: one of SND_SOC_DAIFMT_xxx
+ */
+static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
+{
+ struct psc_i2s *psc_i2s = cpu_dai->private_data;
+ dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+ cpu_dai, format);
+ return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_i2s_dai_template: template CPU Digital Audio Interface
+ */
+static struct snd_soc_dai psc_i2s_dai_template = {
+ .type = SND_SOC_DAI_I2S,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PSC_I2S_RATES,
+ .formats = PSC_I2S_FORMATS,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PSC_I2S_RATES,
+ .formats = PSC_I2S_FORMATS,
+ },
+ .ops = {
+ .startup = psc_i2s_startup,
+ .hw_params = psc_i2s_hw_params,
+ .hw_free = psc_i2s_hw_free,
+ .shutdown = psc_i2s_shutdown,
+ .trigger = psc_i2s_trigger,
+ },
+ .dai_ops = {
+ .set_sysclk = psc_i2s_set_sysclk,
+ .set_fmt = psc_i2s_set_fmt,
+ },
+};
+
+/* ---------------------------------------------------------------------
+ * The PSC I2S 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .period_bytes_max = 1024 * 1024,
+ .period_bytes_min = 32,
+ .periods_min = 2,
+ .periods_max = 256,
+ .buffer_bytes_max = 2 * 1024 * 1024,
+ .fifo_size = 0,
+};
+
+static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ struct psc_i2s_stream *s;
+
+ dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_i2s->capture;
+ else
+ s = &psc_i2s->playback;
+
+ snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
+
+ s->stream = substream;
+ return 0;
+}
+
+static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ struct psc_i2s_stream *s;
+
+ dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_i2s->capture;
+ else
+ s = &psc_i2s->playback;
+
+ s->stream = NULL;
+ return 0;
+}
+
+static snd_pcm_uframes_t
+psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ struct psc_i2s_stream *s;
+ dma_addr_t count;
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_i2s->capture;
+ else
+ s = &psc_i2s->playback;
+
+ count = s->period_current_pt - s->period_start;
+
+ return bytes_to_frames(substream->runtime, count);
+}
+
+static struct snd_pcm_ops psc_i2s_pcm_ops = {
+ .open = psc_i2s_pcm_open,
+ .close = psc_i2s_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .pointer = psc_i2s_pcm_pointer,
+};
+
+static u64 psc_i2s_pcm_dmamask = 0xffffffff;
+static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
+ int rc = 0;
+
+ dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
+ card, dai, pcm);
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &psc_i2s_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (pcm->streams[0].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
+ &pcm->streams[0].substream->dma_buffer);
+ if (rc)
+ goto playback_alloc_err;
+ }
+
+ if (pcm->streams[1].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
+ &pcm->streams[1].substream->dma_buffer);
+ if (rc)
+ goto capture_alloc_err;
+ }
+
+ return 0;
+
+ capture_alloc_err:
+ if (pcm->streams[0].substream)
+ snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+ playback_alloc_err:
+ dev_err(card->dev, "Cannot allocate buffer(s)\n");
+ return -ENOMEM;
+}
+
+static void psc_i2s_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct snd_pcm_substream *substream;
+ int stream;
+
+ dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+struct snd_soc_platform psc_i2s_pcm_soc_platform = {
+ .name = "mpc5200-psc-audio",
+ .pcm_ops = &psc_i2s_pcm_ops,
+ .pcm_new = &psc_i2s_pcm_new,
+ .pcm_free = &psc_i2s_pcm_free,
+};
+
+/* ---------------------------------------------------------------------
+ * Sysfs attributes for debugging
+ */
+
+static ssize_t psc_i2s_status_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+
+ return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x "
+ "tfnum=%i tfstat=0x%.4x\n",
+ in_be16(&psc_i2s->psc_regs->sr_csr.status),
+ in_be32(&psc_i2s->psc_regs->sicr),
+ in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
+ in_be16(&psc_i2s->fifo_regs->rfstat),
+ in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
+ in_be16(&psc_i2s->fifo_regs->tfstat));
+}
+
+static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name)
+{
+ if (strcmp(name, "playback_underrun") == 0)
+ return &psc_i2s->stats.underrun_count;
+ if (strcmp(name, "capture_overrun") == 0)
+ return &psc_i2s->stats.overrun_count;
+
+ return NULL;
+}
+
+static ssize_t psc_i2s_stat_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+ int *attrib;
+
+ attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
+ if (!attrib)
+ return 0;
+
+ return sprintf(buf, "%i\n", *attrib);
+}
+
+static ssize_t psc_i2s_stat_store(struct device *dev,
+ struct device_attribute *attr,
+ const char *buf,
+ size_t count)
+{
+ struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
+ int *attrib;
+
+ attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
+ if (!attrib)
+ return 0;
+
+ *attrib = simple_strtoul(buf, NULL, 0);
+ return count;
+}
+
+static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
+static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,
+ psc_i2s_stat_store);
+static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
+ psc_i2s_stat_store);
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int __devinit psc_i2s_of_probe(struct of_device *op,
+ const struct of_device_id *match)
+{
+ phys_addr_t fifo;
+ struct psc_i2s *psc_i2s;
+ struct resource res;
+ int size, psc_id, irq, rc;
+ const __be32 *prop;
+ void __iomem *regs;
+
+ dev_dbg(&op->dev, "probing psc i2s device\n");
+
+ /* Get the PSC ID */
+ prop = of_get_property(op->node, "cell-index", &size);
+ if (!prop || size < sizeof *prop)
+ return -ENODEV;
+ psc_id = be32_to_cpu(*prop);
+
+ /* Fetch the registers and IRQ of the PSC */
+ irq = irq_of_parse_and_map(op->node, 0);
+ if (of_address_to_resource(op->node, 0, &res)) {
+ dev_err(&op->dev, "Missing reg property\n");
+ return -ENODEV;
+ }
+ regs = ioremap(res.start, 1 + res.end - res.start);
+ if (!regs) {
+ dev_err(&op->dev, "Could not map registers\n");
+ return -ENODEV;
+ }
+
+ /* Allocate and initialize the driver private data */
+ psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
+ if (!psc_i2s) {
+ iounmap(regs);
+ return -ENOMEM;
+ }
+ spin_lock_init(&psc_i2s->lock);
+ psc_i2s->irq = irq;
+ psc_i2s->psc_regs = regs;
+ psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
+ psc_i2s->dev = &op->dev;
+ psc_i2s->playback.psc_i2s = psc_i2s;
+ psc_i2s->capture.psc_i2s = psc_i2s;
+ snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
+
+ /* Fill out the CPU DAI structure */
+ memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
+ psc_i2s->dai.private_data = psc_i2s;
+ psc_i2s->dai.name = psc_i2s->name;
+ psc_i2s->dai.id = psc_id;
+
+ /* Find the address of the fifo data registers and setup the
+ * DMA tasks */
+ fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
+ psc_i2s->capture.bcom_task =
+ bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
+ psc_i2s->playback.bcom_task =
+ bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
+ if (!psc_i2s->capture.bcom_task ||
+ !psc_i2s->playback.bcom_task) {
+ dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
+ iounmap(regs);
+ kfree(psc_i2s);
+ return -ENODEV;
+ }
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
+ out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */
+ out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */
+ out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
+ out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+
+ /* Configure the serial interface mode; defaulting to CODEC8 mode */
+ psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+ MPC52xx_PSC_SICR_CLKPOL;
+ if (of_get_property(op->node, "fsl,cellslave", NULL))
+ psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE |
+ MPC52xx_PSC_SICR_GENCLK;
+ out_be32(&psc_i2s->psc_regs->sicr,
+ psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
+
+ /* Check for the codec handle. If it is not present then we
+ * are done */
+ if (!of_get_property(op->node, "codec-handle", NULL))
+ return 0;
+
+ /* Set up mode register;
+ * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
+ * Second write: register Normal mode for non loopback
+ */
+ out_8(&psc_i2s->psc_regs->mode, 0);
+ out_8(&psc_i2s->psc_regs->mode, 0);
+
+ /* Set the TX and RX fifo alarm thresholds */
+ out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100);
+ out_8(&psc_i2s->fifo_regs->rfcntl, 0x4);
+ out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100);
+ out_8(&psc_i2s->fifo_regs->tfcntl, 0x7);
+
+ /* Lookup the IRQ numbers */
+ psc_i2s->playback.irq =
+ bcom_get_task_irq(psc_i2s->playback.bcom_task);
+ psc_i2s->capture.irq =
+ bcom_get_task_irq(psc_i2s->capture.bcom_task);
+
+ /* Save what we've done so it can be found again later */
+ dev_set_drvdata(&op->dev, psc_i2s);
+
+ /* Register the SYSFS files */
+ rc = device_create_file(psc_i2s->dev, &dev_attr_status);
+ rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
+ rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
+ if (rc)
+ dev_info(psc_i2s->dev, "error creating sysfs files\n");
+
+ /* Tell the ASoC OF helpers about it */
+ of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
+ &psc_i2s->dai);
+
+ return 0;
+}
+
+static int __devexit psc_i2s_of_remove(struct of_device *op)
+{
+ struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
+
+ dev_dbg(&op->dev, "psc_i2s_remove()\n");
+
+ bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
+ bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
+
+ iounmap(psc_i2s->psc_regs);
+ iounmap(psc_i2s->fifo_regs);
+ kfree(psc_i2s);
+ dev_set_drvdata(&op->dev, NULL);
+
+ return 0;
+}
+
+/* Match table for of_platform binding */
+static struct of_device_id psc_i2s_match[] __devinitdata = {
+ { .compatible = "fsl,mpc5200-psc-i2s", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, psc_i2s_match);
+
+static struct of_platform_driver psc_i2s_driver = {
+ .match_table = psc_i2s_match,
+ .probe = psc_i2s_of_probe,
+ .remove = __devexit_p(psc_i2s_of_remove),
+ .driver = {
+ .name = "mpc5200-psc-i2s",
+ .owner = THIS_MODULE,
+ },
+};
+
+/* ---------------------------------------------------------------------
+ * Module setup and teardown; simply register the of_platform driver
+ * for the PSC in I2S mode.
+ */
+static int __init psc_i2s_init(void)
+{
+ return of_register_platform_driver(&psc_i2s_driver);
+}
+module_init(psc_i2s_init);
+
+static void __exit psc_i2s_exit(void)
+{
+ of_unregister_platform_driver(&psc_i2s_driver);
+}
+module_exit(psc_i2s_exit);
+
+
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
new file mode 100644
index 0000000..94f89de
--- /dev/null
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -0,0 +1,625 @@
+/**
+ * Freescale MPC8610HPCD ALSA SoC Fabric driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed
+ * under the terms of the GNU General Public License version 2. This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+#include <asm/immap_86xx.h>
+
+#include "../codecs/cs4270.h"
+#include "fsl_dma.h"
+#include "fsl_ssi.h"
+
+/**
+ * mpc8610_hpcd_data: fabric-specific ASoC device data
+ *
+ * This structure contains data for a single sound platform device on an
+ * MPC8610 HPCD. Some of the data is taken from the device tree.
+ */
+struct mpc8610_hpcd_data {
+ struct snd_soc_device sound_devdata;
+ struct snd_soc_dai_link dai;
+ struct snd_soc_machine machine;
+ unsigned int dai_format;
+ unsigned int codec_clk_direction;
+ unsigned int cpu_clk_direction;
+ unsigned int clk_frequency;
+ struct ccsr_guts __iomem *guts;
+ struct ccsr_ssi __iomem *ssi;
+ unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */
+ unsigned int ssi_irq;
+ unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */
+ unsigned int dma_irq[2];
+ struct ccsr_dma_channel __iomem *dma[2];
+ unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
+};
+
+/**
+ * mpc8610_hpcd_machine_probe: initalize the board
+ *
+ * This function is called when platform_device_add() is called. It is used
+ * to initialize the board-specific hardware.
+ *
+ * Here we program the DMACR and PMUXCR registers.
+ */
+static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device)
+{
+ struct mpc8610_hpcd_data *machine_data =
+ sound_device->dev.platform_data;
+
+ /* Program the signal routing between the SSI and the DMA */
+ guts_set_dmacr(machine_data->guts, machine_data->dma_id,
+ machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI);
+ guts_set_dmacr(machine_data->guts, machine_data->dma_id,
+ machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI);
+
+ guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id,
+ machine_data->dma_channel_id[0], 0);
+ guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id,
+ machine_data->dma_channel_id[1], 0);
+
+ switch (machine_data->ssi_id) {
+ case 0:
+ clrsetbits_be32(&machine_data->guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI);
+ break;
+ case 1:
+ clrsetbits_be32(&machine_data->guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI);
+ break;
+ }
+
+ return 0;
+}
+
+/**
+ * mpc8610_hpcd_startup: program the board with various hardware parameters
+ *
+ * This function takes board-specific information, like clock frequencies
+ * and serial data formats, and passes that information to the codec and
+ * transport drivers.
+ */
+static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct mpc8610_hpcd_data *machine_data =
+ rtd->socdev->dev->platform_data;
+ int ret = 0;
+
+ /* Tell the CPU driver what the serial protocol is. */
+ ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format);
+ if (ret < 0) {
+ dev_err(substream->pcm->card->dev,
+ "could not set CPU driver audio format\n");
+ return ret;
+ }
+
+ /* Tell the codec driver what the serial protocol is. */
+ ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format);
+ if (ret < 0) {
+ dev_err(substream->pcm->card->dev,
+ "could not set codec driver audio format\n");
+ return ret;
+ }
+
+ /*
+ * Tell the CPU driver what the clock frequency is, and whether it's a
+ * slave or master.
+ */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, 0,
+ machine_data->clk_frequency,
+ machine_data->cpu_clk_direction);
+ if (ret < 0) {
+ dev_err(substream->pcm->card->dev,
+ "could not set CPU driver clock parameters\n");
+ return ret;
+ }
+
+ /*
+ * Tell the codec driver what the MCLK frequency is, and whether it's
+ * a slave or master.
+ */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+ machine_data->clk_frequency,
+ machine_data->codec_clk_direction);
+ if (ret < 0) {
+ dev_err(substream->pcm->card->dev,
+ "could not set codec driver clock params\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+/**
+ * mpc8610_hpcd_machine_remove: Remove the sound device
+ *
+ * This function is called to remove the sound device for one SSI. We
+ * de-program the DMACR and PMUXCR register.
+ */
+int mpc8610_hpcd_machine_remove(struct platform_device *sound_device)
+{
+ struct mpc8610_hpcd_data *machine_data =
+ sound_device->dev.platform_data;
+
+ /* Restore the signal routing */
+
+ guts_set_dmacr(machine_data->guts, machine_data->dma_id,
+ machine_data->dma_channel_id[0], 0);
+ guts_set_dmacr(machine_data->guts, machine_data->dma_id,
+ machine_data->dma_channel_id[1], 0);
+
+ switch (machine_data->ssi_id) {
+ case 0:
+ clrsetbits_be32(&machine_data->guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA);
+ break;
+ case 1:
+ clrsetbits_be32(&machine_data->guts->pmuxcr,
+ CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA);
+ break;
+ }
+
+ return 0;
+}
+
+/**
+ * mpc8610_hpcd_ops: ASoC fabric driver operations
+ */
+static struct snd_soc_ops mpc8610_hpcd_ops = {
+ .startup = mpc8610_hpcd_startup,
+};
+
+/**
+ * mpc8610_hpcd_machine: ASoC machine data
+ */
+static struct snd_soc_machine mpc8610_hpcd_machine = {
+ .probe = mpc8610_hpcd_machine_probe,
+ .remove = mpc8610_hpcd_machine_remove,
+ .name = "MPC8610 HPCD",
+ .num_links = 1,
+};
+
+/**
+ * mpc8610_hpcd_probe: OF probe function for the fabric driver
+ *
+ * This function gets called when an SSI node is found in the device tree.
+ *
+ * Although this is a fabric driver, the SSI node is the "master" node with
+ * respect to audio hardware connections. Therefore, we create a new ASoC
+ * device for each new SSI node that has a codec attached.
+ *
+ * FIXME: Currently, we only support one DMA controller, so if there are
+ * multiple SSI nodes with codecs, only the first will be supported.
+ *
+ * FIXME: Even if we did support multiple DMA controllers, we have no
+ * mechanism for assigning DMA controllers and channels to the individual
+ * SSI devices. We also probably aren't compatible with the generic Elo DMA
+ * device driver.
+ */
+static int mpc8610_hpcd_probe(struct of_device *ofdev,
+ const struct of_device_id *match)
+{
+ struct device_node *np = ofdev->node;
+ struct device_node *codec_np = NULL;
+ struct device_node *guts_np = NULL;
+ struct device_node *dma_np = NULL;
+ struct device_node *dma_channel_np = NULL;
+ const phandle *codec_ph;
+ const char *sprop;
+ const u32 *iprop;
+ struct resource res;
+ struct platform_device *sound_device = NULL;
+ struct mpc8610_hpcd_data *machine_data;
+ struct fsl_ssi_info ssi_info;
+ struct fsl_dma_info dma_info;
+ int ret = -ENODEV;
+ unsigned int playback_dma_channel;
+ unsigned int capture_dma_channel;
+
+ machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
+ if (!machine_data)
+ return -ENOMEM;
+
+ memset(&ssi_info, 0, sizeof(ssi_info));
+ memset(&dma_info, 0, sizeof(dma_info));
+
+ ssi_info.dev = &ofdev->dev;
+
+ /*
+ * We are only interested in SSIs with a codec phandle in them, so let's
+ * make sure this SSI has one.
+ */
+ codec_ph = of_get_property(np, "codec-handle", NULL);
+ if (!codec_ph)
+ goto error;
+
+ codec_np = of_find_node_by_phandle(*codec_ph);
+ if (!codec_np)
+ goto error;
+
+ /* The MPC8610 HPCD only knows about the CS4270 codec, so reject
+ anything else. */
+ if (!of_device_is_compatible(codec_np, "cirrus,cs4270"))
+ goto error;
+
+ /* Get the device ID */
+ iprop = of_get_property(np, "cell-index", NULL);
+ if (!iprop) {
+ dev_err(&ofdev->dev, "cell-index property not found\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ machine_data->ssi_id = *iprop;
+ ssi_info.id = *iprop;
+
+ /* Get the serial format and clock direction. */
+ sprop = of_get_property(np, "fsl,mode", NULL);
+ if (!sprop) {
+ dev_err(&ofdev->dev, "fsl,mode property not found\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ if (strcasecmp(sprop, "i2s-slave") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+
+ /*
+ * In i2s-slave mode, the codec has its own clock source, so we
+ * need to get the frequency from the device tree and pass it to
+ * the codec driver.
+ */
+ iprop = of_get_property(codec_np, "clock-frequency", NULL);
+ if (!iprop || !*iprop) {
+ dev_err(&ofdev->dev, "codec bus-frequency property "
+ "is missing or invalid\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ machine_data->clk_frequency = *iprop;
+ } else if (strcasecmp(sprop, "i2s-master") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "lj-slave") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "lj-master") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "rj-slave") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "rj-master") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else if (strcasecmp(sprop, "ac97-slave") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
+ } else if (strcasecmp(sprop, "ac97-master") == 0) {
+ machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
+ machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+ } else {
+ dev_err(&ofdev->dev,
+ "unrecognized fsl,mode property \"%s\"\n", sprop);
+ ret = -EINVAL;
+ goto error;
+ }
+
+ if (!machine_data->clk_frequency) {
+ dev_err(&ofdev->dev, "unknown clock frequency\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ /* Read the SSI information from the device tree */
+ ret = of_address_to_resource(np, 0, &res);
+ if (ret) {
+ dev_err(&ofdev->dev, "could not obtain SSI address\n");
+ goto error;
+ }
+ if (!res.start) {
+ dev_err(&ofdev->dev, "invalid SSI address\n");
+ goto error;
+ }
+ ssi_info.ssi_phys = res.start;
+
+ machine_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi));
+ if (!machine_data->ssi) {
+ dev_err(&ofdev->dev, "could not map SSI address %x\n",
+ ssi_info.ssi_phys);
+ ret = -EINVAL;
+ goto error;
+ }
+ ssi_info.ssi = machine_data->ssi;
+
+
+ /* Get the IRQ of the SSI */
+ machine_data->ssi_irq = irq_of_parse_and_map(np, 0);
+ if (!machine_data->ssi_irq) {
+ dev_err(&ofdev->dev, "could not get SSI IRQ\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ ssi_info.irq = machine_data->ssi_irq;
+
+
+ /* Map the global utilities registers. */
+ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts");
+ if (!guts_np) {
+ dev_err(&ofdev->dev, "could not obtain address of GUTS\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ machine_data->guts = of_iomap(guts_np, 0);
+ of_node_put(guts_np);
+ if (!machine_data->guts) {
+ dev_err(&ofdev->dev, "could not map GUTS\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ /* Find the DMA channels to use. Both SSIs need to use the same DMA
+ * controller, so let's use DMA#1.
+ */
+ for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") {
+ iprop = of_get_property(dma_np, "cell-index", NULL);
+ if (iprop && (*iprop == 0)) {
+ of_node_put(dma_np);
+ break;
+ }
+ }
+ if (!dma_np) {
+ dev_err(&ofdev->dev, "could not find DMA node\n");
+ ret = -EINVAL;
+ goto error;
+ }
+ machine_data->dma_id = *iprop;
+
+ /* SSI1 needs to use DMA Channels 0 and 1, and SSI2 needs to use DMA
+ * channels 2 and 3. This is just how the MPC8610 is wired
+ * internally.
+ */
+ playback_dma_channel = (machine_data->ssi_id == 0) ? 0 : 2;
+ capture_dma_channel = (machine_data->ssi_id == 0) ? 1 : 3;
+
+ /*
+ * Find the DMA channels to use.
+ */
+ while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) {
+ iprop = of_get_property(dma_channel_np, "cell-index", NULL);
+ if (iprop && (*iprop == playback_dma_channel)) {
+ /* dma_channel[0] and dma_irq[0] are for playback */
+ dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0);
+ dma_info.dma_irq[0] =
+ irq_of_parse_and_map(dma_channel_np, 0);
+ machine_data->dma_channel_id[0] = *iprop;
+ continue;
+ }
+ if (iprop && (*iprop == capture_dma_channel)) {
+ /* dma_channel[1] and dma_irq[1] are for capture */
+ dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0);
+ dma_info.dma_irq[1] =
+ irq_of_parse_and_map(dma_channel_np, 0);
+ machine_data->dma_channel_id[1] = *iprop;
+ continue;
+ }
+ }
+ if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] ||
+ !dma_info.dma_irq[0] || !dma_info.dma_irq[1]) {
+ dev_err(&ofdev->dev, "could not find DMA channels\n");
+ ret = -EINVAL;
+ goto error;
+ }
+
+ dma_info.ssi_stx_phys = ssi_info.ssi_phys +
+ offsetof(struct ccsr_ssi, stx0);
+ dma_info.ssi_srx_phys = ssi_info.ssi_phys +
+ offsetof(struct ccsr_ssi, srx0);
+
+ /* We have the DMA information, so tell the DMA driver what it is */
+ if (!fsl_dma_configure(&dma_info)) {
+ dev_err(&ofdev->dev, "could not instantiate DMA device\n");
+ ret = -EBUSY;
+ goto error;
+ }
+
+ /*
+ * Initialize our DAI data structure. We should probably get this
+ * information from the device tree.
+ */
+ machine_data->dai.name = "CS4270";
+ machine_data->dai.stream_name = "CS4270";
+
+ machine_data->dai.cpu_dai = fsl_ssi_create_dai(&ssi_info);
+ machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */
+ machine_data->dai.ops = &mpc8610_hpcd_ops;
+
+ mpc8610_hpcd_machine.dai_link = &machine_data->dai;
+
+ /* Allocate a new audio platform device structure */
+ sound_device = platform_device_alloc("soc-audio", -1);
+ if (!sound_device) {
+ dev_err(&ofdev->dev, "platform device allocation failed\n");
+ ret = -ENOMEM;
+ goto error;
+ }
+
+ machine_data->sound_devdata.machine = &mpc8610_hpcd_machine;
+ machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
+ machine_data->sound_devdata.platform = &fsl_soc_platform;
+
+ sound_device->dev.platform_data = machine_data;
+
+
+ /* Set the platform device and ASoC device to point to each other */
+ platform_set_drvdata(sound_device, &machine_data->sound_devdata);
+
+ machine_data->sound_devdata.dev = &sound_device->dev;
+
+
+ /* Tell ASoC to probe us. This will call mpc8610_hpcd_machine.probe(),
+ if it exists. */
+ ret = platform_device_add(sound_device);
+
+ if (ret) {
+ dev_err(&ofdev->dev, "platform device add failed\n");
+ goto error;
+ }
+
+ dev_set_drvdata(&ofdev->dev, sound_device);
+
+ return 0;
+
+error:
+ of_node_put(codec_np);
+ of_node_put(guts_np);
+ of_node_put(dma_np);
+ of_node_put(dma_channel_np);
+
+ if (sound_device)
+ platform_device_unregister(sound_device);
+
+ if (machine_data->dai.cpu_dai)
+ fsl_ssi_destroy_dai(machine_data->dai.cpu_dai);
+
+ if (ssi_info.ssi)
+ iounmap(ssi_info.ssi);
+
+ if (ssi_info.irq)
+ irq_dispose_mapping(ssi_info.irq);
+
+ if (dma_info.dma_channel[0])
+ iounmap(dma_info.dma_channel[0]);
+
+ if (dma_info.dma_channel[1])
+ iounmap(dma_info.dma_channel[1]);
+
+ if (dma_info.dma_irq[0])
+ irq_dispose_mapping(dma_info.dma_irq[0]);
+
+ if (dma_info.dma_irq[1])
+ irq_dispose_mapping(dma_info.dma_irq[1]);
+
+ if (machine_data->guts)
+ iounmap(machine_data->guts);
+
+ kfree(machine_data);
+
+ return ret;
+}
+
+/**
+ * mpc8610_hpcd_remove: remove the OF device
+ *
+ * This function is called when the OF device is removed.
+ */
+static int mpc8610_hpcd_remove(struct of_device *ofdev)
+{
+ struct platform_device *sound_device = dev_get_drvdata(&ofdev->dev);
+ struct mpc8610_hpcd_data *machine_data =
+ sound_device->dev.platform_data;
+
+ platform_device_unregister(sound_device);
+
+ if (machine_data->dai.cpu_dai)
+ fsl_ssi_destroy_dai(machine_data->dai.cpu_dai);
+
+ if (machine_data->ssi)
+ iounmap(machine_data->ssi);
+
+ if (machine_data->dma[0])
+ iounmap(machine_data->dma[0]);
+
+ if (machine_data->dma[1])
+ iounmap(machine_data->dma[1]);
+
+ if (machine_data->dma_irq[0])
+ irq_dispose_mapping(machine_data->dma_irq[0]);
+
+ if (machine_data->dma_irq[1])
+ irq_dispose_mapping(machine_data->dma_irq[1]);
+
+ if (machine_data->guts)
+ iounmap(machine_data->guts);
+
+ kfree(machine_data);
+ sound_device->dev.platform_data = NULL;
+
+ dev_set_drvdata(&ofdev->dev, NULL);
+
+ return 0;
+}
+
+static struct of_device_id mpc8610_hpcd_match[] = {
+ {
+ .compatible = "fsl,mpc8610-ssi",
+ },
+ {}
+};
+MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match);
+
+static struct of_platform_driver mpc8610_hpcd_of_driver = {
+ .owner = THIS_MODULE,
+ .name = "mpc8610_hpcd",
+ .match_table = mpc8610_hpcd_match,
+ .probe = mpc8610_hpcd_probe,
+ .remove = mpc8610_hpcd_remove,
+};
+
+/**
+ * mpc8610_hpcd_init: fabric driver initialization.
+ *
+ * This function is called when this module is loaded.
+ */
+static int __init mpc8610_hpcd_init(void)
+{
+ int ret;
+
+ printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n");
+
+ ret = of_register_platform_driver(&mpc8610_hpcd_of_driver);
+
+ if (ret)
+ printk(KERN_ERR
+ "mpc8610-hpcd: failed to register platform driver\n");
+
+ return ret;
+}
+
+/**
+ * mpc8610_hpcd_exit: fabric driver exit
+ *
+ * This function is called when this driver is unloaded.
+ */
+static void __exit mpc8610_hpcd_exit(void)
+{
+ of_unregister_platform_driver(&mpc8610_hpcd_of_driver);
+}
+
+module_init(mpc8610_hpcd_init);
+module_exit(mpc8610_hpcd_exit);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c
new file mode 100644
index 0000000..0382fda
--- /dev/null
+++ b/sound/soc/fsl/soc-of-simple.c
@@ -0,0 +1,171 @@
+/*
+ * OF helpers for ALSA SoC Layer
+ *
+ * Copyright (C) 2008, Secret Lab Technologies Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/of.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+#include <sound/initval.h>
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings");
+
+static DEFINE_MUTEX(of_snd_soc_mutex);
+static LIST_HEAD(of_snd_soc_device_list);
+static int of_snd_soc_next_index;
+
+struct of_snd_soc_device {
+ int id;
+ struct list_head list;
+ struct snd_soc_device device;
+ struct snd_soc_machine machine;
+ struct snd_soc_dai_link dai_link;
+ struct platform_device *pdev;
+ struct device_node *platform_node;
+ struct device_node *codec_node;
+};
+
+static struct snd_soc_ops of_snd_soc_ops = {
+};
+
+static struct of_snd_soc_device *
+of_snd_soc_get_device(struct device_node *codec_node)
+{
+ struct of_snd_soc_device *of_soc;
+
+ list_for_each_entry(of_soc, &of_snd_soc_device_list, list) {
+ if (of_soc->codec_node == codec_node)
+ return of_soc;
+ }
+
+ of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL);
+ if (!of_soc)
+ return NULL;
+
+ /* Initialize the structure and add it to the global list */
+ of_soc->codec_node = codec_node;
+ of_soc->id = of_snd_soc_next_index++;
+ of_soc->machine.dai_link = &of_soc->dai_link;
+ of_soc->machine.num_links = 1;
+ of_soc->device.machine = &of_soc->machine;
+ of_soc->dai_link.ops = &of_snd_soc_ops;
+ list_add(&of_soc->list, &of_snd_soc_device_list);
+
+ return of_soc;
+}
+
+static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc)
+{
+ struct platform_device *pdev;
+ int rc;
+
+ /* Only register the device if both the codec and platform have
+ * been registered */
+ if ((!of_soc->device.codec_data) || (!of_soc->platform_node))
+ return;
+
+ pr_info("platform<-->codec match achieved; registering machine\n");
+
+ pdev = platform_device_alloc("soc-audio", of_soc->id);
+ if (!pdev) {
+ pr_err("of_soc: platform_device_alloc() failed\n");
+ return;
+ }
+
+ pdev->dev.platform_data = of_soc;
+ platform_set_drvdata(pdev, &of_soc->device);
+ of_soc->device.dev = &pdev->dev;
+
+ /* The ASoC device is complete; register it */
+ rc = platform_device_add(pdev);
+ if (rc) {
+ pr_err("of_soc: platform_device_add() failed\n");
+ return;
+ }
+
+}
+
+int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev,
+ void *codec_data, struct snd_soc_dai *dai,
+ struct device_node *node)
+{
+ struct of_snd_soc_device *of_soc;
+ int rc = 0;
+
+ pr_info("registering ASoC codec driver: %s\n", node->full_name);
+
+ mutex_lock(&of_snd_soc_mutex);
+ of_soc = of_snd_soc_get_device(node);
+ if (!of_soc) {
+ rc = -ENOMEM;
+ goto out;
+ }
+
+ /* Store the codec data */
+ of_soc->device.codec_data = codec_data;
+ of_soc->device.codec_dev = codec_dev;
+ of_soc->dai_link.name = (char *)node->name;
+ of_soc->dai_link.stream_name = (char *)node->name;
+ of_soc->dai_link.codec_dai = dai;
+
+ /* Now try to register the SoC device */
+ of_snd_soc_register_device(of_soc);
+
+ out:
+ mutex_unlock(&of_snd_soc_mutex);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(of_snd_soc_register_codec);
+
+int of_snd_soc_register_platform(struct snd_soc_platform *platform,
+ struct device_node *node,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct of_snd_soc_device *of_soc;
+ struct device_node *codec_node;
+ const phandle *handle;
+ int len, rc = 0;
+
+ pr_info("registering ASoC platform driver: %s\n", node->full_name);
+
+ handle = of_get_property(node, "codec-handle", &len);
+ if (!handle || len < sizeof(handle))
+ return -ENODEV;
+ codec_node = of_find_node_by_phandle(*handle);
+ if (!codec_node)
+ return -ENODEV;
+ pr_info("looking for codec: %s\n", codec_node->full_name);
+
+ mutex_lock(&of_snd_soc_mutex);
+ of_soc = of_snd_soc_get_device(codec_node);
+ if (!of_soc) {
+ rc = -ENOMEM;
+ goto out;
+ }
+
+ of_soc->platform_node = node;
+ of_soc->dai_link.cpu_dai = cpu_dai;
+ of_soc->device.platform = platform;
+ of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+
+ /* Now try to register the SoC device */
+ of_snd_soc_register_device(of_soc);
+
+ out:
+ mutex_unlock(&of_snd_soc_mutex);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
new file mode 100644
index 0000000..8b7766b
--- /dev/null
+++ b/sound/soc/omap/Kconfig
@@ -0,0 +1,23 @@
+config SND_OMAP_SOC
+ tristate "SoC Audio for the Texas Instruments OMAP chips"
+ depends on ARCH_OMAP && SND_SOC
+
+config SND_OMAP_SOC_MCBSP
+ tristate
+ select OMAP_MCBSP
+
+config SND_OMAP_SOC_N810
+ tristate "SoC Audio support for Nokia N810"
+ depends on SND_OMAP_SOC && MACH_NOKIA_N810
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TLV320AIC3X
+ help
+ Say Y if you want to add support for SoC audio on Nokia N810.
+
+config SND_OMAP_SOC_OSK5912
+ tristate "SoC Audio support for omap osk5912"
+ depends on SND_OMAP_SOC && MACH_OMAP_OSK
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TLV320AIC23
+ help
+ Say Y if you want to add support for SoC audio on osk5912.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
new file mode 100644
index 0000000..e09d1f2
--- /dev/null
+++ b/sound/soc/omap/Makefile
@@ -0,0 +1,13 @@
+# OMAP Platform Support
+snd-soc-omap-objs := omap-pcm.o
+snd-soc-omap-mcbsp-objs := omap-mcbsp.o
+
+obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
+obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+
+# OMAP Machine Support
+snd-soc-n810-objs := n810.o
+snd-soc-osk5912-objs := osk5912.o
+
+obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
new file mode 100644
index 0000000..fae3ad3
--- /dev/null
+++ b/sound/soc/omap/n810.c
@@ -0,0 +1,391 @@
+/*
+ * n810.c -- SoC audio for Nokia N810
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic3x.h"
+
+#define N810_HEADSET_AMP_GPIO 10
+#define N810_SPEAKER_AMP_GPIO 101
+
+static struct clk *sys_clkout2;
+static struct clk *sys_clkout2_src;
+static struct clk *func96m_clk;
+
+static int n810_spk_func;
+static int n810_jack_func;
+static int n810_dmic_func;
+
+static void n810_ext_control(struct snd_soc_codec *codec)
+{
+ if (n810_spk_func)
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
+
+ if (n810_jack_func)
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+ if (n810_dmic_func)
+ snd_soc_dapm_enable_pin(codec, "DMic");
+ else
+ snd_soc_dapm_disable_pin(codec, "DMic");
+
+ snd_soc_dapm_sync(codec);
+}
+
+static int n810_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ n810_ext_control(codec);
+ return clk_enable(sys_clkout2);
+}
+
+static void n810_shutdown(struct snd_pcm_substream *substream)
+{
+ clk_disable(sys_clkout2);
+}
+
+static int n810_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int err;
+
+ /* Set codec DAI configuration */
+ err = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0)
+ return err;
+
+ /* Set cpu DAI configuration */
+ err = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0)
+ return err;
+
+ /* Set the codec system clock for DAC and ADC */
+ err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
+ SND_SOC_CLOCK_IN);
+
+ return err;
+}
+
+static struct snd_soc_ops n810_ops = {
+ .startup = n810_startup,
+ .hw_params = n810_hw_params,
+ .shutdown = n810_shutdown,
+};
+
+static int n810_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = n810_spk_func;
+
+ return 0;
+}
+
+static int n810_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (n810_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ n810_spk_func = ucontrol->value.integer.value[0];
+ n810_ext_control(codec);
+
+ return 1;
+}
+
+static int n810_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = n810_jack_func;
+
+ return 0;
+}
+
+static int n810_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (n810_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ n810_jack_func = ucontrol->value.integer.value[0];
+ n810_ext_control(codec);
+
+ return 1;
+}
+
+static int n810_get_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = n810_dmic_func;
+
+ return 0;
+}
+
+static int n810_set_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (n810_dmic_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ n810_dmic_func = ucontrol->value.integer.value[0];
+ n810_ext_control(codec);
+
+ return 1;
+}
+
+static int n810_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
+ else
+ gpio_set_value(N810_SPEAKER_AMP_GPIO, 0);
+
+ return 0;
+}
+
+static int n810_jack_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
+ else
+ gpio_set_value(N810_HEADSET_AMP_GPIO, 0);
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
+ SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
+ SND_SOC_DAPM_MIC("DMic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "HPLOUT"},
+ {"Headphone Jack", NULL, "HPROUT"},
+
+ {"Ext Spk", NULL, "LLOUT"},
+ {"Ext Spk", NULL, "RLOUT"},
+
+ {"DMic Rate 64", NULL, "Mic Bias 2V"},
+ {"Mic Bias 2V", NULL, "DMic"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *jack_function[] = {"Off", "Headphone"};
+static const char *input_function[] = {"ADC", "Digital Mic"};
+static const struct soc_enum n810_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+};
+
+static const struct snd_kcontrol_new aic33_n810_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", n810_enum[0],
+ n810_get_spk, n810_set_spk),
+ SOC_ENUM_EXT("Jack Function", n810_enum[1],
+ n810_get_jack, n810_set_jack),
+ SOC_ENUM_EXT("Input Select", n810_enum[2],
+ n810_get_input, n810_set_input),
+};
+
+static int n810_aic33_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ /* Not connected */
+ snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
+ snd_soc_dapm_nc_pin(codec, "HPLCOM");
+ snd_soc_dapm_nc_pin(codec, "HPRCOM");
+
+ /* Add N810 specific controls */
+ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&aic33_n810_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add N810 specific widgets */
+ snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
+ ARRAY_SIZE(aic33_dapm_widgets));
+
+ /* Set up N810 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link n810_dai = {
+ .name = "TLV320AIC33",
+ .stream_name = "AIC33",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &aic3x_dai,
+ .init = n810_aic33_init,
+ .ops = &n810_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_n810 = {
+ .name = "N810",
+ .dai_link = &n810_dai,
+ .num_links = 1,
+};
+
+/* Audio private data */
+static struct aic3x_setup_data n810_aic33_setup = {
+ .i2c_bus = 2,
+ .i2c_address = 0x18,
+ .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
+ .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device n810_snd_devdata = {
+ .machine = &snd_soc_machine_n810,
+ .platform = &omap_soc_platform,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &n810_aic33_setup,
+};
+
+static struct platform_device *n810_snd_device;
+
+static int __init n810_soc_init(void)
+{
+ int err;
+ struct device *dev;
+
+ if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
+ return -ENODEV;
+
+ n810_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!n810_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(n810_snd_device, &n810_snd_devdata);
+ n810_snd_devdata.dev = &n810_snd_device->dev;
+ *(unsigned int *)n810_dai.cpu_dai->private_data = 1; /* McBSP2 */
+ err = platform_device_add(n810_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &n810_snd_device->dev;
+
+ sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
+ if (IS_ERR(sys_clkout2_src)) {
+ dev_err(dev, "Could not get sys_clkout2_src clock\n");
+ err = PTR_ERR(sys_clkout2_src);
+ goto err2;
+ }
+ sys_clkout2 = clk_get(dev, "sys_clkout2");
+ if (IS_ERR(sys_clkout2)) {
+ dev_err(dev, "Could not get sys_clkout2\n");
+ err = PTR_ERR(sys_clkout2);
+ goto err3;
+ }
+ /*
+ * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
+ * 96 MHz as its parent in order to get 12 MHz
+ */
+ func96m_clk = clk_get(dev, "func_96m_ck");
+ if (IS_ERR(func96m_clk)) {
+ dev_err(dev, "Could not get func 96M clock\n");
+ err = PTR_ERR(func96m_clk);
+ goto err4;
+ }
+ clk_set_parent(sys_clkout2_src, func96m_clk);
+ clk_set_rate(sys_clkout2, 12000000);
+
+ if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
+ BUG();
+ if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
+ BUG();
+ gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
+ gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
+
+ return 0;
+err4:
+ clk_put(sys_clkout2);
+err3:
+ clk_put(sys_clkout2_src);
+err2:
+ platform_device_del(n810_snd_device);
+err1:
+ platform_device_put(n810_snd_device);
+
+ return err;
+}
+
+static void __exit n810_soc_exit(void)
+{
+ gpio_free(N810_SPEAKER_AMP_GPIO);
+ gpio_free(N810_HEADSET_AMP_GPIO);
+ clk_put(sys_clkout2_src);
+ clk_put(sys_clkout2);
+ clk_put(func96m_clk);
+
+ platform_device_unregister(n810_snd_device);
+}
+
+module_init(n810_soc_init);
+module_exit(n810_soc_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("ALSA SoC Nokia N810");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
new file mode 100644
index 0000000..8485a8a
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -0,0 +1,500 @@
+/*
+ * omap-mcbsp.c -- OMAP ALSA SoC DAI driver using McBSP port
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/control.h>
+#include <mach/dma.h>
+#include <mach/mcbsp.h>
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_KNOT)
+
+struct omap_mcbsp_data {
+ unsigned int bus_id;
+ struct omap_mcbsp_reg_cfg regs;
+ unsigned int fmt;
+ /*
+ * Flags indicating is the bus already activated and configured by
+ * another substream
+ */
+ int active;
+ int configured;
+};
+
+#define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id)
+
+static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
+
+/*
+ * Stream DMA parameters. DMA request line and port address are set runtime
+ * since they are different between OMAP1 and later OMAPs
+ */
+static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2];
+
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+static const int omap1_dma_reqs[][2] = {
+ { OMAP_DMA_MCBSP1_TX, OMAP_DMA_MCBSP1_RX },
+ { OMAP_DMA_MCBSP2_TX, OMAP_DMA_MCBSP2_RX },
+ { OMAP_DMA_MCBSP3_TX, OMAP_DMA_MCBSP3_RX },
+};
+static const unsigned long omap1_mcbsp_port[][2] = {
+ { OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
+ { OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
+ { OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DRR1 },
+};
+#else
+static const int omap1_dma_reqs[][2] = {};
+static const unsigned long omap1_mcbsp_port[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
+static const int omap24xx_dma_reqs[][2] = {
+ { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
+ { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+ { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
+ { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
+ { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
+#endif
+};
+#else
+static const int omap24xx_dma_reqs[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP2420)
+static const unsigned long omap2420_mcbsp_port[][2] = {
+ { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
+ { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1,
+ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
+};
+#else
+static const unsigned long omap2420_mcbsp_port[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP2430)
+static const unsigned long omap2430_mcbsp_port[][2] = {
+ { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap2430_mcbsp_port[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP34XX)
+static const unsigned long omap34xx_mcbsp_port[][2] = {
+ { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+ { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+ OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap34xx_mcbsp_port[][2] = {};
+#endif
+
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int err = 0;
+
+ if (!cpu_dai->active)
+ err = omap_mcbsp_request(mcbsp_data->bus_id);
+
+ return err;
+}
+
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+
+ if (!cpu_dai->active) {
+ omap_mcbsp_free(mcbsp_data->bus_id);
+ mcbsp_data->configured = 0;
+ }
+}
+
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!mcbsp_data->active++)
+ omap_mcbsp_start(mcbsp_data->bus_id);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (!--mcbsp_data->active)
+ omap_mcbsp_stop(mcbsp_data->bus_id);
+ break;
+ default:
+ err = -EINVAL;
+ }
+
+ return err;
+}
+
+static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+ int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+ int wlen;
+ unsigned long port;
+
+ if (cpu_class_is_omap1()) {
+ dma = omap1_dma_reqs[bus_id][substream->stream];
+ port = omap1_mcbsp_port[bus_id][substream->stream];
+ } else if (cpu_is_omap2420()) {
+ dma = omap24xx_dma_reqs[bus_id][substream->stream];
+ port = omap2420_mcbsp_port[bus_id][substream->stream];
+ } else if (cpu_is_omap2430()) {
+ dma = omap24xx_dma_reqs[bus_id][substream->stream];
+ port = omap2430_mcbsp_port[bus_id][substream->stream];
+ } else if (cpu_is_omap343x()) {
+ dma = omap24xx_dma_reqs[bus_id][substream->stream];
+ port = omap34xx_mcbsp_port[bus_id][substream->stream];
+ } else {
+ return -ENODEV;
+ }
+ omap_mcbsp_dai_dma_params[id][substream->stream].name =
+ substream->stream ? "Audio Capture" : "Audio Playback";
+ omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
+ omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
+ cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
+
+ if (mcbsp_data->configured) {
+ /* McBSP already configured by another stream */
+ return 0;
+ }
+
+ switch (params_channels(params)) {
+ case 2:
+ /* Set 1 word per (McBPSP) frame and use dual-phase frames */
+ regs->rcr2 |= RFRLEN2(1 - 1) | RPHASE;
+ regs->rcr1 |= RFRLEN1(1 - 1);
+ regs->xcr2 |= XFRLEN2(1 - 1) | XPHASE;
+ regs->xcr1 |= XFRLEN1(1 - 1);
+ break;
+ default:
+ /* Unsupported number of channels */
+ return -EINVAL;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ /* Set word lengths */
+ wlen = 16;
+ regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16);
+ regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16);
+ regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16);
+ regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16);
+ break;
+ default:
+ /* Unsupported PCM format */
+ return -EINVAL;
+ }
+
+ /* Set FS period and length in terms of bit clock periods */
+ switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr1 |= FWID(wlen - 1);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr1 |= FWID(wlen * 2 - 2);
+ break;
+ }
+
+ omap_mcbsp_config(bus_id, &mcbsp_data->regs);
+ mcbsp_data->configured = 1;
+
+ return 0;
+}
+
+/*
+ * This must be called before _set_clkdiv and _set_sysclk since McBSP register
+ * cache is initialized here
+ */
+static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+
+ if (mcbsp_data->configured)
+ return 0;
+
+ mcbsp_data->fmt = fmt;
+ memset(regs, 0, sizeof(*regs));
+ /* Generic McBSP register settings */
+ regs->spcr2 |= XINTM(3) | FREE;
+ regs->spcr1 |= RINTM(3);
+ regs->rcr2 |= RFIG;
+ regs->xcr2 |= XFIG;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* 1-bit data delay */
+ regs->rcr2 |= RDATDLY(1);
+ regs->xcr2 |= XDATDLY(1);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* 0-bit data delay */
+ regs->rcr2 |= RDATDLY(0);
+ regs->xcr2 |= XDATDLY(0);
+ break;
+ default:
+ /* Unsupported data format */
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ /* McBSP master. Set FS and bit clocks as outputs */
+ regs->pcr0 |= FSXM | FSRM |
+ CLKXM | CLKRM;
+ /* Sample rate generator drives the FS */
+ regs->srgr2 |= FSGM;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* McBSP slave */
+ break;
+ default:
+ /* Unsupported master/slave configuration */
+ return -EINVAL;
+ }
+
+ /* Set bit clock (CLKX/CLKR) and FS polarities */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ /*
+ * Normal BCLK + FS.
+ * FS active low. TX data driven on falling edge of bit clock
+ * and RX data sampled on rising edge of bit clock.
+ */
+ regs->pcr0 |= FSXP | FSRP |
+ CLKXP | CLKRP;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ regs->pcr0 |= CLKXP | CLKRP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ regs->pcr0 |= FSXP | FSRP;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+
+ if (div_id != OMAP_MCBSP_CLKGDV)
+ return -ENODEV;
+
+ regs->srgr1 |= CLKGDV(div - 1);
+
+ return 0;
+}
+
+static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
+ int clk_id)
+{
+ int sel_bit;
+ u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1;
+
+ if (cpu_class_is_omap1()) {
+ /* OMAP1's can use only external source clock */
+ if (unlikely(clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK))
+ return -EINVAL;
+ else
+ return 0;
+ }
+
+ if (cpu_is_omap2420() && mcbsp_data->bus_id > 1)
+ return -EINVAL;
+
+ if (cpu_is_omap343x())
+ reg_devconf1 = OMAP343X_CONTROL_DEVCONF1;
+
+ switch (mcbsp_data->bus_id) {
+ case 0:
+ reg = OMAP2_CONTROL_DEVCONF0;
+ sel_bit = 2;
+ break;
+ case 1:
+ reg = OMAP2_CONTROL_DEVCONF0;
+ sel_bit = 6;
+ break;
+ case 2:
+ reg = reg_devconf1;
+ sel_bit = 0;
+ break;
+ case 3:
+ reg = reg_devconf1;
+ sel_bit = 2;
+ break;
+ case 4:
+ reg = reg_devconf1;
+ sel_bit = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK)
+ omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
+ else
+ omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
+
+ return 0;
+}
+
+static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq,
+ int dir)
+{
+ struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+ int err = 0;
+
+ switch (clk_id) {
+ case OMAP_MCBSP_SYSCLK_CLK:
+ regs->srgr2 |= CLKSM;
+ break;
+ case OMAP_MCBSP_SYSCLK_CLKS_FCLK:
+ case OMAP_MCBSP_SYSCLK_CLKS_EXT:
+ err = omap_mcbsp_dai_set_clks_src(mcbsp_data, clk_id);
+ break;
+
+ case OMAP_MCBSP_SYSCLK_CLKX_EXT:
+ regs->srgr2 |= CLKSM;
+ case OMAP_MCBSP_SYSCLK_CLKR_EXT:
+ regs->pcr0 |= SCLKME;
+ break;
+ default:
+ err = -ENODEV;
+ }
+
+ return err;
+}
+
+#define OMAP_MCBSP_DAI_BUILDER(link_id) \
+{ \
+ .name = "omap-mcbsp-dai-(link_id)", \
+ .id = (link_id), \
+ .type = SND_SOC_DAI_I2S, \
+ .playback = { \
+ .channels_min = 2, \
+ .channels_max = 2, \
+ .rates = OMAP_MCBSP_RATES, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, \
+ }, \
+ .capture = { \
+ .channels_min = 2, \
+ .channels_max = 2, \
+ .rates = OMAP_MCBSP_RATES, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, \
+ }, \
+ .ops = { \
+ .startup = omap_mcbsp_dai_startup, \
+ .shutdown = omap_mcbsp_dai_shutdown, \
+ .trigger = omap_mcbsp_dai_trigger, \
+ .hw_params = omap_mcbsp_dai_hw_params, \
+ }, \
+ .dai_ops = { \
+ .set_fmt = omap_mcbsp_dai_set_dai_fmt, \
+ .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \
+ .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \
+ }, \
+ .private_data = &mcbsp_data[(link_id)].bus_id, \
+}
+
+struct snd_soc_dai omap_mcbsp_dai[] = {
+ OMAP_MCBSP_DAI_BUILDER(0),
+ OMAP_MCBSP_DAI_BUILDER(1),
+#if NUM_LINKS >= 3
+ OMAP_MCBSP_DAI_BUILDER(2),
+#endif
+#if NUM_LINKS == 5
+ OMAP_MCBSP_DAI_BUILDER(3),
+ OMAP_MCBSP_DAI_BUILDER(4),
+#endif
+};
+
+EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("OMAP I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
new file mode 100644
index 0000000..df7ad13
--- /dev/null
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -0,0 +1,55 @@
+/*
+ * omap-mcbsp.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_I2S_H__
+#define __OMAP_I2S_H__
+
+/* Source clocks for McBSP sample rate generator */
+enum omap_mcbsp_clksrg_clk {
+ OMAP_MCBSP_SYSCLK_CLKS_FCLK, /* Internal FCLK */
+ OMAP_MCBSP_SYSCLK_CLKS_EXT, /* External CLKS pin */
+ OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */
+ OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */
+ OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */
+};
+
+/* McBSP dividers */
+enum omap_mcbsp_div {
+ OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */
+};
+
+#if defined(CONFIG_ARCH_OMAP2420)
+#define NUM_LINKS 2
+#endif
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+#undef NUM_LINKS
+#define NUM_LINKS 3
+#endif
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#undef NUM_LINKS
+#define NUM_LINKS 5
+#endif
+
+extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
+
+#endif
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
new file mode 100644
index 0000000..acd68ef
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.c
@@ -0,0 +1,359 @@
+/*
+ * omap-pcm.c -- ALSA PCM interface for the OMAP SoC
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/dma.h>
+#include "omap-pcm.h"
+
+static const struct snd_pcm_hardware omap_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = 64 * 1024,
+ .periods_min = 2,
+ .periods_max = 255,
+ .buffer_bytes_max = 128 * 1024,
+};
+
+struct omap_runtime_data {
+ spinlock_t lock;
+ struct omap_pcm_dma_data *dma_data;
+ int dma_ch;
+ int period_index;
+};
+
+static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
+{
+ struct snd_pcm_substream *substream = data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ unsigned long flags;
+
+ if (cpu_is_omap1510()) {
+ /*
+ * OMAP1510 doesn't support DMA chaining so have to restart
+ * the transfer after all periods are transferred
+ */
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (prtd->period_index >= 0) {
+ if (++prtd->period_index == runtime->periods) {
+ prtd->period_index = 0;
+ omap_start_dma(prtd->dma_ch);
+ }
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ }
+
+ snd_pcm_period_elapsed(substream);
+}
+
+/* this may get called several times by oss emulation */
+static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
+ int err = 0;
+
+ if (!dma_data)
+ return -ENODEV;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ if (prtd->dma_data)
+ return 0;
+ prtd->dma_data = dma_data;
+ err = omap_request_dma(dma_data->dma_req, dma_data->name,
+ omap_pcm_dma_irq, substream, &prtd->dma_ch);
+ if (!err & !cpu_is_omap1510()) {
+ /*
+ * Link channel with itself so DMA doesn't need any
+ * reprogramming while looping the buffer
+ */
+ omap_dma_link_lch(prtd->dma_ch, prtd->dma_ch);
+ }
+
+ return err;
+}
+
+static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+
+ if (prtd->dma_data == NULL)
+ return 0;
+
+ if (!cpu_is_omap1510())
+ omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
+ omap_free_dma(prtd->dma_ch);
+ prtd->dma_data = NULL;
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ return 0;
+}
+
+static int omap_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ struct omap_pcm_dma_data *dma_data = prtd->dma_data;
+ struct omap_dma_channel_params dma_params;
+
+ memset(&dma_params, 0, sizeof(dma_params));
+ /*
+ * Note: Regardless of interface data formats supported by OMAP McBSP
+ * or EAC blocks, internal representation is always fixed 16-bit/sample
+ */
+ dma_params.data_type = OMAP_DMA_DATA_TYPE_S16;
+ dma_params.trigger = dma_data->dma_req;
+ dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ dma_params.src_amode = OMAP_DMA_AMODE_POST_INC;
+ dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT;
+ dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC;
+ dma_params.src_start = runtime->dma_addr;
+ dma_params.dst_start = dma_data->port_addr;
+ dma_params.dst_port = OMAP_DMA_PORT_MPUI;
+ } else {
+ dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT;
+ dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
+ dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC;
+ dma_params.src_start = dma_data->port_addr;
+ dma_params.dst_start = runtime->dma_addr;
+ dma_params.src_port = OMAP_DMA_PORT_MPUI;
+ }
+ /*
+ * Set DMA transfer frame size equal to ALSA period size and frame
+ * count as no. of ALSA periods. Then with DMA frame interrupt enabled,
+ * we can transfer the whole ALSA buffer with single DMA transfer but
+ * still can get an interrupt at each period bounary
+ */
+ dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2;
+ dma_params.frame_count = runtime->periods;
+ omap_set_dma_params(prtd->dma_ch, &dma_params);
+
+ omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+
+ return 0;
+}
+
+static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ int ret = 0;
+
+ spin_lock_irq(&prtd->lock);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ prtd->period_index = 0;
+ omap_start_dma(prtd->dma_ch);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ prtd->period_index = -1;
+ omap_stop_dma(prtd->dma_ch);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ spin_unlock_irq(&prtd->lock);
+
+ return ret;
+}
+
+static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd = runtime->private_data;
+ dma_addr_t ptr;
+ snd_pcm_uframes_t offset;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ptr = omap_get_dma_src_pos(prtd->dma_ch);
+ else
+ ptr = omap_get_dma_dst_pos(prtd->dma_ch);
+
+ offset = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+
+ return offset;
+}
+
+static int omap_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct omap_runtime_data *prtd;
+ int ret;
+
+ snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware);
+
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ spin_lock_init(&prtd->lock);
+ runtime->private_data = prtd;
+
+out:
+ return ret;
+}
+
+static int omap_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ kfree(runtime->private_data);
+ return 0;
+}
+
+static int omap_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+struct snd_pcm_ops omap_pcm_ops = {
+ .open = omap_pcm_open,
+ .close = omap_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = omap_pcm_hw_params,
+ .hw_free = omap_pcm_hw_free,
+ .prepare = omap_pcm_prepare,
+ .trigger = omap_pcm_trigger,
+ .pointer = omap_pcm_pointer,
+ .mmap = omap_pcm_mmap,
+};
+
+static u64 omap_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = omap_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &omap_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+ if (dai->playback.channels_min) {
+ ret = omap_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = omap_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+
+out:
+ return ret;
+}
+
+struct snd_soc_platform omap_soc_platform = {
+ .name = "omap-pcm-audio",
+ .pcm_ops = &omap_pcm_ops,
+ .pcm_new = omap_pcm_new,
+ .pcm_free = omap_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(omap_soc_platform);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_DESCRIPTION("OMAP PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
new file mode 100644
index 0000000..e4369bd
--- /dev/null
+++ b/sound/soc/omap/omap-pcm.h
@@ -0,0 +1,35 @@
+/*
+ * omap-pcm.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_PCM_H__
+#define __OMAP_PCM_H__
+
+struct omap_pcm_dma_data {
+ char *name; /* stream identifier */
+ int dma_req; /* DMA request line */
+ unsigned long port_addr; /* transmit/receive register */
+};
+
+extern struct snd_soc_platform omap_soc_platform;
+
+#endif
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
new file mode 100644
index 0000000..0fe7337
--- /dev/null
+++ b/sound/soc/omap/osk5912.c
@@ -0,0 +1,232 @@
+/*
+ * osk5912.c -- SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS <arunks@mistralsolutions.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 12000000
+
+static struct clk *tlv320aic23_mclk;
+
+static int osk_startup(struct snd_pcm_substream *substream)
+{
+ return clk_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+ clk_disable(tlv320aic23_mclk);
+}
+
+static int osk_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int err;
+
+ /* Set codec DAI configuration */
+ err = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return err;
+ }
+
+ /* Set cpu DAI configuration */
+ err = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return err;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ err =
+ snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return err;
+ }
+
+ return err;
+}
+
+static struct snd_soc_ops osk_ops = {
+ .startup = osk_startup,
+ .hw_params = osk_hw_params,
+ .shutdown = osk_shutdown,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ {"LLINEIN", NULL, "Line In"},
+ {"RLINEIN", NULL, "Line In"},
+
+ {"MICIN", NULL, "Mic Jack"},
+};
+
+static int osk_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+ /* Add osk5912 specific widgets */
+ snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+ /* Set up osk5912 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link osk_dai = {
+ .name = "TLV320AIC23",
+ .stream_name = "AIC23",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &tlv320aic23_dai,
+ .init = osk_tlv320aic23_init,
+ .ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_osk = {
+ .name = "OSK5912",
+ .dai_link = &osk_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device osk_snd_devdata = {
+ .machine = &snd_soc_machine_osk,
+ .platform = &omap_soc_platform,
+ .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *osk_snd_device;
+
+static int __init osk_soc_init(void)
+{
+ int err;
+ u32 curRate;
+ struct device *dev;
+
+ if (!(machine_is_omap_osk()))
+ return -ENODEV;
+
+ osk_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!osk_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
+ osk_snd_devdata.dev = &osk_snd_device->dev;
+ *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */
+ err = platform_device_add(osk_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &osk_snd_device->dev;
+
+ tlv320aic23_mclk = clk_get(dev, "mclk");
+ if (IS_ERR(tlv320aic23_mclk)) {
+ printk(KERN_ERR "Could not get mclk clock\n");
+ return -ENODEV;
+ }
+
+ if (clk_get_usecount(tlv320aic23_mclk) > 0) {
+ /* MCLK is already in use */
+ printk(KERN_WARNING
+ "MCLK in use at %d Hz. We change it to %d Hz\n",
+ (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+ }
+
+ /*
+ * Configure 12 MHz output on MCLK.
+ */
+ curRate = (uint) clk_get_rate(tlv320aic23_mclk);
+ if (curRate != CODEC_CLOCK) {
+ if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
+ printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
+ err = -ECANCELED;
+ goto err1;
+ }
+ }
+
+ printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
+ (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
+ clk_get_usecount(tlv320aic23_mclk));
+
+ return 0;
+err1:
+ clk_put(tlv320aic23_mclk);
+ platform_device_del(osk_snd_device);
+ platform_device_put(osk_snd_device);
+
+ return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+ platform_device_unregister(osk_snd_device);
+}
+
+module_init(osk_soc_init);
+module_exit(osk_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC OSK 5912");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
new file mode 100644
index 0000000..f8c1cdd
--- /dev/null
+++ b/sound/soc/pxa/Kconfig
@@ -0,0 +1,77 @@
+config SND_PXA2XX_SOC
+ tristate "SoC Audio for the Intel PXA2xx chip"
+ depends on ARCH_PXA
+ select SND_PXA2XX_LIB
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the PXA2xx AC97, I2S or SSP interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_PXA2XX_AC97
+ tristate
+ select SND_AC97_CODEC
+
+config SND_PXA2XX_SOC_AC97
+ tristate
+ select AC97_BUS
+ select SND_ARM
+ select SND_PXA2XX_LIB_AC97
+ select SND_SOC_AC97_BUS
+
+config SND_PXA2XX_SOC_I2S
+ tristate
+
+config SND_PXA2XX_SOC_CORGI
+ tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
+ depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-C7x0 models (Corgi, Shepherd, Husky).
+
+config SND_PXA2XX_SOC_SPITZ
+ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
+ depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8750
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
+
+config SND_PXA2XX_SOC_POODLE
+ tristate "SoC Audio support for Poodle"
+ depends on SND_PXA2XX_SOC && MACH_POODLE
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8731
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-5600 model (Poodle).
+
+config SND_PXA2XX_SOC_TOSA
+ tristate "SoC AC97 Audio support for Tosa"
+ depends on SND_PXA2XX_SOC && MACH_TOSA
+ depends on MFD_TC6393XB
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on Sharp
+ Zaurus SL-C6000x models (Tosa).
+
+config SND_PXA2XX_SOC_E800
+ tristate "SoC AC97 Audio support for e800"
+ depends on SND_PXA2XX_SOC && MACH_E800
+ select SND_SOC_WM9712
+ select SND_PXA2XX_SOC_AC97
+ help
+ Say Y if you want to add support for SoC audio on the
+ Toshiba e800 PDA
+
+config SND_PXA2XX_SOC_EM_X270
+ tristate "SoC Audio support for CompuLab EM-x270"
+ depends on SND_PXA2XX_SOC && MACH_EM_X270
+ select SND_PXA2XX_SOC_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for SoC audio on
+ CompuLab EM-x270.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
new file mode 100644
index 0000000..5bc8edf
--- /dev/null
+++ b/sound/soc/pxa/Makefile
@@ -0,0 +1,23 @@
+# PXA Platform Support
+snd-soc-pxa2xx-objs := pxa2xx-pcm.o
+snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
+snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
+obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
+obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+
+# PXA Machine Support
+snd-soc-corgi-objs := corgi.o
+snd-soc-poodle-objs := poodle.o
+snd-soc-tosa-objs := tosa.o
+snd-soc-e800-objs := e800_wm9712.o
+snd-soc-spitz-objs := spitz.o
+snd-soc-em-x270-objs := em-x270.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
+obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
+obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
+obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
+obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
new file mode 100644
index 0000000..2718eaf
--- /dev/null
+++ b/sound/soc/pxa/corgi.c
@@ -0,0 +1,372 @@
+/*
+ * corgi.c -- SoC audio for Corgi
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/corgi.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define CORGI_HP 0
+#define CORGI_MIC 1
+#define CORGI_LINE 2
+#define CORGI_HEADSET 3
+#define CORGI_HP_OFF 4
+#define CORGI_SPK_ON 0
+#define CORGI_SPK_OFF 1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define CORGI_AUDIO_CLOCK 12288000
+
+static int corgi_jack_func;
+static int corgi_spk_func;
+
+static void corgi_ext_control(struct snd_soc_codec *codec)
+{
+ /* set up jack connection */
+ switch (corgi_jack_func) {
+ case CORGI_HP:
+ /* set = unmute headphone */
+ gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ break;
+ case CORGI_MIC:
+ /* reset = mute headphone */
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 0);
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ break;
+ case CORGI_LINE:
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 0);
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ break;
+ case CORGI_HEADSET:
+ gpio_set_value(CORGI_GPIO_MUTE_L, 0);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
+ break;
+ }
+
+ if (corgi_spk_func == CORGI_SPK_ON)
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(codec);
+}
+
+static int corgi_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ corgi_ext_control(codec);
+ return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on corgi */
+static int corgi_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* set = unmute headphone */
+ gpio_set_value(CORGI_GPIO_MUTE_L, 1);
+ gpio_set_value(CORGI_GPIO_MUTE_R, 1);
+ return 0;
+}
+
+static int corgi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops corgi_ops = {
+ .startup = corgi_startup,
+ .hw_params = corgi_hw_params,
+ .shutdown = corgi_shutdown,
+};
+
+static int corgi_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = corgi_jack_func;
+ return 0;
+}
+
+static int corgi_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (corgi_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ corgi_jack_func = ucontrol->value.integer.value[0];
+ corgi_ext_control(codec);
+ return 1;
+}
+
+static int corgi_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = corgi_spk_func;
+ return 0;
+}
+
+static int corgi_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (corgi_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ corgi_spk_func = ucontrol->value.integer.value[0];
+ corgi_ext_control(codec);
+ return 1;
+}
+
+static int corgi_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+static int corgi_mic_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+ return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event),
+SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event),
+SND_SOC_DAPM_LINE("Line Jack", NULL),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Corgi machine audio map (connections to the codec pins) */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* headset Jack - in = micin, out = LHPOUT*/
+ {"Headset Jack", NULL, "LHPOUT"},
+
+ /* headphone connected to LHPOUT1, RHPOUT1 */
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ /* speaker connected to LOUT, ROUT */
+ {"Ext Spk", NULL, "ROUT"},
+ {"Ext Spk", NULL, "LOUT"},
+
+ /* mic is connected to MICIN (via right channel of headphone jack) */
+ {"MICIN", NULL, "Mic Jack"},
+
+ /* Same as the above but no mic bias for line signals */
+ {"MICIN", NULL, "Line Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum corgi_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
+ SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack,
+ corgi_set_jack),
+ SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk,
+ corgi_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int corgi_wm8731_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ snd_soc_dapm_nc_pin(codec, "LLINEIN");
+ snd_soc_dapm_nc_pin(codec, "RLINEIN");
+
+ /* Add corgi specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add corgi specific widgets */
+ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ ARRAY_SIZE(wm8731_dapm_widgets));
+
+ /* Set up corgi specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8731_dai,
+ .init = corgi_wm8731_init,
+ .ops = &corgi_ops,
+};
+
+/* corgi audio machine driver */
+static struct snd_soc_machine snd_soc_machine_corgi = {
+ .name = "Corgi",
+ .dai_link = &corgi_dai,
+ .num_links = 1,
+};
+
+/* corgi audio private data */
+static struct wm8731_setup_data corgi_wm8731_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1b,
+};
+
+/* corgi audio subsystem */
+static struct snd_soc_device corgi_snd_devdata = {
+ .machine = &snd_soc_machine_corgi,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8731,
+ .codec_data = &corgi_wm8731_setup,
+};
+
+static struct platform_device *corgi_snd_device;
+
+static int __init corgi_init(void)
+{
+ int ret;
+
+ if (!(machine_is_corgi() || machine_is_shepherd() ||
+ machine_is_husky()))
+ return -ENODEV;
+
+ corgi_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!corgi_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(corgi_snd_device, &corgi_snd_devdata);
+ corgi_snd_devdata.dev = &corgi_snd_device->dev;
+ ret = platform_device_add(corgi_snd_device);
+
+ if (ret)
+ platform_device_put(corgi_snd_device);
+
+ return ret;
+}
+
+static void __exit corgi_exit(void)
+{
+ platform_device_unregister(corgi_snd_device);
+}
+
+module_init(corgi_init);
+module_exit(corgi_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Corgi");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
new file mode 100644
index 0000000..6781c5b
--- /dev/null
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -0,0 +1,89 @@
+/*
+ * e800-wm9712.c -- SoC audio for e800
+ *
+ * Based on tosa.c
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_machine e800;
+
+static struct snd_soc_dai_link e800_dai[] = {
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+},
+};
+
+static struct snd_soc_machine e800 = {
+ .name = "Toshiba e800",
+ .dai_link = e800_dai,
+ .num_links = ARRAY_SIZE(e800_dai),
+};
+
+static struct snd_soc_device e800_snd_devdata = {
+ .machine = &e800,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *e800_snd_device;
+
+static int __init e800_init(void)
+{
+ int ret;
+
+ if (!machine_is_e800())
+ return -ENODEV;
+
+ e800_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!e800_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(e800_snd_device, &e800_snd_devdata);
+ e800_snd_devdata.dev = &e800_snd_device->dev;
+ ret = platform_device_add(e800_snd_device);
+
+ if (ret)
+ platform_device_put(e800_snd_device);
+
+ return ret;
+}
+
+static void __exit e800_exit(void)
+{
+ platform_device_unregister(e800_snd_device);
+}
+
+module_init(e800_init);
+module_exit(e800_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e800");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
new file mode 100644
index 0000000..e6ff692
--- /dev/null
+++ b/sound/soc/pxa/em-x270.c
@@ -0,0 +1,102 @@
+/*
+ * em-x270.c -- SoC audio for EM-X270
+ *
+ * Copyright 2007 CompuLab, Ltd.
+ *
+ * Author: Mike Rapoport <mike@compulab.co.il>
+ *
+ * Copied from tosa.c:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_dai_link em_x270_dai[] = {
+ {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ },
+ {
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ },
+};
+
+static struct snd_soc_machine em_x270 = {
+ .name = "EM-X270",
+ .dai_link = em_x270_dai,
+ .num_links = ARRAY_SIZE(em_x270_dai),
+};
+
+static struct snd_soc_device em_x270_snd_devdata = {
+ .machine = &em_x270,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *em_x270_snd_device;
+
+static int __init em_x270_init(void)
+{
+ int ret;
+
+ if (!machine_is_em_x270())
+ return -ENODEV;
+
+ em_x270_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!em_x270_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata);
+ em_x270_snd_devdata.dev = &em_x270_snd_device->dev;
+ ret = platform_device_add(em_x270_snd_device);
+
+ if (ret)
+ platform_device_put(em_x270_snd_device);
+
+ return ret;
+}
+
+static void __exit em_x270_exit(void)
+{
+ platform_device_unregister(em_x270_snd_device);
+}
+
+module_init(em_x270_init);
+module_exit(em_x270_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mike Rapoport");
+MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
new file mode 100644
index 0000000..4d9930c
--- /dev/null
+++ b/sound/soc/pxa/poodle.c
@@ -0,0 +1,341 @@
+/*
+ * poodle.c -- SoC audio for Poodle
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/locomo.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/poodle.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm8731.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define POODLE_HP 1
+#define POODLE_HP_OFF 0
+#define POODLE_SPK_ON 1
+#define POODLE_SPK_OFF 0
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define POODLE_AUDIO_CLOCK 12288000
+
+static int poodle_jack_func;
+static int poodle_spk_func;
+
+static void poodle_ext_control(struct snd_soc_codec *codec)
+{
+ /* set up jack connection */
+ if (poodle_jack_func == POODLE_HP) {
+ /* set = unmute headphone */
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 1);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 1);
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ } else {
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 0);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 0);
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ }
+
+ /* set the enpoints to their new connetion states */
+ if (poodle_spk_func == POODLE_SPK_ON)
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(codec);
+}
+
+static int poodle_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ poodle_ext_control(codec);
+ return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on poodle */
+static void poodle_shutdown(struct snd_pcm_substream *substream)
+{
+ /* set = unmute headphone */
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 1);
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 1);
+}
+
+static int poodle_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops poodle_ops = {
+ .startup = poodle_startup,
+ .hw_params = poodle_hw_params,
+ .shutdown = poodle_shutdown,
+};
+
+static int poodle_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = poodle_jack_func;
+ return 0;
+}
+
+static int poodle_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (poodle_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ poodle_jack_func = ucontrol->value.integer.value[0];
+ poodle_ext_control(codec);
+ return 1;
+}
+
+static int poodle_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = poodle_spk_func;
+ return 0;
+}
+
+static int poodle_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (poodle_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ poodle_spk_func = ucontrol->value.integer.value[0];
+ poodle_ext_control(codec);
+ return 1;
+}
+
+static int poodle_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 0);
+ else
+ locomo_gpio_write(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 1);
+
+ return 0;
+}
+
+/* poodle machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
+};
+
+/* Corgi machine connections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* headphone connected to LHPOUT1, RHPOUT1 */
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ /* speaker connected to LOUT, ROUT */
+ {"Ext Spk", NULL, "ROUT"},
+ {"Ext Spk", NULL, "LOUT"},
+};
+
+static const char *jack_function[] = {"Off", "Headphone"};
+static const char *spk_function[] = {"Off", "On"};
+static const struct soc_enum poodle_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
+ SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
+ poodle_set_jack),
+ SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
+ poodle_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int poodle_wm8731_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ snd_soc_dapm_nc_pin(codec, "LLINEIN");
+ snd_soc_dapm_nc_pin(codec, "RLINEIN");
+ snd_soc_dapm_enable_pin(codec, "MICIN");
+
+ /* Add poodle specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add poodle specific widgets */
+ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+ ARRAY_SIZE(wm8731_dapm_widgets));
+
+ /* Set up poodle specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+/* poodle digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link poodle_dai = {
+ .name = "WM8731",
+ .stream_name = "WM8731",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8731_dai,
+ .init = poodle_wm8731_init,
+ .ops = &poodle_ops,
+};
+
+/* poodle audio machine driver */
+static struct snd_soc_machine snd_soc_machine_poodle = {
+ .name = "Poodle",
+ .dai_link = &poodle_dai,
+ .num_links = 1,
+};
+
+/* poodle audio private data */
+static struct wm8731_setup_data poodle_wm8731_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1b,
+};
+
+/* poodle audio subsystem */
+static struct snd_soc_device poodle_snd_devdata = {
+ .machine = &snd_soc_machine_poodle,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8731,
+ .codec_data = &poodle_wm8731_setup,
+};
+
+static struct platform_device *poodle_snd_device;
+
+static int __init poodle_init(void)
+{
+ int ret;
+
+ if (!machine_is_poodle())
+ return -ENODEV;
+
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_AMP_ON, 0);
+ /* should we mute HP at startup - burning power ?*/
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_L, 0);
+ locomo_gpio_set_dir(&poodle_locomo_device.dev,
+ POODLE_LOCOMO_GPIO_MUTE_R, 0);
+
+ poodle_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!poodle_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(poodle_snd_device, &poodle_snd_devdata);
+ poodle_snd_devdata.dev = &poodle_snd_device->dev;
+ ret = platform_device_add(poodle_snd_device);
+
+ if (ret)
+ platform_device_put(poodle_snd_device);
+
+ return ret;
+}
+
+static void __exit poodle_exit(void)
+{
+ platform_device_unregister(poodle_snd_device);
+}
+
+module_init(poodle_init);
+module_exit(poodle_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Poodle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
new file mode 100644
index 0000000..a7a3a9c
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -0,0 +1,232 @@
+/*
+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
+ *
+ * Author: Nicolas Pitre
+ * Created: Dec 02, 2004
+ * Copyright: MontaVista Software Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/ac97_codec.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ pxa2xx_ac97_try_warm_reset(ac97);
+
+ pxa2xx_ac97_finish_reset(ac97);
+}
+
+static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ pxa2xx_ac97_try_cold_reset(ac97);
+
+ pxa2xx_ac97_finish_reset(ac97);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = pxa2xx_ac97_read,
+ .write = pxa2xx_ac97_write,
+ .warm_reset = pxa2xx_ac97_warm_reset,
+ .reset = pxa2xx_ac97_cold_reset,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
+ .name = "AC97 PCM Stereo out",
+ .dev_addr = __PREG(PCDR),
+ .drcmr = &DRCMR(12),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
+ .name = "AC97 PCM Stereo in",
+ .dev_addr = __PREG(PCDR),
+ .drcmr = &DRCMR(11),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
+ .name = "AC97 Aux PCM (Slot 5) Mono out",
+ .dev_addr = __PREG(MODR),
+ .drcmr = &DRCMR(10),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
+ .name = "AC97 Aux PCM (Slot 5) Mono in",
+ .dev_addr = __PREG(MODR),
+ .drcmr = &DRCMR(9),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
+ .name = "AC97 Mic PCM (Slot 6) Mono in",
+ .dev_addr = __PREG(MCDR),
+ .drcmr = &DRCMR(8),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+#ifdef CONFIG_PM
+static int pxa2xx_ac97_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ return pxa2xx_ac97_hw_suspend();
+}
+
+static int pxa2xx_ac97_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ return pxa2xx_ac97_hw_resume();
+}
+
+#else
+#define pxa2xx_ac97_suspend NULL
+#define pxa2xx_ac97_resume NULL
+#endif
+
+static int pxa2xx_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ return pxa2xx_ac97_hw_probe(pdev);
+}
+
+static void pxa2xx_ac97_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ pxa2xx_ac97_hw_remove(pdev);
+}
+
+static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
+ else
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+ return 0;
+}
+
+static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+ else
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+ return 0;
+}
+
+static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+ else
+ cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
+
+ return 0;
+}
+
+#define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+/*
+ * There is only 1 physical AC97 interface for pxa2xx, but it
+ * has extra fifo's that can be used for aux DACs and ADCs.
+ */
+struct snd_soc_dai pxa_ac97_dai[] = {
+{
+ .name = "pxa2xx-ac97",
+ .id = 0,
+ .type = SND_SOC_DAI_AC97,
+ .probe = pxa2xx_ac97_probe,
+ .remove = pxa2xx_ac97_remove,
+ .suspend = pxa2xx_ac97_suspend,
+ .resume = pxa2xx_ac97_resume,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = pxa2xx_ac97_hw_params,},
+},
+{
+ .name = "pxa2xx-ac97-aux",
+ .id = 1,
+ .type = SND_SOC_DAI_AC97,
+ .playback = {
+ .stream_name = "AC97 Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Aux Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = pxa2xx_ac97_hw_aux_params,},
+},
+{
+ .name = "pxa2xx-ac97-mic",
+ .id = 2,
+ .type = SND_SOC_DAI_AC97,
+ .capture = {
+ .stream_name = "AC97 Mic Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = PXA2XX_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = pxa2xx_ac97_hw_mic_params,},
+},
+};
+
+EXPORT_SYMBOL_GPL(pxa_ac97_dai);
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
new file mode 100644
index 0000000..e390de8
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -0,0 +1,22 @@
+/*
+ * linux/sound/soc/pxa/pxa2xx-ac97.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_AC97_H
+#define _PXA2XX_AC97_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_AC97_HIFI 0
+#define PXA2XX_DAI_AC97_AUX 1
+#define PXA2XX_DAI_AC97_MIC 2
+
+extern struct snd_soc_dai pxa_ac97_dai[3];
+
+/* platform data */
+extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
new file mode 100644
index 0000000..e758034
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -0,0 +1,410 @@
+/*
+ * pxa2xx-i2s.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * lrg@slimlogic.co.uk
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+struct pxa2xx_gpio {
+ u32 sys;
+ u32 rx;
+ u32 tx;
+ u32 clk;
+ u32 frm;
+};
+
+/*
+ * I2S Controller Register and Bit Definitions
+ */
+#define SACR0 __REG(0x40400000) /* Global Control Register */
+#define SACR1 __REG(0x40400004) /* Serial Audio I 2 S/MSB-Justified Control Register */
+#define SASR0 __REG(0x4040000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */
+#define SAIMR __REG(0x40400014) /* Serial Audio Interrupt Mask Register */
+#define SAICR __REG(0x40400018) /* Serial Audio Interrupt Clear Register */
+#define SADIV __REG(0x40400060) /* Audio Clock Divider Register. */
+#define SADR __REG(0x40400080) /* Serial Audio Data Register (TX and RX FIFO access Register). */
+
+#define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */
+#define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */
+#define SACR0_EFWR (1 << 4) /* Enable EFWR Function */
+#define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */
+#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */
+#define SACR0_ENB (1 << 0) /* Enable I2S Link */
+#define SACR1_ENLBF (1 << 5) /* Enable Loopback */
+#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */
+#define SACR1_DREC (1 << 3) /* Disable Recording Function */
+#define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */
+
+#define SASR0_I2SOFF (1 << 7) /* Controller Status */
+#define SASR0_ROR (1 << 6) /* Rx FIFO Overrun */
+#define SASR0_TUR (1 << 5) /* Tx FIFO Underrun */
+#define SASR0_RFS (1 << 4) /* Rx FIFO Service Request */
+#define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */
+#define SASR0_BSY (1 << 2) /* I2S Busy */
+#define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */
+#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */
+
+#define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */
+#define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */
+
+#define SAIMR_ROR (1 << 6) /* Enable Rx FIFO Overrun Condition Interrupt */
+#define SAIMR_TUR (1 << 5) /* Enable Tx FIFO Underrun Condition Interrupt */
+#define SAIMR_RFS (1 << 4) /* Enable Rx FIFO Service Interrupt */
+#define SAIMR_TFS (1 << 3) /* Enable Tx FIFO Service Interrupt */
+
+struct pxa_i2s_port {
+ u32 sadiv;
+ u32 sacr0;
+ u32 sacr1;
+ u32 saimr;
+ int master;
+ u32 fmt;
+};
+static struct pxa_i2s_port pxa_i2s;
+static struct clk *clk_i2s;
+
+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
+ .name = "I2S PCM Stereo out",
+ .dev_addr = __PREG(SADR),
+ .drcmr = &DRCMR(3),
+ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
+ .name = "I2S PCM Stereo in",
+ .dev_addr = __PREG(SADR),
+ .drcmr = &DRCMR(2),
+ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
+ DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_gpio gpio_bus[] = {
+ { /* I2S SoC Slave */
+ .rx = GPIO29_SDATA_IN_I2S_MD,
+ .tx = GPIO30_SDATA_OUT_I2S_MD,
+ .clk = GPIO28_BITCLK_IN_I2S_MD,
+ .frm = GPIO31_SYNC_I2S_MD,
+ },
+ { /* I2S SoC Master */
+ .rx = GPIO29_SDATA_IN_I2S_MD,
+ .tx = GPIO30_SDATA_OUT_I2S_MD,
+ .clk = GPIO28_BITCLK_OUT_I2S_MD,
+ .frm = GPIO31_SYNC_I2S_MD,
+ },
+};
+
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (IS_ERR(clk_i2s))
+ return PTR_ERR(clk_i2s);
+
+ if (!cpu_dai->active) {
+ SACR0 |= SACR0_RST;
+ SACR0 = 0;
+ }
+
+ return 0;
+}
+
+/* wait for I2S controller to be ready */
+static int pxa_i2s_wait(void)
+{
+ int i;
+
+ /* flush the Rx FIFO */
+ for(i = 0; i < 16; i++)
+ SADR;
+ return 0;
+}
+
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ pxa_i2s.fmt = 0;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ pxa_i2s.fmt = SACR1_AMSL;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ pxa_i2s.master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ pxa_i2s.master = 0;
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ if (clk_id != PXA2XX_I2S_SYSCLK)
+ return -ENODEV;
+
+ if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT)
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
+
+ return 0;
+}
+
+static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
+ pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
+ BUG_ON(IS_ERR(clk_i2s));
+ clk_enable(clk_i2s);
+ pxa_i2s_wait();
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
+ else
+ cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+ /* is port used by another stream */
+ if (!(SACR0 & SACR0_ENB)) {
+
+ SACR0 = 0;
+ SACR1 = 0;
+ if (pxa_i2s.master)
+ SACR0 |= SACR0_BCKD;
+
+ SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1);
+ SACR1 |= pxa_i2s.fmt;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ SAIMR |= SAIMR_TFS;
+ else
+ SAIMR |= SAIMR_RFS;
+
+ switch (params_rate(params)) {
+ case 8000:
+ SADIV = 0x48;
+ break;
+ case 11025:
+ SADIV = 0x34;
+ break;
+ case 16000:
+ SADIV = 0x24;
+ break;
+ case 22050:
+ SADIV = 0x1a;
+ break;
+ case 44100:
+ SADIV = 0xd;
+ break;
+ case 48000:
+ SADIV = 0xc;
+ break;
+ case 96000: /* not in manual and possibly slightly inaccurate */
+ SADIV = 0x6;
+ break;
+ }
+
+ return 0;
+}
+
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ SACR0 |= SACR0_ENB;
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ SACR1 |= SACR1_DRPL;
+ SAIMR &= ~SAIMR_TFS;
+ } else {
+ SACR1 |= SACR1_DREC;
+ SAIMR &= ~SAIMR_RFS;
+ }
+
+ if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
+ SACR0 &= ~SACR0_ENB;
+ pxa_i2s_wait();
+ clk_disable(clk_i2s);
+ }
+
+ clk_put(clk_i2s);
+}
+
+#ifdef CONFIG_PM
+static int pxa2xx_i2s_suspend(struct platform_device *dev,
+ struct snd_soc_dai *dai)
+{
+ if (!dai->active)
+ return 0;
+
+ /* store registers */
+ pxa_i2s.sacr0 = SACR0;
+ pxa_i2s.sacr1 = SACR1;
+ pxa_i2s.saimr = SAIMR;
+ pxa_i2s.sadiv = SADIV;
+
+ /* deactivate link */
+ SACR0 &= ~SACR0_ENB;
+ pxa_i2s_wait();
+ return 0;
+}
+
+static int pxa2xx_i2s_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ if (!dai->active)
+ return 0;
+
+ pxa_i2s_wait();
+
+ SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
+ SACR1 = pxa_i2s.sacr1;
+ SAIMR = pxa_i2s.saimr;
+ SADIV = pxa_i2s.sadiv;
+ SACR0 |= SACR0_ENB;
+
+ return 0;
+}
+
+#else
+#define pxa2xx_i2s_suspend NULL
+#define pxa2xx_i2s_resume NULL
+#endif
+
+#define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+struct snd_soc_dai pxa_i2s_dai = {
+ .name = "pxa2xx-i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .suspend = pxa2xx_i2s_suspend,
+ .resume = pxa2xx_i2s_resume,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = PXA2XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .startup = pxa2xx_i2s_startup,
+ .shutdown = pxa2xx_i2s_shutdown,
+ .trigger = pxa2xx_i2s_trigger,
+ .hw_params = pxa2xx_i2s_hw_params,},
+ .dai_ops = {
+ .set_fmt = pxa2xx_i2s_set_dai_fmt,
+ .set_sysclk = pxa2xx_i2s_set_dai_sysclk,
+ },
+};
+
+EXPORT_SYMBOL_GPL(pxa_i2s_dai);
+
+static int pxa2xx_i2s_probe(struct platform_device *dev)
+{
+ clk_i2s = clk_get(&dev->dev, "I2SCLK");
+ return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+}
+
+static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
+{
+ clk_put(clk_i2s);
+ clk_i2s = ERR_PTR(-ENOENT);
+ return 0;
+}
+
+static struct platform_driver pxa2xx_i2s_driver = {
+ .probe = pxa2xx_i2s_probe,
+ .remove = __devexit_p(pxa2xx_i2s_remove),
+
+ .driver = {
+ .name = "pxa2xx-i2s",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init pxa2xx_i2s_init(void)
+{
+ if (cpu_is_pxa27x())
+ gpio_bus[1].sys = GPIO113_I2S_SYSCLK_MD;
+ else
+ gpio_bus[1].sys = GPIO32_SYSCLK_I2S_MD;
+
+ clk_i2s = ERR_PTR(-ENOENT);
+ return platform_driver_register(&pxa2xx_i2s_driver);
+}
+
+static void __exit pxa2xx_i2s_exit(void)
+{
+ platform_driver_unregister(&pxa2xx_i2s_driver);
+}
+
+module_init(pxa2xx_i2s_init);
+module_exit(pxa2xx_i2s_exit);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
new file mode 100644
index 0000000..e2def44
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -0,0 +1,20 @@
+/*
+ * linux/sound/soc/pxa/pxa2xx-i2s.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_I2S_H
+#define _PXA2XX_I2S_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_I2S 0
+
+/* I2S clock */
+#define PXA2XX_I2S_SYSCLK 0
+
+extern struct snd_soc_dai pxa_i2s_dai;
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
new file mode 100644
index 0000000..afcd892
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -0,0 +1,123 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: (C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+
+#include "pxa2xx-pcm.h"
+#include "../../arm/pxa2xx-pcm.h"
+
+static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pxa2xx_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+ int ret;
+
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
+ if (!dma)
+ return 0;
+
+ /* this may get called several times by oss emulation
+ * with different params */
+ if (prtd->params == NULL) {
+ prtd->params = dma;
+ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ pxa2xx_pcm_dma_irq, substream);
+ if (ret < 0)
+ return ret;
+ prtd->dma_ch = ret;
+ } else if (prtd->params != dma) {
+ pxa_free_dma(prtd->dma_ch);
+ prtd->params = dma;
+ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ pxa2xx_pcm_dma_irq, substream);
+ if (ret < 0)
+ return ret;
+ prtd->dma_ch = ret;
+ }
+
+ return __pxa2xx_pcm_hw_params(substream, params);
+}
+
+static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+ __pxa2xx_pcm_hw_free(substream);
+
+ if (prtd->dma_ch) {
+ pxa_free_dma(prtd->dma_ch);
+ prtd->dma_ch = 0;
+ }
+
+ return 0;
+}
+
+struct snd_pcm_ops pxa2xx_pcm_ops = {
+ .open = __pxa2xx_pcm_open,
+ .close = __pxa2xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pxa2xx_pcm_hw_params,
+ .hw_free = pxa2xx_pcm_hw_free,
+ .prepare = __pxa2xx_pcm_prepare,
+ .trigger = pxa2xx_pcm_trigger,
+ .pointer = pxa2xx_pcm_pointer,
+ .mmap = pxa2xx_pcm_mmap,
+};
+
+static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK;
+
+static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &pxa2xx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+ if (dai->playback.channels_min) {
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+struct snd_soc_platform pxa2xx_soc_platform = {
+ .name = "pxa2xx-audio",
+ .pcm_ops = &pxa2xx_pcm_ops,
+ .pcm_new = pxa2xx_soc_pcm_new,
+ .pcm_free = pxa2xx_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-pcm.h b/sound/soc/pxa/pxa2xx-pcm.h
new file mode 100644
index 0000000..60c3b20
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-pcm.h
@@ -0,0 +1,19 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author: Nicolas Pitre
+ * Created: Nov 30, 2004
+ * Copyright: MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_PCM_H
+#define _PXA2XX_PCM_H
+
+/* platform data */
+extern struct snd_soc_platform pxa2xx_soc_platform;
+
+#endif
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
new file mode 100644
index 0000000..d307b67
--- /dev/null
+++ b/sound/soc/pxa/spitz.c
@@ -0,0 +1,375 @@
+/*
+ * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/spitz.h>
+#include "../codecs/wm8750.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define SPITZ_HP 0
+#define SPITZ_MIC 1
+#define SPITZ_LINE 2
+#define SPITZ_HEADSET 3
+#define SPITZ_HP_OFF 4
+#define SPITZ_SPK_ON 0
+#define SPITZ_SPK_OFF 1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define SPITZ_AUDIO_CLOCK 12288000
+
+static int spitz_jack_func;
+static int spitz_spk_func;
+
+static void spitz_ext_control(struct snd_soc_codec *codec)
+{
+ if (spitz_spk_func == SPITZ_SPK_ON)
+ snd_soc_dapm_enable_pin(codec, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(codec, "Ext Spk");
+
+ /* set up jack connection */
+ switch (spitz_jack_func) {
+ case SPITZ_HP:
+ /* enable and unmute hp jack, disable mic bias */
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
+ break;
+ case SPITZ_MIC:
+ /* enable mic jack and bias, mute hp */
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+ break;
+ case SPITZ_LINE:
+ /* enable line jack, disable mic bias and mute hp */
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_enable_pin(codec, "Line Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+ break;
+ case SPITZ_HEADSET:
+ /* enable and unmute headset jack enable mic bias, mute L hp */
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
+ break;
+ case SPITZ_HP_OFF:
+
+ /* jack removed, everything off */
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ snd_soc_dapm_disable_pin(codec, "Mic Jack");
+ snd_soc_dapm_disable_pin(codec, "Line Jack");
+ gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
+ gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
+ break;
+ }
+ snd_soc_dapm_sync(codec);
+}
+
+static int spitz_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ spitz_ext_control(codec);
+ return 0;
+}
+
+static int spitz_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret = 0;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops spitz_ops = {
+ .startup = spitz_startup,
+ .hw_params = spitz_hw_params,
+};
+
+static int spitz_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spitz_jack_func;
+ return 0;
+}
+
+static int spitz_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (spitz_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ spitz_jack_func = ucontrol->value.integer.value[0];
+ spitz_ext_control(codec);
+ return 1;
+}
+
+static int spitz_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = spitz_spk_func;
+ return 0;
+}
+
+static int spitz_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (spitz_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ spitz_spk_func = ucontrol->value.integer.value[0];
+ spitz_ext_control(codec);
+ return 1;
+}
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (machine_is_borzoi() || machine_is_spitz())
+ gpio_set_value(SPITZ_GPIO_MIC_BIAS,
+ SND_SOC_DAPM_EVENT_ON(event));
+
+ if (machine_is_akita())
+ gpio_set_value(AKITA_GPIO_MIC_BIAS,
+ SND_SOC_DAPM_EVENT_ON(event));
+
+ return 0;
+}
+
+/* spitz machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_LINE("Line Jack", NULL),
+
+ /* headset is a mic and mono headphone */
+ SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Spitz machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* headphone connected to LOUT1, ROUT1 */
+ {"Headphone Jack", NULL, "LOUT1"},
+ {"Headphone Jack", NULL, "ROUT1"},
+
+ /* headset connected to ROUT1 and LINPUT1 with bias (def below) */
+ {"Headset Jack", NULL, "ROUT1"},
+
+ /* ext speaker connected to LOUT2, ROUT2 */
+ {"Ext Spk", NULL , "ROUT2"},
+ {"Ext Spk", NULL , "LOUT2"},
+
+ /* mic is connected to input 1 - with bias */
+ {"LINPUT1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Mic Jack"},
+
+ /* line is connected to input 1 - no bias */
+ {"LINPUT1", NULL, "Line Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum spitz_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
+ SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack,
+ spitz_set_jack),
+ SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk,
+ spitz_set_spk),
+};
+
+/*
+ * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device
+ */
+static int spitz_wm8750_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ /* NC codec pins */
+ snd_soc_dapm_nc_pin(codec, "RINPUT1");
+ snd_soc_dapm_nc_pin(codec, "LINPUT2");
+ snd_soc_dapm_nc_pin(codec, "RINPUT2");
+ snd_soc_dapm_nc_pin(codec, "LINPUT3");
+ snd_soc_dapm_nc_pin(codec, "RINPUT3");
+ snd_soc_dapm_nc_pin(codec, "OUT3");
+ snd_soc_dapm_nc_pin(codec, "MONO1");
+
+ /* Add spitz specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* Add spitz specific widgets */
+ snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+ ARRAY_SIZE(wm8750_dapm_widgets));
+
+ /* Set up spitz specific audio paths */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+/* spitz digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link spitz_dai = {
+ .name = "wm8750",
+ .stream_name = "WM8750",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8750_dai,
+ .init = spitz_wm8750_init,
+ .ops = &spitz_ops,
+};
+
+/* spitz audio machine driver */
+static struct snd_soc_machine snd_soc_machine_spitz = {
+ .name = "Spitz",
+ .dai_link = &spitz_dai,
+ .num_links = 1,
+};
+
+/* spitz audio private data */
+static struct wm8750_setup_data spitz_wm8750_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1b,
+};
+
+/* spitz audio subsystem */
+static struct snd_soc_device spitz_snd_devdata = {
+ .machine = &snd_soc_machine_spitz,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8750,
+ .codec_data = &spitz_wm8750_setup,
+};
+
+static struct platform_device *spitz_snd_device;
+
+static int __init spitz_init(void)
+{
+ int ret;
+
+ if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
+ return -ENODEV;
+
+ spitz_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!spitz_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(spitz_snd_device, &spitz_snd_devdata);
+ spitz_snd_devdata.dev = &spitz_snd_device->dev;
+ ret = platform_device_add(spitz_snd_device);
+
+ if (ret)
+ platform_device_put(spitz_snd_device);
+
+ return ret;
+}
+
+static void __exit spitz_exit(void)
+{
+ platform_device_unregister(spitz_snd_device);
+}
+
+module_init(spitz_init);
+module_exit(spitz_exit);
+
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Spitz");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
new file mode 100644
index 0000000..afefe41
--- /dev/null
+++ b/sound/soc/pxa/tosa.c
@@ -0,0 +1,292 @@
+/*
+ * tosa.c -- SoC audio for Tosa
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * GPIO's
+ * 1 - Jack Insertion
+ * 5 - Hookswitch (headset answer/hang up switch)
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/tosa.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_machine tosa;
+
+#define TOSA_HP 0
+#define TOSA_MIC_INT 1
+#define TOSA_HEADSET 2
+#define TOSA_HP_OFF 3
+#define TOSA_SPK_ON 0
+#define TOSA_SPK_OFF 1
+
+static int tosa_jack_func;
+static int tosa_spk_func;
+
+static void tosa_ext_control(struct snd_soc_codec *codec)
+{
+ /* set up jack connection */
+ switch (tosa_jack_func) {
+ case TOSA_HP:
+ snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ break;
+ case TOSA_MIC_INT:
+ snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_disable_pin(codec, "Headset Jack");
+ break;
+ case TOSA_HEADSET:
+ snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+ snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Headset Jack");
+ break;
+ }
+
+ if (tosa_spk_func == TOSA_SPK_ON)
+ snd_soc_dapm_enable_pin(codec, "Speaker");
+ else
+ snd_soc_dapm_disable_pin(codec, "Speaker");
+
+ snd_soc_dapm_sync(codec);
+}
+
+static int tosa_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->socdev->codec;
+
+ /* check the jack status at stream startup */
+ tosa_ext_control(codec);
+ return 0;
+}
+
+static struct snd_soc_ops tosa_ops = {
+ .startup = tosa_startup,
+};
+
+static int tosa_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = tosa_jack_func;
+ return 0;
+}
+
+static int tosa_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (tosa_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ tosa_jack_func = ucontrol->value.integer.value[0];
+ tosa_ext_control(codec);
+ return 1;
+}
+
+static int tosa_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = tosa_spk_func;
+ return 0;
+}
+
+static int tosa_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (tosa_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ tosa_spk_func = ucontrol->value.integer.value[0];
+ tosa_ext_control(codec);
+ return 1;
+}
+
+/* tosa dapm event handlers */
+static int tosa_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0);
+ return 0;
+}
+
+/* tosa machine dapm widgets */
+static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* tosa audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ /* headphone connected to HPOUTL, HPOUTR */
+ {"Headphone Jack", NULL, "HPOUTL"},
+ {"Headphone Jack", NULL, "HPOUTR"},
+
+ /* ext speaker connected to LOUT2, ROUT2 */
+ {"Speaker", NULL, "LOUT2"},
+ {"Speaker", NULL, "ROUT2"},
+
+ /* internal mic is connected to mic1, mic2 differential - with bias */
+ {"MIC1", NULL, "Mic Bias"},
+ {"MIC2", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Mic (Internal)"},
+
+ /* headset is connected to HPOUTR, and LINEINR with bias */
+ {"Headset Jack", NULL, "HPOUTR"},
+ {"LINEINR", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Jack"},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+ "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum tosa_enum[] = {
+ SOC_ENUM_SINGLE_EXT(5, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new tosa_controls[] = {
+ SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
+ tosa_set_jack),
+ SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
+ tosa_set_spk),
+};
+
+static int tosa_ac97_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ snd_soc_dapm_nc_pin(codec, "OUT3");
+ snd_soc_dapm_nc_pin(codec, "MONOOUT");
+
+ /* add tosa specific controls */
+ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&tosa_controls[i],codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* add tosa specific widgets */
+ snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
+ ARRAY_SIZE(tosa_dapm_widgets));
+
+ /* set up tosa specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+static struct snd_soc_dai_link tosa_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ .init = tosa_ac97_init,
+ .ops = &tosa_ops,
+},
+{
+ .name = "AC97 Aux",
+ .stream_name = "AC97 Aux",
+ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ .ops = &tosa_ops,
+},
+};
+
+static struct snd_soc_machine tosa = {
+ .name = "Tosa",
+ .dai_link = tosa_dai,
+ .num_links = ARRAY_SIZE(tosa_dai),
+};
+
+static struct snd_soc_device tosa_snd_devdata = {
+ .machine = &tosa,
+ .platform = &pxa2xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *tosa_snd_device;
+
+static int __init tosa_init(void)
+{
+ int ret;
+
+ if (!machine_is_tosa())
+ return -ENODEV;
+
+ ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
+ if (ret)
+ return ret;
+ gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
+
+ tosa_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!tosa_snd_device) {
+ ret = -ENOMEM;
+ goto err_alloc;
+ }
+
+ platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata);
+ tosa_snd_devdata.dev = &tosa_snd_device->dev;
+ ret = platform_device_add(tosa_snd_device);
+
+ if (!ret)
+ return 0;
+
+ platform_device_put(tosa_snd_device);
+
+err_alloc:
+ gpio_free(TOSA_GPIO_L_MUTE);
+
+ return ret;
+}
+
+static void __exit tosa_exit(void)
+{
+ platform_device_unregister(tosa_snd_device);
+ gpio_free(TOSA_GPIO_L_MUTE);
+}
+
+module_init(tosa_init);
+module_exit(tosa_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Tosa");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
new file mode 100644
index 0000000..b9f2353
--- /dev/null
+++ b/sound/soc/s3c24xx/Kconfig
@@ -0,0 +1,46 @@
+config SND_S3C24XX_SOC
+ tristate "SoC Audio for the Samsung S3C24XX chips"
+ depends on ARCH_S3C2410
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the S3C24XX AC97, I2S or SSP interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_S3C24XX_SOC_I2S
+ tristate
+
+config SND_S3C2412_SOC_I2S
+ tristate
+
+config SND_S3C2443_SOC_AC97
+ tristate
+ select AC97_BUS
+ select SND_SOC_AC97_BUS
+
+config SND_S3C24XX_SOC_NEO1973_WM8753
+ tristate "SoC I2S Audio support for NEO1973 - WM8753"
+ depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01
+ select SND_S3C24XX_SOC_I2S
+ select SND_SOC_WM8753
+ help
+ Say Y if you want to add support for SoC audio on smdk2440
+ with the WM8753.
+
+config SND_S3C24XX_SOC_SMDK2443_WM9710
+ tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
+ depends on SND_S3C24XX_SOC && MACH_SMDK2443
+ select SND_S3C2443_SOC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ Say Y if you want to add support for SoC audio on smdk2443
+ with the WM9710.
+
+config SND_S3C24XX_SOC_LN2440SBC_ALC650
+ tristate "SoC AC97 Audio support for LN2440SBC - ALC650"
+ depends on SND_S3C24XX_SOC
+ select SND_S3C2443_SOC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ Say Y if you want to add support for SoC audio on ln2440sbc
+ with the ALC650.
+
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
new file mode 100644
index 0000000..0aa5fb0
--- /dev/null
+++ b/sound/soc/s3c24xx/Makefile
@@ -0,0 +1,19 @@
+# S3c24XX Platform Support
+snd-soc-s3c24xx-objs := s3c24xx-pcm.o
+snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
+snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
+snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
+
+obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
+obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
+obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
+obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
+
+# S3C24XX Machine Support
+snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
+snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
+snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+
+obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
+obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
+obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
diff --git a/sound/soc/s3c24xx/lm4857.h b/sound/soc/s3c24xx/lm4857.h
new file mode 100644
index 0000000..0cf5b70
--- /dev/null
+++ b/sound/soc/s3c24xx/lm4857.h
@@ -0,0 +1,32 @@
+/*
+ * lm4857.h -- ALSA Soc Audio Layer
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Revision history
+ * 18th Jun 2007 Initial version.
+ */
+
+#ifndef LM4857_H_
+#define LM4857_H_
+
+/* The register offsets in the cache array */
+#define LM4857_MVOL 0
+#define LM4857_LVOL 1
+#define LM4857_RVOL 2
+#define LM4857_CTRL 3
+
+/* the shifts required to set these bits */
+#define LM4857_3D 5
+#define LM4857_WAKEUP 5
+#define LM4857_EPGAIN 4
+
+#endif /*LM4857_H_*/
+
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
new file mode 100644
index 0000000..4eab2c1
--- /dev/null
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -0,0 +1,85 @@
+/*
+ * SoC audio for ln2440sbc
+ *
+ * Copyright 2007 KonekTel, a.s.
+ * Author: Ivan Kuten
+ * ivan.kuten@promwad.com
+ *
+ * Heavily based on smdk2443_wm9710.c
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/ac97.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-ac97.h"
+
+static struct snd_soc_machine ln2440sbc;
+
+static struct snd_soc_dai_link ln2440sbc_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &s3c2443_ac97_dai[0],
+ .codec_dai = &ac97_dai,
+},
+};
+
+static struct snd_soc_machine ln2440sbc = {
+ .name = "LN2440SBC",
+ .dai_link = ln2440sbc_dai,
+ .num_links = ARRAY_SIZE(ln2440sbc_dai),
+};
+
+static struct snd_soc_device ln2440sbc_snd_ac97_devdata = {
+ .machine = &ln2440sbc,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_ac97,
+};
+
+static struct platform_device *ln2440sbc_snd_ac97_device;
+
+static int __init ln2440sbc_init(void)
+{
+ int ret;
+
+ ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+ if (!ln2440sbc_snd_ac97_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(ln2440sbc_snd_ac97_device,
+ &ln2440sbc_snd_ac97_devdata);
+ ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev;
+ ret = platform_device_add(ln2440sbc_snd_ac97_device);
+
+ if (ret)
+ platform_device_put(ln2440sbc_snd_ac97_device);
+
+ return ret;
+}
+
+static void __exit ln2440sbc_exit(void)
+{
+ platform_device_unregister(ln2440sbc_snd_ac97_device);
+}
+
+module_init(ln2440sbc_init);
+module_exit(ln2440sbc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ivan Kuten");
+MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
new file mode 100644
index 0000000..87ddfef
--- /dev/null
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -0,0 +1,722 @@
+/*
+ * neo1973_wm8753.c -- SoC audio for Neo1973
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include <asm/mach-types.h>
+#include <asm/hardware/scoop.h>
+#include <mach/regs-clock.h>
+#include <mach/regs-gpio.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
+#include <linux/io.h>
+#include <mach/spi-gpio.h>
+
+#include <asm/plat-s3c24xx/regs-iis.h>
+
+#include "../codecs/wm8753.h"
+#include "lm4857.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+/* Debugging stuff */
+#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0
+#if S3C24XX_SOC_NEO1973_WM8753_DEBUG
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x)
+#else
+#define DBG(x...)
+#endif
+
+/* define the scenarios */
+#define NEO_AUDIO_OFF 0
+#define NEO_GSM_CALL_AUDIO_HANDSET 1
+#define NEO_GSM_CALL_AUDIO_HEADSET 2
+#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
+#define NEO_STEREO_TO_SPEAKERS 4
+#define NEO_STEREO_TO_HEADPHONES 5
+#define NEO_CAPTURE_HANDSET 6
+#define NEO_CAPTURE_HEADSET 7
+#define NEO_CAPTURE_BLUETOOTH 8
+
+static struct snd_soc_machine neo1973;
+static struct i2c_client *i2c;
+
+static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int pll_out = 0, bclk = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ DBG("Entered %s\n", __func__);
+
+ iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ pll_out = 12288000;
+ break;
+ case 48000:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 12288000;
+ break;
+ case 96000:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 12288000;
+ break;
+ case 11025:
+ bclk = WM8753_BCLK_DIV_16;
+ pll_out = 11289600;
+ break;
+ case 22050:
+ bclk = WM8753_BCLK_DIV_8;
+ pll_out = 11289600;
+ break;
+ case 44100:
+ bclk = WM8753_BCLK_DIV_4;
+ pll_out = 11289600;
+ break;
+ case 88200:
+ bclk = WM8753_BCLK_DIV_2;
+ pll_out = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set MCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+
+ /* set codec BCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
+ if (ret < 0)
+ return ret;
+
+ /* set prescaler division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(4, 4));
+ if (ret < 0)
+ return ret;
+
+ /* codec PLL input is PCLK/4 */
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+ iis_clkrate / 4, pll_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+ DBG("Entered %s\n", __func__);
+
+ /* disable the PLL */
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+}
+
+/*
+ * Neo1973 WM8753 HiFi DAI opserations.
+ */
+static struct snd_soc_ops neo1973_hifi_ops = {
+ .hw_params = neo1973_hifi_hw_params,
+ .hw_free = neo1973_hifi_hw_free,
+};
+
+static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ unsigned int pcmdiv = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
+
+ DBG("Entered %s\n", __func__);
+
+ iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+ if (params_rate(params) != 8000)
+ return -EINVAL;
+ if (params_channels(params) != 1)
+ return -EINVAL;
+
+ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
+
+ /* todo: gg check mode (DSP_B) against CSR datasheet */
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set codec PCM division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
+ if (ret < 0)
+ return ret;
+
+ /* configue and enable PLL for 12.288MHz output */
+ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+ iis_clkrate / 4, 12288000);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+ DBG("Entered %s\n", __func__);
+
+ /* disable the PLL */
+ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
+static struct snd_soc_ops neo1973_voice_ops = {
+ .hw_params = neo1973_voice_hw_params,
+ .hw_free = neo1973_voice_hw_free,
+};
+
+static int neo1973_scenario;
+
+static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = neo1973_scenario;
+ return 0;
+}
+
+static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
+{
+ DBG("Entered %s\n", __func__);
+
+ switch (neo1973_scenario) {
+ case NEO_AUDIO_OFF:
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ break;
+ case NEO_GSM_CALL_AUDIO_HANDSET:
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Call Mic");
+ break;
+ case NEO_GSM_CALL_AUDIO_HEADSET:
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line In");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ break;
+ case NEO_GSM_CALL_AUDIO_BLUETOOTH:
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_enable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ break;
+ case NEO_STEREO_TO_SPEAKERS:
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ break;
+ case NEO_STEREO_TO_HEADPHONES:
+ snd_soc_dapm_enable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ break;
+ case NEO_CAPTURE_HANDSET:
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_enable_pin(codec, "Call Mic");
+ break;
+ case NEO_CAPTURE_HEADSET:
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_enable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ break;
+ case NEO_CAPTURE_BLUETOOTH:
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ break;
+ default:
+ snd_soc_dapm_disable_pin(codec, "Audio Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+ snd_soc_dapm_disable_pin(codec, "GSM Line In");
+ snd_soc_dapm_disable_pin(codec, "Headset Mic");
+ snd_soc_dapm_disable_pin(codec, "Call Mic");
+ }
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ DBG("Entered %s\n", __func__);
+
+ if (neo1973_scenario == ucontrol->value.integer.value[0])
+ return 0;
+
+ neo1973_scenario = ucontrol->value.integer.value[0];
+ set_scenario_endpoints(codec, neo1973_scenario);
+ return 1;
+}
+
+static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
+
+static void lm4857_write_regs(void)
+{
+ DBG("Entered %s\n", __func__);
+
+ if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
+ printk(KERN_ERR "lm4857: i2c write failed\n");
+}
+
+static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int reg = kcontrol->private_value & 0xFF;
+ int shift = (kcontrol->private_value >> 8) & 0x0F;
+ int mask = (kcontrol->private_value >> 16) & 0xFF;
+
+ DBG("Entered %s\n", __func__);
+
+ ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
+ return 0;
+}
+
+static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int reg = kcontrol->private_value & 0xFF;
+ int shift = (kcontrol->private_value >> 8) & 0x0F;
+ int mask = (kcontrol->private_value >> 16) & 0xFF;
+
+ if (((lm4857_regs[reg] >> shift) & mask) ==
+ ucontrol->value.integer.value[0])
+ return 0;
+
+ lm4857_regs[reg] &= ~(mask << shift);
+ lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
+ lm4857_write_regs();
+ return 1;
+}
+
+static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
+
+ DBG("Entered %s\n", __func__);
+
+ if (value)
+ value -= 5;
+
+ ucontrol->value.integer.value[0] = value;
+ return 0;
+}
+
+static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 value = ucontrol->value.integer.value[0];
+
+ DBG("Entered %s\n", __func__);
+
+ if (value)
+ value += 5;
+
+ if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
+ return 0;
+
+ lm4857_regs[LM4857_CTRL] &= 0xF0;
+ lm4857_regs[LM4857_CTRL] |= value;
+ lm4857_write_regs();
+ return 1;
+}
+
+static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Audio Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Call Mic", NULL),
+};
+
+
+static const struct snd_soc_dapm_route dapm_routes[] = {
+
+ /* Connections to the lm4857 amp */
+ {"Audio Out", NULL, "LOUT1"},
+ {"Audio Out", NULL, "ROUT1"},
+
+ /* Connections to the GSM Module */
+ {"GSM Line Out", NULL, "MONO1"},
+ {"GSM Line Out", NULL, "MONO2"},
+ {"RXP", NULL, "GSM Line In"},
+ {"RXN", NULL, "GSM Line In"},
+
+ /* Connections to Headset */
+ {"MIC1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Mic"},
+
+ /* Call Mic */
+ {"MIC2", NULL, "Mic Bias"},
+ {"MIC2N", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Call Mic"},
+
+ /* Connect the ALC pins */
+ {"ACIN", NULL, "ACOP"},
+};
+
+static const char *lm4857_mode[] = {
+ "Off",
+ "Call Speaker",
+ "Stereo Speakers",
+ "Stereo Speakers + Headphones",
+ "Headphones"
+};
+
+static const struct soc_enum lm4857_mode_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
+};
+
+static const char *neo_scenarios[] = {
+ "Off",
+ "GSM Handset",
+ "GSM Headset",
+ "GSM Bluetooth",
+ "Speakers",
+ "Headphones",
+ "Capture Handset",
+ "Capture Headset",
+ "Capture Bluetooth"
+};
+
+static const struct soc_enum neo_scenario_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios),
+};
+
+static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
+
+static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
+ SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg, stereo_tlv),
+ SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg, stereo_tlv),
+ SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg, mono_tlv),
+ SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
+ lm4857_get_mode, lm4857_set_mode),
+ SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
+ neo1973_get_scenario, neo1973_set_scenario),
+ SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+};
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
+ * neo1973 II. It is missing logic to detect hp/mic insertions and logic
+ * to re-route the audio in such an event.
+ */
+static int neo1973_wm8753_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
+ DBG("Entered %s\n", __func__);
+
+ /* set up NC codec pins */
+ snd_soc_dapm_nc_pin(codec, "LOUT2");
+ snd_soc_dapm_nc_pin(codec, "ROUT2");
+ snd_soc_dapm_nc_pin(codec, "OUT3");
+ snd_soc_dapm_nc_pin(codec, "OUT4");
+ snd_soc_dapm_nc_pin(codec, "LINE1");
+ snd_soc_dapm_nc_pin(codec, "LINE2");
+
+ /* Add neo1973 specific widgets */
+ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+ ARRAY_SIZE(wm8753_dapm_widgets));
+
+ /* set endpoints to default mode */
+ set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+
+ /* add neo1973 specific controls */
+ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8753_neo1973_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+
+ /* set up neo1973 specific audio routes */
+ err = snd_soc_dapm_add_routes(codec, dapm_routes,
+ ARRAY_SIZE(dapm_routes));
+
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+/*
+ * BT Codec DAI
+ */
+static struct snd_soc_dai bt_dai = {
+ .name = "Bluetooth",
+ .id = 0,
+ .type = SND_SOC_DAI_PCM,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+};
+
+static struct snd_soc_dai_link neo1973_dai[] = {
+{ /* Hifi Playback - for similatious use with voice below */
+ .name = "WM8753",
+ .stream_name = "WM8753 HiFi",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
+ .init = neo1973_wm8753_init,
+ .ops = &neo1973_hifi_ops,
+},
+{ /* Voice via BT */
+ .name = "Bluetooth",
+ .stream_name = "Voice",
+ .cpu_dai = &bt_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
+ .ops = &neo1973_voice_ops,
+},
+};
+
+static struct snd_soc_machine neo1973 = {
+ .name = "neo1973",
+ .dai_link = neo1973_dai,
+ .num_links = ARRAY_SIZE(neo1973_dai),
+};
+
+static struct wm8753_setup_data neo1973_wm8753_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x1a,
+};
+
+static struct snd_soc_device neo1973_snd_devdata = {
+ .machine = &neo1973,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8753,
+ .codec_data = &neo1973_wm8753_setup,
+};
+
+static int lm4857_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ DBG("Entered %s\n", __func__);
+
+ i2c = client;
+
+ lm4857_write_regs();
+ return 0;
+}
+
+static int lm4857_i2c_remove(struct i2c_client *client)
+{
+ DBG("Entered %s\n", __func__);
+
+ i2c = NULL;
+
+ return 0;
+}
+
+static u8 lm4857_state;
+
+static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
+{
+ DBG("Entered %s\n", __func__);
+
+ dev_dbg(&dev->dev, "lm4857_suspend\n");
+ lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
+ if (lm4857_state) {
+ lm4857_regs[LM4857_CTRL] &= 0xf0;
+ lm4857_write_regs();
+ }
+ return 0;
+}
+
+static int lm4857_resume(struct i2c_client *dev)
+{
+ DBG("Entered %s\n", __func__);
+
+ if (lm4857_state) {
+ lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
+ lm4857_write_regs();
+ }
+ return 0;
+}
+
+static void lm4857_shutdown(struct i2c_client *dev)
+{
+ DBG("Entered %s\n", __func__);
+
+ dev_dbg(&dev->dev, "lm4857_shutdown\n");
+ lm4857_regs[LM4857_CTRL] &= 0xf0;
+ lm4857_write_regs();
+}
+
+static const struct i2c_device_id lm4857_i2c_id[] = {
+ { "neo1973_lm4857", 0 },
+ { }
+};
+
+static struct i2c_driver lm4857_i2c_driver = {
+ .driver = {
+ .name = "LM4857 I2C Amp",
+ .owner = THIS_MODULE,
+ },
+ .suspend = lm4857_suspend,
+ .resume = lm4857_resume,
+ .shutdown = lm4857_shutdown,
+ .probe = lm4857_i2c_probe,
+ .remove = lm4857_i2c_remove,
+ .id_table = lm4857_i2c_id,
+};
+
+static struct platform_device *neo1973_snd_device;
+
+static int __init neo1973_init(void)
+{
+ int ret;
+
+ DBG("Entered %s\n", __func__);
+
+ if (!machine_is_neo1973_gta01()) {
+ printk(KERN_INFO
+ "Only GTA01 hardware supported by ASoC driver\n");
+ return -ENODEV;
+ }
+
+ neo1973_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!neo1973_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata);
+ neo1973_snd_devdata.dev = &neo1973_snd_device->dev;
+ ret = platform_device_add(neo1973_snd_device);
+
+ if (ret) {
+ platform_device_put(neo1973_snd_device);
+ return ret;
+ }
+
+ ret = i2c_add_driver(&lm4857_i2c_driver);
+
+ if (ret != 0)
+ platform_device_unregister(neo1973_snd_device);
+
+ return ret;
+}
+
+static void __exit neo1973_exit(void)
+{
+ DBG("Entered %s\n", __func__);
+
+ i2c_del_driver(&lm4857_i2c_driver);
+ platform_device_unregister(neo1973_snd_device);
+}
+
+module_init(neo1973_init);
+module_exit(neo1973_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
new file mode 100644
index 0000000..ded7d99
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -0,0 +1,745 @@
+/* sound/soc/s3c24xx/s3c2412-i2s.c
+ *
+ * ALSA Soc Audio Layer - S3C2412 I2S driver
+ *
+ * Copyright (c) 2006 Wolfson Microelectronics PLC.
+ * Graeme Gregory graeme.gregory@wolfsonmicro.com
+ * linux@wolfsonmicro.com
+ *
+ * Copyright (c) 2007, 2004-2005 Simtec Electronics
+ * http://armlinux.simtec.co.uk/
+ * Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <mach/hardware.h>
+
+#include <linux/io.h>
+#include <asm/dma.h>
+
+#include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
+
+#include <mach/regs-gpio.h>
+#include <mach/audio.h>
+#include <mach/dma.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c2412-i2s.h"
+
+#define S3C2412_I2S_DEBUG 0
+#define S3C2412_I2S_DEBUG_CON 0
+
+#if S3C2412_I2S_DEBUG
+#define DBG(x...) printk(KERN_INFO x)
+#else
+#define DBG(x...) do { } while (0)
+#endif
+
+static struct s3c2410_dma_client s3c2412_dma_client_out = {
+ .name = "I2S PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c2412_dma_client_in = {
+ .name = "I2S PCM Stereo in"
+};
+
+static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = {
+ .client = &s3c2412_dma_client_out,
+ .channel = DMACH_I2S_OUT,
+ .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD,
+ .dma_size = 4,
+};
+
+static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = {
+ .client = &s3c2412_dma_client_in,
+ .channel = DMACH_I2S_IN,
+ .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD,
+ .dma_size = 4,
+};
+
+struct s3c2412_i2s_info {
+ struct device *dev;
+ void __iomem *regs;
+ struct clk *iis_clk;
+ struct clk *iis_pclk;
+ struct clk *iis_cclk;
+
+ u32 suspend_iismod;
+ u32 suspend_iiscon;
+ u32 suspend_iispsr;
+};
+
+static struct s3c2412_i2s_info s3c2412_i2s;
+
+#define bit_set(v, b) (((v) & (b)) ? 1 : 0)
+
+#if S3C2412_I2S_DEBUG_CON
+static void dbg_showcon(const char *fn, u32 con)
+{
+ printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn,
+ bit_set(con, S3C2412_IISCON_LRINDEX),
+ bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY),
+ bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY),
+ bit_set(con, S3C2412_IISCON_TXFIFO_FULL),
+ bit_set(con, S3C2412_IISCON_RXFIFO_FULL));
+
+ printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n",
+ fn,
+ bit_set(con, S3C2412_IISCON_TXDMA_PAUSE),
+ bit_set(con, S3C2412_IISCON_RXDMA_PAUSE),
+ bit_set(con, S3C2412_IISCON_TXCH_PAUSE),
+ bit_set(con, S3C2412_IISCON_RXCH_PAUSE));
+ printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn,
+ bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE),
+ bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE),
+ bit_set(con, S3C2412_IISCON_IIS_ACTIVE));
+}
+#else
+static inline void dbg_showcon(const char *fn, u32 con)
+{
+}
+#endif
+
+/* Turn on or off the transmission path. */
+static void s3c2412_snd_txctrl(int on)
+{
+ struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
+ void __iomem *regs = i2s->regs;
+ u32 fic, con, mod;
+
+ DBG("%s(%d)\n", __func__, on);
+
+ fic = readl(regs + S3C2412_IISFIC);
+ con = readl(regs + S3C2412_IISCON);
+ mod = readl(regs + S3C2412_IISMOD);
+
+ DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+ if (on) {
+ con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+ con &= ~S3C2412_IISCON_TXDMA_PAUSE;
+ con &= ~S3C2412_IISCON_TXCH_PAUSE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_TXONLY:
+ case S3C2412_IISMOD_MODE_TXRX:
+ /* do nothing, we are in the right mode */
+ break;
+
+ case S3C2412_IISMOD_MODE_RXONLY:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_TXRX;
+ break;
+
+ default:
+ dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+ }
+
+ writel(con, regs + S3C2412_IISCON);
+ writel(mod, regs + S3C2412_IISMOD);
+ } else {
+ /* Note, we do not have any indication that the FIFO problems
+ * tha the S3C2410/2440 had apply here, so we should be able
+ * to disable the DMA and TX without resetting the FIFOS.
+ */
+
+ con |= S3C2412_IISCON_TXDMA_PAUSE;
+ con |= S3C2412_IISCON_TXCH_PAUSE;
+ con &= ~S3C2412_IISCON_TXDMA_ACTIVE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_TXRX:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_RXONLY;
+ break;
+
+ case S3C2412_IISMOD_MODE_TXONLY:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ con &= ~S3C2412_IISCON_IIS_ACTIVE;
+ break;
+
+ default:
+ dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+ }
+
+ writel(mod, regs + S3C2412_IISMOD);
+ writel(con, regs + S3C2412_IISCON);
+ }
+
+ fic = readl(regs + S3C2412_IISFIC);
+ dbg_showcon(__func__, con);
+ DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+
+static void s3c2412_snd_rxctrl(int on)
+{
+ struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
+ void __iomem *regs = i2s->regs;
+ u32 fic, con, mod;
+
+ DBG("%s(%d)\n", __func__, on);
+
+ fic = readl(regs + S3C2412_IISFIC);
+ con = readl(regs + S3C2412_IISCON);
+ mod = readl(regs + S3C2412_IISMOD);
+
+ DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+ if (on) {
+ con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+ con &= ~S3C2412_IISCON_RXDMA_PAUSE;
+ con &= ~S3C2412_IISCON_RXCH_PAUSE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_TXRX:
+ case S3C2412_IISMOD_MODE_RXONLY:
+ /* do nothing, we are in the right mode */
+ break;
+
+ case S3C2412_IISMOD_MODE_TXONLY:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_TXRX;
+ break;
+
+ default:
+ dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ }
+
+ writel(mod, regs + S3C2412_IISMOD);
+ writel(con, regs + S3C2412_IISCON);
+ } else {
+ /* See txctrl notes on FIFOs. */
+
+ con &= ~S3C2412_IISCON_RXDMA_ACTIVE;
+ con |= S3C2412_IISCON_RXDMA_PAUSE;
+ con |= S3C2412_IISCON_RXCH_PAUSE;
+
+ switch (mod & S3C2412_IISMOD_MODE_MASK) {
+ case S3C2412_IISMOD_MODE_RXONLY:
+ con &= ~S3C2412_IISCON_IIS_ACTIVE;
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ break;
+
+ case S3C2412_IISMOD_MODE_TXRX:
+ mod &= ~S3C2412_IISMOD_MODE_MASK;
+ mod |= S3C2412_IISMOD_MODE_TXONLY;
+ break;
+
+ default:
+ dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ }
+
+ writel(con, regs + S3C2412_IISCON);
+ writel(mod, regs + S3C2412_IISMOD);
+ }
+
+ fic = readl(regs + S3C2412_IISFIC);
+ DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+
+
+/*
+ * Wait for the LR signal to allow synchronisation to the L/R clock
+ * from the codec. May only be needed for slave mode.
+ */
+static int s3c2412_snd_lrsync(void)
+{
+ u32 iiscon;
+ unsigned long timeout = jiffies + msecs_to_jiffies(5);
+
+ DBG("Entered %s\n", __func__);
+
+ while (1) {
+ iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON);
+ if (iiscon & S3C2412_IISCON_LRINDEX)
+ break;
+
+ if (timeout < jiffies) {
+ printk(KERN_ERR "%s: timeout\n", __func__);
+ return -ETIMEDOUT;
+ }
+ }
+
+ return 0;
+}
+
+/*
+ * Check whether CPU is the master or slave
+ */
+static inline int s3c2412_snd_is_clkmaster(void)
+{
+ u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
+
+ DBG("Entered %s\n", __func__);
+
+ iismod &= S3C2412_IISMOD_MASTER_MASK;
+ return !(iismod == S3C2412_IISMOD_SLAVE);
+}
+
+/*
+ * Set S3C2412 I2S DAI format
+ */
+static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ u32 iismod;
+
+
+ DBG("Entered %s\n", __func__);
+
+ iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
+ DBG("hw_params r: IISMOD: %x \n", iismod);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iismod &= ~S3C2412_IISMOD_MASTER_MASK;
+ iismod |= S3C2412_IISMOD_SLAVE;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ iismod &= ~S3C2412_IISMOD_MASTER_MASK;
+ iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
+ break;
+ default:
+ DBG("unknwon master/slave format\n");
+ return -EINVAL;
+ }
+
+ iismod &= ~S3C2412_IISMOD_SDF_MASK;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iismod |= S3C2412_IISMOD_SDF_MSB;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iismod |= S3C2412_IISMOD_SDF_LSB;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ iismod |= S3C2412_IISMOD_SDF_IIS;
+ break;
+ default:
+ DBG("Unknown data format\n");
+ return -EINVAL;
+ }
+
+ writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
+ DBG("hw_params w: IISMOD: %x \n", iismod);
+ return 0;
+}
+
+static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u32 iismod;
+
+ DBG("Entered %s\n", __func__);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out;
+ else
+ rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in;
+
+ /* Working copies of register */
+ iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
+ DBG("%s: r: IISMOD: %x\n", __func__, iismod);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ iismod |= S3C2412_IISMOD_8BIT;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ iismod &= ~S3C2412_IISMOD_8BIT;
+ break;
+ }
+
+ writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
+ DBG("%s: w: IISMOD: %x\n", __func__, iismod);
+ return 0;
+}
+
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+ unsigned long irqs;
+ int ret = 0;
+
+ DBG("Entered %s\n", __func__);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ /* On start, ensure that the FIFOs are cleared and reset. */
+
+ writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH,
+ s3c2412_i2s.regs + S3C2412_IISFIC);
+
+ /* clear again, just in case */
+ writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC);
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!s3c2412_snd_is_clkmaster()) {
+ ret = s3c2412_snd_lrsync();
+ if (ret)
+ goto exit_err;
+ }
+
+ local_irq_save(irqs);
+
+ if (capture)
+ s3c2412_snd_rxctrl(1);
+ else
+ s3c2412_snd_txctrl(1);
+
+ local_irq_restore(irqs);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ local_irq_save(irqs);
+
+ if (capture)
+ s3c2412_snd_rxctrl(0);
+ else
+ s3c2412_snd_txctrl(0);
+
+ local_irq_restore(irqs);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+exit_err:
+ return ret;
+}
+
+/* default table of all avaialable root fs divisors */
+static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 };
+
+int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
+ unsigned int *fstab,
+ unsigned int rate, struct clk *clk)
+{
+ unsigned long clkrate = clk_get_rate(clk);
+ unsigned int div;
+ unsigned int fsclk;
+ unsigned int actual;
+ unsigned int fs;
+ unsigned int fsdiv;
+ signed int deviation = 0;
+ unsigned int best_fs = 0;
+ unsigned int best_div = 0;
+ unsigned int best_rate = 0;
+ unsigned int best_deviation = INT_MAX;
+
+
+ if (fstab == NULL)
+ fstab = s3c2412_iis_fs;
+
+ for (fs = 0;; fs++) {
+ fsdiv = s3c2412_iis_fs[fs];
+
+ if (fsdiv == 0)
+ break;
+
+ fsclk = clkrate / fsdiv;
+ div = fsclk / rate;
+
+ if ((fsclk % rate) > (rate / 2))
+ div++;
+
+ if (div <= 1)
+ continue;
+
+ actual = clkrate / (fsdiv * div);
+ deviation = actual - rate;
+
+ printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
+ fsdiv, div, actual, deviation);
+
+ deviation = abs(deviation);
+
+ if (deviation < best_deviation) {
+ best_fs = fsdiv;
+ best_div = div;
+ best_rate = actual;
+ best_deviation = deviation;
+ }
+
+ if (deviation == 0)
+ break;
+ }
+
+ printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
+ best_fs, best_div, best_rate);
+
+ info->fs_div = best_fs;
+ info->clk_div = best_div;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+
+/*
+ * Set S3C2412 Clock source
+ */
+static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
+
+ DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
+ freq, dir);
+
+ switch (clk_id) {
+ case S3C2412_CLKSRC_PCLK:
+ iismod &= ~S3C2412_IISMOD_MASTER_MASK;
+ iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
+ break;
+ case S3C2412_CLKSRC_I2SCLK:
+ iismod &= ~S3C2412_IISMOD_MASTER_MASK;
+ iismod |= S3C2412_IISMOD_MASTER_EXTERNAL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
+ return 0;
+}
+
+/*
+ * Set S3C2412 Clock dividers
+ */
+static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
+ u32 reg;
+
+ DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div);
+
+ switch (div_id) {
+ case S3C2412_DIV_BCLK:
+ reg = readl(i2s->regs + S3C2412_IISMOD);
+ reg &= ~S3C2412_IISMOD_BCLK_MASK;
+ writel(reg | div, i2s->regs + S3C2412_IISMOD);
+
+ DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+ break;
+
+ case S3C2412_DIV_RCLK:
+ if (div > 3) {
+ /* convert value to bit field */
+
+ switch (div) {
+ case 256:
+ div = S3C2412_IISMOD_RCLK_256FS;
+ break;
+
+ case 384:
+ div = S3C2412_IISMOD_RCLK_384FS;
+ break;
+
+ case 512:
+ div = S3C2412_IISMOD_RCLK_512FS;
+ break;
+
+ case 768:
+ div = S3C2412_IISMOD_RCLK_768FS;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ }
+
+ reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
+ reg &= ~S3C2412_IISMOD_RCLK_MASK;
+ writel(reg | div, i2s->regs + S3C2412_IISMOD);
+ DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+ break;
+
+ case S3C2412_DIV_PRESCALER:
+ if (div >= 0) {
+ writel((div << 8) | S3C2412_IISPSR_PSREN,
+ i2s->regs + S3C2412_IISPSR);
+ } else {
+ writel(0x0, i2s->regs + S3C2412_IISPSR);
+ }
+ DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR));
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+struct clk *s3c2412_get_iisclk(void)
+{
+ return s3c2412_i2s.iis_clk;
+}
+EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
+
+
+static int s3c2412_i2s_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ DBG("Entered %s\n", __func__);
+
+ s3c2412_i2s.dev = &pdev->dev;
+
+ s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
+ if (s3c2412_i2s.regs == NULL)
+ return -ENXIO;
+
+ s3c2412_i2s.iis_pclk = clk_get(&pdev->dev, "iis");
+ if (s3c2412_i2s.iis_pclk == NULL) {
+ DBG("failed to get iis_clock\n");
+ iounmap(s3c2412_i2s.regs);
+ return -ENODEV;
+ }
+
+ s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
+ if (s3c2412_i2s.iis_cclk == NULL) {
+ DBG("failed to get i2sclk clock\n");
+ iounmap(s3c2412_i2s.regs);
+ return -ENODEV;
+ }
+
+ clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
+
+ clk_enable(s3c2412_i2s.iis_pclk);
+ clk_enable(s3c2412_i2s.iis_cclk);
+
+ s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk;
+
+ /* Configure the I2S pins in correct mode */
+ s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
+ s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+
+ s3c2412_snd_txctrl(0);
+ s3c2412_snd_rxctrl(0);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int s3c2412_i2s_suspend(struct platform_device *dev,
+ struct snd_soc_dai *dai)
+{
+ struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
+ u32 iismod;
+
+ if (dai->active) {
+ i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
+ i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
+ i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
+
+ /* some basic suspend checks */
+
+ iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+ if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
+ dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__);
+
+ if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
+ dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__);
+
+ if (iismod & S3C2412_IISCON_IIS_ACTIVE)
+ dev_warn(&dev->dev, "%s: IIS active\n", __func__);
+ }
+
+ return 0;
+}
+
+static int s3c2412_i2s_resume(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
+
+ dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n",
+ dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+
+ if (dai->active) {
+ writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
+ writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
+ writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
+
+ writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
+ i2s->regs + S3C2412_IISFIC);
+
+ ndelay(250);
+ writel(0x0, i2s->regs + S3C2412_IISFIC);
+
+ }
+
+ return 0;
+}
+#else
+#define s3c2412_i2s_suspend NULL
+#define s3c2412_i2s_resume NULL
+#endif /* CONFIG_PM */
+
+#define S3C2412_I2S_RATES \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+struct snd_soc_dai s3c2412_i2s_dai = {
+ .name = "s3c2412-i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .probe = s3c2412_i2s_probe,
+ .suspend = s3c2412_i2s_suspend,
+ .resume = s3c2412_i2s_resume,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C2412_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C2412_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = {
+ .trigger = s3c2412_i2s_trigger,
+ .hw_params = s3c2412_i2s_hw_params,
+ },
+ .dai_ops = {
+ .set_fmt = s3c2412_i2s_set_fmt,
+ .set_clkdiv = s3c2412_i2s_set_clkdiv,
+ .set_sysclk = s3c2412_i2s_set_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
+
+/* Module information */
+MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
new file mode 100644
index 0000000..aac08a2
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -0,0 +1,38 @@
+/* sound/soc/s3c24xx/s3c2412-i2s.c
+ *
+ * ALSA Soc Audio Layer - S3C2412 I2S driver
+ *
+ * Copyright (c) 2007 Simtec Electronics
+ * http://armlinux.simtec.co.uk/
+ * Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+*/
+
+#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H
+#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__
+
+#define S3C2412_DIV_BCLK (1)
+#define S3C2412_DIV_RCLK (2)
+#define S3C2412_DIV_PRESCALER (3)
+
+#define S3C2412_CLKSRC_PCLK (0)
+#define S3C2412_CLKSRC_I2SCLK (1)
+
+extern struct clk *s3c2412_get_iisclk(void);
+
+extern struct snd_soc_dai s3c2412_i2s_dai;
+
+struct s3c2412_rate_calc {
+ unsigned int clk_div; /* for prescaler */
+ unsigned int fs_div; /* for root frame clock */
+};
+
+extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
+ unsigned int *fstab,
+ unsigned int rate, struct clk *clk);
+
+#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
new file mode 100644
index 0000000..19c5c3c
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -0,0 +1,398 @@
+/*
+ * s3c2443-ac97.c -- ALSA Soc Audio Layer
+ *
+ * (c) 2007 Wolfson Microelectronics PLC.
+ * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * Copyright (C) 2005, Sean Choi <sh428.choi@samsung.com>
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+#include <asm/plat-s3c/regs-ac97.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
+#include <asm/dma.h>
+#include <mach/dma.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-ac97.h"
+
+struct s3c24xx_ac97_info {
+ void __iomem *regs;
+ struct clk *ac97_clk;
+};
+static struct s3c24xx_ac97_info s3c24xx_ac97;
+
+static DECLARE_COMPLETION(ac97_completion);
+static u32 codec_ready;
+static DECLARE_MUTEX(ac97_mutex);
+
+static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ u32 ac_glbctrl;
+ u32 ac_codec_cmd;
+ u32 stat, addr, data;
+
+ down(&ac97_mutex);
+
+ codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
+ ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+ ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg);
+ writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+
+ udelay(50);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ wait_for_completion(&ac97_completion);
+
+ stat = readl(s3c24xx_ac97.regs + S3C_AC97_STAT);
+ addr = (stat >> 16) & 0x7f;
+ data = (stat & 0xffff);
+
+ if (addr != reg)
+ printk(KERN_ERR "s3c24xx-ac97: req addr = %02x,"
+ " rep addr = %02x\n", reg, addr);
+
+ up(&ac97_mutex);
+
+ return (unsigned short)data;
+}
+
+static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ u32 ac_glbctrl;
+ u32 ac_codec_cmd;
+
+ down(&ac97_mutex);
+
+ codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
+ ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+ ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val);
+ writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+
+ udelay(50);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ wait_for_completion(&ac97_completion);
+
+ ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+ ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ;
+ writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
+
+ up(&ac97_mutex);
+
+}
+
+static void s3c2443_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_WARMRESET;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = 0;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+}
+
+static void s3c2443_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = 0;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA |
+ S3C_AC97_GLBCTRL_PCMINTM_DMA | S3C_AC97_GLBCTRL_MICINTM_DMA;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+}
+
+static irqreturn_t s3c2443_ac97_irq(int irq, void *dev_id)
+{
+ int status;
+ u32 ac_glbctrl;
+
+ status = readl(s3c24xx_ac97.regs + S3C_AC97_GLBSTAT) & codec_ready;
+
+ if (status) {
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ complete(&ac97_completion);
+ }
+ return IRQ_HANDLED;
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = s3c2443_ac97_read,
+ .write = s3c2443_ac97_write,
+ .warm_reset = s3c2443_ac97_warm_reset,
+ .reset = s3c2443_ac97_cold_reset,
+};
+
+static struct s3c2410_dma_client s3c2443_dma_client_out = {
+ .name = "AC97 PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c2443_dma_client_in = {
+ .name = "AC97 PCM Stereo in"
+};
+
+static struct s3c2410_dma_client s3c2443_dma_client_micin = {
+ .name = "AC97 Mic Mono in"
+};
+
+static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = {
+ .client = &s3c2443_dma_client_out,
+ .channel = DMACH_PCM_OUT,
+ .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
+ .dma_size = 4,
+};
+
+static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = {
+ .client = &s3c2443_dma_client_in,
+ .channel = DMACH_PCM_IN,
+ .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
+ .dma_size = 4,
+};
+
+static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = {
+ .client = &s3c2443_dma_client_micin,
+ .channel = DMACH_MIC_IN,
+ .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA,
+ .dma_size = 4,
+};
+
+static int s3c2443_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ int ret;
+ u32 ac_glbctrl;
+
+ s3c24xx_ac97.regs = ioremap(S3C2440_PA_AC97, 0x100);
+ if (s3c24xx_ac97.regs == NULL)
+ return -ENXIO;
+
+ s3c24xx_ac97.ac97_clk = clk_get(&pdev->dev, "ac97");
+ if (s3c24xx_ac97.ac97_clk == NULL) {
+ printk(KERN_ERR "s3c2443-ac97 failed to get ac97_clock\n");
+ iounmap(s3c24xx_ac97.regs);
+ return -ENODEV;
+ }
+ clk_enable(s3c24xx_ac97.ac97_clk);
+
+ s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET);
+ s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC);
+ s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI);
+ s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = 0;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ msleep(1);
+
+ ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE;
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq,
+ IRQF_DISABLED, "AC97", NULL);
+ if (ret < 0) {
+ printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n");
+ clk_disable(s3c24xx_ac97.ac97_clk);
+ clk_put(s3c24xx_ac97.ac97_clk);
+ iounmap(s3c24xx_ac97.regs);
+ }
+ return ret;
+}
+
+static void s3c2443_ac97_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ free_irq(IRQ_S3C244x_AC97, NULL);
+ clk_disable(s3c24xx_ac97.ac97_clk);
+ clk_put(s3c24xx_ac97.ac97_clk);
+ iounmap(s3c24xx_ac97.regs);
+}
+
+static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out;
+ else
+ cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_in;
+
+ return 0;
+}
+
+static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA;
+ else
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK;
+ else
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK;
+ break;
+ }
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ return 0;
+}
+
+static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+ else
+ cpu_dai->dma_data = &s3c2443_ac97_mic_mono_in;
+
+ return 0;
+}
+
+static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ u32 ac_glbctrl;
+
+ ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK;
+ }
+ writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
+
+ return 0;
+}
+
+#define s3c2443_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai s3c2443_ac97_dai[] = {
+{
+ .name = "s3c2443-ac97",
+ .id = 0,
+ .type = SND_SOC_DAI_AC97,
+ .probe = s3c2443_ac97_probe,
+ .remove = s3c2443_ac97_remove,
+ .playback = {
+ .stream_name = "AC97 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = s3c2443_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "AC97 Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = s3c2443_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = s3c2443_ac97_hw_params,
+ .trigger = s3c2443_ac97_trigger},
+},
+{
+ .name = "pxa2xx-ac97-mic",
+ .id = 1,
+ .type = SND_SOC_DAI_AC97,
+ .capture = {
+ .stream_name = "AC97 Mic Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = s3c2443_AC97_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .hw_params = s3c2443_ac97_hw_mic_params,
+ .trigger = s3c2443_ac97_mic_trigger,},
+},
+};
+EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+MODULE_AUTHOR("Graeme Gregory");
+MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h
new file mode 100644
index 0000000..a96dcad
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx-ac97.h
@@ -0,0 +1,31 @@
+/*
+ * s3c24xx-ac97.c -- ALSA Soc Audio Layer
+ *
+ * (c) 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Revision history
+ * 10th Nov 2006 Initial version.
+ */
+
+#ifndef S3C24XXAC97_H_
+#define S3C24XXAC97_H_
+
+#define AC_CMD_ADDR(x) (x << 16)
+#define AC_CMD_DATA(x) (x & 0xffff)
+
+#ifdef CONFIG_CPU_S3C2440
+#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97
+#else
+#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97
+#endif
+
+extern struct snd_soc_dai s3c2443_ac97_dai[];
+
+#endif /*S3C24XXAC97_H_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
new file mode 100644
index 0000000..ba4476b
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -0,0 +1,483 @@
+/*
+ * s3c24xx-i2s.c -- ALSA Soc Audio Layer
+ *
+ * (c) 2006 Wolfson Microelectronics PLC.
+ * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * (c) 2004-2005 Simtec Electronics
+ * http://armlinux.simtec.co.uk/
+ * Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/jiffies.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
+#include <asm/dma.h>
+#include <mach/dma.h>
+
+#include <asm/plat-s3c24xx/regs-iis.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
+#define S3C24XX_I2S_DEBUG 0
+#if S3C24XX_I2S_DEBUG
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x)
+#else
+#define DBG(x...)
+#endif
+
+static struct s3c2410_dma_client s3c24xx_dma_client_out = {
+ .name = "I2S PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c24xx_dma_client_in = {
+ .name = "I2S PCM Stereo in"
+};
+
+static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = {
+ .client = &s3c24xx_dma_client_out,
+ .channel = DMACH_I2S_OUT,
+ .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO,
+ .dma_size = 2,
+};
+
+static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = {
+ .client = &s3c24xx_dma_client_in,
+ .channel = DMACH_I2S_IN,
+ .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO,
+ .dma_size = 2,
+};
+
+struct s3c24xx_i2s_info {
+ void __iomem *regs;
+ struct clk *iis_clk;
+ u32 iiscon;
+ u32 iismod;
+ u32 iisfcon;
+ u32 iispsr;
+};
+static struct s3c24xx_i2s_info s3c24xx_i2s;
+
+static void s3c24xx_snd_txctrl(int on)
+{
+ u32 iisfcon;
+ u32 iiscon;
+ u32 iismod;
+
+ DBG("Entered %s\n", __func__);
+
+ iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
+ iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
+ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
+
+ DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+
+ if (on) {
+ iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE;
+ iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN;
+ iiscon &= ~S3C2410_IISCON_TXIDLE;
+ iismod |= S3C2410_IISMOD_TXMODE;
+
+ writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
+ writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
+ } else {
+ /* note, we have to disable the FIFOs otherwise bad things
+ * seem to happen when the DMA stops. According to the
+ * Samsung supplied kernel, this should allow the DMA
+ * engine and FIFOs to reset. If this isn't allowed, the
+ * DMA engine will simply freeze randomly.
+ */
+
+ iisfcon &= ~S3C2410_IISFCON_TXENABLE;
+ iisfcon &= ~S3C2410_IISFCON_TXDMA;
+ iiscon |= S3C2410_IISCON_TXIDLE;
+ iiscon &= ~S3C2410_IISCON_TXDMAEN;
+ iismod &= ~S3C2410_IISMOD_TXMODE;
+
+ writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
+ writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
+ writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ }
+
+ DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+}
+
+static void s3c24xx_snd_rxctrl(int on)
+{
+ u32 iisfcon;
+ u32 iiscon;
+ u32 iismod;
+
+ DBG("Entered %s\n", __func__);
+
+ iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
+ iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
+ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
+
+ DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+
+ if (on) {
+ iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE;
+ iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN;
+ iiscon &= ~S3C2410_IISCON_RXIDLE;
+ iismod |= S3C2410_IISMOD_RXMODE;
+
+ writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
+ writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
+ } else {
+ /* note, we have to disable the FIFOs otherwise bad things
+ * seem to happen when the DMA stops. According to the
+ * Samsung supplied kernel, this should allow the DMA
+ * engine and FIFOs to reset. If this isn't allowed, the
+ * DMA engine will simply freeze randomly.
+ */
+
+ iisfcon &= ~S3C2410_IISFCON_RXENABLE;
+ iisfcon &= ~S3C2410_IISFCON_RXDMA;
+ iiscon |= S3C2410_IISCON_RXIDLE;
+ iiscon &= ~S3C2410_IISCON_RXDMAEN;
+ iismod &= ~S3C2410_IISMOD_RXMODE;
+
+ writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
+ writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
+ writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ }
+
+ DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+}
+
+/*
+ * Wait for the LR signal to allow synchronisation to the L/R clock
+ * from the codec. May only be needed for slave mode.
+ */
+static int s3c24xx_snd_lrsync(void)
+{
+ u32 iiscon;
+ int timeout = 50; /* 5ms */
+
+ DBG("Entered %s\n", __func__);
+
+ while (1) {
+ iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
+ if (iiscon & S3C2410_IISCON_LRINDEX)
+ break;
+
+ if (!timeout--)
+ return -ETIMEDOUT;
+ udelay(100);
+ }
+
+ return 0;
+}
+
+/*
+ * Check whether CPU is the master or slave
+ */
+static inline int s3c24xx_snd_is_clkmaster(void)
+{
+ DBG("Entered %s\n", __func__);
+
+ return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
+}
+
+/*
+ * Set S3C24xx I2S DAI format
+ */
+static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ u32 iismod;
+
+ DBG("Entered %s\n", __func__);
+
+ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
+ DBG("hw_params r: IISMOD: %lx \n", iismod);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iismod |= S3C2410_IISMOD_SLAVE;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ iismod &= ~S3C2410_IISMOD_SLAVE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_LEFT_J:
+ iismod |= S3C2410_IISMOD_MSB;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ iismod &= ~S3C2410_IISMOD_MSB;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ DBG("hw_params w: IISMOD: %lx \n", iismod);
+ return 0;
+}
+
+static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u32 iismod;
+
+ DBG("Entered %s\n", __func__);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
+ else
+ rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in;
+
+ /* Working copies of register */
+ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
+ DBG("hw_params r: IISMOD: %lx\n", iismod);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ iismod |= S3C2410_IISMOD_16BIT;
+ break;
+ }
+
+ writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ DBG("hw_params w: IISMOD: %lx\n", iismod);
+ return 0;
+}
+
+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+
+ DBG("Entered %s\n", __func__);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!s3c24xx_snd_is_clkmaster()) {
+ ret = s3c24xx_snd_lrsync();
+ if (ret)
+ goto exit_err;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s3c24xx_snd_rxctrl(1);
+ else
+ s3c24xx_snd_txctrl(1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s3c24xx_snd_rxctrl(0);
+ else
+ s3c24xx_snd_txctrl(0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+exit_err:
+ return ret;
+}
+
+/*
+ * Set S3C24xx Clock source
+ */
+static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
+
+ DBG("Entered %s\n", __func__);
+
+ iismod &= ~S3C2440_IISMOD_MPLL;
+
+ switch (clk_id) {
+ case S3C24XX_CLKSRC_PCLK:
+ break;
+ case S3C24XX_CLKSRC_MPLL:
+ iismod |= S3C2440_IISMOD_MPLL;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ return 0;
+}
+
+/*
+ * Set S3C24xx Clock dividers
+ */
+static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ u32 reg;
+
+ DBG("Entered %s\n", __func__);
+
+ switch (div_id) {
+ case S3C24XX_DIV_BCLK:
+ reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK;
+ writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ break;
+ case S3C24XX_DIV_MCLK:
+ reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS);
+ writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ break;
+ case S3C24XX_DIV_PRESCALER:
+ writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR);
+ reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
+ writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * To avoid duplicating clock code, allow machine driver to
+ * get the clockrate from here.
+ */
+u32 s3c24xx_i2s_get_clockrate(void)
+{
+ return clk_get_rate(s3c24xx_i2s.iis_clk);
+}
+EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
+
+static int s3c24xx_i2s_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ DBG("Entered %s\n", __func__);
+
+ s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
+ if (s3c24xx_i2s.regs == NULL)
+ return -ENXIO;
+
+ s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis");
+ if (s3c24xx_i2s.iis_clk == NULL) {
+ DBG("failed to get iis_clock\n");
+ iounmap(s3c24xx_i2s.regs);
+ return -ENODEV;
+ }
+ clk_enable(s3c24xx_i2s.iis_clk);
+
+ /* Configure the I2S pins in correct mode */
+ s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
+ s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
+ s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+
+ writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON);
+
+ s3c24xx_snd_txctrl(0);
+ s3c24xx_snd_rxctrl(0);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int s3c24xx_i2s_suspend(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ DBG("Entered %s\n", __func__);
+
+ s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
+ s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
+ s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
+ s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR);
+
+ clk_disable(s3c24xx_i2s.iis_clk);
+
+ return 0;
+}
+
+static int s3c24xx_i2s_resume(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ DBG("Entered %s\n", __func__);
+ clk_enable(s3c24xx_i2s.iis_clk);
+
+ writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
+ writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
+ writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
+ writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR);
+
+ return 0;
+}
+#else
+#define s3c24xx_i2s_suspend NULL
+#define s3c24xx_i2s_resume NULL
+#endif
+
+
+#define S3C24XX_I2S_RATES \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+struct snd_soc_dai s3c24xx_i2s_dai = {
+ .name = "s3c24xx-i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .probe = s3c24xx_i2s_probe,
+ .suspend = s3c24xx_i2s_suspend,
+ .resume = s3c24xx_i2s_resume,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C24XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C24XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .trigger = s3c24xx_i2s_trigger,
+ .hw_params = s3c24xx_i2s_hw_params,},
+ .dai_ops = {
+ .set_fmt = s3c24xx_i2s_set_fmt,
+ .set_clkdiv = s3c24xx_i2s_set_clkdiv,
+ .set_sysclk = s3c24xx_i2s_set_sysclk,
+ },
+};
+EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
+
+/* Module information */
+MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h
new file mode 100644
index 0000000..726d91c
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.h
@@ -0,0 +1,37 @@
+/*
+ * s3c24xx-i2s.c -- ALSA Soc Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Revision history
+ * 10th Nov 2006 Initial version.
+ */
+
+#ifndef S3C24XXI2S_H_
+#define S3C24XXI2S_H_
+
+/* clock sources */
+#define S3C24XX_CLKSRC_PCLK 0
+#define S3C24XX_CLKSRC_MPLL 1
+
+/* Clock dividers */
+#define S3C24XX_DIV_MCLK 0
+#define S3C24XX_DIV_BCLK 1
+#define S3C24XX_DIV_PRESCALER 2
+
+/* prescaler */
+#define S3C24XX_PRESCALE(a,b) \
+ (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT))
+
+u32 s3c24xx_i2s_get_clockrate(void);
+
+extern struct snd_soc_dai s3c24xx_i2s_dai;
+
+#endif /*S3C24XXI2S_H_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
new file mode 100644
index 0000000..e13e614
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -0,0 +1,470 @@
+/*
+ * s3c24xx-pcm.c -- ALSA Soc Audio Layer
+ *
+ * (c) 2006 Wolfson Microelectronics PLC.
+ * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * (c) 2004-2005 Simtec Electronics
+ * http://armlinux.simtec.co.uk/
+ * Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+#include <mach/hardware.h>
+#include <mach/dma.h>
+#include <mach/audio.h>
+
+#include "s3c24xx-pcm.h"
+
+#define S3C24XX_PCM_DEBUG 0
+#if S3C24XX_PCM_DEBUG
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x)
+#else
+#define DBG(x...)
+#endif
+
+static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_LE |
+ SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S8,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128*1024,
+ .period_bytes_min = PAGE_SIZE,
+ .period_bytes_max = PAGE_SIZE*2,
+ .periods_min = 2,
+ .periods_max = 128,
+ .fifo_size = 32,
+};
+
+struct s3c24xx_runtime_data {
+ spinlock_t lock;
+ int state;
+ unsigned int dma_loaded;
+ unsigned int dma_limit;
+ unsigned int dma_period;
+ dma_addr_t dma_start;
+ dma_addr_t dma_pos;
+ dma_addr_t dma_end;
+ struct s3c24xx_pcm_dma_params *params;
+};
+
+/* s3c24xx_pcm_enqueue
+ *
+ * place a dma buffer onto the queue for the dma system
+ * to handle.
+*/
+static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
+{
+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
+ dma_addr_t pos = prtd->dma_pos;
+ int ret;
+
+ DBG("Entered %s\n", __func__);
+
+ while (prtd->dma_loaded < prtd->dma_limit) {
+ unsigned long len = prtd->dma_period;
+
+ DBG("dma_loaded: %d\n", prtd->dma_loaded);
+
+ if ((pos + len) > prtd->dma_end) {
+ len = prtd->dma_end - pos;
+ DBG(KERN_DEBUG "%s: corrected dma len %ld\n",
+ __func__, len);
+ }
+
+ ret = s3c2410_dma_enqueue(prtd->params->channel,
+ substream, pos, len);
+
+ if (ret == 0) {
+ prtd->dma_loaded++;
+ pos += prtd->dma_period;
+ if (pos >= prtd->dma_end)
+ pos = prtd->dma_start;
+ } else
+ break;
+ }
+
+ prtd->dma_pos = pos;
+}
+
+static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
+ void *dev_id, int size,
+ enum s3c2410_dma_buffresult result)
+{
+ struct snd_pcm_substream *substream = dev_id;
+ struct s3c24xx_runtime_data *prtd;
+
+ DBG("Entered %s\n", __func__);
+
+ if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR)
+ return;
+
+ prtd = substream->runtime->private_data;
+
+ if (substream)
+ snd_pcm_period_elapsed(substream);
+
+ spin_lock(&prtd->lock);
+ if (prtd->state & ST_RUNNING) {
+ prtd->dma_loaded--;
+ s3c24xx_pcm_enqueue(substream);
+ }
+
+ spin_unlock(&prtd->lock);
+}
+
+static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s3c24xx_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+ unsigned long totbytes = params_buffer_bytes(params);
+ int ret = 0;
+
+ DBG("Entered %s\n", __func__);
+
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
+ if (!dma)
+ return 0;
+
+ /* this may get called several times by oss emulation
+ * with different params -HW */
+ if (prtd->params == NULL) {
+ /* prepare DMA */
+ prtd->params = dma;
+
+ DBG("params %p, client %p, channel %d\n", prtd->params,
+ prtd->params->client, prtd->params->channel);
+
+ ret = s3c2410_dma_request(prtd->params->channel,
+ prtd->params->client, NULL);
+
+ if (ret < 0) {
+ DBG(KERN_ERR "failed to get dma channel\n");
+ return ret;
+ }
+ }
+
+ s3c2410_dma_set_buffdone_fn(prtd->params->channel,
+ s3c24xx_audio_buffdone);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ runtime->dma_bytes = totbytes;
+
+ spin_lock_irq(&prtd->lock);
+ prtd->dma_loaded = 0;
+ prtd->dma_limit = runtime->hw.periods_min;
+ prtd->dma_period = params_period_bytes(params);
+ prtd->dma_start = runtime->dma_addr;
+ prtd->dma_pos = prtd->dma_start;
+ prtd->dma_end = prtd->dma_start + totbytes;
+ spin_unlock_irq(&prtd->lock);
+
+ return 0;
+}
+
+static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
+
+ DBG("Entered %s\n", __func__);
+
+ /* TODO - do we need to ensure DMA flushed */
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ if (prtd->params) {
+ s3c2410_dma_free(prtd->params->channel, prtd->params->client);
+ prtd->params = NULL;
+ }
+
+ return 0;
+}
+
+static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ DBG("Entered %s\n", __func__);
+
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
+ if (!prtd->params)
+ return 0;
+
+ /* channel needs configuring for mem=>device, increment memory addr,
+ * sync to pclk, half-word transfers to the IIS-FIFO. */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ s3c2410_dma_devconfig(prtd->params->channel,
+ S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC |
+ S3C2410_DISRCC_APB, prtd->params->dma_addr);
+
+ s3c2410_dma_config(prtd->params->channel,
+ prtd->params->dma_size,
+ S3C2410_DCON_SYNC_PCLK |
+ S3C2410_DCON_HANDSHAKE);
+ } else {
+ s3c2410_dma_config(prtd->params->channel,
+ prtd->params->dma_size,
+ S3C2410_DCON_HANDSHAKE |
+ S3C2410_DCON_SYNC_PCLK);
+
+ s3c2410_dma_devconfig(prtd->params->channel,
+ S3C2410_DMASRC_HW, 0x3,
+ prtd->params->dma_addr);
+ }
+
+ /* flush the DMA channel */
+ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH);
+ prtd->dma_loaded = 0;
+ prtd->dma_pos = prtd->dma_start;
+
+ /* enqueue dma buffers */
+ s3c24xx_pcm_enqueue(substream);
+
+ return ret;
+}
+
+static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ DBG("Entered %s\n", __func__);
+
+ spin_lock(&prtd->lock);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ prtd->state |= ST_RUNNING;
+ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
+ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ prtd->state &= ~ST_RUNNING;
+ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP);
+ break;
+
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ spin_unlock(&prtd->lock);
+
+ return ret;
+}
+
+static snd_pcm_uframes_t
+s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s3c24xx_runtime_data *prtd = runtime->private_data;
+ unsigned long res;
+ dma_addr_t src, dst;
+
+ DBG("Entered %s\n", __func__);
+
+ spin_lock(&prtd->lock);
+ s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ res = dst - prtd->dma_start;
+ else
+ res = src - prtd->dma_start;
+
+ spin_unlock(&prtd->lock);
+
+ DBG("Pointer %x %x\n", src, dst);
+
+ /* we seem to be getting the odd error from the pcm library due
+ * to out-of-bounds pointers. this is maybe due to the dma engine
+ * not having loaded the new values for the channel before being
+ * callled... (todo - fix )
+ */
+
+ if (res >= snd_pcm_lib_buffer_bytes(substream)) {
+ if (res == snd_pcm_lib_buffer_bytes(substream))
+ res = 0;
+ }
+
+ return bytes_to_frames(substream->runtime, res);
+}
+
+static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s3c24xx_runtime_data *prtd;
+
+ DBG("Entered %s\n", __func__);
+
+ snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
+
+ prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ spin_lock_init(&prtd->lock);
+
+ runtime->private_data = prtd;
+ return 0;
+}
+
+static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s3c24xx_runtime_data *prtd = runtime->private_data;
+
+ DBG("Entered %s\n", __func__);
+
+ if (!prtd)
+ DBG("s3c24xx_pcm_close called with prtd == NULL\n");
+
+ kfree(prtd);
+
+ return 0;
+}
+
+static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ DBG("Entered %s\n", __func__);
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops s3c24xx_pcm_ops = {
+ .open = s3c24xx_pcm_open,
+ .close = s3c24xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = s3c24xx_pcm_hw_params,
+ .hw_free = s3c24xx_pcm_hw_free,
+ .prepare = s3c24xx_pcm_prepare,
+ .trigger = s3c24xx_pcm_trigger,
+ .pointer = s3c24xx_pcm_pointer,
+ .mmap = s3c24xx_pcm_mmap,
+};
+
+static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = s3c24xx_pcm_hardware.buffer_bytes_max;
+
+ DBG("Entered %s\n", __func__);
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+ return 0;
+}
+
+static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ DBG("Entered %s\n", __func__);
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK;
+
+static int s3c24xx_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ DBG("Entered %s\n", __func__);
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &s3c24xx_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ goto out;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ goto out;
+ }
+ out:
+ return ret;
+}
+
+struct snd_soc_platform s3c24xx_soc_platform = {
+ .name = "s3c24xx-audio",
+ .pcm_ops = &s3c24xx_pcm_ops,
+ .pcm_new = s3c24xx_pcm_new,
+ .pcm_free = s3c24xx_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
+
+MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c24xx-pcm.h
new file mode 100644
index 0000000..0088c79
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.h
@@ -0,0 +1,31 @@
+/*
+ * s3c24xx-pcm.h --
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * ALSA PCM interface for the Samsung S3C24xx CPU
+ */
+
+#ifndef _S3C24XX_PCM_H
+#define _S3C24XX_PCM_H
+
+#define ST_RUNNING (1<<0)
+#define ST_OPENED (1<<1)
+
+struct s3c24xx_pcm_dma_params {
+ struct s3c2410_dma_client *client; /* stream identifier */
+ int channel; /* Channel ID */
+ dma_addr_t dma_addr;
+ int dma_size; /* Size of the DMA transfer */
+};
+
+#define S3C24XX_DAI_I2S 0
+
+/* platform data */
+extern struct snd_soc_platform s3c24xx_soc_platform;
+extern struct snd_ac97_bus_ops s3c24xx_ac97_ops;
+
+#endif
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
new file mode 100644
index 0000000..8515d6f
--- /dev/null
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -0,0 +1,81 @@
+/*
+ * smdk2443_wm9710.c -- SoC audio for smdk2443
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/ac97.h"
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-ac97.h"
+
+static struct snd_soc_machine smdk2443;
+
+static struct snd_soc_dai_link smdk2443_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &s3c2443_ac97_dai[0],
+ .codec_dai = &ac97_dai,
+},
+};
+
+static struct snd_soc_machine smdk2443 = {
+ .name = "SMDK2443",
+ .dai_link = smdk2443_dai,
+ .num_links = ARRAY_SIZE(smdk2443_dai),
+};
+
+static struct snd_soc_device smdk2443_snd_ac97_devdata = {
+ .machine = &smdk2443,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_ac97,
+};
+
+static struct platform_device *smdk2443_snd_ac97_device;
+
+static int __init smdk2443_init(void)
+{
+ int ret;
+
+ smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+ if (!smdk2443_snd_ac97_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(smdk2443_snd_ac97_device,
+ &smdk2443_snd_ac97_devdata);
+ smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev;
+ ret = platform_device_add(smdk2443_snd_ac97_device);
+
+ if (ret)
+ platform_device_put(smdk2443_snd_ac97_device);
+
+ return ret;
+}
+
+static void __exit smdk2443_exit(void)
+{
+ platform_device_unregister(smdk2443_snd_ac97_device);
+}
+
+module_init(smdk2443_init);
+module_exit(smdk2443_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
new file mode 100644
index 0000000..54bd604
--- /dev/null
+++ b/sound/soc/sh/Kconfig
@@ -0,0 +1,38 @@
+menu "SoC Audio support for SuperH"
+ depends on SUPERH
+
+config SND_SOC_PCM_SH7760
+ tristate "SoC Audio support for Renesas SH7760"
+ depends on CPU_SUBTYPE_SH7760 && SH_DMABRG
+ help
+ Enable this option for SH7760 AC97/I2S audio support.
+
+
+##
+## Audio unit modules
+##
+
+config SND_SOC_SH4_HAC
+ tristate
+ select AC97_BUS
+ select SND_SOC_AC97_BUS
+
+config SND_SOC_SH4_SSI
+ tristate
+
+
+
+##
+## Boards
+##
+
+config SND_SH7760_AC97
+ tristate "SH7760 AC97 sound support"
+ depends on CPU_SUBTYPE_SH7760 && SND_SOC_PCM_SH7760
+ select SND_SOC_SH4_HAC
+ select SND_SOC_AC97_CODEC
+ help
+ This option enables generic sound support for the first
+ AC97 unit of the SH7760.
+
+endmenu
diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile
new file mode 100644
index 0000000..a8e8ab8
--- /dev/null
+++ b/sound/soc/sh/Makefile
@@ -0,0 +1,14 @@
+## DMA engines
+snd-soc-dma-sh7760-objs := dma-sh7760.o
+obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o
+
+## audio units found on some SH-4
+snd-soc-hac-objs := hac.o
+snd-soc-ssi-objs := ssi.o
+obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o
+obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o
+
+## boards
+snd-soc-sh7760-ac97-objs := sh7760-ac97.o
+
+obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
new file mode 100644
index 0000000..9faa126
--- /dev/null
+++ b/sound/soc/sh/dma-sh7760.c
@@ -0,0 +1,353 @@
+/*
+ * SH7760 ("camelot") DMABRG audio DMA unit support
+ *
+ * Copyright (C) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ * licensed under the terms outlined in the file COPYING at the root
+ * of the linux kernel sources.
+ *
+ * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which
+ * trigger an interrupt when one half of the programmed transfer size
+ * has been xmitted.
+ *
+ * FIXME: little-endian only for now
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <asm/dmabrg.h>
+
+
+/* registers and bits */
+#define BRGATXSAR 0x00
+#define BRGARXDAR 0x04
+#define BRGATXTCR 0x08
+#define BRGARXTCR 0x0C
+#define BRGACR 0x10
+#define BRGATXTCNT 0x14
+#define BRGARXTCNT 0x18
+
+#define ACR_RAR (1 << 18)
+#define ACR_RDS (1 << 17)
+#define ACR_RDE (1 << 16)
+#define ACR_TAR (1 << 2)
+#define ACR_TDS (1 << 1)
+#define ACR_TDE (1 << 0)
+
+/* receiver/transmitter data alignment */
+#define ACR_RAM_NONE (0 << 24)
+#define ACR_RAM_4BYTE (1 << 24)
+#define ACR_RAM_2WORD (2 << 24)
+#define ACR_TAM_NONE (0 << 8)
+#define ACR_TAM_4BYTE (1 << 8)
+#define ACR_TAM_2WORD (2 << 8)
+
+
+struct camelot_pcm {
+ unsigned long mmio; /* DMABRG audio channel control reg MMIO */
+ unsigned int txid; /* ID of first DMABRG IRQ for this unit */
+
+ struct snd_pcm_substream *tx_ss;
+ unsigned long tx_period_size;
+ unsigned int tx_period;
+
+ struct snd_pcm_substream *rx_ss;
+ unsigned long rx_period_size;
+ unsigned int rx_period;
+
+} cam_pcm_data[2] = {
+ {
+ .mmio = 0xFE3C0040,
+ .txid = DMABRGIRQ_A0TXF,
+ },
+ {
+ .mmio = 0xFE3C0060,
+ .txid = DMABRGIRQ_A1TXF,
+ },
+};
+
+#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x)))
+
+/*
+ * set a minimum of 16kb per period, to avoid interrupt-"storm" and
+ * resulting skipping. In general, the bigger the minimum size, the
+ * better for overall system performance. (The SH7760 is a puny CPU
+ * with a slow SDRAM interface and poor internal bus bandwidth,
+ * *especially* when the LCDC is active). The minimum for the DMAC
+ * is 8 bytes; 16kbytes are enough to get skip-free playback of a
+ * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain
+ * reasonable responsiveness in MPlayer.
+ */
+#define DMABRG_PERIOD_MIN 16 * 1024
+#define DMABRG_PERIOD_MAX 0x03fffffc
+#define DMABRG_PREALLOC_BUFFER 32 * 1024
+#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024
+
+/* support everything the SSI supports */
+#define DMABRG_RATES \
+ SNDRV_PCM_RATE_8000_192000
+
+#define DMABRG_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
+
+static struct snd_pcm_hardware camelot_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = DMABRG_FMTS,
+ .rates = DMABRG_RATES,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 8, /* max of the SSI */
+ .buffer_bytes_max = DMABRG_PERIOD_MAX,
+ .period_bytes_min = DMABRG_PERIOD_MIN,
+ .period_bytes_max = DMABRG_PERIOD_MAX / 2,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 128,
+};
+
+static void camelot_txdma(void *data)
+{
+ struct camelot_pcm *cam = data;
+ cam->tx_period ^= 1;
+ snd_pcm_period_elapsed(cam->tx_ss);
+}
+
+static void camelot_rxdma(void *data)
+{
+ struct camelot_pcm *cam = data;
+ cam->rx_period ^= 1;
+ snd_pcm_period_elapsed(cam->rx_ss);
+}
+
+static int camelot_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ int ret, dmairq;
+
+ snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware);
+
+ /* DMABRG buffer half/full events */
+ dmairq = (recv) ? cam->txid + 2 : cam->txid;
+ if (recv) {
+ cam->rx_ss = substream;
+ ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam);
+ if (unlikely(ret)) {
+ pr_debug("audio unit %d irqs already taken!\n",
+ rtd->dai->cpu_dai->id);
+ return -EBUSY;
+ }
+ (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam);
+ } else {
+ cam->tx_ss = substream;
+ ret = dmabrg_request_irq(dmairq, camelot_txdma, cam);
+ if (unlikely(ret)) {
+ pr_debug("audio unit %d irqs already taken!\n",
+ rtd->dai->cpu_dai->id);
+ return -EBUSY;
+ }
+ (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam);
+ }
+ return 0;
+}
+
+static int camelot_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ int dmairq;
+
+ dmairq = (recv) ? cam->txid + 2 : cam->txid;
+
+ if (recv)
+ cam->rx_ss = NULL;
+ else
+ cam->tx_ss = NULL;
+
+ dmabrg_free_irq(dmairq + 1);
+ dmabrg_free_irq(dmairq);
+
+ return 0;
+}
+
+static int camelot_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ int ret;
+
+ ret = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (ret < 0)
+ return ret;
+
+ if (recv) {
+ cam->rx_period_size = params_period_bytes(hw_params);
+ cam->rx_period = 0;
+ } else {
+ cam->tx_period_size = params_period_bytes(hw_params);
+ cam->tx_period = 0;
+ }
+ return 0;
+}
+
+static int camelot_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int camelot_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+
+ pr_debug("PCM data: addr 0x%08ulx len %d\n",
+ (u32)runtime->dma_addr, runtime->dma_bytes);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ BRGREG(BRGATXSAR) = (unsigned long)runtime->dma_area;
+ BRGREG(BRGATXTCR) = runtime->dma_bytes;
+ } else {
+ BRGREG(BRGARXDAR) = (unsigned long)runtime->dma_area;
+ BRGREG(BRGARXTCR) = runtime->dma_bytes;
+ }
+
+ return 0;
+}
+
+static inline void dmabrg_play_dma_start(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* start DMABRG engine: XFER start, auto-addr-reload */
+ BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD;
+}
+
+static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* forcibly terminate data transmission */
+ BRGREG(BRGACR) = acr | ACR_TDS;
+}
+
+static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* start DMABRG engine: recv start, auto-reload */
+ BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD;
+}
+
+static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam)
+{
+ unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS);
+ /* forcibly terminate data receiver */
+ BRGREG(BRGACR) = acr | ACR_RDS;
+}
+
+static int camelot_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (recv)
+ dmabrg_rec_dma_start(cam);
+ else
+ dmabrg_play_dma_start(cam);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (recv)
+ dmabrg_rec_dma_stop(cam);
+ else
+ dmabrg_play_dma_stop(cam);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id];
+ int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1;
+ unsigned long pos;
+
+ /* cannot use the DMABRG pointer register: under load, by the
+ * time ALSA comes around to read the register, it is already
+ * far ahead (or worse, already done with the fragment) of the
+ * position at the time the IRQ was triggered, which results in
+ * fast-playback sound in my test application (ScummVM)
+ */
+ if (recv)
+ pos = cam->rx_period ? cam->rx_period_size : 0;
+ else
+ pos = cam->tx_period ? cam->tx_period_size : 0;
+
+ return bytes_to_frames(runtime, pos);
+}
+
+static struct snd_pcm_ops camelot_pcm_ops = {
+ .open = camelot_pcm_open,
+ .close = camelot_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = camelot_hw_params,
+ .hw_free = camelot_hw_free,
+ .prepare = camelot_prepare,
+ .trigger = camelot_trigger,
+ .pointer = camelot_pos,
+};
+
+static void camelot_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int camelot_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
+ * in MMAP mode (i.e. aplay -M)
+ */
+ snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX);
+
+ return 0;
+}
+
+struct snd_soc_platform sh7760_soc_platform = {
+ .name = "sh7760-pcm",
+ .pcm_ops = &camelot_pcm_ops,
+ .pcm_new = camelot_pcm_new,
+ .pcm_free = camelot_pcm_free,
+};
+EXPORT_SYMBOL_GPL(sh7760_soc_platform);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
new file mode 100644
index 0000000..df7bc34
--- /dev/null
+++ b/sound/soc/sh/hac.c
@@ -0,0 +1,318 @@
+/*
+ * Hitachi Audio Controller (AC97) support for SH7760/SH7780
+ *
+ * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ * licensed under the terms outlined in the file COPYING at the root
+ * of the linux kernel sources.
+ *
+ * dont forget to set IPSEL/OMSEL register bits (in your board code) to
+ * enable HAC output pins!
+ */
+
+/* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only
+ * the FIRST can be used since ASoC does not pass any information to the
+ * ac97_read/write() functions regarding WHICH unit to use. You'll have
+ * to edit the code a bit to use the other AC97 unit. --mlau
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+/* regs and bits */
+#define HACCR 0x08
+#define HACCSAR 0x20
+#define HACCSDR 0x24
+#define HACPCML 0x28
+#define HACPCMR 0x2C
+#define HACTIER 0x50
+#define HACTSR 0x54
+#define HACRIER 0x58
+#define HACRSR 0x5C
+#define HACACR 0x60
+
+#define CR_CR (1 << 15) /* "codec-ready" indicator */
+#define CR_CDRT (1 << 11) /* cold reset */
+#define CR_WMRT (1 << 10) /* warm reset */
+#define CR_B9 (1 << 9) /* the mysterious "bit 9" */
+#define CR_ST (1 << 5) /* AC97 link start bit */
+
+#define CSAR_RD (1 << 19) /* AC97 data read bit */
+#define CSAR_WR (0)
+
+#define TSR_CMDAMT (1 << 31)
+#define TSR_CMDDMT (1 << 30)
+
+#define RSR_STARY (1 << 22)
+#define RSR_STDRY (1 << 21)
+
+#define ACR_DMARX16 (1 << 30)
+#define ACR_DMATX16 (1 << 29)
+#define ACR_TX12ATOM (1 << 26)
+#define ACR_DMARX20 ((1 << 24) | (1 << 22))
+#define ACR_DMATX20 ((1 << 23) | (1 << 21))
+
+#define CSDR_SHIFT 4
+#define CSDR_MASK (0xffff << CSDR_SHIFT)
+#define CSAR_SHIFT 12
+#define CSAR_MASK (0x7f << CSAR_SHIFT)
+
+#define AC97_WRITE_RETRY 1
+#define AC97_READ_RETRY 5
+
+/* manual-suggested AC97 codec access timeouts (us) */
+#define TMO_E1 500 /* 21 < E1 < 1000 */
+#define TMO_E2 13 /* 13 < E2 */
+#define TMO_E3 21 /* 21 < E3 */
+#define TMO_E4 500 /* 21 < E4 < 1000 */
+
+struct hac_priv {
+ unsigned long mmio; /* HAC base address */
+} hac_cpu_data[] = {
+#if defined(CONFIG_CPU_SUBTYPE_SH7760)
+ {
+ .mmio = 0xFE240000,
+ },
+ {
+ .mmio = 0xFE250000,
+ },
+#elif defined(CONFIG_CPU_SUBTYPE_SH7780)
+ {
+ .mmio = 0xFFE40000,
+ },
+#else
+#error "Unsupported SuperH SoC"
+#endif
+};
+
+#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg)))
+
+/*
+ * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906)
+ */
+static int hac_get_codec_data(struct hac_priv *hac, unsigned short r,
+ unsigned short *v)
+{
+ unsigned int to1, to2, i;
+ unsigned short adr;
+
+ for (i = AC97_READ_RETRY; i; i--) {
+ *v = 0;
+ /* wait for HAC to receive something from the codec */
+ for (to1 = TMO_E4;
+ to1 && !(HACREG(HACRSR) & RSR_STARY);
+ --to1)
+ udelay(1);
+ for (to2 = TMO_E4;
+ to2 && !(HACREG(HACRSR) & RSR_STDRY);
+ --to2)
+ udelay(1);
+
+ if (!to1 && !to2)
+ return 0; /* codec comm is down */
+
+ adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT);
+ *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT);
+
+ HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY);
+
+ if (r == adr)
+ break;
+
+ /* manual says: wait at least 21 usec before retrying */
+ udelay(21);
+ }
+ HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY);
+ return i;
+}
+
+static unsigned short hac_read_codec_aux(struct hac_priv *hac,
+ unsigned short reg)
+{
+ unsigned short val;
+ unsigned int i, to;
+
+ for (i = AC97_READ_RETRY; i; i--) {
+ /* send_read_request */
+ local_irq_disable();
+ HACREG(HACTSR) &= ~(TSR_CMDAMT);
+ HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD;
+ local_irq_enable();
+
+ for (to = TMO_E3;
+ to && !(HACREG(HACTSR) & TSR_CMDAMT);
+ --to)
+ udelay(1);
+
+ HACREG(HACTSR) &= ~TSR_CMDAMT;
+ val = 0;
+ if (hac_get_codec_data(hac, reg, &val) != 0)
+ break;
+ }
+
+ return i ? val : ~0;
+}
+
+static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac = &hac_cpu_data[unit_id];
+ unsigned int i, to;
+ /* write_codec_aux */
+ for (i = AC97_WRITE_RETRY; i; i--) {
+ /* send_write_request */
+ local_irq_disable();
+ HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT);
+ HACREG(HACCSDR) = (val << CSDR_SHIFT);
+ HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD);
+ local_irq_enable();
+
+ /* poll-wait for CMDAMT and CMDDMT */
+ for (to = TMO_E1;
+ to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT));
+ --to)
+ udelay(1);
+
+ HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT);
+ if (to)
+ break;
+ /* timeout, try again */
+ }
+}
+
+static unsigned short hac_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac = &hac_cpu_data[unit_id];
+ return hac_read_codec_aux(hac, reg);
+}
+
+static void hac_ac97_warmrst(struct snd_ac97 *ac97)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac = &hac_cpu_data[unit_id];
+ unsigned int tmo;
+
+ HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9;
+ msleep(10);
+ HACREG(HACCR) = CR_ST | CR_B9;
+ for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--)
+ udelay(1);
+
+ if (!tmo)
+ printk(KERN_INFO "hac: reset: AC97 link down!\n");
+ /* settings this bit lets us have a conversation with codec */
+ HACREG(HACACR) |= ACR_TX12ATOM;
+}
+
+static void hac_ac97_coldrst(struct snd_ac97 *ac97)
+{
+ int unit_id = 0 /* ac97->private_data */;
+ struct hac_priv *hac;
+ hac = &hac_cpu_data[unit_id];
+
+ HACREG(HACCR) = 0;
+ HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9;
+ msleep(10);
+ hac_ac97_warmrst(ac97);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = hac_ac97_read,
+ .write = hac_ac97_write,
+ .reset = hac_ac97_coldrst,
+ .warm_reset = hac_ac97_warmrst,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int hac_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id];
+ int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+ switch (params->msbits) {
+ case 16:
+ HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16;
+ HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20;
+ break;
+ case 20:
+ HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16;
+ HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20;
+ break;
+ default:
+ pr_debug("hac: invalid depth %d bit\n", params->msbits);
+ return -EINVAL;
+ break;
+ }
+
+ return 0;
+}
+
+#define AC97_RATES \
+ SNDRV_PCM_RATE_8000_192000
+
+#define AC97_FMTS \
+ SNDRV_PCM_FMTBIT_S16_LE
+
+struct snd_soc_dai sh4_hac_dai[] = {
+{
+ .name = "HAC0",
+ .id = 0,
+ .type = SND_SOC_DAI_AC97,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = {
+ .hw_params = hac_hw_params,
+ },
+},
+#ifdef CONFIG_CPU_SUBTYPE_SH7760
+{
+ .name = "HAC1",
+ .id = 1,
+ .type = SND_SOC_DAI_AC97,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .ops = {
+ .hw_params = hac_hw_params,
+ },
+
+},
+#endif
+};
+EXPORT_SYMBOL_GPL(sh4_hac_dai);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
new file mode 100644
index 0000000..92bfaf4
--- /dev/null
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -0,0 +1,91 @@
+/*
+ * Generic AC97 sound support for SH7760
+ *
+ * (c) 2007 Manuel Lauss
+ *
+ * Licensed under the GPLv2.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/io.h>
+
+#include "../codecs/ac97.h"
+
+#define IPSEL 0xFE400034
+
+/* platform specific structs can be declared here */
+extern struct snd_soc_dai sh4_hac_dai[2];
+extern struct snd_soc_platform sh7760_soc_platform;
+
+static int machine_init(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_sync(codec);
+ return 0;
+}
+
+static struct snd_soc_dai_link sh7760_ac97_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &sh4_hac_dai[0], /* HAC0 */
+ .codec_dai = &ac97_dai,
+ .init = machine_init,
+ .ops = NULL,
+};
+
+static struct snd_soc_machine sh7760_ac97_soc_machine = {
+ .name = "SH7760 AC97",
+ .dai_link = &sh7760_ac97_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device sh7760_ac97_snd_devdata = {
+ .machine = &sh7760_ac97_soc_machine,
+ .platform = &sh7760_soc_platform,
+ .codec_dev = &soc_codec_dev_ac97,
+};
+
+static struct platform_device *sh7760_ac97_snd_device;
+
+static int __init sh7760_ac97_init(void)
+{
+ int ret;
+ unsigned short ipsel;
+
+ /* enable both AC97 controllers in pinmux reg */
+ ipsel = ctrl_inw(IPSEL);
+ ctrl_outw(ipsel | (3 << 10), IPSEL);
+
+ ret = -ENOMEM;
+ sh7760_ac97_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!sh7760_ac97_snd_device)
+ goto out;
+
+ platform_set_drvdata(sh7760_ac97_snd_device,
+ &sh7760_ac97_snd_devdata);
+ sh7760_ac97_snd_devdata.dev = &sh7760_ac97_snd_device->dev;
+ ret = platform_device_add(sh7760_ac97_snd_device);
+
+ if (ret)
+ platform_device_put(sh7760_ac97_snd_device);
+
+out:
+ return ret;
+}
+
+static void __exit sh7760_ac97_exit(void)
+{
+ platform_device_unregister(sh7760_ac97_snd_device);
+}
+
+module_init(sh7760_ac97_init);
+module_exit(sh7760_ac97_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
new file mode 100644
index 0000000..55c3464
--- /dev/null
+++ b/sound/soc/sh/ssi.c
@@ -0,0 +1,399 @@
+/*
+ * Serial Sound Interface (I2S) support for SH7760/SH7780
+ *
+ * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * licensed under the terms outlined in the file COPYING at the root
+ * of the linux kernel sources.
+ *
+ * dont forget to set IPSEL/OMSEL register bits (in your board code) to
+ * enable SSI output pins!
+ */
+
+/*
+ * LIMITATIONS:
+ * The SSI unit has only one physical data line, so full duplex is
+ * impossible. This can be remedied on the SH7760 by using the
+ * other SSI unit for recording; however the SH7780 has only 1 SSI
+ * unit, and its pins are shared with the AC97 unit, among others.
+ *
+ * FEATURES:
+ * The SSI features "compressed mode": in this mode it continuously
+ * streams PCM data over the I2S lines and uses LRCK as a handshake
+ * signal. Can be used to send compressed data (AC3/DTS) to a DSP.
+ * The number of bits sent over the wire in a frame can be adjusted
+ * and can be independent from the actual sample bit depth. This is
+ * useful to support TDM mode codecs like the AD1939 which have a
+ * fixed TDM slot size, regardless of sample resolution.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/io.h>
+
+#define SSICR 0x00
+#define SSISR 0x04
+
+#define CR_DMAEN (1 << 28)
+#define CR_CHNL_SHIFT 22
+#define CR_CHNL_MASK (3 << CR_CHNL_SHIFT)
+#define CR_DWL_SHIFT 19
+#define CR_DWL_MASK (7 << CR_DWL_SHIFT)
+#define CR_SWL_SHIFT 16
+#define CR_SWL_MASK (7 << CR_SWL_SHIFT)
+#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */
+#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */
+#define CR_SCKP (1 << 13) /* I2Sclock polarity */
+#define CR_SWSP (1 << 12) /* LRCK polarity */
+#define CR_SPDP (1 << 11)
+#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */
+#define CR_PDTA (1 << 9) /* fifo data alignment */
+#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */
+#define CR_BREN (1 << 7) /* clock gating in burst mode */
+#define CR_CKDIV_SHIFT 4
+#define CR_CKDIV_MASK (7 << CR_CKDIV_SHIFT) /* bitclock divider */
+#define CR_MUTE (1 << 3) /* SSI mute */
+#define CR_CPEN (1 << 2) /* compressed mode */
+#define CR_TRMD (1 << 1) /* transmit/receive select */
+#define CR_EN (1 << 0) /* enable SSI */
+
+#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg)))
+
+struct ssi_priv {
+ unsigned long mmio;
+ unsigned long sysclk;
+ int inuse;
+} ssi_cpu_data[] = {
+#if defined(CONFIG_CPU_SUBTYPE_SH7760)
+ {
+ .mmio = 0xFE680000,
+ },
+ {
+ .mmio = 0xFE690000,
+ },
+#elif defined(CONFIG_CPU_SUBTYPE_SH7780)
+ {
+ .mmio = 0xFFE70000,
+ },
+#else
+#error "Unsupported SuperH SoC"
+#endif
+};
+
+/*
+ * track usage of the SSI; it is simplex-only so prevent attempts of
+ * concurrent playback + capture. FIXME: any locking required?
+ */
+static int ssi_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+ if (ssi->inuse) {
+ pr_debug("ssi: already in use!\n");
+ return -EBUSY;
+ } else
+ ssi->inuse = 1;
+ return 0;
+}
+
+static void ssi_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+
+ ssi->inuse = 0;
+}
+
+static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ SSIREG(SSICR) |= CR_DMAEN | CR_EN;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ssi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
+ unsigned long ssicr = SSIREG(SSICR);
+ unsigned int bits, channels, swl, recv, i;
+
+ channels = params_channels(params);
+ bits = params->msbits;
+ recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1;
+
+ pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr);
+ pr_debug("bits: %d channels: %d\n", bits, channels);
+
+ ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA |
+ CR_SWL_MASK);
+
+ /* direction (send/receive) */
+ if (!recv)
+ ssicr |= CR_TRMD; /* transmit */
+
+ /* channels */
+ if ((channels < 2) || (channels > 8) || (channels & 1)) {
+ pr_debug("ssi: invalid number of channels\n");
+ return -EINVAL;
+ }
+ ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT;
+
+ /* DATA WORD LENGTH (DWL): databits in audio sample */
+ i = 0;
+ switch (bits) {
+ case 32: ++i;
+ case 24: ++i;
+ case 22: ++i;
+ case 20: ++i;
+ case 18: ++i;
+ case 16: ++i;
+ ssicr |= i << CR_DWL_SHIFT;
+ case 8: break;
+ default:
+ pr_debug("ssi: invalid sample width\n");
+ return -EINVAL;
+ }
+
+ /*
+ * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S
+ * wires. This is usually bits_per_sample x channels/2; i.e. in
+ * Stereo mode the SWL equals DWL. SWL can be bigger than the
+ * product of (channels_per_slot x samplebits), e.g. for codecs
+ * like the AD1939 which only accept 32bit wide TDM slots. For
+ * "standard" I2S operation we set SWL = chans / 2 * DWL here.
+ * Waiting for ASoC to get TDM support ;-)
+ */
+ if ((bits > 16) && (bits <= 24)) {
+ bits = 24; /* these are padded by the SSI */
+ /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */
+ }
+ i = 0;
+ swl = (bits * channels) / 2;
+ switch (swl) {
+ case 256: ++i;
+ case 128: ++i;
+ case 64: ++i;
+ case 48: ++i;
+ case 32: ++i;
+ case 16: ++i;
+ ssicr |= i << CR_SWL_SHIFT;
+ case 8: break;
+ default:
+ pr_debug("ssi: invalid system word length computed\n");
+ return -EINVAL;
+ }
+
+ SSIREG(SSICR) = ssicr;
+
+ pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr);
+ return 0;
+}
+
+static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id];
+
+ ssi->sysclk = freq;
+
+ return 0;
+}
+
+/*
+ * This divider is used to generate the SSI_SCK (I2S bitclock) from the
+ * clock at the HAC_BIT_CLK ("oversampling clock") pin.
+ */
+static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div)
+{
+ struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
+ unsigned long ssicr;
+ int i;
+
+ i = 0;
+ ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK;
+ switch (div) {
+ case 16: ++i;
+ case 8: ++i;
+ case 4: ++i;
+ case 2: ++i;
+ SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT);
+ case 1: break;
+ default:
+ pr_debug("ssi: invalid sck divider %d\n", div);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
+ unsigned long ssicr = SSIREG(SSICR);
+
+ pr_debug("ssi_set_fmt()\nssicr was 0x%08lx\n", ssicr);
+
+ ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP |
+ CR_SWS_MASTER | CR_SCK_MASTER);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ ssicr |= CR_DEL | CR_PDTA;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ssicr |= CR_DEL;
+ break;
+ default:
+ pr_debug("ssi: unsupported format\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
+ case SND_SOC_DAIFMT_CONT:
+ break;
+ case SND_SOC_DAIFMT_GATED:
+ ssicr |= CR_BREN;
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ ssicr |= CR_SCKP; /* sample data at low clkedge */
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ ssicr |= CR_SCKP | CR_SWSP;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ ssicr |= CR_SWSP; /* word select starts low */
+ break;
+ default:
+ pr_debug("ssi: invalid inversion\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ ssicr |= CR_SCK_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ ssicr |= CR_SWS_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ssicr |= CR_SWS_MASTER | CR_SCK_MASTER;
+ break;
+ default:
+ pr_debug("ssi: invalid master/slave configuration\n");
+ return -EINVAL;
+ }
+
+ SSIREG(SSICR) = ssicr;
+ pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr);
+
+ return 0;
+}
+
+/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in
+ * Master mode, so really this is board specific; the SSI can do any
+ * rate with the right bitclk and divider settings.
+ */
+#define SSI_RATES \
+ SNDRV_PCM_RATE_8000_192000
+
+/* the SSI can do 8-32 bit samples, with 8 possible channels */
+#define SSI_FMTS \
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE)
+
+struct snd_soc_dai sh4_ssi_dai[] = {
+{
+ .name = "SSI0",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .playback = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .capture = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .ops = {
+ .startup = ssi_startup,
+ .shutdown = ssi_shutdown,
+ .trigger = ssi_trigger,
+ .hw_params = ssi_hw_params,
+ },
+ .dai_ops = {
+ .set_sysclk = ssi_set_sysclk,
+ .set_clkdiv = ssi_set_clkdiv,
+ .set_fmt = ssi_set_fmt,
+ },
+},
+#ifdef CONFIG_CPU_SUBTYPE_SH7760
+{
+ .name = "SSI1",
+ .id = 1,
+ .type = SND_SOC_DAI_I2S,
+ .playback = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .capture = {
+ .rates = SSI_RATES,
+ .formats = SSI_FMTS,
+ .channels_min = 2,
+ .channels_max = 8,
+ },
+ .ops = {
+ .startup = ssi_startup,
+ .shutdown = ssi_shutdown,
+ .trigger = ssi_trigger,
+ .hw_params = ssi_hw_params,
+ },
+ .dai_ops = {
+ .set_sysclk = ssi_set_sysclk,
+ .set_clkdiv = ssi_set_clkdiv,
+ .set_fmt = ssi_set_fmt,
+ },
+},
+#endif
+};
+EXPORT_SYMBOL_GPL(sh4_ssi_dai);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
new file mode 100644
index 0000000..16c7453
--- /dev/null
+++ b/sound/soc/soc-core.c
@@ -0,0 +1,1891 @@
+/*
+ * soc-core.c -- ALSA SoC Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ * with code, comments and ideas from :-
+ * Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * TODO:
+ * o Add hw rules to enforce rates, etc.
+ * o More testing with other codecs/machines.
+ * o Add more codecs and platforms to ensure good API coverage.
+ * o Support TDM on PCM and I2S
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+/* debug */
+#define SOC_DEBUG 0
+#if SOC_DEBUG
+#define dbg(format, arg...) printk(format, ## arg)
+#else
+#define dbg(format, arg...)
+#endif
+
+static DEFINE_MUTEX(pcm_mutex);
+static DEFINE_MUTEX(io_mutex);
+static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
+
+/*
+ * This is a timeout to do a DAPM powerdown after a stream is closed().
+ * It can be used to eliminate pops between different playback streams, e.g.
+ * between two audio tracks.
+ */
+static int pmdown_time = 5000;
+module_param(pmdown_time, int, 0);
+MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(dwork);
+
+ /* if there was any work waiting then we run it now and
+ * wait for it's completion */
+ if (ret) {
+ schedule_delayed_work(dwork, 0);
+ flush_scheduled_work();
+ }
+ return ret;
+}
+
+#ifdef CONFIG_SND_SOC_AC97_BUS
+/* unregister ac97 codec */
+static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
+{
+ if (codec->ac97->dev.bus)
+ device_unregister(&codec->ac97->dev);
+ return 0;
+}
+
+/* stop no dev release warning */
+static void soc_ac97_device_release(struct device *dev){}
+
+/* register ac97 codec to bus */
+static int soc_ac97_dev_register(struct snd_soc_codec *codec)
+{
+ int err;
+
+ codec->ac97->dev.bus = &ac97_bus_type;
+ codec->ac97->dev.parent = NULL;
+ codec->ac97->dev.release = soc_ac97_device_release;
+
+ dev_set_name(&codec->ac97->dev, "%d-%d:%s",
+ codec->card->number, 0, codec->name);
+ err = device_register(&codec->ac97->dev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Can't register ac97 bus\n");
+ codec->ac97->dev.bus = NULL;
+ return err;
+ }
+ return 0;
+}
+#endif
+
+static inline const char *get_dai_name(int type)
+{
+ switch (type) {
+ case SND_SOC_DAI_AC97_BUS:
+ case SND_SOC_DAI_AC97:
+ return "AC97";
+ case SND_SOC_DAI_I2S:
+ return "I2S";
+ case SND_SOC_DAI_PCM:
+ return "PCM";
+ }
+ return NULL;
+}
+
+/*
+ * Called by ALSA when a PCM substream is opened, the runtime->hw record is
+ * then initialized and any private data can be allocated. This also calls
+ * startup for the cpu DAI, platform, machine and codec DAI.
+ */
+static int soc_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
+ int ret = 0;
+
+ mutex_lock(&pcm_mutex);
+
+ /* startup the audio subsystem */
+ if (cpu_dai->ops.startup) {
+ ret = cpu_dai->ops.startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open interface %s\n",
+ cpu_dai->name);
+ goto out;
+ }
+ }
+
+ if (platform->pcm_ops->open) {
+ ret = platform->pcm_ops->open(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
+ goto platform_err;
+ }
+ }
+
+ if (codec_dai->ops.startup) {
+ ret = codec_dai->ops.startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't open codec %s\n",
+ codec_dai->name);
+ goto codec_dai_err;
+ }
+ }
+
+ if (machine->ops && machine->ops->startup) {
+ ret = machine->ops->startup(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
+ goto machine_err;
+ }
+ }
+
+ /* Check that the codec and cpu DAI's are compatible */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw.rate_min =
+ max(codec_dai->playback.rate_min,
+ cpu_dai->playback.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai->playback.rate_max,
+ cpu_dai->playback.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai->playback.channels_min,
+ cpu_dai->playback.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai->playback.channels_max,
+ cpu_dai->playback.channels_max);
+ runtime->hw.formats =
+ codec_dai->playback.formats & cpu_dai->playback.formats;
+ runtime->hw.rates =
+ codec_dai->playback.rates & cpu_dai->playback.rates;
+ } else {
+ runtime->hw.rate_min =
+ max(codec_dai->capture.rate_min,
+ cpu_dai->capture.rate_min);
+ runtime->hw.rate_max =
+ min(codec_dai->capture.rate_max,
+ cpu_dai->capture.rate_max);
+ runtime->hw.channels_min =
+ max(codec_dai->capture.channels_min,
+ cpu_dai->capture.channels_min);
+ runtime->hw.channels_max =
+ min(codec_dai->capture.channels_max,
+ cpu_dai->capture.channels_max);
+ runtime->hw.formats =
+ codec_dai->capture.formats & cpu_dai->capture.formats;
+ runtime->hw.rates =
+ codec_dai->capture.rates & cpu_dai->capture.rates;
+ }
+
+ snd_pcm_limit_hw_rates(runtime);
+ if (!runtime->hw.rates) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
+ codec_dai->name, cpu_dai->name);
+ goto machine_err;
+ }
+ if (!runtime->hw.formats) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
+ codec_dai->name, cpu_dai->name);
+ goto machine_err;
+ }
+ if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
+ printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
+ codec_dai->name, cpu_dai->name);
+ goto machine_err;
+ }
+
+ dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
+ dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
+ dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+ runtime->hw.channels_max);
+ dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+ runtime->hw.rate_max);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->playback.active = codec_dai->playback.active = 1;
+ else
+ cpu_dai->capture.active = codec_dai->capture.active = 1;
+ cpu_dai->active = codec_dai->active = 1;
+ cpu_dai->runtime = runtime;
+ socdev->codec->active++;
+ mutex_unlock(&pcm_mutex);
+ return 0;
+
+machine_err:
+ if (machine->ops && machine->ops->shutdown)
+ machine->ops->shutdown(substream);
+
+codec_dai_err:
+ if (platform->pcm_ops->close)
+ platform->pcm_ops->close(substream);
+
+platform_err:
+ if (cpu_dai->ops.shutdown)
+ cpu_dai->ops.shutdown(substream);
+out:
+ mutex_unlock(&pcm_mutex);
+ return ret;
+}
+
+/*
+ * Power down the audio subsystem pmdown_time msecs after close is called.
+ * This is to ensure there are no pops or clicks in between any music tracks
+ * due to DAPM power cycling.
+ */
+static void close_delayed_work(struct work_struct *work)
+{
+ struct snd_soc_device *socdev =
+ container_of(work, struct snd_soc_device, delayed_work.work);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_dai *codec_dai;
+ int i;
+
+ mutex_lock(&pcm_mutex);
+ for (i = 0; i < codec->num_dai; i++) {
+ codec_dai = &codec->dai[i];
+
+ dbg("pop wq checking: %s status: %s waiting: %s\n",
+ codec_dai->playback.stream_name,
+ codec_dai->playback.active ? "active" : "inactive",
+ codec_dai->pop_wait ? "yes" : "no");
+
+ /* are we waiting on this codec DAI stream */
+ if (codec_dai->pop_wait == 1) {
+
+ /* Reduce power if no longer active */
+ if (codec->active == 0) {
+ dbg("pop wq D1 %s %s\n", codec->name,
+ codec_dai->playback.stream_name);
+ snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_PREPARE);
+ }
+
+ codec_dai->pop_wait = 0;
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+
+ /* Fall into standby if no longer active */
+ if (codec->active == 0) {
+ dbg("pop wq D3 %s %s\n", codec->name,
+ codec_dai->playback.stream_name);
+ snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_STANDBY);
+ }
+ }
+ }
+ mutex_unlock(&pcm_mutex);
+}
+
+/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and platform are also
+ * shutdown.
+ */
+static int soc_codec_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ mutex_lock(&pcm_mutex);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cpu_dai->playback.active = codec_dai->playback.active = 0;
+ else
+ cpu_dai->capture.active = codec_dai->capture.active = 0;
+
+ if (codec_dai->playback.active == 0 &&
+ codec_dai->capture.active == 0) {
+ cpu_dai->active = codec_dai->active = 0;
+ }
+ codec->active--;
+
+ /* Muting the DAC suppresses artifacts caused during digital
+ * shutdown, for example from stopping clocks.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ if (cpu_dai->ops.shutdown)
+ cpu_dai->ops.shutdown(substream);
+
+ if (codec_dai->ops.shutdown)
+ codec_dai->ops.shutdown(substream);
+
+ if (machine->ops && machine->ops->shutdown)
+ machine->ops->shutdown(substream);
+
+ if (platform->pcm_ops->close)
+ platform->pcm_ops->close(substream);
+ cpu_dai->runtime = NULL;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* start delayed pop wq here for playback streams */
+ codec_dai->pop_wait = 1;
+ schedule_delayed_work(&socdev->delayed_work,
+ msecs_to_jiffies(pmdown_time));
+ } else {
+ /* capture streams can be powered down now */
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+
+ if (codec->active == 0 && codec_dai->pop_wait == 0)
+ snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_STANDBY);
+ }
+
+ mutex_unlock(&pcm_mutex);
+ return 0;
+}
+
+/*
+ * Called by ALSA when the PCM substream is prepared, can set format, sample
+ * rate, etc. This function is non atomic and can be called multiple times,
+ * it can refer to the runtime info.
+ */
+static int soc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_codec *codec = socdev->codec;
+ int ret = 0;
+
+ mutex_lock(&pcm_mutex);
+
+ if (machine->ops && machine->ops->prepare) {
+ ret = machine->ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine prepare error\n");
+ goto out;
+ }
+ }
+
+ if (platform->pcm_ops->prepare) {
+ ret = platform->pcm_ops->prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform prepare error\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->ops.prepare) {
+ ret = codec_dai->ops.prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: codec DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ if (cpu_dai->ops.prepare) {
+ ret = cpu_dai->ops.prepare(substream);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: cpu DAI prepare error\n");
+ goto out;
+ }
+ }
+
+ /* we only want to start a DAPM playback stream if we are not waiting
+ * on an existing one stopping */
+ if (codec_dai->pop_wait) {
+ /* we are waiting for the delayed work to start */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ snd_soc_dapm_stream_event(socdev->codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else {
+ codec_dai->pop_wait = 0;
+ cancel_delayed_work(&socdev->delayed_work);
+ snd_soc_dai_digital_mute(codec_dai, 0);
+ }
+ } else {
+ /* no delayed work - do we need to power up codec */
+ if (codec->bias_level != SND_SOC_BIAS_ON) {
+
+ snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_PREPARE);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+
+ snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
+ snd_soc_dai_digital_mute(codec_dai, 0);
+
+ } else {
+ /* codec already powered - power on widgets */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+
+ snd_soc_dai_digital_mute(codec_dai, 0);
+ }
+ }
+
+out:
+ mutex_unlock(&pcm_mutex);
+ return ret;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
+ int ret = 0;
+
+ mutex_lock(&pcm_mutex);
+
+ if (machine->ops && machine->ops->hw_params) {
+ ret = machine->ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: machine hw_params failed\n");
+ goto out;
+ }
+ }
+
+ if (codec_dai->ops.hw_params) {
+ ret = codec_dai->ops.hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't set codec %s hw params\n",
+ codec_dai->name);
+ goto codec_err;
+ }
+ }
+
+ if (cpu_dai->ops.hw_params) {
+ ret = cpu_dai->ops.hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: interface %s hw params failed\n",
+ cpu_dai->name);
+ goto interface_err;
+ }
+ }
+
+ if (platform->pcm_ops->hw_params) {
+ ret = platform->pcm_ops->hw_params(substream, params);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform %s hw params failed\n",
+ platform->name);
+ goto platform_err;
+ }
+ }
+
+out:
+ mutex_unlock(&pcm_mutex);
+ return ret;
+
+platform_err:
+ if (cpu_dai->ops.hw_free)
+ cpu_dai->ops.hw_free(substream);
+
+interface_err:
+ if (codec_dai->ops.hw_free)
+ codec_dai->ops.hw_free(substream);
+
+codec_err:
+ if (machine->ops && machine->ops->hw_free)
+ machine->ops->hw_free(substream);
+
+ mutex_unlock(&pcm_mutex);
+ return ret;
+}
+
+/*
+ * Free's resources allocated by hw_params, can be called multiple times
+ */
+static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
+ struct snd_soc_codec *codec = socdev->codec;
+
+ mutex_lock(&pcm_mutex);
+
+ /* apply codec digital mute */
+ if (!codec->active)
+ snd_soc_dai_digital_mute(codec_dai, 1);
+
+ /* free any machine hw params */
+ if (machine->ops && machine->ops->hw_free)
+ machine->ops->hw_free(substream);
+
+ /* free any DMA resources */
+ if (platform->pcm_ops->hw_free)
+ platform->pcm_ops->hw_free(substream);
+
+ /* now free hw params for the DAI's */
+ if (codec_dai->ops.hw_free)
+ codec_dai->ops.hw_free(substream);
+
+ if (cpu_dai->ops.hw_free)
+ cpu_dai->ops.hw_free(substream);
+
+ mutex_unlock(&pcm_mutex);
+ return 0;
+}
+
+static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
+ int ret;
+
+ if (codec_dai->ops.trigger) {
+ ret = codec_dai->ops.trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (platform->pcm_ops->trigger) {
+ ret = platform->pcm_ops->trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (cpu_dai->ops.trigger) {
+ ret = cpu_dai->ops.trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+ return 0;
+}
+
+/* ASoC PCM operations */
+static struct snd_pcm_ops soc_pcm_ops = {
+ .open = soc_pcm_open,
+ .close = soc_codec_close,
+ .hw_params = soc_pcm_hw_params,
+ .hw_free = soc_pcm_hw_free,
+ .prepare = soc_pcm_prepare,
+ .trigger = soc_pcm_trigger,
+};
+
+#ifdef CONFIG_PM
+/* powers down audio subsystem for suspend */
+static int soc_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int i;
+
+ /* Due to the resume being scheduled into a workqueue we could
+ * suspend before that's finished - wait for it to complete.
+ */
+ snd_power_lock(codec->card);
+ snd_power_wait(codec->card, SNDRV_CTL_POWER_D0);
+ snd_power_unlock(codec->card);
+
+ /* we're going to block userspace touching us until resume completes */
+ snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);
+
+ /* mute any active DAC's */
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
+ if (dai->dai_ops.digital_mute && dai->playback.active)
+ dai->dai_ops.digital_mute(dai, 1);
+ }
+
+ /* suspend all pcms */
+ for (i = 0; i < machine->num_links; i++)
+ snd_pcm_suspend_all(machine->dai_link[i].pcm);
+
+ if (machine->suspend_pre)
+ machine->suspend_pre(pdev, state);
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
+ cpu_dai->suspend(pdev, cpu_dai);
+ if (platform->suspend)
+ platform->suspend(pdev, cpu_dai);
+ }
+
+ /* close any waiting streams and save state */
+ run_delayed_work(&socdev->delayed_work);
+ codec->suspend_bias_level = codec->bias_level;
+
+ for (i = 0; i < codec->num_dai; i++) {
+ char *stream = codec->dai[i].playback.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_SUSPEND);
+ stream = codec->dai[i].capture.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_SUSPEND);
+ }
+
+ if (codec_dev->suspend)
+ codec_dev->suspend(pdev, state);
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
+ cpu_dai->suspend(pdev, cpu_dai);
+ }
+
+ if (machine->suspend_post)
+ machine->suspend_post(pdev, state);
+
+ return 0;
+}
+
+/* deferred resume work, so resume can complete before we finished
+ * setting our codec back up, which can be very slow on I2C
+ */
+static void soc_resume_deferred(struct work_struct *work)
+{
+ struct snd_soc_device *socdev = container_of(work,
+ struct snd_soc_device,
+ deferred_resume_work);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+ struct snd_soc_codec *codec = socdev->codec;
+ struct platform_device *pdev = to_platform_device(socdev->dev);
+ int i;
+
+ /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
+ * so userspace apps are blocked from touching us
+ */
+
+ dev_info(socdev->dev, "starting resume work\n");
+
+ if (machine->resume_pre)
+ machine->resume_pre(pdev);
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
+ cpu_dai->resume(pdev, cpu_dai);
+ }
+
+ if (codec_dev->resume)
+ codec_dev->resume(pdev);
+
+ for (i = 0; i < codec->num_dai; i++) {
+ char *stream = codec->dai[i].playback.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_RESUME);
+ stream = codec->dai[i].capture.stream_name;
+ if (stream != NULL)
+ snd_soc_dapm_stream_event(codec, stream,
+ SND_SOC_DAPM_STREAM_RESUME);
+ }
+
+ /* unmute any active DACs */
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
+ if (dai->dai_ops.digital_mute && dai->playback.active)
+ dai->dai_ops.digital_mute(dai, 0);
+ }
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
+ cpu_dai->resume(pdev, cpu_dai);
+ if (platform->resume)
+ platform->resume(pdev, cpu_dai);
+ }
+
+ if (machine->resume_post)
+ machine->resume_post(pdev);
+
+ dev_info(socdev->dev, "resume work completed\n");
+
+ /* userspace can access us now we are back as we were before */
+ snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0);
+}
+
+/* powers up audio subsystem after a suspend */
+static int soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ dev_info(socdev->dev, "scheduling resume work\n");
+
+ if (!schedule_work(&socdev->deferred_resume_work))
+ dev_err(socdev->dev, "work item may be lost\n");
+
+ return 0;
+}
+
+#else
+#define soc_suspend NULL
+#define soc_resume NULL
+#endif
+
+/* probes a new socdev */
+static int soc_probe(struct platform_device *pdev)
+{
+ int ret = 0, i;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+
+ if (machine->probe) {
+ ret = machine->probe(pdev);
+ if (ret < 0)
+ return ret;
+ }
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->probe) {
+ ret = cpu_dai->probe(pdev, cpu_dai);
+ if (ret < 0)
+ goto cpu_dai_err;
+ }
+ }
+
+ if (codec_dev->probe) {
+ ret = codec_dev->probe(pdev);
+ if (ret < 0)
+ goto cpu_dai_err;
+ }
+
+ if (platform->probe) {
+ ret = platform->probe(pdev);
+ if (ret < 0)
+ goto platform_err;
+ }
+
+ /* DAPM stream work */
+ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
+#ifdef CONFIG_PM
+ /* deferred resume work */
+ INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred);
+#endif
+
+ return 0;
+
+platform_err:
+ if (codec_dev->remove)
+ codec_dev->remove(pdev);
+
+cpu_dai_err:
+ for (i--; i >= 0; i--) {
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->remove)
+ cpu_dai->remove(pdev, cpu_dai);
+ }
+
+ if (machine->remove)
+ machine->remove(pdev);
+
+ return ret;
+}
+
+/* removes a socdev */
+static int soc_remove(struct platform_device *pdev)
+{
+ int i;
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_platform *platform = socdev->platform;
+ struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+
+ run_delayed_work(&socdev->delayed_work);
+
+ if (platform->remove)
+ platform->remove(pdev);
+
+ if (codec_dev->remove)
+ codec_dev->remove(pdev);
+
+ for (i = 0; i < machine->num_links; i++) {
+ struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+ if (cpu_dai->remove)
+ cpu_dai->remove(pdev, cpu_dai);
+ }
+
+ if (machine->remove)
+ machine->remove(pdev);
+
+ return 0;
+}
+
+/* ASoC platform driver */
+static struct platform_driver soc_driver = {
+ .driver = {
+ .name = "soc-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = soc_probe,
+ .remove = soc_remove,
+ .suspend = soc_suspend,
+ .resume = soc_resume,
+};
+
+/* create a new pcm */
+static int soc_new_pcm(struct snd_soc_device *socdev,
+ struct snd_soc_dai_link *dai_link, int num)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_dai *codec_dai = dai_link->codec_dai;
+ struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
+ struct snd_soc_pcm_runtime *rtd;
+ struct snd_pcm *pcm;
+ char new_name[64];
+ int ret = 0, playback = 0, capture = 0;
+
+ rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
+ if (rtd == NULL)
+ return -ENOMEM;
+
+ rtd->dai = dai_link;
+ rtd->socdev = socdev;
+ codec_dai->codec = socdev->codec;
+
+ /* check client and interface hw capabilities */
+ sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
+ get_dai_name(cpu_dai->type), num);
+
+ if (codec_dai->playback.channels_min)
+ playback = 1;
+ if (codec_dai->capture.channels_min)
+ capture = 1;
+
+ ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
+ capture, &pcm);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
+ codec->name);
+ kfree(rtd);
+ return ret;
+ }
+
+ dai_link->pcm = pcm;
+ pcm->private_data = rtd;
+ soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
+ soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
+ soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
+ soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
+ soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
+ soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
+ soc_pcm_ops.page = socdev->platform->pcm_ops->page;
+
+ if (playback)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
+
+ if (capture)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
+
+ ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: platform pcm constructor failed\n");
+ kfree(rtd);
+ return ret;
+ }
+
+ pcm->private_free = socdev->platform->pcm_free;
+ printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+ cpu_dai->name);
+ return ret;
+}
+
+/* codec register dump */
+static ssize_t codec_reg_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct snd_soc_device *devdata = dev_get_drvdata(dev);
+ struct snd_soc_codec *codec = devdata->codec;
+ int i, step = 1, count = 0;
+
+ if (!codec->reg_cache_size)
+ return 0;
+
+ if (codec->reg_cache_step)
+ step = codec->reg_cache_step;
+
+ count += sprintf(buf, "%s registers\n", codec->name);
+ for (i = 0; i < codec->reg_cache_size; i += step) {
+ count += sprintf(buf + count, "%2x: ", i);
+ if (count >= PAGE_SIZE - 1)
+ break;
+
+ if (codec->display_register)
+ count += codec->display_register(codec, buf + count,
+ PAGE_SIZE - count, i);
+ else
+ count += snprintf(buf + count, PAGE_SIZE - count,
+ "%4x", codec->read(codec, i));
+
+ if (count >= PAGE_SIZE - 1)
+ break;
+
+ count += snprintf(buf + count, PAGE_SIZE - count, "\n");
+ if (count >= PAGE_SIZE - 1)
+ break;
+ }
+
+ /* Truncate count; min() would cause a warning */
+ if (count >= PAGE_SIZE)
+ count = PAGE_SIZE - 1;
+
+ return count;
+}
+static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
+
+/**
+ * snd_soc_new_ac97_codec - initailise AC97 device
+ * @codec: audio codec
+ * @ops: AC97 bus operations
+ * @num: AC97 codec number
+ *
+ * Initialises AC97 codec resources for use by ad-hoc devices only.
+ */
+int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
+ struct snd_ac97_bus_ops *ops, int num)
+{
+ mutex_lock(&codec->mutex);
+
+ codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
+ if (codec->ac97 == NULL) {
+ mutex_unlock(&codec->mutex);
+ return -ENOMEM;
+ }
+
+ codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
+ if (codec->ac97->bus == NULL) {
+ kfree(codec->ac97);
+ codec->ac97 = NULL;
+ mutex_unlock(&codec->mutex);
+ return -ENOMEM;
+ }
+
+ codec->ac97->bus->ops = ops;
+ codec->ac97->num = num;
+ mutex_unlock(&codec->mutex);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
+
+/**
+ * snd_soc_free_ac97_codec - free AC97 codec device
+ * @codec: audio codec
+ *
+ * Frees AC97 codec device resources.
+ */
+void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
+{
+ mutex_lock(&codec->mutex);
+ kfree(codec->ac97->bus);
+ kfree(codec->ac97);
+ codec->ac97 = NULL;
+ mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
+
+/**
+ * snd_soc_update_bits - update codec register bits
+ * @codec: audio codec
+ * @reg: codec register
+ * @mask: register mask
+ * @value: new value
+ *
+ * Writes new register value.
+ *
+ * Returns 1 for change else 0.
+ */
+int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
+ unsigned short mask, unsigned short value)
+{
+ int change;
+ unsigned short old, new;
+
+ mutex_lock(&io_mutex);
+ old = snd_soc_read(codec, reg);
+ new = (old & ~mask) | value;
+ change = old != new;
+ if (change)
+ snd_soc_write(codec, reg, new);
+
+ mutex_unlock(&io_mutex);
+ return change;
+}
+EXPORT_SYMBOL_GPL(snd_soc_update_bits);
+
+/**
+ * snd_soc_test_bits - test register for change
+ * @codec: audio codec
+ * @reg: codec register
+ * @mask: register mask
+ * @value: new value
+ *
+ * Tests a register with a new value and checks if the new value is
+ * different from the old value.
+ *
+ * Returns 1 for change else 0.
+ */
+int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
+ unsigned short mask, unsigned short value)
+{
+ int change;
+ unsigned short old, new;
+
+ mutex_lock(&io_mutex);
+ old = snd_soc_read(codec, reg);
+ new = (old & ~mask) | value;
+ change = old != new;
+ mutex_unlock(&io_mutex);
+
+ return change;
+}
+EXPORT_SYMBOL_GPL(snd_soc_test_bits);
+
+/**
+ * snd_soc_new_pcms - create new sound card and pcms
+ * @socdev: the SoC audio device
+ *
+ * Create a new sound card based upon the codec and interface pcms.
+ *
+ * Returns 0 for success, else error.
+ */
+int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_machine *machine = socdev->machine;
+ int ret = 0, i;
+
+ mutex_lock(&codec->mutex);
+
+ /* register a sound card */
+ codec->card = snd_card_new(idx, xid, codec->owner, 0);
+ if (!codec->card) {
+ printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
+ codec->name);
+ mutex_unlock(&codec->mutex);
+ return -ENODEV;
+ }
+
+ codec->card->dev = socdev->dev;
+ codec->card->private_data = codec;
+ strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
+
+ /* create the pcms */
+ for (i = 0; i < machine->num_links; i++) {
+ ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: can't create pcm %s\n",
+ machine->dai_link[i].stream_name);
+ mutex_unlock(&codec->mutex);
+ return ret;
+ }
+ }
+
+ mutex_unlock(&codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
+
+/**
+ * snd_soc_register_card - register sound card
+ * @socdev: the SoC audio device
+ *
+ * Register a SoC sound card. Also registers an AC97 device if the
+ * codec is AC97 for ad hoc devices.
+ *
+ * Returns 0 for success, else error.
+ */
+int snd_soc_register_card(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_machine *machine = socdev->machine;
+ int ret = 0, i, ac97 = 0, err = 0;
+
+ for (i = 0; i < machine->num_links; i++) {
+ if (socdev->machine->dai_link[i].init) {
+ err = socdev->machine->dai_link[i].init(codec);
+ if (err < 0) {
+ printk(KERN_ERR "asoc: failed to init %s\n",
+ socdev->machine->dai_link[i].stream_name);
+ continue;
+ }
+ }
+ if (socdev->machine->dai_link[i].codec_dai->type ==
+ SND_SOC_DAI_AC97_BUS)
+ ac97 = 1;
+ }
+ snprintf(codec->card->shortname, sizeof(codec->card->shortname),
+ "%s", machine->name);
+ snprintf(codec->card->longname, sizeof(codec->card->longname),
+ "%s (%s)", machine->name, codec->name);
+
+ ret = snd_card_register(codec->card);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
+ codec->name);
+ goto out;
+ }
+
+ mutex_lock(&codec->mutex);
+#ifdef CONFIG_SND_SOC_AC97_BUS
+ if (ac97) {
+ ret = soc_ac97_dev_register(codec);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: AC97 device register failed\n");
+ snd_card_free(codec->card);
+ mutex_unlock(&codec->mutex);
+ goto out;
+ }
+ }
+#endif
+
+ err = snd_soc_dapm_sys_add(socdev->dev);
+ if (err < 0)
+ printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
+
+ err = device_create_file(socdev->dev, &dev_attr_codec_reg);
+ if (err < 0)
+ printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
+
+ mutex_unlock(&codec->mutex);
+
+out:
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
+
+/**
+ * snd_soc_free_pcms - free sound card and pcms
+ * @socdev: the SoC audio device
+ *
+ * Frees sound card and pcms associated with the socdev.
+ * Also unregister the codec if it is an AC97 device.
+ */
+void snd_soc_free_pcms(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+#ifdef CONFIG_SND_SOC_AC97_BUS
+ struct snd_soc_dai *codec_dai;
+ int i;
+#endif
+
+ mutex_lock(&codec->mutex);
+#ifdef CONFIG_SND_SOC_AC97_BUS
+ for (i = 0; i < codec->num_dai; i++) {
+ codec_dai = &codec->dai[i];
+ if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
+ soc_ac97_dev_unregister(codec);
+ goto free_card;
+ }
+ }
+free_card:
+#endif
+
+ if (codec->card)
+ snd_card_free(codec->card);
+ device_remove_file(socdev->dev, &dev_attr_codec_reg);
+ mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_free_pcms);
+
+/**
+ * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
+ * @substream: the pcm substream
+ * @hw: the hardware parameters
+ *
+ * Sets the substream runtime hardware parameters.
+ */
+int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
+ const struct snd_pcm_hardware *hw)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ runtime->hw.info = hw->info;
+ runtime->hw.formats = hw->formats;
+ runtime->hw.period_bytes_min = hw->period_bytes_min;
+ runtime->hw.period_bytes_max = hw->period_bytes_max;
+ runtime->hw.periods_min = hw->periods_min;
+ runtime->hw.periods_max = hw->periods_max;
+ runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
+ runtime->hw.fifo_size = hw->fifo_size;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
+
+/**
+ * snd_soc_cnew - create new control
+ * @_template: control template
+ * @data: control private data
+ * @lnng_name: control long name
+ *
+ * Create a new mixer control from a template control.
+ *
+ * Returns 0 for success, else error.
+ */
+struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
+ void *data, char *long_name)
+{
+ struct snd_kcontrol_new template;
+
+ memcpy(&template, _template, sizeof(template));
+ if (long_name)
+ template.name = long_name;
+ template.index = 0;
+
+ return snd_ctl_new1(&template, data);
+}
+EXPORT_SYMBOL_GPL(snd_soc_cnew);
+
+/**
+ * snd_soc_info_enum_double - enumerated double mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a double enumerated
+ * mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
+ uinfo->value.enumerated.items = e->max;
+
+ if (uinfo->value.enumerated.item > e->max - 1)
+ uinfo->value.enumerated.item = e->max - 1;
+ strcpy(uinfo->value.enumerated.name,
+ e->texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
+
+/**
+ * snd_soc_get_enum_double - enumerated double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a double enumerated mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val, bitmask;
+
+ for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+ ;
+ val = snd_soc_read(codec, e->reg);
+ ucontrol->value.enumerated.item[0]
+ = (val >> e->shift_l) & (bitmask - 1);
+ if (e->shift_l != e->shift_r)
+ ucontrol->value.enumerated.item[1] =
+ (val >> e->shift_r) & (bitmask - 1);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);
+
+/**
+ * snd_soc_put_enum_double - enumerated double mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a double enumerated mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val;
+ unsigned short mask, bitmask;
+
+ for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+ ;
+ if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ return -EINVAL;
+ val = ucontrol->value.enumerated.item[0] << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
+ if (e->shift_l != e->shift_r) {
+ if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ return -EINVAL;
+ val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
+ }
+
+ return snd_soc_update_bits(codec, e->reg, mask, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
+
+/**
+ * snd_soc_info_enum_ext - external enumerated single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about an external enumerated
+ * single mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = e->max;
+
+ if (uinfo->value.enumerated.item > e->max - 1)
+ uinfo->value.enumerated.item = e->max - 1;
+ strcpy(uinfo->value.enumerated.name,
+ e->texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
+
+/**
+ * snd_soc_info_volsw_ext - external single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single external mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int max = kcontrol->private_value;
+
+ if (max == 1)
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = max;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
+
+/**
+ * snd_soc_info_volsw - single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int max = mc->max;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+
+ if (max == 1)
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+
+ uinfo->count = shift == rshift ? 1 : 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = max;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
+
+/**
+ * snd_soc_get_volsw - single mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg) >> rshift) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ max - ucontrol->value.integer.value[0];
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ max - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
+
+/**
+ * snd_soc_put_volsw - single mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+ unsigned short val, val2, val_mask;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = max - val;
+ val_mask = mask << shift;
+ val = val << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1] & mask);
+ if (invert)
+ val2 = max - val2;
+ val_mask |= mask << rshift;
+ val |= val2 << rshift;
+ }
+ return snd_soc_update_bits(codec, reg, val_mask, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
+
+/**
+ * snd_soc_info_volsw_2r - double mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a double mixer control that
+ * spans 2 codec registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int max = mc->max;
+
+ if (max == 1)
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = max;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
+
+/**
+ * snd_soc_get_volsw_2r - double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a double mixer control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ unsigned int mask = (1<<fls(max))-1;
+ unsigned int invert = mc->invert;
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(codec, reg) >> shift) & mask;
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(codec, reg2) >> shift) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ max - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] =
+ max - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);
+
+/**
+ * snd_soc_put_volsw_2r - double mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a double mixer control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ unsigned int reg2 = mc->rreg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+ int err;
+ unsigned short val, val2, val_mask;
+
+ val_mask = mask << shift;
+ val = (ucontrol->value.integer.value[0] & mask);
+ val2 = (ucontrol->value.integer.value[1] & mask);
+
+ if (invert) {
+ val = max - val;
+ val2 = max - val2;
+ }
+
+ val = val << shift;
+ val2 = val2 << shift;
+
+ err = snd_soc_update_bits(codec, reg, val_mask, val);
+ if (err < 0)
+ return err;
+
+ err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+ return err;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
+
+/**
+ * snd_soc_info_volsw_s8 - signed mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a signed mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int max = mc->max;
+ int min = mc->min;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = max-min;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);
+
+/**
+ * snd_soc_get_volsw_s8 - signed mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a signed mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ int min = mc->min;
+ int val = snd_soc_read(codec, reg);
+
+ ucontrol->value.integer.value[0] =
+ ((signed char)(val & 0xff))-min;
+ ucontrol->value.integer.value[1] =
+ ((signed char)((val >> 8) & 0xff))-min;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8);
+
+/**
+ * snd_soc_put_volsw_sgn - signed mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a signed mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = mc->reg;
+ int min = mc->min;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0]+min) & 0xff;
+ val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
+
+ return snd_soc_update_bits(codec, reg, 0xffff, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
+
+/**
+ * snd_soc_dai_set_sysclk - configure DAI system or master clock.
+ * @dai: DAI
+ * @clk_id: DAI specific clock ID
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the DAI master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ if (dai->dai_ops.set_sysclk)
+ return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
+
+/**
+ * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
+ * @dai: DAI
+ * @clk_id: DAI specific clock divider ID
+ * @div: new clock divisor.
+ *
+ * Configures the clock dividers. This is used to derive the best DAI bit and
+ * frame clocks from the system or master clock. It's best to set the DAI bit
+ * and frame clocks as low as possible to save system power.
+ */
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+ int div_id, int div)
+{
+ if (dai->dai_ops.set_clkdiv)
+ return dai->dai_ops.set_clkdiv(dai, div_id, div);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
+
+/**
+ * snd_soc_dai_set_pll - configure DAI PLL.
+ * @dai: DAI
+ * @pll_id: DAI specific PLL ID
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ if (dai->dai_ops.set_pll)
+ return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
+
+/**
+ * snd_soc_dai_set_fmt - configure DAI hardware audio format.
+ * @dai: DAI
+ * @clk_id: DAI specific clock ID
+ * @fmt: SND_SOC_DAIFMT_ format value.
+ *
+ * Configures the DAI hardware format and clocking.
+ */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ if (dai->dai_ops.set_fmt)
+ return dai->dai_ops.set_fmt(dai, fmt);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
+
+/**
+ * snd_soc_dai_set_tdm_slot - configure DAI TDM.
+ * @dai: DAI
+ * @mask: DAI specific mask representing used slots.
+ * @slots: Number of slots in use.
+ *
+ * Configures a DAI for TDM operation. Both mask and slots are codec and DAI
+ * specific.
+ */
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int mask, int slots)
+{
+ if (dai->dai_ops.set_sysclk)
+ return dai->dai_ops.set_tdm_slot(dai, mask, slots);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
+
+/**
+ * snd_soc_dai_set_tristate - configure DAI system or master clock.
+ * @dai: DAI
+ * @tristate: tristate enable
+ *
+ * Tristates the DAI so that others can use it.
+ */
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ if (dai->dai_ops.set_sysclk)
+ return dai->dai_ops.set_tristate(dai, tristate);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
+
+/**
+ * snd_soc_dai_digital_mute - configure DAI system or master clock.
+ * @dai: DAI
+ * @mute: mute enable
+ *
+ * Mutes the DAI DAC.
+ */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ if (dai->dai_ops.digital_mute)
+ return dai->dai_ops.digital_mute(dai, mute);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
+
+static int __devinit snd_soc_init(void)
+{
+ printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
+ return platform_driver_register(&soc_driver);
+}
+
+static void snd_soc_exit(void)
+{
+ platform_driver_unregister(&soc_driver);
+}
+
+module_init(snd_soc_init);
+module_exit(snd_soc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_DESCRIPTION("ALSA SoC Core");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:soc-audio");
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
new file mode 100644
index 0000000..7351db9
--- /dev/null
+++ b/sound/soc/soc-dapm.c
@@ -0,0 +1,1545 @@
+/*
+ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Features:
+ * o Changes power status of internal codec blocks depending on the
+ * dynamic configuration of codec internal audio paths and active
+ * DAC's/ADC's.
+ * o Platform power domain - can support external components i.e. amps and
+ * mic/meadphone insertion events.
+ * o Automatic Mic Bias support
+ * o Jack insertion power event initiation - e.g. hp insertion will enable
+ * sinks, dacs, etc
+ * o Delayed powerdown of audio susbsystem to reduce pops between a quick
+ * device reopen.
+ *
+ * Todo:
+ * o DAPM power change sequencing - allow for configurable per
+ * codec sequences.
+ * o Support for analogue bias optimisation.
+ * o Support for reduced codec oversampling rates.
+ * o Support for reduced codec bias currents.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/jiffies.h>
+#include <linux/debugfs.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+/* debug */
+#ifdef DEBUG
+#define dump_dapm(codec, action) dbg_dump_dapm(codec, action)
+#else
+#define dump_dapm(codec, action)
+#endif
+
+/* dapm power sequences - make this per codec in the future */
+static int dapm_up_seq[] = {
+ snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
+ snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga,
+ snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
+};
+static int dapm_down_seq[] = {
+ snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
+ snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
+ snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post
+};
+
+static int dapm_status = 1;
+module_param(dapm_status, int, 0);
+MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
+
+static struct dentry *asoc_debugfs;
+
+static u32 pop_time;
+
+static void pop_wait(void)
+{
+ if (pop_time)
+ schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time));
+}
+
+static void pop_dbg(const char *fmt, ...)
+{
+ va_list args;
+
+ va_start(args, fmt);
+
+ if (pop_time) {
+ vprintk(fmt, args);
+ pop_wait();
+ }
+
+ va_end(args);
+}
+
+/* create a new dapm widget */
+static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
+ const struct snd_soc_dapm_widget *_widget)
+{
+ return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
+}
+
+/* set up initial codec paths */
+static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
+ struct snd_soc_dapm_path *p, int i)
+{
+ switch (w->id) {
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer: {
+ int val;
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)
+ w->kcontrols[i].private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+
+ val = snd_soc_read(w->codec, reg);
+ val = (val >> shift) & mask;
+
+ if ((invert && !val) || (!invert && val))
+ p->connect = 1;
+ else
+ p->connect = 0;
+ }
+ break;
+ case snd_soc_dapm_mux: {
+ struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value;
+ int val, item, bitmask;
+
+ for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+ ;
+ val = snd_soc_read(w->codec, e->reg);
+ item = (val >> e->shift_l) & (bitmask - 1);
+
+ p->connect = 0;
+ for (i = 0; i < e->max; i++) {
+ if (!(strcmp(p->name, e->texts[i])) && item == i)
+ p->connect = 1;
+ }
+ }
+ break;
+ /* does not effect routing - always connected */
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_vmid:
+ p->connect = 1;
+ break;
+ /* does effect routing - dynamically connected */
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ p->connect = 0;
+ break;
+ }
+}
+
+/* connect mux widget to it's interconnecting audio paths */
+static int dapm_connect_mux(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
+ struct snd_soc_dapm_path *path, const char *control_name,
+ const struct snd_kcontrol_new *kcontrol)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ int i;
+
+ for (i = 0; i < e->max; i++) {
+ if (!(strcmp(control_name, e->texts[i]))) {
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &dest->sources);
+ list_add(&path->list_source, &src->sinks);
+ path->name = (char*)e->texts[i];
+ dapm_set_path_status(dest, path, 0);
+ return 0;
+ }
+ }
+
+ return -ENODEV;
+}
+
+/* connect mixer widget to it's interconnecting audio paths */
+static int dapm_connect_mixer(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
+ struct snd_soc_dapm_path *path, const char *control_name)
+{
+ int i;
+
+ /* search for mixer kcontrol */
+ for (i = 0; i < dest->num_kcontrols; i++) {
+ if (!strcmp(control_name, dest->kcontrols[i].name)) {
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &dest->sources);
+ list_add(&path->list_source, &src->sinks);
+ path->name = dest->kcontrols[i].name;
+ dapm_set_path_status(dest, path, i);
+ return 0;
+ }
+ }
+ return -ENODEV;
+}
+
+/* update dapm codec register bits */
+static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
+{
+ int change, power;
+ unsigned short old, new;
+ struct snd_soc_codec *codec = widget->codec;
+
+ /* check for valid widgets */
+ if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
+ widget->id == snd_soc_dapm_output ||
+ widget->id == snd_soc_dapm_hp ||
+ widget->id == snd_soc_dapm_mic ||
+ widget->id == snd_soc_dapm_line ||
+ widget->id == snd_soc_dapm_spk)
+ return 0;
+
+ power = widget->power;
+ if (widget->invert)
+ power = (power ? 0:1);
+
+ old = snd_soc_read(codec, widget->reg);
+ new = (old & ~(0x1 << widget->shift)) | (power << widget->shift);
+
+ change = old != new;
+ if (change) {
+ pop_dbg("pop test %s : %s in %d ms\n", widget->name,
+ widget->power ? "on" : "off", pop_time);
+ snd_soc_write(codec, widget->reg, new);
+ pop_wait();
+ }
+ pr_debug("reg %x old %x new %x change %d\n", widget->reg,
+ old, new, change);
+ return change;
+}
+
+/* ramps the volume up or down to minimise pops before or after a
+ * DAPM power event */
+static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power)
+{
+ const struct snd_kcontrol_new *k = widget->kcontrols;
+
+ if (widget->muted && !power)
+ return 0;
+ if (!widget->muted && power)
+ return 0;
+
+ if (widget->num_kcontrols && k) {
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)k->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+
+ if (power) {
+ int i;
+ /* power up has happended, increase volume to last level */
+ if (invert) {
+ for (i = max; i > widget->saved_value; i--)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ } else {
+ for (i = 0; i < widget->saved_value; i++)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ }
+ widget->muted = 0;
+ } else {
+ /* power down is about to occur, decrease volume to mute */
+ int val = snd_soc_read(widget->codec, reg);
+ int i = widget->saved_value = (val >> shift) & mask;
+ if (invert) {
+ for (; i < mask; i++)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ } else {
+ for (; i > 0; i--)
+ snd_soc_update_bits(widget->codec, reg, mask, i);
+ }
+ widget->muted = 1;
+ }
+ }
+ return 0;
+}
+
+/* create new dapm mixer control */
+static int dapm_new_mixer(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *w)
+{
+ int i, ret = 0;
+ char name[32];
+ struct snd_soc_dapm_path *path;
+
+ /* add kcontrol */
+ for (i = 0; i < w->num_kcontrols; i++) {
+
+ /* match name */
+ list_for_each_entry(path, &w->sources, list_sink) {
+
+ /* mixer/mux paths name must match control name */
+ if (path->name != (char*)w->kcontrols[i].name)
+ continue;
+
+ /* add dapm control with long name */
+ snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name);
+ path->long_name = kstrdup (name, GFP_KERNEL);
+ if (path->long_name == NULL)
+ return -ENOMEM;
+
+ path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
+ path->long_name);
+ ret = snd_ctl_add(codec->card, path->kcontrol);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
+ path->long_name);
+ kfree(path->long_name);
+ path->long_name = NULL;
+ return ret;
+ }
+ }
+ }
+ return ret;
+}
+
+/* create new dapm mux control */
+static int dapm_new_mux(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_path *path = NULL;
+ struct snd_kcontrol *kcontrol;
+ int ret = 0;
+
+ if (!w->num_kcontrols) {
+ printk(KERN_ERR "asoc: mux %s has no controls\n", w->name);
+ return -EINVAL;
+ }
+
+ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+ ret = snd_ctl_add(codec->card, kcontrol);
+ if (ret < 0)
+ goto err;
+
+ list_for_each_entry(path, &w->sources, list_sink)
+ path->kcontrol = kcontrol;
+
+ return ret;
+
+err:
+ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
+ return ret;
+}
+
+/* create new dapm volume control */
+static int dapm_new_pga(struct snd_soc_codec *codec,
+ struct snd_soc_dapm_widget *w)
+{
+ struct snd_kcontrol *kcontrol;
+ int ret = 0;
+
+ if (!w->num_kcontrols)
+ return -EINVAL;
+
+ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+ ret = snd_ctl_add(codec->card, kcontrol);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
+ return ret;
+ }
+
+ return ret;
+}
+
+/* reset 'walked' bit for each dapm path */
+static inline void dapm_clear_walk(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_path *p;
+
+ list_for_each_entry(p, &codec->dapm_paths, list)
+ p->walked = 0;
+}
+
+/*
+ * Recursively check for a completed path to an active or physically connected
+ * output widget. Returns number of complete paths.
+ */
+static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_dapm_path *path;
+ int con = 0;
+
+ if (widget->id == snd_soc_dapm_adc && widget->active)
+ return 1;
+
+ if (widget->connected) {
+ /* connected pin ? */
+ if (widget->id == snd_soc_dapm_output && !widget->ext)
+ return 1;
+
+ /* connected jack or spk ? */
+ if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
+ widget->id == snd_soc_dapm_line)
+ return 1;
+ }
+
+ list_for_each_entry(path, &widget->sinks, list_source) {
+ if (path->walked)
+ continue;
+
+ if (path->sink && path->connect) {
+ path->walked = 1;
+ con += is_connected_output_ep(path->sink);
+ }
+ }
+
+ return con;
+}
+
+/*
+ * Recursively check for a completed path to an active or physically connected
+ * input widget. Returns number of complete paths.
+ */
+static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_dapm_path *path;
+ int con = 0;
+
+ /* active stream ? */
+ if (widget->id == snd_soc_dapm_dac && widget->active)
+ return 1;
+
+ if (widget->connected) {
+ /* connected pin ? */
+ if (widget->id == snd_soc_dapm_input && !widget->ext)
+ return 1;
+
+ /* connected VMID/Bias for lower pops */
+ if (widget->id == snd_soc_dapm_vmid)
+ return 1;
+
+ /* connected jack ? */
+ if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+ return 1;
+ }
+
+ list_for_each_entry(path, &widget->sources, list_sink) {
+ if (path->walked)
+ continue;
+
+ if (path->source && path->connect) {
+ path->walked = 1;
+ con += is_connected_input_ep(path->source);
+ }
+ }
+
+ return con;
+}
+
+/*
+ * Handler for generic register modifier widget.
+ */
+int dapm_reg_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ unsigned int val;
+
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ val = w->on_val;
+ else
+ val = w->off_val;
+
+ snd_soc_update_bits(w->codec, -(w->reg + 1),
+ w->mask << w->shift, val << w->shift);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(dapm_reg_event);
+
+/*
+ * Scan each dapm widget for complete audio path.
+ * A complete path is a route that has valid endpoints i.e.:-
+ *
+ * o DAC to output pin.
+ * o Input Pin to ADC.
+ * o Input pin to Output pin (bypass, sidetone)
+ * o DAC to ADC (loopback).
+ */
+static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
+{
+ struct snd_soc_dapm_widget *w;
+ int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
+
+ /* do we have a sequenced stream event */
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ c = ARRAY_SIZE(dapm_up_seq);
+ seq = dapm_up_seq;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ c = ARRAY_SIZE(dapm_down_seq);
+ seq = dapm_down_seq;
+ }
+
+ for(i = 0; i < c; i++) {
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ /* is widget in stream order */
+ if (seq && seq[i] && w->id != seq[i])
+ continue;
+
+ /* vmid - no action */
+ if (w->id == snd_soc_dapm_vmid)
+ continue;
+
+ /* active ADC */
+ if (w->id == snd_soc_dapm_adc && w->active) {
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ w->power = (in != 0) ? 1 : 0;
+ dapm_update_bits(w);
+ continue;
+ }
+
+ /* active DAC */
+ if (w->id == snd_soc_dapm_dac && w->active) {
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ w->power = (out != 0) ? 1 : 0;
+ dapm_update_bits(w);
+ continue;
+ }
+
+ /* pre and post event widgets */
+ if (w->id == snd_soc_dapm_pre) {
+ if (!w->event)
+ continue;
+
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ continue;
+ }
+ if (w->id == snd_soc_dapm_post) {
+ if (!w->event)
+ continue;
+
+ if (event == SND_SOC_DAPM_STREAM_START) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ continue;
+ }
+
+ /* all other widgets */
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ power = (out != 0 && in != 0) ? 1 : 0;
+ power_change = (w->power == power) ? 0: 1;
+ w->power = power;
+
+ if (!power_change)
+ continue;
+
+ /* call any power change event handlers */
+ if (w->event)
+ pr_debug("power %s event for %s flags %x\n",
+ w->power ? "on" : "off",
+ w->name, w->event_flags);
+
+ /* power up pre event */
+ if (power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down pre event */
+ if (!power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* Lower PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && !power)
+ dapm_set_pga(w, power);
+
+ dapm_update_bits(w);
+
+ /* Raise PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && power)
+ dapm_set_pga(w, power);
+
+ /* power up post event */
+ if (power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down post event */
+ if (!power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
+ }
+ }
+ }
+
+ return ret;
+}
+
+#ifdef DEBUG
+static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
+{
+ struct snd_soc_dapm_widget *w;
+ struct snd_soc_dapm_path *p = NULL;
+ int in, out;
+
+ printk("DAPM %s %s\n", codec->name, action);
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ /* only display widgets that effect routing */
+ switch (w->id) {
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ case snd_soc_dapm_vmid:
+ continue;
+ case snd_soc_dapm_mux:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_mixer:
+ if (w->name) {
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ printk("%s: %s in %d out %d\n", w->name,
+ w->power ? "On":"Off",in, out);
+
+ list_for_each_entry(p, &w->sources, list_sink) {
+ if (p->connect)
+ printk(" in %s %s\n", p->name ? p->name : "static",
+ p->source->name);
+ }
+ list_for_each_entry(p, &w->sinks, list_source) {
+ if (p->connect)
+ printk(" out %s %s\n", p->name ? p->name : "static",
+ p->sink->name);
+ }
+ }
+ break;
+ }
+ }
+}
+#endif
+
+/* test and update the power status of a mux widget */
+static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kcontrol, int mask,
+ int mux, int val, struct soc_enum *e)
+{
+ struct snd_soc_dapm_path *path;
+ int found = 0;
+
+ if (widget->id != snd_soc_dapm_mux)
+ return -ENODEV;
+
+ if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
+ return 0;
+
+ /* find dapm widget path assoc with kcontrol */
+ list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ if (path->kcontrol != kcontrol)
+ continue;
+
+ if (!path->name || !e->texts[mux])
+ continue;
+
+ found = 1;
+ /* we now need to match the string in the enum to the path */
+ if (!(strcmp(path->name, e->texts[mux])))
+ path->connect = 1; /* new connection */
+ else
+ path->connect = 0; /* old connection must be powered down */
+ }
+
+ if (found) {
+ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+ dump_dapm(widget->codec, "mux power update");
+ }
+
+ return 0;
+}
+
+/* test and update the power status of a mixer or switch widget */
+static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kcontrol, int reg,
+ int val_mask, int val, int invert)
+{
+ struct snd_soc_dapm_path *path;
+ int found = 0;
+
+ if (widget->id != snd_soc_dapm_mixer &&
+ widget->id != snd_soc_dapm_switch)
+ return -ENODEV;
+
+ if (!snd_soc_test_bits(widget->codec, reg, val_mask, val))
+ return 0;
+
+ /* find dapm widget path assoc with kcontrol */
+ list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+ if (path->kcontrol != kcontrol)
+ continue;
+
+ /* found, now check type */
+ found = 1;
+ if (val)
+ /* new connection */
+ path->connect = invert ? 0:1;
+ else
+ /* old connection must be powered down */
+ path->connect = invert ? 1:0;
+ break;
+ }
+
+ if (found) {
+ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+ dump_dapm(widget->codec, "mixer power update");
+ }
+
+ return 0;
+}
+
+/* show dapm widget status in sys fs */
+static ssize_t dapm_widget_show(struct device *dev,
+ struct device_attribute *attr, char *buf)
+{
+ struct snd_soc_device *devdata = dev_get_drvdata(dev);
+ struct snd_soc_codec *codec = devdata->codec;
+ struct snd_soc_dapm_widget *w;
+ int count = 0;
+ char *state = "not set";
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ /* only display widgets that burnm power */
+ switch (w->id) {
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_mixer:
+ if (w->name)
+ count += sprintf(buf + count, "%s: %s\n",
+ w->name, w->power ? "On":"Off");
+ break;
+ default:
+ break;
+ }
+ }
+
+ switch (codec->bias_level) {
+ case SND_SOC_BIAS_ON:
+ state = "On";
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ state = "Prepare";
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ state = "Standby";
+ break;
+ case SND_SOC_BIAS_OFF:
+ state = "Off";
+ break;
+ }
+ count += sprintf(buf + count, "PM State: %s\n", state);
+
+ return count;
+}
+
+static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
+
+int snd_soc_dapm_sys_add(struct device *dev)
+{
+ int ret = 0;
+
+ if (!dapm_status)
+ return 0;
+
+ ret = device_create_file(dev, &dev_attr_dapm_widget);
+ if (ret != 0)
+ return ret;
+
+ asoc_debugfs = debugfs_create_dir("asoc", NULL);
+ if (!IS_ERR(asoc_debugfs) && asoc_debugfs)
+ debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
+ &pop_time);
+ else
+ asoc_debugfs = NULL;
+
+ return 0;
+}
+
+static void snd_soc_dapm_sys_remove(struct device *dev)
+{
+ if (dapm_status) {
+ device_remove_file(dev, &dev_attr_dapm_widget);
+ }
+
+ if (asoc_debugfs)
+ debugfs_remove_recursive(asoc_debugfs);
+}
+
+/* free all dapm widgets and resources */
+static void dapm_free_widgets(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_widget *w, *next_w;
+ struct snd_soc_dapm_path *p, *next_p;
+
+ list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) {
+ list_del(&w->list);
+ kfree(w);
+ }
+
+ list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) {
+ list_del(&p->list);
+ kfree(p->long_name);
+ kfree(p);
+ }
+}
+
+static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
+ char *pin, int status)
+{
+ struct snd_soc_dapm_widget *w;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ if (!strcmp(w->name, pin)) {
+ pr_debug("dapm: %s: pin %s\n", codec->name, pin);
+ w->connected = status;
+ return 0;
+ }
+ }
+
+ pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+ return -EINVAL;
+}
+
+/**
+ * snd_soc_dapm_sync - scan and power dapm paths
+ * @codec: audio codec
+ *
+ * Walks all dapm audio paths and powers widgets according to their
+ * stream or path usage.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_sync(struct snd_soc_codec *codec)
+{
+ int ret = dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+ dump_dapm(codec, "sync");
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
+
+static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
+ const char *sink, const char *control, const char *source)
+{
+ struct snd_soc_dapm_path *path;
+ struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
+ int ret = 0;
+
+ /* find src and dest widgets */
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+ if (!wsink && !(strcmp(w->name, sink))) {
+ wsink = w;
+ continue;
+ }
+ if (!wsource && !(strcmp(w->name, source))) {
+ wsource = w;
+ }
+ }
+
+ if (wsource == NULL || wsink == NULL)
+ return -ENODEV;
+
+ path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL);
+ if (!path)
+ return -ENOMEM;
+
+ path->source = wsource;
+ path->sink = wsink;
+ INIT_LIST_HEAD(&path->list);
+ INIT_LIST_HEAD(&path->list_source);
+ INIT_LIST_HEAD(&path->list_sink);
+
+ /* check for external widgets */
+ if (wsink->id == snd_soc_dapm_input) {
+ if (wsource->id == snd_soc_dapm_micbias ||
+ wsource->id == snd_soc_dapm_mic ||
+ wsink->id == snd_soc_dapm_line ||
+ wsink->id == snd_soc_dapm_output)
+ wsink->ext = 1;
+ }
+ if (wsource->id == snd_soc_dapm_output) {
+ if (wsink->id == snd_soc_dapm_spk ||
+ wsink->id == snd_soc_dapm_hp ||
+ wsink->id == snd_soc_dapm_line ||
+ wsink->id == snd_soc_dapm_input)
+ wsource->ext = 1;
+ }
+
+ /* connect static paths */
+ if (control == NULL) {
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &wsink->sources);
+ list_add(&path->list_source, &wsource->sinks);
+ path->connect = 1;
+ return 0;
+ }
+
+ /* connect dynamic paths */
+ switch(wsink->id) {
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_pga:
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_vmid:
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &wsink->sources);
+ list_add(&path->list_source, &wsource->sinks);
+ path->connect = 1;
+ return 0;
+ case snd_soc_dapm_mux:
+ ret = dapm_connect_mux(codec, wsource, wsink, path, control,
+ &wsink->kcontrols[0]);
+ if (ret != 0)
+ goto err;
+ break;
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer:
+ ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
+ if (ret != 0)
+ goto err;
+ break;
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_spk:
+ list_add(&path->list, &codec->dapm_paths);
+ list_add(&path->list_sink, &wsink->sources);
+ list_add(&path->list_source, &wsource->sinks);
+ path->connect = 0;
+ return 0;
+ }
+ return 0;
+
+err:
+ printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source,
+ control, sink);
+ kfree(path);
+ return ret;
+}
+
+/**
+ * snd_soc_dapm_connect_input - connect dapm widgets
+ * @codec: audio codec
+ * @sink: name of target widget
+ * @control: mixer control name
+ * @source: name of source name
+ *
+ * Connects 2 dapm widgets together via a named audio path. The sink is
+ * the widget receiving the audio signal, whilst the source is the sender
+ * of the audio signal.
+ *
+ * This function has been deprecated in favour of snd_soc_dapm_add_routes().
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
+ const char *control, const char *source)
+{
+ return snd_soc_dapm_add_route(codec, sink, control, source);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
+
+/**
+ * snd_soc_dapm_add_routes - Add routes between DAPM widgets
+ * @codec: codec
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * Connects 2 dapm widgets together via a named audio path. The sink is
+ * the widget receiving the audio signal, whilst the source is the sender
+ * of the audio signal.
+ *
+ * Returns 0 for success else error. On error all resources can be freed
+ * with a call to snd_soc_card_free().
+ */
+int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+ const struct snd_soc_dapm_route *route, int num)
+{
+ int i, ret;
+
+ for (i = 0; i < num; i++) {
+ ret = snd_soc_dapm_add_route(codec, route->sink,
+ route->control, route->source);
+ if (ret < 0) {
+ printk(KERN_ERR "Failed to add route %s->%s\n",
+ route->source,
+ route->sink);
+ return ret;
+ }
+ route++;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
+
+/**
+ * snd_soc_dapm_new_widgets - add new dapm widgets
+ * @codec: audio codec
+ *
+ * Checks the codec for any new dapm widgets and creates them if found.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_widget *w;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list)
+ {
+ if (w->new)
+ continue;
+
+ switch(w->id) {
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer:
+ dapm_new_mixer(codec, w);
+ break;
+ case snd_soc_dapm_mux:
+ dapm_new_mux(codec, w);
+ break;
+ case snd_soc_dapm_adc:
+ case snd_soc_dapm_dac:
+ case snd_soc_dapm_pga:
+ dapm_new_pga(codec, w);
+ break;
+ case snd_soc_dapm_input:
+ case snd_soc_dapm_output:
+ case snd_soc_dapm_micbias:
+ case snd_soc_dapm_spk:
+ case snd_soc_dapm_hp:
+ case snd_soc_dapm_mic:
+ case snd_soc_dapm_line:
+ case snd_soc_dapm_vmid:
+ case snd_soc_dapm_pre:
+ case snd_soc_dapm_post:
+ break;
+ }
+ w->new = 1;
+ }
+
+ dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
+
+/**
+ * snd_soc_dapm_get_volsw - dapm mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a dapm mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ unsigned int invert = mc->invert;
+ unsigned int mask = (1 << fls(max)) - 1;
+
+ /* return the saved value if we are powered down */
+ if (widget->id == snd_soc_dapm_pga && !widget->power) {
+ ucontrol->value.integer.value[0] = widget->saved_value;
+ return 0;
+ }
+
+ ucontrol->value.integer.value[0] =
+ (snd_soc_read(widget->codec, reg) >> shift) & mask;
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ (snd_soc_read(widget->codec, reg) >> rshift) & mask;
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ max - ucontrol->value.integer.value[0];
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ max - ucontrol->value.integer.value[1];
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
+
+/**
+ * snd_soc_dapm_put_volsw - dapm mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a dapm mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
+ int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+ unsigned short val, val2, val_mask;
+ int ret;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+
+ if (invert)
+ val = max - val;
+ val_mask = mask << shift;
+ val = val << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1] & mask);
+ if (invert)
+ val2 = max - val2;
+ val_mask |= mask << rshift;
+ val |= val2 << rshift;
+ }
+
+ mutex_lock(&widget->codec->mutex);
+ widget->value = val;
+
+ /* save volume value if the widget is powered down */
+ if (widget->id == snd_soc_dapm_pga && !widget->power) {
+ widget->saved_value = val;
+ mutex_unlock(&widget->codec->mutex);
+ return 1;
+ }
+
+ dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert);
+ if (widget->event) {
+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+ ret = widget->event(widget, kcontrol,
+ SND_SOC_DAPM_PRE_REG);
+ if (ret < 0) {
+ ret = 1;
+ goto out;
+ }
+ }
+ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
+ if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+ ret = widget->event(widget, kcontrol,
+ SND_SOC_DAPM_POST_REG);
+ } else
+ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
+
+out:
+ mutex_unlock(&widget->codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
+
+/**
+ * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a dapm enumerated double mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val, bitmask;
+
+ for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+ ;
+ val = snd_soc_read(widget->codec, e->reg);
+ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+ if (e->shift_l != e->shift_r)
+ ucontrol->value.enumerated.item[1] =
+ (val >> e->shift_r) & (bitmask - 1);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
+
+/**
+ * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a dapm enumerated double mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val, mux;
+ unsigned short mask, bitmask;
+ int ret = 0;
+
+ for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+ ;
+ if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ return -EINVAL;
+ mux = ucontrol->value.enumerated.item[0];
+ val = mux << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
+ if (e->shift_l != e->shift_r) {
+ if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ return -EINVAL;
+ val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
+ }
+
+ mutex_lock(&widget->codec->mutex);
+ widget->value = val;
+ dapm_mux_update_power(widget, kcontrol, mask, mux, val, e);
+ if (widget->event) {
+ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+ ret = widget->event(widget,
+ kcontrol, SND_SOC_DAPM_PRE_REG);
+ if (ret < 0)
+ goto out;
+ }
+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+ if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+ ret = widget->event(widget,
+ kcontrol, SND_SOC_DAPM_POST_REG);
+ } else
+ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+
+out:
+ mutex_unlock(&widget->codec->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
+
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @codec: audio codec
+ * @widget: widget template
+ *
+ * Creates a new dapm control based upon the template.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
+ const struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_dapm_widget *w;
+
+ if ((w = dapm_cnew_widget(widget)) == NULL)
+ return -ENOMEM;
+
+ w->codec = codec;
+ INIT_LIST_HEAD(&w->sources);
+ INIT_LIST_HEAD(&w->sinks);
+ INIT_LIST_HEAD(&w->list);
+ list_add(&w->list, &codec->dapm_widgets);
+
+ /* machine layer set ups unconnected pins and insertions */
+ w->connected = 1;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
+
+/**
+ * snd_soc_dapm_new_controls - create new dapm controls
+ * @codec: audio codec
+ * @widget: widget array
+ * @num: number of widgets
+ *
+ * Creates new DAPM controls based upon the templates.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+ const struct snd_soc_dapm_widget *widget,
+ int num)
+{
+ int i, ret;
+
+ for (i = 0; i < num; i++) {
+ ret = snd_soc_dapm_new_control(codec, widget);
+ if (ret < 0)
+ return ret;
+ widget++;
+ }
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
+
+
+/**
+ * snd_soc_dapm_stream_event - send a stream event to the dapm core
+ * @codec: audio codec
+ * @stream: stream name
+ * @event: stream event
+ *
+ * Sends a stream event to the dapm core. The core then makes any
+ * necessary widget power changes.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
+ char *stream, int event)
+{
+ struct snd_soc_dapm_widget *w;
+
+ if (stream == NULL)
+ return 0;
+
+ mutex_lock(&codec->mutex);
+ list_for_each_entry(w, &codec->dapm_widgets, list)
+ {
+ if (!w->sname)
+ continue;
+ pr_debug("widget %s\n %s stream %s event %d\n",
+ w->name, w->sname, stream, event);
+ if (strstr(w->sname, stream)) {
+ switch(event) {
+ case SND_SOC_DAPM_STREAM_START:
+ w->active = 1;
+ break;
+ case SND_SOC_DAPM_STREAM_STOP:
+ w->active = 0;
+ break;
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ if (w->active)
+ w->suspend = 1;
+ w->active = 0;
+ break;
+ case SND_SOC_DAPM_STREAM_RESUME:
+ if (w->suspend) {
+ w->active = 1;
+ w->suspend = 0;
+ }
+ break;
+ case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
+ break;
+ case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
+ break;
+ }
+ }
+ }
+ mutex_unlock(&codec->mutex);
+
+ dapm_power_widgets(codec, event);
+ dump_dapm(codec, __func__);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
+
+/**
+ * snd_soc_dapm_set_bias_level - set the bias level for the system
+ * @socdev: audio device
+ * @level: level to configure
+ *
+ * Configure the bias (power) levels for the SoC audio device.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+ struct snd_soc_machine *machine = socdev->machine;
+ int ret = 0;
+
+ if (machine->set_bias_level)
+ ret = machine->set_bias_level(machine, level);
+ if (ret == 0 && codec->set_bias_level)
+ ret = codec->set_bias_level(codec, level);
+
+ return ret;
+}
+
+/**
+ * snd_soc_dapm_enable_pin - enable pin.
+ * @snd_soc_codec: SoC codec
+ * @pin: pin name
+ *
+ * Enables input/output pin and it's parents or children widgets iff there is
+ * a valid audio route and active audio stream.
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin)
+{
+ return snd_soc_dapm_set_pin(codec, pin, 1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
+
+/**
+ * snd_soc_dapm_disable_pin - disable pin.
+ * @codec: SoC codec
+ * @pin: pin name
+ *
+ * Disables input/output pin and it's parents or children widgets.
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
+{
+ return snd_soc_dapm_set_pin(codec, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
+
+/**
+ * snd_soc_dapm_nc_pin - permanently disable pin.
+ * @codec: SoC codec
+ * @pin: pin name
+ *
+ * Marks the specified pin as being not connected, disabling it along
+ * any parent or child widgets. At present this is identical to
+ * snd_soc_dapm_disable_pin() but in future it will be extended to do
+ * additional things such as disabling controls which only affect
+ * paths through the pin.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin)
+{
+ return snd_soc_dapm_set_pin(codec, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
+
+/**
+ * snd_soc_dapm_get_pin_status - get audio pin status
+ * @codec: audio codec
+ * @pin: audio signal pin endpoint (or start point)
+ *
+ * Get audio pin status - connected or disconnected.
+ *
+ * Returns 1 for connected otherwise 0.
+ */
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin)
+{
+ struct snd_soc_dapm_widget *w;
+
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ if (!strcmp(w->name, pin))
+ return w->connected;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
+
+/**
+ * snd_soc_dapm_free - free dapm resources
+ * @socdev: SoC device
+ *
+ * Free all dapm widgets and resources.
+ */
+void snd_soc_dapm_free(struct snd_soc_device *socdev)
+{
+ struct snd_soc_codec *codec = socdev->codec;
+
+ snd_soc_dapm_sys_remove(socdev->dev);
+ dapm_free_widgets(codec);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
+MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
+MODULE_LICENSE("GPL");
diff --git a/sound/sound_core.c b/sound/sound_core.c
new file mode 100644
index 0000000..10ba421
--- /dev/null
+++ b/sound/sound_core.c
@@ -0,0 +1,593 @@
+/*
+ * Sound core. This file is composed of two parts. sound_class
+ * which is common to both OSS and ALSA and OSS sound core which
+ * is used OSS or emulation of it.
+ */
+
+/*
+ * First, the common part.
+ */
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/err.h>
+
+#ifdef CONFIG_SOUND_OSS_CORE
+static int __init init_oss_soundcore(void);
+static void cleanup_oss_soundcore(void);
+#else
+static inline int init_oss_soundcore(void) { return 0; }
+static inline void cleanup_oss_soundcore(void) { }
+#endif
+
+struct class *sound_class;
+EXPORT_SYMBOL(sound_class);
+
+MODULE_DESCRIPTION("Core sound module");
+MODULE_AUTHOR("Alan Cox");
+MODULE_LICENSE("GPL");
+
+static int __init init_soundcore(void)
+{
+ int rc;
+
+ rc = init_oss_soundcore();
+ if (rc)
+ return rc;
+
+ sound_class = class_create(THIS_MODULE, "sound");
+ if (IS_ERR(sound_class)) {
+ cleanup_oss_soundcore();
+ return PTR_ERR(sound_class);
+ }
+
+ return 0;
+}
+
+static void __exit cleanup_soundcore(void)
+{
+ cleanup_oss_soundcore();
+ class_destroy(sound_class);
+}
+
+module_init(init_soundcore);
+module_exit(cleanup_soundcore);
+
+
+#ifdef CONFIG_SOUND_OSS_CORE
+/*
+ * OSS sound core handling. Breaks out sound functions to submodules
+ *
+ * Author: Alan Cox <alan@lxorguk.ukuu.org.uk>
+ *
+ * Fixes:
+ *
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version
+ * 2 of the License, or (at your option) any later version.
+ *
+ * --------------------
+ *
+ * Top level handler for the sound subsystem. Various devices can
+ * plug into this. The fact they don't all go via OSS doesn't mean
+ * they don't have to implement the OSS API. There is a lot of logic
+ * to keeping much of the OSS weight out of the code in a compatibility
+ * module, but it's up to the driver to rember to load it...
+ *
+ * The code provides a set of functions for registration of devices
+ * by type. This is done rather than providing a single call so that
+ * we can hide any future changes in the internals (eg when we go to
+ * 32bit dev_t) from the modules and their interface.
+ *
+ * Secondly we need to allocate the dsp, dsp16 and audio devices as
+ * one. Thus we misuse the chains a bit to simplify this.
+ *
+ * Thirdly to make it more fun and for 2.3.x and above we do all
+ * of this using fine grained locking.
+ *
+ * FIXME: we have to resolve modules and fine grained load/unload
+ * locking at some point in 2.3.x.
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/smp_lock.h>
+#include <linux/types.h>
+#include <linux/kernel.h>
+#include <linux/sound.h>
+#include <linux/major.h>
+#include <linux/kmod.h>
+
+#define SOUND_STEP 16
+
+struct sound_unit
+{
+ int unit_minor;
+ const struct file_operations *unit_fops;
+ struct sound_unit *next;
+ char name[32];
+};
+
+#ifdef CONFIG_SOUND_MSNDCLAS
+extern int msnd_classic_init(void);
+#endif
+#ifdef CONFIG_SOUND_MSNDPIN
+extern int msnd_pinnacle_init(void);
+#endif
+
+/*
+ * Low level list operator. Scan the ordered list, find a hole and
+ * join into it. Called with the lock asserted
+ */
+
+static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top)
+{
+ int n=low;
+
+ if (index < 0) { /* first free */
+
+ while (*list && (*list)->unit_minor<n)
+ list=&((*list)->next);
+
+ while(n<top)
+ {
+ /* Found a hole ? */
+ if(*list==NULL || (*list)->unit_minor>n)
+ break;
+ list=&((*list)->next);
+ n+=SOUND_STEP;
+ }
+
+ if(n>=top)
+ return -ENOENT;
+ } else {
+ n = low+(index*16);
+ while (*list) {
+ if ((*list)->unit_minor==n)
+ return -EBUSY;
+ if ((*list)->unit_minor>n)
+ break;
+ list=&((*list)->next);
+ }
+ }
+
+ /*
+ * Fill it in
+ */
+
+ s->unit_minor=n;
+ s->unit_fops=fops;
+
+ /*
+ * Link it
+ */
+
+ s->next=*list;
+ *list=s;
+
+
+ return n;
+}
+
+/*
+ * Remove a node from the chain. Called with the lock asserted
+ */
+
+static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit)
+{
+ while(*list)
+ {
+ struct sound_unit *p=*list;
+ if(p->unit_minor==unit)
+ {
+ *list=p->next;
+ return p;
+ }
+ list=&(p->next);
+ }
+ printk(KERN_ERR "Sound device %d went missing!\n", unit);
+ return NULL;
+}
+
+/*
+ * This lock guards the sound loader list.
+ */
+
+static DEFINE_SPINLOCK(sound_loader_lock);
+
+/*
+ * Allocate the controlling structure and add it to the sound driver
+ * list. Acquires locks as needed
+ */
+
+static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev)
+{
+ struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL);
+ int r;
+
+ if (!s)
+ return -ENOMEM;
+
+ spin_lock(&sound_loader_lock);
+ r = __sound_insert_unit(s, list, fops, index, low, top);
+ spin_unlock(&sound_loader_lock);
+
+ if (r < 0)
+ goto fail;
+ else if (r < SOUND_STEP)
+ sprintf(s->name, "sound/%s", name);
+ else
+ sprintf(s->name, "sound/%s%d", name, r / SOUND_STEP);
+
+ device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor),
+ NULL, s->name+6);
+ return r;
+
+ fail:
+ kfree(s);
+ return r;
+}
+
+/*
+ * Remove a unit. Acquires locks as needed. The drivers MUST have
+ * completed the removal before their file operations become
+ * invalid.
+ */
+
+static void sound_remove_unit(struct sound_unit **list, int unit)
+{
+ struct sound_unit *p;
+
+ spin_lock(&sound_loader_lock);
+ p = __sound_remove_unit(list, unit);
+ spin_unlock(&sound_loader_lock);
+ if (p) {
+ device_destroy(sound_class, MKDEV(SOUND_MAJOR, p->unit_minor));
+ kfree(p);
+ }
+}
+
+/*
+ * Allocations
+ *
+ * 0 *16 Mixers
+ * 1 *8 Sequencers
+ * 2 *16 Midi
+ * 3 *16 DSP
+ * 4 *16 SunDSP
+ * 5 *16 DSP16
+ * 6 -- sndstat (obsolete)
+ * 7 *16 unused
+ * 8 -- alternate sequencer (see above)
+ * 9 *16 raw synthesizer access
+ * 10 *16 unused
+ * 11 *16 unused
+ * 12 *16 unused
+ * 13 *16 unused
+ * 14 *16 unused
+ * 15 *16 unused
+ */
+
+static struct sound_unit *chains[SOUND_STEP];
+
+/**
+ * register_sound_special_device - register a special sound node
+ * @fops: File operations for the driver
+ * @unit: Unit number to allocate
+ * @dev: device pointer
+ *
+ * Allocate a special sound device by minor number from the sound
+ * subsystem. The allocated number is returned on succes. On failure
+ * a negative error code is returned.
+ */
+
+int register_sound_special_device(const struct file_operations *fops, int unit,
+ struct device *dev)
+{
+ const int chain = unit % SOUND_STEP;
+ int max_unit = 128 + chain;
+ const char *name;
+ char _name[16];
+
+ switch (chain) {
+ case 0:
+ name = "mixer";
+ break;
+ case 1:
+ name = "sequencer";
+ if (unit >= SOUND_STEP)
+ goto __unknown;
+ max_unit = unit + 1;
+ break;
+ case 2:
+ name = "midi";
+ break;
+ case 3:
+ name = "dsp";
+ break;
+ case 4:
+ name = "audio";
+ break;
+ case 8:
+ name = "sequencer2";
+ if (unit >= SOUND_STEP)
+ goto __unknown;
+ max_unit = unit + 1;
+ break;
+ case 9:
+ name = "dmmidi";
+ break;
+ case 10:
+ name = "dmfm";
+ break;
+ case 12:
+ name = "adsp";
+ break;
+ case 13:
+ name = "amidi";
+ break;
+ case 14:
+ name = "admmidi";
+ break;
+ default:
+ {
+ __unknown:
+ sprintf(_name, "unknown%d", chain);
+ if (unit >= SOUND_STEP)
+ strcat(_name, "-");
+ name = _name;
+ }
+ break;
+ }
+ return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit,
+ name, S_IRUSR | S_IWUSR, dev);
+}
+
+EXPORT_SYMBOL(register_sound_special_device);
+
+int register_sound_special(const struct file_operations *fops, int unit)
+{
+ return register_sound_special_device(fops, unit, NULL);
+}
+
+EXPORT_SYMBOL(register_sound_special);
+
+/**
+ * register_sound_mixer - register a mixer device
+ * @fops: File operations for the driver
+ * @dev: Unit number to allocate
+ *
+ * Allocate a mixer device. Unit is the number of the mixer requested.
+ * Pass -1 to request the next free mixer unit. On success the allocated
+ * number is returned, on failure a negative error code is returned.
+ */
+
+int register_sound_mixer(const struct file_operations *fops, int dev)
+{
+ return sound_insert_unit(&chains[0], fops, dev, 0, 128,
+ "mixer", S_IRUSR | S_IWUSR, NULL);
+}
+
+EXPORT_SYMBOL(register_sound_mixer);
+
+/**
+ * register_sound_midi - register a midi device
+ * @fops: File operations for the driver
+ * @dev: Unit number to allocate
+ *
+ * Allocate a midi device. Unit is the number of the midi device requested.
+ * Pass -1 to request the next free midi unit. On success the allocated
+ * number is returned, on failure a negative error code is returned.
+ */
+
+int register_sound_midi(const struct file_operations *fops, int dev)
+{
+ return sound_insert_unit(&chains[2], fops, dev, 2, 130,
+ "midi", S_IRUSR | S_IWUSR, NULL);
+}
+
+EXPORT_SYMBOL(register_sound_midi);
+
+/*
+ * DSP's are registered as a triple. Register only one and cheat
+ * in open - see below.
+ */
+
+/**
+ * register_sound_dsp - register a DSP device
+ * @fops: File operations for the driver
+ * @dev: Unit number to allocate
+ *
+ * Allocate a DSP device. Unit is the number of the DSP requested.
+ * Pass -1 to request the next free DSP unit. On success the allocated
+ * number is returned, on failure a negative error code is returned.
+ *
+ * This function allocates both the audio and dsp device entries together
+ * and will always allocate them as a matching pair - eg dsp3/audio3
+ */
+
+int register_sound_dsp(const struct file_operations *fops, int dev)
+{
+ return sound_insert_unit(&chains[3], fops, dev, 3, 131,
+ "dsp", S_IWUSR | S_IRUSR, NULL);
+}
+
+EXPORT_SYMBOL(register_sound_dsp);
+
+/**
+ * unregister_sound_special - unregister a special sound device
+ * @unit: unit number to allocate
+ *
+ * Release a sound device that was allocated with
+ * register_sound_special(). The unit passed is the return value from
+ * the register function.
+ */
+
+
+void unregister_sound_special(int unit)
+{
+ sound_remove_unit(&chains[unit % SOUND_STEP], unit);
+}
+
+EXPORT_SYMBOL(unregister_sound_special);
+
+/**
+ * unregister_sound_mixer - unregister a mixer
+ * @unit: unit number to allocate
+ *
+ * Release a sound device that was allocated with register_sound_mixer().
+ * The unit passed is the return value from the register function.
+ */
+
+void unregister_sound_mixer(int unit)
+{
+ sound_remove_unit(&chains[0], unit);
+}
+
+EXPORT_SYMBOL(unregister_sound_mixer);
+
+/**
+ * unregister_sound_midi - unregister a midi device
+ * @unit: unit number to allocate
+ *
+ * Release a sound device that was allocated with register_sound_midi().
+ * The unit passed is the return value from the register function.
+ */
+
+void unregister_sound_midi(int unit)
+{
+ sound_remove_unit(&chains[2], unit);
+}
+
+EXPORT_SYMBOL(unregister_sound_midi);
+
+/**
+ * unregister_sound_dsp - unregister a DSP device
+ * @unit: unit number to allocate
+ *
+ * Release a sound device that was allocated with register_sound_dsp().
+ * The unit passed is the return value from the register function.
+ *
+ * Both of the allocated units are released together automatically.
+ */
+
+void unregister_sound_dsp(int unit)
+{
+ sound_remove_unit(&chains[3], unit);
+}
+
+
+EXPORT_SYMBOL(unregister_sound_dsp);
+
+/*
+ * Now our file operations
+ */
+
+static int soundcore_open(struct inode *, struct file *);
+
+static const struct file_operations soundcore_fops=
+{
+ /* We must have an owner or the module locking fails */
+ .owner = THIS_MODULE,
+ .open = soundcore_open,
+};
+
+static struct sound_unit *__look_for_unit(int chain, int unit)
+{
+ struct sound_unit *s;
+
+ s=chains[chain];
+ while(s && s->unit_minor <= unit)
+ {
+ if(s->unit_minor==unit)
+ return s;
+ s=s->next;
+ }
+ return NULL;
+}
+
+static int soundcore_open(struct inode *inode, struct file *file)
+{
+ int chain;
+ int unit = iminor(inode);
+ struct sound_unit *s;
+ const struct file_operations *new_fops = NULL;
+
+ lock_kernel ();
+
+ chain=unit&0x0F;
+ if(chain==4 || chain==5) /* dsp/audio/dsp16 */
+ {
+ unit&=0xF0;
+ unit|=3;
+ chain=3;
+ }
+
+ spin_lock(&sound_loader_lock);
+ s = __look_for_unit(chain, unit);
+ if (s)
+ new_fops = fops_get(s->unit_fops);
+ if (!new_fops) {
+ spin_unlock(&sound_loader_lock);
+ /*
+ * Please, don't change this order or code.
+ * For ALSA slot means soundcard and OSS emulation code
+ * comes as add-on modules which aren't depend on
+ * ALSA toplevel modules for soundcards, thus we need
+ * load them at first. [Jaroslav Kysela <perex@jcu.cz>]
+ */
+ request_module("sound-slot-%i", unit>>4);
+ request_module("sound-service-%i-%i", unit>>4, chain);
+ spin_lock(&sound_loader_lock);
+ s = __look_for_unit(chain, unit);
+ if (s)
+ new_fops = fops_get(s->unit_fops);
+ }
+ if (new_fops) {
+ /*
+ * We rely upon the fact that we can't be unloaded while the
+ * subdriver is there, so if ->open() is successful we can
+ * safely drop the reference counter and if it is not we can
+ * revert to old ->f_op. Ugly, indeed, but that's the cost of
+ * switching ->f_op in the first place.
+ */
+ int err = 0;
+ const struct file_operations *old_fops = file->f_op;
+ file->f_op = new_fops;
+ spin_unlock(&sound_loader_lock);
+ if(file->f_op->open)
+ err = file->f_op->open(inode,file);
+ if (err) {
+ fops_put(file->f_op);
+ file->f_op = fops_get(old_fops);
+ }
+ fops_put(old_fops);
+ unlock_kernel();
+ return err;
+ }
+ spin_unlock(&sound_loader_lock);
+ unlock_kernel();
+ return -ENODEV;
+}
+
+MODULE_ALIAS_CHARDEV_MAJOR(SOUND_MAJOR);
+
+static void cleanup_oss_soundcore(void)
+{
+ /* We have nothing to really do here - we know the lists must be
+ empty */
+ unregister_chrdev(SOUND_MAJOR, "sound");
+}
+
+static int __init init_oss_soundcore(void)
+{
+ if (register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops)==-1) {
+ printk(KERN_ERR "soundcore: sound device already in use.\n");
+ return -EBUSY;
+ }
+
+ return 0;
+}
+
+#endif /* CONFIG_SOUND_OSS_CORE */
diff --git a/sound/sound_firmware.c b/sound/sound_firmware.c
new file mode 100644
index 0000000..96deaef
--- /dev/null
+++ b/sound/sound_firmware.c
@@ -0,0 +1,79 @@
+#include <linux/vmalloc.h>
+#include <linux/module.h>
+#include <linux/fs.h>
+#include <linux/mm.h>
+#include <linux/slab.h>
+#include <linux/sched.h>
+#include <asm/uaccess.h>
+#include "oss/sound_firmware.h"
+
+static int do_mod_firmware_load(const char *fn, char **fp)
+{
+ struct file* filp;
+ long l;
+ char *dp;
+ loff_t pos;
+
+ filp = filp_open(fn, 0, 0);
+ if (IS_ERR(filp))
+ {
+ printk(KERN_INFO "Unable to load '%s'.\n", fn);
+ return 0;
+ }
+ l = filp->f_path.dentry->d_inode->i_size;
+ if (l <= 0 || l > 131072)
+ {
+ printk(KERN_INFO "Invalid firmware '%s'\n", fn);
+ filp_close(filp, current->files);
+ return 0;
+ }
+ dp = vmalloc(l);
+ if (dp == NULL)
+ {
+ printk(KERN_INFO "Out of memory loading '%s'.\n", fn);
+ filp_close(filp, current->files);
+ return 0;
+ }
+ pos = 0;
+ if (vfs_read(filp, dp, l, &pos) != l)
+ {
+ printk(KERN_INFO "Failed to read '%s'.\n", fn);
+ vfree(dp);
+ filp_close(filp, current->files);
+ return 0;
+ }
+ filp_close(filp, current->files);
+ *fp = dp;
+ return (int) l;
+}
+
+/**
+ * mod_firmware_load - load sound driver firmware
+ * @fn: filename
+ * @fp: return for the buffer.
+ *
+ * Load the firmware for a sound module (up to 128K) into a buffer.
+ * The buffer is returned in *fp. It is allocated with vmalloc so is
+ * virtually linear and not DMAable. The caller should free it with
+ * vfree when finished.
+ *
+ * The length of the buffer is returned on a successful load, the
+ * value zero on a failure.
+ *
+ * Caution: This API is not recommended. Firmware should be loaded via
+ * request_firmware.
+ */
+
+int mod_firmware_load(const char *fn, char **fp)
+{
+ int r;
+ mm_segment_t fs = get_fs();
+
+ set_fs(get_ds());
+ r = do_mod_firmware_load(fn, fp);
+ set_fs(fs);
+ return r;
+}
+EXPORT_SYMBOL(mod_firmware_load);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/sparc/Kconfig b/sound/sparc/Kconfig
new file mode 100644
index 0000000..d75deba
--- /dev/null
+++ b/sound/sparc/Kconfig
@@ -0,0 +1,41 @@
+# ALSA Sparc drivers
+
+menuconfig SND_SPARC
+ bool "Sparc sound devices"
+ depends on SPARC
+ default y
+ help
+ Support for sound devices specific to Sun SPARC architectures.
+
+if SND_SPARC
+
+config SND_SUN_AMD7930
+ tristate "Sun AMD7930"
+ depends on SBUS
+ select SND_PCM
+ help
+ Say Y here to include support for AMD7930 sound device on Sun.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sun-amd7930.
+
+config SND_SUN_CS4231
+ tristate "Sun CS4231"
+ select SND_PCM
+ help
+ Say Y here to include support for CS4231 sound device on Sun.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sun-cs4231.
+
+config SND_SUN_DBRI
+ tristate "Sun DBRI"
+ depends on SBUS
+ select SND_PCM
+ help
+ Say Y here to include support for DBRI sound device on Sun.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-sun-dbri.
+
+endif # SND_SPARC
diff --git a/sound/sparc/Makefile b/sound/sparc/Makefile
new file mode 100644
index 0000000..3cd89c6
--- /dev/null
+++ b/sound/sparc/Makefile
@@ -0,0 +1,12 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2002 by David S. Miller <davem@redhat.com>
+#
+
+snd-sun-amd7930-objs := amd7930.o
+snd-sun-cs4231-objs := cs4231.o
+snd-sun-dbri-objs := dbri.o
+
+obj-$(CONFIG_SND_SUN_AMD7930) += snd-sun-amd7930.o
+obj-$(CONFIG_SND_SUN_CS4231) += snd-sun-cs4231.o
+obj-$(CONFIG_SND_SUN_DBRI) += snd-sun-dbri.o
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
new file mode 100644
index 0000000..f87933e
--- /dev/null
+++ b/sound/sparc/amd7930.c
@@ -0,0 +1,1094 @@
+/*
+ * Driver for AMD7930 sound chips found on Sparcs.
+ * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net>
+ *
+ * Based entirely upon drivers/sbus/audio/amd7930.c which is:
+ * Copyright (C) 1996,1997 Thomas K. Dyas (tdyas@eden.rutgers.edu)
+ *
+ * --- Notes from Thomas's original driver ---
+ * This is the lowlevel driver for the AMD7930 audio chip found on all
+ * sun4c machines and some sun4m machines.
+ *
+ * The amd7930 is actually an ISDN chip which has a very simple
+ * integrated audio encoder/decoder. When Sun decided on what chip to
+ * use for audio, they had the brilliant idea of using the amd7930 and
+ * only connecting the audio encoder/decoder pins.
+ *
+ * Thanks to the AMD engineer who was able to get us the AMD79C30
+ * databook which has all the programming information and gain tables.
+ *
+ * Advanced Micro Devices' Am79C30A is an ISDN/audio chip used in the
+ * SparcStation 1+. The chip provides microphone and speaker interfaces
+ * which provide mono-channel audio at 8K samples per second via either
+ * 8-bit A-law or 8-bit mu-law encoding. Also, the chip features an
+ * ISDN BRI Line Interface Unit (LIU), I.430 S/T physical interface,
+ * which performs basic D channel LAPD processing and provides raw
+ * B channel data. The digital audio channel, the two ISDN B channels,
+ * and two 64 Kbps channels to the microprocessor are all interconnected
+ * via a multiplexer.
+ * --- End of notes from Thoamas's original driver ---
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/slab.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+
+#include <asm/io.h>
+#include <asm/irq.h>
+#include <asm/prom.h>
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Sun AMD7930 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Sun AMD7930 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Sun AMD7930 soundcard.");
+MODULE_AUTHOR("Thomas K. Dyas and David S. Miller");
+MODULE_DESCRIPTION("Sun AMD7930");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Sun,AMD7930}}");
+
+/* Device register layout. */
+
+/* Register interface presented to the CPU by the amd7930. */
+#define AMD7930_CR 0x00UL /* Command Register (W) */
+#define AMD7930_IR AMD7930_CR /* Interrupt Register (R) */
+#define AMD7930_DR 0x01UL /* Data Register (R/W) */
+#define AMD7930_DSR1 0x02UL /* D-channel Status Register 1 (R) */
+#define AMD7930_DER 0x03UL /* D-channel Error Register (R) */
+#define AMD7930_DCTB 0x04UL /* D-channel Transmit Buffer (W) */
+#define AMD7930_DCRB AMD7930_DCTB /* D-channel Receive Buffer (R) */
+#define AMD7930_BBTB 0x05UL /* Bb-channel Transmit Buffer (W) */
+#define AMD7930_BBRB AMD7930_BBTB /* Bb-channel Receive Buffer (R) */
+#define AMD7930_BCTB 0x06UL /* Bc-channel Transmit Buffer (W) */
+#define AMD7930_BCRB AMD7930_BCTB /* Bc-channel Receive Buffer (R) */
+#define AMD7930_DSR2 0x07UL /* D-channel Status Register 2 (R) */
+
+/* Indirect registers in the Main Audio Processor. */
+struct amd7930_map {
+ __u16 x[8];
+ __u16 r[8];
+ __u16 gx;
+ __u16 gr;
+ __u16 ger;
+ __u16 stgr;
+ __u16 ftgr;
+ __u16 atgr;
+ __u8 mmr1;
+ __u8 mmr2;
+};
+
+/* After an amd7930 interrupt, reading the Interrupt Register (ir)
+ * clears the interrupt and returns a bitmask indicating which
+ * interrupt source(s) require service.
+ */
+
+#define AMR_IR_DTTHRSH 0x01 /* D-channel xmit threshold */
+#define AMR_IR_DRTHRSH 0x02 /* D-channel recv threshold */
+#define AMR_IR_DSRI 0x04 /* D-channel packet status */
+#define AMR_IR_DERI 0x08 /* D-channel error */
+#define AMR_IR_BBUF 0x10 /* B-channel data xfer */
+#define AMR_IR_LSRI 0x20 /* LIU status */
+#define AMR_IR_DSR2I 0x40 /* D-channel buffer status */
+#define AMR_IR_MLTFRMI 0x80 /* multiframe or PP */
+
+/* The amd7930 has "indirect registers" which are accessed by writing
+ * the register number into the Command Register and then reading or
+ * writing values from the Data Register as appropriate. We define the
+ * AMR_* macros to be the indirect register numbers and AM_* macros to
+ * be bits in whatever register is referred to.
+ */
+
+/* Initialization */
+#define AMR_INIT 0x21
+#define AM_INIT_ACTIVE 0x01
+#define AM_INIT_DATAONLY 0x02
+#define AM_INIT_POWERDOWN 0x03
+#define AM_INIT_DISABLE_INTS 0x04
+#define AMR_INIT2 0x20
+#define AM_INIT2_ENABLE_POWERDOWN 0x20
+#define AM_INIT2_ENABLE_MULTIFRAME 0x10
+
+/* Line Interface Unit */
+#define AMR_LIU_LSR 0xA1
+#define AM_LIU_LSR_STATE 0x07
+#define AM_LIU_LSR_F3 0x08
+#define AM_LIU_LSR_F7 0x10
+#define AM_LIU_LSR_F8 0x20
+#define AM_LIU_LSR_HSW 0x40
+#define AM_LIU_LSR_HSW_CHG 0x80
+#define AMR_LIU_LPR 0xA2
+#define AMR_LIU_LMR1 0xA3
+#define AM_LIU_LMR1_B1_ENABL 0x01
+#define AM_LIU_LMR1_B2_ENABL 0x02
+#define AM_LIU_LMR1_F_DISABL 0x04
+#define AM_LIU_LMR1_FA_DISABL 0x08
+#define AM_LIU_LMR1_REQ_ACTIV 0x10
+#define AM_LIU_LMR1_F8_F3 0x20
+#define AM_LIU_LMR1_LIU_ENABL 0x40
+#define AMR_LIU_LMR2 0xA4
+#define AM_LIU_LMR2_DECHO 0x01
+#define AM_LIU_LMR2_DLOOP 0x02
+#define AM_LIU_LMR2_DBACKOFF 0x04
+#define AM_LIU_LMR2_EN_F3_INT 0x08
+#define AM_LIU_LMR2_EN_F8_INT 0x10
+#define AM_LIU_LMR2_EN_HSW_INT 0x20
+#define AM_LIU_LMR2_EN_F7_INT 0x40
+#define AMR_LIU_2_4 0xA5
+#define AMR_LIU_MF 0xA6
+#define AMR_LIU_MFSB 0xA7
+#define AMR_LIU_MFQB 0xA8
+
+/* Multiplexor */
+#define AMR_MUX_MCR1 0x41
+#define AMR_MUX_MCR2 0x42
+#define AMR_MUX_MCR3 0x43
+#define AM_MUX_CHANNEL_B1 0x01
+#define AM_MUX_CHANNEL_B2 0x02
+#define AM_MUX_CHANNEL_Ba 0x03
+#define AM_MUX_CHANNEL_Bb 0x04
+#define AM_MUX_CHANNEL_Bc 0x05
+#define AM_MUX_CHANNEL_Bd 0x06
+#define AM_MUX_CHANNEL_Be 0x07
+#define AM_MUX_CHANNEL_Bf 0x08
+#define AMR_MUX_MCR4 0x44
+#define AM_MUX_MCR4_ENABLE_INTS 0x08
+#define AM_MUX_MCR4_REVERSE_Bb 0x10
+#define AM_MUX_MCR4_REVERSE_Bc 0x20
+#define AMR_MUX_1_4 0x45
+
+/* Main Audio Processor */
+#define AMR_MAP_X 0x61
+#define AMR_MAP_R 0x62
+#define AMR_MAP_GX 0x63
+#define AMR_MAP_GR 0x64
+#define AMR_MAP_GER 0x65
+#define AMR_MAP_STGR 0x66
+#define AMR_MAP_FTGR_1_2 0x67
+#define AMR_MAP_ATGR_1_2 0x68
+#define AMR_MAP_MMR1 0x69
+#define AM_MAP_MMR1_ALAW 0x01
+#define AM_MAP_MMR1_GX 0x02
+#define AM_MAP_MMR1_GR 0x04
+#define AM_MAP_MMR1_GER 0x08
+#define AM_MAP_MMR1_X 0x10
+#define AM_MAP_MMR1_R 0x20
+#define AM_MAP_MMR1_STG 0x40
+#define AM_MAP_MMR1_LOOPBACK 0x80
+#define AMR_MAP_MMR2 0x6A
+#define AM_MAP_MMR2_AINB 0x01
+#define AM_MAP_MMR2_LS 0x02
+#define AM_MAP_MMR2_ENABLE_DTMF 0x04
+#define AM_MAP_MMR2_ENABLE_TONEGEN 0x08
+#define AM_MAP_MMR2_ENABLE_TONERING 0x10
+#define AM_MAP_MMR2_DISABLE_HIGHPASS 0x20
+#define AM_MAP_MMR2_DISABLE_AUTOZERO 0x40
+#define AMR_MAP_1_10 0x6B
+#define AMR_MAP_MMR3 0x6C
+#define AMR_MAP_STRA 0x6D
+#define AMR_MAP_STRF 0x6E
+#define AMR_MAP_PEAKX 0x70
+#define AMR_MAP_PEAKR 0x71
+#define AMR_MAP_15_16 0x72
+
+/* Data Link Controller */
+#define AMR_DLC_FRAR_1_2_3 0x81
+#define AMR_DLC_SRAR_1_2_3 0x82
+#define AMR_DLC_TAR 0x83
+#define AMR_DLC_DRLR 0x84
+#define AMR_DLC_DTCR 0x85
+#define AMR_DLC_DMR1 0x86
+#define AMR_DLC_DMR1_DTTHRSH_INT 0x01
+#define AMR_DLC_DMR1_DRTHRSH_INT 0x02
+#define AMR_DLC_DMR1_TAR_ENABL 0x04
+#define AMR_DLC_DMR1_EORP_INT 0x08
+#define AMR_DLC_DMR1_EN_ADDR1 0x10
+#define AMR_DLC_DMR1_EN_ADDR2 0x20
+#define AMR_DLC_DMR1_EN_ADDR3 0x40
+#define AMR_DLC_DMR1_EN_ADDR4 0x80
+#define AMR_DLC_DMR1_EN_ADDRS 0xf0
+#define AMR_DLC_DMR2 0x87
+#define AMR_DLC_DMR2_RABRT_INT 0x01
+#define AMR_DLC_DMR2_RESID_INT 0x02
+#define AMR_DLC_DMR2_COLL_INT 0x04
+#define AMR_DLC_DMR2_FCS_INT 0x08
+#define AMR_DLC_DMR2_OVFL_INT 0x10
+#define AMR_DLC_DMR2_UNFL_INT 0x20
+#define AMR_DLC_DMR2_OVRN_INT 0x40
+#define AMR_DLC_DMR2_UNRN_INT 0x80
+#define AMR_DLC_1_7 0x88
+#define AMR_DLC_DRCR 0x89
+#define AMR_DLC_RNGR1 0x8A
+#define AMR_DLC_RNGR2 0x8B
+#define AMR_DLC_FRAR4 0x8C
+#define AMR_DLC_SRAR4 0x8D
+#define AMR_DLC_DMR3 0x8E
+#define AMR_DLC_DMR3_VA_INT 0x01
+#define AMR_DLC_DMR3_EOTP_INT 0x02
+#define AMR_DLC_DMR3_LBRP_INT 0x04
+#define AMR_DLC_DMR3_RBA_INT 0x08
+#define AMR_DLC_DMR3_LBT_INT 0x10
+#define AMR_DLC_DMR3_TBE_INT 0x20
+#define AMR_DLC_DMR3_RPLOST_INT 0x40
+#define AMR_DLC_DMR3_KEEP_FCS 0x80
+#define AMR_DLC_DMR4 0x8F
+#define AMR_DLC_DMR4_RCV_1 0x00
+#define AMR_DLC_DMR4_RCV_2 0x01
+#define AMR_DLC_DMR4_RCV_4 0x02
+#define AMR_DLC_DMR4_RCV_8 0x03
+#define AMR_DLC_DMR4_RCV_16 0x01
+#define AMR_DLC_DMR4_RCV_24 0x02
+#define AMR_DLC_DMR4_RCV_30 0x03
+#define AMR_DLC_DMR4_XMT_1 0x00
+#define AMR_DLC_DMR4_XMT_2 0x04
+#define AMR_DLC_DMR4_XMT_4 0x08
+#define AMR_DLC_DMR4_XMT_8 0x0c
+#define AMR_DLC_DMR4_XMT_10 0x08
+#define AMR_DLC_DMR4_XMT_14 0x0c
+#define AMR_DLC_DMR4_IDLE_MARK 0x00
+#define AMR_DLC_DMR4_IDLE_FLAG 0x10
+#define AMR_DLC_DMR4_ADDR_BOTH 0x00
+#define AMR_DLC_DMR4_ADDR_1ST 0x20
+#define AMR_DLC_DMR4_ADDR_2ND 0xa0
+#define AMR_DLC_DMR4_CR_ENABLE 0x40
+#define AMR_DLC_12_15 0x90
+#define AMR_DLC_ASR 0x91
+#define AMR_DLC_EFCR 0x92
+#define AMR_DLC_EFCR_EXTEND_FIFO 0x01
+#define AMR_DLC_EFCR_SEC_PKT_INT 0x02
+
+#define AMR_DSR1_VADDR 0x01
+#define AMR_DSR1_EORP 0x02
+#define AMR_DSR1_PKT_IP 0x04
+#define AMR_DSR1_DECHO_ON 0x08
+#define AMR_DSR1_DLOOP_ON 0x10
+#define AMR_DSR1_DBACK_OFF 0x20
+#define AMR_DSR1_EOTP 0x40
+#define AMR_DSR1_CXMT_ABRT 0x80
+
+#define AMR_DSR2_LBRP 0x01
+#define AMR_DSR2_RBA 0x02
+#define AMR_DSR2_RPLOST 0x04
+#define AMR_DSR2_LAST_BYTE 0x08
+#define AMR_DSR2_TBE 0x10
+#define AMR_DSR2_MARK_IDLE 0x20
+#define AMR_DSR2_FLAG_IDLE 0x40
+#define AMR_DSR2_SECOND_PKT 0x80
+
+#define AMR_DER_RABRT 0x01
+#define AMR_DER_RFRAME 0x02
+#define AMR_DER_COLLISION 0x04
+#define AMR_DER_FCS 0x08
+#define AMR_DER_OVFL 0x10
+#define AMR_DER_UNFL 0x20
+#define AMR_DER_OVRN 0x40
+#define AMR_DER_UNRN 0x80
+
+/* Peripheral Port */
+#define AMR_PP_PPCR1 0xC0
+#define AMR_PP_PPSR 0xC1
+#define AMR_PP_PPIER 0xC2
+#define AMR_PP_MTDR 0xC3
+#define AMR_PP_MRDR 0xC3
+#define AMR_PP_CITDR0 0xC4
+#define AMR_PP_CIRDR0 0xC4
+#define AMR_PP_CITDR1 0xC5
+#define AMR_PP_CIRDR1 0xC5
+#define AMR_PP_PPCR2 0xC8
+#define AMR_PP_PPCR3 0xC9
+
+struct snd_amd7930 {
+ spinlock_t lock;
+ void __iomem *regs;
+ u32 flags;
+#define AMD7930_FLAG_PLAYBACK 0x00000001
+#define AMD7930_FLAG_CAPTURE 0x00000002
+
+ struct amd7930_map map;
+
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ /* Playback/Capture buffer state. */
+ unsigned char *p_orig, *p_cur;
+ int p_left;
+ unsigned char *c_orig, *c_cur;
+ int c_left;
+
+ int rgain;
+ int pgain;
+ int mgain;
+
+ struct of_device *op;
+ unsigned int irq;
+ struct snd_amd7930 *next;
+};
+
+static struct snd_amd7930 *amd7930_list;
+
+/* Idle the AMD7930 chip. The amd->lock is not held. */
+static __inline__ void amd7930_idle(struct snd_amd7930 *amd)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&amd->lock, flags);
+ sbus_writeb(AMR_INIT, amd->regs + AMD7930_CR);
+ sbus_writeb(0, amd->regs + AMD7930_DR);
+ spin_unlock_irqrestore(&amd->lock, flags);
+}
+
+/* Enable chip interrupts. The amd->lock is not held. */
+static __inline__ void amd7930_enable_ints(struct snd_amd7930 *amd)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&amd->lock, flags);
+ sbus_writeb(AMR_INIT, amd->regs + AMD7930_CR);
+ sbus_writeb(AM_INIT_ACTIVE, amd->regs + AMD7930_DR);
+ spin_unlock_irqrestore(&amd->lock, flags);
+}
+
+/* Disable chip interrupts. The amd->lock is not held. */
+static __inline__ void amd7930_disable_ints(struct snd_amd7930 *amd)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&amd->lock, flags);
+ sbus_writeb(AMR_INIT, amd->regs + AMD7930_CR);
+ sbus_writeb(AM_INIT_ACTIVE | AM_INIT_DISABLE_INTS, amd->regs + AMD7930_DR);
+ spin_unlock_irqrestore(&amd->lock, flags);
+}
+
+/* Commit amd7930_map settings to the hardware.
+ * The amd->lock is held and local interrupts are disabled.
+ */
+static void __amd7930_write_map(struct snd_amd7930 *amd)
+{
+ struct amd7930_map *map = &amd->map;
+
+ sbus_writeb(AMR_MAP_GX, amd->regs + AMD7930_CR);
+ sbus_writeb(((map->gx >> 0) & 0xff), amd->regs + AMD7930_DR);
+ sbus_writeb(((map->gx >> 8) & 0xff), amd->regs + AMD7930_DR);
+
+ sbus_writeb(AMR_MAP_GR, amd->regs + AMD7930_CR);
+ sbus_writeb(((map->gr >> 0) & 0xff), amd->regs + AMD7930_DR);
+ sbus_writeb(((map->gr >> 8) & 0xff), amd->regs + AMD7930_DR);
+
+ sbus_writeb(AMR_MAP_STGR, amd->regs + AMD7930_CR);
+ sbus_writeb(((map->stgr >> 0) & 0xff), amd->regs + AMD7930_DR);
+ sbus_writeb(((map->stgr >> 8) & 0xff), amd->regs + AMD7930_DR);
+
+ sbus_writeb(AMR_MAP_GER, amd->regs + AMD7930_CR);
+ sbus_writeb(((map->ger >> 0) & 0xff), amd->regs + AMD7930_DR);
+ sbus_writeb(((map->ger >> 8) & 0xff), amd->regs + AMD7930_DR);
+
+ sbus_writeb(AMR_MAP_MMR1, amd->regs + AMD7930_CR);
+ sbus_writeb(map->mmr1, amd->regs + AMD7930_DR);
+
+ sbus_writeb(AMR_MAP_MMR2, amd->regs + AMD7930_CR);
+ sbus_writeb(map->mmr2, amd->regs + AMD7930_DR);
+}
+
+/* gx, gr & stg gains. this table must contain 256 elements with
+ * the 0th being "infinity" (the magic value 9008). The remaining
+ * elements match sun's gain curve (but with higher resolution):
+ * -18 to 0dB in .16dB steps then 0 to 12dB in .08dB steps.
+ */
+static __const__ __u16 gx_coeff[256] = {
+ 0x9008, 0x8b7c, 0x8b51, 0x8b45, 0x8b42, 0x8b3b, 0x8b36, 0x8b33,
+ 0x8b32, 0x8b2a, 0x8b2b, 0x8b2c, 0x8b25, 0x8b23, 0x8b22, 0x8b22,
+ 0x9122, 0x8b1a, 0x8aa3, 0x8aa3, 0x8b1c, 0x8aa6, 0x912d, 0x912b,
+ 0x8aab, 0x8b12, 0x8aaa, 0x8ab2, 0x9132, 0x8ab4, 0x913c, 0x8abb,
+ 0x9142, 0x9144, 0x9151, 0x8ad5, 0x8aeb, 0x8a79, 0x8a5a, 0x8a4a,
+ 0x8b03, 0x91c2, 0x91bb, 0x8a3f, 0x8a33, 0x91b2, 0x9212, 0x9213,
+ 0x8a2c, 0x921d, 0x8a23, 0x921a, 0x9222, 0x9223, 0x922d, 0x9231,
+ 0x9234, 0x9242, 0x925b, 0x92dd, 0x92c1, 0x92b3, 0x92ab, 0x92a4,
+ 0x92a2, 0x932b, 0x9341, 0x93d3, 0x93b2, 0x93a2, 0x943c, 0x94b2,
+ 0x953a, 0x9653, 0x9782, 0x9e21, 0x9d23, 0x9cd2, 0x9c23, 0x9baa,
+ 0x9bde, 0x9b33, 0x9b22, 0x9b1d, 0x9ab2, 0xa142, 0xa1e5, 0x9a3b,
+ 0xa213, 0xa1a2, 0xa231, 0xa2eb, 0xa313, 0xa334, 0xa421, 0xa54b,
+ 0xada4, 0xac23, 0xab3b, 0xaaab, 0xaa5c, 0xb1a3, 0xb2ca, 0xb3bd,
+ 0xbe24, 0xbb2b, 0xba33, 0xc32b, 0xcb5a, 0xd2a2, 0xe31d, 0x0808,
+ 0x72ba, 0x62c2, 0x5c32, 0x52db, 0x513e, 0x4cce, 0x43b2, 0x4243,
+ 0x41b4, 0x3b12, 0x3bc3, 0x3df2, 0x34bd, 0x3334, 0x32c2, 0x3224,
+ 0x31aa, 0x2a7b, 0x2aaa, 0x2b23, 0x2bba, 0x2c42, 0x2e23, 0x25bb,
+ 0x242b, 0x240f, 0x231a, 0x22bb, 0x2241, 0x2223, 0x221f, 0x1a33,
+ 0x1a4a, 0x1acd, 0x2132, 0x1b1b, 0x1b2c, 0x1b62, 0x1c12, 0x1c32,
+ 0x1d1b, 0x1e71, 0x16b1, 0x1522, 0x1434, 0x1412, 0x1352, 0x1323,
+ 0x1315, 0x12bc, 0x127a, 0x1235, 0x1226, 0x11a2, 0x1216, 0x0a2a,
+ 0x11bc, 0x11d1, 0x1163, 0x0ac2, 0x0ab2, 0x0aab, 0x0b1b, 0x0b23,
+ 0x0b33, 0x0c0f, 0x0bb3, 0x0c1b, 0x0c3e, 0x0cb1, 0x0d4c, 0x0ec1,
+ 0x079a, 0x0614, 0x0521, 0x047c, 0x0422, 0x03b1, 0x03e3, 0x0333,
+ 0x0322, 0x031c, 0x02aa, 0x02ba, 0x02f2, 0x0242, 0x0232, 0x0227,
+ 0x0222, 0x021b, 0x01ad, 0x0212, 0x01b2, 0x01bb, 0x01cb, 0x01f6,
+ 0x0152, 0x013a, 0x0133, 0x0131, 0x012c, 0x0123, 0x0122, 0x00a2,
+ 0x011b, 0x011e, 0x0114, 0x00b1, 0x00aa, 0x00b3, 0x00bd, 0x00ba,
+ 0x00c5, 0x00d3, 0x00f3, 0x0062, 0x0051, 0x0042, 0x003b, 0x0033,
+ 0x0032, 0x002a, 0x002c, 0x0025, 0x0023, 0x0022, 0x001a, 0x0021,
+ 0x001b, 0x001b, 0x001d, 0x0015, 0x0013, 0x0013, 0x0012, 0x0012,
+ 0x000a, 0x000a, 0x0011, 0x0011, 0x000b, 0x000b, 0x000c, 0x000e,
+};
+
+static __const__ __u16 ger_coeff[] = {
+ 0x431f, /* 5. dB */
+ 0x331f, /* 5.5 dB */
+ 0x40dd, /* 6. dB */
+ 0x11dd, /* 6.5 dB */
+ 0x440f, /* 7. dB */
+ 0x411f, /* 7.5 dB */
+ 0x311f, /* 8. dB */
+ 0x5520, /* 8.5 dB */
+ 0x10dd, /* 9. dB */
+ 0x4211, /* 9.5 dB */
+ 0x410f, /* 10. dB */
+ 0x111f, /* 10.5 dB */
+ 0x600b, /* 11. dB */
+ 0x00dd, /* 11.5 dB */
+ 0x4210, /* 12. dB */
+ 0x110f, /* 13. dB */
+ 0x7200, /* 14. dB */
+ 0x2110, /* 15. dB */
+ 0x2200, /* 15.9 dB */
+ 0x000b, /* 16.9 dB */
+ 0x000f /* 18. dB */
+};
+
+/* Update amd7930_map settings and program them into the hardware.
+ * The amd->lock is held and local interrupts are disabled.
+ */
+static void __amd7930_update_map(struct snd_amd7930 *amd)
+{
+ struct amd7930_map *map = &amd->map;
+ int level;
+
+ map->gx = gx_coeff[amd->rgain];
+ map->stgr = gx_coeff[amd->mgain];
+ level = (amd->pgain * (256 + ARRAY_SIZE(ger_coeff))) >> 8;
+ if (level >= 256) {
+ map->ger = ger_coeff[level - 256];
+ map->gr = gx_coeff[255];
+ } else {
+ map->ger = ger_coeff[0];
+ map->gr = gx_coeff[level];
+ }
+ __amd7930_write_map(amd);
+}
+
+static irqreturn_t snd_amd7930_interrupt(int irq, void *dev_id)
+{
+ struct snd_amd7930 *amd = dev_id;
+ unsigned int elapsed;
+ u8 ir;
+
+ spin_lock(&amd->lock);
+
+ elapsed = 0;
+
+ ir = sbus_readb(amd->regs + AMD7930_IR);
+ if (ir & AMR_IR_BBUF) {
+ u8 byte;
+
+ if (amd->flags & AMD7930_FLAG_PLAYBACK) {
+ if (amd->p_left > 0) {
+ byte = *(amd->p_cur++);
+ amd->p_left--;
+ sbus_writeb(byte, amd->regs + AMD7930_BBTB);
+ if (amd->p_left == 0)
+ elapsed |= AMD7930_FLAG_PLAYBACK;
+ } else
+ sbus_writeb(0, amd->regs + AMD7930_BBTB);
+ } else if (amd->flags & AMD7930_FLAG_CAPTURE) {
+ byte = sbus_readb(amd->regs + AMD7930_BBRB);
+ if (amd->c_left > 0) {
+ *(amd->c_cur++) = byte;
+ amd->c_left--;
+ if (amd->c_left == 0)
+ elapsed |= AMD7930_FLAG_CAPTURE;
+ }
+ }
+ }
+ spin_unlock(&amd->lock);
+
+ if (elapsed & AMD7930_FLAG_PLAYBACK)
+ snd_pcm_period_elapsed(amd->playback_substream);
+ else
+ snd_pcm_period_elapsed(amd->capture_substream);
+
+ return IRQ_HANDLED;
+}
+
+static int snd_amd7930_trigger(struct snd_amd7930 *amd, unsigned int flag, int cmd)
+{
+ unsigned long flags;
+ int result = 0;
+
+ spin_lock_irqsave(&amd->lock, flags);
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ if (!(amd->flags & flag)) {
+ amd->flags |= flag;
+
+ /* Enable B channel interrupts. */
+ sbus_writeb(AMR_MUX_MCR4, amd->regs + AMD7930_CR);
+ sbus_writeb(AM_MUX_MCR4_ENABLE_INTS, amd->regs + AMD7930_DR);
+ }
+ } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+ if (amd->flags & flag) {
+ amd->flags &= ~flag;
+
+ /* Disable B channel interrupts. */
+ sbus_writeb(AMR_MUX_MCR4, amd->regs + AMD7930_CR);
+ sbus_writeb(0, amd->regs + AMD7930_DR);
+ }
+ } else {
+ result = -EINVAL;
+ }
+ spin_unlock_irqrestore(&amd->lock, flags);
+
+ return result;
+}
+
+static int snd_amd7930_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ return snd_amd7930_trigger(amd, AMD7930_FLAG_PLAYBACK, cmd);
+}
+
+static int snd_amd7930_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ return snd_amd7930_trigger(amd, AMD7930_FLAG_CAPTURE, cmd);
+}
+
+static int snd_amd7930_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned long flags;
+ u8 new_mmr1;
+
+ spin_lock_irqsave(&amd->lock, flags);
+
+ amd->flags |= AMD7930_FLAG_PLAYBACK;
+
+ /* Setup the pseudo-dma transfer pointers. */
+ amd->p_orig = amd->p_cur = runtime->dma_area;
+ amd->p_left = size;
+
+ /* Put the chip into the correct encoding format. */
+ new_mmr1 = amd->map.mmr1;
+ if (runtime->format == SNDRV_PCM_FORMAT_A_LAW)
+ new_mmr1 |= AM_MAP_MMR1_ALAW;
+ else
+ new_mmr1 &= ~AM_MAP_MMR1_ALAW;
+ if (new_mmr1 != amd->map.mmr1) {
+ amd->map.mmr1 = new_mmr1;
+ __amd7930_update_map(amd);
+ }
+
+ spin_unlock_irqrestore(&amd->lock, flags);
+
+ return 0;
+}
+
+static int snd_amd7930_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int size = snd_pcm_lib_buffer_bytes(substream);
+ unsigned long flags;
+ u8 new_mmr1;
+
+ spin_lock_irqsave(&amd->lock, flags);
+
+ amd->flags |= AMD7930_FLAG_CAPTURE;
+
+ /* Setup the pseudo-dma transfer pointers. */
+ amd->c_orig = amd->c_cur = runtime->dma_area;
+ amd->c_left = size;
+
+ /* Put the chip into the correct encoding format. */
+ new_mmr1 = amd->map.mmr1;
+ if (runtime->format == SNDRV_PCM_FORMAT_A_LAW)
+ new_mmr1 |= AM_MAP_MMR1_ALAW;
+ else
+ new_mmr1 &= ~AM_MAP_MMR1_ALAW;
+ if (new_mmr1 != amd->map.mmr1) {
+ amd->map.mmr1 = new_mmr1;
+ __amd7930_update_map(amd);
+ }
+
+ spin_unlock_irqrestore(&amd->lock, flags);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t snd_amd7930_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(amd->flags & AMD7930_FLAG_PLAYBACK))
+ return 0;
+ ptr = amd->p_cur - amd->p_orig;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_amd7930_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ size_t ptr;
+
+ if (!(amd->flags & AMD7930_FLAG_CAPTURE))
+ return 0;
+
+ ptr = amd->c_cur - amd->c_orig;
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+/* Playback and capture have identical properties. */
+static struct snd_pcm_hardware snd_amd7930_pcm_hw =
+{
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_HALF_DUPLEX),
+ .formats = SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW,
+ .rates = SNDRV_PCM_RATE_8000,
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = (64*1024),
+ .period_bytes_min = 1,
+ .period_bytes_max = (64*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static int snd_amd7930_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ amd->playback_substream = substream;
+ runtime->hw = snd_amd7930_pcm_hw;
+ return 0;
+}
+
+static int snd_amd7930_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ amd->capture_substream = substream;
+ runtime->hw = snd_amd7930_pcm_hw;
+ return 0;
+}
+
+static int snd_amd7930_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+
+ amd->playback_substream = NULL;
+ return 0;
+}
+
+static int snd_amd7930_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_amd7930 *amd = snd_pcm_substream_chip(substream);
+
+ amd->capture_substream = NULL;
+ return 0;
+}
+
+static int snd_amd7930_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int snd_amd7930_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_pcm_ops snd_amd7930_playback_ops = {
+ .open = snd_amd7930_playback_open,
+ .close = snd_amd7930_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_amd7930_hw_params,
+ .hw_free = snd_amd7930_hw_free,
+ .prepare = snd_amd7930_playback_prepare,
+ .trigger = snd_amd7930_playback_trigger,
+ .pointer = snd_amd7930_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_amd7930_capture_ops = {
+ .open = snd_amd7930_capture_open,
+ .close = snd_amd7930_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_amd7930_hw_params,
+ .hw_free = snd_amd7930_hw_free,
+ .prepare = snd_amd7930_capture_prepare,
+ .trigger = snd_amd7930_capture_trigger,
+ .pointer = snd_amd7930_capture_pointer,
+};
+
+static int __devinit snd_amd7930_pcm(struct snd_amd7930 *amd)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(amd->card,
+ /* ID */ "sun_amd7930",
+ /* device */ 0,
+ /* playback count */ 1,
+ /* capture count */ 1, &pcm)) < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_amd7930_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_amd7930_capture_ops);
+
+ pcm->private_data = amd;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, amd->card->shortname);
+ amd->pcm = pcm;
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 64*1024, 64*1024);
+
+ return 0;
+}
+
+#define VOLUME_MONITOR 0
+#define VOLUME_CAPTURE 1
+#define VOLUME_PLAYBACK 2
+
+static int snd_amd7930_info_volume(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+
+ return 0;
+}
+
+static int snd_amd7930_get_volume(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_amd7930 *amd = snd_kcontrol_chip(kctl);
+ int type = kctl->private_value;
+ int *swval;
+
+ switch (type) {
+ case VOLUME_MONITOR:
+ swval = &amd->mgain;
+ break;
+ case VOLUME_CAPTURE:
+ swval = &amd->rgain;
+ break;
+ case VOLUME_PLAYBACK:
+ default:
+ swval = &amd->pgain;
+ break;
+ };
+
+ ucontrol->value.integer.value[0] = *swval;
+
+ return 0;
+}
+
+static int snd_amd7930_put_volume(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_amd7930 *amd = snd_kcontrol_chip(kctl);
+ unsigned long flags;
+ int type = kctl->private_value;
+ int *swval, change;
+
+ switch (type) {
+ case VOLUME_MONITOR:
+ swval = &amd->mgain;
+ break;
+ case VOLUME_CAPTURE:
+ swval = &amd->rgain;
+ break;
+ case VOLUME_PLAYBACK:
+ default:
+ swval = &amd->pgain;
+ break;
+ };
+
+ spin_lock_irqsave(&amd->lock, flags);
+
+ if (*swval != ucontrol->value.integer.value[0]) {
+ *swval = ucontrol->value.integer.value[0] & 0xff;
+ __amd7930_update_map(amd);
+ change = 1;
+ } else
+ change = 0;
+
+ spin_unlock_irqrestore(&amd->lock, flags);
+
+ return change;
+}
+
+static struct snd_kcontrol_new amd7930_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Volume",
+ .index = 0,
+ .info = snd_amd7930_info_volume,
+ .get = snd_amd7930_get_volume,
+ .put = snd_amd7930_put_volume,
+ .private_value = VOLUME_MONITOR,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Volume",
+ .index = 0,
+ .info = snd_amd7930_info_volume,
+ .get = snd_amd7930_get_volume,
+ .put = snd_amd7930_put_volume,
+ .private_value = VOLUME_CAPTURE,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Playback Volume",
+ .index = 0,
+ .info = snd_amd7930_info_volume,
+ .get = snd_amd7930_get_volume,
+ .put = snd_amd7930_put_volume,
+ .private_value = VOLUME_PLAYBACK,
+ },
+};
+
+static int __devinit snd_amd7930_mixer(struct snd_amd7930 *amd)
+{
+ struct snd_card *card;
+ int idx, err;
+
+ if (snd_BUG_ON(!amd || !amd->card))
+ return -EINVAL;
+
+ card = amd->card;
+ strcpy(card->mixername, card->shortname);
+
+ for (idx = 0; idx < ARRAY_SIZE(amd7930_controls); idx++) {
+ if ((err = snd_ctl_add(card,
+ snd_ctl_new1(&amd7930_controls[idx], amd))) < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static int snd_amd7930_free(struct snd_amd7930 *amd)
+{
+ struct of_device *op = amd->op;
+
+ amd7930_idle(amd);
+
+ if (amd->irq)
+ free_irq(amd->irq, amd);
+
+ if (amd->regs)
+ of_iounmap(&op->resource[0], amd->regs,
+ resource_size(&op->resource[0]));
+
+ kfree(amd);
+
+ return 0;
+}
+
+static int snd_amd7930_dev_free(struct snd_device *device)
+{
+ struct snd_amd7930 *amd = device->device_data;
+
+ return snd_amd7930_free(amd);
+}
+
+static struct snd_device_ops snd_amd7930_dev_ops = {
+ .dev_free = snd_amd7930_dev_free,
+};
+
+static int __devinit snd_amd7930_create(struct snd_card *card,
+ struct of_device *op,
+ int irq, int dev,
+ struct snd_amd7930 **ramd)
+{
+ struct snd_amd7930 *amd;
+ unsigned long flags;
+ int err;
+
+ *ramd = NULL;
+ amd = kzalloc(sizeof(*amd), GFP_KERNEL);
+ if (amd == NULL)
+ return -ENOMEM;
+
+ spin_lock_init(&amd->lock);
+ amd->card = card;
+ amd->op = op;
+
+ amd->regs = of_ioremap(&op->resource[0], 0,
+ resource_size(&op->resource[0]), "amd7930");
+ if (!amd->regs) {
+ snd_printk("amd7930-%d: Unable to map chip registers.\n", dev);
+ return -EIO;
+ }
+
+ amd7930_idle(amd);
+
+ if (request_irq(irq, snd_amd7930_interrupt,
+ IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) {
+ snd_printk("amd7930-%d: Unable to grab IRQ %d\n",
+ dev, irq);
+ snd_amd7930_free(amd);
+ return -EBUSY;
+ }
+ amd->irq = irq;
+
+ amd7930_enable_ints(amd);
+
+ spin_lock_irqsave(&amd->lock, flags);
+
+ amd->rgain = 128;
+ amd->pgain = 200;
+ amd->mgain = 0;
+
+ memset(&amd->map, 0, sizeof(amd->map));
+ amd->map.mmr1 = (AM_MAP_MMR1_GX | AM_MAP_MMR1_GER |
+ AM_MAP_MMR1_GR | AM_MAP_MMR1_STG);
+ amd->map.mmr2 = (AM_MAP_MMR2_LS | AM_MAP_MMR2_AINB);
+
+ __amd7930_update_map(amd);
+
+ /* Always MUX audio (Ba) to channel Bb. */
+ sbus_writeb(AMR_MUX_MCR1, amd->regs + AMD7930_CR);
+ sbus_writeb(AM_MUX_CHANNEL_Ba | (AM_MUX_CHANNEL_Bb << 4),
+ amd->regs + AMD7930_DR);
+
+ spin_unlock_irqrestore(&amd->lock, flags);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ amd, &snd_amd7930_dev_ops)) < 0) {
+ snd_amd7930_free(amd);
+ return err;
+ }
+
+ *ramd = amd;
+ return 0;
+}
+
+static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_device_id *match)
+{
+ struct resource *rp = &op->resource[0];
+ static int dev_num;
+ struct snd_card *card;
+ struct snd_amd7930 *amd;
+ int err, irq;
+
+ irq = op->irqs[0];
+
+ if (dev_num >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev_num]) {
+ dev_num++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "AMD7930");
+ strcpy(card->shortname, "Sun AMD7930");
+ sprintf(card->longname, "%s at 0x%02lx:0x%08Lx, irq %d",
+ card->shortname,
+ rp->flags & 0xffL,
+ (unsigned long long)rp->start,
+ irq);
+
+ if ((err = snd_amd7930_create(card, op,
+ irq, dev_num, &amd)) < 0)
+ goto out_err;
+
+ if ((err = snd_amd7930_pcm(amd)) < 0)
+ goto out_err;
+
+ if ((err = snd_amd7930_mixer(amd)) < 0)
+ goto out_err;
+
+ if ((err = snd_card_register(card)) < 0)
+ goto out_err;
+
+ amd->next = amd7930_list;
+ amd7930_list = amd;
+
+ dev_num++;
+
+ return 0;
+
+out_err:
+ snd_card_free(card);
+ return err;
+}
+
+static const struct of_device_id amd7930_match[] = {
+ {
+ .name = "audio",
+ },
+ {},
+};
+
+static struct of_platform_driver amd7930_sbus_driver = {
+ .name = "audio",
+ .match_table = amd7930_match,
+ .probe = amd7930_sbus_probe,
+};
+
+static int __init amd7930_init(void)
+{
+ return of_register_driver(&amd7930_sbus_driver, &of_bus_type);
+}
+
+static void __exit amd7930_exit(void)
+{
+ struct snd_amd7930 *p = amd7930_list;
+
+ while (p != NULL) {
+ struct snd_amd7930 *next = p->next;
+
+ snd_card_free(p->card);
+
+ p = next;
+ }
+
+ amd7930_list = NULL;
+
+ of_unregister_driver(&amd7930_sbus_driver);
+}
+
+module_init(amd7930_init);
+module_exit(amd7930_exit);
diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c
new file mode 100644
index 0000000..d44bf98
--- /dev/null
+++ b/sound/sparc/cs4231.c
@@ -0,0 +1,2129 @@
+/*
+ * Driver for CS4231 sound chips found on Sparcs.
+ * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net>
+ *
+ * Based entirely upon drivers/sbus/audio/cs4231.c which is:
+ * Copyright (C) 1996, 1997, 1998 Derrick J Brashear (shadow@andrew.cmu.edu)
+ * and also sound/isa/cs423x/cs4231_lib.c which is:
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/irq.h>
+#include <linux/io.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/timer.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+
+#ifdef CONFIG_SBUS
+#define SBUS_SUPPORT
+#endif
+
+#if defined(CONFIG_PCI) && defined(CONFIG_SPARC64)
+#define EBUS_SUPPORT
+#include <linux/pci.h>
+#include <asm/ebus_dma.h>
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+/* Enable this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Sun CS4231 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Sun CS4231 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Sun CS4231 soundcard.");
+MODULE_AUTHOR("Jaroslav Kysela, Derrick J. Brashear and David S. Miller");
+MODULE_DESCRIPTION("Sun CS4231");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Sun,CS4231}}");
+
+#ifdef SBUS_SUPPORT
+struct sbus_dma_info {
+ spinlock_t lock; /* DMA access lock */
+ int dir;
+ void __iomem *regs;
+};
+#endif
+
+struct snd_cs4231;
+struct cs4231_dma_control {
+ void (*prepare)(struct cs4231_dma_control *dma_cont,
+ int dir);
+ void (*enable)(struct cs4231_dma_control *dma_cont, int on);
+ int (*request)(struct cs4231_dma_control *dma_cont,
+ dma_addr_t bus_addr, size_t len);
+ unsigned int (*address)(struct cs4231_dma_control *dma_cont);
+#ifdef EBUS_SUPPORT
+ struct ebus_dma_info ebus_info;
+#endif
+#ifdef SBUS_SUPPORT
+ struct sbus_dma_info sbus_info;
+#endif
+};
+
+struct snd_cs4231 {
+ spinlock_t lock; /* registers access lock */
+ void __iomem *port;
+
+ struct cs4231_dma_control p_dma;
+ struct cs4231_dma_control c_dma;
+
+ u32 flags;
+#define CS4231_FLAG_EBUS 0x00000001
+#define CS4231_FLAG_PLAYBACK 0x00000002
+#define CS4231_FLAG_CAPTURE 0x00000004
+
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *playback_substream;
+ unsigned int p_periods_sent;
+ struct snd_pcm_substream *capture_substream;
+ unsigned int c_periods_sent;
+ struct snd_timer *timer;
+
+ unsigned short mode;
+#define CS4231_MODE_NONE 0x0000
+#define CS4231_MODE_PLAY 0x0001
+#define CS4231_MODE_RECORD 0x0002
+#define CS4231_MODE_TIMER 0x0004
+#define CS4231_MODE_OPEN (CS4231_MODE_PLAY | CS4231_MODE_RECORD | \
+ CS4231_MODE_TIMER)
+
+ unsigned char image[32]; /* registers image */
+ int mce_bit;
+ int calibrate_mute;
+ struct mutex mce_mutex; /* mutex for mce register */
+ struct mutex open_mutex; /* mutex for ALSA open/close */
+
+ struct of_device *op;
+ unsigned int irq[2];
+ unsigned int regs_size;
+ struct snd_cs4231 *next;
+};
+
+/* Eventually we can use sound/isa/cs423x/cs4231_lib.c directly, but for
+ * now.... -DaveM
+ */
+
+/* IO ports */
+#include <sound/cs4231-regs.h>
+
+/* XXX offsets are different than PC ISA chips... */
+#define CS4231U(chip, x) ((chip)->port + ((c_d_c_CS4231##x) << 2))
+
+/* SBUS DMA register defines. */
+
+#define APCCSR 0x10UL /* APC DMA CSR */
+#define APCCVA 0x20UL /* APC Capture DMA Address */
+#define APCCC 0x24UL /* APC Capture Count */
+#define APCCNVA 0x28UL /* APC Capture DMA Next Address */
+#define APCCNC 0x2cUL /* APC Capture Next Count */
+#define APCPVA 0x30UL /* APC Play DMA Address */
+#define APCPC 0x34UL /* APC Play Count */
+#define APCPNVA 0x38UL /* APC Play DMA Next Address */
+#define APCPNC 0x3cUL /* APC Play Next Count */
+
+/* Defines for SBUS DMA-routines */
+
+#define APCVA 0x0UL /* APC DMA Address */
+#define APCC 0x4UL /* APC Count */
+#define APCNVA 0x8UL /* APC DMA Next Address */
+#define APCNC 0xcUL /* APC Next Count */
+#define APC_PLAY 0x30UL /* Play registers start at 0x30 */
+#define APC_RECORD 0x20UL /* Record registers start at 0x20 */
+
+/* APCCSR bits */
+
+#define APC_INT_PENDING 0x800000 /* Interrupt Pending */
+#define APC_PLAY_INT 0x400000 /* Playback interrupt */
+#define APC_CAPT_INT 0x200000 /* Capture interrupt */
+#define APC_GENL_INT 0x100000 /* General interrupt */
+#define APC_XINT_ENA 0x80000 /* General ext int. enable */
+#define APC_XINT_PLAY 0x40000 /* Playback ext intr */
+#define APC_XINT_CAPT 0x20000 /* Capture ext intr */
+#define APC_XINT_GENL 0x10000 /* Error ext intr */
+#define APC_XINT_EMPT 0x8000 /* Pipe empty interrupt (0 write to pva) */
+#define APC_XINT_PEMP 0x4000 /* Play pipe empty (pva and pnva not set) */
+#define APC_XINT_PNVA 0x2000 /* Playback NVA dirty */
+#define APC_XINT_PENA 0x1000 /* play pipe empty Int enable */
+#define APC_XINT_COVF 0x800 /* Cap data dropped on floor */
+#define APC_XINT_CNVA 0x400 /* Capture NVA dirty */
+#define APC_XINT_CEMP 0x200 /* Capture pipe empty (cva and cnva not set) */
+#define APC_XINT_CENA 0x100 /* Cap. pipe empty int enable */
+#define APC_PPAUSE 0x80 /* Pause the play DMA */
+#define APC_CPAUSE 0x40 /* Pause the capture DMA */
+#define APC_CDC_RESET 0x20 /* CODEC RESET */
+#define APC_PDMA_READY 0x08 /* Play DMA Go */
+#define APC_CDMA_READY 0x04 /* Capture DMA Go */
+#define APC_CHIP_RESET 0x01 /* Reset the chip */
+
+/* EBUS DMA register offsets */
+
+#define EBDMA_CSR 0x00UL /* Control/Status */
+#define EBDMA_ADDR 0x04UL /* DMA Address */
+#define EBDMA_COUNT 0x08UL /* DMA Count */
+
+/*
+ * Some variables
+ */
+
+static unsigned char freq_bits[14] = {
+ /* 5510 */ 0x00 | CS4231_XTAL2,
+ /* 6620 */ 0x0E | CS4231_XTAL2,
+ /* 8000 */ 0x00 | CS4231_XTAL1,
+ /* 9600 */ 0x0E | CS4231_XTAL1,
+ /* 11025 */ 0x02 | CS4231_XTAL2,
+ /* 16000 */ 0x02 | CS4231_XTAL1,
+ /* 18900 */ 0x04 | CS4231_XTAL2,
+ /* 22050 */ 0x06 | CS4231_XTAL2,
+ /* 27042 */ 0x04 | CS4231_XTAL1,
+ /* 32000 */ 0x06 | CS4231_XTAL1,
+ /* 33075 */ 0x0C | CS4231_XTAL2,
+ /* 37800 */ 0x08 | CS4231_XTAL2,
+ /* 44100 */ 0x0A | CS4231_XTAL2,
+ /* 48000 */ 0x0C | CS4231_XTAL1
+};
+
+static unsigned int rates[14] = {
+ 5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050,
+ 27042, 32000, 33075, 37800, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+};
+
+static int snd_cs4231_xrate(struct snd_pcm_runtime *runtime)
+{
+ return snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &hw_constraints_rates);
+}
+
+static unsigned char snd_cs4231_original_image[32] =
+{
+ 0x00, /* 00/00 - lic */
+ 0x00, /* 01/01 - ric */
+ 0x9f, /* 02/02 - la1ic */
+ 0x9f, /* 03/03 - ra1ic */
+ 0x9f, /* 04/04 - la2ic */
+ 0x9f, /* 05/05 - ra2ic */
+ 0xbf, /* 06/06 - loc */
+ 0xbf, /* 07/07 - roc */
+ 0x20, /* 08/08 - pdfr */
+ CS4231_AUTOCALIB, /* 09/09 - ic */
+ 0x00, /* 0a/10 - pc */
+ 0x00, /* 0b/11 - ti */
+ CS4231_MODE2, /* 0c/12 - mi */
+ 0x00, /* 0d/13 - lbc */
+ 0x00, /* 0e/14 - pbru */
+ 0x00, /* 0f/15 - pbrl */
+ 0x80, /* 10/16 - afei */
+ 0x01, /* 11/17 - afeii */
+ 0x9f, /* 12/18 - llic */
+ 0x9f, /* 13/19 - rlic */
+ 0x00, /* 14/20 - tlb */
+ 0x00, /* 15/21 - thb */
+ 0x00, /* 16/22 - la3mic/reserved */
+ 0x00, /* 17/23 - ra3mic/reserved */
+ 0x00, /* 18/24 - afs */
+ 0x00, /* 19/25 - lamoc/version */
+ 0x00, /* 1a/26 - mioc */
+ 0x00, /* 1b/27 - ramoc/reserved */
+ 0x20, /* 1c/28 - cdfr */
+ 0x00, /* 1d/29 - res4 */
+ 0x00, /* 1e/30 - cbru */
+ 0x00, /* 1f/31 - cbrl */
+};
+
+static u8 __cs4231_readb(struct snd_cs4231 *cp, void __iomem *reg_addr)
+{
+ if (cp->flags & CS4231_FLAG_EBUS)
+ return readb(reg_addr);
+ else
+ return sbus_readb(reg_addr);
+}
+
+static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val,
+ void __iomem *reg_addr)
+{
+ if (cp->flags & CS4231_FLAG_EBUS)
+ return writeb(val, reg_addr);
+ else
+ return sbus_writeb(val, reg_addr);
+}
+
+/*
+ * Basic I/O functions
+ */
+
+static void snd_cs4231_ready(struct snd_cs4231 *chip)
+{
+ int timeout;
+
+ for (timeout = 250; timeout > 0; timeout--) {
+ int val = __cs4231_readb(chip, CS4231U(chip, REGSEL));
+ if ((val & CS4231_INIT) == 0)
+ break;
+ udelay(100);
+ }
+}
+
+static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg,
+ unsigned char value)
+{
+ snd_cs4231_ready(chip);
+#ifdef CONFIG_SND_DEBUG
+ if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT)
+ snd_printdd("out: auto calibration time out - reg = 0x%x, "
+ "value = 0x%x\n",
+ reg, value);
+#endif
+ __cs4231_writeb(chip, chip->mce_bit | reg, CS4231U(chip, REGSEL));
+ wmb();
+ __cs4231_writeb(chip, value, CS4231U(chip, REG));
+ mb();
+}
+
+static inline void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg,
+ unsigned char mask, unsigned char value)
+{
+ unsigned char tmp = (chip->image[reg] & mask) | value;
+
+ chip->image[reg] = tmp;
+ if (!chip->calibrate_mute)
+ snd_cs4231_dout(chip, reg, tmp);
+}
+
+static void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg,
+ unsigned char value)
+{
+ snd_cs4231_dout(chip, reg, value);
+ chip->image[reg] = value;
+ mb();
+}
+
+static unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg)
+{
+ snd_cs4231_ready(chip);
+#ifdef CONFIG_SND_DEBUG
+ if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT)
+ snd_printdd("in: auto calibration time out - reg = 0x%x\n",
+ reg);
+#endif
+ __cs4231_writeb(chip, chip->mce_bit | reg, CS4231U(chip, REGSEL));
+ mb();
+ return __cs4231_readb(chip, CS4231U(chip, REG));
+}
+
+/*
+ * CS4231 detection / MCE routines
+ */
+
+static void snd_cs4231_busy_wait(struct snd_cs4231 *chip)
+{
+ int timeout;
+
+ /* looks like this sequence is proper for CS4231A chip (GUS MAX) */
+ for (timeout = 5; timeout > 0; timeout--)
+ __cs4231_readb(chip, CS4231U(chip, REGSEL));
+
+ /* end of cleanup sequence */
+ for (timeout = 500; timeout > 0; timeout--) {
+ int val = __cs4231_readb(chip, CS4231U(chip, REGSEL));
+ if ((val & CS4231_INIT) == 0)
+ break;
+ msleep(1);
+ }
+}
+
+static void snd_cs4231_mce_up(struct snd_cs4231 *chip)
+{
+ unsigned long flags;
+ int timeout;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_ready(chip);
+#ifdef CONFIG_SND_DEBUG
+ if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT)
+ snd_printdd("mce_up - auto calibration time out (0)\n");
+#endif
+ chip->mce_bit |= CS4231_MCE;
+ timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL));
+ if (timeout == 0x80)
+ snd_printdd("mce_up [%p]: serious init problem - "
+ "codec still busy\n",
+ chip->port);
+ if (!(timeout & CS4231_MCE))
+ __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f),
+ CS4231U(chip, REGSEL));
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+static void snd_cs4231_mce_down(struct snd_cs4231 *chip)
+{
+ unsigned long flags, timeout;
+ int reg;
+
+ snd_cs4231_busy_wait(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+#ifdef CONFIG_SND_DEBUG
+ if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT)
+ snd_printdd("mce_down [%p] - auto calibration time out (0)\n",
+ CS4231U(chip, REGSEL));
+#endif
+ chip->mce_bit &= ~CS4231_MCE;
+ reg = __cs4231_readb(chip, CS4231U(chip, REGSEL));
+ __cs4231_writeb(chip, chip->mce_bit | (reg & 0x1f),
+ CS4231U(chip, REGSEL));
+ if (reg == 0x80)
+ snd_printdd("mce_down [%p]: serious init problem "
+ "- codec still busy\n", chip->port);
+ if ((reg & CS4231_MCE) == 0) {
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return;
+ }
+
+ /*
+ * Wait for auto-calibration (AC) process to finish, i.e. ACI to go low.
+ */
+ timeout = jiffies + msecs_to_jiffies(250);
+ do {
+ spin_unlock_irqrestore(&chip->lock, flags);
+ msleep(1);
+ spin_lock_irqsave(&chip->lock, flags);
+ reg = snd_cs4231_in(chip, CS4231_TEST_INIT);
+ reg &= CS4231_CALIB_IN_PROGRESS;
+ } while (reg && time_before(jiffies, timeout));
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ if (reg)
+ snd_printk(KERN_ERR
+ "mce_down - auto calibration time out (2)\n");
+}
+
+static void snd_cs4231_advance_dma(struct cs4231_dma_control *dma_cont,
+ struct snd_pcm_substream *substream,
+ unsigned int *periods_sent)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ while (1) {
+ unsigned int period_size = snd_pcm_lib_period_bytes(substream);
+ unsigned int offset = period_size * (*periods_sent);
+
+ BUG_ON(period_size >= (1 << 24));
+
+ if (dma_cont->request(dma_cont,
+ runtime->dma_addr + offset, period_size))
+ return;
+ (*periods_sent) = ((*periods_sent) + 1) % runtime->periods;
+ }
+}
+
+static void cs4231_dma_trigger(struct snd_pcm_substream *substream,
+ unsigned int what, int on)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ struct cs4231_dma_control *dma_cont;
+
+ if (what & CS4231_PLAYBACK_ENABLE) {
+ dma_cont = &chip->p_dma;
+ if (on) {
+ dma_cont->prepare(dma_cont, 0);
+ dma_cont->enable(dma_cont, 1);
+ snd_cs4231_advance_dma(dma_cont,
+ chip->playback_substream,
+ &chip->p_periods_sent);
+ } else {
+ dma_cont->enable(dma_cont, 0);
+ }
+ }
+ if (what & CS4231_RECORD_ENABLE) {
+ dma_cont = &chip->c_dma;
+ if (on) {
+ dma_cont->prepare(dma_cont, 1);
+ dma_cont->enable(dma_cont, 1);
+ snd_cs4231_advance_dma(dma_cont,
+ chip->capture_substream,
+ &chip->c_periods_sent);
+ } else {
+ dma_cont->enable(dma_cont, 0);
+ }
+ }
+}
+
+static int snd_cs4231_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ int result = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_STOP:
+ {
+ unsigned int what = 0;
+ struct snd_pcm_substream *s;
+ unsigned long flags;
+
+ snd_pcm_group_for_each_entry(s, substream) {
+ if (s == chip->playback_substream) {
+ what |= CS4231_PLAYBACK_ENABLE;
+ snd_pcm_trigger_done(s, substream);
+ } else if (s == chip->capture_substream) {
+ what |= CS4231_RECORD_ENABLE;
+ snd_pcm_trigger_done(s, substream);
+ }
+ }
+
+ spin_lock_irqsave(&chip->lock, flags);
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ cs4231_dma_trigger(substream, what, 1);
+ chip->image[CS4231_IFACE_CTRL] |= what;
+ } else {
+ cs4231_dma_trigger(substream, what, 0);
+ chip->image[CS4231_IFACE_CTRL] &= ~what;
+ }
+ snd_cs4231_out(chip, CS4231_IFACE_CTRL,
+ chip->image[CS4231_IFACE_CTRL]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ break;
+ }
+ default:
+ result = -EINVAL;
+ break;
+ }
+
+ return result;
+}
+
+/*
+ * CODEC I/O
+ */
+
+static unsigned char snd_cs4231_get_rate(unsigned int rate)
+{
+ int i;
+
+ for (i = 0; i < 14; i++)
+ if (rate == rates[i])
+ return freq_bits[i];
+
+ return freq_bits[13];
+}
+
+static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip, int format,
+ int channels)
+{
+ unsigned char rformat;
+
+ rformat = CS4231_LINEAR_8;
+ switch (format) {
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ rformat = CS4231_ULAW_8;
+ break;
+ case SNDRV_PCM_FORMAT_A_LAW:
+ rformat = CS4231_ALAW_8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ rformat = CS4231_LINEAR_16;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ rformat = CS4231_LINEAR_16_BIG;
+ break;
+ case SNDRV_PCM_FORMAT_IMA_ADPCM:
+ rformat = CS4231_ADPCM_16;
+ break;
+ }
+ if (channels > 1)
+ rformat |= CS4231_STEREO;
+ return rformat;
+}
+
+static void snd_cs4231_calibrate_mute(struct snd_cs4231 *chip, int mute)
+{
+ unsigned long flags;
+
+ mute = mute ? 1 : 0;
+ spin_lock_irqsave(&chip->lock, flags);
+ if (chip->calibrate_mute == mute) {
+ spin_unlock_irqrestore(&chip->lock, flags);
+ return;
+ }
+ if (!mute) {
+ snd_cs4231_dout(chip, CS4231_LEFT_INPUT,
+ chip->image[CS4231_LEFT_INPUT]);
+ snd_cs4231_dout(chip, CS4231_RIGHT_INPUT,
+ chip->image[CS4231_RIGHT_INPUT]);
+ snd_cs4231_dout(chip, CS4231_LOOPBACK,
+ chip->image[CS4231_LOOPBACK]);
+ }
+ snd_cs4231_dout(chip, CS4231_AUX1_LEFT_INPUT,
+ mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]);
+ snd_cs4231_dout(chip, CS4231_AUX1_RIGHT_INPUT,
+ mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]);
+ snd_cs4231_dout(chip, CS4231_AUX2_LEFT_INPUT,
+ mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]);
+ snd_cs4231_dout(chip, CS4231_AUX2_RIGHT_INPUT,
+ mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]);
+ snd_cs4231_dout(chip, CS4231_LEFT_OUTPUT,
+ mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]);
+ snd_cs4231_dout(chip, CS4231_RIGHT_OUTPUT,
+ mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]);
+ snd_cs4231_dout(chip, CS4231_LEFT_LINE_IN,
+ mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]);
+ snd_cs4231_dout(chip, CS4231_RIGHT_LINE_IN,
+ mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]);
+ snd_cs4231_dout(chip, CS4231_MONO_CTRL,
+ mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]);
+ chip->calibrate_mute = mute;
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+static void snd_cs4231_playback_format(struct snd_cs4231 *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char pdfr)
+{
+ unsigned long flags;
+
+ mutex_lock(&chip->mce_mutex);
+ snd_cs4231_calibrate_mute(chip, 1);
+
+ snd_cs4231_mce_up(chip);
+
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT,
+ (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ?
+ (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) :
+ pdfr);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ snd_cs4231_mce_down(chip);
+
+ snd_cs4231_calibrate_mute(chip, 0);
+ mutex_unlock(&chip->mce_mutex);
+}
+
+static void snd_cs4231_capture_format(struct snd_cs4231 *chip,
+ struct snd_pcm_hw_params *params,
+ unsigned char cdfr)
+{
+ unsigned long flags;
+
+ mutex_lock(&chip->mce_mutex);
+ snd_cs4231_calibrate_mute(chip, 1);
+
+ snd_cs4231_mce_up(chip);
+
+ spin_lock_irqsave(&chip->lock, flags);
+ if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) {
+ snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT,
+ ((chip->image[CS4231_PLAYBK_FORMAT]) & 0xf0) |
+ (cdfr & 0x0f));
+ spin_unlock_irqrestore(&chip->lock, flags);
+ snd_cs4231_mce_down(chip);
+ snd_cs4231_mce_up(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+ }
+ snd_cs4231_out(chip, CS4231_REC_FORMAT, cdfr);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ snd_cs4231_mce_down(chip);
+
+ snd_cs4231_calibrate_mute(chip, 0);
+ mutex_unlock(&chip->mce_mutex);
+}
+
+/*
+ * Timer interface
+ */
+
+static unsigned long snd_cs4231_timer_resolution(struct snd_timer *timer)
+{
+ struct snd_cs4231 *chip = snd_timer_chip(timer);
+
+ return chip->image[CS4231_PLAYBK_FORMAT] & 1 ? 9969 : 9920;
+}
+
+static int snd_cs4231_timer_start(struct snd_timer *timer)
+{
+ unsigned long flags;
+ unsigned int ticks;
+ struct snd_cs4231 *chip = snd_timer_chip(timer);
+
+ spin_lock_irqsave(&chip->lock, flags);
+ ticks = timer->sticks;
+ if ((chip->image[CS4231_ALT_FEATURE_1] & CS4231_TIMER_ENABLE) == 0 ||
+ (unsigned char)(ticks >> 8) != chip->image[CS4231_TIMER_HIGH] ||
+ (unsigned char)ticks != chip->image[CS4231_TIMER_LOW]) {
+ snd_cs4231_out(chip, CS4231_TIMER_HIGH,
+ chip->image[CS4231_TIMER_HIGH] =
+ (unsigned char) (ticks >> 8));
+ snd_cs4231_out(chip, CS4231_TIMER_LOW,
+ chip->image[CS4231_TIMER_LOW] =
+ (unsigned char) ticks);
+ snd_cs4231_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1] |
+ CS4231_TIMER_ENABLE);
+ }
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return 0;
+}
+
+static int snd_cs4231_timer_stop(struct snd_timer *timer)
+{
+ unsigned long flags;
+ struct snd_cs4231 *chip = snd_timer_chip(timer);
+
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE;
+ snd_cs4231_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return 0;
+}
+
+static void __init snd_cs4231_init(struct snd_cs4231 *chip)
+{
+ unsigned long flags;
+
+ snd_cs4231_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printdd("init: (1)\n");
+#endif
+ snd_cs4231_mce_up(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE |
+ CS4231_PLAYBACK_PIO |
+ CS4231_RECORD_ENABLE |
+ CS4231_RECORD_PIO |
+ CS4231_CALIB_MODE);
+ chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB;
+ snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ snd_cs4231_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printdd("init: (2)\n");
+#endif
+
+ snd_cs4231_mce_up(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_ALT_FEATURE_1,
+ chip->image[CS4231_ALT_FEATURE_1]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ snd_cs4231_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printdd("init: (3) - afei = 0x%x\n",
+ chip->image[CS4231_ALT_FEATURE_1]);
+#endif
+
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_ALT_FEATURE_2,
+ chip->image[CS4231_ALT_FEATURE_2]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ snd_cs4231_mce_up(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT,
+ chip->image[CS4231_PLAYBK_FORMAT]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ snd_cs4231_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printdd("init: (4)\n");
+#endif
+
+ snd_cs4231_mce_up(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ snd_cs4231_mce_down(chip);
+
+#ifdef SNDRV_DEBUG_MCE
+ snd_printdd("init: (5)\n");
+#endif
+}
+
+static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode)
+{
+ unsigned long flags;
+
+ mutex_lock(&chip->open_mutex);
+ if ((chip->mode & mode)) {
+ mutex_unlock(&chip->open_mutex);
+ return -EAGAIN;
+ }
+ if (chip->mode & CS4231_MODE_OPEN) {
+ chip->mode |= mode;
+ mutex_unlock(&chip->open_mutex);
+ return 0;
+ }
+ /* ok. now enable and ack CODEC IRQ */
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ |
+ CS4231_RECORD_IRQ |
+ CS4231_TIMER_IRQ);
+ snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0);
+ __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */
+ __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */
+
+ snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ |
+ CS4231_RECORD_IRQ |
+ CS4231_TIMER_IRQ);
+ snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ chip->mode = mode;
+ mutex_unlock(&chip->open_mutex);
+ return 0;
+}
+
+static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode)
+{
+ unsigned long flags;
+
+ mutex_lock(&chip->open_mutex);
+ chip->mode &= ~mode;
+ if (chip->mode & CS4231_MODE_OPEN) {
+ mutex_unlock(&chip->open_mutex);
+ return;
+ }
+ snd_cs4231_calibrate_mute(chip, 1);
+
+ /* disable IRQ */
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0);
+ __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */
+ __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */
+
+ /* now disable record & playback */
+
+ if (chip->image[CS4231_IFACE_CTRL] &
+ (CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO |
+ CS4231_RECORD_ENABLE | CS4231_RECORD_PIO)) {
+ spin_unlock_irqrestore(&chip->lock, flags);
+ snd_cs4231_mce_up(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->image[CS4231_IFACE_CTRL] &=
+ ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO |
+ CS4231_RECORD_ENABLE | CS4231_RECORD_PIO);
+ snd_cs4231_out(chip, CS4231_IFACE_CTRL,
+ chip->image[CS4231_IFACE_CTRL]);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ snd_cs4231_mce_down(chip);
+ spin_lock_irqsave(&chip->lock, flags);
+ }
+
+ /* clear IRQ again */
+ snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0);
+ __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */
+ __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ snd_cs4231_calibrate_mute(chip, 0);
+
+ chip->mode = 0;
+ mutex_unlock(&chip->open_mutex);
+}
+
+/*
+ * timer open/close
+ */
+
+static int snd_cs4231_timer_open(struct snd_timer *timer)
+{
+ struct snd_cs4231 *chip = snd_timer_chip(timer);
+ snd_cs4231_open(chip, CS4231_MODE_TIMER);
+ return 0;
+}
+
+static int snd_cs4231_timer_close(struct snd_timer *timer)
+{
+ struct snd_cs4231 *chip = snd_timer_chip(timer);
+ snd_cs4231_close(chip, CS4231_MODE_TIMER);
+ return 0;
+}
+
+static struct snd_timer_hardware snd_cs4231_timer_table = {
+ .flags = SNDRV_TIMER_HW_AUTO,
+ .resolution = 9945,
+ .ticks = 65535,
+ .open = snd_cs4231_timer_open,
+ .close = snd_cs4231_timer_close,
+ .c_resolution = snd_cs4231_timer_resolution,
+ .start = snd_cs4231_timer_start,
+ .stop = snd_cs4231_timer_stop,
+};
+
+/*
+ * ok.. exported functions..
+ */
+
+static int snd_cs4231_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ unsigned char new_pdfr;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ new_pdfr = snd_cs4231_get_format(chip, params_format(hw_params),
+ params_channels(hw_params)) |
+ snd_cs4231_get_rate(params_rate(hw_params));
+ snd_cs4231_playback_format(chip, hw_params, new_pdfr);
+
+ return 0;
+}
+
+static int snd_cs4231_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE |
+ CS4231_PLAYBACK_PIO);
+
+ BUG_ON(runtime->period_size > 0xffff + 1);
+
+ chip->p_periods_sent = 0;
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return 0;
+}
+
+static int snd_cs4231_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ unsigned char new_cdfr;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ new_cdfr = snd_cs4231_get_format(chip, params_format(hw_params),
+ params_channels(hw_params)) |
+ snd_cs4231_get_rate(params_rate(hw_params));
+ snd_cs4231_capture_format(chip, hw_params, new_cdfr);
+
+ return 0;
+}
+
+static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE |
+ CS4231_RECORD_PIO);
+
+
+ chip->c_periods_sent = 0;
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return 0;
+}
+
+static void snd_cs4231_overrange(struct snd_cs4231 *chip)
+{
+ unsigned long flags;
+ unsigned char res;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ res = snd_cs4231_in(chip, CS4231_TEST_INIT);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ /* detect overrange only above 0dB; may be user selectable? */
+ if (res & (0x08 | 0x02))
+ chip->capture_substream->runtime->overrange++;
+}
+
+static void snd_cs4231_play_callback(struct snd_cs4231 *chip)
+{
+ if (chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE) {
+ snd_pcm_period_elapsed(chip->playback_substream);
+ snd_cs4231_advance_dma(&chip->p_dma, chip->playback_substream,
+ &chip->p_periods_sent);
+ }
+}
+
+static void snd_cs4231_capture_callback(struct snd_cs4231 *chip)
+{
+ if (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) {
+ snd_pcm_period_elapsed(chip->capture_substream);
+ snd_cs4231_advance_dma(&chip->c_dma, chip->capture_substream,
+ &chip->c_periods_sent);
+ }
+}
+
+static snd_pcm_uframes_t snd_cs4231_playback_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ struct cs4231_dma_control *dma_cont = &chip->p_dma;
+ size_t ptr;
+
+ if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE))
+ return 0;
+ ptr = dma_cont->address(dma_cont);
+ if (ptr != 0)
+ ptr -= substream->runtime->dma_addr;
+
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static snd_pcm_uframes_t snd_cs4231_capture_pointer(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ struct cs4231_dma_control *dma_cont = &chip->c_dma;
+ size_t ptr;
+
+ if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE))
+ return 0;
+ ptr = dma_cont->address(dma_cont);
+ if (ptr != 0)
+ ptr -= substream->runtime->dma_addr;
+
+ return bytes_to_frames(substream->runtime, ptr);
+}
+
+static int __init snd_cs4231_probe(struct snd_cs4231 *chip)
+{
+ unsigned long flags;
+ int i;
+ int id = 0;
+ int vers = 0;
+ unsigned char *ptr;
+
+ for (i = 0; i < 50; i++) {
+ mb();
+ if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT)
+ msleep(2);
+ else {
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_out(chip, CS4231_MISC_INFO, CS4231_MODE2);
+ id = snd_cs4231_in(chip, CS4231_MISC_INFO) & 0x0f;
+ vers = snd_cs4231_in(chip, CS4231_VERSION);
+ spin_unlock_irqrestore(&chip->lock, flags);
+ if (id == 0x0a)
+ break; /* this is valid value */
+ }
+ }
+ snd_printdd("cs4231: port = %p, id = 0x%x\n", chip->port, id);
+ if (id != 0x0a)
+ return -ENODEV; /* no valid device found */
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ /* clear any pendings IRQ */
+ __cs4231_readb(chip, CS4231U(chip, STATUS));
+ __cs4231_writeb(chip, 0, CS4231U(chip, STATUS));
+ mb();
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ chip->image[CS4231_MISC_INFO] = CS4231_MODE2;
+ chip->image[CS4231_IFACE_CTRL] =
+ chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA;
+ chip->image[CS4231_ALT_FEATURE_1] = 0x80;
+ chip->image[CS4231_ALT_FEATURE_2] = 0x01;
+ if (vers & 0x20)
+ chip->image[CS4231_ALT_FEATURE_2] |= 0x02;
+
+ ptr = (unsigned char *) &chip->image;
+
+ snd_cs4231_mce_down(chip);
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ for (i = 0; i < 32; i++) /* ok.. fill all CS4231 registers */
+ snd_cs4231_out(chip, i, *ptr++);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ snd_cs4231_mce_up(chip);
+
+ snd_cs4231_mce_down(chip);
+
+ mdelay(2);
+
+ return 0; /* all things are ok.. */
+}
+
+static struct snd_pcm_hardware snd_cs4231_playback = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_MU_LAW |
+ SNDRV_PCM_FMTBIT_A_LAW |
+ SNDRV_PCM_FMTBIT_IMA_ADPCM |
+ SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = SNDRV_PCM_RATE_KNOT |
+ SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5510,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 32 * 1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = 32 * 1024,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static struct snd_pcm_hardware snd_cs4231_capture = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
+ .formats = SNDRV_PCM_FMTBIT_MU_LAW |
+ SNDRV_PCM_FMTBIT_A_LAW |
+ SNDRV_PCM_FMTBIT_IMA_ADPCM |
+ SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = SNDRV_PCM_RATE_KNOT |
+ SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 5510,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 32 * 1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = 32 * 1024,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static int snd_cs4231_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ runtime->hw = snd_cs4231_playback;
+
+ err = snd_cs4231_open(chip, CS4231_MODE_PLAY);
+ if (err < 0) {
+ snd_free_pages(runtime->dma_area, runtime->dma_bytes);
+ return err;
+ }
+ chip->playback_substream = substream;
+ chip->p_periods_sent = 0;
+ snd_pcm_set_sync(substream);
+ snd_cs4231_xrate(runtime);
+
+ return 0;
+}
+
+static int snd_cs4231_capture_open(struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ runtime->hw = snd_cs4231_capture;
+
+ err = snd_cs4231_open(chip, CS4231_MODE_RECORD);
+ if (err < 0) {
+ snd_free_pages(runtime->dma_area, runtime->dma_bytes);
+ return err;
+ }
+ chip->capture_substream = substream;
+ chip->c_periods_sent = 0;
+ snd_pcm_set_sync(substream);
+ snd_cs4231_xrate(runtime);
+
+ return 0;
+}
+
+static int snd_cs4231_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+
+ snd_cs4231_close(chip, CS4231_MODE_PLAY);
+ chip->playback_substream = NULL;
+
+ return 0;
+}
+
+static int snd_cs4231_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_cs4231 *chip = snd_pcm_substream_chip(substream);
+
+ snd_cs4231_close(chip, CS4231_MODE_RECORD);
+ chip->capture_substream = NULL;
+
+ return 0;
+}
+
+/* XXX We can do some power-management, in particular on EBUS using
+ * XXX the audio AUXIO register...
+ */
+
+static struct snd_pcm_ops snd_cs4231_playback_ops = {
+ .open = snd_cs4231_playback_open,
+ .close = snd_cs4231_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs4231_playback_hw_params,
+ .hw_free = snd_pcm_lib_free_pages,
+ .prepare = snd_cs4231_playback_prepare,
+ .trigger = snd_cs4231_trigger,
+ .pointer = snd_cs4231_playback_pointer,
+};
+
+static struct snd_pcm_ops snd_cs4231_capture_ops = {
+ .open = snd_cs4231_capture_open,
+ .close = snd_cs4231_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_cs4231_capture_hw_params,
+ .hw_free = snd_pcm_lib_free_pages,
+ .prepare = snd_cs4231_capture_prepare,
+ .trigger = snd_cs4231_trigger,
+ .pointer = snd_cs4231_capture_pointer,
+};
+
+static int __init snd_cs4231_pcm(struct snd_card *card)
+{
+ struct snd_cs4231 *chip = card->private_data;
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(card, "CS4231", 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_cs4231_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_cs4231_capture_ops);
+
+ /* global setup */
+ pcm->private_data = chip;
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+ strcpy(pcm->name, "CS4231");
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ &chip->op->dev,
+ 64 * 1024, 128 * 1024);
+
+ chip->pcm = pcm;
+
+ return 0;
+}
+
+static int __init snd_cs4231_timer(struct snd_card *card)
+{
+ struct snd_cs4231 *chip = card->private_data;
+ struct snd_timer *timer;
+ struct snd_timer_id tid;
+ int err;
+
+ /* Timer initialization */
+ tid.dev_class = SNDRV_TIMER_CLASS_CARD;
+ tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE;
+ tid.card = card->number;
+ tid.device = 0;
+ tid.subdevice = 0;
+ err = snd_timer_new(card, "CS4231", &tid, &timer);
+ if (err < 0)
+ return err;
+ strcpy(timer->name, "CS4231");
+ timer->private_data = chip;
+ timer->hw = snd_cs4231_timer_table;
+ chip->timer = timer;
+
+ return 0;
+}
+
+/*
+ * MIXER part
+ */
+
+static int snd_cs4231_info_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[4] = {
+ "Line", "CD", "Mic", "Mix"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 2;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int snd_cs4231_get_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->lock, flags);
+ ucontrol->value.enumerated.item[0] =
+ (chip->image[CS4231_LEFT_INPUT] & CS4231_MIXS_ALL) >> 6;
+ ucontrol->value.enumerated.item[1] =
+ (chip->image[CS4231_RIGHT_INPUT] & CS4231_MIXS_ALL) >> 6;
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return 0;
+}
+
+static int snd_cs4231_put_mux(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ unsigned short left, right;
+ int change;
+
+ if (ucontrol->value.enumerated.item[0] > 3 ||
+ ucontrol->value.enumerated.item[1] > 3)
+ return -EINVAL;
+ left = ucontrol->value.enumerated.item[0] << 6;
+ right = ucontrol->value.enumerated.item[1] << 6;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left;
+ right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right;
+ change = left != chip->image[CS4231_LEFT_INPUT] ||
+ right != chip->image[CS4231_RIGHT_INPUT];
+ snd_cs4231_out(chip, CS4231_LEFT_INPUT, left);
+ snd_cs4231_out(chip, CS4231_RIGHT_INPUT, right);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return change;
+}
+
+static int snd_cs4231_info_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = (mask == 1) ?
+ SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+
+ return 0;
+}
+
+static int snd_cs4231_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask;
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ if (invert)
+ ucontrol->value.integer.value[0] =
+ (mask - ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int snd_cs4231_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ val = (chip->image[reg] & ~(mask << shift)) | val;
+ change = val != chip->image[reg];
+ snd_cs4231_out(chip, reg, val);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return change;
+}
+
+static int snd_cs4231_info_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ uinfo->type = mask == 1 ?
+ SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+
+ return 0;
+}
+
+static int snd_cs4231_get_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ ucontrol->value.integer.value[0] =
+ (chip->image[left_reg] >> shift_left) & mask;
+ ucontrol->value.integer.value[1] =
+ (chip->image[right_reg] >> shift_right) & mask;
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ (mask - ucontrol->value.integer.value[0]);
+ ucontrol->value.integer.value[1] =
+ (mask - ucontrol->value.integer.value[1]);
+ }
+
+ return 0;
+}
+
+static int snd_cs4231_put_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned long flags;
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change;
+ unsigned short val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1;
+ val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2;
+ change = val1 != chip->image[left_reg];
+ change |= val2 != chip->image[right_reg];
+ snd_cs4231_out(chip, left_reg, val1);
+ snd_cs4231_out(chip, right_reg, val2);
+
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return change;
+}
+
+#define CS4231_SINGLE(xname, xindex, reg, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .index = (xindex), \
+ .info = snd_cs4231_info_single, \
+ .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \
+ .private_value = (reg) | ((shift) << 8) | ((mask) << 16) | ((invert) << 24) }
+
+#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, \
+ shift_right, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .index = (xindex), \
+ .info = snd_cs4231_info_double, \
+ .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \
+ .private_value = (left_reg) | ((right_reg) << 8) | ((shift_left) << 16) | \
+ ((shift_right) << 19) | ((mask) << 24) | ((invert) << 22) }
+
+static struct snd_kcontrol_new snd_cs4231_controls[] __initdata = {
+CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT,
+ CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT,
+ CS4231_RIGHT_OUTPUT, 0, 0, 63, 1),
+CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN,
+ CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
+CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN,
+ CS4231_RIGHT_LINE_IN, 0, 0, 31, 1),
+CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT,
+ CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT,
+ CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1),
+CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT,
+ CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT,
+ CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1),
+CS4231_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1),
+CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1),
+CS4231_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1),
+CS4231_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0),
+CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0,
+ 15, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_cs4231_info_mux,
+ .get = snd_cs4231_get_mux,
+ .put = snd_cs4231_put_mux,
+},
+CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5,
+ 1, 0),
+CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0),
+CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1),
+/* SPARC specific uses of XCTL{0,1} general purpose outputs. */
+CS4231_SINGLE("Line Out Switch", 0, CS4231_PIN_CTRL, 6, 1, 1),
+CS4231_SINGLE("Headphone Out Switch", 0, CS4231_PIN_CTRL, 7, 1, 1)
+};
+
+static int __init snd_cs4231_mixer(struct snd_card *card)
+{
+ struct snd_cs4231 *chip = card->private_data;
+ int err, idx;
+
+ if (snd_BUG_ON(!chip || !chip->pcm))
+ return -EINVAL;
+
+ strcpy(card->mixername, chip->pcm->name);
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_cs4231_controls[idx], chip));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int dev;
+
+static int __init cs4231_attach_begin(struct snd_card **rcard)
+{
+ struct snd_card *card;
+ struct snd_cs4231 *chip;
+
+ *rcard = NULL;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_cs4231));
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "CS4231");
+ strcpy(card->shortname, "Sun CS4231");
+
+ chip = card->private_data;
+ chip->card = card;
+
+ *rcard = card;
+ return 0;
+}
+
+static int __init cs4231_attach_finish(struct snd_card *card)
+{
+ struct snd_cs4231 *chip = card->private_data;
+ int err;
+
+ err = snd_cs4231_pcm(card);
+ if (err < 0)
+ goto out_err;
+
+ err = snd_cs4231_mixer(card);
+ if (err < 0)
+ goto out_err;
+
+ err = snd_cs4231_timer(card);
+ if (err < 0)
+ goto out_err;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto out_err;
+
+ dev_set_drvdata(&chip->op->dev, chip);
+
+ dev++;
+ return 0;
+
+out_err:
+ snd_card_free(card);
+ return err;
+}
+
+#ifdef SBUS_SUPPORT
+
+static irqreturn_t snd_cs4231_sbus_interrupt(int irq, void *dev_id)
+{
+ unsigned long flags;
+ unsigned char status;
+ u32 csr;
+ struct snd_cs4231 *chip = dev_id;
+
+ /*This is IRQ is not raised by the cs4231*/
+ if (!(__cs4231_readb(chip, CS4231U(chip, STATUS)) & CS4231_GLOBALIRQ))
+ return IRQ_NONE;
+
+ /* ACK the APC interrupt. */
+ csr = sbus_readl(chip->port + APCCSR);
+
+ sbus_writel(csr, chip->port + APCCSR);
+
+ if ((csr & APC_PDMA_READY) &&
+ (csr & APC_PLAY_INT) &&
+ (csr & APC_XINT_PNVA) &&
+ !(csr & APC_XINT_EMPT))
+ snd_cs4231_play_callback(chip);
+
+ if ((csr & APC_CDMA_READY) &&
+ (csr & APC_CAPT_INT) &&
+ (csr & APC_XINT_CNVA) &&
+ !(csr & APC_XINT_EMPT))
+ snd_cs4231_capture_callback(chip);
+
+ status = snd_cs4231_in(chip, CS4231_IRQ_STATUS);
+
+ if (status & CS4231_TIMER_IRQ) {
+ if (chip->timer)
+ snd_timer_interrupt(chip->timer, chip->timer->sticks);
+ }
+
+ if ((status & CS4231_RECORD_IRQ) && (csr & APC_CDMA_READY))
+ snd_cs4231_overrange(chip);
+
+ /* ACK the CS4231 interrupt. */
+ spin_lock_irqsave(&chip->lock, flags);
+ snd_cs4231_outm(chip, CS4231_IRQ_STATUS, ~CS4231_ALL_IRQS | ~status, 0);
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * SBUS DMA routines
+ */
+
+static int sbus_dma_request(struct cs4231_dma_control *dma_cont,
+ dma_addr_t bus_addr, size_t len)
+{
+ unsigned long flags;
+ u32 test, csr;
+ int err;
+ struct sbus_dma_info *base = &dma_cont->sbus_info;
+
+ if (len >= (1 << 24))
+ return -EINVAL;
+ spin_lock_irqsave(&base->lock, flags);
+ csr = sbus_readl(base->regs + APCCSR);
+ err = -EINVAL;
+ test = APC_CDMA_READY;
+ if (base->dir == APC_PLAY)
+ test = APC_PDMA_READY;
+ if (!(csr & test))
+ goto out;
+ err = -EBUSY;
+ test = APC_XINT_CNVA;
+ if (base->dir == APC_PLAY)
+ test = APC_XINT_PNVA;
+ if (!(csr & test))
+ goto out;
+ err = 0;
+ sbus_writel(bus_addr, base->regs + base->dir + APCNVA);
+ sbus_writel(len, base->regs + base->dir + APCNC);
+out:
+ spin_unlock_irqrestore(&base->lock, flags);
+ return err;
+}
+
+static void sbus_dma_prepare(struct cs4231_dma_control *dma_cont, int d)
+{
+ unsigned long flags;
+ u32 csr, test;
+ struct sbus_dma_info *base = &dma_cont->sbus_info;
+
+ spin_lock_irqsave(&base->lock, flags);
+ csr = sbus_readl(base->regs + APCCSR);
+ test = APC_GENL_INT | APC_PLAY_INT | APC_XINT_ENA |
+ APC_XINT_PLAY | APC_XINT_PEMP | APC_XINT_GENL |
+ APC_XINT_PENA;
+ if (base->dir == APC_RECORD)
+ test = APC_GENL_INT | APC_CAPT_INT | APC_XINT_ENA |
+ APC_XINT_CAPT | APC_XINT_CEMP | APC_XINT_GENL;
+ csr |= test;
+ sbus_writel(csr, base->regs + APCCSR);
+ spin_unlock_irqrestore(&base->lock, flags);
+}
+
+static void sbus_dma_enable(struct cs4231_dma_control *dma_cont, int on)
+{
+ unsigned long flags;
+ u32 csr, shift;
+ struct sbus_dma_info *base = &dma_cont->sbus_info;
+
+ spin_lock_irqsave(&base->lock, flags);
+ if (!on) {
+ sbus_writel(0, base->regs + base->dir + APCNC);
+ sbus_writel(0, base->regs + base->dir + APCNVA);
+ if (base->dir == APC_PLAY) {
+ sbus_writel(0, base->regs + base->dir + APCC);
+ sbus_writel(0, base->regs + base->dir + APCVA);
+ }
+
+ udelay(1200);
+ }
+ csr = sbus_readl(base->regs + APCCSR);
+ shift = 0;
+ if (base->dir == APC_PLAY)
+ shift = 1;
+ if (on)
+ csr &= ~(APC_CPAUSE << shift);
+ else
+ csr |= (APC_CPAUSE << shift);
+ sbus_writel(csr, base->regs + APCCSR);
+ if (on)
+ csr |= (APC_CDMA_READY << shift);
+ else
+ csr &= ~(APC_CDMA_READY << shift);
+ sbus_writel(csr, base->regs + APCCSR);
+
+ spin_unlock_irqrestore(&base->lock, flags);
+}
+
+static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont)
+{
+ struct sbus_dma_info *base = &dma_cont->sbus_info;
+
+ return sbus_readl(base->regs + base->dir + APCVA);
+}
+
+/*
+ * Init and exit routines
+ */
+
+static int snd_cs4231_sbus_free(struct snd_cs4231 *chip)
+{
+ struct of_device *op = chip->op;
+
+ if (chip->irq[0])
+ free_irq(chip->irq[0], chip);
+
+ if (chip->port)
+ of_iounmap(&op->resource[0], chip->port, chip->regs_size);
+
+ return 0;
+}
+
+static int snd_cs4231_sbus_dev_free(struct snd_device *device)
+{
+ struct snd_cs4231 *cp = device->device_data;
+
+ return snd_cs4231_sbus_free(cp);
+}
+
+static struct snd_device_ops snd_cs4231_sbus_dev_ops = {
+ .dev_free = snd_cs4231_sbus_dev_free,
+};
+
+static int __init snd_cs4231_sbus_create(struct snd_card *card,
+ struct of_device *op,
+ int dev)
+{
+ struct snd_cs4231 *chip = card->private_data;
+ int err;
+
+ spin_lock_init(&chip->lock);
+ spin_lock_init(&chip->c_dma.sbus_info.lock);
+ spin_lock_init(&chip->p_dma.sbus_info.lock);
+ mutex_init(&chip->mce_mutex);
+ mutex_init(&chip->open_mutex);
+ chip->op = op;
+ chip->regs_size = resource_size(&op->resource[0]);
+ memcpy(&chip->image, &snd_cs4231_original_image,
+ sizeof(snd_cs4231_original_image));
+
+ chip->port = of_ioremap(&op->resource[0], 0,
+ chip->regs_size, "cs4231");
+ if (!chip->port) {
+ snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev);
+ return -EIO;
+ }
+
+ chip->c_dma.sbus_info.regs = chip->port;
+ chip->p_dma.sbus_info.regs = chip->port;
+ chip->c_dma.sbus_info.dir = APC_RECORD;
+ chip->p_dma.sbus_info.dir = APC_PLAY;
+
+ chip->p_dma.prepare = sbus_dma_prepare;
+ chip->p_dma.enable = sbus_dma_enable;
+ chip->p_dma.request = sbus_dma_request;
+ chip->p_dma.address = sbus_dma_addr;
+
+ chip->c_dma.prepare = sbus_dma_prepare;
+ chip->c_dma.enable = sbus_dma_enable;
+ chip->c_dma.request = sbus_dma_request;
+ chip->c_dma.address = sbus_dma_addr;
+
+ if (request_irq(op->irqs[0], snd_cs4231_sbus_interrupt,
+ IRQF_SHARED, "cs4231", chip)) {
+ snd_printdd("cs4231-%d: Unable to grab SBUS IRQ %d\n",
+ dev, op->irqs[0]);
+ snd_cs4231_sbus_free(chip);
+ return -EBUSY;
+ }
+ chip->irq[0] = op->irqs[0];
+
+ if (snd_cs4231_probe(chip) < 0) {
+ snd_cs4231_sbus_free(chip);
+ return -ENODEV;
+ }
+ snd_cs4231_init(chip);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &snd_cs4231_sbus_dev_ops)) < 0) {
+ snd_cs4231_sbus_free(chip);
+ return err;
+ }
+
+ return 0;
+}
+
+static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match)
+{
+ struct resource *rp = &op->resource[0];
+ struct snd_card *card;
+ int err;
+
+ err = cs4231_attach_begin(&card);
+ if (err)
+ return err;
+
+ sprintf(card->longname, "%s at 0x%02lx:0x%016Lx, irq %d",
+ card->shortname,
+ rp->flags & 0xffL,
+ (unsigned long long)rp->start,
+ op->irqs[0]);
+
+ err = snd_cs4231_sbus_create(card, op, dev);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ return cs4231_attach_finish(card);
+}
+#endif
+
+#ifdef EBUS_SUPPORT
+
+static void snd_cs4231_ebus_play_callback(struct ebus_dma_info *p, int event,
+ void *cookie)
+{
+ struct snd_cs4231 *chip = cookie;
+
+ snd_cs4231_play_callback(chip);
+}
+
+static void snd_cs4231_ebus_capture_callback(struct ebus_dma_info *p,
+ int event, void *cookie)
+{
+ struct snd_cs4231 *chip = cookie;
+
+ snd_cs4231_capture_callback(chip);
+}
+
+/*
+ * EBUS DMA wrappers
+ */
+
+static int _ebus_dma_request(struct cs4231_dma_control *dma_cont,
+ dma_addr_t bus_addr, size_t len)
+{
+ return ebus_dma_request(&dma_cont->ebus_info, bus_addr, len);
+}
+
+static void _ebus_dma_enable(struct cs4231_dma_control *dma_cont, int on)
+{
+ ebus_dma_enable(&dma_cont->ebus_info, on);
+}
+
+static void _ebus_dma_prepare(struct cs4231_dma_control *dma_cont, int dir)
+{
+ ebus_dma_prepare(&dma_cont->ebus_info, dir);
+}
+
+static unsigned int _ebus_dma_addr(struct cs4231_dma_control *dma_cont)
+{
+ return ebus_dma_addr(&dma_cont->ebus_info);
+}
+
+/*
+ * Init and exit routines
+ */
+
+static int snd_cs4231_ebus_free(struct snd_cs4231 *chip)
+{
+ struct of_device *op = chip->op;
+
+ if (chip->c_dma.ebus_info.regs) {
+ ebus_dma_unregister(&chip->c_dma.ebus_info);
+ of_iounmap(&op->resource[2], chip->c_dma.ebus_info.regs, 0x10);
+ }
+ if (chip->p_dma.ebus_info.regs) {
+ ebus_dma_unregister(&chip->p_dma.ebus_info);
+ of_iounmap(&op->resource[1], chip->p_dma.ebus_info.regs, 0x10);
+ }
+
+ if (chip->port)
+ of_iounmap(&op->resource[0], chip->port, 0x10);
+
+ return 0;
+}
+
+static int snd_cs4231_ebus_dev_free(struct snd_device *device)
+{
+ struct snd_cs4231 *cp = device->device_data;
+
+ return snd_cs4231_ebus_free(cp);
+}
+
+static struct snd_device_ops snd_cs4231_ebus_dev_ops = {
+ .dev_free = snd_cs4231_ebus_dev_free,
+};
+
+static int __init snd_cs4231_ebus_create(struct snd_card *card,
+ struct of_device *op,
+ int dev)
+{
+ struct snd_cs4231 *chip = card->private_data;
+ int err;
+
+ spin_lock_init(&chip->lock);
+ spin_lock_init(&chip->c_dma.ebus_info.lock);
+ spin_lock_init(&chip->p_dma.ebus_info.lock);
+ mutex_init(&chip->mce_mutex);
+ mutex_init(&chip->open_mutex);
+ chip->flags |= CS4231_FLAG_EBUS;
+ chip->op = op;
+ memcpy(&chip->image, &snd_cs4231_original_image,
+ sizeof(snd_cs4231_original_image));
+ strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)");
+ chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER;
+ chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback;
+ chip->c_dma.ebus_info.client_cookie = chip;
+ chip->c_dma.ebus_info.irq = op->irqs[0];
+ strcpy(chip->p_dma.ebus_info.name, "cs4231(play)");
+ chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER;
+ chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback;
+ chip->p_dma.ebus_info.client_cookie = chip;
+ chip->p_dma.ebus_info.irq = op->irqs[1];
+
+ chip->p_dma.prepare = _ebus_dma_prepare;
+ chip->p_dma.enable = _ebus_dma_enable;
+ chip->p_dma.request = _ebus_dma_request;
+ chip->p_dma.address = _ebus_dma_addr;
+
+ chip->c_dma.prepare = _ebus_dma_prepare;
+ chip->c_dma.enable = _ebus_dma_enable;
+ chip->c_dma.request = _ebus_dma_request;
+ chip->c_dma.address = _ebus_dma_addr;
+
+ chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231");
+ chip->p_dma.ebus_info.regs =
+ of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma");
+ chip->c_dma.ebus_info.regs =
+ of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma");
+ if (!chip->port || !chip->p_dma.ebus_info.regs ||
+ !chip->c_dma.ebus_info.regs) {
+ snd_cs4231_ebus_free(chip);
+ snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev);
+ return -EIO;
+ }
+
+ if (ebus_dma_register(&chip->c_dma.ebus_info)) {
+ snd_cs4231_ebus_free(chip);
+ snd_printdd("cs4231-%d: Unable to register EBUS capture DMA\n",
+ dev);
+ return -EBUSY;
+ }
+ if (ebus_dma_irq_enable(&chip->c_dma.ebus_info, 1)) {
+ snd_cs4231_ebus_free(chip);
+ snd_printdd("cs4231-%d: Unable to enable EBUS capture IRQ\n",
+ dev);
+ return -EBUSY;
+ }
+
+ if (ebus_dma_register(&chip->p_dma.ebus_info)) {
+ snd_cs4231_ebus_free(chip);
+ snd_printdd("cs4231-%d: Unable to register EBUS play DMA\n",
+ dev);
+ return -EBUSY;
+ }
+ if (ebus_dma_irq_enable(&chip->p_dma.ebus_info, 1)) {
+ snd_cs4231_ebus_free(chip);
+ snd_printdd("cs4231-%d: Unable to enable EBUS play IRQ\n", dev);
+ return -EBUSY;
+ }
+
+ if (snd_cs4231_probe(chip) < 0) {
+ snd_cs4231_ebus_free(chip);
+ return -ENODEV;
+ }
+ snd_cs4231_init(chip);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &snd_cs4231_ebus_dev_ops)) < 0) {
+ snd_cs4231_ebus_free(chip);
+ return err;
+ }
+
+ return 0;
+}
+
+static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match)
+{
+ struct snd_card *card;
+ int err;
+
+ err = cs4231_attach_begin(&card);
+ if (err)
+ return err;
+
+ sprintf(card->longname, "%s at 0x%lx, irq %d",
+ card->shortname,
+ op->resource[0].start,
+ op->irqs[0]);
+
+ err = snd_cs4231_ebus_create(card, op, dev);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ return cs4231_attach_finish(card);
+}
+#endif
+
+static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match)
+{
+#ifdef EBUS_SUPPORT
+ if (!strcmp(op->node->parent->name, "ebus"))
+ return cs4231_ebus_probe(op, match);
+#endif
+#ifdef SBUS_SUPPORT
+ if (!strcmp(op->node->parent->name, "sbus") ||
+ !strcmp(op->node->parent->name, "sbi"))
+ return cs4231_sbus_probe(op, match);
+#endif
+ return -ENODEV;
+}
+
+static int __devexit cs4231_remove(struct of_device *op)
+{
+ struct snd_cs4231 *chip = dev_get_drvdata(&op->dev);
+
+ snd_card_free(chip->card);
+
+ return 0;
+}
+
+static const struct of_device_id cs4231_match[] = {
+ {
+ .name = "SUNW,CS4231",
+ },
+ {
+ .name = "audio",
+ .compatible = "SUNW,CS4231",
+ },
+ {},
+};
+
+MODULE_DEVICE_TABLE(of, cs4231_match);
+
+static struct of_platform_driver cs4231_driver = {
+ .name = "audio",
+ .match_table = cs4231_match,
+ .probe = cs4231_probe,
+ .remove = __devexit_p(cs4231_remove),
+};
+
+static int __init cs4231_init(void)
+{
+ return of_register_driver(&cs4231_driver, &of_bus_type);
+}
+
+static void __exit cs4231_exit(void)
+{
+ of_unregister_driver(&cs4231_driver);
+}
+
+module_init(cs4231_init);
+module_exit(cs4231_exit);
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
new file mode 100644
index 0000000..23ed6f0
--- /dev/null
+++ b/sound/sparc/dbri.c
@@ -0,0 +1,2706 @@
+/*
+ * Driver for DBRI sound chip found on Sparcs.
+ * Copyright (C) 2004, 2005 Martin Habets (mhabets@users.sourceforge.net)
+ *
+ * Converted to ring buffered version by Krzysztof Helt (krzysztof.h1@wp.pl)
+ *
+ * Based entirely upon drivers/sbus/audio/dbri.c which is:
+ * Copyright (C) 1997 Rudolf Koenig (rfkoenig@immd4.informatik.uni-erlangen.de)
+ * Copyright (C) 1998, 1999 Brent Baccala (baccala@freesoft.org)
+ *
+ * This is the low level driver for the DBRI & MMCODEC duo used for ISDN & AUDIO
+ * on Sun SPARCStation 10, 20, LX and Voyager models.
+ *
+ * - DBRI: AT&T T5900FX Dual Basic Rates ISDN Interface. It is a 32 channel
+ * data time multiplexer with ISDN support (aka T7259)
+ * Interfaces: SBus,ISDN NT & TE, CHI, 4 bits parallel.
+ * CHI: (spelled ki) Concentration Highway Interface (AT&T or Intel bus ?).
+ * Documentation:
+ * - "STP 4000SBus Dual Basic Rate ISDN (DBRI) Transceiver" from
+ * Sparc Technology Business (courtesy of Sun Support)
+ * - Data sheet of the T7903, a newer but very similar ISA bus equivalent
+ * available from the Lucent (formerly AT&T microelectronics) home
+ * page.
+ * - http://www.freesoft.org/Linux/DBRI/
+ * - MMCODEC: Crystal Semiconductor CS4215 16 bit Multimedia Audio Codec
+ * Interfaces: CHI, Audio In & Out, 2 bits parallel
+ * Documentation: from the Crystal Semiconductor home page.
+ *
+ * The DBRI is a 32 pipe machine, each pipe can transfer some bits between
+ * memory and a serial device (long pipes, no. 0-15) or between two serial
+ * devices (short pipes, no. 16-31), or simply send a fixed data to a serial
+ * device (short pipes).
+ * A timeslot defines the bit-offset and no. of bits read from a serial device.
+ * The timeslots are linked to 6 circular lists, one for each direction for
+ * each serial device (NT,TE,CHI). A timeslot is associated to 1 or 2 pipes
+ * (the second one is a monitor/tee pipe, valid only for serial input).
+ *
+ * The mmcodec is connected via the CHI bus and needs the data & some
+ * parameters (volume, output selection) time multiplexed in 8 byte
+ * chunks. It also has a control mode, which serves for audio format setting.
+ *
+ * Looking at the CS4215 data sheet it is easy to set up 2 or 4 codecs on
+ * the same CHI bus, so I thought perhaps it is possible to use the on-board
+ * & the speakerbox codec simultaneously, giving 2 (not very independent :-)
+ * audio devices. But the SUN HW group decided against it, at least on my
+ * LX the speakerbox connector has at least 1 pin missing and 1 wrongly
+ * connected.
+ *
+ * I've tried to stick to the following function naming conventions:
+ * snd_* ALSA stuff
+ * cs4215_* CS4215 codec specific stuff
+ * dbri_* DBRI high-level stuff
+ * other DBRI low-level stuff
+ */
+
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include <linux/irq.h>
+#include <linux/io.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <asm/atomic.h>
+
+MODULE_AUTHOR("Rudolf Koenig, Brent Baccala and Martin Habets");
+MODULE_DESCRIPTION("Sun DBRI");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Sun,DBRI}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+/* Enable this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Sun DBRI soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Sun DBRI soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Sun DBRI soundcard.");
+
+#undef DBRI_DEBUG
+
+#define D_INT (1<<0)
+#define D_GEN (1<<1)
+#define D_CMD (1<<2)
+#define D_MM (1<<3)
+#define D_USR (1<<4)
+#define D_DESC (1<<5)
+
+static int dbri_debug;
+module_param(dbri_debug, int, 0644);
+MODULE_PARM_DESC(dbri_debug, "Debug value for Sun DBRI soundcard.");
+
+#ifdef DBRI_DEBUG
+static char *cmds[] = {
+ "WAIT", "PAUSE", "JUMP", "IIQ", "REX", "SDP", "CDP", "DTS",
+ "SSP", "CHI", "NT", "TE", "CDEC", "TEST", "CDM", "RESRV"
+};
+
+#define dprintk(a, x...) if (dbri_debug & a) printk(KERN_DEBUG x)
+
+#else
+#define dprintk(a, x...) do { } while (0)
+
+#endif /* DBRI_DEBUG */
+
+#define DBRI_CMD(cmd, intr, value) ((cmd << 28) | \
+ (intr << 27) | \
+ value)
+
+/***************************************************************************
+ CS4215 specific definitions and structures
+****************************************************************************/
+
+struct cs4215 {
+ __u8 data[4]; /* Data mode: Time slots 5-8 */
+ __u8 ctrl[4]; /* Ctrl mode: Time slots 1-4 */
+ __u8 onboard;
+ __u8 offset; /* Bit offset from frame sync to time slot 1 */
+ volatile __u32 status;
+ volatile __u32 version;
+ __u8 precision; /* In bits, either 8 or 16 */
+ __u8 channels; /* 1 or 2 */
+};
+
+/*
+ * Control mode first
+ */
+
+/* Time Slot 1, Status register */
+#define CS4215_CLB (1<<2) /* Control Latch Bit */
+#define CS4215_OLB (1<<3) /* 1: line: 2.0V, speaker 4V */
+ /* 0: line: 2.8V, speaker 8V */
+#define CS4215_MLB (1<<4) /* 1: Microphone: 20dB gain disabled */
+#define CS4215_RSRVD_1 (1<<5)
+
+/* Time Slot 2, Data Format Register */
+#define CS4215_DFR_LINEAR16 0
+#define CS4215_DFR_ULAW 1
+#define CS4215_DFR_ALAW 2
+#define CS4215_DFR_LINEAR8 3
+#define CS4215_DFR_STEREO (1<<2)
+static struct {
+ unsigned short freq;
+ unsigned char xtal;
+ unsigned char csval;
+} CS4215_FREQ[] = {
+ { 8000, (1 << 4), (0 << 3) },
+ { 16000, (1 << 4), (1 << 3) },
+ { 27429, (1 << 4), (2 << 3) }, /* Actually 24428.57 */
+ { 32000, (1 << 4), (3 << 3) },
+ /* { NA, (1 << 4), (4 << 3) }, */
+ /* { NA, (1 << 4), (5 << 3) }, */
+ { 48000, (1 << 4), (6 << 3) },
+ { 9600, (1 << 4), (7 << 3) },
+ { 5512, (2 << 4), (0 << 3) }, /* Actually 5512.5 */
+ { 11025, (2 << 4), (1 << 3) },
+ { 18900, (2 << 4), (2 << 3) },
+ { 22050, (2 << 4), (3 << 3) },
+ { 37800, (2 << 4), (4 << 3) },
+ { 44100, (2 << 4), (5 << 3) },
+ { 33075, (2 << 4), (6 << 3) },
+ { 6615, (2 << 4), (7 << 3) },
+ { 0, 0, 0}
+};
+
+#define CS4215_HPF (1<<7) /* High Pass Filter, 1: Enabled */
+
+#define CS4215_12_MASK 0xfcbf /* Mask off reserved bits in slot 1 & 2 */
+
+/* Time Slot 3, Serial Port Control register */
+#define CS4215_XEN (1<<0) /* 0: Enable serial output */
+#define CS4215_XCLK (1<<1) /* 1: Master mode: Generate SCLK */
+#define CS4215_BSEL_64 (0<<2) /* Bitrate: 64 bits per frame */
+#define CS4215_BSEL_128 (1<<2)
+#define CS4215_BSEL_256 (2<<2)
+#define CS4215_MCK_MAST (0<<4) /* Master clock */
+#define CS4215_MCK_XTL1 (1<<4) /* 24.576 MHz clock source */
+#define CS4215_MCK_XTL2 (2<<4) /* 16.9344 MHz clock source */
+#define CS4215_MCK_CLK1 (3<<4) /* Clockin, 256 x Fs */
+#define CS4215_MCK_CLK2 (4<<4) /* Clockin, see DFR */
+
+/* Time Slot 4, Test Register */
+#define CS4215_DAD (1<<0) /* 0:Digital-Dig loop, 1:Dig-Analog-Dig loop */
+#define CS4215_ENL (1<<1) /* Enable Loopback Testing */
+
+/* Time Slot 5, Parallel Port Register */
+/* Read only here and the same as the in data mode */
+
+/* Time Slot 6, Reserved */
+
+/* Time Slot 7, Version Register */
+#define CS4215_VERSION_MASK 0xf /* Known versions 0/C, 1/D, 2/E */
+
+/* Time Slot 8, Reserved */
+
+/*
+ * Data mode
+ */
+/* Time Slot 1-2: Left Channel Data, 2-3: Right Channel Data */
+
+/* Time Slot 5, Output Setting */
+#define CS4215_LO(v) v /* Left Output Attenuation 0x3f: -94.5 dB */
+#define CS4215_LE (1<<6) /* Line Out Enable */
+#define CS4215_HE (1<<7) /* Headphone Enable */
+
+/* Time Slot 6, Output Setting */
+#define CS4215_RO(v) v /* Right Output Attenuation 0x3f: -94.5 dB */
+#define CS4215_SE (1<<6) /* Speaker Enable */
+#define CS4215_ADI (1<<7) /* A/D Data Invalid: Busy in calibration */
+
+/* Time Slot 7, Input Setting */
+#define CS4215_LG(v) v /* Left Gain Setting 0xf: 22.5 dB */
+#define CS4215_IS (1<<4) /* Input Select: 1=Microphone, 0=Line */
+#define CS4215_OVR (1<<5) /* 1: Over range condition occurred */
+#define CS4215_PIO0 (1<<6) /* Parallel I/O 0 */
+#define CS4215_PIO1 (1<<7)
+
+/* Time Slot 8, Input Setting */
+#define CS4215_RG(v) v /* Right Gain Setting 0xf: 22.5 dB */
+#define CS4215_MA(v) (v<<4) /* Monitor Path Attenuation 0xf: mute */
+
+/***************************************************************************
+ DBRI specific definitions and structures
+****************************************************************************/
+
+/* DBRI main registers */
+#define REG0 0x00 /* Status and Control */
+#define REG1 0x04 /* Mode and Interrupt */
+#define REG2 0x08 /* Parallel IO */
+#define REG3 0x0c /* Test */
+#define REG8 0x20 /* Command Queue Pointer */
+#define REG9 0x24 /* Interrupt Queue Pointer */
+
+#define DBRI_NO_CMDS 64
+#define DBRI_INT_BLK 64
+#define DBRI_NO_DESCS 64
+#define DBRI_NO_PIPES 32
+#define DBRI_MAX_PIPE (DBRI_NO_PIPES - 1)
+
+#define DBRI_REC 0
+#define DBRI_PLAY 1
+#define DBRI_NO_STREAMS 2
+
+/* One transmit/receive descriptor */
+/* When ba != 0 descriptor is used */
+struct dbri_mem {
+ volatile __u32 word1;
+ __u32 ba; /* Transmit/Receive Buffer Address */
+ __u32 nda; /* Next Descriptor Address */
+ volatile __u32 word4;
+};
+
+/* This structure is in a DMA region where it can accessed by both
+ * the CPU and the DBRI
+ */
+struct dbri_dma {
+ s32 cmd[DBRI_NO_CMDS]; /* Place for commands */
+ volatile s32 intr[DBRI_INT_BLK]; /* Interrupt field */
+ struct dbri_mem desc[DBRI_NO_DESCS]; /* Xmit/receive descriptors */
+};
+
+#define dbri_dma_off(member, elem) \
+ ((u32)(unsigned long) \
+ (&(((struct dbri_dma *)0)->member[elem])))
+
+enum in_or_out { PIPEinput, PIPEoutput };
+
+struct dbri_pipe {
+ u32 sdp; /* SDP command word */
+ int nextpipe; /* Next pipe in linked list */
+ int length; /* Length of timeslot (bits) */
+ int first_desc; /* Index of first descriptor */
+ int desc; /* Index of active descriptor */
+ volatile __u32 *recv_fixed_ptr; /* Ptr to receive fixed data */
+};
+
+/* Per stream (playback or record) information */
+struct dbri_streaminfo {
+ struct snd_pcm_substream *substream;
+ u32 dvma_buffer; /* Device view of ALSA DMA buffer */
+ int size; /* Size of DMA buffer */
+ size_t offset; /* offset in user buffer */
+ int pipe; /* Data pipe used */
+ int left_gain; /* mixer elements */
+ int right_gain;
+};
+
+/* This structure holds the information for both chips (DBRI & CS4215) */
+struct snd_dbri {
+ int regs_size, irq; /* Needed for unload */
+ struct of_device *op; /* OF device info */
+ spinlock_t lock;
+
+ struct dbri_dma *dma; /* Pointer to our DMA block */
+ u32 dma_dvma; /* DBRI visible DMA address */
+
+ void __iomem *regs; /* dbri HW regs */
+ int dbri_irqp; /* intr queue pointer */
+
+ struct dbri_pipe pipes[DBRI_NO_PIPES]; /* DBRI's 32 data pipes */
+ int next_desc[DBRI_NO_DESCS]; /* Index of next desc, or -1 */
+ spinlock_t cmdlock; /* Protects cmd queue accesses */
+ s32 *cmdptr; /* Pointer to the last queued cmd */
+
+ int chi_bpf;
+
+ struct cs4215 mm; /* mmcodec special info */
+ /* per stream (playback/record) info */
+ struct dbri_streaminfo stream_info[DBRI_NO_STREAMS];
+};
+
+#define DBRI_MAX_VOLUME 63 /* Output volume */
+#define DBRI_MAX_GAIN 15 /* Input gain */
+
+/* DBRI Reg0 - Status Control Register - defines. (Page 17) */
+#define D_P (1<<15) /* Program command & queue pointer valid */
+#define D_G (1<<14) /* Allow 4-Word SBus Burst */
+#define D_S (1<<13) /* Allow 16-Word SBus Burst */
+#define D_E (1<<12) /* Allow 8-Word SBus Burst */
+#define D_X (1<<7) /* Sanity Timer Disable */
+#define D_T (1<<6) /* Permit activation of the TE interface */
+#define D_N (1<<5) /* Permit activation of the NT interface */
+#define D_C (1<<4) /* Permit activation of the CHI interface */
+#define D_F (1<<3) /* Force Sanity Timer Time-Out */
+#define D_D (1<<2) /* Disable Master Mode */
+#define D_H (1<<1) /* Halt for Analysis */
+#define D_R (1<<0) /* Soft Reset */
+
+/* DBRI Reg1 - Mode and Interrupt Register - defines. (Page 18) */
+#define D_LITTLE_END (1<<8) /* Byte Order */
+#define D_BIG_END (0<<8) /* Byte Order */
+#define D_MRR (1<<4) /* Multiple Error Ack on SBus (read only) */
+#define D_MLE (1<<3) /* Multiple Late Error on SBus (read only) */
+#define D_LBG (1<<2) /* Lost Bus Grant on SBus (read only) */
+#define D_MBE (1<<1) /* Burst Error on SBus (read only) */
+#define D_IR (1<<0) /* Interrupt Indicator (read only) */
+
+/* DBRI Reg2 - Parallel IO Register - defines. (Page 18) */
+#define D_ENPIO3 (1<<7) /* Enable Pin 3 */
+#define D_ENPIO2 (1<<6) /* Enable Pin 2 */
+#define D_ENPIO1 (1<<5) /* Enable Pin 1 */
+#define D_ENPIO0 (1<<4) /* Enable Pin 0 */
+#define D_ENPIO (0xf0) /* Enable all the pins */
+#define D_PIO3 (1<<3) /* Pin 3: 1: Data mode, 0: Ctrl mode */
+#define D_PIO2 (1<<2) /* Pin 2: 1: Onboard PDN */
+#define D_PIO1 (1<<1) /* Pin 1: 0: Reset */
+#define D_PIO0 (1<<0) /* Pin 0: 1: Speakerbox PDN */
+
+/* DBRI Commands (Page 20) */
+#define D_WAIT 0x0 /* Stop execution */
+#define D_PAUSE 0x1 /* Flush long pipes */
+#define D_JUMP 0x2 /* New command queue */
+#define D_IIQ 0x3 /* Initialize Interrupt Queue */
+#define D_REX 0x4 /* Report command execution via interrupt */
+#define D_SDP 0x5 /* Setup Data Pipe */
+#define D_CDP 0x6 /* Continue Data Pipe (reread NULL Pointer) */
+#define D_DTS 0x7 /* Define Time Slot */
+#define D_SSP 0x8 /* Set short Data Pipe */
+#define D_CHI 0x9 /* Set CHI Global Mode */
+#define D_NT 0xa /* NT Command */
+#define D_TE 0xb /* TE Command */
+#define D_CDEC 0xc /* Codec setup */
+#define D_TEST 0xd /* No comment */
+#define D_CDM 0xe /* CHI Data mode command */
+
+/* Special bits for some commands */
+#define D_PIPE(v) ((v)<<0) /* Pipe No.: 0-15 long, 16-21 short */
+
+/* Setup Data Pipe */
+/* IRM */
+#define D_SDP_2SAME (1<<18) /* Report 2nd time in a row value received */
+#define D_SDP_CHANGE (2<<18) /* Report any changes */
+#define D_SDP_EVERY (3<<18) /* Report any changes */
+#define D_SDP_EOL (1<<17) /* EOL interrupt enable */
+#define D_SDP_IDLE (1<<16) /* HDLC idle interrupt enable */
+
+/* Pipe data MODE */
+#define D_SDP_MEM (0<<13) /* To/from memory */
+#define D_SDP_HDLC (2<<13)
+#define D_SDP_HDLC_D (3<<13) /* D Channel (prio control) */
+#define D_SDP_SER (4<<13) /* Serial to serial */
+#define D_SDP_FIXED (6<<13) /* Short only */
+#define D_SDP_MODE(v) ((v)&(7<<13))
+
+#define D_SDP_TO_SER (1<<12) /* Direction */
+#define D_SDP_FROM_SER (0<<12) /* Direction */
+#define D_SDP_MSB (1<<11) /* Bit order within Byte */
+#define D_SDP_LSB (0<<11) /* Bit order within Byte */
+#define D_SDP_P (1<<10) /* Pointer Valid */
+#define D_SDP_A (1<<8) /* Abort */
+#define D_SDP_C (1<<7) /* Clear */
+
+/* Define Time Slot */
+#define D_DTS_VI (1<<17) /* Valid Input Time-Slot Descriptor */
+#define D_DTS_VO (1<<16) /* Valid Output Time-Slot Descriptor */
+#define D_DTS_INS (1<<15) /* Insert Time Slot */
+#define D_DTS_DEL (0<<15) /* Delete Time Slot */
+#define D_DTS_PRVIN(v) ((v)<<10) /* Previous In Pipe */
+#define D_DTS_PRVOUT(v) ((v)<<5) /* Previous Out Pipe */
+
+/* Time Slot defines */
+#define D_TS_LEN(v) ((v)<<24) /* Number of bits in this time slot */
+#define D_TS_CYCLE(v) ((v)<<14) /* Bit Count at start of TS */
+#define D_TS_DI (1<<13) /* Data Invert */
+#define D_TS_1CHANNEL (0<<10) /* Single Channel / Normal mode */
+#define D_TS_MONITOR (2<<10) /* Monitor pipe */
+#define D_TS_NONCONTIG (3<<10) /* Non contiguous mode */
+#define D_TS_ANCHOR (7<<10) /* Starting short pipes */
+#define D_TS_MON(v) ((v)<<5) /* Monitor Pipe */
+#define D_TS_NEXT(v) ((v)<<0) /* Pipe no.: 0-15 long, 16-21 short */
+
+/* Concentration Highway Interface Modes */
+#define D_CHI_CHICM(v) ((v)<<16) /* Clock mode */
+#define D_CHI_IR (1<<15) /* Immediate Interrupt Report */
+#define D_CHI_EN (1<<14) /* CHIL Interrupt enabled */
+#define D_CHI_OD (1<<13) /* Open Drain Enable */
+#define D_CHI_FE (1<<12) /* Sample CHIFS on Rising Frame Edge */
+#define D_CHI_FD (1<<11) /* Frame Drive */
+#define D_CHI_BPF(v) ((v)<<0) /* Bits per Frame */
+
+/* NT: These are here for completeness */
+#define D_NT_FBIT (1<<17) /* Frame Bit */
+#define D_NT_NBF (1<<16) /* Number of bad frames to loose framing */
+#define D_NT_IRM_IMM (1<<15) /* Interrupt Report & Mask: Immediate */
+#define D_NT_IRM_EN (1<<14) /* Interrupt Report & Mask: Enable */
+#define D_NT_ISNT (1<<13) /* Configure interface as NT */
+#define D_NT_FT (1<<12) /* Fixed Timing */
+#define D_NT_EZ (1<<11) /* Echo Channel is Zeros */
+#define D_NT_IFA (1<<10) /* Inhibit Final Activation */
+#define D_NT_ACT (1<<9) /* Activate Interface */
+#define D_NT_MFE (1<<8) /* Multiframe Enable */
+#define D_NT_RLB(v) ((v)<<5) /* Remote Loopback */
+#define D_NT_LLB(v) ((v)<<2) /* Local Loopback */
+#define D_NT_FACT (1<<1) /* Force Activation */
+#define D_NT_ABV (1<<0) /* Activate Bipolar Violation */
+
+/* Codec Setup */
+#define D_CDEC_CK(v) ((v)<<24) /* Clock Select */
+#define D_CDEC_FED(v) ((v)<<12) /* FSCOD Falling Edge Delay */
+#define D_CDEC_RED(v) ((v)<<0) /* FSCOD Rising Edge Delay */
+
+/* Test */
+#define D_TEST_RAM(v) ((v)<<16) /* RAM Pointer */
+#define D_TEST_SIZE(v) ((v)<<11) /* */
+#define D_TEST_ROMONOFF 0x5 /* Toggle ROM opcode monitor on/off */
+#define D_TEST_PROC 0x6 /* Microprocessor test */
+#define D_TEST_SER 0x7 /* Serial-Controller test */
+#define D_TEST_RAMREAD 0x8 /* Copy from Ram to system memory */
+#define D_TEST_RAMWRITE 0x9 /* Copy into Ram from system memory */
+#define D_TEST_RAMBIST 0xa /* RAM Built-In Self Test */
+#define D_TEST_MCBIST 0xb /* Microcontroller Built-In Self Test */
+#define D_TEST_DUMP 0xe /* ROM Dump */
+
+/* CHI Data Mode */
+#define D_CDM_THI (1 << 8) /* Transmit Data on CHIDR Pin */
+#define D_CDM_RHI (1 << 7) /* Receive Data on CHIDX Pin */
+#define D_CDM_RCE (1 << 6) /* Receive on Rising Edge of CHICK */
+#define D_CDM_XCE (1 << 2) /* Transmit Data on Rising Edge of CHICK */
+#define D_CDM_XEN (1 << 1) /* Transmit Highway Enable */
+#define D_CDM_REN (1 << 0) /* Receive Highway Enable */
+
+/* The Interrupts */
+#define D_INTR_BRDY 1 /* Buffer Ready for processing */
+#define D_INTR_MINT 2 /* Marked Interrupt in RD/TD */
+#define D_INTR_IBEG 3 /* Flag to idle transition detected (HDLC) */
+#define D_INTR_IEND 4 /* Idle to flag transition detected (HDLC) */
+#define D_INTR_EOL 5 /* End of List */
+#define D_INTR_CMDI 6 /* Command has bean read */
+#define D_INTR_XCMP 8 /* Transmission of frame complete */
+#define D_INTR_SBRI 9 /* BRI status change info */
+#define D_INTR_FXDT 10 /* Fixed data change */
+#define D_INTR_CHIL 11 /* CHI lost frame sync (channel 36 only) */
+#define D_INTR_COLL 11 /* Unrecoverable D-Channel collision */
+#define D_INTR_DBYT 12 /* Dropped by frame slip */
+#define D_INTR_RBYT 13 /* Repeated by frame slip */
+#define D_INTR_LINT 14 /* Lost Interrupt */
+#define D_INTR_UNDR 15 /* DMA underrun */
+
+#define D_INTR_TE 32
+#define D_INTR_NT 34
+#define D_INTR_CHI 36
+#define D_INTR_CMD 38
+
+#define D_INTR_GETCHAN(v) (((v) >> 24) & 0x3f)
+#define D_INTR_GETCODE(v) (((v) >> 20) & 0xf)
+#define D_INTR_GETCMD(v) (((v) >> 16) & 0xf)
+#define D_INTR_GETVAL(v) ((v) & 0xffff)
+#define D_INTR_GETRVAL(v) ((v) & 0xfffff)
+
+#define D_P_0 0 /* TE receive anchor */
+#define D_P_1 1 /* TE transmit anchor */
+#define D_P_2 2 /* NT transmit anchor */
+#define D_P_3 3 /* NT receive anchor */
+#define D_P_4 4 /* CHI send data */
+#define D_P_5 5 /* CHI receive data */
+#define D_P_6 6 /* */
+#define D_P_7 7 /* */
+#define D_P_8 8 /* */
+#define D_P_9 9 /* */
+#define D_P_10 10 /* */
+#define D_P_11 11 /* */
+#define D_P_12 12 /* */
+#define D_P_13 13 /* */
+#define D_P_14 14 /* */
+#define D_P_15 15 /* */
+#define D_P_16 16 /* CHI anchor pipe */
+#define D_P_17 17 /* CHI send */
+#define D_P_18 18 /* CHI receive */
+#define D_P_19 19 /* CHI receive */
+#define D_P_20 20 /* CHI receive */
+#define D_P_21 21 /* */
+#define D_P_22 22 /* */
+#define D_P_23 23 /* */
+#define D_P_24 24 /* */
+#define D_P_25 25 /* */
+#define D_P_26 26 /* */
+#define D_P_27 27 /* */
+#define D_P_28 28 /* */
+#define D_P_29 29 /* */
+#define D_P_30 30 /* */
+#define D_P_31 31 /* */
+
+/* Transmit descriptor defines */
+#define DBRI_TD_F (1 << 31) /* End of Frame */
+#define DBRI_TD_D (1 << 30) /* Do not append CRC */
+#define DBRI_TD_CNT(v) ((v) << 16) /* Number of valid bytes in the buffer */
+#define DBRI_TD_B (1 << 15) /* Final interrupt */
+#define DBRI_TD_M (1 << 14) /* Marker interrupt */
+#define DBRI_TD_I (1 << 13) /* Transmit Idle Characters */
+#define DBRI_TD_FCNT(v) (v) /* Flag Count */
+#define DBRI_TD_UNR (1 << 3) /* Underrun: transmitter is out of data */
+#define DBRI_TD_ABT (1 << 2) /* Abort: frame aborted */
+#define DBRI_TD_TBC (1 << 0) /* Transmit buffer Complete */
+#define DBRI_TD_STATUS(v) ((v) & 0xff) /* Transmit status */
+ /* Maximum buffer size per TD: almost 8KB */
+#define DBRI_TD_MAXCNT ((1 << 13) - 4)
+
+/* Receive descriptor defines */
+#define DBRI_RD_F (1 << 31) /* End of Frame */
+#define DBRI_RD_C (1 << 30) /* Completed buffer */
+#define DBRI_RD_B (1 << 15) /* Final interrupt */
+#define DBRI_RD_M (1 << 14) /* Marker interrupt */
+#define DBRI_RD_BCNT(v) (v) /* Buffer size */
+#define DBRI_RD_CRC (1 << 7) /* 0: CRC is correct */
+#define DBRI_RD_BBC (1 << 6) /* 1: Bad Byte received */
+#define DBRI_RD_ABT (1 << 5) /* Abort: frame aborted */
+#define DBRI_RD_OVRN (1 << 3) /* Overrun: data lost */
+#define DBRI_RD_STATUS(v) ((v) & 0xff) /* Receive status */
+#define DBRI_RD_CNT(v) (((v) >> 16) & 0x1fff) /* Valid bytes in the buffer */
+
+/* stream_info[] access */
+/* Translate the ALSA direction into the array index */
+#define DBRI_STREAMNO(substream) \
+ (substream->stream == \
+ SNDRV_PCM_STREAM_PLAYBACK ? DBRI_PLAY: DBRI_REC)
+
+/* Return a pointer to dbri_streaminfo */
+#define DBRI_STREAM(dbri, substream) \
+ &dbri->stream_info[DBRI_STREAMNO(substream)]
+
+/*
+ * Short data pipes transmit LSB first. The CS4215 receives MSB first. Grrr.
+ * So we have to reverse the bits. Note: not all bit lengths are supported
+ */
+static __u32 reverse_bytes(__u32 b, int len)
+{
+ switch (len) {
+ case 32:
+ b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16);
+ case 16:
+ b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8);
+ case 8:
+ b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4);
+ case 4:
+ b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2);
+ case 2:
+ b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1);
+ case 1:
+ case 0:
+ break;
+ default:
+ printk(KERN_ERR "DBRI reverse_bytes: unsupported length\n");
+ };
+
+ return b;
+}
+
+/*
+****************************************************************************
+************** DBRI initialization and command synchronization *************
+****************************************************************************
+
+Commands are sent to the DBRI by building a list of them in memory,
+then writing the address of the first list item to DBRI register 8.
+The list is terminated with a WAIT command, which generates a
+CPU interrupt to signal completion.
+
+Since the DBRI can run in parallel with the CPU, several means of
+synchronization present themselves. The method implemented here uses
+the dbri_cmdwait() to wait for execution of batch of sent commands.
+
+A circular command buffer is used here. A new command is being added
+while another can be executed. The scheme works by adding two WAIT commands
+after each sent batch of commands. When the next batch is prepared it is
+added after the WAIT commands then the WAITs are replaced with single JUMP
+command to the new batch. The the DBRI is forced to reread the last WAIT
+command (replaced by the JUMP by then). If the DBRI is still executing
+previous commands the request to reread the WAIT command is ignored.
+
+Every time a routine wants to write commands to the DBRI, it must
+first call dbri_cmdlock() and get pointer to a free space in
+dbri->dma->cmd buffer. After this, the commands can be written to
+the buffer, and dbri_cmdsend() is called with the final pointer value
+to send them to the DBRI.
+
+*/
+
+#define MAXLOOPS 20
+/*
+ * Wait for the current command string to execute
+ */
+static void dbri_cmdwait(struct snd_dbri *dbri)
+{
+ int maxloops = MAXLOOPS;
+ unsigned long flags;
+
+ /* Delay if previous commands are still being processed */
+ spin_lock_irqsave(&dbri->lock, flags);
+ while ((--maxloops) > 0 && (sbus_readl(dbri->regs + REG0) & D_P)) {
+ spin_unlock_irqrestore(&dbri->lock, flags);
+ msleep_interruptible(1);
+ spin_lock_irqsave(&dbri->lock, flags);
+ }
+ spin_unlock_irqrestore(&dbri->lock, flags);
+
+ if (maxloops == 0)
+ printk(KERN_ERR "DBRI: Chip never completed command buffer\n");
+ else
+ dprintk(D_CMD, "Chip completed command buffer (%d)\n",
+ MAXLOOPS - maxloops - 1);
+}
+/*
+ * Lock the command queue and return pointer to space for len cmd words
+ * It locks the cmdlock spinlock.
+ */
+static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len)
+{
+ /* Space for 2 WAIT cmds (replaced later by 1 JUMP cmd) */
+ len += 2;
+ spin_lock(&dbri->cmdlock);
+ if (dbri->cmdptr - dbri->dma->cmd + len < DBRI_NO_CMDS - 2)
+ return dbri->cmdptr + 2;
+ else if (len < sbus_readl(dbri->regs + REG8) - dbri->dma_dvma)
+ return dbri->dma->cmd;
+ else
+ printk(KERN_ERR "DBRI: no space for commands.");
+
+ return NULL;
+}
+
+/*
+ * Send prepared cmd string. It works by writing a JUMP cmd into
+ * the last WAIT cmd and force DBRI to reread the cmd.
+ * The JUMP cmd points to the new cmd string.
+ * It also releases the cmdlock spinlock.
+ *
+ * Lock must be held before calling this.
+ */
+static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len)
+{
+ s32 tmp, addr;
+ static int wait_id = 0;
+
+ wait_id++;
+ wait_id &= 0xffff; /* restrict it to a 16 bit counter. */
+ *(cmd) = DBRI_CMD(D_WAIT, 1, wait_id);
+ *(cmd+1) = DBRI_CMD(D_WAIT, 1, wait_id);
+
+ /* Replace the last command with JUMP */
+ addr = dbri->dma_dvma + (cmd - len - dbri->dma->cmd) * sizeof(s32);
+ *(dbri->cmdptr+1) = addr;
+ *(dbri->cmdptr) = DBRI_CMD(D_JUMP, 0, 0);
+
+#ifdef DBRI_DEBUG
+ if (cmd > dbri->cmdptr) {
+ s32 *ptr;
+
+ for (ptr = dbri->cmdptr; ptr < cmd+2; ptr++)
+ dprintk(D_CMD, "cmd: %lx:%08x\n",
+ (unsigned long)ptr, *ptr);
+ } else {
+ s32 *ptr = dbri->cmdptr;
+
+ dprintk(D_CMD, "cmd: %lx:%08x\n", (unsigned long)ptr, *ptr);
+ ptr++;
+ dprintk(D_CMD, "cmd: %lx:%08x\n", (unsigned long)ptr, *ptr);
+ for (ptr = dbri->dma->cmd; ptr < cmd+2; ptr++)
+ dprintk(D_CMD, "cmd: %lx:%08x\n",
+ (unsigned long)ptr, *ptr);
+ }
+#endif
+
+ /* Reread the last command */
+ tmp = sbus_readl(dbri->regs + REG0);
+ tmp |= D_P;
+ sbus_writel(tmp, dbri->regs + REG0);
+
+ dbri->cmdptr = cmd;
+ spin_unlock(&dbri->cmdlock);
+}
+
+/* Lock must be held when calling this */
+static void dbri_reset(struct snd_dbri *dbri)
+{
+ int i;
+ u32 tmp;
+
+ dprintk(D_GEN, "reset 0:%x 2:%x 8:%x 9:%x\n",
+ sbus_readl(dbri->regs + REG0),
+ sbus_readl(dbri->regs + REG2),
+ sbus_readl(dbri->regs + REG8), sbus_readl(dbri->regs + REG9));
+
+ sbus_writel(D_R, dbri->regs + REG0); /* Soft Reset */
+ for (i = 0; (sbus_readl(dbri->regs + REG0) & D_R) && i < 64; i++)
+ udelay(10);
+
+ /* A brute approach - DBRI falls back to working burst size by itself
+ * On SS20 D_S does not work, so do not try so high. */
+ tmp = sbus_readl(dbri->regs + REG0);
+ tmp |= D_G | D_E;
+ tmp &= ~D_S;
+ sbus_writel(tmp, dbri->regs + REG0);
+}
+
+/* Lock must not be held before calling this */
+static void __devinit dbri_initialize(struct snd_dbri *dbri)
+{
+ s32 *cmd;
+ u32 dma_addr;
+ unsigned long flags;
+ int n;
+
+ spin_lock_irqsave(&dbri->lock, flags);
+
+ dbri_reset(dbri);
+
+ /* Initialize pipes */
+ for (n = 0; n < DBRI_NO_PIPES; n++)
+ dbri->pipes[n].desc = dbri->pipes[n].first_desc = -1;
+
+ spin_lock_init(&dbri->cmdlock);
+ /*
+ * Initialize the interrupt ring buffer.
+ */
+ dma_addr = dbri->dma_dvma + dbri_dma_off(intr, 0);
+ dbri->dma->intr[0] = dma_addr;
+ dbri->dbri_irqp = 1;
+ /*
+ * Set up the interrupt queue
+ */
+ spin_lock(&dbri->cmdlock);
+ cmd = dbri->cmdptr = dbri->dma->cmd;
+ *(cmd++) = DBRI_CMD(D_IIQ, 0, 0);
+ *(cmd++) = dma_addr;
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+ dbri->cmdptr = cmd;
+ *(cmd++) = DBRI_CMD(D_WAIT, 1, 0);
+ *(cmd++) = DBRI_CMD(D_WAIT, 1, 0);
+ dma_addr = dbri->dma_dvma + dbri_dma_off(cmd, 0);
+ sbus_writel(dma_addr, dbri->regs + REG8);
+ spin_unlock(&dbri->cmdlock);
+
+ spin_unlock_irqrestore(&dbri->lock, flags);
+ dbri_cmdwait(dbri);
+}
+
+/*
+****************************************************************************
+************************** DBRI data pipe management ***********************
+****************************************************************************
+
+While DBRI control functions use the command and interrupt buffers, the
+main data path takes the form of data pipes, which can be short (command
+and interrupt driven), or long (attached to DMA buffers). These functions
+provide a rudimentary means of setting up and managing the DBRI's pipes,
+but the calling functions have to make sure they respect the pipes' linked
+list ordering, among other things. The transmit and receive functions
+here interface closely with the transmit and receive interrupt code.
+
+*/
+static inline int pipe_active(struct snd_dbri *dbri, int pipe)
+{
+ return ((pipe >= 0) && (dbri->pipes[pipe].desc != -1));
+}
+
+/* reset_pipe(dbri, pipe)
+ *
+ * Called on an in-use pipe to clear anything being transmitted or received
+ * Lock must be held before calling this.
+ */
+static void reset_pipe(struct snd_dbri *dbri, int pipe)
+{
+ int sdp;
+ int desc;
+ s32 *cmd;
+
+ if (pipe < 0 || pipe > DBRI_MAX_PIPE) {
+ printk(KERN_ERR "DBRI: reset_pipe called with "
+ "illegal pipe number\n");
+ return;
+ }
+
+ sdp = dbri->pipes[pipe].sdp;
+ if (sdp == 0) {
+ printk(KERN_ERR "DBRI: reset_pipe called "
+ "on uninitialized pipe\n");
+ return;
+ }
+
+ cmd = dbri_cmdlock(dbri, 3);
+ *(cmd++) = DBRI_CMD(D_SDP, 0, sdp | D_SDP_C | D_SDP_P);
+ *(cmd++) = 0;
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+ dbri_cmdsend(dbri, cmd, 3);
+
+ desc = dbri->pipes[pipe].first_desc;
+ if (desc >= 0)
+ do {
+ dbri->dma->desc[desc].ba = 0;
+ dbri->dma->desc[desc].nda = 0;
+ desc = dbri->next_desc[desc];
+ } while (desc != -1 && desc != dbri->pipes[pipe].first_desc);
+
+ dbri->pipes[pipe].desc = -1;
+ dbri->pipes[pipe].first_desc = -1;
+}
+
+/*
+ * Lock must be held before calling this.
+ */
+static void setup_pipe(struct snd_dbri *dbri, int pipe, int sdp)
+{
+ if (pipe < 0 || pipe > DBRI_MAX_PIPE) {
+ printk(KERN_ERR "DBRI: setup_pipe called "
+ "with illegal pipe number\n");
+ return;
+ }
+
+ if ((sdp & 0xf800) != sdp) {
+ printk(KERN_ERR "DBRI: setup_pipe called "
+ "with strange SDP value\n");
+ /* sdp &= 0xf800; */
+ }
+
+ /* If this is a fixed receive pipe, arrange for an interrupt
+ * every time its data changes
+ */
+ if (D_SDP_MODE(sdp) == D_SDP_FIXED && !(sdp & D_SDP_TO_SER))
+ sdp |= D_SDP_CHANGE;
+
+ sdp |= D_PIPE(pipe);
+ dbri->pipes[pipe].sdp = sdp;
+ dbri->pipes[pipe].desc = -1;
+ dbri->pipes[pipe].first_desc = -1;
+
+ reset_pipe(dbri, pipe);
+}
+
+/*
+ * Lock must be held before calling this.
+ */
+static void link_time_slot(struct snd_dbri *dbri, int pipe,
+ int prevpipe, int nextpipe,
+ int length, int cycle)
+{
+ s32 *cmd;
+ int val;
+
+ if (pipe < 0 || pipe > DBRI_MAX_PIPE
+ || prevpipe < 0 || prevpipe > DBRI_MAX_PIPE
+ || nextpipe < 0 || nextpipe > DBRI_MAX_PIPE) {
+ printk(KERN_ERR
+ "DBRI: link_time_slot called with illegal pipe number\n");
+ return;
+ }
+
+ if (dbri->pipes[pipe].sdp == 0
+ || dbri->pipes[prevpipe].sdp == 0
+ || dbri->pipes[nextpipe].sdp == 0) {
+ printk(KERN_ERR "DBRI: link_time_slot called "
+ "on uninitialized pipe\n");
+ return;
+ }
+
+ dbri->pipes[prevpipe].nextpipe = pipe;
+ dbri->pipes[pipe].nextpipe = nextpipe;
+ dbri->pipes[pipe].length = length;
+
+ cmd = dbri_cmdlock(dbri, 4);
+
+ if (dbri->pipes[pipe].sdp & D_SDP_TO_SER) {
+ /* Deal with CHI special case:
+ * "If transmission on edges 0 or 1 is desired, then cycle n
+ * (where n = # of bit times per frame...) must be used."
+ * - DBRI data sheet, page 11
+ */
+ if (prevpipe == 16 && cycle == 0)
+ cycle = dbri->chi_bpf;
+
+ val = D_DTS_VO | D_DTS_INS | D_DTS_PRVOUT(prevpipe) | pipe;
+ *(cmd++) = DBRI_CMD(D_DTS, 0, val);
+ *(cmd++) = 0;
+ *(cmd++) =
+ D_TS_LEN(length) | D_TS_CYCLE(cycle) | D_TS_NEXT(nextpipe);
+ } else {
+ val = D_DTS_VI | D_DTS_INS | D_DTS_PRVIN(prevpipe) | pipe;
+ *(cmd++) = DBRI_CMD(D_DTS, 0, val);
+ *(cmd++) =
+ D_TS_LEN(length) | D_TS_CYCLE(cycle) | D_TS_NEXT(nextpipe);
+ *(cmd++) = 0;
+ }
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+
+ dbri_cmdsend(dbri, cmd, 4);
+}
+
+#if 0
+/*
+ * Lock must be held before calling this.
+ */
+static void unlink_time_slot(struct snd_dbri *dbri, int pipe,
+ enum in_or_out direction, int prevpipe,
+ int nextpipe)
+{
+ s32 *cmd;
+ int val;
+
+ if (pipe < 0 || pipe > DBRI_MAX_PIPE
+ || prevpipe < 0 || prevpipe > DBRI_MAX_PIPE
+ || nextpipe < 0 || nextpipe > DBRI_MAX_PIPE) {
+ printk(KERN_ERR
+ "DBRI: unlink_time_slot called with illegal pipe number\n");
+ return;
+ }
+
+ cmd = dbri_cmdlock(dbri, 4);
+
+ if (direction == PIPEinput) {
+ val = D_DTS_VI | D_DTS_DEL | D_DTS_PRVIN(prevpipe) | pipe;
+ *(cmd++) = DBRI_CMD(D_DTS, 0, val);
+ *(cmd++) = D_TS_NEXT(nextpipe);
+ *(cmd++) = 0;
+ } else {
+ val = D_DTS_VO | D_DTS_DEL | D_DTS_PRVOUT(prevpipe) | pipe;
+ *(cmd++) = DBRI_CMD(D_DTS, 0, val);
+ *(cmd++) = 0;
+ *(cmd++) = D_TS_NEXT(nextpipe);
+ }
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+
+ dbri_cmdsend(dbri, cmd, 4);
+}
+#endif
+
+/* xmit_fixed() / recv_fixed()
+ *
+ * Transmit/receive data on a "fixed" pipe - i.e, one whose contents are not
+ * expected to change much, and which we don't need to buffer.
+ * The DBRI only interrupts us when the data changes (receive pipes),
+ * or only changes the data when this function is called (transmit pipes).
+ * Only short pipes (numbers 16-31) can be used in fixed data mode.
+ *
+ * These function operate on a 32-bit field, no matter how large
+ * the actual time slot is. The interrupt handler takes care of bit
+ * ordering and alignment. An 8-bit time slot will always end up
+ * in the low-order 8 bits, filled either MSB-first or LSB-first,
+ * depending on the settings passed to setup_pipe().
+ *
+ * Lock must not be held before calling it.
+ */
+static void xmit_fixed(struct snd_dbri *dbri, int pipe, unsigned int data)
+{
+ s32 *cmd;
+ unsigned long flags;
+
+ if (pipe < 16 || pipe > DBRI_MAX_PIPE) {
+ printk(KERN_ERR "DBRI: xmit_fixed: Illegal pipe number\n");
+ return;
+ }
+
+ if (D_SDP_MODE(dbri->pipes[pipe].sdp) == 0) {
+ printk(KERN_ERR "DBRI: xmit_fixed: "
+ "Uninitialized pipe %d\n", pipe);
+ return;
+ }
+
+ if (D_SDP_MODE(dbri->pipes[pipe].sdp) != D_SDP_FIXED) {
+ printk(KERN_ERR "DBRI: xmit_fixed: Non-fixed pipe %d\n", pipe);
+ return;
+ }
+
+ if (!(dbri->pipes[pipe].sdp & D_SDP_TO_SER)) {
+ printk(KERN_ERR "DBRI: xmit_fixed: Called on receive pipe %d\n",
+ pipe);
+ return;
+ }
+
+ /* DBRI short pipes always transmit LSB first */
+
+ if (dbri->pipes[pipe].sdp & D_SDP_MSB)
+ data = reverse_bytes(data, dbri->pipes[pipe].length);
+
+ cmd = dbri_cmdlock(dbri, 3);
+
+ *(cmd++) = DBRI_CMD(D_SSP, 0, pipe);
+ *(cmd++) = data;
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+
+ spin_lock_irqsave(&dbri->lock, flags);
+ dbri_cmdsend(dbri, cmd, 3);
+ spin_unlock_irqrestore(&dbri->lock, flags);
+ dbri_cmdwait(dbri);
+
+}
+
+static void recv_fixed(struct snd_dbri *dbri, int pipe, volatile __u32 *ptr)
+{
+ if (pipe < 16 || pipe > DBRI_MAX_PIPE) {
+ printk(KERN_ERR "DBRI: recv_fixed called with "
+ "illegal pipe number\n");
+ return;
+ }
+
+ if (D_SDP_MODE(dbri->pipes[pipe].sdp) != D_SDP_FIXED) {
+ printk(KERN_ERR "DBRI: recv_fixed called on "
+ "non-fixed pipe %d\n", pipe);
+ return;
+ }
+
+ if (dbri->pipes[pipe].sdp & D_SDP_TO_SER) {
+ printk(KERN_ERR "DBRI: recv_fixed called on "
+ "transmit pipe %d\n", pipe);
+ return;
+ }
+
+ dbri->pipes[pipe].recv_fixed_ptr = ptr;
+}
+
+/* setup_descs()
+ *
+ * Setup transmit/receive data on a "long" pipe - i.e, one associated
+ * with a DMA buffer.
+ *
+ * Only pipe numbers 0-15 can be used in this mode.
+ *
+ * This function takes a stream number pointing to a data buffer,
+ * and work by building chains of descriptors which identify the
+ * data buffers. Buffers too large for a single descriptor will
+ * be spread across multiple descriptors.
+ *
+ * All descriptors create a ring buffer.
+ *
+ * Lock must be held before calling this.
+ */
+static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period)
+{
+ struct dbri_streaminfo *info = &dbri->stream_info[streamno];
+ __u32 dvma_buffer;
+ int desc;
+ int len;
+ int first_desc = -1;
+ int last_desc = -1;
+
+ if (info->pipe < 0 || info->pipe > 15) {
+ printk(KERN_ERR "DBRI: setup_descs: Illegal pipe number\n");
+ return -2;
+ }
+
+ if (dbri->pipes[info->pipe].sdp == 0) {
+ printk(KERN_ERR "DBRI: setup_descs: Uninitialized pipe %d\n",
+ info->pipe);
+ return -2;
+ }
+
+ dvma_buffer = info->dvma_buffer;
+ len = info->size;
+
+ if (streamno == DBRI_PLAY) {
+ if (!(dbri->pipes[info->pipe].sdp & D_SDP_TO_SER)) {
+ printk(KERN_ERR "DBRI: setup_descs: "
+ "Called on receive pipe %d\n", info->pipe);
+ return -2;
+ }
+ } else {
+ if (dbri->pipes[info->pipe].sdp & D_SDP_TO_SER) {
+ printk(KERN_ERR
+ "DBRI: setup_descs: Called on transmit pipe %d\n",
+ info->pipe);
+ return -2;
+ }
+ /* Should be able to queue multiple buffers
+ * to receive on a pipe
+ */
+ if (pipe_active(dbri, info->pipe)) {
+ printk(KERN_ERR "DBRI: recv_on_pipe: "
+ "Called on active pipe %d\n", info->pipe);
+ return -2;
+ }
+
+ /* Make sure buffer size is multiple of four */
+ len &= ~3;
+ }
+
+ /* Free descriptors if pipe has any */
+ desc = dbri->pipes[info->pipe].first_desc;
+ if (desc >= 0)
+ do {
+ dbri->dma->desc[desc].ba = 0;
+ dbri->dma->desc[desc].nda = 0;
+ desc = dbri->next_desc[desc];
+ } while (desc != -1 &&
+ desc != dbri->pipes[info->pipe].first_desc);
+
+ dbri->pipes[info->pipe].desc = -1;
+ dbri->pipes[info->pipe].first_desc = -1;
+
+ desc = 0;
+ while (len > 0) {
+ int mylen;
+
+ for (; desc < DBRI_NO_DESCS; desc++) {
+ if (!dbri->dma->desc[desc].ba)
+ break;
+ }
+
+ if (desc == DBRI_NO_DESCS) {
+ printk(KERN_ERR "DBRI: setup_descs: No descriptors\n");
+ return -1;
+ }
+
+ if (len > DBRI_TD_MAXCNT)
+ mylen = DBRI_TD_MAXCNT; /* 8KB - 4 */
+ else
+ mylen = len;
+
+ if (mylen > period)
+ mylen = period;
+
+ dbri->next_desc[desc] = -1;
+ dbri->dma->desc[desc].ba = dvma_buffer;
+ dbri->dma->desc[desc].nda = 0;
+
+ if (streamno == DBRI_PLAY) {
+ dbri->dma->desc[desc].word1 = DBRI_TD_CNT(mylen);
+ dbri->dma->desc[desc].word4 = 0;
+ dbri->dma->desc[desc].word1 |= DBRI_TD_F | DBRI_TD_B;
+ } else {
+ dbri->dma->desc[desc].word1 = 0;
+ dbri->dma->desc[desc].word4 =
+ DBRI_RD_B | DBRI_RD_BCNT(mylen);
+ }
+
+ if (first_desc == -1)
+ first_desc = desc;
+ else {
+ dbri->next_desc[last_desc] = desc;
+ dbri->dma->desc[last_desc].nda =
+ dbri->dma_dvma + dbri_dma_off(desc, desc);
+ }
+
+ last_desc = desc;
+ dvma_buffer += mylen;
+ len -= mylen;
+ }
+
+ if (first_desc == -1 || last_desc == -1) {
+ printk(KERN_ERR "DBRI: setup_descs: "
+ " Not enough descriptors available\n");
+ return -1;
+ }
+
+ dbri->dma->desc[last_desc].nda =
+ dbri->dma_dvma + dbri_dma_off(desc, first_desc);
+ dbri->next_desc[last_desc] = first_desc;
+ dbri->pipes[info->pipe].first_desc = first_desc;
+ dbri->pipes[info->pipe].desc = first_desc;
+
+#ifdef DBRI_DEBUG
+ for (desc = first_desc; desc != -1;) {
+ dprintk(D_DESC, "DESC %d: %08x %08x %08x %08x\n",
+ desc,
+ dbri->dma->desc[desc].word1,
+ dbri->dma->desc[desc].ba,
+ dbri->dma->desc[desc].nda, dbri->dma->desc[desc].word4);
+ desc = dbri->next_desc[desc];
+ if (desc == first_desc)
+ break;
+ }
+#endif
+ return 0;
+}
+
+/*
+****************************************************************************
+************************** DBRI - CHI interface ****************************
+****************************************************************************
+
+The CHI is a four-wire (clock, frame sync, data in, data out) time-division
+multiplexed serial interface which the DBRI can operate in either master
+(give clock/frame sync) or slave (take clock/frame sync) mode.
+
+*/
+
+enum master_or_slave { CHImaster, CHIslave };
+
+/*
+ * Lock must not be held before calling it.
+ */
+static void reset_chi(struct snd_dbri *dbri,
+ enum master_or_slave master_or_slave,
+ int bits_per_frame)
+{
+ s32 *cmd;
+ int val;
+
+ /* Set CHI Anchor: Pipe 16 */
+
+ cmd = dbri_cmdlock(dbri, 4);
+ val = D_DTS_VO | D_DTS_VI | D_DTS_INS
+ | D_DTS_PRVIN(16) | D_PIPE(16) | D_DTS_PRVOUT(16);
+ *(cmd++) = DBRI_CMD(D_DTS, 0, val);
+ *(cmd++) = D_TS_ANCHOR | D_TS_NEXT(16);
+ *(cmd++) = D_TS_ANCHOR | D_TS_NEXT(16);
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+ dbri_cmdsend(dbri, cmd, 4);
+
+ dbri->pipes[16].sdp = 1;
+ dbri->pipes[16].nextpipe = 16;
+
+ cmd = dbri_cmdlock(dbri, 4);
+
+ if (master_or_slave == CHIslave) {
+ /* Setup DBRI for CHI Slave - receive clock, frame sync (FS)
+ *
+ * CHICM = 0 (slave mode, 8 kHz frame rate)
+ * IR = give immediate CHI status interrupt
+ * EN = give CHI status interrupt upon change
+ */
+ *(cmd++) = DBRI_CMD(D_CHI, 0, D_CHI_CHICM(0));
+ } else {
+ /* Setup DBRI for CHI Master - generate clock, FS
+ *
+ * BPF = bits per 8 kHz frame
+ * 12.288 MHz / CHICM_divisor = clock rate
+ * FD = 1 - drive CHIFS on rising edge of CHICK
+ */
+ int clockrate = bits_per_frame * 8;
+ int divisor = 12288 / clockrate;
+
+ if (divisor > 255 || divisor * clockrate != 12288)
+ printk(KERN_ERR "DBRI: illegal bits_per_frame "
+ "in setup_chi\n");
+
+ *(cmd++) = DBRI_CMD(D_CHI, 0, D_CHI_CHICM(divisor) | D_CHI_FD
+ | D_CHI_BPF(bits_per_frame));
+ }
+
+ dbri->chi_bpf = bits_per_frame;
+
+ /* CHI Data Mode
+ *
+ * RCE = 0 - receive on falling edge of CHICK
+ * XCE = 1 - transmit on rising edge of CHICK
+ * XEN = 1 - enable transmitter
+ * REN = 1 - enable receiver
+ */
+
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+ *(cmd++) = DBRI_CMD(D_CDM, 0, D_CDM_XCE | D_CDM_XEN | D_CDM_REN);
+ *(cmd++) = DBRI_CMD(D_PAUSE, 0, 0);
+
+ dbri_cmdsend(dbri, cmd, 4);
+}
+
+/*
+****************************************************************************
+*********************** CS4215 audio codec management **********************
+****************************************************************************
+
+In the standard SPARC audio configuration, the CS4215 codec is attached
+to the DBRI via the CHI interface and few of the DBRI's PIO pins.
+
+ * Lock must not be held before calling it.
+
+*/
+static __devinit void cs4215_setup_pipes(struct snd_dbri *dbri)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&dbri->lock, flags);
+ /*
+ * Data mode:
+ * Pipe 4: Send timeslots 1-4 (audio data)
+ * Pipe 20: Send timeslots 5-8 (part of ctrl data)
+ * Pipe 6: Receive timeslots 1-4 (audio data)
+ * Pipe 21: Receive timeslots 6-7. We can only receive 20 bits via
+ * interrupt, and the rest of the data (slot 5 and 8) is
+ * not relevant for us (only for doublechecking).
+ *
+ * Control mode:
+ * Pipe 17: Send timeslots 1-4 (slots 5-8 are read only)
+ * Pipe 18: Receive timeslot 1 (clb).
+ * Pipe 19: Receive timeslot 7 (version).
+ */
+
+ setup_pipe(dbri, 4, D_SDP_MEM | D_SDP_TO_SER | D_SDP_MSB);
+ setup_pipe(dbri, 20, D_SDP_FIXED | D_SDP_TO_SER | D_SDP_MSB);
+ setup_pipe(dbri, 6, D_SDP_MEM | D_SDP_FROM_SER | D_SDP_MSB);
+ setup_pipe(dbri, 21, D_SDP_FIXED | D_SDP_FROM_SER | D_SDP_MSB);
+
+ setup_pipe(dbri, 17, D_SDP_FIXED | D_SDP_TO_SER | D_SDP_MSB);
+ setup_pipe(dbri, 18, D_SDP_FIXED | D_SDP_FROM_SER | D_SDP_MSB);
+ setup_pipe(dbri, 19, D_SDP_FIXED | D_SDP_FROM_SER | D_SDP_MSB);
+ spin_unlock_irqrestore(&dbri->lock, flags);
+
+ dbri_cmdwait(dbri);
+}
+
+static __devinit int cs4215_init_data(struct cs4215 *mm)
+{
+ /*
+ * No action, memory resetting only.
+ *
+ * Data Time Slot 5-8
+ * Speaker,Line and Headphone enable. Gain set to the half.
+ * Input is mike.
+ */
+ mm->data[0] = CS4215_LO(0x20) | CS4215_HE | CS4215_LE;
+ mm->data[1] = CS4215_RO(0x20) | CS4215_SE;
+ mm->data[2] = CS4215_LG(0x8) | CS4215_IS | CS4215_PIO0 | CS4215_PIO1;
+ mm->data[3] = CS4215_RG(0x8) | CS4215_MA(0xf);
+
+ /*
+ * Control Time Slot 1-4
+ * 0: Default I/O voltage scale
+ * 1: 8 bit ulaw, 8kHz, mono, high pass filter disabled
+ * 2: Serial enable, CHI master, 128 bits per frame, clock 1
+ * 3: Tests disabled
+ */
+ mm->ctrl[0] = CS4215_RSRVD_1 | CS4215_MLB;
+ mm->ctrl[1] = CS4215_DFR_ULAW | CS4215_FREQ[0].csval;
+ mm->ctrl[2] = CS4215_XCLK | CS4215_BSEL_128 | CS4215_FREQ[0].xtal;
+ mm->ctrl[3] = 0;
+
+ mm->status = 0;
+ mm->version = 0xff;
+ mm->precision = 8; /* For ULAW */
+ mm->channels = 1;
+
+ return 0;
+}
+
+static void cs4215_setdata(struct snd_dbri *dbri, int muted)
+{
+ if (muted) {
+ dbri->mm.data[0] |= 63;
+ dbri->mm.data[1] |= 63;
+ dbri->mm.data[2] &= ~15;
+ dbri->mm.data[3] &= ~15;
+ } else {
+ /* Start by setting the playback attenuation. */
+ struct dbri_streaminfo *info = &dbri->stream_info[DBRI_PLAY];
+ int left_gain = info->left_gain & 0x3f;
+ int right_gain = info->right_gain & 0x3f;
+
+ dbri->mm.data[0] &= ~0x3f; /* Reset the volume bits */
+ dbri->mm.data[1] &= ~0x3f;
+ dbri->mm.data[0] |= (DBRI_MAX_VOLUME - left_gain);
+ dbri->mm.data[1] |= (DBRI_MAX_VOLUME - right_gain);
+
+ /* Now set the recording gain. */
+ info = &dbri->stream_info[DBRI_REC];
+ left_gain = info->left_gain & 0xf;
+ right_gain = info->right_gain & 0xf;
+ dbri->mm.data[2] |= CS4215_LG(left_gain);
+ dbri->mm.data[3] |= CS4215_RG(right_gain);
+ }
+
+ xmit_fixed(dbri, 20, *(int *)dbri->mm.data);
+}
+
+/*
+ * Set the CS4215 to data mode.
+ */
+static void cs4215_open(struct snd_dbri *dbri)
+{
+ int data_width;
+ u32 tmp;
+ unsigned long flags;
+
+ dprintk(D_MM, "cs4215_open: %d channels, %d bits\n",
+ dbri->mm.channels, dbri->mm.precision);
+
+ /* Temporarily mute outputs, and wait 1/8000 sec (125 us)
+ * to make sure this takes. This avoids clicking noises.
+ */
+
+ cs4215_setdata(dbri, 1);
+ udelay(125);
+
+ /*
+ * Data mode:
+ * Pipe 4: Send timeslots 1-4 (audio data)
+ * Pipe 20: Send timeslots 5-8 (part of ctrl data)
+ * Pipe 6: Receive timeslots 1-4 (audio data)
+ * Pipe 21: Receive timeslots 6-7. We can only receive 20 bits via
+ * interrupt, and the rest of the data (slot 5 and 8) is
+ * not relevant for us (only for doublechecking).
+ *
+ * Just like in control mode, the time slots are all offset by eight
+ * bits. The CS4215, it seems, observes TSIN (the delayed signal)
+ * even if it's the CHI master. Don't ask me...
+ */
+ spin_lock_irqsave(&dbri->lock, flags);
+ tmp = sbus_readl(dbri->regs + REG0);
+ tmp &= ~(D_C); /* Disable CHI */
+ sbus_writel(tmp, dbri->regs + REG0);
+
+ /* Switch CS4215 to data mode - set PIO3 to 1 */
+ sbus_writel(D_ENPIO | D_PIO1 | D_PIO3 |
+ (dbri->mm.onboard ? D_PIO0 : D_PIO2), dbri->regs + REG2);
+
+ reset_chi(dbri, CHIslave, 128);
+
+ /* Note: this next doesn't work for 8-bit stereo, because the two
+ * channels would be on timeslots 1 and 3, with 2 and 4 idle.
+ * (See CS4215 datasheet Fig 15)
+ *
+ * DBRI non-contiguous mode would be required to make this work.
+ */
+ data_width = dbri->mm.channels * dbri->mm.precision;
+
+ link_time_slot(dbri, 4, 16, 16, data_width, dbri->mm.offset);
+ link_time_slot(dbri, 20, 4, 16, 32, dbri->mm.offset + 32);
+ link_time_slot(dbri, 6, 16, 16, data_width, dbri->mm.offset);
+ link_time_slot(dbri, 21, 6, 16, 16, dbri->mm.offset + 40);
+
+ /* FIXME: enable CHI after _setdata? */
+ tmp = sbus_readl(dbri->regs + REG0);
+ tmp |= D_C; /* Enable CHI */
+ sbus_writel(tmp, dbri->regs + REG0);
+ spin_unlock_irqrestore(&dbri->lock, flags);
+
+ cs4215_setdata(dbri, 0);
+}
+
+/*
+ * Send the control information (i.e. audio format)
+ */
+static int cs4215_setctrl(struct snd_dbri *dbri)
+{
+ int i, val;
+ u32 tmp;
+ unsigned long flags;
+
+ /* FIXME - let the CPU do something useful during these delays */
+
+ /* Temporarily mute outputs, and wait 1/8000 sec (125 us)
+ * to make sure this takes. This avoids clicking noises.
+ */
+ cs4215_setdata(dbri, 1);
+ udelay(125);
+
+ /*
+ * Enable Control mode: Set DBRI's PIO3 (4215's D/~C) to 0, then wait
+ * 12 cycles <= 12/(5512.5*64) sec = 34.01 usec
+ */
+ val = D_ENPIO | D_PIO1 | (dbri->mm.onboard ? D_PIO0 : D_PIO2);
+ sbus_writel(val, dbri->regs + REG2);
+ dprintk(D_MM, "cs4215_setctrl: reg2=0x%x\n", val);
+ udelay(34);
+
+ /* In Control mode, the CS4215 is a slave device, so the DBRI must
+ * operate as CHI master, supplying clocking and frame synchronization.
+ *
+ * In Data mode, however, the CS4215 must be CHI master to insure
+ * that its data stream is synchronous with its codec.
+ *
+ * The upshot of all this? We start by putting the DBRI into master
+ * mode, program the CS4215 in Control mode, then switch the CS4215
+ * into Data mode and put the DBRI into slave mode. Various timing
+ * requirements must be observed along the way.
+ *
+ * Oh, and one more thing, on a SPARCStation 20 (and maybe
+ * others?), the addressing of the CS4215's time slots is
+ * offset by eight bits, so we add eight to all the "cycle"
+ * values in the Define Time Slot (DTS) commands. This is
+ * done in hardware by a TI 248 that delays the DBRI->4215
+ * frame sync signal by eight clock cycles. Anybody know why?
+ */
+ spin_lock_irqsave(&dbri->lock, flags);
+ tmp = sbus_readl(dbri->regs + REG0);
+ tmp &= ~D_C; /* Disable CHI */
+ sbus_writel(tmp, dbri->regs + REG0);
+
+ reset_chi(dbri, CHImaster, 128);
+
+ /*
+ * Control mode:
+ * Pipe 17: Send timeslots 1-4 (slots 5-8 are read only)
+ * Pipe 18: Receive timeslot 1 (clb).
+ * Pipe 19: Receive timeslot 7 (version).
+ */
+
+ link_time_slot(dbri, 17, 16, 16, 32, dbri->mm.offset);
+ link_time_slot(dbri, 18, 16, 16, 8, dbri->mm.offset);
+ link_time_slot(dbri, 19, 18, 16, 8, dbri->mm.offset + 48);
+ spin_unlock_irqrestore(&dbri->lock, flags);
+
+ /* Wait for the chip to echo back CLB (Control Latch Bit) as zero */
+ dbri->mm.ctrl[0] &= ~CS4215_CLB;
+ xmit_fixed(dbri, 17, *(int *)dbri->mm.ctrl);
+
+ spin_lock_irqsave(&dbri->lock, flags);
+ tmp = sbus_readl(dbri->regs + REG0);
+ tmp |= D_C; /* Enable CHI */
+ sbus_writel(tmp, dbri->regs + REG0);
+ spin_unlock_irqrestore(&dbri->lock, flags);
+
+ for (i = 10; ((dbri->mm.status & 0xe4) != 0x20); --i)
+ msleep_interruptible(1);
+
+ if (i == 0) {
+ dprintk(D_MM, "CS4215 didn't respond to CLB (0x%02x)\n",
+ dbri->mm.status);
+ return -1;
+ }
+
+ /* Disable changes to our copy of the version number, as we are about
+ * to leave control mode.
+ */
+ recv_fixed(dbri, 19, NULL);
+
+ /* Terminate CS4215 control mode - data sheet says
+ * "Set CLB=1 and send two more frames of valid control info"
+ */
+ dbri->mm.ctrl[0] |= CS4215_CLB;
+ xmit_fixed(dbri, 17, *(int *)dbri->mm.ctrl);
+
+ /* Two frames of control info @ 8kHz frame rate = 250 us delay */
+ udelay(250);
+
+ cs4215_setdata(dbri, 0);
+
+ return 0;
+}
+
+/*
+ * Setup the codec with the sampling rate, audio format and number of
+ * channels.
+ * As part of the process we resend the settings for the data
+ * timeslots as well.
+ */
+static int cs4215_prepare(struct snd_dbri *dbri, unsigned int rate,
+ snd_pcm_format_t format, unsigned int channels)
+{
+ int freq_idx;
+ int ret = 0;
+
+ /* Lookup index for this rate */
+ for (freq_idx = 0; CS4215_FREQ[freq_idx].freq != 0; freq_idx++) {
+ if (CS4215_FREQ[freq_idx].freq == rate)
+ break;
+ }
+ if (CS4215_FREQ[freq_idx].freq != rate) {
+ printk(KERN_WARNING "DBRI: Unsupported rate %d Hz\n", rate);
+ return -1;
+ }
+
+ switch (format) {
+ case SNDRV_PCM_FORMAT_MU_LAW:
+ dbri->mm.ctrl[1] = CS4215_DFR_ULAW;
+ dbri->mm.precision = 8;
+ break;
+ case SNDRV_PCM_FORMAT_A_LAW:
+ dbri->mm.ctrl[1] = CS4215_DFR_ALAW;
+ dbri->mm.precision = 8;
+ break;
+ case SNDRV_PCM_FORMAT_U8:
+ dbri->mm.ctrl[1] = CS4215_DFR_LINEAR8;
+ dbri->mm.precision = 8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_BE:
+ dbri->mm.ctrl[1] = CS4215_DFR_LINEAR16;
+ dbri->mm.precision = 16;
+ break;
+ default:
+ printk(KERN_WARNING "DBRI: Unsupported format %d\n", format);
+ return -1;
+ }
+
+ /* Add rate parameters */
+ dbri->mm.ctrl[1] |= CS4215_FREQ[freq_idx].csval;
+ dbri->mm.ctrl[2] = CS4215_XCLK |
+ CS4215_BSEL_128 | CS4215_FREQ[freq_idx].xtal;
+
+ dbri->mm.channels = channels;
+ if (channels == 2)
+ dbri->mm.ctrl[1] |= CS4215_DFR_STEREO;
+
+ ret = cs4215_setctrl(dbri);
+ if (ret == 0)
+ cs4215_open(dbri); /* set codec to data mode */
+
+ return ret;
+}
+
+/*
+ *
+ */
+static __devinit int cs4215_init(struct snd_dbri *dbri)
+{
+ u32 reg2 = sbus_readl(dbri->regs + REG2);
+ dprintk(D_MM, "cs4215_init: reg2=0x%x\n", reg2);
+
+ /* Look for the cs4215 chips */
+ if (reg2 & D_PIO2) {
+ dprintk(D_MM, "Onboard CS4215 detected\n");
+ dbri->mm.onboard = 1;
+ }
+ if (reg2 & D_PIO0) {
+ dprintk(D_MM, "Speakerbox detected\n");
+ dbri->mm.onboard = 0;
+
+ if (reg2 & D_PIO2) {
+ printk(KERN_INFO "DBRI: Using speakerbox / "
+ "ignoring onboard mmcodec.\n");
+ sbus_writel(D_ENPIO2, dbri->regs + REG2);
+ }
+ }
+
+ if (!(reg2 & (D_PIO0 | D_PIO2))) {
+ printk(KERN_ERR "DBRI: no mmcodec found.\n");
+ return -EIO;
+ }
+
+ cs4215_setup_pipes(dbri);
+ cs4215_init_data(&dbri->mm);
+
+ /* Enable capture of the status & version timeslots. */
+ recv_fixed(dbri, 18, &dbri->mm.status);
+ recv_fixed(dbri, 19, &dbri->mm.version);
+
+ dbri->mm.offset = dbri->mm.onboard ? 0 : 8;
+ if (cs4215_setctrl(dbri) == -1 || dbri->mm.version == 0xff) {
+ dprintk(D_MM, "CS4215 failed probe at offset %d\n",
+ dbri->mm.offset);
+ return -EIO;
+ }
+ dprintk(D_MM, "Found CS4215 at offset %d\n", dbri->mm.offset);
+
+ return 0;
+}
+
+/*
+****************************************************************************
+*************************** DBRI interrupt handler *************************
+****************************************************************************
+
+The DBRI communicates with the CPU mainly via a circular interrupt
+buffer. When an interrupt is signaled, the CPU walks through the
+buffer and calls dbri_process_one_interrupt() for each interrupt word.
+Complicated interrupts are handled by dedicated functions (which
+appear first in this file). Any pending interrupts can be serviced by
+calling dbri_process_interrupt_buffer(), which works even if the CPU's
+interrupts are disabled.
+
+*/
+
+/* xmit_descs()
+ *
+ * Starts transmitting the current TD's for recording/playing.
+ * For playback, ALSA has filled the DMA memory with new data (we hope).
+ */
+static void xmit_descs(struct snd_dbri *dbri)
+{
+ struct dbri_streaminfo *info;
+ s32 *cmd;
+ unsigned long flags;
+ int first_td;
+
+ if (dbri == NULL)
+ return; /* Disabled */
+
+ info = &dbri->stream_info[DBRI_REC];
+ spin_lock_irqsave(&dbri->lock, flags);
+
+ if (info->pipe >= 0) {
+ first_td = dbri->pipes[info->pipe].first_desc;
+
+ dprintk(D_DESC, "xmit_descs rec @ TD %d\n", first_td);
+
+ /* Stream could be closed by the time we run. */
+ if (first_td >= 0) {
+ cmd = dbri_cmdlock(dbri, 2);
+ *(cmd++) = DBRI_CMD(D_SDP, 0,
+ dbri->pipes[info->pipe].sdp
+ | D_SDP_P | D_SDP_EVERY | D_SDP_C);
+ *(cmd++) = dbri->dma_dvma +
+ dbri_dma_off(desc, first_td);
+ dbri_cmdsend(dbri, cmd, 2);
+
+ /* Reset our admin of the pipe. */
+ dbri->pipes[info->pipe].desc = first_td;
+ }
+ }
+
+ info = &dbri->stream_info[DBRI_PLAY];
+
+ if (info->pipe >= 0) {
+ first_td = dbri->pipes[info->pipe].first_desc;
+
+ dprintk(D_DESC, "xmit_descs play @ TD %d\n", first_td);
+
+ /* Stream could be closed by the time we run. */
+ if (first_td >= 0) {
+ cmd = dbri_cmdlock(dbri, 2);
+ *(cmd++) = DBRI_CMD(D_SDP, 0,
+ dbri->pipes[info->pipe].sdp
+ | D_SDP_P | D_SDP_EVERY | D_SDP_C);
+ *(cmd++) = dbri->dma_dvma +
+ dbri_dma_off(desc, first_td);
+ dbri_cmdsend(dbri, cmd, 2);
+
+ /* Reset our admin of the pipe. */
+ dbri->pipes[info->pipe].desc = first_td;
+ }
+ }
+
+ spin_unlock_irqrestore(&dbri->lock, flags);
+}
+
+/* transmission_complete_intr()
+ *
+ * Called by main interrupt handler when DBRI signals transmission complete
+ * on a pipe (interrupt triggered by the B bit in a transmit descriptor).
+ *
+ * Walks through the pipe's list of transmit buffer descriptors and marks
+ * them as available. Stops when the first descriptor is found without
+ * TBC (Transmit Buffer Complete) set, or we've run through them all.
+ *
+ * The DMA buffers are not released. They form a ring buffer and
+ * they are filled by ALSA while others are transmitted by DMA.
+ *
+ */
+
+static void transmission_complete_intr(struct snd_dbri *dbri, int pipe)
+{
+ struct dbri_streaminfo *info = &dbri->stream_info[DBRI_PLAY];
+ int td = dbri->pipes[pipe].desc;
+ int status;
+
+ while (td >= 0) {
+ if (td >= DBRI_NO_DESCS) {
+ printk(KERN_ERR "DBRI: invalid td on pipe %d\n", pipe);
+ return;
+ }
+
+ status = DBRI_TD_STATUS(dbri->dma->desc[td].word4);
+ if (!(status & DBRI_TD_TBC))
+ break;
+
+ dprintk(D_INT, "TD %d, status 0x%02x\n", td, status);
+
+ dbri->dma->desc[td].word4 = 0; /* Reset it for next time. */
+ info->offset += DBRI_RD_CNT(dbri->dma->desc[td].word1);
+
+ td = dbri->next_desc[td];
+ dbri->pipes[pipe].desc = td;
+ }
+
+ /* Notify ALSA */
+ spin_unlock(&dbri->lock);
+ snd_pcm_period_elapsed(info->substream);
+ spin_lock(&dbri->lock);
+}
+
+static void reception_complete_intr(struct snd_dbri *dbri, int pipe)
+{
+ struct dbri_streaminfo *info;
+ int rd = dbri->pipes[pipe].desc;
+ s32 status;
+
+ if (rd < 0 || rd >= DBRI_NO_DESCS) {
+ printk(KERN_ERR "DBRI: invalid rd on pipe %d\n", pipe);
+ return;
+ }
+
+ dbri->pipes[pipe].desc = dbri->next_desc[rd];
+ status = dbri->dma->desc[rd].word1;
+ dbri->dma->desc[rd].word1 = 0; /* Reset it for next time. */
+
+ info = &dbri->stream_info[DBRI_REC];
+ info->offset += DBRI_RD_CNT(status);
+
+ /* FIXME: Check status */
+
+ dprintk(D_INT, "Recv RD %d, status 0x%02x, len %d\n",
+ rd, DBRI_RD_STATUS(status), DBRI_RD_CNT(status));
+
+ /* Notify ALSA */
+ spin_unlock(&dbri->lock);
+ snd_pcm_period_elapsed(info->substream);
+ spin_lock(&dbri->lock);
+}
+
+static void dbri_process_one_interrupt(struct snd_dbri *dbri, int x)
+{
+ int val = D_INTR_GETVAL(x);
+ int channel = D_INTR_GETCHAN(x);
+ int command = D_INTR_GETCMD(x);
+ int code = D_INTR_GETCODE(x);
+#ifdef DBRI_DEBUG
+ int rval = D_INTR_GETRVAL(x);
+#endif
+
+ if (channel == D_INTR_CMD) {
+ dprintk(D_CMD, "INTR: Command: %-5s Value:%d\n",
+ cmds[command], val);
+ } else {
+ dprintk(D_INT, "INTR: Chan:%d Code:%d Val:%#x\n",
+ channel, code, rval);
+ }
+
+ switch (code) {
+ case D_INTR_CMDI:
+ if (command != D_WAIT)
+ printk(KERN_ERR "DBRI: Command read interrupt\n");
+ break;
+ case D_INTR_BRDY:
+ reception_complete_intr(dbri, channel);
+ break;
+ case D_INTR_XCMP:
+ case D_INTR_MINT:
+ transmission_complete_intr(dbri, channel);
+ break;
+ case D_INTR_UNDR:
+ /* UNDR - Transmission underrun
+ * resend SDP command with clear pipe bit (C) set
+ */
+ {
+ /* FIXME: do something useful in case of underrun */
+ printk(KERN_ERR "DBRI: Underrun error\n");
+#if 0
+ s32 *cmd;
+ int pipe = channel;
+ int td = dbri->pipes[pipe].desc;
+
+ dbri->dma->desc[td].word4 = 0;
+ cmd = dbri_cmdlock(dbri, NoGetLock);
+ *(cmd++) = DBRI_CMD(D_SDP, 0,
+ dbri->pipes[pipe].sdp
+ | D_SDP_P | D_SDP_C | D_SDP_2SAME);
+ *(cmd++) = dbri->dma_dvma + dbri_dma_off(desc, td);
+ dbri_cmdsend(dbri, cmd);
+#endif
+ }
+ break;
+ case D_INTR_FXDT:
+ /* FXDT - Fixed data change */
+ if (dbri->pipes[channel].sdp & D_SDP_MSB)
+ val = reverse_bytes(val, dbri->pipes[channel].length);
+
+ if (dbri->pipes[channel].recv_fixed_ptr)
+ *(dbri->pipes[channel].recv_fixed_ptr) = val;
+ break;
+ default:
+ if (channel != D_INTR_CMD)
+ printk(KERN_WARNING
+ "DBRI: Ignored Interrupt: %d (0x%x)\n", code, x);
+ }
+}
+
+/* dbri_process_interrupt_buffer advances through the DBRI's interrupt
+ * buffer until it finds a zero word (indicating nothing more to do
+ * right now). Non-zero words require processing and are handed off
+ * to dbri_process_one_interrupt AFTER advancing the pointer.
+ */
+static void dbri_process_interrupt_buffer(struct snd_dbri *dbri)
+{
+ s32 x;
+
+ while ((x = dbri->dma->intr[dbri->dbri_irqp]) != 0) {
+ dbri->dma->intr[dbri->dbri_irqp] = 0;
+ dbri->dbri_irqp++;
+ if (dbri->dbri_irqp == DBRI_INT_BLK)
+ dbri->dbri_irqp = 1;
+
+ dbri_process_one_interrupt(dbri, x);
+ }
+}
+
+static irqreturn_t snd_dbri_interrupt(int irq, void *dev_id)
+{
+ struct snd_dbri *dbri = dev_id;
+ static int errcnt = 0;
+ int x;
+
+ if (dbri == NULL)
+ return IRQ_NONE;
+ spin_lock(&dbri->lock);
+
+ /*
+ * Read it, so the interrupt goes away.
+ */
+ x = sbus_readl(dbri->regs + REG1);
+
+ if (x & (D_MRR | D_MLE | D_LBG | D_MBE)) {
+ u32 tmp;
+
+ if (x & D_MRR)
+ printk(KERN_ERR
+ "DBRI: Multiple Error Ack on SBus reg1=0x%x\n",
+ x);
+ if (x & D_MLE)
+ printk(KERN_ERR
+ "DBRI: Multiple Late Error on SBus reg1=0x%x\n",
+ x);
+ if (x & D_LBG)
+ printk(KERN_ERR
+ "DBRI: Lost Bus Grant on SBus reg1=0x%x\n", x);
+ if (x & D_MBE)
+ printk(KERN_ERR
+ "DBRI: Burst Error on SBus reg1=0x%x\n", x);
+
+ /* Some of these SBus errors cause the chip's SBus circuitry
+ * to be disabled, so just re-enable and try to keep going.
+ *
+ * The only one I've seen is MRR, which will be triggered
+ * if you let a transmit pipe underrun, then try to CDP it.
+ *
+ * If these things persist, we reset the chip.
+ */
+ if ((++errcnt) % 10 == 0) {
+ dprintk(D_INT, "Interrupt errors exceeded.\n");
+ dbri_reset(dbri);
+ } else {
+ tmp = sbus_readl(dbri->regs + REG0);
+ tmp &= ~(D_D);
+ sbus_writel(tmp, dbri->regs + REG0);
+ }
+ }
+
+ dbri_process_interrupt_buffer(dbri);
+
+ spin_unlock(&dbri->lock);
+
+ return IRQ_HANDLED;
+}
+
+/****************************************************************************
+ PCM Interface
+****************************************************************************/
+static struct snd_pcm_hardware snd_dbri_pcm_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .formats = SNDRV_PCM_FMTBIT_MU_LAW |
+ SNDRV_PCM_FMTBIT_A_LAW |
+ SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_5512,
+ .rate_min = 5512,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 1,
+ .period_bytes_max = DBRI_TD_MAXCNT,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static int snd_hw_rule_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_mask fmt;
+
+ snd_mask_any(&fmt);
+ if (c->min > 1) {
+ fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_BE;
+ return snd_mask_refine(f, &fmt);
+ }
+ return 0;
+}
+
+static int snd_hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_interval ch;
+
+ snd_interval_any(&ch);
+ if (!(f->bits[0] & SNDRV_PCM_FMTBIT_S16_BE)) {
+ ch.min = 1;
+ ch.max = 1;
+ ch.integer = 1;
+ return snd_interval_refine(c, &ch);
+ }
+ return 0;
+}
+
+static int snd_dbri_open(struct snd_pcm_substream *substream)
+{
+ struct snd_dbri *dbri = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct dbri_streaminfo *info = DBRI_STREAM(dbri, substream);
+ unsigned long flags;
+
+ dprintk(D_USR, "open audio output.\n");
+ runtime->hw = snd_dbri_pcm_hw;
+
+ spin_lock_irqsave(&dbri->lock, flags);
+ info->substream = substream;
+ info->offset = 0;
+ info->dvma_buffer = 0;
+ info->pipe = -1;
+ spin_unlock_irqrestore(&dbri->lock, flags);
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hw_rule_format, NULL, SNDRV_PCM_HW_PARAM_FORMAT,
+ -1);
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ snd_hw_rule_channels, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ -1);
+
+ cs4215_open(dbri);
+
+ return 0;
+}
+
+static int snd_dbri_close(struct snd_pcm_substream *substream)
+{
+ struct snd_dbri *dbri = snd_pcm_substream_chip(substream);
+ struct dbri_streaminfo *info = DBRI_STREAM(dbri, substream);
+
+ dprintk(D_USR, "close audio output.\n");
+ info->substream = NULL;
+ info->offset = 0;
+
+ return 0;
+}
+
+static int snd_dbri_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dbri *dbri = snd_pcm_substream_chip(substream);
+ struct dbri_streaminfo *info = DBRI_STREAM(dbri, substream);
+ int direction;
+ int ret;
+
+ /* set sampling rate, audio format and number of channels */
+ ret = cs4215_prepare(dbri, params_rate(hw_params),
+ params_format(hw_params),
+ params_channels(hw_params));
+ if (ret != 0)
+ return ret;
+
+ if ((ret = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params))) < 0) {
+ printk(KERN_ERR "malloc_pages failed with %d\n", ret);
+ return ret;
+ }
+
+ /* hw_params can get called multiple times. Only map the DMA once.
+ */
+ if (info->dvma_buffer == 0) {
+ if (DBRI_STREAMNO(substream) == DBRI_PLAY)
+ direction = DMA_TO_DEVICE;
+ else
+ direction = DMA_FROM_DEVICE;
+
+ info->dvma_buffer =
+ dma_map_single(&dbri->op->dev,
+ runtime->dma_area,
+ params_buffer_bytes(hw_params),
+ direction);
+ }
+
+ direction = params_buffer_bytes(hw_params);
+ dprintk(D_USR, "hw_params: %d bytes, dvma=%x\n",
+ direction, info->dvma_buffer);
+ return 0;
+}
+
+static int snd_dbri_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_dbri *dbri = snd_pcm_substream_chip(substream);
+ struct dbri_streaminfo *info = DBRI_STREAM(dbri, substream);
+ int direction;
+
+ dprintk(D_USR, "hw_free.\n");
+
+ /* hw_free can get called multiple times. Only unmap the DMA once.
+ */
+ if (info->dvma_buffer) {
+ if (DBRI_STREAMNO(substream) == DBRI_PLAY)
+ direction = DMA_TO_DEVICE;
+ else
+ direction = DMA_FROM_DEVICE;
+
+ dma_unmap_single(&dbri->op->dev, info->dvma_buffer,
+ substream->runtime->buffer_size, direction);
+ info->dvma_buffer = 0;
+ }
+ if (info->pipe != -1) {
+ reset_pipe(dbri, info->pipe);
+ info->pipe = -1;
+ }
+
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_dbri_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dbri *dbri = snd_pcm_substream_chip(substream);
+ struct dbri_streaminfo *info = DBRI_STREAM(dbri, substream);
+ int ret;
+
+ info->size = snd_pcm_lib_buffer_bytes(substream);
+ if (DBRI_STREAMNO(substream) == DBRI_PLAY)
+ info->pipe = 4; /* Send pipe */
+ else
+ info->pipe = 6; /* Receive pipe */
+
+ spin_lock_irq(&dbri->lock);
+ info->offset = 0;
+
+ /* Setup the all the transmit/receive descriptors to cover the
+ * whole DMA buffer.
+ */
+ ret = setup_descs(dbri, DBRI_STREAMNO(substream),
+ snd_pcm_lib_period_bytes(substream));
+
+ spin_unlock_irq(&dbri->lock);
+
+ dprintk(D_USR, "prepare audio output. %d bytes\n", info->size);
+ return ret;
+}
+
+static int snd_dbri_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_dbri *dbri = snd_pcm_substream_chip(substream);
+ struct dbri_streaminfo *info = DBRI_STREAM(dbri, substream);
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ dprintk(D_USR, "start audio, period is %d bytes\n",
+ (int)snd_pcm_lib_period_bytes(substream));
+ /* Re-submit the TDs. */
+ xmit_descs(dbri);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ dprintk(D_USR, "stop audio.\n");
+ reset_pipe(dbri, info->pipe);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t snd_dbri_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_dbri *dbri = snd_pcm_substream_chip(substream);
+ struct dbri_streaminfo *info = DBRI_STREAM(dbri, substream);
+ snd_pcm_uframes_t ret;
+
+ ret = bytes_to_frames(substream->runtime, info->offset)
+ % substream->runtime->buffer_size;
+ dprintk(D_USR, "I/O pointer: %ld frames of %ld.\n",
+ ret, substream->runtime->buffer_size);
+ return ret;
+}
+
+static struct snd_pcm_ops snd_dbri_ops = {
+ .open = snd_dbri_open,
+ .close = snd_dbri_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_dbri_hw_params,
+ .hw_free = snd_dbri_hw_free,
+ .prepare = snd_dbri_prepare,
+ .trigger = snd_dbri_trigger,
+ .pointer = snd_dbri_pointer,
+};
+
+static int __devinit snd_dbri_pcm(struct snd_card *card)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(card,
+ /* ID */ "sun_dbri",
+ /* device */ 0,
+ /* playback count */ 1,
+ /* capture count */ 1, &pcm)) < 0)
+ return err;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_dbri_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_dbri_ops);
+
+ pcm->private_data = card->private_data;
+ pcm->info_flags = 0;
+ strcpy(pcm->name, card->shortname);
+
+ if ((err = snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 64 * 1024, 64 * 1024)) < 0)
+ return err;
+
+ return 0;
+}
+
+/*****************************************************************************
+ Mixer interface
+*****************************************************************************/
+
+static int snd_cs4215_info_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ if (kcontrol->private_value == DBRI_PLAY)
+ uinfo->value.integer.max = DBRI_MAX_VOLUME;
+ else
+ uinfo->value.integer.max = DBRI_MAX_GAIN;
+ return 0;
+}
+
+static int snd_cs4215_get_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dbri *dbri = snd_kcontrol_chip(kcontrol);
+ struct dbri_streaminfo *info;
+
+ if (snd_BUG_ON(!dbri))
+ return -EINVAL;
+ info = &dbri->stream_info[kcontrol->private_value];
+
+ ucontrol->value.integer.value[0] = info->left_gain;
+ ucontrol->value.integer.value[1] = info->right_gain;
+ return 0;
+}
+
+static int snd_cs4215_put_volume(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dbri *dbri = snd_kcontrol_chip(kcontrol);
+ struct dbri_streaminfo *info =
+ &dbri->stream_info[kcontrol->private_value];
+ unsigned int vol[2];
+ int changed = 0;
+
+ vol[0] = ucontrol->value.integer.value[0];
+ vol[1] = ucontrol->value.integer.value[1];
+ if (kcontrol->private_value == DBRI_PLAY) {
+ if (vol[0] > DBRI_MAX_VOLUME || vol[1] > DBRI_MAX_VOLUME)
+ return -EINVAL;
+ } else {
+ if (vol[0] > DBRI_MAX_GAIN || vol[1] > DBRI_MAX_GAIN)
+ return -EINVAL;
+ }
+
+ if (info->left_gain != vol[0]) {
+ info->left_gain = vol[0];
+ changed = 1;
+ }
+ if (info->right_gain != vol[1]) {
+ info->right_gain = vol[1];
+ changed = 1;
+ }
+ if (changed) {
+ /* First mute outputs, and wait 1/8000 sec (125 us)
+ * to make sure this takes. This avoids clicking noises.
+ */
+ cs4215_setdata(dbri, 1);
+ udelay(125);
+ cs4215_setdata(dbri, 0);
+ }
+ return changed;
+}
+
+static int snd_cs4215_info_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+
+ uinfo->type = (mask == 1) ?
+ SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+ return 0;
+}
+
+static int snd_cs4215_get_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dbri *dbri = snd_kcontrol_chip(kcontrol);
+ int elem = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 1;
+
+ if (snd_BUG_ON(!dbri))
+ return -EINVAL;
+
+ if (elem < 4)
+ ucontrol->value.integer.value[0] =
+ (dbri->mm.data[elem] >> shift) & mask;
+ else
+ ucontrol->value.integer.value[0] =
+ (dbri->mm.ctrl[elem - 4] >> shift) & mask;
+
+ if (invert == 1)
+ ucontrol->value.integer.value[0] =
+ mask - ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int snd_cs4215_put_single(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_dbri *dbri = snd_kcontrol_chip(kcontrol);
+ int elem = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 1;
+ int changed = 0;
+ unsigned short val;
+
+ if (snd_BUG_ON(!dbri))
+ return -EINVAL;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert == 1)
+ val = mask - val;
+ val <<= shift;
+
+ if (elem < 4) {
+ dbri->mm.data[elem] = (dbri->mm.data[elem] &
+ ~(mask << shift)) | val;
+ changed = (val != dbri->mm.data[elem]);
+ } else {
+ dbri->mm.ctrl[elem - 4] = (dbri->mm.ctrl[elem - 4] &
+ ~(mask << shift)) | val;
+ changed = (val != dbri->mm.ctrl[elem - 4]);
+ }
+
+ dprintk(D_GEN, "put_single: mask=0x%x, changed=%d, "
+ "mixer-value=%ld, mm-value=0x%x\n",
+ mask, changed, ucontrol->value.integer.value[0],
+ dbri->mm.data[elem & 3]);
+
+ if (changed) {
+ /* First mute outputs, and wait 1/8000 sec (125 us)
+ * to make sure this takes. This avoids clicking noises.
+ */
+ cs4215_setdata(dbri, 1);
+ udelay(125);
+ cs4215_setdata(dbri, 0);
+ }
+ return changed;
+}
+
+/* Entries 0-3 map to the 4 data timeslots, entries 4-7 map to the 4 control
+ timeslots. Shift is the bit offset in the timeslot, mask defines the
+ number of bits. invert is a boolean for use with attenuation.
+ */
+#define CS4215_SINGLE(xname, entry, shift, mask, invert) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .info = snd_cs4215_info_single, \
+ .get = snd_cs4215_get_single, .put = snd_cs4215_put_single, \
+ .private_value = (entry) | ((shift) << 8) | ((mask) << 16) | \
+ ((invert) << 24) },
+
+static struct snd_kcontrol_new dbri_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Playback Volume",
+ .info = snd_cs4215_info_volume,
+ .get = snd_cs4215_get_volume,
+ .put = snd_cs4215_put_volume,
+ .private_value = DBRI_PLAY,
+ },
+ CS4215_SINGLE("Headphone switch", 0, 7, 1, 0)
+ CS4215_SINGLE("Line out switch", 0, 6, 1, 0)
+ CS4215_SINGLE("Speaker switch", 1, 6, 1, 0)
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Volume",
+ .info = snd_cs4215_info_volume,
+ .get = snd_cs4215_get_volume,
+ .put = snd_cs4215_put_volume,
+ .private_value = DBRI_REC,
+ },
+ /* FIXME: mic/line switch */
+ CS4215_SINGLE("Line in switch", 2, 4, 1, 0)
+ CS4215_SINGLE("High Pass Filter switch", 5, 7, 1, 0)
+ CS4215_SINGLE("Monitor Volume", 3, 4, 0xf, 1)
+ CS4215_SINGLE("Mic boost", 4, 4, 1, 1)
+};
+
+static int __devinit snd_dbri_mixer(struct snd_card *card)
+{
+ int idx, err;
+ struct snd_dbri *dbri;
+
+ if (snd_BUG_ON(!card || !card->private_data))
+ return -EINVAL;
+ dbri = card->private_data;
+
+ strcpy(card->mixername, card->shortname);
+
+ for (idx = 0; idx < ARRAY_SIZE(dbri_controls); idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&dbri_controls[idx], dbri));
+ if (err < 0)
+ return err;
+ }
+
+ for (idx = DBRI_REC; idx < DBRI_NO_STREAMS; idx++) {
+ dbri->stream_info[idx].left_gain = 0;
+ dbri->stream_info[idx].right_gain = 0;
+ }
+
+ return 0;
+}
+
+/****************************************************************************
+ /proc interface
+****************************************************************************/
+static void dbri_regs_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_dbri *dbri = entry->private_data;
+
+ snd_iprintf(buffer, "REG0: 0x%x\n", sbus_readl(dbri->regs + REG0));
+ snd_iprintf(buffer, "REG2: 0x%x\n", sbus_readl(dbri->regs + REG2));
+ snd_iprintf(buffer, "REG8: 0x%x\n", sbus_readl(dbri->regs + REG8));
+ snd_iprintf(buffer, "REG9: 0x%x\n", sbus_readl(dbri->regs + REG9));
+}
+
+#ifdef DBRI_DEBUG
+static void dbri_debug_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_dbri *dbri = entry->private_data;
+ int pipe;
+ snd_iprintf(buffer, "debug=%d\n", dbri_debug);
+
+ for (pipe = 0; pipe < 32; pipe++) {
+ if (pipe_active(dbri, pipe)) {
+ struct dbri_pipe *pptr = &dbri->pipes[pipe];
+ snd_iprintf(buffer,
+ "Pipe %d: %s SDP=0x%x desc=%d, "
+ "len=%d next %d\n",
+ pipe,
+ (pptr->sdp & D_SDP_TO_SER) ? "output" :
+ "input",
+ pptr->sdp, pptr->desc,
+ pptr->length, pptr->nextpipe);
+ }
+ }
+}
+#endif
+
+static void __devinit snd_dbri_proc(struct snd_card *card)
+{
+ struct snd_dbri *dbri = card->private_data;
+ struct snd_info_entry *entry;
+
+ if (!snd_card_proc_new(card, "regs", &entry))
+ snd_info_set_text_ops(entry, dbri, dbri_regs_read);
+
+#ifdef DBRI_DEBUG
+ if (!snd_card_proc_new(card, "debug", &entry)) {
+ snd_info_set_text_ops(entry, dbri, dbri_debug_read);
+ entry->mode = S_IFREG | S_IRUGO; /* Readable only. */
+ }
+#endif
+}
+
+/*
+****************************************************************************
+**************************** Initialization ********************************
+****************************************************************************
+*/
+static void snd_dbri_free(struct snd_dbri *dbri);
+
+static int __devinit snd_dbri_create(struct snd_card *card,
+ struct of_device *op,
+ int irq, int dev)
+{
+ struct snd_dbri *dbri = card->private_data;
+ int err;
+
+ spin_lock_init(&dbri->lock);
+ dbri->op = op;
+ dbri->irq = irq;
+
+ dbri->dma = dma_alloc_coherent(&op->dev,
+ sizeof(struct dbri_dma),
+ &dbri->dma_dvma, GFP_ATOMIC);
+ if (!dbri->dma)
+ return -ENOMEM;
+ memset((void *)dbri->dma, 0, sizeof(struct dbri_dma));
+
+ dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n",
+ dbri->dma, dbri->dma_dvma);
+
+ /* Map the registers into memory. */
+ dbri->regs_size = resource_size(&op->resource[0]);
+ dbri->regs = of_ioremap(&op->resource[0], 0,
+ dbri->regs_size, "DBRI Registers");
+ if (!dbri->regs) {
+ printk(KERN_ERR "DBRI: could not allocate registers\n");
+ dma_free_coherent(&op->dev, sizeof(struct dbri_dma),
+ (void *)dbri->dma, dbri->dma_dvma);
+ return -EIO;
+ }
+
+ err = request_irq(dbri->irq, snd_dbri_interrupt, IRQF_SHARED,
+ "DBRI audio", dbri);
+ if (err) {
+ printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq);
+ of_iounmap(&op->resource[0], dbri->regs, dbri->regs_size);
+ dma_free_coherent(&op->dev, sizeof(struct dbri_dma),
+ (void *)dbri->dma, dbri->dma_dvma);
+ return err;
+ }
+
+ /* Do low level initialization of the DBRI and CS4215 chips */
+ dbri_initialize(dbri);
+ err = cs4215_init(dbri);
+ if (err) {
+ snd_dbri_free(dbri);
+ return err;
+ }
+
+ return 0;
+}
+
+static void snd_dbri_free(struct snd_dbri *dbri)
+{
+ dprintk(D_GEN, "snd_dbri_free\n");
+ dbri_reset(dbri);
+
+ if (dbri->irq)
+ free_irq(dbri->irq, dbri);
+
+ if (dbri->regs)
+ of_iounmap(&dbri->op->resource[0], dbri->regs, dbri->regs_size);
+
+ if (dbri->dma)
+ dma_free_coherent(&dbri->op->dev,
+ sizeof(struct dbri_dma),
+ (void *)dbri->dma, dbri->dma_dvma);
+}
+
+static int __devinit dbri_probe(struct of_device *op, const struct of_device_id *match)
+{
+ struct snd_dbri *dbri;
+ struct resource *rp;
+ struct snd_card *card;
+ static int dev = 0;
+ int irq;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ irq = op->irqs[0];
+ if (irq <= 0) {
+ printk(KERN_ERR "DBRI-%d: No IRQ.\n", dev);
+ return -ENODEV;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct snd_dbri));
+ if (card == NULL)
+ return -ENOMEM;
+
+ strcpy(card->driver, "DBRI");
+ strcpy(card->shortname, "Sun DBRI");
+ rp = &op->resource[0];
+ sprintf(card->longname, "%s at 0x%02lx:0x%016Lx, irq %d",
+ card->shortname,
+ rp->flags & 0xffL, (unsigned long long)rp->start, irq);
+
+ err = snd_dbri_create(card, op, irq, dev);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ dbri = card->private_data;
+ err = snd_dbri_pcm(card);
+ if (err < 0)
+ goto _err;
+
+ err = snd_dbri_mixer(card);
+ if (err < 0)
+ goto _err;
+
+ /* /proc file handling */
+ snd_dbri_proc(card);
+ dev_set_drvdata(&op->dev, card);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto _err;
+
+ printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c)+CS4215(%d)\n",
+ dev, dbri->regs,
+ dbri->irq, op->node->name[9], dbri->mm.version);
+ dev++;
+
+ return 0;
+
+_err:
+ snd_dbri_free(dbri);
+ snd_card_free(card);
+ return err;
+}
+
+static int __devexit dbri_remove(struct of_device *op)
+{
+ struct snd_card *card = dev_get_drvdata(&op->dev);
+
+ snd_dbri_free(card->private_data);
+ snd_card_free(card);
+
+ dev_set_drvdata(&op->dev, NULL);
+
+ return 0;
+}
+
+static const struct of_device_id dbri_match[] = {
+ {
+ .name = "SUNW,DBRIe",
+ },
+ {
+ .name = "SUNW,DBRIf",
+ },
+ {},
+};
+
+MODULE_DEVICE_TABLE(of, dbri_match);
+
+static struct of_platform_driver dbri_sbus_driver = {
+ .name = "dbri",
+ .match_table = dbri_match,
+ .probe = dbri_probe,
+ .remove = __devexit_p(dbri_remove),
+};
+
+/* Probe for the dbri chip and then attach the driver. */
+static int __init dbri_init(void)
+{
+ return of_register_driver(&dbri_sbus_driver, &of_bus_type);
+}
+
+static void __exit dbri_exit(void)
+{
+ of_unregister_driver(&dbri_sbus_driver);
+}
+
+module_init(dbri_init);
+module_exit(dbri_exit);
diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig
new file mode 100644
index 0000000..e6485be
--- /dev/null
+++ b/sound/spi/Kconfig
@@ -0,0 +1,38 @@
+#SPI drivers
+
+menuconfig SND_SPI
+ bool "SPI sound devices"
+ depends on SPI
+ default y
+ help
+ Support for sound devices connected via the SPI bus.
+
+if SND_SPI
+
+config SND_AT73C213
+ tristate "Atmel AT73C213 DAC driver"
+ depends on ATMEL_SSC
+ select SND_PCM
+ help
+ Say Y here if you want to use the Atmel AT73C213 external DAC. This
+ DAC can be found on Atmel development boards.
+
+ This driver requires the Atmel SSC driver for sound sink, a
+ peripheral found on most AT91 and AVR32 microprocessors.
+
+ To compile this driver as a module, choose M here: the module will be
+ called snd-at73c213.
+
+config SND_AT73C213_TARGET_BITRATE
+ int "Target bitrate for AT73C213"
+ depends on SND_AT73C213
+ default "48000"
+ range 8000 50000
+ help
+ Sets the target bitrate for the bitrate calculator in the driver.
+ Limited by hardware to be between 8000 Hz and 50000 Hz.
+
+ Set to 48000 Hz by default.
+
+endif # SND_SPI
+
diff --git a/sound/spi/Makefile b/sound/spi/Makefile
new file mode 100644
index 0000000..026fb73
--- /dev/null
+++ b/sound/spi/Makefile
@@ -0,0 +1,5 @@
+# Makefile for SPI drivers
+
+snd-at73c213-objs := at73c213.o
+
+obj-$(CONFIG_SND_AT73C213) += snd-at73c213.o
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
new file mode 100644
index 0000000..09802e8
--- /dev/null
+++ b/sound/spi/at73c213.c
@@ -0,0 +1,1131 @@
+/*
+ * Driver for AT73C213 16-bit stereo DAC connected to Atmel SSC
+ *
+ * Copyright (C) 2006-2007 Atmel Norway
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published by
+ * the Free Software Foundation.
+ */
+
+/*#define DEBUG*/
+
+#include <linux/clk.h>
+#include <linux/err.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/mutex.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+#include <linux/atmel-ssc.h>
+
+#include <linux/spi/spi.h>
+#include <linux/spi/at73c213.h>
+
+#include "at73c213.h"
+
+#define BITRATE_MIN 8000 /* Hardware limit? */
+#define BITRATE_TARGET CONFIG_SND_AT73C213_TARGET_BITRATE
+#define BITRATE_MAX 50000 /* Hardware limit. */
+
+/* Initial (hardware reset) AT73C213 register values. */
+static u8 snd_at73c213_original_image[18] =
+{
+ 0x00, /* 00 - CTRL */
+ 0x05, /* 01 - LLIG */
+ 0x05, /* 02 - RLIG */
+ 0x08, /* 03 - LPMG */
+ 0x08, /* 04 - RPMG */
+ 0x00, /* 05 - LLOG */
+ 0x00, /* 06 - RLOG */
+ 0x22, /* 07 - OLC */
+ 0x09, /* 08 - MC */
+ 0x00, /* 09 - CSFC */
+ 0x00, /* 0A - MISC */
+ 0x00, /* 0B - */
+ 0x00, /* 0C - PRECH */
+ 0x05, /* 0D - AUXG */
+ 0x00, /* 0E - */
+ 0x00, /* 0F - */
+ 0x00, /* 10 - RST */
+ 0x00, /* 11 - PA_CTRL */
+};
+
+struct snd_at73c213 {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+ struct snd_pcm_substream *substream;
+ struct at73c213_board_info *board;
+ int irq;
+ int period;
+ unsigned long bitrate;
+ struct clk *bitclk;
+ struct ssc_device *ssc;
+ struct spi_device *spi;
+ u8 spi_wbuffer[2];
+ u8 spi_rbuffer[2];
+ /* Image of the SPI registers in AT73C213. */
+ u8 reg_image[18];
+ /* Protect SSC registers against concurrent access. */
+ spinlock_t lock;
+ /* Protect mixer registers against concurrent access. */
+ struct mutex mixer_lock;
+};
+
+#define get_chip(card) ((struct snd_at73c213 *)card->private_data)
+
+static int
+snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val)
+{
+ struct spi_message msg;
+ struct spi_transfer msg_xfer = {
+ .len = 2,
+ .cs_change = 0,
+ };
+ int retval;
+
+ spi_message_init(&msg);
+
+ chip->spi_wbuffer[0] = reg;
+ chip->spi_wbuffer[1] = val;
+
+ msg_xfer.tx_buf = chip->spi_wbuffer;
+ msg_xfer.rx_buf = chip->spi_rbuffer;
+ spi_message_add_tail(&msg_xfer, &msg);
+
+ retval = spi_sync(chip->spi, &msg);
+
+ if (!retval)
+ chip->reg_image[reg] = val;
+
+ return retval;
+}
+
+static struct snd_pcm_hardware snd_at73c213_playback_hw = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .formats = SNDRV_PCM_FMTBIT_S16_BE,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 8000, /* Replaced by chip->bitrate later. */
+ .rate_max = 50000, /* Replaced by chip->bitrate later. */
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024 - 1,
+ .period_bytes_min = 512,
+ .period_bytes_max = 64 * 1024 - 1,
+ .periods_min = 4,
+ .periods_max = 1024,
+};
+
+/*
+ * Calculate and set bitrate and divisions.
+ */
+static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip)
+{
+ unsigned long ssc_rate = clk_get_rate(chip->ssc->clk);
+ unsigned long dac_rate_new, ssc_div;
+ int status;
+ unsigned long ssc_div_max, ssc_div_min;
+ int max_tries;
+
+ /*
+ * We connect two clocks here, picking divisors so the I2S clocks
+ * out data at the same rate the DAC clocks it in ... and as close
+ * as practical to the desired target rate.
+ *
+ * The DAC master clock (MCLK) is programmable, and is either 256
+ * or (not here) 384 times the I2S output clock (BCLK).
+ */
+
+ /* SSC clock / (bitrate * stereo * 16-bit). */
+ ssc_div = ssc_rate / (BITRATE_TARGET * 2 * 16);
+ ssc_div_min = ssc_rate / (BITRATE_MAX * 2 * 16);
+ ssc_div_max = ssc_rate / (BITRATE_MIN * 2 * 16);
+ max_tries = (ssc_div_max - ssc_div_min) / 2;
+
+ if (max_tries < 1)
+ max_tries = 1;
+
+ /* ssc_div must be a power of 2. */
+ ssc_div = (ssc_div + 1) & ~1UL;
+
+ if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) {
+ ssc_div -= 2;
+ if ((ssc_rate / (ssc_div * 2 * 16)) > BITRATE_MAX)
+ return -ENXIO;
+ }
+
+ /* Search for a possible bitrate. */
+ do {
+ /* SSC clock / (ssc divider * 16-bit * stereo). */
+ if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN)
+ return -ENXIO;
+
+ /* 256 / (2 * 16) = 8 */
+ dac_rate_new = 8 * (ssc_rate / ssc_div);
+
+ status = clk_round_rate(chip->board->dac_clk, dac_rate_new);
+ if (status < 0)
+ return status;
+
+ /* Ignore difference smaller than 256 Hz. */
+ if ((status/256) == (dac_rate_new/256))
+ goto set_rate;
+
+ ssc_div += 2;
+ } while (--max_tries);
+
+ /* Not able to find a valid bitrate. */
+ return -ENXIO;
+
+set_rate:
+ status = clk_set_rate(chip->board->dac_clk, status);
+ if (status < 0)
+ return status;
+
+ /* Set divider in SSC device. */
+ ssc_writel(chip->ssc->regs, CMR, ssc_div/2);
+
+ /* SSC clock / (ssc divider * 16-bit * stereo). */
+ chip->bitrate = ssc_rate / (ssc_div * 16 * 2);
+
+ dev_info(&chip->spi->dev,
+ "at73c213: supported bitrate is %lu (%lu divider)\n",
+ chip->bitrate, ssc_div);
+
+ return 0;
+}
+
+static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_at73c213 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ /* ensure buffer_size is a multiple of period_size */
+ err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0)
+ return err;
+ snd_at73c213_playback_hw.rate_min = chip->bitrate;
+ snd_at73c213_playback_hw.rate_max = chip->bitrate;
+ runtime->hw = snd_at73c213_playback_hw;
+ chip->substream = substream;
+
+ return 0;
+}
+
+static int snd_at73c213_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_at73c213 *chip = snd_pcm_substream_chip(substream);
+ chip->substream = NULL;
+ return 0;
+}
+
+static int snd_at73c213_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_at73c213 *chip = snd_pcm_substream_chip(substream);
+ int channels = params_channels(hw_params);
+ int val;
+
+ val = ssc_readl(chip->ssc->regs, TFMR);
+ val = SSC_BFINS(TFMR_DATNB, channels - 1, val);
+ ssc_writel(chip->ssc->regs, TFMR, val);
+
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int snd_at73c213_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_at73c213_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_at73c213 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int block_size;
+
+ block_size = frames_to_bytes(runtime, runtime->period_size);
+
+ chip->period = 0;
+
+ ssc_writel(chip->ssc->regs, PDC_TPR,
+ (long)runtime->dma_addr);
+ ssc_writel(chip->ssc->regs, PDC_TCR,
+ runtime->period_size * runtime->channels);
+ ssc_writel(chip->ssc->regs, PDC_TNPR,
+ (long)runtime->dma_addr + block_size);
+ ssc_writel(chip->ssc->regs, PDC_TNCR,
+ runtime->period_size * runtime->channels);
+
+ return 0;
+}
+
+static int snd_at73c213_pcm_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_at73c213 *chip = snd_pcm_substream_chip(substream);
+ int retval = 0;
+
+ spin_lock(&chip->lock);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ ssc_writel(chip->ssc->regs, IER, SSC_BIT(IER_ENDTX));
+ ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTEN));
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTDIS));
+ ssc_writel(chip->ssc->regs, IDR, SSC_BIT(IDR_ENDTX));
+ break;
+ default:
+ dev_dbg(&chip->spi->dev, "spurious command %x\n", cmd);
+ retval = -EINVAL;
+ break;
+ }
+
+ spin_unlock(&chip->lock);
+
+ return retval;
+}
+
+static snd_pcm_uframes_t
+snd_at73c213_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_at73c213 *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_uframes_t pos;
+ unsigned long bytes;
+
+ bytes = ssc_readl(chip->ssc->regs, PDC_TPR)
+ - (unsigned long)runtime->dma_addr;
+
+ pos = bytes_to_frames(runtime, bytes);
+ if (pos >= runtime->buffer_size)
+ pos -= runtime->buffer_size;
+
+ return pos;
+}
+
+static struct snd_pcm_ops at73c213_playback_ops = {
+ .open = snd_at73c213_pcm_open,
+ .close = snd_at73c213_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_at73c213_pcm_hw_params,
+ .hw_free = snd_at73c213_pcm_hw_free,
+ .prepare = snd_at73c213_pcm_prepare,
+ .trigger = snd_at73c213_pcm_trigger,
+ .pointer = snd_at73c213_pcm_pointer,
+};
+
+static int __devinit snd_at73c213_pcm_new(struct snd_at73c213 *chip, int device)
+{
+ struct snd_pcm *pcm;
+ int retval;
+
+ retval = snd_pcm_new(chip->card, chip->card->shortname,
+ device, 1, 0, &pcm);
+ if (retval < 0)
+ goto out;
+
+ pcm->private_data = chip;
+ pcm->info_flags = SNDRV_PCM_INFO_BLOCK_TRANSFER;
+ strcpy(pcm->name, "at73c213");
+ chip->pcm = pcm;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &at73c213_playback_ops);
+
+ retval = snd_pcm_lib_preallocate_pages_for_all(chip->pcm,
+ SNDRV_DMA_TYPE_DEV, &chip->ssc->pdev->dev,
+ 64 * 1024, 64 * 1024);
+out:
+ return retval;
+}
+
+static irqreturn_t snd_at73c213_interrupt(int irq, void *dev_id)
+{
+ struct snd_at73c213 *chip = dev_id;
+ struct snd_pcm_runtime *runtime = chip->substream->runtime;
+ u32 status;
+ int offset;
+ int block_size;
+ int next_period;
+ int retval = IRQ_NONE;
+
+ spin_lock(&chip->lock);
+
+ block_size = frames_to_bytes(runtime, runtime->period_size);
+ status = ssc_readl(chip->ssc->regs, IMR);
+
+ if (status & SSC_BIT(IMR_ENDTX)) {
+ chip->period++;
+ if (chip->period == runtime->periods)
+ chip->period = 0;
+ next_period = chip->period + 1;
+ if (next_period == runtime->periods)
+ next_period = 0;
+
+ offset = block_size * next_period;
+
+ ssc_writel(chip->ssc->regs, PDC_TNPR,
+ (long)runtime->dma_addr + offset);
+ ssc_writel(chip->ssc->regs, PDC_TNCR,
+ runtime->period_size * runtime->channels);
+ retval = IRQ_HANDLED;
+ }
+
+ ssc_readl(chip->ssc->regs, IMR);
+ spin_unlock(&chip->lock);
+
+ if (status & SSC_BIT(IMR_ENDTX))
+ snd_pcm_period_elapsed(chip->substream);
+
+ return retval;
+}
+
+/*
+ * Mixer functions.
+ */
+static int snd_at73c213_mono_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ mutex_lock(&chip->mixer_lock);
+
+ ucontrol->value.integer.value[0] =
+ (chip->reg_image[reg] >> shift) & mask;
+
+ if (invert)
+ ucontrol->value.integer.value[0] =
+ mask - ucontrol->value.integer.value[0];
+
+ mutex_unlock(&chip->mixer_lock);
+
+ return 0;
+}
+
+static int snd_at73c213_mono_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change, retval;
+ unsigned short val;
+
+ val = (ucontrol->value.integer.value[0] & mask);
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+
+ mutex_lock(&chip->mixer_lock);
+
+ val = (chip->reg_image[reg] & ~(mask << shift)) | val;
+ change = val != chip->reg_image[reg];
+ retval = snd_at73c213_write_reg(chip, reg, val);
+
+ mutex_unlock(&chip->mixer_lock);
+
+ if (retval)
+ return retval;
+
+ return change;
+}
+
+static int snd_at73c213_stereo_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+
+ if (mask == 1)
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = mask;
+
+ return 0;
+}
+
+static int snd_at73c213_stereo_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+
+ mutex_lock(&chip->mixer_lock);
+
+ ucontrol->value.integer.value[0] =
+ (chip->reg_image[left_reg] >> shift_left) & mask;
+ ucontrol->value.integer.value[1] =
+ (chip->reg_image[right_reg] >> shift_right) & mask;
+
+ if (invert) {
+ ucontrol->value.integer.value[0] =
+ mask - ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] =
+ mask - ucontrol->value.integer.value[1];
+ }
+
+ mutex_unlock(&chip->mixer_lock);
+
+ return 0;
+}
+
+static int snd_at73c213_stereo_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol);
+ int left_reg = kcontrol->private_value & 0xff;
+ int right_reg = (kcontrol->private_value >> 8) & 0xff;
+ int shift_left = (kcontrol->private_value >> 16) & 0x07;
+ int shift_right = (kcontrol->private_value >> 19) & 0x07;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ int invert = (kcontrol->private_value >> 22) & 1;
+ int change, retval;
+ unsigned short val1, val2;
+
+ val1 = ucontrol->value.integer.value[0] & mask;
+ val2 = ucontrol->value.integer.value[1] & mask;
+ if (invert) {
+ val1 = mask - val1;
+ val2 = mask - val2;
+ }
+ val1 <<= shift_left;
+ val2 <<= shift_right;
+
+ mutex_lock(&chip->mixer_lock);
+
+ val1 = (chip->reg_image[left_reg] & ~(mask << shift_left)) | val1;
+ val2 = (chip->reg_image[right_reg] & ~(mask << shift_right)) | val2;
+ change = val1 != chip->reg_image[left_reg]
+ || val2 != chip->reg_image[right_reg];
+ retval = snd_at73c213_write_reg(chip, left_reg, val1);
+ if (retval) {
+ mutex_unlock(&chip->mixer_lock);
+ goto out;
+ }
+ retval = snd_at73c213_write_reg(chip, right_reg, val2);
+ if (retval) {
+ mutex_unlock(&chip->mixer_lock);
+ goto out;
+ }
+
+ mutex_unlock(&chip->mixer_lock);
+
+ return change;
+
+out:
+ return retval;
+}
+
+#define snd_at73c213_mono_switch_info snd_ctl_boolean_mono_info
+
+static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+
+ mutex_lock(&chip->mixer_lock);
+
+ ucontrol->value.integer.value[0] =
+ (chip->reg_image[reg] >> shift) & 0x01;
+
+ if (invert)
+ ucontrol->value.integer.value[0] =
+ 0x01 - ucontrol->value.integer.value[0];
+
+ mutex_unlock(&chip->mixer_lock);
+
+ return 0;
+}
+
+static int snd_at73c213_mono_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol);
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 8) & 0xff;
+ int mask = (kcontrol->private_value >> 16) & 0xff;
+ int invert = (kcontrol->private_value >> 24) & 0xff;
+ int change, retval;
+ unsigned short val;
+
+ if (ucontrol->value.integer.value[0])
+ val = mask;
+ else
+ val = 0;
+
+ if (invert)
+ val = mask - val;
+ val <<= shift;
+
+ mutex_lock(&chip->mixer_lock);
+
+ val |= (chip->reg_image[reg] & ~(mask << shift));
+ change = val != chip->reg_image[reg];
+
+ retval = snd_at73c213_write_reg(chip, reg, val);
+
+ mutex_unlock(&chip->mixer_lock);
+
+ if (retval)
+ return retval;
+
+ return change;
+}
+
+static int snd_at73c213_pa_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = ((kcontrol->private_value >> 16) & 0xff) - 1;
+
+ return 0;
+}
+
+static int snd_at73c213_line_capture_volume_info(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ /* When inverted will give values 0x10001 => 0. */
+ uinfo->value.integer.min = 14;
+ uinfo->value.integer.max = 31;
+
+ return 0;
+}
+
+static int snd_at73c213_aux_capture_volume_info(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ /* When inverted will give values 0x10001 => 0. */
+ uinfo->value.integer.min = 14;
+ uinfo->value.integer.max = 31;
+
+ return 0;
+}
+
+#define AT73C213_MONO_SWITCH(xname, xindex, reg, shift, mask, invert) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_at73c213_mono_switch_info, \
+ .get = snd_at73c213_mono_switch_get, \
+ .put = snd_at73c213_mono_switch_put, \
+ .private_value = (reg | (shift << 8) | (mask << 16) | (invert << 24)) \
+}
+
+#define AT73C213_STEREO(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_at73c213_stereo_info, \
+ .get = snd_at73c213_stereo_get, \
+ .put = snd_at73c213_stereo_put, \
+ .private_value = (left_reg | (right_reg << 8) \
+ | (shift_left << 16) | (shift_right << 19) \
+ | (mask << 24) | (invert << 22)) \
+}
+
+static struct snd_kcontrol_new snd_at73c213_controls[] __devinitdata = {
+AT73C213_STEREO("Master Playback Volume", 0, DAC_LMPG, DAC_RMPG, 0, 0, 0x1f, 1),
+AT73C213_STEREO("Master Playback Switch", 0, DAC_LMPG, DAC_RMPG, 5, 5, 1, 1),
+AT73C213_STEREO("PCM Playback Volume", 0, DAC_LLOG, DAC_RLOG, 0, 0, 0x1f, 1),
+AT73C213_STEREO("PCM Playback Switch", 0, DAC_LLOG, DAC_RLOG, 5, 5, 1, 1),
+AT73C213_MONO_SWITCH("Mono PA Playback Switch", 0, DAC_CTRL, DAC_CTRL_ONPADRV,
+ 0x01, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PA Playback Volume",
+ .index = 0,
+ .info = snd_at73c213_pa_volume_info,
+ .get = snd_at73c213_mono_get,
+ .put = snd_at73c213_mono_put,
+ .private_value = PA_CTRL | (PA_CTRL_APAGAIN << 8) | \
+ (0x0f << 16) | (1 << 24),
+},
+AT73C213_MONO_SWITCH("PA High Gain Playback Switch", 0, PA_CTRL, PA_CTRL_APALP,
+ 0x01, 1),
+AT73C213_MONO_SWITCH("PA Playback Switch", 0, PA_CTRL, PA_CTRL_APAON, 0x01, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Aux Capture Volume",
+ .index = 0,
+ .info = snd_at73c213_aux_capture_volume_info,
+ .get = snd_at73c213_mono_get,
+ .put = snd_at73c213_mono_put,
+ .private_value = DAC_AUXG | (0 << 8) | (0x1f << 16) | (1 << 24),
+},
+AT73C213_MONO_SWITCH("Aux Capture Switch", 0, DAC_CTRL, DAC_CTRL_ONAUXIN,
+ 0x01, 0),
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Line Capture Volume",
+ .index = 0,
+ .info = snd_at73c213_line_capture_volume_info,
+ .get = snd_at73c213_stereo_get,
+ .put = snd_at73c213_stereo_put,
+ .private_value = DAC_LLIG | (DAC_RLIG << 8) | (0 << 16) | (0 << 19)
+ | (0x1f << 24) | (1 << 22),
+},
+AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0),
+};
+
+static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip)
+{
+ struct snd_card *card;
+ int errval, idx;
+
+ if (chip == NULL || chip->pcm == NULL)
+ return -EINVAL;
+
+ card = chip->card;
+
+ strcpy(card->mixername, chip->pcm->name);
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_at73c213_controls); idx++) {
+ errval = snd_ctl_add(card,
+ snd_ctl_new1(&snd_at73c213_controls[idx],
+ chip));
+ if (errval < 0)
+ goto cleanup;
+ }
+
+ return 0;
+
+cleanup:
+ for (idx = 1; idx < ARRAY_SIZE(snd_at73c213_controls) + 1; idx++) {
+ struct snd_kcontrol *kctl;
+ kctl = snd_ctl_find_numid(card, idx);
+ if (kctl)
+ snd_ctl_remove(card, kctl);
+ }
+ return errval;
+}
+
+/*
+ * Device functions
+ */
+static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip)
+{
+ /*
+ * Continuous clock output.
+ * Starts on falling TF.
+ * Delay 1 cycle (1 bit).
+ * Periode is 16 bit (16 - 1).
+ */
+ ssc_writel(chip->ssc->regs, TCMR,
+ SSC_BF(TCMR_CKO, 1)
+ | SSC_BF(TCMR_START, 4)
+ | SSC_BF(TCMR_STTDLY, 1)
+ | SSC_BF(TCMR_PERIOD, 16 - 1));
+ /*
+ * Data length is 16 bit (16 - 1).
+ * Transmit MSB first.
+ * Transmit 2 words each transfer.
+ * Frame sync length is 16 bit (16 - 1).
+ * Frame starts on negative pulse.
+ */
+ ssc_writel(chip->ssc->regs, TFMR,
+ SSC_BF(TFMR_DATLEN, 16 - 1)
+ | SSC_BIT(TFMR_MSBF)
+ | SSC_BF(TFMR_DATNB, 1)
+ | SSC_BF(TFMR_FSLEN, 16 - 1)
+ | SSC_BF(TFMR_FSOS, 1));
+
+ return 0;
+}
+
+static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip)
+{
+ int retval;
+ unsigned char dac_ctrl = 0;
+
+ retval = snd_at73c213_set_bitrate(chip);
+ if (retval)
+ goto out;
+
+ /* Enable DAC master clock. */
+ clk_enable(chip->board->dac_clk);
+
+ /* Initialize at73c213 on SPI bus. */
+ retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04);
+ if (retval)
+ goto out_clk;
+ msleep(1);
+ retval = snd_at73c213_write_reg(chip, DAC_RST, 0x03);
+ if (retval)
+ goto out_clk;
+
+ /* Precharge everything. */
+ retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0xff);
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APAPRECH));
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, DAC_CTRL,
+ (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR));
+ if (retval)
+ goto out_clk;
+
+ msleep(50);
+
+ /* Stop precharging PA. */
+ retval = snd_at73c213_write_reg(chip, PA_CTRL,
+ (1<<PA_CTRL_APALP) | 0x0f);
+ if (retval)
+ goto out_clk;
+
+ msleep(450);
+
+ /* Stop precharging DAC, turn on master power. */
+ retval = snd_at73c213_write_reg(chip, DAC_PRECH, (1<<DAC_PRECH_ONMSTR));
+ if (retval)
+ goto out_clk;
+
+ msleep(1);
+
+ /* Turn on DAC. */
+ dac_ctrl = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR)
+ | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR);
+
+ retval = snd_at73c213_write_reg(chip, DAC_CTRL, dac_ctrl);
+ if (retval)
+ goto out_clk;
+
+ /* Mute sound. */
+ retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f);
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f);
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f);
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f);
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11);
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11);
+ if (retval)
+ goto out_clk;
+ retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11);
+ if (retval)
+ goto out_clk;
+
+ /* Enable I2S device, i.e. clock output. */
+ ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN));
+
+ goto out;
+
+out_clk:
+ clk_disable(chip->board->dac_clk);
+out:
+ return retval;
+}
+
+static int snd_at73c213_dev_free(struct snd_device *device)
+{
+ struct snd_at73c213 *chip = device->device_data;
+
+ ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS));
+ if (chip->irq >= 0) {
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ }
+
+ return 0;
+}
+
+static int __devinit snd_at73c213_dev_init(struct snd_card *card,
+ struct spi_device *spi)
+{
+ static struct snd_device_ops ops = {
+ .dev_free = snd_at73c213_dev_free,
+ };
+ struct snd_at73c213 *chip = get_chip(card);
+ int irq, retval;
+
+ irq = chip->ssc->irq;
+ if (irq < 0)
+ return irq;
+
+ spin_lock_init(&chip->lock);
+ mutex_init(&chip->mixer_lock);
+ chip->card = card;
+ chip->irq = -1;
+
+ retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip);
+ if (retval) {
+ dev_dbg(&chip->spi->dev, "unable to request irq %d\n", irq);
+ goto out;
+ }
+ chip->irq = irq;
+
+ memcpy(&chip->reg_image, &snd_at73c213_original_image,
+ sizeof(snd_at73c213_original_image));
+
+ retval = snd_at73c213_ssc_init(chip);
+ if (retval)
+ goto out_irq;
+
+ retval = snd_at73c213_chip_init(chip);
+ if (retval)
+ goto out_irq;
+
+ retval = snd_at73c213_pcm_new(chip, 0);
+ if (retval)
+ goto out_irq;
+
+ retval = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (retval)
+ goto out_irq;
+
+ retval = snd_at73c213_mixer(chip);
+ if (retval)
+ goto out_snd_dev;
+
+ snd_card_set_dev(card, &spi->dev);
+
+ goto out;
+
+out_snd_dev:
+ snd_device_free(card, chip);
+out_irq:
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+out:
+ return retval;
+}
+
+static int __devinit snd_at73c213_probe(struct spi_device *spi)
+{
+ struct snd_card *card;
+ struct snd_at73c213 *chip;
+ struct at73c213_board_info *board;
+ int retval;
+ char id[16];
+
+ board = spi->dev.platform_data;
+ if (!board) {
+ dev_dbg(&spi->dev, "no platform_data\n");
+ return -ENXIO;
+ }
+
+ if (!board->dac_clk) {
+ dev_dbg(&spi->dev, "no DAC clk\n");
+ return -ENXIO;
+ }
+
+ if (IS_ERR(board->dac_clk)) {
+ dev_dbg(&spi->dev, "no DAC clk\n");
+ return PTR_ERR(board->dac_clk);
+ }
+
+ retval = -ENOMEM;
+
+ /* Allocate "card" using some unused identifiers. */
+ snprintf(id, sizeof id, "at73c213_%d", board->ssc_id);
+ card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct snd_at73c213));
+ if (!card)
+ goto out;
+
+ chip = card->private_data;
+ chip->spi = spi;
+ chip->board = board;
+
+ chip->ssc = ssc_request(board->ssc_id);
+ if (IS_ERR(chip->ssc)) {
+ dev_dbg(&spi->dev, "could not get ssc%d device\n",
+ board->ssc_id);
+ retval = PTR_ERR(chip->ssc);
+ goto out_card;
+ }
+
+ retval = snd_at73c213_dev_init(card, spi);
+ if (retval)
+ goto out_ssc;
+
+ strcpy(card->driver, "at73c213");
+ strcpy(card->shortname, board->shortname);
+ sprintf(card->longname, "%s on irq %d", card->shortname, chip->irq);
+
+ retval = snd_card_register(card);
+ if (retval)
+ goto out_ssc;
+
+ dev_set_drvdata(&spi->dev, card);
+
+ goto out;
+
+out_ssc:
+ ssc_free(chip->ssc);
+out_card:
+ snd_card_free(card);
+out:
+ return retval;
+}
+
+static int __devexit snd_at73c213_remove(struct spi_device *spi)
+{
+ struct snd_card *card = dev_get_drvdata(&spi->dev);
+ struct snd_at73c213 *chip = card->private_data;
+ int retval;
+
+ /* Stop playback. */
+ ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS));
+
+ /* Mute sound. */
+ retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f);
+ if (retval)
+ goto out;
+ retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f);
+ if (retval)
+ goto out;
+ retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f);
+ if (retval)
+ goto out;
+ retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f);
+ if (retval)
+ goto out;
+ retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11);
+ if (retval)
+ goto out;
+ retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11);
+ if (retval)
+ goto out;
+ retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11);
+ if (retval)
+ goto out;
+
+ /* Turn off PA. */
+ retval = snd_at73c213_write_reg(chip, PA_CTRL,
+ chip->reg_image[PA_CTRL] | 0x0f);
+ if (retval)
+ goto out;
+ msleep(10);
+ retval = snd_at73c213_write_reg(chip, PA_CTRL,
+ (1 << PA_CTRL_APALP) | 0x0f);
+ if (retval)
+ goto out;
+
+ /* Turn off external DAC. */
+ retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x0c);
+ if (retval)
+ goto out;
+ msleep(2);
+ retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x00);
+ if (retval)
+ goto out;
+
+ /* Turn off master power. */
+ retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0x00);
+ if (retval)
+ goto out;
+
+out:
+ /* Stop DAC master clock. */
+ clk_disable(chip->board->dac_clk);
+
+ ssc_free(chip->ssc);
+ snd_card_free(card);
+ dev_set_drvdata(&spi->dev, NULL);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg)
+{
+ struct snd_card *card = dev_get_drvdata(&spi->dev);
+ struct snd_at73c213 *chip = card->private_data;
+
+ ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS));
+ clk_disable(chip->board->dac_clk);
+
+ return 0;
+}
+
+static int snd_at73c213_resume(struct spi_device *spi)
+{
+ struct snd_card *card = dev_get_drvdata(&spi->dev);
+ struct snd_at73c213 *chip = card->private_data;
+
+ clk_enable(chip->board->dac_clk);
+ ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN));
+
+ return 0;
+}
+#else
+#define snd_at73c213_suspend NULL
+#define snd_at73c213_resume NULL
+#endif
+
+static struct spi_driver at73c213_driver = {
+ .driver = {
+ .name = "at73c213",
+ },
+ .probe = snd_at73c213_probe,
+ .suspend = snd_at73c213_suspend,
+ .resume = snd_at73c213_resume,
+ .remove = __devexit_p(snd_at73c213_remove),
+};
+
+static int __init at73c213_init(void)
+{
+ return spi_register_driver(&at73c213_driver);
+}
+module_init(at73c213_init);
+
+static void __exit at73c213_exit(void)
+{
+ spi_unregister_driver(&at73c213_driver);
+}
+module_exit(at73c213_exit);
+
+MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt@atmel.com>");
+MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC");
+MODULE_LICENSE("GPL");
diff --git a/sound/spi/at73c213.h b/sound/spi/at73c213.h
new file mode 100644
index 0000000..fd8b372
--- /dev/null
+++ b/sound/spi/at73c213.h
@@ -0,0 +1,119 @@
+/*
+ * Driver for the AT73C213 16-bit stereo DAC on Atmel ATSTK1000
+ *
+ * Copyright (C) 2006 - 2007 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation; either version 2 of the
+ * License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
+ * 02111-1307, USA.
+ *
+ * The full GNU General Public License is included in this
+ * distribution in the file called COPYING.
+ */
+
+#ifndef _SND_AT73C213_H
+#define _SND_AT73C213_H
+
+/* DAC control register */
+#define DAC_CTRL 0x00
+#define DAC_CTRL_ONPADRV 7
+#define DAC_CTRL_ONAUXIN 6
+#define DAC_CTRL_ONDACR 5
+#define DAC_CTRL_ONDACL 4
+#define DAC_CTRL_ONLNOR 3
+#define DAC_CTRL_ONLNOL 2
+#define DAC_CTRL_ONLNIR 1
+#define DAC_CTRL_ONLNIL 0
+
+/* DAC left line in gain register */
+#define DAC_LLIG 0x01
+#define DAC_LLIG_LLIG 0
+
+/* DAC right line in gain register */
+#define DAC_RLIG 0x02
+#define DAC_RLIG_RLIG 0
+
+/* DAC Left Master Playback Gain Register */
+#define DAC_LMPG 0x03
+#define DAC_LMPG_LMPG 0
+
+/* DAC Right Master Playback Gain Register */
+#define DAC_RMPG 0x04
+#define DAC_RMPG_RMPG 0
+
+/* DAC Left Line Out Gain Register */
+#define DAC_LLOG 0x05
+#define DAC_LLOG_LLOG 0
+
+/* DAC Right Line Out Gain Register */
+#define DAC_RLOG 0x06
+#define DAC_RLOG_RLOG 0
+
+/* DAC Output Level Control Register */
+#define DAC_OLC 0x07
+#define DAC_OLC_RSHORT 7
+#define DAC_OLC_ROLC 4
+#define DAC_OLC_LSHORT 3
+#define DAC_OLC_LOLC 0
+
+/* DAC Mixer Control Register */
+#define DAC_MC 0x08
+#define DAC_MC_INVR 5
+#define DAC_MC_INVL 4
+#define DAC_MC_RMSMIN2 3
+#define DAC_MC_RMSMIN1 2
+#define DAC_MC_LMSMIN2 1
+#define DAC_MC_LMSMIN1 0
+
+/* DAC Clock and Sampling Frequency Control Register */
+#define DAC_CSFC 0x09
+#define DAC_CSFC_OVRSEL 4
+
+/* DAC Miscellaneous Register */
+#define DAC_MISC 0x0A
+#define DAC_MISC_VCMCAPSEL 7
+#define DAC_MISC_DINTSEL 4
+#define DAC_MISC_DITHEN 3
+#define DAC_MISC_DEEMPEN 2
+#define DAC_MISC_NBITS 0
+
+/* DAC Precharge Control Register */
+#define DAC_PRECH 0x0C
+#define DAC_PRECH_PRCHGPDRV 7
+#define DAC_PRECH_PRCHGAUX1 6
+#define DAC_PRECH_PRCHGLNOR 5
+#define DAC_PRECH_PRCHGLNOL 4
+#define DAC_PRECH_PRCHGLNIR 3
+#define DAC_PRECH_PRCHGLNIL 2
+#define DAC_PRECH_PRCHG 1
+#define DAC_PRECH_ONMSTR 0
+
+/* DAC Auxiliary Input Gain Control Register */
+#define DAC_AUXG 0x0D
+#define DAC_AUXG_AUXG 0
+
+/* DAC Reset Register */
+#define DAC_RST 0x10
+#define DAC_RST_RESMASK 2
+#define DAC_RST_RESFILZ 1
+#define DAC_RST_RSTZ 0
+
+/* Power Amplifier Control Register */
+#define PA_CTRL 0x11
+#define PA_CTRL_APAON 6
+#define PA_CTRL_APAPRECH 5
+#define PA_CTRL_APALP 4
+#define PA_CTRL_APAGAIN 0
+
+#endif /* _SND_AT73C213_H */
diff --git a/sound/synth/Makefile b/sound/synth/Makefile
new file mode 100644
index 0000000..e99fd76
--- /dev/null
+++ b/sound/synth/Makefile
@@ -0,0 +1,20 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-util-mem-objs := util_mem.o
+
+#
+# this function returns:
+# "m" - CONFIG_SND_SEQUENCER is m
+# <empty string> - CONFIG_SND_SEQUENCER is undefined
+# otherwise parameter #1 value
+#
+sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_EMU10K1) += snd-util-mem.o
+obj-$(CONFIG_SND_TRIDENT) += snd-util-mem.o
+obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-util-mem.o
+obj-$(call sequencer,$(CONFIG_SND)) += emux/
diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile
new file mode 100644
index 0000000..b690352
--- /dev/null
+++ b/sound/synth/emux/Makefile
@@ -0,0 +1,20 @@
+#
+# Makefile for ALSA
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
+#
+
+snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \
+ emux_effect.o emux_proc.o emux_hwdep.o soundfont.o \
+ $(if $(CONFIG_SND_SEQUENCER_OSS),emux_oss.o)
+
+#
+# this function returns:
+# "m" - CONFIG_SND_SEQUENCER is m
+# <empty string> - CONFIG_SND_SEQUENCER is undefined
+# otherwise parameter #1 value
+#
+sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
+
+# Toplevel Module Dependencies
+obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emux-synth.o
+obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emux-synth.o
diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c
new file mode 100644
index 0000000..f16a3fc
--- /dev/null
+++ b/sound/synth/emux/emux.c
@@ -0,0 +1,191 @@
+/*
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Routines for control of EMU WaveTable chip
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/wait.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/emux_synth.h>
+#include <linux/init.h>
+#include "emux_voice.h"
+
+MODULE_AUTHOR("Takashi Iwai");
+MODULE_DESCRIPTION("Routines for control of EMU WaveTable chip");
+MODULE_LICENSE("GPL");
+
+/*
+ * create a new hardware dependent device for Emu8000/Emu10k1
+ */
+int snd_emux_new(struct snd_emux **remu)
+{
+ struct snd_emux *emu;
+
+ *remu = NULL;
+ emu = kzalloc(sizeof(*emu), GFP_KERNEL);
+ if (emu == NULL)
+ return -ENOMEM;
+
+ spin_lock_init(&emu->voice_lock);
+ mutex_init(&emu->register_mutex);
+
+ emu->client = -1;
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ emu->oss_synth = NULL;
+#endif
+ emu->max_voices = 0;
+ emu->use_time = 0;
+
+ init_timer(&emu->tlist);
+ emu->tlist.function = snd_emux_timer_callback;
+ emu->tlist.data = (unsigned long)emu;
+ emu->timer_active = 0;
+
+ *remu = emu;
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_emux_new);
+
+/*
+ */
+static int sf_sample_new(void *private_data, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr,
+ const void __user *buf, long count)
+{
+ struct snd_emux *emu = private_data;
+ return emu->ops.sample_new(emu, sp, hdr, buf, count);
+
+}
+
+static int sf_sample_free(void *private_data, struct snd_sf_sample *sp,
+ struct snd_util_memhdr *hdr)
+{
+ struct snd_emux *emu = private_data;
+ return emu->ops.sample_free(emu, sp, hdr);
+
+}
+
+static void sf_sample_reset(void *private_data)
+{
+ struct snd_emux *emu = private_data;
+ emu->ops.sample_reset(emu);
+}
+
+int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, char *name)
+{
+ int err;
+ struct snd_sf_callback sf_cb;
+
+ if (snd_BUG_ON(!emu->hw || emu->max_voices <= 0))
+ return -EINVAL;
+ if (snd_BUG_ON(!card || !name))
+ return -EINVAL;
+
+ emu->card = card;
+ emu->name = kstrdup(name, GFP_KERNEL);
+ emu->voices = kcalloc(emu->max_voices, sizeof(struct snd_emux_voice),
+ GFP_KERNEL);
+ if (emu->voices == NULL)
+ return -ENOMEM;
+
+ /* create soundfont list */
+ memset(&sf_cb, 0, sizeof(sf_cb));
+ sf_cb.private_data = emu;
+ if (emu->ops.sample_new)
+ sf_cb.sample_new = sf_sample_new;
+ if (emu->ops.sample_free)
+ sf_cb.sample_free = sf_sample_free;
+ if (emu->ops.sample_reset)
+ sf_cb.sample_reset = sf_sample_reset;
+ emu->sflist = snd_sf_new(&sf_cb, emu->memhdr);
+ if (emu->sflist == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_emux_init_hwdep(emu)) < 0)
+ return err;
+
+ snd_emux_init_voices(emu);
+
+ snd_emux_init_seq(emu, card, index);
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ snd_emux_init_seq_oss(emu);
+#endif
+ snd_emux_init_virmidi(emu, card);
+
+#ifdef CONFIG_PROC_FS
+ snd_emux_proc_init(emu, card, index);
+#endif
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_emux_register);
+
+/*
+ */
+int snd_emux_free(struct snd_emux *emu)
+{
+ unsigned long flags;
+
+ if (! emu)
+ return -EINVAL;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ if (emu->timer_active)
+ del_timer(&emu->tlist);
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+
+#ifdef CONFIG_PROC_FS
+ snd_emux_proc_free(emu);
+#endif
+ snd_emux_delete_virmidi(emu);
+#ifdef CONFIG_SND_SEQUENCER_OSS
+ snd_emux_detach_seq_oss(emu);
+#endif
+ snd_emux_detach_seq(emu);
+
+ snd_emux_delete_hwdep(emu);
+
+ if (emu->sflist)
+ snd_sf_free(emu->sflist);
+
+ kfree(emu->voices);
+ kfree(emu->name);
+ kfree(emu);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_emux_free);
+
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_emux_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_emux_exit(void)
+{
+}
+
+module_init(alsa_emux_init)
+module_exit(alsa_emux_exit)
diff --git a/sound/synth/emux/emux_effect.c b/sound/synth/emux/emux_effect.c
new file mode 100644
index 0000000..a447218
--- /dev/null
+++ b/sound/synth/emux/emux_effect.c
@@ -0,0 +1,310 @@
+/*
+ * Midi synth routines for the Emu8k/Emu10k1
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Contains code based on awe_wave.c by Takashi Iwai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include "emux_voice.h"
+#include <linux/slab.h>
+
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+/*
+ * effects table
+ */
+
+#define xoffsetof(type,tag) ((long)(&((type)NULL)->tag) - (long)(NULL))
+
+#define parm_offset(tag) xoffsetof(struct soundfont_voice_parm *, tag)
+
+#define PARM_IS_BYTE (1 << 0)
+#define PARM_IS_WORD (1 << 1)
+#define PARM_IS_ALIGNED (3 << 2)
+#define PARM_IS_ALIGN_HI (1 << 2)
+#define PARM_IS_ALIGN_LO (2 << 2)
+#define PARM_IS_SIGNED (1 << 4)
+
+#define PARM_WORD (PARM_IS_WORD)
+#define PARM_BYTE_LO (PARM_IS_BYTE|PARM_IS_ALIGN_LO)
+#define PARM_BYTE_HI (PARM_IS_BYTE|PARM_IS_ALIGN_HI)
+#define PARM_BYTE (PARM_IS_BYTE)
+#define PARM_SIGN_LO (PARM_IS_BYTE|PARM_IS_ALIGN_LO|PARM_IS_SIGNED)
+#define PARM_SIGN_HI (PARM_IS_BYTE|PARM_IS_ALIGN_HI|PARM_IS_SIGNED)
+
+static struct emux_parm_defs {
+ int type; /* byte or word */
+ int low, high; /* value range */
+ long offset; /* offset in parameter record (-1 = not written) */
+ int update; /* flgas for real-time update */
+} parm_defs[EMUX_NUM_EFFECTS] = {
+ {PARM_WORD, 0, 0x8000, parm_offset(moddelay), 0}, /* env1 delay */
+ {PARM_BYTE_LO, 1, 0x80, parm_offset(modatkhld), 0}, /* env1 attack */
+ {PARM_BYTE_HI, 0, 0x7e, parm_offset(modatkhld), 0}, /* env1 hold */
+ {PARM_BYTE_LO, 1, 0x7f, parm_offset(moddcysus), 0}, /* env1 decay */
+ {PARM_BYTE_LO, 1, 0x7f, parm_offset(modrelease), 0}, /* env1 release */
+ {PARM_BYTE_HI, 0, 0x7f, parm_offset(moddcysus), 0}, /* env1 sustain */
+ {PARM_BYTE_HI, 0, 0xff, parm_offset(pefe), 0}, /* env1 pitch */
+ {PARM_BYTE_LO, 0, 0xff, parm_offset(pefe), 0}, /* env1 fc */
+
+ {PARM_WORD, 0, 0x8000, parm_offset(voldelay), 0}, /* env2 delay */
+ {PARM_BYTE_LO, 1, 0x80, parm_offset(volatkhld), 0}, /* env2 attack */
+ {PARM_BYTE_HI, 0, 0x7e, parm_offset(volatkhld), 0}, /* env2 hold */
+ {PARM_BYTE_LO, 1, 0x7f, parm_offset(voldcysus), 0}, /* env2 decay */
+ {PARM_BYTE_LO, 1, 0x7f, parm_offset(volrelease), 0}, /* env2 release */
+ {PARM_BYTE_HI, 0, 0x7f, parm_offset(voldcysus), 0}, /* env2 sustain */
+
+ {PARM_WORD, 0, 0x8000, parm_offset(lfo1delay), 0}, /* lfo1 delay */
+ {PARM_BYTE_LO, 0, 0xff, parm_offset(tremfrq), SNDRV_EMUX_UPDATE_TREMFREQ}, /* lfo1 freq */
+ {PARM_SIGN_HI, -128, 127, parm_offset(tremfrq), SNDRV_EMUX_UPDATE_TREMFREQ}, /* lfo1 vol */
+ {PARM_SIGN_HI, -128, 127, parm_offset(fmmod), SNDRV_EMUX_UPDATE_FMMOD}, /* lfo1 pitch */
+ {PARM_BYTE_LO, 0, 0xff, parm_offset(fmmod), SNDRV_EMUX_UPDATE_FMMOD}, /* lfo1 cutoff */
+
+ {PARM_WORD, 0, 0x8000, parm_offset(lfo2delay), 0}, /* lfo2 delay */
+ {PARM_BYTE_LO, 0, 0xff, parm_offset(fm2frq2), SNDRV_EMUX_UPDATE_FM2FRQ2}, /* lfo2 freq */
+ {PARM_SIGN_HI, -128, 127, parm_offset(fm2frq2), SNDRV_EMUX_UPDATE_FM2FRQ2}, /* lfo2 pitch */
+
+ {PARM_WORD, 0, 0xffff, -1, SNDRV_EMUX_UPDATE_PITCH}, /* initial pitch */
+ {PARM_BYTE, 0, 0xff, parm_offset(chorus), 0}, /* chorus */
+ {PARM_BYTE, 0, 0xff, parm_offset(reverb), 0}, /* reverb */
+ {PARM_BYTE, 0, 0xff, parm_offset(cutoff), SNDRV_EMUX_UPDATE_VOLUME}, /* cutoff */
+ {PARM_BYTE, 0, 15, parm_offset(filterQ), SNDRV_EMUX_UPDATE_Q}, /* resonance */
+
+ {PARM_WORD, 0, 0xffff, -1, 0}, /* sample start */
+ {PARM_WORD, 0, 0xffff, -1, 0}, /* loop start */
+ {PARM_WORD, 0, 0xffff, -1, 0}, /* loop end */
+ {PARM_WORD, 0, 0xffff, -1, 0}, /* coarse sample start */
+ {PARM_WORD, 0, 0xffff, -1, 0}, /* coarse loop start */
+ {PARM_WORD, 0, 0xffff, -1, 0}, /* coarse loop end */
+ {PARM_BYTE, 0, 0xff, -1, SNDRV_EMUX_UPDATE_VOLUME}, /* initial attenuation */
+};
+
+/* set byte effect value */
+static void
+effect_set_byte(unsigned char *valp, struct snd_midi_channel *chan, int type)
+{
+ short effect;
+ struct snd_emux_effect_table *fx = chan->private;
+
+ effect = fx->val[type];
+ if (fx->flag[type] == EMUX_FX_FLAG_ADD) {
+ if (parm_defs[type].type & PARM_IS_SIGNED)
+ effect += *(char*)valp;
+ else
+ effect += *valp;
+ }
+ if (effect < parm_defs[type].low)
+ effect = parm_defs[type].low;
+ else if (effect > parm_defs[type].high)
+ effect = parm_defs[type].high;
+ *valp = (unsigned char)effect;
+}
+
+/* set word effect value */
+static void
+effect_set_word(unsigned short *valp, struct snd_midi_channel *chan, int type)
+{
+ int effect;
+ struct snd_emux_effect_table *fx = chan->private;
+
+ effect = *(unsigned short*)&fx->val[type];
+ if (fx->flag[type] == EMUX_FX_FLAG_ADD)
+ effect += *valp;
+ if (effect < parm_defs[type].low)
+ effect = parm_defs[type].low;
+ else if (effect > parm_defs[type].high)
+ effect = parm_defs[type].high;
+ *valp = (unsigned short)effect;
+}
+
+/* address offset */
+static int
+effect_get_offset(struct snd_midi_channel *chan, int lo, int hi, int mode)
+{
+ int addr = 0;
+ struct snd_emux_effect_table *fx = chan->private;
+
+ if (fx->flag[hi])
+ addr = (short)fx->val[hi];
+ addr = addr << 15;
+ if (fx->flag[lo])
+ addr += (short)fx->val[lo];
+ if (!(mode & SNDRV_SFNT_SAMPLE_8BITS))
+ addr /= 2;
+ return addr;
+}
+
+#ifdef CONFIG_SND_SEQUENCER_OSS
+/* change effects - for OSS sequencer compatibility */
+void
+snd_emux_send_effect_oss(struct snd_emux_port *port,
+ struct snd_midi_channel *chan, int type, int val)
+{
+ int mode;
+
+ if (type & 0x40)
+ mode = EMUX_FX_FLAG_OFF;
+ else if (type & 0x80)
+ mode = EMUX_FX_FLAG_ADD;
+ else
+ mode = EMUX_FX_FLAG_SET;
+ type &= 0x3f;
+
+ snd_emux_send_effect(port, chan, type, val, mode);
+}
+#endif
+
+/* Modify the effect value.
+ * if update is necessary, call emu8000_control
+ */
+void
+snd_emux_send_effect(struct snd_emux_port *port, struct snd_midi_channel *chan,
+ int type, int val, int mode)
+{
+ int i;
+ int offset;
+ unsigned char *srcp, *origp;
+ struct snd_emux *emu;
+ struct snd_emux_effect_table *fx;
+ unsigned long flags;
+
+ emu = port->emu;
+ fx = chan->private;
+ if (emu == NULL || fx == NULL)
+ return;
+ if (type < 0 || type >= EMUX_NUM_EFFECTS)
+ return;
+
+ fx->val[type] = val;
+ fx->flag[type] = mode;
+
+ /* do we need to modify the register in realtime ? */
+ if (! parm_defs[type].update || (offset = parm_defs[type].offset) < 0)
+ return;
+
+#ifdef SNDRV_LITTLE_ENDIAN
+ if (parm_defs[type].type & PARM_IS_ALIGN_HI)
+ offset++;
+#else
+ if (parm_defs[type].type & PARM_IS_ALIGN_LO)
+ offset++;
+#endif
+ /* modify the register values */
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ struct snd_emux_voice *vp = &emu->voices[i];
+ if (!STATE_IS_PLAYING(vp->state) || vp->chan != chan)
+ continue;
+ srcp = (unsigned char*)&vp->reg.parm + offset;
+ origp = (unsigned char*)&vp->zone->v.parm + offset;
+ if (parm_defs[i].type & PARM_IS_BYTE) {
+ *srcp = *origp;
+ effect_set_byte(srcp, chan, type);
+ } else {
+ *(unsigned short*)srcp = *(unsigned short*)origp;
+ effect_set_word((unsigned short*)srcp, chan, type);
+ }
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+
+ /* activate them */
+ snd_emux_update_channel(port, chan, parm_defs[type].update);
+}
+
+
+/* copy wavetable registers to voice table */
+void
+snd_emux_setup_effect(struct snd_emux_voice *vp)
+{
+ struct snd_midi_channel *chan = vp->chan;
+ struct snd_emux_effect_table *fx;
+ unsigned char *srcp;
+ int i;
+
+ if (! (fx = chan->private))
+ return;
+
+ /* modify the register values via effect table */
+ for (i = 0; i < EMUX_FX_END; i++) {
+ int offset;
+ if (! fx->flag[i] || (offset = parm_defs[i].offset) < 0)
+ continue;
+#ifdef SNDRV_LITTLE_ENDIAN
+ if (parm_defs[i].type & PARM_IS_ALIGN_HI)
+ offset++;
+#else
+ if (parm_defs[i].type & PARM_IS_ALIGN_LO)
+ offset++;
+#endif
+ srcp = (unsigned char*)&vp->reg.parm + offset;
+ if (parm_defs[i].type & PARM_IS_BYTE)
+ effect_set_byte(srcp, chan, i);
+ else
+ effect_set_word((unsigned short*)srcp, chan, i);
+ }
+
+ /* correct sample and loop points */
+ vp->reg.start += effect_get_offset(chan, EMUX_FX_SAMPLE_START,
+ EMUX_FX_COARSE_SAMPLE_START,
+ vp->reg.sample_mode);
+
+ vp->reg.loopstart += effect_get_offset(chan, EMUX_FX_LOOP_START,
+ EMUX_FX_COARSE_LOOP_START,
+ vp->reg.sample_mode);
+
+ vp->reg.loopend += effect_get_offset(chan, EMUX_FX_LOOP_END,
+ EMUX_FX_COARSE_LOOP_END,
+ vp->reg.sample_mode);
+}
+
+/*
+ * effect table
+ */
+void
+snd_emux_create_effect(struct snd_emux_port *p)
+{
+ int i;
+ p->effect = kcalloc(p->chset.max_channels,
+ sizeof(struct snd_emux_effect_table), GFP_KERNEL);
+ if (p->effect) {
+ for (i = 0; i < p->chset.max_channels; i++)
+ p->chset.channels[i].private = p->effect + i;
+ } else {
+ for (i = 0; i < p->chset.max_channels; i++)
+ p->chset.channels[i].private = NULL;
+ }
+}
+
+void
+snd_emux_delete_effect(struct snd_emux_port *p)
+{
+ kfree(p->effect);
+ p->effect = NULL;
+}
+
+void
+snd_emux_clear_effect(struct snd_emux_port *p)
+{
+ if (p->effect) {
+ memset(p->effect, 0, sizeof(struct snd_emux_effect_table) *
+ p->chset.max_channels);
+ }
+}
+
+#endif /* SNDRV_EMUX_USE_RAW_EFFECT */
diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c
new file mode 100644
index 0000000..0a53914
--- /dev/null
+++ b/sound/synth/emux/emux_hwdep.c
@@ -0,0 +1,171 @@
+/*
+ * Interface for hwdep device
+ *
+ * Copyright (C) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <sound/core.h>
+#include <sound/hwdep.h>
+#include <asm/uaccess.h>
+#include "emux_voice.h"
+
+/*
+ * open the hwdep device
+ */
+static int
+snd_emux_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+
+/*
+ * close the device
+ */
+static int
+snd_emux_hwdep_release(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+
+#define TMP_CLIENT_ID 0x1001
+
+/*
+ * load patch
+ */
+static int
+snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg)
+{
+ int err;
+ struct soundfont_patch_info patch;
+
+ if (copy_from_user(&patch, arg, sizeof(patch)))
+ return -EFAULT;
+
+ if (patch.type >= SNDRV_SFNT_LOAD_INFO &&
+ patch.type <= SNDRV_SFNT_PROBE_DATA) {
+ err = snd_soundfont_load(emu->sflist, arg, patch.len + sizeof(patch), TMP_CLIENT_ID);
+ if (err < 0)
+ return err;
+ } else {
+ if (emu->ops.load_fx)
+ return emu->ops.load_fx(emu, patch.type, patch.optarg, arg, patch.len + sizeof(patch));
+ else
+ return -EINVAL;
+ }
+ return 0;
+}
+
+/*
+ * set misc mode
+ */
+static int
+snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg)
+{
+ struct snd_emux_misc_mode info;
+ int i;
+
+ if (copy_from_user(&info, arg, sizeof(info)))
+ return -EFAULT;
+ if (info.mode < 0 || info.mode >= EMUX_MD_END)
+ return -EINVAL;
+
+ if (info.port < 0) {
+ for (i = 0; i < emu->num_ports; i++)
+ emu->portptrs[i]->ctrls[info.mode] = info.value;
+ } else {
+ if (info.port < emu->num_ports)
+ emu->portptrs[info.port]->ctrls[info.mode] = info.value;
+ }
+ return 0;
+}
+
+
+/*
+ * ioctl
+ */
+static int
+snd_emux_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_emux *emu = hw->private_data;
+
+ switch (cmd) {
+ case SNDRV_EMUX_IOCTL_VERSION:
+ return put_user(SNDRV_EMUX_VERSION, (unsigned int __user *)arg);
+ case SNDRV_EMUX_IOCTL_LOAD_PATCH:
+ return snd_emux_hwdep_load_patch(emu, (void __user *)arg);
+ case SNDRV_EMUX_IOCTL_RESET_SAMPLES:
+ snd_soundfont_remove_samples(emu->sflist);
+ break;
+ case SNDRV_EMUX_IOCTL_REMOVE_LAST_SAMPLES:
+ snd_soundfont_remove_unlocked(emu->sflist);
+ break;
+ case SNDRV_EMUX_IOCTL_MEM_AVAIL:
+ if (emu->memhdr) {
+ int size = snd_util_mem_avail(emu->memhdr);
+ return put_user(size, (unsigned int __user *)arg);
+ }
+ break;
+ case SNDRV_EMUX_IOCTL_MISC_MODE:
+ return snd_emux_hwdep_misc_mode(emu, (void __user *)arg);
+ }
+
+ return 0;
+}
+
+
+/*
+ * register hwdep device
+ */
+
+int
+snd_emux_init_hwdep(struct snd_emux *emu)
+{
+ struct snd_hwdep *hw;
+ int err;
+
+ if ((err = snd_hwdep_new(emu->card, SNDRV_EMUX_HWDEP_NAME, emu->hwdep_idx, &hw)) < 0)
+ return err;
+ emu->hwdep = hw;
+ strcpy(hw->name, SNDRV_EMUX_HWDEP_NAME);
+ hw->iface = SNDRV_HWDEP_IFACE_EMUX_WAVETABLE;
+ hw->ops.open = snd_emux_hwdep_open;
+ hw->ops.release = snd_emux_hwdep_release;
+ hw->ops.ioctl = snd_emux_hwdep_ioctl;
+ hw->exclusive = 1;
+ hw->private_data = emu;
+ if ((err = snd_card_register(emu->card)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * unregister
+ */
+void
+snd_emux_delete_hwdep(struct snd_emux *emu)
+{
+ if (emu->hwdep) {
+ snd_device_free(emu->card, emu->hwdep);
+ emu->hwdep = NULL;
+ }
+}
diff --git a/sound/synth/emux/emux_nrpn.c b/sound/synth/emux/emux_nrpn.c
new file mode 100644
index 0000000..00fc005
--- /dev/null
+++ b/sound/synth/emux/emux_nrpn.c
@@ -0,0 +1,396 @@
+/*
+ * NRPN / SYSEX callbacks for Emu8k/Emu10k1
+ *
+ * Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include "emux_voice.h"
+#include <sound/asoundef.h>
+
+/*
+ * conversion from NRPN/control parameters to Emu8000 raw parameters
+ */
+
+/* NRPN / CC -> Emu8000 parameter converter */
+struct nrpn_conv_table {
+ int control;
+ int effect;
+ int (*convert)(int val);
+};
+
+/* effect sensitivity */
+
+#define FX_CUTOFF 0
+#define FX_RESONANCE 1
+#define FX_ATTACK 2
+#define FX_RELEASE 3
+#define FX_VIBRATE 4
+#define FX_VIBDEPTH 5
+#define FX_VIBDELAY 6
+#define FX_NUMS 7
+
+/*
+ * convert NRPN/control values
+ */
+
+static int send_converted_effect(struct nrpn_conv_table *table, int num_tables,
+ struct snd_emux_port *port,
+ struct snd_midi_channel *chan,
+ int type, int val, int mode)
+{
+ int i, cval;
+ for (i = 0; i < num_tables; i++) {
+ if (table[i].control == type) {
+ cval = table[i].convert(val);
+ snd_emux_send_effect(port, chan, table[i].effect,
+ cval, mode);
+ return 1;
+ }
+ }
+ return 0;
+}
+
+#define DEF_FX_CUTOFF 170
+#define DEF_FX_RESONANCE 6
+#define DEF_FX_ATTACK 50
+#define DEF_FX_RELEASE 50
+#define DEF_FX_VIBRATE 30
+#define DEF_FX_VIBDEPTH 4
+#define DEF_FX_VIBDELAY 1500
+
+/* effect sensitivities for GS NRPN:
+ * adjusted for chaos 8MB soundfonts
+ */
+static int gs_sense[] =
+{
+ DEF_FX_CUTOFF, DEF_FX_RESONANCE, DEF_FX_ATTACK, DEF_FX_RELEASE,
+ DEF_FX_VIBRATE, DEF_FX_VIBDEPTH, DEF_FX_VIBDELAY
+};
+
+/* effect sensitivies for XG controls:
+ * adjusted for chaos 8MB soundfonts
+ */
+static int xg_sense[] =
+{
+ DEF_FX_CUTOFF, DEF_FX_RESONANCE, DEF_FX_ATTACK, DEF_FX_RELEASE,
+ DEF_FX_VIBRATE, DEF_FX_VIBDEPTH, DEF_FX_VIBDELAY
+};
+
+
+/*
+ * AWE32 NRPN effects
+ */
+
+static int fx_delay(int val);
+static int fx_attack(int val);
+static int fx_hold(int val);
+static int fx_decay(int val);
+static int fx_the_value(int val);
+static int fx_twice_value(int val);
+static int fx_conv_pitch(int val);
+static int fx_conv_Q(int val);
+
+/* function for each NRPN */ /* [range] units */
+#define fx_env1_delay fx_delay /* [0,5900] 4msec */
+#define fx_env1_attack fx_attack /* [0,5940] 1msec */
+#define fx_env1_hold fx_hold /* [0,8191] 1msec */
+#define fx_env1_decay fx_decay /* [0,5940] 4msec */
+#define fx_env1_release fx_decay /* [0,5940] 4msec */
+#define fx_env1_sustain fx_the_value /* [0,127] 0.75dB */
+#define fx_env1_pitch fx_the_value /* [-127,127] 9.375cents */
+#define fx_env1_cutoff fx_the_value /* [-127,127] 56.25cents */
+
+#define fx_env2_delay fx_delay /* [0,5900] 4msec */
+#define fx_env2_attack fx_attack /* [0,5940] 1msec */
+#define fx_env2_hold fx_hold /* [0,8191] 1msec */
+#define fx_env2_decay fx_decay /* [0,5940] 4msec */
+#define fx_env2_release fx_decay /* [0,5940] 4msec */
+#define fx_env2_sustain fx_the_value /* [0,127] 0.75dB */
+
+#define fx_lfo1_delay fx_delay /* [0,5900] 4msec */
+#define fx_lfo1_freq fx_twice_value /* [0,127] 84mHz */
+#define fx_lfo1_volume fx_twice_value /* [0,127] 0.1875dB */
+#define fx_lfo1_pitch fx_the_value /* [-127,127] 9.375cents */
+#define fx_lfo1_cutoff fx_twice_value /* [-64,63] 56.25cents */
+
+#define fx_lfo2_delay fx_delay /* [0,5900] 4msec */
+#define fx_lfo2_freq fx_twice_value /* [0,127] 84mHz */
+#define fx_lfo2_pitch fx_the_value /* [-127,127] 9.375cents */
+
+#define fx_init_pitch fx_conv_pitch /* [-8192,8192] cents */
+#define fx_chorus fx_the_value /* [0,255] -- */
+#define fx_reverb fx_the_value /* [0,255] -- */
+#define fx_cutoff fx_twice_value /* [0,127] 62Hz */
+#define fx_filterQ fx_conv_Q /* [0,127] -- */
+
+static int fx_delay(int val)
+{
+ return (unsigned short)snd_sf_calc_parm_delay(val);
+}
+
+static int fx_attack(int val)
+{
+ return (unsigned short)snd_sf_calc_parm_attack(val);
+}
+
+static int fx_hold(int val)
+{
+ return (unsigned short)snd_sf_calc_parm_hold(val);
+}
+
+static int fx_decay(int val)
+{
+ return (unsigned short)snd_sf_calc_parm_decay(val);
+}
+
+static int fx_the_value(int val)
+{
+ return (unsigned short)(val & 0xff);
+}
+
+static int fx_twice_value(int val)
+{
+ return (unsigned short)((val * 2) & 0xff);
+}
+
+static int fx_conv_pitch(int val)
+{
+ return (short)(val * 4096 / 1200);
+}
+
+static int fx_conv_Q(int val)
+{
+ return (unsigned short)((val / 8) & 0xff);
+}
+
+
+static struct nrpn_conv_table awe_effects[] =
+{
+ { 0, EMUX_FX_LFO1_DELAY, fx_lfo1_delay},
+ { 1, EMUX_FX_LFO1_FREQ, fx_lfo1_freq},
+ { 2, EMUX_FX_LFO2_DELAY, fx_lfo2_delay},
+ { 3, EMUX_FX_LFO2_FREQ, fx_lfo2_freq},
+
+ { 4, EMUX_FX_ENV1_DELAY, fx_env1_delay},
+ { 5, EMUX_FX_ENV1_ATTACK,fx_env1_attack},
+ { 6, EMUX_FX_ENV1_HOLD, fx_env1_hold},
+ { 7, EMUX_FX_ENV1_DECAY, fx_env1_decay},
+ { 8, EMUX_FX_ENV1_SUSTAIN, fx_env1_sustain},
+ { 9, EMUX_FX_ENV1_RELEASE, fx_env1_release},
+
+ {10, EMUX_FX_ENV2_DELAY, fx_env2_delay},
+ {11, EMUX_FX_ENV2_ATTACK, fx_env2_attack},
+ {12, EMUX_FX_ENV2_HOLD, fx_env2_hold},
+ {13, EMUX_FX_ENV2_DECAY, fx_env2_decay},
+ {14, EMUX_FX_ENV2_SUSTAIN, fx_env2_sustain},
+ {15, EMUX_FX_ENV2_RELEASE, fx_env2_release},
+
+ {16, EMUX_FX_INIT_PITCH, fx_init_pitch},
+ {17, EMUX_FX_LFO1_PITCH, fx_lfo1_pitch},
+ {18, EMUX_FX_LFO2_PITCH, fx_lfo2_pitch},
+ {19, EMUX_FX_ENV1_PITCH, fx_env1_pitch},
+ {20, EMUX_FX_LFO1_VOLUME, fx_lfo1_volume},
+ {21, EMUX_FX_CUTOFF, fx_cutoff},
+ {22, EMUX_FX_FILTERQ, fx_filterQ},
+ {23, EMUX_FX_LFO1_CUTOFF, fx_lfo1_cutoff},
+ {24, EMUX_FX_ENV1_CUTOFF, fx_env1_cutoff},
+ {25, EMUX_FX_CHORUS, fx_chorus},
+ {26, EMUX_FX_REVERB, fx_reverb},
+};
+
+
+/*
+ * GS(SC88) NRPN effects; still experimental
+ */
+
+/* cutoff: quarter semitone step, max=255 */
+static int gs_cutoff(int val)
+{
+ return (val - 64) * gs_sense[FX_CUTOFF] / 50;
+}
+
+/* resonance: 0 to 15(max) */
+static int gs_filterQ(int val)
+{
+ return (val - 64) * gs_sense[FX_RESONANCE] / 50;
+}
+
+/* attack: */
+static int gs_attack(int val)
+{
+ return -(val - 64) * gs_sense[FX_ATTACK] / 50;
+}
+
+/* decay: */
+static int gs_decay(int val)
+{
+ return -(val - 64) * gs_sense[FX_RELEASE] / 50;
+}
+
+/* release: */
+static int gs_release(int val)
+{
+ return -(val - 64) * gs_sense[FX_RELEASE] / 50;
+}
+
+/* vibrato freq: 0.042Hz step, max=255 */
+static int gs_vib_rate(int val)
+{
+ return (val - 64) * gs_sense[FX_VIBRATE] / 50;
+}
+
+/* vibrato depth: max=127, 1 octave */
+static int gs_vib_depth(int val)
+{
+ return (val - 64) * gs_sense[FX_VIBDEPTH] / 50;
+}
+
+/* vibrato delay: -0.725msec step */
+static int gs_vib_delay(int val)
+{
+ return -(val - 64) * gs_sense[FX_VIBDELAY] / 50;
+}
+
+static struct nrpn_conv_table gs_effects[] =
+{
+ {32, EMUX_FX_CUTOFF, gs_cutoff},
+ {33, EMUX_FX_FILTERQ, gs_filterQ},
+ {99, EMUX_FX_ENV2_ATTACK, gs_attack},
+ {100, EMUX_FX_ENV2_DECAY, gs_decay},
+ {102, EMUX_FX_ENV2_RELEASE, gs_release},
+ {8, EMUX_FX_LFO1_FREQ, gs_vib_rate},
+ {9, EMUX_FX_LFO1_VOLUME, gs_vib_depth},
+ {10, EMUX_FX_LFO1_DELAY, gs_vib_delay},
+};
+
+
+/*
+ * NRPN events
+ */
+void
+snd_emux_nrpn(void *p, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset)
+{
+ struct snd_emux_port *port;
+
+ port = p;
+ if (snd_BUG_ON(!port || !chan))
+ return;
+
+ if (chan->control[MIDI_CTL_NONREG_PARM_NUM_MSB] == 127 &&
+ chan->control[MIDI_CTL_NONREG_PARM_NUM_LSB] <= 26) {
+ int val;
+ /* Win/DOS AWE32 specific NRPNs */
+ /* both MSB/LSB necessary */
+ val = (chan->control[MIDI_CTL_MSB_DATA_ENTRY] << 7) |
+ chan->control[MIDI_CTL_LSB_DATA_ENTRY];
+ val -= 8192;
+ send_converted_effect
+ (awe_effects, ARRAY_SIZE(awe_effects),
+ port, chan, chan->control[MIDI_CTL_NONREG_PARM_NUM_LSB],
+ val, EMUX_FX_FLAG_SET);
+ return;
+ }
+
+ if (port->chset.midi_mode == SNDRV_MIDI_MODE_GS &&
+ chan->control[MIDI_CTL_NONREG_PARM_NUM_MSB] == 1) {
+ int val;
+ /* GS specific NRPNs */
+ /* only MSB is valid */
+ val = chan->control[MIDI_CTL_MSB_DATA_ENTRY];
+ send_converted_effect
+ (gs_effects, ARRAY_SIZE(gs_effects),
+ port, chan, chan->control[MIDI_CTL_NONREG_PARM_NUM_LSB],
+ val, EMUX_FX_FLAG_ADD);
+ return;
+ }
+}
+
+
+/*
+ * XG control effects; still experimental
+ */
+
+/* cutoff: quarter semitone step, max=255 */
+static int xg_cutoff(int val)
+{
+ return (val - 64) * xg_sense[FX_CUTOFF] / 64;
+}
+
+/* resonance: 0(open) to 15(most nasal) */
+static int xg_filterQ(int val)
+{
+ return (val - 64) * xg_sense[FX_RESONANCE] / 64;
+}
+
+/* attack: */
+static int xg_attack(int val)
+{
+ return -(val - 64) * xg_sense[FX_ATTACK] / 64;
+}
+
+/* release: */
+static int xg_release(int val)
+{
+ return -(val - 64) * xg_sense[FX_RELEASE] / 64;
+}
+
+static struct nrpn_conv_table xg_effects[] =
+{
+ {71, EMUX_FX_CUTOFF, xg_cutoff},
+ {74, EMUX_FX_FILTERQ, xg_filterQ},
+ {72, EMUX_FX_ENV2_RELEASE, xg_release},
+ {73, EMUX_FX_ENV2_ATTACK, xg_attack},
+};
+
+int
+snd_emux_xg_control(struct snd_emux_port *port, struct snd_midi_channel *chan,
+ int param)
+{
+ return send_converted_effect(xg_effects, ARRAY_SIZE(xg_effects),
+ port, chan, param,
+ chan->control[param],
+ EMUX_FX_FLAG_ADD);
+}
+
+/*
+ * receive sysex
+ */
+void
+snd_emux_sysex(void *p, unsigned char *buf, int len, int parsed,
+ struct snd_midi_channel_set *chset)
+{
+ struct snd_emux_port *port;
+ struct snd_emux *emu;
+
+ port = p;
+ if (snd_BUG_ON(!port || !chset))
+ return;
+ emu = port->emu;
+
+ switch (parsed) {
+ case SNDRV_MIDI_SYSEX_GS_MASTER_VOLUME:
+ snd_emux_update_port(port, SNDRV_EMUX_UPDATE_VOLUME);
+ break;
+ default:
+ if (emu->ops.sysex)
+ emu->ops.sysex(emu, buf, len, parsed, chset);
+ break;
+ }
+}
+
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
new file mode 100644
index 0000000..5c47b6c
--- /dev/null
+++ b/sound/synth/emux/emux_oss.c
@@ -0,0 +1,516 @@
+/*
+ * Interface for OSS sequencer emulation
+ *
+ * Copyright (C) 1999 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ * Changes
+ * 19990227 Steve Ratcliffe Made separate file and merged in latest
+ * midi emulation.
+ */
+
+
+#ifdef CONFIG_SND_SEQUENCER_OSS
+
+#include <asm/uaccess.h>
+#include <sound/core.h>
+#include "emux_voice.h"
+#include <sound/asoundef.h>
+
+static int snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure);
+static int snd_emux_close_seq_oss(struct snd_seq_oss_arg *arg);
+static int snd_emux_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd,
+ unsigned long ioarg);
+static int snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format,
+ const char __user *buf, int offs, int count);
+static int snd_emux_reset_seq_oss(struct snd_seq_oss_arg *arg);
+static int snd_emux_event_oss_input(struct snd_seq_event *ev, int direct,
+ void *private, int atomic, int hop);
+static void reset_port_mode(struct snd_emux_port *port, int midi_mode);
+static void emuspec_control(struct snd_emux *emu, struct snd_emux_port *port,
+ int cmd, unsigned char *event, int atomic, int hop);
+static void gusspec_control(struct snd_emux *emu, struct snd_emux_port *port,
+ int cmd, unsigned char *event, int atomic, int hop);
+static void fake_event(struct snd_emux *emu, struct snd_emux_port *port,
+ int ch, int param, int val, int atomic, int hop);
+
+/* operators */
+static struct snd_seq_oss_callback oss_callback = {
+ .owner = THIS_MODULE,
+ .open = snd_emux_open_seq_oss,
+ .close = snd_emux_close_seq_oss,
+ .ioctl = snd_emux_ioctl_seq_oss,
+ .load_patch = snd_emux_load_patch_seq_oss,
+ .reset = snd_emux_reset_seq_oss,
+};
+
+
+/*
+ * register OSS synth
+ */
+
+void
+snd_emux_init_seq_oss(struct snd_emux *emu)
+{
+ struct snd_seq_oss_reg *arg;
+ struct snd_seq_device *dev;
+
+ if (snd_seq_device_new(emu->card, 0, SNDRV_SEQ_DEV_ID_OSS,
+ sizeof(struct snd_seq_oss_reg), &dev) < 0)
+ return;
+
+ emu->oss_synth = dev;
+ strcpy(dev->name, emu->name);
+ arg = SNDRV_SEQ_DEVICE_ARGPTR(dev);
+ arg->type = SYNTH_TYPE_SAMPLE;
+ arg->subtype = SAMPLE_TYPE_AWE32;
+ arg->nvoices = emu->max_voices;
+ arg->oper = oss_callback;
+ arg->private_data = emu;
+
+ /* register to OSS synth table */
+ snd_device_register(emu->card, dev);
+}
+
+
+/*
+ * unregister
+ */
+void
+snd_emux_detach_seq_oss(struct snd_emux *emu)
+{
+ if (emu->oss_synth) {
+ snd_device_free(emu->card, emu->oss_synth);
+ emu->oss_synth = NULL;
+ }
+}
+
+
+/* use port number as a unique soundfont client number */
+#define SF_CLIENT_NO(p) ((p) + 0x1000)
+
+/*
+ * open port for OSS sequencer
+ */
+static int
+snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure)
+{
+ struct snd_emux *emu;
+ struct snd_emux_port *p;
+ struct snd_seq_port_callback callback;
+ char tmpname[64];
+
+ emu = closure;
+ if (snd_BUG_ON(!arg || !emu))
+ return -ENXIO;
+
+ mutex_lock(&emu->register_mutex);
+
+ if (!snd_emux_inc_count(emu)) {
+ mutex_unlock(&emu->register_mutex);
+ return -EFAULT;
+ }
+
+ memset(&callback, 0, sizeof(callback));
+ callback.owner = THIS_MODULE;
+ callback.event_input = snd_emux_event_oss_input;
+
+ sprintf(tmpname, "%s OSS Port", emu->name);
+ p = snd_emux_create_port(emu, tmpname, 32,
+ 1, &callback);
+ if (p == NULL) {
+ snd_printk("can't create port\n");
+ snd_emux_dec_count(emu);
+ mutex_unlock(&emu->register_mutex);
+ return -ENOMEM;
+ }
+
+ /* fill the argument data */
+ arg->private_data = p;
+ arg->addr.client = p->chset.client;
+ arg->addr.port = p->chset.port;
+ p->oss_arg = arg;
+
+ reset_port_mode(p, arg->seq_mode);
+
+ snd_emux_reset_port(p);
+
+ mutex_unlock(&emu->register_mutex);
+ return 0;
+}
+
+
+#define DEFAULT_DRUM_FLAGS ((1<<9) | (1<<25))
+
+/*
+ * reset port mode
+ */
+static void
+reset_port_mode(struct snd_emux_port *port, int midi_mode)
+{
+ if (midi_mode) {
+ port->port_mode = SNDRV_EMUX_PORT_MODE_OSS_MIDI;
+ port->drum_flags = DEFAULT_DRUM_FLAGS;
+ port->volume_atten = 0;
+ port->oss_arg->event_passing = SNDRV_SEQ_OSS_PROCESS_KEYPRESS;
+ } else {
+ port->port_mode = SNDRV_EMUX_PORT_MODE_OSS_SYNTH;
+ port->drum_flags = 0;
+ port->volume_atten = 32;
+ port->oss_arg->event_passing = SNDRV_SEQ_OSS_PROCESS_EVENTS;
+ }
+}
+
+
+/*
+ * close port
+ */
+static int
+snd_emux_close_seq_oss(struct snd_seq_oss_arg *arg)
+{
+ struct snd_emux *emu;
+ struct snd_emux_port *p;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ p = arg->private_data;
+ if (snd_BUG_ON(!p))
+ return -ENXIO;
+
+ emu = p->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+
+ mutex_lock(&emu->register_mutex);
+ snd_emux_sounds_off_all(p);
+ snd_soundfont_close_check(emu->sflist, SF_CLIENT_NO(p->chset.port));
+ snd_seq_event_port_detach(p->chset.client, p->chset.port);
+ snd_emux_dec_count(emu);
+
+ mutex_unlock(&emu->register_mutex);
+ return 0;
+}
+
+
+/*
+ * load patch
+ */
+static int
+snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format,
+ const char __user *buf, int offs, int count)
+{
+ struct snd_emux *emu;
+ struct snd_emux_port *p;
+ int rc;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ p = arg->private_data;
+ if (snd_BUG_ON(!p))
+ return -ENXIO;
+
+ emu = p->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+
+ if (format == GUS_PATCH)
+ rc = snd_soundfont_load_guspatch(emu->sflist, buf, count,
+ SF_CLIENT_NO(p->chset.port));
+ else if (format == SNDRV_OSS_SOUNDFONT_PATCH) {
+ struct soundfont_patch_info patch;
+ if (count < (int)sizeof(patch))
+ rc = -EINVAL;
+ if (copy_from_user(&patch, buf, sizeof(patch)))
+ rc = -EFAULT;
+ if (patch.type >= SNDRV_SFNT_LOAD_INFO &&
+ patch.type <= SNDRV_SFNT_PROBE_DATA)
+ rc = snd_soundfont_load(emu->sflist, buf, count, SF_CLIENT_NO(p->chset.port));
+ else {
+ if (emu->ops.load_fx)
+ rc = emu->ops.load_fx(emu, patch.type, patch.optarg, buf, count);
+ else
+ rc = -EINVAL;
+ }
+ } else
+ rc = 0;
+ return rc;
+}
+
+
+/*
+ * ioctl
+ */
+static int
+snd_emux_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, unsigned long ioarg)
+{
+ struct snd_emux_port *p;
+ struct snd_emux *emu;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ p = arg->private_data;
+ if (snd_BUG_ON(!p))
+ return -ENXIO;
+
+ emu = p->emu;
+ if (snd_BUG_ON(!emu))
+ return -ENXIO;
+
+ switch (cmd) {
+ case SNDCTL_SEQ_RESETSAMPLES:
+ snd_soundfont_remove_samples(emu->sflist);
+ return 0;
+
+ case SNDCTL_SYNTH_MEMAVL:
+ if (emu->memhdr)
+ return snd_util_mem_avail(emu->memhdr);
+ return 0;
+ }
+
+ return 0;
+}
+
+
+/*
+ * reset device
+ */
+static int
+snd_emux_reset_seq_oss(struct snd_seq_oss_arg *arg)
+{
+ struct snd_emux_port *p;
+
+ if (snd_BUG_ON(!arg))
+ return -ENXIO;
+ p = arg->private_data;
+ if (snd_BUG_ON(!p))
+ return -ENXIO;
+ snd_emux_reset_port(p);
+ return 0;
+}
+
+
+/*
+ * receive raw events: only SEQ_PRIVATE is accepted.
+ */
+static int
+snd_emux_event_oss_input(struct snd_seq_event *ev, int direct, void *private_data,
+ int atomic, int hop)
+{
+ struct snd_emux *emu;
+ struct snd_emux_port *p;
+ unsigned char cmd, *data;
+
+ p = private_data;
+ if (snd_BUG_ON(!p))
+ return -EINVAL;
+ emu = p->emu;
+ if (snd_BUG_ON(!emu))
+ return -EINVAL;
+ if (ev->type != SNDRV_SEQ_EVENT_OSS)
+ return snd_emux_event_input(ev, direct, private_data, atomic, hop);
+
+ data = ev->data.raw8.d;
+ /* only SEQ_PRIVATE is accepted */
+ if (data[0] != 0xfe)
+ return 0;
+ cmd = data[2] & _EMUX_OSS_MODE_VALUE_MASK;
+ if (data[2] & _EMUX_OSS_MODE_FLAG)
+ emuspec_control(emu, p, cmd, data, atomic, hop);
+ else
+ gusspec_control(emu, p, cmd, data, atomic, hop);
+ return 0;
+}
+
+
+/*
+ * OSS/AWE driver specific h/w controls
+ */
+static void
+emuspec_control(struct snd_emux *emu, struct snd_emux_port *port, int cmd,
+ unsigned char *event, int atomic, int hop)
+{
+ int voice;
+ unsigned short p1;
+ short p2;
+ int i;
+ struct snd_midi_channel *chan;
+
+ voice = event[3];
+ if (voice < 0 || voice >= port->chset.max_channels)
+ chan = NULL;
+ else
+ chan = &port->chset.channels[voice];
+
+ p1 = *(unsigned short *) &event[4];
+ p2 = *(short *) &event[6];
+
+ switch (cmd) {
+#if 0 /* don't do this atomically */
+ case _EMUX_OSS_REMOVE_LAST_SAMPLES:
+ snd_soundfont_remove_unlocked(emu->sflist);
+ break;
+#endif
+ case _EMUX_OSS_SEND_EFFECT:
+ if (chan)
+ snd_emux_send_effect_oss(port, chan, p1, p2);
+ break;
+
+ case _EMUX_OSS_TERMINATE_ALL:
+ snd_emux_terminate_all(emu);
+ break;
+
+ case _EMUX_OSS_TERMINATE_CHANNEL:
+ /*snd_emux_mute_channel(emu, chan);*/
+ break;
+ case _EMUX_OSS_RESET_CHANNEL:
+ /*snd_emux_channel_init(chset, chan);*/
+ break;
+
+ case _EMUX_OSS_RELEASE_ALL:
+ fake_event(emu, port, voice, MIDI_CTL_ALL_NOTES_OFF, 0, atomic, hop);
+ break;
+ case _EMUX_OSS_NOTEOFF_ALL:
+ fake_event(emu, port, voice, MIDI_CTL_ALL_SOUNDS_OFF, 0, atomic, hop);
+ break;
+
+ case _EMUX_OSS_INITIAL_VOLUME:
+ if (p2) {
+ port->volume_atten = (short)p1;
+ snd_emux_update_port(port, SNDRV_EMUX_UPDATE_VOLUME);
+ }
+ break;
+
+ case _EMUX_OSS_CHN_PRESSURE:
+ if (chan) {
+ chan->midi_pressure = p1;
+ snd_emux_update_channel(port, chan, SNDRV_EMUX_UPDATE_FMMOD|SNDRV_EMUX_UPDATE_FM2FRQ2);
+ }
+ break;
+
+ case _EMUX_OSS_CHANNEL_MODE:
+ reset_port_mode(port, p1);
+ snd_emux_reset_port(port);
+ break;
+
+ case _EMUX_OSS_DRUM_CHANNELS:
+ port->drum_flags = *(unsigned int*)&event[4];
+ for (i = 0; i < port->chset.max_channels; i++) {
+ chan = &port->chset.channels[i];
+ chan->drum_channel = ((port->drum_flags >> i) & 1) ? 1 : 0;
+ }
+ break;
+
+ case _EMUX_OSS_MISC_MODE:
+ if (p1 < EMUX_MD_END)
+ port->ctrls[p1] = p2;
+ break;
+ case _EMUX_OSS_DEBUG_MODE:
+ break;
+
+ default:
+ if (emu->ops.oss_ioctl)
+ emu->ops.oss_ioctl(emu, cmd, p1, p2);
+ break;
+ }
+}
+
+/*
+ * GUS specific h/w controls
+ */
+
+#include <linux/ultrasound.h>
+
+static void
+gusspec_control(struct snd_emux *emu, struct snd_emux_port *port, int cmd,
+ unsigned char *event, int atomic, int hop)
+{
+ int voice;
+ unsigned short p1;
+ short p2;
+ int plong;
+ struct snd_midi_channel *chan;
+
+ if (port->port_mode != SNDRV_EMUX_PORT_MODE_OSS_SYNTH)
+ return;
+ if (cmd == _GUS_NUMVOICES)
+ return;
+ voice = event[3];
+ if (voice < 0 || voice >= port->chset.max_channels)
+ return;
+
+ chan = &port->chset.channels[voice];
+
+ p1 = *(unsigned short *) &event[4];
+ p2 = *(short *) &event[6];
+ plong = *(int*) &event[4];
+
+ switch (cmd) {
+ case _GUS_VOICESAMPLE:
+ chan->midi_program = p1;
+ return;
+
+ case _GUS_VOICEBALA:
+ /* 0 to 15 --> 0 to 127 */
+ chan->control[MIDI_CTL_MSB_PAN] = (int)p1 << 3;
+ snd_emux_update_channel(port, chan, SNDRV_EMUX_UPDATE_PAN);
+ return;
+
+ case _GUS_VOICEVOL:
+ case _GUS_VOICEVOL2:
+ /* not supported yet */
+ return;
+
+ case _GUS_RAMPRANGE:
+ case _GUS_RAMPRATE:
+ case _GUS_RAMPMODE:
+ case _GUS_RAMPON:
+ case _GUS_RAMPOFF:
+ /* volume ramping not supported */
+ return;
+
+ case _GUS_VOLUME_SCALE:
+ return;
+
+ case _GUS_VOICE_POS:
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ snd_emux_send_effect(port, chan, EMUX_FX_SAMPLE_START,
+ (short)(plong & 0x7fff),
+ EMUX_FX_FLAG_SET);
+ snd_emux_send_effect(port, chan, EMUX_FX_COARSE_SAMPLE_START,
+ (plong >> 15) & 0xffff,
+ EMUX_FX_FLAG_SET);
+#endif
+ return;
+ }
+}
+
+
+/*
+ * send an event to midi emulation
+ */
+static void
+fake_event(struct snd_emux *emu, struct snd_emux_port *port, int ch, int param, int val, int atomic, int hop)
+{
+ struct snd_seq_event ev;
+ memset(&ev, 0, sizeof(ev));
+ ev.type = SNDRV_SEQ_EVENT_CONTROLLER;
+ ev.data.control.channel = ch;
+ ev.data.control.param = param;
+ ev.data.control.value = val;
+ snd_emux_event_input(&ev, 0, port, atomic, hop);
+}
+
+#endif /* CONFIG_SND_SEQUENCER_OSS */
diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c
new file mode 100644
index 0000000..687e6a1
--- /dev/null
+++ b/sound/synth/emux/emux_proc.c
@@ -0,0 +1,133 @@
+/*
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Proc interface for Emu8k/Emu10k1 WaveTable synth
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/wait.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/emux_synth.h>
+#include <sound/info.h>
+#include "emux_voice.h"
+
+#ifdef CONFIG_PROC_FS
+
+static void
+snd_emux_proc_info_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buf)
+{
+ struct snd_emux *emu;
+ int i;
+
+ emu = entry->private_data;
+ mutex_lock(&emu->register_mutex);
+ if (emu->name)
+ snd_iprintf(buf, "Device: %s\n", emu->name);
+ snd_iprintf(buf, "Ports: %d\n", emu->num_ports);
+ snd_iprintf(buf, "Addresses:");
+ for (i = 0; i < emu->num_ports; i++)
+ snd_iprintf(buf, " %d:%d", emu->client, emu->ports[i]);
+ snd_iprintf(buf, "\n");
+ snd_iprintf(buf, "Use Counter: %d\n", emu->used);
+ snd_iprintf(buf, "Max Voices: %d\n", emu->max_voices);
+ snd_iprintf(buf, "Allocated Voices: %d\n", emu->num_voices);
+ if (emu->memhdr) {
+ snd_iprintf(buf, "Memory Size: %d\n", emu->memhdr->size);
+ snd_iprintf(buf, "Memory Available: %d\n", snd_util_mem_avail(emu->memhdr));
+ snd_iprintf(buf, "Allocated Blocks: %d\n", emu->memhdr->nblocks);
+ } else {
+ snd_iprintf(buf, "Memory Size: 0\n");
+ }
+ if (emu->sflist) {
+ mutex_lock(&emu->sflist->presets_mutex);
+ snd_iprintf(buf, "SoundFonts: %d\n", emu->sflist->fonts_size);
+ snd_iprintf(buf, "Instruments: %d\n", emu->sflist->zone_counter);
+ snd_iprintf(buf, "Samples: %d\n", emu->sflist->sample_counter);
+ snd_iprintf(buf, "Locked Instruments: %d\n", emu->sflist->zone_locked);
+ snd_iprintf(buf, "Locked Samples: %d\n", emu->sflist->sample_locked);
+ mutex_unlock(&emu->sflist->presets_mutex);
+ }
+#if 0 /* debug */
+ if (emu->voices[0].state != SNDRV_EMUX_ST_OFF && emu->voices[0].ch >= 0) {
+ struct snd_emux_voice *vp = &emu->voices[0];
+ snd_iprintf(buf, "voice 0: on\n");
+ snd_iprintf(buf, "mod delay=%x, atkhld=%x, dcysus=%x, rel=%x\n",
+ vp->reg.parm.moddelay,
+ vp->reg.parm.modatkhld,
+ vp->reg.parm.moddcysus,
+ vp->reg.parm.modrelease);
+ snd_iprintf(buf, "vol delay=%x, atkhld=%x, dcysus=%x, rel=%x\n",
+ vp->reg.parm.voldelay,
+ vp->reg.parm.volatkhld,
+ vp->reg.parm.voldcysus,
+ vp->reg.parm.volrelease);
+ snd_iprintf(buf, "lfo1 delay=%x, lfo2 delay=%x, pefe=%x\n",
+ vp->reg.parm.lfo1delay,
+ vp->reg.parm.lfo2delay,
+ vp->reg.parm.pefe);
+ snd_iprintf(buf, "fmmod=%x, tremfrq=%x, fm2frq2=%x\n",
+ vp->reg.parm.fmmod,
+ vp->reg.parm.tremfrq,
+ vp->reg.parm.fm2frq2);
+ snd_iprintf(buf, "cutoff=%x, filterQ=%x, chorus=%x, reverb=%x\n",
+ vp->reg.parm.cutoff,
+ vp->reg.parm.filterQ,
+ vp->reg.parm.chorus,
+ vp->reg.parm.reverb);
+ snd_iprintf(buf, "avol=%x, acutoff=%x, apitch=%x\n",
+ vp->avol, vp->acutoff, vp->apitch);
+ snd_iprintf(buf, "apan=%x, aaux=%x, ptarget=%x, vtarget=%x, ftarget=%x\n",
+ vp->apan, vp->aaux,
+ vp->ptarget,
+ vp->vtarget,
+ vp->ftarget);
+ snd_iprintf(buf, "start=%x, end=%x, loopstart=%x, loopend=%x\n",
+ vp->reg.start, vp->reg.end, vp->reg.loopstart, vp->reg.loopend);
+ snd_iprintf(buf, "sample_mode=%x, rate=%x\n", vp->reg.sample_mode, vp->reg.rate_offset);
+ }
+#endif
+ mutex_unlock(&emu->register_mutex);
+}
+
+
+void snd_emux_proc_init(struct snd_emux *emu, struct snd_card *card, int device)
+{
+ struct snd_info_entry *entry;
+ char name[64];
+
+ sprintf(name, "wavetableD%d", device);
+ entry = snd_info_create_card_entry(card, name, card->proc_root);
+ if (entry == NULL)
+ return;
+
+ entry->content = SNDRV_INFO_CONTENT_TEXT;
+ entry->private_data = emu;
+ entry->c.text.read = snd_emux_proc_info_read;
+ if (snd_info_register(entry) < 0)
+ snd_info_free_entry(entry);
+ else
+ emu->proc = entry;
+}
+
+void snd_emux_proc_free(struct snd_emux *emu)
+{
+ snd_info_free_entry(emu->proc);
+ emu->proc = NULL;
+}
+
+#endif /* CONFIG_PROC_FS */
diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c
new file mode 100644
index 0000000..335aa2c
--- /dev/null
+++ b/sound/synth/emux/emux_seq.c
@@ -0,0 +1,403 @@
+/*
+ * Midi Sequencer interface routines.
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include "emux_voice.h"
+#include <linux/slab.h>
+
+
+/* Prototypes for static functions */
+static void free_port(void *private);
+static void snd_emux_init_port(struct snd_emux_port *p);
+static int snd_emux_use(void *private_data, struct snd_seq_port_subscribe *info);
+static int snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *info);
+
+/*
+ * MIDI emulation operators
+ */
+static struct snd_midi_op emux_ops = {
+ snd_emux_note_on,
+ snd_emux_note_off,
+ snd_emux_key_press,
+ snd_emux_terminate_note,
+ snd_emux_control,
+ snd_emux_nrpn,
+ snd_emux_sysex,
+};
+
+
+/*
+ * number of MIDI channels
+ */
+#define MIDI_CHANNELS 16
+
+/*
+ * type flags for MIDI sequencer port
+ */
+#define DEFAULT_MIDI_TYPE (SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |\
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM |\
+ SNDRV_SEQ_PORT_TYPE_MIDI_GS |\
+ SNDRV_SEQ_PORT_TYPE_MIDI_XG |\
+ SNDRV_SEQ_PORT_TYPE_HARDWARE |\
+ SNDRV_SEQ_PORT_TYPE_SYNTHESIZER)
+
+/*
+ * Initialise the EMUX Synth by creating a client and registering
+ * a series of ports.
+ * Each of the ports will contain the 16 midi channels. Applications
+ * can connect to these ports to play midi data.
+ */
+int
+snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index)
+{
+ int i;
+ struct snd_seq_port_callback pinfo;
+ char tmpname[64];
+
+ emu->client = snd_seq_create_kernel_client(card, index,
+ "%s WaveTable", emu->name);
+ if (emu->client < 0) {
+ snd_printk("can't create client\n");
+ return -ENODEV;
+ }
+
+ if (emu->num_ports < 0) {
+ snd_printk("seqports must be greater than zero\n");
+ emu->num_ports = 1;
+ } else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) {
+ snd_printk("too many ports."
+ "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS);
+ emu->num_ports = SNDRV_EMUX_MAX_PORTS;
+ }
+
+ memset(&pinfo, 0, sizeof(pinfo));
+ pinfo.owner = THIS_MODULE;
+ pinfo.use = snd_emux_use;
+ pinfo.unuse = snd_emux_unuse;
+ pinfo.event_input = snd_emux_event_input;
+
+ for (i = 0; i < emu->num_ports; i++) {
+ struct snd_emux_port *p;
+
+ sprintf(tmpname, "%s Port %d", emu->name, i);
+ p = snd_emux_create_port(emu, tmpname, MIDI_CHANNELS,
+ 0, &pinfo);
+ if (p == NULL) {
+ snd_printk("can't create port\n");
+ return -ENOMEM;
+ }
+
+ p->port_mode = SNDRV_EMUX_PORT_MODE_MIDI;
+ snd_emux_init_port(p);
+ emu->ports[i] = p->chset.port;
+ emu->portptrs[i] = p;
+ }
+
+ return 0;
+}
+
+
+/*
+ * Detach from the ports that were set up for this synthesizer and
+ * destroy the kernel client.
+ */
+void
+snd_emux_detach_seq(struct snd_emux *emu)
+{
+ if (emu->voices)
+ snd_emux_terminate_all(emu);
+
+ mutex_lock(&emu->register_mutex);
+ if (emu->client >= 0) {
+ snd_seq_delete_kernel_client(emu->client);
+ emu->client = -1;
+ }
+ mutex_unlock(&emu->register_mutex);
+}
+
+
+/*
+ * create a sequencer port and channel_set
+ */
+
+struct snd_emux_port *
+snd_emux_create_port(struct snd_emux *emu, char *name,
+ int max_channels, int oss_port,
+ struct snd_seq_port_callback *callback)
+{
+ struct snd_emux_port *p;
+ int i, type, cap;
+
+ /* Allocate structures for this channel */
+ if ((p = kzalloc(sizeof(*p), GFP_KERNEL)) == NULL) {
+ snd_printk("no memory\n");
+ return NULL;
+ }
+ p->chset.channels = kcalloc(max_channels, sizeof(struct snd_midi_channel), GFP_KERNEL);
+ if (p->chset.channels == NULL) {
+ snd_printk("no memory\n");
+ kfree(p);
+ return NULL;
+ }
+ for (i = 0; i < max_channels; i++)
+ p->chset.channels[i].number = i;
+ p->chset.private_data = p;
+ p->chset.max_channels = max_channels;
+ p->emu = emu;
+ p->chset.client = emu->client;
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ snd_emux_create_effect(p);
+#endif
+ callback->private_free = free_port;
+ callback->private_data = p;
+
+ cap = SNDRV_SEQ_PORT_CAP_WRITE;
+ if (oss_port) {
+ type = SNDRV_SEQ_PORT_TYPE_SPECIFIC;
+ } else {
+ type = DEFAULT_MIDI_TYPE;
+ cap |= SNDRV_SEQ_PORT_CAP_SUBS_WRITE;
+ }
+
+ p->chset.port = snd_seq_event_port_attach(emu->client, callback,
+ cap, type, max_channels,
+ emu->max_voices, name);
+
+ return p;
+}
+
+
+/*
+ * release memory block for port
+ */
+static void
+free_port(void *private_data)
+{
+ struct snd_emux_port *p;
+
+ p = private_data;
+ if (p) {
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ snd_emux_delete_effect(p);
+#endif
+ kfree(p->chset.channels);
+ kfree(p);
+ }
+}
+
+
+#define DEFAULT_DRUM_FLAGS (1<<9)
+
+/*
+ * initialize the port specific parameters
+ */
+static void
+snd_emux_init_port(struct snd_emux_port *p)
+{
+ p->drum_flags = DEFAULT_DRUM_FLAGS;
+ p->volume_atten = 0;
+
+ snd_emux_reset_port(p);
+}
+
+
+/*
+ * reset port
+ */
+void
+snd_emux_reset_port(struct snd_emux_port *port)
+{
+ int i;
+
+ /* stop all sounds */
+ snd_emux_sounds_off_all(port);
+
+ snd_midi_channel_set_clear(&port->chset);
+
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ snd_emux_clear_effect(port);
+#endif
+
+ /* set port specific control parameters */
+ port->ctrls[EMUX_MD_DEF_BANK] = 0;
+ port->ctrls[EMUX_MD_DEF_DRUM] = 0;
+ port->ctrls[EMUX_MD_REALTIME_PAN] = 1;
+
+ for (i = 0; i < port->chset.max_channels; i++) {
+ struct snd_midi_channel *chan = port->chset.channels + i;
+ chan->drum_channel = ((port->drum_flags >> i) & 1) ? 1 : 0;
+ }
+}
+
+
+/*
+ * input sequencer event
+ */
+int
+snd_emux_event_input(struct snd_seq_event *ev, int direct, void *private_data,
+ int atomic, int hop)
+{
+ struct snd_emux_port *port;
+
+ port = private_data;
+ if (snd_BUG_ON(!port || !ev))
+ return -EINVAL;
+
+ snd_midi_process_event(&emux_ops, ev, &port->chset);
+
+ return 0;
+}
+
+
+/*
+ * increment usage count
+ */
+int
+snd_emux_inc_count(struct snd_emux *emu)
+{
+ emu->used++;
+ if (!try_module_get(emu->ops.owner))
+ goto __error;
+ if (!try_module_get(emu->card->module)) {
+ module_put(emu->ops.owner);
+ __error:
+ emu->used--;
+ return 0;
+ }
+ return 1;
+}
+
+
+/*
+ * decrease usage count
+ */
+void
+snd_emux_dec_count(struct snd_emux *emu)
+{
+ module_put(emu->card->module);
+ emu->used--;
+ if (emu->used <= 0)
+ snd_emux_terminate_all(emu);
+ module_put(emu->ops.owner);
+}
+
+
+/*
+ * Routine that is called upon a first use of a particular port
+ */
+static int
+snd_emux_use(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ struct snd_emux_port *p;
+ struct snd_emux *emu;
+
+ p = private_data;
+ if (snd_BUG_ON(!p))
+ return -EINVAL;
+ emu = p->emu;
+ if (snd_BUG_ON(!emu))
+ return -EINVAL;
+
+ mutex_lock(&emu->register_mutex);
+ snd_emux_init_port(p);
+ snd_emux_inc_count(emu);
+ mutex_unlock(&emu->register_mutex);
+ return 0;
+}
+
+/*
+ * Routine that is called upon the last unuse() of a particular port.
+ */
+static int
+snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *info)
+{
+ struct snd_emux_port *p;
+ struct snd_emux *emu;
+
+ p = private_data;
+ if (snd_BUG_ON(!p))
+ return -EINVAL;
+ emu = p->emu;
+ if (snd_BUG_ON(!emu))
+ return -EINVAL;
+
+ mutex_lock(&emu->register_mutex);
+ snd_emux_sounds_off_all(p);
+ snd_emux_dec_count(emu);
+ mutex_unlock(&emu->register_mutex);
+ return 0;
+}
+
+
+/*
+ * attach virtual rawmidi devices
+ */
+int snd_emux_init_virmidi(struct snd_emux *emu, struct snd_card *card)
+{
+ int i;
+
+ emu->vmidi = NULL;
+ if (emu->midi_ports <= 0)
+ return 0;
+
+ emu->vmidi = kcalloc(emu->midi_ports, sizeof(struct snd_rawmidi *), GFP_KERNEL);
+ if (emu->vmidi == NULL)
+ return -ENOMEM;
+
+ for (i = 0; i < emu->midi_ports; i++) {
+ struct snd_rawmidi *rmidi;
+ struct snd_virmidi_dev *rdev;
+ if (snd_virmidi_new(card, emu->midi_devidx + i, &rmidi) < 0)
+ goto __error;
+ rdev = rmidi->private_data;
+ sprintf(rmidi->name, "%s Synth MIDI", emu->name);
+ rdev->seq_mode = SNDRV_VIRMIDI_SEQ_ATTACH;
+ rdev->client = emu->client;
+ rdev->port = emu->ports[i];
+ if (snd_device_register(card, rmidi) < 0) {
+ snd_device_free(card, rmidi);
+ goto __error;
+ }
+ emu->vmidi[i] = rmidi;
+ //snd_printk("virmidi %d ok\n", i);
+ }
+ return 0;
+
+__error:
+ //snd_printk("error init..\n");
+ snd_emux_delete_virmidi(emu);
+ return -ENOMEM;
+}
+
+int snd_emux_delete_virmidi(struct snd_emux *emu)
+{
+ int i;
+
+ if (emu->vmidi == NULL)
+ return 0;
+
+ for (i = 0; i < emu->midi_ports; i++) {
+ if (emu->vmidi[i])
+ snd_device_free(emu->card, emu->vmidi[i]);
+ }
+ kfree(emu->vmidi);
+ emu->vmidi = NULL;
+ return 0;
+}
diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c
new file mode 100644
index 0000000..2cc6f6f
--- /dev/null
+++ b/sound/synth/emux/emux_synth.c
@@ -0,0 +1,981 @@
+/*
+ * Midi synth routines for the Emu8k/Emu10k1
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Contains code based on awe_wave.c by Takashi Iwai
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include "emux_voice.h"
+#include <sound/asoundef.h>
+
+/*
+ * Prototypes
+ */
+
+/*
+ * Ensure a value is between two points
+ * macro evaluates its args more than once, so changed to upper-case.
+ */
+#define LIMITVALUE(x, a, b) do { if ((x) < (a)) (x) = (a); else if ((x) > (b)) (x) = (b); } while (0)
+#define LIMITMAX(x, a) do {if ((x) > (a)) (x) = (a); } while (0)
+
+static int get_zone(struct snd_emux *emu, struct snd_emux_port *port,
+ int *notep, int vel, struct snd_midi_channel *chan,
+ struct snd_sf_zone **table);
+static int get_bank(struct snd_emux_port *port, struct snd_midi_channel *chan);
+static void terminate_note1(struct snd_emux *emu, int note,
+ struct snd_midi_channel *chan, int free);
+static void exclusive_note_off(struct snd_emux *emu, struct snd_emux_port *port,
+ int exclass);
+static void terminate_voice(struct snd_emux *emu, struct snd_emux_voice *vp, int free);
+static void update_voice(struct snd_emux *emu, struct snd_emux_voice *vp, int update);
+static void setup_voice(struct snd_emux_voice *vp);
+static int calc_pan(struct snd_emux_voice *vp);
+static int calc_volume(struct snd_emux_voice *vp);
+static int calc_pitch(struct snd_emux_voice *vp);
+
+
+/*
+ * Start a note.
+ */
+void
+snd_emux_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
+{
+ struct snd_emux *emu;
+ int i, key, nvoices;
+ struct snd_emux_voice *vp;
+ struct snd_sf_zone *table[SNDRV_EMUX_MAX_MULTI_VOICES];
+ unsigned long flags;
+ struct snd_emux_port *port;
+
+ port = p;
+ if (snd_BUG_ON(!port || !chan))
+ return;
+
+ emu = port->emu;
+ if (snd_BUG_ON(!emu || !emu->ops.get_voice || !emu->ops.trigger))
+ return;
+
+ key = note; /* remember the original note */
+ nvoices = get_zone(emu, port, &note, vel, chan, table);
+ if (! nvoices)
+ return;
+
+ /* exclusive note off */
+ for (i = 0; i < nvoices; i++) {
+ struct snd_sf_zone *zp = table[i];
+ if (zp && zp->v.exclusiveClass)
+ exclusive_note_off(emu, port, zp->v.exclusiveClass);
+ }
+
+#if 0 // seems not necessary
+ /* Turn off the same note on the same channel. */
+ terminate_note1(emu, key, chan, 0);
+#endif
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < nvoices; i++) {
+
+ /* set up each voice parameter */
+ /* at this stage, we don't trigger the voice yet. */
+
+ if (table[i] == NULL)
+ continue;
+
+ vp = emu->ops.get_voice(emu, port);
+ if (vp == NULL || vp->ch < 0)
+ continue;
+ if (STATE_IS_PLAYING(vp->state))
+ emu->ops.terminate(vp);
+
+ vp->time = emu->use_time++;
+ vp->chan = chan;
+ vp->port = port;
+ vp->key = key;
+ vp->note = note;
+ vp->velocity = vel;
+ vp->zone = table[i];
+ if (vp->zone->sample)
+ vp->block = vp->zone->sample->block;
+ else
+ vp->block = NULL;
+
+ setup_voice(vp);
+
+ vp->state = SNDRV_EMUX_ST_STANDBY;
+ if (emu->ops.prepare) {
+ vp->state = SNDRV_EMUX_ST_OFF;
+ if (emu->ops.prepare(vp) >= 0)
+ vp->state = SNDRV_EMUX_ST_STANDBY;
+ }
+ }
+
+ /* start envelope now */
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ if (vp->state == SNDRV_EMUX_ST_STANDBY &&
+ vp->chan == chan) {
+ emu->ops.trigger(vp);
+ vp->state = SNDRV_EMUX_ST_ON;
+ vp->ontime = jiffies; /* remember the trigger timing */
+ }
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ if (port->port_mode == SNDRV_EMUX_PORT_MODE_OSS_SYNTH) {
+ /* clear voice position for the next note on this channel */
+ struct snd_emux_effect_table *fx = chan->private;
+ if (fx) {
+ fx->flag[EMUX_FX_SAMPLE_START] = 0;
+ fx->flag[EMUX_FX_COARSE_SAMPLE_START] = 0;
+ }
+ }
+#endif
+}
+
+/*
+ * Release a note in response to a midi note off.
+ */
+void
+snd_emux_note_off(void *p, int note, int vel, struct snd_midi_channel *chan)
+{
+ int ch;
+ struct snd_emux *emu;
+ struct snd_emux_voice *vp;
+ unsigned long flags;
+ struct snd_emux_port *port;
+
+ port = p;
+ if (snd_BUG_ON(!port || !chan))
+ return;
+
+ emu = port->emu;
+ if (snd_BUG_ON(!emu || !emu->ops.release))
+ return;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (ch = 0; ch < emu->max_voices; ch++) {
+ vp = &emu->voices[ch];
+ if (STATE_IS_PLAYING(vp->state) &&
+ vp->chan == chan && vp->key == note) {
+ vp->state = SNDRV_EMUX_ST_RELEASED;
+ if (vp->ontime == jiffies) {
+ /* if note-off is sent too shortly after
+ * note-on, emuX engine cannot produce the sound
+ * correctly. so we'll release this note
+ * a bit later via timer callback.
+ */
+ vp->state = SNDRV_EMUX_ST_PENDING;
+ if (! emu->timer_active) {
+ emu->tlist.expires = jiffies + 1;
+ add_timer(&emu->tlist);
+ emu->timer_active = 1;
+ }
+ } else
+ /* ok now release the note */
+ emu->ops.release(vp);
+ }
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+/*
+ * timer callback
+ *
+ * release the pending note-offs
+ */
+void snd_emux_timer_callback(unsigned long data)
+{
+ struct snd_emux *emu = (struct snd_emux *) data;
+ struct snd_emux_voice *vp;
+ unsigned long flags;
+ int ch, do_again = 0;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (ch = 0; ch < emu->max_voices; ch++) {
+ vp = &emu->voices[ch];
+ if (vp->state == SNDRV_EMUX_ST_PENDING) {
+ if (vp->ontime == jiffies)
+ do_again++; /* release this at the next interrupt */
+ else {
+ emu->ops.release(vp);
+ vp->state = SNDRV_EMUX_ST_RELEASED;
+ }
+ }
+ }
+ if (do_again) {
+ emu->tlist.expires = jiffies + 1;
+ add_timer(&emu->tlist);
+ emu->timer_active = 1;
+ } else
+ emu->timer_active = 0;
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+/*
+ * key pressure change
+ */
+void
+snd_emux_key_press(void *p, int note, int vel, struct snd_midi_channel *chan)
+{
+ int ch;
+ struct snd_emux *emu;
+ struct snd_emux_voice *vp;
+ unsigned long flags;
+ struct snd_emux_port *port;
+
+ port = p;
+ if (snd_BUG_ON(!port || !chan))
+ return;
+
+ emu = port->emu;
+ if (snd_BUG_ON(!emu || !emu->ops.update))
+ return;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (ch = 0; ch < emu->max_voices; ch++) {
+ vp = &emu->voices[ch];
+ if (vp->state == SNDRV_EMUX_ST_ON &&
+ vp->chan == chan && vp->key == note) {
+ vp->velocity = vel;
+ update_voice(emu, vp, SNDRV_EMUX_UPDATE_VOLUME);
+ }
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+
+/*
+ * Modulate the voices which belong to the channel
+ */
+void
+snd_emux_update_channel(struct snd_emux_port *port, struct snd_midi_channel *chan, int update)
+{
+ struct snd_emux *emu;
+ struct snd_emux_voice *vp;
+ int i;
+ unsigned long flags;
+
+ if (! update)
+ return;
+
+ emu = port->emu;
+ if (snd_BUG_ON(!emu || !emu->ops.update))
+ return;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ if (vp->chan == chan)
+ update_voice(emu, vp, update);
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+/*
+ * Modulate all the voices which belong to the port.
+ */
+void
+snd_emux_update_port(struct snd_emux_port *port, int update)
+{
+ struct snd_emux *emu;
+ struct snd_emux_voice *vp;
+ int i;
+ unsigned long flags;
+
+ if (! update)
+ return;
+
+ emu = port->emu;
+ if (snd_BUG_ON(!emu || !emu->ops.update))
+ return;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ if (vp->port == port)
+ update_voice(emu, vp, update);
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+
+/*
+ * Deal with a controller type event. This includes all types of
+ * control events, not just the midi controllers
+ */
+void
+snd_emux_control(void *p, int type, struct snd_midi_channel *chan)
+{
+ struct snd_emux_port *port;
+
+ port = p;
+ if (snd_BUG_ON(!port || !chan))
+ return;
+
+ switch (type) {
+ case MIDI_CTL_MSB_MAIN_VOLUME:
+ case MIDI_CTL_MSB_EXPRESSION:
+ snd_emux_update_channel(port, chan, SNDRV_EMUX_UPDATE_VOLUME);
+ break;
+
+ case MIDI_CTL_MSB_PAN:
+ snd_emux_update_channel(port, chan, SNDRV_EMUX_UPDATE_PAN);
+ break;
+
+ case MIDI_CTL_SOFT_PEDAL:
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ /* FIXME: this is an emulation */
+ if (chan->control[type] >= 64)
+ snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, -160,
+ EMUX_FX_FLAG_ADD);
+ else
+ snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, 0,
+ EMUX_FX_FLAG_OFF);
+#endif
+ break;
+
+ case MIDI_CTL_PITCHBEND:
+ snd_emux_update_channel(port, chan, SNDRV_EMUX_UPDATE_PITCH);
+ break;
+
+ case MIDI_CTL_MSB_MODWHEEL:
+ case MIDI_CTL_CHAN_PRESSURE:
+ snd_emux_update_channel(port, chan,
+ SNDRV_EMUX_UPDATE_FMMOD |
+ SNDRV_EMUX_UPDATE_FM2FRQ2);
+ break;
+
+ }
+
+ if (port->chset.midi_mode == SNDRV_MIDI_MODE_XG) {
+ snd_emux_xg_control(port, chan, type);
+ }
+}
+
+
+/*
+ * terminate note - if free flag is true, free the terminated voice
+ */
+static void
+terminate_note1(struct snd_emux *emu, int note, struct snd_midi_channel *chan, int free)
+{
+ int i;
+ struct snd_emux_voice *vp;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ if (STATE_IS_PLAYING(vp->state) && vp->chan == chan &&
+ vp->key == note)
+ terminate_voice(emu, vp, free);
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+
+/*
+ * terminate note - exported for midi emulation
+ */
+void
+snd_emux_terminate_note(void *p, int note, struct snd_midi_channel *chan)
+{
+ struct snd_emux *emu;
+ struct snd_emux_port *port;
+
+ port = p;
+ if (snd_BUG_ON(!port || !chan))
+ return;
+
+ emu = port->emu;
+ if (snd_BUG_ON(!emu || !emu->ops.terminate))
+ return;
+
+ terminate_note1(emu, note, chan, 1);
+}
+
+
+/*
+ * Terminate all the notes
+ */
+void
+snd_emux_terminate_all(struct snd_emux *emu)
+{
+ int i;
+ struct snd_emux_voice *vp;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ if (STATE_IS_PLAYING(vp->state))
+ terminate_voice(emu, vp, 0);
+ if (vp->state == SNDRV_EMUX_ST_OFF) {
+ if (emu->ops.free_voice)
+ emu->ops.free_voice(vp);
+ if (emu->ops.reset)
+ emu->ops.reset(emu, i);
+ }
+ vp->time = 0;
+ }
+ /* initialize allocation time */
+ emu->use_time = 0;
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+EXPORT_SYMBOL(snd_emux_terminate_all);
+
+/*
+ * Terminate all voices associated with the given port
+ */
+void
+snd_emux_sounds_off_all(struct snd_emux_port *port)
+{
+ int i;
+ struct snd_emux *emu;
+ struct snd_emux_voice *vp;
+ unsigned long flags;
+
+ if (snd_BUG_ON(!port))
+ return;
+ emu = port->emu;
+ if (snd_BUG_ON(!emu || !emu->ops.terminate))
+ return;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ if (STATE_IS_PLAYING(vp->state) &&
+ vp->port == port)
+ terminate_voice(emu, vp, 0);
+ if (vp->state == SNDRV_EMUX_ST_OFF) {
+ if (emu->ops.free_voice)
+ emu->ops.free_voice(vp);
+ if (emu->ops.reset)
+ emu->ops.reset(emu, i);
+ }
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+
+/*
+ * Terminate all voices that have the same exclusive class. This
+ * is mainly for drums.
+ */
+static void
+exclusive_note_off(struct snd_emux *emu, struct snd_emux_port *port, int exclass)
+{
+ struct snd_emux_voice *vp;
+ int i;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ if (STATE_IS_PLAYING(vp->state) && vp->port == port &&
+ vp->reg.exclusiveClass == exclass) {
+ terminate_voice(emu, vp, 0);
+ }
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+/*
+ * terminate a voice
+ * if free flag is true, call free_voice after termination
+ */
+static void
+terminate_voice(struct snd_emux *emu, struct snd_emux_voice *vp, int free)
+{
+ emu->ops.terminate(vp);
+ vp->time = emu->use_time++;
+ vp->chan = NULL;
+ vp->port = NULL;
+ vp->zone = NULL;
+ vp->block = NULL;
+ vp->state = SNDRV_EMUX_ST_OFF;
+ if (free && emu->ops.free_voice)
+ emu->ops.free_voice(vp);
+}
+
+
+/*
+ * Modulate the voice
+ */
+static void
+update_voice(struct snd_emux *emu, struct snd_emux_voice *vp, int update)
+{
+ if (!STATE_IS_PLAYING(vp->state))
+ return;
+
+ if (vp->chan == NULL || vp->port == NULL)
+ return;
+ if (update & SNDRV_EMUX_UPDATE_VOLUME)
+ calc_volume(vp);
+ if (update & SNDRV_EMUX_UPDATE_PITCH)
+ calc_pitch(vp);
+ if (update & SNDRV_EMUX_UPDATE_PAN) {
+ if (! calc_pan(vp) && (update == SNDRV_EMUX_UPDATE_PAN))
+ return;
+ }
+ emu->ops.update(vp, update);
+}
+
+
+#if 0 // not used
+/* table for volume target calculation */
+static unsigned short voltarget[16] = {
+ 0xEAC0, 0xE0C8, 0xD740, 0xCE20, 0xC560, 0xBD08, 0xB500, 0xAD58,
+ 0xA5F8, 0x9EF0, 0x9830, 0x91C0, 0x8B90, 0x85A8, 0x8000, 0x7A90
+};
+#endif
+
+#define LO_BYTE(v) ((v) & 0xff)
+#define HI_BYTE(v) (((v) >> 8) & 0xff)
+
+/*
+ * Sets up the voice structure by calculating some values that
+ * will be needed later.
+ */
+static void
+setup_voice(struct snd_emux_voice *vp)
+{
+ struct soundfont_voice_parm *parm;
+ int pitch;
+
+ /* copy the original register values */
+ vp->reg = vp->zone->v;
+
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ snd_emux_setup_effect(vp);
+#endif
+
+ /* reset status */
+ vp->apan = -1;
+ vp->avol = -1;
+ vp->apitch = -1;
+
+ calc_volume(vp);
+ calc_pitch(vp);
+ calc_pan(vp);
+
+ parm = &vp->reg.parm;
+
+ /* compute filter target and correct modulation parameters */
+ if (LO_BYTE(parm->modatkhld) >= 0x80 && parm->moddelay >= 0x8000) {
+ parm->moddelay = 0xbfff;
+ pitch = (HI_BYTE(parm->pefe) << 4) + vp->apitch;
+ if (pitch > 0xffff)
+ pitch = 0xffff;
+ /* calculate filter target */
+ vp->ftarget = parm->cutoff + LO_BYTE(parm->pefe);
+ LIMITVALUE(vp->ftarget, 0, 255);
+ vp->ftarget <<= 8;
+ } else {
+ vp->ftarget = parm->cutoff;
+ vp->ftarget <<= 8;
+ pitch = vp->apitch;
+ }
+
+ /* compute pitch target */
+ if (pitch != 0xffff) {
+ vp->ptarget = 1 << (pitch >> 12);
+ if (pitch & 0x800) vp->ptarget += (vp->ptarget*0x102e)/0x2710;
+ if (pitch & 0x400) vp->ptarget += (vp->ptarget*0x764)/0x2710;
+ if (pitch & 0x200) vp->ptarget += (vp->ptarget*0x389)/0x2710;
+ vp->ptarget += (vp->ptarget >> 1);
+ if (vp->ptarget > 0xffff) vp->ptarget = 0xffff;
+ } else
+ vp->ptarget = 0xffff;
+
+ if (LO_BYTE(parm->modatkhld) >= 0x80) {
+ parm->modatkhld &= ~0xff;
+ parm->modatkhld |= 0x7f;
+ }
+
+ /* compute volume target and correct volume parameters */
+ vp->vtarget = 0;
+#if 0 /* FIXME: this leads to some clicks.. */
+ if (LO_BYTE(parm->volatkhld) >= 0x80 && parm->voldelay >= 0x8000) {
+ parm->voldelay = 0xbfff;
+ vp->vtarget = voltarget[vp->avol % 0x10] >> (vp->avol >> 4);
+ }
+#endif
+
+ if (LO_BYTE(parm->volatkhld) >= 0x80) {
+ parm->volatkhld &= ~0xff;
+ parm->volatkhld |= 0x7f;
+ }
+}
+
+/*
+ * calculate pitch parameter
+ */
+static unsigned char pan_volumes[256] = {
+0x00,0x03,0x06,0x09,0x0c,0x0f,0x12,0x14,0x17,0x1a,0x1d,0x20,0x22,0x25,0x28,0x2a,
+0x2d,0x30,0x32,0x35,0x37,0x3a,0x3c,0x3f,0x41,0x44,0x46,0x49,0x4b,0x4d,0x50,0x52,
+0x54,0x57,0x59,0x5b,0x5d,0x60,0x62,0x64,0x66,0x68,0x6a,0x6c,0x6f,0x71,0x73,0x75,
+0x77,0x79,0x7b,0x7c,0x7e,0x80,0x82,0x84,0x86,0x88,0x89,0x8b,0x8d,0x8f,0x90,0x92,
+0x94,0x96,0x97,0x99,0x9a,0x9c,0x9e,0x9f,0xa1,0xa2,0xa4,0xa5,0xa7,0xa8,0xaa,0xab,
+0xad,0xae,0xaf,0xb1,0xb2,0xb3,0xb5,0xb6,0xb7,0xb9,0xba,0xbb,0xbc,0xbe,0xbf,0xc0,
+0xc1,0xc2,0xc3,0xc5,0xc6,0xc7,0xc8,0xc9,0xca,0xcb,0xcc,0xcd,0xce,0xcf,0xd0,0xd1,
+0xd2,0xd3,0xd4,0xd5,0xd6,0xd7,0xd7,0xd8,0xd9,0xda,0xdb,0xdc,0xdc,0xdd,0xde,0xdf,
+0xdf,0xe0,0xe1,0xe2,0xe2,0xe3,0xe4,0xe4,0xe5,0xe6,0xe6,0xe7,0xe8,0xe8,0xe9,0xe9,
+0xea,0xeb,0xeb,0xec,0xec,0xed,0xed,0xee,0xee,0xef,0xef,0xf0,0xf0,0xf1,0xf1,0xf1,
+0xf2,0xf2,0xf3,0xf3,0xf3,0xf4,0xf4,0xf5,0xf5,0xf5,0xf6,0xf6,0xf6,0xf7,0xf7,0xf7,
+0xf7,0xf8,0xf8,0xf8,0xf9,0xf9,0xf9,0xf9,0xf9,0xfa,0xfa,0xfa,0xfa,0xfb,0xfb,0xfb,
+0xfb,0xfb,0xfc,0xfc,0xfc,0xfc,0xfc,0xfc,0xfc,0xfd,0xfd,0xfd,0xfd,0xfd,0xfd,0xfd,
+0xfd,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,0xfe,
+0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,
+0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,
+};
+
+static int
+calc_pan(struct snd_emux_voice *vp)
+{
+ struct snd_midi_channel *chan = vp->chan;
+ int pan;
+
+ /* pan & loop start (pan 8bit, MSB, 0:right, 0xff:left) */
+ if (vp->reg.fixpan > 0) /* 0-127 */
+ pan = 255 - (int)vp->reg.fixpan * 2;
+ else {
+ pan = chan->control[MIDI_CTL_MSB_PAN] - 64;
+ if (vp->reg.pan >= 0) /* 0-127 */
+ pan += vp->reg.pan - 64;
+ pan = 127 - (int)pan * 2;
+ }
+ LIMITVALUE(pan, 0, 255);
+
+ if (vp->emu->linear_panning) {
+ /* assuming linear volume */
+ if (pan != vp->apan) {
+ vp->apan = pan;
+ if (pan == 0)
+ vp->aaux = 0xff;
+ else
+ vp->aaux = (-pan) & 0xff;
+ return 1;
+ } else
+ return 0;
+ } else {
+ /* using volume table */
+ if (vp->apan != (int)pan_volumes[pan]) {
+ vp->apan = pan_volumes[pan];
+ vp->aaux = pan_volumes[255 - pan];
+ return 1;
+ }
+ return 0;
+ }
+}
+
+
+/*
+ * calculate volume attenuation
+ *
+ * Voice volume is controlled by volume attenuation parameter.
+ * So volume becomes maximum when avol is 0 (no attenuation), and
+ * minimum when 255 (-96dB or silence).
+ */
+
+/* tables for volume->attenuation calculation */
+static unsigned char voltab1[128] = {
+ 0x63, 0x63, 0x63, 0x63, 0x63, 0x63, 0x63, 0x63, 0x63, 0x63,
+ 0x63, 0x2b, 0x29, 0x28, 0x27, 0x26, 0x25, 0x24, 0x23, 0x22,
+ 0x21, 0x20, 0x1f, 0x1e, 0x1e, 0x1d, 0x1c, 0x1b, 0x1b, 0x1a,
+ 0x19, 0x19, 0x18, 0x17, 0x17, 0x16, 0x16, 0x15, 0x15, 0x14,
+ 0x14, 0x13, 0x13, 0x13, 0x12, 0x12, 0x11, 0x11, 0x11, 0x10,
+ 0x10, 0x10, 0x0f, 0x0f, 0x0f, 0x0e, 0x0e, 0x0e, 0x0e, 0x0d,
+ 0x0d, 0x0d, 0x0c, 0x0c, 0x0c, 0x0c, 0x0c, 0x0b, 0x0b, 0x0b,
+ 0x0b, 0x0a, 0x0a, 0x0a, 0x0a, 0x09, 0x09, 0x09, 0x09, 0x09,
+ 0x08, 0x08, 0x08, 0x08, 0x08, 0x07, 0x07, 0x07, 0x07, 0x06,
+ 0x06, 0x06, 0x06, 0x06, 0x05, 0x05, 0x05, 0x05, 0x05, 0x04,
+ 0x04, 0x04, 0x04, 0x04, 0x03, 0x03, 0x03, 0x03, 0x03, 0x02,
+ 0x02, 0x02, 0x02, 0x02, 0x02, 0x01, 0x01, 0x01, 0x01, 0x01,
+ 0x01, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+};
+
+static unsigned char voltab2[128] = {
+ 0x32, 0x31, 0x30, 0x2f, 0x2e, 0x2d, 0x2c, 0x2b, 0x2a, 0x2a,
+ 0x29, 0x28, 0x27, 0x26, 0x25, 0x24, 0x24, 0x23, 0x22, 0x21,
+ 0x21, 0x20, 0x1f, 0x1e, 0x1e, 0x1d, 0x1c, 0x1c, 0x1b, 0x1a,
+ 0x1a, 0x19, 0x19, 0x18, 0x18, 0x17, 0x16, 0x16, 0x15, 0x15,
+ 0x14, 0x14, 0x13, 0x13, 0x13, 0x12, 0x12, 0x11, 0x11, 0x10,
+ 0x10, 0x10, 0x0f, 0x0f, 0x0f, 0x0e, 0x0e, 0x0e, 0x0d, 0x0d,
+ 0x0d, 0x0c, 0x0c, 0x0c, 0x0b, 0x0b, 0x0b, 0x0b, 0x0a, 0x0a,
+ 0x0a, 0x0a, 0x09, 0x09, 0x09, 0x09, 0x09, 0x08, 0x08, 0x08,
+ 0x08, 0x08, 0x07, 0x07, 0x07, 0x07, 0x07, 0x06, 0x06, 0x06,
+ 0x06, 0x06, 0x06, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05,
+ 0x04, 0x04, 0x04, 0x04, 0x04, 0x04, 0x03, 0x03, 0x03, 0x03,
+ 0x03, 0x03, 0x02, 0x02, 0x02, 0x02, 0x02, 0x02, 0x01, 0x01,
+ 0x01, 0x01, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00
+};
+
+static unsigned char expressiontab[128] = {
+ 0x7f, 0x6c, 0x62, 0x5a, 0x54, 0x50, 0x4b, 0x48, 0x45, 0x42,
+ 0x40, 0x3d, 0x3b, 0x39, 0x38, 0x36, 0x34, 0x33, 0x31, 0x30,
+ 0x2f, 0x2d, 0x2c, 0x2b, 0x2a, 0x29, 0x28, 0x27, 0x26, 0x25,
+ 0x24, 0x24, 0x23, 0x22, 0x21, 0x21, 0x20, 0x1f, 0x1e, 0x1e,
+ 0x1d, 0x1d, 0x1c, 0x1b, 0x1b, 0x1a, 0x1a, 0x19, 0x18, 0x18,
+ 0x17, 0x17, 0x16, 0x16, 0x15, 0x15, 0x15, 0x14, 0x14, 0x13,
+ 0x13, 0x12, 0x12, 0x11, 0x11, 0x11, 0x10, 0x10, 0x0f, 0x0f,
+ 0x0f, 0x0e, 0x0e, 0x0e, 0x0d, 0x0d, 0x0d, 0x0c, 0x0c, 0x0c,
+ 0x0b, 0x0b, 0x0b, 0x0a, 0x0a, 0x0a, 0x09, 0x09, 0x09, 0x09,
+ 0x08, 0x08, 0x08, 0x07, 0x07, 0x07, 0x07, 0x06, 0x06, 0x06,
+ 0x06, 0x05, 0x05, 0x05, 0x04, 0x04, 0x04, 0x04, 0x04, 0x03,
+ 0x03, 0x03, 0x03, 0x02, 0x02, 0x02, 0x02, 0x01, 0x01, 0x01,
+ 0x01, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+};
+
+/*
+ * Magic to calculate the volume (actually attenuation) from all the
+ * voice and channels parameters.
+ */
+static int
+calc_volume(struct snd_emux_voice *vp)
+{
+ int vol;
+ int main_vol, expression_vol, master_vol;
+ struct snd_midi_channel *chan = vp->chan;
+ struct snd_emux_port *port = vp->port;
+
+ expression_vol = chan->control[MIDI_CTL_MSB_EXPRESSION];
+ LIMITMAX(vp->velocity, 127);
+ LIMITVALUE(expression_vol, 0, 127);
+ if (port->port_mode == SNDRV_EMUX_PORT_MODE_OSS_SYNTH) {
+ /* 0 - 127 */
+ main_vol = chan->control[MIDI_CTL_MSB_MAIN_VOLUME];
+ vol = (vp->velocity * main_vol * expression_vol) / (127*127);
+ vol = vol * vp->reg.amplitude / 127;
+
+ LIMITVALUE(vol, 0, 127);
+
+ /* calc to attenuation */
+ vol = snd_sf_vol_table[vol];
+
+ } else {
+ main_vol = chan->control[MIDI_CTL_MSB_MAIN_VOLUME] * vp->reg.amplitude / 127;
+ LIMITVALUE(main_vol, 0, 127);
+
+ vol = voltab1[main_vol] + voltab2[vp->velocity];
+ vol = (vol * 8) / 3;
+ vol += vp->reg.attenuation;
+ vol += ((0x100 - vol) * expressiontab[expression_vol])/128;
+ }
+
+ master_vol = port->chset.gs_master_volume;
+ LIMITVALUE(master_vol, 0, 127);
+ vol += snd_sf_vol_table[master_vol];
+ vol += port->volume_atten;
+
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ if (chan->private) {
+ struct snd_emux_effect_table *fx = chan->private;
+ vol += fx->val[EMUX_FX_ATTEN];
+ }
+#endif
+
+ LIMITVALUE(vol, 0, 255);
+ if (vp->avol == vol)
+ return 0; /* value unchanged */
+
+ vp->avol = vol;
+ if (!SF_IS_DRUM_BANK(get_bank(port, chan))
+ && LO_BYTE(vp->reg.parm.volatkhld) < 0x7d) {
+ int atten;
+ if (vp->velocity < 70)
+ atten = 70;
+ else
+ atten = vp->velocity;
+ vp->acutoff = (atten * vp->reg.parm.cutoff + 0xa0) >> 7;
+ } else {
+ vp->acutoff = vp->reg.parm.cutoff;
+ }
+
+ return 1; /* value changed */
+}
+
+/*
+ * calculate pitch offset
+ *
+ * 0xE000 is no pitch offset at 44100Hz sample.
+ * Every 4096 is one octave.
+ */
+
+static int
+calc_pitch(struct snd_emux_voice *vp)
+{
+ struct snd_midi_channel *chan = vp->chan;
+ int offset;
+
+ /* calculate offset */
+ if (vp->reg.fixkey >= 0) {
+ offset = (vp->reg.fixkey - vp->reg.root) * 4096 / 12;
+ } else {
+ offset = (vp->note - vp->reg.root) * 4096 / 12;
+ }
+ offset = (offset * vp->reg.scaleTuning) / 100;
+ offset += vp->reg.tune * 4096 / 1200;
+ if (chan->midi_pitchbend != 0) {
+ /* (128 * 8192: 1 semitone) ==> (4096: 12 semitones) */
+ offset += chan->midi_pitchbend * chan->gm_rpn_pitch_bend_range / 3072;
+ }
+
+ /* tuning via RPN:
+ * coarse = -8192 to 8192 (100 cent per 128)
+ * fine = -8192 to 8192 (max=100cent)
+ */
+ /* 4096 = 1200 cents in emu8000 parameter */
+ offset += chan->gm_rpn_coarse_tuning * 4096 / (12 * 128);
+ offset += chan->gm_rpn_fine_tuning / 24;
+
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+ /* add initial pitch correction */
+ if (chan->private) {
+ struct snd_emux_effect_table *fx = chan->private;
+ if (fx->flag[EMUX_FX_INIT_PITCH])
+ offset += fx->val[EMUX_FX_INIT_PITCH];
+ }
+#endif
+
+ /* 0xe000: root pitch */
+ offset += 0xe000 + vp->reg.rate_offset;
+ offset += vp->emu->pitch_shift;
+ LIMITVALUE(offset, 0, 0xffff);
+ if (offset == vp->apitch)
+ return 0; /* unchanged */
+ vp->apitch = offset;
+ return 1; /* value changed */
+}
+
+/*
+ * Get the bank number assigned to the channel
+ */
+static int
+get_bank(struct snd_emux_port *port, struct snd_midi_channel *chan)
+{
+ int val;
+
+ switch (port->chset.midi_mode) {
+ case SNDRV_MIDI_MODE_XG:
+ val = chan->control[MIDI_CTL_MSB_BANK];
+ if (val == 127)
+ return 128; /* return drum bank */
+ return chan->control[MIDI_CTL_LSB_BANK];
+
+ case SNDRV_MIDI_MODE_GS:
+ if (chan->drum_channel)
+ return 128;
+ /* ignore LSB (bank map) */
+ return chan->control[MIDI_CTL_MSB_BANK];
+
+ default:
+ if (chan->drum_channel)
+ return 128;
+ return chan->control[MIDI_CTL_MSB_BANK];
+ }
+}
+
+
+/* Look for the zones matching with the given note and velocity.
+ * The resultant zones are stored on table.
+ */
+static int
+get_zone(struct snd_emux *emu, struct snd_emux_port *port,
+ int *notep, int vel, struct snd_midi_channel *chan,
+ struct snd_sf_zone **table)
+{
+ int preset, bank, def_preset, def_bank;
+
+ bank = get_bank(port, chan);
+ preset = chan->midi_program;
+
+ if (SF_IS_DRUM_BANK(bank)) {
+ def_preset = port->ctrls[EMUX_MD_DEF_DRUM];
+ def_bank = bank;
+ } else {
+ def_preset = preset;
+ def_bank = port->ctrls[EMUX_MD_DEF_BANK];
+ }
+
+ return snd_soundfont_search_zone(emu->sflist, notep, vel, preset, bank,
+ def_preset, def_bank,
+ table, SNDRV_EMUX_MAX_MULTI_VOICES);
+}
+
+/*
+ */
+void
+snd_emux_init_voices(struct snd_emux *emu)
+{
+ struct snd_emux_voice *vp;
+ int i;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ for (i = 0; i < emu->max_voices; i++) {
+ vp = &emu->voices[i];
+ vp->ch = -1; /* not used */
+ vp->state = SNDRV_EMUX_ST_OFF;
+ vp->chan = NULL;
+ vp->port = NULL;
+ vp->time = 0;
+ vp->emu = emu;
+ vp->hw = emu->hw;
+ }
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+/*
+ */
+void snd_emux_lock_voice(struct snd_emux *emu, int voice)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ if (emu->voices[voice].state == SNDRV_EMUX_ST_OFF)
+ emu->voices[voice].state = SNDRV_EMUX_ST_LOCKED;
+ else
+ snd_printk("invalid voice for lock %d (state = %x)\n",
+ voice, emu->voices[voice].state);
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+EXPORT_SYMBOL(snd_emux_lock_voice);
+
+/*
+ */
+void snd_emux_unlock_voice(struct snd_emux *emu, int voice)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->voice_lock, flags);
+ if (emu->voices[voice].state == SNDRV_EMUX_ST_LOCKED)
+ emu->voices[voice].state = SNDRV_EMUX_ST_OFF;
+ else
+ snd_printk("invalid voice for unlock %d (state = %x)\n",
+ voice, emu->voices[voice].state);
+ spin_unlock_irqrestore(&emu->voice_lock, flags);
+}
+
+EXPORT_SYMBOL(snd_emux_unlock_voice);
diff --git a/sound/synth/emux/emux_voice.h b/sound/synth/emux/emux_voice.h
new file mode 100644
index 0000000..09711f8
--- /dev/null
+++ b/sound/synth/emux/emux_voice.h
@@ -0,0 +1,96 @@
+#ifndef __EMUX_VOICE_H
+#define __EMUX_VOICE_H
+
+/*
+ * A structure to keep track of each hardware voice
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/wait.h>
+#include <linux/sched.h>
+#include <sound/core.h>
+#include <sound/emux_synth.h>
+
+/* Prototypes for emux_seq.c */
+int snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index);
+void snd_emux_detach_seq(struct snd_emux *emu);
+struct snd_emux_port *snd_emux_create_port(struct snd_emux *emu, char *name,
+ int max_channels, int type,
+ struct snd_seq_port_callback *callback);
+void snd_emux_reset_port(struct snd_emux_port *port);
+int snd_emux_event_input(struct snd_seq_event *ev, int direct, void *private,
+ int atomic, int hop);
+int snd_emux_inc_count(struct snd_emux *emu);
+void snd_emux_dec_count(struct snd_emux *emu);
+int snd_emux_init_virmidi(struct snd_emux *emu, struct snd_card *card);
+int snd_emux_delete_virmidi(struct snd_emux *emu);
+
+/* Prototypes for emux_synth.c */
+void snd_emux_init_voices(struct snd_emux *emu);
+
+void snd_emux_note_on(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_emux_note_off(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_emux_key_press(void *p, int note, int vel, struct snd_midi_channel *chan);
+void snd_emux_terminate_note(void *p, int note, struct snd_midi_channel *chan);
+void snd_emux_control(void *p, int type, struct snd_midi_channel *chan);
+
+void snd_emux_sounds_off_all(struct snd_emux_port *port);
+void snd_emux_update_channel(struct snd_emux_port *port,
+ struct snd_midi_channel *chan, int update);
+void snd_emux_update_port(struct snd_emux_port *port, int update);
+
+void snd_emux_timer_callback(unsigned long data);
+
+/* emux_effect.c */
+#ifdef SNDRV_EMUX_USE_RAW_EFFECT
+void snd_emux_create_effect(struct snd_emux_port *p);
+void snd_emux_delete_effect(struct snd_emux_port *p);
+void snd_emux_clear_effect(struct snd_emux_port *p);
+void snd_emux_setup_effect(struct snd_emux_voice *vp);
+void snd_emux_send_effect_oss(struct snd_emux_port *port,
+ struct snd_midi_channel *chan, int type, int val);
+void snd_emux_send_effect(struct snd_emux_port *port,
+ struct snd_midi_channel *chan, int type, int val, int mode);
+#endif
+
+/* emux_nrpn.c */
+void snd_emux_sysex(void *private_data, unsigned char *buf, int len,
+ int parsed, struct snd_midi_channel_set *chset);
+int snd_emux_xg_control(struct snd_emux_port *port,
+ struct snd_midi_channel *chan, int param);
+void snd_emux_nrpn(void *private_data, struct snd_midi_channel *chan,
+ struct snd_midi_channel_set *chset);
+
+/* emux_oss.c */
+void snd_emux_init_seq_oss(struct snd_emux *emu);
+void snd_emux_detach_seq_oss(struct snd_emux *emu);
+
+/* emux_proc.c */
+#ifdef CONFIG_PROC_FS
+void snd_emux_proc_init(struct snd_emux *emu, struct snd_card *card, int device);
+void snd_emux_proc_free(struct snd_emux *emu);
+#endif
+
+#define STATE_IS_PLAYING(s) ((s) & SNDRV_EMUX_ST_ON)
+
+/* emux_hwdep.c */
+int snd_emux_init_hwdep(struct snd_emux *emu);
+void snd_emux_delete_hwdep(struct snd_emux *emu);
+
+#endif
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
new file mode 100644
index 0000000..36d53bd
--- /dev/null
+++ b/sound/synth/emux/soundfont.c
@@ -0,0 +1,1489 @@
+/*
+ * Soundfont generic routines.
+ * It is intended that these should be used by any driver that is willing
+ * to accept soundfont patches.
+ *
+ * Copyright (C) 1999 Steve Ratcliffe
+ * Copyright (c) 1999-2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+/*
+ * Deal with reading in of a soundfont. Code follows the OSS way
+ * of doing things so that the old sfxload utility can be used.
+ * Everything may change when there is an alsa way of doing things.
+ */
+#include <asm/uaccess.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soundfont.h>
+#include <sound/seq_oss_legacy.h>
+
+/* Prototypes for static functions */
+
+static int open_patch(struct snd_sf_list *sflist, const char __user *data,
+ int count, int client);
+static struct snd_soundfont *newsf(struct snd_sf_list *sflist, int type, char *name);
+static int is_identical_font(struct snd_soundfont *sf, int type, unsigned char *name);
+static int close_patch(struct snd_sf_list *sflist);
+static int probe_data(struct snd_sf_list *sflist, int sample_id);
+static void set_zone_counter(struct snd_sf_list *sflist,
+ struct snd_soundfont *sf, struct snd_sf_zone *zp);
+static struct snd_sf_zone *sf_zone_new(struct snd_sf_list *sflist,
+ struct snd_soundfont *sf);
+static void set_sample_counter(struct snd_sf_list *sflist,
+ struct snd_soundfont *sf, struct snd_sf_sample *sp);
+static struct snd_sf_sample *sf_sample_new(struct snd_sf_list *sflist,
+ struct snd_soundfont *sf);
+static void sf_sample_delete(struct snd_sf_list *sflist,
+ struct snd_soundfont *sf, struct snd_sf_sample *sp);
+static int load_map(struct snd_sf_list *sflist, const void __user *data, int count);
+static int load_info(struct snd_sf_list *sflist, const void __user *data, long count);
+static int remove_info(struct snd_sf_list *sflist, struct snd_soundfont *sf,
+ int bank, int instr);
+static void init_voice_info(struct soundfont_voice_info *avp);
+static void init_voice_parm(struct soundfont_voice_parm *pp);
+static struct snd_sf_sample *set_sample(struct snd_soundfont *sf,
+ struct soundfont_voice_info *avp);
+static struct snd_sf_sample *find_sample(struct snd_soundfont *sf, int sample_id);
+static int load_data(struct snd_sf_list *sflist, const void __user *data, long count);
+static void rebuild_presets(struct snd_sf_list *sflist);
+static void add_preset(struct snd_sf_list *sflist, struct snd_sf_zone *cur);
+static void delete_preset(struct snd_sf_list *sflist, struct snd_sf_zone *zp);
+static struct snd_sf_zone *search_first_zone(struct snd_sf_list *sflist,
+ int bank, int preset, int key);
+static int search_zones(struct snd_sf_list *sflist, int *notep, int vel,
+ int preset, int bank, struct snd_sf_zone **table,
+ int max_layers, int level);
+static int get_index(int bank, int instr, int key);
+static void snd_sf_init(struct snd_sf_list *sflist);
+static void snd_sf_clear(struct snd_sf_list *sflist);
+
+/*
+ * lock access to sflist
+ */
+static void
+lock_preset(struct snd_sf_list *sflist)
+{
+ unsigned long flags;
+ mutex_lock(&sflist->presets_mutex);
+ spin_lock_irqsave(&sflist->lock, flags);
+ sflist->presets_locked = 1;
+ spin_unlock_irqrestore(&sflist->lock, flags);
+}
+
+
+/*
+ * remove lock
+ */
+static void
+unlock_preset(struct snd_sf_list *sflist)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&sflist->lock, flags);
+ sflist->presets_locked = 0;
+ spin_unlock_irqrestore(&sflist->lock, flags);
+ mutex_unlock(&sflist->presets_mutex);
+}
+
+
+/*
+ * close the patch if the patch was opened by this client.
+ */
+int
+snd_soundfont_close_check(struct snd_sf_list *sflist, int client)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&sflist->lock, flags);
+ if (sflist->open_client == client) {
+ spin_unlock_irqrestore(&sflist->lock, flags);
+ return close_patch(sflist);
+ }
+ spin_unlock_irqrestore(&sflist->lock, flags);
+ return 0;
+}
+
+
+/*
+ * Deal with a soundfont patch. Any driver could use these routines
+ * although it was designed for the AWE64.
+ *
+ * The sample_write and callargs pararameters allow a callback into
+ * the actual driver to write sample data to the board or whatever
+ * it wants to do with it.
+ */
+int
+snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
+ long count, int client)
+{
+ struct soundfont_patch_info patch;
+ unsigned long flags;
+ int rc;
+
+ if (count < (long)sizeof(patch)) {
+ snd_printk("patch record too small %ld\n", count);
+ return -EINVAL;
+ }
+ if (copy_from_user(&patch, data, sizeof(patch)))
+ return -EFAULT;
+
+ count -= sizeof(patch);
+ data += sizeof(patch);
+
+ if (patch.key != SNDRV_OSS_SOUNDFONT_PATCH) {
+ snd_printk("'The wrong kind of patch' %x\n", patch.key);
+ return -EINVAL;
+ }
+ if (count < patch.len) {
+ snd_printk("Patch too short %ld, need %d\n", count, patch.len);
+ return -EINVAL;
+ }
+ if (patch.len < 0) {
+ snd_printk("poor length %d\n", patch.len);
+ return -EINVAL;
+ }
+
+ if (patch.type == SNDRV_SFNT_OPEN_PATCH) {
+ /* grab sflist to open */
+ lock_preset(sflist);
+ rc = open_patch(sflist, data, count, client);
+ unlock_preset(sflist);
+ return rc;
+ }
+
+ /* check if other client already opened patch */
+ spin_lock_irqsave(&sflist->lock, flags);
+ if (sflist->open_client != client) {
+ spin_unlock_irqrestore(&sflist->lock, flags);
+ return -EBUSY;
+ }
+ spin_unlock_irqrestore(&sflist->lock, flags);
+
+ lock_preset(sflist);
+ rc = -EINVAL;
+ switch (patch.type) {
+ case SNDRV_SFNT_LOAD_INFO:
+ rc = load_info(sflist, data, count);
+ break;
+ case SNDRV_SFNT_LOAD_DATA:
+ rc = load_data(sflist, data, count);
+ break;
+ case SNDRV_SFNT_CLOSE_PATCH:
+ rc = close_patch(sflist);
+ break;
+ case SNDRV_SFNT_REPLACE_DATA:
+ /*rc = replace_data(&patch, data, count);*/
+ break;
+ case SNDRV_SFNT_MAP_PRESET:
+ rc = load_map(sflist, data, count);
+ break;
+ case SNDRV_SFNT_PROBE_DATA:
+ rc = probe_data(sflist, patch.optarg);
+ break;
+ case SNDRV_SFNT_REMOVE_INFO:
+ /* patch must be opened */
+ if (!sflist->currsf) {
+ snd_printk("soundfont: remove_info: patch not opened\n");
+ rc = -EINVAL;
+ } else {
+ int bank, instr;
+ bank = ((unsigned short)patch.optarg >> 8) & 0xff;
+ instr = (unsigned short)patch.optarg & 0xff;
+ if (! remove_info(sflist, sflist->currsf, bank, instr))
+ rc = -EINVAL;
+ else
+ rc = 0;
+ }
+ break;
+ }
+ unlock_preset(sflist);
+
+ return rc;
+}
+
+
+/* check if specified type is special font (GUS or preset-alias) */
+static inline int
+is_special_type(int type)
+{
+ type &= 0x0f;
+ return (type == SNDRV_SFNT_PAT_TYPE_GUS ||
+ type == SNDRV_SFNT_PAT_TYPE_MAP);
+}
+
+
+/* open patch; create sf list */
+static int
+open_patch(struct snd_sf_list *sflist, const char __user *data,
+ int count, int client)
+{
+ struct soundfont_open_parm parm;
+ struct snd_soundfont *sf;
+ unsigned long flags;
+
+ spin_lock_irqsave(&sflist->lock, flags);
+ if (sflist->open_client >= 0 || sflist->currsf) {
+ spin_unlock_irqrestore(&sflist->lock, flags);
+ return -EBUSY;
+ }
+ spin_unlock_irqrestore(&sflist->lock, flags);
+
+ if (copy_from_user(&parm, data, sizeof(parm)))
+ return -EFAULT;
+
+ if (is_special_type(parm.type)) {
+ parm.type |= SNDRV_SFNT_PAT_SHARED;
+ sf = newsf(sflist, parm.type, NULL);
+ } else
+ sf = newsf(sflist, parm.type, parm.name);
+ if (sf == NULL) {
+ return -ENOMEM;
+ }
+
+ spin_lock_irqsave(&sflist->lock, flags);
+ sflist->open_client = client;
+ sflist->currsf = sf;
+ spin_unlock_irqrestore(&sflist->lock, flags);
+
+ return 0;
+}
+
+/*
+ * Allocate a new soundfont structure.
+ */
+static struct snd_soundfont *
+newsf(struct snd_sf_list *sflist, int type, char *name)
+{
+ struct snd_soundfont *sf;
+
+ /* check the shared fonts */
+ if (type & SNDRV_SFNT_PAT_SHARED) {
+ for (sf = sflist->fonts; sf; sf = sf->next) {
+ if (is_identical_font(sf, type, name)) {
+ return sf;
+ }
+ }
+ }
+
+ /* not found -- create a new one */
+ sf = kzalloc(sizeof(*sf), GFP_KERNEL);
+ if (sf == NULL)
+ return NULL;
+ sf->id = sflist->fonts_size;
+ sflist->fonts_size++;
+
+ /* prepend this record */
+ sf->next = sflist->fonts;
+ sflist->fonts = sf;
+
+ sf->type = type;
+ sf->zones = NULL;
+ sf->samples = NULL;
+ if (name)
+ memcpy(sf->name, name, SNDRV_SFNT_PATCH_NAME_LEN);
+
+ return sf;
+}
+
+/* check if the given name matches to the existing list */
+static int
+is_identical_font(struct snd_soundfont *sf, int type, unsigned char *name)
+{
+ return ((sf->type & SNDRV_SFNT_PAT_SHARED) &&
+ (sf->type & 0x0f) == (type & 0x0f) &&
+ (name == NULL ||
+ memcmp(sf->name, name, SNDRV_SFNT_PATCH_NAME_LEN) == 0));
+}
+
+/*
+ * Close the current patch.
+ */
+static int
+close_patch(struct snd_sf_list *sflist)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sflist->lock, flags);
+ sflist->currsf = NULL;
+ sflist->open_client = -1;
+ spin_unlock_irqrestore(&sflist->lock, flags);
+
+ rebuild_presets(sflist);
+
+ return 0;
+
+}
+
+/* probe sample in the current list -- nothing to be loaded */
+static int
+probe_data(struct snd_sf_list *sflist, int sample_id)
+{
+ /* patch must be opened */
+ if (sflist->currsf) {
+ /* search the specified sample by optarg */
+ if (find_sample(sflist->currsf, sample_id))
+ return 0;
+ }
+ return -EINVAL;
+}
+
+/*
+ * increment zone counter
+ */
+static void
+set_zone_counter(struct snd_sf_list *sflist, struct snd_soundfont *sf,
+ struct snd_sf_zone *zp)
+{
+ zp->counter = sflist->zone_counter++;
+ if (sf->type & SNDRV_SFNT_PAT_LOCKED)
+ sflist->zone_locked = sflist->zone_counter;
+}
+
+/*
+ * allocate a new zone record
+ */
+static struct snd_sf_zone *
+sf_zone_new(struct snd_sf_list *sflist, struct snd_soundfont *sf)
+{
+ struct snd_sf_zone *zp;
+
+ if ((zp = kzalloc(sizeof(*zp), GFP_KERNEL)) == NULL)
+ return NULL;
+ zp->next = sf->zones;
+ sf->zones = zp;
+
+ init_voice_info(&zp->v);
+
+ set_zone_counter(sflist, sf, zp);
+ return zp;
+}
+
+
+/*
+ * increment sample couter
+ */
+static void
+set_sample_counter(struct snd_sf_list *sflist, struct snd_soundfont *sf,
+ struct snd_sf_sample *sp)
+{
+ sp->counter = sflist->sample_counter++;
+ if (sf->type & SNDRV_SFNT_PAT_LOCKED)
+ sflist->sample_locked = sflist->sample_counter;
+}
+
+/*
+ * allocate a new sample list record
+ */
+static struct snd_sf_sample *
+sf_sample_new(struct snd_sf_list *sflist, struct snd_soundfont *sf)
+{
+ struct snd_sf_sample *sp;
+
+ if ((sp = kzalloc(sizeof(*sp), GFP_KERNEL)) == NULL)
+ return NULL;
+
+ sp->next = sf->samples;
+ sf->samples = sp;
+
+ set_sample_counter(sflist, sf, sp);
+ return sp;
+}
+
+/*
+ * delete sample list -- this is an exceptional job.
+ * only the last allocated sample can be deleted.
+ */
+static void
+sf_sample_delete(struct snd_sf_list *sflist, struct snd_soundfont *sf,
+ struct snd_sf_sample *sp)
+{
+ /* only last sample is accepted */
+ if (sp == sf->samples) {
+ sf->samples = sp->next;
+ kfree(sp);
+ }
+}
+
+
+/* load voice map */
+static int
+load_map(struct snd_sf_list *sflist, const void __user *data, int count)
+{
+ struct snd_sf_zone *zp, *prevp;
+ struct snd_soundfont *sf;
+ struct soundfont_voice_map map;
+
+ /* get the link info */
+ if (count < (int)sizeof(map))
+ return -EINVAL;
+ if (copy_from_user(&map, data, sizeof(map)))
+ return -EFAULT;
+
+ if (map.map_instr < 0 || map.map_instr >= SF_MAX_INSTRUMENTS)
+ return -EINVAL;
+
+ sf = newsf(sflist, SNDRV_SFNT_PAT_TYPE_MAP|SNDRV_SFNT_PAT_SHARED, NULL);
+ if (sf == NULL)
+ return -ENOMEM;
+
+ prevp = NULL;
+ for (zp = sf->zones; zp; prevp = zp, zp = zp->next) {
+ if (zp->mapped &&
+ zp->instr == map.map_instr &&
+ zp->bank == map.map_bank &&
+ zp->v.low == map.map_key &&
+ zp->v.start == map.src_instr &&
+ zp->v.end == map.src_bank &&
+ zp->v.fixkey == map.src_key) {
+ /* the same mapping is already present */
+ /* relink this record to the link head */
+ if (prevp) {
+ prevp->next = zp->next;
+ zp->next = sf->zones;
+ sf->zones = zp;
+ }
+ /* update the counter */
+ set_zone_counter(sflist, sf, zp);
+ return 0;
+ }
+ }
+
+ /* create a new zone */
+ if ((zp = sf_zone_new(sflist, sf)) == NULL)
+ return -ENOMEM;
+
+ zp->bank = map.map_bank;
+ zp->instr = map.map_instr;
+ zp->mapped = 1;
+ if (map.map_key >= 0) {
+ zp->v.low = map.map_key;
+ zp->v.high = map.map_key;
+ }
+ zp->v.start = map.src_instr;
+ zp->v.end = map.src_bank;
+ zp->v.fixkey = map.src_key;
+ zp->v.sf_id = sf->id;
+
+ add_preset(sflist, zp);
+
+ return 0;
+}
+
+
+/* remove the present instrument layers */
+static int
+remove_info(struct snd_sf_list *sflist, struct snd_soundfont *sf,
+ int bank, int instr)
+{
+ struct snd_sf_zone *prev, *next, *p;
+ int removed = 0;
+
+ prev = NULL;
+ for (p = sf->zones; p; p = next) {
+ next = p->next;
+ if (! p->mapped &&
+ p->bank == bank && p->instr == instr) {
+ /* remove this layer */
+ if (prev)
+ prev->next = next;
+ else
+ sf->zones = next;
+ removed++;
+ kfree(p);
+ } else
+ prev = p;
+ }
+ if (removed)
+ rebuild_presets(sflist);
+ return removed;
+}
+
+
+/*
+ * Read an info record from the user buffer and save it on the current
+ * open soundfont.
+ */
+static int
+load_info(struct snd_sf_list *sflist, const void __user *data, long count)
+{
+ struct snd_soundfont *sf;
+ struct snd_sf_zone *zone;
+ struct soundfont_voice_rec_hdr hdr;
+ int i;
+
+ /* patch must be opened */
+ if ((sf = sflist->currsf) == NULL)
+ return -EINVAL;
+
+ if (is_special_type(sf->type))
+ return -EINVAL;
+
+ if (count < (long)sizeof(hdr)) {
+ printk("Soundfont error: invalid patch zone length\n");
+ return -EINVAL;
+ }
+ if (copy_from_user((char*)&hdr, data, sizeof(hdr)))
+ return -EFAULT;
+
+ data += sizeof(hdr);
+ count -= sizeof(hdr);
+
+ if (hdr.nvoices <= 0 || hdr.nvoices >= 100) {
+ printk("Soundfont error: Illegal voice number %d\n", hdr.nvoices);
+ return -EINVAL;
+ }
+
+ if (count < (long)sizeof(struct soundfont_voice_info) * hdr.nvoices) {
+ printk("Soundfont Error: patch length(%ld) is smaller than nvoices(%d)\n",
+ count, hdr.nvoices);
+ return -EINVAL;
+ }
+
+ switch (hdr.write_mode) {
+ case SNDRV_SFNT_WR_EXCLUSIVE:
+ /* exclusive mode - if the instrument already exists,
+ return error */
+ for (zone = sf->zones; zone; zone = zone->next) {
+ if (!zone->mapped &&
+ zone->bank == hdr.bank &&
+ zone->instr == hdr.instr)
+ return -EINVAL;
+ }
+ break;
+ case SNDRV_SFNT_WR_REPLACE:
+ /* replace mode - remove the instrument if it already exists */
+ remove_info(sflist, sf, hdr.bank, hdr.instr);
+ break;
+ }
+
+ for (i = 0; i < hdr.nvoices; i++) {
+ struct snd_sf_zone tmpzone;
+
+ /* copy awe_voice_info parameters */
+ if (copy_from_user(&tmpzone.v, data, sizeof(tmpzone.v))) {
+ return -EFAULT;
+ }
+
+ data += sizeof(tmpzone.v);
+ count -= sizeof(tmpzone.v);
+
+ tmpzone.bank = hdr.bank;
+ tmpzone.instr = hdr.instr;
+ tmpzone.mapped = 0;
+ tmpzone.v.sf_id = sf->id;
+ if (tmpzone.v.mode & SNDRV_SFNT_MODE_INIT_PARM)
+ init_voice_parm(&tmpzone.v.parm);
+
+ /* create a new zone */
+ if ((zone = sf_zone_new(sflist, sf)) == NULL) {
+ return -ENOMEM;
+ }
+
+ /* copy the temporary data */
+ zone->bank = tmpzone.bank;
+ zone->instr = tmpzone.instr;
+ zone->v = tmpzone.v;
+
+ /* look up the sample */
+ zone->sample = set_sample(sf, &zone->v);
+ }
+
+ return 0;
+}
+
+
+/* initialize voice_info record */
+static void
+init_voice_info(struct soundfont_voice_info *avp)
+{
+ memset(avp, 0, sizeof(*avp));
+
+ avp->root = 60;
+ avp->high = 127;
+ avp->velhigh = 127;
+ avp->fixkey = -1;
+ avp->fixvel = -1;
+ avp->fixpan = -1;
+ avp->pan = -1;
+ avp->amplitude = 127;
+ avp->scaleTuning = 100;
+
+ init_voice_parm(&avp->parm);
+}
+
+/* initialize voice_parm record:
+ * Env1/2: delay=0, attack=0, hold=0, sustain=0, decay=0, release=0.
+ * Vibrato and Tremolo effects are zero.
+ * Cutoff is maximum.
+ * Chorus and Reverb effects are zero.
+ */
+static void
+init_voice_parm(struct soundfont_voice_parm *pp)
+{
+ memset(pp, 0, sizeof(*pp));
+
+ pp->moddelay = 0x8000;
+ pp->modatkhld = 0x7f7f;
+ pp->moddcysus = 0x7f7f;
+ pp->modrelease = 0x807f;
+
+ pp->voldelay = 0x8000;
+ pp->volatkhld = 0x7f7f;
+ pp->voldcysus = 0x7f7f;
+ pp->volrelease = 0x807f;
+
+ pp->lfo1delay = 0x8000;
+ pp->lfo2delay = 0x8000;
+
+ pp->cutoff = 0xff;
+}
+
+/* search the specified sample */
+static struct snd_sf_sample *
+set_sample(struct snd_soundfont *sf, struct soundfont_voice_info *avp)
+{
+ struct snd_sf_sample *sample;
+
+ sample = find_sample(sf, avp->sample);
+ if (sample == NULL)
+ return NULL;
+
+ /* add in the actual sample offsets:
+ * The voice_info addresses define only the relative offset
+ * from sample pointers. Here we calculate the actual DRAM
+ * offset from sample pointers.
+ */
+ avp->start += sample->v.start;
+ avp->end += sample->v.end;
+ avp->loopstart += sample->v.loopstart;
+ avp->loopend += sample->v.loopend;
+
+ /* copy mode flags */
+ avp->sample_mode = sample->v.mode_flags;
+
+ return sample;
+}
+
+/* find the sample pointer with the given id in the soundfont */
+static struct snd_sf_sample *
+find_sample(struct snd_soundfont *sf, int sample_id)
+{
+ struct snd_sf_sample *p;
+
+ if (sf == NULL)
+ return NULL;
+
+ for (p = sf->samples; p; p = p->next) {
+ if (p->v.sample == sample_id)
+ return p;
+ }
+ return NULL;
+}
+
+
+/*
+ * Load sample information, this can include data to be loaded onto
+ * the soundcard. It can also just be a pointer into soundcard ROM.
+ * If there is data it will be written to the soundcard via the callback
+ * routine.
+ */
+static int
+load_data(struct snd_sf_list *sflist, const void __user *data, long count)
+{
+ struct snd_soundfont *sf;
+ struct soundfont_sample_info sample_info;
+ struct snd_sf_sample *sp;
+ long off;
+
+ /* patch must be opened */
+ if ((sf = sflist->currsf) == NULL)
+ return -EINVAL;
+
+ if (is_special_type(sf->type))
+ return -EINVAL;
+
+ if (copy_from_user(&sample_info, data, sizeof(sample_info)))
+ return -EFAULT;
+
+ off = sizeof(sample_info);
+
+ if (sample_info.size != (count-off)/2)
+ return -EINVAL;
+
+ /* Check for dup */
+ if (find_sample(sf, sample_info.sample)) {
+ /* if shared sample, skip this data */
+ if (sf->type & SNDRV_SFNT_PAT_SHARED)
+ return 0;
+ return -EINVAL;
+ }
+
+ /* Allocate a new sample structure */
+ if ((sp = sf_sample_new(sflist, sf)) == NULL)
+ return -ENOMEM;
+
+ sp->v = sample_info;
+ sp->v.sf_id = sf->id;
+ sp->v.dummy = 0;
+ sp->v.truesize = sp->v.size;
+
+ /*
+ * If there is wave data then load it.
+ */
+ if (sp->v.size > 0) {
+ int rc;
+ rc = sflist->callback.sample_new
+ (sflist->callback.private_data, sp, sflist->memhdr,
+ data + off, count - off);
+ if (rc < 0) {
+ sf_sample_delete(sflist, sf, sp);
+ return rc;
+ }
+ sflist->mem_used += sp->v.truesize;
+ }
+
+ return count;
+}
+
+
+/* log2_tbl[i] = log2(i+128) * 0x10000 */
+static int log_tbl[129] = {
+ 0x70000, 0x702df, 0x705b9, 0x7088e, 0x70b5d, 0x70e26, 0x710eb, 0x713aa,
+ 0x71663, 0x71918, 0x71bc8, 0x71e72, 0x72118, 0x723b9, 0x72655, 0x728ed,
+ 0x72b80, 0x72e0e, 0x73098, 0x7331d, 0x7359e, 0x7381b, 0x73a93, 0x73d08,
+ 0x73f78, 0x741e4, 0x7444c, 0x746b0, 0x74910, 0x74b6c, 0x74dc4, 0x75019,
+ 0x75269, 0x754b6, 0x75700, 0x75946, 0x75b88, 0x75dc7, 0x76002, 0x7623a,
+ 0x7646e, 0x766a0, 0x768cd, 0x76af8, 0x76d1f, 0x76f43, 0x77164, 0x77382,
+ 0x7759d, 0x777b4, 0x779c9, 0x77bdb, 0x77dea, 0x77ff5, 0x781fe, 0x78404,
+ 0x78608, 0x78808, 0x78a06, 0x78c01, 0x78df9, 0x78fef, 0x791e2, 0x793d2,
+ 0x795c0, 0x797ab, 0x79993, 0x79b79, 0x79d5d, 0x79f3e, 0x7a11d, 0x7a2f9,
+ 0x7a4d3, 0x7a6ab, 0x7a880, 0x7aa53, 0x7ac24, 0x7adf2, 0x7afbe, 0x7b188,
+ 0x7b350, 0x7b515, 0x7b6d8, 0x7b899, 0x7ba58, 0x7bc15, 0x7bdd0, 0x7bf89,
+ 0x7c140, 0x7c2f5, 0x7c4a7, 0x7c658, 0x7c807, 0x7c9b3, 0x7cb5e, 0x7cd07,
+ 0x7ceae, 0x7d053, 0x7d1f7, 0x7d398, 0x7d538, 0x7d6d6, 0x7d872, 0x7da0c,
+ 0x7dba4, 0x7dd3b, 0x7ded0, 0x7e063, 0x7e1f4, 0x7e384, 0x7e512, 0x7e69f,
+ 0x7e829, 0x7e9b3, 0x7eb3a, 0x7ecc0, 0x7ee44, 0x7efc7, 0x7f148, 0x7f2c8,
+ 0x7f446, 0x7f5c2, 0x7f73d, 0x7f8b7, 0x7fa2f, 0x7fba5, 0x7fd1a, 0x7fe8d,
+ 0x80000,
+};
+
+/* convert from linear to log value
+ *
+ * conversion: value = log2(amount / base) * ratio
+ *
+ * argument:
+ * amount = linear value (unsigned, 32bit max)
+ * offset = base offset (:= log2(base) * 0x10000)
+ * ratio = division ratio
+ *
+ */
+int
+snd_sf_linear_to_log(unsigned int amount, int offset, int ratio)
+{
+ int v;
+ int s, low, bit;
+
+ if (amount < 2)
+ return 0;
+ for (bit = 0; ! (amount & 0x80000000L); bit++)
+ amount <<= 1;
+ s = (amount >> 24) & 0x7f;
+ low = (amount >> 16) & 0xff;
+ /* linear approxmimation by lower 8 bit */
+ v = (log_tbl[s + 1] * low + log_tbl[s] * (0x100 - low)) >> 8;
+ v -= offset;
+ v = (v * ratio) >> 16;
+ v += (24 - bit) * ratio;
+ return v;
+}
+
+EXPORT_SYMBOL(snd_sf_linear_to_log);
+
+
+#define OFFSET_MSEC 653117 /* base = 1000 */
+#define OFFSET_ABSCENT 851781 /* base = 8176 */
+#define OFFSET_SAMPLERATE 1011119 /* base = 44100 */
+
+#define ABSCENT_RATIO 1200
+#define TIMECENT_RATIO 1200
+#define SAMPLERATE_RATIO 4096
+
+/*
+ * mHz to abscent
+ * conversion: abscent = log2(MHz / 8176) * 1200
+ */
+static int
+freq_to_note(int mhz)
+{
+ return snd_sf_linear_to_log(mhz, OFFSET_ABSCENT, ABSCENT_RATIO);
+}
+
+/* convert Hz to AWE32 rate offset:
+ * sample pitch offset for the specified sample rate
+ * rate=44100 is no offset, each 4096 is 1 octave (twice).
+ * eg, when rate is 22050, this offset becomes -4096.
+ *
+ * conversion: offset = log2(Hz / 44100) * 4096
+ */
+static int
+calc_rate_offset(int hz)
+{
+ return snd_sf_linear_to_log(hz, OFFSET_SAMPLERATE, SAMPLERATE_RATIO);
+}
+
+
+/* calculate GUS envelope time */
+static int
+calc_gus_envelope_time(int rate, int start, int end)
+{
+ int r, p, t;
+ r = (3 - ((rate >> 6) & 3)) * 3;
+ p = rate & 0x3f;
+ t = end - start;
+ if (t < 0) t = -t;
+ if (13 > r)
+ t = t << (13 - r);
+ else
+ t = t >> (r - 13);
+ return (t * 10) / (p * 441);
+}
+
+/* convert envelope time parameter to soundfont parameters */
+
+/* attack & decay/release time table (msec) */
+static short attack_time_tbl[128] = {
+32767, 32767, 5989, 4235, 2994, 2518, 2117, 1780, 1497, 1373, 1259, 1154, 1058, 970, 890, 816,
+707, 691, 662, 634, 607, 581, 557, 533, 510, 489, 468, 448, 429, 411, 393, 377,
+361, 345, 331, 317, 303, 290, 278, 266, 255, 244, 234, 224, 214, 205, 196, 188,
+180, 172, 165, 158, 151, 145, 139, 133, 127, 122, 117, 112, 107, 102, 98, 94,
+90, 86, 82, 79, 75, 72, 69, 66, 63, 61, 58, 56, 53, 51, 49, 47,
+45, 43, 41, 39, 37, 36, 34, 33, 31, 30, 29, 28, 26, 25, 24, 23,
+22, 21, 20, 19, 19, 18, 17, 16, 16, 15, 15, 14, 13, 13, 12, 12,
+11, 11, 10, 10, 10, 9, 9, 8, 8, 8, 8, 7, 7, 7, 6, 0,
+};
+
+static short decay_time_tbl[128] = {
+32767, 32767, 22614, 15990, 11307, 9508, 7995, 6723, 5653, 5184, 4754, 4359, 3997, 3665, 3361, 3082,
+2828, 2765, 2648, 2535, 2428, 2325, 2226, 2132, 2042, 1955, 1872, 1793, 1717, 1644, 1574, 1507,
+1443, 1382, 1324, 1267, 1214, 1162, 1113, 1066, 978, 936, 897, 859, 822, 787, 754, 722,
+691, 662, 634, 607, 581, 557, 533, 510, 489, 468, 448, 429, 411, 393, 377, 361,
+345, 331, 317, 303, 290, 278, 266, 255, 244, 234, 224, 214, 205, 196, 188, 180,
+172, 165, 158, 151, 145, 139, 133, 127, 122, 117, 112, 107, 102, 98, 94, 90,
+86, 82, 79, 75, 72, 69, 66, 63, 61, 58, 56, 53, 51, 49, 47, 45,
+43, 41, 39, 37, 36, 34, 33, 31, 30, 29, 28, 26, 25, 24, 23, 22,
+};
+
+/* delay time = 0x8000 - msec/92 */
+int
+snd_sf_calc_parm_hold(int msec)
+{
+ int val = (0x7f * 92 - msec) / 92;
+ if (val < 1) val = 1;
+ if (val >= 126) val = 126;
+ return val;
+}
+
+/* search an index for specified time from given time table */
+static int
+calc_parm_search(int msec, short *table)
+{
+ int left = 1, right = 127, mid;
+ while (left < right) {
+ mid = (left + right) / 2;
+ if (msec < (int)table[mid])
+ left = mid + 1;
+ else
+ right = mid;
+ }
+ return left;
+}
+
+/* attack time: search from time table */
+int
+snd_sf_calc_parm_attack(int msec)
+{
+ return calc_parm_search(msec, attack_time_tbl);
+}
+
+/* decay/release time: search from time table */
+int
+snd_sf_calc_parm_decay(int msec)
+{
+ return calc_parm_search(msec, decay_time_tbl);
+}
+
+int snd_sf_vol_table[128] = {
+ 255,111,95,86,79,74,70,66,63,61,58,56,54,52,50,49,
+ 47,46,45,43,42,41,40,39,38,37,36,35,34,34,33,32,
+ 31,31,30,29,29,28,27,27,26,26,25,24,24,23,23,22,
+ 22,21,21,21,20,20,19,19,18,18,18,17,17,16,16,16,
+ 15,15,15,14,14,14,13,13,13,12,12,12,11,11,11,10,
+ 10,10,10,9,9,9,8,8,8,8,7,7,7,7,6,6,
+ 6,6,5,5,5,5,5,4,4,4,4,3,3,3,3,3,
+ 2,2,2,2,2,1,1,1,1,1,0,0,0,0,0,0,
+};
+
+
+#define calc_gus_sustain(val) (0x7f - snd_sf_vol_table[(val)/2])
+#define calc_gus_attenuation(val) snd_sf_vol_table[(val)/2]
+
+/* load GUS patch */
+static int
+load_guspatch(struct snd_sf_list *sflist, const char __user *data,
+ long count, int client)
+{
+ struct patch_info patch;
+ struct snd_soundfont *sf;
+ struct snd_sf_zone *zone;
+ struct snd_sf_sample *smp;
+ int note, sample_id;
+ int rc;
+
+ if (count < (long)sizeof(patch)) {
+ snd_printk("patch record too small %ld\n", count);
+ return -EINVAL;
+ }
+ if (copy_from_user(&patch, data, sizeof(patch)))
+ return -EFAULT;
+
+ count -= sizeof(patch);
+ data += sizeof(patch);
+
+ sf = newsf(sflist, SNDRV_SFNT_PAT_TYPE_GUS|SNDRV_SFNT_PAT_SHARED, NULL);
+ if (sf == NULL)
+ return -ENOMEM;
+ if ((smp = sf_sample_new(sflist, sf)) == NULL)
+ return -ENOMEM;
+ sample_id = sflist->sample_counter;
+ smp->v.sample = sample_id;
+ smp->v.start = 0;
+ smp->v.end = patch.len;
+ smp->v.loopstart = patch.loop_start;
+ smp->v.loopend = patch.loop_end;
+ smp->v.size = patch.len;
+
+ /* set up mode flags */
+ smp->v.mode_flags = 0;
+ if (!(patch.mode & WAVE_16_BITS))
+ smp->v.mode_flags |= SNDRV_SFNT_SAMPLE_8BITS;
+ if (patch.mode & WAVE_UNSIGNED)
+ smp->v.mode_flags |= SNDRV_SFNT_SAMPLE_UNSIGNED;
+ smp->v.mode_flags |= SNDRV_SFNT_SAMPLE_NO_BLANK;
+ if (!(patch.mode & (WAVE_LOOPING|WAVE_BIDIR_LOOP|WAVE_LOOP_BACK)))
+ smp->v.mode_flags |= SNDRV_SFNT_SAMPLE_SINGLESHOT;
+ if (patch.mode & WAVE_BIDIR_LOOP)
+ smp->v.mode_flags |= SNDRV_SFNT_SAMPLE_BIDIR_LOOP;
+ if (patch.mode & WAVE_LOOP_BACK)
+ smp->v.mode_flags |= SNDRV_SFNT_SAMPLE_REVERSE_LOOP;
+
+ if (patch.mode & WAVE_16_BITS) {
+ /* convert to word offsets */
+ smp->v.size /= 2;
+ smp->v.end /= 2;
+ smp->v.loopstart /= 2;
+ smp->v.loopend /= 2;
+ }
+ /*smp->v.loopend++;*/
+
+ smp->v.dummy = 0;
+ smp->v.truesize = 0;
+ smp->v.sf_id = sf->id;
+
+ /* set up voice info */
+ if ((zone = sf_zone_new(sflist, sf)) == NULL) {
+ sf_sample_delete(sflist, sf, smp);
+ return -ENOMEM;
+ }
+
+ /*
+ * load wave data
+ */
+ if (sflist->callback.sample_new) {
+ rc = sflist->callback.sample_new
+ (sflist->callback.private_data, smp, sflist->memhdr,
+ data, count);
+ if (rc < 0) {
+ sf_sample_delete(sflist, sf, smp);
+ return rc;
+ }
+ /* memory offset is updated after */
+ }
+
+ /* update the memory offset here */
+ sflist->mem_used += smp->v.truesize;
+
+ zone->v.sample = sample_id; /* the last sample */
+ zone->v.rate_offset = calc_rate_offset(patch.base_freq);
+ note = freq_to_note(patch.base_note);
+ zone->v.root = note / 100;
+ zone->v.tune = -(note % 100);
+ zone->v.low = (freq_to_note(patch.low_note) + 99) / 100;
+ zone->v.high = freq_to_note(patch.high_note) / 100;
+ /* panning position; -128 - 127 => 0-127 */
+ zone->v.pan = (patch.panning + 128) / 2;
+#if 0
+ snd_printk("gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n",
+ (int)patch.base_freq, zone->v.rate_offset,
+ zone->v.root, zone->v.tune, zone->v.low, zone->v.high);
+#endif
+
+ /* detuning is ignored */
+ /* 6points volume envelope */
+ if (patch.mode & WAVE_ENVELOPES) {
+ int attack, hold, decay, release;
+ attack = calc_gus_envelope_time
+ (patch.env_rate[0], 0, patch.env_offset[0]);
+ hold = calc_gus_envelope_time
+ (patch.env_rate[1], patch.env_offset[0],
+ patch.env_offset[1]);
+ decay = calc_gus_envelope_time
+ (patch.env_rate[2], patch.env_offset[1],
+ patch.env_offset[2]);
+ release = calc_gus_envelope_time
+ (patch.env_rate[3], patch.env_offset[1],
+ patch.env_offset[4]);
+ release += calc_gus_envelope_time
+ (patch.env_rate[4], patch.env_offset[3],
+ patch.env_offset[4]);
+ release += calc_gus_envelope_time
+ (patch.env_rate[5], patch.env_offset[4],
+ patch.env_offset[5]);
+ zone->v.parm.volatkhld =
+ (snd_sf_calc_parm_hold(hold) << 8) |
+ snd_sf_calc_parm_attack(attack);
+ zone->v.parm.voldcysus = (calc_gus_sustain(patch.env_offset[2]) << 8) |
+ snd_sf_calc_parm_decay(decay);
+ zone->v.parm.volrelease = 0x8000 | snd_sf_calc_parm_decay(release);
+ zone->v.attenuation = calc_gus_attenuation(patch.env_offset[0]);
+#if 0
+ snd_printk("gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n",
+ zone->v.parm.volatkhld,
+ zone->v.parm.voldcysus,
+ zone->v.parm.volrelease,
+ zone->v.attenuation);
+#endif
+ }
+
+ /* fast release */
+ if (patch.mode & WAVE_FAST_RELEASE) {
+ zone->v.parm.volrelease = 0x807f;
+ }
+
+ /* tremolo effect */
+ if (patch.mode & WAVE_TREMOLO) {
+ int rate = (patch.tremolo_rate * 1000 / 38) / 42;
+ zone->v.parm.tremfrq = ((patch.tremolo_depth / 2) << 8) | rate;
+ }
+ /* vibrato effect */
+ if (patch.mode & WAVE_VIBRATO) {
+ int rate = (patch.vibrato_rate * 1000 / 38) / 42;
+ zone->v.parm.fm2frq2 = ((patch.vibrato_depth / 6) << 8) | rate;
+ }
+
+ /* scale_freq, scale_factor, volume, and fractions not implemented */
+
+ if (!(smp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT))
+ zone->v.mode = SNDRV_SFNT_MODE_LOOPING;
+ else
+ zone->v.mode = 0;
+
+ /* append to the tail of the list */
+ /*zone->bank = ctrls[AWE_MD_GUS_BANK];*/
+ zone->bank = 0;
+ zone->instr = patch.instr_no;
+ zone->mapped = 0;
+ zone->v.sf_id = sf->id;
+
+ zone->sample = set_sample(sf, &zone->v);
+
+ /* rebuild preset now */
+ add_preset(sflist, zone);
+
+ return 0;
+}
+
+/* load GUS patch */
+int
+snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data,
+ long count, int client)
+{
+ int rc;
+ lock_preset(sflist);
+ rc = load_guspatch(sflist, data, count, client);
+ unlock_preset(sflist);
+ return rc;
+}
+
+
+/*
+ * Rebuild the preset table. This is like a hash table in that it allows
+ * quick access to the zone information. For each preset there are zone
+ * structures linked by next_instr and by next_zone. Former is the whole
+ * link for this preset, and latter is the link for zone (i.e. instrument/
+ * bank/key combination).
+ */
+static void
+rebuild_presets(struct snd_sf_list *sflist)
+{
+ struct snd_soundfont *sf;
+ struct snd_sf_zone *cur;
+
+ /* clear preset table */
+ memset(sflist->presets, 0, sizeof(sflist->presets));
+
+ /* search all fonts and insert each font */
+ for (sf = sflist->fonts; sf; sf = sf->next) {
+ for (cur = sf->zones; cur; cur = cur->next) {
+ if (! cur->mapped && cur->sample == NULL) {
+ /* try again to search the corresponding sample */
+ cur->sample = set_sample(sf, &cur->v);
+ if (cur->sample == NULL)
+ continue;
+ }
+
+ add_preset(sflist, cur);
+ }
+ }
+}
+
+
+/*
+ * add the given zone to preset table
+ */
+static void
+add_preset(struct snd_sf_list *sflist, struct snd_sf_zone *cur)
+{
+ struct snd_sf_zone *zone;
+ int index;
+
+ zone = search_first_zone(sflist, cur->bank, cur->instr, cur->v.low);
+ if (zone && zone->v.sf_id != cur->v.sf_id) {
+ /* different instrument was already defined */
+ struct snd_sf_zone *p;
+ /* compare the allocated time */
+ for (p = zone; p; p = p->next_zone) {
+ if (p->counter > cur->counter)
+ /* the current is older.. skipped */
+ return;
+ }
+ /* remove old zones */
+ delete_preset(sflist, zone);
+ zone = NULL; /* do not forget to clear this! */
+ }
+
+ /* prepend this zone */
+ if ((index = get_index(cur->bank, cur->instr, cur->v.low)) < 0)
+ return;
+ cur->next_zone = zone; /* zone link */
+ cur->next_instr = sflist->presets[index]; /* preset table link */
+ sflist->presets[index] = cur;
+}
+
+/*
+ * delete the given zones from preset_table
+ */
+static void
+delete_preset(struct snd_sf_list *sflist, struct snd_sf_zone *zp)
+{
+ int index;
+ struct snd_sf_zone *p;
+
+ if ((index = get_index(zp->bank, zp->instr, zp->v.low)) < 0)
+ return;
+ for (p = sflist->presets[index]; p; p = p->next_instr) {
+ while (p->next_instr == zp) {
+ p->next_instr = zp->next_instr;
+ zp = zp->next_zone;
+ if (zp == NULL)
+ return;
+ }
+ }
+}
+
+
+/*
+ * Search matching zones from preset table.
+ * The note can be rewritten by preset mapping (alias).
+ * The found zones are stored on 'table' array. max_layers defines
+ * the maximum number of elements in this array.
+ * This function returns the number of found zones. 0 if not found.
+ */
+int
+snd_soundfont_search_zone(struct snd_sf_list *sflist, int *notep, int vel,
+ int preset, int bank,
+ int def_preset, int def_bank,
+ struct snd_sf_zone **table, int max_layers)
+{
+ int nvoices;
+ unsigned long flags;
+
+ /* this function is supposed to be called atomically,
+ * so we check the lock. if it's busy, just returns 0 to
+ * tell the caller the busy state
+ */
+ spin_lock_irqsave(&sflist->lock, flags);
+ if (sflist->presets_locked) {
+ spin_unlock_irqrestore(&sflist->lock, flags);
+ return 0;
+ }
+ nvoices = search_zones(sflist, notep, vel, preset, bank,
+ table, max_layers, 0);
+ if (! nvoices) {
+ if (preset != def_preset || bank != def_bank)
+ nvoices = search_zones(sflist, notep, vel,
+ def_preset, def_bank,
+ table, max_layers, 0);
+ }
+ spin_unlock_irqrestore(&sflist->lock, flags);
+ return nvoices;
+}
+
+
+/*
+ * search the first matching zone
+ */
+static struct snd_sf_zone *
+search_first_zone(struct snd_sf_list *sflist, int bank, int preset, int key)
+{
+ int index;
+ struct snd_sf_zone *zp;
+
+ if ((index = get_index(bank, preset, key)) < 0)
+ return NULL;
+ for (zp = sflist->presets[index]; zp; zp = zp->next_instr) {
+ if (zp->instr == preset && zp->bank == bank)
+ return zp;
+ }
+ return NULL;
+}
+
+
+/*
+ * search matching zones from sflist. can be called recursively.
+ */
+static int
+search_zones(struct snd_sf_list *sflist, int *notep, int vel,
+ int preset, int bank, struct snd_sf_zone **table,
+ int max_layers, int level)
+{
+ struct snd_sf_zone *zp;
+ int nvoices;
+
+ zp = search_first_zone(sflist, bank, preset, *notep);
+ nvoices = 0;
+ for (; zp; zp = zp->next_zone) {
+ if (*notep >= zp->v.low && *notep <= zp->v.high &&
+ vel >= zp->v.vellow && vel <= zp->v.velhigh) {
+ if (zp->mapped) {
+ /* search preset mapping (aliasing) */
+ int key = zp->v.fixkey;
+ preset = zp->v.start;
+ bank = zp->v.end;
+
+ if (level > 5) /* too deep alias level */
+ return 0;
+ if (key < 0)
+ key = *notep;
+ nvoices = search_zones(sflist, &key, vel,
+ preset, bank, table,
+ max_layers, level + 1);
+ if (nvoices > 0)
+ *notep = key;
+ break;
+ }
+ table[nvoices++] = zp;
+ if (nvoices >= max_layers)
+ break;
+ }
+ }
+
+ return nvoices;
+}
+
+
+/* calculate the index of preset table:
+ * drums are mapped from 128 to 255 according to its note key.
+ * other instruments are mapped from 0 to 127.
+ * if the index is out of range, return -1.
+ */
+static int
+get_index(int bank, int instr, int key)
+{
+ int index;
+ if (SF_IS_DRUM_BANK(bank))
+ index = key + SF_MAX_INSTRUMENTS;
+ else
+ index = instr;
+ index = index % SF_MAX_PRESETS;
+ if (index < 0)
+ return -1;
+ return index;
+}
+
+/*
+ * Initialise the sflist structure.
+ */
+static void
+snd_sf_init(struct snd_sf_list *sflist)
+{
+ memset(sflist->presets, 0, sizeof(sflist->presets));
+
+ sflist->mem_used = 0;
+ sflist->currsf = NULL;
+ sflist->open_client = -1;
+ sflist->fonts = NULL;
+ sflist->fonts_size = 0;
+ sflist->zone_counter = 0;
+ sflist->sample_counter = 0;
+ sflist->zone_locked = 0;
+ sflist->sample_locked = 0;
+}
+
+/*
+ * Release all list records
+ */
+static void
+snd_sf_clear(struct snd_sf_list *sflist)
+{
+ struct snd_soundfont *sf, *nextsf;
+ struct snd_sf_zone *zp, *nextzp;
+ struct snd_sf_sample *sp, *nextsp;
+
+ for (sf = sflist->fonts; sf; sf = nextsf) {
+ nextsf = sf->next;
+ for (zp = sf->zones; zp; zp = nextzp) {
+ nextzp = zp->next;
+ kfree(zp);
+ }
+ for (sp = sf->samples; sp; sp = nextsp) {
+ nextsp = sp->next;
+ if (sflist->callback.sample_free)
+ sflist->callback.sample_free(sflist->callback.private_data,
+ sp, sflist->memhdr);
+ kfree(sp);
+ }
+ kfree(sf);
+ }
+
+ snd_sf_init(sflist);
+}
+
+
+/*
+ * Create a new sflist structure
+ */
+struct snd_sf_list *
+snd_sf_new(struct snd_sf_callback *callback, struct snd_util_memhdr *hdr)
+{
+ struct snd_sf_list *sflist;
+
+ if ((sflist = kzalloc(sizeof(*sflist), GFP_KERNEL)) == NULL)
+ return NULL;
+
+ mutex_init(&sflist->presets_mutex);
+ spin_lock_init(&sflist->lock);
+ sflist->memhdr = hdr;
+
+ if (callback)
+ sflist->callback = *callback;
+
+ snd_sf_init(sflist);
+ return sflist;
+}
+
+
+/*
+ * Free everything allocated off the sflist structure.
+ */
+void
+snd_sf_free(struct snd_sf_list *sflist)
+{
+ if (sflist == NULL)
+ return;
+
+ lock_preset(sflist);
+ if (sflist->callback.sample_reset)
+ sflist->callback.sample_reset(sflist->callback.private_data);
+ snd_sf_clear(sflist);
+ unlock_preset(sflist);
+
+ kfree(sflist);
+}
+
+/*
+ * Remove all samples
+ * The soundcard should be silet before calling this function.
+ */
+int
+snd_soundfont_remove_samples(struct snd_sf_list *sflist)
+{
+ lock_preset(sflist);
+ if (sflist->callback.sample_reset)
+ sflist->callback.sample_reset(sflist->callback.private_data);
+ snd_sf_clear(sflist);
+ unlock_preset(sflist);
+
+ return 0;
+}
+
+/*
+ * Remove unlocked samples.
+ * The soundcard should be silent before calling this function.
+ */
+int
+snd_soundfont_remove_unlocked(struct snd_sf_list *sflist)
+{
+ struct snd_soundfont *sf;
+ struct snd_sf_zone *zp, *nextzp;
+ struct snd_sf_sample *sp, *nextsp;
+
+ lock_preset(sflist);
+
+ if (sflist->callback.sample_reset)
+ sflist->callback.sample_reset(sflist->callback.private_data);
+
+ /* to be sure */
+ memset(sflist->presets, 0, sizeof(sflist->presets));
+
+ for (sf = sflist->fonts; sf; sf = sf->next) {
+ for (zp = sf->zones; zp; zp = nextzp) {
+ if (zp->counter < sflist->zone_locked)
+ break;
+ nextzp = zp->next;
+ sf->zones = nextzp;
+ kfree(zp);
+ }
+
+ for (sp = sf->samples; sp; sp = nextsp) {
+ if (sp->counter < sflist->sample_locked)
+ break;
+ nextsp = sp->next;
+ sf->samples = nextsp;
+ sflist->mem_used -= sp->v.truesize;
+ if (sflist->callback.sample_free)
+ sflist->callback.sample_free(sflist->callback.private_data,
+ sp, sflist->memhdr);
+ kfree(sp);
+ }
+ }
+
+ sflist->zone_counter = sflist->zone_locked;
+ sflist->sample_counter = sflist->sample_locked;
+
+ rebuild_presets(sflist);
+
+ unlock_preset(sflist);
+ return 0;
+}
diff --git a/sound/synth/util_mem.c b/sound/synth/util_mem.c
new file mode 100644
index 0000000..c85522e
--- /dev/null
+++ b/sound/synth/util_mem.c
@@ -0,0 +1,210 @@
+/*
+ * Copyright (C) 2000 Takashi Iwai <tiwai@suse.de>
+ *
+ * Generic memory management routines for soundcard memory allocation
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/mutex.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/util_mem.h>
+
+MODULE_AUTHOR("Takashi Iwai");
+MODULE_DESCRIPTION("Generic memory management routines for soundcard memory allocation");
+MODULE_LICENSE("GPL");
+
+#define get_memblk(p) list_entry(p, struct snd_util_memblk, list)
+
+/*
+ * create a new memory manager
+ */
+struct snd_util_memhdr *
+snd_util_memhdr_new(int memsize)
+{
+ struct snd_util_memhdr *hdr;
+
+ hdr = kzalloc(sizeof(*hdr), GFP_KERNEL);
+ if (hdr == NULL)
+ return NULL;
+ hdr->size = memsize;
+ mutex_init(&hdr->block_mutex);
+ INIT_LIST_HEAD(&hdr->block);
+
+ return hdr;
+}
+
+/*
+ * free a memory manager
+ */
+void snd_util_memhdr_free(struct snd_util_memhdr *hdr)
+{
+ struct list_head *p;
+
+ if (!hdr)
+ return;
+ /* release all blocks */
+ while ((p = hdr->block.next) != &hdr->block) {
+ list_del(p);
+ kfree(get_memblk(p));
+ }
+ kfree(hdr);
+}
+
+/*
+ * allocate a memory block (without mutex)
+ */
+struct snd_util_memblk *
+__snd_util_mem_alloc(struct snd_util_memhdr *hdr, int size)
+{
+ struct snd_util_memblk *blk;
+ unsigned int units, prev_offset;
+ struct list_head *p;
+
+ if (snd_BUG_ON(!hdr || size <= 0))
+ return NULL;
+
+ /* word alignment */
+ units = size;
+ if (units & 1)
+ units++;
+ if (units > hdr->size)
+ return NULL;
+
+ /* look for empty block */
+ prev_offset = 0;
+ list_for_each(p, &hdr->block) {
+ blk = get_memblk(p);
+ if (blk->offset - prev_offset >= units)
+ goto __found;
+ prev_offset = blk->offset + blk->size;
+ }
+ if (hdr->size - prev_offset < units)
+ return NULL;
+
+__found:
+ return __snd_util_memblk_new(hdr, units, p->prev);
+}
+
+
+/*
+ * create a new memory block with the given size
+ * the block is linked next to prev
+ */
+struct snd_util_memblk *
+__snd_util_memblk_new(struct snd_util_memhdr *hdr, unsigned int units,
+ struct list_head *prev)
+{
+ struct snd_util_memblk *blk;
+
+ blk = kmalloc(sizeof(struct snd_util_memblk) + hdr->block_extra_size,
+ GFP_KERNEL);
+ if (blk == NULL)
+ return NULL;
+
+ if (prev == &hdr->block)
+ blk->offset = 0;
+ else {
+ struct snd_util_memblk *p = get_memblk(prev);
+ blk->offset = p->offset + p->size;
+ }
+ blk->size = units;
+ list_add(&blk->list, prev);
+ hdr->nblocks++;
+ hdr->used += units;
+ return blk;
+}
+
+
+/*
+ * allocate a memory block (with mutex)
+ */
+struct snd_util_memblk *
+snd_util_mem_alloc(struct snd_util_memhdr *hdr, int size)
+{
+ struct snd_util_memblk *blk;
+ mutex_lock(&hdr->block_mutex);
+ blk = __snd_util_mem_alloc(hdr, size);
+ mutex_unlock(&hdr->block_mutex);
+ return blk;
+}
+
+
+/*
+ * remove the block from linked-list and free resource
+ * (without mutex)
+ */
+void
+__snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk)
+{
+ list_del(&blk->list);
+ hdr->nblocks--;
+ hdr->used -= blk->size;
+ kfree(blk);
+}
+
+/*
+ * free a memory block (with mutex)
+ */
+int snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk)
+{
+ if (snd_BUG_ON(!hdr || !blk))
+ return -EINVAL;
+
+ mutex_lock(&hdr->block_mutex);
+ __snd_util_mem_free(hdr, blk);
+ mutex_unlock(&hdr->block_mutex);
+ return 0;
+}
+
+/*
+ * return available memory size
+ */
+int snd_util_mem_avail(struct snd_util_memhdr *hdr)
+{
+ unsigned int size;
+ mutex_lock(&hdr->block_mutex);
+ size = hdr->size - hdr->used;
+ mutex_unlock(&hdr->block_mutex);
+ return size;
+}
+
+
+EXPORT_SYMBOL(snd_util_memhdr_new);
+EXPORT_SYMBOL(snd_util_memhdr_free);
+EXPORT_SYMBOL(snd_util_mem_alloc);
+EXPORT_SYMBOL(snd_util_mem_free);
+EXPORT_SYMBOL(snd_util_mem_avail);
+EXPORT_SYMBOL(__snd_util_mem_alloc);
+EXPORT_SYMBOL(__snd_util_mem_free);
+EXPORT_SYMBOL(__snd_util_memblk_new);
+
+/*
+ * INIT part
+ */
+
+static int __init alsa_util_mem_init(void)
+{
+ return 0;
+}
+
+static void __exit alsa_util_mem_exit(void)
+{
+}
+
+module_init(alsa_util_mem_init)
+module_exit(alsa_util_mem_exit)
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
new file mode 100644
index 0000000..4f0eac9
--- /dev/null
+++ b/sound/usb/Kconfig
@@ -0,0 +1,83 @@
+# ALSA USB drivers
+
+menuconfig SND_USB
+ bool "USB sound devices"
+ depends on USB
+ default y
+ help
+ Support for sound devices connected via the USB bus.
+
+if SND_USB && USB
+
+config SND_USB_AUDIO
+ tristate "USB Audio/MIDI driver"
+ select SND_HWDEP
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for USB audio and USB MIDI
+ devices.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-usb-audio.
+
+config SND_USB_USX2Y
+ tristate "Tascam US-122, US-224 and US-428 USB driver"
+ depends on X86 || PPC || ALPHA
+ select SND_HWDEP
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for Tascam USB Audio/MIDI
+ interfaces or controllers US-122, US-224 and US-428.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-usb-usx2y.
+
+config SND_USB_CAIAQ
+ tristate "Native Instruments USB audio devices"
+ select SND_HWDEP
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for caiaq USB audio interfaces,
+ namely:
+
+ * Native Instruments RigKontrol2
+ * Native Instruments RigKontrol3
+ * Native Instruments Kore Controller
+ * Native Instruments Kore Controller 2
+ * Native Instruments Audio Kontrol 1
+ * Native Instruments Audio 8 DJ
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-usb-caiaq.
+
+config SND_USB_CAIAQ_INPUT
+ bool "enable input device for controllers"
+ depends on SND_USB_CAIAQ
+ depends on INPUT=y || INPUT=SND_USB_CAIAQ
+ help
+ Say Y here to support input controllers like buttons, knobs,
+ alpha dials and analog pedals on the following products:
+
+ * Native Instruments RigKontrol2
+ * Native Instruments RigKontrol3
+ * Native Instruments Kore Controller
+ * Native Instruments Kore Controller 2
+ * Native Instruments Audio Kontrol 1
+
+config SND_USB_US122L
+ tristate "Tascam US-122L USB driver"
+ depends on X86 && EXPERIMENTAL
+ select SND_HWDEP
+ select SND_RAWMIDI
+ help
+ Say Y here to include support for Tascam US-122L USB Audio/MIDI
+ interfaces.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-usb-us122l.
+
+endif # SND_USB
+
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
new file mode 100644
index 0000000..abb288b
--- /dev/null
+++ b/sound/usb/Makefile
@@ -0,0 +1,13 @@
+#
+# Makefile for ALSA
+#
+
+snd-usb-audio-objs := usbaudio.o usbmixer.o
+snd-usb-lib-objs := usbmidi.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usb-lib.o
+obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-lib.o
+obj-$(CONFIG_SND_USB_US122L) += snd-usb-lib.o
+
+obj-$(CONFIG_SND) += usx2y/ caiaq/
diff --git a/sound/usb/caiaq/Makefile b/sound/usb/caiaq/Makefile
new file mode 100644
index 0000000..23dadd5
--- /dev/null
+++ b/sound/usb/caiaq/Makefile
@@ -0,0 +1,4 @@
+snd-usb-caiaq-y := caiaq-device.o caiaq-audio.o caiaq-midi.o caiaq-control.o
+snd-usb-caiaq-$(CONFIG_SND_USB_CAIAQ_INPUT) += caiaq-input.o
+
+obj-$(CONFIG_SND_USB_CAIAQ) += snd-usb-caiaq.o
diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c
new file mode 100644
index 0000000..b3a6033
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-audio.c
@@ -0,0 +1,697 @@
+/*
+ * Copyright (c) 2006-2008 Daniel Mack, Karsten Wiese
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/usb.h>
+#include <linux/spinlock.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <linux/input.h>
+
+#include "caiaq-device.h"
+#include "caiaq-audio.h"
+
+#define N_URBS 32
+#define CLOCK_DRIFT_TOLERANCE 5
+#define FRAMES_PER_URB 8
+#define BYTES_PER_FRAME 512
+#define CHANNELS_PER_STREAM 2
+#define BYTES_PER_SAMPLE 3
+#define BYTES_PER_SAMPLE_USB 4
+#define MAX_BUFFER_SIZE (128*1024)
+#define MAX_ENDPOINT_SIZE 512
+
+#define ENDPOINT_CAPTURE 2
+#define ENDPOINT_PLAYBACK 6
+
+#define MAKE_CHECKBYTE(dev,stream,i) \
+ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1)
+
+static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER),
+ .formats = SNDRV_PCM_FMTBIT_S24_3BE,
+ .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000),
+ .rate_min = 44100,
+ .rate_max = 0, /* will overwrite later */
+ .channels_min = CHANNELS_PER_STREAM,
+ .channels_max = CHANNELS_PER_STREAM,
+ .buffer_bytes_max = MAX_BUFFER_SIZE,
+ .period_bytes_min = 128,
+ .period_bytes_max = MAX_BUFFER_SIZE,
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static void
+activate_substream(struct snd_usb_caiaqdev *dev,
+ struct snd_pcm_substream *sub)
+{
+ if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dev->sub_playback[sub->number] = sub;
+ else
+ dev->sub_capture[sub->number] = sub;
+}
+
+static void
+deactivate_substream(struct snd_usb_caiaqdev *dev,
+ struct snd_pcm_substream *sub)
+{
+ unsigned long flags;
+ spin_lock_irqsave(&dev->spinlock, flags);
+
+ if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dev->sub_playback[sub->number] = NULL;
+ else
+ dev->sub_capture[sub->number] = NULL;
+
+ spin_unlock_irqrestore(&dev->spinlock, flags);
+}
+
+static int
+all_substreams_zero(struct snd_pcm_substream **subs)
+{
+ int i;
+ for (i = 0; i < MAX_STREAMS; i++)
+ if (subs[i] != NULL)
+ return 0;
+ return 1;
+}
+
+static int stream_start(struct snd_usb_caiaqdev *dev)
+{
+ int i, ret;
+
+ debug("%s(%p)\n", __func__, dev);
+
+ if (dev->streaming)
+ return -EINVAL;
+
+ memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
+ memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
+ dev->input_panic = 0;
+ dev->output_panic = 0;
+ dev->first_packet = 1;
+ dev->streaming = 1;
+
+ for (i = 0; i < N_URBS; i++) {
+ ret = usb_submit_urb(dev->data_urbs_in[i], GFP_ATOMIC);
+ if (ret) {
+ log("unable to trigger read #%d! (ret %d)\n", i, ret);
+ dev->streaming = 0;
+ return -EPIPE;
+ }
+ }
+
+ return 0;
+}
+
+static void stream_stop(struct snd_usb_caiaqdev *dev)
+{
+ int i;
+
+ debug("%s(%p)\n", __func__, dev);
+ if (!dev->streaming)
+ return;
+
+ dev->streaming = 0;
+
+ for (i = 0; i < N_URBS; i++) {
+ usb_kill_urb(dev->data_urbs_in[i]);
+ usb_kill_urb(dev->data_urbs_out[i]);
+ }
+}
+
+static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream)
+{
+ struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream);
+ debug("%s(%p)\n", __func__, substream);
+ substream->runtime->hw = dev->pcm_info;
+ snd_pcm_limit_hw_rates(substream->runtime);
+ return 0;
+}
+
+static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream)
+{
+ struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream);
+
+ debug("%s(%p)\n", __func__, substream);
+ if (all_substreams_zero(dev->sub_playback) &&
+ all_substreams_zero(dev->sub_capture)) {
+ /* when the last client has stopped streaming,
+ * all sample rates are allowed again */
+ stream_stop(dev);
+ dev->pcm_info.rates = dev->samplerates;
+ }
+
+ return 0;
+}
+
+static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub,
+ struct snd_pcm_hw_params *hw_params)
+{
+ debug("%s(%p)\n", __func__, sub);
+ return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params));
+}
+
+static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub)
+{
+ struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub);
+ debug("%s(%p)\n", __func__, sub);
+ deactivate_substream(dev, sub);
+ return snd_pcm_lib_free_pages(sub);
+}
+
+/* this should probably go upstream */
+#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12
+#error "Change this table"
+#endif
+
+static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100,
+ 48000, 64000, 88200, 96000, 176400, 192000 };
+
+static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int bytes_per_sample, bpp, ret, i;
+ int index = substream->number;
+ struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ debug("%s(%p)\n", __func__, substream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
+ else
+ dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE;
+
+ if (dev->streaming)
+ return 0;
+
+ /* the first client that opens a stream defines the sample rate
+ * setting for all subsequent calls, until the last client closed. */
+ for (i=0; i < ARRAY_SIZE(rates); i++)
+ if (runtime->rate == rates[i])
+ dev->pcm_info.rates = 1 << i;
+
+ snd_pcm_limit_hw_rates(runtime);
+
+ bytes_per_sample = BYTES_PER_SAMPLE;
+ if (dev->spec.data_alignment == 2)
+ bytes_per_sample++;
+
+ bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE)
+ * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams;
+
+ if (bpp > MAX_ENDPOINT_SIZE)
+ bpp = MAX_ENDPOINT_SIZE;
+
+ ret = snd_usb_caiaq_set_audio_params(dev, runtime->rate,
+ runtime->sample_bits, bpp);
+ if (ret)
+ return ret;
+
+ ret = stream_start(dev);
+ if (ret)
+ return ret;
+
+ dev->output_running = 0;
+ wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ);
+ if (!dev->output_running) {
+ stream_stop(dev);
+ return -EPIPE;
+ }
+
+ return 0;
+}
+
+static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd)
+{
+ struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ activate_substream(dev, sub);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ deactivate_substream(dev, sub);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
+{
+ int index = sub->number;
+ struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub);
+
+ if (dev->input_panic || dev->output_panic)
+ return SNDRV_PCM_POS_XRUN;
+
+ if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return bytes_to_frames(sub->runtime,
+ dev->audio_out_buf_pos[index]);
+ else
+ return bytes_to_frames(sub->runtime,
+ dev->audio_in_buf_pos[index]);
+}
+
+/* operators for both playback and capture */
+static struct snd_pcm_ops snd_usb_caiaq_ops = {
+ .open = snd_usb_caiaq_substream_open,
+ .close = snd_usb_caiaq_substream_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_usb_caiaq_pcm_hw_params,
+ .hw_free = snd_usb_caiaq_pcm_hw_free,
+ .prepare = snd_usb_caiaq_pcm_prepare,
+ .trigger = snd_usb_caiaq_pcm_trigger,
+ .pointer = snd_usb_caiaq_pcm_pointer
+};
+
+static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev,
+ struct snd_pcm_substream **subs)
+{
+ int stream, pb, *cnt;
+ struct snd_pcm_substream *sub;
+
+ for (stream = 0; stream < dev->n_streams; stream++) {
+ sub = subs[stream];
+ if (!sub)
+ continue;
+
+ pb = frames_to_bytes(sub->runtime,
+ sub->runtime->period_size);
+ cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ &dev->period_out_count[stream] :
+ &dev->period_in_count[stream];
+
+ if (*cnt >= pb) {
+ snd_pcm_period_elapsed(sub);
+ *cnt %= pb;
+ }
+ }
+}
+
+static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev,
+ const struct urb *urb,
+ const struct usb_iso_packet_descriptor *iso)
+{
+ unsigned char *usb_buf = urb->transfer_buffer + iso->offset;
+ struct snd_pcm_substream *sub;
+ int stream, i;
+
+ if (all_substreams_zero(dev->sub_capture))
+ return;
+
+ for (i = 0; i < iso->actual_length;) {
+ for (stream = 0; stream < dev->n_streams; stream++, i++) {
+ sub = dev->sub_capture[stream];
+ if (sub) {
+ struct snd_pcm_runtime *rt = sub->runtime;
+ char *audio_buf = rt->dma_area;
+ int sz = frames_to_bytes(rt, rt->buffer_size);
+ audio_buf[dev->audio_in_buf_pos[stream]++]
+ = usb_buf[i];
+ dev->period_in_count[stream]++;
+ if (dev->audio_in_buf_pos[stream] == sz)
+ dev->audio_in_buf_pos[stream] = 0;
+ }
+ }
+ }
+}
+
+static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev,
+ const struct urb *urb,
+ const struct usb_iso_packet_descriptor *iso)
+{
+ unsigned char *usb_buf = urb->transfer_buffer + iso->offset;
+ unsigned char check_byte;
+ struct snd_pcm_substream *sub;
+ int stream, i;
+
+ for (i = 0; i < iso->actual_length;) {
+ if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) {
+ for (stream = 0;
+ stream < dev->n_streams;
+ stream++, i++) {
+ if (dev->first_packet)
+ continue;
+
+ check_byte = MAKE_CHECKBYTE(dev, stream, i);
+
+ if ((usb_buf[i] & 0x3f) != check_byte)
+ dev->input_panic = 1;
+
+ if (usb_buf[i] & 0x80)
+ dev->output_panic = 1;
+ }
+ }
+ dev->first_packet = 0;
+
+ for (stream = 0; stream < dev->n_streams; stream++, i++) {
+ sub = dev->sub_capture[stream];
+ if (sub) {
+ struct snd_pcm_runtime *rt = sub->runtime;
+ char *audio_buf = rt->dma_area;
+ int sz = frames_to_bytes(rt, rt->buffer_size);
+ audio_buf[dev->audio_in_buf_pos[stream]++] =
+ usb_buf[i];
+ dev->period_in_count[stream]++;
+ if (dev->audio_in_buf_pos[stream] == sz)
+ dev->audio_in_buf_pos[stream] = 0;
+ }
+ }
+ }
+}
+
+static void read_in_urb(struct snd_usb_caiaqdev *dev,
+ const struct urb *urb,
+ const struct usb_iso_packet_descriptor *iso)
+{
+ if (!dev->streaming)
+ return;
+
+ switch (dev->spec.data_alignment) {
+ case 0:
+ read_in_urb_mode0(dev, urb, iso);
+ break;
+ case 2:
+ read_in_urb_mode2(dev, urb, iso);
+ break;
+ }
+
+ if (dev->input_panic || dev->output_panic) {
+ debug("streaming error detected %s %s\n",
+ dev->input_panic ? "(input)" : "",
+ dev->output_panic ? "(output)" : "");
+ }
+}
+
+static void fill_out_urb(struct snd_usb_caiaqdev *dev,
+ struct urb *urb,
+ const struct usb_iso_packet_descriptor *iso)
+{
+ unsigned char *usb_buf = urb->transfer_buffer + iso->offset;
+ struct snd_pcm_substream *sub;
+ int stream, i;
+
+ for (i = 0; i < iso->length;) {
+ for (stream = 0; stream < dev->n_streams; stream++, i++) {
+ sub = dev->sub_playback[stream];
+ if (sub) {
+ struct snd_pcm_runtime *rt = sub->runtime;
+ char *audio_buf = rt->dma_area;
+ int sz = frames_to_bytes(rt, rt->buffer_size);
+ usb_buf[i] =
+ audio_buf[dev->audio_out_buf_pos[stream]];
+ dev->period_out_count[stream]++;
+ dev->audio_out_buf_pos[stream]++;
+ if (dev->audio_out_buf_pos[stream] == sz)
+ dev->audio_out_buf_pos[stream] = 0;
+ } else
+ usb_buf[i] = 0;
+ }
+
+ /* fill in the check bytes */
+ if (dev->spec.data_alignment == 2 &&
+ i % (dev->n_streams * BYTES_PER_SAMPLE_USB) ==
+ (dev->n_streams * CHANNELS_PER_STREAM))
+ for (stream = 0; stream < dev->n_streams; stream++, i++)
+ usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i);
+ }
+}
+
+static void read_completed(struct urb *urb)
+{
+ struct snd_usb_caiaq_cb_info *info = urb->context;
+ struct snd_usb_caiaqdev *dev;
+ struct urb *out;
+ int frame, len, send_it = 0, outframe = 0;
+
+ if (urb->status || !info)
+ return;
+
+ dev = info->dev;
+
+ if (!dev->streaming)
+ return;
+
+ out = dev->data_urbs_out[info->index];
+
+ /* read the recently received packet and send back one which has
+ * the same layout */
+ for (frame = 0; frame < FRAMES_PER_URB; frame++) {
+ if (urb->iso_frame_desc[frame].status)
+ continue;
+
+ len = urb->iso_frame_desc[outframe].actual_length;
+ out->iso_frame_desc[outframe].length = len;
+ out->iso_frame_desc[outframe].actual_length = 0;
+ out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
+
+ if (len > 0) {
+ spin_lock(&dev->spinlock);
+ fill_out_urb(dev, out, &out->iso_frame_desc[outframe]);
+ read_in_urb(dev, urb, &urb->iso_frame_desc[frame]);
+ spin_unlock(&dev->spinlock);
+ check_for_elapsed_periods(dev, dev->sub_playback);
+ check_for_elapsed_periods(dev, dev->sub_capture);
+ send_it = 1;
+ }
+
+ outframe++;
+ }
+
+ if (send_it) {
+ out->number_of_packets = FRAMES_PER_URB;
+ out->transfer_flags = URB_ISO_ASAP;
+ usb_submit_urb(out, GFP_ATOMIC);
+ }
+
+ /* re-submit inbound urb */
+ for (frame = 0; frame < FRAMES_PER_URB; frame++) {
+ urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
+ urb->iso_frame_desc[frame].length = BYTES_PER_FRAME;
+ urb->iso_frame_desc[frame].actual_length = 0;
+ }
+
+ urb->number_of_packets = FRAMES_PER_URB;
+ urb->transfer_flags = URB_ISO_ASAP;
+ usb_submit_urb(urb, GFP_ATOMIC);
+}
+
+static void write_completed(struct urb *urb)
+{
+ struct snd_usb_caiaq_cb_info *info = urb->context;
+ struct snd_usb_caiaqdev *dev = info->dev;
+
+ if (!dev->output_running) {
+ dev->output_running = 1;
+ wake_up(&dev->prepare_wait_queue);
+ }
+}
+
+static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret)
+{
+ int i, frame;
+ struct urb **urbs;
+ struct usb_device *usb_dev = dev->chip.dev;
+ unsigned int pipe;
+
+ pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ?
+ usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) :
+ usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE);
+
+ urbs = kmalloc(N_URBS * sizeof(*urbs), GFP_KERNEL);
+ if (!urbs) {
+ log("unable to kmalloc() urbs, OOM!?\n");
+ *ret = -ENOMEM;
+ return NULL;
+ }
+
+ for (i = 0; i < N_URBS; i++) {
+ urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL);
+ if (!urbs[i]) {
+ log("unable to usb_alloc_urb(), OOM!?\n");
+ *ret = -ENOMEM;
+ return urbs;
+ }
+
+ urbs[i]->transfer_buffer =
+ kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL);
+ if (!urbs[i]->transfer_buffer) {
+ log("unable to kmalloc() transfer buffer, OOM!?\n");
+ *ret = -ENOMEM;
+ return urbs;
+ }
+
+ for (frame = 0; frame < FRAMES_PER_URB; frame++) {
+ struct usb_iso_packet_descriptor *iso =
+ &urbs[i]->iso_frame_desc[frame];
+
+ iso->offset = BYTES_PER_FRAME * frame;
+ iso->length = BYTES_PER_FRAME;
+ }
+
+ urbs[i]->dev = usb_dev;
+ urbs[i]->pipe = pipe;
+ urbs[i]->transfer_buffer_length = FRAMES_PER_URB
+ * BYTES_PER_FRAME;
+ urbs[i]->context = &dev->data_cb_info[i];
+ urbs[i]->interval = 1;
+ urbs[i]->transfer_flags = URB_ISO_ASAP;
+ urbs[i]->number_of_packets = FRAMES_PER_URB;
+ urbs[i]->complete = (dir == SNDRV_PCM_STREAM_CAPTURE) ?
+ read_completed : write_completed;
+ }
+
+ *ret = 0;
+ return urbs;
+}
+
+static void free_urbs(struct urb **urbs)
+{
+ int i;
+
+ if (!urbs)
+ return;
+
+ for (i = 0; i < N_URBS; i++) {
+ if (!urbs[i])
+ continue;
+
+ usb_kill_urb(urbs[i]);
+ kfree(urbs[i]->transfer_buffer);
+ usb_free_urb(urbs[i]);
+ }
+
+ kfree(urbs);
+}
+
+int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
+{
+ int i, ret;
+
+ dev->n_audio_in = max(dev->spec.num_analog_audio_in,
+ dev->spec.num_digital_audio_in) /
+ CHANNELS_PER_STREAM;
+ dev->n_audio_out = max(dev->spec.num_analog_audio_out,
+ dev->spec.num_digital_audio_out) /
+ CHANNELS_PER_STREAM;
+ dev->n_streams = max(dev->n_audio_in, dev->n_audio_out);
+
+ debug("dev->n_audio_in = %d\n", dev->n_audio_in);
+ debug("dev->n_audio_out = %d\n", dev->n_audio_out);
+ debug("dev->n_streams = %d\n", dev->n_streams);
+
+ if (dev->n_streams > MAX_STREAMS) {
+ log("unable to initialize device, too many streams.\n");
+ return -EINVAL;
+ }
+
+ ret = snd_pcm_new(dev->chip.card, dev->product_name, 0,
+ dev->n_audio_out, dev->n_audio_in, &dev->pcm);
+
+ if (ret < 0) {
+ log("snd_pcm_new() returned %d\n", ret);
+ return ret;
+ }
+
+ dev->pcm->private_data = dev;
+ strcpy(dev->pcm->name, dev->product_name);
+
+ memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
+ memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
+
+ memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware,
+ sizeof(snd_usb_caiaq_pcm_hardware));
+
+ /* setup samplerates */
+ dev->samplerates = dev->pcm_info.rates;
+ switch (dev->chip.usb_id) {
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO):
+ dev->samplerates |= SNDRV_PCM_RATE_88200;
+ dev->samplerates |= SNDRV_PCM_RATE_192000;
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
+ dev->samplerates |= SNDRV_PCM_RATE_88200;
+ break;
+ }
+
+ snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_usb_caiaq_ops);
+ snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_usb_caiaq_ops);
+
+ snd_pcm_lib_preallocate_pages_for_all(dev->pcm,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ MAX_BUFFER_SIZE, MAX_BUFFER_SIZE);
+
+ dev->data_cb_info =
+ kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS,
+ GFP_KERNEL);
+
+ if (!dev->data_cb_info)
+ return -ENOMEM;
+
+ for (i = 0; i < N_URBS; i++) {
+ dev->data_cb_info[i].dev = dev;
+ dev->data_cb_info[i].index = i;
+ }
+
+ dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret);
+ if (ret < 0) {
+ kfree(dev->data_cb_info);
+ free_urbs(dev->data_urbs_in);
+ return ret;
+ }
+
+ dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret);
+ if (ret < 0) {
+ kfree(dev->data_cb_info);
+ free_urbs(dev->data_urbs_in);
+ free_urbs(dev->data_urbs_out);
+ return ret;
+ }
+
+ return 0;
+}
+
+void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *dev)
+{
+ debug("%s(%p)\n", __func__, dev);
+ stream_stop(dev);
+ free_urbs(dev->data_urbs_in);
+ free_urbs(dev->data_urbs_out);
+ kfree(dev->data_cb_info);
+}
+
diff --git a/sound/usb/caiaq/caiaq-audio.h b/sound/usb/caiaq/caiaq-audio.h
new file mode 100644
index 0000000..8ab1f8d
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-audio.h
@@ -0,0 +1,7 @@
+#ifndef CAIAQ_AUDIO_H
+#define CAIAQ_AUDIO_H
+
+int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev);
+void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *dev);
+
+#endif /* CAIAQ_AUDIO_H */
diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c
new file mode 100644
index 0000000..798ca12
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-control.c
@@ -0,0 +1,315 @@
+/*
+ * Copyright (c) 2007 Daniel Mack
+ * friendly supported by NI.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/usb.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/control.h>
+#include <linux/input.h>
+
+#include "caiaq-device.h"
+#include "caiaq-control.h"
+
+#define CNT_INTVAL 0x10000
+
+static int control_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol);
+ struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
+ int pos = kcontrol->private_value;
+ int is_intval = pos & CNT_INTVAL;
+
+ uinfo->count = 1;
+ pos &= ~CNT_INTVAL;
+
+ if (dev->chip.usb_id ==
+ USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)
+ && (pos == 0)) {
+ /* current input mode of A8DJ */
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 2;
+ return 0;
+ }
+
+ if (is_intval) {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 64;
+ } else {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ }
+
+ return 0;
+}
+
+static int control_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol);
+ struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
+ int pos = kcontrol->private_value;
+
+ if (pos & CNT_INTVAL)
+ ucontrol->value.integer.value[0]
+ = dev->control_state[pos & ~CNT_INTVAL];
+ else
+ ucontrol->value.integer.value[0]
+ = !!(dev->control_state[pos / 8] & (1 << pos % 8));
+
+ return 0;
+}
+
+static int control_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol);
+ struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
+ int pos = kcontrol->private_value;
+
+ if (pos & CNT_INTVAL) {
+ dev->control_state[pos & ~CNT_INTVAL]
+ = ucontrol->value.integer.value[0];
+ snd_usb_caiaq_send_command(dev, EP1_CMD_DIMM_LEDS,
+ dev->control_state, sizeof(dev->control_state));
+ } else {
+ if (ucontrol->value.integer.value[0])
+ dev->control_state[pos / 8] |= 1 << (pos % 8);
+ else
+ dev->control_state[pos / 8] &= ~(1 << (pos % 8));
+
+ snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO,
+ dev->control_state, sizeof(dev->control_state));
+ }
+
+ return 1;
+}
+
+static struct snd_kcontrol_new kcontrol_template __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .index = 0,
+ .info = control_info,
+ .get = control_get,
+ .put = control_put,
+ /* name and private_value filled later */
+};
+
+struct caiaq_controller {
+ char *name;
+ int index;
+};
+
+static struct caiaq_controller ak1_controller[] = {
+ { "LED left", 2 },
+ { "LED middle", 1 },
+ { "LED right", 0 },
+ { "LED ring", 3 }
+};
+
+static struct caiaq_controller rk2_controller[] = {
+ { "LED 1", 5 },
+ { "LED 2", 4 },
+ { "LED 3", 3 },
+ { "LED 4", 2 },
+ { "LED 5", 1 },
+ { "LED 6", 0 },
+ { "LED pedal", 6 },
+ { "LED 7seg_1b", 8 },
+ { "LED 7seg_1c", 9 },
+ { "LED 7seg_2a", 10 },
+ { "LED 7seg_2b", 11 },
+ { "LED 7seg_2c", 12 },
+ { "LED 7seg_2d", 13 },
+ { "LED 7seg_2e", 14 },
+ { "LED 7seg_2f", 15 },
+ { "LED 7seg_2g", 16 },
+ { "LED 7seg_3a", 17 },
+ { "LED 7seg_3b", 18 },
+ { "LED 7seg_3c", 19 },
+ { "LED 7seg_3d", 20 },
+ { "LED 7seg_3e", 21 },
+ { "LED 7seg_3f", 22 },
+ { "LED 7seg_3g", 23 }
+};
+
+static struct caiaq_controller rk3_controller[] = {
+ { "LED 7seg_1a", 0 + 0 },
+ { "LED 7seg_1b", 0 + 1 },
+ { "LED 7seg_1c", 0 + 2 },
+ { "LED 7seg_1d", 0 + 3 },
+ { "LED 7seg_1e", 0 + 4 },
+ { "LED 7seg_1f", 0 + 5 },
+ { "LED 7seg_1g", 0 + 6 },
+ { "LED 7seg_1p", 0 + 7 },
+
+ { "LED 7seg_2a", 8 + 0 },
+ { "LED 7seg_2b", 8 + 1 },
+ { "LED 7seg_2c", 8 + 2 },
+ { "LED 7seg_2d", 8 + 3 },
+ { "LED 7seg_2e", 8 + 4 },
+ { "LED 7seg_2f", 8 + 5 },
+ { "LED 7seg_2g", 8 + 6 },
+ { "LED 7seg_2p", 8 + 7 },
+
+ { "LED 7seg_3a", 16 + 0 },
+ { "LED 7seg_3b", 16 + 1 },
+ { "LED 7seg_3c", 16 + 2 },
+ { "LED 7seg_3d", 16 + 3 },
+ { "LED 7seg_3e", 16 + 4 },
+ { "LED 7seg_3f", 16 + 5 },
+ { "LED 7seg_3g", 16 + 6 },
+ { "LED 7seg_3p", 16 + 7 },
+
+ { "LED 7seg_4a", 24 + 0 },
+ { "LED 7seg_4b", 24 + 1 },
+ { "LED 7seg_4c", 24 + 2 },
+ { "LED 7seg_4d", 24 + 3 },
+ { "LED 7seg_4e", 24 + 4 },
+ { "LED 7seg_4f", 24 + 5 },
+ { "LED 7seg_4g", 24 + 6 },
+ { "LED 7seg_4p", 24 + 7 },
+
+ { "LED 1", 32 + 0 },
+ { "LED 2", 32 + 1 },
+ { "LED 3", 32 + 2 },
+ { "LED 4", 32 + 3 },
+ { "LED 5", 32 + 4 },
+ { "LED 6", 32 + 5 },
+ { "LED 7", 32 + 6 },
+ { "LED 8", 32 + 7 },
+ { "LED pedal", 32 + 8 }
+};
+
+static struct caiaq_controller kore_controller[] = {
+ { "LED F1", 8 | CNT_INTVAL },
+ { "LED F2", 12 | CNT_INTVAL },
+ { "LED F3", 0 | CNT_INTVAL },
+ { "LED F4", 4 | CNT_INTVAL },
+ { "LED F5", 11 | CNT_INTVAL },
+ { "LED F6", 15 | CNT_INTVAL },
+ { "LED F7", 3 | CNT_INTVAL },
+ { "LED F8", 7 | CNT_INTVAL },
+ { "LED touch1", 10 | CNT_INTVAL },
+ { "LED touch2", 14 | CNT_INTVAL },
+ { "LED touch3", 2 | CNT_INTVAL },
+ { "LED touch4", 6 | CNT_INTVAL },
+ { "LED touch5", 9 | CNT_INTVAL },
+ { "LED touch6", 13 | CNT_INTVAL },
+ { "LED touch7", 1 | CNT_INTVAL },
+ { "LED touch8", 5 | CNT_INTVAL },
+ { "LED left", 18 | CNT_INTVAL },
+ { "LED right", 22 | CNT_INTVAL },
+ { "LED up", 16 | CNT_INTVAL },
+ { "LED down", 20 | CNT_INTVAL },
+ { "LED stop", 23 | CNT_INTVAL },
+ { "LED play", 21 | CNT_INTVAL },
+ { "LED record", 19 | CNT_INTVAL },
+ { "LED listen", 17 | CNT_INTVAL },
+ { "LED lcd", 30 | CNT_INTVAL },
+ { "LED menu", 28 | CNT_INTVAL },
+ { "LED sound", 31 | CNT_INTVAL },
+ { "LED esc", 29 | CNT_INTVAL },
+ { "LED view", 27 | CNT_INTVAL },
+ { "LED enter", 24 | CNT_INTVAL },
+ { "LED control", 26 | CNT_INTVAL }
+};
+
+static struct caiaq_controller a8dj_controller[] = {
+ { "Current input mode", 0 | CNT_INTVAL },
+ { "GND lift for TC Vinyl mode", 24 + 0 },
+ { "GND lift for TC CD/Line mode", 24 + 1 },
+ { "GND lift for phono mode", 24 + 2 },
+ { "GND lift for TC Vinyl mode", 24 + 3 },
+ { "Software lock", 40 }
+};
+
+int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev)
+{
+ int i;
+ struct snd_kcontrol *kc;
+
+ switch (dev->chip.usb_id) {
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
+ for (i = 0; i < ARRAY_SIZE(ak1_controller); i++) {
+ struct caiaq_controller *c = ak1_controller + i;
+ kcontrol_template.name = c->name;
+ kcontrol_template.private_value = c->index;
+ kc = snd_ctl_new1(&kcontrol_template, dev);
+ snd_ctl_add(dev->chip.card, kc);
+ }
+
+ break;
+
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2):
+ for (i = 0; i < ARRAY_SIZE(rk2_controller); i++) {
+ struct caiaq_controller *c = rk2_controller + i;
+ kcontrol_template.name = c->name;
+ kcontrol_template.private_value = c->index;
+ kc = snd_ctl_new1(&kcontrol_template, dev);
+ snd_ctl_add(dev->chip.card, kc);
+ }
+
+ break;
+
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
+ for (i = 0; i < ARRAY_SIZE(rk3_controller); i++) {
+ struct caiaq_controller *c = rk3_controller + i;
+ kcontrol_template.name = c->name;
+ kcontrol_template.private_value = c->index;
+ kc = snd_ctl_new1(&kcontrol_template, dev);
+ snd_ctl_add(dev->chip.card, kc);
+ }
+
+ break;
+
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER):
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2):
+ for (i = 0; i < ARRAY_SIZE(kore_controller); i++) {
+ struct caiaq_controller *c = kore_controller + i;
+ kcontrol_template.name = c->name;
+ kcontrol_template.private_value = c->index;
+ kc = snd_ctl_new1(&kcontrol_template, dev);
+ snd_ctl_add(dev->chip.card, kc);
+ }
+
+ break;
+
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
+ for (i = 0; i < ARRAY_SIZE(a8dj_controller); i++) {
+ struct caiaq_controller *c = a8dj_controller + i;
+ kcontrol_template.name = c->name;
+ kcontrol_template.private_value = c->index;
+ kc = snd_ctl_new1(&kcontrol_template, dev);
+ snd_ctl_add(dev->chip.card, kc);
+ }
+
+ break;
+ }
+
+ return 0;
+}
+
diff --git a/sound/usb/caiaq/caiaq-control.h b/sound/usb/caiaq/caiaq-control.h
new file mode 100644
index 0000000..2e7ab1a
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-control.h
@@ -0,0 +1,6 @@
+#ifndef CAIAQ_CONTROL_H
+#define CAIAQ_CONTROL_H
+
+int snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev);
+
+#endif /* CAIAQ_CONTROL_H */
diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
new file mode 100644
index 0000000..8317508
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-device.c
@@ -0,0 +1,506 @@
+/*
+ * caiaq.c: ALSA driver for caiaq/NativeInstruments devices
+ *
+ * Copyright (c) 2007 Daniel Mack <daniel@caiaq.de>
+ * Karsten Wiese <fzu@wemgehoertderstaat.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/usb.h>
+#include <linux/input.h>
+#include <linux/spinlock.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/control.h>
+
+#include "caiaq-device.h"
+#include "caiaq-audio.h"
+#include "caiaq-midi.h"
+#include "caiaq-control.h"
+
+#ifdef CONFIG_SND_USB_CAIAQ_INPUT
+#include "caiaq-input.h"
+#endif
+
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.8");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
+ "{Native Instruments, RigKontrol3},"
+ "{Native Instruments, Kore Controller},"
+ "{Native Instruments, Kore Controller 2},"
+ "{Native Instruments, Audio Kontrol 1},"
+ "{Native Instruments, Audio 8 DJ},"
+ "{Native Instruments, Session I/O}}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
+static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+static int snd_card_used[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the caiaq sound device");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the caiaq soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable the caiaq soundcard.");
+
+enum {
+ SAMPLERATE_44100 = 0,
+ SAMPLERATE_48000 = 1,
+ SAMPLERATE_96000 = 2,
+ SAMPLERATE_192000 = 3,
+ SAMPLERATE_88200 = 4,
+ SAMPLERATE_INVALID = 0xff
+};
+
+enum {
+ DEPTH_NONE = 0,
+ DEPTH_16 = 1,
+ DEPTH_24 = 2,
+ DEPTH_32 = 3
+};
+
+static struct usb_device_id snd_usb_id_table[] = {
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_RIGKONTROL2
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_RIGKONTROL3
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_KORECONTROLLER
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_KORECONTROLLER2
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_AK1
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_AUDIO8DJ
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = USB_VID_NATIVEINSTRUMENTS,
+ .idProduct = USB_PID_SESSIONIO
+ },
+ { /* terminator */ }
+};
+
+static void usb_ep1_command_reply_dispatch (struct urb* urb)
+{
+ int ret;
+ struct snd_usb_caiaqdev *dev = urb->context;
+ unsigned char *buf = urb->transfer_buffer;
+
+ if (urb->status || !dev) {
+ log("received EP1 urb->status = %i\n", urb->status);
+ return;
+ }
+
+ switch(buf[0]) {
+ case EP1_CMD_GET_DEVICE_INFO:
+ memcpy(&dev->spec, buf+1, sizeof(struct caiaq_device_spec));
+ dev->spec.fw_version = le16_to_cpu(dev->spec.fw_version);
+ debug("device spec (firmware %d): audio: %d in, %d out, "
+ "MIDI: %d in, %d out, data alignment %d\n",
+ dev->spec.fw_version,
+ dev->spec.num_analog_audio_in,
+ dev->spec.num_analog_audio_out,
+ dev->spec.num_midi_in,
+ dev->spec.num_midi_out,
+ dev->spec.data_alignment);
+
+ dev->spec_received++;
+ wake_up(&dev->ep1_wait_queue);
+ break;
+ case EP1_CMD_AUDIO_PARAMS:
+ dev->audio_parm_answer = buf[1];
+ wake_up(&dev->ep1_wait_queue);
+ break;
+ case EP1_CMD_MIDI_READ:
+ snd_usb_caiaq_midi_handle_input(dev, buf[1], buf + 3, buf[2]);
+ break;
+ case EP1_CMD_READ_IO:
+ if (dev->chip.usb_id ==
+ USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) {
+ if (urb->actual_length > sizeof(dev->control_state))
+ urb->actual_length = sizeof(dev->control_state);
+ memcpy(dev->control_state, buf + 1, urb->actual_length);
+ wake_up(&dev->ep1_wait_queue);
+ break;
+ }
+#ifdef CONFIG_SND_USB_CAIAQ_INPUT
+ case EP1_CMD_READ_ERP:
+ case EP1_CMD_READ_ANALOG:
+ snd_usb_caiaq_input_dispatch(dev, buf, urb->actual_length);
+#endif
+ break;
+ }
+
+ dev->ep1_in_urb.actual_length = 0;
+ ret = usb_submit_urb(&dev->ep1_in_urb, GFP_ATOMIC);
+ if (ret < 0)
+ log("unable to submit urb. OOM!?\n");
+}
+
+int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev,
+ unsigned char command,
+ const unsigned char *buffer,
+ int len)
+{
+ int actual_len;
+ struct usb_device *usb_dev = dev->chip.dev;
+
+ if (!usb_dev)
+ return -EIO;
+
+ if (len > EP1_BUFSIZE - 1)
+ len = EP1_BUFSIZE - 1;
+
+ if (buffer && len > 0)
+ memcpy(dev->ep1_out_buf+1, buffer, len);
+
+ dev->ep1_out_buf[0] = command;
+ return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1),
+ dev->ep1_out_buf, len+1, &actual_len, 200);
+}
+
+int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev,
+ int rate, int depth, int bpp)
+{
+ int ret;
+ char tmp[5];
+
+ switch (rate) {
+ case 44100: tmp[0] = SAMPLERATE_44100; break;
+ case 48000: tmp[0] = SAMPLERATE_48000; break;
+ case 88200: tmp[0] = SAMPLERATE_88200; break;
+ case 96000: tmp[0] = SAMPLERATE_96000; break;
+ case 192000: tmp[0] = SAMPLERATE_192000; break;
+ default: return -EINVAL;
+ }
+
+ switch (depth) {
+ case 16: tmp[1] = DEPTH_16; break;
+ case 24: tmp[1] = DEPTH_24; break;
+ default: return -EINVAL;
+ }
+
+ tmp[2] = bpp & 0xff;
+ tmp[3] = bpp >> 8;
+ tmp[4] = 1; /* packets per microframe */
+
+ debug("setting audio params: %d Hz, %d bits, %d bpp\n",
+ rate, depth, bpp);
+
+ dev->audio_parm_answer = -1;
+ ret = snd_usb_caiaq_send_command(dev, EP1_CMD_AUDIO_PARAMS,
+ tmp, sizeof(tmp));
+
+ if (ret)
+ return ret;
+
+ if (!wait_event_timeout(dev->ep1_wait_queue,
+ dev->audio_parm_answer >= 0, HZ))
+ return -EPIPE;
+
+ if (dev->audio_parm_answer != 1)
+ debug("unable to set the device's audio params\n");
+
+ return dev->audio_parm_answer == 1 ? 0 : -EINVAL;
+}
+
+int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev,
+ int digital, int analog, int erp)
+{
+ char tmp[3] = { digital, analog, erp };
+ return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG,
+ tmp, sizeof(tmp));
+}
+
+static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
+{
+ int ret;
+ char val[4];
+
+ /* device-specific startup specials */
+ switch (dev->chip.usb_id) {
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2):
+ /* RigKontrol2 - display centered dash ('-') */
+ val[0] = 0x00;
+ val[1] = 0x00;
+ val[2] = 0x01;
+ snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 3);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
+ /* RigKontrol2 - display two centered dashes ('--') */
+ val[0] = 0x00;
+ val[1] = 0x40;
+ val[2] = 0x40;
+ val[3] = 0x00;
+ snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 4);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
+ /* Audio Kontrol 1 - make USB-LED stop blinking */
+ val[0] = 0x00;
+ snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, val, 1);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
+ /* Audio 8 DJ - trigger read of current settings */
+ dev->control_state[0] = 0xff;
+ snd_usb_caiaq_set_auto_msg(dev, 1, 0, 0);
+ snd_usb_caiaq_send_command(dev, EP1_CMD_READ_IO, NULL, 0);
+
+ if (!wait_event_timeout(dev->ep1_wait_queue,
+ dev->control_state[0] != 0xff, HZ))
+ return;
+
+ /* fix up some defaults */
+ if ((dev->control_state[1] != 2) ||
+ (dev->control_state[2] != 3) ||
+ (dev->control_state[4] != 2)) {
+ dev->control_state[1] = 2;
+ dev->control_state[2] = 3;
+ dev->control_state[4] = 2;
+ snd_usb_caiaq_send_command(dev,
+ EP1_CMD_WRITE_IO, dev->control_state, 6);
+ }
+
+ break;
+ }
+
+ if (dev->spec.num_analog_audio_out +
+ dev->spec.num_analog_audio_in +
+ dev->spec.num_digital_audio_out +
+ dev->spec.num_digital_audio_in > 0) {
+ ret = snd_usb_caiaq_audio_init(dev);
+ if (ret < 0)
+ log("Unable to set up audio system (ret=%d)\n", ret);
+ }
+
+ if (dev->spec.num_midi_in +
+ dev->spec.num_midi_out > 0) {
+ ret = snd_usb_caiaq_midi_init(dev);
+ if (ret < 0)
+ log("Unable to set up MIDI system (ret=%d)\n", ret);
+ }
+
+#ifdef CONFIG_SND_USB_CAIAQ_INPUT
+ ret = snd_usb_caiaq_input_init(dev);
+ if (ret < 0)
+ log("Unable to set up input system (ret=%d)\n", ret);
+#endif
+
+ /* finally, register the card and all its sub-instances */
+ ret = snd_card_register(dev->chip.card);
+ if (ret < 0) {
+ log("snd_card_register() returned %d\n", ret);
+ snd_card_free(dev->chip.card);
+ }
+
+ ret = snd_usb_caiaq_control_init(dev);
+ if (ret < 0)
+ log("Unable to set up control system (ret=%d)\n", ret);
+}
+
+static struct snd_card* create_card(struct usb_device* usb_dev)
+{
+ int devnum;
+ struct snd_card *card;
+ struct snd_usb_caiaqdev *dev;
+
+ for (devnum = 0; devnum < SNDRV_CARDS; devnum++)
+ if (enable[devnum] && !snd_card_used[devnum])
+ break;
+
+ if (devnum >= SNDRV_CARDS)
+ return NULL;
+
+ card = snd_card_new(index[devnum], id[devnum], THIS_MODULE,
+ sizeof(struct snd_usb_caiaqdev));
+ if (!card)
+ return NULL;
+
+ dev = caiaqdev(card);
+ dev->chip.dev = usb_dev;
+ dev->chip.card = card;
+ dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor),
+ le16_to_cpu(usb_dev->descriptor.idProduct));
+ spin_lock_init(&dev->spinlock);
+ snd_card_set_dev(card, &usb_dev->dev);
+
+ return card;
+}
+
+static int __devinit init_card(struct snd_usb_caiaqdev *dev)
+{
+ char *c;
+ struct usb_device *usb_dev = dev->chip.dev;
+ struct snd_card *card = dev->chip.card;
+ int err, len;
+
+ if (usb_set_interface(usb_dev, 0, 1) != 0) {
+ log("can't set alt interface.\n");
+ return -EIO;
+ }
+
+ usb_init_urb(&dev->ep1_in_urb);
+ usb_init_urb(&dev->midi_out_urb);
+
+ usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev,
+ usb_rcvbulkpipe(usb_dev, 0x1),
+ dev->ep1_in_buf, EP1_BUFSIZE,
+ usb_ep1_command_reply_dispatch, dev);
+
+ usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev,
+ usb_sndbulkpipe(usb_dev, 0x1),
+ dev->midi_out_buf, EP1_BUFSIZE,
+ snd_usb_caiaq_midi_output_done, dev);
+
+ init_waitqueue_head(&dev->ep1_wait_queue);
+ init_waitqueue_head(&dev->prepare_wait_queue);
+
+ if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0)
+ return -EIO;
+
+ err = snd_usb_caiaq_send_command(dev, EP1_CMD_GET_DEVICE_INFO, NULL, 0);
+ if (err)
+ return err;
+
+ if (!wait_event_timeout(dev->ep1_wait_queue, dev->spec_received, HZ))
+ return -ENODEV;
+
+ usb_string(usb_dev, usb_dev->descriptor.iManufacturer,
+ dev->vendor_name, CAIAQ_USB_STR_LEN);
+
+ usb_string(usb_dev, usb_dev->descriptor.iProduct,
+ dev->product_name, CAIAQ_USB_STR_LEN);
+
+ usb_string(usb_dev, usb_dev->descriptor.iSerialNumber,
+ dev->serial, CAIAQ_USB_STR_LEN);
+
+ /* terminate serial string at first white space occurence */
+ c = strchr(dev->serial, ' ');
+ if (c)
+ *c = '\0';
+
+ strcpy(card->driver, MODNAME);
+ strcpy(card->shortname, dev->product_name);
+
+ len = snprintf(card->longname, sizeof(card->longname),
+ "%s %s (serial %s, ",
+ dev->vendor_name, dev->product_name, dev->serial);
+
+ if (len < sizeof(card->longname) - 2)
+ len += usb_make_path(usb_dev, card->longname + len,
+ sizeof(card->longname) - len);
+
+ card->longname[len++] = ')';
+ card->longname[len] = '\0';
+ setup_card(dev);
+ return 0;
+}
+
+static int __devinit snd_probe(struct usb_interface *intf,
+ const struct usb_device_id *id)
+{
+ int ret;
+ struct snd_card *card;
+ struct usb_device *device = interface_to_usbdev(intf);
+
+ card = create_card(device);
+
+ if (!card)
+ return -ENOMEM;
+
+ dev_set_drvdata(&intf->dev, card);
+ ret = init_card(caiaqdev(card));
+ if (ret < 0) {
+ log("unable to init card! (ret=%d)\n", ret);
+ snd_card_free(card);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void snd_disconnect(struct usb_interface *intf)
+{
+ struct snd_usb_caiaqdev *dev;
+ struct snd_card *card = dev_get_drvdata(&intf->dev);
+
+ debug("%s(%p)\n", __func__, intf);
+
+ if (!card)
+ return;
+
+ dev = caiaqdev(card);
+ snd_card_disconnect(card);
+
+#ifdef CONFIG_SND_USB_CAIAQ_INPUT
+ snd_usb_caiaq_input_free(dev);
+#endif
+ snd_usb_caiaq_audio_free(dev);
+
+ usb_kill_urb(&dev->ep1_in_urb);
+ usb_kill_urb(&dev->midi_out_urb);
+
+ snd_card_free(card);
+ usb_reset_device(interface_to_usbdev(intf));
+}
+
+
+MODULE_DEVICE_TABLE(usb, snd_usb_id_table);
+static struct usb_driver snd_usb_driver = {
+ .name = MODNAME,
+ .probe = snd_probe,
+ .disconnect = snd_disconnect,
+ .id_table = snd_usb_id_table,
+};
+
+static int __init snd_module_init(void)
+{
+ return usb_register(&snd_usb_driver);
+}
+
+static void __exit snd_module_exit(void)
+{
+ usb_deregister(&snd_usb_driver);
+}
+
+module_init(snd_module_init)
+module_exit(snd_module_exit)
+
diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h
new file mode 100644
index 0000000..ab56e73
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-device.h
@@ -0,0 +1,129 @@
+#ifndef CAIAQ_DEVICE_H
+#define CAIAQ_DEVICE_H
+
+#include "../usbaudio.h"
+
+#define USB_VID_NATIVEINSTRUMENTS 0x17cc
+
+#define USB_PID_RIGKONTROL2 0x1969
+#define USB_PID_RIGKONTROL3 0x1940
+#define USB_PID_KORECONTROLLER 0x4711
+#define USB_PID_KORECONTROLLER2 0x4712
+#define USB_PID_AK1 0x0815
+#define USB_PID_AUDIO8DJ 0x1978
+#define USB_PID_SESSIONIO 0x1915
+
+#define EP1_BUFSIZE 64
+#define CAIAQ_USB_STR_LEN 0xff
+#define MAX_STREAMS 32
+
+//#define SND_USB_CAIAQ_DEBUG
+
+#define MODNAME "snd-usb-caiaq"
+#define log(x...) snd_printk(KERN_WARNING MODNAME" log: " x)
+
+#ifdef SND_USB_CAIAQ_DEBUG
+#define debug(x...) snd_printk(KERN_WARNING MODNAME " debug: " x)
+#else
+#define debug(x...) do { } while(0)
+#endif
+
+#define EP1_CMD_GET_DEVICE_INFO 0x1
+#define EP1_CMD_READ_ERP 0x2
+#define EP1_CMD_READ_ANALOG 0x3
+#define EP1_CMD_READ_IO 0x4
+#define EP1_CMD_WRITE_IO 0x5
+#define EP1_CMD_MIDI_READ 0x6
+#define EP1_CMD_MIDI_WRITE 0x7
+#define EP1_CMD_AUDIO_PARAMS 0x9
+#define EP1_CMD_AUTO_MSG 0xb
+#define EP1_CMD_DIMM_LEDS 0xc
+
+struct caiaq_device_spec {
+ unsigned short fw_version;
+ unsigned char hw_subtype;
+ unsigned char num_erp;
+ unsigned char num_analog_in;
+ unsigned char num_digital_in;
+ unsigned char num_digital_out;
+ unsigned char num_analog_audio_out;
+ unsigned char num_analog_audio_in;
+ unsigned char num_digital_audio_out;
+ unsigned char num_digital_audio_in;
+ unsigned char num_midi_out;
+ unsigned char num_midi_in;
+ unsigned char data_alignment;
+} __attribute__ ((packed));
+
+struct snd_usb_caiaq_cb_info;
+
+struct snd_usb_caiaqdev {
+ struct snd_usb_audio chip;
+
+ struct urb ep1_in_urb;
+ struct urb midi_out_urb;
+ struct urb **data_urbs_in;
+ struct urb **data_urbs_out;
+ struct snd_usb_caiaq_cb_info *data_cb_info;
+
+ unsigned char ep1_in_buf[EP1_BUFSIZE];
+ unsigned char ep1_out_buf[EP1_BUFSIZE];
+ unsigned char midi_out_buf[EP1_BUFSIZE];
+
+ struct caiaq_device_spec spec;
+ spinlock_t spinlock;
+ wait_queue_head_t ep1_wait_queue;
+ wait_queue_head_t prepare_wait_queue;
+ int spec_received, audio_parm_answer;
+ int midi_out_active;
+
+ char vendor_name[CAIAQ_USB_STR_LEN];
+ char product_name[CAIAQ_USB_STR_LEN];
+ char serial[CAIAQ_USB_STR_LEN];
+
+ int n_streams, n_audio_in, n_audio_out;
+ int streaming, first_packet, output_running;
+ int audio_in_buf_pos[MAX_STREAMS];
+ int audio_out_buf_pos[MAX_STREAMS];
+ int period_in_count[MAX_STREAMS];
+ int period_out_count[MAX_STREAMS];
+ int input_panic, output_panic;
+ char *audio_in_buf, *audio_out_buf;
+ unsigned int samplerates;
+
+ struct snd_pcm_substream *sub_playback[MAX_STREAMS];
+ struct snd_pcm_substream *sub_capture[MAX_STREAMS];
+
+ /* Controls */
+ unsigned char control_state[64];
+
+ /* Linux input */
+#ifdef CONFIG_SND_USB_CAIAQ_INPUT
+ struct input_dev *input_dev;
+ char phys[64]; /* physical device path */
+ unsigned short keycode[64];
+#endif
+
+ /* ALSA */
+ struct snd_pcm *pcm;
+ struct snd_pcm_hardware pcm_info;
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_substream *midi_receive_substream;
+ struct snd_rawmidi_substream *midi_out_substream;
+};
+
+struct snd_usb_caiaq_cb_info {
+ struct snd_usb_caiaqdev *dev;
+ int index;
+};
+
+#define caiaqdev(c) ((struct snd_usb_caiaqdev*)(c)->private_data)
+
+int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, int rate, int depth, int bbp);
+int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, int digital, int analog, int erp);
+int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev,
+ unsigned char command,
+ const unsigned char *buffer,
+ int len);
+
+#endif /* CAIAQ_DEVICE_H */
diff --git a/sound/usb/caiaq/caiaq-input.c b/sound/usb/caiaq/caiaq-input.c
new file mode 100644
index 0000000..f743847
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-input.c
@@ -0,0 +1,363 @@
+/*
+ * Copyright (c) 2006,2007 Daniel Mack, Tim Ruetz
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/input.h>
+#include <linux/usb.h>
+#include <linux/usb/input.h>
+#include <linux/spinlock.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+#include <sound/pcm.h>
+#include "caiaq-device.h"
+#include "caiaq-input.h"
+
+static unsigned short keycode_ak1[] = { KEY_C, KEY_B, KEY_A };
+static unsigned short keycode_rk2[] = { KEY_1, KEY_2, KEY_3, KEY_4,
+ KEY_5, KEY_6, KEY_7 };
+static unsigned short keycode_rk3[] = { KEY_1, KEY_2, KEY_3, KEY_4,
+ KEY_5, KEY_6, KEY_7, KEY_5, KEY_6 };
+
+static unsigned short keycode_kore[] = {
+ KEY_FN_F1, /* "menu" */
+ KEY_FN_F7, /* "lcd backlight */
+ KEY_FN_F2, /* "control" */
+ KEY_FN_F3, /* "enter" */
+ KEY_FN_F4, /* "view" */
+ KEY_FN_F5, /* "esc" */
+ KEY_FN_F6, /* "sound" */
+ KEY_FN_F8, /* array spacer, never triggered. */
+ KEY_RIGHT,
+ KEY_DOWN,
+ KEY_UP,
+ KEY_LEFT,
+ KEY_SOUND, /* "listen" */
+ KEY_RECORD,
+ KEY_PLAYPAUSE,
+ KEY_STOP,
+ BTN_4, /* 8 softkeys */
+ BTN_3,
+ BTN_2,
+ BTN_1,
+ BTN_8,
+ BTN_7,
+ BTN_6,
+ BTN_5,
+ KEY_BRL_DOT4, /* touch sensitive knobs */
+ KEY_BRL_DOT3,
+ KEY_BRL_DOT2,
+ KEY_BRL_DOT1,
+ KEY_BRL_DOT8,
+ KEY_BRL_DOT7,
+ KEY_BRL_DOT6,
+ KEY_BRL_DOT5
+};
+
+#define DEG90 (range / 2)
+#define DEG180 (range)
+#define DEG270 (DEG90 + DEG180)
+#define DEG360 (DEG180 * 2)
+#define HIGH_PEAK (268)
+#define LOW_PEAK (-7)
+
+/* some of these devices have endless rotation potentiometers
+ * built in which use two tapers, 90 degrees phase shifted.
+ * this algorithm decodes them to one single value, ranging
+ * from 0 to 999 */
+static unsigned int decode_erp(unsigned char a, unsigned char b)
+{
+ int weight_a, weight_b;
+ int pos_a, pos_b;
+ int ret;
+ int range = HIGH_PEAK - LOW_PEAK;
+ int mid_value = (HIGH_PEAK + LOW_PEAK) / 2;
+
+ weight_b = abs(mid_value - a) - (range / 2 - 100) / 2;
+
+ if (weight_b < 0)
+ weight_b = 0;
+
+ if (weight_b > 100)
+ weight_b = 100;
+
+ weight_a = 100 - weight_b;
+
+ if (a < mid_value) {
+ /* 0..90 and 270..360 degrees */
+ pos_b = b - LOW_PEAK + DEG270;
+ if (pos_b >= DEG360)
+ pos_b -= DEG360;
+ } else
+ /* 90..270 degrees */
+ pos_b = HIGH_PEAK - b + DEG90;
+
+
+ if (b > mid_value)
+ /* 0..180 degrees */
+ pos_a = a - LOW_PEAK;
+ else
+ /* 180..360 degrees */
+ pos_a = HIGH_PEAK - a + DEG180;
+
+ /* interpolate both slider values, depending on weight factors */
+ /* 0..99 x DEG360 */
+ ret = pos_a * weight_a + pos_b * weight_b;
+
+ /* normalize to 0..999 */
+ ret *= 10;
+ ret /= DEG360;
+
+ if (ret < 0)
+ ret += 1000;
+
+ if (ret >= 1000)
+ ret -= 1000;
+
+ return ret;
+}
+
+#undef DEG90
+#undef DEG180
+#undef DEG270
+#undef DEG360
+#undef HIGH_PEAK
+#undef LOW_PEAK
+
+
+static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev,
+ const unsigned char *buf,
+ unsigned int len)
+{
+ struct input_dev *input_dev = dev->input_dev;
+
+ switch (dev->chip.usb_id) {
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2):
+ input_report_abs(input_dev, ABS_X, (buf[4] << 8) | buf[5]);
+ input_report_abs(input_dev, ABS_Y, (buf[0] << 8) | buf[1]);
+ input_report_abs(input_dev, ABS_Z, (buf[2] << 8) | buf[3]);
+ input_sync(input_dev);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
+ input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]);
+ input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]);
+ input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]);
+ input_sync(input_dev);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER):
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2):
+ input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]);
+ input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]);
+ input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]);
+ input_sync(input_dev);
+ break;
+ }
+}
+
+static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev,
+ const char *buf, unsigned int len)
+{
+ struct input_dev *input_dev = dev->input_dev;
+ int i;
+
+ switch (dev->chip.usb_id) {
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
+ i = decode_erp(buf[0], buf[1]);
+ input_report_abs(input_dev, ABS_X, i);
+ input_sync(input_dev);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER):
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2):
+ i = decode_erp(buf[7], buf[5]);
+ input_report_abs(input_dev, ABS_HAT0X, i);
+ i = decode_erp(buf[12], buf[14]);
+ input_report_abs(input_dev, ABS_HAT0Y, i);
+ i = decode_erp(buf[15], buf[13]);
+ input_report_abs(input_dev, ABS_HAT1X, i);
+ i = decode_erp(buf[0], buf[2]);
+ input_report_abs(input_dev, ABS_HAT1Y, i);
+ i = decode_erp(buf[3], buf[1]);
+ input_report_abs(input_dev, ABS_HAT2X, i);
+ i = decode_erp(buf[8], buf[10]);
+ input_report_abs(input_dev, ABS_HAT2Y, i);
+ i = decode_erp(buf[11], buf[9]);
+ input_report_abs(input_dev, ABS_HAT3X, i);
+ i = decode_erp(buf[4], buf[6]);
+ input_report_abs(input_dev, ABS_HAT3Y, i);
+ input_sync(input_dev);
+ break;
+ }
+}
+
+static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev,
+ char *buf, unsigned int len)
+{
+ struct input_dev *input_dev = dev->input_dev;
+ unsigned short *keycode = input_dev->keycode;
+ int i;
+
+ if (!keycode)
+ return;
+
+ if (input_dev->id.product == USB_PID_RIGKONTROL2)
+ for (i = 0; i < len; i++)
+ buf[i] = ~buf[i];
+
+ for (i = 0; i < input_dev->keycodemax && i < len * 8; i++)
+ input_report_key(input_dev, keycode[i],
+ buf[i / 8] & (1 << (i % 8)));
+
+ if (dev->chip.usb_id ==
+ USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER) ||
+ dev->chip.usb_id ==
+ USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2))
+ input_report_abs(dev->input_dev, ABS_MISC, 255 - buf[4]);
+
+ input_sync(input_dev);
+}
+
+void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev,
+ char *buf,
+ unsigned int len)
+{
+ if (!dev->input_dev || len < 1)
+ return;
+
+ switch (buf[0]) {
+ case EP1_CMD_READ_ANALOG:
+ snd_caiaq_input_read_analog(dev, buf + 1, len - 1);
+ break;
+ case EP1_CMD_READ_ERP:
+ snd_caiaq_input_read_erp(dev, buf + 1, len - 1);
+ break;
+ case EP1_CMD_READ_IO:
+ snd_caiaq_input_read_io(dev, buf + 1, len - 1);
+ break;
+ }
+}
+
+int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev)
+{
+ struct usb_device *usb_dev = dev->chip.dev;
+ struct input_dev *input;
+ int i, ret;
+
+ input = input_allocate_device();
+ if (!input)
+ return -ENOMEM;
+
+ usb_make_path(usb_dev, dev->phys, sizeof(dev->phys));
+ strlcat(dev->phys, "/input0", sizeof(dev->phys));
+
+ input->name = dev->product_name;
+ input->phys = dev->phys;
+ usb_to_input_id(usb_dev, &input->id);
+ input->dev.parent = &usb_dev->dev;
+
+ switch (dev->chip.usb_id) {
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2):
+ input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS);
+ input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) |
+ BIT_MASK(ABS_Z);
+ BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_rk2));
+ memcpy(dev->keycode, keycode_rk2, sizeof(keycode_rk2));
+ input->keycodemax = ARRAY_SIZE(keycode_rk2);
+ input_set_abs_params(input, ABS_X, 0, 4096, 0, 10);
+ input_set_abs_params(input, ABS_Y, 0, 4096, 0, 10);
+ input_set_abs_params(input, ABS_Z, 0, 4096, 0, 10);
+ snd_usb_caiaq_set_auto_msg(dev, 1, 10, 0);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
+ input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS);
+ input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) |
+ BIT_MASK(ABS_Z);
+ BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_rk3));
+ memcpy(dev->keycode, keycode_rk3, sizeof(keycode_rk3));
+ input->keycodemax = ARRAY_SIZE(keycode_rk3);
+ input_set_abs_params(input, ABS_X, 0, 1024, 0, 10);
+ input_set_abs_params(input, ABS_Y, 0, 1024, 0, 10);
+ input_set_abs_params(input, ABS_Z, 0, 1024, 0, 10);
+ snd_usb_caiaq_set_auto_msg(dev, 1, 10, 0);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
+ input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS);
+ input->absbit[0] = BIT_MASK(ABS_X);
+ BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_ak1));
+ memcpy(dev->keycode, keycode_ak1, sizeof(keycode_ak1));
+ input->keycodemax = ARRAY_SIZE(keycode_ak1);
+ input_set_abs_params(input, ABS_X, 0, 999, 0, 10);
+ snd_usb_caiaq_set_auto_msg(dev, 1, 0, 5);
+ break;
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER):
+ case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2):
+ input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS);
+ input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) |
+ BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) |
+ BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) |
+ BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) |
+ BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) |
+ BIT_MASK(ABS_Z);
+ input->absbit[BIT_WORD(ABS_MISC)] |= BIT_MASK(ABS_MISC);
+ BUILD_BUG_ON(sizeof(dev->keycode) < sizeof(keycode_kore));
+ memcpy(dev->keycode, keycode_kore, sizeof(keycode_kore));
+ input->keycodemax = ARRAY_SIZE(keycode_kore);
+ input_set_abs_params(input, ABS_HAT0X, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_HAT0Y, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_HAT1X, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_HAT1Y, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_HAT2X, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_HAT2Y, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_HAT3X, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_HAT3Y, 0, 999, 0, 10);
+ input_set_abs_params(input, ABS_X, 0, 4096, 0, 10);
+ input_set_abs_params(input, ABS_Y, 0, 4096, 0, 10);
+ input_set_abs_params(input, ABS_Z, 0, 4096, 0, 10);
+ input_set_abs_params(input, ABS_MISC, 0, 255, 0, 1);
+ snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5);
+ break;
+ default:
+ /* no input methods supported on this device */
+ input_free_device(input);
+ return 0;
+ }
+
+ input->keycode = dev->keycode;
+ input->keycodesize = sizeof(unsigned short);
+ for (i = 0; i < input->keycodemax; i++)
+ __set_bit(dev->keycode[i], input->keybit);
+
+ ret = input_register_device(input);
+ if (ret < 0) {
+ input_free_device(input);
+ return ret;
+ }
+
+ dev->input_dev = input;
+ return 0;
+}
+
+void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev)
+{
+ if (!dev || !dev->input_dev)
+ return;
+
+ input_unregister_device(dev->input_dev);
+ dev->input_dev = NULL;
+}
+
diff --git a/sound/usb/caiaq/caiaq-input.h b/sound/usb/caiaq/caiaq-input.h
new file mode 100644
index 0000000..ced5355
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-input.h
@@ -0,0 +1,8 @@
+#ifndef CAIAQ_INPUT_H
+#define CAIAQ_INPUT_H
+
+void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev, char *buf, unsigned int len);
+int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev);
+void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev);
+
+#endif
diff --git a/sound/usb/caiaq/caiaq-midi.c b/sound/usb/caiaq/caiaq-midi.c
new file mode 100644
index 0000000..f19fd36
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-midi.c
@@ -0,0 +1,180 @@
+/*
+ * Copyright (c) 2006,2007 Daniel Mack
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/usb.h>
+#include <linux/input.h>
+#include <linux/spinlock.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+#include <sound/pcm.h>
+
+#include "caiaq-device.h"
+#include "caiaq-midi.h"
+
+
+static int snd_usb_caiaq_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static int snd_usb_caiaq_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_usb_caiaqdev *dev = substream->rmidi->private_data;
+
+ if (!dev)
+ return;
+
+ dev->midi_receive_substream = up ? substream : NULL;
+}
+
+
+static int snd_usb_caiaq_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static int snd_usb_caiaq_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_usb_caiaqdev *dev = substream->rmidi->private_data;
+ if (dev->midi_out_active) {
+ usb_kill_urb(&dev->midi_out_urb);
+ dev->midi_out_active = 0;
+ }
+ return 0;
+}
+
+static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev,
+ struct snd_rawmidi_substream *substream)
+{
+ int len, ret;
+
+ dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE;
+ dev->midi_out_buf[1] = 0; /* port */
+ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3,
+ EP1_BUFSIZE - 3);
+
+ if (len <= 0)
+ return;
+
+ dev->midi_out_buf[2] = len;
+ dev->midi_out_urb.transfer_buffer_length = len+3;
+
+ ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC);
+ if (ret < 0)
+ log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed,"
+ "ret=%d, len=%d\n",
+ substream, ret, len);
+ else
+ dev->midi_out_active = 1;
+}
+
+static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_usb_caiaqdev *dev = substream->rmidi->private_data;
+
+ if (up) {
+ dev->midi_out_substream = substream;
+ if (!dev->midi_out_active)
+ snd_usb_caiaq_midi_send(dev, substream);
+ } else {
+ dev->midi_out_substream = NULL;
+ }
+}
+
+
+static struct snd_rawmidi_ops snd_usb_caiaq_midi_output =
+{
+ .open = snd_usb_caiaq_midi_output_open,
+ .close = snd_usb_caiaq_midi_output_close,
+ .trigger = snd_usb_caiaq_midi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_usb_caiaq_midi_input =
+{
+ .open = snd_usb_caiaq_midi_input_open,
+ .close = snd_usb_caiaq_midi_input_close,
+ .trigger = snd_usb_caiaq_midi_input_trigger,
+};
+
+void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev,
+ int port, const char *buf, int len)
+{
+ if (!dev->midi_receive_substream)
+ return;
+
+ snd_rawmidi_receive(dev->midi_receive_substream, buf, len);
+}
+
+int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device)
+{
+ int ret;
+ struct snd_rawmidi *rmidi;
+
+ ret = snd_rawmidi_new(device->chip.card, device->product_name, 0,
+ device->spec.num_midi_out,
+ device->spec.num_midi_in,
+ &rmidi);
+
+ if (ret < 0)
+ return ret;
+
+ strcpy(rmidi->name, device->product_name);
+
+ rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->private_data = device;
+
+ if (device->spec.num_midi_out > 0) {
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_usb_caiaq_midi_output);
+ }
+
+ if (device->spec.num_midi_in > 0) {
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_usb_caiaq_midi_input);
+ }
+
+ device->rmidi = rmidi;
+
+ return 0;
+}
+
+void snd_usb_caiaq_midi_output_done(struct urb* urb)
+{
+ struct snd_usb_caiaqdev *dev = urb->context;
+
+ dev->midi_out_active = 0;
+ if (urb->status != 0)
+ return;
+
+ if (!dev->midi_out_substream)
+ return;
+
+ snd_usb_caiaq_midi_send(dev, dev->midi_out_substream);
+}
+
diff --git a/sound/usb/caiaq/caiaq-midi.h b/sound/usb/caiaq/caiaq-midi.h
new file mode 100644
index 0000000..9d16db0
--- /dev/null
+++ b/sound/usb/caiaq/caiaq-midi.h
@@ -0,0 +1,8 @@
+#ifndef CAIAQ_MIDI_H
+#define CAIAQ_MIDI_H
+
+int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *dev);
+void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len);
+void snd_usb_caiaq_midi_output_done(struct urb* urb);
+
+#endif /* CAIAQ_MIDI_H */
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
new file mode 100644
index 0000000..a51190e
--- /dev/null
+++ b/sound/usb/usbaudio.c
@@ -0,0 +1,3783 @@
+/*
+ * (Tentative) USB Audio Driver for ALSA
+ *
+ * Main and PCM part
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * Many codes borrowed from audio.c by
+ * Alan Cox (alan@lxorguk.ukuu.org.uk)
+ * Thomas Sailer (sailer@ife.ee.ethz.ch)
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ *
+ * NOTES:
+ *
+ * - async unlink should be used for avoiding the sleep inside lock.
+ * 2.4.22 usb-uhci seems buggy for async unlinking and results in
+ * oops. in such a cse, pass async_unlink=0 option.
+ * - the linked URBs would be preferred but not used so far because of
+ * the instability of unlinking.
+ * - type II is not supported properly. there is no device which supports
+ * this type *correctly*. SB extigy looks as if it supports, but it's
+ * indeed an AC3 stream packed in SPDIF frames (i.e. no real AC3 stream).
+ */
+
+
+#include <linux/bitops.h>
+#include <linux/init.h>
+#include <linux/list.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <linux/usb.h>
+#include <linux/vmalloc.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+
+#include "usbaudio.h"
+
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("USB Audio");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Generic,USB Audio}}");
+
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */
+/* Vendor/product IDs for this card */
+static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 };
+static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 };
+static int nrpacks = 8; /* max. number of packets per urb */
+static int async_unlink = 1;
+static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/
+static int ignore_ctl_error;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the USB audio adapter.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the USB audio adapter.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable USB audio adapter.");
+module_param_array(vid, int, NULL, 0444);
+MODULE_PARM_DESC(vid, "Vendor ID for the USB audio device.");
+module_param_array(pid, int, NULL, 0444);
+MODULE_PARM_DESC(pid, "Product ID for the USB audio device.");
+module_param(nrpacks, int, 0644);
+MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB.");
+module_param(async_unlink, bool, 0444);
+MODULE_PARM_DESC(async_unlink, "Use async unlink mode.");
+module_param_array(device_setup, int, NULL, 0444);
+MODULE_PARM_DESC(device_setup, "Specific device setup (if needed).");
+module_param(ignore_ctl_error, bool, 0444);
+MODULE_PARM_DESC(ignore_ctl_error,
+ "Ignore errors from USB controller for mixer interfaces.");
+
+/*
+ * debug the h/w constraints
+ */
+/* #define HW_CONST_DEBUG */
+
+
+/*
+ *
+ */
+
+#define MAX_PACKS 20
+#define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */
+#define MAX_URBS 8
+#define SYNC_URBS 4 /* always four urbs for sync */
+#define MIN_PACKS_URB 1 /* minimum 1 packet per urb */
+
+struct audioformat {
+ struct list_head list;
+ snd_pcm_format_t format; /* format type */
+ unsigned int channels; /* # channels */
+ unsigned int fmt_type; /* USB audio format type (1-3) */
+ unsigned int frame_size; /* samples per frame for non-audio */
+ int iface; /* interface number */
+ unsigned char altsetting; /* corresponding alternate setting */
+ unsigned char altset_idx; /* array index of altenate setting */
+ unsigned char attributes; /* corresponding attributes of cs endpoint */
+ unsigned char endpoint; /* endpoint */
+ unsigned char ep_attr; /* endpoint attributes */
+ unsigned int maxpacksize; /* max. packet size */
+ unsigned int rates; /* rate bitmasks */
+ unsigned int rate_min, rate_max; /* min/max rates */
+ unsigned int nr_rates; /* number of rate table entries */
+ unsigned int *rate_table; /* rate table */
+};
+
+struct snd_usb_substream;
+
+struct snd_urb_ctx {
+ struct urb *urb;
+ unsigned int buffer_size; /* size of data buffer, if data URB */
+ struct snd_usb_substream *subs;
+ int index; /* index for urb array */
+ int packets; /* number of packets per urb */
+};
+
+struct snd_urb_ops {
+ int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
+ int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
+ int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
+ int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
+};
+
+struct snd_usb_substream {
+ struct snd_usb_stream *stream;
+ struct usb_device *dev;
+ struct snd_pcm_substream *pcm_substream;
+ int direction; /* playback or capture */
+ int interface; /* current interface */
+ int endpoint; /* assigned endpoint */
+ struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */
+ unsigned int cur_rate; /* current rate (for hw_params callback) */
+ unsigned int period_bytes; /* current period bytes (for hw_params callback) */
+ unsigned int format; /* USB data format */
+ unsigned int datapipe; /* the data i/o pipe */
+ unsigned int syncpipe; /* 1 - async out or adaptive in */
+ unsigned int datainterval; /* log_2 of data packet interval */
+ unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */
+ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */
+ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */
+ unsigned int freqmax; /* maximum sampling rate, used for buffer management */
+ unsigned int phase; /* phase accumulator */
+ unsigned int maxpacksize; /* max packet size in bytes */
+ unsigned int maxframesize; /* max packet size in frames */
+ unsigned int curpacksize; /* current packet size in bytes (for capture) */
+ unsigned int curframesize; /* current packet size in frames (for capture) */
+ unsigned int fill_max: 1; /* fill max packet size always */
+ unsigned int fmt_type; /* USB audio format type (1-3) */
+ unsigned int packs_per_ms; /* packets per millisecond (for playback) */
+
+ unsigned int running: 1; /* running status */
+
+ unsigned int hwptr_done; /* processed frame position in the buffer */
+ unsigned int transfer_done; /* processed frames since last period update */
+ unsigned long active_mask; /* bitmask of active urbs */
+ unsigned long unlink_mask; /* bitmask of unlinked urbs */
+
+ unsigned int nurbs; /* # urbs */
+ struct snd_urb_ctx dataurb[MAX_URBS]; /* data urb table */
+ struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */
+ char *syncbuf; /* sync buffer for all sync URBs */
+ dma_addr_t sync_dma; /* DMA address of syncbuf */
+
+ u64 formats; /* format bitmasks (all or'ed) */
+ unsigned int num_formats; /* number of supported audio formats (list) */
+ struct list_head fmt_list; /* format list */
+ struct snd_pcm_hw_constraint_list rate_list; /* limited rates */
+ spinlock_t lock;
+
+ struct snd_urb_ops ops; /* callbacks (must be filled at init) */
+};
+
+
+struct snd_usb_stream {
+ struct snd_usb_audio *chip;
+ struct snd_pcm *pcm;
+ int pcm_index;
+ unsigned int fmt_type; /* USB audio format type (1-3) */
+ struct snd_usb_substream substream[2];
+ struct list_head list;
+};
+
+
+/*
+ * we keep the snd_usb_audio_t instances by ourselves for merging
+ * the all interfaces on the same card as one sound device.
+ */
+
+static DEFINE_MUTEX(register_mutex);
+static struct snd_usb_audio *usb_chip[SNDRV_CARDS];
+
+
+/*
+ * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
+ * this will overflow at approx 524 kHz
+ */
+static inline unsigned get_usb_full_speed_rate(unsigned int rate)
+{
+ return ((rate << 13) + 62) / 125;
+}
+
+/*
+ * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
+ * this will overflow at approx 4 MHz
+ */
+static inline unsigned get_usb_high_speed_rate(unsigned int rate)
+{
+ return ((rate << 10) + 62) / 125;
+}
+
+/* convert our full speed USB rate into sampling rate in Hz */
+static inline unsigned get_full_speed_hz(unsigned int usb_rate)
+{
+ return (usb_rate * 125 + (1 << 12)) >> 13;
+}
+
+/* convert our high speed USB rate into sampling rate in Hz */
+static inline unsigned get_high_speed_hz(unsigned int usb_rate)
+{
+ return (usb_rate * 125 + (1 << 9)) >> 10;
+}
+
+
+/*
+ * prepare urb for full speed capture sync pipe
+ *
+ * fill the length and offset of each urb descriptor.
+ * the fixed 10.14 frequency is passed through the pipe.
+ */
+static int prepare_capture_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned char *cp = urb->transfer_buffer;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->iso_frame_desc[0].length = 3;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = subs->freqn >> 2;
+ cp[1] = subs->freqn >> 10;
+ cp[2] = subs->freqn >> 18;
+ return 0;
+}
+
+/*
+ * prepare urb for high speed capture sync pipe
+ *
+ * fill the length and offset of each urb descriptor.
+ * the fixed 12.13 frequency is passed as 16.16 through the pipe.
+ */
+static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned char *cp = urb->transfer_buffer;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->iso_frame_desc[0].length = 4;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = subs->freqn;
+ cp[1] = subs->freqn >> 8;
+ cp[2] = subs->freqn >> 16;
+ cp[3] = subs->freqn >> 24;
+ return 0;
+}
+
+/*
+ * process after capture sync complete
+ * - nothing to do
+ */
+static int retire_capture_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ return 0;
+}
+
+/*
+ * prepare urb for capture data pipe
+ *
+ * fill the offset and length of each descriptor.
+ *
+ * we use a temporary buffer to write the captured data.
+ * since the length of written data is determined by host, we cannot
+ * write onto the pcm buffer directly... the data is thus copied
+ * later at complete callback to the global buffer.
+ */
+static int prepare_capture_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ int i, offs;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ offs = 0;
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ for (i = 0; i < ctx->packets; i++) {
+ urb->iso_frame_desc[i].offset = offs;
+ urb->iso_frame_desc[i].length = subs->curpacksize;
+ offs += subs->curpacksize;
+ }
+ urb->transfer_buffer_length = offs;
+ urb->number_of_packets = ctx->packets;
+ return 0;
+}
+
+/*
+ * process after capture complete
+ *
+ * copy the data from each desctiptor to the pcm buffer, and
+ * update the current position.
+ */
+static int retire_capture_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned long flags;
+ unsigned char *cp;
+ int i;
+ unsigned int stride, len, oldptr;
+ int period_elapsed = 0;
+
+ stride = runtime->frame_bits >> 3;
+
+ for (i = 0; i < urb->number_of_packets; i++) {
+ cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
+ if (urb->iso_frame_desc[i].status) {
+ snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
+ // continue;
+ }
+ len = urb->iso_frame_desc[i].actual_length / stride;
+ if (! len)
+ continue;
+ /* update the current pointer */
+ spin_lock_irqsave(&subs->lock, flags);
+ oldptr = subs->hwptr_done;
+ subs->hwptr_done += len;
+ if (subs->hwptr_done >= runtime->buffer_size)
+ subs->hwptr_done -= runtime->buffer_size;
+ subs->transfer_done += len;
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ period_elapsed = 1;
+ }
+ spin_unlock_irqrestore(&subs->lock, flags);
+ /* copy a data chunk */
+ if (oldptr + len > runtime->buffer_size) {
+ unsigned int cnt = runtime->buffer_size - oldptr;
+ unsigned int blen = cnt * stride;
+ memcpy(runtime->dma_area + oldptr * stride, cp, blen);
+ memcpy(runtime->dma_area, cp + blen, len * stride - blen);
+ } else {
+ memcpy(runtime->dma_area + oldptr * stride, cp, len * stride);
+ }
+ }
+ if (period_elapsed)
+ snd_pcm_period_elapsed(subs->pcm_substream);
+ return 0;
+}
+
+/*
+ * Process after capture complete when paused. Nothing to do.
+ */
+static int retire_paused_capture_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ return 0;
+}
+
+
+/*
+ * prepare urb for full speed playback sync pipe
+ *
+ * set up the offset and length to receive the current frequency.
+ */
+
+static int prepare_playback_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->iso_frame_desc[0].length = 3;
+ urb->iso_frame_desc[0].offset = 0;
+ return 0;
+}
+
+/*
+ * prepare urb for high speed playback sync pipe
+ *
+ * set up the offset and length to receive the current frequency.
+ */
+
+static int prepare_playback_sync_urb_hs(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->iso_frame_desc[0].length = 4;
+ urb->iso_frame_desc[0].offset = 0;
+ return 0;
+}
+
+/*
+ * process after full speed playback sync complete
+ *
+ * retrieve the current 10.14 frequency from pipe, and set it.
+ * the value is referred in prepare_playback_urb().
+ */
+static int retire_playback_sync_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned int f;
+ unsigned long flags;
+
+ if (urb->iso_frame_desc[0].status == 0 &&
+ urb->iso_frame_desc[0].actual_length == 3) {
+ f = combine_triple((u8*)urb->transfer_buffer) << 2;
+ if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) {
+ spin_lock_irqsave(&subs->lock, flags);
+ subs->freqm = f;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ }
+ }
+
+ return 0;
+}
+
+/*
+ * process after high speed playback sync complete
+ *
+ * retrieve the current 12.13 frequency from pipe, and set it.
+ * the value is referred in prepare_playback_urb().
+ */
+static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned int f;
+ unsigned long flags;
+
+ if (urb->iso_frame_desc[0].status == 0 &&
+ urb->iso_frame_desc[0].actual_length == 4) {
+ f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff;
+ if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) {
+ spin_lock_irqsave(&subs->lock, flags);
+ subs->freqm = f;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ }
+ }
+
+ return 0;
+}
+
+/*
+ * process after E-Mu 0202/0404/Tracker Pre high speed playback sync complete
+ *
+ * These devices return the number of samples per packet instead of the number
+ * of samples per microframe.
+ */
+static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned int f;
+ unsigned long flags;
+
+ if (urb->iso_frame_desc[0].status == 0 &&
+ urb->iso_frame_desc[0].actual_length == 4) {
+ f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff;
+ f >>= subs->datainterval;
+ if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) {
+ spin_lock_irqsave(&subs->lock, flags);
+ subs->freqm = f;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ }
+ }
+
+ return 0;
+}
+
+/* determine the number of frames in the next packet */
+static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
+{
+ if (subs->fill_max)
+ return subs->maxframesize;
+ else {
+ subs->phase = (subs->phase & 0xffff)
+ + (subs->freqm << subs->datainterval);
+ return min(subs->phase >> 16, subs->maxframesize);
+ }
+}
+
+/*
+ * Prepare urb for streaming before playback starts or when paused.
+ *
+ * We don't have any data, so we send a frame of silence.
+ */
+static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ unsigned int i, offs, counts;
+ struct snd_urb_ctx *ctx = urb->context;
+ int stride = runtime->frame_bits >> 3;
+
+ offs = 0;
+ urb->dev = ctx->subs->dev;
+ urb->number_of_packets = subs->packs_per_ms;
+ for (i = 0; i < subs->packs_per_ms; ++i) {
+ counts = snd_usb_audio_next_packet_size(subs);
+ urb->iso_frame_desc[i].offset = offs * stride;
+ urb->iso_frame_desc[i].length = counts * stride;
+ offs += counts;
+ }
+ urb->transfer_buffer_length = offs * stride;
+ memset(urb->transfer_buffer,
+ subs->cur_audiofmt->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
+ offs * stride);
+ return 0;
+}
+
+/*
+ * prepare urb for playback data pipe
+ *
+ * Since a URB can handle only a single linear buffer, we must use double
+ * buffering when the data to be transferred overflows the buffer boundary.
+ * To avoid inconsistencies when updating hwptr_done, we use double buffering
+ * for all URBs.
+ */
+static int prepare_playback_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ int i, stride, offs;
+ unsigned int counts;
+ unsigned long flags;
+ int period_elapsed = 0;
+ struct snd_urb_ctx *ctx = urb->context;
+
+ stride = runtime->frame_bits >> 3;
+
+ offs = 0;
+ urb->dev = ctx->subs->dev; /* we need to set this at each time */
+ urb->number_of_packets = 0;
+ spin_lock_irqsave(&subs->lock, flags);
+ for (i = 0; i < ctx->packets; i++) {
+ counts = snd_usb_audio_next_packet_size(subs);
+ /* set up descriptor */
+ urb->iso_frame_desc[i].offset = offs * stride;
+ urb->iso_frame_desc[i].length = counts * stride;
+ offs += counts;
+ urb->number_of_packets++;
+ subs->transfer_done += counts;
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ period_elapsed = 1;
+ if (subs->fmt_type == USB_FORMAT_TYPE_II) {
+ if (subs->transfer_done > 0) {
+ /* FIXME: fill-max mode is not
+ * supported yet */
+ offs -= subs->transfer_done;
+ counts -= subs->transfer_done;
+ urb->iso_frame_desc[i].length =
+ counts * stride;
+ subs->transfer_done = 0;
+ }
+ i++;
+ if (i < ctx->packets) {
+ /* add a transfer delimiter */
+ urb->iso_frame_desc[i].offset =
+ offs * stride;
+ urb->iso_frame_desc[i].length = 0;
+ urb->number_of_packets++;
+ }
+ break;
+ }
+ }
+ /* finish at the frame boundary at/after the period boundary */
+ if (period_elapsed &&
+ (i & (subs->packs_per_ms - 1)) == subs->packs_per_ms - 1)
+ break;
+ }
+ if (subs->hwptr_done + offs > runtime->buffer_size) {
+ /* err, the transferred area goes over buffer boundary. */
+ unsigned int len = runtime->buffer_size - subs->hwptr_done;
+ memcpy(urb->transfer_buffer,
+ runtime->dma_area + subs->hwptr_done * stride,
+ len * stride);
+ memcpy(urb->transfer_buffer + len * stride,
+ runtime->dma_area,
+ (offs - len) * stride);
+ } else {
+ memcpy(urb->transfer_buffer,
+ runtime->dma_area + subs->hwptr_done * stride,
+ offs * stride);
+ }
+ subs->hwptr_done += offs;
+ if (subs->hwptr_done >= runtime->buffer_size)
+ subs->hwptr_done -= runtime->buffer_size;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ urb->transfer_buffer_length = offs * stride;
+ if (period_elapsed)
+ snd_pcm_period_elapsed(subs->pcm_substream);
+ return 0;
+}
+
+/*
+ * process after playback data complete
+ * - nothing to do
+ */
+static int retire_playback_urb(struct snd_usb_substream *subs,
+ struct snd_pcm_runtime *runtime,
+ struct urb *urb)
+{
+ return 0;
+}
+
+
+/*
+ */
+static struct snd_urb_ops audio_urb_ops[2] = {
+ {
+ .prepare = prepare_nodata_playback_urb,
+ .retire = retire_playback_urb,
+ .prepare_sync = prepare_playback_sync_urb,
+ .retire_sync = retire_playback_sync_urb,
+ },
+ {
+ .prepare = prepare_capture_urb,
+ .retire = retire_capture_urb,
+ .prepare_sync = prepare_capture_sync_urb,
+ .retire_sync = retire_capture_sync_urb,
+ },
+};
+
+static struct snd_urb_ops audio_urb_ops_high_speed[2] = {
+ {
+ .prepare = prepare_nodata_playback_urb,
+ .retire = retire_playback_urb,
+ .prepare_sync = prepare_playback_sync_urb_hs,
+ .retire_sync = retire_playback_sync_urb_hs,
+ },
+ {
+ .prepare = prepare_capture_urb,
+ .retire = retire_capture_urb,
+ .prepare_sync = prepare_capture_sync_urb_hs,
+ .retire_sync = retire_capture_sync_urb,
+ },
+};
+
+/*
+ * complete callback from data urb
+ */
+static void snd_complete_urb(struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+ struct snd_usb_substream *subs = ctx->subs;
+ struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
+ int err = 0;
+
+ if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) ||
+ !subs->running || /* can be stopped during retire callback */
+ (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 ||
+ (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+ clear_bit(ctx->index, &subs->active_mask);
+ if (err < 0) {
+ snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err);
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ }
+ }
+}
+
+
+/*
+ * complete callback from sync urb
+ */
+static void snd_complete_sync_urb(struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+ struct snd_usb_substream *subs = ctx->subs;
+ struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
+ int err = 0;
+
+ if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) ||
+ !subs->running || /* can be stopped during retire callback */
+ (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 ||
+ (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+ clear_bit(ctx->index + 16, &subs->active_mask);
+ if (err < 0) {
+ snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err);
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ }
+ }
+}
+
+
+/* get the physical page pointer at the given offset */
+static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs,
+ unsigned long offset)
+{
+ void *pageptr = subs->runtime->dma_area + offset;
+ return vmalloc_to_page(pageptr);
+}
+
+/* allocate virtual buffer; may be called more than once */
+static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+ if (runtime->dma_area) {
+ if (runtime->dma_bytes >= size)
+ return 0; /* already large enough */
+ vfree(runtime->dma_area);
+ }
+ runtime->dma_area = vmalloc(size);
+ if (!runtime->dma_area)
+ return -ENOMEM;
+ runtime->dma_bytes = size;
+ return 0;
+}
+
+/* free virtual buffer; may be called more than once */
+static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs)
+{
+ struct snd_pcm_runtime *runtime = subs->runtime;
+
+ vfree(runtime->dma_area);
+ runtime->dma_area = NULL;
+ return 0;
+}
+
+
+/*
+ * unlink active urbs.
+ */
+static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
+{
+ unsigned int i;
+ int async;
+
+ subs->running = 0;
+
+ if (!force && subs->stream->chip->shutdown) /* to be sure... */
+ return -EBADFD;
+
+ async = !can_sleep && async_unlink;
+
+ if (!async && in_interrupt())
+ return 0;
+
+ for (i = 0; i < subs->nurbs; i++) {
+ if (test_bit(i, &subs->active_mask)) {
+ if (!test_and_set_bit(i, &subs->unlink_mask)) {
+ struct urb *u = subs->dataurb[i].urb;
+ if (async)
+ usb_unlink_urb(u);
+ else
+ usb_kill_urb(u);
+ }
+ }
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ if (test_bit(i+16, &subs->active_mask)) {
+ if (!test_and_set_bit(i+16, &subs->unlink_mask)) {
+ struct urb *u = subs->syncurb[i].urb;
+ if (async)
+ usb_unlink_urb(u);
+ else
+ usb_kill_urb(u);
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+
+static const char *usb_error_string(int err)
+{
+ switch (err) {
+ case -ENODEV:
+ return "no device";
+ case -ENOENT:
+ return "endpoint not enabled";
+ case -EPIPE:
+ return "endpoint stalled";
+ case -ENOSPC:
+ return "not enough bandwidth";
+ case -ESHUTDOWN:
+ return "device disabled";
+ case -EHOSTUNREACH:
+ return "device suspended";
+ case -EINVAL:
+ case -EAGAIN:
+ case -EFBIG:
+ case -EMSGSIZE:
+ return "internal error";
+ default:
+ return "unknown error";
+ }
+}
+
+/*
+ * set up and start data/sync urbs
+ */
+static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime)
+{
+ unsigned int i;
+ int err;
+
+ if (subs->stream->chip->shutdown)
+ return -EBADFD;
+
+ for (i = 0; i < subs->nurbs; i++) {
+ if (snd_BUG_ON(!subs->dataurb[i].urb))
+ return -EINVAL;
+ if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) {
+ snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i);
+ goto __error;
+ }
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ if (snd_BUG_ON(!subs->syncurb[i].urb))
+ return -EINVAL;
+ if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) {
+ snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i);
+ goto __error;
+ }
+ }
+ }
+
+ subs->active_mask = 0;
+ subs->unlink_mask = 0;
+ subs->running = 1;
+ for (i = 0; i < subs->nurbs; i++) {
+ err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot submit datapipe "
+ "for urb %d, error %d: %s\n",
+ i, err, usb_error_string(err));
+ goto __error;
+ }
+ set_bit(i, &subs->active_mask);
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot submit syncpipe "
+ "for urb %d, error %d: %s\n",
+ i, err, usb_error_string(err));
+ goto __error;
+ }
+ set_bit(i + 16, &subs->active_mask);
+ }
+ }
+ return 0;
+
+ __error:
+ // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
+ deactivate_urbs(subs, 0, 0);
+ return -EPIPE;
+}
+
+
+/*
+ * wait until all urbs are processed.
+ */
+static int wait_clear_urbs(struct snd_usb_substream *subs)
+{
+ unsigned long end_time = jiffies + msecs_to_jiffies(1000);
+ unsigned int i;
+ int alive;
+
+ do {
+ alive = 0;
+ for (i = 0; i < subs->nurbs; i++) {
+ if (test_bit(i, &subs->active_mask))
+ alive++;
+ }
+ if (subs->syncpipe) {
+ for (i = 0; i < SYNC_URBS; i++) {
+ if (test_bit(i + 16, &subs->active_mask))
+ alive++;
+ }
+ }
+ if (! alive)
+ break;
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+ if (alive)
+ snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive);
+ return 0;
+}
+
+
+/*
+ * return the current pcm pointer. just return the hwptr_done value.
+ */
+static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_usb_substream *subs;
+ snd_pcm_uframes_t hwptr_done;
+
+ subs = (struct snd_usb_substream *)substream->runtime->private_data;
+ spin_lock(&subs->lock);
+ hwptr_done = subs->hwptr_done;
+ spin_unlock(&subs->lock);
+ return hwptr_done;
+}
+
+
+/*
+ * start/stop playback substream
+ */
+static int snd_usb_pcm_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ subs->ops.prepare = prepare_playback_urb;
+ return 0;
+ case SNDRV_PCM_TRIGGER_STOP:
+ return deactivate_urbs(subs, 0, 0);
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ subs->ops.prepare = prepare_nodata_playback_urb;
+ return 0;
+ default:
+ return -EINVAL;
+ }
+}
+
+/*
+ * start/stop capture substream
+ */
+static int snd_usb_pcm_capture_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ subs->ops.retire = retire_capture_urb;
+ return start_urbs(subs, substream->runtime);
+ case SNDRV_PCM_TRIGGER_STOP:
+ return deactivate_urbs(subs, 0, 0);
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ subs->ops.retire = retire_paused_capture_urb;
+ return 0;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ subs->ops.retire = retire_capture_urb;
+ return 0;
+ default:
+ return -EINVAL;
+ }
+}
+
+
+/*
+ * release a urb data
+ */
+static void release_urb_ctx(struct snd_urb_ctx *u)
+{
+ if (u->urb) {
+ if (u->buffer_size)
+ usb_buffer_free(u->subs->dev, u->buffer_size,
+ u->urb->transfer_buffer,
+ u->urb->transfer_dma);
+ usb_free_urb(u->urb);
+ u->urb = NULL;
+ }
+}
+
+/*
+ * release a substream
+ */
+static void release_substream_urbs(struct snd_usb_substream *subs, int force)
+{
+ int i;
+
+ /* stop urbs (to be sure) */
+ deactivate_urbs(subs, force, 1);
+ wait_clear_urbs(subs);
+
+ for (i = 0; i < MAX_URBS; i++)
+ release_urb_ctx(&subs->dataurb[i]);
+ for (i = 0; i < SYNC_URBS; i++)
+ release_urb_ctx(&subs->syncurb[i]);
+ usb_buffer_free(subs->dev, SYNC_URBS * 4,
+ subs->syncbuf, subs->sync_dma);
+ subs->syncbuf = NULL;
+ subs->nurbs = 0;
+}
+
+/*
+ * initialize a substream for plaback/capture
+ */
+static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int period_bytes,
+ unsigned int rate, unsigned int frame_bits)
+{
+ unsigned int maxsize, n, i;
+ int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned int npacks[MAX_URBS], urb_packs, total_packs, packs_per_ms;
+
+ /* calculate the frequency in 16.16 format */
+ if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
+ subs->freqn = get_usb_full_speed_rate(rate);
+ else
+ subs->freqn = get_usb_high_speed_rate(rate);
+ subs->freqm = subs->freqn;
+ /* calculate max. frequency */
+ if (subs->maxpacksize) {
+ /* whatever fits into a max. size packet */
+ maxsize = subs->maxpacksize;
+ subs->freqmax = (maxsize / (frame_bits >> 3))
+ << (16 - subs->datainterval);
+ } else {
+ /* no max. packet size: just take 25% higher than nominal */
+ subs->freqmax = subs->freqn + (subs->freqn >> 2);
+ maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3))
+ >> (16 - subs->datainterval);
+ }
+ subs->phase = 0;
+
+ if (subs->fill_max)
+ subs->curpacksize = subs->maxpacksize;
+ else
+ subs->curpacksize = maxsize;
+
+ if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH)
+ packs_per_ms = 8 >> subs->datainterval;
+ else
+ packs_per_ms = 1;
+ subs->packs_per_ms = packs_per_ms;
+
+ if (is_playback) {
+ urb_packs = nrpacks;
+ urb_packs = max(urb_packs, (unsigned int)MIN_PACKS_URB);
+ urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
+ } else
+ urb_packs = 1;
+ urb_packs *= packs_per_ms;
+
+ /* decide how many packets to be used */
+ if (is_playback) {
+ unsigned int minsize;
+ /* determine how small a packet can be */
+ minsize = (subs->freqn >> (16 - subs->datainterval))
+ * (frame_bits >> 3);
+ /* with sync from device, assume it can be 12% lower */
+ if (subs->syncpipe)
+ minsize -= minsize >> 3;
+ minsize = max(minsize, 1u);
+ total_packs = (period_bytes + minsize - 1) / minsize;
+ /* round up to multiple of packs_per_ms */
+ total_packs = (total_packs + packs_per_ms - 1)
+ & ~(packs_per_ms - 1);
+ /* we need at least two URBs for queueing */
+ if (total_packs < 2 * MIN_PACKS_URB * packs_per_ms)
+ total_packs = 2 * MIN_PACKS_URB * packs_per_ms;
+ } else {
+ total_packs = MAX_URBS * urb_packs;
+ }
+ subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
+ if (subs->nurbs > MAX_URBS) {
+ /* too much... */
+ subs->nurbs = MAX_URBS;
+ total_packs = MAX_URBS * urb_packs;
+ }
+ n = total_packs;
+ for (i = 0; i < subs->nurbs; i++) {
+ npacks[i] = n > urb_packs ? urb_packs : n;
+ n -= urb_packs;
+ }
+ if (subs->nurbs <= 1) {
+ /* too little - we need at least two packets
+ * to ensure contiguous playback/capture
+ */
+ subs->nurbs = 2;
+ npacks[0] = (total_packs + 1) / 2;
+ npacks[1] = total_packs - npacks[0];
+ } else if (npacks[subs->nurbs-1] < MIN_PACKS_URB * packs_per_ms) {
+ /* the last packet is too small.. */
+ if (subs->nurbs > 2) {
+ /* merge to the first one */
+ npacks[0] += npacks[subs->nurbs - 1];
+ subs->nurbs--;
+ } else {
+ /* divide to two */
+ subs->nurbs = 2;
+ npacks[0] = (total_packs + 1) / 2;
+ npacks[1] = total_packs - npacks[0];
+ }
+ }
+
+ /* allocate and initialize data urbs */
+ for (i = 0; i < subs->nurbs; i++) {
+ struct snd_urb_ctx *u = &subs->dataurb[i];
+ u->index = i;
+ u->subs = subs;
+ u->packets = npacks[i];
+ u->buffer_size = maxsize * u->packets;
+ if (subs->fmt_type == USB_FORMAT_TYPE_II)
+ u->packets++; /* for transfer delimiter */
+ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
+ if (!u->urb)
+ goto out_of_memory;
+ u->urb->transfer_buffer =
+ usb_buffer_alloc(subs->dev, u->buffer_size, GFP_KERNEL,
+ &u->urb->transfer_dma);
+ if (!u->urb->transfer_buffer)
+ goto out_of_memory;
+ u->urb->pipe = subs->datapipe;
+ u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
+ u->urb->interval = 1 << subs->datainterval;
+ u->urb->context = u;
+ u->urb->complete = snd_complete_urb;
+ }
+
+ if (subs->syncpipe) {
+ /* allocate and initialize sync urbs */
+ subs->syncbuf = usb_buffer_alloc(subs->dev, SYNC_URBS * 4,
+ GFP_KERNEL, &subs->sync_dma);
+ if (!subs->syncbuf)
+ goto out_of_memory;
+ for (i = 0; i < SYNC_URBS; i++) {
+ struct snd_urb_ctx *u = &subs->syncurb[i];
+ u->index = i;
+ u->subs = subs;
+ u->packets = 1;
+ u->urb = usb_alloc_urb(1, GFP_KERNEL);
+ if (!u->urb)
+ goto out_of_memory;
+ u->urb->transfer_buffer = subs->syncbuf + i * 4;
+ u->urb->transfer_dma = subs->sync_dma + i * 4;
+ u->urb->transfer_buffer_length = 4;
+ u->urb->pipe = subs->syncpipe;
+ u->urb->transfer_flags = URB_ISO_ASAP |
+ URB_NO_TRANSFER_DMA_MAP;
+ u->urb->number_of_packets = 1;
+ u->urb->interval = 1 << subs->syncinterval;
+ u->urb->context = u;
+ u->urb->complete = snd_complete_sync_urb;
+ }
+ }
+ return 0;
+
+out_of_memory:
+ release_substream_urbs(subs, 0);
+ return -ENOMEM;
+}
+
+
+/*
+ * find a matching audio format
+ */
+static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned int format,
+ unsigned int rate, unsigned int channels)
+{
+ struct list_head *p;
+ struct audioformat *found = NULL;
+ int cur_attr = 0, attr;
+
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *fp;
+ fp = list_entry(p, struct audioformat, list);
+ if (fp->format != format || fp->channels != channels)
+ continue;
+ if (rate < fp->rate_min || rate > fp->rate_max)
+ continue;
+ if (! (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)) {
+ unsigned int i;
+ for (i = 0; i < fp->nr_rates; i++)
+ if (fp->rate_table[i] == rate)
+ break;
+ if (i >= fp->nr_rates)
+ continue;
+ }
+ attr = fp->ep_attr & EP_ATTR_MASK;
+ if (! found) {
+ found = fp;
+ cur_attr = attr;
+ continue;
+ }
+ /* avoid async out and adaptive in if the other method
+ * supports the same format.
+ * this is a workaround for the case like
+ * M-audio audiophile USB.
+ */
+ if (attr != cur_attr) {
+ if ((attr == EP_ATTR_ASYNC &&
+ subs->direction == SNDRV_PCM_STREAM_PLAYBACK) ||
+ (attr == EP_ATTR_ADAPTIVE &&
+ subs->direction == SNDRV_PCM_STREAM_CAPTURE))
+ continue;
+ if ((cur_attr == EP_ATTR_ASYNC &&
+ subs->direction == SNDRV_PCM_STREAM_PLAYBACK) ||
+ (cur_attr == EP_ATTR_ADAPTIVE &&
+ subs->direction == SNDRV_PCM_STREAM_CAPTURE)) {
+ found = fp;
+ cur_attr = attr;
+ continue;
+ }
+ }
+ /* find the format with the largest max. packet size */
+ if (fp->maxpacksize > found->maxpacksize) {
+ found = fp;
+ cur_attr = attr;
+ }
+ }
+ return found;
+}
+
+
+/*
+ * initialize the picth control and sample rate
+ */
+static int init_usb_pitch(struct usb_device *dev, int iface,
+ struct usb_host_interface *alts,
+ struct audioformat *fmt)
+{
+ unsigned int ep;
+ unsigned char data[1];
+ int err;
+
+ ep = get_endpoint(alts, 0)->bEndpointAddress;
+ /* if endpoint has pitch control, enable it */
+ if (fmt->attributes & EP_CS_ATTR_PITCH_CONTROL) {
+ data[0] = 1;
+ if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR,
+ USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
+ PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) {
+ snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n",
+ dev->devnum, iface, ep);
+ return err;
+ }
+ }
+ return 0;
+}
+
+static int init_usb_sample_rate(struct usb_device *dev, int iface,
+ struct usb_host_interface *alts,
+ struct audioformat *fmt, int rate)
+{
+ unsigned int ep;
+ unsigned char data[3];
+ int err;
+
+ ep = get_endpoint(alts, 0)->bEndpointAddress;
+ /* if endpoint has sampling rate control, set it */
+ if (fmt->attributes & EP_CS_ATTR_SAMPLE_RATE) {
+ int crate;
+ data[0] = rate;
+ data[1] = rate >> 8;
+ data[2] = rate >> 16;
+ if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR,
+ USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
+ SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) {
+ snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep 0x%x\n",
+ dev->devnum, iface, fmt->altsetting, rate, ep);
+ return err;
+ }
+ if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR,
+ USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN,
+ SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) {
+ snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep 0x%x\n",
+ dev->devnum, iface, fmt->altsetting, ep);
+ return 0; /* some devices don't support reading */
+ }
+ crate = data[0] | (data[1] << 8) | (data[2] << 16);
+ if (crate != rate) {
+ snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+ // runtime->rate = crate;
+ }
+ }
+ return 0;
+}
+
+/*
+ * find a matching format and set up the interface
+ */
+static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
+{
+ struct usb_device *dev = subs->dev;
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ struct usb_interface *iface;
+ unsigned int ep, attr;
+ int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+ int err;
+
+ iface = usb_ifnum_to_if(dev, fmt->iface);
+ if (WARN_ON(!iface))
+ return -EINVAL;
+ alts = &iface->altsetting[fmt->altset_idx];
+ altsd = get_iface_desc(alts);
+ if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting))
+ return -EINVAL;
+
+ if (fmt == subs->cur_audiofmt)
+ return 0;
+
+ /* close the old interface */
+ if (subs->interface >= 0 && subs->interface != fmt->iface) {
+ if (usb_set_interface(subs->dev, subs->interface, 0) < 0) {
+ snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n",
+ dev->devnum, fmt->iface, fmt->altsetting);
+ return -EIO;
+ }
+ subs->interface = -1;
+ subs->format = 0;
+ }
+
+ /* set interface */
+ if (subs->interface != fmt->iface || subs->format != fmt->altset_idx) {
+ if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) {
+ snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n",
+ dev->devnum, fmt->iface, fmt->altsetting);
+ return -EIO;
+ }
+ snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting);
+ subs->interface = fmt->iface;
+ subs->format = fmt->altset_idx;
+ }
+
+ /* create a data pipe */
+ ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK;
+ if (is_playback)
+ subs->datapipe = usb_sndisocpipe(dev, ep);
+ else
+ subs->datapipe = usb_rcvisocpipe(dev, ep);
+ if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH &&
+ get_endpoint(alts, 0)->bInterval >= 1 &&
+ get_endpoint(alts, 0)->bInterval <= 4)
+ subs->datainterval = get_endpoint(alts, 0)->bInterval - 1;
+ else
+ subs->datainterval = 0;
+ subs->syncpipe = subs->syncinterval = 0;
+ subs->maxpacksize = fmt->maxpacksize;
+ subs->fill_max = 0;
+
+ /* we need a sync pipe in async OUT or adaptive IN mode */
+ /* check the number of EP, since some devices have broken
+ * descriptors which fool us. if it has only one EP,
+ * assume it as adaptive-out or sync-in.
+ */
+ attr = fmt->ep_attr & EP_ATTR_MASK;
+ if (((is_playback && attr == EP_ATTR_ASYNC) ||
+ (! is_playback && attr == EP_ATTR_ADAPTIVE)) &&
+ altsd->bNumEndpoints >= 2) {
+ /* check sync-pipe endpoint */
+ /* ... and check descriptor size before accessing bSynchAddress
+ because there is a version of the SB Audigy 2 NX firmware lacking
+ the audio fields in the endpoint descriptors */
+ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 ||
+ (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ get_endpoint(alts, 1)->bSynchAddress != 0)) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
+ dev->devnum, fmt->iface, fmt->altsetting);
+ return -EINVAL;
+ }
+ ep = get_endpoint(alts, 1)->bEndpointAddress;
+ if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
+ (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+ snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
+ dev->devnum, fmt->iface, fmt->altsetting);
+ return -EINVAL;
+ }
+ ep &= USB_ENDPOINT_NUMBER_MASK;
+ if (is_playback)
+ subs->syncpipe = usb_rcvisocpipe(dev, ep);
+ else
+ subs->syncpipe = usb_sndisocpipe(dev, ep);
+ if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ get_endpoint(alts, 1)->bRefresh >= 1 &&
+ get_endpoint(alts, 1)->bRefresh <= 9)
+ subs->syncinterval = get_endpoint(alts, 1)->bRefresh;
+ else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
+ subs->syncinterval = 1;
+ else if (get_endpoint(alts, 1)->bInterval >= 1 &&
+ get_endpoint(alts, 1)->bInterval <= 16)
+ subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1;
+ else
+ subs->syncinterval = 3;
+ }
+
+ /* always fill max packet size */
+ if (fmt->attributes & EP_CS_ATTR_FILL_MAX)
+ subs->fill_max = 1;
+
+ if ((err = init_usb_pitch(dev, subs->interface, alts, fmt)) < 0)
+ return err;
+
+ subs->cur_audiofmt = fmt;
+
+#if 0
+ printk("setting done: format = %d, rate = %d..%d, channels = %d\n",
+ fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels);
+ printk(" datapipe = 0x%0x, syncpipe = 0x%0x\n",
+ subs->datapipe, subs->syncpipe);
+#endif
+
+ return 0;
+}
+
+/*
+ * hw_params callback
+ *
+ * allocate a buffer and set the given audio format.
+ *
+ * so far we use a physically linear buffer although packetize transfer
+ * doesn't need a continuous area.
+ * if sg buffer is supported on the later version of alsa, we'll follow
+ * that.
+ */
+static int snd_usb_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+ struct audioformat *fmt;
+ unsigned int channels, rate, format;
+ int ret, changed;
+
+ ret = snd_pcm_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (ret < 0)
+ return ret;
+
+ format = params_format(hw_params);
+ rate = params_rate(hw_params);
+ channels = params_channels(hw_params);
+ fmt = find_format(subs, format, rate, channels);
+ if (!fmt) {
+ snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n",
+ format, rate, channels);
+ return -EINVAL;
+ }
+
+ changed = subs->cur_audiofmt != fmt ||
+ subs->period_bytes != params_period_bytes(hw_params) ||
+ subs->cur_rate != rate;
+ if ((ret = set_format(subs, fmt)) < 0)
+ return ret;
+
+ if (subs->cur_rate != rate) {
+ struct usb_host_interface *alts;
+ struct usb_interface *iface;
+ iface = usb_ifnum_to_if(subs->dev, fmt->iface);
+ alts = &iface->altsetting[fmt->altset_idx];
+ ret = init_usb_sample_rate(subs->dev, subs->interface, alts, fmt, rate);
+ if (ret < 0)
+ return ret;
+ subs->cur_rate = rate;
+ }
+
+ if (changed) {
+ /* format changed */
+ release_substream_urbs(subs, 0);
+ /* influenced: period_bytes, channels, rate, format, */
+ ret = init_substream_urbs(subs, params_period_bytes(hw_params),
+ params_rate(hw_params),
+ snd_pcm_format_physical_width(params_format(hw_params)) * params_channels(hw_params));
+ }
+
+ return ret;
+}
+
+/*
+ * hw_free callback
+ *
+ * reset the audio format and release the buffer
+ */
+static int snd_usb_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+
+ subs->cur_audiofmt = NULL;
+ subs->cur_rate = 0;
+ subs->period_bytes = 0;
+ if (!subs->stream->chip->shutdown)
+ release_substream_urbs(subs, 0);
+ return snd_pcm_free_vmalloc_buffer(substream);
+}
+
+/*
+ * prepare callback
+ *
+ * only a few subtle things...
+ */
+static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usb_substream *subs = runtime->private_data;
+
+ if (! subs->cur_audiofmt) {
+ snd_printk(KERN_ERR "usbaudio: no format is specified!\n");
+ return -ENXIO;
+ }
+
+ /* some unit conversions in runtime */
+ subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize);
+ subs->curframesize = bytes_to_frames(runtime, subs->curpacksize);
+
+ /* reset the pointer */
+ subs->hwptr_done = 0;
+ subs->transfer_done = 0;
+ subs->phase = 0;
+
+ /* clear urbs (to be sure) */
+ deactivate_urbs(subs, 0, 1);
+ wait_clear_urbs(subs);
+
+ /* for playback, submit the URBs now; otherwise, the first hwptr_done
+ * updates for all URBs would happen at the same time when starting */
+ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+ subs->ops.prepare = prepare_nodata_playback_urb;
+ return start_urbs(subs, runtime);
+ } else
+ return 0;
+}
+
+static struct snd_pcm_hardware snd_usb_hardware =
+{
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE,
+ .buffer_bytes_max = 1024 * 1024,
+ .period_bytes_min = 64,
+ .period_bytes_max = 512 * 1024,
+ .periods_min = 2,
+ .periods_max = 1024,
+};
+
+/*
+ * h/w constraints
+ */
+
+#ifdef HW_CONST_DEBUG
+#define hwc_debug(fmt, args...) printk(KERN_DEBUG fmt, ##args)
+#else
+#define hwc_debug(fmt, args...) /**/
+#endif
+
+static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audioformat *fp)
+{
+ struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+ /* check the format */
+ if (!snd_mask_test(fmts, fp->format)) {
+ hwc_debug(" > check: no supported format %d\n", fp->format);
+ return 0;
+ }
+ /* check the channels */
+ if (fp->channels < ct->min || fp->channels > ct->max) {
+ hwc_debug(" > check: no valid channels %d (%d/%d)\n", fp->channels, ct->min, ct->max);
+ return 0;
+ }
+ /* check the rate is within the range */
+ if (fp->rate_min > it->max || (fp->rate_min == it->max && it->openmax)) {
+ hwc_debug(" > check: rate_min %d > max %d\n", fp->rate_min, it->max);
+ return 0;
+ }
+ if (fp->rate_max < it->min || (fp->rate_max == it->min && it->openmin)) {
+ hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min);
+ return 0;
+ }
+ return 1;
+}
+
+static int hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ struct list_head *p;
+ struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ unsigned int rmin, rmax;
+ int changed;
+
+ hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max);
+ changed = 0;
+ rmin = rmax = 0;
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *fp;
+ fp = list_entry(p, struct audioformat, list);
+ if (!hw_check_valid_format(params, fp))
+ continue;
+ if (changed++) {
+ if (rmin > fp->rate_min)
+ rmin = fp->rate_min;
+ if (rmax < fp->rate_max)
+ rmax = fp->rate_max;
+ } else {
+ rmin = fp->rate_min;
+ rmax = fp->rate_max;
+ }
+ }
+
+ if (!changed) {
+ hwc_debug(" --> get empty\n");
+ it->empty = 1;
+ return -EINVAL;
+ }
+
+ changed = 0;
+ if (it->min < rmin) {
+ it->min = rmin;
+ it->openmin = 0;
+ changed = 1;
+ }
+ if (it->max > rmax) {
+ it->max = rmax;
+ it->openmax = 0;
+ changed = 1;
+ }
+ if (snd_interval_checkempty(it)) {
+ it->empty = 1;
+ return -EINVAL;
+ }
+ hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed);
+ return changed;
+}
+
+
+static int hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ struct list_head *p;
+ struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ unsigned int rmin, rmax;
+ int changed;
+
+ hwc_debug("hw_rule_channels: (%d,%d)\n", it->min, it->max);
+ changed = 0;
+ rmin = rmax = 0;
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *fp;
+ fp = list_entry(p, struct audioformat, list);
+ if (!hw_check_valid_format(params, fp))
+ continue;
+ if (changed++) {
+ if (rmin > fp->channels)
+ rmin = fp->channels;
+ if (rmax < fp->channels)
+ rmax = fp->channels;
+ } else {
+ rmin = fp->channels;
+ rmax = fp->channels;
+ }
+ }
+
+ if (!changed) {
+ hwc_debug(" --> get empty\n");
+ it->empty = 1;
+ return -EINVAL;
+ }
+
+ changed = 0;
+ if (it->min < rmin) {
+ it->min = rmin;
+ it->openmin = 0;
+ changed = 1;
+ }
+ if (it->max > rmax) {
+ it->max = rmax;
+ it->openmax = 0;
+ changed = 1;
+ }
+ if (snd_interval_checkempty(it)) {
+ it->empty = 1;
+ return -EINVAL;
+ }
+ hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed);
+ return changed;
+}
+
+static int hw_rule_format(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_usb_substream *subs = rule->private;
+ struct list_head *p;
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ u64 fbits;
+ u32 oldbits[2];
+ int changed;
+
+ hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]);
+ fbits = 0;
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *fp;
+ fp = list_entry(p, struct audioformat, list);
+ if (!hw_check_valid_format(params, fp))
+ continue;
+ fbits |= (1ULL << fp->format);
+ }
+
+ oldbits[0] = fmt->bits[0];
+ oldbits[1] = fmt->bits[1];
+ fmt->bits[0] &= (u32)fbits;
+ fmt->bits[1] &= (u32)(fbits >> 32);
+ if (!fmt->bits[0] && !fmt->bits[1]) {
+ hwc_debug(" --> get empty\n");
+ return -EINVAL;
+ }
+ changed = (oldbits[0] != fmt->bits[0] || oldbits[1] != fmt->bits[1]);
+ hwc_debug(" --> %x:%x (changed = %d)\n", fmt->bits[0], fmt->bits[1], changed);
+ return changed;
+}
+
+#define MAX_MASK 64
+
+/*
+ * check whether the registered audio formats need special hw-constraints
+ */
+static int check_hw_params_convention(struct snd_usb_substream *subs)
+{
+ int i;
+ u32 *channels;
+ u32 *rates;
+ u32 cmaster, rmaster;
+ u32 rate_min = 0, rate_max = 0;
+ struct list_head *p;
+ int err = 1;
+
+ channels = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL);
+ rates = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL);
+ if (!channels || !rates) {
+ err = -ENOMEM;
+ goto __out;
+ }
+
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *f;
+ f = list_entry(p, struct audioformat, list);
+ /* unconventional channels? */
+ if (f->channels > 32)
+ goto __out;
+ /* continuous rate min/max matches? */
+ if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) {
+ if (rate_min && f->rate_min != rate_min)
+ goto __out;
+ if (rate_max && f->rate_max != rate_max)
+ goto __out;
+ rate_min = f->rate_min;
+ rate_max = f->rate_max;
+ }
+ /* combination of continuous rates and fixed rates? */
+ if (rates[f->format] & SNDRV_PCM_RATE_CONTINUOUS) {
+ if (f->rates != rates[f->format])
+ goto __out;
+ }
+ if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) {
+ if (rates[f->format] && rates[f->format] != f->rates)
+ goto __out;
+ }
+ channels[f->format] |= (1 << f->channels);
+ rates[f->format] |= f->rates;
+ /* needs knot? */
+ if (f->rates & SNDRV_PCM_RATE_KNOT)
+ goto __out;
+ }
+ /* check whether channels and rates match for all formats */
+ cmaster = rmaster = 0;
+ for (i = 0; i < MAX_MASK; i++) {
+ if (cmaster != channels[i] && cmaster && channels[i])
+ goto __out;
+ if (rmaster != rates[i] && rmaster && rates[i])
+ goto __out;
+ if (channels[i])
+ cmaster = channels[i];
+ if (rates[i])
+ rmaster = rates[i];
+ }
+ /* check whether channels match for all distinct rates */
+ memset(channels, 0, MAX_MASK * sizeof(u32));
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *f;
+ f = list_entry(p, struct audioformat, list);
+ if (f->rates & SNDRV_PCM_RATE_CONTINUOUS)
+ continue;
+ for (i = 0; i < 32; i++) {
+ if (f->rates & (1 << i))
+ channels[i] |= (1 << f->channels);
+ }
+ }
+ cmaster = 0;
+ for (i = 0; i < 32; i++) {
+ if (cmaster != channels[i] && cmaster && channels[i])
+ goto __out;
+ if (channels[i])
+ cmaster = channels[i];
+ }
+ err = 0;
+
+ __out:
+ kfree(channels);
+ kfree(rates);
+ return err;
+}
+
+/*
+ * If the device supports unusual bit rates, does the request meet these?
+ */
+static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime,
+ struct snd_usb_substream *subs)
+{
+ struct audioformat *fp;
+ int count = 0, needs_knot = 0;
+ int err;
+
+ list_for_each_entry(fp, &subs->fmt_list, list) {
+ if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)
+ return 0;
+ count += fp->nr_rates;
+ if (fp->rates & SNDRV_PCM_RATE_KNOT)
+ needs_knot = 1;
+ }
+ if (!needs_knot)
+ return 0;
+
+ subs->rate_list.count = count;
+ subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL);
+ subs->rate_list.mask = 0;
+ count = 0;
+ list_for_each_entry(fp, &subs->fmt_list, list) {
+ int i;
+ for (i = 0; i < fp->nr_rates; i++)
+ subs->rate_list.list[count++] = fp->rate_table[i];
+ }
+ err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &subs->rate_list);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * set up the runtime hardware information.
+ */
+
+static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs)
+{
+ struct list_head *p;
+ int err;
+
+ runtime->hw.formats = subs->formats;
+
+ runtime->hw.rate_min = 0x7fffffff;
+ runtime->hw.rate_max = 0;
+ runtime->hw.channels_min = 256;
+ runtime->hw.channels_max = 0;
+ runtime->hw.rates = 0;
+ /* check min/max rates and channels */
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *fp;
+ fp = list_entry(p, struct audioformat, list);
+ runtime->hw.rates |= fp->rates;
+ if (runtime->hw.rate_min > fp->rate_min)
+ runtime->hw.rate_min = fp->rate_min;
+ if (runtime->hw.rate_max < fp->rate_max)
+ runtime->hw.rate_max = fp->rate_max;
+ if (runtime->hw.channels_min > fp->channels)
+ runtime->hw.channels_min = fp->channels;
+ if (runtime->hw.channels_max < fp->channels)
+ runtime->hw.channels_max = fp->channels;
+ if (fp->fmt_type == USB_FORMAT_TYPE_II && fp->frame_size > 0) {
+ /* FIXME: there might be more than one audio formats... */
+ runtime->hw.period_bytes_min = runtime->hw.period_bytes_max =
+ fp->frame_size;
+ }
+ }
+
+ /* set the period time minimum 1ms */
+ /* FIXME: high-speed mode allows 125us minimum period, but many parts
+ * in the current code assume the 1ms period.
+ */
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME,
+ 1000 * MIN_PACKS_URB,
+ /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX);
+
+ err = check_hw_params_convention(subs);
+ if (err < 0)
+ return err;
+ else if (err) {
+ hwc_debug("setting extra hw constraints...\n");
+ if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_rate, subs,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ -1)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_channels, subs,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ SNDRV_PCM_HW_PARAM_RATE,
+ -1)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_format, subs,
+ SNDRV_PCM_HW_PARAM_RATE,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ -1)) < 0)
+ return err;
+ if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction)
+{
+ struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usb_substream *subs = &as->substream[direction];
+
+ subs->interface = -1;
+ subs->format = 0;
+ runtime->hw = snd_usb_hardware;
+ runtime->private_data = subs;
+ subs->pcm_substream = substream;
+ return setup_hw_info(runtime, subs);
+}
+
+static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
+{
+ struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
+ struct snd_usb_substream *subs = &as->substream[direction];
+
+ if (subs->interface >= 0) {
+ usb_set_interface(subs->dev, subs->interface, 0);
+ subs->interface = -1;
+ }
+ subs->pcm_substream = NULL;
+ return 0;
+}
+
+static int snd_usb_playback_open(struct snd_pcm_substream *substream)
+{
+ return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK);
+}
+
+static int snd_usb_playback_close(struct snd_pcm_substream *substream)
+{
+ return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK);
+}
+
+static int snd_usb_capture_open(struct snd_pcm_substream *substream)
+{
+ return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE);
+}
+
+static int snd_usb_capture_close(struct snd_pcm_substream *substream)
+{
+ return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE);
+}
+
+static struct snd_pcm_ops snd_usb_playback_ops = {
+ .open = snd_usb_playback_open,
+ .close = snd_usb_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_usb_hw_params,
+ .hw_free = snd_usb_hw_free,
+ .prepare = snd_usb_pcm_prepare,
+ .trigger = snd_usb_pcm_playback_trigger,
+ .pointer = snd_usb_pcm_pointer,
+ .page = snd_pcm_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops snd_usb_capture_ops = {
+ .open = snd_usb_capture_open,
+ .close = snd_usb_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_usb_hw_params,
+ .hw_free = snd_usb_hw_free,
+ .prepare = snd_usb_pcm_prepare,
+ .trigger = snd_usb_pcm_capture_trigger,
+ .pointer = snd_usb_pcm_pointer,
+ .page = snd_pcm_get_vmalloc_page,
+};
+
+
+
+/*
+ * helper functions
+ */
+
+/*
+ * combine bytes and get an integer value
+ */
+unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size)
+{
+ switch (size) {
+ case 1: return *bytes;
+ case 2: return combine_word(bytes);
+ case 3: return combine_triple(bytes);
+ case 4: return combine_quad(bytes);
+ default: return 0;
+ }
+}
+
+/*
+ * parse descriptor buffer and return the pointer starting the given
+ * descriptor type.
+ */
+void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype)
+{
+ u8 *p, *end, *next;
+
+ p = descstart;
+ end = p + desclen;
+ for (; p < end;) {
+ if (p[0] < 2)
+ return NULL;
+ next = p + p[0];
+ if (next > end)
+ return NULL;
+ if (p[1] == dtype && (!after || (void *)p > after)) {
+ return p;
+ }
+ p = next;
+ }
+ return NULL;
+}
+
+/*
+ * find a class-specified interface descriptor with the given subtype.
+ */
+void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype)
+{
+ unsigned char *p = after;
+
+ while ((p = snd_usb_find_desc(buffer, buflen, p,
+ USB_DT_CS_INTERFACE)) != NULL) {
+ if (p[0] >= 3 && p[2] == dsubtype)
+ return p;
+ }
+ return NULL;
+}
+
+/*
+ * Wrapper for usb_control_msg().
+ * Allocates a temp buffer to prevent dmaing from/to the stack.
+ */
+int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request,
+ __u8 requesttype, __u16 value, __u16 index, void *data,
+ __u16 size, int timeout)
+{
+ int err;
+ void *buf = NULL;
+
+ if (size > 0) {
+ buf = kmemdup(data, size, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+ }
+ err = usb_control_msg(dev, pipe, request, requesttype,
+ value, index, buf, size, timeout);
+ if (size > 0) {
+ memcpy(data, buf, size);
+ kfree(buf);
+ }
+ return err;
+}
+
+
+/*
+ * entry point for linux usb interface
+ */
+
+static int usb_audio_probe(struct usb_interface *intf,
+ const struct usb_device_id *id);
+static void usb_audio_disconnect(struct usb_interface *intf);
+
+#ifdef CONFIG_PM
+static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message);
+static int usb_audio_resume(struct usb_interface *intf);
+#else
+#define usb_audio_suspend NULL
+#define usb_audio_resume NULL
+#endif
+
+static struct usb_device_id usb_audio_ids [] = {
+#include "usbquirks.h"
+ { .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS),
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL },
+ { } /* Terminating entry */
+};
+
+MODULE_DEVICE_TABLE (usb, usb_audio_ids);
+
+static struct usb_driver usb_audio_driver = {
+ .name = "snd-usb-audio",
+ .probe = usb_audio_probe,
+ .disconnect = usb_audio_disconnect,
+ .suspend = usb_audio_suspend,
+ .resume = usb_audio_resume,
+ .id_table = usb_audio_ids,
+};
+
+
+#if defined(CONFIG_PROC_FS) && defined(CONFIG_SND_VERBOSE_PROCFS)
+
+/*
+ * proc interface for list the supported pcm formats
+ */
+static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct snd_info_buffer *buffer)
+{
+ struct list_head *p;
+ static char *sync_types[4] = {
+ "NONE", "ASYNC", "ADAPTIVE", "SYNC"
+ };
+
+ list_for_each(p, &subs->fmt_list) {
+ struct audioformat *fp;
+ fp = list_entry(p, struct audioformat, list);
+ snd_iprintf(buffer, " Interface %d\n", fp->iface);
+ snd_iprintf(buffer, " Altset %d\n", fp->altsetting);
+ snd_iprintf(buffer, " Format: 0x%x\n", fp->format);
+ snd_iprintf(buffer, " Channels: %d\n", fp->channels);
+ snd_iprintf(buffer, " Endpoint: %d %s (%s)\n",
+ fp->endpoint & USB_ENDPOINT_NUMBER_MASK,
+ fp->endpoint & USB_DIR_IN ? "IN" : "OUT",
+ sync_types[(fp->ep_attr & EP_ATTR_MASK) >> 2]);
+ if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) {
+ snd_iprintf(buffer, " Rates: %d - %d (continuous)\n",
+ fp->rate_min, fp->rate_max);
+ } else {
+ unsigned int i;
+ snd_iprintf(buffer, " Rates: ");
+ for (i = 0; i < fp->nr_rates; i++) {
+ if (i > 0)
+ snd_iprintf(buffer, ", ");
+ snd_iprintf(buffer, "%d", fp->rate_table[i]);
+ }
+ snd_iprintf(buffer, "\n");
+ }
+ // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize);
+ // snd_iprintf(buffer, " EP Attribute = 0x%x\n", fp->attributes);
+ }
+}
+
+static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer)
+{
+ if (subs->running) {
+ unsigned int i;
+ snd_iprintf(buffer, " Status: Running\n");
+ snd_iprintf(buffer, " Interface = %d\n", subs->interface);
+ snd_iprintf(buffer, " Altset = %d\n", subs->format);
+ snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs);
+ for (i = 0; i < subs->nurbs; i++)
+ snd_iprintf(buffer, "%d ", subs->dataurb[i].packets);
+ snd_iprintf(buffer, "]\n");
+ snd_iprintf(buffer, " Packet Size = %d\n", subs->curpacksize);
+ snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n",
+ snd_usb_get_speed(subs->dev) == USB_SPEED_FULL
+ ? get_full_speed_hz(subs->freqm)
+ : get_high_speed_hz(subs->freqm),
+ subs->freqm >> 16, subs->freqm & 0xffff);
+ } else {
+ snd_iprintf(buffer, " Status: Stop\n");
+ }
+}
+
+static void proc_pcm_format_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_usb_stream *stream = entry->private_data;
+
+ snd_iprintf(buffer, "%s : %s\n", stream->chip->card->longname, stream->pcm->name);
+
+ if (stream->substream[SNDRV_PCM_STREAM_PLAYBACK].num_formats) {
+ snd_iprintf(buffer, "\nPlayback:\n");
+ proc_dump_substream_status(&stream->substream[SNDRV_PCM_STREAM_PLAYBACK], buffer);
+ proc_dump_substream_formats(&stream->substream[SNDRV_PCM_STREAM_PLAYBACK], buffer);
+ }
+ if (stream->substream[SNDRV_PCM_STREAM_CAPTURE].num_formats) {
+ snd_iprintf(buffer, "\nCapture:\n");
+ proc_dump_substream_status(&stream->substream[SNDRV_PCM_STREAM_CAPTURE], buffer);
+ proc_dump_substream_formats(&stream->substream[SNDRV_PCM_STREAM_CAPTURE], buffer);
+ }
+}
+
+static void proc_pcm_format_add(struct snd_usb_stream *stream)
+{
+ struct snd_info_entry *entry;
+ char name[32];
+ struct snd_card *card = stream->chip->card;
+
+ sprintf(name, "stream%d", stream->pcm_index);
+ if (!snd_card_proc_new(card, name, &entry))
+ snd_info_set_text_ops(entry, stream, proc_pcm_format_read);
+}
+
+#else
+
+static inline void proc_pcm_format_add(struct snd_usb_stream *stream)
+{
+}
+
+#endif
+
+/*
+ * initialize the substream instance.
+ */
+
+static void init_substream(struct snd_usb_stream *as, int stream, struct audioformat *fp)
+{
+ struct snd_usb_substream *subs = &as->substream[stream];
+
+ INIT_LIST_HEAD(&subs->fmt_list);
+ spin_lock_init(&subs->lock);
+
+ subs->stream = as;
+ subs->direction = stream;
+ subs->dev = as->chip->dev;
+ if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) {
+ subs->ops = audio_urb_ops[stream];
+ } else {
+ subs->ops = audio_urb_ops_high_speed[stream];
+ switch (as->chip->usb_id) {
+ case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
+ case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
+ case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
+ subs->ops.retire_sync = retire_playback_sync_urb_hs_emu;
+ break;
+ }
+ }
+ snd_pcm_set_ops(as->pcm, stream,
+ stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &snd_usb_playback_ops : &snd_usb_capture_ops);
+
+ list_add_tail(&fp->list, &subs->fmt_list);
+ subs->formats |= 1ULL << fp->format;
+ subs->endpoint = fp->endpoint;
+ subs->num_formats++;
+ subs->fmt_type = fp->fmt_type;
+}
+
+
+/*
+ * free a substream
+ */
+static void free_substream(struct snd_usb_substream *subs)
+{
+ struct list_head *p, *n;
+
+ if (!subs->num_formats)
+ return; /* not initialized */
+ list_for_each_safe(p, n, &subs->fmt_list) {
+ struct audioformat *fp = list_entry(p, struct audioformat, list);
+ kfree(fp->rate_table);
+ kfree(fp);
+ }
+ kfree(subs->rate_list.list);
+}
+
+
+/*
+ * free a usb stream instance
+ */
+static void snd_usb_audio_stream_free(struct snd_usb_stream *stream)
+{
+ free_substream(&stream->substream[0]);
+ free_substream(&stream->substream[1]);
+ list_del(&stream->list);
+ kfree(stream);
+}
+
+static void snd_usb_audio_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_usb_stream *stream = pcm->private_data;
+ if (stream) {
+ stream->pcm = NULL;
+ snd_usb_audio_stream_free(stream);
+ }
+}
+
+
+/*
+ * add this endpoint to the chip instance.
+ * if a stream with the same endpoint already exists, append to it.
+ * if not, create a new pcm stream.
+ */
+static int add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp)
+{
+ struct list_head *p;
+ struct snd_usb_stream *as;
+ struct snd_usb_substream *subs;
+ struct snd_pcm *pcm;
+ int err;
+
+ list_for_each(p, &chip->pcm_list) {
+ as = list_entry(p, struct snd_usb_stream, list);
+ if (as->fmt_type != fp->fmt_type)
+ continue;
+ subs = &as->substream[stream];
+ if (!subs->endpoint)
+ continue;
+ if (subs->endpoint == fp->endpoint) {
+ list_add_tail(&fp->list, &subs->fmt_list);
+ subs->num_formats++;
+ subs->formats |= 1ULL << fp->format;
+ return 0;
+ }
+ }
+ /* look for an empty stream */
+ list_for_each(p, &chip->pcm_list) {
+ as = list_entry(p, struct snd_usb_stream, list);
+ if (as->fmt_type != fp->fmt_type)
+ continue;
+ subs = &as->substream[stream];
+ if (subs->endpoint)
+ continue;
+ err = snd_pcm_new_stream(as->pcm, stream, 1);
+ if (err < 0)
+ return err;
+ init_substream(as, stream, fp);
+ return 0;
+ }
+
+ /* create a new pcm */
+ as = kzalloc(sizeof(*as), GFP_KERNEL);
+ if (!as)
+ return -ENOMEM;
+ as->pcm_index = chip->pcm_devs;
+ as->chip = chip;
+ as->fmt_type = fp->fmt_type;
+ err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs,
+ stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0,
+ stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1,
+ &pcm);
+ if (err < 0) {
+ kfree(as);
+ return err;
+ }
+ as->pcm = pcm;
+ pcm->private_data = as;
+ pcm->private_free = snd_usb_audio_pcm_free;
+ pcm->info_flags = 0;
+ if (chip->pcm_devs > 0)
+ sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs);
+ else
+ strcpy(pcm->name, "USB Audio");
+
+ init_substream(as, stream, fp);
+
+ list_add(&as->list, &chip->pcm_list);
+ chip->pcm_devs++;
+
+ proc_pcm_format_add(as);
+
+ return 0;
+}
+
+
+/*
+ * check if the device uses big-endian samples
+ */
+static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp)
+{
+ switch (chip->usb_id) {
+ case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */
+ if (fp->endpoint & USB_DIR_IN)
+ return 1;
+ break;
+ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+ if (device_setup[chip->index] == 0x00 ||
+ fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3)
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * parse the audio format type I descriptor
+ * and returns the corresponding pcm format
+ *
+ * @dev: usb device
+ * @fp: audioformat record
+ * @format: the format tag (wFormatTag)
+ * @fmt: the format type descriptor
+ */
+static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp,
+ int format, unsigned char *fmt)
+{
+ int pcm_format;
+ int sample_width, sample_bytes;
+
+ /* FIXME: correct endianess and sign? */
+ pcm_format = -1;
+ sample_width = fmt[6];
+ sample_bytes = fmt[5];
+ switch (format) {
+ case 0: /* some devices don't define this correctly... */
+ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n",
+ chip->dev->devnum, fp->iface, fp->altsetting);
+ /* fall-through */
+ case USB_AUDIO_FORMAT_PCM:
+ if (sample_width > sample_bytes * 8) {
+ snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n",
+ chip->dev->devnum, fp->iface, fp->altsetting,
+ sample_width, sample_bytes);
+ }
+ /* check the format byte size */
+ switch (fmt[5]) {
+ case 1:
+ pcm_format = SNDRV_PCM_FORMAT_S8;
+ break;
+ case 2:
+ if (is_big_endian_format(chip, fp))
+ pcm_format = SNDRV_PCM_FORMAT_S16_BE; /* grrr, big endian!! */
+ else
+ pcm_format = SNDRV_PCM_FORMAT_S16_LE;
+ break;
+ case 3:
+ if (is_big_endian_format(chip, fp))
+ pcm_format = SNDRV_PCM_FORMAT_S24_3BE; /* grrr, big endian!! */
+ else
+ pcm_format = SNDRV_PCM_FORMAT_S24_3LE;
+ break;
+ case 4:
+ pcm_format = SNDRV_PCM_FORMAT_S32_LE;
+ break;
+ default:
+ snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n",
+ chip->dev->devnum, fp->iface,
+ fp->altsetting, sample_width, sample_bytes);
+ break;
+ }
+ break;
+ case USB_AUDIO_FORMAT_PCM8:
+ pcm_format = SNDRV_PCM_FORMAT_U8;
+
+ /* Dallas DS4201 workaround: it advertises U8 format, but really
+ supports S8. */
+ if (chip->usb_id == USB_ID(0x04fa, 0x4201))
+ pcm_format = SNDRV_PCM_FORMAT_S8;
+ break;
+ case USB_AUDIO_FORMAT_IEEE_FLOAT:
+ pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE;
+ break;
+ case USB_AUDIO_FORMAT_ALAW:
+ pcm_format = SNDRV_PCM_FORMAT_A_LAW;
+ break;
+ case USB_AUDIO_FORMAT_MU_LAW:
+ pcm_format = SNDRV_PCM_FORMAT_MU_LAW;
+ break;
+ default:
+ snd_printk(KERN_INFO "%d:%u:%d : unsupported format type %d\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, format);
+ break;
+ }
+ return pcm_format;
+}
+
+
+/*
+ * parse the format descriptor and stores the possible sample rates
+ * on the audioformat table.
+ *
+ * @dev: usb device
+ * @fp: audioformat record
+ * @fmt: the format descriptor
+ * @offset: the start offset of descriptor pointing the rate type
+ * (7 for type I and II, 8 for type II)
+ */
+static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioformat *fp,
+ unsigned char *fmt, int offset)
+{
+ int nr_rates = fmt[offset];
+
+ if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n",
+ chip->dev->devnum, fp->iface, fp->altsetting);
+ return -1;
+ }
+
+ if (nr_rates) {
+ /*
+ * build the rate table and bitmap flags
+ */
+ int r, idx;
+
+ fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
+ if (fp->rate_table == NULL) {
+ snd_printk(KERN_ERR "cannot malloc\n");
+ return -1;
+ }
+
+ fp->nr_rates = 0;
+ fp->rate_min = fp->rate_max = 0;
+ for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) {
+ unsigned int rate = combine_triple(&fmt[idx]);
+ if (!rate)
+ continue;
+ /* C-Media CM6501 mislabels its 96 kHz altsetting */
+ if (rate == 48000 && nr_rates == 1 &&
+ (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
+ chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
+ fp->altsetting == 5 && fp->maxpacksize == 392)
+ rate = 96000;
+ fp->rate_table[fp->nr_rates] = rate;
+ if (!fp->rate_min || rate < fp->rate_min)
+ fp->rate_min = rate;
+ if (!fp->rate_max || rate > fp->rate_max)
+ fp->rate_max = rate;
+ fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+ fp->nr_rates++;
+ }
+ if (!fp->nr_rates) {
+ hwc_debug("All rates were zero. Skipping format!\n");
+ return -1;
+ }
+ } else {
+ /* continuous rates */
+ fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
+ fp->rate_min = combine_triple(&fmt[offset + 1]);
+ fp->rate_max = combine_triple(&fmt[offset + 4]);
+ }
+ return 0;
+}
+
+/*
+ * parse the format type I and III descriptors
+ */
+static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp,
+ int format, unsigned char *fmt)
+{
+ int pcm_format;
+
+ if (fmt[3] == USB_FORMAT_TYPE_III) {
+ /* FIXME: the format type is really IECxxx
+ * but we give normal PCM format to get the existing
+ * apps working...
+ */
+ switch (chip->usb_id) {
+
+ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+ if (device_setup[chip->index] == 0x00 &&
+ fp->altsetting == 6)
+ pcm_format = SNDRV_PCM_FORMAT_S16_BE;
+ else
+ pcm_format = SNDRV_PCM_FORMAT_S16_LE;
+ break;
+ default:
+ pcm_format = SNDRV_PCM_FORMAT_S16_LE;
+ }
+ } else {
+ pcm_format = parse_audio_format_i_type(chip, fp, format, fmt);
+ if (pcm_format < 0)
+ return -1;
+ }
+ fp->format = pcm_format;
+ fp->channels = fmt[4];
+ if (fp->channels < 1) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, fp->channels);
+ return -1;
+ }
+ return parse_audio_format_rates(chip, fp, fmt, 7);
+}
+
+/*
+ * prase the format type II descriptor
+ */
+static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp,
+ int format, unsigned char *fmt)
+{
+ int brate, framesize;
+ switch (format) {
+ case USB_AUDIO_FORMAT_AC3:
+ /* FIXME: there is no AC3 format defined yet */
+ // fp->format = SNDRV_PCM_FORMAT_AC3;
+ fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */
+ break;
+ case USB_AUDIO_FORMAT_MPEG:
+ fp->format = SNDRV_PCM_FORMAT_MPEG;
+ break;
+ default:
+ snd_printd(KERN_INFO "%d:%u:%d : unknown format tag 0x%x is detected. processed as MPEG.\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, format);
+ fp->format = SNDRV_PCM_FORMAT_MPEG;
+ break;
+ }
+ fp->channels = 1;
+ brate = combine_word(&fmt[4]); /* fmt[4,5] : wMaxBitRate (in kbps) */
+ framesize = combine_word(&fmt[6]); /* fmt[6,7]: wSamplesPerFrame */
+ snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize);
+ fp->frame_size = framesize;
+ return parse_audio_format_rates(chip, fp, fmt, 8); /* fmt[8..] sample rates */
+}
+
+static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp,
+ int format, unsigned char *fmt, int stream)
+{
+ int err;
+
+ switch (fmt[3]) {
+ case USB_FORMAT_TYPE_I:
+ case USB_FORMAT_TYPE_III:
+ err = parse_audio_format_i(chip, fp, format, fmt);
+ break;
+ case USB_FORMAT_TYPE_II:
+ err = parse_audio_format_ii(chip, fp, format, fmt);
+ break;
+ default:
+ snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]);
+ return -1;
+ }
+ fp->fmt_type = fmt[3];
+ if (err < 0)
+ return err;
+#if 1
+ /* FIXME: temporary hack for extigy/audigy 2 nx/zs */
+ /* extigy apparently supports sample rates other than 48k
+ * but not in ordinary way. so we enable only 48k atm.
+ */
+ if (chip->usb_id == USB_ID(0x041e, 0x3000) ||
+ chip->usb_id == USB_ID(0x041e, 0x3020) ||
+ chip->usb_id == USB_ID(0x041e, 0x3061)) {
+ if (fmt[3] == USB_FORMAT_TYPE_I &&
+ fp->rates != SNDRV_PCM_RATE_48000 &&
+ fp->rates != SNDRV_PCM_RATE_96000)
+ return -1;
+ }
+#endif
+ return 0;
+}
+
+static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno);
+static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
+{
+ struct usb_device *dev;
+ struct usb_interface *iface;
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ int i, altno, err, stream;
+ int format;
+ struct audioformat *fp;
+ unsigned char *fmt, *csep;
+ int num;
+
+ dev = chip->dev;
+
+ /* parse the interface's altsettings */
+ iface = usb_ifnum_to_if(dev, iface_no);
+
+ num = iface->num_altsetting;
+
+ /*
+ * Dallas DS4201 workaround: It presents 5 altsettings, but the last
+ * one misses syncpipe, and does not produce any sound.
+ */
+ if (chip->usb_id == USB_ID(0x04fa, 0x4201))
+ num = 4;
+
+ for (i = 0; i < num; i++) {
+ alts = &iface->altsetting[i];
+ altsd = get_iface_desc(alts);
+ /* skip invalid one */
+ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
+ altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
+ (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING &&
+ altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) ||
+ altsd->bNumEndpoints < 1 ||
+ le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0)
+ continue;
+ /* must be isochronous */
+ if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) !=
+ USB_ENDPOINT_XFER_ISOC)
+ continue;
+ /* check direction */
+ stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ?
+ SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ altno = altsd->bAlternateSetting;
+
+ /* audiophile usb: skip altsets incompatible with device_setup
+ */
+ if (chip->usb_id == USB_ID(0x0763, 0x2003) &&
+ audiophile_skip_setting_quirk(chip, iface_no, altno))
+ continue;
+
+ /* get audio formats */
+ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL);
+ if (!fmt) {
+ snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+
+ if (fmt[0] < 7) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+
+ format = (fmt[6] << 8) | fmt[5]; /* remember the format value */
+
+ /* get format type */
+ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE);
+ if (!fmt) {
+ snd_printk(KERN_ERR "%d:%u:%d : no FORMAT_TYPE desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+ if (fmt[0] < 8) {
+ snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n",
+ dev->devnum, iface_no, altno);
+ continue;
+ }
+
+ csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+ /* Creamware Noah has this descriptor after the 2nd endpoint */
+ if (!csep && altsd->bNumEndpoints >= 2)
+ csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
+ if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) {
+ snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
+ " class specific endpoint descriptor\n",
+ dev->devnum, iface_no, altno);
+ csep = NULL;
+ }
+
+ fp = kzalloc(sizeof(*fp), GFP_KERNEL);
+ if (! fp) {
+ snd_printk(KERN_ERR "cannot malloc\n");
+ return -ENOMEM;
+ }
+
+ fp->iface = iface_no;
+ fp->altsetting = altno;
+ fp->altset_idx = i;
+ fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+ fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
+ fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
+ * (fp->maxpacksize & 0x7ff);
+ fp->attributes = csep ? csep[3] : 0;
+
+ /* some quirks for attributes here */
+
+ switch (chip->usb_id) {
+ case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */
+ /* Optoplay sets the sample rate attribute although
+ * it seems not supporting it in fact.
+ */
+ fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE;
+ break;
+ case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
+ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+ /* doesn't set the sample rate attribute, but supports it */
+ fp->attributes |= EP_CS_ATTR_SAMPLE_RATE;
+ break;
+ case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
+ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
+ an older model 77d:223) */
+ /*
+ * plantronics headset and Griffin iMic have set adaptive-in
+ * although it's really not...
+ */
+ fp->ep_attr &= ~EP_ATTR_MASK;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ fp->ep_attr |= EP_ATTR_ADAPTIVE;
+ else
+ fp->ep_attr |= EP_ATTR_SYNC;
+ break;
+ }
+
+ /* ok, let's parse further... */
+ if (parse_audio_format(chip, fp, format, fmt, stream) < 0) {
+ kfree(fp->rate_table);
+ kfree(fp);
+ continue;
+ }
+
+ snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint);
+ err = add_audio_endpoint(chip, stream, fp);
+ if (err < 0) {
+ kfree(fp->rate_table);
+ kfree(fp);
+ return err;
+ }
+ /* try to set the interface... */
+ usb_set_interface(chip->dev, iface_no, altno);
+ init_usb_pitch(chip->dev, iface_no, alts, fp);
+ init_usb_sample_rate(chip->dev, iface_no, alts, fp, fp->rate_max);
+ }
+ return 0;
+}
+
+
+/*
+ * disconnect streams
+ * called from snd_usb_audio_disconnect()
+ */
+static void snd_usb_stream_disconnect(struct list_head *head)
+{
+ int idx;
+ struct snd_usb_stream *as;
+ struct snd_usb_substream *subs;
+
+ as = list_entry(head, struct snd_usb_stream, list);
+ for (idx = 0; idx < 2; idx++) {
+ subs = &as->substream[idx];
+ if (!subs->num_formats)
+ return;
+ release_substream_urbs(subs, 1);
+ subs->interface = -1;
+ }
+}
+
+/*
+ * parse audio control descriptor and create pcm/midi streams
+ */
+static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
+{
+ struct usb_device *dev = chip->dev;
+ struct usb_host_interface *host_iface;
+ struct usb_interface *iface;
+ unsigned char *p1;
+ int i, j;
+
+ /* find audiocontrol interface */
+ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0];
+ if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) {
+ snd_printk(KERN_ERR "cannot find HEADER\n");
+ return -EINVAL;
+ }
+ if (! p1[7] || p1[0] < 8 + p1[7]) {
+ snd_printk(KERN_ERR "invalid HEADER\n");
+ return -EINVAL;
+ }
+
+ /*
+ * parse all USB audio streaming interfaces
+ */
+ for (i = 0; i < p1[7]; i++) {
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ j = p1[8 + i];
+ iface = usb_ifnum_to_if(dev, j);
+ if (!iface) {
+ snd_printk(KERN_ERR "%d:%u:%d : does not exist\n",
+ dev->devnum, ctrlif, j);
+ continue;
+ }
+ if (usb_interface_claimed(iface)) {
+ snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, j);
+ continue;
+ }
+ alts = &iface->altsetting[0];
+ altsd = get_iface_desc(alts);
+ if ((altsd->bInterfaceClass == USB_CLASS_AUDIO ||
+ altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) &&
+ altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) {
+ if (snd_usb_create_midi_interface(chip, iface, NULL) < 0) {
+ snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j);
+ continue;
+ }
+ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+ continue;
+ }
+ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
+ altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) ||
+ altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) {
+ snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, j, altsd->bInterfaceClass);
+ /* skip non-supported classes */
+ continue;
+ }
+ if (snd_usb_get_speed(dev) == USB_SPEED_LOW) {
+ snd_printk(KERN_ERR "low speed audio streaming not supported\n");
+ continue;
+ }
+ if (! parse_audio_endpoints(chip, j)) {
+ usb_set_interface(dev, j, 0); /* reset the current interface */
+ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+ }
+ }
+
+ return 0;
+}
+
+/*
+ * create a stream for an endpoint/altsetting without proper descriptors
+ */
+static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ struct audioformat *fp;
+ struct usb_host_interface *alts;
+ int stream, err;
+ unsigned *rate_table = NULL;
+
+ fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
+ if (! fp) {
+ snd_printk(KERN_ERR "cannot memdup\n");
+ return -ENOMEM;
+ }
+ if (fp->nr_rates > 0) {
+ rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL);
+ if (!rate_table) {
+ kfree(fp);
+ return -ENOMEM;
+ }
+ memcpy(rate_table, fp->rate_table, sizeof(int) * fp->nr_rates);
+ fp->rate_table = rate_table;
+ }
+
+ stream = (fp->endpoint & USB_DIR_IN)
+ ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ err = add_audio_endpoint(chip, stream, fp);
+ if (err < 0) {
+ kfree(fp);
+ kfree(rate_table);
+ return err;
+ }
+ if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber ||
+ fp->altset_idx >= iface->num_altsetting) {
+ kfree(fp);
+ kfree(rate_table);
+ return -EINVAL;
+ }
+ alts = &iface->altsetting[fp->altset_idx];
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ usb_set_interface(chip->dev, fp->iface, 0);
+ init_usb_pitch(chip->dev, fp->iface, alts, fp);
+ init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max);
+ return 0;
+}
+
+/*
+ * create a stream for an interface with proper descriptors
+ */
+static int create_standard_audio_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ int err;
+
+ alts = &iface->altsetting[0];
+ altsd = get_iface_desc(alts);
+ err = parse_audio_endpoints(chip, altsd->bInterfaceNumber);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot setup if %d: error %d\n",
+ altsd->bInterfaceNumber, err);
+ return err;
+ }
+ /* reset the current interface */
+ usb_set_interface(chip->dev, altsd->bInterfaceNumber, 0);
+ return 0;
+}
+
+/*
+ * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface.
+ * The only way to detect the sample rate is by looking at wMaxPacketSize.
+ */
+static int create_uaxx_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ static const struct audioformat ua_format = {
+ .format = SNDRV_PCM_FORMAT_S24_3LE,
+ .channels = 2,
+ .fmt_type = USB_FORMAT_TYPE_I,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ };
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ struct audioformat *fp;
+ int stream, err;
+
+ /* both PCM and MIDI interfaces have 2 or more altsettings */
+ if (iface->num_altsetting < 2)
+ return -ENXIO;
+ alts = &iface->altsetting[1];
+ altsd = get_iface_desc(alts);
+
+ if (altsd->bNumEndpoints == 2) {
+ static const struct snd_usb_midi_endpoint_info ua700_ep = {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ };
+ static const struct snd_usb_audio_quirk ua700_quirk = {
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = &ua700_ep
+ };
+ static const struct snd_usb_midi_endpoint_info uaxx_ep = {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ };
+ static const struct snd_usb_audio_quirk uaxx_quirk = {
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = &uaxx_ep
+ };
+ if (chip->usb_id == USB_ID(0x0582, 0x002b))
+ return snd_usb_create_midi_interface(chip, iface,
+ &ua700_quirk);
+ else
+ return snd_usb_create_midi_interface(chip, iface,
+ &uaxx_quirk);
+ }
+
+ if (altsd->bNumEndpoints != 1)
+ return -ENXIO;
+
+ fp = kmalloc(sizeof(*fp), GFP_KERNEL);
+ if (!fp)
+ return -ENOMEM;
+ memcpy(fp, &ua_format, sizeof(*fp));
+
+ fp->iface = altsd->bInterfaceNumber;
+ fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+ fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+
+ switch (fp->maxpacksize) {
+ case 0x120:
+ fp->rate_max = fp->rate_min = 44100;
+ break;
+ case 0x138:
+ case 0x140:
+ fp->rate_max = fp->rate_min = 48000;
+ break;
+ case 0x258:
+ case 0x260:
+ fp->rate_max = fp->rate_min = 96000;
+ break;
+ default:
+ snd_printk(KERN_ERR "unknown sample rate\n");
+ kfree(fp);
+ return -ENXIO;
+ }
+
+ stream = (fp->endpoint & USB_DIR_IN)
+ ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ err = add_audio_endpoint(chip, stream, fp);
+ if (err < 0) {
+ kfree(fp);
+ return err;
+ }
+ usb_set_interface(chip->dev, fp->iface, 0);
+ return 0;
+}
+
+/*
+ * Create a stream for an Edirol UA-1000 interface.
+ */
+static int create_ua1000_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ static const struct audioformat ua1000_format = {
+ .format = SNDRV_PCM_FORMAT_S32_LE,
+ .fmt_type = USB_FORMAT_TYPE_I,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ };
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ struct audioformat *fp;
+ int stream, err;
+
+ if (iface->num_altsetting != 2)
+ return -ENXIO;
+ alts = &iface->altsetting[1];
+ altsd = get_iface_desc(alts);
+ if (alts->extralen != 11 || alts->extra[1] != USB_DT_CS_INTERFACE ||
+ altsd->bNumEndpoints != 1)
+ return -ENXIO;
+
+ fp = kmemdup(&ua1000_format, sizeof(*fp), GFP_KERNEL);
+ if (!fp)
+ return -ENOMEM;
+
+ fp->channels = alts->extra[4];
+ fp->iface = altsd->bInterfaceNumber;
+ fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+ fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ fp->rate_max = fp->rate_min = combine_triple(&alts->extra[8]);
+
+ stream = (fp->endpoint & USB_DIR_IN)
+ ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ err = add_audio_endpoint(chip, stream, fp);
+ if (err < 0) {
+ kfree(fp);
+ return err;
+ }
+ /* FIXME: playback must be synchronized to capture */
+ usb_set_interface(chip->dev, fp->iface, 0);
+ return 0;
+}
+
+/*
+ * Create a stream for an Edirol UA-101 interface.
+ * Copy, paste and modify from Edirol UA-1000
+ */
+static int create_ua101_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ static const struct audioformat ua101_format = {
+ .format = SNDRV_PCM_FORMAT_S32_LE,
+ .fmt_type = USB_FORMAT_TYPE_I,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ };
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ struct audioformat *fp;
+ int stream, err;
+
+ if (iface->num_altsetting != 2)
+ return -ENXIO;
+ alts = &iface->altsetting[1];
+ altsd = get_iface_desc(alts);
+ if (alts->extralen != 18 || alts->extra[1] != USB_DT_CS_INTERFACE ||
+ altsd->bNumEndpoints != 1)
+ return -ENXIO;
+
+ fp = kmemdup(&ua101_format, sizeof(*fp), GFP_KERNEL);
+ if (!fp)
+ return -ENOMEM;
+
+ fp->channels = alts->extra[11];
+ fp->iface = altsd->bInterfaceNumber;
+ fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+ fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]);
+
+ stream = (fp->endpoint & USB_DIR_IN)
+ ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+ err = add_audio_endpoint(chip, stream, fp);
+ if (err < 0) {
+ kfree(fp);
+ return err;
+ }
+ /* FIXME: playback must be synchronized to capture */
+ usb_set_interface(chip->dev, fp->iface, 0);
+ return 0;
+}
+
+static int snd_usb_create_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk);
+
+/*
+ * handle the quirks for the contained interfaces
+ */
+static int create_composite_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ int probed_ifnum = get_iface_desc(iface->altsetting)->bInterfaceNumber;
+ int err;
+
+ for (quirk = quirk->data; quirk->ifnum >= 0; ++quirk) {
+ iface = usb_ifnum_to_if(chip->dev, quirk->ifnum);
+ if (!iface)
+ continue;
+ if (quirk->ifnum != probed_ifnum &&
+ usb_interface_claimed(iface))
+ continue;
+ err = snd_usb_create_quirk(chip, iface, quirk);
+ if (err < 0)
+ return err;
+ if (quirk->ifnum != probed_ifnum)
+ usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+ }
+ return 0;
+}
+
+static int ignore_interface_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ return 0;
+}
+
+
+/*
+ * boot quirks
+ */
+
+#define EXTIGY_FIRMWARE_SIZE_OLD 794
+#define EXTIGY_FIRMWARE_SIZE_NEW 483
+
+static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interface *intf)
+{
+ struct usb_host_config *config = dev->actconfig;
+ int err;
+
+ if (le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_OLD ||
+ le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_NEW) {
+ snd_printdd("sending Extigy boot sequence...\n");
+ /* Send message to force it to reconnect with full interface. */
+ err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0),
+ 0x10, 0x43, 0x0001, 0x000a, NULL, 0, 1000);
+ if (err < 0) snd_printdd("error sending boot message: %d\n", err);
+ err = usb_get_descriptor(dev, USB_DT_DEVICE, 0,
+ &dev->descriptor, sizeof(dev->descriptor));
+ config = dev->actconfig;
+ if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err);
+ err = usb_reset_configuration(dev);
+ if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err);
+ snd_printdd("extigy_boot: new boot length = %d\n",
+ le16_to_cpu(get_cfg_desc(config)->wTotalLength));
+ return -ENODEV; /* quit this anyway */
+ }
+ return 0;
+}
+
+static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
+{
+ u8 buf = 1;
+
+ snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a,
+ USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+ 0, 0, &buf, 1, 1000);
+ if (buf == 0) {
+ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29,
+ USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+ 1, 2000, NULL, 0, 1000);
+ return -ENODEV;
+ }
+ return 0;
+}
+
+/*
+ * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely
+ * documented in the device's data sheet.
+ */
+static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value)
+{
+ u8 buf[4];
+ buf[0] = 0x20;
+ buf[1] = value & 0xff;
+ buf[2] = (value >> 8) & 0xff;
+ buf[3] = reg;
+ return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION,
+ USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT,
+ 0, 0, &buf, 4, 1000);
+}
+
+static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
+{
+ /*
+ * Enable line-out driver mode, set headphone source to front
+ * channels, enable stereo mic.
+ */
+ return snd_usb_cm106_write_int_reg(dev, 2, 0x8004);
+}
+
+
+/*
+ * Setup quirks
+ */
+#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */
+#define AUDIOPHILE_SET_DTS 0x02 /* if set, enable DTS Digital Output */
+#define AUDIOPHILE_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
+#define AUDIOPHILE_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */
+#define AUDIOPHILE_SET_DI 0x10 /* if set, enable Digital Input */
+#define AUDIOPHILE_SET_MASK 0x1F /* bit mask for setup value */
+#define AUDIOPHILE_SET_24B_48K_DI 0x19 /* value for 24bits+48KHz+Digital Input */
+#define AUDIOPHILE_SET_24B_48K_NOTDI 0x09 /* value for 24bits+48KHz+No Digital Input */
+#define AUDIOPHILE_SET_16B_48K_DI 0x11 /* value for 16bits+48KHz+Digital Input */
+#define AUDIOPHILE_SET_16B_48K_NOTDI 0x01 /* value for 16bits+48KHz+No Digital Input */
+
+static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip,
+ int iface, int altno)
+{
+ /* Reset ALL ifaces to 0 altsetting.
+ * Call it for every possible altsetting of every interface.
+ */
+ usb_set_interface(chip->dev, iface, 0);
+
+ if (device_setup[chip->index] & AUDIOPHILE_SET) {
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS)
+ && altno != 6)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_96K)
+ && altno != 1)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_24B_48K_DI && altno != 2)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_16B_48K_DI && altno != 4)
+ return 1; /* skip this altsetting */
+ if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) ==
+ AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5)
+ return 1; /* skip this altsetting */
+ }
+ return 0; /* keep this altsetting */
+}
+
+/*
+ * audio-interface quirks
+ *
+ * returns zero if no standard audio/MIDI parsing is needed.
+ * returns a postive value if standard audio/midi interfaces are parsed
+ * after this.
+ * returns a negative value at error.
+ */
+static int snd_usb_create_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ typedef int (*quirk_func_t)(struct snd_usb_audio *, struct usb_interface *,
+ const struct snd_usb_audio_quirk *);
+ static const quirk_func_t quirk_funcs[] = {
+ [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk,
+ [QUIRK_COMPOSITE] = create_composite_quirk,
+ [QUIRK_MIDI_STANDARD_INTERFACE] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_FIXED_ENDPOINT] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_RAW] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_CME] = snd_usb_create_midi_interface,
+ [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
+ [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
+ [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk,
+ [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk,
+ [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk
+ };
+
+ if (quirk->type < QUIRK_TYPE_COUNT) {
+ return quirk_funcs[quirk->type](chip, iface, quirk);
+ } else {
+ snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
+ return -ENXIO;
+ }
+}
+
+
+/*
+ * common proc files to show the usb device info
+ */
+static void proc_audio_usbbus_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_usb_audio *chip = entry->private_data;
+ if (!chip->shutdown)
+ snd_iprintf(buffer, "%03d/%03d\n", chip->dev->bus->busnum, chip->dev->devnum);
+}
+
+static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+{
+ struct snd_usb_audio *chip = entry->private_data;
+ if (!chip->shutdown)
+ snd_iprintf(buffer, "%04x:%04x\n",
+ USB_ID_VENDOR(chip->usb_id),
+ USB_ID_PRODUCT(chip->usb_id));
+}
+
+static void snd_usb_audio_create_proc(struct snd_usb_audio *chip)
+{
+ struct snd_info_entry *entry;
+ if (!snd_card_proc_new(chip->card, "usbbus", &entry))
+ snd_info_set_text_ops(entry, chip, proc_audio_usbbus_read);
+ if (!snd_card_proc_new(chip->card, "usbid", &entry))
+ snd_info_set_text_ops(entry, chip, proc_audio_usbid_read);
+}
+
+/*
+ * free the chip instance
+ *
+ * here we have to do not much, since pcm and controls are already freed
+ *
+ */
+
+static int snd_usb_audio_free(struct snd_usb_audio *chip)
+{
+ kfree(chip);
+ return 0;
+}
+
+static int snd_usb_audio_dev_free(struct snd_device *device)
+{
+ struct snd_usb_audio *chip = device->device_data;
+ return snd_usb_audio_free(chip);
+}
+
+
+/*
+ * create a chip instance and set its names.
+ */
+static int snd_usb_audio_create(struct usb_device *dev, int idx,
+ const struct snd_usb_audio_quirk *quirk,
+ struct snd_usb_audio **rchip)
+{
+ struct snd_card *card;
+ struct snd_usb_audio *chip;
+ int err, len;
+ char component[14];
+ static struct snd_device_ops ops = {
+ .dev_free = snd_usb_audio_dev_free,
+ };
+
+ *rchip = NULL;
+
+ if (snd_usb_get_speed(dev) != USB_SPEED_LOW &&
+ snd_usb_get_speed(dev) != USB_SPEED_FULL &&
+ snd_usb_get_speed(dev) != USB_SPEED_HIGH) {
+ snd_printk(KERN_ERR "unknown device speed %d\n", snd_usb_get_speed(dev));
+ return -ENXIO;
+ }
+
+ card = snd_card_new(index[idx], id[idx], THIS_MODULE, 0);
+ if (card == NULL) {
+ snd_printk(KERN_ERR "cannot create card instance %d\n", idx);
+ return -ENOMEM;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (! chip) {
+ snd_card_free(card);
+ return -ENOMEM;
+ }
+
+ chip->index = idx;
+ chip->dev = dev;
+ chip->card = card;
+ chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
+ le16_to_cpu(dev->descriptor.idProduct));
+ INIT_LIST_HEAD(&chip->pcm_list);
+ INIT_LIST_HEAD(&chip->midi_list);
+ INIT_LIST_HEAD(&chip->mixer_list);
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+ snd_usb_audio_free(chip);
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->driver, "USB-Audio");
+ sprintf(component, "USB%04x:%04x",
+ USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id));
+ snd_component_add(card, component);
+
+ /* retrieve the device string as shortname */
+ if (quirk && quirk->product_name) {
+ strlcpy(card->shortname, quirk->product_name, sizeof(card->shortname));
+ } else {
+ if (!dev->descriptor.iProduct ||
+ usb_string(dev, dev->descriptor.iProduct,
+ card->shortname, sizeof(card->shortname)) <= 0) {
+ /* no name available from anywhere, so use ID */
+ sprintf(card->shortname, "USB Device %#04x:%#04x",
+ USB_ID_VENDOR(chip->usb_id),
+ USB_ID_PRODUCT(chip->usb_id));
+ }
+ }
+
+ /* retrieve the vendor and device strings as longname */
+ if (quirk && quirk->vendor_name) {
+ len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
+ } else {
+ if (dev->descriptor.iManufacturer)
+ len = usb_string(dev, dev->descriptor.iManufacturer,
+ card->longname, sizeof(card->longname));
+ else
+ len = 0;
+ /* we don't really care if there isn't any vendor string */
+ }
+ if (len > 0)
+ strlcat(card->longname, " ", sizeof(card->longname));
+
+ strlcat(card->longname, card->shortname, sizeof(card->longname));
+
+ len = strlcat(card->longname, " at ", sizeof(card->longname));
+
+ if (len < sizeof(card->longname))
+ usb_make_path(dev, card->longname + len, sizeof(card->longname) - len);
+
+ strlcat(card->longname,
+ snd_usb_get_speed(dev) == USB_SPEED_LOW ? ", low speed" :
+ snd_usb_get_speed(dev) == USB_SPEED_FULL ? ", full speed" :
+ ", high speed",
+ sizeof(card->longname));
+
+ snd_usb_audio_create_proc(chip);
+
+ *rchip = chip;
+ return 0;
+}
+
+
+/*
+ * probe the active usb device
+ *
+ * note that this can be called multiple times per a device, when it
+ * includes multiple audio control interfaces.
+ *
+ * thus we check the usb device pointer and creates the card instance
+ * only at the first time. the successive calls of this function will
+ * append the pcm interface to the corresponding card.
+ */
+static void *snd_usb_audio_probe(struct usb_device *dev,
+ struct usb_interface *intf,
+ const struct usb_device_id *usb_id)
+{
+ const struct snd_usb_audio_quirk *quirk = (const struct snd_usb_audio_quirk *)usb_id->driver_info;
+ int i, err;
+ struct snd_usb_audio *chip;
+ struct usb_host_interface *alts;
+ int ifnum;
+ u32 id;
+
+ alts = &intf->altsetting[0];
+ ifnum = get_iface_desc(alts)->bInterfaceNumber;
+ id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
+ le16_to_cpu(dev->descriptor.idProduct));
+
+ if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum)
+ goto __err_val;
+
+ /* SB Extigy needs special boot-up sequence */
+ /* if more models come, this will go to the quirk list. */
+ if (id == USB_ID(0x041e, 0x3000)) {
+ if (snd_usb_extigy_boot_quirk(dev, intf) < 0)
+ goto __err_val;
+ }
+ /* SB Audigy 2 NX needs its own boot-up magic, too */
+ if (id == USB_ID(0x041e, 0x3020)) {
+ if (snd_usb_audigy2nx_boot_quirk(dev) < 0)
+ goto __err_val;
+ }
+
+ /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */
+ if (id == USB_ID(0x10f5, 0x0200)) {
+ if (snd_usb_cm106_boot_quirk(dev) < 0)
+ goto __err_val;
+ }
+
+ /*
+ * found a config. now register to ALSA
+ */
+
+ /* check whether it's already registered */
+ chip = NULL;
+ mutex_lock(&register_mutex);
+ for (i = 0; i < SNDRV_CARDS; i++) {
+ if (usb_chip[i] && usb_chip[i]->dev == dev) {
+ if (usb_chip[i]->shutdown) {
+ snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n");
+ goto __error;
+ }
+ chip = usb_chip[i];
+ break;
+ }
+ }
+ if (! chip) {
+ /* it's a fresh one.
+ * now look for an empty slot and create a new card instance
+ */
+ for (i = 0; i < SNDRV_CARDS; i++)
+ if (enable[i] && ! usb_chip[i] &&
+ (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) &&
+ (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) {
+ if (snd_usb_audio_create(dev, i, quirk, &chip) < 0) {
+ goto __error;
+ }
+ snd_card_set_dev(chip->card, &intf->dev);
+ break;
+ }
+ if (!chip) {
+ printk(KERN_ERR "no available usb audio device\n");
+ goto __error;
+ }
+ }
+
+ err = 1; /* continue */
+ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) {
+ /* need some special handlings */
+ if ((err = snd_usb_create_quirk(chip, intf, quirk)) < 0)
+ goto __error;
+ }
+
+ if (err > 0) {
+ /* create normal USB audio interfaces */
+ if (snd_usb_create_streams(chip, ifnum) < 0 ||
+ snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) {
+ goto __error;
+ }
+ }
+
+ /* we are allowed to call snd_card_register() many times */
+ if (snd_card_register(chip->card) < 0) {
+ goto __error;
+ }
+
+ usb_chip[chip->index] = chip;
+ chip->num_interfaces++;
+ mutex_unlock(&register_mutex);
+ return chip;
+
+ __error:
+ if (chip && !chip->num_interfaces)
+ snd_card_free(chip->card);
+ mutex_unlock(&register_mutex);
+ __err_val:
+ return NULL;
+}
+
+/*
+ * we need to take care of counter, since disconnection can be called also
+ * many times as well as usb_audio_probe().
+ */
+static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
+{
+ struct snd_usb_audio *chip;
+ struct snd_card *card;
+ struct list_head *p;
+
+ if (ptr == (void *)-1L)
+ return;
+
+ chip = ptr;
+ card = chip->card;
+ mutex_lock(&register_mutex);
+ chip->shutdown = 1;
+ chip->num_interfaces--;
+ if (chip->num_interfaces <= 0) {
+ snd_card_disconnect(card);
+ /* release the pcm resources */
+ list_for_each(p, &chip->pcm_list) {
+ snd_usb_stream_disconnect(p);
+ }
+ /* release the midi resources */
+ list_for_each(p, &chip->midi_list) {
+ snd_usbmidi_disconnect(p);
+ }
+ /* release mixer resources */
+ list_for_each(p, &chip->mixer_list) {
+ snd_usb_mixer_disconnect(p);
+ }
+ usb_chip[chip->index] = NULL;
+ mutex_unlock(&register_mutex);
+ snd_card_free_when_closed(card);
+ } else {
+ mutex_unlock(&register_mutex);
+ }
+}
+
+/*
+ * new 2.5 USB kernel API
+ */
+static int usb_audio_probe(struct usb_interface *intf,
+ const struct usb_device_id *id)
+{
+ void *chip;
+ chip = snd_usb_audio_probe(interface_to_usbdev(intf), intf, id);
+ if (chip) {
+ dev_set_drvdata(&intf->dev, chip);
+ return 0;
+ } else
+ return -EIO;
+}
+
+static void usb_audio_disconnect(struct usb_interface *intf)
+{
+ snd_usb_audio_disconnect(interface_to_usbdev(intf),
+ dev_get_drvdata(&intf->dev));
+}
+
+#ifdef CONFIG_PM
+static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
+{
+ struct snd_usb_audio *chip = dev_get_drvdata(&intf->dev);
+ struct list_head *p;
+ struct snd_usb_stream *as;
+
+ if (chip == (void *)-1L)
+ return 0;
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ if (!chip->num_suspended_intf++) {
+ list_for_each(p, &chip->pcm_list) {
+ as = list_entry(p, struct snd_usb_stream, list);
+ snd_pcm_suspend_all(as->pcm);
+ }
+ }
+
+ return 0;
+}
+
+static int usb_audio_resume(struct usb_interface *intf)
+{
+ struct snd_usb_audio *chip = dev_get_drvdata(&intf->dev);
+
+ if (chip == (void *)-1L)
+ return 0;
+ if (--chip->num_suspended_intf)
+ return 0;
+ /*
+ * ALSA leaves material resumption to user space
+ * we just notify
+ */
+
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
+
+ return 0;
+}
+#endif /* CONFIG_PM */
+
+static int __init snd_usb_audio_init(void)
+{
+ if (nrpacks < MIN_PACKS_URB || nrpacks > MAX_PACKS) {
+ printk(KERN_WARNING "invalid nrpacks value.\n");
+ return -EINVAL;
+ }
+ return usb_register(&usb_audio_driver);
+}
+
+
+static void __exit snd_usb_audio_cleanup(void)
+{
+ usb_deregister(&usb_audio_driver);
+}
+
+module_init(snd_usb_audio_init);
+module_exit(snd_usb_audio_cleanup);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
new file mode 100644
index 0000000..36e4f7a
--- /dev/null
+++ b/sound/usb/usbaudio.h
@@ -0,0 +1,251 @@
+#ifndef __USBAUDIO_H
+#define __USBAUDIO_H
+/*
+ * (Tentative) USB Audio Driver for ALSA
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+/*
+ */
+
+#define USB_SUBCLASS_AUDIO_CONTROL 0x01
+#define USB_SUBCLASS_AUDIO_STREAMING 0x02
+#define USB_SUBCLASS_MIDI_STREAMING 0x03
+#define USB_SUBCLASS_VENDOR_SPEC 0xff
+
+#define HEADER 0x01
+#define INPUT_TERMINAL 0x02
+#define OUTPUT_TERMINAL 0x03
+#define MIXER_UNIT 0x04
+#define SELECTOR_UNIT 0x05
+#define FEATURE_UNIT 0x06
+#define PROCESSING_UNIT 0x07
+#define EXTENSION_UNIT 0x08
+
+#define AS_GENERAL 0x01
+#define FORMAT_TYPE 0x02
+#define FORMAT_SPECIFIC 0x03
+
+#define EP_GENERAL 0x01
+
+#define MS_GENERAL 0x01
+#define MIDI_IN_JACK 0x02
+#define MIDI_OUT_JACK 0x03
+
+/* endpoint attributes */
+#define EP_ATTR_MASK 0x0c
+#define EP_ATTR_ASYNC 0x04
+#define EP_ATTR_ADAPTIVE 0x08
+#define EP_ATTR_SYNC 0x0c
+
+/* cs endpoint attributes */
+#define EP_CS_ATTR_SAMPLE_RATE 0x01
+#define EP_CS_ATTR_PITCH_CONTROL 0x02
+#define EP_CS_ATTR_FILL_MAX 0x80
+
+/* Audio Class specific Request Codes */
+
+#define SET_CUR 0x01
+#define GET_CUR 0x81
+#define SET_MIN 0x02
+#define GET_MIN 0x82
+#define SET_MAX 0x03
+#define GET_MAX 0x83
+#define SET_RES 0x04
+#define GET_RES 0x84
+#define SET_MEM 0x05
+#define GET_MEM 0x85
+#define GET_STAT 0xff
+
+/* Terminal Control Selectors */
+
+#define COPY_PROTECT_CONTROL 0x01
+
+/* Endpoint Control Selectors */
+
+#define SAMPLING_FREQ_CONTROL 0x01
+#define PITCH_CONTROL 0x02
+
+/* Format Types */
+#define USB_FORMAT_TYPE_I 0x01
+#define USB_FORMAT_TYPE_II 0x02
+#define USB_FORMAT_TYPE_III 0x03
+
+/* type I */
+#define USB_AUDIO_FORMAT_PCM 0x01
+#define USB_AUDIO_FORMAT_PCM8 0x02
+#define USB_AUDIO_FORMAT_IEEE_FLOAT 0x03
+#define USB_AUDIO_FORMAT_ALAW 0x04
+#define USB_AUDIO_FORMAT_MU_LAW 0x05
+
+/* type II */
+#define USB_AUDIO_FORMAT_MPEG 0x1001
+#define USB_AUDIO_FORMAT_AC3 0x1002
+
+/* type III */
+#define USB_AUDIO_FORMAT_IEC1937_AC3 0x2001
+#define USB_AUDIO_FORMAT_IEC1937_MPEG1_LAYER1 0x2002
+#define USB_AUDIO_FORMAT_IEC1937_MPEG2_NOEXT 0x2003
+#define USB_AUDIO_FORMAT_IEC1937_MPEG2_EXT 0x2004
+#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER1_LS 0x2005
+#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER23_LS 0x2006
+
+
+/* maximum number of endpoints per interface */
+#define MIDI_MAX_ENDPOINTS 2
+
+/* handling of USB vendor/product ID pairs as 32-bit numbers */
+#define USB_ID(vendor, product) (((vendor) << 16) | (product))
+#define USB_ID_VENDOR(id) ((id) >> 16)
+#define USB_ID_PRODUCT(id) ((u16)(id))
+
+/*
+ */
+
+struct snd_usb_audio {
+ int index;
+ struct usb_device *dev;
+ struct snd_card *card;
+ u32 usb_id;
+ int shutdown;
+ int num_interfaces;
+ int num_suspended_intf;
+
+ struct list_head pcm_list; /* list of pcm streams */
+ int pcm_devs;
+
+ struct list_head midi_list; /* list of midi interfaces */
+ int next_midi_device;
+
+ struct list_head mixer_list; /* list of mixer interfaces */
+};
+
+/*
+ * Information about devices with broken descriptors
+ */
+
+/* special values for .ifnum */
+#define QUIRK_NO_INTERFACE -2
+#define QUIRK_ANY_INTERFACE -1
+
+enum quirk_type {
+ QUIRK_IGNORE_INTERFACE,
+ QUIRK_COMPOSITE,
+ QUIRK_MIDI_STANDARD_INTERFACE,
+ QUIRK_MIDI_FIXED_ENDPOINT,
+ QUIRK_MIDI_YAMAHA,
+ QUIRK_MIDI_MIDIMAN,
+ QUIRK_MIDI_NOVATION,
+ QUIRK_MIDI_RAW,
+ QUIRK_MIDI_EMAGIC,
+ QUIRK_MIDI_CME,
+ QUIRK_MIDI_US122L,
+ QUIRK_AUDIO_STANDARD_INTERFACE,
+ QUIRK_AUDIO_FIXED_ENDPOINT,
+ QUIRK_AUDIO_EDIROL_UA1000,
+ QUIRK_AUDIO_EDIROL_UA101,
+ QUIRK_AUDIO_EDIROL_UAXX,
+
+ QUIRK_TYPE_COUNT
+};
+
+struct snd_usb_audio_quirk {
+ const char *vendor_name;
+ const char *product_name;
+ int16_t ifnum;
+ uint16_t type;
+ const void *data;
+};
+
+/* data for QUIRK_MIDI_FIXED_ENDPOINT */
+struct snd_usb_midi_endpoint_info {
+ int8_t out_ep; /* ep number, 0 autodetect */
+ uint8_t out_interval; /* interval for interrupt endpoints */
+ int8_t in_ep;
+ uint8_t in_interval;
+ uint16_t out_cables; /* bitmask */
+ uint16_t in_cables; /* bitmask */
+};
+
+/* for QUIRK_MIDI_YAMAHA, data is NULL */
+
+/* for QUIRK_MIDI_MIDIMAN, data points to a snd_usb_midi_endpoint_info
+ * structure (out_cables and in_cables only) */
+
+/* for QUIRK_COMPOSITE, data points to an array of snd_usb_audio_quirk
+ * structures, terminated with .ifnum = -1 */
+
+/* for QUIRK_AUDIO_FIXED_ENDPOINT, data points to an audioformat structure */
+
+/* for QUIRK_AUDIO/MIDI_STANDARD_INTERFACE, data is NULL */
+
+/* for QUIRK_AUDIO_EDIROL_UA700_UA25/UA1000, data is NULL */
+
+/* for QUIRK_IGNORE_INTERFACE, data is NULL */
+
+/* for QUIRK_MIDI_NOVATION and _RAW, data is NULL */
+
+/* for QUIRK_MIDI_EMAGIC, data points to a snd_usb_midi_endpoint_info
+ * structure (out_cables and in_cables only) */
+
+/* for QUIRK_MIDI_CME, data is NULL */
+
+/*
+ */
+
+#define combine_word(s) ((*s) | ((unsigned int)(s)[1] << 8))
+#define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16))
+#define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24))
+
+unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size);
+
+void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype);
+void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsubtype);
+
+int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe,
+ __u8 request, __u8 requesttype, __u16 value, __u16 index,
+ void *data, __u16 size, int timeout);
+
+int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
+ int ignore_error);
+void snd_usb_mixer_disconnect(struct list_head *p);
+
+int snd_usb_create_midi_interface(struct snd_usb_audio *chip, struct usb_interface *iface,
+ const struct snd_usb_audio_quirk *quirk);
+void snd_usbmidi_input_stop(struct list_head* p);
+void snd_usbmidi_input_start(struct list_head* p);
+void snd_usbmidi_disconnect(struct list_head *p);
+
+/*
+ * retrieve usb_interface descriptor from the host interface
+ * (conditional for compatibility with the older API)
+ */
+#ifndef get_iface_desc
+#define get_iface_desc(iface) (&(iface)->desc)
+#define get_endpoint(alt,ep) (&(alt)->endpoint[ep].desc)
+#define get_ep_desc(ep) (&(ep)->desc)
+#define get_cfg_desc(cfg) (&(cfg)->desc)
+#endif
+
+#ifndef snd_usb_get_speed
+#define snd_usb_get_speed(dev) ((dev)->speed)
+#endif
+
+#endif /* __USBAUDIO_H */
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
new file mode 100644
index 0000000..343f896
--- /dev/null
+++ b/sound/usb/usbmidi.c
@@ -0,0 +1,1841 @@
+/*
+ * usbmidi.c - ALSA USB MIDI driver
+ *
+ * Copyright (c) 2002-2007 Clemens Ladisch
+ * All rights reserved.
+ *
+ * Based on the OSS usb-midi driver by NAGANO Daisuke,
+ * NetBSD's umidi driver by Takuya SHIOZAKI,
+ * the "USB Device Class Definition for MIDI Devices" by Roland
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions, and the following disclaimer,
+ * without modification.
+ * 2. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * Alternatively, this software may be distributed and/or modified under the
+ * terms of the GNU General Public License as published by the Free Software
+ * Foundation; either version 2 of the License, or (at your option) any later
+ * version.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ */
+
+#include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/bitops.h>
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+#include <linux/string.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/timer.h>
+#include <linux/usb.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+#include <sound/asequencer.h>
+#include "usbaudio.h"
+
+
+/*
+ * define this to log all USB packets
+ */
+/* #define DUMP_PACKETS */
+
+/*
+ * how long to wait after some USB errors, so that khubd can disconnect() us
+ * without too many spurious errors
+ */
+#define ERROR_DELAY_JIFFIES (HZ / 10)
+
+
+MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
+MODULE_DESCRIPTION("USB Audio/MIDI helper module");
+MODULE_LICENSE("Dual BSD/GPL");
+
+
+struct usb_ms_header_descriptor {
+ __u8 bLength;
+ __u8 bDescriptorType;
+ __u8 bDescriptorSubtype;
+ __u8 bcdMSC[2];
+ __le16 wTotalLength;
+} __attribute__ ((packed));
+
+struct usb_ms_endpoint_descriptor {
+ __u8 bLength;
+ __u8 bDescriptorType;
+ __u8 bDescriptorSubtype;
+ __u8 bNumEmbMIDIJack;
+ __u8 baAssocJackID[0];
+} __attribute__ ((packed));
+
+struct snd_usb_midi_in_endpoint;
+struct snd_usb_midi_out_endpoint;
+struct snd_usb_midi_endpoint;
+
+struct usb_protocol_ops {
+ void (*input)(struct snd_usb_midi_in_endpoint*, uint8_t*, int);
+ void (*output)(struct snd_usb_midi_out_endpoint*);
+ void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t);
+ void (*init_out_endpoint)(struct snd_usb_midi_out_endpoint*);
+ void (*finish_out_endpoint)(struct snd_usb_midi_out_endpoint*);
+};
+
+struct snd_usb_midi {
+ struct snd_usb_audio *chip;
+ struct usb_interface *iface;
+ const struct snd_usb_audio_quirk *quirk;
+ struct snd_rawmidi *rmidi;
+ struct usb_protocol_ops* usb_protocol_ops;
+ struct list_head list;
+ struct timer_list error_timer;
+ spinlock_t disc_lock;
+
+ struct snd_usb_midi_endpoint {
+ struct snd_usb_midi_out_endpoint *out;
+ struct snd_usb_midi_in_endpoint *in;
+ } endpoints[MIDI_MAX_ENDPOINTS];
+ unsigned long input_triggered;
+ unsigned char disconnected;
+};
+
+struct snd_usb_midi_out_endpoint {
+ struct snd_usb_midi* umidi;
+ struct urb* urb;
+ int urb_active;
+ int max_transfer; /* size of urb buffer */
+ struct tasklet_struct tasklet;
+
+ spinlock_t buffer_lock;
+
+ struct usbmidi_out_port {
+ struct snd_usb_midi_out_endpoint* ep;
+ struct snd_rawmidi_substream *substream;
+ int active;
+ uint8_t cable; /* cable number << 4 */
+ uint8_t state;
+#define STATE_UNKNOWN 0
+#define STATE_1PARAM 1
+#define STATE_2PARAM_1 2
+#define STATE_2PARAM_2 3
+#define STATE_SYSEX_0 4
+#define STATE_SYSEX_1 5
+#define STATE_SYSEX_2 6
+ uint8_t data[2];
+ } ports[0x10];
+ int current_port;
+};
+
+struct snd_usb_midi_in_endpoint {
+ struct snd_usb_midi* umidi;
+ struct urb* urb;
+ struct usbmidi_in_port {
+ struct snd_rawmidi_substream *substream;
+ u8 running_status_length;
+ } ports[0x10];
+ u8 seen_f5;
+ u8 error_resubmit;
+ int current_port;
+};
+
+static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep);
+
+static const uint8_t snd_usbmidi_cin_length[] = {
+ 0, 0, 2, 3, 3, 1, 2, 3, 3, 3, 3, 3, 2, 2, 3, 1
+};
+
+/*
+ * Submits the URB, with error handling.
+ */
+static int snd_usbmidi_submit_urb(struct urb* urb, gfp_t flags)
+{
+ int err = usb_submit_urb(urb, flags);
+ if (err < 0 && err != -ENODEV)
+ snd_printk(KERN_ERR "usb_submit_urb: %d\n", err);
+ return err;
+}
+
+/*
+ * Error handling for URB completion functions.
+ */
+static int snd_usbmidi_urb_error(int status)
+{
+ switch (status) {
+ /* manually unlinked, or device gone */
+ case -ENOENT:
+ case -ECONNRESET:
+ case -ESHUTDOWN:
+ case -ENODEV:
+ return -ENODEV;
+ /* errors that might occur during unplugging */
+ case -EPROTO:
+ case -ETIME:
+ case -EILSEQ:
+ return -EIO;
+ default:
+ snd_printk(KERN_ERR "urb status %d\n", status);
+ return 0; /* continue */
+ }
+}
+
+/*
+ * Receives a chunk of MIDI data.
+ */
+static void snd_usbmidi_input_data(struct snd_usb_midi_in_endpoint* ep, int portidx,
+ uint8_t* data, int length)
+{
+ struct usbmidi_in_port* port = &ep->ports[portidx];
+
+ if (!port->substream) {
+ snd_printd("unexpected port %d!\n", portidx);
+ return;
+ }
+ if (!test_bit(port->substream->number, &ep->umidi->input_triggered))
+ return;
+ snd_rawmidi_receive(port->substream, data, length);
+}
+
+#ifdef DUMP_PACKETS
+static void dump_urb(const char *type, const u8 *data, int length)
+{
+ snd_printk(KERN_DEBUG "%s packet: [", type);
+ for (; length > 0; ++data, --length)
+ printk(" %02x", *data);
+ printk(" ]\n");
+}
+#else
+#define dump_urb(type, data, length) /* nothing */
+#endif
+
+/*
+ * Processes the data read from the device.
+ */
+static void snd_usbmidi_in_urb_complete(struct urb* urb)
+{
+ struct snd_usb_midi_in_endpoint* ep = urb->context;
+
+ if (urb->status == 0) {
+ dump_urb("received", urb->transfer_buffer, urb->actual_length);
+ ep->umidi->usb_protocol_ops->input(ep, urb->transfer_buffer,
+ urb->actual_length);
+ } else {
+ int err = snd_usbmidi_urb_error(urb->status);
+ if (err < 0) {
+ if (err != -ENODEV) {
+ ep->error_resubmit = 1;
+ mod_timer(&ep->umidi->error_timer,
+ jiffies + ERROR_DELAY_JIFFIES);
+ }
+ return;
+ }
+ }
+
+ urb->dev = ep->umidi->chip->dev;
+ snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
+}
+
+static void snd_usbmidi_out_urb_complete(struct urb* urb)
+{
+ struct snd_usb_midi_out_endpoint* ep = urb->context;
+
+ spin_lock(&ep->buffer_lock);
+ ep->urb_active = 0;
+ spin_unlock(&ep->buffer_lock);
+ if (urb->status < 0) {
+ int err = snd_usbmidi_urb_error(urb->status);
+ if (err < 0) {
+ if (err != -ENODEV)
+ mod_timer(&ep->umidi->error_timer,
+ jiffies + ERROR_DELAY_JIFFIES);
+ return;
+ }
+ }
+ snd_usbmidi_do_output(ep);
+}
+
+/*
+ * This is called when some data should be transferred to the device
+ * (from one or more substreams).
+ */
+static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep)
+{
+ struct urb* urb = ep->urb;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ep->buffer_lock, flags);
+ if (ep->urb_active || ep->umidi->chip->shutdown) {
+ spin_unlock_irqrestore(&ep->buffer_lock, flags);
+ return;
+ }
+
+ urb->transfer_buffer_length = 0;
+ ep->umidi->usb_protocol_ops->output(ep);
+
+ if (urb->transfer_buffer_length > 0) {
+ dump_urb("sending", urb->transfer_buffer,
+ urb->transfer_buffer_length);
+ urb->dev = ep->umidi->chip->dev;
+ ep->urb_active = snd_usbmidi_submit_urb(urb, GFP_ATOMIC) >= 0;
+ }
+ spin_unlock_irqrestore(&ep->buffer_lock, flags);
+}
+
+static void snd_usbmidi_out_tasklet(unsigned long data)
+{
+ struct snd_usb_midi_out_endpoint* ep = (struct snd_usb_midi_out_endpoint *) data;
+
+ snd_usbmidi_do_output(ep);
+}
+
+/* called after transfers had been interrupted due to some USB error */
+static void snd_usbmidi_error_timer(unsigned long data)
+{
+ struct snd_usb_midi *umidi = (struct snd_usb_midi *)data;
+ int i;
+
+ spin_lock(&umidi->disc_lock);
+ if (umidi->disconnected) {
+ spin_unlock(&umidi->disc_lock);
+ return;
+ }
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
+ struct snd_usb_midi_in_endpoint *in = umidi->endpoints[i].in;
+ if (in && in->error_resubmit) {
+ in->error_resubmit = 0;
+ in->urb->dev = umidi->chip->dev;
+ snd_usbmidi_submit_urb(in->urb, GFP_ATOMIC);
+ }
+ if (umidi->endpoints[i].out)
+ snd_usbmidi_do_output(umidi->endpoints[i].out);
+ }
+ spin_unlock(&umidi->disc_lock);
+}
+
+/* helper function to send static data that may not DMA-able */
+static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep,
+ const void *data, int len)
+{
+ int err;
+ void *buf = kmemdup(data, len, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+ dump_urb("sending", buf, len);
+ err = usb_bulk_msg(ep->umidi->chip->dev, ep->urb->pipe, buf, len,
+ NULL, 250);
+ kfree(buf);
+ return err;
+}
+
+/*
+ * Standard USB MIDI protocol: see the spec.
+ * Midiman protocol: like the standard protocol, but the control byte is the
+ * fourth byte in each packet, and uses length instead of CIN.
+ */
+
+static void snd_usbmidi_standard_input(struct snd_usb_midi_in_endpoint* ep,
+ uint8_t* buffer, int buffer_length)
+{
+ int i;
+
+ for (i = 0; i + 3 < buffer_length; i += 4)
+ if (buffer[i] != 0) {
+ int cable = buffer[i] >> 4;
+ int length = snd_usbmidi_cin_length[buffer[i] & 0x0f];
+ snd_usbmidi_input_data(ep, cable, &buffer[i + 1], length);
+ }
+}
+
+static void snd_usbmidi_midiman_input(struct snd_usb_midi_in_endpoint* ep,
+ uint8_t* buffer, int buffer_length)
+{
+ int i;
+
+ for (i = 0; i + 3 < buffer_length; i += 4)
+ if (buffer[i + 3] != 0) {
+ int port = buffer[i + 3] >> 4;
+ int length = buffer[i + 3] & 3;
+ snd_usbmidi_input_data(ep, port, &buffer[i], length);
+ }
+}
+
+/*
+ * Buggy M-Audio device: running status on input results in a packet that has
+ * the data bytes but not the status byte and that is marked with CIN 4.
+ */
+static void snd_usbmidi_maudio_broken_running_status_input(
+ struct snd_usb_midi_in_endpoint* ep,
+ uint8_t* buffer, int buffer_length)
+{
+ int i;
+
+ for (i = 0; i + 3 < buffer_length; i += 4)
+ if (buffer[i] != 0) {
+ int cable = buffer[i] >> 4;
+ u8 cin = buffer[i] & 0x0f;
+ struct usbmidi_in_port *port = &ep->ports[cable];
+ int length;
+
+ length = snd_usbmidi_cin_length[cin];
+ if (cin == 0xf && buffer[i + 1] >= 0xf8)
+ ; /* realtime msg: no running status change */
+ else if (cin >= 0x8 && cin <= 0xe)
+ /* channel msg */
+ port->running_status_length = length - 1;
+ else if (cin == 0x4 &&
+ port->running_status_length != 0 &&
+ buffer[i + 1] < 0x80)
+ /* CIN 4 that is not a SysEx */
+ length = port->running_status_length;
+ else
+ /*
+ * All other msgs cannot begin running status.
+ * (A channel msg sent as two or three CIN 0xF
+ * packets could in theory, but this device
+ * doesn't use this format.)
+ */
+ port->running_status_length = 0;
+ snd_usbmidi_input_data(ep, cable, &buffer[i + 1], length);
+ }
+}
+
+/*
+ * CME protocol: like the standard protocol, but SysEx commands are sent as a
+ * single USB packet preceded by a 0x0F byte.
+ */
+static void snd_usbmidi_cme_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
+{
+ if (buffer_length < 2 || (buffer[0] & 0x0f) != 0x0f)
+ snd_usbmidi_standard_input(ep, buffer, buffer_length);
+ else
+ snd_usbmidi_input_data(ep, buffer[0] >> 4,
+ &buffer[1], buffer_length - 1);
+}
+
+/*
+ * Adds one USB MIDI packet to the output buffer.
+ */
+static void snd_usbmidi_output_standard_packet(struct urb* urb, uint8_t p0,
+ uint8_t p1, uint8_t p2, uint8_t p3)
+{
+
+ uint8_t* buf = (uint8_t*)urb->transfer_buffer + urb->transfer_buffer_length;
+ buf[0] = p0;
+ buf[1] = p1;
+ buf[2] = p2;
+ buf[3] = p3;
+ urb->transfer_buffer_length += 4;
+}
+
+/*
+ * Adds one Midiman packet to the output buffer.
+ */
+static void snd_usbmidi_output_midiman_packet(struct urb* urb, uint8_t p0,
+ uint8_t p1, uint8_t p2, uint8_t p3)
+{
+
+ uint8_t* buf = (uint8_t*)urb->transfer_buffer + urb->transfer_buffer_length;
+ buf[0] = p1;
+ buf[1] = p2;
+ buf[2] = p3;
+ buf[3] = (p0 & 0xf0) | snd_usbmidi_cin_length[p0 & 0x0f];
+ urb->transfer_buffer_length += 4;
+}
+
+/*
+ * Converts MIDI commands to USB MIDI packets.
+ */
+static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port,
+ uint8_t b, struct urb* urb)
+{
+ uint8_t p0 = port->cable;
+ void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t) =
+ port->ep->umidi->usb_protocol_ops->output_packet;
+
+ if (b >= 0xf8) {
+ output_packet(urb, p0 | 0x0f, b, 0, 0);
+ } else if (b >= 0xf0) {
+ switch (b) {
+ case 0xf0:
+ port->data[0] = b;
+ port->state = STATE_SYSEX_1;
+ break;
+ case 0xf1:
+ case 0xf3:
+ port->data[0] = b;
+ port->state = STATE_1PARAM;
+ break;
+ case 0xf2:
+ port->data[0] = b;
+ port->state = STATE_2PARAM_1;
+ break;
+ case 0xf4:
+ case 0xf5:
+ port->state = STATE_UNKNOWN;
+ break;
+ case 0xf6:
+ output_packet(urb, p0 | 0x05, 0xf6, 0, 0);
+ port->state = STATE_UNKNOWN;
+ break;
+ case 0xf7:
+ switch (port->state) {
+ case STATE_SYSEX_0:
+ output_packet(urb, p0 | 0x05, 0xf7, 0, 0);
+ break;
+ case STATE_SYSEX_1:
+ output_packet(urb, p0 | 0x06, port->data[0], 0xf7, 0);
+ break;
+ case STATE_SYSEX_2:
+ output_packet(urb, p0 | 0x07, port->data[0], port->data[1], 0xf7);
+ break;
+ }
+ port->state = STATE_UNKNOWN;
+ break;
+ }
+ } else if (b >= 0x80) {
+ port->data[0] = b;
+ if (b >= 0xc0 && b <= 0xdf)
+ port->state = STATE_1PARAM;
+ else
+ port->state = STATE_2PARAM_1;
+ } else { /* b < 0x80 */
+ switch (port->state) {
+ case STATE_1PARAM:
+ if (port->data[0] < 0xf0) {
+ p0 |= port->data[0] >> 4;
+ } else {
+ p0 |= 0x02;
+ port->state = STATE_UNKNOWN;
+ }
+ output_packet(urb, p0, port->data[0], b, 0);
+ break;
+ case STATE_2PARAM_1:
+ port->data[1] = b;
+ port->state = STATE_2PARAM_2;
+ break;
+ case STATE_2PARAM_2:
+ if (port->data[0] < 0xf0) {
+ p0 |= port->data[0] >> 4;
+ port->state = STATE_2PARAM_1;
+ } else {
+ p0 |= 0x03;
+ port->state = STATE_UNKNOWN;
+ }
+ output_packet(urb, p0, port->data[0], port->data[1], b);
+ break;
+ case STATE_SYSEX_0:
+ port->data[0] = b;
+ port->state = STATE_SYSEX_1;
+ break;
+ case STATE_SYSEX_1:
+ port->data[1] = b;
+ port->state = STATE_SYSEX_2;
+ break;
+ case STATE_SYSEX_2:
+ output_packet(urb, p0 | 0x04, port->data[0], port->data[1], b);
+ port->state = STATE_SYSEX_0;
+ break;
+ }
+ }
+}
+
+static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint* ep)
+{
+ struct urb* urb = ep->urb;
+ int p;
+
+ /* FIXME: lower-numbered ports can starve higher-numbered ports */
+ for (p = 0; p < 0x10; ++p) {
+ struct usbmidi_out_port* port = &ep->ports[p];
+ if (!port->active)
+ continue;
+ while (urb->transfer_buffer_length + 3 < ep->max_transfer) {
+ uint8_t b;
+ if (snd_rawmidi_transmit(port->substream, &b, 1) != 1) {
+ port->active = 0;
+ break;
+ }
+ snd_usbmidi_transmit_byte(port, b, urb);
+ }
+ }
+}
+
+static struct usb_protocol_ops snd_usbmidi_standard_ops = {
+ .input = snd_usbmidi_standard_input,
+ .output = snd_usbmidi_standard_output,
+ .output_packet = snd_usbmidi_output_standard_packet,
+};
+
+static struct usb_protocol_ops snd_usbmidi_midiman_ops = {
+ .input = snd_usbmidi_midiman_input,
+ .output = snd_usbmidi_standard_output,
+ .output_packet = snd_usbmidi_output_midiman_packet,
+};
+
+static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = {
+ .input = snd_usbmidi_maudio_broken_running_status_input,
+ .output = snd_usbmidi_standard_output,
+ .output_packet = snd_usbmidi_output_standard_packet,
+};
+
+static struct usb_protocol_ops snd_usbmidi_cme_ops = {
+ .input = snd_usbmidi_cme_input,
+ .output = snd_usbmidi_standard_output,
+ .output_packet = snd_usbmidi_output_standard_packet,
+};
+
+/*
+ * Novation USB MIDI protocol: number of data bytes is in the first byte
+ * (when receiving) (+1!) or in the second byte (when sending); data begins
+ * at the third byte.
+ */
+
+static void snd_usbmidi_novation_input(struct snd_usb_midi_in_endpoint* ep,
+ uint8_t* buffer, int buffer_length)
+{
+ if (buffer_length < 2 || !buffer[0] || buffer_length < buffer[0] + 1)
+ return;
+ snd_usbmidi_input_data(ep, 0, &buffer[2], buffer[0] - 1);
+}
+
+static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep)
+{
+ uint8_t* transfer_buffer;
+ int count;
+
+ if (!ep->ports[0].active)
+ return;
+ transfer_buffer = ep->urb->transfer_buffer;
+ count = snd_rawmidi_transmit(ep->ports[0].substream,
+ &transfer_buffer[2],
+ ep->max_transfer - 2);
+ if (count < 1) {
+ ep->ports[0].active = 0;
+ return;
+ }
+ transfer_buffer[0] = 0;
+ transfer_buffer[1] = count;
+ ep->urb->transfer_buffer_length = 2 + count;
+}
+
+static struct usb_protocol_ops snd_usbmidi_novation_ops = {
+ .input = snd_usbmidi_novation_input,
+ .output = snd_usbmidi_novation_output,
+};
+
+/*
+ * "raw" protocol: used by the MOTU FastLane.
+ */
+
+static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint* ep,
+ uint8_t* buffer, int buffer_length)
+{
+ snd_usbmidi_input_data(ep, 0, buffer, buffer_length);
+}
+
+static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint* ep)
+{
+ int count;
+
+ if (!ep->ports[0].active)
+ return;
+ count = snd_rawmidi_transmit(ep->ports[0].substream,
+ ep->urb->transfer_buffer,
+ ep->max_transfer);
+ if (count < 1) {
+ ep->ports[0].active = 0;
+ return;
+ }
+ ep->urb->transfer_buffer_length = count;
+}
+
+static struct usb_protocol_ops snd_usbmidi_raw_ops = {
+ .input = snd_usbmidi_raw_input,
+ .output = snd_usbmidi_raw_output,
+};
+
+static void snd_usbmidi_us122l_input(struct snd_usb_midi_in_endpoint *ep,
+ uint8_t *buffer, int buffer_length)
+{
+ if (buffer_length != 9)
+ return;
+ buffer_length = 8;
+ while (buffer_length && buffer[buffer_length - 1] == 0xFD)
+ buffer_length--;
+ if (buffer_length)
+ snd_usbmidi_input_data(ep, 0, buffer, buffer_length);
+}
+
+static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep)
+{
+ int count;
+
+ if (!ep->ports[0].active)
+ return;
+ count = ep->urb->dev->speed == USB_SPEED_HIGH ? 1 : 2;
+ count = snd_rawmidi_transmit(ep->ports[0].substream,
+ ep->urb->transfer_buffer,
+ count);
+ if (count < 1) {
+ ep->ports[0].active = 0;
+ return;
+ }
+
+ memset(ep->urb->transfer_buffer + count, 0xFD, 9 - count);
+ ep->urb->transfer_buffer_length = count;
+}
+
+static struct usb_protocol_ops snd_usbmidi_122l_ops = {
+ .input = snd_usbmidi_us122l_input,
+ .output = snd_usbmidi_us122l_output,
+};
+
+/*
+ * Emagic USB MIDI protocol: raw MIDI with "F5 xx" port switching.
+ */
+
+static void snd_usbmidi_emagic_init_out(struct snd_usb_midi_out_endpoint* ep)
+{
+ static const u8 init_data[] = {
+ /* initialization magic: "get version" */
+ 0xf0,
+ 0x00, 0x20, 0x31, /* Emagic */
+ 0x64, /* Unitor8 */
+ 0x0b, /* version number request */
+ 0x00, /* command version */
+ 0x00, /* EEPROM, box 0 */
+ 0xf7
+ };
+ send_bulk_static_data(ep, init_data, sizeof(init_data));
+ /* while we're at it, pour on more magic */
+ send_bulk_static_data(ep, init_data, sizeof(init_data));
+}
+
+static void snd_usbmidi_emagic_finish_out(struct snd_usb_midi_out_endpoint* ep)
+{
+ static const u8 finish_data[] = {
+ /* switch to patch mode with last preset */
+ 0xf0,
+ 0x00, 0x20, 0x31, /* Emagic */
+ 0x64, /* Unitor8 */
+ 0x10, /* patch switch command */
+ 0x00, /* command version */
+ 0x7f, /* to all boxes */
+ 0x40, /* last preset in EEPROM */
+ 0xf7
+ };
+ send_bulk_static_data(ep, finish_data, sizeof(finish_data));
+}
+
+static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint* ep,
+ uint8_t* buffer, int buffer_length)
+{
+ int i;
+
+ /* FF indicates end of valid data */
+ for (i = 0; i < buffer_length; ++i)
+ if (buffer[i] == 0xff) {
+ buffer_length = i;
+ break;
+ }
+
+ /* handle F5 at end of last buffer */
+ if (ep->seen_f5)
+ goto switch_port;
+
+ while (buffer_length > 0) {
+ /* determine size of data until next F5 */
+ for (i = 0; i < buffer_length; ++i)
+ if (buffer[i] == 0xf5)
+ break;
+ snd_usbmidi_input_data(ep, ep->current_port, buffer, i);
+ buffer += i;
+ buffer_length -= i;
+
+ if (buffer_length <= 0)
+ break;
+ /* assert(buffer[0] == 0xf5); */
+ ep->seen_f5 = 1;
+ ++buffer;
+ --buffer_length;
+
+ switch_port:
+ if (buffer_length <= 0)
+ break;
+ if (buffer[0] < 0x80) {
+ ep->current_port = (buffer[0] - 1) & 15;
+ ++buffer;
+ --buffer_length;
+ }
+ ep->seen_f5 = 0;
+ }
+}
+
+static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep)
+{
+ int port0 = ep->current_port;
+ uint8_t* buf = ep->urb->transfer_buffer;
+ int buf_free = ep->max_transfer;
+ int length, i;
+
+ for (i = 0; i < 0x10; ++i) {
+ /* round-robin, starting at the last current port */
+ int portnum = (port0 + i) & 15;
+ struct usbmidi_out_port* port = &ep->ports[portnum];
+
+ if (!port->active)
+ continue;
+ if (snd_rawmidi_transmit_peek(port->substream, buf, 1) != 1) {
+ port->active = 0;
+ continue;
+ }
+
+ if (portnum != ep->current_port) {
+ if (buf_free < 2)
+ break;
+ ep->current_port = portnum;
+ buf[0] = 0xf5;
+ buf[1] = (portnum + 1) & 15;
+ buf += 2;
+ buf_free -= 2;
+ }
+
+ if (buf_free < 1)
+ break;
+ length = snd_rawmidi_transmit(port->substream, buf, buf_free);
+ if (length > 0) {
+ buf += length;
+ buf_free -= length;
+ if (buf_free < 1)
+ break;
+ }
+ }
+ if (buf_free < ep->max_transfer && buf_free > 0) {
+ *buf = 0xff;
+ --buf_free;
+ }
+ ep->urb->transfer_buffer_length = ep->max_transfer - buf_free;
+}
+
+static struct usb_protocol_ops snd_usbmidi_emagic_ops = {
+ .input = snd_usbmidi_emagic_input,
+ .output = snd_usbmidi_emagic_output,
+ .init_out_endpoint = snd_usbmidi_emagic_init_out,
+ .finish_out_endpoint = snd_usbmidi_emagic_finish_out,
+};
+
+
+static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_usb_midi* umidi = substream->rmidi->private_data;
+ struct usbmidi_out_port* port = NULL;
+ int i, j;
+
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
+ if (umidi->endpoints[i].out)
+ for (j = 0; j < 0x10; ++j)
+ if (umidi->endpoints[i].out->ports[j].substream == substream) {
+ port = &umidi->endpoints[i].out->ports[j];
+ break;
+ }
+ if (!port) {
+ snd_BUG();
+ return -ENXIO;
+ }
+ substream->runtime->private_data = port;
+ port->state = STATE_UNKNOWN;
+ return 0;
+}
+
+static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct usbmidi_out_port* port = (struct usbmidi_out_port*)substream->runtime->private_data;
+
+ port->active = up;
+ if (up) {
+ if (port->ep->umidi->chip->shutdown) {
+ /* gobble up remaining bytes to prevent wait in
+ * snd_rawmidi_drain_output */
+ while (!snd_rawmidi_transmit_empty(substream))
+ snd_rawmidi_transmit_ack(substream, 1);
+ return;
+ }
+ tasklet_hi_schedule(&port->ep->tasklet);
+ }
+}
+
+static int snd_usbmidi_input_open(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static int snd_usbmidi_input_close(struct snd_rawmidi_substream *substream)
+{
+ return 0;
+}
+
+static void snd_usbmidi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+ struct snd_usb_midi* umidi = substream->rmidi->private_data;
+
+ if (up)
+ set_bit(substream->number, &umidi->input_triggered);
+ else
+ clear_bit(substream->number, &umidi->input_triggered);
+}
+
+static struct snd_rawmidi_ops snd_usbmidi_output_ops = {
+ .open = snd_usbmidi_output_open,
+ .close = snd_usbmidi_output_close,
+ .trigger = snd_usbmidi_output_trigger,
+};
+
+static struct snd_rawmidi_ops snd_usbmidi_input_ops = {
+ .open = snd_usbmidi_input_open,
+ .close = snd_usbmidi_input_close,
+ .trigger = snd_usbmidi_input_trigger
+};
+
+/*
+ * Frees an input endpoint.
+ * May be called when ep hasn't been initialized completely.
+ */
+static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint* ep)
+{
+ if (ep->urb) {
+ usb_buffer_free(ep->umidi->chip->dev,
+ ep->urb->transfer_buffer_length,
+ ep->urb->transfer_buffer,
+ ep->urb->transfer_dma);
+ usb_free_urb(ep->urb);
+ }
+ kfree(ep);
+}
+
+/*
+ * Creates an input endpoint.
+ */
+static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* ep_info,
+ struct snd_usb_midi_endpoint* rep)
+{
+ struct snd_usb_midi_in_endpoint* ep;
+ void* buffer;
+ unsigned int pipe;
+ int length;
+
+ rep->in = NULL;
+ ep = kzalloc(sizeof(*ep), GFP_KERNEL);
+ if (!ep)
+ return -ENOMEM;
+ ep->umidi = umidi;
+
+ ep->urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!ep->urb) {
+ snd_usbmidi_in_endpoint_delete(ep);
+ return -ENOMEM;
+ }
+ if (ep_info->in_interval)
+ pipe = usb_rcvintpipe(umidi->chip->dev, ep_info->in_ep);
+ else
+ pipe = usb_rcvbulkpipe(umidi->chip->dev, ep_info->in_ep);
+ length = usb_maxpacket(umidi->chip->dev, pipe, 0);
+ buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL,
+ &ep->urb->transfer_dma);
+ if (!buffer) {
+ snd_usbmidi_in_endpoint_delete(ep);
+ return -ENOMEM;
+ }
+ if (ep_info->in_interval)
+ usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer,
+ length, snd_usbmidi_in_urb_complete, ep,
+ ep_info->in_interval);
+ else
+ usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer,
+ length, snd_usbmidi_in_urb_complete, ep);
+ ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
+
+ rep->in = ep;
+ return 0;
+}
+
+static unsigned int snd_usbmidi_count_bits(unsigned int x)
+{
+ unsigned int bits;
+
+ for (bits = 0; x; ++bits)
+ x &= x - 1;
+ return bits;
+}
+
+/*
+ * Frees an output endpoint.
+ * May be called when ep hasn't been initialized completely.
+ */
+static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep)
+{
+ if (ep->urb) {
+ usb_buffer_free(ep->umidi->chip->dev, ep->max_transfer,
+ ep->urb->transfer_buffer,
+ ep->urb->transfer_dma);
+ usb_free_urb(ep->urb);
+ }
+ kfree(ep);
+}
+
+/*
+ * Creates an output endpoint, and initializes output ports.
+ */
+static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* ep_info,
+ struct snd_usb_midi_endpoint* rep)
+{
+ struct snd_usb_midi_out_endpoint* ep;
+ int i;
+ unsigned int pipe;
+ void* buffer;
+
+ rep->out = NULL;
+ ep = kzalloc(sizeof(*ep), GFP_KERNEL);
+ if (!ep)
+ return -ENOMEM;
+ ep->umidi = umidi;
+
+ ep->urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!ep->urb) {
+ snd_usbmidi_out_endpoint_delete(ep);
+ return -ENOMEM;
+ }
+ if (ep_info->out_interval)
+ pipe = usb_sndintpipe(umidi->chip->dev, ep_info->out_ep);
+ else
+ pipe = usb_sndbulkpipe(umidi->chip->dev, ep_info->out_ep);
+ if (umidi->chip->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */
+ /* FIXME: we need more URBs to get reasonable bandwidth here: */
+ ep->max_transfer = 4;
+ else
+ ep->max_transfer = usb_maxpacket(umidi->chip->dev, pipe, 1);
+ buffer = usb_buffer_alloc(umidi->chip->dev, ep->max_transfer,
+ GFP_KERNEL, &ep->urb->transfer_dma);
+ if (!buffer) {
+ snd_usbmidi_out_endpoint_delete(ep);
+ return -ENOMEM;
+ }
+ if (ep_info->out_interval)
+ usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer,
+ ep->max_transfer, snd_usbmidi_out_urb_complete,
+ ep, ep_info->out_interval);
+ else
+ usb_fill_bulk_urb(ep->urb, umidi->chip->dev,
+ pipe, buffer, ep->max_transfer,
+ snd_usbmidi_out_urb_complete, ep);
+ ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
+
+ spin_lock_init(&ep->buffer_lock);
+ tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep);
+
+ for (i = 0; i < 0x10; ++i)
+ if (ep_info->out_cables & (1 << i)) {
+ ep->ports[i].ep = ep;
+ ep->ports[i].cable = i << 4;
+ }
+
+ if (umidi->usb_protocol_ops->init_out_endpoint)
+ umidi->usb_protocol_ops->init_out_endpoint(ep);
+
+ rep->out = ep;
+ return 0;
+}
+
+/*
+ * Frees everything.
+ */
+static void snd_usbmidi_free(struct snd_usb_midi* umidi)
+{
+ int i;
+
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
+ struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
+ if (ep->out)
+ snd_usbmidi_out_endpoint_delete(ep->out);
+ if (ep->in)
+ snd_usbmidi_in_endpoint_delete(ep->in);
+ }
+ kfree(umidi);
+}
+
+/*
+ * Unlinks all URBs (must be done before the usb_device is deleted).
+ */
+void snd_usbmidi_disconnect(struct list_head* p)
+{
+ struct snd_usb_midi* umidi;
+ int i;
+
+ umidi = list_entry(p, struct snd_usb_midi, list);
+ /*
+ * an URB's completion handler may start the timer and
+ * a timer may submit an URB. To reliably break the cycle
+ * a flag under lock must be used
+ */
+ spin_lock_irq(&umidi->disc_lock);
+ umidi->disconnected = 1;
+ spin_unlock_irq(&umidi->disc_lock);
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
+ struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
+ if (ep->out)
+ tasklet_kill(&ep->out->tasklet);
+ if (ep->out && ep->out->urb) {
+ usb_kill_urb(ep->out->urb);
+ if (umidi->usb_protocol_ops->finish_out_endpoint)
+ umidi->usb_protocol_ops->finish_out_endpoint(ep->out);
+ }
+ if (ep->in)
+ usb_kill_urb(ep->in->urb);
+ /* free endpoints here; later call can result in Oops */
+ if (ep->out) {
+ snd_usbmidi_out_endpoint_delete(ep->out);
+ ep->out = NULL;
+ }
+ if (ep->in) {
+ snd_usbmidi_in_endpoint_delete(ep->in);
+ ep->in = NULL;
+ }
+ }
+ del_timer_sync(&umidi->error_timer);
+}
+
+static void snd_usbmidi_rawmidi_free(struct snd_rawmidi *rmidi)
+{
+ struct snd_usb_midi* umidi = rmidi->private_data;
+ snd_usbmidi_free(umidi);
+}
+
+static struct snd_rawmidi_substream *snd_usbmidi_find_substream(struct snd_usb_midi* umidi,
+ int stream, int number)
+{
+ struct list_head* list;
+
+ list_for_each(list, &umidi->rmidi->streams[stream].substreams) {
+ struct snd_rawmidi_substream *substream = list_entry(list, struct snd_rawmidi_substream, list);
+ if (substream->number == number)
+ return substream;
+ }
+ return NULL;
+}
+
+/*
+ * This list specifies names for ports that do not fit into the standard
+ * "(product) MIDI (n)" schema because they aren't external MIDI ports,
+ * such as internal control or synthesizer ports.
+ */
+static struct port_info {
+ u32 id;
+ short int port;
+ short int voices;
+ const char *name;
+ unsigned int seq_flags;
+} snd_usbmidi_port_info[] = {
+#define PORT_INFO(vendor, product, num, name_, voices_, flags) \
+ { .id = USB_ID(vendor, product), \
+ .port = num, .voices = voices_, \
+ .name = name_, .seq_flags = flags }
+#define EXTERNAL_PORT(vendor, product, num, name) \
+ PORT_INFO(vendor, product, num, name, 0, \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \
+ SNDRV_SEQ_PORT_TYPE_HARDWARE | \
+ SNDRV_SEQ_PORT_TYPE_PORT)
+#define CONTROL_PORT(vendor, product, num, name) \
+ PORT_INFO(vendor, product, num, name, 0, \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \
+ SNDRV_SEQ_PORT_TYPE_HARDWARE)
+#define ROLAND_SYNTH_PORT(vendor, product, num, name, voices) \
+ PORT_INFO(vendor, product, num, name, voices, \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM2 | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GS | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_XG | \
+ SNDRV_SEQ_PORT_TYPE_HARDWARE | \
+ SNDRV_SEQ_PORT_TYPE_SYNTHESIZER)
+#define SOUNDCANVAS_PORT(vendor, product, num, name, voices) \
+ PORT_INFO(vendor, product, num, name, voices, \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GM2 | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_GS | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_XG | \
+ SNDRV_SEQ_PORT_TYPE_MIDI_MT32 | \
+ SNDRV_SEQ_PORT_TYPE_HARDWARE | \
+ SNDRV_SEQ_PORT_TYPE_SYNTHESIZER)
+ /* Roland UA-100 */
+ CONTROL_PORT(0x0582, 0x0000, 2, "%s Control"),
+ /* Roland SC-8850 */
+ SOUNDCANVAS_PORT(0x0582, 0x0003, 0, "%s Part A", 128),
+ SOUNDCANVAS_PORT(0x0582, 0x0003, 1, "%s Part B", 128),
+ SOUNDCANVAS_PORT(0x0582, 0x0003, 2, "%s Part C", 128),
+ SOUNDCANVAS_PORT(0x0582, 0x0003, 3, "%s Part D", 128),
+ EXTERNAL_PORT(0x0582, 0x0003, 4, "%s MIDI 1"),
+ EXTERNAL_PORT(0x0582, 0x0003, 5, "%s MIDI 2"),
+ /* Roland U-8 */
+ EXTERNAL_PORT(0x0582, 0x0004, 0, "%s MIDI"),
+ CONTROL_PORT(0x0582, 0x0004, 1, "%s Control"),
+ /* Roland SC-8820 */
+ SOUNDCANVAS_PORT(0x0582, 0x0007, 0, "%s Part A", 64),
+ SOUNDCANVAS_PORT(0x0582, 0x0007, 1, "%s Part B", 64),
+ EXTERNAL_PORT(0x0582, 0x0007, 2, "%s MIDI"),
+ /* Roland SK-500 */
+ SOUNDCANVAS_PORT(0x0582, 0x000b, 0, "%s Part A", 64),
+ SOUNDCANVAS_PORT(0x0582, 0x000b, 1, "%s Part B", 64),
+ EXTERNAL_PORT(0x0582, 0x000b, 2, "%s MIDI"),
+ /* Roland SC-D70 */
+ SOUNDCANVAS_PORT(0x0582, 0x000c, 0, "%s Part A", 64),
+ SOUNDCANVAS_PORT(0x0582, 0x000c, 1, "%s Part B", 64),
+ EXTERNAL_PORT(0x0582, 0x000c, 2, "%s MIDI"),
+ /* Edirol UM-880 */
+ CONTROL_PORT(0x0582, 0x0014, 8, "%s Control"),
+ /* Edirol SD-90 */
+ ROLAND_SYNTH_PORT(0x0582, 0x0016, 0, "%s Part A", 128),
+ ROLAND_SYNTH_PORT(0x0582, 0x0016, 1, "%s Part B", 128),
+ EXTERNAL_PORT(0x0582, 0x0016, 2, "%s MIDI 1"),
+ EXTERNAL_PORT(0x0582, 0x0016, 3, "%s MIDI 2"),
+ /* Edirol UM-550 */
+ CONTROL_PORT(0x0582, 0x0023, 5, "%s Control"),
+ /* Edirol SD-20 */
+ ROLAND_SYNTH_PORT(0x0582, 0x0027, 0, "%s Part A", 64),
+ ROLAND_SYNTH_PORT(0x0582, 0x0027, 1, "%s Part B", 64),
+ EXTERNAL_PORT(0x0582, 0x0027, 2, "%s MIDI"),
+ /* Edirol SD-80 */
+ ROLAND_SYNTH_PORT(0x0582, 0x0029, 0, "%s Part A", 128),
+ ROLAND_SYNTH_PORT(0x0582, 0x0029, 1, "%s Part B", 128),
+ EXTERNAL_PORT(0x0582, 0x0029, 2, "%s MIDI 1"),
+ EXTERNAL_PORT(0x0582, 0x0029, 3, "%s MIDI 2"),
+ /* Edirol UA-700 */
+ EXTERNAL_PORT(0x0582, 0x002b, 0, "%s MIDI"),
+ CONTROL_PORT(0x0582, 0x002b, 1, "%s Control"),
+ /* Roland VariOS */
+ EXTERNAL_PORT(0x0582, 0x002f, 0, "%s MIDI"),
+ EXTERNAL_PORT(0x0582, 0x002f, 1, "%s External MIDI"),
+ EXTERNAL_PORT(0x0582, 0x002f, 2, "%s Sync"),
+ /* Edirol PCR */
+ EXTERNAL_PORT(0x0582, 0x0033, 0, "%s MIDI"),
+ EXTERNAL_PORT(0x0582, 0x0033, 1, "%s 1"),
+ EXTERNAL_PORT(0x0582, 0x0033, 2, "%s 2"),
+ /* BOSS GS-10 */
+ EXTERNAL_PORT(0x0582, 0x003b, 0, "%s MIDI"),
+ CONTROL_PORT(0x0582, 0x003b, 1, "%s Control"),
+ /* Edirol UA-1000 */
+ EXTERNAL_PORT(0x0582, 0x0044, 0, "%s MIDI"),
+ CONTROL_PORT(0x0582, 0x0044, 1, "%s Control"),
+ /* Edirol UR-80 */
+ EXTERNAL_PORT(0x0582, 0x0048, 0, "%s MIDI"),
+ EXTERNAL_PORT(0x0582, 0x0048, 1, "%s 1"),
+ EXTERNAL_PORT(0x0582, 0x0048, 2, "%s 2"),
+ /* Edirol PCR-A */
+ EXTERNAL_PORT(0x0582, 0x004d, 0, "%s MIDI"),
+ EXTERNAL_PORT(0x0582, 0x004d, 1, "%s 1"),
+ EXTERNAL_PORT(0x0582, 0x004d, 2, "%s 2"),
+ /* Edirol UM-3EX */
+ CONTROL_PORT(0x0582, 0x009a, 3, "%s Control"),
+ /* M-Audio MidiSport 8x8 */
+ CONTROL_PORT(0x0763, 0x1031, 8, "%s Control"),
+ CONTROL_PORT(0x0763, 0x1033, 8, "%s Control"),
+ /* MOTU Fastlane */
+ EXTERNAL_PORT(0x07fd, 0x0001, 0, "%s MIDI A"),
+ EXTERNAL_PORT(0x07fd, 0x0001, 1, "%s MIDI B"),
+ /* Emagic Unitor8/AMT8/MT4 */
+ EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"),
+ EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"),
+ EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"),
+};
+
+static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_usbmidi_port_info); ++i) {
+ if (snd_usbmidi_port_info[i].id == umidi->chip->usb_id &&
+ snd_usbmidi_port_info[i].port == number)
+ return &snd_usbmidi_port_info[i];
+ }
+ return NULL;
+}
+
+static void snd_usbmidi_get_port_info(struct snd_rawmidi *rmidi, int number,
+ struct snd_seq_port_info *seq_port_info)
+{
+ struct snd_usb_midi *umidi = rmidi->private_data;
+ struct port_info *port_info;
+
+ /* TODO: read port flags from descriptors */
+ port_info = find_port_info(umidi, number);
+ if (port_info) {
+ seq_port_info->type = port_info->seq_flags;
+ seq_port_info->midi_voices = port_info->voices;
+ }
+}
+
+static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi,
+ int stream, int number,
+ struct snd_rawmidi_substream ** rsubstream)
+{
+ struct port_info *port_info;
+ const char *name_format;
+
+ struct snd_rawmidi_substream *substream = snd_usbmidi_find_substream(umidi, stream, number);
+ if (!substream) {
+ snd_printd(KERN_ERR "substream %d:%d not found\n", stream, number);
+ return;
+ }
+
+ /* TODO: read port name from jack descriptor */
+ port_info = find_port_info(umidi, number);
+ name_format = port_info ? port_info->name : "%s MIDI %d";
+ snprintf(substream->name, sizeof(substream->name),
+ name_format, umidi->chip->card->shortname, number + 1);
+
+ *rsubstream = substream;
+}
+
+/*
+ * Creates the endpoints and their ports.
+ */
+static int snd_usbmidi_create_endpoints(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* endpoints)
+{
+ int i, j, err;
+ int out_ports = 0, in_ports = 0;
+
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
+ if (endpoints[i].out_cables) {
+ err = snd_usbmidi_out_endpoint_create(umidi, &endpoints[i],
+ &umidi->endpoints[i]);
+ if (err < 0)
+ return err;
+ }
+ if (endpoints[i].in_cables) {
+ err = snd_usbmidi_in_endpoint_create(umidi, &endpoints[i],
+ &umidi->endpoints[i]);
+ if (err < 0)
+ return err;
+ }
+
+ for (j = 0; j < 0x10; ++j) {
+ if (endpoints[i].out_cables & (1 << j)) {
+ snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, out_ports,
+ &umidi->endpoints[i].out->ports[j].substream);
+ ++out_ports;
+ }
+ if (endpoints[i].in_cables & (1 << j)) {
+ snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, in_ports,
+ &umidi->endpoints[i].in->ports[j].substream);
+ ++in_ports;
+ }
+ }
+ }
+ snd_printdd(KERN_INFO "created %d output and %d input ports\n",
+ out_ports, in_ports);
+ return 0;
+}
+
+/*
+ * Returns MIDIStreaming device capabilities.
+ */
+static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* endpoints)
+{
+ struct usb_interface* intf;
+ struct usb_host_interface *hostif;
+ struct usb_interface_descriptor* intfd;
+ struct usb_ms_header_descriptor* ms_header;
+ struct usb_host_endpoint *hostep;
+ struct usb_endpoint_descriptor* ep;
+ struct usb_ms_endpoint_descriptor* ms_ep;
+ int i, epidx;
+
+ intf = umidi->iface;
+ if (!intf)
+ return -ENXIO;
+ hostif = &intf->altsetting[0];
+ intfd = get_iface_desc(hostif);
+ ms_header = (struct usb_ms_header_descriptor*)hostif->extra;
+ if (hostif->extralen >= 7 &&
+ ms_header->bLength >= 7 &&
+ ms_header->bDescriptorType == USB_DT_CS_INTERFACE &&
+ ms_header->bDescriptorSubtype == HEADER)
+ snd_printdd(KERN_INFO "MIDIStreaming version %02x.%02x\n",
+ ms_header->bcdMSC[1], ms_header->bcdMSC[0]);
+ else
+ snd_printk(KERN_WARNING "MIDIStreaming interface descriptor not found\n");
+
+ epidx = 0;
+ for (i = 0; i < intfd->bNumEndpoints; ++i) {
+ hostep = &hostif->endpoint[i];
+ ep = get_ep_desc(hostep);
+ if ((ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK &&
+ (ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ continue;
+ ms_ep = (struct usb_ms_endpoint_descriptor*)hostep->extra;
+ if (hostep->extralen < 4 ||
+ ms_ep->bLength < 4 ||
+ ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
+ ms_ep->bDescriptorSubtype != MS_GENERAL)
+ continue;
+ if ((ep->bEndpointAddress & USB_ENDPOINT_DIR_MASK) == USB_DIR_OUT) {
+ if (endpoints[epidx].out_ep) {
+ if (++epidx >= MIDI_MAX_ENDPOINTS) {
+ snd_printk(KERN_WARNING "too many endpoints\n");
+ break;
+ }
+ }
+ endpoints[epidx].out_ep = ep->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ if ((ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+ endpoints[epidx].out_interval = ep->bInterval;
+ else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW)
+ /*
+ * Low speed bulk transfers don't exist, so
+ * force interrupt transfers for devices like
+ * ESI MIDI Mate that try to use them anyway.
+ */
+ endpoints[epidx].out_interval = 1;
+ endpoints[epidx].out_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1;
+ snd_printdd(KERN_INFO "EP %02X: %d jack(s)\n",
+ ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack);
+ } else {
+ if (endpoints[epidx].in_ep) {
+ if (++epidx >= MIDI_MAX_ENDPOINTS) {
+ snd_printk(KERN_WARNING "too many endpoints\n");
+ break;
+ }
+ }
+ endpoints[epidx].in_ep = ep->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ if ((ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+ endpoints[epidx].in_interval = ep->bInterval;
+ else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW)
+ endpoints[epidx].in_interval = 1;
+ endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1;
+ snd_printdd(KERN_INFO "EP %02X: %d jack(s)\n",
+ ep->bEndpointAddress, ms_ep->bNumEmbMIDIJack);
+ }
+ }
+ return 0;
+}
+
+/*
+ * On Roland devices, use the second alternate setting to be able to use
+ * the interrupt input endpoint.
+ */
+static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi)
+{
+ struct usb_interface* intf;
+ struct usb_host_interface *hostif;
+ struct usb_interface_descriptor* intfd;
+
+ intf = umidi->iface;
+ if (!intf || intf->num_altsetting != 2)
+ return;
+
+ hostif = &intf->altsetting[1];
+ intfd = get_iface_desc(hostif);
+ if (intfd->bNumEndpoints != 2 ||
+ (get_endpoint(hostif, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK ||
+ (get_endpoint(hostif, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ return;
+
+ snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n",
+ intfd->bAlternateSetting);
+ usb_set_interface(umidi->chip->dev, intfd->bInterfaceNumber,
+ intfd->bAlternateSetting);
+}
+
+/*
+ * Try to find any usable endpoints in the interface.
+ */
+static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* endpoint,
+ int max_endpoints)
+{
+ struct usb_interface* intf;
+ struct usb_host_interface *hostif;
+ struct usb_interface_descriptor* intfd;
+ struct usb_endpoint_descriptor* epd;
+ int i, out_eps = 0, in_eps = 0;
+
+ if (USB_ID_VENDOR(umidi->chip->usb_id) == 0x0582)
+ snd_usbmidi_switch_roland_altsetting(umidi);
+
+ if (endpoint[0].out_ep || endpoint[0].in_ep)
+ return 0;
+
+ intf = umidi->iface;
+ if (!intf || intf->num_altsetting < 1)
+ return -ENOENT;
+ hostif = intf->cur_altsetting;
+ intfd = get_iface_desc(hostif);
+
+ for (i = 0; i < intfd->bNumEndpoints; ++i) {
+ epd = get_endpoint(hostif, i);
+ if ((epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK &&
+ (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ continue;
+ if (out_eps < max_endpoints &&
+ (epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) == USB_DIR_OUT) {
+ endpoint[out_eps].out_ep = epd->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ if ((epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+ endpoint[out_eps].out_interval = epd->bInterval;
+ ++out_eps;
+ }
+ if (in_eps < max_endpoints &&
+ (epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) == USB_DIR_IN) {
+ endpoint[in_eps].in_ep = epd->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ if ((epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)
+ endpoint[in_eps].in_interval = epd->bInterval;
+ ++in_eps;
+ }
+ }
+ return (out_eps || in_eps) ? 0 : -ENOENT;
+}
+
+/*
+ * Detects the endpoints for one-port-per-endpoint protocols.
+ */
+static int snd_usbmidi_detect_per_port_endpoints(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* endpoints)
+{
+ int err, i;
+
+ err = snd_usbmidi_detect_endpoints(umidi, endpoints, MIDI_MAX_ENDPOINTS);
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
+ if (endpoints[i].out_ep)
+ endpoints[i].out_cables = 0x0001;
+ if (endpoints[i].in_ep)
+ endpoints[i].in_cables = 0x0001;
+ }
+ return err;
+}
+
+/*
+ * Detects the endpoints and ports of Yamaha devices.
+ */
+static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* endpoint)
+{
+ struct usb_interface* intf;
+ struct usb_host_interface *hostif;
+ struct usb_interface_descriptor* intfd;
+ uint8_t* cs_desc;
+
+ intf = umidi->iface;
+ if (!intf)
+ return -ENOENT;
+ hostif = intf->altsetting;
+ intfd = get_iface_desc(hostif);
+ if (intfd->bNumEndpoints < 1)
+ return -ENOENT;
+
+ /*
+ * For each port there is one MIDI_IN/OUT_JACK descriptor, not
+ * necessarily with any useful contents. So simply count 'em.
+ */
+ for (cs_desc = hostif->extra;
+ cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2;
+ cs_desc += cs_desc[0]) {
+ if (cs_desc[1] == USB_DT_CS_INTERFACE) {
+ if (cs_desc[2] == MIDI_IN_JACK)
+ endpoint->in_cables = (endpoint->in_cables << 1) | 1;
+ else if (cs_desc[2] == MIDI_OUT_JACK)
+ endpoint->out_cables = (endpoint->out_cables << 1) | 1;
+ }
+ }
+ if (!endpoint->in_cables && !endpoint->out_cables)
+ return -ENOENT;
+
+ return snd_usbmidi_detect_endpoints(umidi, endpoint, 1);
+}
+
+/*
+ * Creates the endpoints and their ports for Midiman devices.
+ */
+static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
+ struct snd_usb_midi_endpoint_info* endpoint)
+{
+ struct snd_usb_midi_endpoint_info ep_info;
+ struct usb_interface* intf;
+ struct usb_host_interface *hostif;
+ struct usb_interface_descriptor* intfd;
+ struct usb_endpoint_descriptor* epd;
+ int cable, err;
+
+ intf = umidi->iface;
+ if (!intf)
+ return -ENOENT;
+ hostif = intf->altsetting;
+ intfd = get_iface_desc(hostif);
+ /*
+ * The various MidiSport devices have more or less random endpoint
+ * numbers, so we have to identify the endpoints by their index in
+ * the descriptor array, like the driver for that other OS does.
+ *
+ * There is one interrupt input endpoint for all input ports, one
+ * bulk output endpoint for even-numbered ports, and one for odd-
+ * numbered ports. Both bulk output endpoints have corresponding
+ * input bulk endpoints (at indices 1 and 3) which aren't used.
+ */
+ if (intfd->bNumEndpoints < (endpoint->out_cables > 0x0001 ? 5 : 3)) {
+ snd_printdd(KERN_ERR "not enough endpoints\n");
+ return -ENOENT;
+ }
+
+ epd = get_endpoint(hostif, 0);
+ if ((epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_IN ||
+ (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) {
+ snd_printdd(KERN_ERR "endpoint[0] isn't interrupt\n");
+ return -ENXIO;
+ }
+ epd = get_endpoint(hostif, 2);
+ if ((epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_OUT ||
+ (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK) {
+ snd_printdd(KERN_ERR "endpoint[2] isn't bulk output\n");
+ return -ENXIO;
+ }
+ if (endpoint->out_cables > 0x0001) {
+ epd = get_endpoint(hostif, 4);
+ if ((epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_OUT ||
+ (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK) {
+ snd_printdd(KERN_ERR "endpoint[4] isn't bulk output\n");
+ return -ENXIO;
+ }
+ }
+
+ ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.out_interval = 0;
+ ep_info.out_cables = endpoint->out_cables & 0x5555;
+ err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
+ if (err < 0)
+ return err;
+
+ ep_info.in_ep = get_endpoint(hostif, 0)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.in_interval = get_endpoint(hostif, 0)->bInterval;
+ ep_info.in_cables = endpoint->in_cables;
+ err = snd_usbmidi_in_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
+ if (err < 0)
+ return err;
+
+ if (endpoint->out_cables > 0x0001) {
+ ep_info.out_ep = get_endpoint(hostif, 4)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ ep_info.out_cables = endpoint->out_cables & 0xaaaa;
+ err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[1]);
+ if (err < 0)
+ return err;
+ }
+
+ for (cable = 0; cable < 0x10; ++cable) {
+ if (endpoint->out_cables & (1 << cable))
+ snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, cable,
+ &umidi->endpoints[cable & 1].out->ports[cable].substream);
+ if (endpoint->in_cables & (1 << cable))
+ snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, cable,
+ &umidi->endpoints[0].in->ports[cable].substream);
+ }
+ return 0;
+}
+
+static struct snd_rawmidi_global_ops snd_usbmidi_ops = {
+ .get_port_info = snd_usbmidi_get_port_info,
+};
+
+static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi,
+ int out_ports, int in_ports)
+{
+ struct snd_rawmidi *rmidi;
+ int err;
+
+ err = snd_rawmidi_new(umidi->chip->card, "USB MIDI",
+ umidi->chip->next_midi_device++,
+ out_ports, in_ports, &rmidi);
+ if (err < 0)
+ return err;
+ strcpy(rmidi->name, umidi->chip->card->shortname);
+ rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+ rmidi->ops = &snd_usbmidi_ops;
+ rmidi->private_data = umidi;
+ rmidi->private_free = snd_usbmidi_rawmidi_free;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usbmidi_output_ops);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usbmidi_input_ops);
+
+ umidi->rmidi = rmidi;
+ return 0;
+}
+
+/*
+ * Temporarily stop input.
+ */
+void snd_usbmidi_input_stop(struct list_head* p)
+{
+ struct snd_usb_midi* umidi;
+ int i;
+
+ umidi = list_entry(p, struct snd_usb_midi, list);
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
+ struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i];
+ if (ep->in)
+ usb_kill_urb(ep->in->urb);
+ }
+}
+
+static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep)
+{
+ if (ep) {
+ struct urb* urb = ep->urb;
+ urb->dev = ep->umidi->chip->dev;
+ snd_usbmidi_submit_urb(urb, GFP_KERNEL);
+ }
+}
+
+/*
+ * Resume input after a call to snd_usbmidi_input_stop().
+ */
+void snd_usbmidi_input_start(struct list_head* p)
+{
+ struct snd_usb_midi* umidi;
+ int i;
+
+ umidi = list_entry(p, struct snd_usb_midi, list);
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
+ snd_usbmidi_input_start_ep(umidi->endpoints[i].in);
+}
+
+/*
+ * Creates and registers everything needed for a MIDI streaming interface.
+ */
+int snd_usb_create_midi_interface(struct snd_usb_audio* chip,
+ struct usb_interface* iface,
+ const struct snd_usb_audio_quirk* quirk)
+{
+ struct snd_usb_midi* umidi;
+ struct snd_usb_midi_endpoint_info endpoints[MIDI_MAX_ENDPOINTS];
+ int out_ports, in_ports;
+ int i, err;
+
+ umidi = kzalloc(sizeof(*umidi), GFP_KERNEL);
+ if (!umidi)
+ return -ENOMEM;
+ umidi->chip = chip;
+ umidi->iface = iface;
+ umidi->quirk = quirk;
+ umidi->usb_protocol_ops = &snd_usbmidi_standard_ops;
+ init_timer(&umidi->error_timer);
+ spin_lock_init(&umidi->disc_lock);
+ umidi->error_timer.function = snd_usbmidi_error_timer;
+ umidi->error_timer.data = (unsigned long)umidi;
+
+ /* detect the endpoint(s) to use */
+ memset(endpoints, 0, sizeof(endpoints));
+ switch (quirk ? quirk->type : QUIRK_MIDI_STANDARD_INTERFACE) {
+ case QUIRK_MIDI_STANDARD_INTERFACE:
+ err = snd_usbmidi_get_ms_info(umidi, endpoints);
+ if (chip->usb_id == USB_ID(0x0763, 0x0150)) /* M-Audio Uno */
+ umidi->usb_protocol_ops =
+ &snd_usbmidi_maudio_broken_running_status_ops;
+ break;
+ case QUIRK_MIDI_US122L:
+ umidi->usb_protocol_ops = &snd_usbmidi_122l_ops;
+ /* fall through */
+ case QUIRK_MIDI_FIXED_ENDPOINT:
+ memcpy(&endpoints[0], quirk->data,
+ sizeof(struct snd_usb_midi_endpoint_info));
+ err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1);
+ break;
+ case QUIRK_MIDI_YAMAHA:
+ err = snd_usbmidi_detect_yamaha(umidi, &endpoints[0]);
+ break;
+ case QUIRK_MIDI_MIDIMAN:
+ umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops;
+ memcpy(&endpoints[0], quirk->data,
+ sizeof(struct snd_usb_midi_endpoint_info));
+ err = 0;
+ break;
+ case QUIRK_MIDI_NOVATION:
+ umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
+ err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+ break;
+ case QUIRK_MIDI_RAW:
+ umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
+ err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+ break;
+ case QUIRK_MIDI_EMAGIC:
+ umidi->usb_protocol_ops = &snd_usbmidi_emagic_ops;
+ memcpy(&endpoints[0], quirk->data,
+ sizeof(struct snd_usb_midi_endpoint_info));
+ err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1);
+ break;
+ case QUIRK_MIDI_CME:
+ umidi->usb_protocol_ops = &snd_usbmidi_cme_ops;
+ err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+ break;
+ default:
+ snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
+ err = -ENXIO;
+ break;
+ }
+ if (err < 0) {
+ kfree(umidi);
+ return err;
+ }
+
+ /* create rawmidi device */
+ out_ports = 0;
+ in_ports = 0;
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
+ out_ports += snd_usbmidi_count_bits(endpoints[i].out_cables);
+ in_ports += snd_usbmidi_count_bits(endpoints[i].in_cables);
+ }
+ err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports);
+ if (err < 0) {
+ kfree(umidi);
+ return err;
+ }
+
+ /* create endpoint/port structures */
+ if (quirk && quirk->type == QUIRK_MIDI_MIDIMAN)
+ err = snd_usbmidi_create_endpoints_midiman(umidi, &endpoints[0]);
+ else
+ err = snd_usbmidi_create_endpoints(umidi, endpoints);
+ if (err < 0) {
+ snd_usbmidi_free(umidi);
+ return err;
+ }
+
+ list_add(&umidi->list, &umidi->chip->midi_list);
+
+ for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
+ snd_usbmidi_input_start_ep(umidi->endpoints[i].in);
+ return 0;
+}
+
+EXPORT_SYMBOL(snd_usb_create_midi_interface);
+EXPORT_SYMBOL(snd_usbmidi_input_stop);
+EXPORT_SYMBOL(snd_usbmidi_input_start);
+EXPORT_SYMBOL(snd_usbmidi_disconnect);
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
new file mode 100644
index 0000000..a492461
--- /dev/null
+++ b/sound/usb/usbmixer.c
@@ -0,0 +1,2076 @@
+/*
+ * (Tentative) USB Audio Driver for ALSA
+ *
+ * Mixer control part
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * Many codes borrowed from audio.c by
+ * Alan Cox (alan@lxorguk.ukuu.org.uk)
+ * Thomas Sailer (sailer@ife.ee.ethz.ch)
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/bitops.h>
+#include <linux/init.h>
+#include <linux/list.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <linux/usb.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/hwdep.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+
+#include "usbaudio.h"
+
+/*
+ */
+
+/* ignore error from controls - for debugging */
+/* #define IGNORE_CTL_ERROR */
+
+/*
+ * Sound Blaster remote control configuration
+ *
+ * format of remote control data:
+ * Extigy: xx 00
+ * Audigy 2 NX: 06 80 xx 00 00 00
+ * Live! 24-bit: 06 80 xx yy 22 83
+ */
+static const struct rc_config {
+ u32 usb_id;
+ u8 offset;
+ u8 length;
+ u8 packet_length;
+ u8 min_packet_length; /* minimum accepted length of the URB result */
+ u8 mute_mixer_id;
+ u32 mute_code;
+} rc_configs[] = {
+ { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */
+ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */
+ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */
+};
+
+struct usb_mixer_interface {
+ struct snd_usb_audio *chip;
+ unsigned int ctrlif;
+ struct list_head list;
+ unsigned int ignore_ctl_error;
+ struct urb *urb;
+ struct usb_mixer_elem_info **id_elems; /* array[256], indexed by unit id */
+
+ /* Sound Blaster remote control stuff */
+ const struct rc_config *rc_cfg;
+ unsigned long rc_hwdep_open;
+ u32 rc_code;
+ wait_queue_head_t rc_waitq;
+ struct urb *rc_urb;
+ struct usb_ctrlrequest *rc_setup_packet;
+ u8 rc_buffer[6];
+
+ u8 audigy2nx_leds[3];
+};
+
+
+struct usb_audio_term {
+ int id;
+ int type;
+ int channels;
+ unsigned int chconfig;
+ int name;
+};
+
+struct usbmix_name_map;
+
+struct mixer_build {
+ struct snd_usb_audio *chip;
+ struct usb_mixer_interface *mixer;
+ unsigned char *buffer;
+ unsigned int buflen;
+ DECLARE_BITMAP(unitbitmap, 256);
+ struct usb_audio_term oterm;
+ const struct usbmix_name_map *map;
+ const struct usbmix_selector_map *selector_map;
+};
+
+struct usb_mixer_elem_info {
+ struct usb_mixer_interface *mixer;
+ struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */
+ struct snd_ctl_elem_id *elem_id;
+ unsigned int id;
+ unsigned int control; /* CS or ICN (high byte) */
+ unsigned int cmask; /* channel mask bitmap: 0 = master */
+ int channels;
+ int val_type;
+ int min, max, res;
+ u8 initialized;
+};
+
+
+enum {
+ USB_FEATURE_NONE = 0,
+ USB_FEATURE_MUTE = 1,
+ USB_FEATURE_VOLUME,
+ USB_FEATURE_BASS,
+ USB_FEATURE_MID,
+ USB_FEATURE_TREBLE,
+ USB_FEATURE_GEQ,
+ USB_FEATURE_AGC,
+ USB_FEATURE_DELAY,
+ USB_FEATURE_BASSBOOST,
+ USB_FEATURE_LOUDNESS
+};
+
+enum {
+ USB_MIXER_BOOLEAN,
+ USB_MIXER_INV_BOOLEAN,
+ USB_MIXER_S8,
+ USB_MIXER_U8,
+ USB_MIXER_S16,
+ USB_MIXER_U16,
+};
+
+enum {
+ USB_PROC_UPDOWN = 1,
+ USB_PROC_UPDOWN_SWITCH = 1,
+ USB_PROC_UPDOWN_MODE_SEL = 2,
+
+ USB_PROC_PROLOGIC = 2,
+ USB_PROC_PROLOGIC_SWITCH = 1,
+ USB_PROC_PROLOGIC_MODE_SEL = 2,
+
+ USB_PROC_3DENH = 3,
+ USB_PROC_3DENH_SWITCH = 1,
+ USB_PROC_3DENH_SPACE = 2,
+
+ USB_PROC_REVERB = 4,
+ USB_PROC_REVERB_SWITCH = 1,
+ USB_PROC_REVERB_LEVEL = 2,
+ USB_PROC_REVERB_TIME = 3,
+ USB_PROC_REVERB_DELAY = 4,
+
+ USB_PROC_CHORUS = 5,
+ USB_PROC_CHORUS_SWITCH = 1,
+ USB_PROC_CHORUS_LEVEL = 2,
+ USB_PROC_CHORUS_RATE = 3,
+ USB_PROC_CHORUS_DEPTH = 4,
+
+ USB_PROC_DCR = 6,
+ USB_PROC_DCR_SWITCH = 1,
+ USB_PROC_DCR_RATIO = 2,
+ USB_PROC_DCR_MAX_AMP = 3,
+ USB_PROC_DCR_THRESHOLD = 4,
+ USB_PROC_DCR_ATTACK = 5,
+ USB_PROC_DCR_RELEASE = 6,
+};
+
+#define MAX_CHANNELS 10 /* max logical channels */
+
+
+/*
+ * manual mapping of mixer names
+ * if the mixer topology is too complicated and the parsed names are
+ * ambiguous, add the entries in usbmixer_maps.c.
+ */
+#include "usbmixer_maps.c"
+
+/* get the mapped name if the unit matches */
+static int check_mapped_name(struct mixer_build *state, int unitid, int control, char *buf, int buflen)
+{
+ const struct usbmix_name_map *p;
+
+ if (! state->map)
+ return 0;
+
+ for (p = state->map; p->id; p++) {
+ if (p->id == unitid && p->name &&
+ (! control || ! p->control || control == p->control)) {
+ buflen--;
+ return strlcpy(buf, p->name, buflen);
+ }
+ }
+ return 0;
+}
+
+/* check whether the control should be ignored */
+static int check_ignored_ctl(struct mixer_build *state, int unitid, int control)
+{
+ const struct usbmix_name_map *p;
+
+ if (! state->map)
+ return 0;
+ for (p = state->map; p->id; p++) {
+ if (p->id == unitid && ! p->name &&
+ (! control || ! p->control || control == p->control)) {
+ // printk("ignored control %d:%d\n", unitid, control);
+ return 1;
+ }
+ }
+ return 0;
+}
+
+/* get the mapped selector source name */
+static int check_mapped_selector_name(struct mixer_build *state, int unitid,
+ int index, char *buf, int buflen)
+{
+ const struct usbmix_selector_map *p;
+
+ if (! state->selector_map)
+ return 0;
+ for (p = state->selector_map; p->id; p++) {
+ if (p->id == unitid && index < p->count)
+ return strlcpy(buf, p->names[index], buflen);
+ }
+ return 0;
+}
+
+/*
+ * find an audio control unit with the given unit id
+ */
+static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit)
+{
+ unsigned char *p;
+
+ p = NULL;
+ while ((p = snd_usb_find_desc(state->buffer, state->buflen, p,
+ USB_DT_CS_INTERFACE)) != NULL) {
+ if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT && p[3] == unit)
+ return p;
+ }
+ return NULL;
+}
+
+
+/*
+ * copy a string with the given id
+ */
+static int snd_usb_copy_string_desc(struct mixer_build *state, int index, char *buf, int maxlen)
+{
+ int len = usb_string(state->chip->dev, index, buf, maxlen - 1);
+ buf[len] = 0;
+ return len;
+}
+
+/*
+ * convert from the byte/word on usb descriptor to the zero-based integer
+ */
+static int convert_signed_value(struct usb_mixer_elem_info *cval, int val)
+{
+ switch (cval->val_type) {
+ case USB_MIXER_BOOLEAN:
+ return !!val;
+ case USB_MIXER_INV_BOOLEAN:
+ return !val;
+ case USB_MIXER_U8:
+ val &= 0xff;
+ break;
+ case USB_MIXER_S8:
+ val &= 0xff;
+ if (val >= 0x80)
+ val -= 0x100;
+ break;
+ case USB_MIXER_U16:
+ val &= 0xffff;
+ break;
+ case USB_MIXER_S16:
+ val &= 0xffff;
+ if (val >= 0x8000)
+ val -= 0x10000;
+ break;
+ }
+ return val;
+}
+
+/*
+ * convert from the zero-based int to the byte/word for usb descriptor
+ */
+static int convert_bytes_value(struct usb_mixer_elem_info *cval, int val)
+{
+ switch (cval->val_type) {
+ case USB_MIXER_BOOLEAN:
+ return !!val;
+ case USB_MIXER_INV_BOOLEAN:
+ return !val;
+ case USB_MIXER_S8:
+ case USB_MIXER_U8:
+ return val & 0xff;
+ case USB_MIXER_S16:
+ case USB_MIXER_U16:
+ return val & 0xffff;
+ }
+ return 0; /* not reached */
+}
+
+static int get_relative_value(struct usb_mixer_elem_info *cval, int val)
+{
+ if (! cval->res)
+ cval->res = 1;
+ if (val < cval->min)
+ return 0;
+ else if (val >= cval->max)
+ return (cval->max - cval->min + cval->res - 1) / cval->res;
+ else
+ return (val - cval->min) / cval->res;
+}
+
+static int get_abs_value(struct usb_mixer_elem_info *cval, int val)
+{
+ if (val < 0)
+ return cval->min;
+ if (! cval->res)
+ cval->res = 1;
+ val *= cval->res;
+ val += cval->min;
+ if (val > cval->max)
+ return cval->max;
+ return val;
+}
+
+
+/*
+ * retrieve a mixer value
+ */
+
+static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret)
+{
+ unsigned char buf[2];
+ int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1;
+ int timeout = 10;
+
+ while (timeout-- > 0) {
+ if (snd_usb_ctl_msg(cval->mixer->chip->dev,
+ usb_rcvctrlpipe(cval->mixer->chip->dev, 0),
+ request,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+ validx, cval->mixer->ctrlif | (cval->id << 8),
+ buf, val_len, 100) >= val_len) {
+ *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len));
+ return 0;
+ }
+ }
+ snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
+ request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type);
+ return -EINVAL;
+}
+
+static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value)
+{
+ return get_ctl_value(cval, GET_CUR, validx, value);
+}
+
+/* channel = 0: master, 1 = first channel */
+static inline int get_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int *value)
+{
+ return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value);
+}
+
+/*
+ * set a mixer value
+ */
+
+static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set)
+{
+ unsigned char buf[2];
+ int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1;
+ int timeout = 10;
+
+ value_set = convert_bytes_value(cval, value_set);
+ buf[0] = value_set & 0xff;
+ buf[1] = (value_set >> 8) & 0xff;
+ while (timeout -- > 0)
+ if (snd_usb_ctl_msg(cval->mixer->chip->dev,
+ usb_sndctrlpipe(cval->mixer->chip->dev, 0),
+ request,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+ validx, cval->mixer->ctrlif | (cval->id << 8),
+ buf, val_len, 100) >= 0)
+ return 0;
+ snd_printdd(KERN_ERR "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n",
+ request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type, buf[0], buf[1]);
+ return -EINVAL;
+}
+
+static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value)
+{
+ return set_ctl_value(cval, SET_CUR, validx, value);
+}
+
+static inline int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int value)
+{
+ return set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, value);
+}
+
+/*
+ * TLV callback for mixer volume controls
+ */
+static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *_tlv)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ DECLARE_TLV_DB_SCALE(scale, 0, 0, 0);
+
+ if (size < sizeof(scale))
+ return -ENOMEM;
+ /* USB descriptions contain the dB scale in 1/256 dB unit
+ * while ALSA TLV contains in 1/100 dB unit
+ */
+ scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256;
+ scale[3] = (convert_signed_value(cval, cval->res) * 100) / 256;
+ if (copy_to_user(_tlv, scale, sizeof(scale)))
+ return -EFAULT;
+ return 0;
+}
+
+/*
+ * parser routines begin here...
+ */
+
+static int parse_audio_unit(struct mixer_build *state, int unitid);
+
+
+/*
+ * check if the input/output channel routing is enabled on the given bitmap.
+ * used for mixer unit parser
+ */
+static int check_matrix_bitmap(unsigned char *bmap, int ich, int och, int num_outs)
+{
+ int idx = ich * num_outs + och;
+ return bmap[idx >> 3] & (0x80 >> (idx & 7));
+}
+
+
+/*
+ * add an alsa control element
+ * search and increment the index until an empty slot is found.
+ *
+ * if failed, give up and free the control instance.
+ */
+
+static int add_control_to_empty(struct mixer_build *state, struct snd_kcontrol *kctl)
+{
+ struct usb_mixer_elem_info *cval = kctl->private_data;
+ int err;
+
+ while (snd_ctl_find_id(state->chip->card, &kctl->id))
+ kctl->id.index++;
+ if ((err = snd_ctl_add(state->chip->card, kctl)) < 0) {
+ snd_printd(KERN_ERR "cannot add control (err = %d)\n", err);
+ return err;
+ }
+ cval->elem_id = &kctl->id;
+ cval->next_id_elem = state->mixer->id_elems[cval->id];
+ state->mixer->id_elems[cval->id] = cval;
+ return 0;
+}
+
+
+/*
+ * get a terminal name string
+ */
+
+static struct iterm_name_combo {
+ int type;
+ char *name;
+} iterm_names[] = {
+ { 0x0300, "Output" },
+ { 0x0301, "Speaker" },
+ { 0x0302, "Headphone" },
+ { 0x0303, "HMD Audio" },
+ { 0x0304, "Desktop Speaker" },
+ { 0x0305, "Room Speaker" },
+ { 0x0306, "Com Speaker" },
+ { 0x0307, "LFE" },
+ { 0x0600, "External In" },
+ { 0x0601, "Analog In" },
+ { 0x0602, "Digital In" },
+ { 0x0603, "Line" },
+ { 0x0604, "Legacy In" },
+ { 0x0605, "IEC958 In" },
+ { 0x0606, "1394 DA Stream" },
+ { 0x0607, "1394 DV Stream" },
+ { 0x0700, "Embedded" },
+ { 0x0701, "Noise Source" },
+ { 0x0702, "Equalization Noise" },
+ { 0x0703, "CD" },
+ { 0x0704, "DAT" },
+ { 0x0705, "DCC" },
+ { 0x0706, "MiniDisk" },
+ { 0x0707, "Analog Tape" },
+ { 0x0708, "Phonograph" },
+ { 0x0709, "VCR Audio" },
+ { 0x070a, "Video Disk Audio" },
+ { 0x070b, "DVD Audio" },
+ { 0x070c, "TV Tuner Audio" },
+ { 0x070d, "Satellite Rec Audio" },
+ { 0x070e, "Cable Tuner Audio" },
+ { 0x070f, "DSS Audio" },
+ { 0x0710, "Radio Receiver" },
+ { 0x0711, "Radio Transmitter" },
+ { 0x0712, "Multi-Track Recorder" },
+ { 0x0713, "Synthesizer" },
+ { 0 },
+};
+
+static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm,
+ unsigned char *name, int maxlen, int term_only)
+{
+ struct iterm_name_combo *names;
+
+ if (iterm->name)
+ return snd_usb_copy_string_desc(state, iterm->name, name, maxlen);
+
+ /* virtual type - not a real terminal */
+ if (iterm->type >> 16) {
+ if (term_only)
+ return 0;
+ switch (iterm->type >> 16) {
+ case SELECTOR_UNIT:
+ strcpy(name, "Selector"); return 8;
+ case PROCESSING_UNIT:
+ strcpy(name, "Process Unit"); return 12;
+ case EXTENSION_UNIT:
+ strcpy(name, "Ext Unit"); return 8;
+ case MIXER_UNIT:
+ strcpy(name, "Mixer"); return 5;
+ default:
+ return sprintf(name, "Unit %d", iterm->id);
+ }
+ }
+
+ switch (iterm->type & 0xff00) {
+ case 0x0100:
+ strcpy(name, "PCM"); return 3;
+ case 0x0200:
+ strcpy(name, "Mic"); return 3;
+ case 0x0400:
+ strcpy(name, "Headset"); return 7;
+ case 0x0500:
+ strcpy(name, "Phone"); return 5;
+ }
+
+ for (names = iterm_names; names->type; names++)
+ if (names->type == iterm->type) {
+ strcpy(name, names->name);
+ return strlen(names->name);
+ }
+ return 0;
+}
+
+
+/*
+ * parse the source unit recursively until it reaches to a terminal
+ * or a branched unit.
+ */
+static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term)
+{
+ unsigned char *p1;
+
+ memset(term, 0, sizeof(*term));
+ while ((p1 = find_audio_control_unit(state, id)) != NULL) {
+ term->id = id;
+ switch (p1[2]) {
+ case INPUT_TERMINAL:
+ term->type = combine_word(p1 + 4);
+ term->channels = p1[7];
+ term->chconfig = combine_word(p1 + 8);
+ term->name = p1[11];
+ return 0;
+ case FEATURE_UNIT:
+ id = p1[4];
+ break; /* continue to parse */
+ case MIXER_UNIT:
+ term->type = p1[2] << 16; /* virtual type */
+ term->channels = p1[5 + p1[4]];
+ term->chconfig = combine_word(p1 + 6 + p1[4]);
+ term->name = p1[p1[0] - 1];
+ return 0;
+ case SELECTOR_UNIT:
+ /* call recursively to retrieve the channel info */
+ if (check_input_term(state, p1[5], term) < 0)
+ return -ENODEV;
+ term->type = p1[2] << 16; /* virtual type */
+ term->id = id;
+ term->name = p1[9 + p1[0] - 1];
+ return 0;
+ case PROCESSING_UNIT:
+ case EXTENSION_UNIT:
+ if (p1[6] == 1) {
+ id = p1[7];
+ break; /* continue to parse */
+ }
+ term->type = p1[2] << 16; /* virtual type */
+ term->channels = p1[7 + p1[6]];
+ term->chconfig = combine_word(p1 + 8 + p1[6]);
+ term->name = p1[12 + p1[6] + p1[11 + p1[6]]];
+ return 0;
+ default:
+ return -ENODEV;
+ }
+ }
+ return -ENODEV;
+}
+
+
+/*
+ * Feature Unit
+ */
+
+/* feature unit control information */
+struct usb_feature_control_info {
+ const char *name;
+ unsigned int type; /* control type (mute, volume, etc.) */
+};
+
+static struct usb_feature_control_info audio_feature_info[] = {
+ { "Mute", USB_MIXER_INV_BOOLEAN },
+ { "Volume", USB_MIXER_S16 },
+ { "Tone Control - Bass", USB_MIXER_S8 },
+ { "Tone Control - Mid", USB_MIXER_S8 },
+ { "Tone Control - Treble", USB_MIXER_S8 },
+ { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */
+ { "Auto Gain Control", USB_MIXER_BOOLEAN },
+ { "Delay Control", USB_MIXER_U16 },
+ { "Bass Boost", USB_MIXER_BOOLEAN },
+ { "Loudness", USB_MIXER_BOOLEAN },
+};
+
+
+/* private_free callback */
+static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
+{
+ kfree(kctl->private_data);
+ kctl->private_data = NULL;
+}
+
+
+/*
+ * interface to ALSA control for feature/mixer units
+ */
+
+/*
+ * retrieve the minimum and maximum values for the specified control
+ */
+static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
+{
+ /* for failsafe */
+ cval->min = default_min;
+ cval->max = cval->min + 1;
+ cval->res = 1;
+
+ if (cval->val_type == USB_MIXER_BOOLEAN ||
+ cval->val_type == USB_MIXER_INV_BOOLEAN) {
+ cval->initialized = 1;
+ } else {
+ int minchn = 0;
+ if (cval->cmask) {
+ int i;
+ for (i = 0; i < MAX_CHANNELS; i++)
+ if (cval->cmask & (1 << i)) {
+ minchn = i + 1;
+ break;
+ }
+ }
+ if (get_ctl_value(cval, GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 ||
+ get_ctl_value(cval, GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
+ snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n",
+ cval->id, cval->mixer->ctrlif, cval->control, cval->id);
+ return -EINVAL;
+ }
+ if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) {
+ cval->res = 1;
+ } else {
+ int last_valid_res = cval->res;
+
+ while (cval->res > 1) {
+ if (set_ctl_value(cval, SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0)
+ break;
+ cval->res /= 2;
+ }
+ if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0)
+ cval->res = last_valid_res;
+ }
+ if (cval->res == 0)
+ cval->res = 1;
+
+ /* Additional checks for the proper resolution
+ *
+ * Some devices report smaller resolutions than actually
+ * reacting. They don't return errors but simply clip
+ * to the lower aligned value.
+ */
+ if (cval->min + cval->res < cval->max) {
+ int last_valid_res = cval->res;
+ int saved, test, check;
+ get_cur_mix_value(cval, minchn, &saved);
+ for (;;) {
+ test = saved;
+ if (test < cval->max)
+ test += cval->res;
+ else
+ test -= cval->res;
+ if (test < cval->min || test > cval->max ||
+ set_cur_mix_value(cval, minchn, test) ||
+ get_cur_mix_value(cval, minchn, &check)) {
+ cval->res = last_valid_res;
+ break;
+ }
+ if (test == check)
+ break;
+ cval->res *= 2;
+ }
+ set_cur_mix_value(cval, minchn, saved);
+ }
+
+ cval->initialized = 1;
+ }
+ return 0;
+}
+
+
+/* get a feature/mixer unit info */
+static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+
+ if (cval->val_type == USB_MIXER_BOOLEAN ||
+ cval->val_type == USB_MIXER_INV_BOOLEAN)
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = cval->channels;
+ if (cval->val_type == USB_MIXER_BOOLEAN ||
+ cval->val_type == USB_MIXER_INV_BOOLEAN) {
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ } else {
+ if (! cval->initialized)
+ get_min_max(cval, 0);
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max =
+ (cval->max - cval->min + cval->res - 1) / cval->res;
+ }
+ return 0;
+}
+
+/* get the current value from feature/mixer unit */
+static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ int c, cnt, val, err;
+
+ if (cval->cmask) {
+ cnt = 0;
+ for (c = 0; c < MAX_CHANNELS; c++) {
+ if (cval->cmask & (1 << c)) {
+ err = get_cur_mix_value(cval, c + 1, &val);
+ if (err < 0) {
+ if (cval->mixer->ignore_ctl_error) {
+ ucontrol->value.integer.value[0] = cval->min;
+ return 0;
+ }
+ snd_printd(KERN_ERR "cannot get current value for control %d ch %d: err = %d\n", cval->control, c + 1, err);
+ return err;
+ }
+ val = get_relative_value(cval, val);
+ ucontrol->value.integer.value[cnt] = val;
+ cnt++;
+ }
+ }
+ } else {
+ /* master channel */
+ err = get_cur_mix_value(cval, 0, &val);
+ if (err < 0) {
+ if (cval->mixer->ignore_ctl_error) {
+ ucontrol->value.integer.value[0] = cval->min;
+ return 0;
+ }
+ snd_printd(KERN_ERR "cannot get current value for control %d master ch: err = %d\n", cval->control, err);
+ return err;
+ }
+ val = get_relative_value(cval, val);
+ ucontrol->value.integer.value[0] = val;
+ }
+ return 0;
+}
+
+/* put the current value to feature/mixer unit */
+static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ int c, cnt, val, oval, err;
+ int changed = 0;
+
+ if (cval->cmask) {
+ cnt = 0;
+ for (c = 0; c < MAX_CHANNELS; c++) {
+ if (cval->cmask & (1 << c)) {
+ err = get_cur_mix_value(cval, c + 1, &oval);
+ if (err < 0) {
+ if (cval->mixer->ignore_ctl_error)
+ return 0;
+ return err;
+ }
+ val = ucontrol->value.integer.value[cnt];
+ val = get_abs_value(cval, val);
+ if (oval != val) {
+ set_cur_mix_value(cval, c + 1, val);
+ changed = 1;
+ }
+ get_cur_mix_value(cval, c + 1, &val);
+ cnt++;
+ }
+ }
+ } else {
+ /* master channel */
+ err = get_cur_mix_value(cval, 0, &oval);
+ if (err < 0 && cval->mixer->ignore_ctl_error)
+ return 0;
+ if (err < 0)
+ return err;
+ val = ucontrol->value.integer.value[0];
+ val = get_abs_value(cval, val);
+ if (val != oval) {
+ set_cur_mix_value(cval, 0, val);
+ changed = 1;
+ }
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new usb_feature_unit_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "", /* will be filled later manually */
+ .info = mixer_ctl_feature_info,
+ .get = mixer_ctl_feature_get,
+ .put = mixer_ctl_feature_put,
+};
+
+
+/*
+ * build a feature control
+ */
+
+static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
+ unsigned int ctl_mask, int control,
+ struct usb_audio_term *iterm, int unitid)
+{
+ unsigned int len = 0;
+ int mapped_name = 0;
+ int nameid = desc[desc[0] - 1];
+ struct snd_kcontrol *kctl;
+ struct usb_mixer_elem_info *cval;
+
+ control++; /* change from zero-based to 1-based value */
+
+ if (control == USB_FEATURE_GEQ) {
+ /* FIXME: not supported yet */
+ return;
+ }
+
+ if (check_ignored_ctl(state, unitid, control))
+ return;
+
+ cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+ if (! cval) {
+ snd_printk(KERN_ERR "cannot malloc kcontrol\n");
+ return;
+ }
+ cval->mixer = state->mixer;
+ cval->id = unitid;
+ cval->control = control;
+ cval->cmask = ctl_mask;
+ cval->val_type = audio_feature_info[control-1].type;
+ if (ctl_mask == 0)
+ cval->channels = 1; /* master channel */
+ else {
+ int i, c = 0;
+ for (i = 0; i < 16; i++)
+ if (ctl_mask & (1 << i))
+ c++;
+ cval->channels = c;
+ }
+
+ /* get min/max values */
+ get_min_max(cval, 0);
+
+ kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
+ if (! kctl) {
+ snd_printk(KERN_ERR "cannot malloc kcontrol\n");
+ kfree(cval);
+ return;
+ }
+ kctl->private_free = usb_mixer_elem_free;
+
+ len = check_mapped_name(state, unitid, control, kctl->id.name, sizeof(kctl->id.name));
+ mapped_name = len != 0;
+ if (! len && nameid)
+ len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name));
+
+ switch (control) {
+ case USB_FEATURE_MUTE:
+ case USB_FEATURE_VOLUME:
+ /* determine the control name. the rule is:
+ * - if a name id is given in descriptor, use it.
+ * - if the connected input can be determined, then use the name
+ * of terminal type.
+ * - if the connected output can be determined, use it.
+ * - otherwise, anonymous name.
+ */
+ if (! len) {
+ len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 1);
+ if (! len)
+ len = get_term_name(state, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 1);
+ if (! len)
+ len = snprintf(kctl->id.name, sizeof(kctl->id.name),
+ "Feature %d", unitid);
+ }
+ /* determine the stream direction:
+ * if the connected output is USB stream, then it's likely a
+ * capture stream. otherwise it should be playback (hopefully :)
+ */
+ if (! mapped_name && ! (state->oterm.type >> 16)) {
+ if ((state->oterm.type & 0xff00) == 0x0100) {
+ len = strlcat(kctl->id.name, " Capture", sizeof(kctl->id.name));
+ } else {
+ len = strlcat(kctl->id.name + len, " Playback", sizeof(kctl->id.name));
+ }
+ }
+ strlcat(kctl->id.name + len, control == USB_FEATURE_MUTE ? " Switch" : " Volume",
+ sizeof(kctl->id.name));
+ if (control == USB_FEATURE_VOLUME) {
+ kctl->tlv.c = mixer_vol_tlv;
+ kctl->vd[0].access |=
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+ }
+ break;
+
+ default:
+ if (! len)
+ strlcpy(kctl->id.name, audio_feature_info[control-1].name,
+ sizeof(kctl->id.name));
+ break;
+ }
+
+ /* quirk for UDA1321/N101 */
+ /* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */
+ /* is not very clear from datasheets */
+ /* I hope that the min value is -15360 for newer firmware --jk */
+ switch (state->chip->usb_id) {
+ case USB_ID(0x0471, 0x0101):
+ case USB_ID(0x0471, 0x0104):
+ case USB_ID(0x0471, 0x0105):
+ case USB_ID(0x0672, 0x1041):
+ if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
+ cval->min == -15616) {
+ snd_printk(KERN_INFO "using volume control quirk for the UDA1321/N101 chip\n");
+ cval->max = -256;
+ }
+ }
+
+ snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
+ cval->id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res);
+ add_control_to_empty(state, kctl);
+}
+
+
+
+/*
+ * parse a feature unit
+ *
+ * most of controlls are defined here.
+ */
+static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsigned char *ftr)
+{
+ int channels, i, j;
+ struct usb_audio_term iterm;
+ unsigned int master_bits, first_ch_bits;
+ int err, csize;
+
+ if (ftr[0] < 7 || ! (csize = ftr[5]) || ftr[0] < 7 + csize) {
+ snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid);
+ return -EINVAL;
+ }
+
+ /* parse the source unit */
+ if ((err = parse_audio_unit(state, ftr[4])) < 0)
+ return err;
+
+ /* determine the input source type and name */
+ if (check_input_term(state, ftr[4], &iterm) < 0)
+ return -EINVAL;
+
+ channels = (ftr[0] - 7) / csize - 1;
+
+ master_bits = snd_usb_combine_bytes(ftr + 6, csize);
+ if (channels > 0)
+ first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize);
+ else
+ first_ch_bits = 0;
+ /* check all control types */
+ for (i = 0; i < 10; i++) {
+ unsigned int ch_bits = 0;
+ for (j = 0; j < channels; j++) {
+ unsigned int mask = snd_usb_combine_bytes(ftr + 6 + csize * (j+1), csize);
+ if (mask & (1 << i))
+ ch_bits |= (1 << j);
+ }
+ if (ch_bits & 1) /* the first channel must be set (for ease of programming) */
+ build_feature_ctl(state, ftr, ch_bits, i, &iterm, unitid);
+ if (master_bits & (1 << i))
+ build_feature_ctl(state, ftr, 0, i, &iterm, unitid);
+ }
+
+ return 0;
+}
+
+
+/*
+ * Mixer Unit
+ */
+
+/*
+ * build a mixer unit control
+ *
+ * the callbacks are identical with feature unit.
+ * input channel number (zero based) is given in control field instead.
+ */
+
+static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc,
+ int in_pin, int in_ch, int unitid,
+ struct usb_audio_term *iterm)
+{
+ struct usb_mixer_elem_info *cval;
+ unsigned int input_pins = desc[4];
+ unsigned int num_outs = desc[5 + input_pins];
+ unsigned int i, len;
+ struct snd_kcontrol *kctl;
+
+ if (check_ignored_ctl(state, unitid, 0))
+ return;
+
+ cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+ if (! cval)
+ return;
+
+ cval->mixer = state->mixer;
+ cval->id = unitid;
+ cval->control = in_ch + 1; /* based on 1 */
+ cval->val_type = USB_MIXER_S16;
+ for (i = 0; i < num_outs; i++) {
+ if (check_matrix_bitmap(desc + 9 + input_pins, in_ch, i, num_outs)) {
+ cval->cmask |= (1 << i);
+ cval->channels++;
+ }
+ }
+
+ /* get min/max values */
+ get_min_max(cval, 0);
+
+ kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
+ if (! kctl) {
+ snd_printk(KERN_ERR "cannot malloc kcontrol\n");
+ kfree(cval);
+ return;
+ }
+ kctl->private_free = usb_mixer_elem_free;
+
+ len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name));
+ if (! len)
+ len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0);
+ if (! len)
+ len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
+ strlcat(kctl->id.name + len, " Volume", sizeof(kctl->id.name));
+
+ snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
+ cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
+ add_control_to_empty(state, kctl);
+}
+
+
+/*
+ * parse a mixer unit
+ */
+static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, unsigned char *desc)
+{
+ struct usb_audio_term iterm;
+ int input_pins, num_ins, num_outs;
+ int pin, ich, err;
+
+ if (desc[0] < 11 || ! (input_pins = desc[4]) || ! (num_outs = desc[5 + input_pins])) {
+ snd_printk(KERN_ERR "invalid MIXER UNIT descriptor %d\n", unitid);
+ return -EINVAL;
+ }
+ /* no bmControls field (e.g. Maya44) -> ignore */
+ if (desc[0] <= 10 + input_pins) {
+ snd_printdd(KERN_INFO "MU %d has no bmControls field\n", unitid);
+ return 0;
+ }
+
+ num_ins = 0;
+ ich = 0;
+ for (pin = 0; pin < input_pins; pin++) {
+ err = parse_audio_unit(state, desc[5 + pin]);
+ if (err < 0)
+ return err;
+ err = check_input_term(state, desc[5 + pin], &iterm);
+ if (err < 0)
+ return err;
+ num_ins += iterm.channels;
+ for (; ich < num_ins; ++ich) {
+ int och, ich_has_controls = 0;
+
+ for (och = 0; och < num_outs; ++och) {
+ if (check_matrix_bitmap(desc + 9 + input_pins,
+ ich, och, num_outs)) {
+ ich_has_controls = 1;
+ break;
+ }
+ }
+ if (ich_has_controls)
+ build_mixer_unit_ctl(state, desc, pin, ich,
+ unitid, &iterm);
+ }
+ }
+ return 0;
+}
+
+
+/*
+ * Processing Unit / Extension Unit
+ */
+
+/* get callback for processing/extension unit */
+static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ int err, val;
+
+ err = get_cur_ctl_value(cval, cval->control << 8, &val);
+ if (err < 0 && cval->mixer->ignore_ctl_error) {
+ ucontrol->value.integer.value[0] = cval->min;
+ return 0;
+ }
+ if (err < 0)
+ return err;
+ val = get_relative_value(cval, val);
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+}
+
+/* put callback for processing/extension unit */
+static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ int val, oval, err;
+
+ err = get_cur_ctl_value(cval, cval->control << 8, &oval);
+ if (err < 0) {
+ if (cval->mixer->ignore_ctl_error)
+ return 0;
+ return err;
+ }
+ val = ucontrol->value.integer.value[0];
+ val = get_abs_value(cval, val);
+ if (val != oval) {
+ set_cur_ctl_value(cval, cval->control << 8, val);
+ return 1;
+ }
+ return 0;
+}
+
+/* alsa control interface for processing/extension unit */
+static struct snd_kcontrol_new mixer_procunit_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "", /* will be filled later */
+ .info = mixer_ctl_feature_info,
+ .get = mixer_ctl_procunit_get,
+ .put = mixer_ctl_procunit_put,
+};
+
+
+/*
+ * predefined data for processing units
+ */
+struct procunit_value_info {
+ int control;
+ char *suffix;
+ int val_type;
+ int min_value;
+};
+
+struct procunit_info {
+ int type;
+ char *name;
+ struct procunit_value_info *values;
+};
+
+static struct procunit_value_info updown_proc_info[] = {
+ { USB_PROC_UPDOWN_SWITCH, "Switch", USB_MIXER_BOOLEAN },
+ { USB_PROC_UPDOWN_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
+ { 0 }
+};
+static struct procunit_value_info prologic_proc_info[] = {
+ { USB_PROC_PROLOGIC_SWITCH, "Switch", USB_MIXER_BOOLEAN },
+ { USB_PROC_PROLOGIC_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
+ { 0 }
+};
+static struct procunit_value_info threed_enh_proc_info[] = {
+ { USB_PROC_3DENH_SWITCH, "Switch", USB_MIXER_BOOLEAN },
+ { USB_PROC_3DENH_SPACE, "Spaciousness", USB_MIXER_U8 },
+ { 0 }
+};
+static struct procunit_value_info reverb_proc_info[] = {
+ { USB_PROC_REVERB_SWITCH, "Switch", USB_MIXER_BOOLEAN },
+ { USB_PROC_REVERB_LEVEL, "Level", USB_MIXER_U8 },
+ { USB_PROC_REVERB_TIME, "Time", USB_MIXER_U16 },
+ { USB_PROC_REVERB_DELAY, "Delay", USB_MIXER_U8 },
+ { 0 }
+};
+static struct procunit_value_info chorus_proc_info[] = {
+ { USB_PROC_CHORUS_SWITCH, "Switch", USB_MIXER_BOOLEAN },
+ { USB_PROC_CHORUS_LEVEL, "Level", USB_MIXER_U8 },
+ { USB_PROC_CHORUS_RATE, "Rate", USB_MIXER_U16 },
+ { USB_PROC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 },
+ { 0 }
+};
+static struct procunit_value_info dcr_proc_info[] = {
+ { USB_PROC_DCR_SWITCH, "Switch", USB_MIXER_BOOLEAN },
+ { USB_PROC_DCR_RATIO, "Ratio", USB_MIXER_U16 },
+ { USB_PROC_DCR_MAX_AMP, "Max Amp", USB_MIXER_S16 },
+ { USB_PROC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 },
+ { USB_PROC_DCR_ATTACK, "Attack Time", USB_MIXER_U16 },
+ { USB_PROC_DCR_RELEASE, "Release Time", USB_MIXER_U16 },
+ { 0 }
+};
+
+static struct procunit_info procunits[] = {
+ { USB_PROC_UPDOWN, "Up Down", updown_proc_info },
+ { USB_PROC_PROLOGIC, "Dolby Prologic", prologic_proc_info },
+ { USB_PROC_3DENH, "3D Stereo Extender", threed_enh_proc_info },
+ { USB_PROC_REVERB, "Reverb", reverb_proc_info },
+ { USB_PROC_CHORUS, "Chorus", chorus_proc_info },
+ { USB_PROC_DCR, "DCR", dcr_proc_info },
+ { 0 },
+};
+
+/*
+ * build a processing/extension unit
+ */
+static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned char *dsc, struct procunit_info *list, char *name)
+{
+ int num_ins = dsc[6];
+ struct usb_mixer_elem_info *cval;
+ struct snd_kcontrol *kctl;
+ int i, err, nameid, type, len;
+ struct procunit_info *info;
+ struct procunit_value_info *valinfo;
+ static struct procunit_value_info default_value_info[] = {
+ { 0x01, "Switch", USB_MIXER_BOOLEAN },
+ { 0 }
+ };
+ static struct procunit_info default_info = {
+ 0, NULL, default_value_info
+ };
+
+ if (dsc[0] < 13 || dsc[0] < 13 + num_ins || dsc[0] < num_ins + dsc[11 + num_ins]) {
+ snd_printk(KERN_ERR "invalid %s descriptor (id %d)\n", name, unitid);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < num_ins; i++) {
+ if ((err = parse_audio_unit(state, dsc[7 + i])) < 0)
+ return err;
+ }
+
+ type = combine_word(&dsc[4]);
+ for (info = list; info && info->type; info++)
+ if (info->type == type)
+ break;
+ if (! info || ! info->type)
+ info = &default_info;
+
+ for (valinfo = info->values; valinfo->control; valinfo++) {
+ /* FIXME: bitmap might be longer than 8bit */
+ if (! (dsc[12 + num_ins] & (1 << (valinfo->control - 1))))
+ continue;
+ if (check_ignored_ctl(state, unitid, valinfo->control))
+ continue;
+ cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+ if (! cval) {
+ snd_printk(KERN_ERR "cannot malloc kcontrol\n");
+ return -ENOMEM;
+ }
+ cval->mixer = state->mixer;
+ cval->id = unitid;
+ cval->control = valinfo->control;
+ cval->val_type = valinfo->val_type;
+ cval->channels = 1;
+
+ /* get min/max values */
+ if (type == USB_PROC_UPDOWN && cval->control == USB_PROC_UPDOWN_MODE_SEL) {
+ /* FIXME: hard-coded */
+ cval->min = 1;
+ cval->max = dsc[15];
+ cval->res = 1;
+ cval->initialized = 1;
+ } else
+ get_min_max(cval, valinfo->min_value);
+
+ kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
+ if (! kctl) {
+ snd_printk(KERN_ERR "cannot malloc kcontrol\n");
+ kfree(cval);
+ return -ENOMEM;
+ }
+ kctl->private_free = usb_mixer_elem_free;
+
+ if (check_mapped_name(state, unitid, cval->control, kctl->id.name, sizeof(kctl->id.name)))
+ ;
+ else if (info->name)
+ strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name));
+ else {
+ nameid = dsc[12 + num_ins + dsc[11 + num_ins]];
+ len = 0;
+ if (nameid)
+ len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name));
+ if (! len)
+ strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
+ }
+ strlcat(kctl->id.name, " ", sizeof(kctl->id.name));
+ strlcat(kctl->id.name, valinfo->suffix, sizeof(kctl->id.name));
+
+ snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
+ cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
+ if ((err = add_control_to_empty(state, kctl)) < 0)
+ return err;
+ }
+ return 0;
+}
+
+
+static int parse_audio_processing_unit(struct mixer_build *state, int unitid, unsigned char *desc)
+{
+ return build_audio_procunit(state, unitid, desc, procunits, "Processing Unit");
+}
+
+static int parse_audio_extension_unit(struct mixer_build *state, int unitid, unsigned char *desc)
+{
+ return build_audio_procunit(state, unitid, desc, NULL, "Extension Unit");
+}
+
+
+/*
+ * Selector Unit
+ */
+
+/* info callback for selector unit
+ * use an enumerator type for routing
+ */
+static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ char **itemlist = (char **)kcontrol->private_value;
+
+ if (snd_BUG_ON(!itemlist))
+ return -EINVAL;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = cval->max;
+ if ((int)uinfo->value.enumerated.item >= cval->max)
+ uinfo->value.enumerated.item = cval->max - 1;
+ strcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+/* get callback for selector unit */
+static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ int val, err;
+
+ err = get_cur_ctl_value(cval, 0, &val);
+ if (err < 0) {
+ if (cval->mixer->ignore_ctl_error) {
+ ucontrol->value.enumerated.item[0] = 0;
+ return 0;
+ }
+ return err;
+ }
+ val = get_relative_value(cval, val);
+ ucontrol->value.enumerated.item[0] = val;
+ return 0;
+}
+
+/* put callback for selector unit */
+static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_elem_info *cval = kcontrol->private_data;
+ int val, oval, err;
+
+ err = get_cur_ctl_value(cval, 0, &oval);
+ if (err < 0) {
+ if (cval->mixer->ignore_ctl_error)
+ return 0;
+ return err;
+ }
+ val = ucontrol->value.enumerated.item[0];
+ val = get_abs_value(cval, val);
+ if (val != oval) {
+ set_cur_ctl_value(cval, 0, val);
+ return 1;
+ }
+ return 0;
+}
+
+/* alsa control interface for selector unit */
+static struct snd_kcontrol_new mixer_selectunit_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "", /* will be filled later */
+ .info = mixer_ctl_selector_info,
+ .get = mixer_ctl_selector_get,
+ .put = mixer_ctl_selector_put,
+};
+
+
+/* private free callback.
+ * free both private_data and private_value
+ */
+static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl)
+{
+ int i, num_ins = 0;
+
+ if (kctl->private_data) {
+ struct usb_mixer_elem_info *cval = kctl->private_data;
+ num_ins = cval->max;
+ kfree(cval);
+ kctl->private_data = NULL;
+ }
+ if (kctl->private_value) {
+ char **itemlist = (char **)kctl->private_value;
+ for (i = 0; i < num_ins; i++)
+ kfree(itemlist[i]);
+ kfree(itemlist);
+ kctl->private_value = 0;
+ }
+}
+
+/*
+ * parse a selector unit
+ */
+static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsigned char *desc)
+{
+ unsigned int num_ins = desc[4];
+ unsigned int i, nameid, len;
+ int err;
+ struct usb_mixer_elem_info *cval;
+ struct snd_kcontrol *kctl;
+ char **namelist;
+
+ if (! num_ins || desc[0] < 5 + num_ins) {
+ snd_printk(KERN_ERR "invalid SELECTOR UNIT descriptor %d\n", unitid);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < num_ins; i++) {
+ if ((err = parse_audio_unit(state, desc[5 + i])) < 0)
+ return err;
+ }
+
+ if (num_ins == 1) /* only one ? nonsense! */
+ return 0;
+
+ if (check_ignored_ctl(state, unitid, 0))
+ return 0;
+
+ cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+ if (! cval) {
+ snd_printk(KERN_ERR "cannot malloc kcontrol\n");
+ return -ENOMEM;
+ }
+ cval->mixer = state->mixer;
+ cval->id = unitid;
+ cval->val_type = USB_MIXER_U8;
+ cval->channels = 1;
+ cval->min = 1;
+ cval->max = num_ins;
+ cval->res = 1;
+ cval->initialized = 1;
+
+ namelist = kmalloc(sizeof(char *) * num_ins, GFP_KERNEL);
+ if (! namelist) {
+ snd_printk(KERN_ERR "cannot malloc\n");
+ kfree(cval);
+ return -ENOMEM;
+ }
+#define MAX_ITEM_NAME_LEN 64
+ for (i = 0; i < num_ins; i++) {
+ struct usb_audio_term iterm;
+ len = 0;
+ namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL);
+ if (! namelist[i]) {
+ snd_printk(KERN_ERR "cannot malloc\n");
+ while (i--)
+ kfree(namelist[i]);
+ kfree(namelist);
+ kfree(cval);
+ return -ENOMEM;
+ }
+ len = check_mapped_selector_name(state, unitid, i, namelist[i],
+ MAX_ITEM_NAME_LEN);
+ if (! len && check_input_term(state, desc[5 + i], &iterm) >= 0)
+ len = get_term_name(state, &iterm, namelist[i], MAX_ITEM_NAME_LEN, 0);
+ if (! len)
+ sprintf(namelist[i], "Input %d", i);
+ }
+
+ kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval);
+ if (! kctl) {
+ snd_printk(KERN_ERR "cannot malloc kcontrol\n");
+ kfree(namelist);
+ kfree(cval);
+ return -ENOMEM;
+ }
+ kctl->private_value = (unsigned long)namelist;
+ kctl->private_free = usb_mixer_selector_elem_free;
+
+ nameid = desc[desc[0] - 1];
+ len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name));
+ if (len)
+ ;
+ else if (nameid)
+ snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name));
+ else {
+ len = get_term_name(state, &state->oterm,
+ kctl->id.name, sizeof(kctl->id.name), 0);
+ if (! len)
+ strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
+
+ if ((state->oterm.type & 0xff00) == 0x0100)
+ strlcat(kctl->id.name, " Capture Source", sizeof(kctl->id.name));
+ else
+ strlcat(kctl->id.name, " Playback Source", sizeof(kctl->id.name));
+ }
+
+ snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",
+ cval->id, kctl->id.name, num_ins);
+ if ((err = add_control_to_empty(state, kctl)) < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * parse an audio unit recursively
+ */
+
+static int parse_audio_unit(struct mixer_build *state, int unitid)
+{
+ unsigned char *p1;
+
+ if (test_and_set_bit(unitid, state->unitbitmap))
+ return 0; /* the unit already visited */
+
+ p1 = find_audio_control_unit(state, unitid);
+ if (!p1) {
+ snd_printk(KERN_ERR "usbaudio: unit %d not found!\n", unitid);
+ return -EINVAL;
+ }
+
+ switch (p1[2]) {
+ case INPUT_TERMINAL:
+ return 0; /* NOP */
+ case MIXER_UNIT:
+ return parse_audio_mixer_unit(state, unitid, p1);
+ case SELECTOR_UNIT:
+ return parse_audio_selector_unit(state, unitid, p1);
+ case FEATURE_UNIT:
+ return parse_audio_feature_unit(state, unitid, p1);
+ case PROCESSING_UNIT:
+ return parse_audio_processing_unit(state, unitid, p1);
+ case EXTENSION_UNIT:
+ return parse_audio_extension_unit(state, unitid, p1);
+ default:
+ snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]);
+ return -EINVAL;
+ }
+}
+
+static void snd_usb_mixer_free(struct usb_mixer_interface *mixer)
+{
+ kfree(mixer->id_elems);
+ if (mixer->urb) {
+ kfree(mixer->urb->transfer_buffer);
+ usb_free_urb(mixer->urb);
+ }
+ usb_free_urb(mixer->rc_urb);
+ kfree(mixer->rc_setup_packet);
+ kfree(mixer);
+}
+
+static int snd_usb_mixer_dev_free(struct snd_device *device)
+{
+ struct usb_mixer_interface *mixer = device->device_data;
+ snd_usb_mixer_free(mixer);
+ return 0;
+}
+
+/*
+ * create mixer controls
+ *
+ * walk through all OUTPUT_TERMINAL descriptors to search for mixers
+ */
+static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
+{
+ unsigned char *desc;
+ struct mixer_build state;
+ int err;
+ const struct usbmix_ctl_map *map;
+ struct usb_host_interface *hostif;
+
+ hostif = &usb_ifnum_to_if(mixer->chip->dev, mixer->ctrlif)->altsetting[0];
+ memset(&state, 0, sizeof(state));
+ state.chip = mixer->chip;
+ state.mixer = mixer;
+ state.buffer = hostif->extra;
+ state.buflen = hostif->extralen;
+
+ /* check the mapping table */
+ for (map = usbmix_ctl_maps; map->id; map++) {
+ if (map->id == state.chip->usb_id) {
+ state.map = map->map;
+ state.selector_map = map->selector_map;
+ mixer->ignore_ctl_error = map->ignore_ctl_error;
+ break;
+ }
+ }
+
+ desc = NULL;
+ while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) {
+ if (desc[0] < 9)
+ continue; /* invalid descriptor? */
+ set_bit(desc[3], state.unitbitmap); /* mark terminal ID as visited */
+ state.oterm.id = desc[3];
+ state.oterm.type = combine_word(&desc[4]);
+ state.oterm.name = desc[8];
+ err = parse_audio_unit(&state, desc[7]);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer,
+ int unitid)
+{
+ struct usb_mixer_elem_info *info;
+
+ for (info = mixer->id_elems[unitid]; info; info = info->next_id_elem)
+ snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ info->elem_id);
+}
+
+static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer,
+ int unitid)
+{
+ if (!mixer->rc_cfg)
+ return;
+ /* unit ids specific to Extigy/Audigy 2 NX: */
+ switch (unitid) {
+ case 0: /* remote control */
+ mixer->rc_urb->dev = mixer->chip->dev;
+ usb_submit_urb(mixer->rc_urb, GFP_ATOMIC);
+ break;
+ case 4: /* digital in jack */
+ case 7: /* line in jacks */
+ case 19: /* speaker out jacks */
+ case 20: /* headphones out jack */
+ break;
+ /* live24ext: 4 = line-in jack */
+ case 3: /* hp-out jack (may actuate Mute) */
+ if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040))
+ snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id);
+ break;
+ default:
+ snd_printd(KERN_DEBUG "memory change in unknown unit %d\n", unitid);
+ break;
+ }
+}
+
+static void snd_usb_mixer_status_complete(struct urb *urb)
+{
+ struct usb_mixer_interface *mixer = urb->context;
+
+ if (urb->status == 0) {
+ u8 *buf = urb->transfer_buffer;
+ int i;
+
+ for (i = urb->actual_length; i >= 2; buf += 2, i -= 2) {
+ snd_printd(KERN_DEBUG "status interrupt: %02x %02x\n",
+ buf[0], buf[1]);
+ /* ignore any notifications not from the control interface */
+ if ((buf[0] & 0x0f) != 0)
+ continue;
+ if (!(buf[0] & 0x40))
+ snd_usb_mixer_notify_id(mixer, buf[1]);
+ else
+ snd_usb_mixer_memory_change(mixer, buf[1]);
+ }
+ }
+ if (urb->status != -ENOENT && urb->status != -ECONNRESET) {
+ urb->dev = mixer->chip->dev;
+ usb_submit_urb(urb, GFP_ATOMIC);
+ }
+}
+
+/* create the handler for the optional status interrupt endpoint */
+static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
+{
+ struct usb_host_interface *hostif;
+ struct usb_endpoint_descriptor *ep;
+ void *transfer_buffer;
+ int buffer_length;
+ unsigned int epnum;
+
+ hostif = &usb_ifnum_to_if(mixer->chip->dev, mixer->ctrlif)->altsetting[0];
+ /* we need one interrupt input endpoint */
+ if (get_iface_desc(hostif)->bNumEndpoints < 1)
+ return 0;
+ ep = get_endpoint(hostif, 0);
+ if ((ep->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_IN ||
+ (ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ return 0;
+
+ epnum = ep->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ buffer_length = le16_to_cpu(ep->wMaxPacketSize);
+ transfer_buffer = kmalloc(buffer_length, GFP_KERNEL);
+ if (!transfer_buffer)
+ return -ENOMEM;
+ mixer->urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!mixer->urb) {
+ kfree(transfer_buffer);
+ return -ENOMEM;
+ }
+ usb_fill_int_urb(mixer->urb, mixer->chip->dev,
+ usb_rcvintpipe(mixer->chip->dev, epnum),
+ transfer_buffer, buffer_length,
+ snd_usb_mixer_status_complete, mixer, ep->bInterval);
+ usb_submit_urb(mixer->urb, GFP_KERNEL);
+ return 0;
+}
+
+static void snd_usb_soundblaster_remote_complete(struct urb *urb)
+{
+ struct usb_mixer_interface *mixer = urb->context;
+ const struct rc_config *rc = mixer->rc_cfg;
+ u32 code;
+
+ if (urb->status < 0 || urb->actual_length < rc->min_packet_length)
+ return;
+
+ code = mixer->rc_buffer[rc->offset];
+ if (rc->length == 2)
+ code |= mixer->rc_buffer[rc->offset + 1] << 8;
+
+ /* the Mute button actually changes the mixer control */
+ if (code == rc->mute_code)
+ snd_usb_mixer_notify_id(mixer, rc->mute_mixer_id);
+ mixer->rc_code = code;
+ wmb();
+ wake_up(&mixer->rc_waitq);
+}
+
+static int snd_usb_sbrc_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+
+ if (test_and_set_bit(0, &mixer->rc_hwdep_open))
+ return -EBUSY;
+ return 0;
+}
+
+static int snd_usb_sbrc_hwdep_release(struct snd_hwdep *hw, struct file *file)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+
+ clear_bit(0, &mixer->rc_hwdep_open);
+ smp_mb__after_clear_bit();
+ return 0;
+}
+
+static long snd_usb_sbrc_hwdep_read(struct snd_hwdep *hw, char __user *buf,
+ long count, loff_t *offset)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+ int err;
+ u32 rc_code;
+
+ if (count != 1 && count != 4)
+ return -EINVAL;
+ err = wait_event_interruptible(mixer->rc_waitq,
+ (rc_code = xchg(&mixer->rc_code, 0)) != 0);
+ if (err == 0) {
+ if (count == 1)
+ err = put_user(rc_code, buf);
+ else
+ err = put_user(rc_code, (u32 __user *)buf);
+ }
+ return err < 0 ? err : count;
+}
+
+static unsigned int snd_usb_sbrc_hwdep_poll(struct snd_hwdep *hw, struct file *file,
+ poll_table *wait)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+
+ poll_wait(file, &mixer->rc_waitq, wait);
+ return mixer->rc_code ? POLLIN | POLLRDNORM : 0;
+}
+
+static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer)
+{
+ struct snd_hwdep *hwdep;
+ int err, len, i;
+
+ for (i = 0; i < ARRAY_SIZE(rc_configs); ++i)
+ if (rc_configs[i].usb_id == mixer->chip->usb_id)
+ break;
+ if (i >= ARRAY_SIZE(rc_configs))
+ return 0;
+ mixer->rc_cfg = &rc_configs[i];
+
+ len = mixer->rc_cfg->packet_length;
+
+ init_waitqueue_head(&mixer->rc_waitq);
+ err = snd_hwdep_new(mixer->chip->card, "SB remote control", 0, &hwdep);
+ if (err < 0)
+ return err;
+ snprintf(hwdep->name, sizeof(hwdep->name),
+ "%s remote control", mixer->chip->card->shortname);
+ hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC;
+ hwdep->private_data = mixer;
+ hwdep->ops.read = snd_usb_sbrc_hwdep_read;
+ hwdep->ops.open = snd_usb_sbrc_hwdep_open;
+ hwdep->ops.release = snd_usb_sbrc_hwdep_release;
+ hwdep->ops.poll = snd_usb_sbrc_hwdep_poll;
+
+ mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!mixer->rc_urb)
+ return -ENOMEM;
+ mixer->rc_setup_packet = kmalloc(sizeof(*mixer->rc_setup_packet), GFP_KERNEL);
+ if (!mixer->rc_setup_packet) {
+ usb_free_urb(mixer->rc_urb);
+ mixer->rc_urb = NULL;
+ return -ENOMEM;
+ }
+ mixer->rc_setup_packet->bRequestType =
+ USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE;
+ mixer->rc_setup_packet->bRequest = GET_MEM;
+ mixer->rc_setup_packet->wValue = cpu_to_le16(0);
+ mixer->rc_setup_packet->wIndex = cpu_to_le16(0);
+ mixer->rc_setup_packet->wLength = cpu_to_le16(len);
+ usb_fill_control_urb(mixer->rc_urb, mixer->chip->dev,
+ usb_rcvctrlpipe(mixer->chip->dev, 0),
+ (u8*)mixer->rc_setup_packet, mixer->rc_buffer, len,
+ snd_usb_soundblaster_remote_complete, mixer);
+ return 0;
+}
+
+#define snd_audigy2nx_led_info snd_ctl_boolean_mono_info
+
+static int snd_audigy2nx_led_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = mixer->audigy2nx_leds[index];
+ return 0;
+}
+
+static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ int value = ucontrol->value.integer.value[0];
+ int err, changed;
+
+ if (value > 1)
+ return -EINVAL;
+ changed = value != mixer->audigy2nx_leds[index];
+ err = snd_usb_ctl_msg(mixer->chip->dev,
+ usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
+ USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+ value, index + 2, NULL, 0, 100);
+ if (err < 0)
+ return err;
+ mixer->audigy2nx_leds[index] = value;
+ return changed;
+}
+
+static struct snd_kcontrol_new snd_audigy2nx_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "CMSS LED Switch",
+ .info = snd_audigy2nx_led_info,
+ .get = snd_audigy2nx_led_get,
+ .put = snd_audigy2nx_led_put,
+ .private_value = 0,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Power LED Switch",
+ .info = snd_audigy2nx_led_info,
+ .get = snd_audigy2nx_led_get,
+ .put = snd_audigy2nx_led_put,
+ .private_value = 1,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Dolby Digital LED Switch",
+ .info = snd_audigy2nx_led_info,
+ .get = snd_audigy2nx_led_get,
+ .put = snd_audigy2nx_led_put,
+ .private_value = 2,
+ },
+};
+
+static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer)
+{
+ int i, err;
+
+ for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) {
+ if (i > 1 && /* Live24ext has 2 LEDs only */
+ mixer->chip->usb_id == USB_ID(0x041e, 0x3040))
+ break;
+ err = snd_ctl_add(mixer->chip->card,
+ snd_ctl_new1(&snd_audigy2nx_controls[i], mixer));
+ if (err < 0)
+ return err;
+ }
+ mixer->audigy2nx_leds[1] = 1; /* Power LED is on by default */
+ return 0;
+}
+
+static void snd_audigy2nx_proc_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ static const struct sb_jack {
+ int unitid;
+ const char *name;
+ } jacks_audigy2nx[] = {
+ {4, "dig in "},
+ {7, "line in"},
+ {19, "spk out"},
+ {20, "hph out"},
+ {-1, NULL}
+ }, jacks_live24ext[] = {
+ {4, "line in"}, /* &1=Line, &2=Mic*/
+ {3, "hph out"}, /* headphones */
+ {0, "RC "}, /* last command, 6 bytes see rc_config above */
+ {-1, NULL}
+ };
+ const struct sb_jack *jacks;
+ struct usb_mixer_interface *mixer = entry->private_data;
+ int i, err;
+ u8 buf[3];
+
+ snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname);
+ if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020))
+ jacks = jacks_audigy2nx;
+ else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040))
+ jacks = jacks_live24ext;
+ else
+ return;
+
+ for (i = 0; jacks[i].name; ++i) {
+ snd_iprintf(buffer, "%s: ", jacks[i].name);
+ err = snd_usb_ctl_msg(mixer->chip->dev,
+ usb_rcvctrlpipe(mixer->chip->dev, 0),
+ GET_MEM, USB_DIR_IN | USB_TYPE_CLASS |
+ USB_RECIP_INTERFACE, 0,
+ jacks[i].unitid << 8, buf, 3, 100);
+ if (err == 3 && (buf[0] == 3 || buf[0] == 6))
+ snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]);
+ else
+ snd_iprintf(buffer, "?\n");
+ }
+}
+
+int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
+ int ignore_error)
+{
+ static struct snd_device_ops dev_ops = {
+ .dev_free = snd_usb_mixer_dev_free
+ };
+ struct usb_mixer_interface *mixer;
+ int err;
+
+ strcpy(chip->card->mixername, "USB Mixer");
+
+ mixer = kzalloc(sizeof(*mixer), GFP_KERNEL);
+ if (!mixer)
+ return -ENOMEM;
+ mixer->chip = chip;
+ mixer->ctrlif = ctrlif;
+ mixer->ignore_ctl_error = ignore_error;
+ mixer->id_elems = kcalloc(256, sizeof(*mixer->id_elems), GFP_KERNEL);
+ if (!mixer->id_elems) {
+ kfree(mixer);
+ return -ENOMEM;
+ }
+
+ if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
+ (err = snd_usb_mixer_status_create(mixer)) < 0)
+ goto _error;
+
+ if ((err = snd_usb_soundblaster_remote_init(mixer)) < 0)
+ goto _error;
+
+ if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) ||
+ mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) {
+ struct snd_info_entry *entry;
+
+ if ((err = snd_audigy2nx_controls_create(mixer)) < 0)
+ goto _error;
+ if (!snd_card_proc_new(chip->card, "audigy2nx", &entry))
+ snd_info_set_text_ops(entry, mixer,
+ snd_audigy2nx_proc_read);
+ }
+
+ err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops);
+ if (err < 0)
+ goto _error;
+ list_add(&mixer->list, &chip->mixer_list);
+ return 0;
+
+_error:
+ snd_usb_mixer_free(mixer);
+ return err;
+}
+
+void snd_usb_mixer_disconnect(struct list_head *p)
+{
+ struct usb_mixer_interface *mixer;
+
+ mixer = list_entry(p, struct usb_mixer_interface, list);
+ usb_kill_urb(mixer->urb);
+ usb_kill_urb(mixer->rc_urb);
+}
diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c
new file mode 100644
index 0000000..d755be0
--- /dev/null
+++ b/sound/usb/usbmixer_maps.c
@@ -0,0 +1,316 @@
+/*
+ * Additional mixer mapping
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+
+struct usbmix_name_map {
+ int id;
+ const char *name;
+ int control;
+};
+
+struct usbmix_selector_map {
+ int id;
+ int count;
+ const char **names;
+};
+
+struct usbmix_ctl_map {
+ u32 id;
+ const struct usbmix_name_map *map;
+ const struct usbmix_selector_map *selector_map;
+ int ignore_ctl_error;
+};
+
+/*
+ * USB control mappers for SB Exitigy
+ */
+
+/*
+ * Topology of SB Extigy (see on the wide screen :)
+
+USB_IN[1] --->FU[2]------------------------------+->MU[16]-->PU[17]-+->FU[18]--+->EU[27]--+->EU[21]-->FU[22]--+->FU[23] > Dig_OUT[24]
+ ^ | | | |
+USB_IN[3] -+->SU[5]-->FU[6]--+->MU[14] ->PU[15]->+ | | | +->FU[25] > Dig_OUT[26]
+ ^ ^ | | | |
+Dig_IN[4] -+ | | | | +->FU[28]---------------------> Spk_OUT[19]
+ | | | |
+Lin-IN[7] -+-->FU[8]---------+ | | +----------------------------------------> Hph_OUT[20]
+ | | |
+Mic-IN[9] --+->FU[10]----------------------------+ |
+ || |
+ || +----------------------------------------------------+
+ VV V
+ ++--+->SU[11]-->FU[12] --------------------------------------------------------------------------------------> USB_OUT[13]
+*/
+
+static struct usbmix_name_map extigy_map[] = {
+ /* 1: IT pcm */
+ { 2, "PCM Playback" }, /* FU */
+ /* 3: IT pcm */
+ /* 4: IT digital in */
+ { 5, NULL }, /* DISABLED: this seems to be bogus on some firmware */
+ { 6, "Digital In" }, /* FU */
+ /* 7: IT line */
+ { 8, "Line Playback" }, /* FU */
+ /* 9: IT mic */
+ { 10, "Mic Playback" }, /* FU */
+ { 11, "Capture Input Source" }, /* SU */
+ { 12, "Capture" }, /* FU */
+ /* 13: OT pcm capture */
+ /* 14: MU (w/o controls) */
+ /* 15: PU (3D enh) */
+ /* 16: MU (w/o controls) */
+ { 17, NULL, 1 }, /* DISABLED: PU-switch (any effect?) */
+ { 17, "Channel Routing", 2 }, /* PU: mode select */
+ { 18, "Tone Control - Bass", USB_FEATURE_BASS }, /* FU */
+ { 18, "Tone Control - Treble", USB_FEATURE_TREBLE }, /* FU */
+ { 18, "Master Playback" }, /* FU; others */
+ /* 19: OT speaker */
+ /* 20: OT headphone */
+ { 21, NULL }, /* DISABLED: EU (for what?) */
+ { 22, "Digital Out Playback" }, /* FU */
+ { 23, "Digital Out1 Playback" }, /* FU */ /* FIXME: corresponds to 24 */
+ /* 24: OT digital out */
+ { 25, "IEC958 Optical Playback" }, /* FU */
+ { 26, "IEC958 Optical Playback" }, /* OT */
+ { 27, NULL }, /* DISABLED: EU (for what?) */
+ /* 28: FU speaker (mute) */
+ { 29, NULL }, /* Digital Input Playback Source? */
+ { 0 } /* terminator */
+};
+
+/* Sound Blaster MP3+ controls mapping
+ * The default mixer channels have totally misleading names,
+ * e.g. no Master and fake PCM volume
+ * Pavel Mihaylov <bin@bash.info>
+ */
+static struct usbmix_name_map mp3plus_map[] = {
+ /* 1: IT pcm */
+ /* 2: IT mic */
+ /* 3: IT line */
+ /* 4: IT digital in */
+ /* 5: OT digital out */
+ /* 6: OT speaker */
+ /* 7: OT pcm capture */
+ { 8, "Capture Input Source" }, /* FU, default PCM Capture Source */
+ /* (Mic, Input 1 = Line input, Input 2 = Optical input) */
+ { 9, "Master Playback" }, /* FU, default Speaker 1 */
+ /* { 10, "Mic Capture", 1 }, */ /* FU, Mic Capture */
+ /* { 10, "Mic Capture", 2 }, */ /* FU, Mic Capture */
+ { 10, "Mic Boost", 7 }, /* FU, default Auto Gain Input */
+ { 11, "Line Capture" }, /* FU, default PCM Capture */
+ { 12, "Digital In Playback" }, /* FU, default PCM 1 */
+ /* { 13, "Mic Playback" }, */ /* FU, default Mic Playback */
+ { 14, "Line Playback" }, /* FU, default Speaker */
+ /* 15: MU */
+ { 0 } /* terminator */
+};
+
+/* Topology of SB Audigy 2 NX
+
+ +----------------------------->EU[27]--+
+ | v
+ | +----------------------------------->SU[29]---->FU[22]-->Dig_OUT[24]
+ | | ^
+USB_IN[1]-+------------+ +->EU[17]->+->FU[11]-+
+ | v | v |
+Dig_IN[4]---+->FU[6]-->MU[16]->FU[18]-+->EU[21]->SU[31]----->FU[30]->Hph_OUT[20]
+ | ^ | |
+Lin_IN[7]-+--->FU[8]---+ +->EU[23]->FU[28]------------->Spk_OUT[19]
+ | | v
+ +--->FU[12]------------------------------------->SU[14]--->USB_OUT[15]
+ | ^
+ +->FU[13]--------------------------------------+
+*/
+static struct usbmix_name_map audigy2nx_map[] = {
+ /* 1: IT pcm playback */
+ /* 4: IT digital in */
+ { 6, "Digital In Playback" }, /* FU */
+ /* 7: IT line in */
+ { 8, "Line Playback" }, /* FU */
+ { 11, "What-U-Hear Capture" }, /* FU */
+ { 12, "Line Capture" }, /* FU */
+ { 13, "Digital In Capture" }, /* FU */
+ { 14, "Capture Source" }, /* SU */
+ /* 15: OT pcm capture */
+ /* 16: MU w/o controls */
+ { 17, NULL }, /* DISABLED: EU (for what?) */
+ { 18, "Master Playback" }, /* FU */
+ /* 19: OT speaker */
+ /* 20: OT headphone */
+ { 21, NULL }, /* DISABLED: EU (for what?) */
+ { 22, "Digital Out Playback" }, /* FU */
+ { 23, NULL }, /* DISABLED: EU (for what?) */
+ /* 24: OT digital out */
+ { 27, NULL }, /* DISABLED: EU (for what?) */
+ { 28, "Speaker Playback" }, /* FU */
+ { 29, "Digital Out Source" }, /* SU */
+ { 30, "Headphone Playback" }, /* FU */
+ { 31, "Headphone Source" }, /* SU */
+ { 0 } /* terminator */
+};
+
+static struct usbmix_selector_map audigy2nx_selectors[] = {
+ {
+ .id = 14, /* Capture Source */
+ .count = 3,
+ .names = (const char*[]) {"Line", "Digital In", "What-U-Hear"}
+ },
+ {
+ .id = 29, /* Digital Out Source */
+ .count = 3,
+ .names = (const char*[]) {"Front", "PCM", "Digital In"}
+ },
+ {
+ .id = 31, /* Headphone Source */
+ .count = 2,
+ .names = (const char*[]) {"Front", "Side"}
+ },
+ { 0 } /* terminator */
+};
+
+/* Creative SoundBlaster Live! 24-bit External */
+static struct usbmix_name_map live24ext_map[] = {
+ /* 2: PCM Playback Volume */
+ { 5, "Mic Capture" }, /* FU, default PCM Capture Volume */
+ { 0 } /* terminator */
+};
+
+/* LineX FM Transmitter entry - needed to bypass controls bug */
+static struct usbmix_name_map linex_map[] = {
+ /* 1: IT pcm */
+ /* 2: OT Speaker */
+ { 3, "Master" }, /* FU: master volume - left / right / mute */
+ { 0 } /* terminator */
+};
+
+static struct usbmix_name_map maya44_map[] = {
+ /* 1: IT line */
+ { 2, "Line Playback" }, /* FU */
+ /* 3: IT line */
+ { 4, "Line Playback" }, /* FU */
+ /* 5: IT pcm playback */
+ /* 6: MU */
+ { 7, "Master Playback" }, /* FU */
+ /* 8: OT speaker */
+ /* 9: IT line */
+ { 10, "Line Capture" }, /* FU */
+ /* 11: MU */
+ /* 12: OT pcm capture */
+ { }
+};
+
+/* Section "justlink_map" below added by James Courtier-Dutton <James@superbug.demon.co.uk>
+ * sourced from Maplin Electronics (http://www.maplin.co.uk), part number A56AK
+ * Part has 2 connectors that act as a single output. (TOSLINK Optical for digital out, and 3.5mm Jack for Analogue out.)
+ * The USB Mixer publishes a Microphone and extra Volume controls for it, but none exist on the device,
+ * so this map removes all unwanted sliders from alsamixer
+ */
+
+static struct usbmix_name_map justlink_map[] = {
+ /* 1: IT pcm playback */
+ /* 2: Not present */
+ { 3, NULL}, /* IT mic (No mic input on device) */
+ /* 4: Not present */
+ /* 5: OT speacker */
+ /* 6: OT pcm capture */
+ { 7, "Master Playback" }, /* Mute/volume for speaker */
+ { 8, NULL }, /* Capture Switch (No capture inputs on device) */
+ { 9, NULL }, /* Capture Mute/volume (No capture inputs on device */
+ /* 0xa: Not present */
+ /* 0xb: MU (w/o controls) */
+ { 0xc, NULL }, /* Mic feedback Mute/volume (No capture inputs on device) */
+ { 0 } /* terminator */
+};
+
+/* TerraTec Aureon 5.1 MkII USB */
+static struct usbmix_name_map aureon_51_2_map[] = {
+ /* 1: IT USB */
+ /* 2: IT Mic */
+ /* 3: IT Line */
+ /* 4: IT SPDIF */
+ /* 5: OT SPDIF */
+ /* 6: OT Speaker */
+ /* 7: OT USB */
+ { 8, "Capture Source" }, /* SU */
+ { 9, "Master Playback" }, /* FU */
+ { 10, "Mic Capture" }, /* FU */
+ { 11, "Line Capture" }, /* FU */
+ { 12, "IEC958 In Capture" }, /* FU */
+ { 13, "Mic Playback" }, /* FU */
+ { 14, "Line Playback" }, /* FU */
+ /* 15: MU */
+ {} /* terminator */
+};
+
+/*
+ * Control map entries
+ */
+
+static struct usbmix_ctl_map usbmix_ctl_maps[] = {
+ {
+ .id = USB_ID(0x041e, 0x3000),
+ .map = extigy_map,
+ .ignore_ctl_error = 1,
+ },
+ {
+ .id = USB_ID(0x041e, 0x3010),
+ .map = mp3plus_map,
+ },
+ {
+ .id = USB_ID(0x041e, 0x3020),
+ .map = audigy2nx_map,
+ .selector_map = audigy2nx_selectors,
+ },
+ {
+ .id = USB_ID(0x041e, 0x3040),
+ .map = live24ext_map,
+ },
+ {
+ /* Hercules DJ Console (Windows Edition) */
+ .id = USB_ID(0x06f8, 0xb000),
+ .ignore_ctl_error = 1,
+ },
+ {
+ /* Hercules DJ Console (Macintosh Edition) */
+ .id = USB_ID(0x06f8, 0xd002),
+ .ignore_ctl_error = 1,
+ },
+ {
+ .id = USB_ID(0x08bb, 0x2702),
+ .map = linex_map,
+ .ignore_ctl_error = 1,
+ },
+ {
+ .id = USB_ID(0x0a92, 0x0091),
+ .map = maya44_map,
+ },
+ {
+ .id = USB_ID(0x0c45, 0x1158),
+ .map = justlink_map,
+ },
+ {
+ .id = USB_ID(0x0ccd, 0x0028),
+ .map = aureon_51_2_map,
+ },
+ { 0 } /* terminator */
+};
+
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
new file mode 100644
index 0000000..9211575
--- /dev/null
+++ b/sound/usb/usbquirks.h
@@ -0,0 +1,2039 @@
+/*
+ * ALSA USB Audio Driver
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>,
+ * Clemens Ladisch <clemens@ladisch.de>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/*
+ * The contents of this file are part of the driver's id_table.
+ *
+ * In a perfect world, this file would be empty.
+ */
+
+/*
+ * Use this for devices where other interfaces are standard compliant,
+ * to prevent the quirk being applied to those interfaces. (To work with
+ * hotplugging, bDeviceClass must be set to USB_CLASS_PER_INTERFACE.)
+ */
+#define USB_DEVICE_VENDOR_SPEC(vend, prod) \
+ .match_flags = USB_DEVICE_ID_MATCH_VENDOR | \
+ USB_DEVICE_ID_MATCH_PRODUCT | \
+ USB_DEVICE_ID_MATCH_INT_CLASS, \
+ .idVendor = vend, \
+ .idProduct = prod, \
+ .bInterfaceClass = USB_CLASS_VENDOR_SPEC
+
+/* Creative/E-Mu devices */
+{
+ USB_DEVICE(0x041e, 0x3010),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Creative Labs",
+ .product_name = "Sound Blaster MP3+",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+{
+ /* E-Mu 0202 USB */
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x041e,
+ .idProduct = 0x3f02,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
+{
+ /* E-Mu 0404 USB */
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x041e,
+ .idProduct = 0x3f04,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
+{
+ /* E-Mu Tracker Pre */
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x041e,
+ .idProduct = 0x3f0a,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+},
+
+/*
+ * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
+ * class matches do not take effect without an explicit ID match.
+ */
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x046d,
+ .idProduct = 0x0850,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+},
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x046d,
+ .idProduct = 0x08ae,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+},
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x046d,
+ .idProduct = 0x08c6,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+},
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x046d,
+ .idProduct = 0x08f0,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+},
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x046d,
+ .idProduct = 0x08f5,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+},
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x046d,
+ .idProduct = 0x08f6,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL
+},
+
+/*
+ * Yamaha devices
+ */
+
+#define YAMAHA_DEVICE(id, name) { \
+ USB_DEVICE(0x0499, id), \
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
+ .vendor_name = "Yamaha", \
+ .product_name = name, \
+ .ifnum = QUIRK_ANY_INTERFACE, \
+ .type = QUIRK_MIDI_YAMAHA \
+ } \
+}
+#define YAMAHA_INTERFACE(id, intf, name) { \
+ USB_DEVICE_VENDOR_SPEC(0x0499, id), \
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
+ .vendor_name = "Yamaha", \
+ .product_name = name, \
+ .ifnum = intf, \
+ .type = QUIRK_MIDI_YAMAHA \
+ } \
+}
+YAMAHA_DEVICE(0x1000, "UX256"),
+YAMAHA_DEVICE(0x1001, "MU1000"),
+YAMAHA_DEVICE(0x1002, "MU2000"),
+YAMAHA_DEVICE(0x1003, "MU500"),
+YAMAHA_INTERFACE(0x1004, 3, "UW500"),
+YAMAHA_DEVICE(0x1005, "MOTIF6"),
+YAMAHA_DEVICE(0x1006, "MOTIF7"),
+YAMAHA_DEVICE(0x1007, "MOTIF8"),
+YAMAHA_DEVICE(0x1008, "UX96"),
+YAMAHA_DEVICE(0x1009, "UX16"),
+YAMAHA_INTERFACE(0x100a, 3, "EOS BX"),
+YAMAHA_DEVICE(0x100c, "UC-MX"),
+YAMAHA_DEVICE(0x100d, "UC-KX"),
+YAMAHA_DEVICE(0x100e, "S08"),
+YAMAHA_DEVICE(0x100f, "CLP-150"),
+YAMAHA_DEVICE(0x1010, "CLP-170"),
+YAMAHA_DEVICE(0x1011, "P-250"),
+YAMAHA_DEVICE(0x1012, "TYROS"),
+YAMAHA_DEVICE(0x1013, "PF-500"),
+YAMAHA_DEVICE(0x1014, "S90"),
+YAMAHA_DEVICE(0x1015, "MOTIF-R"),
+YAMAHA_DEVICE(0x1016, "MDP-5"),
+YAMAHA_DEVICE(0x1017, "CVP-204"),
+YAMAHA_DEVICE(0x1018, "CVP-206"),
+YAMAHA_DEVICE(0x1019, "CVP-208"),
+YAMAHA_DEVICE(0x101a, "CVP-210"),
+YAMAHA_DEVICE(0x101b, "PSR-1100"),
+YAMAHA_DEVICE(0x101c, "PSR-2100"),
+YAMAHA_DEVICE(0x101d, "CLP-175"),
+YAMAHA_DEVICE(0x101e, "PSR-K1"),
+YAMAHA_DEVICE(0x101f, "EZ-J24"),
+YAMAHA_DEVICE(0x1020, "EZ-250i"),
+YAMAHA_DEVICE(0x1021, "MOTIF ES 6"),
+YAMAHA_DEVICE(0x1022, "MOTIF ES 7"),
+YAMAHA_DEVICE(0x1023, "MOTIF ES 8"),
+YAMAHA_DEVICE(0x1024, "CVP-301"),
+YAMAHA_DEVICE(0x1025, "CVP-303"),
+YAMAHA_DEVICE(0x1026, "CVP-305"),
+YAMAHA_DEVICE(0x1027, "CVP-307"),
+YAMAHA_DEVICE(0x1028, "CVP-309"),
+YAMAHA_DEVICE(0x1029, "CVP-309GP"),
+YAMAHA_DEVICE(0x102a, "PSR-1500"),
+YAMAHA_DEVICE(0x102b, "PSR-3000"),
+YAMAHA_DEVICE(0x102e, "ELS-01/01C"),
+YAMAHA_DEVICE(0x1030, "PSR-295/293"),
+YAMAHA_DEVICE(0x1031, "DGX-205/203"),
+YAMAHA_DEVICE(0x1032, "DGX-305"),
+YAMAHA_DEVICE(0x1033, "DGX-505"),
+YAMAHA_DEVICE(0x1034, NULL),
+YAMAHA_DEVICE(0x1035, NULL),
+YAMAHA_DEVICE(0x1036, NULL),
+YAMAHA_DEVICE(0x1037, NULL),
+YAMAHA_DEVICE(0x1038, NULL),
+YAMAHA_DEVICE(0x1039, NULL),
+YAMAHA_DEVICE(0x103a, NULL),
+YAMAHA_DEVICE(0x103b, NULL),
+YAMAHA_DEVICE(0x103c, NULL),
+YAMAHA_DEVICE(0x103d, NULL),
+YAMAHA_DEVICE(0x103e, NULL),
+YAMAHA_DEVICE(0x103f, NULL),
+YAMAHA_DEVICE(0x1040, NULL),
+YAMAHA_DEVICE(0x1041, NULL),
+YAMAHA_DEVICE(0x1042, NULL),
+YAMAHA_DEVICE(0x1043, NULL),
+YAMAHA_DEVICE(0x1044, NULL),
+YAMAHA_DEVICE(0x1045, NULL),
+YAMAHA_INTERFACE(0x104e, 0, NULL),
+YAMAHA_DEVICE(0x104f, NULL),
+YAMAHA_DEVICE(0x1050, NULL),
+YAMAHA_DEVICE(0x1051, NULL),
+YAMAHA_DEVICE(0x1052, NULL),
+YAMAHA_DEVICE(0x2000, "DGP-7"),
+YAMAHA_DEVICE(0x2001, "DGP-5"),
+YAMAHA_DEVICE(0x2002, NULL),
+YAMAHA_DEVICE(0x5000, "CS1D"),
+YAMAHA_DEVICE(0x5001, "DSP1D"),
+YAMAHA_DEVICE(0x5002, "DME32"),
+YAMAHA_DEVICE(0x5003, "DM2000"),
+YAMAHA_DEVICE(0x5004, "02R96"),
+YAMAHA_DEVICE(0x5005, "ACU16-C"),
+YAMAHA_DEVICE(0x5006, "NHB32-C"),
+YAMAHA_DEVICE(0x5007, "DM1000"),
+YAMAHA_DEVICE(0x5008, "01V96"),
+YAMAHA_DEVICE(0x5009, "SPX2000"),
+YAMAHA_DEVICE(0x500a, "PM5D"),
+YAMAHA_DEVICE(0x500b, "DME64N"),
+YAMAHA_DEVICE(0x500c, "DME24N"),
+YAMAHA_DEVICE(0x500d, NULL),
+YAMAHA_DEVICE(0x500e, NULL),
+YAMAHA_DEVICE(0x500f, NULL),
+YAMAHA_DEVICE(0x7000, "DTX"),
+YAMAHA_DEVICE(0x7010, "UB99"),
+#undef YAMAHA_DEVICE
+#undef YAMAHA_INTERFACE
+
+/*
+ * Roland/RolandED/Edirol/BOSS devices
+ */
+{
+ USB_DEVICE(0x0582, 0x0000),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "UA-100",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = & (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S16_LE,
+ .channels = 4,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x01,
+ .ep_attr = 0x09,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ }
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = & (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S16_LE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = EP_CS_ATTR_FILL_MAX,
+ .endpoint = 0x81,
+ .ep_attr = 0x05,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ }
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0007,
+ .in_cables = 0x0007
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0002),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UM-4",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x000f,
+ .in_cables = 0x000f
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0003),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "SC-8850",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x003f,
+ .in_cables = 0x003f
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0004),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "U-8",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0005,
+ .in_cables = 0x0005
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* Has ID 0x0099 when not in "Advanced Driver" mode.
+ * The UM-2EX has only one input, but we cannot detect this. */
+ USB_DEVICE(0x0582, 0x0005),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UM-2",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0007),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "SC-8820",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0013,
+ .in_cables = 0x0013
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0008),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "PC-300",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x009d when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0009),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UM-1",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x000b),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "SK-500",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0013,
+ .in_cables = 0x0013
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* thanks to Emiliano Grilli <emillo@libero.it>
+ * for helping researching this data */
+ USB_DEVICE(0x0582, 0x000c),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "SC-D70",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = & (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S24_3LE,
+ .channels = 2,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x01,
+ .ep_attr = 0x01,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ }
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = & (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S24_3LE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x81,
+ .ep_attr = 0x01,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ }
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0007,
+ .in_cables = 0x0007
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{ /*
+ * This quirk is for the "Advanced Driver" mode of the Edirol UA-5.
+ * If the advanced mode switch at the back of the unit is off, the
+ * UA-5 has ID 0x0582/0x0011 and is standard compliant (no quirks),
+ * but offers only 16-bit PCM.
+ * In advanced mode, the UA-5 will output S24_3LE samples (two
+ * channels) at the rate indicated on the front switch, including
+ * the 96kHz sample rate.
+ */
+ USB_DEVICE(0x0582, 0x0010),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-5",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x0013 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0012),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "XV-5050",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* has ID 0x0015 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0014),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UM-880",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x01ff,
+ .in_cables = 0x01ff
+ }
+ }
+},
+{
+ /* has ID 0x0017 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0016),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "SD-90",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x000f,
+ .in_cables = 0x000f
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x001c when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x001b),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "MMP-2",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x001e when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x001d),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "V-SYNTH",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* has ID 0x0024 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0023),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UM-550",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x003f,
+ .in_cables = 0x003f
+ }
+ }
+},
+{
+ /*
+ * This quirk is for the "Advanced Driver" mode. If off, the UA-20
+ * has ID 0x0026 and is standard compliant, but has only 16-bit PCM
+ * and no MIDI.
+ */
+ USB_DEVICE(0x0582, 0x0025),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-20",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = & (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S24_3LE,
+ .channels = 2,
+ .iface = 1,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x01,
+ .ep_attr = 0x01,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ }
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = & (const struct audioformat) {
+ .format = SNDRV_PCM_FORMAT_S24_3LE,
+ .channels = 2,
+ .iface = 2,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x82,
+ .ep_attr = 0x01,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ }
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x0028 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0027),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "SD-20",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0007
+ }
+ }
+},
+{
+ /* has ID 0x002a when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0029),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "SD-80",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x000f,
+ .in_cables = 0x000f
+ }
+ }
+},
+{ /*
+ * This quirk is for the "Advanced" modes of the Edirol UA-700.
+ * If the sample format switch is not in an advanced setting, the
+ * UA-700 has ID 0x0582/0x002c and is standard compliant (no quirks),
+ * but offers only 16-bit PCM and no MIDI.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x002b),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-700",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x002e when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x002d),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "XV-2020",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* has ID 0x0030 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x002f),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "VariOS",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0007,
+ .in_cables = 0x0007
+ }
+ }
+},
+{
+ /* has ID 0x0034 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0033),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "PCR",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0007
+ }
+ }
+},
+ /* TODO: add Roland M-1000 support */
+{
+ /*
+ * Has ID 0x0038 when not in "Advanced Driver" mode;
+ * later revisions use IDs 0x0054 and 0x00a2.
+ */
+ USB_DEVICE(0x0582, 0x0037),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "Digital Piano",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /*
+ * This quirk is for the "Advanced Driver" mode. If off, the GS-10
+ * has ID 0x003c and is standard compliant, but has only 16-bit PCM
+ * and no MIDI.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x003b),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "BOSS",
+ .product_name = "GS-10",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = & (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x0041 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0040),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "GI-20",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* has ID 0x0043 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0042),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "RS-70",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0044),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "UA-1000",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_EDIROL_UA1000
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_EDIROL_UA1000
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x0049 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0047),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "EDIROL", */
+ /* .product_name = "UR-80", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ /* in the 96 kHz modes, only interface 1 is there */
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x004a when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0048),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "EDIROL", */
+ /* .product_name = "UR-80", */
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0007
+ }
+ }
+},
+ /* TODO: add Edirol M-100FX support */
+{
+ /* has ID 0x004e when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x004c),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "PCR-A",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x004f when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x004d),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "PCR-A",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0007
+ }
+ }
+},
+{
+ /*
+ * This quirk is for the "Advanced Driver" mode. If off, the UA-3FX
+ * is standard compliant, but has only 16-bit PCM.
+ */
+ USB_DEVICE(0x0582, 0x0050),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-3FX",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0052),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UM-1SX",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x0060),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "EXR Series",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+{
+ /* has ID 0x0067 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0065),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "PCR-1",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0003
+ }
+ }
+},
+{
+ /* has ID 0x006b when not in "Advanced Driver" mode */
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x006a),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "SP-606",
+ .ifnum = 3,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* has ID 0x006e when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x006d),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "FANTOM-X",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{ /*
+ * This quirk is for the "Advanced" modes of the Edirol UA-25.
+ * If the switch is not in an advanced setting, the UA-25 has
+ * ID 0x0582/0x0073 and is standard compliant (no quirks), but
+ * offers only 16-bit PCM at 44.1 kHz and no MIDI.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0074),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-25",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x0076 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0075),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "BOSS",
+ .product_name = "DR-880",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* has ID 0x007b when not in "Advanced Driver" mode */
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x007a),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ /* "RD" or "RD-700SX"? */
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ }
+},
+/* Roland UA-101 in High-Speed Mode only */
+{
+ USB_DEVICE(0x0582, 0x007d),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "UA-101",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_EDIROL_UA101
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_EDIROL_UA101
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* has ID 0x0081 when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x0080),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "G-70",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+ /* TODO: add Roland V-SYNTH XT support */
+ /* TODO: add BOSS GT-PRO support */
+{
+ /* has ID 0x008c when not in "Advanced Driver" mode */
+ USB_DEVICE(0x0582, 0x008b),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "PC-50",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+ /* TODO: add Edirol PC-80 support */
+{
+ USB_DEVICE(0x0582, 0x0096),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-1EX",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0582, 0x009a),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UM-3EX",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x000f,
+ .in_cables = 0x000f
+ }
+ }
+},
+{
+ /*
+ * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX
+ * is standard compliant, but has only 16-bit PCM and no MIDI.
+ */
+ USB_DEVICE(0x0582, 0x00a3),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-4FX",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+ /* TODO: add Edirol MD-P1 support */
+{
+ USB_DEVICE(0x582, 0x00a6),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "Juno-G",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ /* Roland SH-201 */
+ USB_DEVICE(0x0582, 0x00ad),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "SH-201",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* Roland SonicCell */
+ USB_DEVICE(0x0582, 0x00c2),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Roland",
+ .product_name = "SonicCell",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* BOSS GT-10 */
+ USB_DEVICE(0x0582, 0x00da),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /* Advanced modes of the Edirol UA-25EX.
+ * For the standard mode, UA-25EX has ID 0582:00e7, which
+ * offers only 16-bit PCM at 44.1 kHz and no MIDI.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "EDIROL",
+ .product_name = "UA-25EX",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_EDIROL_UAXX
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+
+/* Guillemot devices */
+{
+ /*
+ * This is for the "Windows Edition" where the external MIDI ports are
+ * the only MIDI ports; the control data is reported through HID
+ * interfaces. The "Macintosh Edition" has ID 0xd002 and uses standard
+ * compliant USB MIDI ports for external MIDI and controls.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x06f8, 0xb000),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Hercules",
+ .product_name = "DJ Console (WE)",
+ .ifnum = 4,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+
+/* Midiman/M-Audio devices */
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x1002),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "MidiSport 2x2",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x1011),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "MidiSport 1x1",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x1015),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "Keystation",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x1021),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "MidiSport 4x4",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x000f,
+ .in_cables = 0x000f
+ }
+ }
+},
+{
+ /*
+ * For hardware revision 1.05; in the later revisions (1.10 and
+ * 1.21), 0x1031 is the ID for the device without firmware.
+ * Thanks to Olaf Giesbrecht <Olaf_Giesbrecht@yahoo.de>
+ */
+ USB_DEVICE_VER(0x0763, 0x1031, 0x0100, 0x0109),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "MidiSport 8x8",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x01ff,
+ .in_cables = 0x01ff
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x1033),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "MidiSport 8x8",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x01ff,
+ .in_cables = 0x01ff
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x1041),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "MidiSport 2x4",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x000f,
+ .in_cables = 0x0003
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2001),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "Quattro",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = & (const struct snd_usb_audio_quirk[]) {
+ /*
+ * Interfaces 0-2 are "Windows-compatible", 16-bit only,
+ * and share endpoints with the other interfaces.
+ * Ignore them. The other interfaces can do 24 bits,
+ * but captured samples are big-endian (see usbaudio.c).
+ */
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 4,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 5,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 6,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 7,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 8,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 9,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2003),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "AudioPhile",
+ .ifnum = 6,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2008),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "Ozone",
+ .ifnum = 3,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0763, 0x200d),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "M-Audio",
+ .product_name = "OmniStudio",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = & (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 4,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 5,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 6,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 7,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 8,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 9,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ USB_DEVICE(0x0763, 0x2019),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ /* .vendor_name = "M-Audio", */
+ /* .product_name = "Ozone Academic", */
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = & (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_MIDIMAN,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+
+/* Casio devices */
+{
+ USB_DEVICE(0x07cf, 0x6801),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Casio",
+ .product_name = "PL-40R",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_YAMAHA
+ }
+},
+{
+ /* this ID is used by several devices without a product ID */
+ USB_DEVICE(0x07cf, 0x6802),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Casio",
+ .product_name = "Keyboard",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_YAMAHA
+ }
+},
+
+/* Mark of the Unicorn devices */
+{
+ /* thanks to Robert A. Lerche <ral 'at' msbit.com> */
+ .match_flags = USB_DEVICE_ID_MATCH_VENDOR |
+ USB_DEVICE_ID_MATCH_PRODUCT |
+ USB_DEVICE_ID_MATCH_DEV_SUBCLASS,
+ .idVendor = 0x07fd,
+ .idProduct = 0x0001,
+ .bDeviceSubClass = 2,
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "MOTU",
+ .product_name = "Fastlane",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = & (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_MIDI_RAW
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+
+/* Emagic devices */
+{
+ USB_DEVICE(0x086a, 0x0001),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Emagic",
+ /* .product_name = "Unitor8", */
+ .ifnum = 2,
+ .type = QUIRK_MIDI_EMAGIC,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x80ff,
+ .in_cables = 0x80ff
+ }
+ }
+},
+{
+ USB_DEVICE(0x086a, 0x0002),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Emagic",
+ /* .product_name = "AMT8", */
+ .ifnum = 2,
+ .type = QUIRK_MIDI_EMAGIC,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x80ff,
+ .in_cables = 0x80ff
+ }
+ }
+},
+{
+ USB_DEVICE(0x086a, 0x0003),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Emagic",
+ /* .product_name = "MT4", */
+ .ifnum = 2,
+ .type = QUIRK_MIDI_EMAGIC,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x800f,
+ .in_cables = 0x8003
+ }
+ }
+},
+
+/* TerraTec devices */
+{
+ USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "TerraTec",
+ .product_name = "PHASE 26",
+ .ifnum = 3,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0013),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "TerraTec",
+ .product_name = "PHASE 26",
+ .ifnum = 3,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0014),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "TerraTec",
+ .product_name = "PHASE 26",
+ .ifnum = 3,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+{
+ USB_DEVICE(0x0ccd, 0x0035),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Miditech",
+ .product_name = "Play'n Roll",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_CME
+ }
+},
+
+/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */
+{
+ USB_DEVICE(0x103d, 0x0100),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Stanton",
+ .product_name = "ScratchAmp",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+{
+ USB_DEVICE(0x103d, 0x0101),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Stanton",
+ .product_name = "ScratchAmp",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+
+/* Novation EMS devices */
+{
+ USB_DEVICE_VENDOR_SPEC(0x1235, 0x0001),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Novation",
+ .product_name = "ReMOTE Audio/XStation",
+ .ifnum = 4,
+ .type = QUIRK_MIDI_NOVATION
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x1235, 0x0002),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Novation",
+ .product_name = "Speedio",
+ .ifnum = 3,
+ .type = QUIRK_MIDI_NOVATION
+ }
+},
+{
+ USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Novation",
+ .product_name = "ReMOTE25",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_NOVATION
+ }
+},
+
+/* */
+{
+ /* aka. Serato Scratch Live DJ Box */
+ USB_DEVICE(0x13e5, 0x0001),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Rane",
+ .product_name = "SL-1",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+
+/* Miditech devices */
+{
+ USB_DEVICE(0x4752, 0x0011),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Miditech",
+ .product_name = "Midistart-2",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_CME
+ }
+},
+
+/* Central Music devices */
+{
+ /* this ID used by both Miditech MidiStudio-2 and CME UF-x */
+ USB_DEVICE(0x7104, 0x2202),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .ifnum = 0,
+ .type = QUIRK_MIDI_CME
+ }
+},
+
+{
+ /*
+ * Some USB MIDI devices don't have an audio control interface,
+ * so we have to grab MIDI streaming interfaces here.
+ */
+ .match_flags = USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_MIDI_STREAMING,
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ }
+},
+
+#undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/usx2y/Makefile b/sound/usb/usx2y/Makefile
new file mode 100644
index 0000000..7489330
--- /dev/null
+++ b/sound/usb/usx2y/Makefile
@@ -0,0 +1,5 @@
+snd-usb-usx2y-objs := usbusx2y.o usX2Yhwdep.o usx2yhwdeppcm.o
+snd-usb-us122l-objs := us122l.o
+
+obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-usx2y.o
+obj-$(CONFIG_SND_USB_US122L) += snd-usb-us122l.o
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
new file mode 100644
index 0000000..c2515b6
--- /dev/null
+++ b/sound/usb/usx2y/us122l.c
@@ -0,0 +1,693 @@
+/*
+ * Copyright (C) 2007, 2008 Karsten Wiese <fzu@wemgehoertderstaat.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY
+ * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
+ * for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <sound/core.h>
+#include <sound/hwdep.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#define MODNAME "US122L"
+#include "usb_stream.c"
+#include "../usbaudio.h"
+#include "us122l.h"
+
+MODULE_AUTHOR("Karsten Wiese <fzu@wemgehoertderstaat.de>");
+MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.5");
+MODULE_LICENSE("GPL");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
+ /* Enable this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for "NAME_ALLCAPS".");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for "NAME_ALLCAPS".");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable "NAME_ALLCAPS".");
+
+static int snd_us122l_card_used[SNDRV_CARDS];
+
+
+static int us122l_create_usbmidi(struct snd_card *card)
+{
+ static struct snd_usb_midi_endpoint_info quirk_data = {
+ .out_ep = 4,
+ .in_ep = 3,
+ .out_cables = 0x001,
+ .in_cables = 0x001
+ };
+ static struct snd_usb_audio_quirk quirk = {
+ .vendor_name = "US122L",
+ .product_name = NAME_ALLCAPS,
+ .ifnum = 1,
+ .type = QUIRK_MIDI_US122L,
+ .data = &quirk_data
+ };
+ struct usb_device *dev = US122L(card)->chip.dev;
+ struct usb_interface *iface = usb_ifnum_to_if(dev, 1);
+
+ return snd_usb_create_midi_interface(&US122L(card)->chip,
+ iface, &quirk);
+}
+
+/*
+ * Wrapper for usb_control_msg().
+ * Allocates a temp buffer to prevent dmaing from/to the stack.
+ */
+static int us122l_ctl_msg(struct usb_device *dev, unsigned int pipe,
+ __u8 request, __u8 requesttype,
+ __u16 value, __u16 index, void *data,
+ __u16 size, int timeout)
+{
+ int err;
+ void *buf = NULL;
+
+ if (size > 0) {
+ buf = kmemdup(data, size, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+ }
+ err = usb_control_msg(dev, pipe, request, requesttype,
+ value, index, buf, size, timeout);
+ if (size > 0) {
+ memcpy(data, buf, size);
+ kfree(buf);
+ }
+ return err;
+}
+
+static void pt_info_set(struct usb_device *dev, u8 v)
+{
+ int ret;
+
+ ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
+ 'I',
+ USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+ v, 0, NULL, 0, 1000);
+ snd_printdd(KERN_DEBUG "%i\n", ret);
+}
+
+static void usb_stream_hwdep_vm_open(struct vm_area_struct *area)
+{
+ struct us122l *us122l = area->vm_private_data;
+ atomic_inc(&us122l->mmap_count);
+ snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count));
+}
+
+static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area,
+ struct vm_fault *vmf)
+{
+ unsigned long offset;
+ struct page *page;
+ void *vaddr;
+ struct us122l *us122l = area->vm_private_data;
+ struct usb_stream *s;
+
+ mutex_lock(&us122l->mutex);
+ s = us122l->sk.s;
+ if (!s)
+ goto unlock;
+
+ offset = vmf->pgoff << PAGE_SHIFT;
+ if (offset < PAGE_ALIGN(s->read_size))
+ vaddr = (char *)s + offset;
+ else {
+ offset -= PAGE_ALIGN(s->read_size);
+ if (offset >= PAGE_ALIGN(s->write_size))
+ goto unlock;
+
+ vaddr = us122l->sk.write_page + offset;
+ }
+ page = virt_to_page(vaddr);
+
+ get_page(page);
+ mutex_unlock(&us122l->mutex);
+
+ vmf->page = page;
+
+ return 0;
+unlock:
+ mutex_unlock(&us122l->mutex);
+ return VM_FAULT_SIGBUS;
+}
+
+static void usb_stream_hwdep_vm_close(struct vm_area_struct *area)
+{
+ struct us122l *us122l = area->vm_private_data;
+ atomic_dec(&us122l->mmap_count);
+ snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count));
+}
+
+static struct vm_operations_struct usb_stream_hwdep_vm_ops = {
+ .open = usb_stream_hwdep_vm_open,
+ .fault = usb_stream_hwdep_vm_fault,
+ .close = usb_stream_hwdep_vm_close,
+};
+
+
+static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+ struct us122l *us122l = hw->private_data;
+ struct usb_interface *iface;
+ snd_printdd(KERN_DEBUG "%p %p\n", hw, file);
+ if (hw->used >= 2)
+ return -EBUSY;
+
+ if (!us122l->first)
+ us122l->first = file;
+ iface = usb_ifnum_to_if(us122l->chip.dev, 1);
+ usb_autopm_get_interface(iface);
+ return 0;
+}
+
+static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file)
+{
+ struct us122l *us122l = hw->private_data;
+ struct usb_interface *iface = usb_ifnum_to_if(us122l->chip.dev, 1);
+ snd_printdd(KERN_DEBUG "%p %p\n", hw, file);
+ usb_autopm_put_interface(iface);
+ if (us122l->first == file)
+ us122l->first = NULL;
+ mutex_lock(&us122l->mutex);
+ if (us122l->master == file)
+ us122l->master = us122l->slave;
+
+ us122l->slave = NULL;
+ mutex_unlock(&us122l->mutex);
+ return 0;
+}
+
+static int usb_stream_hwdep_mmap(struct snd_hwdep *hw,
+ struct file *filp, struct vm_area_struct *area)
+{
+ unsigned long size = area->vm_end - area->vm_start;
+ struct us122l *us122l = hw->private_data;
+ unsigned long offset;
+ struct usb_stream *s;
+ int err = 0;
+ bool read;
+
+ offset = area->vm_pgoff << PAGE_SHIFT;
+ mutex_lock(&us122l->mutex);
+ s = us122l->sk.s;
+ read = offset < s->read_size;
+ if (read && area->vm_flags & VM_WRITE) {
+ err = -EPERM;
+ goto out;
+ }
+ snd_printdd(KERN_DEBUG "%lu %u\n", size,
+ read ? s->read_size : s->write_size);
+ /* if userspace tries to mmap beyond end of our buffer, fail */
+ if (size > PAGE_ALIGN(read ? s->read_size : s->write_size)) {
+ snd_printk(KERN_WARNING "%lu > %u\n", size,
+ read ? s->read_size : s->write_size);
+ err = -EINVAL;
+ goto out;
+ }
+
+ area->vm_ops = &usb_stream_hwdep_vm_ops;
+ area->vm_flags |= VM_RESERVED;
+ area->vm_private_data = us122l;
+ atomic_inc(&us122l->mmap_count);
+out:
+ mutex_unlock(&us122l->mutex);
+ return err;
+}
+
+static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw,
+ struct file *file, poll_table *wait)
+{
+ struct us122l *us122l = hw->private_data;
+ struct usb_stream *s = us122l->sk.s;
+ unsigned *polled;
+ unsigned int mask;
+
+ poll_wait(file, &us122l->sk.sleep, wait);
+
+ switch (s->state) {
+ case usb_stream_ready:
+ if (us122l->first == file)
+ polled = &s->periods_polled;
+ else
+ polled = &us122l->second_periods_polled;
+ if (*polled != s->periods_done) {
+ *polled = s->periods_done;
+ mask = POLLIN | POLLOUT | POLLWRNORM;
+ break;
+ }
+ /* Fall through */
+ mask = 0;
+ break;
+ default:
+ mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR;
+ break;
+ }
+ return mask;
+}
+
+static void us122l_stop(struct us122l *us122l)
+{
+ struct list_head *p;
+ list_for_each(p, &us122l->chip.midi_list)
+ snd_usbmidi_input_stop(p);
+
+ usb_stream_stop(&us122l->sk);
+ usb_stream_free(&us122l->sk);
+}
+
+static int us122l_set_sample_rate(struct usb_device *dev, int rate)
+{
+ unsigned int ep = 0x81;
+ unsigned char data[3];
+ int err;
+
+ data[0] = rate;
+ data[1] = rate >> 8;
+ data[2] = rate >> 16;
+ err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR,
+ USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
+ SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000);
+ if (err < 0)
+ snd_printk(KERN_ERR "%d: cannot set freq %d to ep 0x%x\n",
+ dev->devnum, rate, ep);
+ return err;
+}
+
+static bool us122l_start(struct us122l *us122l,
+ unsigned rate, unsigned period_frames)
+{
+ struct list_head *p;
+ int err;
+ unsigned use_packsize = 0;
+ bool success = false;
+
+ if (us122l->chip.dev->speed == USB_SPEED_HIGH) {
+ /* The us-122l's descriptor defaults to iso max_packsize 78,
+ which isn't needed for samplerates <= 48000.
+ Lets save some memory:
+ */
+ switch (rate) {
+ case 44100:
+ use_packsize = 36;
+ break;
+ case 48000:
+ use_packsize = 42;
+ break;
+ case 88200:
+ use_packsize = 72;
+ break;
+ }
+ }
+ if (!usb_stream_new(&us122l->sk, us122l->chip.dev, 1, 2,
+ rate, use_packsize, period_frames, 6))
+ goto out;
+
+ err = us122l_set_sample_rate(us122l->chip.dev, rate);
+ if (err < 0) {
+ us122l_stop(us122l);
+ snd_printk(KERN_ERR "us122l_set_sample_rate error \n");
+ goto out;
+ }
+ err = usb_stream_start(&us122l->sk);
+ if (err < 0) {
+ us122l_stop(us122l);
+ snd_printk(KERN_ERR "us122l_start error %i \n", err);
+ goto out;
+ }
+ list_for_each(p, &us122l->chip.midi_list)
+ snd_usbmidi_input_start(p);
+ success = true;
+out:
+ return success;
+}
+
+static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
+ unsigned cmd, unsigned long arg)
+{
+ struct usb_stream_config *cfg;
+ struct us122l *us122l = hw->private_data;
+ unsigned min_period_frames;
+ int err = 0;
+ bool high_speed;
+
+ if (cmd != SNDRV_USB_STREAM_IOCTL_SET_PARAMS)
+ return -ENOTTY;
+
+ cfg = kmalloc(sizeof(*cfg), GFP_KERNEL);
+ if (!cfg)
+ return -ENOMEM;
+
+ if (copy_from_user(cfg, (void *)arg, sizeof(*cfg))) {
+ err = -EFAULT;
+ goto free;
+ }
+ if (cfg->version != USB_STREAM_INTERFACE_VERSION) {
+ err = -ENXIO;
+ goto free;
+ }
+ high_speed = us122l->chip.dev->speed == USB_SPEED_HIGH;
+ if ((cfg->sample_rate != 44100 && cfg->sample_rate != 48000 &&
+ (!high_speed ||
+ (cfg->sample_rate != 88200 && cfg->sample_rate != 96000))) ||
+ cfg->frame_size != 6 ||
+ cfg->period_frames > 0x3000) {
+ err = -EINVAL;
+ goto free;
+ }
+ switch (cfg->sample_rate) {
+ case 44100:
+ min_period_frames = 48;
+ break;
+ case 48000:
+ min_period_frames = 52;
+ break;
+ default:
+ min_period_frames = 104;
+ break;
+ }
+ if (!high_speed)
+ min_period_frames <<= 1;
+ if (cfg->period_frames < min_period_frames) {
+ err = -EINVAL;
+ goto free;
+ }
+
+ snd_power_wait(hw->card, SNDRV_CTL_POWER_D0);
+
+ mutex_lock(&us122l->mutex);
+ if (!us122l->master)
+ us122l->master = file;
+ else if (us122l->master != file) {
+ if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) {
+ err = -EIO;
+ goto unlock;
+ }
+ us122l->slave = file;
+ }
+ if (!us122l->sk.s ||
+ memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) ||
+ us122l->sk.s->state == usb_stream_xrun) {
+ us122l_stop(us122l);
+ if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames))
+ err = -EIO;
+ else
+ err = 1;
+ }
+unlock:
+ mutex_unlock(&us122l->mutex);
+free:
+ kfree(cfg);
+ return err;
+}
+
+#define SND_USB_STREAM_ID "USB STREAM"
+static int usb_stream_hwdep_new(struct snd_card *card)
+{
+ int err;
+ struct snd_hwdep *hw;
+ struct usb_device *dev = US122L(card)->chip.dev;
+
+ err = snd_hwdep_new(card, SND_USB_STREAM_ID, 0, &hw);
+ if (err < 0)
+ return err;
+
+ hw->iface = SNDRV_HWDEP_IFACE_USB_STREAM;
+ hw->private_data = US122L(card);
+ hw->ops.open = usb_stream_hwdep_open;
+ hw->ops.release = usb_stream_hwdep_release;
+ hw->ops.ioctl = usb_stream_hwdep_ioctl;
+ hw->ops.ioctl_compat = usb_stream_hwdep_ioctl;
+ hw->ops.mmap = usb_stream_hwdep_mmap;
+ hw->ops.poll = usb_stream_hwdep_poll;
+
+ sprintf(hw->name, "/proc/bus/usb/%03d/%03d/hwdeppcm",
+ dev->bus->busnum, dev->devnum);
+ return 0;
+}
+
+
+static bool us122l_create_card(struct snd_card *card)
+{
+ int err;
+ struct us122l *us122l = US122L(card);
+
+ err = usb_set_interface(us122l->chip.dev, 1, 1);
+ if (err) {
+ snd_printk(KERN_ERR "usb_set_interface error \n");
+ return false;
+ }
+
+ pt_info_set(us122l->chip.dev, 0x11);
+ pt_info_set(us122l->chip.dev, 0x10);
+
+ if (!us122l_start(us122l, 44100, 256))
+ return false;
+
+ err = us122l_create_usbmidi(card);
+ if (err < 0) {
+ snd_printk(KERN_ERR "us122l_create_usbmidi error %i \n", err);
+ us122l_stop(us122l);
+ return false;
+ }
+ err = usb_stream_hwdep_new(card);
+ if (err < 0) {
+/* release the midi resources */
+ struct list_head *p;
+ list_for_each(p, &us122l->chip.midi_list)
+ snd_usbmidi_disconnect(p);
+
+ us122l_stop(us122l);
+ return false;
+ }
+ return true;
+}
+
+static struct snd_card *usx2y_create_card(struct usb_device *device)
+{
+ int dev;
+ struct snd_card *card;
+ for (dev = 0; dev < SNDRV_CARDS; ++dev)
+ if (enable[dev] && !snd_us122l_card_used[dev])
+ break;
+ if (dev >= SNDRV_CARDS)
+ return NULL;
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE,
+ sizeof(struct us122l));
+ if (!card)
+ return NULL;
+ snd_us122l_card_used[US122L(card)->chip.index = dev] = 1;
+
+ US122L(card)->chip.dev = device;
+ US122L(card)->chip.card = card;
+ mutex_init(&US122L(card)->mutex);
+ init_waitqueue_head(&US122L(card)->sk.sleep);
+ INIT_LIST_HEAD(&US122L(card)->chip.midi_list);
+ strcpy(card->driver, "USB "NAME_ALLCAPS"");
+ sprintf(card->shortname, "TASCAM "NAME_ALLCAPS"");
+ sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)",
+ card->shortname,
+ le16_to_cpu(device->descriptor.idVendor),
+ le16_to_cpu(device->descriptor.idProduct),
+ 0,
+ US122L(card)->chip.dev->bus->busnum,
+ US122L(card)->chip.dev->devnum
+ );
+ snd_card_set_dev(card, &device->dev);
+ return card;
+}
+
+static void *us122l_usb_probe(struct usb_interface *intf,
+ const struct usb_device_id *device_id)
+{
+ struct usb_device *device = interface_to_usbdev(intf);
+ struct snd_card *card = usx2y_create_card(device);
+
+ if (!card)
+ return NULL;
+
+ if (!us122l_create_card(card) ||
+ snd_card_register(card) < 0) {
+ snd_card_free(card);
+ return NULL;
+ }
+
+ usb_get_dev(device);
+ return card;
+}
+
+static int snd_us122l_probe(struct usb_interface *intf,
+ const struct usb_device_id *id)
+{
+ struct snd_card *card;
+ snd_printdd(KERN_DEBUG"%p:%i\n",
+ intf, intf->cur_altsetting->desc.bInterfaceNumber);
+ if (intf->cur_altsetting->desc.bInterfaceNumber != 1)
+ return 0;
+
+ card = us122l_usb_probe(usb_get_intf(intf), id);
+
+ if (card) {
+ usb_set_intfdata(intf, card);
+ return 0;
+ }
+
+ usb_put_intf(intf);
+ return -EIO;
+}
+
+static void snd_us122l_disconnect(struct usb_interface *intf)
+{
+ struct snd_card *card;
+ struct us122l *us122l;
+ struct list_head *p;
+
+ card = usb_get_intfdata(intf);
+ if (!card)
+ return;
+
+ snd_card_disconnect(card);
+
+ us122l = US122L(card);
+ mutex_lock(&us122l->mutex);
+ us122l_stop(us122l);
+ mutex_unlock(&us122l->mutex);
+ us122l->chip.shutdown = 1;
+
+/* release the midi resources */
+ list_for_each(p, &us122l->chip.midi_list) {
+ snd_usbmidi_disconnect(p);
+ }
+
+ usb_put_intf(intf);
+ usb_put_dev(US122L(card)->chip.dev);
+
+ while (atomic_read(&us122l->mmap_count))
+ msleep(500);
+
+ snd_card_free(card);
+}
+
+static int snd_us122l_suspend(struct usb_interface *intf, pm_message_t message)
+{
+ struct snd_card *card;
+ struct us122l *us122l;
+ struct list_head *p;
+
+ card = dev_get_drvdata(&intf->dev);
+ if (!card)
+ return 0;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ us122l = US122L(card);
+ if (!us122l)
+ return 0;
+
+ list_for_each(p, &us122l->chip.midi_list)
+ snd_usbmidi_input_stop(p);
+
+ mutex_lock(&us122l->mutex);
+ usb_stream_stop(&us122l->sk);
+ mutex_unlock(&us122l->mutex);
+
+ return 0;
+}
+
+static int snd_us122l_resume(struct usb_interface *intf)
+{
+ struct snd_card *card;
+ struct us122l *us122l;
+ struct list_head *p;
+ int err;
+
+ card = dev_get_drvdata(&intf->dev);
+ if (!card)
+ return 0;
+
+ us122l = US122L(card);
+ if (!us122l)
+ return 0;
+
+ mutex_lock(&us122l->mutex);
+ /* needed, doesn't restart without: */
+ err = usb_set_interface(us122l->chip.dev, 1, 1);
+ if (err) {
+ snd_printk(KERN_ERR "usb_set_interface error \n");
+ goto unlock;
+ }
+
+ pt_info_set(us122l->chip.dev, 0x11);
+ pt_info_set(us122l->chip.dev, 0x10);
+
+ err = us122l_set_sample_rate(us122l->chip.dev,
+ us122l->sk.s->cfg.sample_rate);
+ if (err < 0) {
+ snd_printk(KERN_ERR "us122l_set_sample_rate error \n");
+ goto unlock;
+ }
+ err = usb_stream_start(&us122l->sk);
+ if (err)
+ goto unlock;
+
+ list_for_each(p, &us122l->chip.midi_list)
+ snd_usbmidi_input_start(p);
+unlock:
+ mutex_unlock(&us122l->mutex);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ return err;
+}
+
+static struct usb_device_id snd_us122l_usb_id_table[] = {
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x0644,
+ .idProduct = USB_ID_US122L
+ },
+/* { */ /* US-144 maybe works when @USB1.1. Untested. */
+/* .match_flags = USB_DEVICE_ID_MATCH_DEVICE, */
+/* .idVendor = 0x0644, */
+/* .idProduct = USB_ID_US144 */
+/* }, */
+ { /* terminator */ }
+};
+
+MODULE_DEVICE_TABLE(usb, snd_us122l_usb_id_table);
+static struct usb_driver snd_us122l_usb_driver = {
+ .name = "snd-usb-us122l",
+ .probe = snd_us122l_probe,
+ .disconnect = snd_us122l_disconnect,
+ .suspend = snd_us122l_suspend,
+ .resume = snd_us122l_resume,
+ .reset_resume = snd_us122l_resume,
+ .id_table = snd_us122l_usb_id_table,
+ .supports_autosuspend = 1
+};
+
+
+static int __init snd_us122l_module_init(void)
+{
+ return usb_register(&snd_us122l_usb_driver);
+}
+
+static void __exit snd_us122l_module_exit(void)
+{
+ usb_deregister(&snd_us122l_usb_driver);
+}
+
+module_init(snd_us122l_module_init)
+module_exit(snd_us122l_module_exit)
diff --git a/sound/usb/usx2y/us122l.h b/sound/usb/usx2y/us122l.h
new file mode 100644
index 0000000..3d10c4b
--- /dev/null
+++ b/sound/usb/usx2y/us122l.h
@@ -0,0 +1,27 @@
+#ifndef US122L_H
+#define US122L_H
+
+
+struct us122l {
+ struct snd_usb_audio chip;
+ int stride;
+ struct usb_stream_kernel sk;
+
+ struct mutex mutex;
+ struct file *first;
+ unsigned second_periods_polled;
+ struct file *master;
+ struct file *slave;
+
+ atomic_t mmap_count;
+};
+
+
+#define US122L(c) ((struct us122l *)(c)->private_data)
+
+#define NAME_ALLCAPS "US-122L"
+
+#define USB_ID_US122L 0x800E
+#define USB_ID_US144 0x800F
+
+#endif
diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c
new file mode 100644
index 0000000..1558a5c
--- /dev/null
+++ b/sound/usb/usx2y/usX2Yhwdep.c
@@ -0,0 +1,280 @@
+/*
+ * Driver for Tascam US-X2Y USB soundcards
+ *
+ * FPGA Loader + ALSA Startup
+ *
+ * Copyright (c) 2003 by Karsten Wiese <annabellesgarden@yahoo.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/interrupt.h>
+#include <linux/usb.h>
+#include <sound/core.h>
+#include <sound/memalloc.h>
+#include <sound/pcm.h>
+#include <sound/hwdep.h>
+#include "usx2y.h"
+#include "usbusx2y.h"
+#include "usX2Yhwdep.h"
+
+int usX2Y_hwdep_pcm_new(struct snd_card *card);
+
+
+static int snd_us428ctls_vm_fault(struct vm_area_struct *area,
+ struct vm_fault *vmf)
+{
+ unsigned long offset;
+ struct page * page;
+ void *vaddr;
+
+ snd_printdd("ENTER, start %lXh, pgoff %ld\n",
+ area->vm_start,
+ vmf->pgoff);
+
+ offset = vmf->pgoff << PAGE_SHIFT;
+ vaddr = (char*)((struct usX2Ydev *)area->vm_private_data)->us428ctls_sharedmem + offset;
+ page = virt_to_page(vaddr);
+ get_page(page);
+ vmf->page = page;
+
+ snd_printdd("vaddr=%p made us428ctls_vm_fault() page %p\n",
+ vaddr, page);
+
+ return 0;
+}
+
+static struct vm_operations_struct us428ctls_vm_ops = {
+ .fault = snd_us428ctls_vm_fault,
+};
+
+static int snd_us428ctls_mmap(struct snd_hwdep * hw, struct file *filp, struct vm_area_struct *area)
+{
+ unsigned long size = (unsigned long)(area->vm_end - area->vm_start);
+ struct usX2Ydev *us428 = hw->private_data;
+
+ // FIXME this hwdep interface is used twice: fpga download and mmap for controlling Lights etc. Maybe better using 2 hwdep devs?
+ // so as long as the device isn't fully initialised yet we return -EBUSY here.
+ if (!(us428->chip_status & USX2Y_STAT_CHIP_INIT))
+ return -EBUSY;
+
+ /* if userspace tries to mmap beyond end of our buffer, fail */
+ if (size > PAGE_ALIGN(sizeof(struct us428ctls_sharedmem))) {
+ snd_printd( "%lu > %lu\n", size, (unsigned long)sizeof(struct us428ctls_sharedmem));
+ return -EINVAL;
+ }
+
+ if (!us428->us428ctls_sharedmem) {
+ init_waitqueue_head(&us428->us428ctls_wait_queue_head);
+ if(!(us428->us428ctls_sharedmem = snd_malloc_pages(sizeof(struct us428ctls_sharedmem), GFP_KERNEL)))
+ return -ENOMEM;
+ memset(us428->us428ctls_sharedmem, -1, sizeof(struct us428ctls_sharedmem));
+ us428->us428ctls_sharedmem->CtlSnapShotLast = -2;
+ }
+ area->vm_ops = &us428ctls_vm_ops;
+ area->vm_flags |= VM_RESERVED | VM_DONTEXPAND;
+ area->vm_private_data = hw->private_data;
+ return 0;
+}
+
+static unsigned int snd_us428ctls_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait)
+{
+ unsigned int mask = 0;
+ struct usX2Ydev *us428 = hw->private_data;
+ struct us428ctls_sharedmem *shm = us428->us428ctls_sharedmem;
+ if (us428->chip_status & USX2Y_STAT_CHIP_HUP)
+ return POLLHUP;
+
+ poll_wait(file, &us428->us428ctls_wait_queue_head, wait);
+
+ if (shm != NULL && shm->CtlSnapShotLast != shm->CtlSnapShotRed)
+ mask |= POLLIN;
+
+ return mask;
+}
+
+
+static int snd_usX2Y_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+static int snd_usX2Y_hwdep_release(struct snd_hwdep *hw, struct file *file)
+{
+ return 0;
+}
+
+static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_status *info)
+{
+ static char *type_ids[USX2Y_TYPE_NUMS] = {
+ [USX2Y_TYPE_122] = "us122",
+ [USX2Y_TYPE_224] = "us224",
+ [USX2Y_TYPE_428] = "us428",
+ };
+ struct usX2Ydev *us428 = hw->private_data;
+ int id = -1;
+
+ switch (le16_to_cpu(us428->chip.dev->descriptor.idProduct)) {
+ case USB_ID_US122:
+ id = USX2Y_TYPE_122;
+ break;
+ case USB_ID_US224:
+ id = USX2Y_TYPE_224;
+ break;
+ case USB_ID_US428:
+ id = USX2Y_TYPE_428;
+ break;
+ }
+ if (0 > id)
+ return -ENODEV;
+ strcpy(info->id, type_ids[id]);
+ info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code
+ if (us428->chip_status & USX2Y_STAT_CHIP_INIT)
+ info->chip_ready = 1;
+ info->version = USX2Y_DRIVER_VERSION;
+ return 0;
+}
+
+
+static int usX2Y_create_usbmidi(struct snd_card *card)
+{
+ static struct snd_usb_midi_endpoint_info quirk_data_1 = {
+ .out_ep = 0x06,
+ .in_ep = 0x06,
+ .out_cables = 0x001,
+ .in_cables = 0x001
+ };
+ static struct snd_usb_audio_quirk quirk_1 = {
+ .vendor_name = "TASCAM",
+ .product_name = NAME_ALLCAPS,
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = &quirk_data_1
+ };
+ static struct snd_usb_midi_endpoint_info quirk_data_2 = {
+ .out_ep = 0x06,
+ .in_ep = 0x06,
+ .out_cables = 0x003,
+ .in_cables = 0x003
+ };
+ static struct snd_usb_audio_quirk quirk_2 = {
+ .vendor_name = "TASCAM",
+ .product_name = "US428",
+ .ifnum = 0,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = &quirk_data_2
+ };
+ struct usb_device *dev = usX2Y(card)->chip.dev;
+ struct usb_interface *iface = usb_ifnum_to_if(dev, 0);
+ struct snd_usb_audio_quirk *quirk =
+ le16_to_cpu(dev->descriptor.idProduct) == USB_ID_US428 ?
+ &quirk_2 : &quirk_1;
+
+ snd_printdd("usX2Y_create_usbmidi \n");
+ return snd_usb_create_midi_interface(&usX2Y(card)->chip, iface, quirk);
+}
+
+static int usX2Y_create_alsa_devices(struct snd_card *card)
+{
+ int err;
+
+ do {
+ if ((err = usX2Y_create_usbmidi(card)) < 0) {
+ snd_printk(KERN_ERR "usX2Y_create_alsa_devices: usX2Y_create_usbmidi error %i \n", err);
+ break;
+ }
+ if ((err = usX2Y_audio_create(card)) < 0)
+ break;
+ if ((err = usX2Y_hwdep_pcm_new(card)) < 0)
+ break;
+ if ((err = snd_card_register(card)) < 0)
+ break;
+ } while (0);
+
+ return err;
+}
+
+static int snd_usX2Y_hwdep_dsp_load(struct snd_hwdep *hw,
+ struct snd_hwdep_dsp_image *dsp)
+{
+ struct usX2Ydev *priv = hw->private_data;
+ int lret, err = -EINVAL;
+ snd_printdd( "dsp_load %s\n", dsp->name);
+
+ if (access_ok(VERIFY_READ, dsp->image, dsp->length)) {
+ struct usb_device* dev = priv->chip.dev;
+ char *buf = kmalloc(dsp->length, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+ if (copy_from_user(buf, dsp->image, dsp->length)) {
+ kfree(buf);
+ return -EFAULT;
+ }
+ err = usb_set_interface(dev, 0, 1);
+ if (err)
+ snd_printk(KERN_ERR "usb_set_interface error \n");
+ else
+ err = usb_bulk_msg(dev, usb_sndbulkpipe(dev, 2), buf, dsp->length, &lret, 6000);
+ kfree(buf);
+ }
+ if (err)
+ return err;
+ if (dsp->index == 1) {
+ msleep(250); // give the device some time
+ err = usX2Y_AsyncSeq04_init(priv);
+ if (err) {
+ snd_printk(KERN_ERR "usX2Y_AsyncSeq04_init error \n");
+ return err;
+ }
+ err = usX2Y_In04_init(priv);
+ if (err) {
+ snd_printk(KERN_ERR "usX2Y_In04_init error \n");
+ return err;
+ }
+ err = usX2Y_create_alsa_devices(hw->card);
+ if (err) {
+ snd_printk(KERN_ERR "usX2Y_create_alsa_devices error %i \n", err);
+ snd_card_free(hw->card);
+ return err;
+ }
+ priv->chip_status |= USX2Y_STAT_CHIP_INIT;
+ snd_printdd("%s: alsa all started\n", hw->name);
+ }
+ return err;
+}
+
+
+int usX2Y_hwdep_new(struct snd_card *card, struct usb_device* device)
+{
+ int err;
+ struct snd_hwdep *hw;
+
+ if ((err = snd_hwdep_new(card, SND_USX2Y_LOADER_ID, 0, &hw)) < 0)
+ return err;
+
+ hw->iface = SNDRV_HWDEP_IFACE_USX2Y;
+ hw->private_data = usX2Y(card);
+ hw->ops.open = snd_usX2Y_hwdep_open;
+ hw->ops.release = snd_usX2Y_hwdep_release;
+ hw->ops.dsp_status = snd_usX2Y_hwdep_dsp_status;
+ hw->ops.dsp_load = snd_usX2Y_hwdep_dsp_load;
+ hw->ops.mmap = snd_us428ctls_mmap;
+ hw->ops.poll = snd_us428ctls_poll;
+ hw->exclusive = 1;
+ sprintf(hw->name, "/proc/bus/usb/%03d/%03d", device->bus->busnum, device->devnum);
+ return 0;
+}
+
diff --git a/sound/usb/usx2y/usX2Yhwdep.h b/sound/usb/usx2y/usX2Yhwdep.h
new file mode 100644
index 0000000..c095d5b
--- /dev/null
+++ b/sound/usb/usx2y/usX2Yhwdep.h
@@ -0,0 +1,6 @@
+#ifndef USX2YHWDEP_H
+#define USX2YHWDEP_H
+
+int usX2Y_hwdep_new(struct snd_card *card, struct usb_device* device);
+
+#endif
diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
new file mode 100644
index 0000000..ff23cc1
--- /dev/null
+++ b/sound/usb/usx2y/usb_stream.c
@@ -0,0 +1,761 @@
+/*
+ * Copyright (C) 2007, 2008 Karsten Wiese <fzu@wemgehoertderstaat.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY
+ * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
+ * for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <linux/usb.h>
+
+#include "usb_stream.h"
+
+
+/* setup */
+
+static unsigned usb_stream_next_packet_size(struct usb_stream_kernel *sk)
+{
+ struct usb_stream *s = sk->s;
+ sk->out_phase_peeked = (sk->out_phase & 0xffff) + sk->freqn;
+ return (sk->out_phase_peeked >> 16) * s->cfg.frame_size;
+}
+
+static void playback_prep_freqn(struct usb_stream_kernel *sk, struct urb *urb)
+{
+ struct usb_stream *s = sk->s;
+ unsigned l = 0;
+ int pack;
+
+ urb->iso_frame_desc[0].offset = 0;
+ urb->iso_frame_desc[0].length = usb_stream_next_packet_size(sk);
+ sk->out_phase = sk->out_phase_peeked;
+ urb->transfer_buffer_length = urb->iso_frame_desc[0].length;
+
+ for (pack = 1; pack < sk->n_o_ps; pack++) {
+ l = usb_stream_next_packet_size(sk);
+ if (s->idle_outsize + urb->transfer_buffer_length + l >
+ s->period_size)
+ goto check;
+
+ sk->out_phase = sk->out_phase_peeked;
+ urb->iso_frame_desc[pack].offset = urb->transfer_buffer_length;
+ urb->iso_frame_desc[pack].length = l;
+ urb->transfer_buffer_length += l;
+ }
+ snd_printdd(KERN_DEBUG "%i\n", urb->transfer_buffer_length);
+
+check:
+ urb->number_of_packets = pack;
+ s->idle_outsize += urb->transfer_buffer_length - s->period_size;
+ snd_printdd(KERN_DEBUG "idle=%i ul=%i ps=%i\n", s->idle_outsize,
+ urb->transfer_buffer_length, s->period_size);
+}
+
+static void init_pipe_urbs(struct usb_stream_kernel *sk, unsigned use_packsize,
+ struct urb **urbs, char *transfer,
+ struct usb_device *dev, int pipe)
+{
+ int u, p;
+ int maxpacket = use_packsize ?
+ use_packsize : usb_maxpacket(dev, pipe, usb_pipeout(pipe));
+ int transfer_length = maxpacket * sk->n_o_ps;
+
+ for (u = 0; u < USB_STREAM_NURBS;
+ ++u, transfer += transfer_length) {
+ struct urb *urb = urbs[u];
+ struct usb_iso_packet_descriptor *desc;
+ urb->transfer_flags = URB_ISO_ASAP;
+ urb->transfer_buffer = transfer;
+ urb->dev = dev;
+ urb->pipe = pipe;
+ urb->number_of_packets = sk->n_o_ps;
+ urb->context = sk;
+ urb->interval = 1;
+ if (usb_pipeout(pipe))
+ continue;
+
+ urb->transfer_buffer_length = transfer_length;
+ desc = urb->iso_frame_desc;
+ desc->offset = 0;
+ desc->length = maxpacket;
+ for (p = 1; p < sk->n_o_ps; ++p) {
+ desc[p].offset = desc[p - 1].offset + maxpacket;
+ desc[p].length = maxpacket;
+ }
+ }
+}
+
+static void init_urbs(struct usb_stream_kernel *sk, unsigned use_packsize,
+ struct usb_device *dev, int in_pipe, int out_pipe)
+{
+ struct usb_stream *s = sk->s;
+ char *indata = (char *)s + sizeof(*s) +
+ sizeof(struct usb_stream_packet) *
+ s->inpackets;
+ int u;
+
+ for (u = 0; u < USB_STREAM_NURBS; ++u) {
+ sk->inurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL);
+ sk->outurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL);
+ }
+
+ init_pipe_urbs(sk, use_packsize, sk->inurb, indata, dev, in_pipe);
+ init_pipe_urbs(sk, use_packsize, sk->outurb, sk->write_page, dev,
+ out_pipe);
+}
+
+
+/*
+ * convert a sampling rate into our full speed format (fs/1000 in Q16.16)
+ * this will overflow at approx 524 kHz
+ */
+static inline unsigned get_usb_full_speed_rate(unsigned rate)
+{
+ return ((rate << 13) + 62) / 125;
+}
+
+/*
+ * convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
+ * this will overflow at approx 4 MHz
+ */
+static inline unsigned get_usb_high_speed_rate(unsigned rate)
+{
+ return ((rate << 10) + 62) / 125;
+}
+
+void usb_stream_free(struct usb_stream_kernel *sk)
+{
+ struct usb_stream *s;
+ unsigned u;
+
+ for (u = 0; u < USB_STREAM_NURBS; ++u) {
+ usb_free_urb(sk->inurb[u]);
+ sk->inurb[u] = NULL;
+ usb_free_urb(sk->outurb[u]);
+ sk->outurb[u] = NULL;
+ }
+
+ s = sk->s;
+ if (!s)
+ return;
+
+ free_pages((unsigned long)sk->write_page, get_order(s->write_size));
+ sk->write_page = NULL;
+ free_pages((unsigned long)s, get_order(s->read_size));
+ sk->s = NULL;
+}
+
+struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk,
+ struct usb_device *dev,
+ unsigned in_endpoint, unsigned out_endpoint,
+ unsigned sample_rate, unsigned use_packsize,
+ unsigned period_frames, unsigned frame_size)
+{
+ int packets, max_packsize;
+ int in_pipe, out_pipe;
+ int read_size = sizeof(struct usb_stream);
+ int write_size;
+ int usb_frames = dev->speed == USB_SPEED_HIGH ? 8000 : 1000;
+ int pg;
+
+ in_pipe = usb_rcvisocpipe(dev, in_endpoint);
+ out_pipe = usb_sndisocpipe(dev, out_endpoint);
+
+ max_packsize = use_packsize ?
+ use_packsize : usb_maxpacket(dev, in_pipe, 0);
+
+ /*
+ t_period = period_frames / sample_rate
+ iso_packs = t_period / t_iso_frame
+ = (period_frames / sample_rate) * (1 / t_iso_frame)
+ */
+
+ packets = period_frames * usb_frames / sample_rate + 1;
+
+ if (dev->speed == USB_SPEED_HIGH)
+ packets = (packets + 7) & ~7;
+
+ read_size += packets * USB_STREAM_URBDEPTH *
+ (max_packsize + sizeof(struct usb_stream_packet));
+
+ max_packsize = usb_maxpacket(dev, out_pipe, 1);
+ write_size = max_packsize * packets * USB_STREAM_URBDEPTH;
+
+ if (read_size >= 256*PAGE_SIZE || write_size >= 256*PAGE_SIZE) {
+ snd_printk(KERN_WARNING "a size exceeds 128*PAGE_SIZE\n");
+ goto out;
+ }
+
+ pg = get_order(read_size);
+ sk->s = (void *) __get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO, pg);
+ if (!sk->s) {
+ snd_printk(KERN_WARNING "couldn't __get_free_pages()\n");
+ goto out;
+ }
+ sk->s->cfg.version = USB_STREAM_INTERFACE_VERSION;
+
+ sk->s->read_size = read_size;
+
+ sk->s->cfg.sample_rate = sample_rate;
+ sk->s->cfg.frame_size = frame_size;
+ sk->n_o_ps = packets;
+ sk->s->inpackets = packets * USB_STREAM_URBDEPTH;
+ sk->s->cfg.period_frames = period_frames;
+ sk->s->period_size = frame_size * period_frames;
+
+ sk->s->write_size = write_size;
+ pg = get_order(write_size);
+
+ sk->write_page =
+ (void *)__get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO, pg);
+ if (!sk->write_page) {
+ snd_printk(KERN_WARNING "couldn't __get_free_pages()\n");
+ usb_stream_free(sk);
+ return NULL;
+ }
+
+ /* calculate the frequency in 16.16 format */
+ if (dev->speed == USB_SPEED_FULL)
+ sk->freqn = get_usb_full_speed_rate(sample_rate);
+ else
+ sk->freqn = get_usb_high_speed_rate(sample_rate);
+
+ init_urbs(sk, use_packsize, dev, in_pipe, out_pipe);
+ sk->s->state = usb_stream_stopped;
+out:
+ return sk->s;
+}
+
+
+/* start */
+
+static bool balance_check(struct usb_stream_kernel *sk, struct urb *urb)
+{
+ bool r;
+ if (unlikely(urb->status)) {
+ if (urb->status != -ESHUTDOWN && urb->status != -ENOENT)
+ snd_printk(KERN_WARNING "status=%i\n", urb->status);
+ sk->iso_frame_balance = 0x7FFFFFFF;
+ return false;
+ }
+ r = sk->iso_frame_balance == 0;
+ if (!r)
+ sk->i_urb = urb;
+ return r;
+}
+
+static bool balance_playback(struct usb_stream_kernel *sk, struct urb *urb)
+{
+ sk->iso_frame_balance += urb->number_of_packets;
+ return balance_check(sk, urb);
+}
+
+static bool balance_capture(struct usb_stream_kernel *sk, struct urb *urb)
+{
+ sk->iso_frame_balance -= urb->number_of_packets;
+ return balance_check(sk, urb);
+}
+
+static void subs_set_complete(struct urb **urbs, void (*complete)(struct urb *))
+{
+ int u;
+
+ for (u = 0; u < USB_STREAM_NURBS; u++) {
+ struct urb *urb = urbs[u];
+ urb->complete = complete;
+ }
+}
+
+int usb_stream_prepare_playback(struct usb_stream_kernel *sk, struct urb *inurb)
+{
+ struct usb_stream *s = sk->s;
+ struct urb *io;
+ struct usb_iso_packet_descriptor *id, *od;
+ int p, l = 0;
+
+ io = sk->idle_outurb;
+ od = io->iso_frame_desc;
+ io->transfer_buffer_length = 0;
+
+ for (p = 0; s->sync_packet < 0; ++p, ++s->sync_packet) {
+ struct urb *ii = sk->completed_inurb;
+ id = ii->iso_frame_desc +
+ ii->number_of_packets + s->sync_packet;
+ l = id->actual_length;
+
+ od[p].length = l;
+ od[p].offset = io->transfer_buffer_length;
+ io->transfer_buffer_length += l;
+ }
+
+ for (;
+ s->sync_packet < inurb->number_of_packets && p < sk->n_o_ps;
+ ++p, ++s->sync_packet) {
+ l = inurb->iso_frame_desc[s->sync_packet].actual_length;
+
+ if (s->idle_outsize + io->transfer_buffer_length + l >
+ s->period_size)
+ goto check_ok;
+
+ od[p].length = l;
+ od[p].offset = io->transfer_buffer_length;
+ io->transfer_buffer_length += l;
+ }
+
+check_ok:
+ s->sync_packet -= inurb->number_of_packets;
+ if (s->sync_packet < -2 || s->sync_packet > 0) {
+ snd_printk(KERN_WARNING "invalid sync_packet = %i;"
+ " p=%i nop=%i %i %x %x %x > %x\n",
+ s->sync_packet, p, inurb->number_of_packets,
+ s->idle_outsize + io->transfer_buffer_length + l,
+ s->idle_outsize, io->transfer_buffer_length, l,
+ s->period_size);
+ return -1;
+ }
+ if (io->transfer_buffer_length % s->cfg.frame_size) {
+ snd_printk(KERN_WARNING"invalid outsize = %i\n",
+ io->transfer_buffer_length);
+ return -1;
+ }
+ s->idle_outsize += io->transfer_buffer_length - s->period_size;
+ io->number_of_packets = p;
+ if (s->idle_outsize > 0) {
+ snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize);
+ return -1;
+ }
+ return 0;
+}
+
+static void prepare_inurb(int number_of_packets, struct urb *iu)
+{
+ struct usb_iso_packet_descriptor *id;
+ int p;
+
+ iu->number_of_packets = number_of_packets;
+ id = iu->iso_frame_desc;
+ id->offset = 0;
+ for (p = 0; p < iu->number_of_packets - 1; ++p)
+ id[p + 1].offset = id[p].offset + id[p].length;
+
+ iu->transfer_buffer_length =
+ id[0].length * iu->number_of_packets;
+}
+
+static int submit_urbs(struct usb_stream_kernel *sk,
+ struct urb *inurb, struct urb *outurb)
+{
+ int err;
+ prepare_inurb(sk->idle_outurb->number_of_packets, sk->idle_inurb);
+ err = usb_submit_urb(sk->idle_inurb, GFP_ATOMIC);
+ if (err < 0) {
+ snd_printk(KERN_ERR "%i\n", err);
+ return err;
+ }
+ sk->idle_inurb = sk->completed_inurb;
+ sk->completed_inurb = inurb;
+ err = usb_submit_urb(sk->idle_outurb, GFP_ATOMIC);
+ if (err < 0) {
+ snd_printk(KERN_ERR "%i\n", err);
+ return err;
+ }
+ sk->idle_outurb = sk->completed_outurb;
+ sk->completed_outurb = outurb;
+ return 0;
+}
+
+#ifdef DEBUG_LOOP_BACK
+/*
+ This loop_back() shows how to read/write the period data.
+ */
+static void loop_back(struct usb_stream *s)
+{
+ char *i, *o;
+ int il, ol, l, p;
+ struct urb *iu;
+ struct usb_iso_packet_descriptor *id;
+
+ o = s->playback1st_to;
+ ol = s->playback1st_size;
+ l = 0;
+
+ if (s->insplit_pack >= 0) {
+ iu = sk->idle_inurb;
+ id = iu->iso_frame_desc;
+ p = s->insplit_pack;
+ } else
+ goto second;
+loop:
+ for (; p < iu->number_of_packets && l < s->period_size; ++p) {
+ i = iu->transfer_buffer + id[p].offset;
+ il = id[p].actual_length;
+ if (l + il > s->period_size)
+ il = s->period_size - l;
+ if (il <= ol) {
+ memcpy(o, i, il);
+ o += il;
+ ol -= il;
+ } else {
+ memcpy(o, i, ol);
+ singen_6pack(o, ol);
+ o = s->playback_to;
+ memcpy(o, i + ol, il - ol);
+ o += il - ol;
+ ol = s->period_size - s->playback1st_size;
+ }
+ l += il;
+ }
+ if (iu == sk->completed_inurb) {
+ if (l != s->period_size)
+ printk(KERN_DEBUG"%s:%i %i\n", __func__, __LINE__,
+ l/(int)s->cfg.frame_size);
+
+ return;
+ }
+second:
+ iu = sk->completed_inurb;
+ id = iu->iso_frame_desc;
+ p = 0;
+ goto loop;
+
+}
+#else
+static void loop_back(struct usb_stream *s)
+{
+}
+#endif
+
+static void stream_idle(struct usb_stream_kernel *sk,
+ struct urb *inurb, struct urb *outurb)
+{
+ struct usb_stream *s = sk->s;
+ int l, p;
+ int insize = s->idle_insize;
+ int urb_size = 0;
+
+ s->inpacket_split = s->next_inpacket_split;
+ s->inpacket_split_at = s->next_inpacket_split_at;
+ s->next_inpacket_split = -1;
+ s->next_inpacket_split_at = 0;
+
+ for (p = 0; p < inurb->number_of_packets; ++p) {
+ struct usb_iso_packet_descriptor *id = inurb->iso_frame_desc;
+ l = id[p].actual_length;
+ if (unlikely(l == 0 || id[p].status)) {
+ snd_printk(KERN_WARNING "underrun, status=%u\n",
+ id[p].status);
+ goto err_out;
+ }
+ s->inpacket_head++;
+ s->inpacket_head %= s->inpackets;
+ if (s->inpacket_split == -1)
+ s->inpacket_split = s->inpacket_head;
+
+ s->inpacket[s->inpacket_head].offset =
+ id[p].offset + (inurb->transfer_buffer - (void *)s);
+ s->inpacket[s->inpacket_head].length = l;
+ if (insize + l > s->period_size &&
+ s->next_inpacket_split == -1) {
+ s->next_inpacket_split = s->inpacket_head;
+ s->next_inpacket_split_at = s->period_size - insize;
+ }
+ insize += l;
+ urb_size += l;
+ }
+ s->idle_insize += urb_size - s->period_size;
+ if (s->idle_insize < 0) {
+ snd_printk(KERN_WARNING "%i\n",
+ (s->idle_insize)/(int)s->cfg.frame_size);
+ goto err_out;
+ }
+ s->insize_done += urb_size;
+
+ l = s->idle_outsize;
+ s->outpacket[0].offset = (sk->idle_outurb->transfer_buffer -
+ sk->write_page) - l;
+
+ if (usb_stream_prepare_playback(sk, inurb) < 0)
+ goto err_out;
+
+ s->outpacket[0].length = sk->idle_outurb->transfer_buffer_length + l;
+ s->outpacket[1].offset = sk->completed_outurb->transfer_buffer -
+ sk->write_page;
+
+ if (submit_urbs(sk, inurb, outurb) < 0)
+ goto err_out;
+
+ loop_back(s);
+ s->periods_done++;
+ wake_up_all(&sk->sleep);
+ return;
+err_out:
+ s->state = usb_stream_xrun;
+ wake_up_all(&sk->sleep);
+}
+
+static void i_capture_idle(struct urb *urb)
+{
+ struct usb_stream_kernel *sk = urb->context;
+ if (balance_capture(sk, urb))
+ stream_idle(sk, urb, sk->i_urb);
+}
+
+static void i_playback_idle(struct urb *urb)
+{
+ struct usb_stream_kernel *sk = urb->context;
+ if (balance_playback(sk, urb))
+ stream_idle(sk, sk->i_urb, urb);
+}
+
+static void stream_start(struct usb_stream_kernel *sk,
+ struct urb *inurb, struct urb *outurb)
+{
+ struct usb_stream *s = sk->s;
+ if (s->state >= usb_stream_sync1) {
+ int l, p, max_diff, max_diff_0;
+ int urb_size = 0;
+ unsigned frames_per_packet, min_frames = 0;
+ frames_per_packet = (s->period_size - s->idle_insize);
+ frames_per_packet <<= 8;
+ frames_per_packet /=
+ s->cfg.frame_size * inurb->number_of_packets;
+ frames_per_packet++;
+
+ max_diff_0 = s->cfg.frame_size;
+ if (s->cfg.period_frames >= 256)
+ max_diff_0 <<= 1;
+ if (s->cfg.period_frames >= 1024)
+ max_diff_0 <<= 1;
+ max_diff = max_diff_0;
+ for (p = 0; p < inurb->number_of_packets; ++p) {
+ int diff;
+ l = inurb->iso_frame_desc[p].actual_length;
+ urb_size += l;
+
+ min_frames += frames_per_packet;
+ diff = urb_size -
+ (min_frames >> 8) * s->cfg.frame_size;
+ if (diff < max_diff) {
+ snd_printdd(KERN_DEBUG "%i %i %i %i\n",
+ s->insize_done,
+ urb_size / (int)s->cfg.frame_size,
+ inurb->number_of_packets, diff);
+ max_diff = diff;
+ }
+ }
+ s->idle_insize -= max_diff - max_diff_0;
+ s->idle_insize += urb_size - s->period_size;
+ if (s->idle_insize < 0) {
+ snd_printk("%i %i %i\n",
+ s->idle_insize, urb_size, s->period_size);
+ return;
+ } else if (s->idle_insize == 0) {
+ s->next_inpacket_split =
+ (s->inpacket_head + 1) % s->inpackets;
+ s->next_inpacket_split_at = 0;
+ } else {
+ unsigned split = s->inpacket_head;
+ l = s->idle_insize;
+ while (l > s->inpacket[split].length) {
+ l -= s->inpacket[split].length;
+ if (split == 0)
+ split = s->inpackets - 1;
+ else
+ split--;
+ }
+ s->next_inpacket_split = split;
+ s->next_inpacket_split_at =
+ s->inpacket[split].length - l;
+ }
+
+ s->insize_done += urb_size;
+
+ if (usb_stream_prepare_playback(sk, inurb) < 0)
+ return;
+
+ } else
+ playback_prep_freqn(sk, sk->idle_outurb);
+
+ if (submit_urbs(sk, inurb, outurb) < 0)
+ return;
+
+ if (s->state == usb_stream_sync1 && s->insize_done > 360000) {
+ /* just guesswork ^^^^^^ */
+ s->state = usb_stream_ready;
+ subs_set_complete(sk->inurb, i_capture_idle);
+ subs_set_complete(sk->outurb, i_playback_idle);
+ }
+}
+
+static void i_capture_start(struct urb *urb)
+{
+ struct usb_iso_packet_descriptor *id = urb->iso_frame_desc;
+ struct usb_stream_kernel *sk = urb->context;
+ struct usb_stream *s = sk->s;
+ int p;
+ int empty = 0;
+
+ if (urb->status) {
+ snd_printk(KERN_WARNING "status=%i\n", urb->status);
+ return;
+ }
+
+ for (p = 0; p < urb->number_of_packets; ++p) {
+ int l = id[p].actual_length;
+ if (l < s->cfg.frame_size) {
+ ++empty;
+ if (s->state >= usb_stream_sync0) {
+ snd_printk(KERN_WARNING "%i\n", l);
+ return;
+ }
+ }
+ s->inpacket_head++;
+ s->inpacket_head %= s->inpackets;
+ s->inpacket[s->inpacket_head].offset =
+ id[p].offset + (urb->transfer_buffer - (void *)s);
+ s->inpacket[s->inpacket_head].length = l;
+ }
+#ifdef SHOW_EMPTY
+ if (empty) {
+ printk(KERN_DEBUG"%s:%i: %i", __func__, __LINE__,
+ urb->iso_frame_desc[0].actual_length);
+ for (pack = 1; pack < urb->number_of_packets; ++pack) {
+ int l = urb->iso_frame_desc[pack].actual_length;
+ printk(" %i", l);
+ }
+ printk("\n");
+ }
+#endif
+ if (!empty && s->state < usb_stream_sync1)
+ ++s->state;
+
+ if (balance_capture(sk, urb))
+ stream_start(sk, urb, sk->i_urb);
+}
+
+static void i_playback_start(struct urb *urb)
+{
+ struct usb_stream_kernel *sk = urb->context;
+ if (balance_playback(sk, urb))
+ stream_start(sk, sk->i_urb, urb);
+}
+
+int usb_stream_start(struct usb_stream_kernel *sk)
+{
+ struct usb_stream *s = sk->s;
+ int frame = 0, iters = 0;
+ int u, err;
+ int try = 0;
+
+ if (s->state != usb_stream_stopped)
+ return -EAGAIN;
+
+ subs_set_complete(sk->inurb, i_capture_start);
+ subs_set_complete(sk->outurb, i_playback_start);
+ memset(sk->write_page, 0, s->write_size);
+dotry:
+ s->insize_done = 0;
+ s->idle_insize = 0;
+ s->idle_outsize = 0;
+ s->sync_packet = -1;
+ s->inpacket_head = -1;
+ sk->iso_frame_balance = 0;
+ ++try;
+ for (u = 0; u < 2; u++) {
+ struct urb *inurb = sk->inurb[u];
+ struct urb *outurb = sk->outurb[u];
+ playback_prep_freqn(sk, outurb);
+ inurb->number_of_packets = outurb->number_of_packets;
+ inurb->transfer_buffer_length =
+ inurb->number_of_packets *
+ inurb->iso_frame_desc[0].length;
+ preempt_disable();
+ if (u == 0) {
+ int now;
+ struct usb_device *dev = inurb->dev;
+ frame = usb_get_current_frame_number(dev);
+ do {
+ now = usb_get_current_frame_number(dev);
+ ++iters;
+ } while (now > -1 && now == frame);
+ }
+ err = usb_submit_urb(inurb, GFP_ATOMIC);
+ if (err < 0) {
+ preempt_enable();
+ snd_printk(KERN_ERR"usb_submit_urb(sk->inurb[%i])"
+ " returned %i\n", u, err);
+ return err;
+ }
+ err = usb_submit_urb(outurb, GFP_ATOMIC);
+ if (err < 0) {
+ preempt_enable();
+ snd_printk(KERN_ERR"usb_submit_urb(sk->outurb[%i])"
+ " returned %i\n", u, err);
+ return err;
+ }
+ preempt_enable();
+ if (inurb->start_frame != outurb->start_frame) {
+ snd_printd(KERN_DEBUG
+ "u[%i] start_frames differ in:%u out:%u\n",
+ u, inurb->start_frame, outurb->start_frame);
+ goto check_retry;
+ }
+ }
+ snd_printdd(KERN_DEBUG "%i %i\n", frame, iters);
+ try = 0;
+check_retry:
+ if (try) {
+ usb_stream_stop(sk);
+ if (try < 5) {
+ msleep(1500);
+ snd_printd(KERN_DEBUG "goto dotry;\n");
+ goto dotry;
+ }
+ snd_printk(KERN_WARNING"couldn't start"
+ " all urbs on the same start_frame.\n");
+ return -EFAULT;
+ }
+
+ sk->idle_inurb = sk->inurb[USB_STREAM_NURBS - 2];
+ sk->idle_outurb = sk->outurb[USB_STREAM_NURBS - 2];
+ sk->completed_inurb = sk->inurb[USB_STREAM_NURBS - 1];
+ sk->completed_outurb = sk->outurb[USB_STREAM_NURBS - 1];
+
+/* wait, check */
+ {
+ int wait_ms = 3000;
+ while (s->state != usb_stream_ready && wait_ms > 0) {
+ snd_printdd(KERN_DEBUG "%i\n", s->state);
+ msleep(200);
+ wait_ms -= 200;
+ }
+ }
+
+ return s->state == usb_stream_ready ? 0 : -EFAULT;
+}
+
+
+/* stop */
+
+void usb_stream_stop(struct usb_stream_kernel *sk)
+{
+ int u;
+ if (!sk->s)
+ return;
+ for (u = 0; u < USB_STREAM_NURBS; ++u) {
+ usb_kill_urb(sk->inurb[u]);
+ usb_kill_urb(sk->outurb[u]);
+ }
+ sk->s->state = usb_stream_stopped;
+ msleep(400);
+}
diff --git a/sound/usb/usx2y/usb_stream.h b/sound/usb/usx2y/usb_stream.h
new file mode 100644
index 0000000..4dd74ab
--- /dev/null
+++ b/sound/usb/usx2y/usb_stream.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (C) 2007, 2008 Karsten Wiese <fzu@wemgehoertderstaat.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY
+ * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
+ * for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#define USB_STREAM_INTERFACE_VERSION 2
+
+#define SNDRV_USB_STREAM_IOCTL_SET_PARAMS \
+ _IOW('H', 0x90, struct usb_stream_config)
+
+struct usb_stream_packet {
+ unsigned offset;
+ unsigned length;
+};
+
+
+struct usb_stream_config {
+ unsigned version;
+ unsigned sample_rate;
+ unsigned period_frames;
+ unsigned frame_size;
+};
+
+struct usb_stream {
+ struct usb_stream_config cfg;
+ unsigned read_size;
+ unsigned write_size;
+
+ int period_size;
+
+ unsigned state;
+
+ int idle_insize;
+ int idle_outsize;
+ int sync_packet;
+ unsigned insize_done;
+ unsigned periods_done;
+ unsigned periods_polled;
+
+ struct usb_stream_packet outpacket[2];
+ unsigned inpackets;
+ unsigned inpacket_head;
+ unsigned inpacket_split;
+ unsigned inpacket_split_at;
+ unsigned next_inpacket_split;
+ unsigned next_inpacket_split_at;
+ struct usb_stream_packet inpacket[0];
+};
+
+enum usb_stream_state {
+ usb_stream_invalid,
+ usb_stream_stopped,
+ usb_stream_sync0,
+ usb_stream_sync1,
+ usb_stream_ready,
+ usb_stream_running,
+ usb_stream_xrun,
+};
+
+#if __KERNEL__
+
+#define USB_STREAM_NURBS 4
+#define USB_STREAM_URBDEPTH 4
+
+struct usb_stream_kernel {
+ struct usb_stream *s;
+
+ void *write_page;
+
+ unsigned n_o_ps;
+
+ struct urb *inurb[USB_STREAM_NURBS];
+ struct urb *idle_inurb;
+ struct urb *completed_inurb;
+ struct urb *outurb[USB_STREAM_NURBS];
+ struct urb *idle_outurb;
+ struct urb *completed_outurb;
+ struct urb *i_urb;
+
+ int iso_frame_balance;
+
+ wait_queue_head_t sleep;
+
+ unsigned out_phase;
+ unsigned out_phase_peeked;
+ unsigned freqn;
+};
+
+struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk,
+ struct usb_device *dev,
+ unsigned in_endpoint, unsigned out_endpoint,
+ unsigned sample_rate, unsigned use_packsize,
+ unsigned period_frames, unsigned frame_size);
+void usb_stream_free(struct usb_stream_kernel *);
+int usb_stream_start(struct usb_stream_kernel *);
+void usb_stream_stop(struct usb_stream_kernel *);
+
+
+#endif
diff --git a/sound/usb/usx2y/usbus428ctldefs.h b/sound/usb/usx2y/usbus428ctldefs.h
new file mode 100644
index 0000000..b864e7e
--- /dev/null
+++ b/sound/usb/usx2y/usbus428ctldefs.h
@@ -0,0 +1,104 @@
+/*
+ *
+ * Copyright (c) 2003 by Karsten Wiese <annabellesgarden@yahoo.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+enum E_In84{
+ eFader0 = 0,
+ eFader1,
+ eFader2,
+ eFader3,
+ eFader4,
+ eFader5,
+ eFader6,
+ eFader7,
+ eFaderM,
+ eTransport,
+ eModifier = 10,
+ eFilterSelect,
+ eSelect,
+ eMute,
+
+ eSwitch = 15,
+ eWheelGain,
+ eWheelFreq,
+ eWheelQ,
+ eWheelPan,
+ eWheel = 20
+};
+
+#define T_RECORD 1
+#define T_PLAY 2
+#define T_STOP 4
+#define T_F_FWD 8
+#define T_REW 0x10
+#define T_SOLO 0x20
+#define T_REC 0x40
+#define T_NULL 0x80
+
+
+struct us428_ctls {
+ unsigned char Fader[9];
+ unsigned char Transport;
+ unsigned char Modifier;
+ unsigned char FilterSelect;
+ unsigned char Select;
+ unsigned char Mute;
+ unsigned char UNKNOWN;
+ unsigned char Switch;
+ unsigned char Wheel[5];
+};
+
+struct us428_setByte {
+ unsigned char Offset,
+ Value;
+};
+
+enum {
+ eLT_Volume = 0,
+ eLT_Light
+};
+
+struct usX2Y_volume {
+ unsigned char Channel,
+ LH,
+ LL,
+ RH,
+ RL;
+};
+
+struct us428_lights {
+ struct us428_setByte Light[7];
+};
+
+struct us428_p4out {
+ char type;
+ union {
+ struct usX2Y_volume vol;
+ struct us428_lights lights;
+ } val;
+};
+
+#define N_us428_ctl_BUFS 16
+#define N_us428_p4out_BUFS 16
+struct us428ctls_sharedmem{
+ struct us428_ctls CtlSnapShot[N_us428_ctl_BUFS];
+ int CtlSnapShotDiffersAt[N_us428_ctl_BUFS];
+ int CtlSnapShotLast, CtlSnapShotRed;
+ struct us428_p4out p4out[N_us428_p4out_BUFS];
+ int p4outLast, p4outSent;
+};
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
new file mode 100644
index 0000000..e5981a6
--- /dev/null
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -0,0 +1,460 @@
+/*
+ * usbusy2y.c - ALSA USB US-428 Driver
+ *
+2005-04-14 Karsten Wiese
+ Version 0.8.7.2:
+ Call snd_card_free() instead of snd_card_free_in_thread() to prevent oops with dead keyboard symptom.
+ Tested ok with kernel 2.6.12-rc2.
+
+2004-12-14 Karsten Wiese
+ Version 0.8.7.1:
+ snd_pcm_open for rawusb pcm-devices now returns -EBUSY if called without rawusb's hwdep device being open.
+
+2004-12-02 Karsten Wiese
+ Version 0.8.7:
+ Use macro usb_maxpacket() for portability.
+
+2004-10-26 Karsten Wiese
+ Version 0.8.6:
+ wake_up() process waiting in usX2Y_urbs_start() on error.
+
+2004-10-21 Karsten Wiese
+ Version 0.8.5:
+ nrpacks is runtime or compiletime configurable now with tested values from 1 to 4.
+
+2004-10-03 Karsten Wiese
+ Version 0.8.2:
+ Avoid any possible racing while in prepare callback.
+
+2004-09-30 Karsten Wiese
+ Version 0.8.0:
+ Simplified things and made ohci work again.
+
+2004-09-20 Karsten Wiese
+ Version 0.7.3:
+ Use usb_kill_urb() instead of deprecated (kernel 2.6.9) usb_unlink_urb().
+
+2004-07-13 Karsten Wiese
+ Version 0.7.1:
+ Don't sleep in START/STOP callbacks anymore.
+ us428 channels C/D not handled just for this version, sorry.
+
+2004-06-21 Karsten Wiese
+ Version 0.6.4:
+ Temporarely suspend midi input
+ to sanely call usb_set_interface() when setting format.
+
+2004-06-12 Karsten Wiese
+ Version 0.6.3:
+ Made it thus the following rule is enforced:
+ "All pcm substreams of one usX2Y have to operate at the same rate & format."
+
+2004-04-06 Karsten Wiese
+ Version 0.6.0:
+ Runs on 2.6.5 kernel without any "--with-debug=" things.
+ us224 reported running.
+
+2004-01-14 Karsten Wiese
+ Version 0.5.1:
+ Runs with 2.6.1 kernel.
+
+2003-12-30 Karsten Wiese
+ Version 0.4.1:
+ Fix 24Bit 4Channel capturing for the us428.
+
+2003-11-27 Karsten Wiese, Martin Langer
+ Version 0.4:
+ us122 support.
+ us224 could be tested by uncommenting the sections containing USB_ID_US224
+
+2003-11-03 Karsten Wiese
+ Version 0.3:
+ 24Bit support.
+ "arecord -D hw:1 -c 2 -r 48000 -M -f S24_3LE|aplay -D hw:1 -c 2 -r 48000 -M -f S24_3LE" works.
+
+2003-08-22 Karsten Wiese
+ Version 0.0.8:
+ Removed EZUSB Firmware. First Stage Firmwaredownload is now done by tascam-firmware downloader.
+ See:
+ http://usb-midi-fw.sourceforge.net/tascam-firmware.tar.gz
+
+2003-06-18 Karsten Wiese
+ Version 0.0.5:
+ changed to compile with kernel 2.4.21 and alsa 0.9.4
+
+2002-10-16 Karsten Wiese
+ Version 0.0.4:
+ compiles again with alsa-current.
+ USB_ISO_ASAP not used anymore (most of the time), instead
+ urb->start_frame is calculated here now, some calls inside usb-driver don't need to happen anymore.
+
+ To get the best out of this:
+ Disable APM-support in the kernel as APM-BIOS calls (once each second) hard disable interrupt for many precious milliseconds.
+ This helped me much on my slowish PII 400 & PIII 500.
+ ACPI yet untested but might cause the same bad behaviour.
+ Use a kernel with lowlatency and preemptiv patches applied.
+ To autoload snd-usb-midi append a line
+ post-install snd-usb-us428 modprobe snd-usb-midi
+ to /etc/modules.conf.
+
+ known problems:
+ sliders, knobs, lights not yet handled except MASTER Volume slider.
+ "pcm -c 2" doesn't work. "pcm -c 2 -m direct_interleaved" does.
+ KDE3: "Enable full duplex operation" deadlocks.
+
+
+2002-08-31 Karsten Wiese
+ Version 0.0.3: audio also simplex;
+ simplifying: iso urbs only 1 packet, melted structs.
+ ASYNC_UNLINK not used anymore: no more crashes so far.....
+ for alsa 0.9 rc3.
+
+2002-08-09 Karsten Wiese
+ Version 0.0.2: midi works with snd-usb-midi, audio (only fullduplex now) with i.e. bristol.
+ The firmware has been sniffed from win2k us-428 driver 3.09.
+
+ * Copyright (c) 2002 - 2004 Karsten Wiese
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*/
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/usb.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+
+#include <sound/rawmidi.h>
+#include "usx2y.h"
+#include "usbusx2y.h"
+#include "usX2Yhwdep.h"
+
+
+
+MODULE_AUTHOR("Karsten Wiese <annabellesgarden@yahoo.de>");
+MODULE_DESCRIPTION("TASCAM "NAME_ALLCAPS" Version 0.8.7.2");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{TASCAM(0x1604), "NAME_ALLCAPS"(0x8001)(0x8005)(0x8007) }}");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
+static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for "NAME_ALLCAPS".");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for "NAME_ALLCAPS".");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable "NAME_ALLCAPS".");
+
+
+static int snd_usX2Y_card_used[SNDRV_CARDS];
+
+static void usX2Y_usb_disconnect(struct usb_device* usb_device, void* ptr);
+static void snd_usX2Y_card_private_free(struct snd_card *card);
+
+/*
+ * pipe 4 is used for switching the lamps, setting samplerate, volumes ....
+ */
+static void i_usX2Y_Out04Int(struct urb *urb)
+{
+#ifdef CONFIG_SND_DEBUG
+ if (urb->status) {
+ int i;
+ struct usX2Ydev *usX2Y = urb->context;
+ for (i = 0; i < 10 && usX2Y->AS04.urb[i] != urb; i++);
+ snd_printdd("i_usX2Y_Out04Int() urb %i status=%i\n", i, urb->status);
+ }
+#endif
+}
+
+static void i_usX2Y_In04Int(struct urb *urb)
+{
+ int err = 0;
+ struct usX2Ydev *usX2Y = urb->context;
+ struct us428ctls_sharedmem *us428ctls = usX2Y->us428ctls_sharedmem;
+
+ usX2Y->In04IntCalls++;
+
+ if (urb->status) {
+ snd_printdd("Interrupt Pipe 4 came back with status=%i\n", urb->status);
+ return;
+ }
+
+ // printk("%i:0x%02X ", 8, (int)((unsigned char*)usX2Y->In04Buf)[8]); Master volume shows 0 here if fader is at max during boot ?!?
+ if (us428ctls) {
+ int diff = -1;
+ if (-2 == us428ctls->CtlSnapShotLast) {
+ diff = 0;
+ memcpy(usX2Y->In04Last, usX2Y->In04Buf, sizeof(usX2Y->In04Last));
+ us428ctls->CtlSnapShotLast = -1;
+ } else {
+ int i;
+ for (i = 0; i < 21; i++) {
+ if (usX2Y->In04Last[i] != ((char*)usX2Y->In04Buf)[i]) {
+ if (diff < 0)
+ diff = i;
+ usX2Y->In04Last[i] = ((char*)usX2Y->In04Buf)[i];
+ }
+ }
+ }
+ if (0 <= diff) {
+ int n = us428ctls->CtlSnapShotLast + 1;
+ if (n >= N_us428_ctl_BUFS || n < 0)
+ n = 0;
+ memcpy(us428ctls->CtlSnapShot + n, usX2Y->In04Buf, sizeof(us428ctls->CtlSnapShot[0]));
+ us428ctls->CtlSnapShotDiffersAt[n] = diff;
+ us428ctls->CtlSnapShotLast = n;
+ wake_up(&usX2Y->us428ctls_wait_queue_head);
+ }
+ }
+
+
+ if (usX2Y->US04) {
+ if (0 == usX2Y->US04->submitted)
+ do
+ err = usb_submit_urb(usX2Y->US04->urb[usX2Y->US04->submitted++], GFP_ATOMIC);
+ while (!err && usX2Y->US04->submitted < usX2Y->US04->len);
+ } else
+ if (us428ctls && us428ctls->p4outLast >= 0 && us428ctls->p4outLast < N_us428_p4out_BUFS) {
+ if (us428ctls->p4outLast != us428ctls->p4outSent) {
+ int j, send = us428ctls->p4outSent + 1;
+ if (send >= N_us428_p4out_BUFS)
+ send = 0;
+ for (j = 0; j < URBS_AsyncSeq && !err; ++j)
+ if (0 == usX2Y->AS04.urb[j]->status) {
+ struct us428_p4out *p4out = us428ctls->p4out + send; // FIXME if more then 1 p4out is new, 1 gets lost.
+ usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->chip.dev,
+ usb_sndbulkpipe(usX2Y->chip.dev, 0x04), &p4out->val.vol,
+ p4out->type == eLT_Light ? sizeof(struct us428_lights) : 5,
+ i_usX2Y_Out04Int, usX2Y);
+ err = usb_submit_urb(usX2Y->AS04.urb[j], GFP_ATOMIC);
+ us428ctls->p4outSent = send;
+ break;
+ }
+ }
+ }
+
+ if (err)
+ snd_printk(KERN_ERR "In04Int() usb_submit_urb err=%i\n", err);
+
+ urb->dev = usX2Y->chip.dev;
+ usb_submit_urb(urb, GFP_ATOMIC);
+}
+
+/*
+ * Prepare some urbs
+ */
+int usX2Y_AsyncSeq04_init(struct usX2Ydev *usX2Y)
+{
+ int err = 0,
+ i;
+
+ if (NULL == (usX2Y->AS04.buffer = kmalloc(URB_DataLen_AsyncSeq*URBS_AsyncSeq, GFP_KERNEL))) {
+ err = -ENOMEM;
+ } else
+ for (i = 0; i < URBS_AsyncSeq; ++i) {
+ if (NULL == (usX2Y->AS04.urb[i] = usb_alloc_urb(0, GFP_KERNEL))) {
+ err = -ENOMEM;
+ break;
+ }
+ usb_fill_bulk_urb( usX2Y->AS04.urb[i], usX2Y->chip.dev,
+ usb_sndbulkpipe(usX2Y->chip.dev, 0x04),
+ usX2Y->AS04.buffer + URB_DataLen_AsyncSeq*i, 0,
+ i_usX2Y_Out04Int, usX2Y
+ );
+ }
+ return err;
+}
+
+int usX2Y_In04_init(struct usX2Ydev *usX2Y)
+{
+ if (! (usX2Y->In04urb = usb_alloc_urb(0, GFP_KERNEL)))
+ return -ENOMEM;
+
+ if (! (usX2Y->In04Buf = kmalloc(21, GFP_KERNEL))) {
+ usb_free_urb(usX2Y->In04urb);
+ return -ENOMEM;
+ }
+
+ init_waitqueue_head(&usX2Y->In04WaitQueue);
+ usb_fill_int_urb(usX2Y->In04urb, usX2Y->chip.dev, usb_rcvintpipe(usX2Y->chip.dev, 0x4),
+ usX2Y->In04Buf, 21,
+ i_usX2Y_In04Int, usX2Y,
+ 10);
+ return usb_submit_urb(usX2Y->In04urb, GFP_KERNEL);
+}
+
+static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S)
+{
+ int i;
+ for (i = 0; i < URBS_AsyncSeq; ++i) {
+ if (S[i].urb) {
+ usb_kill_urb(S->urb[i]);
+ usb_free_urb(S->urb[i]);
+ S->urb[i] = NULL;
+ }
+ }
+ kfree(S->buffer);
+}
+
+
+static struct usb_device_id snd_usX2Y_usb_id_table[] = {
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x1604,
+ .idProduct = USB_ID_US428
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x1604,
+ .idProduct = USB_ID_US122
+ },
+ {
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x1604,
+ .idProduct = USB_ID_US224
+ },
+ { /* terminator */ }
+};
+
+static struct snd_card *usX2Y_create_card(struct usb_device *device)
+{
+ int dev;
+ struct snd_card * card;
+ for (dev = 0; dev < SNDRV_CARDS; ++dev)
+ if (enable[dev] && !snd_usX2Y_card_used[dev])
+ break;
+ if (dev >= SNDRV_CARDS)
+ return NULL;
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct usX2Ydev));
+ if (!card)
+ return NULL;
+ snd_usX2Y_card_used[usX2Y(card)->chip.index = dev] = 1;
+ card->private_free = snd_usX2Y_card_private_free;
+ usX2Y(card)->chip.dev = device;
+ usX2Y(card)->chip.card = card;
+ init_waitqueue_head(&usX2Y(card)->prepare_wait_queue);
+ mutex_init(&usX2Y(card)->prepare_mutex);
+ INIT_LIST_HEAD(&usX2Y(card)->chip.midi_list);
+ strcpy(card->driver, "USB "NAME_ALLCAPS"");
+ sprintf(card->shortname, "TASCAM "NAME_ALLCAPS"");
+ sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)",
+ card->shortname,
+ le16_to_cpu(device->descriptor.idVendor),
+ le16_to_cpu(device->descriptor.idProduct),
+ 0,//us428(card)->usbmidi.ifnum,
+ usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum
+ );
+ snd_card_set_dev(card, &device->dev);
+ return card;
+}
+
+
+static void *usX2Y_usb_probe(struct usb_device *device, struct usb_interface *intf, const struct usb_device_id *device_id)
+{
+ int err;
+ struct snd_card * card;
+ if (le16_to_cpu(device->descriptor.idVendor) != 0x1604 ||
+ (le16_to_cpu(device->descriptor.idProduct) != USB_ID_US122 &&
+ le16_to_cpu(device->descriptor.idProduct) != USB_ID_US224 &&
+ le16_to_cpu(device->descriptor.idProduct) != USB_ID_US428) ||
+ !(card = usX2Y_create_card(device)))
+ return NULL;
+ if ((err = usX2Y_hwdep_new(card, device)) < 0 ||
+ (err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return NULL;
+ }
+ return card;
+}
+
+/*
+ * new 2.5 USB kernel API
+ */
+static int snd_usX2Y_probe(struct usb_interface *intf, const struct usb_device_id *id)
+{
+ void *chip;
+ chip = usX2Y_usb_probe(interface_to_usbdev(intf), intf, id);
+ if (chip) {
+ dev_set_drvdata(&intf->dev, chip);
+ return 0;
+ } else
+ return -EIO;
+}
+
+static void snd_usX2Y_disconnect(struct usb_interface *intf)
+{
+ usX2Y_usb_disconnect(interface_to_usbdev(intf),
+ dev_get_drvdata(&intf->dev));
+}
+
+MODULE_DEVICE_TABLE(usb, snd_usX2Y_usb_id_table);
+static struct usb_driver snd_usX2Y_usb_driver = {
+ .name = "snd-usb-usx2y",
+ .probe = snd_usX2Y_probe,
+ .disconnect = snd_usX2Y_disconnect,
+ .id_table = snd_usX2Y_usb_id_table,
+};
+
+static void snd_usX2Y_card_private_free(struct snd_card *card)
+{
+ kfree(usX2Y(card)->In04Buf);
+ usb_free_urb(usX2Y(card)->In04urb);
+ if (usX2Y(card)->us428ctls_sharedmem)
+ snd_free_pages(usX2Y(card)->us428ctls_sharedmem, sizeof(*usX2Y(card)->us428ctls_sharedmem));
+ if (usX2Y(card)->chip.index >= 0 && usX2Y(card)->chip.index < SNDRV_CARDS)
+ snd_usX2Y_card_used[usX2Y(card)->chip.index] = 0;
+}
+
+/*
+ * Frees the device.
+ */
+static void usX2Y_usb_disconnect(struct usb_device *device, void* ptr)
+{
+ if (ptr) {
+ struct snd_card *card = ptr;
+ struct usX2Ydev *usX2Y = usX2Y(card);
+ struct list_head *p;
+ usX2Y->chip.shutdown = 1;
+ usX2Y->chip_status = USX2Y_STAT_CHIP_HUP;
+ usX2Y_unlinkSeq(&usX2Y->AS04);
+ usb_kill_urb(usX2Y->In04urb);
+ snd_card_disconnect(card);
+ /* release the midi resources */
+ list_for_each(p, &usX2Y->chip.midi_list) {
+ snd_usbmidi_disconnect(p);
+ }
+ if (usX2Y->us428ctls_sharedmem)
+ wake_up(&usX2Y->us428ctls_wait_queue_head);
+ snd_card_free(card);
+ }
+}
+
+static int __init snd_usX2Y_module_init(void)
+{
+ return usb_register(&snd_usX2Y_usb_driver);
+}
+
+static void __exit snd_usX2Y_module_exit(void)
+{
+ usb_deregister(&snd_usX2Y_usb_driver);
+}
+
+module_init(snd_usX2Y_module_init)
+module_exit(snd_usX2Y_module_exit)
diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h
new file mode 100644
index 0000000..456b5fd
--- /dev/null
+++ b/sound/usb/usx2y/usbusx2y.h
@@ -0,0 +1,83 @@
+#ifndef USBUSX2Y_H
+#define USBUSX2Y_H
+#include "../usbaudio.h"
+#include "usbus428ctldefs.h"
+
+#define NRURBS 2
+
+
+#define URBS_AsyncSeq 10
+#define URB_DataLen_AsyncSeq 32
+struct snd_usX2Y_AsyncSeq {
+ struct urb *urb[URBS_AsyncSeq];
+ char *buffer;
+};
+
+struct snd_usX2Y_urbSeq {
+ int submitted;
+ int len;
+ struct urb *urb[0];
+};
+
+#include "usx2yhwdeppcm.h"
+
+struct usX2Ydev {
+ struct snd_usb_audio chip;
+ int stride;
+ struct urb *In04urb;
+ void *In04Buf;
+ char In04Last[24];
+ unsigned In04IntCalls;
+ struct snd_usX2Y_urbSeq *US04;
+ wait_queue_head_t In04WaitQueue;
+ struct snd_usX2Y_AsyncSeq AS04;
+ unsigned int rate,
+ format;
+ int chip_status;
+ struct mutex prepare_mutex;
+ struct us428ctls_sharedmem *us428ctls_sharedmem;
+ int wait_iso_frame;
+ wait_queue_head_t us428ctls_wait_queue_head;
+ struct snd_usX2Y_hwdep_pcm_shm *hwdep_pcm_shm;
+ struct snd_usX2Y_substream *subs[4];
+ struct snd_usX2Y_substream * volatile prepare_subs;
+ wait_queue_head_t prepare_wait_queue;
+};
+
+
+struct snd_usX2Y_substream {
+ struct usX2Ydev *usX2Y;
+ struct snd_pcm_substream *pcm_substream;
+
+ int endpoint;
+ unsigned int maxpacksize; /* max packet size in bytes */
+
+ atomic_t state;
+#define state_STOPPED 0
+#define state_STARTING1 1
+#define state_STARTING2 2
+#define state_STARTING3 3
+#define state_PREPARED 4
+#define state_PRERUNNING 6
+#define state_RUNNING 8
+
+ int hwptr; /* free frame position in the buffer (only for playback) */
+ int hwptr_done; /* processed frame position in the buffer */
+ int transfer_done; /* processed frames since last period update */
+
+ struct urb *urb[NRURBS]; /* data urb table */
+ struct urb *completed_urb;
+ char *tmpbuf; /* temporary buffer for playback */
+};
+
+
+#define usX2Y(c) ((struct usX2Ydev *)(c)->private_data)
+
+int usX2Y_audio_create(struct snd_card *card);
+
+int usX2Y_AsyncSeq04_init(struct usX2Ydev *usX2Y);
+int usX2Y_In04_init(struct usX2Ydev *usX2Y);
+
+#define NAME_ALLCAPS "US-X2Y"
+
+#endif
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
new file mode 100644
index 0000000..9a608fa
--- /dev/null
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -0,0 +1,1024 @@
+/*
+ * US-X2Y AUDIO
+ * Copyright (c) 2002-2004 by Karsten Wiese
+ *
+ * based on
+ *
+ * (Tentative) USB Audio Driver for ALSA
+ *
+ * Main and PCM part
+ *
+ * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de>
+ *
+ * Many codes borrowed from audio.c by
+ * Alan Cox (alan@lxorguk.ukuu.org.uk)
+ * Thomas Sailer (sailer@ife.ee.ethz.ch)
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+
+#include <linux/interrupt.h>
+#include <linux/usb.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "usx2y.h"
+#include "usbusx2y.h"
+
+#define USX2Y_NRPACKS 4 /* Default value used for nr of packs per urb.
+ 1 to 4 have been tested ok on uhci.
+ To use 3 on ohci, you'd need a patch:
+ look for "0000425-linux-2.6.9-rc4-mm1_ohci-hcd.patch.gz" on
+ "https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=0000425"
+ .
+ 1, 2 and 4 work out of the box on ohci, if I recall correctly.
+ Bigger is safer operation,
+ smaller gives lower latencies.
+ */
+#define USX2Y_NRPACKS_VARIABLE y /* If your system works ok with this module's parameter
+ nrpacks set to 1, you might as well comment
+ this #define out, and thereby produce smaller, faster code.
+ You'd also set USX2Y_NRPACKS to 1 then.
+ */
+
+#ifdef USX2Y_NRPACKS_VARIABLE
+ static int nrpacks = USX2Y_NRPACKS; /* number of packets per urb */
+ #define nr_of_packs() nrpacks
+ module_param(nrpacks, int, 0444);
+ MODULE_PARM_DESC(nrpacks, "Number of packets per URB.");
+#else
+ #define nr_of_packs() USX2Y_NRPACKS
+#endif
+
+
+static int usX2Y_urb_capt_retire(struct snd_usX2Y_substream *subs)
+{
+ struct urb *urb = subs->completed_urb;
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ unsigned char *cp;
+ int i, len, lens = 0, hwptr_done = subs->hwptr_done;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+
+ for (i = 0; i < nr_of_packs(); i++) {
+ cp = (unsigned char*)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
+ if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */
+ snd_printk(KERN_ERR "active frame status %i. "
+ "Most propably some hardware problem.\n",
+ urb->iso_frame_desc[i].status);
+ return urb->iso_frame_desc[i].status;
+ }
+ len = urb->iso_frame_desc[i].actual_length / usX2Y->stride;
+ if (! len) {
+ snd_printd("0 == len ERROR!\n");
+ continue;
+ }
+
+ /* copy a data chunk */
+ if ((hwptr_done + len) > runtime->buffer_size) {
+ int cnt = runtime->buffer_size - hwptr_done;
+ int blen = cnt * usX2Y->stride;
+ memcpy(runtime->dma_area + hwptr_done * usX2Y->stride, cp, blen);
+ memcpy(runtime->dma_area, cp + blen, len * usX2Y->stride - blen);
+ } else {
+ memcpy(runtime->dma_area + hwptr_done * usX2Y->stride, cp,
+ len * usX2Y->stride);
+ }
+ lens += len;
+ if ((hwptr_done += len) >= runtime->buffer_size)
+ hwptr_done -= runtime->buffer_size;
+ }
+
+ subs->hwptr_done = hwptr_done;
+ subs->transfer_done += lens;
+ /* update the pointer, call callback if necessary */
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ snd_pcm_period_elapsed(subs->pcm_substream);
+ }
+ return 0;
+}
+/*
+ * prepare urb for playback data pipe
+ *
+ * we copy the data directly from the pcm buffer.
+ * the current position to be copied is held in hwptr field.
+ * since a urb can handle only a single linear buffer, if the total
+ * transferred area overflows the buffer boundary, we cannot send
+ * it directly from the buffer. thus the data is once copied to
+ * a temporary buffer and urb points to that.
+ */
+static int usX2Y_urb_play_prepare(struct snd_usX2Y_substream *subs,
+ struct urb *cap_urb,
+ struct urb *urb)
+{
+ int count, counts, pack;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+
+ count = 0;
+ for (pack = 0; pack < nr_of_packs(); pack++) {
+ /* calculate the size of a packet */
+ counts = cap_urb->iso_frame_desc[pack].actual_length / usX2Y->stride;
+ count += counts;
+ if (counts < 43 || counts > 50) {
+ snd_printk(KERN_ERR "should not be here with counts=%i\n", counts);
+ return -EPIPE;
+ }
+ /* set up descriptor */
+ urb->iso_frame_desc[pack].offset = pack ?
+ urb->iso_frame_desc[pack - 1].offset +
+ urb->iso_frame_desc[pack - 1].length :
+ 0;
+ urb->iso_frame_desc[pack].length = cap_urb->iso_frame_desc[pack].actual_length;
+ }
+ if (atomic_read(&subs->state) >= state_PRERUNNING)
+ if (subs->hwptr + count > runtime->buffer_size) {
+ /* err, the transferred area goes over buffer boundary.
+ * copy the data to the temp buffer.
+ */
+ int len;
+ len = runtime->buffer_size - subs->hwptr;
+ urb->transfer_buffer = subs->tmpbuf;
+ memcpy(subs->tmpbuf, runtime->dma_area +
+ subs->hwptr * usX2Y->stride, len * usX2Y->stride);
+ memcpy(subs->tmpbuf + len * usX2Y->stride,
+ runtime->dma_area, (count - len) * usX2Y->stride);
+ subs->hwptr += count;
+ subs->hwptr -= runtime->buffer_size;
+ } else {
+ /* set the buffer pointer */
+ urb->transfer_buffer = runtime->dma_area + subs->hwptr * usX2Y->stride;
+ if ((subs->hwptr += count) >= runtime->buffer_size)
+ subs->hwptr -= runtime->buffer_size;
+ }
+ else
+ urb->transfer_buffer = subs->tmpbuf;
+ urb->transfer_buffer_length = count * usX2Y->stride;
+ return 0;
+}
+
+/*
+ * process after playback data complete
+ *
+ * update the current position and call callback if a period is processed.
+ */
+static void usX2Y_urb_play_retire(struct snd_usX2Y_substream *subs, struct urb *urb)
+{
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ int len = urb->actual_length / subs->usX2Y->stride;
+
+ subs->transfer_done += len;
+ subs->hwptr_done += len;
+ if (subs->hwptr_done >= runtime->buffer_size)
+ subs->hwptr_done -= runtime->buffer_size;
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ snd_pcm_period_elapsed(subs->pcm_substream);
+ }
+}
+
+static int usX2Y_urb_submit(struct snd_usX2Y_substream *subs, struct urb *urb, int frame)
+{
+ int err;
+ if (!urb)
+ return -ENODEV;
+ urb->start_frame = (frame + NRURBS * nr_of_packs()); // let hcd do rollover sanity checks
+ urb->hcpriv = NULL;
+ urb->dev = subs->usX2Y->chip.dev; /* we need to set this at each time */
+ if ((err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+ snd_printk(KERN_ERR "usb_submit_urb() returned %i\n", err);
+ return err;
+ }
+ return 0;
+}
+
+static inline int usX2Y_usbframe_complete(struct snd_usX2Y_substream *capsubs,
+ struct snd_usX2Y_substream *playbacksubs,
+ int frame)
+{
+ int err, state;
+ struct urb *urb = playbacksubs->completed_urb;
+
+ state = atomic_read(&playbacksubs->state);
+ if (NULL != urb) {
+ if (state == state_RUNNING)
+ usX2Y_urb_play_retire(playbacksubs, urb);
+ else if (state >= state_PRERUNNING)
+ atomic_inc(&playbacksubs->state);
+ } else {
+ switch (state) {
+ case state_STARTING1:
+ urb = playbacksubs->urb[0];
+ atomic_inc(&playbacksubs->state);
+ break;
+ case state_STARTING2:
+ urb = playbacksubs->urb[1];
+ atomic_inc(&playbacksubs->state);
+ break;
+ }
+ }
+ if (urb) {
+ if ((err = usX2Y_urb_play_prepare(playbacksubs, capsubs->completed_urb, urb)) ||
+ (err = usX2Y_urb_submit(playbacksubs, urb, frame))) {
+ return err;
+ }
+ }
+
+ playbacksubs->completed_urb = NULL;
+
+ state = atomic_read(&capsubs->state);
+ if (state >= state_PREPARED) {
+ if (state == state_RUNNING) {
+ if ((err = usX2Y_urb_capt_retire(capsubs)))
+ return err;
+ } else if (state >= state_PRERUNNING)
+ atomic_inc(&capsubs->state);
+ if ((err = usX2Y_urb_submit(capsubs, capsubs->completed_urb, frame)))
+ return err;
+ }
+ capsubs->completed_urb = NULL;
+ return 0;
+}
+
+
+static void usX2Y_clients_stop(struct usX2Ydev *usX2Y)
+{
+ int s, u;
+
+ for (s = 0; s < 4; s++) {
+ struct snd_usX2Y_substream *subs = usX2Y->subs[s];
+ if (subs) {
+ snd_printdd("%i %p state=%i\n", s, subs, atomic_read(&subs->state));
+ atomic_set(&subs->state, state_STOPPED);
+ }
+ }
+ for (s = 0; s < 4; s++) {
+ struct snd_usX2Y_substream *subs = usX2Y->subs[s];
+ if (subs) {
+ if (atomic_read(&subs->state) >= state_PRERUNNING) {
+ snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
+ }
+ for (u = 0; u < NRURBS; u++) {
+ struct urb *urb = subs->urb[u];
+ if (NULL != urb)
+ snd_printdd("%i status=%i start_frame=%i\n",
+ u, urb->status, urb->start_frame);
+ }
+ }
+ }
+ usX2Y->prepare_subs = NULL;
+ wake_up(&usX2Y->prepare_wait_queue);
+}
+
+static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y,
+ struct snd_usX2Y_substream *subs, struct urb *urb)
+{
+ snd_printk(KERN_ERR "ep=%i stalled with status=%i\n", subs->endpoint, urb->status);
+ urb->status = 0;
+ usX2Y_clients_stop(usX2Y);
+}
+
+static void usX2Y_error_sequence(struct usX2Ydev *usX2Y,
+ struct snd_usX2Y_substream *subs, struct urb *urb)
+{
+ snd_printk(KERN_ERR "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n"
+ KERN_ERR "Most propably some urb of usb-frame %i is still missing.\n"
+ KERN_ERR "Cause could be too long delays in usb-hcd interrupt handling.\n",
+ usb_get_current_frame_number(usX2Y->chip.dev),
+ subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out",
+ usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame);
+ usX2Y_clients_stop(usX2Y);
+}
+
+static void i_usX2Y_urb_complete(struct urb *urb)
+{
+ struct snd_usX2Y_substream *subs = urb->context;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+
+ if (unlikely(atomic_read(&subs->state) < state_PREPARED)) {
+ snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n",
+ usb_get_current_frame_number(usX2Y->chip.dev),
+ subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out",
+ urb->status, urb->start_frame);
+ return;
+ }
+ if (unlikely(urb->status)) {
+ usX2Y_error_urb_status(usX2Y, subs, urb);
+ return;
+ }
+ if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF)))
+ subs->completed_urb = urb;
+ else {
+ usX2Y_error_sequence(usX2Y, subs, urb);
+ return;
+ }
+ {
+ struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE],
+ *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
+ if (capsubs->completed_urb &&
+ atomic_read(&capsubs->state) >= state_PREPARED &&
+ (playbacksubs->completed_urb ||
+ atomic_read(&playbacksubs->state) < state_PREPARED)) {
+ if (!usX2Y_usbframe_complete(capsubs, playbacksubs, urb->start_frame))
+ usX2Y->wait_iso_frame += nr_of_packs();
+ else {
+ snd_printdd("\n");
+ usX2Y_clients_stop(usX2Y);
+ }
+ }
+ }
+}
+
+static void usX2Y_urbs_set_complete(struct usX2Ydev * usX2Y,
+ void (*complete)(struct urb *))
+{
+ int s, u;
+ for (s = 0; s < 4; s++) {
+ struct snd_usX2Y_substream *subs = usX2Y->subs[s];
+ if (NULL != subs)
+ for (u = 0; u < NRURBS; u++) {
+ struct urb * urb = subs->urb[u];
+ if (NULL != urb)
+ urb->complete = complete;
+ }
+ }
+}
+
+static void usX2Y_subs_startup_finish(struct usX2Ydev * usX2Y)
+{
+ usX2Y_urbs_set_complete(usX2Y, i_usX2Y_urb_complete);
+ usX2Y->prepare_subs = NULL;
+}
+
+static void i_usX2Y_subs_startup(struct urb *urb)
+{
+ struct snd_usX2Y_substream *subs = urb->context;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ struct snd_usX2Y_substream *prepare_subs = usX2Y->prepare_subs;
+ if (NULL != prepare_subs)
+ if (urb->start_frame == prepare_subs->urb[0]->start_frame) {
+ usX2Y_subs_startup_finish(usX2Y);
+ atomic_inc(&prepare_subs->state);
+ wake_up(&usX2Y->prepare_wait_queue);
+ }
+
+ i_usX2Y_urb_complete(urb);
+}
+
+static void usX2Y_subs_prepare(struct snd_usX2Y_substream *subs)
+{
+ snd_printdd("usX2Y_substream_prepare(%p) ep=%i urb0=%p urb1=%p\n",
+ subs, subs->endpoint, subs->urb[0], subs->urb[1]);
+ /* reset the pointer */
+ subs->hwptr = 0;
+ subs->hwptr_done = 0;
+ subs->transfer_done = 0;
+}
+
+
+static void usX2Y_urb_release(struct urb **urb, int free_tb)
+{
+ if (*urb) {
+ usb_kill_urb(*urb);
+ if (free_tb)
+ kfree((*urb)->transfer_buffer);
+ usb_free_urb(*urb);
+ *urb = NULL;
+ }
+}
+/*
+ * release a substreams urbs
+ */
+static void usX2Y_urbs_release(struct snd_usX2Y_substream *subs)
+{
+ int i;
+ snd_printdd("usX2Y_urbs_release() %i\n", subs->endpoint);
+ for (i = 0; i < NRURBS; i++)
+ usX2Y_urb_release(subs->urb + i,
+ subs != subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]);
+
+ kfree(subs->tmpbuf);
+ subs->tmpbuf = NULL;
+}
+/*
+ * initialize a substream's urbs
+ */
+static int usX2Y_urbs_allocate(struct snd_usX2Y_substream *subs)
+{
+ int i;
+ unsigned int pipe;
+ int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
+ struct usb_device *dev = subs->usX2Y->chip.dev;
+
+ pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) :
+ usb_rcvisocpipe(dev, subs->endpoint);
+ subs->maxpacksize = usb_maxpacket(dev, pipe, is_playback);
+ if (!subs->maxpacksize)
+ return -EINVAL;
+
+ if (is_playback && NULL == subs->tmpbuf) { /* allocate a temporary buffer for playback */
+ subs->tmpbuf = kcalloc(nr_of_packs(), subs->maxpacksize, GFP_KERNEL);
+ if (NULL == subs->tmpbuf) {
+ snd_printk(KERN_ERR "cannot malloc tmpbuf\n");
+ return -ENOMEM;
+ }
+ }
+ /* allocate and initialize data urbs */
+ for (i = 0; i < NRURBS; i++) {
+ struct urb **purb = subs->urb + i;
+ if (*purb) {
+ usb_kill_urb(*purb);
+ continue;
+ }
+ *purb = usb_alloc_urb(nr_of_packs(), GFP_KERNEL);
+ if (NULL == *purb) {
+ usX2Y_urbs_release(subs);
+ return -ENOMEM;
+ }
+ if (!is_playback && !(*purb)->transfer_buffer) {
+ /* allocate a capture buffer per urb */
+ (*purb)->transfer_buffer = kmalloc(subs->maxpacksize * nr_of_packs(), GFP_KERNEL);
+ if (NULL == (*purb)->transfer_buffer) {
+ usX2Y_urbs_release(subs);
+ return -ENOMEM;
+ }
+ }
+ (*purb)->dev = dev;
+ (*purb)->pipe = pipe;
+ (*purb)->number_of_packets = nr_of_packs();
+ (*purb)->context = subs;
+ (*purb)->interval = 1;
+ (*purb)->complete = i_usX2Y_subs_startup;
+ }
+ return 0;
+}
+
+static void usX2Y_subs_startup(struct snd_usX2Y_substream *subs)
+{
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ usX2Y->prepare_subs = subs;
+ subs->urb[0]->start_frame = -1;
+ wmb();
+ usX2Y_urbs_set_complete(usX2Y, i_usX2Y_subs_startup);
+}
+
+static int usX2Y_urbs_start(struct snd_usX2Y_substream *subs)
+{
+ int i, err;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+
+ if ((err = usX2Y_urbs_allocate(subs)) < 0)
+ return err;
+ subs->completed_urb = NULL;
+ for (i = 0; i < 4; i++) {
+ struct snd_usX2Y_substream *subs = usX2Y->subs[i];
+ if (subs != NULL && atomic_read(&subs->state) >= state_PREPARED)
+ goto start;
+ }
+
+ start:
+ usX2Y_subs_startup(subs);
+ for (i = 0; i < NRURBS; i++) {
+ struct urb *urb = subs->urb[i];
+ if (usb_pipein(urb->pipe)) {
+ unsigned long pack;
+ if (0 == i)
+ atomic_set(&subs->state, state_STARTING3);
+ urb->dev = usX2Y->chip.dev;
+ urb->transfer_flags = URB_ISO_ASAP;
+ for (pack = 0; pack < nr_of_packs(); pack++) {
+ urb->iso_frame_desc[pack].offset = subs->maxpacksize * pack;
+ urb->iso_frame_desc[pack].length = subs->maxpacksize;
+ }
+ urb->transfer_buffer_length = subs->maxpacksize * nr_of_packs();
+ if ((err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
+ snd_printk (KERN_ERR "cannot submit datapipe for urb %d, err = %d\n", i, err);
+ err = -EPIPE;
+ goto cleanup;
+ } else
+ if (i == 0)
+ usX2Y->wait_iso_frame = urb->start_frame;
+ urb->transfer_flags = 0;
+ } else {
+ atomic_set(&subs->state, state_STARTING1);
+ break;
+ }
+ }
+ err = 0;
+ wait_event(usX2Y->prepare_wait_queue, NULL == usX2Y->prepare_subs);
+ if (atomic_read(&subs->state) != state_PREPARED)
+ err = -EPIPE;
+
+ cleanup:
+ if (err) {
+ usX2Y_subs_startup_finish(usX2Y);
+ usX2Y_clients_stop(usX2Y); // something is completely wroong > stop evrything
+ }
+ return err;
+}
+
+/*
+ * return the current pcm pointer. just return the hwptr_done value.
+ */
+static snd_pcm_uframes_t snd_usX2Y_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_usX2Y_substream *subs = substream->runtime->private_data;
+ return subs->hwptr_done;
+}
+/*
+ * start/stop substream
+ */
+static int snd_usX2Y_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_usX2Y_substream *subs = substream->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_printdd("snd_usX2Y_pcm_trigger(START)\n");
+ if (atomic_read(&subs->state) == state_PREPARED &&
+ atomic_read(&subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]->state) >= state_PREPARED) {
+ atomic_set(&subs->state, state_PRERUNNING);
+ } else {
+ snd_printdd("\n");
+ return -EPIPE;
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_printdd("snd_usX2Y_pcm_trigger(STOP)\n");
+ if (atomic_read(&subs->state) >= state_PRERUNNING)
+ atomic_set(&subs->state, state_PREPARED);
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+/*
+ * allocate a buffer, setup samplerate
+ *
+ * so far we use a physically linear buffer although packetize transfer
+ * doesn't need a continuous area.
+ * if sg buffer is supported on the later version of alsa, we'll follow
+ * that.
+ */
+static struct s_c2
+{
+ char c1, c2;
+}
+ SetRate44100[] =
+{
+ { 0x14, 0x08}, // this line sets 44100, well actually a little less
+ { 0x18, 0x40}, // only tascam / frontier design knows the further lines .......
+ { 0x18, 0x42},
+ { 0x18, 0x45},
+ { 0x18, 0x46},
+ { 0x18, 0x48},
+ { 0x18, 0x4A},
+ { 0x18, 0x4C},
+ { 0x18, 0x4E},
+ { 0x18, 0x50},
+ { 0x18, 0x52},
+ { 0x18, 0x54},
+ { 0x18, 0x56},
+ { 0x18, 0x58},
+ { 0x18, 0x5A},
+ { 0x18, 0x5C},
+ { 0x18, 0x5E},
+ { 0x18, 0x60},
+ { 0x18, 0x62},
+ { 0x18, 0x64},
+ { 0x18, 0x66},
+ { 0x18, 0x68},
+ { 0x18, 0x6A},
+ { 0x18, 0x6C},
+ { 0x18, 0x6E},
+ { 0x18, 0x70},
+ { 0x18, 0x72},
+ { 0x18, 0x74},
+ { 0x18, 0x76},
+ { 0x18, 0x78},
+ { 0x18, 0x7A},
+ { 0x18, 0x7C},
+ { 0x18, 0x7E}
+};
+static struct s_c2 SetRate48000[] =
+{
+ { 0x14, 0x09}, // this line sets 48000, well actually a little less
+ { 0x18, 0x40}, // only tascam / frontier design knows the further lines .......
+ { 0x18, 0x42},
+ { 0x18, 0x45},
+ { 0x18, 0x46},
+ { 0x18, 0x48},
+ { 0x18, 0x4A},
+ { 0x18, 0x4C},
+ { 0x18, 0x4E},
+ { 0x18, 0x50},
+ { 0x18, 0x52},
+ { 0x18, 0x54},
+ { 0x18, 0x56},
+ { 0x18, 0x58},
+ { 0x18, 0x5A},
+ { 0x18, 0x5C},
+ { 0x18, 0x5E},
+ { 0x18, 0x60},
+ { 0x18, 0x62},
+ { 0x18, 0x64},
+ { 0x18, 0x66},
+ { 0x18, 0x68},
+ { 0x18, 0x6A},
+ { 0x18, 0x6C},
+ { 0x18, 0x6E},
+ { 0x18, 0x70},
+ { 0x18, 0x73},
+ { 0x18, 0x74},
+ { 0x18, 0x76},
+ { 0x18, 0x78},
+ { 0x18, 0x7A},
+ { 0x18, 0x7C},
+ { 0x18, 0x7E}
+};
+#define NOOF_SETRATE_URBS ARRAY_SIZE(SetRate48000)
+
+static void i_usX2Y_04Int(struct urb *urb)
+{
+ struct usX2Ydev *usX2Y = urb->context;
+
+ if (urb->status)
+ snd_printk(KERN_ERR "snd_usX2Y_04Int() urb->status=%i\n", urb->status);
+ if (0 == --usX2Y->US04->len)
+ wake_up(&usX2Y->In04WaitQueue);
+}
+
+static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate)
+{
+ int err = 0, i;
+ struct snd_usX2Y_urbSeq *us = NULL;
+ int *usbdata = NULL;
+ struct s_c2 *ra = rate == 48000 ? SetRate48000 : SetRate44100;
+
+ if (usX2Y->rate != rate) {
+ us = kzalloc(sizeof(*us) + sizeof(struct urb*) * NOOF_SETRATE_URBS, GFP_KERNEL);
+ if (NULL == us) {
+ err = -ENOMEM;
+ goto cleanup;
+ }
+ usbdata = kmalloc(sizeof(int) * NOOF_SETRATE_URBS, GFP_KERNEL);
+ if (NULL == usbdata) {
+ err = -ENOMEM;
+ goto cleanup;
+ }
+ for (i = 0; i < NOOF_SETRATE_URBS; ++i) {
+ if (NULL == (us->urb[i] = usb_alloc_urb(0, GFP_KERNEL))) {
+ err = -ENOMEM;
+ goto cleanup;
+ }
+ ((char*)(usbdata + i))[0] = ra[i].c1;
+ ((char*)(usbdata + i))[1] = ra[i].c2;
+ usb_fill_bulk_urb(us->urb[i], usX2Y->chip.dev, usb_sndbulkpipe(usX2Y->chip.dev, 4),
+ usbdata + i, 2, i_usX2Y_04Int, usX2Y);
+#ifdef OLD_USB
+ us->urb[i]->transfer_flags = USB_QUEUE_BULK;
+#endif
+ }
+ us->submitted = 0;
+ us->len = NOOF_SETRATE_URBS;
+ usX2Y->US04 = us;
+ wait_event_timeout(usX2Y->In04WaitQueue, 0 == us->len, HZ);
+ usX2Y->US04 = NULL;
+ if (us->len)
+ err = -ENODEV;
+ cleanup:
+ if (us) {
+ us->submitted = 2*NOOF_SETRATE_URBS;
+ for (i = 0; i < NOOF_SETRATE_URBS; ++i) {
+ struct urb *urb = us->urb[i];
+ if (urb->status) {
+ if (!err)
+ err = -ENODEV;
+ usb_kill_urb(urb);
+ }
+ usb_free_urb(urb);
+ }
+ usX2Y->US04 = NULL;
+ kfree(usbdata);
+ kfree(us);
+ if (!err)
+ usX2Y->rate = rate;
+ }
+ }
+
+ return err;
+}
+
+
+static int usX2Y_format_set(struct usX2Ydev *usX2Y, snd_pcm_format_t format)
+{
+ int alternate, err;
+ struct list_head* p;
+ if (format == SNDRV_PCM_FORMAT_S24_3LE) {
+ alternate = 2;
+ usX2Y->stride = 6;
+ } else {
+ alternate = 1;
+ usX2Y->stride = 4;
+ }
+ list_for_each(p, &usX2Y->chip.midi_list) {
+ snd_usbmidi_input_stop(p);
+ }
+ usb_kill_urb(usX2Y->In04urb);
+ if ((err = usb_set_interface(usX2Y->chip.dev, 0, alternate))) {
+ snd_printk(KERN_ERR "usb_set_interface error \n");
+ return err;
+ }
+ usX2Y->In04urb->dev = usX2Y->chip.dev;
+ err = usb_submit_urb(usX2Y->In04urb, GFP_KERNEL);
+ list_for_each(p, &usX2Y->chip.midi_list) {
+ snd_usbmidi_input_start(p);
+ }
+ usX2Y->format = format;
+ usX2Y->rate = 0;
+ return err;
+}
+
+
+static int snd_usX2Y_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ int err = 0;
+ unsigned int rate = params_rate(hw_params);
+ snd_pcm_format_t format = params_format(hw_params);
+ struct snd_card *card = substream->pstr->pcm->card;
+ struct list_head *list;
+
+ snd_printdd("snd_usX2Y_hw_params(%p, %p)\n", substream, hw_params);
+ // all pcm substreams off one usX2Y have to operate at the same rate & format
+ list_for_each(list, &card->devices) {
+ struct snd_device *dev;
+ struct snd_pcm *pcm;
+ int s;
+ dev = snd_device(list);
+ if (dev->type != SNDRV_DEV_PCM)
+ continue;
+ pcm = dev->device_data;
+ for (s = 0; s < 2; ++s) {
+ struct snd_pcm_substream *test_substream;
+ test_substream = pcm->streams[s].substream;
+ if (test_substream && test_substream != substream &&
+ test_substream->runtime &&
+ ((test_substream->runtime->format &&
+ test_substream->runtime->format != format) ||
+ (test_substream->runtime->rate &&
+ test_substream->runtime->rate != rate)))
+ return -EINVAL;
+ }
+ }
+ if (0 > (err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)))) {
+ snd_printk(KERN_ERR "snd_pcm_lib_malloc_pages(%p, %i) returned %i\n",
+ substream, params_buffer_bytes(hw_params), err);
+ return err;
+ }
+ return 0;
+}
+
+/*
+ * free the buffer
+ */
+static int snd_usX2Y_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usX2Y_substream *subs = runtime->private_data;
+ mutex_lock(&subs->usX2Y->prepare_mutex);
+ snd_printdd("snd_usX2Y_hw_free(%p)\n", substream);
+
+ if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) {
+ struct snd_usX2Y_substream *cap_subs = subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE];
+ atomic_set(&subs->state, state_STOPPED);
+ usX2Y_urbs_release(subs);
+ if (!cap_subs->pcm_substream ||
+ !cap_subs->pcm_substream->runtime ||
+ !cap_subs->pcm_substream->runtime->status ||
+ cap_subs->pcm_substream->runtime->status->state < SNDRV_PCM_STATE_PREPARED) {
+ atomic_set(&cap_subs->state, state_STOPPED);
+ usX2Y_urbs_release(cap_subs);
+ }
+ } else {
+ struct snd_usX2Y_substream *playback_subs = subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
+ if (atomic_read(&playback_subs->state) < state_PREPARED) {
+ atomic_set(&subs->state, state_STOPPED);
+ usX2Y_urbs_release(subs);
+ }
+ }
+ mutex_unlock(&subs->usX2Y->prepare_mutex);
+ return snd_pcm_lib_free_pages(substream);
+}
+/*
+ * prepare callback
+ *
+ * set format and initialize urbs
+ */
+static int snd_usX2Y_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usX2Y_substream *subs = runtime->private_data;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ struct snd_usX2Y_substream *capsubs = subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE];
+ int err = 0;
+ snd_printdd("snd_usX2Y_pcm_prepare(%p)\n", substream);
+
+ mutex_lock(&usX2Y->prepare_mutex);
+ usX2Y_subs_prepare(subs);
+// Start hardware streams
+// SyncStream first....
+ if (atomic_read(&capsubs->state) < state_PREPARED) {
+ if (usX2Y->format != runtime->format)
+ if ((err = usX2Y_format_set(usX2Y, runtime->format)) < 0)
+ goto up_prepare_mutex;
+ if (usX2Y->rate != runtime->rate)
+ if ((err = usX2Y_rate_set(usX2Y, runtime->rate)) < 0)
+ goto up_prepare_mutex;
+ snd_printdd("starting capture pipe for %s\n", subs == capsubs ? "self" : "playpipe");
+ if (0 > (err = usX2Y_urbs_start(capsubs)))
+ goto up_prepare_mutex;
+ }
+
+ if (subs != capsubs && atomic_read(&subs->state) < state_PREPARED)
+ err = usX2Y_urbs_start(subs);
+
+ up_prepare_mutex:
+ mutex_unlock(&usX2Y->prepare_mutex);
+ return err;
+}
+
+static struct snd_pcm_hardware snd_usX2Y_2c =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (2*128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0
+};
+
+
+
+static int snd_usX2Y_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_usX2Y_substream *subs = ((struct snd_usX2Y_substream **)
+ snd_pcm_substream_chip(substream))[substream->stream];
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (subs->usX2Y->chip_status & USX2Y_STAT_CHIP_MMAP_PCM_URBS)
+ return -EBUSY;
+
+ runtime->hw = snd_usX2Y_2c;
+ runtime->private_data = subs;
+ subs->pcm_substream = substream;
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 1000, 200000);
+ return 0;
+}
+
+
+
+static int snd_usX2Y_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usX2Y_substream *subs = runtime->private_data;
+
+ subs->pcm_substream = NULL;
+
+ return 0;
+}
+
+
+static struct snd_pcm_ops snd_usX2Y_pcm_ops =
+{
+ .open = snd_usX2Y_pcm_open,
+ .close = snd_usX2Y_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_usX2Y_pcm_hw_params,
+ .hw_free = snd_usX2Y_pcm_hw_free,
+ .prepare = snd_usX2Y_pcm_prepare,
+ .trigger = snd_usX2Y_pcm_trigger,
+ .pointer = snd_usX2Y_pcm_pointer,
+};
+
+
+/*
+ * free a usb stream instance
+ */
+static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream)
+{
+ kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
+ usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
+
+ kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]);
+ usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL;
+}
+
+static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm)
+{
+ struct snd_usX2Y_substream **usX2Y_stream = pcm->private_data;
+ if (usX2Y_stream)
+ usX2Y_audio_stream_free(usX2Y_stream);
+}
+
+static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, int capture_endpoint)
+{
+ struct snd_pcm *pcm;
+ int err, i;
+ struct snd_usX2Y_substream **usX2Y_substream =
+ usX2Y(card)->subs + 2 * usX2Y(card)->chip.pcm_devs;
+
+ for (i = playback_endpoint ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+ i <= SNDRV_PCM_STREAM_CAPTURE; ++i) {
+ usX2Y_substream[i] = kzalloc(sizeof(struct snd_usX2Y_substream), GFP_KERNEL);
+ if (NULL == usX2Y_substream[i]) {
+ snd_printk(KERN_ERR "cannot malloc\n");
+ return -ENOMEM;
+ }
+ usX2Y_substream[i]->usX2Y = usX2Y(card);
+ }
+
+ if (playback_endpoint)
+ usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]->endpoint = playback_endpoint;
+ usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]->endpoint = capture_endpoint;
+
+ err = snd_pcm_new(card, NAME_ALLCAPS" Audio", usX2Y(card)->chip.pcm_devs,
+ playback_endpoint ? 1 : 0, 1,
+ &pcm);
+ if (err < 0) {
+ usX2Y_audio_stream_free(usX2Y_substream);
+ return err;
+ }
+
+ if (playback_endpoint)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usX2Y_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usX2Y_pcm_ops);
+
+ pcm->private_data = usX2Y_substream;
+ pcm->private_free = snd_usX2Y_pcm_private_free;
+ pcm->info_flags = 0;
+
+ sprintf(pcm->name, NAME_ALLCAPS" Audio #%d", usX2Y(card)->chip.pcm_devs);
+
+ if ((playback_endpoint &&
+ 0 > (err = snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 64*1024, 128*1024))) ||
+ 0 > (err = snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 64*1024, 128*1024))) {
+ snd_usX2Y_pcm_private_free(pcm);
+ return err;
+ }
+ usX2Y(card)->chip.pcm_devs++;
+
+ return 0;
+}
+
+/*
+ * create a chip instance and set its names.
+ */
+int usX2Y_audio_create(struct snd_card *card)
+{
+ int err = 0;
+
+ INIT_LIST_HEAD(&usX2Y(card)->chip.pcm_list);
+
+ if (0 > (err = usX2Y_audio_stream_new(card, 0xA, 0x8)))
+ return err;
+ if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) == USB_ID_US428)
+ if (0 > (err = usX2Y_audio_stream_new(card, 0, 0xA)))
+ return err;
+ if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) != USB_ID_US122)
+ err = usX2Y_rate_set(usX2Y(card), 44100); // Lets us428 recognize output-volume settings, disturbs us122.
+ return err;
+}
diff --git a/sound/usb/usx2y/usx2y.h b/sound/usb/usx2y/usx2y.h
new file mode 100644
index 0000000..7e59263
--- /dev/null
+++ b/sound/usb/usx2y/usx2y.h
@@ -0,0 +1,51 @@
+/*
+ * Driver for Tascam US-X2Y USB soundcards
+ *
+ * Copyright (c) 2003 by Karsten Wiese <annabellesgarden@yahoo.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_USX2Y_COMMON_H
+#define __SOUND_USX2Y_COMMON_H
+
+
+#define USX2Y_DRIVER_VERSION 0x0100 /* 0.1.0 */
+
+
+/* hwdep id string */
+#define SND_USX2Y_LOADER_ID "USX2Y Loader"
+#define SND_USX2Y_USBPCM_ID "USX2Y USBPCM"
+
+/* hardware type */
+enum {
+ USX2Y_TYPE_122,
+ USX2Y_TYPE_224,
+ USX2Y_TYPE_428,
+ USX2Y_TYPE_NUMS
+};
+
+#define USB_ID_US122 0x8007
+#define USB_ID_US224 0x8005
+#define USB_ID_US428 0x8001
+
+/* chip status */
+enum {
+ USX2Y_STAT_CHIP_INIT = (1 << 0), /* all operational */
+ USX2Y_STAT_CHIP_MMAP_PCM_URBS = (1 << 1), /* pcm transport over mmaped urbs */
+ USX2Y_STAT_CHIP_HUP = (1 << 31), /* all operational */
+};
+
+#endif /* __SOUND_USX2Y_COMMON_H */
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c
new file mode 100644
index 0000000..117946f
--- /dev/null
+++ b/sound/usb/usx2y/usx2yhwdeppcm.c
@@ -0,0 +1,793 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/* USX2Y "rawusb" aka hwdep_pcm implementation
+
+ Its usb's unableness to atomically handle power of 2 period sized data chuncs
+ at standard samplerates,
+ what led to this part of the usx2y module:
+ It provides the alsa kernel half of the usx2y-alsa-jack driver pair.
+ The pair uses a hardware dependant alsa-device for mmaped pcm transport.
+ Advantage achieved:
+ The usb_hc moves pcm data from/into memory via DMA.
+ That memory is mmaped by jack's usx2y driver.
+ Jack's usx2y driver is the first/last to read/write pcm data.
+ Read/write is a combination of power of 2 period shaping and
+ float/int conversation.
+ Compared to mainline alsa/jack we leave out power of 2 period shaping inside
+ snd-usb-usx2y which needs memcpy() and additional buffers.
+ As a side effect possible unwanted pcm-data coruption resulting of
+ standard alsa's snd-usb-usx2y period shaping scheme falls away.
+ Result is sane jack operation at buffering schemes down to 128frames,
+ 2 periods.
+ plain usx2y alsa mode is able to achieve 64frames, 4periods, but only at the
+ cost of easier triggered i.e. aeolus xruns (128 or 256frames,
+ 2periods works but is useless cause of crackling).
+
+ This is a first "proof of concept" implementation.
+ Later, funcionalities should migrate to more apropriate places:
+ Userland:
+ - The jackd could mmap its float-pcm buffers directly from alsa-lib.
+ - alsa-lib could provide power of 2 period sized shaping combined with int/float
+ conversation.
+ Currently the usx2y jack driver provides above 2 services.
+ Kernel:
+ - rawusb dma pcm buffer transport should go to snd-usb-lib, so also snd-usb-audio
+ devices can use it.
+ Currently rawusb dma pcm buffer transport (this file) is only available to snd-usb-usx2y.
+*/
+
+#include <linux/delay.h>
+#include "usbusx2yaudio.c"
+
+#if defined(USX2Y_NRPACKS_VARIABLE) || (!defined(USX2Y_NRPACKS_VARIABLE) && USX2Y_NRPACKS == 1)
+
+#include <sound/hwdep.h>
+
+
+static int usX2Y_usbpcm_urb_capt_retire(struct snd_usX2Y_substream *subs)
+{
+ struct urb *urb = subs->completed_urb;
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ int i, lens = 0, hwptr_done = subs->hwptr_done;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ if (0 > usX2Y->hwdep_pcm_shm->capture_iso_start) { //FIXME
+ int head = usX2Y->hwdep_pcm_shm->captured_iso_head + 1;
+ if (head >= ARRAY_SIZE(usX2Y->hwdep_pcm_shm->captured_iso))
+ head = 0;
+ usX2Y->hwdep_pcm_shm->capture_iso_start = head;
+ snd_printdd("cap start %i\n", head);
+ }
+ for (i = 0; i < nr_of_packs(); i++) {
+ if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */
+ snd_printk(KERN_ERR "activ frame status %i. Most propably some hardware problem.\n", urb->iso_frame_desc[i].status);
+ return urb->iso_frame_desc[i].status;
+ }
+ lens += urb->iso_frame_desc[i].actual_length / usX2Y->stride;
+ }
+ if ((hwptr_done += lens) >= runtime->buffer_size)
+ hwptr_done -= runtime->buffer_size;
+ subs->hwptr_done = hwptr_done;
+ subs->transfer_done += lens;
+ /* update the pointer, call callback if necessary */
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ snd_pcm_period_elapsed(subs->pcm_substream);
+ }
+ return 0;
+}
+
+static inline int usX2Y_iso_frames_per_buffer(struct snd_pcm_runtime *runtime,
+ struct usX2Ydev * usX2Y)
+{
+ return (runtime->buffer_size * 1000) / usX2Y->rate + 1; //FIXME: so far only correct period_size == 2^x ?
+}
+
+/*
+ * prepare urb for playback data pipe
+ *
+ * we copy the data directly from the pcm buffer.
+ * the current position to be copied is held in hwptr field.
+ * since a urb can handle only a single linear buffer, if the total
+ * transferred area overflows the buffer boundary, we cannot send
+ * it directly from the buffer. thus the data is once copied to
+ * a temporary buffer and urb points to that.
+ */
+static int usX2Y_hwdep_urb_play_prepare(struct snd_usX2Y_substream *subs,
+ struct urb *urb)
+{
+ int count, counts, pack;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ struct snd_usX2Y_hwdep_pcm_shm *shm = usX2Y->hwdep_pcm_shm;
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+
+ if (0 > shm->playback_iso_start) {
+ shm->playback_iso_start = shm->captured_iso_head -
+ usX2Y_iso_frames_per_buffer(runtime, usX2Y);
+ if (0 > shm->playback_iso_start)
+ shm->playback_iso_start += ARRAY_SIZE(shm->captured_iso);
+ shm->playback_iso_head = shm->playback_iso_start;
+ }
+
+ count = 0;
+ for (pack = 0; pack < nr_of_packs(); pack++) {
+ /* calculate the size of a packet */
+ counts = shm->captured_iso[shm->playback_iso_head].length / usX2Y->stride;
+ if (counts < 43 || counts > 50) {
+ snd_printk(KERN_ERR "should not be here with counts=%i\n", counts);
+ return -EPIPE;
+ }
+ /* set up descriptor */
+ urb->iso_frame_desc[pack].offset = shm->captured_iso[shm->playback_iso_head].offset;
+ urb->iso_frame_desc[pack].length = shm->captured_iso[shm->playback_iso_head].length;
+ if (atomic_read(&subs->state) != state_RUNNING)
+ memset((char *)urb->transfer_buffer + urb->iso_frame_desc[pack].offset, 0,
+ urb->iso_frame_desc[pack].length);
+ if (++shm->playback_iso_head >= ARRAY_SIZE(shm->captured_iso))
+ shm->playback_iso_head = 0;
+ count += counts;
+ }
+ urb->transfer_buffer_length = count * usX2Y->stride;
+ return 0;
+}
+
+
+static inline void usX2Y_usbpcm_urb_capt_iso_advance(struct snd_usX2Y_substream *subs,
+ struct urb *urb)
+{
+ int pack;
+ for (pack = 0; pack < nr_of_packs(); ++pack) {
+ struct usb_iso_packet_descriptor *desc = urb->iso_frame_desc + pack;
+ if (NULL != subs) {
+ struct snd_usX2Y_hwdep_pcm_shm *shm = subs->usX2Y->hwdep_pcm_shm;
+ int head = shm->captured_iso_head + 1;
+ if (head >= ARRAY_SIZE(shm->captured_iso))
+ head = 0;
+ shm->captured_iso[head].frame = urb->start_frame + pack;
+ shm->captured_iso[head].offset = desc->offset;
+ shm->captured_iso[head].length = desc->actual_length;
+ shm->captured_iso_head = head;
+ shm->captured_iso_frames++;
+ }
+ if ((desc->offset += desc->length * NRURBS*nr_of_packs()) +
+ desc->length >= SSS)
+ desc->offset -= (SSS - desc->length);
+ }
+}
+
+static inline int usX2Y_usbpcm_usbframe_complete(struct snd_usX2Y_substream *capsubs,
+ struct snd_usX2Y_substream *capsubs2,
+ struct snd_usX2Y_substream *playbacksubs,
+ int frame)
+{
+ int err, state;
+ struct urb *urb = playbacksubs->completed_urb;
+
+ state = atomic_read(&playbacksubs->state);
+ if (NULL != urb) {
+ if (state == state_RUNNING)
+ usX2Y_urb_play_retire(playbacksubs, urb);
+ else if (state >= state_PRERUNNING)
+ atomic_inc(&playbacksubs->state);
+ } else {
+ switch (state) {
+ case state_STARTING1:
+ urb = playbacksubs->urb[0];
+ atomic_inc(&playbacksubs->state);
+ break;
+ case state_STARTING2:
+ urb = playbacksubs->urb[1];
+ atomic_inc(&playbacksubs->state);
+ break;
+ }
+ }
+ if (urb) {
+ if ((err = usX2Y_hwdep_urb_play_prepare(playbacksubs, urb)) ||
+ (err = usX2Y_urb_submit(playbacksubs, urb, frame))) {
+ return err;
+ }
+ }
+
+ playbacksubs->completed_urb = NULL;
+
+ state = atomic_read(&capsubs->state);
+ if (state >= state_PREPARED) {
+ if (state == state_RUNNING) {
+ if ((err = usX2Y_usbpcm_urb_capt_retire(capsubs)))
+ return err;
+ } else if (state >= state_PRERUNNING)
+ atomic_inc(&capsubs->state);
+ usX2Y_usbpcm_urb_capt_iso_advance(capsubs, capsubs->completed_urb);
+ if (NULL != capsubs2)
+ usX2Y_usbpcm_urb_capt_iso_advance(NULL, capsubs2->completed_urb);
+ if ((err = usX2Y_urb_submit(capsubs, capsubs->completed_urb, frame)))
+ return err;
+ if (NULL != capsubs2)
+ if ((err = usX2Y_urb_submit(capsubs2, capsubs2->completed_urb, frame)))
+ return err;
+ }
+ capsubs->completed_urb = NULL;
+ if (NULL != capsubs2)
+ capsubs2->completed_urb = NULL;
+ return 0;
+}
+
+
+static void i_usX2Y_usbpcm_urb_complete(struct urb *urb)
+{
+ struct snd_usX2Y_substream *subs = urb->context;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ struct snd_usX2Y_substream *capsubs, *capsubs2, *playbacksubs;
+
+ if (unlikely(atomic_read(&subs->state) < state_PREPARED)) {
+ snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n",
+ usb_get_current_frame_number(usX2Y->chip.dev),
+ subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out",
+ urb->status, urb->start_frame);
+ return;
+ }
+ if (unlikely(urb->status)) {
+ usX2Y_error_urb_status(usX2Y, subs, urb);
+ return;
+ }
+ if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF)))
+ subs->completed_urb = urb;
+ else {
+ usX2Y_error_sequence(usX2Y, subs, urb);
+ return;
+ }
+
+ capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE];
+ capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2];
+ playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
+ if (capsubs->completed_urb && atomic_read(&capsubs->state) >= state_PREPARED &&
+ (NULL == capsubs2 || capsubs2->completed_urb) &&
+ (playbacksubs->completed_urb || atomic_read(&playbacksubs->state) < state_PREPARED)) {
+ if (!usX2Y_usbpcm_usbframe_complete(capsubs, capsubs2, playbacksubs, urb->start_frame))
+ usX2Y->wait_iso_frame += nr_of_packs();
+ else {
+ snd_printdd("\n");
+ usX2Y_clients_stop(usX2Y);
+ }
+ }
+}
+
+
+static void usX2Y_hwdep_urb_release(struct urb **urb)
+{
+ usb_kill_urb(*urb);
+ usb_free_urb(*urb);
+ *urb = NULL;
+}
+
+/*
+ * release a substream
+ */
+static void usX2Y_usbpcm_urbs_release(struct snd_usX2Y_substream *subs)
+{
+ int i;
+ snd_printdd("snd_usX2Y_urbs_release() %i\n", subs->endpoint);
+ for (i = 0; i < NRURBS; i++)
+ usX2Y_hwdep_urb_release(subs->urb + i);
+}
+
+static void usX2Y_usbpcm_subs_startup_finish(struct usX2Ydev * usX2Y)
+{
+ usX2Y_urbs_set_complete(usX2Y, i_usX2Y_usbpcm_urb_complete);
+ usX2Y->prepare_subs = NULL;
+}
+
+static void i_usX2Y_usbpcm_subs_startup(struct urb *urb)
+{
+ struct snd_usX2Y_substream *subs = urb->context;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ struct snd_usX2Y_substream *prepare_subs = usX2Y->prepare_subs;
+ if (NULL != prepare_subs &&
+ urb->start_frame == prepare_subs->urb[0]->start_frame) {
+ atomic_inc(&prepare_subs->state);
+ if (prepare_subs == usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]) {
+ struct snd_usX2Y_substream *cap_subs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2];
+ if (cap_subs2 != NULL)
+ atomic_inc(&cap_subs2->state);
+ }
+ usX2Y_usbpcm_subs_startup_finish(usX2Y);
+ wake_up(&usX2Y->prepare_wait_queue);
+ }
+
+ i_usX2Y_usbpcm_urb_complete(urb);
+}
+
+/*
+ * initialize a substream's urbs
+ */
+static int usX2Y_usbpcm_urbs_allocate(struct snd_usX2Y_substream *subs)
+{
+ int i;
+ unsigned int pipe;
+ int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
+ struct usb_device *dev = subs->usX2Y->chip.dev;
+
+ pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) :
+ usb_rcvisocpipe(dev, subs->endpoint);
+ subs->maxpacksize = usb_maxpacket(dev, pipe, is_playback);
+ if (!subs->maxpacksize)
+ return -EINVAL;
+
+ /* allocate and initialize data urbs */
+ for (i = 0; i < NRURBS; i++) {
+ struct urb **purb = subs->urb + i;
+ if (*purb) {
+ usb_kill_urb(*purb);
+ continue;
+ }
+ *purb = usb_alloc_urb(nr_of_packs(), GFP_KERNEL);
+ if (NULL == *purb) {
+ usX2Y_usbpcm_urbs_release(subs);
+ return -ENOMEM;
+ }
+ (*purb)->transfer_buffer = is_playback ?
+ subs->usX2Y->hwdep_pcm_shm->playback : (
+ subs->endpoint == 0x8 ?
+ subs->usX2Y->hwdep_pcm_shm->capture0x8 :
+ subs->usX2Y->hwdep_pcm_shm->capture0xA);
+
+ (*purb)->dev = dev;
+ (*purb)->pipe = pipe;
+ (*purb)->number_of_packets = nr_of_packs();
+ (*purb)->context = subs;
+ (*purb)->interval = 1;
+ (*purb)->complete = i_usX2Y_usbpcm_subs_startup;
+ }
+ return 0;
+}
+
+/*
+ * free the buffer
+ */
+static int snd_usX2Y_usbpcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usX2Y_substream *subs = runtime->private_data,
+ *cap_subs2 = subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2];
+ mutex_lock(&subs->usX2Y->prepare_mutex);
+ snd_printdd("snd_usX2Y_usbpcm_hw_free(%p)\n", substream);
+
+ if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) {
+ struct snd_usX2Y_substream *cap_subs = subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE];
+ atomic_set(&subs->state, state_STOPPED);
+ usX2Y_usbpcm_urbs_release(subs);
+ if (!cap_subs->pcm_substream ||
+ !cap_subs->pcm_substream->runtime ||
+ !cap_subs->pcm_substream->runtime->status ||
+ cap_subs->pcm_substream->runtime->status->state < SNDRV_PCM_STATE_PREPARED) {
+ atomic_set(&cap_subs->state, state_STOPPED);
+ if (NULL != cap_subs2)
+ atomic_set(&cap_subs2->state, state_STOPPED);
+ usX2Y_usbpcm_urbs_release(cap_subs);
+ if (NULL != cap_subs2)
+ usX2Y_usbpcm_urbs_release(cap_subs2);
+ }
+ } else {
+ struct snd_usX2Y_substream *playback_subs = subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
+ if (atomic_read(&playback_subs->state) < state_PREPARED) {
+ atomic_set(&subs->state, state_STOPPED);
+ if (NULL != cap_subs2)
+ atomic_set(&cap_subs2->state, state_STOPPED);
+ usX2Y_usbpcm_urbs_release(subs);
+ if (NULL != cap_subs2)
+ usX2Y_usbpcm_urbs_release(cap_subs2);
+ }
+ }
+ mutex_unlock(&subs->usX2Y->prepare_mutex);
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static void usX2Y_usbpcm_subs_startup(struct snd_usX2Y_substream *subs)
+{
+ struct usX2Ydev * usX2Y = subs->usX2Y;
+ usX2Y->prepare_subs = subs;
+ subs->urb[0]->start_frame = -1;
+ smp_wmb(); // Make sure above modifications are seen by i_usX2Y_subs_startup()
+ usX2Y_urbs_set_complete(usX2Y, i_usX2Y_usbpcm_subs_startup);
+}
+
+static int usX2Y_usbpcm_urbs_start(struct snd_usX2Y_substream *subs)
+{
+ int p, u, err,
+ stream = subs->pcm_substream->stream;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+
+ if (SNDRV_PCM_STREAM_CAPTURE == stream) {
+ usX2Y->hwdep_pcm_shm->captured_iso_head = -1;
+ usX2Y->hwdep_pcm_shm->captured_iso_frames = 0;
+ }
+
+ for (p = 0; 3 >= (stream + p); p += 2) {
+ struct snd_usX2Y_substream *subs = usX2Y->subs[stream + p];
+ if (subs != NULL) {
+ if ((err = usX2Y_usbpcm_urbs_allocate(subs)) < 0)
+ return err;
+ subs->completed_urb = NULL;
+ }
+ }
+
+ for (p = 0; p < 4; p++) {
+ struct snd_usX2Y_substream *subs = usX2Y->subs[p];
+ if (subs != NULL && atomic_read(&subs->state) >= state_PREPARED)
+ goto start;
+ }
+
+ start:
+ usX2Y_usbpcm_subs_startup(subs);
+ for (u = 0; u < NRURBS; u++) {
+ for (p = 0; 3 >= (stream + p); p += 2) {
+ struct snd_usX2Y_substream *subs = usX2Y->subs[stream + p];
+ if (subs != NULL) {
+ struct urb *urb = subs->urb[u];
+ if (usb_pipein(urb->pipe)) {
+ unsigned long pack;
+ if (0 == u)
+ atomic_set(&subs->state, state_STARTING3);
+ urb->dev = usX2Y->chip.dev;
+ urb->transfer_flags = URB_ISO_ASAP;
+ for (pack = 0; pack < nr_of_packs(); pack++) {
+ urb->iso_frame_desc[pack].offset = subs->maxpacksize * (pack + u * nr_of_packs());
+ urb->iso_frame_desc[pack].length = subs->maxpacksize;
+ }
+ urb->transfer_buffer_length = subs->maxpacksize * nr_of_packs();
+ if ((err = usb_submit_urb(urb, GFP_KERNEL)) < 0) {
+ snd_printk (KERN_ERR "cannot usb_submit_urb() for urb %d, err = %d\n", u, err);
+ err = -EPIPE;
+ goto cleanup;
+ } else {
+ snd_printdd("%i\n", urb->start_frame);
+ if (u == 0)
+ usX2Y->wait_iso_frame = urb->start_frame;
+ }
+ urb->transfer_flags = 0;
+ } else {
+ atomic_set(&subs->state, state_STARTING1);
+ break;
+ }
+ }
+ }
+ }
+ err = 0;
+ wait_event(usX2Y->prepare_wait_queue, NULL == usX2Y->prepare_subs);
+ if (atomic_read(&subs->state) != state_PREPARED)
+ err = -EPIPE;
+
+ cleanup:
+ if (err) {
+ usX2Y_subs_startup_finish(usX2Y); // Call it now
+ usX2Y_clients_stop(usX2Y); // something is completely wroong > stop evrything
+ }
+ return err;
+}
+
+/*
+ * prepare callback
+ *
+ * set format and initialize urbs
+ */
+static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usX2Y_substream *subs = runtime->private_data;
+ struct usX2Ydev *usX2Y = subs->usX2Y;
+ struct snd_usX2Y_substream *capsubs = subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE];
+ int err = 0;
+ snd_printdd("snd_usX2Y_pcm_prepare(%p)\n", substream);
+
+ if (NULL == usX2Y->hwdep_pcm_shm) {
+ if (NULL == (usX2Y->hwdep_pcm_shm = snd_malloc_pages(sizeof(struct snd_usX2Y_hwdep_pcm_shm), GFP_KERNEL)))
+ return -ENOMEM;
+ memset(usX2Y->hwdep_pcm_shm, 0, sizeof(struct snd_usX2Y_hwdep_pcm_shm));
+ }
+
+ mutex_lock(&usX2Y->prepare_mutex);
+ usX2Y_subs_prepare(subs);
+// Start hardware streams
+// SyncStream first....
+ if (atomic_read(&capsubs->state) < state_PREPARED) {
+ if (usX2Y->format != runtime->format)
+ if ((err = usX2Y_format_set(usX2Y, runtime->format)) < 0)
+ goto up_prepare_mutex;
+ if (usX2Y->rate != runtime->rate)
+ if ((err = usX2Y_rate_set(usX2Y, runtime->rate)) < 0)
+ goto up_prepare_mutex;
+ snd_printdd("starting capture pipe for %s\n", subs == capsubs ?
+ "self" : "playpipe");
+ if (0 > (err = usX2Y_usbpcm_urbs_start(capsubs)))
+ goto up_prepare_mutex;
+ }
+
+ if (subs != capsubs) {
+ usX2Y->hwdep_pcm_shm->playback_iso_start = -1;
+ if (atomic_read(&subs->state) < state_PREPARED) {
+ while (usX2Y_iso_frames_per_buffer(runtime, usX2Y) >
+ usX2Y->hwdep_pcm_shm->captured_iso_frames) {
+ snd_printdd("Wait: iso_frames_per_buffer=%i,"
+ "captured_iso_frames=%i\n",
+ usX2Y_iso_frames_per_buffer(runtime, usX2Y),
+ usX2Y->hwdep_pcm_shm->captured_iso_frames);
+ if (msleep_interruptible(10)) {
+ err = -ERESTARTSYS;
+ goto up_prepare_mutex;
+ }
+ }
+ if (0 > (err = usX2Y_usbpcm_urbs_start(subs)))
+ goto up_prepare_mutex;
+ }
+ snd_printdd("Ready: iso_frames_per_buffer=%i,captured_iso_frames=%i\n",
+ usX2Y_iso_frames_per_buffer(runtime, usX2Y),
+ usX2Y->hwdep_pcm_shm->captured_iso_frames);
+ } else
+ usX2Y->hwdep_pcm_shm->capture_iso_start = -1;
+
+ up_prepare_mutex:
+ mutex_unlock(&usX2Y->prepare_mutex);
+ return err;
+}
+
+static struct snd_pcm_hardware snd_usX2Y_4c =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 4,
+ .buffer_bytes_max = (2*128*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0
+};
+
+
+
+static int snd_usX2Y_usbpcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_usX2Y_substream *subs = ((struct snd_usX2Y_substream **)
+ snd_pcm_substream_chip(substream))[substream->stream];
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (!(subs->usX2Y->chip_status & USX2Y_STAT_CHIP_MMAP_PCM_URBS))
+ return -EBUSY;
+
+ runtime->hw = SNDRV_PCM_STREAM_PLAYBACK == substream->stream ? snd_usX2Y_2c :
+ (subs->usX2Y->subs[3] ? snd_usX2Y_4c : snd_usX2Y_2c);
+ runtime->private_data = subs;
+ subs->pcm_substream = substream;
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 1000, 200000);
+ return 0;
+}
+
+
+static int snd_usX2Y_usbpcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_usX2Y_substream *subs = runtime->private_data;
+
+ subs->pcm_substream = NULL;
+ return 0;
+}
+
+
+static struct snd_pcm_ops snd_usX2Y_usbpcm_ops =
+{
+ .open = snd_usX2Y_usbpcm_open,
+ .close = snd_usX2Y_usbpcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_usX2Y_pcm_hw_params,
+ .hw_free = snd_usX2Y_usbpcm_hw_free,
+ .prepare = snd_usX2Y_usbpcm_prepare,
+ .trigger = snd_usX2Y_pcm_trigger,
+ .pointer = snd_usX2Y_pcm_pointer,
+};
+
+
+static int usX2Y_pcms_lock_check(struct snd_card *card)
+{
+ struct list_head *list;
+ struct snd_device *dev;
+ struct snd_pcm *pcm;
+ int err = 0;
+ list_for_each(list, &card->devices) {
+ dev = snd_device(list);
+ if (dev->type != SNDRV_DEV_PCM)
+ continue;
+ pcm = dev->device_data;
+ mutex_lock(&pcm->open_mutex);
+ }
+ list_for_each(list, &card->devices) {
+ int s;
+ dev = snd_device(list);
+ if (dev->type != SNDRV_DEV_PCM)
+ continue;
+ pcm = dev->device_data;
+ for (s = 0; s < 2; ++s) {
+ struct snd_pcm_substream *substream;
+ substream = pcm->streams[s].substream;
+ if (substream && SUBSTREAM_BUSY(substream))
+ err = -EBUSY;
+ }
+ }
+ return err;
+}
+
+
+static void usX2Y_pcms_unlock(struct snd_card *card)
+{
+ struct list_head *list;
+ struct snd_device *dev;
+ struct snd_pcm *pcm;
+ list_for_each(list, &card->devices) {
+ dev = snd_device(list);
+ if (dev->type != SNDRV_DEV_PCM)
+ continue;
+ pcm = dev->device_data;
+ mutex_unlock(&pcm->open_mutex);
+ }
+}
+
+
+static int snd_usX2Y_hwdep_pcm_open(struct snd_hwdep *hw, struct file *file)
+{
+ // we need to be the first
+ struct snd_card *card = hw->card;
+ int err = usX2Y_pcms_lock_check(card);
+ if (0 == err)
+ usX2Y(card)->chip_status |= USX2Y_STAT_CHIP_MMAP_PCM_URBS;
+ usX2Y_pcms_unlock(card);
+ return err;
+}
+
+
+static int snd_usX2Y_hwdep_pcm_release(struct snd_hwdep *hw, struct file *file)
+{
+ struct snd_card *card = hw->card;
+ int err = usX2Y_pcms_lock_check(card);
+ if (0 == err)
+ usX2Y(hw->card)->chip_status &= ~USX2Y_STAT_CHIP_MMAP_PCM_URBS;
+ usX2Y_pcms_unlock(card);
+ return err;
+}
+
+
+static void snd_usX2Y_hwdep_pcm_vm_open(struct vm_area_struct *area)
+{
+}
+
+
+static void snd_usX2Y_hwdep_pcm_vm_close(struct vm_area_struct *area)
+{
+}
+
+
+static int snd_usX2Y_hwdep_pcm_vm_fault(struct vm_area_struct *area,
+ struct vm_fault *vmf)
+{
+ unsigned long offset;
+ void *vaddr;
+
+ offset = vmf->pgoff << PAGE_SHIFT;
+ vaddr = (char*)((struct usX2Ydev *)area->vm_private_data)->hwdep_pcm_shm + offset;
+ vmf->page = virt_to_page(vaddr);
+ get_page(vmf->page);
+ return 0;
+}
+
+
+static struct vm_operations_struct snd_usX2Y_hwdep_pcm_vm_ops = {
+ .open = snd_usX2Y_hwdep_pcm_vm_open,
+ .close = snd_usX2Y_hwdep_pcm_vm_close,
+ .fault = snd_usX2Y_hwdep_pcm_vm_fault,
+};
+
+
+static int snd_usX2Y_hwdep_pcm_mmap(struct snd_hwdep * hw, struct file *filp, struct vm_area_struct *area)
+{
+ unsigned long size = (unsigned long)(area->vm_end - area->vm_start);
+ struct usX2Ydev *usX2Y = hw->private_data;
+
+ if (!(usX2Y->chip_status & USX2Y_STAT_CHIP_INIT))
+ return -EBUSY;
+
+ /* if userspace tries to mmap beyond end of our buffer, fail */
+ if (size > PAGE_ALIGN(sizeof(struct snd_usX2Y_hwdep_pcm_shm))) {
+ snd_printd("%lu > %lu\n", size, (unsigned long)sizeof(struct snd_usX2Y_hwdep_pcm_shm));
+ return -EINVAL;
+ }
+
+ if (!usX2Y->hwdep_pcm_shm) {
+ return -ENODEV;
+ }
+ area->vm_ops = &snd_usX2Y_hwdep_pcm_vm_ops;
+ area->vm_flags |= VM_RESERVED | VM_DONTEXPAND;
+ area->vm_private_data = hw->private_data;
+ return 0;
+}
+
+
+static void snd_usX2Y_hwdep_pcm_private_free(struct snd_hwdep *hwdep)
+{
+ struct usX2Ydev *usX2Y = hwdep->private_data;
+ if (NULL != usX2Y->hwdep_pcm_shm)
+ snd_free_pages(usX2Y->hwdep_pcm_shm, sizeof(struct snd_usX2Y_hwdep_pcm_shm));
+}
+
+
+int usX2Y_hwdep_pcm_new(struct snd_card *card)
+{
+ int err;
+ struct snd_hwdep *hw;
+ struct snd_pcm *pcm;
+ struct usb_device *dev = usX2Y(card)->chip.dev;
+ if (1 != nr_of_packs())
+ return 0;
+
+ if ((err = snd_hwdep_new(card, SND_USX2Y_USBPCM_ID, 1, &hw)) < 0)
+ return err;
+
+ hw->iface = SNDRV_HWDEP_IFACE_USX2Y_PCM;
+ hw->private_data = usX2Y(card);
+ hw->private_free = snd_usX2Y_hwdep_pcm_private_free;
+ hw->ops.open = snd_usX2Y_hwdep_pcm_open;
+ hw->ops.release = snd_usX2Y_hwdep_pcm_release;
+ hw->ops.mmap = snd_usX2Y_hwdep_pcm_mmap;
+ hw->exclusive = 1;
+ sprintf(hw->name, "/proc/bus/usb/%03d/%03d/hwdeppcm", dev->bus->busnum, dev->devnum);
+
+ err = snd_pcm_new(card, NAME_ALLCAPS" hwdep Audio", 2, 1, 1, &pcm);
+ if (err < 0) {
+ return err;
+ }
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usX2Y_usbpcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usX2Y_usbpcm_ops);
+
+ pcm->private_data = usX2Y(card)->subs;
+ pcm->info_flags = 0;
+
+ sprintf(pcm->name, NAME_ALLCAPS" hwdep Audio");
+ if (0 > (err = snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 64*1024, 128*1024)) ||
+ 0 > (err = snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream,
+ SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 64*1024, 128*1024))) {
+ return err;
+ }
+
+
+ return 0;
+}
+
+#else
+
+int usX2Y_hwdep_pcm_new(struct snd_card *card)
+{
+ return 0;
+}
+
+#endif
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.h b/sound/usb/usx2y/usx2yhwdeppcm.h
new file mode 100644
index 0000000..c3382fd
--- /dev/null
+++ b/sound/usb/usx2y/usx2yhwdeppcm.h
@@ -0,0 +1,20 @@
+#define MAXPACK 50
+#define MAXBUFFERMS 100
+#define MAXSTRIDE 3
+
+#define SSS (((MAXPACK*MAXBUFFERMS*MAXSTRIDE + 4096) / 4096) * 4096)
+struct snd_usX2Y_hwdep_pcm_shm {
+ char playback[SSS];
+ char capture0x8[SSS];
+ char capture0xA[SSS];
+ volatile int playback_iso_head;
+ int playback_iso_start;
+ struct {
+ int frame,
+ offset,
+ length;
+ } captured_iso[128];
+ volatile int captured_iso_head;
+ volatile unsigned captured_iso_frames;
+ int capture_iso_start;
+};
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